| 00:00.10 | test34 | jserve, that bug is still open |
| 00:01.16 | test34 | open -> not fixed |
| 00:01.29 | *** join/#asterisk Soul (n=Soul@87-196-6-86.net.novis.pt) |
| 00:04.46 | jserve | Ahh ok, then I have not overlook anything... |
| 00:07.42 | [TK]D-Fender | saftsack : What about them? |
| 00:08.05 | *** join/#asterisk bjohnson (n=bjohnson@i209-195-120-21.cia.com) |
| 00:09.28 | aptura | common reason why cli does not come up after verbose level 3? |
| 00:09.39 | aptura | all modules load. |
| 00:12.19 | *** part/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net) |
| 00:12.27 | saftsack | i want to ask if its possible to let the phones play another dialtone before first pressing the 0 |
| 00:12.58 | *** join/#asterisk ShadowHntr (i=sentinel@wikipedia/Shadowhntr) |
| 00:19.12 | *** join/#asterisk backblue (n=moo@87-196-11-204.net.novis.pt) |
| 00:28.32 | *** join/#asterisk BOLIVIAN (n=klkl@201.222.98.226) |
| 00:37.30 | *** join/#asterisk CrazyTux (n=CrazyTux@64.95.219.140) |
| 00:41.33 | *** join/#asterisk karmatronic (n=karmatro@84.77.151.228) |
| 00:51.40 | *** part/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 00:51.41 | *** join/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 00:52.16 | *** join/#asterisk lorinc (n=ang@caracas-0240.adsl.interware.hu) |
| 00:53.55 | *** join/#asterisk RoyK (n=roy@217-175-39.100710.adsl.tele2.no) |
| 00:56.02 | Bhaal_ | hey Qwell, I am in your fan club ;) |
| 00:56.03 | Bhaal_ | heh |
| 00:57.08 | *** join/#asterisk yassine (n=yassine@xdsl-87-78-22-204.netcologne.de) |
| 00:57.27 | *** join/#asterisk CrazyTux (n=CrazyTux@64.95.219.140) |
| 01:04.16 | *** join/#asterisk yassine (n=yassine@dsl.voicint.com) |
| 01:05.56 | *** join/#asterisk daveb_ (n=daveb@c-69-243-145-154.hsd1.in.comcast.net) |
| 01:06.41 | daveb_ | is anyone using asterisk 1.4 with suse linux 10.2? |
| 01:08.50 | jpe-nyc | philip - that return to ivr in announcements is new? |
| 01:08.57 | jpe-nyc | your the man.. |
| 01:09.08 | weazahl | could someone just give me a kick explination of how a highly available asterisk system is acomplished? how does a backup system take over? is it at the OS level or application level? does asterisk handle heartbeat itself? |
| 01:09.23 | weazahl | quick not kick... |
| 01:09.37 | weazahl | yeah someone give me a kick too. |
| 01:09.44 | weazahl | thanks |
| 01:09.57 | jpe-nyc | brb /away |
| 01:10.36 | jpe-nyc | you might be better off asking that Q in the asterisk channel tho... |
| 01:10.45 | daveb_ | can anyone help me with a few asterisk questions? I'm a total newby to Linux & asterisk |
| 01:11.07 | weazahl | um i am in that channel |
| 01:11.15 | *** join/#asterisk dseeb_ (n=dcb@CPE-124-177-0-178.vic.bigpond.net.au) |
| 01:11.48 | daveb_ | is anyone using asterisk 1.40 ? |
| 01:12.17 | daveb_ | brb |
| 01:12.31 | tzafrir_home | daveb_, ask more specific questions. If you encountered a specific problem, please describe it |
| 01:14.02 | markit | I would like incoming calls be put on "queue" if the secretary is already busy with some other call, but don't want her to be forced to login as agent at the beginning of the day... how could I do? |
| 01:14.57 | *** join/#asterisk DJS_2_6 (n=djstillm@cpe-066-057-115-255.nc.res.rr.com) |
| 01:15.36 | daveb_ | I guess I don't have a specific problem, yet. I'm trying to put together my first asterisk system but am working with my first Linux system as well |
| 01:16.00 | daveb_ | I was wondering if asterisk 1.4 would work with suse linux 10.2.... |
| 01:16.11 | tzafrir_home | Why don't you use the asterisk packages from suse? |
| 01:16.17 | Qwell | Linux is Linux |
| 01:16.23 | markit | daveb_: you are really brave if you are new to GNU/Linux and Asterisk as well :) |
| 01:16.26 | Qwell | if an app works in one place, it should work everywhere else |
| 01:16.55 | daveb_ | hmmm...didn't know suse had asterisk packages...I'll look into that....Thanks! |
| 01:16.57 | tzafrir_home | 10.2 has a fairly recent version of Asterisk. Not to mention zaptel. It will save you much of the initial grief |
| 01:17.49 | daveb_ | just loaded this PC with linux teh otehr night, wondered why it took only 3 of the 5 CD's |
| 01:17.58 | inv_arp[work] | Last night, my girlfriend came home and ran upstairs to me and said, "Take off my dress." |
| 01:18.02 | inv_arp[work] | So I did. Then she said, "Take off my bra and panties." |
| 01:18.04 | daveb_ | maybe asterisjk is buried on teh otehr 2 somewhere? |
| 01:18.05 | inv_arp[work] | So I did. Then she said, "Stop wearing my clothes." |
| 01:18.18 | markit | lol |
| 01:18.19 | *** join/#asterisk Az_au (n=az@216.127.73.119) |
| 01:18.22 | DJS_2_6 | Hello. Finding an abundance of info on Asterisk, and have xBSD/Linux experience. Just hoping to find out the hardware scaling for Asterisk (i.e., 32MB ram and 100MB storage for every 100 users, etc...) |
| 01:18.39 | DJS_2_6 | inv_arp[work] - lol |
| 01:19.08 | Qwell | DJS_2_6: It depends |
| 01:19.10 | daveb_ | wow, thought this channel was dead when I forst got on....livening up now :-) |
| 01:19.31 | DJS_2_6 | daveb_ - That is the way of IRC... |
| 01:20.27 | DJS_2_6 | Qwell - Ok, so what would be a good base config for say, 100 users with most or all of the goodies running CISCO Voip phones? |
| 01:21.13 | CrashHD | hey guys, I have a situation in 1.2.14 over a vpn (phones at one end of the vpn and server at the other) where phones will continue to ring after the server things they ened to stop |
| 01:21.30 | CrashHD | s/things/thinks |
| 01:21.40 | daveb_ | I probably wouldn't have been as interested inknowing linux but the test software that I support in my job is moving from QNX to linux...thought asterisk would be an interesting way to dive in :-) |
| 01:23.07 | *** join/#asterisk wubba (n=kmurrey@cable-76-215.sssnet.com) |
| 01:26.20 | Opperior | markit - have you concidered send the calls bound to the secretary to a queue and just have her extension set as a static agent? |
| 01:27.21 | wubba | Do you know why a deleted extension would still show up in -= Registered Asterisk Dial Plan Hints =- |
| 01:28.46 | *** join/#asterisk Iamnacho (i=Iamnacho@ip68-103-140-124.ks.ok.cox.net) |
| 01:28.54 | *** part/#asterisk russellb (i=russellb@asterisk/developer-and-stable-maintainer/drumkilla) |
| 01:30.13 | *** join/#asterisk ariel_ (n=ariel_@dsl-20-177.cofs.net) |
| 01:36.10 | inv_arp[work] | ./ ../ FVT/ PDT/ PPT/ QRT/ UAT/ |
| 01:36.10 | inv_arp[work] | ocovtst@hosat021:/ama/core > df -h . |
| 01:36.32 | inv_arp[work] | arrgh.. putty |
| 01:37.43 | *** join/#asterisk weazahl (n=weazahl@adsl-66-142-41-124.dsl.kscymo.swbell.net) |
| 01:38.47 | *** join/#asterisk bmg505 (n=leon@c1-241-7.rndf.isadsl.co.za) |
| 01:41.45 | *** join/#asterisk aptura (n=jondoe@S010600a0c93f6f7e.vs.shawcable.net) |
| 01:45.22 | *** join/#asterisk teknoprep (n=chris@unaffiliated/teknoprep) |
| 01:45.31 | SplasPood | Hey anyone know if 1.2.15 fixes the attended transfer bug? |
| 01:45.41 | SplasPood | I didn't see anything that jumped out in the ChangeLog |
| 01:47.02 | *** join/#asterisk wedhorn (n=damien@58.6.89.11) |
| 01:48.20 | *** join/#asterisk EyeCue (n=eyecue@miranda/eyecue) |
| 01:51.05 | ariel_ | attended transfer bug? |
| 01:51.40 | *** part/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 01:51.41 | *** join/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 01:53.31 | Nugget | http://techdigest.tv/pcmaclinux.jpg |
| 01:54.45 | *** join/#asterisk shodan (n=shodan@ip210.99-113-216.pppoe4.joliette.intermonde.net) |
| 01:56.52 | *** join/#asterisk gerphimum (i=Trekkie@207.190.58.83) |
| 02:00.40 | *** part/#asterisk wubba (n=kmurrey@cable-76-215.sssnet.com) |
| 02:02.50 | *** join/#asterisk Flauto (n=zhao@adsl-68-253-253-78.dsl.emhril.ameritech.net) |
| 02:07.04 | *** join/#asterisk antlers (n=antlers@ip70-189-186-184.lv.lv.cox.net) |
| 02:07.11 | SplasPood | wow |
| 02:07.22 | SplasPood | 1.2.15 actually shits itself on an attended transfer |
| 02:07.22 | aptura | wow what |
| 02:07.26 | SplasPood | whereas before it'd just hang up. |
| 02:08.38 | *** join/#asterisk cnbrk (n=cnbrk@senolsun.user.msu.edu) |
| 02:08.41 | cnbrk | hi guys |
| 02:08.48 | gerphimum | hi. |
| 02:08.52 | cnbrk | i have a really annoying problem with asterisk |
| 02:09.00 | cnbrk | i can't use my sip phone on campus |
| 02:09.19 | aptura | via ethernet? |
| 02:09.20 | gerphimum | probably a firewall on your campuses router |
| 02:09.25 | aptura | or wifi |
| 02:09.35 | cnbrk | i have the asterisk server at my home country and everybody in my country, even some guys from other country are using my sip server in my country to talk within each other |
| 02:09.46 | cnbrk | however I can call them but no sound at both sides |
| 02:09.57 | cnbrk | voip.brujula.net works on my campus |
| 02:10.01 | aptura | cnbrk did you configure it that way? |
| 02:10.23 | cnbrk | it's an private network over the internet |
| 02:10.45 | cnbrk | we use it to reduce company phone costs |
| 02:10.56 | cnbrk | all the branches are connected to the network |
| 02:12.53 | cnbrk | how do you think other voip services work on campus and mine doesn't |
| 02:13.11 | cnbrk | i get real public ip when i plug my laptop to network outlet on the wall |
| 02:13.25 | cnbrk | no nat involved i think |
| 02:14.12 | *** join/#asterisk Damin (n=damin@nucleus.nacs.net) |
| 02:18.47 | SplasPood | ok this is crazy, now 1.2.14 is crapping out too |
| 02:23.35 | cnbrk | asterisk drives me crazy |
| 02:24.38 | SplasPood | Its starting to drive me back around to sane again.. it farkin lapped me! |
| 02:27.46 | tessier_ | asterisk has lapped me many times over the last 3 years |
| 02:31.07 | cnbrk | i have a fuckin firewall problem and i can't solve it |
| 02:31.23 | cnbrk | my system doesnt work on campus however vonage works |
| 02:32.01 | gerphimum | sounds like a firewall problem.. |
| 02:32.05 | cnbrk | yes but why |
| 02:32.19 | cnbrk | my asterisk server doesnt have a firewall nor behind nat |
| 02:32.31 | gerphimum | im guessing theres a firewall on your campus, bro. |
| 02:32.45 | cnbrk | yeah but how come vonage passes thru it |
| 02:32.46 | gerphimum | you might get a public ip, but that doesnt mean its wide open on all ports or types of packets |
| 02:33.36 | *** join/#asterisk matiasds19 (n=matiasds@host181.201-252-18.telecom.net.ar) |
| 02:33.46 | matiasds19 | hello |
| 02:33.59 | gerphimum | hi. |
| 02:34.14 | cnbrk | yeah but still the question vonage works mine doesn't |
| 02:34.20 | matiasds19 | im having some problems with asterisk....am i in the right channel? |
| 02:35.39 | gerphimum | yep. |
| 02:35.55 | matiasds19 | ok, so, there it goes |
| 02:36.33 | matiasds19 | i have a sip provider, and the asterisk is dropping the incoming calls |
| 02:38.07 | *** join/#asterisk foxxtrot (n=craig@c-67-185-0-172.hsd1.wa.comcast.net) |
| 02:38.42 | matiasds19 | i dont know why, because i tested it with a PAP2 and the line is working |
| 02:41.29 | matiasds19 | someone can help me? |
| 02:41.43 | *** join/#asterisk coppice (n=chatzill@118.202.17.210.dyn.pacific.net.hk) |
| 02:49.04 | matiasds19 | someone? |
| 02:51.40 | *** part/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 02:51.41 | *** join/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 02:54.55 | weazahl | can hudlite server be install on something other than trixbox? i cant find where to d/l it. |
| 02:57.38 | *** join/#asterisk h0 (n=fakhir@unaffiliated/fakhir) |
| 02:58.58 | *** join/#asterisk pirulo (n=andres_p@70.56.223.76) |
| 03:08.20 | *** join/#asterisk lowlevel (n=Stuart@CPE000e0c057fad-CM000f9f7d6742.cpe.net.cable.rogers.com) |
| 03:09.00 | *** join/#asterisk apardo (n=apardo@87.217.145.48) |
| 03:12.09 | *** join/#asterisk bpiper (n=bpiper@user-142gior.cable.mindspring.com) |
| 03:13.55 | bpiper | Guys, ever seen this message? I get it when trying to access my vmail.cgi script... |
| 03:13.55 | bpiper | Can't locate Time/HiRes.pm in @INC (@INC contains:..... |
| 03:13.55 | bpiper | BEGIN failed--compilation aborted at /var/www/cgi-bin/vmail.cgi line 22. |
| 03:15.19 | *** join/#asterisk mhayk (n=mhayk@200.199.33.204) |
| 03:18.46 | *** part/#asterisk bpiper (n=bpiper@user-142gior.cable.mindspring.com) |
| 03:18.59 | *** join/#asterisk bpiper (n=bpiper@user-142gior.cable.mindspring.com) |
| 03:20.40 | Nugget | you're missing the Time::HiRes perl module. |
| 03:20.47 | Nugget | run cpan and type "install Time::HiRes" |
| 03:20.58 | Nugget | or use whatever packaging system your flavor of unix uses |
| 03:29.01 | aptura | what is the default port for iax under [general] in iax.conf 4569 ? |
| 03:30.00 | *** join/#asterisk h0 (n=fakhir@unaffiliated/fakhir) |
| 03:30.22 | aptura | figured it out |
| 03:32.03 | bpiper | thanks Nugget |
| 03:32.48 | aptura | huu this is stumping me. While i never use iax for calls outside my network i configured my client to log in externally and the dial plan does not init. |
| 03:34.59 | *** join/#asterisk Iamnacho (i=Iamnacho@ip68-103-140-124.ks.ok.cox.net) |
| 03:37.00 | hads | telnet? |
| 03:37.00 | Nugget | telnet is eeeeeeevil! |
| 03:38.45 | *** join/#asterisk Winkie (i=sd@87-194-8-125.bethere.co.uk) |
| 03:44.15 | aptura | nugget you on |
| 03:45.01 | aptura | does iaxprov.conf need to be edited. |
| 03:47.10 | *** join/#asterisk h0 (n=fakhir@unaffiliated/fakhir) |
| 03:48.05 | *** join/#asterisk mtgh (n=chatzill@dsl093-001-038.det1.dsl.speakeasy.net) |
| 03:49.50 | ttuttle | HEY! |
| 03:49.51 | ttuttle | Has anyone here heard of TellMe Networks' 1-800-555-TELL service? |
| 03:50.45 | ttuttle | Good. Well read this, and digg it: http://digg.com/tech_news/Festivus_Easter_Egg_in_TellMe_1_800_555_TELL_voice_portal_TRY_IT |
| 03:50.57 | ttuttle | There's an easter egg in the service that plays a quote from Seinfeld. |
| 03:51.40 | *** part/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 03:51.41 | *** join/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 03:52.39 | ttuttle | Anyone rea dit? |
| 03:52.43 | ttuttle | s/rea dit/read it/; |
| 03:53.08 | Bobthehunter | old nws |
| 03:53.09 | Bobthehunter | news |
| 03:53.46 | ttuttle | Bobthehunter: It is? |
| 03:53.57 | ttuttle | Bobthehunter: Google couldn't find it. |
| 03:54.22 | ttuttle | Bobthehunter: I think I'm the first one to discover, or at least blog about, it. |
| 03:54.45 | ttuttle | Bobthehunter: Have you heard of it before? |
| 03:54.47 | Qwell | Don't misdial 555-tell as 555-8344 |
| 03:54.52 | ttuttle | Qwell: Oh. |
| 03:54.55 | ttuttle | Qwell: Digg my story =D |
| 03:55.10 | ttuttle | Qwell: (Please ;-) |
| 03:55.28 | ttuttle | Yay, 2 diggs! |
| 03:56.07 | Qwell | nice |
| 03:56.21 | ttuttle | Qwell: Dugg? |
| 03:58.03 | ttuttle | Qwell: ? |
| 03:58.51 | aptura | Qwell your a iax master right? I have mine configured for internel access and it works. externally it does not. All ports forwarded and externaip and other info in general almost a mirror copy of sip.conf with the exception the port number is 4569. What may i be missing? all sip configurations work perfectly both internally and externally. |
| 03:59.51 | aptura | or if anyone else could answer this. |
| 04:01.04 | ttuttle | aptura: firewall? |
| 04:01.19 | aptura | port forwarded to my server with 4569 |
| 04:01.36 | aptura | sip forwarded to and works well. |
| 04:01.37 | ttuttle | aptura: ah |
| 04:01.49 | ttuttle | aptura: Is Asterisk receiving the connections at all? |
| 04:01.57 | aptura | no dial plan action. |
| 04:02.14 | aptura | so my asumption is it could be a firewall issue but not sure. |
| 04:02.29 | aptura | it should be right. |
| 04:06.17 | aptura | it says its registered. |
| 04:06.20 | aptura | odd. |
| 04:06.59 | ttuttle | aptura: So it's registered, but can't place or receive calls? |
| 04:07.14 | aptura | not externally outside the fw |
| 04:07.18 | aptura | internally it can. |
| 04:07.29 | aptura | more then likly a fw issue which i dont see how. |
| 04:08.49 | aptura | ttruttle I have been using asterisk for a long time so know it pretty well this though is a stumper :) |
| 04:09.22 | *** join/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net) |
| 04:09.22 | *** mode/#asterisk [+o mog] by ChanServ |
| 04:11.04 | *** join/#asterisk adker_ (n=chatzill@74-33-198-79.br1.glv.ny.frontiernet.net) |
| 04:22.53 | *** join/#asterisk h0 (n=fakhir@unaffiliated/fakhir) |
| 04:23.43 | *** join/#asterisk Johnsie (n=jdlewis@jdlewis.org) |
| 04:24.00 | Johnsie | Hi folks. |
| 04:24.44 | Johnsie | I am stumped, I have a problem where I'd like to essentially do database manipulation (get/put/delete) over ODBC, is there any way to do this in Asterisk 1.4? |
| 04:26.12 | mog | func_odbc |
| 04:26.18 | mog | bam |
| 04:26.19 | mog | or agi |
| 04:27.02 | Johnsie | Bless you. |
| 04:27.11 | Johnsie | I've spent hours searching and it was right under my nose. |
| 04:28.23 | Johnsie | Thank you so much. |
| 04:28.53 | matiasds19 | Hello, can someone help me? |
| 04:29.02 | mog | sure whats up |
| 04:29.02 | matiasds19 | my asterisk is dropping incoming calls |
| 04:29.11 | matiasds19 | iv got a sip provider |
| 04:29.18 | matiasds19 | i configured it like this |
| 04:29.27 | matiasds19 | at the [global] section |
| 04:29.46 | matiasds19 | register => user:pass@provider.com |
| 04:30.14 | matiasds19 | then, i wrote a section for outgoing calls |
| 04:30.23 | *** join/#asterisk DocHolliday (i=RgRabbit@gateway/gpg-tor/key-0x0E4F6D6C) |
| 04:30.31 | matiasds19 | named [provider] and the type is peer |
| 04:31.10 | DocHolliday | any providers out there that offer free call to tollfree? |
| 04:31.11 | matiasds19 | i can make outbound calls, but when someone calls me, the asterisk finishes the call, i answers, and it hangs up |
| 04:31.19 | *** part/#asterisk Johnsie (n=jdlewis@jdlewis.org) |
| 04:31.46 | matiasds19 | sorry about my english, its not my native language |
| 04:32.18 | DocHolliday | your english is great but i cant help you :P |
| 04:33.18 | *** join/#asterisk mtgh (n=chatzill@dsl093-001-038.det1.dsl.speakeasy.net) |
| 04:33.23 | matiasds19 | thank you anyway... |
| 04:33.42 | matiasds19 | anyone that can help please... |
| 04:34.06 | matiasds19 | i cant find anything at google, i dont know where else to search |
| 04:34.11 | mog | whats the output on the console, for why it hangsup |
| 04:35.29 | matiasds19 | no, all it says is = Spawn extension (proximo_saliente, s, 1) exited non-zero on 'SIP/706035-081e2228' |
| 04:36.29 | matiasds19 | but i know that asterisk is dropping it beacause i tested the line with a linksys pap2 and it worked fine |
| 04:36.32 | mog | you should get more than that |
| 04:36.43 | mog | do you have an extension there? |
| 04:37.24 | matiasds19 | yeah, the s extension answers and try to playback the congratulations sound |
| 04:37.36 | matiasds19 | but if i ansewer with a phone the sames happens... |
| 04:37.51 | mog | there is something else on the output or you arent telling us something, paste bin that section of extensions.conf |
| 04:38.56 | matiasds19 | [proximo] |
| 04:39.05 | mog | dont paste it to the channel |
| 04:39.06 | matiasds19 | exten => s,1,Answer(1) |
| 04:39.09 | mog | use pastebin |
| 04:39.15 | matiasds19 | sorry im new |
| 04:39.21 | matiasds19 | whats is pasebin? |
| 04:39.29 | mog | pastebin.ca |
| 04:39.31 | robl^ | ~pastebin |
| 04:39.32 | jbot | it has been said that pastebin is a place to paste your stuff without flooding the channel - try http://pastebin.ca, or http://channels.debian.net/paste, or http://rafb.net/paste/ |
| 04:39.50 | matiasds19 | oh thanks |
| 04:40.40 | matiasds19 | http://www.pastebin.ca/348454 |
| 04:40.57 | matiasds19 | thats the extensions.conf part |
| 04:41.08 | mog | okay pastebin what happens when you call |
| 04:41.14 | *** join/#asterisk bkruse_home (n=kruz@69.73.127.92) |
| 04:42.36 | matiasds19 | http://www.pastebin.ca/348456 |
| 04:43.11 | mog | do a sip show peers |
| 04:43.13 | mog | pastebin that |
| 04:44.00 | matiasds19 | http://www.pastebin.ca/348457 |
| 04:44.26 | matiasds19 | the provider is proximo |
| 04:44.48 | aptura | hi mog |
| 04:44.49 | aptura | :) |
| 04:44.52 | matiasds19 | there are two useres, because i have two lines |
| 04:45.26 | mog | hi aptura |
| 04:45.34 | mog | can you call between boxes locally? |
| 04:45.41 | mog | and can you do a sip debug and pastebin that |
| 04:45.44 | aptura | mog your a iax mistro right? |
| 04:46.08 | mog | i like to think of myself as more a jack of all trades |
| 04:46.16 | aptura | :) |
| 04:46.23 | mog | whats up |
| 04:46.27 | aptura | okay this is a stumper for me. |
| 04:47.14 | aptura | iax works internally in network. When i change the iax clients ip for my public ip it does not pass though the firewall. 4560 like 5060 is port forwarded to my firewall. |
| 04:47.19 | aptura | 4569 I mean |
| 04:47.34 | aptura | I am scratching my head on this one. |
| 04:47.58 | aptura | no dial plan action calling back into my network with a aix client. |
| 04:48.01 | aptura | iax client. |
| 04:48.10 | mog | you forwarding udp? |
| 04:48.15 | mog | and not tcp by accident? |
| 04:48.15 | *** part/#asterisk bpiper (n=bpiper@user-142gior.cable.mindspring.com) |
| 04:48.17 | mog | is it up now |
| 04:48.24 | mog | can you give me ip i could test for ya |
| 04:48.24 | aptura | perhaps. i can check. |
| 04:48.33 | aptura | let me check first |
| 04:49.41 | bkruse_home | please give mogorman root access. |
| 04:49.44 | aptura | okay that might have been it testing now. |
| 04:50.17 | mog | awesome |
| 04:50.25 | aptura | dam |
| 04:50.27 | aptura | that was it |
| 04:51.00 | bkruse_home | mog: get back to work! |
| 04:51.01 | aptura | thanks mog. I didnt even think about that. |
| 04:51.06 | mog | it happens |
| 04:51.11 | mog | oh i need to do that |
| 04:51.12 | mog | now! |
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| 04:51.41 | *** join/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 04:53.39 | aptura | mog where are you |
| 04:53.57 | mog | huntsville alabama |
| 04:54.21 | aptura | nasa! |
| 04:54.22 | aptura | :) |
| 04:54.26 | mog | close |
| 04:54.28 | aptura | dinner. |
| 04:54.35 | mog | a stone throw from where i live |
| 04:54.40 | mog | if i was a giant that is |
| 04:54.46 | mog | or a rocket stone |
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| 04:55.58 | matiasds19 | sorry, my internet conection went down |
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| 05:01.23 | matiasds19 | hey mog, still you there? |
| 05:01.35 | mog | nope |
| 05:01.43 | matiasds19 | hehe |
| 05:04.26 | matiasds19 | someone that can help me? |
| 05:05.20 | putzz | I want to terminate my own lines to pstn, how would I be able to do it cheaper then paying this expensive fees from voip providers? |
| 05:07.46 | Opperior | there are several boards that will allow you to plug a standard phone line into an Asterisk system |
| 05:08.55 | putzz | I dont want to terminate just 1 line |
| 05:08.56 | putzz | ;) |
| 05:08.58 | putzz | thats easy |
| 05:09.12 | Opperior | how many and what kind? |
| 05:09.40 | putzz | I want to terminate sip to pstn |
| 05:10.06 | putzz | but I will need more then 1 line to terminate lots of calls |
| 05:10.48 | Opperior | ok, well, Digium a couple of boards. One supports 4 lines, one support, 24 I think |
| 05:11.07 | fetcher | putzz: have you looked into any pay-as-you-go VoIP providers? Depending on calling patterns, those can often be less expensive than telco lines |
| 05:11.52 | putzz | well I was thinking of setting up a little calling card company, going with a pay-as-you-go voip provider will make the rates too high |
| 05:12.31 | Opperior | then you need more lines then can be done with analog lines. A couple PRI lines might do better |
| 05:13.24 | putzz | how much do PRI lines go for and how many do they support? |
| 05:13.28 | Opperior | depending on what you expect for call volume, of course |
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| 05:14.13 | Opperior | one PRI will support 23 simultanious calls. And they're alot easier to deal with |
| 05:14.26 | putzz | how much would u say that would cost? |
| 05:14.31 | putzz | depends on the company? |
| 05:14.57 | Qwell | putzz: it varies wildly |
| 05:15.16 | Qwell | from as low as $200-300 in some areas, to well over $1k |
| 05:15.19 | Qwell | per month |
| 05:15.36 | Opperior | depending on features and competition in the area |
| 05:15.45 | putzz | is it something charged by the minute? where would I get it exising phone companies? or like MCI |
| 05:15.46 | putzz | ..? |
| 05:16.27 | Opperior | phone companies could provide one, though I would steer cleat of Verizon |
| 05:16.41 | Opperior | er, steer clear |
| 05:16.47 | putzz | well im in canada |
| 05:16.54 | putzz | I want to stay away from crazy taxes |
| 05:16.54 | putzz | heheh |
| 05:16.56 | Opperior | then I haven't a clue |
| 05:16.59 | putzz | so I need some suggestions |
| 05:17.10 | Opperior | :P |
| 05:18.05 | putzz | sorry for all the questions just cant find much info about it |
| 05:18.17 | matiasds19 | ive got an asterisk dropping me the incoming calls, can someone help??? |
| 05:18.32 | putzz | PRI charges by the minute? |
| 05:20.13 | Opperior | around here in New Hampshire they do, but it's cheap $.03 to .04 long distance. But there is a monthly service fee as well |
| 05:21.43 | putzz | Opperior: do you have PRI? |
| 05:21.51 | *** join/#asterisk twisted[asteria] (n=twisted@pdpc/supporter/active/twisted) |
| 05:21.51 | *** mode/#asterisk [+o twisted[asteria]] by ChanServ |
| 05:22.22 | Opperior | I've set up a few of my clients with it, but I wasn't really involved in the billing side |
| 05:22.53 | putzz | right on |
| 05:22.59 | putzz | well thanks I'll look into the rest |
| 05:23.02 | putzz | ;-) |
| 05:24.00 | Opperior | :) |
| 05:25.08 | putzz | man some calling card companies have such cheap prices I dont know how its possible |
| 05:25.08 | putzz | hehe |
| 05:25.14 | putzz | I guess volume |
| 05:26.39 | Opperior | I think volume is the key. It also lets them work some special deals with other carriers. |
| 05:27.11 | *** join/#asterisk ryguillian (i=rhayes@numbertwo.midphase.com) |
| 05:27.17 | putzz | + they start many companies at once making more volume |
| 05:27.40 | *** join/#asterisk ryguillian (i=rhayes@numbertwo.midphase.com) |
| 05:28.36 | putzz | what is the difference between PRI and a T1 voice |
| 05:28.39 | *** part/#asterisk ryguillian (i=rhayes@numbertwo.midphase.com) |
| 05:30.14 | aptura | mog you here? |
| 05:30.36 | mog | yeah |
| 05:30.43 | aptura | like aircraft? |
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| 05:30.54 | mog | i wouldnt mine having one |
| 05:31.26 | aptura | mog, let me show you something that I would fly on and work on occationally. I also trained as a mechanic to work on in the usaf. |
| 05:32.04 | mog | spiffy, i have a few friends in usaf |
| 05:32.08 | aptura | http://www.mh-53pavelow.com/ go here and click on pav low movie. |
| 05:32.23 | aptura | on the right hand side of the url. |
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| 05:32.29 | aptura | err left hand side. |
| 05:32.31 | mog | yeah |
| 05:32.35 | mog | saw it |
| 05:32.40 | aptura | did you |
| 05:32.45 | aptura | been there before? |
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| 05:33.03 | mog | that looks intense |
| 05:33.04 | mog | no |
| 05:33.09 | mog | i just have quick eyes |
| 05:33.12 | putzz | aptura: can I pvt u? |
| 05:33.17 | aptura | sure |
| 05:33.30 | mog | how many people can it hold? |
| 05:33.30 | aptura | mog i am sure you have not seen the movie? |
| 05:33.50 | aptura | 40 troops. |
| 05:34.04 | aptura | I think |
| 05:34.11 | aptura | 35 to be more relistic |
| 05:34.13 | Opperior | PRI is a T1 where one channel is used for management. This lets you do things like have multiple DIDs, the ability to set your own caller ID, and a few other things |
| 05:35.05 | aptura | I was primarly the crewchief on the sikorsky H-3 the older sib of the 53. |
| 05:35.21 | x86 | Opperior: s/management/signalling |
| 05:35.31 | x86 | Opperior: also, you can have multiple DIDs on a CAS T1 |
| 05:35.34 | aptura | A h3 came up for auction at jame g mcmurphy auction once. |
| 05:35.35 | bkruse_home | x86: thank you |
| 05:35.47 | aptura | went for a rediculios 9k usd. |
| 05:35.48 | putzz | Opperior: does does PRI work is it like 1 line per channel, or just a channel to transmit data to the carrier? |
| 05:36.04 | aptura | noramlly that aircraft was 9 million dollars. |
| 05:36.32 | mog | so 12 million on black market |
| 05:36.36 | mog | hhmmmm one day |
| 05:36.36 | aptura | :) |
| 05:36.44 | aptura | done watching the vidio? |
| 05:36.59 | mog | yeah |
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| 05:37.02 | aptura | :) |
| 05:37.23 | aptura | it was some times called the widow maker. |
| 05:37.58 | Opperior | ignoring data, one channel means one active call |
| 05:38.10 | aptura | Once a crewchief was onboard and looked up at the main gear box. He saw a 1/2 inch gap between the transmission and frame mounts. He about crapped his pants. |
| 05:38.21 | aptura | the bolts were comming out. |
| 05:38.46 | aptura | if thay did. main rotor head and transmission would seperate from aircraft and everyone would be killed. |
| 05:39.11 | aptura | mog. now this you will really like. |
| 05:39.47 | aptura | mog, goto this and look at the movie trailer. |
| 05:39.50 | aptura | http://www.transformersmovie.com/ |
| 05:40.11 | mog | no flash |
| 05:40.17 | aptura | ahh no? |
| 05:40.18 | putzz | Opperior: if I got a PRI would it cost me as much as a t1 data line? or more? |
| 05:40.23 | aptura | ya goto see it. |
| 05:40.24 | aptura | :) |
| 05:40.36 | Opperior | depends on the carrier |
| 05:40.37 | aptura | its a really nice suprise. Especially for me :) |
| 05:40.53 | bkruse_home | mog: flash is lame |
| 05:41.03 | aptura | bkruse_home this site requires it. |
| 05:41.04 | bkruse_home | flash = system_load * 10 |
| 05:41.06 | mog | yeah i will |
| 05:41.14 | mog | bkruse_home, thought you were sleeping |
| 05:41.29 | putzz | last questions opperior: buying PRI would that force me to buy 23 channels or I can choose how many I need and upgrade it as I need it? |
| 05:41.29 | *** part/#asterisk bkruse_home (n=kruz@69.73.127.92) |
| 05:41.39 | aptura | It was executive produced by Steven Spielberg |
| 05:42.10 | aptura | Cannot wait to tell my nieces and nephews about it. |
| 05:42.11 | aptura | :) |
| 05:43.27 | Opperior | normaly you can decide how many channels you want (at least the carriers I've delt with let you), but not always. Usually the cost savings isn't worth it, though |
| 05:43.40 | aptura | It also pisses me off that I dont have any pictures of me standing next to it or my aircraft. Security Police would prevent us from taking pics and it was stupid. I know we were special ops but it would have been great to have my pic taken ;) |
| 05:44.40 | mog | heh national secrets be damned |
| 05:45.02 | aptura | yea |
| 05:45.06 | aptura | I know |
| 05:45.35 | Opperior | bed time. later all |
| 05:45.48 | aptura | audio visual would come out to take pictures for training vidios and SP's would get pissed saying to stop photographing the aircraft and we would yell back at them. |
| 05:46.20 | aptura | Aperently I am in one of the pictures and dont know about it. |
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| 06:05.45 | ^^Sidetrack^^ | just a quick question... I am running Trixbox 2.0, and the asterisk version is 1.2.13. I would like to upgrade the asterisk to 1.40.0 and I was wondering if I use the asterisk upgrade will it over write my existing config files? |
| 06:07.48 | ^^Sidetrack^^ | any advice would be greatly appreciated.. thanks in advance |
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| 06:16.39 | bpiper | Sorry for asking a non asterisk question but, I suck with setting permissions & I'm confused... |
| 06:16.39 | bpiper | I'm trying to run a php script. I can run it locally on the box via regular user but not via web. How do I set it so I can run it via web? |
| 06:16.58 | mog | #php might be able to help |
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| 06:38.53 | putzz | exten => _9.,1,Macro(dialout-trunk,3,${EXTEN:1},,) |
| 06:38.53 | putzz | exten => _9.,n,Macro(outisbusy,) |
| 06:39.01 | putzz | exten:1 means remove the 9 correct? |
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| 06:44.53 | dj-fu | yea |
| 06:44.55 | dj-fu | remove the first letter |
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| 06:52.19 | putzz | anyone know where I can get cheap sip phones? they r so expensive sick of spending so much money on it ;-) they should be so cheap been out for years now |
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| 07:10.14 | mosty | putzz, do you want cheap or good? |
| 07:10.38 | mosty | there's lots of cheap sip phones, but they're cheap because they're crap |
| 07:16.20 | Gershwin | is there a particular "expensive" sip phone that's exceptionally good? |
| 07:16.54 | Gershwin | the highest-end sip phone i've looked at is a cisco model.. perhaps there are others |
| 07:19.07 | putzz | cisco has always been the best |
| 07:19.08 | putzz | heh |
| 07:19.54 | Gershwin | I'm not familiar with, nor do I have experience with any other than Cisco |
| 07:20.09 | Gershwin | Do you have some fairly extensive experience with other brands and models putzz? |
| 07:21.10 | putzz | 90% cisco |
| 07:21.31 | Gershwin | I'll take that as a "no" |
| 07:21.57 | putzz | 10% others |
| 07:21.58 | putzz | heh |
| 07:22.02 | aptura | man its late |
| 07:22.20 | aptura | Ger, so your cisco experainced? got your ccna? |
| 07:27.43 | [TK]D-Fender | Gershwin : If you're looking for SIP, forget Cisco and go Polycom. |
| 07:29.09 | L|NUX | i have strange problem with my Cisco ATA 186 |
| 07:29.17 | L|NUX | it will register to asterisk |
| 07:29.28 | [TK]D-Fender | Cisco is a great phone physically speaking, but their SIP implementation is flakey and lacking. |
| 07:29.30 | L|NUX | but after 120 sec there will be no dial tone :( |
| 07:29.36 | L|NUX | any one aware of this issue ? |
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| 07:33.19 | Gershwin | I was just poking around the Polycom website and brought up the brochure of the Polycom SoundPoint IP 650 |
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| 07:40.25 | Op3r | brb gotta set the videoram to 64 |
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| 08:11.30 | matiasds19 | mog are you available now? |
| 08:12.51 | matiasds19 | someone who speaks spanish? |
| 08:13.49 | matiasds19 | anyone who can help me? |
| 08:18.21 | matiasds19 | someone who can help...plase...i cant find anything about my problem at google |
| 08:18.25 | *** join/#asterisk af_ (n=getsmart@ip-179-53.sn1.eutelia.it) |
| 08:20.53 | [TK]D-Fender | matiasds19 : Stop saying how much trouble you are having trying to solve you problem and just TELL US WHAT IT IS. |
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| 08:24.22 | matiasds19 | my asterisk is rejecting the incoming calls from a sip provider |
| 08:24.47 | matiasds19 | here is my sip debug from the incoming call |
| 08:24.53 | matiasds19 | http://www.pastebin.ca/348640 |
| 08:26.03 | matiasds19 | do you need sip.conf or extensions.conf? |
| 08:27.25 | [TK]D-Fender | matiasds19 : Ok, I'm not too sure exactly whree the problem is, but at least now someon else might be able to. |
| 08:28.23 | matiasds19 | oh...ok... |
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| 10:01.47 | xMOe | Hello guys we have problem in our callerID and we need to hair an expert to work on this issue remotly |
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| 10:04.08 | JT | xMOe: what sort of issue? |
| 10:04.09 | xMOe | anyone interesting for the position ? online dev for our company |
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| 11:04.37 | suma | when i do a ztcfg -vvvv i get an error, ZT_CHANCONFIG failed on channel 1: No such device or address (6) , some one please help ? |
| 11:08.01 | mafkees | is the correct module loaded ? |
| 11:09.06 | *** join/#asterisk markit (n=konversa@host119-245-static.72-81-b.business.telecomitalia.it) |
| 11:09.47 | markit | hi, how can I control inside dialplan if an extension is busy (talking) or not? |
| 11:16.15 | suma | mafkees: yes, module loads without any problem |
| 11:16.59 | suma | mafkees: i have sent you in private |
| 11:18.07 | mafkees | what is your /etc/zaptel.conf ? |
| 11:18.13 | mafkees | can you pastebin that one ? |
| 11:20.37 | suma | mafkees: i have only two lines |
| 11:20.39 | suma | fxsks=1-4 |
| 11:20.39 | suma | loadzone = us |
| 11:22.15 | markit | or how can I tell the state of a channel from the dialplan? |
| 11:26.52 | JT | markit: there's a variable or a function iirc |
| 11:27.03 | JT | check out README.variables in the docs directory |
| 11:28.00 | JT | markit: why do you need to do so, anyway? |
| 11:28.47 | mafkees | hhmm |
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| 11:29.38 | mafkees | suma: you need to add "channels=1-4" |
| 11:33.49 | suma | oh |
| 11:33.55 | suma | thanks i will check with that now |
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| 11:38.16 | markit | JT like in incoming call, if the extension 203 is not busy, ring extension 202, otherwise ring 404, or something like that |
| 11:38.42 | markit | a sort of "poke" without really dial to an extension |
| 11:38.50 | *** join/#asterisk [l1gh7-5p33d-\] (n=sanity@220-253-52-76.VIC.netspace.net.au) |
| 11:39.43 | mafkees | you can use the group() funtions for that |
| 11:39.50 | markit | I need a function like "extensionStatus(extension, status_query)" ---> extensionStatus("202", "BUSY") |
| 11:40.15 | markit | mafkees: no, I don't want to ring 203, just check, it has never to ring |
| 11:40.22 | mafkees | ok |
| 11:40.31 | mafkees | but busy means the device is dialed |
| 11:40.33 | mafkees | right ? |
| 11:40.43 | markit | yes |
| 11:40.54 | mafkees | so incoming and outgoing calls from a phone should set the group() |
| 11:41.00 | mafkees | then, you can check against that |
| 11:41.15 | mafkees | without dialing the phone |
| 11:41.21 | markit | is it a variable? |
| 11:41.29 | mafkees | it's a dialplan function |
| 11:41.42 | mafkees | on asterisk cli: show function group |
| 11:42.07 | mafkees | I think |
| 11:42.18 | mafkees | might be: show function GROUP |
| 11:42.33 | markit | mmm ok, that could help a lot, thanks indeed |
| 11:45.23 | markit | I've isdn bri, and I want to limint incominc calls to 1, while letting outgoing to reach 2, but if one incoming call is for fax, let come the other if for speech |
| 11:49.27 | *** join/#asterisk parag_ast (n=Administ@dxb-b18896.alshamil.net.ae) |
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| 11:51.41 | *** join/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 11:53.26 | *** join/#asterisk apardo (n=apardo@87.217.144.23) |
| 12:01.13 | *** join/#asterisk chat_jokey (n=acehunky@59.181.111.61) |
| 12:01.16 | chat_jokey | hello ... |
| 12:01.36 | chat_jokey | does any one know wats the setting in asterisk to redirect RTP traffic directly to provider .. |
| 12:02.14 | *** join/#asterisk zotz (n=zotz@24.244.163.157) |
| 12:02.37 | chat_jokey | nat=no and canreinvite=yes has to be put in provider setting or the user setting ? |
| 12:05.13 | *** join/#asterisk Faithful (n=Faithful@ns.linuxterminal.com) |
| 12:05.51 | mafkees | both |
| 12:07.45 | *** part/#asterisk sasch (n=sasch@82.107.30.102) |
| 12:08.56 | chat_jokey | umm ok .. |
| 12:09.28 | chat_jokey | also wondering what insecure=very used for ? |
| 12:10.32 | mafkees | that will allow calls without username/secret |
| 12:11.17 | chat_jokey | thanks mafkees ... but i still have problems .. my rtp traffic is still getting propogated through my server :( |
| 12:11.27 | chat_jokey | instead of only signalling |
| 12:11.40 | chat_jokey | are there any other setting required to be done in general context of sip.conf ? |
| 12:12.37 | mafkees | hhmm, the phone is on your lan ? |
| 12:12.51 | *** join/#asterisk L|NUX (n=linux@unaffiliated/lnux/x-10290) |
| 12:16.06 | chat_jokey | My server is in middle of my customer and my provider |
| 12:16.06 | chat_jokey | customer has VoIP gateway and is throwing traffic to me over Public IP |
| 12:16.06 | chat_jokey | so i want to be able to brigde RTP traffic between Customers Public IP and Providers Public IP |
| 12:16.08 | chat_jokey | my server only handles the billing part ... |
| 12:16.12 | chat_jokey | and the signalling offcourse .. |
| 12:16.16 | mafkees | hhmm |
| 12:16.18 | mafkees | no idea |
| 12:20.25 | chat_jokey | :( |
| 12:25.00 | *** join/#asterisk Jared_Leto (n=Lostprop@80-89-104-241.DSL.ycn.com) |
| 12:26.56 | tzafrir_laptop | chat_jokey, Asterisk is really not the software for that. If you just want to proxy calls, use e.g. openser |
| 12:27.07 | tzafrir_laptop | Do you need to process them in any way? |
| 12:29.20 | *** join/#asterisk UlbabraB (n=salama@host241-43-static.72-81-b.business.telecomitalia.it) |
| 12:35.30 | chat_jokey | tzafrir_laptop umm no processing apart from the billing ... coz my billing app is inside asterisk |
| 12:35.43 | chat_jokey | as an application to asterisk |
| 12:37.44 | tzafrir_laptop | chat_jokey, maybe you can set up the client to use a different media proxy? |
| 12:37.50 | tzafrir_laptop | What client is it? |
| 12:38.12 | tzafrir_laptop | Not sure if this is actually useful, though |
| 12:39.59 | *** join/#asterisk heh_v_water (n=heh_v_wa@71-210-49-107.hlna.qwest.net) |
| 12:41.34 | heh_v_water | aagh |
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| 12:51.41 | *** join/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 13:02.07 | suma | i could not configure my TDM400P card with FXO modules, can some one please help ? |
| 13:08.15 | *** join/#asterisk danalien (n=danalien@unaffiliated/danalien) |
| 13:12.14 | *** join/#asterisk PakiPenguin (n=uppal@linuxpakistan/admin/pakipenguin) |
| 13:12.18 | PakiPenguin | hello everyone |
| 13:25.14 | *** join/#asterisk yassine (n=yassine@dsl.voicint.com) |
| 13:26.14 | tzafrir_laptop | hi |
| 13:26.25 | suma | hi |
| 13:26.33 | tzafrir_laptop | suma, what exacty is the problem? |
| 13:27.09 | suma | when i do ztcfg -vvv |
| 13:27.11 | suma | i get |
| 13:27.20 | suma | ZT_CHANCONFIG failed on channel 1: No such device or address (6) |
| 13:27.39 | suma | It is a TDM400P card configured with 4 FXO modules |
| 13:29.32 | mafkees | can you check /proc/zaptel ? |
| 13:29.46 | mafkees | there should be some dirs |
| 13:29.51 | mafkees | 1 2 3 and 4 |
| 13:30.08 | suma | nothing like that there |
| 13:30.31 | suma | the directory /proc/zaptel is empty |
| 13:30.42 | mafkees | hhmm |
| 13:30.54 | mafkees | that's not correct I think |
| 13:31.08 | mafkees | maybe someone with more zaptel experience can help you |
| 13:31.23 | mafkees | all I did with zaptel is configuring sangoma E1 cards |
| 13:31.58 | suma | i c |
| 13:32.09 | tzafrir_laptop | lsmod | grep zaptel |
| 13:32.41 | suma | zaptel 216648 1 wctdm |
| 13:32.42 | *** join/#asterisk riddlebox (n=james@75-132-205-166.dhcp.stls.mo.charter.com) |
| 13:33.01 | tzafrir_laptop | So the module is loaded. Please check the kernel logs |
| 13:33.08 | tzafrir_laptop | e.g: dmesg| tail |
| 13:33.40 | suma | there is a problem |
| 13:34.02 | suma | let me get it to pastebin.ca |
| 13:34.12 | suma | http://www.pastebin.ca/348960 |
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| 13:35.00 | suma | wctdm: probe of 0000:00:18.0 failed with error -5 |
| 13:35.05 | suma | and few other lines |
| 13:36.36 | tzafrir_laptop | The driver could not talk with the card properly through the PCI bus |
| 13:36.44 | tzafrir_laptop | something at that level is bad |
| 13:36.53 | suma | i c |
| 13:37.00 | tzafrir_laptop | Others here are probably more familiar with this situation |
| 13:43.58 | *** join/#asterisk karmatronic (n=karmatro@84.77.170.211) |
| 13:44.13 | chat_jokey | i think you have to load the correct conf to determine which FXO module is on which port for your TDM400p card |
| 13:44.20 | chat_jokey | how many fxo / fxs you got suma ? |
| 13:44.35 | suma | chat_jokey: i have 4 fxo modules |
| 13:44.53 | suma | all of them are fxo and there is no fxs |
| 13:45.19 | chat_jokey | ok so what does your zaptel.conf read like ? |
| 13:45.19 | chat_jokey | also post ztcfg -vvvv |
| 13:46.04 | suma | http://www.pastebin.ca/348970 |
| 13:46.15 | suma | i will get the zaptel.conf now |
| 13:47.07 | suma | fxsks=1-4 |
| 13:47.07 | suma | loadzone = us |
| 13:47.07 | suma | defaultzone = us |
| 13:47.07 | suma | channels=1-4 |
| 13:49.34 | *** join/#asterisk kink0 (n=k@161.pool62-37-205.static.orange.es) |
| 13:50.02 | kink0 | hello, anybody gets asterfax running ? I got a lot of problems compiling app_txfax.so |
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| 13:51.41 | *** join/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 13:52.18 | suma | chat_jokey: you need any info please ? |
| 13:52.20 | *** join/#asterisk alfredh (n=alfredh@cm-84.209.226.067.chello.no) |
| 13:52.27 | suma | chat_jokey: please help |
| 13:52.43 | chat_jokey | umm hold |
| 13:52.49 | suma | ok |
| 13:59.32 | chat_jokey | yeah that sounds ok . |
| 13:59.42 | chat_jokey | ztcfg -vvv output of that .. wat does it say ? |
| 14:00.08 | chat_jokey | # |
| 14:00.08 | chat_jokey | ZT_CHANCONFIG failed on channel 1: No such device or address (6) |
| 14:00.13 | chat_jokey | <PROTECTED> |
| 14:00.21 | chat_jokey | IRQ conflicts |
| 14:00.30 | chat_jokey | cat /proc/interuppts |
| 14:00.45 | chat_jokey | <PROTECTED> |
| 14:01.00 | suma | one moment |
| 14:03.14 | suma | http://www.pastebin.ca/348970 |
| 14:08.07 | chat_jokey | duh it doesnt have output for interrupts |
| 14:09.29 | suma | one moment |
| 14:09.36 | suma | just rebooted |
| 14:10.12 | *** join/#asterisk olsen (n=diego@200.61.236.33) |
| 14:12.40 | suma | chat_jokey: http://www.pastebin.ca/349002 |
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| 14:14.12 | mafkees | now it has an interupt |
| 14:15.21 | suma | hey cool, yes, the dmesg | tail did not give any error |
| 14:15.26 | suma | problem with loose connection ? |
| 14:16.01 | chat_jokey | umm i suspect that you have put the card in 3.3V - 64bit PCI slot |
| 14:16.06 | chat_jokey | is it a server board by any chance ? |
| 14:16.21 | suma | chat_jokey: it is an embedded board, efika |
| 14:16.32 | chat_jokey | just google for "3.3V PCI SLOT" |
| 14:16.42 | chat_jokey | and you shall see how to identify 3.3V PCI slots |
| 14:16.44 | suma | yes it can take only 3.3V PCI |
| 14:17.02 | suma | TDM 400P is also 3.3v and 5v compatible right ? |
| 14:17.03 | chat_jokey | well this card i suspect only support 5V properly .. |
| 14:17.19 | chat_jokey | i have seen problems with 3.3V slots |
| 14:17.21 | *** join/#asterisk fholmes (n=fholmes@cpe-72-177-234-192.houston.res.rr.com) |
| 14:17.31 | suma | i c |
| 14:17.54 | suma | external hard disk input will do ? |
| 14:18.01 | *** join/#asterisk niekie (n=niekie@cc725705-a.roden1.dr.home.nl) |
| 14:18.09 | chat_jokey | how does hard disk relate to PCI slot ? |
| 14:18.17 | suma | there is a hard disk power connector in the TDM card |
| 14:18.34 | chat_jokey | you dont need it .. |
| 14:18.44 | chat_jokey | u only have FXO ports |
| 14:19.00 | suma | will that make the card to work well in 3.3v ? , since the embedded board can support only 3.3v |
| 14:19.33 | chat_jokey | well i dont know the exact answer .. since its not working with you ... i suspect that the answer is no |
| 14:19.59 | suma | i c |
| 14:20.00 | chat_jokey | put it in a standard PC with 5V PCI Slot |
| 14:20.26 | chat_jokey | and if it still doesnt work then you gotta call the RMA guys @ digium |
| 14:20.48 | suma | sure, thanks |
| 14:20.51 | chat_jokey | does the Green LED comes up on the card ? |
| 14:20.55 | suma | yes |
| 14:21.01 | suma | GREEN LEDs are on in the card |
| 14:21.04 | chat_jokey | well then the card hardware is all ok :) |
| 14:21.05 | suma | it is working fine now |
| 14:21.29 | chat_jokey | the card is working fine now u mean to say ? |
| 14:21.34 | suma | yes |
| 14:21.38 | chat_jokey | duh .. |
| 14:21.41 | suma | before there was no LED |
| 14:21.55 | suma | after this reboot, it just wotked |
| 14:22.33 | chat_jokey | u didnt update the chat :) |
| 14:22.43 | mafkees | always nice to see how a reboot can solve stuff |
| 14:22.51 | chat_jokey | i was under the impression that the card is still not working |
| 14:22.59 | suma | hey cool, yes, the dmesg | tail did not give any error |
| 14:23.05 | suma | i said this before |
| 14:23.11 | chat_jokey | umm okk |
| 14:23.20 | suma | i forgot to mention that it is working |
| 14:23.26 | mafkees | indeed |
| 14:23.43 | *** join/#asterisk Solaris444 (n=chatzill@203.161.84.80.static.amnet.net.au) |
| 14:23.52 | suma | thanks a lot |
| 14:23.58 | Solaris444 | hi fellas. |
| 14:24.08 | chat_jokey | oh btw mafkees after canreinvite=yes and nat=no on both the ends .. and also after making sure that i m not using S,t,T in my dial string ... i get 50% of RTP load and customer gets 50% of load .. |
| 14:24.10 | suma | for your time, i will surely check if i have similar problems |
| 14:24.13 | Solaris444 | I was just wondering if someone can suggest a good VoIP client that will work with asterisk. |
| 14:24.21 | chat_jokey | so in reality canreinvite is not passing off 100% RTP load |
| 14:24.32 | Solaris444 | my users are on windows mac and linux. |
| 14:24.42 | chat_jokey | Solaris444 wat is a VoIP client ? |
| 14:25.01 | chat_jokey | u mean SIP client ? |
| 14:25.02 | Solaris444 | I client software that will connect to an asterisk server. |
| 14:25.04 | chat_jokey | IAX client ? |
| 14:25.14 | Solaris444 | IAX I think. |
| 14:25.18 | chat_jokey | there is no such thing as VoIP client AFAIK |
| 14:25.35 | mafkees | there are iax softphones |
| 14:25.40 | Solaris444 | A client that supports IAX. |
| 14:25.51 | Solaris444 | Someone mentioned to me I should choose IAX if I wanted secure communications. |
| 14:26.11 | mafkees | ehm |
| 14:26.18 | mafkees | iax2 is not encrypted |
| 14:26.31 | Solaris444 | ok. |
| 14:26.42 | Solaris444 | It is important the communications are encrypted. |
| 14:26.48 | mafkees | if you want to go secure, setup vpn to asterisk box |
| 14:26.55 | chat_jokey | well asterisk natively doesnt support secure RTP |
| 14:26.58 | mafkees | Solaris444: nah, not really |
| 14:27.18 | chat_jokey | Solaris444 why would you want the communication to be encrypted ? |
| 14:27.29 | Solaris444 | Because it is a requirement. |
| 14:27.36 | chat_jokey | your asterisk admin can any ways barge in .. into your conversations :D |
| 14:27.39 | Solaris444 | I don't make the requirements I just have to meet them. |
| 14:27.53 | Solaris444 | eg, we have a jabber server. |
| 14:28.01 | Solaris444 | All communications on that are encrypted. |
| 14:28.07 | Solaris444 | It uses SSL. |
| 14:28.09 | chat_jokey | any ways .. so moral of story .. you need 2 setup a VPN |
| 14:28.17 | Solaris444 | Is that the only way? |
| 14:28.23 | mafkees | yes |
| 14:28.27 | chat_jokey | or use SRTP patch which is not official supported on this channel .. |
| 14:28.39 | Solaris444 | Oh. |
| 14:29.03 | Solaris444 | Well let me ask this, can traffic using IAX2 be intercepted and decoded by anyone in the middle? |
| 14:29.05 | chat_jokey | umm but you also require SRTP on the client end as well .. |
| 14:29.17 | chat_jokey | yeah .. everything is possible :) |
| 14:29.22 | mafkees | uhhuh |
| 14:29.26 | mafkees | even when encrypted |
| 14:29.30 | Solaris444 | ok i know this. |
| 14:29.37 | chat_jokey | well IAX2 aint encrypted by default .. |
| 14:29.42 | Solaris444 | right. |
| 14:29.47 | mafkees | neither is sip |
| 14:29.52 | chat_jokey | right .. |
| 14:29.59 | mafkees | nor h323 |
| 14:30.00 | Solaris444 | Hmmm, someone here told me IAX2 was... |
| 14:30.02 | mafkees | nor skinny |
| 14:30.04 | chat_jokey | he he he ... :P |
| 14:30.19 | chat_jokey | mafkees ROTFL |
| 14:30.30 | mafkees | :) |
| 14:30.42 | Solaris444 | What is the relative difficulty in decoding IAX2 for someone in the middle. |
| 14:30.51 | Solaris444 | can it be tapped easily. |
| 14:30.58 | chat_jokey | money :) and the desire to tapp .. |
| 14:31.09 | Solaris444 | desire = high. |
| 14:31.18 | *** join/#asterisk saftsack (n=oliver@p54A7D839.dip.t-dialin.net) |
| 14:31.23 | saftsack | hi |
| 14:31.36 | saftsack | will the polycom phones write on their bootp directory? |
| 14:31.41 | chat_jokey | well if that desire can lead to immediate financial gain to someone .. then you need 2 look @ VPN |
| 14:31.59 | Solaris444 | Alright I tell you guys what... can you suggest some softphones that support IAX2 |
| 14:32.06 | Solaris444 | I will look into the rest. |
| 14:32.24 | mafkees | Solaris444: for encryption stuff in asterisk look here: http://www.voip-info.org/wiki/view/Asterisk+encryption |
| 14:32.27 | chat_jokey | Google is a great friend of everyone "IAX2 soft phones " |
| 14:32.42 | Solaris444 | yes but I want you OPINIONS on which ones are good. |
| 14:32.56 | Solaris444 | *your |
| 14:32.59 | chat_jokey | which ones have you tried yet ? |
| 14:33.04 | Solaris444 | None. |
| 14:33.10 | chat_jokey | IDEFSK sounds good |
| 14:33.22 | mafkees | idefisk is ok |
| 14:33.59 | *** join/#asterisk UlbabraB (n=salama@host241-43-static.72-81-b.business.telecomitalia.it) |
| 14:34.14 | chat_jokey | http://www.voip-info.org/wiki/view/IAX_OpenVPN mebbe this shall help you Solaris444 |
| 14:34.29 | Solaris444 | thankyou. |
| 14:34.38 | Solaris444 | sorry I am a bit slow replying. |
| 14:34.48 | Solaris444 | All my machines are running at full load right now. |
| 14:34.58 | chat_jokey | ugggh .. this RTP is gonna eat up my entire Bandwidth |
| 14:35.29 | chat_jokey | does any one has any solution for setting up media path directly .. ? |
| 14:35.42 | *** join/#asterisk wunderkin (i=kev@ip72-208-3-221.ph.ph.cox.net) |
| 14:35.53 | chat_jokey | i cant use OpenSER / SER or any SIP Proxy .. coz my billing gets done from asterisk |
| 14:36.10 | Solaris444 | ok idefisk looks ok. any other suggestions for softphones? |
| 14:36.15 | chat_jokey | canreinvite=yes is screwing up big time .. |
| 14:36.24 | mafkees | chat_jokey: hhmm |
| 14:36.43 | mafkees | I did the trick with canreinvite=yes, nat=no |
| 14:36.50 | chat_jokey | any chan_sip developer on the channel ? |
| 14:36.50 | mafkees | and it's working there |
| 14:37.01 | chat_jokey | yeah i read that page on wiki |
| 14:37.04 | mafkees | but asterisk and all the phones are on the same net |
| 14:37.14 | chat_jokey | and remove the S,tT that i was passing on the dial string |
| 14:37.14 | mafkees | internal network |
| 14:37.20 | mafkees | maybe that's the main issue |
| 14:37.25 | mafkees | indeed |
| 14:37.41 | Solaris444 | in any case, thanks fellas. |
| 14:37.48 | mafkees | ur welcome |
| 14:40.55 | chat_jokey | umm the situation is that |
| 14:41.11 | chat_jokey | i have clients who are behind nat of customers's Asterisk Server .. |
| 14:41.17 | chat_jokey | and my asterisk server is doing billing |
| 14:41.27 | chat_jokey | and provider server is where we need 2 throw traffic .. |
| 14:41.55 | chat_jokey | so ideally i want an RTP bridge between Customer's Asterisk Server ==> Providers SIP Proxy |
| 14:42.17 | chat_jokey | and my Asterisk Server only handles the billing part of the signalling |
| 14:44.58 | *** join/#asterisk stkn (n=stkn@gentoo/developer/pdpc.active.stkn) |
| 14:46.11 | mafkees | uhhuh, I understand what you want to do |
| 14:51.18 | *** join/#asterisk chat_jokey (n=acehunky@59.181.111.61) |
| 14:51.24 | chat_jokey | duh got disconnected |
| 14:51.32 | chat_jokey | so mafkees any clues ? |
| 14:51.40 | *** part/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 14:51.41 | *** join/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 14:53.59 | mafkees | nope |
| 14:56.38 | *** join/#asterisk stkn (n=stkn@gentoo/developer/pdpc.active.stkn) [NETSPLIT VICTIM] |
| 14:56.50 | suma | my tdm400p cards is not starting the channels automatically |
| 14:57.01 | suma | only when i do ztcfg, it works fine |
| 14:57.17 | suma | do i need to do ztcfg whenever i do modprobe ? |
| 14:57.23 | suma | http://www.pastebin.ca/349042 |
| 14:57.39 | suma | It says changing from Unused to FXS Kewlstart |
| 14:58.14 | mafkees | you have to run ztcfg when you load the module |
| 14:58.17 | *** join/#asterisk greendisease (n=jack@fedora/greendisease) |
| 14:58.58 | suma | i c |
| 14:59.03 | suma | mafkees, thanks |
| 14:59.10 | mafkees | :) |
| 14:59.23 | *** join/#asterisk kore (i=kore@mindwipe.org) [NETSPLIT VICTIM] |
| 14:59.46 | suma | I'm very glad, since it is working very fine with unusual problems |
| 15:00.14 | suma | mafkees, there is a problem when i use a PCI riser and if i connect the tdm it doesn't work, is that normal ? |
| 15:00.47 | mafkees | I have no idea |
| 15:01.02 | suma | ok |
| 15:01.12 | Qwell | pci risers can cause problems with any hardware |
| 15:01.49 | mafkees | ah |
| 15:02.03 | suma | Qwell: the efika says it will work fine, i just now noticed, that it is not at all booting when i use pci risers |
| 15:03.02 | Qwell | of course it says it'll work fine :) |
| 15:03.20 | suma | the one which i used is a Fleexible riser, not the 90 degree one |
| 15:03.24 | Qwell | They aren't going to say "our cards suck, don't buy them" |
| 15:03.34 | suma | :) true |
| 15:03.50 | sweeper | I love the smell of ruby in the morning~ |
| 15:08.36 | *** join/#asterisk s1gny|wrk (n=s1gny@p54915CEE.dip.t-dialin.net) |
| 15:08.51 | *** part/#asterisk s1gny|wrk (n=s1gny@p54915CEE.dip.t-dialin.net) |
| 15:11.04 | saftsack | hi, has someone the newest polycom (sip 2.1) firmware for me? |
| 15:11.16 | *** join/#asterisk kFuQ (n=somedude@c-67-185-123-34.hsd1.wa.comcast.net) |
| 15:11.24 | saftsack | because dunno how2 get this firmware on the weekend |
| 15:11.51 | The_DoC^ | what style riser is it, one that has jumpers and plugs into a single pci slot or one that has connectors that plug into more than one pci slot |
| 15:19.28 | *** join/#asterisk Iamnacho (i=Iamnacho@ip68-103-140-124.ks.ok.cox.net) |
| 15:19.56 | *** join/#asterisk funkmaster (n=funky1@vhe-364501.sshn.net) |
| 15:21.09 | *** join/#asterisk heison (n=heison@CPE000a01d49e6f-CM00111a59bec2.cpe.net.cable.rogers.com) |
| 15:21.12 | funkmaster | hi there ppl, i got a question, i got a asterisk on my debian machine and a grandstream 486, i configured my voip providers in the sip and extensions.conf, so far the phone rings when any1 calls me on any of my numbers and i can talk |
| 15:21.29 | funkmaster | but there is one account i have problems with, it's skypho, italian voip provider |
| 15:21.43 | funkmaster | when i call ppl with using skypho extensions, it's fine |
| 15:21.51 | funkmaster | but when some1 calls me we can not hear each other |
| 15:22.07 | funkmaster | also the console shows no incoming call or outgoing call when i use skypho |
| 15:22.15 | funkmaster | any got an idea why that could be? |
| 15:25.31 | *** join/#asterisk pardove (n=pardove@80.191.117.98) |
| 15:26.22 | *** join/#asterisk eltech (i=G00Ds@ool-457c93b6.dyn.optonline.net) |
| 15:26.56 | pardove | i have problem installing TDM2400, system waits on modprobe wctdm24xxp forever. |
| 15:28.00 | mog | it has to power up all the modules |
| 15:28.06 | mog | or do you mean really forever |
| 15:28.15 | pardove | yes forever |
| 15:28.24 | pardove | nothing happens |
| 15:28.25 | mog | what does dmesg say |
| 15:28.52 | pardove | it waits on "Resetting the modules..." |
| 15:30.20 | Qwell | mog: can you check email? ;/ |
| 15:30.32 | mog | i was just about to ask in redbull |
| 15:30.33 | mog | i cant |
| 15:30.33 | Qwell | it's not liking me today |
| 15:30.34 | Qwell | oh |
| 15:31.20 | funkmaster | what can be the reason that an incoming call makes the phone ring, but it not shown in the asterisk console? |
| 15:31.33 | pardove | mog: what could be the problem? |
| 15:31.37 | mog | asterisk verbosity is turned down or off |
| 15:31.40 | funkmaster | although when having an incoming call on another number it shows it |
| 15:31.55 | saftsack | i search the polycom ip sip 2.1 firmware |
| 15:32.14 | mog | is it going through asterisk |
| 15:32.16 | funkmaster | also an verbosity issues if it shows other incoming calls on a different number? |
| 15:32.20 | funkmaster | yep |
| 15:32.22 | mog | saftsack, no piracy in here |
| 15:32.38 | Qwell | screw this... |
| 15:32.42 | mog | ? |
| 15:32.45 | Qwell | next year, I'm hiring a hobo to do my taxes |
| 15:32.50 | mog | lol |
| 15:33.02 | Qwell | they can't possibly screw up any worse than others have |
| 15:33.02 | mog | tina did mine last year |
| 15:33.02 | saftsack | mog, what has this to do with piracy? i have a polycom telephone but i dont want to wait for the support |
| 15:33.06 | mog | im gonna as her again |
| 15:33.12 | saftsack | or is it somehow possible to download those firmware from my phone? |
| 15:33.19 | mog | it is illegal to distribute the firmware |
| 15:33.20 | funkmaster | i got 4 numbers registered with my asterisk and it only shows the incoming call on the console for one number not the others.. |
| 15:33.33 | mog | if i had polycom firmware 2.1 i cant give it to you |
| 15:33.47 | mafkees | indeed |
| 15:34.17 | mog | hey pardove can you see if the card is taking interrupts |
| 15:34.21 | saftsack | mog, is there a chance to get it on the weekend the legal way? |
| 15:34.32 | mog | i have no clue |
| 15:34.36 | Qwell | saftsack: if the retailer you bought it from is open on the weekends |
| 15:34.42 | pardove | mog: 82: 1154 239 IO-APIC-level wctdm24xxp |
| 15:34.47 | mog | i just know that a lot of people come in here looking for firmware |
| 15:34.50 | mog | and its not right |
| 15:34.56 | mog | err legal |
| 15:35.01 | mog | probably is morally right |
| 15:35.04 | Qwell | mog: neither is forcing people to get it stupidly :P |
| 15:35.08 | Qwell | but I digress |
| 15:35.12 | saftsack | so it is also possible that someone of you will give me this file ... i mean polycom hasnt to know it |
| 15:35.30 | funkmaster | so someone knows why it does not appear in the console? is there anthing i have to set in the sip or extensions.conf to make it appear? |
| 15:35.30 | mog | pardove, are you sure it never gets to During Resetting the modules |
| 15:36.33 | *** kick/#asterisk [terrapen!i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net] by mog (no reason to curse) |
| 15:36.37 | *** join/#asterisk terrapen (n=cjs@66.29.164.42.static.utahbroadband.com) |
| 15:36.38 | saftsack | are you bought by polycom? |
| 15:36.48 | shido6 | ZzZZzz |
| 15:36.48 | mog | no he is kidding |
| 15:36.53 | Qwell | ha, hp.com takes paypal now |
| 15:36.57 | mog | we just dont support piracy in the channel |
| 15:37.12 | pardove | it gets to "During Resetting the modules..." and "After resetting the modules..." after a while but thats all and modprobe doesn't exit |
| 15:37.18 | pardove | mog: it gets to "During Resetting the modules..." and "After resetting the modules..." after a while but thats all and modprobe doesn't exit |
| 15:37.46 | mog | never gets to proslic anything? |
| 15:40.58 | *** join/#asterisk L|NUX (n=linux@unaffiliated/lnux/x-10290) |
| 15:41.45 | pardove | mog: my board also has VPM module |
| 15:43.47 | funkmaster | what does this mean? chan_sip.c:3030 update_call_counter: Call from user 'gs486' rejected due to usage limit of 1 |
| 15:43.59 | funkmaster | or where can i set the usage limit? |
| 15:44.00 | pardove | mog: when i kill modprobe proc. dmesg says: Port 1: Not installed, ... |
| 15:48.47 | *** join/#asterisk Koshatul (n=evangeli@ppp188-33.static.internode.on.net) |
| 15:51.40 | *** part/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 15:51.41 | *** join/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 15:52.46 | pardove | I need help on installing TDM2400 |
| 15:53.06 | *** join/#asterisk Kronuz (i=Kronuz@189.168.23.31) |
| 15:53.12 | Kronuz | hello |
| 15:56.06 | pardove | I need help on installing TDM2400 |
| 15:58.32 | Kronuz | hey, I just found out about PBX and Asterisk, but I don't yet fully understand what PBX is for... :( |
| 15:59.07 | Kronuz | I've read the wikipedia descriptiosn and all, but I'm still not sure if it does all I think it does... seems too good to be true |
| 16:00.19 | Kronuz | what hardware do one needs to get Asterisk doing something? |
| 16:00.27 | Nugget | A computer running Unix. |
| 16:00.37 | Kronuz | but a modem too? or what? |
| 16:00.42 | Nugget | A computer running Unix. |
| 16:00.44 | niekie | Kronuz, if you use Softphones, no. |
| 16:00.51 | L|NUX | hello |
| 16:00.56 | L|NUX | can some one help me with Cisco ATA |
| 16:01.26 | L|NUX | it register after 120 sec there will be no dial tone :( |
| 16:01.28 | L|NUX | what to do ? |
| 16:02.01 | Kronuz | niekie: okay, but what's the basic idea? it connects to the internet and then one can "call" asterisk using the proper VoIP protocols and then Asterkisk answers and what? or what does it do? |
| 16:02.18 | L|NUX | and i have to restart it |
| 16:02.49 | niekie | Kronuz, you can handle the calls in any way you desire. |
| 16:03.07 | Kronuz | (this is the first time I ever even hear about PBX with that name, but I guess that's what most companies with extensions, mailboxes, conferences, etc. have) |
| 16:03.09 | niekie | Queue them, transfer them, park them, tell them not to call anymore based on caller ID, or whatever. |
| 16:04.00 | Nugget | If you're unsatisfied with how reliable and simple it is to work with the telephones you have in your life then perhaps you'll enjoy asterisk. It combines the legendary complexity and instability of Unix with the breathtakingly obtuse and arcane telecommunications world. |
| 16:04.35 | niekie | Nugget, haha. |
| 16:05.43 | Nugget | In just under 12 hours of learning and configuration you can turn a $1200 home computer into a $9 answering machine. |
| 16:08.57 | mafkees | gheh |
| 16:10.36 | *** join/#asterisk smace (n=smace_br@200.149.32.180) |
| 16:10.53 | smace | how do I find out my Asterisk version? |
| 16:11.52 | mafkees | show version |
| 16:11.53 | smace | 1.0.7 - Is it good or bad? |
| 16:11.53 | Nugget | or run "asterisk -V" at the unix command prompt. |
| 16:11.53 | Nugget | 1.0.7 is ancient. |
| 16:11.53 | mafkees | or: core show version |
| 16:12.08 | Nugget | You should be running 1.2.15 or possibly 1.4.0 |
| 16:12.32 | Nugget | quite a number of significant improvements |
| 16:12.34 | mafkees | I wouldn't run 1.4.0 in production yet |
| 16:12.49 | Nugget | I do, but mafkees is correct that it's not a simple decision |
| 16:13.10 | Nugget | I'd wager that 1.4.0 is still more stable than 1.0.7 :) |
| 16:13.16 | smace | hehe |
| 16:13.23 | smace | Debian does not seem to think so :( |
| 16:13.29 | pardove | mog: are u online? |
| 16:13.32 | mafkees | debian sarge you mean ? |
| 16:13.34 | Qwell | debian still uses apache 0.9 |
| 16:13.40 | Nugget | feh, if you listen to debian they'll have you using smoke signals. |
| 16:13.43 | *** join/#asterisk networkjedi (n=networkj@f3c35.gpcom.net) |
| 16:13.49 | mafkees | lol |
| 16:13.49 | smace | lol |
| 16:13.53 | mafkees | all the debian hate |
| 16:13.57 | mafkees | tsk tsk |
| 16:14.11 | smace | what distro do you consider updated today? |
| 16:14.17 | pardove | Qwell: i have problem installing TDM2400, system waits on modprobe wctdm24xxp forever. |
| 16:14.18 | mafkees | debian etch |
| 16:14.20 | Qwell | debian, if you do it right |
| 16:14.25 | Qwell | pardove: Call support |
| 16:14.36 | *** join/#asterisk [TK]D-Fender (n=joe@64.235.216.2) |
| 16:14.48 | smace | @Qwell, do you use debian? |
| 16:15.01 | Qwell | smace: no, debian is too elite |
| 16:15.09 | Qwell | s/e$/ist/ |
| 16:15.17 | pardove | Qwell: support@digium.com ? |
| 16:16.03 | mafkees | debian is really nice |
| 16:16.08 | mafkees | in my opinion |
| 16:16.48 | *** join/#asterisk eltech (i=G00Ds@ool-457c93b6.dyn.optonline.net) |
| 16:20.23 | Nugget | I use Slackware, but that's not an endorsement. |
| 16:21.31 | smace | is there any Asterisk oficial repository for Debian? |
| 16:21.49 | Nugget | You should just compile asterisk by hand no matter which distro you select. |
| 16:21.51 | *** join/#asterisk sevard (n=sev@adsl-71-129-115-244.dsl.irvnca.pacbell.net) |
| 16:21.57 | Nugget | really, it's the better approach |
| 16:22.05 | smace | fine. I'll make it then. |
| 16:22.16 | smace | shoud I remove old asterisk before? |
| 16:22.28 | test34 | Nugget, why is it better? if there is a package of the latest version |
| 16:22.41 | Nugget | generally speaking, no, but for a jump from 1.0.7 to 1.2.15 it might be wise. |
| 16:23.01 | Nugget | for smaller jumps, though, just doing another "make install" on top of the old is perfectly reasonable |
| 16:23.43 | Nugget | test34: it's unlikely that any binary-packaged asterisk is going to be built the way you'd prefer. |
| 16:24.32 | Nugget | especially with more recent asterisks where there's a lot of decisionmaking that takes place when you're doing the "make menuselect" |
| 16:25.53 | shido6 | Amen. |
| 16:30.11 | *** join/#asterisk h0 (n=fakhir@unaffiliated/fakhir) |
| 16:32.00 | *** join/#asterisk Tili (n=tili@56.Red-83-53-145.dynamicIP.rima-tde.net) |
| 16:32.04 | Tili | what is xpp in zaptel |
| 16:32.17 | Tili | i cudn't get it to work on kernel 2.6.20 |
| 16:32.27 | Tili | so disabled it in Makefile |
| 16:33.58 | *** join/#asterisk lorinc (n=ang@caracas-4604.adsl.interware.hu) |
| 16:34.32 | niekie | sevard, it isn't of much use if I want to make long calls though. |
| 16:34.35 | niekie | 1 minute only ;) |
| 16:34.41 | niekie | But oh well. |
| 16:34.47 | mafkees | Tili: it's zaptel drivers for the xorcom hardware |
| 16:35.01 | Tili | mafkees: yeah found that out on google. thanks |
| 16:35.04 | mafkees | you have to bug tzafrir for it |
| 16:35.07 | Tili | i dont need that |
| 16:35.11 | Tili | nah its ok |
| 16:35.32 | Tili | i think i need to downgrade my kernel. 2.6.20 is not working with wanpipe either |
| 16:35.32 | mafkees | actually, you should post a bugreport on bugs.digium.com about it |
| 16:35.42 | sweeper | arg |
| 16:35.47 | mafkees | 2.6.20 will make a lot of stuff stop |
| 16:35.53 | tzafrir_laptop | or bug tzafrir |
| 16:36.05 | mafkees | lol tzafrir_laptop |
| 16:36.18 | Tili | yeah |
| 16:36.26 | tzafrir_laptop | Tili, what version is it? zaptel 2.6.12? |
| 16:36.41 | Tili | its 1.2.6 |
| 16:36.42 | mafkees | wow |
| 16:36.47 | Tili | all asteirsk 1.2 |
| 16:36.48 | Tili | stuff |
| 16:36.48 | mafkees | zaptel 2.6.12 is out ? |
| 16:36.55 | Tili | will fetch that |
| 16:36.56 | tzafrir_laptop | 1.2.13 is out |
| 16:37.05 | mafkees | ;) |
| 16:37.21 | *** join/#asterisk florz (n=florz@2002:58c6:2592:1:0:0:0:2) |
| 16:37.22 | funkmaster | ok i think i have a setting problem with my codecs, if i call someone, then there is no problem, we can hear each other, but if someone calls me, nobody hears anything, this is due to what? |
| 16:37.44 | tzafrir_laptop | Tili, if you have zaptel 1.2.6, just don't build xpp. The xpp driver there is quite old |
| 16:37.45 | funkmaster | phones are ringing of course |
| 16:38.03 | Tili | tzafrir_laptop: yeah did that. i dont need it anyway. using Sangoma HW |
| 16:38.57 | Tili | thanks |
| 16:39.05 | funkmaster | anyone can give me pointer please? |
| 16:40.32 | *** join/#asterisk stoffell (n=stoffell@d51A4D661.access.telenet.be) |
| 16:41.34 | Tili | funkmaster: cud be NAT or codec. look at asterisk console what it says. |
| 16:41.57 | tzafrir_laptop | funkmaster, if this would be a codec mismatch, the call setup would fail |
| 16:42.33 | tzafrir_laptop | if all else fails, use a sniffer (such as wireshark) |
| 16:42.44 | tzafrir_laptop | to see where the packets are actually going |
| 16:45.26 | *** join/#asterisk dahunter3 (n=dahunter@pool-71-177-150-211.lsanca.fios.verizon.net) |
| 16:50.41 | sweeper | "Rejected connect attempt from 204.8.40.221, who was trying to reach '1005@'" <-- wtf mang? |
| 16:51.18 | sweeper | I have it all set up nicely, and the remote iax box registers with the central server just fine, but that's what happens when I get outgoing calls :/ |
| 16:51.40 | *** part/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 16:51.41 | *** join/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 16:51.51 | sweeper | *try |
| 16:53.38 | *** join/#asterisk BSDTech (n=RNeese@ppp-69-238-49-231.dsl.irvnca.pacbell.net) |
| 16:54.14 | *** join/#asterisk joelsolanki (i=joelsola@220.224.42.184) |
| 17:02.58 | *** join/#asterisk Tr4d3r (n=tr4d3r@exchange.slmsistemas.net) |
| 17:03.00 | Tr4d3r | Hello |
| 17:03.51 | mafkees | hey |
| 17:05.08 | Tr4d3r | i have 4 x100p clone cards, and i'm using g.729 and gsm for sip calls, when some ext with g.729 want to use a zap trunk, hive a busy tone, but if one with gsm do, work correctly, i think is some transcoding issue, there is a way to troubleshoting this? thanks. |
| 17:05.08 | joelsolanki | Hello JT |
| 17:06.45 | Tr4d3r | Someone awake ? hehehe |
| 17:07.28 | BSDTech | you have transcoding issues |
| 17:07.41 | BSDTech | zap does not use g729 it uses ulaw |
| 17:07.57 | BSDTech | and x100p clones suck |
| 17:08.18 | BSDTech | what g729 codec are you using |
| 17:08.19 | mafkees | indeed |
| 17:08.23 | Tr4d3r | hehehe Yes, i understand, but, can i do that transcoding? |
| 17:08.30 | joelsolanki | BSDTech: Hi |
| 17:08.38 | BSDTech | you have to get a g729 license |
| 17:08.42 | Tr4d3r | i'm using the one that comes with asterisk in trixbox |
| 17:08.53 | BSDTech | and download the codec onto your system |
| 17:09.35 | BSDTech | morning Joe |
| 17:09.46 | Tr4d3r | ok i'm understand ... if i go to g.723 is the same? |
| 17:09.53 | joelsolanki | good morning |
| 17:11.52 | Qwell | Tr4d3r: asterisk doesn't support g.723 |
| 17:12.00 | *** join/#asterisk visba (n=dca[lapt@c-24-8-53-17.hsd1.co.comcast.net) |
| 17:12.13 | BSDTech | you have for g723 |
| 17:12.18 | BSDTech | yes it does |
| 17:12.24 | BSDTech | but you have to buy it |
| 17:12.36 | Qwell | There are no g723 codecs for asterisk |
| 17:12.45 | BSDTech | intel makes one |
| 17:12.53 | Qwell | for asterisk? no |
| 17:12.57 | BSDTech | yes |
| 17:13.00 | Tr4d3r | <PROTECTED> |
| 17:13.00 | Tr4d3r | -------------------------------------------------------------------------------- |
| 17:13.00 | Tr4d3r | <PROTECTED> |
| 17:13.00 | Tr4d3r | <PROTECTED> |
| 17:13.00 | Tr4d3r | <PROTECTED> |
| 17:13.09 | BSDTech | and there is a free one for home use only |
| 17:13.12 | Tr4d3r | this is my asterisk |
| 17:13.24 | Qwell | BSDTech: No, there isn't. :) |
| 17:13.26 | Qwell | Not in the US |
| 17:13.27 | BSDTech | and asterisk has a limited ione |
| 17:13.42 | joelsolanki | BSDTech: I m trying to build some wholesale platform using asterisk. so i m behind finding how many sip g729 calls can asterisk on P4 with 1 Gb ram. Concurrent calls. I am using SER as frontend and Asterisk has backend. |
| 17:14.29 | BSDTech | never used serv so could not tell you |
| 17:14.45 | *** part/#asterisk visba (n=dca[lapt@c-24-8-53-17.hsd1.co.comcast.net) |
| 17:14.58 | BSDTech | bbiab breakfast is ready |
| 17:15.36 | joelsolanki | hmm ok. but how many Asterisk can handle ? |
| 17:15.49 | Qwell | joelsolanki: it depends |
| 17:16.26 | alfredh | anyone got speex working with Asterisk 1.4 ? |
| 17:16.26 | joelsolanki | Qwell: can u plz explain little more. what factors plays role ? |
| 17:16.33 | Qwell | joelsolanki: everything the box does |
| 17:19.00 | joelsolanki | Qwell: everything box does means ? |
| 17:19.12 | mafkees | every process |
| 17:19.15 | mafkees | every daemon |
| 17:19.30 | mafkees | load, memusage,bandwidth |
| 17:19.33 | mafkees | all that |
| 17:20.13 | *** join/#asterisk bkruse_home (n=kruz@69.73.127.92) |
| 17:20.45 | Kronuz | hey, what do you think: Digium TDM400P or Sangoma A200 ? |
| 17:20.47 | joelsolanki | yes |
| 17:21.00 | bkruse_home | Kronuz: tdm400p. |
| 17:21.03 | Qwell | Kronuz: I'm biased, but Digium |
| 17:21.16 | joelsolanki | asterisk is on dedicated P4 server with 1 GB ra |
| 17:21.43 | joelsolanki | i have seen P4 1 gb ram handling 30 calls at a time g729 calls. It is concurrent calls. |
| 17:21.55 | mafkees | 30 ? |
| 17:22.05 | Kronuz | hmm.. I heard some Sangoma devices have hardware encoding of voice and hardware assisted stuff... |
| 17:22.06 | mafkees | that's not enough |
| 17:22.28 | joelsolanki | yes 30 calls. |
| 17:23.22 | joelsolanki | anything wrong ? |
| 17:24.09 | Kronuz | now, about those IP phones... if there are several VoIP protocols, how do they know or handle the calls? |
| 17:24.21 | Kronuz | do all IP phones handle many protocols? |
| 17:24.49 | *** join/#asterisk drako (n=ljd@unaffiliated/luisjose) |
| 17:26.06 | drako | hi |
| 17:26.06 | Qwell | Kronuz: most handle one protocol |
| 17:26.09 | *** join/#asterisk Dovid (n=Dovid@85.159.160.207) |
| 17:26.13 | Qwell | get SIP phones, if anything |
| 17:26.20 | Qwell | (or analog, of course) |
| 17:26.29 | Dovid | what is libiax used for ? and does it hep asterisk ? |
| 17:27.06 | Kronuz | what protocol does Asterisk use? |
| 17:27.12 | Kronuz | (for VoIP) |
| 17:27.17 | Qwell | Kronuz: several |
| 17:27.21 | Kronuz | oh |
| 17:27.27 | Kronuz | so the most common is SIP ? |
| 17:27.28 | Qwell | SIP, h323, skinny, mgcp, jabber |
| 17:27.34 | Qwell | oh, iax2, heh |
| 17:27.48 | Kronuz | why are there so many :P |
| 17:27.48 | Qwell | am I forgetting any? |
| 17:27.53 | Dovid | Qwell: what purpose does libiax serve ? |
| 17:27.59 | Kronuz | which is the better one? |
| 17:28.04 | Qwell | Dovid: it's used in third-party apps, like I think iaxcomm uses it |
| 17:28.06 | Dovid | IAX2 |
| 17:28.14 | Qwell | Kronuz: it depends |
| 17:28.20 | drako | Im trying to connect asterisk to mysql to store the cdr info. |
| 17:28.34 | drako | using mysql module |
| 17:28.39 | Dovid | Qwell: at what point in asterisk install should it be installed ? |
| 17:28.47 | Qwell | Dovid: asterisk doesn't use libiax |
| 17:28.52 | Dovid | ah ok |
| 17:29.01 | Dovid | for 3rd party apps that need to connect to asterisk ? |
| 17:29.10 | Dovid | or just general iax support ? |
| 17:29.15 | Qwell | both |
| 17:29.34 | Qwell | I think it's a bit out of date though... I know very little about it |
| 17:29.43 | Dovid | still in beta ? |
| 17:29.47 | Dovid | or release ? |
| 17:30.30 | funkmaster | is anyone using webcalldirect with asterisk? |
| 17:30.30 | Qwell | neither probably |
| 17:30.30 | funkmaster | i can't figure out the right settings for it in the extensions.conf or the sip.conf... |
| 17:30.52 | drako | do I need to set up odbc? i mean, i can't connect without use odbc to asterisk with mysql? |
| 17:31.09 | Kronuz | does anyone know if there are drivers for the TDM400P in FreeBSD ? |
| 17:31.17 | bkruse_home | funkmaster: did you voip-info it? |
| 17:31.40 | funkmaster | yeah looked there but couldn't find aynthign useful |
| 17:32.17 | Dovid | Kronuz: u can connect to mysql without the odbc connextion |
| 17:32.28 | Dovid | Kronuz: what r u trying to accomplish ? |
| 17:32.42 | Dovid | funkmaster: what r u having a problem with ? |
| 17:32.50 | Kronuz | nono, I'm not trying to connect to mysql, it's drako :P |
| 17:33.20 | Kronuz | I want to use digium's TDM400P in FreeBSD :) |
| 17:33.34 | Dovid | Kronuz: cant help u there. sorry |
| 17:33.39 | funkmaster | Dovid: i took the sip settings from their webpage, but then when i try to make a call i get this: |
| 17:33.40 | funkmaster | [Feb 10 19:24:54] WARNING[8125]: chan_sip.c:2719 create_addr: No such host: webcalldirect-out |
| 17:33.40 | funkmaster | [Feb 10 19:24:54] WARNING[8125]: app_dial.c:1081 dial_exec_full: Unable to create channel of type 'SIP' (cause 3 - No route to destination) |
| 17:33.40 | Dovid | have a look at the users list archive |
| 17:33.57 | Dovid | funkmaster: what web page ? |
| 17:33.59 | Qwell | Kronuz: I wouldn't recommend it - and you won't get any support from Digium |
| 17:34.03 | funkmaster | webcalldirect.com |
| 17:34.15 | Kronuz | Qwell: I see |
| 17:34.16 | Dovid | funkmaster: please post ur sip.conf on pb |
| 17:34.16 | bkruse_home | funkmaster: do you have a peer/user/friend in users.conf called webcalldirect-out ? |
| 17:34.16 | funkmaster | in the faq it gives u info to configure the sip device |
| 17:34.17 | Dovid | ~pb |
| 17:34.19 | jbot | i guess pb is a place to paste your stuff without flooding the channel - try http://pastebin.ca, or http://channels.debian.net/paste |
| 17:34.30 | funkmaster | in users.conf? no but in sip.conf |
| 17:34.38 | funkmaster | will pase it just a sec |
| 17:34.42 | Qwell | Kronuz: Linux is supported, and that's what all the devs use... I would highly recommend it |
| 17:34.43 | Kronuz | Qwell: the Sangoma's does support FreeBSD tho' |
| 17:35.14 | Kronuz | I just don't feel comfortable enough in Linux... but I guess I could get used to it |
| 17:35.24 | Dovid | Kronuz: never too late to learn |
| 17:35.28 | Qwell | use gentoo, you'll be right at home :p |
| 17:35.42 | Qwell | portage > ports |
| 17:35.51 | bkruse_home | funkmaster: either or |
| 17:35.54 | Qwell | of course, colonoscopy > ports, but... |
| 17:35.55 | *** join/#asterisk teknoprep (n=chris@unaffiliated/teknoprep) |
| 17:35.56 | Kronuz | Qwell: really? is it the closest to FreeBSD? |
| 17:35.57 | Dovid | funkmaster: did u out it in pb ? |
| 17:36.03 | bkruse_home | users.conf hassip and hasregister == sip.conf friend entry |
| 17:36.09 | Qwell | Kronuz: nah, slackware is probably the "closest" |
| 17:36.18 | Qwell | but, I think you'd prefer gentoo over slack |
| 17:36.30 | Kronuz | I'll give it a try |
| 17:36.32 | toresbe | Qwell: I'd disagree. |
| 17:36.42 | Qwell | toresbe: which part? |
| 17:36.53 | toresbe | Qwell: colonoscopy > ports. :) |
| 17:36.55 | Qwell | I said at least 4 things which might be considered "unpopular" :) |
| 17:37.02 | Qwell | ahh, no, I'm gonna stick to my guns on that one |
| 17:37.09 | Qwell | have you ever actually USED ports? |
| 17:37.10 | tzafrir_laptop | Kronuz, there are zaptel drivers for FreeBSD. Not for other BSDs, I believe |
| 17:37.28 | Kronuz | zaptel? |
| 17:37.29 | Qwell | I mean, come on, it still uses cvs |
| 17:37.31 | kink0 | hello, anybody gets asterfax running ? I got a lot of problems compiling app_txfax.so |
| 17:37.48 | Qwell | tzafrir_laptop: but like I said, it's unsupported |
| 17:37.51 | tzafrir_laptop | drivers for the card you mentioned |
| 17:38.26 | Kronuz | tzafrir_laptop: do zaptel make the drivers? |
| 17:38.45 | Kronuz | 'cause the card is digium's, isn't it? |
| 17:38.50 | Qwell | Kronuz: zaptel is maintained by Digium |
| 17:38.55 | Kronuz | oh |
| 17:38.56 | *** join/#asterisk Vec (n=Vector@dsl-243-86-187.telkomadsl.co.za) |
| 17:38.59 | Vec | Anyone know where I can get some extension.conf examples, I find the best way to learn how to create a neat dial plan is to learn from others ? |
| 17:39.04 | Qwell | freebsd's zaptel is not the same code base though, really |
| 17:39.07 | funkmaster | Dovid: this is it http://pastebin.ca/349247 |
| 17:39.17 | Qwell | I mean, some of it is, I guess, but it's quite different |
| 17:39.19 | Dovid | Vec: make samples in /usr/src |
| 17:39.28 | tzafrir_laptop | Kronuz, no. See http://www.zapatatelephony.org/ for some history |
| 17:39.43 | Dovid | oop |
| 17:39.54 | Dovid | in the asterisk directory that u used to compile asterisk |
| 17:40.48 | Dovid | funkmaster: sorry didnt mention extensions.conf as wel |
| 17:40.50 | Dovid | well* |
| 17:40.55 | funkmaster | ok just a sec |
| 17:41.22 | tzafrir_laptop | Kronuz, However unlike the Tormenta cards referred there, the TDM400P is Digium's |
| 17:41.47 | toresbe | Is "Tormenta" really that good a name for a product? |
| 17:41.48 | Kronuz | but still Zaptel makes drivers for it.... I see |
| 17:43.11 | sweeper | toresbe: why not? |
| 17:43.20 | *** join/#asterisk angryuser (n=Miranda@d05v-212-195-196-21.d4.club-internet.fr) |
| 17:43.23 | funkmaster | Dovid: ok both r in this now http://pastebin.ca/349253 |
| 17:43.30 | mmbl13 | any ideas how to build the libipt_random.so module on 2.6.9-42.0.3.EL-i686 ? |
| 17:43.31 | sweeper | plenty of products with Storm in the name |
| 17:43.38 | toresbe | sweeper: Rather strong connotation to "torment", not? |
| 17:43.45 | mmbl13 | cannot find IPT_RANDOM in the kernel-config |
| 17:43.52 | Qwell | toresbe: there's clearly an a at the end :p |
| 17:44.04 | Qwell | mmbl13: what's that got to do with asterisk? |
| 17:44.25 | mmbl13 | Qwell: oh sorry, wrong channel, wanted to go to window 4, #centos - sorry |
| 17:44.25 | angryuser | evening, does ChanIsAvail() For sip peer takes in consideration call-limit option? |
| 17:44.25 | Qwell | toresbe: apply your logic to the word "therapist" |
| 17:44.26 | toresbe | Qwell: Well, would you buy a frozen pizza named "disgusta"? |
| 17:44.43 | sweeper | interestingly, Tormenta actually does mean "(he/she/it) torments" as well as "storm" in spanish |
| 17:45.00 | Kronuz | what's the "Asterisk Appliance" |
| 17:45.01 | tzafrir_laptop | mmbl13, is it part of pwlib (of openh323)? |
| 17:45.10 | sweeper | Kronuz: pretty branding \o\ |
| 17:45.11 | Qwell | Kronuz: sec, I'll get a link |
| 17:45.18 | angryuser | quit |
| 17:45.19 | Qwell | it's slick :) |
| 17:45.20 | toresbe | Qwell: "Therapist" isn't a brand name, and does not have a double entendre unless you actually change the two words |
| 17:45.26 | Qwell | Kronuz: http://www.digium.com/en/products/hardware/aadk.php |
| 17:45.39 | mmbl13 | tzafrir_laptop: don't think so, it is an iptables module |
| 17:45.55 | Kronuz | is it like a server with Asterisk installed? like a hardware Asterisk box? |
| 17:46.02 | tzafrir_laptop | mmbl13, so it is something .ko |
| 17:46.02 | Qwell | Kronuz: see link |
| 17:46.09 | Qwell | that explains it all |
| 17:46.14 | Kronuz | ok |
| 17:46.38 | Vec | Dovid : thats are a little to simple for what I am looking for. |
| 17:46.43 | bkruse_home | insmod + ztcfg == friend |
| 17:46.50 | mmbl13 | tzafrir_laptop: i need to build that module first, but my kernel config ha sno entry |
| 17:47.04 | *** join/#asterisk errr (n=errr@fedora/errr) |
| 17:47.10 | Dovid | funkmaster: whats the s doing there in the begining ? |
| 17:47.15 | tzafrir_laptop | bkruse_home, why do you use insmod directly? |
| 17:47.21 | Dovid | also did u reload ur asterisk after making changes |
| 17:47.23 | Qwell | insmod is hot |
| 17:47.28 | Dovid | Vec: did u read the book ? |
| 17:47.30 | tzafrir_laptop | use modprobe. run ztcfg only after all modules were probed |
| 17:47.31 | bkruse_home | tzafrir_laptop: why not! |
| 17:47.32 | Dovid | ~book |
| 17:47.36 | jbot | from memory, book is a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 17:47.36 | bkruse_home | Qwell: agreed. |
| 17:47.51 | bkruse_home | tzafrir_laptop: it runs ztcfg also, but its misleading if you do not have something already modprobed |
| 17:47.53 | bkruse_home | example. |
| 17:48.10 | bkruse_home | modprobe zaptel && modprobe zttranscode will return a ztcfg error because you might have a quad span card |
| 17:48.17 | bkruse_home | thus, a support call originates for no reason. |
| 17:48.19 | bkruse_home | :X |
| 17:48.23 | tzafrir_laptop | bkruse_home, because then you'll need to insmod zaptel manually, and ditto for ztdynamic, xpp and probably some others |
| 17:48.25 | *** join/#asterisk adker (n=chatzill@74-33-198-79.br1.glv.ny.frontiernet.net) |
| 17:48.32 | *** join/#asterisk nuonguy (n=john@c-24-6-175-26.hsd1.ca.comcast.net) |
| 17:48.33 | kink0 | ~asterfax |
| 17:48.39 | *** join/#asterisk santiago (n=santiago@debian/developer/santiago) |
| 17:48.42 | funkmaster | Dovid: yes reloaded it, and which s do u mean? |
| 17:48.48 | Kronuz | Qwell: I still don't get what exactly is the "Asterisk Appliance" :( |
| 17:48.56 | Qwell | Kronuz: it's an appliance which runs asterisk |
| 17:48.58 | Dovid | exten => _5.,1,Dial(SIP/s{EXTEN:1}@webcalldirect-out,30,r) |
| 17:49.00 | bkruse_home | tzafrir_laptop: nah, just zaptel and zttranscode, tis all youll ever need |
| 17:49.08 | bkruse_home | Qwell: props for the answer |
| 17:49.32 | Kronuz | looks to me that instead of having a server with Asterisk installed one could have the Asterisk Appliance without any server, just directly connected to the FXO lines and Internet |
| 17:49.36 | tzafrir_laptop | the error from modprobe is because zaptel installs a really dumb configuration in modprobe.conf to run ztcfg automatically |
| 17:49.39 | Qwell | bkruse_home: I swear I'll get that rrdtool stuff done for you soon :p |
| 17:49.55 | funkmaster | Dovid: ah ok will change it and try |
| 17:50.02 | Qwell | Kronuz: technically, the appliance is a "server" |
| 17:50.10 | Qwell | it "serves" packets |
| 17:50.14 | Dovid | funkmaster: i dont think that is the issue, but worth trying |
| 17:50.27 | angryuser | Does ChainIsavail take in consideration call-limit option for peers? |
| 17:50.31 | bkruse_home | Qwell: Really?!?! |
| 17:50.32 | Dovid | funkmaster: nm |
| 17:50.33 | bkruse_home | im excited :D |
| 17:50.38 | Qwell | bkruse_home: some day - totally |
| 17:50.39 | Kronuz | Qwell: that's what I mean, instead of having a regular computer acting as an Asterisk server, you just have the Asterisk Appliance |
| 17:50.40 | Dovid | u have s instea d of $ |
| 17:50.42 | sweeper | the asterisk appliance is a facking asterisk box |
| 17:50.43 | Kronuz | right? |
| 17:50.45 | bkruse_home | Qwell: :D |
| 17:50.45 | Qwell | Kronuz: correct |
| 17:50.45 | bkruse_home | <3 |
| 17:50.52 | sweeper | with some fxo and fxs ports |
| 17:50.54 | Kronuz | :) |
| 17:50.56 | tzafrir_laptop | bkruse_home, plus, even 'modprobe xpp_usb; ztcfg' will not work: xpp takes some time until it registers with zaptel. Or it may require manual operation to regiter with zaptel, to ensure registration in the correct order |
| 17:51.00 | bkruse_home | its pretty pimp if i might say. |
| 17:51.06 | sweeper | it's jsut a really small computer :P |
| 17:51.10 | sweeper | and no moving parts |
| 17:51.15 | bkruse_home | tzafrir_laptop: exactly, thats why i use insmod |
| 17:51.15 | Qwell | sweeper: exactly |
| 17:51.22 | Kronuz | now there's the Asterisk Appliance Developer Kit |
| 17:51.24 | sweeper | which has pros and cons |
| 17:51.25 | bkruse_home | modprobe wct4xxp wont work |
| 17:51.39 | bkruse_home | because its like "omg udev sucks and /dev/zap/channelshere does not exists!!!" |
| 17:51.39 | sweeper | you won't be able to do much transcoding |
| 17:51.39 | Kronuz | which I'm not sure why it's a DK |
| 17:51.40 | *** part/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 17:51.41 | *** join/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 17:51.44 | Dovid | but the devel kit has not been released yet as a stnad alone device :( |
| 17:51.49 | Qwell | Kronuz: in the retail product, it will be basically the same hardware |
| 17:51.53 | sweeper | Kronuz: because they're not selling it yet |
| 17:51.55 | tzafrir_laptop | bkruse_home, so you work around a bug inflicted directly by zaptel's misconfiguration. Instead of fixing that misconfiguration |
| 17:51.57 | Dovid | i know |
| 17:51.57 | Qwell | the aadk comes with training/support/etc |
| 17:52.07 | Dovid | hence the :( :( :( and may i add :( |
| 17:52.09 | Kronuz | oh |
| 17:52.15 | Qwell | it's basically to allow devs to add stuff before the actual release |
| 17:52.19 | Dovid | Qwell: any idea when it wll be out ? |
| 17:52.22 | Kronuz | sweeper: it's a "work in progress" |
| 17:52.28 | tzafrir_laptop | Now am I the only one who thinks this is a bug? |
| 17:52.34 | Qwell | Dovid: I am unable to comment. :) |
| 17:52.34 | sweeper | Kronuz: exactly. |
| 17:52.52 | Kronuz | so later sale the Asterisk Appliance as a hardware fully tested :) |
| 17:53.01 | Qwell | Kronuz: right |
| 17:53.09 | bkruse_home | tzafrir_laptop: no, i work around the time it takes to make and allocate /dev/zap/channels |
| 17:53.10 | Dovid | Qwell: so when its ready u guys will supprise us - how seet |
| 17:53.11 | *** join/#asterisk PakiPenguin (n=uppal@linuxpakistan/admin/pakipenguin) |
| 17:53.16 | PakiPenguin | hello everyone |
| 17:53.26 | funkmaster | Dovid:indeed that did not solve the problem |
| 17:53.32 | tzafrir_laptop | bkruse_home, you work around the fact that you don't use a proper init.d script |
| 17:53.36 | Dovid | funkmaster: did u see my last ? |
| 17:53.42 | tzafrir_laptop | you don't have to run ztcfg immediately |
| 17:53.44 | Dovid | u needed a $ sign instead of the s |
| 17:53.46 | bkruse_home | tzafrir_laptop: thats rhel4 for you. |
| 17:54.10 | bkruse_home | thats what the default modprobe.d configuration is for zaptel, as of now |
| 17:54.10 | Qwell | bkruse_home: a big bug in the form of a distro? |
| 17:54.10 | bkruse_home | its an OS problem |
| 17:54.10 | sweeper | way too low end for my markets anyways. can't think of an instance where I'd prefer to have an AA instead of a full * 1u, where simply using SIP phones wouldn't suffice |
| 17:54.10 | bkruse_home | Qwell: I couldnt have said it better |
| 17:54.37 | Kronuz | Qwell: I see one can get the Asterisk Appliance hardware with the AADK |
| 17:54.43 | Dovid | funkmaster: also did u try to ping the domain - makybe ur box cant get to it ? |
| 17:54.53 | tzafrir_laptop | Qwell, it is a bug inflicted by zaptel's installer. |
| 17:54.53 | Qwell | Kronuz: well, yeah, an appliance is part of the aadk |
| 17:54.57 | funkmaster | Dovid: yes i changed the s to $ |
| 17:55.06 | funkmaster | ok will try |
| 17:55.09 | Kronuz | Qwell: not of the standard one, or is it? |
| 17:55.13 | sweeper | does anyone make a standalone t.38 ATA-type device? |
| 17:55.16 | tzafrir_laptop | Qwell, in fact, packages of zaptel do not include this modprobe config |
| 17:55.25 | Qwell | Kronuz: the kit is the hardware, training, support, etc |
| 17:55.41 | Dovid | funkmaster: try to ping it - the error is that it cant get to that host - or that it dosent exist - can u get to any other hosts |
| 17:55.46 | bkruse_home | zaptel goes in /etc/modprobe.d |
| 17:55.58 | Kronuz | but there's professional, administrator and standard (they all include the appliance?) |
| 17:56.00 | bkruse_home | the modprobe script n e wayz, not the whole zaptel, obvious |
| 17:56.09 | sweeper | oh, and good news, chan_tdmoip.so should be in the works in a month or two~ |
| 17:56.19 | bkruse_home | Kronuz: the appliance is just pimp, I have 3 ! |
| 17:57.01 | funkmaster | Dovid:indeed not pingable, guess the info on the site is wrong then.. |
| 17:57.20 | Dovid | funkmaster: then that was ur issue |
| 17:57.30 | bkruse_home | Qwell: is one day, possibly, a day in the near future? (near future == 1 month) ? |
| 17:57.47 | Qwell | 1 month, give or take 1 year |
| 17:57.57 | funkmaster | Dovid: hm ok thx a lot, that is already something, does anyone know the correct settings for webcalldirec then? |
| 17:58.03 | Dovid | nope |
| 17:58.07 | Dovid | whats on thier site ? |
| 17:58.19 | matiasds19 | some who colud know why my asterisk is droping incoming calls from a sip provider? |
| 17:58.25 | matiasds19 | here is my sip debug from the incoming call |
| 17:58.27 | matiasds19 | http://www.pastebin.ca/348640 |
| 17:58.40 | funkmaster | sip.webcalldirect.com |
| 17:58.50 | bkruse_home | Qwell: :X! |
| 17:58.57 | bkruse_home | its fine, i got tons of stuff to do now |
| 17:59.05 | funkmaster | that's what they have on their site, on a forum i also found that connectionserver.webcalldirect.com should work, but does not either.. |
| 17:59.10 | bkruse_home | im just dev'n on my own box, hoping that i dont include any commands that arent there! |
| 17:59.42 | funkmaster | guess i will try and mail them, but those wankers barely reply to anything, annoying betamax... |
| 18:00.19 | bkruse_home | haha@ the word wankers |
| 18:00.24 | bkruse_home | permission to start using that funkmaster? |
| 18:00.37 | Dovid | funkmaster: its resolving on my dns |
| 18:00.46 | Dovid | its an issue on ur dns |
| 18:00.53 | Kronuz | okay, think one have two Asterisk boxes, one in the US and one in Italy (both with a real telephone line connected to a FXO module); can one have Asterisk answer the call in Italy, convert the information to VoIP and send the call to a phone in the US using the line connected to the other Asterisk box in the US for "free" without using an external VoIP service provider? |
| 18:01.05 | funkmaster | Dovid: which one is resolvin sip or the other? |
| 18:01.06 | Dovid | funkmaster: have a look here |
| 18:01.07 | Dovid | http://www.dnsstuff.com/tools/dnstime.ch?name=sip.webcalldirect.com&type=A |
| 18:01.20 | bkruse_home | Kronuz: iax!!! |
| 18:01.29 | Kronuz | iax? |
| 18:01.52 | bkruse_home | iax2! |
| 18:01.55 | Kronuz | Asterix's protocol? |
| 18:01.55 | bkruse_home | inter-asterisk-eXchange |
| 18:01.58 | bkruse_home | yes |
| 18:02.05 | bkruse_home | its SUPER easy to setup, and its all over the interweb |
| 18:02.17 | funkmaster | Dovid: that's weird, hm so how can it be that i can not ping them? |
| 18:02.34 | bkruse_home | you can have your italy asterisk box answer the call, and its context will be exten => 1,1,Dial( |
| 18:02.34 | matiasds19 | some who colud know why my asterisk is droping incoming calls from a sip provider? |
| 18:02.36 | bkruse_home | crap |
| 18:02.40 | Dovid | funkmaster: u can try to use the IP instea |
| 18:02.45 | drako | How I can connect asterisk with mysql without use odbc? |
| 18:02.57 | funkmaster | Dovid: good idea, trying now |
| 18:03.00 | Dovid | drako: asteirsk-addons |
| 18:03.10 | bkruse_home | exten => 1,1,Dial(IAX2/theusbox/520@italy-incoming) |
| 18:03.10 | Dovid | funkmaster: only issue ur gona have is if they switch IP's |
| 18:03.23 | Kronuz | bkruse_home: oh, so it is possible... and then if I want to make a call to other country or city or receive a call from other country or city, then there's when I need a VoIP service provider, right? |
| 18:03.37 | Kronuz | (to avoid long distance calls) |
| 18:03.42 | bkruse_home | no |
| 18:03.50 | Kronuz | o_O |
| 18:03.51 | bkruse_home | is that box connected to the internet? |
| 18:04.06 | bkruse_home | are the boxen connected to the internet, rather |
| 18:04.22 | bkruse_home | Qwell: you think the rrdtool stuff looks cool? worth some time? |
| 18:04.35 | Kronuz | both Asterisk boxes would |
| 18:04.36 | Qwell | yeah... I don't have much though :( |
| 18:04.42 | funkmaster | Dovid: can also not ping the IP |
| 18:04.50 | bkruse_home | Qwell: me either, its stuff i did in my off time |
| 18:04.52 | Dovid | funkmaster: than its an isp issue |
| 18:04.59 | Dovid | funkmaster: where r u located ? |
| 18:05.03 | Dovid | located* |
| 18:05.09 | funkmaster | Dovid: in NL |
| 18:05.09 | bkruse_home | Kronuz: then IAX2 goes over the internet |
| 18:05.13 | Dovid | hmm |
| 18:05.15 | bkruse_home | so you DONT need a voip provider |
| 18:05.17 | funkmaster | Dovid: but i think the issues is something esle |
| 18:05.19 | Dovid | leme see if i can ping from here |
| 18:05.24 | funkmaster | cuz they have free calls on their site |
| 18:05.26 | bkruse_home | you only need a VOIP provider if you need a number |
| 18:05.27 | Dovid | funkmaster: it can be a FW issue |
| 18:05.28 | funkmaster | without registering |
| 18:05.32 | bkruse_home | aka, you dont have zap hardware |
| 18:05.38 | funkmaster | and that does not work for me anymore as i used them all |
| 18:05.38 | *** join/#asterisk anthm (n=anthm@CPE-72-131-113-50.wi.res.rr.com) |
| 18:05.38 | *** mode/#asterisk [+o anthm] by ChanServ |
| 18:05.41 | funkmaster | so now i have to login |
| 18:05.44 | Qwell | bkruse_home: ahh |
| 18:05.53 | Kronuz | bkruse_home: but I'd have them in certain cities in the US and Italy, and if I want to call a real phone number outside those cities I'd have to make the long distance call from the line in the boxes to the real phone |
| 18:05.54 | funkmaster | so maybe they also block my pings etc without a login from my ip |
| 18:06.06 | bkruse_home | ya |
| 18:06.08 | bkruse_home | OR |
| 18:06.14 | Dovid | possibly |
| 18:06.20 | bkruse_home | have the local box call the long distance box over the internet, to a local phone line. |
| 18:06.22 | Dovid | have a look on thier site at thier faq |
| 18:06.27 | Kronuz | (same if somesone calls from other different city from where the asterisk box is) |
| 18:06.38 | funkmaster | Dovid: ok don't worry i'll figure it out, thx a lot for ur help though :D |
| 18:06.48 | Dovid | np |
| 18:07.05 | bkruse_home | Kronuz: its SO easy! |
| 18:07.08 | bkruse_home | lol |
| 18:07.25 | bkruse_home | do you need me to write it out for you in a vector drawing program? |
| 18:07.27 | *** join/#asterisk Koshatul (n=evangeli@ppp188-33.static.internode.on.net) |
| 18:07.32 | *** join/#asterisk h0 (n=fakhir@unaffiliated/fakhir) |
| 18:07.39 | bkruse_home | :P |
| 18:07.41 | Kronuz | hehe |
| 18:08.07 | bkruse_home | Kronuz: does that make sense though? |
| 18:08.18 | bkruse_home | if not, i can throw down a little more in detail rundown real quick, no big |
| 18:08.45 | Kronuz | bkruse_home: example, say I have a box in San Diego, CA; one in Rome, Italy and someone from Madrid, Spain wants to call someone in Argentina from and to a real phone |
| 18:09.05 | Kronuz | then there's when I mean I'd need a VoIP service, right? |
| 18:09.11 | Dovid | ~itsp |
| 18:09.14 | jbot | extra, extra, read all about it, itsp is Internet Telephony Service Provider. An ITSP is a "VoIP Phone Company" |
| 18:09.29 | Kronuz | :P |
| 18:09.31 | bkruse_home | Kronuz: i thought you said you had zap hardware? |
| 18:09.49 | Kronuz | zap hardware? is that what the FXO modules are? |
| 18:10.05 | bkruse_home | well, ya..... |
| 18:10.11 | Kronuz | (I'm really new to all this, I just found this PBX thing this morning) |
| 18:10.15 | bkruse_home | fxo modules plug into phone lines to RECEIVE calls |
| 18:10.18 | bkruse_home | ha, cool |
| 18:10.42 | drako | i keep getting this |
| 18:10.43 | drako | ............................................................................................Feb 10 15:10:41 WARNING[5018]: cdr_odbc.c:257 odbc_load_module: cdr_odbc: Unable to load config for ODBC CDR's: cdr_odbc.conf |
| 18:10.56 | Kronuz | bkruse_home: I think I understand the basics tho' |
| 18:11.00 | bkruse_home | drako: touch /etc/asterisk/cdr_odbc.conf? |
| 18:11.00 | drako | i don't want to use odbc i want to use mysql |
| 18:11.14 | bkruse_home | so, you call the LOCAL phone line, and then have asterisk dial IAX to the other box, in another country, and its free!!!! |
| 18:11.20 | bkruse_home | gota go though, you wana email me? |
| 18:11.20 | Dovid | drako: did u install asterisk-addons ? |
| 18:11.30 | bkruse_home | PM me, and i cna give you a lil more of a rundown |
| 18:12.13 | Kronuz | bkruse_home: but I'd have to have a box in every source/destination with a local phone line |
| 18:12.29 | Dovid | kronuz; nope |
| 18:12.29 | Kronuz | and if I don't that's when I'd pay someone who does have the infrastructure in many cities |
| 18:12.39 | Kronuz | aghrr!! I'm confused |
| 18:12.44 | Dovid | kronuz: have u read the book ? |
| 18:12.47 | Dovid | ~book |
| 18:12.49 | jbot | i heard book is a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 18:12.52 | *** join/#asterisk foxxtrot (n=craig@c-67-185-0-172.hsd1.wa.comcast.net) |
| 18:12.53 | Kronuz | the Asterisk book? |
| 18:12.57 | Kronuz | I just downloaded it :) |
| 18:13.08 | Dovid | kronuz: start reading.... |
| 18:13.09 | drako | Dovid, im using debian packages |
| 18:13.12 | Dovid | it will fill in a lot fo ru |
| 18:13.13 | Kronuz | I guess I'll read it :) |
| 18:13.31 | bkruse_home | Kronuz: then there is no need for a VOIP provider |
| 18:13.34 | *** part/#asterisk bkruse_home (n=kruz@69.73.127.92) |
| 18:13.44 | drako | i don't have cdr_mysql.so, so i guess ill be forced to use odbc then. |
| 18:13.55 | Kronuz | thanks... I'll be back with more questions after I've checked the book :D |
| 18:14.53 | Dovid | np |
| 18:15.22 | Dovid | draki: debian = unix, correct ? |
| 18:16.36 | drako | Dovid, yes. |
| 18:16.44 | drako | Dovid, but im not installing asterisk from source |
| 18:18.21 | tzafrir_laptop | drako, what debian? |
| 18:18.41 | tzafrir_laptop | etch? |
| 18:18.43 | angryuser | when i do ChainIsAvail on Unreachable peer with j option instead of jumping n+101 asterisk is still trying to call by that peer, any help? |
| 18:19.11 | angryuser | aster 1.4 |
| 18:19.30 | drako | tzafrir_laptop, yes |
| 18:19.31 | tzafrir_laptop | try the packages from pkg-voip, I guess |
| 18:19.42 | drako | tzafrir_laptop, im using it |
| 18:20.04 | angryuser | and.. Chain is avail returns allways 0 on ${AVAILSTATUS} |
| 18:20.17 | tzafrir_laptop | any change modules.conf explicitly unloads res_mysql_config.so? |
| 18:20.46 | tzafrir_laptop | angryuser, maybe this was removedin 1.4? |
| 18:20.48 | drako | tzafrir, i don't that this module the .so file |
| 18:21.03 | tzafrir_laptop | drako, what exactly is the problem? |
| 18:21.17 | tzafrir_laptop | dpkg -l asterisk-mysql |
| 18:21.24 | tzafrir_laptop | dpkg -l asterisk-mysql ^grep ^i |
| 18:21.37 | angryuser | tzafrir_laptop:how to verify status of Sip peer on aster 1.4 then? |
| 18:22.33 | drako | tzafrir_laptop, no packages |
| 18:23.12 | tzafrir_laptop | angryuser, you can do that through the value of AVAILCHAN I guess. But you're right. j is still listed as an option |
| 18:24.01 | mafkees | talking about chanisavail |
| 18:24.16 | mafkees | it does not tell you the channel is not available when a sip phone is not registered |
| 18:24.31 | mafkees | how can I check that without the dundi stuff |
| 18:24.38 | angryuser | http://bugs.digium.com/view.php?id=7433, |
| 18:24.39 | mafkees | I have 3 boxes in 3 locations |
| 18:24.45 | angryuser | found it |
| 18:24.50 | tzafrir_laptop | drako, I do see asterisk-mysql packages in the pool of pkg-voip.buildserver.net |
| 18:24.51 | mafkees | and people take phones from 1 location to the other |
| 18:25.24 | mafkees | I want to know wether I can call locally or use IAX to some remote box |
| 18:25.25 | *** part/#asterisk sevard (n=sev@adsl-71-129-115-244.dsl.irvnca.pacbell.net) |
| 18:25.25 | *** join/#asterisk karmatronic (n=karmatro@84.77.170.211) |
| 18:25.32 | mafkees | I thought chanisavail could help me |
| 18:25.47 | drako | deb http://pkg-voip.buildserver.net/debian etch main |
| 18:25.52 | *** join/#asterisk lorinc (n=ang@caracas-4604.adsl.interware.hu) |
| 18:25.53 | mafkees | but the variables are always the same, wether the phone is registered or not |
| 18:25.55 | drako | tzafrir_laptop, thats my entry in source.list |
| 18:26.28 | drako | wait |
| 18:26.34 | drako | apt-get is getting it |
| 18:26.43 | drako | i think i didnt refresh dselect |
| 18:26.45 | Kronuz | generally, how stable is Asterisk's code in the svn? |
| 18:26.47 | tzafrir_laptop | your entire sources.list? you should have standrd debian sources, I hope |
| 18:26.59 | tzafrir_laptop | don't use dselect. Use apt-get and/or aptitude |
| 18:27.03 | drako | tzafrir_laptop, that too |
| 18:27.18 | Qwell | dselect...ugh |
| 18:27.21 | Qwell | what an abomination |
| 18:27.26 | drako | heh ok but kinda offtopic whats wrong with dselect? |
| 18:27.35 | Qwell | it's horrible |
| 18:27.53 | tzafrir_laptop | I think it implements a number of things independently, and hence buggy |
| 18:27.56 | Qwell | takes like 9 hours to pick the packages you want, then if it fails...TFB! start over |
| 18:27.59 | mafkees | everything is wrong with dselect |
| 18:28.12 | drako | Ok |
| 18:28.12 | tzafrir_laptop | and it doesn't feature a minesweeper game |
| 18:28.14 | drako | get it. |
| 18:28.20 | Qwell | tzafrir_laptop: does apt? |
| 18:28.31 | drako | just last time i used aptitude it unstalled system v init script it was a total mess |
| 18:28.39 | drako | but its cool i think ill keep trying. |
| 18:28.54 | tzafrir_laptop | Apt is basically OK. Much more limited support of the moo command than aptitude, though |
| 18:29.55 | tzafrir_laptop | aptitude tends to be smarter. For instance, it tracks the packages you have installed directly, vs. ones that you installed because they were a dependency |
| 18:30.24 | mafkees | sometimes that sux |
| 18:30.30 | mafkees | specially with -dev stuff |
| 18:32.45 | *** join/#asterisk dansmith (n=dan@gw0.danplanet.com) |
| 18:39.50 | *** join/#asterisk puzzled (n=patrick@puzzled.xs4all.nl) |
| 18:40.13 | dansmith | is anyone using an SPA3102 with asterisk? |
| 18:40.50 | dansmith | I have a generic FXO card, and was wondering if the SPA3102 would be a better FXO or if I should stick with what I have |
| 18:43.06 | angryuser | tzafrir_laptop:after a bit of sarch, whatever the state of my peer ChainIsavail's ${availstatus} = 0 in any cases, it is broken;((( |
| 18:44.00 | tzafrir_laptop | dansmith, "generic fxo card": X100P or alike? |
| 18:44.14 | dansmith | tzafrir_laptop: yup |
| 18:44.47 | tzafrir_laptop | the SPA3102 has an FXS port. Do you need that? |
| 18:45.27 | dansmith | well, I need to get an FXS, so I was going to get a PAP2-NA, but then I saw the SPA3102 and thought it might be a better FXO, in addition to supplying my FXS needs |
| 18:46.06 | dansmith | recommendations are appreciated :) |
| 18:47.20 | [TK]D-Fender | dansmith : It works pretty well |
| 18:47.56 | *** part/#asterisk BSDTech (n=RNeese@ppp-69-238-49-231.dsl.irvnca.pacbell.net) |
| 18:48.06 | dansmith | [TK]D-Fender: ok, but is it better than my X100P? because having two FXS would be cool, and I'd rather have that if the FXO on the SPA3102 isn't going to be any better |
| 18:48.34 | [TK]D-Fender | dansmith : What problems are you experiencing with your X100P? |
| 18:49.05 | dansmith | well, nothing really.. the quality doesn't seem perfect, but I guess it's fine |
| 18:49.25 | dansmith | I've seen people claim that they're sub-optimal, so I just thought I might be better in the long run with something else |
| 18:49.37 | dansmith | I've only had this setup for about 48 hours, so I don't have long-term experience with it |
| 18:49.59 | dansmith | I was just going to buy an FXS interface and wanted some advice about whether I should get an FXO+FXS |
| 18:50.02 | [TK]D-Fender | Neither will seem perfect. That'll cost you. If you need an FXS, then get the 3102, you can't go wrong, and if you don't like the FXO on it, you can inor it for now and just use the FXS |
| 18:50.09 | dansmith | if people are generally happy with the X100P, then I'll stick with it for a while |
| 18:50.16 | [TK]D-Fender | Or save the unit for remote deployment which is kinda cool |
| 18:50.34 | [TK]D-Fender | dansmith : Qustion is, are YOU happy enough with it right now? |
| 18:50.58 | dansmith | heh, well, I haven't made enough phone calls to know, I guess. It seems like I am, though, yes :) |
| 18:51.01 | [TK]D-Fender | dansmith : But again, if you still need jsut 1 more FXS, then get the 3102. |
| 18:51.16 | drako | ok i need some stuff to clean up. |
| 18:51.17 | [TK]D-Fender | dansmith : and treat it accordingly. |
| 18:51.19 | drako | http://pastie.caboo.se/39374 |
| 18:51.26 | drako | why i can get rid of these warnings? |
| 18:51.31 | dansmith | well, I don't have any FXS now, and it would be cool to have two, so I can have a separate extension in the office |
| 18:51.40 | *** part/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 18:51.41 | *** join/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 18:52.12 | [TK]D-Fender | drako : Congrats... no DSN for ODBC or direct MySQL connection... |
| 18:52.32 | [TK]D-Fender | drako : the errors are pretty obvious... |
| 18:52.44 | drako | [TK]D-Fender, im using cdr_mysql.so |
| 18:52.52 | matiasds19 | hey guys i need help with my asterisk...someone can help? |
| 18:52.56 | drako | or trying to use. |
| 18:53.09 | *** join/#asterisk Gunnar (n=gunnar@nat.sigmasoft.com) |
| 18:54.03 | *** join/#asterisk dmz (n=dmz@64.20.147.237) |
| 18:54.27 | dmz | hey y'all, anyone know if there is anyone bulding 1.4 packages for debian? |
| 18:54.41 | Qwell | tzafrir is/was, I believe |
| 18:55.06 | dmz | hmm i'll go see what google finds w/that :) |
| 18:56.42 | tzafrir_laptop | dmz, still busy with 1.2, hope to get to 1.4 later on. But I was hope 1.4.1 to come soon and save me the need for a few patches |
| 18:56.51 | dmz | ah |
| 18:57.27 | dmz | I was looking into web-meetme and it says it's for 1.4 so I thought i'd see what 1.4 has that's worth having |
| 18:57.49 | dmz | is it worth me just compiling it myself, is there any reason to not use 1.4? |
| 18:59.03 | dansmith | Has anyone used one of these: http://www.voiplink.com/BudgeTone_101_p/grstrmbt101.htm ? |
| 18:59.23 | dansmith | I'm sure it's very cheap-o, but if it would work to play around with, that'd be cool |
| 19:01.42 | *** join/#asterisk h0 (n=fakhir@unaffiliated/fakhir) |
| 19:04.34 | [TK]D-Fender | matiasds19 : Provide that pastebin again you gave last night |
| 19:05.02 | [TK]D-Fender | dansmith : GrandSuck should be avoided with extreme prejudice |
| 19:05.14 | dansmith | hehe, good to know :) |
| 19:05.28 | dansmith | any recommendations for the cheapest starter SIP phone? :) |
| 19:05.34 | Qwell | unless you're looking to buy a phone for an 8 year old girl |
| 19:05.46 | dansmith | heh |
| 19:06.16 | mafkees | hhmm |
| 19:06.27 | mafkees | is there a fwd irc channel ? |
| 19:06.48 | Qwell | danalien: I personally like the polycom IP430 |
| 19:06.58 | Qwell | it's relatively cheap I believe |
| 19:07.12 | Qwell | there's also the IP301, which has a few less features |
| 19:07.52 | [TK]D-Fender | dansmith : Where are you located? |
| 19:07.56 | dansmith | [TK]D-Fender: in the US |
| 19:08.20 | dansmith | the IP430 is $150 at first glance.. is that the ballpark we're talking about for the cheapest acceptable IP phone? |
| 19:08.22 | [TK]D-Fender | dansmith : Ok, not planning on PoE are you? |
| 19:08.31 | Qwell | danalien: I think the 301 is slightly less |
| 19:08.37 | Qwell | listen to [TK]D-Fender though, he's the man |
| 19:08.47 | dansmith | if so, I think I better use just FXS extensions for a while so I can convince the wife :) |
| 19:08.53 | dansmith | [TK]D-Fender: nope |
| 19:09.11 | [TK]D-Fender | dansmith : What are you using now? |
| 19:09.19 | dansmith | ekiga :P |
| 19:09.20 | [TK]D-Fender | dansmith : As far as phones go |
| 19:09.39 | Qwell | well, even a barbietones would be a step up from that :P |
| 19:09.40 | [TK]D-Fender | dansmith : Ok, well I'd suggest an IP 501 for you. $170 USD. |
| 19:09.41 | Qwell | barely |
| 19:09.52 | Qwell | [TK]D-Fender: no more 301 recommendations? |
| 19:09.55 | [TK]D-Fender | dansmith : A purchase you won't regret downt he road. |
| 19:10.07 | dansmith | well, I'll order the FXS for now and use analog phones for a bit |
| 19:10.28 | dansmith | I'd like to get a nice IP phone for the office, so I'll keep the IP501 in my back pocket for down the road |
| 19:10.28 | [TK]D-Fender | Qwell : 301 is good for general business, but if we're talking a god "master" phone for primary use byt he owner, then I'd say spend a bit more |
| 19:10.47 | [TK]D-Fender | Qwell : I always judge based on the user, and the deployment |
| 19:10.53 | Qwell | sure |
| 19:11.05 | Qwell | having a wife trumps all that though :) |
| 19:11.06 | dansmith | yea, the 301 is only $134, so not much less than $170 |
| 19:11.21 | [TK]D-Fender | Qwell : Wife > survivability of purchase :) |
| 19:11.33 | [TK]D-Fender | dansmith : You can get a 301 for $115 |
| 19:11.50 | dansmith | ok, cool |
| 19:12.30 | [TK]D-Fender | http://www.telephonydepot.com/Polycom_s/25.htm |
| 19:12.32 | dansmith | so, I think I'll get the PAP2 for now, so I can have two analog extensions and continue using my X100P for now |
| 19:12.44 | dansmith | awesome.. thanks a lot for the recommendations |
| 19:12.53 | [TK]D-Fender | But I'd get the 501 if I were you. Much bigger screen, 3 line keys, MicroBrowser, Speakerphone, etc... |
| 19:12.55 | dansmith | talking to live people on IRC for recommendations is always better than google :) |
| 19:13.06 | Qwell | man, I can't believe the 650 is only $279 |
| 19:13.25 | [TK]D-Fender | dansmith : though again, I'd suggest the SPA-3102 over the PAP-2 based on what you mentioned. |
| 19:13.42 | Qwell | [TK]D-Fender: have you used a 650 yet? |
| 19:13.46 | [TK]D-Fender | Qwell : Yeah, i just noticed the drop |
| 19:13.56 | [TK]D-Fender | Qwell : Nope... still not worth the difference :) |
| 19:14.02 | dansmith | [TK]D-Fender: wait, why? I thought you said that if I'm happy with the X100, that I wouldn't necessarily like the SPA3102 any better? |
| 19:14.03 | Qwell | it totally is |
| 19:14.11 | [TK]D-Fender | Actually.... 30$.. hmm |
| 19:14.12 | [TK]D-Fender | :) |
| 19:14.14 | Qwell | especially if you were gonna buy a 601 |
| 19:14.21 | [TK]D-Fender | with that drop in mind, uhhh yeah :) |
| 19:14.42 | Qwell | it's a freaking amazing phone :P |
| 19:15.05 | [TK]D-Fender | dansmith : And I said if you only needed 1 FXS reall, you'd be better to give the SPA-3102 a shot, and you can use the FXO ro not... your choice. Also good for REMOTE deployments. |
| 19:15.07 | Qwell | I swear, you can have it on speaker, be across the room (a large room), and talk to somebody else, and it will be perfectly clear on the other end |
| 19:15.25 | [TK]D-Fender | Qwell : that can be said of jsut about every Polycom Speakerphone :0 |
| 19:15.48 | Qwell | this is true, but the speaker/mic are even higher quality, because it's HD |
| 19:16.00 | Qwell | even ulaw sounds better |
| 19:16.12 | dansmith | [TK]D-Fender: oh, by remote deployments, you mean putting the device somewhere near the phone tap for the house, instead of right near the computer? |
| 19:16.20 | dansmith | everything comes into my server room anyway, so that's not a problem |
| 19:16.29 | Qwell | I'm waiting for a soundpoint 4050 :D |
| 19:16.35 | Qwell | erm, soundstation |
| 19:17.17 | *** join/#asterisk drfreeze (n=Jim@www.freeze.org) |
| 19:17.50 | drfreeze | For a VOiP only connection, is ingress mgt of QoS needed? |
| 19:17.55 | [TK]D-Fender | dansmith : that, and lets say you have a remote office, you can plug that in-line to hook to another PBX, or onto a phone line in another city/country, etc . |
| 19:18.30 | dansmith | yup, good point |
| 19:18.47 | [TK]D-Fender | Qwell : I still suggest the SoundStation 2W + ATA for wireless conference goodness :) |
| 19:18.58 | Qwell | 2W? |
| 19:19.16 | Qwell | wireless analog conf phone? |
| 19:19.19 | PakiPenguin | http://www.youtube.com/watch?v=M62s18UAo4I :) |
| 19:19.20 | [TK]D-Fender | Qwell : yup |
| 19:19.28 | Qwell | kinda silly, if you ask me |
| 19:19.41 | Qwell | replacing batteries, and really, how often do conf rooms move? |
| 19:20.05 | dansmith | what's the difference between an SPA-3000 and SPA-3102? Just a newer model? |
| 19:20.58 | [TK]D-Fender | Qwell : its rechargeable. |
| 19:21.04 | Qwell | still |
| 19:21.06 | [TK]D-Fender | Qwell : on its own NiMH battery |
| 19:21.21 | [TK]D-Fender | Qwell : And has a rather enormous battery life |
| 19:21.28 | Qwell | just seems kinda silly to have a need for a wireless/cordless conf phone |
| 19:21.38 | [TK]D-Fender | Qwell : depends if you want to run meeting in smaller offices |
| 19:21.38 | Qwell | I'm sure there are some places that might, but...meh |
| 19:21.55 | [TK]D-Fender | Qwell : Sure if you have a super-fixed conference room, whatever.... |
| 19:22.12 | [TK]D-Fender | Qwell : I also got mine before I sold my company on * :) |
| 19:22.28 | [TK]D-Fender | Qwell : Instead of getting the Norstar specific one :) |
| 19:22.43 | *** part/#asterisk karmatronic (n=karmatro@84.77.170.211) |
| 19:23.01 | Qwell | http://polycom.com/investor_relations/1,1434,pw-180-17262,00.html |
| 19:23.04 | Qwell | excessive! |
| 19:23.51 | *** join/#asterisk drako (n=ljd@unaffiliated/luisjose) |
| 19:24.47 | *** join/#asterisk aptura (n=jondoe@S010600a0c93f6f7e.vs.shawcable.net) |
| 19:25.26 | angryuser | I have installed latest asterisk from svn, and still no use ChanIsAvail when is used to check if Sip/xxx exist(even if it does no exist at all) still try to call by it, notmally it should jump to n+101 with j option, ;( crap |
| 19:25.53 | [TK]D-Fender | angryuser : Pastebin your code & the attempt |
| 19:26.30 | angryuser | [TK]D-Fender: 2 mins |
| 19:27.56 | *** join/#asterisk PupenoR (n=pupeno@200.123.183.91) |
| 19:28.38 | *** join/#asterisk obnauticus (i=admin@c-24-21-116-29.hsd1.mn.comcast.net) |
| 19:29.07 | Qwell | [TK]D-Fender: what are all those keys on the left side of a polycom 601/601? |
| 19:30.35 | [TK]D-Fender | Ummmm.. line keys? |
| 19:30.59 | [TK]D-Fender | or do you men the big ones below them? |
| 19:31.07 | *** join/#asterisk Lurchtoke (i=professi@adsl-75-37-75-140.dsl.frs2ca.sbcglobal.net) |
| 19:31.08 | Qwell | the bigger black keys |
| 19:31.10 | angryuser | ~pb |
| 19:31.25 | jbot | somebody said pb was a place to paste your stuff without flooding the channel - try http://pastebin.ca, or http://channels.debian.net/paste |
| 19:31.26 | Lurchtoke | hello everyone |
| 19:31.26 | Lurchtoke | :) |
| 19:31.40 | Lurchtoke | ~book |
| 19:31.56 | jbot | extra, extra, read all about it, book is a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 19:31.58 | [TK]D-Fender | 6 small staggered line key up top, then a few larger fixed function keys for Transfer, Conference, Hold, Services (MicroBrowser), Directories, etc. |
| 19:32.07 | Qwell | ahh |
| 19:32.26 | angryuser | pastebin.ca is dead? |
| 19:32.47 | Lurchtoke | hey fender....where can i find a good primer on how to put my spa-2000 adapters behind linksys vpn routers? |
| 19:32.49 | [TK]D-Fender | Qwell : The fixed function stuff is a waste as the soft-keys are 100% usable.... |
| 19:33.24 | angryuser | http://channels.debian.net/paste/5334 [TK]D-Fender: here we go |
| 19:33.27 | [TK]D-Fender | Lurchtoke : Nothing special to do.... |
| 19:33.42 | Lurchtoke | I set them to dhcp and tried to get them to see my server but they wouldnt grab a 192....ip |
| 19:33.44 | [TK]D-Fender | angryuser : You're calling it wrong. PERIOD |
| 19:33.52 | Lurchtoke | in my lan |
| 19:33.53 | [TK]D-Fender | angryuser : "show application chanisavail" |
| 19:34.00 | *** join/#asterisk NoCarrier (n=NoCarrie@unaffiliated/badpacket) |
| 19:34.43 | aptura | btw when is digium going to add Octasic echo cancellation to there cards? |
| 19:35.00 | [TK]D-Fender | aptura : wake up to 1 year ago... |
| 19:35.05 | aptura | :) |
| 19:35.36 | aptura | has it pretty much eliminated the eco issue? |
| 19:35.50 | [TK]D-Fender | aptura : For that you'd have to ask someone who actually USES one :) |
| 19:36.34 | aptura | Or just order one. |
| 19:36.35 | *** join/#asterisk Dovid (n=Dovid@l192-117-114-1.broadband.actcom.net.il) |
| 19:36.46 | [TK]D-Fender | aptura : At which point you can ask yourself :) |
| 19:36.53 | Qwell | and then we can direct others to you |
| 19:36.57 | Qwell | win-win |
| 19:37.09 | aptura | sure. |
| 19:41.00 | *** join/#asterisk s1gny|wrk (n=s1gny@p54915CEE.dip.t-dialin.net) |
| 19:41.08 | *** part/#asterisk s1gny|wrk (n=s1gny@p54915CEE.dip.t-dialin.net) |
| 19:41.31 | *** join/#asterisk PupenoR (n=pupeno@200.123.183.91) |
| 19:41.40 | Lurchtoke | ok..stupid question...how do you get a spa-2000 to grab an ip on a dhcp network? and shouldnt I program the default gateway in the spa to (192.168.1.100)? |
| 19:41.58 | denon | set it to dhcp |
| 19:42.00 | Lurchtoke | I set it in the config menu on the spa |
| 19:42.05 | denon | it'll get the default gateway from dhcp as well |
| 19:42.17 | denon | you shouldn't hard-set the gateway |
| 19:42.25 | Lurchtoke | yes |
| 19:42.53 | Lurchtoke | when you set it to dhcp it give you the error msg....so it auto grabs the ip...gateway...but what dns should i set? |
| 19:43.09 | denon | it gets dns from dhcp too |
| 19:43.09 | Qwell | it should also get dns from dhcp, if your dhcpd is configured right |
| 19:43.18 | denon | heh, there is that |
| 19:43.26 | [TK]D-Fender | Lurchtoke : If its on the other side of a VPN, it should get an address on the OTHER side |
| 19:43.39 | denon | though most consumer routers (he said 192.168.1.x - read, linksys) do it properly by default |
| 19:43.48 | Qwell | denon: indeed |
| 19:44.02 | Qwell | denon: have we ever actually met, btw? I can't place you :p |
| 19:44.16 | denon | not in person I don't believe |
| 19:44.25 | drfreeze | Anyone here have QoS experience? |
| 19:44.33 | Qwell | denon: hell, I don't even know who you are on the lists :P |
| 19:44.36 | denon | drfreeze: there's a pretty vague question |
| 19:44.51 | denon | Qwell: Im not on the lists much lately, mostly irc |
| 19:45.06 | denon | surely you remember me from the past several years on irc |
| 19:45.08 | drfreeze | denon: Just looking for info if I need to do traffic shaping on inbound traffic, or just outbound |
| 19:45.15 | Qwell | well, yeah...but that's the only place ;) |
| 19:45.32 | Qwell | many of the other people (and all of the ops), I've actually met |
| 19:45.35 | denon | drfreeze: depends on your traffic needs, but yes, inbound and outbound is preferrable |
| 19:45.50 | denon | Qwell: well, send me some plane money and I'll come shake your hands |
| 19:45.51 | drfreeze | denon: k |
| 19:45.55 | Qwell | actually, I take that back - I've never met anthm |
| 19:46.00 | *** join/#asterisk drako (n=ljd@unaffiliated/luisjose) |
| 19:46.10 | denon | hehe |
| 19:46.18 | drfreeze | Anyone have a QoS script for openwrt they wouldn't mind sharing |
| 19:46.37 | denon | drfreeze: you know, if your PBX is dedicated to being a PBX, you could just prioritize everything to and from it's IP |
| 19:46.52 | drfreeze | denon: that was my plan |
| 19:46.54 | denon | you could maybe set svn or ftp at a lower priority, so your updates dont hose voice |
| 19:46.57 | denon | but other than that .. yeah |
| 19:47.18 | drfreeze | denon: but I've never done this before. Just following some tutorials online |
| 19:47.35 | denon | drfreeze: nbd has some good QoS stuff for openwrt already |
| 19:47.35 | drfreeze | Thought a real config file might be helpful |
| 19:47.44 | drfreeze | denon: nbd? |
| 19:47.47 | denon | I dont remember where it's buried, but it's discussed on the wiki |
| 19:48.02 | denon | on the openwrt wiki |
| 19:48.15 | drfreeze | denon: yeah, that's what I am following |
| 19:48.50 | drfreeze | denon: seems you are saying this isn't going to be that difficult |
| 19:49.03 | *** part/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 19:49.03 | *** join/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 19:49.03 | drfreeze | ...or as difficult as I have imagined. :) |
| 19:49.12 | denon | drfreeze: difficulty is relative |
| 19:49.18 | denon | shouldnt be too bad |
| 19:49.18 | drfreeze | :) |
| 19:49.26 | drfreeze | denon: thanks |
| 19:49.30 | denon | if you've got openwrt installed, you probably have some clue .. |
| 19:50.10 | drfreeze | denon: yup, got all the openwrt goodies, iptraf, mtr. Now adding QoS |
| 19:50.18 | PakiPenguin | <PROTECTED> |
| 19:50.33 | denon | I much prefer openwrt over dd-wrt |
| 19:50.36 | drfreeze | PakiPenguin: I hear it is much less stable than openwrt |
| 19:51.11 | drfreeze | PakiPenguin: the word on the net is that dd-wrt has spent more time ont he web front end and less on the backend stability |
| 19:51.30 | drfreeze | just the opposite for openwrt |
| 19:51.39 | *** join/#asterisk aerys (n=aerys@85.137.121.99) |
| 19:51.57 | drfreeze | the openwrt web iface is pretty thin |
| 19:54.25 | PakiPenguin | yeah , but it works fine for me ( ~ 15 wireless users/some servers ) |
| 19:54.36 | PakiPenguin | never tried openwrt though |
| 19:55.54 | drfreeze | PakiPenguin: I don't know from personal experience. Just hearsay |
| 19:58.09 | aptura | very interesting. |
| 19:58.39 | aptura | I guess this is a way to relieve the cpu of the transcoding load. http://www.digium.com/en/products/hardware/tc400b.php |
| 20:03.07 | *** join/#asterisk drako (n=ljd@unaffiliated/luisjose) |
| 20:03.42 | niekie | Hmm, does anyone have Gizmo Project SIP here? |
| 20:03.48 | drako | tzafrir_laptop, any idea why ARI does not show the voicemail? it let me log in and show the monitor info but does not show the voicemail |
| 20:03.49 | [TK]D-Fender | angryuser : So, have you fixed it yet? |
| 20:04.14 | angryuser | [TK]D-Fender: i have inserted it in gotoif() |
| 20:04.23 | tzafrir_laptop | niekie, their client, or their ip service (sipphone.com)? |
| 20:04.35 | [TK]D-Fender | angryuser : You don't need any GotoIf.... |
| 20:04.36 | niekie | tzafrir_laptop, doesn't matter much. |
| 20:04.54 | [TK]D-Fender | angryuser : And you don't need AVAILCHAn |
| 20:05.02 | niekie | I set GizmoProject up on my Asterisk system, though someone who had it told me he couldn't reach it. |
| 20:05.03 | tzafrir_laptop | drako, do you use apache2? |
| 20:05.09 | niekie | And, to be honest, I think it's him. |
| 20:05.48 | niekie | tzafrir_laptop, if you have it, would you be able to make a call to my Gizmo # if possible to diagnose it? |
| 20:06.00 | tzafrir_laptop | niekie, you set up a local gizmo client to register with Asterisk? |
| 20:06.07 | niekie | tzafrir_laptop, nope. |
| 20:06.14 | niekie | I just use an IAX client locally. |
| 20:06.37 | tzafrir_laptop | drako? what httpd do you use? |
| 20:06.58 | drako | tzafrir, apache2 |
| 20:07.03 | angryuser | [TK]D-Fender: well my goai is simple, i need to che if Peer1 is reachable if yes>>call, if no go to peer2, another condition is that i need the call-limit=X to be takek in consideration, and i dont really dont see why you told me that i am calling it wrong |
| 20:07.26 | tzafrir_laptop | drako, so take a look at the logs, under /var/log/apache2/ |
| 20:07.37 | tzafrir_laptop | Anything relevant in the error.log ? |
| 20:07.49 | [TK]D-Fender | angryuser : Well its not working, and you want to not call it if tis busy right? |
| 20:08.07 | *** join/#asterisk zotz (n=zotz@24.244.163.157) |
| 20:08.40 | drako | tzafrir_laptop, no. |
| 20:08.53 | niekie | Hmm... Using the SIPBroker PSTN number to call my GizmoProject # works. |
| 20:08.57 | angryuser | [TK]D-Fender: if it is not working dial out with isdn, but it is just details, wright now i am unable to use ChanIsAvail... |
| 20:09.02 | *** join/#asterisk blitzrage (n=blitzrag@asterisk/documenteur-extraordinaire/blitzrage) |
| 20:09.04 | niekie | Would that definitely mean something is wrong with his connection? |
| 20:09.05 | tzafrir_laptop | drako, the problem is to login? |
| 20:09.20 | tzafrir_laptop | drako, what exactly is the problem? |
| 20:09.20 | drako | tzafrir_laptop, no error besides it can't find a favicon |
| 20:09.44 | drako | tzafrir_laptop, the problem is that it doesnt show the voicemail i mean i can log and even see the call monitor |
| 20:10.00 | drako | but when i go to voicemail is empty when there are 2 msgs |
| 20:10.10 | angryuser | [TK]D-Fender: i got cal-limit set to 2 on 3 peers so =6 lines max |
| 20:10.19 | tzafrir_laptop | where is that mailbox defined? |
| 20:11.15 | angryuser | [TK]D-Fender: i need to choose automacly free lines, along with 'reachable nor rechable' check |
| 20:11.16 | drako | voicemail.conf |
| 20:11.18 | [TK]D-Fender | angryuser : I don't believe ChanIsAvail respects and call-limits in a peer... |
| 20:11.45 | [TK]D-Fender | angryuser : For "in use at all" sure... might work for unreachable... |
| 20:13.05 | [TK]D-Fender | angryuser : You might have to do some sort of extreme AGI for this |
| 20:13.07 | *** join/#asterisk sharp (n=sharp@c-68-46-30-7.hsd1.pa.comcast.net) |
| 20:13.24 | angryuser | [TK]D-Fender: whatever i will set a $var or something, but wright no i got $availstatus =0 each time, and even if i do Chanisavail on inexistant peer, it gives me $availchan with the name of that peer! |
| 20:14.36 | [TK]D-Fender | angryuser : can you pastebin yoursip.conf as well... |
| 20:14.56 | *** join/#asterisk topping (n=topping@204.152.96.238) |
| 20:16.10 | tzafrir_laptop | drako, are you sure you got the mailbox name correctly? in the right context? |
| 20:17.23 | angryuser | [TK]D-Fender: i did test just now, set Chanisavail(Sip/Totalcrap|j) and i got ${AVAILSTATUS}=Totalcrap+session |
| 20:17.48 | drako | tzafrir_laptop, what do you mean? |
| 20:18.03 | angryuser | [TK]D-Fender: ok pasting |
| 20:18.06 | drako | tzafrir_laptop, i can see the folder with the voicemail |
| 20:18.11 | drako | i can call and leave a message |
| 20:22.17 | angryuser | http://channels.debian.net/paste/5335 [TK]D-Fender:got it all |
| 20:24.09 | *** join/#asterisk RoyK (n=roy@217-175-222.100710.adsl.tele2.no) |
| 20:24.45 | dansmith | so, I just had a longer phone call using my X100P to call out to my cell |
| 20:25.06 | dansmith | it was perfect quality in the beginning, but started to degrade in volume and clarity, and then ended up completely inaudible |
| 20:25.18 | dansmith | any clues on where to look for the issue? |
| 20:26.19 | *** join/#asterisk jpe-nyc (n=jpe-nyc@p77-37.acedsl.com) |
| 20:27.45 | *** join/#asterisk yassine (n=yassine@dsl.voicint.com) |
| 20:30.50 | angryuser | afk 5 min |
| 20:32.05 | aptura | dansmith, hard to say. |
| 20:32.22 | dansmith | actually, |
| 20:32.24 | drako | tzafrir_laptop, its on the right context. |
| 20:32.26 | dansmith | I think it might be ekiga |
| 20:32.29 | drako | tzafrir_laptop, still nothing |
| 20:32.30 | dansmith | er, I'm sure it is |
| 20:32.37 | tzafrir_laptop | trying to configure it here |
| 20:32.43 | dansmith | because I did an ekiga call between two PCs and it did the same thing |
| 20:33.45 | *** join/#asterisk J4k3 (i=jsuter@dhcp-12-197-128-58.intrastar.net) |
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| 20:38.52 | *** join/#asterisk zmef420 (n=zmef420@metarb3-pool3-130.mtco.com) |
| 20:41.37 | drako | tzafrir_laptop, any luck? |
| 20:41.47 | tzafrir_laptop | I'm trying to login... |
| 20:42.14 | tzafrir_laptop | <PROTECTED> |
| 20:43.20 | drako | tzafrir_laptop, im not using that file, i using the main.conf.php on the ari's folder |
| 20:44.36 | *** join/#asterisk jpe-nyc (n=jpe-nyc@p77-37.acedsl.com) |
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| 20:54.25 | *** join/#asterisk Givemelove (n=bozoo@cre94-1-81-57-163-2.fbx.proxad.net) |
| 20:54.29 | Givemelove | Hi guys |
| 20:54.47 | Givemelove | I really need help onto an 1.2 -> 1.4 upgrade |
| 20:54.53 | Givemelove | I have an issue with zaptel |
| 20:54.58 | Givemelove | I have the following error: |
| 20:55.07 | Givemelove | <PROTECTED> |
| 20:55.20 | Givemelove | anybody encountered that already? |
| 20:57.16 | *** join/#asterisk PupenoR (n=pupeno@200.123.183.91) |
| 20:59.21 | Givemelove | any hint? |
| 21:01.29 | *** join/#asterisk Gershwin (n=Bonzi@c-76-16-183-209.hsd1.il.comcast.net) |
| 21:03.05 | sweeper | w00t |
| 21:03.07 | sweeper | RAGI rawks |
| 21:05.29 | niekie | RAGI? |
| 21:05.40 | drako | Ruby AGI |
| 21:09.02 | *** join/#asterisk santiago (n=santiago@debian/developer/santiago) |
| 21:09.37 | niekie | Ah. |
| 21:15.59 | *** join/#asterisk drako (n=ljd@unaffiliated/luisjose) |
| 21:16.53 | tzafrir_laptop | Givemelove, it means that /etc/zaptel.conf and the current state in your system (look at /proc/zaptel/* ) don't agree on channel 25 |
| 21:20.23 | *** join/#asterisk antlers (n=antlers@ip70-189-186-184.lv.lv.cox.net) |
| 21:23.19 | tzafrir_laptop | drako, ari seems to have a parsing error |
| 21:23.28 | drako | tzafrir, oh really? |
| 21:23.33 | tzafrir_laptop | It does not know that a '=' is a valid separator |
| 21:23.35 | drako | tzafrir_laptop, any idea ? |
| 21:23.43 | drako | any fix? |
| 21:24.05 | tzafrir_laptop | I changed ' 2004=1234,phone1,1234,' to '2004=>1234,phone1,1234,' |
| 21:24.18 | tzafrir_laptop | and suddenly my mailbox was recognised |
| 21:24.41 | Givemelove | guys need help with that damn zaptel |
| 21:25.13 | Givemelove | tzafrir_laptop -> my cat /proc/zaptel shows all the channels |
| 21:25.28 | Givemelove | the inappropriate ioctl for device is for channel 1 |
| 21:25.31 | Givemelove | not 25 |
| 21:25.36 | Givemelove | 25 is the error code |
| 21:26.48 | drako | tzafrir_laptop, where is taht? |
| 21:27.19 | drako | this is my entry in voicemail.conf |
| 21:27.21 | drako | 10=>1234,Luis Jose,luisjoseve@gmail.com |
| 21:27.22 | *** join/#asterisk Splat (n=splat@220-253-136-53.TAS.netspace.net.au) |
| 21:27.27 | tzafrir_laptop | drako, no, that is an unrelated problem. But only now I was able to replicate your problem |
| 21:27.41 | drako | tzafrir_laptop, oh |
| 21:28.00 | drako | tzafrir_laptop, oh well, maybe a manual installation? im sure ARI works for other ppl. |
| 21:28.13 | mafkees | ARI ? |
| 21:28.31 | Givemelove | Asterisk Recording Interface |
| 21:28.38 | Givemelove | cf voip-info.org |
| 21:29.44 | Corydon76-home | We have a recording interface? |
| 21:30.18 | *** join/#asterisk atlantia (n=scott@64.20.159.149) |
| 21:31.40 | drako | ill try with te latest and see |
| 21:31.41 | atlantia | hrrmph.. well.. installed asterisknow, using x100p card for test setup, detected it fine, worked great. Reinstalled with the same .iso and now i get "No analog device" and |
| 21:31.42 | atlantia | 00:08.0 Communication controller: Motorola: Unknown device 5608 |
| 21:31.42 | *** join/#asterisk haaseg (n=haaseg@74.92.154.217) |
| 21:32.03 | atlantia | wondering what could be diff about this install than last |
| 21:38.59 | haaseg | are very basic questions permitted? I'm in the explarotory "what if I installed Asterisk" phase |
| 21:39.17 | haaseg | or "exploratory" even |
| 21:42.01 | tzafrir_laptop | drako, another thing: ari seem to simply ignore the concept of contetxts |
| 21:42.20 | tzafrir_laptop | and destar puts messages by default in a different context |
| 21:42.20 | florz | haaseg: Given that they are a little above the level of "may I ask a question?" =:-) |
| 21:42.37 | haaseg | hah! |
| 21:43.06 | *** join/#asterisk lowlevel (n=Stuart@CPE000e0c057fad-CM000f9f7d6742.cpe.net.cable.rogers.com) |
| 21:43.12 | tzafrir_laptop | try something of the sort of: ln -s default /var/spool/asterisk/voicemail/pbx1 |
| 21:43.30 | sweeper | haaseg: ask first, find out if you'll get flamed later |
| 21:43.34 | tzafrir_laptop | now it shows me the mailbox |
| 21:43.50 | haaseg | Okay. I have voip, and I was thinking I would have to buy a modem card or something to connect to my gizmo, and then I thought - hey, can't I just connect directly with asterisk? |
| 21:44.05 | atlantia | anyone have an idea of how i can get my linux box to recognize this card? |
| 21:44.39 | tzafrir_laptop | BTW: zaptel 1.2.13 and asterisk 1.2.15 broke the ukcid patch again. Commiting fixes to Debian soon |
| 21:44.43 | *** join/#asterisk ttuttle (n=tom@gentoo/contributor/ttuttle) |
| 21:45.35 | ttuttle | What's a good, cheap, pay-per-minute VoIP provider that offers SIP and IAX2, incoming numbers in most area codes, and allows you to pre-load funds into your account rather than being billed monthly? (I'm using Vitelity but the quality's pretty bad.) |
| 21:47.07 | atlantia | meh these x100ps suck... but why the heck would the same install succeed the first time and fail this time? lspci shows the card as not recognized, but dangit, it worked two days ago |
| 21:47.52 | ttuttle | Anyone? |
| 21:48.22 | florz | atlantia: same version of lspci? |
| 21:49.03 | *** part/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 21:49.03 | *** join/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 21:49.15 | atlantia | florz, same exact asterisknow install disk |
| 21:49.35 | atlantia | florz, trying diff pci slots, thats the only thing that could have changed |
| 21:49.49 | atlantia | damn scientific method doesn't apply to el magi computers |
| 21:49.52 | atlantia | magic* |
| 21:49.55 | florz | atlantia: then I'd try to take the card out, boot once without it, then plug it back in |
| 21:50.09 | ttuttle | I'm sure some of you use VoIP. This is #Asterisk, right? /me double checks. |
| 21:50.11 | Givemelove | guys, any hint on the following issue with zaptel? ZT_CHANCONFIG failed on channel 1: Inappropriate ioctl for device (25) |
| 21:50.40 | *** join/#asterisk phatmonkey (i=nobody@81.2.121.150) |
| 21:50.47 | atlantia | florz ! |
| 21:50.51 | atlantia | florz, thats the diff! |
| 21:51.01 | atlantia | i orignally installed before i had the card |
| 21:51.03 | atlantia | :D |
| 21:51.13 | phatmonkey | i want to have a dial timeout on an outgoing POTS zap channel, but obviously the zap channel answers before ringing. any ideas how to get around this? |
| 21:51.56 | phatmonkey | i think callprogress might be what i want - will this work in the UK? |
| 21:52.27 | florz | atlantia: Doesn't quite sound like that should be a problem =:-) |
| 21:53.23 | ttuttle | Does anyone here use Vitelity? |
| 21:53.30 | *** join/#asterisk matiasds19 (n=matiasdo@host203.201-252-49.telecom.net.ar) |
| 21:54.26 | sweeper | haaseg: gizmo? huh? |
| 21:54.37 | matiasds19 | hey guys still with the same problem, someone who can help, im going crazy with this thing |
| 21:54.50 | sweeper | haaseg: also, please be more explicit with "I have voip" |
| 21:54.59 | haaseg | Okay... let me try again |
| 21:55.44 | *** join/#asterisk Corydon76-home (i=grey@pdpc/supporter/sustaining/Corydon76-home) |
| 21:55.44 | *** mode/#asterisk [+o Corydon76-home] by ChanServ |
| 21:56.10 | matiasds19 | guys pleaseeee...someone who can check my config and tell me if im wrong |
| 21:56.20 | atlantia | florz, you know, you'd think... i figure (kudzoo?) is having some issue.. but yeah, last time, fresh install, ordered garbage x100p (authentic! yeah right) card from ebay, installed, worked, tested. So something had to change, or the card crapped the bed |
| 21:56.37 | haaseg | I have Sunrocket VOIP. They gave me this linksys spa2102 (they call a "Gizmo"). I thought I would have a modem card in my server and have Asterisk connect to that - but then it wouldn't make much sense. I was wondering - is it common/possible to have asterisk connect directly to an external VoIP provider |
| 21:56.38 | atlantia | brb |
| 21:57.39 | haaseg | And if I did that, could I config the spa to connect to asterisk instead of directly to sunrocket |
| 21:57.59 | sweeper | haaseg: yes, it's common, and very possible |
| 21:58.09 | haaseg | sweeper, thank you |
| 21:58.58 | drako | tzafrir_laptop, |
| 21:58.59 | drako | Feb 10 18:58:55 WARNING[3210]: db.c:67 dbinit: Unable to open Asterisk database |
| 21:59.18 | drako | tzafrir_laptop, i get that when i refresh the ARI webpage |
| 21:59.51 | tzafrir_laptop | I disabled some features (e.g: monitoring) to avoid the need of using a database |
| 22:00.08 | tzafrir_laptop | hmm... this is from asteris, however |
| 22:00.20 | tzafrir_laptop | ls -ld /var/spool/asterisk |
| 22:00.32 | tzafrir_laptop | ls -ld /var/lib/asterisk |
| 22:00.40 | tzafrir_laptop | (the second one) |
| 22:00.59 | haaseg | sweeper, this is done in sip.conf? |
| 22:01.18 | sweeper | haaseg: mostly, yes |
| 22:01.28 | drako | drwxr-xr-x 3 asterisk asterisk 4096 2007-02-10 16:42 /var/lib/asterisk |
| 22:01.42 | drako | tzafrir_laptop, i think is related with that msg i just pasted |
| 22:01.44 | drako | about the dbinit |
| 22:01.57 | sweeper | you'll have an entry for your gizmo, and one for your voip provider |
| 22:02.21 | sweeper | and you'll have to configure extensions.conf to make them talk to eachother |
| 22:02.40 | *** join/#asterisk PupenoR (n=pupeno@200.123.183.91) |
| 22:02.44 | haaseg | okay |
| 22:03.18 | haaseg | I'm going to have to start reading a lot |
| 22:03.21 | tzafrir_laptop | drako, do you get the same error from 'database show' in the asterisk CLI? |
| 22:04.26 | drako | tzafrir_laptop, heh yes |
| 22:05.34 | tzafrir_laptop | ls -l /var/lib/asterisk/astdb |
| 22:05.59 | *** join/#asterisk anthm (n=anthm@CPE-72-131-113-50.wi.res.rr.com) |
| 22:05.59 | *** mode/#asterisk [+o anthm] by ChanServ |
| 22:07.09 | tzafrir_laptop | anyway, I'm off for now |
| 22:08.38 | drako | it was root owned |
| 22:08.41 | drako | changed to asterisk |
| 22:11.58 | *** join/#asterisk digix84 (n=digix@72-48-74-116.dyn.grandenetworks.net) |
| 22:12.57 | digix84 | hey all, i hate to keep coming here with just problems, but i could really use some help on this one... |
| 22:13.16 | *** join/#asterisk Corydon76-home (i=two@pdpc/supporter/sustaining/Corydon76-home) |
| 22:13.16 | *** mode/#asterisk [+o Corydon76-home] by ChanServ |
| 22:13.29 | digix84 | i have two servers linked with dundi, and the remote extensions are not accessible through the IVRs |
| 22:14.11 | digix84 | is there anything special i need to do in order to allow each server to see the remote extensions in the IVRs? |
| 22:17.26 | *** join/#asterisk Jared_Leto (n=Lostprop@80-89-104-241.DSL.ycn.com) |
| 22:18.01 | Givemelove | How to include chan_zap in the asterisk compilation? |
| 22:20.48 | atlantia | meh |
| 22:20.57 | Givemelove | ? |
| 22:21.09 | atlantia | reinstalled, tried everything, still getting unknown device for this card |
| 22:21.26 | atlantia | sorry Givemelove meh-ing in general |
| 22:22.05 | Givemelove | ok |
| 22:22.29 | atlantia | ha |
| 22:22.31 | atlantia | now it works |
| 22:22.34 | atlantia | damn magic |
| 22:23.52 | *** join/#asterisk Ebola (n=Ebola@host86-137-4-80.range86-137.btcentralplus.com) |
| 22:28.31 | *** join/#asterisk nny (n=scott@64.20.159.149) |
| 22:28.41 | nny | ok so heres what i am trying to accomplish |
| 22:28.56 | nny | basically welcome to blah foo, press 1 for residential, 2 for commercial |
| 22:29.06 | nny | 1 = forward to cell phone X |
| 22:29.16 | nny | 2 = forward to cell phone X, or Y |
| 22:29.20 | nny | er Y or Z |
| 22:29.21 | [TK]D-Fender | Givemelove : Compile and install Zaptel first |
| 22:29.27 | nny | thats it |
| 22:29.40 | nny | anyone have a sugesstion to the quickes path to that scenario? |
| 22:29.54 | [TK]D-Fender | nny : All basic stuff.... |
| 22:29.55 | [TK]D-Fender | ~book |
| 22:30.10 | jbot | book is, like, a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 22:30.10 | [TK]D-Fender | ~wikis |
| 22:30.15 | jbot | [wikis] http://www.voip-info.org |
| 22:30.18 | [TK]D-Fender | on the WIKi you can lookup IVR tips and will link a good page to start |
| 22:30.52 | nny | ok, once i have that, our provider offers a forward service, which is [flash] number [hang-up]. is this a way to do the cell phone part i mentioned? |
| 22:31.15 | [TK]D-Fender | nny : not really. |
| 22:31.57 | nny | [TK]D-Fender, why? can you explain? with an analog phone, it seems fairly simple, it works, and when it is forwarded, the line is freed up |
| 22:32.17 | nny | we only need this sytem to route calls to our cell phones |
| 22:33.31 | *** join/#asterisk markit (n=konversa@host119-245-static.72-81-b.business.telecomitalia.it) |
| 22:33.36 | [TK]D-Fender | nny : there is no way to grab a line and send a flash to it after |
| 22:34.11 | [TK]D-Fender | nny : yes is SOUNDS simple, its just that * has no automation capacity for that. |
| 22:34.34 | [TK]D-Fender | nny : this IS possible with excessive programming and some small circuit building, but frankly just not worthit |
| 22:34.38 | markit | hi, I call an asterisk machine (ivoice.it), the IVR answers, I hang up. Then when I re-call it, I get on the CLI: "Call on SIP/ivoice-out-0821c668 left from hold" and I can't hear any voice anymore.... I'm clueless :( |
| 22:34.54 | nny | so asterisk can't send a flash to the line, pause for X, dial number, hang up? |
| 22:35.10 | kink0 | hello, anybody gets asterfax running ? I got a lot of problems compiling app_txfax.so |
| 22:35.23 | [TK]D-Fender | nny : Correct |
| 22:35.27 | *** join/#asterisk anothy_x (n=anthony@pool-71-163-212-152.washdc.east.verizon.net) |
| 22:37.39 | nny | thats strange |
| 22:40.25 | nny | [TK]D-Fender, i want to understand why, any advice? |
| 22:41.12 | [TK]D-Fender | nny : The dialplan executes 1 command at a time, and once you'd ahve * dial out, there is no way to send a flash. |
| 22:42.02 | [TK]D-Fender | nny : As for advice, either make a remotely triggerable flash device to plug in-line with your line and get programming, or come up with another way to get what you want. |
| 22:44.06 | *** join/#asterisk [Mr_X] (i=1000@88.118.88.110) |
| 22:49.03 | *** part/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 22:49.04 | *** join/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 22:51.40 | *** join/#asterisk dj-fu (n=deejay@202-89-157-131.dyn-dsl.ubs.qsi.net.nz) |
| 22:53.41 | digix84 | so any dundi experts in here that can help with my IVR problem? |
| 22:54.44 | *** join/#asterisk EmleyMoor (i=ejabberd@hallam.tinsleyviaduct.com) |
| 22:56.09 | EmleyMoor | I am trying to compile zaptel modules against a new kernel, all installed using the Debian package system. I have unpacked the source and made a minor change. I cannot get it to compile - in fact, m-a -k <headers dir> prep fails with "Bad kernel version specification" |
| 22:56.16 | EmleyMoor | How do I proceed? |
| 22:57.29 | [TK]D-Fender | EmleyMoor : Sounds like you don't have the right headers for your kernel installed |
| 22:57.53 | EmleyMoor | Yes, I agree it sounds like it. It is not however the case |
| 22:58.36 | drako | i want get rid of the res_odbc and res_mysql warnings. |
| 23:01.17 | EmleyMoor | I think I'm getting somewhere... |
| 23:04.52 | digix84 | i have two servers linked with dundi, everything works fine calling between servers and all that, but the remote extensions are not availalable from the IVRs; when dialing direct it just says that its not a valid extension |
| 23:05.22 | digix84 | is there anything special i need to do to get the servers to see the remote extensions for this? |
| 23:07.01 | digix84 | ive read all the documentation i can find on dundi and the ivr, but i just cant seem to find out how to get this working |
| 23:10.13 | *** join/#asterisk bkw__ (n=brian@adsl-70-143-63-77.dsl.tul2ok.sbcglobal.net) |
| 23:12.55 | *** join/#asterisk ttuttle (n=tom@gentoo/contributor/ttuttle) |
| 23:13.09 | ttuttle | Do I need to include ISDN4Linux to get support for regular non-ISDN voice modems? |
| 23:14.02 | ttuttle | I'm getting "No channel type registered for 'Modem'" |
| 23:15.13 | *** join/#asterisk Gamercjm (n=chris@pool-71-254-184-57.lsanca.fios.verizon.net) |
| 23:15.31 | ttuttle | Has chan_modem been removed from Asterisk? |
| 23:16.55 | ttuttle | Is there any way to use a standard V.92 modem as an Asterisk channel? |
| 23:17.35 | *** join/#asterisk dseeb_ (n=dcb@CPE-124-179-244-123.vic.bigpond.net.au) |
| 23:18.31 | ttuttle | "./CHANGES: 2. chan_modem_* and related modules are gone because the kernel support for those interfaces is old, buggy and unsupported" Dammit! How am I supposed to use my voice modem? |
| 23:18.52 | ttuttle | Kernel support? It's simple! Open /dev/modem and do stuff! |
| 23:19.07 | ttuttle | And why is there still a sample modem.conf if it doesn't work? |
| 23:19.48 | blitzrage | you can't just use any modem you want for an FXO port |
| 23:20.28 | ttuttle | blitzrage: But I *could* use an AOpen or <I forget>-compatible voice modem, until they removed it. |
| 23:20.32 | ttuttle | blitzrage: Couldn't I at least try? |
| 23:20.45 | ttuttle | blitzrage: It doesn't make sense to throw out code. |
| 23:20.47 | blitzrage | and chan_modem isn't what you use for it -- it's the wcfxo driver in zaptel |
| 23:21.03 | ttuttle | blitzrage: Isn't there a voice modem standard? |
| 23:21.08 | blitzrage | no |
| 23:21.27 | ttuttle | MODEM |
| 23:21.29 | ttuttle | <PROTECTED> |
| 23:21.29 | ttuttle | <PROTECTED> |
| 23:21.31 | ttuttle | whoops |
| 23:21.41 | ttuttle | This is what I meant to paste: http://www.voip-info.org/wiki/view/Asterisk+readme.channels |
| 23:21.59 | ttuttle | It appears there *used* to be a "Generic Voice Modem Channel Driver". So someone tried, at least. |
| 23:22.24 | ttuttle | Okay, how about winmodems? Is there any way to directly interface with a Conexant Winmodem, if it uses ALSA? |
| 23:22.37 | blitzrage | right, but that doesn't mean it is supported now |
| 23:22.53 | blitzrage | you have to use a modem with a certain Motorola chipset |
| 23:22.57 | *** join/#asterisk zotz (n=zotz@24.244.163.157) |
| 23:23.19 | blitzrage | I can't remember the exact chipset from memory |
| 23:23.22 | ttuttle | Why would support for it be removed? If it's a voice modem, it runs over a serial port, and the protocol won't change over time. So the code should work with the modems that it works with, forever. |
| 23:23.45 | ttuttle | How far back would I have to go to get support for it? |
| 23:24.09 | blitzrage | if it's not being supported, there is a reason for it. Give up. |
| 23:24.25 | ttuttle | No. |
| 23:24.37 | ttuttle | (With all due respect.) |
| 23:24.43 | blitzrage | ok that's fine. have fun |
| 23:24.48 | ttuttle | Thanks. |
| 23:25.09 | ttuttle | Seriously, though, if someone were to write a new driver, for the current version of *... is there a voice modem protocol that would work with many modems? |
| 23:26.26 | blitzrage | digix84: sounds like a dialplan issue. All DUNDi is for is to lookup information on a remote server. |
| 23:26.26 | EmleyMoor | Successful upgrade to etch |
| 23:33.04 | *** join/#asterisk orkid_ (n=orkid@bas1-barrie18-1242375743.dsl.bell.ca) |
| 23:38.37 | ttuttle | s/instlal/install/; |
| 23:40.52 | digix84 | blitzrage, dialplan where? on the inbound route? |
| 23:41.05 | blitzrage | digix84: yes |
| 23:41.25 | blitzrage | DUNDi only transmits information... it's all about the dialplan to handle the incoming and outgoing connections |
| 23:41.45 | blitzrage | pastebin some examples of your dialplan and what you're doing (including errors, and such) |
| 23:41.54 | EmleyMoor | Is there a way (I will need this soon) to cheerfully ignore (though perhaps grab the caller ID) a ringing Zap line? |
| 23:42.07 | [TK]D-Fender | ttuttle : Lets put it this way : No one cares to write a channel driver to support every crappy win-modem ever produced, nor is it worth the effort for you to be able to save 15$ because you're even CHEAPER than that and desperate to have whatever gear you have already become "magically" compatible. |
| 23:43.02 | *** join/#asterisk fnordus (n=dnall@24.85.128.203) |
| 23:44.15 | Qwell | dseeb_: heard any reports of moto V series el-cheapo phones working? |
| 23:44.18 | ttuttle | [TK]D-Fender: No. But you'd probably get thousands more users if everyone with one of a few crappy winmodems (there aren't that many truly different ones) could use Asterisk without buying hardware. |
| 23:44.24 | digix84 | blitzrage, i see no spot to enter a dialplan for the inbound in freepbx, i just accept all incoming and direct it to the IVR |
| 23:44.34 | Qwell | motorola says it supports the handsfree profile :D |
| 23:44.51 | ttuttle | [TK]D-Fender: Seriously. I can't help but think that Digium doesn't want crappy voice modems stealing business from their cards. |
| 23:45.04 | blitzrage | digix84: oh -- freepbx isn't supported in this channel, sorry. |
| 23:45.15 | Qwell | ttuttle: Do you have any idea how long it would take to write drivers for every winmodem out there? |
| 23:45.27 | ttuttle | Qwell: How many are there? |
| 23:45.34 | digix84 | you cant be serious... |
| 23:45.35 | Qwell | ttuttle: If you would like somebody to implement it, please feel free to hire somebody |
| 23:45.53 | blitzrage | digix84: see topic |
| 23:46.08 | ttuttle | Qwell: Off the top of my head (in terms of Linux support), there are Smartlink and Conexant. Those two would be a good start, and both use ALSA for audio already. |
| 23:46.18 | Qwell | that's *TWO* |
| 23:46.38 | [TK]D-Fender | ttuttle : And how many of those specific winmodem users world-wide will actualyl be USING *? Divide that byt eh % that actualyl want to use it for analog termination (many have NO special hardware at all and are pure VoIP), and the multiply it by the time someone is going to have to waste building a driver. |
| 23:46.40 | ttuttle | Qwell: It wouldn't take long for ones that use ALSA, since there could be a generic ALSA-Modem implementation, and then little bits of code for the other stuff. |
| 23:46.56 | ttuttle | [TK]D-Fender: None, right now. |
| 23:47.11 | blitzrage | and the people using shitty, cheap modems are not a great demographic to try and support |
| 23:47.24 | blitzrage | it just doesn't make sense |
| 23:47.26 | ttuttle | [TK]D-Fender: I would have been using it years ago with my older laptop if it supported voice modems. |
| 23:47.33 | digix84 | fucking ridiculous |
| 23:47.35 | [TK]D-Fender | "But...but... I have a modem, why won't it work!??!" |
| 23:47.43 | ttuttle | blitzrage: Open source software isn't about whether it's a "good demographic to support". |
| 23:47.44 | blitzrage | digix84: that's the spirit! |
| 23:47.47 | Qwell | ttuttle: bugs.digium.com - please feel free to post a patch when you're done writing it |
| 23:48.01 | Qwell | but, it has to be generic enough to support every winmodem |
| 23:48.05 | sweeper | damn, I keep forgetting to patch that stupid awk bug |
| 23:48.39 | ttuttle | Qwell: Extensible enough to have support added. |
| 23:48.41 | [TK]D-Fender | ttuttle : And who's going to wastetime building drivers for hardware that demographics don't support? If its OSS, they sure as hell aren't being PAID for it. |
| 23:49.00 | sweeper | hmmm |
| 23:49.03 | *** part/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 23:49.03 | *** join/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 23:49.28 | blitzrage | ok... GO LEAFS GO! |
| 23:49.40 | sweeper | I dunno, sounds like someone wants you to buy a $40 ata or FXO card, instead of a $5 winmodem :D |
| 23:49.49 | ttuttle | sweeper: Yeah. |
| 23:49.57 | blitzrage | hardware sucks -- just use SIP |
| 23:49.59 | Qwell | It'll cost SUBSTANTIALLY more than $40 to write the support |
| 23:50.02 | ttuttle | blitzrage: I have SIP too. |
| 23:50.06 | ttuttle | Qwell: I don't care, I like coding. |
| 23:50.16 | sweeper | ttuttle: I suggest trying #openpbx |
| 23:50.18 | Qwell | Then like I said, please feel free to post a patch to bugs.digium.com |
| 23:50.28 | ttuttle | Qwell: I'll see what I can do. |
| 23:50.34 | [TK]D-Fender | ttuttle : You'd have been using * years ago if your LAPTOPS modem was supported by Zaptel? Lame excuse. If you were tryuely interested you'd have invested something into the hardware you'd use. Evidently this small bump in the road is enough to cause your progress to come to a crashing halt. |
| 23:51.08 | sweeper | [TK]D-Fender: if modems were supported, you'd have kids that were hacking * since they were 12 |
| 23:51.31 | sweeper | so much fun shit * can do, and almost EVERYONE has a modem |
| 23:51.31 | ttuttle | [TK]D-Fender: I didn't (and still don't) have any excuse to spend a bunch of money on hardware to play around with Asterisk. Supporting existing hardware is a good idea. |
| 23:51.32 | [TK]D-Fender | sweeper : Oh noes! H4X0rzS!?! |
| 23:51.35 | sweeper | [TK]D-Fender: .... |
| 23:51.42 | sweeper | I meant in a good way :/ |
| 23:51.55 | blitzrage | if lack of support for any old hardware you have laying around instead of spending the $40 (or less) on something that is supported is stopping you from using Asterisk.... then yah... well, good luck |
| 23:52.24 | Qwell | blitzrage: how come the Tandy that's been sitting in my basement for 20 years isn't supported by Linux? |
| 23:52.27 | test34 | sweeper, theres modems which are supported |
| 23:52.34 | sweeper | test34: I realize this |
| 23:52.34 | [TK]D-Fender | ttuttle : You're right. Which is why we don't have 100BT fiber in North America, like so much of southern Asia |
| 23:52.36 | Qwell | I *DEMAND* that open source developers spend time writing support for it |
| 23:52.40 | sweeper | I happen to use them |
| 23:52.47 | blitzrage | Qwell: I don't understand why my 33.6 USR ISA modem isn't supported either |
| 23:53.01 | *** join/#asterisk wubba (n=kmurrey@cable-76-215.sssnet.com) |
| 23:53.13 | *** part/#asterisk wubba (n=kmurrey@cable-76-215.sssnet.com) |
| 23:53.16 | ttuttle | Qwell: I'm not demanding that anyone support it, I'm just angry that support was removed when there was already a driver. I think it should have been marked deprecated, but maintained. |
| 23:53.22 | sweeper | somehow, I doubt winmodems are more highly variable than network cards |
| 23:53.26 | [TK]D-Fender | blitzrage : I have a smoke signal generator right here that screams "chan_zap.so" all over it! Why isn't it supported?!?! |
| 23:53.28 | blitzrage | ttuttle: who is going to maintain it? |
| 23:53.33 | *** join/#asterisk fnordus (n=dnall@24.85.128.203) |
| 23:53.42 | blitzrage | [TK]D-Fender: because asterisk developers are lazy obviously |
| 23:53.45 | Qwell | ttuttle: well, how's this for irony... |
| 23:53.58 | Qwell | ttuttle: in order for us to maintain it...we would have to go out and *BUY* hardware |
| 23:54.26 | blitzrage | crappy hardware at that |
| 23:54.26 | [TK]D-Fender | ttuttle : Nobody CARES to maintin it. We have all come to the conclusion that worthy equipment WILL be maintained, and older random stuff will fall to the wayside. Welcome to the world of Natural Selection. |
| 23:54.42 | Qwell | s/Natural Selection/Open Source/ |
| 23:54.47 | Qwell | brb, food run |
| 23:54.50 | Qwell | umm |
| 23:54.55 | Qwell | I don't even want to know |
| 23:55.04 | [TK]D-Fender | :O |
| 23:55.44 | [TK]D-Fender | Qwell : (Chan_smoke_signal.so joke) |
| 23:56.24 | ttuttle | Alright, I admit it. |
| 23:56.31 | ttuttle | It's not worth their trouble to support it. |
| 23:56.36 | denon | chan_carrier_pigeons.so |
| 23:56.57 | denon | though, latency is an issue |
| 23:57.22 | blitzrage | there are plenty of other pieces of supported hardware that do a better job than a random modem |
| 23:57.22 | [TK]D-Fender | denon : they would clash with my channel driver.... Bbirds + smoke = asphyxiation |
| 23:57.29 | blitzrage | it just isn't necessary to support old / random hardware |
| 23:57.44 | blitzrage | anyways... hockey game is about to start. Time to drink some beer and cheer on the Leafs |
| 23:57.51 | denon | [TK]D-Fender: nah, these are special carrier pigeons, they fly at 30k feet, and register with air traffic control |
| 23:57.52 | blitzrage | [TK]D-Fender: you coming down for Linuxworld in April? |
| 23:58.57 | [TK]D-Fender | denon : Dont forget their oxygen masks! |
| 23:59.02 | [TK]D-Fender | blitzrage : Where? |
| 23:59.17 | denon | [TK]D-Fender: really big lungs .. active hibernation |
| 23:59.54 | blitzrage | [TK]D-Fender: Toronto |
