| 00:00.06 | r0d3nt | encompassing everythign |
| 00:00.09 | r0d3nt | they patented the idea of moving voice to packets |
| 00:00.13 | markit | yes, software patents are a madness, they have to be forbidden |
| 00:00.16 | r0d3nt | and voip call forwarding specifically.... |
| 00:00.37 | CJLinst | So they calim to have, say, RTP patented? |
| 00:00.40 | r0d3nt | markit: i differ in opinion... but yes.. this situation is a clusterfuck that needs to be changed quickly.... |
| 00:00.47 | gambolputty | next thing you know, Verizon will say that no one can use SIP, IAX, or RTP |
| 00:00.52 | r0d3nt | CJLinst: something like that... go read the patents.. it's scary... |
| 00:00.57 | r0d3nt | gambolputty: yup.... |
| 00:01.02 | markit | r0d3nt: how can you be in favour of software patents?!?!? |
| 00:01.18 | r0d3nt | eventually once they shut down vonage.. they'll go after everyone else once they have legal precedence... |
| 00:01.49 | gambolputty | that means this vonage/verizon things needs to be shot down |
| 00:02.05 | gambolputty | else voip is dead |
| 00:02.12 | Fieldy | maybe in the usa |
| 00:02.38 | markit | Fieldy: good shot :) |
| 00:02.47 | r0d3nt | gambolputty: correct.... |
| 00:03.04 | cr4z3d | what's the best choice for MOH? |
| 00:03.29 | r0d3nt | heavy metal. |
| 00:03.39 | r0d3nt | clients love iron maiden and slayer |
| 00:03.39 | r0d3nt | IMHO. |
| 00:03.48 | r0d3nt | maybe some Pantera |
| 00:04.36 | markit | gambolputty: you will be sued for using it, and also your customers, if in USA ;) |
| 00:04.51 | cr4z3d | wait what? |
| 00:04.57 | [TK]D-Fender | "legal precedence" just means soeone found a dumb judge. |
| 00:05.02 | r0d3nt | markit: in a few years.. |
| 00:05.05 | r0d3nt | [TK]D-Fender: agreed |
| 00:05.58 | cr4z3d | so in order to use MOH i need to get format_mp3 or a different mp3 player? |
| 00:07.16 | r0d3nt | mpg123 |
| 00:07.20 | r0d3nt | and some mp3's |
| 00:07.24 | [TK]D-Fender | r0d3nt, "Give me the sense to wonder... to wonder if I'm freee.... give me sence to wonder ..... to know I can believe .... " |
| 00:07.54 | [TK]D-Fender | r0d3nt, "Can I Play With Madness" - Iron Maiden :D |
| 00:07.54 | r0d3nt | [TK]D-Fender: unconfuse me sir... |
| 00:07.55 | cr4z3d | i'm confused too it says mpg123 was used on pre 1.2 and format_mp3 on post |
| 00:07.59 | *** join/#asterisk ShadowHntr (i=sentinel@wikipedia/Shadowhntr) |
| 00:07.59 | r0d3nt | [TK]D-Fender: HAHAhahAHHA AH |
| 00:08.29 | r0d3nt | cr4z3d: ahhh.. i don't use 1.4-dev anymore.. 1.2-stable for life yo' |
| 00:08.31 | [TK]D-Fender | r0d3nt, I think I'm going to freak everyone out and replace my "best of Mozart" with "Best of the Beast" :D |
| 00:08.41 | r0d3nt | [TK]D-Fender: NIiiiiiiiiiice =))) |
| 00:09.01 | r0d3nt | six six six !! the number of the beast ! ! hell and fire !!1 was born to be released !!! |
| 00:09.22 | [TK]D-Fender | "You wanna know the truth, son? Lord I'll tell you the truth .... You're souls gonna burn in a lake if fiiiiirrrreeeee |
| 00:11.19 | [TK]D-Fender | r0d3nt, They are fun to play..... |
| 00:11.53 | r0d3nt | so really.. i don't wanna be a whiny pissy bitch.. but i'm seriously concerned about this vonage/verizon stuff..... it's quite sad... once *nix and asterisk gets off the ground, someone (verizon) kicks it back into the mud.. |
| 00:11.54 | r0d3nt | ((( |
| 00:11.56 | r0d3nt | =(((* |
| 00:12.01 | r0d3nt | this makes for a sad ratman.... |
| 00:12.12 | [TK]D-Fender | r0d3nt, SIP has been around a LONG time before Vonage.... |
| 00:12.17 | r0d3nt | agreed |
| 00:12.21 | r0d3nt | no argument from me... |
| 00:12.35 | [TK]D-Fender | r0d3nt, RFC's are all public, and I'm sure its limited to something stupid on their part |
| 00:12.55 | CJLinst | Trouble is, it doesn't sound like it's about SIP. |
| 00:13.07 | [TK]D-Fender | if it has any grounds at all that is... |
| 00:14.06 | [TK]D-Fender | RTP is jsut as old... older technically as H.3232 used it as well |
| 00:14.06 | r0d3nt | as i read it, maybe i need to refesh my memory on the patents, it was dealing with voip gateways, call forwarding and the transformation of voice to packet technology.. basically they're saying they invented it all and never used it.... |
| 00:14.06 | CunningPike | [TK]D-Fender: RTP is older than VOIP isn't it? Realaudio etc? |
| 00:17.05 | [TK]D-Fender | CunningPike, Don't know anything past H.323 :) |
| 00:17.13 | CunningPike | Heh |
| 00:17.45 | CJLinst | 'nite |
| 00:17.53 | CunningPike | Well, Verizon are going to have to sue 100Ks of people - you can't put the lid back on the box |
| 00:18.14 | mcab | what's VZN doing now? |
| 00:18.40 | CunningPike | Patented IP packets, apparently |
| 00:18.47 | mcab | hahahahahaha |
| 00:19.24 | *** join/#asterisk RoyK (n=roy@ti211310a341-1551.bb.online.no) |
| 00:20.21 | r0d3nt | irc.2600.net/#asterisk 90+ voip related rss feeds 24/7 realtime.... |
| 00:20.31 | r0d3nt | http://today.reuters.com/news/newsarticle.aspx?type=technologyNews&storyid=2007-03-23T231934Z_01_WAT007199_RTRUKOC_0_US-VONAGE-VERIZON.xml |
| 00:21.21 | cr4z3d | Mar 23 17:18:43 WARNING[2901]: file.c:512 ast_openstream_full: File /usr/share/asterisk/mohmp3/bitterbeings does not exist in any format |
| 00:21.21 | cr4z3d | Mar 23 17:18:43 WARNING[2901]: res_musiconhold.c:231 ast_moh_files_next: Unable to open file '/usr/share/asterisk/mohmp3/bitterbeings': No such file or directory |
| 00:21.25 | markit | r0d3nt: http://press.ffii.org/Software_patent_news |
| 00:21.35 | mcab | sucks for Vonage, but there are too many other big players in VoIP these days for it to put a huge crimp in things |
| 00:21.38 | r0d3nt | do they have a rss feed ?? i'll add it |
| 00:21.56 | mcab | although, it means Vzn can walk around with a big stick and bully people for a while :-7 |
| 00:22.02 | r0d3nt | yup |
| 00:22.06 | cr4z3d | what exactly does itmean "does not exist in any format" |
| 00:22.10 | cr4z3d | it's an mp3 file |
| 00:22.45 | CunningPike | cr4z3d: It may be, but it's not where asterisk is looking for it |
| 00:22.53 | CunningPike | cr4z3d: Check permissions |
| 00:23.01 | markit | r0d3nt: if you mean FFII site, they have rss |
| 00:23.10 | markit | (even if M$ patented rss recently) |
| 00:23.24 | cr4z3d | should i do a chmod +x on /usr/share/asterisk/mohmp3 |
| 00:23.46 | r0d3nt | mp3's don't need to be executable. no. |
| 00:23.58 | CunningPike | cr4z3d: What does ls -l show |
| 00:24.32 | r0d3nt | hehe |
| 00:24.35 | cr4z3d | CunningPike, -rw-r--r-- 1 cr4z3d cr4z3d 3235049 2007-03-23 17:15 bitterbeings.mp3 |
| 00:25.18 | CunningPike | cr4z3d: That's in /usr/share/asterisk/mohmp3, then? |
| 00:25.22 | [TK]D-Fender | cr4z3d, Now you need to wake up and install asterisk-addons ;) |
| 00:25.29 | CunningPike | cr4z3d: What are the permissions on that folder |
| 00:25.36 | [TK]D-Fender | cr4z3d, Without which MP3's are nothing but error messages ;) |
| 00:25.52 | cr4z3d | oh alright i thought i did that but i guess not |
| 00:26.00 | cr4z3d | pleeease be an ubuntu package haha |
| 00:26.07 | CunningPike | [TK]D-Fender: Good call |
| 00:26.14 | CunningPike | And on that note........ |
| 00:26.37 | [TK]D-Fender | cr4z3d, I'm going to try to be nice about this.... |
| 00:26.41 | r0d3nt | ... |
| 00:26.43 | r0d3nt | yeah..... |
| 00:26.43 | cr4z3d | damn.. i need to use source |
| 00:26.55 | [TK]D-Fender | cr4z3d, GET YOUR SORRY ASS OFF THAT UBUNTU CRAP AND USE A REAL DISTRO LIKE THE REST OF US! |
| 00:27.05 | [TK]D-Fender | cr4z3d, wait.... that didn't work out so well ;) |
| 00:27.09 | cr4z3d | haha |
| 00:27.17 | cr4z3d | what's wrong with ubuntu it's easy |
| 00:27.25 | [TK]D-Fender | cr4z3d, EASIER uhh? :) |
| 00:27.51 | [TK]D-Fender | cr4z3d, And thats wy you're back here every 2 seconds with new dependency problems, fear of no ZTDUMMY, etc :) |
| 00:28.05 | *** join/#asterisk Fieldy (i=vrBUCZkr@gentoo/contributor/Fieldy) |
| 00:28.29 | cr4z3d | haha i guess that's true.. i'm just scared of building from source because i know i'll screw something up big time |
| 00:28.42 | [TK]D-Fender | getting all the keys bits of * installed is a 5 minute job on CentOS. |
| 00:28.59 | [TK]D-Fender | cr4z3d, At least this way you won't see the problems coming ;) |
| 00:29.20 | [TK]D-Fender | "How hard could it be on Ubuntu? .... *BOOM*" |
| 00:29.37 | cr4z3d | haha oh great |
| 00:30.12 | [TK]D-Fender | cr4z3d, to tell you the trust I wouldn't be surprised if you CAN'T get it by default |
| 00:30.27 | [TK]D-Fender | cr4z3d, In that youcan't get MP3 support direct from them normally either... |
| 00:31.25 | cr4z3d | yeah especially since the default stuff that they say on voip-info.org and the directories on ubuntu seem to be different |
| 00:31.50 | *** join/#asterisk riddlebox (n=james@75-132-205-166.dhcp.stls.mo.charter.com) |
| 00:32.39 | cr4z3d | [TK]D-Fender, if i don't install asterisk-addons and just use a 3rd party mp3 player that supports 8khz playback is that possible? |
| 00:36.48 | Mercestes | cr4z3d, Should be. |
| 00:37.22 | Mercestes | use madplay likely |
| 00:39.13 | cr4z3d | how do i install it from source? |
| 00:39.29 | Qwell | why not just convert your stuff to ulaw or something? |
| 00:39.33 | Qwell | make the whole thing moot |
| 00:39.38 | r0d3nt | http://www.flixxy.com/medieval-tech-support.htm |
| 00:40.07 | cr4z3d | Qwell, moot? |
| 00:40.16 | Qwell | moot |
| 00:40.42 | cr4z3d | what's moot |
| 00:40.46 | cr4z3d | and if i convert everything to ulaw i dont' have to install asterisk-addons? |
| 00:42.31 | Qwell | correct |
| 00:42.36 | Qwell | and that's also the definition of "moot" ;) |
| 00:42.56 | [TK]D-Fender | God I love Novell's new "Linux vs PC vs Mac" ads..... |
| 00:42.58 | [TK]D-Fender | http://blog.wired.com/cultofmac/2007/03/novell_launches.html |
| 00:43.11 | [TK]D-Fender | That chick is imminently bang-able.... |
| 00:43.56 | *** join/#asterisk garreel (n=pippo@host195-31-dynamic.17-87-r.retail.telecomitalia.it) |
| 00:45.52 | *** part/#asterisk garreel (n=pippo@host195-31-dynamic.17-87-r.retail.telecomitalia.it) |
| 00:47.50 | *** join/#asterisk garreel (n=garreel@host195-31-dynamic.17-87-r.retail.telecomitalia.it) |
| 00:48.23 | Qwell | imminently? |
| 00:48.36 | garreel | hi all people |
| 00:48.39 | Mercestes | interesting adjective. |
| 00:49.02 | Mercestes | she's not all that hot either. |
| 00:49.45 | [TK]D-Fender | Mercestes, Fine... so its quite likely she ISN'T a virgin.... |
| 00:50.20 | [TK]D-Fender | And yes, I have a large "vocabulary" and am not afraid to use it ;) |
| 00:50.39 | Mercestes | define imminently. |
| 00:51.47 | [TK]D-Fender | http://dictionary.reference.com/browse/imminently |
| 00:52.13 | [TK]D-Fender | #1 |
| 00:52.29 | Mercestes | Good. you can spell dictionary. NOw, state your senteance again, replacing iminently with the appropriate definition. |
| 00:52.35 | [TK]D-Fender | As in "in a heart-beat" or "like a screen-door in a hurricane" ;) |
| 00:53.02 | [TK]D-Fender | Mercestes, My spelling is PERFECT... my typing sometimes leaves a little to be desired ;) |
| 00:53.11 | *** part/#asterisk drako (n=ljd@nelug/coreteam/luisjose) |
| 00:53.18 | Mercestes | I said nothing about your spelling. I question yoru usage of the wored. |
| 00:53.35 | Mercestes | that woman is like a heart beat bangable. that woman is like a screen door in a hurricane bangable. |
| 00:53.49 | Mercestes | I don't think "imminent" is the word you wanted in this case. |
| 00:53.53 | Fieldy | anyone know if Record can use speex? if so, what's the extension for Record(whatever.whatever)? |
| 00:55.46 | garreel | anyone can tell me how to change default codec for meetme conferences? |
| 00:55.59 | garreel | I know it's u-law |
| 00:56.00 | Qwell | garreel: meetme must use slin |
| 00:56.00 | [TK]D-Fender | Mercestes, Indeed it is. I've used it in several other contexts and it is quite correct |
| 00:56.00 | garreel | slin? |
| 00:56.04 | Qwell | signed linear |
| 00:56.04 | Mercestes | [TK]D-Fender I think you wanted undeniably or inexplicably or something like that... |
| 00:56.05 | Mercestes | [TK]D-FEnder but replace your word with a definition and see if your sentance still makes sense. I bet it doesn't. |
| 00:56.05 | garreel | I'd like to use A-law for meetme conferences |
| 00:56.22 | Qwell | garreel: You can't. It must use slin. You'll have to transcode |
| 00:56.28 | garreel | since in europe it's the default codec on pri lines |
| 00:57.00 | Mercestes | That woman is predestined bangable. That woman is guaranteed to happen in the future bangable. |
| 00:57.08 | Mercestes | that woman is unavoidable bangable. |
| 00:57.14 | garreel | no way?... neither patching sources? |
| 00:57.15 | *** join/#asterisk Hansin321 (i=Eric@c-67-176-40-243.hsd1.co.comcast.net) |
| 00:57.20 | Qwell | garreel: no way. sorry |
| 00:57.24 | Mercestes | That woman will imminently be banged. Now that's a sentance. |
| 00:57.27 | JT | Mercestes: am now |
| 00:57.29 | Qwell | but, transcoding between alaw and slin is trivial |
| 00:57.35 | Qwell | Mercestes: is it |
| 00:57.44 | [TK]D-Fender | Mercestes, You can't just substitute a word with its definition verbatim you SHCMUCK (go look that up while you're at it) :p |
| 00:58.09 | Mercestes | [TK]D-Fender, You usually can. :P |
| 00:58.16 | [TK]D-Fender | im·mi·nent·ly, adverb |
| 00:58.17 | [TK]D-Fender | not |
| 00:58.25 | Mercestes | JT: We think it's a bad card. ./patlooptest gave ok on span 1 and "Device busy" on 2, 3, 4. |
| 00:58.31 | [TK]D-Fender | im·mi·nent –adjective |
| 00:58.48 | JT | Mercestes: hrm ok |
| 00:58.55 | garreel | since I have alaw on all system I'd like to have that one also on meetme to avoid transcoding at all |
| 00:59.09 | [TK]D-Fender | Mercestes, Conjugate dumb-ass! |
| 00:59.13 | [TK]D-Fender | :D |
| 00:59.28 | Qwell | verbify |
| 00:59.30 | [TK]D-Fender | pwned. TOTALY. You can go cry now ;) |
| 00:59.39 | Mercestes | Uh....not really. |
| 00:59.43 | [TK]D-Fender | With an exta "L" even ;) |
| 00:59.47 | Mercestes | imminently and "banged" still don't go to gether. |
| 01:00.07 | Mercestes | imminently meaning "guaranteed to happen in the future, unavoidable" and "banged" being past tense. |
| 01:00.50 | [TK]D-Fender | Mercestes, Sigh.. there's jsut no hope for you.... |
| 01:00.52 | garreel | thank you anyway |
| 01:01.04 | Mercestes | mebbe not. |
| 01:01.11 | Mercestes | That woman is unavoidably bangable. |
| 01:01.21 | Mercestes | yea, just doesn't work. |
| 01:01.55 | JT | garreel: slin to alaw is hardly any load |
| 01:03.27 | [TK]D-Fender | Mercestes, think "I'd do her in a heart-beat". You have to consider the word the adverb is ENHANCING. |
| 01:03.54 | Mercestes | ...that's not what immiently means I don't think. |
| 01:04.44 | florz | ... plus mixing is somewhat difficult to do with alaw ... |
| 01:06.08 | cr4z3d | if i have an IVR context and i through in include => exts will all the extens in exts be available? |
| 01:06.23 | garreel | I had no load... |
| 01:06.55 | garreel | but since now it's doing.... alaw--- > meetme --- > alaw |
| 01:06.58 | [TK]D-Fender | Anyways.. off to play pool, back in a few hours... |
| 01:07.22 | garreel | I was wondering if it's possible o avoid transcoding |
| 01:07.48 | garreel | anyway no problem... |
| 01:09.23 | garreel | http://lists.digium.com/pipermail/asterisk-dev/2006-May/020430.html |
| 01:09.56 | garreel | reading this post I was thinking it was possible |
| 01:10.47 | florz | garreel: The transcoding is just like < 10 instructions per sample, so basically nothing. |
| 01:11.37 | Qwell | garreel: if you search further on that list, you'll find an explanation of why it can't be done |
| 01:12.01 | garreel | will search better |
| 01:22.44 | cr4z3d | is it possible for me to listen in on someone that's making a call from my asterisk |
| 01:23.33 | garreel | cr4z3d: try ZapBarge application |
| 01:24.02 | cr4z3d | eh... not a huge fan of 3rd party apps i'm just wondering if it's possible in asterisk itself |
| 01:24.12 | cr4z3d | i'm just playing around with asterisk not using it for anything important |
| 01:24.26 | garreel | it's included in asterisk |
| 01:24.32 | cr4z3d | oh |
| 01:24.34 | cr4z3d | ooh nm |
| 01:25.31 | garreel | I used that one to make a joke ;-) |
| 01:25.40 | *** join/#asterisk sharp (n=sharp@c-68-46-126-37.hsd1.pa.comcast.net) |
| 01:26.20 | *** join/#asterisk mosty (i=mostyn@60-241-198-194.static.tpgi.com.au) |
| 01:27.38 | cr4z3d | garreel, anything for sip? i don't use zap |
| 01:27.53 | Qwell | chanspy, monitor |
| 01:29.02 | cr4z3d | ok and if they're calling out i can still use chanspy? |
| 01:33.28 | mosty | i have a problem connecting via sip to a pbx which is behind nat. i can make calls, but the pbx doesn't get my dtmf (using the Read cmd). i have tried all three dtmf sip options. what else could be wrong? |
| 01:33.43 | *** join/#asterisk phillipk (n=pkey@216.248.143.77) |
| 01:33.43 | *** join/#asterisk HockeyInJune (n=HockeyIn@pool-68-161-182-123.ny325.east.verizon.net) |
| 01:34.03 | *** join/#asterisk HockeyInJune (n=HockeyIn@pool-68-161-182-123.ny325.east.verizon.net) |
| 01:37.38 | Mercestes | mosty: canreinvite=yes, dtmfmode=auto |
| 01:37.54 | cr4z3d | can anyone help me add monitor to an AGI script? |
| 01:40.02 | RA25 | mosty: echotest works fine...? Im trying to resolve one-way audio problem right now... :-) |
| 01:40.52 | *** join/#asterisk zirman (i=zirman@ip194.207.107.216.seg.NET) |
| 01:43.00 | RA25 | BTW... Has anyone here asterisk on private ip behind NAT ? Cant use my phone on my private AP and some other AP's with the same settings :-( |
| 01:43.12 | *** join/#asterisk RoyK (n=roy@217-175-152.100710.adsl.tele2.no) |
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| 01:47.48 | cr4z3d | hey can i put DYNAMIC_FEATURES=automon |
| 01:47.52 | cr4z3d | in globals? |
| 01:56.41 | [jwb] | anyone know if it's possible to supply a polycom phone with HTTP config info via DHCP? |
| 01:57.50 | *** join/#asterisk dwclarkNU (n=dwclark@c-24-91-40-208.hsd1.ma.comcast.net) |
| 01:58.09 | dwclarkNU | hi, im trying to get asterisknow to work with broadvoice, has anyone had any success at this? |
| 01:58.53 | *** join/#asterisk dalbaech (i=dalbaech@serverchimps.org) |
| 02:00.51 | dalbaech | Good "day" everyone. |
| 02:04.14 | cr4z3d | i'm trying to exec monitor from an AGI script |
| 02:04.35 | cr4z3d | i have EXEC MONITOR and that works but how would i put the parameters in it |
| 02:04.38 | mosty | RA25: actually i have only checked that audio works one-way |
| 02:04.51 | mosty | cr4z3d: yes |
| 02:04.58 | cr4z3d | yes? |
| 02:05.06 | cr4z3d | oh to that question |
| 02:05.24 | mosty | except i think you need => instead of = |
| 02:05.30 | cr4z3d | yeah i read that on the wiki |
| 02:06.03 | cr4z3d | mosty, in agi script i have print "EXEC MONITOR\n"; and that works fine but how would i pass arguments to it |
| 02:06.16 | cr4z3d | EXEC MONITOR wav, filenam? |
| 02:09.17 | *** join/#asterisk Assid (n=assid@203.212.204.107) |
| 02:14.12 | mosty | http://www.voip-info.org/wiki/view/exec |
| 02:15.08 | *** join/#asterisk techie (i=gus@voip.routedsystems.com) |
| 02:15.23 | RA25 | mosty: then, of course any inband DTMF won't work. You can hear but, but you cant talk, right...? |
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| 02:18.57 | *** join/#asterisk eald_home (n=eald@189.157.104.40) |
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| 02:23.13 | *** join/#asterisk MrTelephone (n=DeaLER25@bas13-toronto63-1242371764.dsl.bell.ca) |
| 02:23.20 | MrTelephone | norstar = crap |
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| 02:24.33 | *** mode/#asterisk [+o Qwell] by ChanServ |
| 02:30.55 | gambolputty | ok |
| 02:32.37 | *** join/#asterisk ltd (n=z@202-161-2-97.dyn.iinet.net.au) |
| 02:35.36 | antlers | is there a way to debug incoming calls on a fxo interface? |
| 02:37.47 | JerJer | there isn't much to debug |
| 02:38.10 | *** join/#asterisk cvschleich (n=cvs@65.127.122.56) |
| 02:39.06 | cvschleich | Anybody know if 911 ANI recieved at a PSAP on an analog trunk is any different from standard caller-id? |
| 02:39.27 | *** join/#asterisk sharp (n=sharp@c-68-46-126-37.hsd1.pa.comcast.net) |
| 02:39.54 | JerJer | if its a 'legacy' telco they would send the proper information to the PSAP |
| 02:41.44 | MrTelephone | have any of you tried any t1 over ip bridges? |
| 02:43.29 | *** join/#asterisk dj-fu (i=ajc@unaffiliated/dj-fu) |
| 02:45.30 | MrTelephone | just got lmsensors to work finally |
| 02:46.15 | *** join/#asterisk arsuceno (i=ircap@180.pool85-59-111.dynamic.orange.es) |
| 02:46.23 | arsuceno | hola |
| 02:50.28 | arsuceno | somebody know something about the ETSI FSK standard and caller ids? |
| 02:51.31 | florz | arsuceno: Is that what T-Com uses? |
| 02:52.56 | arsuceno | i don't know about T-Com, but in Spain (where I live) is supposed it is the standard |
| 02:53.12 | arsuceno | i cannot get caller id in a pstn line :( |
| 02:54.23 | *** join/#asterisk ShadowHntr (i=sentinel@wikipedia/Shadowhntr) |
| 02:54.39 | arsuceno | i have been reading about it, appears ETSI FSK is standard in Spain, France, Germany, Italy... |
| 02:54.47 | florz | arsuceno: hmm, well, I do know how the stuff that T-Com uses works, if that's of any help, but no clue what standards that's according to |
| 02:55.05 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 02:55.32 | florz | T-Com as in .de, that is, but I don't think they are offering POTS in any other countries |
| 02:59.27 | arsuceno | umm |
| 02:59.44 | arsuceno | and asterisk get caller id in T-Com lines? |
| 03:00.08 | cvschleich | JerJer: We're looking into replacing a 15 year old 911 system (backup center only, handles about 6 calls a year) with a Digium card, if it will pull the ANI from the telco the same way. |
| 03:00.19 | florz | arsuceno: on ISDN it certainly does, on POTS no clue |
| 03:00.27 | arsuceno | oh, oks |
| 03:00.37 | arsuceno | pots=pstn ? |
| 03:00.40 | JerJer | i don't think there is ANI on pots |
| 03:01.10 | florz | arsuceno: analog line |
| 03:01.45 | JerJer | but there is the FSK data for calling party |
| 03:01.58 | JerJer | so i presume one could send whatever is necessary |
| 03:02.07 | JerJer | that is if you are in control of both sides of the loop |
| 03:02.47 | cvschleich | The current arrangement is 4 analog trunks into a Proctor brand system that pulls ANI from the lines, so it's there, I just don't know if it's the same as caller-id, or if the information is encoded/decoded differently... |
| 03:04.04 | dalbaech | ANI and callerid are signalled differently I believe |
| 03:04.50 | cvschleich | That was my suspicion, but I don't know where to go to find out for sure... |
| 03:05.41 | florz | cvschleich: use a sound card to record the signal and have a look :-) |
| 03:06.39 | arsuceno | oooh.. oks |
| 03:06.59 | cvschleich | There's a thought. What do I compare it to? |
| 03:07.29 | arsuceno | i am thinking that if in uk uses a kind of FKS caller id... maybe it is "adaptable" to Spain |
| 03:08.36 | dalbaech | I'm sorry, I'm trying to find more information cvschleich |
| 03:08.59 | cvschleich | I appreciate the effort!!! |
| 03:09.14 | florz | arsuceno: IIRC the BT stuff works pretty similarly to what I have on T-Com POTS, if that helps you |
| 03:09.17 | dalbaech | CallerID is really just bellcore signalled in the US |
| 03:09.29 | dalbaech | ANI is transmitted in digital |
| 03:09.52 | dalbaech | most people just have analog (I believe you would need a digital line to capture the caller's true ANI) |
| 03:10.15 | dalbaech | [assuming the telco actually sends it to the adapter] |
| 03:10.36 | cvschleich | Yes, but I'm specifically referring the the 911 PSAP where the calls are received. |
| 03:11.00 | dalbaech | is it an analog line? |
| 03:11.19 | cvschleich | Yes, there are four analog lines into the PSAP. |
| 03:11.41 | dalbaech | then the ANI for that individual line should be sent to the 911. |
| 03:11.57 | cvschleich | The Proctor docs speak of TSPS (wink and no wink), and 1A-ESS. Are those standard caller id protocols? |
| 03:12.02 | dalbaech | regaredless of what you set the callerid to... it's dependent on the telco |
| 03:13.17 | cvschleich | I'm just looking to replace the 15 year old Proctor equipment with a digium card and asterisk, if I can get the ANI out of the lines the same way the Proctor equipment does.... |
| 03:13.27 | dalbaech | cvschleich : http://www.tech-faq.com/caller-id.shtml |
| 03:14.06 | cvschleich | Outstanding, thanks very much! I'll dig into that. |
| 03:14.31 | dalbaech | cvschleich : is this for a PSAP or for a client with the possibility of calling an emergency station? |
| 03:14.56 | dalbaech | most 911 carriers get both, caller id and ani |
| 03:15.14 | arsuceno | another question (i have a lot of!), what means BATTERY on wctdm debugs? |
| 03:15.19 | dalbaech | as with E-911, it's callerid information that is sent, then at the analog end, it's converted |
| 03:15.34 | dalbaech | (spoofed so to speak by the telco) |
| 03:17.21 | cvschleich | dalbaech: This is the PSAP. |
| 03:18.00 | dalbaech | ahh |
| 03:18.08 | dalbaech | what type of card or carrier? |
| 03:18.29 | cvschleich | ???? |
| 03:18.33 | dalbaech | I'd say convert to digital / TDM. |
| 03:18.43 | dalbaech | or is it already? |
| 03:19.08 | cvschleich | We have 4 single pair analog lines from the telco. |
| 03:19.33 | dalbaech | that's all? |
| 03:19.40 | dalbaech | how small is the town? |
| 03:19.44 | *** join/#asterisk robl^ (n=robl@pdpc/supporter/monthlybyte/robl) |
| 03:20.49 | dalbaech | well, if you got ISDN (two, so four channels.) The telco could send ANI and callerid |
| 03:21.16 | cvschleich | Yep. Been in place for 15 years. We're in a town of about 15000, about another 15000 throughout the rest of the county. Total of about 30000 population, this is the backup center, there's a full-blown new CAD system going into the main site. |
| 03:22.05 | dalbaech | ok, had me a little scared for a second. |
| 03:22.05 | cvschleich | No, we have to work with the lines we have. |
| 03:22.05 | dalbaech | ok, then you'll receive callerid |
| 03:22.17 | dalbaech | I'm not sure if ANI can be sent or not over plain analog |
| 03:22.32 | cvschleich | The system currently in place precedes caller-id, I'm just wondering if it's different than modern caller-id. |
| 03:22.33 | dalbaech | well, you could get the telco to translate ANI to callerid |
| 03:23.03 | cvschleich | Did you catch the protocols listed above? TSPS and 1A-ESS? |
| 03:23.17 | cvschleich | Do you know anything about those? |
| 03:23.18 | dalbaech | I'm confused on the ANI bit. one sec. |
| 03:23.43 | dalbaech | I'm already accustomned to SIP by itself. :) |
| 03:23.47 | dalbaech | *sp* |
| 03:25.24 | dalbaech | do you have the model # for the card or system you have now? |
| 03:25.58 | cvschleich | Yes, but not where I can get to it right now.... |
| 03:27.03 | dalbaech | k |
| 03:27.11 | dalbaech | I will continue reading until I get too dizzy. |
| 03:27.23 | dalbaech | (hydrocodone) |
| 03:29.00 | *** join/#asterisk dngcy2k (n=dngcy2k@124.82.7.2) |
| 03:29.39 | dalbaech | gotta love old systems |
| 03:30.19 | MrTelephone | someone want to try and send me a fax? |
| 03:30.29 | cvschleich | Proctor model 45505 Mini ANI-LINK mini-cabinet |
| 03:30.33 | dalbaech | Note 2. Public telephones are interfaced to TSPS (Traffic Service Position System). If you call an 800 number and whistle off using 2600 Hz, the distant toll office sends a wink back signal (a short on-hook) indicating it is ready to receive pulsing. TSPS responds to this wink back by printing out the original number called, the number called from, and the number MFed after the wink back. This print out goes to the billing and security departments. |
| 03:31.01 | dngcy2k | anyone know what is flexible extension logic |
| 03:31.02 | cvschleich | Where's you find that???? |
| 03:31.12 | dalbaech | but that's old data. |
| 03:31.29 | cvschleich | That sounds like the right thing. Yeah, it's all old stuff.... |
| 03:32.19 | dalbaech | hehe |
| 03:32.51 | *** join/#asterisk asteriskguy (n=petern@206.166.206.34) |
| 03:33.10 | dalbaech | so in reality; it's just four single analog lines |
| 03:33.33 | *** join/#asterisk Ifaistos (n=stelios@ipa226.211.tellas.gr) |
| 03:33.38 | cvschleich | yep. |
| 03:33.39 | MrTelephone | technology is getting out of hand |
| 03:33.48 | [TK]D-Fender | gambolputty, OLD SCHOOL... |
| 03:33.58 | MrTelephone | im looking for an adit 600 |
| 03:34.06 | [TK]D-Fender | gambolputty, I remember my early days fo Telecom reading 2600.... |
| 03:34.25 | cvschleich | So do you think I can get the ANI off these lines with a digium card?\ |
| 03:34.48 | MrTelephone | ANI? |
| 03:35.03 | [TK]D-Fender | cvschleich, Sure.... if you rewrite the drivers practically from scratch yourself... |
| 03:35.08 | MrTelephone | how do you get the cisco ata186 to stop putitng quotes around the callerid? |
| 03:35.20 | cvschleich | Yeah, that;s what I'm afraid of..... |
| 03:35.31 | dalbaech | well, cvschleich , you may have to get the telco's help.... |
| 03:35.32 | *** join/#asterisk coppice (n=chatzill@90.203.17.210.dyn.pacific.net.hk) |
| 03:35.57 | dalbaech | if you have the analog adapter for a box, it may only be able to receive callerid because of the firmware |
| 03:36.10 | dalbaech | so maybe the telco could translate ANI and rewrite callerid before signalling. |
| 03:36.22 | cvschleich | <snorts coke through nose>Yeah, I don't think the telco's going to help. |
| 03:36.46 | dalbaech | they may have to if it's for 911 service. |
| 03:36.48 | *** join/#asterisk sharp (n=sharp@c-68-46-126-37.hsd1.pa.comcast.net) |
| 03:36.57 | MrTelephone | check out this error |
| 03:36.59 | dalbaech | that's what the universal regulatory fee is for in the US |
| 03:37.00 | MrTelephone | [app_txfax.so]Mar 23 23:36:42 WARNING[13943]: loader.c:325 __load_resource: libspandsp.so.0: cannot open shared object file: No such file or directory |
| 03:37.07 | dalbaech | eeek |
| 03:37.12 | MrTelephone | but its in my /usr/local/lib/ |
| 03:37.29 | dalbaech | well, prolly missing a symlink |
| 03:37.57 | cvschleich | Our fallback is to continue to use these proctor boxes, I guess that's what we'll continue to use. Thanks for your help, all, especially dalbaech! |
| 03:38.22 | MrTelephone | lrwxrwxrwx 1 root staff 19 2007-03-23 23:23 libspandsp.so -> libspandsp.so.0.0.1 |
| 03:38.22 | MrTelephone | lrwxrwxrwx 1 root staff 19 2007-03-23 23:23 libspandsp.so.0 -> libspandsp.so.0.0.1 |
| 03:38.22 | MrTelephone | -rwxr-xr-x 1 root staff 910029 2007-03-23 23:23 libspandsp.so.0.0.1 |
| 03:38.26 | [TK]D-Fender | MrTelephone, You need to add the path to /etc/ld.so.conf and rerun ldconfig |
| 03:38.27 | dalbaech | cvschleich : sorry, but, sometimes technology has to go backwards for it to work |
| 03:38.31 | MrTelephone | ok |
| 03:39.00 | *** part/#asterisk cvschleich (n=cvs@65.127.122.56) |
| 03:40.42 | MrTelephone | someitmes people should stop making ten differnt lib folders |
| 03:40.46 | MrTelephone | pick one and stick with it |
| 03:40.57 | MrTelephone | god |
| 03:41.11 | MrTelephone | usr local lib usr lib usr lib modules lib lib lib lib |
| 03:41.36 | [TK]D-Fender | but Linux is about... choice! |
| 03:42.28 | MrTelephone | i though it was about awesomeness |
| 03:42.31 | MrTelephone | thought |
| 03:42.32 | MrTelephone | heh |
| 03:42.44 | MrTelephone | but still, one lib folder should be enough? |
| 03:43.05 | MrTelephone | ./etc is the best folder anyone ever invented |
| 03:44.00 | *** join/#asterisk Strom_C (n=strom@24.96.136.141) |
| 03:44.13 | MrTelephone | feel like sending me a fax? |
| 03:44.25 | MrTelephone | fender bender |
| 03:44.44 | MrTelephone | gotta test out the old spanddsp |
| 03:46.28 | MrTelephone | wow spandsp is pretty cool |
| 03:48.33 | *** part/#asterisk steve___ (n=steve@kit-dhcp1.porchlight.ca) |
| 03:50.51 | cr4z3d | hmm i have an AGI script running and i'm trying to execute a shell script at the end to merge recordings together |
| 03:51.02 | cr4z3d | i know the shell script works by it self |
| 03:51.15 | cr4z3d | i tried adding exec "shell.sh" |
| 03:51.22 | cr4z3d | but that didn't work any suggestions? |
| 03:52.15 | *** join/#asterisk bmg505 (n=leon@c1-11-15.rndf.isadsl.co.za) |
| 03:52.51 | MrTelephone | good question |
| 03:53.12 | mosty | cr4z3d: file/dir permissions problem |
| 03:53.14 | mosty | ? |
| 03:53.26 | MrTelephone | sometimes there is 2 commands for executing |
| 03:53.37 | MrTelephone | you want to use the command that keeps the current shell running in the bg? |
| 03:53.57 | MrTelephone | is the agi perl? |
| 03:53.59 | cr4z3d | mosty, chmod +x on shell script |
| 03:54.01 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 03:54.01 | cr4z3d | agi is perl |
| 03:54.14 | MrTelephone | did u try system() function? |
| 03:54.18 | mosty | cr4z3d: i meant on the files it's trying to access |
| 03:54.21 | cr4z3d | no i didn't |
| 03:54.34 | cr4z3d | oh they should be the same i think |
| 03:54.43 | cr4z3d | they get created by the monitor |
| 03:54.50 | MrTelephone | oh |
| 03:55.08 | cr4z3d | would system work better than exec? |
| 03:55.32 | MrTelephone | system() function waits until the child process has |
| 03:55.32 | MrTelephone | > > exited. |
| 03:55.37 | MrTelephone | i dunno it might |
| 03:55.46 | cr4z3d | oh system sounds like more of what i want |
| 03:55.54 | MrTelephone | asterisk probably dog bangs the child script your running when the parents closes |
| 03:56.01 | cr4z3d | so system would wait until the perl agi is done? |
| 03:56.20 | cr4z3d | and would system run it as root? |
| 03:56.28 | MrTelephone | system is a function in perl |
| 03:56.41 | cr4z3d | yeah |
| 03:56.45 | MrTelephone | i think you use it like system(path/to/child/program) |
| 03:56.56 | MrTelephone | give it a quick try anyhow |
| 03:57.00 | mosty | cr4z3d: if you install sox, monitor'ed channels will be merged automatically |
| 03:57.03 | MrTelephone | i had some problem with exec() before |
| 03:57.24 | cr4z3d | mosty, really? i have sox installed |
| 03:57.34 | MrTelephone | sox is just a conversion utility? |
| 03:57.41 | cr4z3d | yeah soxmix |
| 03:57.42 | MrTelephone | you still have to run a script or something |
| 03:57.46 | cr4z3d | i wrote a shell script to run it |
| 03:57.57 | cr4z3d | merge.sh basename |
| 03:58.09 | MrTelephone | that sounds like an interesting thing your doing |
| 03:58.14 | mosty | monitor uses sox automatically here, after the monitoring ends |
| 03:58.14 | MrTelephone | i havn't tried monitoring a channel before |
| 03:58.28 | MrTelephone | what are you merging? |
| 03:58.29 | cr4z3d | mosty what version of asterisk you using? |
| 03:58.35 | cr4z3d | two wav files |
| 03:58.43 | MrTelephone | same phone call? |
| 03:58.59 | cr4z3d | yeah |
| 03:59.06 | cr4z3d | one for each side |
| 03:59.09 | MrTelephone | are you spying on your wifes phone calls? |
| 03:59.10 | MrTelephone | j/k |
| 03:59.11 | *** join/#asterisk ManxPower (n=manxpowe@71-8-56-64.dhcp.leds.al.charter.com) |
| 03:59.23 | cr4z3d | haha no just have it setup on my spoof script |
| 03:59.41 | MrTelephone | thats over my head |
| 03:59.46 | MrTelephone | are you having spoof issues? |
| 03:59.51 | cr4z3d | no not at all |
| 04:00.05 | cr4z3d | everything works fine except merging the two wav files |
| 04:00.32 | MrTelephone | back to windows and cooledit pro |
| 04:00.33 | MrTelephone | lol |
| 04:00.46 | cr4z3d | haha fuck that |
| 04:01.04 | cr4z3d | soxmix file1.wav file2.wav file.wav |
| 04:01.06 | cr4z3d | done |
| 04:01.19 | MrTelephone | i havn't dicked around with sound at all because I'm too lazy to get the sound drivers working in linux |
| 04:01.27 | cr4z3d | HAHA |
| 04:01.30 | cr4z3d | whoops stupid caps |
| 04:01.32 | MrTelephone | thats it? |
| 04:01.34 | cr4z3d | i used to do that |
| 04:01.38 | cr4z3d | yeah |
| 04:01.46 | MrTelephone | kick yourself |
| 04:01.58 | cr4z3d | kick myself? |
| 04:02.03 | MrTelephone | monitoring is the same as recording a call? |
| 04:02.18 | cr4z3d | yeah |
| 04:02.35 | mosty | cr4z3d: 1.2.whatever |
| 04:02.39 | MrTelephone | I build a nice server with 2gig of ram.. thinking ok, i never had a problem with memory and linux |
| 04:02.46 | MrTelephone | i look today and theres 14mb left |
| 04:03.17 | MrTelephone | where does it all go :( |
| 04:03.39 | MrTelephone | nevermind there is only 1 gig in there |
| 04:03.54 | cr4z3d | mosty, hmm me too.. wonder why it's not doing it automatically |
| 04:04.00 | cr4z3d | do i have to set anything in a conf file |
| 04:04.14 | MrTelephone | i used to have non-ecc memory but i opened up one of my favorite nudey pictures and the breasts were missing :( |
| 04:04.16 | MrTelephone | geek joke |
| 04:07.19 | cr4z3d | OH mosty found it. i have to set the m flag in monitor |
| 04:07.33 | dalbaech | yep |
| 04:07.41 | dalbaech | (mixes incoming/outgoing) |
| 04:08.12 | dalbaech | good "night" everyone... |
| 04:13.52 | MrTelephone | any of you use an adaptec raid card? |
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| 04:20.47 | cr4z3d | ah i'm so happy i got everything working the way i want it to for today |
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| 04:35.36 | JT | MrTelephone: you have described normal *nix operation |
| 04:35.41 | JT | unused ram is wasted ram |
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| 04:52.22 | Qwell | stupid udev |
| 04:52.50 | Strom_C | catsecks |
| 04:53.26 | Qwell | udev just went ahead and decided that forcedeth was no longer eth0 |
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| 04:58.28 | Mportnoy | Hello anyone knows how to TEst the IVR in a trixbox box? |
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| 05:35.28 | infinity1 | i have a .call file, asterisk makes a call, connects. and starts playing the ivr menu, but guess what, even though asterisk says the call is connected, my phone never rings! |
| 05:36.03 | infinity1 | anyone experience this? |
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| 05:53.51 | ManxPower | infinity1: no |
| 05:54.28 | ManxPower | MY guess would be a problem with the .call file |
| 05:54.39 | ManxPower | I'll be you are making someone very angry |
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| 06:21.25 | tzafrir_laptop | infinity1, in that case, you can start debugging the problem by: set verbose 3 |
| 06:21.42 | tzafrir_laptop | if you want help, pastebin some relevant information |
| 06:22.01 | tzafrir_laptop | e.g: the trace you get from such a call in the CLI |
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| 07:01.09 | mosty | qwell: use nameif in your init scripts |
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| 08:07.42 | Mahmoud | "Any way they let you use it untill you exeed the use and then they simply sue you. so be carefull. |
| 08:07.42 | Mahmoud | If you know Arabic i advice you to check Alkhaleej news paper by the date og 12 March 2006. the sentens 7 Pakistani, 1 1 ndian and 1 europian not mention from which country. they got uptill 3 years in jail and a fine of 13000 DHR (around 3K$) and deport after that, so be carful mate. |
| 08:07.47 | Mahmoud | " |
| 08:07.53 | Mahmoud | Crime? used VoIP! |
| 08:08.46 | Mahmoud | http://www.voipuser.org/forum_topic_5212.html |
| 08:12.43 | JT | what country? |
| 08:13.27 | Mahmoud | UAE |
| 08:13.45 | Mahmoud | i didn't check that news paper yet, once i check it, i'll know if it's a lie or not |
| 08:14.02 | Mahmoud | but it's horrible! 3 years jail! deport! 13000 AED fine! |
| 08:14.21 | Mahmoud | they were merely sending RTP packets containing their ugly voice :/ |
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| 08:16.28 | *** part/#asterisk kindernacht (n=alwaysbl@cpe-69-205-13-132.nycap.res.rr.com) |
| 08:24.31 | tzafrir_laptop | that's an old topic. I guess it is true, otherwise omeone would have commented on it |
| 08:24.54 | tzafrir_laptop | From what I know, people are indeed forced to use VPNs and such there |
| 08:25.43 | tzafrir_laptop | The crime is basically to practice in communication without a license, I guess. |
| 08:32.13 | *** join/#asterisk uwe (n=uwe1@dogbert.palnet.com) |
| 08:48.01 | *** join/#asterisk fbffff (n=fbffff@r74-193-79-81.pfvlcmta01.grtntx.tl.dh.suddenlink.net) |
| 08:48.43 | *** join/#asterisk ltd (n=z@202-161-2-97.dyn.iinet.net.au) |
| 08:51.47 | *** join/#asterisk kate21de (n=Miranda@ppp-82-135-80-171.dynamic.mnet-online.de) |
| 08:52.03 | kate21de | #join ##c++ |
| 08:53.42 | *** join/#asterisk lorinc (n=ang@pool-5535.adsl.interware.hu) |
| 08:59.17 | *** join/#asterisk vlt|home (n=daniel@dslb-088-073-232-105.pools.arcor-ip.net) |
| 09:08.05 | tzafrir_laptop | anybody here uses gtalk? |
| 09:08.11 | *** join/#asterisk fbffff (n=fbffff@r74-193-79-81.pfvlcmta01.grtntx.tl.dh.suddenlink.net) |
| 09:08.20 | tzafrir_laptop | with Asterisk? |
| 09:10.09 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 09:19.08 | *** join/#asterisk [Airwolf] (n=airwolf@86.92.174.235) |
| 09:19.56 | *** join/#asterisk friedrich| (n=friedric@e177251107.adsl.alicedsl.de) |
| 09:20.35 | *** join/#asterisk dj-fu (n=ajc@unaffiliated/dj-fu) |
| 09:45.19 | *** join/#asterisk adde (n=adde@c-96bae055.1169-1-64736c11.cust.bredbandsbolaget.se) |
| 09:46.25 | adde | Ports to connect a sip softphone is only 5060 and 10001-20000 UDP - is that correct? or have i missed something |
| 09:47.10 | JT | 10000-20000 udp |
| 09:50.25 | adde | ? |
| 09:59.46 | adde | please coul somone help in right direction im really stuck |
| 09:59.51 | vlt|home | Hello. I tried using "_X.#" to tell Asterisk that "#" finishes the number. But it doesn't work as expected: It waits for the timeout to end and then continues. What's wrong here? |
| 10:00.08 | JT | adde: you haven't told us what else you've done.. |
| 10:02.35 | *** join/#asterisk ghenry (n=ghenry@suretecsystems.plus.com) |
| 10:04.26 | *** join/#asterisk amessina (n=amessina@h-66-166-108-202.chcgilgm.covad.net) |
| 10:05.17 | amessina | reading through the bug submission guidelines at bugs.digium.com, it's suggested that for feature requests, they are floated here first. |
| 10:07.03 | amessina | anyway, i would like to ask if it would be possible to have the patch at http://www.voip-info.org/wiki/index.php?page=Asterisk+zaptel+pulse+dialing added as a configuration option in the config file, rather than to compile the module specifically for my tdm400 card. |
| 10:19.38 | DarKnesS_WolF | adde: ur right |
| 10:20.54 | DarKnesS_WolF | adde: 5060 is signaling port and 10000-20000 media path |
| 10:20.54 | DarKnesS_WolF | vlt|home: i think this issue of # is phone issue not asterisk " not sure " but for me it works with GiPTL and grandstream phones |
| 10:21.10 | adde | I have opened this and forwarded them through FW. Set up two sip accounts, when i dial from when two another the calls go through but no sound is transfered from any end....Any ideaa? |
| 10:21.31 | adde | r 25 12:49:03] WARNING[2458]: chan_sip.c:6460 determine_firstline_parts: Bad request protocol E9E |
| 10:21.33 | DarKnesS_WolF | adde: the phones behinds the nat ? |
| 10:21.41 | adde | keep getting this in Asterisk |
| 10:22.10 | DarKnesS_WolF | adde: try to add this nat=yes and canreinvite=no in both sip accounts |
| 10:22.21 | adde | My sip phone is on the local network, behind the firewall where i opened all ports. My father is in india, with no firewall... |
| 10:23.16 | DarKnesS_WolF | adde: so it's like that yourfather --->internetl ---->firewall"urs"----->asterisk<---- your phone ? |
| 10:23.33 | adde | yes |
| 10:23.35 | DarKnesS_WolF | then for ur father do the settings i just told u and try again. |
| 10:23.49 | *** join/#asterisk af_ (n=getsmart@ip-156-32.sn2.eutelia.it) |
| 10:23.50 | DarKnesS_WolF | make sure u did forward the UDP ports not the TCP |
| 10:24.38 | adde | http://www.pastebin.ca/407789 |
| 10:25.25 | vlt|home | DarKnesS_WolF: No, it recognized the "#" correctly but seems to match it as the ".", so I solved using several fixed length extensions (from 1 to 15 digits) all ending with "#" ... |
| 10:25.32 | DarKnesS_WolF | hmm never seen somehting like that adde |
| 10:25.56 | DarKnesS_WolF | vlt|home: hehe too much hasel ;-) |
| 10:26.04 | *** join/#asterisk RoyK (n=roy@217-175-152.100710.adsl.tele2.no) |
| 10:26.23 | adde | it just started... :S |
| 10:26.36 | adde | I think its when the Xlite softphones are connected... |
| 10:26.54 | DarKnesS_WolF | adde: wanna my own opinion ? |
| 10:26.57 | DarKnesS_WolF | use IAX |
| 10:27.07 | DarKnesS_WolF | one port u do fwd and softphones here and tehre and ur done ;-) |
| 10:27.50 | adde | hmmm are there IAX for WM5 ? |
| 10:28.30 | DarKnesS_WolF | what is WM5 ? |
| 10:28.36 | adde | Windows Mobile 5 |
| 10:28.54 | DarKnesS_WolF | oh hmm i think so |
| 10:29.07 | DarKnesS_WolF | check voip-info.org searching for IAX clients |
| 10:29.34 | adde | What do i need to do to get IAX working, i have a standard install of Asterisk... |
| 10:30.18 | DarKnesS_WolF | adde: copy the accounts from sip.conf to iax.conf |
| 10:30.33 | DarKnesS_WolF | change ur extension.conf to use IAX protcol instade of SIP |
| 10:30.43 | DarKnesS_WolF | and fwd port 4569 UDP from ur firewall |
| 10:30.44 | DarKnesS_WolF | and ur done. |
| 10:31.31 | adde | [10011] |
| 10:31.31 | adde | type=friend |
| 10:31.31 | adde | username=10011 |
| 10:31.31 | adde | secret=banana |
| 10:31.32 | adde | host=dynamic |
| 10:31.32 | adde | context=internal |
| 10:31.34 | adde | nat=yes |
| 10:31.36 | adde | canreinvite=no |
| 10:31.39 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 10:31.39 | adde | that in IAX.COnf ? |
| 10:31.52 | RoyK | ~pb |
| 10:32.02 | jbot | extra, extra, read all about it, pb is a place to paste your stuff without flooding the channel - try http://pastebin.ca, or http://channels.debian.net/paste |
| 10:32.13 | adde | sorry m8 |
| 10:32.22 | DarKnesS_WolF | adde: yes |
| 10:32.25 | DarKnesS_WolF | thats it |
| 10:32.52 | *** part/#asterisk RoyK (n=roy@217-175-152.100710.adsl.tele2.no) |
| 10:34.51 | adde | Any recomended IAX software to just test its working for WIndows...? |
| 10:35.15 | sbingner | adde, iaxphone? |
| 10:35.21 | adde | ...and shalkl i remove the users from SIP.CONF |
| 10:37.13 | *** join/#asterisk Vec (n=Vec@dsl-243-89-41.telkomadsl.co.za) |
| 10:38.00 | Vec | Does the order I put exten and macro's etc in extensions.conf matter ? |
| 10:38.46 | sbingner | Vec, yes |
| 10:38.57 | Vec | sbingner : how does it matter ? |
| 10:39.13 | sbingner | it'll use the first match first |
| 10:39.40 | JT | are you sure about that? |
| 10:39.54 | sbingner | unless something changed recently... |
| 10:39.58 | sbingner | oh |
| 10:40.07 | Vec | I am more concerned about macro's then pattern matching. |
| 10:40.08 | JT | i thought the most specific match usually came first |
| 10:40.10 | sbingner | does it use the last match first? lol |
| 10:40.16 | JT | most specific |
| 10:40.26 | sbingner | JT, I don't think so |
| 10:40.47 | sbingner | hence the problems with people putting in _. and breaking everything |
| 10:41.08 | sbingner | Vec, macros it doesn't matter |
| 10:41.10 | DarKnesS_WolF | adde: yes remove them from sip.conf " comment " them |
| 10:41.16 | Vec | _. does not seem very clever |
| 10:41.22 | sbingner | it's not... |
| 10:41.23 | JT | sbingner: _. is a silly match :) |
| 10:41.23 | sbingner | heh |
| 10:41.34 | sbingner | but if it was most specific it wouldn't have broken things |
| 10:41.46 | sbingner | -=I=- didn't do it... |
| 10:43.31 | adde | What port in firewall for IAX ? |
| 10:44.16 | JT | 4569 udp |
| 10:46.16 | JT | adde: usually sip is workable too |
| 10:55.52 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 10:59.45 | *** join/#asterisk kate21de (n=MirandaM@ppp-82-135-80-171.dynamic.mnet-online.de) |
| 11:02.12 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 11:07.10 | *** join/#asterisk littleball (n=littleba@bb220-255-154-211.singnet.com.sg) |
| 11:07.56 | littleball | hello, who can recommend a software or web link for me to do a text to speech conversion? i need to create a few voice prompt file for my IVR system |
| 11:09.26 | JT | err if you're making permanent voice prompts they should be made by a human |
| 11:09.50 | littleball | JT, why? |
| 11:10.39 | JT | because customers will hate computer generated ones |
| 11:10.43 | JT | they sound inhuman :) |
| 11:11.17 | littleball | JT, AT&T is good, just cannot slow down the speed |
| 11:11.43 | JT | text to speech is only good for real time information that changes |
| 11:11.48 | JT | what? |
| 11:13.20 | sbingner | I want to find some way to un-invent speech-to-text for all those bastards that remove the number options from their IVR systems |
| 11:13.25 | sbingner | I refuse to talk to a damned computer |
| 11:13.44 | JT | speech recognition is AWFUL |
| 11:14.09 | sbingner | I have switched cell companies as a result of them implimenting that crap |
| 11:14.25 | sbingner | I did find that if I said "fuck you" it would say "I'm sorry you're having problems, let me get you an operator" |
| 11:14.48 | JT | the worst ones are the ones that won't take you to an operator |
| 11:14.53 | JT | or even give dtmf options |
| 11:14.57 | sbingner | I think I had to say it 5 times |
| 11:15.01 | JT | just keep asking you to repeat it |
| 11:15.02 | sbingner | lol |
| 11:15.11 | sbingner | yea I've had that once or twiecx |
| 11:15.12 | sbingner | I hung upo |
| 11:37.32 | *** join/#asterisk coppice (n=chatzill@90.203.17.210.dyn.pacific.net.hk) |
| 11:38.01 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 11:40.28 | *** join/#asterisk cpm (n=Chip@pdpc/supporter/active/cpm) |
| 11:48.25 | *** join/#asterisk Ryushin (i=proxy@windwalker.openinnovations.com) |
| 11:48.27 | adde | When i get a callin and dial an extension it is just quiet until someone answers... how do i get a ringing tone? |
| 11:49.11 | *** join/#asterisk dalbaech (i=dalbaech@serverchimps.org) |
| 11:49.21 | dalbaech | are there any asterisk ninjas in the house? |
| 11:51.41 | *** join/#asterisk zotz (n=zotz@24.244.163.157) |
| 12:00.45 | *** join/#asterisk slima (n=slima@unaffiliated/slima) |
| 12:01.43 | dalbaech | meep |
| 12:03.52 | *** join/#asterisk d00gster (n=doughant@bas1-toronto12-1088929080.dsl.bell.ca) |
| 12:05.16 | vlt|home | Hello. Is there way to Return() from a Gosub() to a certain exten in the "parent" context? |
| 12:06.43 | vlt|home | Something easier than Set(foo=<new_exten>) in the sub context and Goto(${foo},1) in the parent one? |
| 12:07.34 | Gido-E | i thought there was a pop and push function? but i am not sure. |
| 12:10.18 | *** join/#asterisk ToyMan (n=Stuart@user-0cevdmv.cable.mindspring.com) |
| 12:14.12 | *** join/#asterisk L|NUX (n=linux@unaffiliated/lnux/x-10290) |
| 12:19.33 | adde | "ast_rtp_read: Unknown RTP codec 126 received from '1 " - could somone tell me what this is? |
| 12:25.07 | *** join/#asterisk Xen^ (n=linux@unaffiliated/lnux/x-10290) |
| 12:45.26 | *** join/#asterisk Strom_M (n=strom@12-189-87-2.att-inc.com) |
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| 12:51.54 | *** join/#asterisk Ebola (n=Ebola@host86-143-156-147.range86-143.btcentralplus.com) |
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| 12:52.52 | cweiske | When trying to do "mISDN start" to get mISDN running for asterisk, I get the following error: |
| 12:52.59 | *** join/#asterisk ceph_ (i=taco@c-68-58-26-120.hsd1.in.comcast.net) |
| 12:53.01 | cweiske | >> /sbin/modprobe --ignore-install hfcmulti type= protocol= layermask= poll=128 debug=0 timer=0 |
| 12:53.02 | cweiske | FATAL: Error inserting hfcmulti (/lib/modules/2.6.20.3/extra/hfcmulti.ko): Invalid argument |
| 12:53.14 | cweiske | what can I do there? |
| 12:53.17 | ceph_ | should I have my asterisk box behind NAT or not? |
| 12:53.25 | ceph_ | for using SIP |
| 12:54.17 | Strom_M | ceph_: running it behind NAT makes things more hairy if you plan to have SIP phones behind entirely different NATs |
| 12:54.42 | *** join/#asterisk eltech (i=G00Ds@ool-457c93b6.dyn.optonline.net) |
| 12:56.21 | JT | cweiske: i'm guessing one of the options is invalid |
| 12:56.45 | cweiske | the misdn config wiki does not tell me what to specify |
| 12:57.06 | cweiske | it says that it should work directly after doing a mISDN config |
| 12:58.32 | JT | well there's 3 parameters there with null arguments |
| 12:58.35 | JT | seems pointless |
| 12:59.26 | ceph_ | well, i'm having problems receiving incoming calls with my SIP provider, I can call out fine, but it's like my phone isn't even connected when it comes to incoming |
| 12:59.32 | cweiske | yes, mISDN does that |
| 12:59.36 | ceph_ | what should i troubleshoot first |
| 13:01.02 | Strom_M | ceph_: describe your network setup please |
| 13:01.05 | cweiske | does someone have a working mISDN.conf for a hfc usb card? |
| 13:02.50 | JT | err |
| 13:02.51 | ceph_ | just a simple NAT...phone and asterisk server behind the same router |
| 13:03.02 | JT | cweiske: how many ports are on this hfc usb module? |
| 13:03.16 | cweiske | isdn ports? just one |
| 13:03.30 | ceph_ | do I need to forward any ports to the asterisk server? |
| 13:03.37 | JT | then hfcmulti would be totally the wrong driver to use |
| 13:04.01 | cweiske | hfcmulti has been added automatically by the config wizard |
| 13:04.03 | Strom_M | ceph_: yes |
| 13:04.07 | JT | look |
| 13:04.13 | Strom_M | ceph_: forward ports 5060 UDP and 10000-20000 UDP |
| 13:04.18 | JT | clearly you're going to need to do some things manually |
| 13:04.31 | JT | stop blaming everything on less than magic automagic :) |
| 13:04.46 | cweiske | ok |
| 13:05.02 | ceph_ | ah, thanks |
| 13:05.38 | cweiske | that's why I asked if anyone had a working config.. |
| 13:05.48 | JT | not with misdn |
| 13:05.52 | JT | or a usb hfc device |
| 13:05.57 | JT | i avoid usb where possible |
| 13:06.16 | cweiske | my server has only 1 pci slot, and this is occupied with a raid card |
| 13:06.25 | cweiske | so I have to use the usb adapter |
| 13:06.31 | JT | i see |
| 13:07.53 | *** join/#asterisk malverian (n=malveria@gentoo/developer/malverian) |
| 13:11.36 | cweiske | ok, now I did not use "mISDN config" but the misdn-init script that was in the sources |
| 13:11.41 | cweiske | it seems to work now |
| 13:12.02 | cweiske | great, misdnportinfo shows a line! |
| 13:13.22 | *** join/#asterisk gambolputty (n=gambolpu@cblmdm72-240-185-4.buckeyecom.net) |
| 13:13.38 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 13:18.39 | mvanbaak | hey |
| 13:18.51 | mvanbaak | is the AEL2-1.2 branch up-to-date ? |
| 13:19.17 | mvanbaak | my asterisk-trunk installation is not working nice with my IAX2 provider |
| 13:19.22 | mvanbaak | so I want to go back to 1.2 |
| 13:19.43 | mvanbaak | but would hate to redo my extensions.ael in AEL1 version |
| 13:20.14 | Strom_M | what about going to 1.4 branch? |
| 13:21.10 | mvanbaak | for some reason the IAX2 link between my telco (using asterisk 1.0.X) and 1.4 is not working correctly |
| 13:21.34 | Strom_M | which cretinous telco is still using asterisk 1.0.x? |
| 13:21.50 | mvanbaak | .nl provider |
| 13:23.00 | mvanbaak | I hate to go back, because it means I have to look for chan_sccp again and stuff |
| 13:23.17 | mvanbaak | but my homesystem is not reachable 90% of the time |
| 13:23.20 | mvanbaak | and that sux even more |
| 13:23.50 | Strom_M | which codec are you trying to use? |
| 13:24.51 | *** part/#asterisk cweiske (n=cweiske@dslb-088-074-134-071.pools.arcor-ip.net) |
| 13:27.35 | mvanbaak | alaw |
| 13:27.39 | mvanbaak | then ulaw |
| 13:27.41 | mvanbaak | then gsm |
| 13:28.04 | mvanbaak | I have this: |
| 13:28.07 | mvanbaak | disallow=all |
| 13:28.12 | mvanbaak | allow=alaw |
| 13:28.16 | mvanbaak | allow=ulaw |
| 13:28.19 | mvanbaak | allow=gsm |
| 13:30.45 | mvanbaak | hhmm |
| 13:30.52 | mvanbaak | I can fix it some other way I think |
| 13:31.12 | mvanbaak | put asterisk 1.2 on my gateway registering to the IAX2 provider |
| 13:31.18 | mvanbaak | and make my -trunk box register there |
| 13:31.25 | mvanbaak | and route calls that way |
| 13:31.44 | mvanbaak | put voicemail on the 1.2 box |
| 13:32.00 | mvanbaak | so if I break my -trunk box callers will still get the voicemail |
| 13:32.18 | mvanbaak | and all my cisco phones can be configured with 'backup callmanager' |
| 13:34.16 | crashev | anyone knows what is this -> res_musiconhold.c:526 monmp3thread: Request to schedule in the past?!?! <- it spawns mpg123 and it seems to be stuck? |
| 13:34.17 | *** join/#asterisk AdamCrowe (n=adam@82-35-250-182.cable.ubr04.dals.blueyonder.co.uk) |
| 13:34.31 | AdamCrowe | I NEED HELP!!! |
| 13:34.58 | AdamCrowe | Can anybody answer a TrixBox question??? |
| 13:35.35 | mvanbaak | dont use trixbox |
| 13:35.54 | Strom_M | see topic |
| 13:37.26 | *** join/#asterisk viking78 (i=viking78@66-168-102-94.dhcp.jcsn.tn.charter.com) |
| 13:37.36 | AdamCrowe | I don't think it's a TrixBox problem, altough I am using it :/ |
| 13:37.51 | AdamCrowe | I keep getting Kernel Panic on inbound calls... |
| 13:37.54 | Vec | Can I do a include => context1,context2,context3 ? |
| 13:37.57 | Strom_M | it's a trixbox problem |
| 13:38.49 | AdamCrowe | Sure? |
| 13:39.21 | *** join/#asterisk Exhar (n=Roy@213-73-139-87.cable.quicknet.nl) |
| 13:39.34 | Strom_M | sounds like it to me |
| 13:39.48 | Strom_M | go kvetch at them and/or install something more sane |
| 13:40.17 | drfreeze | OT anyone here ever use idrive.com or xdrive.com? |
| 13:40.25 | drfreeze | ..for backup |
| 13:40.36 | AdamCrowe | More sane as in only Asterisk? |
| 13:41.36 | mvanbaak | AdamCrowe: indeed |
| 13:41.49 | Strom_M | indeed...asterisk on a basic linux install |
| 13:42.04 | AdamCrowe | What linux do you recommend? |
| 13:42.13 | mvanbaak | anything you are familiar with |
| 13:42.35 | Strom_M | personally, I like debian |
| 13:42.41 | mvanbaak | me too |
| 13:43.00 | AdamCrowe | I've simply had so many problems with this thing already, I'm getting very affraid of thying anything new. |
| 13:43.13 | AdamCrowe | * trying |
| 13:43.23 | AdamCrowe | I am using a B410P card :/ |
| 13:43.28 | mvanbaak | it's because of trixbox |
| 13:44.05 | AdamCrowe | Anybody here from London area? |
| 13:44.08 | *** join/#asterisk [Airwolf] (n=airwolf@86.92.174.235) |
| 13:45.47 | AdamCrowe | Does Asterisk 1.4.2 require chan_misdn for B410P? |
| 13:46.37 | Strom_M | AdamCrowe: i believe so |
| 13:47.10 | AdamCrowe | So I would still need to recompile Zaptel and Asterisk after installing chan_misdn... |
| 13:48.04 | Strom_M | AdamCrowe: you can type "make b410p" in /usr/src/asterisk and it will do everything for you |
| 13:49.55 | AdamCrowe | I'm doing it manually because the computer is not connected to the Internet. |
| 13:51.52 | AdamCrowe | and I think you mean in /usr/src/zaptel/ |
| 13:53.34 | *** join/#asterisk Fieldy (i=eltGDrAj@gentoo/contributor/Fieldy) |
| 13:54.15 | *** join/#asterisk eltech (i=G00Ds@ool-457c93b6.dyn.optonline.net) |
| 13:55.08 | Fieldy | hm. i put an MP3 in the moh directory, it's readable, has no spaces in its name, and when i start asterisk I get "Segmentation fault" and it dies. any ideas? |
| 13:55.58 | AdamCrowe | If I install Debian with Asterisk, would I still be able to use some web-based configuration "panel" like FreePBX? |
| 13:59.56 | *** join/#asterisk nighty^ (n=nighty@211.154.202.98) |
| 14:01.28 | *** join/#asterisk kate21de (n=Miranda@ppp-82-135-12-5.dynamic.mnet-online.de) |
| 14:02.03 | adde | To run a AGI script from commandline how do i go about i saw somone who wrote in an example php myagi.agi but i dont have command php |
| 14:02.14 | adde | i installed php4 just now |
| 14:03.55 | adde | please |
| 14:04.42 | AdamCrowe | If I install Debian with Asterisk, would I still be able to use some web-based configuration "panel" like FreePBX? |
| 14:05.13 | adde | adam: yes |
| 14:05.24 | adde | but you have to install the gui seperate |
| 14:06.17 | adde | I did that... but to be honest i preffered the conf files... at least you know what changes are made then.... |
| 14:06.36 | adde | so i removed it as fast as it was installed |
| 14:08.37 | *** join/#asterisk garreel (n=garreel@host195-31-dynamic.17-87-r.retail.telecomitalia.it) |
| 14:10.27 | AdamCrowe | Thanks |
| 14:10.34 | adde | np |
| 14:11.12 | AdamCrowe | I'm looking at the Debian ISO's... The last time I installed Debian, it was ony ONE ISO file, NOT 15! Are they all necessary??? |
| 14:11.29 | *** join/#asterisk robl^ (n=robl@pdpc/supporter/monthlybyte/robl) |
| 14:12.08 | adde | i installed from 1 DVD |
| 14:12.27 | adde | but no, you can use net-inst |
| 14:12.35 | adde | 150 mb or something |
| 14:15.15 | AdamCrowe | Still makes no sense... How did go up to two DVDs?! |
| 14:18.25 | sysreq | The additionnal packages; you can now basically install any package you want without ever using you Internet connection. |
| 14:19.35 | sysreq | I mean, if you can have a working environment (without a GUI, of course) that fits into 400MB with the net-inst, it has to be the extra packages. |
| 14:21.12 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 14:21.14 | AdamCrowe | I don't have an Internet connection available yet on that computer. |
| 14:21.38 | AdamCrowe | Which means that I need to download the whole two DVDs... |
| 14:22.45 | adde | When i redirect a call to a new extension its quiet until answered, is it possible to make it ring? |
| 14:23.14 | crashev | what could be the reason that music on hold does not work? mpg123 is spawned but I cant hear anything on the other machine? |
| 14:26.43 | Vec | Does anyone here use extensions.ael rather than extensions.conf ? |
| 14:29.36 | adde | Adam: you still only need one DVD |
| 14:29.43 | adde | at least i only did. |
| 14:29.58 | adde | i suppose you not installing X on your PBX |
| 14:30.19 | tzafrir_laptop | crashev, why do you need mpg123 for moh? |
| 14:31.23 | tzafrir_laptop | AdamCrowe, if you have a decent network connection, just install from the netinst media or the first CD alone |
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| 14:32.29 | tzafrir_laptop | The installer itself is on the first CD. The rest of the disks include parts of the huge Debian packages repository. Most of it are things you won't need anyway |
| 14:32.48 | adde | [15:21] <AdamCrowe> I don't have an Internet connection available yet on that computer. |
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| 15:27.31 | voltagex | hi, is there a guide somewhere to setting up FWD with Asterisk? |
| 15:30.02 | [TK]D-Fender | http://www.voip-info.org/wiki/view/Asterisk+How+to+connect+to+FWD |
| 15:31.08 | voltagex | sorry, should have made myself a little more clear. Is it possible for my FWD number to become the number for my asterisk box? |
| 15:31.26 | Qwell | that is exactly what the link explains how to do |
| 15:31.31 | voltagex | oh |
| 15:32.18 | [TK]D-Fender | qwell, big print never works, nor URLS that kinda shove the answer right in your face :) |
| 15:32.38 | voltagex | be gentle, I'm running out of caffeine |
| 15:33.28 | [TK]D-Fender | voltagex, Means you're not out yet.. no slacking! be warned that guide is using a some deprecated syntax. |
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| 15:41.48 | voltagex | [TK]D-Fender: but will still work with 1.2? |
| 15:42.03 | *** join/#asterisk Fieldy (i=LLyP7vfF@gentoo/contributor/Fieldy) |
| 15:42.10 | adde | Has anyone succeeded to call MSN users from asterisk? |
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| 15:43.14 | [TK]D-Fender | voltagex, Likely, but you should tweak it a bit anyways. things were deprecated in 1.2 and removed entirely from 1.4 |
| 15:43.34 | [TK]D-Fender | voltagex, but it looked 1.2 capabale pretty much as-is |
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| 15:48.58 | tzafrir_laptop | anybody managed to talk with gtlak users through asterisk? |
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| 15:50.08 | tzafrir_laptop | gtalk , that is. And is also happens to know if the Asterisk user should appear as "on-line" for other jabber clients? |
| 15:50.41 | tzafrir_laptop | adde, what protocol does MSN use? |
| 15:51.30 | voltagex | [TK]D-Fender: how do I test if the Asterisk config worked? |
| 15:52.33 | [TK]D-Fender | voltagex, TRY IT <- |
| 15:52.46 | [TK]D-Fender | duh :) |
| 15:52.50 | voltagex | does asterisk need to be restarted? |
| 15:53.12 | [TK]D-Fender | voltagex, No, but you should know what needs to be reloaded.... |
| 15:53.27 | voltagex | I don't :/ |
| 15:53.27 | *** join/#asterisk L|NUX (n=linux@unaffiliated/lnux/x-10290) |
| 15:54.56 | [TK]D-Fender | ~book |
| 15:55.07 | jbot | rumour has it, book is a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 15:55.16 | [TK]D-Fender | voltagex, But as a priner, once you're done editing your configs, "relaod" |
| 15:55.18 | [TK]D-Fender | reload* |
| 15:55.48 | *** part/#asterisk amessina (n=amessina@h-66-166-108-202.chcgilgm.covad.net) |
| 15:55.54 | voltagex | send HUP to the asterisk process? |
| 15:56.29 | Sweeper | no XD |
| 15:56.41 | Sweeper | get into the cli, type reload |
| 15:56.55 | adde | GTalk-To-VoIP supports calling MSN users.... |
| 15:57.46 | adde | so an implementation of this with * would enable calling MSN,Jabber,GTalk etc... |
| 15:57.52 | [TK]D-Fender | voltagex, No, you just type "reload" into the * CLI |
| 15:58.20 | voltagex | [TK]D-Fender: done, just working out where the configs have word wrapped where they're not supposed to |
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| 16:20.20 | tzafrir_laptop | voltagex, reduce verbosity and reload |
| 16:20.37 | tzafrir_laptop | This will give you more focused warnings |
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| 16:57.44 | adde | I have a SIP callin number is there a way to receive fax on this number? |
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| 17:02.47 | ManxPower | adde: FaxOverVoiceOverIPOverSIP usually does not work very well |
| 17:05.42 | [[blah]asfd | i have a new system that I have set up... I have about 10 people on the phone concurrently and all of the phones sound like crap!!! They sound like they are under water. Only on outbound calls though... this is connected via point to point t1 to another server which is connected via PRI. They are using sip protocol and gsm codec. |
| 17:05.53 | [[blah]asfd | how can I fix it. |
| 17:06.17 | adde | ManXPower: Could you please explain? |
| 17:06.22 | ManxPower | [[blah]asfd: very carefully |
| 17:06.31 | ManxPower | adde: running fax over SIP does not work very well |
| 17:06.46 | [[blah]asfd | I am not seeing any cli errors.... |
| 17:06.58 | [[blah]asfd | and my output of top looks really good. |
| 17:07.04 | [[blah]asfd | 3 users sound fine. |
| 17:07.05 | ManxPower | in fact it works so badly they invented a totally new protocol called T.38 to do fax over ip. Unfortunatly almost nothing supports T.38 |
| 17:07.13 | adde | Manxpower: In what way doesnt it work very well... ? |
| 17:07.22 | ManxPower | [[blah]asfd: "sip show channels" confirm the codec is REALLY GSM |
| 17:07.28 | ManxPower | adde: failed faxes |
| 17:07.56 | adde | 50% ? 1% or 90% |
| 17:08.11 | adde | i mean if one in 10 fails its ok... they will just have to resend... |
| 17:08.12 | ManxPower | if you are LUCKY you might get only a 20% failure rate. Most people get closer to 50% - 90% failure rate |
| 17:08.50 | [[blah]asfd | apparenly not all are gsm... some ulaw |
| 17:09.03 | adde | I have been looking for a tutorial to set this up...no luck... everything seems to be done for Reall lines... |
| 17:09.04 | ManxPower | I manage 7 asterisk servers with at least 200 users total at 7 locations and we install a local fax line direct to the telco at each location, not even running thru asterisk |
| 17:09.09 | ManxPower | [[blah]asfd: there is part of your problem |
| 17:09.19 | [[blah]asfd | http://pastebin.ca/408155 |
| 17:09.39 | ManxPower | [[blah]asfd: disallow=all and allow=gsm in sip.conf |
| 17:09.49 | ManxPower | [[blah]asfd: what phones are you using? |
| 17:09.53 | [[blah]asfd | yeah, i allow more than just that... is that the issue? |
| 17:09.58 | [[blah]asfd | linksys spa942 |
| 17:10.05 | ManxPower | [[blah]asfd: yes. asterisk will always pick ulaw if it can |
| 17:10.26 | Qwell | I think g722 might be higher on the list than ulaw actually |
| 17:10.42 | ManxPower | [[blah]asfd: make SURE your rtp packet size is .2 and not .3 If small numbers of calls sound fine then your problem is prolly not rtp packet size |
| 17:11.03 | ManxPower | Qwell: does anything actually support G722? |
| 17:11.09 | Qwell | polycom 650/550 |
| 17:11.17 | [[blah]asfd | ok... so over point to point to another system.. would gsm or ulaw be the preference? |
| 17:11.33 | ManxPower | Qwell: Ok. In %99 of cases Asterisk will pick ulaw if it can |
| 17:11.36 | Qwell | :p |
| 17:11.52 | ManxPower | [[blah]asfd: it all comes down to banwidth usage .vs. voice quality .vs. cpu usage |
| 17:12.19 | [[blah]asfd | cpu is not an issue and bandwidth is not an issue |
| 17:12.25 | [[blah]asfd | quality is... ualw then? |
| 17:12.47 | ManxPower | generallty the less bandwidth you want to use the lower the audio quality, but using lots of CPU can improve call quality |
| 17:12.49 | adde | manxpower: could you hint me what to search for to find more info? |
| 17:13.00 | ManxPower | [[blah]asfd: ULAW/ALAW is what the telcos use. |
| 17:13.16 | ManxPower | adde: not really since I have never tried to send a fax over IP. Most people don't |
| 17:13.20 | ManxPower | you could try the Wiki |
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| 17:13.46 | adde | i only wanna receive |
| 17:13.52 | ManxPower | GSM is low bandwidth, low cpu, but has call quality about that of a good cell phone connection |
| 17:14.18 | ManxPower | iLBC and Speex are low bandwidth, good quality and high bandiwdth useage |
| 17:14.48 | ManxPower | very few devices suport ilbc or speex |
| 17:14.50 | Qwell | lpc10 is low bandwidth, low cpu, low quality |
| 17:14.55 | Qwell | it totally wins |
| 17:16.17 | ManxPower | I could run a couple of calls over that at least |
| 17:16.54 | ManxPower | [[blah]asfd: first make all your calls use GSM. |
| 17:17.19 | ManxPower | The only codec that should sound like "underwater" might be lpc10 and most people say THAT sounds like "Mr. Roboto" |
| 17:17.38 | *** join/#asterisk chema (n=chema@66.Red-88-11-92.dynamicIP.rima-tde.net) |
| 17:17.54 | ManxPower | [[blah]asfd: a ulaw/alaw/G711 voip call would use around 80Kbps of bandwidth, maybe 82 |
| 17:17.56 | Qwell | all my calls start off underwater |
| 17:17.59 | Qwell | until I answer |
| 17:18.09 | ManxPower | Qwell: that's because you live in Atlantis |
| 17:18.22 | Qwell | But that's only because I use Playback(Super_Mario_Bros-Underwater) :P |
| 17:18.26 | *** join/#asterisk shinux__ (n=shinux@86.62.8.178) |
| 17:18.27 | Qwell | erm, Playtones |
| 17:18.27 | *** part/#asterisk mtaht4 (n=m@c-71-198-23-124.hsd1.ca.comcast.net) |
| 17:18.36 | ManxPower | Qwell: Playtones? Pervert |
| 17:18.40 | *** part/#asterisk chema (n=chema@66.Red-88-11-92.dynamicIP.rima-tde.net) |
| 17:18.43 | Qwell | playtones rocks |
| 17:19.04 | Qwell | If I actually had any musical talent, I'd be making more of those :p |
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| 17:28.55 | b0on | can anyone point me a way to allow storage of dtmf decoded tones to be stored in a database, based upon user prompts in the IVR? |
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| 17:53.49 | [[blah]asfd | ManxPower: that fixed my issue by setting it to allow just one codec... thank you very much. |
| 17:55.15 | canapa | for a asterisk box with ~ 100 users, how good would u recomend the hardware (as in cpu, ram, etc) |
| 17:56.29 | *** join/#asterisk Strom_M (n=strom@65.14.229.26) |
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| 17:59.51 | Qwell | Strom_M: 65.14? |
| 18:00.36 | Strom_M | the Charlotte airport wifi network |
| 18:00.40 | *** join/#asterisk cr4z3d (n=cr4z3d@ip70-162-106-15.ph.ph.cox.net) |
| 18:00.40 | Qwell | ahh |
| 18:00.56 | Qwell | a bit out of the way, no? |
| 18:01.07 | Strom_M | this is where US Airways has a hub, it seems |
| 18:01.11 | Qwell | ahh |
| 18:01.14 | Qwell | still |
| 18:01.16 | Strom_M | yes |
| 18:01.32 | Strom_M | US Airways: the nameless faceless airline you never think about but somehow wind up flying anyway |
| 18:01.32 | Qwell | You'd think they'd have somewhere a little more centrally located |
| 18:01.50 | Strom_M | nope - they have Phoenix, Las Vegas, Charlotte, and Baltimore |
| 18:03.25 | *** join/#asterisk xtr-II (i=01928375@S0106000c41ed11e1.vf.shawcable.net) |
| 18:06.41 | *** join/#asterisk hijacked (i=LtY3@cerebus.clandestineresearch.com) |
| 18:06.46 | b0on | can anyone point me to a good agi guide? |
| 18:07.11 | *** part/#asterisk [[blah]asfd (n=ckwall@c-71-195-199-149.hsd1.ut.comcast.net) |
| 18:07.20 | *** join/#asterisk drako (n=ljd@nelug/coreteam/luisjose) |
| 18:07.32 | *** part/#asterisk mog (n=mog@c-71-207-200-130.hsd1.al.comcast.net) |
| 18:12.18 | cr4z3d | b0on, it's just perl scripting for the most part |
| 18:12.38 | cr4z3d | and you do print "APPLICATION arg1|arg2|arg3"; |
| 18:13.31 | b0on | i'm thinking that might get me pretty close to what i'm looking for |
| 18:16.23 | ManxPower | b0on: usually you would use an AGI library to do all the hard stuff for you |
| 18:16.47 | ManxPower | they are available for most languages. asterisk-perl (or maybe perl-asterisk) is, oddly, for perl |
| 18:17.35 | sevard | iirc it's asterisk-perl |
| 18:18.06 | sevard | and there's plenty of information about various scripting languages and agi (including perl) on the wiki |
| 18:18.53 | *** join/#asterisk enex (n=roman@essn-4db6971d.pool.einsundeins.de) |
| 18:19.02 | enex | hello |
| 18:21.02 | *** join/#asterisk cr4z3d (n=cr4z3d@ip70-162-117-14.ph.ph.cox.net) |
| 18:21.17 | cr4z3d | man i hate when i time out and i can't get my nick |
| 18:25.33 | *** join/#asterisk shido6 (n=shido6@d221-68-200.commercial.cgocable.net) |
| 18:28.00 | cpm | ghost |
| 18:28.08 | cr4z3d | ghost? |
| 18:28.15 | Qwell | /msg nickserv help ghost |
| 18:29.00 | cr4z3d | ohalright nice |
| 18:29.06 | cr4z3d | i used recover |
| 18:29.09 | cr4z3d | but it takes forever |
| 18:29.47 | cr4z3d | if i want to have a user enter a number followed by the # i'm guessing i need to use agi scripting or is there a way to do it without that |
| 18:30.32 | *** join/#asterisk Mw3 (n=mw3@ip59934bd1.rubicom.hu) |
| 18:36.17 | wunderkin | yeah use read |
| 18:36.33 | Qwell | cr4z3d: recover and release |
| 18:36.34 | cr4z3d | oh read nice didn't know the had something like that |
| 18:37.01 | cr4z3d | Qwell, yeah i tried that but i must have done something wrong kept getting a retarded "invalid password" error |
| 18:37.24 | wunderkin | ghost is more appropriate |
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| 18:42.04 | cr4z3d | what's some good hold music |
| 18:42.12 | Qwell | slayer |
| 18:42.23 | cr4z3d | haha |
| 18:42.44 | cr4z3d | alright might as well through some slayer in there |
| 18:42.50 | Qwell | or an audiobook of Stephen Hawking reading A Brief History of Time |
| 18:43.07 | Qwell | and yes, they exist |
| 18:43.18 | cr4z3d | haha wow |
| 18:43.37 | Strom_M | come on |
| 18:43.38 | ManxPower | cr4z3d: "show applications" is your friend |
| 18:43.49 | Strom_M | the latest craze in hold music is all disco, all the time |
| 18:44.06 | ManxPower | Strom_M: Well that is one way to get callers to hang up |
| 18:44.25 | cr4z3d | haha |
| 18:44.26 | sevard | doodoodoodoot do the hold music dootodootodoot |
| 18:44.32 | Strom_M | bahahahahahha |
| 18:49.13 | *** part/#asterisk nitrox_77 (n=nitrox@I82-143-6-208I.adsl.topnet.it) |
| 18:52.30 | [TK]D-Fender | b0Nah.. Iron Maiden > ALL |
| 18:53.06 | cr4z3d | fuck her gently by tenacious d is my current song for hold |
| 18:54.06 | gambolputty | sped up 40% of course :) |
| 18:54.40 | enex | ..ooOO(ah, ok, so i've joined a channel of mad people...uh..ok ;-) |
| 18:55.01 | cr4z3d | mad people? |
| 18:55.19 | enex | jufging by ringtone-suggestions ;-) |
| 18:55.24 | enex | *judging |
| 18:55.45 | cr4z3d | this is the greatest and best song in the world |
| 18:55.56 | enex | hehe, ok ok |
| 18:56.17 | enex | I'll try to refine my taste of music some more then ;-) |
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| 18:58.01 | *** join/#asterisk rad07 (n=radoslav@64-126-95-37.static.everestkc.net) |
| 19:08.02 | SkramX | anyone using a Cisco 7940 and Asterisk togethr? |
| 19:08.40 | Strom_M | SkramX: I'm using a cisco 7960 and Asterisk |
| 19:08.43 | Strom_M | close enough though :) |
| 19:08.50 | SkramX | right |
| 19:08.58 | *** join/#asterisk emergence (n=harko@CPE000c6e52557e-CM000e5c709370.cpe.net.cable.rogers.com) |
| 19:09.01 | SkramX | So.. I flashed my phone to the SIP firmware, 6.3 |
| 19:09.32 | SkramX | setup the SIP server, line info, etc. and edited sip.conf apropriately and it's not seeming to work |
| 19:09.40 | SkramX | brb, going to try one other thing |
| 19:09.53 | SkramX | Strom_M: can you maybe post your sip.conf context for your phone? |
| 19:10.00 | *** join/#asterisk lucifr (n=chatzill@cpe-75-85-54-211.socal.res.rr.com) |
| 19:10.07 | emergence | Did chan_gtalk replace chan_jingle? As I do not see the channel driver chan_jingle when I "make menuselect"...... |
| 19:10.08 | Strom_M | SkramX: it's a fairly basic setup |
| 19:10.29 | Strom_M | what do you mean by "it doesnt work" |
| 19:10.53 | SkramX | err, I'll try some more things and then come back; sorry for prematurely asking |
| 19:11.12 | Strom_M | .....ok then |
| 19:11.22 | cr4z3d | man i'm bored.. i need something interesting to do |
| 19:11.24 | lucifr | Hi |
| 19:11.41 | Strom_M | hi |
| 19:11.43 | lucifr | I need help with my ASterisk Installation |
| 19:11.50 | lucifr | first time doing it |
| 19:11.52 | emergence | cr4z3d, go buy an nslu2 and check out the nslu2 linux community |
| 19:12.00 | lucifr | Asterisk* |
| 19:12.07 | cr4z3d | buy an nslu2? |
| 19:12.21 | Strom_M | lucifr: the answer is cheese |
| 19:12.38 | wunderkin | goatsex? |
| 19:12.43 | lucifr | I got a VoIP provider and I have added to my Porviders list and no luck. Any odeas? |
| 19:13.00 | cr4z3d | lucifr, what provider |
| 19:13.14 | lucifr | BroadVoice :( |
| 19:13.46 | [TK]D-Fender | lucifr, and what is this "Providers List" to which you added them? |
| 19:14.12 | cr4z3d | eh nlu2 seems useless for me |
| 19:14.32 | lucifr | in the "Service Providers" in AsteriskNOW |
| 19:14.39 | lucifr | I added throught the GUI |
| 19:14.44 | lucifr | it* |
| 19:14.44 | quidpro | Hmm... is there a way for * to determine if there is Vmail on an analog line (telco vmail system) using Sangoma A200? |
| 19:14.47 | emergence | Anyone can help me? Is the chan_jingle channel driver supposed to come with asterisk? |
| 19:15.06 | emergence | cr4z3d, you can plug in a tv tuner and turn it into a |
| 19:15.07 | emergence | PVR |
| 19:15.23 | [TK]D-Fender | quidpro, Not AFAIK. there is no zaptel functionality for this. |
| 19:15.28 | emergence | you can even install asterisk on it |
| 19:15.31 | cr4z3d | already have a computer dedicated to that |
| 19:15.40 | emergence | oh ok |
| 19:15.55 | cr4z3d | but installing asterisk on something other than my laptop would probably be useful |
| 19:15.56 | quidpro | TK: Damn. I'm wondering whats the best setup... if all 4 analog channels are busy, to have it dump to telco voicemail or just give a busy signal |
| 19:16.17 | Strom_M | quidpro: or forward to a VoIP DID |
| 19:16.26 | Strom_M | and do your own VM |
| 19:16.31 | quidpro | Strom: Not a bad idea at all. |
| 19:16.43 | cr4z3d | voip is the shiznit |
| 19:16.57 | emergence | Im trying to setup gtalk with asterisk and I'm having trouble |
| 19:17.00 | emergence | anyone had any success? |
| 19:17.09 | quidpro | might have to look into that, might be a decent alternative than going partial PRI. |
| 19:17.15 | *** join/#asterisk Modcuts (n=Moducts@88-111-66-94.dynamic.dsl.as9105.com) |
| 19:17.19 | lucifr | Where does AsteriskNOW write its configuration? I opened the sip.conf file and I didn't see my VoIP provided I added fron the GUI. |
| 19:17.20 | [TK]D-Fender | quidpro, Busy it is |
| 19:17.21 | quidpro | TK: BTW, Telus never e-mail me back. |
| 19:17.46 | cr4z3d | where's a good place to find a list of voip providers that allow you to specify the caller id and use pre paid |
| 19:17.50 | [TK]D-Fender | quidpro, Telus = coplete ass-hats |
| 19:18.05 | SkramX | Strom_M: the "context" variable would be the outbound dialplan for the phone |
| 19:18.11 | wunderkin | they have hats for their ass? |
| 19:18.29 | quidpro | TK: THey are probably still thinking "we only offer P-R-I service and not PRI" |
| 19:18.35 | Strom_M | SkramX: I'm not an idiot; I know how asterisk works |
| 19:18.58 | SkramX | ;) I forget to append ", correct"? |
| 19:19.00 | *** join/#asterisk j-voip (n=j@124.29.253.36) |
| 19:19.18 | SkramX | thanks though |
| 19:19.35 | j-voip | hey i had a quick question -- is SIP over TCP supported yet, or are there any plans on the horizon for this? |
| 19:19.38 | lucifr | anybody? |
| 19:19.49 | [TK]D-Fender | lucifr, Check the "users.conf" file. |
| 19:19.49 | quidpro | j-voip: It's out there, AFAIK... thogh not widespread |
| 19:19.54 | Strom_M | j-voip: why the hell would you want to do SIP over TCP? :) |
| 19:19.56 | lucifr | I did |
| 19:20.00 | j-voip | lol! |
| 19:20.03 | Strom_M | SkramX: what specific problem are you having? |
| 19:20.14 | j-voip | unfortunately, this one piece of software i need to get running ONLY supports SIP over TCP |
| 19:20.15 | [TK]D-Fender | lucifr, That is one of the useless new add-ons of 1.4 Leads to much confusion and running in circles... |
| 19:20.17 | j-voip | if you can imagine that |
| 19:20.19 | lucifr | I added my provider manually in the sip.conf file and no luck |
| 19:20.30 | j-voip | quidpro: any idea where i can get it from? |
| 19:20.50 | quidpro | j-voip: Nope. I've seen mention of it.. but can't recall where. |
| 19:21.01 | [TK]D-Fender | j-voip, * does not do SIP over TCP yet. no idea if there are any hopes/placs to |
| 19:21.07 | Modcuts | Does anybody know if you can or can't use includes with realtime extensions? |
| 19:21.17 | [TK]D-Fender | perhaps on Mantis somewhere... |
| 19:21.29 | SkramX | Strom_M: I don't even know if my phone and the * box are talking to eachother. After I get that working, I want to be able to press 9 and playback a soundfile (just to test it all) |
| 19:21.49 | [TK]D-Fender | Strom_M, SIP over TCP is a great way around NAT, etc... |
| 19:22.05 | Strom_M | SkramX: what happens when you type "sip show peers" at the CLI? |
| 19:22.09 | SkramX | Strom_M: http://pastie.caboo.se/49285 |
| 19:22.33 | Strom_M | your phone isnt talking to asterisk |
| 19:22.38 | Strom_M | are they on the same network segment? |
| 19:22.42 | SkramX | nope |
| 19:22.50 | *** join/#asterisk flewid (n=flewid@mail.allgood-domains.com) |
| 19:22.52 | flewid | sup |
| 19:22.52 | SkramX | asterisk box is in a datacenter; phone is here at my house |
| 19:22.57 | Strom_M | which one is behind a NAT? |
| 19:23.13 | Strom_M | or are both of them behind a NAT? |
| 19:23.15 | SkramX | if anything, my phone would be |
| 19:23.18 | SkramX | I think |
| 19:23.29 | cr4z3d | i bet they both are |
| 19:23.33 | SkramX | okay, maybe so |
| 19:23.47 | Strom_M | SkramX: is this that virtual server nonsense you were going on about a few months ago? |
| 19:23.47 | SkramX | haha |
| 19:23.49 | flewid | got a quick question - one of my clients has a box colo'd at his datacenter, we have 2 DID providers coming in to the box, so in the situation that for instance: caller calls in over IP using one of his DID's, then his deskphone rings over IP out from the * box, but he doesn't answer, then it rings his cell phone and he picks up the call on his cellphone |
| 19:23.52 | Qwell | Strom_M: do any cell companies not suck? |
| 19:23.55 | Qwell | any, at all |
| 19:24.02 | quidpro | Grr, I hate corporate cost cutting... the newest Polycom 430 I have, the keys feel "looser" and cheaper than the other 3 phones sitting in front of me. |
| 19:24.04 | Strom_M | Qwell: t-mobile seems to be fairly ok |
| 19:24.06 | flewid | now, whenever that happens and he's talking to a client on his cell, the call quality is always piss poor |
| 19:24.14 | Qwell | Strom_M: t-mobile is raising sms rates |
| 19:24.15 | flewid | is there any tips or tricks to make it sound better for him? |
| 19:24.17 | SkramX | Strom_M: nonsense? |
| 19:24.33 | flewid | (i've tried implicit setting gsm vs ulaw, seemed to make no difference, and he claims that ulaw actually sounds better anyway) |
| 19:24.43 | Qwell | and I found out last night that customer service is good until you have a problem |
| 19:24.43 | Strom_M | well, ulaw does sound better |
| 19:24.48 | flewid | but there is a lot of "talkover" and his clients saying "you still there" etc |
| 19:24.58 | SkramX | Asterisk works swell in a VPS; this is my first time dealing with physical phones. I have customers doing it for their datacenter solutions with phones in S. America and the servers are in Dallas |
| 19:25.01 | flewid | strom: yah, no shit :) but i mean, if it was to a cell, gsm should work just as good no? |
| 19:25.16 | Qwell | they told me that I wasn't an important enough customer to talk to the retention dept :D |
| 19:25.17 | Strom_M | flewid: the fewer transcodes you do, the better |
| 19:25.34 | flewid | strom: yah, that's why i tried gsm the whole way, and ulaw the whole way |
| 19:25.35 | Qwell | "We're sorry sir, that department is reserved for long-time customers" |
| 19:25.40 | Strom_M | flewid: ulaw -> gsm -> ulaw -> gsm is going to sound shittyu |
| 19:25.45 | flewid | obviously to his cisco it should be ulaw cause that's internal |
| 19:26.01 | flewid | okay yeah, and right now it's ulaw -> gsm (unless i'm mistaken on the path?) |
| 19:26.09 | Strom_M | becasuse dont forget that the connection to the mobile phone company is going to be ulaw over a DSx |
| 19:26.17 | flewid | okay |
| 19:26.19 | flewid | so if i have |
| 19:26.37 | flewid | ip (did) ulaw -> * box (ulaw) -> cell phone (ulaw) |
| 19:26.44 | flewid | what else could be a factor of suckyness? :) |
| 19:26.45 | Strom_M | Qwell: I send like five SMSes a month. i dont really care |
| 19:26.54 | Qwell | I send like 2, but that's not the point |
| 19:27.10 | Qwell | the fact that they won't let customers cancel, because it's an "optional" service |
| 19:27.17 | flewid | he's about 20ms from the server, so I don't think it's lag in any way |
| 19:27.20 | Qwell | (nevermind the fact that you can't deny incoming messages) |
| 19:27.26 | flewid | any calls direct to /from his deskphones work great |
| 19:27.31 | flewid | it's just inbound through * to his cell that suck |
| 19:27.50 | Strom_M | what about calls from the deskphone to the mobile phone? |
| 19:28.04 | flewid | they sound fine too, but that's not the same path |
| 19:28.08 | Qwell | Strom_M: Did that law ever pass where you have a 30 day grace period on new cell contracts? |
| 19:28.11 | flewid | cause he's dialing out of a sep DID into his cell on their network |
| 19:28.18 | flewid | his * box isn't involved on the inbound, only out |
| 19:28.33 | Strom_M | flewid: ok, maybe i'm not understanding you |
| 19:28.39 | j-voip | thanks all |
| 19:28.42 | j-voip | bye! |
| 19:28.43 | *** part/#asterisk lucifr (n=chatzill@cpe-75-85-54-211.socal.res.rr.com) |
| 19:28.46 | flewid | heh, yeah it's sorta confusing |
| 19:28.59 | Strom_M | no, you're just not explaining it clearly |
| 19:29.14 | Strom_M | do all direct-dialed calls from the deskphone route the same way? |
| 19:29.29 | flewid | my question is this - besides transcoding and codecs, what other areas of * can i look at to see why he's getting lag/talkover on calls to his cell from the * box |
| 19:29.36 | flewid | yah they do |
| 19:29.42 | Strom_M | i'm trying to help you troubleshoot. |
| 19:30.00 | Strom_M | so, direct-dialed calls go out over a VoIP provider? |
| 19:30.00 | flewid | let me see if i can articulate this a bit better for you :) |
| 19:30.01 | flewid | just a sec |
| 19:30.03 | SkramX | brb |
| 19:30.09 | Strom_M | do forwarded calls route the same way? |
| 19:30.29 | Strom_M | flewid: don't bother re-explaining |
| 19:30.32 | Strom_M | just answer my questions |
| 19:31.53 | flewid | haha fine :) i was drawing you a diagram to make it super quick |
| 19:32.04 | flewid | but yeah, yes, they are going out over voip, or in over voip |
| 19:32.24 | flewid | so if he gets a call to *, but he doesn't answer, it then dials back out over the same provider and calls his cell |
| 19:32.30 | Strom_M | do forwarded calls route the same way as direct dialed calls from the asterisk box forward? |
| 19:32.33 | flewid | so it's voip -> * -> voip -> cell |
| 19:32.55 | flewid | yes |
| 19:33.08 | flewid | we've also tried setting the out to go over a different provider, but that made no diff either |
| 19:33.18 | *** join/#asterisk Zefk (i=Zefk@81.181.249.106) |
| 19:33.24 | Strom_M | exactly how are you doing the forwarding on the asterisk box? |
| 19:33.29 | *** join/#asterisk Hansin321 (i=Eric@c-67-176-41-27.hsd1.co.comcast.net) |
| 19:34.43 | flewid | exten => s,n,Dial(SIP/lesnet_peer/<NUMBERHERE>&IAX2/tony&SIP/7000,32,twTm) |
| 19:34.46 | flewid | is an example of one of them |
| 19:34.55 | flewid | it's just dialing all three and he picks up whichever one he's at |
| 19:34.58 | *** join/#asterisk xtr-II (n=94752345@S0106000c41ed11e1.vf.shawcable.net) |
| 19:35.42 | Strom_M | try turning off teh twTm nonsense |
| 19:35.50 | flewid | already have :/ |
| 19:36.11 | Strom_M | what version of asterisk is this? |
| 19:36.15 | flewid | 1.2.9.1 |
| 19:36.31 | flewid | no 1.2.16 |
| 19:36.36 | flewid | sorry forgot i had upgraded it |
| 19:36.37 | *** join/#asterisk svenna_ (n=svenna@p548d0c8e.dip0.t-ipconnect.de) |
| 19:36.46 | Strom_M | what happens if you have the call tandem through asterisk directly without dialing the other phones as well? |
| 19:37.20 | flewid | sorry, misunderstand, like make it dial in and just route right through and outdial to his cell? |
| 19:37.22 | flewid | same diff |
| 19:37.31 | Strom_M | yes |
| 19:37.35 | Strom_M | have you actually tried that? |
| 19:37.35 | crashev | I recorded my temporary greetings message for voicemail but after it I still hear default vm-intro - anyone knows how to disable this? |
| 19:37.38 | flewid | yup |
| 19:37.57 | Strom_M | which provider is this? |
| 19:38.03 | flewid | he's using uhm cingular |
| 19:38.03 | Strom_M | and are you using IAX2 or SIP? |
| 19:38.05 | Strom_M | no |
| 19:38.07 | Strom_M | voip provider |
| 19:38.10 | flewid | ah, lesnet |
| 19:38.26 | flewid | and iax2 or sip, makes no diff (we have a few providers we've tried to debug this with) |
| 19:38.37 | Strom_M | what is the total bandwidth of the circuit you're using for entrance facilities? |
| 19:38.49 | flewid | i haven't actually heard his phone myself (i'm 4000 miles away) so it COULD be just him being too anal |
| 19:39.02 | *** join/#asterisk L|NUX (n=linux@unaffiliated/lnux/x-10290) |
| 19:39.12 | Strom_M | uhm |
| 19:39.21 | *** join/#asterisk Cybertoy (n=cybertoy@dsl254-123-112.nyc1.dsl.speakeasy.net) |
| 19:39.24 | Strom_M | have you tried troubleshooting this with your OWN telephone? |
| 19:39.53 | flewid | why? that would be worse no? calling out of his box 4000 miles away over voip to my cell here? |
| 19:40.13 | Strom_M | have you /tried it/? |
| 19:40.21 | flewid | no i haven't. |
| 19:40.26 | Strom_M | why don't you do that. |
| 19:40.48 | Cybertoy | hi .. I'm using AgentCallbackLogin for my callcenter agents... anyone know if it's possible to play an audio file before connecting the customer to the agent - i.e. play it to the agent "you are receiving a call from our callcenter now" ?? |
| 19:41.06 | flewid | hrm, k, sec |
| 19:44.41 | *** join/#asterisk plasmid (n=noway@c-24-127-166-193.hsd1.pa.comcast.net) |
| 19:46.34 | wunderkin | 24-48 hours |
| 19:47.02 | flewid | hmm, interesting, yeah, there's like 3 seconds of lag to my cell |
| 19:47.08 | *** join/#asterisk nortex (n=chatzill@adsl-69-149-171-69.dsl.amrltx.swbell.net) |
| 19:47.33 | Strom_M | that's fairly abysmal |
| 19:47.44 | Strom_M | and yet if you call directly from the PBX, what is the lag? |
| 19:48.14 | flewid | interesting, i had my gf talking on the cell and the 7970 on speaker, and if she would just talk, there was about 1 second of lag |
| 19:48.24 | flewid | but once i started to talk back as well, it jumped up to like 4 |
| 19:48.44 | flewid | coudl this be box speed related? |
| 19:48.46 | Strom_M | um |
| 19:48.47 | ManxPower | flewid: now THAT sounds like a bandwidth problem |
| 19:48.58 | Strom_M | it sounds like You Have Issues(tm) |
| 19:49.04 | Strom_M | and also, I should get lunch |
| 19:49.05 | Strom_M | adios |
| 19:49.09 | flewid | thanks strom :) |
| 19:49.14 | ManxPower | Strom_M: more than just VoIP related ones too |
| 19:49.30 | flewid | manx: yah, tell me about it, mind you, i am half way across the country from him so, like i said it's not like this wasn't expcted |
| 19:49.44 | flewid | i'm not wondering if it's box speed related tho, he's got this running off an old laptop |
| 19:49.56 | *** join/#asterisk michaelm2 (n=spam@c-75-69-12-225.hsd1.vt.comcast.net) |
| 19:50.18 | [TK]D-Fender | crashev, Quite likely the way you are calling your box. apste the line. |
| 19:50.28 | flewid | and hmm, just did it again, and my call sounded fine this time too |
| 19:50.39 | flewid | meh, thanks guys |
| 19:51.31 | nortex | Does anyone have chan_cellphone working on Centos 4.3? |
| 19:52.49 | *** join/#asterisk mog (n=mog@c-71-207-200-130.hsd1.al.comcast.net) |
| 19:52.49 | *** mode/#asterisk [+o mog] by ChanServ |
| 19:53.25 | michaelm2 | I'm looking for help with tuning my fxo cards? I've tried ztmonitor 1004hz test phone number and fxotune but am fustrated that I am doing both wrong |
| 19:53.40 | michaelm2 | I'm a newbie and would appreciate help |
| 19:54.22 | michaelm2 | can somebody please help me |
| 19:56.34 | [TK]D-Fender | michaelm2, Perhaps you could tell us whats actually wrong at this point... |
| 19:58.07 | *** join/#asterisk bkruse_home (n=kruz@69.73.127.92) |
| 19:59.59 | michaelm2 | Matching the rxgain with ztmonitor to the quantative 14500 on a 1004hz 0db test leaves a constant hiss/fuzz |
| 20:00.46 | michaelm2 | I believe the number is too high |
| 20:01.26 | michaelm2 | what I'm not sure of is if the tone is actually 1004hz |
| 20:01.42 | michaelm2 | the phone company tech called it a quiet line |
| 20:02.45 | michaelm2 | but searching on google for other 1004hz test numbers that are further in prxcimity seem to sound the same by ear |
| 20:02.56 | crashev | [TK]D-Fender: I'm calling one of the users -> it plays Playing '/var/spool/asterisk/voicemail/default/1234/temp' and thats ok, but after that it also plays Playing 'vm-intro' , how to remove vm-intro from playing when the voicemail responds? |
| 20:03.12 | michaelm2 | my number is (201) 666-9966 |
| 20:03.23 | [TK]D-Fender | crashev, "s" <- |
| 20:03.31 | michaelm2 | meaning the local quiet line |
| 20:04.20 | *** join/#asterisk Xen^ (n=linux@unaffiliated/lnux/x-10290) |
| 20:04.22 | [TK]D-Fender | michaelm2, Hrm, not much I can say for this... |
| 20:07.16 | crashev | [TK]D-Fender: "s" ? |
| 20:07.43 | crashev | a ok |
| 20:07.44 | crashev | works |
| 20:07.46 | crashev | thx |
| 20:08.49 | [TK]D-Fender | <PROTECTED> |
| 20:12.45 | michaelm2 | Ok... I have a question regarding fxotune -> What is meant by "clear the dialtone" |
| 20:15.39 | *** join/#asterisk Faithful (n=Faithful@ns.linuxterminal.com) |
| 20:19.53 | *** join/#asterisk syzygyBSD (n=chatzill@66.226.228.204.cpe.speedyquick.net) |
| 20:31.30 | *** join/#asterisk Assid (n=assid@59.183.15.205) |
| 20:39.21 | *** join/#asterisk SuperID (n=gary@ldlwvtbas01-pool2-a135.ldlwvt.tds.net) |
| 20:50.18 | *** join/#asterisk ilan (i=ilan@69.60.110.251) |
| 20:51.58 | ilan | hello, anyone have issues with FWD? seems that IAX registration stops working every few days. it shows as "timed out". |
| 21:01.08 | d00gster | guys, how do I check whether or not my sip trunking is working? |
| 21:02.44 | mvanbaak | try it |
| 21:02.50 | *** join/#asterisk r0d3nt (i=nobody@foster.stonedcoder.org) |
| 21:02.55 | bkruse_home | mvanbaak: thats what i was thinking :P |
| 21:02.55 | mvanbaak | place and receive calls |
| 21:02.57 | bkruse_home | sup r0d3nt |
| 21:03.05 | mvanbaak | bkruse_home: ;( |
| 21:03.08 | mvanbaak | ;) |
| 21:03.12 | mvanbaak | hhmm |
| 21:03.14 | mvanbaak | coffeeeeeeeeeeeeeeeee |
| 21:03.20 | bkruse_home | YAY |
| 21:03.28 | bkruse_home | coffee++ |
| 21:03.33 | bkruse_home | jbot: coffee++ |
| 21:03.39 | bkruse_home | ~karma coffee |
| 21:03.39 | jbot | coffee has karma of 1 |
| 21:03.41 | bkruse_home | :D |
| 21:03.48 | mvanbaak | while (true) cofee++; |
| 21:04.04 | bkruse_home | jbot: coffee++++++++++++++++++++++++++++++++++++++++++ |
| 21:04.06 | bkruse_home | ~karma coffee |
| 21:04.06 | jbot | coffee has karma of 1 |
| 21:04.09 | bkruse_home | ouch |
| 21:04.29 | mvanbaak | hhmm |
| 21:04.32 | mvanbaak | coffee++ |
| 21:05.12 | *** join/#asterisk nuonguy (n=john@c-24-6-175-26.hsd1.ca.comcast.net) |
| 21:05.25 | *** join/#asterisk r0d3nt (i=nobody@foster.stonedcoder.org) |
| 21:10.08 | *** join/#asterisk frenzy_ (n=frenzy@unaffiliated/frenzy) |
| 21:16.00 | [TK]D-Fender | ~karma [TK]D-Fender |
| 21:16.00 | jbot | [tk]d-fender has karma of 5 |
| 21:16.08 | [TK]D-Fender | Me > coffee :) |
| 21:16.24 | Qwell | jbot: karma Qwell |
| 21:16.24 | jbot | qwell has karma of 7 |
| 21:16.27 | Qwell | :D |
| 21:16.42 | [TK]D-Fender | Qwell, My karma ran over your dogma :D |
| 21:16.47 | cr4z3d | ~karma cr4z3d |
| 21:16.47 | jbot | cr4z3d has neutral karma |
| 21:16.51 | Qwell | jbot: dogma Qwell |
| 21:17.05 | cr4z3d | what's karma |
| 21:22.55 | *** join/#asterisk nuonguy (n=john@c-24-6-175-26.hsd1.ca.comcast.net) |
| 21:25.38 | garreel | anyone known if it's possible to have a soft sip phone to receive calls without setting host = dynamic in sip.conf? |
| 21:26.40 | garreel | I tried assigning a static ip but it doesn't receive |
| 21:26.55 | cr4z3d | what's wrong with host = dynamic |
| 21:27.37 | garreel | I want to assing a particular extension to a particular ip |
| 21:27.47 | garreel | so i have static ips |
| 21:28.12 | [TK]D-Fender | <PROTECTED> |
| 21:28.42 | *** join/#asterisk beluga9 (n=redbelug@74.12.130.193) |
| 21:29.26 | garreel | but i don't that extension picked up by a different pc |
| 21:29.47 | *** join/#asterisk Eil (n=dotpage@207-148-194-23.static.sfldmi.bullseyetelecom.net) |
| 21:29.51 | [TK]D-Fender | garreel, And why would that PC be using the user & pass that belongs to the one i question? |
| 21:30.18 | [TK]D-Fender | garreel, And you can allow/deny IP's if you really feel you HAVE to, but this is sounding a littel paranoid... |
| 21:30.37 | garreel | but I'm a lot paranoid ;-) |
| 21:30.49 | cr4z3d | if i had multiple people using xlite couldn't i do [xlite] user=blah secrete=blah and have multiple [xlite] |
| 21:31.19 | garreel | already doing that |
| 21:32.20 | [TK]D-Fender | cr4z3d, No, you should be doing it more like [xlite1] , secret=1234, [xlite2], secret=5678, etc |
| 21:32.24 | garreel | but someone (an agent) could steal the secret from xlite conf |
| 21:32.27 | [TK]D-Fender | cr4z3d, not using "user" |
| 21:32.47 | cr4z3d | oh.. hm i have it setup w/ user right now i was under the impression i had to |
| 21:33.11 | garreel | it's not that difficult to steal a password from xlite conf |
| 21:33.40 | cr4z3d | well if it's only accessible to root on a linux box |
| 21:33.45 | [TK]D-Fender | garreel, Ok, so have you tried host=1.2.3.4? |
| 21:33.47 | cr4z3d | you have more things to worry about wouldn't you |
| 21:33.49 | garreel | oh.. no... |
| 21:34.00 | garreel | yes I tried host= 1.2.3.4 |
| 21:34.06 | garreel | i can place calls |
| 21:34.12 | garreel | but i cannot receive |
| 21:34.32 | [TK]D-Fender | garreel, then leave as dynamic and try this : http://www.voip-info.org/wiki/index.php?page=Asterisk+sip+permit-deny-mask |
| 21:35.18 | garreel | thanks ;-) |
| 21:40.19 | *** join/#asterisk eltech (i=G00Ds@ool-457c94a3.dyn.optonline.net) |
| 21:43.13 | garreel | another question... |
| 21:43.22 | garreel | anyone knows what is this stuff: |
| 21:43.26 | garreel | !! Unknown IE 26 (cs6, Unknown Information Element) |
| 21:43.37 | Qwell | Eil: by asking repeatedly, obviously |
| 21:43.47 | [TK]D-Fender | Eil, We heard you the first dozen times. repeating yourself like that will not get you anywhere productive. |
| 21:43.57 | *** part/#asterisk frenzy_ (n=frenzy@unaffiliated/frenzy) |
| 21:44.41 | Eil | sorry |
| 21:44.55 | garreel | I obtain this on cli when placing calls on zap channels |
| 21:44.57 | Eil | thought I wasn't being heard somehow |
| 21:45.34 | garreel | starting when the far endpoint answer the call |
| 21:45.42 | [TK]D-Fender | Eil, appraently you CAN'T. They say how often * should reregister and it does it. end of story. |
| 21:45.50 | garreel | and then every about 25 seconds |
| 21:46.17 | Eil | ty |
| 21:46.51 | garreel | see this: |
| 21:46.57 | garreel | [Mar 24 16:06:58] -- Zap/1-1 is ringing |
| 21:46.57 | garreel | [Mar 24 16:07:04] !! Unknown IE 26 (cs6, Unknown Information Element) |
| 21:46.57 | garreel | [Mar 24 16:07:04] -- Zap/1-1 answered SIP/192.168.1.254-0822d958 |
| 21:46.57 | garreel | [Mar 24 16:07:04] -- fixed jitterbuffer created on channel Zap/1-1 |
| 21:46.57 | garreel | [Mar 24 16:07:30] !! Unknown IE 26 (cs6, Unknown Information Element) |
| 21:46.57 | garreel | [Mar 24 16:07:56] !! Unknown IE 26 (cs6, Unknown Information Element) |
| 21:46.59 | garreel | [Mar 24 16:08:22] !! Unknown IE 26 (cs6, Unknown Information Element) |
| 21:47.01 | garreel | ... |
| 21:47.17 | garreel | [Mar 24 16:17:49] !! Unknown IE 26 (cs6, Unknown Information Element) |
| 21:47.17 | garreel | [Mar 24 16:18:00] -- Hungup 'Zap/1-1' |
| 21:47.17 | garreel | [Mar 24 16:18:00] -- fixed jitterbuffer destroyed on channel Zap/1-1 |
| 21:48.22 | garreel | anyone saw such stuff before? |
| 21:55.38 | [TK]D-Fender | garreel, http://bugs.digium.com/view.php?id=6152 |
| 21:56.30 | [TK]D-Fender | garreel, And next time do NOT spam like that use a pastebin |
| 21:56.30 | [TK]D-Fender | ~pb |
| 21:56.32 | jbot | [pb] a place to paste your stuff without flooding the channel - try http://pastebin.ca, or http://channels.debian.net/paste |
| 21:57.22 | garreel | :-) |
| 21:57.24 | garreel | sorry |
| 21:57.45 | garreel | never heard of a pastebin |
| 21:57.48 | garreel | :-) |
| 21:59.33 | garreel | so it's a still open bug? |
| 22:01.04 | [TK]D-Fender | <PROTECTED> |
| 22:01.47 | garreel | I already tried that... but find little explanation |
| 22:03.21 | [TK]D-Fender | garreel, Funny... I found one in the first 3 google results which implicate your regions signalling. Try HARDER. |
| 22:03.33 | garreel | :-) |
| 22:03.46 | garreel | found also some patches |
| 22:04.20 | garreel | but was thinking can have here some clear hints |
| 22:04.59 | garreel | found this one... |
| 22:05.02 | garreel | http://bugs.digium.com/view.php?id=7494 |
| 22:05.41 | garreel | but still wasn't committed to 1.4 since I'm using 1.4.2 |
| 22:08.28 | plasmid | anyone use a Hitachi IP5000 here? |
| 22:10.21 | SkramX | is there a key press to have my Cisco phone reload/whatever and try to reconnect to the SIP proxy? |
| 22:12.56 | emergence | anyone got asterisk to work with gtalk? |
| 22:14.30 | hijacked | SkramX: yes, it's a button that has a "0" with a "1" super-imposed in it. |
| 22:14.43 | SkramX | :p |
| 22:16.54 | *** join/#asterisk Dovid (n=Dovid@l192-117-114-1.broadband.actcom.net.il) |
| 22:18.57 | [TK]D-Fender | SkramX, Yes, try that switch over there on your wall.... |
| 22:22.55 | SkramX | ;( |
| 22:22.57 | SkramX | brb |
| 22:25.43 | *** join/#asterisk CrashHD (n=crashhd@c-67-166-155-233.hsd1.ca.comcast.net) |
| 22:26.16 | CrashHD | anyway to send sip auth info in a dial string? |
| 22:26.26 | CrashHD | SIP/user:pass@host:5060/exten? |
| 22:27.29 | *** join/#asterisk darylvoip (n=darylvoi@c-71-224-42-97.hsd1.pa.comcast.net) |
| 22:28.40 | Dovid | CrashHD: I dont believe so - y dont u use the standard method of using sip.conf ? |
| 22:29.08 | CrashHD | because I'm not doing something standard |
| 22:29.09 | CrashHD | ? |
| 22:29.27 | CrashHD | looking to dynamically dial different ip addresses for testing purposes |
| 22:29.34 | CrashHD | I can setup a general user on each switch |
| 22:29.42 | CrashHD | then with a numeric entry have my phone dial in |
| 22:29.42 | *** join/#asterisk quidpro (n=quid@CPE00131078ba5d-CM000f9f7eff1e.cpe.net.cable.rogers.com) |
| 22:29.44 | CrashHD | hit disa |
| 22:29.52 | CrashHD | to test dial plans as if I were a phone |
| 22:30.04 | CrashHD | but I need to pass auth info in a sip dial string |
| 22:30.19 | CrashHD | so that it remains secure |
| 22:30.19 | Dovid | never tried it that way |
| 22:30.27 | Dovid | what happend when u tried it ? |
| 22:30.34 | CrashHD | I get auth failed |
| 22:30.40 | CrashHD | haven't debugged it yet |
| 22:30.46 | CrashHD | was just curious if anyone had ran in to it |
| 22:32.49 | quidpro | hmm... anybody recommend a multi-line telephone-line surge protector? Seems like all the "all in one" solutions have one or two lines, need something for 4+ lines, |
| 22:34.11 | CrashHD | voip? |
| 22:34.12 | CrashHD | heh |
| 22:34.38 | CrashHD | can't beat the protectiveness of not having analog lines? |
| 22:34.39 | CrashHD | :) |
| 22:34.58 | florz | High voltage packets are dangerous! |
| 22:35.18 | darylvoip | Have you looked into the type that work as a bridge clip on your 110 block? (if you happen to have access to that) |
| 22:35.34 | Qwell | 8 block? heh |
| 22:35.37 | darylvoip | You can even get them with nifty lights that come on when the line is in use. |
| 22:36.02 | darylvoip | 8 block? what do you mean? |
| 22:36.26 | [TK]D-Fender | CrashHD, Yes, that format looks fine |
| 22:37.16 | CrashHD | [TK]D-Fender, it's the standard sip uri format but asterisk doesn't seem to use it? |
| 22:37.17 | [TK]D-Fender | quidpro, 4 cheap surge power bars :) |
| 22:37.29 | [TK]D-Fender | CrashHD, You should be fine with it. TRY |
| 22:37.38 | CrashHD | I am trying? |
| 22:37.55 | CrashHD | wouldn't be asking if I wasn't running in to problems I couldn't explain |
| 22:38.02 | darylvoip | You could actually use just two...most of them are "2 line" |
| 22:38.09 | CrashHD | your time is too precious to waste |
| 22:39.45 | CrashHD | it's not even matching the user |
| 22:40.13 | CrashHD | http://www.pastebin.ca/408562 |
| 22:40.18 | quidpro | darylvoip: Yeah I have access to the block... haven't seen those bridge clip types |
| 22:40.42 | darylvoip | quidpro: trying to find the supplier I've gotten them from before fo you |
| 22:40.50 | quidpro | appreciated |
| 22:41.57 | darylvoip | Sorry...not thinking straight today....I mean 66 block, not 110. I don't know that you could get them on a 110 block, but we're probably still talking about the same thing. |
| 22:42.28 | quidpro | TK: Is there a way to add a label to softkeys, if you remap them to another function on the Polycoms... would be nice, but doesn't look possible. |
| 22:42.49 | quidpro | daryl: yeah, if they make one for 66, they make it for 110 |
| 22:43.36 | [TK]D-Fender | quidpro, Nope |
| 22:43.39 | darylvoip | ITW Link... |
| 22:43.46 | darylvoip | here's some random place to get them online |
| 22:43.48 | darylvoip | http://www.allthingspos.com/itw-linx/up3b-235-ultralinx-66-block.htm |
| 22:43.54 | darylvoip | I think I got them from graybar |
| 22:44.00 | quidpro | Daryl: Thanks. |
| 22:45.08 | Dovid | anyone see this update ? |
| 22:45.09 | Dovid | http://news.zdnet.com/2100-1035_22-6169950.html?part=rss&tag=feed&subj=zdnn |
| 22:45.39 | *** join/#asterisk clive- (n=pirch@dsl-242-191-13.telkomadsl.co.za) |
| 22:45.59 | clive- | can anyone help me, what is wrong with this? :.. svn checkout http://svn.digium.com/svn/zaptel/branches/1.2 zaptel-1.2.16 |
| 22:47.16 | Qwell | clive-: well, that isn't 1.2.16 for one thing |
| 22:47.49 | clive- | qwell:), hi, it isnt? |
| 22:47.58 | Qwell | no, it isn't |
| 22:48.20 | clive- | dazed and confused:) |
| 22:49.14 | clive- | does 1.2.16 not exist in the 1.2 tree? |
| 22:49.28 | Qwell | no it doesn't |
| 22:49.42 | Qwell | svn co http://svn.digium.com/svn/zaptel/tags/1.2.16 zaptel-1.2.16 |
| 22:50.48 | clive- | thanks....let me try that..:) |
| 22:53.08 | clive- | it wants me to do a svn update rather... how do I keep a copy of the old version for a fallback |
| 22:55.11 | Gido-E | make a copy |
| 22:56.31 | *** join/#asterisk tb0301s (n=sowa@brln-4db1100f.pool.einsundeins.de) |
| 22:57.14 | clive- | gido, hi,,,it seems the tags have changed and update doesnt want to work... |
| 22:57.25 | clive- | oh well... |
| 22:57.53 | *** part/#asterisk SuperID (n=gary@ldlwvtbas01-pool2-a135.ldlwvt.tds.net) |
| 23:01.00 | CrashHD | anyone lend some help on getting a simple system to system peer call going (without having two sip.conf entries)? |
| 23:04.26 | Dovid | CrashHD: try the users list. u can also do it based on IP |
| 23:04.38 | *** join/#asterisk luke-jr (n=luke-jr@2002:1891:f663:0:20e:a6ff:fec4:4e5d) |
| 23:08.23 | *** join/#asterisk sabakas1 (n=solapus@66.90.121.129) |
| 23:09.30 | clive- | i am feeling quite stupid,,,svn doesnt want to checkout the code.. |
| 23:09.36 | *** join/#asterisk shido6 (n=shido6@d221-68-200.commercial.cgocable.net) |
| 23:12.26 | *** join/#asterisk lesouvage (n=lesouvag@cc341200-a.assen1.dr.home.nl) |
| 23:12.49 | clive- | qwell any more pointers will be appreciared |
| 23:13.04 | clive- | its giving me a message about a running verison |
| 23:13.27 | clive- | version, but...when I try svn update the running version, it gives an error about the tags |
| 23:14.15 | lesouvage | I build Asterisk 1.4 from source and comfigured mailboxes. Mailing the messages doesn't work. I found this in maillog " DSN: Data format error". Any idea? |
| 23:18.01 | clive- | doing the tarball method and finished |
| 23:23.29 | *** join/#asterisk j3g (i=nobody@201.73.174.143) |
| 23:24.36 | j3g | i am trying to get broadvoice to work with asterisk ... when I try dialing out, asterisk shows: Called Broadvoice SIP/XXXXXXXX (phone #) |
| 23:25.10 | j3g | and stays like that for a while... then drops saying that no channels were avail. |
| 23:26.13 | j3g | sip show channels show: |
| 23:26.38 | j3g | the channel |
| 23:26.47 | j3g | how can I tell why the call is being dropped? |
| 23:32.08 | clive- | ok, ...I think I got thinsg figured out...is there a way to check which version of zaptel is currently in use? |
| 23:36.38 | CrashHD | heh asterisk doesn't consider 1.1.1.001 a valid address |
| 23:36.38 | clive- | please stop shouting all at once |
| 23:36.45 | clive- | :) |
| 23:38.08 | *** join/#asterisk fnordus (n=dnall@24.85.128.203) |
| 23:49.17 | CrashHD | $[${EXTEN:3} ~ ([1-9]+)] |
| 23:49.30 | CrashHD | with EXTEN = *50001 |
| 23:49.33 | CrashHD | is returning 001 |
| 23:49.34 | CrashHD | not 1 |
| 23:49.38 | CrashHD | any one have any idea why? |
| 23:50.03 | clive- | crash try exten:1 |
| 23:50.16 | CrashHD | I don't want it to match any of the *50 |
| 23:51.08 | CrashHD | hmm oops I'm using the wrong operator |
| 23:51.14 | CrashHD | let me double check it |
| 23:51.38 | *** join/#asterisk Dovid (n=Dovid@l192-117-114-1.broadband.actcom.net.il) |
| 23:51.52 | CrashHD | hmm |
| 23:52.01 | CrashHD | using =~ it matches just 0 now |
| 23:52.02 | CrashHD | lol |
| 23:53.39 | clive- | i am not so good with dialplan stuff like that., so sorry, |
| 23:54.40 | CrashHD | no worries |
| 23:54.42 | CrashHD | anyone else? |
| 23:55.22 | clive- | its very quiet in here, I was asking my own silly questions earlier with like almost zero response |
| 23:56.02 | CrashHD | saturday, everyone is off relaxing |
| 23:56.53 | clive- | i just upgraded my zaptel to teh new version, but I am not sure that it is the new version that is being used...:) |
| 23:57.19 | MrTelephone | rmmod zaptel |
| 23:57.24 | MrTelephone | modprobe zaptel |
| 23:57.35 | MrTelephone | make sure you make install first |
| 23:58.10 | clive- | mrtelephone, hi, I have done all that, ..I was like hoping for some thing like "show zap version" or something |
| 23:59.00 | MrTelephone | ahhh |
| 23:59.02 | MrTelephone | wierd |