| 00:00.07 | za3toor | my internal extentions are almost the same as the outbound ones |
| 00:00.29 | za3toor | so for example my outbound calls are 1519xxxxxxx |
| 00:00.45 | za3toor | my extentions are something like 1519xxxxxxy |
| 00:03.04 | JayTee52 | if * can't find a match in any of the other contexts it will look at the default context and you can put a exten => _1519XXXXXXX, Dial(${outboundtrunk}/${EXTEN}) |
| 00:03.38 | JayTee52 | but that's a workaround, you really need to read or re-read the Dialplan basics in The Book |
| 00:03.42 | JayTee52 | ~book |
| 00:03.43 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com, or see ~buybook |
| 00:04.09 | za3toor | thank you very much |
| 00:04.28 | za3toor | i will do thaat for sure... |
| 00:05.16 | JayTee52 | or you could create a context called [providername] and put an include => providername in each context you want to have outbound calling available. |
| 00:05.42 | JayTee52 | it will try to match against anything in the context first then it will try the included context. |
| 00:06.02 | rob0 | ISTM easier to make internal extensions shorter, easier to dial |
| 00:06.04 | za3toor | oh ic |
| 00:06.18 | JayTee52 | yeah, I stick to 4 digit extensions |
| 00:06.39 | za3toor | ok |
| 00:06.45 | JayTee52 | you can always add an NXX to it in the Dial app |
| 00:06.58 | za3toor | yes... |
| 00:26.35 | *** join/#asterisk mackes (n=root@cpe-24-198-43-238.buffalo.res.rr.com) |
| 00:37.59 | *** join/#asterisk ming_zym (n=ming_zym@nat/yahoo/x-43578097bd15afaa) |
| 00:41.20 | *** join/#asterisk shido6 (n=shido6@74-130-51-159.dhcp.insightbb.com) |
| 00:45.04 | *** join/#asterisk mltlnx (n=mltlnx@cpe-24-193-11-14.nyc.res.rr.com) |
| 00:45.08 | *** join/#asterisk DarkRift (i=dark@bas10-montreal02-1177582051.dsl.bell.ca) |
| 00:52.35 | *** join/#asterisk obnauticus (n=obnautic@c-67-160-187-7.hsd1.wa.comcast.net) |
| 00:52.48 | mackes | Has anyone online attended Digium Bootcamp? |
| 00:54.55 | Maliuta | didn't we do this one a few hours ago? |
| 00:56.26 | drmessano | The first rule of Digium Bootcamp is, don't talk about Digium Bootcamp |
| 00:56.58 | drmessano | It's like Fight Club, only, it's SIP enabled |
| 00:57.14 | Maliuta | but they prefer to use IAX |
| 00:57.50 | Maliuta | and if you get out of line they hit you with a jitter buffer |
| 00:58.21 | rob0 | yikes |
| 00:58.50 | drmessano | Could be worse |
| 00:59.24 | drmessano | I went to X100P bootcamp.. The instructor kept repeating repeating himself himself |
| 01:00.06 | rob0 | but on the bright side ... it only cost $10 |
| 01:01.15 | drmessano | Yes.. It was kinda odd... the instructor looked a lot like Russell Bryant, but his name was Bussel Ryant |
| 01:01.16 | tzanger | drmessano: haha |
| 01:01.19 | drmessano | Damn clones |
| 01:02.09 | *** join/#asterisk mikong (n=gones@203.193.37.251) |
| 01:02.41 | Maliuta | what happens at asterisknow camp? is it a shoddy facade that allows you only part access to asterisk bootcamp? |
| 01:03.30 | drmessano | I went to Grandstream camp too.. |
| 01:04.19 | jameswf-home | I went to band camp |
| 01:04.32 | drmessano | The instructor sounded like a 10 pack-a-day smoker.. He fell over 10 times over the weekend and broke 3 of his fingers, his ankle, both wrists, and his collarbone. Guess he wasn't built very well. |
| 01:04.41 | rob0 | The proceedings at Zaptel camp were hard to follow, what with the 1000 interrupts per second. |
| 01:05.13 | jameswf-home | heh zoooom |
| 01:05.23 | Maliuta | pfft, who doesn't use 1000hz timing |
| 01:06.02 | drmessano | Cisco camp was a ripoff.. I paid $18000 for a 2 hour course. We walked in, sat down, the instructor spoke a language none of us could understand, then told us we could come back next year if we sign a support contract. |
| 01:06.32 | tzanger | man we got comedians in there tonight |
| 01:06.54 | drmessano | Trixbox camp was worse |
| 01:06.55 | gitguy | drmessano: that's sad |
| 01:06.56 | Maliuta | drmessano: sounds similar to linsys camp |
| 01:07.08 | Maliuta | linksys even |
| 01:07.49 | Maliuta | tzanger: it's not night though ... it's 11am |
| 01:08.23 | rob0 | Digium Standard Time |
| 01:08.27 | drmessano | Trixbox camp was a whole weekend of hearing how nothing they use is actually supported by them, but that Trixbox is the greatest PBX distro in the world. Then Kerry Garrison made us watch slides of his vacation last summer to the Grand Canyon. |
| 01:08.58 | rob0 | oooh now there would be a challenge ... |
| 01:09.19 | rob0 | ... tin cans with string, all the way across the Grand Canyon |
| 01:10.33 | Maliuta | rob0: that was skype camp |
| 01:10.47 | rob0 | haha |
| 01:10.50 | Maliuta | rob0: except the ran the string via equador |
| 01:11.19 | drmessano | Skype Camp is more like getting drunk and watching Anime on YouTube |
| 01:11.26 | Maliuta | something about needing to get around the rails near the edge |
| 01:11.42 | *** join/#asterisk craigk (n=craigk@58.174.150.119) |
| 01:12.31 | drmessano | You could get croaked on grain alcohol, watch 3 hours of "My Little Pony", and never have to guess what Skype is like again |
| 01:12.50 | *** join/#asterisk Ebola (n=Ebola@unaffiliated/ebola) |
| 01:13.13 | JayTee52 | the most frequently entered question in AsteriskNOW, "Hey guys! Anyone here?" |
| 01:13.41 | tzanger | skype has just not been working for me... I'm not talking the voice part of it, I've neve rused it for that |
| 01:13.45 | tzanger | I'm talking straight IM with skype |
| 01:14.12 | drmessano | Probably works fine in Pidgin |
| 01:14.15 | drmessano | Which is.. ironic |
| 01:14.18 | *** join/#asterisk Mavvie (n=edwin@ppp121-44-49-247.lns10.syd7.internode.on.net) |
| 01:14.41 | drmessano | An app that Mark Spencer wrote handles Skype IM better than their client O.o |
| 01:14.55 | jblack | Skype is such a mess. |
| 01:18.48 | *** join/#asterisk tengulre (n=tengulre@125.71.208.16) |
| 01:19.44 | mackes | yep. I did ask the same question this morning- I was hoping some others might be online- so <mackes> Has anyone online attended Digium Bootcamp? |
| 01:22.05 | *** join/#asterisk lmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 01:22.05 | *** mode/#asterisk [+o lmadsen] by ChanServ |
| 01:22.14 | Maliuta | drmessano: the pidgin to skype thing simply calls the skype client |
| 01:22.40 | Maliuta | drmessano: I know, I had to use the piece of shite in my last job |
| 01:25.47 | drmessano | Hmmm |
| 01:25.59 | drmessano | I didnt know that.. |
| 01:26.22 | Maliuta | I wish I didn't |
| 01:27.18 | *** join/#asterisk mvanbaak (n=michiel@vanbaak.xs4all.nl) |
| 01:29.35 | *** join/#asterisk znoG_ (n=gs@host54.190-30-22.telecom.net.ar) |
| 01:29.41 | *** join/#asterisk cmantito (n=gphreak@pool-71-188-82-138.cmdnnj.east.verizon.net) |
| 01:45.03 | *** join/#asterisk tomcontr3 (n=gcontrer@248-79-20-190.adsl.terra.cl) |
| 01:45.55 | tomcontr3 | hi... Im trying to get to Servers conected using and IAX2 Ext. in Server1 and a IAX2 Trunk in server2... the problem is taht server 2 is not getting the DID from the Ext. |
| 01:46.04 | tomcontr3 | any idea why this might be happening? |
| 01:47.00 | *** join/#asterisk jeffspeff (n=jeff@c-68-53-81-14.hsd1.ky.comcast.net) |
| 01:50.10 | tomcontr3 | any idea? |
| 01:50.19 | shido6 | whats the debug say |
| 01:50.57 | *** join/#asterisk profounded (n=Bryan_Ru@cpe-74-66-233-176.nyc.res.rr.com) |
| 01:51.40 | tomcontr3 | <PROTECTED> |
| 01:51.51 | tomcontr3 | I need to match the incoming calls |
| 01:52.04 | tomcontr3 | so I can route them to an spesific ext. |
| 01:52.49 | shido6 | err |
| 01:52.58 | shido6 | thats something u created :) |
| 01:53.21 | shido6 | can u see what number is being sent? :) |
| 01:54.02 | tomcontr3 | but... If I would need to do this: what would be a correct Ext. Configuration? |
| 01:59.20 | *** join/#asterisk Iamnacho (i=Iamnacho@ip68-103-153-140.ks.ok.cox.net) |
| 02:01.36 | *** join/#asterisk esaym (n=user@cpe-70-120-89-6.satx.res.rr.com) |
| 02:03.51 | *** join/#asterisk drehlecom (i=ircbnc@unaffiliated/drehlecom) |
| 02:05.35 | *** join/#asterisk Kumbang (n=dsp@rusnas.paume.itb.ac.id) |
| 02:15.05 | *** join/#asterisk BeeBuu (n=beebuu@218.13.81.208) |
| 02:42.05 | *** join/#asterisk drummond_ (n=rsd@c-71-224-187-182.hsd1.pa.comcast.net) |
| 02:55.51 | *** join/#asterisk jebreen (n=john@elephant.hgait.net.au) |
| 03:00.51 | *** join/#asterisk gitguy (n=diego@adsl-134-171.click.com.py) |
| 03:28.21 | *** join/#asterisk coppice (n=chatzill@99.166.17.210.dyn.pacific.net.hk) |
| 03:45.25 | *** join/#asterisk DaveCanoe (n=Dave@canoe404.dclg.ca) |
| 03:46.27 | *** join/#asterisk ftp3 (n=none@pool-71-117-212-7.ptldor.fios.verizon.net) |
| 03:52.29 | *** join/#asterisk stkn_ (n=stkn@gentoo/developer/pdpc.active.stkn) |
| 04:03.59 | *** join/#asterisk Micko113 (n=micko113@c-67-183-170-166.hsd1.wa.comcast.net) |
| 04:05.17 | jameswf-home | heh http://unix.rulez.org/~calver/pictures/hax_war.gif |
| 04:06.59 | jaytee | hehehe, that was good |
| 04:09.26 | drmessano | http://www.xkcd.com/419/ |
| 04:11.36 | jaytee | this one is my favorite: http://www.xkcd.com/418/ |
| 04:11.49 | drmessano | lol yeah |
| 04:14.27 | drmessano | http://xkcd.com/276/ |
| 04:15.20 | jaytee | http://i299.photobucket.com/albums/mm318/DasutinD/ebay_shower.jpg |
| 04:15.41 | *** join/#asterisk vector (n=vector@host-178-246-220-24.midco.net) |
| 04:15.56 | *** join/#asterisk ming_zym (n=ming_zym@nat/yahoo/x-c2f93cc14fe965f3) |
| 04:21.30 | jbeez | hahaha wtf |
| 04:33.47 | *** part/#asterisk gitguy (n=diego@adsl-134-171.click.com.py) |
| 04:46.42 | *** join/#asterisk gmaruzz (n=gmaruzz@213-140-6-122.ip.fastwebnet.it) |
| 04:47.37 | *** part/#asterisk gmaruzz (n=gmaruzz@213-140-6-122.ip.fastwebnet.it) |
| 04:49.43 | *** join/#asterisk gmaruzz (n=gmaruzz@213-140-6-122.ip.fastwebnet.it) |
| 04:54.51 | jblack | I miss everyoen loves eric raymond. |
| 04:55.59 | jaytee | y'know, i watched several episodes but I could just never get to the point of loving Raymond. |
| 04:56.05 | Strom_M | unix jokes are the funniest ever </sarcasm> |
| 04:56.32 | drmessano | Unix is a joke.. 25 years to fix a bug |
| 04:56.36 | jblack | No, the comic. Everyone loves Eric Raymond, but me |
| 04:57.12 | jblack | http://geekz.co.uk/lovesraymond/ |
| 04:57.13 | *** join/#asterisk Frogzoo (n=Frogzoo@124.184.18.213) |
| 04:57.38 | coppice | whatever his success, I don't think Eric Raymond is actually trying to be a comic |
| 04:58.15 | jblack | He makes a great one, considering his IP whoring proclivities |
| 04:59.27 | jaytee | nite all |
| 04:59.32 | *** part/#asterisk jaytee (n=jforde05@unaffiliated/jaytee) |
| 05:10.56 | *** join/#asterisk CrashSys1 (n=kumba@azrael.crashsys.com) |
| 05:14.26 | *** join/#asterisk obnauticus (n=obnautic@c-67-160-187-7.hsd1.wa.comcast.net) |
| 05:33.14 | *** join/#asterisk mltlnx (n=mltlnx@cpe-24-193-11-14.nyc.res.rr.com) |
| 05:38.47 | *** part/#asterisk gmaruzz (n=gmaruzz@213-140-6-122.ip.fastwebnet.it) |
| 05:40.46 | *** join/#asterisk apollonx (n=admin@193.19.189.38.STATIC.ISP.KZ) |
| 05:45.10 | *** join/#asterisk angeldavid (n=angeldav@nelug/coreteam/pepo) |
| 05:57.36 | *** join/#asterisk orcimrepus (n=orcimrep@74-130-123-207.dhcp.insightbb.com) |
| 06:00.40 | *** part/#asterisk pikachu2000 (n=pikachu2@196-209-10-21-ndn-esr-2.dynamic.isadsl.co.za) |
| 06:02.53 | *** join/#asterisk gmaruzz (n=gmaruzz@213-140-6-122.ip.fastwebnet.it) |
| 06:09.18 | *** join/#asterisk sergee (n=serg@voip1.west-call.com) |
| 06:13.19 | *** join/#asterisk shinao1 (n=shinao1@smtp.gtbank.com) |
| 06:19.49 | *** join/#asterisk bluequijote (n=chatzill@75-57-4-233.lightspeed.sndgca.sbcglobal.net) |
| 06:24.00 | *** join/#asterisk scardinal (n=supreme@port816.ds1-suoe.adsl.cybercity.dk) |
| 06:24.29 | *** join/#asterisk rcy (n=rcy@24.207.122.28) |
| 06:24.41 | *** join/#asterisk DaveCanoe (n=Dave@canoe404.dclg.ca) |
| 06:29.46 | *** join/#asterisk cmantito (n=gphreak@pool-71-188-82-138.cmdnnj.east.verizon.net) |
| 06:59.08 | *** join/#asterisk lsodi (n=root@213.168.26.50) |
| 07:00.46 | *** join/#asterisk mort_gib (n=mjensen@16.Red-83-36-63.staticIP.rima-tde.net) |
| 07:10.41 | *** join/#asterisk psk (n=psk@golia.caltanet.it) |
| 07:12.53 | *** join/#asterisk pa (n=pa@unaffiliated/pa) |
| 07:17.49 | *** join/#asterisk rolnd (n=rolnd@S0106006097940f68.vw.shawcable.net) |
| 07:18.28 | rolnd | how do I force remote codec on remote, it seems remote always sees requested format = unknown ? |
| 07:20.23 | *** join/#asterisk oej (n=olle@ns.webway.se) |
| 07:33.09 | *** join/#asterisk erojasv (n=erojasv@190.40.58.115) |
| 07:38.45 | *** join/#asterisk darkskiez (n=mhb@cpc1-broo3-0-0-cust659.renf.cable.ntl.com) |
| 07:41.55 | *** join/#asterisk kamanashisroy (n=kamanash@202.56.7.193) |
| 07:56.20 | jblack | ~book |
| 07:56.21 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com, or see ~buybook |
| 07:59.06 | *** join/#asterisk fordfrog (n=fordfrog@gentoo/developer/fordfrog) |
| 08:00.53 | *** join/#asterisk ghenry (n=ghenry@suretecsystems.plus.com) |
| 08:02.08 | *** join/#asterisk _gm (n=gmustafa@117.20.28.50) |
| 08:08.30 | *** part/#asterisk rolnd (n=rolnd@S0106006097940f68.vw.shawcable.net) |
| 08:13.11 | *** join/#asterisk gr0mit (n=tim@dhcp3.zuk40.mot-tools.co.uk) |
| 08:14.28 | *** join/#asterisk darviria (n=darviria@87-194-177-180.bethere.co.uk) |
| 08:16.54 | *** part/#asterisk BeeBuu (n=beebuu@218.13.81.208) |
| 08:17.07 | *** join/#asterisk af_ (n=getsmart@88-149-230-6.dynamic.ngi.it) |
| 08:19.54 | *** join/#asterisk ming_zym (n=ming_zym@nat/yahoo/x-9171ac95b5e0b4c8) |
| 08:25.53 | ikevin | hello |
| 08:25.54 | *** join/#asterisk igascream (n=igascrea@80.179.192.178.cable.012.net.il) |
| 08:26.23 | igascream | Hi all need some help ... |
| 08:26.39 | *** join/#asterisk jivco (n=jivco@85.187.217.6) |
| 08:26.44 | gr0mit | igascream, what sort of help? |
| 08:26.55 | igascream | Is it posible to make asterisk detect DTMF while ringing? |
| 08:27.48 | gr0mit | it all depends, but probably not |
| 08:28.50 | igascream | my boss is about to kill me I have some problem with hangup detection and that's the only way I can fix it |
| 08:29.23 | gr0mit | pls explain your config in more detail! |
| 08:30.55 | Strom_M | igascream: what exactly are you trying to do? what is the exact problem you're trying to solve/ |
| 08:31.57 | igascream | I have no disconnect supervision on my analog line so the only way I can detect hanging up is DTMF signals but if caller hangup while ringing it doesn't detect it and still calling |
| 08:32.53 | *** join/#asterisk fcois (n=fcois@bagnolet.acropolistelecom.net) |
| 08:33.01 | igascream | some idea???? |
| 08:33.03 | fcois | hello all |
| 08:33.15 | fcois | for what? |
| 08:33.32 | JT | igascream: best solution is to get polarity reverse on disconnection, or digital lines |
| 08:34.55 | igascream | JT, but I can't change my telephone provider properties |
| 08:35.13 | Strom_M | igascream: sure you can. call the business office and request they change it. |
| 08:35.13 | JT | then too bad |
| 08:35.46 | rob0 | Vote with your feet, telcos are becoming less relevant every day. |
| 08:36.19 | igascream | so you think there is no way to make * detect DTMF while ringing |
| 08:36.28 | Strom_M | rob0: yeah right...it's the telcos who carry virtually all telephone and internet traffic. |
| 08:37.03 | gr0mit | igascream, or a k-break signal |
| 08:37.19 | gr0mit | this is the one most readily detected by asteirsk |
| 08:37.26 | gr0mit | but best is to use ISDN |
| 08:37.38 | fcois | I need help for an asterisk appliance aa50 from digium |
| 08:38.22 | fcois | I need to recompil the busybox with a tftpd server |
| 08:38.32 | fcois | but I don't really know how to do :( |
| 08:38.57 | igascream | but k-break signal is available only in UK and USA or not? |
| 08:40.00 | JT | no idea what k-break is |
| 08:40.17 | JT | but polarity reverse on idle condition is available in heaps of places |
| 08:40.25 | *** join/#asterisk tzafrir_laptop (n=tzafrir@local.xorcom.com) |
| 08:40.29 | coppice | I think its a breakfast cereal |
| 08:41.01 | tzafrir_laptop | anybody noticed the spam in the edit page in voip-info.org ? |
| 08:41.09 | tzafrir_laptop | try to edit a page |
| 08:41.41 | JT | tzafrir_laptop: is polarity reverse on disconnect available on POTS lines in .il? |
| 08:42.02 | igascream | yeah it is already on in my config but still doesn't work |
| 08:42.26 | JT | igascream: you need to have the service enabled |
| 08:42.45 | tzafrir_laptop | JT: not AFAIK |
| 08:43.12 | JT | i guess isdn is the only option for igascream then |
| 08:43.12 | tzafrir_laptop | AFAIK some landlines have KS |
| 08:43.30 | gr0mit | k-break is a short break in line current when the calling party hangs up |
| 08:43.46 | gr0mit | isdn will be the only proper solution then |
| 08:44.06 | Strom_M | gr0mit: well, technically, it's when the other party hangs up |
| 08:44.13 | Strom_M | since the called party could be the one doing the hanging up too |
| 08:44.15 | gr0mit | gave up with analogue trunks a loooong time ago. they are just way too much hassle |
| 08:44.20 | igascream | tzafrir knows my problem .... |
| 08:44.30 | gr0mit | in uk we have calling party clearing |
| 08:44.31 | JT | yeah analogue lines are made of fail |
| 08:44.49 | gr0mit | for any biz use, forget analogue |
| 08:45.05 | tzafrir_laptop | igascream does not use a landlane. He uses an ATA from the cables company, right? |
| 08:45.06 | gr0mit | they are just waaaay too problematic |
| 08:45.34 | igascream | yeah from MP-202 device |
| 08:45.48 | gr0mit | ugh - voip to analogue to voip. |
| 08:46.05 | igascream | yeah but I have no choice((( |
| 08:46.08 | JT | yeah forget about it |
| 08:46.11 | gr0mit | why not? |
| 08:46.14 | JT | that's too bad |
| 08:46.21 | gr0mit | you always have a choice! |
| 08:46.27 | JT | there's no magic that will make hangup detection start to work |
| 08:46.48 | *** join/#asterisk MatBoy (n=MatBoy@wiljewelwetenhe.xs4all.nl) |
| 08:46.50 | gr0mit | basically analogue was designed over 100 years ago |
| 08:47.10 | gr0mit | you really really do not want to use it |
| 08:47.34 | JT | except in an emergency |
| 08:47.42 | tzafrir_laptop | gr0mit: well, 50 years ago is probably more accurate |
| 08:47.46 | igascream | ok thanx all |
| 08:48.26 | coppice | nope. 100 is about right |
| 08:48.31 | gr0mit | tzafrir, the first automatic telephone exchanges were about 1900 |
| 08:48.42 | coppice | even strowger dates from 1929 |
| 08:49.06 | gr0mit | earlier, Coppice |
| 08:51.12 | fcois | someone can give help for 'asterisk appliance aa50' ??? |
| 08:51.24 | JT | fcois: digium can |
| 08:51.58 | coppice | well, strowger the elder died well before that, but something significant in 1929 made the exchange the one we know now. I just can't remember what it was :-\ |
| 08:51.58 | fcois | JT: digium has no time for me |
| 08:52.05 | fcois | no response by email |
| 08:52.29 | fcois | JT: I work with the source and they don't want to give the source |
| 08:52.39 | fcois | JT: but it is 'open source...' |
| 08:52.57 | JT | well you bought a commercial source version |
| 08:54.04 | *** join/#asterisk dbmoodb (n=david@unaffiliated/dbmoodb) |
| 08:54.26 | dbmoodb | hi there i was getting 403 errors from my voip provider -- any hints ? |
| 08:59.10 | gr0mit | dbmoodb, paid your bill ;-)? |
| 09:00.15 | dbmoodb | i have |
| 09:01.00 | dbmoodb | the bill is not an issue -- i am a first time asterisk user trying to set my server up --- i have a modem that does voip and is an ata and i want to use it - i have successfully registered with my isps service |
| 09:02.05 | lsodi | Hi, I have two digium cards in asterisk server, one is connected to teleco and acting as cpe and second one is connected to another pbx and acting as net, when I call extension 333 I get following notice on asterisk cli: http://pastebin.com/dede557e |
| 09:03.01 | lsodi | what might cause this? |
| 09:16.03 | jblack | Some day, I will figure out how to send faxes over asterisk. |
| 09:17.44 | dbmoodb | meh it was working before my question is how to add my ata too it -- my server |
| 09:18.04 | *** join/#asterisk Delvar (n=Delvar@77.240.56.22) |
| 09:19.21 | tzafrir_laptop | fcois: no response to email since? |
| 09:19.41 | fcois | tzafrir_laptop : no response |
| 09:19.55 | tzafrir_laptop | right, but when did you send it? |
| 09:20.30 | fcois | tzafrir_laptop : friday, I could phone the support and said that I need to speak to the project responsable, but he was at a meeting ... |
| 09:20.51 | fcois | I send it on friday afternoon, this morning |
| 09:21.18 | fcois | but Im in France and digium wake up when we are at 3pm in france! |
| 09:21.21 | tzafrir_laptop | lsodi: "Extension 's' in context 'vvvrec1' from 'XXXXXXX' does not exist." |
| 09:22.14 | fcois | I could contact them for others problems and I could have responses in the day, but with that problem no responses |
| 09:28.48 | fcois | tzafrir_laptop : you have a contact in digium ? |
| 09:29.29 | lsodi | tzafrir_laptop: I have exten => 333,1,Goto(minu,s,1) and [minu] has 's' |
| 09:29.31 | steliosk | lsodi : You call comes in with CID of XXXXXXX but your dialplan expects a 333, and then it fails as there is none |
| 09:29.33 | tzafrir | fcois, none special |
| 09:29.50 | steliosk | tzafrir : hi ! |
| 09:30.12 | rob0 | Why: 08:38 < fcois> I need to recompil the busybox with a tftpd server |
| 09:30.31 | rob0 | You bought an embedded appliance which did not work? |
| 09:30.44 | tzafrir | fcois, "this morning" is still "night" there |
| 09:30.49 | fcois | rob0, because, I need to have autoprovisioning |
| 09:30.54 | rob0 | or you just want to enhance the functionality? |
| 09:31.19 | tzafrir | Unless they have a support centre at a saner time zone :-) |
| 09:31.22 | fcois | tzafrir, yes I know but this afternoon for me I hop to have a response |
| 09:31.48 | rob0 | 3.5 hours until they open |
| 09:31.50 | fcois | rob0, I just need to have this functionality! (tftpd) |
| 09:33.09 | fcois | but last week, I sent some emails without responses! |
| 09:33.37 | fcois | if I received an email at night, it doesnt matter! |
| 09:33.40 | Uatec | hi there, i have an incoming call from my VOIP provider and i'm trying to route it to my SIP proxy, when i Dial() to my sip proxy i get this error: http://rafb.net/p/XDnQBn97.html |
| 09:34.02 | Uatec | i don't know why it's using that URI to authenticate |
| 09:34.42 | Uatec | i've specified a username, password and domain in my sip.conf file |
| 09:34.46 | tzafrir | lsodi, what does context [vvvrec1] have? Could you pastebin it (and every context it include=> -s)? |
| 09:34.54 | Uatec | but it's not using them to auth, so my SIP proxy isn't accepting the connection |
| 09:35.02 | Uatec | how can i make it try to auth with the correct details |
| 09:35.04 | Uatec | ? |
| 09:35.34 | tzafrir | steliosk, what's up? |
| 09:35.45 | *** join/#asterisk whymarkwh (n=dsfsdfsd@196.211.34.2) |
| 09:36.00 | whymarkwh | hi there anyone alive? |
| 09:36.17 | Uatec | nope, nobody |
| 09:36.48 | steliosk | tzafrir : playing around with a Philips AP200 sip/dect bridge |
| 09:37.20 | steliosk | and currently updating a windows xp machine ;) as all setup software is windows only.... |
| 09:37.32 | whymarkwh | downloaded iso of adminsparadise distro with asterisl 1.4 and webgui for fax mail and basic gui for adding extensions where can one find documentation regarding this i tried everywhere |
| 09:37.54 | whymarkwh | it looks to me ike their site got hacked you can only buy viagra there now |
| 09:38.12 | rob0 | life is hard |
| 09:38.56 | tzafrir | You don't have top make them so hard |
| 09:40.01 | *** join/#asterisk oej (n=olle@ns.webway.se) |
| 09:42.00 | lsodi | tzafrir: http://pastebin.com/dede557e under extensions.conf starting at line 33 |
| 09:43.42 | Uatec | does anybody have any idea how to make my asterisk use the correct username and password as specified in the sip.conf file? |
| 09:44.36 | *** join/#asterisk Rico29 (n=Rico@vau75-12-88-181-4-88.fbx.proxad.net) |
| 09:45.12 | *** join/#asterisk talntid (n=t@c-67-185-237-158.hsd1.wa.comcast.net) |
| 09:45.13 | tzafrir | lsodi, well, this is quite obvious. If a call comes in through span 2 whose number is not 333, it won't be handled |
| 09:45.51 | tzafrir | You only handle extension 333 in the context vvvrec1 |
| 09:47.42 | *** join/#asterisk shinao1 (n=shinao1@41.219.232.129) |
| 09:49.38 | *** join/#asterisk mikkel (n=mikkel@84-238-113-66.u.parknet.dk) |
| 09:49.45 | fcois | rob0: you have a solution to add a tftps server? |
| 09:49.47 | lsodi | tzafrir: pri debug output http://pastebin.com/d599e71c2 line 20. should there be dialed extension? |
| 09:50.06 | jblack | I'm trying to make iaxmodem work. I can get it to register to my * server, but atdt anything results in an error |
| 09:50.23 | tzafrir | you actually intended to dial 333? |
| 09:50.48 | lsodi | yes |
| 09:51.07 | tzafrir | Here's a simple way to debug this: |
| 09:51.38 | tzafrir | exten => _X.,1,NoOp(Got a call to number ${EXTEN}) |
| 09:52.17 | tzafrir | add that line to the context in question |
| 09:52.47 | rob0 | fcois, I would be qualified to do that, indeed. But I don't have time to do it. |
| 09:53.26 | fcois | rob0, ok it dont help me :( |
| 09:53.39 | fcois | rob0, and you can give me an advice? |
| 09:53.44 | rob0 | hire someone |
| 09:54.03 | Uatec | :'( sourceforge have broken them selves |
| 09:54.07 | Uatec | i'm trying to download wireshark |
| 09:54.13 | Uatec | but i can't get past the stupid advert page |
| 09:54.19 | Uatec | I DON'T WANT YOUR CRAPPY SHIRT, I WANT WIRESHARK |
| 09:56.27 | *** join/#asterisk RoyK (n=roy@ip-177-22-149-91.dialup.ice.no) |
| 10:06.28 | *** join/#asterisk lvl- (n=lvl@101.248.dhcp.lht.hhs.nl) |
| 10:09.27 | *** join/#asterisk ming_zym (n=ming_zym@nat/yahoo/x-8bdc35b08f2015d3) |
| 10:11.05 | *** join/#asterisk BipBip (n=BipBip@194.65.5.235) |
| 10:11.29 | *** join/#asterisk ming_zym (n=ming_zym@nat/yahoo/x-06e9d4ed35dfa41b) |
| 10:14.17 | tzafrir | Uatec, isn't there a direct link there? |
| 10:14.56 | tzafrir | They used to be completely broken, but now they actually do provide a decent redirect that wget understands |
| 10:16.24 | tzafrir | http://www.wireshark.org/download.html - some mirrors |
| 10:18.13 | tzafrir | http://heanet.dl.sourceforge.net/sourceforge/wireshark/wireshark-1.0.0.tar.gz |
| 10:21.41 | *** join/#asterisk dakol (n=dakol@vbo91-2-82-239-204-13.fbx.proxad.net) |
| 10:21.46 | dakol | hello *body |
| 10:22.18 | lsodi | tzafrir: no help adding noop function, still same output with zap device, when I add same context to sip device it works, so it has to do something with zap configuration. |
| 10:25.44 | steliosk | lsodi : this is a dialplan issue not a zaptel one |
| 10:26.41 | steliosk | lsodi : change the 333 exten to something like _x. to verify it |
| 10:27.07 | *** join/#asterisk svenna_ (n=svenna@p548D038F.dip0.t-ipconnect.de) |
| 10:28.47 | *** join/#asterisk cpm (n=Chip@pdpc/supporter/active/cpm) |
| 10:34.08 | lsodi | steliosk: changed 333 to _X. and still same result, calling from zap device I get "Extension 's' in context 'vvvrec1'...." |
| 10:34.31 | *** join/#asterisk keulin (n=cray@247.126.64-86.rev.gaoland.net) |
| 10:34.41 | lsodi | but no problems when using sip device and same context |
| 10:35.45 | lsodi | < Called Number (len= 3) [ Ext: 1 TON: Unknown Number Type (0) NPI: ISDN/Telephony Numbering Plan (E.164/E.163) (1) '' ] |
| 10:35.57 | lsodi | should this line contain dialed extension? |
| 10:37.54 | whymarkwh | lsodi: put a NoOp to debug*to see what digits you are getting from the call. |
| 10:38.21 | whymarkwh | paste your dialplan |
| 10:41.10 | *** join/#asterisk Ebola (n=Ebola@unaffiliated/ebola) |
| 10:41.26 | lsodi | http://pastebin.com/d193dffb3 |
| 10:42.26 | *** join/#asterisk gr0mit (n=tim@CSC.ge4-0-0.401.ar1.BBS1.gblx.net) |
| 10:42.40 | Uatec | hi, i'm dialing my sip proxy, and asterisk is receiving from it a WWW Challenge, to which it's supposed to then send the username and password. But it's not, i'm just getting "Failed to authenticate on INVITE" in my CLI |
| 10:42.43 | rob0 | exten => cpm,1,NoOp(Zap/${CPM} SIP/${coffee}) |
| 10:42.57 | Uatec | how can i tell asterisk just to try again, as it does with my provider? |
| 10:45.32 | cpm | hmmmm, IAX/${coffee} I think |
| 10:45.46 | cpm | peers with coffee |
| 10:45.52 | cpm | or is that pees? |
| 10:46.02 | rob0 | IAX coffee is icky |
| 10:46.15 | cpm | hugs IAX |
| 10:46.26 | *** join/#asterisk dvnull (i=dvnull@cpe-74-72-223-73.nyc.res.rr.com) |
| 10:46.31 | cpm | pours it a pot at a time, coffee trunking |
| 10:46.48 | rob0 | good idea! |
| 10:46.52 | dvnull | does anyone here know how to do callerid spoofing via asterisk |
| 10:46.59 | dakol | juste to be sure, a IAX2 trunk can not be registered like a SIP trunk ? |
| 10:47.18 | rob0 | core show function CALLERID |
| 10:47.56 | *** join/#asterisk Mavvie (n=edwin@ppp121-44-49-247.lns10.syd7.internode.on.net) |
| 10:51.59 | dakol | got a question (a problem as usual :), i have 2 PBX connected via a IAX2-trunk. On the first on PBX-1, i have a ring-group which points to Extension@PBX-2, if no answer, the call is forwarded to a local extesnion. When a call is made, if the first phone (on PBX-2) is not connected, i am instantanly forwarded to the voicemail |
| 10:52.56 | dakol | is there a way to make PBX-1 aware of availables phones on PBX-2 ? |
| 10:53.23 | tzafrir_laptop | dakol: 'switch =>' in the dialplan |
| 10:53.41 | dakol | nota: i use FreePBX |
| 10:54.29 | tzafrir_laptop | dakol: here we answer Asterisk questions, FreePBX questions go to #freepbx. I answered your Asterisk question |
| 10:54.35 | dakol | okix |
| 10:54.52 | tzafrir_laptop | Another keyword: dundi |
| 10:54.56 | dakol | understood, i have got exten=> and not switch=> |
| 10:54.59 | dakol | thank you |
| 10:55.18 | *** join/#asterisk sysreq (n=sysreq@unaffiliated/sysreq) |
| 10:56.26 | lsodi | problem solved other end of e1 line didn't send dialed number |
| 10:58.09 | Uatec | WTF? my sip proxy is sending back 401: www-authenticate |
| 10:58.12 | Uatec | and asterisk is just ACK |
| 10:58.15 | Uatec | and then failing |
| 10:58.16 | Uatec | WTF? |
| 10:58.18 | Uatec | STUPID asterisk |
| 10:58.24 | Uatec | don't ack it, send the bloody credentials |
| 10:58.31 | *** join/#asterisk mort_gib (n=mjensen@8.Red-81-35-165.dynamicIP.rima-tde.net) |
| 11:01.16 | *** join/#asterisk lesouvage (n=lesouvag@cc341200-a.assen1.dr.home.nl) |
| 11:02.37 | Uatec | does anybody know why this SIP response: |
| 11:03.19 | *** join/#asterisk lmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 11:03.19 | *** mode/#asterisk [+o lmadsen] by ChanServ |
| 11:03.34 | Uatec | http://rafb.net/p/4Cm75P74.html |
| 11:03.55 | Uatec | would result in asterisk just ACKing it, rather than replying with the correct credentials? |
| 11:05.32 | yang | hello lmadsen tzafrir_laptop |
| 11:05.40 | lmadsen | morning |
| 11:05.58 | Uatec | hi there |
| 11:06.24 | *** part/#asterisk bsaxon (n=bryantsa@71-8-14-108.dhcp.leds.al.charter.com) |
| 11:12.44 | *** join/#asterisk sysreq (n=sysreq@unaffiliated/sysreq) |
| 11:15.23 | *** join/#asterisk oej (n=olle@ns.webway.se) |
| 11:15.29 | JT | coppice: ping |
| 11:15.44 | coppice | pong |
| 11:15.55 | talntid | pizzong!!! |
| 11:16.10 | talntid | i'm the echo from the walls. sorry. |
| 11:16.25 | talntid | you may enable echo cancellation if you like. |
| 11:18.18 | Uatec | enables talntid cancellation |
| 11:18.25 | Uatec | ping? |
| 11:18.28 | talntid | :( |
| 11:19.02 | Uatec | disables talntid cancellation and gives talntid a jam doughnut with extra sugar |
| 11:19.06 | talntid | I'm the pong to your ping, baby! |
| 11:19.45 | Uatec | lol |
| 11:20.11 | talntid | great pickup line at a lan party. |
| 11:21.03 | lmadsen | talntid: the kind of people who go to LAN parties are not the kind of people I *want* to pick up... |
| 11:21.19 | Uatec | i've only ever seen 3 girls at a lan party |
| 11:21.32 | Uatec | one was my mates GF, one was a moose and the 3rd was 11 |
| 11:21.36 | talntid | my GF plays COD4, on occasion :) |
| 11:21.41 | Uatec | none of whom i'm going to be using that chat up line on |
| 11:21.47 | lmadsen | Uatec: heh |
| 11:21.56 | talntid | lol. you sure? |
| 11:22.07 | lmadsen | depends how hot the friends GF is |
| 11:22.15 | Uatec | lol |
| 11:22.24 | talntid | righto. |
| 11:22.29 | Uatec | she's alright, big babs, but not my style |
| 11:22.39 | Uatec | also, hitting on friend's girlfriends is not my style |
| 11:22.42 | talntid | but, it probably wouldn't matter, if she's hot she won't understand that line. |
| 11:22.58 | lmadsen | female friend of mine plays rainbow six and kids guys butts on Live :) |
| 11:23.03 | coppice | Uatec: you have style and still spend your time on IRC? :-\ |
| 11:23.15 | talntid | zing! |
| 11:23.32 | lmadsen | has *a* style |
| 11:23.39 | talntid | Score: coppice, 1 Uatec, -0 |
| 11:23.57 | lmadsen | hey guess who hates MS based VPns |
| 11:24.12 | talntid | ooh, ooh!@ pick me! |
| 11:24.15 | *** join/#asterisk DagMoller (n=aguirre@unaffiliated/dagmoller) |
| 11:24.39 | talntid | jumps up and down excitedly. |
| 11:24.44 | lmadsen | hehehe |
| 11:24.45 | Uatec | lol, thanks of the scoring talntid |
| 11:24.54 | Uatec | but i'm being paid to be on IRC right at the moment |
| 11:24.56 | talntid | pick meeeeee!!!!! |
| 11:25.20 | Uatec | also, i didn't say i had style, i just said that that particular style wasn't mine.. |
| 11:25.26 | lmadsen | I'm only here to watch the crazyness while I constantly keep running the VPN connect script waiting for one of these times for it to actually connect and work |
| 11:25.29 | talntid | no. you're getting paid to do your job, just so happens your boss has no idea you're on IRC, and doesn't really care what you do |
| 11:25.45 | Uatec | he pays me to do asterisk stuff |
| 11:25.55 | lmadsen | my boss is an asshole |
| 11:25.57 | Uatec | and he can see my screen from here |
| 11:25.58 | Uatec | lol |
| 11:25.59 | lmadsen | is self employed |
| 11:26.02 | Uatec | lol |
| 11:26.02 | talntid | my boss rocks. |
| 11:26.18 | talntid | no set hours. salary. |
| 11:26.23 | talntid | so long as shit works. it's all good. |
| 11:26.34 | talntid | must spend 5 hours per week at office |
| 11:26.54 | talntid | other than that, if shit runs good. i can do whatever. :) |
| 11:27.09 | talntid | god i hope asterisk doesn't turn out to be a headache :P |
| 11:28.26 | Uatec | lol |
| 11:28.31 | Uatec | Asterisk Business Edition sure is one |
| 11:28.39 | talntid | blah |
| 11:28.42 | Uatec | so much so that i persuaded my boss to let me go Open source with this project |
| 11:28.44 | talntid | i'm using free one |
| 11:28.49 | lmadsen | talntid: you must not be doing much with asterisk then |
| 11:28.50 | Uatec | except i need to use Radius for authentication |
| 11:28.56 | Uatec | which asterisk seems incapable of doing |
| 11:28.59 | Uatec | so i'm using openser |
| 11:29.01 | Uatec | which is a nightmare too |
| 11:29.06 | Uatec | and doesn't have a decent IRC channel |
| 11:29.12 | talntid | why do you say that, lmadsen? |
| 11:29.23 | talntid | running 27 SIP phones over a PRI |
| 11:29.37 | lmadsen | talntid: because if things are going so smoothly, then you can't be doing anything other than what has been tested throughly :) |
| 11:29.38 | talntid | 30,000 outbound calls per month |
| 11:29.55 | tzafrir_laptop | talntid: so clients can call you 24h/day even when you're at home? |
| 11:30.14 | rob0 | ugh, telephone spam |
| 11:30.18 | talntid | i only have 1 client, tzafrir_laptop |
| 11:30.35 | talntid | rob0, outbound call center for businesses. yeah. |
| 11:30.42 | talntid | only call businesses though |
| 11:31.24 | Uatec | ARGH |
| 11:31.26 | Uatec | talntid, i hate you |
| 11:31.34 | Uatec | you're the one who spams me at work |
| 11:31.38 | talntid | tzafrir_home, and they only work from 7:00 to 3:30 |
| 11:31.44 | talntid | oh? :) |
| 11:31.46 | lmadsen | talntid: don't worry... I hate you too... but I just hate everyone equally |
| 11:31.53 | talntid | unlikely, what does your company do? :) |
| 11:31.57 | tzafrir_laptop | talntid: so I guess that when that client wants to call you they will get you |
| 11:32.02 | Uatec | IT support |
| 11:32.05 | Uatec | what does your company do? |
| 11:32.12 | talntid | we don't call you then |
| 11:32.17 | Uatec | what does your company do? |
| 11:32.22 | talntid | we put all the ads on the back of grocery store reciepts |
| 11:32.27 | talntid | you know, the coupons and crap. |
| 11:32.45 | Uatec | lol |
| 11:32.59 | Uatec | so you could infact call us up and try to sell us advertsing |
| 11:33.27 | talntid | we filter by popular advertisers... |
| 11:33.37 | talntid | lube & oil, car wash, nail salons... |
| 11:33.43 | talntid | resturaunts.. |
| 11:33.57 | *** join/#asterisk ming_zym (n=ming_zym@123.103.29.229) |
| 11:34.01 | talntid | but. if you'd like to give me your phone #, i can put you in the list. hahah ;) |
| 11:35.09 | lmadsen | 4163653.... oh wait |
| 11:36.08 | talntid | ;) |
| 11:37.02 | Uatec | lol NO! |
| 11:37.07 | talntid | hey, Uatec, you can't hate me much |
| 11:37.12 | talntid | i have hot chicks on my website. |
| 11:37.15 | talntid | :) |
| 11:37.25 | *** join/#asterisk jm|home (n=jm|home@dilbert.jamiem.com) |
| 11:37.33 | coppice | its an H5N1 information site? |
| 11:37.54 | tzafrir_laptop | lmadsen: not nice of you to give them my number |
| 11:38.05 | lmadsen | tzafrir_home: who said I was a nice guy? :) |
| 11:38.51 | talntid | You should be expecting about 80 phone calls tomorrow. |
| 11:39.40 | JT | coppice: is it still possible to compile spandsp with asterisk? |
| 11:40.22 | talntid | if you use the glue compiler, yes |
| 11:40.24 | coppice | as far as I know 0.0.4 works with the stuff in add ons, and the agx add ons at sourceforge |
| 11:40.30 | talntid | well, glue++ |
| 11:41.05 | tzafrir_laptop | agx says his stuff works with 0.0.4pre16 |
| 11:41.27 | Uatec | talntid, i dont' know you to hate you... |
| 11:41.33 | Uatec | but i don't like you just becuase you run a porn site |
| 11:41.40 | *** join/#asterisk erojasv (n=erojasv@190.40.53.52) |
| 11:42.02 | coppice | it should work with any of the recent 0.0.4pre<somethings>. However, APIs are changing slightly in the 0.0.5pre<something> series |
| 11:42.03 | talntid | not just any porn site. |
| 11:43.03 | coppice | what's wrong with running a porn site.... at least until someone works out a second method of making money from the internet |
| 11:43.21 | talntid | Uatec, you know what? I don't like you either. |
| 11:43.35 | talntid | Uatec, Your name should be spelled with a V at the front, instead of a U. |
| 11:44.14 | talntid | AND. you wear spongebob underwear. |
| 11:47.00 | *** join/#asterisk keulin (n=cray@247.126.64-86.rev.gaoland.net) |
| 11:47.24 | jblack | wtf? |
| 11:48.16 | jblack | talntid: You do know that lmadsen wrote the book on asterisk.. literally? |
| 11:48.30 | lmadsen | lies! |
| 11:48.39 | jblack | I seen it! In the black and white! |
| 11:48.43 | jblack | well, blue and white |
| 11:48.52 | jblack | Yeah. black and white |
| 11:49.00 | talntid | there you are mr james. |
| 11:49.10 | rob0 | suggests getting the color version |
| 11:49.14 | jblack | Causing some hate and discontent? |
| 11:49.25 | talntid | nah. i'm being good. :) |
| 11:49.37 | jblack | rob0: That's what threw me. The Asterisk is reverse blue/white. The authors are black and white. |
| 11:49.48 | talntid | lmadsen didn't write the book on Asterisk... asterisk wrote the book on lmadsen... :) |
| 11:50.04 | rob0 | Book 'em, Danno. |
| 11:51.09 | talntid | Hey James. Living in the city sucks. |
| 11:51.26 | talntid | last 4 hours, sirens from police like 6 houses down. |
| 11:51.29 | jblack | Tell me about it. |
| 11:51.35 | talntid | some car accident. drunk driver vs house. |
| 11:52.00 | jblack | I would be far, far, far from civilization if I could get a good connection. |
| 11:52.17 | jblack | Who won? |
| 11:52.27 | talntid | well, i'd say neither. |
| 11:52.30 | *** join/#asterisk lirakis_work (n=lirakis@65.200.191.241) |
| 11:52.51 | *** join/#asterisk gr0mit (n=tim@82.58.187.81.in-addr.arpa) |
| 11:52.54 | talntid | the car is now a living room conversation piece. |
| 11:53.11 | jblack | Yay! Drive-ins are coming back! |
| 11:53.11 | rob0 | The doctors and lawyers won, of course. |
| 11:53.25 | talntid | neat thing is, the police are STILL going door to door |
| 11:53.30 | talntid | asking if someone has seen the driver. |
| 11:53.41 | jblack | lol. |
| 11:53.53 | talntid | evidently, it was a stolen car, and the driver is nowhere to be found. they have been here 3 times asking if I have seen them. |
| 11:54.05 | jblack | oh, geeze |
| 11:54.13 | talntid | I have 3 business cards from the same detective. saying "If you see them, call me!" |
| 11:54.32 | jblack | If you see them, does that mean you'll need to call thrice? |
| 11:54.50 | talntid | i'm unsure, when he comes back, i'll ask the question and get back to you on that. |
| 11:55.46 | rob0 | It could be done easily with some simple dialplan logic. |
| 11:56.03 | talntid | rob0 speaks the truth. |
| 11:56.25 | lmadsen | jblack: I still haven't determined why you needed to mention that I helped write the book :) |
| 11:56.34 | talntid | also, I'm quite disappointed, jblack. |
| 11:56.43 | jblack | lmadsen: Because I'm jealous. =) |
| 11:56.50 | talntid | 16meg down, 2 up at home. nowhere near the bandwidth work has. |
| 11:57.00 | talntid | also comcast |
| 11:57.22 | jblack | That's better than what I have. |
| 11:57.43 | talntid | yes, but still. i figured it would be at least somewhat like the work connection |
| 11:57.57 | jblack | I run a vpn over carrier pigeons. |
| 11:57.59 | talntid | nope. it downloads at about 900 KB/s |
| 11:58.08 | talntid | sneakernet VPN. |
| 11:58.09 | rob0 | RFC 1149 |
| 11:58.14 | jblack | Yeah. Triple what I get on a good day. |
| 11:58.27 | jblack | Yup. The throughput is great, but the latency _sucks_ |
| 11:58.37 | *** join/#asterisk Shazaum (n=shazaum@200.175.61.250.static.gvt.net.br) |
| 11:59.17 | talntid | uh. rob0. that's disturbing you knew that off the top of your head. |
| 11:59.24 | rob0 | I have 5 cats, and I'll tell you, cat5 is VERY bad for RFC 1149 transport. |
| 11:59.26 | jblack | Who doesn't? |
| 11:59.41 | mgdm | How are cats 1 to 4? |
| 11:59.50 | rob0 | cats 1 through for aren't very nice either |
| 12:00.00 | rob0 | s/for/4/ |
| 12:00.01 | mgdm | :( |
| 12:00.59 | rob0 | They like the yummy birdies, however; less cat food to buy. |
| 12:01.13 | jblack | I can't imagine how bad the dropped packets are on cats5 |
| 12:01.59 | talntid | ttl=9 |
| 12:02.22 | jblack | I did manage to route a few gigabytes through id-10-t though. That only worked because he didn't realize it was a test |
| 12:02.36 | talntid | was that me? |
| 12:02.41 | jblack | No. My brother |
| 12:03.10 | jblack | He's a... artist.. free spirit... Bum, when you get down to it |
| 12:05.11 | jblack | The last time I saw him, he "borrowed" 300 bucks from me. |
| 12:06.03 | *** part/#asterisk dakol (n=dakol@vbo91-2-82-239-204-13.fbx.proxad.net) |
| 12:06.59 | JT | coppice: hrm, where abouts are app_txfax and rxfax kept these days? |
| 12:08.06 | *** join/#asterisk shido6 (n=shido6@74-130-51-159.dhcp.insightbb.com) |
| 12:08.37 | coppice | try sourceforge.net/projects/agx-ast-addons |
| 12:10.31 | *** join/#asterisk oktay (n=oktay@85.98.52.238) |
| 12:10.56 | oktay | howdy. i have asterisk running but nothing is listening on port 5060.. is this normal? |
| 12:11.17 | *** join/#asterisk [TK]D-Fender (n=joe@216.191.106.163) |
| 12:13.35 | jblack | morning |
| 12:14.20 | jblack | oktay: I suppose it might be, if sip is disabled. |
| 12:14.27 | *** join/#asterisk waKKu (n=ugabuga@unaffiliated/wakku) |
| 12:16.44 | JT | coppice: ah, so the those apps are no longer provided with spandsp? |
| 12:16.48 | *** join/#asterisk bsaxon (n=bsaxon@12.68.234.174) |
| 12:17.01 | *** join/#asterisk ming_zym (n=ming_zym@123.103.29.229) |
| 12:17.39 | jblack | I've been trying to get iaxmodem working, so that I can fax out. I'm having a problem with it erroring out on ATDT. Has anyone had any luck with it? |
| 12:17.50 | coppice | JT: they are no longer provided. from asterisk 1.6 i understand there are apps for spandsp in the addons for * itself |
| 12:18.01 | *** part/#asterisk ming_zym (n=ming_zym@123.103.29.229) |
| 12:19.40 | JT | ah right |
| 12:19.43 | JT | thanks coppice |
| 12:20.35 | coppice | those apps can probably move from addons into asterisk, as I'm making the licence LGPL for new versions of spandsp |
| 12:21.08 | *** join/#asterisk talntid (n=t@c-67-185-237-158.hsd1.wa.comcast.net) |
| 12:21.10 | *** join/#asterisk mltlnx (n=mltlnx@pool-96-232-207-89.nycmny.east.verizon.net) |
| 12:23.15 | JT | cool :) |
| 12:26.18 | *** join/#asterisk docelmo (n=chatzill@206.248.239.194) |
| 12:28.00 | *** join/#asterisk sergee (n=serg@voip1.west-call.com) |
| 12:28.01 | *** join/#asterisk gr0mit (n=tim@144.187.4.30) |
| 12:30.16 | *** join/#asterisk _ys (i=yuri@91.151.196.254) |
| 12:30.17 | *** join/#asterisk klimonso (n=eddy@dxb-b12099.alshamil.net.ae) |
| 12:30.40 | klimonso | hi, when i record using an extension , how can i check the recording? where can i check it? |
| 12:33.18 | *** join/#asterisk msetim (n=msetim@200.195.161.164) |
| 12:38.18 | klimonso | hi, when i record using an extension , how can i check the recording? where can i check it? |
| 12:38.31 | klimonso | is there anyone that is alive? |
| 12:38.44 | fcois | I never do it |
| 12:38.56 | fcois | I always insert a mp3 or other... |
| 12:39.01 | *** join/#asterisk vector (n=vector@host-178-246-220-24.midco.net) |
| 12:39.11 | klimonso | i record it by dialing *77 |
| 12:39.16 | klimonso | and i check it using *99 |
| 12:39.26 | fcois | I dont know never do it |
| 12:39.27 | klimonso | but where i can find the file? |
| 12:39.43 | fcois | find / -name name_file |
| 12:39.59 | klimonso | where 2 find it |
| 12:40.02 | fcois | sounds are in /var/lib/asterisk/sounds/ |
| 12:42.41 | klimonso | thanks man |
| 12:44.43 | klimonso | oops |
| 12:44.53 | klimonso | but i cant find recording shit |
| 12:44.56 | klimonso | :S |
| 12:46.13 | [TK]D-Fender | klimonso: Where did you MAKE those 2 extensions record their files? |
| 12:47.30 | klimonso | it is by default, where is it usually? |
| 12:47.38 | *** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 12:48.15 | *** join/#asterisk sergee (n=serg@voip1.west-call.com) |
| 12:48.27 | [TK]D-Fender | klimonso: where fcois told you. And what does your Record line look like? |
| 12:49.25 | *** join/#asterisk tobias (n=tobias@user-0c998nt.cable.mindspring.com) |
| 12:49.53 | klimonso | i cant find it |
| 12:50.19 | klimonso | see what i am doing, i am picking up , dialing *77 speaking and then hang up.. i dial *99 and i can listen to it |
| 12:50.23 | klimonso | but i need that file |
| 12:52.10 | [TK]D-Fender | klimonso: Sorry, but *77 and *99 are not part of Asterisk, but rather from whatever generated your dialplan. If you don't even know where that is, then its pretty much pinned as being from a GUI which is not supported here. |
| 12:57.04 | *** join/#asterisk bobbym (n=bob@unaffiliated/bobbym) |
| 12:57.39 | *** join/#asterisk [intra]lanman (n=lanman@209.85.58.2) |
| 12:58.50 | *** join/#asterisk sergee (n=serg@voip1.west-call.com) |
| 13:00.11 | fcois | klimonso : execute asterisk -r with debug and verbose to see where are the files... |
| 13:02.44 | *** join/#asterisk GiantPickle (n=GiantPic@S01060016b600537f.gv.shawcable.net) |
| 13:03.30 | *** join/#asterisk PodMan99a (n=PodMan99@78-86-189-73.zone2.bethere.co.uk) |
| 13:04.07 | PodMan99a | hey all... these may not be classed as good.. but are there any good free gui based asterisk setups out there that can be recomended? |
| 13:06.30 | [TK]D-Fender | PodMan99a: Nope. all have rather polrized flaws. |
| 13:07.21 | *** join/#asterisk sergee (n=serg@voip1.west-call.com) |
| 13:07.39 | PodMan99a | damn.... any suggestions... lol... have some small issues which I do not know how to combat i.e. cannot dial some users from my sip but can dial others |
| 13:07.46 | PodMan99a | all are setup the same |
| 13:08.12 | *** join/#asterisk arbuser (n=jonathan@arbitrary.frogfoot.net) |
| 13:08.51 | arbuser | quiet in here ;) |
| 13:09.12 | *** join/#asterisk anonymouz666 (n=anonymou@201.19.205.41) |
| 13:09.19 | [TK]D-Fender | PodMan99a: What are you using now? |
| 13:09.43 | PodMan99a | asterisk now but changed to asterisk all configs are done from files |
| 13:09.52 | PodMan99a | installed about 6 months ago but only now playing with it |
| 13:10.48 | [TK]D-Fender | PodMan99a: where are these suers located relative to *? |
| 13:10.53 | [TK]D-Fender | users* |
| 13:11.06 | *** join/#asterisk gandhijee (n=root@host-66-202-34-165.spr.choiceone.net) |
| 13:11.20 | PodMan99a | server in rack (datacenter) .. me office sip twinkle |
| 13:11.22 | *** join/#asterisk sergee (n=serg@voip1.west-call.com) |
| 13:11.43 | PodMan99a | really... either need a good gui or idiots guide to things |
| 13:12.05 | PodMan99a | although think i have configs kinda under control just cant debug and expand |
| 13:12.20 | [TK]D-Fender | PodMan99a: Well I'm going out on a limb here guessing you have NAT issues. Read up : |
| 13:12.22 | [TK]D-Fender | ~sipnat |
| 13:12.23 | jbot | [~sipnat] Quick guide on configuring * + SIP behind NAT : http://www.aocomputing.net/?p=3 , otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
| 13:12.32 | arbuser | Hi All, I'd appreciate a point in the right direction. I'd like to write a phone app that allows people to call it, DTMF a pin number and then listen to a text to speech message. The entire call must be recorded. What should I use and where should I go next? |
| 13:12.35 | [TK]D-Fender | PodMan99a: And for anything GUI, go find their support channel. |
| 13:13.10 | [TK]D-Fender | arbuser: What are you looking to "record" So far the other end isn't talking. |
| 13:13.19 | PodMan99a | not using gui at the moment but would like one... only to assist in manageing /monitoring calls... creation of users ??... |
| 13:13.57 | arbuser | Fender, the other end is going to say their name and leave a message. |
| 13:14.07 | [TK]D-Fender | PodMan99a: *-GUI would probably be the closest thing. |
| 13:14.24 | rob0 | A GUI might help with the high-level stuff (managing extensions and users), but not with the low-level details. |
| 13:14.26 | [TK]D-Fender | arbuser: "core show application record" |
| 13:15.23 | arbuser | Fender, other than the fact that those are all english words I have no idea what you are saying... Can you point me to a good "I'm a noob" tutorial? |
| 13:15.28 | *** join/#asterisk zeedo (n=zeedo@reboot-robot.net) |
| 13:16.00 | arbuser | Fender, also, is Asterisk Now (tm) a good way forward for my level of noobness considering what I want to achieve? |
| 13:16.40 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 13:16.43 | [TK]D-Fender | arbuser: thats a "I should be capable of recognizing this is an * CLI command that will give me the exact instructions on the dialplan application that will let me do what I want" |
| 13:17.01 | [TK]D-Fender | ^ :) |
| 13:18.22 | [TK]D-Fender | arbuser: And you said "listen to a text to speech message", and later alluded to having to actually record something (While listening? Or separate?) |
| 13:18.40 | [TK]D-Fender | arbuser: And No, GUI's would not be a good bet for custom stuff. |
| 13:19.01 | arbuser | Fender, I haven't even got an instance of Asterisk running yet... What I'm after is a good kick in the right direction... |
| 13:19.15 | *** join/#asterisk mcfloppy (n=info@88-134-186-152-dynip.superkabel.de) |
| 13:19.17 | mcfloppy | hello |
| 13:19.34 | [TK]D-Fender | arbuser: For the right direction : |
| 13:19.36 | [TK]D-Fender | ~book |
| 13:19.36 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com, or see ~buybook |
| 13:19.38 | [TK]D-Fender | ^^ |
| 13:19.49 | [TK]D-Fender | arbuser: And go install * and get playing. |
| 13:19.58 | arbuser | cool |
| 13:20.51 | arbuser | One last question, Will i be able to script this sort of thing successfully on a virtualised OS running under VMWare? |
| 13:21.20 | mcfloppy | i have a clean asterisk. i can call the server and it says me hello world. now i try to use callfiles. how can i enable the outgoing calls? i will call my home internal isdn phone with id *16 |
| 13:21.30 | *** join/#asterisk jaytee (n=jforde@unaffiliated/jaytee) |
| 13:23.17 | *** join/#asterisk tobias (n=tobias@cpe-066-026-084-121.nc.res.rr.com) |
| 13:24.08 | [TK]D-Fender | arbuser: So far, probably. |
| 13:24.33 | [TK]D-Fender | arbuser: Dependsing on how you intend to get the call into * in the first place. PCI hardware is pretty much out. |
| 13:24.35 | mcfloppy | [May 12 15:16:58] NOTICE[9601]: chan_local.c:597 local_alloc: No such extension/context *16@default creating local channel |
| 13:24.44 | arbuser | Fender, what do you think of VXML? |
| 13:24.59 | mcfloppy | first row in the callfile: Channel: Local/16 |
| 13:25.05 | [TK]D-Fender | arbuser: Not part of * yet and not needed. |
| 13:25.16 | [TK]D-Fender | mcfloppy: PASTEBIN is your friend... use it. |
| 13:25.18 | [TK]D-Fender | ~pb |
| 13:25.19 | jbot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste |
| 13:25.20 | [TK]D-Fender | ^^^^^^^^^^^^^ |
| 13:26.01 | Uatec | hi, my asterisk implementation is inviting my SIP proxy, and the proxy is logically returning 401 unauthorised. Now, instead of sending along the credentials, asterisk is just saying ACK, and considering it a fail. |
| 13:26.14 | Uatec | Howe can i persuade asterisk to send the credentials the second time? |
| 13:26.18 | mcfloppy | ok |
| 13:26.23 | *** join/#asterisk muiro (n=muiro@cpe-76-189-84-108.neo.res.rr.com) |
| 13:27.26 | mcfloppy | [TK]D-Fender here: http://pastebin.com/m45e5d3d8 |
| 13:27.26 | [TK]D-Fender | Uatec: 401 = too late |
| 13:27.47 | *** join/#asterisk sergee (n=serg@voip1.west-call.com) |
| 13:28.02 | *** join/#asterisk mort_gib (n=mjensen@adsl-2-234.gibnet.gi) |
| 13:28.30 | [TK]D-Fender | mcfloppy: And please patebin your entire [defaut] context including the context header... |
| 13:28.32 | Uatec | [TK]D-Fender, but i'm sending "WWW-Authenticate:" in my reply |
| 13:28.36 | Uatec | it's not supposed to be too late |
| 13:29.34 | arbuser | Fender, thanks. Will be doing some reading ;) |
| 13:29.34 | Uatec | my VOIP provider returns 401 first time, then asterisk sends credentials and it's al ok |
| 13:29.35 | Uatec | all |
| 13:30.17 | [TK]D-Fender | Uatec: Not sure what to tell you at this poitn... |
| 13:30.45 | mcfloppy | http://www.rsp-design.de/extensions.ael |
| 13:30.47 | *** join/#asterisk caio1982 (i=caio1982@CAcert-br/caio1982) |
| 13:30.54 | mcfloppy | here is the complete extensions.ael |
| 13:31.40 | [TK]D-Fender | mcfloppy: do "dialplan show default" |
| 13:32.20 | mcfloppy | http://pastebin.com/m2fda2680 |
| 13:32.56 | *** join/#asterisk gandhijee (n=user@mail.win-ent.com) |
| 13:33.06 | [TK]D-Fender | mcfloppy: ... and "demo" please... |
| 13:33.28 | *** join/#asterisk anthm (n=anthm@CPE-72-131-113-50.wi.res.rr.com) |
| 13:34.00 | mcfloppy | http://pastebin.com/m224a4c23 |
| 13:34.05 | *** join/#asterisk Assid (n=assid@unaffiliated/assid) |
| 13:35.03 | [TK]D-Fender | mcfloppy: So what in there is supposed to match *16 / 16? |
| 13:35.57 | *** join/#asterisk moy (n=moyhu@nat/ibm/x-e3beafe682635463) |
| 13:36.04 | mcfloppy | should i add something? i have no expirience with asterisk |
| 13:37.00 | [TK]D-Fender | mcfloppy: what is *16 or 16 supposed to even represent? |
| 13:37.16 | *** join/#asterisk gr0mit (n=tim@CSC.ge4-0-0.401.ar1.BBS1.gblx.net) |
| 13:37.37 | mcfloppy | the internal number of my isdn phone |
| 13:38.07 | *** join/#asterisk sysreq (n=sysreq@unaffiliated/sysreq) |
| 13:39.07 | [TK]D-Fender | mcfloppy: Well "Local" refers to your dialplan. You are telling * to dial an EXTENSIONS in a specific place in your dialplan and this one doesn't happen to match anything. So go pick something that actually exists |
| 13:41.18 | mcfloppy | hmmm how can i say in the callfile i want to call the extern number 16? |
| 13:41.41 | Katty | ohai |
| 13:41.45 | Katty | hugs [TK]D-Fender |
| 13:41.55 | Katty | waves at mcfloppy |
| 13:42.24 | Uatec | so, [TK]D-Fender, i thought that the way it worked was that the client makes the INVITE request, and then the server sends bcak 401, then the client sends credentials, and everybody gets along |
| 13:42.26 | mcfloppy | hello Katty |
| 13:44.02 | *** join/#asterisk bbryant (n=brett@216.207.245.1) |
| 13:45.52 | *** join/#asterisk mgroman (n=stansmit@adsl-68-252-62-180.dsl.wotnoh.ameritech.net) |
| 13:46.46 | mgroman | Hello, May someone recommend inexpensive hardphones for use with VOIP? (How is Linksys?) |
| 13:46.56 | *** join/#asterisk putnopvut (n=putnopvu@216.207.245.1) |
| 13:46.56 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 13:47.12 | *** join/#asterisk mltlnx (n=mltlnx@m3a5f36d0.tmodns.net) |
| 13:47.15 | *** join/#asterisk sergee (n=serg@voip1.west-call.com) |
| 13:47.23 | Uatec | how is the sip server supposed to ask for credentials? |
| 13:48.13 | Uatec | mgroman, we SPA922s, they're PoE and relatively cheap (£90 or something) and pretty good handsets |
| 13:48.32 | *** join/#asterisk mknerd (i=3f951603@gateway/web/ajax/mibbit.com/x-d3814b3cfbda49df) |
| 13:48.47 | mknerd | anyone doing any AGI on the aa50? |
| 13:48.50 | mgroman | Uatec: Thank you |
| 13:48.54 | mgroman | googles |
| 13:50.34 | ManxPower | ~phones |
| 13:50.34 | jbot | [phones] http://bani.anime.net/phones/. While personal preference will dictate which phone works best for you, general consensus on a rough order of quality and suggestibility is as follows: Polycom (any), Aastra 480i, Aastra 5i Series, Cisco 7940+, Linksys SPA-9XX, Snom, and finally everything else. Do not consider Grandstream phones. Ever. places like such as |
| 13:51.07 | mknerd | anyone here use the AA50 at all? |
| 13:51.22 | ManxPower | mknerd: Chances are the answer is "no". |
| 13:51.32 | rob0 | actually there was one here |
| 13:51.38 | rob0 | fcois: ^^ |
| 13:51.53 | mknerd | manxpower: and what makes you say that? |
| 13:51.55 | ManxPower | Almost everyone here uses the open source version of Asterisk, not the commercial version of Asterisk. |
| 13:52.00 | fcois | rob0 : ^^ |
| 13:52.31 | mknerd | manxpower: i could really care less about what version of asterisk, I am more interested in agi on the embedded appliance |
| 13:52.45 | ManxPower | mknerd: Yeah, but we care. |
| 13:52.47 | fcois | mknerd: you use a aa50 incredible to find one! |
| 13:53.00 | rob0 | And BTW Digium should be open by now, albeit not fully caffeinated. |
| 13:53.18 | ManxPower | mknerd: edit extensions.conf, add your AGI to an extension, run it. I recommend using an AGI library, as it makes programming AGIs much earier. |
| 13:53.29 | fcois | mknerd: what is your pb with aa50 ? |
| 13:53.34 | mknerd | manxpower, and what should I script with? |
| 13:53.43 | ManxPower | mknerd: any language will work. |
| 13:53.48 | mknerd | there is no perl, no bash, no c |
| 13:53.55 | mknerd | no php |
| 13:53.57 | ManxPower | mknerd: perl and PHP seem to have the best supported AGI wrapper libs |
| 13:54.11 | mknerd | yeah, i want to use perl, but its no available |
| 13:54.12 | ManxPower | mknerd: Well, that's not really our problem. |
| 13:54.22 | mknerd | ok, now your just being a jerk |
| 13:54.30 | rob0 | It's on a CF card, no? |
| 13:54.32 | fcois | mknerd: my problem in aa50 is to have the tftpd command ! |
| 13:54.36 | rob0 | the rootfs? |
| 13:54.47 | ManxPower | You asked about Asterisk. I gave you info about Asterisk. If you have AA50 specific questions then you should contact Digium. |
| 13:54.53 | mknerd | no the rootfs is on 16mb internal |
| 13:55.09 | rob0 | 16mb ... what? |
| 13:55.17 | fcois | flash |
| 13:55.22 | mknerd | its editable |
| 13:55.36 | rob0 | so the answer is yes to this: 13:54 < rob0> It's on a CF card, no? |
| 13:55.47 | mknerd | no,its internal |
| 13:55.49 | rob0 | if 16MB isn't enough, buy a bigger one |
| 13:56.01 | mknerd | there is also a CF card |
| 13:56.04 | *** join/#asterisk gr0mit (n=tim@144.187.4.30) |
| 13:56.28 | *** join/#asterisk Defraz (i=t0tal@69.92.19.83) |
| 13:56.29 | [TK]D-Fender | mcfloppy: point your "channel" line to an actual exten that will do what you want. |
| 13:56.36 | fcois | there is flash memory and CF memory it isnt the same |
| 13:56.37 | mknerd | i have a 1gig installed, but the libraries for perl would have to be put on the 16mb |
| 13:56.59 | mknerd | mknerdno the rootfs is on 16mb internal |
| 13:57.00 | rob0 | I guess that the aa50 was designed to be a "works in most cases" solution, and not to be a fully-customizable PBX. |
| 13:57.11 | fcois | the problem with aa50 is that we can't have a system modified after a reboot! |
| 13:57.12 | mknerd | rob0, yeah, 1800$ later |
| 13:57.21 | ManxPower | mknerd: The "Appliance" part of Asterisk Appliance for the most part means "user does not modify" |
| 13:57.23 | mcfloppy | thx |
| 13:57.25 | mcfloppy | i try |
| 13:57.27 | mknerd | fcois, not true, you can save_config |
| 13:57.48 | fcois | yes I know but not all I want to do |
| 13:57.53 | *** join/#asterisk gbr_ (n=gbr@200.103.96.98) |
| 13:58.20 | mknerd | fcois: i know what you mean, I have a pretty lenghty script that runs and sets things up on reboot |
| 13:58.22 | fcois | I could translate it to french but I had to create a frecnh button in home.html and have home.html in save_config |
| 13:58.44 | fcois | after, I execute the script in the french button and thjat copy from CF to / |
| 13:58.52 | *** join/#asterisk puzzled (n=patrick@puzzled.xs4all.nl) |
| 13:59.01 | fcois | how can do it on boot??? |
| 13:59.10 | fcois | explain it to me :-) |
| 13:59.23 | mknerd | fcois : /etc/config/rc.locl |
| 13:59.26 | mknerd | er rc.local |
| 13:59.37 | fcois | oh thank you :-) |
| 13:59.46 | rob0 | You can always make symlinks to bigger files/directories outside the rootfs, as long as those are not needed for booting. |
| 13:59.53 | puzzled | hi |
| 13:59.55 | mknerd | yeah, I use it to unmount and remount the CF somewhere else |
| 13:59.59 | fcois | and other question, how can you add tftpd command, have an idea? |
| 14:00.14 | ManxPower | rob0: You are encouraging them |
| 14:00.16 | mknerd | what for/ |
| 14:00.38 | mknerd | manxpower, are you always half-empty/ |
| 14:00.39 | mknerd | ? |
| 14:00.43 | fcois | for /var/lib/asterisk/static-http/config |
| 14:00.51 | ManxPower | mknerd: only when people insist on trying to be off topic. |
| 14:00.52 | mknerd | fcois: polycoms? |
| 14:01.05 | mknerd | its the asterisk appliance... in the asterisk channel |
| 14:01.15 | mknerd | geez your tight |
| 14:01.16 | fcois | oh no I need tftpd to do autoprovisioning for others phones |
| 14:01.21 | fcois | like cisco thomson... |
| 14:01.33 | mknerd | yeah, don't know about that, I use ftp for my phones |
| 14:01.43 | ManxPower | mknerd: Correct, the Asterisk channel, not the Asterisk Applicance channel, not the Asterisk GUI channel, no the Trixbox channel, not the AMP channel. This is the Asterisk channel. |
| 14:02.02 | mknerd | im not interrupting you am I? |
| 14:02.25 | mknerd | there is not an aa50 channel, so this will have to do |
| 14:02.39 | mknerd | but there should be |
| 14:02.41 | fcois | if you use ftp, it is to upload it to a tftp server ... ? |
| 14:02.48 | *** join/#asterisk mindCrime (n=chatzill@216.27.62.2) |
| 14:02.52 | mknerd | no tftp at all |
| 14:03.01 | mknerd | the polycoms that I have support ftp |
| 14:03.08 | fcois | ok and to provisioning ? |
| 14:03.16 | fcois | *for |
| 14:03.27 | mknerd | they provision over regular ftp, not tftp |
| 14:03.31 | *** join/#asterisk mltlnx (n=mltlnx@68.236.180.175) |
| 14:03.39 | fcois | ok lucky! |
| 14:03.44 | fcois | my phones use tftp ! |
| 14:03.46 | [TK]D-Fender | mknerd: What models? |
| 14:04.49 | fcois | mknerd, I could do scripts for thomson ST2030 and ciscos 7906 7940 7960 7970... PAP2 |
| 14:05.25 | Uatec | my phones use TFTP and will connect to my windows TFTP server but not my linux one on my asterisk box |
| 14:05.27 | fcois | when I add a user I can choose the phone model and that execute the script... |
| 14:05.36 | mknerd | I have the 501's |
| 14:06.00 | mknerd | they are pretty nice, will do tftp and ftp |
| 14:06.07 | mknerd | ftp is nicer and more sucure |
| 14:06.49 | fcois | but if I need tftpd I have to use other server :( |
| 14:07.12 | fcois | because I could see digium they said 'ok we will help you to dev'... |
| 14:07.29 | fcois | and no responses since wenesday |
| 14:08.33 | mknerd | yeah, I just spoke with them, they do not support and AGI scripting on the AA50, but would transfer me to their Dev department to discuss building me a custom solution |
| 14:09.01 | [TK]D-Fender | mknerd: So they support ftp/tftp/http. What is your AA lacking for you to host their provisioning files? |
| 14:09.21 | mknerd | AA is lacking perl/c/php/bash |
| 14:09.23 | mknerd | pick one |
| 14:09.32 | ManxPower | Good for them. |
| 14:09.33 | mknerd | I want to interface with a SQL database on a second box |
| 14:09.35 | [TK]D-Fender | mknerd: Ok, and for provisioning? |
| 14:09.46 | mknerd | fcois wants tftpd for provisioning |
| 14:09.47 | ManxPower | mknerd: you should never have purchased the AA |
| 14:09.57 | mknerd | manxpower, please, just don't talk to me |
| 14:10.01 | tzafrir_laptop | mknerd: is does have a /bin/sh |
| 14:10.08 | ManxPower | mknerd: put me on /ignore |
| 14:10.16 | mknerd | it does, I could script it via netcat |
| 14:10.22 | fcois | and digium want that I do it without help from them!!! |
| 14:10.22 | [TK]D-Fender | mknerd: Well... the AA50 is a toaster. Always was, always will be. You are now expecting Ferrari features out of a Lada. |
| 14:10.23 | mknerd | but thats kinda lame dont you think |
| 14:10.37 | *** join/#asterisk Koshatul (n=evangeli@ppp188-33.static.internode.on.net) |
| 14:10.45 | mknerd | has anyone seen the Pika Warp appliance? |
| 14:10.49 | mknerd | it has perl |
| 14:11.00 | mknerd | 256mb of internal flash memory |
| 14:11.07 | mknerd | and its 1/3 the cost |
| 14:11.12 | rob0 | I think it's rather lame to buy something which does what it claims to do, and then gripe because it doesn't do everything under the sun. |
| 14:11.15 | mknerd | I ordered one last week |
| 14:11.29 | [TK]D-Fender | mknerd: mknerd http://www.soekris.com/net5501.htm <- Here, go build your own. |
| 14:11.37 | fcois | I need that david from digium call me back! |
| 14:12.10 | ManxPower | [TK]D-Fender: I wonder if Digium is starting to regret the AA. Seems like the user support needs would eat any potential profit. |
| 14:12.19 | mknerd | rob0, actually, the provider told me that it could do perl, but then told me after I got it that I would have to pay them to roll it on there |
| 14:12.41 | [TK]D-Fender | ManxPower: Could be, I' haven't really keps up, but I've heard little overal praise for it, only troubles here for that its lacking in. |
| 14:12.48 | [TK]D-Fender | keep* |
| 14:13.24 | rob0 | ManxPower, I doubt it; they just tell buyers who had unrealistic expectations what it would cost to have those expectations met. |
| 14:13.35 | mknerd | I don't think that scripting is asking too much, its well within the potential of the machine |
| 14:14.33 | ManxPower | [TK]D-Fender: classic appliance and gui problems |
| 14:14.42 | rob0 | Still probably well under the cost of many other PBX appliances, I bet. |
| 14:14.58 | ManxPower | rob0: these people don't want an appliance. |
| 14:15.04 | *** join/#asterisk fskrotzki (n=fskrotzk@host.textwise.com) |
| 14:15.14 | fcois | mknerd: are you shure about the /etc/config/rc.local? |
| 14:15.15 | [TK]D-Fender | ManxPower: Mostly CF / OS issues (aside from the nasty heat issues) |
| 14:15.27 | lmadsen | everyone wants a gui that does everything. I've never heard of a GUI that people didn't complain about (but I don't use GUIs, so I don't even know what it's lacking) |
| 14:15.28 | mknerd | rob0, actually the Pika is 800$ fully loaded with 5 fxs and 4 fxo, the aa50 is 1800 |
| 14:15.35 | fcois | mknerd: have you created it or edit it because I dont have it |
| 14:15.36 | *** part/#asterisk lmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 14:15.49 | mknerd | fcois, you have to create it |
| 14:15.51 | vader-- | Have any of you guys bought postini for a small organization from their website? I am wondering if there is a setup fee if you go through the website? |
| 14:15.55 | mknerd | i dont want a gui |
| 14:16.20 | mknerd | lmadsen, who said anything about wanting a gui? |
| 14:16.51 | fcois | ok |
| 14:17.04 | fcois | is it possible to have a copy of your? |
| 14:17.06 | mknerd | fcois: make sure you save_config after you edit it |
| 14:17.18 | jaytee | I want a gui that does my laundry and pours me a beer and I want it installed by some guy who comes to the house and installs it for me and while he's here he mows my lawn and does all that for free! |
| 14:17.20 | fcois | ok I can see how to do after |
| 14:17.25 | mknerd | fcois: do you know normal shell scripting .. that is all that is |
| 14:17.43 | fcois | ok I will try thank you |
| 14:17.58 | *** join/#asterisk shawdog22 (n=shaw@pacman.oaklandcorp.com) |
| 14:18.29 | *** join/#asterisk ChkDigit (n=mrw@static24-89-65-166.regina.accesscomm.ca) |
| 14:19.04 | shawdog22 | Wondering if I can get some pointers on trapping some odd behavior of my asterisk system? |
| 14:19.36 | *** join/#asterisk acxty (n=acxty@201.220.132.138) |
| 14:19.48 | rob0 | jaytee: s/guy/lovely young maiden/ s/ he/ she/ |
| 14:20.24 | jaytee | rob0, hmmm yeah. I wasn't thinkin that far ahead. |
| 14:20.46 | [TK]D-Fender | shawdog22: Perhaps you could actually describe your scenario... |
| 14:21.06 | rob0 | Glad to be of assistance. |
| 14:21.30 | shawdog22 | Sorry, didn't know what the actual rules of the room where. |
| 14:21.34 | *** join/#asterisk tobias (n=tobias@user-0c998nt.cable.mindspring.com) |
| 14:22.36 | shawdog22 | I've got queues set up with SIP agents, and I've noticed that some of the agents go into a state of "(paused)(Not in use)". And don't come directly out of it. |
| 14:23.35 | [TK]D-Fender | shawdog22: pastebin your queues setup (all related config) |
| 14:23.38 | [TK]D-Fender | ~pb |
| 14:23.39 | jbot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste |
| 14:23.40 | [TK]D-Fender | ^^^^^ |
| 14:24.11 | shawdog22 | I actually think I may have found the problem. |
| 14:24.42 | fcois | mknerd : thank you for the rc.local |
| 14:25.23 | shawdog22 | I'm using OpenFire (XMPP based instant messenger) with an Asterisk plugin. Looks like when the IM user goes into an away mode the SIP channel goes into a Paused State. |
| 14:25.36 | mknerd | fcois: yep, no problem, someone else helped me with that |
| 14:25.56 | *** join/#asterisk zxd (n=XoX@213.31.43.2) |
| 14:26.01 | zxd | say |
| 14:26.01 | fcois | mknerd : who are these someone? |
| 14:26.50 | zxd | when registering to Asterisk , is the data stream RTP also routed via asterisk , or is directly p2p with the remote end? |
| 14:26.50 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 14:26.50 | fcois | mknerd: because I just need that (tftpd) :-) |
| 14:28.40 | mknerd | fcois: tftpd is not even a busybox command, so I imagine that it would not be a simple port |
| 14:28.42 | shawdog22 | Is there any way to capture the events that are getting sent via the Manager Interface? |
| 14:28.46 | *** join/#asterisk jtknapp (n=skip@65-126-63-1.dia.static.qwest.net) |
| 14:29.13 | fcois | mknerd: in the website of busybox, I could see that there is tftpd! |
| 14:29.28 | *** join/#asterisk russellb (n=russell@asterisk/developer-and-stable-maintainer/drumkilla) |
| 14:29.28 | *** mode/#asterisk [+o russellb] by ChanServ |
| 14:29.43 | mknerd | fcois, show me a link, all I saw was tftp |
| 14:30.06 | fcois | I have a look.. |
| 14:30.21 | *** join/#asterisk rupa (i=rupa@gw.rupa.com) |
| 14:32.40 | fcois | mknerd, I was shure to see it but I think it is not possible if we dont do somethings in the sources before compiling... |
| 14:33.10 | *** join/#asterisk mltlnx (n=mltlnx@pool-96-232-207-89.nycmny.east.verizon.net) |
| 14:33.54 | *** part/#asterisk shawdog22 (n=shaw@pacman.oaklandcorp.com) |
| 14:34.33 | fcois | mknerd : http://busybox.net/screenshot.html |
| 14:35.10 | *** join/#asterisk acxty (n=acxty@201.220.132.138) |
| 14:38.06 | *** join/#asterisk leanshen (n=bob@ool-43546a71.dyn.optonline.net) |
| 14:38.45 | mknerd | fcois: thats strange, its not in the command list |
| 14:38.47 | *** join/#asterisk gr0mit (n=tim@85.58.187.81.in-addr.arpa) |
| 14:39.08 | fcois | yes its what i could see |
| 14:39.20 | mknerd | your going beyond my technical skills to walk you through a busybox recompile on the aa50 |
| 14:39.49 | b11d` | i cant believe i just learned of busybox last night |
| 14:39.52 | fcois | I never compil a busybox ^^ |
| 14:40.30 | mknerd | b11d its pretty sweet eh |
| 14:40.38 | b11d` | yeah it has its uses, thats for sure |
| 14:41.40 | fcois | yes think too |
| 14:43.07 | *** join/#asterisk flush (n=SYN_SENT@ip216-239-83-43.vif.net) |
| 14:43.18 | oktay | jblack: sorry about the late response. i had stepped out. |
| 14:43.41 | *** part/#asterisk MikeJ (n=MikeJ@33.203.64.208.static.accentrainc.com) |
| 14:43.43 | oktay | i created a generic SIP extension.. but still no port 5060 on the server |
| 14:43.46 | *** join/#asterisk Defraz (n=T0tal@69.92.19.83) |
| 14:43.59 | flush | hey white wire is ground and black is phase or its the inverse ? |
| 14:44.41 | *** join/#asterisk lesouvage (n=lesouvag@cc341200-a.assen1.dr.home.nl) |
| 14:45.48 | *** join/#asterisk KickServer (n=picachu@host-static-89-41-72-225.moldtelecom.md) |
| 14:46.03 | KickServer | hey. Is it possible to call macro from AGI ? |
| 14:46.25 | ManxPower | oktay: Are you going to step out again, or are you going to stick around for help this time? |
| 14:48.01 | ManxPower | Apparently not. I'll go back to paying work then. |
| 14:48.10 | *** join/#asterisk jpeeler (n=jpeeler@216.207.245.1) |
| 14:48.34 | oktay | easy man |
| 14:48.37 | *** join/#asterisk af_ (n=getsmart@88-149-241-145.dynamic.ngi.it) |
| 14:48.56 | oktay | i had to attend a meeting before. |
| 14:49.07 | fcois | mknerd : I can imagine that a tftpd command without compil juste .sh for example ... |
| 14:50.22 | ManxPower | oktay: That does not answer my question. Nobody here is going to try to help if you keep leaving. |
| 14:50.43 | ManxPower | KickServer: Maybe, but don't be surprised if the dialplan never returns to your script. |
| 14:50.53 | *** join/#asterisk hfb (n=hfb@pool-71-118-254-245.lsanca.dsl-w.verizon.net) |
| 14:51.21 | *** join/#asterisk hi365 (n=hi365@bzq-219-141-66.static.bezeqint.net) |
| 14:51.22 | ManxPower | [TK]D-Fender: It's going to be one of those days, isn't it? |
| 14:51.42 | *** join/#asterisk b11d` (n=no@234-200-29-134.hcc.mnscu.edu) |
| 14:52.22 | [TK]D-Fender | ManxPower: At least you see it coming.. |
| 14:53.04 | ManxPower | [TK]D-Fender: nod. |
| 14:53.18 | mcfloppy | http://pastebin.com/m1bb448c5 what must i do, to call over isdn? |
| 14:53.27 | b11d` | hey TK.. do you know of anything similar to the SPA-8000 but is FXO instead of FXS? |
| 14:53.35 | b11d` | I see linksys makes a 2 port FXO.. but.. meh |
| 14:53.44 | oktay | ManxPower: :) |
| 14:53.56 | ManxPower | mcfloppy: you mean mISDN. |
| 14:54.16 | mcfloppy | ManxPower yes i mean misdn |
| 14:54.36 | ManxPower | mcfloppy: Did you build Asterisk after you installed mISDN -dev packages? |
| 14:54.37 | *** join/#asterisk freezey (i=hidden-u@gw.mypublisher.com) |
| 14:55.40 | mcfloppy | ManxPower yes... the other way works... call from the ISDNphone *29 -> SIPphone rings |
| 14:56.19 | [TK]D-Fender | b11d`: They Do? What model? |
| 14:56.29 | b11d` | brb.. let me dig it up |
| 14:56.57 | ManxPower | mcfloppy: It must be a formatting problem of your Dial line. I don't know the format for mISDN. Hopefully someone here will be familiar with it. |
| 14:57.03 | b11d` | err its the 4 port SPA400 |
| 14:57.14 | mcfloppy | ManxPower okay.. |
| 14:57.55 | b11d` | its the SPA-2100 im thinking of.. |
| 14:57.56 | [TK]D-Fender | b11d`: it has "issues". |
| 14:57.58 | b11d` | 2 port fxs.. |
| 14:58.00 | b11d` | err fdo |
| 14:58.01 | b11d` | DOH |
| 14:58.01 | b11d` | fxo |
| 14:58.06 | b11d` | oh really? figures.. |
| 14:58.13 | b11d` | i need somethign like the spa-8000 with 8 fxo ports.. |
| 14:58.59 | ManxPower | b11d`: You mean CHEAP spa-8000 with 8 fxo ports |
| 14:59.14 | ManxPower | I doubt you will find anything like that. What's wrong with a T-1 card and channel bank? |
| 14:59.22 | b11d` | i got my spa-8000 for $215.. didnt think it was that expensive |
| 14:59.32 | ManxPower | b11d`: FXO ports are expensive |
| 14:59.36 | b11d` | yeah im realizing that |
| 14:59.52 | b11d` | well i got one business here with five lines.. i dont need a t1 and a CB for them |
| 14:59.53 | b11d` | thats for sure |
| 14:59.58 | nny_1 | anyone interested in helping update this caller-ID reverse lookup AGI let me know.. removing some old url juju and fixed some variables.. still debugging why it returns "0" to asterisk |
| 15:00.04 | ManxPower | b11d`: that's not for sure. |
| 15:00.12 | b11d` | ?? explain! |
| 15:00.12 | b11d` | :) |
| 15:00.28 | b11d` | maybe a fractional T1 or something would do the trick |
| 15:00.29 | ManxPower | You can have a T-1 card + channel bank for as little as one analog line. |
| 15:00.36 | b11d` | what?? |
| 15:00.39 | b11d` | how do you figure? |
| 15:00.39 | mcfloppy | hmmmm |
| 15:00.45 | ManxPower | if you went with a tact T-1 then you could of course get rid of the channel bank, |
| 15:01.13 | mcfloppy | i tried this: Dial(mISDN/1/*16,10,Ttr);, now it rings in the speaker from the sipphone, but the isdnphone dont ring |
| 15:01.16 | *** join/#asterisk marlow (n=marlow@loke.sca.airwire.ie) |
| 15:01.22 | ManxPower | b11d`: Do you even understand what a channel bank is? |
| 15:01.29 | b11d` | yes i do.. i have a bunch of them |
| 15:01.34 | *** join/#asterisk fcois (n=fcois@bagnolet.acropolistelecom.net) |
| 15:01.51 | ManxPower | I don't see why you are confused then. |
| 15:02.02 | b11d` | i am talking about five lines OUT to the telco.. they have 12 internal extensions.. |
| 15:02.13 | b11d` | why would I get a T1 for those five outbound lines?> |
| 15:02.18 | ManxPower | b11d`: I am also talking about line IN/OUT to/from telco. |
| 15:02.21 | b11d` | fuck |
| 15:02.22 | b11d` | :) |
| 15:02.27 | b11d` | i must be retarted or somethign again.. |
| 15:02.30 | ManxPower | b11d`: I'm NOT TELLING YOU TO GET A T-!!!!! |
| 15:02.35 | b11d` | oh! |
| 15:02.35 | b11d` | :) |
| 15:02.39 | [TK]D-Fender | b11d`: SPA-400 treats all lines dentically and gives you little control. |
| 15:02.43 | ManxPower | telco analog line -> channel bank -> t-1 card in Asterisk |
| 15:02.52 | b11d` | yeah that makes sense to me! |
| 15:03.21 | ManxPower | obviously a better setup would be telco T-1 -> T-1 card in Asterisk, but if you don't want that then you could use a channel bank with FXO ports. |
| 15:03.41 | [TK]D-Fender | b11d`: http://www.telephonydepot.com/product_p/105-066-118-fxo.htm |
| 15:03.46 | *** join/#asterisk dkwiebe (n=darren@h66-112-187-16.mcsnet.ca) |
| 15:03.55 | *** join/#asterisk SirThomas (n=tomc@mail.kendeco.com) |
| 15:03.55 | [TK]D-Fender | b11d`: decent value |
| 15:04.12 | b11d` | 8 fxo too.. |
| 15:05.14 | b11d` | ManxPower.. thanks for explaining that to me! It is appreciated.. you are correct.. I wish I could just get them a T1 though.. I'd wager a CB and a T1 card would be cheaper than two 4 port FXO digium cards.. |
| 15:05.53 | *** join/#asterisk mort_gib (n=mjensen@adsl-2-234.gibnet.gi) |
| 15:06.02 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 15:07.08 | [TK]D-Fender | b11d`: Nope. PCI = cheaper. |
| 15:07.09 | ManxPower | b11d`: Channel banks are expensive. You could purchase two from ebay for less than a single new channel bank |
| 15:07.40 | [TK]D-Fender | b11d`: But this device is probably the best direct value for the ports it provides. |
| 15:12.01 | mcfloppy | how can i reload the extensions.acl? |
| 15:12.14 | mcfloppy | ael |
| 15:14.43 | *** join/#asterisk clive- (n=pirch@41.242.156.73) |
| 15:15.11 | clive- | Hi guys. Is it ok to use 1.4.19.1 or should I rather svn down 1.4.20-rc2 ? |
| 15:15.19 | oktay | good night boys and girls.. |
| 15:15.43 | b11d` | clive-.. where do you intend to run this? |
| 15:15.58 | b11d` | cause if its just for mucking around.. i'd say rc2.. but if not.. then 19.1 |
| 15:16.06 | clive- | on a production box |
| 15:16.21 | b11d` | yeah then dont mess with rc2 unless you desperately need a bugfix thats in there or something |
| 15:16.49 | *** join/#asterisk RoyK (n=roy@062249179121.customer.alfanett.no) |
| 15:16.50 | clive- | I was really wondering if there any important bugfixes in the 20-rc2 version |
| 15:16.59 | b11d` | check out the changelog |
| 15:17.13 | clive- | goes surfing the changelog |
| 15:17.22 | b11d` | I know theres some neat IAX improvements in 4.20 |
| 15:18.23 | Ritzerisk | hey hey does anyone know a good dialer for asterisk |
| 15:21.20 | waKKu | idefisk/zoiper ? |
| 15:21.31 | Ritzerisk | ? |
| 15:21.42 | *** part/#asterisk lsodi (n=root@213.168.26.50) |
| 15:21.53 | clive- | ok, I see some performance changes that I need... seems like some bad code crept into 1.4.19 that was only fixed in 1.4.20 |
| 15:21.57 | *** join/#asterisk esaym (n=user@cpe-70-120-89-6.satx.res.rr.com) |
| 15:23.37 | *** join/#asterisk gr0mit (n=tim@82.58.187.81.in-addr.arpa) |
| 15:23.47 | b11d` | well i'd still wait for 4.20 to finish its RC cycle before using it in production |
| 15:23.52 | b11d` | use at your own risk :) |
| 15:24.00 | *** join/#asterisk klictel (n=klictel@atelka.info) |
| 15:25.23 | *** join/#asterisk beek (n=klinebl@65.211.106.242) |
| 15:25.55 | *** join/#asterisk drummond_ (n=rsd@h-67-103-23-130.phlapafg.covad.net) |
| 15:26.46 | clive- | b11d thanks for your help |
| 15:28.08 | b11d` | any time |
| 15:28.14 | drummond_ | i see they are releasing new versions all the time, 1.4.19, 1.4.19.1, etc |
| 15:28.30 | drummond_ | is there a rule of thumb on how often to migrate to them? |
| 15:29.40 | b11d` | migrate when necessary.. |
| 15:29.42 | zxd | when registering to Asterisk , is the data stream RTP also routed via asterisk , or is directly p2p with the remote end? |
| 15:30.19 | rob0 | depends on canreinvite and related settings for the channel, IIUC. |
| 15:30.23 | file | zxd: do you mean "when a call is placed through Asterisk between two SIP endpoints" |
| 15:30.37 | zxd | yes |
| 15:30.48 | nny_1 | heh this callerid_shell.agi 's curl and grep juju is outdated as far as what data is spit back by google, anywho etc |
| 15:31.01 | zxd | all sip calls are relayed through asterisk no? |
| 15:31.21 | rob0 | unless they're not, sure. :) |
| 15:31.22 | nny_1 | still hacking through it, will post the "update" (as in I beat the crap out of it and that came out) to voip-info |
| 15:31.32 | file | points to what rob0 said about canreinvite |
| 15:31.47 | zxd | i read that asterisk only relays sdp information between the two SIP clients , then they can establish direct RTP channel between them ? |
| 15:31.58 | zxd | ok |
| 15:32.18 | zxd | reads what canreinvite means |
| 15:32.20 | zxd | what is IIUC |
| 15:32.53 | [TK]D-Fender | zxd: If I Understand Correctly |
| 15:32.58 | clive- | zxd all calls are routed through asterisk unless you have canreinvite=yes but you will lose CDR information |
| 15:33.24 | jaytee | My telco is passing me 7 digits on incoming calls, I want to try and match the last 4 digits of the extension with any of my local extensions and if it doesn't match then pass the call to another trunk that connects to another PBX. |
| 15:33.28 | zxd | clive-, including RTP traffic ? |
| 15:33.37 | jaytee | is GotoIf the best way to do that? |
| 15:33.54 | clive- | zxd yes |
| 15:34.00 | *** join/#asterisk ScarEye (n=scareye@12.27.87.66) |
| 15:34.31 | *** join/#asterisk hmmhesays (n=hmmhesay@24-119-176-74.cpe.cableone.net) |
| 15:34.50 | [TK]D-Fender | jaytee: ChanisAvail (Local) followed by GotoIf. |
| 15:35.12 | jaytee | [TK]D-Fender, thanks |
| 15:36.43 | nny_1 | most of these sources (anywho, google, 411.com) seem unreliable, as the site API could change tomorrow and the lookup method shats from that point out |
| 15:36.54 | nny_1 | hence the issue why it is outdated now |
| 15:37.11 | nny_1 | (+/- some bad asterisk variable names from 1.2) |
| 15:37.44 | nny_1 | is there a "better" way or a service that is designed to return values to a system like asterisk? (Seems "gotname.com" is dead) |
| 15:39.42 | Ritzerisk | or does anyone know of a type of asterisk system that can use the auto dialer |
| 15:39.48 | mintee | I'm assuming that an #include is only read once during startup or a reload, correct? |
| 15:40.12 | mintee | Just wondering if I could do like an #include extensions_${EXTEN}.conf or something |
| 15:40.36 | ManxPower | mintee: it is read only on startup or reload |
| 15:40.51 | mintee | k, just as i thought, thanks ManxPower |
| 15:40.54 | ManxPower | You cant' do what you want, as EXTEN is a channel variable |
| 15:41.05 | *** join/#asterisk mackes-Office (n=root@74.10.229.35) |
| 15:41.10 | b11d` | can anyone recommend a good IAX provider in the USA? |
| 15:41.25 | dkwiebe | Ritzerisk: We built a program to place calls based on a list in the database that does basically what you want. www.astpp.org and look for "autocall" |
| 15:41.36 | mackes-Office | Vitelity does a great job |
| 15:41.39 | ManxPower | ~itsp |
| 15:41.40 | jbot | [~itsp] An ITSP is an Internet telephony Service Provider or "VoIP Telephone company". The allow you to either PLACE calls to the PSTN (called Termination), RECEIVE calls from the PSTN (called Origination), or both. Some offer fixed rates, others $/min. Enter ~itsplist-us (USA) or ~itsplist-ca (Canada) for a listing of popular ITSPs |
| 15:41.48 | nny_1 | whats the easiest way to strip the IP address of the server from the callerid? My polycom shows NUMBER@192.168.100.10 each time |
| 15:42.10 | mackes-Office | Set your caller id in your extension config |
| 15:42.22 | b11d` | sweet |
| 15:42.23 | b11d` | thanks |
| 15:42.30 | mackes-Office | your welcome |
| 15:42.59 | *** join/#asterisk russellb (n=russellb@asterisk/developer-and-stable-maintainer/drumkilla) |
| 15:42.59 | *** mode/#asterisk [+o russellb] by ChanServ |
| 15:44.01 | nny_1 | mackes-Office: yeah figured, do I need to strip that IP off or does Set(CALLERID(num)=${EXTEN}) work? |
| 15:45.02 | *** join/#asterisk mknerd (i=3f951603@gateway/web/ajax/mibbit.com/x-114d61bca7d5c4b1) |
| 15:45.22 | rob0 | Something @ the wiki implied that it wouldn't be good to invoke a macro within a macro. I have one macro which sets callerID, and another which places the call (to US tollfree, so I have separate patterns for 1800, 1866, 1877 and 1888. |
| 15:45.24 | mort_gib | How do I find out what's going on here: chan_sip.c: Maximum retries exceeded on transmission 3c34cf3abbcc-4f9568kw88gb |
| 15:45.31 | mackes-Office | The IP is being sent because one of two things- 1. It doesnt sound as if your Polycom phone is regestered with your Asterisk box -it is just making calls to it, and your Asterisk Box allows for Anonymous access of outside devices |
| 15:45.34 | *** join/#asterisk jpsharp (i=269@cruncher.psychoses.org) |
| 15:46.04 | nny_1 | mackes-Office: phone is registered and no by install doesn't |
| 15:46.10 | mackes-Office | If it was regestered, your SIP.CONF entry for the phone would set your caller ID for it. |
| 15:46.20 | nny_1 | wha? |
| 15:46.22 | *** join/#asterisk gr0mit (n=tim@CSC.ge4-0-0.401.ar1.BBS1.gblx.net) |
| 15:46.23 | nny_1 | that's crazy |
| 15:46.23 | mackes-Office | so you can set it in your SIP.conf for it |
| 15:46.26 | rob0 | The question then being, should I call the callerID macro from the dialing one, or use separate priorities in each pattern? |
| 15:46.39 | nny_1 | how would sip.conf set the callerid for the incoming call from an outside source? |
| 15:46.40 | jpsharp | I need a NAT aware TFTP server for Linux. There was one written in Java, but I can't find it again. |
| 15:47.07 | mackes-Office | Is your phone an outside source, or an extension on the system? |
| 15:47.30 | nny_1 | extension in systemk |
| 15:47.41 | mintee | in extensions, must a context be defined before a call for it? |
| 15:47.41 | rob0 | NAT aware tftpd ... hmmm, /me is trying to digest that concept |
| 15:48.02 | mackes-Office | registered extensions on the system have their caller ID set in the SIP.conf |
| 15:48.06 | mintee | Hum, that's kinda confusing... I mean are extensions like firewall rules and followed by the order they are defined? |
| 15:48.08 | jpsharp | It exists. I've used it, but I can't find the program again. |
| 15:48.10 | nny_1 | isn't tftp udp based? |
| 15:48.29 | jpsharp | yes |
| 15:48.29 | mackes-Office | Hmmm... |
| 15:48.43 | nny_1 | er wait misunderstood the q |
| 15:49.05 | nny_1 | mackes-Office: outside source, incoming call from outside source |
| 15:49.24 | *** join/#asterisk Skarmeth (n=Skarmeth@iris.aspec.com.br) |
| 15:49.25 | mackes-Office | When your phone regesters, asterisk takes the information in the SIP.conf for that phone, and uses it for all calls placed on that device. |
| 15:49.35 | Nasra | stupid question: |
| 15:49.59 | nny_1 | mackes-Office: i get the number but it looks like |
| 15:49.59 | nny_1 | sip:8433425901@192.168.100.10 HILTONHEAD,SC |
| 15:49.59 | nny_1 | <PROTECTED> |
| 15:50.11 | Nasra | I installed Asterisk in my system...now how do I unpack it or where do I go ? |
| 15:50.14 | mintee | Narsa, Mine? |
| 15:50.18 | nny_1 | the latter is a nanpa.txt lookup I am working on from an old agiscript |
| 15:50.20 | mintee | oh |
| 15:50.35 | nny_1 | mackes-Office: the sip:blah@foo has always been there |
| 15:50.54 | mackes-Office | you can define caller ID changes in the extension.conf, however you should not have to do that to set the caller ID of a phone placing a call that is regestered on the system |
| 15:51.08 | mackes-Office | ok, for example |
| 15:51.09 | nny_1 | looks up |
| 15:51.16 | mackes-Office | I have a polycom phone on my desk |
| 15:51.17 | nny_1 | mackes-Office: from outside source |
| 15:51.20 | nny_1 | not from another phone |
| 15:51.49 | mackes-Office | registered with Asterisk |
| 15:52.04 | mackes-Office | Here is what I see when I do a lookup from the CLI |
| 15:52.21 | ManxPower | I ALWAYS set the callerid info for all phones in sip.conf |
| 15:52.55 | mackes-Office | <PROTECTED> |
| 15:52.55 | mackes-Office | <PROTECTED> |
| 15:52.55 | mackes-Office | <PROTECTED> |
| 15:52.55 | mackes-Office | <PROTECTED> |
| 15:52.55 | mackes-Office | <PROTECTED> |
| 15:52.56 | mackes-Office | <PROTECTED> |
| 15:52.58 | mackes-Office | <PROTECTED> |
| 15:53.00 | mackes-Office | <PROTECTED> |
| 15:53.02 | mackes-Office | <PROTECTED> |
| 15:53.04 | mackes-Office | <PROTECTED> |
| 15:53.06 | Kobaz | /k |
| 15:53.06 | mackes-Office | <PROTECTED> |
| 15:53.06 | ManxPower | mackes-Office: USE PASTEBIN!!!!!!!!!!!!!!!!!!!!!!!!!!!! |
| 15:53.08 | mackes-Office | <PROTECTED> |
| 15:53.08 | rob0 | flood-- |
| 15:53.10 | mackes-Office | <PROTECTED> |
| 15:53.12 | mackes-Office | <PROTECTED> |
| 15:53.14 | mackes-Office | <PROTECTED> |
| 15:53.16 | mackes-Office | <PROTECTED> |
| 15:53.17 | nny_1 | lol |
| 15:53.18 | mackes-Office | <PROTECTED> |
| 15:53.20 | mackes-Office | <PROTECTED> |
| 15:53.20 | Uatec | lol |
| 15:53.21 | nny_1 | Fail: 1 |
| 15:53.21 | Uatec | paste bin |
| 15:53.22 | mackes-Office | <PROTECTED> |
| 15:53.26 | mackes-Office | <PROTECTED> |
| 15:53.26 | ManxPower | *** mackes-Office has been added to /IGNORE list. |
| 15:53.26 | Kobaz | beats mackes-Office with a large trout |
| 15:53.28 | mackes-Office | <PROTECTED> |
| 15:53.28 | Uatec | someone's on ignore now |
| 15:53.28 | matnel | pastebin |
| 15:53.30 | mackes-Office | <PROTECTED> |
| 15:53.32 | mackes-Office | <PROTECTED> |
| 15:53.34 | mackes-Office | <PROTECTED> |
| 15:53.36 | mackes-Office | <PROTECTED> |
| 15:53.38 | mackes-Office | <PROTECTED> |
| 15:53.38 | rob0 | You blew it bud. |
| 15:53.40 | mackes-Office | <PROTECTED> |
| 15:53.41 | ManxPower | someone kick him |
| 15:53.42 | mackes-Office | <PROTECTED> |
| 15:53.44 | mackes-Office | <PROTECTED> |
| 15:53.46 | Uatec | is he still going? |
| 15:53.46 | mackes-Office | <PROTECTED> |
| 15:53.48 | mackes-Office | <PROTECTED> |
| 15:53.50 | mackes-Office | <PROTECTED> |
| 15:53.52 | mackes-Office | <PROTECTED> |
| 15:53.54 | Kobaz | mackes-Office: /quit |
| 15:53.54 | rob0 | yes!! |
| 15:53.56 | mackes-Office | <PROTECTED> |
| 15:53.58 | mackes-Office | <PROTECTED> |
| 15:54.00 | mackes-Office | <PROTECTED> |
| 15:54.02 | mackes-Office | <PROTECTED> |
| 15:54.04 | mackes-Office | what? |
| 15:54.04 | nny_1 | End: NEVER! |
| 15:54.06 | mackes-Office | My god... did that really cause you all some much harm? |
| 15:54.11 | ManxPower | mackes-Office: yes |
| 15:54.16 | Kobaz | mackes-Office: it's pointless and annoying |
| 15:54.26 | rob0 | This is a busy channel. |
| 15:54.28 | jpsharp | Ahhah. Found it again: http://tanesha.net/projects/tftpd/tftpd-server-2.0.0.zip |
| 15:54.37 | ManxPower | rob0: well at least it was 8-) |
| 15:54.50 | mackes-Office | How do you have a conversation without passing information back and forth? |
| 15:54.51 | nny_1 | ManxPower: so Set(CALLERID(num)=${EXTEN} ? or should I mangle it to remove the channel and IP? |
| 15:54.53 | Kobaz | mackes-Office: feel free to paste stuff, but keep it 3 lines or less |
| 15:54.55 | ManxPower | ~pb |
| 15:54.56 | jbot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste |
| 15:55.05 | ManxPower | mackes-Office: the same way as everyone else on the channel |
| 15:55.18 | nny_1 | well ${EXTEN}) * |
| 15:55.21 | mackes-Office | ok. Sorry Folks. My Mistake |
| 15:55.44 | ManxPower | nny_1: no, that will set the callerid number to be the current value of EXTEN (normall the dialed digits) |
| 15:56.12 | Uatec | Why is it that when asterisk registers with my SIP proxy it tries, gets 401 then tries again with credentials and gets through... |
| 15:56.17 | draygon | anyone here do any IVR recording? |
| 15:56.20 | Uatec | but when it tries to invite, it tries, gets 301, then gives up |
| 15:56.24 | nny_1 | ManxPower: kk I can research what goes on the other side of the = ty |
| 15:56.25 | rob0 | I had a question about 4-5 pages up in the scroll :( |
| 15:56.27 | Uatec | 401, sorry |
| 15:56.32 | Uatec | does anybody know anything about that? |
| 15:56.49 | ManxPower | Uatec: all SIP devices will try without auth first. |
| 15:57.06 | Uatec | ManxPower, that's fine |
| 15:57.16 | Uatec | but asterisk isn't trying WITH auth, second |
| 15:57.22 | Uatec | the conversation goes |
| 15:57.41 | Uatec | INVITE -> WWW_Challenge -> OK -> BYE |
| 15:57.43 | Uatec | WTF? |
| 15:58.59 | b11d` | if you had to install some analog phones, and you wanted very high quality audio, like something you would get using the Polycoms with HD voice.. what phone would you use? |
| 15:59.20 | *** join/#asterisk DaneM (n=dane@adsl-76-236-27-148.dsl.chi2ca.sbcglobal.net) |
| 15:59.34 | ManxPower | b11d`: nothing. analog phones are not HD |
| 16:00.04 | b11d` | damnit thats what I was hoping to not hear :) |
| 16:00.08 | b11d` | thanks though |
| 16:01.07 | ManxPower | b11d`: you really need to learn telecom |
| 16:01.33 | b11d` | i am doing my best... not going as fast as i'd like, but am learning.. |
| 16:01.44 | b11d` | im not afraid to ask a dumb question i guess.. |
| 16:01.44 | hmmhesays | b11d`, you won't get that out of an analog phone |
| 16:01.56 | hmmhesays | they won't support 16khz sampling |
| 16:01.57 | nny_1 | Gah! |
| 16:02.02 | nny_1 | <-- hates AMP |
| 16:02.04 | b11d` | I dont know about every product under the sun, so i ask.. just in case its out there.. |
| 16:02.09 | nny_1 | AMPORTAL/ FREEPBS etc |
| 16:02.09 | mackes-Office | Has anyone here attended BootCamp or The Sip Master training- I am interested as to if it is worth while- |
| 16:02.12 | nny_1 | pbx* |
| 16:02.18 | DaneM | Hello, all. I'm getting a strange installation error when I try to use pbuilder to create an Ubuntu package. The compile works fine, but when it gets to the installation phase, I get this: build_tools/mkpkgconfig: 34: cannot create /usr/lib/pkgconfig/asterisk.pc: Permission denied . I've googled and looked on the Asterisk forums, and I just can't figure it out. Any suggestions? |
| 16:02.26 | rupa | ... HD Voice ? What codec do you use with that? |
| 16:02.33 | hmmhesays | sip master training sounds like a sales gimmick |
| 16:02.34 | nny_1 | DaneM: are you sudo or root? |
| 16:02.47 | nny_1 | hmmhesays: no it's Master training, there is none higher |
| 16:02.51 | nny_1 | :) |
| 16:02.56 | hmmhesays | I call bs on that one |
| 16:03.01 | DaneM | I've tried it as sudo and as root, calling pbuilder like so: pbuilder --build --basetgz /var/cache/pbuilder/base-i386.tgz ../asterisk_1.4.19.1-1.dsc |
| 16:03.10 | DaneM | (@nny_1) |
| 16:03.10 | nny_1 | can't there be only one true master?' |
| 16:03.16 | nny_1 | maybe it's like highlander |
| 16:03.38 | nny_1 | they let you decapitate the other attendees at the end, winner take all |
| 16:03.47 | hmmhesays | that would be worth my time |
| 16:03.58 | *** join/#asterisk mort_gib (n=mjensen@adsl-2-234.gibnet.gi) |
| 16:04.14 | mort_gib | When do I need ztdummy running?? |
| 16:04.24 | nny_1 | mort_gib: when you have no hardware as a timing source |
| 16:04.37 | nny_1 | mort_gib: like a digium card, or FXO/FXS card T1 card etc |
| 16:04.57 | mort_gib | So a Sangoma A200 would work as a timing device?? |
| 16:05.01 | mort_gib | HOw do I know?? |
| 16:05.13 | hmmhesays | yes it will |
| 16:05.24 | mort_gib | OK, thanks! |
| 16:05.30 | nny_1 | DaneM: does the /usr/lib/pkconfig/ dir exist? |
| 16:05.47 | DaneM | let me see. (logging into the pbuilder tgz...) |
| 16:05.56 | mort_gib | I have some pretty strange behavior on a * server, and I'm grasping for straws! |
| 16:06.52 | DaneM | nny_1: hmmm...no it doesn't. Creating it now |
| 16:06.54 | nny_1 | back to my callerid fun.. I am also mangling this old calleridshell agi to handle the updated web sources if anyone wants to have a stab at it.. got it to use the nanpa.txt, but all the web sources it uses look like they have made modifications to their format since it's inception, |
| 16:09.18 | ManxPower | nny_1: why don't you just pay for callerid Name service from your telco?> |
| 16:10.10 | ManxPower | Seems to be a lot of hassle for something that Just Works |
| 16:10.16 | nny_1 | ManxPower: good point, not sure if they even offer it lol |
| 16:10.47 | [TK]D-Fender | b11d`: HD Voice = G.722 = only useful directly between IP phones. Second you hit the PSTN you're dragged back to the LCD... (G.711) |
| 16:10.51 | nny_1 | ManxPower: hah it worked for a residential call :D |
| 16:10.52 | ManxPower | heck if you have a PRI, many times it's enabled by default |
| 16:11.16 | b11d` | i see.. |
| 16:11.37 | b11d` | well even g711 sounds better on those 550 HD receivers than on the 501's.. |
| 16:12.21 | hmmhesays | it won't if you're calling any analog endpoint |
| 16:12.35 | b11d` | it does though.. ive got one.. |
| 16:12.36 | ManxPower | analog or digital does not matter. |
| 16:12.37 | b11d` | it sounds amazing |
| 16:12.48 | b11d` | im sure its not as good as pure g722.. |
| 16:12.49 | mort_gib | I'm still struggling with dropped calls, I would really appreciate fresh ideas... |
| 16:12.53 | DaneM | nny_1: OK. I've tried compiling it again, with the directory created in the chroot, and it still gives me the error: build_tools/mkpkgconfig: 34: cannot create /usr/lib/pkgconfig/asterisk.pc: Permission denied |
| 16:12.54 | b11d` | but it does sound better than the 501 handsets |
| 16:12.57 | ManxPower | b11d`: then your phone is providing you the max quality available for G711 |
| 16:13.03 | b11d` | yep |
| 16:13.16 | ManxPower | mort_gib: turn off busydetect and callprogress if you have them enabled. |
| 16:13.25 | mort_gib | Hang on |
| 16:14.00 | mort_gib | -In sip.conf |
| 16:14.20 | ManxPower | no, in zapata.conf |
| 16:14.59 | ManxPower | Since you did not bother to provide any additional information, I have to assume Zap |
| 16:15.23 | *** join/#asterisk outtolunc (n=me@adsl-66-218-53-172.dslextreme.com) |
| 16:15.40 | mort_gib | ManxPower: What info do you need?? All phones are SIP, SNOM 300-370 |
| 16:15.54 | mort_gib | * Dell PE840, decent spec, CentOS 4.4 |
| 16:15.58 | ManxPower | how do you connect to the telco? |
| 16:16.01 | nny_1 | DaneM: phone one sec |
| 16:16.07 | DaneM | ok. Thanks |
| 16:16.24 | mort_gib | BRI (Sangoma A500 card) but dropped calls are also internal |
| 16:16.41 | ManxPower | mort_gib: Lets deal with the external first. |
| 16:16.52 | mort_gib | -Sure |
| 16:16.58 | ManxPower | you need to make sure callprogress and busydetect are not configured for that card. |
| 16:17.09 | *** join/#asterisk gitguy (n=diego@adsl-134-171.click.com.py) |
| 16:17.11 | mgroman | Can anyone here suggest a decent wireless voip hardphone? |
| 16:17.40 | [TK]D-Fender | mgroman: ... |
| 16:17.43 | DaneM | mgroman: try here:  |
| 16:17.43 | DaneM | http://www.voipsupply.com/index.php?cPath=95_115 |
| 16:17.44 | [TK]D-Fender | ~wifivoip |
| 16:17.45 | jbot | [~wifivoip] Wi-Fi (802.11a/b/g) VoIP phones on the market may possess any of the following drawbacks : poor battery life, limited range, lack of call features (many lack transfer, etc), poor or no NAT traversal, shoddy construction, etc. Because of this they are usually not recommended. Some alternatives are : VoIP ATA + cordless, DECT w/ VoIP base, etc |
| 16:17.53 | gitguy | hi, what do you guys think of this: http://digg.com/software/How_does_FreeSWITCH_compare_to_Asterisk -- i don't meat to troll or anything, just curious... |
| 16:18.01 | gitguy | meant* |
| 16:18.14 | rupa | mgroman, better off going with dect... |
| 16:18.29 | [TK]D-Fender | mgroman: most seem to think the Hitachi suck less than the rest. |
| 16:18.39 | mort_gib | ManxPower: They are not turned on, in fact they are not in the config files |
| 16:18.42 | mgroman | rupa: [TK]D-Fender: thanks just doing some preliminary research |
| 16:18.48 | mgroman | DaneM: self-plug? |
| 16:19.01 | [TK]D-Fender | mgroman: But DECT + SIP base is a better idea. ATA + Cordless also. |
| 16:19.07 | ManxPower | mort_gib: 90% of dropped call issues are cause by those two settings. |
| 16:19.20 | DaneM | mgroman: huh? I'm not any kind of expert on the matter. I just saw the site this morning. |
| 16:19.28 | ManxPower | Best of luck diagnosing your problem -- it falls in that other 10& |
| 16:19.29 | mgroman | DaneM: I was just kidding, thanks for the link |
| 16:19.36 | DaneM | hehe no prob |
| 16:19.44 | mort_gib | OK, but as mentioned I get dropped calls on both internal and external |
| 16:20.04 | mort_gib | So maybe a default setting?? |
| 16:20.28 | ManxPower | they are NEVER EVER enabled by default |
| 16:20.46 | mort_gib | I'm using Sangomas smg stuff |
| 16:21.06 | DaneM | I'm kind-of curious as to what a good cord-free solution for a SIP phone would be as well. I'm looking into setting up as SIP server soon, and I don't really know what works. (which is why I was looking at the above link :-) |
| 16:21.07 | mort_gib | Works great, only I could do without the dropped calls... |
| 16:22.09 | DaneM | I'm trying to get away from analog hardware... |
| 16:22.21 | mgroman | !spa922 |
| 16:22.28 | *** part/#asterisk jivco (n=jivco@85.187.217.6) |
| 16:22.29 | mgroman | ~spa922 |
| 16:23.00 | mort_gib | DaneM: The Snom M3 is okay.... |
| 16:23.12 | keith4 | so, I can put a bunch of sip users in a call group and a pickup group. is there any way to dial that group, in the dialplan? e.g, is there an analog to the Zap/g1 syntax? |
| 16:23.21 | DaneM | mort_gib: thanks. I'll look into it. |
| 16:24.03 | [TK]D-Fender | mgroman: No point to Linksys phones in North America typically. |
| 16:25.23 | rupa | mort_gib, can one base station support multiple extentions? |
| 16:26.14 | mgroman | [TK]D-Fender: they are garbage in North America? |
| 16:26.39 | mort_gib | rupa: -Yes |
| 16:26.44 | mort_gib | I think up to 8 |
| 16:26.45 | [TK]D-Fender | mgroman: No, not garbage, just inferior. |
| 16:26.52 | rupa | ponders |
| 16:26.59 | [TK]D-Fender | mgroman: For the price Polycom can't really be beat. |
| 16:27.05 | mort_gib | And you can map multiple identities (I think) but that I haven't tried ;-) |
| 16:27.26 | mort_gib | so one identity rings on handset 1, another on handset 2 etc |
| 16:27.56 | ManxPower | keith4: yes |
| 16:28.01 | mgroman | googles Polycom |
| 16:28.10 | ManxPower | Dial(SIP/1234&Zap/1) |
| 16:28.57 | *** join/#asterisk Hawk36 (n=me@modemcable202.30-70-69.static.videotron.ca) |
| 16:29.01 | mort_gib | Anyone seen this msg?? The CT_C8_A clock behavior does not conform to the H.100 spec! |
| 16:29.02 | Hawk36 | Hi |
| 16:29.32 | [TK]D-Fender | mgroman: http://www.telephonydepot.com/Polycom_s/25.htm |
| 16:29.39 | Hawk36 | I am completely new to asterisk and I would like to talk in private with someone willing to answer basic questions and help me setup my first system |
| 16:29.53 | keith4 | ~book |
| 16:29.54 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com, or see ~buybook |
| 16:29.57 | nny_1 | is using the 962 phone.. so far it weighs against the polycom, my .02 |
| 16:30.01 | keith4 | Hawk36: ^^ |
| 16:30.02 | rupa | mort_gib, interesting. I'll havec to get a few when I have the spare cash lying around. the extra handsets are a bit pricey fora dect phone. |
| 16:30.20 | *** join/#asterisk djs26 (n=djs@unaffiliated/djs26) |
| 16:30.29 | keith4 | ManxPower: well, yes. but that gets tedious for lots of SIP/whatever. so, then I use a macro, but it requires constant updating. It seems that there isn't a similar way of doing "Zap/g2" with sip then |
| 16:30.34 | mort_gib | Yeah, and it's a bit "plasticy" for my taste, but the functionality is ok |
| 16:30.48 | [TK]D-Fender | mgroman: Gor the grade you're looking at, that'd be the IP 320/330 |
| 16:30.54 | b11d` | so.. who makes a good analog telephone then? I am looking at avaya.. |
| 16:30.59 | mackes-Office | Dial(SIP/1234&Zap/1) --- I have had trouble with Zap and SIP in the same dial command- because the Zap always replies with Answered instantly- stopping the SIP from ringing |
| 16:31.16 | ManxPower | mackes-Office: that only happens on FXO ports |
| 16:31.22 | mackes-Office | Ahhh |
| 16:31.26 | ManxPower | Since he is dialing phones, he's calling FXS ports |
| 16:31.29 | mackes-Office | Yep |
| 16:31.37 | [TK]D-Fender | mackes-Office: You could enable call progress detection on your Zap channel, but that optiosn is synonymous with "drop my calls at random" |
| 16:31.50 | mackes-Office | I do notices that with my PRI connections |
| 16:31.58 | *** join/#asterisk jmesquita (n=jmesquit@201.7.117.114) |
| 16:32.07 | ManxPower | *nod* PRI does not answer as soon as dialing is done |
| 16:32.32 | Hawk36 | I have troubles setting up my first asterisk on CentOS5.0 any help |
| 16:32.53 | [TK]D-Fender | Hawk36: Describe your problem(s) |
| 16:32.59 | Hawk36 | Sorry 5.1 |
| 16:33.11 | Hawk36 | Here directly on the forum? |
| 16:33.24 | [TK]D-Fender | Hawk36: justr get on with it... |
| 16:33.31 | Hawk36 | Sorry new to this |
| 16:33.39 | Hawk36 | It seems to work |
| 16:34.05 | Hawk36 | However I cannot register a linksys SPA962 |
| 16:34.15 | Hawk36 | It just says fails |
| 16:34.15 | [TK]D-Fender | Wow.. IP 4000 upgraded to HD & no ver with Microbrowser, etc.. |
| 16:34.49 | [TK]D-Fender | Hawk36: ok, this has nothing to do with INSTALLING *. As for failure, pastebin the CLI output with SIP DEBUG enabled showing the failure along with your configs. |
| 16:34.55 | [TK]D-Fender | ~pb |
| 16:34.56 | jbot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste |
| 16:34.57 | [TK]D-Fender | ^^^^^^^^^^^^ |
| 16:35.08 | *** join/#asterisk hesco (n=hesco@c-76-17-99-50.hsd1.ga.comcast.net) |
| 16:35.26 | Hawk36 | Actually when I have SIP debug enabled and the device tries to connect I get nothing at all |
| 16:36.00 | [TK]D-Fender | Hawk36: If you get nothing then you either have a firewall/netowrking issue, or your phone isn't talking to the right box at all. |
| 16:36.07 | ManxPower | hands [TK]D-Fender a drink. Here, you'll needthis. |
| 16:36.17 | [TK]D-Fender | ManxPower: Nah, this should be short. |
| 16:36.21 | keith4 | Hawk36: at least have a look at /var/log/asterisk/messages |
| 16:36.26 | *** join/#asterisk pikachu2000 (n=pikachu2@196-209-10-21-ndn-esr-2.dynamic.isadsl.co.za) |
| 16:36.30 | [TK]D-Fender | keith4 : Wasted effort. |
| 16:36.47 | hesco | I had a power failure this morning and after I reset my network, copying a .call file to the outgoing spool yields the following error: |
| 16:36.47 | [TK]D-Fender | keith4 : No SIP debug = no communication = nothing to see. |
| 16:36.49 | hesco | pbx_spool.c:346 scan_service: Unable to open /var/spool/asterisk/outgoing/17707551543.call: Permission denied, deleting |
| 16:36.51 | ManxPower | If there's nothing in SIP debug, then packets are never reaching Asterisk |
| 16:37.01 | hesco | What would that be about? |
| 16:37.09 | keith4 | [TK]D-Fender: maybe a failed registration attempt, no? |
| 16:37.11 | ManxPower | hesco: It's a permissions problem, just like it says |
| 16:37.14 | [TK]D-Fender | hesco: You don't "copy" cal files, you MOVE then |
| 16:37.27 | rob0 | Okay, I restated my question from last hour: http://pastebin.com/d612db27d |
| 16:37.28 | keith4 | hesco: or use cp -a |
| 16:37.31 | [TK]D-Fender | keith4 : he just said "nothing" on SIP debug. |
| 16:37.41 | DaneM | does anybody have an idea on my post-compile installation error with pbuilder on Ubuntu?: build_tools/mkpkgconfig: 34: cannot create /usr/lib/pkgconfig/asterisk.pc: Permission denied |
| 16:37.48 | rob0 | about macro() and general dialplanning |
| 16:37.56 | keith4 | [TK]D-Fender: do you believe him? |
| 16:38.27 | Hawk36 | Why would I lie |
| 16:38.38 | ManxPower | rob0: Yes, you can call a macro from within a macro. |
| 16:38.42 | [TK]D-Fender | keith4 : Well its convenient to in this case. Nothing for us to waste time thinking about in believing it. |
| 16:38.45 | gitguy | <PROTECTED> |
| 16:38.47 | b11d` | so.. who makes a good analog telephone then? I am looking at aastra.. avaya doenst seem to make a 2-line analog.. |
| 16:38.51 | ManxPower | Just remember Macro does NOT clear ANY variables. |
| 16:38.55 | gitguy | what is that, i see it when i sip show peer foo |
| 16:38.56 | keith4 | [TK]D-Fender: fair 'nuff |
| 16:38.59 | [TK]D-Fender | Hawk36: Go check your phone's config again and check your firewall. |
| 16:39.03 | hesco | whether I cp or mv them, I get the same error. |
| 16:39.05 | ManxPower | So if you had an ARG1 set in macro1, it will also be set in macro2 |
| 16:39.10 | Hawk36 | I know my firewall is fine |
| 16:39.25 | ManxPower | hesco: are you really going to make us hold your hand as we solve your permission problem? |
| 16:39.25 | [TK]D-Fender | Hawk36: Then I guess you have all the answers. best of luck with this. |
| 16:39.49 | keith4 | Hawk36: i wasn't suggest that you're lying... but different people have different ideas of "nothing" |
| 16:39.59 | keith4 | sometimes it means "nothing that I can make sense of" |
| 16:40.00 | ManxPower | hesco: What user is Asterisk running as? What is the owner of the .call file. What are the permissions on the .call files |
| 16:40.02 | Hawk36 | I followed step by step the Book Asterisk 2nd edition |
| 16:40.05 | keith4 | sometimes it means "i'm looking in the wrong place" |
| 16:40.13 | gitguy | what is the Expire thing on peers |
| 16:40.13 | keith4 | or "i didn't actually turn on sip debug" |
| 16:40.14 | gitguy | .... |
| 16:40.26 | Hawk36 | keith4, I understand |
| 16:40.28 | ManxPower | keith4: Then he is FAR beyond our help |
| 16:40.29 | [TK]D-Fender | Hawk36: Meaningless description. Show us SIP debug. If you can't get that, then the problem is as I've described before. |
| 16:40.42 | rob0 | okay, I can handle that, thanks. |
| 16:41.55 | Hawk36 | asterisk*CLI> sip set debug |
| 16:41.55 | Hawk36 | SIP Debugging re-enabled |
| 16:42.08 | Hawk36 | Then I try to register and nothing occurs |
| 16:42.21 | [TK]D-Fender | Hawk36: then you're left with the scenario I already layed out for you. |
| 16:42.34 | keith4 | phone no talky to asterisk box |
| 16:42.35 | Hawk36 | Firewall issue right? |
| 16:42.37 | [TK]D-Fender | Hawk36: Go check your phone, and go check all of the networking and firewalls in between. |
| 16:42.42 | [TK]D-Fender | ^^^^^^^^^^ |
| 16:42.59 | ManxPower | Registration only uses UDP/5060, but audio uses many more ports. |
| 16:43.10 | Hawk36 | Ok for the firewall issue, I know it is fine since I opened up the port 5060 |
| 16:43.15 | *** part/#asterisk codefreeze-lap (n=murf@216.166.159.235) |
| 16:43.43 | ManxPower | Hawk36: TCP, UDP, or both? |
| 16:43.44 | Hawk36 | I set up the phone using the specification given by the book and through my research on the web |
| 16:43.47 | ManxPower | Source or dest? |
| 16:43.52 | Hawk36 | UDP |
| 16:43.55 | hesco | I had started asterisk with sudo, as myself. When I did a sudo su - ; then started asterisk, it worked. |
| 16:44.15 | keith4 | doesn't it need 5060 tcp? |
| 16:44.24 | ManxPower | hesco: *nod* That would be expected. If you want my help start answering my questions. |
| 16:44.56 | ManxPower | ManxPower: hesco: What user is Asterisk running as? What is the owner of the .call file. What are the permissions on the .call files |
| 16:44.59 | ManxPower | I won't ask a 3rd time. |
| 16:45.28 | ManxPower | hesco: your problem is so dirt simple any decent systems admin could have it fixed in 5 mins. |
| 16:45.39 | [TK]D-Fender | Hawk36: Sorry, but you haven't added anything of value. |
| 16:45.42 | *** join/#asterisk nny_1 (n=scott@64.203.239.83) |
| 16:45.59 | Hawk36 | Sorry my mistake |
| 16:45.59 | *** join/#asterisk codefreeze-lap (n=murf@216.166.159.235) |
| 16:46.09 | Hawk36 | Just checked under my firewall |
| 16:46.23 | Qwell | packets can go under it? |
| 16:46.24 | Hawk36 | I had added a new port 5060 using TCP |
| 16:46.49 | Hawk36 | It shows that SIP is added to the other ports |
| 16:46.52 | [TK]D-Fender | Hawk36: Where is your phone located relative to *? |
| 16:46.59 | ManxPower | As SIP uses UDP, not TCP...... |
| 16:47.04 | [TK]D-Fender | Hawk36: And you won't get voice with only 5060 |
| 16:47.15 | DaneM | Hi, nny_1. I'm still stuck. Are you up to working with my problem some more? |
| 16:47.17 | Hawk36 | Ok |
| 16:47.30 | Hawk36 | Then there is something I truly lost here |
| 16:47.44 | nny_1 | DaneM: yeah sure sorry had to change OS, in Ubuntu now |
| 16:47.53 | nny_1 | DaneM: so still permission denied? |
| 16:48.18 | DaneM | nny_1: s'ok. Yup. I edited the Makefile to add the directory in the build environment, and I'm trying to re-re-re-re...compile. |
| 16:48.23 | DaneM | :-p |
| 16:48.44 | DaneM | I'll know in a sec whether it worked. |
| 16:49.04 | ManxPower | Hawk36: First you say you added SIP/UDP 5060 to your firewall, then you say you added SIP/TCP 5060, which is it? |
| 16:49.16 | *** join/#asterisk mltlnx (n=mltlnx@209.10.153.194) |
| 16:49.20 | Hawk36 | It was my mistake |
| 16:49.26 | ManxPower | why not just turn off the damn firewall until you manage to fix the problem? |
| 16:49.41 | ManxPower | Hawk36: Which do you have? |
| 16:49.43 | Hawk36 | I had sip tcp and just put udp and I finally got green light |
| 16:50.07 | ManxPower | Now, aren't you glad we didn't believe you when you said it's not a firewall problem |
| 16:50.08 | nny_1 | DaneM: whats the error you are getting? |
| 16:50.20 | nny_1 | Lies! |
| 16:50.27 | DaneM | nny_1: hmmm...looks like my workaround didn't work. It says that the directory was already there. One min...I'll re-post my error. |
| 16:50.39 | nny_1 | the firewall is all poweful and knowing, this is blasphemy! |
| 16:50.40 | Hawk36 | guys I'm sorry I don't understand completly |
| 16:50.48 | nny_1 | :P |
| 16:50.54 | ManxPower | next time don't argue with the experts |
| 16:51.05 | ManxPower | It pissed them off and you are good with ketchup |
| 16:51.12 | Hawk36 | I thought this was the place to ask, if not can you please direct me to the right channal so I don't bother anyone with my newbie questions |
| 16:51.14 | jaytee | and crunchy |
| 16:51.44 | ManxPower | Hawk36: newbie questions are OK. Newbies telling the experts "my firewall is fine" is just arguing with the experts. |
| 16:51.49 | Hawk36 | Well never intended to argue, just tring to understand |
| 16:51.55 | rob0 | NO place is the right place to ask until you have RTFM'ed and RTFwiki'ed. |
| 16:52.03 | [TK]D-Fender | Hawk36: this is the place to ask, but you showed nothing and left us blank assurances that "of course my firewall is perfect". |
| 16:52.10 | nny_1 | hey experts, it is better to set the callerid of the incoming caller in my incoming/default/ophshitcalldontdie context, correct? |
| 16:52.12 | [TK]D-Fender | Words to remember : |
| 16:52.15 | [TK]D-Fender | ~assume |
| 16:52.15 | jbot | About assumptions : http://www.moviewavs.com/php/sounds/?id=gog&media=WAVS&type=Movies&movie=Long_Kiss_Goodnight"e=assume.txt&file=assume.wav |
| 16:52.16 | [TK]D-Fender | ^^^^^ |
| 16:52.19 | Hawk36 | Sorry if that is what I left as an impression |
| 16:52.35 | DaneM | nny_1: there it is: build_tools/mkpkgconfig: 34: cannot create /usr/lib/pkgconfig/asterisk.pc: Permission denied |
| 16:52.45 | Hawk36 | Can we start over? ;) |
| 16:52.51 | [TK]D-Fender | nny_1: You don't normally set the caller ID of the call.. it jsut comes "in". |
| 16:52.52 | ManxPower | DaneM: you are of course root when you run this, right? |
| 16:53.17 | [TK]D-Fender | Hawk36: Yes, now try to keep an open mind. |
| 16:53.19 | DaneM | ManxPower: I've been doing sudo for the most part, although I've tried root. I'm using pbuilder/fakeroot, if that matters. |
| 16:53.37 | DaneM | I'll try as root (not sudo) again, just to be sure. |
| 16:53.42 | Hawk36 | I will do my best and if I screw up I'm sure you guys will bring me back lol |
| 16:53.47 | ManxPower | DaneM: you are having a distro specific problem. Why are you not asking on the support channel for your distro? |
| 16:53.50 | nny_1 | [TK]D-Fender: yeah that works, although i get channel:NUMBER@IP right now on my poly 601 |
| 16:53.54 | *** join/#asterisk stkn|work (n=stkn|wor@gentoo/developer/pdpc.active.stkn) |
| 16:53.54 | nny_1 | was trying to clean it up a bit |
| 16:54.17 | ManxPower | nny_1: then you didn't get callerid |
| 16:54.19 | [TK]D-Fender | nny_1: so only looks funny on the phone itself, right? |
| 16:54.20 | DaneM | ManxPower: I guess I'll give it a shot there. I wasn't sure whether any of you had noticed any similar problem. |
| 16:54.25 | Hawk36 | So you said I would not get voice using UDP 5060, why is it recommended by the book? |
| 16:54.32 | nny_1 | ManxPower: replace NUMBER with the correct number |
| 16:54.39 | [TK]D-Fender | Hawk36: SIP = call signalling, RTP = VOICE <- |
| 16:54.42 | nny_1 | [TK]D-Fender: yeah, maybe i should test another phone heh |
| 16:54.46 | ManxPower | DaneM: I did, but since you are having a distro specific problem, I just ignored them. |
| 16:54.53 | [TK]D-Fender | nny_1: No... NoOp it |
| 16:55.02 | DaneM | OK. |
| 16:55.14 | Hawk36 | What dialplan do you guys recommend |
| 16:55.23 | [TK]D-Fender | Hawk36: extensions.conf <- |
| 16:55.47 | ManxPower | Hawk36: Call setup/teardown is done using SIP (UDP/5060), Audio is done using the RTP protocol (/etc/asterisk/rtp.conf sets the RTP ports) |
| 16:55.48 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 16:55.51 | Hawk36 | I guess my question was bad |
| 16:56.05 | nny_1 | [TK]D-Fender: roger that |
| 16:56.09 | ManxPower | RTP is all UDP, but does not have specific defined port numbers. |
| 16:56.17 | *** join/#asterisk eXistenZ (n=existenz@unaffiliated/existenz) |
| 16:56.56 | DaneM | Thanks for your tips, all. Have a good one. |
| 16:57.00 | *** part/#asterisk DaneM (n=dane@adsl-76-236-27-148.dsl.chi2ca.sbcglobal.net) |
| 16:57.00 | Hawk36 | So if I understand I need to open some RTP ports if I wish to have voice |
| 16:57.18 | [TK]D-Fender | Hawk36: 10000-20000 typical range. |
| 16:57.32 | Hawk36 | Do I need one portr per phone? |
| 16:57.40 | ManxPower | Hawk36: 2 ports per call |
| 16:57.49 | *** join/#asterisk b1shop (n=b1shop@dsl081-149-253.chi1.dsl.speakeasy.net) |
| 16:57.57 | ManxPower | set them to whatever you want in /etc/asterisk/rtp.conf |
| 16:57.57 | Hawk36 | ok so on a 6 phone system I would need 12 ports correct? |
| 16:58.14 | ManxPower | Hawk36: I just SAID it had nothing to do with phones. |
| 16:58.21 | Hawk36 | sorry |
| 16:58.34 | ManxPower | How many calls do you want to do at once outside the system? |
| 16:58.35 | Hawk36 | per call |
| 16:58.41 | Hawk36 | I see |
| 16:59.01 | Hawk36 | so lets use our company as an exemple |
| 16:59.06 | Hawk36 | We have 4 lines right now |
| 16:59.17 | Hawk36 | So that would mean we do 4 simultan calls |
| 16:59.23 | Hawk36 | So I would need 8 ports correct |
| 16:59.28 | ManxPower | By "lines" I assume you mean analog lines from the telco. |
| 16:59.30 | [TK]D-Fender | Hawk36: Nope. |
| 16:59.33 | Hawk36 | to have those 4 calls |
| 16:59.37 | Hawk36 | yes |
| 16:59.42 | Hawk36 | analog |
| 16:59.56 | ManxPower | Hawk36: so you would need 8 ports for the analog lines. I assume you never have two phones call each other? |
| 17:00.08 | ManxPower | (well two phones that are not on the local LAN with the Asterisk server, at least) |
| 17:00.19 | [TK]D-Fender | ManxPower: Then semantic here... this is going into "stupid" territory fast on assumptions... |
| 17:00.29 | [TK]D-Fender | think* |
| 17:00.53 | ManxPower | [TK]D-Fender: he doesn't even know enough to ask the right questions -- I assume he doesn't know enough to provide the right answers too. |
| 17:01.05 | Hawk36 | Probably |
| 17:01.15 | Hawk36 | Beleive me I'm trying |
| 17:01.17 | ManxPower | Hawk36: and why do you even care to limit the ports so much. |
| 17:01.34 | Hawk36 | I do not care, just trying to understand |
| 17:01.35 | ManxPower | just open 100 ports, set rtp.conf, allow them in your firewwall, be done with it |
| 17:02.06 | ManxPower | unless you expect to have more than 50 calls at any one time. |
| 17:02.06 | Hawk36 | let says I open 100 ports that means I can do 50 calls right? |
| 17:02.10 | [TK]D-Fender | Hawk36: What exact hardware is letting your take in your analog lines? How many PHONES are you looking at getting. What KIND of phones? Connected how? |
| 17:02.47 | Hawk36 | Fender, I was just asking as an exemple |
| 17:02.57 | Hawk36 | To understand RTP |
| 17:03.03 | [TK]D-Fender | Hawk36: And your example has more holes than a block of swiss cheese |
| 17:03.12 | Hawk36 | I can see |
| 17:03.22 | Hawk36 | Please be a little patient |
| 17:03.22 | rob0 | mmmm cheese |
| 17:03.23 | *** join/#asterisk Kimitaka (n=swiceje@cpe-065-184-219-014.ec.res.rr.com) |
| 17:03.30 | [TK]D-Fender | Hawk36: it makes plenty of assumptions I'm not even going to attempt to "assume" just to feed you an answer. |
| 17:03.36 | Hawk36 | It is hard when you are new to all this |
| 17:04.12 | Kimitaka | how would you put a door phone in the dial plan, like where you pick up the phone and it automatically calls several extensions without being dialed |
| 17:04.18 | [TK]D-Fender | Hawk36: the devil is in the details, and seeing how much of a miser you are on your firewalling, you are likely to amke a setup that will come back and bite your ass right off a-la-Jaws. |
| 17:04.36 | destructure | dorsal fin and all |
| 17:04.43 | rob0 | with cheese? |
| 17:04.55 | Hawk36 | That is why I am testing and learning |
| 17:05.03 | [TK]D-Fender | Kimitaka: ATA with "dial on pickup" or zap analog channel + "immediate=yes" |
| 17:05.20 | Hawk36 | I have another bad question |
| 17:05.39 | Hawk36 | I thought It was not needed to have an analog card |
| 17:06.29 | Hawk36 | Can I dial an analog line using a sip phone? |
| 17:06.55 | ManxPower | Hawk36: Yes. |
| 17:07.11 | ManxPower | Hawk36: Why you don't just go read The Book and save everyone time? |
| 17:07.16 | Hawk36 | Manx can I private you? |
| 17:07.17 | ManxPower | ~book |
| 17:07.17 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com, or see ~buybook |
| 17:07.31 | rob0 | I would take that as a "no". |
| 17:07.31 | ManxPower | Hawk36: only if it includes a credit card number to pay for my consulting service |
| 17:07.38 | Hawk36 | lol |
| 17:07.56 | Hawk36 | Sorry I guess, I'll have to wait a bit |
| 17:08.15 | Hawk36 | Manx that is the book I am reading and going through |
| 17:08.21 | [TK]D-Fender | Hawk36: By what miracle did you think your analog line would be able to send its calls to * without a card or similar deice? |
| 17:08.36 | Hawk36 | That it not what I wanted |
| 17:08.52 | Hawk36 | Never did I mention I wanted to dial out with my analog lines using asterisk |
| 17:08.59 | *** join/#asterisk doke (n=doke@unaffiliated/emrah) |
| 17:09.11 | doke | Hello hello! |
| 17:09.23 | doke | Is there anybody here that undersand sip well? |
| 17:09.24 | Hawk36 | I now have a dial tone but can't access any lines |
| 17:09.34 | doke | I'm experiencing a very strange behavious |
| 17:09.37 | doke | behaviour* |
| 17:09.41 | doke | with Asterisk |
| 17:09.47 | [TK]D-Fender | Hawk36: Asterisk processes calls. The can come in from any number of different devices. * can, in that processing, DIAL a given device as you tell it to. |
| 17:09.48 | doke | http://pastebin.ca/1015950 |
| 17:09.57 | *** part/#asterisk gitguy (n=diego@adsl-134-171.click.com.py) |
| 17:10.03 | doke | I'm trying to register a CP7975G |
| 17:10.12 | keith4 | Hawk36: you're lucky. you caught [TK]D-Fender on a good day |
| 17:10.13 | [TK]D-Fender | Hawk36: Well what device would LET * use your "lines"? |
| 17:10.45 | Hawk36 | Fender, I guess I must purchase some kind of plan no? |
| 17:10.52 | [TK]D-Fender | keith4 : No, this is just "average", not good or bad. |
| 17:10.54 | rupa | ponders |
| 17:11.06 | ManxPower | Hawk36: plan? No, if you want to use your analog lines you must purchase an analog card |
| 17:11.07 | [TK]D-Fender | Hawk36: No, tio interface with your lines, you need HARDWARE. |
| 17:11.09 | keith4 | it sounds like he hasn't even read the first chapter of the book |
| 17:11.23 | Hawk36 | Ok, I do not wish to use my analog lines for now |
| 17:11.24 | ManxPower | keith4: I think I shall just stop helping. |
| 17:11.31 | rupa | ... or a SIP provider |
| 17:11.37 | doke | please? |
| 17:11.45 | [TK]D-Fender | Hawk36: If I want to use an analog line, I can buy a 10$ cheap ass POS phone and plug it in. That sure isn't a "plan", its a DEVICE. |
| 17:11.46 | Hawk36 | I just wish to be able to call out using asterisk for now |
| 17:12.01 | [TK]D-Fender | Hawk36: call out on WHAT then? |
| 17:12.02 | doke | I'm trying to register a sip device and Asterisk behaves totally very strangely or something is wrong on my side |
| 17:12.11 | Nasra | ManxPower : don't til you help me... |
| 17:12.12 | Hawk36 | My IP Phone |
| 17:12.18 | [TK]D-Fender | Hawk36: WRONG |
| 17:12.35 | [TK]D-Fender | Hawk36: Your IP phone doesn't magically connect to the PSTN. |
| 17:12.45 | Hawk36 | Ok |
| 17:12.49 | ManxPower | Hawk36: what number do you want to dial? |
| 17:12.52 | doke | I have a subnet where Asterisk listens on 10.0.0.8. Then a tunnel on 10.8.0.0/255.255.255.0 and the IP phone on subnet 172.16.24.0 |
| 17:12.56 | Hawk36 | any number |
| 17:12.57 | *** join/#asterisk pikachu2000 (n=pikachu2@pikachu.cyberdesigns.co.za) |
| 17:13.02 | [TK]D-Fender | Hawk36: A phone doesn't let you place calls to the PSTN. A phones conencted to a LINE does. |
| 17:13.27 | ManxPower | [TK]D-Fender: don't get stressed, get evin |
| 17:13.31 | ManxPower | even, even |
| 17:13.32 | Hawk36 | I thought I was able to purchase an IP line and use it on my IP phone |
| 17:13.33 | doke | http://pastebin.ca/1015950 |
| 17:13.37 | [TK]D-Fender | Hawk36: So again, what HARDWARE or SERVICE are you looking to use to let you place calls to the PSTN? |
| 17:13.51 | [TK]D-Fender | Hawk36: Now we might be getting somewhere. |
| 17:13.59 | rob0 | doke, if the OpenVPN peer isn't the default gateway for the 172.16.24.0 network, you need routes there to get back to 10.0.0.x. |
| 17:14.06 | [TK]D-Fender | Hawk36: the ther you're loking for is ITSP |
| 17:14.08 | [TK]D-Fender | ~itsp |
| 17:14.08 | jbot | [~itsp] An ITSP is an Internet telephony Service Provider or "VoIP Telephone company". The allow you to either PLACE calls to the PSTN (called Termination), RECEIVE calls from the PSTN (called Origination), or both. Some offer fixed rates, others $/min. Enter ~itsplist-us (USA) or ~itsplist-ca (Canada) for a listing of popular ITSPs |
| 17:14.25 | doke | rob0: I can register no problem with PJSIP |
| 17:14.25 | ManxPower | doke: put the [2104] section of sip.conf on pastebin.ca |
| 17:14.28 | *** join/#asterisk argos73 (n=mike@w010.z208036240.chi-il.dsl.cnc.net) |
| 17:14.28 | Hawk36 | I have zero hardware so I guess I need an IP service that can dial outside lines, is that it? |
| 17:14.33 | *** join/#asterisk inv_arp (n=junya@c-76-26-24-197.hsd1.fl.comcast.net) |
| 17:14.39 | [TK]D-Fender | Hawk36: And you you can pay for service with one to place/receive calls all via a VoIP protocol |
| 17:14.46 | Hawk36 | Yes |
| 17:14.59 | Hawk36 | That was my qurestion before |
| 17:15.00 | [TK]D-Fender | Hawk36: if you don't have the hardware, time to connect to someone who does.. |
| 17:15.10 | rob0 | Sorry, I saw the 10.8.0.0 and automatically figured it was someone who didn't understand IP routing. :) |
| 17:15.11 | *** join/#asterisk russellb_ (n=russell@asterisk/developer-and-stable-maintainer/drumkilla) |
| 17:15.11 | *** mode/#asterisk [+o russellb_] by ChanServ |
| 17:15.17 | rupa | is there a test # that is always busy? |
| 17:15.22 | rupa | PSTN # |
| 17:15.27 | Hawk36 | I asked if you guys knew of a good dial plan, I should of said a good service plan |
| 17:15.51 | ManxPower | Hawk36: you should not be on this channel, you should be concentrating on reading the book |
| 17:15.53 | keith4 | oy |
| 17:15.57 | [TK]D-Fender | Hawk36: No, you should ask "Hey, who's a decent ITSP in area {x}" |
| 17:16.07 | rob0 | area 51? |
| 17:16.13 | Hawk36 | Manx thanks |
| 17:16.43 | *** join/#asterisk kamanashisroy (n=kamanash@202.56.7.142) |
| 17:16.43 | Hawk36 | Hey, who's a decent ITSP in area 514 |
| 17:17.07 | keith4 | smacks his forehead |
| 17:17.13 | doke | ManxPower: http://pastebin.ca/1015966 |
| 17:17.20 | Nasra | Hawk36 ...you can google it |
| 17:17.30 | [TK]D-Fender | Hawk36: les.net and unlimitel.ca |
| 17:17.38 | Hawk36 | I did, just wanted to see if anyone recommends one more than another |
| 17:17.48 | Hawk36 | Thanks Fender |
| 17:17.49 | nny_1 | vitelity is good too |
| 17:17.53 | ManxPower | doke: This is not valid "callerid=("Emrah KAVUN" <2104>)" |
| 17:17.59 | Hawk36 | Sorry I dodn't know exactly how to ask it |
| 17:18.06 | nny_1 | well as long as you dont upset there draconian billing check mothods |
| 17:18.09 | nny_1 | methods* |
| 17:18.10 | Hawk36 | didn't |
| 17:18.12 | ManxPower | you want callerid=Emrah KAVUN <2104> no quotes, nothing extran |
| 17:18.27 | doke | rob0: all my devices can connect together... I'm currently in a Uni network in England trying to setup this phone for my girl friend.. She has a WRT54gl connected to Switzerland ;) |
| 17:18.28 | ManxPower | and cisco phones are very picky about callerid |
| 17:18.34 | nny_1 | their* gah me fail english |
| 17:18.41 | [TK]D-Fender | nny_1: pricey |
| 17:18.51 | doke | ManxPower: I can login to this peer with other softphones |
| 17:18.54 | nny_1 | [TK]D-Fender: really? i checked les.net whats the other? |
| 17:18.54 | Hawk36 | Fender, do you recommend one more than the other or they are identical in your POV |
| 17:19.00 | doke | no problem at all |
| 17:19.05 | nny_1 | [TK]D-Fender: was getting .015 per minute US and Canada |
| 17:19.14 | [TK]D-Fender | Hawk36: Service is much the same from what I hear |
| 17:19.20 | Hawk36 | Pricing? |
| 17:19.30 | rupa | Hawk36, they have websites |
| 17:19.31 | Hawk36 | We need CAN_US calls a lot |
| 17:20.06 | [TK]D-Fender | Hawk36: well you can use 1 providers for outbound calls in an area with a cheaper service. |
| 17:20.23 | [TK]D-Fender | Hawk36: You know. Go read. |
| 17:20.28 | [TK]D-Fender | ~itsplist-ca |
| 17:20.29 | jbot | [~itsplist-ca] Here are some popular ITSPs (Canada) starting with the more respected ones : http://www.unlimitel.com , http://www.les.net , http://www.babytel.ca |
| 17:20.30 | [TK]D-Fender | ~itsplist-us |
| 17:20.31 | jbot | [~itsplist-us] Here are some popular ITSPs (USA) starting with the more respected ones : http://www.teliax.com , http://connect.voicepulse.com , http://www.nufone.net , http://www.broadvoice.com, http://www.jnctn.com , http://www.bandwidth.com , http://vitelity.net |
| 17:20.31 | doke | ManxPower: any further suggestion? |
| 17:20.32 | ManxPower | doke: then your phone config is wrong |
| 17:20.38 | Hawk36 | What is a red line across the screen mean? |
| 17:20.53 | doke | have you looked at my pastebin SIP dialog? |
| 17:20.55 | ManxPower | Hawk36: nothing, Asterisk does not print a red line on the screen |
| 17:21.14 | Hawk36 | no here on the cannal |
| 17:21.18 | Hawk36 | chanel |
| 17:21.22 | ManxPower | Hawk36: then you should have said that. |
| 17:21.34 | Hawk36 | Is that a private message? |
| 17:21.35 | ManxPower | Hawk36: check the docs for your IRC client. |
| 17:21.45 | Qwell | sort of off-topic - but does anybody happen to know how cell GSM registrations work? apparently, you can have two phones with the same DID, and have incoming calls ring both |
| 17:21.50 | nny_1 | [TK]D-Fender: i realized that NoOp won't work cause the original assumption (no need to ~assume I view this as an error by someone else i can blame) is that the CALLERID was set.. I am updating the extension dialplan for good solid proper callerid, adding some nanpa.txt love for location, and pondering using the full AGI script for reverse lookups.. but I digress.. right now I am looking in the book and online for proper incoming callerid context |
| 17:21.57 | nny_1 | wow i wrote a book |
| 17:21.59 | nny_1 | !mybook |
| 17:22.27 | florz | Qwell: not really in detail - but yes, that is possible, absolutely |
| 17:22.30 | rob0 | nny_1, autograph it for us |
| 17:22.48 | florz | Qwell: I mean, rather obviously that must be possible =:-) |
| 17:23.00 | [TK]D-Fender | ManxPower: just something X-Chat inserts occasionally.. not sure on the triger. |
| 17:23.01 | nny_1 | heh signed Kenneth Rodriguez Consuello Johnson Egbert Douchebag Bonapart |
| 17:23.06 | nny_1 | the third |
| 17:23.17 | ManxPower | [TK]D-Fender: text since you last had the window to the foreground |
| 17:23.28 | *** join/#asterisk lesouvage (n=lesouvag@cc341200-a.assen1.dr.home.nl) |
| 17:23.41 | [TK]D-Fender | ManxPower: Could be. Never really bothered to verify :) |
| 17:24.04 | nny_1 | so yeah back to my book, no need [TK]D-Fender i just need to relearn how to properly do callerid first |
| 17:24.15 | *** join/#asterisk Nasra (n=Nasra@CPE001839494bc9-CM00111ade9528.cpe.net.cable.rogers.com) |
| 17:24.25 | *** join/#asterisk oej (n=olle@ns.webway.se) |
| 17:25.06 | [TK]D-Fender | nny_1: Your descriptions and pastebins are all vague. You need some coherence... |
| 17:28.13 | nny_1 | [TK]D-Fender: yeah you are right... let me learn more about how asterisk interprets and sends the CID, as well as why it says "asterisk" when no callerid is available, and I'll get back to you. I need to figure out how it works before I can address the formatting issue |
| 17:29.01 | rob0 | Hey, here's a dumb question about Zapateller - does it have a way to detect that it's a non-human caller, or do I have to put that logic in my dialplan somehow? |
| 17:32.09 | rob0 | I figure putting them to a menu is good enough, they rarely if ever will dial past Allison's this-call-may-be-monitored-or-recorded. |
| 17:32.09 | [TK]D-Fender | rob0: check out some of the answering machine detection code, etc. you could also jsut through them into an IVR and force them to make a choice. |
| 17:32.13 | rob0 | yup |
| 17:32.29 | [TK]D-Fender | rob0: Looks like you've already got your answer then. |
| 17:33.14 | rob0 | I used to have that, but this line here is FXO-less. We never gave out the number to anyone, and yet it gets spammed all the time. |
| 17:33.41 | [TK]D-Fender | rob0: ... huh? |
| 17:33.58 | [TK]D-Fender | rob0: "line here is FXO-less." <- wtf? |
| 17:33.59 | rob0 | phone line only for DSL |
| 17:34.24 | [TK]D-Fender | rob0: then whats answering it? |
| 17:34.41 | [TK]D-Fender | rob0: And you don't ahve to give it out.. that what phone book lists are for. |
| 17:34.47 | rob0 | most of the time, nothing/nobody, but we do have a phone plugged in now. |
| 17:35.01 | Hawk36 | Fender thanks I will try les.net |
| 17:35.12 | rupa | rob0, so unplug the phone |
| 17:35.23 | [TK]D-Fender | ^^^ |
| 17:35.26 | [TK]D-Fender | Sounds good to me |
| 17:35.30 | rob0 | we have done that, too :) |
| 17:35.53 | rob0 | but I have an unused TDM11B which I am going to bring here |
| 17:36.31 | jaytee | [TK]D-Fender, is this a valid statement if I'm trying to pass the last 4 digits to another context? exten => _NXXXXXX,1,Goto(directory,${EXTEN:-4,4}) |
| 17:37.29 | ManxPower | jaytee: ${EXTEN:3} would do what you want |
| 17:39.36 | [TK]D-Fender | exten => _NXXXXXX,1,Goto(directory,${EXTEN:3},1) |
| 17:39.55 | jaytee | ManxPower, I need to do matching against a list of 4 digit extensions, so exten => _NXXXXXX,1,Goto(directory,$EXTEN:3} would work and try to match against any of the 4 digit extensions there? |
| 17:40.11 | nny_1 | anyone have an opinion on HPEC vs. Digium Hardware echo cancel? |
| 17:40.21 | ManxPower | jaytee: no, ${EXTEN:3} will remove the first 3 digits of the value of EXTEN |
| 17:40.44 | ManxPower | jaytee: no "matching" involved |
| 17:42.27 | jaytee | ManxPower, sorry if I didn't make myself clear, I have one incoming context that normally won't get calls coming in. When a call comes into my * box from that trunk I am being passed 7 digits but all my extensions are 4 digit. |
| 17:43.16 | ManxPower | jaytee: I understand. |
| 17:43.28 | ManxPower | your question was answered. |
| 17:44.01 | [TK]D-Fender | jaytee: You should ask your telco to send you 10 digits IMO... |
| 17:44.08 | jaytee | ah, so when the call goes to the other context the value of ${EXTEN} has already had the 3 digits stripped? |
| 17:44.29 | *** part/#asterisk Shazaum (n=shazaum@200.175.61.250.static.gvt.net.br) |
| 17:44.45 | ManxPower | jaytee: ${EXTEN:3} will strip off the first three digits. What does that leave you with? |
| 17:45.00 | *** join/#asterisk fskrotzki (n=fskrotzk@host198.textwise.com) |
| 17:45.05 | jaytee | the last 4 |
| 17:45.10 | ManxPower | Exactly. |
| 17:45.27 | ManxPower | now TRY it. |
| 17:45.49 | *** join/#asterisk sysreq (n=sysreq@unaffiliated/sysreq) |
| 17:46.09 | jaytee | yep, gonna do that tonight. have to wait till after hours to test but I'm typing up the extensions.conf ahead of time. |
| 17:46.26 | jaytee | ManxPower, thanks! I think I always try to make it harder than it is :-) |
| 17:46.27 | ManxPower | We'll see you tomorrow then. |
| 17:47.06 | *** join/#asterisk hohum (n=dcorbe@64.214.185.114) |
| 17:48.28 | *** join/#asterisk mog (n=mog@c-68-62-219-86.hsd1.al.comcast.net) |
| 17:48.28 | *** mode/#asterisk [+o mog] by ChanServ |
| 17:48.35 | *** join/#asterisk eXistenZ (n=existenz@unaffiliated/existenz) |
| 17:51.02 | *** join/#asterisk djs26 (n=djs@unaffiliated/djs26) |
| 17:57.55 | nny_1 | is there another way to see an agi script work besides agi debug and running it standalone? |
| 17:58.24 | *** join/#asterisk Defraz (i=t0tal@69.92.19.83) |
| 18:12.35 | *** join/#asterisk M1s3ry (n=M1s3ry@216.207.245.1) |
| 18:14.19 | [TK]D-Fender | nny_1: What more are you expecting? |
| 18:15.38 | *** join/#asterisk rolnd (n=rolnd@S0106006097940f68.vw.shawcable.net) |
| 18:15.57 | hmmhesays | when you are using extconfig for voicemail do you still use the static conf file for general settings? |
| 18:16.51 | rolnd | how can one tell remote end that certain codec should be used, remote end always gets Accepting AUTHENTICATED call from ... requested format = unknown |
| 18:17.40 | ManxPower | rolnd: you configure the remote end for only one codec |
| 18:17.41 | *** join/#asterisk CunningPike (n=arodgers@204.239.12.183) |
| 18:18.34 | rolnd | ManxPower, what if I don't have ability to configure remote end, but it has list of 4 codecs |
| 18:18.47 | ManxPower | rolnd: then you can tell Asterisk to only accept one codec. |
| 18:19.00 | rolnd | ManxPower, remote always defaults to first codec |
| 18:19.13 | rolnd | ManxPower, because it never sees my preferred one being negotiated |
| 18:19.17 | ManxPower | rolnd: exactly how did you configure Asterisk to only accept 1 codec. |
| 18:19.45 | ManxPower | rolnd: Asterisk TELLS the phone what codecs it supports when the call comes in. If your remote device is ignoring that then there's nothin Asterisk can do about it. |
| 18:19.49 | rolnd | ManxPower, remote has 4 codecs, gsm being the first one, g729 second and so on |
| 18:19.57 | *** join/#asterisk Goldfisch (n=gturnqui@158-147-54-92.harris.com) |
| 18:20.04 | rolnd | ManxPower, local one has 2 codecs, g729 being the first one, gsm second and so on |
| 18:20.16 | rolnd | ManxPower, the problem is remote always uses *GSM* |
| 18:20.29 | ManxPower | rolnd: if you can't configure it, there really isn't much we need to know about the devices. |
| 18:20.30 | rolnd | ManxPower, because it sees requested format = unknown |
| 18:20.37 | ManxPower | Now, ANSWER my question. |
| 18:20.50 | ManxPower | rolnd: that is a normal thing on many systems, depending on your setup. |
| 18:20.53 | *** part/#asterisk Goldfisch (n=gturnqui@158-147-54-92.harris.com) |
| 18:21.11 | ManxPower | Registrations would have it, as would IAX2 switch => statement |
| 18:21.12 | rolnd | ManxPower, there is two asterisk boxes |
| 18:21.18 | rolnd | ManxPower, remote and local |
| 18:21.25 | rolnd | ManxPower, I don't have control of remote |
| 18:21.46 | ManxPower | rolnd: I will ask you one more time before I give up on you. "exactly how did you configure Asterisk to only accept 1 codec?" |
| 18:22.10 | *** join/#asterisk cyrilrbt (n=crobert@65.39.228.5) |
| 18:22.19 | rolnd | ManxPower, I did not configure asterisk to accept only one codec |
| 18:22.22 | cyrilrbt | hi everyone |
| 18:22.31 | rolnd | ManxPower, again I do not have control of remote just local box |
| 18:22.40 | ManxPower | rolnd: THEN DO IT ON THE BOX YOU CONTROL |
| 18:23.49 | ManxPower | I already know you don't control the remote side. I won't tell you to change it. |
| 18:24.13 | ManxPower | rolnd: you are making a dead simple task into a complex confusing thing. |
| 18:25.13 | hmmhesays | anyone? |
| 18:27.52 | *** join/#asterisk lesouvage (n=lesouvag@cc341200-a.assen1.dr.home.nl) |
| 18:28.26 | *** join/#asterisk iamhrh (n=iamhrh@office.amsvans.com) |
| 18:28.28 | mknerd | 14:18rolndManxPower, local one has 2 codecs, g729 being the first one, gsm second and so on 14:18rolndManxPower, the problem is remote always uses *GSM*, turn off gsm on the local one |
| 18:28.44 | mknerd | just turn on only the one you want on the local one |
| 18:29.39 | rob0 | GLOBAL() "get or set global variables", I can get, but not seeing an example how to set. |
| 18:29.57 | iamhrh | when members are added to a queue via dynamic realtime, should * be calling them when they already have a call? |
| 18:30.08 | ManxPower | rolnd: put the sip.conf from the box you control on pastebin.ca, masking only passwords |
| 18:30.22 | rob0 | I could try Set(VARIABLE|g) but that would be cheating :) |
| 18:30.57 | seanbright | hmmhesays: |
| 18:31.11 | hmmhesays | yes? |
| 18:31.13 | seanbright | err, fat fingered, sorry |
| 18:31.33 | seanbright | was h<tab>ing in another terminal |
| 18:31.35 | seanbright | :) |
| 18:32.38 | *** join/#asterisk tecnico (n=tecnico@user-24-214-56-217.knology.net) |
| 18:34.26 | ManxPower | mknerd: I don't think he really wants help\ |
| 18:35.23 | *** join/#asterisk mltlnx (n=mltlnx@209.10.153.194) |
| 18:35.53 | seanbright | hmmhesays: just glancing at the app_voicemail code, it looks like general settings are read from voicemail.conf even when using realtime |
| 18:36.03 | hmmhesays | yeah |
| 18:36.06 | hmmhesays | figured it out |
| 18:36.13 | seanbright | cool |
| 18:42.43 | *** join/#asterisk BrokenNoze (i=BrokenNo@79-75-233-86.dynamic.dsl.as9105.com) |
| 18:43.13 | BrokenNoze | Hi does anyone here use polycom's in depth? |
| 18:43.33 | rolnd | ManxPower, no it works if I use the single one, however my question was related to codec order |
| 18:44.20 | rolnd | ManxPower, if I do disallow=all,allow=g729,allow=gsm, shouldn't gsm be used first |
| 18:44.24 | rolnd | sorry |
| 18:44.27 | rolnd | g729 I mean |
| 18:44.35 | *** join/#asterisk outtolunc (n=me@adsl-66-218-53-172.dslextreme.com) |
| 18:44.36 | BrokenNoze | I am having serious issues with the 650's and using Auto answer, it only appears to work if 1 line key is ringing, anymore and the phone doesn't auto answer |
| 18:45.42 | *** join/#asterisk unbkbl (n=work@static-adsl201-232-88-87.epm.net.co) |
| 18:45.46 | ManxPower | BrokenNoze: You read the Wiki page? |
| 18:46.19 | ManxPower | rolnd: I can't help your further until you put your sip.conf on pastebin.ca, masking only the passwords |
| 18:46.37 | unbkbl | hello, i want to remove freepbx so that i can install again is there any way to uninstall it? nobody gave me an answer in #freepbx |
| 18:46.54 | ManxPower | unbkbl: format the system and reinstall your OS |
| 18:47.00 | unbkbl | hahaha |
| 18:47.03 | unbkbl | no... really |
| 18:47.10 | [TK]D-Fender | unbkbl: Sorry,but they are the ones who're supposed to know. |
| 18:47.10 | ManxPower | unbkbl: that was not a joke. |
| 18:47.17 | *** join/#asterisk luke-jr (n=luke-jr@2002:440d:6de2:0:20e:a6ff:fec4:4e5d) |
| 18:47.28 | luke-jr | Is it possible to get Asterisk to connect a call directly to a Zaptel line? |
| 18:47.29 | BrokenNoze | ManxPower: on voip-info? |
| 18:47.33 | luke-jr | eg, so they get a dialtone? |
| 18:48.00 | unbkbl | :( |
| 18:48.02 | unbkbl | ok |
| 18:48.07 | rob0 | "There must be ... fifty ways to leave your lover" |
| 18:49.40 | iamhrh | should members of a queue be receiving additional calls when they are already on one? |
| 18:50.04 | ManxPower | iamhrh: yes, if your system design and phones allow that |
| 18:50.23 | ManxPower | BrokenNoze: yes |
| 18:51.32 | *** join/#asterisk angryuser (n=Miranda@df01t2-212-195-203-22.d4.club-internet.fr) |
| 18:51.41 | angryuser | tzafrir here? |
| 18:51.53 | tzafrir | yes |
| 18:52.04 | iamhrh | ManxPower: is there a way to limit it to one at a time? I'd rather it pass up someone who is already busy, not quite sure where to look though. Right now calls to queue members are being routed through Local/member@queudialer (there is some logic involved in deciding what SIP extension to call for each member) |
| 18:52.29 | ManxPower | iamhrh: turn off call waiting on your phones |
| 18:53.50 | *** join/#asterisk gandhijee (n=root@host-66-202-34-165.spr.choiceone.net) |
| 18:53.56 | iamhrh | Manx: 10-4, will try that out |
| 18:55.58 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 18:56.15 | angryuser | here is my feedback about the problem described yesterday, when you have old tdm400p and new tdm410p and b410p in one system, and the driver of misdn taking ypur tdm400p card , firs you need to blacklist it at /etc/modprobe.d/blacklist "blacklist netjetpci" then at system start unload wctdm and wctdm24xxp(driver of tdm410p) and load them again to respect the load order of pci , tzafrir |
| 18:56.24 | BrokenNoze | ManxPoer : yes, and i have it working. though if i have multiple line keys ringing there's no way to prioritize the Auto Answer call. so it just has to wait until the other lines stop ringing, which defeats the idea of aout anwer in the first place |
| 18:57.34 | tzafrir | angryuser, the tdm410p (and maybe even some later 400-s) should have a PCI ID that does not trigger that that mISDN module |
| 18:58.06 | rob0 | "locate netjetpci" comes up empty here. |
| 18:59.31 | angryuser | exactly, but in my case i had ald configs from prevous instal with zaptel channels defined, and only tdm410p was detected on boot, so it mesed my my zap order |
| 19:03.05 | *** join/#asterisk killmel8tr (n=IceChat7@ip-198-22-65-131.quickconnectusa.com) |
| 19:03.27 | *** join/#asterisk shido6 (n=shido6@74-130-51-159.dhcp.insightbb.com) |
| 19:03.54 | *** part/#asterisk ds2 (i=noinf@netblock-66-245-251-24.dslextreme.com) |
| 19:05.32 | rob0 | Um, I guess my point is that this "blacklist netjetpci" advice might be specific to your distro. My kernel.org kernel doesn't have that driver. |
| 19:05.34 | ManxPower | BrokenNoze: only if you were silly enough to use one registration for all line keys |
| 19:06.38 | angryuser | rob0: install misdn you will have it ;) |
| 19:07.06 | angryuser | rob0:and my distro is debian |
| 19:07.11 | mwalling | ~gs |
| 19:07.12 | jbot | GrandSuck phones & gateways are cheap junk which should be avoided with extreme prejudice. |
| 19:08.15 | ManxPower | rolnd: I'm not helping you anymore. You either don't have the time or the desire to spend the required time on this problem. I'm going back to paid work |
| 19:09.02 | keith4 | angryuser: you could write udev rules to enforce the load order |
| 19:09.38 | *** join/#asterisk angom (n=angom@201.170.65.143) |
| 19:09.49 | angryuser | keith4: yes , if i was not a noob in that matter |
| 19:10.09 | keith4 | yah, it's a real pain |
| 19:10.26 | keith4 | i had a similar problem with tuner cards in my myth backend |
| 19:11.50 | keith4 | fwiw, i don't have netjetpci (in debian) modules, but I do see "netjet" under "hisax" |
| 19:13.48 | *** join/#asterisk keulin (n=cray@247.126.64-86.rev.gaoland.net) |
| 19:16.17 | mintee | is there a way to give temp control to a context and return to the original context depending on some variables? |
| 19:16.32 | mintee | without specifically calling the context on the return |
| 19:17.49 | [TK]D-Fender | mintee: "core show application gosub" |
| 19:18.17 | mintee | cool, thanks |
| 19:19.19 | *** join/#asterisk lanning (n=lanning@ip67-152-85-190.z85-152-67.customer.algx.net) |
| 19:20.03 | *** join/#asterisk jsolis (n=jimmy@190.41.82.1) |
| 19:20.32 | *** join/#asterisk RoyK (n=roy@ti211310a080-7540.bb.online.no) |
| 19:20.34 | *** join/#asterisk b1shop (n=b1shop@dsl081-149-253.chi1.dsl.speakeasy.net) |
| 19:27.50 | *** join/#asterisk jetlagmk2 (i=jetlag@pool-70-17-48-245.pskn.east.verizon.net) |
| 19:34.07 | *** join/#asterisk gr0mit (n=tim@extrt.txrx.org.uk) |
| 19:36.39 | *** join/#asterisk Avero (n=Avero@216.186.253.120) |
| 19:38.42 | rolnd | anyone knows how to reset password on azatel ipcall104 |
| 19:40.19 | RoyK | hands rolnd a sledgehammer |
| 19:40.29 | rolnd | heh |
| 19:40.46 | ManxPower | RoyK: he ignores questions, does not provide the requested info, leaves in the middle of troubleshooting. Not worth your time. |
| 19:40.49 | mintee | is there a way to set a variable outside of an extension? IE; Without the SET() function? |
| 19:41.01 | ManxPower | mintee: only global variables |
| 19:41.07 | mintee | hum |
| 19:41.12 | outtolunc | or the manager interface |
| 19:41.24 | ManxPower | you could also it in sip.conf if you need to. |
| 19:41.30 | ManxPower | I think in 1.6 Zap also allows that. |
| 19:42.29 | RoyK | what's really the difference between zap 1.2/1.4/1.6? |
| 19:42.41 | ManxPower | RoyK: heck if I know. |
| 19:42.51 | ManxPower | I guess if I really wanted to know I'd read the changelog |
| 19:43.16 | RoyK | seems to me the changes are minor, but they've pushed the version up along with asstrix |
| 19:43.27 | *** join/#asterisk ryant (n=ryant@c-98-223-72-69.hsd1.in.comcast.net) |
| 19:48.54 | ManxPower | RoyK: 1.6 has "SS7 support in chan_zap (via libss7 library)" |
| 19:49.29 | ManxPower | setvar support for zapata.conf, "auto" mode for analog cards to autodetect port type |
| 19:49.57 | tzafrir | and support for bri |
| 19:50.32 | tzafrir | not complete, but quite good enough for many |
| 19:50.59 | *** join/#asterisk lotho (n=lotho@static.69.46.46.78.clients.your-server.de) |
| 19:51.22 | tzafrir | but that is Asterisk's chan_zap . zaptel is still 1.4 |
| 19:51.32 | tzafrir | and not going to be 1.6 |
| 19:52.17 | *** join/#asterisk gego (n=gego@dyndsl-091-096-099-132.ewe-ip-backbone.de) |
| 19:53.50 | *** join/#asterisk gbr_ (n=gbr@200.103.96.98) |
| 19:54.04 | x86 | when is 1.6 going to be ready for production? |
| 19:54.24 | tzafrir | x86, depends who you'll ask |
| 19:54.39 | tzafrir | for some I think 1.4 will be ready in a year or two |
| 19:54.44 | *** join/#asterisk RoyK (n=roy@ti211310a080-7540.bb.online.no) |
| 19:54.44 | x86 | been almost a month now since the latest Asterisk release was made, which seems like a long time looking back on previous releases |
| 19:55.00 | ManxPower | I don't 1.4 will ever be ready for any of my production servers |
| 19:56.33 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 19:57.22 | RoyK | ManxPower: in these days, most telcos support SIP termination and even so with SIP-T - are there any SIP-T support in the pipeline? |
| 19:57.55 | ManxPower | RoyK: I do not work for Digium and am not a developer. |
| 19:58.16 | ManxPower | I come from the system admin / corporate enviroment |
| 19:59.26 | RoyK | ManxPower: I just wondered if you knew |
| 20:00.38 | clive- | how stable is 1.4 in a production environment? |
| 20:00.54 | keith4 | "most telcos support SIP termination"? that's an awfully broad generalization |
| 20:02.41 | *** part/#asterisk nny_1 (n=scott@64.203.239.83) |
| 20:03.04 | *** join/#asterisk nny_1 (n=scott@64.203.239.83) |
| 20:05.12 | nny_1 | does anyone know of a good explanation of how to set up the callerid to strip the channel and server IP? right now my incoming context doesn't have any Set(CALLERID(num) etc in it for incoming, which I suspect is the larger issue. |
| 20:08.27 | *** join/#asterisk ice_croft (n=nolan@213.132.86.246) |
| 20:08.31 | ice_croft | hi ppl |
| 20:08.41 | ice_croft | need urgent help :( |
| 20:08.53 | nny_1 | ice_croft: I can try to help whats up |
| 20:09.01 | draygon | ask your question |
| 20:09.34 | ice_croft | i updated planet-156 firmware, and whan im callin with ulaw codec -- it reports SIP 488 error |
| 20:09.38 | b11d` | does anyone know of a wireless headset that works with a polycom 501/550 that DOESNT require a Lifter? |
| 20:09.41 | b11d` | i hate lifters! |
| 20:09.47 | [TK]D-Fender | nny_1: You are still looking at the fromt he wrong angle. NoOp it. Do you SEE the IP? You definitely shouldn't. |
| 20:10.05 | ice_croft | if i set vip primary codec to alaw, it calls well |
| 20:10.06 | [TK]D-Fender | b11d`: Jabra's got some. |
| 20:10.11 | b11d` | thanks TK |
| 20:10.18 | ice_croft | where to dig? |
| 20:10.21 | nny_1 | [TK]D-Fender: sorry i wasn't sure what step of the dialplan to NoOp |
| 20:10.38 | [TK]D-Fender | nny_1: How about at a point where you CARE about it? |
| 20:11.14 | [TK]D-Fender | ice_croft: 488 = codec mismatch |
| 20:11.38 | ice_croft | [TK]D-Fender> its set to ulaw both sides :( |
| 20:11.52 | *** join/#asterisk dlynes_laptop (n=chatzill@dsl-vlan468-66-18-244-66.nucleus.com) |
| 20:11.52 | [TK]D-Fender | ice_croft: And where's our pastebin? |
| 20:12.01 | ice_croft | omg, wait a min |
| 20:12.04 | clive- | manxpower why dont you use 1.4 in production? |
| 20:12.39 | *** join/#asterisk dr_gogeta86 (n=gogeta@ppp-232-249.32-151.iol.it) |
| 20:12.46 | dlynes_laptop | I'm having an issue getting mediatrix to call mediatrix, aastra to call mediatrix, anything to call mediatrix....I've got a sip debug log of the transaction I was wondering if anyone could take a look at to determine why it's not happening? |
| 20:12.51 | nny_1 | [TK]D-Fender: I dont see the IP there.. and I just realized our 962 doesn't do it.. so I am guessing this is phone specific |
| 20:12.51 | dlynes_laptop | The log is at http://pastebin.ca/1016163 |
| 20:13.13 | dlynes_laptop | clive-: probably because 1.4 isn't rock solid stable |
| 20:13.14 | [TK]D-Fender | nny_1: Good... no go read up its support docs & links. |
| 20:13.16 | [TK]D-Fender | now* |
| 20:13.42 | dlynes_laptop | clive-: if I didn't need the features in 1.4, I wouldn't be using it, either |
| 20:14.01 | x86 | dlynes_laptop: I hear ya.... 1.2 was _much_ more stable |
| 20:14.24 | nny_1 | [TK]D-Fender: roger that |
| 20:14.35 | x86 | dlynes_laptop: but I also need the features of 1.4, and I'm waiting for a couple new features in 1.6 also |
| 20:14.41 | [TK]D-Fender | dlynes_laptop: You're not getting answered. either IP is wrong, firewall/networking is wrong, etc |
| 20:14.45 | dlynes_laptop | x86: yeah..unfortunately, i've got a boss that __must__ have blf in all possible configurations, and shared line appearance |
| 20:14.55 | clive- | dlynes is it so unstable? |
| 20:15.11 | dlynes_laptop | clive-: yes...but it depends on what features you use in it, too |
| 20:15.50 | dlynes_laptop | [TK]D-Fender: could it be a mismatch in timings for the codecs by any chance, instead? |
| 20:15.53 | x86 | dlynes_laptop: isnt BLF the same thing as SLA? |
| 20:15.55 | clive- | dlynes i am basically just using it for iax2 to sip conversionsis it so unstable? |
| 20:16.02 | dlynes_laptop | [TK]D-Fender: i.e. it using 100ms timings instead of 30ms timings? |
| 20:16.09 | ice_croft | [TK]D-Fender> http://pastebin.ca/1016175 |
| 20:16.15 | [TK]D-Fender | dlynes_laptop: No... you are getting NO response whatsoever in there |
| 20:16.18 | dlynes_laptop | x86: no...SLA needs BLF to work |
| 20:16.27 | ice_croft | [TK]D-Fender> http://pastebin.ca/1016177 |
| 20:16.33 | x86 | dlynes_laptop: not seeing how they are different |
| 20:16.39 | dlynes_laptop | [TK]D-Fender: oh, great...so basically I might have a couple of bricked mediatrix boxes |
| 20:17.13 | clive- | dlynes maybe i should top and go back to 1.2 ... 1.4 is supposed to be more cpu efficient ad fit more calls in the same hardware |
| 20:17.16 | dlynes_laptop | [TK]D-Fender: do you know of a good tutorial for being able to read sip debug logs by any chance? |
| 20:17.37 | ice_croft | damn |
| 20:17.48 | [TK]D-Fender | ice_croft: Capabilities: us - 0x4 (ulaw), peer - audio=0x8 (alaw)/video=0x0 (nothing), combined - (ulaw) <- mismatch of "us" vs "them" |
| 20:17.51 | ice_croft | Capabilities: us - 0x4 (ulaw), peer - audio=0x8 (alaw)/video=0x0 (nothing), combined - (ulaw) |
| 20:17.51 | ice_croft | Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) |
| 20:17.56 | ice_croft | i saw that |
| 20:18.01 | ice_croft | sorry Fender |
| 20:18.07 | ice_croft | gotta hit the wall |
| 20:18.10 | [TK]D-Fender | dlynes_laptop: Here's a tip : No packets with the name "Mediatrix" in them ;) |
| 20:18.48 | dlynes_laptop | [TK]D-Fender: yeah..I seen that...but I figured I wouldn't get any until the rtp finished negotiating or something |
| 20:19.34 | dlynes_laptop | [TK]D-Fender: and it's kinda weird because I can send calls to a mediatrix, but I can't make any calls out from it |
| 20:20.24 | dlynes_laptop | [TK]D-Fender: that log you see is from the mediatrix trying to make a call to an aastra |
| 20:20.45 | [TK]D-Fender | dlynes_laptop: RTP? ixnay <- |
| 20:20.47 | dlynes_laptop | [TK]D-Fender: so it's starting the audio stream, but never sends any sip messages I guess? |
| 20:20.57 | [TK]D-Fender | dlynes_laptop: You have no response To >>SIP<< trying to SET UP the call. |
| 20:21.13 | [TK]D-Fender | dlynes_laptop: Stop counting your chickens so damn early. |
| 20:21.19 | *** join/#asterisk s0lid (n=s0lid@210.213.199.2) |
| 20:22.41 | tzanger | brawwwwwwwk bawk bawk bawk |
| 20:23.29 | b11d` | TK.. do you have a reputable vendor for Jabra headsets? |
| 20:25.34 | *** join/#asterisk RobH (n=RobH@cpe-72-184-129-136.tampabay.res.rr.com) |
| 20:25.59 | ice_croft | [TK]D-Fender> look, plz |
| 20:26.01 | ice_croft | [TK]D-Fender> Capabilities: us - 0x4 (ulaw), peer - audio=0x0 (nothing)/video=0x0 (nothing), combined - (ulaw) |
| 20:26.01 | ice_croft | Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) |
| 20:26.25 | *** join/#asterisk Braxus (n=braxus@netblock-68-183-228-155.dslextreme.com) |
| 20:26.47 | ice_croft | [TK]D-Fender> audio nothing, and its set to ulaw on the peer. should i just trash it? |
| 20:28.48 | [TK]D-Fender | ice_croft: you should get a clue and actually configure the codecs on each side properly. |
| 20:29.04 | *** join/#asterisk anonymouz666 (n=anonymou@201.19.80.140) |
| 20:29.38 | ice_croft | [TK]D-Fender> theres nothing advanced to configure on the fxs, actually.. |
| 20:29.47 | ice_croft | [TK]D-Fender> thanx man |
| 20:29.59 | [TK]D-Fender | ok, heading home, later all. |
| 20:31.51 | *** join/#asterisk mltlnx (n=mltlnx@71.4.175.198.ptr.us.xo.net) |
| 20:34.57 | *** join/#asterisk sysreq (n=sysreq@unaffiliated/sysreq) |
| 20:36.47 | killmel8tr | man people are so non-understanding, its like they want thier phones to work all the time and they dont understand that if everything "just worked" all the time people like us wouldnt have jobs... lol |
| 20:36.50 | *** join/#asterisk sharp (n=sharp@stereotheism.org) |
| 20:37.02 | sharp | does anybody know anything about ztxen and where i might find it? |
| 20:39.13 | x86 | ztxen? |
| 20:41.14 | sharp | http://bugs.digium.com/view.php?id=9592 |
| 20:41.25 | sharp | a ztdummy for xen vm's |
| 20:41.42 | sharp | thats the link i was looking for |
| 20:42.38 | *** join/#asterisk theHub (n=theHub@69.177.93.21) |
| 20:44.05 | x86 | ah |
| 20:44.12 | *** part/#asterisk iamhrh (n=iamhrh@office.amsvans.com) |
| 20:44.32 | x86 | I don't virtualize my mission critical phone systems, and I certainly have no need for ztdummy as I use real hardware |
| 20:45.42 | anonymouz666 | anyone in here using the patch (DTMF/FSK callerid detection) from mantis issue 9096? |
| 20:47.23 | *** join/#asterisk ice_croft (n=nolan@213.132.86.246) |
| 20:50.24 | *** join/#asterisk dlynes (n=chatzill@dsl-vlan468-66-18-244-66.nucleus.com) |
| 20:50.58 | *** join/#asterisk keulin (n=cray@247.126.64-86.rev.gaoland.net) |
| 20:55.27 | *** join/#asterisk aiksa[LV] (n=aiks@mx.fiveplus.lv) |
| 20:56.13 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 20:58.58 | Ritzerisk | or does anyone know of a type of asterisk system that can use the auto dialer |
| 21:00.22 | *** join/#asterisk Maliuta (n=nikolai@kiev.lusan.id.au) |
| 21:00.23 | aiksa[LV] | Ritzerisk: for an outbound callcenter type of auto dialer? |
| 21:00.42 | aiksa[LV] | or like an announcment system? |
| 21:04.41 | ice_croft | f#ckin planet |
| 21:06.06 | mintee | so from out of the box, should MusicOnHold() work? |
| 21:06.17 | *** part/#asterisk clive- (n=pirch@41.242.156.73) |
| 21:06.19 | mintee | cause I can't seem to get it working |
| 21:10.49 | *** join/#asterisk RoyK (n=roy@ip-177-22-149-91.dialup.ice.no) |
| 21:11.18 | dlynes | ice_croft: planet brand network equipment? |
| 21:11.45 | doke | please does anyone has a Cisco 7975 connected to Asterisk here? |
| 21:11.59 | doke | This phone is driving me crazy since a week |
| 21:12.10 | ice_croft | dlynes> yes |
| 21:12.18 | doke | if anyone happends to know well SIP I have a dump of the dialog |
| 21:12.18 | dlynes | ice_croft: complete crap isn't it? |
| 21:12.29 | ice_croft | dlynes> true, man |
| 21:12.52 | dlynes | ice_croft: i've been fighting with one of their switches lately, and the network cards are even worse |
| 21:13.19 | ice_croft | dlynes> thanx to God i never saw their nc's |
| 21:17.37 | mintee | [May 12 17:15:47] WARNING[9068]: mp3/interface.c:215 decodeMP3: Junk at the beginning of frame 49443303 |
| 21:17.38 | mintee | O_O |
| 21:20.12 | *** join/#asterisk shido6 (n=shido6@74-130-51-159.dhcp.insightbb.com) |
| 21:20.54 | muiro | two questions: is there a dialaplan equivalent to isnumeric() or similar -and- can I write my own functions ("applications") in dialplan without using macros? |
| 21:22.23 | mocker | Guh, was trying an invite to GrandCentral and I selected a number that needs a 1 in front of it from the same area code. |
| 21:22.32 | mocker | Now I need another invite because you can't switch numbers. :( |
| 21:23.41 | *** part/#asterisk mgroman (n=stansmit@adsl-68-252-62-180.dsl.wotnoh.ameritech.net) |
| 21:28.34 | *** join/#asterisk rsc-232 (n=mrdigita@65-78-101-58.c3-0.drf-ubr2.atw-drf.pa.cable.rcn.com) |
| 21:28.47 | *** join/#asterisk implicit (n=implicit@ip72-211-213-26.oc.oc.cox.net) |
| 21:30.13 | Katty | hmmph. |
| 21:32.04 | rob0 | I just set up this nifty failover for tollfree calls between sip.tollfreegateway.com and proxy01.sipphone.com, and all I got is this T-shirt. |
| 21:32.33 | ice_croft | lol |
| 21:32.34 | ice_croft | )) |
| 21:32.43 | rob0 | it was my greatest dialplan achievement ever, and sip.tollfreegateway.com is 503'ing me. |
| 21:32.47 | *** join/#asterisk Exstatica (i=Exstatic@freenode/staff/exstatica) |
| 21:33.21 | rob0 | the failover to Sipphone then works nicely |
| 21:34.17 | muiro | is it possible to write custom functions for asterisk? |
| 21:34.25 | draygon | of course. |
| 21:34.38 | Exstatica | anyhone know of a good windows osd sip client? |
| 21:34.42 | muiro | draygon: can you point me in the direction of documentation? |
| 21:35.25 | rob0 | Just wondering, is sip.tollfreegateway.com not a reasonable solution for tollfree termination? |
| 21:35.37 | rob0 | it worked when I tried it yesterday |
| 21:36.35 | *** join/#asterisk ice_croft (n=nolan@213.132.86.246) |
| 21:40.30 | *** join/#asterisk dlynes_laptop (n=chatzill@S01060016b68219f1.vs.shawcable.net) |
| 21:41.33 | *** join/#asterisk ice_croft (n=nolan@213.132.86.246) |
| 21:44.46 | *** join/#asterisk [TK]D-Fender (i=Joe@64.235.218.194) |
| 21:45.52 | ice_croft | so i switched it to alaw |
| 21:46.15 | ice_croft | it cant detect ulaw, damn crap |
| 21:46.45 | *** join/#asterisk AsteriskRo (n=rosadesa@190.36.187.144) |
| 21:46.57 | AsteriskRo | hello all |
| 21:47.01 | aiksa[LV] | Hi |
| 21:47.18 | AsteriskRo | please, any help on variables on asterisk dialplan¿? |
| 21:47.31 | muiro | AsteriskRo: what do you need help with? |
| 21:47.52 | [TK]D-Fender | AsteriskRo, Ask a specific question and you might get a specific answer. |
| 21:48.11 | AsteriskRo | i need to set the CALLERID with a variable that contain the trunk username |
| 21:48.39 | [TK]D-Fender | AsteriskRo, Set(CALLERID(name)=whateverthehellyouwant) |
| 21:48.53 | muiro | AsteriskRo: http://www.voip-info.org/wiki/view/Setting+Callerid |
| 21:49.08 | AsteriskRo | ok, but i need that the value is one of my trunks on users.conf username |
| 21:49.42 | AsteriskRo | for example Set(CALLERID(name)=trunk5/username |
| 21:50.09 | AsteriskRo | and trunk5 is defined on users.conf |
| 21:50.23 | *** join/#asterisk svf (n=svf@wsip-24-234-74-156.lv.lv.cox.net) |
| 21:51.13 | aiksa[LV] | hmm, and this is the user making the call |
| 21:51.24 | AsteriskRo | yeap |
| 21:51.39 | AsteriskRo | i need it to get permission to make the call |
| 21:51.51 | [TK]D-Fender | AsteriskRo, then do "SetVar=whatiwantforCID" in your users.conf entry and use that as the value to set the callerid to. |
| 21:51.52 | aiksa[LV] | why dont you just add callerid to the users.conf (or was that iax.conf and sip.conf specific funcionality)? |
| 21:52.16 | aiksa[LV] | [TK]D-Fender: thinking aprox. about the same |
| 21:52.41 | aiksa[LV] | AsteriskRo: or you could do another very ugly thing if number of users is limited |
| 21:52.41 | AsteriskRo | yes, but i'd like to use a variable instead a value |
| 21:52.46 | [TK]D-Fender | aiksa[LV], I'm just thinking he probably wants to treat "inside" calls different from "outside" |
| 21:52.56 | aiksa[LV] | [TK]D-Fender: looks like that |
| 21:52.59 | [TK]D-Fender | aiksa[LV], otherwise Yeah, I'd say set it direct as "callerid=" |
| 21:54.04 | *** join/#asterisk mltlnx (n=mltlnx@pool-96-232-207-89.nycmny.east.verizon.net) |
| 21:54.09 | aiksa[LV] | AsteriskRo: doesnt that "SetVar=whatiwantforCID" mean that you would have avriable/ |
| 21:54.17 | aiksa[LV] | would have variable |
| 21:54.42 | aiksa[LV] | you could do another very ugly thing |
| 21:54.45 | AsteriskRo | ok, but...which is that variable that has the value of my trunk username??? |
| 21:55.17 | [TK]D-Fender | AsteriskRo, well the "username" could be different from the [sectionname] in users.conf. |
| 21:55.44 | [TK]D-Fender | AsteriskRo, So its a questio of whether the [sectionname] is trustworthy. I didn't assume they were always the same |
| 21:56.05 | aiksa[LV] | sorry [TK]D-Fender Ill leave this to you. (off to bed, its too late here) |
| 21:56.12 | *** part/#asterisk codefreeze-lap (n=murf@216.166.159.235) |
| 21:56.14 | [TK]D-Fender | aiksa[LV], np |
| 21:56.22 | *** join/#asterisk jtknapp (n=skip@65-126-63-1.dia.static.qwest.net) |
| 21:56.26 | *** join/#asterisk codefreeze-lap (n=murf@216.166.159.235) |
| 21:56.54 | AsteriskRo | my sectionname is [trunk_5] |
| 21:57.12 | *** join/#asterisk bminish (n=bminish@rb.brenbox.home.minish.org) |
| 21:57.18 | [TK]D-Fender | AsteriskRo, and is that what you want to use as the "callerid"? |
| 21:57.23 | *** join/#asterisk mort___ (n=mort@user-3e8886cc.tcl115.dsl.pol.co.uk) |
| 21:58.09 | *** join/#asterisk colinm_ (n=colinm@VDSL-130-13-116-41.PHNX.QWEST.NET) |
| 21:58.12 | AsteriskRo | no, i want to use the username of that trunk |
| 21:58.55 | [TK]D-Fender | AsteriskRo, I'd suggest setting a variable to the same value in their entry then. |
| 21:59.30 | muiro | I think he means that when he connects to his trunk etc. he uses a username/password. When he's dialing through a trunk he wants it to show up as that username. I think. |
| 21:59.41 | muiro | maybe... |
| 22:00.17 | AsteriskRo | i'm a "she" not a "he" :) |
| 22:01.05 | AsteriskRo | yes, i need it to authentication/authorization |
| 22:01.31 | _ShrikE | ~seen kronos |
| 22:01.34 | jbot | kronos <n=kronos@85.204.66.113> was last seen on IRC in channel #kde, 303d 3h 47m 49s ago, saying: 'Sho_: any options in xorg.conf? that could be made to avoid maximize over 2 monitors?'. |
| 22:01.45 | AsteriskRo | if i don't use a trunk/username the number that tries to make the call is the extension number (i'm using it on a local pbx) |
| 22:02.02 | [TK]D-Fender | AsteriskRo, if you are doing this based on the device you are calling OUT of, and not the CALLING end (like I just described), then just set it right before you dial. |
| 22:03.00 | muiro | AsteriskRo: do you want the outgoing callerid to match the username for the trunk that was used to dial into your pbx? |
| 22:03.28 | muiro | AsteriskRo: or is it that each trunk that you might dial out of requires a username in the callerid? |
| 22:03.36 | AsteriskRo | yes, it has to mach with the sip trunk username to be autorized to make outbound calls |
| 22:04.07 | muiro | AsteriskRo: it's also going out of the sip trunk? |
| 22:04.29 | [TK]D-Fender | AsteriskRo, then jsut set it right before you dial. You already know the peer you're dialing out of so there doesn't have to be anything "variable" about it. |
| 22:06.12 | AsteriskRo | ok, now i have it set with the plain value before i dial, and it works fine...but i wanted to use a variable |
| 22:07.01 | [TK]D-Fender | AsteriskRo, No point. If you're doing this based on what you're going to dial out on, then that isn't something based on the caller, and * isn't psychic. So you're already doing it the way you have to and this entire exercise has been a waste of time. |
| 22:07.26 | AsteriskRo | sorry |
| 22:08.11 | AsteriskRo | i just wanted to have it on a variable, so if i have to make changes i don't have to change it on users.conf and extensions.conf |
| 22:08.21 | AsteriskRo | sorry to make you waste your time :$ |
| 22:08.47 | _ShrikE | ~seen krhonos |
| 22:08.49 | jbot | _ShrikE: i haven't seen 'krhonos' |
| 22:09.54 | [TK]D-Fender | AsteriskRo, wel DUH you have to change it in users.conf, and * STILL won't know you're about to dial out of them anyways |
| 22:16.22 | *** join/#asterisk RoyK (n=roy@ip-7-17-149-91.dialup.ice.no) |
| 22:18.51 | *** join/#asterisk anthm (n=anthm@mb10736d0.tmodns.net) |
| 22:20.05 | *** join/#asterisk unbkbl (n=work@static-adsl201-232-88-87.epm.net.co) |
| 22:20.52 | unbkbl | hello, i've installed freepbx but when i click in 'FreePBX administrator' link it shows a forbbiden webmessage, i know this is not the freepbx channel but there nobody give me an answer |
| 22:21.34 | [TK]D-Fender | unbkbl, Doesn't matter. Just because your auto-mechanic doesn't tell you how to fix a leading head gasket doesn't mean you should ask that in here either. |
| 22:21.45 | unbkbl | somebody know what could be the problem? |
| 22:21.48 | [TK]D-Fender | unbkbl, we do not support their scripts in here. |
| 22:22.21 | [TK]D-Fender | unbkbl, They have plenty of support message boards & their own channel. Use them. |
| 22:24.49 | unbkbl | ok thnx for nothing |
| 22:27.19 | rob0 | http://sweet.nodns4.us/ |
| 22:32.27 | *** part/#asterisk AsteriskRo (n=rosadesa@190.36.187.144) |
| 22:33.55 | *** join/#asterisk craigk (n=craigk@58.174.150.119) |
| 22:35.52 | mintee | O |
| 22:35.55 | mintee | err |
| 22:36.06 | mintee | I've setup a rather simple extension to test out transfers... |
| 22:36.41 | mintee | Basically call comes in, and rings my cell phone. When I answer the call, I can hit # and it will put the caller into a MusicOnHold while my end says "Transfer" |
| 22:37.19 | mintee | however, I can't determine how to transfer the call. Any thing I put it, it looks for that extension in my original Zap context. |
| 22:37.51 | ManxPower | mintee: it should look in whatever context the Dial happened in. Did you look at "core show application dial"? |
| 22:38.10 | ManxPower | also channelvariables.txt will contain some useful information for you. |
| 22:38.39 | mintee | Well, the dial happened in [transfer_me] |
| 22:39.06 | mintee | however, when I dial an extension, it;s looking for it in [from-pstn] my main zap channel |
| 22:39.08 | ManxPower | mintee: then transfer_me *should* be where it's looking for matching extension. Is that not happening? |
| 22:39.23 | ManxPower | mintee: My two suggestions stand. |
| 22:40.02 | ManxPower | mintee: put a Noop(CONTEXT=${CONTEXT} before the Dial line to make SURE it's coming from transfer_me |
| 22:40.14 | ManxPower | But remember the closing ) |
| 22:40.28 | *** part/#asterisk mog (n=mog@c-68-62-219-86.hsd1.al.comcast.net) |
| 22:40.28 | *** part/#asterisk BrokenNoze (i=BrokenNo@79-75-233-86.dynamic.dsl.as9105.com) |
| 22:40.32 | [TK]D-Fender | mintee, Set(TRANSFER_CONTEXT=where my zaptel channel should let me transfer based on. |
| 22:41.07 | ManxPower | mintee: you will learn that when you follow my suggestions 8-) |
| 22:45.04 | mintee | http://pastebin.ca/1016355 |
| 22:45.35 | *** join/#asterisk Strom_M (n=strom@208.127.172.112) |
| 22:45.43 | mintee | Line 10 |
| 22:46.22 | [TK]D-Fender | mintee, se the variable I told you to and read the docs. |
| 22:46.27 | mintee | yep yep |
| 22:46.41 | mintee | so that can't be a dynamic variable? |
| 22:47.04 | ManxPower | mintee: next time use the correct variable names |
| 22:47.30 | mintee | curses at voip-info.org |
| 22:49.26 | mintee | HAH! Awesome guys, thanks so much |
| 22:49.55 | mintee | setting the TRANSFER_CONTEXT to ${EXTEN} worked fine. |
| 22:50.09 | [TK]D-Fender | mintee, ... um.... |
| 22:50.13 | [TK]D-Fender | mintee, wtf? |
| 22:50.13 | mintee | seems that voip-info.org is more trouble than it's worth sometimes... |
| 22:50.23 | mintee | O_o |
| 22:50.27 | mintee | ? |
| 22:50.49 | [TK]D-Fender | mintee, since when is your EXTEN your target CONTEXT NAME? |
| 22:51.16 | [TK]D-Fender | mintee, -- Executing [s@transfer_me:8] Dial("Zap/1-1", "Zap/g1/215370xxxx|15|rt") in new stack <-- sure looks like "s" to me. |
| 22:51.17 | mintee | lol, err exten => s,n,Set(TRANSFER_CONTEXT=${CONTEXT}) |
| 22:51.46 | mintee | [TK]D-Fender, yeah, it's a s inside another context... |
| 22:51.47 | [TK]D-Fender | mintee, and what is ${CONTEXT} ? |
| 22:52.12 | mintee | specifically there it's called [transfer_me] |
| 22:52.56 | mintee | currently inside [transfer_me] i only have an s exten and a 1 |
| 22:53.22 | mintee | i was using the s to call my cell phone, then once answered hitting #1 to transfer me to 1 |
| 22:54.17 | [TK]D-Fender | mintee, ... nevermind. |
| 22:54.19 | mintee | it's just a test to get get it working... |
| 22:55.00 | mintee | i don't follow? Lemme know if I should be doing it another way... |
| 22:55.11 | *** part/#asterisk RoyK (n=roy@ip-7-17-149-91.dialup.ice.no) |
| 23:04.58 | mintee | goes home for the day. |
| 23:15.32 | *** join/#asterisk jm|laptop (n=jm|home@zen.jamiem.com) |
| 23:21.06 | *** join/#asterisk mindCrime (n=chatzill@cpe-065-190-188-124.nc.res.rr.com) |
| 23:22.17 | *** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 23:24.38 | *** join/#asterisk hmmhesays (n=hmmhesay@24-119-176-74.cpe.cableone.net) |
| 23:29.51 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 23:40.26 | *** join/#asterisk lowtone (n=tj@ip68-99-193-165.ph.ph.cox.net) |
| 23:40.44 | NovceGuru | So I don't think asterisk is the answer for when someone requests a system to be able to tell when someone is on the line? |
| 23:41.14 | NovceGuru | This guy is all setup with cisco 7940's and a hosted service and is complaining he can't tell if someone is on the line before xfering a call |
| 23:41.24 | rob0 | My job is on the line ... |
| 23:41.55 | _ShrikE | NovceGuru: thats incorrect, asterisk does support presence. |
| 23:42.55 | NovceGuru | So thats the word I needed to google |
| 23:56.20 | *** join/#asterisk outtolunc (n=me@c-67-170-211-86.hsd1.ca.comcast.net) |
