| 00:00.00 | jameswf-home | a dirt exchange if you will |
| 00:00.04 | jameswf-home | :) |
| 00:01.20 | jameswf-home | the real question is how much wood could a wood-chuck chuck if a wood-chuck could chuck wood |
| 00:01.39 | drmessano | Security holes are funny.. It can be critical, or critical, and somehow have two different sizes |
| 00:01.57 | *** join/#asterisk MaliutaBris (n=nikolai@kiev.lusan.id.au) |
| 00:02.28 | MaartenB | what security hole are you guys talking about? |
| 00:02.32 | drmessano | HUMUNGOID GIGANTO SECURITY WORMHOLE FOUND IN LINUX KERNEL |
| 00:02.52 | drmessano | http://it.slashdot.org/article.pl?sid=08/02/10/2011257 |
| 00:02.57 | *** join/#asterisk MaliutaWrk (n=nikolai@kiev.lusan.id.au) |
| 00:03.16 | drmessano | EVERYONE, QUICK.. SHUTDOWN -NOW |
| 00:03.29 | CVirus | huh |
| 00:03.50 | drmessano | I think this was making the rounds last night |
| 00:03.54 | drmessano | So it's a day old |
| 00:03.58 | *** part/#asterisk PepOSX (n=angeldav@190.72.132.46) |
| 00:04.03 | Greek-Boy | what is the best practice of calling voicemail for internal users and to avoid them having to put in their mailbox number? VoiceMailMain(${CALLERID(num)}) ? |
| 00:04.43 | Greek-Boy | what happens if a user without a mailbox dials that? will it ask for a mailbox number in that case? |
| 00:07.48 | lmadsen | Greek-Boy: yes |
| 00:08.11 | Greek-Boy | thanks, sorry for all the questions |
| 00:08.18 | Greek-Boy | I am checking out the wiki |
| 00:08.18 | lmadsen | you could always just try it... |
| 00:08.26 | Greek-Boy | but some stuff are outdated |
| 00:08.32 | lmadsen | I wish more people would try things out and learn |
| 00:08.35 | Greek-Boy | especially with all the deprecated stuff |
| 00:08.43 | lmadsen | yes... the wiki is super outdated |
| 00:08.56 | drmessano | If only there was a good book |
| 00:09.00 | lmadsen | if only |
| 00:09.06 | drmessano | *sigh* |
| 00:09.11 | hmodes | someone should write one... |
| 00:09.17 | lmadsen | might as well just switch to trixbox |
| 00:09.17 | drmessano | YES! |
| 00:09.19 | lmadsen | asterisk is useless |
| 00:09.28 | drmessano | :( |
| 00:09.29 | lmadsen | and trixbox has the worlds largest asterisk community |
| 00:09.44 | lmadsen | at least that's what their advertising says |
| 00:09.49 | ManxPower | "A sucker is born every minute." -- PT Barnum |
| 00:10.15 | drmessano | Yes, trixbox has more asterisk users than asterisk |
| 00:10.35 | drmessano | *sigh* |
| 00:10.50 | drmessano | I just installed asterisk... now what? |
| 00:11.02 | Greek-Boy | :) |
| 00:11.48 | completely_phukt | phukin' polaks found the hole in linux. I am compiling the code now.... |
| 00:11.48 | drmessano | lmadsen, wouldn't someone be better off writing a trixbox book? |
| 00:14.46 | hax | lmadsen: it doesn't look like i'm going to be able to get ztdummy to run... is there a list of things i won't be able to do without that module? |
| 00:15.12 | drmessano | Why cant you get it to run? |
| 00:15.18 | lmadsen | anything with timing... which is mostly: app_meetme, iax2 trunking, and probably something else |
| 00:15.23 | *** join/#asterisk ahbritto (n=guest@adsl-69-104-3-183.dsl.pltn13.pacbell.net) |
| 00:15.26 | lmadsen | but it shouldn't build fine, even on a xen enabled kernel |
| 00:15.46 | hax | drmessano: google tells me it won't work on a uml server |
| 00:16.07 | JT | uml.... |
| 00:16.14 | JT | don't run any real time stuff on that |
| 00:16.17 | JT | or near real time |
| 00:16.20 | hax | ? |
| 00:16.40 | JT | it's a much lesser form of virtualisation than xen |
| 00:16.42 | JT | etc |
| 00:17.19 | hax | it seems to work fine, i host with linode.com, and i haven't seen anything unexpected or laggy yet |
| 00:17.30 | JT | yes i have a linod too |
| 00:17.33 | JT | +e |
| 00:17.38 | JT | would never use it for voip |
| 00:17.42 | JT | uml is completely unsuitable |
| 00:17.53 | teknoprep | JT, from what i hear para-virtualization with asterisk is actually pretty good |
| 00:17.58 | JT | you might get away with it for very light load |
| 00:18.00 | JT | teknoprep: yes |
| 00:18.12 | JT | teknoprep: but uml is nothing like proper paravritualisation |
| 00:18.25 | jameswf-home | anyone seen this.... http://www.venturevoip.com/news.php?rssid=801 |
| 00:18.30 | teknoprep | JT, xen is so much better then vmware server |
| 00:18.31 | *** join/#asterisk tzafrir_laptop (n=tzafrir@62.90.10.53) |
| 00:18.41 | teknoprep | JT, but vmware infrastucture 3.0 is the shiznuts |
| 00:18.48 | *** join/#asterisk tzafrir (n=tzafrir@62.90.10.53) |
| 00:18.52 | hax | JT: i think it's probably worth a try, i mean, i'll know if it doesn't work, right? |
| 00:19.05 | drmessano | Its already doesnt work |
| 00:19.08 | JT | hax: but it might work today and not tomorrow |
| 00:19.09 | drmessano | You cant use ztdummy |
| 00:19.13 | JT | if the shared load goes up |
| 00:19.21 | teknoprep | hax, honestly i would try using Xen or VMware server 2.0 beta |
| 00:19.38 | teknoprep | hax, although XEN is much better for paravirt |
| 00:19.49 | teknoprep | hax, vmware server 2.0 beta supports para-virt |
| 00:20.22 | completely_phukt | hey guys, don't worry about the hole in linux. it ONLY works if u r not root or member of root user group. So, everybody QUICKLY log on as root |
| 00:20.38 | completely_phukt | if not already |
| 00:21.33 | jameswf-home | whew ok i am root now wha |
| 00:22.01 | hax | teknoprep: from what i can tell, ztdummy doesn't want to run on xen either |
| 00:22.07 | drmessano | lol |
| 00:22.21 | drmessano | MAH BOXEN R SAFE NOW |
| 00:22.26 | drmessano | ASTERISK CAT APPROVES |
| 00:22.43 | JT | i have co-located servers for voip stuff |
| 00:23.56 | teknoprep | hax, it runs on a para-virtualized xen install |
| 00:24.40 | JT | perhaps use stuff that does need zap timing ;) |
| 00:24.47 | JT | makes life so much easier |
| 00:25.22 | hax | teknoprep: is there more than one kind of xen install? |
| 00:25.39 | teknoprep | hax, you can either use para-virtualization or full-virtualization |
| 00:25.52 | *** join/#asterisk [Latino] (n=rabs@212.40.232.9.static.user.ono.com) |
| 00:25.53 | JT | you need cpu support for paravirtualisation |
| 00:25.59 | JT | it's far superior |
| 00:26.00 | [Latino] | hi all |
| 00:26.15 | JT | new xeons and opterons and probably some others have said support |
| 00:26.16 | teknoprep | hax, i suggest you install CentOS 5.1 with the gnome server-gui and virtualization check boxes checked |
| 00:26.29 | teknoprep | JT, i thought it was for full virtualization you needed special CPU's |
| 00:26.35 | teknoprep | JT, amd-v or intel-v |
| 00:26.40 | hax | teknoprep: i can't, i'm just a lowly VPS subscriber, not a provider :) |
| 00:26.42 | teknoprep | JT, para is supported on almost ALL cpu's |
| 00:26.51 | JT | maybe i've got my terms mixed up |
| 00:26.53 | teknoprep | hax, centos 5.1 is free |
| 00:26.59 | teknoprep | JT, i am pretty sure you do |
| 00:27.22 | teknoprep | JT, xen requires the hardware hypervisor for full-virtualiztion |
| 00:27.46 | hax | teknoprep: yeah, but my provider uses UML |
| 00:27.55 | teknoprep | hax, use a better provider ? |
| 00:27.59 | hax | heh |
| 00:28.18 | teknoprep | hax, use a better provider for just one server ? |
| 00:28.31 | teknoprep | hax, to tell you the truth.. i would just use a colocated "real" box for you service |
| 00:28.49 | hax | teknoprep: yeah, i know, but all tha is expensive, and i'm trying to be cheap |
| 00:28.49 | teknoprep | hax, if you want colocation for asterisk |
| 00:28.57 | teknoprep | hax, it is very cheap to do this |
| 00:29.10 | teknoprep | hax, talking less than 100$ per month at the planet |
| 00:29.37 | hax | yeah, plus a server |
| 00:29.38 | teknoprep | wow this FRIS vodka is really really good |
| 00:29.45 | JT | the planet, didn't they have a massive outage? |
| 00:29.48 | JT | or was that rackspace? |
| 00:29.55 | hax | that was theplanet |
| 00:30.07 | teknoprep | hax, i run an office with 15 VoIP channels from Bandwidth.com on a dual p3 866 server |
| 00:30.09 | JT | yeah, lame power backup design |
| 00:30.22 | *** join/#asterisk hmm-home (n=hmm-home@24-119-176-74.cpe.cableone.net) |
| 00:30.23 | *** join/#asterisk putnopvut (n=putnopvu@user-24-214-112-81.knology.net) |
| 00:30.48 | JT | seriously... chillers going offline during a power outage... ... |
| 00:30.50 | teknoprep | hax, you can get a nice hp LP p3 1ghz machine for 130$ on ebay |
| 00:30.54 | [Latino] | could someone tellme if there is any way of setting the source IP for the RTP of and expecific peer ? |
| 00:31.05 | hax | hmm |
| 00:31.08 | jameswf-home | ~fish |
| 00:31.09 | jbot | i guess fish is FISHFISHFISH! DO THE FISH DANCE! "Give a man a fish and you'll feed him a day. Teach him how to fish and he'll feed himself for the rest of his life." This is so appropriate, instead of asking us to tell you exactly what to do, why not read some docs, then come back and ask specific questions which aren't covered?, or ... |
| 00:31.41 | hmm-home | oh trilian how great are thee |
| 00:31.55 | lmadsen | what a bitch... had to run rpmbuild on the php src rpm along with a million dependencies to build the mssql.so module so that I could write a vm-pin-change.php script for my ODBC enabled voicemail so when someone updates their voicemail pin from app_voicemail it'll also update the MSSQL database so that change is saved. Working though, w00t :) |
| 00:32.25 | drmessano | lol |
| 00:32.54 | jameswf-home | lmadsen: Ron paul could have done than in 5 minutes in pen |
| 00:33.04 | teknoprep | lol |
| 00:33.09 | lmadsen | I'm certainly no Ron Paul :) |
| 00:33.10 | jameswf-home | s/than/that/ |
| 00:33.13 | drmessano | on a napkin |
| 00:33.18 | teknoprep | 60 minutes has hillary on |
| 00:33.19 | *** part/#asterisk Greek-Boy (n=email@41.221.58.4) |
| 00:33.20 | tzanger | jameswf-home: heh, I like "give a man a password and he'll log in for the day. Teach him to hack and he'll log in whenever he wants." |
| 00:33.29 | hax | hmm |
| 00:33.35 | teknoprep | i want obama but i wouldn't mind if hillary wins |
| 00:33.45 | drmessano | Ron Paul can compile a napkin with code written on it |
| 00:33.46 | jameswf-home | I can hack iny box with 3 minutes and a sharp axe |
| 00:33.51 | lmadsen | I'm just happy one of the Bush daughters isn't running for president |
| 00:34.03 | teknoprep | lol |
| 00:34.12 | jameswf-home | huh huh bush daughters huh huh |
| 00:34.25 | teknoprep | i am happy they didn't overturn the admendment that only allows for 2 full terms |
| 00:34.37 | *** part/#asterisk tuxfoo (n=tuxfoo@pool-72-65-149-192.chrlwv.east.verizon.net) |
| 00:34.54 | teknoprep | i would have to leave the country if they re-elected bush |
| 00:35.00 | jameswf-home | december 31st 2008 bush calls marshal law..... ohhhh snap |
| 00:35.10 | drmessano | Yep |
| 00:35.14 | drmessano | I was just gonna say lol |
| 00:35.26 | teknoprep | lol |
| 00:35.27 | drmessano | It ain't over yet |
| 00:35.30 | putnopvut | January 1st, an armed rebellion burns Washington to the ground... |
| 00:35.47 | coppice | bush is a wishy washy liberal. a strong leader would have scraped elections :-) |
| 00:35.49 | drmessano | Don't count your chicken.. because the CIA already is |
| 00:35.53 | teknoprep | we need like 10 million american's to change our government |
| 00:35.56 | jameswf-home | January 2nd Ron paul new president |
| 00:36.15 | drmessano | We need a DIGGolution |
| 00:36.25 | teknoprep | we need voting booths that work? |
| 00:36.33 | lmadsen | we need smarter americans |
| 00:36.38 | jameswf-home | damn a hanging chad |
| 00:36.56 | teknoprep | smarter american... thats a horrible oxymoron |
| 00:36.58 | lmadsen | the impossible dream I suppose |
| 00:37.14 | drmessano | We need voting machines with better security than a Wal Mart filing cabinet |
| 00:37.31 | teknoprep | haha |
| 00:37.57 | coppice | americans are no dumber than the people of any other country. they just have a strong cultural need to behave dumber |
| 00:38.01 | teknoprep | honestly.. the govn't can't come up with something better then the security of windows 95 directly connected to the inet ? |
| 00:38.02 | drmessano | I'm not sure where the joke is there.. the lack of security, or the fact that they use the same key |
| 00:38.04 | lmadsen | welp, I'm done working for the day finally... time to chill on the couch with mary and maybe some of the UK version of The Office |
| 00:38.23 | putnopvut | lmadsen, which series? |
| 00:38.31 | lmadsen | putnopvut: uhh.... The Office... :) |
| 00:38.46 | lmadsen | the UK version was like... 6 episodes or something... and is about a million times better than the US version |
| 00:38.48 | putnopvut | Yeah, series is what the Brits call a season. |
| 00:38.51 | lmadsen | (us version was based on it) |
| 00:39.02 | lmadsen | putnopvut: ahhh.. I thought there was only 1? |
| 00:39.09 | putnopvut | Nope, two. |
| 00:39.12 | putnopvut | I've seen both. |
| 00:39.24 | lmadsen | ooo... ! then there will be some I haven't seen then! |
| 00:39.31 | lmadsen | I've only seen the 1st one |
| 00:39.41 | lmadsen | this one seems to have a christmas special too |
| 00:39.51 | drmessano | I still think Roy, the guy that works in the warehouse, is the funniest dude ever |
| 00:39.53 | putnopvut | Yep, there was a Christmas special after the second one too. |
| 00:40.01 | lmadsen | hawtness |
| 00:40.16 | putnopvut | I think the second season of the American Office was about as funny as any show I've seen in recent memory. |
| 00:40.35 | drmessano | Crap |
| 00:40.38 | drmessano | Not Roy, Darryl |
| 00:40.47 | drmessano | Him and Creed |
| 00:41.50 | drmessano | Darryl asking Michael for a raise.. and he shows Darryl his own paystub to jusitfy not giving him one.. so Darryl snaps a camera phone shot of it and sends it to one of his buddies |
| 00:41.53 | drmessano | Hardcore |
| 00:42.31 | *** part/#asterisk completely_phukt (n=chatzill@static-72-77-217-74.tampfl.dsl-w.verizon.net) |
| 00:44.19 | *** join/#asterisk PepOSX (n=angeldav@190.72.132.46) |
| 00:48.08 | jameswf-home | office space is on woohooo |
| 00:48.21 | JT | yeah |
| 00:48.31 | JT | if you could just go right ahead and keep watching |
| 00:48.33 | JT | that'd be great |
| 00:50.00 | drmessano | Two chicks at once |
| 00:50.34 | sbingner | chicks with dicks? |
| 00:50.44 | teknoprep | now that sounds really fun |
| 00:50.47 | sbingner | idk why i said that |
| 00:50.48 | sbingner | wtf |
| 00:52.26 | hax | JT: so if i find a xen box that loads ztdummy... that should be an acceptable platform to run the pbx on, right? |
| 00:53.18 | JT | hax: are you using something that requires zaptel timing? |
| 00:53.54 | hax | JT: well, i don't really know, i guess i'd like to be able to do a conference call |
| 00:54.14 | JT | there's app_conference |
| 00:54.53 | hax | hmm, that looks promising |
| 00:55.02 | hax | maybe i actually can get away with not need ztdummy |
| 00:55.35 | hax | *not needing |
| 01:03.30 | jameswf-home | so there it was 2 girls and 1 cup |
| 01:03.43 | *** part/#asterisk techie (n=techie@adsl-76-214-27-74.dsl.lsan03.sbcglobal.net) |
| 01:04.24 | jameswf-home | there is no paperjam... |
| 01:04.27 | *** join/#asterisk techie (n=techie@adsl-76-214-27-74.dsl.lsan03.sbcglobal.net) |
| 01:11.35 | JT | jameswf-home: i was utterly disappointed by 2 girls 1 cup |
| 01:11.41 | JT | everyone talked it up too much |
| 01:11.57 | hax | JT: is there something like app_conference but for music on hold, which wouldn't require ztdummy? |
| 01:12.05 | *** join/#asterisk mmurdock (n=TGA@c-24-10-190-87.hsd1.co.comcast.net) |
| 01:12.07 | pkunkra | watched five seconds..... wanted to vomit |
| 01:12.15 | JT | hax: i don't think so |
| 01:12.17 | JT | pkunkra: lame |
| 01:12.21 | hax | pkunkra: you're new to the internet, eh? |
| 01:12.26 | JT | it's just chocolate moouse |
| 01:12.32 | JT | it was obviously fake |
| 01:12.37 | pkunkra | oh |
| 01:13.01 | pkunkra | as i said.... watched it for five seconds. |
| 01:13.03 | drmessano | Best comment on Digg ever.. when the ran the story about it being fake: |
| 01:13.05 | hax | i don't know if that makes it any better |
| 01:13.07 | pkunkra | not enough time to tell. |
| 01:13.13 | drmessano | "You mean I threw up in my trashcan at work for nothing??" |
| 01:13.23 | jameswf-home | 2 girls 1 finger? |
| 01:13.35 | pkunkra | drmessano, nah..... utterly grossed out. |
| 01:13.51 | *** part/#asterisk techie (n=techie@adsl-76-214-27-74.dsl.lsan03.sbcglobal.net) |
| 01:13.56 | drmessano | 2 girls, 1 PBX |
| 01:14.04 | *** join/#asterisk techie (n=techie@adsl-76-214-27-74.dsl.lsan03.sbcglobal.net) |
| 01:14.17 | JT | i guess some people just have weak stomachs |
| 01:14.22 | pkunkra | i heard of a professor that gave a student an F because he wrote about that video. |
| 01:14.29 | putnopvut | lol |
| 01:14.49 | drmessano | All that crap is fake |
| 01:14.53 | pkunkra | drmessano, i think i'd be much more interested in 2 girls, 1 PBX. |
| 01:15.06 | *** join/#asterisk russellb (n=russell@asterisk/developer-and-stable-maintainer/drumkilla) |
| 01:15.06 | *** mode/#asterisk [+o russellb] by ChanServ |
| 01:15.08 | drmessano | "Take the case of 2 girls, 1 cup for example..." <--- Fake |
| 01:15.08 | JT | that would be easy if gumstix based |
| 01:15.12 | pkunkra | 2 girls trying to hack up an asterisk dialplan. |
| 01:15.20 | pkunkra | now, that's sexy! |
| 01:15.54 | pkunkra | i think they'd give up and start talking about their hair instead. |
| 01:17.43 | jameswf-home | 2girls 1 pbx http://itknowledgeexchange.techtarget.com/networkhub/files/2007/10/200_trixbox.jpg |
| 01:18.04 | drmessano | HA |
| 01:18.05 | pkunkra | nice. looks like sales bunnies. |
| 01:18.07 | *** join/#asterisk esaym (n=user@cpe-72-183-198-134.satx.res.rr.com) |
| 01:18.55 | russellb | nothing like whoring out your girlfriend at a tradeshow to get attention to your booth |
| 01:19.17 | jameswf-home | lol yeah |
| 01:20.27 | pkunkra | russellb, tried and true... ;-) |
| 01:21.03 | jameswf-home | sadly I didnt see what that picture had t o do with the article http://itknowledgeexchange.techtarget.com/networkhub/files/2007/10/200_trixbox.jpg |
| 01:22.45 | jameswf-home | she bent over and pooped out a ringgroup |
| 01:27.02 | *** join/#asterisk metfan2007 (n=metfan20@189.180.217.155) |
| 01:27.16 | metfan2007 | hi all!! anyone has implemented VICIDIAL? |
| 01:27.53 | drmessano | ROFL |
| 01:28.31 | *** join/#asterisk AndyGraybeal (n=andy@node54.32.251.72.1dial.com) |
| 01:29.42 | *** join/#asterisk egypcio (n=vinicius@unaffiliated/egypcio) |
| 01:29.55 | jameswf-home | wtf is david kullmann |
| 01:30.46 | drmessano | http://www.davidkullmann.com/ <-- that guy |
| 01:32.12 | jameswf-home | yeah wtf is he |
| 01:32.43 | russellb | ewww, the green thing blinded me |
| 01:33.57 | drmessano | I didnt realize trixbox rocked that hard |
| 01:34.00 | drmessano | Wait, no |
| 01:35.49 | metfan2007 | do you know if vicidial supports asterisk 1.4? |
| 01:36.22 | hmm-home | why would you want to even try that sounds like a nightmare |
| 01:37.02 | metfan2007 | why do you think so? |
| 01:37.10 | jameswf-home | I cant wait to use the idolizer accros 2 t1's |
| 01:37.16 | metfan2007 | vicidial is a great app, or do you know something better? |
| 01:37.48 | hmm-home | i didn't say vicidial was bad, I said trying to use it with versions other than what they recommend sounds like a nightmare |
| 01:38.06 | *** join/#asterisk phix (n=threat@123-243-44-131.tpgi.com.au) |
| 01:38.14 | *** join/#asterisk Frogzoo (n=Frogzoo@CPE-121-218-175-103.nsw.bigpond.net.au) |
| 01:39.57 | *** join/#asterisk ZX81 (n=ZX81@202.49.106.158) |
| 01:40.43 | ZX81 | hi all, does anyone know how to do an ISDN redirect (i.e. some hangup cause I expect) |
| 01:41.06 | jameswf-home | ISDN is so 1972 |
| 01:41.12 | JT | ... |
| 01:41.21 | jameswf-home | oh wait we are in america carry on |
| 01:41.27 | JT | jameswf-home: except that it's used by all most big businesses |
| 01:41.35 | JT | s/all most/most/ |
| 01:42.14 | *** join/#asterisk Wi_Fi (n=OUT@cpe-76-169-21-21.socal.res.rr.com) |
| 01:42.30 | jameswf-home | allot of big business still have dos based merlin dishwashers doesnt make it good |
| 01:42.55 | JT | i hope you're joking |
| 01:42.58 | JT | PRI == ISDN |
| 01:43.11 | JT | what's wrong with pri? |
| 01:43.34 | jameswf-home | well its better than nothing... |
| 01:43.51 | jameswf-home | pri isnt as bad as bri I guess |
| 01:44.06 | jameswf-home | all that work for 2 channels holy crap |
| 01:45.06 | drmessano | Coming in 2009: |
| 01:45.12 | drmessano | Larry the Cable Guy in |
| 01:45.15 | drmessano | Did Ya'll Call me? - The Story of Asterisk |
| 01:45.42 | *** join/#asterisk theron (n=theron@dsl.76.240.networkiowa.com) |
| 01:46.59 | theron | Hi all, I'm looking for a way to get two extensions to ring at the same time. any simple examples? |
| 01:47.08 | jameswf-home | larry has open sores? |
| 01:47.18 | drmessano | HA |
| 01:47.29 | jameswf-home | theron: google asterisk ring group |
| 01:47.36 | jameswf-home | ~ringgroup |
| 01:47.40 | theron | thanks jameswf-home |
| 01:47.53 | JT | jameswf-home: what work? bri is far better than POTS |
| 01:48.17 | JT | jameswf-home: so there's nothing wrong with PRI then i take it? ;) |
| 01:49.08 | *** join/#asterisk vn (n=nostalge@modemcable083.222-80-70.mc.videotron.ca) |
| 01:49.35 | jameswf-home | its hokey, expensive and a waste of bandwith. |
| 01:49.45 | *** join/#asterisk Kumbang (n=dsp@167.205.24.69) |
| 01:49.57 | vn | hi, why would my voip line suddenly half-working? I call a number, it dials but doesnt ring and it makes contact with the other number...but there's no voice |
| 01:49.58 | jameswf-home | All telecom except the last mile is voip now anyway |
| 01:50.22 | jameswf-home | vn firewall |
| 01:50.54 | vn | jameswf-home: I thought about that but even if I put the device in the DMZ, same thing |
| 01:53.16 | JT | jameswf-home: what's a waste of bandwidth? |
| 01:53.52 | JT | i think you'll find quite a lot of telecom is still TDM |
| 01:55.04 | [Latino] | anyone knows if the multi-homed problem it's resolved ? .. I mean an * machine with more than 1 IP on one interface |
| 01:55.56 | JT | also, VoIPoI isn't a good replacement for TDM |
| 01:56.04 | Frogzoo | asterisk takes a couple of rings to pass an incoming ring from the fxo line to a handset attached to an fxs port - any ideas? |
| 01:56.27 | JT | and if it's a dedicated dsl link for VoIP without Internet, why bother with VoIP at all? just go tdm |
| 01:57.02 | ManxPower | Frogzoo: it's waiting for callerid info |
| 01:57.52 | jameswf-home | JT what do you do exactly |
| 01:58.17 | jameswf-home | ever been in a cross box |
| 01:58.27 | *** join/#asterisk AndyGraybeal_ (n=andy@node144.39.251.72.1dial.com) |
| 01:58.29 | JT | i'm not familiar with that term |
| 01:58.58 | Frogzoo | ManxPower: so I need to disable caller id? |
| 01:59.29 | JT | jameswf-home: what is a cross box? |
| 02:00.41 | jameswf-home | I have worked on the business residential and telco side of telecommunications... I work now for a manufacturer, I have been in c/os and in the cross boxes and all over all telecom at somepoint is voip. just because its copper at your dmark doesnt mean it is 1500 feet away |
| 02:01.25 | jameswf-home | the last mile is all thats copper |
| 02:01.31 | JT | yes obviously |
| 02:01.36 | JT | it's converted to TDM at the exchange |
| 02:01.55 | JT | then carrier over pdh or sdh |
| 02:03.24 | JT | i'd be surprised if telecomms infrastructure is substantially different between our contries, apart from a few standard and acronyms |
| 02:03.33 | JT | the principles are similar |
| 02:04.41 | *** join/#asterisk riddlebox (n=james@75-132-225-75.dhcp.stls.mo.charter.com) |
| 02:04.43 | jameswf-home | T1 hasnt changed since 1961 yeah cant beat that |
| 02:05.08 | russellb | it's pretty darn solid. |
| 02:05.09 | JT | jameswf-home: are you saying that in us telecomms they have pretty much eliminated pdh and sdh/sonet in the carrier networks? |
| 02:05.16 | JT | jameswf-home: E1 can beat it ;) |
| 02:06.20 | jameswf-home | If the eu and us would catch up with japan we could have fttc and pure voip with no last mile would be a reality |
| 02:07.20 | JT | i don't see what the attraction to pure voip is |
| 02:07.31 | JT | especially if you're trying to do fax or modem signals |
| 02:07.56 | jameswf-home | jt we are talking about leaving the 70's in the 70's |
| 02:08.21 | jameswf-home | let go |
| 02:10.27 | drmessano | fax or modem signals? |
| 02:11.03 | JT | jameswf-home: it's nothing to do with what decade we're in |
| 02:11.09 | JT | it's just a matter of right tool for the job |
| 02:11.10 | drmessano | good god.. You mean to tell me you can't send facsimilies of documents or surf the web over IP? |
| 02:11.28 | JT | tdm is the most appropriate way to send a constant stream of realtime data like voice |
| 02:11.47 | JT | packet is the most appropriate way to send random data at random intervals |
| 02:12.15 | drmessano | Complaining that fax over VoIP doesn't work is saying that fax has a place in modern technology.. it doesn't |
| 02:12.43 | jameswf-home | let go |
| 02:12.49 | jameswf-home | this is the future |
| 02:12.52 | JT | fax isn't going to go away any time soon |
| 02:13.00 | drmessano | ..for that reason |
| 02:13.06 | drmessano | Dinosaurs holding onto it |
| 02:13.06 | jameswf-home | no need to make packets audio to make em packets again |
| 02:13.14 | drmessano | If the transport died, Fax would be replaced as it should be |
| 02:13.28 | jameswf-home | fax works over IP asterisk may not like it but it works |
| 02:13.40 | drmessano | Dialup internet died, people came off their quarters and got DSL and Cable |
| 02:13.45 | JT | bah, you can chant this is the future until the cows come home, it won'tchange the fact that tdm is still the best for certain applications |
| 02:13.54 | JT | totally different scenarios |
| 02:13.56 | drmessano | People will learn to scan and email when Fax becomes a problem |
| 02:14.08 | jameswf-home | you have yet to give an application thats valid |
| 02:14.18 | JT | voice. |
| 02:14.31 | JT | voip uses more bandwidth than tdm |
| 02:14.35 | JT | for the same codec |
| 02:14.45 | jameswf-home | voice including radio and telivision all digital, even pots runs over voip next |
| 02:15.06 | JT | gar |
| 02:15.12 | JT | yes, digital |
| 02:15.14 | JT | but TDM |
| 02:15.17 | JT | not VoIP |
| 02:15.21 | JT | i am not advocating POTS |
| 02:15.29 | jameswf-home | the US actualy says all television has to be digital by 2008 |
| 02:15.32 | JT | POTS is reliable but sucks balls |
| 02:15.34 | JT | sure |
| 02:15.53 | jameswf-home | *2009 |
| 02:16.09 | JT | what does that have to do with voip? :) |
| 02:16.55 | jameswf-home | voip is a generic term voip is used to describe voice in any packet stream waether ip or not |
| 02:17.33 | JT | ok, but you realise that most carrier networks, whilst the voice is digitised, is not travelling over voip? |
| 02:17.55 | JT | voip does actually specify ip, but yeah |
| 02:17.59 | jameswf-home | are you really hung up on a protocol??? |
| 02:18.08 | JT | dude |
| 02:18.17 | JT | there's significant technical and functional differences |
| 02:18.30 | JT | i just think i'm not explaining well |
| 02:18.44 | JT | maybe i'm not that easy to comprehend |
| 02:18.45 | JT | shrug |
| 02:18.57 | jameswf-home | if i drop a voice call across a novell network its still voip. |
| 02:19.12 | drmessano | VoIPX ;) |
| 02:19.14 | jameswf-home | regaurdless of routing |
| 02:19.19 | JT | but if it's tdm, it's digital but not voip |
| 02:19.38 | JT | circuit switched data is completely different to packet switched data |
| 02:19.41 | jameswf-home | digital = packet streams see where we are going |
| 02:19.46 | JT | no |
| 02:20.00 | JT | digital = quantitised into specific defined states |
| 02:20.09 | JT | digital may or may not be packet based |
| 02:20.28 | jameswf-home | Well we will agree to disagree you hang out in 1984 I will sit in today |
| 02:20.28 | JT | i think this is where the confusion is |
| 02:20.54 | JT | how am i hanging out in 1984? |
| 02:23.14 | JT | i love digital |
| 02:23.29 | JT | i just like to use the most appropriate tool for the job :) |
| 02:23.52 | *** join/#asterisk nvrpunk (n=root@81.90.21.227) |
| 02:24.42 | nvrpunk | if I am trying to dial back to the USA over an IAX trunk using a SIP phone, is there anything special I need to do to make SIP go over to IAX? |
| 02:25.57 | JT | do you have an asterisk server between the sip phone and the iax? |
| 02:26.01 | nvrpunk | I have the IAX dialplan setup |
| 02:26.06 | nvrpunk | yes |
| 02:26.35 | JT | asterisk should automatically transport voice between different channel drivers as needed |
| 02:27.02 | nvrpunk | ok, so it's not like I need to put the same dial plan in the SIP config then |
| 02:27.11 | nvrpunk | that the IAX one has to them |
| 02:27.15 | JT | dialplans go in extensions.conf |
| 02:28.25 | *** join/#asterisk obnauticus (n=obnautic@c-67-160-183-109.hsd1.or.comcast.net) |
| 02:28.41 | nvrpunk | ok |
| 02:28.50 | *** join/#asterisk AndyGraybeal (n=andy@node144.39.251.72.1dial.com) |
| 02:29.08 | nvrpunk | what I have right now is two test phones setup for internal SIP to SIP |
| 02:29.19 | nvrpunk | do those need to be reconfigured in the sip.conf |
| 02:29.29 | nvrpunk | so they have a real number and extension? |
| 02:29.31 | JT | no |
| 02:29.33 | JT | well |
| 02:29.34 | nvrpunk | ok |
| 02:29.36 | jameswf-home | I am filtering through vendor pages man some stuff is really outdated... spent 45 minutes doing wikis |
| 02:29.40 | JT | they should have extensions |
| 02:29.50 | JT | jameswf-home: pm |
| 02:29.53 | nvrpunk | they are 1001 and 1002 |
| 02:30.12 | vn | um...anyone happens to know how i can reset RTP ports on a SPA2102? |
| 02:36.03 | nvrpunk | Anyone sending calls must set a VALID ANI CallerID. You may not deliberately set it blank or to a false number |
| 02:36.24 | nvrpunk | JT, does that mean I have to set a full NPANXXXXXXX? |
| 02:36.36 | nvrpunk | err NPA NXXXXXX |
| 02:36.52 | drmessano | reset RTP ports on a SPA2102? |
| 02:37.04 | JT | nvrpunk: i don't know, probably |
| 02:38.36 | vn | drmessano: nevermind, my provider was the problem |
| 02:40.16 | jameswf-home | wow hillary changed horses mid-stream |
| 02:40.18 | teknoprep | i set my CID to 0000000911 |
| 02:41.16 | JT | jameswf-home: hi |
| 02:41.33 | jameswf-home | Every election year the people should watch wag the dog |
| 02:43.23 | *** join/#asterisk clandmeter (n=Carlo@81.175.82.2) |
| 02:43.44 | *** join/#asterisk DarkRift (i=dark@bas10-montreal02-1177584495.dsl.bell.ca) |
| 02:43.55 | *** join/#asterisk AndyGraybeal_ (n=andy@node144.39.251.72.1dial.com) |
| 02:44.49 | *** part/#asterisk vn (n=nostalge@modemcable083.222-80-70.mc.videotron.ca) |
| 02:48.33 | b11d | jameswf-home.. i was just thinking about that movie not five mins ago.. nice call :) |
| 02:49.09 | cy3o3 | http://www.rowtow.com/2008/02/10/everytime-you-pay-for-windows-2000-xp-or-vista-you-fund-the-church-of-scientology |
| 02:49.26 | *** join/#asterisk InsolentDreams (n=Insolent@p54B9DFBD.dip.t-dialin.net) |
| 02:50.47 | InsolentDreams | Hey all, anyone know a easy way to query the state of a time condition from the console? Eg, I have a GotoIfTime(090-170|mon-fri|1-31|jan-dec?app-announcement-2,s,1), and would like to write a script or have a command to tell me if that will goto right now or not. |
| 02:51.17 | InsolentDreams | Well, that copy/paste got a little muffled, the time is actually right on my side. ;\ |
| 02:52.03 | ZX81 | heh just figured out why my xchat keeps flashing at me for no reason - you said announce :) |
| 02:52.17 | ZX81 | why not just ring that extension |
| 02:52.21 | ZX81 | and see what it does |
| 02:53.42 | *** join/#asterisk [gnubie] (n=[gnubie]@cm160.gamma187.maxonline.com.sg) |
| 02:54.38 | *** join/#asterisk osiris (n=osiris@c-71-205-29-230.hsd1.mi.comcast.net) |
| 02:56.23 | *** join/#asterisk clandmeter (n=Carlo@81.175.82.2) |
| 02:57.40 | jameswf-home | So if the writers are on strike who is writing the agreement... and if others can write who needs writers |
| 02:58.11 | pkunkra | but these are smart and funny writers... for the most part. |
| 02:58.31 | jameswf-home | I watch CNN for funny |
| 02:58.32 | pkunkra | you can hire anyone.... but you may get what you pay for. |
| 02:59.06 | jameswf-home | if you want funny elect ron paul |
| 02:59.08 | pkunkra | interesting.... i watch comedy central for funny. |
| 02:59.16 | *** join/#asterisk _ShrikE-LT (n=_ShrikE-@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 02:59.54 | drmessano | ~ron paul |
| 03:00.17 | drmessano | Forget 1.6, someone needs to work on that bot |
| 03:00.32 | jameswf-home | uh oh someone forgot ron paul |
| 03:00.37 | pkunkra | ~asterisk |
| 03:00.38 | jbot | it has been said that asterisk is the best free PBX in the world, or #asterisk on irc.freenode.net, or http://www.asterisk.org |
| 03:00.44 | drmessano | ZOMG |
| 03:00.47 | drmessano | ~ron paul |
| 03:01.04 | drmessano | jbot: ron paul is Never Forget! |
| 03:01.04 | jbot | okay, drmessano |
| 03:01.09 | drmessano | ~ron paul |
| 03:01.10 | jbot | hmm... ron paul is Never Forget! |
| 03:01.28 | pkunkra | ~jbot |
| 03:01.28 | jbot | jbot is probably a hack!, or known to have only said one useful thing. a tool, or dating the fembots, or [TK]D-Fender's b*tch |
| 03:02.04 | Nugget | 20-Jan-2008 19:16 <jbot> ron paul is, like, my choice for president in 2008 |
| 03:02.04 | jameswf-home | ~no ron paul is <reply> Ron paul could kick chuck Noris; Arse |
| 03:02.05 | jbot | okay, jameswf-home |
| 03:02.26 | Nugget | 02-Feb-2008 18:01 <jbot> it has been said that rupaul is less ghey than Ron Paul |
| 03:02.27 | b11d | too bad Ron Paul is pretty much withdrawing.. too bad too.. he actually made sense. |
| 03:02.33 | *** join/#asterisk coldstea1 (n=coldstea@unaffiliated/coldsteal) |
| 03:02.34 | b11d | I dont like that he wanted to withdraw from the UN though.. |
| 03:02.40 | jameswf-home | ~dropdatabase; |
| 03:02.56 | jameswf-home | jbot dropdatabase; |
| 03:02.57 | jbot | So you think you could kill me, did you. You thought you could come in here, in my own home, and take me down without a fight. I would have given you a chance, really. But now, you have unleashed hell. May god have mercy on your soul. |
| 03:03.18 | drmessano | ~drmessano |
| 03:03.18 | jbot | hmm... drmessano is the leading cause of censorship in #asterisk |
| 03:03.20 | pkunkra | ~show tables; |
| 03:03.31 | pkunkra | hah. worth a try. |
| 03:03.34 | drmessano | damn right I am |
| 03:03.36 | b11d | ~b11d |
| 03:03.36 | jbot | b11d is a constant source of misinformation... |
| 03:03.36 | coldstea1 | I have a problem I come home toay and I can't call out and now one can call in but I have a dialtone |
| 03:03.39 | b11d | haha true |
| 03:03.40 | coldstea1 | and internet |
| 03:03.50 | Nugget | ~'; drop database jbot; |
| 03:04.00 | jameswf-home | jameswf |
| 03:04.09 | jameswf-home | ~jameswf |
| 03:04.09 | jbot | i guess jameswf is he has way to much time on his hands, or a GOD |
| 03:04.35 | coldstea1 | *I have a problem I come home today and I can't call out and now one can call in but I have a dial tone and my asterisk server can reach the Internet |
| 03:04.40 | coldstea1 | how could I fix it? |
| 03:04.48 | b11d | gotta go wax my car.. bbl all |
| 03:05.16 | Nugget | I did that yesterday |
| 03:06.08 | coldstea1 | anyone? |
| 03:06.18 | coldstea1 | I forget what I'm supposed to post |
| 03:06.30 | jameswf-home | ~ask |
| 03:06.30 | jbot | Questions in the channel should be specific, informative, complete, concise, and on-topic. Don't ask if you can ask a question first. Don't ask if a person is there, just ask what you intended to ask them. Better questions more frequently yield better answers. We are all here voluntarily or against our will. |
| 03:06.59 | pkunkra | who is here against their will? |
| 03:07.00 | drmessano | coldstea1: Maybe you need to defrag and scandisk |
| 03:07.06 | drmessano | I am |
| 03:07.30 | coldstea1 | I posted the question |
| 03:08.17 | coldstea1 | drmessano: i don't know how to defrag on linux and it was working fine a few hours ago |
| 03:09.16 | pkunkra | coldstea1, we'll need more info. |
| 03:09.18 | *** join/#asterisk tuxfoo (n=tuxfoo@pool-72-65-149-192.chrlwv.east.verizon.net) |
| 03:09.38 | coldstea1 | pkunkra: okay what would you like and ill get w/e log you need or w/e |
| 03:09.39 | pkunkra | the question you posted is equivalent of "It doesn't work. Why? |
| 03:10.09 | coldstea1 | pkunkra: okay what info would you like so far that's all I know |
| 03:10.25 | coldstea1 | and I'm loged in the asterisk cli right now |
| 03:10.39 | pkunkra | first tell me what problems you're having and some steps you took to try to solve it. |
| 03:11.02 | coldstea1 | okay I can't call and no one can call me |
| 03:11.12 | coldstea1 | I logged in and restarted asterisk |
| 03:11.18 | *** join/#asterisk __freedom__lover (n=eduardo@201-92-88-113.dsl.telesp.net.br) |
| 03:11.21 | coldstea1 | and I still have the same problem |
| 03:11.44 | JT | coldstea1: pb your config for a start |
| 03:11.57 | *** join/#asterisk angryuser (i=nononon@df01t2-212-194-108-123.d4.club-internet.fr) |
| 03:12.16 | pkunkra | try bumping up debug and verbose |
| 03:12.23 | ZX81 | oh well - everyone's left the office - time for shareazza |
| 03:12.27 | pkunkra | that should yield some answers. |
| 03:12.47 | coldstea1 | I can see it does try to call out but it ends up with circuit busy |
| 03:12.59 | ZX81 | via VoIP? |
| 03:13.02 | coldstea1 | pkunkra: okay give me a sec |
| 03:13.12 | ZX81 | try configuring a soft phone to talk directly to the provider |
| 03:13.13 | coldstea1 | ZX81: via the cli |
| 03:13.22 | ZX81 | who says circuit buys? |
| 03:13.25 | ZX81 | *busy |
| 03:13.25 | jameswf-home | coldstea1: pastebin the last 500 lines of your log |
| 03:13.37 | [gnubie] | in the auto-attendant, how will you instruct only those extension numbers in a particular context must be dialed during the WaitExten() on the callers side? |
| 03:13.38 | ZX81 | ~pastebin |
| 03:13.38 | jbot | [~pastebin] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste |
| 03:13.55 | ZX81 | [gnubie], happens automatically |
| 03:14.17 | ZX81 | [gnubie], goes to exten => i,1, otherwise |
| 03:14.24 | ZX81 | or hangs up if there isn't one |
| 03:14.30 | coldstea1 | jameswf-home: which log? |
| 03:14.34 | ZX81 | oh |
| 03:14.40 | ZX81 | unless you mean a different context |
| 03:14.43 | ZX81 | in which case |
| 03:14.52 | ZX81 | include => mycontext |
| 03:14.52 | coldstea1 | pkunkra: let me take out the passwords from my conf |
| 03:14.54 | jameswf-home | asterisk/full or asterisk/messages |
| 03:14.58 | hax | about how much ram does the asterisk process usually take up? |
| 03:15.19 | [gnubie] | ZX81: you mean, there's no possibility that the caller will key in 9. where ignorepat=9 and EXTEN:1? |
| 03:15.25 | ZX81 | hax, really depends on what you're doing - but a linux machine should take all ram for caching eventually |
| 03:15.30 | jameswf-home | hax asterisk is like a woman it will take all it wants |
| 03:15.46 | hax | so... like <100mb? |
| 03:15.51 | ZX81 | it can run on openwrt though which has like 8mb or some such |
| 03:16.02 | ZX81 | there is a define for low memory |
| 03:16.07 | ZX81 | in make menuconfig |
| 03:16.23 | ZX81 | [gnubie], you need to pastebin your conf |
| 03:16.43 | ZX81 | :( shareaza crashed |
| 03:16.45 | [gnubie] | ZX81: ok.. for a while.. |
| 03:17.00 | ZX81 | like if you have |
| 03:17.03 | ZX81 | [context] |
| 03:17.08 | ZX81 | exten => s,1,Answer |
| 03:17.10 | hax | interesting |
| 03:17.16 | ZX81 | exten => s,n,WaitExten() |
| 03:17.26 | ZX81 | exten => 3,1,NoOp(3 pressed) |
| 03:17.31 | ZX81 | and someone presses 4 |
| 03:17.39 | ZX81 | it will go to exten => i,1, |
| 03:17.45 | hax | also, anyone aware of a cheap hosted asterisk service that doesn't suck? |
| 03:17.51 | ZX81 | nah |
| 03:17.54 | ZX81 | mine :) |
| 03:17.58 | ZX81 | if you're in New Zealand |
| 03:17.59 | ZX81 | :) |
| 03:18.01 | *** join/#asterisk tclark (n=TC@S0106000f66c5d294.gv.shawcable.net) |
| 03:18.07 | hax | heh |
| 03:18.08 | coldstea1 | jameswf-home: http://rafb.net/p/bfQxoV52.html |
| 03:18.44 | ZX81 | :) |
| 03:18.46 | teknoprep | so i have been skrewing with polycom digitplan's .. this one seems to be perfect |
| 03:18.47 | teknoprep | <digitmap dialplan.digitmap="911|0T|011xxx.T|[2-9]xxxxxxxxx|1[2-9]xxxxxxxxx|1xxx|*xx" |
| 03:18.52 | ZX81 | coldstea1, um not much useful in there |
| 03:18.56 | ZX81 | see if you can find the error |
| 03:19.22 | ZX81 | (x.) is fine |
| 03:19.22 | ZX81 | :) |
| 03:19.24 | teknoprep | why ? |
| 03:19.29 | ZX81 | just annoying |
| 03:19.30 | teknoprep | its really nice to have that |
| 03:19.34 | ZX81 | I always forget something |
| 03:19.35 | teknoprep | it auto dials when it matches |
| 03:19.38 | ZX81 | PSTN feature codes etc |
| 03:19.42 | ZX81 | we have |
| 03:19.43 | ZX81 | 126 |
| 03:19.45 | ZX81 | 127 |
| 03:19.46 | ZX81 | 123 |
| 03:19.49 | ZX81 | 1956 |
| 03:19.51 | ZX81 | 1957 |
| 03:19.53 | ZX81 | 083210 |
| 03:19.54 | ZX81 | etc etc |
| 03:20.00 | ZX81 | and there's always one more |
| 03:20.01 | ZX81 | :) |
| 03:20.02 | InsolentDreams | The sanity man... |
| 03:20.04 | teknoprep | thats stupid |
| 03:20.07 | InsolentDreams | Use commas instead of returns, ugh |
| 03:20.09 | teknoprep | your pbx is setup stupid |
| 03:20.11 | ZX81 | indeed! |
| 03:20.14 | ZX81 | lol |
| 03:20.17 | ZX81 | that's the PSTN |
| 03:20.18 | coldstea1 | ZX81: the only error I see is [Feb 10 21:16:33] ERROR[8851] res_config_pgsql.c: Postgresql RealTime: Failed to connect database server asterisk on 127.0.0.1. Check debug for more info. |
| 03:20.18 | ZX81 | :) |
| 03:20.22 | ZX81 | not the PBX |
| 03:20.22 | ZX81 | :) |
| 03:20.38 | ZX81 | coldstea1, you said it said circuit busy |
| 03:20.43 | ZX81 | so |
| 03:20.49 | ZX81 | I guess warning |
| 03:20.50 | teknoprep | well you could always end it with x. |
| 03:20.54 | coldstea1 | yeah I ment in the file I posted |
| 03:20.54 | ZX81 | yeah |
| 03:20.58 | ZX81 | but then I get calls |
| 03:21.00 | teknoprep | that way if you dial say ... 911 it goes through right away |
| 03:21.06 | tclark | is there any one here or who knows anyone that has a iax pbx test end pt some where in costa rica near san jose central valley on a dsl/cable service |
| 03:21.09 | ZX81 | saying I normally don't have to press call |
| 03:21.16 | ZX81 | why do I have to press it for "x" feature |
| 03:21.28 | teknoprep | i am adding 10|11|12|13|14|15|16|17|18|19 |
| 03:21.31 | riddlebox | I wish I could speed up my grandstream phones, so it wouldnt take 11 seconds hear a call ring |
| 03:21.32 | ZX81 | or "why does it take 6 seconds to dial my x" |
| 03:21.33 | teknoprep | for my parking lots |
| 03:21.45 | ZX81 | tclark, not i |
| 03:21.49 | teknoprep | polycom phones are the BEST |
| 03:21.53 | ZX81 | riddlebox, 11 seconds! |
| 03:21.54 | ZX81 | :) |
| 03:22.07 | ZX81 | you could shave 3 or 4 by disabling callerid if you're not using it |
| 03:22.09 | *** join/#asterisk asr33 (n=asr33@dialin-209-183-21-133.tor.primus.ca) |
| 03:22.12 | teknoprep | i am never buying another cisco phone again |
| 03:22.13 | ZX81 | and are using an analogue trunk |
| 03:22.19 | ZX81 | 11 seconds is a long time though |
| 03:22.26 | ZX81 | teknoprep, really? |
| 03:22.30 | ZX81 | xml good? |
| 03:22.30 | teknoprep | yes really |
| 03:22.34 | teknoprep | i love it |
| 03:22.47 | ZX81 | do they have xml browsery things? |
| 03:22.57 | teknoprep | you mean web page |
| 03:23.04 | teknoprep | mini browser |
| 03:23.07 | ZX81 | yeah |
| 03:23.14 | teknoprep | i think the ip650 does |
| 03:23.14 | ZX81 | for stock tickers etc |
| 03:23.22 | ZX81 | prices good? |
| 03:23.27 | teknoprep | www.froogle.com |
| 03:23.32 | ZX81 | yeah |
| 03:23.34 | teknoprep | sound quality is the best i have ever heard |
| 03:23.47 | teknoprep | i bought an IP320 and i love it for my house |
| 03:24.00 | teknoprep | i have a 4port POE switch in the basement and it really is nice |
| 03:24.08 | ZX81 | yeah - although I kinda like my microphone - condensor - plugged into mixer with speakers in the other room |
| 03:24.12 | ZX81 | great quality |
| 03:24.14 | teknoprep | well 8port switch 4port poe |
| 03:24.17 | ZX81 | but can only talk to the other room |
| 03:24.18 | ZX81 | :) |
| 03:24.31 | teknoprep | you can buy a polycom pc speakerphone |
| 03:24.43 | ZX81 | yeah if it did wifi :) |
| 03:24.51 | ZX81 | I have a CTU |
| 03:24.55 | ZX81 | and use my cell |
| 03:25.04 | teknoprep | ZX81, http://www.google.com/products?q=polycom+communicator+c100&btnG=Search+Products |
| 03:25.16 | ZX81 | brb ciggy |
| 03:25.21 | ZX81 | shit |
| 03:25.27 | ZX81 | will read that when I get back |
| 03:25.27 | ZX81 | :) |
| 03:25.28 | coldstea1 | this is what I get in the cli when I make a call |
| 03:25.29 | coldstea1 | http://rafb.net/p/HFwXUp30.html |
| 03:25.47 | ZX81 | canreinvite=no |
| 03:26.02 | ZX81 | and try to make the call using the reg details in a softphone |
| 03:26.03 | ZX81 | brb |
| 03:26.10 | teknoprep | coldstea1, are you behind a NAT ? |
| 03:26.15 | *** join/#asterisk mmurdock (n=TGA@c-24-10-190-87.hsd1.ut.comcast.net) |
| 03:26.35 | coldstea1 | teknoprep: yes |
| 03:26.42 | teknoprep | coldstea1, did you setup your nat settings ? |
| 03:26.49 | teknoprep | !nat |
| 03:26.51 | teknoprep | ?nat |
| 03:26.52 | coldstea1 | yeah |
| 03:26.59 | jameswf-home | maybe outside ip changed |
| 03:27.06 | teknoprep | externip=outside.ip |
| 03:27.13 | teknoprep | localnet=10.0.0.0/255.0.0.0 |
| 03:27.13 | coldstea1 | where do I put this canreinvite |
| 03:27.15 | teknoprep | nat=yes |
| 03:27.59 | asr33 | I've read the ~book it was very enjoyable, I now think should learn greater detail about SIP, there are many books on the subject, can somebody recommend an easy to understand book about SIP? |
| 03:28.01 | teknoprep | canreinvite=yes ; put this in your settings for your SIP connection to your ITSP |
| 03:28.20 | asr33 | ~book |
| 03:28.21 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com |
| 03:28.26 | teknoprep | ~nat |
| 03:28.27 | jbot | nat is probably Network Address Translation Usable in Asterisk sip.conf file with externip, localnet, and localmask setup properly. See docs. |
| 03:28.58 | jameswf-home | http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
| 03:29.07 | jameswf-home | ~buybook |
| 03:29.07 | jbot | You can buy "Asterisk the Future of Telephony" at http://www.oreilly.com/catalog/9780596510480/ so go buy it SERIOUSLY |
| 03:29.23 | [gnubie] | ZX81: still there? my dialplan is here already => http://www.privatepaste.com/b61vgXWokk |
| 03:31.11 | coldstea1 | I already have all thoese settings in my sip.conf |
| 03:31.24 | ZX81 | back |
| 03:31.24 | ZX81 | sec |
| 03:31.42 | ZX81 | so what context am I looking at? |
| 03:31.42 | [gnubie] | ZX81: i want that when an incoming call asked to key in the extension number to call, i just want that my pbx will only accept extension numbers of my [family] context and other than that, invalid already |
| 03:32.02 | ZX81 | so |
| 03:32.06 | ZX81 | [incoming] |
| 03:32.11 | ZX81 | exten => s,1,Answer() |
| 03:32.24 | ZX81 | exten => s,n,Background(dialanumber) |
| 03:32.34 | ZX81 | exten => s,n,WaitExten() |
| 03:32.42 | ZX81 | include => extensions |
| 03:32.52 | ZX81 | where extensions is a context that has your extensions in it |
| 03:32.57 | coldstea1 | I can't ping sip.axvoice.com |
| 03:33.02 | [gnubie] | ZX81: please take a look at my [menu] context |
| 03:33.07 | ZX81 | k |
| 03:33.33 | ZX81 | you will need to include the contexts that have your extensions |
| 03:33.48 | ZX81 | i.e. family, relatives, friends |
| 03:34.02 | ZX81 | then it should work fine |
| 03:34.28 | ZX81 | so add in include => family |
| 03:34.31 | ZX81 | include => friends |
| 03:34.31 | [gnubie] | basically, my [menu] and [inbound_trunks].. and the [family] context must be the only recognize extension numbers when keyed-in during the [menu] 's WaitExten() |
| 03:34.45 | ZX81 | yeah so include whatever you want to be recognized |
| 03:35.01 | [gnubie] | ZX81: where shall i insert it? |
| 03:35.08 | ZX81 | just after [menu] |
| 03:35.32 | ZX81 | gotta run - be good |
| 03:35.33 | ZX81 | :) |
| 03:35.41 | *** join/#asterisk adjohn (n=adjohn@p5182-ipad71marunouchi.tokyo.ocn.ne.jp) |
| 03:35.45 | [gnubie] | oh.. |
| 03:36.08 | coldstea1 | oh wait I can ping it but the time is real bad |
| 03:36.09 | coldstea1 | $ ping sip.axvoice.com -c1 |
| 03:36.09 | coldstea1 | PING sip.axvoice.com (216.143.130.36) 56(84) bytes of data. |
| 03:36.09 | coldstea1 | 64 bytes from 216.143.130.36: icmp_seq=1 ttl=44 time=57.6 ms |
| 03:36.09 | coldstea1 | --- sip.axvoice.com ping statistics --- |
| 03:36.09 | coldstea1 | 1 packets transmitted, 1 received, 0% packet loss, time 0ms |
| 03:36.11 | coldstea1 | rtt min/avg/max/mdev = 57.691/57.691/57.691/0.000 ms |
| 03:36.13 | coldstea1 | oO |
| 03:36.24 | coldstea1 | I didn't think that was going to be that big sorry |
| 03:37.18 | JT | 57ms is real bad? |
| 03:37.41 | coldstea1 | it takes a while to respond |
| 03:37.59 | JT | 57ms is less than 1/10th of a second |
| 03:38.04 | teknoprep | nah |
| 03:38.06 | teknoprep | 57ms is fine |
| 03:38.16 | teknoprep | thats .057 seconds |
| 03:38.31 | [gnubie] | is there such thing as "exclude => blah" in the asterisk's dialplan? |
| 03:38.49 | *** part/#asterisk theron (n=theron@dsl.76.240.networkiowa.com) |
| 03:39.24 | lmadsen | [gnubie]: nada |
| 03:39.48 | [gnubie] | i see.. thanks.. |
| 03:39.52 | lmadsen | you shouldn't need it |
| 03:40.17 | coldstea1 | okay I got it working |
| 03:40.38 | [gnubie] | lmadsen: were you able to read our discussions with ZX81 earlier regarding my problem with the dialplan? |
| 03:40.46 | coldstea1 | I unplugged my ATA and plugged it back in |
| 03:40.48 | lmadsen | sorry, I didn't scroll back |
| 03:40.59 | lmadsen | I'm actually just checkin' the email quick then off to read |
| 03:41.56 | [gnubie] | lmadsen: this is my extensions.conf => http://www.privatepaste.com/b61vgXWokk |
| 03:43.49 | [gnubie] | lmadsen: my problem is whenever a caller from an inbound_trunks asked to key in an extension number to call during my [menu]'s WaitExten() , it should only accept those extension numbers under the [family] context and other than that must be invalid |
| 03:44.16 | jameswf-home | ~rob |
| 03:44.17 | jbot | somebody said rob was (Radically Omnipotent Boy) A tall, bearded, long-haired person who is scarily intelligent and often dangerous (they have been known to be photographed with axes and chainsaws). A ROB normally likes to drive really fast cars and eat lots of pizza. ROBs also frequently contribute nearly 100% of the content of on-line Internet webzines. |
| 03:45.44 | jameswf-home | ~fsck |
| 03:45.45 | jbot | No devices specified to be checked! |
| 03:45.55 | jameswf-home | ~fsck hda1 |
| 03:45.55 | jbot | e2fsck /dev/hda1 : warning! filesystem contains helpdesk workers! |
| 03:45.59 | *** part/#asterisk coldstea1 (n=coldstea@unaffiliated/coldsteal) |
| 03:46.07 | jameswf-home | ~fsck sda1 |
| 03:46.08 | jbot | e2fsck /dev/sda1 : warning! filesystem contains dumbasses! |
| 03:46.15 | jameswf-home | ~fsck sda2 |
| 03:46.15 | jbot | e2fsck /dev/sda2 : warning! filesystem contains helpdesk workers! |
| 03:46.48 | [gnubie] | lmadsen: ZX81's suggestion was to add "include => family" below the [menu] line.. does it mean that it will automatically exclude the other contexts and whenever a number that is not a member of the [family] context are invalid? |
| 03:47.49 | lmadsen | [gnubie]: you probably don't want to use an include there then because there is no way to block the transitive properties. What I'd do is use a separate pattern match in the [menu] to explicitly match what you want, and control access to the other context that way |
| 03:48.43 | [gnubie] | lmadsen: sample please from my existing dialplan if you don't mind? |
| 03:48.58 | [gnubie] | lmadsen: http://www.privatepaste.com/b61vgXWokk/download |
| 03:49.18 | lmadsen | [gnubie]: sorry, that was pretty straight forward and I only give out so much free consulting a day ;) |
| 03:49.52 | [gnubie] | ok. thanks anyway. |
| 03:50.28 | lmadsen | bascially I'm saying remove the include entirely and create a separate pattern match with a Goto() |
| 03:50.41 | lmadsen | exten => _4XXX,1,Goto(${EXTEN},some_context) |
| 03:50.45 | lmadsen | in your menu context |
| 03:50.48 | lmadsen | and with that... I'm out! |
| 03:52.07 | [gnubie] | lmadsen: ok.. thanks again. |
| 03:55.51 | *** part/#asterisk lmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 04:14.28 | *** join/#asterisk angryuser[A] (i=nononon@df01t2-212-194-108-123.d4.club-internet.fr) |
| 04:28.08 | *** join/#asterisk Igbothom_III (n=Hilton@office.quarkgroup.com.au) |
| 04:32.48 | *** join/#asterisk L|NUX (i=L_NUX@unaffiliated/lnux/x-10290) |
| 04:52.12 | *** join/#asterisk adjohn (n=adjohn@p5182-ipad71marunouchi.tokyo.ocn.ne.jp) |
| 04:52.21 | *** join/#asterisk Daejeo (n=chatzill@211.211.234.7) |
| 04:55.08 | *** join/#asterisk craigk (n=craigk@58.174.150.119) |
| 05:00.21 | *** join/#asterisk apollonx (n=admin@89.218.144.26) |
| 05:03.42 | *** join/#asterisk metfan2007 (n=metfan20@189.180.217.155) |
| 05:04.00 | metfan2007 | Hi, how do I find the version number in a zaptel instalation? |
| 05:04.35 | jameswf-home | modinfo |
| 05:04.52 | metfan2007 | great, thanks |
| 05:06.30 | metfan2007 | and for openh323? xD |
| 05:19.32 | *** join/#asterisk MaliutaWrk (n=nikolai@kiev.lusan.id.au) |
| 05:25.02 | *** join/#asterisk CrazyTux[m] (n=CrazyTux@ppp-70-248-182-215.dsl.hstntx.swbell.net) |
| 05:25.30 | *** join/#asterisk Maliuta (n=nikolai@kiev.lusan.id.au) |
| 05:26.19 | *** join/#asterisk dacs (n=haiger@unaffiliated/dacs) |
| 05:27.49 | *** join/#asterisk emist (n=emist@unaffiliated/emist) |
| 05:39.48 | *** join/#asterisk CunningPike (n=CunningP@S010600095b33697f.vc.shawcable.net) |
| 05:44.10 | [gnubie] | is there a default variable in asterisk-1.4 that is similar to "dialed ID" or "called ID" ? |
| 05:44.31 | *** join/#asterisk emist (n=emist@unaffiliated/emist) |
| 05:45.01 | [gnubie] | i know there is a CALLERID variable but that one is the caller.. i'm looking for the callee's id if there exist |
| 05:51.12 | pkunkra | $EXTEN ? |
| 05:55.15 | findlay | [gnubie]: http://www.voip-info.org/wiki/view/CallerID |
| 05:55.33 | findlay | oh, wrong answer |
| 05:57.30 | findlay | http://www.voip-info.org/wiki-Asterisk+variables |
| 05:58.35 | findlay | should be ${EXTEN} |
| 05:58.36 | *** join/#asterisk mmurdock (n=TGA@c-24-10-190-87.hsd1.co.comcast.net) |
| 06:01.28 | [gnubie] | findlay: ok.. thanks.. |
| 06:03.50 | emist | yo guys, in case it hasn't reached you yet, if you're running kernel .17-24 it would be a good time to patch |
| 06:03.51 | emist | http://www.reactivated.net/weblog/archives/2008/02/critical-linux-kernel-vmsplice-security-issues/ |
| 06:06.35 | jameswf-home | omfg get over it |
| 06:06.55 | *** join/#asterisk shtoom (n=godson@59.93.114.10) |
| 06:07.06 | emist | ? |
| 06:07.22 | drmessano | Good god |
| 06:07.25 | drmessano | Nothing to see here |
| 06:07.29 | drmessano | Go back to your families |
| 06:07.43 | jameswf-home | I logged in as root to fix it |
| 06:07.54 | jameswf-home | now everything i do is as root |
| 06:08.03 | drmessano | I closed Internet Explorer, and it was gone |
| 06:08.23 | jameswf-home | rm -rf *.bug |
| 06:08.31 | drmessano | Christ |
| 06:08.39 | jameswf-home | rm -rf *.securitah_holez |
| 06:08.41 | drmessano | This isn't even a news story, it's BLOG SPAM |
| 06:08.51 | drmessano | Put BLOG SPAM on Digg where it belongs |
| 06:09.09 | emist | well, seeing as the latest stable kernel is vulnerable i would call it a news story |
| 06:09.10 | drmessano | Link me something with LINUS or TORVALDS on it, pls |
| 06:09.13 | jameswf-home | no no its true I read it on slashdot |
| 06:09.31 | drmessano | Not the link you posted |
| 06:09.31 | emist | er...i tested it on 19-24 personally |
| 06:09.37 | emist | im lost |
| 06:09.43 | drmessano | yes, you are |
| 06:09.44 | Docfxit | Where can I find the backtrace.txt file in Asterisknow? I have looked in /etc/asterisk/doc. There is no directory called doc within the asterisk directory. |
| 06:09.58 | drmessano | GIVE ME A LINK TO A REAL STORY, NOT SOME ASSHATS BLOG |
| 06:10.02 | drmessano | Blog spam blows |
| 06:10.05 | emist | a real story? |
| 06:10.09 | drmessano | If you got a good link, link it |
| 06:10.16 | Docfxit | I found backtrace.txt on the internet but not loaded on the system. |
| 06:10.19 | emist | http://www.milw0rm.com/exploits/5092 |
| 06:10.41 | jameswf-home | big perm i mean worm |
| 06:10.42 | emist | maybe thats a little more real |
| 06:11.02 | drmessano | That's better.. I'm trying to make you one less person that posts a blog post when a better authority exists for all to see |
| 06:11.07 | jameswf-home | OH NOZ HAXORZ MIL WURMZ |
| 06:11.30 | emist | ----------------------------------- |
| 06:11.30 | emist | <PROTECTED> |
| 06:11.30 | emist | <PROTECTED> |
| 06:11.30 | emist | ----------------------------------- |
| 06:11.30 | emist | [+] mmap: 0x0 .. 0x1000 |
| 06:11.30 | emist | [+] page: 0x0 |
| 06:11.32 | emist | [+] page: 0x20 |
| 06:11.34 | emist | [+] mmap: 0x4000 .. 0x5000 |
| 06:11.36 | emist | [+] page: 0x4000 |
| 06:11.38 | emist | [+] page: 0x4020 |
| 06:11.38 | drmessano | Stop |
| 06:11.40 | emist | [+] mmap: 0x1000 .. 0x2000 |
| 06:11.41 | drmessano | STOP |
| 06:11.42 | emist | [+] page: 0x1000 |
| 06:11.44 | emist | [+] mmap: 0xb7f8c000 .. 0xb7fbe000 |
| 06:11.45 | drmessano | STOP PASTING |
| 06:11.46 | emist | [+] root |
| 06:11.48 | emist | $ whoami |
| 06:11.50 | emist | root |
| 06:11.52 | emist | calm down son |
| 06:12.11 | drmessano | Stop pasting, SON |
| 06:12.33 | emist | i just did son |
| 06:12.47 | jameswf-home | lmao |
| 06:13.08 | jameswf-home | damn dad |
| 06:13.22 | emist | either way, if you have any idea what the kernel is and how to patch it you might want to go for it, if not forget what i just said |
| 06:13.30 | emist | good day |
| 06:13.38 | jameswf-home | lmao trollz |
| 06:13.54 | jameswf-home | ~troll |
| 06:13.54 | jbot | somebody said troll was a race on some muds; a guy under a bridge: an annoying robot; a port scanner, or somebody faking being clueless to be shown the One Linux Way so he can argue against it for his psychology thesis on linux advocates, or an employee of trolltech, or at http://www.kuro5hin.org/story/2001/7/27/51233/2979 or ... |
| 06:14.24 | drmessano | No shit |
| 06:15.10 | jameswf-home | the script secretly rewrites the env to display root when an idiot types whoami woooooooooo |
| 06:15.16 | jameswf-home | bwahhahhahhaaa |
| 06:16.08 | jameswf-home | crap he is probably haxoring me right now |
| 06:17.20 | drmessano | jbot: blogspam is when someone posts a link to some unofficial or self-gratifying blog post on a story where a more official source of the information exists, and is often linked to somewhere in the middle of said blog post, making the link to the blog post nothing more than "Spam" |
| 06:17.21 | jbot | drmessano: okay |
| 06:17.26 | drmessano | ~blogspam |
| 06:17.27 | jbot | [blogspam] when someone posts a link to some unofficial or self-gratifying blog post on a story where a more official source of the information exists, and is often linked to somewhere in the middle of said blog post, making the link to the blog post nothing more than "Spam" |
| 06:19.15 | drmessano | jameswf-home, when they hell are they going to release Web 2.1? There's quite a few bugfixes and new features I am waiting on. |
| 06:19.37 | drmessano | s/they/the/ |
| 06:19.42 | jameswf-home | I dunno I am allready on web 4.0 alpha |
| 06:20.02 | drmessano | lol |
| 06:20.03 | *** join/#asterisk obnauticus (n=obnautic@c-67-160-183-109.hsd1.or.comcast.net) |
| 06:20.55 | drmessano | Web 4.0 will be the the full circle convergence of VoIP + Web where people stop blogging about shit and actually call each other like they did in the 70s and 80s |
| 06:24.36 | drmessano | People will grow tired of phones with a bunch of buttons and long for the days of rotary, making pulse dialing stability the key feature in Asterisk 1.8 |
| 06:26.24 | drmessano | "If you are using a touch tone phone, please hang up, get back into your time capsule, and set a course for 1999.. Rotary is back!" |
| 06:26.41 | jwh | lol |
| 06:28.04 | drmessano | I also want my 12 button cable boxes back too |
| 06:28.30 | drmessano | Push down 2 + 5 + 9 and get playboy |
| 06:28.36 | drmessano | YAY |
| 06:37.48 | [gnubie] | hello all.. |
| 06:38.17 | [gnubie] | kindly check out => http://www.privatepaste.com/f2o3nr7WfW |
| 06:38.52 | [gnubie] | what do you think of the line 6 of my [auto-attendant] context? is it on a proper syntax or not? |
| 06:41.03 | *** join/#asterisk mmmToop (n=michaelt@firewall.datapro.co.za) |
| 06:48.12 | *** part/#asterisk [gnubie] (n=[gnubie]@cm160.gamma187.maxonline.com.sg) |
| 06:49.22 | the_5th_wheel | Is there any reason why someone would be moved later in a queue when they sit in th que? it seems the last people are being helped first |
| 06:50.16 | *** join/#asterisk [T]ank (n=ckwall@c-71-195-199-107.hsd1.ut.comcast.net) |
| 06:50.42 | jwh | the_5th_wheel: what queue? |
| 06:51.18 | *** join/#asterisk adjohn (n=adjohn@p5182-ipad71marunouchi.tokyo.ocn.ne.jp) |
| 06:59.14 | [T]ank | i am looking for a comparison of codecs on sip and how much bandwidth they take. I am trying to decide which codec is the highest quality with the best use of bandwidth. can anyone refer me to a link with this kind of info? |
| 07:05.39 | *** join/#asterisk oej (n=olle@soll4-125.cust.blixtvik.net) |
| 07:29.16 | *** join/#asterisk DaveCanoe (n=Dave@canoe404.dclg.ca) |
| 07:30.00 | daven | <PROTECTED> |
| 07:34.17 | *** join/#asterisk yangvnc (n=yang@cpe-212-18-59-220.dynamic.amis.net) |
| 07:35.40 | yangvnc | Has anyone experience with Grandstream phones + BLF lights + Asterisk, I can use the keys, but they don't blink red on busy signal... |
| 07:38.14 | *** join/#asterisk harpal (n=harpal@124.125.76.195) |
| 07:38.58 | harpal | Hey any one has used earthcaller site? I cant make call with that site, so any one tell me what port I have to open to make it work |
| 07:50.19 | the_5th_wheel | jwh: i have tech support que. its a ringall. it keeps on telling people you are nr3 in the queu, then it says you are nr 5 in the que |
| 07:50.45 | jwh | mm, no console messages? |
| 07:50.53 | jwh | like, indicating change? |
| 07:51.47 | the_5th_wheel | lemme try it |
| 07:53.12 | the_5th_wheel | uhm, its rather difficult to test it without there actually being clients on the que |
| 07:53.16 | *** join/#asterisk Grash (n=grash@cdbil1-a1-2-23.ipcom.comunitel.net) |
| 07:53.43 | jwh | hehe |
| 07:54.58 | the_5th_wheel | http://pastebin.div0.co.za/results/A6GEFG3E6.html <-- this is my que config |
| 07:59.58 | *** join/#asterisk tzafrir (n=tzafrir@62.90.10.53) |
| 08:00.50 | jwh | looks fine |
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| 08:01.00 | *** part/#asterisk Jmlk17 (n=Jmlk17Wi@wikipedia/Jmlk17) |
| 08:02.02 | *** part/#asterisk harpal (n=harpal@124.125.76.195) |
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| 08:24.19 | the_5th_wheel | Can i use an SPA3102 as a LCR? |
| 08:27.39 | *** join/#asterisk qdk (n=qdk@85.235.253.139) |
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| 08:33.05 | krdian_ | yangvnc: have u configured hints ? |
| 08:33.56 | *** join/#asterisk the-moog (n=the-moog@static-195-248-121-199.adsl.hotchilli.net) |
| 08:34.59 | the-moog | Hi, I need some help getting asterisk-addons to compile. |
| 08:35.59 | krdian_ | yangvnc: http://www.grandstream.com/documents/GXP2000BLFwithAsteriskConfiguration.pdf |
| 08:40.30 | *** join/#asterisk sergey (n=sergey@91.189.233.66) |
| 08:40.51 | tzafrir | the-moog, what error do you get? on what platform? what verisons? |
| 08:43.54 | yangvnc | krdian_: yes, I have configured hints |
| 08:44.09 | yangvnc | good Morning tzafrir |
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| 08:44.47 | yangvnc | krdian_: thanks, the link is helpfull |
| 08:45.01 | the-moog | tzafrir: ubuntu 1.7 (kernel 2.6.22-14) on i686 with asterisk 1.4.17. asterisk addons is from the trunk |
| 08:45.41 | the-moog | tzafrir: Applying the 1.4 patch for chan_mobile fails. |
| 08:46.15 | tzafrir | I don't think trunk addons and asterisk 1.4 can build |
| 08:48.20 | the-moog | So why publish a patch, see http://www.chan-mobile.org/?page_id=4 |
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| 08:56.46 | tzafrir | the-moog, http://www.chan-mobile.org/?page_id=5 |
| 08:57.43 | the-moog | Yep, that's the bit that goes wrong, see http://rafb.net/p/i01Sq954.html |
| 08:58.17 | the-moog | It appears that the code in trunk has been updated and the code in the patch not. |
| 08:58.43 | the-moog | I wonder if I can find what version the patch came from |
| 09:06.13 | the-moog | tzafrir |
| 09:06.32 | the-moog | tzafrir: checking out version 454 allows patch to work :) |
| 09:06.59 | yangvnc | krdian_: Are you using the grandstream phone? |
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| 09:11.26 | the_5th_wheel | on my SPA3102, correct me im wrong, but this would make that all calls excluding 08xxxxxxxx and 07xxxxxxxx go to the pstn line? |
| 09:11.29 | the_5th_wheel | (xxxx|08xxxxxxxx|07xxxxxxxx|xxxxxxxxxx:@gw0) |
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| 09:13.58 | *** join/#asterisk Cyorxamp (n=Cyorxamp@212.57.232.254) |
| 09:13.58 | *** mode/#asterisk [-o+b Cyorxamp *!*Cyorxamp@212.57.232.*] by twisted |
| 09:13.58 | *** kick/#asterisk [Cyorxamp!n=root@pdpc/supporter/active/twisted] by twisted (Join flood (4 joins in 28secs of 50secs)) |
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| 09:16.28 | krdian_ | yangvnc: yes, |
| 09:16.34 | krdian_ | yangvnc: unfortunatly |
| 09:17.40 | krdian_ | yangvnc: *unfortunately |
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| 09:21.43 | yangvnc | hehe |
| 09:22.05 | yangvnc | krdian_: ok i did everything correct, but this happens when i try to takeover the call by BLF key http://openpaste.org/en/5074/ |
| 09:23.03 | yangvnc | ast_get_group: Ignoring invalid group 65 (maximum group is 63) |
| 09:23.13 | yangvnc | I wonder what this error relies to |
| 09:23.58 | *** mode/#asterisk [-b *!*Cyorxamp@212.57.232.*] by twisted |
| 09:28.59 | yangvnc | also the call gets totally weird with echoes if i apply that rule |
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| 09:30.39 | *** join/#asterisk Sinar (n=bwoSinar@81.144.146.38) |
| 09:31.30 | Sinar | morning. Is there a way of getting Playback() to access a remote file? Wanting to access a file from a webserver, not on the local machine. I'm using FastAGI so don't have access to the stream. Wanting to play back a custom file, not use this for music on hold streaming |
| 09:32.03 | tzafrir | What type of remote file? |
| 09:32.10 | tzafrir | Generally, no |
| 09:32.17 | tzafrir | But maybe there's a way |
| 09:32.25 | Sinar | ulaw sound file. pre-rendered text-to-speech content. |
| 09:32.42 | Sinar | so that's why I need to use Fast AGI to pass variables around so the dialplan plays back the right file |
| 09:32.53 | tzafrir | Yeah, but how do you get to that file? |
| 09:33.03 | *** join/#asterisk XnOSX (i=d491ac7f@gateway/web/ajax/mibbit.com/x-6958006e161bd850) |
| 09:33.06 | Sinar | I was hoping for http:? :) |
| 09:33.10 | krdian_ | yangvnc: try to use dial instead of pickup |
| 09:33.16 | Sinar | i guess it could be a samba share |
| 09:33.21 | mvanbaak | Sinar: use wget to grab the file |
| 09:34.12 | Sinar | ok I'll look into that. but not sure how that'll work with Fast AGI running on a different box |
| 09:34.15 | Sinar | brb |
| 09:34.22 | yangvnc | krdian_: is there a difference between exten => _**5X,2,Hangup or Hangup() ? |
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| 09:35.19 | tzafrir | If it is a samba or NFS share, you can mount it, and it becomes a local file |
| 09:35.23 | krdian_ | yangvnc: not really |
| 09:35.49 | tzafrir | As for http: you can try grabbing that file beforehand with curl, I guess |
| 09:36.46 | yangvnc | krdian_: do i need to enter exten => _**6X,1,Dial(${EXTEN:2}) or exten => _**6X,1,Dial(SIP/${EXTEN:2}) |
| 09:36.58 | tzafrir | or maybe use something like http://httpfs.sourceforge.net/ |
| 09:37.15 | tzafrir | Sinar, ==^ |
| 09:37.25 | krdian_ | yangvnc: exten => _**6X,1,Dial(SIP/${EXTEN:2}) |
| 09:38.42 | yangvnc | thanks |
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| 09:40.30 | worgil | i am using DVG-1402S and have two line ont this, but when it login to asterisk server only one line be up, other line is waiting to query. how can i do it for ? |
| 09:41.31 | *** join/#asterisk keulin (n=cray@ifth-pdcr2.infotheme.fr) |
| 09:42.45 | yangvnc | krdian_: if i apply DIAL command instead of pickup then the users gets the second call |
| 09:43.05 | yangvnc | if i press the BLF button |
| 09:43.35 | yangvnc | usually the call is being taken with *8 |
| 09:43.38 | yangvnc | which works fine |
| 09:44.32 | worgil | i am using DVG-1402S and it have two line on this, but when it login to asterisk server only one line be up, other line is waiting to query. how can i do it for ? |
| 09:44.39 | krdian_ | yangvnc: oh, sorry right, i didn't check ur dialplan |
| 09:45.22 | yangvnc | I might test this later in the afternoon, everyone is making the calls now |
| 09:47.59 | krdian_ | yangvnc: generaly, if u use just hints in proper context is enough to BLF |
| 09:49.17 | yangvnc | i use exten => 60,hint,SIP/60 ; Jozi |
| 09:49.35 | yangvnc | and in sip.conf i have subscribecontext=BLF |
| 09:49.40 | yangvnc | hints are in BLF |
| 09:50.19 | yangvnc | without the manual you told me about, the lights never went red |
| 09:51.23 | krdian_ | yangvnc: not suer if u can use subscribecontext..., i have context with hints included in context |
| 09:52.54 | krdian_ | yangvnc: try to include 'BLF' in context which u connfigured in sip.conf |
| 09:54.02 | worgil | i am using DVG-1402S and it have two line on this, but when it login to asterisk server only one line be up, other line is waiting to query. how can i do it for ? |
| 09:54.03 | yangvnc | ok |
| 09:56.43 | Sinar | thanks for that tzafrir |
| 09:57.59 | krdian_ | worgil: i don't know ur phone but check call-limit in sip.conf and ringinuse option in queue.conf |
| 10:01.26 | krdian_ | does anybody known any voice stress analysis software similar to liarliar project but working as comand-line ? |
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| 10:16.29 | worgil | krdian_, now can be online two lines but not comieng sound have you any idea ? |
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| 10:19.26 | krdian_ | worgil: no sound ? on both lines ? |
| 10:19.59 | worgil | only one linehave sound |
| 10:21.48 | krdian_ | worgil: no idea, maybe your phone need to has second account to do as you want |
| 10:22.57 | krdian_ | worgil: or uhave to switch accounts somehow, on my grandstream i have to push proper line |
| 10:23.58 | krdian_ | worgil: is ur phone d-link, is it ? |
| 10:24.45 | worgil | yes |
| 10:24.47 | krdian_ | worgil: i guess it worse than grandstream :) sorry i have bad exerience with d-link switches |
| 10:25.38 | krdian_ | worgil: oh, its gatewaay not phone ... |
| 10:25.47 | worgil | krdian_, yes d-link and i can not keep up two line, when i can keep up than only one line use sound |
| 10:26.07 | krdian_ | worgil: i think u have to configure second account on it |
| 10:26.37 | worgil | what must i do krdian_? |
| 10:28.46 | krdian_ | worgil: u have to configure second sip account and agent on it IMO |
| 10:29.59 | krdian_ | worgil: but as i said i dontt know ur equipment :( |
| 10:30.35 | worgil | i did krdian_ |
| 10:30.57 | worgil | i have two acount 1011 and 1012 on d-link |
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| 10:32.05 | krdian_ | worgil: can u call both extension directly ? |
| 10:32.33 | worgil | yes |
| 10:32.38 | worgil | i can |
| 10:34.00 | krdian_ | worgil: so, u have problem when you calling these extension through queue, right ? |
| 10:34.10 | worgil | yes |
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| 10:36.43 | krdian_ | worgil: do u see sth strange in logs ? |
| 10:36.55 | worgil | no krdian_ |
| 10:40.28 | krdian_ | worgil: as i understand u have two analog phones connected to this box ? |
| 10:41.27 | krdian_ | worgil: both account have similar configs |
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| 10:44.10 | worgil | when i call other extension, they are not hear my sound |
| 10:46.13 | phix | hmmm |
| 10:56.37 | *** join/#asterisk Dovid (n=Dovid@bzq-79-178-28-220.red.bezeqint.net) |
| 10:56.48 | Dovid | hi. I am trying to use agentcallbacklogin |
| 10:57.20 | Dovid | i am having an issue where when i try to log in it asks me for a new extension. I dont see that any where in the doc's. can anyone advise ? |
| 11:02.01 | angryuser | ~freepbx |
| 11:02.01 | jbot | [~freepbx] FreePBX is unable to be supported here. It is made up of complex dialplans and scripts which can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking there |
| 11:02.39 | Dovid | angryuser: I am not using freepbx ;) |
| 11:02.54 | Dovid | i got the error Extension '100' is not valid for automatic login of agent '100' |
| 11:02.57 | angryuser | oh it is not your post related ;) |
| 11:03.26 | Dovid | :) |
| 11:03.32 | Dovid | i figured it out. I was missing a , |
| 11:10.01 | *** join/#asterisk MindTheGap (n=MindTheG@c9503f78.bhz.virtua.com.br) |
| 11:11.46 | *** join/#asterisk Azam (n=azamzia@58-65-160-140.nayatel.pk) |
| 11:15.10 | Dovid | another question. If a user is logged in to a queue and they are DND it goes to VM, is there any way if the user is on DND to go to the next available user ? |
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| 11:25.44 | *** join/#asterisk _gm (n=mustafa@221.120.225.186) |
| 11:26.00 | _gm | got a UA, but i'm in state 1 |
| 11:26.44 | _gm | any idea about above message? |
| 11:27.52 | angryuser | <Dovid> hi, i am using snom phones, they got dnd button ;) if it is presed * switch to next agent |
| 11:39.47 | Dovid | angryuser: How do you have ur queue set up ? I have a snom here and a Eye beam. if dnd on eyebeam is enabled it craps out |
| 11:42.42 | Dovid | angryuser: i was over complicating it. works like a charm now ;) |
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| 12:02.31 | _gm | anyone tried compiling sangoma drivers? |
| 12:02.38 | _gm | i m getting invalud arguement (22) |
| 12:02.46 | _gm | when i issue ztcfg -vv |
| 12:04.51 | *** join/#asterisk phix (n=threat@123-243-44-131.tpgi.com.au) |
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| 12:13.47 | tzafrir | what versions? |
| 12:14.13 | tzafrir | _gm, also consider the possibility you still have an older version of zaptel loaded |
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| 12:16.24 | *** join/#asterisk Al_WinKiller (i=Alex_Win@83.139.12.188) |
| 12:17.36 | Al_WinKiller | hi guys, need help I got two Asterisk ( first is on CentOS (1.4) second is on FreeBSD (1.2) ) and some phone connected to them ( soft phones and cisco ip phones ) , they can call each other but I can't call from one server to another,, pls help |
| 12:19.37 | *** join/#asterisk anonymouz666 (n=anonymou@201.19.126.76) |
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| 12:27.24 | mmmToop | anyone know how to find out if a SIP device has done a 302 "Moved Temp.." in the dialplan? |
| 12:27.43 | mmmToop | tried to get SIP headers...no luck |
| 12:28.00 | the_5th_wheel | if i get asterisk top record my phonecalls, how big will the files be per minute? |
| 12:28.36 | mmmToop | depends on the format |
| 12:28.44 | mmmToop | if u use wav49 (gsm) |
| 12:28.55 | Al_WinKiller | can someone help me ? pls ? |
| 12:29.00 | *** join/#asterisk agx (n=AGX@host63-216-static.34-88-b.business.telecomitalia.it) |
| 12:29.30 | *** join/#asterisk russellb (n=russell@asterisk/developer-and-stable-maintainer/drumkilla) |
| 12:29.30 | *** mode/#asterisk [+o russellb] by ChanServ |
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| 12:30.59 | mmmToop | Al_Windkiller...use IAX trunks |
| 12:33.06 | Al_WinKiller | I do |
| 12:33.15 | Al_WinKiller | I got it in my iax conf |
| 12:34.56 | Al_WinKiller | still doesn't work :( ,,, i go for smoke |
| 12:34.58 | Al_WinKiller | :( |
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| 12:37.56 | Zeeek | Splash |
| 12:38.17 | angryuser | <Dovid> forgot to tell you agentcallbacklogin ask's you ext number because system need to assosiate ext/agent_number, from other side it let you change agent's workplac |
| 12:38.27 | angryuser | *workplace |
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| 12:44.33 | *** join/#asterisk nebojsajsimic (n=nebojsaj@cable-89-216-16-106.static.sbb.co.yu) |
| 12:44.38 | nebojsajsimic | Hi all |
| 12:44.56 | nebojsajsimic | can someone give a little help with Festival |
| 12:46.42 | nebojsajsimic | rejected from asterisk not in access list |
| 12:47.00 | nebojsajsimic | i know that it is some little problem for You |
| 12:47.02 | nebojsajsimic | :) |
| 12:47.52 | russellb | not in what access list? |
| 12:48.30 | nebojsajsimic | when i make a call to festival |
| 12:48.38 | nebojsajsimic | <PROTECTED> |
| 12:48.39 | Zeeek | hey russellb |
| 12:48.42 | *** join/#asterisk RypPn (i=TuMbL@rosscom.demon.co.uk) |
| 12:48.54 | nebojsajsimic | from asterisk |
| 12:49.03 | *** join/#asterisk JT (n=j@unaffiliated/jt) |
| 12:49.05 | nebojsajsimic | i get that message |
| 12:49.13 | Zeeek | I'm looking at the asterisk GUI - pretty amazing |
| 12:49.21 | the_5th_wheel | how can i set asterisk up to automatically record phonecalls going to a queue |
| 12:49.24 | russellb | Zeeek: cool :) |
| 12:49.31 | Zeeek | I think I know too much |
| 12:49.54 | Zeeek | but I though you could add modules to the voip-only box? |
| 12:50.05 | nebojsajsimic | russellb can you help ??? |
| 12:50.13 | russellb | nebojsajsimic: probably not, sorry |
| 12:50.19 | nebojsajsimic | ok thx |
| 12:50.28 | russellb | Zeeek: not sure ... |
| 12:50.41 | russellb | theoretically, probably, but you're not supposed to open the box :) |
| 12:50.46 | Zeeek | it's a little hard testing the config with n o phones :) |
| 12:50.59 | Zeeek | I have to bring one back from home |
| 12:51.13 | Zeeek | but the appliance is pretty cool |
| 12:51.25 | russellb | glad you like it |
| 12:51.28 | *** join/#asterisk GBR_ (n=gbr@200.103.96.98) |
| 12:51.42 | Zeeek | I don't know if you know the story of Astricon Paris 2 yrs ago? |
| 12:52.00 | russellb | nope ... wasn't there |
| 12:52.11 | Zeeek | I asked MArk after his keynote (in front of the whole audience) why Digium didn't make a small box with the hardware... |
| 12:52.22 | Zeeek | He looked at me like I was from Mars |
| 12:52.40 | Zeeek | I learned an hour later that they did make such a box but it was top secret |
| 12:53.03 | Zeeek | there was a big circle of people standing around a table and no one was allowed near |
| 12:53.22 | Zeeek | people were being called over: "Psssssst, look at this!" |
| 12:53.32 | Zeeek | Hilarious |
| 12:54.41 | Zeeek | I thought I heard then that you could plug both kinds of modules in. Since the connectors are there, I'm dying of curiosity |
| 12:56.40 | *** join/#asterisk Frogzoo (n=Frogzoo@CPE-121-218-175-103.nsw.bigpond.net.au) |
| 12:56.47 | Greek-Boy | lol Zeeek |
| 12:57.40 | Zeeek | Greek-Boy LO @ the story? |
| 12:57.59 | Greek-Boy | yip |
| 12:58.11 | Zeeek | it was funny, yeah |
| 12:59.13 | Greek-Boy | have u been to every astricon? |
| 12:59.53 | Zeeek | Greek-Boy no, just 2: Madrid and Paris |
| 13:00.12 | *** join/#asterisk ArchSSM (n=tommy@host-81-191-139-130.bluecom.no) |
| 13:00.17 | Zeeek | I can never go to the US ones because we leave just in the beginning of Sept |
| 13:02.32 | Greek-Boy | at least u got to go to 2. |
| 13:02.49 | Greek-Boy | i haven't even been to 1 |
| 13:02.59 | Greek-Boy | hopefully this year will be my first one :) |
| 13:09.39 | Zeeek | oh? Which one? |
| 13:10.34 | Greek-Boy | US |
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| 13:15.41 | *** join/#asterisk [TK]D-Fender (n=joe@MTRLPQ02-1177745839.sdsl.bell.ca) |
| 13:16.08 | Zeeek | cool |
| 13:18.51 | *** join/#asterisk oej (n=olle@147.240.13.217.in-addr.dgcsystems.net) |
| 13:19.02 | *** join/#asterisk lirakis (i=lirakis@66.252.24.133) |
| 13:24.08 | *** join/#asterisk af_ (n=getsmart@88-149-240-211.dynamic.ngi.it) |
| 13:26.22 | *** join/#asterisk corramor (n=corramor@216.207.245.1) |
| 13:38.19 | *** join/#asterisk oej (n=olle@147.240.13.217.in-addr.dgcsystems.net) |
| 13:40.01 | *** join/#asterisk _Krieger_ (n=krieger@193.39.118.158) |
| 13:40.40 | *** join/#asterisk saftsack (n=oliver@p54A7F583.dip.t-dialin.net) |
| 13:41.03 | saftsack | hi, is a via epia 1000mhz fast enough for asterisk with about 6 callers? |
| 13:41.49 | Zeeek | I'sd think so but it depends on things like transcoding |
| 13:41.59 | saftsack | no transcoding. just voicemail |
| 13:42.14 | mmmToop | check out the voip-info dimentioning page |
| 13:42.29 | mmmToop | but will be fine I think |
| 13:42.29 | saftsack | mmmToop, where to find? |
| 13:42.33 | Zeeek | yopu mean dementing page? |
| 13:42.43 | saftsack | yes |
| 13:42.45 | Zeeek | or dimensioning? |
| 13:42.58 | mmmToop | http://www.voip-info.org/wiki/view/Asterisk+dimensioning |
| 13:43.23 | mmmToop | ;) |
| 13:43.37 | *** join/#asterisk stkn|work (n=stkn|wor@gentoo/developer/pdpc.active.stkn) |
| 13:43.41 | *** join/#asterisk lunaphyte_ (n=lunaphyt@70.90.148.3) |
| 13:44.11 | saftsack | thanks ,) |
| 13:45.51 | *** join/#asterisk AndyGraybeal (n=andy@node210.34.251.72.1dial.com) |
| 13:48.05 | tzafrir | saftsack, yes |
| 13:50.25 | *** join/#asterisk AndyGraybeal (n=andy@node210.34.251.72.1dial.com) |
| 13:51.07 | Zeeek | what does Registration from .... failed for ... - ACL error (permit/deny) mean? |
| 13:52.57 | *** join/#asterisk ifnotwhynot (n=davidh@196.211.34.2) |
| 13:59.19 | *** join/#asterisk marlow (n=marlow@loke.sca.airwire.ie) |
| 14:02.29 | *** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 14:04.14 | *** join/#asterisk ddunavant (n=David@pool-71-163-223-147.washdc.east.verizon.net) |
| 14:06.42 | ifnotwhynot | hi there i am trying to get more familiar with the variables used in the asterisk dial plan, tell me could you do something like this _XXXXAXXXXXXXXXXT,1,Set(${from_number} = S{EXTEN:0:4}) _XXXXAXXXXXXXXXXT,2,NoOP(${from_number}) to display the last XXXXXXXXXX? |
| 14:07.11 | Al_WinKiller | guys when I am calling from one asterisk to another I get this from destination |
| 14:07.13 | Al_WinKiller | 83.139.12.187, request '1299501@default' does not exist |
| 14:07.19 | Al_WinKiller | what can I do ? |
| 14:07.25 | Al_WinKiller | cuz lacally number works |
| 14:07.26 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 14:07.29 | ifnotwhynot | if i do i get nothing in the NoOp command(that should print the last XXXX to cli sceen |
| 14:07.39 | Zeeek | does the default context exist and is there an exten 1299501 in it? |
| 14:07.48 | Zeeek | Al_WinKiller ^^^ |
| 14:07.48 | ifnotwhynot | sorry the first numbers XXXX |
| 14:08.08 | ifnotwhynot | no Zeeek you have to create it |
| 14:08.25 | Al_WinKiller | in iax.conf ? |
| 14:08.35 | Zeeek | in your extensons |
| 14:08.46 | ifnotwhynot | yes |
| 14:09.05 | ifnotwhynot | no in your extensions.conf |
| 14:09.28 | ifnotwhynot | what do you want to do zeeek? |
| 14:09.33 | Al_WinKiller | let me see |
| 14:09.35 | [TK]D-Fender | ifnotwhynot: that is broken about 3 different ways |
| 14:10.05 | ifnotwhynot | [TK]D-Fender what do you mean? |
| 14:10.22 | *** join/#asterisk coppice (n=chatzill@175.203.17.210.dyn.pacific.net.hk) |
| 14:10.23 | [TK]D-Fender | ifnotwhynot: You should not have any white-space in your "Set", next, you do not "reference" your variable that you want to set, that pulls it s value, not the name itself. |
| 14:10.45 | [TK]D-Fender | ifnotwhynot: And you really want letters mixed in with your numbers in the pattern? |
| 14:11.21 | [TK]D-Fender | ifnotwhynot: Whats with the "A", "T", and "S" bit in there? |
| 14:11.25 | Zeeek | if he can dial them, they'll work |
| 14:11.36 | ifnotwhynot | but i get letters from the cli string(dtmf) from the pabx |
| 14:11.39 | Zeeek | well, A will |
| 14:11.52 | *** join/#asterisk _mwoodj_ (n=MWoodJ@pdpc/sponsor/digium/hyper-eye) |
| 14:12.05 | [TK]D-Fender | ifnotwhynot: If you say so.... |
| 14:12.14 | Al_WinKiller | thx, works :D |
| 14:12.15 | ifnotwhynot | true |
| 14:13.23 | ifnotwhynot | if i do a noop of the ${EXTEN} i get 8989A0112328676T |
| 14:13.59 | [TK]D-Fender | ifnotwhynot: Like I said, your set was VERY abd. |
| 14:14.03 | [TK]D-Fender | bad* |
| 14:14.07 | ifnotwhynot | i just want to link 8989 to a variable i can call by name |
| 14:16.57 | ifnotwhynot | thank you [TK]D-Fender the spaces was the problem |
| 14:17.24 | [TK]D-Fender | ifnotwhynot: half of the problem. the other is your variable referencing and added chars |
| 14:19.12 | Zeeek | has anyone ever set up an asterisk appliance? |
| 14:19.23 | ifnotwhynot | working now thanks |
| 14:19.25 | *** join/#asterisk lmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 14:19.25 | *** mode/#asterisk [+o lmadsen] by ChanServ |
| 14:20.29 | Zeeek | plus my firends' asterisk is logging in and I'm seeing "Got SIP response 489 "Bad event" back from ...." every few minutes |
| 14:21.00 | *** join/#asterisk dth0 (n=dth0@189-19-45-26.dsl.telesp.net.br) |
| 14:21.12 | Al_WinKiller | it works , but when I cal from 1 asterisk to 2 asterisk and I peak up ( soft phone ) it is still ringing :( |
| 14:21.17 | anonymouz666 | maybe someone is sending a SIP method that Asterisk does not support? |
| 14:21.18 | *** join/#asterisk pabloff (n=pablov@85.136.133.112.dyn.user.ono.com) |
| 14:21.21 | pabloff | hi all |
| 14:21.34 | Zeeek | anonymouz666it's another asterisk |
| 14:22.59 | [TK]D-Fender | Zeeek: like QUALIFY <- |
| 14:23.09 | file | message waiting indication. |
| 14:23.11 | Zeeek | ah |
| 14:23.23 | Al_WinKiller | on first server I use cisco phone on second is zoiper connected ( gsm is used ) |
| 14:23.34 | [TK]D-Fender | file: that too :) |
| 14:23.57 | Zeeek | I've never seen it beofre with a zillion things connected to my box here |
| 14:24.07 | Zeeek | he is running 1.4 and I'm running 1.2 |
| 14:24.24 | *** part/#asterisk dth0 (n=dth0@189-19-45-26.dsl.telesp.net.br) |
| 14:24.25 | Al_WinKiller | yep,,, it is weird :( |
| 14:24.38 | file | Zeeek: Asterisk doesn't support receiving MWI notification over SIP, never has... so if you connect an Asterisk box up to another Asterisk box, set the mailbox option, that'll pop up eventually |
| 14:24.38 | Zeeek | so file, what do you know about the appliance? |
| 14:24.55 | file | I have barely been involved with it... what'cha wanna know |
| 14:25.49 | Zeeek | for one thing I'm getting refused login on my 1.2 asterisk with an ACL error. But what's odd is that AIX is selected and it says "login from <sip:.....> |
| 14:26.00 | Zeeek | also the port in advanced settings is 5060 |
| 14:26.06 | file | the GUI? I have no idea |
| 14:26.22 | Zeeek | should I just go in and edit the confs? |
| 14:26.24 | *** join/#asterisk ManxPower (n=manxpowe@202.sub-70-222-142.myvzw.com) |
| 14:26.29 | Zeeek | I'll bet that would raise some hell |
| 14:26.51 | Zeeek | anyway the whole point was to test the appliance out of the box |
| 14:27.07 | BBHoss | Zeeek, what problems are you having with it |
| 14:27.36 | Zeeek | My biggest problem is resisting the temptation to open the box to see if modules can be plugged in :) |
| 14:28.02 | BBHoss | ive seen the inside of them, and it looks modular, but i can't be sure |
| 14:28.11 | Zeeek | otherwise I was trying to set up my production asterisk 1.2 as a "custom voip service provider" |
| 14:28.34 | Zeeek | only because I don'thave any phoines available here atm |
| 14:28.58 | Zeeek | I wanted to see if it would register |
| 14:29.05 | BBHoss | yeah its really touchy with the custom provider, doesn't give you all the options you need |
| 14:29.06 | Zeeek | so far, no luck |
| 14:29.16 | Zeeek | BBHoss you have one? |
| 14:29.21 | BBHoss | yeah |
| 14:29.29 | Zeeek | voip only? |
| 14:29.29 | *** join/#asterisk FabiOne (n=FabiOne@host107-144-static.59-88-b.business.telecomitalia.it) |
| 14:29.33 | FabiOne | hi all |
| 14:29.50 | BBHoss | 4FXO/4FXS |
| 14:29.53 | *** join/#asterisk AndyGraybeal_ (n=andy@node138.39.251.72.1dial.com) |
| 14:29.58 | Zeeek | like it? |
| 14:30.14 | BBHoss | eh, its alright, but I tend to stay away from GUIs |
| 14:30.24 | Zeeek | I've never used a GUI beofre! |
| 14:30.35 | Zeeek | s/beofre/before/ |
| 14:30.49 | BBHoss | well the nice thing is it won't mangle your config files like other guis will |
| 14:30.58 | BBHoss | most of the time :) |
| 14:31.08 | Zeeek | so I can add stuff like users to the conf? |
| 14:31.25 | BBHoss | should be able to |
| 14:31.31 | *** join/#asterisk worgil (n=worgil@78.166.127.4) |
| 14:31.36 | Zeeek | but that would defeat the purpose of the tests |
| 14:31.45 | BBHoss | they are coming out with an AA-250 soon, for bigger businesses |
| 14:32.14 | Zeeek | I don't need bigger. In all fairness, the GUI is great visually and it looks logical enough |
| 14:32.19 | *** join/#asterisk esaym (n=user@cpe-72-183-198-134.satx.res.rr.com) |
| 14:32.37 | BBHoss | back when i got mine, the gui was still in beta |
| 14:32.50 | BBHoss | the networking side really needs to be worked over though, still not kosher |
| 14:32.58 | Zeeek | in what way? |
| 14:33.02 | BBHoss | no way to port-forward using the gui |
| 14:33.28 | BBHoss | and when you change options in the networking page, sometimes they don't get saved |
| 14:33.37 | BBHoss | especially the Enable SSH option |
| 14:33.52 | Zeeek | that worked here |
| 14:34.07 | BBHoss | yeah, i've just had trouble with it before |
| 14:34.20 | *** part/#asterisk dofear (n=arodef@202-91-197-146.intrapower.net.au) |
| 14:34.30 | BBHoss | i think it had to do with me changing multiple settings on different pages |
| 14:34.30 | *** join/#asterisk dofear (n=arodef@202-91-197-146.intrapower.net.au) |
| 14:35.07 | Zeeek | without using "apply" ? |
| 14:35.18 | *** part/#asterisk lmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 14:35.59 | Zeeek | can the appliance be behind NAT? |
| 14:36.29 | BBHoss | yeah, if you mess with it enough |
| 14:36.35 | Zeeek | heh |
| 14:37.07 | worgil | which softphone can i use for windows ? |
| 14:37.25 | Zeeek | X-Lite, Gizmo Project, SJPhone, Zoiper |
| 14:37.38 | worgil | thanks Zeeek |
| 14:37.43 | Zeeek | np |
| 14:38.13 | draygon | Zeeek Which is the best out of those listed? |
| 14:38.22 | draygon | I usually use xlite |
| 14:38.24 | Zeeek | they're all the same price |
| 14:38.31 | Zeeek | I prefer X-Lite |
| 14:38.32 | draygon | isn't xlite free? |
| 14:38.40 | Zeeek | yes, they're all the same price |
| 14:38.42 | draygon | well they have a paid version |
| 14:38.46 | draygon | ah |
| 14:38.58 | worgil | Zeeek but x-lite not support gsm and g729 codec |
| 14:39.02 | draygon | is anyone willing to setup a simple pbx for me with incoming and outgoing calls? |
| 14:39.05 | Zeeek | X-Lite is very good. |
| 14:39.06 | draygon | I'm willing to pay. |
| 14:39.16 | styelz | xlite has video |
| 14:39.18 | Zeeek | draygon see http://onsip.com |
| 14:39.21 | styelz | or not |
| 14:39.23 | styelz | ? |
| 14:39.42 | Zeeek | or use http://freeworlddialup.com free |
| 14:39.54 | draygon | i want it to run on my server |
| 14:40.01 | styelz | i tried the video with asterisk, works ok |
| 14:40.06 | Zeeek | You can test onsip.com free |
| 14:40.23 | *** join/#asterisk _charly_ (i=kroseneg@sunrise.schmidham.net) |
| 14:40.28 | Zeeek | draygon oh, set it up for you? I'm sure many can help |
| 14:40.55 | draygon | Is there a good guide installing asterisks on CentOS somewhere? |
| 14:41.27 | Zeeek | just use the Book |
| 14:41.31 | BBHoss | ~book |
| 14:41.32 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com, or see ~buybook |
| 14:41.55 | BBHoss | draygon, also, be sure to look at the section that lets you run it as non-root |
| 14:42.07 | worgil | Zeeek can we use Gizme Projet for our asterisk server ? |
| 14:42.18 | worgil | Gizme=Gizmo |
| 14:42.19 | Zeeek | of course you can |
| 14:42.35 | Zeeek | asterisk works with clients that haven't even been invented yet |
| 14:42.42 | Zeeek | Gizmo is a little special though |
| 14:42.52 | worgil | zoiper is not friendly |
| 14:42.56 | Zeeek | You have to set your asterisk server as a "secondary" server |
| 14:43.14 | Zeeek | Zoiper has a nice tiny footprint |
| 14:43.20 | *** join/#asterisk beek (n=klinebl@65.211.106.243) |
| 14:43.39 | worgil | sure but for new user not easy |
| 14:43.43 | Zeeek | Gizmo won't work without using a Gizmo account, but that's free and you can just ignore it |
| 14:44.38 | Zeeek | I guess I'll go home now |
| 14:44.54 | *** part/#asterisk Zeeek (n=Heh@pdpc/supporter/active/Zeeek) |
| 14:45.12 | *** part/#asterisk mintee (n=mintee@72-165-177-90.dia.static.qwest.net) |
| 14:45.15 | *** join/#asterisk mintee (n=mintee@72-165-177-90.dia.static.qwest.net) |
| 14:46.03 | *** part/#asterisk mmmToop (n=michaelt@firewall.datapro.co.za) |
| 14:46.27 | *** join/#asterisk fskrotzki (n=fskrotzk@host198.textwise.com) |
| 14:48.14 | *** join/#asterisk jameswf-home (n=james@dsl093-157-131.phx1.dsl.speakeasy.net) |
| 14:50.23 | *** join/#asterisk putnopvut (n=putnopvu@216.207.245.1) |
| 14:50.56 | *** join/#asterisk drako (n=thinkpad@nelug/coreteam/luisjose) |
| 14:51.40 | drako | [Feb 11 09:47:40] NOTICE[8762]: rtp.c:787 process_rfc3389: Comfort noise support incomplete in Asterisk (RFC 3389). Please turn off on client if possible. Client IP: 0.0.0.0 |
| 14:51.45 | drako | how can i get rid of this ? |
| 14:52.33 | *** join/#asterisk ZaVoid (n=zavoid@66-95-182-90.client.dsl.net) |
| 14:52.36 | ZaVoid | morning guys |
| 14:52.50 | ZaVoid | can someone point me to a function/feature that i can generate a random number in my dialplan? |
| 14:53.03 | draygon | can someone help me install a pbx as non root user please |
| 14:53.08 | ZaVoid | for unique ID purposes |
| 14:53.25 | Frogzoo | ZaVoid: Random() |
| 14:53.37 | ZaVoid | oh |
| 14:54.02 | ZaVoid | [Description] |
| 14:54.02 | ZaVoid | Random([probability]:[[context|]extension|]priority) |
| 14:54.03 | ZaVoid | <PROTECTED> |
| 14:54.03 | ZaVoid | DEPRECATED: Use GotoIf($[${RAND(1,100)} > <number>]?<label>) |
| 14:54.05 | ZaVoid | hmmm |
| 14:54.44 | Frogzoo | deprecated! the * v2 book's getting a little out of date then |
| 14:55.27 | ZaVoid | lol |
| 14:55.45 | ZaVoid | hmm that does a bit more then i want i think |
| 14:57.06 | *** join/#asterisk AndyGraybeal_ (n=andy@node138.39.251.72.1dial.com) |
| 14:57.51 | [TK]D-Fender | drako: Tell your device to stop. |
| 14:58.08 | beek | draygon: http://lists.digium.com/pipermail/asterisk-dev/2003-October/001823.html |
| 14:58.10 | b11d | all of my voicemail is being stored with timezone UTC info, but my server existsin CST6CDT.. Comedian Mail is misreading the voicemail timestamps as a result.. any way to fix this? I didnt see a timezone preference in voicemail.conf and would rather not put the server back to UTC.. |
| 14:58.16 | [TK]D-Fender | drako: looking like a softphone on the server itself I'm guessing byt eh IP |
| 14:58.23 | b11d | although I will if there is no other choice.. not that big of a deal |
| 14:58.58 | drako | [TK]D-Fender, im using twinkle softphone, 2 ATA grandstream and 2 grandstream IP PHONE |
| 14:59.25 | nebojsajsimic | i get SIOD ERROR: unbound variable : tts_textasterisk |
| 14:59.55 | nebojsajsimic | please help with this |
| 15:00.43 | Frogzoo | on an fxo line which I pass through to an fxs handset, it takes a couple of rings on the fxo line before the fxs handset rings - must I disable caller id, or is there a better solution? |
| 15:01.14 | *** join/#asterisk Docfxit (n=none@cpe-72-129-118-135.socal.res.rr.com) |
| 15:01.15 | [TK]D-Fender | Frogzoo: Yes, waiting for CID will delay passing on the ring. |
| 15:01.18 | Qwell | Frogzoo: if you want callerid, you have to wait |
| 15:01.27 | [TK]D-Fender | Frogzoo: You want it, you wait. |
| 15:01.35 | Qwell | though |
| 15:01.39 | Frogzoo | ok, well pffft |
| 15:01.50 | Qwell | actually, yes I do know why. nevermind |
| 15:01.59 | *** join/#asterisk teknoprep (n=Chris@unaffiliated/teknoprep) |
| 15:02.19 | pkunkra | caller id is always sent between the first and second rings in i recall correctly |
| 15:02.29 | Qwell | there could probably be an option added like "yes, I know callerid is coming - send the call on anyways, and notify it when I get the callerid" |
| 15:02.38 | [TK]D-Fender | pkunkra: In North America typically, yes |
| 15:02.47 | Frogzoo | Qwell: that would be desirable |
| 15:02.55 | Qwell | Frogzoo: I can imagine |
| 15:03.05 | [TK]D-Fender | Qwell: Won't work on FXO spillover onto FXS |
| 15:03.10 | nebojsajsimic | how to dial from php ??? can someone give me some example line |
| 15:03.12 | Qwell | [TK]D-Fender: why not? |
| 15:03.12 | nebojsajsimic | thx |
| 15:03.13 | pkunkra | i'm not familar with phone systems outside the U.S. |
| 15:03.16 | [TK]D-Fender | Qwell: SIP can handle this IIRC. |
| 15:03.22 | Qwell | so can FXS :) |
| 15:03.31 | drako | [TK]D-Fender, btw, im having randon consoles disconnect and when i call uptime, it says it just started so i get it is a crash, but there is nothing non-normal at logs |
| 15:03.34 | Qwell | I mean...think about it |
| 15:03.54 | [TK]D-Fender | Qwell: the phone on your FXS has its specific time where you can send it. If you don't ahve it then, you cna't change your mind after. |
| 15:03.58 | Qwell | incoming ring, dial out, outgoing ring, incoming cid, outgoing cid, incoming ring, outgoing ring, rinse repeat |
| 15:04.36 | ManxPower | ACTUALLY, many devices that support Caller*ID (at least USA style) will accept the Caller*ID FSK spill most anytime. |
| 15:04.43 | drako | [Feb 11 10:00:48] WARNING[9154]: chan_iax2.c:9385 build_user: Unable to support trunking on user 'konoko_pe' without zaptel timing |
| 15:04.45 | drako | damn |
| 15:04.56 | ManxPower | ASTERISK, however, does not. |
| 15:05.00 | *** join/#asterisk binary-zero (n=binary--@unaffiliated/binary-zero) |
| 15:05.05 | ManxPower | drako: there ya go! |
| 15:05.06 | Qwell | [TK]D-Fender: the same specific time that you would require it to come in |
| 15:05.14 | binary-zero | hi guys - how can i get rid of Got 200 OK on REGISTER that isn't a register |
| 15:05.25 | drako | ManxPower, but i have ztdummy loaded |
| 15:05.34 | binary-zero | is there any configuration due to which i can have SIP wait for the message a bit longer |
| 15:05.40 | ManxPower | drako: perhaps you did not install zaptel before you installed Asterislk |
| 15:05.53 | Qwell | if we don't have the callerid from the fxo by the time the phone needs the fsk spill, then...we aren't getting callerid |
| 15:06.31 | [TK]D-Fender | Qwell: Exactly my point. You can't ahve the FXS catch up after. |
| 15:06.36 | Qwell | an option like that would, obviously, completely remove your ability to use callerid matching in dialplan |
| 15:06.41 | Qwell | [TK]D-Fender: why not? |
| 15:06.52 | Qwell | we send it the same way we receive it |
| 15:06.58 | binary-zero | any one for 200 OK REGISTER issue ? |
| 15:07.03 | ManxPower | Qwell: Well, I guess you could WRITE support for it. |
| 15:07.12 | ManxPower | binary-zero: It is a harmless message. Ignore it. |
| 15:07.34 | binary-zero | ManxPower: would it cause to stop the incoming calls for some duration |
| 15:07.45 | [TK]D-Fender | Qwell: to tell the FXS channel what the CID is, you can only do it between its first 2 rings, any timing desync causing you not to have parsed it out from FXO will mean you can't pass it on. |
| 15:07.46 | binary-zero | as i can see my sip show register as "REGISTERING" ... |
| 15:07.52 | binary-zero | for a little while |
| 15:08.05 | binary-zero | is there any configuration which can delay re-registration time |
| 15:08.56 | ManxPower | binary-zero: not in my experience. |
| 15:08.57 | *** join/#asterisk SteveTotaro (n=Elizabet@c-69-243-124-5.hsd1.md.comcast.net) |
| 15:09.09 | Frogzoo | binary-zero: in sip.conf ";defaultexpiry=120 ; Default length of incoming/outgoing registration |
| 15:09.15 | Qwell | [TK]D-Fender: sure, but if we screw the timing on getting it, we won't have it anyways |
| 15:09.20 | binary-zero | thanks Frogzoo that was what i looking for |
| 15:09.34 | Qwell | maybe it's just too early and I'm missing something obvious |
| 15:09.35 | ManxPower | binary-zero: don't expect it to fix the issue. |
| 15:09.45 | *** join/#asterisk Knorrie (i=knorrie@kantoor.mendix.nl) |
| 15:10.28 | *** join/#asterisk mindCrime (n=chatzill@66.83.208.218.nw.nuvox.net) |
| 15:12.03 | Qwell | what's the name of the kernel package you need to build zaptel on debian? |
| 15:12.05 | ManxPower | It sounds like this is what Qwell wants to do: http://www.artofhacking.com/files/OB-FAQ.HTM |
| 15:12.08 | ManxPower | Is that correct? |
| 15:12.22 | Frogzoo | Qwell: zaptel-source |
| 15:12.33 | Qwell | ManxPower: no, nothing like that |
| 15:13.07 | ManxPower | Qwell: Are you sure? I was referring to being able to send the CID spill at any time. |
| 15:13.08 | tzafrir | linux-headers-`uname -r` ? |
| 15:13.34 | tzafrir | Qwell, ==^ |
| 15:13.42 | Qwell | tzafrir: is that different from linux-kernel-headers? |
| 15:13.51 | Qwell | it says I have the latest of that, but your command gives me new stuff |
| 15:13.54 | *** join/#asterisk SirThomas (n=tomc@mail.kendeco.com) |
| 15:14.06 | Qwell | yeah, that worked |
| 15:14.11 | tzafrir | there is no single linux-headers . There is a linux-headers package per linux-image package |
| 15:14.26 | drmessano | So the question is.. |
| 15:14.50 | tzafrir | (And now for my question: |
| 15:14.54 | Frogzoo | Qwell: you might like to read: https://wiki.ubuntu.com/AsteriskOnUbuntu |
| 15:15.14 | drmessano | Can you ring the FXS as soon as the FXO recieves a ring, and just wait for the callerID, vs waiting for the FXO to get CID and then ringing the FXS, thus removing the delay? |
| 15:15.18 | Qwell | Frogzoo: I've compiled asterisk once or twice before |
| 15:15.27 | Qwell | drmessano: that's what I'm saying! |
| 15:15.29 | *** join/#asterisk tzafrir_laptop (n=tzafrir@62.90.10.53) |
| 15:15.43 | tzafrir | what exactly does the value of the qualify parameter in sip.conf mean? |
| 15:15.56 | ManxPower | drmessano: Yes, if you write a patch to Asterisk |
| 15:16.00 | tzafrir | time between checks? maximal allowed round-trip-time? |
| 15:16.08 | Frogzoo | Qwell: ooh - sarcasm |
| 15:16.12 | mvanbaak | tzafrir: max-allowed roundtrip time |
| 15:16.17 | Qwell | tzafrir: if I remove a NIC, do I have to do anything special to get the existing NIC to become eth0? |
| 15:16.25 | drmessano | Qwell: I've never seen a device that was smart enough to care when it recieves CallerID.. so Asterisk would be the only issue |
| 15:16.30 | ManxPower | tzafrir: qualify=2000 No more than 2000ms of latency in response to the SIP OPTIONS packet qualify= sends. |
| 15:16.30 | *** join/#asterisk CrazyTux[m] (n=CrazyTux@ppp-70-248-182-215.dsl.hstntx.swbell.net) |
| 15:16.35 | mvanbaak | Qwell: no |
| 15:16.44 | ManxPower | if you have even ONE packet lost, the peer will go offline. (SIP) |
| 15:17.17 | ManxPower | And that is why I don't normally use qualify= |
| 15:17.30 | tzafrir | Qwell, in recent versions: the annoying /etc/udev/rules.d/z45_persistent-net-generator.rules /etc/udev/rules.d/z25_persistent-net.rules |
| 15:19.28 | drako | ManxPower, i see |
| 15:19.37 | drako | ManxPower, and about the noise one |
| 15:22.29 | drako | [Feb 11 10:18:37] NOTICE[9678]: rtp.c:787 process_rfc3389: Comfort noise support incomplete in Asterisk (RFC 3389). Please turn off on client if possible. Client IP: 0.0.0.0 |
| 15:23.43 | ManxPower | drako: turn off VAD/CND on your VoIP client. |
| 15:25.03 | drako | ok |
| 15:27.07 | drako | damn but it keep crashing |
| 15:27.10 | drako | pbx*CLI> |
| 15:27.10 | drako | Disconnected from Asterisk server |
| 15:27.10 | drako | Executing last minute cleanups |
| 15:27.33 | ManxPower | drako: "asterisk -cvvv" that should give you more info on the screen when it crashes. |
| 15:29.03 | *** join/#asterisk CrazyTux[m] (n=CrazyTux@ppp-70-248-182-215.dsl.hstntx.swbell.net) |
| 15:29.29 | *** join/#asterisk puzzled (n=patrick@puzzled.xs4all.nl) |
| 15:31.08 | *** join/#asterisk inv_arp[work] (n=junya@c-76-26-24-197.hsd1.fl.comcast.net) |
| 15:31.51 | _Krieger_ | how to easily limit max number of concurrent calls? |
| 15:32.41 | *** join/#asterisk anthm (n=anthm@CPE-72-131-113-50.wi.res.rr.com) |
| 15:32.41 | *** mode/#asterisk [+o anthm] by ChanServ |
| 15:34.47 | ifnotwhynot | just spend 3 hours reworking my dialplan for app_RxFAXX keaps on shutting down my asterisk server only to realize i dont have fax=yes in my zapata,,,dumb,dumber,MEMEMEMEMEM! |
| 15:36.01 | *** join/#asterisk jtknapp (n=skip@65-126-63-1.dia.static.qwest.net) |
| 15:39.41 | tzafrir | ifnotwhynot, "shuts down my asterisk server" == "segfaults"? |
| 15:39.48 | *** part/#asterisk binary-zero (n=binary--@unaffiliated/binary-zero) |
| 15:40.05 | tzafrir | anyway, get asterisk 1.2.4 and 1.4.6 beta while their hot! :-) |
| 16:07.32 | *** join/#asterisk jbot (i=ibot@pdpc/supporter/active/TimRiker/bot/apt) |
| 16:07.32 | *** topic/#asterisk is Asterisk: The Open Source Telephony Application Platform (asterisk.org) -=- Asterisk, -Addons, Libpri 1.6.0-beta2 (2008/01/28), Asterisk 1.4.18 (2008/02/07), *-Addons 1.4.5 (2007/12/1), Zaptel 1.4.8 (2008/01/14), Libpri 1.4.3 (2007/12/13) -=- #asterisknow or #asterisk-gui for AsteriskNOW (asterisknow.org) -=- #switchvox for Switchvox (switchvox.com) -=- #freepbx for FreePBX/trixbox |
| 16:07.50 | putnopvut | Hmm, do any queue members have a higher penalty than others? |
| 16:08.01 | hi365 | nope |
| 16:08.07 | *** join/#asterisk [T]ank (n=ckwall@c-71-195-199-107.hsd1.ut.comcast.net) |
| 16:08.24 | [T]ank | how many concurrent calls should I be able to do with gsm and sip? |
| 16:08.38 | putnopvut | What happens if you run the command "queue show" from the CLI? Specifically what does it show for the member's statuses? |
| 16:08.50 | hmm-home | [t]ank: what a fantastically vague question |
| 16:08.51 | hi365 | ill look |
| 16:09.20 | x86 | [T]ank: it really depends on: 1. your server hardware, and 2. the amount of bandwidth available between both sites |
| 16:09.52 | x86 | [T]ank: it also depends on if you're having to translate between codecs |
| 16:10.29 | *** join/#asterisk Al_WinKiller (i=Alex_Win@83.139.12.188) |
| 16:10.45 | [T]ank | hmm-home: sorry... let me put it differently. If I have a 1.5 meg point to point t1, between two sites, and I am connecting polycome 301 phones to a hpdl380 server... what could be my limit on channels if I am using gsm the entire way and using sip. |
| 16:10.51 | Al_WinKiller | guys I get this fault ( calling from one to another asterisk (iax) ) |
| 16:10.53 | Al_WinKiller | Operating with different codecs 4[(ulaw)] 14[(gsm|ulaw|alaw)] , can't native bridge.. |
| 16:11.04 | Al_WinKiller | ? |
| 16:11.07 | Al_WinKiller | any help ? pls ? |
| 16:11.57 | hi365 | putnopvut: http://pastebin.ca/899989 |
| 16:11.57 | putnopvut | Al_WinKiller: I think the codec ordering when using IAX is important. So the one that has gsm|ulaw|alaw could be changed to have ulaw first, it might fix the problem. |
| 16:12.15 | [T]ank | is g729 better to use than sip? yes, i know that is also vague... but generally speaking... which is better? |
| 16:12.46 | dkatz334 | Tank, one is a CODEC the other a signalling system, they're not interchangable. |
| 16:13.12 | [T]ank | sorry... i meant to ask g729 vs gsm.... not sip. still tired this morning. |
| 16:13.53 | putnopvut | hi365: Hmm, nothing too terrible in that output. |
| 16:14.02 | hi365 | you would think! |
| 16:14.17 | hi365 | ame config worked fine on another system, here its just stuck... |
| 16:14.22 | Al_WinKiller | ok, i will try |
| 16:14.25 | Al_WinKiller | thnx |
| 16:14.36 | putnopvut | hi365: that makes it all the more perplexing then. |
| 16:14.45 | hi365 | yup |
| 16:14.55 | putnopvut | Same version of Asterisk? |
| 16:15.07 | hi365 | yup |
| 16:15.09 | hi365 | 1.2.26 |
| 16:15.40 | dkatz334 | Tank, I prefer gsm due to licensing issues with g729 |
| 16:15.48 | putnopvut | Ooh, 1.2. I haven't looked at that in ages. |
| 16:15.55 | BBHoss | speex is nice if you don't have lots of calls |
| 16:17.56 | hi365 | was on 1.4.17 but went back cause i though maybe it was causing issues |
| 16:18.43 | BBHoss | hi365, what kind of issues were you having |
| 16:19.09 | hi365 | BBHoss: ringall isnt ringing all |
| 16:19.17 | hi365 | <hi365> sure. got a queue (duh). 20ish extensions. in ringall. call comes in. after x amount of time SOME extnesions start to ring |
| 16:19.17 | hi365 | [18:05:59] <hi365> x can be anywhere from 0 seconds to 3 minutes |
| 16:19.30 | putnopvut | hi365: don't know what to tell you with regards to that queuing issue...if it's happening with 1.4 too, I might be able to help out. |
| 16:21.38 | hi365 | it was happening in 1.4 |
| 16:27.44 | *** join/#asterisk scruz (n=Dell_ope@41.220.73.170) |
| 16:27.54 | scruz | hello all |
| 16:28.08 | *** join/#asterisk b1ch0 (n=ralabiso@static-200-105-209-46.acelerate.net) |
| 16:28.42 | dkatz334 | hello |
| 16:28.52 | BBHoss | sup dog |
| 16:28.57 | *** join/#asterisk uwe (n=uwe@a32-160.adsl.paltel.net) |
| 16:29.05 | drmessano | uwe? |
| 16:29.20 | scruz | nothing much |
| 16:29.24 | uwe | drmessano? |
| 16:29.32 | *** join/#asterisk magumbade (n=magumbad@p5497D8DE.dip.t-dialin.net) |
| 16:29.34 | scruz | trying to get a hang of asterisk and phone systems in general |
| 16:29.58 | drmessano | uwe: I found an old convo of yours in here concerning ringall not working in a queue |
| 16:30.03 | drmessano | Any resolve? |
| 16:30.05 | uwe | i just killed the other uwe |
| 16:30.15 | uwe | oh |
| 16:30.30 | uwe | old conversation here ... how old you mean ? |
| 16:30.35 | drmessano | lol |
| 16:30.37 | BBHoss | scruz: ~book |
| 16:30.46 | drmessano | No clue.. |
| 16:30.54 | BBHoss | ~book |
| 16:30.55 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com, or see ~buybook |
| 16:31.18 | drmessano | http://ircarchive.info/asterisk/2007/3/22/33.html |
| 16:31.27 | uwe | oh , pretty old |
| 16:32.04 | uwe | well |
| 16:32.19 | uwe | drmessano, i think this is simply how the queue work |
| 16:32.42 | dkatz334 | Speaking of books, anybody know of a good book that covers AGIs? |
| 16:32.42 | uwe | the queue retries every for example 10 seconds |
| 16:33.04 | drmessano | ok |
| 16:33.23 | [TK]D-Fender | dkatz334: ATFOT2 covers it. |
| 16:33.43 | uwe | im actually trying to remember exactly what the problem was |
| 16:34.03 | uwe | but at the end i cant confirm it was a problem, rather just queues behave that way |
| 16:34.08 | drmessano | You stated it acted like it was using some other strategy |
| 16:34.14 | [TK]D-Fender | hi365: how about actually showing us the queue's state before the call comes in, another with the call as it enters, and your dialplan. |
| 16:34.52 | dkatz334 | Fender, I was hoping for something more comphrensive than just "Here's what we think it can do. |
| 16:34.59 | [TK]D-Fender | hi365: And while you're at it, your queues.conf |
| 16:35.13 | *** join/#asterisk skirmisha (n=viki@90.154.201.215) |
| 16:35.16 | [TK]D-Fender | dkatz334: The book shows some very solid samples. |
| 16:35.30 | skirmisha | guys can i send IAX as peer only, no need to register |
| 16:35.40 | BBHoss | skirmisha, yeah |
| 16:35.57 | krdian_ | is there any way to play files to connected channel ? |
| 16:36.06 | [TK]D-Fender | skirmisha: Yes |
| 16:36.14 | BBHoss | krdian_, what format |
| 16:36.17 | uwe | i dont remember drmessano |
| 16:36.29 | [TK]D-Fender | krdian_: What is on each end of the call? |
| 16:36.51 | skirmisha | well i see no authority found |
| 16:36.53 | dkatz334 | I'm re-reading now, thanks. |
| 16:36.59 | uwe | thats one year ago ... |
| 16:37.07 | skirmisha | is this because number does not exist |
| 16:37.15 | krdian_ | [TK]D-Fender: i like to inform caller how many minutes he spent on call |
| 16:37.37 | krdian_ | [TK]D-Fender: or send him some information s like that |
| 16:37.51 | krdian_ | [TK]D-Fender: externaalivr ? |
| 16:37.59 | [TK]D-Fender | krdian_: "core show application chanspy" <- |
| 16:38.03 | skirmisha | ???? |
| 16:38.58 | [TK]D-Fender | skirmisha: Could be that, could be an improper context specified, bad call formatting, etc. |
| 16:38.58 | *** join/#asterisk ddunavant (n=David@pool-71-163-223-147.washdc.east.verizon.net) |
| 16:38.58 | krdian_ | [TK]D-Fender: ok, but this way i can send him info thrugh script ? |
| 16:39.00 | skirmisha | let me check |
| 16:39.10 | [TK]D-Fender | krdian_: Have your script use that app. go read. |
| 16:39.11 | drmessano | uwe: thats cool, thanks :) |
| 16:43.08 | *** join/#asterisk putnopvut (n=putnopvu@216.207.245.1) |
| 16:43.30 | krdian_ | [TK]D-Fender: i don't understand how i can use chanspy in for example perl script |
| 16:44.28 | [TK]D-Fender | krdian_: have your script start a local channel that will Chanspy on the the channel you want to platy the notice to and whisper in on it. |
| 16:44.59 | krdian_ | [TK]D-Fender: ah! i see |
| 16:46.39 | krdian_ | [TK]D-Fender: thanks, i'll try it |
| 16:47.12 | *** join/#asterisk cfh (n=luca@host196-137-static.90-82-b.business.telecomitalia.it) |
| 16:47.43 | scruz | thanks. have the book on my desktop and was looking through it this morning |
| 16:47.45 | *** join/#asterisk florz (i=nobody@2001:1a50:503c:0:0:0:0:1) |
| 16:48.09 | cfh | hi all, i have a problem with the hint function in asterisk 1.4 but with 1.2 it works |
| 16:48.12 | cfh | what can i do / |
| 16:48.13 | cfh | ? |
| 16:48.26 | [TK]D-Fender | cfh: SHOW US the problem, and "hint" is not a "function. |
| 16:48.37 | [TK]D-Fender | cfh: Pastebin is your friend |
| 16:48.39 | BBHoss | scruz, if you're really serious about learning asterisk, i recommend getting paper copy, because it becomes tedious to switch between a terminal and pdf for referencing |
| 16:48.39 | [TK]D-Fender | ~pb |
| 16:48.39 | jbot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste |
| 16:48.52 | scruz | guess i was put off because it starts with hardware, power supplies and whatnot. |
| 16:49.04 | [TK]D-Fender | BBHoss: my fingers auto-home on Alt-Tab :p |
| 16:49.04 | *** part/#asterisk reber (n=reber@193.253.213.73) |
| 16:49.28 | BBHoss | [TK]D-Fender, well even me with dual monitors, its still nice to have a paper reference |
| 16:49.35 | scruz | BBHoss: thanks. will see if i can convince anyone to pay for printing the book :D |
| 16:49.58 | BBHoss | scruz, you can buy it from many bookstores |
| 16:50.01 | [TK]D-Fender | BBHoss: Yeah, thats why I printed myself a copy of each from work. |
| 16:50.07 | BBHoss | heh |
| 16:50.12 | BBHoss | that works :) |
| 16:50.28 | pkunkra | i might buy the book. |
| 16:50.32 | pkunkra | i like paper books. |
| 16:50.43 | pkunkra | plus i get to support the author. :-) |
| 16:50.49 | cfh | [TK]D-Fender : if i set exten => 11,hint,SIP/11 and i try to do in asterisk 1.4 "show hints " i see always State:Idle |
| 16:50.59 | scruz | BBHoss: i don't think we're quite on the same page here. i'd most likely have to order it from another country |
| 16:51.06 | BBHoss | yeah |
| 16:51.14 | BBHoss | wasn't sure |
| 16:51.49 | scruz | since the PDF is free, much better printing it at someone else's expense |
| 16:51.59 | BBHoss | wow nigeria? |
| 16:51.59 | uwe | ive setup sip peer with insecure=port,invite and with a host, also, to the same IP i use to make calls to it, i defined it in anther section ... but after that incoming calls from that ip are not assoicated with the first sip account anymore ... any idea how to work around that ? |
| 16:52.02 | [TK]D-Fender | cfh: you should have "type=peer" , "call-limit=99" in your device's sip.conf entry as well. |
| 16:52.47 | cfh | [TK]D-Fender: call-limit=99 in the [general] section ? |
| 16:52.59 | alexcf | and scruz is setting his hilights off! |
| 16:53.05 | *** join/#asterisk defswork (n=andy@mx2.3gcomms.co.uk) |
| 16:53.25 | scruz | ?? |
| 16:53.29 | scruz | :-/ |
| 16:53.33 | alexcf | my name has cruz in it ;) |
| 16:53.38 | [TK]D-Fender | cfh: taht is not what I said. Read it again |
| 16:53.44 | alexcf | 16:50:54 #asterisk:[12]: * scruz is in Nigeria |
| 16:53.45 | alexcf | :p |
| 16:53.48 | pkunkra | scruz: nigeria, eh? no, i'm not interested in helping you move millions of dollars. :-) |
| 16:53.54 | alexcf | lol! |
| 16:54.12 | BBHoss | 419 heh |
| 16:54.13 | scruz | don't worry, i'm not asking :P |
| 16:54.21 | *** join/#asterisk NetForces (n=courchea@67.70.240.2) |
| 16:55.04 | NetForces | Anyone familiar with Allstream PRI and account codes? Manual dial and account codes works but D() or even M() does not work |
| 16:55.14 | cfh | [TK]D-Fender: ok on the phone section , i have try but it doesnt work , are there some limit on the number of hint ? |
| 16:56.11 | [TK]D-Fender | cfh: apstebin everything |
| 16:56.15 | pkunkra | a friend of mine used to reply to the nigerian scammers and messed around with them. used to pretend to wire the money to a western union located two hours away from where the scammer requested it to be sent. made the scammer drive out there ... but no money. |
| 16:56.27 | BBHoss | heh |
| 16:56.49 | scruz | you should search for 'scambaiter' on youtibe |
| 16:56.49 | scruz | *youtube |
| 16:57.04 | BBHoss | http://www.zug.com/pranks/powerbook/ |
| 16:57.05 | scruz | very funny videos there. |
| 16:57.09 | b11d | any idea why asterisk is timestamping my voicemail as UTC, while the OS is set to CST? |
| 16:57.10 | BBHoss | best one |
| 16:57.30 | BBHoss | b11d, you need to define the timezone in the voicemail.conf file i thinks |
| 16:57.33 | b11d | I did that |
| 16:57.36 | b11d | tz = central |
| 16:57.46 | b11d | according to the voicemail.conf.sample |
| 16:58.09 | scruz | did the commands change from asterisk 1.2 to 1.4? i was trying out a few commands on asteriskwin (cygwin built from 1.2), but the commands didn't work |
| 16:58.12 | BBHoss | no idea then |
| 16:58.23 | BBHoss | scruz, yes, a few have |
| 16:58.25 | *** join/#asterisk Corydon76-lap (n=Corydon7@pdpc/supporter/bronze/Corydon76-home) |
| 16:58.25 | *** mode/#asterisk [+o Corydon76-lap] by ChanServ |
| 16:59.44 | NetForces | Anyone knows a way to send DTMF inband on PRI before the call is actually answered? Tried D() and M() and does not work. |
| 17:00.23 | cfh | [TK]D-Fender: http://pastebin.com/m1ba50cae |
| 17:00.44 | Corydon76-lap | NetForces: that's part of the point of the protocol |
| 17:00.59 | *** join/#asterisk twisted (n=root@pdpc/supporter/active/twisted) |
| 17:00.59 | *** mode/#asterisk [+o twisted] by ChanServ |
| 17:01.04 | twisted | yay fiber outages |
| 17:01.16 | NetForces | Cory, not sure what you mean... |
| 17:01.24 | Silent-X | I want fiber =( |
| 17:01.37 | Corydon76-lap | NetForces: you are not permitted to send ANY audio on a PRI until the channel is answered |
| 17:01.47 | *** join/#asterisk qdk (n=qdk@0x573fe57e.bynqu2.broadband.tele.dk) |
| 17:02.27 | Corydon76-lap | NetForces: otherwise, people could have conversations on a PRI without ever being answered (and hence never being billed) |
| 17:02.35 | NetForces | Hmmm. The setup is multiple PRI with allstream. Manual dial you hear a beeeeep and you punch-in your account code (4 digits) and the call gets through. Trying to automize with D() or M() and does not go through. |
| 17:03.51 | *** join/#asterisk tuxfoo (n=tuxfoo@pool-72-65-149-192.chrlwv.east.verizon.net) |
| 17:04.02 | scruz | if you've used asteriskwin and asterisk, could you please compare both? i want to learn asterisk, but i can't sacrifice a computer for linux. the vmware option is good. now if my internet connection would only agree to my wishes... |
| 17:04.30 | BBHoss | scruz, sacrafice? i would say upgrade |
| 17:04.45 | tuxfoo | hahah - Winders |
| 17:04.46 | skirmisha | guys still get no authority found |
| 17:04.50 | skirmisha | where can the problem be |
| 17:04.53 | Silent-X | I havent used asteriskwin, but id have to say asterisk > asteriskwin |
| 17:06.03 | *** join/#asterisk b1shop (n=b1shop@dsl081-149-253.chi1.dsl.speakeasy.net) |
| 17:06.18 | scruz | oh no, they caught up with me! |
| 17:06.36 | *** join/#asterisk outtolunc (n=me@adsl-66-218-53-172.dslextreme.com) |
| 17:06.39 | Silent-X | he knows our plan, we must act quickly |
| 17:07.33 | b11d | ok.. I just had to cold restart asterisk.. I guess updating the timezone while asterisk was running isnt good enough. |
| 17:08.22 | *** join/#asterisk aikanaro79 (n=noone@89-180-11-208.net.novis.pt) |
| 17:08.29 | tzafrir | b11d, why? |
| 17:08.31 | aikanaro79 | hi people |
| 17:08.40 | *** join/#asterisk CunningPike (n=arodgers@204.239.12.183) |
| 17:08.52 | b11d | dont know.. voicemail would just NOT stop timestamping voicemail as UTC despite that my system was set to CST. |
| 17:08.53 | Silent-X | howd aikanaro79 |
| 17:09.04 | tzafrir | Changing the time zone doesn't change the clock |
| 17:09.06 | drako | seg fault with no info |
| 17:09.06 | b11d | but I had started asterisk while it was UTC.. and then I moved to CST, but asterisk didnt catch the change. |
| 17:09.08 | cfh | [TK]D-Fender: what can i do ? |
| 17:09.30 | b11d | i did a pbx_config reload and an entire 'modules reload' to no avail.. |
| 17:09.34 | b11d | had to stop asterisk, and restart it. |
| 17:09.44 | tzafrir | b11d, you set timezone through TZ or through /etc/localtime ? |
| 17:09.47 | aikanaro79 | in my dialplan, if I define an extension like exten=>00_X!, ... it means that any number dialed that starts with 2 zeros and has 1 or more digits will do as this extension bids right? |
| 17:09.49 | b11d | tzsetup |
| 17:10.04 | b11d | which sets localtime |
| 17:10.51 | tzafrir | aikanaro79, a '_' marks an extension as "special" (one where "X", "!" and such are meaningful), |
| 17:10.59 | tzafrir | but only if it is the first character |
| 17:11.00 | *** join/#asterisk stkn|work (n=stkn|wor@gentoo/developer/pdpc.active.stkn) |
| 17:11.06 | *** join/#asterisk saint_ (n=saint@c-69-242-118-124.hsd1.nj.comcast.net) |
| 17:11.15 | [TK]D-Fender | cfh: You did not follow the instructions I jsut gave you, nor did you show me their state was other than listed. |
| 17:11.23 | Kobaz | so what's the best place to get sangoma cards |
| 17:11.33 | tzafrir | aikanaro79, that pattern of your only matches to the explicit "extension" 00_X! |
| 17:11.45 | aikanaro79 | ok |
| 17:11.47 | aikanaro79 | thx |
| 17:12.00 | tzafrir | Use: _00X!, for starters |
| 17:12.09 | [TK]D-Fender | aikanaro79: exten => _00., |
| 17:12.13 | [TK]D-Fender | ^^ |
| 17:12.14 | aikanaro79 | got it |
| 17:12.15 | aikanaro79 | thanks |
| 17:12.17 | aikanaro79 | :) |
| 17:13.50 | cfh | <PROTECTED> |
| 17:13.54 | cfh | or not ? |
| 17:13.57 | [TK]D-Fender | no |
| 17:16.00 | *** part/#asterisk scruz (n=Dell_ope@41.220.73.170) |
| 17:17.23 | NetForces | Sound something interesting in regards to my DTMF and the D() flag... My account code is 0098. If I use Dial(Zap/g0/15144441212,300,D(098)) and when I hear the beeeep I press 0, Asterisk will then execute the D() flag and send 098... |
| 17:18.03 | b11d | i love you guys |
| 17:18.19 | drmessano | back at ya |
| 17:18.25 | b11d | thanks bud |
| 17:18.26 | b11d | :) |
| 17:19.13 | NetForces | any ideas? |
| 17:19.14 | drmessano | oh no.. nobody gets to leave |
| 17:19.53 | drmessano | Ok, time to head to the office.. bah.. |
| 17:21.46 | *** join/#asterisk andresmujica (n=andresmu@correo.seaq.com.co) |
| 17:22.25 | andresmujica | Hi |
| 17:22.31 | b11d | HI@!@!!@!@! |
| 17:22.40 | b11d | whats up andresmujica? |
| 17:22.59 | andresmujica | anyone knows if i can use BLADE servers for asterisk? |
| 17:23.13 | SteveTotaro | what does this indicate? "Retransmitting #4 (no NAT) to 195.123.123.123:5060" |
| 17:23.42 | andresmujica | hmm resending voice control packages |
| 17:23.47 | andresmujica | in SIP |
| 17:23.57 | SteveTotaro | does retranmitting mean there was no reply? |
| 17:24.09 | [TK]D-Fender | NetForces: "D()" does not activate when you push something, it triggers immediately upon answer. |
| 17:27.59 | *** join/#asterisk adjohn (n=adjohn@p5182-ipad71marunouchi.tokyo.ocn.ne.jp) |
| 17:28.03 | andresmujica | not necessarily |
| 17:28.17 | andresmujica | as it's UDP normally there are 3 retransmissions |
| 17:28.40 | andresmujica | there's no ACK at transport level. so the system send multiple times the packets |
| 17:28.42 | *** join/#asterisk jpeeler (n=jpeeler@216.207.245.1) |
| 17:29.04 | andresmujica | the problem is when excessuive retransmission are sent |
| 17:29.07 | hmmhesays | SteveTotaro, that means didn't get any response to your sip invite |
| 17:29.27 | hmmhesays | if memory serves from chan_sip.c it sends 6 |
| 17:30.07 | *** part/#asterisk cfh (n=luca@host196-137-static.90-82-b.business.telecomitalia.it) |
| 17:30.34 | *** join/#asterisk nirz (n=nir@tony09-113-34.inter.net.il) |
| 17:30.44 | NetForces | TKD: Well, in my case it does not... |
| 17:31.18 | NetForces | If I just use the Dial(Zap/g0/15144441212,300,D(098)) it will just stay there then the allstream lady will tell me that it can not complete the call |
| 17:31.30 | hmmhesays | andresmujica, you're right but not about SteveTotaro's error message |
| 17:31.37 | NetForces | If I use again Dial(Zap/g0/15144441212,300,D(098)) and when I hear the beep, I press 0, asterisk sends the 098 |
| 17:32.00 | [TK]D-Fender | NetForces: What are you calling out on? |
| 17:32.06 | NetForces | which ompletes the 0098 account code and the call goes through |
| 17:32.12 | *** join/#asterisk JenniferAkemi (n=akemi@206-248-153-17.dsl.teksavvy.com) |
| 17:32.13 | NetForces | Allstreal PRI |
| 17:32.18 | NetForces | Allstream sorry |
| 17:33.15 | hmmhesays | I use dynamic feature map for that |
| 17:34.25 | NetForces | TK: As soon as I press 0 or any other digit, I see "Sending DTMF '098' to the called party." |
| 17:34.53 | *** join/#asterisk anonymouz666 (n=anonymou@201.19.127.250) |
| 17:35.32 | *** join/#asterisk BadHorsie (n=sebas@201.198.239.167) |
| 17:37.24 | *** join/#asterisk teknoprep (n=Chris@unaffiliated/teknoprep) |
| 17:38.12 | teknoprep | now EVERY time i setup asterisk i use nat with sip channels. i setup the sip_nat.conf file properly with it included in my sip.conf... but here i am getting one-way audio... is there a way to check that asterisk is correctly doing sip nat? |
| 17:39.03 | andresmujica | upps sorry. |
| 17:39.23 | teknoprep | this is the first time i have ever had a problem |
| 17:39.35 | andresmujica | i've sent those messages but the sip com was ok (a litle slow at the beginning) |
| 17:39.38 | teknoprep | i have all my settings done setup the same way as every other time i set it up |
| 17:39.58 | *** join/#asterisk CrazyTux[m] (n=CrazyTux@ppp-70-248-182-215.dsl.hstntx.swbell.net) |
| 17:42.56 | hmmhesays | teknoprep, rtp debug mang |
| 17:43.06 | teknoprep | ok |
| 17:43.24 | NetForces | TK: Any ideas? Looking at the app_dial code, the D() is invoke only is res ir true |
| 17:43.42 | ManxPower | NetForces: DTMF inside Macro or Gosub inside a Dial doesn't work well. |
| 17:43.58 | ManxPower | In fact I never got it to recognize anything but a single DTMF. |
| 17:44.15 | NetForces | ok, but D() should work well no? |
| 17:44.31 | ManxPower | NetForces: D() is Gosub? |
| 17:44.49 | ManxPower | A Gosub and a macro are almost the same thing, BTW. |
| 17:44.51 | teknoprep | [Feb 11 12:44:40] Sent RTP packet to 209.244.42.253:64294 (type 00, seq 029912, ts 210816, len 000160) |
| 17:44.51 | teknoprep | [Feb 11 12:44:40] Got RTP packet from 192.168.15.253:30274 (type 00, seq 001711, ts 210976, len 000160) |
| 17:44.56 | teknoprep | i get that over and over |
| 17:44.56 | NetForces | Not even... I have en "extension" that simply does a Dial wth the D(). |
| 17:45.22 | ManxPower | NetForces: Then it should be sending the DTMF as soon as the call is answered. |
| 17:45.26 | NetForces | I tries with the AMI and same thing. |
| 17:45.43 | ManxPower | Of course ANALOG FXO ports are considered ANSWERED as soon as dialing is done. One of the reasons ANALOG FXOs suckl |
| 17:45.49 | ManxPower | and suck too |
| 17:46.09 | NetForces | I tries on other circuits and that is what it does, but not on this perticular case. It need me to press a key before sending the DTMF flow |
| 17:46.25 | ManxPower | NetForces: paste JUST your Dial(... line |
| 17:46.27 | *** join/#asterisk ferai (n=jefferai@amarok/developer/mitchell) |
| 17:46.56 | NetForces | Dial(Zap/119/15147124064,300,D(098)) |
| 17:47.27 | ManxPower | What type of port is 119? |
| 17:47.31 | teknoprep | hmmhesays, i am not getting ANY rtp packets from bandwidth.com |
| 17:47.41 | teknoprep | hmmhesays, which makes no sense |
| 17:47.48 | ManxPower | teknoprep: sounds like a NAT or firewall problem to me. |
| 17:47.49 | NetForces | last channel of the 5th PRI |
| 17:48.02 | ManxPower | NetForces: I have no idea what would cause that to happen |
| 17:48.07 | hmmhesays | sure does, look at the sip transaction messages |
| 17:48.18 | teknoprep | ManxPower, i have setup NAT many times. i have everything setup correctly in m0n0wall tho |
| 17:48.37 | *** join/#asterisk Fah (i=cynic@paranoia.neverlight.com) |
| 17:48.38 | NetForces | I think I'll open a bug.. |
| 17:48.45 | jameswf | ****YOU REALLY SHOULD READ THIS**** |
| 17:49.02 | ManxPower | jameswf: OK, I read it, now what? |
| 17:49.04 | teknoprep | you know whats really wierd... is VoicePulse works over SIP.. but bandwidth.com doesn't |
| 17:49.18 | teknoprep | also my remote extension on my laptop works over sip.. but bandwidth.com doesn't |
| 17:49.24 | *** part/#asterisk tuxfoo (n=tuxfoo@pool-72-65-149-192.chrlwv.east.verizon.net) |
| 17:49.28 | teknoprep | so i know the firewall is not blocking the ports |
| 17:49.36 | teknoprep | or i wouldn't be able to use my laptop or voicepulse |
| 17:49.37 | jameswf | i dunno but thant like the coolest thing ever |
| 17:49.43 | jameswf | I am easily ammused |
| 17:50.05 | jameswf | be like stuffs not loading? look at your logs anything jump out at you |
| 17:52.04 | drako | *** glibc detected *** corrupted double-linked list: 0x08212e08 *** |
| 17:52.04 | drako | Abort |
| 17:52.39 | drako | and it crashed... |
| 17:53.33 | *** join/#asterisk murdmath (n=vircuser@mail.kimballequipment.com) |
| 17:53.49 | ManxPower | drako: Are all updates for your OS installed? |
| 17:54.20 | bkruse | Hey, what performance testing tools do you guys use? |
| 17:54.26 | bkruse | sipp? Sip bomber? etc etc |
| 17:54.40 | [TK]D-Fender | NetForces: Sounds like the remote side might not be answering the call until DTMF is received due to early audio. |
| 17:54.40 | jameswf | viagra |
| 17:54.55 | tzafrir | bkruse, another asterisk. Or a few other |
| 17:54.57 | tzafrir | s |
| 17:55.13 | bkruse | jameswf: I am sorry you suck so bad to have to use it :[ |
| 17:55.30 | bkruse | tzafrir: Yes, I already have that in the mix, just wondering if there is something out that will spit out results or some sort |
| 17:55.39 | Tebi | vgsm me no debug |
| 17:55.53 | jameswf | my wife isnt... I dont care who you are you cant go 4 hours un assisted.... |
| 17:55.55 | Fah | drako: I saw a very similar probem that happened when a bug in libc around the resolver libraries was getting triggered under load. |
| 17:56.15 | Fah | unfortunately I forget the exact version, this happened a while back |
| 17:56.58 | bkruse | jameswf: 4 hours? |
| 17:57.23 | bkruse | You are doing something wrong my friend |
| 17:57.52 | bkruse | But this is not the room for that, you can go to #sex-help if need be |
| 17:57.58 | BBHoss | ~sex |
| 17:57.59 | jbot | updatedb; locate; talk; date; cd; strip; look; touch; finger; unzip; uptime; gawk; head; apt-get install condom; mount; fsck; gasp; more; yes; yes; yes; more; umount; apt-get remove --purge condom; make clean; sleep, or super extractor, http://sf.net/projects/sex/ |
| 17:58.10 | jameswf | ~~~~ |
| 17:58.11 | jbot | ARGH!!! STOP IT jameswf!!! |
| 17:58.21 | jameswf | ~~ |
| 17:58.22 | jbot | Every moment in which I'm called upon is torture. |
| 17:58.46 | jameswf | ~viagra |
| 17:58.47 | jbot | i guess viagra is the nickname for the Woody Tech Support Crew |
| 18:01.00 | jameswf | ~wood |
| 18:01.01 | jbot | how much wood could a woodchuck chuck if a woodchuck could chuck wood? |
| 18:01.01 | Silent-X | oO |
| 18:01.12 | jameswf | ~dude |
| 18:01.13 | jbot | Be most excellent to each other! |
| 18:01.28 | jameswf | jbot: whats mine say |
| 18:01.35 | jameswf | jbot: what's mine say |
| 18:01.35 | jbot | dude! ... What's Mine Say? |
| 18:01.36 | bkruse | jameswf: /msg jbot and have all the fun you want |
| 18:02.10 | *** join/#asterisk teknoprep (n=Chris@unaffiliated/teknoprep) |
| 18:02.21 | teknoprep | is there a way to make sure that my sip_nat settings are working properly? |
| 18:02.38 | teknoprep | i have my externip= setup and localnet= |
| 18:02.58 | teknoprep | now is there a cli command to see if my nat settings are working for sip? |
| 18:04.54 | *** mode/#asterisk [-v jameswf] by bkruse |
| 18:05.14 | *** mode/#asterisk [+v jameswf] by bkruse |
| 18:05.22 | *** join/#asterisk mjoyce (i=tbl@hick.org) |
| 18:06.24 | jameswf | bkruse: you have * running on openmoko |
| 18:06.26 | tzafrir | he didn't run fast enough, I guess |
| 18:07.10 | *** join/#asterisk worgil (n=worgil@88.231.34.68) |
| 18:07.10 | bkruse | jameswf: Yes |
| 18:07.16 | patrick-- | hey there. im trying to get my FXO channel up but i get a : ZT_CHANCONFIG failed on channel 1: No such device or address (6) |
| 18:07.18 | bkruse | jameswf: and an iaxclient |
| 18:07.19 | patrick-- | any idea? |
| 18:07.21 | bkruse | jameswf: just POC |
| 18:07.32 | bkruse | patrick--: zttool; do you see your card? |
| 18:07.33 | *** join/#asterisk shtoom (n=godson@59.93.118.77) |
| 18:07.44 | tzafrir | patrick--, do you have anything in /proc/zaptel/* ? |
| 18:07.57 | patrick-- | empty |
| 18:08.05 | patrick-- | 01:00.0 ISDN controller: Cologne Chip Designs GmbH ISDN network Controller [HFC-8S] (rev 01) |
| 18:08.09 | patrick-- | 01:01.0 ISDN controller: Cologne Chip Designs GmbH ISDN network Controller [HFC-4S] (rev 01) |
| 18:08.12 | patrick-- | they are there tho ... |
| 18:08.13 | tzafrir | so zaptel is loaded, but the module for the card isn't |
| 18:08.24 | bkruse | tzafrir: correct |
| 18:08.44 | tzafrir | patrick--, two bero.net cards? |
| 18:08.46 | patrick-- | im pretty new to the whole telephony thing... which module is it? |
| 18:08.49 | patrick-- | tzafrir: correct |
| 18:09.00 | patrick-- | BN8s0 and BN4s0 |
| 18:09.03 | tzafrir | shouldn't be relevant |
| 18:09.09 | patrick-- | right |
| 18:09.20 | tzafrir | hmm.... do you have zaptel hardware? |
| 18:09.40 | *** join/#asterisk drmessano-LT (n=nonya@207.230.140.240) |
| 18:10.52 | *** join/#asterisk RypPn_OuT (i=TuMbL@rosscom.demon.co.uk) |
| 18:10.59 | *** join/#asterisk angom (n=angom@200.56.104.87.dsl.dyn.telnor.net) |
| 18:13.33 | BBHoss | they got asterisk to compile on openmoko |
| 18:14.29 | jameswf | other than the coolness factor what would be the point of asterisk on a cell phone |
| 18:14.45 | BBHoss | evidentally bkruse is the one doing it :) |
| 18:15.00 | bkruse | BBHoss: Its done |
| 18:15.11 | bkruse | you can checkout the source from the openmoko repo, or download the ipkg from bkruse.com |
| 18:15.18 | bkruse | no sounds included, but its easy to add them (scp) |
| 18:15.25 | bkruse | since the package bloats a lot with them |
| 18:15.28 | BBHoss | do they have a CDMA version of moko yet? |
| 18:15.34 | bkruse | no no |
| 18:15.45 | bkruse | The next version is GSM, as it should be, but its more iphone like |
| 18:15.58 | bkruse | (not the GTA02 but whole nother phone, dont think freerunner/neo1973) |
| 18:16.08 | bkruse | CDMA will come when/if they sign a contract with someone, most likely |
| 18:16.19 | BBHoss | yeah thats too bad, i have verizon |
| 18:16.32 | bkruse | ya, it is to bad you have verizon :P |
| 18:16.44 | bkruse | I have one though, its a pretty neat little phone, wouldnt use it for everyday, YET |
| 18:17.12 | BBHoss | i heard verizon is transitioning to gsm within the next couple of years, who knows? |
| 18:18.03 | BBHoss | i hate cdma |
| 18:18.30 | jameswf | SOLD OUT! Arhg!! |
| 18:19.00 | NetForces | TK: Any idea of a fix ? |
| 18:19.10 | NetForces | Sorry I was out for lunch |
| 18:19.53 | badcfe | i have two asterisks A and B and observe something for SIP only calls going thru them: when a phone calls A its sent forward to asterisk B -- now, the A reinvites the phone and B, all as configured. but sometimes i see a 491 Request pending from B to A. Actually A retransmits the re-INVITE to B even if B ansers 200 OK to it. So finally B says 491 Request pending. Why does this happen? Is it normal operation when an asterisks re-INVITE |
| 18:20.03 | *** join/#asterisk erago (n=erago@236.Red-81-39-224.dynamicIP.rima-tde.net) |
| 18:20.17 | [TK]D-Fender | NetForces: nope |
| 18:20.58 | patrick-- | Hey, i keep getting this error: http://phpfi.com/295843 can someone help? |
| 18:21.45 | *** join/#asterisk clyrrad (n=darryl@CPE000802212b48-CM0011aea484a4.cpe.net.cable.rogers.com) |
| 18:21.45 | NetForces | TK: Found this bug that might be related: http://bugs.digium.com/view.php?id=5266 |
| 18:23.05 | *** join/#asterisk anonymouz666 (n=anonymou@201.19.127.250) |
| 18:24.13 | clyrrad | I have a variable that I need to CUT the last 'zero' off the end. I have NoOp( ${CUT(myVar,0,11)} ) - which as I read should CUT everything before the 11th zero. I need to cut this string so that its exactly 10 characters. Can anyone help? |
| 18:25.03 | BBHoss | i can't image 100Mbits downstream from a cell connection |
| 18:25.07 | BBHoss | imagine |
| 18:25.10 | [TK]D-Fender | clyrrad: Cut from which end? |
| 18:25.22 | clyrrad | [TK]D-Fender: the right side |
| 18:25.32 | clyrrad | need to strip off the last zero |
| 18:25.37 | [TK]D-Fender | clyrrad: ${MyVar:0:11} |
| 18:25.39 | clyrrad | on the end of the string |
| 18:25.53 | [TK]D-Fender | clyrrad: Time to re-read "Variables 101" |
| 18:27.31 | clyrrad | Oh duh |
| 18:27.35 | clyrrad | i knew that too |
| 18:27.48 | clyrrad | but actually it hsould be ${MyVar:0:10} :D |
| 18:27.53 | clyrrad | thanks [TK]D-Fender |
| 18:29.38 | jameswf | the world needs an IAX client for blackerry |
| 18:29.46 | hmm-home | heh |
| 18:29.47 | hmm-home | why |
| 18:30.04 | hmm-home | give me a sip client for blackberry over iax |
| 18:30.28 | *** join/#asterisk hi365 (n=hi365@mail.pcgeula.co.il) |
| 18:31.50 | patrick-- | http://phpfi.com/295846 |
| 18:31.53 | patrick-- | any idea on this? |
| 18:32.14 | [TK]D-Fender | patrick--: No more than when you asked 10 minutes ago. |
| 18:32.35 | patrick-- | right |
| 18:33.33 | _ShrikE | patrick--: do you have any timing interfaces? |
| 18:33.51 | patrick-- | im not sure |
| 18:33.53 | patrick-- | i guess not |
| 18:34.05 | _ShrikE | if you dont have a zaptel card then load ztdummy |
| 18:34.13 | patrick-- | i want to use asterisk with 2 bero.net cards + misdn |
| 18:34.27 | *** join/#asterisk guillote_GNU (n=guillote@host157.201-253-55.telecom.net.ar) |
| 18:41.29 | *** join/#asterisk nitrus^ (n=nitrus@cpe-76-166-248-27.socal.res.rr.com) |
| 18:41.49 | nitrus^ | anyone know what would prevent audio on a sip<->bridge? |
| 18:42.02 | nitrus^ | the zap channel picks up but neither side can be heard |
| 18:42.14 | nitrus^ | this problem began after going from * 1.2 to 1.4 |
| 18:45.03 | *** join/#asterisk anonymouz666 (n=anonymou@201.19.127.250) |
| 18:45.41 | *** join/#asterisk denon (n=denon@tooth.decay.org) |
| 18:45.41 | *** mode/#asterisk [+o denon] by ChanServ |
| 18:47.33 | patrick-- | [Feb 11 15:58:42] WARNING[31806] chan_iax2.c: Unable to open IAX timing interface: No such device or address |
| 18:47.40 | patrick-- | what do i do if i dont have a timing interface? |
| 18:48.17 | pkunkra | might be looking for the ztdummy device. |
| 18:48.44 | tzafrir | this is needed for trunk mode (or whatever this is called) of iax2, right? |
| 18:48.47 | pkunkra | as far as i recall, some modules use the zaptel drivers for timing. |
| 18:48.51 | *** join/#asterisk drako (n=thinkpad@nelug/coreteam/luisjose) |
| 18:49.12 | [TK]D-Fender | patrick--: set up ZTDUMMY and recompile * accordingly. |
| 18:49.12 | drako | ManxPower, yes, its debian etch and is up to date |
| 18:49.48 | patrick-- | [TK]D-Fender: i dont think i need IAX really |
| 18:50.06 | patrick-- | i only use sip |
| 18:50.35 | [TK]D-Fender | patrick--: Nobody said you did. You jsut asked what we'd do about the message you showed us. Well what'd we'd do is satisfy its requirements. You set it up that way. Typically you try to make it work, not just find some other way around. |
| 18:50.43 | tzafrir | unload => chan_iax.so ; in modules.conf ... |
| 18:51.19 | [TK]D-Fender | patrick--: And you'd only get that warning if you tried setting up a trunked connection int he first place. |
| 18:51.37 | *** join/#asterisk timeshell (n=Khoja@gw.lusi.on.ca) |
| 18:51.50 | patrick-- | im really sorry. im proper beginner when it comes to telephony/asterisk. i just want to get this machine to work. |
| 18:52.09 | [TK]D-Fender | patrick--: Then go install zaptel & ZTDUMMY like you're supposed to. |
| 18:52.34 | patrick-- | do i need zaptel? |
| 18:52.50 | patrick-- | thought it'd work with asterisk + misdn |
| 18:53.44 | ManxPower | You need zaptel if you want MeetMe or IAX2 TRUNKING (trunking is just a way to stuff more calls into the same bandwidth) |
| 18:53.46 | [TK]D-Fender | patrick--: that has nothing to do with the IAX TRUNK you are defining. |
| 18:54.02 | ManxPower | You don't NEED trunking most of the time. |
| 18:54.21 | patrick-- | [TK]D-Fender: lets start over |
| 18:54.27 | patrick-- | i went by the asterisk book |
| 18:54.31 | [TK]D-Fender | patrick--: AKA "Stop talking about your radio when you can see your rear differential 50m back on the road in your rear-view mirror" |
| 18:54.31 | patrick-- | Ive got: |
| 18:55.36 | patrick-- | Beronet BN8S0 + BN4S0 |
| 18:55.41 | patrick-- | what do you suggest me to start with? |
| 18:56.13 | patrick-- | i dont want trouble. im just seeking some help. |
| 18:56.49 | [TK]D-Fender | patrick--: You asked about a very specific warning message. do you CARE about IAX at all? |
| 18:56.59 | ManxPower | patrick--: ignore the message if you are not using iax |
| 18:57.15 | patrick-- | ManxPower: asterisk will not start |
| 18:57.32 | [TK]D-Fender | patrick--: then i highly doubt that message has anything to do with it. |
| 18:57.37 | ManxPower | patrick--: then "mv /etc/asterisk/iax.conf /etc/asterisk/iax.conf.disabled" |
| 18:57.48 | [TK]D-Fender | patrick--: pastebin the ENTIRE startup process when you do it manually |
| 18:57.50 | [TK]D-Fender | ~pb |
| 18:57.51 | jbot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste |
| 18:58.21 | *** join/#asterisk sx|lappy (n=sxpert@abo-180-6-68.ech.modulonet.fr) |
| 18:59.21 | patrick-- | http://phpfi.com/295850 |
| 18:59.28 | patrick-- | im aware of that.. |
| 19:00.21 | shtoom | Hi I am getting WARNING[16108]: callerid.c:217 callerid_get_dtmf: Couldn't detect start-character. CID parsing might be unreliable |
| 19:00.21 | shtoom | that warning on console and first 3 digits of CID are missing , I am using cidsignalling=dtmf and cidstart=ring in zapata.conf |
| 19:00.31 | [TK]D-Fender | patrick--: Right now I'm thinking your misdn init is crashing. add "noload => chan_misdn.so" to modules.conf and try again. if it loads, thats your problem. |
| 19:01.01 | *** join/#asterisk angryuser[A] (i=nononon@df01t2-212-194-216-180.d4.club-internet.fr) |
| 19:01.02 | [TK]D-Fender | shtoom: What are you connecting to? |
| 19:01.19 | patrick-- | [TK]D-Fender: it runs... |
| 19:01.28 | patrick-- | how could that crash?! :O |
| 19:01.41 | shtoom | <PROTECTED> |
| 19:01.45 | [TK]D-Fender | patrick--: if the module fails, it takes out *, jsut like Zaptel issues do. |
| 19:01.52 | patrick-- | okay |
| 19:01.55 | [TK]D-Fender | shtoom: Located in what country? |
| 19:02.06 | shtoom | <PROTECTED> |
| 19:02.10 | [TK]D-Fender | patrick--: Good, we can now ignore your IAX warning. |
| 19:02.19 | [TK]D-Fender | shtoom: Ok, and DTMF is normal there for CID? |
| 19:02.49 | patrick-- | [TK]D-Fender: i really appreciate your help |
| 19:02.54 | patrick-- | So how do i proceed? |
| 19:03.13 | shtoom | <PROTECTED> |
| 19:03.45 | shtoom | <PROTECTED> |
| 19:03.46 | [TK]D-Fender | patrick--: Go through whatever CLI means you have to verify that your card's drivers are correctly configured and running and that everything leading up to * is OK. Then look at your * configs for it to make sure they seem right |
| 19:04.27 | [TK]D-Fender | shtoom: If you have "echotraining" enabled on a zaptel interface, that might be involved. disable that if it isn't already. |
| 19:05.20 | *** join/#asterisk JenniferAkemi (n=akemi@206-248-153-17.dsl.teksavvy.com) |
| 19:05.51 | patrick-- | [TK]D-Fender: it seems all okay |
| 19:05.58 | patrick-- | do you think i should start over with asterisk? |
| 19:06.09 | [TK]D-Fender | patrick--: What do you mean "start over"? |
| 19:06.18 | patrick-- | uninstall and reinstall? |
| 19:06.30 | patrick-- | cause i do think i went wrong on some interceptions |
| 19:06.42 | [TK]D-Fender | patrick--: No, you should investigate your * configs next if everything outside of * checks out OK. |
| 19:06.54 | patrick-- | outside looks okay |
| 19:07.04 | shtoom | <PROTECTED> |
| 19:07.54 | *** join/#asterisk ShadowHntr (n=sentinel@wikipedia/Shadowhntr) |
| 19:08.00 | [TK]D-Fender | shtoom: You can do a sanity check and try disabling EC just to see if CID starts working. Then you might want to try and verify for certain exactly what is being applied to that line. |
| 19:08.27 | *** join/#asterisk ph0ne (n=ph0ne@dsl-207-112-91-102.tor.primus.ca) |
| 19:08.58 | ShadowHntr | got a question. i've read about the UNISTIM module for Asterisk - anyone want to give their feedback on using it with Nortel IP phones? |
| 19:09.37 | patrick-- | [TK]D-Fender: i just think im missing sth. out... the configs look okay |
| 19:09.45 | patrick-- | what are the essential configs for channel configuration? |
| 19:10.45 | shtoom | <PROTECTED> |
| 19:10.56 | *** join/#asterisk kyron (n=kyron@211-217-static-ppp.3menatwork.com) |
| 19:11.51 | [TK]D-Fender | shtoom: Ok, verify the line. |
| 19:11.56 | *** join/#asterisk Greek-Boy (n=email@41.221.58.4) |
| 19:12.03 | [TK]D-Fender | ShadowHntr: Do you have them already? |
| 19:12.30 | [TK]D-Fender | patrick--: You're going to ahve to go read some guides for this... |
| 19:12.33 | *** part/#asterisk Fah (i=cynic@paranoia.neverlight.com) |
| 19:12.41 | Greek-Boy | which dialing pattern(s) would cover all international calls? |
| 19:12.44 | ShadowHntr | [TK]D-Fender: no. just researching for an eventual rollout in a home office environment. |
| 19:12.49 | shtoom | <PROTECTED> |
| 19:12.55 | patrick-- | [TK]D-Fender: i would absolutely love a great guide on beronet cards with misdn |
| 19:12.58 | patrick-- | you got any? |
| 19:13.02 | [TK]D-Fender | ShadowHntr: then forget about UNISTIM if you know whts good for you |
| 19:13.08 | shtoom | I am miles apart from the server |
| 19:13.09 | ShadowHntr | it seems like the Nortel phones are at a good price point, and I've used them in an office before. quality phone construction. |
| 19:13.10 | [TK]D-Fender | patrick--: www.google.com |
| 19:13.20 | [TK]D-Fender | ShadowHntr: Complete waste of time. |
| 19:16.47 | *** join/#asterisk anonymouz666 (n=anonymou@201.19.127.250) |
| 19:16.47 | scooby2 | shtoom: I thought sangoma was the devil until I got a Digium card. Now I really want the sangoma back! |
| 19:16.47 | shtoom | <PROTECTED> |
| 19:16.47 | mjoyce | why did think sangoma was the devil? |
| 19:16.47 | scooby2 | lots of echo issues |
| 19:16.47 | mjoyce | did you have hardware echocan? |
| 19:16.48 | scooby2 | plus the voodoo involved getting them working |
| 19:16.48 | patrick-- | [TK]D-Fender: is there a way to completely remove asterisk? |
| 19:16.48 | shtoom | scooby2:I didn't faced any major problems with sangoma |
| 19:16.48 | scooby2 | mjoyce: supposedly hardware echo cancellation but it liked to turn itself off all the time |
| 19:17.47 | scooby2 | sangoma a103 from 2003 |
| 19:17.47 | nitrus^ | anyone know what tone intercom systems play to get the employees attention at say best buy or something? i have a PA system connected to one of my zap channels and i'd like it to play a tone before the speaker can begin. I was planning on just using a follow me with a remote announce that will play the tone, i just dont know what file to use or where to find the sound. |
| 19:17.47 | mjoyce | 103? |
| 19:17.47 | mjoyce | 3 port digital card? |
| 19:17.47 | mjoyce | never heard of it |
| 19:17.47 | scooby2 | 2 port |
| 19:17.58 | mjoyce | then a102? |
| 19:18.07 | scooby2 | nope, have it in my hand |
| 19:18.09 | *** join/#asterisk Docfxit (n=none@cpe-72-129-118-135.socal.res.rr.com) |
| 19:18.09 | scooby2 | a103 |
| 19:18.14 | scooby2 | rev a |
| 19:18.14 | shtoom | scooby2:how ever here my situation is like I don't have other options due to monopolistic nature of digium distributors here in India |
| 19:18.23 | mjoyce | weird, does it have the octasic chip on it? |
| 19:18.30 | mjoyce | it shoudl be like, under the little wiggily slots in it |
| 19:19.02 | [TK]D-Fender | patrick--: How does uninstalling * solve you issue? |
| 19:19.05 | scooby2 | actually thats the expansion card for the second port |
| 19:19.06 | [TK]D-Fender | your* |
| 19:19.11 | scooby2 | the card itself is a101 |
| 19:19.34 | shtoom | <PROTECTED> |
| 19:19.36 | scooby2 | no octasic chip |
| 19:19.55 | [TK]D-Fender | shtoom: Nope, sorry.... |
| 19:20.13 | patrick-- | [TK]D-Fender: i'd start all over cause im afraid i wont find the mistake |
| 19:20.14 | Greek-Boy | [TK]D-Fender do u know which dialing pattern will cover all international calls? |
| 19:21.09 | [TK]D-Fender | Greek-Boy: Depends what patterns are used wherever you are. You should already know the answer to this. |
| 19:21.26 | [TK]D-Fender | patrick--: jsut wipe your configs and reinstall. |
| 19:21.37 | *** join/#asterisk worgil (n=worgil@88.231.34.68) |
| 19:21.47 | patrick-- | make samples? |
| 19:22.02 | [TK]D-Fender | patrick--: And that is a very sad way considering we haven't even looked at your misdn config no anything supporting it. |
| 19:22.23 | [TK]D-Fender | patrick--: You can look up the sample confi right in your source folder. |
| 19:23.11 | patrick-- | would you like to have a look at it? [TK]D-Fender |
| 19:23.49 | ph0ne | hello, what would be the best book to read about SIP? |
| 19:24.19 | [TK]D-Fender | ph0ne: http://www.ietf.org/rfc/rfc3261.txt |
| 19:24.46 | ph0ne | its very technical- dry even |
| 19:25.08 | [TK]D-Fender | ph0ne: the very definition of "best". |
| 19:25.31 | [TK]D-Fender | ph0ne: Complete, from the point of origin, and irrefutable. |
| 19:25.50 | ph0ne | ok I WILL READ IT |
| 19:25.59 | ph0ne | caps lock |
| 19:26.29 | endre | cruise control for cool |
| 19:28.07 | patrick-- | [TK]D-Fender: http://phpfi.com/295861 |
| 19:30.21 | angryuser[A] | i need for * to wait when EXT is fully dialed, i do WaitExten(8) but when i press any number it stops waiting for next digits and tryed go to that extension, any help ? |
| 19:31.09 | [TK]D-Fender | angryuser[A]: pastebin is your friend. |
| 19:31.11 | [TK]D-Fender | ~pb |
| 19:31.44 | jbot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste |
| 19:31.45 | [TK]D-Fender | angryuser[A]: Show us your code and your CLI output at verbsoe 10 |
| 19:31.45 | [TK]D-Fender | verbose* |
| 19:31.45 | angryuser[A] | k |
| 19:33.11 | *** join/#asterisk hmodes (n=hmodes@2001:470:1f04:59:0:0:0:2) |
| 19:34.21 | styelz | anyone seen this before ? .. -- Got SIP response 400 "Bad Subscription-State or Content header or message body" back from 10.0.0.138 |
| 19:37.33 | patrick-- | [TK]D-Fender: is there anything wrong with my misdn.conf? |
| 19:37.45 | Greek-Boy | in my case it seems that most number dialed out for international are 14 or 15 digits. |
| 19:38.02 | *** join/#asterisk FlatFoot (n=chatzill@80.88.218.4) |
| 19:38.38 | pkunkra | what is a ideal/realistic latency to shoot for in VoIP apps? |
| 19:38.50 | pkunkra | 50ms? 100ms? 200ms? |
| 19:39.04 | Greek-Boy | i'd say 100ms total |
| 19:40.08 | bkruse | Anything over 200ms is asking for trouble, especially in an unreliable means for transfer (eg through java. hehea web interface) |
| 19:40.25 | pkunkra | ok. so no more than 100ms, and definately not more than 200ms. |
| 19:41.37 | pkunkra | i've got pings ranging from about 60ms to 300ms. i have a troublesome router on the route though so that seems to be accounting for most of the fluctuations... |
| 19:41.40 | hmmhesays | it really depends on the caller for acceptable delay |
| 19:41.58 | hmmhesays | To some people that have never had phone service at all, that 500ms satellite connection is great |
| 19:42.10 | pkunkra | hahaha |
| 19:42.31 | hmmhesays | delay really has nothing to do with the "quality" of the sound in the call |
| 19:42.33 | pkunkra | that would sound like i'm calling someone overseas |
| 19:42.51 | hmmhesays | pkunkra, my statement still holds true |
| 19:42.59 | hmmhesays | I've done enough satellite installs to know. |
| 19:43.05 | pkunkra | right, but if the network is jittery, delay makes the problem worse. |
| 19:43.15 | pkunkra | i have some slight jitter in my network. |
| 19:43.44 | hmmhesays | yes if your delta values are crazy it does |
| 19:44.02 | pkunkra | mostly because packets frequently take multiple routers, some routes are really fast. others are terribly slow. |
| 19:44.05 | hmmhesays | but a high round trip delay itself does nothing to the sound quality |
| 19:44.17 | *** join/#asterisk freezey (n=freezey@gw.mypublisher.com) |
| 19:44.25 | pkunkra | right |
| 19:44.27 | mintee | I'm getting some errors building zaptel. basically regarding headers |
| 19:44.30 | mintee | <PROTECTED> |
| 19:44.33 | file | this cogent link from JFK to ORD is terribly slow right now for example... |
| 19:44.42 | *** join/#asterisk CrazyTux[m] (n=CrazyTux@ppp-70-248-182-215.dsl.hstntx.swbell.net) |
| 19:44.53 | hmmhesays | I've had some very good sounding conversations in the 500-600ms range if you can deal with the delay |
| 19:44.55 | pkunkra | delay just means the other end hears your voice later. |
| 19:45.02 | pkunkra | quality is not degraded. |
| 19:45.22 | hmmhesays | and to the people in the grass hut in the middle of nowhere it was some kind of miracle |
| 19:45.23 | hmmhesays | lol |
| 19:45.42 | pkunkra | you've had voip service to folks in a grass hut? |
| 19:45.44 | pkunkra | wow |
| 19:45.52 | pkunkra | ;-) |
| 19:45.53 | hmmhesays | pkunkra, I kid you not |
| 19:46.03 | pkunkra | you're serious? |
| 19:46.13 | hmmhesays | bamboo villages, yup |
| 19:46.14 | angryuser[A] | http://www.pastebin.ca/900245 <[TK]D-Fender> |
| 19:46.22 | pkunkra | wow |
| 19:46.25 | pkunkra | just wow. |
| 19:46.31 | pkunkra | where was this? what country? |
| 19:46.38 | hmmhesays | all through out africa |
| 19:46.54 | hmmhesays | somalia, kenya, nigeria (rural) |
| 19:46.59 | pkunkra | what kinda phones did you give them? |
| 19:47.22 | mintee | has anyone else had problems building zaptel 1.4.8 on a 2.6.24 kernel ? |
| 19:47.22 | hmmhesays | usually quintum multiport fxs |
| 19:47.23 | pkunkra | i assume not any cisco equipment. |
| 19:47.25 | *** part/#asterisk lirakis (i=lirakis@66.252.24.133) |
| 19:47.33 | pkunkra | oh |
| 19:47.41 | hmmhesays | that wasn't the bulk of what I did. But it happened here and there |
| 19:47.43 | pkunkra | regular handsets with an FXS card |
| 19:47.48 | *** join/#asterisk lirakis (i=lirakis@66.252.24.133) |
| 19:47.51 | pkunkra | ok. that makes sense. |
| 19:48.02 | pkunkra | i though there was a real sip phone sitting in their hut. |
| 19:48.06 | angryuser[A] | <[TK]D-Fender> no cli output coz nothing interesting there, it works fine but application WaitExten(8) not waiting till i finish the number. i am trying to make work call-in-out like this pstn/mobile>>>asterisk ivr>>>sip provider |
| 19:48.14 | pkunkra | but its just a regular phone |
| 19:48.22 | hmmhesays | a lot of them at that time where h.323 |
| 19:48.40 | pkunkra | the backend is asterisk based, locked away in a safe telecome closet. |
| 19:48.50 | hmmhesays | I've done some strange installs |
| 19:49.01 | hmmhesays | especially in countries where telecom is gov't regulated |
| 19:49.17 | pkunkra | what's the strangest yet? |
| 19:49.45 | hmmhesays | when you have a guy tell you the gateway you're configuring is locked in a closet in his basement... |
| 19:50.06 | pkunkra | what's so strange about that? |
| 19:50.18 | hmmhesays | and that we can only accept traffic certain times a day otherwise the gov't will become suspicous |
| 19:50.36 | pkunkra | oh |
| 19:50.46 | pkunkra | yeah... smells fishy. |
| 19:51.36 | *** join/#asterisk ZPertee (n=ZPertee@189.sub-75-218-244.myvzw.com) |
| 19:51.39 | hmmhesays | I got out of that biz a couple years ago though, too much of a pita |
| 19:52.14 | [TK]D-Fender | angryuser[A]: is that literally a paste right out of your extensions.conf? |
| 19:52.25 | lunaphyte_ | i had another window partially obscuring this one, and all i saw was ... closet in his basement / ...will become suspicious. |
| 19:52.34 | jameswf | THe us has alot of laws for unregulated telecom |
| 19:52.39 | ZPertee | how can I do a gotoif statement based on which zap channel the call comes in on? |
| 19:53.00 | angryuser[A] | <[TK]D-Fender> no forget about => ;) dont worry they exist ;) |
| 19:53.35 | [TK]D-Fender | ZPertee: look at the channel, and compare it to the channel you care about and do your GotoIf based on it. |
| 19:53.53 | [TK]D-Fender | angryuser[A]: Show me the ENTIRE real picture, including CLI output... |
| 19:54.09 | [TK]D-Fender | angryuser[A]: I want to see that whole context and everything linked to it. |
| 19:54.34 | angryuser[A] | <[TK]D-Fender> ok ok, do you remember how do we copy things from putty client ? |
| 19:55.10 | [TK]D-Fender | angryuser[A]: Grab your mouse. Highlight. Paste. End of story |
| 19:55.19 | *** join/#asterisk mmmToop (n=michaelt@dsl-243-248-143.telkomadsl.co.za) |
| 19:56.26 | angryuser[A] | <[TK]D-Fender> i want to copy content from putty to my windows clipboard |
| 19:56.41 | mintee | weird, the 1.4 trunk build fine. |
| 19:56.46 | [TK]D-Fender | angryuser[A]: I jsut told you. HIGHLIGHT IT with your mouse and jsut PASTE. |
| 19:57.01 | [TK]D-Fender | ~pb |
| 19:57.35 | jbot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste |
| 19:57.36 | bkruse | highlight and right click angryuser[A] to copy, then right click to paste also |
| 19:57.36 | bkruse | 1.4 trunk? |
| 19:58.22 | badcfe | why does zap_channel get used (i use the transcoder wildcard) even for alaw-alaw bridges ? |
| 19:58.39 | angryuser[A] | <[TK]D-Fender> i does the paste in terminal, repeating the hightlighted text |
| 19:58.53 | badcfe | the transcoder counts alaw-alaw bridges 8-( |
| 20:00.06 | angryuser[A] | ok got it |
| 20:00.30 | hmmhesays | this guy testifying is really annoying |
| 20:00.49 | badcfe | i have two asterisks A and B and observe something for SIP only calls going thru them: when a phone calls A its sent forward to asterisk B -- now, the A reinvites the phone and B, all as configured. but sometimes i see a 491 Request pending from B to A. Actually A retransmits the re-INVITE to B even if B ansers 200 OK to it. So finally B says 491 Request pending. Why does this happen? Is it normal operation when an asterisks re-INVITE |
| 20:07.19 | *** join/#asterisk CrazyTux[m] (n=CrazyTux@ppp-70-248-182-215.dsl.hstntx.swbell.net) |
| 20:11.01 | *** part/#asterisk andresmujica (n=andresmu@correo.seaq.com.co) |
| 20:11.47 | tzafrir | <mintee> /usr/src/zaptel-1.4.8/wctc4xxp/base.c:52:26: error: linux/zaptel.h: No such file or directory |
| 20:11.57 | tzafrir | mintee, got that sorted out? |
| 20:12.11 | tzafrir | Are you sure you don't use a modified tarball? |
| 20:12.38 | tzafrir | modified Makefile? Modified base.c? |
| 20:12.51 | tzafrir | missinf -DSTANDALONE_ZAPATA somewhere? |
| 20:13.46 | angryuser[A] | http://www.pastebin.ca/900281 <[TK]D-Fender> ok i commented it a bit to make it easyer |
| 20:14.10 | *** join/#asterisk inadaptado (n=matias@32.59.64.129) |
| 20:15.27 | *** join/#asterisk danalien (n=danalien@unaffiliated/danalien) |
| 20:17.18 | *** join/#asterisk Strom_C (n=strom@208.127.172.112) |
| 20:23.05 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 20:23.41 | [TK]D-Fender | angryuser[A]: your "8" extension is NOT a proper IVR |
| 20:25.09 | patrick-- | can anyone tell me how i can completely remove my asterisk installation? |
| 20:25.38 | [TK]D-Fender | angryuser[A]: but aside from that, you hit "0" as the first digit of your intended exten in the secondary IVR, and there is no match and * is reacting perfectly normally. |
| 20:26.27 | *** join/#asterisk qufk (n=somedude@c-67-185-112-20.hsd1.wa.comcast.net) |
| 20:26.29 | [TK]D-Fender | patrick--: You don't need to yank out * completely. You can do "make samples after removing all your old configs to start from scratch, but then again you are only having problems with ONE aspect of your setup. No point in killing everything. |
| 20:27.11 | patrick-- | :) |
| 20:27.23 | *** join/#asterisk MACscr (n=Mark@adsl-75-23-68-162.dsl.peoril.sbcglobal.net) |
| 20:27.28 | MACscr | php script for remote queue agent logins? |
| 20:28.16 | errr | in our VPN, what ports would we need to forward to our pbx server to allow people to connect to the vpn then use some kind of soft phone to connect to the pbx and be able to send/recv calls? |
| 20:29.25 | angryuser[A] | <[TK]D-Fender> ok, so how to let user dial his number and call out with external sip provider? |
| 20:29.46 | *** join/#asterisk ZaVoid (n=zavoid@c-67-165-25-195.hsd1.pa.comcast.net) |
| 20:29.54 | *** join/#asterisk sweeper (i=sweeper@66.221.78.1) |
| 20:30.50 | patrick-- | http://phpfi.com/295878 <-- thats the current log... the server still doesnt start |
| 20:30.56 | sweeper | hey guys, I need to find myself another voip reseller with nice web billing interface, and CONUS termination at 2 cents or less, any suggestions? we've been having a bad experience with deltathree |
| 20:31.01 | angryuser[A] | from ivr |
| 20:31.54 | sweeper | err, not a voip reseleer, a voip PROVIDEr that I can get a reseller account with |
| 20:32.22 | [TK]D-Fender | angryuser[A]: Give them an EXTENSION they can dial that will let them |
| 20:32.48 | angryuser[A] | i have added a line like this, but it still stops after first 0 dialed exten => _000[12345789]XXXXXXXX.,3,Dial(Sip/${EXTEN:1}@voipprovider,60,t) |
| 20:33.39 | *** join/#asterisk Juggie (i=Juggie@CPE001601df17fb-CM001a6682d7b6.cpe.net.cable.rogers.com) |
| 20:33.49 | [TK]D-Fender | angryuser[A]: You also didn't set your inter-digit timouts, etc. "core show function TIMEOUT". You need to learn how to make proper IVR's, and doing so in a giant messy context the way you are doing it is bad. |
| 20:34.33 | *** part/#asterisk mmmToop (n=michaelt@dsl-243-248-143.telkomadsl.co.za) |
| 20:35.39 | angryuser[A] | <[TK]D-Fender> i know it is old one, i am working to clear then mess, but havent put it online yet |
| 20:35.48 | angryuser[A] | that* |
| 20:36.15 | [TK]D-Fender | errr: if they are VPN'd then typically they will have a local IP on the same private subnet as * is on. This mean you shouldn't need any kind of forwarding because they are already "inside" |
| 20:38.17 | *** join/#asterisk techie (n=techie@adsl-76-214-27-74.dsl.lsan03.sbcglobal.net) |
| 20:39.43 | errr | [TK]D-Fender: well our vpn requires you to forward a source port with a dest port and also to select either tcp or udp |
| 20:40.07 | [TK]D-Fender | errr: wierd, but OK... then follow this : |
| 20:40.09 | [TK]D-Fender | ~sipnat |
| 20:40.09 | jbot | [~sipnat] Quick guide on configuring * + SIP behind NAT : http://www.aocomputing.net/?p=3 otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
| 20:41.32 | *** part/#asterisk shtoom (n=godson@59.93.118.77) |
| 20:42.30 | drmessano-LT | jbot: drmessano-LT is like drmessano, just 1/3 less filler |
| 20:42.31 | jbot | okay, drmessano-LT |
| 20:42.39 | drmessano-LT | ~drmessano-LT |
| 20:42.39 | jbot | rumour has it, drmessano-lt is like drmessano, just 1/3 less filler |
| 20:42.42 | drmessano-LT | bya |
| 20:42.46 | drmessano-LT | byah too |
| 20:42.49 | errr | [TK]D-Fender: thanks |
| 20:43.29 | drmessano-LT | If you have a VPN that requires setting ports, it cant be much of a VPN lol |
| 20:44.05 | drmessano-LT | Sounds more like a walled garden or a decontamination chamber |
| 20:44.18 | [TK]D-Fender | drmessano-LT: it locks what you are even first allowed to access.... extra private! |
| 20:44.56 | x86 | anyone know if a CAC Adit 600 can do PRI on its T1 port? |
| 20:45.03 | drmessano-LT | We lock down subnet access.. To keep someone from having full access to the whole WAN.. but PORTS.. wow |
| 20:45.33 | [TK]D-Fender | x86: keep fighting with those channel banks, you'll win for sure! Double or nothing? ;) |
| 20:45.51 | *** join/#asterisk J4k3 (n=jsuter@openwrt.us) |
| 20:47.20 | errr | drmessano-LT: our vpn sucks. I wish we could get a real solution. |
| 20:47.37 | errr | but our company is to cheap to get what I want (a firepass) |
| 20:48.21 | *** join/#asterisk draygon (i=draygon-@208.76.99.254) |
| 20:48.31 | x86 | [TK]D-Fender: not having any problems with them now... |
| 20:48.45 | x86 | [TK]D-Fender: was just wondering if I should be using PRI... looks like the Adit 600 does not support it |
| 20:49.16 | drmessano-LT | errr: I have issue with calling it VPN.. In the future, should you refer to it here, please call it "BackyardBirthdayPartyWithNoClown" |
| 20:49.18 | [TK]D-Fender | x86: No point anyways.... your analog channel has no progress anyways |
| 20:49.49 | drmessano-LT | BBPWNC as an abreviation will suffice |
| 20:49.55 | errr | ok |
| 20:50.04 | ManxPower | x86: I have never ever heard of a plain channel bank having PRI support, but if you really want to know, CONTACT ADIT |
| 20:50.30 | ManxPower | As [TK]D-Fender said, there really isn't any point. |
| 20:51.04 | x86 | ManxPower: err, i already said I found out ;) |
| 20:51.27 | x86 | ManxPower: you said earlier that it's always better to do PRI? |
| 20:51.35 | x86 | ManxPower: now you're changing your story? :P |
| 20:51.47 | ManxPower | "I was wondering if I should fill up my car using diesel, but my car only runs on gas." |
| 20:51.55 | [TK]D-Fender | x86: keep flogging the deceased equine :) |
| 20:52.08 | x86 | ;) |
| 20:52.08 | ManxPower | x86: It is always better to use PRI, unless you are terminating into something horrid like a channel bank. |
| 20:52.20 | [TK]D-Fender | x86: to TELCO yes, PRI is best. |
| 20:52.31 | [TK]D-Fender | x86: the channel banks, no. |
| 20:52.37 | draygon | What is a good document online for installing a PBX on centos 5/ |
| 20:52.49 | [TK]D-Fender | draygon: ... |
| 20:52.50 | x86 | ManxPower: right, but when I asked this morning (and specifically said channel bank), you told me PRI would be better |
| 20:52.50 | [TK]D-Fender | ~book |
| 20:52.51 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com, or see ~buybook |
| 20:52.53 | [TK]D-Fender | ^^^^ |
| 20:53.18 | draygon | Is there anything more simple? heh |
| 20:53.27 | x86 | [TK]D-Fender: I _wish_ they would let me do PRI to the telco here... |
| 20:53.48 | x86 | [TK]D-Fender: we're an outbound call center, and we get better rates if we do CAS T1 |
| 20:53.58 | x86 | [TK]D-Fender: as CAS is dedicated, and PRI is switched |
| 20:55.21 | x86 | AT&T is our preferred telco vendor (dont ask me why), and we're paying $0.029 per minute with CAS T1 (which is insanity, given our call volume) |
| 20:55.40 | x86 | with a PRI T1, we're looking at around $0.039 per minute at the minimum |
| 20:56.10 | chavigny | hey can anyone tell me how to link two boxes, i have a PRI in one, im trying to link them with SIP users not iax, and then pass the call to that box |
| 20:56.13 | *** join/#asterisk JenniferAkemi (n=akemi@206-248-153-17.dsl.teksavvy.com) |
| 20:57.01 | *** part/#asterisk techie (n=techie@adsl-76-214-27-74.dsl.lsan03.sbcglobal.net) |
| 20:57.31 | x86 | chavigny: you'll get better performance with IAX |
| 20:57.37 | chavigny | ok.. |
| 20:57.48 | x86 | chavigny: sure it's possible either way though |
| 20:58.01 | chavigny | ok can you explain with IAX? please :) |
| 20:58.08 | x86 | chavigny: but if you do an IAX trunk between the two servers, you'll cut your bandwidth usage down |
| 20:58.09 | chavigny | I want to learn something today |
| 20:58.20 | x86 | read Teh Book |
| 20:58.21 | x86 | ~book |
| 20:58.22 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com, or see ~buybook |
| 20:59.01 | *** join/#asterisk [T]ank (n=ckwall@206.71.78.158) |
| 21:00.36 | chavigny | ok but x86, with sip peers couldnt I just pass the call like, exten => 5555555555,1,Dial(SIP/user:pass@ipofotherbox/${EXTEN}) |
| 21:00.51 | chavigny | because its only 1 number |
| 21:01.07 | [TK]D-Fender | chavigny: Are these boxes on a loacl LAN to each other? |
| 21:01.07 | [T]ank | so i have set up a phone in sip.conf with disallow=all then allow=gsm. I am getting an error when I try to dial out from that phone (linksys spa942) that says "Feb 11 13:56:52 NOTICE[4951]: chan_sip.c:3775 process_sdp: No compatible codecs!" |
| 21:01.14 | [T]ank | i thought gsm was pretty universal |
| 21:01.22 | chavigny | yes Fender |
| 21:01.23 | *** join/#asterisk DarWin_vcch (n=daryl@205.241.238.3) |
| 21:02.04 | chavigny | TANK maybe gsm is not avail on your box |
| 21:02.06 | [TK]D-Fender | chavigny: Then SIP is jsut fine, and yes you can pass off calls pretty easy. Go lookup "asterisk dual servers" on the WIKI for some inspiration and keep in mind that some of the code may be deprecated. |
| 21:02.23 | chavigny | or not enabled |
| 21:02.31 | [TK]D-Fender | [T]ank: pastebint he complete call attempt at verbose 10 & SIP debug enabled. |
| 21:02.39 | drmessano-LT | SPA-942s have GSM? |
| 21:02.49 | [TK]D-Fender | drmessano-LT: Nope. |
| 21:02.59 | drmessano-LT | Then that why it go poo poo |
| 21:03.03 | [T]ank | thats more what I was wondering.... |
| 21:03.09 | [T]ank | so spa942 is the issue, not the codec |
| 21:03.28 | drmessano-LT | I didnt think Linksys could spell GSM |
| 21:03.33 | drmessano-LT | I guess they can't |
| 21:03.36 | *** join/#asterisk jm|home (n=jm|home@zen.jamiem.com) |
| 21:04.46 | x86 | indeed |
| 21:05.17 | *** join/#asterisk mishehu (i=1000@crosscreek.cartissolutions.net) |
| 21:07.28 | [T]ank | suck!!!! |
| 21:07.37 | chavigny | gsm sucks |
| 21:07.43 | [T]ank | ok, so of these codecs which would be the most compact? Codecs supported : G177u, G711a, G726, G729a, and G723 |
| 21:07.59 | chavigny | g177 is ulaw but high bandwidth |
| 21:08.06 | chavigny | id use those if your on a pri |
| 21:08.17 | [T]ank | gsm was about 29kb per call. |
| 21:08.41 | [T]ank | which would give me about 48 concurrent calls on a 1.5mb T1 |
| 21:09.19 | *** join/#asterisk seanbright (n=elixer@65.207.74.18) |
| 21:09.36 | chavigny | oh use g729a |
| 21:09.46 | drmessano-LT | Do you have G729 licenses? |
| 21:09.52 | chavigny | might need the license from like digim |
| 21:10.01 | chavigny | digium |
| 21:10.08 | [T]ank | yeah... but dont i have to have one per phone? |
| 21:10.13 | drmessano-LT | No |
| 21:10.15 | chavigny | no just one per pbx |
| 21:10.21 | drmessano-LT | Wut? |
| 21:10.22 | drmessano-LT | No |
| 21:10.24 | drmessano-LT | One per channel |
| 21:10.33 | drmessano-LT | If you're transcoding |
| 21:10.46 | drmessano-LT | Thats not 1 per phone or 1 per PBX |
| 21:10.49 | [T]ank | thats another thing i am kind of confused about... |
| 21:10.54 | [T]ank | how do i know if i am transcoding |
| 21:11.09 | drmessano-LT | If the thing the G729 is talking to is not G729, its transcoding |
| 21:11.21 | *** join/#asterisk jetlagmk2 (i=jetlag@pool-70-17-48-245.pskn.east.verizon.net) |
| 21:11.49 | drmessano-LT | That could be another device, voicemail, a SIP peer, etc |
| 21:11.58 | [T]ank | so if i go from linksys spa942 to an asterisk server with the peer set to use g729... |
| 21:12.14 | drmessano-LT | You're not transcoding |
| 21:12.39 | drmessano-LT | brb.. need to go play engineer |
| 21:14.52 | *** join/#asterisk jetlagmk2 (i=jetlag@pool-70-17-48-245.pskn.east.verizon.net) |
| 21:15.18 | *** join/#asterisk WindBack (n=jorge@host59.190-31-75.telecom.net.ar) |
| 21:15.58 | Greek-Boy | if I use the Authenticate() app before Background() will DTMF input be considered the pin code for Authenticate() ? |
| 21:16.47 | WindBack | In the CDR of asterisk.. what is the field acountcode?? |
| 21:17.15 | timeshell | is there any work in getting asterisk to recognize 2 lines on the same IP and on the same port (ie. 5060) ? |
| 21:17.28 | WindBack | I saw that this field is only used in DISA |
| 21:17.40 | timeshell | I think I saw something like this in the features of 1.6.... is that true? |
| 21:19.30 | x86 | anyone ever play with impedance settings on a channel bank? |
| 21:19.51 | x86 | someone told me that I should be using 600 ohms, but my channel bank is setup to use 900 ohms |
| 21:20.06 | *** join/#asterisk moa_ (n=moa_@216-129-228-165.vnet-inc.com) |
| 21:20.11 | x86 | I'm in the US |
| 21:23.42 | *** join/#asterisk JT (n=j@unaffiliated/jt) |
| 21:23.42 | timeshell | Hey, drmesso |
| 21:23.51 | timeshell | Remember our Windows sucks debate? |
| 21:23.53 | *** join/#asterisk JonMcN (n=Jon@cpc4-sout2-0-0-cust715.sotn.cable.ntl.com) |
| 21:23.57 | timeshell | Check this out |
| 21:23.59 | timeshell | http://www.itworldcanada.com/Pages/Docbase/ViewArticle.aspx?id=idgml-14a08f27-0ec4-42c2&Portal=4fb7319b-aa7c-423a-822d-2f6e24698c71&sub=1503762 |
| 21:24.20 | *** join/#asterisk nvrpunk (n=root@81.90.21.227) |
| 21:24.23 | JonMcN | Hi, If i want to have 1 extension number shared over two physical handsets, how can i do it? |
| 21:24.32 | timeshell | yes |
| 21:24.36 | JonMcN | I'm finding only one will ring :( |
| 21:24.50 | timeshell | what kind of handsets? |
| 21:24.51 | JonMcN | (using RT btw) |
| 21:25.05 | JonMcN | timeshell, Linksys SPA-942 |
| 21:25.19 | JonMcN | but it's not the handset, asterisk only knows about one URI |
| 21:25.31 | nvrpunk | hey, I am trying to get my dialplan setup for a DID number http://zomgoblinz.org/extensions.conf <-- that's my test config anyone mind pointing out whats wrong? |
| 21:25.31 | JonMcN | So is only pushing the call to one handset |
| 21:25.59 | bkruse | I do not know if I would go to a website with 'zomg' in the name |
| 21:26.10 | timeshell | lol |
| 21:26.20 | nvrpunk | bkruse, my fiancee's word :/ |
| 21:26.28 | timeshell | get my messages bk? |
| 21:26.30 | nvrpunk | i thought it was a neat domain name! |
| 21:27.15 | bkruse | timeshell: yep, dont have time today, but will tomorrow, ping me about it |
| 21:27.24 | timeshell | np |
| 21:27.41 | timeshell | Just wanted to make sure you knew the ones I sent were good (at least for me) |
| 21:28.06 | bkruse | timeshell: I believe they are, initial code review was fine (i need to mark that they are good) and then discuss implementation |
| 21:29.23 | timeshell | that's great. I look forward to hearing the result |
| 21:30.21 | *** join/#asterisk BCS-Satori (n=somewher@75.148.21.113) |
| 21:30.24 | timeshell | JonMcN: Not sure what to tell you. I've many times logged in as the same user from multiple devices. |
| 21:31.04 | timeshell | I think... |
| 21:31.06 | timeshell | :p |
| 21:31.17 | BCS-Satori | I am attempting to setup asterisk realtime with mysql, and upon a reload i see this "[Feb 11 16:28:41] ERROR[2776]: res_config_mysql.c:853 mysql_reconnect: MySQL RealTime: Ping failed (2006). Trying an explicit reconnect." I dont see much googling it, could anyone explain this error to me and how to resolve it? |
| 21:31.33 | JonMcN | timeshell, yeah - outbound works fine - just inbound * is only ringing one handset :( |
| 21:31.47 | timeshell | Settings on the phone? |
| 21:31.51 | timeshell | Ringer off? :p |
| 21:32.00 | JonMcN | it's *, i'm sure of it |
| 21:32.19 | timeshell | do both phones show up on sip show registry? |
| 21:33.11 | moa_ | Anyone got a second to explain to me what the "logical span number" inside a spanmap is? |
| 21:34.15 | x86 | anyone? |
| 21:34.32 | x86 | someone told me that I should be using 600 ohms, but my channel bank is setup to use 900 ohms... what would this affect? |
| 21:34.43 | [T]ank | ok, so i set my peer in spi.conf for my linksys spa942 to use g729 and this is what happened: http://pastebin.ca/900380 any ideas on what I need to look at or what I did wrong? |
| 21:35.32 | JonMcN | timeshell, good question - i'll check |
| 21:35.49 | timeshell | [T]ank: You have the g729 codec? |
| 21:35.55 | moa_ | I was just going to say that |
| 21:36.00 | timeshell | [T]ank: It's not included in asterisk |
| 21:36.07 | [T]ank | ahhhhhhhhhh |
| 21:36.09 | [T]ank | :-D |
| 21:36.13 | [T]ank | i assumed ;-) |
| 21:36.33 | [T]ank | i thought i just had to get a license, not that I had to install it also. |
| 21:36.44 | [T]ank | does asterisk have to be reinstalled after I install the codec? |
| 21:37.06 | timeshell | I believe you may need to do a new menuselect and recompile |
| 21:37.16 | timeshell | (although I could be mistaken) |
| 21:37.26 | *** join/#asterisk a1fa (n=a1fa@unaffiliated/a1fa) |
| 21:37.40 | a1fa | ook.. i am getting sick and tired of PAP2 instability |
| 21:37.57 | timeshell | a1fa: instability?? I've not had any problems with mine |
| 21:38.07 | a1fa | what firmware version? |
| 21:38.15 | timeshell | 3.1.6 |
| 21:38.28 | a1fa | where is you * box at? |
| 21:38.32 | timeshell | 1.4.18 |
| 21:38.34 | a1fa | LAN or Internet? |
| 21:38.38 | timeshell | LAN |
| 21:39.01 | *** part/#asterisk lirakis (i=lirakis@66.252.24.133) |
| 21:39.24 | timeshell | And I'm using both lines on mine |
| 21:39.38 | timeshell | Is your's a NA or an unlocked? |
| 21:40.20 | drmessano-LT | a1fa |
| 21:40.37 | drmessano-LT | Is this the same PAP2 you couldn't get working a few weeks back? |
| 21:40.55 | a1fa | myup |
| 21:40.56 | a1fa | yup |
| 21:41.02 | a1fa | it may be over-heating or something |
| 21:41.02 | drmessano-LT | THROW IT IN THE TRASH |
| 21:41.05 | timeshell | lol |
| 21:41.15 | timeshell | better yet, send it to me |
| 21:41.16 | timeshell | :D |
| 21:41.16 | drmessano-LT | ITS NOT UNSTABLE, ITS EFFIN BORKED |
| 21:41.16 | a1fa | i had to power it off and let it cool of for 5min before it started working again |
| 21:41.26 | a1fa | Firmware Version: 3.1.22(LS) |
| 21:41.32 | timeshell | Ah, overheated |
| 21:41.36 | drmessano-LT | THROW IT AWAY |
| 21:41.37 | timeshell | Mine's done that |
| 21:41.43 | a1fa | timeshell : does it stop working? |
| 21:41.51 | timeshell | It's gone weird a couple times |
| 21:41.58 | timeshell | I usually just unplug/replug |
| 21:42.05 | a1fa | yeah |
| 21:42.07 | drmessano-LT | lol |
| 21:42.07 | a1fa | that thing |
| 21:42.12 | x86 | anyone ever use OPS lines? |
| 21:42.17 | x86 | Off-Premise Stations |
| 21:42.19 | a1fa | it just stops resolving dns |
| 21:42.19 | timeshell | But I think it's only happened to me like 2 or 3 times in the past year |
| 21:42.24 | a1fa | and it stops trying to register |
| 21:42.33 | timeshell | Mine isn't using DNS |
| 21:42.33 | drmessano-LT | I have had 1 stop working and needing to be reset |
| 21:42.43 | drmessano-LT | Once |
| 21:42.54 | drmessano-LT | I think your PAP2 is broken.. |
| 21:42.58 | drmessano-LT | Throw it away |
| 21:43.08 | timeshell | I also have it sitting on my other Linksys routers and a lot of heat get's generated |
| 21:43.11 | *** join/#asterisk kyron (n=kyron@211-217-static-ppp.3menatwork.com) |
| 21:43.34 | timeshell | It could be a buggy flash |
| 21:43.45 | timeshell | 3.1.22 huh? |
| 21:43.47 | drmessano-LT | It could be cockroaches |
| 21:43.58 | timeshell | heh |
| 21:44.06 | drmessano-LT | or maybe oompa loompas |
| 21:44.22 | drmessano-LT | or, idk, it could be broke |
| 21:44.24 | timeshell | Or drmessano's cat... |
| 21:44.30 | drmessano-LT | yes |
| 21:44.38 | timeshell | How's Itchy? |
| 21:44.46 | *** part/#asterisk axisys (i=iqbala@outbound.silenceisdefeat.org) |
| 21:44.53 | timeshell | :D |
| 21:44.54 | drmessano-LT | Itchy is scratchy |
| 21:44.57 | timeshell | yes |
| 21:44.57 | drmessano-LT | lol |
| 21:45.00 | timeshell | that's what I meant |
| 21:45.03 | timeshell | :D |
| 21:45.33 | pkunkra | i've heard of lots of cats getting caught in electrical closets and shorting out the equipment |
| 21:45.55 | timeshell | Really? |
| 21:45.59 | timeshell | Mine's never done that |
| 21:46.06 | drmessano-LT | Sorry, drmessano pet peeve #11 If someone tells you something broke, don't show up weeks later complaining of some non-existant stability issues |
| 21:46.09 | pkunkra | as they say... curiosity killed the cat. |
| 21:46.15 | timeshell | I could really use some advice with my polycom 301 |
| 21:46.17 | a1fa | brb |
| 21:46.32 | pkunkra | well, if there is an opening it can crawl into, it will go inside and explore |
| 21:46.33 | timeshell | I need to register 2 lines on it with the same server, but it doesn't seems to know how to use 5061 properly |
| 21:46.34 | a1fa | drmessano: why you gotta be a d1ck ? |
| 21:46.35 | a1fa | :p |
| 21:46.47 | drmessano-LT | "Asterisk 1.4 is so friggin unstable.. This PII 200 should be able to handle 100 concurrent calls" |
| 21:46.47 | timeshell | (or I've forgotten to do something on the asterisk server to let it) |
| 21:47.38 | pkunkra | if there is anything that's exposed that i might rub against... welll. you get the picture. |
| 21:47.43 | pkunkra | it* |
| 21:48.19 | drmessano-LT | Because, a1fa, as Dr. Christian Troy says "You have to possess two things, a steady hand, and a big ____" |
| 21:48.32 | pkunkra | whatever. |
| 21:48.33 | timeshell | You shouldn't rub things the wrong way |
| 21:48.39 | angryuser[A] | <[TK]D-Fender> my autofallthrough=yes and i have added the exten => 9,1,Set(TIMEOUT(digit)=5) and then WaitExten but nothing changes, * does not wait my chain to complete |
| 21:48.51 | clyrrad | Anyone used PrivacyManger()? If so how do you get around the case when caller id is set to "unknown" instead of blank empty string? |
| 21:49.20 | clyrrad | Almost need a way to check if the caller id is numeric |
| 21:51.02 | *** part/#asterisk beek (n=klinebl@65.211.106.243) |
| 21:52.56 | [T]ank | so i am looking at the g729 page on voip-info.org. What is the difference between g729 and g729a? And do i have to use the intel wrapper with this also? I am really confused on this whole thing. |
| 21:53.48 | timeshell | Why don't you use g723, g726 or even ulaw or alaw? |
| 21:53.53 | a1fa | drmessano-LT : who said that |
| 21:54.34 | [T]ank | timeshell: i am looking to increase the number of calls per t1 I can do, and apparently my phone does not support gsm :-( |
| 21:54.53 | [T]ank | how does each of the others compare in bandwidth? |
| 21:55.09 | JonMcN | timeshell, only showing 1 :( |
| 21:56.16 | timeshell | [T]ank: http://www.ozvoip.com/voip-codecs/ |
| 21:56.29 | timeshell | JonMcN: Well, I think that's your issue |
| 21:56.39 | timeshell | The other phone isn't registring |
| 21:57.17 | JonMcN | well it is! |
| 21:57.26 | timeshell | No.. |
| 21:57.33 | timeshell | It doesn't need to register to make calls |
| 21:57.34 | JonMcN | but, using RT, * is only caching 1 |
| 21:57.48 | nvrpunk | hmm, anyone mind looking at my test extensions.conf trying to setup DID on one extension and haven't had any luck |
| 21:57.57 | nvrpunk | http://zomgoblinz.org/extensions.conf |
| 21:58.00 | timeshell | A phone needs to register to receive calls |
| 21:58.02 | jameswf | Ron paul can run 25 calls on a Single Pots line while doinf TDM in hid head |
| 21:58.11 | JonMcN | timeshell, it is |
| 21:58.31 | jameswf | *doing *his |
| 21:58.33 | [T]ank | timeshell:so if i am reading this right.... g723 would be better than gsm. |
| 21:58.36 | [T]ank | right? |
| 21:58.43 | JonMcN | g723 :( |
| 21:58.43 | [T]ank | call quality suck? |
| 21:58.46 | JonMcN | g729 :) |
| 21:59.14 | [T]ank | JonMcN:yeah, thats what I was planning on using... sounds like i was being talked out of it. |
| 21:59.15 | timeshell | I'm not qualified to advise on the best codec. |
| 21:59.25 | [T]ank | i am confused as to how to impliment it. |
| 21:59.37 | [T]ank | looks like i has to either use windows or an intel wrapper. is that correct? |
| 21:59.42 | timeshell | g723 seems to use reasonable bandwidth. |
| 21:59.58 | timeshell | I'd suggest try it first and move along to something else if it doesn't work for you. |
| 22:00.06 | timeshell | At least you know it'd work |
| 22:00.09 | timeshell | :p |
| 22:00.35 | Daviey | g723 uses more cpu and bw than g729 |
| 22:00.38 | Daviey | why use it? |
| 22:00.41 | jameswf | Ron paul can transcode to 32 different codecs while sleeping |
| 22:01.24 | timeshell | nvrpunk: What's your incoming #? |
| 22:01.29 | a1fa | Chuck Norris can transcode to 100 devices |
| 22:02.12 | nvrpunk | 8772629143 |
| 22:02.22 | timeshell | nvrpunk: No 1? |
| 22:02.25 | a1fa | Chuck Norris FTW |
| 22:02.27 | jameswf | Ron Paul can make Chuck Noris his peison b*** with a spoon and have a total of 132 codecs while sleeping |
| 22:02.32 | nvrpunk | well theres a 1 |
| 22:02.34 | nvrpunk | before it |
| 22:02.41 | nvrpunk | usa number |
| 22:02.44 | jameswf | *prison |
| 22:03.30 | [T]ank | so just answer me this so i can keep researching... is g729 and g729a the same thing or two different codecs? |
| 22:03.30 | nvrpunk | timeshell, do I need to change the actuall extension to have the 1 ahead of it? |
| 22:03.42 | *** join/#asterisk fiXXXerMet (n=kjohnson@dsl092-156-002.wdc2.dsl.speakeasy.net) |
| 22:03.43 | nvrpunk | isnt there a and b |
| 22:03.47 | nvrpunk | o0 |
| 22:03.54 | timeshell | Is this a SIP incoming? |
| 22:04.12 | fiXXXerMet | Anyone here use Cisco IP phones? Need some help with getting a 7914 expansion module to work with a 7961 ip phone. |
| 22:04.32 | nvrpunk | timeshell, IAX trunk with an IAX softphone |
| 22:05.04 | nvrpunk | G.729a is compatible with G.729, but requires less computation |
| 22:05.14 | nvrpunk | so different but work together |
| 22:05.15 | timeshell | I've had trouble getting callerid working on a iax2 trunk... |
| 22:05.30 | timeshell | Always showed me the connections id rather than the callerid |
| 22:05.35 | [T]ank | so I would want g729a then |
| 22:06.05 | nvrpunk | timeshell, im just interested in getting it so they can dial my iax2 via that number |
| 22:06.25 | nvrpunk | i have 4 soft phones, 2 sips, 2 iax |
| 22:06.28 | nvrpunk | iax 1 and 2 |
| 22:06.30 | nvrpunk | sip 1 and 2 |
| 22:06.31 | timeshell | use s, then for your incoming |
| 22:08.09 | nvrpunk | so, exten => _1NXXNXXXXXX,s,1,Playback(beep) ? |
| 22:08.12 | timeshell | no |
| 22:08.16 | nvrpunk | err -1 |
| 22:08.24 | timeshell | exten=>s,1,Playback(...) |
| 22:08.33 | *** part/#asterisk marlow (n=marlow@loke.sca.airwire.ie) |
| 22:11.15 | nvrpunk | so in theory, I should be able to call myself and get a busy signal, correct? |
| 22:11.36 | timeshell | What is the # coming infrom? |
| 22:11.46 | timeshell | IAX2 server? PSTN? SIP? |
| 22:11.50 | timeshell | ZAP? |
| 22:11.54 | nvrpunk | PSTN |
| 22:12.00 | timeshell | Using zap? |
| 22:12.09 | nvrpunk | junction networks |
| 22:12.12 | nvrpunk | no clue |
| 22:12.14 | *** join/#asterisk [TK]D-Fender (n=Joe@64.235.216.2) |
| 22:12.18 | nvrpunk | thats not my end |
| 22:12.26 | timeshell | How is it getting to you? |
| 22:12.30 | timeshell | IAX2? |
| 22:12.34 | nvrpunk | yes |
| 22:12.37 | nvrpunk | via them |
| 22:13.00 | timeshell | If you call yourself, unless you can only use one channel, I'd say you'd get ringing. |
| 22:13.34 | nvrpunk | not getting anything |
| 22:13.42 | nvrpunk | did you take a look at my extensions.conf? |
| 22:13.48 | timeshell | yes |
| 22:14.06 | nvrpunk | only think I changed was the _1NXXNXXXXXX to s |
| 22:14.19 | timeshell | What's your console telling youu? |
| 22:14.36 | nvrpunk | how do i check that, im a noob |
| 22:14.45 | timeshell | asterisk -r |
| 22:15.14 | variable_office | would it be possible to have a single ata register as one sip user and then spit out each of the concurrent calls down a different pots line? |
| 22:15.34 | timeshell | uh what? |
| 22:15.49 | timeshell | Doesn't sound like it. |
| 22:16.13 | [TK]D-Fender | variable_office, Umm... what "pots line"? Calls from where to where? |
| 22:16.18 | timeshell | Unless the ATA has multiple ports with individual connection settings |
| 22:16.27 | nvrpunk | timeshell, I did asterisk -r but not seeing anything across the console |
| 22:16.28 | a1fa | [TK]D-Fender ^5 |
| 22:16.36 | timeshell | make the call again after you run asterisk -r |
| 22:16.41 | variable_office | have the incoming calls from SIP go out a different fxs line on the ata |
| 22:16.43 | timeshell | You should see some verbage |
| 22:16.52 | *** join/#asterisk Ryushin (i=proxy@windwalker.openinnovations.com) |
| 22:17.01 | kyron | Q: does anyone have any concise references to setting the optimal MTU for VoIP traffic (µLaw)? I'd guess something around the average packet size of a stream (then again, I could get burned on the IAX trunk side) |
| 22:17.06 | timeshell | Have multiple lines on the ata login with the same user |
| 22:17.11 | timeshell | on different ports |
| 22:17.19 | nvrpunk | [Feb 11 20:29:27] NOTICE[3325]: chan_iax2.c:6031 update_registry: Restricting registration for peer 'iax2' to 60 seconds (requested 1200) |
| 22:17.19 | timeshell | That might do it |
| 22:17.37 | a1fa | hmm |
| 22:17.41 | a1fa | i am eating lill cezars pica |
| 22:17.42 | a1fa | :p |
| 22:17.43 | a1fa | pizza |
| 22:17.44 | a1fa | uhmmm |
| 22:17.57 | timeshell | You know nvrpunk, I'd suggest speaking with your provider on what your settings should be... |
| 22:17.59 | nvrpunk | their pizza's not as good as it used to be |
| 22:18.01 | nvrpunk | back in the day |
| 22:18.07 | a1fa | i dont know |
| 22:18.09 | a1fa | its good $5 |
| 22:18.18 | a1fa | it beats pizza hut at $15 |
| 22:18.22 | timeshell | If you're not even getting something coming in... |
| 22:18.26 | nvrpunk | timeshell, the way it was is the way they stated :p |
| 22:18.30 | russellb | for $5, it's a win |
| 22:18.35 | russellb | IMO |
| 22:18.36 | russellb | :) |
| 22:18.39 | timeshell | nvrpunk: Do you have a register line in your iax2.conf? |
| 22:19.01 | variable_office | [TK]D-Fender, is what i said possible or make sense? |
| 22:19.16 | variable_office | I am just trying to figure out how to deal with concurrent calls on an ata |
| 22:19.17 | timeshell | Did you register your asterisk server with your provider? |
| 22:19.30 | nvrpunk | timeshell, my iax.conf has one |
| 22:19.35 | nvrpunk | and the trunk is registered |
| 22:19.44 | nvrpunk | yeah |
| 22:19.45 | nvrpunk | i can call out |
| 22:19.46 | nvrpunk | just fine |
| 22:19.48 | nvrpunk | through them |
| 22:20.00 | [TK]D-Fender | variable_office, What ATA? |
| 22:20.01 | *** join/#asterisk craigk (n=craigk@58.174.150.119) |
| 22:20.23 | timeshell | When you call your 800# you should at least see something coming in on your console indicating what it is. |
| 22:20.39 | variable_office | [TK]D-Fender, its for our business customers, we havent committed to anything yet, obviously i would like something good and inexpensive as possible; any suggestions on this? |
| 22:21.22 | [TK]D-Fender | variable_office, ok, you are throwing nameless scenarios areound poorly worded. start over. You are not yet clear and your situation not somethign you can "show" use since it doesn't exist yet |
| 22:21.47 | [TK]D-Fender | variable_office, Please be very specific with references to any tech involved with each call you are referring to. |
| 22:23.20 | *** join/#asterisk inadaptado (n=matias@190.3.121.15) |
| 22:23.59 | variable_office | Ok, I want business users, who are use to using a single multi-line phone with say 4 pots lines to be able to migrate to Sip. The easiest way I can think of doing this is giving their ATA a single SIP registration back to our server and set maxcalls=4 and then their phone number can have up to 4 calls similar to their old pots setup. But in order to use the same phone, each call would need to come off a different line so that it could g |
| 22:23.59 | variable_office | o into their old 4 line phone. does that make sense? |
| 22:24.27 | [TK]D-Fender | variable_office, Yes, now it makes sense. |
| 22:24.55 | a1fa | variable_office : you have a pbx at the office? |
| 22:25.15 | [TK]D-Fender | variable_office, HOWEVER.... this is cludgy and means you'll need 2 ATA's for a 4-line phone. That would cost over 110$ USD for which you could get them a NICE phone instead. |
| 22:25.17 | variable_office | not at the customers office, no, i just wanted it to be a drop-in ata |
| 22:25.32 | [TK]D-Fender | variable_office, but yes you can do this and it would involve some dialplan trickery. |
| 22:25.35 | variable_office | [TK]D-Fender, could i not get a 4 line-ata? |
| 22:25.37 | a1fa | [TK]D-Fender : or he could get a nice zap card |
| 22:25.48 | [TK]D-Fender | a1fa, ..... YUCK! |
| 22:25.51 | timeshell | Wouldn't it be better to create a ring group and ring all? |
| 22:25.52 | a1fa | variable_office : grandstream makes an 8 port one |
| 22:25.56 | a1fa | [TK]D-Fender : =) |
| 22:26.06 | a1fa | i |
| 22:26.08 | a1fa | i' |
| 22:26.20 | a1fa | i'd do what d-fender suggests.. buy policom phones |
| 22:26.24 | a1fa | run some cat5 |
| 22:26.31 | a1fa | it will be cheaper + better experience for end user |
| 22:26.32 | variable_office | [TK]D-Fender, so it is not possible to make this a simple ata-level logic, of, for each new call that comes down sipuser1, drop the call onto a new fxs line? |
| 22:26.39 | a1fa | dont use atas dude |
| 22:26.44 | a1fa | i've done that once |
| 22:26.52 | a1fa | hooked ATA to POTS PBX |
| 22:26.59 | a1fa | not a very good idea |
| 22:27.06 | [TK]D-Fender | variable_office, I just said "YES". However just because you can do something doesn't mean you SHOULD. This is an UGLY and non-cost effective solution. |
| 22:27.22 | a1fa | it will cost you more in the long run |
| 22:27.25 | a1fa | if you run atas |
| 22:27.35 | a1fa | because you will see that this solutions sucks |
| 22:27.41 | a1fa | it also adds a lot of overhead management |
| 22:27.48 | [TK]D-Fender | a1fa, ATA's are jsut fine.... if you are using a single port for a single phone you're more than fine. Trying to recycle multi-line analog phones is BS however |
| 22:28.02 | angryuser[A] | i am using application Playtones(!440) and after i execute read, i would like it to stop generate tone after first digit pressed, how? |
| 22:28.15 | a1fa | IP 550 |
| 22:28.20 | variable_office | [TK]D-Fender, ya but try convincing a business owner of that |
| 22:28.20 | a1fa | Polycom does 4 lines |
| 22:28.28 | angryuser[A] | <[TK]D-Fender> it is working now thx |
| 22:28.31 | a1fa | tell him it will be cheaper in the long run |
| 22:28.32 | [TK]D-Fender | IP 550 = waste. Overpriced and no point... |
| 22:28.42 | a1fa | 330? |
| 22:28.45 | a1fa | or 320 then? |
| 22:29.02 | a1fa | what do you have in mind |
| 22:29.17 | [TK]D-Fender | variable_office, easy... no MOH for your analog phones. have to WAIT for CID. No proper conferencing across multiple calls. More complex dialplan and devices to configure |
| 22:29.33 | [TK]D-Fender | variable_office, that idea SUCKS and is to be avoided unless necessary |
| 22:29.57 | [TK]D-Fender | a1fa, yup, IP 320/330 depending on Cat5 availability |
| 22:29.59 | angryuser[A] | what about snoms? nice phones |
| 22:30.05 | [TK]D-Fender | angryuser[A], Bleh |
| 22:30.35 | [TK]D-Fender | angryuser[A], Second rate audio, firmware flakyness, poor LCD usage, etc. |
| 22:30.48 | a1fa | [TK]D-Fender : he wanted 4 lines so i suggested 550 :P |
| 22:30.58 | variable_office | [TK]D-Fender, ok, well say i get them convinced to do sip phone, how would I run it so that each new call goes to a different line then (the goal is to NOT have to mess with my asterisk dialplan for every new user) |
| 22:31.02 | angryuser[A] | <[TK]D-Fender> you got a point on firmware |
| 22:31.17 | a1fa | lol |
| 22:31.22 | a1fa | variable_office : configure your shit right |
| 22:31.45 | a1fa | you wouldnt have to mess with dialplan if you set it right the first time |
| 22:31.58 | a1fa | XXX dials SIP/XXX |
| 22:32.07 | a1fa | thats one way around that problem |
| 22:32.13 | [TK]D-Fender | variable_office, You clearly have never worked with a decent SIP phone before. |
| 22:32.18 | timeshell | Ok, so while we're talking about polycom, how to do you get a polycom 301 to register 2 lines on the same server without getting authentication digest errors when trying to dial out from the second line on the same port? OR, how do you get the polycom to register the second line on a different port? I've tried changing the second line's port to 5061, but it won't register on the Asterisk. |
| 22:32.47 | variable_office | alfa thats not the part i am concerned with |
| 22:33.13 | angryuser[A] | <[TK]D-Fender> but besides that when using 6.2.3 it is working fine, i have another aastra phone, the xfer usage is poor, not user friendy |
| 22:33.17 | [TK]D-Fender | variable_office, there is nothing to configure in * for a normal sip phone to handle multiple calls. |
| 22:33.32 | [TK]D-Fender | angryuser[A], Yes, I hat Aastra for different reasons :) |
| 22:33.34 | [TK]D-Fender | hate* |
| 22:35.24 | variable_office | [TK]D-Fender, but then the sip phone will just put each current call in the same line on the same phone, I want it to be spaced out across multiple phones |
| 22:36.11 | timeshell | That statement doesn't make sense to me |
| 22:36.34 | variable_office | so that, for example, if this is a restaurant that gets calls, I want it so that the customers can dial the same restaurant number, but it will ring a free phone each time |
| 22:36.45 | angryuser[A] | <[TK]D-Fender> on snom you push transfer and blf ocnfigured button, on aastra you can have BLF but no that direct transfer without consulting, on aastra you need to dial manually ext each time, total mess |
| 22:36.50 | jameswf | Why would a 7 foot wookie live on...... it does not make sense |
| 22:37.03 | timeshell | variable_office: Why not use a call queue or a ringgroup then? |
| 22:37.39 | variable_office | because that is stuff that then has to be manually configured on my end |
| 22:38.26 | timeshell | Have your ata log in with multiple lines with the same user on the asterisk server. |
| 22:38.54 | variable_office | timeshell, would asterisk space them out evenly? |
| 22:39.10 | timeshell | THey would all ring |
| 22:39.41 | jameswf | jbot: tell variable_office about buybook |
| 22:39.43 | *** join/#asterisk egecko (n=sam@cpe-76-176-206-245.san.res.rr.com) |
| 22:39.49 | variable_office | and then the first to pick up would win? |
| 22:39.57 | timeshell | If you want to consecutively ring them, you need a queue or a ringgroup |
| 22:40.07 | timeshell | variable_office: basically |
| 22:40.38 | variable_office | i seem to remember trying multiple simultaneous registrations before, but i thought that wasn't correct, you have done that before? |
| 22:41.20 | timeshell | I have done it. I don't know how it would fare as a production configuration, but I've had it work. |
| 22:42.07 | variable_office | anybody using that method in production? |
| 22:42.51 | jameswf | we have individual users with a soft phone and a sip phone both registered but not across multiple stations |
| 22:42.51 | nvrpunk | timeshell, I had to set the whole extension on to 18772629143 |
| 22:43.32 | timeshell | Glad to hear you got it to work. |
| 22:47.32 | variable_office | jameswf, so you had two sip devices registered to the same server as the same sipuser? |
| 22:48.11 | jameswf | yes |
| 22:48.22 | jameswf | not devices |
| 22:48.31 | jameswf | 1 device one soft phone |
| 22:48.54 | ManxPower | jameswf: that is two devices. that won't work. Don't do it. |
| 22:49.17 | ManxPower | The last device to register on that account is the one that will get the calls. |
| 22:49.29 | [TK]D-Fender | variable_office, You would have to make a kludgy dialplan to account for this. it would be messy, but possible |
| 22:49.50 | variable_office | ManxPower, thats what I thought, i dont know if you saw any of the above, but do you have any idea on how to do that? |
| 22:49.58 | [TK]D-Fender | variable_office, seriously though, this idea is STUPID and should be avoided. |
| 22:50.19 | variable_office | [TK]D-Fender, then how can i have multiple phones ring for the same number? |
| 22:50.22 | ManxPower | variable_office: you don't do it. |
| 22:50.41 | ManxPower | exten => 666,1,Dial(SIP/user-a&SIP/user-b) |
| 22:50.42 | [TK]D-Fender | variable_office, That question shows you don't even know the Dial application. |
| 22:50.47 | jameswf | ~ringgroup |
| 22:50.50 | timeshell | variable_office: I have to agree. Although it may work, it's not a preferred way to do it |
| 22:50.52 | [TK]D-Fender | variable_office, You need to find a Clue, and fast... |
| 22:50.56 | [TK]D-Fender | ~cluebat variable_office |
| 22:50.57 | jbot | ACTION pulls out a ClueBat (tm) and thwaps variable_office. |
| 22:51.11 | jameswf | ~book |
| 22:51.11 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com, or see ~buybook |
| 22:51.14 | ManxPower | where user-a and user-b are accounts in sip.conf |
| 22:51.31 | ManxPower | i.e. in sip.conf [user-a] and [user-b] entries. |
| 22:51.47 | [TK]D-Fender | Ringgroup is NOT a real telecom or Asterisk term! Forget it NOW. |
| 22:52.07 | variable_office | uggh, i understand that, I understand the basics of asterisk, that would be easy for a user or two, but i do NOT want to have to reedit the dialplan every single time I add a new user |
| 22:52.10 | ManxPower | variable_office: one of the most important (maybe THE most important) thing you need to understand about VoIP is that an extension is just a number, the PBX maps that number to the real device (sip, IAX, etc) |
| 22:52.25 | variable_office | this isnt for a one time only setup, I want a cookie-cutter repeatable approach |
| 22:52.34 | ManxPower | variable_office: That is another thing you will have to learn. you need to edit the dialplan for every new device. |
| 22:52.52 | ManxPower | You can whine, scream, throw temper tantrums, but that is the way it is. |
| 22:52.57 | variable_office | ManxPower, not the way I have it now, it is all enum and fanciness |
| 22:53.12 | ManxPower | variable_office: and as you can clearly see it does not scale for complex dialplans. |
| 22:53.20 | jameswf | heh http://en.wikipedia.org/wiki/Rings_of_Uranus |
| 22:53.26 | timeshell | variable_office: I have to agree with these guys. If you want to find a fancy way to go about it, I think you're basically on your own. |
| 22:53.53 | timeshell | variable_office; Trial and error . |
| 22:54.07 | ManxPower | using the extension as the device ID is not a good way to design a dialplan. |
| 22:54.33 | timeshell | I agree with that too...so why does *gui do that? |
| 22:54.34 | variable_office | ManxPower, thats not what I am doing, I have each number in enum, and then each user has a customer user id |
| 22:54.37 | ManxPower | Hell, not even the TELCO supports the same number on different lines. |
| 22:54.45 | jameswf | jbot tell variable_office: about freepbx |
| 22:54.53 | timeshell | :D |
| 22:54.59 | jameswf | jbot tell variable_office about freepbx |
| 22:55.40 | [TK]D-Fender | jameswf, FreePBX won't survive the analog craziness he was looking for. |
| 22:55.42 | ManxPower | timeshell: because that is what users want and freepbx does MASSIVE amounts of work and incredibly complex dialplans and configs to make that work to the user. I guarntee you that freepbx does not use the same sip userid for multiple devices. |
| 22:55.43 | variable_office | ManxPower, i agree with that, But I am trying to improve upon that |
| 22:55.54 | jameswf | [TK]D-Fender: will anything |
| 22:56.03 | jameswf | within his power |
| 22:56.10 | ManxPower | variable_office: then you would be the first person to succeed in doing that in the entire history of Asterisk. |
| 22:56.20 | [TK]D-Fender | variable_office, Ok, I'll sum this up nice & quick : You are trying to outsmart * and you will almost certainly fail, and hard. |
| 22:56.24 | [TK]D-Fender | ~wglwat |
| 22:56.25 | jbot | wglwat is, like, well, good luck with all that |
| 22:56.51 | timeshell | Manx: You dont' have to convince me!!! |
| 22:56.52 | timeshell | :p |
| 22:56.55 | variable_office | ughh, i just I could just set calllimit=1 and then if the line is busy drop it off to line b, then line c and so forth like the telco eh? |
| 22:57.02 | jameswf | i saw walmart :) |
| 22:57.11 | ManxPower | Asterisk does not support registrations from different devices to the same SIP UserID. If you don't like that then you are welcome to rewrite chan_sip.c |
| 22:57.30 | jameswf | ~rtfc |
| 22:57.31 | jbot | hmm... rtfc is read the fine code |
| 22:57.54 | ManxPower | variable_office: that is pretty much exactly what we do. We set the SIP UserID to be the MAC address of the phone with -a, -b, -c etc appended to it for 1st line appearance, 2nd line appearance, etc. |
| 22:58.00 | pkunkra | i think they should have used the more colorful version of "fine" |
| 22:58.01 | jameswf | ~rwtfc |
| 22:58.01 | ManxPower | Then we tell the phone to turn off call waiting |
| 22:58.23 | [TK]D-Fender | ManxPower, Yeah, but your situation is typically considered psycho and to be avoided ;) |
| 22:58.52 | ManxPower | [TK]D-Fender: Maybe so, but I can route calls far more finely then you could ever hope to. |
| 22:59.11 | jameswf | oooh fight fight |
| 22:59.27 | [TK]D-Fender | ManxPower, Nope, I have a fraction of your SIP.CONF entries, a far simpler dialplan :) |
| 22:59.45 | ManxPower | [TK]D-Fender: can you route calls to individual line appearances on each phone? |
| 23:00.08 | [TK]D-Fender | ManxPower, When all roads lead to Rome there's no point in giving them different names, just arrows for the direction to head :) |
| 23:00.28 | ManxPower | So the answer is "no". We have a need to do that. |
| 23:00.44 | ManxPower | You (and many people) do not. In which case, things can be much simplier. |
| 23:01.00 | [TK]D-Fender | ManxPower, Were I to have a person who even NEEDED to have multiple identities on a phoen, yes. Then again, I buy each of my users their own and don't force them to commune with each other like rats in a nest :p |
| 23:01.24 | [TK]D-Fender | ManxPower, that is the point of course. |
| 23:01.43 | [TK]D-Fender | ManxPower, You client is a cheap silly twit :) |
| 23:02.27 | ManxPower | Many of my users don't need multiple identities, but enough do that we had to write the scripts for it. |
| 23:02.58 | ManxPower | For example one secretary needs to have the callerid some from 3 different bosses, depending on which line she picks. |
| 23:03.09 | ManxPower | some == come |
| 23:03.50 | [TK]D-Fender | ManxPower, Nope, she could dial OUT with an exten that would set accordingly. |
| 23:03.53 | Igbothom_III | ManxPower, I was gonna suggest receptionist for a place with multiple clients, but the secretary for multiple bosses also works |
| 23:04.08 | [TK]D-Fender | ManxPower, but that would feel less "natural" |
| 23:06.03 | ManxPower | Our users sometimes have trouble mustering the brain power to remember to breath, anything complex on a phone just makes the scream. |
| 23:06.09 | moa_ | Quick question, if I reload zaptel to bring up another PRI. Will I drop calls? |
| 23:06.20 | *** join/#asterisk hi365_m (n=hi365@213.151.62.64) |
| 23:06.41 | ManxPower | moa_: reloading zaptel won't bring up another PRI (changing signaling, adding/removing channels, etc doesn't work on a reload) |
| 23:06.53 | moa_ | bah |
| 23:07.05 | ManxPower | However, a simple "reload" of "reload chan_zap.so" won't drop calls. |
| 23:07.29 | moa_ | what about init.d/zaptel reload |
| 23:07.41 | *** join/#asterisk atis_home (n=chatzill@193.238.213.215) |
| 23:08.05 | ManxPower | moa_: You should expect that to drop all zap calls on the system. It calls ztcfg to reload /etc/zapata.conf that ztcfg drops all calls |
| 23:08.14 | ManxPower | remember, chan_zap.so is NOT zaptel |
| 23:08.46 | moa_ | Thanks, guess this will have to wait until the maintenance window. |
| 23:08.51 | [TK]D-Fender | moa_, Yes, you will lose your calls. |
| 23:08.52 | *** join/#asterisk MaliutaWrk (i=nikolai@119.11.104.19) |
| 23:10.01 | ManxPower | "All the calls dropped? It must be a telco issue, I'll call in a trouble ticket." |
| 23:10.24 | ManxPower | <-- BPAFH |
| 23:10.31 | angryuser[A] | i have a sipprovider, let's say sip.mysipprovider.com dns request on that adress returns 2 ip's , asterisk take sometimes firs one, sometimes second one, and if he is unable to register * dont try to tegister with another ip adress, any way to fix that problem ? |
| 23:10.33 | moa_ | To bad I'm working with the telco. |
| 23:10.34 | ManxPower | Bastard PBX Admin From Hell. |
| 23:10.42 | ManxPower | We are closely related to BOFHs |
| 23:11.55 | angryuser[A] | and as a result i have sip peers offline |
| 23:12.13 | angryuser[A] | any cheapes fxs box dont have that issue |
| 23:12.51 | angryuser[A] | *cheapest |
| 23:13.54 | Corydon76-lap | ManxPower: have you recently said "I see that you have 100 message slots available in your Voicemail INBOX currently..." |
| 23:14.18 | Corydon76-lap | in response to "I need more message slots..." |
| 23:14.48 | Corydon76-lap | "You mean I have 100 more, now?" "Noooooo..." "Augh!" |
| 23:15.02 | jameswf | Cingular drops allot of calls no one cares |
| 23:17.13 | *** join/#asterisk JT (n=j@unaffiliated/jt) |
| 23:18.28 | ManxPower | Corydon76-lap: Nope. I say "You have 100 messages in your INBOX. You won't be able to get any more messages until you clean out your inbox. |
| 23:19.15 | scooby2 | can you weight individual agents? |
| 23:19.58 | *** join/#asterisk nitram (i=nitram@superblob.com) |
| 23:22.58 | *** join/#asterisk anthm (n=anthm@mbb0736d0.tmodns.net) |
| 23:22.58 | *** mode/#asterisk [+o anthm] by ChanServ |
| 23:25.21 | *** join/#asterisk nitram (i=nitram@superblob.com) |
| 23:26.01 | *** join/#asterisk Robba (n=rob@203.56.181.15) |
| 23:26.21 | Robba | Hi |
| 23:26.40 | Robba | Is anyone using the Linksys SPA-941/942's? |
| 23:27.02 | Igbothom_III | I've used one before but don't like them |
| 23:27.12 | Igbothom_III | do like them better than the Netcom phone, tho |
| 23:27.21 | *** join/#asterisk riddlebox (n=james@75-132-225-75.dhcp.stls.mo.charter.com) |
| 23:27.33 | Robba | do you know anything about line configuration with them? |
| 23:27.38 | pkunkra | if you have to ask about a home based router impacting your VoIP communications, then yes. you need to replace it. it is crap. |
| 23:28.30 | pkunkra | sorry. thought the linksys was a router |
| 23:28.39 | Robba | nah |
| 23:28.40 | Igbothom_III | isn't this the Linksys phone? |
| 23:28.44 | Robba | yeah |
| 23:28.58 | pkunkra | yeah, its a phone |
| 23:29.17 | *** join/#asterisk inadaptado (n=matias@190.3.121.15) |
| 23:31.45 | [TK]D-Fender | Robba, www.voxilla.com <- you you need hints on how to set it up. |
| 23:31.50 | *** join/#asterisk JT (n=j@unaffiliated/jt) |
| 23:31.58 | *** join/#asterisk CrazyTux[m] (n=CrazyTux@ppp-70-248-182-215.dsl.hstntx.swbell.net) |
| 23:32.51 | *** join/#asterisk SwK (n=SwK@user-24-214-55-149.knology.net) |
| 23:33.33 | riddlebox | is there a way I can see a call come into a zap channel or any activity on that channel? |
| 23:33.34 | Robba | TK, its not just setting it up |
| 23:33.46 | Robba | its getting the multiple lines to work |
| 23:33.55 | Robba | one line works fine |
| 23:34.16 | Robba | but one or more and it tends to give 486 busy here responses |
| 23:35.17 | Robba | sorry not one or more |
| 23:35.20 | Robba | two or more |
| 23:36.35 | *** join/#asterisk mchou (n=mchou@c-71-198-127-234.hsd1.ca.comcast.net) |
| 23:37.09 | *** join/#asterisk Absorto (n=user@189.141.94.36) |
| 23:37.25 | Absorto | hello! how can I tell which audio files the meetMe app is playing? |
| 23:37.25 | *** join/#asterisk b1shop (n=b1shop@dsl081-149-253.chi1.dsl.speakeasy.net) |
| 23:37.54 | hi365_m | b listening when it palyes it |
| 23:37.55 | tzafrir | Absorto, core set verbose 3 |
| 23:38.06 | hi365_m | or by watching the cli |
| 23:38.12 | [TK]D-Fender | Robba, describe your line usage on it |
| 23:38.24 | Absorto | thanks tzafrir! |
| 23:38.30 | tzafrir | gee, xchat does not complete asterisk CLI commands :-( |
| 23:38.31 | Absorto | no thanks to you, hi365_m! |
| 23:38.54 | Absorto | no, wait: thank you too :) |
| 23:39.16 | *** part/#asterisk Absorto (n=user@189.141.94.36) |
| 23:49.10 | *** join/#asterisk coppice (n=chatzill@175.203.17.210.dyn.pacific.net.hk) |
| 23:49.44 | *** join/#asterisk adeel (n=adeel@c-24-7-132-23.hsd1.ca.comcast.net) |
| 23:50.46 | adeel | my sip provider doesn't terminate my toll free 8xx calls, can anyone point me to a provider who'll terminate my toll free calls for free? |
| 23:51.34 | J4k3 | ~tollfree |
| 23:51.35 | J4k3 | ~800 |
| 23:52.52 | riddlebox | is there a way I can see a call come into a zap channel or any activity on that channel? |
| 23:58.31 | riddlebox | I have configured a tdm card with 4 fxo ports correctly, the card passes ztcfg -vv I have set the context to demo, and there is a demo context in extensions.conf, but it just rings and rings? |
| 23:59.17 | ManxPower | You would see that sort of stuff on the Asterisk CLI |
| 23:59.41 | riddlebox | ManxPower, you talking to me? |
| 23:59.51 | ManxPower | riddlebox: yes |
