| 00:00.02 | Freman | how can I achive this? |
| 00:00.08 | anonobomber | jpablo: I usually check froogle.com and pricegrabber.com |
| 00:00.15 | *** join/#asterisk Micro (n=john@ptr-66-11-88-185.ptr.terago.ca) |
| 00:00.30 | Freman | I'd like to make it dynamic if possible... EG from mysql, or txt file or something |
| 00:02.14 | *** join/#asterisk glm2k (n=GLM@rrcs-24-199-11-45.west.biz.rr.com) |
| 00:02.19 | Tetraboy[Away] | What's everyone's opinion on SPA-841 vs GXP-2000? |
| 00:04.28 | fafnir | it shouldnt be too hard for asterisk to ring a softfone like 3 times, then forward to a cell phone, right? |
| 00:04.50 | *** join/#asterisk SwK (n=SwK@12-219-144-126.client.mchsi.com) |
| 00:04.53 | JerJer | fafnir: trivial |
| 00:05.04 | fafnir | sweet |
| 00:05.05 | JerJer | just set a timeout - based in seconds versus rings |
| 00:05.44 | *** join/#asterisk roulduke_ (i=3fjiqdpy@p508D1413.dip0.t-ipconnect.de) |
| 00:07.18 | Freman | I'm thinking perl is the way to do what I want... |
| 00:07.27 | Freman | would it be better to use cmd perl, or perl as an agi? |
| 00:14.37 | Equinox | Can one dial vonage or packet8 phones via enum? |
| 00:21.05 | *** join/#asterisk pr0m_ (n=pr0methe@24-75-196-70.chvlva.adelphia.net) |
| 00:23.15 | *** join/#asterisk axscode (n=adfdsas@203.131.153.34) |
| 00:23.47 | Himeko | fafnir you can make it ring both |
| 00:24.11 | Himeko | then just choose what to pick up |
| 00:24.16 | Equinox | Anyone use DUNDi or ENUM? |
| 00:24.35 | *** join/#asterisk hardwire (n=hardwire@209-112-186-141-cdsl-rb1.nwc.acsalaska.net) |
| 00:24.37 | hardwire | yay |
| 00:24.43 | hardwire | snom360 extended keypad kinda rules |
| 00:28.00 | *** join/#asterisk Snake-Eyes (n=blog@203.201.96.101) |
| 00:29.44 | *** join/#asterisk eng1neer (i=1000@adsl-68-94-10-164.dsl.rcsntx.swbell.net) |
| 00:30.26 | axscode | want to ask.. anyone tried using ISDN modem on Asterisk with OpenBSD? |
| 00:30.39 | puzzled | anyone know if there are issues with using gcc4 for building asterisk? |
| 00:30.58 | *** join/#asterisk kmj (n=kmj@69-168-15-141.chvlva.adelphia.net) |
| 00:31.25 | anonobomber | puzzled: I would check the Fedora Core mailing list since FC4 shipped with gcc4 |
| 00:31.41 | puzzled | anonobomber: I would if they had a clue about asterisk :) |
| 00:32.35 | *** join/#asterisk da_monumental_1 (n=da_monum@rrcs-24-172-102-198.midsouth.biz.rr.com) |
| 00:32.51 | axscode | what distribution of linux thus asterisk works well? |
| 00:33.24 | eng1neer | how do U get asterisk to send a 302 msg to my sip provider so that I can get them (to preserve the caller ID info) to forward to another number |
| 00:33.30 | anonobomber | axscode: works fine on Debian, Fedora, RHEL (cent os too), gentoo |
| 00:33.46 | axscode | hmmm ok.. Debian |
| 00:34.22 | JerJer | eng1neer: large hammer |
| 00:34.33 | eng1neer | is there not a way to do it? |
| 00:35.14 | ctooley | make[1]: *** No rule to make target `chan_iax2.o', needed by `chan_iax2.so'. Stop. |
| 00:35.14 | ctooley | make[1]: Leaving directory `/usr/src/asterisk/channels' |
| 00:35.14 | ctooley | make: *** [subdirs] Error 1 |
| 00:35.22 | ctooley | sorry, wrong window |
| 00:35.26 | eng1neer | I can forward THRU my asterisk server but I lose the caller id as they dont "trust me" |
| 00:35.33 | ctooley | uhh... can someone point to the problem though? |
| 00:36.12 | *** join/#asterisk websae (n=websae@207-118-143-134.dyn.centurytel.net) |
| 00:36.54 | JunK-Y | ctooley: make clean && make install |
| 00:37.19 | axscode | what happened to www.debian.org ? |
| 00:37.40 | ctooley | JunK-Y did that, that's how I got the compile problem |
| 00:37.59 | JunK-Y | u did a cvs update -d too? |
| 00:38.14 | ctooley | nope |
| 00:38.24 | JunK-Y | try that. |
| 00:38.36 | ctooley | nope |
| 00:39.22 | websae | anyone here using FREEBSD ? |
| 00:39.29 | kmj | yes |
| 00:39.34 | websae | you are kmj? |
| 00:39.39 | kmj | yes |
| 00:39.41 | ai-a | can i use a standard serial modem with asterisk so i can connect to a telephony switch via an analogue line? |
| 00:40.15 | ctooley | ai-a: not if you want to use that modem for phone calls, no. |
| 00:40.40 | ai-a | it will ring, and i want asterisk to answer it. |
| 00:41.16 | ctooley | to do data? |
| 00:41.30 | ctooley | because pppd will do that. |
| 00:41.39 | ctooley | errr mgetty and pppd |
| 00:41.48 | ai-a | no analogue calls. |
| 00:41.54 | ctooley | voice calls? |
| 00:42.00 | ai-a | yes voice. |
| 00:42.05 | ctooley | nope |
| 00:42.10 | ai-a | the modem rings,, asterisk answers it and directs to a specific ext. |
| 00:42.19 | ai-a | so i should write a modem -> iax or something program. |
| 00:44.40 | *** join/#asterisk The_Ball (n=alex@static-227.35.240.220.dsl.comindico.com.au) |
| 00:45.07 | CoaxD | asterisk doesnt speak modem |
| 00:45.15 | *** join/#asterisk pr0m_ (n=pr0methe@24-75-196-70.chvlva.adelphia.net) |
| 00:46.09 | JunK-Y | ctooley: nope? |
| 00:48.13 | ctooley | JunK-Y can't do a cvs update |
| 00:48.54 | axscode | y? |
| 00:49.33 | ctooley | because our app won't work with current CVS HEAD IAX2 is too broken. |
| 00:50.00 | *** join/#asterisk shift_ (i=shift@82-39-34-36.cable.ubr01.benw.blueyonder.co.uk) |
| 00:52.18 | hardwire | blah |
| 00:52.43 | bkw__ | grrr |
| 00:52.49 | bkw__ | this fucker don't see /dev/hda3 |
| 00:52.55 | bkw__ | udev sucks yo |
| 00:53.08 | hardwire | udev is an odd little thing |
| 00:53.12 | hardwire | what distro? |
| 00:53.25 | bkw__ | gentoo livecd |
| 00:53.38 | hardwire | well |
| 00:53.39 | *** join/#asterisk bin13 (n=bin33@dsl-38-117-145-134.pilosoft.com) |
| 00:53.39 | hardwire | nm then |
| 00:53.55 | bkw__ | fun fun fun |
| 00:53.58 | bkw__ | I have never seen such a thing |
| 00:54.02 | hardwire | I spent all day making building plan views of the extensions list |
| 00:54.13 | hardwire | everybody went oooooh... thats much easier to read |
| 00:54.16 | hardwire | I wish I did this 2 days ago |
| 00:55.06 | hardwire | I got mass snom paging down like a fine art. |
| 00:55.47 | hardwire | but I think I need to steal some MoH style mentality and use it vs called playback or calling meetmes |
| 00:55.55 | hardwire | make something more independant w/ less overhead |
| 00:56.10 | *** join/#asterisk adker (n=adker@67-136-213-101.dsl1.glv.ny.frontiernet.net) |
| 00:57.02 | JunK-Y | where's my damn mic! |
| 00:57.24 | *** part/#asterisk adjacent (n=scott@cpe-024-168-216-235.sc.res.rr.com) |
| 01:02.49 | *** join/#asterisk Spike_76 (n=it@69.61.164.126) |
| 01:06.38 | *** part/#asterisk bin13 (n=bin33@dsl-38-117-145-134.pilosoft.com) |
| 01:09.04 | *** join/#asterisk voipheck (n=yakumo@203.87.204.126) |
| 01:10.25 | *** join/#asterisk EvilMetal (n=StorM@213.161.223.40) |
| 01:13.29 | edgecase | So this Steve Underwood character is doing T.38. someone should hook him up with the bounty |
| 01:14.58 | Sedorox | hmm |
| 01:15.41 | *** join/#asterisk Snake-Eyes (n=blog@203.201.96.101) |
| 01:16.43 | glm2k | hmm, he's too bloody practical for that methinks |
| 01:17.29 | *** join/#asterisk apardo (n=w0w0@239.Red-83-46-187.pooles.rima-tde.net) |
| 01:23.33 | *** join/#asterisk jtodd (n=jtodd@dsl027-191-178.sfo1.dsl.speakeasy.net) |
| 01:25.58 | JunK-Y | j'aime pas ta soupe! |
| 01:26.18 | Tetraboy[Away] | What's a good store to buy an ip phone from? |
| 01:27.20 | websae | too bad bestbuy doesn't sell IP phones |
| 01:27.35 | Tetraboy[Away] | Wish newegg did. |
| 01:27.47 | Sedorox | voipsupply.com |
| 01:27.50 | kFuQ | they'd be hella overpriced like all their stuff |
| 01:27.50 | Sedorox | I believe |
| 01:28.02 | kFuQ | Sedorox: yah |
| 01:28.26 | kFuQ | find decent deals on ebay once in a while |
| 01:29.49 | websae | anyone here using FREEBSD on their machine? |
| 01:30.47 | hardwire | sneed to add pickupgroup to queues |
| 01:30.58 | hardwire | I should find a way to do that with pseudo channels |
| 01:31.05 | *** join/#asterisk Sgt_Stedenko (n=piespy@c-67-185-117-105.hsd1.wa.comcast.net) |
| 01:31.06 | Tetraboy[Away] | I've got a pc running freebsd. |
| 01:31.49 | kFuQ | got some here too |
| 01:32.45 | [Airwolf] | Does anyone have a init script for asterisk laying around ? (gentoo) |
| 01:32.45 | *** join/#asterisk Sgt_Stedenko (n=piespy@c-67-185-117-105.hsd1.wa.comcast.net) |
| 01:33.03 | *** join/#asterisk ZX81 (n=ZX81@222-153-80-230.jetstream.xtra.co.nz) |
| 01:33.09 | nick125 | [Airwolf]: i think theres one in the ebuild for it |
| 01:33.42 | *** part/#asterisk ZX81 (n=ZX81@222-153-80-230.jetstream.xtra.co.nz) |
| 01:34.26 | [Airwolf] | nick125, yep I know there is. But I used the cvs for my installation |
| 01:34.46 | nick125 | maybe you could grab the tarball used in the ebuild and find the init script? |
| 01:35.18 | *** join/#asterisk gopherspidey (n=webb@dsl-musc-15-084.machlink.com) |
| 01:35.18 | bkw__ | Qwell, dear those people in #gentoo won't answer you |
| 01:35.30 | *** join/#asterisk chendy (n=Alex_Dot@218.1.218.17) |
| 01:35.31 | Qwell | bkw_: I know. Find me another high volume channel though, where 90% of the users run Linux :p |
| 01:35.53 | [Airwolf] | nick125, I would like to do that, but I have no idea where to find that tarball |
| 01:36.09 | nick125 | [Airwolf]: look at distfiles.gentoo.org, and find the asterisk tarball there |
| 01:36.18 | [Airwolf] | nick125, tnx |
| 01:36.22 | bkw__ | Qwell, hehe |
| 01:36.22 | nick125 | sure |
| 01:36.25 | gopherspidey | Is anyone on the channel that is working on the Manager proxy idea? |
| 01:36.29 | [Airwolf] | Qwell, what problem do you have ? |
| 01:36.44 | bkw__ | Qwell, do a whois and findout who is on from there |
| 01:36.45 | Qwell | [Airwolf]: one that I need an Adelphia customer in southern CA to verify |
| 01:36.50 | Qwell | bkw__: yeah... |
| 01:37.07 | [Airwolf] | Qwell, can't help you with that ;) |
| 01:37.40 | nick125 | i cant help with that either :P |
| 01:38.18 | nick125 | maybe #adelphiainsca could help |
| 01:39.09 | [Airwolf] | hehe |
| 01:40.24 | [Airwolf] | nick125, good choice, you don't want to hear me sing |
| 01:43.30 | [Airwolf] | Qwell, are you running Gentoo at the moment ? |
| 01:43.36 | Qwell | [Airwolf]: I am |
| 01:43.42 | [Airwolf] | with asterisk ? |
| 01:43.45 | Qwell | [Airwolf]: I am |
| 01:43.57 | [Airwolf] | Can you mail me your /etc/init.d/asterisk ? |
| 01:44.10 | Qwell | isn't it in the contrib/ dir? |
| 01:44.33 | [Airwolf] | ehh, last time I looked it wasn't |
| 01:44.38 | nick125 | http://ffd.nick125.com/asterisk |
| 01:44.43 | Qwell | /usr/cvsroot/asterisk/contrib/init.d/rc.gentoo.asterisk |
| 01:44.54 | [Airwolf] | hmm, cool |
| 01:44.57 | stkn | or /usr/portage/net-misc/asterisk/files/asterisk.rc6* |
| 01:49.00 | [Airwolf] | Im really wondering how asterisk is going to preform on a Pentium I 233 |
| 01:50.51 | Qwell | [Airwolf]: poorly. how else? |
| 01:51.11 | *** join/#asterisk shmaltz (n=chatzill@69.28.255.210) |
| 01:51.40 | shmaltz | anybody know how to decipher Zap Show status? |
| 01:53.06 | JunK-Y | shmaltz:??? |
| 01:53.16 | shmaltz | Yep JunK-Y |
| 01:53.29 | *** join/#asterisk Insanity5 (n=feaw@ip68-105-209-13.tl.dl.cox.net) |
| 01:53.32 | shmaltz | I get an output I would like to know what it means? |
| 01:53.45 | Insanity5 | What significance does the realm have, and do I ever need to change it from the default of asterisk? |
| 01:54.04 | JunK-Y | shmaltz: the only thing u should really care, is the Alarms field. |
| 01:54.14 | shmaltz | Thsi is what I get: |
| 01:54.27 | shmaltz | http://pastebin.com/345596 |
| 01:54.42 | shmaltz | But I have no clue what alarm recover means |
| 01:54.45 | JunK-Y | REC? |
| 01:57.57 | shmaltz | Yep, JunK-Y |
| 01:58.21 | shmaltz | what happens if a TE411 is plugged into a 5 Volt slot? |
| 01:58.32 | tzanger | shmaltz: not possible |
| 01:58.33 | tzanger | it's keyed |
| 01:58.43 | DarthClue | yeah, keyed. |
| 01:58.43 | shmaltz | oh, you right |
| 01:58.51 | Hogie | tzanger: maybe he filed it down? |
| 01:58.54 | Hogie | and made it own key? |
| 01:58.55 | Hogie | :P |
| 01:58.58 | DarthClue | but if you break the key, well, um, smoke? |
| 01:59.02 | DarthClue | fire? |
| 01:59.03 | DarthClue | water? |
| 01:59.05 | DarthClue | burn? |
| 01:59.05 | tzanger | Hogie: like this: http://www.mixdown.ca/~asterisk/ |
| 01:59.09 | shmaltz | so why does my zap show status show that it's a TE410 when it's a TE406? |
| 01:59.20 | tzanger | DarthClue: it will likely cease to function |
| 01:59.35 | *** join/#asterisk tengulre (n=tengulre@61.185.238.166) |
| 01:59.49 | DarthClue | shmaltz: talk to digium support. sounds like a driver issue of some sort. |
| 01:59.58 | DarthClue | tzanger: yeah, that was my point. |
| 02:00.31 | *** join/#asterisk rvhi (n=rv@66.175.65.89) |
| 02:01.28 | *** join/#asterisk postwait (n=jesus@pcp0011335392pcs.elkrdg01.md.comcast.net) |
| 02:02.06 | tengulre | which IRC tools that runing under console? |
| 02:02.40 | crash3m | bitchx |
| 02:02.53 | *** join/#asterisk surfdue (n=surfdue@unaffiliated/surfdue) |
| 02:06.42 | *** join/#asterisk J[SS]_ (i=ph33r@smartserv.ipv6.smart-serv.net) |
| 02:07.48 | *** join/#asterisk Defraz (n=t0tal@tim.ibccom.net) |
| 02:08.22 | *** join/#asterisk Insanity5 (n=feaw@ip68-105-209-13.tl.dl.cox.net) |
| 02:08.25 | Insanity5 | As far as my asterisk box is concerned... what's a registration server as compared to a proxy? In xlite, all I put in was proxy and it worked great. |
| 02:08.43 | Nivex | I envisioned the A@H ISO being a lot smaller |
| 02:09.28 | MicC_ | http://www.mixdown.ca/~asterisk/ |
| 02:09.36 | MicC_ | Nivex: its a RH clone |
| 02:09.36 | MicC_ | :P |
| 02:09.42 | MicC_ | kernel is like 30meg alone |
| 02:10.00 | MicC_ | Mysqld, perl, perl modules...etc. |
| 02:10.10 | Nivex | *shudder* |
| 02:10.13 | MicC_ | yeap |
| 02:10.19 | MicC_ | shoulda started with Gentoo |
| 02:10.19 | MicC_ | :P |
| 02:10.39 | nick125 | lol |
| 02:10.42 | *** part/#asterisk edgecase (n=jjackson@ragnarok.coplanar.net) |
| 02:11.11 | *** join/#asterisk Netgeeks (n=Chris@68-185-24-2.static.mdfd.or.charter.com) |
| 02:11.11 | *** join/#asterisk chendy_ (n=Alex_Dot@web1.ningo.net) |
| 02:12.22 | *** join/#asterisk tengulre (n=root@61.185.238.166) |
| 02:12.56 | [Airwolf] | I'm beginning to hate x-lite |
| 02:13.11 | fugitivo | is stanphone having problems? |
| 02:13.28 | [Airwolf] | fugitivo, who or what is stanphone ? |
| 02:14.32 | fugitivo | stanaphone |
| 02:14.34 | fugitivo | sorry |
| 02:15.59 | glm2k | stanaphone says i'm registered |
| 02:16.10 | glm2k | that much i know |
| 02:16.11 | bkw__ | stananphone? |
| 02:16.12 | bkw__ | wtf is that |
| 02:16.24 | *** join/#asterisk zox (n=zox@ip70-176-64-134.ph.ph.cox.net) |
| 02:16.34 | glm2k | basically provides did's in the east coast |
| 02:16.38 | glm2k | sip |
| 02:17.16 | *** join/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 02:17.49 | *** join/#asterisk vuvie (n=vuvie@bb219-74-44-131.singnet.com.sg) |
| 02:19.32 | fugitivo | i'm registered too, but too laggy |
| 02:19.49 | *** part/#asterisk vuvie (n=vuvie@bb219-74-44-131.singnet.com.sg) |
| 02:20.05 | *** join/#asterisk CdtDelta (n=CdtDelta@dsl081-225-161.chi1.dsl.speakeasy.net) |
| 02:21.04 | *** join/#asterisk Lex (n=icechat5@62.249.217.145) |
| 02:21.19 | *** join/#asterisk Eminence (n=achin@cpe-24-198-66-186.maine.res.rr.com) |
| 02:21.55 | Lex | Looking for advice for my first Asterisk setup. I've been running *@H for a little while. |
| 02:22.30 | JerJer | im sorry |
| 02:22.51 | Lex | Which distro would anyone recomend : CentOS (*@H is on 3.5) or Fedora core |
| 02:22.58 | Lex | Thakns./ |
| 02:23.11 | JerJer | slack |
| 02:23.34 | Lex | slackware? |
| 02:25.25 | *** join/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 02:26.08 | *** join/#asterisk SplasPood (n=jwb@brooklyn.paravolve.net) |
| 02:27.30 | *** join/#asterisk drumkilla (n=russell@12.21.243.167) |
| 02:27.50 | Lex | JerJer were you talking to me - sorry if I interupted you. |
| 02:28.19 | *** join/#asterisk techie (n=gus@70.86.57.50) |
| 02:28.27 | fugitivo | Lex: please, don't ask that question :) |
| 02:28.40 | *** join/#asterisk Moc_ (n=mochouin@131-199-0-72-ppp.3menatwork.com) |
| 02:28.52 | DarthClue | Lex: It all depends on what you are comfortable with. I use Fedora, some people use gentoo, others use debian, centos, freebsd, etc. |
| 02:28.54 | Moc_ | Hail |
| 02:29.09 | DarthClue | no, no. It's HELL! |
| 02:29.19 | Lex | Sorry fugitivo. I couldn't see it in the FAQ?!?!?!!? |
| 02:29.55 | Lex | I'm running away from windows :-d |
| 02:30.06 | fugitivo | Lex: Darthclue's answer is the best |
| 02:30.42 | DarthClue | no. My answer is the most PC capable answer I can give without killing myself. |
| 02:30.52 | fugitivo | lol |
| 02:31.13 | Lex | I'm not familiar with any of them really - I used Unix when it was unix (SCO). |
| 02:31.23 | DarthClue | Lex: anything but windows. be comfortable with what you want to use. If you use something that isn't 'mainstream' then you most likely will be on your own. |
| 02:31.51 | DarthClue | Lex: the ASSware is based on RH, so either Fedora or CentOS will be the most familiar. |
| 02:32.13 | Lex | Thanks - I know that * is built on FC1 and I run *@H on centos 3.5. |
| 02:33.03 | *** join/#asterisk ptiggerdine (n=ptiggerd@c210-49-98-194.rochd1.qld.optusnet.com.au) |
| 02:33.09 | ptiggerdine | SCO suck! |
| 02:33.17 | Lex | OK - appologies for asking a stupid question::$ |
| 02:33.20 | glm2k | back then, everything sucked |
| 02:33.30 | ptiggerdine | Lex, dumb looks are still free.. |
| 02:33.32 | DarthClue | Lex: not stupid, just political. |
| 02:33.51 | Lex | 'night darth |
| 02:34.15 | ptiggerdine | DarthClue, is the a colorful youthism that you youth ppl call it? |
| 02:34.16 | Lex | OK. will pick FC3. |
| 02:34.58 | Lex | any probs testing fc3 and *1.2 |
| 02:35.05 | Lex | sorry people! |
| 02:37.09 | Lex | OK. Thanks glm. Happy birthday(^) |
| 02:37.18 | glm2k | lol |
| 02:38.17 | Lex | 'night all. |
| 02:38.43 | *** join/#asterisk docelmo (n=me@149-87.119-70.tampabay.res.rr.com) |
| 02:39.49 | *** part/#asterisk postwait (n=jesus@pcp0011335392pcs.elkrdg01.md.comcast.net) |
| 02:40.06 | hardwire | ok |
| 02:40.10 | hardwire | I have snom indication via Queues |
| 02:40.11 | hardwire | err |
| 02:40.12 | hardwire | for Queues |
| 02:40.19 | hardwire | mapped to PickUp() |
| 02:40.20 | hardwire | this rules |
| 02:40.23 | hardwire | it even blinks |
| 02:40.34 | hardwire | Manager interface kinda rocks my balls |
| 02:40.51 | hardwire | blinking spikes the server.. so I am gonna go with solid :) |
| 02:40.59 | bkw__ | you mean sucks balls? |
| 02:41.15 | hardwire | still not a failing statement. |
| 02:41.28 | *** join/#asterisk drumkilla (n=russell@12.21.243.167) |
| 02:42.01 | *** mode/#asterisk [+o drumkilla] by ChanServ |
| 02:44.07 | *** join/#asterisk hellagony (n=hellagon@200.121.160.34) |
| 02:48.44 | *** join/#asterisk ttyp0 (n=ttyp0@204.Red-83-50-39.pooles.rima-tde.net) |
| 02:50.13 | *** join/#asterisk Sedorox (i=brandon@smartserv/cna/Sedorox) |
| 02:50.40 | [Airwolf] | everyone to bed I think ? |
| 02:51.33 | *** join/#asterisk DIEMACO (i=diemaco@unaffiliated/diemaco) |
| 02:52.07 | *** join/#asterisk Insanity5 (n=feaw@ip68-105-209-13.tl.dl.cox.net) |
| 02:52.12 | fugitivo | no |
| 02:52.15 | fugitivo | but yes |
| 02:52.26 | fugitivo | i'm going to sleep now |
| 02:52.29 | fugitivo | :) |
| 02:52.44 | Insanity5 | Is it possible to transfer a land line to a voip provider to obtain a DID in an area they don't have? Or do they have to have something special / a PRI in that area? |
| 02:55.50 | websae | i have 4 voip phones in the office that will be connected to 2 voice lines from SBC, and using sbc yahoo dsl as my broadband..and i'll have 2 more voip (one at home) and one at another location..what type of transcoding will i need if any? |
| 02:56.18 | *** join/#asterisk litage (n=nick@203.201.96.101) |
| 02:57.06 | [Airwolf] | websae, depends on the bandwith you have |
| 02:57.47 | *** join/#asterisk santiago (n=santiago@63.245.86.163) |
| 02:57.59 | websae | not much |
| 02:58.08 | websae | about 1.5mbs...the sbc yahoo dsl basic |
| 02:58.51 | Moc_ | websae, always calculate on the lowest bandwidth value you have.. |
| 02:59.06 | websae | say 80kps? |
| 02:59.13 | websae | or 90kps? |
| 02:59.14 | websae | like that? |
| 02:59.14 | Moc_ | 1.5download and 640kbits upload, it basicly mean 640 |
| 02:59.54 | Moc_ | ulaw is 82kbits/sec |
| 02:59.58 | Moc_ | with overhead (about that) |
| 03:00.14 | websae | what do you suggest then? |
| 03:00.24 | Moc_ | Im not sugesting anything... |
| 03:00.40 | Moc_ | just giving you info |
| 03:00.47 | websae | have 2 voip phones outside the office that i want to connect to the office(office using sbc yahoo dsl) |
| 03:00.54 | websae | what transcoding would you recommend? |
| 03:01.05 | Moc_ | 640/82= ? ;) |
| 03:01.59 | [Airwolf] | ~8 ? |
| 03:02.25 | tengulre | anybody runing asterisk with a voice modem? |
| 03:03.03 | ManxPower | websae: what codecs do the phones support? |
| 03:04.45 | *** join/#asterisk Insanity5 (n=feaw@ip68-105-209-13.tl.dl.cox.net) |
| 03:04.50 | Insanity5 | Does asterisk log anything by default? |
| 03:05.15 | JunK-Y | Insanity5: see logger.conf |
| 03:05.56 | ManxPower | websae: if you private /msg me again you'll be put on /ignore. |
| 03:06.20 | Qwell | ManxPower: so...remember that log from this morning? |
| 03:06.26 | JerJer | .msg ManxPower ignore me!!! |
| 03:06.26 | *** join/#asterisk rogercharlie (n=charlier@c-69-181-35-83.hsd1.ca.comcast.net) |
| 03:06.27 | JerJer | oh |
| 03:06.31 | Qwell | ManxPower: Adelphia wants me to send them one email per IP. |
| 03:06.38 | ManxPower | Qwell: Ah! |
| 03:07.04 | Qwell | I about laughed myself into a coma when the guy told me that |
| 03:07.15 | ManxPower | JerJer: Naw, you are on the "Only put on /ignore if he's ever actually verbally abusive to me." list. |
| 03:07.37 | puowvip | so do I spend $7k on a vacation or don't I |
| 03:07.51 | ManxPower | puowvip: I did this summer. It was worth it. |
| 03:08.23 | puowvip | manx: only problem is I already spent 8500 on a freagin roof |
| 03:08.36 | puowvip | I don't want to spend more than I earn this year. that would be Bad(tm). |
| 03:08.37 | ManxPower | JerJer: Most people are on the "Put on ignore if they piss me off." list. |
| 03:08.54 | *** join/#asterisk Moc__ (n=mochouin@94-199-0-72-ppp.3menatwork.com) |
| 03:09.52 | *** join/#asterisk Atamido (n=Atamido@user-0ccsqgh.cable.mindspring.com) |
| 03:10.57 | *** join/#asterisk mago (n=maxgluck@200.109.166.172) |
| 03:11.02 | Atamido | hey all |
| 03:11.03 | Insanity5 | Anyone ever used/unlocked a belkin message manager? |
| 03:11.15 | Nugget | Qwell: going to write up the perl script to accomodate their request? :) |
| 03:11.15 | MikeJ[Laptop] | no, no one |
| 03:11.34 | Insanity5 | MikeJ[Laptop] - I had it working on stanaphone once, can't get it to work on my asterisk box :( |
| 03:11.57 | bkw__ | Qwell, what is the problelm? |
| 03:12.09 | bkw__ | how many IP's |
| 03:12.33 | puowvip | pajamatime |
| 03:12.34 | puowvip | ninight |
| 03:13.05 | mago | anyone know if appradius runs on cvs-head? |
| 03:13.09 | Qwell | bkw_: tens of thousands - arp requests |
| 03:13.19 | Qwell | Nugget: I think I am. |
| 03:13.28 | mago | knows |
| 03:13.38 | websae | i am getting a Sipura 481 VOIP phone trying to see if it supports ULAW...can't tell anyone know? |
| 03:13.44 | JerJer | mago: you don't want to know |
| 03:14.01 | Moc__ | websae, all voip phone I know off support ulaw.. |
| 03:14.13 | Moc__ | it the defecto standards for non compress voip call |
| 03:14.48 | websae | how do you trascode? |
| 03:15.00 | Moc__ | websae, why would you transcode ? |
| 03:15.11 | Moc__ | just stay ulaw.. |
| 03:15.18 | mago | Jer: I need to... already have a billing/radius server and appradius is the best I can think of |
| 03:15.46 | websae | so with that dsl connection and those 2 outside voip phones shoulud be fine? |
| 03:16.14 | Moc__ | websae... 640/82=7 .... |
| 03:16.23 | nick125 | it should be fine, yes |
| 03:16.25 | Moc__ | so 7-2 = 5 |
| 03:16.37 | websae | can get 5 more lines |
| 03:16.45 | websae | 5 mor voip that means? |
| 03:16.47 | Moc__ | so as long this calcul is positive you are OK ;) |
| 03:18.45 | websae | what ports does asterisk use? |
| 03:18.48 | *** join/#asterisk Insanity5 (n=feaw@ip68-105-209-13.tl.dl.cox.net) |
| 03:19.01 | nick125 | 10,000-20,000 for sip |
| 03:19.11 | nick125 | and another port i dont remember off hand |
| 03:19.14 | websae | like for asterlink? |
| 03:19.23 | nick125 | (10000-20000 udp) |
| 03:19.31 | *** join/#asterisk r0m (n=SysOp@a81-84-68-89.cpe.netcabo.pt) |
| 03:19.32 | ManxPower | websae: every phone supports ulaw and alaw |
| 03:19.48 | websae | what is alaw, a lower quality? |
| 03:19.57 | r0m | good morning |
| 03:19.59 | Insanity5 | I am trying to debug a registration that is failing and it keeps saying 403 forbidden. |
| 03:20.16 | Insanity5 | How can I dig deeper. I'm running sip debug, ut I really don't see anything useful. |
| 03:20.33 | *** join/#asterisk Kumbang (n=unknown@167.205.24.5) |
| 03:20.47 | ManxPower | websae: find yourself a telecom glossary. Read it. Rinse. Repeat. |
| 03:20.51 | Moc__ | websae depend of channel driver |
| 03:21.23 | ManxPower | alaw is what most places in the world except USA and Canada use instead of ulaw |
| 03:21.36 | brettnem | hey, anyone know how to do a pri debug with a sangoma card? |
| 03:21.54 | rogercharlie | does the export pdf work for anyone with the areski cdr? |
| 03:22.47 | *** join/#asterisk harryvv (n=none@S010600a0c93f6f7e.vs.shawcable.net) |
| 03:23.16 | Insanity5 | Can anyone help me debug why a sip device is not connecting? I am watching sip debug from the console. |
| 03:23.21 | Insanity5 | I need help interpreting what it means. |
| 03:23.39 | Insanity5 | In a nuteshell, its sending back 403 forbidden... but I don't know why. |
| 03:24.10 | rogercharlie | the context doesn't have the number dialed |
| 03:24.11 | ManxPower | Insanity5: what is the specific Asterisk console registration refused message? Paste the 1 line message. |
| 03:24.18 | *** join/#asterisk Lex (n=icechat5@62.249.217.145) |
| 03:24.31 | ManxPower | Ah, not connecting. So yo are not having registration issues? |
| 03:24.52 | ManxPower | If that's the case, put the entry for that SIP device from sip.conf on pastebin.ca, sans password |
| 03:25.33 | Insanity5 | Aug 24 23:27:06 NOTICE[23854]: chan_sip.c:7733 handle_request: Registration from '<sip:2084444444@sip.cainkar.com>' failed for '68.105.209.13' |
| 03:26.00 | ManxPower | Insanity5: And you have a [2084444444] section of sip.conf with a username=2084444444 ? |
| 03:26.25 | Insanity5 | So 2084444444 is the username its sending? It should be xlite1 |
| 03:27.02 | ManxPower | yes. don't worry about it, make it work with my suggested settings. |
| 03:27.15 | ManxPower | once it works, you are on your own as to why it's not sending the username you expect. |
| 03:27.22 | Moc__ | damn I am tired.. |
| 03:27.38 | Insanity5 | that number is set as "sip set tel_num" on the console of the POS. |
| 03:27.52 | Insanity5 | I figured to "sip set reg_id" would be the username... right next to "sip set reg_pwd" :P |
| 03:28.39 | Insanity5 | Ok restarting POS router/voip thingy again. |
| 03:28.45 | Insanity5 | brb |
| 03:30.41 | brookshire | i can give you a fake huntsville address too if she doesn't get anything |
| 03:30.45 | brookshire | oops |
| 03:30.47 | brookshire | :D |
| 03:34.31 | drumkilla | brookshire: !! |
| 03:38.41 | harryvv | off subject |
| 03:38.42 | harryvv | I want a toyota diesel :) |
| 03:41.01 | websae | what do VOIP gateways do? |
| 03:41.21 | Qwell | they open and close ways, for voip |
| 03:43.13 | Nugget | they keep bkw from drunk-dialing you. |
| 03:43.53 | nick125 | debian is offically gonna kill me off |
| 03:44.23 | websae | im afraid my compilatation of asterisk on my FreeBSD machine has stalled and is not goign anywhere..hmm |
| 03:44.59 | *** join/#asterisk FarrisG (n=jrush@h-68-164-19-170.dllatx37.covad.net) |
| 03:46.02 | *** join/#asterisk juice (n=juice@mo-67-77-177-215.dyn.sprint-hsd.net) |
| 03:49.24 | *** join/#asterisk IronHelix (n=irc@ool-45785cfe.dyn.optonline.net) |
| 03:50.56 | websae | how do i restart a compile of asterick on my freebsd machine...it seem to have stalled..anyone know? |
| 03:51.05 | Nugget | make clean and then make again. |
| 03:51.12 | Nugget | you building by hand or from ports? |
| 03:51.19 | websae | after it's done compiling do i do a make clean? |
| 03:51.29 | websae | ports |
| 03:51.30 | Nugget | if it's done compiling, why do you want to do it over? |
| 03:51.40 | websae | just unstalled nevermind |
| 03:51.41 | websae | lol |
| 03:51.44 | websae | slow machine i guess |
| 03:51.44 | Nugget | heh |
| 03:51.48 | websae | you running freebsd? |
| 03:51.57 | Nugget | one of my asterisk servers is freebsd, yeah. |
| 03:53.28 | *** join/#asterisk xcrypto (n=xcrypto@adsl-068-209-115-147.sip.asm.bellsouth.net) |
| 03:53.45 | *** join/#asterisk bmg505 (n=leon@rndf-146-27-21.telkomadsl.co.za) |
| 03:55.22 | websae | has anyone found asterisk to be reliable and well worth the time investment in setting it up? |
| 03:55.26 | *** join/#asterisk [hC] (n=hardcore@c-66-176-181-76.hsd1.fl.comcast.net) |
| 03:55.41 | [hC] | Is there a way to play a sound after the dialed party answers the phone? |
| 03:56.21 | Qwell | websae: a few hours to set it up, to save tens of thousands of dollars... |
| 03:56.23 | [hC] | I have some 7960's set up to auto answer on a second line (for intercom) and i want to play a beep when they first answer. |
| 03:56.24 | Qwell | nah |
| 03:57.04 | Nugget | my main complaint with asterisk is that all the development takes place in the CVS HEAD version. For any sort of worthwhile asterisk use you pretty much have to use the bleeding-edge dev version. |
| 03:57.19 | Qwell | Nugget: 1.2 |
| 03:57.24 | Nugget | there doesn't seem to be anything more than lip service paid to the stable branch or cutting releases on a reliable schedule |
| 03:57.56 | Nugget | I don't think I'd have had any luck at all with asterisk if it weren't for my ability to spend time in here sort of keeping my finger on the pulse of where things were. |
| 03:58.03 | Nugget | it's been a real time commitment |
| 03:58.44 | Nugget | as ManxPower is quick to point out, relying on voice over internet is way more trouble than its worth in any but the most optimal circumstances. |
| 03:59.35 | *** join/#asterisk insomni (n=insomni@x1-6-00-e0-18-6f-34-ad.k455.webspeed.dk) |
| 03:59.53 | *** part/#asterisk twisted (n=twisted@asterisk/friend-and-developer/pdpc.professional.twisted) |
| 04:00.00 | *** join/#asterisk MikeJ[Laptop] (n=ircatjer@d14-69-8-30.try.wideopenwest.com) |
| 04:00.11 | Nugget | asterisk is immature and you shoudl expect to twiddle with it and face a moderate number of annoyances and quirks if you deploy it. |
| 04:00.47 | Nugget | with sufficient planning and work you can build a stable platform, but you'll face conflicting needs being torn between stability and the need to keep upgrading as asterisk improves and matures |
| 04:00.50 | hardwire | what causes the sip notify subscriptions to devalidate when extensions.conf is reloaded? |
| 04:00.58 | hardwire | there aught to be a way to retain all that |
| 04:01.18 | websae | what about this anyone..well i want to set it up like this.....at my parents office want to have 4 VOIP phones that will use the 2 voice lines from SBC...then use my bsd box with asterix with a digium 2 FXO card and then two VOIP phones (one at home to use with office 2 lines) and one in another remote location....will that work? |
| 04:01.55 | *** join/#asterisk twisted (n=twisted@asterisk/friend-and-developer/pdpc.professional.twisted) |
| 04:01.55 | *** mode/#asterisk [+o twisted] by ChanServ |
| 04:02.33 | Qwell | websae: sure |
| 04:02.53 | Nugget | No. Not with bsd. |
| 04:03.00 | Nugget | use linux if you want zaptel cards. |
| 04:03.06 | Qwell | oh, right |
| 04:03.19 | websae | zaptel doesn't work with bsd? |
| 04:03.24 | Qwell | not well...or something |
| 04:03.25 | Nugget | and I'd urge you to use external sipuras instead of zaptel. it'll be bulkier but more stable. |
| 04:03.35 | Nugget | zaptel in freebsd is alpha-quality |
| 04:03.39 | [hC] | Hmm. has anyone found a way to play a short sound after a call is answered? I am trying to implement a notification sound for intercoms |
| 04:04.01 | websae | in the mean time i am going to asterlink with freebsd, that should work fine nugget right? |
| 04:04.04 | *** join/#asterisk Tili (i=Tili@218.19.4.56) |
| 04:04.23 | Nugget | as long as you don't need meetme conferencing or hardware drivers for zaptel, sure. |
| 04:04.40 | Nugget | I'd suggest skipping the port entirely and installing cvs stable or cvs head, too. |
| 04:05.04 | Nugget | it'll compile find by hand without any convincing. just use gmake instead of make and it'll install exactly like the docs say. |
| 04:05.09 | Nugget | s/find/fine/ |
| 04:05.31 | websae | what linux distro should i use? |
| 04:05.38 | Nugget | it doesn't matter in the slightest |
| 04:06.57 | websae | so you don't suggest asterisk? |
| 04:07.28 | Nugget | I don't suggest deploying it in production until you're very familiar with what you're getting in to. |
| 04:07.34 | harryvv | yes |
| 04:07.38 | harryvv | know your dial plan |
| 04:07.51 | Nugget | every phone call I've made for the past 8 months has gone through asterisk. It's obviously capable of doing lots of things. |
| 04:08.03 | Nugget | but it hasn't been easy or simple or problem free, and that's an understatement |
| 04:08.09 | harryvv | nugget, so far for the most part mine been reliable. |
| 04:08.12 | websae | what is the best beginners tutorial in setting this up? |
| 04:08.15 | *** join/#asterisk Insanity5 (n=feaw@ip68-105-209-13.tl.dl.cox.net) |
| 04:08.17 | Insanity5 | Ok. |
| 04:08.18 | Nugget | ~docs |
| 04:08.21 | jbot | from memory, docs is Documentation can be found at http://www.digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or http://www.voip-info.org/wiki-Asterisk or http://www.asteriskdocs.org or http://www.oreilly.com/catalog/asterisk |
| 04:08.35 | Insanity5 | I can dial, phone rings, but the moment I press the last digit on thep hone I can no longer hear anything |
| 04:08.38 | Insanity5 | What might cause this? |
| 04:08.52 | harryvv | that makes no sence |
| 04:09.01 | Insanity5 | And the phone doesn't stop ringing when I hang up. |
| 04:09.23 | harryvv | what phone |
| 04:09.26 | Nugget | most of my asterisk problems have arisen from stubbornly trying to swim upstream and use freebsd instead of linux, and from the zaptel drivers and my tdm400p in linux. |
| 04:09.29 | harryvv | try the same with x-lite |
| 04:09.48 | Nugget | well, and from the crappy call quality I get from internet-routed voip calls from time to time |
| 04:10.00 | harryvv | nugget, i have used fedora and now centos with @home been reliable. |
| 04:10.01 | Insanity5 | harryvv - x-lite works fine. |
| 04:10.05 | Insanity5 | harryvv - belkin message manager. |
| 04:10.15 | Nugget | a linux box and a pair of spa-2000s instead of freebsd and a tdm400p would be orders of magnitude more reliable for you. |
| 04:10.18 | harryvv | insanity mabey its a phone problem |
| 04:10.43 | Insanity5 | the ata is a POS, but I can use it on stanaphone without any problems. |
| 04:11.09 | Qwell | Are there any like SIP PRI interfaces? |
| 04:11.11 | Qwell | or something |
| 04:11.23 | Qwell | similar to an ATA, of course |
| 04:11.31 | websae | nugget-how much are those? |
| 04:11.53 | Nugget | I'm not sure. I bought a tdm400p instead. :) |
| 04:11.57 | Insanity5 | can you force an extension/userid/registration to ring from the asterisk console? |
| 04:12.17 | *** join/#asterisk Cresl1n (n=Cresl1n@digium.com) |
| 04:12.34 | *** join/#asterisk visik7 (n=mierda@unaffiliated/visik7) |
| 04:12.53 | websae | what kind of setup do you have running nugget? |
| 04:13.20 | Nugget | I have a freebsd 5.mumble box running just plain old asterisk with nothing but iax voip connections and sip clients. |
| 04:13.36 | Nugget | and I have a linux box with a tdm400p that has 1 FXO and 1FXS and some sip hard phones at home |
| 04:13.54 | Nugget | (cisco 7960 and a wireless 802.11 thing) |
| 04:14.05 | Qwell | Nugget: bridging your 7960? |
| 04:14.18 | Nugget | no, I mean a wireless phone. |
| 04:14.22 | Qwell | oh, heh |
| 04:14.26 | Nugget | I bought one of those dreadful zyxel things |
| 04:14.30 | Qwell | they no good? |
| 04:14.40 | Nugget | not the zyxel ones (same as the pulver phone) |
| 04:14.43 | Insanity5 | What are the advantages to using an ISDN pair instead of two pots lines for two incoming lines? they cost the same. |
| 04:14.46 | Nugget | I hear good things about the hitachis |
| 04:15.03 | *** join/#asterisk sfarquhar (n=sfarquha@b128D.static.pacific.net.au) |
| 04:15.22 | Nugget | websae: I documented my asterisk experimenting at http://slacker.com/~nugget/asterisk.php |
| 04:16.02 | sfarquhar | is this a place for newbie questions about asterisk? Not technical, just where I can find a vendor to do everything for me? |
| 04:16.07 | Nugget | the only problems I still have are having to cold boot the linux box every month or so to make the tdm400p start working again, and perpetual voip call quality issues. |
| 04:16.26 | *** join/#asterisk NSGN (n=brandonb@cpe-66-69-197-25.austin.res.rr.com) |
| 04:16.34 | NSGN | hello again |
| 04:16.44 | Nugget | if you skip the voip part and only do pstn calls you can surely put together a decent and cheap solution |
| 04:17.03 | websae | whose your provider nugget? |
| 04:17.15 | Nugget | I use nufone and asterlink and voicepulse. |
| 04:17.28 | websae | which is the best? |
| 04:17.33 | Nugget | none of them. :) |
| 04:17.43 | JerJer | Nugget: ur not running windows - just reload the kernel drivers and start asterisk |
| 04:17.44 | websae | which is the best deal? |
| 04:17.55 | harryvv | can asterisk measure voice quality before say, dropping the call then revert to another voip service? |
| 04:17.55 | MikeJ[Laptop] | oh Qwell... |
| 04:17.59 | Nugget | JerJer: I haven't foind that to be effective. |
| 04:18.07 | Nugget | in fact, I can't even reboot the box. it has to be a cold start |
| 04:18.15 | MikeJ[Laptop] | Qwell, 4403! |
| 04:18.21 | Nugget | the best deal depends entirely on what you need. |
| 04:18.37 | JerJer | that was fixed lately - i believe |
| 04:18.38 | Nugget | if you can live with a toll free number and not a local area code number your options will be much better |
| 04:18.48 | Nugget | cool, it's been a good month or so since I cvs upped. |
| 04:19.20 | [hC] | Man.. This CDR caller id bug is driving me batty. |
| 04:19.36 | Qwell | MikeJ[Laptop]: is it closed? |
| 04:19.39 | [hC] | And if i could find a way to play audio immediately after a call is answered, even better. |
| 04:19.51 | MikeJ[Laptop] | Qwell, just got commited |
| 04:19.54 | Qwell | sweet |
| 04:19.57 | MikeJ[Laptop] | yes |
| 04:20.00 | NSGN | hc: cdr caller id bug? |
| 04:20.06 | Qwell | bout time :p |
| 04:20.40 | WilliamK | alot of activity on the CVS finally |
| 04:20.50 | WilliamK | noticed the last week is very active |
| 04:20.50 | Insanity5 | Does anyone use an 800# as your home #? Does it become a problem day-to-day? :) |
| 04:20.52 | [hC] | NSGN: yeah, for some reason people calling in over my PRI, occasionally the caller Id gets stored right, occasionally it stores the caller id of a previous call. ${CALLERIDNUM} is alweays correct as far as the dialplan is concerned, its just the cdr storage function that's sometimes flaky. |
| 04:21.13 | harryvv | Nugget, here is a interesting question. Can asterisk read incomming bell core signaling in the s as in start dial plan and then ring a extention? |
| 04:21.59 | Nugget | I'm too ignorant to answer that. |
| 04:22.22 | harryvv | okay |
| 04:22.29 | Nugget | Insanity5: I tried and met with some resistance, but not insurmpuntable. |
| 04:22.33 | Nugget | insurmountable, even. |
| 04:22.44 | Nugget | my girlfriend's parents totally do not "get" calling her on an 800 number. |
| 04:22.57 | NSGN | heh |
| 04:22.59 | harryvv | problem is, our condo external audio control panel uses dtmf buttons to ring the condo owners phone..but with a bell core ring and no cid. |
| 04:23.01 | Nugget | and I recently changed dentists and it took a few tries to convince them that it was not my work number |
| 04:23.15 | Qwell | another issue I've seen with 800... |
| 04:23.24 | Qwell | if you set your cidnum to 8xx, you can't call certain numbers |
| 04:23.41 | Nugget | oh, really? I haven't encountered that yet |
| 04:23.48 | harryvv | mabey i should get a 1800 did and then give it to my perents ;) |
| 04:24.00 | Qwell | Nugget: do you set an 8xx cidnum on outgoing? I can pm you a number to try |
| 04:24.06 | twisted | wheeeeee |
| 04:24.18 | Nugget | I only recently switched to the 800 number is my cidnum |
| 04:24.27 | Nugget | I've mostly been using my local voicepulse did |
| 04:24.28 | twisted | someone send money to the "twisted needs more beer" fund |
| 04:24.38 | twisted | toastido@toastido.net via paypal :P |
| 04:25.05 | twisted | I promise that all proceeds will go towards beer. |
| 04:25.11 | harryvv | nugget you made that entire dial plan by hand? |
| 04:25.15 | Nugget | yes |
| 04:25.15 | Qwell | twisted: no crack this time? |
| 04:25.19 | harryvv | very nice |
| 04:25.19 | harryvv | ;) |
| 04:25.25 | twisted | Qwell, lol - nah, i'm off that crap |
| 04:25.27 | [hC] | Sweet, i figured out how to play an annoucement upon call connect! |
| 04:25.28 | [hC] | :) |
| 04:25.30 | Qwell | twisted: wussy |
| 04:25.31 | Qwell | :P |
| 04:25.39 | websae | with voice pulse can you transfer your number? |
| 04:25.47 | ManxPower | Reality is for people that can't handle drugs. |
| 04:26.14 | NSGN | hc: nice. so could one then have that announcement be a menu of sorts, asking for one to enter a selection |
| 04:26.21 | NSGN | like press 1 for this 2 for that 3 for them? |
| 04:26.21 | websae | is there number portability with VP? |
| 04:27.19 | harryvv | nugget explain this extention _+1%20512%20NXX-XXXX |
| 04:27.29 | Nugget | total vanity. |
| 04:27.40 | harryvv | total vanity ? |
| 04:27.44 | Nugget | I export my os x address book to a web cgi which I call up from my 7960 web interface thingey |
| 04:27.49 | NSGN | can someone tell me the difference between this |
| 04:27.50 | NSGN | http://www.digitnetworks.com/store/product_info.php?cPath=28&products_id=62 |
| 04:27.51 | ManxPower | Isn't %20 a space? |
| 04:27.51 | NSGN | and |
| 04:27.57 | NSGN | http://www.digitnetworks.com/store/product_info.php?cPath=28&products_id=40 |
| 04:27.58 | NSGN | that |
| 04:27.59 | [hC] | NSGN: well... Yes, you can do that. there are two options you can pass to Dial.. A(x) and M(x) ... A(x) will play x.gsm when the called party answers, and M(x) will jump to macro x after the party answers |
| 04:28.01 | Nugget | in my address book, my numbers are formatted as "+1 xxx nxx-nxxx" |
| 04:28.13 | [hC] | i just wanted it to beep to give them an indicator that an intercom call was coming in |
| 04:28.13 | Nugget | I didn't want to have to sanitize those for asterisk's sake, because it looks ugly |
| 04:28.22 | JerJer | digit networks is evil |
| 04:28.32 | Nugget | so I allow the 7960 to dial those numbers in their pretty format, and clean it up in the dialplan |
| 04:28.53 | harryvv | nugget so +1 is adding the long distance |
| 04:29.03 | Nugget | yeah |
| 04:29.10 | harryvv | interesting |
| 04:29.10 | ManxPower | NSGN: The Sipura has a history of making good hardware and firmware. Grandstream has a history of "almost OK" products and very buggy firmware. |
| 04:29.23 | Nugget | it's ugly, but it lets my phone book stay nice-looking. |
| 04:29.34 | Nugget | http://slacker.com/~nugget/asterisk8.php |
| 04:29.35 | ManxPower | NSGN: try voxzilla.com 's store. |
| 04:29.41 | Nugget | http://slacker.com/photos/nugget/IMG_2455 |
| 04:30.25 | FarrisG | I got my Polycom 501 working with DHCP, and now it's setting the time correct, except that it's two hours off. Will I have to manually set the time zone on each one of these, or is there a way to have it get the TZ and date (or specify an NTP server) from DHCP? |
| 04:30.55 | harryvv | ferris i have the 500 its nice. works good. |
| 04:31.05 | ManxPower | FarrisG: You mean like: option time-offset -21600; |
| 04:31.41 | *** join/#asterisk Insanity5 (n=feaw@ip68-105-209-13.tl.dl.cox.net) |
| 04:32.00 | FarrisG | ManxPower: Yes, but is that a parameter that should be sent by the DHCP server? As in, something I need to configure in dhcpd.conf? |
| 04:32.19 | Insanity5 | What's a subscribe request by an ata mean? |
| 04:32.26 | *** join/#asterisk diegows (n=diegows@201.250.113.103) |
| 04:32.32 | FarrisG | It's getting the proper time (it wasn't before), but it's still getting the wrong date |
| 04:32.47 | ManxPower | FarrisG: yes. see the man page |
| 04:32.47 | harryvv | Farris got all the buttons to work on it? |
| 04:33.10 | harryvv | like services button , conferance button |
| 04:33.12 | ManxPower | FarrisG: chances are it's in GMT/UTC and it's tomorrow there. |
| 04:33.19 | FarrisG | harryvv: Not yet, haven't tried |
| 04:33.31 | FarrisG | ManxPower: Nope. The phone thinks it's October 5 |
| 04:33.33 | FarrisG | :) |
| 04:33.49 | ManxPower | FarrisG: odd. |
| 04:33.51 | diegows | hi |
| 04:34.20 | diegows | little question, what is the syntax to answer calls incomming from iax using alsa? |
| 04:34.22 | twisted | Nugget, :P |
| 04:34.25 | twisted | Nugget, thanks tho :) |
| 04:34.33 | Nugget | don't drink it all in one place! |
| 04:34.39 | Insanity5 | What does this mean? |
| 04:34.39 | Insanity5 | Aug 25 00:34:40 WARNING[23854]: Maximum retries exceeded on call 23ba20-c0a8c801-1-2d@68.105.209.13 for seqno 2 (Non-critical Response) |
| 04:34.48 | twisted | lol, actually, that's the only way it will work :P |
| 04:34.51 | Qwell | diegows: answer |
| 04:35.13 | Tekmaven | hey guys |
| 04:35.14 | twisted | Nugget, but i do appreciate the gesture :) |
| 04:35.16 | diegows | Qwell, only answer? for example: |
| 04:35.20 | diegows | [iaxin] |
| 04:35.28 | diegows | exten => s,1,answer |
| 04:35.29 | ManxPower | Insanity5: That means "The phone went away, I gave up trying". If the phone is behind NAT then add qualify=yes to the sip.conf entry for that device. |
| 04:35.34 | diegows | that's all? |
| 04:35.46 | Qwell | no, answer on the CLI, if thats where its dialing to |
| 04:35.51 | Insanity5 | ManxPower - The phone is not, but might it help? |
| 04:36.03 | ManxPower | Insanity5: I doubt it. |
| 04:36.27 | Insanity5 | ManxPower - How does the phone know what RTP port to connect to the server on? I only opened 40 of them and specified the range in rtp.conf. Works fine for xlite. Would this with/without nat change anything? |
| 04:36.41 | diegows | Qwell, but when ther other person calls to me he enter to some context |
| 04:36.41 | Insanity5 | ManxPower - Well, every time I dial a call that is what happens and nobody can hear anything |
| 04:37.09 | diegows | what i need in the context to give a chance to use the comand answer |
| 04:37.10 | diegows | ? |
| 04:37.37 | Qwell | You'll need to dial the CLI |
| 04:37.42 | Tekmaven | is there any reason why if i put a call on hold the IAXy hangs up? |
| 04:38.16 | ManxPower | diegows: you mean like.... exten => 3511,4,Dial(Console/1) |
| 04:38.26 | *** join/#asterisk mNick34 (n=emcee@208.179.183.106) |
| 04:38.38 | Insanity5 | ManxPower - This is almost the same issue I'm having, minus the xlite phones -- http://lists.digium.com/pipermail/asterisk-users/2004-June/051127.html |
| 04:39.01 | diegows | ManxPower, yes... that's it thanks |
| 04:39.02 | diegows | :) |
| 04:39.07 | diegows | Qwell, thanks too |
| 04:39.14 | MikeJ[Laptop] | Corydon76-home, ping |
| 04:39.25 | Corydon76-home | pong |
| 04:39.54 | Corydon76-home | What's with the "review" flag? |
| 04:40.36 | Cresl1n | eh? |
| 04:41.04 | MikeJ[Laptop] | Corydon76-home, a little impromptu workflow. |
| 04:41.09 | Tekmaven | Qwell: have any insight? :) |
| 04:41.19 | Tekmaven | i defer to your knowledge ;) |
| 04:41.28 | Insanity5 | BRB |
| 04:41.29 | MikeJ[Laptop] | so you got e-mail on 4892 I suppose. |
| 04:41.49 | Corydon76-home | I got an email on quite a few things |
| 04:42.05 | *** join/#asterisk juice (n=juice@mo-67-77-177-215.dyn.sprint-hsd.net) |
| 04:42.55 | FarrisG | Ok, I was wrong, it's not getting the time correctly |
| 04:43.20 | MikeJ[Laptop] | 4892, kp just commented on 5 min ago |
| 04:43.21 | FarrisG | It just happened to look correct because the default time is apparantly 9:30, and it was 9:30 when I turned the phone on :) |
| 04:43.43 | FarrisG | I just added an "option ntp-servers" line to my dhcpd.conf, seeing if that helps on release |
| 04:43.45 | harryvv | nugget, seen the @home dial plan? |
| 04:43.46 | Corydon76-home | Wow, you're allowing me a full 5 minutes to reply? |
| 04:44.05 | ManxPower | <PROTECTED> |
| 04:44.05 | ManxPower | <PROTECTED> |
| 04:44.05 | ManxPower | <PROTECTED> |
| 04:44.07 | *** part/#asterisk santiago (n=santiago@63.245.86.163) |
| 04:44.07 | MikeJ[Laptop] | heh |
| 04:44.16 | MikeJ[Laptop] | just heads up that it was out there... |
| 04:44.27 | MikeJ[Laptop] | not demanding immediate turn around... |
| 04:44.36 | MikeJ[Laptop] | you can have 10 min ;) |
| 04:44.39 | MikeJ[Laptop] | :P |
| 04:44.42 | FarrisG | ha! Not even close |
| 04:44.49 | FarrisG | wait |
| 04:44.51 | FarrisG | yes it is! |
| 04:44.53 | FarrisG | nm :) |
| 04:44.56 | Tekmaven | any of you guys can help me? |
| 04:45.09 | litage | does * support sending SMS messages? |
| 04:45.45 | harryvv | Farris :) |
| 04:45.53 | *** join/#asterisk Insanity5 (n=feaw@ip68-105-209-13.tl.dl.cox.net) |
| 04:46.48 | Qwell | don't msg me |
| 04:46.59 | FarrisG | ok, so I've got this thing all happy time-and-tcp/ip-wise |
| 04:47.13 | FarrisG | Now if I could just get the damn thing SIP/*-happy :) |
| 04:48.03 | *** join/#asterisk mogorman_ (n=mogorman@user-24-236-84-48.knology.net) |
| 04:48.19 | FarrisG | Ack, no! It's still not taking the offset. The date and time are correct, but it's getting GMT |
| 04:48.25 | *** join/#asterisk LogicalOnline (n=LogicalO@m210e36d0.tmodns.net) |
| 04:48.27 | *** join/#asterisk papsow (n=pape@modemcable224.217-130-66.mc.videotron.ca) |
| 04:48.50 | Tekmaven | i can connect.. and * outputs this on the cli: |
| 04:48.52 | Tekmaven | -- SIP/sip.broadvoice.com-5780 answered IAX2/iaxy@iaxy/3 |
| 04:49.20 | Tekmaven | but if i push "hold" on the phone... it hangs up |
| 04:49.22 | Tekmaven | and the CLI says: |
| 04:49.23 | Tekmaven | <PROTECTED> |
| 04:49.26 | harryvv | brb |
| 04:49.36 | *** join/#asterisk mog_work (n=mogorman@user-24-236-84-48.knology.net) |
| 04:49.41 | mog_work | yo yo yo |
| 04:49.57 | Tekmaven | and Qwell: sorry for msging you |
| 04:50.11 | mog_work | hows everyone out here tonight |
| 04:50.24 | Tekmaven | mog_work: nice night here in NY.. |
| 04:50.31 | Tekmaven | just bashing my head with my IAXy ;) |
| 04:50.33 | mog_work | good stuff |
| 04:50.36 | mog_work | oh no |
| 04:50.39 | mog_work | whats the problem |
| 04:50.59 | Cresl1n | mog_work: are yous still here? |
| 04:51.04 | mog_work | come back |
| 04:51.08 | mog_work | im lonely... |
| 04:51.14 | Cresl1n | are you at work? |
| 04:51.19 | mog_work | are you? |
| 04:51.21 | Cresl1n | yeah |
| 04:51.23 | Cresl1n | :-P |
| 04:51.29 | Cresl1n | I guess we don't really know anymore |
| 04:51.33 | Tekmaven | mog_work: IAXy will disconnect when i hit "hold" |
| 04:51.40 | Tekmaven | (on the phone) |
| 04:51.50 | mog_work | how are you connected? |
| 04:51.56 | mog_work | and yeah come back here matt |
| 04:52.02 | mog_work | i have a question for you |
| 04:52.02 | Tekmaven | iaxy is going through an old fasionned PBX system |
| 04:52.09 | Tekmaven | it is like.. from the mid 80s |
| 04:52.10 | mog_work | so iaxy goes to asteirks |
| 04:52.11 | Tekmaven | as a phone line |
| 04:52.13 | mog_work | to old pbx |
| 04:52.15 | mog_work | over what? |
| 04:52.15 | Tekmaven | yeap |
| 04:52.22 | Cresl1n | mog_work: I'm getting ready to leave in a couple of minutes |
| 04:52.29 | Tekmaven | * -> cat5 -> IAXy -> RJ11 |
| 04:52.33 | Tekmaven | -> old pbx |
| 04:52.35 | mog_work | it will only take a second |
| 04:52.47 | mog_work | im confused |
| 04:52.49 | Tekmaven | -> RJ11 -> Phone ;) |
| 04:52.51 | Cresl1n | nope |
| 04:52.56 | Cresl1n | no can do |
| 04:53.01 | mog_work | lame.... |
| 04:53.02 | Cresl1n | I don't believe that you're relaly here |
| 04:53.05 | Cresl1n | :-P |
| 04:53.08 | Tekmaven | heh |
| 04:53.10 | mog_work | yeah |
| 04:53.12 | mog_work | thats true |
| 04:53.15 | mog_work | you can see my ip |
| 04:53.19 | Cresl1n | hehe |
| 04:53.20 | mog_work | im on the couch |
| 04:53.27 | Cresl1n | I could tell |
| 04:53.28 | mog_work | working on my secret project |
| 04:53.31 | Cresl1n | by the way you were typing |
| 04:53.32 | mog_work | which is cooler than yours |
| 04:53.35 | Cresl1n | hehe |
| 04:53.36 | mog_work | ? |
| 04:53.36 | Cresl1n | sure |
| 04:53.38 | Cresl1n | :-P |
| 04:53.43 | mog_work | yeah it is nice on the couch |
| 04:53.50 | mog_work | i need a new laptop |
| 04:54.01 | mog_work | ^_^ |
| 04:54.07 | *** join/#asterisk greendisease (n=jack@fedora/greendisease) |
| 04:54.10 | Cresl1n | go back to help tekmaven |
| 04:54.17 | Cresl1n | he has a need :-P |
| 04:54.35 | *** join/#asterisk juice (n=juice@mo-67-77-177-215.dyn.sprint-hsd.net) |
| 04:54.37 | *** join/#asterisk oOlli (i=www-data@baltz-online.de) |
| 04:55.13 | oOlli | hi folks. i am not able to hear the MusicOnHold-Sound! i have compiled a new rtc-timer (zaptelrtc), too |
| 04:56.10 | oOlli | is it possible that my SIP phone does not SLIN?! |
| 04:56.38 | *** join/#asterisk rking (n=rking@ip68-105-231-56.lu.dl.cox.net) |
| 04:56.56 | mog_work | most dont |
| 04:57.00 | mog_work | but they do ulaw |
| 04:58.32 | oOlli | i am just wondering... this is my log when starting the MOH: --> http://pastebin.com/345682 |
| 04:59.29 | *** join/#asterisk kevin (i=1001@rrcs-67-78-250-234.se.biz.rr.com) |
| 04:59.45 | Vco | one thing i learned when i was there...you can get way cooler stuff in vending machines and arcades http://www.winnipegit.com/vending.jpg |
| 05:00.08 | Insanity5 | Is there any known solution to maximum retries on call for an otherwise unfirewalled device? Is there any debug information that might help find the problem, or is it hopelesS? |
| 05:00.26 | *** join/#asterisk anonobomber (n=anonobom@c-67-170-91-29.hsd1.wa.comcast.net) |
| 05:02.22 | *** join/#asterisk papsow (n=pape@modemcable224.217-130-66.mc.videotron.ca) |
| 05:06.27 | rking | wow, i have to admit, i am definitely overwhelmed by the configuration of asterisk at the moment. i am trying the stuff in iaxclient's quickstart doc, but there are some parts i don't get at all, e.g.: when it says to put some stuff in extensions.conf, which [section] does it mean? also, when i make changes to /etc/asterisk/*, do i have to kill -HUP the server? |
| 05:06.55 | rking | and that's not even asking about this: chan_iax2.c:5761 socket_read: Rejected connect attempt from 127.0.0.1, request '100@default' does not exist =) |
| 05:07.06 | JamesDotCom | you just have to reload your config through the asterisk cli |
| 05:07.11 | Corydon76-home | MikeJ[Laptop]: comment added |
| 05:07.32 | JamesDotCom | 100@default means that extension on that context doesnt exist on the remote server |
| 05:07.55 | rking | JamesDotCom: ok, so the [foo] sections in extensions.conf are "context"s? |
| 05:08.04 | JamesDotCom | yes |
| 05:08.40 | *** join/#asterisk Tili (i=Tili@218.19.4.56) |
| 05:09.10 | rking | how do i know what context it's trying to use? just guess & check? |
| 05:09.59 | Corydon76-home | MikeJ[Laptop]: I'm somewhat afraid that he's going to 'insist', in which case the code won't make it into 1.2... I simply don't have the time for a rework |
| 05:11.13 | mago | how can I put the colors back in asterisk if it lost them? |
| 05:11.53 | JamesDotCom | rking, either dont send it to a context if you dont have access to the remote server |
| 05:12.00 | JamesDotCom | or create a matching one |
| 05:12.02 | JamesDotCom | assuming you do |
| 05:12.08 | JamesDotCom | since you seem to be connecting to 127.0.0.1 |
| 05:13.27 | rking | JamesDotCom: hrmm.. maybe i should go about this this way: do you know what i can dial from iaxcomm into a default asterisk config to test the demos, etc. |
| 05:13.39 | rking | JamesDotCom: maybe i could figure out how to work it all if i could start with the working demo's. =\ |
| 05:14.16 | *** join/#asterisk greendisease (n=jack@fedora/greendisease) |
| 05:14.47 | JamesDotCom | rking: well i'm not sure at all how iaxcomm is setup, and i rarely deal with iax as it is (sip4life, ahah) |
| 05:14.47 | *** join/#asterisk djin_ib (n=djin_ib@gridfox.xs4all.nl) |
| 05:14.51 | JamesDotCom | but |
| 05:15.02 | JamesDotCom | you're just trying to make calls to, or recieve calls from iaxcomm? or both? |
| 05:15.21 | rking | JamesDotCom: what i want, fundamentally, is a replacement for skype 1-on-1 and conferences |
| 05:15.35 | rking | JamesDotCom: i got the feeling that iax > sip, but i'm swayable |
| 05:15.55 | Tekmaven | hey guys |
| 05:15.57 | Tekmaven | just a little update |
| 05:16.00 | Tekmaven | i figured out the hold disconnect problem |
| 05:16.09 | Tekmaven | it was the bloody old PBX that the IAXy was connected too |
| 05:16.11 | rking | JamesDotCom: i'd like my localhost to serve as a central point for conferences or one-on-one calls to me |
| 05:16.52 | rking | Tekmaven: i'm not quite sure what all that means, but i'm glad you've figured it out. =) |
| 05:17.25 | JamesDotCom | okay |
| 05:18.26 | JamesDotCom | so what you'd want to do, in your iax.conf, define iaxcomm as a peer, you will also set an incoming context, and this is where calls FROM iaxcomm go to and define the section the incoming call should go to in extensions.conf |
| 05:18.36 | Tekmaven | rking: i had this crap problem with my IAXy and my phone setup |
| 05:18.43 | Tekmaven | where if i hit "hold" on a phone, the IAXy would hang up the call |
| 05:19.02 | JamesDotCom | also in iax.conf you'll want to put in a register statement so you register to iaxcomm as a user |
| 05:19.05 | Tekmaven | its * -> IAXy -> Old 1980s PBX System -> Phone :) |
| 05:19.21 | Tekmaven | JamesDotCom: while your talking about iax.conf, i have a quicky question |
| 05:19.35 | Tekmaven | one thing i notice about my IAXy calls, is the volume is "too loud" |
| 05:19.54 | JamesDotCom | rking: you're using a sip phone to register to asterisk? |
| 05:19.57 | Tekmaven | if there was a volume setting, and it was set to a "5", i would make it a "4"... if it was a "10", i'd make it an 8 |
| 05:20.13 | JamesDotCom | sorry Tekmaven, no idea ;( |
| 05:20.18 | Tekmaven | ah okay, np |
| 05:20.24 | Tekmaven | thanks for reading the question ;) |
| 05:20.25 | JamesDotCom | i dont really deal with iax |
| 05:20.30 | JamesDotCom | i much prefer sip |
| 05:20.39 | JamesDotCom | so all i use iax for is for a gateway to my sip network |
| 05:20.39 | JamesDotCom | heh |
| 05:20.41 | Tekmaven | heh, even with all the firewall stuff |
| 05:20.56 | Insanity5 | Aug 25 01:22:06 NOTICE[23854]: Rejected connect attempt from 198.22.67.70 |
| 05:21.04 | JamesDotCom | never had an issue with sip and firewalls |
| 05:21.04 | Insanity5 | Supposedly, an iax peer -- incoming call. |
| 05:21.10 | JamesDotCom | i also use SER for SIP though |
| 05:21.19 | Insanity5 | I added to iax.conf and extensions.conf as advised. Any ideas? |
| 05:21.33 | JamesDotCom | Insanity5: iax2 debug blahblah |
| 05:21.38 | rogercharlie | does the areski cdr - pdf export work? |
| 05:21.38 | rking | JamesDotCom: ok... so i did something like this: [iaxcomm] type=peer; username=corp; secret=donkey; peercontext=incoming |
| 05:21.43 | JamesDotCom | check it's the right auth method n stuff |
| 05:22.23 | mago | is there any client that works with * that does file transfer (besides messenger) |
| 05:23.08 | JamesDotCom | context=incoming |
| 05:23.09 | JamesDotCom | i believe |
| 05:23.37 | rking | JamesDotCom: i did auth=md5,plaintext,rsa ... just snarfing it from a commented line from up above |
| 05:23.59 | rking | JamesDotCom: oh, is it specifying how to interpret the secret=... part? |
| 05:24.08 | JamesDotCom | sorry, that was in response to Insanity5 |
| 05:24.10 | Insanity5 | JamesDotCom - Cause: No authority found. |
| 05:24.17 | JamesDotCom | i think if you stick to md5 you should be fine |
| 05:24.22 | rking | JamesDotCom: k |
| 05:24.34 | rking | JamesDotCom: so should the secret=... be md5'd? |
| 05:24.50 | JamesDotCom | i dont think so |
| 05:25.12 | rking | k... |
| 05:25.33 | JamesDotCom | rking: i think you also need a "register => username:password@host.iaxcomm.blah" statement in your iax.conf file too |
| 05:25.58 | rking | JamesDotCom: how did you learn all of this? so far, i can't find any docs read the source? |
| 05:26.23 | rking | JamesDotCom: what would that register => line do? |
| 05:26.32 | JamesDotCom | then if you're using a sip phone to connect to the machine, in your sip extension defintion, set another incoming context.. say "from-sip", then in there put an extension in like "exten => _.,1,Dial(IAX2/iacomm/${EXTEN})" |
| 05:26.41 | JamesDotCom | it registers you as a user to iaxcomm |
| 05:26.46 | JamesDotCom | you login, basically |
| 05:27.05 | rking | this is extremely trippy. =) |
| 05:27.08 | JamesDotCom | and, i dunno, i've been playing with asterisk for years now, so the gist of it comes pretty naturally |
| 05:27.12 | JamesDotCom | there are heeeeeeaps of docs |
| 05:27.16 | JamesDotCom | check www.voip-info.org |
| 05:27.18 | JamesDotCom | and go from there |
| 05:27.36 | Insanity5 | JamesDotCom - Whenever you're done helping him -- it errored out as no authority found on the tx packet :). Thanks. |
| 05:28.04 | *** join/#asterisk litage (n=nick@203.201.96.101) |
| 05:28.17 | JamesDotCom | Insanity5: check out google with that error message, i've been through it once before, there was stacks |
| 05:28.22 | JamesDotCom | hahaha |
| 05:28.27 | JamesDotCom | yeah, my shift's over :P |
| 05:28.29 | Tekmaven | JamesDotCom: i've been working with * for about 3 weeks now, just doing it in my home... and after spending *days+* reading the wiki, i have a basic understanding =) |
| 05:28.32 | Insanity5 | JamesDotCom - Giving me general authentication issues so far =) |
| 05:28.33 | JamesDotCom | no problem |
| 05:29.10 | Tekmaven | * is really sweet, and im happy i put the time in ;).. i totally feel every beginner's pain |
| 05:29.10 | Tekmaven | heh |
| 05:29.26 | Insanity5 | my bad :P. it's in sip.conf and not iax hehe. |
| 05:29.32 | *** join/#asterisk r0d3nt (n=RatMan@telephreak.net) |
| 05:32.28 | Tekmaven | umm, is voip-info down? :X |
| 05:32.28 | Tekmaven | oh no, just slow for a min |
| 05:32.36 | Qwell | ~wikistatus |
| 05:33.06 | Tekmaven | hehe |
| 05:33.21 | Tekmaven | Qwell: is there a way to set rxgain and txgain for an IAXy? |
| 05:33.28 | Tekmaven | and are the gains for "volume"? |
| 05:34.18 | Qwell | got me |
| 05:34.20 | Tekmaven | :-) |
| 05:34.56 | *** join/#asterisk clive- (n=pirch@rndf-146-33-211.telkomadsl.co.za) |
| 05:36.19 | *** join/#asterisk coaxial (i=hobo@gluttony.neverhost.net) |
| 05:37.23 | *** join/#asterisk sycofly (n=syco@sycofly.com) |
| 05:37.25 | *** join/#asterisk remmo (n=rem@smack.isp.net.au) |
| 05:37.30 | *** join/#asterisk oej (n=oej@apollo.webway.se) |
| 05:39.00 | *** join/#asterisk [1]JohnJacob (n=JohnJaco@pcp0011543387pcs.mainf01.in.comcast.net) |
| 05:39.32 | Insanity5 | JamesDotCom - What do tx and rx frame retryies mean? |
| 05:39.38 | Insanity5 | fori ax |
| 05:39.59 | JamesDotCom | transmit and recieve |
| 05:40.00 | *** join/#asterisk papsow (n=pape@modemcable224.217-130-66.mc.videotron.ca) |
| 05:40.04 | JamesDotCom | ? |
| 05:43.43 | kevin | I have a few questions if there is someone out there that can help me :-) |
| 05:45.13 | Tekmaven | hey kevin, i dunno how much i can help, but i will sure try.. |
| 05:45.21 | rking | kevin: it's a pretty helpful channel - i don't know squat, but there are fairly good odds that your question will be answered. there's one catch, though. |
| 05:45.36 | kevin | what's the catch? |
| 05:45.52 | *** join/#asterisk newmember (i=user@S010600036d1139fb.cg.shawcable.net) |
| 05:46.08 | FarrisG | I upgraded the bootROM and sip app on my 501, and it all seemed to go well, but now when the phone starts up it hangs at the "Welcome! Processing configuration" screen |
| 05:46.34 | kevin | ok.. this is my situation... started out with *@home, but have been advised by several people that I sould do it from scratch and it would solve the issues I was having.... |
| 05:47.35 | *** join/#asterisk Insanity5 (n=feaw@ip68-105-209-13.tl.dl.cox.net) |
| 05:47.52 | Insanity5 | Hmmm... can asterisk be made to pick up pulse dialing? |
| 05:47.58 | kevin | so I have the box working, can make and receive calls.. use IAX for my trunk and have 2 DIDs... my first issue is that I need to insert my own caller ID info for making calls over the different DIDs... this worked in *@home, but I can't figure out how to make it insert my CID |
| 05:48.37 | Insanity5 | Or is this a function of the ata? |
| 05:48.57 | kevin | the other issue is that my voicemail works, but I can't access it by dialing *98 or *97, so I have voicemails, but can't get to them.... what do I need to do to get into the voicemail? |
| 05:49.59 | Insanity5 | Oddd.. can dial my 866 # from cell, but not landline. |
| 05:51.55 | rking | kevin: the catch is that you have to ask the question. --- i was going to say this earlier but i got distracted. |
| 05:52.17 | kevin | lol... well I have asked.. waiting now |
| 05:52.43 | *** join/#asterisk Uberbot (n=Uberbot@pcp01880954pcs.sandia01.nm.comcast.net) |
| 05:53.13 | Uberbot | Hi all. |
| 05:53.20 | *** join/#asterisk litage (n=nick@203.201.96.101) |
| 05:53.47 | *** join/#asterisk grimse (n=grimse@p5481E735.dip.t-dialin.net) |
| 05:54.17 | Qwell | kevin: 1) You'll want to use SetCIDNum(12345) (or, with cvs head, I think its like Set(CID(Num)=12345) or something) |
| 05:54.37 | Qwell | kevin: 2) You'll need to setup an extension that does VoicemailMain |
| 05:55.06 | Qwell | I accept cash, credit card, paypal, check, and money order |
| 05:55.20 | Qwell | (not in that order) |
| 05:55.50 | Corydon76-home | Set(CID(number)=12345) |
| 05:55.59 | Qwell | number..I was close |
| 05:56.33 | kevin | ok.. where do I put that?? in the config for the exten in extensions.conf? |
| 05:56.43 | Qwell | extensions |
| 05:56.48 | Corydon76-home | Actually, spelled out |
| 05:57.04 | *** join/#asterisk juice (n=juice@mo-67-77-177-215.dyn.sprint-hsd.net) |
| 05:57.16 | Qwell | Corydon76-home: like Set(CallerID(number)=12345) ? |
| 05:57.29 | kevin | can that be done in globals? |
| 05:57.31 | Corydon76-home | Except the function ALL CAPS |
| 05:57.41 | Qwell | case matters? |
| 05:57.46 | Corydon76-home | No, it has to be done in the dialplan itself |
| 05:57.56 | Corydon76-home | All functions are all caps |
| 05:58.13 | Corydon76-home | Yes, case matters |
| 05:58.16 | Qwell | SET is a function, Answer is not, correct? |
| 05:58.25 | Corydon76-home | No, Set is an application |
| 05:58.34 | Qwell | When what function? |
| 05:58.37 | Corydon76-home | Answer is also an application |
| 05:58.40 | Qwell | then* |
| 05:58.46 | Corydon76-home | CALLERID() is a function |
| 05:58.49 | Qwell | ahh |
| 05:59.43 | Corydon76-home | The really nice thing about functions is that they can be used inline, just like variables |
| 05:59.49 | Qwell | right |
| 05:59.54 | Insanity5 | We don't do reqeusted format gsm, falling back to peer capability 65343 -- how do you tel what 65343 is? |
| 06:00.08 | Corydon76-home | Insanity5: show codec 65343 |
| 06:00.37 | kevin | ok.. so in the dial plan... in the outbound trunk string? |
| 06:00.52 | Insanity5 | Corndawg_ - It's not on the show codecs list. show codec 65343 shows another list |
| 06:01.17 | Corydon76-home | Insanity5: that's correct... 65343 is all of those codecs |
| 06:01.30 | Corydon76-home | Pretty much every codec except slin and lpc10 |
| 06:02.04 | Insanity5 | It's some special mathematically derived coding to tell it that it allows all but gsm? |
| 06:02.24 | Corydon76-home | It's a bitwise code, yes |
| 06:03.09 | Insanity5 | But I guess I'm in the wrong spot to find out what it actually used =) |
| 06:03.10 | Insanity5 | hehe |
| 06:03.15 | Corydon76-home | No, the peer is simply specifying all the codecs it is willing to talk |
| 06:03.39 | Insanity5 | correct |
| 06:03.59 | Corydon76-home | If you do a logical AND between your cap and the peer cap, you'll get all the codecs both of you are willing to talk |
| 06:04.14 | Corydon76-home | bitwise AND, not logical AND |
| 06:04.27 | Insanity5 | And nufone doesn't do alaw? arg |
| 06:04.59 | Corydon76-home | Probably due to the codec taking up so much bandwidth |
| 06:05.09 | Insanity5 | They do for my outgoing calls. |
| 06:05.12 | Qwell | they allow pretty much anything |
| 06:05.16 | Corydon76-home | 64k |
| 06:05.19 | Insanity5 | I dono't know how you can stand gsm :( |
| 06:05.36 | Corydon76-home | Easy, it's the codec used on most cell phones |
| 06:05.43 | Qwell | is there much of a difference between ulaw and alaw? |
| 06:05.51 | Insanity5 | not really |
| 06:05.58 | Corydon76-home | GSM = Global System for Mobile calls |
| 06:05.59 | Insanity5 | alaw has a slightly wider frequency range. |
| 06:06.11 | Insanity5 | alaw - At the cost of smashing stuff together that is diverse :) |
| 06:06.24 | Insanity5 | Corydon-w - My verizon cell phone sounds better than gsm |
| 06:06.49 | gordonjcp | what does it use? |
| 06:06.53 | Corydon76-home | Insanity5: nah, they probably just are just getting more packets through |
| 06:07.05 | Insanity5 | gordonjcp - dunno, probably cdma. |
| 06:07.11 | Insanity5 | Ok, disallow before allow, oops :) |
| 06:07.12 | Insanity5 | hehe |
| 06:07.14 | Qwell | cdma isn't a codec... |
| 06:07.20 | Insanity5 | Qwell - How can I tell? |
| 06:07.27 | Corydon76-home | CDMA is a protocol, not a codec |
| 06:07.30 | gordonjcp | yup |
| 06:07.34 | Corydon76-home | GSM is both |
| 06:07.35 | Qwell | gsm happens to be both |
| 06:07.35 | Insanity5 | I know it sounds 50 times better than my sprint one =) |
| 06:07.40 | Insanity5 | ahh ok -- hehe |
| 06:07.46 | Qwell | Insanity5: sprint just sucks is all |
| 06:07.47 | gordonjcp | is it the same make and model of phone? |
| 06:07.52 | Insanity5 | nope |
| 06:08.09 | Insanity5 | verizon = v60 something |
| 06:08.13 | Insanity5 | sprint = lg pm325 |
| 06:08.15 | gordonjcp | Insanity5: well, there you go then |
| 06:08.21 | gordonjcp | lg phones are shit |
| 06:08.28 | Insanity5 | But I believe it's more of a factor of what the network speaks. |
| 06:08.48 | Qwell | I'd agree with Corydon76-home here |
| 06:08.51 | Corydon76-home | Nah, it's more of a factor of how strong the network is |
| 06:08.55 | Qwell | ^ |
| 06:09.09 | Corydon76-home | Crappy signal is going to give you bad audio, anywhere |
| 06:09.57 | *** join/#asterisk [1]JohnJacob (n=JohnJaco@pcp0011543387pcs.mainf01.in.comcast.net) |
| 06:10.11 | *** join/#asterisk zoo (i=nobody@ip-142-16.travedsl.de) |
| 06:10.20 | Insanity5 | Ok someone call 866-249-2403 -- why is there a fine snap/crackle/pop in the background that isn't present through xlite direct? The number just plays some music. |
| 06:10.42 | Insanity5 | Well maybe it's this pos phone, I'm not sure :P |
| 06:11.04 | gordonjcp | Qwell: how goes the midi madness? |
| 06:11.19 | Qwell | gordonjcp: haven't done any in quite a while |
| 06:11.26 | gordonjcp | man, I can't even remember what analogue phones sound like |
| 06:11.46 | Qwell | I haven't used an analog phone...in like... |
| 06:11.51 | Kumbang | guys, what type is dtmf mode for connecting two asterisk server via SIP |
| 06:11.53 | Corydon76-home | Insanity5: I don't get any music, just 4 rings and a congestion |
| 06:11.53 | Qwell | over a year, at least |
| 06:12.01 | rking | Qwell: that's truly awesome. |
| 06:12.17 | Insanity5 | Corydon-w - Same here. Trying to figure out why! |
| 06:12.17 | Insanity5 | arg |
| 06:12.19 | Qwell | I have one sitting in my room, but it never rings or anything |
| 06:12.35 | Corydon76-home | Insanity5: maybe you're not Answer()ing the line? |
| 06:12.45 | Insanity5 | Corydon-w -I am now its working again wtf |
| 06:12.47 | Insanity5 | try now |
| 06:12.50 | Insanity5 | I didn't do anything |
| 06:12.58 | Insanity5 | I see lots of tx-rx frame retries though |
| 06:14.07 | Insanity5 | The asterisk server is only 7ms away from the providers iax server so hopefully conenctions aren't an issue |
| 06:14.10 | Insanity5 | hehe |
| 06:14.25 | Corydon76-home | Sounds like some of your audio is getting chopped off at the top end |
| 06:14.40 | Insanity5 | Corydon-w - Which is a function of the codec I'm assuming, right? Not *? |
| 06:14.55 | Corydon76-home | How are you streaming the audio? |
| 06:14.57 | Insanity5 | It should be using a/ulaw |
| 06:15.02 | Insanity5 | mp3player |
| 06:15.16 | Corydon76-home | Any zaptel devices? |
| 06:15.22 | Insanity5 | Nope... |
| 06:15.27 | Insanity5 | What do they do? |
| 06:15.32 | Corydon76-home | Provide timing |
| 06:15.38 | Insanity5 | timing? needed? |
| 06:15.42 | Insanity5 | For all callS? |
| 06:15.42 | *** join/#asterisk pbxbart__ (n=pbxbart@ip-80-226-230-74.vodafone-net.de) |
| 06:15.52 | *** part/#asterisk pbxbart__ (n=pbxbart@ip-80-226-230-74.vodafone-net.de) |
| 06:15.56 | Insanity5 | Now I'm getting the busy signal crap again |
| 06:15.58 | Insanity5 | WHY?!?!?! |
| 06:15.59 | Insanity5 | arg!!!!! |
| 06:16.00 | Corydon76-home | Needed for the audio stream |
| 06:16.03 | *** join/#asterisk jeffgus (n=jeffgus@2002:d856:c704:0:0:0:0:1) |
| 06:16.40 | Insanity5 | Ok, come back to the zaptel in a minute |
| 06:16.47 | Insanity5 | but how come I'm getting fast busy signals now? |
| 06:17.46 | Corydon76-home | Probably your network provider is unable to contact your server |
| 06:17.54 | Insanity5 | and now it works again |
| 06:17.54 | Insanity5 | arg |
| 06:17.58 | *** join/#asterisk darkskiez (n=darkskie@host86-133-147-187.range86-133.btcentralplus.com) |
| 06:18.06 | Qwell | behind a nat? Got the port forwarded? |
| 06:18.17 | Insanity5 | transparent firewall, but iax port is opened. |
| 06:18.18 | Qwell | your router might only keep it open for a short time |
| 06:18.20 | Corydon76-home | Or another machine is registering in as that client |
| 06:18.26 | Qwell | or that |
| 06:18.37 | Qwell | here's a dumb question |
| 06:18.46 | Qwell | if I have asterisk and say iaxcomm on the same machine |
| 06:18.51 | Qwell | I register with both... |
| 06:18.57 | Qwell | what gets the call? |
| 06:19.19 | Qwell | nevermind, heh. only one of them could even have the port open |
| 06:20.19 | Insanity5 | Corydon-w - Back to timers. |
| 06:20.24 | Insanity5 | Corydon-w - Doesn't teh comptuer have a clock? :P |
| 06:20.44 | Corydon76-home | Yes, and you could use ztdummy to take the timing off that clock |
| 06:21.08 | Insanity5 | What does zaptel do for me then? |
| 06:21.12 | Insanity5 | Software? |
| 06:21.15 | Qwell | ztdummy isn't the best |
| 06:21.18 | Insanity5 | I see it on asteriskpbx.org |
| 06:21.25 | Insanity5 | Qwell - I'd imagine it's better than nothing, right? |
| 06:21.30 | Qwell | of course |
| 06:21.33 | Corydon76-home | It's a source of very consistent timing |
| 06:21.48 | ender | docs I read said that ztdummy should not be used in production, only in testing stuff. |
| 06:22.09 | Qwell | I wouldn't use it in production...thats just me though |
| 06:22.20 | Qwell | any machine I'd use in production would have digium hardware though |
| 06:22.24 | ender | yeah |
| 06:22.30 | Insanity5 | Production is "my house" :) |
| 06:22.33 | Insanity5 | hehe |
| 06:22.34 | ender | our production boxen ( and even our test boxen ) have digium hardware. |
| 06:22.38 | pygrammer | does anybody here know a good place for getting cheap DIDs? |
| 06:22.51 | Qwell | pygrammer: your telco |
| 06:22.55 | Qwell | on a PRI |
| 06:22.55 | Insanity5 | I wish I could use these four quad T-1 carsd I have siting here :) |
| 06:23.03 | Insanity5 | Not made by digium... I ripped it out of some old voip box. |
| 06:23.06 | Qwell | Insanity5: send them over, I'll put them to good use |
| 06:23.08 | Qwell | oh, meh :p |
| 06:23.09 | pygrammer | Qwell, how about via voip? |
| 06:23.11 | Corydon76-home | x100p is an inexpensive source for zaptel timing |
| 06:23.20 | Qwell | Corydon76-home: agreed |
| 06:23.20 | pygrammer | there are a few services |
| 06:23.25 | pygrammer | just wondered which/what is besty |
| 06:23.27 | pygrammer | *best |
| 06:23.30 | pygrammer | i don't want a PRI right now |
| 06:23.32 | Insanity5 | Qwell - They'rep ci cards, but I have no clue how you would use them. |
| 06:23.45 | Qwell | Insanity5: you'd have to write a driver for them |
| 06:23.48 | pygrammer | nor do I want that many |
| 06:23.50 | Insanity5 | Qwell - They're not digium cards... |
| 06:23.52 | *** join/#asterisk tzafrir_laptop (n=tzafrir@local.xorcom.com) |
| 06:23.58 | Insanity5 | Qwell I have a NT driver on a cd =) |
| 06:24.00 | Insanity5 | hehe |
| 06:24.04 | Qwell | a zaptel driver |
| 06:25.01 | Insanity5 | What do I need for zaptel? |
| 06:25.07 | Insanity5 | The box is 2,000 miles away |
| 06:25.08 | Insanity5 | hehe |
| 06:25.19 | Qwell | ztdummy or hardware |
| 06:25.47 | Insanity5 | special hardware (nothign that might be in teh amchine)? |
| 06:26.02 | Qwell | hardware with a zaptel driver |
| 06:26.19 | rking | if i have a plain old modem with a POTS line running into it, can * be my answering machine? |
| 06:26.25 | Qwell | rking: no |
| 06:26.33 | Qwell | ~fxo |
| 06:26.37 | jbot | foreign exchange office - type of port you need to connect a POTS (Plain Old Telephone Service) line from your telco to a pbx http://www.digium.com/index.php?menu=fxsvfxo |
| 06:26.43 | *** join/#asterisk djin_ib (n=djin_ib@gridfox.xs4all.nl) |
| 06:27.27 | rking | hrm |
| 06:27.52 | Insanity5 | Qwell - Any risk in using/installing ztdummy? I don't want to freeze a remote box. |
| 06:28.21 | *** join/#asterisk qweee (n=vikramhe@210.212.195.134) |
| 06:29.07 | qweee | net2phone has changed the dial plan and it needs a prefix 011 in the header |
| 06:29.14 | qweee | how do i do this |
| 06:29.59 | qweee | earlier calling gulf used to be 9971xxxxxxxxx |
| 06:30.17 | qweee | now it has to be 011971xxxxxxxx |
| 06:30.26 | qweee | can any one help me |
| 06:30.47 | Qwell | edit extensions.conf |
| 06:30.59 | qweee | any other changes |
| 06:31.16 | *** part/#asterisk ipso (n=ipso@d207-81-249-35.bchsia.telus.net) |
| 06:31.24 | Qwell | shouldn't be |
| 06:31.40 | *** join/#asterisk Inv_arp (i=junya@adsl-11-73-99.mia.bellsouth.net) |
| 06:31.49 | qweee | it says call forbidden unauthorised dialed no |
| 06:32.02 | Qwell | bed |
| 06:33.30 | *** part/#asterisk Uberbot (n=Uberbot@pcp01880954pcs.sandia01.nm.comcast.net) |
| 06:35.53 | *** join/#asterisk CleanerX (n=nix@nat-ph3-wh.rz.uni-karlsruhe.de) |
| 06:36.17 | Freman | Hey... I want to dial *# before a number and have it recorded on the cdr as 'work related' how can I do that without duplicating all of my current [internal] exten lines with *# in front of them? |
| 06:37.30 | rking | Freman: i know nothing, but it wouldn't be too bad to write a perl script that automatically made those lines for you. |
| 06:38.13 | Freman | http://pastebin.com/345712 |
| 06:38.16 | rking | Freezer: perl -pe 's/exten => /$& *#/' or something |
| 06:38.28 | Freman | that's my current internal context |
| 06:40.08 | Freman | yah... but I was looking to see if there was a way to check for *# (EG: $[${EXTEN:0:2} = *#]) flat it with setaccount(2) then remove the *# from extension and have it run through the extensions like normal |
| 06:41.33 | *** join/#asterisk redG (i=[U2FsdGV@67.51.185.15) |
| 06:41.34 | *** join/#asterisk litage (n=nick@203.201.97.235) |
| 06:41.34 | redG | <PROTECTED> |
| 06:41.37 | *** join/#asterisk rkvarala (n=rkvarala@59.93.53.194) |
| 06:41.45 | rking | Freman: sounds good - i can't even set up * to give me plain old voip, yet, so i'm not to comment further. =) |
| 06:42.48 | Freman | hehe, only been at it off and on for a month my self (c: |
| 06:42.59 | Freman | maybe if I dial *# and have it prompt me for a number... |
| 06:43.13 | *** join/#asterisk r3d5un (n=oscam@80.121.192.21) |
| 06:46.55 | *** join/#asterisk Snake-Eyes (n=blog@203.201.97.235) |
| 06:47.09 | rkvarala | do we have any provision to send INFO packets to the * externally |
| 06:50.00 | rogercharlie | does anyone use the pdf export from areski's CDR? |
| 06:51.19 | *** join/#asterisk tuxinator_linuxM (n=tuxinato@209.101.237.194) |
| 06:53.14 | *** join/#asterisk stkn_ (i=nobody@gentoo/developer/pdpc.active.stkn) |
| 06:53.27 | Freman | http://pastebin.com/345721 <- what do you think my chances are of *# working the way I hope (c: |
| 06:53.51 | *** join/#asterisk litage (n=nick@203.201.97.235) |
| 06:54.25 | Insanity5 | I can not figure out how to compile the damn ztdummy/zaptel driver. grr |
| 06:56.32 | *** join/#asterisk msan (n=xxxxxxxx@golia.caltanet.it) |
| 06:56.53 | Freman | emerge zaptel |
| 06:56.54 | Freman | (c: |
| 07:00.04 | Insanity5 | Ok are there any alternatives other than zaptel? |
| 07:00.11 | Insanity5 | No way in hell I'm going to find the right kernel source |
| 07:02.42 | *** join/#asterisk litage (n=nick@203.201.97.235) |
| 07:02.50 | mover | ~seen coppice |
| 07:02.56 | jbot | coppice <n=chatzill@226.193.17.210.dyn.pacific.net.hk> was last seen on IRC in channel #asterisk, 14h 33m 34s ago, saying: 'shido6: you do if you want to know how to kick very slowly :-)'. |
| 07:03.18 | *** join/#asterisk m (n=maninder@220.224.39.63) |
| 07:03.48 | mover | any authorisation guru here? |
| 07:05.23 | m | i am maninder |
| 07:05.28 | m | <PROTECTED> |
| 07:05.35 | *** join/#asterisk bjohnson_ (n=bjohnson@i216-58-58-169.cybersurf.com) |
| 07:08.12 | m | halo |
| 07:10.47 | m | halo |
| 07:10.58 | mNick34 | hallo |
| 07:11.17 | m | <PROTECTED> |
| 07:11.42 | m | halo |
| 07:11.44 | *** join/#asterisk insomni_ (n=insomni@x1-6-00-e0-18-6f-34-ad.k455.webspeed.dk) |
| 07:11.46 | m | where ru |
| 07:11.51 | m | hio |
| 07:11.53 | m | hi |
| 07:12.01 | m | insomni |
| 07:14.12 | m | hi |
| 07:14.23 | m | hi |
| 07:15.16 | *** join/#asterisk sigmounte (n=sigmount@www.sighq.net) |
| 07:16.45 | m | hi |
| 07:17.01 | m | hi |
| 07:17.10 | m | hi |
| 07:17.49 | pygrammer | oh my fucking god |
| 07:17.58 | glm2k | hmm? |
| 07:18.05 | pygrammer | i'm getting 401 Unauthorized messages again!!!!!! THE NIGHTMARE HAS RETURNED |
| 07:18.06 | gordonjcp | morning pygrammer, glm2k |
| 07:18.15 | gordonjcp | lol |
| 07:18.17 | glm2k | morning gordonjcp |
| 07:18.19 | pygrammer | hello |
| 07:18.24 | gordonjcp | the 401 that would not die |
| 07:18.29 | pygrammer | really... |
| 07:18.38 | pygrammer | this is fucked up... i didn't change my configuration one bit |
| 07:18.39 | gordonjcp | DUN DUN DUUUHHNNNNN |
| 07:18.49 | pygrammer | i'm beginning to wonder if it's Stanaphone's server configuration |
| 07:19.08 | pygrammer | problem is, it could be so many things |
| 07:19.18 | pygrammer | it's hard, if not impossible, to pinpoint the problem |
| 07:19.18 | glm2k | looks that way to me, if you haven't changed a thing on your end. |
| 07:19.36 | pygrammer | well, yeah. today i called it just to hear my beautiful voice, and what did i hear but the DEMO message playing |
| 07:19.51 | *** join/#asterisk colombus (n=colombus@193.190.210.151) |
| 07:19.53 | pygrammer | needless to say, i almost threw the box out the window |
| 07:19.54 | gordonjcp | pygrammer: try it with another SIP provider |
| 07:20.00 | pygrammer | i had thought i had conquered that shit |
| 07:20.13 | pygrammer | gordonjcp, like who? |
| 07:20.19 | gordonjcp | <shrug> |
| 07:20.21 | pygrammer | stana's free incoming :) |
| 07:20.30 | pygrammer | don't have to pay anything |
| 07:20.33 | gordonjcp | well, you're only looking to test it, right? |
| 07:20.37 | pygrammer | yeah. |
| 07:20.45 | pygrammer | that's why i have stana |
| 07:20.57 | gordonjcp | does it actually matter if they are even in the US? |
| 07:21.06 | gordonjcp | set up a sipgate number |
| 07:21.09 | *** join/#asterisk tuyan (n=tuyan@81.214.255.57) |
| 07:21.22 | *** join/#asterisk _m_ (n=m@fbta199.fbta.uni-karlsruhe.de) |
| 07:22.13 | pygrammer | the heck is sipgate |
| 07:22.35 | glm2k | hehe, i rmember someone asking the same of stana earlier today |
| 07:23.00 | glm2k | or was it yesterday? |
| 07:23.04 | gordonjcp | possibly stanaphone is broken |
| 07:23.11 | pygrammer | gordonjcp, yeah, could gbe |
| 07:23.12 | pygrammer | *be |
| 07:23.51 | pygrammer | glm2k, well, i only know of stanaphone because some friends suggested trying it when it was entirely free as a beta |
| 07:24.02 | *** join/#asterisk Martohtar (i=Martohta@82.196.218.80) |
| 07:24.21 | glm2k | halfway free is still good enough |
| 07:25.32 | pygrammer | :D |
| 07:27.35 | *** part/#asterisk m (n=maninder@220.224.39.63) |
| 07:28.35 | *** join/#asterisk wzl (n=Miranda@82-169-62-42-mx.xdsl.tiscali.nl) |
| 07:28.56 | wzl | good morning all yeah i know .. to late @work :-S |
| 07:30.59 | pygrammer | glm2k, how would I test the sipgate number if I'm not in the UK? |
| 07:31.01 | *** join/#asterisk afv-13 (n=adrianvv@196.37.3.190) |
| 07:31.12 | pygrammer | i don't think i can call internationally on my cell phone, and i don't think i should do that on my land-line |
| 07:31.13 | *** join/#asterisk t-pa (i=tr67200@kielo.uta.fi) |
| 07:31.31 | afv-13 | could someone please reccomend a softphone, i'm getting tired of xlite |
| 07:32.01 | pygrammer | afv-13, make your own ^_^ |
| 07:32.09 | drray | moziax |
| 07:32.11 | RaYmAn-Bx | pygrammer: just get someone in the UK you call the number |
| 07:32.30 | RaYmAn-Bx | you/to |
| 07:32.34 | pygrammer | no, i'm not testing call quality, but debugging this on the CLI |
| 07:33.11 | pygrammer | maybe Stana's having problems |
| 07:33.14 | glm2k | what RaYmAn-Bx said |
| 07:33.21 | pygrammer | that would ideal |
| 07:33.23 | pygrammer | *be |
| 07:33.28 | glm2k | but? |
| 07:33.37 | glm2k | no one you know in the UK? |
| 07:33.56 | pygrammer | not who're on |
| 07:34.04 | *** join/#asterisk m (n=maninder@220.224.39.63) |
| 07:34.17 | m | hi |
| 07:34.26 | RaYmAn-Bx | perhaps someone UK in here might be willing? I've successfully had that happen before when testing my UK numbers |
| 07:34.40 | m | hi |
| 07:34.44 | m | how ru |
| 07:34.48 | m | i am maninder |
| 07:34.53 | m | <PROTECTED> |
| 07:34.57 | glm2k | we heard you the first time m |
| 07:35.22 | *** join/#asterisk Syncros (n=sysop@noc.routermonkey.net) |
| 07:35.28 | m | i am using hi5 frist time |
| 07:35.58 | m | is any body want to become my friend |
| 07:36.06 | glm2k | m: you're using * for the first time? or did i get you wrong? |
| 07:36.24 | pygrammer | or maybe my computer got fried during a power outage a couple days ago ... |
| 07:36.37 | glm2k | pygrammer: not likely |
| 07:36.40 | pygrammer | i was too lazy to hook it up to a surge protector :D |
| 07:36.40 | m | no i am using it frist time |
| 07:36.55 | pygrammer | it's only a 400mhz pII, so i don't really give a shit if it dies |
| 07:36.59 | pygrammer | my * server |
| 07:37.02 | glm2k | hehe |
| 07:37.15 | pygrammer | only 128 megs of RAM, so it's constantly bottle-necked |
| 07:37.26 | m | glm2k:halo |
| 07:37.26 | pygrammer | 124/128 consistently |
| 07:37.54 | m | halo where ru |
| 07:38.03 | *** join/#asterisk t-pa (i=tr67200@kielo.uta.fi) |
| 07:38.07 | m | do you want to chat with me |
| 07:38.29 | pygrammer | about what? |
| 07:39.04 | pygrammer | maybe stanaphone's changing their hardware or something |
| 07:39.28 | gordonjcp | heh |
| 07:39.30 | pygrammer | i had gotten rid of the 401 seemingly permanently, and now it has come back to haunt me |
| 07:39.40 | gordonjcp | if they get working again, then I might register a number there |
| 07:39.40 | pygrammer | haunt/taunt |
| 07:39.51 | pygrammer | site works: http://www.stanaphone.com |
| 07:39.58 | pygrammer | might as well; it's a free NYC number |
| 07:40.04 | pygrammer | not 212 though |
| 07:40.09 | m | pygrammer: hi |
| 07:40.21 | m | hi |
| 07:40.27 | m | how r u |
| 07:40.34 | pygrammer | i found a couple 212 numbers on didx.org |
| 07:40.36 | pygrammer | pretty sweet |
| 07:40.39 | gordonjcp | pygrammer: ah |
| 07:40.59 | pygrammer | m, i'm good |
| 07:41.03 | pygrammer | a bit tired |
| 07:41.12 | gordonjcp | shiiiit |
| 07:41.13 | pygrammer | how about you? |
| 07:41.21 | gordonjcp | 2.5c per minute to call the UK? |
| 07:41.29 | pygrammer | :-o |
| 07:41.31 | pygrammer | you're in the uk? |
| 07:41.33 | gordonjcp | yeah |
| 07:41.36 | m | pygrammer: do you want to chat with me |
| 07:41.41 | gordonjcp | man |
| 07:41.49 | *** join/#asterisk sonja (n=hellowor@thes730a-762.otenet.gr) |
| 07:41.54 | gordonjcp | if the lag wasn't too bad I'd almost be cheaper calling from the US |
| 07:42.21 | pygrammer | 2.5c is cheap |
| 07:42.24 | gordonjcp | yeah |
| 07:42.25 | *** join/#asterisk nagl (n=nagl@rih.zid-nw.wu-wien.ac.at) |
| 07:42.29 | gordonjcp | it's a little over a penny a minute |
| 07:43.15 | gordonjcp | what does it cost to call an NY number from outside, possibly another state? |
| 07:43.30 | pygrammer | heh, it's trivial nowadays; i think my land-line has unlimited long distance |
| 07:43.34 | pygrammer | celly is free |
| 07:43.42 | *** join/#asterisk BadaBing (n=aljishi@c-24-128-136-9.hsd1.ma.comcast.net) |
| 07:43.47 | gordonjcp | cool |
| 07:43.56 | BadaBing | hello guyz |
| 07:44.02 | gordonjcp | hi m |
| 07:44.49 | gordonjcp | damn me, it's cheaper to call a mobile in Iraq than a landline! |
| 07:45.05 | BadaBing | can anyone help when it comes to securing asterisk? |
| 07:45.05 | m | hi gordonjcp |
| 07:45.14 | pygrammer | haha |
| 07:45.34 | m | hi |
| 07:45.35 | gordonjcp | 29.8c/min |
| 07:45.43 | m | pry |
| 07:45.46 | gordonjcp | that's cheaper than off-network mobile-to-mobile |
| 07:45.55 | *** join/#asterisk shaon (n=sss@dsl-58-6-34-163.nsw.westnet.com.au) |
| 07:46.03 | gordonjcp | dammit, I can't phone my gf for less than that! |
| 07:46.39 | *** part/#asterisk sonja (n=hellowor@thes730a-762.otenet.gr) |
| 07:47.19 | *** join/#asterisk MmmmToop (n=chatzill@c1-67-3.rndf.isadsl.co.za) |
| 07:48.33 | shaon | hi |
| 07:48.48 | MmmmToop | hi |
| 07:49.05 | shaon | Help Capi dialing problem |
| 07:49.20 | pygrammer | ?? |
| 07:49.59 | *** join/#asterisk huehner (n=huehner@kt-gmbh.de) |
| 07:50.23 | m | shaon: hi |
| 07:50.43 | shaon | I have c2 card so i want to dial out with any available line how do i do that? if i assign MSN it works great |
| 07:51.48 | gordonjcp | pygrammer: quite tempted to set one up... |
| 07:52.52 | gordonjcp | awww wtf? |
| 07:53.04 | gordonjcp | it requires a US phone number before you can register |
| 07:53.58 | gordonjcp | pygrammer: ping? |
| 07:54.18 | shaon | Help!! Dial(CAPI/???????) |
| 07:54.24 | pygrammer | sorry |
| 07:54.29 | pygrammer | i was looking at asterisk consultants |
| 07:54.32 | pygrammer | back |
| 07:54.33 | pygrammer | ping |
| 07:55.31 | *** join/#asterisk |Fraeggl| (n=Fraeggl@rkom.r-kom.de) |
| 07:56.07 | *** join/#asterisk razu (n=razu@217-159-242-106-dsl.est.estpak.ee) |
| 07:56.31 | Delvar | ~ping |
| 07:56.34 | jbot | pong |
| 07:56.53 | pygrammer | gordonjcp, just put a bogus phone number |
| 07:56.56 | pygrammer | you know those things don't check :) |
| 07:57.03 | pygrammer | 7323491652 |
| 07:57.04 | pygrammer | anything |
| 07:57.13 | gordonjcp | heh |
| 07:57.23 | Delvar | eh! how did you know my number? |
| 07:57.34 | pygrammer | the only thing i've ever seen check was 1&1, which automatically called me when i set up an account |
| 07:57.40 | pygrammer | that's fucking stupid |
| 07:57.44 | gordonjcp | pygrammer: I should put my 50p/min number, in case they send marketing calls to it |
| 07:57.51 | pygrammer | i have about 5 numbers i could use to set up accts with |
| 07:57.56 | pygrammer | they wont' |
| 07:57.58 | pygrammer | but sure |
| 07:58.15 | gordonjcp | I do try to get sales calls on that |
| 07:58.23 | gordonjcp | I mean I actively sign that up for phone spam |
| 07:58.24 | pygrammer | what area code? |
| 07:58.30 | pygrammer | hahah |
| 07:58.38 | gordonjcp | 07031 - looks kind of like a UK mobile number |
| 07:58.50 | pygrammer | that's a US number? |
| 07:58.55 | shaon | help ISDN dialplan!!!!!!!!!!!!!!! |
| 07:59.34 | pygrammer | !!!!!!!!!!!!1111111111111111111 |
| 07:59.36 | gordonjcp | no, that's a UK mobie number, it seems they will take a country code |
| 07:59.49 | gordonjcp | !!!11!!!1llll11!one |
| 08:00.00 | pygrammer | oh, ok |
| 08:00.03 | pygrammer | one |
| 08:00.09 | pygrammer | two |
| 08:00.10 | pygrammer | three |
| 08:00.12 | *** join/#asterisk Freman (n=twitsrus@ppp115-4.lns1.bne3.internode.on.net) |
| 08:00.13 | gordonjcp | pygrammer: ok, next daft question, there are five NY-ish area codes |
| 08:00.31 | pygrammer | don't matter which you get; i got 347 |
| 08:00.39 | pygrammer | 646 is manhattan |
| 08:00.41 | gordonjcp | ok, I was going to ask, what are the areas like? |
| 08:00.42 | pygrammer | 347 is queens |
| 08:00.50 | shaon | CAPI dial help |
| 08:00.58 | pygrammer | don't remember the others/their physical locations |
| 08:01.08 | Freman | ok... I've dialed my extension... how can I match all #'s without stopping other matches? |
| 08:01.10 | gordonjcp | I mean I don't want a phone number that sounds like it terminates in a tin shack with a rusty F150 with no wheels outside |
| 08:01.21 | gordonjcp | ;-) |
| 08:01.48 | pygrammer | that's not NYC for you -- anywhere :) |
| 08:02.10 | *** join/#asterisk zoa2 (n=kkk@83.228.8.96) |
| 08:02.13 | pygrammer | it's more caddies with spinning rimz in front of housing projects |
| 08:02.41 | *** join/#asterisk brainy_ (i=brainy@blub.tnib.de) |
| 08:02.45 | brainy_ | hi there |
| 08:02.53 | pygrammer | hi here |
| 08:03.42 | gordonjcp | pygrammer: hehe, not so very different from where I live here then... |
| 08:03.58 | zoa2 | hi ho |
| 08:04.19 | pygrammer | gordonjcp, that's not manhattan though |
| 08:04.40 | brainy_ | is there know bug in chan_h323 that sometimes don |
| 08:04.43 | pygrammer | that's probably queens or so |
| 08:04.51 | brainy_ | don't recognize hangups? |
| 08:04.56 | pygrammer | oh man, it's late -- 4:04 AM |
| 08:05.04 | pygrammer | or, depending on how one looks at it, early |
| 08:05.11 | gordonjcp | pygrammer: heh |
| 08:05.11 | pygrammer | <PROTECTED> |
| 08:05.17 | gordonjcp | man, I got up at about 4am this morning |
| 08:05.20 | glm2k | damn, 1 am. time to sleep |
| 08:05.21 | gordonjcp | (local 4am) |
| 08:05.29 | pygrammer | lol |
| 08:05.30 | brainy_ | 10am here;) |
| 08:05.40 | pygrammer | you all have screwed-up times :D |
| 08:05.40 | gordonjcp | 'course, I went back to bed for a bit, and got up at 0530 |
| 08:05.50 | glm2k | hmm, brainy_: somewhere in europe? |
| 08:06.06 | pygrammer | i remember when i went to london; i slept for like 15 hours straight since i was so thrown off by the sudden time change |
| 08:06.09 | brainy_ | glm2k: yup, germany |
| 08:06.14 | gordonjcp | pygrammer: heh |
| 08:06.27 | pygrammer | i'm not the biggest fan of london |
| 08:06.32 | gordonjcp | yeah |
| 08:06.35 | gordonjcp | london is a hole |
| 08:06.38 | pygrammer | its food is despicale |
| 08:06.40 | pygrammer | *despicable |
| 08:06.42 | *** join/#asterisk darkskiez (n=darkskie@194.247.78.146) |
| 08:06.52 | pygrammer | i liked cambridge |
| 08:06.56 | pygrammer | and seeing the greenwhich observatory |
| 08:06.56 | glm2k | ok, so working backward.. gordonjcp is in the uk, pygrammer is in the east coast and i'm bordering the pacific ocean. neat. |
| 08:07.12 | pygrammer | glm2k, i used to be on the West |
| 08:07.18 | pygrammer | we're originally from California |
| 08:07.22 | glm2k | we? |
| 08:07.26 | pygrammer | my family |
| 08:07.28 | glm2k | your whole family? |
| 08:07.30 | pygrammer | yes |
| 08:07.31 | glm2k | ah. i see. |
| 08:07.35 | pygrammer | my mom's from SF, dad's from LA |
| 08:07.39 | glm2k | nice |
| 08:07.45 | pygrammer | SF is awesome |
| 08:07.48 | glm2k | aye |
| 08:07.59 | pygrammer | albeit expensive |
| 08:08.04 | glm2k | can't get enough of it. of course, haveing a bunch of fun loving cousins help a lot |
| 08:08.10 | gordonjcp | pygrammer: you should come to Scotland, the food is wonderful |
| 08:08.11 | pygrammer | :) |
| 08:08.25 | pygrammer | gordonjcp, i've always wanted to go to scotland |
| 08:08.29 | pygrammer | i want to play st. andrews :) |
| 08:08.40 | glm2k | gordonjcp: i might just drop by for some scones >:) my parents live in newcastle upon tyne |
| 08:09.05 | pygrammer | gordonjcp, have you been to st. andrewsw? |
| 08:09.12 | pygrammer | bah i can't type at this hour |
| 08:09.12 | gordonjcp | glm2k: I was down there a month or so back |
| 08:10.14 | glm2k | haven't been there myself...yet |
| 08:10.30 | gordonjcp | pygrammer: yes, but I don't play golf |
| 08:10.36 | pygrammer | :( i do |
| 08:10.39 | gordonjcp | I haven't been there for about 5 years though |
| 08:10.42 | gordonjcp | maybe more |
| 08:10.46 | pygrammer | beautiful place, eh? |
| 08:10.48 | gordonjcp | yeah |
| 08:10.53 | *** join/#asterisk Astarta (n=vesi_pz@217-10-251-242-cable.telecablenet.com) |
| 08:10.59 | *** join/#asterisk dexteruk (n=dexteruk@217.165.98.194) |
| 08:11.17 | *** join/#asterisk _m_ (n=m@fbta199.fbta.uni-karlsruhe.de) |
| 08:11.21 | pygrammer | haha |
| 08:11.24 | gordonjcp | now I need to go up to Aberdeen to pick the same bloody PDP-11s up from there |
| 08:11.27 | pygrammer | why were you picking up pdp-11s? |
| 08:11.38 | gordonjcp | 'cos they were getting skipped otherwise |
| 08:11.49 | darkskiez | Coz the women told him to f-off |
| 08:11.54 | pygrammer | lol |
| 08:12.01 | dexteruk | I am using an Digium Analog TDM but it is not disconnecting on busy tone |
| 08:12.24 | gordonjcp | darkskiez: nah, you've got to have the PDP-11s before you get the women |
| 08:12.31 | darkskiez | lo gordonjcp, i got confused as to what channel I was in there! |
| 08:12.37 | gordonjcp | heh |
| 08:12.41 | gordonjcp | are you coming tonight? |
| 08:12.49 | gordonjcp | (just to confuse the issue further) |
| 08:12.54 | glm2k | i'll settle for having the women _on_ the pdp-11s |
| 08:13.26 | pygrammer | what's the industry standard for CRM software? |
| 08:13.26 | darkskiez | i probably should, but i aint gonna learn much as the topic _is_ my job |
| 08:13.28 | pygrammer | anyone know |
| 08:13.35 | pygrammer | what a company like dell or microsoft would uswe? |
| 08:13.37 | pygrammer | *use |
| 08:13.47 | gordonjcp | glm2k: in my case, that would involve raising the ceiling |
| 08:13.56 | glm2k | gordonjcp: lol |
| 08:13.56 | *** join/#asterisk RoyK (n=roy@cm-80.111.22.187.chello.no) |
| 08:13.57 | dexteruk | has anyone used audacity? because i have used this propgram and the busy tone i have checked is 335/400,0/350,335/230,0/530 |
| 08:14.10 | Astarta | yes, I use it |
| 08:14.12 | *** join/#asterisk jordyke (n=xxxx@ip-213-49-85-123.dsl.scarlet.be) |
| 08:14.20 | gordonjcp | pygrammer: we use a weird homebrewed thing running on an AS/400 |
| 08:14.32 | gordonjcp | probably written in Rexx or more likely PL/1 |
| 08:14.39 | pygrammer | who's we? |
| 08:14.39 | Astarta | but I think 'Audio Recorder' is much better |
| 08:14.50 | gordonjcp | dexteruk: sounds about right for the UK congestion tone |
| 08:14.56 | shaon | help CAPI dial application!!!!!!!!!!!!!!!!!!!!!! |
| 08:15.01 | gordonjcp | the second beep should be a little louder too |
| 08:15.04 | *** join/#asterisk r0d3nt (n=RatMan@telephreak.net) |
| 08:15.05 | gordonjcp | shaon: RTFM |
| 08:15.27 | gordonjcp | shaon: or ask a question, rather than just throwing random words out |
| 08:15.32 | dexteruk | On on audacity is the duration is easy but the freq is that the hight of the wave form |
| 08:15.40 | gordonjcp | np |
| 08:15.41 | gordonjcp | no |
| 08:15.47 | shaon | gordonjcp:what is RTFM |
| 08:15.53 | gordonjcp | Read The Fine Manual |
| 08:15.54 | darkskiez | quality |
| 08:16.02 | pygrammer | damn i'm tired |
| 08:16.04 | pygrammer | good night, all |
| 08:16.08 | gordonjcp | shaon: there's this great website that can tell you all about it |
| 08:16.13 | gordonjcp | it's at www.google.com |
| 08:16.19 | *** join/#asterisk musical_Duck (n=kvirc@wblv-146-235-27.telkomadsl.co.za) |
| 08:16.23 | gordonjcp | nn pygrammer |
| 08:16.31 | pygrammer | ;) |
| 08:16.32 | gordonjcp | dexteruk: frequency is how far apart the peaks are |
| 08:16.34 | darkskiez | gordonjcp: you're on form today |
| 08:16.36 | gordonjcp | how *frequent* they are |
| 08:16.38 | shaon | gordonjcp: how to dial use a group in capi cannels |
| 08:16.44 | gordonjcp | darkskiez: ;-) |
| 08:16.59 | darkskiez | gordonjcp: depends if u have audacity in FFT mode or waveform mode. |
| 08:17.02 | gordonjcp | shaon: I don't know what that is, are you asking me a question |
| 08:17.07 | gordonjcp | darkskiez: true enough |
| 08:17.26 | gordonjcp | shaon: or are you describing something, or what? |
| 08:17.43 | dexteruk | top and bottom? or from mid line to the top? |
| 08:18.29 | shaon | gordonjcp: so far i hvent found any information about group dialing in capi. in zap we use Dial(zap/g1/XXXX but capi does not support capi/gq/xxx |
| 08:18.33 | darkskiez | dexteruk: from looking at that tone text, not listening, it looks like network congestion rather than busy |
| 08:18.39 | dexteruk | could i send anyone this wav file to double check my reading? |
| 08:19.38 | darkskiez | dexteruk: in /etc/asterisk/indications.conf has the uk tones listed |
| 08:20.06 | darkskiez | they even have some sarcastic comments about british wordings |
| 08:20.13 | shaon | how to use Dial(CAPI/g1/XXXXXX) |
| 08:20.17 | dexteruk | gordonjcp: would you have time to double check my wav file? |
| 08:20.25 | darkskiez | ; Busy is also called "Engaged" |
| 08:20.25 | darkskiez | busy = 400/375,0/375 |
| 08:21.09 | dexteruk | No im in the UAE and the exchanges are configured differently from place to place in the same city |
| 08:21.27 | darkskiez | dexterUK: sorry, your nick confused me |
| 08:21.56 | dexteruk | Yes it my nick i have had for many years |
| 08:22.40 | dexteruk | I have recorded the busy tone that i am getting, and i have tryed to us audacity to get the reading i just need to double check it |
| 08:23.42 | dexteruk | i get something like 335/400,0/350,335/230,0/530 |
| 08:23.59 | *** join/#asterisk Martohtar (i=Martohta@82.196.218.80) |
| 08:24.19 | dexteruk | darkskiez: could you look at the wav file for me? |
| 08:24.40 | dexteruk | darkskiez: and see what you get? |
| 08:25.04 | Freman | gah |
| 08:25.08 | gordonjcp | dexteruk: I can't at the moment, I'm at work |
| 08:25.09 | darkskiez | dexteruk: not really no, i dont have the tools on this computer |
| 08:25.12 | Freman | dialplans need the ability to match and continue |
| 08:25.13 | darkskiez | and im at wokr |
| 08:25.21 | dexteruk | ok |
| 08:25.24 | gordonjcp | shaon: maybe it doesn't do it yet |
| 08:25.32 | dexteruk | thanks guys |
| 08:25.37 | timecop | did fwd jsut take a shit? |
| 08:25.57 | Freman | IE... something like exten => ^_X.,1,Dosomething - but don't stop matching |
| 08:25.58 | darkskiez | oh timecops back! |
| 08:26.00 | darkskiez | yay |
| 08:26.15 | darkskiez | I always enjoy your eloquence |
| 08:26.49 | timecop | right. |
| 08:28.24 | Freman | is there such a match in asterisk? |
| 08:29.18 | dexteruk | ok in Audacity how do i check the Freq |
| 08:29.32 | darkskiez | dexteruk: put it into FFT mode |
| 08:29.37 | darkskiez | or do FFT analysis |
| 08:29.45 | darkskiez | not used it in ageeess |
| 08:30.06 | dexteruk | Where is FFT mode? |
| 08:30.35 | darkskiez | dexter: i've not used audacity in years, i dont know, but look for fft, it is the key |
| 08:31.30 | *** part/#asterisk jordyke (n=xxxx@ip-213-49-85-123.dsl.scarlet.be) |
| 08:31.55 | dexteruk | ok looking i have fft filters |
| 08:32.44 | dexteruk | got it its the freq analyser |
| 08:34.43 | dexteruk | darkskiez i have the ploted freq |
| 08:40.44 | jalsot | hi |
| 08:41.43 | jalsot | does anybody know what to set to have asterisk core dump files? I'm getting segfault, but no core logged [using safe_asterisk] |
| 08:42.30 | *** join/#asterisk Ogun (n=johangri@h236n2fls34o865.telia.com) |
| 08:42.36 | JerJer | <PROTECTED> |
| 08:42.54 | dexteruk | darkskiez: Ok FFT is the Spectrograph, when you see the sound as a spectrum? |
| 08:43.15 | jalsot | I don't have there |
| 08:43.45 | gordonjcp | not bad, I get 100ms to sip.stanaphone.com |
| 08:43.47 | gordonjcp | usable... |
| 08:43.50 | jalsot | I found one core file, but asterisk crashed more than once [that core dump is from earlier] |
| 08:43.59 | jalsot | if I have core file, what next? |
| 08:45.09 | darkskiez | gdb asterisk corefile; then bt full |
| 08:45.32 | jalsot | No symbol table info available |
| 08:46.03 | darkskiez | then paste the bt full output to pastebin.ca |
| 08:46.10 | jalsot | ok, moment... |
| 08:47.04 | jalsot | http://pastebin.ca/21089 |
| 08:47.51 | *** join/#asterisk fulgas (n=fulgas@213.58.130.46) |
| 08:50.16 | darkskiez | OH |
| 08:50.29 | jalsot | :) |
| 08:50.33 | jalsot | any hint? |
| 08:51.00 | darkskiez | thats not very helpful |
| 08:51.01 | darkskiez | bugger |
| 08:51.14 | darkskiez | warning: exec file is newer than core file. |
| 08:51.28 | darkskiez | you've rebuilt asterisk after that corefile was produced |
| 08:51.35 | jalsot | ah, yes |
| 08:51.37 | darkskiez | it does not make sense anymore |
| 08:51.40 | *** join/#asterisk rkvarala (n=rkvarala@59.93.52.30) |
| 08:51.41 | jalsot | I hoped it will help |
| 08:51.46 | fulgas | hey |
| 08:51.50 | jalsot | but crashes remain :( |
| 08:52.23 | qweee | is there any software to rip the source code of a dialpad |
| 08:52.29 | darkskiez | do u have the older binary? |
| 08:52.42 | darkskiez | qweee: a dialpad ? |
| 08:52.48 | rkvarala | in asterisk do we have any posibility to send info / sip packets |
| 08:52.50 | jalsot | I will take a look |
| 08:53.53 | *** join/#asterisk djin_ib (n=djin_ib@gridfox.xs4all.nl) |
| 08:54.26 | qweee | ues a softphone |
| 08:54.37 | jalsot | darkskiez: is it enough if I use just asterisk binary from oldel version? |
| 08:54.46 | qweee | any open source soft phones there |
| 08:55.12 | djin_ib | I just found some info on 'fxotune'. However it's missing in the zaptel source. Am I overlooking something? |
| 08:55.17 | jalsot | I found it, but under old source tree, so modules are new in the normal filesystem |
| 08:56.00 | *** join/#asterisk [1]JohnJacob (n=JohnJaco@pcp0011543387pcs.mainf01.in.comcast.net) |
| 08:56.32 | darkskiez | might have some luck anyway |
| 08:56.37 | jalsot | darkskiez: new gdb log: http://pastebin.ca/21090 |
| 08:56.42 | *** part/#asterisk Astarta (n=vesi_pz@217-10-251-242-cable.telecablenet.com) |
| 08:56.47 | qweee | coz i want to know the way they froward the calls |
| 08:57.13 | darkskiez | jalson: dont use sqlite to log your CDR |
| 08:57.24 | darkskiez | jalson: thats seems to be whats screwing it |
| 08:57.31 | jalsot | darkskiez: hmm, why :) |
| 08:57.43 | darkskiez | jalson: dont know, but changing that should be a quick fix |
| 08:58.08 | jalsot | cdr_csv can be used? |
| 08:58.22 | jalsot | if yes, I can try to avoid sqlite and we will see |
| 09:00.00 | darkskiez | should be aye |
| 09:00.10 | darkskiez | its the default |
| 09:00.20 | jalsot | ok, will try |
| 09:02.55 | *** join/#asterisk pa (n=Paolo@unaffiliated/pa) |
| 09:03.40 | darkskiez | may still be borked tho, might be a problem with cdr customisations :/ |
| 09:04.49 | *** join/#asterisk paul_ (n=NL-PAUL@83.98.247.211) |
| 09:06.36 | *** join/#asterisk oden (n=oden@194-237-146-22.customer.telia.com) |
| 09:09.23 | gordonjcp | hahaha |
| 09:09.31 | gordonjcp | I have *way* too much time on my hands |
| 09:09.35 | jalsot | darkskiez: many thanks! I hope it will help |
| 09:09.51 | darkskiez | gordonjcp: what provoked that |
| 09:10.16 | gordonjcp | darkskiez: VNC into my desktop |
| 09:10.22 | gordonjcp | tweak a mix in Ardour |
| 09:10.26 | gordonjcp | mp3 it |
| 09:10.37 | gordonjcp | then feed it into MOH so I can phone it up to hear it |
| 09:10.56 | paul_ | After a not planned shutdown of the server (powerdown) ithe wcfxs module is not loading anymore. |
| 09:10.58 | paul_ | First Kudzu comes with the " Tiger Jet Network Inc. Intel 537 (rev 0)." and " Tiger Jet Network Inc. Intel 537 (#2) (rev 0)." |
| 09:10.59 | paul_ | sorry, in a hurry I say yes install :(, ok remoed the cards and hisax driver from modules.conf and /etc/sysconfig/hwconf in Fedora |
| 09:11.01 | paul_ | But when I try "modprobe wcfxs" I get this error message: |
| 09:11.02 | paul_ | /lib/modules/2.4.22-1.2199.nptl/misc/wcfxs.o: init_module: No such device |
| 09:11.04 | paul_ | Hint: insmod errors can be caused by incorrect module parameters, including invalid IO or IRQ parameters. |
| 09:11.05 | paul_ | <PROTECTED> |
| 09:11.07 | paul_ | /lib/modules/2.4.22-1.2199.nptl/misc/wcfxs.o: insmod /lib/modules/2.4.22-1.2199.nptl/misc/wcfxs.o failed |
| 09:11.08 | paul_ | /lib/modules/2.4.22-1.2199.nptl/misc/wcfxs.o: insmod wcfxs failed |
| 09:11.10 | paul_ | From "/proc/pci": |
| 09:11.11 | paul_ | <PROTECTED> |
| 09:11.13 | paul_ | <PROTECTED> |
| 09:11.15 | paul_ | <PROTECTED> |
| 09:11.16 | paul_ | <PROTECTED> |
| 09:11.18 | paul_ | <PROTECTED> |
| 09:11.20 | paul_ | <PROTECTED> |
| 09:11.21 | hugo-v6 | damn STOP |
| 09:11.21 | paul_ | <PROTECTED> |
| 09:11.23 | paul_ | <PROTECTED> |
| 09:11.25 | paul_ | <PROTECTED> |
| 09:11.26 | hugo-v6 | STOPIT |
| 09:11.27 | paul_ | <PROTECTED> |
| 09:11.29 | paul_ | <PROTECTED> |
| 09:11.31 | paul_ | <PROTECTED> |
| 09:11.33 | *** join/#asterisk MmmmToop (n=chatzill@196-11-241-43.mtn.co.za) |
| 09:11.33 | paul_ | There are two card insatalled one E1 single SPAN and one TDM400P card... |
| 09:11.34 | gordonjcp | paul_: PASTEBIN! |
| 09:11.35 | JerJer | grr |
| 09:11.35 | paul_ | Anyone some suggestions ???? |
| 09:11.43 | gordonjcp | paul_: yes, use pastebin in future |
| 09:11.47 | JerJer | yeah, don't be a dumbass |
| 09:11.56 | gordonjcp | floodage! |
| 09:11.59 | gordonjcp | man the pumps! |
| 09:12.03 | syle | why does off-hook mean online |
| 09:12.16 | paul_ | sorry, i will try to lear some more about irc |
| 09:12.17 | gordonjcp | syle: because the "receiver" is off the "hook" |
| 09:12.22 | *** join/#asterisk gambolputty (n=gambolpu@cblmdm72-240-241-108.buckeyecom.net) |
| 09:12.30 | syle | ahh ok |
| 09:12.32 | gordonjcp | paul_: pastebin.ca is your friend |
| 09:13.10 | paul_ | but some who knows somting about htis problem with loading the driver of the TDM400P? |
| 09:13.14 | gordonjcp | syle: on very very old telephones, where the mike was on the front and the earpiece was a big brass-and-bakelite thing on a cable, there was a hook to hang it off on the side of the box |
| 09:13.52 | gordonjcp | syle: eventually it occurred to someone that it would be a clever idea to put a hinge on the hook, and have it switch between telephone and bell when you hung the heavy receiver on it |
| 09:14.07 | gordonjcp | I know it's obvious now, but *someone* had to think of it |
| 09:17.11 | *** join/#asterisk Shoragan (n=shoragan@d072.apm.etc.tu-bs.de) |
| 09:17.37 | *** join/#asterisk litage (n=nick@203.201.99.104) |
| 09:18.09 | djin_ib | fyi, I was looking for fxotune earlier, but it's only in HEAD. |
| 09:19.14 | jalsot | does anybody know why I'm getting this? Aug 23 13:38:43 WARNING[8582] format_gsm.c: Short read (-1) (Bad file descriptor)! |
| 09:19.28 | JerJer | bad file descriptor |
| 09:19.49 | jalsot | also getting 'Bad file descriptor' sometimes starting FastAGI |
| 09:19.58 | *** join/#asterisk _mountie (n=mountie@24.42.99.232) |
| 09:20.34 | jalsot | JerJer: yes, but what is it caused by? |
| 09:26.24 | JerJer | are you attempting to run asterisk as a normal user? |
| 09:26.30 | *** part/#asterisk oden (n=oden@194-237-146-22.customer.telia.com) |
| 09:26.58 | JerJer | do you have an invalid IP address in bindaddr ? |
| 09:30.46 | jalsot | JerJer: yes, normal user, np IP problems - this problem happens only rarely |
| 09:31.50 | JerJer | like non-root ? |
| 09:32.20 | JerJer | if so you have to make sure all of the permissions are proper |
| 09:34.00 | rogercharlie | does the areski cdr - export to pdf work for anyone? |
| 09:34.18 | jalsot | yes, non-root, permissions are ok, it works like a charm, however time-to-time I'm getting that problem |
| 09:34.36 | DannyF | Morning folks |
| 09:34.39 | jalsot | do you think asterisk uses more than 1024 open files and this casuses that problem? |
| 09:34.50 | jalsot | 60 concurrent calls |
| 09:34.50 | JerJer | depends on laod |
| 09:34.52 | JerJer | load |
| 09:35.51 | jalsot | I see only ~260 open fds |
| 09:36.40 | jalsot | there are 60 calls generated by pbx_spool and connected to fastagi server |
| 09:37.04 | *** join/#asterisk [Airwolf] (n=airwolf@attilla.nl) |
| 09:39.01 | h3x0r | look at limit/ulimit |
| 09:39.07 | h3x0r | depending on your shell |
| 09:39.18 | jalsot | h3x0r: 1024 is allowed per user |
| 09:39.55 | jalsot | I tuned up to 4096, but didn't help |
| 09:40.08 | h3x0r | how did you do that |
| 09:40.21 | jalsot | pam+limits.conf |
| 09:40.45 | *** join/#asterisk Tili (i=Tili@219.136.218.164) |
| 09:40.45 | h3x0r | and your shell limit/ulimit is telling you the same number right |
| 09:40.52 | h3x0r | because there are several places that limits occur |
| 09:40.58 | jalsot | if I su to that user and do ulimit -a, it shows 4096 |
| 09:41.09 | h3x0r | did you do su -l |
| 09:41.17 | h3x0r | because if you didnt then... |
| 09:41.20 | Tili | how do we do pre-paid billing on 3 way calls. |
| 09:41.30 | h3x0r | its not the same as logging in as that user |
| 09:41.49 | jalsot | my su doesn't have -l option |
| 09:41.54 | h3x0r | oh |
| 09:42.16 | h3x0r | su - foo |
| 09:42.37 | h3x0r | depends on the distro i guess |
| 09:42.42 | jalsot | ok, did: got 4096 |
| 09:42.59 | h3x0r | alright just wanted to make sure |
| 09:42.59 | jalsot | however from current fd usage 1024 could be far enough |
| 09:43.35 | h3x0r | limits are set by the kernel, then pam or something else in the middle of login depending on the OS, and then by the shell so you never know |
| 09:43.40 | h3x0r | which one has the lowest limit wins |
| 09:44.06 | *** join/#asterisk Lan16`spdy^gTp (n=spdy@altom.net1.nerim.net) |
| 09:44.28 | spdy | hi |
| 09:44.33 | jalsot | any other thing to try? |
| 09:44.48 | h3x0r | are you using head or stable |
| 09:44.56 | jalsot | cvs head |
| 09:45.13 | jalsot | right CVS-NHEAD-08/24/05-13:12:18 |
| 09:45.29 | h3x0r | are you 64 bit |
| 09:45.29 | jalsot | but had the same issue with ~15 days older cvs as well |
| 09:45.44 | jalsot | 64 bit xeons running in 32 bit mode |
| 09:45.51 | h3x0r | ok |
| 09:46.15 | spdy | i've a problem, i use the command parkandannounce to park a call, calls are parked in 801-880, with this cmd i can dial someone when the parked user hear music, when the person take the call it say, transfer 801, and hangup, why its hanging up ? |
| 09:46.24 | Freman | hey |
| 09:46.31 | Freman | my enum returns sucsessfull... |
| 09:46.36 | Freman | even if it has no info )c; |
| 09:46.36 | jalsot | this problem happening only rarely, I wasn't able to find when and why |
| 09:47.06 | h3x0r | what are you doing with it |
| 09:47.20 | Freman | -- Executing EnumLookup("SIP/9100-7e1f", "61754993353") in new stack |
| 09:47.20 | Freman | ENUM got '0' |
| 09:47.20 | Freman | <PROTECTED> |
| 09:47.20 | Freman | <PROTECTED> |
| 09:47.20 | Freman | ^^^ That NoOp should display the value of ${ENUM} |
| 09:47.37 | *** join/#asterisk gambolputty (n=gambolpu@cblmdm72-240-241-108.buckeyecom.net) |
| 09:47.41 | jalsot | h3x0r: right IVR out |
| 09:47.53 | h3x0r | does fastcgi have anything to do with it |
| 09:47.58 | jalsot | h3x0r: call generated by pbx_spool and connected to fastagi server |
| 09:47.58 | h3x0r | are you using manager sockets or anything |
| 09:48.02 | spdy | someone know the pb ? |
| 09:48.06 | jalsot | yes, both |
| 09:48.29 | dexteruk | can anyone help me with indications |
| 09:48.32 | h3x0r | so you are connection pooling manager with fastcgi or just opening a new one every time you do something |
| 09:48.57 | jalsot | I do IVR in fastagi, so the connection is alive during the call |
| 09:49.14 | jalsot | every call has it's own socket than, so max. 60 to fastagi server |
| 09:49.35 | musical_Duck | Can someone help me to connect to kiax clients through my asterisk box? |
| 09:49.53 | h3x0r | hrm |
| 09:50.06 | dexteruk | i have an anaglod TDM and it is not hanging up the Zap channel when it receives busy tone |
| 09:50.18 | dexteruk | analog TDM sorry |
| 09:50.27 | jalsot | I checked res_agi.c but bad file descriptor message comes only after the socket has been successfully connected |
| 09:50.48 | jalsot | when it tryes to write the 1st line, than it is returning with this error message |
| 09:50.50 | h3x0r | i mean the only thing i can think of is your fd went away coz it closed or something |
| 09:51.20 | h3x0r | or maybe theres a bug in asterisk but youd think it would be on mantis |
| 09:51.23 | jalsot | on agiserver I see new connection but no data on it, than just teardown |
| 09:51.45 | jalsot | I checked mantis, but didn't find similar thing, yet |
| 09:52.18 | h3x0r | can you try stable |
| 09:52.51 | jalsot | unfortunately cannot |
| 09:53.18 | spdy | i've a transfer problem with parkandannounce cmd can someone help me ? |
| 09:53.32 | spdy | it dial but when the person take the call it hangup |
| 09:53.46 | jalsot | it can also be a problem with agiserver, but... |
| 09:54.01 | jalsot | but I have that another 'bad file descriptor' with reading gsm file |
| 09:54.33 | jalsot | WARNING[30939] format_gsm.c: Short read (-1) (Bad file descriptor)! |
| 09:54.49 | jalsot | I don't know if those things have any relation or not |
| 09:55.17 | h3x0r | well if it was a limit issue i think it would fail on open instead |
| 09:56.16 | jalsot | yes, probably |
| 09:56.56 | Freman | my enumlookup seems to be broken )c: |
| 09:57.01 | jalsot | right I found another info: ERROR[15878] cdr_custom.c: Unable to re-open master file /var/log/asterisk/cdr-custom/Master.csv : Bad file de |
| 09:57.41 | jalsot | this time it was not able to open |
| 09:57.42 | jalsot | hmmm |
| 09:57.56 | Freman | -- Executing EnumLookup("SIP/9100-9900", "143123489761754281400") in new stack |
| 09:57.56 | Freman | ENUM got '0' |
| 09:58.08 | Freman | and it continues executing )C: |
| 09:58.40 | wzl | hi guys .. i purchased 5 g729 licenses from digium for testing purposes. Registration went well, when i enter show translation on CLI i see g729 is enabled. Also allow=g729 is also entered. but still my extensions will not call with g729 what is it what i do wrong here? |
| 09:59.09 | RoyK | that was one hell of a phone number..... |
| 09:59.31 | RoyK | 22 digits :P |
| 10:00.42 | *** join/#asterisk Dovid (n=dovi5988@pool-141-150-20-167.mad.east.verizon.net) |
| 10:02.49 | wzl | hmm it works now ... |
| 10:03.22 | spdy | i've a transfer problem with parkandannounce cmd can someone help me ? |
| 10:03.50 | wzl | it seems that the SIP phone selected his own codec i will give priority to g729 ofcourse! |
| 10:04.14 | musical_Duck | Is music on hold working in the CVS HEAD? |
| 10:04.26 | *** join/#asterisk Mimmus (n=viggiani@ext.pitagora.it) |
| 10:04.28 | spdy | yes music on hold is working |
| 10:04.30 | jalsot | h3x0r: any other idea? |
| 10:04.34 | musical_Duck | thx |
| 10:04.43 | Freman | RoyK: I know.. I know it's invalid... but enum returns '0' and skips on to priority 4 rather then erroring and going to 104 |
| 10:04.59 | spdy | i've a transfer problem with parkandannounce cmd can someone help me ? |
| 10:05.07 | Mimmus | hi, how can i check the exit code of a 'system' command in the dialplan? |
| 10:05.45 | *** join/#asterisk meppl (n=mephisto@p54AAFE62.dip.t-dialin.net) |
| 10:09.14 | *** join/#asterisk bmg505 (n=leon@rndf-146-8-166.telkomadsl.co.za) |
| 10:16.57 | *** join/#asterisk adelas (n=booger@rrcs-24-199-21-141.west.biz.rr.com) |
| 10:18.49 | *** join/#asterisk MmmmToop_ (n=chatzill@c1-50-2.rndf.isadsl.co.za) |
| 10:21.30 | *** join/#asterisk gr0mit (n=w10277@fw.mot-tools.co.uk) |
| 10:23.58 | *** join/#asterisk dwmw2_gone (i=ctrlprox@baythorne.infradead.org) |
| 10:24.53 | *** join/#asterisk zagaya972 (n=d2s-comp@APointe-a-Pitre-102-1-9-116.w81-248.abo.wanadoo.fr) |
| 10:29.19 | *** join/#asterisk cpm (n=Chip@pdpc/supporter/sustaining/cpm) |
| 10:29.39 | *** join/#asterisk elmar69 (i=foobar@p5088E0A6.dip.t-dialin.net) |
| 10:30.14 | *** part/#asterisk cpm (n=Chip@pdpc/supporter/sustaining/cpm) |
| 10:33.00 | elmar69 | Hi, |
| 10:33.02 | elmar69 | I do have an question regarding queues: |
| 10:33.03 | elmar69 | The incoming call is hearing the "you are first". The phone connected to the queue starts ringing after the announcement is finished. |
| 10:33.05 | elmar69 | How can I get the queue ring the phone immediately - the announcement can be abortet once the call is answered by an human. |
| 10:34.24 | *** join/#asterisk ncjp (n=switch@61.206.115.5.user.ad.il24.net) |
| 10:34.51 | *** join/#asterisk switch (n=switch@61.206.115.5.user.ad.il24.net) |
| 10:36.04 | *** join/#asterisk surfdue (n=surfdue@user-0c6t1g9.cable.mindspring.com) |
| 10:39.05 | shaon | helo CAPI dialplan plz |
| 10:39.13 | shaon | help CAPI dialplan plz |
| 10:41.17 | Freman | why don't that fkn thing work |
| 10:41.25 | Freman | if it's not resolving, it should be n+101 |
| 10:41.43 | *** join/#asterisk ennuyeux72 (n=ennuyeux@host-83-146-53-40.bulldogdsl.com) |
| 10:47.00 | syle | whats better sox or wmin for mixing sound files? |
| 10:47.04 | syle | wmix |
| 10:47.12 | jontow | i <3 sox |
| 10:47.15 | jontow | but its personal preference |
| 10:50.09 | syle | be nice if monitor application did that on its own |
| 10:51.35 | jontow | so write a patch :) |
| 10:51.44 | *** join/#asterisk qwer^^ (n=muja@210.212.195.134) |
| 10:52.51 | syle | you using mpg123 or madplay? |
| 10:53.09 | jontow | mpg123 |
| 10:53.19 | syle | i am just wondering if madplay processes hang if asterisk doesn;t start up properly like mpg123 |
| 10:54.31 | syle | ie: forgetting to load zap modules beforehand , or problem with zap config when loading zap drivers.....hangs mpg123 processes using 99% cpu |
| 10:54.42 | Bhaal | Can someone point me to a good doc on setup iax between 2 servers? |
| 10:54.58 | syle | yeah i can one sec |
| 10:55.22 | syle | http://www.voip-info.org/tiki-index.php?page=Asterisk%20-%20dual%20servers |
| 10:55.38 | Bhaal | Thanks |
| 10:55.38 | syle | i went with example 2 personally |
| 10:55.43 | Bhaal | Ahh |
| 10:55.59 | jontow | i used the samples from install time.. |
| 10:58.54 | Bhaal | syle: I dont need to worry about rsa auth etc? |
| 10:59.08 | *** join/#asterisk surfdue (n=surfdue@unaffiliated/surfdue) |
| 10:59.48 | syle | up to you , standard seems to be md5 |
| 11:00.36 | syle | i wouldn;t even worry about putting anything for auth till you have it working |
| 11:00.48 | syle | then come back to it |
| 11:01.08 | qwer^^ | can someone help me ,we have a problem with our asterisk server ,its forwarding calls to n2p but n2p ios regecting it is tere any way in which i can make is work |
| 11:01.28 | *** join/#asterisk Weezey (i=WeezeyD@206.210.109.233) |
| 11:01.54 | Weezey | if I'm setting up a bunch of users, should they each have their own contexts for outbound dialling? |
| 11:02.10 | *** join/#asterisk surfdue (n=surfdue@user-0c6t1g9.cable.mindspring.com) |
| 11:02.10 | Ogun | Is it possible to forwad a call into a meetme conference? Can't get it to work. |
| 11:02.16 | gordonjcp | Weezey: depends |
| 11:02.21 | *** join/#asterisk colombus (n=colombus@193.190.210.151) |
| 11:02.30 | jontow | weezey; the idea is more what privilege levels you want to set them at, or what specific behaviors they need when dialing |
| 11:02.39 | *** join/#asterisk the_phantom (n=the_phan@59.95.161.223) |
| 11:03.00 | qwer^^ | can someone help me ,we have a problem with our asterisk server ,its forwarding calls to n2p but n2p ios regecting it is tere any way in which i can make is work |
| 11:03.14 | Weezey | gordonjcp: well, that's how I'm doing it now, (or by groups of people for a company), but that doesn't really work with the realtime stuff that I really should implement to make automation easier. |
| 11:03.31 | jontow | qwer; stop repeating yourself.. you're not even fixing the typos between repeats. |
| 11:03.43 | jontow | qwer; and also, your question is too vague |
| 11:04.16 | the_phantom | can anyone tell me which sip rfc asterisk currently implements and if it supports the older rfc's |
| 11:04.30 | qwer^^ | sorry |
| 11:04.53 | qwer^^ | jontow: sorry but can u help me |
| 11:05.08 | Weezey | gordonjcp: I'm thinking that to continue to make it work, I'll just have to develop some code that supports automation of context additions too. |
| 11:05.10 | ai-a | can i install Asterisk@Work's web frontend onto my already installed asterisk? |
| 11:05.26 | jontow | qwer; if the remote end is rejecting it.. then you need to figure out why; analyse the error its giving you.. then check your permissions at least 3 times .. 'n2p' isn't a familiar term to me so i can't really say how.. just that its your best bet at this point |
| 11:05.32 | jontow | qwer; also.. google the heck out of it :) |
| 11:05.49 | *** join/#asterisk zoo (i=nobody@ip-142-16.travedsl.de) |
| 11:05.54 | jontow | ai-a; if you have the energy.. you can do roughly anything. |
| 11:08.02 | qwer^^ | jontow:n2p -net2phone |
| 11:08.19 | jontow | ah.. not familiar indeed :) |
| 11:08.58 | qwer^^ | jontow: we can traceroot te n2p frm my ip but i cant tracert frm asterisk does it mean tat te asterisk ip is blocked |
| 11:09.19 | pa | google talk: do you know it? |
| 11:09.38 | jontow | qwer; very possible.. does the asterisk box have trouble getting to anything else? |
| 11:09.51 | Bhaal | syle: Okey, serverB is rejecting serverA .. Saying no registration for the username I put in... |
| 11:10.04 | jontow | bhaal; forget to 'reload chan_iax2.so' ? |
| 11:10.07 | qwer^^ | no we can even tracert yahoo |
| 11:10.09 | pa | can i use a SIP client to talk in google talk? |
| 11:10.19 | qwer^^ | but not n2p ip |
| 11:10.27 | Mavvie | pa: that's google voip, not google talk :-) |
| 11:11.02 | pa | Mavvie: so google talk is not voip? |
| 11:11.21 | Bhaal | jontow: no I restarted, but I think I spot the problem |
| 11:11.23 | Weezey | pa: google talk is with a keyboard |
| 11:11.26 | jontow | pa; google talk is, iirc, an instant messenger service.. |
| 11:11.35 | Weezey | jabber |
| 11:11.37 | pa | oh ok |
| 11:11.59 | *** join/#asterisk peat (n=pete@flooble.terminus.net.au) |
| 11:12.01 | pa | but i see also a phone icon in the talk.google.com |
| 11:12.09 | Mavvie | pa: http://www.google.com/talk/about.html |
| 11:12.37 | pa | is not "talk with" voip? |
| 11:13.08 | jontow | actually it does appear to deal with voice |
| 11:13.24 | Mavvie | just like that Yahoo! Chat support voice, it still doesn't make it a phone application. |
| 11:13.30 | *** join/#asterisk Mother_ (n=m@53.Red-217-126-93.pooles.rima-tde.net) |
| 11:13.37 | Mother_ | multiple greetings! |
| 11:13.40 | pa | oh ok |
| 11:13.44 | gordonjcp | you could do a kind of CB-like chat by sending gsm files over jabber |
| 11:14.14 | *** join/#asterisk Red15 (n=irc@d54C323D1.access.telenet.be) |
| 11:14.22 | Red15 | hello all |
| 11:14.34 | qwer^^ | hello red15 |
| 11:14.43 | Red15 | i have a quick question |
| 11:14.52 | qwer^^ | red15: ya tell me |
| 11:15.03 | Red15 | i'm trying to install * on a CentOS 4.1 with a AVM Fritz card |
| 11:15.22 | qwer^^ | k |
| 11:15.24 | Red15 | I already got it working using isdn4linux and modem.conf settings |
| 11:15.44 | qwer^^ | k |
| 11:15.50 | Red15 | but it seems the AMP (Asterisk Management Portal) doest not like modem.conf settings and won't allow me to make any such settings |
| 11:15.59 | musical_Duck | I am having some trouble with festival, the first bit of speech gets cut off, any thoughts? |
| 11:16.21 | Red15 | musical_Duck: a little hack : WaitExten(0.5) |
| 11:16.23 | Weezey | pa: http://www.google.com/talk/developer.html -> In the near future, we plan to support SIP signaling. |
| 11:16.31 | musical_Duck | will try thx |
| 11:16.32 | jontow | musical_duck; i had that same problem.. i never found a solution.. only a workaround, and that is add a wait to the beginning.. |
| 11:16.34 | *** join/#asterisk rkvarala (n=rkvarala@59.93.96.90) |
| 11:16.45 | the_phantom | Hi all |
| 11:16.48 | Mother_ | I also have a problem - two * over IAX2, one sees the other just fine, the other reports the one as unavailable |
| 11:16.53 | musical_Duck | I have plain wait(2) thats not the same? |
| 11:16.53 | *** join/#asterisk hellop_pda (n=hellop@cpe-66-8-249-233.hawaii.res.rr.com) |
| 11:16.58 | rkvarala | what is good monitoring tool for asteirsk sip channels |
| 11:17.02 | jontow | musicalduck; no. |
| 11:17.02 | Mother_ | however, any connection works OK over UDP port 4569 so it's not there |
| 11:17.02 | Red15 | musical_Duck: should be though |
| 11:17.13 | Red15 | musical_Duck oh spoke too soon :D |
| 11:17.28 | Red15 | Mother_ any nat translation going on between the two ? |
| 11:17.28 | musical_Duck | thx! will try |
| 11:17.40 | hellop_pda | hi |
| 11:17.48 | Mother_ | Red15: yes, but that works fine, I can netcat from one to the other just fine |
| 11:17.54 | Freman | meh |
| 11:17.59 | Freman | my dial plan > http://forums.whirlpool.net.au/forum-replies.cfm?t=388586&p=-1#bottom |
| 11:18.04 | Freman | I think I finaly got enum working |
| 11:18.13 | Red15 | qwer^^ : am i able to use zapatel drivers to control the AVM Fritz card ? |
| 11:18.15 | Freman | misscounted n+101 |
| 11:18.44 | Red15 | Mother_ try playing around with the nat traversal settings within iax.conf ... are your entries peer/user/friends ? |
| 11:19.40 | Mother_ | Red15: they are friends - problem is it was working fine when I left on holidays, now I get back and they tell me a few days after I went it "stopped working" |
| 11:19.57 | Red15 | Mother_ oh lord ... users ... :D |
| 11:20.03 | Mother_ | so now that I check it seems that one is sending out the pokes and getting back the answer, and the other is not |
| 11:20.05 | Mother_ | LOL! |
| 11:20.07 | Mother_ | indeed |
| 11:20.18 | hellop_pda | iax softphone exists? |
| 11:20.29 | Red15 | yes there is one that i know of |
| 11:20.34 | Red15 | idefisk it is called |
| 11:20.38 | Red15 | but i never got it working |
| 11:20.43 | Red15 | so i stayed with xlite |
| 11:21.02 | Mother_ | firefly also, works OK |
| 11:21.07 | jontow | iaxcomm is decent |
| 11:21.17 | drray | moziax |
| 11:21.24 | hellop_pda | for pocket pc? |
| 11:21.29 | Red15 | didn't firefly require you to register and have your * server publicly available as well ? |
| 11:21.50 | Mother_ | Red15: I run firefly against my own * no problems |
| 11:22.33 | Red15 | Mother_ : and also didn't it require you to register an account with them ? so there is hardly a way to deploy it within a company |
| 11:22.42 | Mother_ | Red15: this is the so-called "third party" firefly, I think there is another version that works as you say |
| 11:22.48 | Mother_ | Red15: nope, no registrations, no nothing |
| 11:23.01 | Mother_ | just setup the account on my *, configure it and away you go |
| 11:23.09 | Red15 | can you transfer calls with firefly ? |
| 11:23.34 | Red15 | so you needed an IAX account to connect your * to their server ? |
| 11:23.37 | Mother_ | I believe you can, but I've not tried |
| 11:23.47 | *** join/#asterisk christo (n=chris@195.82.114.14) |
| 11:24.18 | Mother_ | Red15: Oh, if you wanted to do that then yes, but I don't want to connect to their server, I run SIP on the * extensions plus PSTN outbound so I have all I need |
| 11:24.41 | the_phantom | REGISTER apbcs881gold |
| 11:24.52 | Red15 | ah but i am not really looking to use my server as a voip gateway and use an account to dial over internet |
| 11:25.08 | Mother_ | Red15: https://www.virbiage.com/download.php |
| 11:25.12 | Red15 | i am mostly interested in getting asterisk to function as an ordinary PSTN central works |
| 11:25.20 | musical_Duck | wierd, try adding the waitexten infront of festival but now i loose second word, should it be somewhere else? |
| 11:25.24 | Red15 | with call direction etc |
| 11:25.28 | gordonjcp | Red15: easy enough |
| 11:25.56 | Mother_ | Red15: that's what I am doing, I have * to serve as PSTN PBX, with SIP clients, and I can use it over IAX2 while on the road with firefly |
| 11:25.56 | Red15 | musical_Duck: how about no delay at all ? is there also an answer on the top ? |
| 11:26.14 | Mother_ | that way I can call internal extensions or dial out PSTN using the local rates etc. |
| 11:26.24 | Bhaal | Okey, from the asterisk console I can initate a call right? How would I do that? |
| 11:27.05 | Red15 | Dial(....) |
| 11:27.12 | drray | Bhaal do you have the right sound drivers? |
| 11:28.04 | Bhaal | drray: I dont care about audio, I just wanna test whether the call is going to route properly... |
| 11:28.18 | Red15 | www.voip-info.org/tiki-index.php?page=Asterisk+cmd+Dial |
| 11:28.21 | musical_Duck | mmm seems that festival can't say quick? Using mrbola voice? |
| 11:28.30 | Red15 | musical_Duck: hehe |
| 11:28.32 | drray | kwik |
| 11:28.53 | musical_Duck | I am running a realtime mysql extension system, might this be causing delays? |
| 11:29.28 | musical_Duck | :P I opted for nimble :) |
| 11:29.55 | Red15 | Mother_ you said something about the firefly third party version, what other version does exist ? |
| 11:30.30 | zoa2 | i have very good news |
| 11:30.37 | musical_Duck | do tell |
| 11:30.37 | zoa2 | the sip jb might be fixed |
| 11:30.47 | rkvarala | is nagios is good with sip on asteirsk ? any feedback ? |
| 11:31.14 | Red15 | sip jb ? |
| 11:32.18 | qweee | is there anyway i can block the caller identification of the land lone |
| 11:32.21 | qweee | line |
| 11:32.25 | Bhaal | Red15: Ummm, this is from asterisk -r type console right? |
| 11:32.51 | drray | qwee - set it to something else? |
| 11:32.56 | Red15 | should work as well Bhaal |
| 11:33.08 | Bhaal | Its not, what other consoles are there? |
| 11:33.13 | qweee | but how can i do it |
| 11:33.14 | Mother_ | Red15: I think other versions are customised for VoIP operators etc. so they are 'closed' |
| 11:33.26 | qweee | its with the telecom carrier rt |
| 11:33.27 | zoa2 | sip jitter buffer |
| 11:33.43 | Red15 | Bhaal: can't test it atm but other 'console' would be from an extension dialing it |
| 11:34.02 | Bhaal | yeah, tis cool... I will try when I am at work... |
| 11:34.13 | Mother_ | grrrr this is weird...it seems the UDP checksums on the outbound packets from the asterisk server that doesn't see the other are 'bad' |
| 11:34.22 | Red15 | Mother_ I just installed the firefly and it crashes on launch |
| 11:34.25 | qweee | drray is there any equipment which will prevent them from seeing my cell no |
| 11:34.37 | Mother_ | pffff that sucks |
| 11:34.49 | Mother_ | 13:34:13.307991 192.168.0.2.4569 > x.x.x.x.4569: [bad udp cksum 9237!] udp 12 (DF) |
| 11:35.00 | drray | qwee - I don't follow, if you are calling in, you can set caller ID to whaterver you want to your inside extension |
| 11:35.20 | Red15 | Mother_ : you could turn off chksum setting |
| 11:35.22 | qweee | i am talkin about pstn lines |
| 11:35.26 | drray | me too |
| 11:35.48 | qweee | i am calling outside |
| 11:36.05 | Mother_ | Red15: will look into that |
| 11:36.21 | qweee | may be to ur phone and i dont want u to see my no |
| 11:36.24 | Red15 | Mother_ what kind of PSTN connection board do you use ? |
| 11:36.43 | musical_Duck | wierdness: Can't get music on hold to play, not even with madplay, get reschedule in past warnigs, any thoughts? |
| 11:36.44 | qweee | noramal landline |
| 11:37.11 | *** join/#asterisk Pj (n=pj@fernande.happycoders.org) |
| 11:37.22 | musical_Duck | Was working on non-CVS gentoo version previously. |
| 11:37.26 | Pj | Heya ppl ! |
| 11:37.28 | musical_Duck | lo |
| 11:37.38 | *** join/#asterisk spdy (n=spdy@altom.net1.nerim.net) |
| 11:37.50 | Red15 | lo |
| 11:38.00 | *** join/#asterisk Lan16`spdy^gTp (n=spdy@altom.net1.nerim.net) |
| 11:38.05 | jontow | musical_Duck: timing? ... do you have 'ztdummy' or another zaptel module loaded ? |
| 11:38.15 | musical_Duck | loaded ztdummy |
| 11:38.30 | Mother_ | Red15: four-port TDM400 |
| 11:38.41 | Mother_ | btw where do you turn off checksums? |
| 11:38.43 | Red15 | Mother_: is this a card from digium ? |
| 11:38.51 | Mother_ | yes, digium one |
| 11:39.09 | Red15 | how much did it cost ? I'm still early in tryouts don't want to buy expensive stuff |
| 11:40.34 | drray | Red, there is a dev kit from digium |
| 11:40.40 | musical_Duck | just silence from moh, not even jitters of clicks |
| 11:40.41 | drray | that is 1fxs and 1fxo port |
| 11:41.04 | *** part/#asterisk spdy (n=spdy@altom.net1.nerim.net) |
| 11:41.15 | Red15 | yes i know but buying a dev kit is hardly usable for later production use at our company |
| 11:41.30 | drray | how many fxs and fxo ports would you need? |
| 11:41.32 | Red15 | so you then would have to buy yet another card with more fxo's en fxs' |
| 11:41.39 | Mother_ | Red15: depends on the number of ports you need |
| 11:41.48 | drray | we bought a dev kit to try, and then removed it for a channel bank |
| 11:41.55 | Mother_ | you can swap out the FXO and FXS boards as needed later on |
| 11:41.56 | Red15 | i don't really now what kind of line in from pstn can an FXO handle ? |
| 11:41.57 | drray | it was worth it, rather than buying a channel bank |
| 11:42.03 | musical_Duck | BTW anyone here from South Africa? |
| 11:42.18 | Mother_ | Red15: the TDM will handle analog pstn only |
| 11:42.34 | Red15 | We have 1 ISDN line with 2 MSNs and 2 actual lines and also have 2 Analog lines |
| 11:43.16 | hellop_pda | qwee google |
| 11:43.25 | Mother_ | well you'll need an adapter for the ISDN and then a TDM with 2 FXO ports - do you want your internal phones as analog or will you get them voip? |
| 11:43.29 | *** join/#asterisk yellowsnow (n=yellowsn@dsl-213-134-245-123.solcon.nl) |
| 11:43.34 | Mother_ | I have all the internal network in SIP |
| 11:43.46 | Red15 | my internal phones are all softphones |
| 11:43.46 | drray | we run internal FXS ports and sip |
| 11:43.56 | Mother_ | so the TDM was a cheaper and sufficient solution for me |
| 11:44.21 | Mother_ | Red15: then a channel bank will be overkill for just 2 analog lines |
| 11:44.24 | Red15 | we also already have an option on our phone line which can split our isdn into 2 analog lines |
| 11:44.25 | drray | teh TDM has modules so you can change your mind |
| 11:44.35 | Red15 | so we would then have 4 analog lines |
| 11:44.50 | Mother_ | then get the TDM400P with 4 FXO boards, that will handle your four analog lines |
| 11:45.02 | Mother_ | it's what I have, cost was under $400 |
| 11:45.32 | drray | ebay has them for cheaper, but I would thow digium the $$$, plus they will help you get them working undr linux if you hit a snag |
| 11:45.43 | Druken | Mother_: do you have to use park at home? :O |
| 11:45.46 | dexteruk | Should the Busydetect work at any time ie a call is place to the PBX and goes to voice mail if they hangup the handset should it get disconnected? |
| 11:47.34 | Mother_ | Druken: ? park? as in call parking? |
| 11:47.36 | mover | is there a way to send dedicated messages for hangupcauses to a sip UA? |
| 11:49.37 | Red15 | Mother_ : can i plug an ISDN cable straight into one port of the TDM400P ? |
| 11:49.58 | drray | I don't think you can, you need them split or you need BRI card |
| 11:50.01 | Mother_ | Red15: nope, you can plug the two analog lines that come of the ISDN TA |
| 11:50.09 | drray | which I know zero about |
| 11:50.16 | *** join/#asterisk dwmw2_gone (i=ctrlprox@baythorne.infradead.org) |
| 11:51.13 | Red15 | Mother_ : is there another card that can handle (ie) 4 to 8 isdn lines and convert them itself into 8-16 analog lines ? |
| 11:51.49 | drray | Red, can you get your ISDN lines converted into a PRI? |
| 11:52.07 | *** join/#asterisk puzzled (n=patrick@puzzled.xs4all.nl) |
| 11:52.33 | puzzled | morning all |
| 11:52.45 | Red15 | drray: i'm not really sure. We have a client where there are 8 ISDN boxes which all goto the current (propriatary) PBX |
| 11:53.11 | Red15 | drray: IIRC PRI is like T1 or E1 correct ? |
| 11:53.17 | drray | yes |
| 11:53.40 | drray | 8 ISDN lines would be 16 channels, a e1 is 30 channels |
| 11:53.57 | drray | I assume you are in E1 land? |
| 11:53.58 | mover | i need to do something like that: dial(exten,60Ã after timeout i need to sen a 408 request timeout. Ÿ) |
| 11:54.10 | Red15 | drray: to be honest i really don't know (i live in belgium) |
| 11:54.17 | drray | europe is E1 |
| 11:54.20 | drray | for the most part |
| 11:55.04 | Red15 | so i assume that if i get PRI instead of 8 ISDN channels I would need some other card which probably costs a small fortune :P |
| 11:55.07 | Mother_ | yes, that should be E1 |
| 11:55.10 | drray | $500 |
| 11:55.12 | drray | ish |
| 11:55.24 | drray | and you would not need a channel bank |
| 11:55.32 | *** join/#asterisk zeedo (n=zeedo@www.reboot-robot.net) |
| 11:55.38 | Mother_ | indeed |
| 11:55.40 | drray | and that card would handle 30 lines |
| 11:55.44 | drray | I beleive |
| 11:55.52 | Red15 | then when would you need a channel bank ? |
| 11:55.58 | drray | you would not |
| 11:56.03 | drray | channel banks are for analog |
| 11:56.05 | yellowsnow | hi all |
| 11:56.13 | Mother_ | BLAAAAAAAARGH |
| 11:56.28 | Mother_ | this is REALLY getting on my nerves now |
| 11:56.43 | Red15 | so i'm still kinda puzzled |
| 11:56.50 | Red15 | you probably all know A@H right |
| 11:56.55 | Mother_ | *A sends UDP with bad checksum to *B, *B does not see these |
| 11:57.12 | drray | red a@home is a fork project |
| 11:57.23 | drray | we all mostly use just plain asterisk |
| 11:57.23 | Mother_ | however, when *B sends a POKE to *A, *A sends back the reply with bad checksum too, but *B DOES see this reply |
| 11:57.29 | yellowsnow | drray: don't forget an ISDN30 account is can be much more expensive as multiple ISDN2 lines |
| 11:57.36 | Red15 | so any A@H user would require a FXO capable card ? which home user is buying a $400 card for home phone usage |
| 11:57.57 | drray | yellowsnow - I did not know that, being an ugly american :) |
| 11:58.17 | *** join/#asterisk likwid-- (n=likwid@nc-69-69-189-124.dyn.sprint-hsd.net) |
| 11:58.26 | hellop_pda | http://www.silicontechnix.com/babarnazmi/iaxclient/iaxclientocx.htm iax on pda |
| 11:58.41 | Ogun | Belgium is E1, definately. Atleast last time I was there :) |
| 11:58.43 | drray | Red, you could use a sip based FXO port, or you could use a voip sip/iaxy provider, or you could FXO port it |
| 11:58.45 | yellowsnow | ddray: and living in Belgium? |
| 11:58.56 | Red15 | yellowsnow: that would be me |
| 11:58.58 | drray | no, trying to help REd |
| 11:59.10 | drray | (doing it badly) :) |
| 11:59.15 | Red15 | Ogun: thx for confirming |
| 11:59.28 | dtwilson | Red15: are you simply trying to find a way to connect your analogue phone to a@h? |
| 11:59.31 | yellowsnow | Red15: Sorry, wasn't paying attention :-) |
| 11:59.55 | drray | dtwilson , he has two analog lines and an ISDN line |
| 11:59.58 | Red15 | dtwilson: yea and using AMP for management especially (A@H is short cut, i also have CentOS self-configured) |
| 12:00.19 | *** join/#asterisk pa (n=Paolo@unaffiliated/pa) |
| 12:00.36 | Red15 | so my CentOS running * can dial out using AVM Fritz Card |
| 12:00.53 | Red15 | all self-configured (blood,sweat and tears) |
| 12:01.05 | dtwilson | Red15: ok use bri cards for isdn, x101p clones for the analogue lines and then use sip for internal extensions |
| 12:01.13 | Red15 | but the thing is we want to be able to manage it without much hassle (AMP springs to mind) |
| 12:01.26 | dtwilson | and if you really must use analogue phones just buy an ATA |
| 12:01.30 | Red15 | dtwilson: yep so far that all works |
| 12:01.32 | Red15 | ATA ? |
| 12:01.38 | dtwilson | analogue terminal adapter |
| 12:01.45 | Red15 | like a regular voice modem ? |
| 12:01.55 | dtwilson | nop - google cisco ATA |
| 12:02.05 | dtwilson | they allow you to plug analogue phone into the network for SIP |
| 12:02.13 | Red15 | ah thx |
| 12:02.20 | drray | look for FXO ports though |
| 12:02.24 | Ogun | I am using an old ISDN-to-analog adapter and a HFC card |
| 12:03.09 | dtwilson | Red15: my guess is tho that you would pick up[ a SIP handset cheaper than most ATA's |
| 12:03.40 | dtwilson | iirc telappliant in UK (presuming you're uk) have Grandstreams at around £45 |
| 12:03.45 | Ogun | dtwilson: Agreed. Beronet in germay sells sipura 841:s for about 80 EUR. |
| 12:04.13 | Red15 | thx for all the info i now pretty much know all the hardware i will/can use |
| 12:04.21 | puzzled | dtwilson: but the grandstreams are sucky phones and the sipura is a good product |
| 12:04.29 | Red15 | but i'm still wondering how i can make this * server configurable |
| 12:04.48 | dtwilson | puzzled: true - just giving cheapest options from top of my head |
| 12:05.16 | dtwilson | Red15 try DeStar http://www.holgerschurig.de/index.html |
| 12:05.29 | Red15 | in other words is AMP in any way capable of configuring/using anything other then IAX/VoIP provider/FXO to dial out ? |
| 12:05.43 | puzzled | dtwilson: yup, cheap they are. I had the "pleasure" of messing with one of their pre-production models and boy were they crap |
| 12:05.44 | *** join/#asterisk huehner_ (n=huehner@kt-gmbh.de) |
| 12:06.50 | dtwilson | puzzled: I have the *pleasure* of having 10 of them here :/ |
| 12:06.50 | puzzled | Red15: amp does sip too (iirc) |
| 12:06.55 | puzzled | ouch :) |
| 12:07.07 | Red15 | puzzled: ok so i missed SIP :P |
| 12:07.12 | dtwilson | and a zyxel wifi phone |
| 12:07.20 | Red15 | i just want to configure AMP to dial out using my /dev/ttyI0 |
| 12:08.29 | *** join/#asterisk Tekmaven (n=Tekmaven@ool-182e31c8.dyn.optonline.net) |
| 12:08.30 | Red15 | is there any way to make an isdn adaptor behave like it were some FXO single channel |
| 12:08.31 | syle | can i t and h extensions be used differently in different contexts? |
| 12:08.42 | Red15 | syle: yes |
| 12:09.00 | syle | k cause i thought extensions had to be unique |
| 12:09.09 | puzzled | Red15: afaik amp does not support isdn4linux stuff |
| 12:09.48 | puzzled | syle: think you can stick em at the end of any context you like |
| 12:10.01 | Red15 | syle: i believe you simply overwrite the previously active context-dependant extensions |
| 12:10.12 | *** join/#asterisk zotz (n=zotz@24.231.36.100) |
| 12:10.21 | Red15 | puzzled: how sure are you about AMP and isdn4linux |
| 12:10.35 | syle | i can only hope they aren;t inherited in other contexts that i haven;t used them in |
| 12:11.05 | puzzled | Red15: haven't seen it mentioned anywhere on the amp website just like capi support. but you can always ask on the amp mailinglist |
| 12:11.30 | Red15 | puzzled: will try thx for the help |
| 12:12.01 | *** join/#asterisk nemisus (n=nemisus@203-217-87-135.dyn.iinet.net.au) |
| 12:12.31 | puzzled | Red15: you are possibly better off with bristuff and a hfc-s based isdn card. iirc there were many issues with isdn4linux and afaik it is not really looked at anymore |
| 12:13.01 | Red15 | I tried the CAPI variant but could not even get it working kernel level ... :( |
| 12:13.14 | Red15 | since i do have a capi capable isdn card |
| 12:13.27 | puzzled | yeah it's not that easy. which card do you have? |
| 12:13.37 | Red15 | AVM Fritz |
| 12:13.48 | Red15 | it's a passive card but is CAPI capable |
| 12:14.03 | puzzled | there are some docs on voip-info.org describing the install |
| 12:14.10 | Red15 | yes i tried those |
| 12:14.29 | puzzled | which distro do you use? |
| 12:14.29 | Red15 | but capi gives me errors about not being able to open /dev/capi20.00 etc |
| 12:14.32 | Red15 | CentOS |
| 12:14.34 | Red15 | 4.1 |
| 12:14.44 | Red15 | and i did read those udev notes |
| 12:14.55 | Red15 | specifically those regarding to FC3 and later |
| 12:15.00 | Mimmus | how can I check exit code of 'system' command in a dialplan? |
| 12:15.31 | Red15 | Return codes |
| 12:15.32 | Red15 | Returns -1 on failure to execute the specified command. If the command itself executes but is in error, and if there exists a priority n + 101, where 'n' is the priority of the current instance, then the channel will be setup to continue at that priority level. Otherwise, System returns 0. |
| 12:15.57 | puzzled | Red15: ah, that capi20 issue is easily fixed. You need to add a file with capi info to /etc/udev/rules.d |
| 12:16.02 | Mimmus | Red15: thanks. but how is the syntax of GotoIf to check for '-1'? |
| 12:16.18 | Red15 | Mimmus: store the return code |
| 12:16.18 | ai-a | can i force asterisk to reload from a user level? or how quickly will asterisk reload after i save new extentions? |
| 12:16.24 | Red15 | puzzled: do you have an url on that ? |
| 12:16.30 | puzzled | Red15: check the chan_capi srpm on http://www.laimbock.com/asterisk/ iirc it has the file |
| 12:16.45 | Mimmus | Red15: sorry for ignorance but what is the syntax? |
| 12:17.26 | Mother_ | well, a complete power down cured it, as I expected, but I'm puzzled as to why all of a sudden asterisk starts sending out bad UDP checksums |
| 12:17.41 | Mother_ | even restarting asterisk, setting nochecksum=yes, did nothing |
| 12:17.45 | Red15 | Mimmus: not really sure check this url : http://www.voip-info.org/tiki-index.php?page=Asterisk+cmd+System |
| 12:17.58 | puzzled | Mother_: bad cable? |
| 12:18.20 | Mother_ | puzzled: would a power down and restart fix a bad cable? |
| 12:18.31 | puzzled | nope |
| 12:18.33 | Mother_ | if so I can start making gobs of money :D |
| 12:18.34 | Red15 | maybe if you give it a kiss :P |
| 12:18.35 | Mother_ | hehehe |
| 12:18.47 | Red15 | sweet talk to it |
| 12:18.51 | Mother_ | yeah |
| 12:18.58 | Mimmus | Red15: I saw it, but I don't know Asterisk syntax to store exit code of a command in a var |
| 12:19.15 | Mother_ | well if this was windows I'd say a DLL didn't unload when stopping asterisk, but since it's not, I don't know what could have gone wrong |
| 12:19.22 | Mother_ | time to put a reboot in cron :) |
| 12:20.50 | drray | so from console, I can dial and hangup, can I play a sound file too? |
| 12:21.08 | Mother_ | drray: playback command |
| 12:21.37 | Red15 | Mimmus: your extension skips to line n+101 |
| 12:21.45 | Red15 | where n is return code |
| 12:21.50 | drray | playback is not an available command in console |
| 12:22.20 | Mimmus | Red15: interesting. And if return code is '-1'? extension n+100? |
| 12:22.23 | *** join/#asterisk MajestiK (n=MajestiK@S0106000ea6572b5f.ed.shawcable.net) |
| 12:22.29 | Red15 | n+101 = 100 |
| 12:22.39 | *** join/#asterisk TheCops (n=mdb@206-248-136-187.dsl.teksavvy.com) |
| 12:22.40 | Red15 | -1 + 101 = 100 that is |
| 12:22.41 | TheCops | Hi |
| 12:22.43 | RoyK | n+101 != 100 |
| 12:22.49 | drray | oh shit, it's thecops |
| 12:22.55 | Red15 | lol |
| 12:23.04 | drray | pardon me |
| 12:23.16 | TheCops | Someone have a website with all prefix location for Canada? |
| 12:23.36 | Red15 | http://wherecopsgo.ca :P |
| 12:24.20 | Mimmus | Red15: thanks. I'm trying immediately |
| 12:24.30 | Red15 | Mimmus: let me know |
| 12:24.56 | Red15 | Mimmus: i never had to do something like this before but it is interesting to know |
| 12:26.02 | *** join/#asterisk lathos42 (n=lathos42@65-42-27-66.dowdingindustries.com) |
| 12:26.35 | lathos42 | Hello everyone |
| 12:26.35 | Mimmus | Red15: I'd like to create a more reliable way to send a fax, with result notification to sender |
| 12:26.56 | RoyK | Mimmus: more reliable than what? |
| 12:27.09 | RoyK | :%s/fax/e-mail/gi |
| 12:27.10 | drray | ok, duh, I'm stupid |
| 12:27.13 | Red15 | Mimmus: in this peticular case it would be handy if extensions kept running whitout any sort of SIP connectio or other and have callback on :P |
| 12:27.48 | Mimmus | RoyK: than simple putting a .call fil eunder outgoing and hoping in TxFax... |
| 12:28.41 | Mimmus | RoyK: I'd like notification to sender of result |
| 12:28.43 | *** join/#asterisk sneak (n=sneak@64.220.234.21.ptr.us.xo.net) |
| 12:29.02 | Red15 | oh btw puzzled if you are still here the udev rules for chan_capi are not on the supplied url |
| 12:29.26 | *** join/#asterisk Praktikant01 (n=lars@dsl-084-059-141-181.arcor-ip.net) |
| 12:30.03 | *** join/#asterisk mithro (n=tim@c213-100-42-188.swipnet.se) |
| 12:30.11 | Praktikant01 | i need help with iax, i rgsteredmy asterisk on a other asterisk, but how can i call a extension through this iax-connection ?? |
| 12:30.45 | drray | you need a dialplan, in extension.conf |
| 12:31.17 | puzzled | Red15: let me fire up the centos box and see what they are |
| 12:31.41 | Red15 | puzzled: much obliged |
| 12:32.09 | *** join/#asterisk websae (n=websae@207-118-143-134.dyn.centurytel.net) |
| 12:33.46 | Praktikant01 | drray: i tried the following: Dial(iax2/user:password@target/extension@context) |
| 12:33.49 | RoyK | Mimmus: agi, perhaps? |
| 12:34.39 | websae | can someone tell me what I should do/need to do about this...using freebsd and was trying to compile asterisk |
| 12:34.40 | websae | Verifying install for /usr/local/include/zaptel.h in /usr/ports/misc/zaptel |
| 12:34.40 | websae | >> The zaptel port needs FreeBSD kernel source code to compile. |
| 12:34.40 | websae | >> Please install FreeBSD kernel source code in /usr/src/sys. |
| 12:34.40 | websae | *** Error code 1 |
| 12:34.41 | websae | Stop in /usr/ports/misc/zaptel. |
| 12:34.43 | websae | *** Error code 1 |
| 12:34.45 | websae | Stop in /usr/ports/net/asterisk. |
| 12:34.53 | Mimmus | RoyK: peraphs it should be better... I have no experience in AGI. |
| 12:35.01 | *** join/#asterisk rkvarala (n=rkvarala@59.93.52.96) |
| 12:35.02 | *** join/#asterisk DagMoller (n=DagMolle@tor/session/x-4b4208e7e3f605cf) |
| 12:35.30 | rkvarala | what is the minimum timeout for the registrations in the asterisk? |
| 12:35.47 | rkvarala | can that is configurable ? |
| 12:36.20 | drray | man, every couple of weeks I learn something new about asterisk and it amazes me |
| 12:36.28 | websae | anyone know about my probleM? |
| 12:36.47 | *** part/#asterisk DagMoller (n=DagMolle@tor/session/x-4b4208e7e3f605cf) |
| 12:37.00 | Mother_ | websae: you obviously need to install the kernel source tree |
| 12:37.11 | websae | how do i do that? |
| 12:37.17 | websae | do you use freebsd? |
| 12:37.21 | puzzled | Red15: get http://www.xs4all.nl/~pjl/tmp/10-capi.rules and stick the file in /etc/udev/rules.d |
| 12:37.25 | Mother_ | Google Is Your Friend (tm) |
| 12:37.33 | Red15 | thx puzzled |
| 12:37.39 | puzzled | np |
| 12:37.47 | Mimmus | Red15: it doesn't work. TxFax jumps to hungup when exits |
| 12:38.01 | RoyK | Mimmus: just take a look at the examples |
| 12:38.04 | RoyK | Mimmus: quite simple |
| 12:39.05 | Red15 | Mimmus: ... maybe an undefined error returned ? |
| 12:39.26 | Red15 | or the wrong extension number used |
| 12:39.48 | Red15 | sorry extensions line |
| 12:40.01 | Mimmus | Red15: no no, txwax was successful |
| 12:40.37 | Red15 | Mimmus: can you paste output from console ? |
| 12:40.39 | Mimmus | Red15: I was waiting for a mail confirmation to extension n+100 |
| 12:40.39 | Red15 | oh wait |
| 12:40.56 | Red15 | use www.pastebin.com |
| 12:41.12 | *** join/#asterisk docelm0 (n=docelm0@67.106.194.90.ptr.us.xo.net) |
| 12:41.15 | Red15 | and paste your specific part of extensions.conf |
| 12:42.11 | Mimmus | Red15: this is custom context: http://pastebin.com/345847 |
| 12:42.22 | *** join/#asterisk CaNaBiS (n=CaN@lvrgtnbas02-pool9-a7.lvrgtn.tds.net) |
| 12:42.53 | Red15 | ah yes |
| 12:42.57 | Red15 | you do n+101 |
| 12:43.15 | Red15 | extension line 100 is in case of error |
| 12:43.32 | Red15 | -1 + 101 remember |
| 12:43.41 | Red15 | return code 0 does 0+101 |
| 12:44.14 | Mimmus | Red15: I inserted 100,101,102 only for test but I receive nothing, Asterisk hangups |
| 12:44.52 | *** join/#asterisk ]bodhi[ (n=not@c-3b91e255.1310-2-64736c14.cust.bredbandsbolaget.se) |
| 12:44.53 | Red15 | try removing the line where it says hangup() |
| 12:45.02 | Mimmus | Red15: ok now... |
| 12:45.04 | Red15 | or play some hold music or something and see hat happens |
| 12:45.17 | ]bodhi[ | sup all+ |
| 12:45.48 | *** join/#asterisk bjohnson (n=bjohnson@jecinc.tor.istop.com) |
| 12:46.07 | ]bodhi[ | what is needed (connectionwise) in order to run asterisk? |
| 12:46.22 | Red15 | puzzled: my capiinit start still returns cannot load module kernelcapi |
| 12:46.55 | Red15 | ]bodhi[ anything from analog line to isdn to E1/T1 or only internet connection |
| 12:47.15 | Red15 | in fact you could run asterisk without internet as well and only on lan and use it for intercom purposes |
| 12:47.25 | ]bodhi[ | red15, oki, so I could set it up on a normal adsl line |
| 12:47.44 | Red15 | ]bodhi[ you mean a normal analog (POTS) line ? |
| 12:47.50 | ]bodhi[ | yes, sorry |
| 12:47.53 | Red15 | np |
| 12:48.09 | ]bodhi[ | sweet.. |
| 12:49.19 | *** join/#asterisk FunFusion (i=a@62.162.228.109) |
| 12:49.26 | FunFusion | www.FunFusion.Net New Place 4 Fun (daily updates) |
| 12:49.33 | *** part/#asterisk FunFusion (i=a@62.162.228.109) |
| 12:49.58 | Red15 | im not sure about what hardware you need but if im not mistaken you'd need something like an intel pci modem |
| 12:50.32 | Red15 | more specifically : an x101p clone card |
| 12:52.03 | Mimmus | Red15: no, Asterisk hangups and stop |
| 12:52.29 | Red15 | what did you put behind the system line now ? |
| 12:52.35 | Red15 | nothing or just music ? |
| 12:53.26 | Mimmus | Red15: this is context: http://pastebin.com/345857 |
| 12:53.58 | *** join/#asterisk MikeJ[Laptop] (n=ircatjer@mi.origenfinancial.com) |
| 12:54.01 | Mimmus | Red15: .call file points to it, extension out_fax, priority 1 |
| 12:54.06 | Red15 | where is your system() call ? |
| 12:54.45 | Mimmus | Red15: no, sorry, in this case, I'm interested to TxFax call |
| 12:54.52 | Red15 | ah hold on |
| 12:54.54 | Mimmus | it has same exit codes |
| 12:55.27 | Red15 | hmm i do not have these commands nor do i find any documention on it |
| 12:55.32 | Mimmus | from source code of app_txfax.c: Returns -1 when the user hangs up, or if the file does not exist |
| 12:55.38 | *** join/#asterisk edgecase (n=jjackson@CPE00a0c9841796-CM000f9fa6b7d6.cpe.net.cable.rogers.com) |
| 12:55.38 | Mimmus | Returns 0 otherwise |
| 12:56.34 | *** join/#asterisk oden (n=oden@194-237-146-22.customer.telia.com) |
| 12:56.35 | edgecase | so it seems Steve "Mr. T.38" Underwood has dropped off the net. damnit! |
| 12:57.28 | syle | hmmm |
| 12:57.37 | syle | monitor doesn;t seem to record ringing |
| 12:58.12 | syle | first half of file is always just silence |
| 12:59.18 | Red15 | Mimmus: sorry i can't really help you any further then i have |
| 12:59.41 | Mimmus | Red15: ok, don't worry, thank you very much anyway |
| 12:59.49 | *** join/#asterisk pudo (i=central@202.74.179.207) |
| 13:00.16 | pudo | hi all. here's a newbie question: how do I forward an extension to s specific phone number via sip? |
| 13:00.27 | Katty | morning. |
| 13:00.56 | Red15 | pudo: exten => s,1,Dial(SIP/<extension here>) |
| 13:01.13 | Red15 | morning Katty |
| 13:01.19 | pudo | where extension is any phone number? |
| 13:01.35 | Red15 | pudo : no |
| 13:01.41 | Red15 | pudo: how do you dial out ? |
| 13:01.41 | pudo | i thought so. |
| 13:01.58 | syle | NoOp in dialplan doesn;t seem to log anywhere in /var/log/asterisk----is there a way to do this |
| 13:02.07 | syle | stead of just reading from CLI |
| 13:02.10 | pudo | Red15: via SIP..? |
| 13:02.16 | Red15 | syle: increase verbosity at startup |
| 13:02.41 | Red15 | pudo: do you have a voip provider you pay ? |
| 13:03.08 | *** join/#asterisk hadi57 (n=al_moghr@83.136.8.206) |
| 13:03.10 | pudo | Red15: yeah |
| 13:03.21 | qwer^^ | red15: may i know how do i increase te verbosity |
| 13:03.24 | Red15 | does your voip provider give you any settings you should use ? |
| 13:03.36 | syle | red15: umm i think you didn;t read what i wrote |
| 13:03.48 | Red15 | qwer^^ : at startup : # asterisk -vvvv |
| 13:03.56 | pudo | Red15: I think that is not the problem: i got the sip thing set up all right |
| 13:04.17 | Hmm-home | so what went wrong qwer^^? |
| 13:04.24 | syle | how can we take the verbosity you can read from the CLI and redirect it into a file |
| 13:04.25 | Red15 | or at asterisk running in background already : asterisk -vvv -r |
| 13:04.37 | jontow | syle; its already there.. |
| 13:04.37 | *** join/#asterisk yogurt2ungue (n=yogurt2u@44-170-114-200.fibertel.com.ar) |
| 13:04.38 | qwer^^ | red15: thanx |
| 13:04.45 | Red15 | syle: asterisk should do this ok |
| 13:04.49 | jontow | qwer^^: you can set the verbosity in a running instance with 'set verbose N' |
| 13:04.56 | Hmm-home | did net2phone tell you what was up? |
| 13:04.58 | syle | NoOp is not showing up in any logs |
| 13:05.00 | jontow | where N is a digit (equal to the number of v's on the command line, iirc) |
| 13:05.03 | syle | go try it |
| 13:05.16 | Mother_ | bbl |
| 13:05.25 | Red15 | pudo: one sec |
| 13:05.47 | jontow | bash-2.05b# grep "NoOp" /var/log/asterisk/messages | wc -l |
| 13:05.47 | jontow | <PROTECTED> |
| 13:05.48 | pudo | Red15: thanks |
| 13:06.18 | syle | # grep "NoOp" /var/log/asterisk/messages | wc -l |
| 13:06.18 | syle | <PROTECTED> |
| 13:06.26 | syle | ok thats fucked |
| 13:06.39 | Red15 | pudo please read thourougly : http://www.voip-info.org/tiki-index.php?page=Asterisk+config+sip.conf |
| 13:06.46 | jontow | hehe |
| 13:06.58 | jontow | i run verbose = 12 |
| 13:06.59 | jontow | ;) |
| 13:07.07 | Red15 | yipes :P |
| 13:07.12 | jontow | mine is a development pbx though, so its worthy of doing that |
| 13:07.16 | Red15 | ah |
| 13:07.27 | jontow | on a production run.. i don't think it'd be so wise |
| 13:07.29 | syle | i don;t think putting more than 3 v's makes any difference from what i have read |
| 13:07.32 | jontow | the logs fill pretty quickly like that |
| 13:08.14 | syle | ok jontow: what does your messages line look like in logger.conf? |
| 13:08.40 | jontow | messages => notice,warning,error,debug,verbose |
| 13:08.45 | jontow | ;) |
| 13:08.47 | syle | yeah thats what i thought |
| 13:08.55 | *** join/#asterisk eisman (n=eisman@pool-71-241-82-204.scr.east.verizon.net) |
| 13:08.55 | syle | just added verbose |
| 13:09.03 | jontow | indeed |
| 13:09.10 | eisman | morning |
| 13:09.33 | MmmmToop_ | afternoon ; ) |
| 13:09.38 | syle | there we go |
| 13:09.41 | syle | i;m happy now lol |
| 13:10.07 | syle | well my problem is i run half my crap in screen, so reattaching to session i loose alot of info |
| 13:10.21 | pudo | Red15: ok, that is 15 RTFM points for me. found it |
| 13:10.22 | jontow | btw, from a quick grep through the source.. there are at least 4 different verbose logging prefixes |
| 13:10.29 | Red15 | pudo :D np |
| 13:10.32 | jontow | so .. at least 4 v's will be useful |
| 13:10.35 | jontow | after that im not so sure |
| 13:10.39 | *** join/#asterisk Ariel_ (n=Ariel@adsl-068-157-125-248.sip.mia.bellsouth.net) |
| 13:10.44 | Red15 | pudo sometimes you tend to go faster over docs cause you can't find it |
| 13:10.49 | jontow | syle; no you don't.. ^A: then type scrollback 10000 |
| 13:10.58 | Red15 | pudo then someone has to tell you you specifically have to look there |
| 13:11.20 | jontow | use whatever you want for the scrollback value in screen.. to get to the scrollback buffer, ^A[escape] and pageup/pagedown |
| 13:11.30 | jontow | to get out of copy mode (which is the scrollback buffer) hit escape again |
| 13:11.46 | Mimmus | another question: any idea why Asterisk doesn't disable echo cancellation during fax transmission? |
| 13:12.18 | *** join/#asterisk fjean (n=fjean@201.29.141.127) |
| 13:13.37 | syle | what is ^A |
| 13:14.25 | *** join/#asterisk grimse_ (n=grimse@p5481DAD8.dip.t-dialin.net) |
| 13:14.30 | fjean | hi everybody, I have an E1 digium card, how do i build the cable that will fit in it, is it the same as a regular network cable ? |
| 13:14.37 | syle | ctrl-A escape don;t make a diff |
| 13:14.39 | eisman | Is there a debug dump for the zaptel drives seperate from asterisk? I want to see the call setup progress. Hook flash, wink, ANI/DNS etc... |
| 13:14.43 | fjean | straight trough ? |
| 13:15.04 | syle | wonder if their is a buffer i have to enable |
| 13:15.28 | eisman | fjean, are you conencting to a CSU? |
| 13:15.28 | Mimmus | fjean: to connect to telco, you can use regular network cable |
| 13:16.05 | Mimmus | fjean: to connect to another asterisk (or legacy PBX), you need a cross E1 cable (different from cross ethernet cable) |
| 13:16.37 | fjean | i am connecting to a HDSL modem, i need to figure out this part too. |
| 13:16.58 | fjean | there is two BNC connector on it |
| 13:17.15 | jontow | ^A is the escape character in screen |
| 13:17.15 | fjean | someone told me to go with a ballun |
| 13:17.18 | jontow | ctrl+a |
| 13:17.26 | jontow | the buffer is enabled by default in screen |
| 13:17.29 | jontow | i don't know what the default value is |
| 13:17.32 | Mimmus | fjean: ah, then you need a balun (impedence converter: from 120 to 75 ohm) |
| 13:17.40 | fjean | ok |
| 13:17.45 | jontow | but however you switch screens or detach, the first sequence you hit to do that is your 'escape character' |
| 13:18.05 | fjean | minnus, what about the cable between the two ? |
| 13:18.40 | Mimmus | fjean: simple ethernet cable |
| 13:18.42 | syle | ok i see what you mean |
| 13:18.52 | syle | pageup and pagedown don;t work for me in securecrt |
| 13:18.59 | syle | i have to use arrow keys |
| 13:19.01 | jontow | then fix them |
| 13:19.04 | jontow | ok |
| 13:19.07 | jontow | vi keys work too |
| 13:19.14 | jontow | so ^U and ^D will work for Up and Down |
| 13:19.15 | jontow | ;) |
| 13:19.24 | fjean | mimmus, like if i would connect a network card to a regular hub ? |
| 13:19.41 | musical_Duck | Iax2 to sip dialing takes quite a while, how long does it take for you guys? |
| 13:19.50 | syle | ahh that works |
| 13:19.54 | syle | thx dude |
| 13:19.56 | jontow | np |
| 13:19.58 | Mimmus | fjean: yes, I have a Te410P Digium card and this setup |
| 13:20.02 | jontow | little tricks, man, little tricks ;) |
| 13:20.09 | fjean | mimmus, oh, cool |
| 13:20.28 | Mimmus | fjean: only that I'm in Europe and I have E1 line |
| 13:20.33 | jontow | keep in mind the old adage 'theres more than one way to do it' and then think.. 'in unix, there are more than 10 ways to do it' |
| 13:20.44 | jontow | that'll get ya whatever you need in time |
| 13:20.46 | fjean | mimmus, i am in brazil, i have an E1 too |
| 13:21.16 | Mimmus | fjean: here in Italy, telco device has an hw switch to set 120 or 75 ohm |
| 13:21.27 | fjean | mimmus, ok |
| 13:22.09 | fjean | mimmus, mine has 130 ohm as output |
| 13:22.23 | fjean | mimmus, but let me check |
| 13:22.27 | Mimmus | fjean: the world is beautiful because is various! |
| 13:24.33 | fjean | mimmus, ok there is an adaptor on it, apparently it's 75 ohm... |
| 13:25.17 | fjean | mimmus, is this was you have : http://www.datacabos.com.br/balun/index.html |
| 13:25.42 | Mimmus | fjean: it's this |
| 13:25.59 | Ariel_ | Morning all |
| 13:26.39 | *** join/#asterisk greendisease (n=jack@fedora/greendisease) |
| 13:26.59 | eisman | Is there a debug dump for the zaptel drives seperate from asterisk? I want to see the call setup progress. Hook flash, wink, ANI/DNS etc... |
| 13:27.12 | eisman | err DNIS not DNS |
| 13:28.02 | *** join/#asterisk spackle (n=spackle@209.234.83.19) |
| 13:28.14 | fjean | mimmus, you had to put two cables between the adaptor and the device or you just snapped it ? |
| 13:28.24 | Mimmus | fjean: no cables |
| 13:29.05 | fjean | mimmus, ok |
| 13:29.16 | *** join/#asterisk ai-a (n=gandalfi@81.168.0.204) |
| 13:30.24 | syle | jontow |
| 13:30.25 | *** part/#asterisk oden (n=oden@194-237-146-22.customer.telia.com) |
| 13:30.30 | syle | seem to have locked my screen lol |
| 13:30.31 | *** join/#asterisk MRH2 (n=Mr_happy@fcirc-adsl.demon.co.uk) |
| 13:30.31 | Mimmus | fjean: you need to set exact framing, timing, etc in /etc/zaptel.conf |
| 13:30.56 | syle | how do you make it so you can type in it again |
| 13:31.11 | fjean | mimmus, ok |
| 13:31.24 | MRH2 | ne1 use the TE411P ? |
| 13:31.46 | jontow | :P |
| 13:31.48 | jontow | so unlock it, hehe |
| 13:32.12 | ai-a | when using a softphone to connect to asterisk via IAX, the audio is clicky,, is there a reason why? ive tried codec u-law and GSM,, its on my local lan. |
| 13:32.15 | syle | ctrl-c dopn;t work |
| 13:32.56 | jontow | hit escape to get out of copy mode |
| 13:33.04 | jontow | and if its asking for a password, its your login password |
| 13:33.33 | Red15 | is it possible to put a CAPI capable isdn modem into a ZAP Trunk ? |
| 13:33.34 | fjean | mimmus, can you send me yours ? i got some info from the wiki, but i don't know if they will work |
| 13:34.05 | Red15 | ai-a : what specs do you have on the * computer ? |
| 13:34.16 | Mimmus | fjean: they are different in different countries |
| 13:34.23 | Hmmhesays | <PROTECTED> |
| 13:34.26 | fjean | mimmus, ok |
| 13:34.45 | fjean | mimmus, that makes sens ! |
| 13:34.53 | ai-a | 3.2ghz P4 HP server with 1GB ram |
| 13:35.01 | Mimmus | fjean: in Italy, I have: span=1,1,2,ccs,hdb3,crc4,yellow |
| 13:35.49 | fjean | mimmus, ok |
| 13:35.49 | ai-a | Red15: its not the pc specs :) |
| 13:35.50 | Red15 | ai-a: hmm so it's not that :D |
| 13:35.50 | *** join/#asterisk Tili (i=Tili@59.42.97.223) |
| 13:35.50 | fjean | mimmus, you ran into lots of difficulties to make it work ? |
| 13:36.16 | *** join/#asterisk Emea (n=usr@adsl-62-167-38-104.adslplus.ch) |
| 13:36.27 | *** part/#asterisk MRH2 (n=Mr_happy@fcirc-adsl.demon.co.uk) |
| 13:36.50 | Mimmus | fjean: only with different cable connectors (120 Ohm has a non-standard DB9 connector) |
| 13:36.58 | fjean | mimmus, ok |
| 13:37.03 | Mimmus | fjean: no problems with coax+balun |
| 13:38.36 | ai-a | Red15: is there a way i can connect directly to a demo PBX to hear the quality via IAX? |
| 13:39.02 | Red15 | ai-a you could setup an echo extensions |
| 13:39.09 | ai-a | i have that. |
| 13:39.11 | Red15 | or by default asterisk installation dial 500 |
| 13:39.46 | ai-a | yep,, my voice sounds ok,, when i dial 500 and transfer to the demo it is still clicky,, on IAXComm and FireFly. |
| 13:40.04 | *** join/#asterisk Romik (n=romik_@212.143.5.146) |
| 13:40.20 | Red15 | you mean when your pbx dials to the digium pbx ? |
| 13:40.23 | *** join/#asterisk qwue (n=vikramhe@210.212.195.134) |
| 13:40.28 | ai-a | yep |
| 13:40.45 | Red15 | this would suggest a QoS/Congestion problem |
| 13:40.56 | ai-a | i dont have qos on my networks. |
| 13:41.25 | mutilator | obviosly |
| 13:41.36 | Dovid | QOS is a must for quality |
| 13:41.49 | ai-a | why? msn and skype work fine. |
| 13:42.00 | ai-a | and this is over a local lan with no activity. just 2 ps |
| 13:42.01 | ai-a | *pcs |
| 13:42.02 | Mimmus | another question: any idea why Asterisk doesn't disable echo cancellation during fax transmission? |
| 13:42.18 | Dovid | if u have a lot of file transfers etc. skype and yahoo wont work well |
| 13:42.26 | Red15 | sorry Mimmus no idea, probably you'd be best to check/hack the sourec |
| 13:42.27 | *** join/#asterisk kevin_ (i=1001@rrcs-67-78-250-234.se.biz.rr.com) |
| 13:42.50 | ai-a | Dovid: QoS over a 1GB connection between two pcs? |
| 13:42.57 | ai-a | 1Gbit. |
| 13:43.09 | Dovid | ai-a: do you have a lot transfer ? |
| 13:43.13 | *** join/#asterisk meppl (n=mephisto@p54AAFE62.dip.t-dialin.net) |
| 13:43.15 | qwue | hmmhe i guess we know the prob |
| 13:43.35 | ai-a | Dovid: well im transfering IRC packets. |
| 13:43.40 | Dovid | hehe |
| 13:43.47 | Dovid | i dont know y there should be a problem then |
| 13:43.50 | Red15 | oh lord, the incredible network load of irc :P |
| 13:43.53 | qwue | i need a open source softphone? |
| 13:44.14 | qwue | which will work for net2phone |
| 13:44.26 | qwue | preferably linux based |
| 13:44.28 | ai-a | qwue: open source as in to use it, or to code with? |
| 13:44.48 | *** join/#asterisk dalabera (n=Dalabera@pmr.pmrtechnologies.com) |
| 13:44.54 | qwue | i wanna see the config of the softphone |
| 13:45.09 | qwue | to use it |
| 13:46.42 | ai-a | also, why do i get an increasing delay when someone connects to me on the pbx? doesnt the softphones drop packets to keep up with the stream? it increases to 30 seconds after a while. |
| 13:46.42 | *** join/#asterisk SpaceBass (n=sp@c-24-125-184-203.hsd1.va.comcast.net) |
| 13:46.46 | SpaceBass | morning |
| 13:53.30 | *** join/#asterisk jbot (i=ibot@pdpc/supporter/active/TimRiker/bot/apt) |
| 13:53.30 | *** topic/#asterisk is Preview the new website! http://beta.asterisk.org || Asterisk: The Open Source PBX || Asterisk 1.2 Janitors Needed! http://dev.asteriskdocs.org || Astricon 2005 in Anaheim Oct 12-14 http://www.astricon.net/2005/ - speakers wanted |
| 13:53.42 | ManxPower | not softswitch, opencall. |
| 13:53.43 | SpaceBass | wow... that would be amazing...IM to phone |
| 13:53.43 | edgecase | ok, so where is Steve Underwood? |
| 13:53.56 | cochi | space, inevitable. just the question who will survive |
| 13:54.00 | edgecase | coppice.org is down, no DNS servers responding |
| 13:54.04 | azrishahril | Hmmhesays : i'm new to h323 (voip) ... thought we have to set it if termination partner enable fast start |
| 13:54.06 | Dovid | hmm |
| 13:54.09 | Dovid | dns rundown? |
| 13:54.28 | SpaceBass | cochi I was trying to explain this to a non-techy friend yesterday... its basically the demise of the phone companies...brilliant! |
| 13:54.44 | *** join/#asterisk meppl (n=mephisto@p54AAFE62.dip.t-dialin.net) |
| 13:54.56 | cochi | the big carriers start using VoIP, too. at least on their backbones to move off old tech |
| 13:55.11 | cochi | and at least here in germany it won't be so long until all majors offer voip. some already do |
| 13:55.27 | syle | problem was they didn;t change the prices and invited competition |
| 13:55.28 | edgecase | so if anyone can reach Steve Underwood, they should tell him he's off the net |
| 13:55.30 | cochi | same stuff about "turning markets upside down" as with ebay, amazon, the internet itself etc |
| 13:55.41 | SpaceBass | my carrier offers VoIP but I have to use an ATA, they won't let me directly connect |
| 13:56.01 | syle | so hack the ATA and get the sip settings |
| 13:56.07 | cochi | well asterisk helped me somehow ;) external isdn, internal via zap, happy ;) |
| 13:56.12 | SpaceBass | syle if I had the skill.... |
| 13:56.27 | Hmmhesays | azrishahril, what is the problem you are having |
| 13:57.02 | syle | locking ata;s is just becoming famous for not switch LD providers, big scam |
| 13:57.28 | *** join/#asterisk file[lap1op] (n=jcolp@mctnnbsa30w-156034050174.nb.aliant.net) |
| 13:58.01 | SpaceBass | right now I just use 2 x100ps for a connection from my local carrier... and after all the problems friends have had with their VoIP service, I'm not itching to try it |
| 13:58.07 | *** join/#asterisk muckdog (n=jay@coal.acdmis.com) |
| 13:58.12 | SpaceBass | I'll stick with BV for voip for now |
| 13:59.13 | *** join/#asterisk Nix (n=Nix@81.214.255.57) |
| 13:59.16 | syle | yeah i have a tdm400P for my ILEC |
| 13:59.29 | eisman | I use BV, I am not happy with the call quality to germany. |
| 13:59.38 | azrishahril | Hmmhesays : i'm trying to connect asterisk box to forward calls to AS5x00 via h323 (chan_h323) |
| 14:00.03 | SpaceBass | i just made my fiirst intenational call to Ireland with BV.. I was quite impressed with the quality |
| 14:00.29 | *** join/#asterisk DrWho (n=MIKE@mike-new.tc3net.com) |
| 14:02.02 | *** join/#asterisk elwin (n=yellowsn@dsl-213-134-245-123.solcon.nl) |
| 14:03.58 | Hmmhesays | azrishahril, and you are having issues? |
| 14:04.14 | azrishahril | Hmmhesays : yes |
| 14:04.43 | azrishahril | Hmmhesays : call connect , but no voice appear |
| 14:05.02 | *** join/#asterisk Shizzla (i=steve@d029123.adsl.hansenet.de) |
| 14:06.04 | Hmmhesays | using which h323chan? |
| 14:06.44 | azrishahril | Hmmhesays :chan_h323 hufone |
| 14:06.50 | Hmmhesays | try oh323 |
| 14:07.24 | Hmmhesays | give me some details and I can send a test call with it |
| 14:08.06 | azrishahril | Hmmhesays : oh323 => done it , but wonder how to configure multiple AS5x00 with different FastStart & Tunneling ? |
| 14:08.08 | azrishahril | any idea ? |
| 14:08.39 | Hmmhesays | just leave them all enabled |
| 14:08.54 | Hmmhesays | where calls failing because you have fast start and tunneling enabled? |
| 14:09.48 | azrishahril | Hmmhesays : about to perform packet capture using ethereal ... maybe i can get in-depth info |
| 14:09.51 | musical_Duck | Going crazy, all asterisk generated audio (moh, rings, agentlogin) not working, festival is working tho and all was well on previous version, any thoughts? |
| 14:09.54 | ai-a | Red15 Dovid: this is interresting, when i mean a voice message is nice a smooth |
| 14:10.04 | Katty | Hmmhesays: mew |
| 14:10.09 | Hmmhesays | hey |
| 14:10.19 | Red15 | ai-a ? |
| 14:10.27 | Hmmhesays | azrishahril, I don't know of those settings force fast start and tunneling or not |
| 14:10.29 | RoyK | ai-ai-ai |
| 14:10.32 | ai-a | Red15: this clicking QoS issue. |
| 14:10.43 | RoyK | ai-a: clicking? |
| 14:10.50 | Bhaal | Is anyone else having trouble with FWD iax2 lately? |
| 14:10.51 | Red15 | ai-a yes. when is it smooth ? |
| 14:10.51 | *** join/#asterisk astadmin (n=shafqat@pk-isb-trg-sc01-001.speedcast.com) |
| 14:11.15 | ai-a | i left a voice message on exr 12345, and played it in winamp and its fine.. its just in my ears |
| 14:11.16 | RoyK | if it is SIP, it's the lack of a jitterbuffer that's causing the noise |
| 14:11.22 | ai-a | its IAX. |
| 14:11.36 | RoyK | stable? head? trunk or single? |
| 14:11.36 | Hmmhesays | unless you got zoa's patch |
| 14:11.40 | Hmmhesays | I hear it works pretty well |
| 14:11.52 | RoyK | Hmmhesays: i'm using it on a test rig here |
| 14:11.54 | RoyK | works nice |
| 14:11.58 | RoyK | nicely, even |
| 14:12.30 | Hmmhesays | which patch did you grab RoyK? cause the one I found didn't patch rtp.c real nice |
| 14:12.42 | ai-a | RoyK: i have 2 problems,, 1) when i listen to the demo speech its clicking,, i dont have any QoS, but its over a lan with min activity.. 2) in this IAXComm softphone over the ISP internet,, a delay increases upto and not stopping at 30 seconds.. doesnt it drop packets to keep up with the audio stream? |
| 14:15.07 | *** join/#asterisk file[laptop] (n=jcolp@mctnnbsa30w-156034053056.nb.aliant.net) |
| 14:15.26 | *** join/#asterisk shift (i=shift@82-39-34-36.cable.ubr01.benw.blueyonder.co.uk) |
| 14:16.51 | *** join/#asterisk santiago (n=santiago@63.245.86.163) |
| 14:17.20 | *** part/#asterisk santiago (n=santiago@63.245.86.163) |
| 14:17.57 | *** join/#asterisk meppl (n=mephisto@p54AAFE62.dip.t-dialin.net) |
| 14:18.10 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 14:21.27 | qwer^^ | anyone how do i change te system date in linux server |
| 14:21.39 | kFuQ | date |
| 14:21.42 | SpaceBass | pinfo date |
| 14:21.43 | kFuQ | or ntpdate |
| 14:21.55 | *** join/#asterisk Qwell (n=north@24-50-66-194.vnnyca.adelphia.net) |
| 14:22.20 | *** join/#asterisk MattH (n=MattH@63.174.244.174) |
| 14:23.44 | *** join/#asterisk mkrufky (n=mk@68.160.103.77) |
| 14:26.33 | *** join/#asterisk mhnoyes (n=mhnoyes@user-38lc169.dialup.mindspring.com) |
| 14:30.07 | mutilator | anyone know if there is a way to monitor channels on a T with a cisco? |
| 14:30.36 | *** join/#asterisk pussfeller (n=todd@12.150.129.171) |
| 14:30.42 | bendy24_ | snmp? |
| 14:30.49 | *** join/#asterisk Lan16`spdy^gTp (n=spdy@altom.net1.nerim.net) |
| 14:30.52 | *** join/#asterisk klasstek (n=nunyobiz@sta-206-168-218-206.rockynet.com) |
| 14:36.03 | mutilator | bendy24_: active monitoring tho? |
| 14:36.28 | spdy | hi |
| 14:36.29 | bendy24_ | well, then you would probably need to sniff the packets |
| 14:37.05 | *** join/#asterisk newl (n=newlook@203-59-187-240.dyn.iinet.net.au) |
| 14:37.13 | spdy | i have a problem with ParkAndAnnounce command, when i want to transfer the call, it rings but when i take the call its hanging up, someone know ? |
| 14:37.50 | *** join/#asterisk cliffton_2005 (i=cliffton@adsl-69-105-45-36.dsl.pltn13.pacbell.net) |
| 14:39.30 | *** join/#asterisk file (n=jcolp@mctnnbsa30w-156034056020.nb.aliant.net) |
| 14:39.52 | MattH | quick question here for anyone running voip over the net... sprint is telling me that the jitter I"m seeing on two of their routers is "normal internet traffic changes".. I think it's an overlaoded router... things like spikes from 10ms to 253ms |
| 14:40.28 | dalabera | hello guys, |
| 14:40.47 | *** join/#asterisk file[lap1op] (n=jcolp@156.34.56.20) |
| 14:41.44 | *** join/#asterisk Uther_P (n=uther_p@66.180.120.82) |
| 14:41.56 | dalabera | I have 2 old T400p cards that I would like to use them together on a server. Do I have to change the numeric setting (onboard) on those cards? |
| 14:42.38 | *** join/#asterisk firebalrog (n=Paul_j_b@209-58-245-194.cust.telepacific.net) |
| 14:43.20 | musical_Duck | Cheers, good-bye and good-luck! |
| 14:43.49 | *** join/#asterisk Romik (n=romik_@212.143.5.146) |
| 14:44.48 | Katty | Hmmhesays: you wouldn't happen to know about a list somewhere that shows how many simultaneous phone calls can happen on a particular computer (processor/ram/etc) would you? |
| 14:45.21 | Katty | Hmmhesays: like an Asterisk Tested blah blah blah if you use a 2.8ghz processor with 1 gig of ram you can have blah phone calls at once time outline thingy |
| 14:45.21 | [Airwolf] | Katty, that's something I would like to know to |
| 14:45.34 | file | Katty: it's dependent on too many variables... |
| 14:45.49 | Hmmhesays | file is correct |
| 14:45.50 | Ariel_ | dalabera, yes |
| 14:45.52 | file | ie: what codec, if transcoding is going on, any background applications going as well, if it's just regular calls, etc |
| 14:46.06 | Katty | file: so i've no hope of any information like that? :< |
| 14:46.19 | Hmmhesays | Katty: not really |
| 14:46.35 | Hmmhesays | however, if you have some specifics that you are curious about usually one of us can ballpark it |
| 14:46.47 | Katty | http://www.voip-info.org/wiki-Asterisk+dimensioning <- oooh |
| 14:46.51 | Katty | ^- that helps at least (= |
| 14:47.00 | Hmmhesays | you can get some rules of thumb out of there |
| 14:47.16 | Katty | Hmmhesays: well that's the point. my boss wants a little page that says This Is When Asterisk Explodes type thing |
| 14:47.23 | file | yay hot-n-sexy Hmmhesays |
| 14:47.27 | Hmmhesays | hey file |
| 14:47.29 | Hmmhesays | howsit |
| 14:47.30 | Hogie | heh, I can do 5 calls (ulaw to Zap) with recording on an AMD1900 with 1GB ram with 0.01 userload if that helps Katty |
| 14:47.35 | Katty | Hmmhesays: so if a client calls in and says I Want 100 Lines, we know what to tell them |
| 14:47.37 | file | I'm sad, my workstation restarted and I lost a word document I was working on |
| 14:47.47 | Katty | file: aww |
| 14:47.54 | Hmmhesays | you tell them you need a dual xeon box with 1gig ram and raid |
| 14:47.54 | Hogie | of course,we haven't tried more than that (or had that many more on) |
| 14:48.01 | Hmmhesays | thats no good |
| 14:48.06 | gordonjcp | Katty: it seems to work for lowish call volumes on a PII-350 |
| 14:48.07 | Hmmhesays | autosave every 60 seconds |
| 14:48.19 | gordonjcp | Katty: just for shits and giggles, I plan to install it on my libretto |
| 14:48.38 | Ariel_ | Katty, your running a box right now. |
| 14:49.05 | Hmmhesays | the rule of thumb is 120 sip<-> sip transcoded calls per dual p4 |
| 14:49.06 | Ariel_ | When you get to your busy time most users on the system if you do uptime it will give you a load reading you can use that to figure your needs |
| 14:49.31 | Katty | Ariel_: yes, i am (= |
| 14:50.01 | *** join/#asterisk [1]JohnJacob (n=JohnJaco@pcp0011543387pcs.mainf01.in.comcast.net) |
| 14:51.06 | Ariel_ | 07:50:56 up 149 days, 10:50, 2 users, load average: 0.66, 0.48, 0.31 |
| 14:51.24 | Ariel_ | Katty, this is a dual server running about 90 channels right now |
| 14:51.44 | Hmmhesays | if you are using zaptel though the rules of the game change |
| 14:52.09 | Ariel_ | ture this one box has 2 TE410p with 8 pri's going into it. |
| 14:52.23 | Hmmhesays | you get 96 zap calls come in at once on a quad card and your box is going to go 'fsck joo' |
| 14:52.39 | Ariel_ | Hmmhesays, really |
| 14:52.51 | *** join/#asterisk allanon (i=allanon@c-24-18-189-126.hsd1.wa.comcast.net) |
| 14:52.53 | Ariel_ | humm we have had this box up to 150 channels no problems |
| 14:52.58 | Hmmhesays | chances of that actually happening are slim to none |
| 14:53.17 | Hogie | Ariel: but they are on diff pci busses, right? |
| 14:53.29 | *** join/#asterisk yartelecom (n=no-email@62.33.182.163) |
| 14:53.34 | Hmmhesays | Ariel_: I meant if they came in simultaneous |
| 14:53.37 | Ariel_ | Hogie, yes we disable everythign we don't need |
| 14:53.48 | Hmmhesays | all call setups at once |
| 14:54.04 | Ariel_ | Hmmhesays, they do it's a main server for a small telco |
| 14:54.19 | Hmmhesays | hmm interesting |
| 14:54.29 | Hmmhesays | you wouldn't want to paste me the hardware specs on that box in a pm would you? |
| 14:54.36 | Hogie | heh |
| 14:54.48 | Hogie | Hmmhesays: I bet it has to do with it having multi PCI busses |
| 14:55.12 | Hogie | I think that would help stop IRQ challenges |
| 14:55.13 | Hmmhesays | I bet you are correct |
| 14:55.23 | Ariel_ | Hmmhesays, it's an intel MB, Dual Xeon 2.8 1 gig ram external raid box using QFS there actually 3 boxes being used with hartbeat. |
| 14:55.37 | Hmmhesays | interesting |
| 14:56.30 | Ariel_ | at around 1 pm I will look at the load it's in the west cost so allot of people get on at that time. |
| 14:57.40 | Ariel_ | Hmmhesays, there is no moh on it just for your info |
| 14:58.12 | Hmmhesays | heh I would hope not |
| 14:58.14 | *** join/#asterisk cliffton_2005 (i=cliffton@adsl-69-105-45-36.dsl.pltn13.pacbell.net) |
| 14:58.27 | Hmmhesays | ok kde's lack of the arial font irritates me |
| 14:58.32 | MmmmToop_ | Hmmmmhesays: what are you doing with a Dual Xeon...quite a beast of a machine...? |
| 14:59.31 | Hmmhesays | dual 2.8ghz xeons are not very beastly anymore |
| 14:59.46 | Red15 | Could anyone tell me what the price is for the Digium Development Kit ? |
| 14:59.57 | *** join/#asterisk lot (n=lot@68.148.192.184) |
| 15:00.05 | MmmmToop_ | me thinks they are ; ) |
| 15:00.09 | visik7 | Red15 what a ddk does ? |
| 15:00.19 | Red15 | no the average price |
| 15:00.25 | Hmmhesays | MmmmToop_, well the next dually I will be using is going to be for a hotel, 90 rooms |
| 15:00.33 | visik7 | Red15 is a question for me :) |
| 15:00.44 | Red15 | hmm that's why i asked :P |
| 15:01.00 | visik7 | I asked what a digium developer kit is used for |
| 15:01.02 | MattH | with cvs-head I'm trying to use cdr_odbc and I get cdr_odbc: Error SQLConnet -1 (unable to connect to database: MSSQL-asterisk).... any thoughts? I can connect just fine from the command prompt (telnet server 1433) |
| 15:01.09 | Hmmhesays | voicemail, directory etc... |
| 15:01.13 | MmmmToop_ | cool |
| 15:01.38 | Red15 | visik7 and what did they tell ? |
| 15:01.48 | Hmmhesays | dell has reasonable pricing on their hardware, and in a hotel setup you need to garauntee high availability |
| 15:01.49 | MmmmToop_ | u could pop calling card in there as well while you are at it & earn add. rev.! ; ) |
| 15:02.01 | visik7 | Red15 I asked to you |
| 15:02.09 | Ariel_ | Red15, it's about 190.00 |
| 15:02.12 | Hmmhesays | cause the minute a guest picks up the phone hits 0 and doesn't get the front desk, there will be hell to pay |
| 15:02.27 | Red15 | visik7: oh that's a nice price |
| 15:02.34 | *** join/#asterisk blessen (n=Bless@140.99.23.26) |
| 15:02.42 | blessen | hi |
| 15:02.51 | Ariel_ | Hmmhesays, if your going to setup for an hotel you should look at the Redfone brick. |
| 15:03.14 | blessen | i Need some help..I got astrisk installed..but i cannot connect it with kphone |
| 15:03.24 | MmmmToop_ | saw it on voipsupply...what if the brick fails!!! ; ) |
| 15:03.25 | blessen | i wish to use softphones |
| 15:03.45 | Hmmhesays | Ariel_, dont' need it |
| 15:04.10 | Hmmhesays | vonage will be providing the incoming lines, and for the rooms, i'm punching down 4 1124's in the phone room |
| 15:04.10 | Ariel_ | MmmmToop_, well you can put two on them on the setup. |
| 15:04.29 | Ariel_ | 1124? |
| 15:04.46 | Hmmhesays | mediatrix 1124's |
| 15:04.57 | Hmmhesays | that way nothing in the rooms has to be re-wired |
| 15:05.17 | blessen | me and my friends are using 2 systems and we have an VOIP connection. I wish to get calls on our linux system which has astrisk installed. But i am not sure about the configurations and i am not able to connect it with softphones like kphone. |
| 15:06.01 | *** join/#asterisk slak- (n=alex@dsl093-236-129.hfd1.dsl.speakeasy.net) |
| 15:06.06 | slak- | dudes wtf is up witht his |
| 15:06.08 | slak- | <PROTECTED> |
| 15:06.13 | RoyK | argh. how can i stop mantis emailing me about a bug? |
| 15:06.19 | slak- | and REAL shitty conversation quality |
| 15:06.23 | Katty | when do you hit asterisk's limit? |
| 15:06.26 | Hmmhesays | Ariel_, the 1124 is adapted perfectly for this type of setup |
| 15:06.37 | Katty | voip-info is suggesting that 40 calls on a single box is pushing it |
| 15:06.51 | Katty | but you can combine asterisk servers |
| 15:06.52 | Ariel_ | Katty, they lie |
| 15:06.53 | Hmmhesays | maybe on a dual p3 |
| 15:07.00 | Hmmhesays | transcoding |
| 15:07.02 | Hmmhesays | 800mhz |
| 15:07.05 | Hmmhesays | haha |
| 15:07.06 | Katty | so, if you bridge lots of asterisk servers together... |
| 15:07.20 | slak- | dudes Aug 25 11:47:14 WARNING[18809]: chan_iax2.c:6010 socket_read: Received mini frame before first full voice frame wtf is that |
| 15:07.38 | Hmmhesays | 120 calls on a slow dual p4 is the rule of thumb |
| 15:07.55 | Hmmhesays | after that just keep adding boxes |
| 15:07.55 | MmmmToop_ | mmmm...interesting |
| 15:08.16 | blessen | Hi |
| 15:08.25 | *** join/#asterisk fugitivo (n=ajf@201.255.101.80) |
| 15:09.11 | Hmmhesays | then either use SER to load balance between them, use DNS to load balance or do it manually. someone correct me if i'm wrong please |
| 15:09.51 | *** join/#asterisk twisted (n=twisted@asterisk/friend-and-developer/pdpc.professional.twisted) |
| 15:09.51 | *** mode/#asterisk [+o twisted] by ChanServ |
| 15:10.17 | MikeJ[Laptop] | digium tests business eddition to 120 calls per box |
| 15:10.39 | Ariel_ | I just want to make sure peopld don't think calls are how many device or sip accouts you have |
| 15:10.48 | MikeJ[Laptop] | no, active calls |
| 15:10.58 | MikeJ[Laptop] | so up to 240 "call legs" |
| 15:11.04 | Ariel_ | MikeJ[Laptop], correct |
| 15:11.07 | MikeJ[Laptop] | in zap out sip = 1 call |
| 15:11.23 | Hmmhesays | on what type of hardware? |
| 15:12.28 | *** join/#asterisk [Outcast] (n=knoppix@c-24-218-94-11.hsd1.ma.comcast.net) |
| 15:12.43 | [Outcast] | any status on queue and the deadlock issue? |
| 15:14.31 | visik7 | deadlock where ? |
| 15:16.52 | Red15 | can anyone tell me if there is a lot of trouble installing 2 TDM400's ? |
| 15:17.33 | spackle | Red15: do you have lots of free interrupts, or can you assign them on your motherboard through the bios? |
| 15:17.43 | Red15 | most probably |
| 15:18.01 | spackle | Red15, then probably won't be a problem. |
| 15:18.09 | Red15 | hehe :D |
| 15:18.18 | Red15 | no special driver trix |
| 15:18.49 | visik7 | tdm400 doesn't support irq sharing ? |
| 15:18.49 | spackle | Red15 just defining your channels in zapata.conf and zaptel.conf. |
| 15:19.12 | Red15 | ah so channels go from 1-8 instead of typical 1-4 then ? |
| 15:19.30 | *** join/#asterisk hellagony (n=hellagon@200.121.129.178) |
| 15:19.31 | spackle | yup. Are they FXS or FXO or a mix? |
| 15:19.39 | *** join/#asterisk adjacent (n=scott@cpe-024-168-216-235.sc.res.rr.com) |
| 15:19.55 | *** join/#asterisk tzafrir_laptop (n=tzafrir@bzq-179-75-202.cust.bezeqint.net) |
| 15:20.23 | *** join/#asterisk viLeR (i=1000@ip-47-252.telesat.com.co) |
| 15:21.18 | MmmmToop_ | bye...must go home to wife.. |
| 15:21.28 | sivana | buh bye |
| 15:21.52 | wzl | g927 works great :P |
| 15:22.00 | wzl | g729 |
| 15:22.28 | *** part/#asterisk colombus (n=colombus@193.190.210.151) |
| 15:23.34 | *** join/#asterisk malaiwah (n=malaiwah@Quebec-HSE-ppp242302.qc.sympatico.ca) |
| 15:24.36 | *** join/#asterisk bonez41 (n=aint@c-67-166-77-14.hsd1.ut.comcast.net) |
| 15:25.33 | malaiwah | i'm trying to put my extensions configuration in a mysql database using odbc; how do i include other contexts when i'm in the database ? there is no exten/priority for an include.. |
| 15:26.16 | Hmmhesays | taint bad |
| 15:26.53 | *** join/#asterisk anthm (n=anthmct@CPE-69-76-83-52.wi.res.rr.com) |
| 15:26.53 | *** mode/#asterisk [+o anthm] by ChanServ |
| 15:28.52 | *** join/#asterisk postel (n=jp@unaffiliated/postel) |
| 15:29.47 | *** join/#asterisk santiago (n=santiago@63.245.86.163) |
| 15:31.01 | FarrisG | Anyone know how to get X-Lite for Linux to move to foreground when the phone rings in Gnome? |
| 15:31.42 | [Airwolf] | FarrisG, If you fins out, please let me know. :) |
| 15:31.45 | [Airwolf] | find |
| 15:31.55 | *** join/#asterisk jake1932 (n=jake1932@pool-70-16-142-207.phil.east.verizon.net) |
| 15:35.32 | jake1932 | anyone have a 7960 that reboots when on a call and another one comes in? |
| 15:35.50 | postel | what load you're on? |
| 15:36.44 | kajtzu | skinny or sip |
| 15:36.49 | jake1932 | sip |
| 15:36.56 | postel | jake1932: version? |
| 15:36.58 | kajtzu | doesnt happen with 7.5 at least ;) |
| 15:37.06 | jake1932 | let me see what version |
| 15:37.09 | sivana | anyhone getting -- Got SIP response 500 "Internal Server Error" back from 147.135.20.128 from Broadvoice? |
| 15:37.10 | postel | jake1932: settings - 5 -3 |
| 15:37.19 | jake1932 | 7.03 |
| 15:37.24 | kajtzu | jake1932: upgrade |
| 15:37.31 | jake1932 | will do |
| 15:37.34 | jake1932 | to? |
| 15:37.38 | jake1932 | 7.5? |
| 15:37.40 | kajtzu | 7.5 is the latest and greatest |
| 15:37.52 | jake1932 | tnx |
| 15:38.23 | kajtzu | came out 11-jul-05 |
| 15:39.11 | *** join/#asterisk blessen (n=Bless@140.99.23.26) |
| 15:39.11 | dexteruk | I have Digium Analog TDM 4 Port Card FXS it is not detecting the busy tone |
| 15:39.35 | blessen | i get error when i try to connect kphone to asterisk server |
| 15:39.45 | jake1932 | dexteruk: there are a few otions in the Makefile for detecting busy - did you try different options? |
| 15:39.59 | blessen | erro --> chan_sip.c:9808 handle_request_register: Registration from '"Blessen" <sip:kphone@192.168.15.2>' failed for '192.168.15.2' |
| 15:40.01 | *** join/#asterisk chris78 (n=schulte@gate.sympat.de) |
| 15:40.06 | dexteruk | Which makefile? |
| 15:40.20 | dexteruk | Zaptel or Asterisk? |
| 15:40.33 | jake1932 | don't remeber - check both |
| 15:40.52 | slak- | is asterisk.org down? |
| 15:40.56 | malaiwah | i'm trying to put my extensions configuration in a mysql database using odbc; how do i include other contexts when i'm in the database ? there is no exten/priority for an include.. |
| 15:40.59 | jake1932 | says "busy detect" or something like that |
| 15:41.11 | slak- | is asterisk.org down? |
| 15:41.13 | *** join/#asterisk huehner (n=huehner@kt-gmbh.de) |
| 15:41.30 | jontow | no |
| 15:41.35 | slak- | the websitre |
| 15:41.36 | slak- | -r |
| 15:41.40 | jontow | it was getting hit heavily for a few minutes though |
| 15:41.55 | slak- | ah there we go |
| 15:41.59 | blessen | can anyone help me with this...chan_sip.c:9808 handle_request_register: Registration from '"Blessen" <sip:kphone@192.168.15.2>' failed for '192.168.15.2' |
| 15:42.11 | jontow | looks like it still might be at times |
| 15:42.27 | slak- | yea its crawling |
| 15:42.34 | jontow | yep |
| 15:42.37 | blessen | i get that when i try to connect to asterisk from kphone |
| 15:43.00 | [Outcast] | any status on queue and the deadlock issue? |
| 15:43.11 | blessen | i followed whats said in http://www.voip-info.org/tiki-index.php?page=Asterisk+phone+Kphone |
| 15:43.11 | *** join/#asterisk |dennis| (n=dennis@200.32.215.82) |
| 15:43.37 | dexteruk | ok found it. its using Martin's busydetect at the moment |
| 15:43.57 | dexteruk | I will try the others thanks |
| 15:44.39 | *** join/#asterisk Derkommissar (n=alberto@66.64.215.6.nw.nuvox.net) |
| 15:44.51 | Derkommissar | why is it so hard to order a digium board now days ? |
| 15:45.18 | mutilator | cause you're ordering from the wrong place |
| 15:45.21 | bkw_ | yo Derkommissar |
| 15:45.42 | Derkommissar | sup man |
| 15:45.55 | mutilator | digi techs tell me my te110p will run me 6000ft |
| 15:46.04 | *** join/#asterisk nroej (n=joern@lak-73-8.wohnheime.ruhr-uni-bochum.de) |
| 15:46.06 | nroej | hi |
| 15:46.15 | mutilator | "the specs tell us theoretically it will run 6000ft." |
| 15:46.54 | blessen | hi i have an error..can anyone help me with this... |
| 15:47.13 | blessen | i cannot connect from kphone to asterisk server |
| 15:47.39 | bkw_ | 6000ft? |
| 15:47.40 | dexteruk | jake1932: do you know where it compared the busytones to? which file contains the discriptions? |
| 15:47.41 | bkw_ | BULLSHIT |
| 15:47.44 | bkw_ | their is no way |
| 15:47.47 | *** join/#asterisk shadebob (n=shadebob@ll81-144-114-192-81.ll81.iam.net.ma) |
| 15:48.11 | bkw_ | 6000ft is a little over a mile |
| 15:48.28 | bkw_ | man that would suck |
| 15:48.36 | Nivex | 720ft over to be precise :) |
| 15:48.43 | bkw_ | yep |
| 15:48.57 | mishehu | it'd all be easier if the USA would just frickin use metric |
| 15:49.02 | bkw_ | sure |
| 15:49.02 | Nivex | mishehu: agreed |
| 15:49.17 | bkw_ | but we can't be logical here yo |
| 15:49.29 | bkw_ | while we are at it.. everyone should use NTSC |
| 15:49.29 | mutilator | yeh well |
| 15:49.29 | *** join/#asterisk iCEBrkr_ (i=icebrkr@rrcs-24-129-130-158.se.biz.rr.com) |
| 15:49.29 | mishehu | "An acre is the area of a rectangle whose length is one furlong, and whose width is one chord..." |
| 15:49.31 | Nivex | The United States of Assininity |
| 15:49.32 | mutilator | thats what they said |
| 15:49.43 | bkw_ | down with PAL |
| 15:49.47 | Hmmhesays | haha |
| 15:49.54 | Hmmhesays | NTSC all the way? |
| 15:49.55 | Nivex | bkw_: Never The Same Color ? |
| 15:50.04 | bkw_ | you know most people in the US dont have a clue about NTSC and PAL |
| 15:50.09 | mishehu | there, now do you want to use it? |
| 15:50.13 | Hogie | nope, most people dont |
| 15:50.14 | Hogie | or SECAM |
| 15:50.15 | bkw_ | and they have no clue that the power requirements differ in other coutries |
| 15:50.19 | Hmmhesays | because they don't have to |
| 15:50.24 | mishehu | SECAM is some wonky french thingie |
| 15:50.28 | mutilator | bkw_: most people period |
| 15:50.29 | mishehu | it can be safely ignored |
| 15:50.41 | bkw_ | haha |
| 15:50.43 | Hmmhesays | honestly I don't see why joe blow who works down at the saw mill needs to know about pal period |
| 15:50.45 | Hogie | secam has more lines of resolution than PAL though... |
| 15:50.50 | mishehu | let's use dunams instead of acres while we're at it... 1 sq km is a dunam |
| 15:50.53 | malaiwah | ok.. i'm kinda confused about how does the "Realtime Extensions" works.. I thought I could include all my configuration in the DB but it seems that I must update extensions.conf with a "switch" statement for every new context that I want ? Am I right ? |
| 15:51.12 | *** join/#asterisk jhiver (n=jhiver@aqu33-1-82-66-1-167.fbx.proxad.net) |
| 15:51.14 | mishehu | Hogie: but it's french! |
| 15:51.14 | Hmmhesays | and vice versa of course |
| 15:51.18 | jhiver | hi lads |
| 15:51.28 | Hogie | so? They had to do 1 thing right in their history |
| 15:51.31 | Hogie | it wasnt warfare |
| 15:51.33 | jhiver | is there a way with asterisk to do echo cancel between 2 SIP legs? |
| 15:51.35 | Hogie | so it might as well be TV |
| 15:51.45 | *** join/#asterisk _cleric_ (n=dacleric@p5482B021.dip0.t-ipconnect.de) |
| 15:51.54 | mishehu | yeah, but they think they're better than the rest of us because they have paris. |
| 15:52.04 | mishehu | and all we have is paris hilton |
| 15:52.06 | *** join/#asterisk mhnoyes (n=mhnoyes@user-38lc169.dialup.mindspring.com) |
| 15:52.09 | malaiwah | jhiver: i don't think so, it should not be necessary |
| 15:52.09 | mishehu | and a hilton in paris. |
| 15:52.13 | Pj | and paris texas |
| 15:52.30 | Hogie | I was thinking of flying to paris texas for my first cross country flight |
| 15:52.40 | Hogie | they have a mini tower there.... |
| 15:53.07 | *** join/#asterisk dgilmore (n=dennis@ws-135.royalpublishing.com) |
| 15:53.08 | Hogie | and they have a jesus wearing cowboy boots (like an 8ft tall jesus statue) at some guy's grave |
| 15:53.10 | malaiwah | jhiver: echo cancellation is usually needed when the tx and rx signal are on the same wire, like POTS |
| 15:53.23 | jhiver | malaiwah: k |
| 15:53.24 | mishehu | yeah, well, you know what they say... |
| 15:53.29 | jhiver | shame :( |
| 15:53.39 | mishehu | only in tex-ass. |
| 15:54.14 | slak- | my crazy roomate wants to buy a katamaran |
| 15:54.16 | *** part/#asterisk santiago (n=santiago@63.245.86.163) |
| 15:54.19 | Katty | mew? |
| 15:54.20 | Katty | oh |
| 15:54.22 | slak- | anyone ever rode one of those things before |
| 15:54.23 | Katty | silly hilight |
| 15:54.31 | mishehu | that and "only two kinds come from texas... steers, and queers..." |
| 15:54.40 | syle | hahaa |
| 15:54.57 | syle | only 2 seasons in canada winter and construction |
| 15:55.05 | mishehu | so who wants to play the part of private cowboy? |
| 15:55.06 | syle | vancouver...also nicknamed HONGcouver |
| 15:55.14 | bkw_ | FYI linux threads suck |
| 15:55.28 | Hogie | well no shit bkw |
| 15:55.30 | bkw_ | NPTL rocks |
| 15:55.39 | Katty | your socks. |
| 15:55.49 | tzanger | ugh |
| 15:55.52 | tzanger | get it right people |
| 15:55.54 | mishehu | bkw_: but didn't you say nptl was more difficult to degbu? |
| 15:55.57 | tzanger | ${FOO} rox yer sox |
| 15:55.58 | mishehu | debug even |
| 15:56.19 | bkw_ | mishehu, its linux.. its not harder than pulling your hair out. |
| 15:56.32 | bkw_ | the debug output just looks a tad bit diff |
| 15:56.32 | Katty | mee sha whoo |
| 15:56.42 | *** join/#asterisk Broom (i=Broom@jescobar.ayustar.net) |
| 15:57.00 | bkw_ | is that how you say it? |
| 15:57.01 | Broom | hello all |
| 15:57.01 | mishehu | bkw_: I have a lot of hair on my head still... seems pretty hard for me to pull it out. ;-) |
| 15:57.08 | Katty | bkw_: obviously. |
| 15:57.10 | bkw_ | mishehu, you have short hair |
| 15:57.14 | mishehu | it's mishehu |
| 15:57.16 | Katty | everyone has short hair. |
| 15:57.16 | Hmmhesays | I have a lot of hair, period |
| 15:57.29 | Katty | even Hmmhesays |
| 15:57.29 | Broom | i have a problem with an analog phone conected to an fxs port, instead of a dialtone I get a ring back, any idea? |
| 15:57.29 | Katty | Hmmhesays: lies. |
| 15:57.31 | Katty | Hmmhesays: it does /not/ go to your shoulders. |
| 15:57.35 | Katty | Hmmhesays: i have long hair (+ |
| 15:57.36 | Katty | (= |
| 15:57.45 | Nugget | I have cyan hair. |
| 15:57.47 | Katty | mishehu: mee she who? |
| 15:58.08 | Hmmhesays | mee shey hoo? |
| 15:58.15 | bkw_ | Nugget, did a smurf pee in your hair? |
| 15:58.21 | Nugget | something like that |
| 15:58.27 | *** join/#asterisk Mimmus (n=viggiani@ext.pitagora.it) |
| 15:58.29 | Katty | Nugget is a freak. |
| 15:58.37 | hugo-v6 | .o(hmmmm pee) |
| 15:58.46 | bkw_ | ok hugo-v6 you scare me at times |
| 15:58.47 | Nugget | I'm just doing my part to Keep Austin Weird(tm) |
| 15:58.55 | *** join/#asterisk toddf (n=toddf@adsl-66-141-65-28.dsl.okcyok.swbell.net) |
| 15:59.05 | hugo-v6 | bkw: _you_ say _that_ im amazed :) |
| 15:59.12 | Mimmus | hi, can I ask why TxFax does return always '0' even if fax transmission went bad? |
| 15:59.19 | Hmmhesays | someone dcc me some beer |
| 15:59.21 | *** join/#asterisk Defraz (n=t0tal@tim.ibccom.net) |
| 15:59.34 | hugo-v6 | skoll |
| 15:59.35 | nroej | <PROTECTED> |
| 15:59.48 | Hmmhesays | w00t cheap german beer |
| 15:59.59 | shaon | how to install H323 in asterisk? |
| 16:00.00 | nroej | Hmmhesays: good enogh |
| 16:00.02 | hugo-v6 | Hmmhesays: not that cheap ;) bitburger ;) |
| 16:00.15 | Hmmhesays | nroej, didn't you have the bathtub full yesterday? |
| 16:00.16 | bkw_ | shaon, http://www.pbxfreeware.org/chan_woomera/ |
| 16:00.31 | nroej | Hmmhesays: its still full the party will begin soon |
| 16:00.35 | zoa2 | bkw, we will send you something to read during the weekend :) |
| 16:00.41 | bkw_ | ok |
| 16:00.49 | bkw_ | is it nude photos again? |
| 16:00.54 | *** join/#asterisk shift (i=shift@82-39-34-36.cable.ubr01.benw.blueyonder.co.uk) |
| 16:01.09 | Katty | nude photos? |
| 16:01.18 | hugo-v6 | damn rainy weather. |
| 16:01.24 | bkw_ | Katty, did we corrupt you too? |
| 16:01.25 | Ariel_ | nudes my xchat ping twice |
| 16:01.28 | Hmmhesays | there was a wedding party at playmakers last night, the girls were wearing little pictures of guys with their floppy dongs hanging out |
| 16:01.34 | Katty | bkw_: i was corrupted to start with. |
| 16:01.38 | hugo-v6 | Ariel_: :> |
| 16:01.42 | bkw_ | Katty, kewl |
| 16:01.51 | Katty | Hmmhesays: that is not a pretty mental image. |
| 16:01.58 | *** join/#asterisk xylome (n=asterisk@hg-msq-hol.levigo.de) |
| 16:01.59 | Hmmhesays | indeed it is not |
| 16:02.14 | Hmmhesays | if I want to see that i'll take my pants off |
| 16:02.21 | xylome | <PROTECTED> |
| 16:02.52 | hugo-v6 | xylome: LOL |
| 16:02.52 | Broom | anyone have had that same problem before? |
| 16:03.46 | bkw_ | hahaha |
| 16:03.48 | bkw_ | xylome, |
| 16:04.02 | hugo-v6 | Ariel_: i would do that oo if i got money :/ |
| 16:04.08 | *** join/#asterisk xylome (n=asterisk@hg-msq-hol.levigo.de) |
| 16:04.12 | *** join/#asterisk xylome (n=asterisk@hg-msq-hol.levigo.de) |
| 16:04.15 | cpatry | mouhahah! |
| 16:04.19 | *** join/#asterisk xylome (n=asterisk@hg-msq-hol.levigo.de) |
| 16:04.24 | mishehu | Katty: yep, unlike english, my nickname has no variation of pronounciation. i is like e in me, e is like e in ed, and u is like oo in boom. |
| 16:04.25 | *** join/#asterisk xylome (n=asterisk@hg-msq-hol.levigo.de) |
| 16:04.29 | bkw_ | haha |
| 16:04.33 | Ariel_ | hugo-v6, i did not say anything about money. (does not have any but I an beg). |
| 16:04.37 | bkw_ | xylome, might wanna change your password now |
| 16:04.47 | hugo-v6 | Ariel_: uh ok ;) |
| 16:04.50 | *** join/#asterisk [1]JohnJacob (n=JohnJaco@pcp0011543387pcs.mainf01.in.comcast.net) |
| 16:04.53 | Hmmhesays | on the road again, I just can't wait to be on the road again, my life my love is makin' music with my friends |
| 16:04.57 | *** part/#asterisk slak- (n=alex@dsl093-236-129.hfd1.dsl.speakeasy.net) |
| 16:05.05 | *** join/#asterisk slak- (n=alex@dsl093-236-129.hfd1.dsl.speakeasy.net) |
| 16:05.15 | Katty | me s eh whoooo |
| 16:05.19 | slak- | is this correct to checkout 8/16/2005 code cvs checkout -D "2005/08/16" |
| 16:05.21 | Katty | me sheh whoo! |
| 16:05.33 | Katty | mishehu: me sh eh who? |
| 16:05.38 | hugo-v6 | now going out into the rainy weather to go with my dog "gassi" |
| 16:05.41 | hugo-v6 | l8r pals |
| 16:05.51 | Katty | mishehu: but in three section thingys. |
| 16:05.54 | mishehu | Katty: meee she hu |
| 16:06.10 | mishehu | if you'd like me to pronounce it and send you the wav file it's no problem ;-) |
| 16:06.14 | Katty | mishehu: but does it sound like she or sh eh |
| 16:06.19 | Katty | she a |
| 16:06.56 | Katty | silly chicago people. |
| 16:06.59 | mishehu | hahaha! success! I made the girl explode. |
| 16:07.00 | *** join/#asterisk jero (n=jero@savoirfairelinux.net) |
| 16:07.09 | mishehu | Katty: mishehu has nothing to do with chicago. ;-) |
| 16:07.11 | Katty | ... |
| 16:07.18 | Katty | that's beside the point. |
| 16:07.20 | jero | hi |
| 16:07.24 | mishehu | anyway, you live in Misery ;-) |
| 16:07.26 | Katty | i'll blame whatever i want :P |
| 16:07.31 | Katty | i do :< |
| 16:07.59 | mishehu | Katty: if we put out a ransom demand, would we get millions for you? |
| 16:08.08 | Katty | mishehu: no. |
| 16:08.12 | Katty | ManxPower: get off. |
| 16:08.41 | mishehu | Katty: hmm... so if I kidnap you, what do I get out of it more than an increased grocery bill? |
| 16:08.43 | mishehu | heh |
| 16:08.52 | ManxPower | mishehu: hairballs? |
| 16:08.53 | Katty | ummm. |
| 16:08.56 | ai-a | how do i connect a full-duplex external voice/data v.90 data modem to asterisk? vgetty is working on the modem. |
| 16:09.03 | Katty | mishehu: i play piano and play well with windows most of the time |
| 16:09.17 | ManxPower | ai-a: you don't. |
| 16:09.25 | Katty | mishehu: not much (= |
| 16:10.02 | Katty | umm umm umm |
| 16:10.12 | Katty | k, well, i have nothing to offer then. |
| 16:10.14 | Pj | Is there a way to do p2p with IAX2 ? |
| 16:10.47 | mishehu | Katty: well, can't help ya then. ;-) |
| 16:10.52 | Katty | mishehu: mew :< |
| 16:11.11 | Pj | mishehu: what about pity ? :/ |
| 16:11.14 | Ariel_ | ahh my locat cafeteria is open and I just ordered some good cuban food...(They deliver). |
| 16:11.25 | Ariel_ | locat/local |
| 16:11.33 | ai-a | ManxPower: why not? |
| 16:11.34 | Ariel_ | damm I can't spell correctly today. |
| 16:11.34 | Pj | slocate cafeteria |
| 16:11.46 | *** join/#asterisk Romik (n=romik_@212.143.5.146) |
| 16:12.15 | ManxPower | ai-a: because voice modems are not supported, regardless of that the chan_modem stuff says. |
| 16:12.29 | ManxPower | Cordless Phone -> Sipura 2100 -> SIP -> Asterisk-1 -> IAX2 -> Asterisk-2 -> PRI -> Tellabs -> PRI -> Asterisk-2 -> IAX2 -> Asterisk-1 -> IAX2 -> Asterisk-3 -> SIP -> Polycom IP 500 |
| 16:12.34 | ManxPower | that the call I just made 8-) |
| 16:12.39 | *** join/#asterisk [2]JohnJacob (n=JohnJaco@pcp0011543387pcs.mainf01.in.comcast.net) |
| 16:12.41 | *** join/#asterisk mrtwister (n=mrtwiste@cable-1-32.cgates.lt) |
| 16:12.47 | ai-a | ManxPower: but i could write a c program that sends the modem down a iax channel. |
| 16:12.47 | Katty | lunch time. bye. |
| 16:13.01 | ManxPower | ai-a: you would need to write the driver first. |
| 16:13.01 | Mimmus | hi, can I ask why TxFax does return always '0' even if fax transmission went bad? |
| 16:13.11 | jero | ManxPower: how about latency? |
| 16:13.14 | *** join/#asterisk mhnoyes_ (n=mhnoyes@user-2ivfmui.dialup.mindspring.com) |
| 16:13.18 | Romik | manxPower: what voice quality of such mix? |
| 16:13.29 | ManxPower | jero: very little actually. voice quality was good. |
| 16:13.45 | shaon | bkw : the install process of woomega is not clear |
| 16:13.49 | slak- | how do i drop calls from a certain number |
| 16:13.59 | ManxPower | Granted everything except Asterisk-3 and the Polycom are on the same local lan |
| 16:14.19 | jero | oh ok |
| 16:14.45 | *** join/#asterisk hanchi_ (n=telliott@68-112-44-203.static.sprn.tx.charter.com) |
| 16:14.53 | shaon | help H323 setup |
| 16:15.10 | jero | help: topic not found |
| 16:15.19 | mishehu | shaon: Ivrit? |
| 16:15.45 | shaon | mishehu: what is ivrit? |
| 16:16.32 | Broom | When I pickup an analog phone connected to an fxs port, i get a ring back instead of a dial tone, any ideas? |
| 16:16.35 | shaon | please help to add H323 to asterisk |
| 16:16.57 | ManxPower | Broom: well set immediate=no ! |
| 16:17.08 | mishehu | shaon: ah, that answers my question. was just asking if you spoke that language, as your nick would mean "clock" in it. |
| 16:17.17 | ManxPower | shaon: You are not a native english speaker, are you? |
| 16:17.24 | hanchi_ | Anyone have any ideas on where and hot to integrate an ipcop firewall into an ISDN PRI connection with asterisk, I need to firewall the lan, and with the ISDN PRI connected directly to asterisk is sufficient security if some of the B channels are used for voice, and some for data? |
| 16:17.41 | Broom | it is on no |
| 16:17.44 | *** join/#asterisk ugenn (n=ugenn@cm188.omega71.maxonline.com.sg) |
| 16:18.11 | shaon | need h323 support plz |
| 16:18.18 | ManxPower | Broom: is immediate=no BEFORE or AFTER the channel => lines? |
| 16:18.21 | hugo-v6 | puh. re. |
| 16:18.27 | ManxPower | shaon: You are not a native english speaker, are you? |
| 16:18.34 | shaon | nop |
| 16:18.47 | Broom | Man: before |
| 16:18.52 | ManxPower | shaon: H323 is one of the hardest things to make work with Asterisk. |
| 16:18.56 | hugo-v6 | have forgotten my doorkeys so i had to burgle ionto. |
| 16:19.04 | ManxPower | That's why nobody runs it if they have any other choice. |
| 16:19.12 | ugenn | hi. i am a noob at IVR. where can i find information on setting up a simple automated answering system using a typical PC and phoneline? |
| 16:19.14 | Broom | nevermind, thanks for the help, typo.. :| |
| 16:19.35 | Broom | what does immediate do anyways? |
| 16:19.48 | shaon | ManxPower: please if you can, help me |
| 16:19.49 | *** join/#asterisk christo (n=chris@office.enovi.com) |
| 16:20.10 | ManxPower | shaon: I don't run H323, almost nobody runs H323. |
| 16:20.21 | ManxPower | The ones that do, don't want to take the entire day to help someone for free. |
| 16:20.58 | ManxPower | shaon: perhaps you could look at the docs included with the h323 channel driver, the Wiki, the mailing list archive, etc. |
| 16:21.17 | shaon | ManxPower: just i want to know, is there any built in module in asterisk? |
| 16:21.18 | *** join/#asterisk Lan16`spdy^gTp (n=spdy@altom.net1.nerim.net) |
| 16:23.04 | Broom | another question: how can I verify if i'm sharing IRQs in linux? |
| 16:23.20 | ManxPower | shaon: There are 4 channel drivers, the one included with Asterisk /path/to/asterisk/channels/h323, asterisk-oh323 (download from 3rd party), chan_oh323 (in the asterisk-addons CVS), and chan_woomera |
| 16:23.26 | christo | No channel type registered for 'IAX2' - is this something wrong with the way I have build *? |
| 16:23.28 | ManxPower | Broom: cat /proc/interrupts |
| 16:23.35 | ai-a | Broom: /proc/interrupts |
| 16:23.39 | Broom | thanks |
| 16:23.45 | ManxPower | christo: you might get that if you don't have an iax.conf |
| 16:23.56 | ManxPower | or if you noload'd it in /etc/asterisk/modules.conf |
| 16:24.21 | *** part/#asterisk jluk (n=jon@82-32-208-99.cable.ubr07.newt.blueyonder.co.uk) |
| 16:24.42 | *** join/#asterisk Cresl1n (n=Cresl1n@digium.com) |
| 16:24.43 | Broom | ok, if i have this: 11: 7220504 XT-PIC Intel 82801BA-ICH2, uhci_hcd:usb2, eth0, wctdm, wctdm, means I'm sharing it for wctdm right? |
| 16:24.43 | christo | ManxPower - neither is the case :S |
| 16:24.53 | *** join/#asterisk aVaLaNcHe (n=stecnic@m194-158-70-187.andorpac.ad) |
| 16:25.00 | ManxPower | Broom: yes. |
| 16:25.17 | ManxPower | Broom: your motherboard really sucks. |
| 16:25.21 | Broom | yeah |
| 16:26.38 | aVaLaNcHe | somebody can tell me the syntax for including the command SetTransferCapability('SPEECH') before Dial command ? |
| 16:26.55 | Broom | could that be the problem for which I can not make an outgoing call with the analog phone? it is trying to Attempting native bridge of Zap/8-1 and Zap/1-1 |
| 16:26.56 | cpatry | its already SPEECH by default. |
| 16:27.00 | Broom | but it doesn't go out |
| 16:27.06 | Broom | ? |
| 16:27.17 | shaon | ManxPower: thanks a lot. got lot of info from you. Great.A++++++++++ |
| 16:27.39 | Broom | or would sharing IRQ cause the crackling noise i hear? |
| 16:27.46 | dgilmore | can someone tell me if i can set asterisk so that when im on a call to certain extentions i can send a # and the call wont be transfered i need to send # in dtmf to call monitoring equipment |
| 16:28.02 | fulgas | on comedian mail , folder options, when you press # , * says a warning, that a sound file is missing... anyone knows what sound might be? |
| 16:28.09 | Broom | eliminate the t option in the dial command |
| 16:28.39 | *** join/#asterisk pa (n=Paolo@unaffiliated/pa) |
| 16:28.44 | aVaLaNcHe | cpatry: but when I try to make a call outside it says me , Requested transfer capability: 0x00 - SPEECH, and I get a hangup |
| 16:29.05 | Mimmus | does echocancel need on both incoming and outgoing channels? |
| 16:29.09 | aVaLaNcHe | but I receive calls from outside without any problem |
| 16:29.46 | cpatry | aVaLaNcHe: show me ur dial. |
| 16:30.01 | *** join/#asterisk SpaceBass (n=sp@c-24-125-184-203.hsd1.va.comcast.net) |
| 16:30.16 | *** join/#asterisk [1]JohnJacob (n=JohnJaco@pcp0011543387pcs.mainf01.in.comcast.net) |
| 16:30.36 | Beirdo | ~nickometer aVaLaNcHe |
| 16:30.41 | Beirdo | heh |
| 16:30.59 | Beirdo | how harsh |
| 16:31.21 | aVaLaNcHe | exten => _XXXXXX,1,Dial(Zap/g4/${EXTEN},60,tTr) |
| 16:31.26 | Uther_P | wierd |
| 16:31.32 | Uther_P | ~nickometer Uther_P |
| 16:31.39 | Uther_P | heh cool |
| 16:31.42 | cpatry | activate ur debug and see whats wrong. |
| 16:32.03 | mishehu | ~nickometer Beirdo |
| 16:32.08 | Beirdo | :) |
| 16:32.28 | mishehu | Beirdo: I wonder what one does to get a lame rating. |
| 16:32.30 | blessen | not able to receive calls using asterisk and kphone |
| 16:32.46 | *** part/#asterisk ugenn (n=ugenn@cm188.omega71.maxonline.com.sg) |
| 16:32.48 | Hmmhesays | what does the cli output tell you? |
| 16:32.57 | Pj | mishehu: mixed case, numbers like 666, 42, & co I'd guess |
| 16:32.58 | *** join/#asterisk myiagy (n=myiagy@200.138.215.78) |
| 16:33.05 | aVaLaNcHe | it says Executing Dial("SIP ....."Zap/g4/840064|60|tTr") in new stack |
| 16:33.16 | aVaLaNcHe | Requested transfer capabilyty.... |
| 16:33.17 | *** join/#asterisk The_Ball (n=alex@static-227.35.240.220.dsl.comindico.com.au) |
| 16:33.20 | Hmmhesays | blessen, what does the cli output tell you |
| 16:33.24 | aVaLaNcHe | Called g4/840064 |
| 16:33.26 | blessen | one sec |
| 16:33.36 | cpatry | aVaLaNcHe: thats not debug |
| 16:33.36 | aVaLaNcHe | Channel 0/1, span 4 got hangup |
| 16:33.39 | blessen | res_search: NO result ! |
| 16:33.39 | blessen | SipClient: Sending to '192.168.15.2:5060' |
| 16:33.39 | blessen | SipClient: Receiving message... |
| 16:33.54 | cpatry | u sure the remote end can pickup? |
| 16:33.59 | aVaLaNcHe | yes |
| 16:34.07 | Beirdo | mishehu: mixed case, numbers, _, etc |
| 16:34.11 | aVaLaNcHe | I am sure |
| 16:34.12 | cpatry | i assume u control remote end, right? |
| 16:34.15 | *** join/#asterisk Goshen (n=Goshen@c-67-172-238-57.hsd1.ut.comcast.net) |
| 16:34.26 | aVaLaNcHe | I thing so |
| 16:34.33 | blessen | this is what i receive.. |
| 16:34.35 | blessen | -------------------------------- |
| 16:34.36 | blessen | REGISTER sip:192.168.15.2 SIP/2.0 |
| 16:34.36 | blessen | Via: SIP/2.0/UDP 192.168.15.2:5062;branch=z9hG4bK375CD1A0 |
| 16:34.36 | blessen | CSeq: 9262 REGISTER |
| 16:34.36 | blessen | To: "Blessen Cherian" <sip:kphone@192.168.15.2> |
| 16:34.36 | blessen | Expires: 900 |
| 16:34.38 | blessen | From: "Blessen Cherian" <sip:kphone@192.168.15.2> |
| 16:34.42 | blessen | Call-ID: 1865953309@192.168.15.2 |
| 16:34.42 | cpatry | blessen: use pastebin.ca |
| 16:34.44 | blessen | Content-Length: 0 |
| 16:34.46 | blessen | User-Agent: kphone/4.2 |
| 16:34.48 | blessen | Event: registration |
| 16:34.50 | blessen | Allow-Events: presence |
| 16:34.51 | cpatry | think? |
| 16:34.52 | sivana | nice |
| 16:34.52 | blessen | Contact: "Blessen Cherian" <sip:kphone@192.168.15.2:5062;transport=udp>;methods="INVITE, MESSAGE, INFO, SUBSCRIBE, OPTIONS, BYE, CANCEL, NOTIFY, ACK, REFER" |
| 16:34.55 | blessen | res_search: NO result ! |
| 16:34.57 | blessen | res_search: NO result ! |
| 16:34.59 | blessen | SipClient: Sending to '192.168.15.2:5060' |
| 16:35.00 | Uther_P | I'd stop that |
| 16:35.01 | blessen | SipClient: Receiving message... |
| 16:35.03 | blessen | SipClient: Received: 09:27:10.618 |
| 16:35.20 | sivana | ~pastebin |
| 16:35.22 | jbot | rumour has it, pastebin is a place to paste your stuff without flooding the channel - try http://pastebin.ca/ |
| 16:35.30 | blessen | ok..thanks |
| 16:36.19 | aVaLaNcHe | cpatry: yes I'm in the remote end |
| 16:36.59 | *** join/#asterisk santiago (n=santiago@63.245.86.163) |
| 16:37.00 | blessen | here is the details..http://pastebin.ca/21127 |
| 16:37.16 | *** join/#asterisk shido6 (n=shido6@d57-87-253.home.cgocable.net) |
| 16:38.51 | mishehu | ~nickometer mishehu |
| 16:39.08 | Wonka | ~nickometer jbot |
| 16:39.20 | Wonka | ~nickometer L33tKidd |
| 16:39.37 | Wonka | ok, not everything scores 0% |
| 16:39.38 | Beirdo | ~nickometer _IsUcK69! |
| 16:40.03 | *** join/#asterisk qbertibur (n=kvirc@p54879040.dip0.t-ipconnect.de) |
| 16:40.09 | [Airwolf] | ~nickometer [Airwolf] |
| 16:40.09 | qbertibur | hi |
| 16:40.26 | [Airwolf] | ~nickometer Airwolf]_ |
| 16:40.34 | *** join/#asterisk The_Ball (n=alex@static-227.35.240.220.dsl.comindico.com.au) |
| 16:40.42 | [Airwolf] | ~nickometer Airwolf_ |
| 16:40.56 | [Airwolf] | hehe |
| 16:41.02 | ManxPower | If you are going to play with your bot, it's polite to do it in private, and wash your hands after. |
| 16:41.12 | Beirdo | and clean the channel up too |
| 16:41.15 | Beirdo | heh |
| 16:41.18 | Beirdo | anyways |
| 16:41.21 | [Airwolf] | ManxPower, it's not my bot |
| 16:41.24 | bendy24_ | sick people |
| 16:41.46 | ManxPower | If you are going to play with someone else's bot, it's polite to do it in private, and wash your hands after. |
| 16:42.14 | [Airwolf] | ManxPower, I think you montitor is reflecting somehow |
| 16:42.18 | bendy24_ | Beirdo: why does it always have to degrade like this? |
| 16:42.39 | Beirdo | because we are geeks? |
| 16:42.47 | bendy24_ | naw |
| 16:42.50 | [Airwolf] | Beirdo, good answer |
| 16:43.01 | [Airwolf] | bendy24_, so your not a geek ? |
| 16:43.14 | bendy24_ | im a closet geek |
| 16:43.25 | qbertibur | Is anybody here to talk about asterisk? I've got some questions... |
| 16:43.39 | [Airwolf] | qbertibur, asterisk what is that ? |
| 16:43.41 | spackle | ask away |
| 16:43.47 | qbertibur | *g* |
| 16:43.49 | Beirdo | qbertibur: I'm sure if you ask, people will get back on topic |
| 16:43.55 | JerJer | nope this chat room is only about the disussion of the ascii charector * |
| 16:44.11 | bendy24_ | ** * |
| 16:44.17 | [Airwolf] | *** |
| 16:44.22 | JerJer | ah ascii 42 how I love thee |
| 16:44.22 | bendy24_ | *! |
| 16:44.40 | Beirdo | just keep * away from rm |
| 16:44.41 | qbertibur | I set up a box with asterisk, but I can't connect. |
| 16:44.43 | *** join/#asterisk generalhan (i=general_@63.133.146.82) |
| 16:44.45 | ManxPower | JerJer: Count the ways! Count the ways! |
| 16:44.50 | JerJer | let me count the ways |
| 16:45.04 | *** join/#asterisk amir (n=amir@shield.guindehi.ch) |
| 16:45.04 | *** join/#asterisk CaNaBiS (n=CaN@lvrgtnbas02-pool9-a7.lvrgtn.tds.net) |
| 16:45.07 | generalhan | hey guys |
| 16:45.13 | [Airwolf] | qbertibur, what kind of client ? |
| 16:45.16 | [Airwolf] | sip, iax ? |
| 16:45.22 | JerJer | there are no guys here only marklar |
| 16:45.23 | qbertibur | kphone |
| 16:45.26 | qbertibur | sip |
| 16:45.33 | bendy24_ | marklar? |
| 16:45.39 | [Airwolf] | qbertibur, ok what kind of error message do you get ? |
| 16:46.04 | *** join/#asterisk epoch (i=epoch@octane.breakbeats.org) |
| 16:46.05 | JerJer | i am about to go to the marklar to get some more marklar, anyone want any marklar ? |
| 16:46.08 | shido6 | dont use the p4p800-vm for mac os x intel - tho it has the intel integrated extreme graphics - and you do get quartz express enabled you loose your mouse every few minutes and have to plug the mouse back in... |
| 16:46.39 | qbertibur | It give me a lot of debug output. It shows me the received packeges and says unauthorized. I'm pretty sure I set the correct pass, so I'm confused... |
| 16:46.40 | generalhan | i just bought an Aastra SIP phone and i need to know how to get to the VMBox ... is there some kind of code i can put in the extensions that will allow them to get directly to their VM ? |
| 16:46.47 | [Airwolf] | shido6, just by a mac if you want to use mac os x |
| 16:46.50 | shido6 | its not THAT much of an improvement anyway.. yeah u do get 3d stuff - but I dont use 3d stuff that much |
| 16:46.54 | carrar | <PROTECTED> |
| 16:46.55 | *** part/#asterisk ManxPower (n=eric@dsl-209-16-67-160.datasync.com) |
| 16:47.10 | [Airwolf] | qbertibur, could you paste it private to me ? |
| 16:47.13 | bendy24_ | wtf is marklar? |
| 16:47.18 | shido6 | if I buy a mac now and they switch to intel my box will be phased out and that would make a lot of sense, wouldnt it. |
| 16:47.23 | LoRez | can an IAXY be used as an overhead paging access device? |
| 16:47.33 | bendy24_ | shido6: why not? |
| 16:47.35 | generalhan | Jer Jer: any suggestions? |
| 16:47.35 | shido6 | IAXy's dont auto answer |
| 16:47.44 | JerJer | doesn't ANYONE watch southpark ? |
| 16:47.48 | bendy24_ | shido6: you'd still be able to run osx applications |
| 16:47.51 | carrar | heh |
| 16:48.00 | [Airwolf] | shido6, ppc will be phased out in 2009, your system will be old by then |
| 16:48.01 | carrar | it's back to back SP this week! |
| 16:48.02 | shido6 | but who will still write them |
| 16:48.08 | shido6 | there are PowerPC architecture apps |
| 16:48.15 | shido6 | and there are PowerPC,Intel architecture apps |
| 16:48.24 | bendy24_ | shido6: see Airwolf's response |
| 16:48.33 | slak- | buy my ss10 dial 50mhz |
| 16:48.33 | bendy24_ | your logic is flawed im afraid |
| 16:48.36 | slak- | and run asterisk! |
| 16:48.38 | Beirdo | JerJer: sure do. I'm feeling a bit like Tweek these days |
| 16:48.38 | shido6 | hopefully they'll still write PowerPC compatible apps after the switch |
| 16:48.41 | shido6 | but for how long |
| 16:48.41 | slak- | dual |
| 16:48.53 | bendy24_ | shido6: see Airwolf's response |
| 16:48.56 | [Airwolf] | shido6, one word: universal |
| 16:49.02 | shido6 | unversal binaries |
| 16:49.02 | [Airwolf] | binary's |
| 16:49.04 | shadebob | Is it possible to execute a macro or an app once communication began? |
| 16:49.06 | JerJer | TIMMY |
| 16:49.11 | Beirdo | hehe |
| 16:49.12 | shido6 | yeah thats true - but again, for how long will they write them like that |
| 16:49.18 | carrar | TIMMAH!! |
| 16:49.22 | Beirdo | Cartmen is my hero though |
| 16:49.23 | [Airwolf] | shido6, as long as ppc is supported |
| 16:49.26 | bendy24_ | TIMMMA |
| 16:49.31 | [Airwolf] | untill 2009 |
| 16:49.35 | Beirdo | TIMMAY! |
| 16:49.39 | bendy24_ | Beirdo==Cartman |
| 16:49.39 | shido6 | with the commodore - you couldnt run amiga software on the c64 |
| 16:49.46 | *** part/#asterisk hanchi_ (n=telliott@68-112-44-203.static.sprn.tx.charter.com) |
| 16:49.49 | Beirdo | I'm not THAT fat |
| 16:49.50 | bendy24_ | hes not fat, hes big boned |
| 16:49.51 | shido6 | with the IIGS you couldnt run IIGS on IIE's |
| 16:49.54 | Uther_P | I keek yoo squah in tha nutz |
| 16:50.05 | bendy24_ | IE?! |
| 16:50.08 | bendy24_ | omg run! |
| 16:50.08 | shido6 | LC II on IIGS ( to a point) |
| 16:50.10 | Beirdo | respect ma authoritah! |
| 16:50.28 | Cresl1n | hya! |
| 16:50.36 | Uther_P | thats not how you uphold the law! you gotta hit 'em upside the head, they go down faster |
| 16:50.40 | Beirdo | heheh |
| 16:50.46 | shido6 | I freaked out when I could run PowerPC only apps on osx intel |
| 16:50.51 | LoRez | shido6: could though if digium released a firmware upgrade to do that though, no? |
| 16:50.55 | shido6 | transitive works |
| 16:50.59 | *** join/#asterisk nextime (n=nextime@213-140-6-96.ip.fastwebnet.it) |
| 16:51.00 | Uther_P | and I'd also like to add that reading totaly sucks as |
| 16:51.01 | Uther_P | s |
| 16:51.20 | Beirdo | ooh, that reminds me... pack book for flight. |
| 16:51.45 | bendy24_ | silly Beirdo |
| 16:52.00 | Beirdo | not like I don't have enough books |
| 16:52.25 | bendy24_ | are you going to update us via irc during your trip? |
| 16:52.30 | Beirdo | heh |
| 16:52.33 | Beirdo | not bloody likely |
| 16:52.37 | bendy24_ | you better |
| 16:52.45 | bendy24_ | we'll cheer you on |
| 16:53.14 | Beirdo | and that IAXy better not misbehave down there |
| 16:53.28 | Beirdo | or I'll kick it square in the nuts |
| 16:53.35 | bendy24_ | ! |
| 16:53.37 | generalhan | can some one please help me out with a voicemail question !!??? |
| 16:53.52 | Nugget | only if you ASK LOUDER. |
| 16:54.01 | bendy24_ | TRY YELLING |
| 16:54.22 | bendy24_ | the ops love it when you do that |
| 16:54.25 | generalhan | how can i have my users get to their VM boxes with the push of a button ? is there something i can add to the dial plan or something ? |
| 16:54.46 | bendy24_ | extension? |
| 16:55.07 | SpaceBass | generalhan depends on phone if it has a message button, but if you add the mailbox=<mailbox> to their extension then dialing *98 should do it |
| 16:55.07 | Nugget | doesn't the sample dialplan already have exactly that function in it already? |
| 16:55.28 | Cresl1n | yeah |
| 16:55.30 | bendy24_ | 8500 |
| 16:55.44 | Cresl1n | just make an extension that passes their extension to the voicemail app |
| 16:55.50 | Cresl1n | that way you don't have to authenticate |
| 16:56.03 | Nugget | and make sure that nobody but your internal phones can access that extension |
| 16:56.15 | SpaceBass | on google's blog they appear to be solciting help / partners for SIP with gtalk... someone at digium should suggest a partnership... or at least the developer community |
| 16:56.32 | *** join/#asterisk Goshen (n=Goshen@c-67-172-238-57.hsd1.ut.comcast.net) |
| 16:56.36 | generalhan | the mailboxes work, but i dont want to have to dial my number then listen to the message then press star and enter a password. i have a programmable key open and i just want it to automatically go to the "enter your password" part so i can get my messages faster |
| 16:56.46 | *** join/#asterisk SplasPood (i=jwb@ludicrous.paravolve.net) |
| 16:56.49 | Nugget | generalhan: then do what was just suggested. |
| 16:57.22 | generalhan | Nugget: im looking at the sample dialplan right now and i dont see what youre talking about |
| 16:57.39 | *** join/#asterisk Seggy (i=rbutler@tsss.org) |
| 16:58.03 | Uther_P | generalhan: you can also do VoicemailMain(102), for example, to goto mailbox 102.. it just asks for pw |
| 16:58.41 | slak- | everyone is too lazy to change their vw password |
| 16:58.48 | *** join/#asterisk Ayano (n=erik_lee@adsl-70-245-190-90.dsl.spfdmo.swbell.net) |
| 16:59.14 | Nugget | consult the documentation on the differences between calling Voicemail() and VoicemailMain() |
| 16:59.20 | *** join/#asterisk Nix (n=Nix@81.213.125.220) |
| 16:59.40 | spackle | exit |
| 17:00.15 | spackle | generalhan, you want this: xten => *99,2,VoiceMailMain(s${CALLERIDNUM}) |
| 17:00.23 | *** join/#asterisk websae (n=websae@207-118-143-134.dyn.centurytel.net) |
| 17:00.40 | websae | anyone here running FreeBSD with their asterisk ? |
| 17:00.48 | Nugget | websae: just the same people you asked last night. :) |
| 17:01.08 | websae | hahaha |
| 17:01.14 | websae | hey need some help Nugget |
| 17:01.37 | websae | was compiling this thing for hours...and then got this |
| 17:01.37 | websae | ===> Verifying install for /usr/local/include/zaptel.h in /usr/ports/misc/zaptel |
| 17:01.38 | websae | >> The zaptel port needs FreeBSD kernel source code to compile. |
| 17:01.38 | websae | >> Please install FreeBSD kernel source code in /usr/src/sys. |
| 17:01.38 | websae | *** Error code 1 |
| 17:01.38 | websae | Stop in /usr/ports/misc/zaptel. |
| 17:01.40 | websae | *** Error code 1 |
| 17:01.42 | websae | Stop in /usr/ports/net/asterisk. |
| 17:02.00 | Nugget | please don't paste in here. |
| 17:02.00 | cpatry | ~pastebin |
| 17:02.03 | jbot | methinks pastebin is a place to paste your stuff without flooding the channel - try http://pastebin.ca/ |
| 17:02.27 | Nugget | my advice is to skip zaptel completely. it will never work right for you and it'll cause all sorts of annoying problems. |
| 17:02.45 | Nugget | especially since you're running a release version of freebsd |
| 17:02.51 | websae | how do i skip that? |
| 17:03.00 | Nugget | don't install it. |
| 17:03.10 | Romik | mpg123: is take 99% of CPU, any body can advice any method to put music on hold? |
| 17:03.15 | websae | so i have to do a makeinstall clean now? |
| 17:03.39 | Nugget | a what? |
| 17:03.50 | websae | i was making it from the port |
| 17:03.55 | ast_freak | How can I monitor my zaptel cards? How do I build ztmonitor? |
| 17:03.58 | websae | so do make install clean? |
| 17:04.09 | *** join/#asterisk klasstek (n=nunyobiz@sta-206-168-218-206.rockynet.com) |
| 17:04.19 | Nugget | no, that would be "installing it" |
| 17:04.29 | Nugget | don't worry about zaptel at all. skip it. it's nothing but trouble. |
| 17:04.33 | Nugget | just install asterisk |
| 17:04.38 | websae | how do i do that though |
| 17:04.38 | *** part/#asterisk epoch (i=epoch@octane.breakbeats.org) |
| 17:04.41 | websae | i installed from ports |
| 17:04.47 | websae | and how do i skip it |
| 17:04.49 | Uther_P | Nugget, I think in BSD it trys to install zaptel as a dependancy |
| 17:04.52 | websae | i just did the make command |
| 17:04.56 | websae | yes |
| 17:04.59 | websae | it does |
| 17:05.00 | Nugget | skip the port, then. |
| 17:05.03 | Nugget | it's old anyway |
| 17:05.31 | *** join/#asterisk danny33923 (n=no@66.236.175.18.ptr.us.xo.net) |
| 17:05.35 | websae | hrm there are a lot of dependencies though |
| 17:05.53 | spackle | Romik: use Madplay, there is info on the wiki. |
| 17:06.01 | Nugget | It builds fine without the port. |
| 17:06.05 | danny33923 | hey folks. having trouble setting up a simple if statement... |
| 17:06.27 | danny33923 | exten => 2,1,Macro(rg-group,15,REC,214); |
| 17:06.27 | danny33923 | exten => 2,2,Goto(aa_4,s,1); |
| 17:06.54 | *** join/#asterisk file[laptop] (n=jcolp@mctnnbsa30w-156034056020.nb.aliant.net) |
| 17:07.15 | Katty | is there a wiki how to thingy for to email someone a notification when they get a new voicemail? |
| 17:07.17 | danny33923 | before the 2,2, I want to check if caller id = "xxx-xxx-xxxx" and if so, then Goto something else |
| 17:07.35 | spackle | Katty: yes. |
| 17:07.39 | drray | voicemail.conf will send them an email if it is configured |
| 17:07.46 | Katty | i know it will |
| 17:07.47 | Katty | i want the wiki |
| 17:07.57 | Katty | Hmmhesays: where is the wiki for making voicemail send out an email notification? |
| 17:08.15 | *** part/#asterisk chris78 (n=schulte@gate.sympat.de) |
| 17:08.19 | bendy24_ | tiki wiki |
| 17:08.24 | *** join/#asterisk [1]JohnJacob (n=JohnJaco@pcp0011543387pcs.mainf01.in.comcast.net) |
| 17:09.22 | Katty | anthm: where is teh wiki page on for how to send out an email to $person when they get a new voicemail? |
| 17:09.52 | [Airwolf] | Katty, you know there is a search option on the wiki ? |
| 17:10.07 | danny33923 | anyone know how to do this?? i basically want to make all phones ring if the boss calls |
| 17:10.11 | anthm | not sure if there is a wiki page but I know how since I added the functionality to it that makes it easier |
| 17:10.14 | Katty | [Airwolf]: obviously. |
| 17:10.20 | Katty | anthm: post |
| 17:10.21 | [Airwolf] | Katty, then use it ? |
| 17:10.32 | Katty | [Airwolf]: when i can get anthm to answer me? never. |
| 17:10.54 | Katty | [Airwolf]: besides, anthm will post something better for me. |
| 17:10.58 | blessen | i am getting this |
| 17:11.00 | [Airwolf] | O my god, you're lazy |
| 17:11.01 | blessen | -- Got SIP response 408 "Request Timeout" back from 216.115.25.198 |
| 17:11.06 | Katty | [Airwolf]: obviously |
| 17:11.17 | blessen | what might be the reason for this |
| 17:11.21 | Beirdo | Katty gets away with being lazy :) |
| 17:11.31 | Katty | Beirdo: yes. |
| 17:11.32 | drray | function of having breasts |
| 17:11.43 | bendy24_ | :O |
| 17:11.47 | Katty | Beirdo: but i don't act stupid and not listen to my replies. |
| 17:11.52 | bendy24_ | i want breasts now |
| 17:12.11 | Beirdo | Katty: yeah, you pay attention too. :) heh |
| 17:12.20 | anthm | oh wait I was thinking recording the call and email it |
| 17:12.25 | anthm | voicemail is easy |
| 17:12.28 | hugo-v6 | guys. dont use cheap ram... even its only a devel/play-box |
| 17:12.30 | Katty | anthm: well.... |
| 17:12.39 | Katty | anthm: i don't want to do that cause misterbossman never deletes old emails |
| 17:12.42 | Katty | anthm: ever. |
| 17:12.46 | Katty | anthm: and then server would go boom. |
| 17:12.49 | hugo-v6 | that was the 2nd and last time. damn shit. |
| 17:12.53 | anthm | if they have an email in the config file and email mode is on |
| 17:13.06 | *** join/#asterisk bweschke (n=bweschke@pcp09754274pcs.narlington.nj.comcast.net) |
| 17:13.12 | Katty | anthm: i have teh email in voicemail already. |
| 17:13.25 | Katty | anthm: where do i turn email mode on at? in voicemail.conf? |
| 17:13.36 | anthm | yah |
| 17:13.46 | anthm | [general] |
| 17:14.02 | anthm | format=wav49 or whatever format |
| 17:14.12 | anthm | serveremail=sender@mydomain.com |
| 17:14.18 | anthm | attach=yes |
| 17:14.40 | *** join/#asterisk mcn (n=mcn@ext-gw.newtoncomputing.co.uk) |
| 17:14.45 | anthm | then each user with an email addr defined gets one |
| 17:14.46 | ast_freak | Where can I get newt? |
| 17:15.08 | *** join/#asterisk insomni_ (n=insomni@x1-6-00-e0-18-6f-34-ad.k455.webspeed.dk) |
| 17:15.10 | CoaxD | ast: Need eye of newt? I'll slap it in the mail via fedex. |
| 17:15.17 | Beirdo | ast_freak: try a pet store? |
| 17:15.17 | Katty | anthm: does asterisk have a built in email server? ^_- |
| 17:15.34 | CoaxD | ast: (For the record, 'eye of newt' is an HERB, not an ANIMAL.) |
| 17:15.39 | Beirdo | hehe |
| 17:15.44 | anthm | it uses sendmail binary on the system |
| 17:15.44 | Beirdo | poor newt |
| 17:15.44 | ast_freak | No, for zttool and astman |
| 17:15.53 | Katty | anthm: kk, thanky. |
| 17:16.19 | CoaxD | hehehe. oh, you mean libnewt |
| 17:16.44 | Tili | how can i make outgoing call in outgoing folder to wait a few seconds before trying to call |
| 17:16.51 | ast_freak | Yes. help me out |
| 17:17.16 | CoaxD | ast_freak: Go download it |
| 17:17.18 | Tili | i am trying a callback script but as soon i write file to outgoing folder and call hangup on current channel. it calls. so sometimes i haven't even hungup and it calls back |
| 17:17.23 | mishehu | bah. |
| 17:17.52 | ast_freak | CoaxD: Yeah, trying to find a tarball for it |
| 17:18.17 | CoaxD | ast_freak: Google |
| 17:18.27 | CoaxD | ast_freak: http://www.freshmeat.net |
| 17:18.42 | Katty | anthm: i don't need to configure sendmail, do i? |
| 17:18.58 | anthm | just as an agent |
| 17:19.05 | Katty | k |
| 17:19.14 | Katty | hopefully that's already taken care of. |
| 17:19.16 | CoaxD | ast_freak: But you need the library compiled for your system. not just any random tarball |
| 17:19.19 | *** join/#asterisk jeffgus (n=jeffgus@2002:d856:c704:0:0:0:0:1) |
| 17:19.25 | anthm | you can cheat and make a fake /usr/sbin/sendmail too |
| 17:19.29 | CoaxD | ast_freak: So, you need to find it for your system, or compile it yourself |
| 17:19.42 | anthm | that mimics the behaviour of outgoing dilevery agent |
| 17:19.55 | anthm | cat email | sendmail -t |
| 17:19.57 | anthm | etc |
| 17:20.06 | mishehu | sendmail schendmail |
| 17:20.20 | bkw_ | exim baby.. thats where its at! |
| 17:20.28 | bendy24_ | no no |
| 17:20.31 | danny33923 | does anyone know how to make another LookupBlacklist??? for example... exten,s,1,LookupBOSSlist |
| 17:20.37 | mutilator | anyone recommend any type of dslam that does adsl/sdsl? |
| 17:20.44 | drray | IES-1000 |
| 17:20.49 | drray | zyxel |
| 17:20.52 | puowvip | freenode has an ipv6 server? |
| 17:20.55 | drray | we just deplyed it |
| 17:20.57 | ast_freak | CoaxD: Yeah, found gnewt, but will that work? |
| 17:20.58 | danny33923 | i can recomend what not to get |
| 17:20.59 | drray | er, deployed |
| 17:21.09 | danny33923 | net2net dslams SUCK! |
| 17:21.22 | drray | mutilator - how many ports do you need? |
| 17:21.31 | anthm | need not start mail agent flame war sendmail is hardcoded into voicemail that is the point |
| 17:21.34 | anthm | #define SENDMAIL "/usr/sbin/sendmail -t" |
| 17:21.39 | CoaxD | ast_freak: Dude. learn linux |
| 17:21.41 | puowvip | irc.ipv6.freenode.net |
| 17:21.46 | puowvip | ok |
| 17:21.47 | CoaxD | ast_freak: *then* learn *. |
| 17:22.01 | Beirdo | anthm: you can always create a script called /usr/sbin/sendmail if you are stuck :) |
| 17:22.05 | anthm | so hack the code or make sure /usr/sbin/sendmail is something that reads an email on stdion and delivers it |
| 17:23.08 | mutilator | drumkilla: well the one we have right now is 24 per card and takes 7 cards |
| 17:23.09 | mutilator | but it sucks |
| 17:23.23 | mutilator | copperedge ce200 sdsl/idsl dslam |
| 17:23.38 | ast_freak | CoaxD: Look, I'm no rookie, I just can't find the source tarball for libnewt. Care to actually help me fine it, or do you just want to spout off? |
| 17:23.46 | *** join/#asterisk websae (n=websae@207-118-143-134.dyn.centurytel.net) |
| 17:23.52 | mutilator | drray i mean* |
| 17:23.52 | ast_freak | s/fine/find/ |
| 17:23.59 | drray | :) |
| 17:24.05 | danny33923 | is this #asterisk or some clec room.? |
| 17:24.07 | mutilator | looking for 40 or so ports atleast |
| 17:24.09 | mutilator | per dslam |
| 17:24.17 | mutilator | preferably more |
| 17:24.25 | drray | zyxel makes a abigger IES chasis |
| 17:24.28 | *** join/#asterisk puowvip (i=ircuser@thegrey.diamond.org) |
| 17:24.43 | mutilator | that does adsl? |
| 17:24.47 | drray | we love the ies-1000, it's replacing out paradyne |
| 17:25.05 | drray | I believe it will do a/sh and vdsl |
| 17:25.14 | Katty | anthm: hmm, looks like it didn't work. let me post my voicemail.conf to make sure i did screw it up somehow. |
| 17:25.19 | drray | vdsl is some hybrid thing |
| 17:25.35 | mutilator | wonder what distances ya can get from em |
| 17:25.57 | blessen | i get error res_search: NO result ! |
| 17:26.00 | mishehu | Katty!!! |
| 17:26.04 | blessen | any one any ide about this |
| 17:26.17 | drray | we use the ies-1000 with adsl port |
| 17:26.29 | drray | 16 ports per 1u chasis |
| 17:26.30 | Katty | mishehu: mew? |
| 17:26.36 | *** join/#asterisk ManxPower (n=eric@dsl-209-16-67-160.datasync.com) |
| 17:26.43 | bkw_ | qmail is NOT the answer |
| 17:26.45 | bkw_ | haha |
| 17:26.51 | blessen | can anyone help me with geting asterisk to connect with kphone to pick calls |
| 17:26.59 | *** part/#asterisk ManxPower (n=eric@dsl-209-16-67-160.datasync.com) |
| 17:27.16 | Beirdo | qmail is NEVER the answer |
| 17:27.23 | danny33923 | kphone??? I hear xlite now runs on linux |
| 17:27.42 | *** join/#asterisk _ioscanner (n=ioscanne@c-67-166-249-43.hsd1.tx.comcast.net) |
| 17:27.43 | blessen | yes kphone is a soft phone which runs on linux |
| 17:27.45 | mutilator | something wrong with qmail? |
| 17:27.49 | Beirdo | yup |
| 17:28.02 | Nix | qmail sucks |
| 17:28.05 | *** join/#asterisk rikstahh (n=rick@84.93.141.188.plusnet.pte-ag1.dyn.plus.net) |
| 17:28.05 | Nix | use postfix ;-) |
| 17:28.28 | _ioscanner | I see an error ast_yyerror(): syntax error: syntax error: Input: = ANSWER? |
| 17:28.28 | blessen | but i am not able to receive calls using my kphone and asterisk server |
| 17:28.30 | bjohnson | postfix here |
| 17:28.33 | mutilator | that ies do dhcp on client side drray? |
| 17:28.36 | _ioscanner | Where would I find this error |
| 17:28.43 | mutilator | or is it all routed |
| 17:29.00 | drray | mutilator, I don't know how to answer that, we run a DHCP server that feeds it |
| 17:30.31 | mutilator | so you're vlan |
| 17:30.46 | drray | on each port |
| 17:30.51 | *** join/#asterisk sandra (n=awefawe@200.60.26.44) |
| 17:31.02 | sandra | hi |
| 17:31.10 | drray | It's 2k per chasis, 16 ports, plus $100 each for the modem |
| 17:31.12 | drray | we love it |
| 17:31.15 | sandra | can anyone helme to configure my aah? |
| 17:31.17 | sandra | :( |
| 17:31.21 | FarrisG | I got my Polycom 501 working, but dtmf won't work while in a call. I hear the tones, but they don't register on the other end (voicemail systems, etc.) |
| 17:31.28 | _ioscanner | Any ideas? |
| 17:31.48 | SpaceBass | FarrisG have you tried changing the settings (inband, etc) |
| 17:31.57 | mutilator | see with this crapp coppermountain i can't vlan unless i run a hssi/ds3 wan port into the box |
| 17:32.05 | mutilator | can't do vlan over the ethernet port |
| 17:32.09 | drray | eww |
| 17:32.47 | sandra | can anyone helme to configure my aah? |
| 17:32.49 | mutilator | so right now the dslam is a big router, static routed 248 subnets to each port |
| 17:33.06 | drray | yeah, I can see why you hate that |
| 17:33.30 | FarrisG | SpaceBass: Can't seem to find dtmf settings |
| 17:34.12 | Equinox | You have dtmf_mode=inband in sip.conf? |
| 17:34.49 | blessen | i cannot register a user through kphone on a asterisk server |
| 17:35.06 | Equinox | FarrisG- You have dtmf_mode=inband ? |
| 17:35.50 | _ioscanner | I am getting a blank IVR when I call in to my SIP (broadvoice) line |
| 17:36.10 | SpaceBass | FarrisG its in your extension context for each extension |
| 17:36.27 | *** join/#asterisk MasterYoda (n=mnichols@pdpc/supporter/sustaining/MasterYoda) |
| 17:36.43 | _ioscanner | What would cause this. It would work for a few mintues then not. I see something about saying the format changed. Anyone seen this |
| 17:36.55 | MasterYoda | does the UserEvent application work properly for sending events with more than one header in the body? |
| 17:37.05 | _ioscanner | I can call in internal and it works |
| 17:37.18 | _ioscanner | just sip broadvoice |
| 17:37.18 | slak- | hey anyone got an example of how to block a call based on callerid |
| 17:37.20 | slak- | crazy ex girlfriend |
| 17:37.29 | MasterYoda | it seems to me it would be better as UserEvent(eventname|header1: sweet|header2: even sweeter) |
| 17:37.32 | slak- | keeps calling my cell, killing the battery, and proceeds to my job |
| 17:37.42 | CoaxD | slak: 8001112222/1112223333,1, |
| 17:37.53 | MasterYoda | rather than UserEvent(eventname|header1: sweet\nheader2: sweetless... |
| 17:38.17 | slak- | CoaxD huh? |
| 17:38.22 | CoaxD | slak: c'mon dude |
| 17:38.39 | CoaxD | slak: You just add to your extension's did - '/<num_you_want_blocked' |
| 17:38.52 | *** join/#asterisk rkvarala (n=rkvarala@59.93.52.96) |
| 17:38.55 | slak- | i want it for all extensions :P |
| 17:39.04 | slak- | id hate for this little cunt to call my boss |
| 17:39.12 | CoaxD | exten => 8003942193/4073907227,1,Congestion() |
| 17:39.14 | sandra | can anyone helme to configure my aah? |
| 17:39.16 | rkvarala | is there any provision to run the shell scripts from asteirsk |
| 17:39.23 | bkw_ | agi |
| 17:39.30 | slak- | the 800 is my number right? |
| 17:39.33 | CoaxD | slak: Yea |
| 17:39.37 | *** part/#asterisk Equinox (n=secret@star.l93.com) |
| 17:39.43 | CoaxD | slak: And you dont need it for all extensions. just those that shemight call YOU on |
| 17:39.46 | *** join/#asterisk apeman (n=cmurray@zoidberg.digital-storm.net) |
| 17:40.24 | slak- | exten => */4073907227,1,Congestion() will that work for any number that she might dial? |
| 17:40.41 | CoaxD | no |
| 17:41.00 | CoaxD | for every number you have defined (incoming DIDs) you need to add that |
| 17:41.00 | *** join/#asterisk meppl (n=mephisto@p54AAEF6B.dip.t-dialin.net) |
| 17:41.02 | CoaxD | this is not tough |
| 17:41.04 | CoaxD | just do it |
| 17:41.56 | CoaxD | best part is, if you do that and she's calling in on an iax2-voip did, you wont even get charged for the call. |
| 17:42.06 | bkw_ | zoa wake your ass up |
| 17:42.17 | CoaxD | i.e. if its 800 |
| 17:42.34 | *** join/#asterisk NSGN (n=NSGN@cpe-66-69-197-25.austin.res.rr.com) |
| 17:42.35 | *** join/#asterisk roche (n=roche@obiwan.inalambrica.net) |
| 17:42.43 | NSGN | i'm back! remember me? |
| 17:42.45 | *** join/#asterisk Equinox (n=secret@27-74.202-68.tampabay.res.rr.com) |
| 17:42.46 | NSGN | 0_o |
| 17:42.48 | NSGN | hello |
| 17:43.17 | CoaxD | slak: so, now, she'll only be able to hit your cel |
| 17:43.37 | NSGN | i have a quick question. how much of an asterisk server could i get away with running on a 233mhz amd :-D |
| 17:43.55 | slak- | how much ram? |
| 17:44.00 | SpaceBass | NSGN that |
| 17:44.06 | CoaxD | nsgn: That thing is slow as piss |
| 17:44.11 | SpaceBass | NSGN thats what I use with 512mb and it works fine for home use |
| 17:44.17 | CoaxD | nsgn: 50% of the processor would be used - just playing an mp3 |
| 17:44.20 | NSGN | i could prolly get 380 in it |
| 17:44.26 | SpaceBass | can do at least 3 outbound calls at once |
| 17:44.31 | slak- | will do for home |
| 17:44.34 | SpaceBass | i haven't tested more |
| 17:44.40 | CoaxD | spacebass: All depends on config |
| 17:44.44 | NSGN | ok, well i only have one PSTN line |
| 17:44.45 | CoaxD | splas: I could get 1000 outbound calls on it |
| 17:44.46 | SpaceBass | true |
| 17:44.50 | NSGN | so i'll be alright with it? |
| 17:45.09 | CoaxD | Space: If the calls are sip, and sip redirects are sent so private parties are individually connected |
| 17:45.17 | CoaxD | (i.e. not connected to *) |
| 17:45.40 | slak- | CoaxD uhm dude it didnt work |
| 17:45.44 | SpaceBass | NSGN for what kind of use? home, test, play... its fine |
| 17:45.48 | CoaxD | slak: It DOES work |
| 17:45.55 | NSGN | home, one PSTN line, maybe five extensions |
| 17:45.58 | SpaceBass | CoaxD Im not doing direct connect at the moment |
| 17:46.00 | slak- | exten => myincomingnumber/blockednumber,1,Congestion() |
| 17:46.03 | slak- | that didnt work |
| 17:46.06 | CoaxD | slak: The example i gave you is for my own config |
| 17:46.07 | slak- | i just tested with my cell |
| 17:46.11 | SpaceBass | NSGN I have 10 extensions, 2 pstns, 2 bv sip lines... |
| 17:46.15 | slak- | its in my incoming config |
| 17:46.23 | NSGN | awesome, then i guess i'm good |
| 17:46.26 | CoaxD | slak: You're probably accepting the call before you have that there |
| 17:46.30 | *** join/#asterisk glm2k (n=GLM@rrcs-24-199-11-45.west.biz.rr.com) |
| 17:46.32 | NSGN | thanks. i'm out for now |
| 17:46.42 | slak- | CoaxD can i show you my config? |
| 17:46.59 | CoaxD | slak: I would look at it, except i need to leave |
| 17:47.00 | *** part/#asterisk FarrisG (n=jrush@h-68-164-19-170.dllatx37.covad.net) |
| 17:47.05 | slak- | dude :O |
| 17:47.08 | slak- | its so quick |
| 17:47.15 | CoaxD | slak: search on "ex girlfriend feature asterisk" with google |
| 17:47.19 | CoaxD | slak: Its all out there |
| 17:47.22 | Equinox | Where do you guys get DIDs? |
| 17:47.32 | SpaceBass | BV and pstn here |
| 17:48.40 | bjohnson | no |
| 17:57.11 | *** join/#asterisk adelas` (n=booger@rrcs-24-199-21-141.west.biz.rr.com) |
| 18:01.43 | *** join/#asterisk danny33923 (n=no@66.236.175.18.ptr.us.xo.net) |
| 18:02.27 | danny33923 | does anyone see something wrong with the following |
| 18:02.29 | danny33923 | exten => s,8,GotoIf($["${CALLERIDNUM}" = "9175551234"]?ext-group,4,2:9) |
| 18:03.45 | *** join/#asterisk oej (n=oej@apollo.webway.se) |
| 18:06.04 | *** join/#asterisk mithro (n=tim@c213-100-42-188.swipnet.se) |
| 18:06.55 | lathos42 | Is there anyway to pickup a call that's on hold from another phone? I've got it working to Pickup a call that's ringing elsewhere, but I want to be able to grab it when its on hold |
| 18:07.28 | anthm | try changrab |
| 18:08.40 | danny33923 | anyone know whats wrong with my statement? I wall the boss's phone calls to go into a ring group |
| 18:08.41 | danny33923 | exten => s,8,GotoIf($["${CALLERIDNUM}" = "9175551234"]?ext-group,4,2:9) |
| 18:09.46 | *** join/#asterisk Hmmhesays (n=Hmmm@66.173.103.107) |
| 18:09.53 | Hmmhesays | bind is not hunting for me |
| 18:10.07 | *** join/#asterisk litage (n=nick@203.201.96.203) |
| 18:12.24 | lathos42 | anthm: Will changrab allow me to steal a ringing call as well, so I could just have one extension that would either steal the ringing or on-hold call? |
| 18:12.42 | anthm | yah you can steal any channel |
| 18:14.16 | lathos42 | anthm: Cool.. now to see if I can manage to get it compiled :) I currently have 1.0.9 installed from the Gentoo ebuild |
| 18:16.54 | *** join/#asterisk Sedorox (n=Sed@smartserv/cna/Sedorox) |
| 18:19.35 | *** join/#asterisk Veto (i=mdkuser@cpe-66-69-38-192.satx.res.rr.com) |
| 18:20.25 | stkn | lathos42: it's been cvs head only, but somebody posted a patch in the comments to make it work with 1.0.x |
| 18:20.38 | stkn | lathos42: http://www.pbxfreeware.org/archives/2005/06/app_changrab_ta.html#comments |
| 18:21.19 | *** join/#asterisk kothog (n=kothog@S010600500480af12.gv.shawcable.net) |
| 18:21.42 | *** join/#asterisk Holos (n=asdf@72.1.197.10) |
| 18:21.52 | danny33923 | can anyone help with my GotoIf statement |
| 18:22.11 | lathos42 | Hmm.. Is Changrab CLI only? |
| 18:22.26 | anthm | no app and cli |
| 18:22.28 | Holos | Can anyone help me debug MOH? mpg123 is installed and is linked to /usr/bin but asterisk doesn't start the mpg123 processes even though the musiconhold.conf is set to load them. |
| 18:24.20 | Holos | Nevermind... Asterisk will not load MOH if there is a space between a comma and the last argument. ", -z" will fail, ",-z" will work. |
| 18:27.14 | *** join/#asterisk adjacent (n=scott@cpe-024-168-216-235.sc.res.rr.com) |
| 18:29.14 | *** join/#asterisk gdsm (n=gdsm@e1-1.ns500-1.ts.milt.as9105.net) |
| 18:30.25 | *** part/#asterisk qwue (n=vikramhe@210.212.195.134) |
| 18:30.35 | blessen | any idea..why i get this --error --> -- Got SIP response 408 "Request Timeout" back from 216.115.25.198 |
| 18:31.14 | Hmmhesays | anyone know why when you ping from windows bind doesn't hunt and when you ping from linux it does? |
| 18:31.59 | *** join/#asterisk dswillia (i=dswillia@wsip-68-15-113-253.ok.ok.cox.net) |
| 18:33.23 | dswillia | have a small problem, I have 2 pstn connections comming into 2 x100p clone cards, if a incomming pstn call comes in and gets sent to an extentions vm, if the person hangs up before the VM is rdy to record then the line stays open. I get a voicemail with dialtone and a message "If you would like to make a call" |
| 18:33.55 | dswillia | anyone have any idea how to get the VM to detect a hangup? |
| 18:34.09 | greg_work | dswillia: it's your card that's not detecting it |
| 18:34.35 | dswillia | ok |
| 18:35.31 | dswillia | so where would i need to look to have it detect it? |
| 18:35.56 | Hmmhesays | anyone in here real familiar with bind? i'm kind of a n00b |
| 18:36.58 | *** join/#asterisk ctooley (n=ctooley@rrcs-24-227-212-181.sw.biz.rr.com) |
| 18:37.52 | *** join/#asterisk hadi57 (n=al_moghr@83.136.8.206) |
| 18:39.41 | *** join/#asterisk Shoragan (n=shoragan@d072.apm.etc.tu-bs.de) |
| 18:39.51 | *** join/#asterisk [1]JohnJacob (n=JohnJaco@pcp0011543387pcs.mainf01.in.comcast.net) |
| 18:40.28 | lathos42 | Forgive my ignorance here, but is there any way to easily integrate the output of a diff into a file? |
| 18:41.16 | *** join/#asterisk CleanerX1 (n=nix@nat-ph3-wh.rz.uni-karlsruhe.de) |
| 18:43.44 | Corydon-w | lathos42: patch -p0 < /path/to/patchfile |
| 18:44.06 | Corydon-w | (assuming you're already in the base directory) |
| 18:44.12 | zoa2 | hey ho |
| 18:44.17 | zoa2 | new sip jb patch is online |
| 18:44.24 | Corydon-w | zoa2: hey fag |
| 18:46.42 | *** join/#asterisk L0rdy (i=steve@d127152.adsl.hansenet.de) |
| 18:46.51 | zoa2 | hey wussie |
| 18:47.06 | Corydon-w | hey slut |
| 18:47.49 | Beirdo | jeez |
| 18:48.28 | *** join/#asterisk trimi` (i=Pharrel@62.162.242.206) |
| 18:48.36 | *** join/#asterisk pc4 (n=pc@209.151.52.81) |
| 18:48.51 | pc4 | In a makefile, what is this line performing in order to return a negative/positive? @if ! [ -d $(KSRC) ]; then echo "You do not appear to have the sources for the $(KVERS) kernel installed."; exit 1 ; f |
| 18:48.55 | pc4 | How is it determining (what is it running) that is returning a positive causing a missing kernel sourcem ssage? |
| 18:49.09 | trimi` | err, i got serious problem, sometimes asterisk dont understand some dmtp which comes from a PSTN into the ZAP, how can i slove it ? |
| 18:49.13 | Katty | did someone call me? |
| 18:49.28 | Katty | Hmmhesays: you didn't call me, did you? |
| 18:49.56 | *** join/#asterisk r0m (n=SysOp@a81-84-68-89.cpe.netcabo.pt) |
| 18:51.22 | *** join/#asterisk Falle (i=falstaf@voip-forum.se) |
| 18:52.02 | *** part/#asterisk Moc (n=mochouin@h66-201-214-109.gtconnect.net) |
| 18:52.06 | *** join/#asterisk kg (n=kg@chello062179062077.chello.pl) |
| 18:52.08 | Hmmhesays | Katty: negatory |
| 18:52.18 | *** join/#asterisk bhima (n=gf2e@i13pc168.ilkd.uni-karlsruhe.de) |
| 18:52.24 | zoa2 | help i need more people to test the sip jb |
| 18:52.58 | bhima | I just signed up, and got allocated a DID with nufone, but I haven't gotten an e-mail with config info and I can't find it on their web site. Any idea where I can find that? |
| 18:53.25 | |dennis| | Question.. wouald a p4, 2.6, 512 ram be enough to run a SOHO asterisk setup?? with about 15-20 extensions...4 pstn lines..?? |
| 18:53.28 | Hmmhesays | ~seen ManxPower |
| 18:53.32 | jbot | manxpower <n=eric@dsl-209-16-67-160.datasync.com> was last seen on IRC in channel #asterisk, 2h 8m 47s ago, saying: 'JerJer: Count the ways! Count the ways!'. |
| 18:53.35 | Katty | Hmmhesays: k |
| 18:53.46 | *** join/#asterisk myiagy (n=myiagy@200.138.215.78) |
| 18:53.50 | *** join/#asterisk Delete (n=Bloody@213.226.30.193) |
| 18:54.17 | trimi` | err, i got serious problem, sometimes asterisk dont understand some dmtp which comes from a PSTN into the ZAP, how can i slove it ? |
| 18:54.39 | zoa2 | relaxdtmf |
| 18:54.50 | trimi` | <zoa2> ? |
| 18:55.07 | trimi` | can you explain what you meant ? |
| 18:55.12 | Hmmhesays | can someone give me a hand with bind, this is irritating me |
| 18:55.20 | blessen | bind ? |
| 18:55.28 | drray | dns/bind |
| 18:55.32 | Hmmhesays | aye |
| 18:55.32 | *** join/#asterisk santiago (n=santiago@63.245.86.163) |
| 18:55.40 | zoa2 | google for relaxdmtf |
| 18:55.42 | blessen | yes bind is a software wwhich implements the dns server |
| 18:55.53 | drray | I know enough about bind to not run my own dns server |
| 18:55.54 | zoa2 | |dennis|: yes |
| 18:55.58 | blessen | its also called named |
| 18:56.02 | Hmmhesays | damn thing doesn't rotate my addresses every time |
| 18:56.04 | trimi` | thnx zoa2 |
| 18:56.09 | |dennis| | zoa2: thanks... :) |
| 18:56.14 | blessen | it has files like named.conf which stores the details of the domain zone files |
| 18:56.28 | Hmmhesays | i have a fake domain, booty.call with a subdomain hot |
| 18:56.36 | blessen | and there will be a folder which stores zone files....it can be found from named.conf file |
| 18:56.40 | Hmmhesays | hot.booty.call has 2 address entries |
| 18:56.52 | RaYmAn-Bx | Hmmhesays: afaik the only rr dns records that bind implements is random. |
| 18:57.09 | Hmmhesays | why the fsck do they call it round robin then |
| 18:57.19 | lathos42 | http://pastebin.ca/21140 |
| 18:57.24 | RaYmAn-Bx | who knows :) |
| 18:57.37 | lathos42 | This is the output I get when I try to compile app_changrab with the patch for 1.0.x |
| 18:57.41 | RaYmAn-Bx | I might be remembering wrong though. |
| 18:57.42 | blessen | http://pastebin.ca/21127 |
| 18:57.51 | Hmmhesays | of the docs i read... it sure sounds like round robin |
| 18:58.22 | Katty | Hmmhesays: help. |
| 18:58.25 | RaYmAn-Bx | does it really matter whether it's 100% equally divided? |
| 18:58.26 | Hmmhesays | with? |
| 18:58.28 | Katty | Hmmhesays: exim4 has insaned. |
| 18:58.36 | Hmmhesays | RaymAn-Bx: yes |
| 18:58.39 | Katty | Hmmhesays: i have errrr log. |
| 18:58.45 | Hmmhesays | pastebin it |
| 18:58.47 | Katty | Hmmhesays: with relaying denied. |
| 18:58.48 | Katty | k |
| 18:58.50 | Hmmhesays | www.pastebin.can |
| 18:58.56 | Hmmhesays | *ca |
| 18:59.27 | *** join/#asterisk Lan16`spdy^gTp (n=spdy@lns-th2-15-poi-82-64-226-46.adsl.proxad.net) |
| 19:02.50 | *** join/#asterisk iCEBrkr (i=icebrkr@rrcs-24-129-130-158.se.biz.rr.com) |
| 19:03.10 | Cherbru_ | Anyone have any comments on running Asterisk on a small IBM x206 server with a 2.4Ghz Celeron, 256MB of ram, and a LSI Logic Megaraid 150-6 SATA raid card with RAID1 on the drives? |
| 19:03.21 | Cherbru_ | it's for a small client |
| 19:03.22 | *** join/#asterisk twitchmax (n=twitch@c-24-98-56-147.hsd1.ga.comcast.net) |
| 19:04.09 | twitchmax | greetings all |
| 19:04.18 | RaYmAn-Bx | Cherbru_: it depends on how small and what it's going to be doing.. |
| 19:04.28 | Cherbru_ | lets say 30 users |
| 19:04.29 | twitchmax | was wondering if there is a way to make a h323 trunk with amp? |
| 19:04.36 | Cherbru_ | g711 |
| 19:04.45 | *** part/#asterisk bhima (n=gf2e@i13pc168.ilkd.uni-karlsruhe.de) |
| 19:04.53 | Cherbru_ | but I might be interested in doing g711 to g729 transcoding later |
| 19:05.01 | Cherbru_ | will a Celeron be ok or should I use a P4? |
| 19:05.17 | zoa2 | it will be enough |
| 19:05.40 | zoa2 | but 30 simultaneous calls will be on the edge |
| 19:06.07 | blessen | someone please help me |
| 19:06.11 | blessen | CLI> Aug 25 12:05:22 NOTICE[28586]: chan_sip.c:5782 register_verify: Peer 'kphone' is trying to register, but not configured as host=dynamic |
| 19:06.13 | Cherbru_ | for 30 simultaneous calls I WILL need g729 becuase it's only a 1.5Mbit link |
| 19:06.26 | zoa2 | gsm will also work |
| 19:06.29 | Cherbru_ | but I don't think I'll ever reach 30 calls with 30 users |
| 19:06.38 | zoa2 | yeah also doubt that |
| 19:06.43 | Cherbru_ | my trunk only does g711 and g729 |
| 19:07.05 | lathos42 | Has anyone here compiled app_changrab on 1.0.x before? |
| 19:07.05 | Cherbru_ | but how does a celeron compare to a P4 doing transcoding? |
| 19:07.06 | mutilator | hey all, on a router when i type a command and it exceeds the buffer it says --More--, is there a way with Net::Telnet to find that and send it a space so it continues with the command? |
| 19:07.40 | zoa2 | emulate a bigger screen type |
| 19:10.01 | Cherbru_ | Digium's site says a dual 1.8Ghz Xeon will do 80 calls with g.729 |
| 19:10.10 | Cherbru_ | they don't say what it's transcoding from.. I assume g.711 |
| 19:10.23 | mutilator | .. |
| 19:10.30 | Cherbru_ | er 60 calls that is |
| 19:10.31 | *** part/#asterisk Lan16`spdy^gTp (n=spdy@lns-th2-15-poi-82-64-226-46.adsl.proxad.net) |
| 19:10.40 | Cherbru_ | dual Xeon 2.8 did 80 calls |
| 19:10.52 | *** join/#asterisk websae (n=websae@207-118-143-134.dyn.centurytel.net) |
| 19:11.43 | Cherbru_ | so a 2.4ghz Celeron MIGHT be able to do 30 |
| 19:11.53 | zoa2 | might |
| 19:11.54 | zoa2 | just |
| 19:11.59 | zoa2 | dual xeon could do more |
| 19:12.16 | brookshire | zoa2: omg! you have a clone? |
| 19:12.22 | zoa2 | yes |
| 19:12.30 | brookshire | are you the evil zoa? |
| 19:12.34 | brookshire | or the good zoa |
| 19:12.48 | zoa2 | they copied me, despite my "no part of this body may be reproduced by any means t-shirt" |
| 19:12.57 | zoa2 | i dont know, im the left brain cell |
| 19:12.57 | *** join/#asterisk ADANALI_TURK (n=Yuzuk-4@81.214.220.63) |
| 19:13.14 | brookshire | heh |
| 19:13.40 | *** join/#asterisk naturalblue (n=Adam@195.26.12.229) |
| 19:15.17 | Holos | Cherbru_ You in US or Canada? |
| 19:16.07 | *** join/#asterisk myiagy (n=myiagy@200.138.215.78) |
| 19:16.46 | pc4 | Can anyone help me compile zaptel ztdummy? I am having serious problems with kernel sources |
| 19:16.54 | Hmmhesays | what os? |
| 19:17.44 | Cherbru_ | Holos: US |
| 19:19.12 | pc4 | Hmm-home - suse 9.2 |
| 19:19.19 | pc4 | hmmhesays - suse 9.2 |
| 19:19.41 | pc4 | I've been playing with the MAkefile but I can't get anything to work. |
| 19:19.54 | pc4 | And I can't download/recompile a kernel on the machine. |
| 19:20.21 | Holos | Cherbru_: Look for a non-celeron IBM, if you have other backup plans the x306 is a 1U for $1000 right now, and it's a 3.0Ghz with a 36G SCSI, add a 300GB SATA if you need it. |
| 19:20.43 | stkn | lathos42: mind if i pm you? |
| 19:20.55 | lathos42 | stkn: Go ahead |
| 19:20.59 | Cherbru_ | Holos: I want RAID1 |
| 19:21.31 | Cherbru_ | I can get a 3ghz P3 x206 |
| 19:21.35 | *** join/#asterisk pa (n=Paolo@unaffiliated/pa) |
| 19:21.36 | Cherbru_ | for a few hundred more |
| 19:21.45 | Holos | Cherbru_: Hmm.. That would power a lot more calls... |
| 19:22.35 | *** join/#asterisk criptos (n=criptos@201.137.246.228) |
| 19:22.47 | *** join/#asterisk Optic (i=dfraser@H31.C18.B96.tor.eicat.ca) |
| 19:22.50 | criptos | How a peer is formed at chan_iax2? |
| 19:23.04 | Cherbru_ | I can get the x206 for $722 and then 2 80 gig SATA hot swap drives for $170 each |
| 19:23.12 | Optic | hello |
| 19:23.16 | criptos | hello |
| 19:23.16 | Cherbru_ | and then a LSI MegaRAID 150-6 for $346 |
| 19:23.24 | Cherbru_ | and i'd be arond $1400 |
| 19:23.30 | Optic | we just got a happy new PRI installed here! |
| 19:23.41 | Optic | TE110P... but we're having some echo issues |
| 19:23.49 | *** part/#asterisk santiago (n=santiago@63.245.86.163) |
| 19:23.49 | zoa2 | try the new echo canceller |
| 19:23.55 | Optic | is it better? |
| 19:23.55 | zoa2 | kb1ec |
| 19:23.59 | zoa2 | they claim it is |
| 19:24.04 | pc4 | Can anyone help me compile zaptel ztdummy? I am having serious problems with kernel sources. It keeps erroring out. What exact files does it need? I'm trying to decipher the Makefile, but I can't read it worht a damn. |
| 19:24.08 | Optic | we're getting long echos... like 1/2 second or more |
| 19:24.09 | Cherbru_ | Holos: But you think a 3ghz P4 is that much better than a 2.4ghz Celeron? |
| 19:24.10 | Optic | which is very strange |
| 19:24.28 | *** join/#asterisk luke-jr__ (n=luke-jr@CPE-65-26-132-140.kc.res.rr.com) |
| 19:24.41 | pc4 | Cherbru_ - Sure is for multitasking =). Don't know about asteirsk though. |
| 19:25.07 | *** join/#asterisk dersteer (n=travis@24-236-197-212.dhcp.aldl.mi.charter.com) |
| 19:25.10 | Optic | which I just can't account for |
| 19:25.24 | Cherbru_ | I will most likely put PBXWare on it |
| 19:25.38 | pc4 | holos - Go to tomshardware.com and find a similar speed test in one of his processor benchmarks. Encoding a mp3 or math i/o should be relevant tests. |
| 19:25.56 | pc4 | I doubt he has a voip one. |
| 19:26.12 | infinity1 | is there a way to timestamp the console log? |
| 19:26.35 | Optic | new echo canceller in asterisk cvs? |
| 19:26.36 | Cherbru_ | pc4: Like this? http://www.tomshardware.com/cpu/20020903/p4_celeron-08.html#mp3_encoding_lame_392 |
| 19:26.43 | tzanger | Optic: yup |
| 19:26.46 | tzanger | KB1 |
| 19:26.48 | tzanger | it's good shit |
| 19:26.51 | criptos | if busydetect=no how a zap channel know when to hangup? |
| 19:26.52 | Optic | ok |
| 19:27.04 | tzanger | criptos: CPD or it's told to hang up |
| 19:27.07 | Optic | do you have to make a special define or config prameters? are there docs? |
| 19:27.14 | tzanger | busydetect is only inband "try to hear a hangup" |
| 19:27.33 | tzanger | criptos: keep it in here |
| 19:27.40 | tzanger | CPD = called/calling party disconnect |
| 19:27.49 | websae | anyone here using FreeBSD? |
| 19:27.52 | tzanger | websae: nope |
| 19:28.01 | websae | linux distro heh? |
| 19:28.07 | websae | i am running bsd having a few issues |
| 19:28.07 | tzanger | yup |
| 19:28.18 | criptos | How can I only use busydetect and no CPD hangup detection? |
| 19:28.43 | tzanger | criptos: you don't want that |
| 19:28.49 | tzanger | you want to avoid busy detect if at all possible |
| 19:28.59 | pc4 | Cherbru_ - Yup, should be pretty accurate. |
| 19:28.59 | tzanger | use kewlstart signalling |
| 19:29.31 | pc4 | Cherbru_ - About the same I guess. Doesn't answer the multithreaded question and what nature the reduced cache size has, but it looks like the answer is "not much". |
| 19:29.51 | pc4 | Cherbru_ - 2.4 may have different cache size, but won't matter a whole lot. |
| 19:29.55 | criptos | tzanger: fxsks signalling at zaptel.conf, and I have a LOT of call dropped... |
| 19:30.06 | trimi` | i got a small problem, when a incomming call into my ZAP hang up, the asterisk keep the line open for like 20 or 30 sec. and the realese it so some1 else cann call, is there any option to make the PSTN FREE imediately ???? |
| 19:30.19 | Katty | who's good with exim4? |
| 19:30.48 | Katty | nevermind. |
| 19:31.11 | blessen | SipClient: Received: 12:30:48.711 |
| 19:31.11 | blessen | --------------------------------- |
| 19:31.11 | blessen | SIP/2.0 401 Unauthorized |
| 19:31.27 | *** join/#asterisk neron (n=chatzill@adsl-67-119-149-94.dsl.lsan03.pacbell.net) |
| 19:31.29 | criptos | I`m usning fxs_ks at zapata.conf also, so.. what else can I do to avoid doping calls from a zaptel interface? |
| 19:31.41 | *** join/#asterisk gooagle (n=goldenol@ns2.xoasisnetworks.com) |
| 19:31.44 | blessen | any idea why this error....please help me connect to asterisk using kphone |
| 19:31.45 | gooagle | hi |
| 19:31.50 | criptos | hi |
| 19:31.53 | Cherbru_ | Anyone use wcfxsusb? |
| 19:32.10 | criptos | nope |
| 19:32.10 | neron | Hi everyone! |
| 19:32.20 | gooagle | anyone have any additional tips on getting rid of echo on PRIs? The various MARK2, MARK3, AGGRESSIVE, etc. seem to help but not much... Not sure how echotraining fits in either. |
| 19:32.56 | neron | I just found out about Asterisk last night and I think it's great!!! I can't believe this system is open source! |
| 19:32.58 | *** part/#asterisk twitchmax (n=twitch@c-24-98-56-147.hsd1.ga.comcast.net) |
| 19:33.28 | *** join/#asterisk CharlieHorse (n=MattH@63.174.244.174) |
| 19:34.06 | CharlieHorse | Hi... does anyone here know anything about the RTP jitterbuffer patch? I'm trying to use it but it makes the speech sound like it's been slowed down (coming back to the asterisk server).. alot |
| 19:35.00 | neron | Does Asterisk support legacy PBX system and can it be used only as a voicemail system? |
| 19:35.24 | websae | any FreeBSD users here? |
| 19:35.28 | websae | need some help.. |
| 19:35.35 | trimi` | i got a small problem, when a incomming call into my ZAP hang up, the asterisk keep the line open for like 20 or 30 sec. and the realese it so some1 else cann call, is there any option to make the PSTN FREE imediately ???? |
| 19:36.02 | lathos42 | anthm: What would be proper syntax be for changrab if I wanted to grab the call ringing into the phone. I've got it working fine for grabbing the on-hold call |
| 19:36.26 | anthm | you could try app_intercept for that too |
| 19:36.56 | anthm | it's just changrab(<chan_name>) |
| 19:37.14 | zoa2 | anthm, join #sipjb |
| 19:37.24 | *** join/#asterisk litage (n=nick@203.201.97.50) |
| 19:38.20 | neron | anyone here familar with the Replay Plus voicemail system? |
| 19:38.27 | infinity1 | is there a reason the callerid name isn't given by teliax when calls come in? only the callerid number is populated. |
| 19:38.56 | neron | brb |
| 19:39.04 | *** join/#asterisk dwmw2_gone_ (i=ctrlprox@baythorne.infradead.org) |
| 19:39.12 | lathos42 | whoops! that was a bad thing |
| 19:40.12 | lathos42 | I put it in my dialplan like this .. exten -> #9399,1,changrab(SIP/399) and Asterisk crashed when I tried it |
| 19:40.35 | blessen | My kphone is not talking to my asterisk server |
| 19:40.41 | blessen | :-( please help me.. |
| 19:40.51 | websae | anyone know a LINUX distro that I can make boot floppies and install from their FTP server? |
| 19:41.14 | tzanger | websae: debian |
| 19:41.17 | zoa2 | debian |
| 19:41.18 | tzanger | for sure, not sure who else |
| 19:41.24 | infinity1 | websae: floppy drive? whats that? |
| 19:41.48 | websae | debian has boot floppies |
| 19:41.52 | websae | just 2 you think? |
| 19:41.56 | Cybertank__ | websae just about any of them |
| 19:41.57 | *** join/#asterisk axelilly (n=axe@208.51.242.164) |
| 19:42.34 | websae | looking to install asterisk on a LINUX distro, which is the most popular that is has been installed on? I've tried my FreeBSD machine and not working well |
| 19:42.47 | zoa2 | i thin debian or fc3 or the most popular |
| 19:42.50 | jontow | lots of gentoo, fedora core, and debian |
| 19:42.55 | tzanger | websae: I think you need to start with something known to work |
| 19:42.57 | Cybertank__ | websae... Asterisk@home. :) |
| 19:43.02 | axelilly | tao works too |
| 19:43.10 | tzanger | that's my initial guess too... I personally run slackware but it's not for everyone |
| 19:43.13 | trimi` | i got a small problem, when a incomming call into my ZAP hang up, the asterisk keep the line open for like 20 or 30 sec. and the realese it so some1 else cann call, is there any option to make the PSTN FREE imediately ???? |
| 19:43.18 | jontow | yeah, learn asterisk before learning how to make it work on a officially unsupported OS ;) |
| 19:43.23 | jontow | definitely my advice |
| 19:43.24 | *** join/#asterisk Veto (i=mdkuser@cpe-66-69-38-192.satx.res.rr.com) |
| 19:43.24 | trimi` | no1 dont answer my question :( |
| 19:43.42 | axelilly | does dundi come compiled into asterisk or do I need to DL it from somewhere? |
| 19:43.56 | jontow | trimi; make sure you have a Hangup line at the end of your outbound calls. |
| 19:44.09 | infinity1 | does ${CALLERIDNAME} ever get populated on incoming calls that come through iax? |
| 19:44.13 | jontow | axelilly; CVS HEAD only, if my mind serves me correctly today. |
| 19:44.54 | axelilly | jontow: ah ok, I compiled that last official release, that could explain why it's missing |
| 19:45.03 | jarrod | why would inbound faxing work from cisco -> t38 -> fax, but outbound negotiates t38 then falls back to g711u |
| 19:45.04 | Cybertank__ | infinity1 yes... by default... unless you set an id on the iax trunk. |
| 19:45.07 | axelilly | jontow: is it available as an addon? |
| 19:45.18 | jontow | no |
| 19:45.26 | infinity1 | Cybertank__: hm. |
| 19:45.50 | *** part/#asterisk Nix (n=Nix@81.213.125.220) |
| 19:45.56 | infinity1 | Cybertank__: what do you mean by ID? |
| 19:46.11 | Optic | moo moo |
| 19:46.14 | jarrod | sipSPICompareStreams: negotiated codec changed from T38Fax to g711ulaw |
| 19:46.14 | *** join/#asterisk W|NGNUT (n=wingnut-@062016241059.customer.alfanett.no) |
| 19:46.36 | infinity1 | Cybertank__: all the calls i get from teliax don't have calleridname. i don't see an ID in iax.conf. |
| 19:47.21 | Optic | so to get the new echo canceller what tag should I check out? HEAD? |
| 19:47.38 | zoa2 | yeah head |
| 19:47.45 | Optic | is it pretty stable>? |
| 19:48.05 | `Sauron | when I try to bridge a sip call out my zap/1 interface |
| 19:48.12 | `Sauron | .. I get nothing but silence |
| 19:48.44 | `Sauron | anyone know why this would happen? |
| 19:49.46 | `Sauron | It tells me that Zap/1/<number> answered |
| 19:49.59 | `Sauron | but it's still ringing the other phone |
| 19:51.08 | criptos | so no one know how to only rely on busydectect function to hangup a zap channel? |
| 19:51.20 | axelilly | how do you check out head? |
| 19:51.27 | *** join/#asterisk dwmw2_gone (i=ctrlprox@baythorne.infradead.org) |
| 19:51.43 | criptos | login to cvs and only make cvs checkount asterisk libpri zaptel |
| 19:51.59 | criptos | further instructions at: http://beta.asterisk.org/download |
| 19:52.05 | axelilly | ok, thank you |
| 19:52.10 | *** join/#asterisk file[laptop] (n=jcolp@mctnnbsa30w-156034056020.nb.aliant.net) |
| 19:52.50 | infinity1 | Cybertank__: who are you using for inbound iax? apparently some providers do not supply it. |
| 19:53.05 | JerJer | not a celeron |
| 19:53.30 | *** join/#asterisk ManxPower (n=eric@dsl-209-16-67-160.datasync.com) |
| 19:53.32 | *** join/#asterisk eksffa (n=fbsdbr8@c911c45e.bhz.virtua.com.br) |
| 19:53.38 | CharlieHorse | any idea why this happens when I hangup? |
| 19:53.39 | CharlieHorse | Aug 25 15:53:19 WARNING[1094] cdr.c: CDR on channel 'SIP/400-08bd' not posted |
| 19:53.39 | CharlieHorse | Aug 25 15:53:19 WARNING[1094] cdr.c: CDR on channel 'SIP/400-08bd' lacks end |
| 19:55.05 | ManxPower | ran out of disk space on /var?> |
| 19:55.14 | glm2k | are you using ForkCDR() or ResetCDR() or NoCDR()? |
| 19:55.18 | blessen | can anyone help me configure kphone and asterisk |
| 19:55.22 | blessen | ? |
| 19:55.35 | mutilator | just got a call back from digium from my talk yesterday |
| 19:55.47 | CharlieHorse | glm2k, where would I find that? I thought I was using ResetCDR.. |
| 19:55.50 | mutilator | he confirmed, specs say te110p can go 2000m |
| 19:56.00 | CharlieHorse | ManxPower, there is 36GB free.. /var is part of / |
| 19:56.02 | glm2k | CharlieHorse: extensions.conf |
| 19:56.13 | mutilator | and the adtran ta624 i have can do the same. so i can make a mile link no problem |
| 19:56.45 | CharlieHorse | glm2k, hangupcall is s,1,ResetCDR(w) then s,2,NoCDR() s,3,Wait(5) s,4,Hangup |
| 19:57.12 | jarrod | anyone using cisco 7200 as t38 gateway |
| 19:57.15 | ManxPower | CharlieHorse: Im, if you don't want a CDR then it's not going to post the CDR |
| 19:57.22 | glm2k | just as i thought. i use each in a different context and they spit out the same error. or one of them does. |
| 19:57.27 | ManxPower | CharlieHorse: If you don't want those messages, change logger.conf |
| 19:57.40 | glm2k | what ManxPower said |
| 19:57.42 | CharlieHorse | but I do want a CDR |
| 19:58.06 | CharlieHorse | the issue is not that I don't want those messages... I want the CDR! :) |
| 19:58.09 | ManxPower | CharlieHorse: Then stop running ResetCDR, and NoCDR. |
| 19:58.14 | glm2k | not after NoCDR() you wont |
| 19:58.27 | CharlieHorse | hrmm interesting ok.. I had those on another system and it will still logging them |
| 19:58.37 | CharlieHorse | so I should resetCDR? or just hangup |
| 19:58.45 | glm2k | since i knew what i wanted and didn't want, i just changed logger.conf and pretended it never happened :) |
| 19:58.45 | ManxPower | When I shoot myself in the foot it hurts! |
| 19:58.52 | CharlieHorse | lol |
| 19:58.57 | ManxPower | CharlieHorse: Just don't mess with CDR, the default will work. |
| 19:59.16 | CharlieHorse | ManxPower, so you're saying remove the resetcdr and nocdr and let the system deal |
| 19:59.22 | ManxPower | CharlieHorse: "show application resetcdr" would be the correct place to see the docs for it. |
| 19:59.29 | ManxPower | CharlieHorse: Yeah. |
| 19:59.36 | websae | a lot people here use Debian? |
| 20:00.04 | *** join/#asterisk L|NUX (n=linux@202.5.145.14) |
| 20:01.18 | infinity1 | are there companies similar to teliax? teliax does not support calleridname! |
| 20:01.23 | *** join/#asterisk dwmw2_gone (i=ctrlprox@baythorne.infradead.org) |
| 20:01.53 | *** join/#asterisk Ariel_ (n=Ariel@dsl-20-177.cofs.net) |
| 20:02.07 | *** join/#asterisk fourcheeze (n=rich@westbury.doilywood.org.uk) |
| 20:02.40 | CharlieHorse | ahh there we go |
| 20:02.42 | *** part/#asterisk criptos (n=criptos@201.137.246.228) |
| 20:02.56 | fourcheeze | I'm trying to set the caller id number with an AGI script |
| 20:03.10 | fourcheeze | the debug shows the script is called |
| 20:03.16 | ManxPower | infinity1: most VoIP carriers don't support incoming callerid name, only the really big ones might. |
| 20:03.30 | fourcheeze | but the caller id doesn't get set |
| 20:03.34 | ManxPower | fourcheeze: stop asterisk, start it again as "asterisk -cvvvddd" |
| 20:03.45 | fourcheeze | ManxPower: ok, will do |
| 20:03.56 | *** join/#asterisk dwmw2_gone (i=ctrlprox@baythorne.infradead.org) |
| 20:05.07 | ManxPower | fourcheeze: you'll see more debugging info for AGI that way. |
| 20:05.13 | *** join/#asterisk nax (n=nax@dyn231-119.dsl.ligado.de) |
| 20:05.13 | Ariel_ | Ok Katrina is now a Cat 1 hurricane and it's just 30 miles off shore...It's raining cats and dogs folks here. |
| 20:05.40 | ManxPower | Ariel_: better you than me. |
| 20:05.52 | Ariel_ | ManxPower, it's ok things happen |
| 20:06.01 | Ariel_ | looks like it's going to miss you completely |
| 20:06.05 | *** join/#asterisk [1]JohnJacob (n=JohnJaco@pcp0011543387pcs.mainf01.in.comcast.net) |
| 20:06.14 | ManxPower | Ariel_: Yeah! |
| 20:06.24 | infinity1 | ManxPower: do you know of any that do? |
| 20:06.36 | infinity1 | ManxPower: besides vonage. heh |
| 20:06.39 | *** join/#asterisk dwmw2_gone_ (i=ctrlprox@baythorne.infradead.org) |
| 20:06.53 | ManxPower | They should just close down Florida. It gets too many hurricanes. |
| 20:07.07 | Katty | Ariel_: way to set off my hilight. |
| 20:07.13 | Katty | Ariel_: katrina is a /nick on another server :P |
| 20:07.18 | Katty | Ariel_: two other servers, actually. |
| 20:07.27 | Ariel_ | Katty, sorry |
| 20:07.36 | Katty | Ariel_: s'ok (= |
| 20:07.49 | fourcheeze | ManxPower: good shout |
| 20:07.55 | fourcheeze | I'm getting permission denied on the script |
| 20:08.03 | Ariel_ | The lake in the back is raising fast. |
| 20:08.27 | Beirdo | Katty: well your other /nick caused me to change my flights :) |
| 20:08.32 | glm2k | that doesn't sound good |
| 20:08.53 | Beirdo | well... |
| 20:09.13 | Beirdo | I ain't gonna willingly fly through Miami when I have another choice. |
| 20:09.22 | Beirdo | that storm's gonna mess with the flights |
| 20:09.23 | Optic | our office is OUT OF COFFEE |
| 20:09.27 | Optic | omg |
| 20:09.31 | Ariel_ | fun just got hit by a flying trash can... they really make lots of noise |
| 20:09.37 | Optic | ahh |
| 20:09.42 | Optic | new asterisk requires new libpri |
| 20:09.46 | Optic | this isn't going to be like butter, is it? :) |
| 20:09.48 | glm2k | Ariel_: you should set up a webcam :) |
| 20:10.00 | Beirdo | Ariel_: where ya located? |
| 20:10.37 | *** join/#asterisk sob0l (n=peter@uo166.internetdsl.tpnet.pl) |
| 20:10.38 | mutilator | haha, techsupport just had a voicemail that said "My cell phone's not working" then hung up |
| 20:11.02 | glm2k | rotfl |
| 20:11.29 | mutilator | the person sounded so confused too |
| 20:12.25 | *** join/#asterisk cjk (n=cjk@212.233.32.55) |
| 20:12.52 | Ariel_ | Beirdo, West Kendall |
| 20:13.03 | Ariel_ | about 1 mile north of KTMB |
| 20:13.09 | Ariel_ | ~weather KTMB |
| 20:13.20 | Insanity5 | ~weather ksun |
| 20:13.44 | Ariel_ | lt lies |
| 20:13.52 | Insanity5 | yes it does |
| 20:14.01 | Beirdo | cool |
| 20:14.20 | lathos42 | ~weather AZO |
| 20:14.32 | Beirdo | well, I'm flying to San Juan on Saturday, I don't need to worry about storm avoidance now, flying through Chicago now |
| 20:14.52 | Ariel_ | just look at this weather map:http://www.weather.com/outlook/travel/businesstraveler/map/33196 |
| 20:15.09 | Insanity5 | Compare ksun here to the weather on AVIS @ 1-877-788-2967 |
| 20:15.13 | Insanity5 | =) |
| 20:15.25 | Insanity5 | One of the two is a little bit off =) |
| 20:15.31 | Ariel_ | lathos42, need 4 letters |
| 20:15.39 | lathos42 | ~weather KFPK |
| 20:15.44 | Insanity5 | If it's in north america, add "K" |
| 20:15.51 | Beirdo | ~weather CYTZ |
| 20:15.52 | glm2k | Ariel_: bloody hell, that's some storm |
| 20:15.54 | Insanity5 | lathos42 - So try kazo |
| 20:15.56 | *** join/#asterisk SuperBawlz (i=user@72.22.41.86) |
| 20:16.06 | *** part/#asterisk oej (n=oej@apollo.webway.se) |
| 20:16.13 | Beirdo | anyways, enough bot-spam :) |
| 20:16.14 | Beirdo | hehe |
| 20:16.21 | Insanity5 | Does an ss7 isdn line over a normal one have any advantages to asterisk? |
| 20:16.35 | lathos42 | Yeah, AZO is actually KAZO |
| 20:16.36 | Ariel_ | ss7 isdn??? |
| 20:16.38 | Ariel_ | well |
| 20:17.15 | h3x0r | i think you mean SS7 ISUP |
| 20:17.40 | Insanity5 | might be |
| 20:17.44 | *** join/#asterisk DrWho17 (n=MIKE@179-219-112-64.dsl.tc3net.com) |
| 20:17.52 | SuperBawlz | Quick question, what do in need hardware wise to accept 8 lines of POTS calls and transfer them to a Nortel MICS cabnet? |
| 20:18.02 | *** join/#asterisk dwmw2_gone (i=ctrlprox@baythorne.infradead.org) |
| 20:18.06 | h3x0r | if you dont know if you need it or not then you dont need it |
| 20:18.16 | Insanity5 | :) |
| 20:18.20 | h3x0r | there are few things you can do with isup that you cant do with isdn |
| 20:18.32 | Insanity5 | like? |
| 20:18.45 | h3x0r | talk to peoples switches that dont support isdn |
| 20:18.46 | h3x0r | heh |
| 20:18.57 | h3x0r | there are a few things |
| 20:19.03 | h3x0r | but most of them dont matter coz asterisk cant do it anyway |
| 20:19.09 | h3x0r | class 5 features and shit |
| 20:19.28 | Insanity5 | I need two pots lines -- am I better just ordering one isdn? |
| 20:19.29 | Insanity5 | samep rice |
| 20:19.31 | Insanity5 | about |
| 20:19.53 | Insanity5 | termination hardware is more though :( |
| 20:19.56 | h3x0r | SuperBawlz: Throw the nortel away and use asterisk and voip ohones |
| 20:19.57 | h3x0r | phone |
| 20:19.57 | h3x0r | s |
| 20:19.58 | h3x0r | heh |
| 20:20.04 | SuperBawlz | I would like to. |
| 20:20.10 | SuperBawlz | I can't get this company to do it yet. |
| 20:20.26 | SuperBawlz | If I had my way I would run the whole thing SIP / IAX |
| 20:20.29 | Insanity5 | Keep the nortel on standby and use voip |
| 20:20.32 | Insanity5 | =) |
| 20:20.32 | h3x0r | if you need to do fxo to the nortel |
| 20:20.36 | h3x0r | it will cost more than buying more phones |
| 20:20.40 | h3x0r | seriously |
| 20:20.51 | *** join/#asterisk MRH2 (n=Mr_happy@fcirc-adsl.demon.co.uk) |
| 20:20.57 | Insanity5 | h3x0r - Can you help me compile zaptel/ztdummy? |
| 20:21.00 | Insanity5 | It is been hell |
| 20:21.25 | *** part/#asterisk emp (n=emp@70.57.239.37) |
| 20:21.43 | Insanity5 | and rebooting/recompiling the kernel on this box is not an option. |
| 20:21.45 | dgilmore | anyone here familiar with sipura 2100 |
| 20:21.58 | dgilmore | specifically dtmf handling |
| 20:22.22 | h3x0r | do you really need ztdummy |
| 20:22.34 | h3x0r | you only need it for meetme or iax2 _trunking_ on a box with no zaptel devices |
| 20:22.36 | Insanity5 | h3x0r - on hold music sounds like shit |
| 20:22.49 | h3x0r | i dont think moh uses ztdummy does it |
| 20:22.55 | h3x0r | thats probably because you are using mpg321 not mpg123 |
| 20:22.59 | h3x0r | are you on redhat or something |
| 20:23.02 | *** join/#asterisk oplog9 (n=oplog9@206.222.29.50) |
| 20:23.04 | Insanity5 | using mpg123 |
| 20:23.11 | Insanity5 | I was told it is only needed for on hold / prerecorded stuff |
| 20:23.27 | h3x0r | ls -l `which mpg123` |
| 20:23.28 | Insanity5 | call 1-866-249-2403 and listen :( |
| 20:23.38 | Insanity5 | -rwxr-xr-x 1 root root 126616 Aug 19 21:32 /usr/local/bin/mpg123 |
| 20:23.46 | h3x0r | it sounds great to me |
| 20:23.51 | h3x0r | oh nevermind |
| 20:24.00 | Insanity5 | ? |
| 20:24.05 | h3x0r | i think its just your song |
| 20:24.11 | Insanity5 | Sounds fine via xlite |
| 20:24.15 | Insanity5 | dialign teh extension of the song |
| 20:24.25 | *** join/#asterisk MBD (n=YZY@213.212.193.204) |
| 20:26.21 | h3x0r | keep in mind its resampled to 8khz |
| 20:26.36 | h3x0r | im pretty sure that has nothing to do with zaptel though |
| 20:27.36 | Insanity5 | Why is it resampled to 8khz? |
| 20:27.39 | Insanity5 | And note when using xlite? |
| 20:27.45 | Insanity5 | not |
| 20:28.09 | h3x0r | because thats what a phone line is |
| 20:28.29 | Insanity5 | Over xlite w/ g711a it sounds like 50 times better. |
| 20:28.44 | Insanity5 | the phone link is 8 ms away, and uses g711 |
| 20:28.51 | axelilly | anyone know where any dundi for asterisk documentation is? |
| 20:29.03 | axelilly | I'd like to set it up, but can't find much |
| 20:29.21 | ManxPower | g711 is the same codec the telcos use. |
| 20:29.31 | h3x0r | so |
| 20:29.32 | MRH2 | hi how can i switch of cdr records going into csv files under /var/log/asterisk...i am loging to postgresql so I don;t need these |
| 20:29.46 | h3x0r | g711a or u? |
| 20:29.49 | h3x0r | are you in the US? |
| 20:29.57 | h3x0r | maybe thats what the problem is |
| 20:30.01 | axelilly | anybody using dundi? |
| 20:30.05 | h3x0r | it should be u for ulaw |
| 20:30.09 | h3x0r | if you are in the us, canada, japan |
| 20:30.28 | h3x0r | if you mismatch a and u it will definatelty sound like crap like that |
| 20:30.33 | *** join/#asterisk websae2k (i=websae@207-118-143-134.dyn.centurytel.net) |
| 20:31.03 | Insanity5 | I was using a because it is generally better quality |
| 20:31.11 | h3x0r | what is |
| 20:31.14 | Insanity5 | the sound |
| 20:31.16 | Insanity5 | over u |
| 20:31.18 | h3x0r | A or U dude |
| 20:31.22 | Insanity5 | g711a |
| 20:31.24 | h3x0r | you should not use a |
| 20:31.27 | Insanity5 | g711a to the provider |
| 20:31.29 | tzafrir_laptop | is there any decent mirror for downloading spandsp? |
| 20:31.29 | h3x0r | unless you are in europe or something |
| 20:31.38 | h3x0r | the provider should use u if they are in the us, canada, japan |
| 20:31.41 | tzafrir_laptop | soft-switch.org is down as usual |
| 20:31.43 | Insanity5 | h3x0r - They let me set it. |
| 20:31.45 | ManxPower | Insanity5: ulaw and alaw are similar quality. |
| 20:31.52 | h3x0r | Well where are they at |
| 20:31.55 | Insanity5 | ManxPower - a is a tad bit wider =) |
| 20:32.03 | Insanity5 | h3x0r - detrioit, mi |
| 20:32.08 | Insanity5 | nufone |
| 20:32.12 | axelilly | anyone use dundi? |
| 20:32.14 | h3x0r | ulaw and alaw are identical, they are just different ways to deal with sound encoding |
| 20:32.18 | h3x0r | use ulaw then |
| 20:32.27 | h3x0r | nufone is probably talking ulaw and you are talking alaw and thats why it sounds funny |
| 20:32.32 | ManxPower | Asterisk can have significant issues transcoding between alaw and ulaw, at least that's what people report. |
| 20:32.42 | h3x0r | exactly |
| 20:32.55 | h3x0r | and he is causing it to transcode twice |
| 20:33.03 | tzafrir_laptop | MRH2, asterisk basicaly uses all the logging modules available. So simply remove the ones you don't need |
| 20:33.17 | Insanity5 | Does asterisk pass off the straight ulaw from the pri? |
| 20:33.20 | MRH2 | cool, thanks |
| 20:33.25 | Insanity5 | So it doesn't reincode (like a fx0)? |
| 20:33.35 | axelilly | anyone use dundi with asterisk? |
| 20:33.38 | h3x0r | T1s are encoded in ulaw |
| 20:33.42 | h3x0r | E1s are encoded in alaw |
| 20:34.03 | h3x0r | you can voip with either and the point is to keep from having to transcode |
| 20:34.05 | tzafrir_laptop | ManxPower, isn't it basically the same issue as de-encoding to slinear? |
| 20:34.06 | *** join/#asterisk CaNaBiS` (n=CaN@63.144.16.242) |
| 20:34.12 | *** join/#asterisk telephoneman (n=chatzill@64.207.35.66) |
| 20:34.19 | Insanity5 | h3x0r - but asterisk can pass off the ulaw from the pri (T-1) withou trasncoding, right? |
| 20:34.30 | h3x0r | Yes of course it can but you need to set ulaw not alaw |
| 20:34.34 | h3x0r | for the voip side |
| 20:34.37 | Insanity5 | ok |
| 20:34.48 | *** join/#asterisk cpm_ (n=Chip@orthanc.avwashington.com) |
| 20:34.52 | fourcheeze | can I call a function in an AGI script from within *? |
| 20:35.07 | fourcheeze | rather than calling the whole script |
| 20:35.12 | fourcheeze | trying to get a modular feel going |
| 20:35.12 | tzafrir_laptop | fourcheeze, no. AGI runs in its own process |
| 20:35.31 | fourcheeze | ok |
| 20:35.37 | axelilly | I'm trying to set up dundi, does anyone know anything about it? |
| 20:35.40 | h3x0r | but if you are using nufone you should just use gsm |
| 20:35.44 | Insanity5 | haxor - does it sound any better now? I only havea cell phone to test it on |
| 20:35.45 | *** part/#asterisk cpm_ (n=Chip@orthanc.avwashington.com) |
| 20:35.55 | *** join/#asterisk Uther_P (n=uther_p@66.180.120.82) |
| 20:35.56 | telephoneman | is anyone using the Digium wrapper for Cepstral tts? |
| 20:36.08 | *** join/#asterisk zoa (n=k@213.91.216.136) |
| 20:36.09 | h3x0r | no. did you reload chan_iax2.so or whatever |
| 20:36.15 | Insanity5 | I just did a "reload" |
| 20:36.26 | h3x0r | maybe you need to do a restart |
| 20:36.42 | h3x0r | i donno if it ignores that parameter on reload or not some of them it does |
| 20:36.46 | Insanity5 | done |
| 20:37.12 | h3x0r | iax2 show channels |
| 20:37.16 | ManxPower | unload chan_iax2.so and load chan_iax2.so |
| 20:37.31 | h3x0r | that is another way to do it. heh |
| 20:37.44 | Insanity5 | ok try now |
| 20:37.51 | tzafrir_laptop | anybody with a spandsp for me to download? |
| 20:38.04 | h3x0r | i think its your song |
| 20:38.05 | h3x0r | heh |
| 20:38.09 | Insanity5 | IAX2/NuFone@198.22.6 198.22.67.70 NuFone 00003/00587 00006/00005 00010ms 0003ms 0000ms ulaw |
| 20:38.37 | Insanity5 | It says ulaw |
| 20:38.42 | h3x0r | yep |
| 20:38.45 | h3x0r | try a diff song dude |
| 20:39.03 | Insanity5 | Can you sip direct to it if I get you credentials? |
| 20:39.07 | Insanity5 | give |
| 20:39.12 | Insanity5 | It sounds fine in xlite |
| 20:39.16 | Insanity5 | don't know why |
| 20:39.38 | *** join/#asterisk oe1 (n=oej@apollo.webway.se) |
| 20:40.59 | Insanity5 | exten => 8662492403,2,MP3Player(/var/lib/asterisk/mohmp3/fpm-sunshine.mp3) |
| 20:41.00 | Insanity5 | try now |
| 20:41.02 | Insanity5 | I used the built in mp3 |
| 20:41.09 | *** join/#asterisk dwmw2_gone_ (i=ctrlprox@baythorne.infradead.org) |
| 20:41.45 | *** join/#asterisk SplasPood (i=jwb@ludicrous.paravolve.net) |
| 20:42.04 | spackle | what ever happened to the #996 conference? |
| 20:43.43 | *** join/#asterisk dwmw2_gone_ (i=ctrlprox@baythorne.infradead.org) |
| 20:43.47 | Insanity5 | I used the built in mp3 |
| 20:43.53 | Insanity5 | Is that you haxor? :) |
| 20:44.01 | h3x0r | yeah |
| 20:44.07 | h3x0r | that one seems ok |
| 20:44.15 | axelilly | anyone know anything about dundi? |
| 20:44.22 | Insanity5 | h3x0r - Is that as good as it gets? |
| 20:44.29 | Insanity5 | The stuff off the pri at work is pretty damn good :) |
| 20:44.43 | h3x0r | maybe nufone sucks |
| 20:44.43 | h3x0r | heh |
| 20:44.45 | Insanity5 | Some times there's a crackle/snap/pop |
| 20:44.46 | Insanity5 | hehe |
| 20:44.53 | h3x0r | whats your traceroute look like |
| 20:45.15 | *** join/#asterisk SexyKen (n=ksandell@c-24-5-129-114.hsd1.ca.comcast.net) |
| 20:45.25 | ManxPower | what happens if you Playback a file and not use the unsupported and crappy built in mp3player? |
| 20:45.28 | SexyKen | Anyone know what "480: Temporary unavialable" means? |
| 20:45.34 | SexyKen | The phone wont register and gets that |
| 20:45.48 | *** join/#asterisk dwmw2_gone_ (i=ctrlprox@baythorne.infradead.org) |
| 20:45.50 | Insanity5 | h3x0r - http://pastebin.com/346271 |
| 20:46.47 | MasterYoda | topic needs updating. beta.asterisk.org is live |
| 20:47.11 | h3x0r | that isnt bad |
| 20:47.16 | Insanity5 | h3x0r - And this is the dialout server (they'er different) http://pastebin.com/346274 |
| 20:47.45 | Insanity5 | So the timer won't help? |
| 20:47.52 | *** join/#asterisk dwmw2_gone_ (i=ctrlprox@baythorne.infradead.org) |
| 20:48.22 | rg1_ | bkw_ - you out there my man? |
| 20:48.26 | h3x0r | you have trunk=no in iax.conf right |
| 20:48.29 | h3x0r | for nufone |
| 20:48.55 | Insanity5 | h3x0r - no I do not |
| 20:49.14 | Insanity5 | context, secret, allow, disallow is all. |
| 20:49.21 | rg1_ | I have a PRI-anyone out there that might be able to help me figure out why I can get incoming calls but I get an all-circuits busy message from asterisk when i try to dial out? |
| 20:49.24 | Insanity5 | Just what they e-mailed me. |
| 20:49.30 | *** join/#asterisk cpm (n=Chip@orthanc.avwashington.com) |
| 20:49.35 | h3x0r | one thing before you change anything |
| 20:49.37 | h3x0r | iax2 show peers |
| 20:49.41 | *** join/#asterisk jaxxan (n=jaxxan@202.70.125.66) |
| 20:49.45 | h3x0r | doe sit have a (T) on the line with nufone |
| 20:49.48 | spackle | rg1_: is the line direct to the the telco? |
| 20:49.58 | *** join/#asterisk dwmw2_gone_ (i=ctrlprox@baythorne.infradead.org) |
| 20:50.00 | Insanity5 | NuFone 66.225.202.72 (S) 255.255.255.255 4569 Unmonitored |
| 20:50.06 | Insanity5 | nope |
| 20:50.21 | jaxxan | hey guys |
| 20:50.24 | *** part/#asterisk cpm (n=Chip@pdpc/supporter/sustaining/cpm) |
| 20:50.37 | h3x0r | ok |
| 20:50.46 | *** join/#asterisk jesse_133 (n=jdandr2@miro.voltaiccommerce.com) |
| 20:51.03 | h3x0r | wel it would be a better idea to test it with Playback instead of moh |
| 20:51.14 | Insanity5 | Why and what does trunk do? |
| 20:51.15 | rg1_ | spackle - its a PRI/T1 line coming out of my ADIT 600 channel bank |
| 20:51.26 | *** join/#asterisk Blissex (n=Blissex@82-69-39-138.dsl.in-addr.zen.co.uk) |
| 20:51.26 | rg1_ | i have 7 b channels configured as pri_cep |
| 20:51.30 | rg1_ | pri_cpe |
| 20:51.39 | ManxPower | Insanity5: IAX2 Trunking saves bandwidth when you have more than 2 calls going between the same two Asterisk servers. |
| 20:51.49 | h3x0r | why are you doing pri for a channel bank |
| 20:51.56 | h3x0r | use loop start... well kewl start |
| 20:52.03 | ManxPower | What channel bank supports PRI anyway? |
| 20:52.07 | jesse_133 | one know a softphone that allows you to create dial:8885551212 URI's in html <a href="dial:5551212">call me</a> which when clicked cause the sip phone to call... I'm thinking xten/sjphone? |
| 20:52.09 | h3x0r | they do but its just dumb |
| 20:52.17 | rg1_ | again, they are coming in - i think it might have something to do with the context perhaps of an extension dialing out - not sure where to start looking though |
| 20:52.26 | h3x0r | The reason CLECs configure PRIs for Channel Banks is a loop hole to get around taxes |
| 20:52.26 | *** join/#asterisk nicox (n=nicox@h082218027030.host.wavenet.at) |
| 20:52.27 | *** join/#asterisk pablasso (n=pablasso@dsl-201-128-242-27.prod-infinitum.com.mx) |
| 20:52.28 | rg1_ | manxpower - ADIT 600 |
| 20:52.31 | pablasso | hey there |
| 20:52.34 | ManxPower | h3x0r: like driving a ferarri on a dirt road. |
| 20:52.36 | *** join/#asterisk Insanity4 (i=Insanity@ip68-105-216-37.tl.dl.cox.net) |
| 20:52.39 | nicox | hy guys |
| 20:52.39 | h3x0r | Dont use PRI from Channel Bank to Asterisk |
| 20:52.40 | Insanity4 | sorry |
| 20:52.42 | Insanity4 | interent just took a dump |
| 20:52.44 | Insanity4 | internet |
| 20:52.51 | ManxPower | Insanity5: IAX2 Trunking saves bandwidth when you have more than 2 calls going between the same two Asterisk servers. |
| 20:52.52 | Insanity4 | h3x0r - what did you say after I asked why test it with playback instead of moh? |
| 20:53.05 | Insanity4 | ManxPower - At the cost of? |
| 20:53.06 | h3x0r | insanity, but to use trunking you need ztdummy or a zaptel device |
| 20:53.08 | h3x0r | that is why i was asking |
| 20:53.15 | Insanity4 | Ahhh |
| 20:53.19 | Insanity4 | nope... don't have them. |
| 20:53.26 | Insanity4 | Does it still "trunk" even if only one circuit is open? |
| 20:53.36 | rg1_ | h3x0r - it's basically a "pass-thru" - not really treating the ADIT as a channel bank - the ADIT came from XO Communications so we can split off some of the B channels and use them as internet/data |
| 20:53.38 | h3x0r | Yes, if you had trunk=yes |
| 20:53.47 | Insanity4 | I didn't have it set either way h3xor |
| 20:53.54 | h3x0r | rg1 ok so you mean you are taking in PRI as add-drop |
| 20:54.01 | nicox | did anybody use a asterisk to manage about 50 000 users? |
| 20:55.21 | Insanity4 | h3x0r - Might the trunking issue by why it soudns so damn good for the sip clientms? |
| 20:55.31 | fourcheeze | is it just me or is there very little documentation on using python for AGI? |
| 20:55.39 | spackle | h3x0r, if the fractional t1 is configured that way is there still a 'D' channel? |
| 20:55.40 | rg1_ | h3x04 - not sure what you mean |
| 20:55.44 | h3x0r | insanity, No because there is no (T) |
| 20:55.47 | fourcheeze | simple stuff like how to pass a variable to my script |
| 20:55.54 | *** join/#asterisk dwmw2_gone (i=ctrlprox@baythorne.infradead.org) |
| 20:56.01 | Insanity4 | h3x0r - Meaning, it's not trunking anyways. |
| 20:56.15 | h3x0r | spackle you can not add drop lines out of a PRI because they cant share the D cahnnel |
| 20:56.22 | h3x0r | you can do that with robbed bit signalling |
| 20:57.01 | h3x0r | rg1_: does it go from XO to the channel bank and then the "CPE" or "Add Drop" port to Asterisk? |
| 20:57.03 | MRH2 | anyone know of any way to get the sip extension an agent is on given the agent number, if not does that fall under 'dialplan functions' on mantis? |
| 20:57.14 | spackle | h3x0r, thought that seemed odd. Sounds more like a context issue. |
| 20:57.47 | *** join/#asterisk jets (n=b@guardian.pmt.org) |
| 20:57.54 | spackle | rg1_ what context do you have these lines defined in? |
| 20:57.54 | h3x0r | spackle: hes talking about a transport thing |
| 20:58.09 | ManxPower | h3x0r: our carrier will do PRI, CT1, and data channels all on the same PRI. They put in an Adtran TA850 to split things out before it gets to our equipment. |
| 20:58.16 | h3x0r | the adit has a add/drop port which measn you can specify DS0s to pass through to another port |
| 20:58.31 | nicox | did anybody know if there is a limitation of entrys for asterisk to use extensions.conf from y mysql database? |
| 20:58.44 | h3x0r | ManxPower: yes but most people dont order pri + channelized on the same T |
| 20:59.01 | Insanity4 | brb |
| 20:59.04 | ManxPower | h3x0r: True. We do, however. |
| 20:59.26 | ManxPower | usually only 1 or 2 CT1 channels for fax/modem/alarm |
| 20:59.29 | h3x0r | that is just weird :P |
| 20:59.35 | ManxPower | We have had no end of problems faxing via Asterisk |
| 21:00.08 | nicox | did anybody know if there is a limitation of entrys for asterisk to use extensions.conf from y mysql database? |
| 21:00.10 | neron | Can someone help me answer a few question about Asterisk and telephone is general, please? |
| 21:00.13 | ManxPower | I think it's because we have to lower the txgain so much to eliminate echo. |
| 21:01.13 | nicox | only ask |
| 21:01.20 | neron | ok |
| 21:01.21 | *** join/#asterisk santiago (n=santiago@63.245.87.180) |
| 21:02.20 | MRH2 | any one have an opinion on the 411P is the echo cancellation daughter board worth the extra $$$ |
| 21:02.55 | JerJer | yes |
| 21:03.04 | neron | I currently have a 96 port Panasonic service unit (DBS576) and Replay Plus 6 for voicemail. Can someone tell me if I can use Asterisk to replace the Replay PLus voicemail system? |
| 21:03.42 | spackle | JerJer, did you test the echo can cards before they came out? |
| 21:04.25 | nicox | i think thts not a problem. how muh users do you have to serve? |
| 21:04.26 | *** join/#asterisk harlequin516 (n=sham@dsl01-ppp-4444.fastq.com) |
| 21:04.33 | harlequin516 | msg nickserv identify handy |
| 21:04.40 | drray | hee |
| 21:04.41 | MRH2 | does it solve most of the echo related issues? |
| 21:04.54 | nicox | i thin now you have to change your password |
| 21:05.06 | harlequin516 | Can I use a Digium Wildcard X100P as a modem too? |
| 21:05.18 | *** join/#asterisk cpm (n=Chip@pdpc/supporter/sustaining/cpm) |
| 21:05.19 | *** join/#asterisk mcn (n=mcn@ext-gw.newtoncomputing.co.uk) |
| 21:05.34 | blessen | I am getting this error while registering with my asterisk server |
| 21:05.37 | blessen | res_search: NO result ! |
| 21:05.37 | blessen | SipClient: Sending to '192.168.15.2:5060' |
| 21:05.41 | harlequin516 | <PROTECTED> |
| 21:05.46 | neron | nicox, about 50-70 user in this office |
| 21:05.59 | *** part/#asterisk ManxPower (n=eric@dsl-209-16-67-160.datasync.com) |
| 21:06.01 | blessen | please help |
| 21:06.13 | *** part/#asterisk cpm (n=Chip@pdpc/supporter/sustaining/cpm) |
| 21:06.17 | *** join/#asterisk lilo (i=levin@freenode/staff/pdpc.levin) |
| 21:06.31 | harlequin516 | Anyone know about using this card with TTY devices or Baudot 45.5? |
| 21:07.13 | rg1_ | h3x04 - CPE port to Asterisk I believe |
| 21:07.18 | neron | nicox, but there will more about 30-40 more user with voicemail box only for remote users |
| 21:07.29 | *** join/#asterisk sigmounte (n=sigmount@www.sighq.net) |
| 21:07.48 | rg1_ | anyone have the URL for the wiki for setting up asterisk? |
| 21:07.49 | *** join/#asterisk clive- (n=pirch@rrba-146-97-10.telkomadsl.co.za) |
| 21:08.12 | telephoneman | bkw_: Brian, is there any info about how to run the Cepstral wrapper in Asterisk? |
| 21:08.37 | telephoneman | I have Cepstral working stand-alone |
| 21:08.44 | telephoneman | I have it compiled in Asterisk |
| 21:09.17 | rg1_ | telephoneman - question for you? |
| 21:09.24 | telephoneman | ok |
| 21:09.35 | *** join/#asterisk flewid (n=flewid@CPE000024c43e1c-CM000f9fac6da2.cpe.net.cable.rogers.com) |
| 21:09.36 | flewid | sup |
| 21:09.39 | rg1_ | you ever work with PRI's? |
| 21:09.43 | h3x0r | rg1 |
| 21:09.46 | telephoneman | yes |
| 21:09.54 | h3x0r | are you trying to use some b channels on asterisk and others on the channel bank analog lines |
| 21:10.10 | clive- | does anyone here use monit with asterisk ? |
| 21:10.17 | flewid | quick question - i have two asterisk servers, one on a PRI, about 3000 miles away, and one locally to my offices. |
| 21:10.20 | rg1_ | i have a PRI going into my digium 4-port card - looks to be working |
| 21:10.35 | flewid | I've setup the same extensions on each (6103 for instance), and using the polycom ip500 phones |
| 21:10.38 | rg1_ | just tyring to figure out in my extensions.conf how i would setup for dial-out context |
| 21:10.43 | telephoneman | T1 or E1? |
| 21:10.43 | flewid | the reason is so we can accept/make local/ld calls from either place |
| 21:10.56 | rg1_ | T! |
| 21:10.58 | rg1_ | T1 |
| 21:11.01 | flewid | now the problem is, on the phone when a message is left at remote location or local location, the MWI indicator only beeps on the top extension |
| 21:11.15 | flewid | so it's confusing all of my users when it says they have a message on one server, but in reality it's on another |
| 21:11.28 | cochi | be happy. mine doesnt notify me at all (zaptel, no stutter) |
| 21:11.30 | flewid | is this just due to the phone getting confused and thinking 6103 and 6103 are the same extensions on the same server? |
| 21:11.35 | spackle | rg1_ what context is your phone in? For testing you can just include it in the same one your dial-out lines are in. |
| 21:11.42 | rg1_ | i don'thtkn the [from-trunck] context is right |
| 21:11.54 | rg1_ | so default? |
| 21:12.00 | telephoneman | in zaptel.conf you have "span=1,1,0,esf,b8zs"? |
| 21:12.11 | *** join/#asterisk dudes (n=dudes@12-215-34-84.client.mchsi.com) |
| 21:12.13 | rg1_ | yes |
| 21:12.19 | Hmmhesays | ~ManxPower |
| 21:12.21 | jbot | well, manxpower is your God. |
| 21:12.28 | Hmmhesays | haha |
| 21:12.29 | flewid | heh |
| 21:12.33 | Hmmhesays | ~seen ManxPower |
| 21:12.44 | jbot | manxpower <n=eric@dsl-209-16-67-160.datasync.com> was last seen on IRC in channel #asterisk, 12m 31s ago, saying: 'I think it's because we have to lower the txgain so much to eliminate echo.'. |
| 21:13.04 | axelilly | is anyone actually using dundi? It seems kind of unorganized |
| 21:13.05 | rg1_ | hey telephoneman, can we do a personal-window conversation on this? or whatever its called |
| 21:13.19 | h3x0r | dundi for phones is like WAIS for the internet |
| 21:13.23 | telephoneman | in zapata.conf you have "signalling=pri_cpe" followed by your channel statements |
| 21:13.28 | devel | morning all. |
| 21:13.29 | spackle | rg1_ I don't know what your setup looks like but probably. inclue -> from-trunk right beneath [default] |
| 21:13.38 | rg1_ | yes |
| 21:13.56 | spackle | rg1_ did you restart asterisk or do a reload? |
| 21:13.58 | devel | on occasion my voicemail audio is just "dead air" rather than the message. is this a known issue? |
| 21:14.21 | *** join/#asterisk dwildes (n=dwildes@209.164.237.195) |
| 21:14.30 | *** join/#asterisk jwolfe37 (n=jwolfe37@h-67-103-108-202.cmbrmaor.covad.net) |
| 21:14.36 | axelilly | h3x0r: but it makes no real sense |
| 21:14.44 | rg1_ | spackle - yes |
| 21:14.46 | h3x0r | it was a joke |
| 21:14.56 | h3x0r | dundi is useful when you have like |
| 21:15.07 | axelilly | h3x0r: for instance, if I wanted to use the dundi routing information, there is no clear cut information on how to do that. |
| 21:15.09 | h3x0r | well consider it to be like BGP for extensions |
| 21:15.24 | pc4 | h3x0r - Ok, I'm back. (866 # guy with crappy music quality) |
| 21:15.36 | spackle | rg1_ and you are still getting the same results - can dial in, but not dial out? What does your dial out extension look like (you kow about pastebin.ca?) |
| 21:15.37 | h3x0r | heh |
| 21:15.48 | pc4 | h3x0r - Any other ideas on what might be causing it? |
| 21:15.51 | axelilly | h3x0r: ok, but how do you even get that routing table? |
| 21:16.06 | drray | this cisco 7960 is annoying me, I can call it via my PSTN lines, my sip lines, my iaxy lines.. I just can't reach it via the zap lines... |
| 21:16.20 | h3x0r | axelilly: the point is when you have a cluster of asterisk boxes and extensions registered to different ones |
| 21:16.20 | h3x0r | it can find the other box to send it to |
| 21:16.26 | h3x0r | its also useful to do on-net call routing |
| 21:16.43 | h3x0r | so if someone dials another person that would normally go to an outside line it would redirect it internally |
| 21:17.58 | axelilly | h3x0r: yes, but I heard that there was a dundi network setup to allow for tail end hop off in order to eliminate toll calls |
| 21:18.03 | Delta34 | does anybody know what this warning means? |
| 21:18.04 | Delta34 | Aug 25 14:17:17 WARNING[4536]: channel.c:1449 ast_indicate: Unable to handle indication 3 for 'SIP/4088-dd81' |
| 21:18.22 | *** join/#asterisk JakBeatZ (n=JakBeatZ@trek.tor1.ebit.ca) |
| 21:18.23 | Delta34 | call still works just not sure whats its warning about |
| 21:18.24 | axelilly | Delta34: you have a problem with indication 3 |
| 21:18.34 | Delta34 | in macro? |
| 21:18.46 | axelilly | Delta34: yea, you can disregard it |
| 21:19.30 | pc4 | h3x0r - Any other ideas on what might be causing it (the poor v oice quality) :) |
| 21:19.37 | clive- | does nay one have a file etc/init.d/asterisk ?...Mine is missing, or is that normal ? |
| 21:19.57 | h3x0r | pc4 did you try recording a file and using Playback |
| 21:20.01 | axelilly | h3x0r: so, am I missunderstanding and there is no such network? |
| 21:20.09 | jwolfe37 | I'm having trouble getting qualify=yes (or qualify=anything) to actually send SIP qualify messages, can anyone help? |
| 21:20.27 | JakBeatZ | Anyone know a way to do loops? Like dial an extension for 90 seconds, if no one answers, play a soundfile, the go back to dialing extension for 90 seconds, etc. until answered? |
| 21:20.40 | h3x0r | axelilly: there probably is but i doubt you will get many endpoints |
| 21:21.06 | axelilly | h3x0r: yea, that's how it's looking |
| 21:21.22 | h3x0r | its a concept thing |
| 21:21.28 | h3x0r | thats not what it was really designed for |
| 21:21.34 | pc4 | h3x0r - Can you provide a file? The only ones I have are gsm. Note: mp3player works fine if I sip direct to the * server. |
| 21:21.39 | axelilly | yea, that makes sense |
| 21:21.39 | h3x0r | pc4 |
| 21:21.42 | spackle | JakBeatz: look into goto and priorities on the wiki. In can be done. |
| 21:21.42 | h3x0r | show application Record |
| 21:21.44 | h3x0r | show application Playback |
| 21:22.21 | *** join/#asterisk CaNaBiS (n=CaN@63.144.16.242) |
| 21:22.36 | axelilly | how do you get MOH working? |
| 21:22.51 | axelilly | mine tries to start it but it never works |
| 21:22.52 | JakBeatZ | spackle: would show application goto give me the same info or would I need the wiki |
| 21:23.04 | h3x0r | axelilly: what distro |
| 21:23.19 | axelilly | tao linux |
| 21:23.24 | *** join/#asterisk dsfr (n=dsfr@digium.com) |
| 21:23.28 | h3x0r | ls -l `which mpg123` |
| 21:23.43 | axelilly | NOTICE[15528]: res_musiconhold.c:493 monmp3thread: Request to schedule in the past?!?! |
| 21:23.44 | axelilly | ok |
| 21:24.25 | spackle | Jakbeatz: you can start there (no examples) also see playback and record. |
| 21:24.30 | *** join/#asterisk rg1_ (n=rg1@mail.airlinksystems.com) |
| 21:24.36 | *** join/#asterisk gentooplex (n=thrranee@198.83.126.81) |
| 21:24.46 | spackle | Jakbeatz: and dial too of course. |
| 21:24.54 | gentooplex | hi |
| 21:25.06 | axelilly | I don't have any mpg123 installed |
| 21:25.21 | h3x0r | install the right one |
| 21:25.38 | gentooplex | please I need some quick answers yes no kinda |
| 21:25.38 | jwolfe37 | anyone have experience with qualify=yes? |
| 21:25.40 | axelilly | what is mpg123? |
| 21:25.42 | h3x0r | http://www.voip-info.org/tiki-index.php?page=mpg123 |
| 21:25.44 | rikstahh | is voip-info.org down |
| 21:25.48 | h3x0r | mpg123 version 0.59r is known to work with Asterisk. |
| 21:25.50 | spackle | axelilly: in your asterisk source directory do make mpg123 and then 'make install' and asterisk gets if for you. |
| 21:25.56 | Sedorox | generalhan: yes, no, yes, no, no, no |
| 21:25.56 | Sedorox | :p |
| 21:25.59 | h3x0r | oh shit i know |
| 21:26.03 | h3x0r | hey pc4 |
| 21:26.30 | gentooplex | I am doing my own research on asterisk but I'd like some answers upfront |
| 21:26.33 | rg1_ | telephoneman - you still out there |
| 21:26.37 | h3x0r | pc4: mpg123 -v |
| 21:26.43 | h3x0r | pc4: what does it say the version number is |
| 21:26.43 | telephoneman | yes |
| 21:26.50 | *** join/#asterisk FarrisG (n=jrush@h-68-164-19-170.dllatx37.covad.net) |
| 21:27.14 | *** join/#asterisk jalsot (n=tamas@abacus.eworldcom.hu) |
| 21:27.21 | h3x0r | spackle: i forgot about that |
| 21:27.22 | gentooplex | with asterisk can you call people with regular phones? Cell phones? Can you receive calls from others that use regular phones or cell phones? |
| 21:27.44 | h3x0r | gentooplex: You can only call people that have landlines with asterisk |
| 21:27.45 | FarrisG | Does anyone know why, when I dial 11 digits (9 + external number) with the handset off the hook, the last digit wraps around to the front and screws up the call, but if I dial with the handset down it works fine? |
| 21:27.53 | h3x0r | it dosent support cells yet |
| 21:28.08 | MRH2 | is there any way to stop an agent channel being marked busy/unavailable wheen a call is active |
| 21:28.25 | axelilly | cool! MOH now works |
| 21:28.55 | gentooplex | h3x0r: so there's no way whatsoever to call and receive messages from people with regular phones like Skypes does? |
| 21:29.04 | spackle | axelilly: now you will hate mpg123 and want to investigate using madplay instead. |
| 21:29.22 | h3x0r | gentooplex: Skype got a patent on that, and we're trying to license that technolgy |
| 21:29.34 | h3x0r | <- BOFH |
| 21:29.37 | *** part/#asterisk dwildes (n=dwildes@209.164.237.195) |
| 21:29.45 | Sedorox | h3x0r: very... |
| 21:29.47 | gentooplex | h3x0r: I see... |
| 21:29.58 | axelilly | spackle: why? |
| 21:30.08 | gentooplex | h3x0r: patent??? I thought Asterisk was open source?? GPL thingy... |
| 21:30.17 | gentooplex | isn't a patent against GPL? |
| 21:30.17 | h3x0r | you cant gpl patented stuff |
| 21:30.28 | syle | can you use macro with gotoif? |
| 21:30.28 | gentooplex | h3x0r: k |
| 21:30.31 | Sedorox | gentooplex: all you need is PSTN access, somehow... be it from a VoIP provider... or if you get a card and stick it in your box... and connect the line from your house.. |
| 21:30.31 | blessen | i got a user to register using kphone..but i cannot make any calls from kphone...it gives me error ..Call failed : Not found |
| 21:30.39 | h3x0r | Dude, im totally shitting you... |
| 21:30.51 | h3x0r | of course you can use it with phone lines |
| 21:30.56 | h3x0r | havent you looked at the hardware on digium's site |
| 21:31.11 | axelilly | blessen: check your channels or your extensions.conf |
| 21:31.22 | spackle | axelilly: mpg123 uses more processes, has issues. Details about using madplay are on the wiki. At least it works for now right? Unfortunately you can't read a stream using madplay. |
| 21:31.38 | blessen | this is what i have |
| 21:31.40 | blessen | [from-sip] |
| 21:31.40 | blessen | exten=>1,1,Dial(SIP/kphone,20,tr) |
| 21:31.53 | blessen | in extension.conf |
| 21:31.56 | gentooplex | Sedorox: yeah but that requires an additional account with a Telecom right? So it would be like this cable internet+landline+asterisk=two bills (internet and phone company) right? |
| 21:31.58 | axelilly | ok, is [from-sip] set as the context? |
| 21:32.09 | Sedorox | generalhan: no |
| 21:32.09 | Sedorox | er |
| 21:32.11 | Sedorox | gentooplex: no |
| 21:32.22 | Sedorox | not with telcom anyway... |
| 21:32.27 | h3x0r | You can just not pay anybody like these assholes that switch providers all the time |
| 21:32.43 | Sedorox | you can pay Teliax, and the likes.. and they will give you access to the PSTN.. and if you choose, a number that people can call you with |
| 21:32.57 | spackle | blessen: check your variable, I think you are referencing the variable, not the value it contains. |
| 21:33.01 | gentooplex | h3x0r: rofl but then HOW do you own a phone number if you keep switching? |
| 21:33.12 | blessen | didnt get u |
| 21:33.13 | h3x0r | port it |
| 21:33.13 | h3x0r | heh |
| 21:33.13 | blessen | ? |
| 21:33.18 | rikstahh | is voip-info.org down ? |
| 21:33.39 | rikstahh | i need to get the script to create the CDR database in mysql |
| 21:33.41 | blessen | didnt get u boss: spackle |
| 21:33.46 | h3x0r | rikstahh: Nope, but it is hosted across the street from me |
| 21:33.54 | rikstahh | i can't get to it at all |
| 21:33.57 | rg1_ | anyone - from the CLI> prompt, how do I reload extensions? |
| 21:34.02 | h3x0r | wiltel is probably broken again |
| 21:34.02 | rikstahh | extensinos reload |
| 21:34.06 | rikstahh | extensions reload |
| 21:34.06 | h3x0r | im on the same router as them |
| 21:34.16 | SplasPood | hey, is there any way to convert a nufone IAX account to SIP without bothering JerJer? |
| 21:34.17 | rg1_ | thx |
| 21:34.20 | Sedorox | its working for me |
| 21:34.21 | spackle | blessen: I think you want exten=>1,1,Dial(SIP/kphone,20,tr) |
| 21:34.42 | spackle | blessen: I think you want exten=>1,1,Dial(SIP/${kphone},20,tr) instead. |
| 21:34.43 | *** join/#asterisk supaigtr (n=yurplsl@152.53.16.10) |
| 21:34.46 | blessen | yes..thats the setting i got from net and i am trying it..me not sure how to configure it |
| 21:34.57 | *** part/#asterisk fourcheeze (n=rich@westbury.doilywood.org.uk) |
| 21:35.00 | blessen | ok.. |
| 21:35.30 | *** join/#asterisk tzafrir_laptop (n=tzafrir@bzq-179-75-202.cust.bezeqint.net) |
| 21:35.48 | gentooplex | ok thanks all you for the patience, I am really n000b to all this asterisk business and I was trying to wight the pros and cons between Skype and Asterisk, obviously Asterisk has some serious shortcomings still... |
| 21:35.51 | FarrisG | Has nobody else experienced a problem with the Soundpoint 501 wrapping the last digit of the call to the front? |
| 21:35.58 | blessen | me still getting the same error.... |
| 21:36.02 | riksta | can someone please email me cdr.sql for mysql? |
| 21:36.10 | blessen | shall i show u my configuration : spackle |
| 21:36.12 | clive- | does anyone have a file etc/init.d/asterisk ?...Mine is missing, or is that normal ? |
| 21:36.14 | Sedorox | .. it does have shortcoming? please list? |
| 21:36.46 | spackle | blessen: only if you put it in pastebin.ca ;-) I'll look at it. |
| 21:36.49 | Sedorox | 'cause I haven't seen one.. besides skypebeing extremely easy to use out of the box, persay |
| 21:36.56 | gentooplex | Sedorox: well it doesn't have the capability (YET?) to make phone calls to regular phones and receive phone calls from regular landlines and cell phones |
| 21:37.12 | Sedorox | gentooplex: you talking about different things here |
| 21:37.18 | Sedorox | Asterisk is a Software PBX |
| 21:37.21 | Sedorox | Skype is a service |
| 21:37.32 | Sedorox | to allow asterisk to dial like that.. you need a service |
| 21:37.40 | Sedorox | (tho skype won't work...) |
| 21:37.43 | spackle | clive: check the install docs, there is a 'make init' or something that will create that for you on several distos. |
| 21:37.59 | drray | I'm in the wiki |
| 21:38.12 | clive- | spackle, thanks, I am looking hard:) |
| 21:38.46 | gentooplex | Sedorox: you know I am looking for convergence, it doesn't make sense to have three bills three things skype regular phone asterisk .... So what kinda service is attached or could be attached to ASterisk? |
| 21:38.51 | blessen | here it is boss |
| 21:38.51 | spackle | clive: what distro? |
| 21:38.53 | blessen | http://pastebin.ca/21159 |
| 21:38.58 | blessen | please help me with this... |
| 21:39.23 | Sedorox | any SIP, H323, or IAX service |
| 21:39.27 | supaigtr | Anyone have a working NFAS zapata and zaptel file they could pastebin? |
| 21:39.30 | h3x0r | gentooplex: you are lookijng for the wrong thing. Asterisk is what you use to start your own Skype or replace your PBX at your office |
| 21:39.39 | Sedorox | skype has proprietary stuff in their network... |
| 21:39.44 | blessen | spackle : did u get my message with the link http://pastebin.ca/21159 |
| 21:40.01 | h3x0r | you cant get "free phonelines" anywhere. |
| 21:40.20 | clive- | spackle RH9 |
| 21:40.45 | Sedorox | well you can if you get all your friends to setup asterisk (like) software too :p |
| 21:40.47 | gentooplex | Sedorox: are SIP H323 or IAX expensive? how much per month approc?? |
| 21:41.00 | Sedorox | oi... |
| 21:41.05 | Sedorox | those are protocols... |
| 21:41.08 | gentooplex | Sedorox: I will set up asterisk to as many people as possible |
| 21:41.14 | Sedorox | 0_o |
| 21:41.25 | Sedorox | I wish you luck with that one kimosabi... |
| 21:41.38 | h3x0r | asterisk isnt a client, its a server |
| 21:41.39 | Sedorox | your best bet... |
| 21:41.52 | Sedorox | go read voip-into.org |
| 21:41.55 | h3x0r | you set up an asterisk server and connect voip phones to it if you want to do that |
| 21:42.00 | Sedorox | supaigtr: I probably spelled it wrong :p |
| 21:42.02 | gentooplex | h3x0r: so what's a client for Asterisk server?? |
| 21:42.08 | supaigtr | Looks right to me. |
| 21:42.18 | gentooplex | h3x0r: ok I see... |
| 21:42.19 | h3x0r | xten eyebeam, diaxphone, |
| 21:42.22 | h3x0r | etc etc |
| 21:42.23 | drray | soft phone, zap channel, iax phone, another asterisk server |
| 21:42.39 | gentooplex | Sedorox: sorry, thanks for the patience, I am extremely n000b for asterisk |
| 21:43.02 | Sedorox | :p |
| 21:43.03 | blessen | spackle : Can i mesage you directly . please |
| 21:43.25 | h3x0r | thjeres like 100 different clients or more |
| 21:43.29 | h3x0r | there are hardware voip phones |
| 21:43.30 | Sedorox | I think along with reading the wiki... go catch up on telcom stuff.. and you'll better understand it |
| 21:43.36 | h3x0r | theres hardware to plug in regular phones and lines |
| 21:43.41 | spackle | blessen: sure. |
| 21:44.12 | Katty | what's the name of the package that lets me do "vncviewer ip:port" |
| 21:44.12 | gentooplex | Sedorox: thanks... yesterday I was reading some intro stuff to asterisk |
| 21:44.16 | clive- | spackle are you usre about that "make init" ..cant find much info on that |
| 21:44.26 | Katty | apt-cache search vncviewer does not bring back a plain 'vncviwer' |
| 21:44.29 | spackle | clive: it's make config, should work fine on RH9 |
| 21:44.49 | clive- | spackle, make config will wipe out my other configs afaik |
| 21:45.06 | gentooplex | h3x0r: if a person gains this asterisk knowledge, is there marked out there? How much do companies to have this?? |
| 21:45.07 | *** join/#asterisk _pablasso (n=pablasso@201.138.255.59) |
| 21:45.08 | clive- | oh wait, I am confusing make samples |
| 21:45.13 | h3x0r | Katty: realvnc.com tightvnc.com |
| 21:45.14 | supaigtr | Look familar? pri_dchannel: Ring requested on unconfigured channel 0/1 span 1 |
| 21:45.16 | drray | make samples will kill it, make clean wont |
| 21:45.31 | Sedorox | gentooplex: you really don't need asterisk unless you plan on setting up your own PBX..... and for jobs... it helps as it would be pbx systems.. |
| 21:45.44 | h3x0r | gentooplex: Vonage uses asterisk for voicemail, several other voip providers use only asterisk to run their network |
| 21:45.52 | h3x0r | like nufone, voicepulse, etc. |
| 21:45.55 | Sedorox | but as far as I know.. no one is actually asking for asterisk |
| 21:46.09 | *** part/#asterisk FarrisG (n=jrush@h-68-164-19-170.dllatx37.covad.net) |
| 21:46.16 | gentooplex | h3x0r: I see, so you wouldn't install asterisk manually for a company? |
| 21:46.21 | Katty | h3x0r: i meant the package name for apt-getting purposes. |
| 21:46.23 | h3x0r | but there are other pieces of software that are better at asterisk at some things like SER |
| 21:46.39 | *** join/#asterisk zotz (n=zotz@24.231.36.100) |
| 21:46.50 | h3x0r | Yes, but you need to do a lot on the telecom side of things |
| 21:46.58 | clive- | spackle, thanks...that did the trick |
| 21:46.58 | gentooplex | h3x0r: k |
| 21:47.12 | h3x0r | and its not just plug and play |
| 21:47.17 | h3x0r | a lot of applications you have to code shit for it |
| 21:47.19 | spackle | blessen: spackle at ipmvs dot com |
| 21:47.25 | drray | but it works, once it is up |
| 21:47.26 | gentooplex | h3x0r: that sounds cool, I think the real challenge is to see asterisk being implemented |
| 21:47.32 | h3x0r | if you arent a software developer i wouldnt bother too much |
| 21:47.52 | *** join/#asterisk kpettit (n=keith@user-0cet2fu.cable.mindspring.com) |
| 21:47.52 | blessen | spackle: what?? |
| 21:48.03 | gentooplex | h3x0r: mm... developer? do you mean is not cut and paste?? |
| 21:48.13 | h3x0r | simple configurations are easy |
| 21:48.18 | h3x0r | but what companies ask you to make it do |
| 21:48.18 | gentooplex | borrow a script from here, change the configuration there? |
| 21:48.25 | spackle | blessen: email me here: spackle at ipmvs dot com |
| 21:48.35 | blessen | ok.. |
| 21:48.35 | drray | if you can script in perl, you can do asterisk.. if you can read and follow instructions you can do asterisk |
| 21:48.35 | gentooplex | h3x0r: yeah that's what I was refering to |
| 21:48.54 | h3x0r | me and my guys are developing a custom inbound call center application and ran into a whole bunch of snags |
| 21:49.09 | gentooplex | h3x0r: yaiks... |
| 21:49.24 | spackle | h3x0r: you have guys? Cool. |
| 21:49.26 | gentooplex | and being more experienced than I am... |
| 21:49.39 | gentooplex | h3x0r: can you hire me...? |
| 21:49.43 | h3x0r | heh |
| 21:49.47 | gentooplex | ;-) |
| 21:50.07 | h3x0r | we dont normally do development projects |
| 21:50.57 | h3x0r | for others anyway |
| 21:50.57 | gentooplex | h3x0r: how about internship? do you have any available positions? |
| 21:50.57 | supaigtr | The wiki has confused me.. Is it 1 to accept timing from the telco or 0 ? |
| 21:50.57 | *** join/#asterisk paulc (n=Paul@216.187.75.190.novuscom.net) |
| 21:51.01 | h3x0r | i own www.carrierone.net |
| 21:51.25 | *** part/#asterisk spackle (n=spackle@209.234.83.19) |
| 21:52.06 | gentooplex | h3x0r: I see fiber optic cables on the pic |
| 21:53.22 | puowvip | boring |
| 21:53.56 | *** join/#asterisk jalsot (n=tamas@abacus.eworldcom.hu) |
| 21:54.16 | ai-a | anyone point me to the website selling cheap cards 1-4 port analogue cards supported by asterisk. |
| 21:54.24 | ai-a | UK £ |
| 21:55.07 | *** join/#asterisk meppl (n=mephisto@p54AAEF6B.dip.t-dialin.net) |
| 21:55.32 | gentooplex | cool |
| 21:56.57 | file[laptop] | paulc: poke |
| 21:57.01 | gentooplex | linux needs now, open cell phone, open satellite, open TV, open all... |
| 21:57.14 | paulc | file: poke yerself :-p |
| 21:57.23 | file[laptop] | I do that at night mmmkthx |
| 21:57.24 | supaigtr | Man this NFAS span, logical span, is confusing me.... |
| 21:58.17 | *** join/#asterisk shmaltz (n=chatzill@69.28.255.210) |
| 22:01.27 | harlequin516 | quit |
| 22:01.45 | syle | GotoIf($["${mother}" = "bitch"]?macro-fucking-off^${asswipe},s,1:7) |
| 22:01.46 | *** part/#asterisk mkrufky (n=mk@68.160.103.77) |
| 22:01.50 | syle | how do i pass args? |
| 22:01.55 | syle | in gotoif |
| 22:02.56 | syle | it sees ${asswipe} as literal context here |
| 22:06.36 | shmaltz | syle, you have to fucking create an extension that points to the macro with the right arguments, ok fucker |
| 22:08.08 | syle | you mean point macro that points to extension |
| 22:08.56 | syle | err gotoif that points to extension with macro in it, i mean |
| 22:09.06 | DrWho17 | extension that calls the macro |
| 22:09.23 | *** join/#asterisk bjohnson (n=bjohnson@i216-58-15-165.cybersurf.com) |
| 22:09.43 | shmaltz | syle, like this: |
| 22:09.45 | shmaltz | GotoIf($[${VAR1} = foo]?test,s,1) |
| 22:09.46 | shmaltz | exten => s,1,Macro(whatever,arg,arg2,arg3....) |
| 22:10.15 | DrWho17 | ?s,1 |
| 22:10.16 | syle | yeah i know i can do that, just seeing if i can do it in less lines of code, so it ain;t possible i guess |
| 22:10.35 | syle | they should do something like dial |
| 22:10.36 | DrWho17 | yea, well I try that too, but end up being easier to just do it |
| 22:10.53 | syle | dial blah blah M(blah^var1^var2) |
| 22:11.44 | neron | Hi |
| 22:12.35 | neron | Does Asterisk work with existing PBX units and add voicemail? |
| 22:13.05 | *** join/#asterisk ChzDhippt (n=noway@host-198-149.baldwincountyinternet.com) |
| 22:13.11 | *** join/#asterisk Lathos42 (n=Lathos42@68.77.108.51) |
| 22:13.14 | ChzDhippt | Hi all. |
| 22:13.22 | neron | anyone? |
| 22:13.30 | Lathos42 | Hello again |
| 22:13.40 | DrWho17 | neron: it can, I use it with our pbx |
| 22:14.00 | *** join/#asterisk atmel (n=vlad@ruxi.dynamic.ucsd.edu) |
| 22:14.07 | DrWho17 | just program the pbx call routes to send out T1 channels, in our case |
| 22:15.01 | neron | DrWho17, I want/need/must replace our old Replay Plus voicemail system soon.... Do you think Asterick can do the job? |
| 22:15.28 | ChzDhippt | Can anyone tell me how to make dialogics work with asterisk? I can't find any documents describing the process.. I have asterisk working, and I have the Intel drivers working, but dont see anything to connect them. |
| 22:15.49 | *** join/#asterisk thrraneern (n=thrranee@198.83.126.81) |
| 22:16.15 | Supaplex | if it were only as simple as building a lego fortress |
| 22:17.25 | RaYmAn-Bx | aww, be nice and share, Nugget! |
| 22:17.31 | Ariel_ | argh I wish the power would either shut down completly or just stay on. We are getting lights on 1 minut off 2 minutes, on 2 minute then back off. argh can't this stupid storm make it's mind. |
| 22:17.45 | Nugget | Ida can send us more! |
| 22:17.59 | RaYmAn-Bx | :) |
| 22:18.15 | DrWho17 | neron: sure, it can do the job, it's voicemail is very good |
| 22:18.15 | RaYmAn-Bx | technically she's in UK atm, but otherwise, sure |
| 22:18.20 | Ariel_ | ChzDhippt, dialogic is great to sell on ebay and get your self a real board from digium. |
| 22:18.30 | neron | DrWho17, brb |
| 22:18.33 | DrWho17 | you will just need to tell your pbx to go to asterisk for voicemail |
| 22:18.52 | DrWho17 | mine can, I imagine the mileage will vary amongst other vendors |
| 22:19.10 | ChzDhippt | Well, I have other uses for the dialogic board, but it says on Asterisks site that the card is supported. |
| 22:19.26 | devel | also, is there any support for CAMA trunks on the analog zap card? |
| 22:20.19 | paulc | What kinda PBX we talking anyway? |
| 22:20.36 | DrWho17 | he didn't say |
| 22:21.05 | gooagle | will rxgain/txgain have any effect on a T-1 PRI echo? |
| 22:21.24 | ChzDhippt | I am looking to run asterisk for incoming only on a Dialogic D/240JCT-T1, which the drivers are installed and working. |
| 22:21.56 | Ariel_ | ChzDhippt, good luck |
| 22:22.18 | *** join/#asterisk Mavvie (i=edwin@252-131-222-203.rev.techex.net.au) |
| 22:22.44 | rg1_ | i have the following in my extensions_additional.conf.......................... |
| 22:22.46 | rg1_ | [outrt-001-9_outside] |
| 22:22.46 | rg1_ | include => outrt-001-9_outside-custom |
| 22:22.47 | rg1_ | exten => _9.,1,Macro(dialout-trunk,1,${EXTEN:1}) |
| 22:22.47 | rg1_ | exten => _9.,2,Macro(outisbusy) ; No available circuits |
| 22:22.47 | Ariel_ | ChzDhippt, you know you can get more for that board on ebay then what it will cost you for a te110p |
| 22:23.00 | rg1_ | can someone tell me what "group" this will call on? |
| 22:23.30 | neron | DrWho17, are you doing any VoIP with Asterisk? |
| 22:23.35 | Ariel_ | rg1_, it's going to your dialout-trunk |
| 22:23.41 | rg1_ | and how do you read/interpret ${EXTEN:1} ? |
| 22:23.51 | rg1_ | ariel - the macro |
| 22:24.01 | Ariel_ | exten is the number you dialed the :1 meams remove the 9 |
| 22:24.06 | Ariel_ | or first digit |
| 22:24.21 | ChzDhippt | So is the answer that the card is not actually supported in asterisk, or that you just don't know how to do it. |
| 22:24.33 | Ariel_ | rg1_, yes you need to look at the macro dialout-trunk |
| 22:24.33 | *** part/#asterisk jesse_133 (n=jdandr2@miro.voltaiccommerce.com) |
| 22:24.37 | rg1_ | ariel - ok |
| 22:24.42 | rg1_ | i'm looking at that right now |
| 22:24.49 | rg1_ | and i'm looking at the log |
| 22:25.02 | Ariel_ | ChzDhippt, not supported for free you need drivers that cost money and only works in 1/2 duplex mode. |
| 22:25.05 | *** join/#asterisk veteran (n=veteran@cpe-66-25-30-132.houston.res.rr.com) |
| 22:25.53 | ChzDhippt | 1/2 duplex would be fine, since it will be VM only. But the drivers costing money is kinda a bummer. |
| 22:26.14 | Ariel_ | ChzDhippt, yes it is |
| 22:26.44 | *** join/#asterisk meppl (n=mephisto@p54AAEF6B.dip.t-dialin.net) |
| 22:26.49 | Ariel_ | ChzDhippt, that card is used allot for voicemail systems |
| 22:27.16 | neron | DrWho17, are you still there? |
| 22:27.19 | ChzDhippt | Which is what I intend to use it for also |
| 22:27.42 | ChzDhippt | hmm, no mention of the dialogic drivers on digiums website, nor any phone number to contact them |
| 22:27.43 | naturalblue | i rebuilt my asteriskathome system and when i try to login using the defaults it wont let me |
| 22:27.45 | file[laptop] | ChzDhippt: you have to talk to Digium |
| 22:28.04 | naturalblue | anyone know how to get in |
| 22:29.10 | telephoneman | I'm trying to compile res_cepstral.c and I get "skipping incompatible /usr/swift/libswift.so when searching for -lswift". Is there anything I can do? x86_64 FC3 |
| 22:29.12 | paulc | ChzDhippt: Honestly - for the amount of time, effort, energy, and hassle involved, you'd be WAY better off ditching the Dialogic card and going with a Digium one |
| 22:29.22 | paulc | I love Dialogic to bits, use it all the time.. but it's not always the best solution to a problem |
| 22:29.39 | neron | I have a Replay Plus voicemail system with (2) DSP 4-port PCI card on the computer. How can Asterick can take over this? |
| 22:30.06 | paulc | neron: a 4 port TDM card. Or a bunch of Sipura SPA-3000s, depending on your requirements |
| 22:30.22 | paulc | feature wise, asterisk voicemail is probably not as rich as Replay Plus |
| 22:30.37 | *** join/#asterisk Seba_soy (n=s@64.76.126.29) |
| 22:30.39 | neron | paulc, what! really? |
| 22:30.49 | Seba_soy | HELLO EVERYBODY |
| 22:30.56 | Seba_soy | I am from Argentina |
| 22:31.03 | zoa | hello argentina |
| 22:31.04 | jero | hello hello |
| 22:31.04 | syle | i;m from my mom |
| 22:31.06 | syle | whats up |
| 22:31.06 | Seba_soy | someone speak spanish? |
| 22:31.11 | Seba_soy | :) |
| 22:31.11 | paulc | someone does, sure |
| 22:31.14 | zoa | please give us the points for the eurosongfestival! |
| 22:31.15 | Seba_soy | yes |
| 22:31.24 | Seba_soy | well, I am here looking for some help!! |
| 22:31.27 | Seba_soy | ;) |
| 22:31.30 | zoa | your english is just fine! |
| 22:31.40 | paulc | neron: what do you want? basic voicemail? sure - that works.. but advanced thigns like distribution lists, return receipt etc aren't in Comedian Mail.. yet.. |
| 22:31.42 | syle | hey how do you say lick my balls in spanish? |
| 22:31.54 | Seba_soy | I did my own CALLINGCARD application, it is written in PHP, and runs in DEADAGI |
| 22:32.16 | Seba_soy | but, BUT, I am having an strange situation |
| 22:32.23 | jero | ah AH |
| 22:32.50 | neron | paulc, so you're saying Asterisk is not strong with voicemail? |
| 22:33.00 | Seba_soy | MEMORY USAGE in my LINUX BOS, increases until it reaches 100%, where asterisk can't do nothing, until I restart it |
| 22:33.16 | jero | lol |
| 22:33.19 | paulc | neron: I'm saying that Comedian Mail is pretty good as a basic voicemail system, but it lacks a lot of the features you get in bigger/"voicemail only" type systems |
| 22:33.22 | Seba_soy | look at this... |
| 22:33.28 | syle | wtf neron, you can;t do 1/10th of the shit with skype you can with asterisk |
| 22:33.36 | syle | not sure what drugs your on |
| 22:33.48 | Seba_soy | <PROTECTED> |
| 22:33.48 | zoa | high^h^h five^h^hle |
| 22:33.48 | Seba_soy | 24562 root 16 0 417m 376m 10m S 9.7 75.0 28:52.69 asterisk |
| 22:34.00 | neron | paulc, I see.. thank you |
| 22:34.01 | Seba_soy | :(:(:(:(::( |
| 22:34.01 | ChzDhippt | Apparently the dialogic drivers are in Alpha under NDA, so if I get it working, I wont be able to tell anyone how. :) |
| 22:34.05 | zoa | somebody interested in chan_skype ? |
| 22:34.21 | Seba_soy | somebody can help me with this? |
| 22:34.50 | syle | zoa you coded one? |
| 22:35.04 | zoa | no, i was looking to make one |
| 22:35.05 | *** join/#asterisk dumbhack (i=dumbhack@c-71-57-23-69.hsd1.il.comcast.net) |
| 22:35.10 | zoa | but cost would be 25.000$ |
| 22:35.18 | file[laptop] | 25 dollars? ok |
| 22:35.20 | syle | 25 bucks? |
| 22:35.20 | zoa | so unless someone will give me that, there will be none :( |
| 22:35.22 | syle | ok |
| 22:35.48 | file[laptop] | hop to it boy |
| 22:35.51 | zoa | and if they change the protocol, it had to be rdone from scratch :/ |
| 22:36.06 | dumbhack | What are roughly the minimum x86 system requirements for a minimal Asterisk installation handling a couple SIP devices? |
| 22:36.18 | riksta | 486 dx4 100 |
| 22:36.21 | *** join/#asterisk Kraven (n=Kraven@p548B79CC.dip.t-dialin.net) |
| 22:36.21 | zoa | 486 |
| 22:36.22 | ChzDhippt | dumbhack: probably very low |
| 22:36.29 | zoa | or less |
| 22:36.31 | jero | forget about moh |
| 22:36.33 | jero | lol |
| 22:36.35 | Kraven | yeah: http://www.monstersgame.net/?ac=vid&vid=2003916 |
| 22:36.39 | Seba_soy | <PROTECTED> |
| 22:36.39 | Seba_soy | 24562 root 16 0 417m 376m 10m S 9.7 75.0 28:52.69 asterisk |
| 22:36.44 | Delta34 | anybody got time to help me with my zaptel config, i changed it from pri to t1 |
| 22:36.45 | Seba_soy | look at %MEM |
| 22:36.55 | Delta34 | it was working under pri |
| 22:37.02 | Delta34 | now not workign under t1 |
| 22:37.12 | rg1_ | ariel - here's a line in my extensions.conf for the dialout-trunk that does the dialing |
| 22:37.14 | rg1_ | <PROTECTED> |
| 22:37.28 | rg1_ | Executing Dial("SIP/108-2e71", "ZAP/g0/4269931") in new stack |
| 22:37.33 | devel | Delta34, you're sure all the signaling is right? |
| 22:37.42 | rg1_ | that is the log showing it is dialing out on ZAP/g0 |
| 22:37.51 | *** join/#asterisk baloo_ (n=b@84-73-93-233.dclient.hispeed.ch) |
| 22:37.54 | Delta34 | i believe so, i amd using fxo_ls |
| 22:37.55 | rg1_ | WHERE is it getting the ZAP/g0 from ? |
| 22:38.09 | Delta34 | the telco side is fxs_ls |
| 22:38.30 | Delta34 | using d4,ami |
| 22:38.32 | *** part/#asterisk baloo_ (n=b@84-73-93-233.dclient.hispeed.ch) |
| 22:38.56 | Seba_soy | :( |
| 22:39.10 | Seba_soy | who had some problem as me? |
| 22:39.21 | Seba_soy | same problem |
| 22:39.21 | *** join/#asterisk criptos (n=criptos@201.145.229.183) |
| 22:39.37 | *** part/#asterisk SuperBawlz (i=user@72.22.41.86) |
| 22:39.44 | criptos | hello... |
| 22:39.53 | rg1_ | Executing Macro("SIP/108-2e71", "dialout-trunk|1|4269931") in new stack |
| 22:40.01 | rg1_ | can anyone help interpret something here? |
| 22:40.12 | criptos | I`m starting to toss my tdp400p card.. :( I have issues with random hangups! |
| 22:40.17 | rg1_ | is there somewhere that trunk#s are mapped to zap channels |
| 22:41.10 | criptos | rg1_ have you check the top of your extensions.conf for a definition, maybe at globals section for dialout-trunk ? |
| 22:41.14 | Seba_soy | I am not an expert on linux box, I need to know why asteisk is taking a lot of memory... |
| 22:41.20 | devel | Delta34, did you try fxs_ls instead? the analog stuff at least is "backwards" like that i think |
| 22:41.54 | criptos | seba_soy, especify what do you means with "lot" about memory. |
| 22:41.59 | Seba_soy | <PROTECTED> |
| 22:41.59 | Seba_soy | 24562 root 16 0 417m 376m 10m S 9.7 75.0 28:52.69 asterisk |
| 22:42.01 | Delta34 | so if telco is fxs_ls shoudling i be fso? |
| 22:42.01 | Seba_soy | this |
| 22:42.04 | Seba_soy | %MEM |
| 22:42.13 | Seba_soy | and it is increasing each minute |
| 22:42.21 | rg1_ | criptos |
| 22:42.25 | rg1_ | let me see |
| 22:42.26 | devel | Delta34, other than that, i've got nothing else, as we use e&m wink here.... |
| 22:42.34 | Seba_soy | criptos, I means that... |
| 22:42.35 | Delta34 | k |
| 22:42.41 | criptos | seba what? |
| 22:42.44 | criptos | ? |
| 22:42.54 | Seba_soy | criptos, |
| 22:42.55 | Seba_soy | <PROTECTED> |
| 22:42.55 | Seba_soy | 24562 root 16 0 417m 376m 10m S 9.7 75.0 28:52.69 asterisk |
| 22:42.57 | syle | i;lm guessing cause of something you coded with database seba |
| 22:43.09 | Seba_soy | this is output from my TOP command |
| 22:43.22 | criptos | seba_soy use pastebin.ca |
| 22:43.28 | Seba_soy | ? |
| 22:43.30 | Seba_soy | what is that? |
| 22:43.33 | syle | strace -p 24562 |
| 22:43.48 | rg1_ | criptos |
| 22:43.50 | criptos | is a place where you can paste things and is more polite than pasting here... |
| 22:43.50 | Seba_soy | what is pastebin |
| 22:43.58 | criptos | is a website. |
| 22:44.11 | Seba_soy | can you show me how in private chat? |
| 22:44.32 | criptos | Is that an insuniation? No thanks :D |
| 22:44.40 | Seba_soy | nooooo :):):) |
| 22:45.24 | *** join/#asterisk onid (n=ure@unaffiliated/onid) |
| 22:46.34 | *** part/#asterisk Kraven (n=Kraven@p548B79CC.dip.t-dialin.net) |
| 22:46.48 | criptos | So seba, first of all take a loot at pasebin.com :) |
| 22:46.54 | criptos | there u can cut and pase... |
| 22:47.01 | |dennis| | Question: right now we have a nortel pbx box with 3 pots line. The phone are analog nortel instrumetns with the ability to select anyone of the three lines to use and dialout....can asterisk and aan ip phone like gxp2000 be setup so that when line1 is pressed, it willl dial thru line 1 and when line 2 is pressed, it will dial thru line and so on...if yes..how can this be done? is there any documented soure of wheer this has been done? please help... :) |
| 22:47.22 | criptos | and about your memory usage, is not normal, are u using mysql o a very large list of mp3 for music n hold? |
| 22:47.59 | criptos | dennis, that can be achivied using patter matching and channel groups :) |
| 22:48.16 | criptos | For example, I have 6 incoming lines, 3 from one telco, other 3 for other telco... |
| 22:48.22 | *** part/#asterisk onid (n=ure@unaffiliated/onid) |
| 22:49.12 | |dennis| | thast for incoming.....same with outgoing?? |
| 22:49.38 | criptos | group=1 and group=2 respectively, at dialplan I have exten => _9XXXXXX,1,Dial(Zap/g1/${EXTENSION:1}) and exten =>_8XXXXXXX,1,Dial(Zap/g2/...) |
| 22:50.21 | criptos | So, If I dial 933437744 I get out from one of the 3 lines of telco 1, if I dial 833437745 then, I get out from the one of the other 3 lines of the telco2 |
| 22:51.01 | criptos | So, any with experience with HARD disconnection issues with tdm400p cards? |
| 22:52.12 | *** join/#asterisk mino (n=mino@217.172.177.60) |
| 22:53.14 | jero | criptos : voltage issue ? |
| 22:53.31 | *** join/#asterisk MBD (n=YZY@213.212.193.204) [NETSPLIT VICTIM] |
| 22:53.31 | *** join/#asterisk Chuji (i=Chuji@pcp09930052pcs.tulipgrove.tn.nash.comcast.net) [NETSPLIT VICTIM] |
| 22:53.31 | *** join/#asterisk Cherbru_ (i=jgarland@72.36.136.226) [NETSPLIT VICTIM] |
| 22:53.41 | Optic | ok, I got the latest asterisk... how do I turn on the new echo canceller? :) |
| 22:53.41 | *** join/#asterisk jeobjeobjeob (n=jeobjeob@pool-70-111-97-254.nwrk.east.verizon.net) |
| 22:53.44 | jeobjeobjeob | hey |
| 22:53.56 | |dennis| | criptos: oh okie.....but the buttons on the phone..i know it sounds silly "the buttons" but admin will probably ask me whether that can be done before we move to asterisk |
| 22:53.58 | jeobjeobjeob | i have multiple phones |
| 22:54.07 | jeobjeobjeob | and i try dialing from one phone to the other, and it doesn't work |
| 22:54.12 | jeobjeobjeob | i can dial out to the pstn just fine |
| 22:54.27 | jeobjeobjeob | but when i dial within then etwork, it automatically says everyone is busy |
| 22:54.31 | criptos | jero: I really dont know, And I dont thinks so, I have filters installed, and how can I verify that? |
| 22:54.34 | |dennis| | criptos: the line 1 button need to dial thru line1, the line 2 button onthe phone need to dial thru line 2...and so on.. |
| 22:54.54 | jero | criptos: try loading the driver module with lowpower=1 |
| 22:54.58 | paulc | jeob x 3: are the phones registered? (sip show peers) and does your dialplan/context allow them to dial each other? |
| 22:54.59 | Ariel_ | jeobjeobjeob, what are you using normal dialing rules or amp |
| 22:55.10 | criptos | Dennis: take your phone, plug it to a fxs interface (A green one) dial your magic buttons and see what toes the log says... |
| 22:55.13 | jeobjeobjeob | normal dialing rules |
| 22:55.21 | jeobjeobjeob | sip show peers shows the phone |
| 22:55.22 | jeobjeobjeob | s |
| 22:55.25 | Katty | hi |
| 22:55.26 | jeobjeobjeob | i have a phones context |
| 22:55.33 | jeobjeobjeob | and the dp's all inherit the phones context |
| 22:56.15 | Ariel_ | jeobjeobjeob, ok but when you dial the internal phone what does the cli say about the call? |
| 22:56.15 | criptos | For example, I have one phone, which has a button called "voicemail" when you press this button, the phone dials *811# (dont ask me why), so, I just added voicemail to that extension and voila :) |
| 22:56.21 | Ariel_ | Katty, how are you tonight? |
| 22:56.37 | Ariel_ | Rain,Rain go away and don't come back again..... |
| 22:56.46 | |dennis| | criptos: i will be getting my phones next week i was just asking in advance.....thanks anyways..maybe it will be better to ask when i have everything in place.... |
| 22:57.04 | jeobjeobjeob | it says "executing dial" |
| 22:57.14 | jeobjeobjeob | but then it immediately says all lines are busy |
| 22:57.15 | Ariel_ | |dennis|, don't think of buttons for lines |
| 22:57.29 | criptos | dennis, but you get the idea? |
| 22:57.49 | Ariel_ | jeobjeobjeob, what is your dial rule for internal phones? |
| 22:57.53 | |dennis| | yes guys ..thanks.. |
| 22:57.58 | Katty | Ariel_: consuming falafel |
| 22:58.30 | |dennis| | Ariel_: i agree with you..i can configure * to basically choose which line is available.......and dial out... |
| 22:58.42 | Optic | is mark3 the best echo canceller for remote hybrid imbalance echo? |
| 22:58.50 | rg1_ | anyone know about an error I'm getting ---- " -- DBget: varname=RecEnable, family=RECORD-OUT, key=108 |
| 22:58.50 | rg1_ | <PROTECTED> |
| 22:59.08 | Ariel_ | Katty, so your snacking |
| 22:59.27 | Katty | Ariel_: i consider falafel dinner. |
| 22:59.46 | jeobjeobjeob | for internal phones |
| 22:59.48 | jeobjeobjeob | wait a second |
| 22:59.51 | Ariel_ | Katty, ahh well I guess you could. |
| 22:59.52 | jeobjeobjeob | then dial phone for 20 seconds |
| 23:00.22 | Katty | Ariel_: (= |
| 23:00.30 | jeobjeobjeob | so itd be like exten => 110,1,Wait(1) |
| 23:00.31 | Ariel_ | ~google falatel |
| 23:00.38 | jeobjeobjeob | exten => 110,2,Dial(SIP/cust-u1005-001319C049271,20,r) |
| 23:00.49 | Ariel_ | jeobjeobjeob, what is that |
| 23:01.04 | Ariel_ | your sip device is cust-u1005 |
| 23:01.08 | neron | Anyone familiar with Avaya IP Office products? |
| 23:01.35 | jeobjeobjeob | oh crap |
| 23:02.20 | jeobjeobjeob | yeah it is |
| 23:02.28 | jeobjeobjeob | that's the name in sip.conf |
| 23:02.28 | Ariel_ | jeobjeobjeob, you should not have the r at the end either |
| 23:02.43 | jeobjeobjeob | would that affect the performance? |
| 23:02.49 | Ariel_ | then you would dial sip/cust-u1005,20) |
| 23:03.02 | Ariel_ | then you would dial(sip/cust-u1005,20) |
| 23:03.28 | Ariel_ | the name is the long ass number after the u1005- |
| 23:03.29 | jeobjeobjeob | the name is cust-u1005-001319C049271 |
| 23:03.33 | Ariel_ | argh why? |
| 23:03.40 | jeobjeobjeob | (that's the mac address followed by the line number) |
| 23:03.47 | jeobjeobjeob | this format had been working up until today |
| 23:03.52 | jeobjeobjeob | and then something died |
| 23:03.54 | Ariel_ | jeobjeobjeob, you never done the kiss rule |
| 23:04.07 | jeobjeobjeob | keep it simple stupid? |
| 23:04.13 | criptos | Any other tougts about voltage control? |
| 23:04.32 | jeobjeobjeob | lemme try that |
| 23:04.36 | Ariel_ | Optic, what card do you have |
| 23:04.53 | Ariel_ | criptos, voltage control??? |
| 23:05.12 | criptos | Ariel_ How to verify that? |
| 23:06.04 | Katty | Ariel_: all gone. |
| 23:08.20 | rg1_ | question: We added a SIP extension from the web page, and set it up for recording. x103 - works fine |
| 23:08.29 | twisted[work] | Ariel_, say hello to Katrina :) |
| 23:08.48 | Ariel_ | twisted[work], ok let me just yell hello katrina |
| 23:08.54 | twisted[work] | lol |
| 23:09.05 | Ariel_ | rg1_, your using amp? |
| 23:09.16 | rg1_ | we then simply made changes in the .conf files to add another extension (x108) - the extension looks good and the same as x103 in all the .conf files -BUT in the log it says something about DBGET database error for x108 |
| 23:09.22 | rg1_ | yes |
| 23:09.24 | twisted[work] | Ariel_, just don't go outside to say hi, i hear she likes to throw things at people. |
| 23:09.39 | Ariel_ | she does |
| 23:09.51 | rg1_ | it appears as though maybe something more than just modifying the configs needs to be done to add an extension? |
| 23:09.54 | Ariel_ | rg1_, you need to put the extensions with the amp gui |
| 23:10.03 | rg1_ | bummer |
| 23:10.19 | twisted[work] | it's amazing to me that you still have power / internet access attm |
| 23:10.19 | Ariel_ | or make your own in the extensions_custom.conf but follow there way of doing things |
| 23:10.20 | rg1_ | is that because the amp gui inserts something into the database? |
| 23:10.35 | Ariel_ | twisted[work], it comes and goes |
| 23:10.40 | twisted[work] | Ariel_, ah. |
| 23:10.47 | Ariel_ | laptop has about 2.5 hours left |
| 23:10.52 | Ariel_ | phone lines are the last to go |
| 23:11.01 | dersteer | thats a great run time for a laptop |
| 23:11.06 | Katty | twisted[work]: mrow. |
| 23:11.11 | twisted[work] | hi Katty ;) |
| 23:11.12 | criptos | bah |
| 23:11.13 | dersteer | mine could hardly go 2.5 hrs fully charged |
| 23:11.18 | Ariel_ | I think we lost out sat dish. No signal |
| 23:11.24 | twisted[work] | d'oh |
| 23:11.36 | twisted[work] | sat. TV + hurricane == nothing on |
| 23:11.38 | rg1_ | Ariel - we DID do that in the extensions_custom.confhey ariel - there is NO extensions_custom.conf |
| 23:11.46 | *** join/#asterisk darkskiez (n=darkskie@host86-133-147-187.range86-133.btcentralplus.com) |
| 23:11.55 | rg1_ | there is only extensions.conf and extensions_additional.conf |
| 23:12.18 | Ariel_ | you need to make it there should be a sample one around there. |
| 23:12.27 | dersteer | twisted[work]: you could change dishes and get it no matter what... use a fringe area dish |
| 23:12.29 | Katty | yay tomorrow is friday |
| 23:12.30 | rg1_ | yeah theres a sample |
| 23:12.41 | Ariel_ | rg1_, you see read it and follow it's lead |
| 23:13.24 | Ariel_ | hey ligths are back |
| 23:13.25 | rg1_ | here's what i don't understand |
| 23:14.16 | rg1_ | when i do a "database show" on the CLI> line, it shows different entries for 103 and 108 |
| 23:14.25 | Ariel_ | oh talk too soon there out again. |
| 23:14.30 | rg1_ | like SIP/Registry/103 and /SIP/Registry/108 are there |
| 23:14.42 | *** join/#asterisk W|NGNUT (n=wingnut-@062016241059.customer.alfanett.no) |
| 23:15.25 | Ariel_ | rg1_, why don't you want to use the gui? |
| 23:16.00 | rg1_ | because we want to automatcially do these configs from a script |
| 23:16.41 | supaigtr | In debug mode what does Urgent handler mean? Is that timing slips, interrupts?? |
| 23:17.06 | Ariel_ | rg1_, if you don't want to use amp then look at the mysql database and write it direct to it. |
| 23:17.55 | shmaltz | is there anyway to parse lets say an argument in the dialplan? |
| 23:18.19 | Ariel_ | I am going off for a little while the wind is too high and we need to move to another room. See you all later. |
| 23:19.00 | *** join/#asterisk netsurfer (n=bbjunkie@i-83-67-48-18.freedom2surf.net) |
| 23:19.21 | netsurfer | hi ppl |
| 23:19.54 | *** join/#asterisk Corndawg (i=whoisit@c-66-176-249-51.hsd1.fl.comcast.net) |
| 23:22.00 | rg1_ | Ariel - got it |
| 23:24.37 | *** join/#asterisk Snake-Eyes (n=blog@203.201.97.50) |
| 23:25.05 | *** part/#asterisk _dan__t (n=dant@69.93.33.54) |
| 23:25.44 | *** join/#asterisk Corndawg_ (i=whoisit@c-66-176-249-51.hsd1.fl.comcast.net) |
| 23:28.16 | *** join/#asterisk pablasso (n=pablasso@201.138.255.59) |
| 23:28.28 | pablasso | hi guys |
| 23:28.36 | *** join/#asterisk _cleric_ (n=dacleric@p5482B021.dip0.t-ipconnect.de) |
| 23:29.09 | *** join/#asterisk Moc (n=mochouin@h66-201-214-109.gtconnect.net) |
| 23:29.13 | criptos | hi |
| 23:29.39 | *** join/#asterisk W1NGNUT (n=wingnut-@062016241059.customer.alfanett.no) |
| 23:29.39 | pablasso | 17:42:09 WARNING[30953]: pbx.c:1291 pbx_extension_helper: No application 'rxfax' for extension (fax, s, 2) |
| 23:29.40 | pablasso | <PROTECTED> |
| 23:30.09 | pablasso | > 17:42:09 WARNING[30953]: pbx.c:1291 pbx_extension_helper: No application 'rxfax' for extension (fax, s, 2) |
| 23:30.20 | pablasso | im still fighting with spandsp, what can i do if i get that error? |
| 23:30.21 | visik7 | can I fax through voip ? |
| 23:30.32 | criptos | *sigh* |
| 23:30.32 | pablasso | that is what im trying to do... |
| 23:30.34 | Moc | visik7: yes in perfect conditions |
| 23:30.49 | visik7 | ok thanks |
| 23:31.48 | pablasso | the rxfax is already compiled as module, and i can see it in /usr/lib/asterisk/modules/ so i dont know what it says that the application doesnt exists... |
| 23:32.01 | supaigtr | pablasso: did you install spandsp and do the makefile patch? |
| 23:32.03 | tzanger | pablasso: make install perhaps? |
| 23:32.18 | pablasso | yeah i did.. it patched makefile without errors |
| 23:32.52 | supaigtr | patch makefile, make install spandsp, make install asterisk |
| 23:33.55 | pablasso | lol... i got the error.. pretty stupid mistake... brb |
| 23:34.40 | supaigtr | Anyone know how to debug echo canceling? That is get it to display what it is doing on both vpm eq card and the new kb1 software? |
| 23:34.41 | *** join/#asterisk Legend (n=legend@24.244.142.133) |
| 23:35.22 | *** join/#asterisk ginvent (n=joseph@adsl-67-121-208-105.dsl.sndg02.pacbell.net) |
| 23:35.33 | ginvent | How do I activate On-Demand recording? |
| 23:35.38 | ginvent | I have it turned on in amp. |
| 23:35.57 | ginvent | I assume I can turn on and off recording during the convo? |
| 23:36.41 | ginvent | Hello? |
| 23:36.47 | ginvent | Burrito? |
| 23:38.02 | Tekmaven | hey |
| 23:38.07 | Tekmaven | what ports does IAX use? |
| 23:38.13 | Tekmaven | what should i forward to the asterisk? :) |
| 23:38.25 | ginvent | 4569? |
| 23:38.27 | ginvent | hang on. |
| 23:38.31 | Tekmaven | i dont seem to see it on the wiki |
| 23:38.41 | Tekmaven | ak |
| 23:38.44 | Tekmaven | *found it :) |
| 23:38.47 | Tekmaven | IAX uses a single UDP port 4569, and thus works well in NAT environments (the obsolete IAX1 protocol used port 5036). |
| 23:39.13 | ginvent | I was correct! |
| 23:39.15 | ginvent | :D |
| 23:39.20 | Tekmaven | yup, nice one |
| 23:39.27 | ginvent | Do you know how to activate on-demand recording? |
| 23:39.43 | ginvent | I have it turned on and setup... I have no idea the key to press to get it to record. |
| 23:41.01 | neron | Can Asterisk use an old DSP card? |
| 23:41.24 | ginvent | Depends on which one... you have to search google to tell sometimes. |
| 23:41.25 | pablasso | mmm it didnt worked... im using a zap channel to dial my asterisk server, the fax tried to send the info but got a comm. problem |
| 23:41.54 | pablasso | and i didnt got any error in the asterisk cli just this -> Executing RxFAX("Zap/1-1", "/home/testfax.tif") in new stack and then hungup |
| 23:41.59 | pablasso | u_u |
| 23:42.14 | ginvent | Wait... I had the same problem. |
| 23:42.21 | ginvent | You need to set your tx and rx gains. |
| 23:42.32 | ginvent | Otherwise faxes won't work. |
| 23:42.59 | ginvent | Use the zap config stuff to test and set it. |
| 23:43.06 | pablasso | oks... and whichs specific valor i set? |
| 23:43.12 | pablasso | just test differents? |
| 23:43.29 | hugo-v6 | ginvent: is there a documentation about that topic? |
| 23:43.45 | ginvent | hugo, no... not that I know of... |
| 23:43.57 | ginvent | You have to change your zapata.conf file to have tx and rx gains set. |
| 23:44.17 | pablasso | oks thanks, let me try it |
| 23:44.17 | ginvent | There are some documents on calling a 0 db line thingy that you can set your tx and rx gains. |
| 23:44.29 | ginvent | Pablasso, are you using a x100p clone? |
| 23:44.43 | neron | Can someone tell me the best way for me to implement Asterisk for voicemail on my current setup: PBX already in place with PRI, the old voicemail system needs to be replaced with Asterisk, but the voicemail system has (2) 4 port DSP PCI cards. Can it be done? |
| 23:44.53 | pablasso | ginvent, yes im using it |
| 23:45.03 | ginvent | I had to crank the gain... I use rxgain=8.0 and txgain=16.0 Which is huge. |
| 23:45.15 | ginvent | Best thing is to run ztmon during a fax and see where the levels are. |
| 23:45.21 | pablasso | oks, im gonna try that gains |
| 23:45.43 | ginvent | neron, depends on the cards and the drivers. |
| 23:46.23 | ginvent | I would type into google asterisk and the name of the card... |
| 23:46.26 | neron | ginvent, if I decide to replace those card. What would you recommend? |
| 23:46.45 | ginvent | Well, depends on cash flow... I think overall people like those digium cards. I never used one though. |
| 23:47.25 | ginvent | You can get a card and place a combination of 4 either fxs or fxo ports on it. |
| 23:47.51 | ginvent | or any combo of. |
| 23:48.01 | ginvent | They range from $250 - 500 |
| 23:48.10 | [Outcast] | has anyone notice problem with zaptel channels locking up asterisk? |
| 23:48.14 | ginvent | depending on the # of ports. |
| 23:48.30 | ginvent | Out, not me... My phone crashes though... lol |
| 23:48.40 | ginvent | Not lately since I updated the firmware though. |
| 23:48.44 | *** join/#asterisk Corndawg (i=whoisit@c-66-176-249-51.hsd1.fl.comcast.net) |
| 23:48.57 | ginvent | The sipura spa-841 is pretty cool. |
| 23:49.04 | ginvent | for a 4 line cheapo ip phone. |
| 23:49.05 | neron | ginvent, ok thank you... |
| 23:49.12 | ginvent | Neron, no problam. |
| 23:49.16 | [Outcast] | i think it might be a problem with the board. |
| 23:49.20 | hugo-v6 | ginvent: well i want to use it with isdn anyways... so i hope i get around that |
| 23:49.45 | [Outcast] | ginvent: you are using digium boards right? |
| 23:49.53 | *** join/#asterisk websae (i=websae@207-118-143-134.dyn.centurytel.net) |
| 23:49.58 | ginvent | No, I don't use digium boards... but I have heard good things. |
| 23:50.07 | ginvent | I don't do analog if I can avoid it. |
| 23:50.13 | [Outcast] | k |
| 23:50.21 | [Outcast] | i am using the sangoma boards |
| 23:50.41 | ginvent | IP phones are so cheap now... no reason to go analog. |
| 23:50.44 | neron | ginvent, how about implementing VoIP using Asterisk in different branches to save money on phone calls. Is it possible? Assuming all braches have DSL connection? |
| 23:50.55 | ginvent | neron, I totally do that now. |
| 23:51.00 | ai-a | why do my IAX calls keep cutting off, the longest we had was 55 minutes, but before that they were about 3 minutes long... im getting WARNING[25013] chan_iax2.c: Max retries exceeded to host 192.168.0.3 on IAX2/guest@192.168.0.3:4569-5 (type = 6, subclass = 2, ts=3217655, seqno=35) |
| 23:51.16 | [Outcast] | neron, that is the whole point of voip. |
| 23:51.18 | ginvent | hmm ai-a, don't know... |
| 23:51.40 | ginvent | Neron, It's cool, I can call my local number, patch through to a voip # and save tons of $. |
| 23:51.51 | websae | Im trying to setup gentoo, and i downloaded the minimal install cd, when i boot from it i get the command prompt and type boot, then say "Preparing Live CD...press alt F1 for verbose mode"...it just stays and doesn't go anywhere at that screen, anyone have any ideas? |
| 23:51.57 | ginvent | It's like $0.02 per minute on a super cheapo line. |
| 23:52.02 | ginvent | or free for like $25/mo. |
| 23:52.09 | [Outcast] | ginvent: what about larger scale where you need a PRI |
| 23:52.11 | ginvent | But I don't like monthly... |
| 23:52.14 | ginvent | I buy minutes. |
| 23:52.25 | kg | websae: wow, that's a great off-topic! congrats! |
| 23:52.25 | ginvent | outcast? |
| 23:52.31 | neron | gincent, brb.. I wanna talk to you a bit more |
| 23:52.39 | neron | brb |
| 23:52.39 | websae | i know a lot of gentoo users in here |
| 23:52.42 | *** part/#asterisk neron (n=chatzill@adsl-67-119-149-94.dsl.lsan03.pacbell.net) |
| 23:52.43 | websae | i am setting up gentoo to use asterisk |
| 23:52.52 | websae | but iam having issues with even installing gentoo |
| 23:52.58 | greg_work | websae: try #gentoo |
| 23:53.00 | websae | and wondering if anyone here would know about that problem |
| 23:53.18 | ai-a | ginvent:also getting in the *'s == Spawn extension (macro-stdexten, s, 1) exited non-zero on 'Local/1@default-0ad6,2' in macro 'stdexten' == Spawn extension (default, 1, 1) exited non-zero on 'Local/1@default-0ad6,2' |
| 23:53.29 | ginvent | The amount of data you can get on a 100T line is great. |
| 23:53.33 | syle | there firmware for teh TDM400P? |
| 23:53.37 | [Outcast] | ginvert: you said you avoid analog when possible, what about larger scale PBXs |
| 23:53.51 | ginvent | Outcast, again, what is the need for analog? |
| 23:54.18 | *** join/#asterisk Freman (n=twitsrus@jaguar.wbs.net.au) |
| 23:54.19 | *** join/#asterisk shmaltz (n=chatzill@69.28.255.210) |
| 23:54.29 | *** join/#asterisk Corndawg_ (i=whoisit@c-66-176-249-51.hsd1.fl.comcast.net) |
| 23:54.39 | Freman | hey... anyone got an 80x15 image for asterisk? (for a "powered by") |
| 23:54.47 | [Outcast] | ginvert: let me try another question, who do you use a VOIP provider for larger systems? |
| 23:54.53 | *** join/#asterisk neron (n=chatzill@adsl-67-119-149-94.dsl.lsan03.pacbell.net) |
| 23:55.10 | pablasso | ginvent, with gains rx=8 and tx=16 still got the communication problem in the fax |
| 23:55.11 | [Outcast] | ginvert: i can't type tonight |
| 23:55.29 | neron | ginvent, ok what do you use for a VoIP solution? |
| 23:55.38 | shmaltz | anybody know how to enable the unboard echo can on the TE406? |
| 23:55.47 | shmaltz | unboard = onboard |
| 23:59.13 | ai-a | Freman: a what? |
| 23:59.26 | ai-a | oh ic :) |
