| 00:04.22 | *** join/#asterisk harryv (n=harry@0x55508034.adsl.cybercity.dk) |
| 00:04.55 | harryv | hello. i'm using zoiper to connect to my localhost asterisk. it used to work fine, but now zoiper tells me "Rejected registration for 'harry@localhost' with cause 'interworking'" when i try to register.. |
| 00:11.36 | *** join/#asterisk Darthclue (n=root@li13-84.members.linode.com) |
| 00:11.51 | harryv | ? |
| 00:11.53 | Darthclue | ~Book |
| 00:11.53 | 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 |
| 00:13.34 | *** join/#asterisk MikeJ_ (n=MikeJ@d149-67-220-106.try.wideopenwest.com) |
| 00:14.34 | *** join/#asterisk RoyK (n=roy@ip-75-8-149-91.dialup.ice.no) |
| 00:16.13 | *** join/#asterisk tzafrir_laptop (n=tzafrir@89.130.6.136) |
| 00:19.30 | harryv | any idea why? |
| 00:21.04 | tzafrir_laptop | problem building libpri on latest Debian? http://bugs.debian.org/453161 |
| 00:22.15 | MikeJ_ | tzafrir_laptop: fixed in trunk |
| 00:22.25 | MikeJ_ | q931.c, include stddef.h |
| 00:22.52 | *** part/#asterisk klasstek (n=karl@ip67-154-143-190.z143-154-67.customer.algx.net) |
| 00:22.57 | troy- | is there a SIP/IAX client for blackberry? |
| 00:23.01 | tzafrir_laptop | ok. Should be fixed in 1.4 |
| 00:23.08 | tzafrir_laptop | as well... |
| 00:23.18 | Qwell | it's already fixed in 1.4 |
| 00:23.57 | troy- | Qwell: do you happen to know? |
| 00:23.59 | MikeJ_ | why thehell do people report stuff to debian for other people software.. thats dumb |
| 00:24.26 | *** join/#asterisk klasstek (n=karl@ip67-154-143-190.z143-154-67.customer.algx.net) |
| 00:25.07 | tzafrir_laptop | MikeJ_, it broke a building of a debian package, hence a serius bug for a debian package |
| 00:25.48 | tzafrir_laptop | I haven't had yet time to report it "upstream" |
| 00:26.02 | Qwell | tzafrir_laptop: there's no reason to - it's been fixed for over a month |
| 00:26.25 | tzafrir_laptop | Qwell, not in the version there |
| 00:26.42 | tzafrir_laptop | it's fixed in 1.4.2? |
| 00:27.01 | Qwell | no, it's fixed in 1.4 |
| 00:28.23 | harryv | what would 'interworking' mean? |
| 00:29.47 | dec | I'm using 1.4.4 here and it's registering to a remote SIP server. When that SIP server calls back, it's ignoring the SIP INVITE. What could cause this? |
| 00:30.08 | Qwell | dec: being on a version 10 versions behind |
| 00:30.22 | *** join/#asterisk tzafrir_laptop (n=tzafrir@89.130.6.136) [NETSPLIT VICTIM] |
| 00:30.41 | russellb | Qwell: lol .. |
| 00:30.54 | dec | Qwell: I understand your point but that's not a helpful answer. |
| 00:30.56 | Qwell | russellb: not much point in even trying :) |
| 00:31.09 | Qwell | dec: there have been *hundreds* of bugs fixed since 1.4.4 |
| 00:31.16 | Qwell | dare I even say...over a thousand? |
| 00:31.25 | russellb | Qwell: i'll check |
| 00:31.39 | Qwell | I'm sure it's a ridiculous amount |
| 00:31.49 | dec | Qwell: you're willing to guess that a basic SIP INVITE didn't work in 1.4.4 ? |
| 00:32.02 | dec | That's one hell of a bug if so :) |
| 00:32.03 | Qwell | pretty much, yeah |
| 00:32.17 | russellb | it's likely that it's because whatever destination it is calling doesn't exist in your dialplan |
| 00:32.24 | russellb | that's the most common cause for what you're seeing. |
| 00:32.37 | Qwell | well, it would throw a warning on the console if that happened, wouldn't it? |
| 00:32.45 | russellb | 1059 fixes to 1.4 since 1.4.4 :) |
| 00:33.06 | russellb | Qwell: actually, no ... people complained about that at astricon and i added it, heh |
| 00:33.11 | Qwell | ahh |
| 00:33.12 | Qwell | AND |
| 00:33.15 | Qwell | thus my point |
| 00:33.22 | Qwell | upgrade, and you'll get a warning :) |
| 00:33.30 | russellb | or a NOTICE, or maybe DEBUG |
| 00:33.32 | russellb | i don't remember |
| 00:33.36 | Qwell | See, I knew I was going somewhere with that - I just didn't know where. |
| 00:33.39 | dec | Heh :) |
| 00:34.06 | dec | russellb: so other than upgrading to the latest version, I should check that I have a SIP peer defined for the extension that's being called and also an entry point in the dialplan for that extension? |
| 00:34.09 | Qwell | dec: no, but really, with any problem - even ones that you'd say "no, they wouldn't make a release if that bug existed..." - try the latest version first |
| 00:34.26 | *** part/#asterisk klasstek (n=karl@ip67-154-143-190.z143-154-67.customer.algx.net) |
| 00:34.34 | dec | Qwell: I completely understand... but on a production system its hard to schedule downtime in the middle of the day to upgrade the phone system |
| 00:34.34 | *** part/#asterisk shtoom (n=shtoom@59.93.123.86) |
| 00:34.36 | russellb | dec: yeah |
| 00:34.48 | Qwell | dec: sure, but one should have a dev system :) |
| 00:34.56 | dec | Qwell: I wish :) |
| 00:35.11 | dec | $manager of course doesn't understand that requirement |
| 00:35.21 | Qwell | silly manager people |
| 00:35.37 | dec | until <manager> Why isn't the phone system working? <me> because we don't have a dev system to test these things |
| 00:35.50 | Qwell | dec: sounds like a perfect opportunity |
| 00:36.07 | dec | Qwell: I tried this morning and got denied again. |
| 00:36.10 | Qwell | suck |
| 00:36.33 | dec | Then again we just purchased lots more blade systems and SAN / storage gear, so I'm sure they're feeling a little broke right now. :) |
| 00:36.47 | Qwell | hell, a dev system doesn't have to be anything more than an old desktop |
| 00:37.02 | Qwell | don't need a fancy new blade system for that |
| 00:37.41 | dec | out of curiosity, when asterisk responds to a SIP INVITE, how does it work out what to use as its from address? |
| 00:40.07 | mvanbaak | I dev on an IBM thinkpad, but the real tests run on a 233Mhz OpenBSD machine with 64Meg RAM |
| 00:40.16 | mvanbaak | good enough for most stuff |
| 00:40.38 | mvanbaak | if the OBSD box actually had a VGA card I would use it to dev |
| 00:40.51 | mvanbaak | but PCI VGA cards are hard to get nowedays |
| 00:41.16 | mvanbaak | *OUCH* |
| 00:41.20 | mvanbaak | Thu Nov 29 01:41:20 CET 2007 |
| 00:41.29 | mvanbaak | 6 AM is wakeup time |
| 00:41.44 | mvanbaak | tomorrow will be a tough morning |
| 00:41.49 | *** join/#asterisk RoyK (n=roy@ip-75-8-149-91.dialup.ice.no) |
| 00:41.55 | dec | It's not so bad. |
| 00:42.05 | dec | I'm in the future, I've already experienced 'tomorrow morning' |
| 00:42.37 | mvanbaak | ;) |
| 00:42.49 | mvanbaak | anywayz |
| 00:42.55 | mvanbaak | I'm off to get some sleep |
| 00:43.00 | Darthclue | i experience tomorrow morning at 5:30 am every morning. |
| 00:43.17 | [hC] | I get up at 9-10am |
| 00:43.28 | [hC] | but i also go to bed between 1-2am every day |
| 00:43.31 | Darthclue | i haven't seen the sun in weeks. |
| 00:43.34 | mvanbaak | [hC]: you must be a student |
| 00:43.41 | [hC] | mvanbaak: close, business owner. |
| 00:44.01 | *** join/#asterisk mbit (n=nothing9@218-214-57-65.people.net.au) |
| 00:44.10 | [hC] | mvanbaak: specifically voip for business.. better known as "cant get any work done during the day" |
| 00:44.14 | mbit | hey guys does anyone know how to set the 1 mail box for 2 extensions? |
| 00:44.21 | Darthclue | i go to work before it comes up, i go home after it goes down, and i work in an office with no windows. i've forgotten what it looks like. is it pretty? |
| 00:44.21 | [hC] | i do "support" during the day and "work" from 6pm - 11pm |
| 00:44.37 | *** part/#asterisk harryv (n=harry@0x55508034.adsl.cybercity.dk) |
| 00:45.37 | Yourname`` | Hi. How can I do an attended transfer without features.conf? :S |
| 00:49.33 | Yourname`` | Nevermind. Someone breathing down my neck. |
| 00:49.53 | Yourname`` | Darthclue: Do you work with me? |
| 00:50.30 | MikeJ | its Darthclue ... long time no see |
| 00:53.45 | Darthclue | Yourname``, Not that I know of, but who knows. |
| 00:54.01 | dec | Hmmm.... a working SIP call sends back a 100 Trying response to an INVITE, and yet this broken one sends back a 407 Proxy Auth Required... |
| 00:54.38 | Darthclue | MikeJ, yeah, random phone calls from people who want help with pbx systems mean I have to actually read the book and attempt some form of osmosis so that I can figure what the hell they screwed up (and get paid to fix it!) |
| 00:55.11 | *** join/#asterisk JayTee52 (n=jforde05@c-69-137-243-25.hsd1.in.comcast.net) |
| 00:55.13 | Darthclue | gnite all, I've done my time for now. |
| 00:55.17 | JayTee52 | nite |
| 00:57.25 | *** join/#asterisk ming_zym (i=ming_zym@nat/yahoo/x-a0906f1d754afd22) |
| 00:57.52 | [TK]D-Fender | Yourname``, get a real phone |
| 00:57.57 | *** join/#asterisk RoyKa (n=roy@ip-167-7-149-91.dialup.ice.no) |
| 00:58.06 | troy- | hi, i compiled a 2.6 kernel in centos with grsec using my existing .config, when i boot the OS in vmware i get kernel panics which state they are unable to find the VolGroup00 FS |
| 00:59.08 | blitzrage | sounds like a #linux question.... ? |
| 00:59.18 | troy- | woops wrong channel :( |
| 00:59.20 | *** join/#asterisk teknoprep (n=tekno@unaffiliated/teknoprep) |
| 00:59.20 | blitzrage | :) |
| 00:59.28 | troy- | sorry, irssi does that to me sometimes |
| 00:59.36 | teknoprep | is call quality for sip and iax2 connection over inet any better in asterisk 1.4 ? |
| 00:59.39 | teknoprep | over 1.2 ? |
| 00:59.53 | blitzrage | troy-: it happens :) |
| 01:00.28 | blitzrage | teknoprep: well... perhaps because of the jitterbuffer -- also some people hear a difference in the quality of the audio. I had a customer comment that it was better one day when I upgraded to 1.4. |
| 01:00.49 | blitzrage | so my answer would be... yes w/ caveat |
| 01:01.04 | blitzrage | caveat: if you Internet connection isn't voip quality.. it's gonna still suck |
| 01:01.16 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 01:03.54 | *** join/#asterisk Victor_Yure (n=aaa@201.9.18.85) |
| 01:04.42 | barhom | for some reason the my remote xlite (at a friends house) is registring his local ip as the contact<> to asterisk, <sip:deadel@192.168.0.101...... instead of sip:deadel@HISIP, what could cause this? |
| 01:05.07 | barhom | *-the |
| 01:05.52 | JayTee52 | we're ordering a new Dell Poweredge server for Asterisk. It's going to have an Intel Quad Core cpu, 1333mhz FSB and 4GB of RAM. I can get it with no OS installed or with Red Hat EL 5 or Suse 10 as linux choices. Dell has drivers for their PERC RAID controller for both distros. I'm thinking of going with no OS and loading Fedora on it. Any advice is appreciated. |
| 01:07.26 | [TK]D-Fender | barhom, if he's behind NAT you need to put "nat=yes", "canreinvite=no", and "qualify=yes". * needs to know not to use the IP he send in, but ranter the IP its received from. |
| 01:07.36 | *** join/#asterisk runlvl (n=pc@190.2.40.57) |
| 01:07.38 | runlvl | No se olviden de visitar http://www.asterisk-la.org , Comunidad de Asterisk en espaņol :-) |
| 01:08.04 | JayTee52 | no hablo espanol |
| 01:08.12 | *** join/#asterisk harryv (n=harry@0x55508034.adsl.cybercity.dk) |
| 01:08.15 | harryv | ~book |
| 01:08.16 | 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 |
| 01:08.25 | *** part/#asterisk harryv (n=harry@0x55508034.adsl.cybercity.dk) |
| 01:08.25 | runlvl | harryv, ok ;-) |
| 01:10.56 | blitzrage | JayTee52: skip Fedora -- use RHEL5 or CentOS 5 |
| 01:11.09 | blitzrage | Fedora is too "new" for a server imho |
| 01:11.16 | blitzrage | too many things always changing |
| 01:11.47 | JayTee52 | blitzrage, yeah, that was one reason I was wavering |
| 01:12.10 | blitzrage | just go with no OS and grab a copy of CentOS 5 (unless RHEL5 license is included in your server) |
| 01:13.11 | JayTee52 | well, the PERC controller is the big factor, Dell has drivers for RHEL and Suse but it doesn't list CentOS |
| 01:15.54 | JayTee52 | RHEL 5 is available for an additional $224 preinstalled with one year support. |
| 01:16.16 | [TK]D-Fender | JayTee52, .... CentOS **IS** RHEL |
| 01:16.39 | JayTee52 | so a Dell driver for RHEL would work on CentOS? |
| 01:17.13 | [TK]D-Fender | JayTee52, yup |
| 01:17.28 | JayTee52 | I knew CentOS was based on RHEL. Like Ubuntu is based on Debian but I didn't think they were alike enough to allow using drivers from one on the other. |
| 01:17.48 | blitzrage | JayTee52: ya... it actually is the same distro, just with the RH stuff renamed to CentOS, and no support |
| 01:17.59 | blitzrage | so RHEL5 files will work no problem on CentOS 5 |
| 01:18.00 | *** join/#asterisk MikeJ_ (n=MikeJ@d149-67-220-106.try.wideopenwest.com) |
| 01:18.06 | JayTee52 | cool |
| 01:18.22 | blitzrage | [TK]D-Fender: you over emphasize too many words :) |
| 01:18.47 | [TK]D-Fender | blitzrage, WHAT ... are YOU ........ TALKING ABOUT??!? |
| 01:18.55 | Qwell | aboot |
| 01:18.55 | blitzrage | aboot :) |
| 01:18.58 | Qwell | pwnt |
| 01:18.58 | blitzrage | hahaha |
| 01:18.58 | [TK]D-Fender | :p |
| 01:19.18 | [TK]D-Fender | <- Quebecois! |
| 01:19.30 | JayTee52 | aboot? sounds Scottish! |
| 01:19.32 | Qwell | [TK]D-Fender: why do I doubt that? |
| 01:20.35 | [TK]D-Fender | Qwell, no idea, but I'm sure you'll come up with something to share before log :) |
| 01:20.40 | [TK]D-Fender | long even :) |
| 01:24.03 | *** join/#asterisk RoyK (n=roy@ip-167-7-149-91.dialup.ice.no) |
| 01:24.19 | *** join/#asterisk mog (i=mog@nat/digium/x-4d069178854978d1) |
| 01:24.19 | *** mode/#asterisk [+o mog] by ChanServ |
| 01:24.25 | Qwell | mog: !!! |
| 01:25.50 | mog | Qwell,! |
| 01:30.18 | *** join/#asterisk rene- (n=rene@12.161.84.2) |
| 01:36.35 | *** join/#asterisk |omni| (n=rob@75.148.59.9) |
| 01:45.59 | *** join/#asterisk russellb (n=russell@asterisk/developer-and-stable-maintainer/drumkilla) |
| 01:45.59 | *** mode/#asterisk [+o russellb] by ChanServ |
| 01:52.51 | *** part/#asterisk rpm (n=russell@d154-5-130-82.bchsia.telus.net) |
| 01:57.02 | *** join/#asterisk jm|laptop (n=jm|home@dilbert.jamiem.com) |
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| 02:37.13 | iCEBrkr | Dude. |
| 02:37.28 | iCEBrkr | I have a 7meg log mostly full of unreachable and lagged message to Voicepulse. |
| 02:37.56 | iCEBrkr | and about 5 pages worth of unable to register.. |
| 02:38.15 | iCEBrkr | meanwhile, my other providers don't seem to have a problem |
| 02:40.42 | blitzrage | ~book |
| 02:40.43 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf |
| 02:40.54 | iCEBrkr | blitzrage: I'm offended. |
| 02:41.04 | blitzrage | no no... I just don't have the link memorized :) |
| 02:41.10 | blitzrage | it was for me -- no you :) |
| 02:41.23 | iCEBrkr | :| how's this? |
| 02:41.34 | blitzrage | ummm... you look fairly passive |
| 02:41.38 | iCEBrkr | damn |
| 02:42.06 | Qwell | blitzrage: Will you sign my pdf? |
| 02:42.10 | blitzrage | iCEBrkr: then I might choose not to use voicepulse :) |
| 02:42.14 | blitzrage | Qwell: of course! |
| 02:43.57 | iCEBrkr | Yeah |
| 02:44.12 | iCEBrkr | I may have to find another primary |
| 02:44.25 | iCEBrkr | I've been with voicepulse for 3yrs now. |
| 02:45.35 | iCEBrkr | Are DID's portable? :P |
| 02:50.26 | JayTee52 | yes, they are |
| 02:55.33 | *** join/#asterisk jetlagmk2 (i=jetlag@pool-70-17-58-187.pskn.east.verizon.net) |
| 03:02.58 | *** join/#asterisk tobias (n=tobias@user-0ce2hr3.cable.mindspring.com) |
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| 03:15.16 | *** join/#asterisk shmaltz (n=mybox@mail2.dmaven.com) |
| 03:15.47 | shmaltz | how will function strftime handle daylight savings time? |
| 03:16.14 | shmaltz | what does epoch return? local tz or gmt? |
| 03:17.43 | *** join/#asterisk UnixDog (n=unixdog@adsl-69-234-183-43.dsl.irvnca.pacbell.net) |
| 03:22.20 | Nugget | <pedantic> neither, it is in UTC </pedantic> |
| 03:28.43 | *** join/#asterisk lesterc (n=lesterc@158.27.96.58.exetel.com.au) |
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| 03:52.29 | mmlj4 | where are user-recorded greetings stored on the filesystem? |
| 03:53.39 | shmaltz | Nugget, what is the difference between UTC and GMT? |
| 03:54.10 | shmaltz | Nugget, good one you almost got me there :P |
| 03:54.27 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 03:54.52 | Nugget | there is a difference. |
| 03:54.58 | Nugget | one has leap seconds and the other one doesn't |
| 03:55.13 | shmaltz | Nugget, what does leap seconds mean? |
| 03:55.19 | Nugget | exactly what it sounds like. |
| 03:55.42 | Nugget | 1 second in UTC isn't the same as 1 second in GMT. |
| 03:55.46 | Nugget | and over time they drift apart. |
| 03:55.58 | Nugget | so leap seconds are added to GMT so it can "catch up" to UTC |
| 03:56.21 | *** join/#asterisk zippytech (n=ron@244.zippytech.com) |
| 03:56.22 | Nugget | the difference only really matters to astronomers, but it's real and something to keep in mind for computer time syncs. |
| 03:56.27 | Nugget | NTP is strictly UTC-based. |
| 03:56.42 | Nugget | People use the term "GMT" interchangably with "UTC" but that's incorrect. |
| 03:57.00 | zippytech | hello any way to reduce echo on extention to extention calls |
| 03:57.13 | zippytech | i have very little on pots |
| 03:57.58 | shmaltz | Nugget, thanks for the lesson |
| 03:59.06 | shmaltz | ok, then if I want to take current date and time in the dialplan, how can I make sure that it's using the right tz? I'm trying to extract it from EPOCH |
| 03:59.30 | shmaltz | since we got est and dst, how can I make sure? |
| 03:59.35 | shmaltz | nugget? you still here? |
| 03:59.47 | Nugget | the strftime function takes a timezone as one of the parameters |
| 04:00.35 | shmaltz | Nugget, but AFAIK I can only enter GMT-5 and I want -4 for EDT and -5 EST |
| 04:00.39 | Nugget | for north america you'll want to use something like "EST5EDT" or "CST6CDT" to properly account for daylight saving time |
| 04:01.10 | [TK]D-Fender | zippytech, "extensions" are numbers you dial on your system. perhaps you're asking about your PHONES. Maybe you should tell us what you're using. |
| 04:01.17 | Nugget | grope through /etc/zoneinfo/ for the options you have, but if you're using a normal unix system those are almost certainly options |
| 04:01.35 | [TK]D-Fender | mmlj4, They're wherever you told the recording to go. |
| 04:01.38 | Nugget | er, possibly /usr/share/zoneinfo/ depending on flavor of unix |
| 04:02.15 | shmaltz | Nugget, so if I do: Set(MYDATE=${STRFTIME(${EPOCH},EST5EDT,%d%m%C)}) I will get 281107? |
| 04:02.36 | Nugget | looks right to me |
| 04:02.52 | mmlj4 | [TK]D-Fender: right... where might that be defined? |
| 04:02.58 | shmaltz | Nugget, thank you I will test it now, and let you know :) |
| 04:03.28 | Nugget | well, I think %Y is a better choice than %C, but that's just me. |
| 04:03.39 | [TK]D-Fender | mmlj4, isn't defined anywhere. What did you put in your Record statement? |
| 04:03.45 | Nugget | %C gets you a 20. |
| 04:04.06 | shmaltz | Nugget, I think it should give me 07 checking now |
| 04:04.21 | shmaltz | ouch you are rigth |
| 04:04.22 | Nugget | depending on what you're doing with the value, %Y%m%d might be a better choice because it sorts properly |
| 04:04.33 | Nugget | (if you're creating mixmonitor files or something) |
| 04:04.38 | shmaltz | which one gives me 07? |
| 04:04.46 | zippytech | gpx 2000 and bt 200 |
| 04:04.48 | Nugget | %y |
| 04:04.50 | zippytech | grandstream |
| 04:05.02 | shmaltz | nah, I'm using this to interact with a db that has dates stored in ddmmyy |
| 04:05.13 | Nugget | *nod* |
| 04:05.17 | [TK]D-Fender | zippytech, try different firmwares. they've had all sorts of flakey releases. Go check the WIKI for more info |
| 04:05.24 | Nugget | not a proper date field? nutty. |
| 04:05.25 | shmaltz | thanks for pointing out that C and y |
| 04:05.48 | shmaltz | Nugget, it is a proper date field but the AGI wants it like ddmmyy |
| 04:05.53 | Nugget | *nod* |
| 04:06.13 | Nugget | so it'll break in the year 2101. didn't we learn anything from y2k? :) |
| 04:07.28 | mmlj4 | hrm, i've only got one Record statement, there must be something else going on |
| 04:08.51 | zippytech | so you think the problem is in the phones not the asterisk |
| 04:08.52 | zippytech | ? |
| 04:10.10 | [TK]D-Fender | zippytech, indeed. Nothing miraculously happens to IP between your 2 phones. |
| 04:10.31 | [TK]D-Fender | zippytech, the firmwares have had all sorts of gain issues causing accoustic echo, etc. |
| 04:10.33 | [TK]D-Fender | ~gs |
| 04:10.34 | jbot | GrandSuck phones & gateways are cheap junk which should be avoided with extreme prejudice. |
| 04:10.34 | zippytech | cool thanks , thats a starting point |
| 04:10.36 | [TK]D-Fender | ^^^^^^^^^^^^^ |
| 04:10.45 | *** join/#asterisk PepOSX (n=pepOSX@190.78.220.149) |
| 04:10.48 | [TK]D-Fender | ~grandstream |
| 04:10.49 | jbot | hmm... grandstream is the Yugo of VoIP hardware. Run. Run away now. |
| 04:10.56 | zippytech | lol |
| 04:11.04 | MikeJ | thats not very nice |
| 04:11.13 | mmlj4 | no, but it's very true |
| 04:11.24 | mmlj4 | grandstream is the crap |
| 04:11.34 | zippytech | i have about 10 here and they are 2 +years old and seem to have work fine but the new bt 200 well? |
| 04:12.47 | zippytech | what do you like for phones? |
| 04:13.14 | [TK]D-Fender | Polycom > all |
| 04:15.06 | MikeJ | Polycoms are fine unless you have to configure one.. or talk in the speaker phone :P |
| 04:15.19 | MikeJ | s/in/on/ |
| 04:15.54 | mmlj4 | anyone here used aastra? |
| 04:16.02 | MikeJ | Jbot meant: I can't think for myself and need to mimic others |
| 04:16.17 | *** join/#asterisk luckyone (n=hidden@CPE-65-28-8-148.kc.res.rr.com) |
| 04:16.26 | luckyone | hello everyon |
| 04:16.40 | MikeJ | mmlj4: I think I have one in a box around here somewhere |
| 04:16.57 | luckyone | can anyone help me with asterisk on a debian system? I am trying to connect to my running asterisk process via the cli so I can debug what is wrong |
| 04:17.10 | MikeJ | luckyone: asterisk -r |
| 04:17.42 | luckyone | when I run asterisk -vvvvr I get does /var/run/asterisk/asterisk.ctl exist? |
| 04:17.53 | luckyone | and it doesn't |
| 04:17.54 | MikeJ | does it? |
| 04:18.00 | [TK]D-Fender | MikeJ, Mine are configured jsut fine, never put up the slightest fight, and Polycom is renowned for their speakerphone. |
| 04:18.04 | luckyone | so, I killed the process |
| 04:18.19 | luckyone | made the /var/run/asterisk dir |
| 04:18.25 | luckyone | then started * back up |
| 04:18.25 | MikeJ | [TK]D-Fender: they may be renowned.. but they still suck |
| 04:18.33 | luckyone | still know ctl file |
| 04:18.47 | shmaltz | nugget thanks EST5EDT does the job |
| 04:18.47 | MikeJ | I have a pingtel phone.. thats the best speakerphone I have ever used.. |
| 04:18.49 | [TK]D-Fender | mmlj4, Aastra is "OK" but have their fair share of quirks |
| 04:18.57 | MikeJ | then again. they don't make them anymore... |
| 04:19.00 | Nugget | yay |
| 04:19.05 | [TK]D-Fender | MikeJ, You are almost alone in that opinion, but you're welcome to it |
| 04:19.24 | mmlj4 | i loathe the feel of polycom, but aastra to me is very nice on the eyes and to the touch |
| 04:19.27 | MikeJ | I say today 2 feet from the thing. and the other end had me repeatedly cutting out.. |
| 04:20.04 | [TK]D-Fender | mmlj4, Touch fails on the 5i series, handset has no wieght, rubbery buttons, very low angle of readbility of the screen. |
| 04:20.19 | mmlj4 | hrm |
| 04:20.48 | [TK]D-Fender | mmlj4, and the handset cradles so lightly tha the slightest thing knocks it off-hook |
| 04:20.58 | luckyone | any ideas on how to connect to my * process? |
| 04:21.03 | [TK]D-Fender | mmlj4, the 480i was a better model physically. |
| 04:21.09 | [TK]D-Fender | luckyone, "asterisk -r |
| 04:21.13 | luckyone | I need to figure out why I can't make calls :( |
| 04:21.22 | luckyone | [TK]D-Fender: no dice |
| 04:21.29 | [TK]D-Fender | luckyone, probably ... because asterisk isn't RUNNING <------ |
| 04:21.51 | luckyone | ps -ef | grep asterisk says otherwise |
| 04:21.53 | [TK]D-Fender | luckyone, which sorta explains everything in one shot now doesn't it? |
| 04:21.54 | MikeJ | luckyone: you running as root? |
| 04:22.14 | MikeJ | and what permissions does /var/run/asterisk have? |
| 04:22.17 | [TK]D-Fender | ^^^^^ |
| 04:22.47 | luckyone | running as root |
| 04:23.14 | luckyone | perms on /var/run/asterisk are drwxr-xr-x |
| 04:23.42 | [TK]D-Fender | luckyone, just because you see the process doesn't mean its running properly. kill everything related to it and start it manually to test. "asterisk -gvvvvc" |
| 04:23.44 | MikeJ | when you start asterisk... do you get any errors at all? |
| 04:24.04 | shmaltz | I like this: |
| 04:24.06 | shmaltz | Noop(${STRFTIME($[${EPOCH}+86400]|EST5EDT|%d%m%y)}) |
| 04:24.07 | shmaltz | This shows tomorrows date |
| 04:24.17 | luckyone | [TK]D-Fender: will start manually now |
| 04:24.21 | shmaltz | so the caller can select 1 for today and 2 for tomorrow |
| 04:24.23 | [TK]D-Fender | mmlj4, and Record's default relative path is the sounds folder defined in asterisk.conf (typically /var/lib/asterisk/sounds) |
| 04:26.36 | mmlj4 | thanks |
| 04:27.19 | luckyone | totally worked |
| 04:27.23 | luckyone | wtf |
| 04:27.44 | luckyone | what is wrong with my /etc/init.d/asterisk start script... |
| 04:29.22 | mocker | Half way through a week long course on the Aspect Unified IP platform. |
| 04:29.30 | mocker | woo? |
| 04:31.51 | *** join/#asterisk lmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 04:31.51 | *** mode/#asterisk [+o lmadsen] by ChanServ |
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| 05:08.08 | Hadi- | hello |
| 05:09.28 | *** part/#asterisk MikeJ (n=MikeJ@d149-67-220-106.try.wideopenwest.com) |
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| 05:33.26 | Nukemizer | I am looking for the right macro location to have ACD remote agents and DISA calls pass the "calling number" out my SIP trunks to the receiving phone that is NOT in the * box as an extension.. Would (macro-outbound-callerid) the place to modify or look for a way to change this proccess ? Thank you in advance for any guideance |
| 05:35.17 | *** join/#asterisk jer_ (n=jtregunn@unaffiliated/jer) |
| 05:35.29 | *** part/#asterisk axisys (i=iqbala@outbound.silenceisdefeat.org) |
| 05:36.09 | *** join/#asterisk Corydon76-vcch (n=tilghman@pdpc/supporter/bronze/Corydon76-home) |
| 05:36.09 | *** mode/#asterisk [+o Corydon76-vcch] by ChanServ |
| 05:42.05 | De_Mon | Nukemizer I'm sorry, your trying to do what? |
| 05:43.34 | Nukemizer | outside caller will call in to ACD, a remote Agent on a cell phone will take the call and I want to pass the original Caller ID to the agents phone or at very least mask the outbound Caller ID with a DID from our list od DID's to better identify the call for the Agent |
| 05:43.58 | *** join/#asterisk FireMac (n=firemac@CPE000d88ae88b9-CM0011ae8bb0ee.cpe.net.cable.rogers.com) |
| 05:44.18 | [TK]D-Fender | Nukemizer : remote agents are tricky at best, and usually unworkable. |
| 05:44.20 | De_Mon | you've never touched an asterisk dialplan have you |
| 05:45.30 | Nukemizer | i have in the past, so perhaps I am making this harder than it needs to be ? |
| 05:45.52 | *** join/#asterisk ming_zym (i=ming_zym@nat/yahoo/x-642270ae42988c8b) |
| 05:46.30 | De_Mon | setting the callerID on an outbound call to a sip client is easy. Having identifiable information on the incoming call you can use to set that callerid, is what you have to find |
| 05:46.37 | Nukemizer | i am by no means an expert, for sure a novice, but willing and able to experiment and explore. Just lost at the moment |
| 05:47.02 | [TK]D-Fender | Nukemizer, if you don't know where in the dialplan to do this. you're already lost, because thats the EASY part. |
| 05:47.14 | Nukemizer | De_Mon, this would be sending a call to a Cell phone not a sip phone |
| 05:47.38 | [TK]D-Fender | Nukemizer, Next when you call to the PST you'll have an issue of you send to a phone that for any reason falls to voicemail. These acts will answer the call and cut off distribution. |
| 05:47.42 | De_Mon | so your trying to change the callerid of your own number/trunk |
| 05:47.43 | Nukemizer | the sip part is easy as you indicated and I do get that part |
| 05:48.15 | [TK]D-Fender | Nukemizer, Cell phones are ESPECIALLY bad at this. Imagine if that phone goes out of range. your entire queue will end up in his VM box |
| 05:48.21 | De_Mon | going to a pstn -- cell |
| 05:48.32 | [TK]D-Fender | Nukemizer, So stop worrying about CID, you have REAL PROBLEMS to deal with |
| 05:48.36 | De_Mon | is it even legal to spoof the cid |
| 05:48.44 | [TK]D-Fender | De_Mon, yeah, thats fine) |
| 05:48.44 | Nukemizer | Correct I want to alter the outbound Caller ID for the ACD remote agent notify call only |
| 05:49.29 | De_Mon | [TK]D-Fender he just has to use the same technique I am to get the remote agent into the queue AFTER they acknowledge a real person picked up |
| 05:49.30 | [TK]D-Fender | Nukemizer, Foreg the CID already, thats the EASY part. Next you have to make sure your provider THEMSLEVES to not count your call as answered the moment they accept it. Several DO. |
| 05:49.34 | Nukemizer | hehe, yeah i understand what you mean about the coverage of cell service, but this could be a lind line ACD agent too. |
| 05:49.37 | De_Mon | which, by the way has run into some problems ;) |
| 05:50.02 | [TK]D-Fender | De_Mon, yeah, your way is a perverse approach :) |
| 05:50.48 | [TK]D-Fender | Nukemizer, same problem. Imagine they're on a call, AGAIN every new caller gets dumped to VM. |
| 05:51.02 | [TK]D-Fender | Nukemizer, no control over call waiting, etc. |
| 05:51.10 | [TK]D-Fender | Nukemizer, remote agents = mistake. |
| 05:51.48 | [TK]D-Fender | Nukemizer, better approach would be to use SMS as an "alerter" to tell them they should call into * and log into the queue |
| 05:52.41 | Nukemizer | for my current situationj going to VM is acceptable if the caller ended up in VM .. |
| 05:53.49 | De_Mon | how do I send sms thru asterisk? |
| 05:54.21 | Nukemizer | just wanted to pass the Caller ID our to the agent is all, I know at the telco there are limits and spoffing controlled for just being sent to the ACD Agents, but it sound like Asterisk can not be configured this way from what I am hearing ? |
| 05:55.40 | [TK]D-Fender | Nukemizer, Final time, CID = EASY. |
| 05:56.00 | [TK]D-Fender | Nukemizer, 1 friggen line. |
| 05:56.16 | Nukemizer | If the phone number is a DID that belongs to my DID range the CLID will be past along so I gathered there may be a macro that can mangle the number to be the REALCALLERID |
| 05:56.19 | lmadsen | one!!11 |
| 05:56.19 | [TK]D-Fender | Nukemizer, Worry about the HUGE problem that is CALL PROGRESS |
| 05:56.57 | [TK]D-Fender | Nukemizer, No, there are no magic "macros" and you clearly DON'T understand the dialplan at all. |
| 05:57.04 | Nukemizer | D-Fender, I hear what you are saying, so perhaps you could let me have a hint as to where I might modify or add my line in to what macro for this task ? |
| 05:57.15 | Nukemizer | that is where I am stuc i beleive :) |
| 05:57.16 | lmadsen | you create the macro |
| 05:57.28 | [TK]D-Fender | Nukemizer, Set(CALLERID(num)=8006661234) ; Yippy-kay-yay! |
| 05:57.32 | lmadsen | the dialplan is logic you create to control call flow |
| 05:58.33 | Nukemizer | where does ACD look for this macro ? it this hard coded as to wehre ACD will look for "outdial" |
| 05:59.24 | [TK]D-Fender | Nukemizer, And BTW... FreePBX is NOT supported here and if you think you're going to hack THAT into doing what you want.... |
| 05:59.27 | [TK]D-Fender | ~wglwat |
| 05:59.28 | jbot | wglwat is probably well, good luck with all that |
| 05:59.29 | [TK]D-Fender | ~freepbx |
| 05:59.30 | jbot | [~freepbx] FreePBX is unable to be supported here. It is made up of complex dialplans and scripts which can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking there |
| 06:00.05 | lmadsen | queues.conf controls the members it dials from the queue configuration. They could either be direct connections to an end point via SIP,IAX2, etc... or it could be a loopback into the dialplan via the Local channel which then performs some logic in a macro (or my preference GoSub), that then decides what end point to dial |
| 06:00.28 | Nukemizer | yikes ! did not mean to tip the cart here.. just asking a question. Thank you to all who have helped. |
| 06:00.43 | [TK]D-Fender | Nukemizer, ACD does not look for "a macro". You do not seem to understand the first thing about Asterisk at all and All of your questions are tied to complex junk the EXPERIENCED people don't want to touch. You are in so far over your head you'd better find a scuba tank, FAST. |
| 06:01.34 | lmadsen | Nukemizer: in order to get familiar with how some of the parts in Asterisk work, you may want to read some documentation. It'll help you to start sorting the peices together in your head |
| 06:01.46 | lmadsen | [TK]D-Fender: lay off the caffeine... |
| 06:01.58 | [TK]D-Fender | lmadsen, yeah, but no prayer of integrating it with FreePBX in theo foreseeable future... |
| 06:02.04 | Nukemizer | Well i do have years of backround in PBX work so the concepts do not escape me, it is just the logic of asterisk which I see will come in time. I do thank you for your patience with my questions however |
| 06:02.42 | lmadsen | Nukemizer: if you're using freepbx and need to do custom work -- just stop now. freepbx can do what it does -- beyond that, you need to work with vanilla asterisk |
| 06:03.24 | [TK]D-Fender | yup |
| 06:03.41 | Nukemizer | Understood, will explore and test more |
| 06:04.25 | lmadsen | Nukemizer: sounds good -- good luck! |
| 06:05.01 | Nukemizer | thank you again to all of you who had the patience to help me :) |
| 06:05.32 | linagee | if i were to hook up a regular POTS phone line to a sound card, would i get buzzing + barely audio on the recording? |
| 06:05.44 | linagee | (if there was a phone call in progress) |
| 06:05.57 | [TK]D-Fender | linagee, No, the moment your line rang your card would FRY |
| 06:06.01 | linagee | lol |
| 06:06.21 | [TK]D-Fender | linagee, Aside from the other hilarity. |
| 06:06.33 | linagee | [TK]D-Fender: assuming there was already a call in progress. |
| 06:06.59 | De_Mon | im sorry, how do you hook a pots line into a sound card? |
| 06:07.15 | linagee | very carefully. :) |
| 06:07.16 | [TK]D-Fender | linagee, Get off that crack, you've clearly got no clue how the audio is transposed on a telephone line. just no way to take that straight. |
| 06:07.30 | De_Mon | yeaaaa... |
| 06:07.38 | [TK]D-Fender | De_Mon, You grab the duct tape, I'll get the CrazyGlue (tm)! |
| 06:09.41 | [TK]D-Fender | Alrighty, enough insanity for one night. |
| 06:09.47 | [TK]D-Fender | check-out time... |
| 06:09.47 | linagee | [TK]D-Fender: gnite |
| 06:09.59 | [TK]D-Fender | 'nite |
| 06:10.11 | De_Mon | doesnt look like sms over sip is very practical |
| 06:10.23 | linagee | De_Mon: sms? |
| 06:10.36 | linagee | De_Mon: what's that? something you have to pay providers mondo dollars for? |
| 06:10.38 | linagee | :) |
| 06:11.10 | linagee | De_Mon: good luck getting it to where cell phone users can send text messages to you. to even interface to that network is $$$ |
| 06:11.16 | De_Mon | I'm not really sure how it works... since I can't really specify an outbount desitnation |
| 06:11.30 | De_Mon | looks like an ISDN line is required into * |
| 06:13.08 | De_Mon | yeeeah I'll stick to email |
| 06:13.34 | *** join/#asterisk brut- (n=brut@66.173.4.254) |
| 06:13.44 | De_Mon | off to bed I go, hi-ho! hi-ho! |
| 06:18.35 | *** join/#asterisk ManxPower (n=manxpowe@149.sub-70-221-219.myvzw.com) |
| 06:24.15 | *** join/#asterisk Maliuta (n=nikolai@c210-49-69-241.rochd2.qld.optusnet.com.au) |
| 06:28.21 | *** join/#asterisk rati (n=rati@124.125.255.24) |
| 06:37.40 | *** join/#asterisk Chris-NB (n=chris@ng1.kurtkrenn.com) |
| 06:37.44 | Chris-NB | hi |
| 06:37.52 | *** join/#asterisk harpal (n=Harpal@124.125.255.24) |
| 06:37.53 | Chris-NB | anyone using asterisk 1.4 with queues/agents? |
| 06:38.30 | Chris-NB | I got the strange behavior that If I call a Agent, there is no audio on the line. If I call the SIP Peer everything is fine |
| 06:38.39 | Chris-NB | someone had this before? |
| 06:41.12 | *** join/#asterisk jarod14 (n=jarod14@ns2.viatelecom.com) |
| 06:55.37 | Chris-NB | no one had the problem that there was no audio on a Local channel, created from a Queue to an agent? |
| 06:56.16 | *** join/#asterisk docelmo (n=docelmo@c-76-99-153-242.hsd1.de.comcast.net) |
| 06:56.51 | docelmo | I know this isnt #openser but I have a small question and I was wondering if someone knowledgable would be able to answer it. |
| 07:03.35 | *** join/#asterisk dominic1 (n=dob@213.221.82.242) |
| 07:04.43 | *** join/#asterisk ZX81 (n=matt@121.90.246.191) |
| 07:05.47 | ZX81 | hey anyone know if latest 1.4 has the agi thing fixed? I.E. if I update will I still be able to run DeadAGI and let it do both my routing and billing? |
| 07:06.11 | ZX81 | or can you still not run AGI on dead channel and DeadAGI on live channel? |
| 07:06.38 | Corydon76-dig | ZX81: the change has only been made in trunk |
| 07:07.01 | Corydon76-dig | and DeadAGI will go away |
| 07:07.18 | *** part/#asterisk dominic1 (n=dob@213.221.82.242) |
| 07:07.23 | ZX81 | so it will work in 1.4 or not work in 1.4? |
| 07:07.31 | Corydon76-dig | will not |
| 07:07.37 | ZX81 | ok damn |
| 07:07.43 | ZX81 | still can't update and am getting cores |
| 07:08.09 | ZX81 | in iax_poke_peer |
| 07:08.18 | ZX81 | i.e. http://bugs.digium.com/view.php?id=10551 |
| 07:08.20 | Corydon76-dig | In fact, in 1.6, AGIs will be able to cross the barrier from live channel to dead channel |
| 07:08.32 | ZX81 | like 1.4.0 did :) |
| 07:08.50 | Corydon76-dig | ZX81: No, AGIs could never cross the barrier |
| 07:09.09 | ZX81 | we've been running SVN-branch-1.4-r81367M for ages |
| 07:09.13 | ZX81 | and they just do deadagi |
| 07:09.30 | ZX81 | and when the call finishes they deduct credit based on the first part of the agi |
| 07:09.34 | Corydon76-dig | Once you got the HUP signal, you could never interact with the channel anymore. You had to start a new DeadAGI script |
| 07:09.47 | ZX81 | oh well works here :) |
| 07:09.52 | *** join/#asterisk GoldFingaZ (n=wt5@bas13-ottawa23-1177809155.dsl.bell.ca) |
| 07:10.08 | ZX81 | we were getting dialstatus and answered time |
| 07:10.18 | Corydon76-dig | ZX81: yeah, but running deadagi on a live channel is not supported |
| 07:10.33 | ZX81 | anyway, will write out to vars and hope they make it through to h exten |
| 07:10.58 | ZX81 | thanks for the clarification |
| 07:11.19 | Corydon76-dig | Never was supported, and when we hear from people reporting bugs using DeadAGI on a live channel, we tell them this |
| 07:11.36 | Corydon76-dig | It's not supposed to work that way |
| 07:11.43 | ZX81 | sweet :) |
| 07:12.12 | docelmo | How do you ensure the script ran completely with AGI if AGI will stop the script if the channel hangs up |
| 07:12.14 | Corydon76-dig | Hence why we made it transition cleanly in trunk |
| 07:12.28 | Corydon76-dig | docelmo: AGI doesn't stop the script |
| 07:12.45 | Corydon76-dig | It merely sends a HUP signal, which the script is free to intercept |
| 07:13.17 | Corydon76-dig | but the signal also indicates that AGI can no longer interact with the channel. It must clean up its resources and exit |
| 07:16.38 | ZX81 | night |
| 07:18.58 | docelmo | anyone here know openser? Im having a small issue with BYE messages being passed back to the orignator from the PSTN. UA -> OpenSER -> Asterisk -> PSTN the BYE doesnt make it past OpenSER |
| 07:20.22 | awk | try #openser |
| 07:21.54 | docelmo | did.. everyone dead.. |
| 07:23.20 | *** join/#asterisk xxoxx (n=xxoxx@tor/regular/xxoxx) |
| 07:26.06 | *** join/#asterisk rasix (n=rasix@ip70-110.cbn.net.id) |
| 07:26.22 | *** join/#asterisk jarod14 (n=jarod14@ns2.viatelecom.com) |
| 07:32.03 | *** join/#asterisk reber (n=reber@LNeuilly-152-21-126-73.w193-253.abo.wanadoo.fr) |
| 07:36.56 | *** join/#asterisk syneus (n=syneus@syneus.aemcom.net) |
| 07:38.34 | *** join/#asterisk PC_Clone (n=pc_clone@c-76-16-79-154.hsd1.il.comcast.net) |
| 07:40.20 | *** join/#asterisk klasstek (n=karl@ip67-154-143-190.z143-154-67.customer.algx.net) |
| 07:41.07 | klasstek | good evening/morning |
| 07:41.45 | rasix | morning |
| 07:42.28 | klasstek | I need some ideas for a problem |
| 07:43.29 | klasstek | modem -> channelbank (adit600) -> sangoma a108dm port 4 -> Dial(Zap/g3/${EXTEN}) -> out PSTN on port 3 |
| 07:43.46 | klasstek | echocan turned off in zapata.conf |
| 07:43.57 | *** part/#asterisk agx (n=AGX@host63-216-static.34-88-b.business.telecomitalia.it) |
| 07:44.04 | klasstek | calling a phone I can hear the modem fine, but it never seems to train up |
| 07:44.23 | klasstek | when I call the remote modem |
| 07:44.58 | PC_Clone | anyone around familiar with smartjacks? |
| 07:46.30 | klasstek | PC_Clone: in what way? |
| 07:46.58 | PC_Clone | if the status leds actually mean what they look like they mean |
| 07:47.05 | PC_Clone | :) |
| 07:47.10 | PC_Clone | I have a circuit |
| 07:47.11 | klasstek | usually |
| 07:47.19 | PC_Clone | and the b8zs light is on |
| 07:47.20 | klasstek | los is loss of signal |
| 07:48.54 | PC_Clone | but it's an ami line |
| 07:49.37 | *** join/#asterisk spatulamaan (n=ggilmore@67-40-197-239.tukw.qwest.net) |
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| 08:04.19 | *** join/#asterisk Uzzi (n=andrea@host49-236-dynamic.2-87-r.retail.telecomitalia.it) |
| 08:05.23 | Uzzi | Can Asterisk work with Communication controller: Conexant HCF 56k Data/Fax/Voice/Spkp (w/Handset) Modem? |
| 08:17.39 | *** join/#asterisk Schumie (n=Steve@cmarfw01.marlow.spinvox.com) |
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| 08:30.20 | Uzzi | What is the best gui for asterisk? |
| 08:30.33 | obnauticus | None. |
| 08:31.09 | Uzzi | why? |
| 08:31.42 | Uzzi | obnauticus, Can Asterisk work with Conexant HCF 56k Data/Fax/Voice/Spkp? |
| 08:33.15 | *** join/#asterisk duki (n=duki@host-85-27-58-159.brutele.be) |
| 08:33.42 | obnauticus | %I remember asking that.. |
| 08:33.46 | obnauticus | I was told NOT. |
| 08:33.50 | obnauticus | It works NOT. |
| 08:34.09 | obnauticus | I don't know why it wouldn't work as an FXS |
| 08:37.45 | Uzzi | I've not understood |
| 08:37.49 | Uzzi | sorry |
| 08:37.56 | *** join/#asterisk BeeBuu (n=beebuu@125.95.196.59) |
| 08:39.04 | BeeBuu | <PROTECTED> |
| 08:39.17 | BeeBuu | what's that mean? |
| 08:39.34 | BeeBuu | there is the file in there! |
| 08:40.40 | kaldemar | BeeBuu: what does your playback-line look like? |
| 08:41.57 | BeeBuu | background(my_music) |
| 08:45.25 | kaldemar | that's most certainly not the line from your dialplan. but however, it should work if you have my_music.gsm in /var/lib/asterisk/sounds/. |
| 08:45.50 | obnauticus | Has anyone here tried the Chanskype failure? |
| 08:46.34 | BeeBuu | kaldemar: how can i play a wav file in *? |
| 08:46.46 | obnauticus | Playback(file.wav) ? |
| 08:47.06 | *** join/#asterisk qdk (n=qdk@85.235.253.139) |
| 08:47.10 | BeeBuu | can i? |
| 08:47.21 | BeeBuu | and how ? |
| 08:47.39 | obnauticus | core show application playback |
| 08:48.28 | kaldemar | playback is not what he is looking for if he wants to listen for an extension. |
| 08:48.43 | kaldemar | BeeBuu: what is the name of your file? |
| 08:48.47 | BeeBuu | kaldemar: can i play a wav file in *? |
| 08:49.01 | kaldemar | yes you can. |
| 08:49.11 | BeeBuu | kaldemar: how to do that? |
| 08:50.07 | kaldemar | do you have the format_wav module? |
| 08:50.19 | BeeBuu | i don't know |
| 08:50.31 | BeeBuu | how to know it? |
| 08:50.49 | kaldemar | find out. and find out what format the wav has to be and make your sound file according to it. |
| 08:51.17 | BeeBuu | kaldemar: i don't know find what |
| 08:51.23 | kaldemar | compiled and installed modules are by default in /usr/lib/asterisk/modules/ |
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| 08:51.57 | BeeBuu | format_wav.so ? |
| 08:52.11 | BeeBuu | OR format_wav_gsm.so? |
| 08:52.34 | kaldemar | 'module show like format_wav.so' in cli will give you information on the module if you have it installed and loaded |
| 08:53.14 | BeeBuu | i got format_wav.so and format_wav_gsm.so two files |
| 08:53.23 | *** join/#asterisk apollonx (n=admin@193.19.189.38.STATIC.ISP.KZ) |
| 08:53.23 | BeeBuu | it mean i can play WAV files? |
| 08:53.42 | kaldemar | yes. |
| 08:53.51 | kaldemar | if they're in the right format. |
| 08:53.52 | BeeBuu | with background(mywav.wav)ïž |
| 08:54.46 | BeeBuu | which WAV format require in *? |
| 08:57.12 | kaldemar | the book has instructions on converting to the right format. |
| 08:59.34 | BeeBuu | kaldemar: thanks,i will check that in the book... |
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| 09:10.15 | cjk | hi, should zttest give out some output even if a cable is not connected? |
| 09:10.43 | *** join/#asterisk raggi (n=ra66i@80.84.54.130) |
| 09:10.51 | raggi | can anyone assist with an oslec problem? |
| 09:11.33 | *** join/#asterisk harpal (n=Harpal@124.125.255.24) |
| 09:12.02 | raggi | specifically i'm getting: zaptel: no version for "oslec_echo_can_traintap" found: kernel tainted. |
| 09:12.13 | raggi | I think there's something missing in zapata.conf, but i'm unsure |
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| 09:15.32 | BeeBuu | kaldemar: would you tell me which chapter in the book about WAV format? |
| 09:15.40 | kaldemar | page 301. |
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| 09:18.20 | BeeBuu | you mean the usage of SOX? |
| 09:18.30 | BeeBuu | -r 8000ïž |
| 09:18.59 | kaldemar | yes. |
| 09:19.27 | xrg_ | Any digium dev, please prv. me.. |
| 09:20.54 | BeeBuu | thanks again,kaldemar |
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| 09:21.34 | *** join/#asterisk ik_5 (n=ik@85.64.203.142.dynamic.barak-online.net) |
| 09:23.16 | ik_5 | hello, is there a way to know and set the number of timeout for Dial cmd when the other side (zap or sip trunk) did not answer ? |
| 09:25.24 | kaldemar | 'core show application dial' in cli. |
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| 09:29.45 | ik_5 | kaldemar, so there is a variable for the duation of the call, but how can i set for example 30 seconds from dial started until hangup if no answer signal was given ? |
| 09:30.46 | kaldemar | it's the first mentioned parameter in the example. timeout. |
| 09:31.24 | ik_5 | kaldemar, as i understand the timeout is global call timeout.. or didn't i understand it correctly ? |
| 09:31.51 | kaldemar | you misunderstood. |
| 09:32.02 | ik_5 | thanks :) |
| 09:32.46 | kaldemar | Dial(Tech/number,10) would wait 10 seconds for an answer and then move on to the next priority. |
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| 09:36.24 | ik_5 | kaldemar, thanks |
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| 09:37.36 | ik_5 | under question regarding dail, is there a way to play the ringing tune of the pbx i'm calling to instead of the "r" option, so if there are unique sounds for example, it can be heared instead of just rining sound ? |
| 09:37.58 | ik_5 | s/under/another/ |
| 09:42.28 | kaldemar | don't use the r option, it will give false progress information for the user. |
| 09:42.54 | ik_5 | so what should i use instead ? |
| 09:43.17 | kaldemar | nothing. |
| 09:44.32 | ik_5 | but it will not give the real indication of the other side in sounds.. is there a way to give a real indication of what happens on the other side ? |
| 09:45.59 | kaldemar | that is a concern of the terminating switch. |
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| 09:51.10 | Olobola | can GET DATA return * ? |
| 09:51.39 | Olobola | I can only get 0 through 9 for some reason |
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| 09:54.18 | ik_5 | kaldemar, thanks |
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| 10:00.13 | Uzzi | with analogic line and analogic phones, what is the best economic solution? |
| 10:00.59 | Uzzi | I've only block some numeration from my analogic line |
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| 10:20.17 | *** join/#asterisk funkyhippy (n=notahipp@87.127.49.162) |
| 10:22.13 | funkyhippy | Hi I'm trying to configure a simple queue qith asterisk 1.2. I have the queue defined and joinemtpy=no with 3 members defined, if no agents are logged on i can still enter the queue is this expected behaviour? |
| 10:22.27 | *** join/#asterisk my007ms (i=ircbnc@217.139.17.150) |
| 10:22.37 | my007ms | hello |
| 10:22.51 | funkyhippy | I though joinempty=no would stop calls entering the queue if no agents we logged in |
| 10:26.16 | JayTee52 | funkyhippy, you need to set it to strict instead of no if you want to block calls into the queue. |
| 10:26.26 | my007ms | is there any extra config i need to do in asterisk to make it allow video call |
| 10:26.34 | my007ms | i use eyeBeam softphone |
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| 10:26.50 | JayTee52 | my007ms, yes there is. |
| 10:26.55 | JayTee52 | just a second |
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| 10:31.58 | JayTee52 | my007ms, you need to add videosupport=yes to the [general] section of SIP.CONF and also allow=h263, allow=h263p for the eyeBeam to work. |
| 10:32.26 | my007ms | JayTee52, thanks much for help |
| 10:32.31 | my007ms | i will try and let u know ;) |
| 10:32.34 | JayTee52 | if you're running asterisk 1.4 you can use allow=h264 |
| 10:32.58 | JayTee52 | my007ms, we're doing it at work with just a couple softphones but it works ok on 1.2 |
| 10:33.09 | JayTee52 | a little jitter here and there but mostly smooth |
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| 10:43.06 | funkyhippy | thank you JayTee52 that works brilliantly! |
| 10:43.18 | *** join/#asterisk dandre (n=daniel@was59-3-82-236-48-30.fbx.proxad.net) |
| 10:43.23 | dandre | hello, |
| 10:44.00 | *** join/#asterisk AlienPenguin (n=Miranda@213.188.207.153) |
| 10:44.34 | dandre | I am trying to use asterisk manager interface action=command to send a underlying system command (such as !cat somefile) but this doesn't work |
| 10:44.43 | dandre | is there any workaround? |
| 10:45.07 | JayTee52 | funkyhippy, great! glad to be of help |
| 10:45.20 | AlienPenguin | hi, i have an asterisk conneting over pstn line through a patton gateway and i am experiencing the following problems when a sip phone calls a pstn number and on the other side there is an IVR sometimes the dtmf are correctly recognized, some times not (not the same IVR though) any suggestions on what i might check? |
| 10:51.30 | *** part/#asterisk raggi (n=ra66i@80.84.54.130) |
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| 10:52.56 | JayTee52 | AlienPenguin, try putting relaxdtmf=yes in the [general] section of your SIP.CONF file and then reload |
| 10:53.20 | AlienPenguin | JayTee52: : ok will try, thanks |
| 10:54.37 | JayTee52 | AlienPenguin, however if you get consistent results ok with one IVR system but another outside IVR system fails to detect it could be the PSTN carrier's equipment or the outside IVR that is at fault. |
| 10:55.15 | JayTee52 | does it happen from any SIP phone or is it isolated to one or two that fail to work with the IVR? |
| 10:55.29 | AlienPenguin | JayTee52: in fact i suspect there is something wrong with the patton equipment |
| 10:55.51 | AlienPenguin | but since i did not install or configure it i want to be sure there is nothing wrong with asterisk |
| 10:56.01 | AlienPenguin | all the sip phones fail with that ivr |
| 10:56.21 | *** join/#asterisk Somebee (n=Somebee@80.232.5.97) |
| 10:56.27 | JayTee52 | understood. I haven't used patton equipment, just Digium cards. |
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| 11:31.06 | HaMYaI | i'm using Dial command with the r option but don't get the ring back tone |
| 11:31.09 | HaMYaI | any idea? |
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| 11:37.43 | JayTee52 | HaMYaI, try using the Dial command without the r flag or verify that the call is actually being placed from the CLI with sip debug turned on. |
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| 11:54.32 | HaMYaI | JayTee5, done that |
| 11:58.02 | HaMYaI | JayTee5, here's the outcome http://pastebin.com/d2d9ce2e6 |
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| 12:47.09 | agx | Anyone from germany here that can point me to the regional settings for Linksys PAP2T for that country? |
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| 13:10.14 | dandre | I am trying to use asterisk manager interface action=command to send a underlying system command (such as !cat somefile) but this doesn't work. Is there any workaround? |
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| 13:14.12 | oceanicity | hi peeps |
| 13:14.44 | oceanicity | can i get some help troubleshooting a bad-sounding installation from someone?... |
| 13:16.07 | iratik | what the heck happened? |
| 13:16.24 | iratik | I get on my box this morning and I see like 5 DCC RECV boxes |
| 13:16.27 | iratik | what? |
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| 13:25.18 | oceanicity | anyone here ?? |
| 13:25.56 | *** join/#asterisk [TK]D-Fender (n=joe@MTRLPQ02-1177745839.sdsl.bell.ca) |
| 13:25.59 | *** join/#asterisk hwt (i=hwt@195.139.204.163) |
| 13:26.36 | hwt | hi, i have a dialplan where i want to do a Background(soundfile), but go to the next priority, instead of the digit entered by the user. how can i do that? |
| 13:27.37 | [TK]D-Fender | hwt: Thats exactly what * does. What is your next priority doing? |
| 13:27.51 | hwt | starting meetme. |
| 13:27.57 | hwt | i want to play a disclaimer before starting meetme. |
| 13:28.19 | hwt | but when a user presses any digit, i want it to start meetme() |
| 13:28.49 | hwt | right now it tries to jump to priority 2,3,4, or whatever i press on the keypad. |
| 13:28.53 | [TK]D-Fender | hwt: The use another method. |
| 13:29.16 | hwt | [TK]D-Fender: do you have any suggestions? |
| 13:29.20 | *** join/#asterisk JayTee52 (n=jforde05@207-67-84-185.static.twtelecom.net) |
| 13:29.42 | [TK]D-Fender | hwt: I suggest you take a moment and actually read the applications list... |
| 13:29.49 | *** join/#asterisk CBU[^_^]M`` (n=love@210.213.142.140) |
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| 13:33.29 | hwt | [TK]D-Fender: uhm, i can only find Playback() and Background(). |
| 13:33.35 | hwt | [TK]D-Fender: am i missing something? |
| 13:33.49 | hwt | playback() won't let you interrupt during the playback. |
| 13:33.51 | [TK]D-Fender | hwt: ControlPlayback, etc. |
| 13:33.51 | oceanicity | i'm using the gsm codec and my files sound pretty crap - what would be the best way to improve them?? |
| 13:34.01 | oceanicity | i encoded them to gsm using sox... |
| 13:34.25 | [TK]D-Fender | oceanicity: how do *'s premade recordings sound? |
| 13:34.32 | hwt | hm, that might do it. thanks. |
| 13:34.50 | oceanicity | [TK]D-Fender: fine |
| 13:35.18 | oceanicity | and i've had it call 5 phones at once and play the default menu, and it sounds fine |
| 13:35.50 | oceanicity | was trying to experiment with using slin files instead |
| 13:36.15 | oceanicity | but it complains that there are "no compatible codecs" (in messages) |
| 13:36.32 | oceanicity | even though "core show codecs" lists slin |
| 13:38.01 | [TK]D-Fender | oceanicity: Either your source recording is screwed up then, or Sox is mangling the file |
| 13:38.14 | oceanicity | ok |
| 13:40.36 | dandre | I am trying to use asterisk manager interface action=command to send a underlying system command (such as !cat somefile) but this doesn't work. Is there any workaround? |
| 13:41.06 | *** join/#asterisk mort_gib (n=mjensen@16.Red-83-36-63.staticIP.rima-tde.net) |
| 13:42.13 | [TK]D-Fender | dandre: that is for an ASTERISK CLI command, not an OS SHELL |
| 13:42.58 | dandre | ye I know but in a cli console I can issue ! shell_command |
| 13:43.12 | MrChimpy | why on earth would you want to? |
| 13:43.22 | shido6 | what are you trying to DO, dandre? |
| 13:43.23 | [TK]D-Fender | dandre: AMI runs over an unencrypted plain text readable format. that means anyone can grab a connect packet get the user & pass and WIPE YOUR DRIVE? |
| 13:43.33 | [TK]D-Fender | d^^^^^^^^^^^^^^^^^^ |
| 13:44.01 | [TK]D-Fender | dandre: Use your brain for a second here. PERMITTING what you are thinking is LUDICROUS |
| 13:44.04 | dandre | I just want to cat a file |
| 13:44.22 | [TK]D-Fender | dandre: come up with another way. |
| 13:44.28 | MrChimpy | well why not do that from the program that sends the AMI command? |
| 13:44.37 | [TK]D-Fender | dandre: * is not a remote server administration tool. |
| 13:44.38 | lirakis | dandre: it doesnt matter.. if you can send a cat command.. it means ANYONE can cat /dev/random > /dev/hda1 |
| 13:44.48 | [TK]D-Fender | lirakis: FUN!!! |
| 13:44.57 | dandre | MrChimpy: because it runs on another host |
| 13:45.18 | MrChimpy | dandre: ok then do it sensibly through SSH |
| 13:45.33 | [TK]D-Fender | ^^^ |
| 13:47.15 | dandre | ok |
| 13:49.23 | *** join/#asterisk phillipk (n=pkey@fw.datafax.net) |
| 14:09.19 | *** join/#asterisk Uzzi (n=andrea@host49-236-dynamic.2-87-r.retail.telecomitalia.it) |
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| 14:28.53 | *** join/#asterisk Dr-Linux (n=Nothing@202.125.139.198) |
| 14:28.59 | Dr-Linux | hi guys |
| 14:29.07 | *** join/#asterisk harpal (n=Harpal@124.125.255.24) |
| 14:30.08 | Dr-Linux | i'm recording my queues calls with within the queue with option monitor-format.. etc, but i want to save these calls on different location. |
| 14:30.28 | Dr-Linux | i had a link for it but i unable to find it |
| 14:30.30 | Dr-Linux | anyone can help? |
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| 14:33.57 | afritz | good morning |
| 14:34.07 | *** part/#asterisk luckyone (n=hidden@CPE-65-28-8-148.kc.res.rr.com) |
| 14:35.02 | muiro | question: with mostly a default asterisk setup, how many processes should be running for asterisk? |
| 14:35.07 | *** join/#asterisk bkw_ (n=brian@adsl-70-142-57-3.dsl.tul2ok.sbcglobal.net) |
| 14:36.08 | Corydon76-dig | muiro: one process, multiple threads |
| 14:36.46 | Corydon76-dig | How many threads depends upon your configuration, number of channels in service. |
| 14:37.25 | muiro | Corydon76-dig: thanks. |
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| 14:37.53 | *** mode/#asterisk [+o anthm] by ChanServ |
| 14:38.40 | *** join/#asterisk jcmoore (n=jcmoore@unaffiliated/tgrman) |
| 14:40.30 | Dr-Linux | Corydon76-dig: any clue on my question? |
| 14:42.47 | *** join/#asterisk mbranca (i=daemon@mi-gw1.voismart.net) |
| 14:43.51 | Corydon76-dig | Dr-Linux: give it a full pathname |
| 14:44.48 | Dr-Linux | Corydon76-dig: yes that i understand, maybe in first priority i can do like this: |
| 14:45.05 | Dr-Linux | exten => 111,1,SetVar(CALLFILENAME=/home/callcenter/MCP/TVP/${STRFTIME(${EPOCH},,%Y%m%d)}/${CALLERIDNUM}-${STRFTIME(,,%c)}) |
| 14:45.41 | Dr-Linux | Corydon76-dig: now how i will use this CALLFILENAME variable in 2nd priroity where i'm calling queue? |
| 14:47.15 | Dr-Linux | Corydon76-dig: i had a link for this help, but i lost that, now i'm unable to find |
| 14:49.27 | *** join/#asterisk [hC] (n=hardcore@S01060016b61c8983.vf.shawcable.net) |
| 14:53.48 | *** join/#asterisk blitzrage (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 14:53.48 | *** mode/#asterisk [+o blitzrage] by ChanServ |
| 14:55.06 | e` | how can I set, when a specific number is called, if you press 0, it goes to voicemail? |
| 14:57.21 | blitzrage | huh? |
| 14:57.31 | blitzrage | what are you calling? |
| 14:57.39 | blitzrage | and from which direction |
| 14:58.43 | Dr-Linux | blitzrage: any clue on my question? |
| 14:59.02 | blitzrage | Dr-Linux: what question -- as you may have noticed, I just joined 2 mins ago |
| 14:59.03 | e` | example, when i call extension 555, while it rings, if i press 0, it will go to voicemail |
| 14:59.06 | e` | makes sense/. |
| 14:59.07 | e` | ? |
| 14:59.12 | blitzrage | e`: yes, that makes sense |
| 14:59.22 | Dr-Linux | blitzrage: lemme ask again |
| 14:59.38 | *** join/#asterisk shinao1 (n=shinao1@217.20.242.50) |
| 14:59.55 | Dr-Linux | blitzrage: i'm recording my queues calls with within the queue with option monitor-format.. etc, but i want to save these calls on different location. |
| 15:00.13 | Dr-Linux | <PROTECTED> |
| 15:00.19 | blitzrage | e`: show application dial, and you'll find what you need (it's a flag to Dial that you're looking for) |
| 15:00.34 | Dr-Linux | blitzrage: now how i will use this CALLFILENAME variable in 2nd priroity where i'm calling queue? |
| 15:01.00 | blitzrage | Dr-Linux: sorry, not sure |
| 15:01.01 | Dr-Linux | like here: exten => 111,n,Queue(mcp-support|tT|||600) |
| 15:01.08 | blitzrage | I just woke up 2 mins ago |
| 15:01.15 | Dr-Linux | good morning |
| 15:01.17 | blitzrage | morning :) |
| 15:01.18 | Dr-Linux | :) |
| 15:01.23 | Dr-Linux | here is night |
| 15:01.28 | blitzrage | ya, 10am here |
| 15:01.31 | blitzrage | late start for me |
| 15:01.49 | Dr-Linux | actually i had a link for it, but i lost it and trying to find it since morning |
| 15:01.50 | blitzrage | was up late working on virtual machine auto-provisioning |
| 15:01.57 | Dr-Linux | here 8:01 pm |
| 15:04.35 | davevg-btwtech | Dr-Linux, are you looking for something like this Set(MONITOR_FILENAME=/home/callcenter/......) |
| 15:05.10 | Dr-Linux | davevg-btwtech: that's already i'm doing |
| 15:05.22 | [TK]D-Fender | Dr-Linux: which is clearly written in the queues.conf.sample file which you apparently didn't bother reading |
| 15:05.58 | Dr-Linux | but i'm recording calls from the queue with monitor-format=wav option, but i records calls on default location, i wanna change location |
| 15:06.10 | Dr-Linux | and i paste there the first command which is: |
| 15:06.15 | Dr-Linux | exten => 111,1,SetVar(CALLFILENAME=/home/callcenter/MCP/TVP/${STRFTIME(${EPOCH},,%Y%m%d)}/${CALLERIDNUM}-${STRFTIME(,,%c)}) |
| 15:06.17 | [TK]D-Fender | Dr-Linux: us the proper variable name <- |
| 15:06.22 | [TK]D-Fender | use* |
| 15:06.27 | Dr-Linux | now how i will use this CALLFILENAME variable in 2nd priroity where i'm calling queue? |
| 15:06.30 | Dr-Linux | oo |
| 15:06.40 | badcfe | i start dialplan in Answer and ResetCDR. however when i hangup, the CDR is not posted .. i get message "CDR on channel 'SIP/2227654902-b5f01ca0' not posted" |
| 15:06.43 | [TK]D-Fender | Dr-Linux: MONITOR_FILENAME != CALLFILENAME |
| 15:06.43 | badcfe | why? |
| 15:07.04 | badcfe | .. i want the CDR to be posted anyway (even that its true, that i didnt do any Dial ..) |
| 15:07.13 | badcfe | how? |
| 15:07.31 | blitzrage | badcfe: CDRs are not posted in 1.4.14 if the call was not answered |
| 15:07.46 | badcfe | but but, i do Answer on the incoming chan ... |
| 15:08.31 | blitzrage | badcfe: i still think the call has to bridge somewhere for it to post a CDR |
| 15:08.34 | blitzrage | codefreeze: is that true? |
| 15:08.40 | Dr-Linux | [TK]D-Fender: yes, that i understand but where should i use CALLFILENAME variable here >> exten => 111,n,Queue(mcp-support|tT|||600) ? |
| 15:08.59 | [TK]D-Fender | Dr-Linux: SET THE VARIABLE WE TOLD YOU TO! |
| 15:09.00 | Dr-Linux | that's what bothering me |
| 15:09.14 | Dr-Linux | [TK]D-Fender: i already set that |
| 15:09.16 | anonymouz666 | blitzrage: so how to account the missed calls? |
| 15:09.21 | [TK]D-Fender | Dr-Linux: No, you DIDN'T |
| 15:09.25 | [TK]D-Fender | Dr-Linux: MONITOR_FILENAME != CALLFILENAME <------------- |
| 15:09.31 | Dr-Linux | <PROTECTED> |
| 15:09.33 | Dr-Linux | isn't it? |
| 15:09.46 | blitzrage | anonymouz666: you don't... (until codefreeze adds the old way of posting EVERYTHING in as an option) |
| 15:09.51 | [TK]D-Fender | Dr-Linux: You are NOT setting MONITOR_FILENAME there! |
| 15:10.12 | blitzrage | the change was based on the community, which then promptly said it wanted the old way too... or something like that |
| 15:11.19 | Dr-Linux | [TK]D-Fender: this one portion is already working for me |
| 15:11.59 | Dr-Linux | but i'm not sure where i can put ${CALLFILENAME} var over here >> exten => 111,n,Queue(mcp-support|tT|||600) |
| 15:12.20 | Dr-Linux | sorry, for my bad english, but my question is straight forward |
| 15:13.08 | Dr-Linux | [TK]D-Fender: i know you are talking about this >> Monitor(gsm,${CALLFILENAME}) |
| 15:13.16 | Dr-Linux | but in this case i don't know if i need this |
| 15:13.33 | Dr-Linux | bcoz this thing is beind done by queue |
| 15:13.37 | [TK]D-Fender | Dr-Linux: No, I'm not. |
| 15:14.27 | [TK]D-Fender | Dr-Linux: You. Must. Set. THIS. Variable. BEFORE Entering the Queue : MONITOR_FILENAME <------ This EXACT VARIAABLE. |
| 15:14.40 | [TK]D-Fender | asdffdasdfasdfasfasdfjkl;sdfjgl;skdfjgl;sdfjgkl;kfdgfdgfdgfds |
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| 15:18.52 | Dr-Linux | [TK]D-Fender: ok thanks |
| 15:18.54 | Dr-Linux | done |
| 15:19.01 | bofh80 | hi people, i have installed asteriskNOW but am having troubles getting my X100P to be picked up, the web interface states that no analouge lines have been found :( i have followed kb 34 and 40, to no avail, any help please? |
| 15:19.41 | Dr-Linux | <PROTECTED> |
| 15:20.52 | [TK]D-Fender | bofh80: pastebin the output of "dmesg" , "cat /proc/interrupts" |
| 15:21.18 | [TK]D-Fender | bofh80: And "zaptel.conf & zapata.conf |
| 15:21.20 | [TK]D-Fender | ~pb |
| 15:21.21 | 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 |
| 15:21.22 | [TK]D-Fender | ^^^^^^^^^^^^^^^^^ |
| 15:21.47 | bofh80 | [TK]D-Fender, thanks very much will do and give you links :) |
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| 15:29.44 | blitzrage | anonymouz666: codefreeze has just informed me the new option is in the 1.4 branch if you need it post-1.4.14 |
| 15:30.46 | jameswf | I need to connect 3 boxen 1 master that everything talks to and 2 slave midgits that bear 100 extensions each, do I do a bunch of iax trunking or should i bother with dundi |
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| 15:32.36 | IOscanner | I am having an issue with meetme. It has been working fine ever. I found today that it is recieving the pin, but it says the pin is not valid. |
| 15:32.38 | *** join/#asterisk RoyK (n=roy@213.160.242.90) |
| 15:32.46 | IOscanner | Any ideas how to find the problem? |
| 15:32.52 | IOscanner | I don't see anything in the logs |
| 15:33.22 | [TK]D-Fender | IOscanner: pastebin your configs and CLI output with enough debug to see the DTMF passed. All in a pastebin |
| 15:33.23 | [TK]D-Fender | ~pb |
| 15:33.24 | 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 |
| 15:33.25 | [TK]D-Fender | ^^^^^^^^^^^^^^^^^^ |
| 15:37.02 | IOscanner | http://pastebin.com/d72b2d11 |
| 15:37.10 | IOscanner | First time I have seen this |
| 15:37.29 | IOscanner | DTMF is passed and asterisk even sees it |
| 15:40.02 | RoyK | those 1st gen te410p cards are quite crappy...... |
| 15:40.08 | IOscanner | I updated with the dialplan for the 8103. |
| 15:40.08 | IOscanner | http://pastebin.com/d491ff86b |
| 15:41.17 | Yourname`` | Hi. Need some insight on this error please -> Nov 29 10:41:29 NOTICE[3333]: chan_sip.c:3773 process_sdp: No compatible codecs! |
| 15:42.22 | bofh80 | [TK]D-Fender, all requested info: problem X100P card :) http://pastebin.com/m442cc392 |
| 15:43.03 | *** join/#asterisk tobias (n=tobias@user-0ce2hr3.cable.mindspring.com) |
| 15:43.47 | *** join/#asterisk ManxPower (n=manxpowe@106.sub-75-203-20.myvzw.com) |
| 15:44.05 | [TK]D-Fender | bofh80: try "modprobe wcfxo" and redo "cat /proc/interrupts" |
| 15:44.38 | bofh80 | [TK]D-Fender, i have put the interrupts in there, and the other conf files, keep scrolling :D |
| 15:44.53 | bofh80 | [TK]D-Fender, i'll try that modprobe tho |
| 15:46.24 | bofh80 | [TK]D-Fender, am using the asterisknow. uhm modprobe says bad command |
| 15:46.44 | Yourname`` | Nov 29 10:46:53 NOTICE[3333]: chan_sip.c:3773 process_sdp: No compatible codecs! -> How do I rid this error? |
| 15:47.14 | [TK]D-Fender | bofh80: ok, out of ideas... no idea where that distro leaves your KO's |
| 15:47.37 | [TK]D-Fender | Yourname``: Set your codecs right. And next time apstebin the failed call with sip debug enabled. |
| 15:48.00 | Yourname`` | [TK]D-Fender: ALL codecs are set to ulaw. :S Lemme pb it. |
| 15:48.57 | [TK]D-Fender | Yourname``: and your configs |
| 15:50.04 | ManxPower | Yourname``: Usually that means the phone and Asterisk cannot agree on a common codec to use. |
| 15:50.35 | Yourname`` | ManxPower: That's what I thought, but I set the phone to use pref codec only, which is ulaw, and same with Asterisk. :S |
| 15:50.36 | ManxPower | For example, if the phone only accepts G729 and Asterisk is configured to only accept ulaw, you could get that message |
| 15:50.50 | ManxPower | Yourname``: then you are not doing something right. |
| 15:50.52 | [TK]D-Fender | Yourname``: Stop talking, more Pastebin. |
| 15:51.06 | Yourname`` | [TK]D-Fender: lol gosh, one sec.. you're so hard on me sometimes! |
| 15:51.22 | *** join/#asterisk russellb (i=russell@asterisk/developer-and-stable-maintainer/drumkilla) |
| 15:51.22 | *** mode/#asterisk [+o russellb] by ChanServ |
| 15:51.49 | [TK]D-Fender | Yourname``: Allt he "but I did everything right!" in the world isn't going to help your situation, so move along to something that will. |
| 15:52.15 | bofh80 | [TK]D-Fender, you seem to know what your doing, what's your recommended procedure for installing asterisk ? debian? or what? on it's own machine, only 2 people in my office atm that will be using it. |
| 15:52.52 | [TK]D-Fender | bofh80: I always suggest installing some sane Distro & compiling * and its related bits from source tarball downloads |
| 15:53.36 | bofh80 | [TK]D-Fender, i tried that with ubuntu, but the asterisk service simply starts then stops. no errors or logs to be found anywhere |
| 15:53.45 | *** join/#asterisk fskrotzki (n=fskrot@host198.textwise.com) |
| 15:54.00 | [TK]D-Fender | bofh80: Well you don't have anything to show us right now, and Ubuntu has a lot of "gotchas" |
| 15:54.18 | Yourname`` | ManxPower: I'm hardly ever doing anything right. :( |
| 15:54.37 | bofh80 | [TK]D-Fender, would i be ok to install and use Debian then you think? and go from there? would that be a safe option? i need this running, i bought the X100P sepecifically :( i am at 'home' with debian? |
| 15:54.55 | IOscanner | so has anyone seen why a conference room will not work even if everything is setup correct. It worked until this morning. I think it is a bug, but I have never seen this issue before. http://pastebin.com/d491ff86b |
| 15:55.05 | [TK]D-Fender | bofh80: Debian is a perfectly fine choice. |
| 15:55.11 | Yourname`` | Perfect, for some reason I'm not able to paste it in pastebin! argh |
| 15:55.12 | Yourname`` | ~pb |
| 15:55.13 | 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 |
| 15:55.16 | bofh80 | [TK]D-Fender, then i will start there :) thanks. |
| 15:56.16 | Yourname`` | [TK]D-Fender: Pastebin of sip debug: http://pastebin.ca/801762 |
| 15:57.54 | Yourname`` | Wait, disallow needs to be on top of allow in sip.conf, correcto? |
| 15:58.08 | [TK]D-Fender | Yourname``: I don't see your error in there at all and it si a MESS. Stop using a web process to capture your CLI |
| 15:58.21 | Yourname`` | [TK]D-Fender: It's putty capturing it.. I hate it too. |
| 15:58.46 | *** join/#asterisk putnopvut (i=putnopvu@nat/digium/x-9bc5a659f814233b) |
| 15:58.55 | Yourname`` | Ok, I think I got it. |
| 15:58.57 | Yourname`` | That was it. |
| 15:59.07 | Yourname`` | disallow was ALL the way at the bottom of the [peer] |
| 15:59.14 | Yourname`` | and allow was somewhere at the top. |
| 15:59.18 | Yourname`` | Stupidness. |
| 15:59.34 | ManxPower | you want to disallow=all then allow= only the codec you want. |
| 15:59.38 | Yourname`` | Well, thank you for looking through that garbage in the pb, [TK]D-Fender. |
| 15:59.59 | Yourname`` | ManxPower: Yessir, that's what I did. It was just in the incorrect order, allow was at the top, and disallow was all the way at the bottom. |
| 16:00.29 | ManxPower | Yourname``: I paste from Putty all the time and it does not screw up the new lines |
| 16:00.49 | Yourname`` | ManxPower: Really? Hmm, you go to session logging and log "All output", no? |
| 16:01.38 | ManxPower | no, I copy and paste from the scrollback buffer |
| 16:01.59 | ManxPower | I do have a large scrollback buffer set. |
| 16:02.25 | _Sam-- | just as an aside, the scrollback buffer in putty doesnt work well, if you are running 'screen' |
| 16:02.55 | ManxPower | _Sam--: that is correct. |
| 16:03.05 | ManxPower | One of my annoyances with screen. |
| 16:03.27 | _Sam-- | yeah....it messes me with me alot. oh well. |
| 16:03.42 | Yourname`` | ManxPower: And yo just "Copy all to clipboard"? nice. |
| 16:03.55 | _Sam-- | <PROTECTED> |
| 16:03.55 | _Sam-- | <[TK]D-Fender> Yourname``: Stop talking, more Pastebin. |
| 16:03.55 | _Sam-- | -:- SignOff zippytech: #asterisk (Remote closed the connection) |
| 16:03.55 | _Sam-- | <Yourname``> [TK]D-Fender: lol gosh, one sec.. you're so hard on me sometimes! |
| 16:03.55 | _Sam-- | -:- russellb [i=russell@asterisk/developer-and-stable-maintainer/drumkilla] has joined #asterisk |
| 16:03.56 | _Sam-- | -:- mode/#asterisk [+o russellb] by ChanServ |
| 16:03.58 | _Sam-- | <[TK]D-Fender> Yourname``: Allt he "but I did everything right!" in the world isn't going to help your situation, so move along to something that will. |
| 16:04.01 | _Sam-- | <bofh80> [TK]D-Fender, you seem to know what your doing, what's your recommended procedure for installing asterisk ? debian? or what? on it's own machine, only 2 people in my office atm that will be using |
| 16:04.03 | Nivex | Use Ctrl-A [ to get into copy mode. That will let you scroll back in screen. Then just hit ESC when done |
| 16:04.04 | _Sam-- | <PROTECTED> |
| 16:04.06 | _Sam-- | <[TK]D-Fender> bofh80: I always suggest installing some sane Distro & compiling * and its related bits from source tarball downloads |
| 16:04.09 | _Sam-- | <bofh80> [TK]D-Fender, i tried that with ubuntu, but the asterisk service simply starts then stops. no errors or logs to be found anywhere |
| 16:04.14 | _Sam-- | er damn right click. |
| 16:04.16 | _Sam-- | <sorry> |
| 16:04.23 | Yourname`` | Well, hello to you too _Sam-- ;) |
| 16:04.38 | *** join/#asterisk fordfrog (n=fordfrog@gentoo/developer/fordfrog) |
| 16:04.42 | Yourname`` | Nivex: Nice, thanks. |
| 16:04.42 | _Sam-- | i was messing with my scrolback :) |
| 16:05.01 | Nivex | <3 screen |
| 16:05.06 | _Sam-- | Yourname`` : you got your teliax dtmf fixed? |
| 16:05.39 | ManxPower | The company we bought our new mail server software (open source) seems to have taken customer service lessons from Microsoft. |
| 16:05.57 | Yourname`` | _Sam--: Yes, this was part of that regime, heh. They put me on some proxy that had 'the fix' :) |
| 16:06.08 | Yourname`` | ManxPower: People picking up in India? |
| 16:06.43 | ManxPower | Yourname``: No. It's a small company. E-mail is frequently unanswered, nobody answers their tech support line. |
| 16:06.44 | _Sam-- | Yourname`` : thats not the real fix. i think i mentioned yesterday they had one or two servers still that would work with 1.2, but thats not the 'fix'. but if it works for you, then it works for me. |
| 16:07.10 | ManxPower | Yourname``: they have a good product. |
| 16:07.20 | Yourname`` | ManxPower: Yeah, hence the open source. :( |
| 16:07.34 | Yourname`` | _Sam--: Yeah, they put me on a server that works with 1.2 |
| 16:07.51 | _Sam-- | probably voip-co4 or voip-ny1 like i told you in PM yesterday. |
| 16:08.01 | Yourname`` | yup yup :) |
| 16:08.25 | _Sam-- | you could have done that yourself, using your login , and their web interface. |
| 16:08.36 | file | is this teliax DTMF issue sending to them or receiving from them? |
| 16:08.51 | Yourname`` | _Sam--: .. yeah, should have.. :( |
| 16:08.51 | _Sam-- | sending to them, with 1.2 versions of *, on some of their hosts. |
| 16:09.13 | file | ah, if they upgraded them to 1.4 they probably didn't set the compatibility option |
| 16:09.48 | anonymouz666 | rfc2833compensate? |
| 16:09.55 | _Sam-- | alls i know is they have a fix on their side, if you complain. mine was fixed within 3 minutes of mentioning the problem. |
| 16:10.07 | _Sam-- | i will point that out to them, though -rcf2833compensate. |
| 16:10.09 | *** join/#asterisk csmiga (n=chsmiga@68-114-1-65.dhcp.gsvl.ga.charter.com) |
| 16:11.15 | _Sam-- | they probably didnt compile with it, and now just add it manually to people's sip config |
| 16:11.23 | _Sam-- | is that possible? |
| 16:11.42 | _Sam-- | or the only way that works, is via sip.conf? |
| 16:11.43 | file | you don't compile with it... it's a global or per-user/per-peer option |
| 16:11.48 | _Sam-- | i see |
| 16:12.27 | _Sam-- | they should just add that as an option in their web interface on a per user basis, then |
| 16:12.40 | _Sam-- | "Use Pre-1.4 Asterisk DTMF" |
| 16:12.55 | *** join/#asterisk kkn088 (n=kikoun@77.204.31.201) |
| 16:13.14 | *** join/#asterisk rcahilig (n=user@203.115.187.98) |
| 16:13.47 | _Sam-- | what is the downside to just setting that option globally, for everyone? |
| 16:13.55 | _Sam-- | Juggie: why havent they done that already? |
| 16:13.57 | Yourname`` | Oh, wow, asterisk 1.2.* has been taken off of asterisk.org |
| 16:14.00 | _Sam-- | er , damn nick completion. |
| 16:14.05 | file | DTMF duration is not preserved |
| 16:14.46 | ManxPower | Yourname``: Digium does not like to deal with old versions, even if 1/2 of the Asterisk installs out there have 1.2 somewhere in the mix. |
| 16:15.03 | Yourname`` | heh |
| 16:15.15 | _Sam-- | i found the 1.2 versions there just the other day. |
| 16:15.36 | ManxPower | I sort of agree with their reasons, I just don't agree there is no solution. |
| 16:15.57 | ManxPower | You can still get it via direct download, of course. You can get ANY version that way. |
| 16:16.04 | _Sam-- | right |
| 16:16.07 | rcahilig | Hi, I'm using asterisk 1.2 in a 10 seater call center and I want to record each agent call, how do I do this, were not using predictive dialer. Thanks. |
| 16:16.14 | file | ManxPower: and what is the solution in your mind? |
| 16:16.19 | _Sam-- | http://downloads.digium.com/pub/asterisk/ |
| 16:16.33 | ManxPower | file: I didn't say I had a solution. 8-| |
| 16:17.01 | file | ManxPower: that makes two things O.O |
| 16:17.53 | ManxPower | I also don't have a solution to the War on Drugs, but it is pretty obvious the current methods for that are not working. |
| 16:17.58 | [TK]D-Fender | rcahilig: Please read the queues.conf.sample file as it explain everything you need to know about how to record agent's calls |
| 16:18.54 | [TK]D-Fender | ManxPower: I do. End prohibition. Period. |
| 16:19.50 | rcahilig | [TK]D-Fender:I don't think I need the queue, were outbound call center |
| 16:19.57 | ManxPower | [TK]D-Fender: as much as I like that idea, it won't work all by itself. |
| 16:20.13 | *** join/#asterisk |omni| (n=rob@c-67-185-144-206.hsd1.wa.comcast.net) |
| 16:21.30 | [TK]D-Fender | ManxPower: Sure it will. Can't declare war on a lifestyle. The cost of prohibition is unbelievably stupid, expensive and ineffective. tobacco, alcohol, cars and even bathroom accidents each kill immensely more people every day |
| 16:23.12 | ManxPower | They should legalize all drugs except for antibiotics, tax them, use the tax money to provide free addiction treatment. |
| 16:26.31 | [TK]D-Fender | umm... Anti-biotics are legal... just not "over the counter"... |
| 16:26.53 | [TK]D-Fender | ManxPower: but tahts fine in the interest of trying to save you from poisoning yourself. |
| 16:31.42 | *** join/#asterisk chris_1 (n=chris@ng1.kurtkrenn.com) |
| 16:34.15 | mort_gib | Having a problem logging into my voicemail! |
| 16:34.32 | mort_gib | As anyone is not here, who is? |
| 16:36.32 | *** join/#asterisk UnixDog (n=unixdog@adsl-69-234-183-43.dsl.irvnca.pacbell.net) |
| 16:36.57 | [TK]D-Fender | mort_gib: nope, its just, and perhaps you could proveide some actual details and CLI output and configs so we can SEE your porblem |
| 16:37.03 | [TK]D-Fender | ~pb |
| 16:37.04 | jbot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste |
| 16:37.05 | [TK]D-Fender | ^^^^^^^^^^^ |
| 16:39.11 | *** join/#asterisk outtolunc (n=me@adsl-66-218-53-172.dslextreme.com) |
| 16:39.47 | mort_gib | http://pastebin.com/d7d48ca71 |
| 16:40.13 | mort_gib | No CLI output other than the expected where voicemail is played |
| 16:40.43 | mort_gib | I think it's dtmf that I haven't fully understood yet! |
| 16:41.57 | ManxPower | I still don't see the CLI output of a failed call. |
| 16:42.31 | mort_gib | As in login incorrect... |
| 16:42.59 | ManxPower | mort_gib: there are three DTMF modes. RFC2833, INFO, and INBAND. INFO is not recommended unless you need it, INBAND only works with ulaw and alaw codecs. You must set the same DTMF mode on ASTERISK and on the VOIP DEVICE or it won't work correctly. |
| 16:44.37 | *** join/#asterisk CunningPike (n=arodgers@204.239.12.183) |
| 16:45.58 | *** join/#asterisk xrg_ (n=panos@cpe028189.netone.gr) |
| 16:46.23 | *** join/#asterisk alerios (n=alerios@190.84.254.27) |
| 16:46.43 | *** part/#asterisk alerios (n=alerios@190.84.254.27) |
| 16:47.10 | xrg_ | up: Any asterisk core dev from *digium* please prv. me! |
| 16:47.49 | Corydon76-vcch | xrg_: why? |
| 16:48.46 | ManxPower | Corydon76-vcch: he probably wants to bypass the normal support methods and get his question answered by a developer. |
| 16:49.31 | Corydon76-vcch | Usually, the only reason to take it private at this stage is for the purpose of reporting a security vulnerability |
| 16:49.32 | xrg_ | w8 |
| 16:49.44 | *** join/#asterisk klictel (n=klictel@atelka.info) |
| 16:49.51 | Corydon76-vcch | and even then, security@asterisk.org is a much better idea |
| 16:50.45 | ManxPower | w8? That's a bad sign right there. |
| 16:50.57 | ManxPower | xrg_: then come back when you are not busy. |
| 16:51.34 | nestAr | HAX |
| 16:51.35 | nestAr | OMG |
| 16:52.40 | putnopvut | xrg_: if this is a license concern you have, there is an e-mail address on the license agreement to direct questions to. |
| 16:53.57 | xrg_ | putnopvut: Yes, it is both, licensing and security.. |
| 16:55.49 | anonymouz666 | shhhh... Don't say nothing. Linus Torvalds likes to fix kernel security bugs and don't tell to anyone. |
| 16:55.54 | anonymouz666 | lol |
| 16:56.43 | *** join/#asterisk teknoprep (n=tekno@unaffiliated/teknoprep) |
| 16:56.45 | teknoprep | hey all |
| 16:58.00 | teknoprep | i am having a problem with registering the same extension multiple times on the same phone |
| 16:58.03 | teknoprep | i have 1.4.13 |
| 16:58.08 | anonymouz666 | is it right to fix security bugs without exposing the issue? |
| 16:58.09 | teknoprep | and cisco 7940 phones |
| 16:58.32 | *** join/#asterisk essobi_ (n=essobi_@72.1.94.14) |
| 16:58.36 | *** part/#asterisk essobi_ (n=essobi_@72.1.94.14) |
| 16:59.04 | *** join/#asterisk Koshatul (n=evangeli@ppp188-33.static.internode.on.net) |
| 17:01.26 | xrg_ | anonymouz666: It's about ethics: if you want to *help* the project, you don't just disclose security issues when you know people can't patch their systems. |
| 17:01.50 | *** join/#asterisk Koshatul (n=evangeli@ppp188-33.static.internode.on.net) |
| 17:01.59 | mort_gib | ManxPower: I'm using Llinphone (and a horrible fax/phone from Panasonic) none of them will log in. Linphone will only let me use info or rfc2833 -have tried both |
| 17:02.18 | ManxPower | teknoprep: you can't do that. |
| 17:03.00 | ManxPower | teknoprep: The cisco phones will emulate that if you only register the first line and leave the rest of the lines' registration info empty. |
| 17:03.29 | anonymouz666 | ASA are sent to bugtraq also? |
| 17:04.00 | russellb | yes |
| 17:05.22 | russellb | xrg_: i would be happy to talk about any security issues in private. You don't have to submit code to describe an issue if that's what you want to do. |
| 17:05.54 | russellb | xrg_: Otherwise, if you have license concerns, putnopvut gave you the correct recommendation. You should direct any questions or concerns you have to the email address listed on the license. |
| 17:06.22 | russellb | xrg_: and for discussing security issues, i would prefer encrypted email. |
| 17:06.35 | ManxPower | mvanbaak: My Stroopvaffels order should arrive today. |
| 17:06.39 | tzanger | my zipper is a security issue |
| 17:06.52 | ManxPower | tzanger: only if someone wants to open it. |
| 17:07.17 | ManxPower | Heck, leaving my truck unlocked is a security issue -- except that nobody wants to steal it. |
| 17:09.42 | outtolunc | less than 30 days left to wish hateful thoughts on the person that just bought the last of the item you wanted out from under you |
| 17:12.31 | ManxPower | outtolunc: I usually just track them down and kill them. |
| 17:12.32 | outtolunc | dang, i thought it was friday already.. grr |
| 17:12.40 | *** join/#asterisk LakeSolon (n=blake@64-83-209-86.dhcp.stcd.mn.charter.com) |
| 17:12.45 | blitzrage | outtolunc: me too... burn |
| 17:12.59 | [TK]D-Fender | mort_gib: pastebin your CLI output of the failed attempt and your configs. |
| 17:13.00 | [TK]D-Fender | ~pb |
| 17:13.01 | 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 |
| 17:13.01 | *** join/#asterisk qdk (n=qdk@0x573fe57e.bynqu2.broadband.tele.dk) |
| 17:17.48 | mvanbaak | ManxPower: Stroopwafels ???? YUMMEEEEEEEEEE !!!!!!!!! |
| 17:17.50 | mvanbaak | share! |
| 17:20.05 | *** join/#asterisk loca|host (n=tux@196.203.53.221) |
| 17:20.07 | loca|host | hello all |
| 17:20.51 | loca|host | i've just installed asterisk, but got warning that 3 modules were not installed on my system, these are the modules: app_saycountpl.so chan_ooh323.so format_mp3.so |
| 17:20.53 | [TK]D-Fender | mvanbaak: Indeed.... |
| 17:21.13 | loca|host | but it doent say what are the dependencys for those modules so i can figure it out |
| 17:21.14 | [TK]D-Fender | loca|host: All part of asterisk-addons |
| 17:21.50 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 17:21.54 | loca|host | [TK]D-Fender, so should i ignore that WARNING and go install asterisk-addons ? |
| 17:22.02 | UnixDog | cool my project installs and works |
| 17:22.08 | [TK]D-Fender | loca|host: pastebin this full message |
| 17:22.09 | mvanbaak | loca|host: run 'make menuselect' |
| 17:22.14 | mvanbaak | that will give you the dependecies |
| 17:22.25 | UnixDog | now we have a easy to install asterisk/bsd setp |
| 17:22.29 | [TK]D-Fender | mvanbaak: nope |
| 17:22.38 | UnixDog | much smaller then te linux distros now days |
| 17:22.48 | file | the message says the modules were not installed BY this version of Asterisk... because they are part of -addons |
| 17:22.54 | [TK]D-Fender | mvanbaak: * didn't come with those modules, and its just a warning that it FOUND them. Everything is probably fine as-is |
| 17:23.04 | loca|host | mvanbaak, http://pastebin.ca/801852 |
| 17:23.16 | mvanbaak | <PROTECTED> |
| 17:23.20 | mvanbaak | <PROTECTED> |
| 17:23.24 | mvanbaak | [TK]D-Fender: it does say |
| 17:23.28 | mvanbaak | ah |
| 17:23.29 | mvanbaak | :) |
| 17:23.31 | mvanbaak | lol |
| 17:24.08 | [TK]D-Fender | mvanbaak: Wasn't debating the function of a dependency check, I just devalidated the NEED for it :) |
| 17:25.27 | loca|host | hey guys, how to figure this issue ? |
| 17:25.31 | loca|host | should i go install the addons and ignore the warning ? |
| 17:25.51 | [TK]D-Fender | loca|host: you already have a version installed. is it still compatible? |
| 17:26.54 | mvanbaak | if you upgraded asterisk, the best shot is to recompile the addons as well |
| 17:27.17 | loca|host | i havent installed asterisk in the past |
| 17:27.24 | loca|host | it's the first time i do it on this server |
| 17:27.43 | *** join/#asterisk grEvenX (n=even@pc107-130.ktv.no) |
| 17:29.20 | [TK]D-Fender | loca|host: well *-addons IS installed. |
| 17:29.34 | [TK]D-Fender | loca|host: So just try it out. Probably fine |
| 17:37.13 | *** join/#asterisk GoldFingaZ (n=wt5@bas13-ottawa23-1177809155.dsl.bell.ca) |
| 17:38.09 | loca|host | i've successfully installed Zaptel |
| 17:38.21 | loca|host | and when doing modprobe ztdummy |
| 17:38.28 | loca|host | i got this: FATAL: Module ztdummy not found. |
| 17:39.46 | loca|host | ok i've done this: ln -s /lib/modules/2.6.18 /lib/modules/2.6.18-ovz-028stab045.1/asterisk |
| 17:39.57 | loca|host | now, i have this error: |
| 17:40.04 | loca|host | FATAL: Error inserting ztdummy (/lib/modules/2.6.18-ovz-028stab045.1/asterisk/misc/ztdummy.ko): Invalid module format |
| 17:40.19 | *** join/#asterisk atisss (n=atisss@193.238.212.171) |
| 17:40.34 | [TK]D-Fender | loca|host: recompile zaptel. Looks like your kernel header versiondidn't match |
| 17:41.05 | loca|host | it's being recompiled now |
| 17:41.18 | loca|host | configure: *** Zaptel build successfully configured *** |
| 17:41.26 | loca|host | Zaptel installed successfully. |
| 17:42.30 | loca|host | same error |
| 17:46.10 | loca|host | any idea ? |
| 17:49.22 | *** join/#asterisk Defraz (n=t0tal@fw.fuzecore.com) |
| 17:51.44 | Corydon76-vcch | Make sure the currently running kernel matches the kernel source version you have installed |
| 17:52.14 | *** join/#asterisk shinao1 (n=shinao1@217.20.242.50) |
| 17:52.23 | Corydon76-vcch | and I mean exact string match, not just version number |
| 18:04.11 | Juggie | does cmd BUSY() not set a pri_cause when it hangs up |
| 18:04.33 | *** part/#asterisk dandre (n=daniel@was59-3-82-236-48-30.fbx.proxad.net) |
| 18:07.45 | *** join/#asterisk atomicd (n=Atomicd@74-206-0-80.static-ip.m.telepacific.net) |
| 18:10.38 | loca|host | Codemaster, it seems i got some old headers installed |
| 18:10.43 | loca|host | i'll delete'm all |
| 18:14.13 | loca|host | same problem |
| 18:14.14 | loca|host | :( |
| 18:16.19 | *** join/#asterisk riksta (n=rick@rhamnett.plus.com) |
| 18:16.36 | *** part/#asterisk bkw_ (n=brian@adsl-70-142-57-3.dsl.tul2ok.sbcglobal.net) |
| 18:19.43 | *** join/#asterisk hi365_m (n=hi365@213.151.52.98) |
| 18:20.20 | hi365_m | im having this weird thin where asterisk is reporting a pri on span 2 when in fact it should be on span 1 << PRI span 2/0: Provisioned, In Alarm, Down, Active |
| 18:20.27 | hi365_m | http://pastebin.ca/801899 |
| 18:21.46 | *** join/#asterisk shinao1 (n=shinao1@217.20.242.50) |
| 18:23.10 | blitzrage | does anyone have a really good sample set of configuration files for a 7960? (I don't need the firmware) |
| 18:26.05 | IOscanner | Anyone know of a way to get asterisk to see the zaptel module after asterisk has started. For some reason libc6 was upgraded and zaptel module did build. Asterisk came up but the conference rooms don't work with out it. |
| 18:26.07 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 18:26.33 | IOscanner | Is there a way to resolve it live with out stopping asterisk? |
| 18:26.33 | *** join/#asterisk Runlvl (n=gustavo@190.2.40.57) |
| 18:27.55 | *** join/#asterisk honeybeebuzz (n=bee@206-248-138-47.dsl.teksavvy.com) |
| 18:27.55 | *** join/#asterisk Yourname`` (n=Miranda@unaffiliated/yourname/x-837320) |
| 18:28.14 | *** part/#asterisk honeybeebuzz (n=bee@206-248-138-47.dsl.teksavvy.com) |
| 18:28.39 | *** join/#asterisk oej (n=olle@85.19.87.179) |
| 18:33.47 | hi365_m | im having this weird thin where asterisk is reporting a pri on span 2 when in fact it should be on span 1 "PRI span 2/0: Provisioned, In Alarm, Down, Active" how can I debug?? |
| 18:33.51 | hi365_m | http://pastebin.ca/801899 |
| 18:33.55 | *** join/#asterisk RoyK (n=roy@ip-135-16-149-91.dialup.ice.no) |
| 18:37.00 | *** join/#asterisk alrs (n=lars@pozug.com) |
| 18:38.12 | *** join/#asterisk mackes (n=root@65-121-253-83.dia.static.qwest.net) |
| 18:38.37 | *** join/#asterisk alrs (n=lars@pozug.com) |
| 18:47.27 | *** join/#asterisk UnixDog (n=unixdog@adsl-69-234-183-43.dsl.irvnca.pacbell.net) |
| 18:49.04 | *** join/#asterisk anonymouz666 (n=anonymou@201.19.134.26) |
| 18:50.57 | Yourname`` | Quick dialplan question, I have a use for 12345 in [context1], I have the same use for 12345 in [context2].. there isn't any way they will get mixed, right? Both these extensions are supposed to add an agent to a queue. Except, they add the agent to two different queues. |
| 18:51.32 | Yourname`` | Or does it have to be two different extensions? Like 12345 and 54321? |
| 18:53.05 | polerin | as long as you're clear about how you send them there, I don't see why it would get confused. **shrugs** I'm a newb though |
| 18:54.08 | polerin | anyone have an elegant way to connect a logged in user directly to their mailbox without hard coding? |
| 18:54.30 | Yourname`` | hmm |
| 18:54.33 | polerin | for example, pulling it out of the sip information in sip.conf, etc? |
| 18:58.09 | *** join/#asterisk r0d3nt (i=nobody@foster.stonedcoder.org) |
| 18:58.20 | [TK]D-Fender | polerin: typical ways : use CID when the box # matches. Or use SetVar in the sip.conf entry |
| 18:58.53 | _Sam-- | Yourname`` : you still need asterisk dtmf help, or you are happy with your 'fix'? i got the owner on IM asking if you're ok |
| 18:59.10 | _Sam-- | sorry asterisk dtmf help = teliax dtmf help |
| 18:59.44 | Yourname`` | Yeah, I'm ok. Thank you. :) |
| 19:00.05 | *** join/#asterisk rcahilig (n=user@203.115.187.98) |
| 19:00.37 | polerin | [TK]D-Fender: drawbacks to either? |
| 19:01.17 | [TK]D-Fender | polerin: both obvious |
| 19:02.00 | polerin | [TK]D-Fender: I'm fairly new at this, so humor me? CID problem being .. CID? |
| 19:02.10 | polerin | and setvar being fragile or something? |
| 19:02.11 | Yourname`` | exten=> 6,n,Dial(100,,) -> Does this look ok? |
| 19:02.20 | mvanbaak | Yourname``: no |
| 19:02.22 | [TK]D-Fender | polerin: if your box isn't the same as your callerid well that jsut isn't going to wrok, now is it? |
| 19:02.24 | *** join/#asterisk [hC] (n=hardcore@66.119.167.162) |
| 19:02.29 | [TK]D-Fender | Yourname``: Hell no |
| 19:02.36 | *** join/#asterisk asteriskmonkey (n=philip@69.77.169.14) |
| 19:02.36 | Yourname`` | lol |
| 19:02.38 | *** join/#asterisk SirWhit (n=sirjames@blk-222-38-6.eastlink.ca) |
| 19:02.51 | Yourname`` | Basically, it's just going to dial a local extension 100.. and that's wrong? |
| 19:02.52 | *** join/#asterisk |Rain| (i=rain@2001:440:eeee:fffb:42:0:0:2) |
| 19:02.54 | asteriskmonkey | is there a way to interject audio announcemnts while playing music on hold in 1.2x? |
| 19:02.58 | [TK]D-Fender | Yourname``: 100 isnt' a tech |
| 19:03.21 | SirWhit | Hey there.. I have a question about Meetme |
| 19:03.24 | mvanbaak | asteriskmonkey: not that I know |
| 19:03.38 | SirWhit | Is there a limit to the number of simulataneous conference rooms you can have? |
| 19:03.38 | [TK]D-Fender | asteriskmonkey: use a streaming music source that you have a way to do that to |
| 19:03.39 | Yourname`` | exten=> 6,n,Dial(SIP/100,,) -> Does this look ok? |
| 19:03.41 | Yourname`` | There we go. |
| 19:03.49 | |Rain| | is it possible to have switch() match an empty string? the closest I've come so far, short of quoting everything, is 'case -', but that doesn't work since the default case is shorter (sw-73- vs. sw-73--) and matches first |
| 19:03.51 | mvanbaak | Yourname``: way better |
| 19:03.55 | [TK]D-Fender | Yourname``: better, but no need for trailing ","'s |
| 19:04.07 | polerin | [TK]D-Fender: define box.. you're talking about the phone/softphone, or the server. |
| 19:04.08 | Yourname`` | Ok, thank you guys. :) |
| 19:04.20 | [TK]D-Fender | polerin: VOICEMAIL box # |
| 19:04.26 | polerin | oh |
| 19:04.27 | polerin | heh |
| 19:04.30 | polerin | duh ok :) |
| 19:04.31 | *** join/#asterisk [N00B] (n=ckwall@206.71.78.158) |
| 19:04.32 | Yourname`` | [TK]D-Fender: Quick dialplan question, I have a use for 12345 in [context1], I have the same use for 12345 in [context2].. there isn't any way they will get mixed, right? Both these extensions are supposed to add an agent to a queue. Except, they add the agent to two different queues. |
| 19:04.55 | [N00B] | are the only options for voice mail file format wav49|gsm|wav? Or can I add mp3 to that also? |
| 19:05.05 | [TK]D-Fender | Yourname``: look what CONTEXT is being used. nothing simply dials an extension. It is dailed against a CONTEXT. Go verify WHICH. |
| 19:05.23 | Yourname`` | [TK]D-Fender: The respective contexts are being used. |
| 19:05.27 | mvanbaak | [N00B]: no mp3 support for voicemail |
| 19:05.30 | [TK]D-Fender | [N00B]: no MP#. You can record in and codec you have a "codec_XXX.so" for. |
| 19:05.32 | [N00B] | ok, thanks |
| 19:05.40 | [TK]D-Fender | any* |
| 19:06.06 | [TK]D-Fender | Yourname``: "respective contexts"? Meaningles term. Go look at the exacty device added. |
| 19:06.06 | mvanbaak | [N00B]: but you can run a command when voicemail is left, so you can create some shellscript that runs lame on it |
| 19:06.21 | mvanbaak | but you cant listen to your vm using VoiceMailMain after that |
| 19:06.23 | polerin | mvanbaak: yeah I was about to ask.. thought you could do it ;) |
| 19:06.29 | Yourname`` | ok |
| 19:06.33 | [TK]D-Fender | mvanbaak: Sure you can. |
| 19:06.36 | polerin | mvanbaak: only if you delete the orginal :P |
| 19:06.43 | [TK]D-Fender | mvanbaak: * can still play back the MP3 |
| 19:06.49 | mvanbaak | using voicemailmain ? |
| 19:06.52 | polerin | or.. that :P |
| 19:07.04 | [TK]D-Fender | mvanbaak: Every file that * plays goes through the same selection engine |
| 19:07.09 | [N00B] | its so that my lame marketing people can listen to their voicemail on their iPhones. not worth the effort for me. |
| 19:07.10 | mvanbaak | ah, right |
| 19:07.19 | [N00B] | they cannot figure out how to do wav on the iphone |
| 19:07.22 | [TK]D-Fender | mvanbaak: If you can play 1 MP3 anywhere, you can do them everywhere |
| 19:07.28 | mvanbaak | brb, food |
| 19:07.44 | [TK]D-Fender | [N00B]: wav is perfectly playable by pretty much every platform. |
| 19:07.52 | [TK]D-Fender | [N00B]: there should be no need for MP3 |
| 19:07.56 | [N00B] | my thoughts exactly |
| 19:08.10 | [N00B] | but they cant figure it out on the iphone... which is not my problem. |
| 19:08.26 | [N00B] | i just figured if it was as easy as adding mp3 to the format= line i would do it. |
| 19:08.27 | [N00B] | but its not. |
| 19:08.29 | [N00B] | thanks all |
| 19:08.31 | *** join/#asterisk TedNJ38 (n=HungLad@ool-4573adc7.dyn.optonline.net) |
| 19:08.33 | Qwell | [N00B]: wav, or wav49? |
| 19:08.37 | Qwell | wav should work everywhere |
| 19:08.39 | [N00B] | wav |
| 19:08.49 | polerin | [TK]D-Fender: web streaming wav? yech. |
| 19:09.18 | Qwell | wav49 might be worth a shot |
| 19:09.18 | *** part/#asterisk [N00B] (n=ckwall@206.71.78.158) |
| 19:09.25 | [TK]D-Fender | polerin: Who said anything about streaming WAV? |
| 19:09.26 | Yourname`` | [TK]D-Fender: http://pastebin.com/d22d06cae -> a few snippets.. |
| 19:09.41 | TedNJ38 | I have a question guys. How may I tweak the attributes of a specific extension that don't show up in the web interface (such as language=)? If I change them in sip_additional they get overwritten eventually. |
| 19:10.18 | polerin | [TK]D-Fender: You said there is no need for mp3, and the common thing I can think of for mp3 would be to stream it from a web interface |
| 19:11.12 | [TK]D-Fender | Yourname``: exten=> 4,1,AddQueueMember(frontier) <- this adds the CHANNEL device, not an EXTENSION <-- |
| 19:11.32 | [TK]D-Fender | Yourname``: So if you're calling from SIP/100 well it adds SIP/100, not some sort of exten that LEADS to it. |
| 19:11.53 | _x86_ | ok, I had a station (analog ZAP FXS station) that was working just fine making outbound calls all day |
| 19:11.59 | [TK]D-Fender | polerin: You never said HOW the iPhone was to pick up this "voicemail in the first place. |
| 19:12.03 | Yourname`` | [TK]D-Fender: Won't be calling FROM SIP/100 though |
| 19:12.28 | [TK]D-Fender | Yourname``: Show CLI of the device logging in, a queue dump as well. |
| 19:12.31 | _x86_ | now all of a sudden, that station has really bad static, to the point where you can hardly even hear dial tone |
| 19:12.52 | _x86_ | it is the only station affected, and we've tried multiple phones on the same port, all had the same problem |
| 19:12.57 | *** part/#asterisk |Rain| (i=rain@2001:440:eeee:fffb:42:0:0:2) |
| 19:12.58 | _x86_ | what could be causing this? |
| 19:13.01 | [TK]D-Fender | TedNJ38: .... |
| 19:13.03 | [TK]D-Fender | ~freepbx |
| 19:13.04 | jbot | [~freepbx] FreePBX is unable to be supported here. It is made up of complex dialplans and scripts which can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking there |
| 19:13.05 | [TK]D-Fender | ^^^^^^^^^^ |
| 19:13.20 | Yourname`` | [TK]D-Fender: Haven't set it up yet. Just wondering if an agent dials 4 if asterisk will confused and use [cleanserq] or [queue-test].. that's all. |
| 19:13.41 | polerin | [TK]D-Fender: actually that was n00b, I just said streaming. **shrugs** anyway. |
| 19:13.49 | Yourname`` | And ofcourse the agent is context=cleanserq |
| 19:14.06 | [TK]D-Fender | Yourname``: AddQueueMember(queuename[|interface[|penalty[|options[|membername]]]]): <--- this has nothing to do with using your DIALPLAN to call the agent. |
| 19:14.17 | [TK]D-Fender | Yourname``: the way YOU'RE calling it anyways. |
| 19:14.41 | Yourname`` | oh ok |
| 19:15.12 | [TK]D-Fender | Yourname``: exten=> 4,1,AddQueueMember(frontier) <- this will look at the device placing that call. |
| 19:15.50 | [TK]D-Fender | Yourname``: Oh yeah and... exten => 54321,1,Set(CALLERID(num)=${CALLERID(num)}) <--- VERY useful.... |
| 19:16.59 | Yourname`` | LOL |
| 19:17.09 | Yourname`` | Just overlooked that one, not too purposeful anyway, hehe |
| 19:17.45 | *** join/#asterisk teknoprep (n=teknopre@unaffiliated/teknoprep) |
| 19:17.47 | teknoprep | heya ll |
| 19:17.49 | *** join/#asterisk csm4ch (i=csm4ch@gateway/tor/x-41bf582f68765381) |
| 19:17.52 | Yourname`` | I wanted to do (ALL), but then ?I thought it'd set both num and name. And all I want is the name to be custom, so I put the num to the way it is.. and used name to the custom name |
| 19:17.53 | teknoprep | where do i get nvfax support from? |
| 19:17.59 | teknoprep | is this something i have to compile into asterisk? |
| 19:18.52 | *** join/#asterisk angom (n=angom@200.38.31.239.dsl.dyn.telnor.net) |
| 19:20.28 | [TK]D-Fender | Yourname``: You mean "changed its contents to... what it already contains!". |
| 19:20.55 | [TK]D-Fender | teknoprep: Yes, its 3rd part. Go search the WIKI for it. |
| 19:21.05 | Yourname`` | lol |
| 19:21.10 | Yourname`` | yes yes, I'll change it. |
| 19:21.48 | [TK]D-Fender | Yourname``: *remove* |
| 19:22.18 | Yourname`` | Yes sir, remove it is! |
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| 19:24.37 | *** part/#asterisk TedNJ38 (n=HungLad@ool-4573adc7.dyn.optonline.net) |
| 19:25.01 | teknoprep | [Nov 29 14:24:19] WARNING[4127] pbx.c: No application 'NVFaxDetect' for extension (from-pstn, 14843351444, 7) |
| 19:25.13 | teknoprep | i do have app_rx and txfax.so |
| 19:26.14 | *** join/#asterisk dushantch (n=chatzill@adsl-ppp-0237.yubc.net) |
| 19:26.39 | [TK]D-Fender | teknoprep: that has nothing to do with SpanDSP and is from a completely different author |
| 19:27.16 | teknoprep | hmmm |
| 19:28.03 | RoyK | are there any works in progress making t38 support for * apart from passthrough mode? |
| 19:32.34 | *** join/#asterisk philippel (n=p_lindhe@c-98-203-245-82.hsd1.wa.comcast.net) |
| 19:32.50 | polerin | btw, ASTDB the only way to use the blacklist command? |
| 19:34.32 | bkruse | RoyK: im not sure what I can say |
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| 19:34.37 | bkruse | but im pretty sure thats a yes :] |
| 19:35.36 | RoyK | bkruse: i've heard it's been in the works for a couple of years :P |
| 19:35.56 | bkruse | well, just recently some more talk and interest in it |
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| 19:38.49 | pdd_ | does anybody know how to execute Dial() synchronously? iow, I don't want the context to keep executing until the call is answered |
| 19:38.49 | [TK]D-Fender | polerin: "Blacklist" is a dead feature. This is really up to you to implement and it is remarkably easy via AstDB. |
| 19:39.14 | [TK]D-Fender | pdd_: It won't All dialplan execution for a channel is linear. |
| 19:39.23 | *** join/#asterisk beighto (n=chatzill@12.176.156.130) |
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| 19:40.13 | pdd_ | is there any way to accomplish that? like creating a loop with Goto reading the call status perhaps? |
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| 19:41.34 | pdd_ | to me it seems odd that this is the default behaviour... how do you design your dial plan to play a message after the call is answered? |
| 19:41.47 | docelm0 | Can someone tell me what this means? |
| 19:41.48 | docelm0 | Nov 29 14:38:12 WARNING[5712]: chan_sip.c:1084 __sip_xmit: sip_xmit of 0x8165480 (len 485) to XXX.XXX.XXX.XXX:5060 returned -1: Bad file descriptor |
| 19:42.34 | docelm0 | And most importantly how to resolve it? |
| 19:44.27 | pdd_ | check your sip.conf bindaddr |
| 19:44.28 | docelm0 | pdd_ just resolved it.. Thanks.. |
| 19:44.28 | pdd_ | what was it? |
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| 19:51.10 | beighto | When dialing * on my Polycom phones, the phone sends the request to Asterisk and attempts to Dial *. So when I dial some number followed by * it attempts to dial that number followed by a * and won't let me dial anything after the *. Does anybody know a fix for this? |
| 19:52.34 | Qwell | beighto: fix the dialplan in your polycom |
| 19:54.02 | *** join/#asterisk tobias (n=tobias@user-0ce2hr3.cable.mindspring.com) |
| 19:54.07 | *** join/#asterisk _charly_ (i=kroseneg@sunrise.schmidham.net) |
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| 19:54.35 | *** mode/#asterisk [+o russellb] by ChanServ |
| 19:54.52 | [TK]D-Fender | pdd_: "show application dial" <- for what yuo can do upon answer. |
| 19:55.33 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 19:56.35 | beighto | qwell: What file has the Polycom dialplan? |
| 19:58.04 | *** join/#asterisk dutch_ (n=dutch_@195-144-082-138.dyn.adsl.xs4all.be) |
| 19:58.27 | *** join/#asterisk Nenillo (n=c00l@125.Red-80-34-5.staticIP.rima-tde.net) |
| 19:58.30 | Nenillo | hi |
| 19:58.32 | [TK]D-Fender | beighto: typically sip.cfg |
| 19:58.36 | blitzrage | beighto: you'll need to grab the documentation from Polycom website to understand what to change anyways |
| 19:58.49 | [TK]D-Fender | beighto: unless over-ridden on the the phone level config |
| 19:58.54 | dutch_ | any asterisk freebsd users around? |
| 19:59.33 | beighto | time to do some research... I have a dozen of these phones so I don't think I want to do it phone level. Thanks |
| 19:59.49 | Nenillo | I want to make a simple demo with asterisk to make calls in my LAN with SIP phones, is there any doc explaining how to do something similar to that? |
| 20:00.10 | blitzrage | ~book |
| 20:00.11 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf |
| 20:00.15 | blitzrage | that'll get you started |
| 20:00.17 | Nenillo | I don't want out calls... |
| 20:00.21 | Nenillo | mmm |
| 20:01.03 | Nenillo | well, I see is not so easy as I thought... |
| 20:01.14 | blitzrage | asterisk isn't easy.... |
| 20:01.21 | blitzrage | see freepbx for a simple demo |
| 20:01.27 | blitzrage | (well... not simple... but ya) |
| 20:01.49 | blitzrage | your demo isn't hard to build -- but you need to know how asterisk works first |
| 20:02.17 | Nenillo | yeah... that's the problem |
| 20:02.40 | Nenillo | thanks for the help, I'll look at these resources |
| 20:03.02 | [TK]D-Fender | Nenillo: here : |
| 20:03.04 | [TK]D-Fender | ~jerjerguide |
| 20:03.05 | jbot | [~jerjerguide ] Jeremy McNamara's quick sample guide to setting up * : http://www.jeremy-mcnamara.com/2007/02/26/how-to-configure-asterisk-your-first-installation/ |
| 20:03.26 | Nenillo | thanks! |
| 20:03.34 | lirakis | can i goto a pattern priority in a context...? like Goto(testing,_XXX,1) ? |
| 20:03.49 | blitzrage | lirakis: that makes no sense |
| 20:03.59 | blitzrage | lirakis: I think you want ${EXTEN} |
| 20:04.08 | blitzrage | [TK]D-Fender: ah yes... that guide |
| 20:04.14 | *** join/#asterisk muiro (n=muiro@cpe-76-189-84-108.neo.res.rr.com) |
| 20:05.21 | lirakis | blitzrage: for my i priority.. i want to go to the "start" of my testing context... which begins with a pattern "exten => _XX,1,...." |
| 20:06.23 | lirakis | so .. its an invalid ${EXTEN} .. that they entered... but there is no prompt for the entry... like an exten => s,1,Background(msg) |
| 20:07.36 | blitzrage | lirakis: store the value of ${EXTEN} prior to jumping to i |
| 20:07.47 | blitzrage | then use Goto(testing,${MY_EXTEN},1) |
| 20:07.59 | blitzrage | _XXX,n,Set(MY_EXTEN=${EXTEN}) |
| 20:08.39 | lirakis | blitzrage: ... yeah .. i know.. but they got to i by entering in ${EXTEN} .. so MY_EXTEN would then be bad and not match any pattern... nm .. i need to rework this bit |
| 20:08.58 | lirakis | thanks though |
| 20:09.00 | philippel | anyone ever get "No such Label ..." when using labels in goto branches? The reported error is very clear. The bizare thing is unquestionaly the label is there and defined (show dialplan ...) and changing the label to a different label still fails even if that label different label works from elsewhere in the same dialplan? |
| 20:09.03 | blitzrage | lirakis: huh? you do MY_EXTEN before you even get to 'i' |
| 20:09.12 | lirakis | blitzrage: ha ha no.. dont worry about it |
| 20:09.28 | blitzrage | lirakis: I wasn't worried -- you're probably making it more complicated than necessary -- but go nuts |
| 20:09.36 | blitzrage | philippel: sounds like a typo |
| 20:09.49 | philippel | I can absolutely guarantee it is not |
| 20:09.57 | blitzrage | philippel: need to see the dialplan in question |
| 20:10.07 | blitzrage | you're not doing this from 'h' or a macro are you? |
| 20:10.44 | blitzrage | philippel: please put in a pastebin |
| 20:11.22 | philippel | blitzrage I appreciate it, I don't want to take your time for something that is not a typo, plus I'm in a slightly older asterisk version and would not want it to be something that is already fixed |
| 20:11.34 | blitzrage | philippel: what is 'slightly older' ? |
| 20:11.48 | philippel | 1.2.10 with a few minor security patches installed |
| 20:11.56 | blitzrage | oh ya -- shouldn't be an issue |
| 20:12.08 | blitzrage | at least I don't think so... :) |
| 20:12.38 | blitzrage | philippel: I can read dialplan pretty quick -- if you pastebin the matching dialplan along with the console output, I can at least verify it's not a typo |
| 20:12.49 | philippel | that's what I woudl suspect, however you never know - this is one of the more bizarre one's I have seen. And it is related to the specific goto line |
| 20:12.50 | e` | Ok so i'm trying to set while an extension is ringing, if you press 0, it will go directly to that extens voicemail. I know i can use the Dial cmd with the 'd' option, but not sure on the correct syntax |
| 20:12.52 | blitzrage | I'll get ready for the gym, then check back |
| 20:13.06 | blitzrage | philippel: maybe I can see something you haven't spotted |
| 20:13.08 | philippel | blitzrage I'll pastebin it for grins |
| 20:13.54 | *** join/#asterisk seba_soy (n=chatzill@190.2.63.135) |
| 20:14.05 | seba_soy | Hello... |
| 20:14.13 | seba_soy | some clue about this? |
| 20:14.14 | seba_soy | <PROTECTED> |
| 20:15.12 | blitzrage | e`: the description is pretty clear: Set(EXITCONTEXT=context_i_wanna_match_in) |
| 20:15.21 | blitzrage | [context_i_wanna_match_in] |
| 20:15.30 | blitzrage | exten => 0,1,Voicemail(100@default) |
| 20:15.33 | blitzrage | <PROTECTED> |
| 20:15.45 | blitzrage | [context_i_call_from] |
| 20:15.54 | blitzrage | exten => _X.,1,NoOp() |
| 20:16.08 | blitzrage | exten => _X.,n,Set(EXITCONTEXT=....) |
| 20:16.19 | blitzrage | exten => _X.,n,Dial(SIP/foobar,30,d) |
| 20:19.11 | asteriskmonkey | its there a way to park an incomming call using dial plan in 1.2x and get a variable back from it so you can then pick it up the call later using the variable? |
| 20:21.03 | blitzrage | asteriskmonkey: sorry, no such functionality that I'm aware of |
| 20:21.22 | blitzrage | that was discussed in a conference call recently actually -- not even sure there is a way to do that now |
| 20:22.12 | *** join/#asterisk RipeR-81 (n=ircap8@190.53.33.3) |
| 20:22.18 | asteriskmonkey | I can see using dynamic conferencing and having a set of diaplan logic that wraps around that concept but that would be awefully sloppy |
| 20:22.48 | RipeR-81 | hi everyone im shopping for a PSTN Gateway... which one can you reccomend me? |
| 20:22.56 | blitzrage | asteriskmonkey: parking was never designed to be used without it being attended-transfered |
| 20:23.15 | asteriskmonkey | RipeR-81: an asterisk box with a digium card |
| 20:23.29 | philippel | blitzrage: http://pastebin.com/d32f6c45d |
| 20:23.39 | blitzrage | asteriskmonkey: the way you mentioned above is probably gonna be your best bet... I think I used a Local channel though that called MusicOnHold() |
| 20:23.42 | philippel | blitzrage I just pulled out the highlights |
| 20:23.42 | RipeR-81 | asteriskmonkey i have a asterisk server as a local PBX... but im not in the states |
| 20:23.56 | RipeR-81 | asteriskmonkey i need a PSTN service to route my calls thru internet... |
| 20:24.16 | asteriskmonkey | blitzrage, could always use chan local to tranfer someone to a context that handles dynamic queing and pickup would be essential joining backinto the dynamic conference :P |
| 20:24.25 | *** join/#asterisk tobias (n=tobias@user-0ce2hr3.cable.mindspring.com) |
| 20:24.28 | blitzrage | philippel: try taking the spaces out from after the ? in your GotoIf() |
| 20:24.49 | blitzrage | asteriskmonkey: that sounds like more layers than I did :) |
| 20:24.58 | asteriskmonkey | RipeR-81: what country/city |
| 20:25.16 | asteriskmonkey | blitzrage: oh do tell what you did :) |
| 20:25.28 | philippel | blitzrage I did |
| 20:25.33 | blitzrage | philippel: that version you're running is old enough that asterisk is probably quite sensitive to the spaces.... |
| 20:26.03 | blitzrage | asteriskmonkey: let me see if I can find a snippet... |
| 20:26.36 | [TK]D-Fender | Nov 29 12:15:16 NOTICE[26860] pbx.c: No such label ' a30 ' in extension 's' in context 'macro-dial' <--- spaces = BAD |
| 20:26.58 | Yourname`` | ~ports |
| 20:26.59 | jbot | somebody said ports was http://www.debian.org/ports/, or http://www.isi.edu/in-notes/iana/assignments/port-numbers, or the FreeBSD ports system etc etc, or http://www.portforward.com/routers.htm |
| 20:26.59 | philippel | blitzrage plus the revealing thing is that Asterisk knows the correct label it is trying to go to (from the log) and it is there and yet ..., but let me take a look at the spaces again even though I did as I have spent hours in the past just to find out it is sensitive to spaces sometimes |
| 20:27.18 | Yourname`` | jbot: I meant what ports need to be forwarded for softphone or ATA to have a proper call :( |
| 20:27.19 | jbot | You meant what ports need to be forwarded for softphone or ATA to have a proper call :(? |
| 20:27.30 | Yourname`` | Yes jbot |
| 20:27.32 | Titanous | How do I compile a asterisk app without recompiling asterisk? |
| 20:27.59 | *** join/#asterisk _charly_ (i=kroseneg@sunrise.schmidham.net) |
| 20:29.45 | blitzrage | asteriskmonkey: hrmmm... this might not be useful for you... as I used func_odbc to store the value into a database... but you could rip that out and use astDB I suppose |
| 20:30.35 | asteriskmonkey | blitzrage: great thanks, i can easily rewrite it, just need to wrap my head around possible solutions :/ |
| 20:30.36 | philippel | blitzrage I guess I did not try enough space combos, it was the final trailing space (all the other it was happy with):) |
| 20:30.44 | beek | Titanous: jbot is a 'bot. It's simply repeating what you asked. |
| 20:31.27 | ManxPower | philippel: don't ever add spaces you do not need. |
| 20:31.32 | ManxPower | it causes stuff like you are experiencing |
| 20:32.13 | philippel | ManxPower thank, I know, I've run into it - I was just converting this code from number to labels, so was just doing a replace. It has bit me before to the tune of many hours of troubleshooting |
| 20:32.36 | Titanous | How do I compile a asterisk app without recompiling asterisk? |
| 20:32.49 | blitzrage | asteriskmonkey: http://pastebin.com/d37cde130 |
| 20:32.58 | Titanous | s|app|app/module| |
| 20:33.06 | ManxPower | Titanous: I doubt anyone here knows, as we always just recompile asterisk. It's not like it's going to break anything, as long as you use them same code you originally installed Asterisk with |
| 20:33.10 | blitzrage | philippel: yes -- asterisk 1.2 is very sensitive to spaces |
| 20:33.30 | asteriskmonkey | blitzrage: thanks :) |
| 20:33.32 | blitzrage | Titanous: astxs I think |
| 20:33.51 | blitzrage | Titanous: search the sources for that file -- it's a script I think to compile individual modules |
| 20:33.51 | Titanous | ManxPower, I have no problem recompiling asterisk, It's just I'm not sure how to do that without taking my phone system offline for the duration of the compile |
| 20:34.04 | ManxPower | Titanous: you do not need to stop asterisk. |
| 20:34.06 | [TK]D-Fender | Titanous: You don't have to. |
| 20:34.07 | Titanous | blitzrage, thanks |
| 20:34.11 | ManxPower | you can compile and install Asterisk while it is running. |
| 20:34.13 | philippel | blitzrage thanks for taking a look |
| 20:34.16 | Titanous | ok, I didn't no that |
| 20:34.20 | Titanous | *know |
| 20:34.27 | Titanous | thanks alot |
| 20:34.30 | blitzrage | a lot* |
| 20:36.29 | *** join/#asterisk Ryushin (i=proxy@windwalker.openinnovations.com) |
| 20:39.13 | *** join/#asterisk atisss (n=atisss@193.238.212.171) |
| 20:46.44 | *** join/#asterisk ELBunce (n=erik@kde/developer/bunce) |
| 20:49.40 | *** join/#asterisk grandpapadot (n=null@mail.heavylogic.com) |
| 20:49.51 | grandpapadot | Hi all. What would be the best digium hardware to terminate a single PRI into an asterisk box? |
| 20:50.19 | *** join/#asterisk tdonahue-laptop (n=tdonahue@vonmail.vonworldwide.com) |
| 20:50.22 | tdonahue-laptop | hi all |
| 20:52.31 | SwK | anyone need any good TFs cheap? |
| 20:52.33 | SwK | msg me |
| 20:52.57 | [TK]D-Fender | TFs? |
| 20:53.04 | SwK | TollFrees US/Canadia |
| 21:00.37 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 21:03.15 | RipeR-81 | asteriskmonkey sorry for the delay... im located at managua/Nicaragua |
| 21:04.56 | RipeR-81 | asteriskmonkey i was reading networkjunction |
| 21:07.45 | *** join/#asterisk supjigator (n=sysgod@152.53.16.10) |
| 21:07.54 | supjigator | Anyone using a MAXTNT? |
| 21:14.53 | tdonahue-laptop | how can i get the DTMF input from an extension so I can insert it into a AstDB key? |
| 21:15.57 | *** join/#asterisk darkskiez (n=mhb@cpc1-broo3-0-0-cust659.renf.cable.ntl.com) |
| 21:19.02 | [TK]D-Fender | tdonahue-laptop: "show application read" |
| 21:21.14 | tdonahue-laptop | [TK]D-Fender, thanks, from the short description "Read a variable" i didn't realize that had anything to do with DTMF |
| 21:21.16 | *** join/#asterisk JT (n=j@unaffiliated/jt) |
| 21:23.34 | *** join/#asterisk r0d3nt (i=nobody@foster.stonedcoder.org) |
| 21:26.28 | *** join/#asterisk roe_ (n=roe___@216-164-160-36.c3-0.eas-ubr10.atw-eas.pa.static.cable.rcn.com) |
| 21:27.30 | roe_ | I'm trying to properly size a asterisk deployment, it will have about 50 users (all voIP/sip) and a T-1 trunk |
| 21:27.30 | bkruse | roe_: full t1 voice? |
| 21:27.54 | roe_ | I'm not entirely sure yet will find out tomorrow, I have no problem assuming yes for the time being |
| 21:28.19 | *** join/#asterisk ak_hepcat (n=ak_hepca@patroler.gci.net) |
| 21:28.24 | bkruse | roe_: that would be a simple system with a 1 port (te1xxp) |
| 21:28.28 | bkruse | is it a call center ? or office. |
| 21:28.33 | roe_ | office |
| 21:29.03 | *** part/#asterisk asteriskmonkey (n=philip@69.77.169.14) |
| 21:29.37 | roe_ | I'm concerned about horse power and reliability |
| 21:29.49 | roe_ | a phonesystem can't go down, especially for this size deployment |
| 21:30.36 | roe_ | while commodity hardware is inexpensive, it isn't known for its mtbf |
| 21:30.45 | *** join/#asterisk Koshatul (n=evangeli@ppp188-33.static.internode.on.net) |
| 21:30.55 | blitzrage | roe_: sounds like you're going to have to come up with a clustering/failover strategy |
| 21:31.16 | blitzrage | any single box deployment will go down at some point |
| 21:31.25 | blitzrage | (imho) |
| 21:31.44 | roe_ | perhaps, but that is secondary, how can I properly anticipate load, am I looking at 2cores 4cores 8cores? |
| 21:31.45 | _Sam-- | i think that depends on how often you obsolete your hardware. |
| 21:31.57 | blitzrage | roe_: use sipp and load test a box |
| 21:32.24 | blitzrage | roe_: are you transcoding? are you going to be doing conferencing? how many simultaneous calls? |
| 21:32.29 | blitzrage | those are all questions you need to ask |
| 21:32.31 | roe_ | I can do that, and I have to a small extent, I was wondering if anyone here has a system of that size already |
| 21:32.32 | blitzrage | (and answer) |
| 21:32.42 | blitzrage | roe_: yes -- but every system is different |
| 21:33.13 | blitzrage | 2GB RAM, 2x Xeon processors will probably* do what you want |
| 21:33.28 | blitzrage | probably around a Dell 1950 or so |
| 21:33.45 | roe_ | ok, I'll start there and extrapolate my specifics, thanx. |
| 21:36.01 | *** join/#asterisk oej (n=olle@85.19.87.179) |
| 21:36.02 | *** part/#asterisk lirakis (n=lirakis@65.200.191.253) |
| 21:37.42 | *** join/#asterisk jameswf (n=james@dsl093-157-131.phx1.dsl.speakeasy.net) |
| 21:39.48 | *** join/#asterisk atisss (n=atisss@193.238.212.171) |
| 21:41.33 | *** join/#asterisk edwin_quijada (n=m@25.116.88.200.m.sta.codetel.net.do) |
| 21:41.59 | edwin_quijada | what is the best language for AGI php o perl? |
| 21:43.47 | polerin | edwin_quijada: which do you know better, and what exactly are you trying to do with it. **shrugs** |
| 21:43.51 | polerin | every tool |
| 21:47.37 | *** join/#asterisk oej (n=olle@85.19.87.179) |
| 21:48.17 | edwin_quijada | polerin: perl |
| 21:48.38 | edwin_quijada | I am trying to do a IVR with conection to DB |
| 21:49.35 | polerin | then perl it is |
| 21:58.38 | *** join/#asterisk GoldFingaZ (n=wt5@bas13-ottawa23-1177808409.dsl.bell.ca) |
| 22:03.26 | *** topic/#asterisk by russellb -> Asterisk: The Open Source Telephony Application Platform (asterisk.org) -=- Asterisk 1.4.15 (2007/11/29), *-Addons 1.4.4 (2007/10/16), Zaptel 1.4.6 (2007/10/18), Libpri 1.4.2 (2007/10/16) -=- #asterisknow or #asterisk-gui for AsteriskNOW (asterisknow.org) -=- #switchvox for Switchvox (switchvox.com) -=- #asterisk-commits to monitor svn -=- #freepbx (freepbx.org) or #trixbox for trixbox (trixbox.org) support |
| 22:04.30 | *** join/#asterisk ThatKidKel (n=Kelvin@66.236.241.67.ptr.us.xo.net) |
| 22:04.48 | ThatKidKel | Any recommendations for a cordless SIP phone, 802.11 not required... |
| 22:07.57 | [hC] | So has anyone looked into asterisk supporting UMA? |
| 22:12.05 | xrg_ | UMA as in "vendor-lock phones to GSM providers" ? |
| 22:13.05 | [hC] | I guess. I learned of it 5 minutes ago. I see that the new blackberry 8820 does UMA and not SIP for voip via wifi |
| 22:13.35 | xrg_ | That is it, then.. |
| 22:13.49 | [hC] | UMA traverses GSM only? or can it also go via wifi? |
| 22:13.59 | Qwell | [hC]: I have |
| 22:14.11 | Qwell | [hC]: it would be...non trivial |
| 22:14.14 | Qwell | BUT, possible |
| 22:14.36 | *** join/#asterisk Yourname`` (n=Miranda@unaffiliated/yourname/x-837320) |
| 22:14.47 | Yourname`` | Hi. Does anyone know how to make express talk auto answer? |
| 22:14.48 | [hC] | Qwell: to make UMA work? Im just trying to find the killer solution which is voip in a blackberry, thats all :) |
| 22:15.06 | Qwell | chan_uma? |
| 22:15.10 | Qwell | yeah, non-trivial |
| 22:15.15 | *** join/#asterisk kand (n=kanderso@139.148.188.72.cfl.res.rr.com) |
| 22:15.35 | Qwell | but what you want is asterisk as the server side? |
| 22:15.55 | [hC] | Qwell: yes, so i can place voip calls to an internal enterprise pbx |
| 22:16.12 | Qwell | from the blackberry, over uma, to asterisk? |
| 22:16.13 | kand | hey, I have a SPA-2102 that can make calls but when you answer it the ata sends a bye immediatly |
| 22:16.15 | [hC] | If it makes more sense to put openser, freeswitch, or something similar in front rather than rewrite it, then so be it i guess. |
| 22:17.06 | [hC] | Qwell: Yeah, this new blackberry that came out can do wifi and edge, and its "voip client" supports UMA only. In my ideal world, employees could connect to the company pbx via wifi when in the office or in range of wifi, or call over gsm if not |
| 22:17.15 | kand | Anybody have a sugestion? |
| 22:17.29 | Qwell | [hC]: I doubt you'd even be able to change the IP that it tries to connect to for the UMA server |
| 22:17.38 | [hC] | but it would provide truphone-like cheap/free LD, or provide internal dialing capabilities, and presumably enhanced presence behavior |
| 22:17.54 | [hC] | Qwell: with it connected to a BES server, it should be able to, i'd think |
| 22:18.01 | Qwell | BES server? |
| 22:18.06 | [hC] | Qwell: Blackberry Enterprise server gives you control over pretty much everything |
| 22:18.15 | Qwell | and I guess, worst case scenario, you could just iptables that IP away |
| 22:18.17 | [hC] | Qwell: conduit for tethering blackberries into your company 'domain' |
| 22:18.45 | Qwell | [hC]: somebody would have to write an LGPL GSM stack that asterisk can use :D |
| 22:18.46 | [hC] | Qwell: it links company exchange servers to blackberries and also serves as an IP gateway for data usage.. |
| 22:19.51 | *** join/#asterisk r0d3nt (i=nobody@foster.stonedcoder.org) |
| 22:20.11 | [hC] | Qwell: hmm. I dont really understand how uma works... but it is feasible that asterisk could do it? |
| 22:20.59 | Yourname`` | Ok, fine. What about this error-> [Nov 29 16:44:35] WARNING[32380]: translate.c:163 framein: no samples for ulawtolin |
| 22:21.13 | *** join/#asterisk Schumie (i=SteveWri@cpc1-rdng2-0-0-cust441.winn.cable.ntl.com) |
| 22:21.22 | *** part/#asterisk berniv6 (n=berni@fliwatuet.birkenwald.de) |
| 22:24.19 | Yourname`` | OR |
| 22:24.22 | Yourname`` | handle_response: Remote host can't match request NOTIFY to call |
| 22:25.47 | Qwell | [hC]: not sure - when I was researching it, I was thinking of Asterisk as the client |
| 22:26.02 | Qwell | it's "feasible", but it would take quite a bit of work |
| 22:26.28 | Qwell | as far as I could tell, asterisk would need to implement the GSM protocol (which extends SS7), and wrap that in UMA |
| 22:26.50 | Qwell | oh, and there's the whole problem of the AMR codec being patent encumbered |
| 22:31.40 | kand | Can someone help me configure a spa2102? It can make calls fine but when you answer a ringing call it hangs up within 5ms and the other party hears a fast busy. |
| 22:33.03 | *** join/#asterisk slam (n=slam@DCC.SEND.startkeylogger.000.telephreak.org) |
| 22:33.31 | slam | anybody here with time to help with a asterisk problem |
| 22:35.16 | TJNII | Don't ask to ask, just ask |
| 22:35.52 | slam | I have a problem with disa, it like it doesn't hear the dtmf tones |
| 22:36.17 | slam | any ideas |
| 22:37.02 | TJNII | Check how the device is sending the DTMF tones |
| 22:37.31 | slam | ok, call me stupid but how do I do that |
| 22:37.43 | *** join/#asterisk RoyK (n=roy@ip-135-16-149-91.dialup.ice.no) |
| 22:38.43 | slam | how many ways can it send them? |
| 22:39.09 | slam | dtmf is dtmf I thought |
| 22:39.29 | Corydon76-vcch | inband, SIP INFO, and RFC 2833 |
| 22:39.34 | slam | it obviously hears them |
| 22:39.41 | slam | its IAX2 |
| 22:39.53 | TJNII | slam: What kind of device are you using? |
| 22:39.54 | Corydon76-vcch | Oh, there's only one way on IAX2 |
| 22:39.57 | slam | inband etc. is not vadid |
| 22:40.14 | slam | valid |
| 22:40.23 | RoyK | does iax support other methods than inband? |
| 22:40.35 | russellb | iax only supports one DTMF method |
| 22:40.38 | slam | I have been told no |
| 22:40.38 | blitzrage | doesn't it only support it in signally? |
| 22:40.38 | russellb | and it's _not_ inband |
| 22:40.45 | blitzrage | signalling* |
| 22:40.53 | RoyK | btw, how do you pronounce IAX - Yaiks? |
| 22:41.05 | russellb | "eeks" |
| 22:41.05 | blitzrage | ya -- you can see the DTMF frames if you do an iax2 debug |
| 22:41.09 | blitzrage | "eeks" for me |
| 22:41.47 | slam | I have my box setup to ask for a password that it reconizes but it doesn't hear anything after that |
| 22:41.54 | RoyK | but that means asterisk must decode the dtmf and not send it over with the audio, right? |
| 22:42.46 | slam | I am lost |
| 22:42.56 | TJNII | slam: What is your setup here? What are you trying to do? |
| 22:42.59 | file | RoyK: if DTMF is received from something that does do inband, then it is always decoded and turned into a frame and not sent over the audio... so inband DTMF never reaches chan_iax2 |
| 22:43.02 | darkskiez | i just say I-A-X =] |
| 22:43.38 | *** join/#asterisk BCS-Satori (n=BCS-Sato@68.34.28.26) |
| 22:44.57 | RoyK | file: how does chan_sip do this with 'inband'? does asterisk decode and then encode the stuff? |
| 22:45.00 | BCS-Satori | Does anyone here use TCP instead of UDP for SIP, and could anyone explain to me the negative of doing so? Reason being I seem to loose hints in UDP. |
| 22:45.38 | RoyK | does * support sip over tcp these days? |
| 22:45.41 | TJNII | I don't think SIP supports TCP |
| 22:45.57 | RoyK | TJNII: SIP is supported over UDP, TCP and SCTP by standard |
| 22:46.11 | file | RoyK: DTMF coming in is sent through a DSP which decodes the DTMF and returns it so a frame can be created and sent through the core, for outgoing DTMF chan_sip sends back a return value to the core that says "hey, generate this inband for me" and the core happily does it |
| 22:46.15 | slam | I will be back, I have to go for now |
| 22:46.18 | slam | thanks |
| 22:46.26 | BCS-Satori | I know the Linksys phones support TCP and TLS but i figure there has to be a negitive to it |
| 22:46.41 | file | or rather, audio coming in is sent through |
| 22:46.41 | TJNII | RoyK: Well OK then |
| 22:46.55 | RoyK | file: wouldn't it be better if 'inband' dtmf was sent with the rtp as it was received? |
| 22:47.32 | *** join/#asterisk b_d (n=brian@209.240.42.151) |
| 22:47.44 | file | RoyK: what if the other side doesn't do inband? what if you are talking to an application? |
| 22:48.23 | _Sam-- | does random = > yes work on asterisk 1.2 MOH? |
| 22:48.26 | RoyK | file: well, if you're specifying inband, that should be your problem |
| 22:48.29 | TJNII | BCS-Satori, RoyK: Isn't UDP used to avoid the re-transmission delay times inherent in TCP? |
| 22:48.46 | RoyK | TJNII: yeah, same with SCTP |
| 22:49.06 | TJNII | Okay, then at least I had that much right. ;P |
| 22:49.29 | file | RoyK: ha... |
| 22:49.51 | RoyK | TJNII: only the retransmit stuff shouldn't matter with SIP as the audio is sent over RTP which goes over UDP |
| 22:50.33 | file | RoyK: how is Callweaver going? |
| 22:50.35 | RoyK | file: I was thinking like with SIP/PSTN gatewaying - you don't want the delay for asterisk to recode the stuff, would you? |
| 22:51.00 | file | RoyK: if DTMF is compatible between two SIP endpoints they will be bridged together and DTMF won't be decoded |
| 22:51.01 | RoyK | file: doing good, but there are some sqlite locking bugs that inhibits it to scale |
| 22:51.13 | file | well, if the codecs are compatible too... |
| 22:51.26 | RoyK | obviously :P |
| 22:51.32 | RoyK | but sip/pstn? |
| 22:51.37 | *** part/#asterisk ziimp (n=ziimp@web.fia-net.de) |
| 22:51.42 | RoyK | is that recoded? it seems like it.... |
| 22:51.46 | file | it is. |
| 22:51.57 | RoyK | imho it shouldn't |
| 22:51.59 | TJNII | RoyK: Oh, yea. the actuall sip signalling doesn't matter, but the RDP voice data stream does. Makes sense. |
| 22:52.38 | RoyK | file: any idea how to stop asterisk from recoding dtmf and have it just letting it through with the rtp stream? |
| 22:52.57 | *** join/#asterisk VijayG (i=VijayG@58.68.47.115) |
| 22:53.41 | file | RoyK: sure, change it so that generic bridging has the capability to see if a channel has/takes inband DTMF and if so tell the respective channels to not perform DTMF detection on it |
| 22:53.44 | TJNII | RoyK: Isn't that the "in-audio" option? |
| 22:54.09 | RoyK | TJNII: that's inband |
| 22:54.19 | file | but depending on the hardware it may perform the DTMF detection outside the scope of Asterisk, so you would have to make sure you could disable that as well |
| 22:55.15 | RoyK | file: I was more thinking of turning off the whole dtmf detection. on PRI/E1/PRA/something DTMF is transmitted inband anyway |
| 22:55.17 | VijayG | Hello, i have installed 1.2.10 asterisk on my server, my office people call on VOIP using Xlite soft phones, but the problem is many a times they are required to dial # to get connected to the right person they are calling, but as soon as we press # after the call is connected, soft phone just says transferred, looks like # is meant for transfer in asterisk, is there any way i can keep a control on this, i need this feature to be disabled, or any soft phone |
| 22:55.19 | VijayG | please update |
| 22:56.15 | file | RoyK: that is specific to what you want, what I said would potentially work for everyone |
| 22:57.01 | RoyK | file: what I just asked for was why recode it when pstn relies on audio anyway.... |
| 22:57.08 | RoyK | oh |
| 22:57.31 | RoyK | how exactly do I do that? |
| 22:57.37 | file | do what? |
| 22:57.53 | VijayG | <file> , you think if i make it as inband, my problem will also be resolved |
| 22:57.59 | RoyK | [23:53] <file> RoyK: sure, change it so that generic bridging has the capability to see if a channel has/takes inband DTMF and if so tell the respective channels to not perform DTMF detection on it |
| 22:58.32 | file | that I do not know without doing it |
| 22:58.33 | VijayG | looks good, looks like i got my answer |
| 22:58.43 | VijayG | let me test |
| 22:58.50 | VijayG | it should be working |
| 22:58.58 | file | VijayG: ...wha? I was talking to RoyK |
| 22:59.36 | RoyK | file: I'm just wondering why * should choose to recode something that should just be bridged like audio |
| 22:59.48 | RoyK | ISDN generally doesn't transmit DTMF OOB |
| 23:00.33 | file | maybe it shouldn't be, but nobody has brought it up and nobody has done it... and even then I wouldn't consider it that high a priority |
| 23:00.37 | *** part/#asterisk zerohalo (n=zeroHalo@pool-71-162-97-18.bstnma.east.verizon.net) |
| 23:00.43 | *** join/#asterisk ixx (i=foobar@cpe-70-112-149-89.austin.res.rr.com) |
| 23:00.55 | RoyK | and with recoding you get a delay that isn't necessarily what you want, meaning you need to tell the callie to preeeeeeeeeeeeessssssssssssss the right button |
| 23:01.06 | ixx | Anyone know what reason 8 is for a failed dial through a SIP provider? |
| 23:01.11 | file | delay? |
| 23:01.13 | *** join/#asterisk usam (n=usam@202.91.18.194) |
| 23:01.16 | ixx | I got a 480 try again... |
| 23:01.30 | ixx | what is strange is I dial the number and it says its invalid |
| 23:01.37 | file | DTMF in Asterisk 1.4 is two separate frames, a begin and an end so there shouldn't be any delay between start/finish |
| 23:01.38 | ixx | I was hoping reason 8 would lead to more info... |
| 23:01.51 | ixx | no need to try again when that number is not in service |
| 23:02.08 | usam | hey, seems like people are no longer interested in x100p cards any longer? can any confirm their status of uses of c100p cards in their countries? |
| 23:02.47 | VijayG | inband is not taking DTMF at all |
| 23:02.51 | VijayG | that didn't worked |
| 23:02.58 | VijayG | any other solution you can think ok |
| 23:03.06 | VijayG | we need to pass # as DTMF |
| 23:03.23 | VijayG | but it must not give the signal to transfer to * |
| 23:03.56 | russellb | don't enable transfer? |
| 23:04.21 | VijayG | how can i disable it |
| 23:04.31 | VijayG | Hello, i have installed 1.2.10 asterisk on my server, my office people call on VOIP using Xlite soft phones, but the problem is many a times they are required to dial # to get connected to the right person they are calling, but as soon as we press # after the call is connected, soft phone just says transferred, looks like # is meant for transfer in asterisk, is there any way i can keep a control on this, i need this feature to be disabled, or any soft phone |
| 23:04.57 | russellb | you just pasted that less than 10 minutes ago |
| 23:05.00 | *** mode/#asterisk [+b %VijayG!*@*] by russellb |
| 23:06.06 | file | VijayG: even though you can't speak... remove the t and T options from your Dial line |
| 23:06.29 | file | VijayG: and change back to rfc2833, oh and make sure silence suppression is disabled |
| 23:07.50 | Yourname`` | What are the best VoIP providers out there, who are reliable, almost all calls get through, and good call quality? |
| 23:08.49 | blitzrage | and now... I must eat |
| 23:10.28 | Yourname`` | Must! |
| 23:10.29 | *** join/#asterisk PoDunK (n=biga@207-255-210-182-dhcp.cbe.md.atlanticbb.net) |
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| 23:35.24 | *** mode/#asterisk [+o lmadsen] by ChanServ |
| 23:36.46 | nohup_ | hello |
| 23:37.19 | nohup_ | is there a variable (in extensions.conf) that holds the return code of an AGI script ? |
| 23:37.56 | lmadsen | return codes are not saved to a variable |
| 23:38.01 | lmadsen | they are just used internally in the code |
| 23:39.27 | nohup_ | so there's no way to make the 'flow' for that extension be altered according to the return code of a script ? |
| 23:41.16 | twisted | duh dun dun da da |
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| 23:45.43 | teknoprep | hey all |
| 23:46.07 | Yourname`` | A few odd errors if someone is willing to help a brotha out. |
| 23:46.13 | Yourname`` | [Nov 29 16:44:35] WARNING[32380]: translate.c:163 framein: no samples for ulawtolin |
| 23:46.30 | Yourname`` | [Nov 29 16:40:39] WARNING[26711]: chan_sip.c:12543 handle_response: Remote host can't match request NOTIFY to call '4a5042636c589e7c02c4f1906f1ba6e8@6x.xx.xx.xx'. Giving up. |
| 23:51.01 | *** topic/#asterisk by lmadsen -> Asterisk: The Open Source Telephony Application Platform (asterisk.org) -=- Asterisk 1.4.15 (2007/11/29), *-Addons 1.4.4 (2007/10/16), Zaptel 1.4.7 (2007/11/27), Libpri 1.4.2 (2007/10/16) -=- #asterisknow or #asterisk-gui for AsteriskNOW (asterisknow.org) -=- #switchvox for Switchvox (switchvox.com) -=- #asterisk-commits to monitor svn -=- #freepbx (freepbx.org) or #trixbox for trixbox (trixbox.org) support |
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