| 00:00.07 | Druken | near the nasa training camp? :) |
| 00:00.50 | mog_work | not really |
| 00:00.52 | mog_work | but close |
| 00:00.57 | Druken | hehe |
| 00:01.08 | hardwire | fucking hell |
| 00:01.33 | Druken | you rang? |
| 00:02.39 | kippi | ok cool |
| 00:02.41 | kippi | thanks |
| 00:03.39 | kippi | anyone know what time it is there? and wot time they leave? |
| 00:03.48 | mog_work | it is where? |
| 00:03.49 | mog_work | here |
| 00:03.53 | mog_work | its 6:03 |
| 00:04.10 | mog_work | we officially leave in 57 minutes |
| 00:04.13 | mog_work | we actually leave nevre |
| 00:04.19 | kippi | ah coo |
| 00:04.20 | kippi | l |
| 00:05.02 | mog_work | hey thats my face |
| 00:05.07 | hardwire | heh |
| 00:05.17 | file[laptop] | mog_work: Matttttttt |
| 00:05.23 | *** join/#asterisk TheCops (i=nobody@got.securebinary.com) |
| 00:05.25 | mog_work | Fileeeeeeee |
| 00:05.44 | file[laptop] | mog_work: how was your day? |
| 00:05.54 | mog_work | fantastical |
| 00:06.01 | mog_work | finaly got a straight answer from linksys |
| 00:06.07 | file[laptop] | oh oh oh and |
| 00:06.20 | mog_work | the sip device talking on the lan ports will never be supported. |
| 00:06.31 | Druken | was it truely a straight answer? or a roundabout straight answer? |
| 00:06.50 | mog_work | very direct answer |
| 00:06.59 | hardwire | mofo |
| 00:07.02 | hardwire | ofom |
| 00:07.16 | mog_work | said it would take a hardware upgrade and it wasnt gonna happen, but in reality I imagine it is just software and they arent gonna do it |
| 00:07.37 | mog_work | they did say there would be products int the future etc |
| 00:08.12 | file[laptop] | there's always products in the future, what kind of company would they be if there wasn't |
| 00:09.02 | mog_work | a dead one? |
| 00:09.10 | mog_work | or an oil company? |
| 00:11.21 | *** join/#asterisk phifli (n=mike@user-69-73-44-160.knology.net) |
| 00:15.20 | *** join/#asterisk Faithful (n=Faithful@202-6-145-116.ip.adam.com.au) |
| 00:16.17 | nextime | ls |
| 00:16.19 | nextime | ops |
| 00:16.21 | nextime | sorry |
| 00:19.08 | *** join/#asterisk luke-jr_ (n=luke-jr@user-0c938qu.cable.mindspring.com) |
| 00:19.11 | morale | rm |
| 00:19.13 | morale | -rf |
| 00:19.19 | *** join/#asterisk shmaltz (n=chatzill@69.28.255.210) |
| 00:20.32 | Druken | god damn flying is expensive... |
| 00:20.57 | shmaltz | Druken, where do u want to fly to? |
| 00:21.14 | morale | why do indians like wd-40? because it comes with a straw and shot glass. |
| 00:21.15 | Druken | victoria bc |
| 00:21.24 | morale | victoria sucks., |
| 00:21.39 | Druken | morale: wouldn't know, never been... |
| 00:22.08 | morale | all victoria is good for is getting cheap acid. |
| 00:22.32 | morale | and the wax museum |
| 00:22.38 | Druken | well, no.. if i was flying there for acid, i'd fly in with a cesna |
| 00:22.48 | shmaltz | Druken where from? |
| 00:22.53 | Druken | toronto |
| 00:23.08 | shmaltz | Druken, it should be around $500 Ca am I right? |
| 00:23.25 | shmaltz | you could try with connecting flight thru the states it migth be cheaper |
| 00:23.27 | morale | go to brocket, alberta. thats a happening place, they even got a website www.brocket99.ca |
| 00:23.31 | Druken | west jet posts it at around 350... |
| 00:23.36 | Druken | with a connection in edmonton |
| 00:24.02 | Druken | is that where your from morale? |
| 00:24.38 | morale | druken, no. if you go to brocket you don't come back. |
| 00:25.08 | morale | its one of the small towns on the news with all of the kids huffing gas and stuff. |
| 00:25.24 | morale | im from bc. |
| 00:26.28 | *** join/#asterisk Lurr (i=user@adsl-067-034-122-207.sip.mia.bellsouth.net) |
| 00:26.37 | Druken | :) |
| 00:26.42 | Druken | that's funny that site |
| 00:26.50 | *** join/#asterisk santiago (n=santiago@208.195.215.160) |
| 00:27.27 | morale | it offends 99/100 people. |
| 00:28.16 | docelm0 | YIPPIE! |
| 00:28.22 | TheCops | Someone have tried to install asterisk 1.2.0 on Fedora core 3 ? |
| 00:28.49 | morale | tried i installed asterisk 1.2.0 on fedora core 3 |
| 00:29.52 | bweschke | TheCops: yes |
| 00:30.05 | Druken | morale: why would people get offended? |
| 00:30.19 | TheCops | bweschke, I can't make load the zaptel drive and compiling asterisk |
| 00:30.42 | Druken | not my fault they are a bunch of drunk ass loosers who got nothing better todo than sit on welfare and bitch that "you white man took out land" |
| 00:31.11 | morale | Druken: exactly :) |
| 00:31.20 | morale | We're gonna play some AC/DC for ya. Den we gonna hit it up wit shum more AC/DC. Den maybe after dat we play shum AC/DC for ya. |
| 00:33.39 | Druken | morale: like the hott girls section... hehe |
| 00:33.48 | morale | yeah haha :P |
| 00:33.51 | Druken | i was leary about clicking on it... but it's good :) |
| 00:34.11 | morale | my favorite is "Application to be an Indian" |
| 00:34.29 | Druken | oh i didn't see that... |
| 00:34.41 | Druken | i should fill it out... i'm sure i could pass :) hehe |
| 00:34.55 | morale | haha. "What color is your car" Pink, White, Primer, Rust |
| 00:35.05 | morale | Whos your mudder, Whos your fudder |
| 00:35.06 | Druken | or wait... i'm not a drunk.... shit.. i forgot about that prerequisite |
| 00:35.11 | *** join/#asterisk r0d3nt (i=nobody@wsip-24-234-241-145.lv.lv.cox.net) |
| 00:35.26 | morale | www.brocket99.ca/application.html i think |
| 00:36.51 | Druken | hahaha |
| 00:36.54 | Druken | if living in automobile |
| 00:38.10 | *** join/#asterisk MikeJ[Laptop] (n=ircatjer@d14-69-8-30.try.wideopenwest.com) |
| 00:38.20 | marcus2 | so if i understand this bug 5780 properly, it only affects channels that are using g.729? |
| 00:42.18 | hardwire | poke me if you are an IT geek needing a job |
| 00:42.54 | hardwire | 2+ years of on-site IT experience :) |
| 00:44.22 | mog_work | hrrm hardwire?? |
| 00:44.40 | hardwire | not you.. you need to work at digium and make us happy. |
| 00:44.51 | mog_work | lol |
| 00:45.01 | mog_work | to loyal |
| 00:45.05 | mog_work | and its to nice |
| 00:45.17 | mog_work | but i will do anything for the redbull |
| 00:45.36 | hardwire | they have you by the junk! |
| 00:46.07 | mog_work | heh i suppport my own habit |
| 00:47.04 | *** join/#asterisk brc_ (n=Brian@pdpc/supporter/basic/brc) |
| 00:47.16 | *** join/#asterisk alephcom (n=Miranda@207.34.97.130) |
| 00:47.51 | marcus2 | what sort of "IT geek" ? |
| 00:48.57 | Druken | IT geeks don't need jobs, just more money to blow on electronics and blow up dolls |
| 00:49.29 | mog_work | mmm electronics |
| 00:49.53 | *** join/#asterisk SwK (n=SwK@12-219-151-128.client.mchsi.com) |
| 00:50.33 | orlock | Druken: but you get to play with unaffordable toys when you have a job! |
| 00:51.33 | Druken | orlock: yeah... but you don't always get to do that YOU want with those toys |
| 00:51.55 | orlock | no :-( |
| 00:52.02 | orlock | usually, though :) |
| 00:52.12 | Druken | what the boss don't know won't hurt him... |
| 00:52.22 | Druken | till they get the bandwidth bill... |
| 00:55.30 | *** join/#asterisk zotz (n=zotz@24.231.47.168) |
| 00:57.49 | dos000 | when one does "sip show peers" are you supposed to see all IPs of registered sip endpoints? The problem is that i am only able to see 4 of them |
| 00:58.22 | dos000 | and i am sure the other non registered sip endpoints are making calls .. so i am puzled |
| 00:59.01 | dos000 | BTW all my extensions{.conf} require a passwd |
| 01:03.05 | *** join/#asterisk JunK-Y (n=junky@67.71.158.165) |
| 01:05.09 | docelm0 | Anyone in the need for 1U servers? I have 4 P4 2.8 with 512 of ram |
| 01:05.50 | dos000 | i just got 4 1u ibm delivered :-( |
| 01:07.01 | Druken | how much?? :) |
| 01:09.06 | *** join/#asterisk DrJES (n=macleajb@blk-222-132-53.eastlink.ca) |
| 01:09.38 | Juggie | those arnt servers |
| 01:09.40 | Juggie | they are toys |
| 01:11.40 | dos000 | anyone know why a sip endpoint (ata) would not show up when you do sip show peers ? i am really puzled here |
| 01:12.02 | dos000 | it could be a bug from the looks of it |
| 01:13.24 | *** join/#asterisk jcath (n=skycat@159.226.21.127) |
| 01:13.51 | *** join/#asterisk jeffik (n=Jeff@CPE0011505c92d3-CM014350000760.cpe.net.cable.rogers.com) |
| 01:14.03 | *** part/#asterisk [hC] (n=hardcore@209.153.195.139) |
| 01:16.59 | *** join/#asterisk joat (n=joat@ip70-160-150-20.hr.hr.cox.net) |
| 01:19.00 | docelm0 | 2000 for all or 600 each |
| 01:19.34 | docelm0 | I am getting rid of my colo and I need to get rid of these boxes.. Perfect condition.. Just pulled from production |
| 01:30.23 | *** join/#asterisk brookshire (n=nubb@gateway.digium.com) |
| 01:30.24 | bweschke | docelm0: ya. I'm interested. |
| 01:33.05 | jcath | hey, any one would like to recommend me some command line based sip user agent ? |
| 01:40.35 | *** join/#asterisk Scratchrf (n=scratchr@63-226-200-214.tukw.qwest.net) |
| 01:41.39 | *** part/#asterisk Lurr (i=user@adsl-067-034-122-207.sip.mia.bellsouth.net) |
| 01:45.10 | *** join/#asterisk RoyK (n=roy@55.80-202-161.nextgentel.com) |
| 01:50.27 | *** join/#asterisk r0d3nt (i=nobody@wsip-24-234-241-145.lv.lv.cox.net) |
| 01:53.10 | *** join/#asterisk [hC] (n=hardcore@209.153.195.139) |
| 01:53.11 | *** join/#asterisk rking (n=rking@ip68-1-234-152.dl.dl.cox.net) |
| 01:54.22 | tzafrir_laptop | jcath, for testing? sipp (debian package sip-tester) can be of some use. There is also the terminal-based linphonec . There are some iax sample command-line tools from iaxclient |
| 01:56.46 | *** join/#asterisk santiago (n=santiago@208.195.215.160) |
| 01:58.41 | jcath | thanks, tzafrir |
| 01:59.07 | Scratchrf | anyone here have a jabber im server using the asterisk-im plugin? |
| 01:59.22 | jcath | for testing and using. yesterday I download minisip and related library, compiled and installed |
| 01:59.49 | jcath | every time after I quit minisip, it gives me a segmentation fault |
| 01:59.54 | jcath | :( |
| 02:00.45 | rking | so, i have a recently-installed zaptel FXO, and i'd like to configure * as my answering machine. i'm running asterisk -vvvv -c, tail -f /var/log/{messages,syslog}, but did not see anything when i dialed my own number (although the actual phone did ring, which is routed through the FXO). where should i look to make this puppy go? |
| 02:06.43 | tzafrir_laptop | check /var/log/asterisk/* . However, is zaptel set up properly? |
| 02:07.02 | tzafrir_laptop | e.g: what's the outputof 'ztcfg -vv'? |
| 02:07.21 | trixter | is asterisk logging to /vra/log/asterisk/sopmething ? |
| 02:07.22 | rking | tzafrir_laptop: ... 0 channels configured |
| 02:07.26 | docelm0 | tzafrir_laptop alot of stuff |
| 02:07.27 | docelm0 | :) |
| 02:08.51 | tzafrir_laptop | Scratchrf, what's the asterisk-im plugin? |
| 02:09.15 | rking | tzafrir_laptop: so first i should tinker with /etc/zaptel.conf until ztconfig -vv shows a channel, then /etc/asterisk/zapata.conf until i can see a ping in the asterisk -vvvv -c ? |
| 02:09.59 | tzafrir_laptop | if you have a channel configured in zapata.conf and not actually on the system, Asterisk will generally fail to start |
| 02:10.00 | Scratchrf | it's a plugin for jive messenger server that shows asterisk presence info to the im server.. |
| 02:10.15 | rking | k |
| 02:10.34 | tzafrir_laptop | did you build the zaptel modules for your kernel? |
| 02:11.11 | tzafrir_laptop | cat /proc/zaptel/* |
| 02:12.51 | rking | tzafrir_laptop: Span 1: WCFXO/0 "Generic Clone Board 1" RED; 1 WCFXO/0/0 |
| 02:15.14 | tzafrir_laptop | so what's the error message from ztcfg ? |
| 02:16.02 | Dr-Linux | whats the recommended/latest firware for cisco 7960 ip phone ? |
| 02:16.32 | rking | tzafrir_laptop: there is no error message - it just says "Zaptel Configuration ; ===== ; Channel Map: 0 channels configured" |
| 02:17.06 | rking | tzafrir_laptop: but i'm working on zaptel.conf right now. I'm at: "span=1,1,0," and trying to learn enough to decide the framing and coding |
| 02:18.22 | *** part/#asterisk joat (n=joat@ip70-160-150-20.hr.hr.cox.net) |
| 02:19.55 | tzafrir_laptop | you don't need no span. Just 'fxsks=1' |
| 02:20.31 | rking | oh, cool - i was just looking at the fx.ks's in http://www.voip-info.org/wiki/index.php?page=Asterisk+config+zaptel.conf |
| 02:20.37 | tzafrir_laptop | and loadzone and defaulttonezone, of course |
| 02:21.31 | rking | right - those are already set. now that i've changed this file, do i just do an rmmod wcfxo; modprobe wcfxo? |
| 02:22.05 | *** join/#asterisk [Airwolf] (n=airwolf@attilla.nl) |
| 02:22.27 | rking | ztcfg -vvv == Channel 01: FXS Kewlstart (Default) (Slaves: 01) |
| 02:23.18 | *** part/#asterisk Scratchrf (n=scratchr@63-226-200-214.tukw.qwest.net) |
| 02:23.42 | rking | my zapata.conf says signalling=fxo_ls, that should be fxo_ks to match the zaptel.conf, right? |
| 02:23.54 | rking | tzafrir_laptop++ for helping me, btw - i'm sorry i'm such a cluebie |
| 02:24.19 | rking | fxs_ks, i mean |
| 02:24.32 | docelm0 | Hay Damin you alive? |
| 02:24.44 | _Sam-- | hi i had been using the SIPPs softclient for a while, but when I upgraded to the newest version i get this on the console (and no calls): Nov 22 21:14:15 WARNING[32568]: chan_sip.c:2704 process_sdp: Unknown SDP media type in offer video 1551 RTP/AVP 34 |
| 02:25.46 | _Sam-- | downgraded to their older client, and its fine....in the new client there is no video related settings that i can disable |
| 02:26.35 | _Sam-- | my asterisk is Asterisk CVS-v1-0-11/16/05-10:39:37 |
| 02:28.05 | *** join/#asterisk test34 (i=1000@unaffiliated/test34) |
| 02:29.13 | bbz | Would anyone here be interested in some Polycom 501 phones? We got stuck w/ an extra 20 after a project, and we need to unload them at cost, -- if anyone is interested send me a private message (if i dont respond immediately, leave me your email address, and i can send you info) i have a reference in the channel if you need. -- |
| 02:29.25 | *** join/#asterisk zamsler (n=zamsler@c-67-184-232-149.hsd1.il.comcast.net) |
| 02:30.14 | *** join/#asterisk beernuts (n=mattfox7@CPE-60-228-213-135.qld.bigpond.net.au) |
| 02:31.08 | beernuts | anyone have sample conf file for australian telstra e1 ?? |
| 02:31.38 | tzafrir_laptop | rking, it should be fxs_ks . |
| 02:31.46 | tzafrir_laptop | right |
| 02:32.10 | rking | tzafrir_laptop: here's the non-comment contents of my zapata.conf: http://rafb.net/paste/results/R3xVtF22.txt |
| 02:32.32 | rking | but i'm still not seeing anything on the asterisk -c -vvvv when i dial in |
| 02:33.24 | rking | beernuts: is this bad news?: http://lists.digium.com/pipermail/asterisk-users/2005-January/082641.html |
| 02:33.52 | tzafrir_laptop | what do you see in 'zap show channels' ? |
| 02:34.32 | rking | tzafrir_laptop: ahh, interesting: Chan = pseudo Context = default |
| 02:34.42 | rking | tzafrir_laptop: at one point i had ztdummy going on |
| 02:35.07 | tzafrir_laptop | rking, that's a different card: T1/E1PRI card. However I recall that there were some problems with using the X100P in Australia. Anybody? |
| 02:36.17 | tzafrir_laptop | rking, if the card does not show there (as channel '1'), something went wrong. Maybe you left out 'channel => 1' in zapata.conf? |
| 02:36.24 | *** join/#asterisk milkyflava (n=milkyfla@20-156-237-24-mvl.ewc.gci.net) |
| 02:36.32 | milkyflava | hello |
| 02:37.28 | rking | tzafrir_laptop: hrmm... that link i posted has the config in it - it shows channel => 1 at the bottom... however, the "reload" output only mentions zap in one place: -- Setting global variable 'TRUNK' to 'Zap/g2' |
| 02:37.31 | milkyflava | I have a TDM with four FXO modules, I have 4 phone lines going into the card. The first module I can hear pops and clicks on but the other three are fine. Is there a fix for this? |
| 02:37.38 | *** join/#asterisk Sedorox (i=brandon@smartserv/cna/Sedorox) |
| 02:38.05 | *** join/#asterisk Koshatul (n=evangeli@ip157-65-132.cust.bit.net.au) |
| 02:38.09 | *** join/#asterisk Chuji (i=Chuji@pcp09930052pcs.tulipgrove.tn.nash.comcast.net) |
| 02:38.30 | milkyflava | TDM400P |
| 02:39.40 | milkyflava | Or is it a known issue? I have tried three different cards and all the same problem on the first FXO module. I have even moved the modules to different ports and the problem remains |
| 02:40.18 | *** join/#asterisk firestrm (n=no@S01060013105c1c69.gv.shawcable.net) |
| 02:40.31 | firestrm | hello |
| 02:41.58 | firestrm | anyone here know why v1.20 doesnt respect ring cadence patterns? I just upgraded from V1stable, and now my previuosly working ring patterns dont work |
| 02:42.28 | milkyflava | I have tried them on two diiferent servers with different phone lines in different buildings and the song remains the same |
| 02:43.47 | *** join/#asterisk diverge (n=user@cpe-65-25-16-186.neo.res.rr.com) |
| 02:44.36 | tainted_ | is 50ms latency good enough to run voip off of? |
| 02:44.57 | *** join/#asterisk implicit (n=implicit@ip70-181-114-97.oc.oc.cox.net) |
| 02:45.00 | rking | tainted_: i'd think so |
| 02:45.15 | rking | tainted_: latency isn't /that/ important with voip - bandwidth helps |
| 02:45.34 | tainted_ | 100Mbps fiber |
| 02:45.37 | Druken | anything under 100 is good i'd say |
| 02:45.46 | Sedorox | I run voip over 100ms.... but its also g729 |
| 02:45.50 | Sedorox | works ok... |
| 02:46.44 | tainted_ | hmm |
| 02:46.49 | tainted_ | i mean for production work environment |
| 02:47.23 | *** part/#asterisk ctooley (n=ctooley@rrcs-24-227-212-162.sw.biz.rr.com) |
| 02:48.07 | firestrm | anyone know how to set ring cadence in v 1.2? |
| 02:48.54 | firestrm | the SetVar(_ALERT_INFO=Bellcore-r2) dont work anymore |
| 02:48.55 | Dr-Linux | anybody can help for cisco 7960 IP phone? |
| 02:50.56 | *** join/#asterisk newsmafia (n=newsmafi@wsip-68-15-19-142.sd.sd.cox.net) |
| 02:51.30 | tzafrir_laptop | milkyflava, shared interrupts? |
| 02:51.49 | tzafrir_laptop | grep wctdm /proc/interrupts |
| 02:53.02 | firestrm | i guess no-one knows.. i'l have to pay digium AGAIN!!! to tell me what they screwed up and how to fix it.. |
| 02:53.39 | Druken | i must say... the only money digium got outta me was for a TDM400p... |
| 02:53.45 | Druken | and it's a pos... |
| 02:54.07 | firestrm | Druken, you got off easy.. |
| 02:54.11 | tzafrir_laptop | firestrm, does it show well in 'zap show cadence' ? |
| 02:54.58 | tzafrir_laptop | Are you sure you fully restarted asterisk after your last change to zapata.conf? with chan_zap a reload does nothing. |
| 02:55.03 | firestrm | yes they are there, it just wont use em |
| 02:55.25 | tzafrir_laptop | set verbose 3 |
| 02:55.46 | firestrm | tzafrir_laptop, it worked fine with v1.0... but broke when i went to 1.2 |
| 02:55.49 | tzafrir_laptop | how exactly do you see it is not used? |
| 02:56.20 | firestrm | tzafrir_laptop, it is not ringing the proper cadences when the extension is being rung |
| 02:56.32 | firestrm | even though i set them |
| 02:57.23 | firestrm | Executing Set("Zap/1-1", "ALERT_INFO=Bellcore-r3") in new stack |
| 02:58.34 | *** join/#asterisk bleck (i=kris@dsl-202-72-161-61.wa.westnet.com.au) |
| 02:58.40 | bleck | I have an asterisk machine, and a callmanager express router, how do I get a phone registered with asterisk to dial out through the e1 card on the callmanager router? (is this possible) |
| 02:59.01 | firestrm | tzafrir_laptop, i changed the setvar to set and the _ALERT_INFO to ALERT_INFO from the 1.0 version |
| 02:59.40 | bleck | i presume this is asterisk as in the pbx software? |
| 03:00.09 | firestrm | bleck, welcome to the seventh level of software hell.. |
| 03:01.13 | *** join/#asterisk mmlj4 (n=jkelly@ip70-171-92-106.no.no.cox.net) |
| 03:02.29 | tzafrir_laptop | firestrm, sorry, can't help you further here |
| 03:02.45 | rking | tzafrir_laptop: hey, sweet! i followed your advice to firestrm, about restarting * completely, and it finally had some output on the CLI when i dialed in. now, just to make it actually go to voice mail. =) |
| 03:03.39 | firestrm | rking, at least it worked for you :) im glad... my problems are never simple.. |
| 03:06.16 | Druken | firestrm: any problem isn't simple... |
| 03:06.53 | *** join/#asterisk zobia (n=laura_sh@218.6.242.212) |
| 03:07.08 | *** join/#asterisk lilo (i=levin@freenode/staff/pdpc.levin) |
| 03:07.10 | zobia | hello everyone |
| 03:07.13 | zobia | good morning |
| 03:07.40 | zobia | is there any way for asterisk to detect answer machine in the dialplan? |
| 03:08.06 | Druken | for ourgoing calls? |
| 03:08.34 | zobia | yes. for outbound call |
| 03:08.53 | Druken | it is possible... but *I* don't know how... |
| 03:08.57 | shido6 | use pastebin.ca |
| 03:09.48 | zobia | thanks Druken, at least i know it's possible |
| 03:10.03 | zobia | anyone knows how to detect answer machine in the asterisk? |
| 03:10.21 | firestrm | what does it cost nowadays for a digium support ticket ? |
| 03:12.35 | ptiggerdine | jtodd, just read some of your fine examples and the tutorials on oreily's. Thanks for providing such simple ideas and explaining them. further more to provide working configuration file has made the "J" curve of learning asterisk 10 times easier. thanks once again. |
| 03:12.47 | trixter | is it just my configuration or has asterisk-users and asterisk-biz gotten really quiet this afternoon? |
| 03:14.02 | trixter | like for the last 5 or so hours ... |
| 03:14.03 | bleck | hmm nm, just figured it out. |
| 03:17.55 | firestrm | F(*#&$ F)#)($* F)#$(*$#(, i cant even BUY tech support.. im screwed:( |
| 03:18.18 | trixter | for what? |
| 03:18.41 | firestrm | version 1.2 asterisk, broken cadence bug |
| 03:19.22 | ptiggerdine | isn't 1.2 stable? |
| 03:19.23 | trixter | ahh well you could buy support from digium, but odds are you dont wanna pay for business edition :P |
| 03:19.36 | trixter | as for the cadence bug explian hat problem you are specifically having |
| 03:19.37 | *** join/#asterisk NeonLevel (n=NeonLeve@201.145.221.42) |
| 03:19.47 | trixter | I missed it aparently |
| 03:19.49 | NeonLevel | good evening, iīm using asterisk 1.2.0 and my ip phone provider gave me account info to make calls, he uses username, password and authorization user in order to make the call, and i canīt seem to make calls because i get Forbidden - wrong password on authentication for INVITE but the passowrd is ok, the problem is authorization name but where do i put this??? anyone please!!!!!!!!! |
| 03:19.50 | TheCops | wow, 1.2 rocks |
| 03:20.12 | firestrm | trixter, at this point, i dont see much choice, i will have to bite the bullet, pay for support TOMORROW! and get royally screwed |
| 03:20.50 | firestrm | TheCops, i dont have anything positive to way about 1.2 right now.. |
| 03:20.52 | *** join/#asterisk [TK]D-Fender (n=joe@66.11.164.239) |
| 03:21.08 | file[laptop] | NeonLevel: put it in the peer entry in sip.conf, username=<username> and fromuser=<username> |
| 03:21.10 | firestrm | 1.2 blows afaic |
| 03:21.20 | *** join/#asterisk juanjoc (n=jcomella@40-115-235-201.fibertel.com.ar) |
| 03:21.39 | NeonLevel | thanks file:laptop trying it out.... |
| 03:21.48 | *** join/#asterisk Flauto (n=zhao@c-24-14-197-214.hsd1.il.comcast.net) |
| 03:22.27 | TheCops | firestrm, all my system is working as before with 1.0.9, and I have intercom feature new and some other tricks |
| 03:23.07 | trixter | firestrm: could you explain the problem you are having? I missed it and erased my scrollback buffer before I saw you say that you couldnt even pay for support |
| 03:23.07 | firestrm | TheCops, well dont try to get ring cadence working.. its totally screwed up |
| 03:23.08 | [TK]D-Fender | y0 |
| 03:23.48 | firestrm | TheCops, they have royaly botched it in 1.2 |
| 03:24.02 | TheCops | file[laptop], ring cadence, I'm not using this feature |
| 03:24.02 | rking | what is format_jpeg.so for? |
| 03:24.13 | TheCops | I hope that PRI feature and stuff like that are clean |
| 03:24.20 | *** join/#asterisk znoG (n=gs@OL101-122.fibertel.com.ar) |
| 03:24.43 | firestrm | file[laptop], i think he ment that for me :) |
| 03:25.15 | milkyflava | tzafrir_laptop: Sorry, I was reading on the voipinfo site about the problem. I will check the interrupts when I get there tonight. Thanks for the advice, it is the same thing I am reading on the voipinfo site. |
| 03:25.25 | firestrm | file[laptop], by the way.. long time no chat.. seems im only here when im having a crisis :P |
| 03:25.41 | Qwell | today, I experienced explaining telephony to a chimp |
| 03:25.54 | firestrm | Qwell, managers can be fun |
| 03:26.00 | Qwell | firestrm: worse |
| 03:26.14 | Qwell | Whats dumber then a manager, but usually makes more? |
| 03:26.14 | firestrm | lol.. what can be worse than that:) |
| 03:26.22 | Qwell | I know some of you know... |
| 03:26.25 | trixter | his wife? |
| 03:26.32 | Qwell | tech writer |
| 03:26.36 | file[laptop] | haha |
| 03:26.38 | firestrm | rotfl |
| 03:26.42 | file[laptop] | Qwell: you're mean |
| 03:26.56 | Qwell | nah, ours is dense, heh |
| 03:27.05 | Qwell | "What does Join Empty mean?" |
| 03:27.17 | trixter | I dunno at one place we hired 3 tech writers at about $90/hr but because they were indentured servants from india (although working in america) most of what they were paid went to pay off their school and moving debt |
| 03:27.22 | trixter | so they actually got very little personally |
| 03:27.25 | trixter | :/ |
| 03:27.26 | Qwell | "It means that if somebody tries to join a queue with no agents logged in, it'll failover to something else." |
| 03:27.36 | *** join/#asterisk ryguillian (n=ryguilli@c-67-176-193-252.hsd1.il.comcast.net) |
| 03:27.40 | Qwell | "What's a queue?" |
| 03:27.51 | Qwell | "?! We've been talking about queues for the last hour now..." |
| 03:28.02 | Qwell | "oh, okay. and an agent?" |
| 03:28.13 | trixter | I stood in a queue at the mcdonalds in scotland.. |
| 03:28.16 | file[laptop] | I'd hit my head against the wall and look for a weapon |
| 03:28.28 | trixter | it seemed to go faster than the queues in america |
| 03:28.56 | Qwell | all the while, I was imagining how much easier this might be, if I were explaining it to an Amish person |
| 03:29.03 | milkyflava | the fries are too salty and the burgers go through you faster than the soda |
| 03:29.08 | milkyflava | im lovin it |
| 03:29.17 | file[laptop] | I'm mildly amused |
| 03:29.17 | firestrm | i think im going to go back to 1.0.10.. at least until they get the bugs out of 1.2 |
| 03:29.25 | Qwell | there is a 1.0.10? |
| 03:29.32 | firestrm | yes |
| 03:29.34 | file[laptop] | there's times though where you just crave a McDonalds meal... |
| 03:29.45 | Qwell | firestrm: since when? |
| 03:29.56 | trixter | milkyflava: I do agree the fries were REALLY salty there.. and the drinks are too small |
| 03:29.59 | file[laptop] | actually, that's a lie - I had it in March at Spring VON |
| 03:30.01 | Qwell | oh, last night |
| 03:30.03 | firestrm | its on the digium ftp site.. |
| 03:30.14 | trixter | the pub down the street was way better |
| 03:30.23 | Qwell | firestrm: What bugs in 1.2? |
| 03:30.29 | Damin | Beer is the best. Always. Hands down. |
| 03:30.32 | Qwell | there are likely far more in 1.0 than 1.2 |
| 03:30.42 | Damin | firestrm: 1.2 is perfect. No bugs whatsoever. |
| 03:30.42 | file[laptop] | you're the one that I ever needed, show me love and what it's all about |
| 03:30.44 | firestrm | Qwell, ring cadence is broken |
| 03:30.49 | milkyflava | trixter: VERY! |
| 03:30.49 | file[laptop] | Damin: you're insane |
| 03:30.51 | *** join/#asterisk pigpen2 (n=mark@fw.seamans.cc) |
| 03:31.02 | file[laptop] | btw this is now #asterisk after hours, talk at your own peril :P |
| 03:31.09 | Qwell | Damin: beer > magic flask? |
| 03:31.14 | file[laptop] | because I might make fun of you. |
| 03:31.24 | trixter | milkyflava: they also seemed to over cook the fries every time |
| 03:31.33 | Damin | Qwell: Thanks for reminding me. I'm going to stay w/ the Inlaws for a week, so I'll need to bring the magic flask along.. |
| 03:31.35 | zobia | what's the standard format of sound file that asterisk support? |
| 03:31.38 | Qwell | haha |
| 03:31.38 | trixter | the pub did good burgers and fries though.. quite tasty |
| 03:31.40 | file[laptop] | how can you over cook McDonalds fries? |
| 03:31.45 | file[laptop] | I thought they just set a timer |
| 03:31.50 | Qwell | Damin: Any time. ;] |
| 03:31.51 | Damin | zobia: Yes. It does support standard audio formats. |
| 03:31.53 | milkyflava | they never decay those fries |
| 03:31.54 | trixter | they should just set a timer but they all came out dark brown |
| 03:32.04 | trixter | that was a biased experiment |
| 03:32.08 | Qwell | zobia: any, all...whatever |
| 03:32.10 | trixter | closed jar and all |
| 03:32.17 | firestrm | is there a bug report list/site/procedure for asterisk? |
| 03:32.17 | blitzrage | Qwell: I heard that! |
| 03:32.18 | milkyflava | wasnt it closed on all of them? |
| 03:32.22 | Qwell | blitzrage: did not |
| 03:32.32 | file[laptop] | firestrm: http://bugs.digium.com/ |
| 03:32.33 | trixter | most of that was half science and misleading statements |
| 03:32.42 | Damin | firestrm: Yes. bugs.digium.com. Read the bug submission guidelines linked from the front page. |
| 03:32.42 | firestrm | file[laptop], thanks.. |
| 03:33.01 | blitzrage | Qwell: I definately don't make more than a manager :) |
| 03:33.03 | Qwell | blitzrage: I don't consider you a tech writer though, heh. I'd call you a tech more than anything. :p |
| 03:33.04 | Damin | firestrm: http://www.digium.com/bugguidelines.html |
| 03:33.06 | *** join/#asterisk _DAW (n=bob@adsl-222-51-184.msy.bellsouth.net) |
| 03:33.17 | blitzrage | Qwell: ouch :) |
| 03:33.18 | file[laptop] | poor blitzrage makes enough to get by |
| 03:33.20 | _DAW | hello |
| 03:33.23 | milkyflava | still fun to watch |
| 03:33.26 | Qwell | ouch? |
| 03:33.33 | blitzrage | Qwell: just a tech eh? |
| 03:33.33 | zobia | Qwell, i mean it must be 8 kHz mono 16-bit? |
| 03:33.37 | Qwell | just? |
| 03:33.44 | Qwell | blitzrage: You'd rather be a tech writer? |
| 03:33.45 | file[laptop] | blitzrage is also a teacher |
| 03:33.46 | Damin | blitzrage: Hehehe.. I found all the veiled references to "Someone running a P-133" in the Asterisk book.. :) |
| 03:33.54 | Qwell | well, there you go then |
| 03:34.00 | Damin | blitzrage: I got news for you.. I had lunch w/ a guy who is running it on a P-90. |
| 03:34.10 | blitzrage | Damin: glad you found it :) |
| 03:34.14 | blitzrage | Damin: wow -- a P90 eh?> |
| 03:34.20 | Damin | blitzrage: Over Thanksgiving break, I'm going to get it running on my 486 laptop! ;) |
| 03:34.23 | blitzrage | Damin: seems like a waste of power to me... :) |
| 03:34.28 | blitzrage | Damin: that'd be wicked :) |
| 03:34.36 | file[laptop] | is this going to be one of those "oh yeah? I run my asterisk on a calculator!" things? |
| 03:34.38 | Qwell | I have an old 386 amd... |
| 03:34.46 | Qwell | had...damn |
| 03:34.49 | Damin | blitzrage: We'll see. :) |
| 03:34.55 | milkyflava | I still have my Vic20 |
| 03:35.02 | blitzrage | Qwell: well... I suppose not. I'm just sentitive from my days as tech support -- and I just associate "tech" with "tech support" |
| 03:35.08 | Qwell | oh god no |
| 03:35.19 | *** join/#asterisk Kyreeth (n=ashley@aquila.feathers.net) |
| 03:35.24 | Damin | milkyflava: I've got an Apple II+ sitting in the back room. ;) |
| 03:35.31 | file[laptop] | I'm a tech, but I don't do tech support really |
| 03:35.31 | blitzrage | I want to install Asterisk on my Zaurus |
| 03:35.43 | firestrm | Damin, i did manage to compile it for sparc once, though not terrably useful |
| 03:35.47 | _DAW | can someone tell me if alert_info will work with the page command? |
| 03:35.48 | blitzrage | cross compile it :) |
| 03:35.55 | milkyflava | Damin: If I had known that someone that had one of those I would have given you all my disks |
| 03:36.01 | milkyflava | copyII plus |
| 03:36.10 | *** join/#asterisk SplasPood (i=nobody@paravolve.net) |
| 03:36.18 | file[laptop] | all this talk of ancient machines is making me feel young :P |
| 03:36.19 | Damin | milkyflava: I've got all that crap. I have almost 1/2 gig of Apple stuff. :) |
| 03:36.32 | Damin | milkyflava: Dalton Disk Disintergrator images.. :) |
| 03:36.41 | milkyflava | Damin: lol, a half gig?!?!? |
| 03:36.42 | *** join/#asterisk SplasPood (i=nobody@paravolve.net) |
| 03:36.54 | milkyflava | Damin: you must have it all |
| 03:37.06 | Damin | milkyflava: Yeah.. It's the combined collection of several Apple BBS's from the 80s... |
| 03:37.21 | milkyflava | Damin: that explains it |
| 03:37.31 | Qwell | must've been a massive raid storing all that |
| 03:37.45 | file[laptop] | freestyler! |
| 03:37.53 | Damin | Qwell: No.. it was lots of individual 20 and 30 megabyte hard drives! ;) |
| 03:38.00 | milkyflava | a tape deck and tons of casettes |
| 03:38.04 | dos000 | anyone know why a sip endpoint (ata) would not show up when you do sip show peers ? i am really puzled here |
| 03:38.10 | trixter | gah.. only about 40,000 webpages to go to get all the numbering plan info off this one site.. stupid they chage for the database but let you do http to get at it for free |
| 03:38.17 | file[laptop] | dos000: if it's not registered, it won't show up... |
| 03:38.22 | firestrm | ok.. guidelines read.. its still a bug.. trying to decide between major (because it really screws me up) or minor (because it seems no-one but me is using this feature) |
| 03:38.24 | trixter | just 5 entries per page or something.. so it takes forever to avoid DoSing their server |
| 03:38.24 | Qwell | trixter: script it |
| 03:38.30 | trixter | it is scripted |
| 03:38.35 | trixter | you think I am doing this by hand? |
| 03:38.38 | dos000 | file[laptop], its supposed to be registered .. its making calls ! |
| 03:38.38 | Qwell | wget with a random wait? |
| 03:38.39 | trixter | way too much work |
| 03:38.44 | Damin | firestrm: Put it in as Minor, and then ask for people to corroborate the issue.. |
| 03:38.51 | file[laptop] | dos000: you don't have to be registered to make call, only to receive them |
| 03:39.01 | trixter | not random I only care to avoid DoSing their server, so it sleeps 1 second between hits, that way my load is very very low.. |
| 03:39.08 | firestrm | Damin, ya i think i agree.. |
| 03:39.14 | trixter | cause I cant predict the load that server will have otherwise |
| 03:39.39 | Damin | trixter: What you should do is fire off 1,000 concurrent threads to retrieve everything at once. |
| 03:39.41 | dos000 | file[laptop], how does it know where the server is ? |
| 03:39.44 | trixter | its to add to http://www.0xdecafbad.com/Global-Numbering-Plan.html ... I have about 5500 entries but they have about 250,000 so I wanted to bulk up :D |
| 03:40.00 | trixter | yeah let me DoS their server that would be better |
| 03:40.03 | Damin | dos000: Cause you programmed it to send all it's shit to a specific IP / Proxy / Gateway. |
| 03:40.22 | Qwell | trixter: by the time they figure it out and fix it, you'll be done. :p |
| 03:40.28 | milkyflava | lol |
| 03:40.40 | dos000 | Damin, there is really no way to stop them from making calls if they are not registered then ? |
| 03:41.01 | Damin | dos000: You could DOS them. |
| 03:41.09 | dos000 | darn ... ! |
| 03:41.15 | file[laptop] | Damin: that's mean, but creative |
| 03:41.21 | Damin | dos000: Of force them to enter a passcode to make a call. |
| 03:41.28 | Damin | dos000: Or firewall their IP.. |
| 03:41.49 | dos000 | Damin, all end points (which are sip) are configured with a passwd ! |
| 03:41.51 | Damin | dos000: Or send all their calls, no matter what they dial to 1-800-ANAL-BOY (That's always a fun one..) |
| 03:42.01 | milkyflava | lol |
| 03:42.14 | milkyflava | Now that sounds like fun |
| 03:42.24 | milkyflava | Im gonna make all calls go there |
| 03:42.25 | dos000 | Damin, i just dont know how to differentiate registered and non registered here |
| 03:42.27 | firestrm | Damin, is that an actual working number? |
| 03:42.28 | milkyflava | then quit |
| 03:42.39 | Damin | firestrm: Call it. |
| 03:42.44 | firestrm | lol |
| 03:42.48 | milkyflava | lol |
| 03:43.10 | dos000 | Damin, is there a macro to call to check if they are registered ? |
| 03:43.13 | Damin | firestrm: I've modified the telemarketer script to re-direct them to that number.. |
| 03:43.27 | milkyflava | genius! |
| 03:43.37 | firestrm | rotflmao.. i can have some SERIOUS fun with that one.. |
| 03:44.11 | dos000 | Damin, it maybe eaxctly what they want ... otoh .. i am wondering how you came to know about it :-) |
| 03:44.28 | dos000 | about the # of course |
| 03:44.51 | file[laptop] | Damin has nothing better to do with his time then dial every toll-free number in existence and pester people |
| 03:45.06 | *** join/#asterisk FuriousGeorge (n=brian@ool-44c5a9b8.dyn.optonline.net) |
| 03:45.15 | Damin | dos000: The number? I was pranked with it once.. kept it and we use it all the time. When I'm bored, I'll drop a call file in that bridges that to some of the support techs for fun.. |
| 03:45.21 | NeonLevel | hey file:laptop thank you very much for your help, worked great....!!!!!!!!!!!!! |
| 03:45.30 | dos000 | lol |
| 03:45.39 | file[laptop] | NeonLevel: yes, I rock |
| 03:45.47 | file[laptop] | can't touch this! |
| 03:45.54 | NeonLevel | yeah man you rock!!!!!!!!!!!!!!!!!!!!!!11 |
| 03:45.56 | milkyflava | he's very modest to |
| 03:45.58 | Damin | file: Actually, one night (while drinking, of course) we sat around the speaker phone and just dialed random phrases to see what we would get.. |
| 03:46.20 | file[laptop] | Damin: you and drinking, what a great combo |
| 03:46.31 | file[laptop] | Qwell: meep don't touch that! |
| 03:46.53 | Damin | 1-800-CUNT-LAP is another good one.. |
| 03:46.54 | milkyflava | Damin: just say it wasn't you that came up with that random phrase |
| 03:47.00 | blitzrage | $this->touches; |
| 03:47.21 | file[laptop] | blitzrage: ooh baby touch this harder :P |
| 03:47.31 | Qwell | oh, I see how it is |
| 03:47.41 | Damin | Alright.. so.. I bought a Tyan barebones box w/ an LCD on the front.. |
| 03:47.50 | Qwell | Damin: queue stats |
| 03:47.51 | blitzrage | file[laptop]: you always have to take it one step too far eh? :) |
| 03:47.59 | bweschke | wow. we've gone TV-MA again, huh? |
| 03:48.00 | Qwell | stock ticker |
| 03:48.22 | Damin | They distribute a pre-compiled application that lets you configure the server's IP / GW etc from the front, and do some simple things like shut it down.. |
| 03:48.24 | file[laptop] | blitzrage: yes. |
| 03:48.36 | Damin | But, they mispelled stuff, in classic Chingrish ways.. |
| 03:48.42 | bweschke | lol |
| 03:48.49 | dos000 | Damin, seriously tho .. is there a macro to call to differentiate them. I kind of (maybe) missed what you meant by DOSing them ... Denial Of Service ??? |
| 03:48.50 | Damin | When you shut the system down, it says "Shitting down server" |
| 03:48.55 | Qwell | awesome |
| 03:48.59 | milkyflava | lol |
| 03:49.04 | milkyflava | thats classic |
| 03:49.07 | bweschke | would be call to hack it to show BHCA while asterisk is running or some other stat like that |
| 03:49.09 | file[laptop] | woot Chingrish!!! |
| 03:49.15 | Damin | And instead of saying "TYAN Server" on the banner, it says "TYAN Sever" |
| 03:49.25 | milkyflava | I had cuntlevers instead of cantlevers delivered once |
| 03:49.28 | file[laptop] | okay, that one made me chuckle |
| 03:49.52 | milkyflava | Damin: where do I get one of those |
| 03:49.53 | blitzrage | you want to know whats dangerous? |
| 03:50.04 | Damin | So I looked on their FTP site to get the source code, and found an archive that works, but doesn't have any of their menuing code in it.. |
| 03:50.11 | blitzrage | having a big bowl of chocolate covered almonds on a table right beside your desk |
| 03:50.21 | file[laptop] | blitzrage: oh those are good, so very very good |
| 03:50.29 | Damin | So I ended up HexEditing the damn binary to correct the spelling errors before I send it out to the client! ;) |
| 03:50.33 | blitzrage | file[laptop]: oh -- don't I know it |
| 03:50.37 | Damin | milkyflava: On the web? |
| 03:50.39 | Qwell | Damin: That's great... |
| 03:50.50 | dos000 | lol |
| 03:50.55 | *** join/#asterisk harryvv (n=none@S010600a0c93f6f7e.vs.shawcable.net) |
| 03:50.57 | file[laptop] | blitzrage: are you ProductiveLeif(tm) today? |
| 03:51.00 | file[laptop] | or rather, tonight |
| 03:51.06 | blitzrage | semi... was moreso yesterday |
| 03:51.18 | milkyflava | Damin: I thought it might be a certain version of their system, so just any of them? |
| 03:51.23 | blitzrage | waiting for someone to build be a server and put it in a rack in Miami |
| 03:51.28 | blitzrage | s/be/me |
| 03:51.39 | Damin | Qwell: Well, turns out that the guys from the Bluw Quartz project did the same thing, but actually wrote a perl based daemon / wrapper script that hex-edits the binary w/ the arguements you pass to it, and the re-starts the application to change the display on the screen.. :) |
| 03:51.52 | *** join/#asterisk TheCops (n=henri@206-248-136-187.dsl.teksavvy.com) |
| 03:51.57 | Qwell | hmm |
| 03:52.13 | TheCops | Someone played with Snom and Subscription support of asterisk 1.2 ? |
| 03:52.24 | bweschke | TheCops: yes |
| 03:52.25 | Damin | Talk about cobbling crap together w/ bubblegum! :) |
| 03:52.33 | harryvv | blitz you were one of the editors for the Asterisk: The Future of Telephony book? |
| 03:52.36 | TheCops | bweschke, What did you do to make it work ? |
| 03:52.39 | Qwell | Damin: But what happens when the perl script has typos?! |
| 03:52.43 | TheCops | bweschke, I'm trying, but nothing to do.. |
| 03:52.56 | bweschke | put the hints above the actual extension definition in extensions.conf |
| 03:53.04 | bweschke | made sure I defined subscribecontext in sip.conf |
| 03:53.19 | *** join/#asterisk benjk (n=benjk@f8a01-0357.din.or.jp) |
| 03:53.20 | Damin | Qwell: It's like 2 lines and it does boundary checking to truncate your arguements to 16 chars or less per screen of the LCD.. |
| 03:53.22 | bweschke | and set the destination on the phone for the button I wanted to light and it pretty much worked |
| 03:53.31 | Qwell | ahh, heh |
| 03:53.42 | bweschke | this was on a snom 360 |
| 03:53.52 | blitzrage | harryvv: co-author actually |
| 03:54.26 | *** part/#asterisk NeonLevel (n=NeonLeve@201.145.221.42) |
| 03:54.26 | *** join/#asterisk asylum (n=asylum@dsl-58-6-126-60.qld.westnet.com.au) |
| 03:54.26 | blitzrage | harryvv: but yah :) |
| 03:54.26 | TheCops | bweschke, subscribecontext, in plain text SIP format ? |
| 03:54.29 | Damin | trixter: Did you know that you can get all the US NANP information for free? |
| 03:54.51 | trixter | yes |
| 03:55.01 | Damin | trixter: Just checking. ;) |
| 03:55.05 | trixter | but that wont tell me which mobile provider has which block in the congo republic |
| 03:55.12 | trixter | :) |
| 03:55.21 | bweschke | subscribecontext=localextensions; Only allow SUBSCRIBE for local extensions |
| 03:55.35 | benjk | trixter: hi, how are you |
| 03:55.44 | dos000 | anyone know if there is a macro to call to differentiate between registered and non-registered sip end points ? |
| 03:55.50 | milkyflava | thanks for the help |
| 03:56.14 | Damin | Why aren't you all working on the bug tracker right now? |
| 03:56.29 | Damin | There are bugs that YOU could be helping to track down and fix! :) |
| 03:56.34 | *** join/#asterisk pigpen2 (n=mark@fw.seamans.cc) |
| 03:56.35 | asylum | Just wondering if someone could point me in the right direction, When callers call the DID it automaticly goes to the IVR, when i user selects 1 it rings an extention, Asterisk is pausing about 7 seconds before ringing any extentions, I have made incomming calls go directly to an extention and it still pausing before dialing the extention |
| 03:56.58 | firestrm | Damin, ive allready got one bug:) i dont need more |
| 03:57.01 | Damin | asylum: Change the timeout in the config files for DTMF detection.. |
| 03:57.19 | file[laptop] | Damin: I'm acting social with friends |
| 03:57.23 | file[laptop] | it's scary :D |
| 03:57.35 | asylum | I'll try that now, thanks Damin ;) |
| 03:58.00 | benjk | trixter, are you there? |
| 03:58.23 | trixter | sorta |
| 03:58.27 | trixter | whats up? |
| 03:58.57 | benjk | I made some changes to your sample dialplan |
| 03:59.12 | benjk | the one with the area codes |
| 03:59.25 | benjk | Switzerland was fairly incomplete |
| 03:59.39 | trixter | ok, please email it to trixter@0xdecafbad.com I am getting more data from other sources and will integrate everything after I have that |
| 03:59.46 | TheCops | bweschke do you state the ringing ? |
| 04:00.00 | trixter | yeah I bet a lot of em were, the initial parts came from mci and astbill, I then filled in some specific stuff here and there |
| 04:00.00 | benjk | I am still working on other countries |
| 04:00.14 | trixter | I am adding about 240,000 routes through this new site so it should be much more complete |
| 04:00.16 | benjk | Japan for example |
| 04:00.40 | trixter | well if you dont wanna spend the time why dont you wait, I will have more data then you can see what is missing :) |
| 04:00.46 | trixter | rather than do the same work that I have going automated |
| 04:00.49 | benjk | Also, I am not sure your dialplan logic is correct |
| 04:00.58 | trixter | it works for me |
| 04:01.09 | trixter | aside from the one typo someone pointed out in the UK |
| 04:01.19 | bweschke | do I "state the ringing" ? |
| 04:01.22 | bweschke | not sure what you mean? |
| 04:01.30 | bweschke | the LED does blink when that device is ringing, yes |
| 04:01.36 | *** join/#asterisk asterboy (n=Snake@S01060204ee2b6007.ed.shawcable.net) |
| 04:01.38 | Damin | Every single person should test this patch.. |
| 04:01.39 | Damin | http://bugs.digium.com/view.php?id=5374 |
| 04:01.53 | benjk | I have seen Asterisk getting confused when you don't explicitly code around mobile or premium rate numbers |
| 04:01.55 | bweschke | oh. the good 'ole silence generator |
| 04:02.32 | TheCops | bweschke can I see your sip.conf and extension context for that tricks ? |
| 04:02.50 | Damin | bweschke: That's part of it, but it's more about asychnronous send/receive.. |
| 04:03.04 | tzafrir_laptop | any idea what is AutoAnswer from bristuff? |
| 04:03.13 | bweschke | yep |
| 04:03.14 | benjk | for example if you have _004179X. (swiss mobiles) |
| 04:03.19 | bweschke | cool stuff. |
| 04:03.35 | bweschke | I don't see any reason why it wouldn't go in at this point |
| 04:03.45 | Damin | firestrm: Is this what you are having problems with? http://bugs.digium.com/view.php?id=5826 |
| 04:04.12 | benjk | you will need to do the surrounding land lines like _00417[0-8]X. |
| 04:04.25 | trixter | benjk: I havent had that problem but meh.. I will accept any changes you have |
| 04:04.37 | trixter | I am not going to do it now though so email is the best way so I have a saved copy :) |
| 04:04.52 | benjk | if you have overlapping codes, it is not predictable which one Asterisk will pick to match |
| 04:05.04 | asylum | Damin - Do I have to edit zaptel source code todo this? |
| 04:05.10 | *** join/#asterisk pigpen2 (n=mark@fw.seamans.cc) |
| 04:05.15 | *** join/#asterisk stkn_ (i=nobody@gentoo/developer/pdpc.active.stkn) |
| 04:05.54 | TheCops | bweschke, I'm doing a sip show subscription and nothing happen when a phone is on call |
| 04:05.56 | Damin | asylum: No.. I can't remember where the hell it is, but something allows you to configure how long to listen for DTMF events before you assume the person is done and move on.. |
| 04:05.56 | TheCops | weird |
| 04:06.05 | firestrm | aaaaarrrrggggghghhh!!.. save me from the wee turtles!! |
| 04:06.12 | trixter | I thought it was supposed to match based on the most strict first and the least strict last |
| 04:06.17 | trixter | thought that was a coded feature |
| 04:06.23 | Damin | asylum: I think it might be an option to the "Background" command, which plays an IVR menu and listens for DTMF.. let me check.. |
| 04:06.50 | asylum | Damin - Thanks, I had a look on Google, but i couldn't find anything applicable. |
| 04:06.55 | benjk | trixter: I thought so too, but I found that it's not predictable |
| 04:06.56 | trixter | so like _123. would match anything that doesnt match _12345. ... that type of thing |
| 04:07.00 | *** join/#asterisk Vco (n=Vco@S01060050da6df072.sc.shawcable.net) |
| 04:07.05 | trixter | hrm maybe the docs should be more clear |
| 04:07.52 | trixter | I will work on that a bit, the reason I want this is for LCR scripts the csv |
| 04:08.01 | benjk | I have also observed slightly different behaviour for different releases of Asterisk |
| 04:08.03 | trixter | the dialplan stuff was just to be nice :P |
| 04:08.13 | Damin | asylum: Here it is.. exten => 211,4,DigitTimeout,2 |
| 04:08.29 | asylum | Ta thanks |
| 04:08.31 | Damin | Put that before your "Background" command and it will wait 2 seconds before timing out digits.. |
| 04:08.33 | bweschke | TheCops: does it show anything at all? |
| 04:08.45 | TheCops | bweschke, yeah, I see all phone in Idle mode |
| 04:08.50 | benjk | yeah, I realise that |
| 04:08.54 | TheCops | but when they change state, nothing happening |
| 04:09.00 | Damin | asylum: You'll want to modify it to make sense for your dial-plan of course, but the DigitTimeout command is what you are looking for.. |
| 04:09.00 | TheCops | I mean, make a call, or on a call |
| 04:09.10 | bweschke | can you pastebin your dial plan? |
| 04:09.31 | asylum | Damin: it's weird tho, When i reset the whole box, There is no problems with it for a few days.. then it starts taking it's time.. |
| 04:09.33 | Damin | asylum: igitTimeout(seconds): Set the maximum amount of time permitted between |
| 04:09.34 | Damin | digits when the user is typing in an extension. When this timeout expires, |
| 04:09.34 | Damin | after the user has started to type in an extension, the extension will be |
| 04:09.34 | Damin | considered complete, and will be interpreted. |
| 04:09.52 | Damin | asylum: Just set it. Test it. If it works, send me money. ;) |
| 04:10.02 | asylum | haha, i'll try it now :) |
| 04:10.04 | Damin | asylum: Or beer.. whichever you fancy. |
| 04:10.23 | asylum | haha brb.. i'll give it a go |
| 04:12.05 | benjk | trixter: on a slightly different but somewhat related note ... did you come across information about special service codes in different countries? Things like enabling/disabling of sending caller ID and other such features offered by local telcos |
| 04:12.10 | firestrm | FU$K. its broken in 1.0.10 too.. arrgh.. kill me now and get it over with.. |
| 04:12.13 | TheCops | bweschke, http://www.groupeneotech.ca/extensions.conf.txt |
| 04:12.35 | trixter | you mean like CLASS services for various countries? |
| 04:12.42 | benjk | yes |
| 04:12.52 | trixter | nope havent looked either |
| 04:13.04 | trixter | and I dont know if anyone else is doing it on a national standard so that may be a problem :/ |
| 04:13.09 | benjk | I have some of that info for a few countries, but seek to complete it |
| 04:13.40 | benjk | have been working on an XML database for this |
| 04:13.45 | bweschke | TheCops: is there a reason why you're doing exten = 100,hint,SIP/100 instead of exten => 100,hint,SIP/100 ? |
| 04:13.48 | benjk | along with country codes |
| 04:14.08 | bweschke | also - move the hints to the top of that context. above everything else in local-exten context |
| 04:14.47 | TheCops | bweschke, on voip-info the exten = is shown like that |
| 04:14.52 | TheCops | for the hint priority |
| 04:15.00 | zobia | hello qwell |
| 04:15.26 | zobia | do u know how to dial some dial with two lines? |
| 04:15.40 | TheCops | bweschke, I'll try to move the hint |
| 04:16.27 | TheCops | bweschke, not working... |
| 04:17.45 | Damin | asylum: Well? :) |
| 04:17.48 | bweschke | ok |
| 04:17.49 | bweschke | so.. |
| 04:18.00 | asylum | Damin: Didn't work :/ |
| 04:18.09 | asylum | Why would this happen tho.. |
| 04:18.12 | bweschke | well - I'd do exten => 100,hint,SIP/100 regardless |
| 04:18.14 | asylum | When i rest the system.. |
| 04:18.16 | asylum | works fine.. |
| 04:18.22 | TheCops | bweschke, ok, I tried both anyway |
| 04:18.25 | asylum | up for a few days it pauses before dialing extentions.. |
| 04:18.27 | bweschke | anyways... probably need to get a SIP debug log as a next step |
| 04:18.33 | asylum | reset even |
| 04:18.49 | Damin | bweschke: You wanna look at http://bugs.digium.com/view.php?id=5749 and put your vote in there? I say we close it. |
| 04:19.19 | benjk | trixter: I have also managed to get test this voipbuster thing, had to ask a friend to run the windoze client to sign up for me, Kinkos wouldn't let me run it, they are scared of virus infection, you can only run what they have installed |
| 04:19.24 | bweschke | ya - let's draw the shutters on that one for sure |
| 04:19.48 | TheCops | bweschke, did you change something on the config of the phone ? |
| 04:19.54 | Damin | bweschke: BTW.. I need to capture DTMF events in a SIP session, and I didn't have much luck w/ it yesterday.. any suggestions? I was trying to use etheral. |
| 04:20.15 | bweschke | DTMF logger channel in logger.conf ? |
| 04:20.45 | Damin | bweschke: OK.. Let me look.. |
| 04:20.48 | TheCops | bweschke, like Publish Presence ? it's off on my side.. |
| 04:21.02 | bweschke | TheCops: that shouldn't matter... |
| 04:21.24 | trixter | benjk: yeah I signed up their interconnection in london sucks ass so latency is really high for me, tried a few different originating routes and they all go through the same place |
| 04:21.30 | trixter | like a 100ms jump there |
| 04:21.30 | bweschke | TheCops: because Asterisk is making a decision on the state of the device based on what it's doing with * at that point, not what the device is telling * that it's doing |
| 04:21.42 | TheCops | ok |
| 04:21.48 | Corydon76-home | Ah, I love people who hit "Reply All" when replying to a list posting |
| 04:22.16 | bweschke | I did have to define the configuration of the button to actually have the phone register for notification |
| 04:22.24 | benjk | trixter: well, can't complain about the quality at that price ;-) |
| 04:22.41 | bweschke | but that's why we're going to want to see the SIP debug now... to see whether it's the phone that is not registering for those notifications, or whether it's *, or whatever.... |
| 04:23.27 | Damin | bweschke: Crap.. I'm running 1.0-stable branch on this box.. I don't think it has the DTMF logger patch.. |
| 04:24.04 | benjk | Cory: that's why mailing lists should not reveal any sender addresses |
| 04:24.21 | trixter | heh |
| 04:24.21 | benjk | which is what we do on the Macintosh Asterisk Mailing List |
| 04:24.23 | bweschke | I've seen some reports with the polycoms where subscriptions work initially and then they fall off after a while, but nothing where it just "dpesn't work" |
| 04:24.26 | trixter | yeah but they only do half of what I do :D |
| 04:24.33 | bweschke | Damin: ugh. |
| 04:24.36 | Damin | beschke: What about cherry picking the rfc2833 events out of the SIP stream? |
| 04:24.44 | bweschke | sure. |
| 04:24.47 | TheCops | bweschke, very weird...I'll do a sip debug |
| 04:24.50 | bweschke | you want a quick patch for that? |
| 04:25.10 | bweschke | what do you have? 1.0.10 on that box ? |
| 04:25.30 | bweschke | 2833 would be out of the rtp stream - or are you using sip info there? |
| 04:26.26 | Damin | bweschke: Well, wherever it is, if I'm capturing all shit from/to the destination sip server, it should be in there somewhere, no? ;) |
| 04:26.34 | benjk | trixter: I don't really have any need to call any of the countries in their free list, other than US and perhaps very ocassionally Belgium and Switzerland |
| 04:26.51 | bweschke | no. not in a sip debug. 2833 is in the rtp stream. sip info, yes, for sure. |
| 04:27.13 | bweschke | but - ethereal out to be catching it too if you're watching the RTP |
| 04:27.20 | bweschke | beyond just the SIP ports |
| 04:27.50 | trixter | yeah well 14 countries vs 30 ... hehe |
| 04:28.18 | *** join/#asterisk tessier (n=treed@wsip-68-15-4-13.sd.sd.cox.net) |
| 04:28.38 | benjk | but for the time being, this will be a nice throw-in for customers to whom I want to sell an Asterisk box ;-) |
| 04:29.05 | diverge | how hard would it be to write a script that only answers if a certain caller id is present, if not it forwards all calls |
| 04:29.31 | bweschke | diverge: that's builtin functionality in the DP today |
| 04:29.33 | trixter | if you want you have to pay to get past the 1 minute limit |
| 04:29.46 | *** join/#asterisk andrew` (i=andrew@69-12-136-56.dsl.static.sonic.net) |
| 04:30.06 | benjk | trixter: 10 EUR investment per customer is a bargain |
| 04:30.26 | benjk | and I use paypal, so they don't get my credit card details |
| 04:30.49 | Damin | bweschke: patch -p0 -F3 < ../dtmf-logger.patch |
| 04:30.52 | firestrm | yaaaa. i fixed it.. gotta love undocumented changes |
| 04:31.23 | firestrm | i will have to update the wiki.. |
| 04:31.30 | Damin | bwesh: It applies to |
| 04:31.37 | Damin | 1.0 stable.. |
| 04:31.44 | Damin | Let's see if it compiles! ;) |
| 04:31.45 | diverge | bweschke: im brand new to asterisk, im writing a voip proxy |
| 04:32.11 | bweschke | does it? cool! |
| 04:32.42 | bweschke | that'd be a good way to do it... at least then you'll get all your dtmf events happening.. not just 2833 |
| 04:32.50 | firestrm | exit |
| 04:33.01 | firestrm | oops.. wrong window:) |
| 04:33.22 | Damin | best: Well.. it applies.. doesn't compile.. |
| 04:33.24 | Damin | logger.c: In function `make_logchannel': |
| 04:33.24 | Damin | logger.c:142: syntax error before "else" |
| 04:33.24 | Damin | logger.c: In function `strip_coloring': |
| 04:33.24 | Damin | logger.c:499: `chan' undeclared (first use in this function) |
| 04:33.30 | bweschke | hmm |
| 04:33.31 | bweschke | well |
| 04:33.39 | bweschke | a quick and dirty would be to go into rtp.c |
| 04:34.04 | bweschke | and go down to process_rfc2833(...) |
| 04:34.08 | bweschke | and then just drop in ... |
| 04:34.23 | *** join/#asterisk Qwell (n=north@24-205-180-81.dhcp.wsco.ca.charter.com) |
| 04:34.29 | bweschke | <PROTECTED> |
| 04:34.52 | bweschke | problem is there's no real good way to ... ummm... turn it off once you've done that. :) |
| 04:35.26 | Damin | #if 0 |
| 04:35.26 | Damin | <PROTECTED> |
| 04:35.26 | Damin | #endif |
| 04:35.30 | bweschke | ya |
| 04:35.35 | bweschke | that's the poor man's logger |
| 04:35.36 | bweschke | :) |
| 04:35.42 | bweschke | console logger baby! |
| 04:35.43 | Qwell | Corydon76-home: You around? |
| 04:35.48 | file[laptop] | pesky buggery! |
| 04:35.51 | Damin | How does one define "0" ? ;) |
| 04:35.54 | Corydon76-home | No? |
| 04:35.55 | file[laptop] | cause I need to go, need to get away! |
| 04:38.03 | tzafrir_laptop | #define a 0 ? |
| 04:38.34 | Corydon76-home | Damin: it can be done in Perl... |
| 04:38.42 | Corydon76-home | 0 is defined, but false |
| 04:38.47 | bweschke | you're better off with the ast_log call... at least then you'll get it captured to a logger that's capturing the debug channel |
| 04:39.00 | Corydon76-home | Ditto for the empty string |
| 04:40.11 | bweschke | oh man... here we go... www.linksysone.com |
| 04:40.15 | bweschke | what the hell is that? |
| 04:40.20 | tzafrir_laptop | gee, those automated post-packaging tests in Debian: |
| 04:40.41 | benjk | bweschke, very easy to turn selected logs on and off ... |
| 04:40.45 | tzafrir_laptop | W: asterisk-doc: package-contains-readme-for-other-platform-or-distro usr/share/doc/asterisk-doc/README.cygwin |
| 04:41.23 | benjk | if (flag_file_exists(flag_file_path)) ast_log(... |
| 04:41.53 | benjk | then use touch <flagfilename> to turn it on |
| 04:41.59 | benjk | and rm to turn it off |
| 04:42.24 | bweschke | not in the 1.0 base it's not |
| 04:42.53 | Qwell | Corydon76-home: re: curl and env...you're right. What do you think about the other stuff I said though, that cannot be set with an env var? |
| 04:43.09 | *** join/#asterisk ManxPower (n=ewieling@68-185-236-44.dhcp.slid.la.charter.com) |
| 04:43.42 | Corydon76-home | What other stuff? |
| 04:44.03 | Qwell | like CURLOPT_PROXYHOST or CURLOPT_PROXYUSERPWD |
| 04:44.16 | Qwell | the latter could be pretty important to some people |
| 04:44.28 | Corydon76-home | Do you really need those? |
| 04:44.42 | Qwell | just something so that you can set any of those CURLOPT's, and it would loop through the list of vars, and if any match, it'll be set |
| 04:45.02 | Qwell | I don't, no, but others may, and the code appears to be fairly easy to write |
| 04:45.13 | Corydon76-home | I've admittedly been guilty of feature creep in the past, but let's hold off on those until somebody actually needs them |
| 04:46.16 | Qwell | okay, I'll agree to that |
| 04:46.49 | ManxPower | ~docs |
| 04:46.51 | jbot | docs is probably probably Documentation can be found at http://www.digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or http://www.voip-info.org/wiki-Asterisk or http://www.asteriskdocs.org or http://www.oreilly.com/catalog/asterisk or http://www.asteriskguru.com |
| 04:46.52 | ManxPower | ~mailinglist |
| 04:46.54 | jbot | extra, extra, read all about it, mailinglist is Search Asterisk mailing lists by prepending site:lists.digium.com to your Google search. Browse the mailing list archive at http://lists.digium.com/ or search through it at http://www.asteriskguru.com/archives/search.php |
| 04:47.21 | Qwell | Corydon76-home: You can close that bug then. I somehow missed ENV |
| 04:47.30 | bweschke | what jerks. These guys release the SPA-941, and everyone gets excited and starts talking about it... and now it turns out that there's actually a better phone tied already ready to go GA that is tied to their hosted VoIP solution that no one can get access to. http://newsroom.cisco.com/dlls/2005/eKits/Data_Sheet_IP_Manager_Phone.pdf yep.. definitely NOT the old Sipura we all knew anymore. :( |
| 04:47.48 | benjk | Hi Manx, long time no see ... still into that bot thingie, eh?! ;-) |
| 04:48.08 | Corydon76-home | Qwell: it wasn't until 1.2 that you could actually set ENV |
| 04:48.17 | Damin | bweschke: One word. Polycom. ;) |
| 04:48.25 | Corydon76-home | In 1.0, you could read an environmental variable, but not write it |
| 04:48.46 | ManxPower | hmm? |
| 04:48.54 | bweschke | Damin: ya - I hear you.. but Polycom needs to come out with a color screen now to be as sexy as these phones |
| 04:49.15 | bweschke | sound quality rocks on polycoms.. it's what I recommend to my clients, but the marketing machine is alive and well here |
| 04:50.12 | bweschke | a 16 bit color, backlit display.... so you do it for this phone, but not for the 941 and no option outside for anything that isn't tied to your proprietary call control protocols/appliances |
| 04:50.14 | bweschke | dicks |
| 04:50.26 | ManxPower | benjk, It's just that when I'm gone for a while I figure that posting the info to the channel is a good idea. |
| 04:51.09 | Corydon76-home | bweschke: and people wonder why I don't like Cisco equipment |
| 04:51.11 | ManxPower | Life has been....busy. |
| 04:52.55 | Qwell | Corydon76-home: I use head anyhow. good to know though |
| 04:53.33 | benjk | Manx, sure, you may remember I used to spent hours feeding the original bot when it first appeared in here two or three years ago, it was quite fun for a while, too distracting though ;-) |
| 04:53.45 | Corydon76-home | Qwell: I would highly recommend that you get used to running 1.2, for at least the next 3 months |
| 04:54.32 | bweschke | haha.. ya - us hackers can't wait to get our hands on that codebase and wreck it up good. :) |
| 04:54.43 | Corydon76-home | Qwell: it is our intention to use the next 3 months to break CVS HEAD in new radical ways, before we stabilize it for the 1.4 branch |
| 04:55.56 | Corydon76-home | So it's probably okay to use HEAD for about the next week or so, but after that, don't expect HEAD to be production-quality (at least until after the feature freeze goes into place for 1.4) |
| 04:56.13 | Nugget | will 1.2 go neglected during that time, as 1.0.x did, or will there people people who care about keeping 1.2 stable? |
| 04:56.21 | Nugget | s/people people/be people/ |
| 04:56.43 | Corydon76-home | Nugget: No, we're going to leave 1.2 blowing in the wind |
| 04:56.54 | Nugget | was that supposed to be sarcasm? |
| 04:56.54 | benjk | I don't even expect 1.2 to be production quality until at least 1.2.6 or 1.2.7 |
| 04:57.18 | Corydon76-home | Nugget: you post a troll and expected a serious response? |
| 04:57.23 | Nugget | it wasn't a troll. |
| 04:57.27 | Nugget | it was a legitimate question |
| 04:57.36 | Corydon76-home | 1.0 was maintained |
| 04:57.37 | benjk | 1.0 was a buge disaster until 1.0.7 |
| 04:57.56 | Nugget | we must have different concepts of what "maintained" means, then. |
| 04:57.58 | benjk | s/buge/huge |
| 04:58.01 | bweschke | Nugget: do you upgrade to the new version of Windows as soon as it comes out? |
| 04:58.06 | Nugget | I don't run windows. |
| 04:58.12 | benjk | :-) |
| 04:58.23 | Corydon76-home | Do you upgrade to .0 releases of FreeBSD on your production machines? |
| 04:58.32 | Nugget | but that's really irrelevant. I run plenty of other software that manages to address the needs of users and not just developers and hackers. |
| 04:58.32 | benjk | Nugget: beautiful reply! |
| 04:58.42 | Nugget | asterisk has failed miserably at that goal, from my perspective. |
| 04:58.43 | Corydon76-home | Or God-forbid, did you upgrade to FreeBSD 4.4 on your production equipment? |
| 04:59.02 | Corydon76-home | Nugget: and you're still here, because? |
| 04:59.03 | Nugget | I track the -stable branch with freebsd, which I sorely wish were an option with asterisk. |
| 04:59.05 | bweschke | Nugget: sorry to hear that. I'm quite successful running a business with it |
| 04:59.13 | Nugget | because I run asterisk and I want it to succeed. |
| 04:59.32 | Corydon76-home | Nugget: and you of course contribute patches, right? |
| 04:59.32 | Nugget | but I hate that I've pretty much had to move my servers to HEAD because 1.0.x is such a wasteland of neglect. |
| 04:59.42 | Nugget | Now who is trolling, Corydon76-home? |
| 04:59.59 | Corydon76-home | I'm just responding in kind |
| 05:00.11 | Nugget | No, you really aren't. You're being antagonistic. |
| 05:00.12 | bweschke | define "wasteland of neglect" ? drumkilla wouldn't be happy to hear you say that |
| 05:00.34 | Corydon76-home | Oh, and you aren't? |
| 05:00.43 | Nugget | No. |
| 05:00.44 | *** join/#asterisk `Sauron (i=sauron@h-69-3-12-50.hstqtx02.covad.net) |
| 05:00.59 | bweschke | and with the 1.2 base you're going to find that the folks working ABE are going to take an active interest in making sure that the 1.2 base is as stable as possible |
| 05:01.14 | Corydon76-home | Nice to see that you have different definitions of antagonistic when applied to yourself or others |
| 05:01.22 | Nugget | I'm an asterisk end user who has been forced into running the development branch of the code, which I wish I could avoid, because critical features and interoperability was harmed in the release versions. |
| 05:01.37 | Nugget | Heck, Digium even disavowed the "stable branch" and doesn't even call it such any more. |
| 05:01.50 | *** join/#asterisk marc32422 (n=marc3234@206-248-153-156.dsl.teksavvy.com) |
| 05:01.55 | bweschke | you're reading too much into that statement. |
| 05:02.06 | Corydon76-home | We don't call it stable, because people have certain beliefs about that word |
| 05:02.16 | Corydon76-home | Our release branch is feature-frozen |
| 05:02.20 | Nugget | I've been hopeful that 1.2's release marked a new development philosophy which addressed the needs of users such as myself, but it sounds like you aren't very concerned with actual users. |
| 05:02.51 | Corydon76-home | You got me. I'm wholly unconcerned with my customers. |
| 05:02.57 | marc32422 | voip-info says asterisk realtime db is unstable-- crashes. |
| 05:03.07 | Nugget | I hope that 1.2 doesn't fall into disrepair and neglect as I feel that 1.0.x did as all the development energy was spent on HEAD features and rapid development. |
| 05:03.20 | Qwell | Corydon76-home: I dev, so...it's head, good or bad |
| 05:03.21 | bweschke | Nugget: but I guess that's where I'm a little confused with where this conversation has gone. what is your expectation of the 1.2 branch? |
| 05:03.22 | Corydon76-home | voip-info is editable by literally anybody |
| 05:03.52 | Nugget | I would expect that 1.2 (and subsequent patches to that branch) would continue to be useful and viable no matter what goes in in the development branch. |
| 05:04.03 | Nugget | until such a time as the next major release is readied. |
| 05:04.31 | marc32422 | how do you change the greeting of voicemailmain? |
| 05:04.36 | bweschke | I think that's going to happen... as I said, the next major release of ABE and subsequent minor releases of ABE after that will be based on 1.2 |
| 05:04.56 | Corydon76-home | As the developers, we fully expect to be running 1.2 ourselves, on our production systems, until 1.4 is released. |
| 05:05.08 | Nugget | that's encouraging to hear. |
| 05:05.11 | bweschke | but 1.0.10 was released just a few days back too. so I'm not sure how you could say 1.0 has been completely neglected |
| 05:05.27 | ManxPower | You say that now, but when CVS-HEAD gets cool features you'll all switch to that. |
| 05:05.34 | Corydon76-home | Or at least, until the public betas of 1.4 are released |
| 05:05.48 | bweschke | if those features are stable enough - I might... it depends.. |
| 05:06.05 | ManxPower | bweschke, one developer works part time on 1.0 |
| 05:06.13 | Nugget | cvs head is a lot of work to use. I'd rather not need to be so watchful of the asterisk development process. |
| 05:06.16 | Corydon76-home | Depends on whether an application I need depends on features that are only in HEAD |
| 05:06.20 | ManxPower | I call that neglect |
| 05:06.30 | bweschke | and I guess that's where we differ.... we wouldn't backport new features to 1.2 or 1.0 for that matter |
| 05:06.44 | ManxPower | bweschke, *nod* |
| 05:06.57 | Corydon76-home | and by new features, we mean changing the API |
| 05:07.10 | ManxPower | The SOLUTION to all this is a shorter release cycle. |
| 05:07.13 | Corydon76-home | or adding APIs to the core |
| 05:07.17 | ManxPower | 1.2 is a major disapointment for me. |
| 05:07.18 | bweschke | Digium heard you |
| 05:07.26 | bweschke | they're looking to get to 1.4 by 6/1 |
| 05:07.28 | Corydon76-home | ManxPower: we're looking at 6 months for a release cycle |
| 05:07.28 | bweschke | not a year this time |
| 05:08.19 | Corydon76-home | and Digium is also looking to make control of the project a bit more independant of Digium |
| 05:08.31 | Corydon76-home | So more community based |
| 05:08.43 | docelm0 | DAMNIT! |
| 05:08.50 | docelm0 | DTMF ISSUES! GRRR! |
| 05:09.18 | bweschke | with regard to drumkilla and 1.0 - I think we can all agree that he was much more than a "part time guy" on 1.0 maintenance in the past year. but agree that more should be done.. and I think by nature of what ABE has become, the open source community will benefit by those resources contributing back fixes to the 1.2 branch |
| 05:09.27 | docelm0 | How is it from Voip -> PSTN no dtmf but PSTN -> voip works????? |
| 05:09.48 | ManxPower | docelm0, your rxgain is out of whack? |
| 05:09.57 | Corydon76-home | docelm0: which protocol and which method of passing DTMF? |
| 05:09.59 | docelm0 | no.. Cause its voip to voip |
| 05:10.06 | Corydon76-home | and which codec? |
| 05:10.12 | docelm0 | I am using g729 to asterisk and ULAW out of asterisk |
| 05:10.16 | ManxPower | docelm0, then it's always an issue of mismatched dtmfmodes |
| 05:10.22 | docelm0 | I have both sides set to auto |
| 05:10.38 | Corydon76-home | You need to set both sides to either info or rfc2833 |
| 05:10.43 | ManxPower | I didn't know there was a dtmfmode=auto in sip.con |
| 05:10.47 | Vco | hey... bweschke |
| 05:10.48 | Corydon76-home | but pick one |
| 05:10.48 | bweschke | there is |
| 05:10.51 | bweschke | Vco: hey |
| 05:10.52 | Vco | re: voicemail |
| 05:10.54 | Vco | thanks! |
| 05:10.55 | file[laptop] | it got added awhile ago... |
| 05:11.00 | bweschke | np |
| 05:11.01 | file[laptop] | I hate it. |
| 05:11.17 | bweschke | ya - it's not very intuitive... dtmfmode=auto that is |
| 05:11.31 | docelm0 | All I know is I tried to inband and rfc2833 for both sides and such and it still doesnt work |
| 05:11.35 | ManxPower | docelm0, try forcing it like we did back in my day, where we had to carry each DTMF 2 miles every day. |
| 05:11.37 | docelm0 | I am about fit to be tied.. |
| 05:11.51 | Corydon76-home | Do NOT try to do inband DTMF with g729 |
| 05:11.56 | ManxPower | docelm0, you wans rfc2833 |
| 05:12.01 | ManxPower | all the way, even with ulaw. |
| 05:12.02 | Corydon76-home | It's guaranteed to screw up |
| 05:12.04 | bweschke | agree |
| 05:12.06 | docelm0 | tried.. doesnt work |
| 05:12.10 | bweschke | AVT / 2833 is you're best bet |
| 05:12.24 | ManxPower | docelm0, it's not going to do any good unless your force it on all devices |
| 05:12.27 | docelm0 | I have tried every combo I can think og.. |
| 05:12.28 | bweschke | well - if that's not working for you then - turn on the dtmf logger channel and monitor that you're sending when you should be sending |
| 05:12.50 | docelm0 | so force rfc2833 @ the incoming and force rfc2833 on outgoing? |
| 05:12.51 | benjk | I get immediate hangups if I sent DTMF over a SIP connection to my Japanese VoIP, no matter which version of Asterisk, no matter which DTMF mode I use |
| 05:12.54 | *** join/#asterisk a_phantasm (i=phantasm@pa-nantyglo1b-75.pittpa.adelphia.net) |
| 05:12.56 | docelm0 | what is g711u? |
| 05:12.59 | ManxPower | and if your SIP device isn't sending a telephone event when you press keys then it's not set for rfc2833 |
| 05:13.08 | a_phantasm | Greetings everyone |
| 05:13.10 | docelm0 | err what if |
| 05:13.12 | ManxPower | docelm0, force it ON THE DEVICE. |
| 05:13.18 | wasim | docelm0: its an aberration of good ol' alaw |
| 05:13.44 | docelm0 | I know what it is.. Im saying about passing rfc2833 over ulaw |
| 05:13.55 | benjk | I guess with DTMF over SIP to a provider, you can't really tell where the problem is, it may be further upstream |
| 05:13.56 | docelm0 | I would but I doubt my carrier will be happy |
| 05:13.57 | ManxPower | um, g711u = ulaw = pcmu |
| 05:14.05 | ManxPower | g711a = alaw = pcma |
| 05:14.07 | a_phantasm | Is anyone familiar with why 3 hardware SIP phones can not dial to extentions through asterisk? ie. call extention 1331 and you get the person is on the phone, but they aren't. |
| 05:14.38 | file[laptop] | you don't pass rfc2833 over ulaw, it's just a separate packet type in the RTP stream |
| 05:14.43 | file[laptop] | so it works with any codec |
| 05:15.16 | ManxPower | We use RFC2833 DTMF with ulaw all the time with our polycoms so any transient network hickup won't mess up DTMF |
| 05:15.53 | bbz | Would anyone here be interested in some Polycom 501 phones? We got stuck w/ an extra 20 after a project, and we need to unload them at cost, -- if anyone is interested send me a private message (if i dont respond immediately, leave me your email address, and i can send you info) i have a reference in the channel if you need. -- |
| 05:16.13 | file[laptop] | bbz: dejavu |
| 05:16.21 | ManxPower | The only issue we have ever had with DTMF are: 1) rxgain or txgain out of whack caused odd DTMF screwieness (missed digits, doubled digits, etc), and 2) Asterisk's Zaptel DTMF length is WAY too short, so we set it longer in digits.h |
| 05:16.31 | Corydon76-home | bbz: save them; some of them will break eventually |
| 05:16.40 | bweschke | bbz: what is "cost" ? are they NIB? |
| 05:18.31 | *** join/#asterisk maik_ (n=maik@bfs.cs.uni-sb.de) |
| 05:19.36 | *** join/#asterisk mog_home (n=mogorman@user-24-236-84-48.knology.net) |
| 05:20.27 | *** join/#asterisk CoolAcid (n=jason@216.99.98.39) |
| 05:21.10 | Damin | docelm0: You still alive? |
| 05:21.20 | docelm0 | no |
| 05:21.25 | docelm0 | Im going to sleep pissed off |
| 05:21.35 | bweschke | Damin: how'd you make out with that logging patch? |
| 05:21.45 | *** join/#asterisk beernuts (n=mattfox7@CPE-60-228-213-135.qld.bigpond.net.au) |
| 05:21.56 | docelm0 | ya whats up man? |
| 05:22.05 | ptiggerdine | the sky? |
| 05:22.16 | mog_home | planets? |
| 05:22.22 | mog_home | ceiling |
| 05:22.24 | beernuts | moon |
| 05:22.30 | ptiggerdine | maybe not if your standing on ya nogen |
| 05:22.35 | beernuts | in the southern hemisphere |
| 05:23.16 | ptiggerdine | might make a dkms for zaptel and voicetronix drivers |
| 05:23.24 | ptiggerdine | make my life easy |
| 05:24.29 | Damin | bweschke: I didn't.. |
| 05:24.56 | Damin | docelm0: If I force my SIP peer to use inband DTMF to you, it works. But that isn't a long term solution. |
| 05:25.18 | docelm0 | Then its my provider |
| 05:25.27 | docelm0 | who I am sending to |
| 05:25.32 | bweschke | did you still want to do it, or not worth it? |
| 05:25.34 | docelm0 | they want inband.. those bastards |
| 05:25.50 | *** join/#asterisk maik_ (n=maik@bfs.cs.uni-sb.de) |
| 05:25.52 | Damin | bweschke: Is the patch worth it? I think so.. |
| 05:26.22 | Damin | bweschke: I was going to backport it, but if you have the energy go for it.. :) |
| 05:26.48 | Damin | docelm0: I see an X/O owned IP somewhere in the SIP headers.. some 211.x.x.x address or some such shit.. ;) |
| 05:26.55 | docelm0 | hehe |
| 05:26.59 | docelm0 | yes |
| 05:27.29 | bweschke | well - to just have it on indefinitely - it's cake... to enable rtpdebug and then have it log when you're rtpdebugging.. that's a little bit more... |
| 05:27.35 | beernuts | any australians here??? |
| 05:27.38 | docelm0 | but thats my inband issue.. 800 calls work with dtmf |
| 05:28.41 | Damin | docelm0: Are your asterisk server staying in the media path? |
| 05:28.56 | docelm0 | yes transcode yes.. ulaw should |
| 05:29.09 | docelm0 | I belive I have canreinvite=no on all of them |
| 05:30.09 | Damin | docelm0: So when I send to 72.9.111.28 I'm connecting to an Asterisk server? |
| 05:30.38 | docelm0 | Nope.. SER |
| 05:30.44 | Damin | docelm0: Does your 800 service go out PRI/CT1? |
| 05:30.48 | docelm0 | PRI |
| 05:31.12 | Damin | docelm0: So if I send you an 800 call w/ RFC2833 and it works, then we're sure its between you and X/O? |
| 05:32.06 | newsmafia | Does any one know how the Linksys SPA941 compares to the cisco 79XX? I'd like to get a few for my home system. |
| 05:32.33 | docelm0 | I know its a XO problem.. I dont doubt that.. I had to modify chan_sip.c to get xo to work |
| 05:33.07 | zobia | hello. |
| 05:33.39 | a_phantasm | Is there a reason why asterisk will say a SIP phone is in use when it is not! |
| 05:33.41 | docelm0 | ahh well Im going to bed.. Damin I will catch you online tomorrow.. |
| 05:33.51 | docelm0 | You broke it? |
| 05:34.35 | Damin | docelm0: BTW.. when I send out via you w/ 800 service, rfc2833 works.. |
| 05:34.37 | zobia | hello qwell |
| 05:34.57 | zobia | can i dial one number with 10 lines at the same time. |
| 05:35.15 | *** join/#asterisk oogle_ (n=jart@ool-4578cc36.dyn.optonline.net) |
| 05:36.35 | docelm0 | I know |
| 05:36.47 | docelm0 | its going to a cisco I setup.. I know that shit is right.. I did it.. |
| 05:36.55 | docelm0 | :) Cya tomorrow ASS MONKEY! |
| 05:37.36 | bweschke | Damin: what ver of 1.0 is this? 1.0.10 ? |
| 05:37.49 | bweschke | not that it should matter too much I think |
| 05:38.28 | dos000 | damin, i am still not sure why even after putting a passwd in sip.conf the endpoints are able to call whithout registering ! i apologize if i am missing something 8-; |
| 05:38.35 | Damin | bweschke: Im running 1.0 via CVS. Just updated tonight, so 1.0.10+ should be fair.. |
| 05:38.41 | bweschke | cool |
| 05:38.46 | bweschke | that's what I'm diff'ing against |
| 05:39.12 | Damin | dos000: What does the debug say? |
| 05:40.18 | Qwell | zobia: Now why would you want to do something like that? |
| 05:40.21 | dos000 | i have tehtereal running and i can see them sending invites by .. but never a register |
| 05:41.02 | dos000 | Damin, only 2 of them are sending registers. so sip show peers has them listed |
| 05:41.30 | ptiggerdine | beernuts, I'm a queenslander does that count? |
| 05:41.32 | dos000 | Damin, there is totally no difference in sip.conf between them. they all need passwd |
| 05:41.53 | *** join/#asterisk YoMama (n=r00t@pcp02689850pcs.roylok01.mi.comcast.net) |
| 05:43.09 | Damin | dos000: Don't know. Perhaps you have a context= in your [general] section that is passing through all SIP traffic? |
| 05:43.18 | Damin | dos000: What is the IP address of your Asterisk server? |
| 05:43.21 | *** join/#asterisk twisted (n=twisted@asterisk/friend-and-developer/pdpc.professional.twisted) |
| 05:43.21 | *** mode/#asterisk [+o twisted] by ChanServ |
| 05:43.29 | Damin | twisted: Werd! |
| 05:43.31 | bweschke | Damin: patch is rdy. u want me to email it to you? |
| 05:43.41 | Damin | bweschke: Sure.. |
| 05:43.43 | YoMama | my sip proxy doesn't work consistently...anyone good with extensions.conf and sip.conf? |
| 05:43.51 | YoMama | i'm thinking it might even be a bug |
| 05:43.55 | YoMama | it's just utterly bizarre |
| 05:43.59 | dos000 | Damin, 192.168.15.64 |
| 05:44.17 | dos000 | Damin, they are on the same network |
| 05:44.51 | Damin | dos000: Don't know. Paste your sip.conf file into pastebin.ca and then give me the link to review it. |
| 05:46.12 | *** join/#asterisk Cresl1n (n=matt@user-24-236-124-147.knology.net) |
| 05:46.20 | YoMama | Darrin: wanna have a go at my issue? i had three people boggled this afternoon |
| 05:49.08 | bweschke | damin: try that.. just throw it a "rtp debug" once you've got it compiled in and 2833 events should then start to go to the debug logger channel |
| 05:49.41 | bweschke | to stop it again just issue it "rtp no debug" |
| 05:49.47 | YoMama | hmm...i forgot about the sip debug command... |
| 05:50.44 | YoMama | when you do a sip debug..is ther eanyway to redirect that stuff into a file? |
| 05:51.15 | bweschke | yes |
| 05:51.17 | bweschke | logger.conf |
| 05:51.55 | YoMama | and..how do u turn it off? |
| 05:52.02 | YoMama | sip debug peer <peername> |
| 05:52.07 | YoMama | i turned that on..now how do i turn it off? |
| 05:52.15 | bweschke | sip no debug ? |
| 05:52.32 | YoMama | thanks |
| 05:52.36 | bweschke | np |
| 05:52.42 | YoMama | almost like Cisco IOS..except the no goes first |
| 05:52.47 | bweschke | correct |
| 05:53.24 | bweschke | and not so much like it that you're not left with a smokey hole in your wallet where your money used to be. :) |
| 05:54.01 | *** part/#asterisk jcath (n=skycat@159.226.21.127) |
| 05:54.29 | YoMama | bweschke: ha..i'm just having this bizarre problem with not being able to bridge a Zap with a sip proxy |
| 05:55.17 | trixter | that is strange becuase that normally works |
| 05:55.41 | bweschke | what is the exact problem? call drops? etc? |
| 05:56.27 | YoMama | bweschke: call doesn't go through and i get a 400 Bad Request from the sip provider |
| 05:56.36 | YoMama | the irony is that if i place a call from an internal extension..it works just fine |
| 05:57.17 | bweschke | does your provider have rules about what your from: line in your sip hdr can be? (eg - it must be the DID that's assigned to you and nothing else) |
| 05:57.39 | YoMama | maybe...it's outbound only |
| 05:57.44 | YoMama | but i'm thinking it might be the from header |
| 05:57.55 | YoMama | i'm thinking it's gotta be something about the packet |
| 05:58.02 | YoMama | because why would they reject it |
| 05:58.05 | bweschke | could be. won't know the difference between internal calls and zap originated calls until you capture a sip trace in the file |
| 05:58.18 | YoMama | it's not a service that openly supports asterisk..so i'm not gonna get help from them |
| 05:58.26 | YoMama | good idea |
| 05:58.28 | bweschke | you probably don't need it |
| 05:58.28 | YoMama | so logger.conf |
| 05:58.34 | YoMama | don't need what? |
| 05:58.43 | bweschke | if you've got your internal extensions able to call out, you just mimic that |
| 05:58.54 | bweschke | don't need help from your provider to "support asterisk" |
| 05:59.12 | bweschke | yes.. logger.conf |
| 05:59.27 | YoMama | ok |
| 05:59.29 | YoMama | lemme give this a shot |
| 05:59.31 | bweschke | that's got some good config data in it which will allow you to break out what messages go where |
| 05:59.41 | YoMama | i can affect the header? |
| 05:59.43 | dos000 | Damin, http://pastebin.ca/30512 i am using * 1.0.9 |
| 06:01.02 | Damin | dos000: Get rid of "context=default" in your [general] section. |
| 06:01.25 | YoMama | bweschke: so this thing uses syslog i take it |
| 06:01.46 | bweschke | it can. doesn't have to |
| 06:01.48 | dos000 | Damin, FTR endpoints are behind routers so they need nating ! so i have inet <-> asterisk <-> router <-> sip_ata |
| 06:02.17 | bweschke | by default, it doesn't |
| 06:02.27 | Damin | dos000: Define it for the users only. If anything, create a new context called [denied] that contains extension => s,1,HangUp in it and then put that back in there. |
| 06:03.08 | dos000 | Damin, ok ! lemme try that |
| 06:05.04 | YoMama | does reload reload the logger? |
| 06:06.54 | bweschke | no |
| 06:07.02 | *** join/#asterisk LANmower (n=LANmower@ndn-165-158-234.telkomadsl.co.za) |
| 06:07.03 | YoMama | hehe..i figured that out |
| 06:07.07 | LANmower | mornin' all :) |
| 06:07.16 | YoMama | i uncommented debug..but now it's debugging everything |
| 06:07.19 | LANmower | anyone here with some festival experience |
| 06:07.20 | LANmower | ? |
| 06:07.25 | YoMama | i only turned on sip debug peer |
| 06:07.35 | bweschke | that's ok |
| 06:07.48 | YoMama | eh..screw it..ok..time to do some tests |
| 06:07.49 | wasim | LANmower: i went to octoberfest, mardi gras and spring break, but haven't been to carnival as yet |
| 06:08.02 | bweschke | you're only going to want this on long enough to grab what you need and then you'll disable it again |
| 06:08.05 | wasim | LANmower: and ofcourse basant |
| 06:08.15 | LANmower | er |
| 06:08.36 | LANmower | wasim, thats very nice :), but I dont think that will help me generate voice... |
| 06:09.32 | YoMama | bweschke: hmm..the sip header info isn't int he debug file |
| 06:09.48 | bweschke | it should be... did you grab verbose as well? |
| 06:09.56 | YoMama | i'm on level 3 |
| 06:11.25 | Damin | bweschke: I can think of really evil things to do w/ this code.. |
| 06:11.31 | bweschke | :) |
| 06:12.24 | bweschke | well - it's not quite as bad as logging all the inbound and outbound rtp packets blindly, but it's a bit "chatty" on a busy system. :) |
| 06:12.28 | Damin | bweschke: Post it as a patch for 1.0 on the bugtracker. At least it'll be archivedu p there.. :) |
| 06:12.39 | bweschke | true enuff |
| 06:13.01 | LANmower | I take it no one here really used festival then... |
| 06:14.39 | YoMama | i think i just figured it out :) |
| 06:15.27 | YoMama | bweschke: i think i actually found a bug |
| 06:15.38 | *** part/#asterisk LANmower (n=LANmower@ndn-165-158-234.telkomadsl.co.za) |
| 06:15.53 | *** join/#asterisk LANmower (n=LANmower@ndn-165-158-234.telkomadsl.co.za) |
| 06:16.40 | bweschke | k. shoot |
| 06:17.25 | YoMama | bweschke: I have a SetCallerID("Incoming Landline <9999999999>",a) in my dialplan |
| 06:17.44 | LANmower | bleh |
| 06:17.51 | YoMama | i don't have caller id on my landline and i wanted the screen on my sip phone to read something decent...so i did that |
| 06:17.56 | YoMama | well, my sip header says |
| 06:18.10 | YoMama | ""Incoming Landline" <sip:username@.... |
| 06:18.13 | YoMama | see the problem? |
| 06:18.36 | bweschke | ya |
| 06:18.39 | bweschke | the quotes are off |
| 06:19.00 | YoMama | yeah..i don't know where it's getting the other quote because they're balanced in my extensions file |
| 06:19.12 | YoMama | so i'm thinking someone fatfingered the parser |
| 06:19.16 | bweschke | this isn't actually the first time I've seen this |
| 06:19.33 | bweschke | I've seen some real funky stuff happen with clid handing |
| 06:19.36 | bweschke | handling rather |
| 06:19.44 | YoMama | bah |
| 06:19.54 | YoMama | is there a place where u can force the header? |
| 06:20.05 | bweschke | well - depends what part of the header |
| 06:20.11 | YoMama | the To: |
| 06:20.17 | bweschke | chan_sip probably isn't going to let you mess with To: and From: etc |
| 06:20.28 | YoMama | wait |
| 06:20.31 | YoMama | no..it's the From: |
| 06:20.47 | YoMama | should i report it as a bug? |
| 06:21.15 | bweschke | sure. you can do that. what happens if you use a single tick in setcallerID instead of dbl ticks? |
| 06:21.25 | YoMama | this means that...if i put a caller id line in my sip.conf file for my desk phone..i bet i'll have the same problem |
| 06:21.29 | YoMama | single tick..let's give it a shot |
| 06:22.00 | bweschke | in theory, yes.. but the way those values are read (one from the file and the other from an app argument) is completely different |
| 06:22.31 | YoMama | huh..ok |
| 06:22.53 | bweschke | you could try the SIP_HEADER dial plan function in 1.2. It's supposed to let you set sip headers ... but again.. don't know if it will let you set headers the system wants to set on it's own already |
| 06:23.16 | YoMama | ta da! it works! |
| 06:23.26 | bweschke | single tick works? |
| 06:23.28 | YoMama | yup |
| 06:23.33 | YoMama | if i were gay or a chick..i'd kiss ya |
| 06:23.35 | YoMama | hahaha |
| 06:23.39 | YoMama | thanks man |
| 06:23.41 | bweschke | np |
| 06:23.47 | YoMama | i was up 'til 4 am last night with that |
| 06:24.10 | bweschke | well - at least you get to go to bed earlier tonight with it fixed |
| 06:24.20 | bweschke | which is where I'm headed now I think... g'night everyone |
| 06:24.25 | YoMama | nite nite |
| 06:24.26 | YoMama | thanks again |
| 06:24.43 | bweschke | nite. np |
| 06:24.57 | *** join/#asterisk testmachine (n=assink@ip237-239-58-62.adsl.versatel.nl) |
| 06:26.06 | *** join/#asterisk Damin (n=damin@nucleus.nacs.net) |
| 06:26.11 | Damin | fire: I think the shit that twisted just committed will solve your ring cadence issues. |
| 06:26.16 | *** join/#asterisk scardinal (n=supreme@port816.ds1-suoe.adsl.cybercity.dk) |
| 06:28.03 | twisted | Damin, what was fire's issue? |
| 06:28.08 | twisted | all I did was fix spelling :P |
| 06:28.10 | trixter | yes |
| 06:28.16 | Damin | twisted: Don't know. |
| 06:28.32 | Damin | twisted: But he was babbling about something.. |
| 06:28.41 | trixter | fire had some problem with cadence, he said something eventually about undocumented changes and implied he got it working I dunno though |
| 06:28.41 | twisted | ah ok |
| 06:29.07 | benjk | Any header the config doesn't let you change can easily be changed in the source of chan_sip. I had to do this for a couple of Japanese VoIP services which officially only support their own end user devices. |
| 06:29.09 | Damin | twisted: Buy Ken Rice a beer for me.. |
| 06:29.15 | twisted | oddly enough, it was spelled correctly in zaptel, but not asterisk. |
| 06:29.22 | trixter | whatever the specific problem was he said that 1.2 and 1.0.10 had it.. I thought that 1.0.10 wasnt official ... |
| 06:29.22 | twisted | Damin, what'd he do? |
| 06:29.31 | LANmower | *sigh, festival can be such a hassle... |
| 06:29.34 | Damin | twisted: We were both lamenting the lameness of the AudioCodes gateways.. :) |
| 06:29.39 | twisted | oh |
| 06:29.40 | twisted | haha |
| 06:29.41 | twisted | yeah |
| 06:30.06 | Damin | twisted: We just had a meeting of the minds and came to the conclusion that while they suck, they are still the best thing out there price/performance wise.. |
| 06:30.21 | twisted | *nods* |
| 06:30.22 | Damin | twisted: Flexible little bastards! |
| 06:30.46 | Damin | twisted: Now we just need proper T.38 support in Asterisk.. |
| 06:30.50 | twisted | hehe |
| 06:30.59 | twisted | soon |
| 06:31.05 | twisted | said the birdie to the bee |
| 06:31.18 | Damin | Seriously.. that is a major issue for these SMB deployments.. |
| 06:31.48 | Damin | twisted: We want them COMPLETELY off the PSTN so that SBC doesn't have ANY opportunity to send them advertising crap intheir bill etc.. |
| 06:32.07 | Damin | Does anyone know of any T.38 fax providers? |
| 06:33.51 | YoMama | dammit..i have follow me working |
| 06:35.17 | tainted_ | is it possible to port DIDs from CLEC/ILEC? |
| 06:35.46 | tainted_ | do i have to get PRI with that CLEC/ILEC to retain DID? |
| 06:35.51 | YoMama | hmm |
| 06:35.54 | YoMama | u in the US? |
| 06:35.58 | tainted_ | yea |
| 06:36.01 | wasim | tainted_: you can do it over ss7 as well |
| 06:36.02 | YoMama | i'm thinking that # portability extensions to DIDs as well |
| 06:36.06 | tainted_ | los angeles / verizon |
| 06:36.23 | *** join/#asterisk MGSsancho (n=user@ppp-67-126-241-139.dsl.irvnca.pacbell.net) |
| 06:36.45 | tainted_ | wasim but am I limited to TDM? |
| 06:37.04 | wasim | tainted_: nope, if your clec will give it to you on IP |
| 06:37.22 | tainted_ | know offhand if verizon does? |
| 06:37.36 | tainted_ | i'm sure they want an arm & a leg |
| 06:38.51 | *** join/#asterisk shido6 (n=shido6@d221-68-216.commercial.cgocable.net) |
| 06:42.01 | IronHelix | never fear, for I is here |
| 06:42.04 | IronHelix | who missed me? |
| 06:42.11 | IronHelix | (probably only jbot) |
| 06:42.59 | IronHelix | heh |
| 06:43.00 | IronHelix | sup dude |
| 06:43.16 | mog_home | de nada |
| 06:43.18 | mog_home | e tu |
| 06:43.30 | IronHelix | same, bored :( |
| 06:43.39 | IronHelix | got a bunch of work to do but procrastinating it |
| 06:44.20 | LANmower | oh well, it seems like no one here knows anything about festival. bbl |
| 06:45.06 | IronHelix | not really, but i like the nick tho |
| 06:45.08 | IronHelix | clever++ |
| 06:45.27 | LANmower | thx ;) cheers till later.. |
| 06:45.31 | IronHelix | cya |
| 06:45.36 | *** part/#asterisk LANmower (n=LANmower@ndn-165-158-234.telkomadsl.co.za) |
| 06:45.44 | tainted_ | hey IronHelix |
| 06:45.54 | tainted_ | i got the spec |
| 06:45.55 | IronHelix | sup tainted |
| 06:45.57 | IronHelix | awesome |
| 06:45.59 | IronHelix | mail it up |
| 06:46.15 | tainted_ | k |
| 06:46.19 | tainted_ | gimme about 10 mins |
| 06:46.53 | YoMama | Iron: figured it out..bweschke helped me |
| 06:47.29 | IronHelix | what was wrong? |
| 06:47.35 | IronHelix | cool tainted take your time |
| 06:47.38 | YoMama | Iron: it's a bug... |
| 06:47.48 | YoMama | I had a SetCallerID line in there |
| 06:47.49 | YoMama | like so: |
| 06:48.06 | YoMama | SetCallerID("Incoming Line <1234567890>",a) |
| 06:48.15 | YoMama | just like how it shows it in the manual |
| 06:48.27 | YoMama | however...when it was creating the sip header..it was doing this |
| 06:48.41 | YoMama | From: ""Incoming Landline" <sip:username@domain.tld> |
| 06:48.45 | YoMama | see the problem? |
| 06:48.59 | YoMama | fun lemme tell ya |
| 06:49.02 | *** join/#asterisk insomni (n=insomni@x1-6-00-e0-18-6f-34-ad.k455.webspeed.dk) |
| 06:49.25 | IronHelix | ahhh |
| 06:49.33 | IronHelix | it was listning to the setcallerid and not fromuser |
| 06:49.34 | IronHelix | interesting |
| 06:49.52 | IronHelix | in theory fromuser and fromdomain should override setcallerid |
| 06:50.55 | YoMama | 1,Iron: nope |
| 06:50.58 | YoMama | oops..Iron: nope |
| 06:51.20 | YoMama | for some reason..it tacks on the ANI from SetCallerID |
| 06:51.29 | YoMama | should i report this? |
| 06:51.47 | Qwell | use RPID |
| 06:51.59 | YoMama | RPID? |
| 06:52.08 | Qwell | ~rpid |
| 06:52.09 | jbot | i guess rpid is Remote-Party-ID |
| 06:52.29 | YoMama | well, it works..i just used a ' instead of a " and boom..no more misplaced quotes |
| 06:52.30 | IronHelix | ahhh |
| 06:52.35 | IronHelix | its the ,a that got you |
| 06:52.38 | Qwell | Don't use quotes at all |
| 06:52.50 | YoMama | quotes are in the manual though |
| 06:52.54 | Qwell | It's wrong |
| 06:53.00 | YoMama | hehe |
| 06:53.19 | YoMama | Iron: if you're right tho..somehow...it's passing along the ANI and fromuser is supposed to override it...correct? |
| 06:53.29 | YoMama | i'm just wondering if i should report it as a bug..the damn thing finally works right for me |
| 06:53.34 | Qwell | rpid |
| 06:53.35 | Qwell | :p |
| 06:54.02 | Qwell | SIP + callerid = pfft |
| 06:54.06 | Qwell | sip + rpid = <3 |
| 06:54.31 | twisted | Qwell, you're welcome ;) |
| 06:54.37 | Qwell | for? |
| 06:54.41 | Qwell | and, thanks :p |
| 06:54.45 | twisted | me, pushing rpid into head |
| 06:54.49 | Qwell | ahh, yeah |
| 06:54.50 | twisted | fixing, and pushing. |
| 06:54.59 | Qwell | bkw says it was b0rked for a while :p |
| 06:55.03 | twisted | it never was there |
| 06:55.06 | twisted | it wasn't borked. |
| 06:55.08 | twisted | it wasn't there. |
| 06:55.12 | Qwell | dunno |
| 06:55.27 | IronHelix | hehe |
| 06:55.30 | twisted | we'd respect it if we got it, but we never sent it |
| 06:55.31 | Qwell | I was using his patch, then he said not to use it, then he said he was done contributing code... |
| 06:55.53 | Qwell | I tried not to pay attention to that whole deal... |
| 06:55.56 | Qwell | rpid rocks though |
| 06:55.56 | twisted | his patch was my patch, slightly modified, which was someone else's that was borked. |
| 06:56.00 | Qwell | ahh |
| 06:56.07 | YoMama | if i have like 50 NPANXX's that are local to me..so i used regexp to turn them into exten =>....but do i need to add priority 2 and 102 lines for every single one? or is there a way to make catchalls for a specific context? |
| 06:56.12 | YoMama | s? |
| 06:56.38 | Qwell | You can do lik |
| 06:56.38 | Qwell | e |
| 06:56.47 | Qwell | exten => _12345,1,blah |
| 06:56.52 | Qwell | exten => _NXXXX,2,blah |
| 06:56.55 | Qwell | exten => _12345,3,blah |
| 06:56.59 | Qwell | perfectly valid |
| 06:57.12 | Qwell | except for the _ in the non pattern...heh |
| 06:57.26 | YoMama | so this is legal: |
| 06:57.34 | Qwell | and the whole N not equalling 1 thing |
| 06:57.34 | *** join/#asterisk dos000 (n=dos000@CPE00119572fd49-CM00137186e53a.cpe.net.cable.rogers.com) |
| 06:57.44 | YoMama | exten => _91248999XXXX,1,blah |
| 06:57.46 | Qwell | You get the point though |
| 06:57.51 | dos000 | Damin, seems to be working |
| 06:58.04 | YoMama | exten => _91NXXNXXXXXX,2,blah |
| 06:58.08 | Qwell | YoMama: yep |
| 06:58.10 | YoMama | sweet |
| 06:58.14 | Qwell | people will yell at you for doing it, but... |
| 06:58.34 | benjk | YoMama: you should use macros |
| 06:58.56 | YoMama | yell at me? shit..if i don't do it this way...my 70 NPANXX's will turn into 210 lines |
| 06:58.58 | YoMama | of crap |
| 06:59.04 | Qwell | YoMama: thats my argument |
| 06:59.08 | YoMama | benjk: is that the more kosher way of doing it? |
| 06:59.14 | benjk | [macro-MyDial] |
| 06:59.50 | benjk | exten => s,1,dial(.../${arg1}... |
| 06:59.53 | benjk | etc |
| 07:00.11 | benjk | then in your context from which you want to invoke this do ... |
| 07:00.42 | YoMama | huh...that sounds just as clean... |
| 07:00.44 | benjk | exten => blahblah,1,macro(MyDial,${EXTEN}) |
| 07:01.21 | YoMama | nice |
| 07:01.24 | benjk | where ${EXTEN} becomes ${arg1} in the macro-MyDial section |
| 07:01.56 | YoMama | hmm...that's a clean solution too |
| 07:02.05 | YoMama | there's 95 NPANXX's that are local to me |
| 07:02.09 | YoMama | sheesh |
| 07:02.12 | benjk | you need to prefix the context name with "macro-" to let Asterisk know it's a macro and not an ordinary context |
| 07:02.32 | *** join/#asterisk Rowter (n=SilverDr@201.135.26.195) |
| 07:02.34 | YoMama | benjk: thanks...lemme give that a shot..i'm already using macros for my follow me/caller announce routine |
| 07:02.36 | YoMama | they're cool |
| 07:02.52 | Rowter | if I have some ZAP/pseudo on cdr it means meetme's? |
| 07:03.04 | IronHelix | probably |
| 07:03.26 | Rowter | but definitly it does not mean a call out right? |
| 07:03.40 | benjk | I just did this for all US area codes ... |
| 07:03.41 | benjk | ; USA 2xx |
| 07:03.41 | benjk | exten => _8120[12356789]NXXXXXX,1,Macro(DialViaVoipbusters,${EXTEN:1}) |
| 07:03.42 | benjk | exten => _8121[0-9]NXXXXXX,1,Macro(DialViaVoipbusters,${EXTEN:1}) |
| 07:03.42 | benjk | exten => _8122[49]NXXXXXX,1,Macro(DialViaVoipbusters,${EXTEN:1}) |
| 07:03.43 | benjk | exten => _8123[146]iNXXXXXX,1,Macro(DialViaVoipbusters,${EXTEN:1}) |
| 07:03.43 | benjk | exten => _8124[08]NXXXXXX,1,Macro(DialViaVoipbusters,${EXTEN:1}) |
| 07:03.45 | benjk | exten => _8125[12346]NXXXXXX,1,Macro(DialViaVoipbusters,${EXTEN:1}) |
| 07:03.47 | benjk | exten => _8126[027]NXXXXXX,1,Macro(DialViaVoipbusters,${EXTEN:1}) |
| 07:03.49 | benjk | exten => _8127[068]NXXXXXX,1,Macro(DialViaVoipbusters,${EXTEN:1}) |
| 07:03.51 | benjk | exten => _8128[13]NXXXXXX,1,Macro(DialViaVoipbusters,${EXTEN:1}) |
| 07:03.53 | benjk | ; USA 3xx |
| 07:04.00 | Qwell | ~pb |
| 07:04.01 | jbot | [pastebin] a place to paste your stuff without flooding the channel - try http://pastebin.ca/ |
| 07:04.30 | benjk | ok, ok, I'm sorry |
| 07:04.35 | endre | why is that 'i' at [145]6]? |
| 07:04.39 | endre | [146] |
| 07:04.45 | YoMama | hehe |
| 07:04.50 | Qwell | good question |
| 07:04.52 | YoMama | benjk: voipbusters? is that your voip provider? |
| 07:05.21 | benjk | Not really, I am just testing their free calls |
| 07:05.51 | YoMama | free calls??? |
| 07:06.01 | benjk | which means you have to make your dialplan absolutely tight so you make sure you don't accidentally dial something that they charge you $$$ for |
| 07:06.17 | IronHelix | benjk- can you set your outbound caller id with voipbuster? |
| 07:06.34 | benjk | well, I can set it, but I have no clue if they pass it on |
| 07:07.30 | benjk | YoMama: Voipbusters offer free 1 minute calls to 14 countries, one of which is the US |
| 07:07.32 | YoMama | wait..for voipbusters...u can call anywhere in the US for free? |
| 07:07.36 | YoMama | oh |
| 07:07.38 | YoMama | 1 minute? |
| 07:07.41 | IronHelix | yeah pretty much |
| 07:07.41 | YoMama | augh |
| 07:07.50 | IronHelix | if you send them like 5 euros its unlimited tho |
| 07:07.51 | benjk | if you put 10 EUR credit into your account, they lift the 1 min limit |
| 07:08.19 | benjk | HOWEVER, if you dial a number that is not free, then your credit gets depleted |
| 07:08.24 | YoMama | i'm paying $8/month for unlimited... |
| 07:08.28 | benjk | and the quality is solala |
| 07:08.34 | benjk | not really worth paying for |
| 07:09.05 | benjk | but as a free service for a one time deposit of 10 EUR (=12 USD) it is cool |
| 07:09.49 | YoMama | wait..holy smokes..there's 503 NPANXX's that are local to me |
| 07:09.50 | YoMama | sheesh |
| 07:10.41 | benjk | the limit does get lifted and the calls are free: |
| 07:10.44 | benjk | Date Time Number called Duration Amount |
| 07:10.44 | benjk | 2005-11-22 21:49:39 18777008647 00:03:55 FREE! |
| 07:11.11 | IronHelix | bbl |
| 07:12.23 | YoMama | if a line is busy or in use...it goes n + 101? |
| 07:12.33 | benjk | yes |
| 07:12.39 | YoMama | that's different tho |
| 07:12.48 | Qwell | depends on the version |
| 07:12.54 | YoMama | 1.2.0 |
| 07:12.58 | Qwell | Then no |
| 07:13.08 | YoMama | i want it to try another channel to call out on if the other line is in use |
| 07:13.20 | YoMama | but if they call a party and the other end is busy..so be it |
| 07:13.21 | Qwell | check the DIALSTATUS variable afterwards |
| 07:14.38 | YoMama | ah |
| 07:14.40 | YoMama | nice |
| 07:15.03 | YoMama | so if the channel is in use..u get CHANUNAVAIL |
| 07:15.09 | YoMama | but if it's busy, then you get BUSY |
| 07:15.10 | Qwell | yes |
| 07:19.33 | *** join/#asterisk aminorex (n=tony@71-13-40-131.dhcp.dlth.mn.charter.com) |
| 07:23.05 | *** part/#asterisk pussfeller (n=todd@12.150.129.170) |
| 07:24.26 | *** join/#asterisk pussfeller (n=todd@12.150.129.170) |
| 07:25.32 | *** join/#asterisk sahafeez (n=sahafeez@67.109.14.227.ptr.us.xo.net) |
| 07:29.12 | YoMama | hmm..i think it works |
| 07:29.25 | Qwell | of course it works. ;] |
| 07:29.33 | YoMama | oh..it doesn't always work |
| 07:29.49 | YoMama | so my friend gave me an S100U..should i not bother trying to get it to work properly? |
| 07:29.58 | Qwell | why not? |
| 07:30.05 | YoMama | i have all sorts of problems with wcusb loading properly..and when i do..it's pretty whack |
| 07:30.23 | YoMama | umm...because if i have fxsks => 1 and fxoks => 2 |
| 07:30.33 | benjk | wcusb sucks ballz |
| 07:30.34 | YoMama | and try loading either wcfxo and wcusb..neither of them will load |
| 07:30.39 | YoMama | benjk: yes it does |
| 07:30.52 | benjk | its a total waste of time |
| 07:31.05 | YoMama | benjk: hehe..k |
| 07:31.07 | benjk | that's why Digium got rid of it |
| 07:31.18 | YoMama | yeah.i heard there's lots o' problems with it |
| 07:31.33 | YoMama | so if i wanna hook up this really expensive cordless i got |
| 07:31.35 | benjk | there's nothing BUT problems with it |
| 07:31.36 | tzafrir_laptop | YoMama, signalling=fxsks |
| 07:31.39 | YoMama | maybe i need an ATA |
| 07:31.57 | YoMama | tzafrir: i'm talking in zaptel.conf i can't even get it working right |
| 07:32.08 | YoMama | or eventually buy a TDM400 card |
| 07:32.18 | tzafrir_laptop | fxsks=1 |
| 07:32.26 | tzafrir_laptop | no '>' |
| 07:32.31 | benjk | yeah, get an ATA and give those USB adapters to somebody with a Windows machine |
| 07:32.36 | YoMama | tza fxoks=1 actually for the S100U |
| 07:32.45 | tzafrir_laptop | never tried a wcusb before, though |
| 07:32.53 | YoMama | benjk: wait..they work on a windoze box? |
| 07:33.23 | YoMama | benjk: what does it do if it's plugged into a windoze box |
| 07:33.45 | benjk | There are some answering machine apps for Windows which use the same USB adapters |
| 07:33.57 | benjk | also for MacOS X |
| 07:34.27 | benjk | apparently their drivers are working more reliably than wcusb |
| 07:34.37 | YoMama | ha |
| 07:34.44 | *** join/#asterisk aminorex (n=tony@71-13-40-131.dhcp.dlth.mn.charter.com) |
| 07:34.51 | YoMama | but i can't use it to use my windoze box as an ATA |
| 07:34.57 | YoMama | that's really sick actually..forget i asked that |
| 07:35.44 | benjk | I only know because I have stumbled into it on the web some time ago, don't really know much about it |
| 07:36.39 | benjk | the point is that those adapters are no good as doorstops, not heavy enough |
| 07:36.51 | benjk | so what else is there to do with them than to give them away |
| 07:36.51 | YoMama | haha |
| 07:37.07 | YoMama | what the hell..linksys makes this wireless thing and it's for skype??? |
| 07:37.08 | YoMama | bah |
| 07:37.17 | YoMama | what's the cheapest wifi sip phone? |
| 07:37.24 | YoMama | (that isn't a seirous piece of shit) |
| 07:37.27 | benjk | UT Starcom's F1000 |
| 07:38.25 | YoMama | hmm |
| 07:38.33 | benjk | I haven't tested it yet but I know UT Starcom, they are a good company and the products they make that I know are good quality |
| 07:38.34 | YoMama | i'll haveta upgrade my access point first |
| 07:38.46 | YoMama | it's barely 802.11b compatible :-P |
| 07:40.09 | benjk | Amongst all those WiFi SIP phone manufacturers, UT Starcom are the only ones who actually have experience with wireless handsets |
| 07:40.20 | YoMama | hmm |
| 07:40.44 | benjk | They are the number one wireless handset vendor in China |
| 07:40.47 | Qwell | I don't know...Cisco has experience with wifi, and telephony |
| 07:41.03 | benjk | Cisco knows shit about wireless telephony |
| 07:41.08 | YoMama | haha |
| 07:41.21 | YoMama | u guys know which usb handsets work with x-lite? |
| 07:41.26 | benjk | UT Starcom have sold something like 100 million wireless handsets in China |
| 07:41.31 | Qwell | YoMama: any should |
| 07:41.41 | Qwell | it just shows up as a sound card to the system |
| 07:42.07 | YoMama | oh |
| 07:42.16 | YoMama | here's one...$11 |
| 07:42.17 | YoMama | :) |
| 07:42.31 | YoMama | i wanna use x-lite at a cafe with wifi |
| 07:42.36 | Qwell | If a company can make this - http://tinyurl.com/6d7c9 - how can they possibly go wrong with a wireless headset (or any product for that matter)? |
| 07:42.38 | YoMama | but...i feel like a dork wearing a headset |
| 07:43.25 | YoMama | is there anyway to get x-lite workign with a bluetooth headset? |
| 07:43.32 | Qwell | sure, why not? |
| 07:43.32 | benjk | you could get one of those Bluetooth earpieces |
| 07:43.44 | Qwell | a soundcard is a soundcard |
| 07:44.03 | YoMama | i don't know much about bluetooth..so if i have a bluetooth PC card in my laptop...and a bluetooth headset..it'd show up as a sound device? |
| 07:44.38 | benjk | if you don't have Bluetooth built in, just get a Bluetooth USB dongle |
| 07:45.07 | zobia | hello Qwell. beside repeat to dial 10 time to dial one number , is there any sytax of dialplan to dial the same number 10 times at the same time? |
| 07:45.18 | benjk | Don't know on Windoze, but on OSX it shows up as a sound device automatically |
| 07:45.27 | YoMama | benjk: hmm |
| 07:45.32 | YoMama | how much are those dongles? |
| 07:45.53 | benjk | something like 35-45 USD perhaps? |
| 07:46.00 | benjk | froogle is your friend |
| 07:46.09 | YoMama | i'm froogling it now |
| 07:46.13 | YoMama | that'd be badass |
| 07:46.15 | benjk | ;-) |
| 07:46.17 | YoMama | wow..some of them have 100M range |
| 07:46.23 | YoMama | i could talk to someone while in the john |
| 07:46.24 | benjk | yeah |
| 07:47.20 | YoMama | about $30 |
| 07:47.23 | benjk | I have Bluetooth built-in, I should get one of those ear pieces myself |
| 07:48.05 | YoMama | benjk: u in the US? |
| 07:48.15 | benjk | no in Japan |
| 07:48.18 | YoMama | oh |
| 07:48.35 | YoMama | benjk: radio shack has a sale on a plantronics bluetooth headset the day after thanksgiving |
| 07:48.38 | benjk | well, I live here, but I am very often overseas |
| 07:48.40 | YoMama | after rebates..it's $9.99 |
| 07:48.45 | YoMama | can't beat that |
| 07:48.53 | benjk | yeah, sounds like a good deal |
| 07:49.00 | Qwell | for plantronics? yeah, I'd say so |
| 07:49.15 | Himeko | i tried a plantronics one onc, i didn't car for the sound on the other end |
| 07:49.27 | YoMama | well, for $10..i ain't complaining |
| 07:49.31 | Himeko | i guess |
| 07:49.34 | YoMama | they're usually like $50-100 dependin on which one u get |
| 07:49.37 | benjk | The trouble is that many US mailorder companies don't ship overseas |
| 07:49.50 | YoMama | benjk: well, konichiwa |
| 07:49.54 | Himeko | just the ppl listenign to you will be complaining |
| 07:49.56 | benjk | :-) |
| 07:49.57 | YoMama | that and sushi terms are about all i know..haha |
| 07:50.00 | YoMama | omakase |
| 07:50.07 | YoMama | omikase i think |
| 07:50.09 | YoMama | it's spelled |
| 07:50.16 | benjk | no, omakase was right |
| 07:50.22 | YoMama | hehe..thanks |
| 07:51.05 | benjk | did you know that the 'o' in front of those words is an honorary prefix |
| 07:51.21 | benjk | to be more polite |
| 07:51.23 | Himeko | oh well, time for bed |
| 07:51.28 | Himeko | oyasumi |
| 07:51.39 | benjk | oyasumi nasai |
| 07:52.00 | YoMama | i just know that when u go up to the sushi chef |
| 07:52.05 | benjk | and France is the only country that gets the honorary prefix |
| 07:52.11 | YoMama | and say "omakase"..his face lights up, and u get all sorts of special stuff |
| 07:52.12 | benjk | O-Furansu |
| 07:52.28 | YoMama | france? france sucks..haha |
| 07:52.36 | benjk | not here in Japan |
| 07:52.40 | Himeko | they need to feel better about themselves |
| 07:52.55 | YoMama | why the hell would japanese give the french an honorary anything? |
| 07:53.10 | YoMama | have you seen how much a louis vuitton purse costs? |
| 07:53.12 | Himeko | it's a pity honorific |
| 07:53.12 | FuriousGeorge | Himeko: im gonna go tell #ipcop that youre in here :P |
| 07:53.25 | benjk | it's Igirisu for England, Amerika for the US, Roshia for Russia, ... no honorary |
| 07:53.39 | YoMama | not even japan |
| 07:53.39 | benjk | but its O-Furansu for France |
| 07:53.40 | YoMama | Nippon |
| 07:53.48 | benjk | not even for Japan |
| 07:53.52 | YoMama | haha..that blows |
| 07:54.32 | benjk | that would be blasphemy |
| 07:54.32 | YoMama | O anything? |
| 07:54.32 | YoMama | what about Okinawa |
| 07:54.32 | YoMama | you're telling me that's honorary? |
| 07:54.32 | benjk | you cannot use exaltation for yourself |
| 07:54.32 | *** join/#asterisk deadcandance (i=dawndark@athedsl-25209.otenet.gr) |
| 07:54.32 | benjk | only for your counterpart or a third party |
| 07:54.32 | Himeko | not every o is a prefix |
| 07:54.45 | benjk | not every o is, but the o in O-Furansu is |
| 07:55.03 | YoMama | well, u should petition the japanese government to remove that O |
| 07:55.08 | YoMama | the french don't deserve shit |
| 07:55.16 | benjk | Okinawa is Oki and Nawa |
| 07:55.49 | Himeko | ya, my comment was direct toward okinawa |
| 07:56.21 | benjk | and the O in Osaka is Oh for big |
| 07:56.21 | YoMama | like Okidata |
| 07:56.28 | YoMama | hehehe |
| 07:56.34 | benjk | Correct |
| 07:56.49 | benjk | the Oki in Okidata is the same kanji as the Oki in Okinawa |
| 07:56.52 | YoMama | they got the d-link bluetooth usb dongle at office depot for $10 more than mailorder |
| 07:56.59 | YoMama | after shipping tho...might as well get it at the store |
| 07:57.16 | benjk | in Okinawa? |
| 07:57.58 | Himeko | i dun think they have an office depot there |
| 07:58.08 | benjk | too bad |
| 07:58.13 | Himeko | although i have never been there myself |
| 07:58.37 | YoMama | no..in the US |
| 07:58.42 | benjk | Well, they might |
| 07:58.51 | benjk | there is a large US base on Okinawa |
| 07:58.52 | YoMama | best buy has the kensington one |
| 07:58.59 | Himeko | there are at least 6 costco in japan though |
| 07:59.05 | Himeko | and ikea opens 2006 |
| 07:59.22 | benjk | I thought Ikea was already here |
| 07:59.29 | benjk | somewhere in Saitama |
| 07:59.40 | Himeko | they are opening the 1st one in chiba |
| 07:59.43 | Qwell | I have the most massive freaking Ikea by my house. it's like 500 acres and shit, heh |
| 08:00.13 | benjk | Qwell: that must smell terribly |
| 08:00.14 | Himeko | http://www.ikea.co.jp/ |
| 08:00.46 | *** join/#asterisk milkyflava (n=milkyfla@57-48-237-24.gci.net) |
| 08:00.58 | benjk | I don't think they have Bluetooth ear pieces at Ikea though |
| 08:01.09 | Himeko | newp |
| 08:01.14 | Himeko | but might find one at costco |
| 08:01.21 | benjk | at least I would be surprised if they did |
| 08:01.25 | Himeko | or sofmap |
| 08:01.35 | benjk | Japan is always behind in useful technology |
| 08:01.50 | benjk | they are 10 years ahead in toys |
| 08:02.00 | benjk | but 10 years behind in useful stuff |
| 08:02.10 | Himeko | and in innumerable useless variation of the same thing |
| 08:02.16 | benjk | Bluetooth is only now just beginning to show up here |
| 08:02.33 | benjk | I wanted to get a new mobile phone for years |
| 08:02.33 | bbz | Would anyone here be interested in some Polycom 501 phones? We got stuck w/ an extra 20 after a project, and we need to unload them at cost, -- if anyone is interested send me a private message (if i dont respond immediately, leave me your email address, and i can send you info) i have a reference in the channel if you need. -- |
| 08:02.52 | benjk | one with Bluetooth so I could synchronise my addressbook with the phone |
| 08:03.00 | Himeko | nokia 6630 or some such |
| 08:03.16 | benjk | yes, but they only just hit the streets here |
| 08:03.17 | *** join/#asterisk lorinc (n=ang@caracas-2944.adsl.interware.hu) |
| 08:03.23 | Himeko | vodafone has it, wcdma 2100 plus triband gsm |
| 08:03.24 | benjk | and I have to change providers |
| 08:03.34 | benjk | Vodafone are the only ones who carry Nokia |
| 08:03.36 | Himeko | nah, that one has been out for almost a year |
| 08:03.46 | benjk | with lots of bugs though |
| 08:04.12 | benjk | the new firmware has only been released this month, or at least it was due to be |
| 08:04.22 | Himeko | vodafone has at least 8 phones now that are wcdma plus triband gsm |
| 08:04.46 | benjk | Well I'll get the Nokia |
| 08:04.56 | Himeko | 2 mot's too |
| 08:04.56 | benjk | I don't particularly like the shape of it |
| 08:05.11 | benjk | but I absolutely hate the clamshell types |
| 08:05.23 | benjk | and in Japan there is almost nothing but clamshell phones |
| 08:05.36 | Himeko | i have a 3600 i have the keypad auto lock |
| 08:05.46 | Himeko | called 911 by accident twice |
| 08:06.04 | benjk | that wouldnt be a problem over here |
| 08:06.21 | Himeko | well, it was because there was an emergency speed dial |
| 08:06.23 | benjk | 911 does get you a "wrong number" announcement |
| 08:06.28 | *** join/#asterisk fourcheeze (n=rich@westbury.doilywood.org.uk) |
| 08:06.47 | Himeko | not that it was hitting 911 in my pocket |
| 08:06.48 | benjk | ah ok |
| 08:07.09 | Himeko | that would be hard to do twice |
| 08:07.12 | Himeko | once maybe |
| 08:07.20 | benjk | hehe |
| 08:08.03 | Himeko | my wife has decided to go back to japan for 3 months in the spring and go job hunting |
| 08:08.14 | benjk | Do you know if there are any wcdma-voip adapters yet? |
| 08:08.24 | benjk | like those GSM-voip adapters |
| 08:08.35 | Himeko | heh, i haven't even seen the gsm ones |
| 08:08.52 | Himeko | i think wcdma is debuting at 1900 here though |
| 08:08.53 | Himeko | not 2100 |
| 08:09.18 | benjk | they are little boxes with an antenna and a SIM card slot and an ethernet port (for SIP_ |
| 08:09.38 | benjk | or just analog with RJ11 |
| 08:10.13 | benjk | If I could get one of those for wcdma that worked on Vodafone's network here |
| 08:10.30 | benjk | I'd never have to pay any call charges on mobile ever again |
| 08:11.37 | Himeko | how would that work |
| 08:11.39 | benjk | just sign up for two contracts on a family package on the lowest plan with free calls between them, then let Asterisk provide you with the calls you want to make from your mobile |
| 08:11.58 | benjk | you could even initiate this by SMS |
| 08:12.09 | benjk | and thereby use your phonebook |
| 08:12.31 | benjk | using 'send this contact by SMS' |
| 08:12.50 | wasim | :) |
| 08:12.52 | Himeko | i was considing something similar here with my voip provider |
| 08:13.08 | benjk | Asterisk would check your caller ID and parse the SMS, then call you back and bridge you through |
| 08:13.10 | Himeko | cuz there was a plan with unlimiting free incoming calls |
| 08:13.36 | benjk | yep that works too |
| 08:13.41 | Himeko | so if i wanted a dial tone i could send out an sms or email and it would call me back with a dial tone |
| 08:13.45 | benjk | you can do it in the US with GSM |
| 08:13.56 | benjk | because those GSM adapters are available |
| 08:13.57 | *** join/#asterisk implicit (n=implicit@ip70-181-114-97.oc.oc.cox.net) |
| 08:14.07 | Himeko | don't need the adapter |
| 08:14.09 | benjk | but I haven't seen any of those for wcdma yet |
| 08:14.13 | benjk | no? |
| 08:14.32 | Himeko | the deal is all incomign calls are free |
| 08:14.55 | benjk | but how do you get Asterisk to receive SMS? |
| 08:15.18 | Himeko | providers sms to email gateway |
| 08:15.27 | benjk | that's the big question I am asking myself |
| 08:15.48 | benjk | I wouldn't do that |
| 08:16.08 | benjk | insecure channel |
| 08:16.10 | Himeko | but if you just want to recieve sms then you dount need the voip adapter |
| 08:16.25 | benjk | someone might hijack your DIY dial around service |
| 08:16.57 | Himeko | just a gsm card or phone on the computer |
| 08:16.59 | benjk | well, if you always use the same GSM SIM, then you could prevent abuse |
| 08:17.11 | benjk | but I need this for poor man's roaming |
| 08:17.17 | hypa7ia | oh cisco. why must your phones crash on me all the time :s |
| 08:17.24 | Himeko | well, call charges work different there anyway |
| 08:17.29 | Qwell | hypa7ia: using sip? sucker |
| 08:17.30 | *** part/#asterisk deadcandance (i=dawndark@athedsl-25209.otenet.gr) |
| 08:17.34 | benjk | so I would be sending in SMS from a prepaid SIM I just bought some place |
| 08:17.41 | Himeko | your idea is better suited for there |
| 08:17.52 | hypa7ia | Qwell: yup. 45 minutes then *poof* need to reboot if i want to make another call |
| 08:17.55 | benjk | I'd stick in a PIN number into the SMS to let Asterisk know it's me |
| 08:18.13 | Qwell | hypa7ia: yeah, mine too...never seen that with sccp though |
| 08:18.25 | Qwell | I actually like sccp...a lot |
| 08:18.43 | benjk | the data card is very expensive over here, both initial cost and monthly fees |
| 08:18.44 | Himeko | with my method for here it couldn't be hijacked anyway because it would be hardcoded to call my cell |
| 08:18.46 | hypa7ia | hmm. how well are the various ways of doing sccp with asterisk working right now? |
| 08:18.48 | benjk | not worth it |
| 08:19.00 | hypa7ia | i might try loading that :/ |
| 08:19.04 | Qwell | hypa7ia: I use chan_sccp2, it's great |
| 08:19.15 | Qwell | hypa7ia: If you bug me, I'll even let you use my realtime patch for it. ;] |
| 08:19.15 | hypa7ia | cool, i should give it a try |
| 08:19.19 | hypa7ia | oooo |
| 08:19.21 | hypa7ia | :) |
| 08:19.28 | benjk | yes but if you go overseas and buy a prepaid SIM to avoid roaming charges, then your scheme won't work anymore |
| 08:19.29 | Qwell | I'm actually gonna release it when I get time |
| 08:19.47 | Qwell | I asked on the mailing list, if anybody wanted it...nobody really bit |
| 08:20.06 | Himeko | neither will yours :) |
| 08:20.18 | hypa7ia | in a couple of weeks i'm actually going to have time to play with this stuff, too :) |
| 08:20.26 | Qwell | wonder why? ;] |
| 08:20.28 | benjk | No, if I can get Asterisk to receive SMS, my scheme will work |
| 08:20.43 | hypa7ia | hehe. soooooo excited! |
| 08:20.54 | Himeko | i'm sure the family plan wouldn;t allow free international calls |
| 08:21.06 | benjk | perhaps you missed the part where I said I'd stick a PIN number into the SMS to identify myself and let Asterisk know I am on a new number |
| 08:21.39 | Himeko | i mean the free calls part |
| 08:21.49 | hypa7ia | anywhoo, i'm out, nite Qwell |
| 08:21.53 | Qwell | later |
| 08:22.03 | benjk | It's about 20-30 cents per minute to mobile phones overseas |
| 08:22.14 | benjk | But it is about 2 USD per minute on roaming |
| 08:22.28 | Qwell | just get a voip account in the country you're in |
| 08:22.48 | *** join/#asterisk bmg505 (n=leon@rndf-146-56-25.telkomadsl.co.za) |
| 08:22.55 | benjk | yeah, that would work while I am walking around or sit in a cab |
| 08:23.15 | Himeko | i have sjphone on my pda |
| 08:23.43 | Himeko | make calls from any wifi connection i have access to |
| 08:23.50 | benjk | that won;t help you for incoming alls |
| 08:24.00 | *** join/#asterisk pooh_ (n=hfwang@cust.15.241.adsl.cistron.nl) |
| 08:24.09 | Himeko | sure ppl can call my pda |
| 08:24.23 | benjk | I need incoming calls to my office to be forwarded to a *** mobile *** phone overseas |
| 08:25.00 | Himeko | i have mutiple phone ringing on the same number |
| 08:25.08 | Himeko | cell phonme, pda, deskphones |
| 08:25.18 | benjk | I'd have to get a local contract to get data over GSM (most prepaid plans don't offer that) and then I would have to be connected all the time |
| 08:25.40 | Himeko | then just pick up the cheapset ringing phone i can here |
| 08:25.42 | Himeko | hear |
| 08:25.45 | *** join/#asterisk bugant (n=bugant@host139-82.pool80105.interbusiness.it) |
| 08:26.18 | Himeko | gprs is terriblly priced in canada |
| 08:26.36 | benjk | well, the tricky bit is that you don't know in advance what the number will be of the various prepaid SIM cards you are going to purchase during your trip |
| 08:26.37 | Himeko | my cell phone provider is like $50000/gb |
| 08:27.28 | benjk | so you need a means to *** securely *** send Asterisk a message to identify yourself with a new temporary phone number |
| 08:27.58 | Himeko | just ssh in and change the named it calls |
| 08:28.03 | Himeko | er number |
| 08:28.11 | benjk | and ideally without the hassle of trying to finding an internet cafe and log in |
| 08:28.31 | benjk | I want it to be a 3 step process: |
| 08:28.34 | Himeko | or do it from a more expesive call |
| 08:28.36 | benjk | 1) buy SIM card |
| 08:28.49 | benjk | 2) insert SIM card into your phone |
| 08:28.54 | Himeko | i really have to go to bed |
| 08:28.57 | Himeko | 1:30am now |
| 08:29.02 | benjk | 3) send SMS with PIN number to Asterisk and be done |
| 08:29.13 | nextime | 'morning |
| 08:29.26 | benjk | ok, good night then |
| 08:32.04 | *** join/#asterisk axscode (n=paranoid@203.213.217.123) |
| 08:32.10 | axscode | can i put two LANCARDS for asterisk? |
| 08:32.19 | Qwell | sure |
| 08:32.23 | axscode | one is for LOCAL one is for EXTERNAL? |
| 08:32.25 | Qwell | sure |
| 08:32.44 | axscode | ok nice.. thanks |
| 08:32.49 | axscode | no config needed right? |
| 08:32.58 | Qwell | quite a bit of config needed |
| 08:33.31 | *** join/#asterisk rumba (n=ropawa@cpe-68-201-147-176.sw.res.rr.com) |
| 08:35.54 | axscode | o theres a config needed? |
| 08:35.56 | axscode | what would it be? |
| 08:36.17 | zobia | hello if i want to call several call at the same time simultaneously, and if one of them answered. can other one keep dialing still? |
| 08:36.43 | zobia | i use & in the Dial command to dial several phone together |
| 08:39.51 | benjk | zobia: no problem |
| 08:41.13 | *** join/#asterisk mufa (n=test@mx2.finsiel.ro) |
| 08:41.23 | benjk | wasim: are you there? |
| 08:41.26 | zobia | yes |
| 08:41.43 | zobia | i am here. no what i mean is how to make them not stop dialing another one when one is answered |
| 08:42.05 | zobia | benjk. |
| 08:42.42 | zobia | hello benjk |
| 08:42.47 | benjk | zobia: the first one to pick up the phone will get the incoming call, the other phones are not affected by that and can be used to dial out or receive further incoming calls |
| 08:44.18 | jtodd | <PROTECTED> |
| 08:44.19 | zobia | no. actually i am using this to do outbound call. |
| 08:45.11 | benjk | jtodd: hi there, good to see you. yes, I do have experienced CVS going funny many times |
| 08:45.32 | jtodd | Crap. I need to do an update. Both servers are whacked. |
| 08:45.33 | zobia | benjk. my line is like exten => 8893,1,Dial(${TRUNKIN}/123456&${TRUNKIN}/456789,,G(con1^s^1)) |
| 08:45.58 | zobia | so what i want i when i dial 8893 , it will dial 2 number at the same time. |
| 08:46.15 | *** join/#asterisk [Airwolf] (n=airwolf@attilla.nl) |
| 08:46.32 | benjk | so what's the problem? |
| 08:46.47 | zobia | but i want them to be independant call. if 123456 answered , the 456789 need to go on ringing . |
| 08:47.10 | *** join/#asterisk bon (i=bon@localhost.sk) |
| 08:47.17 | zobia | but the result above is if 123456 answered. 456789 will stopped ring and dial |
| 08:47.24 | benjk | why would you want that? |
| 08:47.40 | benjk | yeah that's how it is supposed to be |
| 08:47.54 | zobia | oh. then i think u use the wrong way to do so. |
| 08:48.01 | zobia | actually i want to do a test. |
| 08:48.23 | benjk | why do you want the second phone to continue ringing? |
| 08:48.36 | zobia | i want to issue a dialing dial the same number with 5 lines simultaniously |
| 08:48.54 | zobia | i want to test how many line could answered it. |
| 08:49.07 | benjk | I don't understand |
| 08:49.15 | zobia | like repeat dial the same number 5 times |
| 08:49.24 | zobia | but issue them at the same time. |
| 08:49.39 | zobia | not dial one by one. |
| 08:50.01 | benjk | that's not what & is for |
| 08:50.04 | zobia | the number i dialed has 5 lines to receive the call. |
| 08:50.06 | ptiggerdine | zobia, are you a telemarketer? |
| 08:50.14 | zobia | no.i am not |
| 08:50.39 | ptiggerdine | explain what you'd want to ring 5 ppl at the same time. |
| 08:50.41 | zobia | but it's same like telemarketings |
| 08:51.07 | *** join/#asterisk shadebob (n=shadebob@ll81-144-114-192-81.ll81.iam.net.ma) |
| 08:51.08 | benjk | if you want to initiate 5 calls at about the same time, use a call file and drop it into /var/spool/asterisk |
| 08:51.41 | zobia | okay, thanks. let me read that. |
| 08:51.56 | zobia | thanks ptiggerdine also. |
| 08:51.56 | *** join/#asterisk chapeaurouge (n=chap@85.201.80.249) |
| 08:51.58 | benjk | it seems he wants to test how many simultaneous calls his setup can handle |
| 08:52.10 | zobia | yes |
| 08:52.28 | zobia | u r great |
| 08:52.33 | benjk | the call files are for that |
| 08:52.51 | ptiggerdine | hmm. I akin telemarketers to teleteriorist |
| 08:52.52 | zobia | okay. thanks a lot benjk. |
| 08:52.53 | benjk | alternatively, you can use the Manager API to initiate the calls |
| 08:53.19 | chapeaurouge | sorry, ill jump in benjk. So you would manually create call files, move them to the directory, and watch how * behaves? |
| 08:53.27 | zobia | i read manager api, but it can only initiate 1 by 1 |
| 08:54.06 | benjk | chap: that's how I do load testing, yes |
| 08:54.39 | benjk | but you don't have to create the call files manually, you can write a script to do it for you |
| 08:55.03 | chapeaurouge | benjk, yes of course. thx for that nice piece of knowledge, right after I wake up :-D |
| 08:55.26 | benjk | welcome |
| 08:55.48 | zobia | okay. i will make a agi to write that file |
| 08:56.27 | benjk | One of these days I'll have to make a GUI to generate test calls |
| 08:57.36 | *** join/#asterisk krisguy (n=krisguy@h216-170-039-057.adsl.navix.net) |
| 08:59.42 | benjk | wasim: are you still around? |
| 09:03.29 | *** join/#asterisk psk (n=psk@golia.caltanet.it) |
| 09:11.34 | *** join/#asterisk Gunnar (n=gunnar@bkkb-gw.bitcon.no) |
| 09:13.32 | Qwell | call files can create more than one channel at a time? |
| 09:15.18 | *** join/#asterisk XTR-II (n=xtr@staff-nat.netnation.com) |
| 09:17.24 | *** join/#asterisk grimse (n=grimse@p5481FA5E.dip.t-dialin.net) |
| 09:21.10 | *** join/#asterisk tobiasWolf (n=konversa@195.162.255.10) |
| 09:23.47 | *** join/#asterisk UlbabraB (n=caplaz@host205-72.pool8174.interbusiness.it) |
| 09:24.42 | Qwell | bahahaha |
| 09:24.51 | Qwell | Asterisk: The Future of Telephony - 404 pages |
| 09:29.05 | fourcheeze | Qwell: and this is funny because...? |
| 09:29.31 | Qwell | $10 says it was planned |
| 09:29.39 | fourcheeze | to have exactly 404? |
| 09:29.44 | Qwell | Yes. |
| 09:29.47 | fourcheeze | I would have gone for 407 |
| 09:30.05 | Qwell | I'll ask blitzrage about it tomorrow. heh |
| 09:32.34 | *** join/#asterisk Nivex (i=kjotte@user-0c8hq5r.cable.mindspring.com) |
| 09:32.58 | chapeaurouge | lol |
| 09:33.05 | chapeaurouge | good book btw. finished reading it yesterday |
| 09:33.16 | Qwell | great book |
| 09:33.18 | Qwell | ~thebook |
| 09:33.20 | jbot | well, thebook is Asterisk: The Future of Telephony, released under the Creative Commons license and available at http://www.asteriskdocs.org << Read the book online! |
| 09:33.33 | chapeaurouge | aaah |
| 09:33.38 | chapeaurouge | i didn't know the license terms :) |
| 09:33.42 | chapeaurouge | niceness |
| 09:34.42 | Qwell | bed time |
| 09:35.06 | chapeaurouge | argh! no bookmark in the pdf |
| 09:35.21 | Qwell | print it! :D |
| 09:35.47 | chapeaurouge | hehe :) I also got it off safari.oreilly.com |
| 09:37.25 | ptiggerdine | is mpg123 still the only util that works with asterisk for musiconhold? |
| 09:37.35 | chapeaurouge | ptiggerdine, no |
| 09:37.38 | Qwell | no, you can install asterisk-addons and use format_mp3 |
| 09:37.50 | ptiggerdine | great |
| 09:37.52 | ptiggerdine | thanks |
| 09:38.04 | ptiggerdine | thank god for that. |
| 09:38.10 | chapeaurouge | or you can convert your mp3 and use native |
| 09:39.00 | ptiggerdine | chapeaurouge, whilst that a good idea I'd much rather not at leave them all in the one format. |
| 09:39.24 | chapeaurouge | sure, whatever suits you best :) |
| 09:39.35 | ptiggerdine | thanks chapeaurouge, Qwell |
| 09:39.54 | Qwell | there is a particular format mp3's should follow. keep that in mind |
| 09:41.11 | zobia | hello benjk |
| 09:41.28 | Qwell | bed...for reason this time |
| 09:41.34 | zobia | i put one call file in the spool to dial outgoing call. but could not http://pastebin.ca/30536 |
| 09:41.41 | Qwell | reason? real. see? |
| 09:41.49 | *** join/#asterisk darkskiez (n=darkskie@194.247.78.146) |
| 09:42.06 | *** join/#asterisk kiwnix (n=egarcia@82.158.158.137) |
| 09:43.15 | zobia | the erros is in http://pastebin.ca/30537 |
| 09:47.07 | testmachine | is there some graphical tool |
| 09:47.15 | testmachine | too config sip.conf? |
| 09:47.20 | testmachine | like adding users and so? |
| 09:51.52 | bon | exten => _.3300XXXX,2,Dial(SIP/${EXTEN}) |
| 09:51.54 | bon | is that correct |
| 09:52.04 | bon | anything+3300XXXX notation? |
| 09:53.51 | zobia | hello. can any one help me with my .call file? |
| 10:03.10 | phifli | on 1.2.0 i cant set my outgoing DID cid/ani if i bridge and i was able before on the older versions... trying to set to my toll free num |
| 10:03.24 | phifli | i changed chan_iax2.c and apps_dial.c seemed like its the privacy option |
| 10:03.30 | phifli | or ani/ndis |
| 10:03.34 | phifli | anyone know about it? |
| 10:04.30 | testmachine | wtf |
| 10:04.31 | testmachine | astwin |
| 10:04.36 | testmachine | what is that |
| 10:04.40 | testmachine | asterisk for windows :/ |
| 10:13.46 | *** join/#asterisk pakipenguin (n=pingu@linuxpakistan/admin/pakipenguin) |
| 10:13.49 | pakipenguin | morning |
| 10:17.30 | *** join/#asterisk Zeeek (n=icechat5@pdpc/supporter/active/Zeeek) |
| 10:18.03 | Zeeek | ~seen kram |
| 10:18.06 | jbot | kram is currently on #asterisk. Has said a total of 34 messages. Is idling for 1d 5h 32m 17s |
| 10:20.24 | *** join/#asterisk Mozzie (n=kevinsco@dsl-203-113-194-10.VIC.netspace.net.au) |
| 10:21.13 | *** join/#asterisk vexorg (n=vexorg@CPE000021ded913-CM001225419164.cpe.net.cable.rogers.com) |
| 10:21.29 | *** join/#asterisk jhiver (n=jhiver@AStDenis-105-1-4-4.w193-253.abo.wanadoo.fr) |
| 10:21.43 | Mozzie | Hey gurus - I have a problem with an ABE server and Quadro Boxes by Epygi ... anyone had any luck integrating to these at all ? |
| 10:22.01 | jhiver | hi lads |
| 10:23.11 | jhiver | Can somebody explain to me the difference between "quad span E1" and "quad E1" ? |
| 10:24.18 | benjk | testmachine: It's not exactly Asterisk for Windows. More precisely, it's Asterisk for Cygwin. |
| 10:24.52 | testmachine | ah |
| 10:24.55 | testmachine | benjk: k |
| 10:25.06 | testmachine | benjk: but why would somebody install it on a windows server |
| 10:25.07 | testmachine | hehe |
| 10:25.13 | testmachine | then download the asterisk @home cd |
| 10:25.14 | testmachine | :) |
| 10:25.27 | benjk | I don't think it is intended as a server |
| 10:25.32 | testmachine | it sounds scary |
| 10:25.36 | *** join/#asterisk jcath (n=skycat@159.226.21.127) |
| 10:25.55 | benjk | I think it is for two things .. |
| 10:26.30 | *** join/#asterisk moo[work] (n=moo@82.102.1.42) |
| 10:26.44 | benjk | 1) say you want to run Asterisk on your Windows notebook, Astwin and AsteriskWin32 would be good for that |
| 10:27.08 | moo[work] | hi, which asterisk web conf gui do asterisk admins prefer? which one have the better developing work? |
| 10:27.54 | benjk | 2) say you have a medium sized company with a Windows centric or Windows only IT department and you want to get them warm up to the idea to run an Asterisk server (using Linux or BSD) |
| 10:28.30 | Mozzie | Moo .. i have seen a good java one .. http://www.lanvik-icu.com/asterisk/dialplanner/index.php |
| 10:29.07 | testmachine | hm |
| 10:29.16 | benjk | the Windows techies in the IT department will feel mor comfortable with the idea if you can give them an actual Asterisk sandbox that runs on their Windows desktops so they can play, familiaris and shed off hostility |
| 10:29.20 | testmachine | benjk: then i would say, get vmware and install linux :P |
| 10:29.39 | testmachine | just this asterisk @home cd |
| 10:29.41 | testmachine | is fine |
| 10:29.49 | benjk | the trick is to make it as natural for them as possible |
| 10:30.11 | testmachine | http://www.tomsnetworking.com/Reviews-219-ProdID-AAH-3.php |
| 10:30.16 | testmachine | isnt that natural enough? |
| 10:31.00 | benjk | if they can download an installer that does everything for them without committing the sacriledge of leaving Windows (not yet) then you will make sure they will at least take a peek at it |
| 10:31.10 | testmachine | :) |
| 10:31.18 | Mozzie | Hey testmachine .. y not do a linux - fedora core 2 install, put 1.20 on it.. and get ipswitchboard, ipspeeddial, ipmanager ( that all runs under WinBlows .Net ) |
| 10:31.46 | benjk | in other words, the idea would be akin to a trojan horse type of approach |
| 10:31.47 | testmachine | :) |
| 10:31.49 | benjk | ;-) |
| 10:32.07 | testmachine | Mozzie: i would never try to use windows for asterisk at all |
| 10:32.14 | testmachine | Mozzie: the others here at work would |
| 10:32.14 | benjk | Mozzie: that would be step 2 |
| 10:32.17 | testmachine | but i wont :P |
| 10:32.44 | *** join/#asterisk YDL (n=gefla@i3ED6E257.versanet.de) |
| 10:33.06 | Mozzie | testmachine : u wouldnt be using winblows for the real work ... the gui would be Winblows |
| 10:33.17 | benjk | first you have to get the WIndows fanboys in IT to shed off hostility to something that is linked to Linux |
| 10:33.46 | benjk | YDL: is that YDL as in Yellow Dog ? |
| 10:33.56 | testmachine | but i have easy to talk |
| 10:33.57 | testmachine | like |
| 10:34.00 | Mozzie | testmachine : I have heard from Microsux staff .. that they themselves use Asterisk as their Voice solution !!! |
| 10:34.04 | testmachine | i am unix administrator |
| 10:34.08 | testmachine | almost only *nix |
| 10:34.16 | testmachine | and we have an windows administrator here |
| 10:34.27 | testmachine | so i may do the cool work :P |
| 10:34.44 | testmachine | Mozzie: wtd |
| 10:34.46 | testmachine | LOL |
| 10:36.38 | benjk | tesmachine: if your environment isn't *nix hostile, then you won't have to go through the hoops of making Windows fanboys buy in |
| 10:36.40 | Mozzie | <=== comes from a formal Microsoft background - converted to Linux for Voice ... !!! |
| 10:36.41 | testmachine | Mozzie: but they even used like bsd servers for hotmail too |
| 10:37.03 | *** join/#asterisk kink0 (n=kinko@pluton.interec.com) |
| 10:37.25 | benjk | but that's not always the case, so these installers that make Asterisk run under Cygwin on a Windows box are a welcome tool |
| 10:37.48 | testmachine | toilet time |
| 10:38.28 | *** join/#asterisk chapeaurouge (n=chap@85.201.80.249) |
| 10:38.45 | Mozzie | Asterisk under a windows box is a great tool for showing that it can work as a demonstration purpose only .. I wouldnt use it as a final server engine |
| 10:38.50 | benjk | ~seen wasim |
| 10:38.52 | jbot | wasim is currently on #asterisk (2d 4h 42m 58s). Has said a total of 15 messages. Is idling for 2h 26m 3s |
| 10:39.39 | Mozzie | anyone enountered Epygi Quadro boxes yet ? |
| 10:39.53 | Mozzie | *encountered |
| 10:40.01 | benjk | Mozzie: well, that's what I said |
| 10:40.06 | Mozzie | damn MS keyboard grrr |
| 10:40.56 | benjk | but also for folks who are unfortunate enough to have to use a Windows notebook |
| 10:41.23 | Mozzie | benjk : ummm ??? is that info for me ? |
| 10:42.00 | testmachine | gheghe |
| 10:42.13 | benjk | I run Asterisk on my Powerbook (under OSX) and its great not only for demos but as a SIP/IAX gateway when I am traveling and the environment doesn't let me use SIP |
| 10:42.39 | Mozzie | <--- can see markster and co but doesnt know if they are alive or asleep ? |
| 10:42.40 | bon | benjk: what do you mean? |
| 10:42.46 | bon | doesnt let you use SIP? |
| 10:43.16 | benjk | well, there are countries where SIP simply cannot and will not work. Period. |
| 10:43.46 | bon | like what? |
| 10:43.51 | benjk | and I do not accept any buts or ifs |
| 10:44.00 | benjk | Like most of the Middle East |
| 10:44.05 | benjk | most of Africa |
| 10:44.15 | bon | shouldn't vpn do the job? |
| 10:44.58 | benjk | the most used VoIP service in the Middle East is by a company in Sweden who have developed a protocol called LTP which is conceptually like IAX |
| 10:45.18 | benjk | bon: VPN is often illegal in these places |
| 10:45.27 | Mozzie | exten => s,1,Dial(SIP/exten@ipaddress,60|tr) 4tw |
| 10:45.48 | benjk | and often VPN doesnt work either |
| 10:45.58 | bon | so what do you do then? |
| 10:46.15 | benjk | then again, which providers offer VPN based VoIP gateways? |
| 10:46.24 | benjk | You use LTP or IAX |
| 10:46.46 | bon | nah i was thinking of having some port restrictions or something |
| 10:46.52 | bon | so you vpn to your machine outside |
| 10:46.58 | bon | and from there you are 'free' |
| 10:47.26 | *** join/#asterisk pooh_ (n=hfwang@cust.15.241.adsl.cistron.nl) |
| 10:47.43 | benjk | I have been using IAX with some parameter tuning and the ILBC codec from places in Africa where the dialup line was so rotten (the copper was literally rotten) that even web browsing and email with attachments was a serious challenge |
| 10:47.50 | Mozzie | http://dns2go.com/ |
| 10:48.20 | moo[work] | anyone have digicom 3200 cards overthere? |
| 10:48.35 | benjk | bon: if you have never been there and tried to make things work you really shouldn't give advice about something you have not the faintest clue about |
| 10:48.58 | bon | i am not giving any advice, you are the one who experienced that so i am just curious and asking :) |
| 10:49.09 | Mozzie | hey benjk : have you integrated Quadro boxes with asterisk servers yet ? |
| 10:49.14 | bbz | Would anyone here be interested in some Polycom 501 phones? We got stuck w/ an extra 20 after a project, and we need to unload them at cost, -- if anyone is interested send me a private message (if i dont respond immediately, leave me your email address, and i can send you info) i have a reference in the channel if you need. -- |
| 10:49.32 | benjk | Not even Cisco was able to get Voip service working for a major organisation in the Arab world when I walked in there with an Asterisk box and IAX did the trick |
| 10:49.50 | testmachine | hm |
| 10:50.00 | testmachine | asterisk @home completed |
| 10:50.10 | testmachine | i wil test that |
| 10:50.11 | testmachine | :) |
| 10:50.35 | Mozzie | Hey benjk : was that because of the codec support in the country ? |
| 10:50.38 | benjk | the Cisco guys were shattered |
| 10:51.19 | benjk | it was due to the most horrible conditions you would not be able to dream up even if you are on crack |
| 10:51.44 | benjk | IP routes that change from one second to the next |
| 10:52.04 | Mozzie | For everyones consumption : Quadro release version 3.0 codec 726/32 is fubar and will not work with Cisco / Asterisk 726/32 because of differing specs.. also the ILBC version the Quadros provide is not quite right .. |
| 10:52.09 | benjk | you do a traceroute there and never ever get the same output twice |
| 10:53.26 | benjk | The ILBC codec I used was because I had a dialup link with rotten copper that would only deliver 18 kbps, that was in the office I used there |
| 10:53.33 | YDL | benjk: no, like in YveDeLarc |
| 10:53.47 | benjk | but even on broadband there were serious issues |
| 10:53.47 | kink0 | off-topic: is possible to swap SIM configuration from a large SIMM ussing AT commands ? |
| 10:54.52 | Mozzie | SNOM 360 related : anyone found these telephones try to do a registration all the time ? |
| 10:55.10 | benjk | when I looked at the connection these guys have over there I thought "Oh no, there is no way I get a stable IAX connection working" |
| 10:55.23 | Mozzie | Asterisk 4tw |
| 10:55.35 | benjk | but every time I thought "this is it, this won't work" IAX suprised me |
| 10:56.19 | Mozzie | well benjk : if you get delays that are constant that is better than having delays that are spiked like a varying dc waveform |
| 10:56.50 | benjk | and I have exchanged experiences with other Asterisk users from South Africa who have been doing work all over Africa, and they had the same experiences |
| 10:57.07 | Mozzie | I have heard of places in the pacific .. that have constant 300ms delays that mean it is possible to still use voip |
| 10:57.13 | *** join/#asterisk santoshr (i=1063@203.199.110.93) |
| 10:57.45 | santoshr | >>>>>>chan_sip.c:1064 __sip_xmit: sip_xmit of 0x81952c0 (len 364) to 192.168.1.196:-1 returned 0: Invalid argument<<<<<<<<< i keep getting this whenever a exten is dialed |
| 10:57.47 | benjk | So long story, short message: I have seen environments where SIP simply didn't work but IAX worked flawlessly |
| 10:58.47 | benjk | and when I say SIP didn't work, I don't mean "I couldn't make it work" but in many places Cisco and Avaya were bidding foer the same contracts and their specialists would not make SIP work |
| 10:58.53 | Mozzie | santoshr : reload your dial plan and look at when you get an invalid argument on the reload .. check ur dial plan |
| 10:59.24 | Mozzie | look at the spot .. and it maybe there is a ; or something not right |
| 10:59.24 | bon | benjk: so you're special .) |
| 10:59.48 | santoshr | http://sial.org/pbot/14495 |
| 10:59.56 | *** join/#asterisk Paramount` (n=Paramoun@ip70-186-17-123.ma.dl.cox.net) |
| 11:02.23 | benjk | bon: I am not. That's the point. With IAX it's a walk in the park. |
| 11:02.39 | Paramount` | can onyone helpe me diagnose why I am not able to take call-wating on my sipura 2002? |
| 11:02.44 | bon | well anyway, nice speaking to you, gotta go, hope to see you soon |
| 11:02.47 | Paramount` | works on a software sip |
| 11:02.51 | Paramount` | not on this phone |
| 11:05.26 | Mozzie | have you tried to debug on that telephones ip ? |
| 11:05.35 | santoshr | ya |
| 11:05.48 | santoshr | wait i will paste everything looks normal |
| 11:05.54 | Mozzie | kewl |
| 11:06.40 | Mozzie | santoshr .... where is the problem occuring when u reload |
| 11:07.02 | *** join/#asterisk lorinc (n=ang@caracas-0780.adsl.interware.hu) |
| 11:07.24 | Paramount` | well Ill be on another line, and someone else tryies calling in, and rather than getting a call-waying beep, in order to 3 way, the go straight to my voicemail |
| 11:07.27 | Paramount` | :\ |
| 11:08.02 | *** join/#asterisk zotz (n=zotz@24.231.47.168) |
| 11:08.22 | Mozzie | hmm whats the dial plan say ? |
| 11:08.26 | santoshr | i am nt able to see any warning whn i reload |
| 11:08.34 | Mozzie | rgr that santos |
| 11:08.44 | santoshr | rgr ? |
| 11:08.51 | Mozzie | ok |
| 11:09.04 | Mozzie | sorry .. using short cut speech |
| 11:09.21 | santoshr | what do u mean by rgr ? |
| 11:09.35 | Mozzie | rgr = roger ( airforce term for okay ) |
| 11:10.01 | santoshr | i know wht roger is captain! |
| 11:10.36 | santoshr | so i was telling tht i am not getting any indications of error when i do a reload |
| 11:10.59 | Mozzie | ok .. but what do you dial to bring up the error ? |
| 11:11.24 | santoshr | dial a extension from a fxs box |
| 11:11.48 | santoshr | Mozzie: http://sial.org/pbot/14495 |
| 11:13.15 | *** join/#asterisk demetrio (n=qershor@host108-146.pool872.interbusiness.it) |
| 11:13.18 | Mozzie | santos : so u have FXS ports ? |
| 11:13.32 | santoshr | yes on both the sides.. |
| 11:13.46 | Mozzie | ok .. is it a TDM card that you are using |
| 11:13.53 | santoshr | if u notice tht pastling .. 63 --> 62 and |
| 11:13.59 | santoshr | no its a 1fxs box |
| 11:14.20 | Mozzie | does it use zaptel drivers ? |
| 11:15.00 | santoshr | no |
| 11:15.01 | santoshr | SIP |
| 11:15.58 | demetrio | when I start asterisk it registers succesfully to several accounts. however after a while (hours) it will unregister from one of them, and subsequent registration attempts will timeout. if I change public IP (restart router) it will register again, and again wait for a while before giving the same problem. any suggestions? |
| 11:16.04 | Mozzie | ok .. |
| 11:16.30 | Mozzie | canreinvite=yes |
| 11:17.06 | Mozzie | type=friend |
| 11:17.52 | demetrio | Mozzie, doesn't work |
| 11:18.23 | Mozzie | what should it be then dem |
| 11:19.24 | demetrio | I don't have a clue. If I try to register directly from a sip phone it works, even while asterisk doesn't (when the IP has already "expired") |
| 11:19.38 | moo[work] | what managing tools do you use arround here? |
| 11:19.41 | benjk | demetrio: do you have qualify=yes ? |
| 11:20.07 | santoshr | any clue abt this people <<<<chan_sip.c:1064 __sip_xmit: sip_xmit of 0x8143fd0 (len 742) to 192.168.1.196:-1 returned 0: Invalid argument<<<<< |
| 11:20.13 | Mozzie | umm santos .. that info i was posting was for you mate |
| 11:20.21 | santoshr | ohh |
| 11:20.25 | demetrio | benjk, now I do, and it still doesn't work |
| 11:20.29 | santoshr | i thought tht was fr demetrio |
| 11:20.35 | *** join/#asterisk Faithful (n=Faithful@202-6-145-116.ip.adam.com.au) [NETSPLIT VICTIM] |
| 11:20.37 | Mozzie | no mate for u |
| 11:20.37 | santoshr | i just tried those two options |
| 11:20.39 | santoshr | still the same |
| 11:20.39 | Paramount | anyone have any ideas for me? I recently used stanaphone, and finally got a sipura 2002 and got my landline workon it... but it dosne't accept call-wating calls.. I just goes straight to my voicemail |
| 11:20.48 | benjk | demetrio: may be related to your router |
| 11:21.09 | demetrio | benjk, how can that be if the sip phone is behind the same router and it does work? |
| 11:21.28 | Paramount | everything worked fine while using stanaphone |
| 11:21.40 | Mozzie | paramount u got exten => number,1,Dial(SIP/EXTEN@ipaddy,60|tr) |
| 11:21.57 | Paramount | in extensions.conf? |
| 11:22.08 | *** join/#asterisk juice (n=juice@mo-69-69-117-72.sta.sprint-hsd.net) [NETSPLIT VICTIM] |
| 11:22.12 | *** join/#asterisk luke-jr_ (n=luke-jr@CPE-24-31-244-49.kc.res.rr.com) [NETSPLIT VICTIM] |
| 11:22.32 | benjk | demetrio: which device is losing its registration where? |
| 11:22.38 | *** join/#asterisk Igbothom_III (n=HiltonT@static-84.217.240.220.dsl.comindico.com.au) [NETSPLIT VICTIM] |
| 11:22.38 | *** join/#asterisk asteriskgeeks (n=SIPdawg@pbxtech.com) [NETSPLIT VICTIM] |
| 11:23.13 | Paramount | I got exten => 100,1,Dial(SIP/nickg) |
| 11:23.25 | demetrio | benjk, the asterisk box is losing its registration to a certain sip account, while others work with no problem whatsoever. |
| 11:23.55 | benjk | ok, but where is that account? |
| 11:24.20 | demetrio | it's an italian voip provider's, called digitaleitalia if you happen to know it |
| 11:24.39 | *** join/#asterisk stkn (i=nobody@gentoo/developer/pdpc.active.stkn) |
| 11:25.00 | benjk | so the sip proxy you are registering with is outside of your NAT, right? |
| 11:25.01 | santoshr | Mozzie: |
| 11:25.19 | Mozzie | paramount: why not after SIP/nickg put a ,time in digits ie 20|tr) |
| 11:25.38 | demetrio | right, but I have set up asterisk with port forwarding and everything |
| 11:25.39 | Mozzie | santos .. ok... |
| 11:25.43 | Paramount | whats the value add, mozzie? |
| 11:25.57 | Paramount | will that fix my call waiting issue? |
| 11:26.01 | Mozzie | paramount .. ie time to wait before going to step 2 |
| 11:26.20 | benjk | demetrio, that's why I am saying the issue may be related to your NAT router |
| 11:26.28 | Mozzie | step 2 could be .. exten => number,2,Congestion |
| 11:26.45 | Mozzie | paramount exten => number,102,Busy |
| 11:26.48 | benjk | I had lots of weirdo things until I started to use proper routing equipment |
| 11:27.08 | demetrio | benjk, how... ? my router's a zyxel, shouldn't be crap |
| 11:27.24 | benjk | especially with Linksys NAT routers I had very bad things happening all the time |
| 11:27.35 | Mozzie | nat forwarding help found here : http://www.asteriskguru.com/tutorials/ in teh Troubleshooting help |
| 11:27.52 | benjk | demetrio: not saying that it is. just that it *may* |
| 11:27.55 | santoshr | Mozzie: Executing Dial("SIP/192.168.1.192-0818af18", "SIP/62|100|Tt") in new stack after this i start getting tht error |
| 11:28.51 | Mozzie | rgr that... |
| 11:29.02 | *** join/#asterisk MuppetMaster (n=MuppetMa@62.37.169.64) |
| 11:29.05 | MuppetMaster | Hello |
| 11:29.09 | demetrio | benjk, I get the point, but I think it would be very improbable, given that other accounts work fine. right? |
| 11:29.18 | MuppetMaster | Anyone heard a rumor that Nortel is suing Digium about IP issues with Asterisk? |
| 11:29.38 | *** join/#asterisk lilo (i=levin@freenode/staff/pdpc.levin) |
| 11:30.17 | benjk | demetrio: that's what I meant when I said "weirdo things" |
| 11:30.46 | Mozzie | santos = try exten => 62,1,Dial(SIP/192.168.1.196,30|tr) |
| 11:31.05 | Mozzie | santos exten => 62,2,Congestion |
| 11:31.19 | benjk | but since you are not using a Linksys worst-routers-in-the-universe router, my hunch may well be wrong |
| 11:31.22 | Mozzie | santos exten => 62,102,Busy |
| 11:31.39 | demetrio | ok, then I'll try talking to my god of choice |
| 11:32.15 | *** join/#asterisk frenzy (n=frenzy@193.220.82.108) |
| 11:32.18 | *** join/#asterisk pa (n=Paolo@unaffiliated/pa) |
| 11:32.20 | Mozzie | Santos do the same for your 63 |
| 11:32.37 | benjk | Muppet: and what IP issues would that be? |
| 11:32.56 | frenzy | I dont know if there a product out there for this |
| 11:33.06 | Mozzie | Intelectual Property issues ? |
| 11:33.07 | frenzy | but has anyone heard of a rate engine? |
| 11:33.22 | benjk | I know what IP in this context means |
| 11:33.37 | benjk | I meant what could Nortel have to sue Digium for? |
| 11:34.25 | benjk | The only think I could imagine Nortel might have a case is Digium's choice of name for the Voicemail system |
| 11:34.31 | benjk | Comdian Mail |
| 11:34.39 | frenzy | Am looking for an applications that would help me manipulate rates from various providers and to create standard rates sheet ? |
| 11:34.41 | Mozzie | isnt it comedian mail |
| 11:34.41 | benjk | er Comedian Mail |
| 11:34.54 | MuppetMaster | benjk: I do not know, just heard the rumors. |
| 11:35.06 | Mozzie | well then they should take comedians to task... as in clowns for the use of the name |
| 11:35.23 | benjk | And since they use the same narrator, it just sounds like your'e connected to a Nortel box and you just heard "Meridian Mail" ;-) |
| 11:35.23 | MuppetMaster | And mimes |
| 11:35.38 | benjk | Muppte: that's not the point |
| 11:35.38 | Zeeek | it's Cameleon MAil |
| 11:35.51 | Mozzie | No she sounds tooo damn sweet to say Meridian Mail... |
| 11:35.55 | benjk | the legal term is "passing off" |
| 11:36.09 | Mozzie | even with my aussie twang .. i hear her good n proper : ) |
| 11:36.57 | benjk | if Nortel can show that Digium did deliberately try to make this appear as close as possible to "Meridan Mail" in order to give callers the impression that it is a Nortel box, then they will have a case against Digium for passing off |
| 11:37.22 | frenzy | has anyone heard of a calling card software ? |
| 11:37.27 | Mozzie | yes |
| 11:37.29 | frenzy | calling rates *** |
| 11:37.37 | benjk | frenzy: astcc |
| 11:37.48 | frenzy | no I meant calling rates |
| 11:38.01 | frenzy | I have messy files with list of terminations |
| 11:38.12 | frenzy | I need to make a standarzied file |
| 11:38.17 | frenzy | so I can have proper LCR |
| 11:38.19 | *** part/#asterisk MuppetMaster (n=MuppetMa@62.37.169.64) |
| 11:38.39 | benjk | Of course Digium would say that the choice of name "Comedian Mail" was just -cough- conincidental |
| 11:38.44 | *** join/#asterisk __a (n=a@85.100.123.10) |
| 11:39.04 | Mozzie | She can say it however she wants /... |
| 11:39.13 | benjk | but whther the court would be inclined to believe this is a different matter |
| 11:39.21 | Mozzie | Now if i could get her to say ... Mozzzie |
| 11:39.44 | benjk | Mozzie: YOu can! For 10 bucks you get 20 words |
| 11:39.54 | benjk | or was it 11 bucks? |
| 11:40.04 | h3x | you are on crack |
| 11:40.09 | Mozzie | awesome ... i dont mind... I have heard she is a hotty .. |
| 11:40.19 | h3x | nortel, nextel, and sprint are suing everybody for turning voice into ip packets |
| 11:40.36 | h3x | probably not even nortel |
| 11:40.39 | benjk | So Digium might then be forced to settle and change the name "Comedian Mail" to something else |
| 11:40.41 | h3x | i think its just sprint/nextel |
| 11:40.51 | benjk | like for eaxmple "Asterisk Mail" |
| 11:40.52 | __a | guys, I'd like to make an FXO backup line for asterisk, so that in case VoIP fails it'll dial some predefined number where it will enter a pin number and dial the original number that the user dialed. Is it possible out of the box with Asterisk or do I need to do custom programming? |
| 11:41.05 | benjk | that would hardly be a loss for Digium |
| 11:41.26 | benjk | in fact it would be an opportunity to get publicity |
| 11:42.08 | Mozzie | <_a> use exten => number,1,Dial(SIP/EXTEN@IPADDY,30|tr) |
| 11:42.10 | benjk | after all, if Nortel sues Digium that means Asterisk has become a threat to Nortel's PBX business and that is an endorsement in favour of Asterisk |
| 11:42.33 | Mozzie | <_a> exten => number,1,Dial(ZAP/EXTEN@IPADDY,30|tr) |
| 11:42.46 | Mozzie | err that should have read number,2 |
| 11:42.53 | Mozzie | that is your fail over |
| 11:43.10 | __a | OMG really? |
| 11:43.21 | Mozzie | whenever the call cant route via the first route it will automatically use the second route |
| 11:43.59 | Mozzie | well my second context statement aint right but thats the sorta thing you need |
| 11:45.07 | Mozzie | it uses number,1,Dial and then the number,2,Dial statement straight out of the box and if u had a thrid route you could enter that the same way |
| 11:45.11 | *** join/#asterisk enemy^x (n=null@enemy.vdsl.no) |
| 11:45.49 | __a | Mozzie: but in zap case, there's an IVR which asks for a number to dial |
| 11:45.55 | enemy^x | Can anyone help me with zapata.conf ? Im trying to make my nt on a quad bri card to call directly out on the te`s. I only get "Extension 's' in context 'isdnnt' from '' does not exist. Rejecting call on channel 0/2, span 6" .... |
| 11:46.31 | __a | Mozzie: does above handle this case as well? |
| 11:46.40 | benjk | enemy: how about creating an extensions 's' in your context then? |
| 11:46.57 | Mozzie | so the ivr reqests u to enter a password ? |
| 11:47.20 | __a | Mozzie: yep |
| 11:47.41 | Mozzie | <_a> do u need the ivr to use that route or can you not just create another context to use the zap channels ? |
| 11:48.30 | __a | Mozzie: but I'd like to solve the easier problem first, i.e. it will prompt me for a number to dial which I can bypass by waiting for some seconds and then dialing the number in $EXTEN |
| 11:48.54 | Mozzie | I guess it really depends on how hard you want to make the failover |
| 11:50.03 | __a | well, the problem is this: you know calling cards? I'd like asterisk to dial a calling card number and then dial the number the user entered, in case dialing out via SIP fails. |
| 11:50.36 | demetrio | uhm ok, I just found out that if I enable SIP ALG in my router & disable externip in sip.conf it works, only other accounts do no more |
| 11:51.13 | *** join/#asterisk future (n=future@virtual.phuture.sk) |
| 11:51.48 | jcath | hello, anyone use fwd with linphone behind a firewall? |
| 11:51.52 | jcath | how to set the parameters? |
| 11:51.57 | Mozzie | <_a> Sounds like you got your work cut out for you ... that is also a topic i have to do some research into ... |
| 11:52.39 | Mozzie | <_a> http://www.asteriskguru.com/tutorials/ might be of some assistance ... |
| 11:53.12 | *** join/#asterisk ful|work (n=fulgas@209.8.233.68) |
| 11:53.21 | __a | Mozzie: I guess I need to modify the dial app so that after establishing a channel it waits for some seconds and then sends the exten number again... |
| 11:53.42 | benjk | jcath: use IAX then |
| 11:53.46 | santoshr | Mozzie: after making the change s tht u mentioned i am getting the following ... a single ring and this SIP/192.168.1.196-93a9 is busy >>> xecuting Congestion("SIP/192.168.1.192-0818b068", "") |
| 11:54.30 | Mozzie | <_a> If the person already authenticates in the first place it should automatically parse the info to the route without having to re-entetr it |
| 11:55.06 | __a | Mozzie: yep, that's why i'm not doing authentication second time. I'm just sending the number the person dialed originally. |
| 11:56.01 | *** part/#asterisk Paramount (n=Paramoun@ip70-186-17-123.ma.dl.cox.net) |
| 11:56.07 | benjk | demetrio: see, I was right then, it is related to your router |
| 11:56.09 | santoshr | Mozzie: dude u aroung |
| 11:56.36 | Mozzie | santoshr .. try 62,1,Dial(SIP/192.168.1.196,60|tr) |
| 11:56.50 | Mozzie | try increasing the timer after the ip |
| 11:56.52 | demetrio | benjk, looks like it. |
| 11:57.13 | demetrio | benjk, what routers do you use? |
| 11:57.17 | benjk | demetrio: I think I know what went wrong there |
| 11:57.26 | benjk | I have seen this before |
| 11:57.44 | benjk | depending on the router used |
| 11:58.10 | demetrio | hmm, what went wrong then? and, most important: how do I solve this? :) |
| 11:58.29 | santoshr | the same a ring and then it drops |
| 11:58.37 | santoshr | executing Congestion |
| 11:58.41 | demetrio | maybe I already did |
| 12:00.11 | demetrio | what I can't get is this: SIP ALG shold be content-munging, in the sense that the router peeks inside SIP headers and translates IP addresses. shouldn't the externip=etc directive do the same? why does one method work and the other doesn't? |
| 12:00.38 | benjk | I don't remember in great detail but I seem to remember that it has to do with the fact that outbound requests to a remote sip proxy sometimes get mapped on the NAT router and sometimes they don't |
| 12:00.50 | benjk | well with some routers |
| 12:00.58 | benjk | or the mapping may change |
| 12:01.00 | Mozzie | Santos , error on my part mate ... exten => 62,1,Dial(SIP/EXTEN@192.168.1.196,60|tr) |
| 12:01.00 | Mozzie | exten => 2611,1,Dial(SIP/2611,20|tr) |
| 12:01.00 | Mozzie | exten => 2611,2,Voicemail(u2610) |
| 12:01.00 | Mozzie | exten => 2611,3,Hangup |
| 12:01.19 | *** join/#asterisk Pazzo (n=Pazzo@host130-250.pool8172.interbusiness.it) |
| 12:01.19 | Mozzie | santos replace 2611 with 62 |
| 12:01.28 | Mozzie | apologies mate |
| 12:02.22 | santoshr | not a problem |
| 12:02.25 | santoshr | wait me trying out |
| 12:03.11 | benjk | for some reason, when you have port forwarding enabled together with outbound registers (which get mapped) then the responses from the SIP proxy may not come back in on the mapped port but they come in via portforwarding |
| 12:03.17 | Mozzie | <-- was thinking more sip @ ip style .. when you are on the same subnet |
| 12:03.27 | benjk | for some reason Asterisk seems to have trouble when that happens |
| 12:03.41 | Mozzie | benjk : have you had problems with DMZ and firewalls ? |
| 12:03.59 | Mozzie | benjk one way audio ? |
| 12:04.10 | benjk | Mozzie: I did |
| 12:04.18 | Mozzie | benjk : howd ya fix it |
| 12:04.24 | benjk | no, one way audio is easy |
| 12:04.36 | demetrio | benjk, that may be solved if asterisk could use different ports for inbound & outbound connections |
| 12:04.47 | Mozzie | benjk : ok how bro .. |
| 12:04.50 | benjk | its more cumbersome when you have issues wuch as demetrio has |
| 12:05.27 | Mozzie | benjk: i have a system that cant receive audio from *COUGH* other systems |
| 12:05.38 | benjk | Mozzie: most of the time qualify=yes and canreinvite=no will resolve the no audio problem |
| 12:05.39 | Mozzie | benjk: Quadro |
| 12:06.01 | demetrio | anyway, I managed to solve the problem turning on the SIP ALG feature my router has, but this way I had to disable port forwarding on 5060 otherwise packets will get lost in outer space. this isn't much of a problem for me right now, but this way my asterisk box cannot accept registrations from users oustiside my NAT. |
| 12:06.05 | Mozzie | benjk: if its a canreinvitre=no then im sorted |
| 12:07.32 | benjk | However, I have to say though, I would not ever advise anybody to try to defeat SIP's BAD and run servers behind NAT |
| 12:07.54 | benjk | SIP simply isn't designed for that. Period. |
| 12:08.17 | demetrio | benjk, I will follow your advice when ipv6 comes :) |
| 12:08.20 | Mozzie | benjk : what about qualify = 3000 ? |
| 12:08.26 | benjk | Yes, there are workarounds, all of which are terribly bad hacks, but a proper solution it is not |
| 12:08.28 | Druken | morning everyone... |
| 12:09.21 | benjk | demetrio: benjk's law of SIP and NAT goes like this ... |
| 12:09.36 | benjk | 1) if you must use SIP, don't use NAT |
| 12:09.46 | h3x | benjk: it looks kinda like a gay porno |
| 12:10.02 | h3x | sip works with nat |
| 12:10.04 | benjk | 2) if you must use NAT, use tunneling |
| 12:10.04 | h3x | it just dosent like double nat |
| 12:10.22 | benjk | 3) if you can, use IAX |
| 12:10.23 | santoshr | Mozzie: i can head it ringing the extension but it is not ringing actualyl. and also i am getting the same erro on the console <<<<chan_sip.c:1064 __sip_xmit: sip_xmit of 0x8195bf8 (len 742) to 192.168.1.196:-1 returned 0: Invalid argument>>>> and then it went into voice mail from tht ,2, |
| 12:10.26 | Druken | h3x: why do you know what a gay porno looks like? |
| 12:10.28 | demetrio | yaknow, not everyone has a class C subnet to play with |
| 12:10.32 | h3x | Er um no |
| 12:10.33 | santoshr | head=har |
| 12:10.36 | santoshr | hear sorry |
| 12:11.00 | Druken | demetrio: all it takes is 1 ip.... |
| 12:11.11 | benjk | h3x: SIP doesn't work with NAT - it can be raped to do NAT in some cases |
| 12:11.25 | benjk | demetrio: there is IAX |
| 12:11.35 | Mozzie | extensions.conf |
| 12:11.35 | Mozzie | exten => 62,1,Dial(SIP/62,100,Tt) |
| 12:11.35 | Mozzie | exten => 63,1,Dial(SIP/63,100,Tt) |
| 12:11.35 | benjk | and there is IPsec |
| 12:11.37 | Druken | if you are going to run a server, that is accessed from the world wide interweb, DON'T HAVE IT BEHIND NAT!!! |
| 12:11.55 | benjk | Druken: A M E N ! ! ! |
| 12:12.26 | demetrio | Druken, all I have is 1 ip... and 5 boxes that need internet access. I'm not an enterprise, I'm just somebody who likes to play. |
| 12:12.45 | santoshr | just tried it.. |
| 12:12.49 | benjk | demetrio: get yourself an old Pentium II and a CompactFlash adapter |
| 12:12.59 | santoshr | the same "sip_xmit of 0x81956f8 (len 742) to 192.168.1.196:-1 returned 0: Invalid argument" |
| 12:12.59 | Druken | demetrio: so get yourself a business class connection with a block of 8 ip's |
| 12:13.02 | benjk | then download Wolverine |
| 12:13.06 | Druken | that'll give ya 5 useable ip's |
| 12:13.22 | Mozzie | Santoshr: hmm there is another way bro ... give me 5 |
| 12:13.29 | santoshr | ya sure |
| 12:13.33 | Druken | shit, i have a block of 32 ip's... i pay.. $200 a month... |
| 12:13.40 | demetrio | Druken, I'm a student, not a rich bastard that can afford 8 ips :D |
| 12:13.40 | santoshr | sorr rgr tht ..;) |
| 12:14.00 | benjk | http://www.coyotelinux.com/products.php?Product=wolverine |
| 12:14.17 | Druken | demetrio: i'm not a rich bastard either... i just know if i want to play... i need ip's |
| 12:14.45 | benjk | then get two additional NICs (RTL8139 at 5 USD a piece will do) for your DIY Wolverine box |
| 12:15.06 | benjk | then put your Asterisk server into the WOlverine's HARDWARE DMZ |
| 12:15.10 | h3x | fuck that |
| 12:15.13 | h3x | use openbsd pf |
| 12:15.16 | h3x | with bridging |
| 12:15.31 | benjk | that will get your Asterisk a global IP address |
| 12:15.42 | benjk | and you still have a NAT for your other gears |
| 12:15.45 | benjk | er gear |
| 12:16.18 | benjk | h3x: sure BSD will work too but you have to know how to configure it |
| 12:16.47 | benjk | Wolverine on the other hand can be configured by just about everybody |
| 12:16.49 | h3x | if you dont know how to configure wolverine |
| 12:16.51 | h3x | you are fucked too |
| 12:16.52 | h3x | heh |
| 12:16.54 | Druken | hey benjk, any idea how i give my customers a proper busy signal when it's busy?? and a congestion when i run out of channels |
| 12:17.02 | santoshr | ya Mozzie can u please visit this page.. http://sial.org/pbot/14496. this has the output of sip debug |
| 12:17.31 | Druken | ~pastebin |
| 12:17.34 | jbot | from memory, pastebin is a place to paste your stuff without flooding the channel - try http://pastebin.ca/ |
| 12:17.39 | benjk | I lean myself out of the window as far as to say that there isn't any networking gear on the planet that's as easy to setup as Wolverine is |
| 12:17.40 | demetrio | yes I thought about that, but I'm happy with bad-looking solutions as long as they work |
| 12:18.28 | benjk | Druken: you can customise those |
| 12:18.36 | benjk | indications.c IIRC |
| 12:18.42 | benjk | or .h |
| 12:19.12 | Druken | benjk: no this is a dialplan thing... either congestion or busy i know that much.... |
| 12:19.25 | benjk | demetrio: that's precisely why you are having thes problems |
| 12:19.51 | benjk | get a proper setup and solve it once and for all |
| 12:19.51 | Mozzie | santos : 62 = 192.168.1.174 ip |
| 12:19.51 | *** join/#asterisk Abbas (n=Abbas@203.81.194.242) |
| 12:20.25 | santoshr | no 174 is the ip which has asterisk runnign |
| 12:20.27 | Druken | ugly setups are just that... ugly... |
| 12:20.38 | benjk | Druken, the trouble is I can't tell the difference between Congestion and Busy |
| 12:20.46 | Mozzie | ahh rgr that ... 174 .. |
| 12:20.52 | Druken | congestion is a faster busy signal |
| 12:21.05 | benjk | so, I reckon you have to customise the underlying tone |
| 12:21.14 | benjk | not on Asterisk |
| 12:21.18 | Druken | yes on asterisk |
| 12:21.23 | Druken | :) |
| 12:21.45 | benjk | the tones Asterisk produces when you say Congestion and when you say Busy are the same |
| 12:21.52 | benjk | at least to my ears |
| 12:21.59 | benjk | but then again, I may be deaf |
| 12:22.41 | santoshr | can this be the issue <<<<<<<<Found no matching peer or user for '192.168.1.192:5060'>>>>>>>> |
| 12:22.43 | Druken | well, let me see if i notice a diffrence |
| 12:22.44 | *** join/#asterisk ptiggerdine (n=ptiggerd@c210-49-98-194.rochd1.qld.optusnet.com.au) |
| 12:22.47 | *** join/#asterisk BhaalWK (i=bhaal@freenode/staff/bhaal) |
| 12:23.41 | Druken | oh yeah... huge diffrence |
| 12:24.16 | benjk | since when? |
| 12:24.35 | Druken | well, i just dialed a busy extension on my phone... |
| 12:25.19 | benjk | so what's your problem then? I thought you wanted a proper busy and fast busy |
| 12:25.25 | benjk | appearently you have it |
| 12:25.56 | Druken | no... i made an extension that goes to a busy... |
| 12:25.58 | Mozzie | yep... |
| 12:26.01 | Druken | strickly to compare the two |
| 12:26.17 | Mozzie | santoshr : what version of card is it |
| 12:26.53 | Druken | benjk: shitty thing is... where do you find a busy line to test it... |
| 12:26.53 | benjk | well you can try to use exten => foo,1,Busy and exten => bar,1.Congestion |
| 12:27.22 | benjk | er , not . before Congestion |
| 12:28.09 | *** join/#asterisk puzzled (n=patrick@puzzled.xs4all.nl) |
| 12:28.13 | puzzled | morning |
| 12:28.32 | *** join/#asterisk oej (n=Olle@apollo.webway.se) |
| 12:28.41 | benjk | hi oej |
| 12:28.56 | oej | Hi Ben! |
| 12:29.27 | oej | benjk: You're sunrise-ben, right? |
| 12:29.39 | benjk | I was working on a response to your mail but got stuck and then distracted with other stuff |
| 12:29.47 | benjk | re res_zeroconf |
| 12:29.55 | oej | Right. |
| 12:30.01 | benjk | there is only one benjk |
| 12:30.05 | oej | It happens, believe me, I know |
| 12:30.05 | benjk | ;-) |
| 12:30.06 | oej | :-) |
| 12:30.21 | Mozzie | santoshr : http://www.welltech.com/support/1501.htm |
| 12:30.59 | santoshr | yes Mozzie thts the device |
| 12:31.18 | Mozzie | santos which one is it |
| 12:31.33 | benjk | the thing is most of the developers contributing to astmasters.net projects don't agree with Digium's disclaim policy |
| 12:31.44 | santoshr | 1501 sip 1asipfxs.108.zip |
| 12:31.51 | Mozzie | santos have u the latest firmware ? |
| 12:31.55 | santoshr | yes |
| 12:31.57 | Mozzie | ok |
| 12:32.00 | tzafrir_laptop | does res_zeroconf work on anything but osx? |
| 12:32.17 | benjk | it works on Linux and BSD and Solaris too |
| 12:32.19 | santoshr | downloaded today /./ |
| 12:32.21 | santoshr | :( |
| 12:32.38 | tzafrir_laptop | what headers does it need for building, then? |
| 12:32.39 | benjk | but you need to download the (BSD licensed) Zeroconf library from Apple |
| 12:32.43 | Mozzie | ok .. ill have a quick read while im playing |
| 12:32.57 | benjk | on OSX that library is already on the system by default |
| 12:33.21 | tzafrir_laptop | hmmm.... is it in Debian yet? |
| 12:33.26 | benjk | the headers should be the same |
| 12:33.53 | oej | benjk: But have they contributed to res_zeroconf? |
| 12:33.55 | benjk | the library is available as source code from Apple, it is for any POSIX system |
| 12:34.20 | benjk | oej: res_zeroconf has two authors, Simon and myself |
| 12:34.32 | oej | Right. And you don't want to disclaim then? |
| 12:35.16 | benjk | we are rather disappointed with Digium's position on zeroconf and service discovery in general |
| 12:35.54 | benjk | one of our community members is the author of the firmware for a device called SitePlayer |
| 12:35.56 | oej | benjk: Asterisk is bigger than Digium. There are a lot of things they don't like, like DNS, DNS SRV, ENUM but that I work on anyway because a lot of users use it. |
| 12:36.26 | benjk | a small serial to Ethernet dongle that lets you ssh into a serial port remotely on servers even if they are off the network |
| 12:36.29 | oej | benjk: If you tell me that you can disclaim this code, I'll work on Digium. Going there next week. |
| 12:36.59 | benjk | anyway, this little marvel of technology has the same microcontroller as the IAXy |
| 12:37.01 | oej | benjk: But if you don't want to do it, it's ok. |
| 12:37.17 | oej | oej: benjk: But it would be stupid to have to create another implementation. |
| 12:37.32 | benjk | I am building up to how I could see a situation where we would disclaim it |
| 12:38.35 | *** part/#asterisk Romik_ (n=romik_@1.fix.netvision.net.il) |
| 12:38.44 | benjk | tzafrir, bkw used the module on Linux and he said he made a few changes, but he never sent us the patches. If you find there are modifications needed for compilation on Linux, please let us know |
| 12:38.53 | santoshr | Mozzie: did u find anything.. google wont say much though |
| 12:39.22 | benjk | oej: Siteplayer has zeroconf and the code takes 2 KByte of the ROM |
| 12:39.38 | oej | benjk: That's cool. Zeroconf and bonjour is cool. |
| 12:39.41 | benjk | Digium said they only have 3-4 KByte left in the IAXY |
| 12:39.56 | benjk | they said it wouls be impossible to do zeroconf in that |
| 12:40.07 | oej | benjk: Let's start with the Asterisk server, then try to make them understand the rest. |
| 12:40.09 | benjk | our guy has done it in 2K and he offered to help |
| 12:40.28 | Druken | benjk: ok, i owe ya an apology, my other server doesn't have a diffrent sound to busy and congestion... |
| 12:40.37 | oej | benjk: But to be clear, do you want to disclaim this regardless of the IAXY and their full love of zeroconf |
| 12:40.55 | benjk | We did a nice Cocoa GUI for an IAXy Configuration Assistant that would use Zeroconf to discover the IAXy and provision it |
| 12:41.08 | Druken | benjk: oh wait a min... the busy is flowing threw the servers.... |
| 12:41.19 | *** join/#asterisk gambolputty (n=gambolpu@cblmdm72-240-242-4.buckeyecom.net) |
| 12:41.31 | benjk | we offered Digium that assistant (once completed) if they out Zeroconf into IAXY |
| 12:41.57 | benjk | I also told them that Apple's Zeroconf tsar was interested in IAXy (I show the device to him and he was excited) |
| 12:42.12 | benjk | that there was a chance that Apple might carry IAXy in their Apple Stores |
| 12:42.18 | oej | benjk: Don't you see it. They do not understand this at all now and you are trying to make them implement it in their product. Start with Asterisk, work slowly. |
| 12:42.47 | oej | The IAXy doesn't even have proper DNS to start with... |
| 12:43.00 | benjk | Now, the reason why this would be of importance to use is that it would put the IAX protocol on Apple's radar and we'd likely get IAX support for iChat |
| 12:43.03 | oej | ...and the wrong protocol. Ha ha ha. |
| 12:43.17 | benjk | Well we agree to disagree there |
| 12:43.28 | benjk | I am all for IAX and LTP |
| 12:43.40 | oej | benjk: I see all the benefits, so let Digium discover this bit by bit. Please disclaim that code. |
| 12:44.37 | benjk | OK, here is the deal I propose to you |
| 12:44.48 | oej | listening... |
| 12:45.07 | benjk | You convince Digium to add zeroconf client support to the IAXy firmware |
| 12:45.34 | benjk | and I convince Simon (who wrote the lion share of the code) that we disclaim the code to Digium |
| 12:46.44 | oej | benjk: Now you're back to square one. They haven't even got DNS, so how can they add a full Bonjour? When we agree that they do not understand this? Let's implement it in Asterisk first, then show them what we can do with that and they will understand themselves. There's now way I can convince them of functionality in IAXy that we haven't proved in Asterisk first. |
| 12:47.40 | benjk | You don't need DNS to do Bonjour |
| 12:47.50 | oej | It's based on multicast DNS. |
| 12:47.50 | *** join/#asterisk warthog (n=nvadekar@66.55.113.93.ppp.northrock.bm) |
| 12:47.54 | benjk | Zeroconf uses DNS format for what it advertises |
| 12:48.07 | benjk | but it doesn't need a full DNS implementations |
| 12:48.19 | oej | Still need a DNS resolver |
| 12:48.21 | oej | Or? |
| 12:48.28 | oej | Not a full DNS server. |
| 12:48.30 | benjk | Alex Karahalios did the entire thing in 2K for the same microcontroller |
| 12:48.41 | Mozzie | santoshr: can u post ur full sip and extensions .conf files apart from the full dial plan... |
| 12:48.46 | Mozzie | ie passwords etfc |
| 12:49.49 | Mozzie | i dont need the passwords etc |
| 12:49.49 | benjk | and he offered assistance to accomplish the same for IAXy |
| 12:49.58 | santoshr | ya i can do tht not an issue and as it is.. i am still very close to the defaults onyl. |
| 12:50.07 | oej | benjk: That's fine, but they are not ready for it. As you said. |
| 12:50.41 | benjk | if somebody like yourself was to talk to Mark or Greg, I am sure they will open up |
| 12:50.57 | Mozzie | kewl |
| 12:51.02 | benjk | I believe this is just an issue of attention span |
| 12:51.35 | benjk | we talk to them through some employee who has 30 seconds to tell them what it is all about |
| 12:52.19 | benjk | but yourself, you should be able to get 10 or 15 minutes and get the message accross |
| 12:53.54 | benjk | so what do you think about the challenge? |
| 12:53.57 | oej | benjk: Everyone has to pick their fights. I don't want to press code into asterisk that way with an clause that they have to do it in the Iaxy to get the code for Asterisk. It's enough of a fight to get it into Asterisk to start with, that'll take *more* than 15 minutes. If you force me to do it for IAXy as well, it is not possible. Why not wait with that issue until we have it in Asterisk and can show people? |
| 12:54.17 | benjk | I wouldn't put it that way |
| 12:55.04 | benjk | I am not the sole owner of the IP |
| 12:56.32 | __a | is there a way to push DTMF into an already established channel? |
| 12:56.59 | benjk | whenever we have discussed this issue before (not just zeroconf) people get very emotional. They don't like to be pressed to sign over their work for somebody else to exploit it commercially while they themselves are bound by the limitations of the GPL. |
| 12:58.10 | oej | That is ok to me if they don't want to do that because of other reasons. But you are playing another game. |
| 12:58.32 | oej | But lets agree to disagree on that topic and move on. Where do I send my changes to zeroconf? |
| 12:58.49 | benjk | That's why I say in the case of Zeroconf, if there was support for a device like IAXy to come out of an agreement, that can do the trick and let people change their mind |
| 12:59.13 | benjk | I am not playing a game |
| 12:59.45 | oej | benjk: That won't happen... I have no way to change specs on the IAXy, however I can add stuff to Asterisk. |
| 13:00.05 | oej | So I guess that if I need zeroconf for Asterisk, I have to write another module then... |
| 13:00.07 | benjk | I am telling you that whoever signs that disclaimer will need to feel that they are happy, that they had done it for a purpose they believe in. |
| 13:00.07 | oej | Too bad. |
| 13:00.41 | benjk | Well, the module is GPL |
| 13:00.47 | benjk | let's not forget that |
| 13:01.17 | benjk | anybody can use the code and bundle it with Asterisk, as long as the entire package is GPL |
| 13:01.33 | benjk | just not for a commercial release of Asterisk |
| 13:01.46 | benjk | for that, you'll need an exception from the authors |
| 13:02.04 | oej | Benjk: I know how it works. |
| 13:02.07 | benjk | which, if it is oej asking for his client somewhere in Sweden |
| 13:02.27 | *** join/#asterisk areski (n=areski@polar.es6.egwn.net) |
| 13:02.34 | benjk | to whom he is sre-elling a commercial release |
| 13:02.43 | benjk | we'd be happy to grant it |
| 13:03.09 | *** part/#asterisk warthog (n=nvadekar@66.55.113.93.ppp.northrock.bm) |
| 13:03.11 | benjk | but not on a blanket basis to anybody who is using dual licensing as a weapon |
| 13:03.14 | oej | Well, you can't do that, benjk. |
| 13:03.30 | *** join/#asterisk pr0m (n=pr0methe@166.170.4.5) |
| 13:03.39 | oej | It's either only GPL, or a disclaimer or license agreement with Digium. |
| 13:03.40 | benjk | Simon and I can grant you any license to the module we please |
| 13:03.58 | oej | benjk: No, it's using the Asterisk API and by doing so, it's covered by GPL |
| 13:04.09 | benjk | if you have a commercial release from Digium that you resell |
| 13:04.16 | oej | So the only customer would be Digium |
| 13:04.17 | benjk | and we grant you an exception |
| 13:04.37 | *** join/#asterisk tdonahue (n=tdonahue@208.51.101.201) |
| 13:04.41 | benjk | then you can copy that module into the modules ditectory and let the commercial asterisk load it |
| 13:04.43 | oej | Well, I'm no american lawyer, but with a commercial license from Digium, yes, that may work. |
| 13:04.56 | __a | oej: mind a technical question? |
| 13:04.58 | benjk | for you there are two cases |
| 13:05.12 | benjk | either you provide a GPL version of Asterisk ot a client |
| 13:05.25 | benjk | in which case you can use the module under GPL terms |
| 13:05.30 | oej | ___a: go ahead |
| 13:05.52 | benjk | or you provide a commercial version of Asterisk to a client (as a reseller of Digium) |
| 13:06.39 | benjk | in which case you can add modules from anybody as long as the owners of the IP of these modules give you permission to do so |
| 13:06.58 | __a | oej: i'd like to make asterisk dial a calling card number via Zap interface, enter a pin and desired number and establish a conversation. Is it doable with out of the box asterisk or do I need to do custom programming here? |
| 13:07.22 | oej | Check the macro argument of the Dial command to do that |
| 13:07.45 | benjk | __a: you can send DTMF out from within the dialplan |
| 13:08.04 | *** join/#asterisk theNOTO (n=biggs@69-165-25-59.clvdoh.adelphia.net) |
| 13:08.16 | __a | benjk: any examples? |
| 13:08.28 | Mozzie | santoshr: which version of asterisk are you using ? |
| 13:08.54 | benjk | __a: as oej said, check the documentation of the dial command |
| 13:09.11 | __a | OK, doing that already. |
| 13:09.16 | benjk | alternatively the applications and available AGIs |
| 13:09.59 | __a | M(x[^arg]) - Execute the Macro for the *called* channel before connecting |
| 13:10.22 | __a | I guess I need to send DTMF *after* connecting, no? |
| 13:11.22 | oej | I think it executes after the outbound channel is answered, but before we bridge the incoming channel to the outgoing channel |
| 13:11.50 | __a | that'd be great. |
| 13:13.19 | Mozzie | http://www.asteriskguru.com/tools/switchboard.php any1 got this working correctly ? |
| 13:13.45 | *** join/#asterisk jmacz (n=jmacz@208.195.215.48) |
| 13:14.46 | zoa | yeah i did |
| 13:14.47 | zoa | :) |
| 13:14.59 | zoa | but then again i have the programmer sitting next to me somewhere |
| 13:15.01 | Druken | i'm gonna download and play with it... :) |
| 13:15.03 | zoa | but it should work |
| 13:15.22 | zoa | its the first version though, dont expect it to be flawless |
| 13:15.38 | fourcheeze | zoa: if you add a peer where does it save that to? |
| 13:15.52 | fourcheeze | is that just in the running conf? |
| 13:16.01 | benjk | __a: *CLI> show applications |
| 13:16.03 | zoa | just in the running conf |
| 13:16.10 | benjk | one of those is called SendDTMF |
| 13:16.11 | zoa | so restart asterisk and its gone |
| 13:16.11 | fourcheeze | so you have to get it to write out |
| 13:16.33 | zoa | the tool only does everything that the manager can do |
| 13:16.34 | zoa | nothing more |
| 13:16.51 | fourcheeze | can asterisk write out somewhere? |
| 13:17.12 | Druken | zoa: the second i connect to asterisk, i get an error... |
| 13:17.20 | Druken | access violation |
| 13:17.28 | benjk | __a: SendDTMF(digits): Sends DTMF digits on a channel. |
| 13:17.29 | benjk | <PROTECTED> |
| 13:17.29 | benjk | <PROTECTED> |
| 13:17.48 | Mozzie | zoa : a small suggestion... write a how to configure the baby so ppl that dont have a clue **WHISTLES** gets info of how to install it |
| 13:18.22 | __a | benjk: I'm aware that I should send DTMF, I just needed to figure out how to do it after the called party has answered. If Macro works as oej said, then my problem is solved. |
| 13:18.38 | benjk | ok |
| 13:18.42 | Druken | otherwise an impressive program... |
| 13:20.52 | *** join/#asterisk synthetiq (n=roger@64.201.13.50) |
| 13:21.59 | zoa | druken, cool |
| 13:22.00 | zoa | :) |
| 13:22.11 | zoa | mozzie, i will someday |
| 13:22.14 | zoa | first fixing the other stuff |
| 13:22.17 | zoa | lots of work here |
| 13:22.47 | *** join/#asterisk aminorex (n=tony@71-13-40-131.dhcp.dlth.mn.charter.com) |
| 13:22.49 | Druken | does the multiserver support work ? |
| 13:23.05 | __a | from the docs, looks like Macro works before the party answers :( |
| 13:23.27 | __a | zoa: any idea how to send dtmf digits after the remote party has answered? |
| 13:24.11 | Druken | oh, excelent.. it does |
| 13:24.34 | *** join/#asterisk DeeJayTwo (n=deejay2@37-179.sh.cgocable.ca) |
| 13:24.43 | Druken | zoa: why does it say everything is down ? |
| 13:26.02 | Mozzie | zoa : np mate - at least yours doesnt need .net ; ) |
| 13:27.15 | Druken | pretty lights too... |
| 13:27.23 | Zeeek | error software caused connection abort |
| 13:27.31 | *** join/#asterisk bjohnson (n=bjohnson@jecinc.tor.istop.com) |
| 13:27.50 | Druken | Zeeek: i had that... to fix it, i had to remove that server connection and put it back in... |
| 13:28.05 | Druken | worked fine afterwards, i got both my servers in, it's a pretty nice tool |
| 13:29.31 | Zeeek | I think also if I were to allow access to the requesting machine ip it might help :) |
| 13:30.45 | demetrio | how can I spoof callerID before forwarding a call to a local sip phone? |
| 13:31.09 | Zeeek | using the Set(CALLERID(num) finction |
| 13:31.21 | Zeeek | or before 1.2 SetCallerIDNum |
| 13:31.33 | demetrio | thanks |
| 13:34.38 | Zeeek | zoa that works well |
| 13:34.46 | docelmo | GOOD MORNING ALL! |
| 13:35.39 | santoshr | Mozzie: workin now.. it was in peer-peer mode and when shifted it to proxy it connected asking for password mismatch.. .. |
| 13:35.47 | santoshr | we are using asterisk v 1.2 |
| 13:35.54 | bbz | Would anyone here be interested in some Polycom 501 phones? We got stuck w/ an extra 20 after a project, and we need to unload them at cost, -- if anyone is interested send me a private message (if i dont respond immediately, leave me your email address, and i can send you info) i have a reference in the channel if you need. -- |
| 13:36.18 | Zeeek | bbz have you posted to the asterisk list? tens of thousands of readers |
| 13:36.54 | santoshr | and then put username , secret stuf in the sip.conf. |
| 13:38.34 | crash3m | can anyone here access http://www.linksys.com ? my connection is being refused :/ |
| 13:39.26 | *** join/#asterisk toddf (n=toddf@net-66-210-104-120.theshop.net) |
| 13:40.13 | Zeeek | me too - so don't worry |
| 13:40.23 | crash3m | heh, but...but...I need firmware! |
| 13:40.35 | Zeeek | "we're sorry, we are currently being hacked and can't allow you web access" |
| 13:40.46 | Zeeek | what firmware? |
| 13:40.46 | crash3m | heh |
| 13:40.51 | crash3m | wrt54g v5 |
| 13:41.24 | Zeeek | prolly won't last long |
| 13:41.31 | Zeeek | the downtime I mean |
| 13:41.37 | crash3m | v5 -w- firmware 1.00.0 breaks sip, trying to see if any of the other releases might fix it |
| 13:41.42 | Mozzie | kewl santoshr |
| 13:41.42 | Mozzie | fantastic matey |
| 13:43.03 | *** join/#asterisk [TK]D-Fender (n=joe@toronto-HSE-ppp4122655.sympatico.ca) |
| 13:46.26 | crash3m | never fails...when I really want to do something, I cant |
| 13:47.10 | Zeeek | that's a common dilemma in life |
| 13:48.40 | DeeJayTwo | heh maybe someone of you guys could know the answer to my question ;) |
| 13:48.49 | DeeJayTwo | <PROTECTED> |
| 13:48.56 | DeeJayTwo | <PROTECTED> |
| 13:49.29 | DeeJayTwo | discarding the fact that it would like to authenticate and relaying it without question? ;) |
| 13:52.44 | tzafrir_laptop | DeeJayTwo, smtp over ssl? if it is over ssl, recall that the certificate should have the exact host name |
| 13:53.58 | tzafrir_laptop | DeeJayTwo, anyway, this is highly configuration-specific |
| 13:54.16 | *** join/#asterisk scardinal (n=supreme@port816.ds1-suoe.adsl.cybercity.dk) |
| 13:55.12 | DeeJayTwo | yeah... well as you already guessed it is for hotspots. There are people using SMTP authenticaion with their provider and other not... |
| 13:55.32 | DeeJayTwo | I was just wondering if doing SMTP redirection would be usefull for ones and a mess for others... ? |
| 13:57.39 | *** join/#asterisk RoyK (n=roy@ti211310a080-4132.bb.online.no) |
| 13:58.02 | DeeJayTwo | anyway, smtp over ssl isn't using port 25? |
| 13:58.20 | DeeJayTwo | so if I'm not redirecting traffic for any other port than 25, it should be ok for everybody.. |
| 14:02.56 | *** join/#asterisk kshumard (n=ksh@gateway.digium.com) |
| 14:05.12 | crash3m | smtps uses 465 |
| 14:06.38 | tzafrir_laptop | anybody here using the tcl asterisk manager bindings? I tried to use their sample to get a call-back function in response to a command, but the call-back is not called. Anybody had success? |
| 14:07.27 | tzafrir_laptop | Alternatively, what do you use for a decent interface over the manager api? |
| 14:13.48 | Delvar | tzafrir_laptop: telnet :) |
| 14:13.49 | *** join/#asterisk jcath (n=skycat@221.216.15.99) |
| 14:16.38 | *** join/#asterisk Dead-Bum (n=Satan@tor/session/x-99ef5f57df054670) |
| 14:17.20 | Dead-Bum | Anyone know why amp is not writing to the .conf files in /etc/asterisk/? I've temporarily chmodded them to 777 and they still aren't being written to. |
| 14:17.34 | tzafrir_laptop | Delvar, the point is that I have to keep track of what is the response to what. Not to mention parse everything. E.g: Asterisk::Manager simply assumes that the first thing that comes after an event is the response to it. This already broke badly for me |
| 14:17.36 | Dead-Bum | asterisk 1.2, amp 1.10.010 |
| 14:17.54 | *** join/#asterisk paryl (n=paryl@216-201-177-82.res.logixcom.net) |
| 14:18.23 | tzafrir_laptop | Dead-Bum, what about the directory /etc/asterisk itself? |
| 14:18.46 | paryl | ok... the newly released stuff is causing me all kinds of issues. now when i compile zaptel drivers from the cvs head i get "zaptel: Unknown symbol crc_ccitt_table" in dmesg |
| 14:18.57 | tzafrir_laptop | also, please avoid the executable bit. try 666. You'll also have nicer colors in ls in the shell |
| 14:19.19 | *** join/#asterisk __a (n=a@85.100.123.10) |
| 14:19.28 | Dead-Bum | tzafrir_laptop: I'll modify accordingly, hold on. |
| 14:19.29 | tzafrir_laptop | paryl, use modprobe rather than insmod. or insmod |
| 14:20.07 | tzafrir_laptop | paryl, anyway, insmod crc_cccitt |
| 14:20.23 | __a | guys, how to make asterisk register to a proxy server with realtime? |
| 14:20.24 | paryl | tzafrir: problem is, make install doesn't seem to install the modules so that modprobe can see them... i always get "unknown module zaptel" |
| 14:20.27 | Dead-Bum | tzafrir_laptop: chmod'd the directory and all the files 666, still won't write. |
| 14:20.56 | tzafrir_laptop | paryl, run 'depmod -a' after adding modules under /lib/modules |
| 14:21.18 | __a | with sip.conf I'd write register => user@proxy, but now that I'm using realtime, I'd like to write this register=> thingy in DB as well. If possible of course. |
| 14:21.42 | __a | paryl: depmod -a |
| 14:22.02 | tzafrir_laptop | Dead-Bum, what distro? |
| 14:22.07 | paryl | __a: yeah... i've tried that... i could never figure it out |
| 14:22.10 | Dead-Bum | tzafrir_laptop: centos 4.2 |
| 14:22.42 | Dead-Bum | tzafrir_laptop: It somehow magically wrote to one of the files yesterday and then mysteriously stopped being able to. |
| 14:23.24 | tzafrir_laptop | paryl, did you install it under the right directory? under /lib/modules/`uname -r` ? |
| 14:23.58 | paryl | the modules get put into /lib/modules/2.6.12/misc automatically, and i have a link to that under the 'uname -r' directory |
| 14:24.26 | paryl | shouldn't that work? |
| 14:25.07 | jlewis | anyone around who can give me some advice on entering an issue the the bug tracker "correctly"? |
| 14:25.15 | paryl | actually... i have my answer... the link wasn't working but copying the misc directory worked... wierd |
| 14:28.40 | Dead-Bum | tzafrir_laptop: Any ideas? |
| 14:31.00 | tzafrir_laptop | Dead-Bum, sorry, no :-( |
| 14:31.16 | tzafrir_laptop | jlewis, I figure you read the guidelines |
| 14:31.22 | Dead-Bum | dang |
| 14:31.23 | Dead-Bum | ok |
| 14:31.24 | Dead-Bum | thanks anyway. |
| 14:31.26 | *** join/#asterisk coppice (n=chatzill@168.155.17.210.dyn.pacific.net.hk) |
| 14:31.49 | benjk | Hi coppice, still in Osaka? or back in HK |
| 14:32.11 | coppice | yes |
| 14:32.27 | *** join/#asterisk funto (n=tomasz@80.50.247.26) |
| 14:32.38 | benjk | so how do you like Nippon? |
| 14:33.04 | coppice | interesting place, and interesting people |
| 14:33.11 | coppice | and great food |
| 14:33.28 | benjk | :-) |
| 14:33.29 | *** join/#asterisk konfuzed (n=KonfuzeD@H129.C72.B0.tor.eicat.ca) |
| 14:33.35 | coppice | i was suprised to find how authentic the japanese restaurants in HK are |
| 14:34.04 | benjk | well, you have to take into account that most of those are there for the Japanese expats |
| 14:34.25 | coppice | not at all |
| 14:34.53 | *** join/#asterisk a_phantasm (n=RK@207-255-7-018-static.jst.pa.atlanticbb.net) |
| 14:34.53 | benjk | perhaps not anymore, but that's how it used to be when the Japanese economy was bubbling |
| 14:35.59 | a_phantasm | Is anyone available that might be able to help me figure out a few headaches? |
| 14:36.42 | benjk | then again, for the ultimate sushi experience you have to come to Tokyo and visit the sushi bars at the fishmarket in Tsukiji |
| 14:36.50 | fourcheeze | I have paracetamol |
| 14:37.08 | *** join/#asterisk anthm (n=anthmct@CPE-69-76-83-52.wi.res.rr.com) |
| 14:37.08 | *** mode/#asterisk [+o anthm] by ChanServ |
| 14:37.24 | benjk | fourcheez: LOL |
| 14:37.45 | a_phantasm | It seems whenever I restart the system, asterisk wont auto start unless I dump all the extentions |
| 14:38.24 | *** join/#asterisk Jzalae (n=sk@dsl-66-63-110-48.gwi.net) |
| 14:38.30 | a_phantasm | Unregistered channel type 'Zap' and so on until Loading module chan_zap.so failed! |
| 14:38.41 | a_phantasm | and can't figure out how to get it to register |
| 14:39.38 | Zeeek | a_phantasm is there a message about running out of stack space ? |
| 14:39.50 | a_phantasm | None that I can see |
| 14:39.58 | Zeeek | and did you run ztcfg ? |
| 14:40.06 | Zeeek | idf so, what is the output |
| 14:40.07 | a_phantasm | Yeah |
| 14:40.11 | Zeeek | if not, why not? |
| 14:40.11 | a_phantasm | One second |
| 14:40.23 | Zeeek | run it with -vvvvv |
| 14:41.51 | *** join/#asterisk drc_ (i=drclaw@wakko.cs.wmich.edu) |
| 14:43.14 | Chotaire | gutentakk! |
| 14:43.21 | drc_ | sorry to join and jump right into a question, but that's why I'm here =(. Does anyone know how I can override the date/time in callerID set by my pstn provider on incoming calls? |
| 14:43.42 | drc_ | oh yeah, also hello |
| 14:43.53 | a_phantasm | [root@asterisk1 root]# ztcfg -vvvvv |
| 14:43.53 | a_phantasm | Zaptel Configuration |
| 14:43.53 | a_phantasm | ====================== |
| 14:43.53 | a_phantasm | Channel map: |
| 14:43.53 | a_phantasm | Channel 01: FXS Kewlstart (Default) (Slaves: 01) |
| 14:43.54 | a_phantasm | Channel 02: FXS Kewlstart (Default) (Slaves: 02) |
| 14:43.56 | a_phantasm | Channel 03: FXS Kewlstart (Default) (Slaves: 03) |
| 14:43.58 | a_phantasm | Channel 04: FXS Kewlstart (Default) (Slaves: 04) |
| 14:44.00 | a_phantasm | Channel 05: FXO Kewlstart (Default) (Slaves: 05) |
| 14:44.04 | jhiver | oh joy |
| 14:44.04 | a_phantasm | Channel 06: FXO Kewlstart (Default) (Slaves: 06) |
| 14:44.05 | a_phantasm | Channel 07: FXO Kewlstart (Default) (Slaves: 07) |
| 14:44.11 | a_phantasm | Channel 08: FXO Kewlstart (Default) (Slaves: 08) |
| 14:44.11 | a_phantasm | 8 channels configured. |
| 14:44.11 | a_phantasm | ztcfg output ^^ |
| 14:44.22 | jhiver | I'm trying to place a call to France and it work neither with VoIPJet nor NuFone :( |
| 14:44.31 | jhiver | <PROTECTED> |
| 14:44.34 | *** join/#asterisk santiago (n=santiago@208.195.215.160) |
| 14:44.42 | jhiver | Nov 23 18:43:39 WARNING[2538]: chan_iax2.c:1480 attempt_transmit: Max retries exceeded to host 216.118.117.46 on IAX2/voipjet/4 (type = 6, subclass = 1, ts=8, seqno=0) |
| 14:44.47 | jhiver | duuuh |
| 14:44.50 | Zeeek | jhiver you want me to try the number? |
| 14:45.00 | Zeeek | a_phantasm looks ok to me |
| 14:45.09 | jhiver | errrr, no actually, it's you know, like a business number |
| 14:45.30 | a_phantasm | Zeeek, One more stupid question |
| 14:45.41 | Chotaire | !seen kapejod |
| 14:45.46 | jhiver | looks like I'll have to modify my dialplan to use Zap as a last resort |
| 14:45.49 | Chotaire | anyone seen kapejod lately? |
| 14:45.54 | jhiver | I'll check that I have some balance left too :) |
| 14:46.06 | benjk | Chotair, two days ago |
| 14:46.08 | a_phantasm | All the lines connected, either SIP or Zap Channel, when you call the extentions, it says they are on the phone... But none of the lines are in use |
| 14:46.27 | drc_ | jhiver: it's not a matter of an exten pattern being matched by something else and then only part of the number being sent to voipjet/nufone for sure right? |
| 14:46.37 | Chotaire | benjk: greetings mate... ok cool, so he's still alive. do you by accident happen to know if chan_capi was fixed for asterisk 1.2.0 finally? when I tried on friday it was still not working properly. |
| 14:47.03 | Chotaire | junghanns told me it was about to be released over the weekend. |
| 14:47.04 | jhiver | IAX2/543@voipjet/011331xxxxxxxx |
| 14:47.08 | drc_ | k |
| 14:47.16 | docelmo | jhiver check out plainvoip.com |
| 14:47.16 | jhiver | (i've changed the real number by xxx) |
| 14:47.22 | docelmo | :) SIP and IAX support |
| 14:47.38 | jhiver | is this worth something? |
| 14:48.01 | benjk | Chotaire: kapejod said something like "/me just finished bristuffing Asterisk 1.2" |
| 14:48.06 | drc_ | jhiver, got me, sorry |
| 14:48.44 | a_phantasm | Why would it say "The person at extention XXXX is on the phone" when no one is using any of the lines? |
| 14:49.00 | jhiver | Dr_Ray, ? |
| 14:49.05 | jhiver | drc_, ? |
| 14:49.10 | jhiver | I don't understand |
| 14:49.16 | drc_ | drc != dr ray |
| 14:49.26 | jhiver | I know, it's stupid tab completion |
| 14:49.41 | jhiver | sorry :) |
| 14:49.43 | Chotaire | bristuff? oh well... |
| 14:49.50 | jhiver | btw lads |
| 14:49.50 | __a | <PROTECTED> |
| 14:49.50 | __a | <PROTECTED> |
| 14:49.50 | __a | <PROTECTED> |
| 14:49.50 | __a | <PROTECTED> |
| 14:49.50 | __a | <PROTECTED> |
| 14:49.56 | Chotaire | i hope its about the same when it comes to usage. |
| 14:50.02 | jhiver | Can somebody explain to me the difference between "quad span E1" and "quad E1" ? |
| 14:50.08 | __a | does anybody know if there's a builtin command that sends DTMF *after* bridging? |
| 14:50.10 | Chotaire | let me try to call him and see if he's done. |
| 14:50.40 | drc_ | anyone have an idea what to look into on my callerid question? (changing the incoming time/date from what the provider sends).. tried googling, but no joy so far |
| 14:51.24 | *** part/#asterisk bugz (n=will@69.15.174.114) |
| 14:52.07 | drc_ | I should probably mention it's on a voip line, not pri or anything |
| 14:53.04 | Zeeek | jhiver what is the problem with voipjet? |
| 14:53.26 | jhiver | It Doesn't Work(tm) |
| 14:53.32 | jhiver | it says: |
| 14:53.35 | Chotaire | bristuff doesnt work with avm cards, right? |
| 14:53.40 | jhiver | Nov 23 18:43:39 WARNING[2538]: chan_iax2.c:1480 attempt_transmit: Max retries exceeded to host 216.118.117.46 on IAX2/voipjet/4 (type = 6, subclass = 1, ts=8, seqno=0) |
| 14:53.44 | Zeeek | you want me to try a call to France with voipjet? |
| 14:53.52 | Zeeek | what zone (first two numbers) |
| 14:53.59 | Zeeek | I'll call myself if it's 01 |
| 14:54.05 | jhiver | 01 |
| 14:54.15 | Zeeek | and you tried nufone too? |
| 14:55.01 | jhiver | yep |
| 14:55.08 | jhiver | <PROTECTED> |
| 14:55.40 | jhiver | I have $36.58 left on nufone |
| 14:55.41 | Zeeek | how are you dialing with nufone? 011331 ? |
| 14:55.52 | Chotaire | hehe, kapejod's mailbox is full. |
| 14:56.04 | jhiver | jhiver@NuFone/011331xxxxxxxxx |
| 14:56.48 | zobia | how can send several outbound call in one time? |
| 14:56.53 | zobia | not dial one by one? |
| 14:57.43 | zobia | hello Zeeek. do u know how to initiate several outbound call in one time? |
| 14:57.56 | jhiver | zobia, what do you mean? |
| 14:58.12 | jhiver | like getting asterisk to call people to send them "VoIP Spam"? |
| 14:58.15 | jhiver | use .call files |
| 14:58.17 | drc_ | zobia: Dial(IAX2/user@provider/5551212&SIP/user@provider/5551212&ZAP/2/5551212) |
| 14:58.23 | jhiver | aaaah I see |
| 14:58.49 | *** join/#asterisk BladeRunner05 (n=feelme@adsl-130-214.37-151.net24.it) |
| 14:59.25 | a_phantasm | Why would it say "The person at extention XXXX is on the phone" when no one is using any of the lines? |
| 14:59.27 | drc_ | or call files like jhiver suggests depending on what your trying to do |
| 14:59.48 | jhiver | .call files are fun methinls |
| 14:59.49 | drc_ | a_phantasm: are you sure you want kewlstart? |
| 14:59.51 | jhiver | methinks |
| 15:00.07 | zobia | hello drc_ |
| 15:00.17 | zobia | what u give me is for inbound. |
| 15:00.32 | drc_ | zobia: no dial() is for outboudn |
| 15:00.32 | _DAW | good morning |
| 15:00.41 | zobia | i want to dial outbound . like if dial several line in the same time. |
| 15:00.43 | Zeeek | jhiver, I'll be back in about 20 min and I can check those for you - I'm interested since I use both (mostly for USA though) |
| 15:01.00 | drc_ | zobia: yeah, what I suggested should work |
| 15:01.07 | *** part/#asterisk Zeeek (n=icechat5@pdpc/supporter/active/Zeeek) |
| 15:01.15 | jhiver | ok Zeeek |
| 15:01.22 | zobia | i did that. but if one of these number answered other call will stopped ringing. |
| 15:01.44 | zobia | but i want each call has their own line, doesn't effect each other. if one answered. other ones still ringing. |
| 15:01.45 | drc_ | ah ok, use call files then if you want to have them all continue to ring |
| 15:01.58 | *** join/#asterisk kFuQ (n=somedude@c-67-185-114-199.hsd1.wa.comcast.net) |
| 15:02.13 | _DAW | can someone tell me if it is possible to pass alert_info with the page cmd in *? |
| 15:02.30 | zobia | how to make call files conitinue to ring? |
| 15:02.58 | drc_ | zobia: you can craft .call files to move into /var/spool/asterisk/outgoing and point them to a context that handles what to do with the call after the party answers |
| 15:02.58 | zobia | i wrote on call file. with one channel. if i want 5 channels. i need to write them 1 bye |
| 15:03.08 | Chotaire | well it seems as if bristuff is NOT for capi 2.0 cards, so in that case chan_capi is still not working with asterisk 1.2.0.. anyone has an idea? |
| 15:03.11 | Chotaire | I can't upgrade ;( |
| 15:03.19 | drc_ | zobia: just write five call files and move them all to the spool at once |
| 15:05.22 | zobia | wow . drc_ can u give an example? |
| 15:05.46 | drc_ | zobia: are you using a voip provider or some kind of fxo'd ata / digium pci card to access outgoing channels? |
| 15:06.18 | zobia | i am using diguium card. |
| 15:06.33 | zobia | like using zap/g1 |
| 15:06.36 | zobia | to dial out |
| 15:07.42 | drc_ | so make a .call file with Channel: zap/g1/NUMBER, and then another call file with Channel: ZAP/g2.. I'm not really sure if there's a better way without enumerating each channel, sorry =( |
| 15:07.48 | RoyK | digigugiuium card |
| 15:08.24 | Chotaire | you have a digigugiuium card? I thought they were only sold in botswana. |
| 15:11.15 | *** join/#asterisk Inkubot (n=inkubot@200.75.4.7) |
| 15:11.28 | *** join/#asterisk _T3_ (n=rposada@200.63.231.210) |
| 15:13.02 | coppice | benjk: I managed to spend tuesday afternoon in Kyoto, but the traffic was so bad we didn't get to see many things :-( |
| 15:13.33 | __a | guys, what does w mean in a DTMF string? |
| 15:13.39 | Damin | coppice: Yeah.. I hate the traffic in Japan. Makes the states look good in comparison. ;) |
| 15:13.52 | Damin | __a: w = 9 |
| 15:14.25 | coppice | seoul makes kyoto look good. bangkok makes seoul look good :-) |
| 15:14.34 | *** join/#asterisk rking (n=rking@ip68-1-234-152.dl.dl.cox.net) |
| 15:14.54 | *** join/#asterisk Pete_Largo (n=PeteLarg@225-196.35-65.tampabay.res.rr.com) |
| 15:15.11 | __a | Damin: what's its purpose when there's normal 9? |
| 15:15.45 | _DAW | does anyone know if it is possible to set alert_info value with the page command? |
| 15:16.03 | chapeaurouge | i have a terminology question. There are 2 ISDN-BRI "lines" at my company. But we have a PBX there (soon to be replaced with *). Should we call these "lines" trunks? |
| 15:16.50 | Pete_Largo | I'm having trouble getting spandsp to compile... I'm installing a new * server and I am getting an error when comiling at app_rxfax.c line 267 the error is "structure has no member named 'logging'" Anyone seen this before? I can't find anything via google... |
| 15:17.42 | coppice | no logging == old version |
| 15:17.42 | Pete_Largo | chapeaurouge: a simple definition is - trunks connect switches to swithes, lines connect switches to phones... |
| 15:18.05 | Pete_Largo | coppice: can you elaborate? |
| 15:18.15 | chapeaurouge | so, we should call them trunk I guess, since a PBX is a switch |
| 15:18.27 | Pete_Largo | chapeaaurouge: I would ;) |
| 15:18.32 | chapeaurouge | thx ;) |
| 15:18.37 | coppice | Pete Largo: there is nothing else to say |
| 15:18.40 | Pete_Largo | but the phone company may disagree with you. |
| 15:18.45 | *** join/#asterisk RoyK (n=roy@ti211310a080-4132.bb.online.no) |
| 15:18.55 | Pete_Largo | coppice: am I using the wrong version of spandsp? |
| 15:20.28 | coppice | if app_rxfax expects logging, and spandsp doesn't provide it, there is a version mismatch between them |
| 15:21.22 | Pete_Largo | hmmph... not nice, I downloaded them from the softswitch.org asterisk 1.2 directory this morning ... |
| 15:21.52 | *** join/#asterisk Zeeek (n=IceChat7@pdpc/supporter/active/Zeeek) |
| 15:22.05 | Zeeek | jhiver voipjet to France is working |
| 15:22.07 | Pete_Largo | should I be using pre 21 c with *1.2 ? |
| 15:22.12 | coppice | If you downloaded pre12c and used the spandsp and app_rxfax in there, you would not have that error |
| 15:22.30 | Pete_Largo | would not, or should not :) |
| 15:23.01 | coppice | so you don't want to solve your problem. bye |
| 15:23.18 | Pete_Largo | whoa, hold up there cowboy... |
| 15:23.22 | Pete_Largo | I'm listening to you |
| 15:23.35 | Pete_Largo | I just deleted the asterisk directory so I can start fresh |
| 15:23.42 | Pete_Largo | touchy touchy |
| 15:23.50 | *** join/#asterisk ast_freak (n=jesse@68-112-134-195.dhcp.stcl.mn.charter.com) |
| 15:28.37 | Pete_Largo | I wonder if I'm doing something wrong in the Makefile... the patch doesn't work so I'm 'patching' it by hand... |
| 15:30.06 | kippi | can someone help me please? I want to remove amp all together and learn how to build a asterisk box without a gui! could somoeone help me? |
| 15:30.35 | *** join/#asterisk azzie (n=az@azzie.net) |
| 15:31.15 | *** join/#asterisk tartar (n=tartar@CPE0004e27b716e-CM014370001917.cpe.net.cable.rogers.com) |
| 15:31.23 | Pete_Largo | kippi: there is a pretty good step-by-step at www.automated.it/guidetoasterisk.html it's what I used when I first started |
| 15:32.02 | *** join/#asterisk desktophero (n=chatzill@ip24-56-30-250.ph.ph.cox.net) |
| 15:32.43 | Pete_Largo | coppice: Steve, you were absolutely right, I must have made a stupid because I started fresh, downloaded everything again, and it seems to have worked fine... I apologize for being a smartass, thanks for your help. |
| 15:35.55 | *** join/#asterisk brc_ (n=Brian@pdpc/supporter/basic/brc) |
| 15:37.16 | rajiv | damn. i'm looking at my cdr. someone put 2 calls on hold for 2 hours each. |
| 15:37.16 | docelmo | Uippie! |
| 15:37.21 | docelmo | err Yippie! |
| 15:37.31 | docelmo | sigh.. Friday's.. |
| 15:38.38 | rajiv | looks like they called 1 person, it was busy or something, pressed flash, got a dial tone, dialed a second #. so the first call was on hold the whole time |
| 15:40.27 | rajiv | i think i need to dump these analog phones |
| 15:41.09 | demetrio | that doesn't make sense. an ATA should either hang the first call up or make the caller aware of the pending call in some way. |
| 15:42.04 | coppice | rajiv: isn't that what should happen? sounds normal |
| 15:42.19 | rajiv | coppice: ya i thikn that is what should happen. |
| 15:42.54 | rajiv | actually no |
| 15:43.11 | rajiv | if the first call out was busy, then * should have terminated the call |
| 15:43.38 | rajiv | the call went out over a zap trunk. so * sat there listening to the busy for 2 hrs? |
| 15:44.24 | rajiv | hmm. i dont have busydetect=yes in zapata.conf |
| 15:44.32 | coppice | rajiv: so you have a busy detect problem. that isn't too rare. Maybe * needs to half decent DSP :-) |
| 15:45.19 | rajiv | heh. is busydetect=yes the default ? it is commented out in the config |
| 15:46.06 | *** part/#asterisk SplasPood (i=nobody@paravolve.net) |
| 15:46.15 | kippi | could someone have a look at my dial plan please? |
| 15:46.32 | iDunno | nope. you haven't given us a URL or *anything*. |
| 15:46.36 | iDunno | <-- useful ;) |
| 15:46.40 | kippi | http://pastebin.ca/30596 |
| 15:46.43 | docelmo | I can look at it.. But what do you wanna know? |
| 15:46.48 | kippi | its just hanging up |
| 15:46.57 | kippi | it not going to voicemail |
| 15:47.26 | iDunno | exten => 478102,3,GoToIf($[$DIALSTATUS) =CHANUNAVAIL]?7) |
| 15:47.27 | iDunno | exten => 478102,4,GoToIf($[$DIALSTATUS) =BUSY]?9) |
| 15:47.32 | iDunno | ^ they look broken. |
| 15:47.52 | docelmo | Why do it that way |
| 15:47.59 | docelmo | Try it the way I am gonna change it to |
| 15:48.14 | rajiv | dial() will jump to n+1 if busy |
| 15:48.23 | rajiv | no need to do the goto if |
| 15:48.44 | oej | rajiv: Please read "show application dial" |
| 15:49.03 | oej | n+1 is *not* busy and the jumping is deprecated, using DIALSTATUS is much better |
| 15:49.12 | rajiv | oops. n+101 |
| 15:49.36 | docelmo | http://pastebin.ca/30600 |
| 15:49.41 | rajiv | i forgot bc i use a macro based on DIALSTATUS |
| 15:50.14 | docelmo | well damnit.. |
| 15:50.19 | *** join/#asterisk SplasPood (i=nobody@paravolve.net) |
| 15:50.24 | docelmo | It still works tho.. I use it @ work |
| 15:51.51 | docelmo | And I am running head from like a month ago or something |
| 15:52.04 | rajiv | ah. busydetect default is no. |
| 15:52.13 | *** join/#asterisk MikeJ[Laptop] (n=ircatjer@mi.origenfinancial.com) |
| 15:53.40 | *** join/#asterisk kpettit (n=keith@69.15.174.114) |
| 15:54.21 | *** join/#asterisk pr0m (n=pr0methe@166.170.0.244) |
| 15:54.56 | __a | guys, is it possible to write register => user@sipproxy with asterisk in realtime? |
| 15:55.43 | docelmo | yes... put it in your sip.conf |
| 15:56.04 | *** join/#asterisk lehel (n=lehel@82.79.20.17) |
| 15:56.12 | lehel | hello |
| 15:56.18 | lehel | in every 5 mins i get: DEBUG[5312]: chan_iax2.c:3800 raw_hangup: Raw Hangup 172.24.4.8:34722, src=0, dst=0 |
| 15:56.20 | docelmo | olleh! |
| 15:56.42 | file | it's debug. |
| 15:56.46 | __a | docelmo: any chance of putting it in db? I'd like it to be editable via web interface |
| 15:57.06 | file | debug messages are not meant to be seen under normal circumstances, you're just freaking because you turned it on and are going wtf?!? these have to be errors! when they aren't. |
| 15:57.13 | __a | docelmo: or is it possible to define a user foo in db and put register => foo sip.conf? |
| 15:57.16 | docelmo | No.. but you can just edit a #include sip_include.conf file |
| 15:57.23 | docelmo | you can do that also |
| 15:57.26 | docelmo | I believe.. |
| 15:57.41 | docelmo | no.. sip.conf variables are only in the sip.conf not channel vars.. |
| 15:57.59 | lehel | file, yea but i don't like it;) especially 'coz i've nothing connected to that 4.8, yes i now about debug. |
| 15:58.43 | docelmo | __a, you could do this.. make your config inside a database then have your script dump the config in the db in a formatted file for sip.conf |
| 15:59.05 | docelmo | Be like AMP.. {PUKE} |
| 15:59.27 | lehel | file: anyway.. it doesn't really matters, what matters is that i have echo between iax-virtual and zap ;\ |
| 15:59.54 | lehel | however, i have everything turned on [echocancel..] |
| 16:00.41 | *** join/#asterisk A-Tuin|work (n=A-Tuin@nat.office.legend.net.uk) |
| 16:01.10 | file | there's so much.. sooo much stuff available online about echo, tweaking, etc |
| 16:01.27 | lehel | file, k .. nm |
| 16:01.39 | file | you are not alone... |
| 16:01.43 | lehel | i know ;) |
| 16:01.58 | lehel | i'm just maybe a bit lazy.. my fault anyway |
| 16:03.09 | rajiv | this whole analog phone world is getting me down. i thought in this day and age we could move past needing to check voltages on the line... |
| 16:03.50 | *** join/#asterisk Utah_Dave (n=boucha@0-2pool130-228.nas28.salt-lake-city1.ut.us.da.qwest.net) |
| 16:04.31 | Katty | file: i need information |
| 16:04.44 | Katty | file: is setting up asterisk with broadband cards anything like setting up asterisk with analog cards? |
| 16:04.50 | Katty | file: or is it insanely more complicated, etc |
| 16:05.23 | file | broadband cards? you mean PRI? |
| 16:05.25 | docelmo | Katty, broadband meaning PRI cards? |
| 16:05.26 | asterboy | Can someone point me to a good SIP resource? |
| 16:05.41 | docelmo | asterboy, define sip resource |
| 16:05.49 | asterboy | faq |
| 16:05.57 | asterboy | need to do some reading. |
| 16:06.02 | asterboy | on how to setup |
| 16:06.04 | docelmo | Read the RFC |
| 16:06.15 | asterboy | 3286? |
| 16:06.26 | docelmo | I think that might be it.. |
| 16:06.42 | docelmo | But that gives hardcore information.. |
| 16:06.53 | asterboy | 3161 |
| 16:06.53 | asterboy | 3261 |
| 16:07.03 | docelmo | ya.. Just googled it. |
| 16:08.29 | asterboy | Is there a more summarized version. |
| 16:08.32 | asterboy | ? |
| 16:08.34 | docelmo | That should keep you busy for awhile |
| 16:08.38 | docelmo | Not really.. |
| 16:08.44 | docelmo | everything you want to know about it is there |
| 16:08.52 | rajiv | try the oreilly book maybe |
| 16:09.03 | LostFrog | I should read the RFCs.. I could use a nice nap. |
| 16:09.03 | *** join/#asterisk msw (n=msw@rdu-nat.rpath.com) |
| 16:09.11 | docelmo | oreilly has a book on sip? |
| 16:09.16 | docelmo | What do they not have a book on |
| 16:09.22 | rajiv | the * book has a section on sip |
| 16:09.27 | rajiv | small section |
| 16:09.35 | Nugget | "O'Reilly Books in a Nutshell" |
| 16:09.45 | asterboy | LostFrog, lol...ya its a sea of text |
| 16:09.50 | rajiv | i dont think they have a book on just sip though |
| 16:12.23 | chapeaurouge | Switching to VoIP from oreilly also talks about SIP |
| 16:12.58 | asterboy | Anyone here from Canada know how to dial a UK number? |
| 16:13.17 | zoa | +44 i think |
| 16:13.37 | a_phantasm | Is there a chan for AAH assistance? |
| 16:13.58 | Zeeek | there's a forum |
| 16:14.14 | a_phantasm | Going through the forum as I type |
| 16:14.21 | Zeeek | http://www.voip-info.org/tiki-index.php?page=Asterisk+at++Home |
| 16:15.51 | *** join/#asterisk scanna (n=scannach@81-174-16-211.f5.ngi.it) |
| 16:16.08 | a_phantasm | yeah, nothing at that page addressing the problem... |
| 16:16.13 | crash3m | can anyone tell me if its possible to force only one polycom (e.g. ignoring sip.ver) to user a certain version of firmware and/or bootrom? |
| 16:16.24 | Zeeek | a_phantasm what problem? |
| 16:16.51 | a_phantasm | If I call any of my extensions, it says they are on the phone and dumps me to vm |
| 16:16.52 | Zeeek | crash3m maybe use a different home directory? |
| 16:17.08 | crash3m | hmmmm |
| 16:17.12 | Zeeek | what does the CLI say? |
| 16:18.08 | a_phantasm | Nov 23 11:17:32 VERBOSE[2210]: -- Executing AGI("SIP/1337-71b6", "dialparties.agi") in new stack |
| 16:18.08 | a_phantasm | Nov 23 11:17:32 VERBOSE[2210]: -- Launched AGI Script /var/lib/asterisk/agi-bin/dialparties.agi |
| 16:18.08 | a_phantasm | Nov 23 11:17:32 VERBOSE[2210]: -- AGI Script dialparties.agi completed, returning 0 |
| 16:18.08 | a_phantasm | Nov 23 11:17:32 DEBUG[2210]: Launching 'NoOp' |
| 16:18.08 | a_phantasm | Nov 23 11:17:32 VERBOSE[2210]: -- Executing NoOp("SIP/1337-71b6", "Returned from dialparties with no extensions to call") in new stack |
| 16:18.10 | *** join/#asterisk gniretar (n=mark@152.160.35.1) |
| 16:18.13 | a_phantasm | I think that is where the issue is |
| 16:18.15 | LostFrog | ~pb? |
| 16:18.18 | jbot | rumour has it, pastebin is a place to paste your stuff without flooding the channel - try http://pastebin.ca/ |
| 16:18.19 | file | you're using AMP... |
| 16:18.46 | Zeeek | you can't even dial a number without AGI in that stuff? |
| 16:18.58 | gniretar | hey guys |
| 16:19.02 | gniretar | quick question |
| 16:19.05 | Zeeek | that kinda sucks |
| 16:19.14 | gniretar | In extenstions.conf |
| 16:19.18 | a_phantasm | http://pastebin.ca/30611 |
| 16:19.29 | gniretar | if a call comes in and doesnt match any extensions in the context |
| 16:19.32 | a_phantasm | thats the output from when I make a call |
| 16:19.43 | gniretar | how to write a line to handle it |
| 16:19.51 | Zeeek | a_phantasm I think you are indeed going to need to go the @hole forum |
| 16:19.53 | gniretar | like the else in a lonf if statement |
| 16:20.06 | oogle | what's the name of that file in /etc/ that tells apps where to look for libraries? |
| 16:20.16 | a_phantasm | I'm thinking about installing slack and going about this the right way |
| 16:20.20 | *** part/#asterisk santoshr (i=1063@203.199.110.93) |
| 16:20.21 | Zeeek | gniretar you want a fallthrough default behavior? |
| 16:20.32 | Zeeek | a_phantasm I'd applaud that decision |
| 16:20.35 | gniretar | Zeeek: yes |
| 16:20.37 | Zeeek | ~Zeeek |
| 16:20.39 | jbot | you are probably someone who once said "learning asterisk using a GUI is like learning sex through masturbation. You'll never get to the good stuff." |
| 16:20.40 | Katty | file: yes pri cards. |
| 16:20.45 | chapeaurouge | oogle, /etc/ld.so.conf ? |
| 16:20.47 | Katty | file: never used them before.. |
| 16:20.53 | oogle | chapeaurouge: thanks! |
| 16:20.58 | `Sauron | Meow :) |
| 16:21.05 | Katty | `Sauron: mew. |
| 16:21.18 | a_phantasm | Zeeek, I agree, but had to try to keep this system idiot proof for the other people around here... Oh well :o) |
| 16:21.39 | Zeeek | my opinion about those is that in the end, you're sorry you installed it |
| 16:22.32 | *** join/#asterisk brookshire (n=nubb@gateway.digium.com) |
| 16:22.34 | bbz | Would anyone here be interested in some Polycom 501 phones? We got stuck w/ an extra 20 after a project, and we need to unload them at cost, -- if anyone is interested send me a private message (if i dont respond immediately, leave me your email address, and i can send you info) i have a reference in the channel if you need. -- |
| 16:22.54 | Zeeek | gniretar you can make ANY sequence of at least 2 digits fire like this: exten => _X.,1,DoSomething() |
| 16:23.24 | file | bbz: deja dejavu |
| 16:23.27 | file | brookshire: Mattttttttt |
| 16:23.28 | gniretar | Zeeek: If i put that at the end will it only be triggered if nothing preceeding is matched? |
| 16:23.35 | file | Katty: they're not that bad, there's documentation and stuffs... |
| 16:23.40 | Zeeek | it's a little more complicated than that |
| 16:23.51 | Zeeek | gniretar - you'll need to actually read the wiki |
| 16:24.15 | gniretar | Zeeek: Glad to do that, what topic do ytou recommend? |
| 16:24.20 | Zeeek | gniretar I'll give a like in a sec |
| 16:25.13 | Zeeek | Start here: http://www.voip-info.org/wiki/index.php?page=Asterisk+Dialplan+Introduction |
| 16:25.22 | Katty | file: i'm not so good at documentations. |
| 16:25.34 | Katty | file: they confuzzle me to pieces with their big words and fuzzy information. |
| 16:26.06 | brookshire | hey file! |
| 16:26.06 | Zeeek | gniretar here's the page you may want: http://www.voip-info.org/wiki/index.php?page=Asterisk+Extension+Matching |
| 16:26.15 | LostFrog | fuzzy information? |
| 16:26.22 | Katty | brookshire: ! |
| 16:26.28 | Katty | brookshire: there is a road out here named brookshire :> |
| 16:26.39 | Katty | brookshire: two blocks from my flat :> |
| 16:26.46 | Zeeek | gniretar and the final link is..... |
| 16:26.49 | Zeeek | http://www.voip-info.org/wiki/index.php?page=Asterisk+config+extensions.conf+sorting |
| 16:26.52 | Zeeek | <PROTECTED> |
| 16:27.03 | Zeeek | damned ech |
| 16:27.33 | file | tonight is the nightfall, I love you! |
| 16:27.35 | file | ooooooooooh |
| 16:27.45 | file | brookshire: I was wondering yesterday where you went to |
| 16:29.21 | brookshire | katty: that is awesome! |
| 16:29.23 | brookshire | i need pics |
| 16:29.27 | brookshire | :D |
| 16:29.37 | brookshire | file: i went crazy |
| 16:29.51 | file | brookshire: but you were already crazy |
| 16:29.55 | file | did you go crazier? |
| 16:30.02 | brookshire | i guess i just realized it |
| 16:30.12 | file | well that's cute |
| 16:30.59 | Katty | brookshire: k, i'll get you one |
| 16:31.04 | Katty | brookshire: on the way back from lunch |
| 16:31.17 | docelmo | w00t! |
| 16:31.19 | docelmo | FRIDAY! |
| 16:31.23 | Katty | docelmo: wednesday |
| 16:31.30 | chapeaurouge | lol |
| 16:31.37 | docelmo | No.. FRIDAY! I dont work the rest of the week |
| 16:31.44 | *** join/#asterisk BhaalWK (i=bhaal@freenode/staff/bhaal) |
| 16:31.52 | docelmo | Well @ one job.. I still have to work for my clients.. :( |
| 16:31.59 | *** join/#asterisk saftsack (n=oliver@p54A7F2A1.dip.t-dialin.net) |
| 16:32.03 | docelmo | Its what pays for my toys.. |
| 16:32.04 | saftsack | hi, asterisk says me didn't find capi device with controller = 1 or group = 0. but i configured controller=1 in capi.conf |
| 16:32.19 | docelmo | Which btw, anyone interested in a X10 interface for asterisk to control their house? |
| 16:32.24 | docelmo | I am thinking about making one |
| 16:32.54 | brookshire | docelmo: i think mark has one |
| 16:33.03 | brookshire | kram is all about some x10 |
| 16:33.06 | Pete_Largo | he would |
| 16:33.22 | docelmo | hehe.. I will have to get with him about it. |
| 16:33.31 | [TK]D-Fender | docelmo : You talking about the AH2 USB one? |
| 16:33.36 | docelmo | I am looking to buy the new X10 platform they have and intergrate it in Asterisk |
| 16:33.37 | docelmo | yes |
| 16:34.04 | brookshire | definately.. i know for sure he has some cgi scripts for x10 |
| 16:34.18 | saftsack | Nov 23 17:34:54 NOTICE[14181]: chan_capi.c:3271 load_module: Unused contr1 |
| 16:34.32 | docelmo | I am talking asterisk intergrated.. So you can call it directly from the dialplan in a conext |
| 16:34.38 | docelmo | Not CGI |
| 16:34.42 | [TK]D-Fender | docelmo : I recall discussing it with you a few weeks ago. I'm still running a CM11A (serial 1st gen) on with HEYU2 to crontl things through *. How is support for the new unit going? I'm still waiting before buying... |
| 16:34.54 | brookshire | mark cgi = c normally |
| 16:34.58 | brookshire | hehe |
| 16:35.23 | docelmo | I found some drivers for it. I am gonna start coding a wrapper for asterisk in the near future I am waiting for my X10 gear to arrive |
| 16:35.55 | gniretar | zeeek: Thanks, what i wanted was an i extention |
| 16:35.58 | [TK]D-Fender | docelmo : I'm hoping to bring Sphinx into the picture and then start having some real fun :) |
| 16:36.01 | *** join/#asterisk wunderkin (i=kev@12-201-105-27.client.mchsi.com) |
| 16:36.08 | docelmo | sphinx? |
| 16:36.21 | file | voice recognition |
| 16:36.28 | [TK]D-Fender | Yeah for voice recognition control of X-10 through * |
| 16:37.06 | brookshire | haha |
| 16:37.41 | brookshire | "ILLUMINATE" "DELUMINATE" |
| 16:37.57 | docelmo | hehe |
| 16:37.58 | [TK]D-Fender | I was working on an "AI" in VB that you would use to actually describe your Z-10 layout to by voice and then allow control of it after. |
| 16:38.11 | brookshire | i forget what movie that is from |
| 16:38.12 | brookshire | :/ |
| 16:38.13 | *** join/#asterisk bweschke (n=bweschke@pcp09754274pcs.narlington.nj.comcast.net) |
| 16:38.24 | [TK]D-Fender | brookshire : I think those 2 choices might lead to VR mis-reads :) |
| 16:38.53 | LostFrog | Clap on, Clap off |
| 16:38.58 | brookshire | [tk]: probably so.. it was just from a movie |
| 16:39.01 | docelmo | X10 CLAPPER! |
| 16:39.06 | asterboy | ok, so I have a UK number 44-87-04906549...do I have to add anything? |
| 16:39.07 | [TK]D-Fender | "ON", "OFF", "LOW "MEDIUM", "HIGH" work well. |
| 16:39.26 | brookshire | doesn't it usually pay attention to the first part ? |
| 16:39.34 | brookshire | D and I sound different |
| 16:39.36 | *** join/#asterisk DannyF (n=dannyf@c-5e4fe353.24-0099-74657210.cust.bredbandsbolaget.se) |
| 16:39.47 | [TK]D-Fender | docelmo : Clapper si doable with the Universal module in relay mode :) |
| 16:39.58 | asterboy | Do I put a 11 in front of international numbers? |
| 16:40.12 | kippi | I am having a small problem, I had a echo problem and I changed the zaptel config and it fixed it, but now when you talk really loud down the phone you get echo and some feed back, any ideas? |
| 16:40.14 | [TK]D-Fender | brookshire : Sure to you and me... tell that to OSS Sphinx ;) |
| 16:40.31 | denon | kippi: turn down your volume :) |
| 16:40.42 | asterboy | kippi, don't talk loud on the phone. |
| 16:40.45 | asterboy | :P |
| 16:40.58 | docelmo | true.. |
| 16:41.02 | kippi | any reason why asterisk is doing this? |
| 16:41.09 | asterboy | feedback |
| 16:41.25 | asterboy | try some better phone equipment |
| 16:41.26 | docelmo | I dont know.. I am planning to kick it out before long.. Once I get it fairly stable I plan to try and add it to cvs as an option |
| 16:41.49 | kippi | what handset equipment? |
| 16:41.49 | asterboy | There are a lot of options to counter echo. |
| 16:41.59 | asterboy | polycom |
| 16:42.02 | asterboy | cisco |
| 16:42.16 | [TK]D-Fender | Well I'll have to get back into X-10 once I get my * server back up and running at home. I just printed all the docs I should need to set up my new ADSL card so I think I'm set. |
| 16:42.37 | [TK]D-Fender | Just need to sit down and do it. |
| 16:42.43 | docelmo | adsl card? |
| 16:42.44 | asterboy | If you equipment has a golden sticker at the bottom labled "Made in Taiwan"...ummm don;t expect great quality. |
| 16:43.01 | [TK]D-Fender | docelmo : Sangoma S518 ADSL PCI card. |
| 16:43.12 | docelmo | Your shoving it in your asterisk box? |
| 16:43.35 | [TK]D-Fender | docelmo : So I can do away with my PPPoE modem, extra NIC, Power bloack, etc... |
| 16:43.39 | [TK]D-Fender | docelmo : yup! |
| 16:43.43 | saftsack | is it possible to get a hfc card for 50bucks? |
| 16:43.43 | kippi | docelmo: its ISDN card |
| 16:43.45 | docelmo | nice.. |
| 16:44.04 | docelmo | I got rid of FIOS PPPoE it suck! |
| 16:44.06 | [TK]D-Fender | MY * box is my "super server". Does it all in 1 box. No NAT issues or anything... |
| 16:44.07 | docelmo | sucks! |
| 16:44.07 | asterboy | So in north America, we dial 1 and area code...what do you do for international? |
| 16:44.16 | asterboy | 011? then whatever? |
| 16:44.16 | [TK]D-Fender | FIOS? |
| 16:45.26 | asterboy | no one here dials international? |
| 16:45.46 | Zeeek | 011 + number |
| 16:45.59 | Zeeek | as in 011 331 4543524 |
| 16:46.00 | [TK]D-Fender | asterboy : rarely, but from north america you are looking to prefix with 011 |
| 16:46.20 | docelmo | [TK]D-Fender, I have Verzon Fios 30/5 internet at my house |
| 16:46.22 | asterboy | ok that seems to work. |
| 16:46.33 | asterboy | just not sure of he number format. |
| 16:46.34 | [TK]D-Fender | docelmo : ADSL modem? |
| 16:46.40 | docelmo | Nope.. Fiber |
| 16:46.43 | Zeeek | if you use voiptalk it's just the country code + number |
| 16:46.44 | [TK]D-Fender | :O |
| 16:46.57 | docelmo | I have 30Mb down and 5Mb up |
| 16:46.59 | [TK]D-Fender | What kind of bandwidth? |
| 16:47.13 | [TK]D-Fender | EEK! |
| 16:47.23 | [TK]D-Fender | OMG, how much does it cost? |
| 16:47.24 | docelmo | ya takes me bout an hour to download shit.. |
| 16:47.36 | docelmo | Im paying 150ish I think |
| 16:47.36 | Zeeek | Fibre to the Home (FTH) 100M up/down |
| 16:47.54 | [TK]D-Fender | define "hour to download shit". |
| 16:48.20 | *** join/#asterisk Corydon-w (i=red@pdpc/supporter/sustaining/Corydon76-home) |
| 16:48.21 | docelmo | Bit Torrent Movies.. Takes about 10 seconds to download Asterisk from CVS w/ addons and sounds |
| 16:48.29 | [TK]D-Fender | wow |
| 16:48.51 | [TK]D-Fender | I'm just happy for my 3mbit/800kbit service |
| 16:48.56 | docelmo | hehe.. |
| 16:49.01 | asterboy | found a good resource on international number dialling conventins: |
| 16:49.03 | asterboy | http://paulm.com/how_to/write_a_phone_number.html |
| 16:49.10 | docelmo | I dont even live near the city.. I live out in the sticks |
| 16:49.13 | [TK]D-Fender | I'm not a leech (I have friends for that :D) so I don't need much. |
| 16:49.16 | *** join/#asterisk fugitivo (n=ajf@209.13.243.47) |
| 16:49.25 | *** join/#asterisk pbd (n=plancomm@12.144.118.36) |
| 16:49.36 | pbd | Good day, all. |
| 16:50.27 | tzafrir_laptop | good evening |
| 16:50.31 | pbd | Anyone out there ever play with the idea of serving up voicemail using pop or imap, a la unified messaging? |
| 16:51.03 | Chotaire | ehrm.. do we have rfc2833 problems with 1.0.10 also? |
| 16:51.23 | morale | whats rfc2833? |
| 16:51.37 | morale | rtp payload.. |
| 16:51.42 | gniretar | i'm getting massive amounts of '!! Got reject for Frame X retransmitting frame X now, updating n_r' |
| 16:51.49 | gniretar | anyone know where this could be coming from? |
| 16:51.53 | file | morale: 101 for asterisk |
| 16:53.09 | darkskiez | can i dial in via a TE110P card ? |
| 16:54.11 | *** join/#asterisk marv[work] (n=timr@border0hsv.asterisksgi.com) |
| 16:54.26 | Chotaire | can anyone assist me with a 1.0.10 problem with dmtfmode rfc2833? |
| 16:54.46 | Chotaire | it seems dtmfmode=rfc2833 in sip.conf will no longer work (it did in 1.0.3) |
| 16:54.57 | Chotaire | sip callers can't press any keys. |
| 16:55.33 | Chotaire | pstn->sip->* that is. |
| 16:55.35 | *** join/#asterisk marv[work] (n=timr@border0hsv.asterisksgi.com) |
| 16:56.51 | Zeeek | Chotaire I did notice something weird about that |
| 16:57.08 | Zeeek | in 1?2 |
| 16:57.11 | Zeeek | 1.2 |
| 16:57.28 | Chotaire | for 1.2.*, see topic. |
| 16:57.37 | Zeeek | what topic? |
| 16:57.44 | Chotaire | however, I run 1.0.10, users can press keys until they die, nothing happens. |
| 16:57.49 | Chotaire | I have no other choice than rfc2833. |
| 16:58.07 | Chotaire | * Topic is 'Asterisk 1.2.0 has been released! -- ftp.digium.com (RFC2833 users see bug 5780) || http://www.asterisk.org' |
| 16:58.23 | Zeeek | ah shit I didn't see it! :) |
| 16:58.50 | Chotaire | someone please help me with this shit, otherwise I will dowgrade to 1.0.3 once again and for good. |
| 17:02.44 | LostFrog | Damn.. you know when I finally do move to 1.2, I will forget to apply that patch. |
| 17:02.55 | LostFrog | Oh wait.. that's the g729 patch I'm thinking of. |
| 17:03.01 | Zeeek | that was easy - now let's see if it worked :) |
| 17:03.06 | *** join/#asterisk Sedorox (i=brandon@smartserv/cna/Sedorox) |
| 17:03.09 | Chotaire | 1.2 came with so many surprises, I downgraded right away. |
| 17:03.29 | Chotaire | chan_capi goes berzerk on 1.2 |
| 17:03.48 | *** join/#asterisk axscode (n=paranoid@203.213.217.123) |
| 17:03.54 | Zeeek | that's funny, now vmail doesn't hear the tones... |
| 17:03.55 | axscode | anyone running asterisk on CentOS? |
| 17:04.11 | Chotaire | zeeek: cool.. not. |
| 17:04.13 | axscode | i downloaded the kernel source.. where should it be placed.. |
| 17:04.22 | Chotaire | omg |
| 17:05.11 | Chotaire | what about /usr/src/linux ? |
| 17:05.18 | axscode | ive done that |
| 17:05.26 | Chotaire | good. |
| 17:05.47 | axscode | /usr/src/linux/2.6.9-22.ELsmp |
| 17:05.56 | axscode | but zaptel is looking for it still |
| 17:05.58 | Chotaire | no, it's /usr/src/linux |
| 17:06.35 | Chotaire | mv /usr/src/linux/2.6.9-22.ELsmp/* /usr/src/linux/ ; rmdir /usr/src/linux/2.6.9-22.ELsmp |
| 17:06.58 | Zeeek | Chotaire - no it works again now that I moved back from inband to rfc. This was how it was in 1.0.9 |
| 17:07.09 | Chotaire | yes, inband is a bitch. |
| 17:07.11 | [TK]D-Fender | or you could try just symlinking it.... |
| 17:07.21 | ikarus | Chotaire: chan_capi and one of the other ISDN channels iirc are known to be buggy with 1.2, annoyingly |
| 17:07.24 | *** join/#asterisk Cresl1n (n=matt@gateway.digium.com) |
| 17:07.24 | Zeeek | but I am heading tones in the polycom instead of them being muted and heard by asterisk |
| 17:07.39 | Chotaire | well, I am ok with 1.2 not working for the m oment... kapejod seems still not done with an update. |
| 17:07.43 | axscode | Chotaire: its not workin |
| 17:07.51 | Chotaire | but still, I want rfc2833 to be working in 1.0.10, this can't be true. |
| 17:07.54 | *** join/#asterisk UlbabraB (n=caplaz@host241-43.pool8172.interbusiness.it) |
| 17:08.01 | axscode | You do not appear to have the sources for the 2.6.9-22.ELsmp kernel installed. |
| 17:08.16 | Chotaire | you did UNPACK the archive, right? |
| 17:08.23 | axscode | drwxr-xr-x 18 root root 4096 Nov 24 01:02 linux |
| 17:08.28 | axscode | of course.. |
| 17:08.47 | Zeeek | Chotaire it does work after the patch - one line to add |
| 17:08.50 | *** part/#asterisk darkskiez (n=darkskie@194.247.78.146) |
| 17:09.01 | Chotaire | directory /usr/src/linux/include exists? |
| 17:09.08 | Chotaire | Zeeek: in 1.0.10? |
| 17:09.20 | Zeeek | I just called a provider and checked vmail. Tones are detected |
| 17:09.28 | axscode | drwxr-xr-x 15 root root 4096 Nov 23 22:28 include |
| 17:09.30 | Zeeek | oops I misread your version , sorry |
| 17:09.45 | Zeeek | it's not the same bug? |
| 17:10.32 | Zeeek | any Polycom users out there? I don't see where to configure the DTMF ? |
| 17:10.49 | tzafrir_laptop | make KSRC=/path/to/kernel/source |
| 17:12.06 | Chotaire | zeeek: there is not even a line that contains annexb in sip.conf |
| 17:12.10 | Chotaire | ehrm chan_sip.c |
| 17:12.29 | Zeeek | ah. That makes it a little harder |
| 17:14.14 | axscode | [root@voip zaptel-1.2.0]# make KSRC=/usr/local/linux/ |
| 17:14.32 | axscode | /usr/local/linux/ |
| 17:14.32 | axscode | You do not appear to have the sources for the 2.6.9-22.ELsmp kernel installed. |
| 17:14.32 | axscode | make: *** [linux26] Error 1 |
| 17:15.20 | *** join/#asterisk litb (n=litb@p54B19007.dip0.t-ipconnect.de) |
| 17:15.27 | litb | what is a sip uri? |
| 17:15.33 | Chotaire | Zeeek: any idea? |
| 17:15.37 | litb | is that my voip number? |
| 17:15.48 | *** join/#asterisk UlbabraB (n=caplaz@host241-43.pool8172.interbusiness.it) |
| 17:22.06 | Zeeek | Chotaire I'm trying to get the features.conf stuff working here - it depended on DTMF obviously |
| 17:22.21 | docelmo | litb a sip URI is the following sip:107@sip.planet-telecom.com |
| 17:22.31 | docelmo | Which is my SIP URI for my office |
| 17:22.46 | Zeeek | seems like there's still some weirdness, I wonder if transfers are best handled by decent phones |
| 17:23.55 | *** part/#asterisk frenzy (n=frenzy@193.220.82.108) |
| 17:24.16 | litb | my VOIP number is "06175309xxx". how can i find out what a sip uri it has? (sorry, i'm a noob) |
| 17:24.24 | *** join/#asterisk h4mm3r` (n=h4mm3r@85-18-14-10.fastres.net) |
| 17:24.58 | litb | i want to use kcall with that phone number |
| 17:26.14 | *** join/#asterisk A-Tuin|work (n=A-Tuin@nat.office.legend.net.uk) |
| 17:28.21 | Chotaire | nice bug hell. |
| 17:28.42 | Chotaire | if I call someone *=>SIP=>PSTN then they can press dialtones and it will properly be detected. |
| 17:28.55 | Chotaire | if someone calls me PSTN=>SIP=>* then dialtones will be undetected. |
| 17:29.00 | Chotaire | this behaviour happens on asterisk 1.0.10 |
| 17:29.03 | Chotaire | anyone has an idea? |
| 17:29.33 | viperdude | Chotaire: do you mean DTMF? |
| 17:29.43 | saftsack | what are the cheapest isdn cards with nt mode? |
| 17:29.44 | Zeeek | what is supposed to happen when they hit tones? Where are youy in the dialplan? |
| 17:30.05 | Zeeek | DISA, read? |
| 17:30.30 | viperdude | check dtmfmode in sip.conf |
| 17:30.30 | chapeaurouge | what's a good and cheap ATA for Europe? |
| 17:30.45 | *** join/#asterisk escribzz (n=escribzz@h-64-105-157-50.phndaz91.covad.net) |
| 17:31.09 | Chotaire | viperdude: yup |
| 17:31.15 | Zeeek | didn't the linksys work out? |
| 17:31.16 | Chotaire | I did check dtmfmode in sip.conf |
| 17:31.24 | Chotaire | dtmfmode=rfc2833 |
| 17:31.25 | Chotaire | as it should be. |
| 17:31.33 | Chotaire | and there is no alternative except downgrading. |
| 17:32.11 | Zeeek | hmmmm I haven't tried my DISA stuff yet in 1.2 |
| 17:32.51 | Zeeek | my home phone bill for the last two months was 0.00 in calls. They al went through the office pbx :) |
| 17:33.03 | Chotaire | omg... |
| 17:33.11 | Chotaire | anyone knows if this bug exists in 1.0.9 also? |
| 17:33.15 | docelmo | Mine has been that way for a year |
| 17:33.22 | Zeeek | I didn't have the problem in 1.0.9 |
| 17:33.24 | docelmo | I have a DID and Termination from work |
| 17:33.29 | Chotaire | Zeeek: did you try 1.0.10?` |
| 17:33.41 | Zeeek | never since I went right to 1.2 |
| 17:33.58 | Chotaire | I dont see any changes regarding dtmf in 1.0.10 |
| 17:34.00 | Zeeek | all was well in 1.0.9 |
| 17:34.13 | Chotaire | and the last change was in 1.0.4 (enable outbound dtmf) |
| 17:34.25 | Zeeek | an undocumented feature? |
| 17:34.37 | litb | ok, 1und1 sip server is sip.1und1.de. but where can i get the chars before @? |
| 17:35.20 | Chotaire | 491234567890:password@sip1.und1.de |
| 17:35.23 | escribzz | Hey does anyone have a crash on zap with chan_zap.so with ulimit set to unlimited? Its crashing my system. |
| 17:35.32 | razu | is there any way to use moh without makeing a queue ? |
| 17:36.05 | demetrio | razu: yes, put the call on hold. |
| 17:37.16 | razu | how can i define it in extension.conf ? |
| 17:37.45 | Chotaire | you put the call on hold ;) |
| 17:38.39 | Chotaire | great shit.. downgrading to 1.0.3 once again. |
| 17:38.49 | Zeeek | why not 1.0.9? |
| 17:38.54 | Chotaire | I dont even bother trying. |
| 17:39.05 | Chotaire | if shit is not documented... |
| 17:39.07 | Zeeek | or 1.2? |
| 17:39.13 | Chotaire | 1.2 doesnt work with chan_capi |
| 17:39.18 | Zeeek | ah. bad |
| 17:39.35 | Chotaire | 1.0.3 was working all properly except one problems with the "reply" feature in voicemail |
| 17:39.58 | Chotaire | when I hit reply, nothing happens or mail gets sent to a sip phonenumber that is defined somewhere in sip.conf (duh!) |
| 17:41.17 | Chotaire | oh, and password changing also doesn't work. |
| 17:43.52 | *** join/#asterisk mkrufky (n=mk@68.160.103.77) |
| 17:44.47 | fugitivo | i have a strange problem with IAX + jitterbuffer |
| 17:45.04 | fugitivo | if jitterbuffer is on, asterisk won't detect when a iax phone is picking a call |
| 17:45.29 | Chotaire | I'm gonna try 1.2.0 again now and see if I can temporarily live without chan_capi.. however, I only can do that if all other features are free of bugs. |
| 17:46.51 | *** join/#asterisk Gunnar (n=gunnar@34.80-203-209.nextgentel.com) |
| 17:47.04 | Zeeek | FWIW I just tried a pstn callin and DISA (hence tone detection) works fine in 1.2 |
| 17:47.20 | Zeeek | I can't remember what capi is |
| 17:48.54 | LostFrog | ISDN |
| 17:49.00 | LostFrog | If I remember correctly. |
| 17:49.21 | Zeeek | DIVAS |
| 17:49.45 | Chotaire | zeeek: ok, will try. |
| 17:49.55 | Chotaire | without chan_capi I cannot take isdn calls or dial out to pstn though |
| 17:50.04 | Chotaire | i hope kapejod fixes this stuff asap. |
| 17:50.13 | Zeeek | yeah if you need that it won't be any good |
| 17:50.28 | Chotaire | for 1-2 weeks, I can live without it. |
| 17:50.34 | Chotaire | but not any longer. |
| 17:50.41 | Zeeek | you have analog lines as well? |
| 17:50.44 | Chotaire | nope |
| 17:50.56 | Zeeek | you'll have no phone except SIP ? |
| 17:51.00 | Chotaire | yup |
| 17:51.02 | Zeeek | OMG |
| 17:51.10 | Chotaire | true |
| 17:51.20 | Zeeek | well, these days we all have cells I guess |
| 17:51.23 | Chotaire | I have a converter that can make an isdn line become analog. |
| 17:51.30 | Zeeek | dirty! |
| 17:51.43 | Chotaire | and then? use a frickin voice modem? how lame is this? |
| 17:51.45 | Zeeek | funny we had ISDN for a year or so and I hated it |
| 17:51.50 | Chotaire | isdn rocks |
| 17:52.29 | Zeeek | but no native driver? or is that just the chosen hardware? |
| 17:52.30 | Chotaire | ok, going 1.2.0 now, let's see what happens. |
| 17:52.39 | Chotaire | I dont have any cards for doing analog lines. |
| 17:53.04 | Chotaire | but I have a hardware pbx that I can use to make isdn calls go to asterisk, and that's what I will be trying. |
| 17:53.27 | Chotaire | I hate updating asterisk in general. |
| 17:53.29 | fugitivo | so, nobody is having a IAX+jitterbuffer problem |
| 17:53.50 | Zeeek | as soon as you have a couple patches that need to be made every time, it's a pain, yes |
| 17:54.02 | Zeeek | fugitivo not using jitterbuffer at the mo |
| 17:54.05 | Chotaire | yup, my 1.0.3 is patched to hell. |
| 17:54.06 | *** join/#asterisk dwmw2_gone (i=ctrlprox@baythorne.infradead.org) |
| 17:54.15 | fugitivo | i had to disable it |
| 17:54.19 | fugitivo | maybe it's a bug |
| 17:54.22 | Zeeek | I'm already trying to remember the patches |
| 17:54.39 | Zeeek | one for changing the ring frequency so my Seimens phones ring |
| 17:54.50 | Zeeek | some screwing around with rxfax |
| 17:55.23 | Zeeek | Has anyone used the new hardware-specific g729 module yet? |
| 17:55.25 | Chotaire | i dont even connect my siemens to * |
| 17:55.28 | LostFrog | Based on what everyone has said in here, I'm not even going to try to use rxfax/spandsp unless I have real problems with HylaFax. |
| 17:55.32 | Chotaire | i know what would happen: headache |
| 17:55.34 | Zeeek | why not, they don't ring ? |
| 17:55.42 | _DAW | does anyone know if it is possible to set alert_info value with the page command? |
| 17:55.59 | Chotaire | the siemens phone I got is known to bug around with both * and my hardware pbx |
| 17:56.08 | Chotaire | so I connect it directly to isdn instead. |
| 17:56.09 | Zeeek | What bug? |
| 17:56.18 | Zeeek | it doesn't ring? |
| 17:56.24 | Chotaire | I dont even remember ;) I might try with 1.2.0 and see if it works |
| 17:56.27 | Chotaire | been a while |
| 17:56.39 | Zeeek | I need new phones anyway, none of the old cordless have CID |
| 17:56.56 | Chotaire | I love my panasonic analog business phones. |
| 17:57.01 | Chotaire | oldskool that works the way it should |
| 17:57.08 | Zeeek | I've always heard panasonic were decent phones |
| 17:57.29 | Zeeek | I'm getting to hate the over high tech of new cordless (and SIP) phones |
| 17:57.42 | Zeeek | the ones where you CAN'T diable missed call lists? |
| 17:58.00 | fugitivo | Zeeek: i have a cordless siemens |
| 17:58.02 | Zeeek | when you have three phones, it isn't necessary to have three leds blinking all day |
| 17:58.13 | Chotaire | ;) |
| 17:58.24 | Zeeek | fugitivo oh yeah? Does it work well with * ? |
| 17:58.28 | fugitivo | Zeeek: with callid, message waiting indicator, and a lot of stuff |
| 17:58.30 | Chotaire | my panasonic phones dont even have CID, but hey... what do I have * for. |
| 17:58.39 | Zeeek | mine are very old, none of the LCD work right anymore |
| 17:58.42 | fugitivo | Zeeek: it works better than my crappy atcom ip phone |
| 17:58.46 | Chotaire | I'Ll make it read the caller ids if I need them. |
| 17:59.11 | Zeeek | those atcom are bottom of the line phones |
| 17:59.24 | fugitivo | are crap |
| 17:59.39 | fugitivo | the siemens works perfect with * |
| 17:59.39 | Zeeek | they have the advantage of doing IAX if you want that |
| 17:59.53 | Zeeek | connecte don FXS? |
| 18:00.04 | fugitivo | using pap2-na |
| 18:00.14 | fugitivo | in my case |
| 18:00.15 | [TK]D-Fender | Hey, the RF 2833 bug # doesn't appear in Manti from what I can see.... is it resolved? |
| 18:00.25 | *** join/#asterisk pr0m (n=pr0methe@166.170.17.144) |
| 18:00.29 | Zeeek | no it was there twenty minutes ago |
| 18:01.16 | Zeeek | <PROTECTED> |
| 18:01.24 | *** join/#asterisk stkn (n=stkn@gentoo/developer/pdpc.active.stkn) |
| 18:01.27 | Zeeek | one line fix |
| 18:02.06 | Zeeek | as the 18 year old beautiful girl said about sex, "I tried it and just loved it!" |
| 18:02.29 | *** join/#asterisk loick (n=loic@APuteaux-151-1-6-121.w82-120.abo.wanadoo.fr) |
| 18:02.33 | *** join/#asterisk DarthClue (i=DarthClu@AC874BD0.ipt.aol.com) |
| 18:02.37 | Zeeek | adding a line to chan_sip.c may give you less of a thrill though - YMMV |
| 18:02.50 | [TK]D-Fender | Zeeek : Is it in CVS-HEAD right now then? |
| 18:03.00 | *** join/#asterisk Frosted (n=Procrast@buzzbud.plus.com) |
| 18:03.05 | Zeeek | dunno - I don't follow HEAD |
| 18:03.26 | Zeeek | I don't usually do the patches either but this was an annoying prob and the patch simple |
| 18:04.49 | Zeeek | I haven't been around here at this houir for a while - seems like there are a lot less people |
| 18:05.06 | Zeeek | also, in the morning when I am around, there are a lot fewer questions |
| 18:05.09 | [TK]D-Fender | ...OR I could just read further down the page to see the big print that its in HEAD now :0 |
| 18:05.18 | Zeeek | does this mean asterisk is hitting a stride rhythm? |
| 18:05.33 | Zeeek | [TK]D-Fender heh |
| 18:05.47 | syzygyBSD | or people are able to find answers to their questions online |
| 18:05.54 | Zeeek | I'm always afraid they brioke something new in HEAD |
| 18:06.02 | *** join/#asterisk dwmw2_gone_ (i=ctrlprox@baythorne.infradead.org) |
| 18:06.21 | *** join/#asterisk justinu (n=j2@72.18.13.48) |
| 18:06.30 | [TK]D-Fender | Zeeek : They DO... but I like having only the most up to date bugs :) No second-hand crashing for me! |
| 18:06.32 | Zeeek | yeah well afetr repeating "Check these documents:" macros thousands of times, that's a GoodThing |
| 18:06.55 | justinu | ~seen oej |
| 18:07.05 | jbot | oej <n=Olle@apollo.webway.se> was last seen on IRC in channel #asterisk, 2h 18m 2s ago, saying: 'n+1 is *not* busy and the jumping is deprecated, using DIALSTATUS is much better'. |
| 18:07.05 | Zeeek | shit I remember when Manxpower used his doc macro a hundred times a day |
| 18:08.15 | Chotaire | zeeek: same shitty bug in 1.2.0 |
| 18:08.18 | Zeeek | the problem with multi-line SIP phones is that when they become unavailable, the messages fill the screen! |
| 18:08.42 | Zeeek | Chotaire what part of the world are you in? |
| 18:08.47 | Chotaire | oh wait.. I forgot the patch ;) |
| 18:08.57 | Zeeek | hmmmmm |
| 18:09.01 | Chotaire | .de |
| 18:09.30 | Zeeek | insert the new line and remake |
| 18:09.35 | Zeeek | you'll love it |
| 18:10.13 | *** join/#asterisk dalbjerg (n=dalbjerg@2001:16d8:ff59:0:fd11:d477:745a:2ecc) |
| 18:10.13 | Chotaire | almost done recompiling. |
| 18:10.15 | Chotaire | I'll see. |
| 18:10.49 | Zeeek | I need to find some newer code snippets. My knowledge is locked at 1.0.1 |
| 18:11.03 | Chotaire | and I thought I was a dino still working with 1.0.3 |
| 18:11.15 | Zeeek | maybe it was 1.0.6 come to think |
| 18:11.31 | Zeeek | got the nerve to move to 1.0.9 and finally a few days ago to 1.2 |
| 18:12.16 | Chotaire | ..testing... |
| 18:12.26 | Chotaire | not working. |
| 18:12.32 | Zeeek | how can that be? |
| 18:12.40 | *** join/#asterisk tld (n=tld@172.80-202-167.nextgentel.com) |
| 18:12.48 | Chotaire | it will not even detect dtmf |
| 18:13.19 | Chotaire | I call from any POTS phone to my SIP phone number and it will not detect dtmf in * |
| 18:13.23 | *** join/#asterisk oej (n=oej@apollo.webway.se) |
| 18:13.40 | Zeeek | but you go through what hardware? |
| 18:13.43 | Chotaire | if I dial out via SIP to any POTS phone number and transfer the call to the dialin context, it WILL work. |
| 18:13.45 | *** join/#asterisk dwmw2_gone (i=ctrlprox@baythorne.infradead.org) |
| 18:14.00 | Chotaire | right now I dont use ANY to make sure nothing else is buggy |
| 18:14.06 | Chotaire | and it doesnt matter anyway, 1.0.3 worked as intended. |
| 18:14.25 | Zeeek | maybe it's an old patch you forgot to apply? |
| 18:14.36 | Chotaire | what should I patch? |
| 18:14.45 | Zeeek | dunno |
| 18:14.48 | Chotaire | me neither |
| 18:14.58 | Zeeek | so pstn calls what, a DID ? |
| 18:15.23 | Chotaire | pstn calls *, * takes call, plays message, waits for dtmf |
| 18:15.41 | Zeeek | how does it reach asterisk if you have no hardware? |
| 18:15.45 | Chotaire | sip |
| 18:16.07 | Chotaire | and no, I also tried other providers. |
| 18:16.09 | Zeeek | if I were in de, I'd call what number (not the exact number but what is it?) |
| 18:16.14 | *** join/#asterisk clinthome (n=clinthom@va-chrvlle-cad1-bdgrp5-8d-209.chvlva.adelphia.net) |
| 18:16.21 | Chotaire | +49-blah |
| 18:16.29 | Zeeek | a provider |
| 18:16.49 | Chotaire | you can test it if you like and you will now |
| 18:16.51 | Chotaire | know |
| 18:16.56 | Chotaire | if you call, shit happens. |
| 18:17.01 | Chotaire | if I call you, you can do everything. |
| 18:17.16 | Chotaire | and I have no other chance than rfc2833 dtmfmode, otherwise stuff will not work as intended. |
| 18:17.19 | chapeaurouge | is astman in the addon pkg? |
| 18:18.10 | Chotaire | omg |
| 18:18.12 | Zeeek | Chotaire when a call comes in from your SIP provider, where does it go? What happens in the context? |
| 18:18.23 | Zeeek | maybe it's an issue with dial? |
| 18:18.28 | Chotaire | background() |
| 18:18.32 | Chotaire | no |
| 18:18.35 | Chotaire | there is no dial |
| 18:18.40 | Chotaire | it doesnt matter |
| 18:18.45 | Chotaire | it has nothing to do with contexts |
| 18:18.49 | Chotaire | nor with hardware |
| 18:18.50 | Zeeek | have you looked at SIP debug? |
| 18:18.50 | Chotaire | nor with providers |
| 18:19.05 | Chotaire | there is nothing to debug because it doesnt detect rfc2833 signals |
| 18:20.21 | Zeeek | let me think, wait yest I can do that test |
| 18:20.35 | Zeeek | I can call a SIP number and check if it hears tones |
| 18:20.46 | Zeeek | that would be the identical situation, right? |
| 18:21.01 | Chotaire | yup |
| 18:21.07 | Chotaire | if you use a POTS phone. |
| 18:21.08 | Zeeek | ok let me rig that up |
| 18:21.09 | Chotaire | not if you use SIP |
| 18:21.15 | Chotaire | sip phones will work |
| 18:21.20 | Zeeek | I'll spend the 0.015 eu for you |
| 18:22.39 | Chotaire | check msg |
| 18:23.11 | syzygyBSD | when recompiling the kernel what should my timer level be set to? I think someone said 1000Hz but 250Hz is the default |
| 18:23.25 | af_ | I am wondering which way I could put an irq for adapter... bios? 2.6? pnp os? |
| 18:23.27 | [TK]D-Fender | Chotaire : CVS-HEAD seems to fix the RFC-2833 so jut use that. |
| 18:23.31 | chapeaurouge | is astman in 1.2.0 ? i can't find the sucker. |
| 18:23.54 | tzafrir_laptop | yes, it's there |
| 18:23.54 | Chotaire | D-Fender: that is not the bug mentioned in topic? |
| 18:23.55 | syzygyBSD | af_: I think it has to be done in the bios |
| 18:24.07 | tzafrir_laptop | find . -name astman.c |
| 18:24.08 | Chotaire | D-Fender: that stuff in topic doesnt help. |
| 18:24.09 | af_ | syzygyBSD, no way there :( |
| 18:24.13 | [TK]D-Fender | af_ : Normally if you are trying to reserve an IRQ for a Digium card you'll do it in your MB's BIOS |
| 18:24.21 | chapeaurouge | tzafrir_laptop, it seems it didn't really compile for me.. i can see the .c file, but nothing else. |
| 18:24.27 | [TK]D-Fender | Chotaire : Yes, that bug. |
| 18:24.31 | syzygyBSD | af_: it would be per pci slot |
| 18:24.35 | Zeeek | Chotaire I'm trying the test but something weird happened, unrelated tot he problem |
| 18:24.41 | Chotaire | D-Fender: well, then there is another bug |
| 18:24.46 | fugitivo | so, nobody is having a IAX+jitterbuffer problem? |
| 18:24.50 | [TK]D-Fender | Chotaire : FIXED in CVS-HEAD, and a documented 1-line fix for whatever other version you want to patch |
| 18:24.54 | tzafrir_laptop | chapeaurouge, do you have newt headers? |
| 18:24.58 | af_ | syzygyBSD, in the irq i can choose only pnp/pci or lecacy ISA |
| 18:24.58 | chapeaurouge | hmmm |
| 18:25.00 | chapeaurouge | checkign |
| 18:25.06 | *** join/#asterisk tainted- (n=somewher@mail.k2usa.com) |
| 18:25.12 | syzygyBSD | ya, it is PCI |
| 18:25.19 | af_ | all is pci |
| 18:25.21 | syzygyBSD | What type of bios is it? |
| 18:25.23 | chapeaurouge | tzafrir_laptop, just the library.. is there a -dev? |
| 18:25.24 | af_ | award |
| 18:25.39 | tainted- | anyone here with socal PRI presence? NPA 562? |
| 18:25.39 | af_ | syzygyBSD, I guess award bois |
| 18:25.46 | chapeaurouge | tzafrir_laptop, yes there is. trying that |
| 18:25.52 | tzafrir_laptop | chapeaurouge, whatdistro? |
| 18:26.02 | chapeaurouge | sarge |
| 18:26.05 | [TK]D-Fender | af_ : You board may not offer the option to fix the IRQ. Your best bet may be to try and disable the other devices sharing IRQ's with it if they are not necessary. Al take a look at your MB manual as it may tell you what devices share IRQ's with which slots |
| 18:26.20 | tzafrir_laptop | on debian: libnewt-dev |
| 18:26.25 | chapeaurouge | yup, just installed it |
| 18:26.35 | Zeeek | Chotaire - you're right. DMTF does NOT work thru pstn->SIP |
| 18:26.44 | *** join/#asterisk tainted- (n=somewher@mail.k2usa.com) |
| 18:26.59 | Zeeek | it works thru SIP or PSTN->ZAP |
| 18:27.15 | af_ | [TK]D-Fender, of course I did, no way: irq 11 is shared between three adapter |
| 18:27.27 | syzygyBSD | af_: arn't their options for slot[1-4] irq? |
| 18:27.45 | af_ | syzygyBSD, I didn't find it |
| 18:27.54 | [TK]D-Fender | af_ : Try changing the slot its in. |
| 18:28.04 | af_ | ther is just ine slot, (two the reiser) |
| 18:28.07 | syzygyBSD | af_: it would be at the top of the page under pci pnp setup |
| 18:28.08 | tzafrir_laptop | chapeaurouge, the package from pkg-voip's SVN builds just fine for me on Sarge |
| 18:28.08 | Zeeek | Chotaire but I never tried that before so maybe it wasn't working in 1.0.9 |
| 18:28.12 | af_ | I have just two (filled) |
| 18:28.13 | tzafrir_laptop | with astman |
| 18:28.18 | chapeaurouge | tzafrir_laptop, http://pastebin.ca/30649 |
| 18:28.37 | syzygyBSD | http://www.lynxstudio.com/awardbiossetup.html last picture on that page, |
| 18:28.39 | [TK]D-Fender | af_ : You may need a new MB then.... |
| 18:29.13 | chapeaurouge | deafneuron:/opt/asterisk-1.2.0/utils# make astman 2>&1|paste2paste.pl |
| 18:29.13 | chapeaurouge | Your paste URL is: http://pastebin.ca/30649 |
| 18:29.32 | Zeeek | Chotaire ya get that? You are not alone |
| 18:30.06 | Chotaire | re |
| 18:30.13 | Chotaire | zeeek: sorry I was busy.. what did I miss=? |
| 18:30.46 | tzafrir_laptop | I don't see astman explicitly mentioned in any debian patch |
| 18:31.03 | syzygyBSD | what about implicitly |
| 18:31.07 | Chotaire | if you are referring to D-Fender, he was speaking about somethign else. |
| 18:31.29 | Chotaire | it seems I am pretty alone with my problem. |
| 18:31.34 | Zeeek | Chotaire no I said a major thing up^higher to you |
| 18:31.39 | af_ | it's new, its' an apia5000 |
| 18:31.42 | Zeeek | YOU ARE NOT ALONE |
| 18:31.45 | af_ | epia5000, pardon |
| 18:32.23 | Zeeek | Chotaire iow, I made the test and you are right, DTMF is NOT detected coming in thru PSTN->SIP |
| 18:32.31 | chapeaurouge | tzafrir_laptop, i forgot to mention im not using debs, but source from asterisk.org (1.2.0 release, not cvs) |
| 18:32.37 | Zeeek | however do I care? no |
| 18:32.44 | Chotaire | well, I do ;) |
| 18:32.47 | Chotaire | downgrading to 1.0.3 for good. |
| 18:32.49 | Chotaire | happy fixing ;) |
| 18:32.52 | Zeeek | that may have been broken for months |
| 18:33.15 | Chotaire | I rather live with broken voicemail. |
| 18:33.24 | Chotaire | all the rest does work as intended in 1.0.3+348793498734 patches |
| 18:33.26 | Zeeek | why not report the bug? |
| 18:33.32 | Chotaire | I will. |
| 18:33.36 | Zeeek | thank you |
| 18:33.37 | tzafrir_laptop | chapeaurouge, those are packages of 1.2.0 |
| 18:34.08 | Chotaire | maybe you can help me with the case that users are unable to change password in 1.0.3 voicemail nor will they be able to reply to a message properly... is reply to message context dependant or something? |
| 18:34.13 | chapeaurouge | tzafrir_laptop, ok. you want paste2paste.pl.. here: http://www.madpenguin.org/blogs/chapeaurouge/?p=92 |
| 18:34.17 | Chotaire | if I get that fixed, I can live with 1.0.3 |
| 18:34.27 | syzygyBSD | tzafrir_laptop: someone wanted to add that to jbot, dont' know if they did |
| 18:36.10 | tzafrir_laptop | jbot, paste2pastebin is a script to paste to the http://pastebin.ca from the standard input (linux/unix CLI) . Can be fetched from http://www.madpenguin.org/blogs/chapeaurouge/?p=92 |
| 18:36.12 | jbot | okay, tzafrir_laptop |
| 18:36.12 | _T3_ | anybody knows about SIP_CODEC variable????? |
| 18:36.17 | Zeeek | Chotaire I don't know anyting about that |
| 18:36.25 | chapeaurouge | oh yea :) |
| 18:36.27 | Zeeek | bye |
| 18:36.30 | *** part/#asterisk Zeeek (n=IceChat7@pdpc/supporter/active/Zeeek) |
| 18:36.34 | tzafrir_laptop | ~paste2pastebin |
| 18:36.36 | jbot | well, paste2pastebin is a script to paste to the http://pastebin.ca from the standard input (linux/unix CLI) . Can be fetched from http://www.madpenguin.org/blogs/chapeaurouge/?p=92 |
| 18:36.43 | tzafrir_laptop | bah, too long |
| 18:36.54 | tzafrir_laptop | (the name) |
| 18:36.56 | chapeaurouge | yea |
| 18:37.04 | chapeaurouge | p2bin ? |
| 18:37.10 | chapeaurouge | paste2bin |
| 18:37.24 | tzafrir_laptop | jbot, p2bin is a script to paste to the http://pastebin.ca from the standard input (linux/unix CLI) . Can be fetched from http://www.madpenguin.org/blogs/chapeaurouge/?p=92 |
| 18:37.26 | jbot | tzafrir_laptop: okay |
| 18:37.34 | tzafrir_laptop | jbot, forget paste2pastebin |
| 18:37.34 | jbot | i forgot paste2pastebin, tzafrir_laptop |
| 18:37.43 | tzafrir_laptop | ~p2bin |
| 18:37.45 | jbot | from memory, p2bin is a script to paste to the http://pastebin.ca from the standard input (linux/unix CLI) . Can be fetched from http://www.madpenguin.org/blogs/chapeaurouge/?p=92 |
| 18:38.54 | *** join/#asterisk loick (n=loick@APuteaux-151-1-6-121.w82-120.abo.wanadoo.fr) |
| 18:39.33 | *** join/#asterisk s34n (n=s34n@ip-206-159-190-125.mvdsl.com) |
| 18:39.45 | tzafrir_laptop | chapeaurouge, any reason why files from /usr/include/asterisk are used, rather than ../include/asterisk/ ? |
| 18:40.24 | chapeaurouge | tzafrir_laptop, no.. i just did a 'make astman' in that directory |
| 18:40.33 | *** part/#asterisk escribzz (n=escribzz@h-64-105-157-50.phndaz91.covad.net) |
| 18:40.39 | chapeaurouge | the Makefile is quite simple, dunno. |
| 18:41.15 | s34n | I just did a cvs update, and now any incoming call crashes asterisk |
| 18:41.26 | gniretar | dont use CVS |
| 18:41.43 | gniretar | asterisk 1.2 release FTW |
| 18:41.57 | s34n | what does that mean? |
| 18:42.03 | gniretar | for the win |
| 18:43.11 | s34n | so why not cvs? |
| 18:43.25 | *** join/#asterisk cjk (n=cjk@80.92.64.103) |
| 18:43.33 | cjk | is there a way in asterisk realtime to send the read and the write querries to different servers? |
| 18:43.41 | gniretar | because, CVS is inherantly unstable |
| 18:43.47 | gniretar | because the code is a work in progress |
| 18:43.58 | gniretar | i'm assuming your getting the latest nit the release |
| 18:44.01 | gniretar | not* |
| 18:44.06 | tzafrir_laptop | chapeaurouge, do you take astman from the directory astman or utils? |
| 18:44.13 | chapeaurouge | utils |
| 18:44.59 | chapeaurouge | tzafrir_laptop, i changed the Makefile to chg directory, but same error |
| 18:45.11 | s34n | gniretar: I would like to update to the latest stable |
| 18:45.25 | gniretar | ah, I always just get the tarball |
| 18:45.45 | tzafrir_laptop | chapeaurouge, they seem to assume that make will be called recursively. I believe they set some values (e.g: CFLAGS) in the main makefile |
| 18:45.58 | chapeaurouge | tzafrir_laptop, ok will check. |
| 18:46.26 | Chotaire | downgraded to 1.0.3, working... |
| 18:46.40 | tzafrir_laptop | in this case you probably need to have -I$(TOPDIR)/include/ in CFLAGS. or something similar |
| 18:49.54 | *** join/#asterisk BladeRunner05 (n=gianni@adsl-208-214.38-151.net24.it) |
| 18:51.52 | chapeaurouge | I |
| 18:51.55 | chapeaurouge | mmm |
| 18:51.57 | chapeaurouge | nm |
| 18:52.12 | s34n | can anybody give me tips on cvs updating to the latest stable? |
| 18:52.27 | syle | make update |
| 18:52.33 | sivana | why would I get this? -- Incoming call: Got SIP response 500 "Internal Server Error" back from 209.91.169.180 |
| 18:53.17 | syle | sip debug |
| 18:55.26 | *** join/#asterisk fugitivo (n=ajf@209.13.243.47) |
| 18:55.41 | chapeaurouge | tzafrir_laptop, same error. btw, all the main Makefile also reference /usr/local/include |
| 18:55.54 | *** join/#asterisk viLeR (i=1000@66.128.47.232) |
| 18:56.06 | chapeaurouge | ill post on the mailing list, never know.. |
| 18:56.16 | *** join/#asterisk clinthome (n=clinthom@va-chrvlle-cad1-bdgrp5-8d-209.chvlva.adelphia.net) |
| 18:56.22 | fugitivo | ok |
| 18:56.52 | fugitivo | iax.conf, jitterbuffer=on, iax phone rings, i pick up the call, but asterisk doesn't detect i pick up the call |
| 18:56.53 | fugitivo | anyone? |
| 18:57.09 | fugitivo | jitterbuffer=off, and everything works |
| 18:58.35 | fugitivo | i'm the only one with that problem? |
| 18:59.58 | Chotaire | http://bugs.digium.com/view.php?id=5838 |
| 19:06.18 | *** part/#asterisk pbd (n=plancomm@12.144.118.36) |
| 19:07.19 | Chotaire | zeeek: that was you calling? ;) |
| 19:09.14 | bbz | Would anyone here be interested in some Polycom 501 phones? We got stuck w/ an extra 20 after a project, and we need to unload them at cost, -- if anyone is interested send me a private message (if i dont respond immediately, leave me your email address, and i can send you info) i have a reference in the channel if you need. -- |
| 19:11.51 | bweschke | bbz: you're starting to look very "bot"ish with that msg |
| 19:11.58 | fugitivo | lol |
| 19:12.00 | chapeaurouge | lol |
| 19:15.15 | tainted- | jane jetson is hot |
| 19:15.27 | LostFrog | Ok.. that's out of the blue. |
| 19:15.43 | tainted- | nah i'm working on some hanna barbara stuff |
| 19:15.51 | tainted- | she's hot |
| 19:16.21 | LostFrog | And he quits out of embarrasement. |
| 19:17.15 | *** join/#asterisk kimc (n=root@pcp04041197pcs.wbrmfd01.mi.comcast.net) |
| 19:17.17 | trixter | tainted always struck me as more of a rosie the robot kinda guy |
| 19:17.32 | bweschke | at least he's not made of metal like bbz |
| 19:17.38 | trixter | heh |
| 19:18.19 | LostFrog | I'm sitting here trying to think of a hot saturday morning animated woman.. |
| 19:18.35 | LostFrog | The pussycats always struck me as hot. |
| 19:18.42 | justinu | cheetara from thundercats |
| 19:18.56 | LostFrog | Didn't do it for me.. neither did the woman from he-man. |
| 19:19.09 | justinu | daphne was kinda hot |
| 19:19.11 | *** join/#asterisk jmacz (n=jmacz@208.195.215.48) |
| 19:19.22 | LostFrog | Daphne was the smart one with glasses? |
| 19:19.27 | justinu | no, that was velma |
| 19:19.39 | LostFrog | Se.. I thought *she* was hot. |
| 19:19.43 | justinu | weird |
| 19:19.44 | LostFrog | <--- strange. |
| 19:20.36 | LostFrog | I prefer Alyson Hannigan to Sarah Michelle Gellar. |
| 19:21.51 | LostFrog | This one time at band camp.... |
| 19:21.52 | LostFrog | lol |
| 19:22.08 | *** join/#asterisk filar (n=what@63.138.198.34) |
| 19:22.22 | filar | hello all can anyone help me with a few questions i have |
| 19:22.50 | justinu | you don't ask to ask |
| 19:22.51 | justinu | you just ask |
| 19:22.57 | justinu | or else we kick your ass |
| 19:23.05 | chapeaurouge | heh |
| 19:23.42 | justinu | i dunno, cheetara still looks pretty good |
| 19:23.43 | justinu | http://www.tcsillustrated.net/cheetara14.html |
| 19:24.07 | _DAW | does anyone here use the page cmd in *? I am wondering if it is possible to set alert_info like you can with Dial. I am using Polycom ip 500's and they need alert info to auto answer. |
| 19:24.30 | *** join/#asterisk razu_ (n=razu@ip61.cab74.mus.starman.ee) |
| 19:26.22 | *** join/#asterisk emakris2 (n=emakris@c-24-128-56-2.hsd1.ma.comcast.net) |
| 19:28.54 | filar | ok i want to know how do i connect multipule servers so i can handle the 500 users i have in my call center |
| 19:29.17 | sivana | heh |
| 19:29.20 | justinu | heh |
| 19:29.29 | trixter | I would connect em with ethernet of some flavo |
| 19:29.33 | trixter | flavor |
| 19:29.39 | filar | so they dont need to be clustered? |
| 19:29.40 | trixter | but that is just me |
| 19:29.42 | justinu | lol |
| 19:30.00 | sivana | filar... why |
| 19:30.12 | sivana | you been reading networking magazines again? |
| 19:30.18 | filar | :( no |
| 19:30.36 | asterboy | Need some help with my SIP setup...sip debug shows 404 not found. |
| 19:30.39 | filar | im just trying to figure out how i can implement all the features that the bosses want |
| 19:30.42 | justinu | filar: honestly, you probably won't get much help here for that |
| 19:30.45 | filar | and still make it work |
| 19:31.01 | justinu | filar: that's like enterprise level asterisk consulting, and people here might know, but won't walk you thru it for free... |
| 19:31.01 | filar | well where can i get some answers to those general questions? |
| 19:31.10 | sivana | filar: it's not general |
| 19:31.14 | filar | dont want a walk through |
| 19:31.20 | filar | just some basic answers |
| 19:31.22 | fugitivo | ~docs |
| 19:31.24 | jbot | it has been said that docs is probably Documentation can be found at http://www.digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or http://www.voip-info.org/wiki-Asterisk or http://www.asteriskdocs.org or http://www.oreilly.com/catalog/asterisk or http://www.asteriskguru.com |
| 19:31.25 | asterboy | Anyone know sip and asterisk pretty good? |
| 19:31.40 | sivana | asterboy: nope |
| 19:31.44 | justinu | asterboy: probably because you don't have the extension you're dialing in the context |
| 19:32.01 | asterboy | thats what I though, but I must be missing something. |
| 19:32.07 | asterboy | thought |
| 19:32.17 | asterboy | I'm using didx.org |
| 19:32.41 | asterboy | setup the endpoint with <name>@ip |
| 19:33.02 | chapeaurouge | asterboy, i had 404 when my incoming context didn't include the other context that had the phone numbers |
| 19:33.08 | chapeaurouge | :) |
| 19:33.28 | asterboy | ah |
| 19:33.35 | asterboy | ok I'll check for that. |
| 19:33.44 | *** join/#asterisk loick (n=loick@APuteaux-151-1-6-121.w82-120.abo.wanadoo.fr) |
| 19:34.34 | kimc | anyone know how to get a tdm-400 fxo to ignore short loss of CO battery ? |
| 19:35.09 | justinu | don't use kewlstart? |
| 19:35.12 | asterboy | I have <name>@IP ----> <name> in sip.conf ----->context = internal ----->internal has -----> phone |
| 19:35.18 | kimc | using kewlstart |
| 19:35.32 | asterboy | Can you give me an example setup? |
| 19:35.32 | justinu | kewlstart takes battery drop as disco signal, iirc |
| 19:35.38 | *** join/#asterisk epablo (n=epablo@200.75.139.188) |
| 19:35.42 | asterboy | I'm missing something, just not sure what. |
| 19:36.16 | epablo | Hi people. How's it going |
| 19:36.22 | chapeaurouge | asterboy, and is phone included in your default context? |
| 19:36.31 | chapeaurouge | i mean internal |
| 19:36.37 | kimc | ahh.. wish I could get it to ignore battery drop.. |
| 19:36.46 | justinu | kimc: try using loopstart. |
| 19:36.46 | asterboy | no tag with phone, I meant my telephone |
| 19:36.55 | chapeaurouge | ah |
| 19:37.02 | justinu | asterboy: paste your sip debug and sip.conf sections (blank passwords) |
| 19:37.04 | kimc | tnx will try loopstart sigs ..brb |
| 19:37.09 | *** join/#asterisk tainted- (n=somewher@mail.k2usa.com) |
| 19:37.37 | chapeaurouge | asterboy, anyway, your default context must be aware of the phone number. if not, it wont find it and you'll get a 404.. |
| 19:38.34 | asterboy | wilco justinu....also, from what you have said...I think I need exten =>phonenumber,1,Answer() ....instead I have exten => 8,1,Answer() |
| 19:38.35 | chapeaurouge | you can use paste2paste.pl ;) |
| 19:38.39 | chapeaurouge | ~p2bin |
| 19:38.40 | jbot | p2bin is, like, a script to paste to the http://pastebin.ca from the standard input (linux/unix CLI) . Can be fetched from http://www.madpenguin.org/blogs/chapeaurouge/?p=92 |
| 19:39.06 | justinu | asterboy: you're right, you need to put the phone number as it comes in from didx.org |
| 19:39.11 | asterboy | ah |
| 19:39.17 | asterboy | thats what I'm missing. |
| 19:39.17 | justinu | you can see what they're sending if you look at the INVITE sip URI |
| 19:40.15 | asterboy | sweet! |
| 19:40.20 | asterboy | thank you. |
| 19:40.23 | justinu | np |
| 19:41.01 | kimc | is loopstart fxols ? |
| 19:41.16 | justinu | i think so |
| 19:41.16 | hardwire | anybody in northern colorado needing a job as voip technician? |
| 19:41.22 | hardwire | give me a poke if so.. |
| 19:42.40 | epablo | I have 2 boxes interconected with IAX, but I loose the caller id when fwd a DID from one to the other. Any idea why that could be? |
| 19:42.56 | hardwire | voip job: http://groups.yahoo.com/group/rmiug-jobs/message/24882 |
| 19:43.22 | emakris2 | anyone having sound issues on FC4 and xlite? |
| 19:43.29 | justinu | what's the pay for that job like? |
| 19:45.35 | *** join/#asterisk shido6 (n=shido6@d221-68-216.commercial.cgocable.net) |
| 19:48.02 | kimc | If I change fxsks to fxsls in zaptel.conf * won't start.. |
| 19:49.32 | Chotaire | dudes.. one little question. |
| 19:49.33 | Chotaire | -- No CID number available, no reply sent |
| 19:50.18 | Chotaire | when someone from mailbox 3001 sends a mail to 3002, hitting "reply" doesnt work. |
| 19:50.41 | Chotaire | any idea how I can include that information? |
| 19:50.58 | Chotaire | it seems that replying to a message will only work for caller id's? |
| 19:52.20 | Chotaire | anyone awake who has run into the same problem=? |
| 19:52.40 | Chotaire | oh, for your information, I am running 1.0.3 again because I ran into severe problems with rfc2833... |
| 19:52.46 | ikarus | Hmmmm, is there any applications availible already to make an app contact Asterisk and make it establish a phone call between two devices (for example SIP/phone1 and SIP/phone2) |
| 19:53.47 | *** join/#asterisk Dr-Linux (n=Nothing@202.125.141.6) |
| 19:53.54 | Dr-Linux | justinu ;) hey |
| 19:53.58 | ikarus | Or do I have to hack something up |
| 19:54.26 | Dr-Linux | from where i can download the cisco firmware???? |
| 19:54.41 | Dr-Linux | for cisco 7960 |
| 19:54.57 | jontow | you can't just download it, without having a contract with cisco |
| 19:54.57 | jontow | its not freeware. |
| 19:55.59 | bweschke | Dr-Linux: from cisco cco |
| 19:55.59 | jontow | in fact, being cisco, its anti-freeware ;) |
| 19:55.59 | Dr-Linux | :S |
| 19:55.59 | Dr-Linux | but our all products are from cisco, should i call them? |
| 19:55.59 | Dr-Linux | what is cco ? |
| 19:56.20 | jontow | do you have a maintenance contract for any of your cisco hardware? |
| 19:56.24 | bweschke | it's their online software download library |
| 19:56.26 | jontow | ie. the one with the login/password to their site? |
| 19:56.30 | ikarus | jontow: ah, yes, Cisco the worst thing for your wallet |
| 19:56.44 | jontow | if so.. you can usually login and find the firmware (be prepared to spend a while hunting on their site..) |
| 19:56.49 | jontow | its not the "proper" way to do it, but it works. |
| 19:57.23 | Dr-Linux | yeah have login password , but that i use to download routers/switches images/versions |
| 19:57.24 | filar | cisco told me 1.8 million to do there voip platform :*( |
| 19:57.27 | *** join/#asterisk Bentley (n=Bentley@S0106000f3d016dd2.cg.shawcable.net) |
| 19:57.37 | filar | thats why im looking @ asterisk |
| 19:57.44 | filar | im hopeful it will only be 600k |
| 19:57.46 | bbz | Would anyone here be interested in some Polycom 501 phones? We got stuck w/ an extra 20 after a project, and we need to unload them at cost---->>> $165/ea <<<---- [sorry alot of people have been missing the cost of these] if anyone is interested send me a private message (if i dont respond immediately, leave me your email address, and i can send you info) i have a reference in the channel if you need. -- |
| 19:57.48 | ikarus | filar: Cisco is a fine company, but NOT for VoIP at any sane price range |
| 19:57.57 | ikarus | filar: erm, 1.8 million what ? |
| 19:58.02 | filar | usd |
| 19:58.09 | *** join/#asterisk Rawplayer (n=kevin@ipc31055d2.oom-killer.org) |
| 19:58.16 | ikarus | filar: what kind of deployment are you consideringg ? |
| 19:58.21 | *** join/#asterisk clinthome (n=clinthom@va-chrvlle-cad1-bdgrp5-8d-209.chvlva.adelphia.net) |
| 19:58.24 | filar | multi site |
| 19:58.28 | ikarus | filar: even for 1000 phones that is ALOT |
| 19:58.38 | filar | its about 250 users |
| 19:58.39 | ikarus | filar: yes, but talk sites * phones |
| 19:58.45 | filar | 200 soft phones |
| 19:58.49 | filar | 50 ip phones |
| 19:59.00 | Bentley | hey all - does any know of _any_ linux softphone that will use ESD (sound daemon)? |
| 19:59.06 | filar | ivr blened dialing |
| 19:59.09 | Rawplayer | hi i have a problem with gsm codec, when i want to start asterisk i get this message(the permissions are fine though(i tried it as root to)) |
| 19:59.12 | endre | ribancok |
| 19:59.12 | Rawplayer | Nov 23 20:53:38 WARNING[19430] loader.c: /usr/lib/asterisk/modules/codec_gsm.so: cannot make segment writable for relocation: |
| 19:59.14 | ikarus | filar: that would be about $40000 at most |
| 19:59.15 | Rawplayer | Permission denied |
| 19:59.18 | Rawplayer | like that |
| 19:59.20 | Rawplayer | Nov 23 20:53:38 WARNING[19430] loader.c: Loading module codec_gsm.so failed! |
| 19:59.22 | ikarus | filar: with an Asterisk consultancy firm |
| 19:59.26 | filar | well |
| 19:59.30 | filar | send me info for one |
| 19:59.32 | ikarus | filar: maybe a little more if it is more then 5 sites |
| 19:59.32 | filar | we r looking |
| 19:59.41 | filar | budget is way more then that |
| 19:59.41 | bweschke | filar: www.btwtech.com <--- I'm biased. :) |
| 19:59.48 | Dr-Linux | i'm in pakistan, cisco ipphone 7960 is in US office, whats the better way to download the SIP firmware on it to register with me PBX ? |
| 19:59.49 | filar | u near nyc? |
| 19:59.51 | ikarus | filar: well, I would say look on the wiki, I myself am based in Europe |
| 19:59.57 | bweschke | yes - across the river |
| 20:00.04 | bweschke | in Secaucus |
| 20:00.10 | filar | well im in south plainfield |
| 20:00.38 | ikarus | filar: but $1.8 million is INSANE for only 250 phones of which only 50 are hardware (btw, why the softphones, in my experience users don't like them) |
| 20:00.38 | bweschke | yep. grew up in Bridgewater... so I know south plainfield |
| 20:00.52 | filar | its a call center |
| 20:01.05 | filar | we want them to not have a hard phone to play around on |
| 20:02.01 | ikarus | filar: ah, for a call center setup maybe at most $100000 depending on the complexity of the IVR and any supporting software |
| 20:02.13 | ikarus | but that is even a HIGH estimate |
| 20:02.43 | filar | ivr isnt really needed |
| 20:02.51 | filar | mostly vectoring |
| 20:02.55 | filar | and linking of the 3 sites |
| 20:03.05 | filar | basic stuff |
| 20:03.10 | ikarus | filar: that is all basic stuff (if I understand the use of the word vectoring) |
| 20:03.20 | ikarus | that should fall way below the lowest price I mentioned |
| 20:03.30 | filar | basically |
| 20:03.30 | *** part/#asterisk Bentley (n=Bentley@S0106000f3d016dd2.cg.shawcable.net) |
| 20:03.44 | filar | there are 3 things sales outbound and customer support |
| 20:03.52 | *** part/#asterisk benjk (n=benjk@f8a01-0357.din.or.jp) |
| 20:03.52 | filar | then a queue for all of those |
| 20:04.12 | ikarus | filar: that is all basic call center stuff |
| 20:04.21 | filar | yes i know |
| 20:04.28 | filar | thats why 1.8 million usd was insane |
| 20:04.31 | ikarus | filar: that really shouldn't cost more then 50000 including one massive support contract |
| 20:05.04 | filar | yeah thats what we r looking for |
| 20:05.20 | ikarus | filar: ofcourse if you want to do it yourself, you could prolly master it in under 2 months, but the issue then is that testing and debugging is more tricky |
| 20:06.36 | asterboy | When calling into a context and connecting to another pots line...I'm just getting dial tone and no response from key press...what am I missing to bridge the call. |
| 20:07.05 | *** part/#asterisk epablo (n=epablo@200.75.139.188) |
| 20:08.37 | syle | everyone hates softphones thats reasonable |
| 20:08.39 | asterboy | exten => 8,1,Answer(), exten => Dial(Zap/5) connects and I get dial tone...what do I need next to make another call out on that line? Key presses are ignored. |
| 20:09.00 | syle | who the hell wants to sit on their computer to talk when they can walk around their house on their cordless hehe |
| 20:09.12 | asterboy | I have an implant. |
| 20:09.17 | trixter | I use a cordless softphone |
| 20:09.23 | syle | no shit |
| 20:09.25 | syle | that exists? |
| 20:09.27 | asterboy | resistance is futile! |
| 20:09.31 | tainted- | trixter how's that PRI working for ya |
| 20:09.31 | trixter | ipaq with wifi |
| 20:09.34 | trixter | yes it does :) |
| 20:09.39 | asterboy | lower your asterisk and prepare... |
| 20:09.45 | filar | thats why we hope to find a constulting firm for 2 months while i learn it all |
| 20:09.48 | ikarus | filar: but seriously, WTF is Cisco smoking |
| 20:09.54 | tainted- | cordless software = decroted piece of crap |
| 20:10.02 | trixter | tainted: it was originally requested for 24 channels, that is not a pri, sorry to burst your bubble |
| 20:10.46 | tainted- | show me the url for a $200/ "virtual t1" |
| 20:10.56 | trixter | I never said virtual |
| 20:11.33 | filar | i wish i knew i htink i need some of it |
| 20:12.04 | tainted- | that guy asked for PRI |
| 20:12.11 | ikarus | filar: hehehe |
| 20:12.50 | trixter | he assked for 24 channels |
| 20:12.59 | tainted- | lemme check |
| 20:13.00 | zoa | cisco is always expensive |
| 20:13.03 | zoa | they are good though |
| 20:13.11 | trixter | which cant be a pri, becuase the only way a pri can do 24 channels is with NFAS which requires more than a single span |
| 20:13.33 | tainted- | 24 channels is 24 channels |
| 20:13.38 | bweschke | hi kram! |
| 20:13.39 | tainted- | what you're implying is 24 US |
| 20:13.45 | tainted- | USABLE channels for voice |
| 20:14.11 | trixter | that is what he was wanting |
| 20:14.36 | filar | ive seen good cisco |
| 20:14.42 | filar | and bad cisco installs |
| 20:14.42 | trixter | Nov 20 22:07:14 nick125 quick question: anyone here know about how much a voice\ |
| 20:14.42 | trixter | <PROTECTED> |
| 20:14.49 | ikarus | trixter: a PRI can easily do 30 channels :P, in Europe |
| 20:14.51 | filar | pri is 23 channels |
| 20:14.57 | trixter | I am not implying anything I was reading what was said and not trying to pretend I know more than I do |
| 20:15.01 | tainted- | ok fine |
| 20:15.04 | trixter | he isnt in europe |
| 20:15.06 | ikarus | filar: or 24 if you don't mind bitrobbing |
| 20:15.12 | tainted- | show me 24 channels for 200/month |
| 20:15.12 | trixter | nor was the question related to europe |
| 20:15.21 | ikarus | filar: which just isn't acceptable |
| 20:15.24 | kram | hey bweschke :) |
| 20:15.29 | tainted- | i quoted 500 - 1000 for PRI depending on his NPA and carrier |
| 20:15.32 | kram | sorry it took me a long time to make sure i spelled it right |
| 20:15.46 | bweschke | lol. |
| 20:15.47 | trixter | Nov 20 22:07:14 nick125 quick question: anyone here know about how much a voice t1 (24 channels) costs per month? |
| 20:15.49 | trixter | Nov 20 22:08:23 tainted_ nick125 if it's in a metro area, i'd say between 500 - 1000 |
| 20:15.53 | Corydon-w | kram: tab-completion |
| 20:15.57 | trixter | you didnt quote for a pri you quoted for a voice TT1 24 channels |
| 20:16.00 | trixter | which is pathetic |
| 20:16.05 | bweschke | maybe should just change it to beej - it's probably easier for everyone |
| 20:16.07 | trixter | and as soon as I sid that seemed high you started in on me |
| 20:16.12 | zoa | hey kram! |
| 20:16.15 | zoa | Que pasa |
| 20:16.20 | *** join/#asterisk pryk (n=tmalkut@fw.orasoft.net.pl) |
| 20:16.21 | trixter | now 2 days later you wont let it go |
| 20:16.21 | kram | hi zoa |
| 20:16.23 | zoa | what did i hear about you liking homebrew |
| 20:16.31 | *** part/#asterisk mog_work (n=mogorman@gateway.digium.com) |
| 20:16.32 | zoa | i hope you dont prefer it over the genuine belgian beer! |
| 20:16.34 | tainted- | i just want to see the 200/month link :( |
| 20:16.45 | trixter | volcano telephone |
| 20:16.47 | tainted- | cause we could use one here at work |
| 20:16.58 | trixter | course about 5-6 years ago they went 100% fiber everywhere so its really cheap for them to provide |
| 20:17.07 | *** join/#asterisk mog_work (n=mogorman@gateway.digium.com) |
| 20:17.07 | *** join/#asterisk darby_t (i=darby_t@dkq86.neoplus.adsl.tpnet.pl) |
| 20:17.24 | filar | i pay 8k for 45mbit |
| 20:17.25 | filar | hehe |
| 20:17.29 | filar | or 600 for 24 |
| 20:17.34 | filar | err 24 channels |
| 20:17.36 | zoa | http://www.ratebeer.com/Ratings/Beer/RateBeer-Best/RateBeer-Best-JUN2005.asp |
| 20:17.50 | trixter | course it wasnt that costly for volcano to do that becuase they only service 2 counties in california. |
| 20:17.53 | tainted- | trixter http://www.volcanocommunications.com/ |
| 20:17.58 | tainted- | ? |
| 20:18.08 | trixter | let me see that doesnt seem right |
| 20:18.38 | ikarus | Hmmm, I wonder how expensive a REAL E1 is here (fake ones, backed with ADSL or SDSL are 250 euro a month) |
| 20:18.45 | trixter | that is one of their pages, they are all very closely tied together |
| 20:18.53 | Rawplayer | whats wrong with this? i cant find it on google or anywhere else |
| 20:18.53 | Rawplayer | <PROTECTED> |
| 20:18.55 | Rawplayer | Nov 23 21:18:34 WARNING[4983]: loader.c:554 load_modules: Loading module codec_gsm.so failed! |
| 20:18.58 | trixter | small ilec so ... |
| 20:19.16 | tainted- | so 200 includes DID bundle? |
| 20:19.24 | tainted- | and origination? |
| 20:19.34 | trixter | course the one thing I never mentioned nor did you ever ask was the usage commitment |
| 20:19.44 | trixter | but you just went all psycho over this at the sound of $200 |
| 20:19.53 | trixter | so it was pointless then to even try to reason with you |
| 20:20.05 | trixter | with the right usage agreement I bet even sbc or qwest would do it for $200/mo |
| 20:20.18 | tainted- | oh? |
| 20:20.32 | tainted- | what usage agreement would that be? |
| 20:20.55 | bbz | http://www.dumpalink.com/media/1131535942/Beer_And_A_Butt NSFW |
| 20:21.53 | trixter | why dont you call and ask.. it is in part on your negotiating skills and area.. volcano for example wont sell you anything unless you are in the local area, the whole thing about being a tiny rural ilec. Normally they cant give it away free, why NJ Bell (before bell atlantic and subsequent verizon buyouts) would do T3 termination for $1/mo |
| 20:22.00 | *** join/#asterisk colloron (n=liran@80.178.91.87.adsl.012.net.il) |
| 20:22.20 | trixter | that was their published rate at the time |
| 20:22.26 | luite | ikarus: kpn isdn30 is a real e1 isn't it? that's 199 a month, ex vat |
| 20:23.03 | *** join/#asterisk mazimp (i=nobody@p54AB3B90.dip0.t-ipconnect.de) |
| 20:23.04 | ikarus | luite: you have to be kidding |
| 20:23.19 | luite | ikarus: just look at the kpn business site |
| 20:23.25 | luite | ikarus: http://www.kpn.com/kpn/show/id=816905/spage=816886/sc=cfed72 |
| 20:23.29 | ikarus | luite: doesn't work without Flash |
| 20:24.11 | ikarus | luite: I was on the phone with Versatel about their E1, they said it was backed with an *DSL link and required you to pay extra for that |
| 20:24.27 | ikarus | luite: so KPN in practice might be cheaper |
| 20:24.27 | luite | ikarus: well, it says 199 a month for isdn30 belbasis, 219 for belbasis extra, setup cost is 1499,50 for the first line, 1270,59 for the other lines |
| 20:24.37 | wunderkin | my pri is delivered over an hdsl line |
| 20:24.40 | *** join/#asterisk oej (n=oej@apollo.webway.se) |
| 20:25.03 | ikarus | wunderkin: I sure hope you get a discount for that |
| 20:25.05 | asterboy | can I make a sip or iax call to my asterbox and bridge the call to my pots line? |
| 20:25.30 | madeddie | bbz: GOD that's awfull :) |
| 20:25.34 | shido6 | asterbox |
| 20:25.35 | shido6 | hehehe |
| 20:25.36 | wunderkin | well, im only using that provider since they were the cheapest for minutes and it wasnt that much worse for the loop charge |
| 20:25.51 | asterboy | or *box |
| 20:26.22 | wunderkin | it would have been nicer if i could have gone in wth a provider that is on-net but they wouldnt work on the minutes with me |
| 20:26.37 | asterboy | well??? |
| 20:27.17 | ikarus | luite: they 89 euro a month for ISDN 20 according to the people I spoke at versatel is only if you also have internet from them and is only disentangled from the internet traffic at their office |
| 20:27.28 | asterboy | SIP ---> ZAP ??? |
| 20:27.34 | asterboy | Zap a sip? |
| 20:27.47 | asterboy | sippy to zappy? |
| 20:27.58 | asterboy | zippy sippy? |
| 20:28.02 | syzygyBSD | asterboy: yes |
| 20:28.15 | asterboy | ok, so I'm not imagining it. |
| 20:29.12 | ikarus | luite: also I discovered BRI cascading takes about 30 minutes to arrange with KPN, just call a few days in advance so they have all data on hand when you want the transfer |
| 20:29.39 | asterboy | I have it so my SIP connects to ZAP and I get a dial tone...BUT I can't dial anything...what do I need in extensions.conf to make this work??? |
| 20:29.55 | luite | ikarus: ah, sounds good, tnx |
| 20:30.16 | ikarus | I wish a PRI was an option for the setup I am currently working on, but a 600 euro card is just not an option, if the entire roll out costs 900 euro with 2x BRI |
| 20:30.38 | ikarus | (that price includes a system to run Asterisk on) |
| 20:31.10 | *** join/#asterisk darby_t (i=darby_t@dkq86.neoplus.adsl.tpnet.pl) |
| 20:31.31 | luite | ikarus: have you already looked at chan_visdn? |
| 20:31.44 | ikarus | luite: no, still don't have the HFC-S cards |
| 20:31.48 | asterboy | exten => context,1,Dial(Zap/5) do I need something else after this??? |
| 20:32.17 | ikarus | luite: also Asterisk 1.0 is very tempting to keep considering some loose ends in 1.2 |
| 20:32.52 | asterboy | I get the dial tone...but it does not repsond to key presses? |
| 20:33.00 | luite | ikarus: there is a patch that makes the channel driver work with 1.2, but thats stil quite.. experimental. and it seems that you need to run asterisk as root to be able to use visdn |
| 20:33.07 | *** join/#asterisk darby_d (i=darby_t@dkl100.neoplus.adsl.tpnet.pl) |
| 20:33.09 | *** join/#asterisk epablo (n=epablo@200.75.139.188) |
| 20:33.24 | asterboy | Can someone please point me to a doc that has an example. |
| 20:33.26 | asterboy | ??? |
| 20:33.34 | ikarus | luite: well, that is always "fixable" unless it uses raw port access |
| 20:33.45 | ikarus | (in which case it just takes more programming) |
| 20:34.12 | asterboy | voip-info.org has no bridge information. |
| 20:34.12 | epablo | Anyone have 5 min to help me out with a DID loosing it callerid through IAX, from one box to another? |
| 20:34.30 | luite | ikarus: it needs the CAP_NET_BIND_SERVICE capability, which apparently is only available for root |
| 20:34.40 | *** join/#asterisk mcadory (n=mcadory@208.149.64.28) |
| 20:34.43 | ikarus | luite: that you can patch kernel level |
| 20:34.59 | luite | true |
| 20:35.04 | ikarus | luite: what kernel module is used with visdn ? |
| 20:35.08 | asterboy | 335 people...and not much info on bridging? |
| 20:35.13 | luite | ikarus: visdn's own module |
| 20:35.33 | luite | -' |
| 20:35.36 | luite | :) |
| 20:35.37 | *** part/#asterisk mcadory (n=mcadory@208.149.64.28) |
| 20:36.02 | asterboy | christ, just the other day we were diving into quantum entaglements. |
| 20:36.12 | luite | ikarus: which only supports hfc-s and its multiport cousins |
| 20:36.21 | ikarus | luite: hmmmmm, erm CAP_NET_BIND_SERVICE is used to be able to bind to TCP/UDP ports below 1024, what is the deal there |
| 20:37.12 | luite | ikarus: hm, don't know, I found it on the mailing list... let me check again |
| 20:37.14 | *** join/#asterisk BrianR___ (i=brianr@c-24-61-206-174.hsd1.ma.comcast.net) |
| 20:37.43 | syle | you need root to bind to ports lower than 1024 |
| 20:38.01 | ikarus | syle: that is not the issue |
| 20:38.46 | luite | ikarus: ah, it seems that they want you to be root to connect to the lapd (D channel) socket |
| 20:39.11 | ikarus | luite: hmmmmm, let me check where that code lives |
| 20:39.39 | luite | it's probably doable to allow another uid to do that as well |
| 20:40.45 | luite | ikarus: https://mailman.uli.it/pipermail/visdn-hackers/2005-November/000101.html |
| 20:42.51 | ikarus | luite: well, you could if I remember how grant a user CAP_NET_BIND_SERVICE capability and run Asterisk as that user (go fine grained restrictions) |
| 20:43.04 | asterboy | please, can someone give me some insite into why I'm getting a dial tone but no response to key presses??? |
| 20:43.28 | tainted- | br0ken phone |
| 20:43.44 | ikarus | luite: or else, lapd/af_lapd.c line 1713 |
| 20:44.07 | luite | ikarus: but that stil is not ideal. it would still be better to have asterisk drop its privileges after opening the socket |
| 20:45.07 | ikarus | luite: well, true, but if the socket ever needs re-opened for what ever reason (I don't know the code on this, but some code is that creepy) |
| 20:45.50 | *** join/#asterisk mazimp (i=nobody@p54AB3B90.dip0.t-ipconnect.de) |
| 20:46.28 | *** join/#asterisk MRH2 (n=Mr_happy@fcirc-adsl.demon.co.uk) |
| 20:46.37 | *** join/#asterisk felipex (n=dsfdsf@85-18-250-142.ip.fastwebnet.it) |
| 20:46.58 | ikarus | luite: I am sure it is possible to drop CAPabilities, so that a quick patch to the visdn code would do the trick, aslong as Asterisk is launched with having those capabilities (in such a way it doesn't loose them for another reason, like becoming non-root due to the init order) |
| 20:54.18 | mazimp | anybody there, who can help me on an problem with incoming SIP calls? |
| 20:55.19 | *** join/#asterisk `erb (n=gen2@69.17.142.245) |
| 20:55.38 | *** join/#asterisk rajiv (n=irc@gentoo/developer/rajiv) |
| 20:55.55 | ikarus | luite: I think I can get it working, if I understand this correctly, I can make a process launched get the capability, which in theory should survive the UID changing, but later on can be dropped |
| 20:56.08 | *** join/#asterisk darby_t (i=darby_t@dkv174.neoplus.adsl.tpnet.pl) |
| 20:56.47 | `erb | hey guys, I am having a problem with my SPA (fxs) making ringtones after an outgoing call is connected |
| 20:58.32 | `erb | its pretty annoying and its the only problem I've left to fix |
| 20:59.10 | _Sam-- | hi i had been using the SIPPs softclient for a while, but when I upgraded to the newest version i get this on the console (and no calls): Nov 22 21:14:15 WARNING[32568]: chan_sip.c:2704 process_sdp: Unknown SDP media type in offer video 1551 RTP/AVP 34 |
| 20:59.14 | *** join/#asterisk Druken (n=druken@CPE00121716da99-CM000e5cde4ca2.cpe.net.cable.rogers.com) |
| 20:59.15 | _Sam-- | downgraded to their older client, and its fine....in the new client there is no video related settings that i can disable |
| 20:59.22 | _Sam-- | my asterisk is Asterisk CVS-v1-0-11/16/05-10:39:37 |
| 21:00.59 | _Sam-- | any ideas? |
| 21:01.46 | ikarus | luite: just a bit unsure about how the current state in the kernel is (just slowly discovered capabilities isn't an entirely standard feature) |
| 21:05.28 | asterboy | Sam; sounds like there is some sort of information being passed along...maybe a codec that is not recognize...just a noob so can't help too much. |
| 21:05.50 | asterboy | Its just a warning though, so you should be ok. |
| 21:06.19 | asterboy | Maybe the company who creates the software has added some future provision for video. |
| 21:06.52 | asterboy | Did you turn on "sip debug" to get more info? |
| 21:07.47 | asterboy | I'm having trouble getting my SIP call to dial out on my POTS line. It makes the bridge and I get dial tone...but it ignores any key presses...not sure what is going there.??? |
| 21:08.02 | ikarus | luite: but one thing is for sure, what you suggest is possible, either by hacking Asterisk (which is the easy route, but might require maintaining a seperate patch) or with the capabilities stuff |
| 21:08.23 | *** part/#asterisk kimc (n=root@pcp04041197pcs.wbrmfd01.mi.comcast.net) |
| 21:08.51 | epablo | asterboy: that a problem with the DTMF. How are you going to POTS? digium HW? |
| 21:09.06 | *** join/#asterisk kimc (n=root@pcp04041197pcs.wbrmfd01.mi.comcast.net) |
| 21:10.41 | asterboy | epablo, yes |
| 21:11.03 | asterboy | It works from internal FXS connected telephones. |
| 21:11.33 | asterboy | But bridging from a sip call it connects with dial tone but ignores keys. |
| 21:11.40 | *** join/#asterisk ast_freak (n=jesse@68-112-134-195.dhcp.stcl.mn.charter.com) |
| 21:11.42 | *** join/#asterisk kimc (n=freenode@pcp04041197pcs.wbrmfd01.mi.comcast.net) |
| 21:11.55 | asterboy | I'm thinking I'm missing something simple in extensions.conf |
| 21:12.38 | asterboy | epablo: Do you do SIP to PSTN bridges or know of an example I could follow? |
| 21:12.40 | syzygyBSD | asterboy, in your sip.conf what is yoru dtmf setting |
| 21:13.04 | *** part/#asterisk oej (n=oej@apollo.webway.se) |
| 21:13.17 | syzygyBSD | *dtmfmode |
| 21:13.25 | asterboy | syzygyBSD: dtmfmode=rfc2833 |
| 21:13.57 | syzygyBSD | try inband, it may break other key presses though |
| 21:13.57 | asterboy | should that be inband? |
| 21:14.14 | asterboy | :P...I owe you a beer. |
| 21:14.44 | *** join/#asterisk n4y (n=tmalkut@fw.orasoft.net.pl) |
| 21:15.05 | syzygyBSD | send it to santa, maybe if he is drunk he will ignore some of the things i have done this year |
| 21:16.29 | ikarus | luite: I'll look into it later, I think I need to recompile the kernel with atleast ACL and Capabilities support, but I don't have a spare system handy here |
| 21:16.52 | asterboy | lol |
| 21:17.12 | asterboy | ok...so changing to inband and restarting asterisk did not seem to make a difference. |
| 21:18.04 | asterboy | I was thinking cause * bridges the line, it would simply pass through anything I pressed, no? |
| 21:19.25 | asterboy | maybe its a codec thing. |
| 21:20.13 | asterboy | Well, how do ATA devices work? The are basically a sip device, (some iax), that wants to grab a PSTN line, no? |
| 21:20.21 | *** join/#asterisk saftsack (n=saftsack@p54A7F0B5.dip.t-dialin.net) |
| 21:21.20 | saftsack | i |
| 21:21.20 | saftsack | hi |
| 21:22.05 | syzygyBSD | your hi? |
| 21:22.21 | *** join/#asterisk cnet2 (n=jjohn@adslnat-sanjose-4.ice.co.cr) |
| 21:22.27 | syzygyBSD | or was the i a mistype? |
| 21:22.38 | saftsack | that was a misstype |
| 21:23.40 | syzygyBSD | asterboy: go back to rfc2833 |
| 21:23.53 | asterboy | I did |
| 21:24.05 | syzygyBSD | how are you trying to dial the number to the zap channel? |
| 21:24.18 | syzygyBSD | can I see your extensions.conf, pastbin it |
| 21:24.25 | asterboy | ok |
| 21:25.19 | *** part/#asterisk santiago (n=santiago@208.195.215.160) |
| 21:25.22 | saftsack | do you know a cheap solution for providing two s0 busses and 4 analog ports on an asterisk pbx? |
| 21:25.27 | *** join/#asterisk bugz (n=bugz@cpe-24-27-67-66.houston.res.rr.com) |
| 21:25.47 | bugz | put a dual xeon up with 16 T1's in it the other day, still shares irq's |
| 21:26.13 | bugz | it has 4 four port t1 cards in it but one of them wants to sit on the same irq as the raid controller |
| 21:26.26 | asterboy | exten => 9,1, Answer() |
| 21:26.34 | zoa | apic ? |
| 21:26.43 | asterboy | exten => 9,2 ,Dial(Zap/5) |
| 21:26.45 | bugz | zoa the bios sucks |
| 21:26.53 | zoa | give it to me :) |
| 21:27.05 | bugz | haha |
| 21:27.06 | asterboy | syzygyBSD: Thats pretty much it...I have a simple file. |
| 21:27.10 | bugz | the box went for 25 grand |
| 21:27.12 | asterboy | syzygyBSD: I'll paste bin |
| 21:27.53 | bugz | but i'll give it to you, right now, for the very low price of $26K |
| 21:28.15 | bugz | which, if you think about it, is a very good deal |
| 21:28.17 | zoa | haha |
| 21:28.32 | Katty | hi lads. |
| 21:28.35 | bugz | considering that 5 more of them will be replacing a 2 million dollar nortel |
| 21:28.53 | zoa | why would you buy a 4 xeon if you can have a dual core dual opteron that is faster, for 5000$ |
| 21:29.00 | syzygyBSD | normally i see something like exten => 9XXXXXXX,1,Dial(Zap/5/${EXTEN:1}) |
| 21:29.04 | bugz | we allready have the part |
| 21:29.21 | bugz | but amd is in the future |
| 21:29.46 | *** join/#asterisk CunningPike (n=arodgers@S010600095b33697f.vc.shawcable.net) |
| 21:29.48 | bugz | the bbox handles 65,000 calls a day with a load of around 2 |
| 21:29.57 | zoa | when you have the amd's donate the old quad xeons to asteriskguru |
| 21:30.05 | zoa | bugz, we do 100.000 on a dual xeon |
| 21:30.07 | bugz | ill see what i can do |
| 21:30.11 | zoa | depends on what you do with the calls :) |
| 21:30.13 | bugz | nice |
| 21:30.16 | bugz | right |
| 21:30.17 | CunningPike | Does anyone know if #asterisk-stable is borked? |
| 21:30.19 | zoa | and how long the calls last |
| 21:31.30 | bugz | zoa: what city is your box in? |
| 21:31.38 | bugz | ours is in houston |
| 21:31.50 | zoa | belgium |
| 21:32.05 | bugz | now there is a support contract for you |
| 21:32.06 | zoa | thats not really a city though :) |
| 21:32.16 | bugz | close enough |
| 21:32.34 | *** join/#asterisk tstoermer (n=Administ@p54B2B720.dip0.t-ipconnect.de) |
| 21:32.45 | bugz | why is asteriskguru always down |
| 21:32.50 | zoa | is it down ? |
| 21:32.53 | zoa | its sometimes down ??? |
| 21:32.57 | zoa | it shouldnt be |
| 21:33.01 | zoa | i visit it all day long |
| 21:33.08 | bugz | zoa: seriously i have alot of trouble getting to it regularly |
| 21:33.23 | zoa | owz |
| 21:33.39 | zoa | maybe its the archives thing taking up too much cpu |
| 21:34.02 | zoa | bugz, next time you have problems reaching it, send me a traceroute |
| 21:34.08 | bugz | will do |
| 21:34.18 | kimc | what is a good alternative to tdm-400 fxo's for > reliability ? |
| 21:34.35 | bugz | kimc: spare cards ;P |
| 21:34.37 | zoa | its on a pretty expensive fiber link, so should be quite good |
| 21:34.40 | kimc | heh.. |
| 21:34.41 | zoa | (not fast though) |
| 21:34.47 | zoa | maybe i will put a mirror in the states |
| 21:34.51 | *** join/#asterisk sm7syx (n=marten@h-83-140-228-136.ip.cust.port80.se) |
| 21:35.03 | kimc | are quintum ports good ? |
| 21:35.55 | *** part/#asterisk tstoermer (n=Administ@p54B2B720.dip0.t-ipconnect.de) |
| 21:36.10 | asterboy | syzygyBSD: Here is the pastebin: http://pastebin.ca/30716 |
| 21:36.45 | sm7syx | Hi, I need help with my dailplan. I would like to answer the call, let it ring for some time and them in background play a file and continue to ring ... Can't get it right. |
| 21:37.12 | *** join/#asterisk Rawplayer (n=kevin@ipc31055d2.oom-killer.org) |
| 21:37.38 | *** join/#asterisk boday_ (n=jeanfran@modemcable194.120-201-24.mc.videotron.ca) |
| 21:37.44 | boday_ | Hi, can someone tell me if SIP mode behind firewall is handle better now |
| 21:38.05 | mog_work | better? |
| 21:38.50 | boday_ | mog_work: i was using it i think version 1.0 and if my server was behind my firewall and i was over the internet, my sip phone was working like shit |
| 21:38.57 | boday_ | i did forward port |
| 21:39.03 | boday_ | rtp, sip 5060 |
| 21:39.12 | boday_ | always work like shit |
| 21:39.21 | boday_ | but i dont know with newer version |
| 21:39.23 | boday_ | since 1.0 |
| 21:39.26 | shido6 | rtp is on diff ports |
| 21:39.33 | boday_ | i know |
| 21:39.43 | shido6 | what do you have there, or did you set them in your * config |
| 21:39.55 | mog_work | you probably never configured it correctly |
| 21:39.56 | boday_ | what should i set |
| 21:40.00 | mog_work | but sip != nat friendly |
| 21:40.01 | boday_ | oh |
| 21:40.02 | *** part/#asterisk CunningPike (n=arodgers@S010600095b33697f.vc.shawcable.net) |
| 21:40.06 | boday_ | yeah i know |
| 21:40.08 | mog_work | but asterisk can bridge it for you |
| 21:40.10 | boday_ | sip != nat |
| 21:40.19 | asterboy | brb |
| 21:40.22 | boday_ | is there a way to make it friendly |
| 21:40.55 | mog_work | Are asterisk and your phone on same network? |
| 21:41.18 | boday_ | nope |
| 21:41.20 | boday_ | asterisk private |
| 21:41.23 | boday_ | phone anywhere |
| 21:41.26 | boday_ | over the internet |
| 21:41.58 | *** join/#asterisk windex (n=jamie@host-69-95-178-154.mil.choiceone.net) |
| 21:42.06 | *** join/#asterisk r0d3nt (n=RatMan@tinfoilhat.net) |
| 21:42.20 | MRH2 | can u config ur phone to use the same port for in and out rtp? |
| 21:42.45 | windex | question. :) my sip phone has a confrence button, but every attempt to use it causes asterisk to ignore the request (as far as i can tell). suggestions? |
| 21:43.04 | syzygyBSD | asterboy: sorry, I am recompiling a kernel and was afk at the server |
| 21:43.20 | syzygyBSD | did you see the dial line i wrote? |
| 21:43.58 | boday_ | x-lite is use as sip phone |
| 21:44.22 | boday_ | MRH2 probably i can |
| 21:44.36 | syzygyBSD | probably depends on the phone |
| 21:45.39 | *** join/#asterisk ReD-MaN (i=redman@dhcp-0-2-b3-9a-4a-5b.cpe.quickclic.net) |
| 21:45.54 | MRH2 | well if it is a soft phone then it could be anything causing quality issues |
| 21:46.23 | *** join/#asterisk Faithful (n=Faithful@202-6-145-116.ip.adam.com.au) |
| 21:46.36 | asterboy | syzygyBSD: yes, I tried it, looks like its just a way to pass the context via a variable to the end of the dial. |
| 21:46.42 | asterboy | syzygyBSD: but it did not work. |
| 21:46.46 | asterboy | syzygyBSD: :( |
| 21:47.08 | asterboy | syzygyBSD: Its like I'm missing the outbound channel. |
| 21:47.34 | asterboy | syzygyBSD: I can connect to whatever context I want when the call comes in...but after that I can't do anything more. |
| 21:48.21 | syzygyBSD | it is a way to pass the phone number into the dial command |
| 21:48.34 | *** join/#asterisk graphyx (n=mike@67.50.46.118) |
| 21:48.56 | graphyx | What is the abbreviation that you use for Thursday using the gotoiftime command? thu? |
| 21:49.11 | syzygyBSD | so if you dial 9(7digitphone number) it should pass the 7 digits to Dial |
| 21:49.12 | asterboy | syzygyBSD: oh I see...well it just a way to automate the process, but regardless the darn thing does not want to take input of anykind. |
| 21:50.00 | syzygyBSD | what did it do when you did that extension? |
| 21:50.15 | asterboy | syzygyBSD: as you said...it trys to pass the info along. |
| 21:50.23 | asterboy | syzygyBSD: still no go though. |
| 21:50.46 | syzygyBSD | did it give you dialtone? fast busy, invalid number? |
| 21:50.57 | graphyx | any idea on the weekday abbreviation for thursday? |
| 21:51.02 | asterboy | syzygyBSD: as before...a dial tone. |
| 21:51.03 | syzygyBSD | Thurs |
| 21:51.08 | graphyx | ok |
| 21:51.09 | syzygyBSD | or Th |
| 21:51.26 | asterboy | syzygyBSD: Th? |
| 21:51.43 | syzygyBSD | not for you, weekday abriviation for thursday |
| 21:51.49 | asterboy | syzygyBSD: ah |
| 21:52.16 | asterboy | syzygyBSD: how does a normal sip phone bridge calls? |
| 21:52.32 | *** part/#asterisk graphyx (n=mike@67.50.46.118) |
| 21:52.44 | asterboy | syzygyBSD: I'm using didx.org with a DID pointing to my system via SIP. |
| 21:53.19 | asterboy | syzygyBSD: So I call the number from a POTS line and it forwards it onto my *box. |
| 21:53.32 | asterboy | syzygyBSD: Then I'm trying to connect out again. |
| 21:53.44 | syzygyBSD | what happend when you used that dial command I gave you |
| 21:53.45 | asterboy | syzygyBSD: So I can make long distance calls from my cell for free. |
| 21:53.45 | *** part/#asterisk windex (n=jamie@host-69-95-178-154.mil.choiceone.net) |
| 21:53.50 | *** join/#asterisk UlbabraB (n=caplaz@host241-43.pool8172.interbusiness.it) |
| 21:53.53 | asterboy | syzygyBSD: same exact thing. |
| 21:54.02 | syzygyBSD | just dialtone? |
| 21:54.05 | asterboy | syzygyBSD: yep |
| 21:54.26 | syzygyBSD | odd |
| 21:54.44 | asterboy | syzygyBSD: yes...I didnn't think it would be that difficult |
| 21:55.11 | syzygyBSD | can you pastebin your zapata.conf without comments? |
| 21:55.16 | syzygyBSD | or the whole thing, |
| 21:55.18 | asterboy | syzygyBSD: Maybe I should try IAX and change my contexts to numbers instead of names so that when they are passed using that variable it actually has a number. |
| 21:56.10 | asterboy | syzygyBSD: http://pastebin.ca/30726 |
| 21:57.57 | syzygyBSD | do you really not have another label in there somewhere? |
| 21:58.19 | syzygyBSD | inbetween the 2 context= |
| 21:58.27 | MRH2 | how about within the context for the inbound call 1 s- answer, 2 playback background "enter the number".... 3 goto2 then extensions dial blah |
| 21:58.50 | *** join/#asterisk PupenoL (n=pupeno@200.123.183.89) |
| 21:58.58 | MRH2 | check the caller id is from ur mobile first |
| 21:59.13 | MRH2 | sorry cell |
| 21:59.33 | *** join/#asterisk blitzrage (n=blitzrag@asterisk/documenteur-extraordinaire/blitzrage) |
| 21:59.35 | syzygyBSD | he can't get zap to recognize any DTMF tones |
| 22:00.09 | syzygyBSD | in that zapata.conf file 3 variables are overwritten with different values |
| 22:00.19 | MRH2 | ah k |
| 22:01.04 | syzygyBSD | I would have him try a exten => 9,1,Dial(Zap/5/1234567); as well |
| 22:01.22 | syzygyBSD | so it isn't dealing with numbers from his sip phone at all |
| 22:01.48 | syzygyBSD | if he is still around |
| 22:01.52 | *** part/#asterisk loick (n=loick@APuteaux-151-1-6-121.w82-120.abo.wanadoo.fr) |
| 22:02.29 | *** join/#asterisk MikeJ[Laptop] (n=ircatjer@d14-69-8-30.try.wideopenwest.com) |
| 22:03.26 | MRH2 | gotta go .... toodles |
| 22:05.14 | asterboy | ok, it works for IAX |
| 22:05.16 | asterboy | just not SIP |
| 22:05.22 | asterboy | for whatever reason. |
| 22:06.28 | asterboy | syzygyBSD: I'll give that a shot later to see if it makes a difference. |
| 22:06.35 | asterboy | syzygyBSD: IAX seems to work great. |
| 22:06.40 | *** join/#asterisk kink0 (n=k@62.37.205.161) |
| 22:06.43 | kink0 | buenas noches !! |
| 22:07.12 | asterboy | syzygyBSD: interesting SIP DTMF tones don't work though...my paste bins, (scroll back), have all my files info. |
| 22:07.18 | asterboy | syzygyBSD: if you find something let me know. |
| 22:07.28 | asterboy | hablo espanola? |
| 22:07.38 | kink0 | I have few questions about AGI: I have created ( C ) a program, and I call them as AGI(programname -c parameter1 -d parameter2 ...) |
| 22:07.47 | asterboy | no spelly espanola very welly |
| 22:07.52 | asterboy | :P |
| 22:07.56 | kink0 | asterboy, si, hablo espaņol |
| 22:08.02 | syzygyBSD | is your zapata.conf right? it looks like there should be another label in there |
| 22:08.03 | asterboy | como esta? |
| 22:08.17 | syzygyBSD | asterboy: you don't know if you speak spanish? |
| 22:08.23 | kink0 | but in public here I speek english so all people can understand ( except for greeting :)) ) |
| 22:08.23 | asterboy | lol |
| 22:08.30 | asterboy | un poco |
| 22:08.56 | syzygyBSD | ya, the last letter of the verb makes a lot of difference |
| 22:08.58 | syzygyBSD | I think... |
| 22:09.01 | asterboy | syzygyBSD: what do you think should be in the zapata.conf |
| 22:09.01 | *** join/#asterisk phifli (n=mike@user-69-73-44-160.knology.net) |
| 22:09.02 | asterboy | syzygyBSD: ?? |
| 22:09.23 | phifli | Why with the new 1.2.0 SetCallerPres() and SetCallerID() won't work as it did before? It kereps showing up as private? |
| 22:09.33 | phifli | ive tried 888 and 1888 |
| 22:09.42 | syzygyBSD | In between your mailbox and context lines should there be another [context] |
| 22:10.15 | syzygyBSD | otherwise you are resetting some values |
| 22:10.41 | kink0 | well, my program ... appears to fail , but if I excute it at shell is ok. |
| 22:10.48 | *** join/#asterisk asteriskmonkey (n=phil@HSE-Toronto-ppp300017.sympatico.ca) |
| 22:11.01 | asteriskmonkey | does zttool always show clock source as INTERNAL? |
| 22:11.03 | syzygyBSD | I would think that sounds like a permissions problem |
| 22:11.30 | kink0 | if I call ussing AGI ... nothing happens, but if I excute the same ussing shell then my program use my mobile phone and start a call |
| 22:12.17 | kink0 | Failed to execute '/usr/serial/serial -p /dev/usb/ttyUSB0 -s 115200 -c d657897123; ': No such file or directory |
| 22:12.17 | kink0 | <PROTECTED> |
| 22:12.17 | kink0 | <PROTECTED> |
| 22:12.21 | asterboy | syzygyBSD: good catch. |
| 22:12.29 | kink0 | where 657897123 is my mobile phone number |
| 22:12.31 | asterboy | syzygyBSD: I think that was accidentally pased there. |
| 22:12.33 | asterboy | syzygyBSD: pasted |
| 22:12.38 | asteriskmonkey | twisted, drumkilla? is zttool always supposed to show internal clock ? |
| 22:12.48 | kink0 | of course if I execute from shell the same sentence my phone is ringing |
| 22:13.18 | syzygyBSD | asterboy: I couldn't help but notice the signalling was different too, that could be causing the issue |
| 22:14.08 | kink0 | I did a little C program just to dial mobile phone ussing several devices |
| 22:14.09 | syzygyBSD | also if you use that Dial() command i did you should also NOOP(${Exten}) before it so the number that will be dialed is printed out in the console |
| 22:14.22 | syzygyBSD | kink0: which is quickest? |
| 22:14.31 | asterboy | syzygyBSD: what is the difference in signalling? |
| 22:14.36 | SplasPood | Can anyone /w Cisco 7960 (SIP) experience suggest why I might only be getting inbound audio, but not outbound.. The 7960 is behind NAT, asterisk is not |
| 22:14.57 | syzygyBSD | other people can handle that better, I just know you use the oposite of the card you have |
| 22:14.59 | *** join/#asterisk pfn (n=pfnguyen@netblock-66-245-252-239.dslextreme.com) |
| 22:15.45 | kink0 | syzygyBSD, when I call my program ussing AGI nothing happens, but if I call it from shell is running ok |
| 22:15.45 | kink0 | I observe this: -- Executing AGI("OSS/dsp", "/usr/serial/serial -p /dev/usb/ttyUSB0 -s 115200 -c d657897123; ") in new stack |
| 22:15.47 | asterboy | syzygyBSD: oh, that I know is working as it should. |
| 22:15.53 | kink0 | but my program is not related with OSS |
| 22:16.13 | asterboy | syzygyBSD: yes and FXO is a telephone and hooked up to an FXS, so you have to tetll your FXS to use FXO signalling. |
| 22:16.26 | syzygyBSD | kink, what are the ownership permissions of your script? |
| 22:16.56 | kink0 | syzygyBSD, root, and asterisk is running as root also , to avoid problem accessing devices. |
| 22:17.34 | kink0 | but is excutable for all: -rwxr-xr-x 1 root root 14040 2005-11-23 20:40 serial |
| 22:18.02 | kink0 | and requires not root or suid permisions, also /dev/usb/ttyUSB0 has ok permisions to access it. |
| 22:18.07 | syzygyBSD | kink0: are you runnign asterisk on the console, or daemon? |
| 22:18.20 | kink0 | syzygyBSD, on console now. |
| 22:18.27 | syzygyBSD | how verbose |
| 22:18.37 | kink0 | many v |
| 22:18.43 | syzygyBSD | good :) |
| 22:18.52 | asterboy | syzygyBSD: thanks for the help...I'll work on it more later. |
| 22:18.59 | asterboy | syzygyBSD: Going to do a service call now. |
| 22:19.09 | kink0 | <PROTECTED> |
| 22:19.09 | kink0 | Failed to execute '/usr/serial/serial -p /dev/usb/ttyUSB0 -s 115200 -c d657897123; ': No such file or directory |
| 22:19.09 | kink0 | <PROTECTED> |
| 22:19.09 | kink0 | <PROTECTED> |
| 22:19.12 | kink0 | that is all I got |
| 22:19.21 | ender | were there any echo improvements between 1.2.0beta1 and 1.2.0 ? |
| 22:19.22 | kink0 | but mobile phone does not start the call |
| 22:19.36 | syzygyBSD | kink0: debug soemthing to stderr or stdout in the first couple lines of your program |
| 22:19.40 | ender | some of my users complain of echo when calling out or getting calls in from our T1 to a Fujitsu PBX. |
| 22:19.43 | syzygyBSD | you should see it in the console |
| 22:19.45 | kink0 | if I run the same program/parameters from shell then my phone does the call |
| 22:20.30 | kink0 | syzygyBSD, I know the communication channel between asterisk and AGI apps are std in/out and stderror |
| 22:21.05 | syzygyBSD | I was just thinkign to see if the program was executing at all from asterisk |
| 22:21.34 | kink0 | syzygyBSD, I thinl is not excuted at all |
| 22:21.38 | kink0 | I think |
| 22:22.04 | kink0 | and at some point is aborted, but that little code just do a control over the serial port and send a number to dial |
| 22:22.05 | syzygyBSD | the python agi i use dumps all of the agi variables on import, so I always know when it is called |
| 22:22.11 | kink0 | and then ends with returns 0 |
| 22:22.14 | kink0 | that is all |
| 22:22.54 | kink0 | and the program runs as start and ends, not still running, just send an ATD command to the serial port and then end |
| 22:23.45 | kink0 | syzygyBSD, my purpose for this is, once I have gain control over how to manage my mobile phones from any console ( i.e. minicom ) , use an externa dialing program |
| 22:23.57 | asteriskmonkey | anyone have problems with asterisk hanging up when a caller is calling a cell phone and it goes to the cell phone vm system? |
| 22:24.01 | kink0 | my "serial" program is just this, a dialing program, more likely as chat or so |
| 22:24.08 | syzygyBSD | kink0: well the only issues like that I have had were permissions issues, it looks like you have those taken care of |
| 22:24.45 | kink0 | syzygyBSD, yes.. may be, but if I run asterisk as root, I think all apps called from AGI would be forked as root, right ? |
| 22:25.20 | syzygyBSD | I don't know that, maybe someone else can tell you for sure |
| 22:25.28 | kink0 | really I hate to run as root, but I try to use it in a way I can ensure there no problem accessing devices |
| 22:25.31 | *** join/#asterisk PupenoL (n=pupeno@200.123.183.89) |
| 22:25.36 | syzygyBSD | I would think that would be the case, but I can't say for sure |
| 22:26.09 | kink0 | syzygyBSD, hmmmm I will try to set permisions for ALL on those involved ( program, devices ) temporally to test it |
| 22:27.13 | kink0 | syzygyBSD, the same, even if I set chmod a+rw /dev/usb/ttyUSB0 |
| 22:27.18 | syzygyBSD | kink0: if it isn't too hard, I would have the first line in your program being a `cout << "First Line"` just to see if it ever hits that |
| 22:27.25 | syzygyBSD | sorry if my c syntax is off |
| 22:28.05 | kink0 | syzygyBSD, ahh ok, I will start then my program even as "strace .... > log.file" to see what is doing |
| 22:28.58 | kink0 | "OSS/dsp", "/usr/serial/serial -p /dev/usb/ttyUSB0 -s 115200 -c d657897123; > /tmp/dial.log" |
| 22:29.08 | kink0 | and dial.log is empty |
| 22:29.13 | kink0 | nothing happens |
| 22:29.25 | kink0 | appears my program is not excuted at all |
| 22:29.55 | kink0 | I have a doubt, the complete verbose report from asterisk says: |
| 22:30.01 | *** join/#asterisk Rawplayer (n=kevin@ipc31055d2.oom-killer.org) |
| 22:30.04 | kink0 | Failed to execute '/usr/serial/serial -p /dev/usb/ttyUSB0 -s 115200 -c d657897123; > /tmp/dial.log': No such file or directory |
| 22:30.04 | kink0 | <PROTECTED> |
| 22:30.12 | kink0 | No such file or directory ??? |
| 22:30.24 | kink0 | and after that, returning 0 ? |
| 22:30.45 | *** join/#asterisk Flauto (n=zhao@c-24-14-197-214.hsd1.il.comcast.net) |
| 22:31.16 | kippi | is anyone using xorcom rapid? |
| 22:31.44 | syzygyBSD | oh, you are trying to pass parameters to your program through astersik? |
| 22:32.13 | syzygyBSD | try to call it without those parameters |
| 22:32.37 | syzygyBSD | create a script to run that command, and call the script, who knows it may just work |
| 22:33.14 | syzygyBSD | can you paste the AGI line from extensions.conf? |
| 22:33.29 | *** join/#asterisk lesouvage (n=lesouvag@cc341200-a.assen1.dr.home.nl) |
| 22:34.36 | lesouvage | I just installed ziaxphone ( http://www.kauss.org/Stephan/ziaxphone/ ) and changed my zaurus pda in a wireless iax phone. The world is full of little miracles. |
| 22:34.52 | *** join/#asterisk PupenoL (n=pupeno@200.123.183.89) |
| 22:35.14 | kink0 | syzygyBSD, exten => _6XXXXXXXX,2,AGI(/usr/serial/serial -p /dev/usb/ttyUSB0 -s 115200 -c d6${EXTEN:${TRUNKMSD}}\; > /tmp/dial.log) |
| 22:35.33 | syzygyBSD | hah, wish i had asked for that earlier |
| 22:35.35 | kink0 | yes, I need to pass at least the extension number i call in asterisk |
| 22:36.09 | *** join/#asterisk copantl (n=galel@205.240.205.192) |
| 22:37.00 | copantl | hello |
| 22:37.33 | copantl | i have a problem with my te110p |
| 22:37.49 | syzygyBSD | in asterisk.conf there is a astagidir setting, the agi commands need to be there, I dont' think they can' be referenced with /usr/, also I don't know if it is possible to pass arguments like that, it may be looking for the agi command named everything |
| 22:37.58 | mog_work | whats wrong copantl |
| 22:38.09 | copantl | my PRI channels give me busy congested state |
| 22:38.33 | copantl | but i ask to my pstn and they say all channels of the E1 are clear |
| 22:38.35 | mog_work | are the bchannels up |
| 22:38.49 | mog_work | i mean you saw d an b channels come up at start of asterisk? |
| 22:39.00 | copantl | i can only get connected 4 channels of the 30 |
| 22:39.50 | copantl | if u type zap show channel 1 |
| 22:39.52 | *** join/#asterisk PupenoL (n=pupeno@200.123.183.89) |
| 22:40.48 | copantl | in the part of Actual HookState: what shuld be the state Onkook? |
| 22:41.14 | kink0 | syzygyBSD, I have copied the files to the astagidir, but the same |
| 22:41.29 | *** join/#asterisk genmud (n=genmud@ip68-98-82-206.ph.ph.cox.net) |
| 22:41.49 | *** join/#asterisk brookshire (n=nubb@gateway.digium.com) |
| 22:41.54 | copantl | im connected to a lucent switch |
| 22:42.19 | copantl | through a PRI/ISDN |
| 22:43.57 | syzygyBSD | kink0: are you still calling it with arguments? asterisk adds its own to agi commands |
| 22:44.22 | bbz | Would anyone here be interested in some Polycom 501 phones? We got stuck w/ an extra 20 after a project, and we need to unload them at cost---->>> <<<---- [sorry alot of people have been missing the cost of these] if anyone is interested send me a private message (if i dont respond immediately, leave me your email address, and i can send you info) i have a reference in the channel if you need. -- |
| 22:44.22 | copantl | mog_work: any idea? |
| 22:44.30 | trixter | the montana puc is kinda lame.. they mail all the LECs with everyone on the CC: header of the email so if one stupid person replkies to all you see everything.. course its notices of rule changes etc.. but meh its anoying plus its more than a page of CC |
| 22:44.40 | kink0 | yes, I call it with arguments, but also I have try to call a script that contains all possible arguments, with the same fail |
| 22:44.50 | chapeaurouge | bbz...a robot with AI.. ;) |
| 22:44.51 | trixter | I am actualy suprised how many clecs there are in montana, but then again I am one of them |
| 22:45.17 | trixter | bbz is more like a real client with a script ... cause he did respond to me after some time |
| 22:45.22 | chapeaurouge | trixter, you know centurytel? |
| 22:45.48 | trixter | vaguely why? |
| 22:45.49 | kink0 | syzygyBSD, I think the problem is not the arguments, I included all arguments in script called "llamar" and then the AGI calls to llamar program without arguments, but the same. |
| 22:46.17 | chapeaurouge | trixter, i used to work for them when I lived in the US, in Louisiana. I know they have quite a few LECs out there |
| 22:46.28 | chapeaurouge | was just curious |
| 22:46.39 | *** join/#asterisk Ivar1974 (n=r@static-70-19-119-112.ny325.east.verizon.net) |
| 22:46.45 | syzygyBSD | kink0: odd, don't know what to tell you |
| 22:46.55 | trixter | I havent done anything with them directly ... |
| 22:47.04 | trixter | I think I talked to em a little when I first formed my clec but that was it |
| 22:47.11 | chapeaurouge | ok |
| 22:47.24 | Ivar1974 | hi guys, I received the following bug where zap channel 17 froze: DEBUG[29324] chan_zap.c: Exception on 32, channel 17 |
| 22:47.42 | *** join/#asterisk shmaltz (n=chatzill@69.28.255.210) |
| 22:48.16 | Ivar1974 | then on the next line: WARNING[29324] chan_zap.c: We're Zap/17-1, not, it repeated this in the logs for about 10 minutes and the extention was dead |
| 22:48.57 | Ivar1974 | then: DEBUG[29868] channel.c: Avoiding initial deadlock for 'Zap/17-1, but it still was dead, I tried to stop, then kill * but couldn't so I just rebooted |
| 22:49.06 | kink0 | syzygyBSD, has you any call to AGI in your extensions ? |
| 22:49.09 | Ivar1974 | now its fine but I just want to know why it happened? |
| 22:49.17 | *** join/#asterisk chandi (n=burni13@modemcable248.1-201-24.mc.videotron.ca) |
| 22:49.33 | syzygyBSD | kink0: many |
| 22:50.17 | trixter | uhhh.. argentina has 15,000 routes :/ course that is a good thing since my original list only had 5500 or so for the whole world |
| 22:50.26 | kink0 | syzygyBSD, and there some of them that are calling your own scripts/programs ? |
| 22:50.40 | syzygyBSD | all of them |
| 22:50.45 | kink0 | may be even a syntax problem, since I got: |
| 22:50.47 | kink0 | Failed to execute '/var/lib/asterisk/agi-bin/serial -p /dev/usb/ttyUSB0 -s 115200 -c d657897123 > /tmp/dial.log': No such file or directory |
| 22:50.58 | kink0 | why not such file ? |
| 22:51.01 | syzygyBSD | exten => _.,3,AGI(ai_operator_busy.py) |
| 22:51.03 | kink0 | is there !! |
| 22:51.05 | *** join/#asterisk stkn (i=nobody@gentoo/developer/pdpc.active.stkn) |
| 22:51.11 | chandi | hey folks. I sometines need to use the application DIAL to dial to a phone gateway to my cell phone. It answers instantly and gives me another dialtone, then I send DTMFs to call my cell phone. Is there a way to hangup that call if the cell phone doesn't answer ?? Since the call has been answered alredy... I guess it's more complicated |
| 22:51.21 | *** part/#asterisk mkrufky (n=mk@68.160.103.77) |
| 22:51.35 | syzygyBSD | kink0: no it isn't, there is no file on your system named that, it is looking at everything as a command name, not a command name with argument |
| 22:51.39 | kink0 | syzygyBSD, ai_operator_busy.py is in /var/lib/asterisk/agi-bin ? |
| 22:51.54 | syzygyBSD | it is in the directory I setup in asterisk.conf |
| 22:52.01 | kink0 | syzygyBSD, but if I supress arguments , the same |
| 22:52.34 | syzygyBSD | what is your agi command when it is in agi-bin? |
| 22:52.39 | syzygyBSD | without arguments |
| 22:52.57 | syzygyBSD | and the error message that run gives |
| 22:53.28 | *** join/#asterisk p1tst0p (n=admin@82-38-106-54.cable.ubr03.donc.blueyonder.co.uk) |
| 22:53.35 | kink0 | yes, I place a program called "llamar" in my /var/lib/asterisk/agi-bin , that is the directory I have for agi in my asterisk.conf |
| 22:53.41 | kink0 | but I still got: |
| 22:54.00 | kink0 | <PROTECTED> |
| 22:54.00 | kink0 | Failed to execute '/var/lib/asterisk/agi-bin/llamar': Exec format error |
| 22:54.27 | p1tst0p | Hi, i wonder if anyone can help.. i cant see for looking whats happening.. i have got a broadvoice account configured in Asterisk, but when i dial the numbers, its missing 2 digits off.. like so ( [context] |
| 22:54.27 | p1tst0p | ;For incoming calls |
| 22:54.27 | p1tst0p | ;This extension line will ring SIP |
| 22:54.27 | p1tst0p | ;extension 2001 for 60 seconds then hang up. Modify as necessary to fit your dialplan |
| 22:54.27 | p1tst0p | exten => s,1,Dial(SIP/2001,60,tr) |
| 22:54.29 | p1tst0p | exten => s,2,hangup |
| 22:54.31 | p1tst0p | For outgoing calls |
| 22:54.33 | p1tst0p | ;Pattern match for local call plan, use appropriate pattern if on nationwide plan. |
| 22:54.35 | p1tst0p | exten => _1NXXNXXXXXX,1,dial(SIP/${EXTEN}@sip.broadvoice.com,30) |
| 22:54.37 | p1tst0p | exten => _1NXXNXXXXXX,2,congestion() |
| 22:54.39 | p1tst0p | oh |
| 22:54.41 | p1tst0p | im really sorry |
| 22:54.43 | kink0 | then says Exec error, even is excutable -rwxr-xr-x 1 root root 65 2005-11-23 23:35 llamar |
| 22:55.37 | p1tst0p | Executing Dial("SIP/4403-783e", "SIP/0114411325727@sip.broadvoice.com|30") in new stack < -- this is what i was meant to paste... i am missing 2 digets when trying to dial to broadvoice ! its giving me a headache ! |
| 22:55.44 | syzygyBSD | kink0: is the first line of that script #!/bin/sh |
| 22:56.10 | kink0 | syzygyBSD, no, is not calling any sh, I can try to run under sh |
| 22:56.34 | syzygyBSD | kink0: it doesn't know how to execute that program, add that and try |
| 22:57.09 | kink0 | apsss !! now appears to run !!! |
| 22:57.18 | syzygyBSD | all of my python files require #!/bin/python2 in order to run, I had errors when they didn't |
| 22:57.34 | kink0 | ok , that was !! asterisk does not know about what shell to use |
| 22:57.35 | syzygyBSD | one problem down |
| 22:57.50 | kink0 | syzygyBSD, not at all... now I need to pass the variables |
| 22:58.02 | syzygyBSD | easy enough |
| 22:58.19 | syzygyBSD | all the channel variables are always passed |
| 22:58.33 | syzygyBSD | so set the arguments as channel variables |
| 22:58.45 | syzygyBSD | can you read python and decypher? |
| 22:58.52 | kink0 | syzygyBSD, hmmmmm... how to get the ${EXTEN:${TRUNKMSD}} value in a C program ? |
| 22:59.07 | asteriskmonkey | anyone in here with pri/asterisk experience? |
| 22:59.18 | kink0 | yes, I can read python, but I have not experience with py programing access to serial ports |
| 22:59.23 | asteriskmonkey | ah tzanger you on? |
| 23:00.48 | sm7syx | anyone that can help with a dailplan? |
| 23:00.50 | *** join/#asterisk nagl (n=nagl@213.235.241.6) |
| 23:02.06 | p1tst0p | has anyone got any ideas why i cant send 15 digets out to my sip provider! asterisk stops me at 13.. |
| 23:02.19 | syzygyBSD | kink0: http://pastebin.ca/30752 that is a couple lines of how to read in all the channel variables (including exten) in python |
| 23:03.19 | syzygyBSD | I think it prefixes them with "agi_" too |
| 23:03.59 | syzygyBSD | asteriskmonkey: what do you need help with |
| 23:04.19 | syzygyBSD | p1tst0p: how does asterisk stop you? |
| 23:05.10 | kink0 | syzygyBSD, I see, thanks... going to touch some lines at my c code and try it. |
| 23:05.15 | p1tst0p | syzygyBSD, as soon as i get to the 13'th digit, it sends, Executing Dial("SIP/4403-783e", "SIP/0114411325727@sip.broadvoice.com|30") in new stack |
| 23:05.16 | *** join/#asterisk bjohnson (n=bjohnson@i216-58-14-207.cybersurf.com) |
| 23:05.32 | *** join/#asterisk Qwell[] (n=chatzill@pool-71-108-28-219.lsanca.dsl-w.verizon.net) |
| 23:05.48 | syzygyBSD | p1tst0p: can you paste the offending lines of your extensions.conf |
| 23:06.51 | _DAW | does anyone here use the page cmd in *? I am wondering if it is possible to set alert_info like you can with Dial. I am using Polycom ip 500's and they need alert info to auto answer. |
| 23:07.24 | p1tst0p | syzygyBSD, this is all im using, dial 8 to go over broadvoice.. exten => _8.,1,Dial(SIP/${EXTEN}@sip.broadvoice.com,60) |
| 23:08.15 | bancus | does anyone know if it's possible to make it so only one person can be using the outgoing broadvoice connection at a time? |
| 23:08.20 | syzygyBSD | hah, that isn't the real one p1tst0p |
| 23:08.32 | syzygyBSD | at the very least it is ${exten:1} |
| 23:08.37 | p1tst0p | ah, ok |
| 23:08.59 | syzygyBSD | maybe . is only catching up to 12 numbers, try .. |
| 23:09.08 | Qwell[] | ..? |
| 23:09.11 | Qwell[] | no |
| 23:09.44 | bancus | exten => _91NXXNXXXXXX, 1, dial(SIP/${EXTEN:1}@sip.broadvoice.com,30) |
| 23:09.54 | bancus | that's part of my broadvoice dialing plan |
| 23:10.04 | bancus | which WFM |
| 23:10.12 | p1tst0p | thanks bancus |
| 23:10.29 | bancus | I have similar rules to catch numbers dialled without area code |
| 23:10.38 | bancus | and a bunch of international calling rules |
| 23:10.49 | *** join/#asterisk bweschke (n=bweschke@pcp09754274pcs.narlington.nj.comcast.net) |
| 23:10.49 | bancus | exten => _9NXXXXXX, 1, dial(SIP/1909${EXTEN:1}@sip.broadvoice.com,30) |
| 23:10.56 | bancus | that's how to do the non-area code |
| 23:11.01 | bancus | replace 909 with your own area code |
| 23:11.09 | Qwell[] | eww, 909? |
| 23:12.07 | p1tst0p | what do u mean, own area code ? you mean country code ? |
| 23:14.36 | syzygyBSD | p1tst0p: his dialplan is setup for the U.S. that was for his issue |
| 23:14.51 | syzygyBSD | I think... |
| 23:14.54 | syzygyBSD | maybe not |
| 23:15.33 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 23:16.12 | asteriskmonkey | anyone know how to pass an ext on and cut of the first digit? |
| 23:16.22 | Nivex | ${EXTEN:1} |
| 23:16.27 | Flauto | hehe |
| 23:16.45 | asteriskmonkey | really? |
| 23:16.54 | file[laptop] | yup |
| 23:17.26 | asteriskmonkey | so ${EXTEN:2} would cut of the first 2 digits? |
| 23:17.33 | GXTi | bingo |
| 23:17.33 | p1tst0p | syzygyBSD, i think its my hardphone, |
| 23:17.35 | file[laptop] | scary eh? |
| 23:17.36 | _DAW | does alert_info work with page cmd? |
| 23:17.55 | _DAW | from what i can tell it doesnt |
| 23:17.57 | p1tst0p | syzygyBSD, because that stops at 12.. and doesnt display anymore |
| 23:19.14 | *** join/#asterisk tessier (n=treed@wsip-68-15-4-13.sd.sd.cox.net) |
| 23:19.25 | syzygyBSD | try changing your line to exten => _9XXXXXXXXXXXXX ... However many digits you want to dial, just to see if it works |
| 23:19.52 | asteriskmonkey | ooo... so for making an 800 dialling context it would like like this, since my local pri can call toll free with no 1's :) exten => _1NXXXXXXXXX,1,Dial(Zap/g0/${EXTEN:1}) ? |
| 23:20.25 | asteriskmonkey | probably better like this exten => _18XXXXXXXXX,1,Dial(Zap/g0/${EXTEN:1}) right? |
| 23:20.50 | syzygyBSD | asteriskmonkey: that should work, though any 11 digit number that starts with a 1 would have it striped |
| 23:21.06 | syzygyBSD | 2nd one looks better |
| 23:23.05 | p1tst0p | syzygyBSD, sorry im confused with _1NX .. what would one dial to make astersisk catch _1NXXXX ? |
| 23:23.05 | asteriskmonkey | should probably make 3 of them to make 800 888 and 877 .. just incase there is ld is 1-8xx |
| 23:24.48 | syzygyBSD | p1tst0p: what do you want it to do, I am following too many questions right now |
| 23:25.34 | Qwell[] | 10 X's? |
| 23:25.37 | Qwell[] | silly people |
| 23:25.47 | mog_work | thats why im all about _X. |
| 23:25.48 | Qwell[] | That would allow me to dial international |
| 23:25.51 | mog_work | bump everything else |
| 23:26.20 | syzygyBSD | mog_work: how many digits does . accept? |
| 23:26.27 | Qwell[] | 0 or 500000000000000000000000 |
| 23:26.28 | syzygyBSD | I would think as many as you type... |
| 23:26.30 | Qwell[] | anywhere in between |
| 23:26.49 | p1tst0p | sorry mate.. im having trouble dialing out to broadvoice.. which if i understand correctly, you need to dial.. 011, then the country code, then the number... so for me im testing say, a number to the uk.. this would be 011441302123456 that makes, 15 digits.. but either my phone, or asterisk is stopping me at 12. |
| 23:26.49 | syzygyBSD | but in his setup it is only accepting 12 |
| 23:26.59 | Qwell[] | syzygyBSD: for other reasons, I'm sure |
| 23:27.07 | Flauto | i think it is his phone |
| 23:27.09 | Qwell[] | p1tst0p: It's your phone |
| 23:27.12 | Flauto | not his dialplan |
| 23:27.47 | p1tst0p | im using a Avaya 4602 IP.. and you guys could be right. cuz the LCD stops when it sends the dial string to asterisk |
| 23:27.49 | syzygyBSD | and the real experts have answered |
| 23:28.02 | Flauto | he has to setup his phone to allow him dialing more digits |
| 23:28.04 | asteriskmonkey | so whats the format for say this digit followed by anynumber of digits? |
| 23:28.05 | Qwell[] | p1tst0p: You can usually change the dialplan on the phones, but Avaya is crap |
| 23:28.11 | Qwell[] | so you may or may not be able to |
| 23:28.24 | Qwell[] | asteriskmonkey: Which digit? |
| 23:28.41 | asteriskmonkey | say 1 followed by any number of digits |
| 23:28.45 | Qwell[] | _1. |
| 23:28.56 | Qwell[] | but that is silly |
| 23:29.05 | Qwell[] | What are you trying to do? |
| 23:29.05 | *** part/#asterisk Ivar1974 (n=r@static-70-19-119-112.ny325.east.verizon.net) |
| 23:29.12 | asteriskmonkey | so _ means make sure it meats exact then . means any number of digits is that right? |
| 23:29.24 | Qwell[] | _ means that it should do pattern matching |
| 23:29.52 | asteriskmonkey | cool so if you have some similar patterns in your dial plan it dosnt matter what order it will pick the closest right? |
| 23:30.00 | Qwell[] | no |
| 23:30.00 | p1tst0p | Qwelll, so is there a solution using the dial plan.. say maybe.. i dunno prefix each country with a digit or 2, then use that, which would send my 011+country code rather than the phone sending it... if you get what i mean ! hehe. |
| 23:30.08 | Qwell[] | It'll pick exact matches only |
| 23:30.20 | Qwell[] | p1tst0p: find the avaya docs |
| 23:30.34 | *** join/#asterisk robl^ (n=robl@dsl093-025-118.hou1.dsl.speakeasy.net) |
| 23:31.08 | sm7syx | How can I get Asterisk to mulitask? |
| 23:31.09 | *** join/#asterisk _mwoodj_ (n=mwoodj@pdpc/sponsor/digium/hyper-eye) |
| 23:31.23 | Qwell[] | sm7syx: rephrase the question |
| 23:32.26 | sm7syx | Qwell[], I would lite te hear it ringing and at the same time play a sound |
| 23:32.38 | Qwell[] | sm7syx: explain the scenario |
| 23:33.05 | wunderkin | ringing and play a sound.. together? |
| 23:33.18 | Qwell[] | ringing is a sound. problem solved. :D |
| 23:33.26 | syzygyBSD | lol |
| 23:33.37 | syzygyBSD | or the Background() command |
| 23:34.01 | sm7syx | Qwell[], Someone is calling me and I want them to hear the ring signal, but I also would like to give some information while they wait. |
| 23:34.29 | sm7syx | Can I post part of my extensions.conf? |
| 23:34.33 | Qwell[] | ~pb |
| 23:34.35 | jbot | extra, extra, read all about it, pastebin is a place to paste your stuff without flooding the channel - try http://pastebin.ca/ |
| 23:35.21 | sm7syx | OK, got the picture, I'm ne wto this.... |
| 23:35.45 | Rawplayer | anyone in here using /etc/init.d/skeleton as start-stop script for asterisk? |
| 23:37.54 | sm7syx | Qwell[] please take a look at http://pastebin.ca/30769 |
| 23:38.41 | Qwell[] | okay, whats it not doing? |
| 23:39.22 | Qwell[] | sm7syx: You could use the M() option to dial to run a macro while the phones are ringing |
| 23:39.32 | *** join/#asterisk Iam8up (n=iam8up@user-0cdv2kk.cable.mindspring.com) |
| 23:39.51 | sm7syx | It starts with the Background not ringing before whitch confuses people. |
| 23:40.03 | Qwell[] | show application dial |
| 23:40.23 | Qwell[] | damnit, I forgot my iaxy again |
| 23:40.29 | sm7syx | Qwell[] M() option ? In dail ?? |
| 23:40.33 | Qwell[] | yes |
| 23:41.12 | sm7syx | Tnx will read and try, U here for some time if I run into problem? |
| 23:41.58 | Qwell[] | probably, but there are others who can help you too |
| 23:43.14 | sm7syx | OK, thanks. |
| 23:44.02 | *** join/#asterisk Sedorox (i=brandon@smartserv/cna/Sedorox) |
| 23:52.32 | p1tst0p | Qwell, sorted it buddy ! |
| 23:52.41 | p1tst0p | Qwell, the Avaya 4602 has a web interface |
| 23:52.58 | p1tst0p | which had a "Dial Plan" option, which was set to, xxxxxxxxxxx ( 12 ) ! |
| 23:55.37 | Pete_Largo | can anyone tell me how to switch voices in festival? |
| 23:55.54 | syzygyBSD | you can do that? |
| 23:55.56 | Pete_Largo | I just read the festival docs and now my head hurts... |
| 23:55.58 | syzygyBSD | hmm |
| 23:56.38 | syzygyBSD | might have to try sometime |
| 23:56.38 | Pete_Largo | well, I assume so since there are about a dozen voices available for download from the festival site... |
| 23:58.31 | Qwell[] | "voices", heh |
| 23:58.47 | Qwell[] | different pitch of monotony |