| 00:03.33 | n0n4m3 | hehe |
| 00:03.48 | n0n4m3 | imagine what changing type=peer to type=friend does :D |
| 00:05.13 | *** join/#asterisk Aeudian (n=chatzill@c-69-250-24-154.hsd1.md.comcast.net) |
| 00:07.52 | n0n4m3 | darn :S |
| 00:12.09 | n0n4m3 | in case i want to 'forward' a call to an internal ip phone do i have to use Answer as the first command in extensions.conf or not? tried with Ringing instead but no luck :/ |
| 00:13.04 | n0n4m3 | http://rula.net/92 |
| 00:13.40 | n0n4m3 | but like i said.. i don't want to 'answer' but just 'ring on' if i make any sense :D |
| 00:14.24 | n0n4m3 | and in case noone answers after 30 sec, forward the call to voicemail and then end the call |
| 00:15.07 | dijungal | when i do a "sip show peers" what does the port column mean? |
| 00:15.26 | fujin | the port that they registered from |
| 00:15.39 | n0n4m3 | the 'port' |
| 00:15.44 | n0n4m3 | usually 5060 |
| 00:15.45 | dijungal | i have a phone on port 25452, i can make a call but I get no audio when the other end picks up |
| 00:15.52 | fujin | nat? |
| 00:15.57 | dijungal | yes |
| 00:16.17 | dijungal | the phone is behind a firewall and the asterisk box is open on the internet |
| 00:16.35 | fujin | lol |
| 00:16.42 | fujin | and how is asterisk supposed to traverse that NAT? |
| 00:16.47 | fujin | err, the phone |
| 00:17.09 | Aeudian | RTP is being blocked |
| 00:17.33 | Aeudian | asterisk uses RTP on ports UDP:10000-20000 |
| 00:17.53 | Aeudian | check your phones to see what they are using, cause Linksys Phones use RT UDP: 16384-16482 |
| 00:17.59 | dijungal | yes.. i have those ports open on the firewall |
| 00:18.00 | GlobeTrotter | ok guys,, i figured out what that first error was about,, now i have this second one that i keep getting;; translate.c:163 framein: no samples for g729tolin |
| 00:18.15 | Aeudian | dijungal: does your phones match the same rtp range? |
| 00:18.17 | dijungal | i'm using eyebeam phone |
| 00:21.24 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 00:21.32 | *** join/#asterisk fujin_ (n=aj@unaffiliated/fujin) |
| 00:21.57 | Aeudian | dijungal: you may also need to look into an external stun server, to traverse your IP externally |
| 00:22.05 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
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| 00:26.48 | *** join/#asterisk h0 (n=Fakhir@unaffiliated/fakhir) |
| 00:28.01 | *** join/#asterisk voiper1 (n=luke@ozvoip.dsl.onthenet.net) |
| 00:28.30 | dijungal | Aeudian: really... |
| 00:28.41 | dijungal | Aeudian: what does the stun server do? |
| 00:29.52 | *** join/#asterisk the_Goat (n=rsd095@c-71-224-187-182.hsd1.pa.comcast.net) |
| 00:30.06 | dijungal | so i've open up all ports to the asterisk server address |
| 00:30.25 | Aeudian | dijungal: STUN enables a device to find out its public IP address and the type of NAT service its sitting behind. for more info read here: http://www.voip-info.org/wiki-STUN |
| 00:33.00 | dijungal | funny thing is.. i've open all ports to the asterisk address on the firewall |
| 00:33.27 | dijungal | and set "nat=yes" in the sip.conf for that sip account |
| 00:34.26 | *** join/#asterisk Snake-eyes (n=hgjg@70.55.220.203.static.comindico.com.au) |
| 00:35.38 | fujin_ | yeah |
| 00:35.43 | *** join/#asterisk DarkRift (i=dark@bas10-montreal02-1177585253.dsl.bell.ca) |
| 00:35.45 | fujin_ | but have you got port forwarding on your firewall device that the phone sits behind |
| 00:35.50 | fujin_ | for the sip and rtp ports? |
| 00:37.58 | *** join/#asterisk Teln1100A (i=hello123@bas2-toronto12-1128737996.dsl.bell.ca) |
| 00:40.00 | Aeudian | I've seen phones try to handle the phone call instead of the voip server which causes nat issues as well |
| 00:40.21 | Aeudian | cause once this happens your nat and port fowarding mean jack |
| 00:40.34 | *** join/#asterisk jingles (n=foo@39.183.dowl.anc.borealisbroadband.net) |
| 00:41.21 | Sweeper | Aeudian: you mean reinvites? |
| 00:41.31 | Sweeper | yea, you need to turn those off :v |
| 00:41.59 | Aeudian | sweeper: i haven't seen it under asterisk yet, but SPA9000 w/ SPA942 phones seem to have this issue |
| 00:43.52 | *** join/#asterisk DarkRift (i=dark@bas10-montreal02-1177585253.dsl.bell.ca) |
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| 00:51.11 | *** join/#asterisk dec (n=tom@unaffiliated/dec) |
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| 00:58.02 | Sweeper | Aeudian: well, you can tell asterisk in sip.conf to disallow reinvites |
| 01:00.49 | n0n4m3 | Checksum: 0x21bd [incorrect, should be 0x7e33 (maybe caused by "UDP checksum offload"?)] |
| 01:00.51 | n0n4m3 | argh :S |
| 01:02.28 | *** join/#asterisk wunderkin (n=wunderki@ip72-223-86-126.ph.ph.cox.net) |
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| 01:14.19 | Teln1100A | how do I get asterisk by making a call from command line? |
| 01:14.22 | Teln1100A | I have added a trunk |
| 01:15.04 | [TK]D-Fender | Teln1100A, "help dial" <- |
| 01:15.18 | n0n4m3 | in case _ANYONE_ knows... how to convince asterisk to use 'proper' sip registering... |
| 01:16.09 | n0n4m3 | i have a problem.. i wouldn't like asterisk to connect as |
| 01:16.09 | n0n4m3 | From: "asterisk" <sip:asterisk@my.ip>;tag=as7dc61a96 |
| 01:16.55 | n0n4m3 | but rather than the user provided in register => |
| 01:19.22 | Teln1100A | terisk@my.ip>;tag=as7dc61a96 |
| 01:19.22 | Teln1100A | [21:16] <n0n4m3> but rather than the user |
| 01:19.33 | Teln1100A | help dial |
| 01:19.53 | Teln1100A | you mean on CLI> |
| 01:19.57 | [TK]D-Fender | Teln1100A, yes |
| 01:20.56 | Teln1100A | like I want to place a call |
| 01:21.03 | Teln1100A | not relate it to a sip phone |
| 01:21.09 | Teln1100A | I want asterisk to call |
| 01:21.22 | Teln1100A | to see if my trunk is functional |
| 01:21.46 | n0n4m3 | Teln1100A help dial is for me? |
| 01:22.15 | *** join/#asterisk guillote_GNU (n=guillote@190.7.27.17) |
| 01:22.37 | Teln1100A | oh |
| 01:22.41 | Teln1100A | that makes sense |
| 01:24.07 | fujin | anyone here have a PHP ui for reading out the cdr_mysql data? |
| 01:24.22 | *** join/#asterisk Juggie (n=Juggie@CPE00c049d9f271-CM000a73a18a20.cpe.net.cable.rogers.com) |
| 01:25.29 | [TK]D-Fender | fujineasy enough to do as its jsut CSV. There's a single function to parse out a line. |
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| 01:34.22 | Teln1100A | I added a sip extension but my asterisk server is not accepting connecitons |
| 01:34.28 | Teln1100A | what do I need to do to enable this? |
| 01:36.56 | [TK]D-Fender | Teln1100A, could you be any MORE vague? |
| 01:40.50 | n0n4m3 | nite |
| 01:45.30 | *** join/#asterisk riddlebox (n=james@ppp-70-225-67-135.dsl.covlil.ameritech.net) |
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| 02:01.48 | WilliamK | who wants to fix the zaptel in svn space.....http://www.pastebin.ca/681740 |
| 02:04.42 | Teln1100A | how do I get asterisk to accept sip connections, I am trying to connect xlite softphone to asterisk |
| 02:05.30 | [TK]D-Fender | Teln1100A, setup your phone properly, make sue the context matches your dialplan context, and that you are dialing a valid exten in it |
| 02:05.31 | *** join/#asterisk Aeudian (n=chatzill@c-69-250-24-154.hsd1.md.comcast.net) |
| 02:06.04 | Aeudian | Anyone have a working "Steal2" script which pickups phone call that is on hold on a specific phone? |
| 02:06.21 | Teln1100A | the issue is on the asterisk server |
| 02:06.30 | Teln1100A | it is new and has never worked before |
| 02:06.39 | *** join/#asterisk dijungal (n=kdaniel@208.0.231.66) |
| 02:06.50 | *** join/#asterisk Tili (n=tili@cm48.gamma244.maxonline.com.sg) |
| 02:08.36 | Teln1100A | I get 404 on softphone |
| 02:08.56 | [TK]D-Fender | Teln1100A, show some SIP debug & CLI output to back it up as well as your dialplan. All in a PASTEBIN please. |
| 02:08.58 | [TK]D-Fender | ~pb |
| 02:08.59 | jbot | A Pastebin is a place to paste your stuff without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste |
| 02:09.17 | [TK]D-Fender | Teln1100A, 404 means youare NOT dialing a valid exten for the context being used |
| 02:09.25 | Teln1100A | please tell me what command you would like me enter on CLI |
| 02:09.32 | [TK]D-Fender | Teln1100A, make sure you're even using the right one |
| 02:09.34 | Teln1100A | 404 is before registration |
| 02:09.41 | Teln1100A | Registration error: 404 is before registration |
| 02:10.05 | [TK]D-Fender | 404 on register? means your phone's credentials don't match the peer you set up. |
| 02:10.14 | *** join/#asterisk thansen|laptop (n=thansen@19.243.sfcn.org) |
| 02:10.27 | Teln1100A | oh |
| 02:10.47 | *** join/#asterisk mordaunt (n=mordaunt@unaffiliated/mordaunt) |
| 02:15.22 | *** join/#asterisk etfonhomey_ (n=chatzill@74-131-136-195.dhcp.insightbb.com) |
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| 02:27.51 | *** join/#asterisk johnadsfsdfdf (n=john@cpe-72-226-69-65.nycap.res.rr.com) |
| 02:28.01 | johnadsfsdfdf | whats the best way to install asterisk 1.4 on debian? |
| 02:28.18 | *** join/#asterisk ManxPower (n=manxpowe@148.sub-70-218-13.myvzw.com) |
| 02:29.12 | *** join/#asterisk Defraz (n=t0tal@24-116-152-177.cpe.cableone.net) |
| 02:29.23 | [TK]D-Fender | johnadsfsdfdf, wget |
| 02:30.19 | johnadsfsdfdf | hi again [TK]D-Fender after yesterday i've decided to just try installing 1.4 to see if that will get my extensions to work |
| 02:30.44 | johnadsfsdfdf | because those conf files were working on a 1.4 asterisk on redhat |
| 02:31.23 | [TK]D-Fender | johnadsfsdfdf, I recall you having some sort of issue, but none of the details... |
| 02:31.52 | johnadsfsdfdf | when you called into the PBX it didnt recognize extensions being dialed, not even buttons being pressed |
| 02:32.14 | johnadsfsdfdf | but when it ran a macro and called out to my cell phone it would read the confirmation for the connection |
| 02:32.24 | ManxPower | 1.2 had DTMF issues with a couple of providers, usually it was dialing OUT to IVRs. |
| 02:32.41 | johnadsfsdfdf | my provider is broadvoice |
| 02:32.52 | ManxPower | and only with SIP |
| 02:33.06 | ManxPower | and mostly with providers who used Level 3 as their provider |
| 02:33.22 | johnadsfsdfdf | i dont know who my providers provider is... lol |
| 02:33.38 | ManxPower | you should. |
| 02:34.08 | johnadsfsdfdf | do you have any recommended providers for sip service? |
| 02:34.37 | fujin | I'm using worldxchange, here in NZ |
| 02:34.48 | fujin | they run broadvoice, everything seems nice so far. |
| 02:35.13 | *** join/#asterisk Gamercjm (n=chris@pool-71-254-179-93.lsanca.fios.verizon.net) |
| 02:35.15 | johnadsfsdfdf | worldxchange.com? |
| 02:35.22 | johnadsfsdfdf | that website is scary |
| 02:36.15 | ManxPower | All providers suck! Some suck less than others. (c)2007 ManxPower |
| 02:36.26 | ManxPower | Teliax is one of those that usually seems to suck much less than most. |
| 02:37.52 | *** join/#asterisk Maliuta (n=nikolai@kiev.lusan.id.au) |
| 02:40.16 | Gamercjm | for musiconhold, how do I use it if im speaking with someone and I want to put them on hold? ive only seen where the user calls and it directs them to a sertain extension till someone answers |
| 02:40.47 | fujin | johnadsfsdfdf, wxc.co.nz |
| 02:40.50 | fujin | not sure of their other ones. |
| 02:40.54 | fujin | I doubt worldxchange.com is them |
| 02:41.07 | ManxPower | Gamercjm: you use the HOLD button on your phone |
| 02:41.53 | Teln1100A | how do you enable sip from asterisk to listen on all ips |
| 02:42.24 | Gamercjm | Im using a Mitel IP phone and when I press HOLD it puts them on hold, but its silent. It doesnt seem to be going to the musiconhold feature |
| 02:42.37 | fujin | Gamercjm, have you defined the musiconhold class in sip.conf? |
| 02:43.18 | Gamercjm | I believe i have, but let me double check |
| 02:45.25 | [TK]D-Fender | Gamercjm, And go make sure you HAVE music you can play for the mode specified and that all of its dependencies are met |
| 02:46.45 | *** join/#asterisk panitaxx (n=i@nelug/developer/panitaxx) |
| 02:51.05 | *** join/#asterisk Krurst (n=me@eth244.wa.adsl.internode.on.net) |
| 02:52.33 | fujin_ | does the console say 'stating music on hold for <blah>'? |
| 03:02.26 | *** join/#asterisk craigk (n=ckowald@58.174.113.53) |
| 03:03.29 | fujin | doh |
| 03:03.37 | fujin | anyone know what 'exited non-zero' means, and if I can drill down any further? |
| 03:03.41 | fujin | getting obscure messages like |
| 03:03.42 | fujin | == Spawn extension (macro-queue_helpdesk, s, 7) exited non-zero on 'SIP/maxnetvoip-b5a0d6e8' in macro 'queue_helpdesk' |
| 03:08.15 | Krurst | it just means a function didn't return 0 when it finished. Sometimes this indicates an error, sometimes its normal behaviour, depends on the function. |
| 03:09.07 | fujin_ | yeah I gathered that much |
| 03:09.11 | fujin_ | just wanted to know why* it was happening |
| 03:09.23 | johnadsfsdfdf | what should the permissions on /etc/init.d scripts be set to? |
| 03:09.38 | fujin_ | root:root 700 |
| 03:09.44 | johnadsfsdfdf | thats what i thought |
| 03:10.13 | johnadsfsdfdf | why doesn't the debian default install do that? |
| 03:10.40 | fujin_ | it's crap? |
| 03:13.13 | Teln1100A | I have a softphone registered to my asterisk server, and a trunk as well, but I can not make any calls. I get the message: Callfailed, Not found |
| 03:13.23 | Teln1100A | How can I fix this? |
| 03:13.32 | fujin_ | oh god |
| 03:14.02 | fujin_ | have you given asterisk a way to handle calls? |
| 03:14.04 | johnadsfsdfdf | oops, i meant to ask that in #debian |
| 03:14.15 | johnadsfsdfdf | oh well, thanks anyways |
| 03:14.21 | Teln1100A | do you mean extensions.conf? |
| 03:14.57 | *** join/#asterisk stkn__ (n=stkn@gentoo/developer/pdpc.active.stkn) |
| 03:15.03 | fujin_ | yes, |
| 03:15.12 | fujin_ | have you given asterisk a way to handle calls, in extensions.conf/ael? |
| 03:16.07 | Teln1100A | think I am missing that |
| 03:16.16 | Teln1100A | thought I had put teliax info somewhere |
| 03:16.34 | fujin_ | yes, well |
| 03:16.38 | fujin_ | without a way to *handle* calls |
| 03:16.42 | fujin_ | registering devices is useless |
| 03:17.06 | Teln1100A | I had put in a trunk at Freepbx web interface |
| 03:17.15 | fujin_ | k |
| 03:17.16 | fujin_ | #freepbx |
| 03:17.27 | fujin_ | we don't support that shit here |
| 03:17.32 | *** join/#asterisk sysreq (n=sysreq@unaffiliated/sysreq) |
| 03:17.40 | Teln1100A | I am open to doing it without that |
| 03:17.49 | Teln1100A | can they not work together? |
| 03:17.53 | fujin_ | no |
| 03:17.59 | Teln1100A | say if I made changes in config files? |
| 03:18.14 | fujin_ | that'd be dumb |
| 03:18.26 | fujin_ | freepbx is overkill for most peoples needs |
| 03:18.28 | Teln1100A | do you recommend any gui for asterisk? |
| 03:18.35 | fujin_ | guis are for muppets and windows weenys |
| 03:18.42 | fujin_ | vim is the only gui you need tbh |
| 03:19.19 | ManxPower | ~zeeek |
| 03:19.19 | jbot | i heard zeeek is someone who once said "learning asterisk using a GUI is like learning sex through masturbation. You'll never get to the good stuff." |
| 03:19.24 | ManxPower | He said it best. |
| 03:19.32 | fujin_ | +1 zeeek |
| 03:19.38 | Teln1100A | what cli command can I use to see if the trunk has been setup correctly? |
| 03:19.54 | fujin_ | by trunk I assume you mean 'peer? |
| 03:20.11 | Teln1100A | teliax account used to make outgoing calls |
| 03:20.12 | ManxPower | Teln1100A: if it works, it is set up correctly and "trunk" is a GUI term has virtually no meaning in the voip world |
| 03:20.37 | Teln1100A | ok |
| 03:20.40 | fujin_ | obviously you need to build the functionality in extensions.conf/ael to dial across your teliax peer |
| 03:20.49 | fujin_ | and then calling will work, and your test wil be succesful |
| 03:20.50 | Teln1100A | I didnt really like freepbx |
| 03:20.54 | fujin_ | there's not really any other way to do it |
| 03:20.57 | *** join/#asterisk CVirus (n=GoD@41.233.134.17) |
| 03:20.57 | fujin_ | what kind of setup do you have? |
| 03:21.02 | Teln1100A | vps server |
| 03:21.06 | fujin_ | I mean |
| 03:21.10 | fujin_ | uh; |
| 03:21.15 | Teln1100A | no DID |
| 03:21.15 | fujin_ | devices, purpose |
| 03:21.22 | Teln1100A | softphone only |
| 03:21.25 | Teln1100A | xlite |
| 03:21.39 | Teln1100A | just getting started with Asterisk |
| 03:22.02 | Teln1100A | one other thing, should I wipe the yum instlal and compile from source? |
| 03:22.13 | Teln1100A | operating system is CentOS 5 |
| 03:22.29 | fujin_ | yes |
| 03:22.47 | fujin_ | buidling asterisk from source is a definite win-win situation |
| 03:22.57 | fujin_ | you get all the documentation, all features enabled (unless you specifically disable them) |
| 03:23.06 | Teln1100A | yea, thought so |
| 03:23.10 | fujin_ | access to the upstream maintainers changelogs |
| 03:23.24 | Teln1100A | Removed: asterisk.i386 1:1.4.11-46.el5 |
| 03:23.31 | fujin_ | a small performance gain by building for your local platform instead of running a package build for someone elses platform |
| 03:23.34 | fujin_ | up to date package, too |
| 03:25.32 | CVirus | Teln1100A: this might help http://www.voip-info.org/wiki/index.php?page=Asterisk+Linux+CentOS |
| 03:26.21 | Teln1100A | since this is a vps, kernel headers are not availble |
| 03:26.38 | Teln1100A | virtuozo proprietary kernel |
| 03:26.44 | Teln1100A | will that be an issue? |
| 03:26.48 | Sweeper | SOL mang |
| 03:26.54 | CVirus | no idea really |
| 03:27.02 | antimoof | server hardwasre is cheap. |
| 03:27.04 | Sweeper | I mean, you could try running sans ztdummy |
| 03:27.12 | Sweeper | but it'll probably be bad |
| 03:27.27 | fujin_ | lol |
| 03:27.28 | Sweeper | since you need a timer for some stuff |
| 03:27.47 | fujin_ | I don't even have ztdummy; running a pure-sip implementation |
| 03:27.52 | fujin_ | haven't noticed any reason to have it, yet ;} |
| 03:28.37 | CVirus | fujin_: you'll need it for the meetme application |
| 03:28.45 | Sweeper | ah, that~ |
| 03:28.46 | fujin_ | meetme can pissoff |
| 03:28.50 | fujin_ | solved |
| 03:28.55 | Sweeper | hmmm |
| 03:28.59 | Sweeper | AHA! |
| 03:29.00 | Sweeper | I know~ |
| 03:29.00 | CVirus | hehe |
| 03:29.02 | russellb | fujin_: well that's not very nice to say to meetme |
| 03:29.08 | fujin_ | russellb, my apologies |
| 03:29.11 | russellb | :-p |
| 03:29.21 | Sweeper | run a small freeswitch instance |
| 03:29.29 | Sweeper | do all your conf there |
| 03:29.52 | fujin_ | not sure that's a good idea |
| 03:29.58 | fujin_ | I've grown quite fond of AEL, anyway :0 |
| 03:30.12 | russellb | nice |
| 03:30.33 | fujin_ | absolutely |
| 03:30.44 | fujin_ | russellb, your devstate has been working great by the way |
| 03:30.48 | Sweeper | fujin_: well, it solves the conf problem, and freeswitch has a pretty small footprint |
| 03:31.06 | russellb | fujin_: good to hear |
| 03:31.11 | fujin_ | Sweeper, mm, true, but I doubt I can do the things with it that I can do with asterisk, I have a pretty complex callcentre setup |
| 03:31.16 | fujin_ | hotdesking, trackable agents |
| 03:31.23 | fujin_ | device state-based queue call delivery |
| 03:31.30 | Sweeper | well, you CAN |
| 03:31.37 | Sweeper | it just takes actual effort ;) |
| 03:31.41 | Sweeper | but that's not what I'm saying |
| 03:31.48 | Sweeper | just have freeswitch do conference calls |
| 03:31.54 | fujin_ | ah |
| 03:31.55 | Sweeper | do everything else with asterisk |
| 03:32.03 | fujin_ | I've just been doing conf calls with the devices |
| 03:32.09 | fujin_ | bridge two calls to the one device, seems to work ok |
| 03:32.13 | fujin_ | 3way calling anyway |
| 03:32.16 | Sweeper | yep |
| 03:32.47 | fujin_ | What's that? |
| 03:32.59 | Sweeper | it's a voip integration library |
| 03:33.13 | russellb | ruby + agi |
| 03:33.15 | Sweeper | currently asterisk-only, but next rev is gonna be freeswitch compatible as well |
| 03:33.44 | Sweeper | so I can whip up all the asterisk apps from scratch in <100 LOC |
| 03:33.52 | Sweeper | each, that is~ |
| 03:33.58 | fujin_ | the joys of ruby, right? :\ |
| 03:34.07 | Sweeper | yea! |
| 03:34.19 | russellb | i don't think you can write a conferencing application in less than 100 lines of ruby. |
| 03:34.28 | fujin_ | have you seen the video fo the guy making the blogging engine in 10 minutes in ruby on rails? |
| 03:34.31 | fujin_ | it's pretty incredible |
| 03:34.40 | Sweeper | russellb: errr |
| 03:34.44 | fujin_ | there must be some massive backend functionality to be able to generate that kind of code |
| 03:35.24 | Sweeper | all you gotta do is...play announces, join people, prompt for pins, log, and allow stuff like kicks/mutes |
| 03:35.45 | Sweeper | probably a 50 line job |
| 03:35.47 | russellb | and do audio mixing? :) |
| 03:35.51 | Sweeper | err |
| 03:35.54 | Teln1100A | is this package necessary for voip only ie no cards ? zaptel-1.4.2.1.tar.gz |
| 03:35.56 | Sweeper | that's what freeswitch does :P |
| 03:36.09 | Sweeper | they've got a mixer |
| 03:36.21 | Sweeper | you've just got to do the logic |
| 03:36.46 | russellb | i can do conferencing with 1 line of extensions.conf :-p |
| 03:38.09 | fujin_ | russellb wins |
| 03:38.28 | osiris | just out of curiosty, does it take anything special on the provider end to register and call with a *box |
| 03:38.48 | Sweeper | yea, but you can't scale, have to WRITE extensions.conf, which sucks, and have to deal with the limitations of meetme |
| 03:39.02 | fujin_ | meh, I don't *ever* write extensions.conf |
| 03:39.04 | osiris | if i have a byod ATA, can i register the *box if i know the auth details |
| 03:39.16 | Sweeper | done properly in ruby, any conf app can be directly extended from the dialplan |
| 03:39.27 | Sweeper | pass blocks, etc |
| 03:44.22 | Teln1100A | done compiling asterisk |
| 03:44.29 | fujin_ | congratulations |
| 03:44.32 | russellb | yay |
| 03:44.36 | Teln1100A | and registered sip |
| 03:44.45 | Teln1100A | now to add teliax for outgoing calls |
| 03:45.07 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 03:45.11 | fujin_ | have fun |
| 03:45.17 | Teln1100A | whats better sip or iax? |
| 03:45.25 | fujin_ | depends on the purpose |
| 03:45.29 | russellb | iax!! |
| 03:45.36 | Teln1100A | ok |
| 03:45.40 | russellb | IAX unless it's not an option :) |
| 03:45.45 | fujin_ | heh |
| 03:45.51 | fujin_ | do the polycoms run IAX yet? that'd be awesome. |
| 03:45.55 | russellb | not yet, no |
| 03:46.14 | Teln1100A | teliax < IAX > asterisk |
| 03:46.21 | russellb | though I have heard of the potential for at least one major manufacturer to start supporting it |
| 03:46.24 | russellb | we'll see! |
| 03:46.26 | Teln1100A | but sip to phones |
| 03:46.52 | fujin_ | telstralear shot me down when I asked about an iax trunk |
| 03:46.58 | fujin_ | because apparently, it's a proprietary protocol |
| 03:47.13 | fujin_ | unfortunately they don't understand that proprietary doesn't mean bad, especially in the case of free/oss -_- |
| 03:47.17 | russellb | fujin_: are you serious? heh |
| 03:47.25 | russellb | it's not proprietary at all |
| 03:47.31 | russellb | there is an RFC draft for it ... |
| 03:47.34 | fujin_ | I see |
| 03:47.45 | fujin_ | well, the name kind of implies that it is, so that is maybe where they got tha tfrom |
| 03:47.51 | fujin_ | generally I'd put it down to ignorance. |
| 03:48.01 | russellb | heh, yeah .. |
| 03:48.12 | Teln1100A | <PROTECTED> |
| 03:52.30 | *** join/#asterisk bmg505 (n=leon@196.209.178.180) |
| 03:55.13 | Teln1100A | what does teliax need in extensions.ael to work? |
| 03:55.15 | *** join/#asterisk thansen|laptop (n=thansen@19.243.sfcn.org) |
| 03:56.06 | scooby2 | why does wanpipe have to be such a pain in the butt |
| 03:57.57 | [TK]D-Fender | Teln1100A, .... |
| 03:57.58 | [TK]D-Fender | ~book |
| 03:57.59 | jbot | [book] a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 03:58.02 | [TK]D-Fender | ^^^^^^^^^^^^^^^^^^^^^^ |
| 03:58.05 | *** join/#asterisk Yourname` (n=chatzill@76-10-128-178.dsl.teksavvy.com) |
| 03:58.18 | [TK]D-Fender | time to find a clue |
| 03:58.23 | Teln1100A | hmm, |
| 03:58.38 | Teln1100A | I compiled asterisk today |
| 03:59.06 | russellb | i did too :) |
| 03:59.14 | Teln1100A | is yours working yet? |
| 03:59.17 | Yourname` | Hi, so using ulaw and each box using approximately 300 channels, and 4 boxes of such.. are utilizing 100mbps of bandwidth. Just wondering what else can I use that will not give quality problems yet be easy on the bandwidth? |
| 03:59.40 | russellb | Teln1100A: sometimes, though i tend to break it throughout the day as I change things ;) |
| 03:59.56 | Teln1100A | [TK]D-Fender besides your link doesnt work |
| 04:00.07 | jql | that's good traffic |
| 04:00.18 | Teln1100A | so many files and options |
| 04:00.25 | Teln1100A | I get really confused |
| 04:00.41 | Teln1100A | what are your thoughts on freepbx or other web interface? |
| 04:00.42 | [TK]D-Fender | ~tfot |
| 04:00.43 | jbot | tfot is, like, "The Future of Telephony", a book about Asterisk from O'Reilly Publishing, ISBN: 0-596-00962-3, click http://www.oreilly.com/catalog/asterisk/ for more details |
| 04:00.55 | [TK]D-Fender | ~thebook |
| 04:00.56 | jbot | it has been said that thebook is a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 04:00.58 | russellb | ~gui |
| 04:00.59 | jbot | gui is probably (Graphical User Interface) Any system that uses graphics to represent the functions of a program. All Windows operating systems are GUI. or point-and-drool, double-click, drag-and-drop, ... or at http://pla-netx.com/linebackn/guis/index.html. Of course Real Programmers use the command line interface. See cli |
| 04:01.09 | russellb | o.O |
| 04:01.13 | Teln1100A | I see |
| 04:01.14 | russellb | not what i was looking for |
| 04:01.33 | Teln1100A | wasnt looking for the die hard everything command line answer |
| 04:01.50 | russellb | Teln1100A: have you taken a look at asterisknow.org ? |
| 04:01.51 | Teln1100A | I realize a lot of things are easier to do with the command line and are important to know how |
| 04:02.05 | Teln1100A | but sometimes theres ease of use in a GUI |
| 04:02.10 | russellb | i like that gui more than freepbx, but i am biased |
| 04:02.34 | Teln1100A | is that an OS? |
| 04:02.52 | russellb | yeah, full linux distro, but you can install the same gui by itself |
| 04:02.56 | Teln1100A | am running on a VPS so Operating system is out of question |
| 04:03.09 | russellb | it's only a few commands to get it installed, actually |
| 04:03.12 | Teln1100A | has to be a package or install |
| 04:03.24 | russellb | because it doesn't require any other packages ... |
| 04:03.36 | Teln1100A | yes, but I am restricted to use my providers OS |
| 04:03.37 | russellb | if you join #asterisk-gui, the commands to install it are in the topic |
| 04:03.46 | Teln1100A | they virtualize |
| 04:03.50 | russellb | the gui doesn't require any other OS or anything |
| 04:03.59 | russellb | it's just a bunch of html / javascript |
| 04:04.07 | Teln1100A | oh ok |
| 04:04.08 | russellb | and it talks to asterisk directly |
| 04:04.18 | Teln1100A | it can be installed on top of asterisk? |
| 04:04.24 | Teln1100A | or rather after asterisk |
| 04:04.30 | russellb | asterisk 1.4 has a built in mini web server, and a management interface over http, which is what the gui uses to manage asterisk |
| 04:04.33 | russellb | correct |
| 04:04.47 | *** join/#asterisk Cresl1n (n=matt@c-68-62-219-187.hsd1.al.comcast.net) |
| 04:04.47 | *** mode/#asterisk [+o Cresl1n] by ChanServ |
| 04:04.59 | russellb | Cresl1n: !!!!!!!!! |
| 04:05.07 | Cresl1n | russellb! |
| 04:05.09 | Cresl1n | no way! |
| 04:05.14 | Cresl1n | it's you!? |
| 04:05.18 | russellb | IT IS |
| 04:05.20 | Teln1100A | does it come in a packaged format |
| 04:05.26 | Teln1100A | or iso only? |
| 04:05.31 | Sweeper | http://www.badmouth.net/graphics/warp_11_Kiki_Sing.php <-- all I can say is "YES" |
| 04:05.42 | russellb | Teln1100A: the gui can be installed by itself, just check the topic of #asterisk-gui |
| 04:05.49 | Cresl1n | check this out, we just got two b channel transfer working in libpri for NI2, 5ESS, and 4ESS |
| 04:05.53 | russellb | Teln1100A: you're just putting some files in the right directory and changing a few asterisk options |
| 04:05.59 | russellb | Cresl1n: nice!!!!! |
| 04:06.02 | Cresl1n | matt florrell confirmed it for me today |
| 04:06.05 | russellb | Cresl1n: i've been seeing you hacking on that code :) |
| 04:06.14 | russellb | nice job! |
| 04:06.21 | Cresl1n | yeah, the man let me have some time to work on it and it works now |
| 04:06.26 | russellb | lol |
| 04:06.27 | Cresl1n | I gotta merge it all back in now |
| 04:06.46 | russellb | dude, i'm really hoping i can make time to learn some driver stuff |
| 04:06.55 | Cresl1n | :-) |
| 04:06.57 | scooby2 | anyone here using wanpipe? |
| 04:06.59 | russellb | i may need your help figuring some things out :) |
| 04:07.00 | Cresl1n | that would be awesome |
| 04:07.05 | Cresl1n | funny you mentioned it |
| 04:07.13 | Cresl1n | I just started writing a brand new driver today |
| 04:07.17 | russellb | oooooh |
| 04:07.22 | russellb | nice! |
| 04:07.27 | Cresl1n | which doesn't happen too often |
| 04:07.32 | russellb | that rocks |
| 04:07.37 | Cresl1n | I'm going to write a zaptel driver for the b410p |
| 04:07.43 | russellb | ooooh |
| 04:07.44 | [TK]D-Fender | scooby2, specific questions tend to get specific answers.... |
| 04:08.17 | russellb | Cresl1n: i read a couple of chapters in a book this morning on interrupy processing and "bottom halves" :) |
| 04:08.23 | russellb | interrupt* |
| 04:08.25 | Cresl1n | it's pretty much a side project though, so I don't know how quickly it's going to go |
| 04:08.27 | Cresl1n | oh, good stuff |
| 04:08.38 | Cresl1n | those are your tools of the trade for drivers |
| 04:08.51 | russellb | Cresl1n: it's not really so much black magic like i thought :-p |
| 04:08.54 | Cresl1n | pretty much all your real work is done with interrupt handlers |
| 04:09.06 | Cresl1n | at least for DMA'ing cards |
| 04:09.41 | Cresl1n | but this driver I think you could do all the fun new stuff you want to with it |
| 04:09.50 | Yourname` | Is there somewhere a tutorial that documents nicely about running asterisk as non-root? The wiki sounds scary for some reason. |
| 04:09.53 | *** join/#asterisk dalbaech (i=narf@youhackme.com) |
| 04:09.58 | Cresl1n | since mISDN only works on fairly recent 2.6 kernels anyways |
| 04:10.02 | russellb | Cresl1n: only support 2.6? |
| 04:10.05 | russellb | niiice |
| 04:10.12 | Cresl1n | so if you want to hack away with cool cleanups, then I think it'd be fair game |
| 04:10.29 | Cresl1n | since it's a new driver |
| 04:10.40 | Cresl1n | you're no worse of than you were before with the old mISDN drivers |
| 04:11.00 | russellb | you mean hack on it later on after you have it written but before it's really released? |
| 04:11.23 | Cresl1n | could be either before or after |
| 04:11.28 | *** part/#asterisk dalbaech (i=narf@youhackme.com) |
| 04:11.29 | scooby2 | [TK]D-Fender: just trying to find out which version people are using. I cannot get wanpipe-2.3.4-13 to compile on centos5 or ubuntu dapper |
| 04:11.47 | [TK]D-Fender | scooby2, pastebin is your friend.... |
| 04:11.48 | Cresl1n | anyways, gotta sleep |
| 04:11.53 | Cresl1n | good luck with the LDD book :-) |
| 04:11.59 | russellb | Cresl1n: alright, well i'll talk to you later about it |
| 04:12.07 | russellb | like ... at work and not the middle of the night :-p |
| 04:12.11 | Cresl1n | heh :-) |
| 04:12.14 | russellb | g'night |
| 04:12.17 | Yourname` | Maybe there should be something in asterisk1.4 that helps people install as non-root from scratch. :) |
| 04:12.18 | Cresl1n | I got an 8:00 class tomorrow |
| 04:12.24 | russellb | yikes |
| 04:12.25 | *** join/#asterisk Qapf (n=Qapf@mail.oldworlddoor.com) |
| 04:12.27 | Cresl1n | amen to that |
| 04:12.28 | Cresl1n | nite all! |
| 04:12.29 | russellb | good luck |
| 04:12.31 | russellb | WilliamK: 8 |
| 04:12.36 | russellb | WilliamK: ignore that ... |
| 04:12.55 | Yourname` | Class at 8? I have to be up at 5am. And it's 12.12am right now.. :( |
| 04:14.02 | WilliamK | hiya russellb |
| 04:14.06 | WilliamK | long time no see |
| 04:14.10 | russellb | hey :) |
| 04:14.54 | russellb | i type your name sometimes accidentally because i use a console IRC client, and I type "/wi<tab> <some number> <enter>", and miss the '/' |
| 04:15.12 | WilliamK | ah! |
| 04:15.14 | russellb | and it completes to your nick instead of the command, heh |
| 04:15.49 | WilliamK | and here I thought you were wanting to fix0r my zaptel prob with the svn :) |
| 04:15.56 | russellb | ha |
| 04:16.29 | WilliamK | xmas wish I guess? :) |
| 04:16.41 | russellb | put it in the bug tracker :) |
| 04:17.13 | *** join/#asterisk mitcheloc (n=mitchel@adsl-68-120-230-95.dsl.irvnca.pacbell.net) |
| 04:17.14 | WilliamK | to be quite honest, I've never ever done anything in bug tracker |
| 04:17.15 | WilliamK | =) |
| 04:17.39 | WilliamK | and all I can really say is it's broke :) |
| 04:17.52 | russellb | ha |
| 04:17.57 | russellb | most people that report bugs aren't, it's fine |
| 04:18.04 | russellb | but we do need more than "it's broke" :-p |
| 04:18.18 | Yourname` | codefreeze once helped me do the bug thing. |
| 04:18.33 | Yourname` | And I was all sissy thinking "Maybe it's me! Maybe I'm doing something wrong" lol |
| 04:18.46 | Sweeper | russellb: there's a bad bit of assembly in the tor2 kernel module |
| 04:19.01 | russellb | oh yeah? |
| 04:19.04 | WilliamK | I've used zaptel enough I know it's not me personally, just don't know howto fix |
| 04:19.08 | Sweeper | /usr/src/zaptel/tor2.c:603: error: impossible constraint in â..asmâ.. |
| 04:19.23 | WilliamK | http://www.pastebin.ca/681740 |
| 04:19.26 | russellb | i don't think that code has changed recently ... |
| 04:19.27 | Sweeper | well, assuming that refers to asm, maybe it's just a function |
| 04:19.30 | WilliamK | that's my entire screenshot |
| 04:21.08 | Yourname` | And I'm trying another 1.2 to 1.4 upgrade, and it gives me this error in the end: /usr/bin/install: cannot stat `app_addon_sql_mysql.so': No such file or directory |
| 04:21.10 | Yourname` | Along with others. |
| 04:21.21 | Yourname` | I mv'd the old modules directory to something else so it doesn't conflict. |
| 04:21.36 | Yourname` | Even configured the addons directory.. to no avail. |
| 04:21.41 | Yourname` | What is wrong? |
| 04:24.28 | russellb | WilliamK: that's enough to post to the bug tracker |
| 04:24.56 | russellb | WilliamK: the important other stuff is kernel version (uname -a), and distribution info |
| 04:25.23 | russellb | i'm thinking it's a distro or kernel version specific issue |
| 04:26.38 | Yourname` | Can someone please point me to the right direction? |
| 04:28.53 | Yourname` | Errors -> /usr/bin/install: cannot stat `app_addon_sql_mysql.so': No such file or directory |
| 04:28.55 | russellb | Yourname`: that's odd ... on "make install" ? |
| 04:28.59 | Yourname` | Yessir. |
| 04:29.07 | russellb | did you run "make" by itself? |
| 04:29.08 | *** join/#asterisk aris_g (n=andres@190.25.97.227) |
| 04:29.10 | Yourname` | Yup. |
| 04:29.17 | aris_g | Hello . |
| 04:29.32 | russellb | Yourname`: hrm, try "make distclean ; ./configure ; make ; make install" |
| 04:29.45 | Yourname` | russelb: In the addons? |
| 04:30.31 | russellb | yep |
| 04:31.01 | aris_g | Do you know if exist some firmware telephone IP that can be modified...? |
| 04:31.13 | Yourname` | russellb: One sec.. |
| 04:31.25 | aris_g | some firmware with source code.....any idea? |
| 04:31.50 | WilliamK | russell: 2 different kernels, only thing common is the distro |
| 04:32.12 | jql | the snom runs linux, but I don't know if its telecom software is open |
| 04:32.18 | jql | probably not |
| 04:32.45 | jql | although it'd be funny if someone ported asterisk to it |
| 04:32.46 | russellb | WilliamK: and compiler version i suppose ... |
| 04:32.49 | russellb | WilliamK: gcc --version |
| 04:33.28 | Yourname` | russellb: make install ended this way -> http://pastebin.ca/681825 |
| 04:33.29 | WilliamK | 2 different versions of gcc |
| 04:33.31 | WilliamK | :) |
| 04:33.46 | Yourname` | And I see no mention of mysql stuff. :( |
| 04:34.28 | aris_g | i see... thanks jql |
| 04:35.02 | russellb | Yourname`: http://bugs.digium.com/file_download.php?file_id=14534&type=bug |
| 04:35.05 | russellb | Yourname`: make that change |
| 04:35.16 | russellb | and try again .. |
| 04:35.19 | Yourname` | Oh great.. I don't know how to.. :S |
| 04:35.29 | russellb | ok, then do ..... |
| 04:35.38 | Teln1100A | I keep getting == Connect attempt from '127.0.0.1' unable to authenticate on CLI trying Asterisk GUI |
| 04:35.51 | *** part/#asterisk dec (n=tom@unaffiliated/dec) |
| 04:37.14 | russellb | $ cd src/asterisk-addons ; for n in Makefile.am Makefile.in ; do sed -i -e 's/libchan_h323.so.1/libchan_h323.1/' asterisk-ooh323c/${n} ; done |
| 04:37.32 | russellb | Teln1100A: run "make checkconfig" in the GUI directory where you installed it from |
| 04:37.42 | russellb | Teln1100A: it will help you with the necessary asterisk config changes |
| 04:38.18 | Teln1100A | did that |
| 04:38.22 | Teln1100A | it said all ok |
| 04:39.08 | Yourname` | russellb: Done, so I do the make distclean in the addons dir and do it again? |
| 04:39.08 | russellb | i've got to go bed ... |
| 04:39.27 | russellb | Yourname`: try just "make && make install" |
| 04:40.11 | Yourname` | russellb: Seems to have worked!!! |
| 04:40.15 | russellb | yay! |
| 04:40.18 | Yourname` | russellb: Let me try the install now, lol |
| 04:40.20 | russellb | k |
| 04:40.22 | Yourname` | Thanks so much, one sec. |
| 04:41.00 | WilliamK | hey russellb, in bugtracker should I be listing the catagory as zaptel or ? -- the whole package fails the entire make proccess |
| 04:41.33 | russellb | zaptel/general probably |
| 04:41.55 | WilliamK | and should the thing be listed as block or major for severity? |
| 04:42.48 | russellb | WilliamK: i would say "minor" because come to think of it, you can easily disable that one module from building "assumign you don't need it" |
| 04:42.53 | *** join/#asterisk techie (n=techie@c-67-181-184-170.hsd1.ca.comcast.net) |
| 04:43.01 | russellb | run "make menuselect", turn off the tor2 module, hit 'x' to save and quit |
| 04:43.07 | scooby2 | http://pastebin.ca/681829 |
| 04:43.09 | WilliamK | k, lemme try that |
| 04:43.11 | *** part/#asterisk techie (n=techie@c-67-181-184-170.hsd1.ca.comcast.net) |
| 04:43.27 | scooby2 | what i'm getting from latest wanpipe (beta and stable) on centos5 and ubuntu dapper |
| 04:43.44 | russellb | i'm off to bed ... good luck ... |
| 04:45.24 | WilliamK | thanks, it failed again even turning that off also |
| 04:45.25 | Yourname` | russellb: Good night.. and thank you. :) |
| 04:49.59 | WilliamK | okie, finally got it to work |
| 04:50.02 | WilliamK | 3 mods are bad |
| 05:01.20 | Yourname` | And I still can't get it to work, sigh |
| 05:01.23 | Yourname` | Good night errbody. |
| 05:02.47 | *** part/#asterisk aris_g (n=andres@190.25.97.227) |
| 05:03.11 | WilliamK | so who thinks I should list all 3 broken mods on 1 report and who thinks I should list them separately? |
| 05:04.32 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 05:07.44 | *** join/#asterisk workaphobia (n=workapho@magneton-35.dynamic.rpi.edu) |
| 05:08.02 | *** join/#asterisk grndslm (n=grndslm@24-116-87-97.cpe.cableone.net) |
| 05:15.44 | *** join/#asterisk fujin_ (n=aj@unaffiliated/fujin) |
| 05:16.04 | *** join/#asterisk mvanbaak (n=mafkees@vanbaak.xs4all.nl) |
| 05:16.20 | *** part/#asterisk johnadsfsdfdf (n=john@cpe-72-226-69-65.nycap.res.rr.com) |
| 05:19.16 | *** join/#asterisk fujin (n=aj@unaffiliated/fujin) |
| 05:36.24 | pkunkra | you know, it really seems to be the case that every voip reseller just sucks. |
| 05:36.44 | pkunkra | i think i'm starting to get really jaded. |
| 05:38.52 | *** part/#asterisk workaphobia (n=workapho@magneton-35.dynamic.rpi.edu) |
| 05:39.01 | WilliamK | so what's the newest complaint? |
| 05:39.31 | pkunkra | the voip provider is screwing up the tones when the call goes to voice mail |
| 05:39.53 | pkunkra | turns into this loud screech |
| 05:40.15 | pkunkra | i hear my callers saying "ouch" as their first word. |
| 05:40.43 | pkunkra | you know "you have reached xyz. leave a message. beep." |
| 05:40.48 | pkunkra | they screw up the beep. |
| 05:41.58 | pkunkra | maybe i need to tell them about "dtmfmode=rfc2833" |
| 05:42.00 | pkunkra | :-) |
| 05:42.12 | *** join/#asterisk rat1101 (n=rat1101@pool-71-163-254-224.washdc.fios.verizon.net) |
| 05:43.13 | *** join/#asterisk jarod14 (n=jarod14@mail.viatelecom.com) |
| 05:43.33 | *** join/#asterisk workaphobia (n=workapho@magneton-35.dynamic.rpi.edu) |
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| 05:44.28 | *** part/#asterisk workaphobia (n=workapho@magneton-35.dynamic.rpi.edu) |
| 05:44.28 | pkunkra | that probably isn't the issue though |
| 05:45.46 | *** join/#asterisk henkoegema (n=henkoege@d54C552E4.access.telenet.be) |
| 05:46.01 | henkoegema | <PROTECTED> |
| 05:54.08 | *** join/#asterisk WilliamK (n=noc@static-71-170-144-3.dllstx.fios.verizon.net) |
| 05:54.16 | WilliamK | power hits... |
| 06:05.39 | *** join/#asterisk blinkbot2k (n=me@c-71-230-47-244.hsd1.pa.comcast.net) |
| 06:15.50 | *** join/#asterisk shtoom (n=shtoom@59.93.120.20) |
| 06:18.24 | *** join/#asterisk dominic1 (n=dob@213.221.82.242) |
| 06:18.37 | *** part/#asterisk dominic1 (n=dob@213.221.82.242) |
| 06:21.07 | awk | hmm, in older releases of asterisk the CLI had a dial command, how could I issue a dial command from the CLI now? |
| 06:22.48 | *** join/#asterisk syneus (n=syneus@syneus.aemcom.net) |
| 06:30.47 | *** join/#asterisk Daejeo1 (n=chatzill@211.177.189.25) |
| 06:31.33 | Daejeo1 | anyone have sip firmware 8.6/8.7 for cp 7961g? |
| 06:36.21 | mvanbaak | Daejeo1: you need to have a smartnet account for that |
| 06:36.30 | mvanbaak | then you can download it from cisco website |
| 06:36.34 | Daejeo1 | i know |
| 06:36.38 | Daejeo1 | but |
| 06:36.43 | Daejeo1 | i can't pay] |
| 06:37.36 | *** join/#asterisk bird_of_Luck (n=melifaro@secured.by.ipfw.ru) |
| 06:40.47 | *** join/#asterisk fordfrog (n=fordfrog@gentoo/developer/fordfrog) |
| 06:42.33 | *** join/#asterisk DaveCanoe (n=Dave@canoe404.dclg.ca) |
| 06:43.17 | *** part/#asterisk rat1101 (n=rat1101@pool-71-163-254-224.washdc.fios.verizon.net) |
| 06:44.51 | pkunkra | ah. i finally know why the music is screwed up everytime i'm on hold at another company's pbx |
| 06:45.16 | pkunkra | i was like "geez, if they're gonna put me on hold, at least make the music sound ok" |
| 06:45.29 | mvanbaak | they were playing /dev/urandom to you ? |
| 06:45.48 | pkunkra | yeah, /dev/urandom might have sounded better. |
| 06:46.00 | pkunkra | i should play that for my callers. |
| 06:46.15 | pkunkra | it might make the charts |
| 06:46.22 | mvanbaak | who knows |
| 06:46.38 | mvanbaak | at least it's royalty free |
| 06:46.50 | pkunkra | that's true. |
| 06:47.05 | pkunkra | might keep the phone lines clear too. |
| 06:49.26 | *** join/#asterisk yassaccan (n=yassacca@admin186.hgo.se) |
| 06:49.30 | *** join/#asterisk saftsack (n=oliver@p54A7D2B0.dip.t-dialin.net) |
| 06:51.18 | *** join/#asterisk af_ (n=getsmart@81-174-47-128.dynamic.ngi.it) |
| 06:53.47 | *** join/#asterisk zeeesh (i=zeeesh@202.166.161.45) |
| 06:54.14 | pkunkra | god, my phones sound worse. |
| 06:54.26 | pkunkra | gotta fix that codec. |
| 06:54.32 | pkunkra | its really not meant for music |
| 06:55.18 | pkunkra | forget mp3's if gsm can't handle the bundled moh files. |
| 06:55.35 | *** join/#asterisk tzafrir_laptop (n=tzafrir@62.90.10.53) |
| 06:55.35 | pkunkra | acutally |
| 06:55.45 | pkunkra | no, its getting encoded twice. |
| 06:56.05 | *** join/#asterisk asterisknerdscom (n=logger@66.7.122.93) |
| 06:56.05 | JT | funnily enough, voice codecs aren't made for music |
| 06:56.19 | asterisknerdscom | <PROTECTED> |
| 06:56.26 | *** join/#asterisk kiscokid (n=Ron_Laut@adsl-216-101-109-187.dsl.snfc21.pacbell.net) |
| 06:56.39 | *** join/#asterisk FlatFoot (n=simon@80.88.192.83) |
| 06:56.43 | awk | grrr |
| 06:56.48 | awk | I have a simens routing through us |
| 06:56.57 | awk | right, but its not passing ditis for some reason |
| 06:57.29 | awk | any ideas? |
| 06:57.31 | awk | digits |
| 06:58.36 | *** join/#asterisk r0d3nt (i=nobody@punk.valuetel.net) |
| 07:01.47 | *** join/#asterisk asterisknerdscom (n=logger@66.7.122.93) |
| 07:02.01 | asterisknerdscom | <PROTECTED> |
| 07:02.21 | FlatFoot | good morning all |
| 07:08.06 | *** join/#asterisk asterisknerdscom (n=logger@66.7.122.93) |
| 07:08.20 | asterisknerdscom | <PROTECTED> |
| 07:09.31 | awk | if I do an intense debug I can see it sending through 1 digit at a time |
| 07:09.38 | awk | doesnt seem to be passing the whole string |
| 07:12.03 | awk | < Called Number (len= 4) [ Ext: 1 TON: Unknown Number Type (0) NPI: ISDN/Telephony Numbering Plan (E.164/E.163) (1) '3' ] |
| 07:12.03 | awk | Sending Receiver Ready (44) |
| 07:12.06 | awk | something like this |
| 07:12.26 | awk | and where you see the 3 the next digit is like 7, etc etc |
| 07:12.34 | awk | but I cant work out this unknwon number type |
| 07:12.57 | JT | unknown is good |
| 07:13.02 | JT | as opposed to national, etc |
| 07:13.22 | awk | thing is it was working |
| 07:13.43 | *** join/#asterisk asterisknerdscom (n=logger@66.7.122.93) |
| 07:13.47 | *** join/#asterisk KDan (n=KDan@87-194-122-30.bethere.co.uk) |
| 07:13.57 | asterisknerdscom | <PROTECTED> |
| 07:14.21 | *** join/#asterisk kkn088 (n=kikoun@84.4.216.243) |
| 07:16.08 | *** join/#asterisk KDan (n=KDan@87-194-122-30.bethere.co.uk) |
| 07:16.30 | *** join/#asterisk steliosk (n=Stelios@ipa226.211.tellas.gr) |
| 07:20.21 | *** join/#asterisk asterisknerdscom (n=logger@66.7.122.93) |
| 07:20.35 | asterisknerdscom | <PROTECTED> |
| 07:22.09 | *** join/#asterisk saftsack (n=oliver@p54A7C789.dip.t-dialin.net) |
| 07:27.26 | *** join/#asterisk asterisknerdscom (n=logger@66.7.122.93) |
| 07:31.33 | *** join/#asterisk psk (n=psk@golia.caltanet.it) |
| 07:32.48 | *** join/#asterisk asterisknerdscom (n=logger@66.7.122.93) |
| 07:35.00 | tzafrir_laptop | isn't the channel already logged elsewhere? |
| 07:35.03 | tzafrir_laptop | ~log |
| 07:35.03 | jbot | extra, extra, read all about it, log is as piece of wood, or a record, or the opposite of exponentiation |
| 07:35.15 | *** join/#asterisk Igbothom_III (n=Hilton@office.quarkgroup.com.au) |
| 07:35.34 | tzafrir_laptop | ~logs |
| 07:35.35 | jbot | hmm... logs is apt/ibot/infobot/jbot/purl all log daily to http://ibot.rikers.org/<channelname>/ where channelname is html encoded ie: %23debian | lines that start with a space are not shown | some channels have stats at http://ibot.rikers.org/stats/<channelname>.html.gz |
| 07:37.21 | tzafrir_laptop | http://ibot.rikers.org/%23asterisk/ |
| 07:37.32 | tzafrir_laptop | wow, I didn't know that |
| 07:38.14 | *** join/#asterisk asterisknerdscom (n=logger@66.7.122.93) |
| 07:42.01 | *** join/#asterisk |YonahW| (n=kvirc@84.229.151.80) |
| 07:42.35 | Renacor | asterisk's sip protocol is 2.0 right? |
| 07:43.44 | *** join/#asterisk asterisknerdscom (n=logger@66.7.122.93) |
| 07:44.49 | *** join/#asterisk Igbothom_3rd (n=Hilton@office.quarkgroup.com.au) |
| 07:48.01 | Nugget | On intel machines sometimes it's sip 2.0000000000012 |
| 07:48.58 | |YonahW| | I have a snom 300 which was registering via sip on asterisk fine until the power went out, now I can see the registration requests hitting the asterisk box but no response and it would seem that asterisk does not get the request |
| 07:49.10 | *** join/#asterisk asterisknerdscom (n=logger@66.7.122.93) |
| 07:49.11 | |YonahW| | anyone know have any tips on figuring this out? |
| 07:49.24 | asterisknerdscom | <PROTECTED> |
| 07:50.43 | *** join/#asterisk Igbothom_3rd (n=Hilton@office.quarkgroup.com.au) |
| 07:51.01 | |YonahW| | My iptables allow all udp so I dont think the problem is there |
| 07:55.06 | tzafrir_laptop | asterisknerdscom, no need to announce it every two seconds |
| 07:55.30 | tzafrir_laptop | put it in a nice factoid and/or in the channel's topic |
| 07:55.55 | |YonahW| | tzafrir_laptop: how ya doing? |
| 07:55.56 | tzafrir_laptop | I'm not sure if any of the channel's admins are awake at this time of day |
| 07:56.09 | tzafrir_laptop | |YonahW|, very well |
| 07:56.33 | tzafrir_laptop | asterisknerdscom, you know how to edit jbot's factoids? |
| 07:57.12 | *** join/#asterisk Alexus265 (n=alexus@gw.vdel.ru) |
| 07:57.13 | tzafrir_laptop | jbot, extralogs are somewhere under http://www.asterisknerds.com |
| 07:57.13 | jbot | tzafrir_laptop: okay |
| 07:57.25 | tzafrir_laptop | ~extralogs |
| 07:57.25 | jbot | extralogs are somewhere under http://www.asterisknerds.com |
| 07:59.31 | tzafrir | jbot, no, extralogs are at http://www.asterisknerds.com |
| 07:59.32 | jbot | okay, tzafrir |
| 07:59.36 | Alexus265 | Hi all. How can I set timeout for agents in a queue while using roundrobin? timeout = [sec] in queue scope gives me timeout for the whole queue. Thanx. |
| 08:01.26 | |YonahW| | Alexus265: are you looking to set a timeout on the dial? on the agent registration? |
| 08:03.09 | Alexus265 | |YonahW|: I want call to be passed to another agent if current one didn't answer for a certain time. |
| 08:03.34 | tzafrir | asterisknerdscom, http://www.asterisknerds.com/cgi/irclogger_log/asterisk?date=2007-09-05,Wed gives me an error for "Redirection loop" |
| 08:04.16 | *** join/#asterisk [Xwire] (i=Administ@74.210.19.215) |
| 08:04.24 | [Xwire] | phone broke |
| 08:06.51 | [Xwire] | PHONE BROKE! |
| 08:07.32 | *** join/#asterisk lowlevel (n=Stuart@CPE0017f2e2a3e8-CM000f9f7d6742.cpe.net.cable.rogers.com) |
| 08:07.57 | [Xwire] | would it be possible to use this with teh new blackberry |
| 08:08.07 | [Xwire] | ? |
| 08:09.02 | kaldemar | hello, i'm getting "Module '<module>' did not register itself during load" with asterisk 1.4.11. i compiled the modules myself and the moduledir is right. would anyone happen to know any other reason for that except the modules being of wrong version? |
| 08:09.39 | [Xwire] | no |
| 08:10.24 | |YonahW| | Alexus265: I thought that was what the timeout option was for |
| 08:10.25 | *** join/#asterisk Archssm (n=tommy@85.19.215.250) |
| 08:10.58 | |YonahW| | is it possible that your whole queue is timing out at the same time because there are no other agents logged in to that queue? |
| 08:11.47 | Archssm | Is there a way for GXP2000 phones to automatically register as an agent? I cannot seem to find any pause function with the Grandstream unit. |
| 08:12.29 | [Xwire] | does anybody know if it would be possible to use this system with the new blackberry UMA phones |
| 08:13.16 | Archssm | [Xwire] : 'This system' ? |
| 08:13.38 | [Xwire] | well asterix |
| 08:13.44 | [Xwire] | well asterisk |
| 08:14.16 | Archssm | Sure. The blackberries support SIP, right? |
| 08:14.29 | [Xwire] | we are looking to replace a nortel box |
| 08:14.40 | [Xwire] | unfortunatally not, it' |
| 08:14.45 | [Xwire] | it's UMA |
| 08:15.26 | [Xwire] | they have a model that does SIP but it has no cellular capabilities |
| 08:16.16 | Archssm | Typical... |
| 08:16.22 | Archssm | Let me check it out. |
| 08:16.29 | [Xwire] | yea they are provider driven. |
| 08:16.52 | [Xwire] | well what we really want is a seamless handover between the wifi and cell network |
| 08:18.18 | [Xwire] | we will likely be using aruba mobility controller and access points |
| 08:18.46 | Alexus265 | |YonahW|: queue exits after [timeout] seconds, i have several members in a queue with different penalty and using roundrobin strategy. When the call is passed to a member, it's ringing for several minutes without being passed to another. If [timeout] expires, queue exits, and if even i call Queue again, call is passed to the same member, because there is only one member with the least penalty. |
| 08:20.14 | *** join/#asterisk qdk (n=qdk@213.150.62.32) |
| 08:23.42 | Alexus265 | |YonahW|: I mistakenly mentioned agents, I use only members (local SIP and external numbers via member => Zap/g2/<my_mobile> as well) |
| 08:24.00 | *** join/#asterisk asterisknerdscom (n=logger@66.7.122.93) |
| 08:25.17 | |YonahW| | Alexus265: sorry I have never used queues like that |
| 08:25.33 | *** join/#asterisk Strom_M (n=strom@208.127.172.112) |
| 08:26.20 | |YonahW| | Alexus265: did you see my sip registration problem above? |
| 08:26.49 | Alexus265 | |YonahW|: no, I've recently joined |
| 08:28.19 | |YonahW| | I have a snom 300 which was registering fine with asterisk until the power went out now it wont register but it seems to me like the problem in on asterisk's end |
| 08:28.53 | |YonahW| | i can see the registration request packets hitting the asterisk box but it would seem that asterisk is not receiving them and i dont see anything wrong with my iptables |
| 08:29.26 | *** join/#asterisk asterisknerdscom (n=logger@66.7.122.93) |
| 08:29.31 | *** join/#asterisk Formater (i=Formater@dial-111.041net.co.yu) |
| 08:29.33 | Formater | hi |
| 08:31.09 | Formater | asterisk 1.2.x, clients are stored in db (sip_buddies). is there a way to see from astersik console if there are registered or not? |
| 08:34.52 | *** join/#asterisk asterisknerdscom (n=logger@66.7.122.93) |
| 08:37.39 | *** join/#asterisk kiscokid (n=Ron_Laut@adsl-216-101-109-187.dsl.snfc21.pacbell.net) |
| 08:38.42 | Alexus265 | |YonahW|: Is asterisk on udp:5060? I don't remember whether snom 300 is tftp capable, does it recieve tftp parameters via dhcp and fetch needed files in case it is? Can other clients register with this asterisk? |
| 08:39.16 | *** join/#asterisk Igbothom_III (n=Hilton@office.quarkgroup.com.au) |
| 08:40.14 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 08:40.19 | *** join/#asterisk asterisknerdscom (n=logger@66.7.122.93) |
| 08:40.24 | Nichtwirklich | Formater: sip show peers ? |
| 08:40.41 | *** part/#asterisk kiscokid (n=Ron_Laut@adsl-216-101-109-187.dsl.snfc21.pacbell.net) |
| 08:44.02 | *** mode/#asterisk [+b *!n=logger@66.7.122.93] by Corydon76-dig |
| 08:44.45 | |YonahW| | Alexus265: asterisk is on udp:5060. the snom is tftp capable however i am not utilizing that, and the settings look fine. other clients are currently registered to this asterisk also snom 300s |
| 08:45.14 | [Xwire] | maybe someone can answer this.. why would someone use asterisk over say a shoretel or nortel box |
| 08:45.35 | Nugget | flexibility. |
| 08:45.38 | *** join/#asterisk WildPikachu (n=WildPika@about/linux/staff/wildpikachu) |
| 08:45.51 | Formater | Nichtwirklich: that shows only clients from sip.conf, and not shows the users from sip_buddies. |
| 08:45.55 | WildPikachu | i wonder if its possible to get my grandstream gxp2000 to show if a call is an internal transfer before I answer it? |
| 08:46.31 | WildPikachu | at present it shows "asterisk\nasterisk" |
| 08:46.46 | Nugget | WildPikachu: so make it show something else. |
| 08:47.10 | WildPikachu | i can set the caller id to the extension dialing me, is that right? |
| 08:47.21 | Nugget | you can set the callerid to anything you want |
| 08:47.51 | WildPikachu | *sigh* |
| 08:50.44 | *** join/#asterisk Pazzo (n=ugelt@reserved-225136.rol.raiffeisen.net) |
| 08:54.36 | *** join/#asterisk puzzled (n=patrick@puzzled.xs4all.nl) |
| 08:55.58 | puzzled | hi |
| 08:56.31 | puzzled | tzafrir: ping |
| 08:56.36 | tzafrir | pong |
| 08:57.37 | Formater | sip show peers shows only clients from sip.conf, and not shows the users from sip_buddies. is there a way to see which users are regsitered from sip_buddies too? |
| 08:57.52 | puzzled | tzafrir: morning. A guy I know has a debian box with a junghanns 4bri card and last century asterisk release. do you have a repo with a recent asterisk release? possibly with bristuff built-in? |
| 08:58.36 | tzafrir | standard debian packages |
| 08:59.19 | tzafrir | backports of recent asterisk is available from http://buildserver.net/ , but this is a bleeding-edge backport from Unstable. So requires more testing |
| 09:00.02 | puzzled | tzafrir: I see. he mentioned the box runs 2.6.18-4-686. Is that an old debian version for which there possibly isn't a recent asterisk release available? |
| 09:00.58 | tzafrir | It's the current Etch (Stable) version. |
| 09:01.03 | tzafrir | Well supported |
| 09:01.22 | puzzled | ok good. and Etch has 1.2.24 debs? |
| 09:05.16 | WildPikachu | hrmmmmmm |
| 09:05.34 | WildPikachu | i spose i should be getting callerid on my grandstream when someone transfers a call to me |
| 09:06.38 | puzzled | WildPikachu: if it is being set in the first place then I spose so yes. I have seen callerid being displayed on their old models |
| 09:07.13 | WildPikachu | see ... when i get a call transferred to me by one of my staff who answers a call in the queue, I get "asterisk" displayed as the caller id |
| 09:07.22 | WildPikachu | callerid is not set in the first place on inbound calls as my telecom is dumb |
| 09:07.41 | puzzled | then kick those idiots that they need to pass clid :) |
| 09:07.45 | Nugget | so set callerid. I thought we went over this 10 minutes ago. |
| 09:08.09 | WildPikachu | just trying to make sure i'll do the right thing here .... |
| 09:08.09 | Nugget | "asterisk" showing up means you're either not setting it, or you're trying to set it to something asterisk can't parse. |
| 09:08.19 | WildPikachu | aha, there we go :) |
| 09:08.21 | Nugget | You don't have to be a rocket surgeon. |
| 09:08.30 | WildPikachu | lol, rocket surgeon |
| 09:08.39 | WildPikachu | biomechanical rockets? |
| 09:08.44 | puzzled | but you do have to be a brain scientist |
| 09:08.47 | |YonahW| | do rockets actually require surgeons? |
| 09:09.12 | puzzled | prolly the borg vessel needs a few |
| 09:09.29 | WildPikachu | so the caller id on a transfer ... should be the person who's doing the transfer to me? which i must set in my dialpna? (just making sure) |
| 09:09.30 | [Xwire] | can anyone tell me how complete the GUI is? |
| 09:09.38 | WildPikachu | *dialplan |
| 09:09.48 | Nugget | WildPikachu: are you setting caller id for the extensions themselves (in sip.conf)? |
| 09:09.53 | Nugget | that *ought* to do it |
| 09:10.17 | WildPikachu | yes .... callerid=1001 callerid=1002 ... etc |
| 09:10.40 | WildPikachu | when i get a transferred call ringing by me with attended transfer, i still get asterisk thats my prob that i'm trying to resolve |
| 09:10.43 | Nugget | that's setting callerid name but not number, I think. |
| 09:10.52 | WildPikachu | aha |
| 09:10.57 | Nugget | I'd expect to see "callerid=WildPikachu <1001>" |
| 09:11.12 | WildPikachu | aha! ... so asterisk can't parse my 1001 |
| 09:11.45 | Nugget | you should read the example configs in the sip.conf file. |
| 09:12.07 | *** join/#asterisk zeedo (n=zeedo@reboot-robot.net) |
| 09:12.20 | [Xwire] | si the GUI for asterisk any good? |
| 09:12.27 | WildPikachu | thats all you needed to do, point me at the docs :) ... but then again i was dumb |
| 09:12.36 | *** part/#asterisk shtoom (n=shtoom@59.93.120.20) |
| 09:13.14 | Nugget | [Xwire]: the asterisk gui is adequate for delegating administrative tasks to other people. if you're expecting that the gui means you won't actually have to read the docs or understand the config files you will be really unhappy. |
| 09:13.49 | [Xwire] | thank you! ... one more |
| 09:13.50 | puzzled | [Xwire]: it's still in beta. join #asterisk-gui and ask there |
| 09:13.52 | WildPikachu | ok, on inbound calls entering the queue, i can set callerid=Support <0> ... that should then show up when the queue rings my phone if i'm in the queue? |
| 09:14.13 | [Xwire] | how difficult is it to upgrade version to version |
| 09:14.26 | Nugget | WildPikachu: yes, although id you're going to set the callerid in the dial plan the syntax is different (as you'd expect) |
| 09:14.46 | WildPikachu | excellent, /me goes to read more |
| 09:15.40 | [Xwire] | the phone systems i've used you generally flash a rom, but this is different |
| 09:15.57 | [Xwire] | does it break between versions |
| 09:17.24 | Nugget | define "break" |
| 09:17.28 | *** join/#asterisk KermitTheFragger (n=siepkes@118-197.bbned.dsl.internl.net) |
| 09:17.48 | Nugget | often upgrades involve changes (documented in the ChangeLog) which will cause some degree of breakage if you aren't paying attention. |
| 09:17.59 | [Xwire] | well for example, on talkswitch it migrates the config between versions |
| 09:18.00 | Nugget | that might be a minor thing or a major thing |
| 09:18.18 | Nugget | there's no such automated migration in asterisk |
| 09:18.50 | [Xwire] | so it could mean major down time |
| 09:18.58 | Nugget | that's probably an overstatement. |
| 09:19.10 | Nugget | first off, it never requires downtime. |
| 09:19.14 | Nugget | just planning |
| 09:19.30 | Nugget | and secondly, the changes are normally minimal from a configuration/local code standpoint |
| 09:19.58 | Nugget | if you choose to upgrade without reading the documentation or changelogs then, sure, you should expect downtime and problems. |
| 09:20.11 | [Xwire] | well i am assuming that the new version means new config files, then all the settings have got to be put in the new files |
| 09:20.40 | Nugget | that's probably not a safe assumption, and the phrasing itself sort of indicates an unfamiliarity with asterisk |
| 09:20.51 | *** join/#asterisk steliosk (n=Stelios@ipa226.211.tellas.gr) |
| 09:21.21 | [Xwire] | i have never used it, i am considering it as an option to shoretell or nortel |
| 09:21.24 | Nugget | asterisk's "config files" are way more complicated than I'm suspecting you imagine them to be, and the notion of "all new" is not really valid. |
| 09:21.49 | Nugget | asterisk is software, it's not an appliance like your nortel or shoretell solutions are |
| 09:22.04 | Nugget | the dialplan is more code than it is a config file. |
| 09:22.10 | [Xwire] | well there is little difference |
| 09:22.35 | Nugget | says the guy who has never used it? :) |
| 09:22.42 | *** join/#asterisk darkskiez (n=mbryars@195-11-205-216.suip.mezzonet.net) |
| 09:22.58 | [Xwire] | i mean between an appliance and software |
| 09:23.03 | Nugget | yes, I mean that too. |
| 09:23.52 | [Xwire] | well my main concern is the migration path |
| 09:24.19 | [Xwire] | and upgrades |
| 09:24.46 | Nugget | I think you're wise to be concerned about both of those things. They represent the areas where asterisk is least mature, for sure. |
| 09:24.52 | [Xwire] | well like a barrcudda for example it runs a full linux distro, but the upgrades are inplace and seamless |
| 09:25.19 | Nugget | asterisk is several years away from that level of abstraction |
| 09:25.44 | Nugget | the benefit is that asterisk is considerably more flexible if you have a reason to put in the effort |
| 09:26.31 | [Xwire] | yes, and we do, we want to get blackberries to roam between cell and and wifi |
| 09:26.42 | [Xwire] | and if anything will be able to it is this |
| 09:26.53 | [Xwire] | because nothing else can at present |
| 09:26.57 | Nugget | it's that sort of application where asterisk is really worhtwhile |
| 09:28.01 | Formater | rtcachefriends = yes|no : Cache realtime friends by adding them to the internal list just like friends added from the config file. I changed this to yes, and still i do not see users from sip_buddies with sip show peers :( |
| 09:28.20 | *** join/#asterisk syneus (n=syneus@syneus.aemcom.net) |
| 09:28.24 | [Xwire] | thanks nugget! you have been very helpfull |
| 09:28.28 | Nugget | happy to help |
| 09:44.40 | *** join/#asterisk stkn|work (n=stkn|wor@gentoo/developer/pdpc.active.stkn) |
| 09:44.43 | WildPikachu | Nugget, thanks, it worked |
| 09:44.59 | WildPikachu | i assume the callerid number can be text aswell? |
| 09:48.47 | Renacor | anybody know what could possibly cause this Unable to create channel of type 'SIP' (cause 3 - No route to destination) |
| 09:53.51 | WildPikachu | Renacor, i got that when the other phone was off or not registered |
| 09:54.10 | Renacor | it is registered |
| 09:54.13 | Renacor | however look at this |
| 09:54.14 | Renacor | http://pastebin.ca/682058 |
| 09:54.33 | Renacor | also getting channel.c:804 channel_find_locked: Avoided initial deadlock for '0xb6300d30', 9 retries! right before it |
| 09:55.29 | kaldemar | what does your sip.conf, 'sip show peers' and extensions.conf (the dial line) look like? |
| 09:57.55 | Renacor | one sec |
| 09:58.31 | Renacor | don't have sip show peers or anything like that in sip.conf |
| 09:58.58 | puzzled | Renacor: it's a command you type in on the asterisk command line |
| 09:58.59 | kaldemar | sip show peers is a cli command. |
| 09:59.05 | Renacor | oh |
| 09:59.27 | Renacor | host shows as unspecified |
| 09:59.32 | Renacor | interesting |
| 09:59.57 | kaldemar | what made you think the endpoint is registered? |
| 10:00.01 | Renacor | the ones that work show ip addresses |
| 10:00.10 | Renacor | cause I can get to their web interface |
| 10:00.13 | Renacor | and I can call out on them |
| 10:00.40 | kaldemar | getting to a phone's web page has nothing to do with it being registered to asterisk. |
| 10:00.53 | Renacor | right but being able to call out |
| 10:01.04 | Renacor | from the phoneI would assume u are registered |
| 10:01.33 | kaldemar | a phone doesn't have to be registered for you to be able to dial out from it. |
| 10:02.01 | kaldemar | it just needs the right authentication parameters and asterisk's ip address. |
| 10:02.02 | Renacor | k how would it be registered |
| 10:03.30 | kaldemar | i don't know how to use a phone i don't know. |
| 10:04.11 | Renacor | nm kaldemar i think I figured it out, thanks for the clarification though |
| 10:04.51 | *** join/#asterisk engrxyz (i=1000@host81-150-247-250.in-addr.btopenworld.com) |
| 10:05.57 | Renacor | k i keep getting this though: Sep 5 15:10:36 WARNING[8391]: channel.c:804 channel_find_locked: Avoided initial deadlock for '0xb6300d30', 9 retries! |
| 10:06.03 | Renacor | is that something to worry about? |
| 10:07.53 | n0n4m3 | anyone of you guys ever configured vood 322 with asterisk successfuly? |
| 10:07.55 | n0n4m3 | ll |
| 10:14.05 | Alexus265 | |YonahW|: can't imagine why this can happen, sorry. Didn't you solve this yet? |
| 10:14.49 | *** join/#asterisk BrokenNoze (n=root@62.253.194.107) |
| 10:15.37 | BrokenNoze | Hi, Does anyone know what seqno 1 (Critical Response) means? my phone works fine for first 5 minutes or so then I start getting this on the console? any ideas? |
| 10:16.34 | *** join/#asterisk sergee (n=serg@voip1.west-call.com) |
| 10:17.19 | *** join/#asterisk codefreeze (n=steve_mu@216.166.159.235) |
| 10:17.19 | *** mode/#asterisk [+o codefreeze] by ChanServ |
| 10:17.29 | *** join/#asterisk MrMister2 (n=mrmister@195-23-105-234.net.novis.pt) |
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| 10:23.33 | MrMister2 | Hi. I'm trying to get a vanilla * to work. I'm having trouble sending a call from a trunk to a extension where it plays a message and hangsup. I can get it to pickup and hangup but the message doesn't play. According to some people I may have my sip.conf badly configured. Can anyone give a hand? |
| 10:24.33 | MrMister2 | http://pastebin.ca/682087 |
| 10:28.53 | n0n4m3 | are there any voip2gsm modules? |
| 10:29.29 | n0n4m3 | like vood322 is for voip2pots and patton smart-dta for voip2isdn |
| 10:30.21 | puzzled | n0n4m3: look at beronet.com or junghanns.net |
| 10:30.38 | n0n4m3 | thanks |
| 10:31.11 | n0n4m3 | hope the products are compatible with asterisk |
| 10:33.49 | *** join/#asterisk cheGGo (n=snafu__@gate.goobernetworks.com) |
| 10:33.59 | cheGGo | hi there |
| 10:43.47 | *** join/#asterisk folkob (n=c1c94244@212.152.60.24) |
| 10:43.57 | folkob | hello |
| 10:45.05 | *** join/#asterisk tsurko (n=tsurko@213.91.216.130) |
| 10:45.14 | cheGGo | hi folkob |
| 10:46.26 | folkob | i've got problem connecting * with CCM using ooh323 , with cisco IP phones connected to CCM all work perfectly , but if call number wich belons to PBX after two rings i hear bisy tone and error 'every one bisy\concested this time' |
| 10:47.00 | folkob | but if called person picj up phone immideatly after ring all work nice |
| 10:47.20 | folkob | could it be * problem or CCM problem ? |
| 10:49.25 | cheGGo | which ringtime parameter r u using in your DIAL command? |
| 10:51.38 | folkob | don't use it ... now dial string look like : Dial,ooh323/${EXTEN}@IP |
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| 10:58.36 | henkoegema | <PROTECTED> |
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| 11:17.08 | henkoegema | <PROTECTED> |
| 11:17.39 | cheGGo | whazzup henkoegema? |
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| 12:11.15 | *** join/#asterisk whywontitwork (n=d@196.211.34.2) |
| 12:16.11 | whywontitwork | here is my problem: i am receiving faxes using asterisk (works Fine) ever so often someone phones the fax number by mistake and dont send a fax, asterisk however still answers the call as a fax then mails me, with no attachemnt(I know because no fax was received) how can i stop this from happening? |
| 12:22.38 | shido6 | nice nick |
| 12:22.39 | whywontitwork | here is my problem: i am receiving faxes using asterisk (works Fine) ever so often someone phones the fax number by mistake and dont send a fax, asterisk however still answers the call as a fax then mails me, with no attachemnt(I know because no fax was received) how can i stop this from happening? |
| 12:22.47 | whywontitwork | thx |
| 12:22.48 | shido6 | pastebin your dialplan |
| 12:23.16 | whywontitwork | where you not allowed to paste in channel |
| 12:23.51 | shido6 | pastebin.ca |
| 12:26.04 | *** join/#asterisk Deeewayne (i=dwayne@nat/digium/x-c21fcd569c105280) |
| 12:26.04 | *** mode/#asterisk [+o Deeewayne] by ChanServ |
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| 12:29.32 | cheGGo | hi Deeewayne :) |
| 12:30.36 | whywontitwork | thxhttp://pastebin.com/d6b7105c5 |
| 12:31.33 | ManxPower | try checking if the file exists before e-mailing it. |
| 12:31.44 | ManxPower | also, do you delete the on-disk file after e-mailing it? |
| 12:31.58 | whywontitwork | that the problem |
| 12:32.42 | whywontitwork | there is no fax file, people dial the number then no fax is received how do you tel asterisk to stop mailing it if there is no file? |
| 12:33.43 | ManxPower | whywontitwork: what you are doing in your dialplan is ALWAYS on EVERY call e-mail a file. |
| 12:34.07 | whywontitwork | sorry that makes no sence; here goes somewhere you must be able to tell asterisk if no fax is detected hangup |
| 12:34.44 | ManxPower | whywontitwork: Um, you are not even trying to detect a fax, you are blindly running rxfax for all calls that come into the number. |
| 12:35.04 | *** join/#asterisk tcseke (n=chatzill@22-36.adsl.etel.hu) |
| 12:35.04 | whywontitwork | no only for if ${EXTEN} = 2550,2551,2552,2553,2554 |
| 12:35.13 | cheGGo | thats what he said |
| 12:35.28 | cheGGo | but u have to check if that incoming call is really a fax |
| 12:35.42 | whywontitwork | if the above aplies it must be a fax for it is our fax numbers |
| 12:35.46 | ManxPower | I suggest that you replace the "email" program with a shell script that tests to see if it should even send a message |
| 12:35.51 | cheGGo | thats not right |
| 12:35.52 | whywontitwork | how does one detect fax? |
| 12:36.05 | cheGGo | u can call a fax from a normal phone |
| 12:36.17 | whywontitwork | yes i know |
| 12:36.23 | ManxPower | whywontitwork: in your case, because you have dedicated fax numbers, I would not bother try to detect faxes. |
| 12:36.34 | whywontitwork | is the asterisk command to detect fax? |
| 12:36.36 | ManxPower | THIS IS NOT AN ASTERISK ISSUE. |
| 12:36.47 | cheGGo | hehe |
| 12:36.52 | cheGGo | calm down ;) |
| 12:37.39 | cheGGo | ManxPower, do you ever used reinvites? |
| 12:37.43 | whywontitwork | k k k k k i will google it some more, keep yo |
| 12:37.51 | ManxPower | cheGGo: yes, but never with NAT. |
| 12:38.20 | sheppard | If i have a phone line I want to use to call out on using asterisk, I need a fxo card right? |
| 12:38.32 | cheGGo | ok, no problem, may u can tell me if this behaviour of asterisk is normal |
| 12:38.41 | *** join/#asterisk [TK]D-Fender (n=joe@MTRLPQ02-1177745839.sdsl.bell.ca) |
| 12:38.45 | ManxPower | sheppard: correct |
| 12:38.56 | cheGGo | i do a callback via callfiles, and send reinvites after the call is bridged |
| 12:39.07 | cheGGo | that works fine, and its exactly what i want |
| 12:39.24 | *** join/#asterisk saftsack (n=saftsack@pD9E07188.dip.t-dialin.net) |
| 12:39.26 | sheppard | ManxPower: is therea cheap one I can put in a bsd or linux box, or would is there a cheap voip -> fso appliance |
| 12:39.30 | cheGGo | but, if someone of both callers hangup |
| 12:39.42 | ManxPower | sheppard: it's telephony -- nothing is cheap. |
| 12:39.53 | cheGGo | asterisk try to fetch back the rtpstream of that channel who did NOT the hangup |
| 12:40.00 | ManxPower | sheppard: you could use an ATA w/FXO port, but it is a big hassle. |
| 12:40.43 | sheppard | ok any particular card I should use then? |
| 12:40.48 | sheppard | something with the zaptel chipset? |
| 12:41.10 | ManxPower | A T-1 card for a nortel box is about $4000. If you want PRI protocol on that T-1 card it will cost another $4,000. |
| 12:41.19 | ManxPower | Digium and others sell T-1 cards for about $500 |
| 12:41.25 | cheGGo | then, asterisk realized that the call finished, and THEN he send the bye packet for the left opened channel |
| 12:41.39 | ManxPower | in telephony $500 is cheap. |
| 12:41.47 | cheGGo | ManxPower, do you know is this behaviour is normal? |
| 12:42.00 | ManxPower | sheppard: you need to use a zaptel compatable card like Digium or Sangoma |
| 12:42.18 | ManxPower | cheGGo: I don't know. |
| 12:42.21 | cheGGo | :( |
| 12:43.06 | ManxPower | <PROTECTED> |
| 12:43.06 | ManxPower | Not good |
| 12:44.30 | cheGGo | thats nothing :P |
| 12:44.42 | cheGGo | my maximum load was 260.50 :D |
| 12:45.04 | sheppard | wow |
| 12:45.06 | sheppard | that's not cheap |
| 12:45.25 | *** join/#asterisk angler (i=angler@pdpc/sponsor/digium/angler) |
| 12:45.25 | *** mode/#asterisk [+o angler] by ChanServ |
| 12:45.57 | cheGGo | load @ 260.50 u even get ping timeouts on local loopback connections |
| 12:45.58 | cheGGo | :D |
| 12:48.36 | sheppard | ManxPower: whats wrong with http://www.ncix.com/products/index.php?sku=21919&vpn=SPA3102-NA&manufacture=Linksys besides the fact I can't use asterisk |
| 12:49.52 | ManxPower | sheppard: that should work, but all the linksys/sipura products are complicated to make the FXO work the way you want. |
| 12:50.11 | sheppard | ahh k |
| 12:50.21 | sheppard | cause all of the local suppliers i've checked out so far |
| 12:50.31 | sheppard | want $800 or more for a fxo card, and it's all pci-x based |
| 12:50.41 | ManxPower | I'm getting spambombed. |
| 12:51.05 | ManxPower | Huh? Digium TDM400P w/FXO should be well under $200 |
| 12:51.34 | JT | sheppard: i think you mean pci |
| 12:52.07 | ManxPower | sheppard: you need a card that works with Asterisk and that usually means Digium or Sangoma if you want a card |
| 12:52.28 | sheppard | JT: no the cards my two suppliers were listing were t1/e1 cards for the most part and were pci-x based |
| 12:52.53 | JT | sheppard: i see |
| 12:52.58 | JT | perhaps you mean pci-e |
| 12:53.10 | JT | no-one makes pci-x zaptel cards that i know of |
| 12:53.14 | ManxPower | sheppard: WHAT BRAND OF CARD? |
| 12:53.57 | *** part/#asterisk Alexus265 (n=alexus@gw.vdel.ru) |
| 12:54.37 | sheppard | http://www.directdial.com/ca/shop/go/go.asp?new_header_r7_c15.x=0&new_header_r7_c15.y=0&new_header_r7_c15=Search&OrderBy=Mfgr_Code&OBS=on&RL=none&DisplayStyle=BigSpecials&SearchString=sangoma |
| 12:54.45 | sheppard | however |
| 12:54.48 | *** join/#asterisk tr2x (n=alvar@80-218-185-55.dclient.hispeed.ch) |
| 12:54.51 | sheppard | ebay is returning some acceptable results |
| 12:54.57 | *** join/#asterisk mrbond82 (n=bondo@d221-91-164.commercial.cgocable.net) |
| 12:55.51 | ManxPower | sheppard: those are T-1/E-1 cards, not analog cards |
| 12:56.30 | *** join/#asterisk _ShrikE (n=ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 12:57.17 | sheppard | yeah i know |
| 12:57.17 | mrbond82 | If I have my "internal network" defined in 1 context and someone is on one of their lines and I call them from my line, they come up as busy. Why when I get a multiple incoming calls from another context (one that defines external handling) does it not give a busy signal and my phone actually shows them as "on hold" (on that same line), I'd like to get it to behave with a busy signal, is this something to do with the context? |
| 12:57.17 | sheppard | like i said, all my supplier was listing was t1/e1 cards |
| 12:57.17 | sheppard | but ebay is working out nicely |
| 12:57.28 | *** join/#asterisk jmesquita (n=jmesquit@c915230f.virtua.com.br) |
| 12:59.50 | ManxPower | Just beware of the cheap "X100P clone" cards. |
| 12:59.55 | ManxPower | They suck. |
| 12:59.55 | *** join/#asterisk waKKu (n=eurulo@unaffiliated/wakku) |
| 13:00.42 | [TK]D-Fender | Sangoma A200D-X PCI Express Hardware echo canceller <----- Analog |
| 13:01.29 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 13:01.32 | Wonka | [TK]D-Fender: PCIe != PCI-X |
| 13:02.04 | JT | Wonka: the Sangoma -X cards are PCI Express. |
| 13:02.24 | [TK]D-Fender | Wonka: Correct, and I never suggested otherwise. |
| 13:02.25 | Wonka | JT: PCI-X is something different |
| 13:02.47 | JT | Wonka: i know the difference, scroll up. |
| 13:02.53 | Wonka | seen |
| 13:02.57 | JT | i just said they were different a couple of minutes ago. |
| 13:03.00 | Wonka | but sheppard talked about PCI-X... |
| 13:03.07 | *** join/#asterisk geoff_k (n=geoff_k_@host81-152-90-217.range81-152.btcentralplus.com) |
| 13:03.09 | JT | so why would you need to tell ME again? |
| 13:03.41 | [TK]D-Fender | Wonka: Go caffeinate |
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| 13:06.48 | ManxPower | I can't imagine that a person that wants a single analog fxo port would ever need anything other than plain old PCI |
| 13:08.05 | Wonka | PCI has enough bang for about 1GBit/s - that's about 512 E1 lines... |
| 13:08.33 | Wonka | and that's 32Bit at 33MHz |
| 13:10.18 | JT | won't handle the interrupts |
| 13:11.26 | *** join/#asterisk af_ (n=getsmart@81-174-47-128.dynamic.ngi.it) |
| 13:11.43 | [TK]D-Fender | SPA-3102 <--- cheaper, less hassle, more fliexible. End of story. |
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| 13:19.02 | sheppard | ManxPower: I think i'm going to go with your reccomendation on the TDM400p card. I don't see any 1 port versions, so I guess i'll have 3 ports i'll never use |
| 13:19.18 | [TK]D-Fender | sheppard: SPA-3102 <------- |
| 13:19.42 | mrbond82 | does anyone have any clue about my context question? |
| 13:20.13 | sheppard | [TK]D-Fender: apparently those are quesitonable to get configured the way you want |
| 13:20.20 | sheppard | plus I'm going to have an odd setup |
| 13:20.32 | sheppard | my cell -> office wifi -> internet -> my house -> free LD line |
| 13:20.35 | [TK]D-Fender | mrbond82: PASTEBIN is your friend.... |
| 13:20.37 | [TK]D-Fender | ~pb |
| 13:20.38 | jbot | A Pastebin is a place to paste your stuff without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste |
| 13:20.38 | awk | hrm, anyone here any good with perl or python |
| 13:20.40 | [TK]D-Fender | ^^^^^^^^^^^^^^^ |
| 13:20.42 | awk | and have the ability to test some code |
| 13:20.48 | awk | tell me whats wrong? |
| 13:21.02 | awk | i cant work out what im doing wrong |
| 13:21.25 | mrbond82 | My q was too long??? |
| 13:22.39 | [TK]D-Fender | mrbond82: No, I think you should be showing us "sip show peers", your dialplan, and the full CLI output of your failed call at verbose 10 with SIP debug enabled (if using SIP phones) |
| 13:23.02 | [TK]D-Fender | mrbond82: because right now you seem to think we're psychic. |
| 13:24.07 | mrbond82 | sorry I was just thinking there was some silly internal thing regarding contexts and if dialing within a context to an already in use extension will give a busy signal (again, an * internal thing) and if on the line within 1 context, and another context calls the sip phone, it will ring through and let the phone handle it |
| 13:24.20 | mrbond82 | also I thought you guys were pretty sharp |
| 13:24.53 | *** join/#asterisk konqi_ (n=konqi@217.193.163.2) |
| 13:25.58 | [TK]D-Fender | mrbond82: Us pretty sharp? You show us nothing and expect us to know the nitty-gritty of your problem. |
| 13:26.15 | shido6 | psychics |
| 13:26.16 | mrbond82 | like I said, I thought my problem was a common thing because I'm new |
| 13:26.39 | shido6 | asterrisk psi -guru oracle |
| 13:26.43 | [TK]D-Fender | mrbond82: And as for "busy", thats a grey statement as several of the messages * COULD put on CLI may LOOK that way, but not mean what you think it does depending on verbose levels and channel debug. |
| 13:26.51 | MrMister2 | sheppard: I have a TDM400P installed on my * server. It works OK |
| 13:27.09 | *** join/#asterisk santiago (i=santiago@debian/developer/santiago) |
| 13:27.23 | [TK]D-Fender | mrbond82: So lets quit burning karma and get to pastebinning, shall we? |
| 13:27.43 | MrMister2 | Haven't used a SPA-3102 so no idea on what it can do. It could be a better device for you. |
| 13:27.48 | mrbond82 | ~pb ? |
| 13:27.48 | jbot | A Pastebin is a place to paste your stuff without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste |
| 13:27.56 | mrbond82 | too lazy to type |
| 13:27.58 | [TK]D-Fender | YES |
| 13:28.37 | mrbond82 | what files would you like to see ? |
| 13:29.07 | MrMister2 | [TK]D-Fender: A quick question. Is it possible to do a attended transfer and still keep the CID of the original caller? |
| 13:31.15 | [TK]D-Fender | MrMister2: not through most sip phones. * won't know tis a transfer until the phone actually tries to pass off the call. For DTMF based attended transfers you always modify the source..... |
| 13:31.28 | [TK]D-Fender | mrbond82: I jsut gave you a very specific list of things to provide... |
| 13:31.46 | henkoegema | can somebody test my ENUM (http://pastebin.com/d50461f13) ? |
| 13:33.39 | MrMister2 | [TK]D-Fender: I'm using X-Lite to transfer the call. If I do a unattended transfer the CID does get passed to another X-Lite, if I do a attended it doesnt. |
| 13:34.15 | [TK]D-Fender | MrMister2: I just ANSWERED this for you. |
| 13:35.55 | mrbond82 | wow |
| 13:38.30 | whywontitwork | TK how does one detect a fax? |
| 13:38.46 | whywontitwork | in your dialplan that is! |
| 13:39.12 | cybertooth | Text-to-Speech. Anyone have any recommendations? I'm looking at using Festival right now. |
| 13:39.39 | chemikk | <PROTECTED> |
| 13:39.40 | [TK]D-Fender | whywontitwork: lookup the "fax" Standard Extension. |
| 13:39.53 | whywontitwork | thx TK |
| 13:39.55 | [TK]D-Fender | whywontitwork: IIRC it only works on Zap channels. |
| 13:40.09 | whywontitwork | k thx |
| 13:40.19 | BrokenNoze | Hi, asked the question earlier but, anyone know why my phone will work fine on * for 30 minutes, then start failing with a seqno 1 Critical Response error on the server? The only way round it seems to be to change the IP address of the phone? using a polycom 350 'm sure it must be me doing something wrong |
| 13:40.50 | [TK]D-Fender | cybertooth: Festival works, but I'm told is noticably inferior to Cepstral (which is pretty affordable per channel last I heard) |
| 13:41.28 | *** join/#asterisk putnopvut (i=putnopvu@nat/digium/x-6002c2461f147cf1) |
| 13:41.28 | [TK]D-Fender | BrokenNoze: Please describe the networking between your phone and your * server in detail. |
| 13:41.39 | cybertooth | Hmm, I bought a dev license for Cepstral about a year back... It worked about the same for me as festival - but only in lab conditions. |
| 13:42.08 | cybertooth | Thanks [TK]D-Fender , I'll take another look at Cepstral. |
| 13:42.37 | [TK]D-Fender | cybertooth: I don't have any personal experience with either, just recounting the opinions of several others I respect in here. |
| 13:42.53 | BrokenNoze | Fender : OK.. then if it's a network issue I can understand. My * is on a remote managed server. my handset is on a private network but i've put in the DMZ |
| 13:42.55 | cybertooth | Danke. |
| 13:43.18 | [TK]D-Fender | BrokenNoze: please breakdown that path in detail.... |
| 13:43.25 | BrokenNoze | however access to the internet is though a bloody managed exit. which i have NO control over |
| 13:44.22 | BrokenNoze | Phone ---» switch ---» router and firwall ----» CLOUD OF CRAP FROM MANAGED OFFICES ---» uk2net.com managed server |
| 13:45.05 | creativx | ive heard those crapclouds can be hard to talk through. |
| 13:45.11 | [TK]D-Fender | BrokenNoze: Is the * server effectively on a public IP? |
| 13:45.18 | BrokenNoze | I'm not really sure why this would fail though i suppose as we've had SipGate working through the cloud |
| 13:45.26 | BrokenNoze | Fedner: Yep |
| 13:45.47 | BrokenNoze | static |
| 13:45.50 | *** join/#asterisk Ebola (n=Ebola@host86-138-6-48.range86-138.btcentralplus.com) |
| 13:45.53 | [TK]D-Fender | BrokenNoze: then your phone's sip.conf entry should have "qualify=yes", "canreinvite=no", and "nat=yes" |
| 13:46.03 | [TK]D-Fender | BrokenNoze: And you should not be forwarding anything to it. |
| 13:46.05 | cybertooth | BrokenNoze, sounds like the "firwall" has some form of flood protection turned on, and it sees your RTP packets as a flood. Thus it blocks the IP... of course it's just a theory. |
| 13:46.09 | BrokenNoze | ah... qualify, thats a new one |
| 13:46.12 | [TK]D-Fender | BrokenNoze: (remove from DMZ) |
| 13:46.29 | BrokenNoze | OK.. |
| 13:47.05 | BrokenNoze | cheers guys I'll look up qualify |
| 13:47.10 | [TK]D-Fender | BrokenNoze: Go test and let us know. Also there is no Polycom "350", which did you actually mean? |
| 13:47.33 | BrokenNoze | will do, and it's a 330. sorry |
| 13:47.50 | BrokenNoze | crap phone btw. not impressed at all |
| 13:47.55 | [TK]D-Fender | BrokenNoze: Just thought I'd ask... not that its terribly pertinent to your problem :) |
| 13:48.00 | [TK]D-Fender | BrokenNoze: :O |
| 13:48.08 | [TK]D-Fender | Crap in what way? |
| 13:48.36 | BrokenNoze | well firstly i don't understand why polycom insist on continuing to make phones without backlights |
| 13:48.55 | [TK]D-Fender | BrokenNoze: IP 550/650 have that... but are pricey |
| 13:49.07 | [TK]D-Fender | BrokenNoze: So on to legitmate gripes... :) |
| 13:49.45 | JT | almost no ip phones have backlights, stupid reason to call a phone crap |
| 13:49.52 | *** join/#asterisk Ryushin (i=proxy@windwalker.openinnovations.com) |
| 13:50.13 | BrokenNoze | mmm |
| 13:50.28 | BrokenNoze | OK. well i suppose i'm just used to the 650s |
| 13:50.52 | [TK]D-Fender | BrokenNoze: Oh, so you're juse SPOILED them? :) |
| 13:51.04 | [TK]D-Fender | BrokenNoze: Oh, so you're just SPOILED them? :) |
| 13:51.07 | BrokenNoze | so perhaps i was being a little unfair there :-) |
| 13:51.08 | [TK]D-Fender | ahdhhfasfdghfglyuidfygioewrt |
| 13:51.28 | [TK]D-Fender | BrokenNoze: only in a Ferrari vs Lada kind of way.... |
| 13:51.42 | BrokenNoze | I guess |
| 13:51.47 | [TK]D-Fender | Jeez... I can't type today... |
| 13:51.51 | BrokenNoze | I withdraw my comment |
| 13:52.09 | BrokenNoze | :) |
| 13:53.29 | *** join/#asterisk jfitzgibbon (n=NADT@64.72.237.130) |
| 13:53.47 | [TK]D-Fender | file: MUFFINS dammit! |
| 13:53.55 | shido6 | muffins? |
| 13:53.57 | file | [TK]D-Fender: those you have to earn |
| 13:54.04 | shido6 | so hungry |
| 13:54.37 | [TK]D-Fender | file: So gimme a shout for next week's schedule! |
| 13:54.47 | JT | ah, the good old Soggy Sao ;) |
| 13:54.55 | file | [TK]D-Fender: aight |
| 13:55.54 | file | [TK]D-Fender: so far I only have the regular day stuff planned and dinner with a friend, so time slots are available! call now and receive a free steak knife |
| 13:56.28 | [TK]D-Fender | JT : ..... EW......... just... EW!!!!!!! |
| 13:56.50 | WilliamK | morning file |
| 13:57.31 | JT | [TK]D-Fender: i take it you understand the reference, even being in north america? :D |
| 13:58.11 | WilliamK | do ya'll prefer sub-modules that are borked on the same bug tracker report or do you like them broken down to individual reports? appears to be the same coding error (zaptel svn) |
| 13:58.24 | [TK]D-Fender | ~[TK]D-Fender |
| 13:58.25 | jbot | [TK]D-Fender is the Zen Master of the blatantly obvious, and #asterisk 's resident Google-proxy. |
| 13:58.27 | [TK]D-Fender | ^^^^^^^^^^^^^^^^^^^^ |
| 13:58.32 | Qwell | nice |
| 13:58.34 | JT | ah, google |
| 13:58.39 | [TK]D-Fender | JT : And now jsut a little more damaged for that search.... |
| 13:58.40 | file | WilliamK: what version of zaptel? |
| 13:58.46 | WilliamK | latest |
| 13:58.56 | JT | [TK]D-Fender: yeah i've heard that Limp Bizkit was actually named after Soggy Sao |
| 13:59.00 | JT | unsure how true that is |
| 13:59.02 | WilliamK | I even repulled the svn last night |
| 13:59.06 | file | there's 3 branches :D latest can mean 3 different things, and will yield 3 different answers from me |
| 13:59.47 | WilliamK | svn checkout http://svn.digium.com/svn/zaptel/trunk zaptel |
| 13:59.53 | WilliamK | that help? |
| 14:00.14 | cheGGo | heyas file :-) |
| 14:00.20 | file | yes, trunk is going to be wiped away soon |
| 14:00.35 | WilliamK | ah |
| 14:00.40 | file | because it is a mess |
| 14:00.43 | WilliamK | which one should I be pulling from? |
| 14:00.43 | Qwell | trunk has, for good reason, not been kept up to date |
| 14:00.44 | file | cheGGo: hola |
| 14:00.47 | file | 1.4 |
| 14:01.19 | cheGGo | file, hi :) u processed my "issue report" which wasn an issue ;) |
| 14:01.28 | cheGGo | with reinvite and sip |
| 14:01.51 | file | cheGGo: ooh, neat, I looked for you on here cause I vaguely remembered you talking when I woke up but your nick didn't match your Mantis |
| 14:02.30 | cheGGo | yes, indeed, sorry for bothering you ;) |
| 14:02.41 | file | don't be sorry, 'tis my job |
| 14:02.57 | file | did you need further clarification on something? |
| 14:03.49 | cheGGo | little bit... i searched in the rtp.c for the function which handled this behaviour |
| 14:04.10 | cheGGo | do you know if its a lot of work to change it mysql for my use case |
| 14:04.20 | cheGGo | mysql=myself ;) |
| 14:04.29 | file | set_rtp_peer is called from bridge_native_loop, it tells chan_sip to change the location where media should be sent... and chan_sip then sends out a reinvite |
| 14:05.05 | file | the places where it calls set_rtp_peer with NULL values and 0 values is where it is telling chan_sip to bring the audio back to the Asterisk box |
| 14:05.10 | file | removing those should get rid of the reinvite |
| 14:05.25 | cheGGo | ah, kkk... just comment out? |
| 14:05.36 | cheGGo | ah |
| 14:05.46 | file | remove... comment out... does the same thing |
| 14:06.06 | cheGGo | ok... i will check it tonigh/tomorrow |
| 14:06.23 | cheGGo | it possible to contact u tomorror again, for 1 or 2 questions |
| 14:06.29 | WilliamK | file: thanks! that one compiles correctly |
| 14:06.52 | file | cheGGo: I'll probably be here |
| 14:07.20 | *** join/#asterisk usam (n=alx@ppp-124.120.64.107.revip2.asianet.co.th) |
| 14:07.59 | cheGGo | file, very nice... thank u for ur qualified help :-) |
| 14:08.28 | cheGGo | i will check out the source :) |
| 14:09.48 | *** join/#asterisk snook3r (n=snook3r@bzq-219-46-202.isdn.bezeqint.net) |
| 14:11.04 | *** join/#asterisk fakhir (n=Fakhir@unaffiliated/fakhir) |
| 14:11.06 | *** join/#asterisk Aeudian (n=Aeudian@74.92.134.190) |
| 14:12.18 | Aeudian | Anyone know the old centrex *xx number for direct extension stealing. like *37 is direct pickup for a ringing extension |
| 14:13.09 | *** join/#asterisk mocker (n=user@198.247.173.227) |
| 14:13.54 | mocker | Awesome, I'm working with a person in Bulgaria and asked him what type of SIP phone he has. He responded, "Cosco, it's a very good copy of Cisco" |
| 14:14.07 | Qwell | great |
| 14:14.18 | etfonhomey_ | sweet |
| 14:14.40 | etfonhomey_ | Isn't that a grocery store? |
| 14:15.09 | mocker | In the US it is. |
| 14:15.15 | mvanbaak | gheh |
| 14:15.22 | cheGGo | bye bye |
| 14:15.33 | mocker | Anyone going to Astricon? |
| 14:15.44 | Qwell | mocker: #astricon |
| 14:15.53 | mocker | Qwell: Sweet. |
| 14:15.59 | Qwell | everybody going should be in there |
| 14:18.12 | *** join/#asterisk Dan0maN_Work (n=dschuh@64.149.174.136) |
| 14:18.16 | chemikk | hello i have problem, help me please: http://pastebin.com/d41c53091 |
| 14:19.49 | chemikk | problem with application DISA |
| 14:20.13 | chemikk | do not redirecting to tyoed number |
| 14:20.22 | chemikk | sorry for my english |
| 14:20.34 | *** part/#asterisk jarod14 (n=jarod14@mail.viatelecom.com) |
| 14:20.59 | BrokenNoze | Fender: Hey.. Looks like thats fixed it!!! thanks for your help. didn't know what qualify did but looked it up now and looks like the root of the problem. |
| 14:21.41 | [TK]D-Fender | chemikk: What did you try to dial exactly? |
| 14:21.50 | [TK]D-Fender | BrokenNoze: good to hear. |
| 14:23.12 | *** part/#asterisk mocker (n=user@198.247.173.227) |
| 14:23.18 | *** join/#asterisk mocker (n=user@198.247.173.227) |
| 14:25.15 | chemikk | [TK]D-Fender: number 800123456, and i need redirecting call to another sip operator: exten => s,20,Dial(SIP/552308181/${ARG1}) |
| 14:25.48 | chemikk | [TK]D-Fender: sorry im absolutly begginer |
| 14:27.28 | *** join/#asterisk Daviey (n=dave@ubuntu/member/daviey) |
| 14:27.34 | [TK]D-Fender | chemikk: Well you are telling DISA to use the [trymat] in which you can only dial 100 |
| 14:27.43 | [TK]D-Fender | ooops, strike that. |
| 14:28.02 | Daviey | Hi, any recommendations for ISDN30e / PRI service providers in the UK? |
| 14:28.16 | *** join/#asterisk qdk (n=qdk@0x50c627be.bynxx11.adsl-dhcp.tele.dk) |
| 14:28.25 | [TK]D-Fender | chemikk: Go do a "Read" before your DISA to make sure you are detecting DTMF properly |
| 14:30.23 | *** join/#asterisk santiago (i=santiago@debian/developer/santiago) |
| 14:30.33 | chemikk | ok |
| 14:30.41 | chemikk | i try it |
| 14:31.23 | *** join/#asterisk Infested (n=infested@24.148.112.10) |
| 14:31.44 | mocker | [TK]D-Fender: You going to be at astricon? |
| 14:31.54 | *** join/#asterisk CVirus (n=GoD@41.233.160.215) |
| 14:31.59 | mrbond82 | Is there a way to turn down the volume of sound that asterisk sends to people received from the sip phone? |
| 14:32.40 | [TK]D-Fender | mocker: Nope. Too far, too $$, and no passport (yet) |
| 14:32.52 | [TK]D-Fender | mrbond82: Nope. |
| 14:33.20 | [TK]D-Fender | mrbond82: audio is supposed to be normalized by each SIP endpoint. |
| 14:34.16 | mrbond82 | I'm surprised I didn't have to send you my sip.conf, extensions.conf, debug output from the console and my bank account info to get that answer out of you |
| 14:34.21 | mrbond82 | :) |
| 14:34.38 | anonymouz666 | hahahaha |
| 14:35.17 | etfonhomey_ | mrbond82, why bite the hand that feeds you? |
| 14:36.30 | *** join/#asterisk CVirus (n=GoD@41.233.160.215) |
| 14:37.08 | chemikk | [TK]D-Fender: i try use read and DTMF is not detecting |
| 14:37.35 | *** join/#asterisk Daejeo1 (n=chatzill@211.177.189.25) |
| 14:38.25 | Daejeo1 | anyone have sip firmware 8.6/8.7 for CP 7961g? |
| 14:38.27 | [TK]D-Fender | mrbond82: I'm (not) surprised its been an HOUR now since this was requested for your previous problem an you haven't told us you've resolved it, nor have you PROVIDED this simple request for information so we can HELP YOU. |
| 14:39.06 | [TK]D-Fender | mrbond82: Guess you're too lazy to do anything to help yourself and expect us to be able and willing to spoon-feed you from beginning to end. |
| 14:39.51 | *** join/#asterisk Juggie (n=Juggie@CPE00c049d9f271-CM000a73a18a20.cpe.net.cable.rogers.com) |
| 14:39.52 | [TK]D-Fender | chemikk: Ok, go verify what dtmfmode you should be using with your SIP channels, correct them and retest. |
| 14:40.02 | *** join/#asterisk datachomper (n=russ@ool-43509aa5.dyn.optonline.net) |
| 14:43.04 | *** join/#asterisk ToyMan (n=Stuart@72.168.167.241) |
| 14:45.33 | *** join/#asterisk hmmhesays (n=Neg@24-119-176-74.cpe.cableone.net) |
| 14:46.30 | hmmhesays | hello folks |
| 14:48.48 | *** join/#asterisk rmayorga (n=rmayorga@unaffiliated/rmayorga) |
| 14:52.20 | syzygyBSD | hello hmmhesays |
| 14:52.32 | *** part/#asterisk mitcheloc (n=mitchel@adsl-68-120-230-95.dsl.irvnca.pacbell.net) |
| 14:52.35 | syzygyBSD | though I don't consider myself a "folk" |
| 14:54.57 | *** join/#asterisk zerohalo (n=zeroHalo@pool-71-255-173-195.bstnma.east.verizon.net) |
| 15:00.57 | *** join/#asterisk jtknapp (n=skip@65-126-63-1.dia.static.qwest.net) |
| 15:01.23 | *** join/#asterisk russellb (i=russellb@asterisk/developer-and-stable-maintainer/drumkilla) |
| 15:01.23 | *** mode/#asterisk [+o russellb] by ChanServ |
| 15:02.21 | chemikk | [TK]D-Fender: right dtmfmode is inband, this dtmf is function with application "background" but no with "read", i dont understand this |
| 15:02.22 | *** join/#asterisk ming_zym (n=ming_zym@f0-0-tep-rtr1.corp.cnb.yahoo.com) |
| 15:03.04 | [TK]D-Fender | chemikk: inband is usually bad... the vast majority of services use rfc2833. give it a try |
| 15:05.46 | *** join/#asterisk lisandropm (n=lisandro@cpe-22-76.bvconline.com.ar) |
| 15:06.15 | lisandropm | Hello |
| 15:06.53 | lisandropm | Has anyone been able to connect a Hicom 300 (not E nor H) with a DIUS2 board to a server running asterisk using an E1 board? |
| 15:07.50 | chemikk | setting dtmfmode in sip.conf is setting how SEND dtmf and where is setting how asterisk DETECT dtmf with incoming call? |
| 15:08.06 | chemikk | its right? |
| 15:09.23 | [TK]D-Fender | chemikk: its the same. its for send AND receive for a given call |
| 15:09.31 | hmmhesays | syzygyBSD: why not? |
| 15:09.47 | syzygyBSD | cuz I am not over 50? |
| 15:10.51 | hmmhesays | tell that to the poor amish kid weaving those baskets they sell |
| 15:14.03 | Yourname` | Has anyone had any experience with gastman on windows?? |
| 15:14.05 | *** join/#asterisk jcmoore (n=jcmoore@unaffiliated/tgrman) |
| 15:22.13 | *** join/#asterisk Rinner (n=raid@h8441151249.dsl.speedlinq.nl) |
| 15:22.17 | Rinner | hello |
| 15:22.21 | Rinner | hello Hello, uhmn can i ask something about astrix and avaya ? |
| 15:22.22 | chemikk | how i show dtmf in cli? |
| 15:22.32 | Qwell | Rinner: I know what avaya is, but what is astrix? |
| 15:22.39 | Rinner | anybody into Avaya Clan card ? |
| 15:22.52 | Qwell | avaya is a cult! I knew it! |
| 15:22.54 | Rinner | Qwell. i ment astrisk |
| 15:22.59 | Qwell | What's astrisk? |
| 15:23.32 | shido6 | cousin game to a$trix |
| 15:23.33 | Rinner | Qwell. great, avaya did buy lucent, something like that, i have a C-Lan board, i need speeddialing on digital phones |
| 15:23.35 | Rinner | working |
| 15:23.51 | Rinner | Qwell. do i need to install a CTI Server software in order to talk to the C-lan board ? |
| 15:24.04 | Qwell | I don't know what a c-lan board is |
| 15:24.21 | Rinner | ow okay, a CTI board ? |
| 15:24.28 | [TK]D-Fender | Rinner: this is NOT an Avaya support channel. Youa re in the wrong place. |
| 15:24.49 | Rinner | TD , it is a telephone issue |
| 15:24.56 | Rinner | okay forget the question |
| 15:24.56 | Qwell | next somebody is going to ask about a voicemail board |
| 15:24.57 | Rinner | sorry. |
| 15:25.01 | [TK]D-Fender | Rinner: how... GENERIC. |
| 15:25.13 | Rinner | what about a mediaprocessorboard in a PC ? |
| 15:25.28 | [TK]D-Fender | Rinner: Again, nothing to do with us here. |
| 15:25.33 | Rinner | o oki |
| 15:29.24 | datachomper | Somehow tt-weasels found its way into one of my production IVRs |
| 15:29.45 | Qwell | datachomper: it has a tendency to do that |
| 15:34.25 | Yourname` | Oh, great. After installing SVN checkout of ast1.4, and stopping and starting asterisk, I still don't see Asterisk SVN being used in version. What did I do wrong now? |
| 15:35.28 | *** join/#asterisk wunderkin (n=wunderki@ip72-223-86-126.ph.ph.cox.net) |
| 15:36.33 | chemikk | bad life |
| 15:37.15 | *** join/#asterisk citats (n=james@mrplow.gnuinternet.com) |
| 15:37.42 | Yourname` | lol bad life yup |
| 15:41.09 | *** join/#asterisk cpm (n=Chip@pdpc/supporter/active/cpm) |
| 15:41.27 | *** join/#asterisk funxion (n=nunya@63.214.236.169) |
| 15:42.11 | Yourname` | Hmm, should I be even getting SVN from asterisk/branches/1.4 or /trunk? |
| 15:42.38 | *** join/#asterisk DrAk0 (n=ljd@nelug/coreteam/luisjose) |
| 15:42.43 | *** join/#asterisk dasuberdavid (i=david@nat/digium/x-12365217743545d7) |
| 15:42.49 | funxion | exten => _.,32,GotoIf($[${LEN(${CALLERIDNUM})} < "4"]?47:33) ;<--Does anyone see a problem with this? when I pass a call with a callerid with length 1 it jumps to 33 and I cant find a problem with the syntax |
| 15:44.09 | *** join/#asterisk DrAk0 (n=ljd@nelug/coreteam/luisjose) |
| 15:45.03 | [TK]D-Fender | funxion: whitespace + quotes |
| 15:45.20 | [TK]D-Fender | funxion: And you really should be using the CALLERID function as well |
| 15:45.54 | [TK]D-Fender | exten => _.,32,GotoIf($[${LEN(${CALLERID(num)})}<4]?47) <---- better |
| 15:46.03 | funxion | exten => _.,32,GotoIf($[${LEN("${CALLERIDNUM}")} < "4"]?47:33) I've tried this I've also tried the callerid(num) function as well |
| 15:46.17 | [TK]D-Fender | funxion: And as I said... QUOTES <--------- |
| 15:46.42 | [TK]D-Fender | funxion: also "_." is a terrible thing to do... matches dangerous extens... |
| 15:47.33 | Daejeo1 | nyone have sip firmware 8.6/8.7 for CP 7961g? |
| 15:47.52 | funxion | its a closed system and only specific calls have access into this context |
| 15:48.32 | [TK]D-Fender | funxion: Ok, fine, sure. |
| 15:48.35 | jfitzgibbon | funxion: he means that it matches 'i', 'h', 'a', 'o', and all those special Asterisk extensions |
| 15:49.30 | funxion | I got ya |
| 15:49.41 | funxion | really working on proof of concept |
| 15:49.54 | funxion | not finalized dialplan |
| 15:50.01 | funxion | thnx for the warning though |
| 15:50.17 | funxion | tk I tried what you gave me and got the same results |
| 15:50.18 | funxion | -- Executing GotoIf("Zap/1-1", "0<4?47:33") in new stack |
| 15:50.18 | funxion | -- Goto (seamobileJ,7321,33) |
| 15:50.46 | funxion | do I have the labels messed up? |
| 15:52.07 | datachomper | So, asterisk used to ignore the first sound file I would send to the channel, after I initialized an agi ivr. Now it's not ignoring them anymore. |
| 15:52.07 | funxion | how does that make sense |
| 15:52.16 | wunderkin | Yourname`: 1.4 |
| 15:52.34 | Yourname` | Thanks wunderkin |
| 15:52.57 | [TK]D-Fender | funxion: pastebin "dialplan show [context]" |
| 15:54.27 | funxion | http://pastebin.ca/682489 |
| 15:54.38 | *** join/#asterisk t3rror (n=t3rror@gateway.sscgp.com) |
| 15:56.26 | *** join/#asterisk caio1982 (i=caio1982@CAcert-br/caio1982) |
| 15:56.54 | [TK]D-Fender | GotoIf($[${LEN(${CALLERID(num)})}<4]?47:33) [pbx_config] |
| 15:56.55 | [TK]D-Fender | hrm |
| 15:57.04 | chemikk | [TK]D-Fender: please watch: http://pastebin.com/d11d21d27 |
| 15:57.40 | funxion | am I missing something? |
| 15:57.46 | [TK]D-Fender | funxion: somethings fishy... |
| 15:57.52 | funxion | yeah I realize |
| 15:57.56 | [TK]D-Fender | funxion: its like it ignored the $[] eval... |
| 15:58.04 | funxion | its * 1.2 |
| 15:58.27 | [TK]D-Fender | funxion: syntax hasn't changed from 1.0 even |
| 15:58.33 | [TK]D-Fender | funxion: SHOULD be fine |
| 15:58.43 | funxion | it works in other areas |
| 15:59.09 | funxion | i even tried tried GotoIf($[${LEN(${CALLERID(num)})}!=0]?47:33) |
| 15:59.14 | funxion | it didnt work either |
| 16:00.11 | datachomper | I <3 english. |
| 16:00.37 | [TK]D-Fender | chemikk: no idea... |
| 16:00.51 | *** join/#asterisk VJFROMGT (n=vjfromgt@static-68-161-227-229.ny325.east.verizon.net) |
| 16:00.55 | Qwell | use spaces |
| 16:01.10 | [TK]D-Fender | Qwell: news to me... since when? |
| 16:01.15 | VJFROMGT | i have question regarding ASR, when calculating ASR is a ring out a sucess or a fail? |
| 16:01.15 | Qwell | $[1<2] is a string |
| 16:01.17 | Renacor | k this is weird i do a read() for an extension, and put it in a variable, read says it "User entered '301'", however I do a NoOp() to show the variable, and it shows "s" whats going on?? |
| 16:01.20 | Qwell | since always :D |
| 16:01.28 | Qwell | except maybe in 1.4 |
| 16:01.30 | [TK]D-Fender | funxion: You heard him... |
| 16:01.36 | funxion | no |
| 16:01.38 | Qwell | codefreeze: ^^? |
| 16:01.38 | funxion | qwell? |
| 16:01.54 | [TK]D-Fender | funxion: correct... add aspace aroun < |
| 16:01.54 | funxion | thnx |
| 16:01.58 | codefreeze | Qwell: you rang? |
| 16:02.10 | Qwell | codefreeze: $[1<2] is evaluated as a string, yes? |
| 16:02.20 | Qwell | rather than math |
| 16:02.26 | chemikk | i have bad day |
| 16:02.27 | *** join/#asterisk [hC] (n=hardcore@ip67-90-234-94.z234-90-67.customer.algx.net) |
| 16:02.35 | [TK]D-Fender | Renacor: Probably because you think you can read to EXTEN with is READ-ONLY. |
| 16:02.49 | codefreeze | Qwell: should be mathmatical. Both numbers are pure digits |
| 16:02.55 | [TK]D-Fender | Renacor: Don't try to mess with the exten you're IN |
| 16:02.56 | Renacor | oh it's a predefined variable? |
| 16:03.04 | Qwell | codefreeze: what about in 1.2? I thought there was a problem with that at one point |
| 16:03.04 | funxion | yay it werkd |
| 16:03.09 | [TK]D-Fender | Renacor: its "Wher you are" in the dialplan. |
| 16:03.10 | funxion | thats tk and qwell |
| 16:03.17 | [hC] | [TK]D-Fender: Hey, you were mentioning that the new polycom firmware, 2.2.0, has that ring-when-busy feature to audibly ring on the base of the phone for call waiting. Is that on by default or something? I seem to have lost call waiting indications after upgrading. |
| 16:03.20 | funxion | thanks is what I meant to say |
| 16:03.31 | Renacor | ahh i see |
| 16:03.51 | codefreeze | Qwell: hmmmm. 1.2. I'd have to look; but I'd think it'll still eval to "1" there, too. |
| 16:03.53 | [TK]D-Fender | [hC]: Maybe if you tried using an old config template that overrides the key its in. Did you rebuild from scratch? |
| 16:04.13 | funxion | qwell you are correct |
| 16:04.15 | Qwell | funxion: You may want to verify that it's working as expected... |
| 16:04.22 | funxion | it is |
| 16:04.29 | codefreeze | Qwell: waitaminute... either way, it should eval to 1, shouldn't it? |
| 16:04.39 | Qwell | codefreeze: 1<2 was just an example |
| 16:04.45 | Qwell | GotoIf($[${LEN(${CALLERID(num)})}!=0]?47:33) |
| 16:04.48 | funxion | codefreeze it wasnt working |
| 16:04.48 | Qwell | That's what he was trying |
| 16:05.08 | Qwell | I assume it was always going to 47 |
| 16:05.21 | codefreeze | Qwell: in 1.2, are we still using the old expr parser? then you would have to say 1 < 2 |
| 16:05.27 | Qwell | k |
| 16:05.35 | codefreeze | with a single space around each token...? |
| 16:05.38 | [TK]D-Fender | Renacor: ok, lunch time, back in a few. |
| 16:05.51 | Renacor | ? |
| 16:05.52 | [TK]D-Fender | codefreeze: either side of an operator |
| 16:05.58 | Renacor | [TK]D-Fender: figured it out |
| 16:06.04 | Renacor | [TK]D-Fender: thanks for the info |
| 16:06.07 | funxion | 32,GotoIf($[${LEN("${CALLERIDNUM}")} < "4"]?47:33) is what I was using originally but it wasnt working |
| 16:06.08 | [TK]D-Fender | ok, lunch time, back in a few. |
| 16:06.15 | [TK]D-Fender | Renacor: wasn't meant to be directed to you |
| 16:06.21 | Renacor | k |
| 16:06.33 | [TK]D-Fender | funxion: yeah, the quotes were killink it. |
| 16:06.58 | funxion | yup |
| 16:07.05 | [TK]D-Fender | funxion: GotoIf($[${LEN(${CALLERID(num)})} < 4]?47) <- there |
| 16:07.08 | [hC] | [TK]D-Fender: i used an older config template but i do so by means of separate includable files.. my overrides only touch registration details and some small config things like one touch voicemail, etc.. do you know what config setting it is, for the call waiting selection? |
| 16:07.35 | slima | I have 2 accounts from the same sip provider, and inbound calls match always the first register, why? and how to fix it? my configs http://pastebin.com/d64bef9a0 |
| 16:07.35 | [TK]D-Fender | [hC]: Don't know offhand... go DL the admin guide :) |
| 16:07.49 | funxion | thanks |
| 16:07.58 | slima | sorry for my english.. |
| 16:08.16 | [hC] | [TK]D-Fender: already downloading :) |
| 16:08.55 | anonymouz666 | DEBUG[1069] chan_zap.c: DTMF digit: f on Zap/1-1 |
| 16:08.59 | codefreeze | funxion: [TK]D-Fender: right. compare as nums, not strings. "201" will be less than "4" |
| 16:09.01 | anonymouz666 | how can I send the 'f' digit? |
| 16:09.37 | anonymouz666 | how chan_zap detected that? |
| 16:10.12 | anonymouz666 | and after: -- Redirecting Zap/1-1 to fax extension |
| 16:11.11 | *** join/#asterisk etfonhomey (n=chatzill@12.169.248.226) |
| 16:12.59 | funxion | does anyone have any experience with thomson phones? |
| 16:14.25 | VJFROMGT | hos do i terminate a call from cli? |
| 16:15.48 | dasuberdavid | soft hangup |
| 16:16.05 | Qwell | dnubb! |
| 16:16.07 | *** join/#asterisk shinao1 (n=shinao1@196.207.1.30) |
| 16:16.14 | dasuberdavid | Qwell: whats up |
| 16:16.29 | slima | any idea? |
| 16:16.36 | Qwell | dasuberdavid: trolling our blog :p |
| 16:16.48 | dasuberdavid | Qwell: ha ha ! |
| 16:17.00 | russellb | my blog > digium blog |
| 16:17.01 | russellb | lol |
| 16:17.03 | anonymouz666 | it's very strange the dial does not have the 't' option and asterisk is still interpreting the dtmf sent by called party |
| 16:17.15 | Qwell | russellb: Does YOUR blog ...umm...do stuff? |
| 16:17.16 | *** join/#asterisk nny (n=nny@64.203.239.83.static-pool-4.pool.hargray.net) |
| 16:17.20 | Qwell | I've got nothin' |
| 16:17.29 | russellb | Qwell: it has a sweet turtle picture on it. |
| 16:17.32 | *** join/#asterisk dpc_clyde (n=clyde_@port-212-202-71-89.dynamic.qsc.de) |
| 16:17.38 | dpc_clyde | hi |
| 16:17.41 | Qwell | is it the ipv6 turtle? |
| 16:17.44 | Qwell | oh, meh, it isn't |
| 16:17.50 | anonymouz666 | fax sux |
| 16:17.54 | russellb | it's the ... turtle i found next to my hose |
| 16:18.01 | Qwell | You found a turtle next to your house? |
| 16:18.03 | Qwell | here? |
| 16:18.04 | Deeewayne | Is it the tortoiseCVS turtle ? |
| 16:18.14 | Qwell | kame turtle++ |
| 16:18.24 | nny | hi, have an asterisk box that seems to be having vm issues. left a vm, client went to check it, logged in with password, and hit key to play vm. instead of playing vm, it gives fast busy signal |
| 16:18.25 | Qwell | only real men can see kame dance |
| 16:18.30 | russellb | no! it is my own turtle ... that i keep in a box. |
| 16:18.31 | Qwell | http://www.kame.net/ |
| 16:18.32 | russellb | not really. |
| 16:18.56 | Qwell | oh, hmm, speaking of ipv6 |
| 16:19.07 | Qwell | I *just* remembered why I replaced my linksys router years ago with a sparcstation |
| 16:19.21 | Deeewayne | russellb: you should bring your turtle to my house and let him ride on Reba's back |
| 16:19.27 | dpc_clyde | short question, how can i set that asterisk after 20sek sends busy when a caller from extern call over sip? |
| 16:19.36 | Daejeo1 | anyone have sip firmware 8.6/8.7 for CP 7961g? |
| 16:19.37 | russellb | Deeewayne: he ran away :( |
| 16:19.51 | Qwell | Daejeo1: You have to buy it from Cisco, basically |
| 16:19.58 | Qwell | nobody here is going to give it to you |
| 16:20.06 | Deeewayne | next time: always tether your turtle |
| 16:20.28 | file | Deeewayne: his terms of service does not allow tethering |
| 16:20.54 | russellb | ~thwap file |
| 16:20.55 | jbot | ACTION thwaps file on the nose with a 2 by 4 |
| 16:21.13 | file | such hate |
| 16:21.36 | russellb | dpc_clyde: just set the Dial() application timeout |
| 16:21.42 | russellb | Dial(SIP/whatever|| |
| 16:21.44 | russellb | errr ... |
| 16:21.54 | russellb | Dial(SIP/whatever|20) |
| 16:21.55 | Yourname` | Does anyone know of a provider that lets atleast 20-30 channels on a single inbound DID? |
| 16:22.03 | *** join/#asterisk krdian_ (i=krdian@killer.radom.net) |
| 16:22.03 | Qwell | Yourname`: any of the per minute providers should |
| 16:22.14 | *** join/#asterisk DaveCanoe (n=Dave@canoe404.dclg.ca) |
| 16:22.24 | dpc_clyde | it works intern, but if someone from extern calls he recivied after 20 sek a busy signal..... |
| 16:22.29 | Yourname` | Qwell: I see gafachi lets only 2 channels on an inbound tollfree number.. |
| 16:22.40 | Qwell | Yourname`: well, that's silly |
| 16:22.44 | Yourname` | Qwell: Almost everybody is like that, for some reason. |
| 16:22.48 | Yourname` | Yeah, tell me about it lol |
| 16:22.48 | Qwell | nufone.net lets you |
| 16:22.52 | Qwell | or, they used to |
| 16:23.06 | Yourname` | I think I check nufone, let me look again. Harder. |
| 16:24.40 | Renacor | is there a place to globally set the callerid for all outbound calls (outside the pbx) |
| 16:28.01 | Yourname` | Qwell: IT doesn't say anywhere, lol.. so I guess I'll have to contact these guyd |
| 16:28.54 | *** join/#asterisk tsurko (n=tsurko@78.90.100.214) |
| 16:32.07 | nny | anyone have a preferred windows app for playing gsm files? |
| 16:32.21 | slima | winamp |
| 16:33.45 | nny | thanbks |
| 16:33.47 | nny | thanks* |
| 16:34.42 | etfonhomey | Looking for an * consultant to provide installation/support services in the New York City area (Florham Park, NJ). Msg me if interested. |
| 16:35.24 | nny | i have a server that is not recording vms properly. Worked fine up until last week. audio wav files are blank and when i leave a message, it just plays silence. Any advice? |
| 16:35.29 | [TK]D-Fender | Yourname`: Most of the per-minute and places that that charge per channel as well |
| 16:35.52 | slima | nny: you require http://www.mlkj.net/gsm/winamp_plugin_gsm_codec.php |
| 16:35.59 | [TK]D-Fender | nny: Use "Record" and do a seperate test to make sure * is getting the audio. |
| 16:36.09 | Yourname` | [TK]D-Fender: Ah.. any no hassle signup and start providers you know of? |
| 16:36.33 | [TK]D-Fender | Yourname`: I don't about the hassle factor, and it depends where you want to DID's from |
| 16:36.49 | nny | [TK]D-Fender: thanks |
| 16:36.58 | Yourname` | [TK]D-Fender: Toll free or anywhere in the US at all. Doesn't matter, really. |
| 16:37.22 | [TK]D-Fender | Yourname`: Check out VoicePulse Connect |
| 16:37.38 | Yourname` | ok |
| 16:37.45 | *** join/#asterisk Olgem (n=Olgem@host-69-144-136-61.bln-mt.client.bresnan.net) |
| 16:41.42 | Nugget | http://connect.voicepulse.com/ |
| 16:41.56 | Nugget | if you go to the "real" voicepulse site (at www, not connect) it will only confuse you |
| 16:46.51 | *** join/#asterisk NightMonkey (n=NightMon@pdpc/supporter/sustaining/NightMonkey) |
| 16:47.52 | *** part/#asterisk dpc_clyde (n=clyde_@port-212-202-71-89.dynamic.qsc.de) |
| 16:50.29 | neverblue | is * 1.2.18 a very insecure server, or are the latest updates minimal ? |
| 16:50.33 | nny | hmm confusion here. I play /var/spool/asterisk/voicemail/user/100/inavail.wav or .gsm locally and it works, but when I call to hit VM it goes to the stock unavail and the messages are all dead air, like it can't read or write to the 100 folder... |
| 16:50.54 | nny | it says in console it is playing the corret file, no errors evident |
| 16:50.59 | nny | correct* |
| 16:55.32 | Corydon76-dig | nny: maybe it's trying to access 100@default instead of 100@user |
| 16:55.57 | Corydon76-dig | btw, that should be unavail, not inavail |
| 16:56.06 | nny | oh yeah unavail typo |
| 16:56.23 | Corydon76-dig | dumber things have happened than a simple typo |
| 16:56.42 | nny | working with dev on issue now, let you all know once he gets into it. He is a lot more experienced with it than i am :) |
| 17:02.47 | *** join/#asterisk ta^3 (n=tacvbo@189.136.41.204) |
| 17:06.17 | *** join/#asterisk ptblank (n=MURDER1@cpe-75-84-218-175.socal.res.rr.com) |
| 17:09.58 | *** join/#asterisk caio1982 (i=caio1982@CAcert-br/caio1982) |
| 17:15.11 | *** join/#asterisk ming_zym (n=ming_zym@f0-0-tep-rtr1.corp.cnb.yahoo.com) |
| 17:15.40 | Yourname` | Thanks, Nugget. |
| 17:15.48 | *** join/#asterisk CVirus (n=GoD@41.233.160.215) |
| 17:15.59 | Yourname` | I just found sellvoip.net too. It's like 0.003 per min on tollfree inbounds. |
| 17:16.24 | Qwell | a third of a cent per minute? |
| 17:16.28 | Yourname` | Yessir. |
| 17:16.31 | Qwell | no |
| 17:16.51 | Yourname` | http://www.sellvoip.net/NewRateForm.php |
| 17:17.11 | Qwell | $0.030 |
| 17:17.24 | *** join/#asterisk mog (i=mog@nat/digium/x-78196c0e7fa11d52) |
| 17:17.24 | *** mode/#asterisk [+o mog] by ChanServ |
| 17:17.43 | Yourname` | Better than 3.000 at most places! |
| 17:17.45 | Yourname` | lol |
| 17:17.50 | *** join/#asterisk Op3r (n=Op3r@121.97.145.174) |
| 17:18.01 | *** join/#asterisk grantm (n=grantm@kolob.wingateservices.com) |
| 17:18.45 | Op3r | somebody tells me a good billing software for asterisk |
| 17:20.21 | *** join/#asterisk Op3r (n=Op3r@121.97.145.174) |
| 17:20.34 | GlobeTrotter | helloo,, i have this error on my * box.. i get the error right after the call is sent to the queue |
| 17:20.35 | GlobeTrotter | translate.c:163 framein: no samples for g729tolin |
| 17:20.59 | GlobeTrotter | anyone had this error before? |
| 17:21.06 | GlobeTrotter | what does it mean? |
| 17:21.27 | Qwell | GlobeTrotter: sounds like you have VAD on or something |
| 17:22.20 | GlobeTrotter | VAD on my asterisk? |
| 17:22.37 | Qwell | on your client |
| 17:23.48 | GlobeTrotter | i am using grandstream phones and i doen have that set,, nor do i have silence suppresion on |
| 17:24.05 | GlobeTrotter | is that an error to be concerned about? |
| 17:24.40 | Qwell | VAD is silence suppression |
| 17:25.20 | *** join/#asterisk Defraz (n=t0tal@fw.fuzecore.com) |
| 17:27.13 | *** join/#asterisk jcmoore (n=jcmoore@unaffiliated/tgrman) |
| 17:27.58 | *** join/#asterisk generalhan (n=asd@ip67-90-64-2.z64-90-67.customer.algx.net) |
| 17:28.01 | generalhan | hey all ! |
| 17:28.29 | *** join/#asterisk Ebola (n=Ebola@host86-138-6-48.range86-138.btcentralplus.com) |
| 17:29.12 | generalhan | anyone here worked with an Aastra 480i SIP Phone ? im trying to figure out how to put a line monitor on one of the softkeys, so that i can see if a certain line is active or available |
| 17:32.24 | [TK]D-Fender | generalhan: IINM its jsut like the 5i series for web administration. Is that how you're adding them? |
| 17:34.02 | generalhan | well i dont have the phone here yet, its opn its way, so im not really doing it at all. i just want to setup something like HINTS so that i can have something on the display to show if another line is busy,ringing,or available |
| 17:35.03 | [TK]D-Fender | generalhan: "hints are how you do the * side. adding a subscription via the web interface is dead easy. I've SEEN the provisioning and it looked pretty simle there too... |
| 17:35.13 | generalhan | but i also have a 57i and sidecar coming as well, so i guess i can learn them together ! |
| 17:35.23 | [TK]D-Fender | generalhan: Actually I HAVE provisioned a 480i once myself... forgot about it it was so long ago... |
| 17:35.32 | generalhan | ok so it needs to be done via the web interface not the config files |
| 17:35.38 | [TK]D-Fender | generalhan: Side-car=really nice, but the PHONE.... bleh |
| 17:35.47 | generalhan | eww ... really ? |
| 17:35.49 | [TK]D-Fender | generalhan: can be via EITHER |
| 17:36.26 | generalhan | is there another 5*i that is better than the 57 ? or are all the 5i's "bleh" ? |
| 17:36.28 | [TK]D-Fender | generalhan: handset has NO weight. hug LCD with SHIT USAGE. inferior call handling, 2nd rate audio quality, ^#%$ RUBBER BUTTONS. |
| 17:36.44 | [TK]D-Fender | generalhan: 5i = al of the 5Xi series |
| 17:36.47 | [TK]D-Fender | all |
| 17:37.20 | generalhan | boo, i really want that phone for the sidecar ! lol, plus its WAY less expensive than a Cisco with expansion |
| 17:37.20 | [TK]D-Fender | generalhan: I was using a 57i CT here as my desk phone. I'd much have preserred my bed-side Polycom IP301 over it. |
| 17:37.50 | [TK]D-Fender | generalhan: For a single identity receptionist it might do, but I wouldn't want to use one personally. |
| 17:38.47 | generalhan | well thats what the 5i is for ... a receptionist that will need to see about 30 users' availablility. the 480i is for another receptionist that only needs to monitor 2 users' availability |
| 17:39.32 | [TK]D-Fender | generalhan: 480i at lease doesn't use the 5i shit-for-all rubber buttons |
| 17:39.36 | [TK]D-Fender | least* |
| 17:39.47 | generalhan | how did you like the cordless peice ? cuase we were thinking about that one for the 480i. that way this person can get up and roam around without missing calls |
| 17:40.07 | generalhan | [TK]D-Fender: if i could get a sidecar for the 480 i would just get 2 of those !!! |
| 17:40.08 | [TK]D-Fender | generalhan: soft-keys are the one thing that Aastra did AWESOMELY |
| 17:40.38 | [TK]D-Fender | generalhan: Cordless is decent as long as its for the SAME reg as the base (single for both) |
| 17:40.55 | generalhan | sweet, thats how it will be setup ! |
| 17:41.21 | generalhan | and its not an everyday thing ... but if she needs to run to the copy machine or whatever, i dont have to hear anyone complaining about missed calls |
| 17:42.49 | [TK]D-Fender | generalhan: you might be happy with it then. Get a good large belt-clip pouch for it. don't bet on the little clip it comes with to last |
| 17:43.19 | generalhan | i havent used any of the more advanced Aastra phones, but i LOVE the 9133i phones. we have about 50 of them and havent had a single problem yet |
| 17:43.24 | funxion | does anyone have any experience with thomson phones? |
| 17:43.42 | generalhan | im hoping that these next two have the same trac-record with us ! |
| 17:44.22 | [TK]D-Fender | funxion: Only really have a presence in Europe, but I hear they're pretty decent. |
| 17:44.39 | funxion | I'm using them in paris |
| 17:44.58 | funxion | our IT guy in paris insisted on them |
| 17:45.11 | funxion | he loaded some fuinky firmware and now were having voice issues |
| 17:45.19 | funxion | quality issues |
| 17:45.38 | [TK]D-Fender | funxion: The go undo the upgrade! |
| 17:45.43 | funxion | was hoping someone had a good experience with them and miht be able to suggest a revision |
| 17:47.03 | [TK]D-Fender | funxion: Sorry can't help you there, and there is VERY little talk about them in here... |
| 17:47.15 | [TK]D-Fender | funxion: jsut that what little I've heard has been positive |
| 17:50.11 | *** join/#asterisk zotz (n=zotz@24.244.163.157) |
| 17:52.09 | cpm | wondering aloud, want to try to use hylafax (with a *real* modem) on my one phone line. Would be neat if hylafax could poll asterisk to see if the fxo port was in use before attempting to send a fax. |
| 17:52.28 | *** join/#asterisk saftsack (n=saftsack@pD9E07188.dip.t-dialin.net) |
| 17:52.54 | cpm | but I may be using broken thought processes here, (as I often do) |
| 17:53.33 | *** join/#asterisk saftsack (n=saftsack@pD9E07188.dip.t-dialin.net) |
| 17:53.34 | *** join/#asterisk ajohnson (n=ajohnson@corpex.pivotal.televerde.com) |
| 17:53.44 | *** join/#asterisk MdeP (n=mdep@236-151-180.adsl.din.tie.cl) |
| 17:54.02 | ajohnson | Does anyone know if you can pass a named priority in a manager originate event? |
| 17:54.45 | *** join/#asterisk saftsack (n=saftsack@pD9E07188.dip.t-dialin.net) |
| 17:58.06 | [TK]D-Fender | cpm: You can block it if you want, but I'm not sure about "notifying" it |
| 17:58.36 | *** join/#asterisk Xarion (n=xarion@c1-34-6.rndf.isadsl.co.za) |
| 17:58.54 | Xarion | bmg505 is kwaai |
| 17:59.01 | [TK]D-Fender | ajohnson: I don't think so. When the dialplan is loaded everything is parsed out and priority numbers are evaluated. |
| 18:02.26 | *** join/#asterisk drwelby (n=mpfister@mail.enplan.com) |
| 18:06.18 | cybertooth | cpm, I think Hylafax can do this. |
| 18:06.47 | cpm | cybertooth, yeah, but the more I think about it, it sounds like a rubegoldberg approach |
| 18:06.58 | cybertooth | ... and if not, you can set it to login remotely to the asterisk server and poll that channel to see if it is in use. |
| 18:07.25 | cybertooth | rubegoldberg <== my hero. |
| 18:07.58 | Sweeper | and they want to do it over t.38.... |
| 18:08.31 | Nivex | Facsimilie: some the finest technology the 80's had to offer |
| 18:08.37 | *** join/#asterisk etfonhomey (n=chatzill@12.169.248.226) |
| 18:10.01 | cybertooth | Dude. In the early days we used to intercept, decode and resend via our own fax server - now we just resell a local Analog line and charge them more for Faxing. |
| 18:10.49 | cybertooth | The trouble-ticket load is not worth the amount of money you make by supporting faxing via VoIP. |
| 18:12.32 | *** join/#asterisk JerJer (n=PhatJ@pdpc/supporter/bronze/jerjer) |
| 18:12.33 | ajohnson | [TK]D-Fender: Thanks for the response, that makes sense |
| 18:13.11 | JerJer | has anyone ever attempted to parse a channel variable into two or more variables ? |
| 18:13.22 | ajohnson | yes, using cut? |
| 18:13.41 | JerJer | ajohnson: with variable length data |
| 18:14.03 | JerJer | perhaps like the perl split |
| 18:14.03 | ajohnson | Any delimiter? |
| 18:14.03 | *** join/#asterisk bkruse (i=bkruse@nat/digium/x-08ec0ece99a4db41) |
| 18:14.14 | JerJer | yes csv data |
| 18:14.21 | JerJer | simple csv data |
| 18:14.37 | ajohnson | I have set variables in sip.conf entries using a delimiter and then split them up using cut |
| 18:14.43 | Corydon76-dig | JerJer: show function CUT |
| 18:14.48 | ajohnson | csv should be relatively easy |
| 18:14.49 | JerJer | great! |
| 18:15.04 | JerJer | i figured cut might work, but wanted to be sure before I committed :) |
| 18:15.15 | Corydon76-dig | JerJer: works very similar to the cut(1) command, which is what it was modeled after |
| 18:15.42 | Corydon76-dig | Well, cut(1) with the -d option, anyway |
| 18:15.47 | *** join/#asterisk DrukenHME (n=jdumais@CPE000e08cb2a29-CM00137189cb0c.cpe.net.cable.rogers.com) |
| 18:16.29 | *** join/#asterisk Grash (n=grash@207.144.216.87.dynamic.jazztel.es) |
| 18:17.10 | Grash | Hi! |
| 18:17.17 | JerJer | low |
| 18:17.26 | Grash | hi |
| 18:17.51 | JerJer | Corydon76-dig: have you ever utilized sql stored procedures in func_odbc or func_mysql ? |
| 18:18.01 | JerJer | like call them from... |
| 18:18.09 | *** join/#asterisk hfb (n=hfb@pool-71-116-255-84.lsanca.dsl-w.verizon.net) |
| 18:19.13 | *** join/#asterisk zapp-branigan (n=zapp-bra@9.218.216.87.static.jazztel.es) |
| 18:19.35 | zapp-branigan | hi i have a proble compiling asterisk addons : configure.in:32: error: possibly undefined macro: AC_PROG_LIBTOOL |
| 18:19.35 | zapp-branigan | <PROTECTED> |
| 18:19.35 | zapp-branigan | <PROTECTED> |
| 18:20.24 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 18:20.59 | Corydon76-dig | JerJer: " { CALL proc(args); } ", I think |
| 18:21.08 | *** join/#asterisk RypPn (i=TuMbL@rosscom.demon.co.uk) |
| 18:21.26 | Corydon76-dig | Might just be " { proc(args); } " |
| 18:21.28 | JerJer | yes - but have you / anyone actually done that ? |
| 18:21.31 | JerJer | its call |
| 18:21.52 | Corydon76-dig | Yes. You have to wrap it in the curly braces, though |
| 18:21.55 | JerJer | i had to tweak a few different things in openser to make it work with stored procedures |
| 18:22.30 | JerJer | like in mysql_real_connect and how the queries were dealt with - in the mysql C api |
| 18:23.58 | bmg505 | Xarion, wat maak jy hier? |
| 18:24.12 | Xarion | =] |
| 18:24.23 | Xarion | need help with this damn sipura |
| 18:24.27 | JerJer | spraken ze engrish plz |
| 18:25.05 | denon | hehe JerJer learned german from watching hogan's heroes |
| 18:25.23 | JerJer | the great escape :) |
| 18:27.06 | bmg505 | actually its a language called afrikaans |
| 18:27.13 | bmg505 | which is spoken in south africa |
| 18:27.15 | denon | great movie |
| 18:27.51 | Strom_M | hoe gaan dit |
| 18:28.07 | *** join/#asterisk doughecka (n=doug@unaffiliated/doughecka) |
| 18:28.13 | denon | JerJer: you ever see Stalag 17? |
| 18:28.40 | *** join/#asterisk Slimey (n=simon@virtual.bogons.net) |
| 18:29.09 | JerJer | ok - next topic... how does one get mysql development libs into an Asterisk BE / Rpath system ? conary sucks |
| 18:29.38 | JerJer | i spoze i could call but i hate the phone |
| 18:30.35 | cybertooth | The Rpath guys are very responsive (and conary does suck, but less than almost any other package management system) |
| 18:30.50 | JerJer | apt works great for me |
| 18:31.10 | JerJer | but i'm not the run running BE |
| 18:31.18 | cybertooth | Yep. But do you package your own apt packages? |
| 18:31.25 | JerJer | cybertooth: nope |
| 18:31.43 | JerJer | well if i happen to need a custom kernel, yes |
| 18:31.47 | cybertooth | Bless those that do create them. |
| 18:31.56 | JerJer | asterisk / openser i compile myself |
| 18:32.14 | cybertooth | meir auch. |
| 18:32.28 | JerJer | but the rest of my LAMP crap is all given to me via apt-get |
| 18:32.58 | cybertooth | Understood. The cutting edge that has to be right - you do, the rest can all come as it is. |
| 18:33.06 | JerJer | granted I am my own debian mirror, so yeah know |
| 18:33.10 | Slimey | anyone here understand asterisk internals? :) |
| 18:33.39 | JerJer | Slimey: nobody does - its just an ever evolving suite of plugins |
| 18:33.42 | pkunkra | ok. i officially love asterisk now. |
| 18:33.45 | Slimey | bah |
| 18:33.52 | cybertooth | its about time. |
| 18:34.01 | Slimey | I need to make some code changes in chan_sip.c |
| 18:34.19 | ajohnson | ruh roh |
| 18:34.19 | Slimey | but, I need to pass some info in from the dialplan |
| 18:34.20 | JerJer | Slimey: have you examined http://bugs.digium.com/ |
| 18:34.30 | cybertooth | Slimey, I do that all the time. |
| 18:34.36 | JerJer | Slimey: why make changes ? use channel variables |
| 18:34.40 | Slimey | ok |
| 18:34.51 | Slimey | how do I access a channel variable from C? :) |
| 18:35.02 | Slimey | that's what I've not worked out yet |
| 18:35.05 | JerJer | there is an api function |
| 18:35.14 | Corydon76-dig | pbx_builtin_getvar_helper() |
| 18:35.18 | JerJer | wor |
| 18:35.18 | JerJer | d |
| 18:35.19 | pkunkra | asterisk told me to turn off comfort noise generation on my phone. that solved a huge issue i was having for a while. |
| 18:35.22 | Slimey | cool |
| 18:35.30 | *** join/#asterisk kkn088 (n=kikoun@88-136-53-187.adslgp.cegetel.net) |
| 18:35.46 | *** join/#asterisk CrazyTux[m] (n=CrazyTux@h460e93db.area3.spcsdns.net) |
| 18:35.50 | Slimey | so, I'm just about to write a patch to chan_sip |
| 18:36.04 | Slimey | that allows you to specify a prefix for the call-id that gets generated |
| 18:36.04 | JerJer | Slimey: grep around for its usage - it has a few specific requirements |
| 18:36.18 | Slimey | do you think anyone else will be interested? :) |
| 18:36.32 | ajohnson | I wonder if they plan on implementing the ability to create new functions in the dialplan |
| 18:36.42 | JerJer | like adding the system name? |
| 18:36.49 | chemikk | <PROTECTED> |
| 18:36.51 | JerJer | ^^^ Slimey |
| 18:37.02 | tzafrir_laptop | Actually with any decent package management system it shouldn't be difficult to modify and rebuild your own packages |
| 18:37.15 | Slimey | jerjer: I need to pass some info to our prepay system |
| 18:37.29 | Slimey | and the callid is the only SIP header which gets passed to it that I can change |
| 18:37.31 | JerJer | Slimey: then use a CDR varible |
| 18:37.42 | JerJer | or set a custom SIP header |
| 18:37.49 | Slimey | can't |
| 18:37.59 | JerJer | then your prepay system is crap |
| 18:37.59 | cybertooth | tzafrir_laptop is right, but it is all about the best tool for the job. Sometimes the best tool is the one that fits *your* hand well. |
| 18:38.07 | Slimey | JerJer: Yes, it is :) |
| 18:38.28 | Slimey | we've implemented a prepay system on a broadworks platform |
| 18:38.32 | JerJer | Slimey: write a new one using Adhearsion |
| 18:38.41 | Slimey | except they found that broadworks can't do IVR |
| 18:38.47 | Slimey | so I've bolted Asterisk on the front end |
| 18:39.11 | Slimey | $1m voice platform, and need to use asterisk to make it useful :) |
| 18:39.12 | *** join/#asterisk ToTo (n=ToTo@host223-91-dynamic.56-82-r.retail.telecomitalia.it) |
| 18:39.23 | cybertooth | Hear, hear! |
| 18:39.33 | JerJer | i coulda put that 1m to much better use |
| 18:39.42 | Slimey | so could I |
| 18:39.44 | *** join/#asterisk ToTo (n=ToTo@host223-91-dynamic.56-82-r.retail.telecomitalia.it) |
| 18:40.04 | ajohnson | Like the lots of awesome stuff for ajohnson fund |
| 18:40.16 | Slimey | all through the deployment, when broadsoft got stuck on stuff, I kept saying "I can do that in Asterisk" |
| 18:40.30 | Slimey | eventually they relented and let me use asterisk for the IVRs |
| 18:40.39 | cybertooth | We have to mod chan_sip.c to allow our Asterisk app server and gateways to interact nicely with the Genband T6000 |
| 18:41.10 | cybertooth | All our cutting edge apps run via Asterisk, but the core is still Genband. |
| 18:41.42 | VJFROMGT | i have 1 gig or ram on a p4 , do i have enough ram to do 20 concurent calls with all sort of transcoding? |
| 18:41.57 | ajohnson | yes |
| 18:42.09 | VJFROMGT | darn,, |
| 18:42.17 | VJFROMGT | i cant figure out why my calls are breaking up |
| 18:42.25 | VJFROMGT | only outbound sound breaks up |
| 18:42.32 | VJFROMGT | bandwith looks like its fine |
| 18:42.35 | ajohnson | 128 Megs would probably be more than sufficient |
| 18:42.46 | ajohnson | What are you transcoding from/to? |
| 18:42.55 | VJFROMGT | ulaw to g729 mostly |
| 18:43.18 | ajohnson | are you recording any of it to disk? |
| 18:43.23 | VJFROMGT | none |
| 18:43.26 | ajohnson | have you run top to see what the io load is? |
| 18:43.36 | VJFROMGT | minimal,, mostly idle |
| 18:44.04 | VJFROMGT | what do u know,, it is recording |
| 18:44.14 | VJFROMGT | sip.conf specifies not recording |
| 18:44.19 | VJFROMGT | what else can be doing this? |
| 18:44.34 | ajohnson | mixmonitor? |
| 18:44.36 | ajohnson | monitor |
| 18:44.43 | ajohnson | The rest would be done through extensions.conf |
| 18:44.58 | VJFROMGT | allright,, i will take a look at that,, tahnks |
| 18:45.02 | ajohnson | np |
| 18:45.21 | VJFROMGT | its just recording one extension though |
| 18:46.12 | ajohnson | well... |
| 18:46.19 | ajohnson | single processor or dual? |
| 18:46.44 | ajohnson | when you run top you will see a section that says: load average: 0.05, 0.01, 0.00 |
| 18:46.59 | VJFROMGT | single |
| 18:47.22 | VJFROMGT | Cpu(s): 14.0% us, 3.2% sy, 0.0% ni, 82.0% id, 0.8% wa, 0.0% hi, 0.0% si |
| 18:47.26 | zapp-branigan | hi i have a problem compiling the asterisk addons and i have installed the libtools @LIBTOOL@: command not found |
| 18:47.38 | ajohnson | What's the load average? |
| 18:48.48 | VJFROMGT | <PROTECTED> |
| 18:49.10 | ajohnson | ok, shouldn't be a problem if it is below 1.00 |
| 18:49.51 | VJFROMGT | so everything points to bandwidht now? |
| 18:50.01 | ajohnson | Specifically watch the load average and see if you see any spikes. Other than that, they system should be more than powerful enough to handle the transcoding |
| 18:50.11 | wwalker | I've read the 7 or 8 pages on meetme in the book and looked at the docs in the code, but I don't see any docs on the DTMF controls inside meetme. Where are those doc'd? |
| 18:50.26 | VJFROMGT | hmm |
| 18:55.39 | *** join/#asterisk dropshot (n=porkypig@adsl-69-107-86-17.dsl.pltn13.pacbell.net) |
| 18:55.44 | dropshot | morning |
| 18:55.55 | dropshot | i just came across asterisk while i was browsing the web |
| 18:56.23 | *** join/#asterisk roxy_ (n=roxy_@4.249.97-84.rev.gaoland.net) |
| 18:56.46 | roxy_ | ~seen dhenry |
| 18:56.48 | jbot | i haven't seen 'dhenry', roxy_ |
| 18:56.58 | roxy_ | ~seen ghenry |
| 18:56.59 | jbot | ghenry <n=ghenry@212.159.59.85> was last seen on IRC in channel #asterisk, 40d 4h 32m 52s ago, saying: 'Polycom ip501 a safe bet?'. |
| 18:58.27 | *** join/#asterisk engrxyz (n=fgsfgfs@82-34-18-23.cable.ubr03.basi.blueyonder.co.uk) |
| 18:59.30 | *** join/#asterisk NOT_guru (n=chatzill@24-241-103-142.static.stls.mo.charter.com) |
| 18:59.55 | roxy_ | what is the best ml to ask for help on asterisk ? (i want to incorporate ldap's rt module and would like to know how to do it) |
| 19:02.25 | Sweeper | man, theo is rubbing off on me |
| 19:04.34 | Aeudian | Is there a good web GUI that allows the ability to record voice menus, such as an auto attendant prompts? |
| 19:04.36 | *** join/#asterisk lesouvage (n=lesouvag@cc341200-a.assen1.dr.home.nl) |
| 19:06.47 | [TK]D-Fender | Aeudian: just use Record, or do them on a PC and convert the files to something * can use. |
| 19:06.55 | lesouvage | The one making an inbound call doesn't hear the phone ring but just silence while I use the r parameter in the dialstring en the cli output indicates that there is ringing going on. WIth an internal call it works fine. Any clue? (asterisk 1.4.11) |
| 19:07.57 | [TK]D-Fender | lesouvage: "r" = evil and should be avoided. |
| 19:08.06 | *** join/#asterisk kombi (n=kombi@213.160.14.18) |
| 19:08.12 | Aeudian | [TK]D-Fender: AsteriskNow has a gui that lets you create a file name and then the system calls a phone to record the message and when the user hangs up the message is saved? theres nothing else like that? |
| 19:08.14 | *** join/#asterisk saftsack (n=saftsack@pD9E07188.dip.t-dialin.net) |
| 19:09.10 | *** join/#asterisk kiscokid (n=Ron_Laut@adsl-216-101-109-187.dsl.snfc21.pacbell.net) |
| 19:09.25 | [TK]D-Fender | Aeudian: Sure. Go install FreePBX, or any of the other GUI's which all do the exact same thing. |
| 19:09.46 | [TK]D-Fender | Aeudian: or just do it in your own dialplan like the rest of us. |
| 19:09.55 | wwalker | so, is all of the IO muxing for meetme done in the zaptel driver? |
| 19:09.57 | lesouvage | [TK]D-Fender: I will try it without. Why is it evil, I have always used it (but until now noboby informed me about the evil nature of this parameter) |
| 19:10.26 | *** join/#asterisk guillote_GNU (n=bancaria@host185.190-136-203.telecom.net.ar) |
| 19:10.55 | [TK]D-Fender | lesouvage: "r" tries passing the ringing as AUDIO, not as an INDICATION state. If you're dealing with a channel where you normally don't have audio during that phase (like most SIP) you won't get anything. Its useful in only a select few scenarios. |
| 19:11.16 | [TK]D-Fender | wwalker: yup. |
| 19:16.45 | *** join/#asterisk saftsack (n=saftsack@pD9E07188.dip.t-dialin.net) |
| 19:17.36 | *** join/#asterisk nny (n=nny@64.203.239.83.static-pool-4.pool.hargray.net) |
| 19:17.40 | nny | well |
| 19:17.43 | kombi | what's the best way to move a caller into another context on pressing a key? |
| 19:17.49 | lesouvage | [TK]D-Fender: It's still complete silence when calling in. What should I do so the one calling in hears the phone instead of using the r parameter? |
| 19:18.25 | kombi | read into variable, then decide with gotoif? |
| 19:19.10 | lesouvage | Kombi: Using the Read cmd is one of the options followed by a GotoIf() |
| 19:19.27 | kombi | lesouvage: thanks! |
| 19:19.44 | nny | Me and our dev are at a wall with this issue. We call, the voicemail app says its playing "unavail", goes to vm-intro, lets me leave messages. All messages consist of either fast dial tone, steady dial tone or silence. When I call and someone picks up the line via SIP phone, all is normal. We had some shady weather here over the weekend. We tested btoh lines indepently of each other, same issue. I have the exact same system locally |
| 19:20.30 | nny | btw unavail is not heard on the line though |
| 19:20.39 | wwalker | how many simultaneous callers can meetme handle per conference? per machine? (assuming all g711, no tdm) |
| 19:21.03 | kombi | lesouvage: I had the suspicion there was something even simpler.. it is to give callers the option to leave a message while waiting in a queue |
| 19:21.07 | nny | we have tried different unavail messages and they are not heard on the call |
| 19:21.15 | Qwell | wwalker: somewhere between 1 and a million, depending on your hardware |
| 19:21.16 | denon | wwalker: that probably depends a lot on the machine |
| 19:21.31 | Qwell | If you've got a cray, probably somewhere on the top end of that |
| 19:21.40 | Qwell | if you have a 100mhz P1...maybe 1 |
| 19:21.48 | denon | tdm? probably 2 |
| 19:21.48 | denon | :) |
| 19:22.02 | lesouvage | wwalker: I have once did some testing with 30 concurrent calls. It depends on the quality of the connections used. |
| 19:22.04 | nny | Is it better for me to broadcast this issue in forums, here or an alternative wya? |
| 19:22.05 | nny | way* |
| 19:22.27 | denon | oh, no tdm |
| 19:22.39 | kombi | nny: try and narrow down the issue, try playback, record etc separately, try vm locally |
| 19:23.31 | nny | did a playback(/var/spool/asterisk/voicemail/atlantiatech/100/unavail) and it worked. |
| 19:23.38 | kombi | good! |
| 19:23.40 | *** join/#asterisk Bashtoni (n=sam@82-69-174-69.dsl.in-addr.zen.co.uk) |
| 19:23.42 | *** join/#asterisk Xarion (n=xarion@c1-34-6.rndf.isadsl.co.za) |
| 19:24.16 | Bashtoni | Has anyone had a problem where dialing out via PRI only works if the leading number is dropped (in this case a 0)? |
| 19:24.43 | kombi | Bashtoni: that must be something in your dialplan |
| 19:25.18 | roxy_ | what are the res_* file in the asterisk source ? are they part of asterisk or all separate module ? |
| 19:25.26 | Bashtoni | kombi: Dialplay is simply Dial(ZAP/g0/${EXTEN:1}) |
| 19:25.33 | kombi | nny: do more of those tests, crank up verbosity on cli, reproduce, isolate, reproduce |
| 19:25.35 | nny | kombi: can try Record |
| 19:25.45 | Bashtoni | kombi: If I remove the :1 so it dials the full number it doesn't dial out |
| 19:25.47 | kombi | Bashtony: and there you have it.. |
| 19:25.56 | kombi | remove :1 |
| 19:26.12 | Bashtoni | kombi: When I do that, it dials the inital digit, which *should* work, but doesn't |
| 19:26.31 | Bashtoni | kombi: Which is the whole problem.. |
| 19:26.54 | kombi | you mean it only dials out when you leave out the first digit? |
| 19:26.59 | Bashtoni | kombi: Yup |
| 19:27.02 | kombi | that is weird |
| 19:27.10 | Bashtoni | kombi: You're telling me :) |
| 19:27.31 | kombi | bri you say, have you tried a normal phone? |
| 19:27.39 | Bashtoni | kombi: PRI |
| 19:28.07 | Bashtoni | kombi: And not got anything but an Asterisk box with this Sangoma card to try with |
| 19:28.57 | kombi | Bashtoni: try the PRI connection in some other way to make sure where the issue is |
| 19:29.36 | *** join/#asterisk nny (n=nny@64.203.239.83.static-pool-4.pool.hargray.net) |
| 19:31.10 | roxy_ | is there a chance that might work: get the source for asterisk 1.2 (debian testing), add in it the file res_ldap, compile, add res_ldap.so the /usr/lib/asterisk. Would that give me ldap support ? |
| 19:31.11 | wwalker | Qwell: lesouvage: denon: assume a dual proc with 2 GB RAM and 2 AMD 2216s (2.4 GHz Opterons)? |
| 19:31.28 | zapp-branigan | hi i have compiled the ztdummy and when i do modprobe ztdummy work ok but when i load asterisk Unable to support trunking on peer without zaptel timing please is neeed something else ? |
| 19:33.34 | *** join/#asterisk nny (n=nny@64.203.239.83.static-pool-4.pool.hargray.net) |
| 19:33.37 | nny | sorry! |
| 19:33.52 | nny | pidgin SUCKS as an IRC client |
| 19:34.32 | nny | http://pastebin.com/m5776567 |
| 19:34.50 | nny | hopefully the IRC server kicked me before that sick flood of garbage |
| 19:35.31 | nny | that pastebin is DEBUG output, I have tested Record and Playback |
| 19:36.22 | nny | both Record and Playback work fine, this seems to be something with the voicemail app |
| 19:36.34 | *** join/#asterisk Mercestes (n=Merceste@c-76-30-153-176.hsd1.tx.comcast.net) |
| 19:36.44 | Mercestes | Helllloooooo #ASTERISK! |
| 19:36.49 | Mercestes | did ya miss me? |
| 19:36.49 | tristanbob | hello Mercestes |
| 19:36.53 | zapp-branigan | hi i have compiled the ztdummy and when i do modprobe ztdummy work ok but when i load asterisk Unable to support trunking on peer without zaptel timing please is neeed something else ? |
| 19:37.21 | [TK]D-Fender | Mercestes: Yes, but our aim is improving! |
| 19:37.34 | Mercestes | [TK]D-Fender, I believe in you. Keep practicing. |
| 19:37.41 | *** join/#asterisk thansen|laptop (n=thansen@19.243.sfcn.org) |
| 19:37.42 | *** join/#asterisk ReDNeQ- (i=ReDNeQ@cpe-66-68-37-190.austin.res.rr.com) |
| 19:37.42 | Mercestes | hey, got an interesting problem. |
| 19:37.54 | [TK]D-Fender | Mercestes: #drphil |
| 19:38.02 | Mercestes | I meant with asterisk. |
| 19:38.08 | nny | ha |
| 19:38.14 | funxion | Im trying to limit the maximum duration of outbound calls, I'm currrently using func absolutetimeout,seconds but this includes the inbound time not just the connected time does anyone have any suggestions as to how I could do this? |
| 19:38.19 | doughecka | heh |
| 19:38.32 | [TK]D-Fender | zapp-branigan: You have to recompile * AFTER zaptel to it picks up that its available |
| 19:38.35 | *** join/#asterisk Ebola (n=Ebola@host86-138-6-48.range86-138.btcentralplus.com) |
| 19:38.51 | zapp-branigan | i have compiled first zaptel |
| 19:39.11 | thansen|laptop | with agi, how can I do post processing after the party hangs up? is it possible? |
| 19:39.13 | Mercestes | Asterisk 1.4.5, when I record a voicemail, it says it plays unavail, but I hear nothign, it plays vm-intro, and that works, and I record a voicemail, but it just plays me either a: silence, b: a fast busy, or C:, a dial tone. |
| 19:39.15 | zapp-branigan | and second asterisk |
| 19:39.32 | [TK]D-Fender | zapp-branigan: do zaptel, then modprobe it, then comiple * and reinstall it. |
| 19:39.41 | Mercestes | I can do a playback(/var/spool/asterisk/voicemail/context/user/unavail) and that works. |
| 19:39.45 | zapp-branigan | :D |
| 19:39.47 | Sweeper | wow, asterisk-biz is just full of good fun today |
| 19:40.08 | Mercestes | I even did a record(/var/spool/asterisk/blah/blah/test) and a playback(/var/spool/asterisk/blah/blah/test) (blah equals the appropriate context and user of course) and that even works. |
| 19:40.10 | [TK]D-Fender | Mercestes: Actually I think I've recently herd exactly that from a few others in here recently. |
| 19:40.17 | nny | heh |
| 19:40.21 | Mercestes | [TK]D-Fender, any fixes? |
| 19:40.36 | [TK]D-Fender | Mercestes: Not that I'd heard. Could be in SVN.... |
| 19:40.49 | Mercestes | Well, it's 1.4.5 on ubuntu. |
| 19:41.02 | Mercestes | my half-assed response was "upgrade" but...can't promise that will fix anything. |
| 19:41.22 | Mercestes | this is on a 2 line analog setup soo....I'm no analog expert...it almost sounds like to me the lines are bleeding over (recording dialtone and all). |
| 19:41.40 | Mercestes | they can even hit # to end the recording of the dialtone. =/ and conversations are fine, main menu is fine, just voicemail debauchery |
| 19:42.16 | Mercestes | any brilliant ideas? |
| 19:42.51 | Sweeper | nothing constructive :D |
| 19:43.12 | *** join/#asterisk iBuMp- (i=ReDNeQ@cpe-66-68-37-190.austin.res.rr.com) |
| 19:43.43 | Mercestes | Sweeper, lol. |
| 19:43.48 | Mercestes | anyone got anything constructive I can try? |
| 19:43.50 | funxion | Im trying to limit the maximum duration of outbound calls, I'm currrently using func absolutetimeout,seconds but this includes the inbound time not just the connected time does anyone have any suggestions as to how I could do this? |
| 19:45.22 | *** join/#asterisk CunningPike (n=CunningP@dhcp-10-234.district.north-van.bc.ca) |
| 19:46.08 | Mercestes | Gah, nothing eh? |
| 19:46.18 | [TK]D-Fender | Mercestes: **downgrade** |
| 19:46.23 | Mercestes | 1.2.13? |
| 19:46.24 | Mercestes | >.> |
| 19:46.42 | [TK]D-Fender | Mercestes: As far as you wish to :) |
| 19:46.49 | roxy_ | [TK]D-Fender: do I have a chance to make res_ldap(in svn atm) work with 1.2 ? |
| 19:47.03 | [TK]D-Fender | roxy_: Of course. |
| 19:47.08 | Mercestes | Might give 1.2.17 a try |
| 19:47.14 | Mercestes | or CCM >.> |
| 19:47.18 | *** join/#asterisk potsboy (n=chrisg@vc-196-207-32-230.3g.vodacom.co.za) |
| 19:47.20 | nny | **note "shoud be shot" is deprecated in 1.4 and should be replace with *shot, they should be" |
| 19:47.27 | jfitzgibbon | funxion: show application Dial, look at the L() option |
| 19:47.35 | Mercestes | nny: LMAO |
| 19:47.40 | roxy_ | [TK]D-Fender: could you be kind enough to give me the basic step to follow ? |
| 19:47.41 | funxion | lol |
| 19:47.46 | funxion | thnx |
| 19:48.02 | [TK]D-Fender | nny: talks Mercestes does funny, hmmmMMMMMM!?!??!? |
| 19:48.06 | Mercestes | Warning: Shoot dev has been depcrecated in 1.4 and will be removed in future releases. Please use "core shoot dev" instead. |
| 19:48.14 | [TK]D-Fender | roxy_: Nope. |
| 19:49.24 | funxion | jfitzgibbon that looks like its just what I needed thanx |
| 19:50.58 | *** join/#asterisk pgarcia (n=root@shiva.kanatek.com) |
| 19:52.42 | zapp-branigan | [TK]D-Fender a lot of thanks now i can do trunk :D |
| 19:53.15 | mocker | $10/mo for 200G |
| 19:53.21 | mocker | I wonder if they'll suspend me |
| 19:56.45 | Sweeper | why would they? |
| 19:56.58 | Sweeper | I mean, the service really sucks... |
| 19:57.26 | *** join/#asterisk Arno[Slack] (n=hellSOUN@gre92-1-81-57-177-108.fbx.proxad.net) |
| 20:01.51 | roxy_ | most of the file in astrisk use ast_variable_new with 2 args bu res_ldap_config.c uses 3 args. How can I get around it ? |
| 20:03.37 | [TK]D-Fender | roxy_: those are questions for #asterisk-dev , not here |
| 20:04.07 | [TK]D-Fender | roxy_: and 1.2 is a dead end. I'd move on if I were you |
| 20:04.34 | Qwell | Did Apple just drop the price of the iphone by like $200? |
| 20:04.43 | ajohnson | Why, 1.2 is the only version stable enough to use... ;-) |
| 20:04.52 | [TK]D-Fender | Qwell: Yup |
| 20:05.01 | Qwell | crazy |
| 20:05.05 | Qwell | are they still locked to AT&T? |
| 20:05.11 | [TK]D-Fender | Qwell: Yup |
| 20:05.16 | Qwell | for now |
| 20:05.25 | Qwell | what else did they announce? |
| 20:05.28 | [TK]D-Fender | Qwell: "officially" |
| 20:05.30 | doughecka | fat nano |
| 20:05.37 | *** join/#asterisk DrAk0 (n=ljd@nelug/coreteam/luisjose) |
| 20:05.38 | doughecka | iphone minus phone |
| 20:05.38 | [TK]D-Fender | Qwell: http://www.engadget.com/ |
| 20:05.42 | doughecka | :) |
| 20:06.21 | roxy_ | [TK]D-Fender: ok, thanks |
| 20:07.29 | t3rror | jobs just f'd all of those fanboys |
| 20:09.18 | [TK]D-Fender | t3rror: how so? Gave them what they wanted/expected//// |
| 20:09.54 | doughecka | I want an iphone with 160GB drive |
| 20:10.10 | t3rror | the early adopters |
| 20:10.25 | [TK]D-Fender | I want the Rev 2 OpenMoko running *! |
| 20:10.29 | generalhan | doughecka: do you also want a forklift to carry it around ? |
| 20:10.33 | Qwell | [TK]D-Fender: buy me a rev 1 and it will |
| 20:10.42 | doughecka | pah |
| 20:10.45 | generalhan | lol |
| 20:10.50 | Qwell | (that offer goes for anybody) |
| 20:10.52 | [TK]D-Fender | t3rror: No he didn't.... they F'd THEMSELVES months ago. |
| 20:10.55 | doughecka | I already carry around a vx6700 with extended battery |
| 20:11.00 | doughecka | ANYTHING is smaller than that |
| 20:11.12 | generalhan | doughecka: i use that same phone ! :) i LOVE it despite the size |
| 20:11.15 | t3rror | there are basically 1 million customers who will mostly be heated to know they paid too much |
| 20:11.17 | doughecka | oh, it works |
| 20:11.23 | generalhan | you use verizon? |
| 20:11.29 | doughecka | yup |
| 20:11.36 | generalhan | the 6800 is coming out next month, i already have mine ordered ! |
| 20:11.39 | Qwell | t3rror: They are apple fanboys. They don't care that they paid more. |
| 20:11.42 | t3rror | i do like the ipod touch |
| 20:11.42 | doughecka | oh? |
| 20:11.57 | t3rror | Qwell: you are right |
| 20:12.02 | generalhan | yea, if you want to see it check out "the mogul" from sprint its the same phone |
| 20:12.07 | Nugget | Oh my god! the price of electronics goes down! Who could have predicted that? |
| 20:12.23 | [TK]D-Fender | doughecka: I helped a friend of mine upgrade his to WM6. 10x better he says |
| 20:12.27 | *** join/#asterisk luisjose (n=ljd@nelug/coreteam/luisjose) |
| 20:12.27 | t3rror | not $200 in 3 months |
| 20:12.29 | doughecka | ?!? |
| 20:12.33 | Qwell | 2 months. heh |
| 20:12.38 | doughecka | upgrade the xv6700 to wm6? |
| 20:12.43 | [TK]D-Fender | doughecka: thex v6700 |
| 20:12.46 | [TK]D-Fender | doughecka: yup |
| 20:12.47 | generalhan | [TK]D-Fender: thats what ive heard ... the 6800 is shipping with WM6 so im looking forward to playing around with it |
| 20:12.47 | doughecka | please tell! |
| 20:12.50 | doughecka | <PROTECTED> |
| 20:12.58 | Nugget | *shrug* |
| 20:13.08 | Nugget | sometimes it is $200 in 3 months. |
| 20:13.14 | *** join/#asterisk mtaht4 (n=m@200.62.111.173) |
| 20:13.26 | t3rror | Nugget: when was the last time that happened? |
| 20:13.33 | Nugget | fiik. why? |
| 20:13.39 | Kurin- | Is there a way either to play an audio file given its full path, rather than one of the files in the sounds directory, OR to play a given user's voicemail greeting? |
| 20:13.42 | [TK]D-Fender | doughecka: http://www.engadget.com/2007/08/31/cooked-winmo-6-rom-verified-for-ppc-6700-xv6700/ |
| 20:13.48 | [TK]D-Fender | doughecka: Merry Christmas |
| 20:13.57 | doughecka | awesome, thanks! :) |
| 20:14.28 | [TK]D-Fender | Kurin-: Yes, just SPECIFY the whole path |
| 20:14.49 | Kurin- | That didn't work, but maybe it was my agi program |
| 20:14.54 | [TK]D-Fender | Kurin-: Playback(/var/spool/asterisk/voicemail/default/100/unavail) |
| 20:15.04 | [TK]D-Fender | Kurin-: or any other file |
| 20:15.07 | Nugget | ask all the people who paid $2000 for a PS3 off ebay. Just tells me that Apple is better at pricing their product than Sony was. |
| 20:15.18 | funxion | tk remeber early my problem with GotoIf($[${LEN(${CALLERID(num)})} < 4]?47:33) well I never tested the other side of it which doesnt work now |
| 20:15.21 | funxion | I dont understand |
| 20:15.28 | funxion | now all calls go to 47 |
| 20:15.30 | Kurin- | Well I get "file.c:563 ast_openstream_full: File /var/spool/asterisk/voicemail/voicemail/114/greet.wav does not exist in any format" |
| 20:15.32 | [TK]D-Fender | funxion: You know what to do.... |
| 20:15.33 | Kurin- | except it does |
| 20:15.36 | t3rror | Nugget: that was different, sony wasn't charging $2k |
| 20:15.47 | Nugget | it's the same effect for consumers. |
| 20:15.57 | Nugget | like it said, it just tells me that apple is better at it than sony |
| 20:15.59 | Qwell | If you want early access to something, you are going to pay more |
| 20:16.01 | [TK]D-Fender | Kurin-: As you SHOULD well know, you CANNOT specify the EXTENSION on the file <------ |
| 20:16.06 | *** join/#asterisk DrukenLPY (n=jdumais@CPE000e08cb2a29-CM00137189cb0c.cpe.net.cable.rogers.com) |
| 20:16.20 | Nugget | iPhone pricing was spot on -- they just barely ran out when they released them and have been selling them as they're made since. |
| 20:16.24 | Nugget | now that's slowed, the price drops |
| 20:16.26 | [TK]D-Fender | Kurin-: (ditch the .wav) |
| 20:16.27 | Kurin- | I didn't know that |
| 20:16.30 | Kurin- | yes I see |
| 20:16.32 | [TK]D-Fender | :p |
| 20:16.52 | Nugget | that tells me that the iPhone costs what it's worth to people |
| 20:16.53 | t3rror | actually people realize that the phone sucks and they just wanted an ipod in that form dactor |
| 20:16.55 | Nugget | and that price is lower now |
| 20:17.04 | t3rror | they made it and are selling it lower than the iphone |
| 20:17.08 | Nugget | I like mine just fine. It's the best phone I've ever owned. |
| 20:17.11 | Qwell | t3rror: sounds like both announcements are win-win then |
| 20:17.24 | Qwell | I'll buy an iphone when I can use it with tmobile |
| 20:17.28 | [TK]D-Fender | I'm going to a local OpenMoko demo this weekend.... screw Apple :p |
| 20:17.30 | t3rror | for apple and steve it is win-win for the consumers, they got pwned |
| 20:17.39 | Qwell | [TK]D-Fender: get me one while you're there, kthx |
| 20:17.43 | Nugget | How did I get pwned? |
| 20:17.50 | Qwell | t3rror: how? The customers are getting exactly what they want |
| 20:17.52 | Nugget | I have a great phone and I feel I paid a fair price for it |
| 20:18.37 | Nugget | now they cost less -- that's great, it means more people will be able to buy one |
| 20:18.39 | funxion | Sep 5 23:24:58 DEBUG[22988]: pbx.c:1274 pbx_extension_helper: Launching 'NoOp' |
| 20:18.39 | funxion | <PROTECTED> |
| 20:18.43 | t3rror | and it was worth the premium you paid to use it for the past three months? |
| 20:18.43 | funxion | is what I get from |
| 20:18.51 | funxion | NoOp(${LEN(${CALLERID(num)})}) |
| 20:18.52 | Qwell | Nugget: which means more people can work on unlock hacks? :P |
| 20:18.59 | Nugget | It was worth what I paid for it, otherwise I wouldn't have bought it |
| 20:19.08 | Nugget | that's a sunk cost. |
| 20:19.17 | funxion | but I see callerid in cdr |
| 20:19.32 | ajohnson | caller id number I assume |
| 20:19.55 | *** join/#asterisk ToyMan (n=Stuart@user-160uamh.cable.mindspring.com) |
| 20:20.47 | codefreeze | funxion: the ANI is given preference in CDR records. If it's there, you get that... (2cents worth) |
| 20:21.29 | [TK]D-Fender | NoOp(CID is "${CALLERID(num)}" Length is "${LEN(${CALLERID(num)})}") |
| 20:21.35 | [TK]D-Fender | funxion: ^^^^^^^^ |
| 20:21.48 | [TK]D-Fender | funxion: have you validated whats coming IN? |
| 20:22.08 | funxion | yes |
| 20:22.23 | [TK]D-Fender | funxion: do as above and test |
| 20:22.58 | Slimey | ok... Does anyone want a patch to chan_sip to allow you to specify a string prefix to the callid (via a chanvar) ? |
| 20:23.26 | Slimey | or should I keep it as a local hack? |
| 20:24.01 | funxion | tk i get Executing NoOp("Zap/1-1", "CID is "" Length is "0"") in new stack |
| 20:24.22 | *** join/#asterisk sof78a (n=sof78a@atelka.info) |
| 20:24.38 | funxion | but like I said I see clid in cdr |
| 20:24.44 | codefreeze | funxion: so, try the above, but use ani instead of num... |
| 20:25.33 | generalhan | Everyone: how do you handle your inbound/outbound faxing ?? im researching my best options and i need some feedback on solutions others are working with ! im so tired of having a PRI bill from one company, and an analog bill from stupid qwest. |
| 20:25.50 | funxion | same results |
| 20:25.57 | funxion | -- Executing NoOp("Zap/1-1", "CID is "" Length is "0"") in new stack |
| 20:26.39 | Nugget | funxion: is it a PRI? I had this problem on our PRI -- the CID info was coming in a second after the call, so I had to put a wait(1) in the incoming dialplan to capture it |
| 20:26.40 | sof78a | Hi, we have a dual xeon 2ghz and we are trying to do many meetme , we have made a test and after 30-40 simultaneous plain gsm meetme the sound starts to get choppy , we are using asterisk 1.2.14 ... Is it normal if not is there a way to improve the voice quality |
| 20:27.07 | funxion | it is a pri |
| 20:27.09 | codefreeze | funxion: um,.... try ANI instead of ani |
| 20:27.50 | funxion | I think its just the version Im using CALLERIDNUM werks |
| 20:28.00 | funxion | I know its deprecated |
| 20:28.06 | codefreeze | funxion: are you using 1.2? |
| 20:28.23 | funxion | 1.0.2 |
| 20:28.31 | funxion | thats the problem |
| 20:28.57 | [TK]D-Fender | Oh god. |
| 20:29.00 | codefreeze | funxion: ooooooooh. Yeah. Check your docs. I don't think CALLERID() existed way back then. |
| 20:29.04 | Nugget | yow 1.0.2? |
| 20:29.07 | [TK]D-Fender | funxion: Yeah, forget that function |
| 20:29.12 | funxion | I plan to update to 1.4 |
| 20:29.24 | funxion | but dont have the time to update all the code atm |
| 20:29.25 | Nugget | Are you going to install indoor plumbing too? :) |
| 20:29.30 | funxion | haha |
| 20:29.45 | funxion | hey 1.0.2 has been stable as hell for me |
| 20:29.53 | [TK]D-Fender | Nugget: I'm IT for a distributer, I could get his set up cheap ;) |
| 20:29.56 | funxion | running for 2+years with no problems |
| 20:30.06 | codefreeze | funxion: that'll be a JUMP. scan your config files maybe 3 times before running on 1.4! |
| 20:30.22 | funxion | codefreeze I understand |
| 20:30.29 | [TK]D-Fender | ok, checkout time here. later all! |
| 20:30.35 | funxion | thnx tk |
| 20:31.05 | *** part/#asterisk dasuberdavid (i=david@nat/digium/x-12365217743545d7) |
| 20:31.15 | *** join/#asterisk dasuberdavid (i=david@nat/digium/x-12365217743545d7) |
| 20:31.52 | generalhan | does anyone handle faxes through asterisk ? or passed somewhere else via asterisk ? |
| 20:33.59 | syzygyBSD | generalhan: lots of people |
| 20:34.34 | syzygyBSD | I pass them all to a fax machine that passes them back to asterisk with error detection |
| 20:35.28 | generalhan | see i think i want to do some form of fax server. we waste so much money just for the toner in our 3 fax machines, that i would rather have the stored electronically and print them out on our monster printer that is designed to print that many pages a day |
| 20:35.45 | funxion | generalhan are you using sip or zap? |
| 20:35.57 | generalhan | currently ? |
| 20:36.02 | funxion | to handle faxes |
| 20:36.11 | generalhan | we have a PRI for phones, and Analog lines for fax machines |
| 20:36.19 | generalhan | 2 different providers |
| 20:36.32 | funxion | what are you trying to accomplish? |
| 20:37.05 | generalhan | a fax server solution that will allow me to port our fax numbers to our PRI lines and then pass them through asterisk to the fax server (or something similar) |
| 20:37.24 | funxion | fax server? are you trying to email your faxes? |
| 20:37.27 | generalhan | faxing is the 2nd most important aspect of this company though, so it NEEDS to be very reliable |
| 20:37.42 | funxion | or pass them to a fax machine? |
| 20:37.55 | generalhan | i want to stop using all of our fax machines |
| 20:38.09 | generalhan | except maybe for outbound faxes, in which case i would save only one |
| 20:38.40 | funxion | have you read the wiki? |
| 20:38.52 | funxion | http://www.voip-info.org/wiki-Asterisk+fax |
| 20:38.54 | *** join/#asterisk saftsack (n=saftsack@pD9E07188.dip.t-dialin.net) |
| 20:39.00 | funxion | tells you how to do pretty much that |
| 20:39.21 | generalhan | i have heard many horror stories about using Asterisk for faxing |
| 20:39.50 | syzygyBSD | I have heard many success stories |
| 20:39.51 | generalhan | which is why i was asking, to see if people have had good results (near perfect, if not perfect) using it that way. or if people are using a different solution |
| 20:40.06 | syzygyBSD | it really depends on how you have it setup |
| 20:40.14 | generalhan | hmmm |
| 20:40.35 | funxion | I'm using it that way but Im doing it on ABE and only for about a month now and it seems to be working just fine |
| 20:41.06 | syzygyBSD | I have found there are more issues when the faxes are sent over public phone lines, with some fax machines that don't have error checking, etc |
| 20:41.15 | generalhan | i work for a law firm ... we get 1000-2000 pages a day. most of them are "life and death" important, so i need to be sure that how ever i do this, that it will be 99.99% perfect ! lol |
| 20:41.41 | syzygyBSD | that is why I have them sent to a machine I control, that forwards it into asterisk. It is a way to get asterisk to reliably handle faxes for me |
| 20:41.47 | generalhan | well we have no problems on the old POTS lines, but it just seems so ... old school to me ! lol |
| 20:41.54 | t3rror | normally you pay a premium for 4 9's > thus keep paying for toner |
| 20:42.35 | generalhan | it sux so bad changing toner in 3 fax machines once a week, i want to shoot myself ... and i may |
| 20:42.36 | generalhan | lol |
| 20:43.37 | syzygyBSD | hmm, I wonder if just a fax/modem would work |
| 20:44.14 | t3rror | instead i would try to get your clients to start sending you documents electronically instead of in analog format |
| 20:44.28 | `Sean | generalhan save money get a thermal fax machine :) |
| 20:44.40 | generalhan | syzygyBSD: you know i tried using WinFax with a Brooktrout, and it worked great at first ... but then it seemed to start getting overloaded or something. every other fax just disconnected or something ... it was garbage ! |
| 20:45.07 | generalhan | haha electronic documents ... 75% of our clients cant spell electronic ! |
| 20:45.30 | *** join/#asterisk CoolGuy21 (n=Tilt@cpe-76-175-234-137.socal.res.rr.com) |
| 20:45.47 | CoolGuy21 | can i make asterisk authenticate using mac address also? |
| 20:46.58 | syzygyBSD | that would limit asterisk to only your LAN coolguy |
| 20:47.05 | CoolGuy21 | oh ok |
| 20:47.19 | `Sean | I wonder how much it would cost to get another version of spandsp developed for asterisk or callweaver |
| 20:47.45 | syzygyBSD | mac addresses are layer 2 I believe, and are not routed outside of a subnet.... |
| 20:48.11 | CoolGuy21 | how can i make it so the same extension can only have one session |
| 20:48.21 | CoolGuy21 | so the users cant login from 2 places with one extension |
| 20:48.33 | syzygyBSD | using sip? |
| 20:48.35 | CoolGuy21 | yes |
| 20:50.49 | CoolGuy21 | syzygyBSD any ideaS? |
| 20:51.03 | syzygyBSD | there is an option in sip.conf, finding the exact one |
| 20:52.00 | syzygyBSD | that, or there is only one allowed per sip registry anyway... |
| 20:52.05 | syzygyBSD | I think that is the case |
| 20:52.21 | syzygyBSD | when #2 registers it removes the previous one |
| 20:52.39 | CoolGuy21 | k |
| 20:52.42 | CoolGuy21 | how do i do that? |
| 20:53.02 | syzygyBSD | it already acts that way I believe |
| 20:53.27 | CoolGuy21 | it does? |
| 20:53.49 | *** join/#asterisk limbje (n=root@limbique.xs4all.nl) |
| 20:53.51 | limbje | hi |
| 20:54.17 | limbje | anyone can help me with my sip.conf? |
| 20:58.10 | shido6 | ? |
| 20:58.12 | shido6 | whats up? |
| 20:59.30 | *** part/#asterisk shido6 (n=shido6@204.126.120.132) |
| 21:00.19 | limbje | hi |
| 21:00.29 | limbje | yes... (switching consoles sorry) |
| 21:00.43 | limbje | can't get my asterisk register on my sip provider |
| 21:01.03 | limbje | can't see anything in logging/ sip show registry/ sip debug |
| 21:01.17 | *** join/#asterisk ygguh2 (n=ftp@ool-44c57e6c.dyn.optonline.net) |
| 21:03.10 | limbje | so i can't see anything whats wrong.. looks like he ignore my register command |
| 21:03.34 | ygguh2 | Im unable to load the ztdummy drivers, zaptel-1.4.1, asterisk-1.4.11, on redhat 2.6 fc4. modprobe ztdummy not found. |
| 21:03.45 | ygguh2 | any idea? |
| 21:04.05 | ygguh2 | [2A |
| 21:04.24 | limbje | is there a way to see any register attempts? |
| 21:05.17 | chemikk | why asterisk nor use unix socket with communication with postgresql? |
| 21:05.47 | *** join/#asterisk fujin_ (n=aj@unaffiliated/fujin) |
| 21:06.17 | ygguh2 | 2.6.17-1.2142_FC4, /lib/modules/2.6.17-1.2142_FC4/misc/ztdummy.ko, updatdb done |
| 21:06.38 | *** join/#asterisk CunningPike_ (n=arodgers@204.239.12.183) |
| 21:07.53 | *** join/#asterisk Lawbringer (n=Lawbring@212.183.136.194) |
| 21:08.05 | fujin_ | morning asteriskers |
| 21:08.10 | limbje | hi |
| 21:08.18 | limbje | 23:00 here :) |
| 21:09.06 | ygguh2 | insmod: error inserting 'ztdummy.ko': -1 Unknown symbol in module |
| 21:09.12 | ygguh2 | interesting |
| 21:10.33 | ygguh2 | I ran the make install yesterday |
| 21:10.54 | dasuberdavid | I would try running make distclean then running make and make install again |
| 21:11.38 | ygguh2 | im doing that now. |
| 21:13.05 | ber123 | any good links for how to monitor your SIP service provider to make sure they are functioning for termination/origination? |
| 21:13.51 | *** join/#asterisk xxoxx (n=haoyu@tor/regular/xxoxx) |
| 21:13.59 | *** part/#asterisk Olgem (n=Olgem@host-69-144-136-61.bln-mt.client.bresnan.net) |
| 21:14.17 | ygguh2 | interesting, im receiving the same unknown symbol in module error after the build |
| 21:14.52 | *** join/#asterisk kFuQ (n=somedude@c-67-185-112-20.hsd1.wa.comcast.net) |
| 21:17.16 | limbje | :S why can't i see anything if my register command is working :S |
| 21:17.25 | limbje | more.. why it's not working... |
| 21:18.25 | limbje | i can connect to my provider with a softphone (x-lite |
| 21:18.30 | *** join/#asterisk jfitzgibbon (n=chatzill@64.72.237.130) |
| 21:18.51 | limbje | i can call between phones (connected thru cisco 186 ata |
| 21:19.07 | ygguh2 | dasuberdavid, thanks for the help. I have to leave now. I'll try again later. |
| 21:19.17 | limbje | but i can't get my asterisk registered on my provider :S |
| 21:19.29 | *** join/#asterisk orvux (n=orvux@200.77.223.187.cable.dyn.cableonline.com.mx) |
| 21:20.36 | orvux | Hi everybody, some one know if there is a iax2 problem when you |
| 21:20.48 | orvux | connect a 1.2 with a 1.4 ??? |
| 21:21.11 | limbje | how to check the current version of asterisk? |
| 21:21.46 | limbje | found |
| 21:21.54 | limbje | running 1.2.13 |
| 21:21.59 | t3rror | do i really need to have zaptel installed if i am just using asterisk as a iax trunk to itsp and a sip connection to a local ata? |
| 21:22.59 | t3rror | or are the zaptel drivers just used for hardware that is installed in the server? |
| 21:23.33 | _ShrikE | t3rror: IAX trunking needs zaptel for timing |
| 21:23.34 | limbje | i know there is a ztdummy |
| 21:23.47 | limbje | only used for timing (correct me if i'm wrong) |
| 21:23.48 | t3rror | ok, i will recompile the kernel then |
| 21:24.05 | _ShrikE | thats right. Use ztdummy if you dont have a zaptel card. |
| 21:24.10 | t3rror | i didn't have the CC stuff modularized |
| 21:24.19 | limbje | how can i see if i have it? |
| 21:25.43 | limbje | i dont use iax... do i need ztdummy? |
| 21:26.01 | _ShrikE | zaptel is also required for meetme |
| 21:26.06 | jfitzgibbon | limbje: a timer is only *needed* for meetme and iax trunking |
| 21:26.23 | datachomper | How is PRI and CCS related? |
| 21:26.39 | limbje | tyvm |
| 21:28.17 | jfitzgibbon | datachomper: CCS is the framing on the line. PRI is the signaling protocol |
| 21:30.09 | *** join/#asterisk SA007 (n=sa007@ip565e0006.direct-adsl.nl) |
| 21:30.39 | *** join/#asterisk notoriousrab1982 (n=chatzill@207.47.34.74.static.nextweb.net) |
| 21:31.09 | notoriousrab1982 | anyone here had any experience connecting intertel - 6822 phones to asterisk - they appear to be SIP |
| 21:31.48 | SA007 | i'm busy with my phone again, turned on sip debug, and avaery reply i get from the busgetphone sip server is 'SIP/2.0 407 Proxy Authentication Required', but it is registering ok (that does give a 200 ok) |
| 21:31.59 | SA007 | avaery -> every |
| 21:34.33 | datachomper | jfitzgibbon, thanks. |
| 21:37.25 | *** join/#asterisk anthony[ (n=anthony@fl-71-49-118-147.dhcp.embarqhsd.net) |
| 21:37.29 | *** join/#asterisk jgomo3 (n=jgomez@200.62.25.82) |
| 21:37.58 | jgomo3 | Greetings. |
| 21:39.03 | jgomo3 | I'm developing a really small asterisk billing system on PHP basede on simples queries over cdr and two support tables |
| 21:40.45 | *** join/#asterisk NightMonkey (n=NightMon@pdpc/supporter/sustaining/NightMonkey) |
| 21:40.48 | CoolGuy21 | jgomo3 why not use A2Billing? |
| 21:42.51 | jgomo3 | CoolGuy21: The first impresion was that it was to complex for what we needed. The true reason, is becouse my boos |
| 21:43.38 | jgomo3 | CoolGuy21: But I understand him. What we need is so simple, that he tough we could do it (excuse my english) |
| 21:44.23 | jgomo3 | CoolGuy21: He is the owner of a callcenter. In other words: people come here, make a call and then they pay according to the time and destiny |
| 21:45.05 | CoolGuy21 | ah ok |
| 21:46.24 | *** join/#asterisk linagee (n=linagee@about/linux/staff/linagee) |
| 21:46.27 | *** join/#asterisk linagee_ (n=linagee@about/linux/staff/linagee) |
| 21:52.36 | *** join/#asterisk jesselang|laptop (n=jesse@h75-100-164-127.75-100.unk.tds.net) |
| 21:53.32 | jgomo3 | CoolGuy21: Still you think we did a good aproach? I mean: should we use A2Billing instead? |
| 21:54.10 | jesselang|laptop | Hello. I'm using AGI to dial a server via IAX using qualify=yes. I would expect the Dial() to return immediately when the server is unreachable, but it blocks, Dial() doesn't return. Could anyone help me? |
| 21:55.03 | jesselang|laptop | Dialing within the dialplan, I receive this message: |
| 21:55.05 | jesselang|laptop | [Sep 5 14:53:04] WARNING[4885]: app_dial.c:1106 dial_exec_full: Unable to create channel of type 'IAX2' (cause 3 - No route to destination) |
| 21:55.34 | *** join/#asterisk holiday42 (n=no@spike.wcta.net) |
| 21:55.36 | fujin_ | jesselang|laptop, it should go to n+1 |
| 21:56.37 | jgomo3 | The problem: I make a call from phone A to phone B. I don't answer on phone B. Phone B stop riginig. The sound of phone A change from the "waiting for answer" tone to the "Busy" tone. I hang phone A. The call is registered in cdr as Asnwered. |
| 21:56.44 | fujin_ | unless there is no timeout specified, obviously |
| 21:57.02 | jgomo3 | why? |
| 21:57.09 | fujin_ | not you |
| 21:57.11 | fujin_ | lol |
| 21:57.12 | jesselang|laptop | fujin_: in the dialplan, it does, but how can I get it to return from Dial () within the AGI, so it can continue execution? |
| 21:57.33 | fujin_ | not sure, I'm not familiar with AGI - haven't had to use it for anything yet. |
| 21:58.46 | fujin_ | AEL has proven more than capable for all of the advanced stuff I've tried to do |
| 21:58.46 | jesselang|laptop | fujin_: thanks for trying to help. |
| 21:58.54 | jesselang|laptop | Can anyone else help me? |
| 22:03.40 | *** join/#asterisk jwh (i=jwh@62.84.188.34) |
| 22:04.18 | jwh | Hi all |
| 22:05.57 | jwh | has the realtime database structure changed dramatically in 1.4? it doesn't even appear to be trying the database, no output from the console, even with debug+verbose |
| 22:06.01 | codefreeze | jgomo3: what version of Asterisk? |
| 22:06.20 | jgomo3 | codefreeze:2.2 |
| 22:06.37 | codefreeze | jgomo3: you mean 1.2, right? |
| 22:06.48 | *** join/#asterisk Koshatul (n=evangeli@ppp188-33.static.internode.on.net) |
| 22:07.39 | codefreeze | jgomo3: which exact version of 1.2? |
| 22:07.53 | jgomo3 | codefreeze: 1.2.18 |
| 22:08.38 | codefreeze | jgomo3: can you tell me exactly what the args to the Dial() app are? |
| 22:08.42 | SA007 | wow, major break here, just succeeded in dialing an external numer, and my cellphone actually ringing |
| 22:09.19 | SA007 | i just don't get any audio, what are the known remedies for that (no audio in both ways, no nat) |
| 22:10.22 | *** join/#asterisk fakhir (n=Fakhir@unaffiliated/fakhir) |
| 22:10.50 | jgomo3 | codefreeze: Prior, one more detail: The problem hapen when the call go over a TRUNK ZAP, but it works good over the SIP TRUNK |
| 22:11.39 | codefreeze | jgomo3: so, you are dialing out over an FXO card? |
| 22:11.55 | *** join/#asterisk ReD-MaN (i=daemon@CPE0002b38bce8b-CM0018c0b357cc.cpe.net.cable.rogers.com) |
| 22:11.55 | jgomo3 | codefreeze: Yes |
| 22:13.14 | *** join/#asterisk CrazyTux (n=CrazyTux@216-110-94-231.static.twtelecom.net) |
| 22:13.15 | codefreeze | jgomo3: the zaptel cards can't tell when things are answered at the far end. You got "Answered" in the cdr because the phone co. gave you dialtone. |
| 22:13.57 | CrazyTux | Has anyone here messed with polycoms and provisioning? I'm wondering if there is a simple way to access the polycom's mac address via some variable they provide to pass off via HTTP get method? |
| 22:14.06 | codefreeze | jgomo3: if you want more accurate outgoing call info, you need to use PRI or somesuch. |
| 22:15.21 | *** join/#asterisk tc3driver (n=huh@rrcs-67-52-113-254.west.biz.rr.com) |
| 22:16.07 | jwh | is no one able to confirm that there have been no major changes? |
| 22:17.13 | jwh | as I can't see anything in console/log files, even with debug on, for mysql/realtime queries, as there are clearly users in the database, but its returning not found |
| 22:17.17 | jgomo3 | codefreeze: Ok, I'm just digesting that right now... ;-( |
| 22:19.17 | fujin_ | CrazyTux, if they're local to the server (on the same LAN/broadcast) you could just arp the IP address |
| 22:19.28 | fujin_ | if not, I'm sure there is some variable. My spa942's make use of $MAC$ |
| 22:19.44 | jgomo3 | codefreeze: Thanx for the info, by the way |
| 22:19.54 | CrazyTux | fujin_, Yes, the SPA's have the $MAC, which is what I'm trying to find similiar for polycom. |
| 22:20.55 | *** join/#asterisk lancey (i=lancey@support.net1.cc) |
| 22:20.56 | lancey | hi all |
| 22:21.10 | lancey | is there a way to switch asterisk cdr generation to the old mode - single cdr per call? |
| 22:21.48 | codefreeze | lancey: 1.4, right? |
| 22:22.12 | jgomo3 | codefreeze: We can't afford a PRI, we have a two simples PSTN lines... how can I tell? Is there another way to tell if the destiny didn't answered. |
| 22:22.25 | jesselang|laptop | Can anyone offer a hand with this Dial () problem using AGI? |
| 22:22.27 | lancey | codefreeze: yup. it logs "s" and "h" cdrs for each call |
| 22:22.38 | *** join/#asterisk lincoln6e (n=lincoln6@ip68-227-216-225.dc.dc.cox.net) |
| 22:22.40 | lancey | and for each fork of it alsoo, if it dials multiple destinations |
| 22:23.42 | lancey | been searching on the voip-wiki, nothing found yet :/ |
| 22:24.08 | lincoln6e | hello newbie here, Digium card - works with PCI Express slots on Dell rack server? |
| 22:24.40 | fujin_ | depends on the server |
| 22:24.49 | fujin_ | our 2950s' have standard pci slots |
| 22:25.24 | codefreeze | jgomo3: I can't think of anything... zaptel has some options, but I've heard they aren't very... dependable... |
| 22:25.55 | lincoln6e | has anyone installed the TE212P on a Dell 1U rack server? |
| 22:26.11 | codefreeze | lancey: Yes, it's irritating. I've been thinking of a way to reduce the noise.... but haven't attacked it yet. |
| 22:26.52 | *** join/#asterisk Strom_C (n=strom@208.127.172.112) |
| 22:27.16 | codefreeze | lancey: although I've done some stuff in 1.6 (the team/murf/CDRfix5 branch) to calm things down... |
| 22:27.36 | codefreeze | er, /1.6/trunk/ |
| 22:27.57 | lancey | well, it might be useful to be this way, but i think this needs to be configurable and one being able to turn it off |
| 22:28.27 | lancey | okay, thanks, i've asked because i think i missed something |
| 22:28.43 | jgomo3 | codefreeze: Well, I'll try. Thanx for everythink |
| 22:28.50 | lincoln6e | fujin: was that a 3.3v (short) slot in your 2950? |
| 22:30.47 | jgomo3 | codefreeze: I'm playing with reverse polarity to see if it works |
| 22:32.57 | *** part/#asterisk SA007 (n=sa007@ip565e0006.direct-adsl.nl) |
| 22:33.12 | mcab | CrazyTux: new polycom software (2.2.0/4.0.0) have a setting that will have the Polycom include the MAC in the User-Agent string when making HTTP requests |
| 22:33.28 | *** join/#asterisk ZaVoid (n=zavoid@c-67-165-25-195.hsd1.ct.comcast.net) |
| 22:36.10 | *** join/#asterisk nighty^ (n=nighty@sushi.rural-networks.com) |
| 22:37.25 | fujin_ | lincoln6e, not sure sorry, standard pci cards work thoug |
| 22:39.01 | *** join/#asterisk frenzy (n=frenzy@unaffiliated/frenzy) |
| 22:39.12 | neverblue | Sep 5 15:00:48 WARNING[8224]: chan_iax2.c:710 jb_warning_output: Resyncing the jb. last_delay -311, this delay -14320, threshold 1622, new offset -4 |
| 22:41.38 | *** join/#asterisk MaartenB (n=Maarten@213-73-177-32.cable.quicknet.nl) |
| 22:41.52 | MaartenB | hello everyone |
| 22:42.07 | *** join/#asterisk keulin (n=cray@ifth-pdcr2.infotheme.fr) |
| 22:42.09 | MaartenB | I was wondering if someone could explain the automon function to me, I have not had succesfull results with it :( |
| 22:45.38 | fujin_ | any way to tell exactly what causes ' == Spawn extension (macro-queue_helpdesk, s, 7) exited non-zero on 'SIP/maxnetvoip-b5d40708''? |
| 22:45.45 | fujin_ | if it was a hangup, the channel dying |
| 22:48.29 | lincoln6e | fujin, thanks, what was the model of your Digium card? |
| 22:48.36 | *** join/#asterisk MaartenB_ (n=Maarten@h8441243087.dsl.speedlinq.nl) |
| 22:49.03 | jwh | hm, what am I doing wrong... I have sipusers with table 'sipusers' and sippeers as 'sippeers', registration for phones fails, but if both users and peers is set to the same table it works, does it fail if the first lookup fails, or am I doing something wrong? |
| 22:49.43 | *** part/#asterisk mtaht4 (n=m@200.62.111.173) |
| 22:50.38 | *** join/#asterisk NirS (n=NirS@87.68.158.123) |
| 22:50.42 | NirS | hello all |
| 22:51.35 | NirS | anyone here with Voicemail + ODBC experience ? |
| 22:52.06 | *** join/#asterisk cheGGo (n=cheGGo__@dslb-084-059-044-218.pools.arcor-ip.net) |
| 22:52.19 | cheGGo | hi there :o) |
| 22:52.29 | NirS | hi che |
| 22:56.49 | *** join/#asterisk lesouvage (n=lesouvag@cc341200-a.assen1.dr.home.nl) |
| 22:58.41 | *** join/#asterisk the_Goat (n=rsd@c-71-224-187-182.hsd1.pa.comcast.net) |
| 22:59.43 | *** join/#asterisk Ebola (n=Ebola@host86-138-6-48.range86-138.btcentralplus.com) |
| 22:59.44 | *** join/#asterisk zotz (n=zotz@24.244.163.157) |
| 22:59.56 | *** join/#asterisk the_Goat (n=rsd@c-71-224-187-182.hsd1.pa.comcast.net) |
| 23:01.12 | *** join/#asterisk grimsy (n=chatzill@203.14.171.102) |
| 23:03.13 | *** join/#asterisk darkskiez (n=mbryars@bb-87-81-62-203.ukonline.co.uk) |
| 23:05.57 | NirS | anyone here with Voicemail + ODBC experience ? |
| 23:06.42 | fujin_ | storing voicemail in a database just seems very wrong to me |
| 23:06.52 | fujin_ | if you could store them on a samba share, that'd be awesome |
| 23:07.23 | NirS | well, nothing prevents you from doing that |
| 23:07.28 | the_Goat | no odbc and voicemail exp sorry..... |
| 23:07.40 | the_Goat | anyone using cisco 7940/7960 phones with asterisk? |
| 23:07.45 | NirS | not here |
| 23:07.58 | MaartenB_ | does somebody know the difference betwen Monitor() and automon? |
| 23:08.16 | jwh | hm, do any of you guys use extensions in realtime; ie; using it to enable/disable dialling prefixes per context? |
| 23:08.24 | jwh | for multi user purposes |
| 23:08.24 | fujin_ | nothing gives me that functionality, though :) |
| 23:08.27 | the_Goat | i am having an issue when i park/transfer a call. when i pickup the cisco phone, from the call or xfr, the phone can transfer, but i can hear nothing out of the cisco phone |
| 23:09.51 | cheGGo | NirS, realtime integration? |
| 23:10.08 | NirS | no |
| 23:10.22 | NirS | I want to store voicemail messages on a mySQL resource |
| 23:10.30 | cheGGo | uhh |
| 23:10.34 | cheGGo | dirty =) |
| 23:10.38 | NirS | more or less |
| 23:10.52 | NirS | i'm trying to integrate Asterisk with an external resource that works with MySQL |
| 23:11.03 | NirS | and ODBC for voicemail makes the easiest choice |
| 23:11.17 | jwh | NirS: do you use mysql for sip/extensions at all? |
| 23:12.04 | NirS | no |
| 23:12.30 | jwh | :( |
| 23:12.38 | jwh | I can't find a sensible way to do it |
| 23:12.56 | jwh | as there is potential for asterisk to pull extensions out of the database in the wrong order |
| 23:13.17 | jwh | but my extensions.conf is getting out of hand, currently at about 2500 lines |
| 23:13.44 | jwh | unless I just have a catchall for everyone and then just match based on user, but not sure how to go about it |
| 23:18.20 | *** join/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 23:21.05 | Nichtwirklich | jwh: extension.conf is a mess |
| 23:21.37 | jwh | it is that |
| 23:21.41 | *** part/#asterisk jgomo3 (n=jgomez@200.62.25.82) |
| 23:22.03 | Nichtwirklich | after a while, they all look like basic on a c64 |
| 23:22.03 | *** part/#asterisk jesselang|laptop (n=jesse@h75-100-164-127.75-100.unk.tds.net) |
| 23:22.09 | jwh | hehe |
| 23:22.27 | jwh | the plan behind what im trying to do, is enable dialling prefixes per context, which is actually per user |
| 23:22.34 | jwh | ie; each user is dumped into their own context |
| 23:23.00 | rvhi | hi, has anyone used ragi? |
| 23:23.13 | jwh | without having to sit and copy/paste into the file manually, and reload |
| 23:23.18 | lincoln6e | the Goat: yes we are, Asterisk with Cisco 7940 |
| 23:23.42 | Nichtwirklich | do you have an example? |
| 23:24.03 | jwh | Nichtwirklich: sorry, are you asking me? |
| 23:24.24 | Nichtwirklich | jwh: yes, I don't have a Cisco phone ;-) |
| 23:24.31 | jwh | oh |
| 23:24.36 | *** join/#asterisk ReD-MaN (i=daemon@CPE0002b38bce8b-CM0018c0b357cc.cpe.net.cable.rogers.com) |
| 23:24.58 | jwh | I meant generally, are you trying to use a cisco phone with asterisk or? |
| 23:25.13 | Nichtwirklich | <jwh> ie; each user is dumped into their own context |
| 23:25.17 | Nichtwirklich | <Nichtwirklich> do you have an example? |
| 23:25.46 | jwh | oh, well the end user equipment wouldn't matter, what exactly are you after an example of? |
| 23:25.50 | jwh | im somewhat confused |
| 23:26.17 | Nichtwirklich | ... enable dialling prefixes per context ... each user is dumped into their own context |
| 23:26.21 | jwh | oh |
| 23:26.52 | jwh | like, just specify context=username in sip.conf, as for dialling prefixes, ie; _555X., _800X. etc |
| 23:27.15 | jwh | obviously not the above as i'm .uk, but the principal is the same |
| 23:27.42 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 23:28.01 | Nichtwirklich | for home or office? |
| 23:28.07 | jwh | carrier |
| 23:28.18 | Nichtwirklich | okay, so _really_ big |
| 23:28.27 | jwh | stupidly big |
| 23:28.27 | jwh | yes |
| 23:28.50 | jwh | just need something abit more practical than flatfiles |
| 23:29.22 | jwh | as I want the customers to be able to do things themselves, rather than having to get staff to manually make changes |
| 23:29.40 | Nichtwirklich | user 45433 gets an own context, how to dial 0800 number (i think in the eu, the are 0900 now) |
| 23:29.59 | fujin_ | 0800 is freecall, 0900 is charge, here in NZ |
| 23:30.09 | jwh | ok so, instead of specifing a default outbound extension (for example you would use _X.) |
| 23:30.12 | nny | HA |
| 23:30.22 | nny | just figured out our voicemail issue |
| 23:30.30 | *** join/#asterisk fujin (n=aj@unaffiliated/fujin) |
| 23:30.32 | nny | if anyone was here when Mercestes was looking arounf |
| 23:30.35 | nny | around* |
| 23:30.42 | Nichtwirklich | jwh: yes ... |
| 23:30.44 | jwh | you specify _0800X. and they can only dial those numbers |
| 23:30.59 | Nichtwirklich | jwh: still yes |
| 23:31.12 | jwh | so you can restrict what that user dials, which isn't possible in mysql without some sort of ordering |
| 23:31.24 | jwh | from what I can see |
| 23:31.38 | jwh | for example, if asterisk pulls in 07X before 070X |
| 23:31.43 | jwh | it will patch 07X |
| 23:31.46 | jwh | err, match |
| 23:31.57 | Nichtwirklich | it will |
| 23:32.05 | jwh | which is the real problem |
| 23:32.14 | Nichtwirklich | okay normalisation, basic course |
| 23:32.24 | Nichtwirklich | table one: |
| 23:32.54 | Nichtwirklich | prefix-id, prefix, prefix-sort-order |
| 23:32.58 | Nichtwirklich | table two: |
| 23:33.14 | Nichtwirklich | user-id, prefix-id (which he gets) |
| 23:33.29 | jwh | well, thats it, there doesn't appear to be a way to tell asterisk any order, or process for selecting |
| 23:33.41 | Nichtwirklich | "table" three is a view |
| 23:34.11 | jwh | unless asterisk did lots of queries, to try a more specific extension first, but that would be alot of overhead |
| 23:34.40 | jwh | hm |
| 23:35.18 | Sweeper | jwh: use adhearsion, abandon the heathen ways of extensions.conf ;) |
| 23:35.27 | Nichtwirklich | select prefix from table1, table2 where table1.prefix-id=table2.prefix-id and where user-id=$USER order by prefix-sort-order |
| 23:35.45 | jwh | yeah |
| 23:35.54 | *** part/#asterisk orvux (n=orvux@200.77.223.187.cable.dyn.cableonline.com.mx) |
| 23:36.04 | Nichtwirklich | something like this in a few seconds, I mean this was free and not a 2k EUR consulting ;-) |
| 23:36.09 | jwh | Sweeper: ooh, that loks quite nice |
| 23:36.12 | jwh | looks* |
| 23:36.26 | jwh | Nichtwirklich: hm, I need to play with it I think |
| 23:36.38 | fujin_ | use AEL, abandon the unheathen ways of extensions.conf AND ruby |
| 23:37.00 | Sweeper | does AEL have an ORM? |
| 23:37.12 | jwh | well, ideally I want it pulled from mysql on demand, to avoid reloads/changes/whatever |
| 23:37.18 | Sweeper | or....good syntax? |
| 23:37.22 | fujin_ | orm, no |
| 23:37.25 | fujin_ | good syntax, yes |
| 23:37.26 | Sweeper | jwh: adhearsion has diaplan caching ;) |
| 23:38.09 | Nichtwirklich | he still has the problem, that he has to write down somewhere the order of his extensions, and with maybe 1000 prefixes, mysql should be still pretty fast |
| 23:38.17 | jwh | yeah |
| 23:38.22 | jwh | its ordering, as you do in the extensions |
| 23:38.34 | jwh | (file) |
| 23:38.58 | jwh | I would assume the ordering is done by uniqueid, but that doesn't really help if you are removing/adding prefixes |
| 23:39.13 | jwh | as obviously when you add an extension, it will have a higher ID |
| 23:39.17 | Nichtwirklich | prefix-id, prefix, prefix-sort-order |
| 23:39.30 | jwh | yeah |
| 23:39.44 | Nichtwirklich | first is unique id, prefix is 0800, sort order is 200 (whatever) |
| 23:39.44 | jwh | need to modify asterisk a little I think, unless there is a way to specify the query? |
| 23:39.53 | jwh | or am I being blind |
| 23:40.09 | Nichtwirklich | for this you should use the view |
| 23:40.20 | fujin | yeah, make a view that asterisk selects from |
| 23:40.28 | fujin | specifying the query in the client is the wrong place to do it |
| 23:40.41 | Nichtwirklich | if you store the view in the db, from asterisk its just select * from table3 where user-id = 45665 |
| 23:40.47 | jwh | hm |
| 23:40.52 | fujin | learn2database |
| 23:41.02 | jwh | hadn't thought of doing it like thart |
| 23:41.02 | nny | Sweeper: so, local telco implements Centrex VM last week, it answers EXACTLY the same time as * |
| 23:41.04 | jwh | that* |
| 23:41.20 | *** join/#asterisk saftsack (n=saftsack@ip-90-187-79-190.web.vodafone.de) |
| 23:41.21 | Nichtwirklich | jwh: do you have a user three lions on a shirt? jk |
| 23:41.27 | nny | Sweeper: Mercestes and me were banging our heads as to why vm was broken |
| 23:41.30 | jwh | Nichtwirklich: lol |
| 23:41.38 | nny | seems telco was beating us to the call by .00001 ms |
| 23:41.49 | nny | so asterisk got sloppy seconds, and a dial tone |
| 23:42.05 | nny | diagnosed: tried it with asterisk *off*... something was picking up the line |
| 23:42.19 | Nichtwirklich | good luck against israel and extonia ... YOU'LL NEED IT ;-) |
| 23:43.07 | nny | mind you the system directly would show the vm-intro etc. in console as the telco's prompt played, which, is also the same as * |
| 23:43.10 | nny | wow |
| 23:43.12 | nny | beer time |
| 23:44.51 | Nichtwirklich | yeah |
| 23:45.01 | *** part/#asterisk nny (n=nny@64.203.239.83.static-pool-4.pool.hargray.net) |
| 23:45.42 | *** join/#asterisk Yourname` (n=chatzill@76-10-128-178.dsl.teksavvy.com) |
| 23:46.01 | *** join/#asterisk [TK]D-Fender (n=Joe@64.235.216.2) |
| 23:48.31 | saftsack | has someone a a500 card? |
| 23:48.33 | saftsack | from sangoma? |
| 23:49.18 | Sweeper | nny: amazing |
| 23:49.33 | luisjose | i need an sniffer |
| 23:49.37 | *** join/#asterisk ReD-MaN (i=daemon@CPE0002b38bce8b-CM0018c0b357cc.cpe.net.cable.rogers.com) |
| 23:49.40 | luisjose | a sniffer even |
| 23:53.36 | Sweeper | luisjose: wireshark or tcpdump |
| 23:54.06 | luisjose | Sweeper, no X and tcpdump doesnt look like give me any information just do nothing when i set the host |
| 23:54.33 | Sweeper | then you're doing it wrong, or there's nothing there :P |
| 23:55.14 | luisjose | Sweeper, doing it wrong im sure |
| 23:55.27 | luisjose | tcpdump -vvv -ttt host <the ip> |
| 23:55.30 | luisjose | but it says nothing |
| 23:56.22 | Sweeper | might want to specify an interface ;) |
| 23:59.01 | luisjose | Sweeper, good, i have activity now |
| 23:59.09 | luisjose | Sweeper, but how i can get contents? |
| 23:59.15 | luisjose | im getting like just headers |