| 00:03.30 | *** part/#asterisk ManxPower (n=manxpowe@208.sub-70-222-74.myvzw.com) |
| 00:03.44 | MDK2MDK | what that mean ztdummy driver ?? |
| 00:04.06 | *** join/#asterisk ManxPower (n=manxpowe@208.sub-70-222-74.myvzw.com) |
| 00:04.23 | MDK2MDK | is there body here can help me plz ?????????????,,,, |
| 00:10.18 | JayTee52 | MDK2MDK, download and read this, then come back |
| 00:10.21 | JayTee52 | ~book |
| 00:10.22 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com, or see ~buybook |
| 00:10.47 | JayTee52 | the first link is the downloadable book. It has an index and in it there is a reference to ztdummy. |
| 00:11.24 | MDK2MDK | im reading it now bat some thins i dident understund them |
| 00:11.34 | MDK2MDK | i'm a new user of linux |
| 00:11.47 | mwalling | then consult your distro documentation |
| 00:12.01 | MDK2MDK | but i'm a good devlopper in java and .net langages |
| 00:12.29 | MDK2MDK | ok thx :) i'll try |
| 00:13.00 | ManxPower | MDK2MDK: To really set up Asterisk you need a good working knowledge of Linux, networking, telecom, NAT, SIP, and Asterisk |
| 00:13.54 | JayTee52 | and don't forget two things, Don't Panic and always bring a towel |
| 00:14.59 | MDK2MDK | ok :) |
| 00:15.19 | MDK2MDK | i'll try all of that tkx :) |
| 00:15.26 | Nasra | ManxPower: how much is good working knowledge for you ....I don't know much about ...and I almost ready to go for it.... |
| 00:15.27 | Nasra | thanks |
| 00:16.17 | *** join/#asterisk bronson (n=bronson@adsl-68-122-117-135.dsl.pltn13.pacbell.net) |
| 00:17.12 | *** join/#asterisk bronson (n=bronson@adsl-68-122-117-135.dsl.pltn13.pacbell.net) |
| 00:34.03 | jeev | is crazy |
| 00:35.23 | *** part/#asterisk voipman (i=ccrites@minibar.rackmount.org) |
| 00:39.44 | *** part/#asterisk xai (n=pasta@about/networking/0.0.0.0/xai) |
| 00:42.53 | *** join/#asterisk Great_Randew (n=Andrew@stjhnbsu84w-156034168181.nb.aliant.net) |
| 00:44.59 | *** join/#asterisk nighty^ (n=nighty@210.188.173.246) |
| 00:48.05 | *** join/#asterisk dlynes (n=dlynes@216.251.149.69) |
| 00:48.51 | *** join/#asterisk bsaxon (n=bsaxon@adsl-068-209-196-209.sip.bhm.bellsouth.net) |
| 00:48.58 | *** join/#asterisk jim4voice (n=chatzill@199.93.187.81.in-addr.arpa) |
| 00:49.19 | luke-jr | I love how Asterisk 1.4.19 adds new bugs -.- |
| 00:50.16 | _ShrikE | looks at luke-jr and agrees |
| 00:50.41 | *** join/#asterisk andresmujica (n=andresmu@190.25.102.22) |
| 00:51.08 | luke-jr | _ShrikE: wish I knew how it got past QA |
| 00:51.21 | dlynes | Did asterisk 1.4.14 or later fix a nasty bug in the voicemail in 1.4.13, where the thread would crash when retrieving certain voicemail messages? |
| 00:51.48 | dlynes | I've got a bunch of 60 byte voicemails that when the user goes to retrieve them, asterisk hangs up on them |
| 00:52.11 | JunK-Y | dlynes: can I close http://bugs.digium.com/view.php?id=11072 ? |
| 00:52.51 | jeev | hey guys, http://pastebin.com/m74192ede ... i can't seem to figure out call parking OR transfer via softphone. where would i put the Tt/tT in that set up? |
| 00:53.15 | dlynes | JunK-Y: I'm guessing it's been added to 1.6? |
| 00:53.23 | _ShrikE | jeev: If you are using a softphone. The Tt options should not be necessary. |
| 00:53.39 | jeev | _ShrikE, X-Lite does not have transfer :/ |
| 00:54.06 | _ShrikE | ahh.. you are correct |
| 00:54.26 | jeev | wish it did man.. but riht now, im planning on rnning everything on X-Lite, then 2 months later, changing to IP Phones. |
| 00:54.32 | JunK-Y | dlynes: yes, but I also wrote a patch for 1.4 |
| 00:55.01 | dlynes | JunK-Y: Ok, cool |
| 00:55.08 | dlynes | JunK-Y: go ahead and close it, then...thanks |
| 00:56.09 | jeev | _ShrikE, anything? :/ |
| 00:56.09 | JunK-Y | i will see if I could create a backport for 1.4 |
| 00:56.27 | *** join/#asterisk bsaxon (n=bsaxon@adsl-068-209-196-209.sip.bhm.bellsouth.net) |
| 00:56.59 | dlynes | luke-jr: what new bugs did asterisk 1.4.19 add? |
| 00:57.08 | luke-jr | http://bugs.digium.com/view.php?id=12427 at least |
| 00:58.08 | JunK-Y | luke-jr: is that 1.4.19-rc3? |
| 00:58.19 | luke-jr | 1.4.19 final |
| 00:58.22 | *** join/#asterisk NirS (n=NirS@87.68.3.201.cable.012.net.il) |
| 00:59.06 | JunK-Y | high volume? |
| 00:59.12 | luke-jr | ? |
| 00:59.21 | JunK-Y | do you have a really high volume? |
| 01:00.29 | Qwell | what, no debug logs? |
| 01:00.36 | Qwell | useful bug report there |
| 01:00.37 | luke-jr | JunK-Y: whatever defaults are? |
| 01:00.49 | *** join/#asterisk korihor (n=humberto@190.74.120.245) |
| 01:01.16 | JunK-Y | there's default for volume? i didnt know that. |
| 01:01.28 | Qwell | default is 0 |
| 01:01.37 | jeev | hey guys, http://pastebin.com/m74192ede ... i can't seem to figure out call parking OR transfer via softphone. where would i put the Tt/tT in that set up? i'm using X-lite softphone and transfer isn't enabled. |
| 01:01.50 | drmessano | If anyone is interested Openfire enterprise is now open source.. so some of the cool asterisk functionality is now free |
| 01:01.56 | JunK-Y | Qwell: i totally agree that bug means _nothing_, since theres not infos at all. |
| 01:02.06 | Qwell | drmessano: eh? |
| 01:02.14 | *** join/#asterisk d00gster (n=doughant@bas1-cooksville01-1279552398.dsl.bell.ca) |
| 01:02.26 | drmessano | Openfire XMPP server.. |
| 01:02.27 | Qwell | oh |
| 01:02.30 | Qwell | neat |
| 01:02.32 | drmessano | Yeah |
| 01:04.25 | plik | erk... what could suddenly cause outgoing calls to fail with "CHANUNAVAIL" ? |
| 01:04.48 | Qwell | plik: the channel being unavailable |
| 01:05.22 | Kyoshi | heheh |
| 01:05.24 | NirS | is going to bed, it's 4am aleady |
| 01:05.34 | Kyoshi | could be lack of network connectivity |
| 01:05.37 | NirS | is going to bed, it's 4am already <- zZzZzZzZzZzZzZzZzZzZzZzZzZzZ |
| 01:05.45 | Kyoshi | could be the sip host could be rejecting you |
| 01:05.46 | plik | I kinda guessed that, but no idea why it would suddently appear, no changes to outgoing calls section of dialplan |
| 01:06.15 | plik | hmm... internet works, incoming calls work |
| 01:06.25 | Kyoshi | dialplan has nothing to do with it |
| 01:11.05 | *** join/#asterisk FireMac (n=firemac@CPE000d88ae88b9-CM0011ae8bb0ee.cpe.net.cable.rogers.com) |
| 01:27.34 | *** join/#asterisk nhuismanwork (n=nhuisman@dhcp79.IfA.Hawaii.Edu) |
| 01:28.17 | nhuismanwork | I have some cisco 7940/60's that recently started displaing an incorrect time. They were correct up till maybe a week ago or so. Now they are all 1 hour ahead. |
| 01:28.19 | nhuismanwork | any ideas? |
| 01:28.45 | mwalling | DST-fu? |
| 01:28.45 | nhuismanwork | The time on the asterisk server is correct |
| 01:28.58 | nhuismanwork | my first thought was DST |
| 01:29.03 | nhuismanwork | but i'm in hawaii and we don't have DST |
| 01:29.21 | mwalling | do your phones know that? |
| 01:29.40 | nhuismanwork | I would think they should know that HST is not affected by DST |
| 01:31.15 | nhuismanwork | I don't have any dstoffset stuff set in the SIPDefault.cnf |
| 01:32.17 | mwalling | shrugs.... it was a hunch |
| 01:32.25 | nhuismanwork | you could be right |
| 01:32.28 | nhuismanwork | i'm turning off dst_auto_adjust: 1 |
| 01:32.32 | nhuismanwork | maybe it's just retarded |
| 01:32.40 | nhuismanwork | I THINK that should turn off dst |
| 01:33.50 | *** join/#asterisk m4sk4r4 (n=m4sk4r4@189.13.73.82) |
| 01:44.58 | *** join/#asterisk DarkRift (i=dark@bas10-montreal02-1177584618.dsl.bell.ca) |
| 01:46.02 | profounded | does anyone know of any good sip providers for calling the UK and Canada? |
| 01:46.22 | JunK-Y | for canada, unlimitel is good. |
| 01:47.07 | profounded | oh really??? what r the rates like? ill go there now |
| 01:47.10 | plik | profounded: check out voiptel.org |
| 01:47.14 | profounded | thanks! |
| 01:47.39 | plik | profounded: sorry, that should be voiptalk.org |
| 01:47.47 | profounded | voiptel? boy i must have hit lottery 2nite! |
| 01:47.57 | profounded | ok tks! |
| 01:49.47 | plik | voiptalk, voiptalk :) |
| 01:52.02 | ManxPower | The time on the Asterisk server has nothing to do with the Cisco phones if they are running SIP |
| 01:56.33 | *** join/#asterisk m4sk4r4 (n=m4sk4r4@189.13.73.82) |
| 02:05.39 | ManxPower | The cisco phones get their time (in UTC) from their NTP server. Any local timezone stuff is done on the phone, moving the displayed time forward or back the correct number of hours |
| 02:10.29 | *** join/#asterisk FireMac (n=firemac@CPE000d88ae88b9-CM0011ae8bb0ee.cpe.net.cable.rogers.com) |
| 02:13.27 | *** part/#asterisk comprookie2000 (n=david@adsl-065-012-210-216.sip.bct.bellsouth.net) |
| 02:14.13 | *** join/#asterisk RypPn (i=TuMbL@rosscom.demon.co.uk) |
| 02:15.14 | luke-jr | Does IAX2 use RTP? |
| 02:15.17 | luke-jr | at all? |
| 02:15.30 | file | no. |
| 02:19.58 | luke-jr | the diff between 1.4.18.1 and 1.4.19 is too huge â¹ |
| 02:21.14 | JunK-Y | luke-jr: the equivalence of the RTP in IAX2 is the mini-frame. |
| 02:21.40 | luke-jr | but that wouldn't use rtp.c code, would it? |
| 02:22.01 | JunK-Y | no, like file just said. |
| 02:22.15 | luke-jr | yeah, so the marker bit change is unrelated I guess |
| 02:22.21 | luke-jr | which leave me no traces â¹ |
| 02:22.51 | JunK-Y | u need to take a trace of the iax2 debug |
| 02:23.18 | luke-jr | even though the problem is independent of IAX2? |
| 02:23.30 | luke-jr | (occurs in both SIP and IAX2 calls) |
| 02:27.22 | JunK-Y | u need to trace a bit luke-jr, i really suspect theres a bug in both drivers, otherwise a LOT of ppl would report that same bug. |
| 02:27.46 | luke-jr | actually, I think I might be wrong |
| 02:27.51 | luke-jr | it might be SIP only |
| 02:28.04 | luke-jr | i'm not 100% sure those IAX2 calls were actually IAX2 |
| 02:28.30 | luke-jr | is there a way to disable the marker bit thing in 1.4.19? |
| 02:30.50 | UnixDog | luke-jr: come over to the dark side |
| 02:31.33 | file | you can comment out the code, recompile, and see if that fixes it |
| 02:31.55 | luke-jr | file: then I need to setup a dev environment |
| 02:35.58 | luke-jr | anyone know what revs 1.4.18.1 and 1.4.19 are from in branches/1.4? |
| 02:36.34 | file | you don't need to know. |
| 02:36.38 | file | tags/1.4.18.1 and tags/1.4.19 |
| 02:37.41 | JunK-Y | file: seriously, that question is asked many many time and I also think that could help ppl, when they apply patches here and there. |
| 02:37.51 | file | what question? |
| 02:38.21 | JunK-Y | 1.4.x is based on which specific Revision. |
| 02:38.33 | file | svn info http://svn.digium.com/svn/asterisk/tags/<version> will tell you the revision, or just check out the tag directly and get that version |
| 02:38.34 | luke-jr | file: easier to narrow the bug to a specific rev |
| 02:38.54 | luke-jr | but that requires working with the branch |
| 02:39.09 | *** join/#asterisk flynux (n=flynux@2a01:38:0:0:0:0:0:1) |
| 02:40.01 | JunK-Y | but getting a specific tag, you have then to do lot of diff for some users. |
| 02:40.51 | file | so do what I said and get the revision |
| 02:41.19 | luke-jr | which is? |
| 02:41.36 | file | Last Changed Rev: 112286 |
| 02:43.25 | luke-jr | Last Changed != Tagged/Copied |
| 02:43.27 | *** join/#asterisk icebourg (n=abourg@c-76-25-92-95.hsd1.co.comcast.net) |
| 02:43.38 | *** join/#asterisk Nasra (n=Nasra@CPE001217b1920e-CM00111ade9528.cpe.net.cable.rogers.com) |
| 02:44.42 | file | blinks |
| 02:46.34 | luke-jr | hm, that's interesting⦠PBX decided to randomly reboot |
| 02:47.06 | *** join/#asterisk fnordus (n=dnall@24.84.160.227) |
| 02:50.36 | jbeez | what kind of pbx |
| 02:55.13 | luke-jr | jbeez: eh, AMD Duron |
| 02:55.58 | jbeez | your asterisk server? |
| 02:56.07 | luke-jr | yeah |
| 02:59.45 | JayTee52 | asterisk on a Duron? wow! "My name's Forrest, Forrest Gump" |
| 03:01.59 | *** join/#asterisk LakeSolon (n=blake@12-202-198-20.client.mchsi.com) |
| 03:03.05 | luke-jr | ⦠|
| 03:03.19 | luke-jr | the Duron replaced a K6-2 |
| 03:08.43 | JayTee52 | had two Durons at work, they were never reliable and always ran extremely hot. |
| 03:09.59 | JayTee52 | I've had * running on a P3 1ghz that ran way better than a 1.4ghz Duron. |
| 03:10.12 | luke-jr | shrug |
| 03:10.16 | luke-jr | this one does its job well |
| 03:10.16 | luke-jr | usually |
| 03:12.09 | drmessano | Hmm |
| 03:13.39 | luke-jr | I'm scared to wonder what that reboot issue was |
| 03:13.49 | luke-jr | I could clearly hear the hard drive reset from across the room |
| 03:13.55 | luke-jr | before the bootup beep |
| 03:14.28 | luke-jr | and I recall Linux successfully reiniting IDE a few days ago :/ |
| 03:15.44 | JayTee52 | all this bargain basement hardware is gonna give * a bad name |
| 03:16.59 | luke-jr | meh, * gives me more problems than the hardware |
| 03:17.31 | JayTee52 | how do you know it's really * and not the hardware causing the problems? |
| 03:17.55 | luke-jr | because downgrading * fixes it |
| 03:18.08 | JayTee52 | downgrading from what to what? |
| 03:18.21 | luke-jr | 1.4.19 to 1.4.18.1 |
| 03:18.25 | JayTee52 | ah |
| 03:19.07 | JayTee52 | I'm running 1.4.18.1 on CentOS 5.1 just fine and 1.4.11 on Red Hat EL5 64bit just fine |
| 03:19.33 | luke-jr | ew, redhats :þ |
| 03:19.52 | JayTee52 | except for a little bit of jitter on my damn cheap POS Grandstreams |
| 03:20.12 | JayTee52 | what are you running for linux? |
| 03:20.16 | luke-jr | Gentoo |
| 03:20.25 | JayTee52 | pfffft |
| 03:21.13 | JayTee52 | we have several at work, they're really fast swimmers but I like the King penguins better |
| 03:21.26 | luke-jr | haha |
| 03:21.37 | *** join/#asterisk esaym (n=user@cpe-70-120-89-6.satx.res.rr.com) |
| 03:21.37 | JayTee52 | seriously! I have live penguins where I work |
| 03:22.25 | JayTee52 | and lions, tigers, dolphins, elephants, lemurs, rhinos, giraffes, baboons and soon koalas. |
| 03:22.46 | *** join/#asterisk lesouvage (n=lesouvag@62.140.137.125) |
| 03:23.26 | luke-jr | 103000-106635 |
| 03:23.34 | luke-jr | somewhere in those revs is the bug |
| 03:23.54 | luke-jr | I wish * compiled faster |
| 03:24.48 | JayTee52 | so are you running on Gentoo so you can optimize the kernel or just because you like Gentoo and are used to it? |
| 03:25.53 | JayTee52 | my 64bit build compiled on a Quad Core Xeon in less than 2 minutes. |
| 03:26.52 | luke-jr | JayTee52: because it lets me run testing/unstable apps on a stable OS |
| 03:27.14 | luke-jr | something not even Debian offers me |
| 03:27.24 | JayTee52 | it certainly lends itself well to kernel optimizing |
| 03:27.36 | luke-jr | Gentoo does nothing for kernel optimizing, actually |
| 03:28.05 | JayTee52 | I was thinking it would be nice to build a fine tuned low-latency kernel in it specifically for * on older CPUs like a P3 or early P4 |
| 03:28.06 | luke-jr | in that regard it is the same as RH/Debian |
| 03:28.21 | luke-jr | you can build a custom kernel for them just as easily as for Gentoo |
| 03:28.53 | JayTee52 | I've done it on Ubuntu but all the tweak freaks seem to like Gentoo the best. |
| 03:29.59 | *** join/#asterisk blq (n=Bl@dslb-088-064-146-089.pools.arcor-ip.net) |
| 03:30.23 | *** join/#asterisk blq (n=Bl@dslb-088-064-146-089.pools.arcor-ip.net) |
| 03:30.26 | *** join/#asterisk BeeBuu (n=beebuu@219.132.188.242) |
| 03:30.32 | BeeBuu | hi,all |
| 03:31.20 | BeeBuu | is this work? while($[ 1 = 1] and $[ 2 =2 ]) |
| 03:33.16 | luke-jr | JayTee52: that's because Gentoo automates tweaking everything EXCEPT the kernel |
| 03:33.56 | JayTee52 | ok, well it's late so I'm off. Nite everyone |
| 03:35.40 | *** join/#asterisk _charly_ (i=kroseneg@sunrise.schmidham.net) |
| 03:48.19 | *** join/#asterisk stkn_ (n=stkn@gentoo/developer/pdpc.active.stkn) |
| 03:50.07 | *** join/#asterisk L|NUX (n=linux@unaffiliated/lnux/x-10290) |
| 03:51.46 | *** join/#asterisk s0lid (n=s0lid@210.213.242.60) |
| 03:57.06 | profounded | Has anyone have any experience with the sip provider phonosip? |
| 03:57.26 | profounded | voice quality? performnace... etc???? |
| 04:00.39 | ManxPower | ~itsp |
| 04:00.39 | 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 $/min. Enter ~itsplist-us (USA) or ~itsplist-ca (Canada) for a listing of popular ITSPs |
| 04:00.49 | ManxPower | All ITSPs suck. Some, however, suck more than others. |
| 04:02.20 | ManxPower | BeeBuu: I would have to try it, but I suspect it would have to be more like while($[$[1 = 1] & $[2 = 3]]) |
| 04:02.34 | ManxPower | You should read /path/to/src/asterisk/docs/channelvariables.txt |
| 04:02.46 | ManxPower | (might be "doc" instead of "docs" |
| 04:04.04 | profounded | hey thanks for the feedback |
| 04:04.25 | profounded | its like a shot in the dark for me. |
| 04:04.31 | *** join/#asterisk paci`` (n=paci@cpe-075-182-072-065.nc.res.rr.com) |
| 04:04.32 | paci`` | [Apr 12 04:02:45] NOTICE[98732]: chan_sip.c:17677 handle_request_register: Registration from '<sip:win@rmyou.org>' failed for '72.65.73.23' - No matching peer found |
| 04:04.37 | ManxPower | I recommend Vitelity and Teliax |
| 04:04.40 | paci`` | I have it in sip.conf |
| 04:04.43 | *** join/#asterisk Kage` (n=Kage@pool-72-65-73-23.clrk.east.verizon.net) |
| 04:04.53 | paci`` | See any problem that is commenly overlooked? |
| 04:05.12 | luke-jr | I recommend Voipjet! |
| 04:05.14 | ManxPower | I suspect the device is registering as [72.65.73.23] |
| 04:05.28 | luke-jr | whatever you do, avoid Sellvoip |
| 04:05.38 | paci`` | ManxPower, I don't think so |
| 04:05.43 | paci`` | I set the username as win |
| 04:05.50 | paci`` | though |
| 04:05.52 | paci`` | that does look backwards |
| 04:06.18 | paci`` | ManxPower, http://pastebin.com/m4c62641b |
| 04:06.39 | ManxPower | paci``: I'm about 3 beers past being to help you. 8-) |
| 04:06.46 | paci`` | lol |
| 04:07.48 | paci`` | nope |
| 04:09.00 | paci`` | anyone know? |
| 04:11.39 | *** join/#asterisk mwalling_ (i=mwalling@you.dontlike.us) |
| 04:22.10 | *** join/#asterisk davevg-btwtech (n=davevg-b@nj-67-76-177-147.sta.embarqhsd.net) |
| 04:44.18 | *** join/#asterisk cmantito (n=gphreak@pool-71-188-82-138.cmdnnj.east.verizon.net) |
| 04:58.36 | *** join/#asterisk PepOSX (n=angeldav@200.93.28.168) |
| 05:14.11 | jeev | hey guys, http://pastebin.com/m74192ede ... i can't seem to figure out call parking OR transfer via softphone. where would i put the Tt/tT in that set up? i'm using X-lite softphone and transfer isn't enabled. |
| 05:20.49 | paci`` | jeev, its due to too many nigawatts being inputted |
| 05:21.06 | jeev | 0_o |
| 05:25.31 | paci`` | je |
| 05:25.32 | paci`` | jeev, |
| 05:25.34 | paci`` | i told you dude |
| 05:25.38 | paci`` | those niggabyte hdd's |
| 05:25.43 | paci`` | they have capacity |
| 05:25.47 | paci`` | but you can hardly ever get them to wor |
| 05:25.48 | paci`` | +k |
| 05:26.17 | jeev | alright dood |
| 05:26.20 | jeev | you smoked too much |
| 05:26.35 | paci`` | not really |
| 05:26.37 | paci`` | im just tired |
| 05:27.14 | C4away | jeev, are you using a gui like Freepbx? |
| 05:27.48 | C4away | actually, nevermind |
| 05:28.02 | C4away | it just goes in the dial options portion of the Dial() command |
| 05:28.11 | jeev | i use to use asteriskNOW |
| 05:28.37 | *** join/#asterisk dimas (n=ds@84.53.210.46) |
| 05:28.51 | C4away | Dial(SIP/dude@somewhere,30,tTwWrR) or whatever |
| 05:29.02 | jeev | i realize |
| 05:29.09 | jeev | but my config is weird. |
| 05:29.10 | C4away | look up the dial command, I may be wrong on the argument number and position |
| 05:29.24 | C4away | well that's where they go, however you get them there |
| 05:29.26 | jeev | i did.. but there are multiple bro, i'm lost. |
| 05:29.32 | jeev | http://pastebin.com/m74192ede |
| 05:29.34 | C4away | on FreePBX it is on the General Settings tab |
| 05:29.34 | jeev | you know what i mean ? |
| 05:29.38 | jeev | i'm using asterisk. |
| 05:29.52 | jeev | [macro-stdexten] |
| 05:29.52 | jeev | exten => s,1,Dial(${ARG2},20) |
| 05:29.58 | jeev | is that it? ,20,Tr |
| 05:30.05 | C4away | The page cannot be displayed |
| 05:30.18 | C4away | Cannot find server or DNS Error |
| 05:30.41 | C4away | hmm.. it came up this time |
| 05:31.35 | C4away | for example line 18 make it exten => s,1,Dial(${ARG2},20,Tt) |
| 05:32.42 | C4away | for line 84 make it exten => 1020,1,Dial(SIP/1020,10,tT) |
| 05:33.05 | C4away | for extension => extension dialing you have to have both T and t or only one side will be able to transfer |
| 05:33.19 | C4away | you might want to create a context for exten->exten |
| 05:33.24 | C4away | and an out-dial context |
| 05:33.27 | C4away | or something like that |
| 05:34.07 | jeev | k gimme a sec |
| 05:34.23 | C4away | anyway, gotta run to the store, put tT in anywhere you are dialing and see if transfer works |
| 05:34.33 | C4away | I'll be back in 30 let me know if it works |
| 05:35.14 | jeev | k thanks bro |
| 05:35.16 | jeev | im about to get off the phone |
| 05:37.59 | *** join/#asterisk qdk (n=qdk@0x573fe57e.bynqu2.broadband.tele.dk) |
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| 06:05.55 | jeev | C4away ? |
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| 06:55.30 | Jumpie | hey guys, how hard is it to make like a calling card service? |
| 06:55.35 | Jumpie | where i can give like 3 friends a pin and can call |
| 06:56.10 | carrar | Anyone have issues with forcename & forcegreetings when running voicemail out of a DB? |
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| 07:00.20 | drmessano | hmmm |
| 07:02.30 | scooby2 | the docs say AGENTDUMP means the agent dumped the caller during the announcement. How does that work? |
| 07:03.04 | scooby2 | or does it mean the agent immediately hungup? |
| 07:03.18 | *** join/#asterisk implicit (n=implicit@ip68-105-92-210.sd.sd.cox.net) |
| 07:03.27 | C4away | Jumpie: a2billing is a pre-made calling card app |
| 07:03.45 | C4away | can be a bit of a pain to get working, and then another bit of a pain to configure, but it does its job |
| 07:05.18 | C4away | if you just want to track a few people it is probably easier to use an accountcode for each and just pull their calls from the CDR |
| 07:05.28 | C4away | bill them once a month |
| 07:05.47 | C4away | you won't have the ability to cut them off when they reach a specified limit though |
| 07:07.38 | *** join/#asterisk jfg_ (n=jfg@dyn-88-121-30-169.ppp.tiscali.fr) |
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| 07:16.40 | carrar | hahah i figured out my issue |
| 07:16.41 | carrar | w0t0 |
| 07:19.21 | drmessano | w0t0? |
| 07:19.28 | drmessano | Lemme guess, your issue was a type |
| 07:19.30 | drmessano | Lemme guess, your issue was a typo |
| 07:19.33 | drmessano | Crap |
| 07:19.36 | drmessano | :( |
| 07:19.51 | jeev | C4away, you there dood |
| 07:20.10 | drmessano | I think he's AWAY |
| 07:20.14 | drmessano | hence the name |
| 07:20.21 | jeev | scroll up. |
| 07:20.37 | drmessano | I don't need to scroll up |
| 07:20.51 | drmessano | C4away - c4 = away |
| 07:22.31 | jeev | [12:05am] <C4away> bill them once a month |
| 07:22.31 | jeev | [12:05am] <C4away> you won't have the ability to cut them off when they reach a specified limit though |
| 07:22.34 | jeev | [12:19am] <jeev> C4away, you there dood |
| 07:22.48 | drmessano | [03:20] <drmessano> C4away - c4 = away |
| 07:22.54 | jeev | lol |
| 07:22.58 | jeev | so maybe it's |
| 07:23.03 | drmessano | I can paste too |
| 07:23.25 | jeev | c4 away = acronym for something? "always watching asterisk y*" |
| 07:23.36 | drmessano | ~failburger |
| 07:23.37 | jbot | You fail at life. Have a failburger with fail fries and a large diet fail. |
| 07:33.52 | carrar | Fatburgers are good |
| 07:46.10 | Jumpie | bah cant sleep |
| 07:46.15 | Jumpie | popped some benedryl lol |
| 07:46.30 | Jumpie | i was readin c4, about a2billing |
| 07:46.34 | Jumpie | is this somethin extra i have to instlal |
| 07:46.49 | drmessano | Did you google it? |
| 07:47.05 | jeev | Jumpie, drmessano is some type of bot, it's also a failburger. |
| 07:47.37 | *** join/#asterisk lesouvage (n=lesouvag@62.140.137.125) |
| 07:47.40 | drmessano | jeev: Don't you have to wake up early tomorrow for saturday school? |
| 07:47.44 | drmessano | .... |
| 07:47.56 | jeev | i Wish |
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| 07:48.01 | C4away | Jumpie: yes you would have to install the a2billing application |
| 07:48.30 | jeev | hi c4 :D |
| 07:48.35 | jeev | ahem drmessano. |
| 07:48.43 | C4away | hello jeev |
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| 07:49.28 | jeev | hey C4away, i haven't done what you asked, i'll try right now, you gonna be here for a while ? |
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| 08:15.21 | bougie | hello :p |
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| 09:38.49 | santoshr | How to dial out using h323 channel. Dial(H323/number@ip) is this right. because asterisk doesnt do anything. i mean no packets being sent out |
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| 09:48.55 | sxpert | can I use ipv6 with asterisk ? (surely by now that should be working) ? |
| 09:49.07 | *** join/#asterisk klauwhamer (n=felixdhc@ipd50af070.speed.planet.nl) |
| 09:49.28 | *** join/#asterisk Unipz (n=derka@sudbury-cable-69-60-230-233.unitz.ca) |
| 09:51.55 | Unipz | is there anyway to pipe asterisk through exsisting rj11 jacks in your house? id like to keep my pre-exsisting wiring |
| 09:54.29 | *** join/#asterisk fordfrog (n=fordfrog@gentoo/developer/fordfrog) |
| 09:57.04 | Guggemand | Unipz ypu can use an ATA |
| 09:57.12 | Guggemand | ypo/you |
| 09:57.36 | Unipz | how much does one run for |
| 09:58.14 | Guggemand | i imagine google knows |
| 09:58.37 | Unipz | What make would you recommend? |
| 09:58.46 | Unipz | I usually stick with cisco for voip. |
| 09:58.59 | Guggemand | i dont know, ive never used one |
| 09:59.12 | Guggemand | i stick to real ip phones |
| 10:00.29 | Unipz | It's just a project im doing at home I don't really feel like running 7 ethernet cords around my house or spending a ton on wireless ip phones |
| 10:01.43 | Guggemand | remember to check if the ata you get can handle 7 phones |
| 10:01.54 | Unipz | ty, will do. |
| 10:03.12 | *** join/#asterisk planio (n=user@p548F3C5A.dip.t-dialin.net) |
| 10:04.26 | planio | i have a question concerning the line "AGI Script foo.php completed, returning 0". is it true that this line always ends with 0 in asterisk 1.4? |
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| 10:14.17 | santoshr | can i collect call from a gnugk and route it another gk with asterisk in between |
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| 10:15.03 | santoshr | gk ----> asterisk ----> remote IP is this possible asterisk is currently registered to the gk sending the call. can i make it register to two gatekeeper or something |
| 10:26.37 | santoshr | anybody with any experience on chan_h323 |
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| 10:40.24 | klauwhamer | wondering if it is possible to connect a analog telephone in a best modem to call voip with asterisk? |
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| 11:25.30 | appel-- | Is there somebody here who has experience with asterisk acting as a client of a Linksys SPA-9000 voip gateway? |
| 11:34.07 | *** join/#asterisk _charly_ (i=kroseneg@sunrise.schmidham.net) |
| 11:43.04 | klauwhamer | another question is it possible to call with a analog modem (pci card) plugged in a analog telephone on a asterisk box over the voip? |
| 11:51.41 | bminish | yes if you use ulaw / alaw codec (choose appropriate one for your region ) |
| 11:52.28 | marlow | and don't expect the best results .. your internet connectivy better be good |
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| 13:28.25 | *** join/#asterisk MDK2MDK (n=NanoTec@41.248.236.198) |
| 13:31.04 | MDK2MDK | hello, i want to ask if asterisk can make for example 20 calls in the same time with 1 analogique line ??? |
| 13:31.11 | MDK2MDK | using converters for exp |
| 13:32.17 | *** join/#asterisk RoyK (n=roy@ip-29-6-149-91.dialup.ice.no) |
| 13:34.32 | MDK2MDK | any help plzzzzzzzzzz |
| 13:35.46 | marlow | MDK2MDK : you'd need to define better, what you want to do |
| 13:36.22 | marlow | MDK2MDK : let's say .. do you have one line out and you want to pass 20 calls at the same time ? |
| 13:36.39 | marlow | MDK2MDK : that won't work .. because the one line can only handle one call at any given time |
| 13:36.46 | marlow | MDK2MDK : asterisk won't change that for you |
| 13:39.43 | MDK2MDK | and how call centers work |
| 13:39.44 | MDK2MDK | ?? |
| 13:39.55 | MDK2MDK | because call centers had only one line |
| 13:40.11 | MDK2MDK | let say that it a intenet line |
| 13:40.12 | marlow | one PRI |
| 13:40.16 | marlow | which is 30 lines |
| 13:40.17 | MDK2MDK | for exaple |
| 13:40.29 | marlow | and with one internet line, that's ok .. then your provider has many lines |
| 13:40.49 | marlow | the limit there is the size of your internet line |
| 13:41.07 | MDK2MDK | aaah know i see |
| 13:41.29 | marlow | one analogue line, one call at a time |
| 13:41.39 | marlow | one BRI, two lines at a time |
| 13:41.42 | marlow | one PRI, 30 calles at a time |
| 13:41.56 | marlow | one internet connection, as many calls as there is bandwidth |
| 13:42.37 | MDK2MDK | so if i have a internet connection whith 20 Mb |
| 13:42.53 | MDK2MDK | i can make 20 calls for exp in the same time |
| 13:43.18 | marlow | 20 mbit ? |
| 13:43.26 | marlow | nah .. there should be place for a lot more there |
| 13:43.41 | marlow | let's say .. 64 kbit codec .. + overhead .. uses about 80-90 kbit |
| 13:43.46 | marlow | that's one call |
| 13:43.57 | MDK2MDK | ook that good :) |
| 13:44.11 | marlow | so, you take your internet connection, you take what you use for other traffic |
| 13:44.20 | marlow | and subtract that |
| 13:44.28 | marlow | and you split the rest bu let's say 90 kbit |
| 13:44.39 | MDK2MDK | ok i see |
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| 13:44.45 | marlow | that's the amount of calls .. if you use a codec, that uses less bandwidth, you can pass more calls |
| 13:45.08 | marlow | like .. with G726 you'd be able to pass 2 x as many calls as with ulaw/alaw |
| 13:45.59 | MDK2MDK | ok i understand that thx a loooooooot :) |
| 13:46.04 | MDK2MDK | but in hardware, if a chose to use the internet connection wath should i have ?? |
| 13:50.31 | marlow | MDK2MDK : to talk to the outside world ? a network card |
| 13:56.30 | *** join/#asterisk MDK2MDK (n=NanoTec@41.248.236.198) |
| 13:57.01 | MDK2MDK | sorry connection problem |
| 13:57.03 | MDK2MDK | yes i need to talk outside to france for exp |
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| 14:16.34 | jorgeraidel | hi |
| 14:16.36 | jorgeraidel | :) |
| 14:16.51 | jorgeraidel | Listen i want to know something |
| 14:16.52 | jorgeraidel | :) |
| 14:17.21 | jorgeraidel | when i send one call since quintum to asterisk i receive this log |
| 14:17.23 | jorgeraidel | frame.c:202 __ast_smoother_feed: Dropping extra frame of G.729 since we already have a VAD frame at the end |
| 14:17.51 | jorgeraidel | but if send son linksys or snom is ok |
| 14:17.52 | jorgeraidel | :( |
| 14:17.57 | jorgeraidel | some suggestion? |
| 14:18.46 | jorgeraidel | hello |
| 14:19.53 | *** join/#asterisk xenonex (n=xenonex@92.47.0.5) |
| 14:22.01 | jorgeraidel | hi |
| 14:22.04 | jorgeraidel | :) |
| 14:23.13 | edwin_quijada | Hi! |
| 14:23.24 | edwin_quijada | we can implement QoS in asterisk? |
| 14:25.51 | marlow | it's not asterisk, that need QoS |
| 14:25.51 | marlow | it's your routers |
| 14:25.51 | marlow | asterisk sets the right packet-marking already |
| 14:27.42 | *** join/#asterisk rotozip (n=rotozip@c-68-34-139-139.hsd1.mo.comcast.net) |
| 14:29.06 | rotozip | This $99 Shuttle PC looks like a great Asterisk box http://tinyurl.com/4xkjz3. I just need to convince the wife that I need yet another computer. |
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| 14:51.30 | edwin_quijada | I have 2 t1 CARD that server will be appropiate for this |
| 14:52.13 | edwin_quijada | I think in Core Duo with 2gg RAM , HD sata 200gb Dell |
| 14:52.44 | edwin_quijada | 48 simultaneously calls? It is enough? |
| 14:54.26 | Maliuta | QoS is outside the realms of * |
| 14:54.28 | Maliuta | you need to do it at a router level |
| 14:54.32 | Maliuta | I recommend OpenBSD and pf |
| 14:55.15 | Maliuta | depends on your 'net connection, codec choice and what * is doing |
| 14:56.17 | UnixDog | PFsense |
| 14:56.30 | UnixDog | www.pfsense.org |
| 15:01.00 | mvanbaak | or use stock OpenBSD with nsh |
| 15:01.26 | mvanbaak | or just stock OpenBSD and your favourite editor |
| 15:01.29 | mvanbaak | that's what I do |
| 15:02.06 | mvanbaak | pfsense is freebsd. pf is an OpenBSD project so I think it's best to go for OpenBSD |
| 15:02.09 | mvanbaak | it's more complete |
| 15:03.20 | *** join/#asterisk inv_arp (n=junya@c-76-26-24-197.hsd1.fl.comcast.net) |
| 15:04.37 | Maliuta | OpenBSD + GNUs + vim |
| 15:06.35 | MDK2MDK | ztdummy what that mean , they says that it work as a timing device but a dident understand well |
| 15:13.03 | UnixDog | ztdummy = timeing source for meetme and musiconhold |
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| 15:18.03 | RoyK | http://karlsbakk.net/fun/redneck_electric_toothbrush.jpg |
| 15:22.05 | klauwhamer | another question is it possible to call with a analog modem (pci card) plugged in a analog telephone on a asterisk box over the voip? |
| 15:22.26 | *** join/#asterisk luke-jr (n=luke-jr@2002:440d:6de2:0:20e:a6ff:fec4:4e5d) |
| 15:22.46 | mvanbaak | klauwhamer: no |
| 15:22.53 | mvanbaak | klauwhamer: you need a TDM card for that |
| 15:23.14 | *** join/#asterisk tobias (n=tobias@user-0ce2hpk.cable.mindspring.com) |
| 15:23.29 | klauwhamer | mvanbaak: what is TDM |
| 15:24.01 | mvanbaak | it's a card you can buy from digium that is supported by asterisk and allows you to connect phonelines and phones |
| 15:24.37 | klauwhamer | mvanbaak: ok |
| 15:25.12 | klauwhamer | mvanbaak: can i call with voip pstn lines ? |
| 15:25.26 | mvanbaak | pstn != voip |
| 15:25.31 | mvanbaak | so I have no idea what you mean |
| 15:26.41 | klauwhamer | mvanbaak: i mean the old telephone network |
| 15:27.02 | mvanbaak | you can use that if you have the TDM card yes |
| 15:27.09 | Maliuta | mvanbaak: that's a very simple way of looking at it |
| 15:27.42 | mvanbaak | Maliuta: I know. but for most users it's true |
| 15:28.07 | klauwhamer | mvanbaak: can i call everyone on analog phonelines free with voip ? |
| 15:28.21 | mvanbaak | klauwhamer: no |
| 15:28.31 | klauwhamer | what do i need more ? |
| 15:28.38 | Maliuta | klauwhamer: you are far from a clue |
| 15:29.04 | mvanbaak | I give up |
| 15:29.41 | mvanbaak | klauwhamer: why do you think phonecalls using voip will be free to every landline in the world ? |
| 15:29.51 | *** join/#asterisk bl4q (n=Bl@dslb-088-064-155-115.pools.arcor-ip.net) |
| 15:30.44 | klauwhamer | mvanbaak: so i dont need a account just a internet connection am i right? |
| 15:30.54 | mvanbaak | wrong |
| 15:31.34 | klauwhamer | mvanbaak: can i call everyone who is using voip ? |
| 15:31.43 | mvanbaak | if you want to do voip to landlines you either need a landline yourself and connect that to asterisk, or get an account at an ITSP |
| 15:32.05 | mvanbaak | klauwhamer: you can call ppl who allow anonymous voip calls yeah. but you have to know how to reach them |
| 15:32.07 | klauwhamer | i see |
| 15:32.30 | plik | mvanbaak: to be fair, a lot of basic consumer level marketing does seem to suggest that "voip means free calls" period. |
| 15:32.45 | mvanbaak | for example: you can call me on IAX2/guest@lunteren.vanbaak.info/michiel |
| 15:33.02 | mvanbaak | but you cant call my parents (who are using voip as well, but dont allow anonymous calls) |
| 15:33.13 | klauwhamer | ok |
| 15:34.36 | mvanbaak | plik: yeah, think so. it's so darn stupid |
| 15:34.39 | klauwhamer | mvanbaak: it is just like email adresses but more secure ? am i right? |
| 15:35.08 | mvanbaak | huh ? |
| 15:35.15 | mvanbaak | why would voip be more secure then email ? |
| 15:35.57 | klauwhamer | mvanbaak: well you said not everyone allows anonemous calls |
| 15:36.22 | mvanbaak | almost noone is, simply because they have no clue and cannot fix it |
| 15:36.35 | mvanbaak | look, you need something that can accept incoming voip calls |
| 15:36.46 | drmessano | Hmmm |
| 15:36.49 | mvanbaak | most home users that use voip simply have a phone that registers with some provider |
| 15:37.04 | drmessano | So if my mail server rejects all outside mail, then I have secure email? |
| 15:37.10 | drmessano | Fascinating.. and useless. |
| 15:37.11 | plik | lol |
| 15:37.16 | mvanbaak | lol drmessano |
| 15:37.49 | mvanbaak | klauwhamer: I really think you need to study what voip is |
| 15:38.13 | klauwhamer | mvanbaak: yeah i think you are right |
| 15:38.16 | plik | drmessano: does HappyClownPBX[TM] allows secure anonymous pstn voip calls? |
| 15:38.16 | drmessano | VoIP isn't "Free calls forever, sticking it to the man, free kevin?" |
| 15:38.45 | mvanbaak | voip is overrated |
| 15:38.53 | mvanbaak | it's buggy, unstable, and useless |
| 15:38.58 | drmessano | HappyClownPBX[TM] only allows anonymous MGCP calls |
| 15:39.04 | *** join/#asterisk iamhrh (n=iamhrh@office.amsvans.com) |
| 15:39.32 | mvanbaak | well, at least MGCP supports setting a parkinglot in the new multiparking branch of asterisk ;) |
| 15:39.38 | mvanbaak | I just put that in 2 days ago |
| 15:39.38 | drmessano | lol |
| 15:39.41 | drmessano | cool |
| 15:39.42 | plik | heh |
| 15:40.02 | drmessano | I may have to fork HCPBX |
| 15:40.13 | klauwhamer | mvanbaak: what technology should replace voip then ? |
| 15:40.14 | mvanbaak | yeah |
| 15:40.24 | mvanbaak | klauwhamer: pidgins |
| 15:40.33 | drmessano | Start a branch that does allow SIP anonymous and call it "ClownKilledMyDadPBX[TM]" |
| 15:40.40 | *** join/#asterisk JayTee52 (n=jforde05@c-69-243-161-112.hsd1.in.comcast.net) |
| 15:41.03 | iamhrh | can anybody point me in the right direction here... I'm trying to create a program using iaxclient that makes a call to *, and then plays various files over the call. I thought iaxc_play_sound would do what I was looking for - but it seems to just play audio out over the output devices |
| 15:41.13 | drmessano | The reference comes from an SNL skit.. Deep Thoughts |
| 15:41.29 | iamhrh | i'm just looking for the right documentation or examples - or something :-) |
| 15:41.44 | drmessano | "I don't know why I am afraid of clowns. I think it may go back to the time that we went to the circus when I was a kid, and a clown killed my dad" |
| 15:41.59 | klauwhamer | mvanbaak: so the market is making a move to voip, so you suggest that is a stupid move |
| 15:41.59 | errr | lol! |
| 15:42.06 | errr | I love deep thoughts |
| 15:42.20 | drmessano | That was some of the funniest crap ever |
| 15:42.26 | errr | indeed |
| 15:42.49 | JayTee52 | "If you're returning from space in a shuttle with your dog, don't let him stick his head out the window during re-entry or it'll burn off." - Deep Thoughts by Jack Handy |
| 15:43.01 | drmessano | Yep lol |
| 15:43.03 | mvanbaak | klauwhamer: it all depends on the usecase, and the reason to switch to voip |
| 15:43.09 | drmessano | Love that one |
| 15:43.13 | drmessano | Alright.. I need to head out for a bit.. BB to terrorize all later |
| 15:43.20 | JayTee52 | later |
| 15:43.25 | *** join/#asterisk shinao1 (n=shinao1@smtp.gtbank.com) |
| 15:43.26 | klauwhamer | mvanbaak: ok |
| 15:43.29 | mvanbaak | if ppl switch to voip because they think they get free calls, they should switch to pidgins instead |
| 15:44.16 | plik | klauwhamer: he didn't say stupid, just overrated, buggy, unstable and useless -- like so much modern stuff these days |
| 15:44.22 | JayTee52 | if ppl switch to VOIP because they think it's easier they should switch to carrier pidgeon instead. |
| 15:44.37 | *** join/#asterisk UnixDog (n=UnixDog@ppp-69-238-167-52.dsl.irvnca.pacbell.net) |
| 15:44.44 | mvanbaak | JayTee52: that's what I said |
| 15:45.16 | plik | s/free calls /easier/ |
| 15:45.28 | mvanbaak | plik: and specially since most networks still have *shruk* hubs in their infrastructure .... |
| 15:45.37 | klauwhamer | mvanbaak: ill have to study pidgeon too |
| 15:45.44 | JayTee52 | not really, you said free, I said easy, you said pidgin, I said carrier pidgeon as in the bird with note on the leg :-) |
| 15:46.19 | mvanbaak | I meant the bird as well |
| 15:46.26 | JayTee52 | hehe |
| 15:46.26 | mvanbaak | it's even in an RFC |
| 15:46.34 | mvanbaak | aviation carrier or something |
| 15:46.39 | klauwhamer | ill have to study pidgeon/pidgins |
| 15:46.51 | *** join/#asterisk inv_arp (n=junya@c-76-26-24-197.hsd1.fl.comcast.net) |
| 15:46.52 | klauwhamer | is that supported by asterisk |
| 15:46.59 | *** join/#asterisk qdk (n=qdk@195.242.194.42) |
| 15:47.01 | JayTee52 | the downside to that is the maintenance costs of cleaning all the poop off your roof |
| 15:47.23 | mvanbaak | <PROTECTED> |
| 15:47.46 | mvanbaak | klauwhamer: look here: http://www.blug.linux.no/rfc1149/ |
| 15:47.48 | JayTee52 | someone had way too much time on their hands |
| 15:48.04 | klauwhamer | mvanbaak: ok |
| 15:48.12 | mvanbaak | JayTee52: yup |
| 15:48.59 | JayTee52 | hahaha, http://img.4chan.org/b/src/1208014237125.jpg |
| 15:49.39 | jbeez | hahha |
| 15:50.30 | mvanbaak | hahaha |
| 15:50.44 | JayTee52 | I just sent that link to my buddy in Helsinki |
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| 15:55.15 | *** mode/#asterisk [+o anthm] by ChanServ |
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| 15:57.46 | mvanbaak | brb |
| 15:59.37 | klauwhamer | k |
| 16:09.32 | *** part/#asterisk jivco (n=jivco@85.187.217.6) |
| 16:15.08 | klauwhamer | mvanbaak: it has not mutch to say |
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| 16:17.41 | *** join/#asterisk Strom_C (n=strom@208.127.172.112) |
| 16:19.52 | Jumpie | yay my cheap fxo card just came :) |
| 16:20.24 | UnixDog | have fun getting it to work |
| 16:23.00 | Jumpie | it'l work |
| 16:23.04 | Jumpie | and its just for myh home |
| 16:23.18 | Jumpie | isnt there a basic utility you run to detect what you have? |
| 16:32.14 | *** join/#asterisk jumpie (n=jumpie@pool-96-231-155-171.washdc.fios.verizon.net) |
| 16:36.03 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 16:37.45 | jumpie | just wondering, if i have my phone line into my fxo card, but also have a home phone plugged into another jack in my house, on an incoming call, which would win? |
| 16:38.06 | C4away | if you have an IVR or something on the pbx then it will answer right away |
| 16:38.18 | C4away | or if you have fax detect enabled it will answer and play ringing tones listening for the fax tone |
| 16:38.20 | jumpie | no ivr... |
| 16:38.57 | C4away | but if you just have it ring another line, it shouldn't pick up until a device on the server answers and seizes the line |
| 16:39.00 | jumpie | i just hoping i can avoid unpluggedin al my phones |
| 16:39.09 | C4away | what card do you have? |
| 16:39.19 | jumpie | its a cheap clone, i had alrady bought it a week ago |
| 16:39.24 | jumpie | t100p or somethin |
| 16:39.28 | C4away | x100p clone? |
| 16:39.30 | Darthclue | typically, the fxo will win ... my analog phone rings once, then the box picks up |
| 16:39.34 | C4away | install and run oslec |
| 16:39.35 | jumpie | but it supposedly detects as a wildcard |
| 16:39.41 | C4away | that will get rid of most of the echo problems |
| 16:39.50 | jumpie | oslec? is that a library i can get right from yum? |
| 16:40.02 | C4away | not sure if it is available in yum |
| 16:40.06 | jumpie | k |
| 16:40.09 | C4away | google it, you should find the source and be able to comile it |
| 16:40.10 | jumpie | i gotta get it detected first |
| 16:40.12 | C4away | compile rather |
| 16:40.13 | jumpie | thanks |
| 16:40.21 | C4away | yea, just write that down, you'll probably need it |
| 16:40.31 | jumpie | I think that the zaptel hardware you have on your system is: |
| 16:40.31 | jumpie | pci:0000:03:0a.0 wcfxo+ 1057:5608 Wildcard X100P |
| 16:40.35 | jumpie | not bad eh :) |
| 16:40.40 | jumpie | ok thanks C4away |
| 16:40.59 | C4away | if by not bad you mean that it guessed it correctly, sure |
| 16:41.02 | *** join/#asterisk ManxPower (n=manxpowe@15.sub-70-222-77.myvzw.com) |
| 16:41.12 | C4away | if you are talking about the card itself, I don't know, never used one |
| 16:41.18 | jumpie | right but dont most clones not really show up as a wildcard |
| 16:41.22 | C4away | I actually want to get my hands on one to see if it is really as bad as everyone says |
| 16:41.23 | jumpie | ah k |
| 16:41.35 | C4away | depending on the modem chipset |
| 16:41.37 | ManxPower | C4away: The X100P clones? |
| 16:41.47 | MDK2MDK | if we chose to call whith asterisk using voip , do wi need a zaptel hardware ? |
| 16:41.50 | jumpie | ffs..i had this problem last time, chkconfig doenst seem to want to work, even though its in my path |
| 16:41.56 | C4away | if they use the same exact chipset as digium used then the detect script can't tell the difference, and there isn't much anyway |
| 16:42.03 | ManxPower | MDK2MDK: Generally no. |
| 16:42.16 | MDK2MDK | ah oky |
| 16:42.18 | MDK2MDK | thx |
| 16:42.29 | jumpie | is chkconfig outdated? |
| 16:42.37 | jumpie | i remember having this problem last time and cant remember how i got it fixed |
| 16:43.11 | C4away | ManxPower: yea, I'd like to play with an x100p at some point just to see if it really is so bad |
| 16:43.38 | jumpie | ah...hmm i have to type /sbin./chkconfig |
| 16:43.43 | jumpie | that doesnt make sense because sbin is in my path |
| 16:44.14 | ManxPower | C4away: The chipset used in the X100Ps have not been made in several years |
| 16:44.45 | ManxPower | jumpie: chkconfig is part of your Distro, not Asterisk |
| 16:44.58 | jumpie | ManxPower, i know....the problem is somehow /sbin isnt in my path i suppose |
| 16:45.05 | jumpie | service start didnt work either |
| 16:45.10 | jumpie | had to /sbin |
| 16:45.23 | jumpie | i mean i can fix it, just strange i figured that was a default path dir |
| 16:45.45 | ManxPower | jumpie: sounds like your OS is screwed up |
| 16:46.09 | jumpie | hah |
| 16:46.13 | jumpie | doesnt seem to be other than that :) |
| 16:46.17 | jumpie | everything installed and compiled just fine |
| 16:46.28 | jumpie | i hope not |
| 16:48.44 | ManxPower | what distro are you using? |
| 16:48.53 | jumpie | centos 5 |
| 16:49.03 | jumpie | i fixed the path, just hope its not cleared at nex reboot |
| 16:49.26 | ManxPower | Did you fix it in your .rc files? |
| 16:49.48 | jumpie | ah no |
| 16:49.51 | jumpie | .bashrc? |
| 16:50.24 | jumpie | will do |
| 16:53.04 | *** join/#asterisk mwalling (i=mwalling@you.dontlike.us) |
| 17:00.07 | *** join/#asterisk comprookie2000 (n=david@adsl-065-012-210-216.sip.bct.bellsouth.net) |
| 17:03.58 | jumpie | ok this is confusing to me, in zapata.conf, it says in the comments that fxs is the default for signalling, yet fxo_ls is set |
| 17:06.46 | MDK2MDK | hello |
| 17:07.10 | plik | jumpie: for an fxo card connected to a phone line, you want fxs signalling, prolly with ks - the book describes the difference |
| 17:07.27 | MDK2MDK | i use the zttool and the stcfg but i had some errors |
| 17:07.30 | jumpie | yea, mentions to do that in the zaptel.conf as well |
| 17:07.39 | MDK2MDK | what's the problem?? |
| 17:07.40 | jumpie | just curiosu why the default wasnt what was set, wanted to be sure lol |
| 17:08.21 | MDK2MDK | unable to open master device /dev/zap/ctl |
| 17:09.08 | plik | because the same conf file also configures FXS ports for analog phones toplug in to which use fxo signalling |
| 17:09.23 | jumpie | ok well this file has a buttload of config options |
| 17:09.30 | jumpie | i guess i can ignore all the span/timing stuff eh |
| 17:09.46 | plik | certainly at first yes |
| 17:11.47 | MDK2MDK | some help pls |
| 17:13.26 | *** join/#asterisk colinm_ (n=colol@VDSL-130-13-122-158.PHNX.QWEST.NET) |
| 17:14.21 | jumpie | plik in zapata.conf the context= |
| 17:14.26 | jumpie | is pinting to extensions.conf right? |
| 17:14.57 | plik | yes |
| 17:15.05 | jumpie | thanks just verifyin |
| 17:16.23 | jeev | toot toot. |
| 17:16.30 | jumpie | ok after i did the modprobe of the correct drivers, and i did a ztcfg -vv |
| 17:16.50 | jumpie | i got http://jumpie.pastebin.org/29249 |
| 17:16.53 | jumpie | so thats all good right |
| 17:17.21 | jeev | ya looks good |
| 17:17.25 | jeev | lool ,like i know what to look for |
| 17:17.55 | jumpie | hehe |
| 17:18.47 | pa | is it normal that if i specify sippeers and iaxpeers in extconfig.conf, static iax users in iax.conf still work? |
| 17:19.49 | pa | and also, get no debug from realtime asterisk odbc module? |
| 17:22.45 | *** join/#asterisk shinao1 (n=shinao1@smtp.gtbplc.com) |
| 17:28.04 | pa | also |
| 17:28.04 | pa | iaxcomm keep saying |
| 17:28.04 | pa | PortAudio error at Unable to open streams: Illegal error number |
| 17:30.46 | jumpie | did you install the driver? |
| 17:30.49 | jumpie | what car do you have? |
| 17:30.53 | jumpie | card |
| 17:34.24 | pa | sec maybe i have some problem with the linux audio driver |
| 17:34.24 | pa | i want to try on a windows machine |
| 17:34.24 | pa | because |
| 17:34.28 | pa | i have a strange problem: |
| 17:34.57 | pa | when i call, (and if iaxcomm works), i see in the asterisk debug, it plays the file i told him, (in gsm format) but i cant hear anythig |
| 17:35.20 | jumpie | hey C4away u there man |
| 17:37.00 | jumpie | or plik |
| 17:37.02 | jumpie | :P |
| 17:37.23 | pa | now it's funny.. if i try to call non-existent extension, debug shows "Rejected connect attempt from 192.168.1.2, request '555@default' does not exist" |
| 17:37.23 | pa | if i try to call existent extension |
| 17:37.24 | pa | debug shows nothing |
| 17:37.59 | pa | and goes timeout |
| 17:45.37 | pa | full of nicks here.. just parked.. |
| 17:45.45 | *** join/#asterisk rickross (n=rickross@supporter/active/rickross) |
| 17:46.18 | *** join/#asterisk mecredis (n=fcb@cpe-74-73-135-105.nyc.res.rr.com) |
| 17:46.26 | mecredis | hi, I'm having issues setting up call files |
| 17:46.36 | mecredis | I have a php script that generates a call file |
| 17:46.44 | mecredis | and asterisk pays attention to it when I run it from the CLI |
| 17:46.55 | mecredis | but when I try to access it from a browser |
| 17:46.57 | mecredis | it won't execute |
| 17:47.55 | jumpie | my outbound calling would be Zap/G1 right? |
| 17:48.04 | jumpie | considering group 1 is default and channel 1 as well |
| 17:51.07 | jumpie | argh wtf |
| 17:51.15 | pa | moreover why the heck doesnt asterisk have a line in asterisk.,conf for sounds???????????????????? |
| 17:51.16 | jumpie | it says ignoring signalling when i go tor eload |
| 17:51.24 | jumpie | what do you mean pa? |
| 17:51.31 | jumpie | the sounds are in a specific directory |
| 17:51.36 | jumpie | you call it with background or playback |
| 17:53.32 | pa | yes |
| 17:53.57 | pa | jumpie: where do you specify this directory? |
| 18:01.12 | jumpie | what are you trying to do? |
| 18:01.31 | jumpie | i think its normally in /var/lib/asterisk/sounds |
| 18:01.49 | jumpie | http://jumpie.pastebin.org/29271 im having this problem with my channels |
| 18:02.34 | pa | jumpie, no, it's actually in /usr/share/asterisk/sound |
| 18:02.36 | pa | but |
| 18:02.43 | pa | WHERE can i specify it? |
| 18:02.45 | pa | set |
| 18:02.47 | pa | declare |
| 18:02.49 | pa | whatever |
| 18:03.10 | pa | is that hardcoded like in the worst crappy pieces of software? |
| 18:07.24 | bkw_ | what are we talkin bout? |
| 18:08.24 | bkw_ | Jumpie: you can't change the signalling when you reload chan_zap.. thats a dangerous operation |
| 18:09.00 | jumpie | wat...what do you mean? |
| 18:09.15 | jumpie | i did what i was supposed to, edited the files, THEN reloaded it |
| 18:09.20 | bkw_ | pa: its usually off /var/lib/asterisk |
| 18:09.47 | bkw_ | Jumpie: well if you get a message saying ignoring signaling thats why.. you can't change signaling without restarting asterisk |
| 18:10.03 | jumpie | oh duh |
| 18:10.05 | bkw_ | because if you get it wrong and they allowed that it would segfault.. thats a very messy operation |
| 18:10.06 | *** join/#asterisk Nasra (n=Nasra@CPE001217b1920e-CM00111ade9528.cpe.net.cable.rogers.com) |
| 18:10.08 | bkw_ | lots of moving parts |
| 18:10.11 | bkw_ | not wise to allow it |
| 18:10.15 | *** join/#asterisk oej (n=olle@ns.webway.se) |
| 18:10.21 | jumpie | well i reloaded extensions.conf and did chan zap |
| 18:10.33 | bkw_ | reloading things is what starts the biggest part of segfaults |
| 18:10.52 | jumpie | ok, so what do i need to do? when i ztcfg it shows its there, but instead of saying configured, it says 'to be configured' |
| 18:10.56 | jumpie | even though i have edited all what i need |
| 18:11.06 | bkw_ | well stop asterisk |
| 18:11.08 | bkw_ | ztcfg -vvv |
| 18:11.08 | jumpie | k |
| 18:11.11 | bkw_ | start asterisk back |
| 18:11.20 | bkw_ | if everything is where it should be then it should work |
| 18:11.51 | jumpie | ok, i stopped it |
| 18:12.01 | jumpie | and when i ran ztcfg -vvv it still says 'to configure' instead of 'configured' |
| 18:12.13 | jumpie | which i dont get, because there's realy only 3 main things i needed to edit,l which i did |
| 18:12.17 | bkw_ | what exactly are you trying to connfigure? |
| 18:12.35 | jumpie | signaling, loadzone, default zone |
| 18:12.50 | bkw_ | in zaptel.conf? |
| 18:12.53 | jumpie | yea |
| 18:13.04 | bkw_ | try removing all the zaptel modules |
| 18:13.06 | bkw_ | and reloading them |
| 18:13.18 | ectospasm | jumpie: don't get caught up in "to configure" and "configured"... those messages mean the same thing... |
| 18:13.21 | jumpie | you mean physically take the card out? |
| 18:13.24 | jumpie | ectospasm, oh.. |
| 18:13.31 | jumpie | oh..ok hold on then |
| 18:13.34 | bkw_ | Jumpie: no the kernel modules |
| 18:13.43 | bkw_ | sorry should have been more clear |
| 18:13.43 | jumpie | bkw, i had recompiled the zaptel |
| 18:13.55 | bkw_ | did you remove the existing ones that were loaded |
| 18:13.55 | jumpie | and it was successful, because earlier i ahd isntalled it WITHOUT the card present |
| 18:14.02 | bkw_ | rmod zaptel |
| 18:14.04 | jumpie | hmm, no |
| 18:14.05 | *** join/#asterisk jm|laptop (n=jm|home@dilbert.jamiem.com) |
| 18:14.05 | jumpie | ok |
| 18:14.07 | jumpie | shit |
| 18:14.17 | bkw_ | well if they were loaded and you recompiled then I highly doubt the new ones are loaded |
| 18:14.31 | jumpie | it think it was, because it detected the wildcard |
| 18:14.37 | jumpie | but i can start over just to be sure |
| 18:14.40 | ectospasm | you have to modprobe the drivers after you build them... make install doesn't do that |
| 18:14.55 | ectospasm | ...the zaptel init script will do that for you... |
| 18:14.55 | jumpie | i did that |
| 18:15.18 | jumpie | modprobe zaptel, modprobe wcfxo, modprobe wcusb |
| 18:15.20 | jumpie | no errors |
| 18:15.38 | ectospasm | you actually have those devices? |
| 18:15.51 | jumpie | i have the wcfxo |
| 18:15.57 | jumpie | book told me to do all 3 |
| 18:16.17 | jumpie | i can back out, but rmod is file not found |
| 18:16.21 | ectospasm | modprobe wcfxo should automatically load zaptel, since it's a dependency |
| 18:16.26 | jumpie | ah k |
| 18:16.30 | ectospasm | modprobe -r |
| 18:16.30 | jumpie | wel. it seemed to work, didnt get an error |
| 18:16.32 | ectospasm | or rmmod |
| 18:16.33 | jumpie | ok |
| 18:16.48 | jumpie | so what all should id o, modprobe -r zaptel, or wcfxo, or boht? |
| 18:16.55 | jumpie | both sorry |
| 18:17.03 | ectospasm | if you modprobe -r wcfxo it should work |
| 18:17.29 | jumpie | k sec |
| 18:17.31 | ectospasm | repeat that, lsmod | grep zaptel until it has no output |
| 18:17.47 | ectospasm | er... you may have to give an explicit modprobe |
| 18:17.49 | jumpie | ok |
| 18:18.08 | ectospasm | but if you're using the init script, you should be able to say service zaptel stop (or /etc/init.d/zaptel stop) |
| 18:18.08 | jumpie | zaptel 190212 9 wcusb,xpp,wctdm,wctdm24xxp,wcte11xp,wct1xxp,wcte12xp,wct4xxp,tor2 |
| 18:18.09 | jumpie | crc_ccitt 6337 1 zaptel |
| 18:18.17 | jumpie | which i did, service zaptel start |
| 18:18.18 | ectospasm | you're loading way too many drivers |
| 18:18.22 | jumpie | ffs |
| 18:18.27 | jumpie | somethin did it automatically then |
| 18:18.31 | ectospasm | service zaptel stop |
| 18:18.33 | jumpie | all i did manually was wcusb zaptel and fxo |
| 18:18.34 | jumpie | k |
| 18:18.40 | jumpie | done |
| 18:18.42 | ectospasm | edit /etc/sysconfig/zaptel, only add the drivers you need |
| 18:19.09 | ectospasm | I don't understand why you need wcusb... do you have the hardware to go with it? |
| 18:19.11 | jumpie | so what automatically loaded so many drivers? when i did chkconfig zaptel or what |
| 18:19.16 | ectospasm | I don't even remember what that driver is for |
| 18:19.19 | jumpie | ectospasm, no, i may have just been mistaken in reading this |
| 18:19.22 | jumpie | it may haver been ane xample |
| 18:19.54 | jumpie | holy shit they are all there..yep |
| 18:19.58 | ectospasm | jumpie: /etc/sysconfig/zaptel tells /etc/init.d/zaptel (service zaptel...) what to load |
| 18:20.01 | jumpie | why wasnt this explained at all |
| 18:20.05 | jumpie | yea i get it..just a sec |
| 18:20.27 | ectospasm | what book are you reading? |
| 18:20.46 | *** join/#asterisk inv_arp (n=junya@c-76-26-24-197.hsd1.fl.comcast.net) |
| 18:20.48 | jumpie | i was going off a wiki, asterisk for dummies, and the "~book" |
| 18:20.57 | jumpie | all have been invaluable and i've had no problems until now |
| 18:21.10 | jumpie | btw wcusb is a usb single fxo..so definately not it |
| 18:21.37 | jumpie | ok its changed to just load wcfxo |
| 18:22.13 | jumpie | now i modprobe wcfxo and thast it right |
| 18:22.31 | ectospasm | what do you mean isn't right? |
| 18:22.38 | ectospasm | what card do you have? |
| 18:22.44 | jumpie | i have x100p |
| 18:22.50 | jumpie | no , i was responding to what you said about wcusb |
| 18:22.52 | jumpie | you didnt know what it was |
| 18:23.01 | jumpie | i was reading in the config and it says its a usb adapter single fxo port, which i dont ahve |
| 18:23.10 | jumpie | so just reiterating you were right about having way too many drivers |
| 18:23.11 | ectospasm | I thought it was that... |
| 18:23.36 | ectospasm | you do know that the x100p is obsolete...? |
| 18:24.04 | jumpie | lol yeah |
| 18:24.07 | jumpie | it was a cheap card i got on ebay |
| 18:24.09 | jumpie | strictly for lab purposes |
| 18:24.14 | jumpie | no way in hell ill use it for production |
| 18:24.18 | *** join/#asterisk riddlebox (n=james@75-132-225-75.dhcp.stls.mo.charter.com) |
| 18:24.27 | jumpie | but the firmware is recognzied as wildcard not a clone |
| 18:24.29 | jumpie | so i guess thats a plus |
| 18:24.38 | jumpie | ok so modprobe wcfxo? or somethin else first? |
| 18:24.40 | *** join/#asterisk Great_Randew (n=Andrew@stjhnbsu84w-156034183064.nb.aliant.net) |
| 18:25.06 | ectospasm | I dunno, I would assume so... what's the problem you get when you modprobe/zaptel start with just wcfxo? |
| 18:25.32 | jumpie | havnet yet |
| 18:25.33 | jumpie | will do now |
| 18:25.42 | jumpie | it just goes to next line, no errors, message |
| 18:25.46 | jumpie | i assume no news=good news :) |
| 18:25.57 | jumpie | oh..kewl |
| 18:26.04 | jumpie | Changing signalling on channel 1 from Unused to FXS Kewlstart |
| 18:26.23 | ectospasm | no output from modprobe is a good sign |
| 18:26.34 | jumpie | ok |
| 18:26.37 | *** join/#asterisk mwalling_ (i=mwalling@you.dontlike.us) |
| 18:26.39 | jumpie | so try to reload asterisk now? |
| 18:26.40 | ectospasm | so ztcfg returns no erros |
| 18:26.43 | jumpie | correct |
| 18:26.52 | ectospasm | yeah, launch asterisk now |
| 18:26.53 | jumpie | [14:25:50] [root@ippbx: /etc]$ lsmod | grep zaptel |
| 18:26.53 | jumpie | zaptel 190212 1 wcfxo |
| 18:26.55 | jumpie | just one :D |
| 18:27.11 | jumpie | ok..no errors starting asterisk |
| 18:27.34 | jumpie | is there away to just see if i get dial tone? |
| 18:27.40 | jumpie | or just try an inbound call eh |
| 18:28.44 | jumpie | do i need to unplug the other phones in my house? i ahve dt on them still |
| 18:28.53 | jumpie | i got zero input in asterisk |
| 18:29.01 | jumpie | and nothin rang, not my analog phones, or my soft phone |
| 18:29.03 | mvanbaak | you dont have to unplug them |
| 18:29.07 | jumpie | k |
| 18:29.32 | jumpie | context=fios-line |
| 18:29.46 | jumpie | signalling=fxs_ks |
| 18:29.53 | jumpie | channel => 1 |
| 18:30.04 | jumpie | thast really all thats relevant right? of course the other caller id,3 way calling, etc defaults |
| 18:30.04 | ectospasm | did you watch on the CLI (asterisk -rvvvvvvvvvvvv) when the call came in? |
| 18:30.10 | jumpie | not that many v lol |
| 18:30.14 | jumpie | lemme try again |
| 18:30.23 | mvanbaak | or do: |
| 18:30.25 | mvanbaak | asterisk -r |
| 18:30.30 | mvanbaak | core set verbose 255 |
| 18:30.33 | jumpie | howly shit |
| 18:30.34 | jumpie | k |
| 18:31.01 | *** join/#asterisk solar_ant (n=John@122.164.227.234) |
| 18:31.13 | jumpie | doesnt like that |
| 18:31.13 | ectospasm | heh, for a while I tried to memorize the signed 32bit integer limit... 2bn, or so |
| 18:31.22 | jumpie | er hold on a sec |
| 18:31.42 | jumpie | not good wtf |
| 18:31.50 | jumpie | Unable to connect to remote asterisk (does /var/run/asterisk.ctl exist?) |
| 18:31.55 | jumpie | i didnt get that until just now |
| 18:32.02 | jumpie | it starts to connect |
| 18:32.25 | ectospasm | is asterisk still running? |
| 18:32.28 | jumpie | this happened after i did a stop now |
| 18:32.33 | ectospasm | of course |
| 18:32.35 | ectospasm | You stopped asterisk |
| 18:32.47 | jumpie | right, i thought asterisk -r restarted if its not running |
| 18:32.50 | jumpie | sorry i feel silly |
| 18:32.53 | ectospasm | if you're in a -r session, you can type exit to get out, without stopping asterisk |
| 18:33.02 | jumpie | i have to service asterisk start? |
| 18:33.12 | ectospasm | not necessarily |
| 18:33.13 | jumpie | right i know |
| 18:33.16 | jumpie | but you ahd said to sotp asterisk |
| 18:33.19 | jumpie | so i did |
| 18:33.45 | ectospasm | I don't recall telling you to stop asterisk |
| 18:33.47 | *** join/#asterisk sione (i=sione@ocs.net) |
| 18:34.30 | sione | I upgraded to asterisk 1.6 and now my hint not showing when line in-use or ringing as it did with asterisk 1.4 |
| 18:34.41 | jumpie | ectospasm> service zaptel stop |
| 18:34.43 | jumpie | lol |
| 18:35.00 | jumpie | its ok i know what to do |
| 18:35.06 | ectospasm | zaptel is independent of asterisk |
| 18:35.14 | ectospasm | hence zaptel.conf is not in /etc/asterisk |
| 18:35.21 | jumpie | i know, but i gues i had thouhgt you said stop it |
| 18:35.23 | jumpie | its all good ir estarted it |
| 18:35.32 | jumpie | verbose is now 255 |
| 18:35.53 | jumpie | hah |
| 18:35.55 | jumpie | works now :) |
| 18:35.59 | jumpie | woooooooooooooooooooot |
| 18:36.04 | ectospasm | w00t |
| 18:36.14 | jumpie | would you min calling me just to check for echo? |
| 18:36.20 | jumpie | i cant have a convo from my cell phone to my xlite |
| 18:36.23 | jumpie | :) |
| 18:36.39 | jumpie | you said to get something else later too |
| 18:36.40 | jumpie | oclec? |
| 18:36.45 | sione | to test echo I dial google-411 :) |
| 18:37.01 | ectospasm | there's the Echo diaplan app |
| 18:37.25 | mvanbaak | jumpie: oslec |
| 18:37.37 | mvanbaak | it's an opensource echo cancel thingie |
| 18:37.40 | mvanbaak | it's really great |
| 18:37.46 | jumpie | k |
| 18:37.52 | jumpie | ectospasm, but so..that has nothin to do with distance or what? |
| 18:37.56 | jumpie | thanks mvan |
| 18:38.00 | jumpie | im gonna do a test cdall before and after |
| 18:38.06 | mvanbaak | ok, food |
| 18:38.30 | ectospasm | wait, I was thinking of something else... the Echo application just echoes sounds back to you... no good test of echo |
| 18:38.40 | jumpie | right |
| 18:38.41 | ectospasm | because it explicitly HAS echo |
| 18:38.41 | jumpie | lol |
| 18:38.49 | jumpie | im haviing a friend call me now, then ill install oslec |
| 18:38.51 | jumpie | and see difference |
| 18:39.34 | sione | anyone know how to get hint working in asterisk 1.6 link it did with asterisk 1.4? |
| 18:39.39 | sione | er like |
| 18:44.26 | sione | sighs |
| 18:46.56 | *** join/#asterisk mwalling (i=mwalling@you.dontlike.us) |
| 18:49.39 | ManxPower | sione: call-limit= |
| 18:49.47 | ManxPower | or calllimit= I don't recall which |
| 18:50.11 | *** join/#asterisk ccvp (n=Owner@user-24-214-126-81.knology.net) |
| 18:50.25 | sione | I have it set to 2 as i did with asterisk 1.4 |
| 18:50.52 | *** join/#asterisk steliosk (n=Stelios@athedsl-105743.home.otenet.gr) |
| 18:50.55 | *** join/#asterisk jeffgus (n=jeffgus@216.86.199.4) |
| 18:51.54 | *** join/#asterisk UnixDog (n=UnixDog@ppp-69-238-167-52.dsl.irvnca.pacbell.net) |
| 18:52.34 | ManxPower | perhaps it's broken. 1.6 is BETA |
| 18:52.40 | sione | bummer |
| 18:52.49 | sione | oh well |
| 18:53.24 | lirakis | does anyone know of an online fax sending service... where i can upload an image and it will fax it to a number |
| 18:53.36 | lirakis | ... im trying to test my rxfax installation |
| 18:53.47 | lirakis | .. and i dont have a fax machine to test with |
| 18:54.14 | ManxPower | I did not say is IS, I said MAYBE |
| 18:54.25 | sione | ya |
| 18:54.33 | luke-jr | ManxPower: 1.4 is broken ;) |
| 18:54.54 | MDK2MDK | can somme one help me , when a want to start Zaptel an asterisk services i have this problem : Error: missing /dev/zap! |
| 18:54.56 | UnixDog | why is 1.4 broken |
| 18:55.03 | UnixDog | it works fine here |
| 18:55.09 | UnixDog | I have 1.4.19 |
| 18:55.11 | luke-jr | UnixDog: as of 1.4.19 anyhow |
| 18:55.21 | UnixDog | whats broken |
| 18:55.28 | luke-jr | SIP |
| 18:55.32 | UnixDog | nope works fine |
| 18:55.36 | luke-jr | not here |
| 18:55.50 | UnixDog | then you have a build issue |
| 18:55.56 | UnixDog | whats is broken about it |
| 18:56.01 | luke-jr | rebuilt it many times |
| 18:56.06 | luke-jr | it's half-duplex |
| 18:56.12 | UnixDog | did you update all the deps |
| 18:56.18 | luke-jr | the remote party can't hear me |
| 18:56.20 | sione | echo cancle? |
| 18:56.25 | sione | oh |
| 18:56.29 | UnixDog | thats a nat issue |
| 18:56.34 | UnixDog | are you behind a router |
| 18:56.36 | luke-jr | UnixDog: it's a 1.4.19 issue |
| 18:56.42 | UnixDog | no its not |
| 18:56.49 | luke-jr | it works fine with 1.4.18.1 |
| 18:56.49 | UnixDog | I have 1.4.19 working fine |
| 18:56.58 | sione | or both ends using the same codec? |
| 18:57.18 | UnixDog | I have 1.4.19 on over 200 boxes |
| 18:57.22 | UnixDog | and they all work fine |
| 18:57.33 | UnixDog | no issues |
| 18:57.35 | luke-jr | the packet flow looks the same between both |
| 18:57.44 | UnixDog | and I am even usuing freebsd and asterisk |
| 18:57.49 | sione | you sniff on both sides? |
| 18:58.04 | UnixDog | it shound like he has nat issues |
| 18:58.15 | sione | nat/firewall |
| 18:58.15 | luke-jr | sione: I only have control of one side |
| 18:58.40 | sione | or codec missmatch that one side does not support the codec its reciving |
| 18:59.38 | UnixDog | ~sipnat |
| 18:59.39 | 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 |
| 18:59.42 | luke-jr | no NAT involved, BTW |
| 18:59.57 | sione | I never had one way audio issues on any of my asterisk running 1.4 or 1.6 |
| 19:00.01 | UnixDog | make a call |
| 19:00.08 | UnixDog | then do a sip show channels |
| 19:00.16 | UnixDog | and see what codecs are being used |
| 19:00.27 | UnixDog | and that you have support enabled for them |
| 19:01.08 | luke-jr | UnixDog: ulaw |
| 19:01.24 | *** join/#asterisk bminish (n=bminish@rb.brenbox.home.minish.org) |
| 19:01.28 | sione | luke-jr: on both sides? |
| 19:01.52 | luke-jr | I only can see one side |
| 19:02.29 | sione | sip show channels |
| 19:02.31 | sione | will show both |
| 19:02.40 | luke-jr | both channels are ulaw, yes |
| 19:02.47 | sione | ok good |
| 19:03.15 | *** join/#asterisk kyron (n=kyron@modemcable086.140-70-69.static.videotron.ca) |
| 19:03.49 | luke-jr | while there is no NAT involved, it might be worth noting that the calls go ITSP -> Internet -> PBX -> LAN -> PAP2 |
| 19:04.18 | *** join/#asterisk [hC] (n=hardcore@S01060016b6b53c0c.vc.shawcable.net) |
| 19:04.29 | luke-jr | but like I said, the packets all look the same between 1.4.18.1 and 1.4.19, for the most part |
| 19:04.41 | sione | hmmm |
| 19:04.55 | sione | the PAP2 have a routable IP? |
| 19:05.02 | *** join/#asterisk inv_arp (n=junya@c-76-26-24-197.hsd1.fl.comcast.net) |
| 19:05.15 | luke-jr | sione: LAN routable, 192.168.77.11 |
| 19:05.19 | sione | its probaly trying to make an audio connection to the remote end direct |
| 19:05.45 | *** join/#asterisk bmg505 (n=leon@196-209-79-122-tbnb-esr-2.dynamic.isadsl.co.za) |
| 19:05.47 | luke-jr | it shouldn't be |
| 19:05.54 | sione | you have it configured to use outbound proxy as the asterisk server? |
| 19:05.59 | luke-jr | even if something changed in the SIP logic for that, I am using Monitor |
| 19:06.06 | sione | it will if you not forcing it to use outbound proxy |
| 19:06.07 | luke-jr | and Monitor blocks all reinvites |
| 19:06.40 | luke-jr | notes the PAP2 has not changed, only Asterisk has |
| 19:07.05 | luke-jr | Proxy is the * box, Outbound Proxy is blank |
| 19:08.10 | *** join/#asterisk [TK]D-Fender (n=Joe@64.235.216.2) |
| 19:08.40 | sione | you will need the asterisk to nat for the PAP2 |
| 19:08.54 | luke-jr | shouldn't |
| 19:09.02 | luke-jr | didn't need to for 1.4.18.1 or earlier |
| 19:09.07 | sione | the 192.168 IP not going to work on the internet |
| 19:09.16 | luke-jr | the internet should never be talking directly to the PAP2 |
| 19:09.45 | *** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 19:09.45 | jumpie | -- Executing [s@fios-line:1] Dial("Zap/1-1" |
| 19:09.48 | luke-jr | and never tries |
| 19:09.50 | jumpie | why is it saying 1-1 ? |
| 19:09.50 | sione | thahts why you need astreisk to handle all its outgoing calls for it and must be strickly use outbound proxy as the astreisk |
| 19:10.07 | mecredis | why does Asterisk reset the file permissions in /var/lib/asterisk/agi-bin/dir/ ?? |
| 19:10.31 | UnixDog | it doesnt |
| 19:10.43 | mecredis | ok that was helpful |
| 19:10.48 | UnixDog | if your using trixbox/freepbx amportal does |
| 19:10.54 | mecredis | neither |
| 19:11.05 | mecredis | I set chmod 755 to a file in a subdir of agi-bin |
| 19:11.09 | mecredis | and then run something |
| 19:11.11 | jumpie | hey i just did a 20 minute phone call over my fios line into my x100p |
| 19:11.16 | jumpie | it was perfect quality, no delay/echo |
| 19:11.21 | jumpie | does this mean i still should install OSLEC? |
| 19:11.22 | mecredis | then eventually, somewhere, something just resets it to 644 |
| 19:11.32 | [TK]D-Fender | jumpie, Zap/1 is a device. Zap/1-1 is a channel. a given device might be used for multiple simultaneous calls. Hence Zap/1-1 means the first call on device Zap/1 |
| 19:11.33 | sione | if you dont have outbound proxy set on the pap2 it will try to make a direct connection to the remote party |
| 19:11.49 | luke-jr | mecredis: chown it to someone else and see what errors |
| 19:11.54 | mecredis | ok |
| 19:11.56 | jumpie | fender, ah, well since its a 1 port fxo, the most it'll ever be is zap/1-1 then right |
| 19:12.08 | [TK]D-Fender | jumpie, this is only really applicable to Zaptel FXS (phones) whre the phone might be using call-waiting, 3way, etc and thus be on 2 "calls" at a time. |
| 19:12.09 | *** join/#asterisk comprookie2000 (n=david@adsl-065-012-210-216.sip.bct.bellsouth.net) |
| 19:12.13 | sione | lunch time |
| 19:12.15 | luke-jr | sione: it doesn't |
| 19:12.17 | jumpie | gotcha |
| 19:12.29 | jumpie | intersting though, when the call came in, i saw the CID on xlite |
| 19:12.34 | jumpie | but i didnt show in the cli |
| 19:12.39 | mecredis | still getting Permission denied |
| 19:13.11 | jumpie | im apprehensive to get oslec since i seem to be fine |
| 19:13.23 | mecredis | k |
| 19:13.26 | mecredis | chown seems to have worked |
| 19:13.28 | mecredis | thanks luke-jr |
| 19:13.36 | luke-jr | ⦠|
| 19:13.44 | mecredis | ... |
| 19:13.48 | jumpie | oooh, i can now filter a list of knokwn collectors/telemarkters and play the 'this number has been disconnected' message |
| 19:13.49 | jumpie | hahaha |
| 19:14.07 | luke-jr | jumpie: that's boring |
| 19:14.13 | luke-jr | jumpie: go for a torture service |
| 19:14.53 | jumpie | what exactly is entailed in that? :) |
| 19:15.07 | jumpie | there are some peole i want it to sound official so they think im gone and sotp calling |
| 19:15.09 | jumpie | telemarkters i can bug |
| 19:15.19 | *** join/#asterisk fakhir_ (n=fakhir@unaffiliated/fakhir) |
| 19:15.33 | *** join/#asterisk RoyK (n=roy@ip-29-6-149-91.dialup.ice.no) |
| 19:15.37 | jumpie | i have 2 debts i have paid in full, have the letters, have sent them cease and desist letters, filed a FTC complaint, but still call |
| 19:15.41 | luke-jr | jumpie: http://www.voip-info.org/wiki/view/Asterisk+AEL+Telemarketer+Torture |
| 19:15.44 | jumpie | thanks |
| 19:16.10 | luke-jr | jumpie: heh, threaten to sue for your money back? ;) |
| 19:16.12 | *** join/#asterisk obnauticus (n=obnautic@c-67-160-183-109.hsd1.wa.comcast.net) |
| 19:16.39 | jumpie | they dont care |
| 19:16.42 | jumpie | i mean im not worried |
| 19:16.44 | jumpie | they have no recourse |
| 19:16.46 | jumpie | im just tired of them |
| 19:17.05 | luke-jr | jumpie: your credit rating? |
| 19:17.39 | jumpie | its removed from my credit |
| 19:17.48 | [TK]D-Fender | jumpie, record the call, take the callerID, their names, attestations of who they are placing the claim for, etc, and see a lawyer. He'll take the case as its easy money for the winnings. |
| 19:17.51 | jumpie | past statute of limitations and i hired an attorney :D |
| 19:18.05 | jumpie | hmm yea, shit i guess it was a matter of gettin around to it |
| 19:18.06 | jumpie | but your right |
| 19:18.11 | jumpie | fcra violations are like 1k a pop right? |
| 19:18.32 | jumpie | now i have better call accounting with asterisk than a simple caller id screen my wife likes to randomly erase :) |
| 19:18.36 | luke-jr | notes you need to inform them you are recording the call in some states. |
| 19:19.00 | jumpie | is that for residential? or just biz |
| 19:19.40 | luke-jr | everyone |
| 19:19.42 | jumpie | k |
| 19:19.46 | luke-jr | in some states* |
| 19:19.50 | jumpie | well sometimes just calling in general, regardless of what they say, is a violation |
| 19:19.55 | jumpie | but thast a good point |
| 19:20.03 | luke-jr | but so is recording |
| 19:20.13 | luke-jr | and illegal recordings can't be used in court |
| 19:21.31 | jumpie | haha man, reading this script |
| 19:21.33 | jumpie | wonderful |
| 19:21.48 | jumpie | where is the audio though? |
| 19:22.01 | luke-jr | I guess you're supposed to record it |
| 19:22.17 | luke-jr | LOL @ telemarket-exception |
| 19:23.17 | jumpie | haha yeah |
| 19:23.20 | jumpie | just gettin to that |
| 19:23.44 | luke-jr | I love how the political party list has Republicans and Democrats at the very bottom |
| 19:23.48 | *** part/#asterisk planio (n=user@p548F3C5A.dip.t-dialin.net) |
| 19:26.31 | jumpie | it looks lke they have different options for politicla party |
| 19:26.34 | jumpie | you just change the number |
| 19:26.35 | jumpie | lol |
| 19:32.07 | *** join/#asterisk RoyK (n=roy@ip-29-6-149-91.dialup.ice.no) |
| 19:43.12 | jumpie | Hey guys in outbound dialing |
| 19:43.17 | jumpie | is zap case sensitive? |
| 19:43.19 | jumpie | i.e. Zap, ZAP |
| 19:44.02 | jumpie | nm ZAP worked |
| 19:45.53 | ManxPower | jumpie: If you use the same caps as every example of dialing via Zap. you won't go wrong. |
| 19:45.59 | luke-jr | jumpie: 9 for more |
| 19:46.11 | jumpie | well heres my thing |
| 19:46.18 | jumpie | since i have my card now and i have unlimited fios, i use that as primary outbound |
| 19:46.25 | jumpie | but i want it to revert to call with us if thats in use |
| 19:46.39 | jumpie | but yet i cant have 2 outbound rules at the same time diff providers can i |
| 19:46.50 | luke-jr | what? |
| 19:47.06 | jumpie | or would it be like exten, blah, dial(fios), then next line dial (cwu) ?> |
| 19:47.16 | jumpie | clearly truncated |
| 19:47.19 | ManxPower | jumpie: I don't know about "rules" but you can do failover quite easily |
| 19:47.28 | jumpie | i meant like, outbound dialplan |
| 19:47.33 | jumpie | sorry gotta get the terminology |
| 19:47.33 | luke-jr | if ("${DIALSTATUS}" = "BUSY") |
| 19:47.50 | ManxPower | jumpie: You want to check the value of DIALSTATUS after each Dial and determine if you need to try a different dialstring(dest) |
| 19:47.53 | luke-jr | if it's busy, you don't want to try again |
| 19:47.58 | jumpie | but i have callwaiting/3way calling, so if im on the line it won't neccesarily detect it as busy would it? |
| 19:48.08 | jumpie | hm k |
| 19:48.09 | ManxPower | jumpie: you must turn off callwaiting |
| 19:48.11 | luke-jr | jumpie: outbound calls |
| 19:48.29 | luke-jr | if the attempt said the destination line was busy, trying another route is futile |
| 19:48.46 | jumpie | why? thats exactly what i want |
| 19:48.51 | jumpie | if fios = busy, go callwithus |
| 19:49.02 | ManxPower | jumpie: there are TWO KINDS OF BUSY |
| 19:49.09 | jumpie | ok |
| 19:49.31 | ManxPower | If the destination number is BUSY then you don't want to try a different route. If the line is CHANUNAVAIL, then you want to try a different route. |
| 19:49.46 | jumpie | ahah :) |
| 19:49.47 | jumpie | gotcha |
| 19:49.56 | ManxPower | also, you want to check of the line was ANSWERED if so you don't want to call the same number again |
| 19:50.15 | luke-jr | eh |
| 19:50.24 | jumpie | i think i was gettin confused with busy, but yes, i want it to see if zap channel is 'in use' aka chanunvail, to dial sip/callwithus |
| 19:50.27 | luke-jr | if it's answered, Dial terminates the procedure |
| 19:50.29 | jumpie | i see what you're saying |
| 19:58.48 | jumpie | are includes processed top to bottom? |
| 19:59.33 | jumpie | im basically creating a dialplan that falls fios, if unavailable, go to context cwu-outbound |
| 19:59.39 | *** join/#asterisk lmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 19:59.39 | *** mode/#asterisk [+o lmadsen] by ChanServ |
| 19:59.45 | jumpie | but on the sip peers, i have both included in one called fromhome |
| 19:59.48 | jumpie | i wanna be sure it not confused |
| 20:03.22 | [TK]D-Fender | ManxPower, highly unlikely.... |
| 20:04.11 | *** join/#asterisk paci`` (n=paci@cpe-075-182-072-065.nc.res.rr.com) |
| 20:04.12 | paci`` | [Apr 12 20:02:35] NOTICE[81491]: rtp.c:1008 process_rfc3389: Comfort noise support incomplete in Asterisk (RFC 3389). Please turn off on client if possible. Client IP: 216.246.61.6 |
| 20:04.14 | paci`` | :\ |
| 20:04.21 | jumpie | hmm |
| 20:08.29 | paci`` | is there a way i can end someone's call from the panel |
| 20:08.36 | jumpie | hmm i tried to check on ChanIsAvail |
| 20:08.49 | jumpie | then have a goto to another context, but it's still trying to create another zap channel |
| 20:09.24 | lmadsen | paci``: soft hangup |
| 20:09.35 | paci`` | lmadsen, i mean a certain persons |
| 20:09.54 | lmadsen | paci``: ya... end the channel that is associated with who you want to terminate |
| 20:10.21 | paci`` | No such command 'soft end' (type 'help' for help) |
| 20:12.15 | lmadsen | funny how I didn't say 'soft end' |
| 20:12.27 | paci`` | oh |
| 20:12.28 | paci`` | lol |
| 20:12.37 | mvanbaak | lol lmadsen |
| 20:12.41 | lmadsen | :) |
| 20:12.46 | jumpie | fender would you mind lookin at this http://jumpie.pastebin.org/29293 |
| 20:12.54 | jumpie | im close, but i think im missing somethin :) |
| 20:12.58 | paci`` | mm |
| 20:12.59 | mvanbaak | it's like saying 'run this: rm -rf /tmp/*' and they type 'rm -rf /bin |
| 20:13.02 | *** join/#asterisk ectospasm (n=ectospas@c-71-207-229-248.hsd1.al.comcast.net) |
| 20:13.02 | paci`` | how do I find their channel, lmadsen |
| 20:13.08 | lmadsen | show channels |
| 20:13.26 | paci`` | rootwired*CLI> show channels |
| 20:13.26 | paci`` | No such command 'show channels' (type 'help' for help) |
| 20:13.26 | paci`` | rootwired*CLI> |
| 20:13.26 | mvanbaak | snow channels |
| 20:13.34 | mvanbaak | paci``: core show channels |
| 20:13.39 | paci`` | ah |
| 20:13.47 | lmadsen | I guess I assumed you were running 1.4 |
| 20:13.57 | mvanbaak | 1.4 is really old |
| 20:14.00 | lmadsen | lol |
| 20:14.00 | mvanbaak | ;) |
| 20:14.05 | lmadsen | its so last year |
| 20:14.08 | lmadsen | literall :) |
| 20:14.10 | lmadsen | y |
| 20:14.19 | mvanbaak | real men run trunk |
| 20:14.30 | mvanbaak | oh wait |
| 20:14.38 | mvanbaak | real men run team/group/multiparking |
| 20:14.39 | mvanbaak | ;) |
| 20:14.47 | paci`` | lmadsen, what if there are more than one |
| 20:14.50 | paci`` | of the same channel |
| 20:14.51 | luke-jr | s/trunk/freeswitch/ |
| 20:14.52 | luke-jr | runs |
| 20:15.11 | mvanbaak | luke: now that is the dark side of the force |
| 20:15.16 | luke-jr | lol |
| 20:16.24 | mvanbaak | 1.4 is soooooooooooo 2007 |
| 20:16.27 | lmadsen | paci``: then you're probably scuppered |
| 20:16.33 | jumpie | scuppered |
| 20:16.34 | jumpie | ? |
| 20:16.34 | jumpie | <PROTECTED> |
| 20:16.45 | lmadsen | a nice way of saying 'you're fucked' |
| 20:16.46 | *** join/#asterisk pa (n=pa@unaffiliated/pa) |
| 20:16.46 | lmadsen | :) |
| 20:16.58 | mvanbaak | lmadsen: TFOT v3 needs to include multiparking |
| 20:17.10 | mvanbaak | multiple parkinglots configured in features.conf |
| 20:17.12 | lmadsen | mvanbaak: it probably will when its done :) |
| 20:17.19 | lmadsen | (the multiparking) |
| 20:17.20 | mvanbaak | lmadsen: it's close |
| 20:17.28 | lmadsen | that's what they all say |
| 20:17.37 | lmadsen | ok, going offline, gotta move equipment around |
| 20:17.40 | mvanbaak | I fixed the first version, and jpeeler converted it to astobj2 |
| 20:17.51 | drmessano | I have stacked parking lots in a little feature I like to call a "parking deck" |
| 20:19.40 | *** join/#asterisk grEvenX (n=even@pc107-130.ktv.no) |
| 20:19.56 | paci`` | how would i hang up based on caller id? |
| 20:20.02 | mvanbaak | drmessano: in this new setup you can add parking lots in features.conf |
| 20:20.16 | mvanbaak | something like: [parkinglot_mvanbaak] |
| 20:20.28 | mvanbaak | with all the callpark features like numbers etc |
| 20:20.36 | mvanbaak | and in the sip/iax/zap.conf |
| 20:20.36 | drmessano | That's cool |
| 20:20.44 | mvanbaak | you can specify the lot per channel |
| 20:20.52 | mvanbaak | parkinglot => mvanbaak |
| 20:21.14 | mvanbaak | you can overwrite it with a channelvariable |
| 20:21.19 | paci`` | how would i hang up based on caller id, with soft hangup |
| 20:21.20 | *** join/#asterisk FireMac (n=firemac@CPE000d88ae88b9-CM0011ae8bb0ee.cpe.net.cable.rogers.com) |
| 20:21.26 | drmessano | So I can tell the customer "Sorry I lost your call, we had it parked in the wrong lot" |
| 20:21.32 | mvanbaak | exten => blaat,n,Set(PARKINGLOT=blaat) |
| 20:21.34 | drmessano | Sounds like a restaurant I no longer visit |
| 20:21.36 | drmessano | j/k |
| 20:21.46 | [TK]D-Fender | paci``, hangup when? |
| 20:21.50 | drmessano | I see |
| 20:21.53 | jumpie | drmessano, or fender can you help me out? |
| 20:21.55 | jumpie | http://jumpie.pastebin.org/29293 |
| 20:21.56 | paci`` | [TK]D-Fender, I want to terminate a certian users call |
| 20:22.11 | jumpie | is chanisavail being used wrong? its never hitting my goto and tryin to dial another zap |
| 20:22.13 | mvanbaak | drmessano: this will prevent $moron_at_company_a to dial '701' all the time to steal a parked call from $company_b |
| 20:22.18 | *** join/#asterisk solar_ant (n=John@122.164.233.126) |
| 20:22.32 | drmessano | I like that a lot |
| 20:22.45 | paci`` | im not sure how to find his channel id though, [TK]D-Fender |
| 20:22.49 | jumpie | lol @ blaat |
| 20:23.03 | [TK]D-Fender | paci``, You'd make an AGI that would read the list of active channels (via AMI or RX call) and then could issue the hangup based on CID you input, or that you pick through some sort of IVR you'd present |
| 20:23.14 | drmessano | Hate to run.. but going to the in-laws for dinner.. and my delaying this any longer is losing me karma |
| 20:23.20 | drmessano | So.. bbiab |
| 20:23.22 | mvanbaak | paci``: exten => blaat,n,GotoIf($(CALLERID(num)=bar?hangup( |
| 20:23.28 | [TK]D-Fender | drmessano, My karma ran over your dogma... |
| 20:23.31 | paci`` | no, no not like that |
| 20:23.33 | paci`` | we have a conference call |
| 20:23.35 | drmessano | ha |
| 20:23.38 | paci`` | I want to kick someone off |
| 20:23.38 | drmessano | later! |
| 20:23.44 | paci`` | by killing their call |
| 20:23.46 | paci`` | sip show channels |
| 20:23.52 | paci`` | how do I kill a certain one off? |
| 20:23.59 | [TK]D-Fender | paci``, then issue a "soft hangup" |
| 20:24.00 | mvanbaak | paci``: soft hangup <call> |
| 20:24.10 | jumpie | hah |
| 20:24.13 | paci`` | mvanbaak, yeah, but what is <call> |
| 20:24.25 | *** join/#asterisk Iamnacho (i=Iamnacho@ip68-103-153-140.ks.ok.cox.net) |
| 20:24.32 | [TK]D-Fender | paci``, SIP/1000-ef25 <- theres a sampe of a channel. |
| 20:24.50 | paci`` | 66.54.---.-6 646--------- 7d8f76973291c42 0x4 (ulaw) No Rx: ACK |
| 20:24.54 | [TK]D-Fender | paci``, like I said, SCAN the open channels the see which one you want to kill via AMI |
| 20:25.00 | paci`` | an example from 'sip show channels' |
| 20:25.11 | [TK]D-Fender | paci``, "core show channels", not "sip show channels" |
| 20:25.14 | paci`` | ah |
| 20:25.32 | paci`` | [TK]D-Fender, yeah, but that doesnt show based by CID |
| 20:25.50 | paci`` | SIP/66.54.---46-087 1@default:2 Up Dial(SIP/--) |
| 20:25.53 | [TK]D-Fender | paci``, that will get your the channel list. You would take that and look at the details channel by channel. |
| 20:26.03 | paci`` | ah |
| 20:26.09 | [TK]D-Fender | paci``, "show channel [channel]" |
| 20:26.10 | mvanbaak | with 'core show channel <channel>' |
| 20:26.14 | [TK]D-Fender | yup |
| 20:26.21 | paci`` | ah |
| 20:28.19 | jumpie | im a bit confused with this chanisavail |
| 20:28.21 | jumpie | do i need more? |
| 20:31.12 | ManxPower | jumpie: Dude, Dial will set DIALSTATUS to CHANUNAVAIL, no need to use a separate function/app. |
| 20:31.33 | jumpie | oh...hah |
| 20:31.40 | mvanbaak | what ManxPower said ;) |
| 20:31.49 | jumpie | so as long as there is another path to take outbound |
| 20:31.51 | jumpie | itll do it |
| 20:32.00 | jumpie | my concern ManxPower , was how does asterisk know priority? |
| 20:32.07 | jumpie | order i list? |
| 20:32.22 | mvanbaak | dial will continue at the next priority after a dial |
| 20:32.34 | mvanbaak | so put them in the correct order there |
| 20:32.38 | jumpie | ok...ah thanks |
| 20:32.46 | jumpie | man i tried to be slick and tripped myself |
| 20:32.51 | mvanbaak | and if you need, you can add prio which will check the DIALSTATUS var |
| 20:38.39 | jumpie | mvanbaak, its still not working |
| 20:38.44 | jumpie | its still trying to dial out on a 2nd zap |
| 20:39.10 | jumpie | even though my next line specifics to go to another context, which then specifics the sip connection |
| 20:40.25 | jumpie | its like its totaling not even going to the next line |
| 20:40.57 | ManxPower | jumpie: did you gforget priority 1 again? |
| 20:41.21 | ManxPower | jumpie: paste the Goto statement you are using |
| 20:41.49 | jumpie | putting it on pastebin now lol |
| 20:42.26 | jumpie | http://jumpie.pastebin.org/29296 |
| 20:43.18 | jumpie | no priority 1 is good, it seems like its just not getting to the next line, and trying another zap |
| 20:43.21 | *** join/#asterisk dacs (n=haiger@unaffiliated/dacs) |
| 20:44.33 | ManxPower | jumpie: you NEVER Goto a pattern |
| 20:44.40 | ManxPower | you go to a number, a exten pattern will match |
| 20:45.07 | jumpie | hmm really? i was trying to follow this book :L) |
| 20:45.15 | jumpie | how do i push it to cwu-outbound then? |
| 20:45.24 | ManxPower | NOT exten => _1NXXNXXXXXX,100,Goto(cwu-outbound,_1NXXNXXXXXX,1) but exten => _1NXXNXXXXXX,100,Goto(cwu-outbound,${EXTEN},1) |
| 20:45.26 | ManxPower | in this case |
| 20:45.34 | jumpie | ooh |
| 20:45.40 | ManxPower | and stop using numbered priorities! |
| 20:45.54 | jumpie | sorry |
| 20:45.55 | ManxPower | priority 1 is the only numbered priority you want |
| 20:46.05 | jumpie | yea i noticed what happened if you forgot that :) |
| 20:46.08 | ManxPower | if you need to go to a specific priority use a (label) |
| 20:46.34 | jumpie | ooh, so it's still passing the _1NXX....., i was thinking i had to push it to 'that area' of cwu-outbound |
| 20:46.38 | jumpie | thats interesting thank you |
| 20:47.04 | UnixDog | jumie its the weekend relax |
| 20:47.13 | UnixDog | you haev all week to fix issues |
| 20:47.19 | jumpie | lol |
| 20:47.22 | jumpie | im anal |
| 20:47.22 | UnixDog | the weekend is for resting |
| 20:47.33 | jumpie | im about to go see a movie |
| 20:47.38 | jumpie | smart people |
| 20:47.46 | jumpie | not starring jumpie |
| 20:48.18 | jumpie | other than that ManxPower i should be good? |
| 20:48.42 | jumpie | also i cant STAND the gsm buzz from my tmobile phones on every freakin speaker |
| 20:49.09 | jumpie | ManxPower, still failing |
| 20:49.14 | jumpie | it wants to dial out a 2nd zap trunk |
| 20:50.02 | ManxPower | jumpie: I did not look at the rest |
| 20:50.36 | jumpie | ManxPower, nm |
| 20:50.39 | jumpie | i forgot about the priority |
| 20:50.45 | jumpie | i fixed form 100 back to n and it worked |
| 20:51.56 | jumpie | interestingly tho, i still got the error |
| 20:52.02 | jumpie | but i think thats what you have to get, for it to go to the goto |
| 20:52.13 | jumpie | [Apr 12 16:50:08] WARNING[8412]: app_dial.c:1183 dial_exec_full: Unable to create channel of type 'ZAP' (cause 0 - Unknown) |
| 20:52.13 | jumpie | <PROTECTED> |
| 20:52.13 | jumpie | <PROTECTED> |
| 20:52.21 | jumpie | weeeeeeeee |
| 20:52.23 | jumpie | thanks man |
| 20:57.10 | jumpie | what was that calling card app again? |
| 20:57.13 | jumpie | a2accounting or something? |
| 20:59.39 | UnixDog | a2billing |
| 20:59.53 | UnixDog | a2killing |
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| 21:08.57 | paci`` | hey |
| 21:09.01 | paci`` | what do i specify the type= |
| 21:09.03 | paci`` | in sip.conf |
| 21:09.07 | paci`` | for it to register |
| 21:13.23 | jumpie | hmm i want to make it so im not charged for my callwithus connection |
| 21:13.46 | jumpie | i would like to somehow call them back and bridge a call or somethin on my fios line |
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| 21:27.52 | jumpie | ugh a2billing doesnt look super simple |
| 21:27.56 | jumpie | i think ill save it for later :) |
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| 21:41.22 | jumpie | anyone know much about php? |
| 21:41.31 | jbeez | whats php? |
| 21:41.41 | jbeez | :P |
| 21:49.37 | jumpie | personal hygeine proposal |
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| 21:58.06 | paci`` | jumpie, |
| 21:58.07 | paci`` | i KNOW PHP |
| 21:58.09 | paci`` | -caps |
| 21:58.10 | paci`` | i <3 php |
| 21:59.15 | jumpie | i mean i know of it |
| 21:59.24 | jumpie | and it has to do with data extraction, databases, alot of web programming |
| 21:59.28 | jumpie | but what is it 'in a nutshell' |
| 21:59.35 | paci`` | uh |
| 21:59.42 | jumpie | lol |
| 21:59.43 | paci`` | i don't use it for web stuff mutch |
| 21:59.50 | paci`` | I actually have asterisk using it alot |
| 21:59.59 | paci`` | its awesome |
| 22:00.00 | paci`` | thats about it |
| 22:01.14 | jumpie | lol |
| 22:01.19 | jumpie | im tryin to get everything installed for a2biling |
| 22:01.23 | jumpie | but then actualy try to configure it later |
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| 22:15.23 | jumpie | ugh these a2billing instructions are terrible |
| 22:15.26 | jumpie | they are all over the place |
| 22:17.45 | UnixDog | yes they are |
| 22:18.01 | UnixDog | its a pain in the arse to do |
| 22:18.50 | jumpie | also , if english isnt your second language, dont right a technical faq in english |
| 22:18.51 | jumpie | lol |
| 22:18.54 | jumpie | er first |
| 22:19.15 | jumpie | http://trac.asterisk2billing.org/cgi-bin/trac.cgi/wiki/Installation%20Guide is what im going by |
| 22:19.20 | jumpie | its saying mysql this and postgresql that |
| 22:19.25 | jumpie | saying things that are centos related, that arent |
| 22:19.45 | UnixDog | well a2 billing was made to be used with trixbox |
| 22:19.51 | jumpie | now i edited the files it said, and postgresql wont even start now, says failed, but wont tell me why |
| 22:19.52 | UnixDog | and there for it uses mysql |
| 22:19.57 | jumpie | well i wanted to use mysql |
| 22:20.01 | jumpie | and i have those packages |
| 22:21.15 | jumpie | mysqld is running |
| 22:21.17 | jumpie | httpd is running |
| 22:21.33 | jumpie | but its saying postgres stuff in the mysql portion, i think he got sidetracked and started repeating or left somethin out |
| 22:22.03 | paci`` | lol' |
| 22:22.13 | UnixDog | dump all the tables |
| 22:22.14 | paci`` | maybe its just an entire sql section |
| 22:22.16 | *** join/#asterisk inv_arp (n=junya@c-76-26-24-197.hsd1.fl.comcast.net) |
| 22:22.17 | UnixDog | and redo them |
| 22:23.46 | jumpie | im so clueless on this crap |
| 22:23.56 | jumpie | i followed this stuff to a t |
| 22:23.58 | jumpie | and got all the missing packages |
| 22:24.10 | jumpie | i dont know wha tyou mean by dump the tables lol |
| 22:24.23 | jumpie | at this pint i dont even know if its ok that postgresql is failing to start |
| 22:25.15 | *** join/#asterisk qdk_ (n=qdk@195.242.194.42) |
| 22:25.32 | jumpie | and where is the log file that says WHY it failed, so i can be led to wtf is wrong lol |
| 22:40.36 | tzanger | that's awesome... domino's pizza in my area uses asterisk |
| 22:42.52 | Darthclue | tzanger: and just how do you know this? |
| 22:43.00 | tzanger | Darthclue: the hold music, and one of hte prompts |
| 22:45.38 | Darthclue | tzanger: they aren't using the monkeys one are they? or asking what you're wearing? cause i'm not sure i'd want pizza from them if they did. |
| 22:49.15 | tzanger | haha |
| 22:49.17 | tzanger | no |
| 22:49.39 | *** join/#asterisk lesouvage (n=lesouvag@62.140.137.125) |
| 22:50.39 | UnixDog | anyonelooking for a sangoma a200d 2/2 |
| 22:51.31 | Darthclue | likes to ask telemarketers what they are wearing...usually stops the call dead :) |
| 22:51.36 | *** join/#asterisk sun_moon (n=RaviRaja@ip72-206-113-190.om.om.cox.net) |
| 22:53.39 | *** join/#asterisk staralfur15 (n=Viktor@c-67-184-41-58.hsd1.il.comcast.net) |
| 22:54.03 | UnixDog | lol |
| 22:54.19 | UnixDog | likes to ask them what they get into sexualy |
| 22:55.03 | UnixDog | and hear the click |
| 22:55.27 | jbeez | or a moan |
| 22:55.34 | *** join/#asterisk jm|home (n=jm|home@zen.jamiem.com) |
| 22:55.36 | jbeez | then they hear a click |
| 22:55.48 | mwalling | i think that might be illegal... |
| 22:55.55 | mwalling | luid conduct or somethin |
| 22:55.59 | UnixDog | nope |
| 22:56.04 | UnixDog | only on thier side |
| 22:56.11 | UnixDog | they are the on calling |
| 22:56.21 | UnixDog | on/one |
| 22:56.45 | UnixDog | you can be as rude and lude as you want on your phone |
| 22:56.56 | UnixDog | you just cant call osme one a ask that |
| 22:57.12 | UnixDog | I believe |
| 22:57.15 | jbeez | or you can just hang up |
| 22:57.16 | jbeez | heh |
| 22:57.18 | UnixDog | I will double check |
| 22:57.24 | UnixDog | yeah |
| 22:57.53 | UnixDog | but its more fun to give them a taste of what they are doing to you back |
| 22:58.06 | UnixDog | just like them calling me at 9 at night |
| 22:58.52 | UnixDog | I get rude and say look asshole its after 8pm stop calling me |
| 22:59.36 | jbeez | wanna hear something funny? |
| 22:59.45 | Qwell | OR, you could just add your number to the DNC list |
| 23:00.04 | jbeez | I saved a voicemail I got from some telemarketer, they thought they had hung up after going to my vm but they didn't, and you can hear them talking to their co-workers about ways to scam people |
| 23:00.55 | jbeez | http://www.jbeez.net/misc/1-866-297-6734-voicemail.wav |
| 23:01.50 | jblack | click click click |
| 23:02.11 | jbeez | jblack: I thought I typed that for a second |
| 23:02.14 | jbeez | im like, wtf |
| 23:02.18 | jbeez | I didn't say that |
| 23:03.02 | *** part/#asterisk sun_moon (n=RaviRaja@ip72-206-113-190.om.om.cox.net) |
| 23:05.02 | jbeez | hrm, i should really trim that wave down |
| 23:05.07 | jbeez | alot of junk in the front |
| 23:05.14 | jblack | I was expecting something bigger |
| 23:05.24 | jbeez | i think its a "compressed" wav |
| 23:05.30 | Darthclue | is on the DNC list, i still get calls on occassion |
| 23:05.50 | Darthclue | usually from companies who refuse to identify themselves and use fake caller ids |
| 23:05.57 | jblack | I don't get calls, probably because I tell companies that I don't have a phone number. |
| 23:06.28 | jbeez | you hear how they are trying to get people |
| 23:06.35 | jblack | yeah. I caught that |
| 23:06.54 | mvanbaak | I dont get calls because I route everything to voicemail |
| 23:07.25 | *** join/#asterisk Great_Randew (n=Andrew@74.210.4.101) |
| 23:08.34 | jbeez | DND? |
| 23:09.36 | staralfur15 | hey guys |
| 23:09.52 | staralfur15 | i'm a little confused as to what asterisk is capable of doing |
| 23:10.00 | staralfur15 | can anybody answer a few simple questions? |
| 23:10.00 | Darthclue | I have a whole section of numbers that get an immediate disconnect which kills most of the known telemarketer tricks including some of the fake cids they use |
| 23:10.32 | staralfur15 | can asterisk be put on a computer to act as a voip system, like skype, or magic jack or w/e |
| 23:10.42 | staralfur15 | or do you need a seperate voip provider |
| 23:12.24 | mvanbaak | staralfur15: what do you want asterisk to do ? |
| 23:12.31 | tzafrir_home | staralfur15, skype is just a client. It registers with a server |
| 23:12.51 | staralfur15 | i want to be able to call other land lines |
| 23:12.53 | staralfur15 | using asterisk |
| 23:12.57 | tzafrir_home | Asterisk can connect to other servers without an external voip provider |
| 23:12.57 | staralfur15 | and no other service |
| 23:13.03 | staralfur15 | except for high speed internet |
| 23:13.05 | tzafrir_home | You just have to know where to connect to |
| 23:13.25 | tzafrir_home | If you have two servers in two offices - sure - no problem |
| 23:13.31 | staralfur15 | okay |
| 23:13.37 | staralfur15 | so there is no way then |
| 23:13.47 | staralfur15 | using one server |
| 23:13.51 | tzafrir_home | No way to do what? |
| 23:13.58 | staralfur15 | to be able to reach somebody on a cellular phone |
| 23:14.17 | staralfur15 | basically, it is not a replacement for a home telephone service? |
| 23:14.22 | tzafrir_home | to reach a cellular phone you have to connect to the PSTN in one way or the other |
| 23:15.07 | tzafrir_home | There are plenty of providers who will sell you this service through VoIP |
| 23:15.25 | tzafrir_home | (with varying parameters of price, relibility, etc.) |
| 23:15.38 | staralfur15 | so i cannot be my own provider? |
| 23:15.41 | tzafrir_home | You can also connect directly to some PSTN provider |
| 23:15.52 | staralfur15 | sorry i know i must sound stupid |
| 23:15.53 | tzafrir_home | e.g: a PRI line to your telco |
| 23:16.20 | staralfur15 | how expesnive would a PRI line be |
| 23:16.54 | tzafrir_home | You'll have to ask others, I really don't know prices |
| 23:18.01 | staralfur15 | okay |
| 23:18.04 | staralfur15 | thanks for the help |
| 23:20.10 | mwalling | staralfur15: no, asterisk cannot magicly connect to the PSTN, you need a provider |
| 23:21.57 | jbeez | but but, penn and teller said it can |
| 23:22.26 | *** join/#asterisk paci`` (n=paci@cpe-075-182-072-065.nc.res.rr.com) |
| 23:22.28 | paci`` | Hey |
| 23:22.30 | paci`` | question |
| 23:22.41 | paci`` | how would i transfer a certain channel to another extension |
| 23:22.45 | mwalling | staralfur15: you're going to need someone who is connected to the PSTN to bridge your calls (that someone could be you) |
| 23:23.10 | staralfur15 | okay |
| 23:23.11 | paci`` | like how soft hangup kills it |
| 23:23.14 | staralfur15 | so let me ask this |
| 23:23.14 | paci`` | how would i transfer it |
| 23:23.53 | staralfur15 | i pay 30 dollars per month for my telephone service |
| 23:23.59 | staralfur15 | can asterisk cut this cost in any way |
| 23:24.31 | staralfur15 | how do you guys connect to the PSTN |
| 23:24.44 | Qwell | cell phone |
| 23:25.54 | paci`` | is it possible? |
| 23:34.44 | ManxPower | staralfur15: all you need is an ATA and a service provider, no need for Asterisk. |
| 23:35.06 | ManxPower | Unless you LIKE building a working PBX from the Asterisk toolkit. |
| 23:35.39 | staralfur15 | i just misunderstood exactly what its function was |
| 23:37.21 | mwalling | ManxPower: some of us are sadists after all |
| 23:42.21 | cmantito | staralfur15: you can connect to the PSTN through a) a SIP or IAX2 provider who you simply place calls through and htye do the work for you, b) an FXO card, connected to your existing analogue telephone line, c) a PRI card, and a PRI to go with it (most expenive option :P) d) bluetooth -> cellular phone |
| 23:42.36 | cmantito | there are other options as well, in fact, there's almost limiteless options, but those are the "main" choices |
| 23:43.08 | cmantito | if you went with option A, you wouldn't necessarily need asterisk-- you could potentially use an adapter or IP phone with the provider directly. |
| 23:43.33 | jbeez | for 1 phone? |
| 23:43.51 | cmantito | huh? |
| 23:44.03 | jbeez | I pay for a 500minute plan with vonage because I rarely use my home phone, $20/month after all the crap fees and taxes |
| 23:44.43 | cmantito | right |
| 23:46.18 | luke-jr | Vonage is crap |
| 23:46.22 | cmantito | agreed |
| 23:46.32 | UnixDog | vonage is your friend come on |
| 23:46.46 | jbeez | works for me |
| 23:46.46 | UnixDog | just because they are not like other voip providers |
| 23:46.52 | luke-jr | Vonage is no different from AT&T et al |
| 23:46.54 | jbeez | i pick up the phone, i have dialtone |
| 23:47.00 | luke-jr | except they leech off existing networks |
| 23:47.02 | jbeez | i call or get calls, works fine |
| 23:47.06 | UnixDog | and their service is not open like most providers |
| 23:47.15 | cmantito | vonage is my competition ;) |
| 23:47.47 | luke-jr | jbeez: same as with AT&T et al |
| 23:50.46 | UnixDog | at&t / CHarter / TImeWarner / vonage/ verizon/ and loads of other sip/iax providers |
| 23:51.23 | luke-jr | none of those are SIP/IAX providers⦠|
| 23:52.19 | UnixDog | yes they are ou just have to find out what port they use . but most of them are using sip or mgcp |
| 23:52.31 | *** join/#asterisk olinux (n=olinux@ip68-107-9-43.sd.sd.cox.net) |
| 23:52.39 | UnixDog | most I know have sip on port 5069 and 5072 |
| 23:52.52 | UnixDog | at&t uses port 5072 |
| 23:52.54 | olinux | polycom 501 one model says something about wireless handset |
| 23:52.58 | olinux | anyone know about it? |
| 23:53.10 | *** join/#asterisk esaym (n=user@cpe-70-120-89-6.satx.res.rr.com) |
| 23:53.28 | UnixDog | you have to buy a blutooth kit |
| 23:53.41 | UnixDog | it does not have wireless or bluetooth builtin |
| 23:54.18 | olinux | thenerds.net has two 501 models, one is listed as having wireless handset |
| 23:54.47 | UnixDog | they must be including a wireless ket |
| 23:54.49 | UnixDog | kit |
| 23:54.51 | luke-jr | UnixDog: they *use* SIP, they don't provide it |
| 23:55.05 | luke-jr | UnixDog: even if you figured out the port, you'd never figure out the password |
| 23:55.53 | *** join/#asterisk comprookie2000 (n=david@adsl-065-012-210-216.sip.bct.bellsouth.net) |
| 23:56.03 | *** join/#asterisk aaawrekng (n=mark@c-76-121-221-213.hsd1.wa.comcast.net) |
| 23:58.21 | UnixDog | bbiab |
| 23:59.02 | *** join/#asterisk implicit (n=implicit@ip68-105-92-210.sd.sd.cox.net) |
