| 00:00.11 | riddlebox | ManxPower, I dont see anything come across using verbose 10 |
| 00:00.49 | ManxPower | riddlebox: then you have a problem with /etc/asterisk/zapata.conf or with your board or the drivers |
| 00:01.02 | ManxPower | riddlebox: What is the output of "core zap show channels" |
| 00:01.09 | [TK]D-Fender | riddlebox, or quite possibly you built zaptel AFTER Asterisk and have no chan_zap.so |
| 00:01.27 | [TK]D-Fender | riddlebox, perhaps you should show us your configs and the backup that everything else looks fine.... |
| 00:01.28 | [TK]D-Fender | ~pb |
| 00:01.29 | jbot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste |
| 00:01.30 | [TK]D-Fender | ^^^^^^^^^^ |
| 00:02.18 | riddlebox | ManxPower, no such command 'core zap' |
| 00:04.28 | riddlebox | ManxPower, http://pastebin.ca/900551 |
| 00:05.30 | ManxPower | riddlebox: then you did not build zaptel before you built Asterisk |
| 00:06.08 | riddlebox | hrmm I could sworn I did, but I guess the proof is there |
| 00:06.56 | ManxPower | riddlebox: you are running 1.4, right? |
| 00:07.22 | riddlebox | ManxPower, yes |
| 00:09.12 | *** join/#asterisk angom_w (n=angom@200.56.104.87.dsl.dyn.telnor.net) |
| 00:09.31 | *** join/#asterisk EduardoBaret (n=chatzill@201.229.176.252) |
| 00:10.42 | adeel | if i set canreinvite=yes, will * handle dtmf properly? |
| 00:11.14 | *** join/#asterisk angom_w (n=angom@200.56.104.87.dsl.dyn.telnor.net) |
| 00:11.49 | *** join/#asterisk tecnico (n=tecnico@user-24-214-56-217.knology.net) |
| 00:12.44 | ManxPower | adeel: RFC2833 and INFO DTMF yes. |
| 00:13.12 | *** join/#asterisk angom_w (n=angom@200.56.104.87.dsl.dyn.telnor.net) |
| 00:14.27 | *** join/#asterisk lmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 00:14.27 | *** mode/#asterisk [+o lmadsen] by ChanServ |
| 00:15.23 | coppice | don't use RFC2833. its the work of satan |
| 00:16.56 | Robba | Ok what i need to do is run 2 SIP accounts on the One Device |
| 00:17.57 | Robba | eg. i need to have EXT 7626 and 7637 running on 2 seperate lines on the phone |
| 00:20.35 | Robba | so i have configured line one to use 7626 and line four to use 7637 |
| 00:21.00 | Robba | from SIP/7625 to SIP/7626 it works fine |
| 00:21.33 | Robba | from SIP/7625 to SIP/7637 it responds 486 BUSY HERE |
| 00:33.05 | riddlebox | [TK]D-Fender, I did a make uninstall all in the asterisk source directory, the I reinstalled zaptel drivers but I still dont get anything on the zap show channels |
| 00:34.48 | *** join/#asterisk CVirus (n=GoD@82.201.174.232) |
| 00:36.40 | [TK]D-Fender | riddlebox, and you still haven't shown us anything of value. |
| 00:36.54 | riddlebox | ManxPower, http://pastebin.ca/900551 |
| 00:37.04 | riddlebox | [TK]D-Fender, ^^^^^ |
| 00:37.14 | [TK]D-Fender | riddlebox, and you still haven't shown us anything of value. |
| 00:37.16 | [TK]D-Fender | ^^^^^^^^^^ |
| 00:38.02 | [TK]D-Fender | riddlebox, Where are your configs? Where is the output of "ztcfg -vvvv"? Where is the output of your attempt to manually load chan_zap.so? |
| 00:40.44 | tzafrir | riddlebox, what is the output of: cat /proc/zaptel/* # all's well at the kernel level? |
| 00:42.50 | Robba | is it possible to run 2 SIP account on 1 device? |
| 00:43.09 | [TK]D-Fender | Robba, Certainly, depending on the device |
| 00:46.40 | *** join/#asterisk saftsack (n=oliver@p54A7C929.dip.t-dialin.net) |
| 00:46.50 | saftsack | hi where to download asterisk with wget? |
| 00:47.28 | scooby2 | asterisk.org |
| 00:47.49 | adeel | ManxPower, hmmm...well i seem to be having issues when using RFC2833 and INFO while using canreinvite=yes...it doesn't seem to work |
| 00:47.52 | scooby2 | http://downloads.digium.com/pub/asterisk/releases/ |
| 00:47.54 | saftsack | scooby2, downloading there is just possible over a ref link which doesnt work with wget |
| 00:48.11 | scooby2 | which version do you want? |
| 00:48.11 | alrs | there is probably a pristine version in the Debian archives that doesn't require registration |
| 00:48.24 | adeel | ManxPower, does it work on any network config? including being behind a NAT? |
| 00:48.24 | scooby2 | saftsack: you just pull all the crud off the front of the link |
| 00:48.28 | saftsack | i want more than one file so i just need a ftp directory ... a real one |
| 00:48.35 | saftsack | and then the filename directly? |
| 00:48.35 | [TK]D-Fender | saftsack, cut the referral part off and it works just fine |
| 00:48.37 | saftsack | does this work? |
| 00:49.17 | saftsack | works :) |
| 00:49.21 | riddlebox | [TK]D-Fender, http://pastebin.ca/900595 |
| 00:49.25 | scooby2 | :) |
| 00:49.38 | saftsack | so why this ref crap |
| 00:49.56 | scooby2 | not sure |
| 00:50.20 | alrs | saftsack: because Debian shut down their FTP server so that they can collect marketing info on everyone that downloads Asterisk |
| 00:50.43 | alrs | saftsack: I get telemarketing Asterisk calls from their local reseller because of this |
| 00:51.01 | alrs | saftsack: er, I mean digium, not debian |
| 00:51.25 | scooby2 | alrs: i was wondering there for a second |
| 00:51.59 | saftsack | american marketing ^^ |
| 00:52.44 | alrs | http://article.gmane.org/gmane.comp.telephony.pbx.asterisk.user/190484/match=ftp+shut+down |
| 00:52.59 | adeel | is there a way to get low level statistics on a zap channel...e.g. noise levels, etc? |
| 00:54.05 | *** part/#asterisk a1fa (n=a1fa@unaffiliated/a1fa) |
| 00:54.42 | scooby2 | zap show channel # shows a decent amount but not real low level |
| 00:56.26 | adeel | scooby2, it doesn't show the noise level does it? |
| 00:56.52 | phix | hey, how do I stop this message from dispalying? --> Feb 12 11:56:23 WARNING[2439]: dsp.c:1424 ast_dsp_process: Inband DTMF is not supported on codec g729. Use RFC2833 |
| 00:56.53 | tzafrir | downloads.digium.com does not require registration |
| 00:57.02 | tzafrir | It does require manual editing of URLs |
| 00:57.19 | scooby2 | adeel: does not appear so |
| 00:58.11 | adeel | that's unfortunate |
| 00:58.38 | tzafrir | alrs, anyway, as you mentioned Debian, this should give you proper links: |
| 00:58.38 | saftsack | it is smart to compile asterisk without zaptel if i use it just as a pbx without an interface card? (external gateway) |
| 00:58.42 | tzafrir | http://pkg-voip.alioth.debian.org/cgi-bin/digium.cgi?q=asterisk/releases |
| 00:59.02 | phix | [TK]D-Fender: you know the answer don't you? you know, the answer that doesn't require me to recompile asterisk? :) |
| 00:59.14 | riddlebox | saftsack, you will at least need the ztdummy module for timing |
| 00:59.29 | tzafrir | saftsack, you may need zaptel for: (1) meetme, (2) IAX trunking and (3) generally a time source |
| 00:59.31 | saftsack | so i have to compile zaptel ....:/ ^^ |
| 01:00.12 | saftsack | my poor 4gb compact flash card xD |
| 01:00.42 | *** join/#asterisk eric2 (n=nobody@sudbury-cable-69-60-240-185.unitz.ca) |
| 01:01.04 | tzafrir | use a different build machine... |
| 01:03.22 | adeel | saftsack, i have * with zaptel running on a 2gig flash no problems |
| 01:04.15 | saftsack | adeel, this was a joke ;) |
| 01:04.38 | saftsack | but im angry about the poor reading performance (5mb/s) with an sandisk extrem III card |
| 01:04.50 | *** join/#asterisk outtolunc (n=me@c-67-174-216-60.hsd1.ca.comcast.net) |
| 01:05.48 | tzafrir | Well, it's a small disk. Nothing much to read. |
| 01:06.26 | scooby2 | whats the 1.2 equivalent of monitor-type=mixmonitor? |
| 01:06.28 | saftsack | hopefully. but i have just 256mb of ram+ |
| 01:09.53 | J4k3 | saftsack: wtfbbq, those are rated 40MB/sec read! |
| 01:10.06 | J4k3 | saftsack: make sure the kernel is detecting the right IDE modes |
| 01:10.10 | J4k3 | if not, force them |
| 01:10.40 | J4k3 | make sure you're not swapping to CF.. that ain't gonna fly |
| 01:10.54 | saftsack | J4k3, have no swap parition |
| 01:11.50 | Robba | with the 2 SIP account on one device, will each account need to be on another port? eg. 7626 bindport=5060 and 7637 bindport=5061? |
| 01:11.51 | *** join/#asterisk FireMac (n=firemac@CPE000d88ae88b9-CM0011ae8bb0ee.cpe.net.cable.rogers.com) |
| 01:12.07 | scooby2 | Robba: yes |
| 01:12.28 | Robba | ok i'm going with these phones are junk then |
| 01:13.08 | saftsack | J4k3, it seems that just pio3 is activated |
| 01:13.17 | saftsack | pio6, sorry |
| 01:13.50 | *** join/#asterisk ph0ne (n=ph0ne@dsl-207-112-91-102.tor.primus.ca) |
| 01:14.41 | ManxPower | adeel: reinvites do not work with nat |
| 01:15.28 | *** join/#asterisk RypPn (i=TuMbL@rosscom.demon.co.uk) |
| 01:15.31 | *** join/#asterisk nvrpunk (n=root@81.90.21.227) |
| 01:15.45 | nvrpunk | got a question about fixing Jitter |
| 01:15.58 | nvrpunk | having a problem with chop using g729 |
| 01:16.02 | nvrpunk | via IAX |
| 01:16.35 | nvrpunk | set the jitter buffer and at max there is a difference of 80ms jitter. Not often though |
| 01:16.36 | saftsack | J4k3, hdparm -d 1 /dev/hda doesnt help |
| 01:16.56 | nvrpunk | what settings would help? and what codecs deal with jitter better than others |
| 01:17.09 | nvrpunk | also would realtime help? |
| 01:18.50 | riddlebox | [TK]D-Fender, did you get a chance to look at that pastebin? |
| 01:19.36 | *** join/#asterisk r0d3nt (i=nobody@foster.stonedcoder.org) |
| 01:20.30 | *** join/#asterisk masca123 (n=blabla@bd219f80.virtua.com.br) |
| 01:20.58 | masca123 | hi, I'm need with asterix and linux, Where can I setup asterix firewall |
| 01:20.59 | masca123 | ? |
| 01:21.43 | r0d3nt | asterisk isn't a firewall. |
| 01:21.46 | saftsack | my asterisk succesfully starts without the ztdummy module |
| 01:22.10 | *** join/#asterisk Kumbang (n=dsp@167.205.24.69) |
| 01:23.46 | masca123 | I'm trying to make asterix to work in and simple way.. first just with local x-lite phone but when I try to connect show an error 503 - Service unavailable |
| 01:23.51 | masca123 | can you help me? |
| 01:24.02 | riddlebox | saftsack, anything that needs timing depends on the ztdummy or zaptel module |
| 01:24.04 | masca123 | I'm using asterixnow |
| 01:24.25 | saftsack | riddlebox, so the sip protocoll too .... ? |
| 01:25.19 | riddlebox | saftsack, I dont think so, but conferencing and moh and others too |
| 01:25.49 | saftsack | ok so i will compile it in |
| 01:27.05 | *** join/#asterisk emist (n=emist@unaffiliated/emist) |
| 01:27.35 | masca123 | When I boot asterixnow there is a error too, iptables firewall rules failed |
| 01:28.06 | riddlebox | masca123, I believe its asterisknow |
| 01:30.30 | masca123 | But, I red the forums and a lot of people have asterixnow working fine. I'm trying to do the most easy part of setup |
| 01:31.36 | riddlebox | masca123, I think the topic says to try #asterisk-gui for AsteriskNOW |
| 01:31.57 | ph0ne | ~asterisknow |
| 01:31.58 | jbot | extra, extra, read all about it, asterisknow is based on Asterisk, but it is not Asterisk, and it is unlikely to live up to Asterisk's standards. Only Asterisk is supported on #asterisk. Use #AsteriskNow instead. Even if the channel happens to be less helpful, support for systems other than Asterisk is offtopic on #asterisk |
| 01:32.01 | riddlebox | or #asterisnow |
| 01:32.19 | riddlebox | yeah what he said |
| 01:32.34 | Jake[work] | not a he - an it |
| 01:32.50 | masca123 | ok, sorry |
| 01:33.13 | riddlebox | that too |
| 01:40.51 | *** join/#asterisk Frogzoo (n=Frogzoo@CPE-121-218-175-103.nsw.bigpond.net.au) |
| 01:47.03 | *** join/#asterisk tzafrir_laptop (n=tzafrir@62.90.10.53) |
| 01:49.17 | riddlebox | tzafrir, got any ideas on my problem? |
| 01:50.42 | tzafrir | I asked you for th output of a specific command |
| 01:50.51 | saftsack | does someone know this zaptel errormsg? scripts/genksyms/genksyms: No such file or directory |
| 01:52.07 | *** join/#asterisk Telamon (n=telamon@blk-7-252-110.eastlink.ca) |
| 01:52.16 | saftsack | solved make prepare scripts helped |
| 01:52.35 | tzafrir | where do you get that error from? |
| 01:53.01 | Telamon | Does anyone know of repositories/websites for current RPMs of Asterisk? |
| 01:53.09 | Telamon | Google doesn't seem to bring anything up. |
| 01:53.18 | saftsack | from zaptel-1.4.8 . i downloaded the kernel source matching to my kernel but i didnt run make prepare scripts. i just did make prepare |
| 01:53.45 | lmadsen | saftsack: huh? |
| 01:53.48 | tzafrir | saftsack, what distro do you use? |
| 01:53.55 | lmadsen | sounds like a gentoo thing though |
| 01:53.56 | saftsack | ubuntu 7.10 server |
| 01:54.12 | tzafrir | do you run a custom kernel? |
| 01:54.15 | lmadsen | Telamon: compiling isn't very difficult, and there is even documentation :) |
| 01:54.27 | saftsack | no the generic kernel image (via epia) |
| 01:54.50 | tzafrir | apt-get install linux-heaaders-`uname -r` |
| 01:54.54 | Telamon | lmadsen: Yeah, but I'm going freaking nuts trying to build RPMs. I am not a RedHat fan (not my server.) And I want to be able to upgrade/downgrade quickly. |
| 01:55.04 | tzafrir | that's what you need. It's already configured |
| 01:55.37 | tzafrir | Telamon, there are now packages in Fedora |
| 01:55.43 | tzafrir | what distri do you use? |
| 01:56.07 | Telamon | CentOS 5, same as RHEL 5. |
| 01:56.10 | tzafrir | (there are also decent packages in SuSE and PLD, and some contrib packages in Mandriva) |
| 01:56.20 | lmadsen | Telamon: just grab the sources you need for the versions you want to downgrade/upgrade from, compile them all, then whenever you want to switch, just do: rm -f /usr/lib/asterisk/modules ; make install |
| 01:56.23 | saftsack | tzafrir, yes installed them too. but i did the link to the sources. |
| 01:56.33 | lmadsen | as long as you don't run 'make clean', everything will already be compiled |
| 01:56.36 | saftsack | im talking about the /usr/src/linux link |
| 01:56.51 | tzafrir | saftsack, you don't need a link to /usr/src/linux |
| 01:56.56 | Telamon | lmadsen: Hmm, I suppose. |
| 01:57.08 | saftsack | tzafrir, just installing the kernel headers didnt work for me :-P |
| 01:57.14 | tzafrir | what matters is /lib/modules/`uname -r`/build . And it's probably already there |
| 01:57.25 | tzafrir | linux-headers |
| 01:57.26 | lmadsen | Telamon: I don't think you're going to find RPMs unless you roll your own |
| 01:57.51 | Telamon | lmadsen: Yeah, doing that now. RPMs are even worse than ebuilds... |
| 01:58.01 | tzafrir | saftsack, what error do you get with them? |
| 01:58.17 | *** join/#asterisk egypcio (n=vinicius@unaffiliated/egypcio) |
| 01:59.02 | lmadsen | Telamon: really? gentoo makes me puke |
| 01:59.14 | saftsack | actually i get no error but i downloaded the wrong sources *duck* |
| 01:59.49 | Telamon | lmadsen: On a modern day machine, it really runs nicely. Especially if you don't think rpm --force is a good idea. |
| 02:00.05 | lmadsen | I've never had to do rpm --force :) |
| 02:00.51 | tzafrir | or roll your own rpms (or debs) |
| 02:03.38 | saftsack | gn8 |
| 02:04.13 | lmadsen | I've not seen an rpmbuild script unfortunately (and the one in the asterisk repo doesn't work -- bkruse and I tried to get it working one night, but neither of us had the skillz to fix it) |
| 02:05.26 | *** join/#asterisk adjohn (n=adjohn@p5182-ipad71marunouchi.tokyo.ocn.ne.jp) |
| 02:07.15 | *** part/#asterisk nitrus^ (n=nitrus@cpe-76-166-248-27.socal.res.rr.com) |
| 02:07.36 | *** join/#asterisk tzafrir_laptop (n=tzafrir@62.90.10.53) |
| 02:12.07 | *** join/#asterisk javar (n=javar@69.79.134.24) |
| 02:14.40 | *** join/#asterisk Olobola (i=Olobola@207.sub-70-210-181.myvzw.com) |
| 02:15.07 | *** join/#asterisk tzafrir_laptop (n=tzafrir@62.90.10.53) |
| 02:18.07 | Olobola | I am unable to install on my current host's server.. any suggestions? |
| 02:18.24 | Olobola | on a virtual dedicated |
| 02:22.51 | lmadsen | Olobola: I think more informatoin is needed |
| 02:23.09 | lmadsen | have you tried installing a non-root? asterisk probably trying to install to places that it isn't allowed to |
| 02:23.21 | lmadsen | ~thebook |
| 02:23.22 | 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 |
| 02:23.29 | lmadsen | some information about installing as non-root is there |
| 02:25.36 | Olobola | that's the problem. I will try that first, thanks. |
| 02:35.13 | *** join/#asterisk nighty^ (n=nighty@210.188.173.246) |
| 02:37.32 | *** join/#asterisk thomas_newbie__ (n=thomas@CPE0014bf493235-CM00140493ede8.cpe.net.cable.rogers.com) |
| 02:38.17 | riddlebox | can anyone help me get my tdm card working, so far, this is what I have, http://pastebin.ca/900685 |
| 02:45.19 | outtolunc | eject |
| 02:45.24 | outtolunc | whoops <G> |
| 02:48.23 | *** join/#asterisk ming_zym (n=ming_zym@f0-0-tep-rtr1.corp.cnb.yahoo.com) |
| 02:57.45 | *** join/#asterisk ZaVoid (n=zavoid@c-67-165-25-195.hsd1.pa.comcast.net) |
| 02:57.47 | ZaVoid | hey al |
| 02:58.13 | *** join/#asterisk terracon (n=greisky@CPE0050da822b70-CM0012254076d6.cpe.net.cable.rogers.com) |
| 02:58.13 | ZaVoid | i've got a dialplan question... lets say i'm making a dial plan exten => s,n,Read(arg4) |
| 02:58.27 | ZaVoid | AND I'm getting the dtmf issues where aserisk is telling me "user entered nothing" |
| 02:59.05 | ZaVoid | exten => s,n,GotoIf($[${ARG4} = 'xxxxxxx']?macro-startver,s,1) |
| 02:59.13 | ZaVoid | what is xxxx there for user entered nothing? 0? |
| 02:59.15 | b11d | ahh.. finally got faxing solved.. all campus faxes on a Rhino CB connected via T1 and everythings just tits.. |
| 02:59.21 | ZaVoid | can't seem to figure that out |
| 03:03.20 | *** join/#asterisk mtaht4 (n=m@190.212.11.64) |
| 03:03.44 | *** part/#asterisk mtaht4 (n=m@190.212.11.64) |
| 03:03.59 | *** join/#asterisk Olobola (i=Olobola@54.sub-75-208-132.myvzw.com) |
| 03:04.47 | *** join/#asterisk h3x0r (n=hex@65.107.59.98) |
| 03:04.47 | Olobola | configure: error: C++ preprocessor "/lib/cpp" fails sanity check |
| 03:05.08 | Olobola | my virtual account won't let me yum install |
| 03:05.22 | Olobola | any suggestions? |
| 03:08.31 | *** join/#asterisk oldmantyree (n=heath@74-60-94-233.stc.clearwire-dns.net) |
| 03:08.47 | oldmantyree | asterisk just did some crazy time travelling |
| 03:11.22 | oldmantyree | conversations from the past |
| 03:11.33 | oldmantyree | came back from a quantum tunnel |
| 03:11.37 | oldmantyree | i'm serious |
| 03:12.27 | *** join/#asterisk mindCrime (n=chatzill@cpe-065-190-188-124.nc.res.rr.com) |
| 03:12.29 | angryuser[A] | dr Zorg ate your mind |
| 03:14.00 | [TK]D-Fender | ZaVoid, exten => s,n,GotoIf($["${ARG4}"=""]?macro-startver,s,1) |
| 03:14.11 | ZaVoid | thanks fender |
| 03:14.17 | ZaVoid | so "" is equiv to null? |
| 03:14.22 | ZaVoid | i think i'm saying that right? |
| 03:14.30 | [TK]D-Fender | ZaVoid, nope.... |
| 03:14.48 | [TK]D-Fender | ZaVoid, *'s evaluation parser is almost as dumb as dirt |
| 03:15.08 | ZaVoid | oh? |
| 03:15.48 | [TK]D-Fender | ZaVoid, it literally takes the quotes surrounding the variable reference, and compares that to a plain set of quotes. The quotes themselves are counted as part of a text string being compared. |
| 03:16.01 | ZaVoid | really |
| 03:16.25 | ZaVoid | so arg4 is well null so i'm basically double quoting null.. i'm saying that wrong but i see what your saying |
| 03:16.35 | ZaVoid | and " no ' ? |
| 03:16.51 | ZaVoid | exten => s,n,GotoIf($[${DNISYES} = '2']?macro-fenderrox,s,1) |
| 03:16.56 | ZaVoid | i have that now.. single ' |
| 03:17.00 | [TK]D-Fender | ZaVoid, See because it takes everything literally if you did this : GotoIf($[${ARG4}=""]?macro-startver,s,1) : it would compare (BLANK) to "" which is no good. also not good is that there would be NOTHING on the left of the = sign |
| 03:17.03 | ZaVoid | bah dog is barking brb |
| 03:17.23 | [TK]D-Fender | ZaVoid, exten => s,n,GotoIf($[${DNISYES} = '2']?macro-fenderrox,s,1) <- bad |
| 03:17.33 | [TK]D-Fender | ZaVoid, exten => s,n,GotoIf($["${DNISYES}" = "2"]?macro-fenderrox,s,1) <- good |
| 03:18.50 | [TK]D-Fender | ZaVoid, because DNISYES *night* be blank. the surronding quotes have nothing to do with trying to represent either side of this comparison as "text", but rather just to ensure that there is SOMETHING on each side of the = side so *'s stupid parser doesn't have a shit-fit |
| 03:22.00 | riddlebox | [TK]D-Fender, did you get a chance to look at that pastebin? |
| 03:22.20 | *** join/#asterisk puzzled (n=patrick@puzzled.xs4all.nl) |
| 03:22.38 | [TK]D-Fender | riddlebox, you showed everything but the last attempt to load chan_zap, and your dialplan, and call attempt |
| 03:23.09 | riddlebox | [TK]D-Fender, nothing happens when there is a call attempt |
| 03:23.17 | ZaVoid | so why is '2' bad? because it could be blank.. thats why right? |
| 03:23.26 | [TK]D-Fender | riddlebox, pastebin everything and include core debug 10, verbose 10 |
| 03:23.40 | [TK]D-Fender | ZaVoid, you need '' on the LEFT side too, thats why |
| 03:23.56 | [TK]D-Fender | ZaVoid, the chars are LITERALLY counted into the comparison |
| 03:23.58 | riddlebox | I thought I pasted the load chan_zap.so in there, and the dialplan is the demo that is included from make samples |
| 03:24.11 | ZaVoid | ahh well my way is working now if 1 or 2 is entered.. not on blank though :) |
| 03:24.15 | [TK]D-Fender | riddlebox, I will not accept an assumption when things are not working. |
| 03:28.57 | ZaVoid | so this would be right then fender for user entered nothing? s,n,GotoIf($["${DNISYES}" = ""] |
| 03:29.06 | ZaVoid | with macro after it.. blah blah |
| 03:30.32 | ZaVoid | I CAN TAKE out the spaces i have in there too right? |
| 03:31.00 | [TK]D-Fender | ZaVoid, Yes, I generally suggest it as well |
| 03:31.28 | [TK]D-Fender | ZaVoid, *'s low-end parsing means extra white-space will usually come back to bite you in the ass |
| 03:32.19 | ZaVoid | gotcha |
| 03:32.24 | ZaVoid | thanks again man |
| 03:32.29 | ZaVoid | gonna go test it now |
| 03:36.36 | *** join/#asterisk [hC] (n=hardcore@S01060016b6b53c0c.vc.shawcable.net) |
| 03:37.03 | *** join/#asterisk Wayhigh (i=noid@www.kevinlynn.com) |
| 03:41.42 | Frogzoo | trying to build a deb for 1.6b2, and for some reason, it wants to write to /usr/lib/pkgconfig/asterisk.pc ? Makefile bug ? |
| 03:42.23 | riddlebox | [TK]D-Fender, http://pastebin.ca/900730 |
| 03:43.17 | [TK]D-Fender | riddlebox, all of your other configs again please, and "zap show channels". |
| 03:43.31 | *** join/#asterisk Igbothom_III (n=Hilton@office.quarkgroup.com.au) |
| 03:43.46 | *** join/#asterisk timeshell (n=Khoja@206.248.136.108) |
| 03:44.10 | riddlebox | [TK]D-Fender, http://pastebin.ca/900685 |
| 03:46.21 | [hC] | any idea what would cause asterisk to acknowledge dtmf no problem, until trying to use DISA, which then keeps playing dialtone infinitely no matter if you dial dtmf or not? ( Using a Zap channel ) |
| 03:47.00 | Wayhigh | any idea why this thing wouldn't see the soundfile directory "[Feb 11 22:45:34] WARNING[12567] file.c: File silence/1 does not exist in any format |
| 03:47.32 | Wayhigh | It's bizarre.. the files are definitely in /var/lib/asterisk/sounds/silence and it's just not seeing them |
| 03:47.48 | Wayhigh | permissions are readable.. owned by the asterisk user.. |
| 03:48.08 | *** join/#asterisk [TK]D-Fender (n=Joe@64.235.216.2) |
| 03:48.14 | [TK]D-Fender | ouch |
| 03:49.03 | [TK]D-Fender | riddlebox, and I said "zap show channels", not "zap show status" |
| 03:51.06 | riddlebox | [TK]D-Fender, http://pastebin.ca/900740 |
| 03:51.57 | [TK]D-Fender | riddlebox, try "module reload chan_zap.so" then repear |
| 03:51.59 | [TK]D-Fender | repeat* |
| 03:55.50 | riddlebox | [TK]D-Fender, http://pastebin.ca/900745 |
| 03:56.24 | *** join/#asterisk b1shop (n=b1shop@c-24-7-202-70.hsd1.il.comcast.net) |
| 03:57.37 | [TK]D-Fender | riddlebox, Ok, something is just plain wrong... not sure what. Go recompile * |
| 03:58.03 | riddlebox | [TK]D-Fender, I have, I did a make uninstall-all then downloaded 1.4.18 and installed that version |
| 03:58.40 | *** join/#asterisk g0mb0 (n=root@external.micom.mng.net) |
| 03:58.41 | [TK]D-Fender | riddlebox, Ok, if you've done that after setting up zaptel completely I'm at a bit of a loss.... |
| 03:58.42 | ZaVoid | so fender if i want to set the channel language from the setting in sip_buddies i would use.. exten => s,n,Set(CHANNEL(language)=${LANG}) ? except thats not working right |
| 03:59.11 | ZaVoid | i never use zaptel riddlebox or i'd try to help too sorry :( |
| 03:59.18 | [TK]D-Fender | ZaVoid, that will set it based on that var, but I have no idea what "sip_buddies" is that you are referring to. |
| 03:59.26 | ZaVoid | usign realtime |
| 03:59.39 | ZaVoid | i have the realtime fields setup in my db |
| 03:59.47 | [TK]D-Fender | ZaVoid, Ok, can't help you much there, go look at the actual CLI executed. |
| 03:59.54 | ZaVoid | k |
| 04:00.02 | g0mb0 | Hi, I have some patches made for chan_ooh323 (asterisk-addons-1.4.5) which is compatible when codec negotiation patch applied to asterisk-1.4.7 |
| 04:00.17 | ZaVoid | exten => s,n,Set(CHANNEL(language)=en) works for english... i'll check the var |
| 04:00.22 | g0mb0 | can somebody check it for me whether it is correct or not? |
| 04:00.45 | g0mb0 | http://lists.digium.com/pipermail/asterisk-users/attachments/20080212/7ec8a960/attachment.obj |
| 04:00.53 | g0mb0 | http://lists.digium.com/pipermail/asterisk-users/attachments/20080212/7ec8a960/attachment-0001.obj |
| 04:01.10 | riddlebox | [TK]D-Fender, I guess I will just reload, everything tomorrow and see if that helps, and if that doesnt then maybe the card has a problem |
| 04:01.42 | riddlebox | [TK]D-Fender, you know what, my boss had the power adaptor plugged into it, would that cause a problem if there are 4 fxo ports in the card? |
| 04:01.58 | [TK]D-Fender | riddlebox, shouldn't |
| 04:02.06 | [TK]D-Fender | riddlebox, go verify the LINES |
| 04:02.44 | riddlebox | yeah I have, before I left work I made sure that it was ringing |
| 04:02.55 | [TK]D-Fender | riddlebox, though the fact that "zap show channels" DOESN'T show anythign concerns me. I can't seem to find something misconfigured |
| 04:03.14 | riddlebox | yeah I pretty much copied what I have here at my house |
| 04:03.21 | riddlebox | from one that works |
| 04:14.45 | sweeper | [TK]D-Fender: so who's your current favorit for CONUS termination? I need some decent prices and some redundancy |
| 04:15.08 | ManxPower | riddlebox: what happens when you do a "unload chan_zap.so" then "load chan_zap.so" You might need "core" in front of the "load" |
| 04:15.21 | *** join/#asterisk jameswf-home (n=james@ip72-204-221-181.ph.ph.cox.net) |
| 04:16.56 | riddlebox | ManxPower, still nothing |
| 04:17.31 | jameswf-home | ~idnms |
| 04:17.31 | jbot | Why would a Wookiee, an eight-foot tall Wookiee, want to live on Endor, with a bunch of two-foot tall Ewoks? That does not make sense! But more important, you have to ask yourself: What does this have to do with this case? Nothing. |
| 04:17.50 | riddlebox | ManxPower, the only thing is that maybe the ring voltage is not strong enough, it is coming from an analog station on a pbx? |
| 04:18.35 | *** join/#asterisk Robba (n=rob@203.56.181.15) |
| 04:19.22 | *** join/#asterisk Maxous (n=Maxous@76.97.3.24) |
| 04:20.00 | *** join/#asterisk WilliamK (n=noc@static-71-170-144-3.dllstx.fios.verizon.net) |
| 04:20.38 | *** join/#asterisk hmodes (i=hmodes@B1-66ER.matrix.gs) |
| 04:24.43 | sweeper | ManxPower: so I'm looking for termination under 2 cents a minute, I've got les.net, but I want a backup option for some redundancy |
| 04:25.23 | WilliamK | hiya sweepy |
| 04:25.25 | WilliamK | :) |
| 04:26.54 | sweeper | WilliamK: why hello thar |
| 04:30.38 | J4k3 | sweeper: termination can be had for mad cheap |
| 04:31.25 | sweeper | J4k3: yea, that's what I'm looking for. I know l3 does permination, I also know you need to buy a fuckton of minutes and they don't list actual prices on their site |
| 04:31.40 | sweeper | les.net is .15 cents with no minimums |
| 04:31.44 | sweeper | 1.5 |
| 04:32.43 | [TK]D-Fender | sweeper, voicepulse connect isn't too bad |
| 04:33.16 | pkunkra | nice. my provider offers E911 services now. |
| 04:33.30 | pkunkra | i should send them a ticket asking for it. |
| 04:34.02 | sweeper | oh that IS nice |
| 04:34.08 | hmodes | imo arrange a test call with your psap if you actually plan to rely on it in an emergency ;p |
| 04:34.26 | pkunkra | right |
| 04:34.53 | pkunkra | i have to call it. and tell them i'm just testing the 911 services. |
| 04:35.01 | pkunkra | yeah, don't hang up if you do that. |
| 04:35.04 | pkunkra | they'll call you back |
| 04:35.12 | hmodes | don't just call them blind |
| 04:35.19 | sweeper | we're currently using delta3, but they're assholes and have no device visibility, and charge too much, so boss said "find bulk minutes, install gateway" |
| 04:35.24 | hmodes | look up the local admin number and arrange a time to call |
| 04:35.30 | pkunkra | what do you mean, don't call them blind? |
| 04:35.48 | pkunkra | local admin number? |
| 04:35.52 | hmodes | if you just dial 911 out of the blue, you risk preempting actual emergencies |
| 04:35.56 | pkunkra | oh |
| 04:36.02 | *** join/#asterisk L|NUX (i=L_NUX@unaffiliated/lnux/x-10290) |
| 04:36.05 | pkunkra | oh, in case the call fails. |
| 04:36.07 | *** join/#asterisk angryuser (i=nononon@df01t2-212-194-216-180.d4.club-internet.fr) |
| 04:36.09 | jql | yeah, every psap has an "office" number |
| 04:36.11 | hmodes | you can look up the local office number and call them on a non-emergency line to arrange a time for a test call |
| 04:36.20 | pkunkra | ok |
| 04:36.44 | hmodes | they'll generally be able to tell you when their slowest time is, and the operator will be _much_ less annoyed if you immediately tell them you're placing a scheduled test call :) |
| 04:37.05 | pkunkra | haha |
| 04:37.06 | pkunkra | ok |
| 04:37.27 | pkunkra | i wonder if they get a lot of those? |
| 04:37.35 | pkunkra | considering there aren't very many emergencies. |
| 04:37.36 | hmodes | not enough ;p |
| 04:37.46 | pkunkra | i've only dialed 911 once in my life. |
| 04:37.47 | hmodes | you'd be surprised |
| 04:38.23 | hmodes | being a 911 operator does not pay particularly well in most cases ;p |
| 04:38.36 | *** join/#asterisk ming_zym (n=ming_zym@f0-0-tep-rtr1.corp.cnb.yahoo.com) |
| 04:38.41 | hmodes | if one thing goes significantly wrong, they can get flooded fast |
| 04:38.41 | ZaVoid | exten => s,n,Macro(fenderrules,s,10) <-- this should send my dialplan to step 10 of fenderrules macro right? |
| 04:38.43 | jql | there's a reason for that |
| 04:38.49 | jql | they work 20+ hours of overtime |
| 04:38.56 | jql | their hourly takes that into account. :) |
| 04:39.27 | hmodes | still, it's not like there's massive callcenters waiting for you to order something ;p |
| 04:39.39 | [TK]D-Fender | ZaVoid, Nope.... that'd be the [fenderrules] context. |
| 04:39.49 | ZaVoid | err duh |
| 04:39.56 | pkunkra | hmmm. |
| 04:40.04 | ZaVoid | wait |
| 04:40.05 | pkunkra | you figure there wouldn't be a wait. |
| 04:40.13 | pkunkra | actually, i wonder if 911 has a queue |
| 04:40.20 | ZaVoid | exten => s,n,Macro(fenderrules) drops me right into the start of fenderrules though |
| 04:40.28 | jql | there is a queue |
| 04:40.35 | jql | I've been put in the queue |
| 04:40.42 | pkunkra | wow |
| 04:40.47 | pkunkra | how long was the wait? |
| 04:41.02 | hmodes | yeah, all the more reason to keep the unnecessary calls to a minimum ;p |
| 04:41.06 | hmodes | being in queue for 911 _sucks_ |
| 04:41.07 | jql | waited 2 minutes and gave up. the highway patrol handles all cell-phone 911 calls in cali |
| 04:41.08 | jameswf-home | Thankyou for calling 911 please hold and a representitive will help you shortly |
| 04:41.11 | jql | in a central office |
| 04:41.17 | [TK]D-Fender | ZaVoid, thats because it pushes the args into the vars, and then does a goto to "macro-[themacronamespecified]" |
| 04:41.21 | jql | so it's easy for a queue to form |
| 04:41.30 | ZaVoid | ahh |
| 04:41.36 | jameswf-home | if you are being murdered press 1 |
| 04:41.41 | pkunkra | hahahahaa |
| 04:41.42 | pkunkra | 1 |
| 04:41.45 | [TK]D-Fender | ZaVoid, and also applies rules so that when "s" runs out, it goes back |
| 04:41.53 | jql | if you're being murdered, use a land-line for immediate response |
| 04:42.10 | pkunkra | *attempts to get to landline.... dies in the process* |
| 04:42.12 | jameswf-home | so you are being murdered if this is correct press 1 to go back press 2 |
| 04:42.13 | jql | your cell is being redirected to.... /Sacramento/ ... please stay on the line |
| 04:42.17 | ZaVoid | yah but its not going back... i jumped into a macro to play some stuff from a a GoToIf statement |
| 04:42.23 | ZaVoid | but now its not returning to the original macro |
| 04:42.25 | jameswf-home | you selected o back are you sure |
| 04:42.34 | hmodes | 'i couldn't understand you, did you say 'HELP!'?' |
| 04:42.43 | [TK]D-Fender | ZaVoid, you should not jump into or out of macros...... |
| 04:42.53 | ZaVoid | ? |
| 04:42.56 | ZaVoid | why? |
| 04:43.01 | ZaVoid | the 7 macro limit? |
| 04:43.07 | hmodes | 'i'm sorry, i still couldn't understand you, did you say 'OMG HE'S GOT A GUN!'? |
| 04:43.30 | [TK]D-Fender | ZaVoid, it'll mess stuff up when you end up running out of "s" in another context and stuf starts acting wierd. |
| 04:43.32 | ZaVoid | exten => s,n,GotoIf($["${DNISYES}" = "2"]?macro-nodtmf,s,5) |
| 04:43.34 | jameswf-home | Sorry we couldnt understand you call back later |
| 04:43.36 | *** join/#asterisk MrTelephone (n=test@S010600195b946fa3.ls.shawcable.net) |
| 04:43.40 | [TK]D-Fender | ZaVoid, these are things you just don't do. |
| 04:43.43 | ZaVoid | lol |
| 04:43.50 | ZaVoid | could you reccomend a better way to do this then? |
| 04:43.54 | [TK]D-Fender | ZaVoid, you are asking for trouble... |
| 04:43.59 | jameswf-home | ~e911 |
| 04:44.01 | ZaVoid | basically the thing we did earlier |
| 04:44.09 | MrTelephone | has anyone yet experiences brief cross channel talk when you try and dial out on a busy system? |
| 04:44.12 | [TK]D-Fender | ZaVoid, pastebin the whole big picture so I can get a better idea of what you're trying to butcher :p |
| 04:44.18 | ZaVoid | yyeah |
| 04:44.59 | jameswf-home | ~e911 is <reply> for generic 911 test info visit: http://www.vtwiki.org/wiki/index.php/E-911_Information |
| 04:45.00 | jbot | okay, jameswf-home |
| 04:45.31 | ZaVoid | hmm i got an idea |
| 04:47.45 | ZaVoid | bah it doesn't like me jumping out of macro a to macro b and back to a lol |
| 04:50.15 | jameswf-home | I think I will write an e911 article on voipinfo |
| 04:50.40 | MrTelephone | how come? |
| 04:50.51 | Nugget | http://macnugget.org/carrera/ <-- my e911 |
| 04:50.52 | chavigny | :> |
| 04:51.26 | sbingner | that looks liek a p911 to me |
| 04:51.26 | MrTelephone | my friend bought a cadillac escalade for 31k |
| 04:51.29 | MrTelephone | its from the states |
| 04:51.31 | Nugget | heh |
| 04:51.37 | MrTelephone | is it one of those ones in hurricane katrina? |
| 04:52.56 | hmodes | my question is what is anyone in their right mind doing importing an escalade? |
| 04:53.08 | Nugget | ew |
| 04:53.48 | MrTelephone | no idea |
| 04:53.49 | MrTelephone | the price |
| 04:54.46 | ZaVoid | Nugget: |
| 04:55.01 | ZaVoid | your car depreciates 2018 dollars a month? |
| 04:55.12 | *** join/#asterisk adjohn (n=adjohn@219.106.248.145) |
| 04:55.23 | hmodes | worth every penny. |
| 04:55.36 | MrTelephone | its a 2006 |
| 04:55.50 | MrTelephone | they are running 44k average but im sure you can get down to 44 |
| 04:56.27 | ZaVoid | i thought they usally apreciate.. or depreciate very little |
| 04:56.56 | MrTelephone | first 2 years is the worst |
| 04:57.17 | J4k3 | escalades are eh |
| 04:57.23 | MrTelephone | if you leased it your looking at 40% of the value x taxes / 36 months |
| 04:57.25 | J4k3 | shipping crate with plush seats |
| 04:57.25 | drmessano | Who the ef wants an Escalade |
| 04:57.27 | ZaVoid | my buddy has a 84 1/2.. or 85 1/2? which year had the 1/2 |
| 04:57.33 | ZaVoid | we call it the poooorsche |
| 04:57.41 | ZaVoid | but hes slowly restoring it |
| 04:57.59 | MrTelephone | i took it for a spin an its an amazing ride |
| 04:58.01 | drmessano | Escalades are ugly, expensive excuse for a suburban |
| 04:58.02 | J4k3 | I like listening to the tires scream pushing through 100+ |
| 04:58.32 | drmessano | All an Escalade screams is "Look at me, I have a small penis" |
| 04:58.35 | MrTelephone | we couldn't get the tires to spin on ice with 400 horsepower |
| 04:59.21 | J4k3 | welcome to traction control. |
| 04:59.28 | hmodes | because the entire thing is designed to be slushy and remove you from any actual involvement with the road ;p |
| 04:59.31 | J4k3 | and if the torque converter stalls at 1800 rpm |
| 04:59.35 | J4k3 | you're getting about ~80 ho |
| 04:59.36 | J4k3 | er hp |
| 05:00.01 | drmessano | My KIA doesn't slip on ice, can I win now? |
| 05:00.13 | J4k3 | my hyundai slips when I ask it to |
| 05:00.23 | J4k3 | my dodge is more fun, though. |
| 05:00.28 | MrTelephone | the buttons probably fell off your deck too |
| 05:00.31 | jql | your KIA probably slips in a stiff wind |
| 05:00.46 | drmessano | Naah |
| 05:00.49 | MrTelephone | kick the fuse panel a couple times, it will start |
| 05:01.12 | MrTelephone | make sure you buy our 12/litre kia oil or you void your warranty |
| 05:01.16 | drmessano | KIA's rock |
| 05:01.45 | Frogzoo | would anyone have a sample debian/rules for packaging asterisk? |
| 05:01.54 | J4k3 | heh, girlfriend's SLS has a couple broken switches, etc... my crappy santa fe still has all its interior parts at the same age and mileage. |
| 05:02.02 | *** part/#asterisk Maxous (n=Maxous@76.97.3.24) |
| 05:02.13 | drmessano | You get a KIA made in the last 3 or 4 years and it's just as good as anything coming out of Japan |
| 05:02.15 | J4k3 | but you won't find yourself at 100 mph in zero time in the hyundai |
| 05:02.25 | Nugget | ZaVoid: yeah, buying new is a bitch. :) |
| 05:02.37 | ZaVoid | i like my mustang convertible |
| 05:02.37 | Kobaz | how would i go about getting a serial port bound to a pri channel.... ie: dial into a serial console via a did on the pri |
| 05:02.43 | MrTelephone | your street racing in ontario if your doing more than 90mph |
| 05:02.46 | ZaVoid | gets me from point a to b looking decent and having my top down :) |
| 05:03.01 | ZaVoid | nothing fancy but a convertbile :) |
| 05:03.03 | J4k3 | MrTelephone: in texas, you automatically can take defensive driving to dismiss a ticket for anything under 90 mph |
| 05:03.06 | drmessano | Mustangs have no room for my tools lol |
| 05:03.08 | J4k3 | above 90 you must speak to the judge |
| 05:03.20 | J4k3 | friend got a ticket for 149 in a 65 (mph!) and didn't even go to jail |
| 05:03.24 | ZaVoid | drmessano: thats why my friends have trucks.. to get my stuff from homedepot and carry my stuff haha |
| 05:03.27 | pkunkra | hah |
| 05:03.27 | Nugget | convertible > * |
| 05:03.30 | drmessano | HA! |
| 05:03.35 | pkunkra | 149 is definate jail. |
| 05:03.36 | MrTelephone | oh well |
| 05:03.44 | pkunkra | that's a misdemeanor. reckless driving |
| 05:03.48 | MrTelephone | im tired of people crashing because they drive to fast and kill people |
| 05:03.59 | drmessano | Forget fast |
| 05:03.59 | J4k3 | pkunkra: well, his uncle was also head of the TXDOT |
| 05:04.01 | drmessano | Cell phone users |
| 05:04.09 | pkunkra | oh... |
| 05:04.10 | J4k3 | word. |
| 05:04.11 | MrTelephone | cell phone users, woman doing make up |
| 05:04.14 | pkunkra | i agree with drmessano. |
| 05:04.20 | Nugget | In texas it's only reckless driving at officer's discretion. |
| 05:04.25 | pkunkra | cell phones kill more than fast drivers do... |
| 05:04.26 | J4k3 | yep |
| 05:04.41 | J4k3 | left lane campers kill more people than speeders |
| 05:04.44 | J4k3 | cellphone users for sure |
| 05:04.45 | drmessano | WUT DO U MEAN I CANUT EATS MY CHEEZBURGER AND TXT WILE DRIVERING MR POLICEMANBADGEGUY? |
| 05:04.45 | MrTelephone | up here in the north you got morons running slicks in the winter |
| 05:04.46 | pkunkra | however..... if i see someone driving fast... and ... talking on their cell. i will shoot them myself. |
| 05:04.59 | Nugget | in texas that's justifiable homicide. :) |
| 05:05.22 | MrTelephone | speaking of texas how come most of the First 48 episodes are filmed in dallas |
| 05:05.26 | sbingner | pkunkra, why? I do that all the time |
| 05:05.28 | MrTelephone | do lots of people get kille there? |
| 05:05.42 | J4k3 | dallas is a shithole |
| 05:05.49 | sbingner | pkunkra, cell phones don't make bad drivers... bad drivers make bad drivers |
| 05:06.02 | J4k3 | good drivers don't talk on the goddamn cellphone |
| 05:06.02 | drmessano | J.R. EWING ruined that town |
| 05:06.05 | Nugget | http://macnugget.org/photos/tws200711/DSC_1672 <-- texas |
| 05:06.08 | J4k3 | thats why god made voicemail |
| 05:06.31 | J4k3 | porch |
| 05:06.46 | hmodes | i dunno, i'm not so sure about the whole 'don't mess with texas' thing... |
| 05:06.53 | sbingner | J4k3, psh... it doesn't take two hands to drive |
| 05:06.55 | J4k3 | it was a warning to michael jackson |
| 05:06.55 | *** part/#asterisk MACscr (n=Mark@adsl-75-23-68-162.dsl.peoril.sbcglobal.net) |
| 05:07.01 | hmodes | i've messed with texas a lot, and they mostly seem like friendly and accomodating folks ;p |
| 05:07.13 | J4k3 | sbingner: MT equipped vehicles it certainly does. |
| 05:07.21 | drmessano | God invented Asterisk so I can open Outlook on my laptop with an Aircard and listen to an MP3 voicemail sent to me instead of doing something stupid like dialing in on my cell |
| 05:07.30 | drmessano | oh, wait |
| 05:07.33 | J4k3 | drmessano: word. |
| 05:07.35 | sbingner | J4k3, generally I agree there.. but most have autos |
| 05:07.47 | sbingner | J4k3, but I've driven manual with one hand... just have to steer with a knee :p |
| 05:07.49 | J4k3 | sbingner: people who drive autos are usually too asleep to pay attention to two things at once. |
| 05:08.05 | sbingner | but that's just when you have to shift |
| 05:08.10 | sbingner | lol |
| 05:08.14 | ZaVoid | thanks again [TK]D-Fender ! |
| 05:08.15 | J4k3 | sbingner: we've all done it, but its most certainly not as safe as driving without the bs distraction |
| 05:08.21 | sbingner | *nod* |
| 05:08.43 | jameswf-home | ~no e911 is <reply> for 911 info visit http://www.fcc.gov/pshs/services/911-services/ |
| 05:08.44 | jbot | jameswf-home: okay |
| 05:08.44 | sbingner | but cell phone on auto trans, to me is about the same as walking and talking |
| 05:08.48 | sbingner | anyway |
| 05:09.03 | drmessano | ~112 |
| 05:09.07 | drmessano | :( |
| 05:09.10 | jameswf-home | ~420 |
| 05:09.11 | jbot | extra, extra, read all about it, 420 is something some stoner made up |
| 05:09.25 | Olobola | ~thebook |
| 05:09.26 | 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 |
| 05:09.37 | J4k3 | ~omgwtfbbq |
| 05:09.38 | jbot | i heard omgwtfbbq is Oh man guys, where's the freakin' barbeque? |
| 05:10.08 | drmessano | J4k3: Cellphones should only work in an approved cradle or holder at +40MPH or with a headset |
| 05:10.15 | J4k3 | drmessano: yep |
| 05:11.12 | sbingner | I rather enjoy calling a friend to let them know I'm doubling the speed limit... ;) |
| 05:11.14 | sbingner | j/k |
| 05:11.25 | J4k3 | thats when you break out the cellcam |
| 05:11.29 | J4k3 | and start filming footage |
| 05:11.36 | hmodes | my bluetooth cradle from spain would beg to differ with that sentiment... |
| 05:11.43 | sbingner | haha |
| 05:12.15 | sbingner | I don't see how handsfree is any safer than using a hand to hold a cell |
| 05:12.19 | sbingner | unless you have a manual |
| 05:12.47 | hmodes | <-- manual + hands free, still outright hangs up on people while merging |
| 05:12.49 | drmessano | Its not 100% safer, but a good degree safer |
| 05:13.12 | hmodes | the solution is not regulation, the solution is people not being idiots ;p |
| 05:13.15 | hmodes | sadly that'll never happen |
| 05:13.51 | drmessano | Ive noticed that I personally drive worse with this new phone I have due to the effort it takes to keep it up against my ear |
| 05:14.01 | pkunkra | i just say shoot the really bad offenders on the spot. :-) |
| 05:14.09 | pkunkra | will solve the problem real quick. |
| 05:14.18 | pkunkra | like 120+ gets the shotgun. |
| 05:14.24 | drmessano | I make it a point to use it less when I am in heavy traffic, because I know I am impared with it more |
| 05:14.27 | Olobola | I can't run yum on my virtual dedicateded account.. does this mean I need to hunt for a million RPM's? my error: configure: error: C++ preprocessor "/lib/cpp" fails sanity check |
| 05:14.29 | pkunkra | they're going to kill someone eventually. |
| 05:14.29 | sbingner | see the solution is enforcing reckles driving laws |
| 05:14.40 | sbingner | if you're too stupid to talk on the cell and drive... get a ticket for reckless driving |
| 05:14.42 | sbingner | if not.... great |
| 05:14.47 | Olobola | centos |
| 05:14.55 | sbingner | whether they're on the cell or not is irrelevant |
| 05:15.03 | MrTelephone | how come asterisk muffles remote induced dtmf? |
| 05:15.19 | ManxPower | MrTelephone: It doesn't. |
| 05:15.47 | hmodes | yeah, i dunno, i think of my bluetooth kit the same way i think of my cup holder |
| 05:16.03 | hmodes | useful tool when everything is on autopilot |
| 05:16.03 | MrTelephone | when you get handed off by a remote answering system usually the dtmf sounds as if its trying to be suppresse |
| 05:16.06 | drmessano | sbingner: Just the like the answer to drunk driving problems isnt "If he wrecks and kills himself, problem solved" or "if he gets caught, he'll go to jail and figure it out".. but this is America.. we have to prevent everything |
| 05:16.11 | ManxPower | hmodes: always filling up with change and misc stuff? |
| 05:16.13 | hmodes | when not on auto pilot, it can wait |
| 05:16.21 | drmessano | Pregnancy, drunk driving, cell phone use, being able to eat a 3000 calorie taco |
| 05:16.27 | hmodes | haha, that too |
| 05:16.31 | sbingner | drm, *nod* |
| 05:16.34 | ManxPower | MrTelephone: perhaps you are using inband DTMF with a compressed codec, and even my cat knows that won't work. |
| 05:17.14 | sbingner | ManxPower, damn... my pet iguana told me it would |
| 05:17.15 | drmessano | cat man asterisk dtmf ? |
| 05:17.20 | hmodes | but can your cat meow in dtmf? |
| 05:17.28 | MrTelephone | sometimes i feel like im on a planet by myself |
| 05:17.34 | pkunkra | i can sing in dtmf |
| 05:17.36 | drmessano | MrTelephone: You are |
| 05:18.09 | outtolunc | did anyone hear something? |
| 05:18.17 | pkunkra | that was me singing.... |
| 05:18.21 | drmessano | I can pee in ASCII |
| 05:18.44 | outtolunc | hmm thought i heard a phone ring on a far away planet |
| 05:18.46 | pkunkra | pee in ascii? I can't even imagine how one would do that. |
| 05:19.08 | drmessano | pkunkra: It takes skill, and a control key |
| 05:19.21 | MrTelephone | i should phone someone an just get them to punch a couple digits |
| 05:20.02 | outtolunc | you can phone the *&^*&^ whose car alarm is going off as they are not turning it off |
| 05:20.22 | hmodes | relaxed or oob dtmf ftw? |
| 05:20.41 | outtolunc | with oob BFG |
| 05:21.10 | ManxPower | MrTelephone: You left out something critical. |
| 05:21.15 | outtolunc | sorry, it is driving me insane |
| 05:21.19 | MrTelephone | what |
| 05:21.31 | ManxPower | Any useful information about the setup. |
| 05:21.42 | MrTelephone | just a standard setup |
| 05:21.47 | drmessano | lol |
| 05:21.49 | MrTelephone | i don't have problems with dtmf |
| 05:21.57 | hmodes | there's a standard setup? |
| 05:22.07 | ManxPower | Polycom Phone -> Asterisk (w/ Digium or Sangoma card) -> PRI -> telco |
| 05:22.15 | MrTelephone | when i callthe microsoft helpline orsomething and you get passed off you here a couple bad dtmfs |
| 05:22.15 | ManxPower | That's MY standard setup. |
| 05:22.24 | hmodes | oh, well i fail at that standard then ;p |
| 05:22.25 | MrTelephone | yeah same here |
| 05:22.36 | ManxPower | MrTelephone: exactly the same? |
| 05:22.42 | hmodes | someone wanna buy me a pri? |
| 05:22.43 | MrTelephone | i have a cisco 6960 at my desk and i hate it |
| 05:22.43 | drmessano | Grandstream phone > Trixbox Appliance > X100P > Noisy POTS line |
| 05:22.49 | hmodes | i'll spring for the polycom |
| 05:22.51 | ManxPower | And your Polycom phones are using the ulaw codec? |
| 05:22.51 | MrTelephone | 7960 i mean |
| 05:23.02 | pkunkra | i like my cisco 7912 |
| 05:23.14 | MrTelephone | yeah i set everything for ulaw im pretty sure.. i'll have to double check tomorrow |
| 05:23.14 | pkunkra | wish it had more codecs though |
| 05:23.23 | pkunkra | debating buying 729 |
| 05:23.32 | ManxPower | Then we can continue to help you...tomorrow. |
| 05:23.33 | MrTelephone | noisy pots line caused me a lot of trouble |
| 05:24.06 | drmessano | Im sure that being Mister Telephone, you got it knocked right out though |
| 05:24.24 | MrTelephone | had to buy a pri |
| 05:24.58 | drmessano | That'll show em |
| 05:25.35 | MrTelephone | it sure did |
| 05:26.02 | drmessano | "Can't fix my POTS line? Screw you guys, i'm gonna buy a PRI from you" |
| 05:26.05 | MrTelephone | im just waitin to be sued by the phone company for stealing all their customers |
| 05:26.08 | drmessano | Stickin it to the MAN |
| 05:26.13 | MrTelephone | hahha |
| 05:27.46 | hmodes | in soviet canada, phone company sues you! |
| 05:27.54 | hmodes | oh wait, no, that's the US |
| 05:27.55 | hmodes | sorry |
| 05:27.57 | drmessano | LOL |
| 05:28.12 | drmessano | That reminds me of how I had problems with dialup internet from AT&T |
| 05:28.18 | drmessano | They pissed me off BIG TIME |
| 05:28.20 | drmessano | So I was all like |
| 05:28.27 | MrTelephone | that patentusonline website or whatever is the biggest crock of shit i've ever seen |
| 05:28.31 | drmessano | "FINE, IM ORDERING DSL... FROM YOU THEN" |
| 05:28.47 | MrTelephone | im pantenting the idea of vacumming so if you vacuum your rug you have to pay me 35 cents |
| 05:28.51 | drmessano | Yay.. showed them |
| 05:29.11 | MrTelephone | don't get mad at me because your kia broke down |
| 05:29.28 | drmessano | My KIA's never broken down lol |
| 05:30.01 | MrTelephone | why do you have to pull a bruce willis against your service provider |
| 05:30.06 | hmodes | MrTelephone: sorry, already patented cleaning and pressure differentials, that'll be $200 plzkthx, do not pass go. |
| 05:30.16 | MrTelephone | everyone who has dialup should buy highspeed from the opposing companY? |
| 05:30.23 | hmodes | PRIOR ART! |
| 05:30.30 | hmodes | PRIOR ARRRRRRT! |
| 05:30.38 | MrTelephone | i'll paypal you the money |
| 05:31.07 | MrTelephone | i dunno if its true but i remember someone saying that bill gates tried to patent binary code or something |
| 05:31.10 | drmessano | I'm pretty sure that buying a different service from the same company who couldn't fix another problem is... douche |
| 05:31.31 | drmessano | IMHO, of course |
| 05:31.39 | MrTelephone | why is your line noisy then drmessano? |
| 05:31.51 | MrTelephone | you mean they wouldn't run 500k worth the new copper so you can have a clearer phone line? |
| 05:31.55 | drmessano | I don't have noisy lines |
| 05:32.06 | MrTelephone | whats wrong with them damn |
| 05:32.22 | drmessano | If I have a line problem they fix it |
| 05:33.27 | MrTelephone | the tdm cards just don't work well overall |
| 05:33.46 | MrTelephone | never tried them near the central office though |
| 05:33.53 | drmessano | Yep, TDM cards suck |
| 05:33.57 | drmessano | X100P FTFW |
| 05:34.23 | MrTelephone | i had a tdm2400 with echo cancel an it couldn't handle a slight hum on the line |
| 05:34.40 | MrTelephone | what are the telcos using for echo cancel |
| 05:34.44 | MrTelephone | their shit must be goo |
| 05:34.49 | MrTelephone | d |
| 05:35.21 | sbingner | um... telcos don't use voip? |
| 05:35.29 | sbingner | well now they do I suppose |
| 05:36.00 | drmessano | Proper impedence matching |
| 05:36.33 | hmodes | they also own a small apartment in your neighborhood for their gear |
| 05:36.33 | *** join/#asterisk fnordus (n=dnall@24.84.160.227) |
| 05:36.51 | MrTelephone | their wiring is usally a rats nest |
| 05:36.55 | MrTelephone | a few circuit boards |
| 05:37.07 | pkunkra | What do you recommend instead of TDM cards? |
| 05:37.16 | JT | it doesn't need to be neat, they have cable tracers |
| 05:37.31 | drmessano | pkunkra: TDM cards a book on Asterisk go a long way |
| 05:37.34 | hmodes | actually, my local co is one of the most neatly cabled facilities i've ever seen |
| 05:37.39 | drmessano | pkunkra: TDM cards + a book on Asterisk go a long way |
| 05:37.56 | hmodes | to the point of having a fire vault so if the building burns to the ground they still have all the important cable termination... |
| 05:38.08 | hmodes | 100yrs of monopoly buys some niceties |
| 05:38.26 | MrTelephone | when you make 100 billion a year you can buy some nice shit |
| 05:38.35 | *** join/#asterisk [hC] (n=hardcore@S01060016b6b53c0c.vc.shawcable.net) |
| 05:38.38 | chavigny | lol |
| 05:38.43 | JT | sbingner: hmm, but echo isn't specifically a voip issue? |
| 05:39.05 | MrTelephone | theres extra latency in the computing of voip traffic |
| 05:39.07 | pkunkra | i was eventually planning on buying TDM cards for my pbx |
| 05:39.21 | drmessano | pkunkra: TDM cards work fine |
| 05:39.30 | MrTelephone | pkunkra, clear the dial tone on your lines to see if they are nice and clean first |
| 05:39.34 | pkunkra | right now i haven an external provider did the calls in over the network. |
| 05:40.28 | MrTelephone | i don't remember having problems on a sip - sip call with echo |
| 05:40.38 | MrTelephone | always tdm->ip switching seems to be a problem |
| 05:40.51 | MrTelephone | but that isn't news for anyone |
| 05:41.39 | JT | MrTelephone: the problem is in the analogue section |
| 05:42.34 | hmodes | pesky electrons |
| 05:43.09 | hmodes | all echo == bit decay, imo |
| 05:43.15 | BBHoss | anyone know any network cards that aren't host-based |
| 05:43.39 | hmodes | once your electrons start getting out of phase, ho boy look out |
| 05:43.59 | MrTelephone | no kiddin |
| 05:44.19 | MrTelephone | im tryin out these sip clients built into cable modems and sometimes people get a ringback when they hang up |
| 05:44.24 | pkunkra | time to reboot all my servers |
| 05:44.31 | drmessano | We switched to protons, and all our echo went away |
| 05:44.36 | MrTelephone | some people had that trouble with vonage.. three way calling? |
| 05:44.36 | pkunkra | for that linux kernel exploit |
| 05:44.55 | MrTelephone | there's an exploit? |
| 05:44.59 | pkunkra | yes |
| 05:45.13 | hmodes | oh dear god you said my run and hide word |
| 05:45.21 | drmessano | ha |
| 05:45.22 | MrTelephone | local exploit or network type |
| 05:45.25 | drmessano | hmodes |
| 05:45.33 | pkunkra | http://www.securityfocus.com/bid/27704/info |
| 05:45.35 | drmessano | Pee Wee's Playhouse.. the secret word |
| 05:45.38 | BBHoss | pkunkra, what does rebooting do, have they released a patch yet? |
| 05:45.43 | hmodes | AHHHHHHHHHHHHHHHH |
| 05:45.47 | pkunkra | local exploit |
| 05:45.50 | drmessano | Todays Secret Word is: EXPLOIT |
| 05:45.54 | drmessano | .. |
| 05:45.56 | drmessano | EXPLOIT |
| 05:46.01 | drmessano | AHHHHHHHHHHHHHHHHHHHHHH |
| 05:46.09 | pkunkra | its not a remote exploit |
| 05:46.19 | BBHoss | still critical though |
| 05:46.22 | hmodes | AAAAAAGGGGGHHHHHHHHHHH |
| 05:46.27 | BBHoss | r00t access from a npu |
| 05:46.47 | drmessano | Its a "IF MAH USERS GNU LINIX, AND HAD LOGINS, I WUD BE SO FUXORED ZOMG" |
| 05:46.56 | MrTelephone | i got charged for getting root on some university servers a long time ago |
| 05:47.02 | BBHoss | heh |
| 05:47.14 | MrTelephone | running someone elses buffer overflow |
| 05:47.17 | pkunkra | charge as in pay or as in jail? |
| 05:47.36 | BBHoss | charge doesn't mean either of those |
| 05:47.37 | MrTelephone | they were nice an just made me write an apology letter |
| 05:47.48 | BBHoss | convicted does though |
| 05:48.34 | MrTelephone | did you check out the source in that exploit? |
| 05:48.34 | drmessano | I taught Kevin Mitnick how to dumpster dive.. they never caught me though |
| 05:48.35 | MrTelephone | hah |
| 05:48.36 | MrTelephone | <PROTECTED> |
| 05:49.08 | hmodes | LOL I'M USING THE INTERNET! |
| 05:49.31 | MrTelephone | i like the "thanks jen for finding the vulnerbilities" yeah thank you for cracking half the worlds systems.. thanks man |
| 05:50.08 | drmessano | Yeah man, takin down the effin MATRIX |
| 05:50.17 | MrTelephone | what a burnout |
| 05:50.39 | MrTelephone | finding an exploit is like finding a needle in a haystack |
| 05:50.47 | MrTelephone | i guess you find them when your writing code or something |
| 05:52.10 | MrTelephone | thats a sweet exploit to work on all those kernels |
| 05:53.20 | drmessano | Two chicks at once |
| 05:53.28 | MrTelephone | no kidding |
| 05:53.39 | MrTelephone | he has a function in there called gimmieroot(), how cute |
| 05:53.59 | *** join/#asterisk obnauticus (n=obnautic@c-67-160-183-109.hsd1.wa.comcast.net) |
| 05:54.12 | obnauticus | Does anyone here know how to fix this error with a new asterisk-realtime install |
| 05:54.14 | obnauticus | [Feb 12 00:51:44] WARNING[4658]: config.c:1316 find_engine: Realtime mapping for 'sippeers' found to engine 'mysql', but the engine is not available |
| 05:54.32 | pkunkra | drmessano, i don't find much that's really useful in dumpsters though. |
| 05:54.38 | obnauticus | relevant extconfig.conf line: sippeers => mysql,asterisk,sip_buddies |
| 05:55.08 | *** join/#asterisk MerK--- (i=sadad@cable-41-220-107-101.intelvision.net) |
| 05:56.15 | drmessano | Most companies don't throw decent stuff away anymore |
| 05:56.30 | drmessano | Losers like me who smash hard drives before tossing them |
| 05:56.32 | MrTelephone | what are you looking for |
| 05:56.53 | MrTelephone | i have 3 sata raid cards that i don't know what to do with |
| 05:56.58 | drmessano | Phone records, root exploits, empty cup o noodles |
| 05:57.02 | MerK--- | hi, i'm having an issue with the queue_stats? can i get help here or is their a specific channel that i can get help |
| 06:01.38 | phix | Power alarm on module 1 |
| 06:01.40 | phix | wtf? |
| 06:01.53 | phix | hardware is tdm400p |
| 06:02.27 | JT | with what modules? |
| 06:03.09 | MrTelephone | if you have a cisco mgc ss7 switch can you be a facility based clec? |
| 06:04.42 | phix | JT: 3xFXS |
| 06:04.52 | phix | (TDM30B) |
| 06:05.16 | JT | phix: is the molex connector plugged in? |
| 06:05.44 | phix | JT: yes |
| 06:06.08 | phix | JT: I have dial tone too, well only on 2 modules, module 3 doesn't have a dial tone |
| 06:06.19 | phix | module 1 does though |
| 06:07.02 | JT | hmm |
| 06:07.40 | phix | JT: interesting problem ay. |
| 06:10.05 | *** join/#asterisk dchipman (n=david@CPE0014bf4ab5ca-CM000f212f943c.cpe.net.cable.rogers.com) |
| 06:11.17 | dchipman | Hi all, I'm tryign to get asterisk working on FC8. It says it can't write the asterisk.pid file to the /var/run directory. What could be the problem.? |
| 06:13.56 | *** join/#asterisk WilliamK (n=noc@static-71-170-144-3.dllstx.fios.verizon.net) |
| 06:14.06 | sbingner | dchipman, permissions |
| 06:14.40 | dchipman | On what, sbingner? |
| 06:14.52 | sbingner | dchipman, where can't it write? |
| 06:15.12 | dchipman | It can't write in /var/run |
| 06:15.26 | sbingner | so where would you think permissions could be the problem? |
| 06:15.42 | dchipman | I don't know. NOthing else has this problem.. |
| 06:15.54 | drmessano | Hmm /var/run maybe |
| 06:16.07 | drmessano | ~idk |
| 06:16.13 | drmessano | :( |
| 06:16.14 | dchipman | set them to what, though? |
| 06:16.39 | sbingner | dchipman, well you need to be able to write to /var/run/asterisk.pid - that would depend on how you want to permit that |
| 06:17.09 | sbingner | generally /var/run is world writable |
| 06:17.10 | dchipman | THe /var/run ls -l output is drwxr-xr-x 33 root root 4096 2008-02-12 00:56 /var/run/ |
| 06:17.14 | dchipman | OK |
| 06:17.17 | sbingner | er |
| 06:17.24 | sbingner | only writable by root |
| 06:17.26 | dchipman | What? |
| 06:17.32 | dchipman | I see that.. |
| 06:17.41 | obnauticus | Does anyone here know why i am getting this error with realtime |
| 06:17.47 | obnauticus | MySQL RealTime: Failed to connect database server asterisk on (err 2002). Check debug for more info. |
| 06:17.47 | sbingner | you may want to just make asterisk.pid and make it writtable by * |
| 06:17.54 | obnauticus | I have specified the host, but it's not even showing in the cli |
| 06:19.19 | dchipman | There, now it's running! |
| 06:20.41 | *** join/#asterisk sorush20 (n=sorush20@87-194-133-2.bethere.co.uk) |
| 06:20.46 | sorush20 | hi |
| 06:20.52 | obnauticus | anyone :( |
| 06:20.53 | *** part/#asterisk dacs (n=haiger@unaffiliated/dacs) |
| 06:21.00 | sorush20 | do I need hardware to be able to record my telephone conversations |
| 06:21.05 | obnauticus | No |
| 06:21.13 | obnauticus | lookup Application Dial and monitoring. |
| 06:21.23 | obnauticus | the M flag i think |
| 06:22.24 | *** join/#asterisk justdave (n=dave@unaffiliated/justdave) |
| 06:23.50 | phix | ok I swaped module 1 and 3 around, and module 3 doesn't work |
| 06:24.17 | phix | is there some limitation on how many ports will work under asterisk (from debian sarge)? |
| 06:24.28 | phix | (or zaptel drivers under debian testing)? |
| 06:24.47 | sbingner | heh no |
| 06:24.54 | phix | hmmmm |
| 06:25.03 | phix | so what would be stopping ports 3 - 4 from not working |
| 06:25.11 | sbingner | not being configured properly? |
| 06:25.20 | sbingner | not being plugged in properly? |
| 06:26.02 | phix | configured in what way? /etc/zaptel.conf and /etc/asterisk/zapata.conf, are there any other config files? |
| 06:26.10 | fujin | phix: I trust you're aware that specific ports have different signalling types? |
| 06:26.20 | phix | the configuration of channel 3 is the same as channel 1 and 2, channels 1 and 2 work |
| 06:26.21 | fujin | e.g.; port 3 usually won't work while plugged into 1 |
| 06:26.36 | fujin | You've probably got a 4port fxo/fxs card - that is, two ports fxo, two ports fxs |
| 06:26.41 | fujin | are they differnt colours? (green, orange) |
| 06:26.44 | phix | different signalling types? I have a 3 port FXS card |
| 06:26.49 | fujin | ah |
| 06:26.51 | phix | all green |
| 06:27.01 | fujin | odd indeed |
| 06:27.20 | phix | ports 1 - 2 work, port 3 has no dial tone |
| 06:27.23 | fujin | so, all 3 ports have green modules, and you've got them all configured with the correct signalling type in zapata.conf? |
| 06:27.39 | phix | yes |
| 06:27.42 | fujin | do the relevant zt* commands produce the correct output (forgive me, I'm not familiar with it) |
| 06:27.54 | phix | dmesg picks all three up |
| 06:27.56 | sbingner | ztcfg -v |
| 06:28.00 | phix | when the module loads |
| 06:28.21 | fujin | what about zttool / ztcfg -v? |
| 06:28.27 | phix | 3 channels to configure, registered tone zone 1 (Australia) |
| 06:28.42 | phix | zaptel ver: 1.4.7.1 |
| 06:28.54 | phix | does it matter what zaptel vers I use compared with asterisk? |
| 06:29.11 | phix | although to me it seems half of the TDM board is dead |
| 06:29.36 | sbingner | you could always try the module in port 3 in port 4 too... but you didn't show the entire ztcfg -v |
| 06:29.42 | sbingner | http://rafb.net/paste |
| 06:31.08 | *** join/#asterisk [hC] (n=hardcore@S0106001346a4b813.vc.shawcable.net) |
| 06:31.16 | *** join/#asterisk oej (n=olle@soll4-125.cust.blixtvik.net) |
| 06:33.12 | phix | sbingner: I am in cmd line |
| 06:33.16 | phix | no x on this machine |
| 06:33.25 | obnauticus | How do i enable debugging on realtime |
| 06:33.26 | obnauticus | ? |
| 06:34.35 | phix | sbingner: it says: zaptel version: 1.4.7.1\n Echo canceller: MG2\n Configuration\n =====================\n\n\n 3 channels to configure.\n\n Registered tone zone 1 (Australia) |
| 06:34.40 | phix | sbingner: that is the complete output |
| 06:35.33 | phix | sbingner: using ztcfg -vv tells me that the ports use FXO kewlstart signalling |
| 06:35.43 | phix | which is correct |
| 06:43.12 | phix | ok I just used ztmonitor to record from module 3, and nothing |
| 06:44.11 | phix | anything else I can try? |
| 06:44.11 | pkunkra | all done with patches |
| 06:44.25 | phix | JT? sbingner? |
| 06:44.34 | pkunkra | hmodes: "exploit!" |
| 06:45.53 | phix | pkunkra: ? |
| 06:47.15 | drmessano | AHHHHHHHHHHHHHHHHH! |
| 06:47.41 | drmessano | patches, we don't need no stinking patches |
| 06:47.47 | phix | :\ |
| 06:48.35 | pkunkra | haha |
| 06:48.59 | pkunkra | i'm reminded of monty python.... |
| 06:49.37 | drmessano | Asterisk is the holy hand grenade? |
| 06:50.08 | phix | no, the evil rabiit |
| 06:50.15 | pkunkra | that too. but i was thinking of "we are the knights that say..." |
| 06:50.23 | phix | BRB, taking this card out to do more testing |
| 06:50.27 | phix | NI! |
| 06:50.37 | drmessano | lol |
| 06:50.47 | pkunkra | i thought it was spelled "nee!" |
| 06:50.56 | drmessano | Do not count to 2, unless thou proceedeth to 3 |
| 06:51.13 | drmessano | Thou shall not count to 4 |
| 06:51.15 | J4k3 | keep your patches, I want diffs. |
| 06:51.16 | drmessano | 5 is right out |
| 06:52.05 | drmessano | A SHRUBBERY! |
| 06:52.53 | J4k3 | you can keep the shrubbery, its bad enough we got george bush |
| 06:53.19 | drmessano | I command you to cut down the tallest tree in the forest, with.. |
| 06:53.35 | pkunkra | i'd would have voted for the shrubbery if it ran against bush. |
| 06:53.39 | drmessano | A HERRING |
| 06:53.46 | pkunkra | it probably would have made a better president. |
| 06:54.06 | drmessano | President Boxwood |
| 06:54.36 | J4k3 | pkunkra: come on, john ketchup wasn't good enough for you?! |
| 06:54.41 | drmessano | HA |
| 06:55.00 | drmessano | His running mate, Peter Mustard, was a big turn off |
| 06:55.22 | drmessano | Thats a damn good porn name |
| 06:55.57 | pkunkra | oh my. what lovely discussions on #asterisk at 2am |
| 06:57.23 | drmessano | lol |
| 06:57.36 | drmessano | It's not just 2AM |
| 06:57.37 | *** join/#asterisk [T]ank (n=ckwall@c-71-195-199-107.hsd1.ut.comcast.net) |
| 06:58.03 | [T]ank | i am looking for the link on the digium website where i pay for the g729 license. I am not finding it. anyone know where I can find it? |
| 06:58.20 | adeel | why am i getting this output when i try sending DTMF tones after a call is connected? Packet2Packet bridging SIP/foo and SIP/bar |
| 07:00.01 | drmessano | http://store.digium.com/productview.php?category_id=5&product_code=8G729CODEC&main_category_id=5 |
| 07:00.16 | drmessano | I clicked STORE |
| 07:00.20 | drmessano | and then G729 Codec |
| 07:12.58 | *** join/#asterisk exothermc (n=miles@68-189-133-163.dhcp.wlwl.wa.charter.com) |
| 07:13.29 | exothermc | how are vendors pulling off FMC with cell networks while not utilizing dual mode phones? |
| 07:14.34 | ManxPower | fmc? |
| 07:18.04 | drmessano | Like Cell + Wifi |
| 07:18.42 | drmessano | IF (Wifi = Very Yes) { wifi on | cell off } |
| 07:20.15 | ManxPower | FMC freddie Mac Corporation |
| 07:24.01 | J4k3 | exothermc: dect |
| 07:24.11 | *** join/#asterisk erojasv (n=erojasv@201.240.80.49) |
| 07:25.12 | exothermc | ManxPower: FMC is fixed mobile convergence. |
| 07:29.52 | *** join/#asterisk jblack (n=jblack@pool-71-173-53-239.sctnpa.east.verizon.net) |
| 07:30.15 | jblack | I woke up tonight not knowing what year it was. |
| 07:30.56 | sweeper | 2040, obviously |
| 07:33.41 | drmessano | 2032 |
| 07:35.22 | obnauticus | jblack not as bad as what i did one time |
| 07:35.31 | obnauticus | i woke up and tried to smell out of my ear |
| 07:35.31 | yangvnc | Do you know a good forum for minute reselling ? |
| 07:35.32 | exothermc | J4k3: hmm no there are some like newstep that pull if off with existing infrastructure. http://www.newstep.com/products%20and%20solutions/video.asp |
| 07:36.24 | drmessano | Uh |
| 07:36.28 | drmessano | The video also shows NewStep’s industry leading Dual-Mode Handoff solution and demonstrates its seamless cellular/Wi-Fi capabilities......... |
| 07:36.49 | obnauticus | The cake is a lie. |
| 07:37.58 | jblack | When I was a child, I woke up at my bus stop at 2am. |
| 07:39.10 | J4k3 | exothermc: I've seen it in use.. the handoffs are pretty damned hard under real-world circumstance |
| 07:39.26 | J4k3 | if you slooooowly lose service off the wifi network you might not experience more than a standard GSM handoff, but thats not typical for wifi |
| 07:39.41 | J4k3 | you walk out the front door and close its big metal self and suddenly you go from -60 to dead. |
| 07:40.05 | J4k3 | (at least thats the experience with my wifi-equipped pocketpc and using SIP) |
| 07:45.10 | *** join/#asterisk Al_WinKiller (i=Alex_Win@83.139.12.188) |
| 07:46.59 | Al_WinKiller | hi guys, I am facing a problem cisco phone ( sip ) ---> Asterisk_1 <----- IAX2 -----> Asterisk_2 ---> softphone (iax2) , so my question is how do I make my softphone work with sip not iax2 ? cuz I want to replace it with another sip phone |
| 07:47.01 | Al_WinKiller | ? |
| 07:47.32 | drmessano | Get a sip softphone |
| 07:48.23 | Al_WinKiller | I got one ) and more ) and sip gw too ) the problem is how to make iax2 work between servers ONLY and clients ( on both sides with SIP ) |
| 07:48.25 | Al_WinKiller | ? |
| 07:48.49 | drmessano | The phones have nothing to do with the trunks |
| 07:49.33 | Al_WinKiller | ok, can you explain me that ? pls ? |
| 07:49.44 | drmessano | Theres nothing to explain |
| 07:49.55 | drmessano | You have IAX2 between boxes |
| 07:49.57 | Al_WinKiller | I have done it by this manual http://www.voip-info.org/wiki-Asterisk+-+dual+servers |
| 07:50.01 | Al_WinKiller | example 2 |
| 07:50.05 | drmessano | You can use whatever you want on each Asterisk |
| 07:50.12 | Al_WinKiller | ok |
| 07:50.17 | *** join/#asterisk agx (n=AGX@host63-216-static.34-88-b.business.telecomitalia.it) |
| 07:50.50 | Al_WinKiller | so,, when I register phone ( soft phone as a SIP client ) on second server ( asterisk_2 ),,, it says that it can't reach it :( |
| 07:51.10 | drmessano | Do you have a SIP peer defines for that phone? |
| 07:51.13 | drmessano | defined |
| 07:51.33 | Al_WinKiller | for soft phone in sip.conf ? |
| 07:52.00 | drmessano | The same way you set the other one up on the other box |
| 07:52.04 | drmessano | Its no different |
| 07:52.22 | *** join/#asterisk Frogzoo (n=Frogzoo@CPE-58-168-48-62.nsw.bigpond.net.au) |
| 07:52.39 | Al_WinKiller | may I past it here ? 12 rows ? |
| 07:52.43 | drmessano | NO |
| 07:52.56 | drmessano | You have a SIP phone working on one box, no? |
| 07:53.21 | Al_WinKiller | yes, on asterisk_1 ( it is cisco phone ) |
| 07:53.31 | drmessano | So set up asterisk 2 to have SIP phone too |
| 07:53.36 | drmessano | The SAME way you did it on box 1 |
| 07:53.36 | *** join/#asterisk reber (n=reber@193.253.213.73) |
| 07:53.42 | Al_WinKiller | and same thing on Asterisk_2 ( but in iax.conf ) |
| 07:53.47 | Al_WinKiller | ok |
| 07:53.50 | drmessano | No |
| 07:53.51 | Al_WinKiller | hold on a second |
| 07:53.51 | *** part/#asterisk reber (n=reber@193.253.213.73) |
| 07:53.52 | drmessano | NOT IAX |
| 07:53.55 | drmessano | SIP |
| 07:53.59 | drmessano | You want a SIP phone |
| 07:54.03 | *** join/#asterisk stimpie (n=stimpie@84-104-5-115.cable.quicknet.nl) |
| 07:54.07 | yangvnc | I have these 2 extensions : |
| 07:54.08 | drmessano | Thats what you asked about |
| 07:54.08 | yangvnc | exten => 5863162,1,Dial(SIP/62&SIP/82,7,rtk) |
| 07:54.08 | yangvnc | exten => 5863162,2,Dial(SIP/gsm/041334765,30,rtk) |
| 07:54.35 | yangvnc | How could I define that 62 keeps ringing, simply by adding &SIP/62 in second context? |
| 07:55.05 | sweeper | http://jicksta.com/articles/2008/02/11/menu-command <-- oh this is sexy, although I'm so adding an interdigit timeout option asap |
| 07:55.33 | *** join/#asterisk Frogzoo (n=Frogzoo@CPE-58-168-48-62.nsw.bigpond.net.au) |
| 07:56.37 | exothermc | J4k3: Ya you are talking about the wifi handoff, I'm not concerned with that at all. I'm talking about the ability to use the cell network to call 4 digit extensions etc. |
| 07:56.38 | Al_WinKiller | look what I got |
| 07:56.42 | Al_WinKiller | <PROTECTED> |
| 07:56.42 | Al_WinKiller | Feb 12 11:41:41 WARNING[29701]: chan_iax2.c:2785 create_addr: No such host: 1299501 |
| 07:57.01 | Al_WinKiller | but when 1299501 is in iax.conf it works good :( |
| 07:57.22 | drmessano | ~book |
| 07:57.23 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com, or see ~buybook |
| 07:57.31 | exothermc | Does someone have a good page which lists the major improvements to 1.6? |
| 07:57.42 | *** join/#asterisk adjohn (n=adjohn@219.106.248.145) |
| 07:58.19 | *** join/#asterisk Maliuta (n=nikolai@kiev.lusan.id.au) |
| 07:59.32 | Al_WinKiller | DUDE IT WORKS !!!! stupid fout :D in extensions.conf :D |
| 07:59.40 | Al_WinKiller | thnx a lot ) |
| 07:59.41 | drmessano | yep |
| 07:59.44 | obnauticus | <3 |
| 07:59.51 | Al_WinKiller | GOD BLESS INTERNET !!!!! despite I am an ateist :D |
| 07:59.55 | Al_WinKiller | ha ha ha |
| 07:59.57 | obnauticus | lol |
| 08:00.30 | obnauticus | Just got my 43U server rack |
| 08:00.35 | obnauticus | pretty nice for 25 duckets. |
| 08:01.03 | obnauticus | http://img208.imageshack.us/img208/1656/dsc0045wc3.jpg |
| 08:01.21 | drmessano | cool |
| 08:01.52 | *** join/#asterisk ShadowHntr (n=sentinel@wikipedia/Shadowhntr) |
| 08:03.48 | J4k3 | I have a a couple racks that I use for shelving units |
| 08:03.57 | J4k3 | they're AWESOME for that, assuming you have good shelves |
| 08:05.45 | yangvnc | How could I define that 62 keeps ringing, simply by adding exten => 5863162,2,Dial(SIP/62&SIP/gsm/041334765,30,rtk) in second extension? |
| 08:05.48 | yangvnc | I have these 2 extensions : |
| 08:05.51 | yangvnc | exten => 5863162,1,Dial(SIP/62&SIP/82,7,rtk) |
| 08:05.55 | yangvnc | exten => 5863162,2,Dial(SIP/gsm/041334765,30,rtk) |
| 08:06.15 | *** join/#asterisk Alexandre_fr (n=alex@nat04-bzr.dys1.com) |
| 08:06.15 | *** join/#asterisk esaym (n=user@72.183.198.134) |
| 08:15.57 | yangvnc | I am getting also this error with Pickup string - http://openpaste.org/en/5083/ |
| 08:18.13 | *** join/#asterisk adjohn (n=adjohn@219.106.248.145) |
| 08:21.13 | *** join/#asterisk Xen^ (n=linux@unaffiliated/lnux/x-10290) |
| 08:21.23 | *** join/#asterisk mort_gib (n=mjensen@16.Red-83-36-63.staticIP.rima-tde.net) |
| 08:23.45 | *** join/#asterisk stimpie (n=stimpie@84-104-5-115.cable.quicknet.nl) |
| 08:24.23 | *** join/#asterisk stimpie (n=stimpie@84-104-5-115.cable.quicknet.nl) |
| 08:25.16 | *** join/#asterisk L|NUX (n=linux@unaffiliated/lnux/x-10290) |
| 08:25.33 | *** join/#asterisk stimpie (n=stimpie@84-104-5-115.cable.quicknet.nl) |
| 08:26.18 | *** join/#asterisk steliosk (n=Stelios@85.75.208.185) |
| 08:27.43 | *** join/#asterisk stimpie (n=stimpie@84-104-5-115.cable.quicknet.nl) |
| 08:28.35 | jblack | Smart kids are dumb. |
| 08:28.53 | jblack | And the smarter the are, the dumber they are. |
| 08:29.36 | drmessano | If Stephen Hawking was so smart, he would figure out a way to get out of that wheelchair |
| 08:29.46 | jblack | Exactly! |
| 08:30.32 | *** join/#asterisk stimpie (n=stimpie@84-104-5-115.cable.quicknet.nl) |
| 08:31.06 | jblack | Three days of ?OMG WTF OVER? Because my damn kid couldn't bother to tell me that she had *1* late assignment. |
| 08:31.15 | *** join/#asterisk adjohn (n=adjohn@219.106.248.145) |
| 08:31.28 | jblack | Turned a 99 average into a B. |
| 08:31.58 | drmessano | niice |
| 08:32.17 | jblack | Yeah. Straight As for everything but one class... a B on the quarter, and a C on the midterm. |
| 08:32.42 | drmessano | Damn |
| 08:33.05 | jblack | A normally straight A student that gets a C on anything is like a frigging road flare going up on a moonless night. |
| 08:33.13 | *** join/#asterisk incog_afk (n=nospam@cyberarmy.com) |
| 08:33.21 | drmessano | I would suspect drugs |
| 08:33.26 | jblack | Nope. |
| 08:33.39 | sweeper | I would suspect being bored :P |
| 08:34.17 | drmessano | I almost flunked out of high school |
| 08:34.19 | incog_afk | Well... if you're bored... :) I have questions about Asterisk as a SIP client to FWD... :) |
| 08:34.19 | drmessano | lol |
| 08:34.21 | jblack | The quarter grade was her not turning in a single, trival, project on time. Once it was late, she hid it from me, which made it hard for her to get it done. So it was late, then late again, then late again. |
| 08:34.47 | jblack | The midterm grade was because she was cocksure. She didn't need to study it at all! She already knew everything! |
| 08:35.08 | jblack | dumbass. |
| 08:35.17 | incog_afk | ah... I used to finish assignments in class, then lose them before the instructor was willing to let me turn them in... |
| 08:36.00 | jblack | I was so pissed on monday, that I shouted at her "You better not DARE to put too much salt on my french fries once you end up at mcdonalds!" |
| 08:36.14 | drmessano | ha |
| 08:36.27 | drmessano | I hear McDonalds has a networking school now |
| 08:36.39 | incog_afk | Careful. You'll give her ideas. She'll start working there after school instead of doing her homework... |
| 08:37.05 | jblack | "That midterm grade is so bad, they might not let you into Subway university" |
| 08:37.26 | incog_afk | Mickey-D's does a lot of small-LAN-type networking - the whole order-entry/POS system is network-based... |
| 08:37.30 | drmessano | You're laughing now, wait until shes in the Computer Science program at Hamburger Univeristy |
| 08:37.44 | jblack | ugh. |
| 08:38.02 | drmessano | Im well are of the network infrastructure of McDonalds |
| 08:38.09 | drmessano | Im well aware of the network infrastructure of McDonalds |
| 08:38.11 | drmessano | grrr |
| 08:38.14 | jblack | Dont' tell her, but at this moment I'm happy it's just a case of two independant fuckups, rather than a mass implosion. |
| 08:38.23 | incog_afk | I can't wait until I'm up against someone with a Bacelor's degree from HU, for a networking job... :) |
| 08:38.39 | jblack | Is there a Hamburger University at mcd's ? |
| 08:38.43 | drmessano | yes |
| 08:38.59 | jblack | I'll print that out for her. |
| 08:39.01 | drmessano | http://www.mcdonalds.com/corp/career/hamburger_university.html |
| 08:39.13 | jblack | yeah, found it. |
| 08:39.26 | incog_afk | It's a good way to recover the cash from their cheeseburgery scholarship program... run the school, and you're only paying yourself... |
| 08:40.17 | incog_afk | Anyone here actually use Asterisk? :-P |
| 08:41.02 | incog_afk | I shoulda known... :-P |
| 08:41.05 | drmessano | Whats asterisk? |
| 08:41.35 | jblack | It's that funny character just over the 8 on the keyboard, right? |
| 08:41.49 | incog_afk | It's a VoIP PBX system... :) |
| 08:42.03 | jblack | Hamburger university is a VoIP PBX system? |
| 08:42.08 | Silent-X | isnt "*" asterisk? |
| 08:42.15 | jblack | Can I get them to hold the pickles? |
| 08:42.20 | Silent-X | I thought this channel was all about the * key |
| 08:42.26 | drmessano | Over the 2 on the Commodore 64 keyboard |
| 08:42.40 | jblack | Silent-X: Sure.... * rocks much more than } |
| 08:42.56 | Silent-X | corse, everyone loves the key that makes wildcards possible |
| 08:43.11 | drmessano | @ had it's own key on the C64 keyboard.. revolutionary bastards |
| 08:43.14 | incog_afk | No... this channel is supposed to be about the Asterisk open-source PBX project (see http://www.asterisk.org/ ) |
| 08:43.30 | drmessano | WTF is VOIP? |
| 08:43.44 | incog_afk | Voice-over-Internet-Protocol |
| 08:43.45 | Silent-X | Video Of Internet Porn? |
| 08:43.57 | jblack | It's the sound that comes from vinyl rubbing against latex. |
| 08:44.19 | jblack | Not that I would know how that sounds |
| 08:44.25 | Silent-X | I wonder |
| 08:44.30 | drmessano | Voice over internet? |
| 08:44.34 | drmessano | Sounds... stupid |
| 08:45.06 | ShadowHntr | must... resist... feeding... the trolls... |
| 08:45.08 | incog_afk | Eh. It's a way of making free long-distance calls, to most people that use it. |
| 08:45.15 | drmessano | WHAT? |
| 08:45.17 | *** join/#asterisk sx|lappy (n=sxpert@lgit-1225.obs.ujf-grenoble.fr) |
| 08:45.20 | drmessano | FREE LONG DISTANCE? |
| 08:45.32 | drmessano | Why the effing dookieville didn't you say so? |
| 08:45.40 | Silent-X | that might improve my pranking |
| 08:45.42 | drmessano | Now THAT tickles my fancy |
| 08:46.01 | drmessano | Over the internet? |
| 08:46.07 | drmessano | Thats CRAZYTALK |
| 08:46.20 | drmessano | Is that what Skype is? |
| 08:46.27 | drmessano | Free Long Distance Thingo? |
| 08:46.39 | Silent-X | why not, the internet is just a series of tubes, im sure someone could yell down them loud enough for it to travel |
| 08:46.54 | incog_afk | Well... Asterisk is a lot more involved than just that. VoIP is comparatively simple, but the various uses for it can get complicated. I have one that is particularly complex, and I was here looking for help setting it up... |
| 08:46.58 | drmessano | Silent-X: thats effin brilliant |
| 08:47.11 | incog_afk | Skype is a VoIP provider, yes... |
| 08:47.17 | drmessano | tell me more about this asterisk |
| 08:47.23 | drmessano | I am infangled |
| 08:47.28 | incog_afk | So is FreeWorldDialup, and many, many others... |
| 08:47.35 | drmessano | Free? |
| 08:47.38 | drmessano | Effin A |
| 08:47.54 | drmessano | So where do I buy askerist? |
| 08:47.59 | drmessano | asterisk, sorry |
| 08:48.29 | incog_afk | it's a free download from Asterisk.org |
| 08:48.35 | drmessano | Cool |
| 08:48.41 | drmessano | I hope it works with Vista |
| 08:48.54 | Silent-X | your using vista? im still on 98 |
| 08:49.04 | ShadowHntr | 98? i'm still on 3.11 ! |
| 08:49.09 | drmessano | Does Asterisk work on 98? |
| 08:49.14 | Silent-X | I hope so |
| 08:49.31 | Silent-X | im tired of getting the shiv just because I dont want to upgrade |
| 08:49.38 | incog_afk | you might want to look into X-lite from CounterPath, instead of starting with Asterisk. |
| 08:49.46 | drmessano | Maybe I need to swith to that.. I am only using Vista because this new Gateway I got comes with Vista |
| 08:49.55 | *** join/#asterisk stimpie (n=stimpie@84-104-5-115.cable.quicknet.nl) |
| 08:50.00 | drmessano | Why not Asterisk? |
| 08:50.07 | drmessano | Is it that bad? |
| 08:50.31 | incog_afk | Asterisk is a lot more complicated than you are likely to need, and it's made to run on Linux. |
| 08:50.38 | drmessano | Oh |
| 08:50.44 | drmessano | :( |
| 08:50.56 | drmessano | How long have you been using Asterisk? |
| 08:51.18 | Silent-X | whats linux? |
| 08:51.23 | incog_afk | A couple of years, on and off. |
| 08:51.25 | Silent-X | is that the new microsoft OS? |
| 08:51.32 | nixguy | Silent-X: yes. but no |
| 08:51.37 | sergee | jblack: jpink? :) |
| 08:51.38 | *** join/#asterisk stimpie (n=stimpie@84-104-5-115.cable.quicknet.nl) |
| 08:51.45 | drmessano | Silent-X: i think it's something they use on Mainframes and ATMs |
| 08:51.50 | incog_afk | Linux is an open-source operating system that has nothing to do with Microsoft. |
| 08:51.52 | drmessano | It's like DOS |
| 08:52.08 | sergee | linux is not like a DOS |
| 08:52.12 | sergee | thanks god |
| 08:52.14 | drmessano | Pretty shitty if you ask me |
| 08:52.19 | nixguy | :) |
| 08:52.36 | nixguy | well the interaction with the "os" is quite similair to DOS at first glance |
| 08:52.58 | stimpie | what is the similairity? |
| 08:53.06 | stimpie | you can enter commands? |
| 08:53.07 | drmessano | _ <---- |
| 08:53.15 | *** part/#asterisk Sinar (n=bwoSinar@81.144.146.38) |
| 08:53.17 | nixguy | yes |
| 08:53.30 | nixguy | it has a textbased interface |
| 08:53.32 | drmessano | You type stuff, it does stuff |
| 08:53.47 | drmessano | Like ./configure |
| 08:53.52 | drmessano | make && make install |
| 08:53.53 | drmessano | etc |
| 08:53.55 | stimpie | so the rollercoaster I operated runs on something like dos? |
| 08:53.55 | sergee | drmessano: you pres Alt+FN and what you get? |
| 08:54.01 | Silent-X | rm -Rf /* ftw |
| 08:54.10 | mort_gib | Kindergarten meet today?? |
| 08:54.11 | incog_afk | Linux is principally a command-line environment - although there are a series of windowing environments that can run within Linux... |
| 08:54.32 | stimpie | linux is an operating system in which you can run a shell (graphical or commandline) |
| 08:54.48 | drmessano | Like a turtle? |
| 08:54.56 | Silent-X | I like turtles |
| 08:54.57 | drmessano | Is Asterisk a turtle? |
| 08:55.08 | nixguy | isnt it mor like a disc? |
| 08:55.11 | nixguy | with a turtle beneath? |
| 08:55.14 | sergee | no, it's more like a shark |
| 08:55.16 | stimpie | thats it |
| 08:55.16 | drmessano | Ohhh |
| 08:55.24 | drmessano | A TURTLESHARK |
| 08:55.29 | sergee | yeah |
| 08:55.32 | Silent-X | genesplicing at its finest |
| 08:55.39 | drmessano | Dont eff with a TURTLESHARK |
| 08:55.43 | sergee | turtle with a head of shark |
| 08:55.56 | sergee | nuclear power, mutations.. you know.. |
| 08:56.10 | Silent-X | I bet godzilla could be a turtleshark |
| 08:56.16 | drmessano | incog_afk: This Asterisk, does it use that SIP thing? |
| 08:56.27 | Silent-X | straw? |
| 08:56.53 | drmessano | SIP: Some Internet Port <-- It's a phone thing |
| 08:57.13 | drmessano | It's ICP/TC port 5070 |
| 08:57.28 | drmessano | I can google :) |
| 08:57.30 | stimpie | I thought SIP was a nickname for skype, Stupid Internet Phone |
| 08:57.45 | incog_afk | Yes, drmessano... And I'm trying to use that SIP thing to communicate through that VoIP thing, to FreeWorldDialup... |
| 08:57.57 | drmessano | OHHH |
| 08:58.08 | drmessano | Maybe I can help u? |
| 08:58.27 | incog_afk | Better not to worry about what SIP stands for. The thing that it stands for won't mean any more than the acronym... |
| 08:58.52 | drmessano | Is that like a Vontage? |
| 08:59.53 | incog_afk | Maybe... If you know anything about Asterisk. And yes - a little. FreeWorldDialup and Vonage are both SIP-based VoIP providers. My problem is specific to FreeWorldDialup, however... |
| 09:00.08 | drmessano | Well, FWD sucks |
| 09:00.17 | drmessano | One day it works, next it doesn't |
| 09:00.25 | incog_afk | Everything sucks when it isn't working... :) |
| 09:00.37 | *** join/#asterisk shtoom (n=godson@59.93.102.161) |
| 09:00.59 | drmessano | Sipphone sucks less |
| 09:01.11 | drmessano | ALA Gizmo |
| 09:01.33 | drmessano | Same idea, less "ZOMG IT WORKED 2 HOURS AGO" |
| 09:01.50 | drmessano | Alright, its way past my bedtime |
| 09:01.55 | drmessano | Im out |
| 09:01.59 | Silent-X | night drmessano, was fun |
| 09:02.05 | drmessano | ;) |
| 09:02.07 | incog_afk | I have a terminal adapter, and a subscription VoIP provider. But I can deliver DTMF tones over FWD, and I can't, over ViaTalk... |
| 09:02.36 | shtoom | Hi I am getting yellow alarm on PRI channels every now and then and all the calls are getting dropped any idea y this is happening ? |
| 09:02.53 | incog_afk | I'm trying to set up an IVR menu, and I really need DTMF tones for that... |
| 09:03.16 | shtoom | I've searched over the lists and asteriskguru but things are not quite clear on what to do with this |
| 09:03.19 | *** join/#asterisk vgster (n=vgster@cpc2-ledn1-0-0-cust826.leed.cable.ntl.com) |
| 09:03.48 | incog_afk | what is sending you the alarm? It doesn't sound like an Asterisk issue - sounds like a hardware or telco issue... |
| 09:05.02 | shtoom | it shows a warning on console chan_zap.c Detected alarm on channel 1 : Yellow alarm |
| 09:05.15 | shtoom | like that is shows for all the channels |
| 09:05.21 | incog_afk | so... it's a zaptel card? |
| 09:05.25 | shtoom | and the calls are dropped |
| 09:06.06 | *** join/#asterisk Dovid (n=Dovid@bzq-79-182-143-236.red.bezeqint.net) |
| 09:07.25 | shtoom | Wildcard TE12xP |
| 09:08.41 | *** join/#asterisk psk (n=psk@golia.caltanet.it) |
| 09:08.48 | incog_afk | Here's my off-the-cuff guess... It sounds like your PRI span is dropping in and out of alarm. A common cause for this would be a timing synchronization problem. Do you have CPE equipment from the phone company, that you tie into? |
| 09:08.56 | *** join/#asterisk eharris_ (n=eharris@75-43-20-21.lightspeed.austtx.sbcglobal.net) |
| 09:10.05 | incog_afk | (I don't actually have any telephone equipment, on my Asterisk configuration, so I'm drawing from memory, from when I used to be a telco guy. I'm an unemployed nobody, today...) |
| 09:10.15 | Dovid | when using this script (http://bugs.digium.com/view.php?id=10523) which one is the correct one to use ? |
| 09:10.33 | shtoom | there is a box given by telco to which we plug into |
| 09:10.46 | shtoom | the line is working fine with siemens PBX equipment |
| 09:10.55 | *** part/#asterisk oej (n=olle@soll4-125.cust.blixtvik.net) |
| 09:10.58 | shtoom | the only problem is with asterisk |
| 09:11.23 | shtoom | asterisk guru says to replace the card but they are not sure about the solution they have given |
| 09:11.25 | incog_afk | do you have more than one zaptel card? a spare, perhaps? |
| 09:11.52 | shtoom | no, thats the only card available at that location |
| 09:11.55 | incog_afk | I was just going to suggest that the zaptel timing source may be flaky... |
| 09:12.33 | incog_afk | Dovid - on Debian, you really are better off using aptitude to get the package dependencies that you need... |
| 09:12.57 | shtoom | when i am checking with zttool is shows Sync Source: Inernally Clocked |
| 09:13.02 | shtoom | is that a good thing |
| 09:13.02 | incog_afk | Zaptel cards rely upon an internal timing source... |
| 09:13.21 | incog_afk | It's a good thing, as long as the timing source is stable. |
| 09:13.45 | incog_afk | It's absolutely required, for conferencing... |
| 09:13.58 | Dovid | icong_afk: I am using CentOS. |
| 09:14.10 | Dovid | the bug tracker seems to say that is for CentOS as well |
| 09:14.24 | adeel | is it possible to boost the volume on zap-to-sip call, other than increasing the gain on the zapata channel? |
| 09:15.53 | shtoom | incog_afk:if the card is internally clocked why people say to sync with remote end |
| 09:16.54 | shtoom | I am talking about timing source option in zaptel.conf , Sync Soure and timing source are they two different things ? |
| 09:18.28 | incog_afk | It says that, yes... here's why I think that it's better to use aptitude, when possible - this applies to other systems, as well. A lot of times, these scripts make assumptions about version numbering, that are no longer valid for the version of Asterisk that you have installed. I usually look at the make-setup (am I remembering the name correctly?) menu to see what dependencies I'm missing, and then install whatever ius current for my... |
| 09:19.00 | incog_afk | ...operating system, myself. That ensures that I get whatever the most current versions are, for my environment. |
| 09:19.51 | *** join/#asterisk Azam (n=azamzia@58-65-160-140.nayatel.pk) |
| 09:20.49 | incog_afk | shtoom - typically, in a 'synchronous' circuit (like a digital PRI), both ends of the circuit need to know how long a signal should last, and must maintain synchronized timing of each time period. Let me try to explain that... |
| 09:20.57 | Azam | Hi can we use asterisk behind NAT? my asterisk is trying to send a call to another asterisk server on global ip. |
| 09:21.51 | shtoom | incog_afk:ok |
| 09:22.25 | incog_afk | You know how long a second is. You need to tell someone to do something, and you have to send them a tone, for a one-second duration. They are listening, but they are examining each second separately. If you send the tone at the wrong time, they'll here two half-second tones, and not one full second tone... |
| 09:23.25 | incog_afk | So, you need to start the tone at the beginning of one of their one-second intervals, and end it at the end of the interval. To do that, you need to know when those intervals begin and end. That's call 'timing synchronization'... |
| 09:23.28 | *** join/#asterisk Grizzy (n=Generic@adsl-76-204-180-205.dsl.pltn13.sbcglobal.net) |
| 09:24.24 | shtoom | ok |
| 09:24.51 | incog_afk | If both of you have separate timing sources, and they are synchronized to one another, then you're good - as long as they are reliable. If one of them starts to think that a second is a little longer than the other, then it is said to be 'drifting' - and it needs to be resynchronized. |
| 09:24.52 | *** join/#asterisk mmmToop (n=michaelt@firewall.datapro.co.za) |
| 09:25.25 | shtoom | ok |
| 09:26.02 | *** join/#asterisk ToTo (n=ToTo@209.8.41.79) |
| 09:26.37 | incog_afk | All timing sources have *some* drift. As a result, all synchronous circuits will resynchronize, from time to time. If a systems timing drifts too far, too fast, then resynchronization fails, and a synchronous circuit will fail, while it tries to reinitial itself (as if it was just activated). |
| 09:26.50 | incog_afk | That is typically where you get your alarm from. |
| 09:27.19 | shtoom | ok |
| 09:27.41 | incog_afk | Does that help you to understand what might be going wrong? Remember, this is all off-the-cuff, since I haven't actually seen what's going on in your installation... |
| 09:28.26 | shtoom | that really helps to understand a few things about synchronous circuits |
| 09:28.49 | shtoom | so do you expect this yellow alarm is a result of drifting ? |
| 09:28.50 | *** join/#asterisk Sajjad_Ali_Musht (n=Sajjad_A@octroi.enst-bretagne.fr) |
| 09:29.00 | incog_afk | And just think - the job that the world seems to think that I'm best qualified for, is 'truck driver'... :-P |
| 09:29.52 | incog_afk | That's a wild-assed guess, but it's a common cause, yes. To test it, I'd start by replacing the timing source - it's in a daughter-card on the Zaptel, as I recall... |
| 09:31.43 | *** join/#asterisk psavva (n=psavva@athedsl-123937.home.otenet.gr) |
| 09:31.47 | *** join/#asterisk stkn|work (n=stkn|wor@gentoo/developer/pdpc.active.stkn) |
| 09:32.10 | *** part/#asterisk psavva (n=psavva@athedsl-123937.home.otenet.gr) |
| 09:32.22 | shtoom | what is exactly meant by yellow alarm ? when you say red alarm no line is connected right? |
| 09:32.31 | incog_afk | You can tell from looking at the card, sometimes. A lot of timing sources rely upon a crystal - it'll look like a little metal cylinder, usually next to some sort of a chip. You used to be able to just look for a 555 timer, but any more, they build them into other circuits, on other chips, and it's hard to tell where they're hiding. |
| 09:33.13 | JT | incog_afk: btw, PRI is plesiochronous not synchronous |
| 09:33.35 | incog_afk | Red usually means that something is not connected at all. A carriet signal is down, or some voltage is not present - something like that, that the circuit does not think that it can recover from, on it's own. |
| 09:34.16 | incog_afk | Jump on in, any time, JT, and explain the difference... :) |
| 09:34.26 | incog_afk | *carrier |
| 09:34.28 | Grizzy | does plesiochronous mean it has sync bits? |
| 09:34.48 | JT | no |
| 09:35.02 | JT | it means it's "nearly" synchronised |
| 09:35.20 | incog_afk | Remember... I came here with a question, too - so I certainly don't know it all... :) |
| 09:35.52 | shtoom | <PROTECTED> |
| 09:35.53 | JT | the equipment in a plesiochronous network is meant to be able to handle the odd bit slip |
| 09:36.48 | incog_afk | I'd guess that your circuit is drifting more than the odd bit slip... :) But it's still just a raw guess, not having seen anything... |
| 09:37.00 | tzafrir | A yellow alarm means that the other side reports a red alarm ("line disconnected") |
| 09:37.03 | jblack | plesiochronous sounds like a kind of dinosaur. Did they eat velociraptors? |
| 09:37.30 | Grizzy | snort-laugh |
| 09:37.47 | incog_afk | The yellow alarm implies an error that the circuit can recover from, on it's own. It may interrupt service, but the interruption is expected to be temporary. |
| 09:38.15 | incog_afk | That's why it points (often) to timing... |
| 09:38.22 | tzafrir | incog_afk, hmm.... I'm not really sure |
| 09:38.31 | JT | don't trust zaptel to necessarily follow conventions re alarms |
| 09:38.45 | tzafrir | it can also be plain bad cable |
| 09:39.51 | incog_afk | Since I haven't got a zaptel card to play with, I can't try to reproduce the conditions. And, if zaptel does not follow the convention in alarm labelling, well - I can't determine that, either, without having seen one. |
| 09:41.03 | shtoom | tzafrir: thanks for yellow alarm explanation |
| 09:41.34 | incog_afk | Bad cable could cause it, yes, or a badly-seated daughter card - or a variety of other things. But if it was working before, and it's not working, now, replacing the card usually runs you through most of these other possibilities. You'll reseat the card, but replacing it, and reseat the daughter card, but putting in a newly populated one, etc... |
| 09:42.18 | incog_afk | If it's a production system, I'd rather do my analysis, on the bench - and get the phone system back online as fast as possible... :) |
| 09:43.32 | incog_afk | Most places lose a lot more money from having phones offline for an hour, then the equipment to support them would cost... |
| 09:45.27 | incog_afk | So... anyone wanna help me to figure out what the heck is wrong with my SIP client configuration for FWD? :) |
| 09:48.04 | incog_afk | Feh. You people suck... :-P |
| 09:49.59 | jblack | incog_afk: It's fwd that sucks. |
| 09:50.11 | shtoom | <PROTECTED> |
| 09:51.12 | shtoom | the yellow value in the zaptel.conf in span definition got any thing to do with this ? |
| 09:52.58 | incog_afk | I'm trying to get incoming calls from FreeWorldDialup... I have Asterisk behind NAT, and I can't seem to get *any* inbound traffic to the Asterisk box on an inbound call, so I have no packet trace to look at. I have it working with ViaTalk, but not FreeWorldDialup... :( |
| 09:54.00 | jblack | incog_afk: Try ~sipnat |
| 09:54.08 | incog_afk | no idea about the zaptel.conf. Never used it, because I haven't got any cards... :( |
| 09:54.22 | incog_afk | where, jblack? |
| 09:54.27 | jblack | FWD is a pain under the most simple of setups. Multiplying the problem with nat won't make it easier. |
| 09:54.31 | jblack | ~sipnat |
| 09:54.31 | 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 |
| 09:56.25 | incog_afk | excellent... I'll examine that. I'm actually using IAX for one of my FWD accounts, and that's working great. But I can't seem to get FWD to authorize my other account, so that I can register it with their IAX server - so I'm forced to try to get SIP working... |
| 09:56.49 | jblack | I wasn't able to get IAX working with fwd on my three fwd accounts. |
| 09:57.09 | jblack | People regularly report fwd does iax for some accounts, and doesn't do it for others. |
| 09:57.53 | shtoom | <PROTECTED> |
| 09:58.14 | jblack | Thus why most people won't spend any effort to help you get fwd working. Their systems are a mess. |
| 09:58.18 | incog_afk | I was fortunate to have requested it on one of my accounts in 2004. I guess that they finally got around to it, sometime between then and last week, when I finally tried to authenticate into it, again. I don't think I had tried since 2005... |
| 09:59.02 | *** join/#asterisk mikkel (n=mikkel@84.238.113.66) |
| 10:00.23 | incog_afk | I shouldn't need to port-forward. I have SIP working, with NAT, on ViaTalk - so it's obviously possible. Oh, and I'm on the FWD SIP phone test team, so if it's just that FWD is messed up, then all I have to do is find out how they are misconfigured, at their end, and I can get it fixed. |
| 10:03.37 | incog_afk | I'm going to do a write-up on configuring Asterisk for SIP from behind a firewall, when I get it working. I *should* be able to get it added to the configuration guides, on the website... assuming that I can get something to work... :) |
| 10:03.58 | jblack | That's what ~sipnat is |
| 10:05.09 | incog_afk | If only they would link to it, from the FWD site... :) I hope that this works. It wouldn't be the first such write-up that didn't. There was one like it, on voip-info.org, that was a huge waste of time and energy... |
| 10:05.55 | *** join/#asterisk sergey (n=sergey@91.189.233.66) |
| 10:08.24 | incog_afk | yeah... I should not need to port-forward. I don't need to, for ViaTalk (BYoDVoIP)... This tutorial tells me I have to port-forward, and I'm not seeing any reason why that would be necessary for one VoIP provider, and not the other... |
| 10:08.29 | jblack | Like I said, fwd is a big mess. |
| 10:08.29 | Alexandre_fr | Hello |
| 10:09.19 | incog_afk | Okay... what exactly are they doing differently, that makes port-forwarding necessary? If I can determine that, I can pass the word back to them, and they can fix it... |
| 10:09.45 | incog_afk | If it's a missing RTP packet header, or something, that can be fixed... |
| 10:09.57 | jblack | Nothing that I know of. SIP for them works fine for me. |
| 10:10.05 | Alexandre_fr | i'm using softphone, and I would like to know if there is a way to indicate to the peaple in a same callgroup-pickupgroup that there is a call that they can pickup |
| 10:10.32 | Alexandre_fr | beacause with softphone we don't here the phone ringing |
| 10:10.33 | XnOSX | anyone here have a nice sh file for /etc/init.d refer asterisk? |
| 10:10.40 | jblack | type=friend insecure=port allow=ulaw username=myusername secret=mypassword |
| 10:10.46 | incog_afk | ordinarily, you ring the other phones simultaneously with yours... |
| 10:11.02 | jblack | And for register... register => username:password@fwd.pulver.com |
| 10:11.06 | jblack | It works fine for me. |
| 10:11.07 | *** join/#asterisk KermitTheFragger (n=siepkes@118-197.bbned.dsl.internl.net) |
| 10:11.12 | XnOSX | i want to run my asterisk service inmediately when my debian start near init.d! |
| 10:11.25 | XnOSX | anyone have a nice .sh for these! |
| 10:11.27 | jblack | xnosx: Leave it starting at runlevel 2, please. |
| 10:11.53 | jblack | starting it in runlevel 1 isn't going to give you what you want. |
| 10:12.20 | XnOSX | ummm |
| 10:12.37 | incog_afk | are you running asterisk behind NAT, jblack? |
| 10:12.39 | Silent-X | What are some ready to run asterisk systems other then AsteriskNow and Trixbox? |
| 10:12.45 | shtoom | asterisk is good at working with clients behind NAT but when it comes to asterisk being behind NAT you'll hit the hard ground. Last time when I've checked asterisk don't have full STUN support. |
| 10:12.49 | jblack | incog_afk: Nope. I have enough headaches. |
| 10:13.50 | shtoom | and most of clients like xlite , eyebeam would work behind NAT with FWD cause they have good STUN support. |
| 10:14.26 | shtoom | however there are a few hacks that you can do I guess like look for externip option in sip.conf |
| 10:14.30 | incog_afk | Asterisk has no STUN client, shtoom, you're right. But it *does* support outbound proxying, and to be honest, the only thing I needed to do for ViaTalk's SIP server, was enable nat=yes in the relevant section of sip.conf... |
| 10:15.33 | incog_afk | Try Asterisk@Home, Silent-X? |
| 10:18.16 | shtoom | <PROTECTED> |
| 10:20.34 | incog_afk | It seemed to, at least for ViaTalk. You have to set it up as a peer, though, rather than a friend - I'm guessing that's what makes it two-way communication. But for some reason, FWD isn't sending me INVITE requests (or anything else) on inbound calls... |
| 10:23.04 | incog_afk | I'm still registered - I can see that in the CLI. I'm just not getting any traffic on the inbound call... |
| 10:23.27 | incog_afk | (It's pissing me off, so I know it's something stupid...) |
| 10:23.56 | Dovid | asterisk trunk is what ? (is it testing or is it stable ?) |
| 10:25.38 | incog_afk | 1.4.17 - I haven't installed 1.4.18 yet. I tried 1.6.0-beta2, but it didn't run my extensions.conf the same way as 1.4, and there doesn't seem to be adequate documentation on the changes, to permit me to update my configuration, so I had to downgrade back down... |
| 10:29.29 | *** join/#asterisk inadaptado (n=matias@32.59.64.129) |
| 10:29.36 | Dovid | ? |
| 10:29.49 | inadaptado | ! |
| 10:34.41 | tzafrir | 1.6.* releases will be made directly from trunk (as has been with the betas) |
| 10:35.01 | *** join/#asterisk ming_zym (n=ming_zym@124.14.232.25) |
| 10:35.36 | tzafrir | Testing has 1.4.17. Stable has 1.2.13 ;-) |
| 10:36.15 | *** join/#asterisk mkl1525 (n=qwertz@89.246.168.197) |
| 10:37.12 | mkl1525 | Hi, (1.4) having problems using AgentCallBackLogin. When there's already a caller in the queue and an agent tries to login * says " -- Agent/203 is busy |
| 10:38.06 | mkl1525 | and logs him out - that's not wantedt cause so as long as a caller is in the queue no new agent can login - anybody knows this and has a fix/workaround? |
| 10:39.48 | incog_afk | New for upcoming Asterisk v1.4.0 release: (July 2006) Due to various issues with AgentCallbackLogin this feature is likely to be deprecated by Digium (according to Kevin P. Fleming). Similar functionality can be achieved through existing dialplan functionatliy using dynamic members. |
| 10:40.18 | incog_afk | from http://www.voip-info.org/tiki-index.php?page=Asterisk%20cmd%20AgentCallbackLogin |
| 10:41.40 | incog_afk | just to cover the obvious... you do have a zaptel card installed, right? |
| 10:44.05 | mkl1525 | incog_afk thanks I know the statement, problem is that this is a working environment where I've to fix the login problem |
| 10:44.47 | incog_afk | this is a new problem in an existing (previously working) environment? |
| 10:46.43 | mkl1525 | we experienced it just yet, don't know if it previously existed |
| 10:47.44 | *** join/#asterisk sergey (n=sergey@91.189.233.66) |
| 10:49.14 | incog_afk | hmmm... Have you done a debug trace? I'm curious as to what exactly is failing, at the protocol level... |
| 10:49.59 | incog_afk | Lots of things can result in a 'busy' condition... :-| |
| 10:50.25 | Dovid | hi: I am trying to apply the patch http://bugs.digium.com/view.php?id=11662 |
| 10:50.25 | *** join/#asterisk fugitivo (n=ajf@201-212-152-100.cab.prima.net.ar) |
| 10:50.33 | fugitivo | hello |
| 10:50.47 | Dovid | i am using asterisk trunk. i am lost as to which file to use to patch asterisk with |
| 10:51.20 | *** join/#asterisk Greek-Boy (n=email@41.221.58.4) |
| 10:53.43 | incog_afk | Those look like souce patches, Dovid... I'd guess that they have to be applied against the source tree. Did you install Asterisk from source, or as something like a package? |
| 10:53.57 | incog_afk | *source patches |
| 10:54.03 | Dovid | i didnt install it yet. I got the latest from SVN Trunk |
| 10:54.24 | Dovid | i got it from http://svn.digium.com/svn/asterisk/trunk/ |
| 10:54.28 | Dovid | vai svn checkout |
| 10:55.28 | Dovid | via* |
| 10:58.41 | incog_afk | Look at each patch - the very first line. That tells you what file the patch is supposed to be applied against. |
| 10:59.52 | incog_afk | I believe that you can run all of these patches from the root of the Asterisk source distribution tree... |
| 11:01.38 | *** join/#asterisk Woifi1988 (n=anon@M1237P001.adsl.highway.telekom.at) |
| 11:01.43 | Woifi1988 | hi all |
| 11:02.11 | *** join/#asterisk psk (n=psk@golia.caltanet.it) |
| 11:02.34 | incog_afk | For example: patch -p0 <app_voicemail.c.patch |
| 11:03.07 | Woifi1988 | anyone who know how to test asterisk load when converting codecs? Please don't say sipp ;-> i tried that for a long time |
| 11:03.19 | Silent-X | incog_afk: didnt asterisk@home become trixbox or something, used it long ago |
| 11:03.56 | Woifi1988 | Silent-X: yes thats right |
| 11:03.58 | *** join/#asterisk fordfrog (n=fordfrog@gentoo/developer/fordfrog) |
| 11:04.20 | incog_afk | Dunno - I just know that I have a CD in my library for it... Never used it, to be honest - one of my freinds did, and insisted I get a copy... |
| 11:04.28 | *** join/#asterisk sim-NIX (n=user@82-204-21-111.dsl.bbeyond.nl) |
| 11:08.36 | *** join/#asterisk steliosk (n=Stelios@athedsl-121101.home.otenet.gr) |
| 11:13.52 | *** join/#asterisk steliosk (n=Stelios@athedsl-121101.home.otenet.gr) |
| 11:16.14 | *** join/#asterisk Frogzoo (n=Frogzoo@CPE-124-184-225-88.nsw.bigpond.net.au) |
| 11:16.44 | Frogzoo | anyone tried building a .deb from 1.6? |
| 11:16.47 | badcfe | isnt it possible to enable sip debug for a set of more than _one_ ip address (without tracing it alltogether) |
| 11:16.51 | badcfe | ? |
| 11:17.12 | Woifi1988 | Frogzoo whats the problem? |
| 11:17.22 | Woifi1988 | rpm -i ... |
| 11:17.41 | Frogzoo | open_source/asterisk-1.6.0-beta2/include/asterisk/autoconfig.h -march=i486 " sh build_tools/mkpkgconfig /usr/lib/pkgconfig; fi |
| 11:17.41 | *** join/#asterisk J4zen (n=Jvan4zen@a82-95-153-17.adsl.xs4all.nl) |
| 11:17.44 | Frogzoo | build_tools/mkpkgconfig: 33: cannot create /usr/lib/pkgconfig/asterisk.pc: Permission denied |
| 11:17.59 | Woifi1988 | did you made a sudo su - first? |
| 11:18.06 | Woifi1988 | "sudo su -" |
| 11:18.22 | Frogzoo | Woifi1988: I'm building from 'dpkg-buildpackage -rfakeroot' |
| 11:18.38 | Woifi1988 | ok |
| 11:19.02 | Woifi1988 | why not "sudo su -" ? |
| 11:20.29 | Frogzoo | well, I understand the debian build hardly at all - so I'm working from the debian maintainers guide |
| 11:21.21 | Frogzoo | but clearly when mkpkgconfig is called without setting $(DESTDIR) to anything, meaning mkpkgconfig tries to write to /usr/lib/pkgconfig |
| 11:21.23 | Woifi1988 | and why do you not compile * at your own? |
| 11:21.29 | Woifi1988 | that increases performace |
| 11:21.56 | Frogzoo | Woifi1988: well, part of the .deb build process is to compile source to binaries |
| 11:22.22 | *** join/#asterisk guillote_GNU (n=guillote@host157.201-253-55.telecom.net.ar) |
| 11:23.05 | incog_afk | I can see doing a single build, and transporting it to other machines, to support a server farm without waiting for dozens of parallel compile runs to complete... |
| 11:23.08 | *** join/#asterisk IPGHOST (n=IPGHOST@203.215.176.186) |
| 11:24.06 | Frogzoo | the real advantage to me of a .deb is version control |
| 11:27.28 | incog_afk | I found that I needed to recompile for various module and hardware support... |
| 11:27.52 | *** join/#asterisk flush (n=SYN_SENT@ip216-239-88-218.vif.net) |
| 11:27.56 | flush | ahoyyyy |
| 11:28.04 | *** join/#asterisk GBR_ (n=gbr@200.103.96.98) |
| 11:28.04 | flush | anyone alive arrr |
| 11:28.08 | incog_afk | Not all hosts are equal, in terms of hardware, or functional requirements... :) |
| 11:28.10 | flush | i got questions |
| 11:28.17 | incog_afk | Me too... :-P |
| 11:28.20 | *** join/#asterisk nvrpunk (n=root@81.90.21.227) |
| 11:28.26 | flush | what about that TDM400P pci card |
| 11:28.36 | nvrpunk | anyone know if astbill will work with asterisk 1.4? |
| 11:28.41 | flush | should i set it 4 FXS modules, and use my voxzone x100p for FXO |
| 11:28.57 | flush | or use 3 FXS and 1 FXO on the TDM400P and throw that voxzone FXO card at the garbage |
| 11:29.12 | incog_afk | depends upon your needs, flush... |
| 11:29.25 | flush | i never set up a pbx box before |
| 11:29.34 | flush | i heard asterisk is the cool, so here i come |
| 11:30.04 | flush | my dream would be you know.. like people call home and theres a choice like "lahblah press 1, bahlbhlah press 2", when i put on hold there is music.. is it possible |
| 11:30.05 | incog_afk | I'm pretty sure that you can only do 4/0, 2/2, or 0/4 port FXS/FXO configurations on that card, anyhow... |
| 11:30.28 | flush | oh |
| 11:30.30 | incog_afk | Those tasks are all possible, with Asterisk, yes... |
| 11:30.46 | flush | no matter of the card ? i mean, it would work with the tdm400p |
| 11:30.58 | incog_afk | The card has two daughter-card slots, for the FXS/FXO configurations... |
| 11:31.03 | flush | whats the deal in using another pci card like my voxzone for FXO ? |
| 11:31.26 | flush | hrm.. i see |
| 11:32.18 | flush | but.. if i use 4 FXS modules on the TDM400P and get the voxzone x100p which has one FXO.. would there be any issue or it would work all good just like if i had the TDM400P card with a FXO module on it |
| 11:33.14 | flush | my point is is there any problem in using 2 cards, one for FXO and one for FXS if they are both 100% asterisk compatible.. can i loose some options or whatever cause it's not a single hardware ? |
| 11:33.31 | incog_afk | I'm not aware of any such issues, no. |
| 11:34.00 | flush | nice, so do you think this is good; |
| 11:34.18 | flush | i get the TDM400P with 4 FXS modules to use normal phones and plug them into it, and the voxzone x100p to connect to the pstn |
| 11:35.11 | flush | brb need coffee, its 6h35 am here |
| 11:36.40 | incog_afk | Sounds good to me. I wish I had the equipment. I'm running entirely on VoIP, for my configuration (which sounds very similar, in terms of the IVR application). |
| 11:37.50 | *** join/#asterisk McPaul (i=MrPaul@79.81.57.210) |
| 11:38.02 | *** part/#asterisk McPaul (i=MrPaul@79.81.57.210) |
| 11:39.36 | Frogzoo | tdm400p supports any combination of fxo/fxs - 3x fxs / 1 x fxo is what I have here |
| 11:40.43 | incog_afk | really? I thought it only had two daughter-card slots? Am I thinking of a different card? |
| 11:40.55 | Frogzoo | I think so - 4 daughter cards |
| 11:41.53 | flush | nice, so 4 fxs on the tdm400p and my 1 fxo voxzone card is all good for what i want to do? |
| 11:41.57 | flush | im about to bought it on ebay |
| 11:42.23 | Frogzoo | no idea what a voxzone card is |
| 11:42.39 | Greek-Boy | if I use Authenticate() before Background() will dtmf digits be considered the pin code? |
| 11:43.27 | flush | http://www.daniweb.com/forums/thread64965.html |
| 11:43.36 | flush | this is the voxzone card.. |
| 11:43.43 | flush | its 35 bucks on ebay |
| 11:43.47 | tzafrir | a x100p |
| 11:43.56 | flush | yes.. |
| 11:43.58 | tzafrir | clone |
| 11:44.02 | tzafrir | whatever |
| 11:44.10 | flush | thats okay or bad |
| 11:44.35 | tzafrir | I don't really care about the word "clone" here. |
| 11:44.42 | flush | they say its not a clone.. |
| 11:44.44 | flush | http://cgi.ebay.ca/Voxzone-X100P-OEM-FXO-PCI-Card-for-Asterisk-PBX_W0QQitemZ220198514300QQihZ012QQcategoryZ61839QQssPageNameZWDVWQQrdZ1QQcmdZViewItem |
| 11:44.46 | tzafrir | The problem is that the original wasn't the grandest |
| 11:44.57 | incog_afk | I'm impressed - but I'm also unemployed, so $35 is either groceries for a couple of weeks, or a cool card. Guess which way I'm going to go... |
| 11:45.18 | flush | i know what youre talking about man |
| 11:45.26 | stimpie | groceries are so overrated |
| 11:45.27 | tzafrir | lately I heard good success stories regarding the echo with x100p-s |
| 11:45.34 | tzafrir | using oslec |
| 11:46.01 | incog_afk | The kids get mad when I buy toys for me instead of food for them... |
| 11:46.06 | badcfe | i have a mysterious observation on sip reinvite between a couple of asterisks. where should i pastebin the sip trace? |
| 11:46.11 | flush | lmao |
| 11:46.17 | tzafrir | flush, BTW: if the 35$ does not include the delivery, then that card is probably over-priced |
| 11:46.35 | flush | actually its something like 20$ + 10$ shipping to canada |
| 11:46.54 | incog_afk | On eBay, the delivery charge is usually where the seller makes their profit margin... |
| 11:47.37 | flush | and if i get the tdm400p with 4 fxs modules, can i later buy a single fxo module and plug it onto it? is it how it works? |
| 11:48.26 | tzafrir | looks OK, basically |
| 11:49.01 | tzafrir | regarding the "Wildcard X100P vs. "Generic clone" - that's just playing with PCI IDs |
| 11:49.39 | incog_afk | modules are basically daughter-cards. Remove and replace, like PCI cards in a PC... |
| 11:50.10 | flush | awesome |
| 11:50.10 | Frogzoo | flush: maximum 4 slots on the tdm400p |
| 11:50.17 | flush | yea |
| 11:50.32 | flush | cool folks thanks a lot, im going to courses before im late, bbl have a nice day |
| 11:50.54 | tzafrir | in fact, the two extra lines they include at the top are not even from the loading of the wcfxo driver |
| 11:51.46 | tzafrir | hmm.... hang on.... it's from singapure? |
| 11:52.10 | tzafrir | it reminds me of the x100p card I bought from ebay |
| 11:52.10 | incog_afk | I'm not convinced that I trust the seller... |
| 11:52.12 | *** join/#asterisk chris_1 (n=chris@ng1.kurtkrenn.com) |
| 11:52.19 | tzafrir | and that was actually substandard |
| 11:52.20 | badcfe | is pastebin.ca the way to go? |
| 11:52.40 | tzafrir | based on si3035 rather than on si3034 |
| 11:52.57 | tzafrir | flush, have you already ordered that card? |
| 11:53.11 | incog_afk | I think he ran off for classes... |
| 11:53.32 | chris_1 | i looking for a working example how to use txfax |
| 11:56.37 | badcfe | anyway, heres my problem a trace and description. http://rafb.net/p/m7WgUf78.html |
| 11:57.20 | Greek-Boy | I want to use Authenticate() and play a sound file but if a user inputs the password/pin it should stop playing and go to the next step |
| 11:57.40 | Greek-Boy | similair to Background() but with the input expected for Authenticate() |
| 11:58.41 | fugitivo | Greek-Boy: use background() |
| 11:58.44 | *** join/#asterisk keulin (n=cray@ifth-pdcr2.infotheme.fr) |
| 11:59.24 | Dovid | anyone here from Israel ? |
| 12:00.39 | Greek-Boy | fugitivo: before or after Authenticate() ? |
| 12:01.18 | tzafrir | Besides me and my other two instances? |
| 12:01.29 | incog_afk | instead of Authenticate, Greek-Boy... |
| 12:02.05 | *** join/#asterisk michael-i (n=michael-@141.41.40.225) |
| 12:02.48 | *** join/#asterisk J4zen (n=Jvan4zen@a82-95-153-17.adsl.xs4all.nl) |
| 12:03.00 | *** join/#asterisk saftsack (n=oliver@p54A7C703.dip.t-dialin.net) |
| 12:03.41 | incog_afk | I don't see where this is failing, badcfe... |
| 12:04.23 | *** part/#asterisk adeel (n=adeel@c-24-7-132-23.hsd1.ca.comcast.net) |
| 12:06.31 | Greek-Boy | incog_afk: in that case how will I secure the extension? I need the user to Authenticate |
| 12:06.41 | *** join/#asterisk saftsack (n=oliver@p54A7CD71.dip.t-dialin.net) |
| 12:07.24 | *** join/#asterisk ruied (n=ruied@pal-213-228-184-21.netvisao.pt) |
| 12:07.26 | *** join/#asterisk FlatFoot (n=bigflatf@80.88.192.83) |
| 12:07.32 | michael-i | Greetings everyone...I was wondering if anyone has experienced SIP registration problems since moving from 1.4.17 to 1.4.18? |
| 12:07.35 | incog_afk | use a GotoIF to change the call processing flow, based upon whether the value entered by the user, matches your Authenticate value... |
| 12:08.07 | incog_afk | Haven't upgraded, yet. IT just came out, like this weekend... :) |
| 12:08.30 | *** join/#asterisk cpm (n=Chip@pdpc/supporter/active/cpm) |
| 12:08.45 | michael-i | I know :) i just replaced the core of AskoziaPBX with 1.4.18 and now some people are reporting registration failure. Naturally, I can't reproduce this myself... |
| 12:09.41 | incog_afk | Naturally... :) |
| 12:09.56 | *** join/#asterisk ming_zym (n=ming_zym@123.103.29.252) |
| 12:10.12 | incog_afk | How are they backing up these reports? Just as observations on user SIP phones? |
| 12:10.41 | incog_afk | I mean, you should be able to see these failures, in your CLI on the Asterisk box... |
| 12:10.59 | *** join/#asterisk windback (n=windback@proxy.unc.edu.ar) |
| 12:11.15 | michael-i | It only seems to be happening with provider accounts. The only submissions I've received are "chan_sip.c:7425 in sip_reg_timeout:" messages. |
| 12:11.52 | *** join/#asterisk af_ (n=getsmart@88-149-240-211.dynamic.ngi.it) |
| 12:12.30 | incog_afk | Have you done a "sip set debug", to see what's timing out? |
| 12:13.10 | michael-i | I haven't had the chance yet as these aren't my boxes. Hopefully more debug info will come in today. |
| 12:14.39 | incog_afk | You're going to need it... :-| If the same configuration gives new results, because of a version change, then there is probably a change in the behavior of one of your configuration items, in the upgrade. To find it, you'll want to know exactly how it's failing... |
| 12:17.34 | *** join/#asterisk DrAk0 (n=thinkpad@nelug/coreteam/luisjose) |
| 12:18.31 | *** join/#asterisk SteveTotaro (n=Elizabet@c-69-243-124-5.hsd1.md.comcast.net) |
| 12:21.07 | *** join/#asterisk Woifi1988 (n=anon@M1237P001.adsl.highway.telekom.at) |
| 12:21.18 | Woifi1988 | when i dial a number and immediatly hang up my asterisk goes on with dialing and never kills the call! Any solution for that? |
| 12:21.42 | *** join/#asterisk waKKu (n=eurulo@unaffiliated/wakku) |
| 12:21.43 | SteveTotaro | ~pb |
| 12:21.43 | jbot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste |
| 12:22.49 | incog_afk | Dial(SIP/somecontext/extension,60,r) - the "60" refers to the number of seconds to try before timing out. |
| 12:23.13 | Woifi1988 | what does the r do? |
| 12:24.05 | incog_afk | it forces a rignback tone, so that you don't hear things like a rign before a busy signal, etc... |
| 12:24.11 | incog_afk | *rangback |
| 12:24.14 | incog_afk | **ringback |
| 12:24.45 | Woifi1988 | ah i see! |
| 12:25.00 | Woifi1988 | but why doesn't asterisk just hang up when i hang up? |
| 12:26.14 | incog_afk | Most likely, Asterisk is not aware of the dropped call at your end. This is often the result of a terminal adapter that doesn't send the disconnect signalling unless the call has been supervised, or some such similar issue. |
| 12:27.17 | *** join/#asterisk saftsack (n=oliver@p54A73ADF.dip0.t-ipconnect.de) |
| 12:27.26 | saftsack | hi, how is the standard asterisk webgui called? |
| 12:27.49 | Woifi1988 | incog_afk: But it's just a softphone which terminates the call! |
| 12:27.49 | incog_afk | AsteriskNOW? |
| 12:28.04 | saftsack | incog_afk, thanks :) |
| 12:28.39 | incog_afk | So, how sure are you that the softphone will send a disconnection request on a call that was never supervised? |
| 12:29.04 | incog_afk | no problem, saftsack... :) |
| 12:29.12 | *** join/#asterisk ^shark_ (n=^shark_@217.194.147.193) |
| 12:29.28 | Woifi1988 | incog_afk Very sure! I have sniffed to proove! |
| 12:29.42 | Woifi1988 | i also works when i hang up some time later |
| 12:30.00 | *** join/#asterisk windback (n=windback@proxy.unc.edu.ar) |
| 12:30.53 | *** join/#asterisk Al_WinKiller (n=alex_pas@mgm.cornet.am) |
| 12:32.24 | Al_WinKiller | hi guys,, I have two asterisk server asterisk_1 and asterisk_2 with some phone connected to them,,, I have established connection between server with sip ( have tried with IAX , worked good, but need SIP ),,,, and when I dial from server 1 to server 2 it says that extension is wrong :( can someone help me ? |
| 12:33.16 | Al_WinKiller | on server 1 I put this in extensions.conf |
| 12:33.22 | incog_afk | My bet would be that Asterisk is not getting this signalling. Not having your equipment here, I can't prove that, but if you watch a debug trace, you should be able to find it. If the signalling really is getting through, Asterisk *should* drop the Dial channel. If it's not doing so, you may have a bug report to make, to the developers. It's vaguely possible that Asterisk doesn't drop the Dial channel if it hasn't been supervised, yet... |
| 12:33.35 | Al_WinKiller | exten => 1299501,2,Dial(SIP/@83.139.051/1299501,30,r) |
| 12:33.35 | Al_WinKiller | exten => 1299501,3,Congestion |
| 12:34.36 | saftsack | incog_afk, asterisknow is a complete distribution. i search for the web interface because i already compiled asterisk |
| 12:35.15 | incog_afk | Extension problems usually come from mismatched registration entries and extension.conf configurations... |
| 12:36.22 | Al_WinKiller | hm,, I did the same what I did by IAX,, but,, it doesn't work :( |
| 12:36.44 | *** join/#asterisk coppice (n=chatzill@175.203.17.210.dyn.pacific.net.hk) |
| 12:37.26 | incog_afk | I don't think that they come separately, saftsack - the web gui relies upon a clear knowledge of the condition of the config files, and the version of the Asterisk software. I believe that the AsteriskNOW software is actually customized, for the GUI interface, such that the standard Asterisk distribution would not work with their GUI... |
| 12:37.52 | *** join/#asterisk Dovid[Laptop] (n=Dovid@bzq-79-182-143-236.red.bezeqint.net) |
| 12:38.20 | saftsack | incog_afk, hmm but i have in mind that the first asterisk 1.4.X versions came with an optional to compile webinterface |
| 12:38.21 | incog_afk | IAX is different, Al_WinKiller - the remote server specifies the extension to handle the inbound call, in the IAX headers. SIP does not do this... |
| 12:38.51 | Al_WinKiller | ok, ,,, how do I fix this problem ? :( |
| 12:39.11 | Greek-Boy | incog_afk so I should use Background() first and then Read() and then Gotoif() ? |
| 12:40.02 | Al_WinKiller | loog, I put this in extensions.conf |
| 12:40.03 | Al_WinKiller | exten => 1299501,1,Dial(SIP/CentOS:test123@83.139.0.51/1299501,30,r) |
| 12:40.03 | Al_WinKiller | exten => 1299501,2,Congestion |
| 12:40.22 | Al_WinKiller | and I got this in debug |
| 12:40.23 | Al_WinKiller | [Feb 12 15:29:34] WARNING[14037]: chan_sip.c:2898 create_addr: No such host: 83.139.0.51/1299501 |
| 12:40.23 | Al_WinKiller | [Feb 12 15:29:34] WARNING[14037]: app_dial.c:1191 dial_exec_full: Unable to create channel of type 'SIP' (cause 3 - No route to destination) |
| 12:41.05 | *** join/#asterisk anonymouz666 (n=anonymou@201.19.127.250) |
| 12:41.14 | incog_afk | I'm not aware of that interface, saftsack - if there is one, it will be configurable in the make menuselect... |
| 12:41.30 | saftsack | didnt found it there :( |
| 12:41.47 | Al_WinKiller | man,,, I need a cigarette :( |
| 12:41.57 | incog_afk | Background isn't necessary, Greek-Boy - Read will play the sound file, for you, too... |
| 12:42.48 | incog_afk | strip the authentication information out of your registration line, in sip.conf, Al_WinKiller, and show me that. |
| 12:43.30 | *** join/#asterisk grEvenX (n=even@1elt2pn.ip.hipercom.no) |
| 12:43.41 | *** join/#asterisk hugolp (n=jo@unaffiliated/hugolp) |
| 12:44.13 | Al_WinKiller | dude,, pls hold on a second,, I really need a smoke :( ( ASTERISK is good,, but hard to configure ),, brb ,, hope you be here and help,, cuz I really need a help |
| 12:44.35 | anonymouz666 | ~book |
| 12:44.35 | 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 |
| 12:44.47 | anonymouz666 | you should learn this |
| 12:44.50 | anonymouz666 | read |
| 12:44.53 | incog_afk | I am curious about 83.139.0.51 - the error seems to indicate that this isn't where your dialed SIP server is... |
| 12:46.40 | incog_afk | If I had to make a bet. anonymouz666, I'd say that English is not his first language. Reading technical books, in English, when you English is spotty, is surprisingly difficult... |
| 12:47.11 | anonymouz666 | my first language is portuguese |
| 12:47.25 | anonymouz666 | so I don't agree with that |
| 12:48.30 | Frogzoo | the book is surprisingly good |
| 12:50.03 | incog_afk | I'd say that your English is considerably better than his, anonymouz666... |
| 12:50.37 | anonymouz666 | I missed too many english classes :D |
| 12:52.38 | kaldemar | Al_WinKiller: your SIP/XXXX... is the problem. you can't use same syntax as with IAX. http://www.voip-info.org/tiki-index.php?page=Asterisk%20SIP%20Channels |
| 12:53.40 | Al_WinKiller | let me chek it |
| 12:56.40 | Al_WinKiller | look, I put in my extensions.conf on server 1 this "exten => s,1,Dial(SIP/83.139.0.51:9999,20)" |
| 12:56.43 | *** join/#asterisk RoyK (n=roy@fw.fortel.no) |
| 12:56.45 | Al_WinKiller | and I get this fout |
| 12:56.46 | *** join/#asterisk dominic1 (n=dob@213.221.82.242) |
| 12:56.49 | dominic1 | tenbelegung nicht? |
| 12:57.01 | *** part/#asterisk dominic1 (n=dob@213.221.82.242) |
| 12:57.02 | Al_WinKiller | *CLI> [Feb 12 15:45:49] NOTICE[14119]: chan_sip.c:13859 handle_request_invite: Call from '1299401' to extension '1299501' rejected because extension not found |
| 12:57.18 | Al_WinKiller | it doesn't reach server 2 |
| 12:57.22 | *** join/#asterisk ZaVoid (n=zavoid@66-95-182-90.client.dsl.net) |
| 12:57.46 | SteveTotaro | ~book |
| 12:57.47 | 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 |
| 12:59.50 | incog_afk | you need to define extension 1299401 in whatever context is supposed to handle incoming calls... |
| 12:59.57 | *** join/#asterisk javar (n=javar@69.79.134.24) |
| 13:00.29 | *** join/#asterisk nebojsajsimic (n=nebojsaj@cable-89-216-16-106.static.sbb.co.yu) |
| 13:01.06 | incog_afk | Most likely, you are expecting calls on some other extension - probably whatever you were getting from IAX inbound calls... |
| 13:01.16 | *** join/#asterisk flujan (n=flujan@200-160-115-020.static.spo.ctbc.com.br) |
| 13:01.45 | Al_WinKiller | 1299401 is on server 1 ( it works file ) 1299501 is on server 2,,, both of them are OK,, but can't call from one to another via servers ( sip ) |
| 13:02.12 | Al_WinKiller | hm |
| 13:03.22 | incog_afk | I was referring to the error message. Whichever server you are getting it on, is telling you that it doesn't know how to handle the inbound call. Check to make sure that the call is being handled in the context that you think ait should be, and make sure that the extension is defined *in that context*... |
| 13:04.21 | Al_WinKiller | the problem is ,, it doesn't reach second server ( Asterisk 2) it is an error msg form Asterisk 1 ( I am making call from ) |
| 13:04.40 | *** join/#asterisk erojasv (n=erojasv@201.240.231.84) |
| 13:05.42 | Al_WinKiller | look on server Asterisk 1 I put this in extensions.conf exten => s,1,Dial(SIP/1299501@83.139.0.51) |
| 13:05.49 | Al_WinKiller | so it should send it to Asterisk 2 |
| 13:06.01 | Al_WinKiller | but it doesn't cuz I see nothing on debug ( on Asterisk 2 ) |
| 13:06.17 | Al_WinKiller | and Asterisk 1 is registred on Asterisk 2 |
| 13:06.28 | *** join/#asterisk af_ (n=getsmart@88-149-240-211.dynamic.ngi.it) |
| 13:07.52 | Al_WinKiller | man,, I need a cigarette ( cigarettes gonna kill me ,, I know :( ) |
| 13:07.55 | windback | I'm triyng to store CDR on a SQLite dbase |
| 13:07.57 | Al_WinKiller | brb 2 min |
| 13:09.27 | windback | I'm triyng to store CDR on a SQLite dbase, the wiki say: "The module will automatically compile if you have the SQLite library and header files installed" I installed de SQLite package in Debian, but... what is the header file??? |
| 13:09.31 | *** join/#asterisk lirakis (i=lirakis@66.252.24.133) |
| 13:09.42 | incog_afk | This is looking like Asterisk 1 is never sending the call out - looks like you're trying to reach a nonexistent extension on Asterisk 1. Perhaps you meant to reach and extension that would then set up the call with the other server...? |
| 13:10.29 | *** join/#asterisk PepOSX (n=angeldav@190.79.252.29) |
| 13:11.20 | *** join/#asterisk reber (n=reber@193.253.213.73) |
| 13:11.35 | *** part/#asterisk reber (n=reber@193.253.213.73) |
| 13:12.09 | Al_WinKiller | I don't get you ,, asterisk 1 should just send the call,, doesn't it ? and the rest suppose to handle asterisk 2 ? am I right ? |
| 13:13.23 | Al_WinKiller | all it ( asterisk 1 )has to do is send the call out |
| 13:13.24 | Al_WinKiller | ? |
| 13:13.30 | Al_WinKiller | what you think ? |
| 13:13.50 | SteveTotaro | ~pb |
| 13:13.50 | jbot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste |
| 13:13.51 | Al_WinKiller | according to this extension exten => s,1,Dial(SIP/1299501@83.139.0.51 ,20) |
| 13:14.11 | SteveTotaro | remove s |
| 13:14.17 | SteveTotaro | and replace with extension |
| 13:14.21 | SteveTotaro | or pattern match |
| 13:14.38 | SteveTotaro | make sure the incoming call is going to the proper context |
| 13:16.12 | SteveTotaro | s is used if no extension is sent |
| 13:16.24 | Al_WinKiller | ok, hold on a second |
| 13:16.29 | *** join/#asterisk agx (n=AGX@88.34.216.63) |
| 13:17.26 | incog_afk | It looks like he has one set, SteveTotaro, based upon the error... |
| 13:17.28 | *** join/#asterisk sergee (n=serg@voip1.west-call.com) |
| 13:17.59 | SteveTotaro | yes, that is why i said s will not work |
| 13:18.45 | SteveTotaro | he either needs a pattern match or exact extension |
| 13:19.26 | Al_WinKiller | i put exten => 1299501,1,Dial(SIP/1299501@83.139.0.51 ,20) |
| 13:19.30 | Al_WinKiller | and it says |
| 13:19.50 | Al_WinKiller | [Feb 12 16:09:06] WARNING[14426]: chan_sip.c:2898 create_addr: No such host: 83.139.0.51 |
| 13:19.51 | Al_WinKiller | [Feb 12 16:09:06] WARNING[14426]: app_dial.c:1191 dial_exec_full: Unable to create channel of type 'SIP' (cause 3 - No route to destination) |
| 13:19.51 | Al_WinKiller | <PROTECTED> |
| 13:20.14 | Al_WinKiller | :( |
| 13:20.54 | Al_WinKiller | I don't get this one "No such host: 83.139.0.51" |
| 13:21.03 | Al_WinKiller | cuz he is registred by 83.139.0.51 |
| 13:21.04 | Al_WinKiller | :( |
| 13:22.15 | Al_WinKiller | look on Asterisk 1 |
| 13:22.32 | Al_WinKiller | from sip.conf "register => CentOS:123456@83.139.0.51" |
| 13:22.49 | Al_WinKiller | and 83.139.0.51 has accepted him ( Asterisk 2 ) |
| 13:22.50 | Al_WinKiller | :( |
| 13:23.30 | *** join/#asterisk [TK]D-Fender (n=joe@MTRLPQ02-1177745839.sdsl.bell.ca) |
| 13:23.36 | Al_WinKiller | ppl ? help ? pls ? :) |
| 13:24.09 | jameswf-home | ~pb |
| 13:24.10 | jbot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste |
| 13:24.12 | jameswf-home | gah |
| 13:25.29 | incog_afk | I'm curious about that, too, Al_WinKiller - It *looks* as though that ip address isn't where your other Asterisk box is... |
| 13:26.33 | Al_WinKiller | it is Asterisk 1 is on 83.139.12.187 and Asterisk 2 is on 83.139.0.51 |
| 13:26.38 | Al_WinKiller | very siple |
| 13:26.41 | Al_WinKiller | simple |
| 13:27.03 | *** part/#asterisk hugolp (n=jo@unaffiliated/hugolp) |
| 13:28.20 | Al_WinKiller | first one is linux second is bsd |
| 13:29.31 | Al_WinKiller | man,,, I will smoke another cigarette,, and go down ( basement ) and kill the server,, but first I am gonna kick it (servers) ass :D:D:D: |
| 13:31.24 | badcfe | http://rafb.net/p/m7WgUf78.html |
| 13:33.56 | *** join/#asterisk dream_th (n=dream_th@82.114.68.154) |
| 13:35.39 | incog_afk | Is there a NAT router in between these Asterisk boxes, Al_WinKiller? Maybe one in front of each? As if you were trying to link up PBXes for offices in different cities? |
| 13:35.51 | windback | When I try to include the module cdr_sqlite in the make menuselect from tha asterisk compilation, it have XXX and tell me that it depends on sqlite(E), What it means??? |
| 13:36.19 | Al_WinKiller | no, no NAT |
| 13:36.33 | windback | I have installed sqlite package, but nothings hapened |
| 13:36.42 | SteveTotaro | windback it means you have to install dependancies |
| 13:36.53 | SteveTotaro | you are missing something |
| 13:36.56 | michael-i | windback: the sqlite-dev package as well? |
| 13:36.56 | Al_WinKiller | no, both of them are downstairs ( datacenter ) I am working in ISP |
| 13:37.02 | dream_th | Hi, how to resctrict a trunk/outbound route so only specific extensions can use |
| 13:37.40 | Al_WinKiller | if I just got an error on Asterisk 2,, |
| 13:37.41 | Al_WinKiller | :( |
| 13:38.33 | [TK]D-Fender | dream_th: Put them in different contexts |
| 13:38.47 | *** join/#asterisk jm|home (n=jm|home@zen.jamiem.com) |
| 13:39.29 | *** join/#asterisk russellb (n=russell@asterisk/developer-and-stable-maintainer/drumkilla) |
| 13:39.30 | *** mode/#asterisk [+o russellb] by ChanServ |
| 13:39.36 | dream_th | where can i read about that |
| 13:40.10 | [TK]D-Fender | dream_th: ... |
| 13:40.11 | [TK]D-Fender | ~book |
| 13:40.12 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com, or see ~buybook |
| 13:40.14 | [TK]D-Fender | ^^^^ |
| 13:40.28 | [TK]D-Fender | dream_th: This is all part of Dialplan basics... |
| 13:40.39 | dream_th | okay thank you |
| 13:41.50 | windback | SteveTotaro, what dependencies?? |
| 13:42.07 | windback | SteveTotaro, I've installed everything |
| 13:42.16 | SteveTotaro | dunno, never installed it |
| 13:42.25 | SteveTotaro | why not just use mysql? |
| 13:43.07 | Al_WinKiller | almost work )))))) got error shit on server 2 :D |
| 13:45.00 | incog_afk | what'd you change? |
| 13:45.15 | tzanger | morning everyone |
| 13:46.43 | incog_afk | morning... |
| 13:49.27 | Al_WinKiller | in sip.conf |
| 13:49.37 | Al_WinKiller | wait, it is not working yet |
| 13:49.47 | Al_WinKiller | I will fix it and will tell you |
| 13:50.11 | incog_afk | ah, but your call is getting routed from one machine to the other, now - that's something... |
| 13:50.34 | incog_afk | It's actually related to the problem I'm having, which is why I'm so interested... |
| 13:51.30 | *** join/#asterisk ghenry (n=ghenry@suretecsystems.plus.com) |
| 13:52.27 | *** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 13:52.38 | Al_WinKiller | IT WORKS !!!!!!!!!!!!!!!!!!!!! |
| 13:52.41 | Al_WinKiller | AAAAA !!!!!!!!!!!! |
| 13:52.46 | Al_WinKiller | I am so HAPPY !!!!!!!!!!!!! |
| 13:52.50 | Al_WinKiller | like a kid :) |
| 13:52.54 | Al_WinKiller | !!!!!!!!!!!!! |
| 13:53.12 | Al_WinKiller | GOD BLESS INTERNET FUCK MICROSOFT !!!! KILL BILL AND SAVE PENGUIN !!!!!!!!!!!!! |
| 13:53.17 | Al_WinKiller | sorry for flooding :) |
| 13:53.42 | *** join/#asterisk shido6 (n=shido6@204.126.120.132) |
| 13:54.08 | lirakis | morning |
| 13:54.23 | incog_afk | How'd you fix it? |
| 13:55.02 | Al_WinKiller | ok |
| 13:55.05 | Al_WinKiller | look |
| 13:55.14 | Al_WinKiller | Asterisk 1 is registred in Asterisk 2 |
| 13:55.20 | Al_WinKiller | in sip.conf |
| 13:56.31 | Al_WinKiller | and in sip.conf I got ( on Asterisk 1 ) I got some info about Asterisk 2 ( username, host, peer and more ) |
| 13:56.51 | Al_WinKiller | on Asterisk 2 I put in extensions.conf 1299501 under default |
| 13:56.55 | Al_WinKiller | and it works |
| 13:57.23 | *** join/#asterisk lunaphyte_ (n=lunaphyt@70.90.148.3) |
| 13:57.59 | Al_WinKiller | ok, I have to do some more job, cuz Asterisk 2 gonna be mail server ( for operators ) |
| 13:58.10 | Al_WinKiller | and do some more things with billing on asterisk ! |
| 13:58.17 | Al_WinKiller | mean asterisk 1 |
| 13:58.45 | incog_afk | ha! So the problem was that the extension wasn't defined in the context that it was supposed to be, right? Just like I said? :-P |
| 13:59.00 | *** join/#asterisk anthm (n=anthm@CPE-72-131-113-50.wi.res.rr.com) |
| 13:59.00 | *** mode/#asterisk [+o anthm] by ChanServ |
| 13:59.23 | Al_WinKiller | yep ) I am new to Asterisk ( have worked with h323 ) |
| 13:59.33 | Al_WinKiller | you were right :-P |
| 13:59.42 | *** part/#asterisk flujan (n=flujan@200-160-115-020.static.spo.ctbc.com.br) |
| 13:59.45 | Al_WinKiller | by the way,, do you mind if I ask you whre you from ? |
| 14:01.11 | *** join/#asterisk anthm (n=anthm@CPE-72-131-113-50.wi.res.rr.com) |
| 14:01.11 | *** mode/#asterisk [+o anthm] by ChanServ |
| 14:01.11 | nebojsajsimic | where to download phpagi for asterisk??? |
| 14:01.44 | incog_afk | I'm from the United States... :) You? |
| 14:03.04 | Al_WinKiller | from Armenia ) but I lived 15 years ( and study ) abroad :D U.S. is cool ) silicon valley :D |
| 14:03.43 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 14:03.47 | anonymouz666 | anyone in here has experience with Siemens HG equipament? |
| 14:03.53 | *** join/#asterisk ^shark_ (n=^shark_@217.194.147.193) |
| 14:03.54 | Al_WinKiller | listen,, by the way,, when I cal from one node ( sip phone or soft phone, doesn't matter ) and I disconet from the device I have called ,, it is still connected :( |
| 14:04.29 | Al_WinKiller | has to do something with signaling or dtmf ? |
| 14:05.03 | *** join/#asterisk flujan (n=flujan@200-160-115-020.static.spo.ctbc.com.br) |
| 14:05.58 | incog_afk | I used to work in the Silicon Valley, as a network programmer, until that work all went to India. And that's a new problem... |
| 14:05.59 | *** join/#asterisk dbtid (i=m1bmcfmr@cpe-65-24-173-154.columbus.res.rr.com) |
| 14:06.11 | dbtid | what's the goal behind asterisk 1.6? is it a radical change from 1.4? |
| 14:06.25 | incog_afk | How are you determining that the remote device is still connected? |
| 14:07.22 | incog_afk | not sure, dbtid - I just know that the new features are poorly documented, so far, and therefore hard to implement and take advantage of. How badly do you want them? |
| 14:07.31 | dbtid | i don't |
| 14:07.35 | dbtid | necessarily |
| 14:07.48 | dbtid | i saw tht it was available; i just got 1.4.18 and am going to try to build it for a powerpc |
| 14:07.58 | dbtid | i expect quite a learning curve :) |
| 14:08.19 | *** join/#asterisk hmmhesays (n=hmmhesay@24-119-176-74.cpe.cableone.net) |
| 14:09.09 | incog_afk | I just climbed it (for the second time; I did it once before, about two years ago), and I had to downgrade from 1.6 back to 1.4, because of the lack of documentation to use, for the climb... |
| 14:09.15 | Al_WinKiller | I can see that cuz I got cisco phone ( 1299401 ) connected to Asterisk 1 and Alvarion GW (1299501 ) connected to Asterisk 2 and also X-Lite is connected to Asterisk 2 ( behind NAT ) |
| 14:10.01 | Al_WinKiller | incog_aft by the way what languages you know ? ( mean programming languages ) ? |
| 14:10.58 | *** join/#asterisk SteveTotaro (n=Elizabet@c-69-243-124-5.hsd1.md.comcast.net) |
| 14:11.20 | incog_afk | About 20 of them. The most useful of which are C/C++, Perl, PHP, SQL... |
| 14:12.10 | *** join/#asterisk timeshell (n=Khoja@gw.lusi.on.ca) |
| 14:12.40 | x86 | <PROTECTED> |
| 14:12.42 | x86 | >:) |
| 14:12.56 | Al_WinKiller | maaaaan,,, you are cool )))) I know shell scripting ( bash ),,, some basic of SQL ,, want to lear Perl ( very usefull language ) :D |
| 14:13.15 | incog_afk | So... when you are checking to see if a phone is still connected, what exactly are you checking? |
| 14:14.45 | *** join/#asterisk scruz (n=Dell_ope@41.220.73.170) |
| 14:14.51 | Al_WinKiller | I can see that , for sample I call from 1299401 to 1299501 and pick it up,, when I put down the phone ( 1299501) 1299401 is still in progress |
| 14:14.51 | incog_afk | Don't get too excited about learning all about programming. My job, today, is to have a doctor swab my sinuses, to see if I have a specific virus that he wants to study. I've been unemployed for about 5 months. Apparently, all this knowledge only qualifies me for human medical experimentation. |
| 14:14.54 | scruz | hello |
| 14:15.29 | incog_afk | When you pick up the phone again, do you get a dial tone? Is there a call on hold? |
| 14:16.22 | hmmhesays | why does smsq have to be dependent on libpopt |
| 14:16.24 | *** join/#asterisk ManxPower (n=manxpowe@34.sub-70-220-73.myvzw.com) |
| 14:16.26 | Al_WinKiller | :D :D :D I don't know,, I like it ) programming is cool,,,,, and so exciting )))) even when I create some simple script,,, I feel good,,, it is nice feelling,, I don't know how to describe,, hope you undestand me |
| 14:16.27 | hmmhesays | what a freaking pita |
| 14:16.29 | Al_WinKiller | no,, no dial tone |
| 14:16.37 | Al_WinKiller | just is like it connected |
| 14:16.39 | Al_WinKiller | no voice |
| 14:16.54 | Al_WinKiller | nothing, and I can see ( on indicator ) that is still connected |
| 14:18.00 | incog_afk | Sounds like you might be misinterpreting the log. The call has terminated. The SIP device is still logged on, but it should stay logged on, so that it can recieve other calls, and make other calls... |
| 14:18.15 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 14:18.27 | Al_WinKiller | ok, let me see that on debug |
| 14:18.32 | incog_afk | But if you have no dial tone... hmmm... |
| 14:19.21 | Al_WinKiller | no |
| 14:19.23 | Al_WinKiller | look |
| 14:19.26 | incog_afk | In your extensions.conf, are you ending the handling of the extension with a Hangup or something else? |
| 14:19.41 | Al_WinKiller | when I pickup again ( without calling ) it is still connected :( |
| 14:19.49 | timeshell | Greetings |
| 14:19.59 | incog_afk | Salutations |
| 14:20.07 | incog_afk | Felicitations |
| 14:20.22 | incog_afk | (and other permutations of "Hello"...) |
| 14:20.22 | Al_WinKiller | I got this in extensions.conf ( Asterisk 1 ) |
| 14:20.23 | *** join/#asterisk ddunavant (n=David@pool-71-178-103-218.washdc.east.verizon.net) |
| 14:20.24 | Al_WinKiller | exten => 1299501,1,Dial(SIP/1299501@83.139.0.51,30,r) |
| 14:20.25 | ^shark_ | hi guys |
| 14:20.26 | Al_WinKiller | ? |
| 14:20.30 | Al_WinKiller | hi dude |
| 14:20.38 | incog_afk | That's sip.conf... |
| 14:20.46 | ^shark_ | i have a cisco 7912 ip phone that has issues with firmware |
| 14:20.48 | incog_afk | no, wait... it's not... |
| 14:21.07 | ^shark_ | i have got the firmware but i dont know how to proceed properly |
| 14:21.08 | incog_afk | It's just not the end of the entension handling... |
| 14:21.12 | timeshell | I have a Polycom 301 that won't register the second line on port 5061 |
| 14:21.12 | Al_WinKiller | perhaps ,30,r ? |
| 14:21.23 | Al_WinKiller | has to do something with that ? |
| 14:22.03 | *** join/#asterisk puzzled (n=patrick@puzzled.xs4all.nl) |
| 14:22.06 | ManxPower | Al_WinKiller: don't use "r" |
| 14:22.10 | incog_afk | when the call ends, Al_WinKiller, what are you doing, next? Usually, this is where you want to do a Hangup... |
| 14:22.26 | incog_afk | (The 'r' isn't hurting anything...) |
| 14:22.28 | Al_WinKiller | yes |
| 14:22.35 | Al_WinKiller | agree with you |
| 14:22.44 | Al_WinKiller | so ? what is a solution ? what you think ? |
| 14:23.00 | ManxPower | Don'y worry. "r" will come back to bite you eventually. |
| 14:23.04 | *** join/#asterisk b1ch0 (i=b1ch0@static-200-105-150-60.acelerate.net) |
| 14:23.15 | Al_WinKiller | :D |
| 14:23.26 | incog_afk | It just forces a ringback tone, ManxPower - it's not relevant to his problem... |
| 14:24.04 | Al_WinKiller | but when I drop phone ( from I have called ) it disconnect OK |
| 14:24.12 | incog_afk | add exten => 1299501,n,Hangup() after the Dial line, Al_WinKiller... |
| 14:24.21 | ManxPower | incog_afk: Correct. It is not relevant to THIS problem. "r" does not just provide ringback. Reingback is enabled by default. "r" gives you ringback when you should be getting some other tone like a busy tone, for example. |
| 14:25.14 | incog_afk | Yes, ManxPower. For now, however, it's fine. He's just learning this stuff. I'm trying not to confuse him with too many problems at once... |
| 14:26.17 | Al_WinKiller | like this |
| 14:26.20 | Al_WinKiller | exten => 1299501,1,Dial(SIP/1299501@83.139.0.51,30,n,HUNGUP() ? |
| 14:26.24 | ManxPower | incog_afk: I don't think more confusion is really going to make a difference. |
| 14:26.25 | incog_afk | Right now, he hasn't decided to do Voicemail, or busy signalling, or hunt groups, or anything. He's not even hanging up the phone at the end of a call, for goodness' sake... :) |
| 14:26.43 | ManxPower | Al_WinKiller: Please step away from the keyboard and read The Book |
| 14:26.46 | ManxPower | ~book |
| 14:26.46 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com, or see ~buybook |
| 14:26.54 | Al_WinKiller | wich one ? |
| 14:27.04 | ManxPower | that one |
| 14:27.32 | incog_afk | no, Al_WinKiller... :) There will be two lines od extension handling, for exten => 1299501 - the Dial line and the Hangup line... |
| 14:27.33 | Al_WinKiller | ok, I will :D thnx |
| 14:27.53 | incog_afk | It is definitely worth examining the book, yes. |
| 14:28.14 | Al_WinKiller | I will ) |
| 14:28.16 | incog_afk | Thankfully, it's a pdf, so you can save it locally, and read it at your leisure... |
| 14:28.33 | Al_WinKiller | can you give me example of hungup line ? |
| 14:28.45 | Al_WinKiller | cuz I have to finish this project |
| 14:29.28 | ManxPower | Al_WinKiller: You won't be finishing this project for weeks. Asterisk is a PBX Toolkit, not a bicycle. |
| 14:29.52 | Al_WinKiller | I agree ) but,,,it has to work ) |
| 14:30.08 | incog_afk | exten => 1299501,n,Hangup() |
| 14:30.14 | ManxPower | It will work eventually. Just don't expect to even know enough to make it work for a while. |
| 14:30.35 | Al_WinKiller | what you mean ? |
| 14:30.45 | ManxPower | And if you really mean't "It has to work right away." Well..maybe you should start looking for another job then. |
| 14:30.57 | ManxPower | Because it is not our problem. |
| 14:31.05 | Al_WinKiller | thnx |
| 14:31.28 | Al_WinKiller | thnx ManxPower you are very "kind" |
| 14:31.28 | incog_afk | I spent two weeks of fulltime effort just setting up one Asterisk box, two softphones, a terminal adapter, and two VoIP service provider accounts, Al_WinKiller - you will most likely not finish this project before you have read that book... |
| 14:31.30 | ManxPower | The Book is a good start for you. It should contain all the info you need. |
| 14:31.36 | ManxPower | Al_WinKiller: I'm a realist. |
| 14:31.39 | Al_WinKiller | I will read it |
| 14:32.01 | timeshell | I keep getting a "Got SIP response 489 "Bad Event" Back from ... " between 2 of my servers. Any idea why? They seem to be communicating between themselves fine other than that. They are both 1.4.18. Calls in both directions work. |
| 14:32.10 | timeshell | This message is just annoying me. |
| 14:32.34 | incog_afk | set sip debug, and find out what the 'Bad Event' is... |
| 14:33.13 | file | timeshell: message waiting indication. |
| 14:33.40 | timeshell | file: No voicemail on the extensions |
| 14:34.00 | file | doesn't me an it's not configured... |
| 14:34.04 | file | erm mean |
| 14:34.07 | incog_afk | Is a 'message waiting' considered a 'Bad Event'? That's just stupid... |
| 14:34.14 | timeshell | Agreed |
| 14:34.20 | timeshell | Doesn't make must sense |
| 14:34.25 | *** join/#asterisk kkn088 (n=kikoun@77.205.52.191) |
| 14:34.29 | timeshell | must=much* |
| 14:34.30 | file | Asterisk doesn't accept receiving MWI over SIP from a remote server, thus it sends back a Bad Event |
| 14:34.49 | incog_afk | aha... THAT makes some sense... |
| 14:35.02 | incog_afk | Now I have to go screw with that... |
| 14:35.02 | timeshell | The remote server is registered to the one that's giving the error |
| 14:35.08 | *** join/#asterisk b1shop (n=b1shop@c-24-7-202-70.hsd1.il.comcast.net) |
| 14:35.35 | timeshell | It shouldn't be providing any message indicators to it |
| 14:35.42 | incog_afk | Does Asterisk accept MWI over IAX, or is that a 'Bad Event', too? |
| 14:36.01 | file | timeshell: well, it is |
| 14:36.16 | ManxPower | file: Maybe it should say "Asterisk does not support MWI over SIP between servers." instead? |
| 14:36.56 | file | the message is being displayed on the server that sent out the MWI. |
| 14:37.13 | ManxPower | Ah. |
| 14:37.51 | timeshell | file: Why would the server that is registering to another server be sending MWI back to the server it is registering against? |
| 14:37.58 | timeshell | That doesn't make any sense at all.... |
| 14:38.12 | b1ch0 | hi guys, to add an fxs module to my card do i just add it to zaptel.conf (fxoks=8 it is last port) |
| 14:38.25 | Al_WinKiller | thnx guys,,, see you around , see ya |
| 14:38.32 | file | timeshell: I didn't configure your stuff. |
| 14:38.44 | timeshell | file: Exactly my point. |
| 14:38.48 | timeshell | file: You're wrong |
| 14:38.49 | timeshell | :p |
| 14:39.06 | ManxPower | Our file is never wrong. |
| 14:39.06 | incog_afk | Are you sure, file? I could swear I saw you doing it, on the sly... |
| 14:39.22 | *** join/#asterisk fskrotzki (n=fskrotzk@host198.textwise.com) |
| 14:39.24 | timeshell | file: It must be something else |
| 14:39.43 | timeshell | file: Perhaps has to do with latency or natting? |
| 14:39.55 | file | timeshell: just do a sip debug, see it's a NOTIFY packet carrying voicemail read and unread information probably of 0... and go figure out why |
| 14:40.07 | timeshell | file: I'm doing that as we speak. |
| 14:40.33 | ManxPower | timeshell: Server A sends MWI (via mailbox= in sip.conf) to Server B. Server B responds to Server A with a "Bad Event" message. Server A displays the message on it's console. |
| 14:40.50 | *** join/#asterisk ming_zym (n=ming_zym@f0-0-tep-rtr1.corp.cnb.yahoo.com) |
| 14:40.51 | timeshell | file: "Really destroying SIP dialog 'whole bunch of hex&ipaddress'" |
| 14:41.02 | file | incog_afk: as for you... MWI information for chan_iax2 is carried over the packet used for registration, so while Asterisk will display it on the CLI it doesn't use it internally |
| 14:41.08 | file | timeshell: which is perfectly normal... |
| 14:41.27 | *** join/#asterisk JenniferAkemi (n=akemi@76-10-147-54.dsl.teksavvy.com) |
| 14:41.45 | incog_afk | Okay... but if he isn't doing voicmail, how is he generating an MWI for the other server to choke on, and send this message back to his console? |
| 14:42.36 | file | the only reason it would do that, is if it was configured. |
| 14:42.48 | timeshell | OK, it is MWI, from the registered server. |
| 14:42.58 | timeshell | I apologize... my bad |
| 14:43.28 | timeshell | Wouldn't it be better if it just ignored such a message? |
| 14:43.29 | incog_afk | ^^^ "<ManxPower> Our file is never wrong." |
| 14:44.07 | file | timeshell: let me rephrase your sentence to what you probably meant, "Wouldn't it be better if it didn't display the message on the CLI?" |
| 14:44.13 | incog_afk | The better question is what's wrong with your config, that is causing it to generate MWI to begin with...? |
| 14:44.31 | incog_afk | file beat me to it... |
| 14:44.39 | coppice | why not arrogantly apologise? |
| 14:44.54 | ManxPower | incog_afk: we know what is causing it. |
| 14:45.13 | incog_afk | Do tell! He's got me all excited and curious, now... |
| 14:45.18 | timeshell | coppice: When one replies in arrogance, one needs to apologize with humility. |
| 14:45.38 | timeshell | coppice: So I apologize for my arrogance as well. |
| 14:45.38 | *** join/#asterisk freezey (n=freezey@gw.mypublisher.com) |
| 14:46.00 | timeshell | One of my personality flaws. :( |
| 14:46.05 | *** join/#asterisk jameswf-home (n=james@dsl093-157-131.phx1.dsl.speakeasy.net) |
| 14:46.32 | ManxPower | incog_afk: mailbox= is what controls MWI. |
| 14:47.39 | jameswf-home | yet another reason not to use BSD anti american bastards http://www.kesq.com/Global/story.asp?S=7858527&nav=menu191_2 |
| 14:48.43 | *** join/#asterisk anthm (n=anthm@CPE-72-131-113-50.wi.res.rr.com) |
| 14:48.43 | *** mode/#asterisk [+o anthm] by ChanServ |
| 14:49.06 | russellb | incog_afk: you don't seem very afk to me |
| 14:49.11 | ManxPower | jameswf-home: Wow, you must be really bored to care about that. |
| 14:49.28 | jameswf-home | its 8am what do you want :) |
| 14:49.41 | incog_afk | I know... I registered the nick not realizing that my status was afk... :) Now I'm stuck with it, if I want to be here... :) |
| 14:50.34 | tzafrir_laptop | Is BSD actually related to Berkeley (besides some old copyrights)? |
| 14:50.44 | windback | Somebody can helpme to include the CDR_sqlite module |
| 14:50.53 | coppice | i thought every tax payer supported troops |
| 14:51.00 | incog_afk | A lot less of them will volunteer, without being 'supported' - and a lot more will die. It's worth remembering that it is possible to support your troops the way that you support your local fire department... |
| 14:51.05 | ManxPower | coppice: it's treason not to, you know. |
| 14:51.06 | tzafrir_laptop | windback, what version of Asterisk? |
| 14:51.22 | windback | tzafrir, 1.4.13 |
| 14:51.38 | jameswf-home | *dance |
| 14:51.40 | ManxPower | incog_afk: But I believe in what the local fire department does. 8-) |
| 14:51.41 | jameswf-home | lol |
| 14:51.42 | russellb | _please_ do not talk about any military / troops realted stuff |
| 14:51.51 | incog_afk | A lot of development work still comes out of Berkeley, and other California state-funded universities... |
| 14:52.13 | russellb | np ... |
| 14:52.20 | jameswf-home | alot of general research is still done in universities... cheap labor |
| 14:52.29 | drmessano | I believe in Asterisk, and the Easter Bunny |
| 14:52.29 | windback | tzafrir_laptop, I have installed the sqlite package in debian, but when I try to add it in the make menuselect from asterisk it is XXX |
| 14:52.38 | drmessano | I am behind the Easter Bunny 100% |
| 14:52.51 | jameswf-home | mmmmmmmmm rabbit stew |
| 14:52.52 | coppice | what are you doing back there? |
| 14:52.58 | tzafrir | windback, I suppose you need something of the sort of libsqlite-dev |
| 14:53.02 | tzafrir | or libsqlite3-dev |
| 14:53.24 | windback | tzafrir, yes I have installed it too |
| 14:53.25 | coppice | chillied rabbit |
| 14:53.34 | jameswf-home | I told my kids we ate the easter bunny for dinner last year |
| 14:53.40 | tzafrir | IIRC the sqlite_cdr there is sqlite2, which needs libsqlite-dev . did you run ./configure afterwards? |
| 14:53.42 | jameswf-home | silly rabbit |
| 14:53.55 | incog_afk | And what exactly are you doing, behind the Easter Bunny? ;-P |
| 14:54.09 | jameswf-home | dirty |
| 14:54.18 | x86 | lol |
| 14:54.20 | jameswf-home | no politics no beastiality |
| 14:54.33 | jameswf-home | unless both involve hillary |
| 14:54.39 | jameswf-home | at the same time |
| 14:55.05 | jameswf | puts on SFW face |
| 14:55.05 | tzafrir | polyticks? |
| 14:55.12 | x86 | harhar |
| 14:55.33 | jameswf | yes tzafrir ticks with 2 heads |
| 14:55.57 | jameswf | mean little bast*rds |
| 14:56.03 | windback | tzafrir, I have to run ./configure before make menuselect |
| 14:56.09 | windback | tzafrir, i did it |
| 14:56.32 | tzafrir | can you now build cdr_sqlite? |
| 14:56.56 | jameswf | oh i is ok to talk About ron paul as long as all refrences are stated in the form of a chuck noris fact |
| 14:57.04 | jameswf | *it |
| 14:57.18 | russellb | jameswf: only if you can relate it to asterisk |
| 14:57.36 | Qwell | Chuck Norris once round-housed Ron Paul through app_meetme |
| 14:57.40 | windback | tzafrir, noo, |
| 14:57.48 | Qwell | russellb: what do I win? :p |
| 14:57.52 | jameswf | oh but you can see what qwell just did |
| 14:57.59 | russellb | Qwell: nothing! |
| 14:58.03 | Qwell | lame1 |
| 14:58.03 | jameswf | doh |
| 14:58.05 | Qwell | ! too |
| 14:58.51 | ManxPower | Asterisk is to a PBX exactly as a model airplane kit is to a model airplane. |
| 14:58.51 | jameswf | ron paul runs asterisk on his ipod and can write AGI in his sleep |
| 14:59.04 | *** join/#asterisk adjohn (n=adjohn@p5182-ipad71marunouchi.tokyo.ocn.ne.jp) |
| 14:59.12 | windback | tzafrir, perhaps I have to do make uninstall on /usr/src/asterisk-1.4.xx ??? |
| 14:59.36 | incog_afk | what's VOCAL, then, ManxPower? |
| 14:59.52 | ManxPower | windback: I removed the source and unpacked it from the tar file to get 1.4 to see new libs. |
| 15:00.12 | tzafrir | windback, no. That is not necessary |
| 15:00.22 | ManxPower | incog_afk: VOCAL is more like a kit for a house with all the instructions in Klingon |
| 15:00.45 | windback | ManxPower, ok, I will probe |
| 15:01.26 | tzafrir | hmmm.... any reason you use 1.4.13 rather than latest? |
| 15:01.32 | tzafrir | windback, ==^ |
| 15:02.22 | jameswf | I would have suggested 1.4.13 before 1.4.18 but with new release process 1.4.18 should be gold |
| 15:03.18 | DrAk0 | im having this real bad problem , when i try to go out via IAX after few seconds it cuts |
| 15:03.27 | DrAk0 | the sound |
| 15:03.32 | DrAk0 | after 20 seconds or so |
| 15:03.56 | ManxPower | DrAk0: notransfer=yes in iax.conf |
| 15:05.09 | DrAk0 | ManxPower, whats that? |
| 15:05.32 | ManxPower | DrAk0: that is the option that might fix your problem. |
| 15:05.53 | DrAk0 | let me try, reload is enough or do i have to restart |
| 15:05.58 | *** join/#asterisk Dovid (n=Dovid@bzq-79-178-84-138.red.bezeqint.net) |
| 15:07.55 | *** join/#asterisk docelmo (n=vircuser@h74.196.180.12.cable.rstb.cablerocket.net) |
| 15:08.20 | *** join/#asterisk methods (n=daquino@c-68-36-237-152.hsd1.nj.comcast.net) |
| 15:08.54 | methods | does asterisk add the local area code to? or does the phone company add the lines area code ? NXXXXXX |
| 15:11.42 | x86 | methods: no and no |
| 15:12.08 | *** join/#asterisk tnt_ (n=tnt_@8.253-244-81.adsl-static.isp.belgacom.be) |
| 15:12.38 | [TK]D-Fender | methods: * does what you tell it to in your dialplan. |
| 15:13.06 | yangvnc | hey [TK]D-Fender |
| 15:13.12 | _ShrikE | ha.. google "Find Chuck Norris" but click I'm feeling lucky instead of Google search. |
| 15:13.27 | tnt_ | Hi everyone. While looking in the doc of polycom, I see they support directed call pickup. For a group pickup you press *8 on the phone but what to press for a directed pickup ? |
| 15:13.29 | eric2 | methods, you're going to have to add the area code in your dial plan |
| 15:14.18 | eric2 | methods: extensions.conf is where you'll make asterisk sing, dance and scream! :) |
| 15:14.35 | yangvnc | I am getting this error with Pickup string on Grandstream phones - http://openpaste.org/en/5083/ |
| 15:14.39 | *** join/#asterisk bkw_ (n=brian@adsl-70-142-54-162.dsl.tul2ok.sbcglobal.net) |
| 15:16.52 | *** part/#asterisk scruz (n=Dell_ope@41.220.73.170) |
| 15:17.41 | hmmhesays | anyone know of any non ip, pstn style phones that are sms capable? |
| 15:18.21 | anthm | probably ones in uk? |
| 15:18.29 | yangvnc | hmmhesays: I never heard of those |
| 15:18.45 | incog_afk | The mechanics of doing that make it impossible on the current PSTN, hmmhesays... |
| 15:19.16 | hmmhesays | well I can currently send sms messages to the pstn line connected to this asterisk box, there is an sms gateway provided |
| 15:20.02 | *** part/#asterisk methods (n=daquino@c-68-36-237-152.hsd1.nj.comcast.net) |
| 15:20.30 | [TK]D-Fender | yangvnc: "core show application pickup" |
| 15:20.35 | incog_afk | There is no channel on the PSTN to carry SMS messages. Where are they going? |
| 15:22.46 | *** join/#asterisk simbol76 (n=simbol@ip-212-18.sn1.eutelia.it) |
| 15:22.50 | *** join/#asterisk jtknapp (n=skip@65-126-63-1.dia.static.qwest.net) |
| 15:23.15 | ManxPower | incog_afk: you must be american |
| 15:23.32 | ManxPower | SMS over the PSTN is FSK |
| 15:23.55 | tzafrir | In Israel there are some "SMS-capable" analog phones |
| 15:23.57 | bkw_ | ManxPower: Well its a bit of a tricky one.. SMS for mobile isn't the same as SMS on fixed line. |
| 15:24.06 | simbol76 | hello |
| 15:24.07 | tzafrir | and look ad app_sms |
| 15:24.21 | simbol76 | hi chat |
| 15:24.44 | simbol76 | I have the problem with asterisk1.4 |
| 15:24.44 | tzafrir | bkw_, ends up the same |
| 15:24.51 | incog_afk | I'm American. I keep thinking that we're some kind of technology leaders, and not the third-world country that we apparently really are... |
| 15:25.08 | bkw_ | tzafrir: it doesn't.. you can't sms a mobile device using app_sms |
| 15:25.12 | simbol76 | features.conf dont't work with chan_agent |
| 15:25.22 | tzafrir | bkw_, sadly I already have |
| 15:25.25 | bkw_ | tzafrir: sms for mobile phones are carried via ss7 |
| 15:25.33 | drmessano | Cool, I got accepted into the computer science program at hamburger university |
| 15:25.40 | simbol76 | it's possibile this???? |
| 15:25.47 | simbol76 | is very problem for me |
| 15:25.47 | incog_afk | And you have SS7 all the way out to the handset? |
| 15:25.57 | tzafrir | yes. And my telco probably knows how to connect to ss7 |
| 15:25.58 | coppice | bkw_ you can SMS a mobile device with app_sms in many places |
| 15:26.07 | hmmhesays | tzafrir, care to point me to one? |
| 15:26.16 | bkw_ | coppice: but they translate it .. its not direct like it is on the analog side is it? |
| 15:26.24 | coppice | although app_sms only implements one of the two ETSI land line SMS protocols |
| 15:26.28 | tzafrir | hmmhesays, app_sms or actually: smsq |
| 15:26.56 | coppice | bkw_: there's no translation to do. land line SMS sends more or less the same packets as GSM |
| 15:27.06 | bkw_ | coppice: interesting |
| 15:27.06 | incog_afk | okay... now I have to mess with SMS messaging... THAT would be COOL! |
| 15:27.18 | bkw_ | coppice: too bad that won't work here in the states |
| 15:27.24 | hmmhesays | tzafrir, oh i meant an sms capable phone |
| 15:28.09 | tzafrir | asterisk (with an FXO zap adapter) is one such "phone" :-) |
| 15:28.17 | coppice | I have a Chinese SMS phone sitting right beside me. I can send an SMS to my cellphone with it |
| 15:28.33 | hmmhesays | tzafrir, I'm compiling smsq for this blackfin, but it depends on libpopt which requires libgettext |
| 15:28.47 | hmmhesays | which equals a pain in my behind |
| 15:28.47 | bkw_ | coppice: Guess I learn something every day... :P |
| 15:29.08 | bkw_ | hmmhesays: you're in the united states.. I don't think that sms app can do you much good. |
| 15:29.15 | tzafrir | wow. the blackfin has no rpm natively (popt) :-) |
| 15:29.32 | hmmhesays | bkw_, this blackfin however is not in the united states |
| 15:29.40 | bkw_ | ah |
| 15:29.42 | bkw_ | I get ya |
| 15:30.45 | simbol76 | I set up a customized features features.conf But I read on the Internet that chan_agent of asterisk does not support this feature. |
| 15:30.45 | simbol76 | I set up a customized features features.conf But I read on the Internet that |
| 15:31.09 | coppice | hmmhesays: astfin has a package for spandsp on the blackfin now |
| 15:31.17 | simbol76 | If an agent typed features asterisk does not seem to notice anything. |
| 15:31.35 | hmmhesays | coppice, oh really? does rxfax work on it? |
| 15:31.52 | coppice | no. they are using it only for CLID right now |
| 15:32.20 | coppice | but they do have it building and installing, which I think was causing you trouble |
| 15:32.30 | simbol76 | Someone tell me knows how to create a type features for a features.conf Agent? |
| 15:32.53 | hmmhesays | it was for a bit, but I solved the problem I was having |
| 15:32.54 | *** join/#asterisk methods (n=daquino@c-68-36-237-152.hsd1.nj.comcast.net) |
| 15:33.08 | methods | is the first digit of an area code never 0 or 1 ? |
| 15:33.28 | simbol76 | 0 |
| 15:33.37 | hmmhesays | I have built installed and running, but like those logs I showed you, the soft fpu just can't keep up |
| 15:34.00 | incog_afk | I'm a relative newcomer, here, but I'm pretty sure that your question doesn't make sense, simbol76... maybe you should explain it to someone with better English skills, and have them ask the question in the room? |
| 15:34.20 | *** join/#asterisk stimpie (n=stimpie@84-104-5-115.cable.quicknet.nl) |
| 15:34.38 | coppice | hmmhesays: I wonder what digium run on those blackfin based appliances. quite a lot in * does heavy floating point |
| 15:34.40 | simbol76 | sorry but my english is very bad |
| 15:34.47 | [TK]D-Fender | simbol76: Show us what you're doing exactly by way of features.conf stuff, and your agent setups and all related dialplan. PASTEBIN is your friend. |
| 15:34.49 | [TK]D-Fender | ~pb |
| 15:34.49 | jbot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste |
| 15:34.52 | [TK]D-Fender | ^^^^^^^^^^^^ |
| 15:35.05 | *** join/#asterisk docelmo (n=vircuser@h74.196.180.12.cable.rstb.cablerocket.net) |
| 15:35.32 | simbol76 | I have to create a features for an agent in asterisk 1.4 |
| 15:35.45 | hmmhesays | coppice: I have no idea there. |
| 15:36.36 | coppice | astfin couldn't even get CLID to run. now they use the fixed point code from spandsp, they hardly notice the CLID running |
| 15:36.40 | *** join/#asterisk anthm (n=anthm@CPE-72-131-113-50.wi.res.rr.com) |
| 15:36.40 | *** mode/#asterisk [+o anthm] by ChanServ |
| 15:36.54 | simbol76 | chan_agent Does not support features made features.conf |
| 15:37.19 | simbol76 | on Asterisk 1.4 |
| 15:37.32 | simbol76 | it's possible?? |
| 15:37.54 | hmmhesays | are they doing something as timing senstive as faxing is though is the question |
| 15:38.02 | simbol76 | It is really strange |
| 15:38.15 | incog_afk | Then we'll need to know what you're talking about, from your features.conf, simbol76. And probably the extensions.conf, and anything else that seems relevant... |
| 15:38.46 | [TK]D-Fender | simbol76: ....PASTEBIN please.... |
| 15:39.08 | hmmhesays | coppice: you wouldn't have a link to one of these sms phones you have would you? |
| 15:39.09 | incog_afk | yeah... if you paste it into the room, you'll flood the room, and that will be bad... |
| 15:39.29 | coppice | <--- its just over there |
| 15:40.16 | *** join/#asterisk oej (n=olle@soll4-125.cust.blixtvik.net) |
| 15:40.16 | coppice | you want a Chinese SMS phone? |
| 15:41.03 | simbol76 | !#features.conf |
| 15:41.23 | simbol76 | Agent => 0,self/callee,Macro,test |
| 15:41.57 | simbol76 | #extensions.conf |
| 15:42.15 | simbol76 | exten => XXXXXXXX,1,Set(DYNAMIC_FEATURES=Agent) |
| 15:42.26 | simbol76 | but don't work :( |
| 15:43.30 | hmmhesays | Well an sms phone period, I'm coming up short on my google searches |
| 15:43.36 | methods | so what is + and 011. in the dial plan ? |
| 15:43.53 | coppice | this one was supplied by the telco. |
| 15:44.02 | methods | talking to me ? |
| 15:44.03 | [TK]D-Fender | simbol76: PASTEBIN everything I have requested please. |
| 15:44.05 | [TK]D-Fender | ~pb |
| 15:44.05 | jbot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste |
| 15:44.08 | [TK]D-Fender | ^^^^^^^^^^^^^^^ |
| 15:44.13 | simbol76 | If you do not use chan_agent works |
| 15:44.28 | simbol76 | ok |
| 15:44.47 | [TK]D-Fender | methods: Go read Chapter 5 again |
| 15:44.49 | [TK]D-Fender | ~book |
| 15:44.49 | 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 |
| 15:44.52 | [TK]D-Fender | ^^^^^^^ |
| 15:45.35 | DrAk0 | ManxPower, keep doing the same |
| 15:46.26 | simbol76 | ok Thanks!!! |
| 15:46.37 | *** join/#asterisk zelip (n=zelip@nat/hp/x-bd82d8c15a473f03) |
| 15:47.09 | *** join/#asterisk glen2 (n=glen@212.54.184.253) |
| 15:48.06 | glen2 | Hello. What's a good card for connecting Asterisk to a English ISDN30? |
| 15:48.21 | zelip | hi guys.. i have a ISP that has a recording when a wrong number is called.. Anyone knows of a way to detect this?.. I know is not easy. :S |
| 15:49.04 | ManxPower | zelip: Your telco should be sending back a VoIP or PRI message indicating the number is not in service. |
| 15:49.24 | ManxPower | Then you handle it in your dialplan, but since you didn't give us any additional information..... |
| 15:49.50 | ManxPower | glen2: any Digium or Sangoma T-1/E-1 card |
| 15:49.58 | *** join/#asterisk mort_gib (n=mjensen@16.Red-83-36-63.staticIP.rima-tde.net) |
| 15:50.07 | jameswf | or Rhino t1/e1/ji |
| 15:50.08 | jameswf | gah |
| 15:50.42 | zelip | ManxPower.. maybe I'm not getting that PRI message |
| 15:50.46 | zelip | the line are analog lines.. |
| 15:50.54 | *** join/#asterisk b1shop (n=b1shop@dsl081-149-253.chi1.dsl.speakeasy.net) |
| 15:50.56 | ManxPower | zelip: or you have progressinband=yes in zapata.conf |
| 15:50.58 | zelip | is there a tone before the recording i might be missing? |
| 15:51.08 | ManxPower | zelip: Ah. You can't do any of that on analog lines. |
| 15:51.29 | zelip | :( |
| 15:51.31 | zelip | jej |
| 15:51.34 | [TK]D-Fender | zelip: First you say ISP (which reads as ISp for us), then you mention PRI, then you mention analog. Please some coherence FAST. |
| 15:51.38 | [TK]D-Fender | ISTSP* |
| 15:51.47 | [TK]D-Fender | find* |
| 15:51.53 | [TK]D-Fender | BLARG... I can't type for shit today |
| 15:51.54 | zelip | jaja.. ok |
| 15:52.01 | zelip | i have some analog lines.. |
| 15:52.11 | zelip | so i get a recording when a wrong number is called. |
| 15:52.14 | ManxPower | zelip: you can't detect disconnected numbers when using analog lines. |
| 15:52.27 | zelip | jmm ok.. that all then |
| 15:52.27 | *** join/#asterisk [T]ank (n=ckwall@206.71.78.158) |
| 15:52.32 | zelip | that's |
| 15:53.15 | [T]ank | i am looking for the link on digiums website where i can purchase a g729 license but am only finding the install instructions. Could anyone point me in the right direction? |
| 15:53.46 | russellb | store.digium.com |
| 15:54.06 | glen2 | Can anyone recomend an analog interface card for about four lines? |
| 15:54.12 | Alexandre_fr | I don't understand how to use private key and public key between asterisk and an softphone-hardphone |
| 15:54.47 | jameswf | [T} |
| 15:54.51 | *** join/#asterisk docelm0 (n=vircuser@h74.196.180.12.cable.rstb.cablerocket.net) |
| 15:54.54 | jameswf | [T]ank: http://www.digium.com/en/products/g729codec.php |
| 15:55.09 | jameswf | glen2: R4FXO |
| 15:55.20 | Alexandre_fr | I did that between to asterisk but between asterisk and softphone or hardphone i don't know how to do that |
| 15:55.42 | russellb | glen2: TDM410 with 4 FXO modules |
| 15:55.43 | ManxPower | glen2: I don't think any of us can recommend analog. However, Digium's TDM400P w/modules would work, as well as the similar Sangoma card. |
| 15:56.19 | russellb | ManxPower: s/400/410/ |
| 15:56.24 | russellb | just got replaced |
| 15:56.26 | jameswf | ManxPower: no Sangoma... cheeky bastards.. glen2 digium or rhino |
| 15:56.37 | anonymouz666 | Digium sells g723.1 codec? |
| 15:56.44 | russellb | anonymouz666: only on a ard |
| 15:56.47 | russellb | card |
| 15:56.48 | ManxPower | jameswf: Rhino is such a new card virtually nobody has them and nobody can help if he has problems. |
| 15:57.02 | [T]ank | thanks all. |
| 15:57.03 | ManxPower | anonymouz666: almost nobody actually needs G723.1 |
| 15:57.06 | jameswf | ManxPower: wtf are you talking about |
| 15:57.09 | glen2 | Thanks. |
| 15:57.24 | anonymouz666 | ManxPower: unless you are talking with some Siemens HG equipament |
| 15:57.30 | ManxPower | jameswf: the Rhino card is what? less than a year old? |
| 15:57.38 | *** part/#asterisk Alexandre_fr (n=alex@nat04-bzr.dys1.com) |
| 15:57.47 | ManxPower | anonymouz666: and that equipment does not support G729, ulaw, or alaw? |
| 15:57.56 | anonymouz666 | g713.1 and ulaw |
| 15:58.00 | anonymouz666 | ops |
| 15:58.03 | anonymouz666 | g723.1 |
| 15:58.08 | jameswf | its a few years old has thousands in the field full free support should i go on |
| 15:58.10 | ManxPower | anonymouz666: It sucks to be you. |
| 15:58.33 | ManxPower | Does anyone here use the Rhino zaptel compatible cards? |
| 15:58.37 | *** join/#asterisk putnopvut (n=putnopvu@216.207.245.1) |
| 15:59.01 | anonymouz666 | putnopvut the app_queue master! |
| 15:59.14 | putnopvut | How's it going? |
| 15:59.22 | ManxPower | Maybe they are all trixbox users. |
| 16:00.48 | *** join/#asterisk CunningPike (n=arodgers@vpn.dnv.org) |
| 16:02.15 | jameswf | heck arizona a desert team ( i know ice hockey in the desert wtf) can kick the crap out of canadian hockey |
| 16:03.23 | JenniferAkemi | what matters is how many of the players on the AZ team are canadians :P |
| 16:04.08 | jameswf | Yeah who would want to live in 40 below when you could play in the desert |
| 16:04.40 | eric2 | it was only -25c here this morning |
| 16:04.51 | eric2 | yesterday it was -34c |
| 16:04.57 | eric2 | today is warm :) |
| 16:04.59 | jameswf | Its going to be 77F here today |
| 16:05.17 | JenniferAkemi | heh i just guessed, but a quick google reveals that about 50% of the coyotes are canadian by birth |
| 16:06.08 | JenniferAkemi | only 5 of them are born in the us |
| 16:06.15 | hmmhesays | I probably could have gone my entire life without knowing that |
| 16:06.29 | jameswf | If the U.S. is so crappy why do they poor in from canada (the worlds perfect country) and Mexico (the opisite of the other) |
| 16:06.31 | JenniferAkemi | :P |
| 16:06.40 | JenniferAkemi | who said the US was crappy? |
| 16:06.51 | FlatFoot | afternoon all |
| 16:06.52 | JenniferAkemi | they got the money |
| 16:06.59 | ManxPower | jameswf: The Canadians don't "pour into" the USA. |
| 16:07.16 | jameswf | *pour in to play hockey) |
| 16:07.21 | cpm | yeah, I would say it was rather the other way around |
| 16:07.27 | ManxPower | Mostly they just go shopping there, then return home |
| 16:07.31 | cpm | US folks pouring into canada |
| 16:07.32 | FlatFoot | anyone managed to get BLF working on a snom 360/6.5.15 ? |
| 16:07.51 | [TK]D-Fender | jameswf: Hockey is our greatest export next to beer, maple syrup, and Celine Dion (you can KEEP her however...) |
| 16:08.09 | cpm | no no! she's yours, you take her back! |
| 16:08.23 | coppice | i would think canada would be very keen to export celine dion |
| 16:08.25 | FlatFoot | [TK]D-Fender .... atke her back pleaseeeeeeeee |
| 16:08.27 | cpm | Terrance and Phillip otoh |
| 16:08.29 | FlatFoot | *take |
| 16:08.52 | lunaphyte_ | yeah, we don't want her. |
| 16:08.59 | jameswf | I think Celine is going back, now vegas will have barbra strisand ick |
| 16:09.16 | *** join/#asterisk simbol76 (n=simbol@ip-212-18.sn1.eutelia.it) |
| 16:09.26 | lunaphyte_ | we already put up with the barenaked ladies, we don't need any more. |
| 16:10.24 | JenniferAkemi | i would give up beer, hockey and anything else for amazon.com |
| 16:11.33 | jameswf | We would like to expory all the super liberals and 75% of california to canada |
| 16:11.33 | jameswf | *export |
| 16:12.11 | jameswf | the movie "the day after tomorrow" showed when global warming triggers the next ice age canada is screwed |
| 16:12.28 | JenniferAkemi | hopefully i'll be dead by then |
| 16:12.37 | tnt_ | Are the s-BUSY, s-NOANSWER ... thingies supposed to be working ... |
| 16:12.40 | tnt_ | ? |
| 16:12.52 | iCEBrkr | 'supposed to be working' |
| 16:12.58 | jameswf | according to al gore it is going to happen next week, and he invented the internes so he knows all |
| 16:13.16 | tnt_ | iCEBrkr: What I meant is are they deprecated for eg and replaced by smtg else ? |
| 16:13.17 | jameswf | *internet |
| 16:13.26 | [TK]D-Fender | jameswf: And most of the US didn't actually make it, they just fgured we didnt' stand a chance, but we understand cold here :p |
| 16:13.28 | jameswf | damn korean keyboard |
| 16:13.33 | iCEBrkr | tnt_: If you read the macro, you'll notice it's a combination of a status code. |
| 16:13.37 | iCEBrkr | s-{status} |
| 16:13.55 | iCEBrkr | I have bad gas today.. I blame global wamring |
| 16:13.56 | jameswf | I am as far south as you can be without going in to mexico I am good :) |
| 16:13.56 | simbol76 | for all |
| 16:13.58 | simbol76 | I have an asterisk 1.4. |
| 16:13.58 | simbol76 | Using both queue that Agent Login. |
| 16:14.14 | [TK]D-Fender | iCEBrkr: your bad gas CAUSES global warming... |
| 16:14.19 | simbol76 | I need to make sure that when an agent responds to a client, can run a macro typing 0 in conversation. |
| 16:14.26 | tnt_ | iCEBrkr: I actually used the macro macro-stdexten 'as-is' but somehow I never endup on voice mail. |
| 16:14.36 | simbol76 | With 1.2 asterisk this was possible with features.conf. |
| 16:14.36 | simbol76 | In asterisk 1.4 I read that the chan_agent not work with features.conf. |
| 16:14.36 | simbol76 | How do I do this? |
| 16:14.50 | [TK]D-Fender | simbol76: Where's my pastebin? |
| 16:14.55 | iCEBrkr | [TK]D-Fender: oh!! |
| 16:15.10 | iCEBrkr | tnt_: core set verbose 9 |
| 16:15.24 | iCEBrkr | tnt_: and watch the status that's returned from Dial() |
| 16:16.26 | *** join/#asterisk watchy (n=watchy@h200.176.255.206.cable.cmdn.cablelynx.com) |
| 16:16.27 | *** join/#asterisk mindCrime (n=chatzill@66.83.208.218.nw.nuvox.net) |
| 16:16.44 | watchy | anyone have a problem when forwarding a call to another sip phone dtmf doesnt work? |
| 16:17.21 | tnt_ | iCEBrkr: http://pastebin.com/m4067a82e and it just return BUSY status to the SIP caller instead of redirecting to voicemail. |
| 16:17.34 | iCEBrkr | w00t |
| 16:18.00 | nvrpunk | how do i set the password to mysqladmin |
| 16:18.00 | iCEBrkr | tnt_: Then the call isn't being handled correctly.. It should be returning NOANSWER |
| 16:18.05 | nvrpunk | it keeps denying me :( |
| 16:19.03 | simbol76 | What you post to pastebin? D-Fender |
| 16:19.11 | tnt_ | iCEBrkr: No ... to speed things up I set up DND mode on the called phone. But I still should end up on voicemail |
| 16:19.35 | iCEBrkr | Well, then set up a s-BUSY to land in VoiceMailMain() |
| 16:19.44 | *** join/#asterisk jpeeler (n=jpeeler@216.207.245.1) |
| 16:19.49 | iCEBrkr | exten => s-BUSY,1,Voicemail(${ARG1}|sb) |
| 16:20.49 | Dovid | i am half alseep so please forgive me. how do I get the stsus of a phone if it is set to DND and how can I set it from the dial plan ? |
| 16:21.00 | *** join/#asterisk scardinal (n=supreme@port816.ds1-suoe.adsl.cybercity.dk) |
| 16:21.00 | tnt_ | iCEBrkr: There is one ... that's what I don't understand ... |
| 16:21.33 | iCEBrkr | Everyone is busy/congested at this time (1:1/0/0) |
| 16:21.40 | iCEBrkr | That to me, doesn't look like it's right |
| 16:21.41 | tnt_ | iCEBrkr: http://pastebin.com/m7fa6abda That's the macro |
| 16:22.22 | iCEBrkr | When DND is off, the phone rings? |
| 16:22.37 | [TK]D-Fender | tnt_: You are MISSING the GOTO that was in the demo sample |
| 16:23.00 | iCEBrkr | haha |
| 16:23.00 | iCEBrkr | yeah |
| 16:23.13 | [TK]D-Fender | tnt_: exten => s,2,Goto(s-${DIALSTATUS},1) |
| 16:23.13 | iCEBrkr | exten => s,3,Goto(s-${DIALSTATUS},1) |
| 16:23.23 | [TK]D-Fender | iCEBrkr: almost :p |
| 16:23.26 | iCEBrkr | well it's 3rd in mine |
| 16:23.36 | [TK]D-Fender | iCEBrkr: It'd be second in HIS ;) |
| 16:23.37 | Dovid | TK: did u see my question ? |
| 16:23.38 | iCEBrkr | I got an announcment at the top |
| 16:23.46 | iCEBrkr | [TK]D-Fender: yeah, most likely. |
| 16:23.49 | iCEBrkr | oh yeah |
| 16:23.50 | iCEBrkr | duh |
| 16:24.00 | [TK]D-Fender | Dovid: You can't |
| 16:24.06 | Dovid | :( |
| 16:24.09 | Dovid | thanks TK |
| 16:24.33 | iCEBrkr | Friday can't get here any sooner.. |
| 16:24.38 | [TK]D-Fender | Dovid: Cheer up! http://icanhascheezburger.com/2007/01/16/go-cry-emo-kid/ |
| 16:24.39 | tnt_ | [TK]D-Fender: Oh ... thanks ... I must deleted it without realizing ... |
| 16:25.27 | file | ZaVoid: over here. |
| 16:25.36 | file | ZaVoid: provide an rtp debug |
| 16:25.39 | Dovid | TK: :) |
| 16:25.40 | ZaVoid | sure |
| 16:25.55 | ZaVoid | you want me to run a tethereal trace for the rtp debug? |
| 16:25.56 | *** join/#asterisk moa_ (n=moa_@216-129-228-165.vnet-inc.com) |
| 16:26.24 | file | no, you type rtp debug in the CLI and pastebin the output |
| 16:26.31 | moa_ | Hello guys. |
| 16:26.34 | ZaVoid | oh |
| 16:26.45 | moa_ | Has anyone every setup NFAS before? |
| 16:27.02 | Dovid | TK: My issue is that if i have a SIP account that is registerd as two agents for two queues if they are on a call from Queue1 i dont want a call from queue2 to come to them till they are off the phone with queue1 |
| 16:27.34 | [TK]D-Fender | Dovid: that has nothing to do with DND |
| 16:28.01 | iCEBrkr | 15 |
| 16:28.04 | iCEBrkr | Did I hit? |
| 16:28.11 | iCEBrkr | THAC0 please? |
| 16:28.51 | [TK]D-Fender | iCEBrkr: your swing went wild and hit a tree, please roll a 1D4 tos ee the direction that it falls as you cleaved right through... |
| 16:28.52 | Dovid | TK: I was trying to use DND as a quick fix ;) |
| 16:29.10 | [TK]D-Fender | Dovid: "show application chanisavail" |
| 16:29.22 | Dovid | TK: tried that. didnt work |
| 16:29.27 | iCEBrkr | [TK]D-Fender: hah |
| 16:29.32 | [TK]D-Fender | Dovid: you = fail. |
| 16:29.34 | Dovid | if they were on the phone still worked. let em look at it again |
| 16:29.37 | iCEBrkr | Dovid: epic fail |
| 16:29.46 | Dovid | iCEBrkr: ? |
| 16:29.54 | DrAk0 | [Feb 12 11:25:40] NOTICE[5983]: chan_iax2.c:2986 __auto_congest: Auto-congesting call due to slow response |
| 16:30.21 | Dovid | ahhhhhhhhhhhhhhh |
| 16:30.27 | watchy | tk: ever seen dtmf not work if the call is forwarded/transfered? |
| 16:30.51 | [TK]D-Fender | watchy: only if the device you're forwarding to wasn't set up right in the first place. |
| 16:31.03 | simbol76 | http://pastebin.com/m32eb62a2 |
| 16:31.05 | watchy | hrm |
| 16:31.09 | [TK]D-Fender | watchy: And you should be a little clearer as to exactly how you are testing... |
| 16:31.23 | watchy | well i have an ata on a gate, it used to work fine but now all of a sudden it doesnt work |
| 16:31.29 | iCEBrkr | I wonder if I can get a DID from my soon to be new employer. |
| 16:31.32 | iCEBrkr | >:) |
| 16:31.34 | watchy | the ata dials ext 301 and you can use dtmf on it |
| 16:31.45 | watchy | but if the call is forwarded to say 302 dtmf doesnt work |
| 16:31.53 | iCEBrkr | Because I'm kinda not liking VoicePulse anymore |
| 16:31.54 | *** join/#asterisk teknoprep (n=Chris@unaffiliated/teknoprep) |
| 16:32.07 | [TK]D-Fender | simbol76: No, features.conf will not work in Agentlogin(). Almost nobody uses it, but rather "AddQueueMember" etc... |
| 16:32.12 | ZaVoid | file: http://pastebin.com/d1b501816 |
| 16:32.19 | ZaVoid | i think thats what you want |
| 16:32.30 | ZaVoid | rfc2833 is set for all the endpoints too |
| 16:33.04 | DrAk0 | what other alternative for H.323 besides OOH323 ? |
| 16:33.36 | ZaVoid | DrAk0: send calls outbount SIP to a SBC and convert there or on a media gw :) |
| 16:33.49 | file | ZaVoid: and rfc2833compensate did not work? |
| 16:33.51 | simbol76 | thanks!! D-Fender |
| 16:34.08 | ZaVoid | compensate? |
| 16:34.11 | ZaVoid | you mean relaxdtmf? |
| 16:34.16 | file | no, rfc2833compensate |
| 16:34.27 | DrAk0 | ZaVoid, and from asterisk itself? |
| 16:34.30 | Dovid | TK: I am using sj option. still ringing |
| 16:34.53 | [TK]D-Fender | simbol76: You might be able to do "exten => 1234,1,Set(DYNAMIC_FEATURES=Agent)" , "exten => 1234,2,Dial(Local/s@login_normal/n,,Ww) as well... you could try that in the meantime. |
| 16:35.12 | ZaVoid | thats for 1.2 asterisk file right? |
| 16:35.14 | ZaVoid | i'm on 1.4.x |
| 16:35.19 | [TK]D-Fender | Dovid: And you are not SHOWING me anything. Do you think I trust your coding? :0 |
| 16:35.33 | file | it is an option on 1.4 for some funky RFC2833 implementations |
| 16:35.42 | [TK]D-Fender | Dovid: You would be sadly mistaken I assure you! |
| 16:35.57 | [TK]D-Fender | trust-- |
| 16:36.02 | Dovid | TK: I dont blame you based on the past ;) |
| 16:36.08 | ZaVoid | ok i'll try that |
| 16:36.16 | [TK]D-Fender | Dovid: Pastebin it all up already! |
| 16:36.50 | ZaVoid | how do i turn that on? rfc2833compensate=yes in general in sip.conf? |
| 16:36.57 | file | yes |
| 16:36.59 | ZaVoid | k |
| 16:37.38 | [TK]D-Fender | Dovid: And its not just you, I don't trust ANYBODY's setup when its not working. |
| 16:38.00 | [TK]D-Fender | Dovid: So we go through the motions of picking it all apart in as direct a path as possible |
| 16:38.09 | ZaVoid | dtmfmode=RFC2833 <-- case matter here? |
| 16:38.10 | [TK]D-Fender | Dovid: Do hurry up though, I'm checking outta here in a few minutes. |
| 16:38.20 | Dovid | http://pastebin.ca/901370 |
| 16:38.23 | *** join/#asterisk thomas_newbie__ (n=thomas@CPE0014bf493235-CM00140493ede8.cpe.net.cable.rogers.com) |
| 16:38.28 | Dovid | u can see it has a call and then i call and it rings |
| 16:38.42 | [TK]D-Fender | ZaVoid: wHaT ArE YoU TaLkInG AbOuT???!? |
| 16:38.54 | file | ZaVoid: shouldn't. |
| 16:39.05 | [TK]D-Fender | Dovid: dialplaN PLEASE. |
| 16:39.08 | *** join/#asterisk esaym (n=user@cpe-72-183-198-134.satx.res.rr.com) |
| 16:39.15 | ZaVoid | thought so file |
| 16:39.38 | Dovid | http://pastebin.ca/901373 |
| 16:40.29 | Dovid | TK: I know its not clean. it is a hack job ;) but it should work |
| 16:40.41 | *** join/#asterisk [T]ank (n=ckwall@206.71.78.158) |
| 16:41.01 | [TK]D-Fender | Dovid: Umm... looks like 100 wasn't on the phone for #2 |
| 16:41.20 | [TK]D-Fender | Dovid: == Spawn extension (ah-queue, 100, 1) exited non-zero on 'Local/100@ah-queue-97c9,2' |
| 16:41.24 | [TK]D-Fender | #13 |
| 16:41.38 | ZaVoid | file: that seemed to make it worse |
| 16:41.42 | [T]ank | ok, so i bought the g729 license and have installed it. when i dial an extension set to use g729 I get the following error on the cli> [Feb 12 09:39:26] WARNING[19014]: chan_sip.c:2963 sip_call: No audio format found to offer. Cancelling call to 1724 |
| 16:41.42 | [T]ank | <PROTECTED> |
| 16:41.44 | ZaVoid | now i don't see it in the debu at all |
| 16:41.45 | Dovid | but the call was. TK: look at this |
| 16:41.49 | Dovid | i will PB |
| 16:42.02 | [TK]D-Fender | Dovid: better PB please. |
| 16:42.18 | *** join/#asterisk kaii (n=kai@ciphron.de) |
| 16:42.20 | file | ZaVoid: then I suggest filing a bug after making sure the already open DTMF bugs do not apply to you |
| 16:42.31 | ZaVoid | ok |
| 16:42.43 | Dovid | http://pastebin.ca/901376 |
| 16:42.48 | ZaVoid | could it be that dtmf is coming from a phone in qatar? |
| 16:42.52 | ZaVoid | different tone settings? |
| 16:42.57 | Dovid | i use the 2222 extension |
| 16:43.02 | Dovid | twie from 101 to 100 |
| 16:43.08 | Dovid | twice* |
| 16:43.16 | Dovid | second call should not go through |
| 16:44.44 | [TK]D-Fender | Dovid: interesting. after chanisavail, noop the channel vars it sets and take a look. |
| 16:44.49 | [TK]D-Fender | But have to go for now, BBIAB |
| 16:45.08 | Dovid | Thanks TK |
| 16:45.42 | DrAk0 | any idea how to get out via h323 besides OOH323 ? |
| 16:46.30 | ManxPower | DrAk0: There are at least FOUR H323 drivers for Asterisk. |
| 16:46.36 | watchy | well tk i upgraded * and it fixed it |
| 16:46.42 | watchy | i guess 1.2.1.9 was kinda old |
| 16:47.05 | DrAk0 | ManxPower, well OOH323 crash me real bad |
| 16:47.25 | DrAk0 | ManxPower, so i would like another alternative with success experience |
| 16:47.50 | ManxPower | there is one in asterisk-addons, there is one included with Asterisk |
| 16:48.03 | ManxPower | BUT, H323 is one of the hardest things to make work with Asterisk. |
| 16:48.54 | *** join/#asterisk outtolunc (n=me@adsl-66-218-53-172.dslextreme.com) |
| 16:49.03 | ZaVoid | file: i took relaxdtmf out and then put it back in and now seems to be working right |
| 16:49.36 | Dovid | Zavoid: u set relaxd dtmf in ooh323 ? |
| 16:50.10 | DrAk0 | ManxPower, yeah , ooh323 is the one on the asterisk-addons and is very unstable |
| 16:50.21 | ZaVoid | no sip.confg Dovid different issue sorry |
| 16:50.28 | anonymouz666 | drak0: which is the most stable? |
| 16:50.43 | ZaVoid | were are you trying to go with h3232 dovid? |
| 16:51.41 | DrAk0 | anonymouz666, thats my question ... |
| 16:52.34 | *** join/#asterisk DaveCanoe (n=Dave@canoe404.dclg.ca) |
| 16:53.31 | Dovid | Zavoid: had some DTMF issues with it |
| 16:54.20 | ZaVoid | ahh |
| 16:56.26 | moa_ | I'm unable to get nfas to work. Anyone want to look at http://pastebin.ca/901392 and see what I've done wrong? |
| 16:58.10 | simbol76 | http://pastebin.com/d1148c271 |
| 16:58.15 | *** join/#asterisk af_ (n=getsmart@88-149-240-211.dynamic.ngi.it) |
| 17:01.32 | Dovid | Zavoid: seems that cisco does not like asterisk implemintation of ooh323 |
| 17:01.43 | *** join/#asterisk jwh (i=jwh@kelley.ber.rewt.org.uk) |
| 17:01.51 | DrAk0 | Dovid, problem with DTMF and OOH323? |
| 17:02.13 | simbol76 | Don't work property D-Fender |
| 17:06.35 | *** join/#asterisk drmessano-LT (n=nonya@207.230.140.240) |
| 17:06.54 | yangvnc | What is the difference between NT & TE ISDN cards? |
| 17:07.32 | ZaVoid | Dovid: which cisco? 5350? |
| 17:07.43 | bkw_ | yangvnc: one is network side and one is the customers side |
| 17:07.48 | ZaVoid | file: it seems to be working sporadically.. i'll contact my carriers i guess |
| 17:07.50 | bkw_ | NT is network.. TE is customer if i'm correct |
| 17:07.59 | yangvnc | ok |
| 17:08.13 | bkw_ | ZaVoid: what kind of problem are you seeing? |
| 17:08.23 | ZaVoid | spotty dtmf reconition from |
| 17:08.23 | Dovid | Zavoid: i dont know which one |
| 17:08.30 | Dovid | there are lots of issues |
| 17:08.31 | ZaVoid | is it a vendor Dovid ? |
| 17:08.33 | bkw_ | ZaVoid: those cisco boxen can be an issue |
| 17:08.53 | *** join/#asterisk Greek-B0y (n=email@41.221.58.4) |
| 17:09.24 | ZaVoid | nah mine isn't cisco related |
| 17:09.58 | yangvnc | bkw_: TE is client side ? http://en.wikipedia.org/wiki/Network_termination |
| 17:10.06 | bkw_ | yah figured |
| 17:10.12 | bkw_ | client customer |
| 17:10.14 | bkw_ | same thing in my mind |
| 17:10.22 | yangvnc | ok |
| 17:11.48 | *** part/#asterisk methods (n=daquino@c-68-36-237-152.hsd1.nj.comcast.net) |
| 17:12.57 | *** part/#asterisk SteveTotaro (n=Elizabet@c-69-243-124-5.hsd1.md.comcast.net) |
| 17:14.08 | tzafrir | yangvnc, NT: the side of the telco. This is what you need to e.g. connect an ISDN phone. |
| 17:14.17 | tzafrir | TE: what you need to connect to the telco |
| 17:14.53 | yangvnc | tzafrir: thanks |
| 17:14.53 | ManxPower | that would depend on what COUNTRY you are in. |
| 17:15.08 | ManxPower | In the USA/Canada just forget about ISDN BRI |
| 17:15.55 | Qwell | ManxPower: I think Strom got ISDN BRI working in LA |
| 17:16.28 | ManxPower | Strom is a telco guru. |
| 17:16.33 | Qwell | this is true |
| 17:16.43 | ManxPower | Chances I could get it working too, but it would be a lot of work. |
| 17:16.54 | tzafrir | ManxPower, that reads: "stuck with analog" |
| 17:18.11 | *** join/#asterisk techie (n=techie@adsl-76-214-27-74.dsl.lsan03.sbcglobal.net) |
| 17:19.55 | *** join/#asterisk egecko (n=sam@cpe-76-176-206-245.san.res.rr.com) |
| 17:20.23 | *** part/#asterisk FlatFoot (n=bigflatf@80.88.192.83) |
| 17:22.05 | Silent-X | anyone here messed with counterpath's Bria? |
| 17:22.15 | egecko | trying to understand something here .. a provider that gives me a DID number .. I need to obviously pay the per minute charges for them to "send" that number to an asterisk box. Now are there providers with their own DIDs and are also a hosting provider? Shouldn't this setup reduce incoming rates? |
| 17:22.49 | *** join/#asterisk fnordus (n=dnall@24.84.160.227) |
| 17:22.56 | yangvnc | tzafrir: Which part is the ISDN PCI card = TE ? |
| 17:23.13 | tzafrir | they are TE by default |
| 17:23.48 | tzafrir | I think you can pass a parameter to them to behave as NT. But then you'll also need some slightly different wiring |
| 17:24.08 | tzafrir | (single-span HFC-S -based ISDN cards) |
| 17:24.30 | yangvnc | ok |
| 17:25.42 | yangvnc | egecko: I suggest voip.ms , which has international DID's no charges on incoming (but there are many others) |
| 17:26.17 | jameswf | our bri cards have jumpers to set te nt |
| 17:26.38 | jameswf | some have dip switches |
| 17:27.36 | *** join/#asterisk fnordus (n=dnall@24.84.160.227) |
| 17:27.45 | tzafrir | yeah, all HFC-[48]S -based cards (and sadly: also our device) have those |
| 17:27.45 | egecko | yangvnc: thanks I'll check them out. I'd prefer to go with a company that has experience hosting asterisk boxes as opposed to a regular dedicated server host |
| 17:28.18 | tzafrir | The nice thing is that you don't need any special cables then - just a flat cat5 one |
| 17:29.24 | yangvnc | egecko: they let you set the incoming number on your asterisk either on usa or u.k pop |
| 17:29.43 | DrAk0 | having bad problems with |
| 17:29.46 | yangvnc | egecko: and they got very easy to navigate menus |
| 17:30.01 | DrAk0 | [Feb 12 12:25:12] NOTICE[2835]: chan_iax2.c:2986 __auto_congest: Auto-congesting call due to slow response |
| 17:30.39 | egecko | yangvnc: I guess that's the part I'm trying to figure out. Do I host my asterisk box at any dedicated host company or go with a provider that will offer me both the DIDs and the hosting |
| 17:32.47 | bkw_ | tzafrir: in the US we use ISND-BRI-U so the wire doesn't matter how its hooked up since its only two wires :P |
| 17:34.16 | *** join/#asterisk wwalker (n=wwalker@pdpc/supporter/sustaining/wwalker) |
| 17:34.45 | zelip | hi.. anyone know what CTE stands for? (is used for a type of end user phone). like proper and mobile |
| 17:35.05 | wwalker | any negatives if the dial plan answers() a call that was already answered by an AGI? (The AGI may or may not answer during it's run) |
| 17:35.21 | ZaVoid | what does this mean? [2008-02-12 17:34:37] DEBUG[14377] app_macro.c: Oooh, got something to jump out with ('9')! |
| 17:36.48 | *** join/#asterisk |dennis| (n=dennis@200.32.236.10) |
| 17:37.21 | ZaVoid | gah this ignoring the dtmf is driving me nuts |
| 17:37.24 | variable_office | is there a command that simply checks if a sip user exists? |
| 17:38.24 | ZaVoid | Asterisk 1.4.13 might be the problem |
| 17:40.52 | ZaVoid | is ther any dtmf fixes from 1.4.13 to 1.4.15 or .18? |
| 17:41.48 | anonymouz666 | ZaVoid: there is no way for you to change the DTMF mode? |
| 17:42.00 | ZaVoid | change to what? its already rfc2833 |
| 17:42.04 | anonymouz666 | SIP INFO |
| 17:43.08 | ZaVoid | ugh |
| 17:43.10 | ZaVoid | that never works |
| 17:44.06 | ZaVoid | i'll try it though |
| 17:44.15 | *** join/#asterisk bsdwarrior (n=mfahey@fahey.enter.net) |
| 17:44.58 | bsdwarrior | How can I forward an extension from one to the other without using the button on the phone. |
| 17:46.24 | jameswf | To the canadians .. anyone have farend disconect issues on bell canada pots |
| 17:50.14 | *** join/#asterisk admin0 (n=admin@116.90.228.34) |
| 17:53.23 | *** join/#asterisk angom (n=angom@200.56.104.87.dsl.dyn.telnor.net) |
| 17:54.21 | *** join/#asterisk admin0 (n=admin@116.90.228.34) |
| 18:00.04 | tnt_ | Is there anyway to display the time on the polycom phones in 24h format ? |
| 18:00.31 | defswork | jameswf: what fxo card are you using ? |
| 18:02.44 | [TK]D-Fender | tnt_: Yes, go look on the phone itself |
| 18:03.12 | [TK]D-Fender | tnt_: incredibly obvious |
| 18:04.07 | the_5th_wheel | zIf i want to upgrade BRISTuff, do i just install it over? |
| 18:04.25 | [TK]D-Fender | variable_office: "core show function SIPPEER" |
| 18:11.08 | jameswf | lol not trouble shooting looking for folks with issues |
| 18:11.26 | tzafrir | the_5th_wheel, basically: yes |
| 18:11.33 | jameswf | bell canada doesnt apparently employ engineers |
| 18:11.41 | drmessano-LT | jameswf: Is this a trixbox system? |
| 18:11.45 | tzafrir | just run ./install.sh |
| 18:11.57 | drmessano-LT | jameswf: Have you rebooted? |
| 18:11.58 | tzafrir | this reinstalls the software. Doesn't touch configuration |
| 18:14.06 | slima | asterisk: symbol lookup error: /usr/lib/asterisk/modules/chan_sip.so: undefined symbol: ast_pickup_ext - which module is missing? |
| 18:14.39 | jameswf | good to know bell canada support is as helpful as any of the US baby bells |
| 18:14.59 | [TK]D-Fender | jameswf: They do. "Entourage" previously spun off as a separate support entity, but were bought back and allowed Bell to get their old techs back ast a fraction of their previous pay. |
| 18:15.28 | *** join/#asterisk lmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 18:15.28 | *** mode/#asterisk [+o lmadsen] by ChanServ |
| 18:16.08 | drmessano-LT | The Bell System: Using the excuse "Hey, that's copper for ya!" since 1892 |
| 18:16.17 | ManxPower | Wow! That's a good deal for...someone |
| 18:17.46 | slima | ah. res_features.so |
| 18:18.43 | *** join/#asterisk puzzled (n=patrick@puzzled.xs4all.nl) |
| 18:19.01 | jameswf | I call bell canada and say I need specs for bell ca lines cpc timing etc... they say line specs depend on the dmark.. I ask so you dont have any base line specs or the cpc timing he said no if a customer has an issue we just go out and look at the line.... if you dont have a base line how do you know if its broke..... ummmm you have to have the customer make a ticket |
| 18:19.02 | jameswf | gah |
| 18:20.22 | jameswf | and my guess like any of the baby bells they go out and say all looks good must be the hardware |
| 18:20.59 | ManxPower | jameswf: Do you have a "princess phone", i.e. one with the lighted keypad, powered by the line? |
| 18:21.25 | jameswf | ManxPower: dude ITS NOT MY LINE I AM NOT AN END USER |
| 18:21.40 | jameswf | I AM NOT TROUBLESHOOTING |
| 18:21.42 | jameswf | bah |
| 18:21.45 | ManxPower | If so, call someone on it, tell them to hang up, time how long it takes for the light on the phone to go dark for a moment, that is when disconnect happens. |
| 18:21.55 | drmessano-LT | lol |
| 18:21.58 | ManxPower | jameswf: then why are you babbleing about CPC? |
| 18:22.06 | drmessano-LT | jameswf: Is the power on? |
| 18:22.24 | jameswf | because bell canada sucks and we cant program for non existant specs |
| 18:22.32 | *** join/#asterisk mishehu (i=1000@crosscreek.cartissolutions.net) |
| 18:23.03 | jameswf | and end user estimations are no substitute for the real thing |
| 18:23.04 | ManxPower | You should expect 500ms batter drop for USA and Canada ILECs and for most CLECs |
| 18:23.37 | ManxPower | 99% of PSTN/Telco POTS lines would have the 500ms drop. |
| 18:23.47 | jameswf | Rule of thumb you should never expect anything from a bell company |
| 18:24.33 | jameswf | standards are kinda a joke in north america |
| 18:25.28 | drmessano-LT | Yes |
| 18:25.40 | drmessano-LT | Depends on their mood |
| 18:26.11 | ManxPower | sorry, the standard says 800ms, not 500ms |
| 18:26.22 | *** join/#asterisk guillote_GNU (n=guillote@host157.201-253-55.telecom.net.ar) |
| 18:26.24 | jameswf | The US is a superpower with 3rcd maybe 4th world infrastructure |
| 18:26.29 | ManxPower | http://www.tscm.com/phone/cpc.html |
| 18:26.53 | jameswf | Manx dude stop googling get over it |
| 18:27.06 | ManxPower | I was going to suggest the same to you. |
| 18:27.15 | jameswf | just because its a standard doesnt mean bell uses it |
| 18:27.24 | ManxPower | It doesn't mean they don't. |
| 18:27.45 | jameswf | if they did stanards compliant stuff would work |
| 18:27.57 | ManxPower | I give up. |
| 18:27.59 | jameswf | bell is like microsoft standards dont apply |
| 18:28.32 | ManxPower | But at least go whine about it on a different channel. |
| 18:29.02 | jameswf | lmao or ignore ManxPowerfor not knowing who people are |
| 18:29.04 | jameswf | poof |
| 18:29.41 | jameswf | ~newb |
| 18:29.41 | jbot | Don't bother telling us you're a "newb" or a "n00b". We can tell. |
| 18:29.46 | drmessano-LT | I'm pretty sure jameswf has dealt with this problem more than most have |
| 18:29.54 | *** join/#asterisk Docfxit (n=none@cpe-72-129-118-135.socal.res.rr.com) |
| 18:29.59 | drmessano-LT | 10x more than most lol |
| 18:30.20 | jameswf | I have a key to the prencesses chastity belt :) |
| 18:30.27 | drmessano-LT | heh |
| 18:30.48 | jameswf | if only i could type on the korean keyboard |
| 18:30.57 | drmessano-LT | The standard is, there is no standard |
| 18:31.02 | jameswf | konversation needs auto sspeel chek |
| 18:31.32 | *** join/#asterisk oej (n=olle@soll4-125.cust.blixtvik.net) |
| 18:31.40 | jameswf | oh check it out it has auto spell check |
| 18:31.43 | jameswf | werd |
| 18:31.47 | drmessano-LT | werd lol |
| 18:31.47 | Docfxit | I'm trying to install Zaptel in Ubuntu. Is it ok to ask for help here? |
| 18:32.20 | jameswf | but only highlights red to say dude your an idiot no correction bah |
| 18:32.29 | jameswf | yes dox |
| 18:32.41 | Docfxit | Tx |
| 18:34.30 | Docfxit | The first instruction I have to install Zaptel is ./configure I'm getting an error saying C compiler cannot create executables. |
| 18:34.56 | Docfxit | I'm in the terminal window . |
| 18:35.12 | *** part/#asterisk techie (n=techie@adsl-76-214-27-74.dsl.lsan03.sbcglobal.net) |
| 18:35.36 | *** join/#asterisk metfan2007 (n=metfan20@fw.grupositel.com.mx) |
| 18:35.39 | Docfxit | New Install of Ubuntu |
| 18:36.12 | jameswf | do you have build essentials installed |
| 18:37.05 | Docfxit | I didn't. I'm installing them now. |
| 18:37.13 | metfan2007 | hi all!!, it is possible to know the IP used in my Asterisk server to receive inbound h323 calls in the dialplan? |
| 18:38.10 | [TK]D-Fender | metfan2007: why don't you pastebin an inbound call at verbose 10 and show us... |
| 18:38.12 | [TK]D-Fender | ~pb |
| 18:38.13 | jbot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste |
| 18:39.33 | metfan2007 | [TK]D-Fender, http://pastebin.com/m1767d221 |
| 18:40.01 | [TK]D-Fender | metfan2007: Dunno about you, but I see the IP pretty clearly there... |
| 18:40.50 | metfan2007 | [TK]D-Fender, here is the challenge, I'm receiving H323 trunks, and then I bridged the calls to another H323 trunk, that's the other party IP, I need to know witch of my Asterisk IP is connected to... |
| 18:41.57 | [TK]D-Fender | metfan2007: -- Executing [56120983@default:7] Dial("H323/ip$172.16.2.16:20677/15324", "H323/856120983@172.16.2.16:1720|40") in new stack |
| 18:42.01 | Docfxit | To install Zaptel I ran ./configure then make then make install I'm getting an error saying zaptel.rules: Permission denied |
| 18:42.52 | metfan2007 | [TK]D-Fender, the issue here is that I'm not able to read the CallerID sent by the other party (Avaya), and I need to get some identifier sent by Avaya, about witch campaign is.... the IP that you see is the Avaya IP |
| 18:43.27 | *** join/#asterisk d00gster (n=doughant@bas1-toronto12-1088937109.dsl.bell.ca) |
| 18:43.53 | [TK]D-Fender | metfan2007: well you said you wanted to know where the call came from... well, there it is.. |
| 18:45.04 | [TK]D-Fender | Docfxit: http://www.chr05210084.com/content/install-asterisk-1.4-and-asterisk-gui-ubuntu-7.04-server |
| 18:45.42 | *** part/#asterisk oej (n=olle@soll4-125.cust.blixtvik.net) |
| 18:45.47 | *** join/#asterisk iratik (n=itariki@12-226-116-3.client.mchsi.com) |
| 18:45.52 | metfan2007 | [TK]D-Fender, the first option I tried was to identify the campaign by CallerID from Avaya, but that's not working, the second option that I'm trying is to setup a few Virtual IPs in Asterisk, so if Avaya is trying to Dial from campaign 1 use the Asterisk IP 1, if it is trying to dial a campaign 2 number then use Asterisk IP 2... and so on... so I need to figure out witch of Asterisk IPs is using.... |
| 18:46.22 | ManxPower | metfan2007: Asterisk does not pass the info to the outgoing call. |
| 18:46.27 | *** join/#asterisk abaci (n=IceChat7@ool-4b7fc532.static.optonline.net) |
| 18:46.29 | [TK]D-Fender | metfan2007: AH, wanting to know which IP the call is coming in ON... well that... No, I don't believe you'll have a way... |
| 18:46.42 | ManxPower | If you were using SIP, you should be able to get out of the headers. |
| 18:46.50 | [TK]D-Fender | indeed |
| 18:47.01 | metfan2007 | I now that, but we are using H323... xD |
| 18:47.10 | ManxPower | metfan2007: It sucks to be you. |
| 18:47.37 | tzafrir | Docfxit, can you please pastebin a more complete trace from the error and the commands around it? |
| 18:49.04 | *** join/#asterisk Tond (n=t@CPE0014bf30c190-CM00194747ae5e.cpe.net.cable.rogers.com) |
| 18:49.58 | Tond | Hi, I am using asterisk 1.2 and getting some performance issues such as choppy voice a few minutes into the conversations, etc |
| 18:50.19 | Tond | i did a top to look at my ssystem's usge and it doesn't seem very loaded |
| 18:50.33 | Tond | results of top is here: http://pastebin.com/d16b00b5e |
| 18:50.45 | Tond | there are 8 instances of mpg123 that is each taking 27% cpu usage |
| 18:51.07 | Tond | and most of CPU resource is at the kernel level abouts 75% |
| 18:51.22 | [TK]D-Fender | Tond: ditch mpg123 and use Native MoH |
| 18:52.05 | Tond | I had tried to use it b4 and it didn't work for some reason... |
| 18:52.57 | [TK]D-Fender | Tond: Try it now and we'll see about getting it working like it should. Clearly MPG123 is ont he "offenders" list for your setup |
| 18:53.19 | Tond | :) |
| 18:53.38 | Tond | k, thanks.. for native MOH i need to convert my MP3 files right? |
| 18:55.27 | *** part/#asterisk zelip (n=zelip@nat/hp/x-bd82d8c15a473f03) |
| 18:56.17 | Docfxit | tzafrir » http://pastebin.com/m4e1468b1 |
| 18:57.30 | tzafrir | you need to run 'make install' as root |
| 18:57.48 | Tond | where can i get a sample native MOH file? |
| 18:58.01 | Tond | or will any mp3 file do just fine? |
| 18:58.08 | Docfxit | How can I get to the root? |
| 18:58.21 | Tond | Doc> su - |
| 18:58.22 | tzafrir | sudo |
| 18:58.27 | tzafrir | either |
| 18:58.34 | Docfxit | Tx |
| 19:00.37 | metfan2007 | [TK]D-Fender, ManxPower, is there any way to use Wireshark to know what I'm receiving as CallerID from a H323 trunk? |
| 19:01.04 | ManxPower | metfan2007: I have no idea. I spent 2 years in therapy the last time I tried H323. |
| 19:01.30 | metfan2007 | ManxPower, hahaha..... |
| 19:01.31 | Tond | metfan> have you got h323 workign stable with asterisk? the last tie i tried it it made the box very unstable |
| 19:02.35 | metfan2007 | Tond, At this moment it is bridging 10 simultaneous calls in a desktop PC xD, but It is only bridging calls... |
| 19:02.49 | Tond | i see |
| 19:03.20 | Tond | metfan> so it just connect them, no codec translation, etc right? |
| 19:03.51 | patrick-- | Hi all. I cant seem to find out why my asterisk is giving me those error messages: mISDN_close: fid(18) isize(131072) inbuf(0xb71f4008) irp(0xb71f4008) iend(0xb71f4008) |
| 19:03.52 | metfan2007 | But that is only a test equipment, I'm working right now in a production state, with racks servers and 700 calls :S |
| 19:03.58 | Tond | metfan> is it cpu intesive? have you had any crashes? which docs did you follow and did u have to make twicks and adjustments? |
| 19:03.59 | metfan2007 | yep, no codec translation |
| 19:04.32 | metfan2007 | I'm using H323 channel, no OH323, no OOH323 |
| 19:05.03 | DrAk0 | metfan2007, how did you get H323 to compile? |
| 19:05.17 | Tond | metfan> and you didn't have to do anything out of the ordinary or make any adjustments out of the doc right? |
| 19:05.34 | Tond | cause maybe I will give it a try too if it si straight forward and stable now |
| 19:05.59 | DrAk0 | Tond, been getting lots of crashes with OOH323 |
| 19:06.13 | Tond | i see |
| 19:06.15 | metfan2007 | well, I used some specific versions of OpenH323 and Pwlib |
| 19:06.31 | DrAk0 | metfan2007, i tried but openh323 does not compile |
| 19:06.49 | metfan2007 | I'm writing in my blog the way I did, asteriskmx.blogspot.com |
| 19:06.58 | DrAk0 | metfan2007, http://pastebin.ca/901532 |
| 19:07.01 | metfan2007 | wait it till night |
| 19:07.02 | Tond | tnx |
| 19:07.35 | *** join/#asterisk seanbright (n=elixer@65.207.74.18) |
| 19:07.38 | Tond | dude it's in spanish |
| 19:07.38 | Tond | lol |
| 19:08.20 | metfan2007 | make sure u have openssl-devel and expat-devel |
| 19:08.21 | bkruse | lmadsen: I remember our rpmbuild excursion |
| 19:08.26 | lmadsen | lol |
| 19:08.34 | lmadsen | wow, you did some serious scrollback :) |
| 19:08.38 | DrAk0 | lol i read spanish :D |
| 19:08.43 | metfan2007 | hehehehehe |
| 19:08.51 | lmadsen | bkruse: too bad it didn't end up in success |
| 19:09.29 | bkruse | lmadsen: We should try that again. And yes, some serious scrollback :] |
| 19:09.32 | Docfxit | I'm trying to install Asterisk in Ubuntu. I started with the command ./configure . I'm getting an error saying *** termcap support not found. Any ideas why? |
| 19:09.36 | Tond | meta, dark> yall lucky.. we were taught French as second language not spanish |
| 19:09.40 | lmadsen | bkruse: indeed, I'd love to figure out how to do it... |
| 19:09.46 | russellb | Docfxit: hmmmmmmmm ... |
| 19:09.47 | Tond | perhpas i'll try to use translation |
| 19:09.51 | russellb | it might be because termcap support wasn't found |
| 19:09.52 | russellb | :) |
| 19:09.58 | lmadsen | Docfxit: you're missing the termcap libraries? :) |
| 19:09.59 | russellb | sudo apt-get install libncurses-dev |
| 19:10.08 | bkruse | apt-get install libncurses-dev |
| 19:10.09 | bkruse | dangit |
| 19:10.09 | Docfxit | Tx |
| 19:10.15 | russellb | bkruse: i win |
| 19:10.15 | lmadsen | bkruse: sloooooooooooooooow |
| 19:10.19 | bkruse | russellb: totally. |
| 19:10.24 | lmadsen | yum install libncurses-devel :) |
| 19:10.27 | bkruse | I included libncurses and was backspacing :[ |
| 19:10.31 | metfan2007 | DrAk0, use pwlib v1.10.0 and openh323 v1.18.0 |
| 19:10.39 | bkruse | yum? what is yum? apt-get is life |
| 19:10.59 | jameswf | yum is the RHEL pac man |
| 19:11.00 | DrAk0 | im using openh323 1.18 |
| 19:11.10 | DrAk0 | and ptlib_v1_12_0 |
| 19:11.18 | jameswf | pacman is pretty cool |
| 19:11.35 | bkruse | apt-get remove --purge jameswf |
| 19:11.36 | jameswf | I think only arch uses it |
| 19:11.43 | bkruse | :] |
| 19:11.47 | metfan2007 | not all versions of pwlib works with all versions of openh323 |
| 19:12.15 | jameswf | bah on package managers be a man work from source |
| 19:12.25 | Docfxit | I'm getting a message saying libncurses-dev has no installation candidate |
| 19:12.38 | Qwell | libncurses5-dev |
| 19:12.45 | metfan2007 | DrAk0, in your pastebin I see you are using 1.19 |
| 19:12.48 | metfan2007 | /usr/src/src-asterisk/openh323/openh323_v1_19_0_1/include/h4601.h: In member function ‘H460_FeatureContent::operator H460_FeatureTable*()’: |
| 19:12.52 | Docfxit | Tx |
| 19:12.55 | Qwell | russellb, bkruse: You both lose :P |
| 19:13.10 | jameswf | I think it took a week for our engineer to emerge kde... |
| 19:13.17 | russellb | Qwell: whyfor? |
| 19:13.22 | Qwell | russellb: missed the 5 :D |
| 19:13.30 | russellb | i thought it was aliased ... |
| 19:13.31 | russellb | that's sad |
| 19:13.45 | Qwell | the regular libncurses package might b |
| 19:13.46 | Qwell | e |
| 19:14.19 | Docfxit | I'm getting an error saying libncurses5-dev couldn't find package |
| 19:14.33 | Qwell | Docfxit: debian, right? |
| 19:14.40 | Docfxit | yes |
| 19:15.30 | kyron | jameswf, a week, your engeneers are week unless they were doing something "special" or totaly masochistic like using ICC :P |
| 19:15.34 | jameswf | The following NEW packages will be installed: |
| 19:15.34 | jameswf | <PROTECTED> |
| 19:15.52 | *** join/#asterisk bsdwarrior (n=mfahey@fahey.enter.net) |
| 19:16.05 | jameswf | nah emerge over a crap arse t1 with all the deps and kde is a bad idea |
| 19:16.08 | bsdwarrior | how can I forward one ext to another without doing this on the phones ? |
| 19:16.17 | Tond | <[TK]D-Fender> got the native MOH working... Hopefuly that would bring the load down |
| 19:16.36 | Tond | but i still see mpg123 instances in the mem.. can i kill those? |
| 19:16.49 | *** join/#asterisk dlynes (n=dlynes@mail.247communications.com) |
| 19:17.04 | kyron | jameswf, even on a crap arse T1, unless it's as bad as a 56.6kbps modem :P |
| 19:17.40 | Qwell | Docfxit: don't know what to tell you - maybe one of the debian folks (tzafrir?) are around and have an idea |
| 19:17.51 | jameswf | well people around here like to open 10 connections to pull an iso and with that you watch the bandwith go to hell |
| 19:18.24 | jameswf | I do heavy downloading from home 8+ meg pipe |
| 19:18.29 | bkruse | Docfxit: apt-get update; apt-cache search libncurses |
| 19:18.56 | seanbright | Docfxit: maybe 'apt-get install ncurses-dev' |
| 19:18.57 | bkruse | there is also a termcap package for legacy use (which is NOT the libncurses library) (termcap-compat) |
| 19:18.57 | jameswf | they say 12megs but i think that is some crap for windoze |
| 19:19.01 | patrick-- | Hi all. I cant seem to find out why my asterisk is giving me those error messages: mISDN_close: fid(18) isize(131072) inbuf(0xb71f4008) irp(0xb71f4008) iend(0xb71f4008) |
| 19:19.04 | bkruse | seanbright: good idea |
| 19:19.20 | Qwell | Docfxit: apt-cache search ncurses, pastebin the results, maybe we can find a good one |
| 19:19.36 | Docfxit | When I go into Synaptic it says libncurses5 is installed. |
| 19:19.48 | dlynes | seanbright: Have you been able to get unixodbc cdr working in asterisk, under debian etch? |
| 19:19.53 | tzafrir | Docfxit, http://packages.ubuntu.com/cgi-bin/search_packages.pl?keywords=libncurses&searchon=names&subword=1&version=feisty&release=all |
| 19:19.56 | jameswf | Docfxit: -DEV ? |
| 19:20.05 | seanbright | dlynes: i don't run debian. |
| 19:20.16 | dlynes | Docfxit: also make sure to run configure after doing the apt-get installs |
| 19:20.16 | tzafrir | Are you sure you have complete apt sources? |
| 19:20.19 | bkruse | Hey guys, if I have a global variable defined in [globals], do I have to reference it by ${GLOBAL(varname)} ? because ${varname} works fine too |
| 19:20.29 | Qwell | bkruse: GLOBAL() is for setting |
| 19:20.35 | dlynes | Docfxit: before you do the make menuselect |
| 19:20.37 | bkruse | Docfxit: pastebin sources.list |
| 19:20.41 | dlynes | seanbright: oh |
| 19:20.56 | bkruse | Qwell: Ahhh!! But when I did Noop(${GLOBAL(varname)}) it worked? |
| 19:21.02 | dlynes | bkruse: Have you been able to get unixodbc cdr working in asterisk, under debian etch? |
| 19:21.05 | Qwell | bkruse: yeah, no reason for it not to have a .read also |
| 19:21.12 | Qwell | it's unnecessary though |
| 19:21.20 | bkruse | Qwell: Perfect |
| 19:21.30 | Qwell | I would recommend using it if you intend to use the global, but.. |
| 19:21.40 | bkruse | But it would not be standard to read global vars like that then, just reference them by name ${varname} right? |
| 19:22.00 | bkruse | dlynes: I have not messed around with it to much, sorry :[ |
| 19:22.04 | seanbright | i just use Set(__FOO=bar) |
| 19:22.05 | *** join/#asterisk keulin (n=cray@ifth-pdcr2.infotheme.fr) |
| 19:22.09 | dlynes | bkruse: ok, thanks |
| 19:22.10 | bkruse | From what I did do, just used a voip wiki page |
| 19:23.30 | dlynes | bkruse: yeah...that's what I used also, but it didn't seem to work |
| 19:23.43 | bkruse | Qwell: I might run some dialplan by you later, if you do not mind, on doing trunk failover for the gui |
| 19:23.46 | dlynes | bkruse: didn't have any errors, but it just kept using the csv's, and not the database |
| 19:24.12 | bkruse | Hmmm, are you sure everything is connecting (to the database) ? |
| 19:24.23 | dlynes | bkruse: yeah...tested it using isql |
| 19:24.36 | bkruse | Ya, but make sure asterisk is not complaining about accessing it |
| 19:24.57 | kyron | jameswf, yeah well I am starting to think my ISP is screwing with me as I am supposed to have a supposed 5MBPS link but it seems to stall at 1.8... |
| 19:24.58 | dlynes | bkruse: yeah...asterisk was complaining steady...not sure why though...isql never had a problem with it |
| 19:25.36 | dlynes | bkruse: you just reminded me that there was errors...just errors that didn't make any sense, considering how isql never had a problem |
| 19:26.49 | bkruse | If the connection cannot be established, I believe it forgets about the connection and goes and falls back.....to Master.csv (or whatever is defined otherwise) |
| 19:27.08 | Docfxit | @ bkruse» Okay |
| 19:29.41 | bkruse | Docfxit: Anything about ncurses that you tried to install? |
| 19:30.02 | Docfxit | I'll have to continue this later. Thank you all for your help. |
| 19:30.10 | bkruse | np, cya |
| 19:38.28 | [TK]D-Fender | kyron: Who are you with, and how much $/mo? |
| 19:39.02 | [TK]D-Fender | bkruse: Screw globals.... AstDB FTW |
| 19:44.07 | *** join/#asterisk simbol76 (n=simbol@host242-226-dynamic.35-79-r.retail.telecomitalia.it) |
| 19:44.38 | *** join/#asterisk arekm (i=arekm@pld-linux/arekm) |
| 19:44.59 | *** join/#asterisk teknoprep (n=Chris@unaffiliated/teknoprep) |
| 19:45.00 | *** join/#asterisk angom (n=angom@200.56.104.87.dsl.dyn.telnor.net) |
| 19:50.44 | *** join/#asterisk ph0ne (n=ph0ne@dsl-207-112-91-102.tor.primus.ca) |
| 19:51.05 | *** join/#asterisk simbol76 (n=simbol@host242-226-dynamic.35-79-r.retail.telecomitalia.it) |
| 19:51.30 | simbol76 | Italiani?? |
| 19:51.51 | *** join/#asterisk jblack (n=jblack@pool-71-173-53-239.sctnpa.east.verizon.net) |
| 19:52.07 | dlynes | bkruse: do you know if you can have both unixodbc cdr and csv cdr running at the same time? |
| 19:52.16 | Qwell | dlynes: you can |
| 19:52.28 | *** join/#asterisk worgil (n=worgil@88.231.37.24) |
| 19:52.32 | Qwell | it writes to all loaded(registered) cdr routines |
| 19:52.35 | dlynes | Qwell: ok, thanks...just wanted to try and get it working, and not lose track of any calls in the process |
| 19:53.00 | *** join/#asterisk mintee (n=mintee@72-165-177-90.dia.static.qwest.net) |
| 19:53.58 | JenniferAkemi | which codec would you recommend using if i'm going to be sending calls out on a PRI |
| 19:54.04 | JenniferAkemi | on the inbound i mean |
| 19:54.10 | dlynes | Qwell: Any idea what would cause it to get a SQLConnect return value of -1, then? |
| 19:54.30 | [TK]D-Fender | JenniferAkemi: G.711 |
| 19:55.13 | *** join/#asterisk yahya (n=yahya@ner-as26589.alshamil.net.ae) |
| 19:55.21 | JenniferAkemi | that's pretty high bw though. |
| 19:55.53 | JenniferAkemi | i was wondering if we should go 729a |
| 19:56.12 | JenniferAkemi | calls will be coming in from the internet too so that's why bw is a consideration |
| 19:56.38 | *** join/#asterisk weazahl (n=jeremy@adsl-76-230-116-17.dsl.ksc2mo.sbcglobal.net) |
| 19:56.45 | *** join/#asterisk simbol76 (n=simbol@host242-226-dynamic.35-79-r.retail.telecomitalia.it) |
| 19:57.06 | weazahl | anyone in here know MERLIN? i cant get into it for programming |
| 19:57.16 | *** part/#asterisk wwalker (n=wwalker@pdpc/supporter/sustaining/wwalker) |
| 19:57.26 | *** join/#asterisk jm|laptop (n=jm|home@dilbert.jamiem.com) |
| 19:57.29 | teknoprep | for some reason my polycom phone isn't pickup up the TFTP boot server from DHCP |
| 19:58.16 | [TK]D-Fender | teknoprep: make sure you told it to look for one in your BR setup |
| 19:58.30 | teknoprep | [TK]D-Fender, BR setup ? |
| 19:58.35 | [TK]D-Fender | JenniferAkemi: You asked for "best" for your PRI, and G.711 is it. |
| 19:58.41 | *** join/#asterisk simbol76 (n=simbol@host242-226-dynamic.35-79-r.retail.telecomitalia.it) |
| 19:58.44 | teknoprep | [TK]D-Fender, oh i know what you are talking about |
| 19:59.02 | [TK]D-Fender | JenniferAkemi: What you've done is like asking "Whats the best car I can buy? I'm going to buy a Lada anyways...." |
| 19:59.10 | [TK]D-Fender | teknoprep: Opt66 <--- |
| 19:59.11 | arekm | hello, zap show channels shows me one channel doubled but once it's marked as "pseudo" - http://pld.pastebin.com/f56ec665e . Question: what's "pseudo" channel? |
| 19:59.29 | JenniferAkemi | ok |
| 19:59.32 | [TK]D-Fender | arekm: ztdummy |
| 19:59.32 | JenniferAkemi | sorry |
| 19:59.33 | Qwell | arekm: pseudo channels are used for timing, for things like meetme |
| 19:59.42 | Qwell | ignore them, basically |
| 19:59.44 | [TK]D-Fender | arekm: yeah that ;) |
| 20:00.22 | jblack | Today, it rained in my bedroom. |
| 20:00.50 | arekm | I'm having a problem with it - all outgoing calls going via this zap end with Unable to create channel of type 'Zap' (cause 34 - Circuit/channel congestion). Incoming work fine. Unfortunately this is bristuffed asterisk/zaptel and quadgsm card |
| 20:01.25 | JenniferAkemi | what's the best choice to get voice quality that doesn't make people want to hang up the voip phone and call back on a landline, but that takes into consideration internet bw isn't free |
| 20:01.53 | cpm | get rid of the landline |
| 20:01.58 | cpm | all better now. |
| 20:02.24 | yahya | How do I match numbers dialed with a + prefix? |
| 20:03.14 | chavigny | yahya |
| 20:03.18 | chavigny | you want it to add 00? |
| 20:03.19 | dlynes | Is there any way to get more debug information out of the asterisk cdr_odbc module? |
| 20:04.23 | worgil | how many cal i can do call with g729 codec same time ? |
| 20:04.41 | yahya | I have a sip capable cellphone and all my numbers are stored with a + prefix. right now when I dial 1800 222 2222 it works. but now when I do +1800 222 22222 |
| 20:08.56 | yahya | when I dial +1800 222 2222 for example, asterisk doesn't match _1NXXNXXXXXX, but when I put _.,1,SayDigits(${EXTEN}), it matched and read digits starting from 1800.. |
| 20:09.06 | simbol76 | Qualcuno italiano??? |
| 20:09.08 | *** join/#asterisk Nivex (n=kjotte@user-0c8hvoj.cable.mindspring.com) |
| 20:09.21 | simbol76 | hi chat!! |
| 20:09.25 | dlynes | yahya: _+1NXXNXXXXXX, ... |
| 20:09.33 | simbol76 | I have the problem with asterisk 1.4 |
| 20:09.42 | simbol76 | |
| 20:09.42 | simbol76 | I have an Asterisk 1.4 and use the command AgentLogin (). |
| 20:09.42 | simbol76 | I come from version 1.2 and have found that chan_agent does not support features.conf. |
| 20:09.42 | simbol76 | Someone knows how to implement chan_agent.c to enable features.conf? |
| 20:09.56 | simbol76 | anyidea???? |
| 20:09.57 | yahya | dlynes: oh the pattern can take + too ? cool |
| 20:10.57 | simbol76 | or another solution?? |
| 20:13.36 | kyron | [TK]D-Fender, Radioactif.com, got static IP, no blocked ports, ~35$/mo |
| 20:13.45 | dlynes | yahya: it can take a '*', too |
| 20:13.59 | dlynes | yahya: in case you need to match *69, *67, *980, ... |
| 20:14.00 | yahya | dlynes: thanks. it works now |
| 20:14.15 | kyron | [TK]D-Fender, http://www.radioactif.com/forfaits_internet/ |
| 20:14.39 | [TK]D-Fender | kyron: OH.... DSL.... |
| 20:14.48 | kyron | [TK]D-Fender, yeah... |
| 20:15.01 | [TK]D-Fender | kyron: I was thinking fiber for a moment there... |
| 20:15.23 | kyron | [TK]D-Fender, LOL, I wish, this is at home ;) |
| 20:16.38 | *** join/#asterisk codazoda (n=chatzill@207.155.179.56) |
| 20:17.16 | [TK]D-Fender | kyron: www.pubnix.net |
| 20:17.22 | [TK]D-Fender | kyron: Good people. |
| 20:18.14 | Tond | <[TK]D-Fender> i am using native MOH now but i sitll see mpg123 in my memory |
| 20:18.21 | Tond | any idea why? |
| 20:18.44 | kyron | [TK]D-Fender, your provider? |
| 20:18.54 | codazoda | Hi. I've got a CentOS box that won't start asterisk automatically. When I run 'chkconfig --list | grep asterisk' I see '0:off 1:off 2:on 3:on 4:on 5:on 6:off'. If I do 'service asterisk start' it says 'OK' but asterisk isn't running. If I run safe_asterisk, everything works. |
| 20:19.04 | [TK]D-Fender | kyron: Yup, and I gett 100% of the 5000/800 I'm paying for |
| 20:19.09 | kyron | [TK]D-Fender, from what I see, they need a good translator ;) |
| 20:19.36 | [TK]D-Fender | codazoda: make sure its not bombing for not having zaptel on startup, etc... |
| 20:19.43 | kyron | [TK]D-Fender, at the moment, I am attempting to identify the culprit: is it my modem/distance to CO/the ISP??? |
| 20:19.47 | [TK]D-Fender | kyron: They don't focus on that obviously |
| 20:19.52 | *** join/#asterisk scardinal (n=supreme@port816.ds1-suoe.adsl.cybercity.dk) |
| 20:20.14 | *** join/#asterisk glen2 (n=glen@87-194-2-134.bethere.co.uk) |
| 20:21.33 | *** join/#asterisk mamacito62 (n=XD@201.208.84.21) |
| 20:22.05 | kyron | [TK]D-Fender, 114$/Month?? O_o.. with a 300$ setup fee O_o!! |
| 20:22.07 | slima | hehehe, niez³y opis na allegro: |
| 20:22.09 | slima | Witam. Mam dzi¶ do sprzedania Switch firmy D-Link na 8 portów. Switch jest praktycznie nowy, gdy¿ nied³ugo po jego zakupie niektórzy u¿ytkownicy po prostu zrezygnowali z bycia w sieci. Switch byl w u¿ytku zaledwie kilka dni. |
| 20:22.33 | [TK]D-Fender | kyron: Wrong service. Mine is $45 w fixed IP, dry loop & ulim BW |
| 20:22.35 | slima | to chyba kiepski skoro z jego(?) powodu zrezygnowali z sieci. ;P |
| 20:22.53 | kyron | oops.. let me check that out.. |
| 20:23.01 | slima | OOOPS sorry! Wrong window! |
| 20:23.03 | slima | ;) |
| 20:23.07 | [TK]D-Fender | Tond: have you restarted * after changing to native? |
| 20:23.58 | kyron | [TK]D-Fender, oh...I checked ADSL...didn't realise it was a different link to DSL...which is weird (so which one is DSL, symmetric?) |
| 20:25.14 | codazoda | If I use 'service asterisk start' I get the following in the /var/log/messages file '/usr/sbin/safe_asterisk: line 111: 5857 Aborted (core dumped)'. But, if I use 'safe_asterisk', all is fine. |
| 20:26.03 | *** join/#asterisk danalien (n=danalien@unaffiliated/danalien) |
| 20:26.57 | [TK]D-Fender | kyron: difference is PPPoE vs non-PPPoE |
| 20:26.58 | *** join/#asterisk zobia (n=laurashr@222.212.64.155) |
| 20:27.14 | simbol76 | Qualcuno parla italiano??? |
| 20:27.16 | kyron | [TK]D-Fender, ah yeah, that would make sense |
| 20:27.35 | simbol76 | Avrei un problema |
| 20:27.39 | simbol76 | con asteriak 1.4 |
| 20:27.57 | J4k3 | who cares how fast it is, its all about an effective SLA |
| 20:28.00 | simbol76 | uso nel dialplan il comnado AgentLogin() |
| 20:28.12 | simbol76 | solo che con la versione 1.4 |
| 20:28.18 | codazoda | 'which safe_asterisk' returns '/usr/sbin/safe_asterisk', so I'm executing the same one. Maybe some type of permissions problem (I'm root but services isn't)? |
| 20:28.22 | simbol76 | quando uso chan_agent |
| 20:28.46 | simbol76 | non mi funzionano e feature che configure in features.conf |
| 20:29.00 | simbol76 | cercando su internet |
| 20:29.18 | simbol76 | ho trovato che in asterisk 1.4 le features sono disabilitate per gli agenti |
| 20:29.22 | simbol76 | allora chiedo |
| 20:29.42 | simbol76 | se io uso agentlogin e devo far digitare qualcosa agli agenti mentre stanno parlando |
| 20:29.53 | simbol76 | come faccio con 1.4???? |
| 20:30.12 | tzafrir | wow. There is actually #asterisk-it . Empty :-( |
| 20:30.36 | simbol76 | ok :) |
| 20:30.53 | tzafrir | Please use English here |
| 20:30.57 | simbol76 | |
| 20:30.57 | simbol76 | I have an Asterisk 1.4 and use the command AgentLogin (). |
| 20:30.57 | simbol76 | I come from version 1.2 and have found that chan_agent does not support features.conf. |
| 20:30.57 | simbol76 | Someone knows how to implement chan_agent.c to enable features.conf? |
| 20:31.17 | tzafrir | You've already asked that |
| 20:31.23 | simbol76 | yes |
| 20:31.36 | simbol76 | but no resolve |
| 20:32.22 | *** join/#asterisk simbol76 (n=simbol@host242-226-dynamic.35-79-r.retail.telecomitalia.it) |
| 20:32.59 | codazoda | Since safe_asterisk works fine, is there any harm in shutting off the asterisk service and loading safe_asterisk from rc.d? |
| 20:33.57 | zobia | hello everyone |
| 20:34.49 | *** join/#asterisk enjay5150 (n=chatzill@63.226.63.66) |
| 20:34.55 | simbol76 | hello zobia |
| 20:35.16 | zobia | hi . i have a problem with cisco 7960 |
| 20:35.23 | enjay5150 | experiencing an odd issue installing zaptel.. uname spits out the running kernel version but when I make install of zaptel it installs it in the wrong /lib/modules/kernelversion |
| 20:35.40 | zobia | i configure 2 lines with 7960. but when there are 2 calls come in. always can find the first line. |
| 20:35.47 | enjay5150 | Im running 2.6.12-9-386 yet it installs it in /lib/modules/2.6.12/misc instead of /lib/modules/2.6.12-9.386/misc |
| 20:35.54 | zobia | if the first line is busy . the second phone will go to voicemail |
| 20:35.55 | simbol76 | don't know cisco |
| 20:35.55 | mvanbaak | codazoda: the only harm is that you wont know when asterisk dies because safe_asterisk will restart it for you |
| 20:36.00 | enjay5150 | so modprobe zaptel comes back with a FATAL: Module zaptel not found. |
| 20:36.04 | enjay5150 | any ideas? |
| 20:36.23 | mvanbaak | enjay5150: did you run 'depmod -a' ? |
| 20:36.48 | zobia | simbo176: thank you for answering. anyone else has idea? |
| 20:36.50 | codazoda | mvanbaak: I think that's a good thing... Doesn't the service do the same? |
| 20:36.55 | simbol76 | enjay5150 you have compile zaptel??? |
| 20:37.03 | codazoda | If not, maybe I should do it this way anywayz. |
| 20:37.06 | mvanbaak | codazoda: I like to know when stuff breaks |
| 20:37.10 | enjay5150 | yes I did run depmod and I did compile zaptel.. |
| 20:37.22 | *** join/#asterisk af_ (n=getsmart@88-149-240-211.dynamic.ngi.it) |
| 20:37.34 | simbol76 | version??? |
| 20:37.40 | enjay5150 | 1.2.23 |
| 20:37.44 | codazoda | mvanbaak: Me too, but that's what log files are for. I'd rather use those than have a production server bite the dust randomly. :-) |
| 20:38.04 | codazoda | I mean, it will still be random, but the next call will come through (probably). |
| 20:38.25 | simbol76 | you have a dgium card?? |
| 20:38.27 | codazoda | K, I'll try this route for now. Thanks for the help. |
| 20:38.38 | enjay5150 | yes TDM04B |
| 20:38.42 | simbol76 | you have any card installed?? |
| 20:38.47 | simbol76 | ok |
| 20:38.57 | mvanbaak | hhmm, my asterisk never dies |
| 20:38.58 | simbol76 | now |
| 20:39.07 | simbol76 | you compile zaptel first |
| 20:39.11 | simbol76 | with |
| 20:39.17 | simbol76 | make clean |
| 20:39.21 | enjay5150 | right make clean |
| 20:39.23 | enjay5150 | make |
| 20:39.24 | enjay5150 | make install |
| 20:39.27 | simbol76 | ok |
| 20:39.35 | enjay5150 | depmod -a |
| 20:39.35 | mvanbaak | now run: depmod -a |
| 20:39.38 | enjay5150 | modprobe zaptel |
| 20:39.45 | enjay5150 | FATAL: Module zaptel not found |
| 20:40.07 | enjay5150 | its very bazaar.. |
| 20:40.22 | simbol76 | yes |
| 20:40.33 | mvanbaak | hhmm |
| 20:40.46 | *** join/#asterisk Yourname`` (n=Miranda@unaffiliated/yourname/x-837320) |
| 20:40.51 | mvanbaak | the modules are not in /lib/modules/`uname -r` ? |
| 20:40.51 | enjay5150 | even odder zaptel.ko is only found in /lib/modules/2.6.12/misc/zaptel.ko not in /lib/modules/2.6.12-9.386/misc which is the kernel Im running.. |
| 20:40.57 | mvanbaak | ok |
| 20:41.10 | mvanbaak | did you install the kernel headers for 2.6.12-9.386 ? |
| 20:41.59 | enjay5150 | trying something else real quick.. |
| 20:41.59 | simbol76 | OS & Version kernel??? |
| 20:42.25 | enjay5150 | moving 2.6.12 to 2.6.12.old and doing a symbolic link from 2.6.12-9-386 to 2.6.12 |
| 20:43.52 | enjay5150 | sweet that worked.. |
| 20:44.06 | enjay5150 | what a pita... I hate fixing other peoples half ass installs.. |
| 20:44.16 | enjay5150 | thanks much guys... |
| 20:44.37 | [TK]D-Fender | enjay5150: With an equally hack-ish half-assed fix ;) |
| 20:44.51 | enjay5150 | hahah true true... |
| 20:45.15 | sweeper | smlink? what the hell...just change your bootloader :v |
| 20:45.15 | enjay5150 | I just want them up and running.. I'll rebuild their system later not on ubuntu.. |
| 20:45.22 | *** join/#asterisk gutz|work (n=mark@gateway.meteor-web.com) |
| 20:45.29 | gutz|work | hi all |
| 20:45.45 | gutz|work | where can i download the latest version of the IAXy provisioning software? |
| 20:47.50 | BBHoss | gutz|work, svn |
| 20:48.04 | BBHoss | or ftp either one |
| 20:48.17 | gutz|work | hmmm, where is the FTP server? i havent been able to find a link |
| 20:48.21 | Yourname`` | How can I simulate a NoOp in the CLI? |
| 20:48.23 | BBHoss | ints in the users manual |
| 20:49.10 | Qwell | Yourname``: press enter |
| 20:49.11 | gutz|work | the users manual i lost? |
| 20:49.23 | Yourname`` | Qwell: You mean type and pres enter? |
| 20:49.27 | BBHoss | heh |
| 20:49.28 | Qwell | no, I mean press enter |
| 20:49.35 | Qwell | noop means do...nothing |
| 20:49.41 | seanbright | "no operation" |
| 20:50.05 | BBHoss | gutz|work, svn co http://svn.digium.com/iaxyprov/trunk/ iaxyprov |
| 20:50.11 | BBHoss | that should get you the files |
| 20:50.29 | *** join/#asterisk hmodes (i=hmodes@B1-66ER.matrix.gs) |
| 20:50.42 | *** join/#asterisk dmz (n=dmz@64.253.5.180.dyn-cm-pool75.pool.hargray.net) |
| 20:50.54 | Yourname`` | Basically, one of the extensions has a NoOp(Phone number:) thing going. But to test from manager interface, I wanted to just make NoOp show up in the CLI. (Logging using manager) |
| 20:50.57 | gutz|work | hmm, that link didnt work |
| 20:51.04 | gutz|work | 405 method not allowed |
| 20:51.13 | *** join/#asterisk lvl- (n=lvl@145.52.248.101) |
| 20:51.19 | BBHoss | follow this guide: https://www.voiptalk.org/products/digium-iaxy-s101i-setup |
| 20:51.33 | gutz|work | OK, reading |
| 20:51.58 | dmz | hwody y'all, quick problem here...i have 2 asterisk boxes, one in colo receiving / sending calls to SIP provider, other here @ house connected via vpn/iax. things use to work but i upgraded from 1.2 -> 1.4 (debian) and now whenever a call goes into the queue it ever comes out. it rings on an angents desk but never bridges or transers the call over. any suggestions? this use to work :( |
| 20:52.10 | arekm | if I change order of span= in zaptel.conf, will that change mapping to physical card too? like this http://pld.pastebin.com/m7c73e15b |
| 20:52.23 | tzafrir | arekm, no problem |
| 20:52.29 | Yourname`` | Qwell: One of the extensions does a noop and shows the phone number of the caller on CLI. An AGI is using manager to log those phone numbers shown on the CLI. Without having to actually call, I'm trying to simulate a NoOp on the CLI. |
| 20:52.35 | BBHoss | dmz, did you read upgrade.txt and make sure you aren't using anything thats changed |
| 20:52.46 | JenniferAkemi | i think i might be missing something here, but when for some reason asterisk always wants to transcode my 726 to pcm. both phones can handle 726, but when i look at the channels in the CLI it says the outbound one is pcm and since it's transcoded it sounds awful. Is there somewhere I am supposed to set it to not always send pcm? |
| 20:52.53 | dmz | yeah i went through everythig and it "was" working after the upgrade |
| 20:53.00 | tzafrir | it's the content that matters. |
| 20:53.00 | dmz | it stopped working sometime last week |
| 20:53.05 | BBHoss | dmz, what changed? |
| 20:53.16 | dmz | not sure what's wrong. i can call direct extensions but anything in the queue doesn't work |
| 20:53.36 | dmz | i had lots of variable changes, functions that aren't used in 1.4 anymroe, took me a while to upgrade my extensions.conf |
| 20:53.47 | BBHoss | hmm |
| 20:53.49 | JenniferAkemi | when i do a direct ip call (on the phone it's a grandstream 2000) to the other phone it works fine and stays 726 |
| 20:53.51 | mvanbaak | 21:50 < BBHoss> gutz|work, svn co http://svn.digium.com/iaxyprov/trunk/ iaxyprov |
| 20:54.07 | BBHoss | ? |
| 20:54.11 | mvanbaak | that should be: svn co http://svn.digium.com/svn/iaxyprov/trunk iaxyprov |
| 20:54.17 | arekm | tzafrir: so physical card 3 becomes span 4 as seen by asterisk? Because I did that and still span 4 had "congestion" error |
| 20:54.20 | Yourname`` | brb |
| 20:54.23 | tzafrir | arekm, you practically changed almost nothing. |
| 20:54.45 | arekm | tzafrir: changed ordering of 2 last span= entries |
| 20:55.12 | tzafrir | it's not the number. It's a priority for being a sync source |
| 20:55.26 | gutz|work | OK, i have the new version |
| 20:55.31 | mvanbaak | digium did not use <Location /> (webdav and dav_svn stuff here) </Location> |
| 20:55.32 | gutz|work | but it still isnt working |
| 20:55.35 | arekm | tzafrir: ah, so it's impossible then ;/ |
| 20:55.37 | gutz|work | an asterisk debug reveals this |
| 20:55.38 | gutz|work | http://pastebin.com/m7f98d346 |
| 20:55.40 | mvanbaak | the did use: <Location /svn>..... |
| 20:55.48 | tzafrir | arekm, what do you want to do? |
| 20:56.10 | arekm | tzafrir: make 3 physical card be seen as span 4 in asterisk and vice versa (physical card 4 seen as span 3) |
| 20:56.22 | simbol76 | enjay5150 |
| 20:56.36 | simbol76 | http://pastebin.com/d539ddc30 |
| 20:56.41 | simbol76 | for you!!! |
| 20:57.09 | tzafrir | arekm, maybe you can abuse the section trunkgroups in zapata.conf for that . I'm not really sure |
| 20:57.27 | tzafrir | But why do you need that switching? |
| 20:58.01 | arekm | tzafrir: to verify that there is no some "off by one" bug that prevents span 4 (always this one - 4) from working |
| 20:58.02 | egecko | centos 4 or 5 for an * server? |
| 20:58.18 | tzafrir | Maybe you should just use groups instad of single channels |
| 20:58.33 | mvanbaak | egecko: whatever you feel most comfortable with |
| 20:58.45 | gutz|work | i cant get my iaxy to talk to my asterisk server. the iax2 debug i ran showed this error but i have no idea what it means: http://pastebin.com/m7f98d346 |
| 20:58.48 | simbol76 | Slackware :-) |
| 20:58.49 | tzafrir | At this stage, I figure 5 is nicer |
| 20:58.55 | egecko | I'm confortable with whatever works best :-) |
| 20:59.04 | simbol76 | work fine for me |
| 20:59.08 | simbol76 | very stable |
| 20:59.16 | egecko | getting a quote from theplanet.com for a server |
| 20:59.23 | *** join/#asterisk jmesquita (n=jmesquit@201.7.117.114) |
| 20:59.34 | tzafrir | very stable, but becoming old. You want that PBX to last for a few years |
| 20:59.39 | simbol76 | enjay..... |
| 21:00.13 | tzafrir | arekm, what off-by-one? |
| 21:00.14 | gutz|work | best idea, stay away from anything related to Red Hat or SuSe |
| 21:00.20 | *** part/#asterisk simbol76 (n=simbol@host242-226-dynamic.35-79-r.retail.telecomitalia.it) |
| 21:00.25 | *** join/#asterisk ferai (n=jefferai@amarok/developer/mitchell) |
| 21:00.43 | *** join/#asterisk simbol76 (n=simbol@79.35.226.242) |
| 21:00.53 | mvanbaak | gutz|work: amen |
| 21:01.22 | dlynes | bkruse: well, got it working |
| 21:01.34 | tzafrir | gutz|work, do you hint that Centos is RedHat[tm]? ;-) |
| 21:01.35 | mvanbaak | if I'm forced to use PRI/BRI hardware I use Debian, otherwise I use OpenBSD |
| 21:01.36 | dlynes | bkruse: had to write my own debug code into cdr_odbc.c to find out why it was failing |
| 21:01.37 | arekm | tzafrir: in bristuff code, I'm just guessing ;/ |
| 21:01.58 | JenniferAkemi | i wonder if i need to set g726nonstandard = yes? |
| 21:02.02 | dlynes | bkruse: pretty ridiculous that cdr_odbc.c doesn't have enough information to let you know why the connection didn't work |
| 21:02.05 | gutz|work | i certainly do! |
| 21:02.07 | tzafrir | arekm, can you be more specific? |
| 21:03.10 | arekm | tzafrir: my not working span is 4, the last one. spans 1, 2 and 3 are operating correctly. span 4 doesn't work for outgoing only, incoming works fine |
| 21:04.01 | tzafrir | "not working" == ? what error do you get? |
| 21:04.06 | tzafrir | Can you pastebin a trace? |
| 21:05.05 | arekm | tzafrir: http://pld.pastebin.com/m4c04b347 (but as I said earlier, it's bristuffed asterisk so who knows what can hide in it) |
| 21:05.09 | gutz|work | hey, sorry for repeating again but is there anyone who can help me with my iaxy not talking to my * box? Here is the debug link: http://pastebin.com/m7f98d346 |
| 21:05.47 | mvanbaak | make sure the [door] entry in iax.conf is correct |
| 21:05.55 | BBHoss | gutz|work, check your usernames |
| 21:06.13 | tzafrir | arekm, what happens if you swap the lines between port 3 and port 4? |
| 21:06.31 | ferai | Can anyone give me some information about making Asterisk work with local sound cards? I know that it can, but I'm trying to figure out how selectable things are... |
| 21:06.33 | tzafrir | Is it a problem of the line or of your local settings/port? |
| 21:07.16 | gutz|work | BBHoss: they would appear to all match |
| 21:07.32 | arekm | tzafrir: this is (quad) gsm card. Replacing SIMs doesn't change anything - still span 4 fails |
| 21:08.04 | tzafrir | arekm, another sanity check: are you sure that there is something in group 4? can you pastebin your zapata.conf? |
| 21:09.08 | BBHoss | gutz|work, pastebin your iax.conf file and the iaxyprov provisioning file, also the debug information, scrub the passwords if you like ;) |
| 21:10.43 | gutz|work | BBHoss Updated with iaxy.conf working on iax.conf http://pastebin.com/d6be4f039 |
| 21:10.51 | BBHoss | k |
| 21:11.11 | arekm | tzafrir: http://pld.pastebin.com/m1eb39ebe |
| 21:11.53 | gutz|work | BBHoss: updated with the relevant piece of iax.conf |
| 21:12.07 | tzafrir | arekm, right. you need to put the group= line before the channel => line |
| 21:12.34 | *** join/#asterisk teknoprep (n=Chris@unaffiliated/teknoprep) |
| 21:12.59 | BBHoss | gutz|work, and you provisioned this file to the iaxy? |
| 21:13.03 | tzafrir | A 'channel' line applies current values. a 'group' line sets current group value |
| 21:13.06 | gutz|work | yes |
| 21:13.34 | BBHoss | did you do a reload? |
| 21:13.43 | gutz|work | yes |
| 21:14.08 | tzafrir | arekm, BTW: I wonder if genzaptelconf would have made a working config for your system |
| 21:16.26 | BBHoss | gutz|work, does door show up in iax2 show peers/registry etc |
| 21:16.34 | gutz|work | yes |
| 21:16.48 | BBHoss | hmm |
| 21:16.50 | BBHoss | no idea |
| 21:16.57 | BBHoss | lemme look at one of my configs |
| 21:16.59 | gutz|work | k |
| 21:18.35 | BBHoss | gutz|work, try replacing the username and passwords with ONLY numbers instead of text |
| 21:18.43 | gutz|work | how do i use the 'extra debugging' on the iaxy? |
| 21:18.47 | gutz|work | just a netcat? |
| 21:19.02 | BBHoss | it broadcasts to port 9999 |
| 21:19.15 | *** join/#asterisk cmantito (n=gphreak@pool-71-188-82-138.cmdnnj.east.verizon.net) |
| 21:19.26 | ferai | tzafrir: do you know much about using Asterisk with sound cards? |
| 21:19.27 | worgil | how can i set dtmf time ? |
| 21:19.51 | teknoprep | yo |
| 21:20.01 | tzafrir | ferai, I figure others here know a bit more. Ask |
| 21:20.04 | teknoprep | my polycom ip650 phone keeps rebooting on its own? |
| 21:20.14 | teknoprep | says Reboot Initialized on the screen |
| 21:20.37 | gutz|work | BBHoss: same error |
| 21:21.04 | worgil | how can i set dtmf time for asterisk 1.4 ? |
| 21:21.16 | BBHoss | can you try dhcp instead of static, that seems to be the only difference between my and your configs |
| 21:21.24 | gutz|work | i ahve no DHCP server here |
| 21:21.26 | gutz|work | have* |
| 21:21.28 | gutz|work | so no |
| 21:21.29 | BBHoss | hmm |
| 21:21.34 | ferai | I'm wondering how flexible Asterisk is working with ALSA (or possibly with JACK). If I have a multi-input/output sound card with several headsets connected to it, I'd like to be able to have an audio channel from the card pulled into Asterisk to use as one side of a VoIP call |
| 21:21.44 | gutz|work | im installing ethereal right now |
| 21:21.52 | gutz|work | so that i can try and look at this port 9999 stuff |
| 21:22.00 | ferai | So that essentially, multiple people on headsets with mics and some remote site can communicate in a sort of conference |
| 21:22.42 | BBHoss | gutz|work, there is also an iaxydebug program, not sure what it does |
| 21:22.44 | JenniferAkemi | can someone help me figure out why i can't make a g726 codec call from one grandstream to another? |
| 21:22.54 | arekm | tzafrir: making channel = last one for each span part didn't fix the problem unfortunately |
| 21:22.55 | Yourname`` | Hi, One of the extensions does a noop and shows the phone number of the caller on CLI. An AGI is using manager to log those phone numbers shown on the CLI. Without having to actually call, I'm trying to simulate a NoOp on the CLI. How can I do so? |
| 21:23.16 | arekm | tzafrir: genzaptelconf creates bad config according to example one provided with that quadgsm card |
| 21:23.20 | *** join/#asterisk grantm (n=grant@thenetrouter.users.xmission.com) |
| 21:23.36 | BBHoss | gutz|work, looks like it just prints port 9999 out to the screen |
| 21:23.38 | tzafrir | well, I never really tested it. I could only speculate :-) |
| 21:24.01 | tzafrir | arekm, did you reload? |
| 21:24.08 | tzafrir | group should update on reload |
| 21:24.17 | BBHoss | gutz|work, also, you must enable the debug option in iaxy.conf |
| 21:24.27 | gutz|work | k |
| 21:24.34 | gutz|work | brb, gotta enable, reprov and restart |
| 21:24.38 | BBHoss | sure |
| 21:24.46 | arekm | tzafrir: restart even |
| 21:25.25 | tzafrir | can you pastebin your zapata.conf again, please? |
| 21:26.11 | *** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 21:26.54 | arekm | tzafrir: http://pld.pastebin.com/m5a6d32cc |
| 21:28.35 | tzafrir | looks ok |
| 21:28.56 | tzafrir | if you use Zap/7 directly, instead of Zap/G4 ? |
| 21:29.26 | tzafrir | It's kind of pointless to use groups of single channels :-) |
| 21:29.42 | *** join/#asterisk ruied (n=ruied@bl7-219-99.dsl.telepac.pt) |
| 21:30.59 | *** join/#asterisk nny_2 (n=Scott_My@64.203.239.83) |
| 21:31.17 | nny_2 | so anyone have an opinion on using gsm for certain situations over ulaw? |
| 21:31.28 | Qwell | nny_2: bandwidth constraints |
| 21:31.34 | nny_2 | Qwell yeah |
| 21:31.35 | Qwell | that's the only reason to not use ulaw (or better) |
| 21:31.40 | nny_2 | Qwell thats the reason :) |
| 21:31.46 | Qwell | then yeah, gsm is fine |
| 21:32.01 | JenniferAkemi | is gsm better than g726? |
| 21:32.02 | Qwell | it's not great, but what can you expect? |
| 21:32.05 | nny_2 | Wanted to test quality here, but it seems my fancy polycom doesn't support gsm |
| 21:32.08 | Qwell | JenniferAkemi: dunno |
| 21:32.09 | *** join/#asterisk keulin (n=cray@ifth-pdcr2.infotheme.fr) |
| 21:32.14 | mvanbaak | <-- prefers [u|a]law |
| 21:32.15 | nny_2 | 501 polycom* |
| 21:32.16 | Qwell | nny_2: most hardware phones do g729 rather than gsm |
| 21:32.19 | nny_2 | yeah /agree |
| 21:32.24 | nny_2 | Qwell: ahh ok |
| 21:32.35 | JenniferAkemi | g726 sounds pretty good, i just can't get it to work when i go phone -> asterisk box -> phone |
| 21:32.36 | Qwell | use g729 if the phones support it - it's good, but non-free |
| 21:32.45 | nny_2 | i am working on a 800 phone network design with a DOCSIS 2.0 infrastructure |
| 21:33.55 | JenniferAkemi | i can't even figure out how to change the codec for some reason. is there somewhere other than the allow, disallow lines in users.conf and setting it in the phone that I have to change it? |
| 21:34.25 | JenniferAkemi | the system is getting the g726 call, and then converting it to ulaw before sending it on |
| 21:34.38 | JenniferAkemi | and i don't know why |
| 21:36.51 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 21:38.22 | nny_2 | cool will do some testing here thanks for the opinions/advice! |
| 21:43.31 | arekm | tzafrir: whoah, Zap/7 works! |
| 21:43.36 | teknoprep | does anyone know any astersik 3rd party software that integrates with ACT ? so that CID of calls comming in can bring up the account inside of ACT ? |
| 21:44.12 | teknoprep | i know of auto-dialing software for ACT that allows for easy dialing from ACT... but i was looking for inbound ease of use applications for bringing accounts up faster |
| 21:44.26 | arekm | tzafrir: but G4 also works hmm (but there was machine restart in meantime) |
| 21:44.45 | tzafrir | arekm, maybe it is Zap vs. ZAP ? |
| 21:44.56 | jameswf | teknoprep: if you find one let me know cause I hear that question alot but never really have an answer |
| 21:45.05 | teknoprep | jameswf, will do |
| 21:46.51 | JenniferAkemi | is it possible that g726 just doesn't work for some resaon with grandstream phones? |
| 21:46.59 | Qwell | ~gs |
| 21:46.59 | jbot | GrandSuck phones & gateways are cheap junk which should be avoided with extreme prejudice. |
| 21:47.01 | Qwell | a lot is possible |
| 21:47.12 | JenniferAkemi | oh |
| 21:47.20 | mvanbaak | niiiiiiiiiiiiice phone |
| 21:47.27 | arekm | tzafrir: my dialplan has "Dial(Zap/GX/...)" everywhere |
| 21:47.47 | JenniferAkemi | that'w hat i have mvanbaak |
| 21:48.27 | JenniferAkemi | so it sounds like my phone just sucks, i'm not misconfiguring anything? |
| 21:48.40 | JenniferAkemi | i can't even find WHERE to configure the stuff |
| 21:48.43 | JenniferAkemi | in asterisk |
| 21:48.53 | Qwell | sip.conf, allow= |
| 21:49.38 | mvanbaak | what Qwell said |
| 21:49.44 | JenniferAkemi | Qwell, i am using asteriskNOW and it put all my users stuff in users.conf, but it has the appropriate allow=g726 line |
| 21:49.51 | JenniferAkemi | should that work also? |
| 21:49.55 | *** join/#asterisk Winkie (n=urmom@ur.fa.gs) |
| 21:50.15 | Qwell | sure |
| 21:50.21 | JenniferAkemi | well damn. |
| 21:51.09 | *** join/#asterisk craigk (n=craigk@58.174.150.119) |
| 21:53.30 | gutz|work | BBHoss: well, i got it working. I put the iaxy on the same network as the asterisk server and it worked... it makes no sense to me though. IAX should have no problem traversing my router |
| 21:53.38 | gutz|work | i have no access control or NAT set up |
| 21:53.47 | JenniferAkemi | hm. do i have to make sure the codecs are installed in a module or something somewhere |
| 21:57.34 | *** join/#asterisk chavigny (n=nrp@c-67-171-147-26.hsd1.or.comcast.net) |
| 22:02.18 | nny_2 | woudl it work if i set allow=g729 in sip.conf |
| 22:02.19 | nny_2 | ? |
| 22:02.27 | nny_2 | seem to be getting translator path errors |
| 22:03.16 | JenniferAkemi | you have to buy that one |
| 22:04.18 | *** join/#asterisk glen2 (n=glen@87-194-2-134.bethere.co.uk) |
| 22:04.33 | JenniferAkemi | nny_2: http://forums.digium.com/viewtopic.php?t=14108 |
| 22:05.03 | nny_2 | JenniferAkemi: much obliged |
| 22:05.14 | nny_2 | JenniferAkemi: hmm licenses |
| 22:05.24 | JenniferAkemi | nny_2: np. are you able to get g726 working? |
| 22:05.32 | JenniferAkemi | nny_2: it's not too bad, $10 a channel |
| 22:05.38 | ManxPower | g729 licenses from Digium are cheap |
| 22:06.07 | nny_2 | thats not too bad.. checking to see if polycom 501 supports g726 |
| 22:06.21 | JenniferAkemi | g726 is supposed to sound almost as good as pcm (711) but it's half the bw |
| 22:07.01 | *** part/#asterisk arekm (i=arekm@pld-linux/arekm) |
| 22:08.09 | nny_2 | hmm seems polycom 501 doesn't like that codec, it appears all it supports is u/alaw and 729 |
| 22:08.20 | nny_2 | will have to try it out with another phone or softphone |
| 22:08.41 | JenniferAkemi | you only need 729 license if you're going 729 to something else i think |
| 22:08.47 | *** join/#asterisk husimon (n=nhuisman@aeko.IfA.Hawaii.Edu) |
| 22:09.09 | husimon | hey I have some pap2t but none of them seem to have a power led when I plug them in? are they all dead or am I missing something |
| 22:09.43 | nny_2 | JenniferAkemi: hmm |
| 22:10.09 | nny_2 | JenniferAkemi: let me try to set 729 in polycom as first coded, it may have tried to translate due to codec hierarchy |
| 22:10.15 | nny_2 | first codec* |
| 22:11.03 | nny_2 | lol is it wronng I think the snom phones have a leg up on polycom? |
| 22:11.10 | nny_2 | wrong* |
| 22:11.10 | ManxPower | If you don't want G729 then don't enable it in sip.conf |
| 22:11.28 | nny_2 | i am actually trying to test 729 atm ManxPower |
| 22:11.36 | ManxPower | Don't restrict the codec on the phone, restrict it in Asterisk. |
| 22:11.45 | ManxPower | nny_2: then you want to disable all codecs except G729 |
| 22:11.51 | nny_2 | yeah I did |
| 22:12.05 | nny_2 | but the polycom i think was trying to use a translation from 729 |
| 22:12.15 | nny_2 | so i reset the phone with *only* 729 to test |
| 22:13.08 | nny_2 | ahh |
| 22:13.12 | nny_2 | duh |
| 22:13.17 | ManxPower | The polycom won't try to use a codec you didn't allow= in sip.con |
| 22:13.18 | nny_2 | translate from *729* to ZAP |
| 22:13.31 | nny_2 | i misunderstood the translation error, so yeah i need a license |
| 22:13.37 | nny_2 | pick up one tomorrow for testing |
| 22:14.44 | *** join/#asterisk [TK]D-Fender (n=Joe@64.235.216.2) |
| 22:15.39 | nny_2 | also is* EVERYONE picked up a phone at once, which is unlikely, but still physically somewhat possible |
| 22:15.49 | nny_2 | will figure an average useage % later |
| 22:16.12 | nny_2 | bandwidth calculator on asteriskguru rocks my socks |
| 22:18.33 | worgil | which softphone i can use zoiper, x-lite else ? |
| 22:18.54 | timeshell | zoiper has closing issues |
| 22:20.11 | cmantito | SJphone is OK. on the mac anyway, not sure how well the Windows one works. |
| 22:20.31 | worgil | cmantito, i need for windows |
| 22:20.32 | Silent-X | I like x-lite/eyebeam |
| 22:20.45 | cmantito | it exists for windows, I imagine if the mac one works well the windows one should too. |
| 22:21.23 | Silent-X | id like to try out bria, but they dont have a trial, and $60 is quite a bit to decide if you like it or not |
| 22:21.42 | [TK]D-Fender | Ekiga |
| 22:23.24 | timeshell | 3cx |
| 22:23.30 | timeshell | Not great, but an alternative |
| 22:23.38 | timeshell | Saves multiple profiles |
| 22:23.54 | lirakis | later |
| 22:23.56 | *** part/#asterisk lirakis (i=lirakis@66.252.24.133) |
| 22:24.05 | timeshell | I have problems with dtmf and xlite |
| 22:27.48 | *** join/#asterisk jm|home (n=jm|home@zen.jamiem.com) |
| 22:33.21 | husimon | man the advanced config for linksys pap2t is seriously in depth |
| 22:36.51 | [TK]D-Fender | husimon, thats nothing compared to a MediaTrix or AudioCodes unit.... |
| 22:37.20 | husimon | I just wasn't expecting to see the level of detail they have in there. |
| 22:37.41 | husimon | I'm crossing my fingers that the defaults work ok :) |
| 22:37.59 | husimon | the guides i have read mostly just say change the sip user settings. |
| 22:38.13 | [TK]D-Fender | husimon, a quick start on one of those involves setting 4 fields, and the dialplan |
| 22:38.44 | *** join/#asterisk lunaphyte (n=lunaphyt@static-71-120-128-10.gdrpmi.dsl-w.verizon.net) |
| 22:39.09 | husimon | [TK]D-Fender, yeah that's what i've done so far. |
| 22:39.41 | [TK]D-Fender | husimon, user, auth user, pass, server & dialplan. Thats about all you need. |
| 22:40.11 | husimon | [TK]D-Fender, i noticed they have three way calling but in the advanced settings it allows for a conference bridge of up to 10. You ever try that many? |
| 22:40.54 | [TK]D-Fender | husimon, I don't think thats managed on the unit, it depends on the server for the work. |
| 22:40.58 | [TK]D-Fender | husimon, AKA useless |
| 22:41.01 | husimon | ahh |
| 22:41.06 | husimon | hence the conference bridge url |
| 22:41.13 | [TK]D-Fender | indeed |
| 22:46.17 | Yourname`` | Hi. A call comes in to extension 54321, goes to a queue, agent 30 takes it, it has all the callerID attributes of the caller. Then, when the call is transferred to another extension, like 9, all the callerID attributes are gone. Instead it shows Agent 30's callerID, for example "Agent" <30>. How can I make it so that the callerID attribs are not lost? I have a Set(CALLERID(num)) upon the call entering extension 9.. but it doesn't do t |
| 22:47.07 | *** join/#asterisk xp_prg (n=wgheath@69.12.128.128) |
| 22:47.26 | xp_prg | anyone know of a software based call center solution I can kind of rent for quick setup? |
| 22:47.43 | *** join/#asterisk pkunkra (n=chris@cpe-74-65-226-183.nyc.res.rr.com) |
| 22:49.07 | lmadsen | Yourname``: you need to pass the callerID information through using the 'o' option to Dial() |
| 22:51.25 | Yourname`` | lmadsen: But there isn't a Dial() in there though.. |
| 22:51.46 | lmadsen | ummm.... when you transfer to another extention... there needs to be a Dial() somewhere.... |
| 22:52.32 | lmadsen | call leg 1 --> asterisk --> dial --> agent. agent transfers --> asterisk --> dial --> end point |
| 22:52.59 | Yourname`` | Ok, one sec.. lemme type this out. |
| 22:53.27 | lmadsen | pastebin is better |
| 22:53.37 | lmadsen | if you're going to be typing a bunch of info :) |
| 22:58.06 | *** join/#asterisk teknoprep (n=Chris@unaffiliated/teknoprep) |
| 22:59.11 | Yourname`` | http://pastebin.ca/901800 |
| 23:00.28 | tzafrir | What's supposed to be special about http://www.youtube.com/watch?v=oMFa2XWjUOA ? |
| 23:00.50 | Yourname`` | lmadsen: They just get transferred without the dial.. :S |
| 23:01.06 | tzafrir | (the one linked from voip-info.org: GreenfieldTech presents a proof-of-concept for the worlds fastest Asterisk based dialer) |
| 23:01.59 | lmadsen | Yourname``: uhhh... you're doing the Set() part wrong... you're just setting the callerID to the same value it already was. When the first call comes in, you need to set the callerID to a variable and make it transitive, then set the value of that to the callerID |
| 23:02.15 | lmadsen | Set(_ORIGINAL_CID=${CALLERID(num)}) |
| 23:02.19 | tzafrir | I just replicated something similar by piping 30 "originate" commands to asterisk.ctl |
| 23:02.29 | lmadsen | then when you do the 54321, you do Set(CALLERID(num)=${ORIGINAL_CID}) |
| 23:02.31 | tzafrir | (on a junk PII 350) |
| 23:02.51 | tzafrir | why is it supposed to be difficult? |
| 23:03.18 | puzzled | tzafrir: does zaptel 1.4.8 with your oslec patch compile for you? I get zaptel-base.c:3741: error: implicit declaration of function 'oslec_echo_can_identify |
| 23:03.48 | tzafrir | puzzled, current Debian package builds for me |
| 23:03.55 | *** join/#asterisk msolomos (n=ms@ppp55-201.adsl.forthnet.gr) |
| 23:04.02 | puzzled | tzafrir: and 2 errors about readv and writev |
| 23:04.05 | tzafrir | I also slightly changed the patch |
| 23:04.24 | msolomos | hi |
| 23:04.49 | tzafrir | puzzled, what version do you use? |
| 23:04.50 | Yourname`` | lmadsen: I got the part where I need to have another variable, but you lost me after that. |
| 23:04.52 | Yourname`` | :S |
| 23:05.05 | *** join/#asterisk angom (n=angom@200.56.104.87.dsl.dyn.telnor.net) |
| 23:05.21 | msolomos | any questions? |
| 23:05.50 | lmadsen | Yourname``: when the call originally comes in... set the CID to that variable. Make it transitive by prefixing the variabel name with and underscore, Then that variable will be available on the other channel you create when you do the transfer, and you can use that variable to set the CallerID before you send it to the Queue() |
| 23:06.11 | *** join/#asterisk rangerelf (n=gustavo@CableLink222-147.telefonia.InterCable.net) |
| 23:07.01 | puzzled | tzafrir: zaptel 1.4.8 and the 1.4.8 patch in oslec svn |
| 23:07.41 | rangerelf | Hello all, good afternoon. There's something I want to do, but I haven't been able to find how to do it on all the googlings I've done. |
| 23:08.48 | _ShrikE | rangerelf: Just go ahead and ask your question. |
| 23:09.19 | puzzled | tzafrir: svn rev 1333 |
| 23:09.33 | *** join/#asterisk draygon2 (n=lokbo@cpe-76-185-106-151.tx.res.rr.com) |
| 23:09.37 | Yourname`` | lmadsen: So, the first time the call comes in -> Set(_ORIGINAL_CID=${CALLERID(num)}) then before it goes to the Queue(), I Set(${CALLERID(num)}=${_ORIGINAL_CID}) .. and that'll take care of it before going to extension 9, correcto? |
| 23:09.38 | rangerelf | I want all extensions which go through the default outgoing route, to go through a queue first, instead of directly trying for one (of my two) zap channels. They're congested most of the time, so a little music until the call goes through is nice. |
| 23:09.51 | draygon2 | Hi is anyone willing to install a simple PBX for pay"? |
| 23:09.54 | draygon2 | Please PM me. |
| 23:09.55 | draygon2 | thank you. |
| 23:12.08 | jameswf | I need to train someone around here to make rpms or script the process bah |
| 23:13.35 | lmadsen | Yourname``: you don't need the underscore in the 2nd Set() -- look at my first example |
| 23:14.07 | *** join/#asterisk teknoprep (n=Chris@unaffiliated/teknoprep) |
| 23:14.08 | tzafrir | puzzled, that function is included in the oslec_wrapper.c in the deb |
| 23:14.18 | Yourname`` | lmadsen: Sorry, didn't know what a transitive variable was. |
| 23:14.31 | puzzled | tzafrir: thanks. lemme check here |
| 23:14.33 | lmadsen | check out the book... I think we write about it... |
| 23:14.39 | tzafrir | But if you don't want to change oslec, just add it (echo_can_identify) as an inline |
| 23:14.40 | lmadsen | Yourname``: no problem -- they are super handy |
| 23:14.50 | tzafrir | static inline |
| 23:15.17 | jameswf | damn spam |
| 23:15.21 | puzzled | tzafrir: that's way over my head. my C skills suck |
| 23:15.31 | Yourname`` | I'll check out the 2nd ed of the book then! :) |
| 23:15.47 | lmadsen | :) |
| 23:16.16 | rangerelf | Any takers? None? :-( |
| 23:16.54 | lmadsen | rangerelf: figure out how to use Local channels |
| 23:17.07 | lmadsen | then use a Local channel to control the outgoing call from the queue |
| 23:17.17 | lmadsen | I've thought of the same thing, but not implemented it |
| 23:17.31 | lmadsen | it's not going to be trivial, but it's totally possible |
| 23:17.57 | rangerelf | lmadsen: I suppose it's possible, just didn't have any idea how to implement it. Local channels, then. |
| 23:18.09 | lmadsen | yeppers. Use Local channels and static members in the queue.conf file |
| 23:18.22 | lmadsen | that'lll give you a start anyways |
| 23:18.31 | rangerelf | lmadsen: ok, thanks :-) |
| 23:18.48 | lmadsen | feel free to ask questions when you get stuck, but I don't hand hold for free :) |
| 23:19.08 | jameswf | lmadsen: will you hold my hand :) |
| 23:19.19 | lmadsen | jameswf: only if you promise..... nevermind |
| 23:19.39 | jameswf | sorry |
| 23:20.58 | lmadsen | hahaha |
| 23:21.51 | jameswf | if you find lmadsen helpfull please buy ten coppies of ~book so he can buy a double cheeseburger |
| 23:22.05 | lmadsen | yes... I'm hungry |
| 23:22.41 | jameswf | I got a weight problem... cant wait to eat |
| 23:23.25 | rangerelf | lmadsen: what's your book titled? |
| 23:23.43 | jameswf | jbot: tell rangerelf about book |
| 23:25.24 | teknoprep | HEY all ... i am having a problem with a polycom ip650 phone i just bought... it randomly reboots |
| 23:25.28 | teknoprep | quite often actually |
| 23:25.37 | teknoprep | any ideas ? |
| 23:29.05 | rangerelf | OK cool :-) Will do! bye! |
| 23:30.05 | *** part/#asterisk Knorrie (i=knorrie@kantoor.mendix.nl) |
| 23:30.27 | jameswf | after adding new flags to your spec file remember to set them apropriately |
| 23:31.58 | *** join/#asterisk adorah (n=Michael@87.69.130.248) |
| 23:33.02 | *** join/#asterisk mindCrime (n=chatzill@cpe-065-190-188-124.nc.res.rr.com) |
| 23:34.36 | drmessano | HAAAAAAAAA |
| 23:34.47 | drmessano | 2008 WAS going to be the year of Click To Call |
| 23:34.49 | drmessano | http://blogs.zdnet.com/ip-telephony/?p=3216 |
| 23:34.52 | drmessano | Bye Bye |
| 23:34.57 | *** join/#asterisk [T]ank (n=ckwall@c-71-195-199-107.hsd1.ut.comcast.net) |
| 23:36.12 | *** join/#asterisk hi365_m (n=hi365@213.151.62.64) |
| 23:37.18 | *** join/#asterisk adeel (n=adeel@c-24-7-132-23.hsd1.ca.comcast.net) |
| 23:37.25 | lmadsen | patents are dumb |
| 23:37.40 | lmadsen | some make sense.. but stuff like that shows just how naive the patent people are about technology |
| 23:37.54 | drmessano | yep |
| 23:38.10 | SwK | oh oh oh ... I know.... |
| 23:38.41 | SwK | lets just take all the old phone patents and insert 'over the internet' in them |
| 23:38.42 | adorah | appalling |
| 23:38.52 | SwK | and then lets take all the old web and email patents and s/email/voice over ip/g |
| 23:39.07 | SwK | s/world wide web/voice over ip/g |
| 23:39.09 | lmadsen | SwK: you're a GENIUS! |
| 23:39.10 | adeel | on my * 1.2 install, i'm constantly getting these warnings, Feb 12 15:35:26 WARNING[1377] channel.c: Avoided initial deadlock for '0x8364050', 9 retries! , and now recently, i've had calls randomly drop on sip & zap |
| 23:39.35 | drmessano | If you think about it, its a bit surprising that someone hasnt come forward before now |
| 23:39.46 | drmessano | Nextel got everyone with Push To Talk over Cellular |
| 23:40.22 | SwK | adeel, i see that all the time even with no calls up |
| 23:40.37 | bkw_ | haha |
| 23:40.39 | adeel | SwK, so it shouldn't have anything to do with my calls being dropped? |
| 23:40.48 | SwK | it could |
| 23:41.17 | SwK | dunno the reasons calls can get dropped are wide and varied from the far end dropping them to asterisk screwing up |
| 23:41.40 | SwK | that message tho typically doesnt end in a dropped call |
| 23:41.43 | adeel | well, the calls on zap dropping indicate to me that it's a local * problem |
| 23:42.14 | SwK | adeel, i'm not saying its not... i'm saying the reasons are wide and varied... |
| 23:42.49 | SwK | whats up lmadsen |
| 23:42.56 | [T]ank | so... i thought i installed the g729 codec correctly, but i run the command "show g729" from the cli and it says that is not a valid command. those were the instructions from the digium site. could anyone assist me with this? |
| 23:43.05 | [T]ank | i ran the ./register command successfully. |
| 23:43.23 | SwK | [T]ank, did you restart asterisk or manually load the codec module afterwards? |
| 23:43.27 | adeel | yeah, i know....the only things that i changed last night was changing the echo cancellor, and resetting the echo co-efficients using fxotune |
| 23:43.32 | Qwell | [T]ank: please call Digium support. |
| 23:43.33 | [T]ank | i did restart asterisk |
| 23:43.33 | lmadsen | SwK: nada much mang... j00? |
| 23:43.49 | SwK | lmadsen, working too much |
| 23:43.53 | lmadsen | SwK: I hear that |
| 23:43.55 | SwK | gots to get paid t0 |
| 23:43.57 | SwK | y0 |
| 23:44.54 | lmadsen | indubitably |
| 23:45.11 | *** join/#asterisk PepOSX (n=angeldav@190.79.252.29) |
| 23:49.11 | adeel | is there a way to ensure that VAD isn't happening? |
| 23:50.43 | jameswf | ~idnms |
| 23:50.44 | jbot | Why would a Wookiee, an eight-foot tall Wookiee, want to live on Endor, with a bunch of two-foot tall Ewoks? That does not make sense! But more important, you have to ask yourself: What does this have to do with this case? Nothing. |
| 23:51.37 | drmessano | Chewbacca defense FTW |
| 23:52.04 | *** join/#asterisk troy- (n=troy@prospero.tauri.ca) |
| 23:52.12 | troy- | where can i get up 2 date firmware for my cisco 7941? |
| 23:52.33 | drmessano | Cisco |
| 23:52.59 | *** join/#asterisk awannabe (n=brad@ip24-251-152-67.ph.ph.cox.net) |
| 23:53.25 | awannabe | hi guys, anyone have experience with adtran total access 750 channel banks using them as a channel bank with *? |
| 23:53.56 | troy- | drmessano where specifically on the site? |
| 23:55.28 | *** join/#asterisk Docfxit (n=none@cpe-72-129-118-135.socal.res.rr.com) |
| 23:55.49 | drmessano | You need to sign in with valid Cisco support options for your hardware and you will have access to the firmware |
| 23:56.00 | Qwell | drmessano: within 6-8 weeks |
| 23:56.06 | drmessano | yes lol |
| 23:56.21 | drmessano | Welcome to Cisco |
| 23:56.32 | drmessano | Cisco "We made a switch complicated" |
| 23:56.51 | awannabe | and thank goodness they did :) |
| 23:57.03 | drmessano | oh yes |
| 23:57.20 | troy- | are there any known sound quality issues with the 8-2-0-55 firmware? |
| 23:57.47 | jameswf | how can i sun cisco on windoze wtf where is the gui ahhhh sweet sweet gui |
| 23:57.54 | jameswf | *run |
| 23:58.20 | drmessano | Cisco are the kind of assholes to market a 4 port managed switch |
| 23:58.28 | Qwell | 4? 1. |
| 23:58.31 | jameswf | Cisco Its like a linksys only much much more expensive |
| 23:58.32 | drmessano | LOL |
| 23:58.50 | awannabe | your sayng cisco has no more features then linksys? |
| 23:59.08 | drmessano | Linksys 4 port: $39, managed Cisco 4 port, $289.. PoE option $599 |
| 23:59.13 | Qwell | awannabe: with or without openwrt? |
| 23:59.17 | drmessano | Who puts PoE on a 4 port switch? Cisco |
| 23:59.22 | Qwell | with, no, they don't have any more features |
| 23:59.27 | awannabe | lol, sorry, but Cisco does't even compete to linksys |
| 23:59.35 | drmessano | Uhhh |
| 23:59.43 | drmessano | Yeah, Linksys stuff has so many more features |
| 23:59.49 | awannabe | you saying openwrt is now a layer3 switch? |
