| 00:00.30 | generalhan | i just dont understand why it was happening to begin with ... and now changing something that should have no effect has magically seemed to help |
| 00:01.03 | JT | have you checked with your provider as to whether or not they are having technical difficulties? |
| 00:01.59 | generalhan | yes... but they are freaking liars ... when i had an iccodent a while back that i had someone come in and look at for me ... they PROVED it was the provider and they claimed everything was perfect ... so i cant count on them to fess up to anything |
| 00:02.15 | JT | heh ok |
| 00:02.48 | JT | make sure there's no interrupt sharing |
| 00:03.22 | generalhan | JT: how would i go about that ? |
| 00:03.35 | JT | cat /proc/interrupts |
| 00:04.18 | *** join/#asterisk DrAk0 (n=ljd@nelug/coreteam/luisjose) |
| 00:04.44 | *** join/#asterisk drako (n=ljd@nelug/coreteam/luisjose) |
| 00:06.34 | generalhan | JT: http://generalhan.pastebin.ca/649286 |
| 00:06.41 | generalhan | looks ok to me |
| 00:07.16 | JT | looks fine there |
| 00:08.00 | generalhan | JT: ok ran it for a while .... |
| 00:08.19 | generalhan | JT: and saw something under what you told me was bad: |
| 00:08.20 | generalhan | Best: 100.000000 -- Worst: 99.963379 -- Average: 99.987335 |
| 00:09.03 | JT | did you see it, or did it only show up in the totals? |
| 00:09.45 | generalhan | only in the totals ... i have been watching it and never saw that number anywhere |
| 00:10.15 | JT | so have there been any T1 flaps during this whole time? |
| 00:10.53 | generalhan | one person said they dropped a call ... but i typically write that off as someone hanging up on them :) so im not sure |
| 00:11.03 | JT | ... |
| 00:11.05 | generalhan | anytime it has happened before there was like 10 people that said it all at once |
| 00:11.12 | JT | it's easy to see if the T1 flaps |
| 00:11.15 | JT | watch the console. |
| 00:11.22 | generalhan | i was watching the zttest |
| 00:11.28 | JT | well do both |
| 00:11.32 | JT | or you're wasting time |
| 00:11.46 | generalhan | and what am i looking for in console ? |
| 00:12.03 | JT | a screen full of Red Alarms and D channel errors |
| 00:12.06 | JT | should be hard to miss |
| 00:12.42 | generalhan | well that has not happened since i was running that, else messages would have caught it |
| 00:13.03 | JT | then the problem hasn't occured |
| 00:14.15 | generalhan | right ... so should i just keep zttest running for hours and see if i get the red alarms again ? |
| 00:14.55 | *** join/#asterisk GoldFingaz (n=whoohoo1@bas13-ottawa23-1088841481.dsl.bell.ca) |
| 00:15.26 | generalhan | i just got a 99.975, but no "T1 flapping" : 99.987793% 100.000000% 99.975586% 99.987793% 99.987793 |
| 00:15.27 | GoldFingaz | hi...what is the difference between asterisk 1.4.10 and 1.2.24 |
| 00:15.37 | GoldFingaz | can seem to find any docs on it |
| 00:16.56 | GoldFingaz | anyone? |
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| 00:17.10 | *** mode/#asterisk [+o mog] by ChanServ |
| 00:18.04 | JT | 99.975 is just okay |
| 00:18.47 | JT | generalhan: i guess so, it's ideal if you're able to check zttest while it fails |
| 00:19.04 | JT | generalhan: but at this point i'm thinking it's a provider issue |
| 00:19.18 | generalhan | ok .. well ill keep watching ... but of course its not going to fail while im paying attention :( |
| 00:19.31 | JT | may be worth seeing if you have any LEDs on your SHDSL modem/smartjack |
| 00:19.34 | GoldFingaz | am i in the right place for asking newbie questions? |
| 00:19.36 | JT | when it has a red alarm |
| 00:20.05 | JT | GoldFingaz: 1.4 is the newer branch, but 1.2 is slightly more stable |
| 00:20.16 | generalhan | JT: im on vacation right now .. im 1000 miles from the office. i will be back in the office on thursday and i will check then |
| 00:20.24 | JT | ah ok |
| 00:20.28 | *** join/#asterisk techie (n=techie@c-67-181-184-170.hsd1.ca.comcast.net) |
| 00:21.19 | GoldFingaz | thanks |
| 00:29.16 | ChulJin | 252 other people and I'll still wind up talking to myself, but worth a shot: |
| 00:29.29 | ChulJin | anyone else use res_jabber (and possibly chan_gtalk)? |
| 00:30.10 | ChulJin | I got it going last night, and it ran continuously since just before noon today, now the connection is flapping at like .5Hz |
| 00:34.27 | *** join/#asterisk ltdwk (n=z@203-173-10-9.perm.iinet.net.au) |
| 00:37.04 | JT | ChulJin: probably no-one uses it |
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| 00:39.03 | snuff-work | anyway to ask if a phone is busy without using 'dial' ? |
| 00:39.45 | JT | chanisavail or something |
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| 00:42.14 | snuff-work | nah chanisavail just checks can it be reached :( |
| 00:42.43 | CoaxD | blah. |
| 00:42.51 | generalhan | snuff-work: chanisavail returns information about the status of that line. it will return BUSY if it is in use |
| 00:43.31 | generalhan | snuff-work: i use it constantly in my dialplan to "disable" call waiting on my user's phones |
| 00:47.04 | snuff-work | mm.. |
| 00:47.30 | snuff-work | must have done somethin not quite right in my tests |
| 00:49.19 | generalhan | snuff-work: look into the s option "s - Consider the channel unavailable if the channel is in use at all" |
| 00:49.36 | snuff-work | ah probably missing that |
| 00:49.43 | generalhan | that means even if its currently ringing, its not avail |
| 00:49.44 | *** join/#asterisk sysreq (n=sysreq@modemcable064.127-81-70.mc.videotron.ca) |
| 00:50.23 | generalhan | JT: im going cross-eyed staring at the CLI and the zttest output ! lol. and of course not one single hiccup yet |
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| 01:06.06 | *** join/#asterisk CrashHD (n=crashhd@70.96.98.65) |
| 01:06.35 | CrashHD | Hello, Anyone know where I can find a visual of the voicemail tree in asterisk? |
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| 01:52.39 | x86 | heya |
| 01:52.57 | x86 | anyone know of a decent SIP IM app for Mac? |
| 01:55.51 | x86 | Adium is supposed to support it, but wont connect to Asterisk |
| 02:09.49 | SwK | ichat? |
| 02:12.41 | *** join/#asterisk CVirus (n=GoD@196.218.101.88) |
| 02:14.27 | x86 | ichat can talk SIP? |
| 02:14.38 | x86 | iDidnt know that ;-) |
| 02:14.40 | SwK | how do you think it does video? |
| 02:14.46 | x86 | h323? |
| 02:15.13 | SwK | SIP |
| 02:15.25 | x86 | so i can add a SIP account to iChat? |
| 02:16.02 | x86 | only supported account types are AIM and .Mac |
| 02:16.12 | x86 | how do you add a SIP account? |
| 02:16.26 | SwK | well it also does jabber, and bonjour other then AIM and .mac |
| 02:16.32 | SwK | can you directly add a sip account dunno |
| 02:16.59 | SwK | but there are things like xMeeting but I dunno which ones do SIMPLE |
| 02:17.24 | x86 | yeah the only SIP/SIMPLE clients I can find requre a XMPP account |
| 02:17.41 | x86 | except the Adium plugin, which doesn't work |
| 02:17.56 | SwK | so use XMPP |
| 02:18.40 | x86 | really wanted to use a pure SIMPLE client |
| 02:19.26 | SwK | its over rated |
| 02:19.31 | SwK | eh |
| 02:19.33 | SwK | heh |
| 02:19.45 | SwK | (sorry nothing personal thats jsut the way I feel) |
| 02:20.28 | x86 | heh |
| 02:20.46 | Nivex | much love for XMPP |
| 02:20.47 | Nivex | :) |
| 02:21.09 | SwK | much love or what ever you want to pay me to do :P |
| 02:22.29 | x86 | ok, next question -- anyone know of a free Java applet (as in web applet) that acts as a SIP client? |
| 02:23.07 | SwK | i wish |
| 02:23.20 | SwK | i want to find one in flash personally |
| 02:24.25 | Nivex | IAX would be easier to work with IMHO |
| 02:24.56 | *** join/#asterisk mihinomenest (i=FV2C@66.255.220.17) |
| 02:25.00 | SwK | no |
| 02:25.11 | SwK | i refuse to run iax on my network |
| 02:25.23 | *** join/#asterisk jcaceres (n=jcaceres@190.41.82.1) |
| 02:27.02 | *** join/#asterisk [T]ank (n=ckwall@c-71-195-196-198.hsd1.ut.comcast.net) |
| 02:27.19 | x86 | SwK: that sucks for you then ;) |
| 02:27.31 | x86 | IAX2 is superior to SIP imho |
| 02:27.33 | SwK | what sux for me? refusing to run IAX? |
| 02:27.36 | SwK | hah |
| 02:27.43 | SwK | tyou have never ran it in a very large scale network |
| 02:27.49 | x86 | single port, less overhead, easier with NAT, etc, etc |
| 02:28.02 | [T]ank | i have an iax provider coming into my phone system. I am noticing that I do not get callwaiting caller id. is there a setting for that in iax.conf? i am not finding any information that supports that. |
| 02:28.04 | SwK | true, false, false, etc etc |
| 02:28.55 | SwK | [T]ank, you should be getting it by default there is no setting for it... and call-waiting is a function of your end device (ata sip phone whatever) not a function of the trunks coming in via sip/iax whatever |
| 02:29.03 | jcaceres | hello i have installed asterisk successfully in debian, but against my other experiences in this week this time asterisk did not create a file in /etc/int.d/ to start automatically asterisk and zaptel, even after rebooting......??? any idea of how could have change this time? |
| 02:29.34 | [T]ank | ok... so i am pushing my calls out to zap for my analog phones. Is that where it would need to be set? |
| 02:29.44 | jcaceres | why is it? |
| 02:29.59 | *** join/#asterisk phix (i=threat@60.240.43.214) |
| 02:30.19 | SwK | [T]ank, yes that is correct check your zap configs |
| 02:30.39 | JT | x86: there is no real superiority of iax over sip |
| 02:30.43 | phix | hey, does version 1.2.13 allow me to change the rfc2833 payload type to 96 in a configuration file yet? |
| 02:30.46 | JT | especially on large scale networks/ITSPs |
| 02:30.50 | phix | or do I need to patch asterisk? |
| 02:31.29 | phix | JT: I call being able to trunk calls superior |
| 02:31.44 | SwK | phix: still not superior in large scale networking |
| 02:31.50 | phix | JT: and handle NAT well also superior |
| 02:32.00 | phix | SwK: ok |
| 02:32.06 | JT | phix: it's not superior on a large scale |
| 02:32.06 | jcaceres | any body has experience with debian an asterisk? |
| 02:32.10 | [T]ank | thnks SwK |
| 02:32.20 | JT | and anyone can make sip work over most NAT unless they are an idiot |
| 02:32.23 | SwK | phix: you can get the same 'reduction in bandwidth' by adjusting the samples per RTP packet to something other then 20ms |
| 02:32.29 | SwK | and what JT just said |
| 02:32.36 | drako | jcaceres, a bit, yes |
| 02:33.02 | SwK | not to mention with SIP signal path vs media path are optionally the same thing... you can never say taht about iax... |
| 02:33.11 | phix | JT: heh, yes but it is easier to setup NAT with AIX than SIP |
| 02:33.23 | JT | AIX zomg IBM UNIX |
| 02:33.29 | phix | JT: any way, change of subject a bit, changing the rfc2833 payload type to 96 |
| 02:33.30 | SwK | sure you can native transfer, but you always start out with the media going thru the call switch box meaing higher loads on the call router |
| 02:33.33 | jcaceres | drako, i have installed asterisk, but this time it did not create the file to load it automatically, what do you think that happened there? |
| 02:33.47 | drako | jcaceres, 1.4 ? |
| 02:33.50 | jcaceres | yes |
| 02:33.56 | drako | make config |
| 02:33.57 | drako | ? |
| 02:34.05 | jcaceres | the aste version, i just downloaded |
| 02:34.20 | JT | phix: almost no difference in setup difficulty over NAT from an end user perspective |
| 02:34.36 | jcaceres | is it the same as make menuconfig? or make menuselect? |
| 02:35.01 | jcaceres | i normally do |
| 02:35.02 | drako | jcaceres, just type make config on the asterisk src directory |
| 02:35.16 | *** join/#asterisk ltdwk (n=z@203-173-10-9.perm.iinet.net.au) |
| 02:35.38 | jcaceres | .configure, then make menuconfig, then make, an finally make install |
| 02:35.48 | jcaceres | i'll follow your steps |
| 02:35.54 | drako | make config |
| 02:35.57 | phix | hmmm |
| 02:36.21 | drako | it will install a rc start up script on /etc/init.d |
| 02:37.05 | phix | how does licences work with the G729 codec? |
| 02:37.13 | phix | it reads some file? |
| 02:38.53 | jcaceres | btw, which kind of g729 codec is the one that can used free of charge ? |
| 02:39.33 | *** part/#asterisk [T]ank (n=ckwall@c-71-195-196-198.hsd1.ut.comcast.net) |
| 02:40.04 | SwK | jcaceres, in the US G729 is patented... using the "free ones" is illegal and discouraged... i |
| 02:41.07 | *** join/#asterisk Kumbang (n=kumbang@router-wmi.paume.itb.ac.id) |
| 02:41.11 | jcaceres | SwK, yes, i know, but i heard that there was a kind of g729 that could be used for educational purposes |
| 02:41.36 | SwK | intel has one |
| 02:42.10 | SwK | but its been my experience that people asking for that sort of thing want more then just educational use... if you want that just spend $20 on digiums website and order 2 licenses |
| 02:42.33 | phix | hmmm |
| 02:42.43 | phix | am I asking stupid questions? or am I just being ignored? |
| 02:42.44 | jcaceres | i do not know why is it proprietary, i thought CELP was free to be used |
| 02:43.34 | SwK | phix: go read the g729 info on digiums website its explained there |
| 02:43.58 | SwK | jcaceres, CELP might be but G729 and G723 are convered by numerous patents |
| 02:45.04 | phix | SwK: no I am refering to my question regarding changing the rfc2833 payload type to 96 |
| 02:45.23 | jcaceres | and finally who patented, digium? |
| 02:45.27 | SwK | phix: oh ... sorry... probably requires a patch |
| 02:45.38 | phix | SwK: would you know where this patch is? |
| 02:45.50 | SwK | jcaceres, digium does not own the patents to 729 thats owned by several people who set up a consortium to license it... |
| 02:46.04 | SwK | phix, nope... its usually easier to change the equipment you are talking too |
| 02:46.07 | *** join/#asterisk CVirus (n=GoD@196.218.101.88) |
| 02:46.27 | *** join/#asterisk eatmypiano (n=eatmypia@CPE00195b4be0d6-CM0019477f689a.cpe.net.cable.rogers.com) |
| 02:46.36 | SwK | jcaceres, digium just spent a nice sized 5digit number to get the rights to resell it |
| 02:46.59 | phix | SwK: sure I will ring up my VoIP provider and tell them to replace all of their cisco equipment. |
| 02:47.03 | SwK | jcaceres, not including whatever they invested in getting the codec code itself |
| 02:47.15 | SwK | phix, cisco by default does 101 |
| 02:47.33 | phix | SwK: well this place uses 96 |
| 02:47.45 | SwK | oh well look in chan sip |
| 02:47.49 | phix | I am assuming they are using ciscos |
| 02:47.58 | phix | chan sip ay |
| 02:48.02 | JT | so you don't even know? |
| 02:48.20 | phix | JT: do you know how to change rtp payload type? |
| 02:48.29 | JT | you edit chan_sip.c |
| 02:48.37 | phix | ok |
| 02:48.44 | *** join/#asterisk mutilator (i=WebChat@the.drinkproject.com) |
| 02:49.10 | JT | cisco being defective, how unusual ;) |
| 02:50.25 | *** join/#asterisk Strom_M (n=strom@adsl-69-105-23-47.dsl.irvnca.pacbell.net) |
| 02:50.32 | phix | JT: how difficult would it be to create a configuration directive to make the payload type variable? (without needing to recompile each time) |
| 02:51.15 | JT | no idea, how easy do you find coding C? |
| 02:51.27 | SwK | man |
| 02:52.56 | phix | JT: I can code in C |
| 02:53.09 | *** join/#asterisk ltdwk (n=z@203-173-10-9.perm.iinet.net.au) |
| 02:53.11 | phix | printf("woot!"); |
| 02:53.23 | phix | :P |
| 02:54.52 | Strom_M | you forgot int main() { |
| 02:55.10 | JT | } |
| 02:55.19 | *** join/#asterisk ltdwk (n=z@203-173-10-9.perm.iinet.net.au) |
| 02:55.33 | jer | char*f="char*f=%c%s%c;main() printf(f,34,f,34,10);%c";main()printf(f,34,f,34,10); |
| 02:55.37 | jer | or something like that |
| 02:57.05 | russellb | o.O |
| 03:00.26 | SwK | russellb, |
| 03:00.37 | jer | (if that was wrong, excuse me... been a while since i wrote a quine in C, and i just did that one from memory |
| 03:01.00 | russellb | jer: heh, it's all good |
| 03:01.57 | phix | Strom_M: :( I fail! |
| 03:02.10 | phix | Strom_M: I also forgot stdio.h but ssshhh :) |
| 03:02.35 | Strom_M | hi / cocks protocol (rfc 4373) |
| 03:02.58 | *** join/#asterisk mtaht4 (n=m@66.153.18.42) |
| 03:03.12 | generalhan | JT: thanks for all your help earlier ... i really think that it HAD to have been provider error |
| 03:03.34 | JT | no probs |
| 03:03.38 | JT | i'd think it would be |
| 03:04.05 | generalhan | JT: i left it running for a LONG time and hit a pretty big low ... but for the most part the average is pretty good ! |
| 03:04.07 | generalhan | --- Results after 9672 passes --- Best: 100.000000 -- Worst: 99.938965 -- Average: 99.996384 |
| 03:04.38 | JT | hmm ok |
| 03:04.46 | JT | i don't trust the totals anyway |
| 03:05.21 | generalhan | i actually say that 99.93 and there was no T1 issues at the time, and i still have not had any Red Alarm errors since the round this after noon |
| 03:05.27 | generalhan | s/say/saw/ |
| 03:05.36 | phix | JT: would you happen to know which #define / variable I change? :) |
| 03:05.58 | JT | nope |
| 03:06.46 | phix | JT: so it is definitly in chan_sip.c? or would it be in a header file? |
| 03:07.26 | *** join/#asterisk kiscokid (n=ron@208.106.33.66) |
| 03:07.33 | *** join/#asterisk stkn__ (n=stkn@gentoo/developer/pdpc.active.stkn) |
| 03:08.48 | kiscokid | Is there any way to simulate a PRI so I can test my Sangoma to see if it works with Asterisk? |
| 03:09.10 | *** join/#asterisk CVirus (n=GoD@196.218.101.88) |
| 03:12.12 | *** join/#asterisk tuxd00d (n=tuxinato@128.187.128.131) |
| 03:12.32 | phix | JT: it was rtp.c, no chan_sip.c |
| 03:12.37 | phix | no = not |
| 03:13.19 | JT | cool |
| 03:17.18 | phix | http://www.asteriskguru.com/tutorials/unknown_codec_received.html |
| 03:18.04 | phix | JT: add that to the channel bot :) |
| 03:19.22 | *** join/#asterisk jmacz (n=jmacz@190.25.32.84) |
| 03:25.53 | *** join/#asterisk red9012 (n=marc3234@206-248-132-147.dsl.teksavvy.com) |
| 03:26.09 | red9012 | are dtmf problems finally fixed with asterisk? |
| 03:26.35 | JT | doesn't look like the problem of vague questions has been |
| 03:26.57 | Sweeper | JT: 1, Noobs: 0 |
| 03:27.09 | Sweeper | I think we need more cheerleaders |
| 03:27.20 | russellb | lol |
| 03:27.27 | russellb | red9012: yeah, they're all fixed, actually |
| 03:27.33 | Sweeper | preferably Dallas Cowboys cheerleaders looking to get a bit intellectual |
| 03:27.36 | russellb | just finished fixing them this morning |
| 03:28.02 | russellb | there was this last one with the '5' digit that kept us stumped for months |
| 03:28.18 | [hC] | oh man how did you solve that russel? |
| 03:28.22 | [hC] | I still cant get my 5s to work |
| 03:28.27 | [hC] | ive tried EVERYTHING |
| 03:28.30 | russellb | lost many nights of sleep |
| 03:29.04 | russellb | good plan |
| 03:29.07 | *** join/#asterisk Joe_CoT (i=joe_cot@ubuntu/member/joeterranova) |
| 03:29.17 | JT | that's another reason we all buy polycom |
| 03:29.25 | JT | you can reasign the number keys |
| 03:29.30 | russellb | ha, can you really? |
| 03:29.33 | Joe_CoT | question: is there a place where I can get a list of sip providers? perhaps comparisons between them? |
| 03:29.34 | JT | apparently |
| 03:29.35 | russellb | that's crazy |
| 03:29.49 | JT | an undocumented feature |
| 03:29.52 | Sweeper | I'm gonna have to do that to someone as a practical joke |
| 03:30.14 | Sweeper | make them a special provisioning file that maps them all backwards |
| 03:30.19 | phix | <PROTECTED> |
| 03:30.52 | JT | Please smash your cisco with a hammer (RFC 3389). |
| 03:31.10 | phix | russellb: hehe |
| 03:31.20 | nick125 | JT: You have some kind of hatrid against ciscos, don't you? |
| 03:32.05 | phix | JT: comfort noise is static right? |
| 03:32.09 | JT | nick125: too many fanboys, too many dollars |
| 03:32.13 | JT | phix: more or less |
| 03:32.25 | JT | phix: it means the other side is doing RTP silence supression |
| 03:33.13 | phix | JT: ok, so I have no access to my VoIP providers SIP hardware, is there a way to turn off the wanring at least? it is annoying |
| 03:33.41 | Strom_M | use a codec that doesn't have VAD |
| 03:33.45 | Strom_M | i.e. not g729 |
| 03:33.51 | JT | phix: probably edit the source again :) |
| 03:34.06 | JT | Strom_M: some providers do RTP silence supression even on g.711 |
| 03:34.10 | phix | Strom_M: I have a choice of using g729, alaw or ulaw. |
| 03:34.26 | phix | JT: hehe, aww but I am compiling already! |
| 03:34.29 | Strom_M | ulaw if north america |
| 03:34.40 | Strom_M | alaw if not north america |
| 03:34.49 | JT | phix: what's more annoying than the warning is a dead sounding phone call |
| 03:34.53 | phix | Strom_M: oh, well they support it :) they use American hardware |
| 03:35.01 | phix | JT: I suppose |
| 03:35.23 | JT | phix: people on your side will always wonder if the other end has dropped out |
| 03:35.30 | phix | I will get a static generator |
| 03:35.44 | JT | and promise to comit it to mantis :) |
| 03:35.53 | Sweeper | phix: I've got a cheap one I can sell you |
| 03:36.06 | Sweeper | I made it myself! |
| 03:36.17 | phix | Sweeper: heh, does it go buuuzzzzz or hiiizizizz or a combination of both? |
| 03:36.18 | Sweeper | you have to have long hair to make it work well tho |
| 03:36.31 | Sweeper | more of a crackling.... |
| 03:36.36 | phix | I only like hiiziz sounds |
| 03:36.40 | Sweeper | ahhh |
| 03:36.41 | phix | :) |
| 03:36.52 | phix | is there a filter for that? |
| 03:37.07 | Sweeper | well, what you really want is gaussian white noise |
| 03:37.31 | phix | I am allergic to mathematical algorithms |
| 03:37.37 | Sweeper | :P |
| 03:39.39 | kiscokid | Anyone know where I could borrow a PRI simulator? |
| 03:40.27 | JT | my garage |
| 03:40.34 | JT | don't think that's useful though |
| 03:40.35 | JT | :P |
| 03:40.41 | JT | to you anyway |
| 03:40.52 | kiscokid | where is it located? |
| 03:40.59 | JT | sydney, australia |
| 03:41.12 | kiscokid | not immediately useful |
| 03:41.28 | JT | what card do you have? |
| 03:41.34 | JT | and why do you want to simulate? |
| 03:41.44 | *** join/#asterisk VoipMasta (n=fabio@dial-148-240-58-190.zone-2.dial.net.mx) |
| 03:41.47 | kiscokid | Sangoma A101d |
| 03:41.49 | VoipMasta | Hi there |
| 03:42.11 | kiscokid | Just want to test my configuration before we get our real PRI |
| 03:42.17 | VoipMasta | Does anyone know how can I know the IAX registration state/status from an AGI? |
| 03:42.27 | JT | shrug, as if you need to test ;) |
| 03:42.41 | JT | tell us what sort of connection you're getting, and pastebin your configs |
| 03:43.15 | kiscokid | ok |
| 03:44.19 | *** join/#asterisk Ciber311 (n=Ciber311@user-12ld42j.cable.mindspring.com) |
| 03:46.27 | *** join/#asterisk LakeSolon (n=blake@64-83-205-22.dhcp.stcd.mn.charter.com) |
| 03:55.10 | Getafix | hey, anyone familiar with queues.conf/timeoutrestart? |
| 03:55.46 | Getafix | the docs on the voip-info wiki say that timeoutrestart is set to yes, the timeout for an agent to answer is reset if a BUSY or CONGESTION is received |
| 03:55.57 | Getafix | well, I have it turned to =yes, and I am sending back a BUSY |
| 03:56.05 | Getafix | but it is dropping to voicemail |
| 03:56.16 | Getafix | instead of timing out |
| 03:57.36 | *** join/#asterisk CBU[^_^]M`` (n=love@210.213.139.218) |
| 04:01.42 | *** part/#asterisk kiscokid (n=ron@208.106.33.66) |
| 04:04.56 | SplasPood | Getafix: what is dropping to vm |
| 04:06.20 | phix | What is a good tool for converting wav / pcm files to g729? |
| 04:07.24 | phix | also, I have one g729 licence, if there is a phone conversation already in progress and somebody else calls, will I need another licence to play a busy message which is in g729 format? |
| 04:07.24 | Getafix | SplasPood: the queue caller |
| 04:07.38 | Getafix | SplasPood: I assumed that with that turned on, and with BUSY being sent back from the phone, the caller would remain in the queue |
| 04:08.07 | Getafix | instead of going to the agents voicemail |
| 04:14.02 | Getafix | I can't work out why the ACD is causing Executing Macro("Local/703@maxnet-users-da24,2", "stdexten|703|SIP/703") in new stack |
| 04:14.06 | Getafix | to happen |
| 04:14.11 | VoipMasta | phix: yes, you'll need a second license to answer another channel in g729 |
| 04:14.12 | Getafix | can I specify what it does when it calls a user? |
| 04:14.39 | VoipMasta | phix: if you have g729 transcoding (since I guess you have as you own a license) you don't have to convert it, Asterisk will do it for you |
| 04:14.54 | Getafix | what type of channel is Local |
| 04:15.43 | JT | VoipMasta: you're completely missing the whole point |
| 04:15.44 | Getafix | and why does ACD use it |
| 04:16.02 | VoipMasta | JT: I am? |
| 04:16.07 | JT | phix: no, you will not need a licence if the file is already in G.729 format I believe. |
| 04:16.39 | VoipMasta | JT: Ok, you can save a license if you limit yourself to just play a message (no voicemail, forwarding, etc.) |
| 04:16.50 | JT | VoipMasta: G.729 format file playing without transcoding SHOULD == no licence use |
| 04:17.00 | VoipMasta | I believe there's an audio converting tool in asteriskguru right? |
| 04:17.09 | JT | probably |
| 04:19.06 | VoipMasta | phix: http://www.asteriskguru.com/tools/audio_conversion.php |
| 04:19.46 | VoipMasta | however I still think that for $10 it's worth it being able to "do something else" with the call besides playing a message |
| 04:20.01 | JT | VoipMasta: again, you really should read what he said :) |
| 04:20.04 | JT | he HAS a licence |
| 04:20.07 | Getafix | anyone know if it's possible to configure *how* asterisk ACD delivers calls to agents? |
| 04:20.16 | VoipMasta | JT: he has ONE license |
| 04:20.17 | JT | and is talking about what to do with second calls coming in |
| 04:20.46 | JT | VoipMasta: correct |
| 04:21.17 | VoipMasta | JT: What I mean is that for another $10 he could "do more" with the second call than just playing a message |
| 04:21.39 | VoipMasta | JT: like sending it to a voicemail, using call-waiting, etc. |
| 04:21.42 | phix | JT: nice, i will do that :) |
| 04:21.45 | JT | true |
| 04:21.57 | JT | maybe he could just not use evil G.729 |
| 04:22.19 | VoipMasta | some devices do only support G729 as low bw codec |
| 04:22.22 | VoipMasta | no GSM :( |
| 04:22.39 | phix | no gsm here :( |
| 04:22.57 | Getafix | anyone know what timeoutrestart actually does? |
| 04:23.13 | VoipMasta | we've been using Cisco and Linksys and we have to stick to G.729a or G.723 |
| 04:23.35 | VoipMasta | I would be the happiest man on earth if they started adding GSM as an option to their devices |
| 04:23.54 | JT | VoipMasta: they don't do g.711? |
| 04:24.04 | VoipMasta | yes, but again... for low bw users... |
| 04:24.31 | JT | pesky users ;) |
| 04:24.43 | VoipMasta | we have users in countries where DSL connections aren't symmetric... they get maybe 2mbps download but only 256kbps upload |
| 04:25.15 | VoipMasta | and they "promise" 256kbps, the truth is that hey rarely reach over 200 |
| 04:25.18 | phix | VoipMasta: I am in one of those countries :( |
| 04:25.23 | JT | dsl is usually never symmetric |
| 04:25.28 | JT | phix: you're on Earth ;) |
| 04:25.31 | phix | VoipMasta: synced DSL is too expensive |
| 04:25.32 | VoipMasta | now if they want to have more than one call at the same time... |
| 04:25.43 | phix | JT: heh |
| 04:25.55 | JT | it also massively raises the noise floor in copper loop cables |
| 04:25.55 | VoipMasta | we need to use something less bandwidth consuming than 711 |
| 04:26.15 | JT | (symmetrical dsl that is) |
| 04:26.38 | phix | JT: hmmm if I wanted to talk to a computer on Mars, what protocol should I use? :) since there is a 8minute lag (16minute to send then receive) |
| 04:26.56 | JT | carrier pidgeon |
| 04:27.03 | Getafix | concur |
| 04:27.06 | VoipMasta | here in Mexico it's almost impossible to get something over 256kbps upstream while keeping the cost under $100/mo |
| 04:27.18 | phix | JT: heh, that is a great RFC, has anyone actually implemented it? |
| 04:27.31 | Getafix | the people who wrote the RFC implemented it, afaik |
| 04:27.37 | phix | heh |
| 04:27.41 | JT | we can get 24000/1200kbit/s dsl here for less than $100/mo |
| 04:28.13 | phix | JT: ditto, although there it is capped at 50 or 60Gb or something tiny like that |
| 04:28.17 | VoipMasta | here in Mexico a 2000/256 dsl costs about 60/mo |
| 04:28.45 | phix | VoipMasta: oohhh do you own a huge hat? |
| 04:28.57 | VoipMasta | huge hat? |
| 04:29.12 | VoipMasta | not really |
| 04:29.34 | phix | sombrero |
| 04:29.49 | phix | aawww |
| 04:29.57 | JT | phix: pfft, 60GB isn't that bad |
| 04:30.01 | JT | i rarely use over 10GB |
| 04:30.11 | *** part/#asterisk Joe_CoT (i=joe_cot@ubuntu/member/joeterranova) |
| 04:30.12 | phix | JT: I use 100Gb or more |
| 04:30.19 | VoipMasta | as a matter of fact, I've always thought that all hats, caps and similar things are for the working masses, I do only justify using them while exercising |
| 04:30.30 | phix | JT: I hate caps regardless if I wasn't using that much |
| 04:30.35 | JT | phix: leecher :P |
| 04:30.37 | JT | uhuh |
| 04:31.23 | phix | JT: :P I leech linux isos and other licence free software and media :) |
| 04:31.25 | Getafix | anyone know what lets the Local channel decide what to do? |
| 04:31.30 | JT | phix: sure you do |
| 04:31.38 | Getafix | I need to work out what is making ACD work, and where to change it. |
| 04:31.41 | phix | JT: yep :) that is my story and I am sticking to it |
| 04:32.10 | VoipMasta | phix: Then shut down your P2P client... you don't need it for open source software |
| 04:32.30 | phix | I hate P2P, I do not use it |
| 04:33.05 | phix | http, ftp+tls/ssl, and sftp are my protocols of choice. |
| 04:33.10 | VoipMasta | phix: there was a time when I downloaded every single linux iso as it was released... then I saw the light |
| 04:33.33 | phix | VoipMasta: :) |
| 04:33.49 | VoipMasta | phix: and switched to BSD, which I install over the internet :) |
| 04:38.41 | Getafix | oh crhist |
| 04:38.44 | Getafix | I'm fucking incredible |
| 04:38.49 | Getafix | I'm like *the* best. |
| 04:41.33 | phix | VoipMasta: BSD is nice although I know more about Linux than BSD |
| 04:42.12 | Getafix | anyone super familiar with agentcallbacklogin ? |
| 04:42.19 | Getafix | I need to make it so that it doesn't ask for the username |
| 04:42.41 | phix | I have read up on it from the BSD handbooks, they are usefull, but alot of things that are simply in linux, linux volume management (LVM/EVMS) seem a bit hacky in BSD |
| 04:43.42 | VoipMasta | phix: true, but once you work with it for a while, you start doing everything "naturally" and linux becomes more hacky :) |
| 04:44.28 | phix | ok :) |
| 04:44.43 | phix | I wouldn't mind learn Solaris either |
| 04:45.13 | VoipMasta | phix: as a matter of fact, right now I'm setting up a small pbx using a small freebsd box, with a set of php scripts to manage it over the web |
| 04:45.31 | phix | nice |
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| 05:17.06 | phix | VoipMasta: how well does Asterisk run on FreeBSD? |
| 05:17.11 | phix | BSD even |
| 05:20.14 | Mavvie | I see it in the ports tree, which makes me believe it works good enough. |
| 05:21.19 | JT | where good == not so good for some zaptel hardware |
| 05:22.41 | *** join/#asterisk fujin (n=fujin@unaffiliated/fujin) |
| 05:22.47 | fujin | hi there, anyone know what would cause Aug 8 17:22:18 WARNING[22711]: channel.c:787 channel_find_locked: Avoided initial deadlock for '0xb6e3b3c0', 10 retries! |
| 05:23.01 | fujin | I have my phone set to DND, and whenever someone calls it I see that in the console |
| 05:23.25 | JT | how does your phone handle DND? |
| 05:23.43 | fujin | 486 busy here |
| 05:24.07 | fujin | I would have assumed that I would see the callflow though |
| 05:24.09 | fujin | instead of just that |
| 05:24.36 | JT | well you might with sip debug on |
| 05:24.42 | fujin | I do |
| 05:24.53 | fujin | now what I want to know is why I don't see it with verbose 3 |
| 05:25.09 | JT | try verbose 10 |
| 05:25.29 | fujin | ah |
| 05:25.30 | fujin | lol |
| 05:25.32 | fujin | I'm such a tard |
| 05:25.35 | fujin | had verbose 0 |
| 05:25.41 | fujin | still, I'm interested to know what the deadlock error is |
| 05:25.42 | fujin | any ideas? |
| 05:26.47 | JT | yes, see what the sip and rtp behaviour is like |
| 05:33.17 | fujin | hrm, I'm trying to get *good* queue behaviour with my phones |
| 05:33.18 | fujin | by using DND |
| 05:33.29 | fujin | and the Busy application |
| 05:33.43 | fujin | but I'd like it to only apply when it's a ACD call calling the phone |
| 05:33.59 | fujin | anyway I can pass a variable or something to the macro which actually dials the Local? |
| 05:35.43 | fujin | or like check if a variable exists in my Macro |
| 05:35.55 | fujin | if ${i_am_an_acd_call} |
| 05:35.56 | fujin | :( |
| 05:35.58 | fujin | any ideas JT ? |
| 05:40.22 | phix | JT knows all |
| 05:45.40 | JT | you can pass whatever variables you wish to a macro |
| 05:45.43 | JT | but you must pass them |
| 05:46.11 | JT | it does not inherit |
| 05:47.50 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 05:57.14 | fujin | JT: sure, I understand that |
| 05:57.27 | fujin | but what variables does asterisk pass to the Macro, when it calls an agent |
| 05:57.30 | fujin | that's what I wanna know. |
| 06:00.13 | JT | i have no idea |
| 06:00.43 | Sweeper | ok, who's going to hire me for a bit of coding tonight? |
| 06:00.58 | Sweeper | I'm at stopping points for all my other projects |
| 06:01.00 | fujin | I'll hire you if you can tell me what asterisk passes when it delivers an ACD call to an agent |
| 06:01.03 | *** join/#asterisk dongs (i=500@l212168.ppp.asahi-net.or.jp) |
| 06:01.08 | fujin | I'll give you five dollars, via paypal. |
| 06:01.09 | fujin | ;\ |
| 06:01.14 | dongs | does zaptel 1.2.19 compile with 2.6.22.x kernel |
| 06:01.18 | JT | Sweeper: can you reprogram all errors to say "Error 101: Nub." ? |
| 06:01.30 | fujin | <PROTECTED> |
| 06:01.31 | Sweeper | JT: the outlook is good |
| 06:02.26 | tzafrir | dongs, AFAIK, yes |
| 06:02.39 | dongs | okey. cuz 1.2.13 or something didnt |
| 06:02.43 | dongs | .19 still compiling |
| 06:03.16 | VoipMasta | phix: sorry, it runs pretty well |
| 06:04.10 | dongs | looks working |
| 06:04.11 | dongs | great |
| 06:04.37 | phix | VoipMasta: ok |
| 06:05.00 | VoipMasta | phix: if you ask me, I've found it to be far more reliable on BSD |
| 06:05.27 | fujin | is there a way to print all current variables? |
| 06:05.36 | phix | VoipMasta: interesting |
| 06:05.37 | *** join/#asterisk cthorner (n=cthorner@72-254-9-98.client.stsn.net) |
| 06:06.15 | VoipMasta | phix: my average asterisk uptime is 2 months on BSD |
| 06:06.30 | VoipMasta | without any leaks nor overloading the cpu |
| 06:06.32 | dongs | since when does asterisk run on BSD? |
| 06:06.37 | Sweeper | dongs: a long time |
| 06:06.47 | Sweeper | just not very good zaptel support |
| 06:06.59 | VoipMasta | dongs: it was ported a while ago... maybe something like 2 years |
| 06:07.56 | JT | VoipMasta: 2 months isn't that great |
| 06:08.01 | JT | VoipMasta: linux can do that too |
| 06:08.07 | dongs | heh i see |
| 06:08.16 | dongs | includeing zaptel hardware? |
| 06:08.34 | Sweeper | JT: yes, but BSD is sexier |
| 06:08.50 | VoipMasta | JT: Yes, I know, but it was about 400 SIP devices registering every day. Maybe it would last longer, but it's a production environment so I don't want to risk it. |
| 06:08.59 | VoipMasta | "it has" |
| 06:13.40 | JT | VoipMasta: so you reboot it? |
| 06:13.42 | *** join/#asterisk sergee (n=serg@voip1.west-call.com) |
| 06:19.21 | flenders | pbx*CLI> core show uptime |
| 06:19.22 | flenders | System uptime: 10 weeks, 2 days, 1 hour, 45 minutes, 43 seconds |
| 06:20.05 | JT | i think pbx uptimes should be measured in years |
| 06:20.56 | JT | heh |
| 06:21.11 | VoipMasta | JT: I restart asterisk |
| 06:21.25 | VoipMasta | JT: as I said, 2 months is my average asterisk uptime, not server uptime |
| 06:22.52 | JT | VoipMasta: why do you restart for no reason? |
| 06:23.10 | dongs | hmm |
| 06:23.10 | dongs | System uptime: 29 weeks, 5 days, 15 hours, 18 minutes, 13 seconds |
| 06:23.20 | dongs | heh |
| 06:23.22 | Sweeper | JT: he doesn't want the outage to take place during critical times? |
| 06:23.42 | JT | Sweeper: trying to gamble with randomness? ;) |
| 06:23.48 | VoipMasta | JT: From what I've seen, asterisk has small memory leaks when it comes to config reloading, those small leaks add up and sooner or later make it start having issues. |
| 06:23.56 | Sweeper | JT: yes! |
| 06:23.57 | Sweeper | :D |
| 06:24.11 | Sweeper | there's randomness, and there's the creeping memory leaks |
| 06:24.23 | JT | VoipMasta: not a good inditment on asterisk ;) |
| 06:24.46 | Sweeper | might as well have a scheduled outage every two months at 3am if it'll reduce chances of screaming death during business hours |
| 06:24.52 | VoipMasta | JT: by restarting asterisk every 2 months I get 10 more minutes of peaceful sleep every day :) |
| 06:25.40 | JT | wonder when asterisk will be stable ;) |
| 06:26.26 | Sweeper | JT: I hear it's a lot more stable when you run it on the MS unified communications server |
| 06:26.33 | VoipMasta | I think it's stable enough... there isn't a 100% stable piece of software, even your server's bios might become unstable under certain circumstances. |
| 06:27.08 | *** join/#asterisk remmo (n=junk@smack.isp.net.au) |
| 06:28.03 | JT | VoipMasta: err, my server hardware uptime is measured in multiple months or years, so that doesn't come into play for me |
| 06:28.12 | JT | i'm talking about software uptime |
| 06:28.34 | VoipMasta | JT: I've also reached an uptime of nearly 2 years... but as I said "under certain circumstances" |
| 06:28.52 | JT | how will it ever improve if people continue to have an attitude of "stable enough" |
| 06:28.54 | *** join/#asterisk syneus (n=syneus@syneus.aemcom.net) |
| 06:29.10 | JT | also, the stability of asterisk seems to vary wildly depending on what you do with it |
| 06:29.21 | JT | low demand setups seem to stay up longer |
| 06:29.23 | VoipMasta | I just had an issue with an X101p clone... it was making the bios go berzerk! some IRQ problems I guess... since we replaced that clone everything went back to normal |
| 06:30.02 | JT | well yeah, using junk does that |
| 06:30.13 | VoipMasta | JT: I agree, my small office PBX could stay online forever... but it's never stressed out. One of our asterisk servers handles a lot of calls and some heavy transcoding |
| 06:30.52 | VoipMasta | junk, hardware failures, power variations, heat, shocks... again "certain circumstances" |
| 06:31.41 | JT | x101p == junk |
| 06:32.52 | VoipMasta | I didn't say x101p!=junk, I just said that junky hardware is just one of many circumstances than can affect a piece of software, even something as "basic" as the BIOS |
| 06:32.54 | *** join/#asterisk dominic1 (n=dob@213.221.82.242) |
| 06:33.31 | JT | again, a lot of guessing here instead of reality |
| 06:33.40 | JT | BIOSes VERY rarely cause issues on modern servers |
| 06:34.17 | VoipMasta | mmm I still have issues with a specific BIOS version and Ubuntu... (AMD64) |
| 06:34.43 | VoipMasta | and we've faced the need to update (flash) the BIOS more than once on brand servers |
| 06:35.00 | VoipMasta | JT: how many servers do you own (not lease)? |
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| 06:37.45 | *** part/#asterisk dongs (i=500@l212168.ppp.asahi-net.or.jp) |
| 06:53.40 | *** join/#asterisk implicit_ (n=implicit@vc240168.vpn.uci.edu) |
| 06:54.57 | JT | VoipMasta: i'm not sure what your point is. you say you have high end server yet use X101Ps |
| 06:57.04 | *** join/#asterisk TheNewAndy (n=TheNewAn@144.131.134.181) |
| 06:59.18 | *** join/#asterisk menil (n=meni@bzq-179-153-168.static.bezeqint.net) |
| 07:01.20 | implicit_ | maybe he's just not a VoipMasta? |
| 07:01.22 | VoipMasta | JT: yes, we have high end servers for several purposes, not just voip... and yes, we use x101p clones for small pbx systems |
| 07:01.23 | *** join/#asterisk zeedo (n=zeedo@reboot-robot.net) |
| 07:02.52 | JT | VoipMasta: imho they shouldn't be used for anything |
| 07:04.47 | *** join/#asterisk GaryH (n=chatzill@2001:618:42d:101:213:72ff:fecf:8262) |
| 07:04.47 | VoipMasta | JT: I do agree, but there are some customers (really small businesses and SOHOs) that like their $500.00 USD PBX systems |
| 07:05.10 | VoipMasta | JT: and it's impossible to sell something with real hardware (you name it, sangoma, digium, etc.) for that price |
| 07:05.20 | JT | linksys |
| 07:06.51 | *** join/#asterisk vgster (n=vgster@host217-45-221-53.in-addr.btopenworld.com) |
| 07:06.56 | VoipMasta | JT: FXO? |
| 07:07.08 | JT | yes |
| 07:07.16 | JT | better than a X***P |
| 07:07.46 | *** join/#asterisk gzero (n=gzero@81.175.82.2) |
| 07:08.00 | VoipMasta | but can you get 2xFXO + 4xFXS + IVR + Voicemail + all hardware for $500.00? |
| 07:09.31 | JT | how do you provide 4 FXS ports with X101Ps? |
| 07:10.13 | VoipMasta | no, I do so with 2 x PAP2-NA |
| 07:10.31 | JT | i see |
| 07:11.03 | *** part/#asterisk gzero (n=gzero@81.175.82.2) |
| 07:11.04 | JT | selling a PBX on price alone is stupid |
| 07:11.46 | *** join/#asterisk psk (n=psk@golia.caltanet.it) |
| 07:12.25 | *** join/#asterisk clandmeter (n=gzero@81.175.82.2) |
| 07:14.01 | *** part/#asterisk TheNewAndy (n=TheNewAn@144.131.134.181) |
| 07:14.46 | VoipMasta | JT: may I ask why? |
| 07:15.28 | JT | because it's a crap way to make a living unless all your stuff is mass produced really cheaply |
| 07:15.35 | JT | like in china |
| 07:15.47 | *** part/#asterisk clandmeter (n=gzero@81.175.82.2) |
| 07:15.51 | *** join/#asterisk zeeesh (i=zeeesh@202.38.55.121) |
| 07:16.16 | VoipMasta | we sell those PBXs dirt cheap because we provide call termination and origination to those customers |
| 07:16.17 | JT | it also devalues the potential valued add that asterisk and other open source solutions can provide :) |
| 07:16.42 | JT | well then you're not actually in the PBX market there, you're in the ITSP market |
| 07:17.21 | VoipMasta | we're not a hardware vendor |
| 07:17.45 | VoipMasta | but each PBX we sell at 500.00 generates about 300/mo profits from phone services ;) |
| 07:18.05 | JT | that's totally different then |
| 07:18.20 | JT | and you'd still want the device to be rock solid, to avoid support costs |
| 07:18.25 | JT | and customer frustration |
| 07:20.13 | VoipMasta | what I've experienced is that once an X101p is fully working it will rarely fail |
| 07:20.25 | VoipMasta | they usually fail within the first few hours of use |
| 07:21.03 | JT | you realise they are discontinued |
| 07:21.47 | VoipMasta | yes |
| 07:21.55 | VoipMasta | but I have about 300 of them here |
| 07:22.06 | VoipMasta | x101p clones |
| 07:22.20 | JT | the X101P is the clone |
| 07:22.27 | JT | digium never had a product called the X101P |
| 07:22.33 | VoipMasta | when I found them and saw that I could buy them really cheap (7.50 each) I bought as many as possible |
| 07:23.55 | *** join/#asterisk af_ (n=getsmart@81-174-47-190.dynamic.ngi.it) |
| 07:25.14 | JT | i see |
| 07:25.31 | JT | is it some sort of embedded box? |
| 07:25.41 | VoipMasta | no, a mini-tower |
| 07:25.46 | JT | oh... |
| 07:25.55 | VoipMasta | again, $500.00 |
| 07:26.05 | JT | yeah, and |
| 07:26.15 | JT | you'd want it to be embedded given the business model ;) |
| 07:27.09 | VoipMasta | I'm buying PAP2T-NA at 55 each, so it's 110 for 2 ATAs... about 200 for the mini-tower... 15 for 2 X101P |
| 07:27.12 | VoipMasta | that's 345 |
| 07:27.54 | VoipMasta | plus what I have to pay a guy to go to the customer's office and install it |
| 07:28.07 | VoipMasta | and a small profit for my company (there has to be a profit) |
| 07:28.22 | JT | the profit bit is optional if it's tied into telephony |
| 07:29.00 | VoipMasta | it's not optional when we replace faulty hardware at our expense |
| 07:29.31 | VoipMasta | and sometimes our technicians have to do some small RJ-11 RJ-45 wiring |
| 07:29.43 | VoipMasta | ohh and a 4 port switch (used to connect both ATA's to the PC) |
| 07:30.45 | JT | some embedded boards are pretty cheap and have multiple ethernet ports |
| 07:31.05 | VoipMasta | but most embedded boards don't have PCI slots. they have mini-pci |
| 07:31.29 | s0ck | morning |
| 07:32.19 | JT | soekris has pci slots |
| 07:32.25 | JT | and if you play it differently |
| 07:32.42 | JT | ie, ATA for FXO, you can get embedded board + ATA for <$200 |
| 07:33.16 | VoipMasta | which ATA? |
| 07:33.24 | VoipMasta | 2 x FXO? |
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| 07:35.40 | s0ck | by 'embedded' do you mean using an sd card for *? |
| 07:36.12 | JT | s0ck: no, using an embedded board |
| 07:36.23 | VoipMasta | s0ck: yes, in an embedded system you would use an SD card as a storage device |
| 07:36.40 | s0ck | i've toyed with this idea but not sure how well they perform...? |
| 07:36.46 | JT | they usually use CF cards |
| 07:36.56 | JT | or onboard NAND flahs |
| 07:36.57 | VoipMasta | s0ck: an embedded board is a single board with cpu, ram, ethernet, etc. |
| 07:36.58 | JT | flash |
| 07:37.06 | s0ck | seems to be the way forward really, eliminate the hdd from the equation (moving parts et al) |
| 07:37.18 | JT | s0ck: not good for transcoding, but fine otherwise |
| 07:37.19 | VoipMasta | JT: my mistake, they are actually CFs |
| 07:37.27 | VoipMasta | I have 2 WRAPs here |
| 07:37.31 | VoipMasta | but they are way too expensive |
| 07:37.42 | s0ck | i remember looking up the spec on those soekris units |
| 07:37.44 | JT | WRAPs are some of the most expensive in the market |
| 07:37.46 | JT | and discontinued |
| 07:37.48 | s0ck | does * actualy run tidy on them? |
| 07:37.52 | JT | yes |
| 07:37.54 | VoipMasta | JT: which ATA do you know that provides 2 FXO ports? |
| 07:38.02 | s0ck | p3 etc init? |
| 07:38.06 | VoipMasta | JT: They are not discontinued... |
| 07:38.31 | JT | they are, there is a new series coming out |
| 07:38.41 | JT | the chipset was discontinued, so so was the WRAP |
| 07:42.13 | JT | VoipMasta: non have 2 FXO that i've seen |
| 07:42.18 | JT | not at that price range |
| 07:42.27 | JT | SPA-3102 = $70 |
| 07:42.37 | JT | 2 * 70 + 55 = 195 |
| 07:43.16 | JT | embedded gear would probably be $150 all up |
| 07:43.46 | JT | maybe more maybe less, depends on what you get |
| 07:44.11 | VoipMasta | the SPA3102 is 1 FXS / 1 FXO right? |
| 07:45.22 | VoipMasta | so 2 x 20 + 55 = 195 |
| 07:45.31 | VoipMasta | embedded = 50 |
| 07:45.34 | VoipMasta | 150 |
| 07:45.41 | VoipMasta | power supply = 10 |
| 07:45.49 | VoipMasta | case = 25 |
| 07:46.10 | VoipMasta | so far $380 |
| 07:46.13 | JT | s/20/70/ but yeah |
| 07:46.33 | VoipMasta | CF card: $15 |
| 07:46.42 | JT | 150 was a rough estimate, depends what you get |
| 07:46.58 | VoipMasta | ok we're approaching the 400 range |
| 07:47.06 | VoipMasta | now, there are a few things to consider... |
| 07:47.14 | JT | i think it could be done for under $350 |
| 07:47.32 | VoipMasta | 1. I don't know any reliable provider of embedded boards in Mexico, so I would have to import them myself like I did with these WRAPs |
| 07:47.54 | JT | soekris.com or routerboard.com |
| 07:48.03 | VoipMasta | 2. $25 for a case is assuming that I can find a "similar" case that can be suited for this project, otherwise it would be more expensive |
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| 07:48.24 | *** join/#asterisk tzafrir (n=tzafrir@62.90.10.53) |
| 07:48.45 | VoipMasta | 3. I need to sell the ~300 X101P that I have here |
| 07:49.36 | VoipMasta | 4. When a non-techie customer sees our PBX as a "full box" they feel that they are getting more hardware for their money than if we deliver a small embedded system (again, non-tech customers) |
| 07:49.56 | VoipMasta | so maybe we will start doing it with embedded in the near future, but for now we have to stick with the mini-tower model |
| 07:50.05 | JT | if your box looks like a pc, most customers think "wtf a pc, where's the pbx" |
| 07:50.13 | creativx | put a sticker of a phone on it. |
| 07:51.21 | VoipMasta | we use some "uncommon" cases (all black) and we put a 5.25" LCD display that shows CPU temp, fan RPMs, etc. (yeah, it's BS but it makes it look more "pro") |
| 07:51.38 | JT | VoipMasta: do you do any faxing? |
| 07:51.40 | VoipMasta | and we have room on the inside to place the ATAs |
| 07:51.46 | VoipMasta | nope |
| 07:51.55 | VoipMasta | I haven't found a reliable t.38 termination provider |
| 07:52.35 | JT | VoipMasta: do any customers use ip phones? |
| 07:53.00 | VoipMasta | only a few |
| 07:53.04 | VoipMasta | most of them use regular analog phones |
| 07:53.12 | VoipMasta | that's why we use ATAs |
| 07:54.33 | JT | well i assumed the FXS ports weren't there just to fill up space |
| 07:55.17 | *** join/#asterisk c0dz3r0 (n=d@cpe-74-72-105-63.nyc.res.rr.com) |
| 07:55.20 | VoipMasta | ip phones are hard to find here in mexico |
| 07:55.36 | VoipMasta | most "ip phones" you'll find are really usb headsets or skype phones |
| 07:55.44 | JT | eww |
| 07:56.01 | JT | they're not things that are in every corner store anyway |
| 07:56.04 | VoipMasta | and a real avaya or cisco ip phone would cost about 400 usd |
| 07:56.06 | LakeSolon | They're not exactly growing from trees up here either... just order 'em online. |
| 07:56.21 | c0dz3r0 | im trying to use a different menu system from VoicemailMain(), would you suggest retwritting the actual application on writing a mix of AEL and Perl AGI to get it to work? |
| 07:56.23 | JT | yeah but who would but avaya or cisco overpriced crap? |
| 07:56.25 | JT | LakeSolon: indeed |
| 07:56.25 | VoipMasta | but products from china have a very high import tax |
| 07:56.40 | LakeSolon | JT - What's your brand of choice? |
| 07:56.42 | JT | what about thailand? |
| 07:56.44 | JT | polycom |
| 07:56.59 | VoipMasta | is polycom manufactured in thailand |
| 07:57.00 | VoipMasta | ? |
| 07:57.02 | c0dz3r0 | im trying to use a different menu system from VoicemailMain(), would you suggest retwritting the actual application on writing a mix of AEL and Perl AGI to get it to work? |
| 07:57.03 | JT | yes |
| 07:57.03 | *** join/#asterisk vgster (n=vgster@host217-45-221-53.in-addr.btopenworld.com) |
| 07:57.26 | VoipMasta | I didn't know that, that could be a solution |
| 07:57.45 | LakeSolon | So, I've got an odd one for you... |
| 07:57.57 | JT | the IP320 is seriously close in price to the grandstream GXP2000 |
| 07:58.01 | JT | and they are worlds apart |
| 07:58.08 | JT | the grandstream being junk, of course |
| 07:58.43 | LakeSolon | Whenever I use one of my DID's as the outbound CID for a call (to my cell phone) I get a message "The number you have dialed is unallocated", and /then/ it dials my cell phone and rings properly. |
| 07:58.56 | LakeSolon | If I use a totally bogus CID entry, it rings directly to the cell phone. |
| 07:59.15 | JT | LakeSolon: does the bogus CID appear? |
| 07:59.20 | LakeSolon | Yu'up |
| 07:59.28 | JT | bloody america |
| 07:59.30 | *** join/#asterisk qdk (n=qdk@213.150.62.32) |
| 07:59.31 | JT | heh |
| 07:59.47 | JT | do you send all digits of the DID? |
| 08:00.29 | LakeSolon | The traditional 10, though I suppose I could try the 1 prefix and make it 11 |
| 08:01.26 | JT | it may be something to speak to your telco about |
| 08:01.47 | LakeSolon | Ya, I'm starting to wonder about it... |
| 08:01.50 | LakeSolon | Vitelity, fwiw. |
| 08:01.56 | JT | oh, ITSP |
| 08:02.20 | LakeSolon | ITSP? |
| 08:02.43 | JT | ~itsp |
| 08:02.44 | jbot | An ITSP (Internet Telephony Service Provider.) is a "VoIP Phone Company". Example : Vonage, Broadvoice, Teliax, VoicePulse, etc. "All ITSPs suck. Some suck less than others." (tm) (c) 2007 ManxPower |
| 08:03.04 | LakeSolon | ya, just read that. |
| 08:03.21 | LakeSolon | You seemed to be correcting me when I said Vitelity, which IS a voip provider. |
| 08:03.30 | LakeSolon | Thus the confusion =) |
| 08:04.16 | JT | heh |
| 08:04.41 | *** join/#asterisk Kumbang (n=kumbang@router-wmi.paume.ITB.ac.id) |
| 08:04.55 | VoipMasta | well I gotta go |
| 08:04.59 | VoipMasta | it's getting kinda late |
| 08:05.15 | VoipMasta | it's been nice talking to you JT |
| 08:05.18 | JT | 1804 here |
| 08:05.21 | JT | seeya |
| 08:05.49 | VoipMasta | bye |
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| 08:27.45 | zeeesh | i just installed "asterisk-addons-1.4.2" for realtime asterisk ..where do i need to copy this file "res_config_mysql.c" ? |
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| 08:42.43 | *** join/#asterisk kv0s (n=kv0s@p4FD222EC.dip0.t-ipconnect.de) |
| 08:42.54 | kv0s | Hi! |
| 08:46.19 | jcaceres | JI |
| 08:46.25 | jcaceres | hi |
| 08:47.14 | kv0s | Does anybody has an idea why my date/time in my voicemailmessages is -2 hours from my realtime at my linuxbox? |
| 08:48.23 | tzafrir | real time according to what, exactly? date ? |
| 08:48.53 | JT | kv0s: what is your timezone offset from UTC? |
| 08:49.54 | *** join/#asterisk clandmeter (n=Carlo@81.175.82.2) |
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| 08:58.14 | *** join/#asterisk foxpdll (n=dima@195.218.213.138) |
| 08:58.34 | foxpdll | Доброе всем время суток |
| 08:58.59 | JT | Parse error. |
| 08:59.04 | foxpdll | не могли бы наличиствующие сдесь гуру проконсультировать тупого юзера по одному очень скромному вопросу? |
| 08:59.10 | JT | Parse error. |
| 09:00.58 | foxpdll | еуые |
| 09:01.02 | foxpdll | test |
| 09:01.29 | foxpdll | Hello! Can anyone help me in asterisk trouble? |
| 09:01.55 | JT | depends if you have a question |
| 09:02.05 | foxpdll | A need to connect D-Link DPH140s sip phone whith asterisk |
| 09:02.36 | *** join/#asterisk dickyjoe (n=richardl@dsl-124-149-123-133.nsw.westnet.com.au) |
| 09:03.01 | foxpdll | but i recieve this error message [Aug 8 13:02:43] NOTICE[10406]: chan_sip.c:14758 handle_request_register: Registration from '<sip:503@192.168.7.130:5060>' failed for '192.168.7.8' - Device does not match ACL |
| 09:03.07 | dickyjoe | Hi akk |
| 09:03.09 | dickyjoe | all |
| 09:03.31 | foxpdll | hi |
| 09:04.15 | *** join/#asterisk shinao1 (n=shinao1@41.205.185.13) |
| 09:04.17 | dickyjoe | Any Aussies on here that run a TDM400P with a FXO module? |
| 09:04.31 | JT | sometimes there are |
| 09:04.49 | dickyjoe | Hi Jon |
| 09:05.21 | foxpdll | Has anyone make asterisk working thru realtime from database? |
| 09:05.30 | foxpdll | sqlitr fot example |
| 09:05.34 | dickyjoe | Just wondering the best way of setting opermode=AUSTRALIA |
| 09:05.47 | dickyjoe | running 2.6 kernal |
| 09:05.48 | JT | hi |
| 09:06.34 | foxpdll | sqlite |
| 09:09.40 | kv0s | JT: -2 hours see Mi 8. Aug 09:09:17 UTC 2007 |
| 09:09.40 | kv0s | irrsee:~# date |
| 09:09.40 | kv0s | Mi 8. Aug 11:09:24 CEST 2007 |
| 09:10.28 | JT | kv0s: well clearly you're getting UTC output |
| 09:10.29 | kv0s | If i check my voicemail with ari, the correct time is displayed. If i check my voicemail with the sending mails the time is offset about -2 hours. |
| 09:10.31 | JT | which is normal |
| 09:10.45 | kv0s | Mhm. |
| 09:10.56 | kv0s | ARI displays the correct date & time. |
| 09:11.00 | kv0s | The Mail not. |
| 09:11.08 | kv0s | So i must set my time to UTC? |
| 09:11.16 | kv0s | Without any offset? |
| 09:12.06 | *** join/#asterisk guomi (n=francois@c2cpc3.camptocamp.com) |
| 09:12.12 | *** join/#asterisk Uatec_ (n=uatecuk@adsl.ntsols.com) |
| 09:12.28 | foxpdll | can you help me? |
| 09:12.28 | foxpdll | I have asterisk and S-Link DPH-140s sip phone |
| 09:12.28 | foxpdll | i need to authorise phone on asterisk and make call throo it |
| 09:12.28 | foxpdll | but i need make this thru realtime database - ldap |
| 09:12.28 | foxpdll | or throo sqlite database |
| 09:12.29 | foxpdll | had authorise softphone SJphone but D-Link DPH-140s cant |
| 09:12.31 | foxpdll | asterisk write this [Aug 8 13:02:43] NOTICE[10406]: chan_sip.c:14758 handle_request_register: Registration from '<sip:503@192.168.7.130:5060>' failed for '192.168.7.8' - Device does not match ACL |
| 09:13.02 | Uatec_ | i wonder if it's possible to program my phone to play music over the headset while the phone is not in use |
| 09:14.33 | *** part/#asterisk jarod14 (n=jarod14@81.253.51.14) |
| 09:14.53 | creativx | Uatec_: "winamp" |
| 09:14.54 | creativx | :) |
| 09:20.53 | *** join/#asterisk stkn|work (n=stkn|wor@gentoo/developer/pdpc.active.stkn) |
| 09:21.29 | Uatec_ | lol |
| 09:21.35 | Uatec_ | winamp doesn't play through my telephone's headset |
| 09:21.37 | Uatec_ | :( |
| 09:25.32 | *** join/#asterisk Polis_ttt (n=your@194-237-172-225-no48.business.telia.com) |
| 09:27.52 | kv0s | Uatec_: I think this feature doesn't matter asterisk .. ,-) |
| 09:28.31 | *** join/#asterisk shadebob (n=chatzill@84.16.28.38) |
| 09:29.16 | shadebob | Hi, I seach how to convert .gsm to .g729 with the command line... Is this tool exist ? |
| 09:29.19 | foxpdll | had anyone make asterisk worked with realtime database driver |
| 09:31.11 | *** join/#asterisk steliosk (n=Stelios@ipa226.211.tellas.gr) |
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| 09:33.42 | *** join/#asterisk frood (n=dms@hal9000.org.uk) |
| 09:34.13 | tzafrir | shadebob, it will work if your asterisk can transcode gsm to g729 |
| 09:35.10 | frood | I'm using asterisk 1.2 at the moment. When i transfer calls I end up with two sets of WAVs. One for the original call made and one for the transfer with no way of linking the two. Is it possible to record the whole thing as one file? |
| 09:39.07 | Uatec_ | I wonder, how can i test the call quality of a GSM vs alaw? |
| 09:39.31 | Uatec_ | we use alaw athe moment, but somebody suggested that we should move to GSM cos it's smaller and fits down ADSL lines better. |
| 09:39.45 | Uatec_ | i've just got to come up with a way of testing the two |
| 09:40.05 | frood | I use GSM. tiny file size, sounds fine |
| 09:40.06 | creativx | record() |
| 09:40.10 | creativx | or mixmonitor |
| 09:40.21 | creativx | call in yourself and record to gsm and to alaw |
| 09:40.25 | creativx | then.. listen? |
| 09:41.22 | Uatec_ | can you name a windows gsm player? or alaw player? |
| 09:41.45 | creativx | asterisk itself.. playback() |
| 09:41.52 | *** join/#asterisk KermitTheFragger (n=siepkes@118-197.bbned.dsl.internl.net) |
| 09:41.52 | creativx | why not listen to it through the phone |
| 09:42.49 | Uatec_ | i just figured out how to force individual phones to use gsm vs alaw |
| 09:48.46 | *** join/#asterisk shtoom (n=shtoom@221.128.190.158) |
| 09:51.56 | s0ck | i found gsm to be very good quality |
| 09:52.21 | *** join/#asterisk Zhad (n=tom@cpc1-sout6-0-0-cust691.sotn.cable.ntl.com) |
| 09:52.22 | s0ck | and the footprint is like 4KB/s afaik |
| 09:52.39 | Uatec_ | weird |
| 09:52.52 | Uatec_ | i put disallow=all allow=gsm in my sip.conf |
| 09:53.02 | Uatec_ | that should force that sip device to use gsm, right? |
| 09:53.30 | *** join/#asterisk jm|home (n=jm|home@zen.jamiem.com) |
| 09:53.47 | s0ck | might have to tell the phone to use gsm as a priority too |
| 09:54.11 | Uatec_ | but it wouldn't cause the phone to break or anything, |
| 09:54.12 | Uatec_ | would it? |
| 09:54.55 | Zhad | a long long time ago, I saw an article describing how to store queue information into an SQL database (so that if you needed to restart the daemon process, then you wouldn't need to re-login dynamic agents). I'm damned if I can find it now. Does anyone know where I can find this information? |
| 09:54.57 | s0ck | the handsets usually have a list of protocols they will try, ulaw/alaw being top of that list |
| 09:55.37 | Zhad | you should only need to set gsm as an allowed codec in the phone. |
| 09:55.55 | Zhad | if you do a sip debug, it will tell you which bit is going wrong. |
| 09:56.37 | Uatec_ | i'm looking on my phone's config page |
| 09:56.42 | Uatec_ | it says "SDP Payload Type" |
| 09:56.45 | Uatec_ | but doesn't list GSM |
| 09:56.57 | *** join/#asterisk jql (n=jql@12.9a.344a.static.theplanet.com) |
| 09:57.16 | Uatec_ | oh wait |
| 09:57.18 | Uatec_ | found Audio Configuration |
| 09:57.21 | Uatec_ | Prefered Code: |
| 09:57.37 | Uatec_ | loads of G7XXX codecs |
| 09:57.39 | Uatec_ | but no GSM |
| 09:57.43 | Uatec_ | not even ulaw or alaw |
| 09:57.51 | Zhad | G.711u is ulaw |
| 09:57.56 | Zhad | G.711a is alaw |
| 09:59.01 | Zhad | it's masically the signalling differences between American and European ISDN standards (if you mix them up then apparently you can sound like a Dalek). |
| 10:03.00 | Uatec_ | ok |
| 10:03.08 | Uatec_ | lol |
| 10:03.13 | Uatec_ | i'm going to put mine on to ulaw :P |
| 10:03.15 | mamep | hello, i'm having some problems with incoming calls.. anyone can help me? |
| 10:04.01 | Zhad | what's happening? |
| 10:04.25 | mamep | i get an error message in asterisk console |
| 10:04.28 | mamep | Unknown SDP media type in offer: video 10768 RTP/AVP 26 31 34 103 |
| 10:04.41 | Zhad | do you have videosupport=yes in sip.conf? |
| 10:04.48 | mamep | i think it's something with codecs |
| 10:04.52 | mamep | i don't think so |
| 10:04.54 | mamep | let me check |
| 10:04.57 | Zhad | (assuming by the message that this is an incoming video call). |
| 10:05.10 | mamep | videosupport=no |
| 10:05.23 | mamep | under general |
| 10:05.28 | Zhad | what are you using to make the call? |
| 10:05.35 | Uatec_ | I DON'T sound like a dalek |
| 10:05.43 | mamep | local phone number |
| 10:06.02 | Zhad | Uatec> It's very difficult to do with asterisk, if it finds mis-matched codecs it will transcode them. |
| 10:06.03 | Uatec_ | Zhad, which codec is just plain old gsm then? |
| 10:06.12 | Uatec_ | lol, ok then |
| 10:06.13 | Zhad | gsm is gsm sadly. |
| 10:06.17 | adeel | how do i set an incoming 'catchall' context without using freepbx? |
| 10:06.36 | Uatec_ | but ulaw is G.711u, alaw is G.711a |
| 10:06.41 | Uatec_ | doesn't gsm have a name like that? |
| 10:06.54 | Zhad | no |
| 10:06.57 | Uatec_ | cos myphone doesn't offer "gsm" |
| 10:07.00 | Zhad | well it has an ITU reference |
| 10:07.06 | Uatec_ | WTF? how can a phone not offer gsm? |
| 10:07.08 | Uatec_ | i don't understand that |
| 10:07.22 | adeel | Uatec_, there's no law saying that every phone has to offer the same codecs |
| 10:07.38 | adeel | grand stream offers gsm, while polycoms don't |
| 10:07.45 | adeel | just the way the world works |
| 10:07.46 | Uatec_ | i know i know |
| 10:07.52 | Uatec_ | do you know about linksys? |
| 10:07.54 | Zhad | Sadly a lot of new phones don't support G.729 (because of the new licencensing restrictions. |
| 10:08.02 | Uatec_ | particularly the sipura spa922 ? |
| 10:08.14 | adeel | nope, don't use linksys |
| 10:08.42 | Zhad | mamep> what are you using to place the call? |
| 10:08.53 | Uatec_ | i'm already using linksys |
| 10:08.59 | Uatec_ | these phones were the best of the ones we tried |
| 10:09.01 | Zhad | then don't use gsm |
| 10:09.08 | Uatec_ | it looks like i'm not going to be able to |
| 10:09.14 | Zhad | or get asterisk to transcode to gsm if you need to use if for termination |
| 10:09.34 | mamep | zhad? |
| 10:09.45 | Uatec_ | well, i'm making a deal with an ISP, they will provide me with Fancy pants ADSL, and an IAX trunk |
| 10:09.54 | Uatec_ | but they say i should use gsm if i want more calls down the ADSL line |
| 10:10.01 | Zhad | namep> It looks like the client is trying to start a video call. |
| 10:10.24 | mamep | yeah i can get that but why? |
| 10:10.25 | Zhad | gsm or G.726 or G.729 |
| 10:10.42 | mamep | i mean i have a local call in number from my sip provider |
| 10:11.12 | Zhad | and if they support it, (and you have a timing source) set trunking=yes in iax.conf. |
| 10:11.46 | adeel | if someone could take a second and explain something to me real quick, it'd be appreciated....according to the asterisk book, i should use different contexts for incoming and outgoing calls....how do i assign users to use 1 context for incoming and a totally different one for outgoing? |
| 10:11.48 | Zhad | namep> make a sip debug of the incoming call and put it in pastebin. |
| 10:12.13 | mamep | just a sec |
| 10:12.36 | adeel | Uatec_, by using GSM you're doing more compression than is done with G711 |
| 10:12.37 | Zhad | adeel> set incoming as your default codec in iax/conf and sip.conf and set the outgoing codecs in peer declarations. |
| 10:12.48 | Uatec_ | Zhad, they said GSM |
| 10:12.49 | jcaceres | hello, i am trying co to connect two asterisk using two te120p cards, which signalling and switch type should i use? |
| 10:12.52 | Uatec_ | adeel, i know that |
| 10:13.22 | Zhad | Uatec_> Use ulaw on the phone, and gsm with the provider, asterisk will transcode the calls for you. |
| 10:13.39 | Zhad | Uatec_> It can do that (that's why they're called codecs). |
| 10:13.40 | adeel | Zhad, ahhhh...thanks, that definitely makes more sense now |
| 10:14.40 | mamep | Zhad : is there any way to get it in a file? |
| 10:14.40 | mamep | http://www.pastebin.ca/649696 |
| 10:15.42 | adeel | Uatec_, you can also get an idea of how much time it'll take for asterisk to transcode from one codec to another by issuing a 'show translation' |
| 10:16.00 | Zhad | SIP/2.0 404 Not Found is more interesting. |
| 10:16.12 | Uatec_ | lol, Zhad, i know |
| 10:16.24 | Zhad | though from ulaw to gsm and back is very little time at all |
| 10:16.27 | Uatec_ | but i wanted to use gsm internally, to find out what the call quality is like |
| 10:16.40 | adeel | Uatec_, gsm is your cell phone standard quality |
| 10:16.53 | adeel | well, roughly anyway |
| 10:16.54 | Zhad | I also notice the hated ATA |
| 10:17.53 | LakeSolon | Does anyone know if there's a male voice "The number you have dialed is unallocated" recording included w/ Asterisk? I just want to be as sure as possible I'm not generating this error message internally =p |
| 10:17.59 | Zhad | Uatec> you will know what the quality is like when somone calls you. |
| 10:18.00 | LakeSolon | (trixbox, specifically) |
| 10:18.09 | Uatec_ | adeel, yeah, but i wanted some practical exposure to it |
| 10:18.14 | Zhad | the quality wont improve when it is being transcoded back. |
| 10:18.31 | krdian_ | hi |
| 10:18.53 | adeel | Uatec_, ever use a cell phone? that's pretty much what the call will sound like...at best |
| 10:19.00 | Zhad | If you need higher quality with the same sort of bandwidth then G.726/G.729 are abetter bet, but it's at the expense of runtime (at both ends). |
| 10:19.07 | Uatec_ | Zhad, i want to know, before i have to sign a contact though,... don't i |
| 10:19.08 | Uatec_ | contract |
| 10:19.41 | Uatec_ | Zhad, yes i know the quality wont improve again... :P |
| 10:19.42 | adeel | Uatec_, some providers allow you to do inter-op testing and all...but what you should be more concerned about is QoS and if your provider can gaurentee it |
| 10:19.44 | Zhad | you could use a free provider to have a play. |
| 10:20.38 | Uatec_ | adeel, i'm concerned about that too |
| 10:20.44 | Uatec_ | why can't i be concerned with more than one thing? |
| 10:21.12 | krdian_ | is that normal that manager show eventq shoving long list of events ? how to free memory from that ? |
| 10:21.12 | Zhad | mamep> do you have a 2115777777 extension in the context that the peer declaration for i-call |
| 10:21.23 | Zhad | + states. |
| 10:21.42 | adeel | Uatec_, no one is saying you can't be concerned with multiple things...but in my opinion, the QoS is more important that GSM |
| 10:21.53 | Uatec_ | well ok then |
| 10:22.00 | Uatec_ | but someone else is dealing with the QoS of the provider |
| 10:22.01 | Zhad | agreed. |
| 10:24.17 | dickyjoe | can the goto command call another exten in a different context? |
| 10:24.25 | Zhad | yeas |
| 10:24.37 | Zhad | Goto (context,extension,priority) |
| 10:24.51 | Zhad | works as well as Goto(extension,priority). |
| 10:25.11 | adeel | i keep getting this warning message in my CLI output.....res_config_pgsql.c:192 realtime_pgsql: Postgresql RealTime: Could not find any rows in table sip_buddies |
| 10:25.25 | adeel | even though there are rows in sip_buddies |
| 10:25.30 | dickyjoe | ok thanks, what about if its originating from the s extension |
| 10:25.44 | Zhad | dickyjoe> still works |
| 10:26.06 | Zhad | (well, certainly does here). |
| 10:26.48 | dickyjoe | i get: Channel 'Zap/4-1' sent into invalid extension 'internal' in context 'incoming', but no invalid handler |
| 10:28.21 | dickyjoe | ok i didn't enter the priority |
| 10:29.41 | Zhad | :-) |
| 10:30.19 | Zhad | still sounds strange. |
| 10:32.32 | jcaceres | hello, i am trying co to connect two asterisk using two te120p cards, which signalling and switch type should i use? |
| 10:33.28 | JT | jcaceres: what type of circuit and switchtype are you connecting to? |
| 10:33.36 | JT | oh |
| 10:33.39 | JT | two asterisk boxes |
| 10:33.46 | JT | well |
| 10:33.53 | JT | you can choose T1 or E1 |
| 10:33.58 | LakeSolon | Does the number 213-291-1900 hold any special significance to anyone? |
| 10:34.31 | mamep | Zhad : no my number is not 2115777777 |
| 10:35.46 | Zhad | from that sip debug it looks like the provider is trying to forward the call to 2115777777 on your box |
| 10:36.29 | mamep | shit i know what is it.. |
| 10:36.52 | mamep | thx Zhad |
| 10:37.02 | mamep | i used |
| 10:37.04 | mamep | to register |
| 10:37.06 | mamep | in sip.conf |
| 10:37.11 | Zhad | :-) |
| 10:37.29 | Zhad | doh! |
| 10:37.37 | mamep | register => user:pass@sip.i-call.gr/2115777777 |
| 10:37.39 | mamep | that's why |
| 10:37.44 | mamep | now changed to my number |
| 10:37.45 | mamep | :P |
| 10:37.46 | Zhad | that'll do it |
| 10:37.50 | mamep | :) |
| 10:37.52 | mamep | thx bro |
| 10:38.27 | Zhad | with all providers (exckluding sipgate) you can chanis it one that asterisk calls if you accidentally cock up your local calls context so that it you can change the /blahblah to anything you want to reference in extensions.conf if it makes it easier to read. |
| 10:38.48 | jcaceres | JT, i am using E1, but i mean in the zapta configuration file |
| 10:38.51 | Zhad | most providers don't even mind if it's written text (asterisk certainly doesn't). |
| 10:39.05 | jcaceres | <PROTECTED> |
| 10:39.38 | mamep | Zhad : btw is there any way to hide your number when calling out? |
| 10:40.44 | phearless | hey guys |
| 10:40.47 | Zhad | namep> Set(CALLERID(num)='') may work |
| 10:40.50 | phearless | i've got a weird question |
| 10:40.58 | phearless | we've got one office in france |
| 10:41.03 | phearless | one in UK |
| 10:41.15 | frood | can someone explain to me what "lastdata" is in the master.csv file? |
| 10:41.22 | phearless | we've got a french guy in the UK |
| 10:41.37 | phearless | and he wants to be in the "queue" of the french office |
| 10:41.39 | JT | jcaceres: one side takes timing, the other side provides it |
| 10:41.42 | mamep | where i can do thaT? |
| 10:41.50 | phearless | but to receive call the the UK, |
| 10:42.01 | phearless | people need to dial the UK number and then the extension |
| 10:42.26 | phearless | so how can he receive phone calls from the phone "queue" in france ? |
| 10:42.49 | Zhad | phearless> AddQueueMember(<queuename>|howtocallhiminuk). |
| 10:42.52 | Zhad | eg. |
| 10:43.16 | phearless | no, because there is no direct dial to the UK |
| 10:43.26 | Zhad | there doesn't need to be |
| 10:43.33 | phearless | can you explain me ? |
| 10:43.36 | Zhad | set up a trunk. to the uk. |
| 10:43.40 | Zhad | or similar. |
| 10:43.58 | mvanbaak | you mean you have to dial a number in UK, wait for it to be answered and hit some numbers to reach the person ? |
| 10:44.07 | phearless | mvanbaak: exactly |
| 10:44.31 | mvanbaak | phearless: you should add a local channel to the queue and use a dial there which can do this |
| 10:44.53 | mvanbaak | AddQueueMember(<queuename>|Local/uk@agents) |
| 10:45.13 | *** join/#asterisk cpm (n=Chip@pdpc/supporter/active/cpm) |
| 10:46.06 | mamep | shadow*CLI> Set(CALLERID(num)='') |
| 10:46.07 | mamep | No such command 'Set(CALLERID(num)='')' (type 'help' for help) |
| 10:46.08 | phearless | "use a dial there" what do you mean ? |
| 10:46.37 | mvanbaak | in the local you can do: Dial(Zap/g1/<uknumber>,45,D<number to reach person>) |
| 10:46.42 | phearless | mamep: exten => 4,1,Set(CALLERID(name)="Accounting ${CALLERID(name)}") |
| 10:46.47 | phearless | mamep: just an example |
| 10:47.18 | mamep | hmm |
| 10:47.22 | phearless | ok mvanbaak I will think about this... |
| 10:47.27 | mamep | it should be placed in extensions |
| 10:47.27 | mamep | ok |
| 10:49.23 | phearless | thank you mvanbaak , I have checked http://www.voip-info.org/wiki/view/Asterisk+cmd+Dial and the "D" feature seems to be useful |
| 10:51.42 | *** join/#asterisk mcoxall (n=marcc@host217-45-221-53.in-addr.btopenworld.com) |
| 10:53.04 | *** join/#asterisk Tili (n=tili@169.16.221.87.dynamic.jazztel.es) |
| 10:58.36 | dickyjoe | How are the busy, dial and NU tone types changed in asterisk from US to australia |
| 10:59.10 | JT | indications.conf |
| 10:59.34 | mamep | i'm getting this error.. |
| 10:59.35 | mamep | Aug 8 13:10:27 ERROR[65200]: chan_sip.c:11086 handle_request_subscribe: Got SUBSCRIBE for extension 004753769397@external-nikosaei from 147.52.78.15, but there is no hint for that extension |
| 11:01.51 | Zhad | what is 147.52.78.15 ? |
| 11:02.01 | Zhad | other than helios.edu.uoc.gr |
| 11:02.25 | Zhad | actually fwiw, I get that from one of my providers when a registration completes, never did find out why |
| 11:02.29 | mamep | it's the same pc |
| 11:02.38 | mamep | helios is dns entry |
| 11:02.58 | Zhad | weird |
| 11:04.21 | mamep | what? |
| 11:06.19 | Zhad | oh bugger just types restart now into wrong console. |
| 11:06.39 | Zhad | I hope no-one was on a call |
| 11:07.28 | mamep | btw where i can find additional sound to asterisk? |
| 11:11.16 | Zhad | what do you mean? |
| 11:11.46 | Zhad | when building use make menuselect to choose different sound codec packs |
| 11:12.36 | Zhad | and any of the additional sound packages (which can also be downloaded |
| 11:13.02 | Zhad | there are other languages on the internet (iirc freepbx.org has some). |
| 11:13.38 | Zhad | failing that, you can pay allison to make customised ones, or record them yourself. |
| 11:13.57 | *** join/#asterisk masus (n=tet@88.248.14.186) |
| 11:23.07 | *** join/#asterisk r0d3nt (i=nobody@foster.stonedcoder.org) |
| 11:25.11 | *** join/#asterisk dharrigan (n=david@host12.williamhill.co.uk) |
| 11:26.35 | masus | is somtihign like this possible pbx*CLI>dial 110@outbound > 115@outbound |
| 11:27.07 | creativx | JT: still around? |
| 11:27.14 | masus | so the dial is do from the cli and it rings on two extensions |
| 11:28.02 | JT | yes, sometimes |
| 11:28.53 | creativx | JT: i was just wondering what kind of hardphones you like.. that supports SIP |
| 11:29.02 | *** join/#asterisk boch (n=fran@190.48.240.189) |
| 11:29.02 | creativx | i need to get rid of these ip10's from swissvoice, piece of unstable crap |
| 11:29.13 | JT | trick question? ;) |
| 11:29.16 | JT | polycom of course |
| 11:30.35 | creativx | heh no. i just need some phones that doesnt drop calls, hang, fall off the lan, or lock up aka torture the users |
| 11:31.39 | drako | mISDN: INTERNAL ERROR in /usr/src/1_4_asterisk/zaptel-1.4.4/mISDN-1_1_3/drivers/isdn/hardware/mISDN/stack.c:1151 |
| 11:31.39 | drako | release_l1 id 100 |
| 11:31.39 | drako | mISDNd: addr(f0000) prim(f1980) failed err(-92) |
| 11:32.36 | JT | creativx: so yeah |
| 11:32.49 | *** join/#asterisk coppice (n=chatzill@93.195.17.210.dyn.pacific.net.hk) |
| 11:34.54 | creativx | JT: imma look up polys.. i need a simple, cheap, stable one |
| 11:35.13 | creativx | no need for bells and whistles, since all the fun is handled on the pc screen anyways |
| 11:39.27 | Sweeper | creativx: polycom 320 \o\ |
| 11:40.03 | Sweeper | $87 |
| 11:45.07 | *** join/#asterisk lirakis (n=lirakis@cpe-68-175-38-65.nyc.res.rr.com) |
| 11:47.06 | tzanger | morning |
| 11:48.35 | adeel | i keep getting ' Looking for s in incoming' as a sip message...how do i get rid of that? |
| 11:51.47 | drako | is it possible to have a RDSI card with a TDM together? |
| 11:54.00 | *** join/#asterisk kieran491 (n=ehh@c211-28-159-50.brasd1.vic.optusnet.com.au) |
| 11:54.14 | *** part/#asterisk kieran491 (n=ehh@c211-28-159-50.brasd1.vic.optusnet.com.au) |
| 12:00.35 | lirakis | ~pattern |
| 12:00.40 | *** join/#asterisk jesselang|laptop (n=jesse@h75-100-164-249.75-100.unk.tds.net) |
| 12:01.18 | jesselang|laptop | Hello. quick q: Are the AGI changes from r76707 included in 1.4.10? |
| 12:01.48 | jesselang|laptop | I don't see them mentioned in the ChangeLog. |
| 12:01.56 | *** join/#asterisk skeffling (n=andrew@andrew.1ec.aaisp.net.uk) |
| 12:02.50 | lirakis | pattrerns in ANI recogition ..?? exten => 212123456/_212111XXXX ????? |
| 12:02.52 | lirakis | is that right? |
| 12:03.57 | *** join/#asterisk zotz (n=zotz@24.244.163.157) |
| 12:06.29 | *** join/#asterisk [TK]D-Fender (n=joe@MTRLPQ02-1177745839.sdsl.bell.ca) |
| 12:07.45 | skeffling | Hi, I'm looking at using MixMonitor so as to reduce disk io. Is it more efficient to record as native alaw than wav? (we're using alaw on out sip clients) |
| 12:09.27 | JT | not really |
| 12:10.45 | lirakis | morning [TK]D-Fender |
| 12:11.33 | [TK]D-Fender | &yawn& |
| 12:11.44 | *** part/#asterisk jesselang|laptop (n=jesse@h75-100-164-249.75-100.unk.tds.net) |
| 12:12.07 | *** join/#asterisk bird_of_Luck (n=melifaro@secured.by.ipfw.ru) |
| 12:14.04 | drako | [TK]D-Fender, what do you need to get misdn commands on asterisk? |
| 12:15.24 | [TK]D-Fender | drako: chan_misn.so |
| 12:15.28 | [TK]D-Fender | drako: chan_misdn.so |
| 12:15.51 | zeenix | hey guys |
| 12:16.21 | lirakis | morning zeenix |
| 12:16.26 | zeenix | is there anyway i can make asterisk be able to convert gsm to ilbc |
| 12:16.42 | drako | [TK]D-Fender, i installed a b410p and the TDM stopped work, now i don't have neither zap nor misdn on asterisk :/ |
| 12:16.47 | [TK]D-Fender | zeenix: it does this automatically when it has to. |
| 12:17.07 | JT | IF iLBC is installed. |
| 12:17.08 | zeenix | [TK]D-Fender: not for me :( |
| 12:17.14 | [TK]D-Fender | drako: clarify "TMD" please |
| 12:17.21 | [TK]D-Fender | TDM* |
| 12:17.25 | drako | TDM400 |
| 12:17.26 | zeenix | let me fetch the actual error.. |
| 12:17.31 | JT | TDM400P |
| 12:17.38 | drako | yah |
| 12:17.47 | [TK]D-Fender | drako: do your modprobe's again. |
| 12:17.58 | drako | i did and i get error now |
| 12:18.26 | drako | ZT_CHANCONFIG failed on channel 1: No such device or address (6) |
| 12:18.26 | drako | open error on /dev/zap/1: No such file or directory |
| 12:18.42 | *** join/#asterisk shinao1 (n=shinao1@41.205.185.13) |
| 12:18.48 | [TK]D-Fender | drako : go modprobe all of your modules again |
| 12:19.53 | drako | [TK]D-Fender, http://pastie.caboo.se/85946 |
| 12:20.01 | hmmhesays | yes not being able to sleep |
| 12:20.02 | hmmhesays | great fun! |
| 12:20.37 | zeenix | [TK]D-Fender: http://www.paste2.org/p/5614 |
| 12:21.01 | *** part/#asterisk masus (n=tet@88.248.14.186) |
| 12:21.45 | [TK]D-Fender | drako: Verify that its listed in dmesg, if so recompile zaptel and make sure the modules are all listed. |
| 12:22.51 | [TK]D-Fender | zeenix: "show translation" & "show modules like codec" |
| 12:26.33 | hmmhesays | so tired |
| 12:27.13 | tzanger | heh |
| 12:29.28 | zeenix | [TK]D-Fender: the row and column for iblc is only dashes :) |
| 12:29.46 | zeenix | in the out of 'show translation' |
| 12:30.02 | [TK]D-Fender | zeenix: check your modules folder. then redo "make menuconfig" for * and checkt he codec listings |
| 12:30.21 | zeenix | oh! ilbc doesn't seem to be installed :( |
| 12:30.22 | *** join/#asterisk NamNguyen (n=NamNguye@cm246.delta196.maxonline.com.sg) |
| 12:30.23 | NamNguyen | hi |
| 12:30.35 | NamNguyen | is there any FXO/FXS card that i can plug into a laptop? |
| 12:30.40 | [TK]D-Fender | zeenix: Perhaps you were missing a dependency..... |
| 12:31.10 | [TK]D-Fender | NamNguyen: Nothing I've heard of. Go grab a Linksys SPA-3102 instead |
| 12:31.35 | lirakis | i keep seeing " Got SUBSCRIBE for extension 2006@sip-general from xxx.xxx.xxx.xxx, but there is no hint for that extension" in my cli .. but i only get it for 2006 .. none of my other extensions.. what have i missed? |
| 12:31.49 | zeenix | [TK]D-Fender: are you talking about asterisk source dir? |
| 12:31.54 | drako | i don't think any TDM entry on dmesg.... maybe they are on conflicts? |
| 12:32.03 | zeenix | [TK]D-Fender: cause i actually installed it from debian packages |
| 12:32.13 | [TK]D-Fender | lirakis: its not lying, go check your dialplan |
| 12:32.24 | [TK]D-Fender | zeenix: ... |
| 12:32.27 | [TK]D-Fender | ~wglwat |
| 12:32.27 | jbot | wglwat is, like, well, good luck with all that |
| 12:32.58 | *** part/#asterisk NamNguyen (n=NamNguye@cm246.delta196.maxonline.com.sg) |
| 12:32.59 | [TK]D-Fender | drako: Maybe, or maybe the card isn't well seated (got moved while installing the other card, etc.) |
| 12:32.59 | tzanger | ha |
| 12:33.17 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 12:33.28 | hmmhesays | so my cable network is officially going insane |
| 12:33.37 | lirakis | <PROTECTED> |
| 12:34.18 | [TK]D-Fender | lirakis: Thats whats use for PRESENCE. On of your phones it trying to track the status of another extension and you didn't set that up in * then. |
| 12:34.39 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 12:34.44 | lirakis | hmm.. okay |
| 12:35.16 | lirakis | .. so maybe on my gxp-2000 (extension 2006) i have blf set up on it.. and the extension its monitoring doesnt exist |
| 12:35.29 | lirakis | i should disable that |
| 12:36.41 | hmmhesays | or set up that extension |
| 12:36.49 | [TK]D-Fender | lirakis: Or perhaps you should ENABLE you system to let him :) |
| 12:37.04 | tzafrir | zeenix, ilbc is generally removed explicitly from Debian packages, due to licensing issues |
| 12:37.12 | lirakis | hmmhesays: right.. well it must be an old extension that was removed |
| 12:37.16 | [TK]D-Fender | lirakis: Unless that impedes your ability to efficiently slack-off ;) |
| 12:37.49 | lirakis | [TK]D-Fender: lol .. no .. 2006 is a phone that i put in my bedroom.. so .. it doesnt really need any blf set.. lol |
| 12:37.56 | hmmhesays | thats what I miss most about having a day job |
| 12:38.06 | hmmhesays | massive slackery |
| 12:43.16 | lirakis | hrm |
| 12:43.22 | lirakis | i dont seem to have BLF set at all in the phone |
| 12:44.49 | lirakis | ah i got it |
| 12:44.55 | lirakis | .. it was from a different phone.. |
| 12:45.02 | lirakis | that had blf set for 2006 |
| 12:45.23 | *** join/#asterisk Lawbringer (n=Lawbring@84-45-215-247.no-dns-yet.enta.net) |
| 12:48.29 | x86 | morning |
| 12:50.40 | *** join/#asterisk saftsack (n=saftsack@pD9E05769.dip.t-dialin.net) |
| 12:50.50 | hmmhesays | hello |
| 12:53.33 | *** join/#asterisk vgster (n=vgster@host217-45-221-53.in-addr.btopenworld.com) |
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| 12:55.42 | Sci_05 | morning all |
| 12:55.45 | hmmhesays | morning |
| 12:58.00 | *** join/#asterisk waKKu (n=eurulo@unaffiliated/wakku) |
| 13:01.51 | hmmhesays | slow day in here today I guess |
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| 13:07.15 | *** join/#asterisk _bobweever_ (n=_bobweev@adsl-074-185-215-060.sip.msy.bellsouth.net) |
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| 13:13.54 | *** mode/#asterisk [+o lmadsen] by ChanServ |
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| 13:18.30 | *** mode/#asterisk [+o anthm] by ChanServ |
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| 13:18.57 | Sci_05 | ya it sure is |
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| 13:49.44 | lirakis | .. damn subways ... |
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| 13:55.16 | TechBlazer | [TK]D-Fender: Thanks for the info yesterday. After a week or so of trying to get AsteriskNow to work I dumped it, reformated, installed Ubuntu and Asterisk, edited three files and made my first call. |
| 13:55.50 | [TK]D-Fender | TechBlazer: Good to hear... |
| 13:57.27 | JunK-Y | <PROTECTED> |
| 13:58.18 | TechBlazer | My next question is this: For adding/managing users, would it be easier/faster to use either FreePBX or the AsteriskGUI or stay with editing text files? |
| 14:01.40 | *** join/#asterisk mindCrime (n=chatzill@66.83.208.219.nw.nuvox.net) |
| 14:02.00 | krdian_ | hi |
| 14:02.12 | lirakis | TechBlazer: .. when you use either of those.. you buy into "the system" they use |
| 14:02.25 | *** join/#asterisk JunK-Y (n=junky@modemcable223.205-56-74.mc.videotron.ca) |
| 14:02.55 | [TK]D-Fender | TechBlazer: just stick with text files unless you end up with a reall large system where you intend on delegating its managment to a schmuck |
| 14:02.58 | lirakis | TechBlazer: it depends .. on how much/many people will be modifying stuff... and if it will be customized really |
| 14:03.57 | krdian_ | is there any way to flush eventq from memory ? |
| 14:05.57 | TechBlazer | Good ideas. I'll have ~30 users. The text files route seems like a cleaner way of doing things too. |
| 14:07.08 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 14:09.12 | [TK]D-Fender | TechBlazer: Mostly copy& paste anyways. |
| 14:09.42 | [TK]D-Fender | TechBlazer: somet hings may be best split across included files. |
| 14:12.22 | TechBlazer | Nice. I was just reading about include files. Where is a good place to find some examples of config files? |
| 14:13.05 | *** join/#asterisk pigpen (n=mark@fw.seamans.cc) |
| 14:13.47 | pigpen | So..anyone out there with knowledge of the blackberry 7270 and asterisk? Just trying to see how well it works. |
| 14:15.11 | *** join/#asterisk brodiem (n=brodiem@67.110.68.66.ptr.us.xo.net) |
| 14:22.59 | [TK]D-Fender | TechBlazer: there are the samples & docs that * comes with, the BOOK, and the WIKI. |
| 14:23.14 | *** join/#asterisk Mercestes (n=Merceste@rrcs-71-41-157-70.sw.biz.rr.com) |
| 14:28.43 | s0ck | does anyone use cisco 29* series catalysts? |
| 14:35.43 | *** join/#asterisk techie (n=techie@c-67-181-184-170.hsd1.ca.comcast.net) |
| 14:39.17 | GromiTM | Is there a possibility to the change the behavior of the asterisk relatet to callqueues? |
| 14:39.50 | GromiTM | A call agent has to log in to the queue and then he "telephones" with the queue |
| 14:40.23 | GromiTM | He gets automaticly connected to a calling client. |
| 14:41.11 | GromiTM | I would like to have it similar like: Agent logs into the queue and the Telephone ring, if there is some client in the queue. |
| 14:41.17 | GromiTM | Is it possible? |
| 14:41.44 | *** join/#asterisk javar (n=javar@69.79.134.24) |
| 14:44.26 | Ethon | marcus: Das Faxgeraet ist da |
| 14:44.36 | *** join/#asterisk flujan (n=flujan@200-160-115-020.static.spo.ctbc.com.br) |
| 14:44.38 | Ethon | sorry, wrong window |
| 14:46.25 | *** join/#asterisk ManxPower (n=manxpowe@015-854-218.area5.spcsdns.net) |
| 14:47.21 | flujan | [TK]D-Fender: hi... sorry for the disturbance [TK]D-Fender. :) Tomorrow I googled a bit about using hints on the dialplan... so I could ensure a show hints as you said... |
| 14:47.38 | zeenix | [TK]D-Fender: ok! now i have asterisk built (myself) with ilbc installed :) |
| 14:47.53 | flujan | because I am using agi scripts with asterisk, I couldn't do that right? So, there is other option to get a channel state? |
| 14:49.41 | [TK]D-Fender | flujan: You can call 'asterisk -rx "show hints"' or use AMI in there to get them. |
| 14:49.53 | [TK]D-Fender | zeenix: Lesson : Packed * SUCKS |
| 14:50.00 | [TK]D-Fender | packaged* |
| 14:50.22 | [TK]D-Fender | GromiTM: * queues can already work this way. |
| 14:50.34 | zeenix | yup! now do i also need to install the sound files from the tarball? |
| 14:50.34 | flujan | [TK]D-Fender: which command from the AMI do you recommend? I am trying the extensionstate but it is always showing the -1. :( |
| 14:50.46 | zeenix | i get this error: File enter-ext-of-person does not exist in any format |
| 14:51.07 | [TK]D-Fender | zeenix: You can probably just keep using the ones you previously installed, or they should get retrieved automatically during your install |
| 14:51.19 | [TK]D-Fender | flujan: COMMAND - show hints |
| 14:51.20 | zeenix | although /usr/share/asterisk/sounds/enter-ext-of-person.gsm is there |
| 14:51.41 | [TK]D-Fender | flujan: You really need to read some the VERy fine manuals out there ;) |
| 14:51.42 | GromiTM | [TK]D-Fender: ok ... then I did not really understand the documentation. Were should I look to determine the behavior? |
| 14:52.37 | [TK]D-Fender | GromiTM: the sample queues.conf will tell you how to let a caller enter the queue when its empty, and whenever an agent joins the queue they become eligable to be called. |
| 14:52.55 | zeenix | [TK]D-Fender: do i need to tell asterisk where to find the sound files? |
| 14:52.56 | flujan | [TK]D-Fender: hum... but as I said the show hints outputs nothing to me... this is the main question... |
| 14:53.07 | [TK]D-Fender | zeenix: be very careful of your paths. typicaly * stores its sounds in /var/lib/asterisk/sounds |
| 14:53.18 | GromiTM | [TK]D-Fender: ok .... |
| 14:53.21 | GromiTM | thnx |
| 14:53.35 | [TK]D-Fender | zeenix: Look in /etc/asterisk/asterisk.conf this is the file that lays out all of the important paths |
| 14:53.36 | flujan | [TK]D-Fender: when I receive a call I send it using deadagi to my perl script... it connects to the database and contruct the dialplan "on the fly" |
| 14:53.37 | zeenix | [TK]D-Fender: what '*' ? |
| 14:53.54 | [TK]D-Fender | zeenix: 8 = ASTERISK <- |
| 14:53.56 | [TK]D-Fender | * |
| 14:54.17 | ManxPower | Splat: The Next Generation of PBX |
| 14:55.02 | [TK]D-Fender | ManxPower: Prophetic, isn't it? :) |
| 14:56.11 | ManxPower | I don't use * as a shorthand for Asterisk, it confuses people. |
| 14:56.31 | flujan | [TK]D-Fender: because of that I can set up hints on the dialplan... I research a bit and see that to use hints i need something like this: |
| 14:56.55 | flujan | exten => _.XXXXXXX,HINT,do something... |
| 14:56.56 | flujan | right? |
| 14:57.11 | [TK]D-Fender | flujan: No, you cannot use a pattern match. it has to be a FIXED entry |
| 14:57.54 | flujan | [TK]D-Fender: hum... this restriction alone kicks my entire dialplan... I use a lot of pattern matching... :( |
| 14:58.10 | *** join/#asterisk cthorner (n=cthorner@72-254-9-98.client.stsn.net) |
| 14:58.28 | [TK]D-Fender | flujan: this is for your internal extensions related to PHONES. hardcode the stupid things and be done with it. |
| 14:59.49 | *** join/#asterisk fakhir (n=Fakhir@unaffiliated/fakhir) |
| 15:01.04 | *** part/#asterisk dominic1 (n=dob@213.221.82.242) |
| 15:04.06 | drako | how do I uninstall zaptel |
| 15:05.37 | lmadsen | flujan: try #exec which you can use to have a script generate the hints for you instead of typing them in by hand |
| 15:05.54 | ManxPower | flujan: you should generally not use pattern matching for extensions |
| 15:06.00 | ManxPower | It limits you in too many ways |
| 15:06.49 | [TK]D-Fender | flujan: and WONDERFUL things happen when you assume SIP devices & voicemail boxes, etc exist when they don't just because you want to be lazy.... |
| 15:07.45 | ManxPower | WONDERFUL things happen when you realize that an extension is not a device or phone |
| 15:08.01 | *** join/#asterisk hfb (n=hfb@pool-72-87-254-188.lsanca.dsl-w.verizon.net) |
| 15:11.27 | lirakis | what kind of swap space do youguys setup for your asterisk boxes? |
| 15:11.34 | flujan | [TK]D-Fender: the voicemail system and the handle of all calls is sent to a agi script... I create a new system of voicemail messages by my onw, to I record the files and the users can access them via a web interface... :) |
| 15:11.36 | lirakis | or .. disk partition in general |
| 15:11.45 | [TK]D-Fender | lirakis: Whatever is starndard for a linux server. |
| 15:11.53 | lirakis | <PROTECTED> |
| 15:11.55 | flujan | [TK]D-Fender: ManxPower http://pastie.caboo.se/85998 |
| 15:12.12 | flujan | here goes my extensions.conf... How can enable hints on it? |
| 15:12.16 | ManxPower | lirakis: if your machine is swapping much asterisk isn't going to work very well |
| 15:12.23 | flujan | that is my key point. :D |
| 15:12.23 | lirakis | [TK]D-Fender: .. well that varies widely.. some say half the memory.. but I wasnt sure that since asterisk is a very "real time" service that swap is not going to do much good |
| 15:12.33 | [TK]D-Fender | <PROTECTED> |
| 15:12.37 | lirakis | ManxPower: thats my thought.. which is why i ask |
| 15:12.41 | [TK]D-Fender | _. <------------- |
| 15:13.22 | ManxPower | flujan: your dialplan looks like it was written by someone that has never used Asterisk |
| 15:13.30 | Zhad | SuSE used to recommend 4 x the physical memory as a swap partition |
| 15:13.53 | ManxPower | Dialplans are complex things, the dialplan you have will not work in the real world. |
| 15:14.13 | flujan | hum... every call that enters my pbx is routed to the agi... the agi do all the stuff... |
| 15:14.28 | ManxPower | lirakis: I use 2x physical memory, but that is just so if I have to do something that isn't asterisk on the system (like compiling a new verison of Asterisk) |
| 15:14.54 | ManxPower | flujan: you will find more and more things you cannot do with a design like that. |
| 15:15.05 | flujan | using the callerid and the exten incoming.. .I have all the information stored in a database. |
| 15:15.34 | flujan | [TK]D-Fender, ManxPower: which are you sugestion about this dialplan? |
| 15:15.43 | flujan | remove the ._... ok, i can do this... |
| 15:15.53 | flujan | it will not crush my app... :) |
| 15:15.57 | ManxPower | flujan: I suggest you put some hint lines in there. |
| 15:17.51 | flujan | ManxPower: could please show me a example of where and which line i can put to have hints on it? |
| 15:18.08 | ManxPower | flujan: show me a SIP userid |
| 15:18.36 | flujan | all them are five digits numbers |
| 15:18.39 | flujan | 40003 |
| 15:18.42 | flujan | 50000 |
| 15:18.45 | flujan | and so on. |
| 15:19.28 | ManxPower | exten => 40003,1,Hint(SIP/40003) Assuming your extension is 40003 and your SIP userid as listed in sip.conf is 40003 |
| 15:20.09 | [TK]D-Fender | ManxPower: ummmmm....... |
| 15:20.13 | flujan | ManxPower: hum... |
| 15:20.17 | [TK]D-Fender | ManxPower: Go caffeinate :) |
| 15:20.18 | ManxPower | WAIT! |
| 15:20.20 | flujan | [TK]D-Fender: lol |
| 15:20.23 | ManxPower | Let me get some more coffee. |
| 15:20.26 | flujan | ehehe |
| 15:20.27 | ManxPower | [TK]D-Fender: you show him |
| 15:20.32 | b11d|bbl | . |
| 15:20.59 | [TK]D-Fender | b11d: You've made your point ;) |
| 15:21.09 | [TK]D-Fender | flujan: exten => 40003,hint,SIP/40003 |
| 15:21.10 | flujan | ManxPower: [TK]D-Fender : ok... but i have 300 sip peers... i really cannot use the pattern matching? |
| 15:21.14 | b11d | I just cant drop it ;) |
| 15:21.21 | Mercestes | <PROTECTED> |
| 15:21.25 | [TK]D-Fender | flujan: No, you CAN'T. |
| 15:21.26 | b11d | Thank you :) |
| 15:21.36 | Mercestes | np. |
| 15:21.45 | b11d | NOW WHIP ME |
| 15:21.47 | flujan | [TK]D-Fender: this is the kind of trouble i don't want to get into... |
| 15:21.53 | [TK]D-Fender | b11d: ! <- there... I've banged your point... happy? ;) |
| 15:21.58 | b11d | LOL |
| 15:22.06 | b11d | finally some real satisfaction |
| 15:22.16 | ManxPower | Here is an ACTUAL hint from one of my production boxes.: |
| 15:22.17 | ManxPower | exten => 3523,hint,SIP/0004f200cf85-a&SIP/0004f200cf85-b&SIP/0004f200cf85-c |
| 15:22.41 | flujan | to every sip peer i need to set up that line in the extensions.conf? |
| 15:22.56 | ManxPower | flujan: Asking over and over is not going to change the answer. |
| 15:23.06 | flujan | I have something about 40 new users coming or leaving the company per month... |
| 15:23.11 | [TK]D-Fender | ManxPower: Why is ti again that you reg each line-key differently? Was it your co that "shared" phones amongst multiple people? |
| 15:23.35 | [TK]D-Fender | flujan: Do their EXTENSIONS change? |
| 15:24.18 | flujan | [TK]D-Fender: unfortunately yes... I keep all users in the realtime sip_tables to billing purposes... |
| 15:24.27 | ManxPower | [TK]D-Fender: the most basic reason is so *I* control what line appearances calls ring on and on which order they ring on. |
| 15:24.50 | flujan | for billing purposes... :( |
| 15:25.02 | flujan | for instance bob joins the company works here for a month... |
| 15:25.03 | ManxPower | Some of the phones are shared between different people. |
| 15:25.24 | flujan | I bill after bob leave the company, so I need to keep the user bob... :( |
| 15:25.28 | [TK]D-Fender | ManxPower: Imperfect solutions for a less than ideal work environment I guess.... |
| 15:25.54 | flujan | [TK]D-Fender: In a ideal work environment phones will be shared? |
| 15:26.14 | [TK]D-Fender | flujan: Then make a script to generate the hints and reload on some regular basis. Maybe as part of a SQL trigger on DB change. |
| 15:26.31 | [TK]D-Fender | flujan: No, they wouldn't. |
| 15:26.38 | ManxPower | We have "bull pen" aka "diamond mines" which are areas of open desks where 2 people share the same desk and phone. We also have a person with an assistant (or assistants) that have complex call routing patterns, we also have operator phones which also have complex call routing requirements. |
| 15:27.09 | [TK]D-Fender | flujan: For billing you should export your CDR's, etc and create a more long-term tracking code sperate from SIP. |
| 15:27.52 | flujan | [TK]D-Fender: will... good point to the script... I can put all the extensions.conf in a database table right? |
| 15:28.00 | ManxPower | [TK]D-Fender: some users want their first line to ring, if there is no answer go to voicemail, if busy roll to a different extension. |
| 15:28.04 | flujan | [TK]D-Fender: so it will make life easies... :) |
| 15:28.12 | *** join/#asterisk Yourname` (n=chatzill@76-10-128-178.dsl.teksavvy.com) |
| 15:28.13 | ManxPower | You can't really do that with one registration per phone. |
| 15:28.49 | ManxPower | [TK]D-Fender: with my scripts and macros, doing it that way adds very little complexity to the dialplan |
| 15:28.49 | [TK]D-Fender | flujan: No, what I would do is a *nix script (lang of your choice) that will read your other configs and auto-generate a SECONDARY file with your hints that gets INCLUDED by your main. It would then trigger the extensions reload. |
| 15:29.15 | drako | great, now none of the cards work |
| 15:29.38 | [TK]D-Fender | ManxPower: You know... what you jsut described is the NATURAL use of 3 line-keys to a single reg. |
| 15:29.48 | [TK]D-Fender | ManxPower: and requires NO dialplan :) |
| 15:29.51 | Yourname` | Hello, what will I have to look at to be able to make the agent call another number, give a few words, and then TRANSFER the current call? Much like 3way.. except the first party cannot hear the agent talking to someone else. |
| 15:30.04 | [TK]D-Fender | drako: Starting to look like its time to call on Digiu support. |
| 15:30.33 | ManxPower | [TK]D-Fender: Huh? How do you make a 1-reg/n-line phone come back BUSY when only 1 line is in use? |
| 15:30.37 | [TK]D-Fender | Yourname`: That is what is called an ATTENDED RANSFER, and is something normally handled by your PHONE. |
| 15:30.59 | drako | [TK]D-Fender, good idea. |
| 15:31.20 | [TK]D-Fender | ManxPower: OH, you want to INTERPRET it as busy for VM do you? |
| 15:31.29 | [TK]D-Fender | ManxPower: Even though you DO roll-over? |
| 15:32.05 | ManxPower | [TK]D-Fender: actually we need to know of the busy to send the call to some other extension on a different phone |
| 15:32.19 | Yourname` | [TK]D-Fender: Meaning asterisk is not capable of doing that? The agent, while on call with a customer, on the softphone can't dial a number, talk to the person who picks up, and then patch the current call through? |
| 15:32.24 | Mercestes | ChanIsAvail() maybe? |
| 15:33.16 | [TK]D-Fender | ManxPower: So basically try to ring if they are busy and roll-over to another person.... understandable... you COULD do taht with a simple ChanIsAvail Check though and save the Reg complexity. But then again, you have the "shared phone" deal to account for as well... |
| 15:33.42 | ManxPower | [TK]D-Fender: also it is harder to change on the fly. |
| 15:34.06 | [TK]D-Fender | Yourname`: Yes, they CAN, and you should do this on your PHONE. Attended transfers is a PHON job. * can handle it if your phone SUCKS and strangely does not offer this perfectly normal feature. |
| 15:34.23 | ManxPower | I set channel variables, run my macro. The variables can be wet by AGI or govt mind control rays -- my script does not care. |
| 15:34.45 | Yourname` | [TK]D-Fender: Hmm, how about a 3 way and then the agent leaving the 3way? |
| 15:34.46 | ManxPower | wet == sent |
| 15:34.58 | Mercestes | sent == set |
| 15:35.12 | RealBorg | what do I need to enable client authentication? |
| 15:35.19 | [TK]D-Fender | Yourname`: again your PHONE usually determines this. |
| 15:35.20 | Mercestes | RealBorg, the book. |
| 15:35.28 | [TK]D-Fender | Yourname`: You have just described 2 different things |
| 15:35.43 | ManxPower | RealBorg: set context=INVALID in [general], then put the correct context= line for each of the entries in sip.conf |
| 15:35.47 | flujan | [TK]D-Fender: ok, I will try this script. :D |
| 15:35.50 | Yourname` | [TK]D-Fender: Meaning Xlite or Express talk softphones? |
| 15:35.55 | RealBorg | ekiga logs tells me it registered but when I try to make a call I always get "SIP/2.0 407 Proxy Authentication Required" |
| 15:36.13 | ManxPower | make sure you don't actually have a [INVALID] context in extensions.conf |
| 15:36.29 | [TK]D-Fender | Yourname`: 1) and attended transfer. A calls B and asks to pass off their call. B accepts, adn A passses the call onwards. 2) a 3-way call. 3) at the end of a 3-way call, A wants to hang up and leave B and his call TALKING. |
| 15:36.34 | ManxPower | RealBorg: that is normal, the phone will then try the call with auth info |
| 15:36.40 | [TK]D-Fender | Yourname`: meaning whatever the hell you are using. |
| 15:36.50 | Mercestes | well, you can have an invalid, but it'd have to read something like [invalid] _x.,1,Playback(tt-monkeys) _x.,2,Hangup() |
| 15:36.51 | ManxPower | Yourname`: WHAT PHONE DO YOU HAVE? |
| 15:36.52 | [TK]D-Fender | Yourname`: Polycom phones do ALL of these functions remarkably well. |
| 15:37.02 | [TK]D-Fender | ~softphones |
| 15:37.04 | [TK]D-Fender | ~softphone |
| 15:37.04 | jbot | something that should be drug out into the street and shot |
| 15:37.09 | [TK]D-Fender | ^^^^^^^^^^^^^^^^^^^^^^^^^ |
| 15:37.29 | [TK]D-Fender | Yourname`: X-Lite is a flaming piece of shit when it comes to call handling. |
| 15:37.41 | [TK]D-Fender | Yourname`: You wanna be cheap, prepare to BURN for it. |
| 15:37.43 | ManxPower | Yourname`: X-lite does not have an attended transfer feature, IIRC, you need to non-free version for that feature. |
| 15:38.09 | [TK]D-Fender | ManxPower: I seem to specifically recall that as well... |
| 15:38.50 | RealBorg | ManxPower, it does for register but not for invite |
| 15:39.29 | ManxPower | RealBorg: Your problem is a client problem, not an Asterisk problem |
| 15:39.38 | *** join/#asterisk pifiu (n=someone@216.5.79.1) |
| 15:39.44 | Yourname` | [TK]D-Fender: Ah, so softphones ARE capable of doing attended transfers! So, ManxPower, you say Xlite has it in the paid version? |
| 15:40.04 | pifiu | hell everyone |
| 15:40.09 | RealBorg | ManxPower, client works ok with sipgate.at |
| 15:40.17 | Mercestes | pifiu: don't bother correcting it. your right on as it is |
| 15:40.17 | ManxPower | RealBorg: The ONLY thing registration does is inform the server of the IP for that user/password pair. This is so Asterisk can send calls to the correct IP address. It has NOTHING to do with sending calls TO asterisk |
| 15:40.23 | brettnem | Hey all |
| 15:40.41 | all | holy shit |
| 15:40.45 | all | now I'll get mad greetz :) |
| 15:40.48 | ManxPower | Yourname`: as we keep saying you READ YOUR DAMN PHONE DOCUMENTATION |
| 15:41.02 | brettnem | anyone know why some calls being setup on my Asterisk 1.4 setup take about 1-2 seconds before rtp is established.. Happens with a couple of locations on the same box |
| 15:41.15 | Yourname` | ManxPower: We so far used ExpressTalk and XLite |
| 15:41.19 | b11d | are your default routes correct brettnem? |
| 15:41.40 | brettnem | yeah, they are |
| 15:41.42 | b11d | seen that once where there was a fucked up routing table in the core switch |
| 15:41.49 | [TK]D-Fender | Yourname`: SIP is SIP. There are better implementations and worse ones. |
| 15:42.01 | [TK]D-Fender | RealBorg: ... |
| 15:42.03 | [TK]D-Fender | ~sipreg |
| 15:42.06 | [TK]D-Fender | ~sipregister |
| 15:42.06 | jbot | [~sipregister] SIP Registration is to tell your provider as to what IP address & EXTEN to send INCOMING calls to. Some ITSP's let you use a fixed IP or host rather that register. Registration is NOT normally needed to PLACE a call, as those are typically auth'd independently. Others accept UN-AUTH'D calls once you are registered (saves on negotiation BW) |
| 15:42.29 | brettnem | b11d: it happens on several soft phones in a couple different locations... |
| 15:42.34 | Yourname` | [TK]D-Fender: I didn't know it was a phone feature rather than an asterisk feature. Now, my quest is to find the best softphone that does this attended transfer. |
| 15:42.41 | brettnem | b11d: I was wondering if SPI could be causing problems |
| 15:42.57 | ManxPower | [TK]D-Fender: are you shacked up with jbot again? I thought you said he was "emotionally unavailable". |
| 15:43.00 | [TK]D-Fender | Yourname`: Why is it you're hung up on soft-phones? |
| 15:43.17 | Yourname` | [TK]D-Fender: I think for 400 agents, that's the best.. no? |
| 15:43.29 | *** join/#asterisk ZaVoid (n=zavoid@host-24-225-239-34.patmedia.net) |
| 15:43.33 | [TK]D-Fender | Yourname`: No, I don't |
| 15:43.35 | ZaVoid | morning all |
| 15:43.37 | ZaVoid | hey fender |
| 15:43.59 | [TK]D-Fender | Yourname`: I think that for the job you should give an appropriate tool. |
| 15:44.18 | pifiu | What do you call the feature where someone can pickup a phone, talk into it and all of the other phones act as speakers? Sort of like in a target, or walmart, when they pickup a phone and say "call for bob on line 1, come to the front please" |
| 15:44.23 | [TK]D-Fender | Yourname`: A person stuck on the phone all day should have a good REAL phone to hold and use |
| 15:44.33 | cpm | pifiu, paging |
| 15:44.35 | [TK]D-Fender | pifiu: Pageing |
| 15:44.43 | [TK]D-Fender | pifiu: "show application page" |
| 15:44.46 | Yourname` | [TK]D-Fender: Takes up too much time in picking up, hanging up. |
| 15:44.51 | pifiu | and this is possible with asterisk and polycom 501s? |
| 15:45.01 | ManxPower | Yourname`: The larger the number of users you have, the less softphones are right for the job. |
| 15:45.06 | [TK]D-Fender | Yourname`: 1 button on my CSR's phone. |
| 15:45.27 | [TK]D-Fender | Yourname`: And its even lit up when in use. |
| 15:45.35 | [TK]D-Fender | pifiu: All Polycom's |
| 15:45.40 | zeeesh | my server already authenticates through callerid number .. i hv already complete data at mysql ... do i need to just add sip.conf and extensinos.conf at .. mysql ???? |
| 15:46.47 | ManxPower | zeeesh: Saying that something "authenticates thru callerid" is like saying someone authenticates thru the color of their hair". |
| 15:46.58 | Yourname` | [TK]D-Fender: I guess one day, I'll do that. Use some Polycoms for this. |
| 15:47.23 | ZaVoid | anyone ever use sox much? |
| 15:47.30 | Yourname` | However, in your knowledge, does anyone know what softphones have the attended transfer feature? |
| 15:48.15 | *** join/#asterisk wunderkin (n=wunderki@ip72-223-86-126.ph.ph.cox.net) |
| 15:48.34 | [TK]D-Fender | Yourname`: eyeBeam |
| 15:48.54 | *** join/#asterisk Defraz (n=t0tal@24-116-152-177.cpe.cableone.net) |
| 15:49.37 | Yourname` | [TK]D-Fender: And they refer to it as "attended transfer" or what shall I be looking for exactly? |
| 15:50.03 | ManxPower | Yourname`: there are two names for the feature. "Attended Transfer" and "Supervised Transfer" |
| 15:50.31 | ManxPower | That is what you should be looking for. |
| 15:50.33 | zeeesh | <ManxPower>: so what to do ... for adding all of my mysql data ... for using .. realtime asterisk ... i think there is just missing sip.conf and extensins.con and iax.conf ... ? |
| 15:50.33 | Defraz | Weird issue and I might be missing something: When I call number that is disconnected, It seems to ring and ring and ring, I am using a VoIP Provider using SIP trunks. When I call it with a land line or cell phone it gives me the disconnect notice. |
| 15:50.38 | Defraz | Am I missing some config. |
| 15:51.50 | ManxPower | zeeesh: I really can't understand a word you are saying. I cannot help with realtime or databases |
| 15:51.53 | Yourname` | ManxPower: Thanks. |
| 15:52.04 | Yourname` | Hmm, looks like eyeBeam 1.5 basic should cut it. |
| 15:54.28 | *** join/#asterisk kombi (n=kombi@213.160.14.18) |
| 15:54.45 | *** join/#asterisk mintee (n=mintee@72-165-177-90.dia.static.qwest.net) |
| 15:55.15 | kombi | capi hanging up in state 4 <- what might it mean? |
| 15:55.51 | kombi | ...avm b1 card working nicely in msn mode but not in did mode |
| 15:56.06 | flujan | [TK]D-Fender: ManxPower : [TK]D-Fender based on your sugestion... here goes the pastie... |
| 15:56.13 | zeeesh | <ManxPower>: will u pls guide how to configure realtime asterisk with using mysql... |
| 15:56.17 | flujan | [TK]D-Fender ManxPower: http://pastie.caboo.se/86024 |
| 15:56.26 | flujan | does it will set up hints? |
| 15:56.47 | ManxPower | zeeesh: No. |
| 15:56.58 | [TK]D-Fender | flujan: I never said you had to do more than make HINTS for your extens. I said noting about FIXING the fact you pattern match what you DIAL for them. |
| 15:57.24 | kombi | how does one switch on debug logging again? |
| 15:57.42 | [TK]D-Fender | zeeesh: This is all documented on the WIKI, in BOOK, and on a dozen other guides on-line. Get off your ass and get reading. No-one is going to want to hand hold you through this for free. |
| 15:57.43 | *** join/#asterisk gammah (n=gammah@70-253-197-131.ded.swbell.net) |
| 15:58.03 | gammah | Im trying to find docs/rfcs on sccp - the skinny protocol |
| 15:58.10 | gammah | anyone have any resources? |
| 15:58.28 | [TK]D-Fender | gammah: Not sure on RFC... its a proprietary protocol. |
| 15:58.31 | kombi | skinny is cisco |
| 15:58.39 | *** join/#asterisk fordfrog (n=fordfrog@gentoo/developer/fordfrog) |
| 15:58.43 | gammah | well, documentation then |
| 15:58.45 | flujan | [TK]D-Fender: ok, so only this exten => user.name,hint,SIP/user.name will be enough? in the users.conf file? |
| 15:58.46 | lirakis | gammah: skinny = cisco = closed |
| 15:58.53 | gammah | I know it's cisco |
| 15:58.59 | lirakis | cisco = closed |
| 15:59.17 | gammah | lirakis: I GET IT |
| 15:59.23 | zeeesh | <[TK]D-Fender>: thanks .. and my pleasure .. how ... nice guys with best attitude ... i was just asking a weblink ... not telling to teach me ... |
| 15:59.25 | [TK]D-Fender | flujan: sip.conf, FORGET users.conf it is another flaming pile of crap. |
| 15:59.27 | gammah | doesn't mean someone hasn't reversed it enuff to know about it |
| 15:59.56 | *** join/#asterisk phillipk (n=pkey@216.248.143.87) |
| 16:00.00 | gammah | didn't know if someone who worked on channels/chan)skinny.c might have some pointers |
| 16:00.29 | flujan | [TK]D-Fender: I do not understand... the sip.conf is already used by asterisk... the users.conf file is to include in the extensions.conf to enable hints... |
| 16:00.43 | Mercestes | Zeesh: Might I suggest a consultant? |
| 16:00.50 | gammah | what? |
| 16:00.53 | gammah | oh nm |
| 16:01.01 | mintee | can asterisk act as a standalone voicemail box? Users can call in to their number, setup a voicemail, and it's only for incoming voicemails that they can check reguarly? Answering on the first or second ring? |
| 16:01.04 | Mercestes | Zeeesh, gammah.....i could see how those are simliar. |
| 16:01.12 | [TK]D-Fender | flujan: users.conf is an completely separate *1.4 config file which should not be used. |
| 16:01.19 | Mercestes | They both end in h. |
| 16:01.27 | gammah | <PROTECTED> |
| 16:01.32 | [TK]D-Fender | mintee: You can do this iwth *, yes... |
| 16:01.37 | Mercestes | gammah, blame society |
| 16:01.37 | flujan | [TK]D-Fender: ok... will use other filename |
| 16:01.43 | gammah | I blame your momma |
| 16:01.49 | mintee | [TK]D-Fender, cool, thanks |
| 16:02.20 | Mercestes | gammah, ....so you're...what? 14? 15? |
| 16:02.41 | gammah | :) |
| 16:02.42 | ZaVoid | fender? |
| 16:03.53 | gammah | ah sweet, the messages at least are all spec'd as structs in chan_skinny |
| 16:04.59 | ZaVoid | little confused |
| 16:04.59 | flujan | [TK]D-Fender: puff... here, I changed it again... :) hope it is right now. |
| 16:05.02 | flujan | [TK]D-Fender: http://pastie.caboo.se/86024 |
| 16:05.05 | ZaVoid | does asterisk use .wav by default or .gsm? |
| 16:05.06 | ZaVoid | http://www.voip-info.org/wiki/view/Asterisk+sound+files |
| 16:06.10 | ManxPower | ZaVoid: If the file exists in the same format as the calling channel, it will use that format, otherwise it will transcode whatever format the file is in. |
| 16:06.20 | ZaVoid | thats what i thought |
| 16:06.20 | [TK]D-Fender | flujan: Yead, that looks a lot better, though I might suggest you give it a more meaningful name like "extensions-hints.conf" |
| 16:06.26 | [TK]D-Fender | flujan: Just a thought... |
| 16:06.27 | ZaVoid | maybe the wav format i recorded in isn't working right |
| 16:06.35 | flujan | [TK]D-Fender: thanks... |
| 16:06.36 | ZaVoid | manx you ever use sox to convert? |
| 16:06.48 | ManxPower | ZaVoid: it needs to be mono, not stereo and 8Khz |
| 16:06.52 | ZaVoid | (except for g.723) |
| 16:07.04 | [TK]D-Fender | ZaVoid: Generally its easiest when you USE * to make your recordings. |
| 16:07.12 | pifiu | fender any info on where to read more up on paging on polycoms? |
| 16:07.35 | pifiu | http://www.voip-info.org/wiki/view/Asterisk+Paging+and+Intercom |
| 16:08.20 | ManxPower | ZaVoid: I used sox once to convert |
| 16:08.49 | ZaVoid | yeah i can't use asteirsk in this case.... |
| 16:08.56 | [TK]D-Fender | pifiu: Go follow that... you |
| 16:09.01 | [TK]D-Fender | re on the right trail. |
| 16:09.02 | ZaVoid | mono 8kh ok hold on |
| 16:09.03 | ZaVoid | thanks manx |
| 16:09.46 | lirakis | ZaVoid: its quick to use sox to "clean" a wav |
| 16:10.08 | ZaVoid | oh? |
| 16:10.17 | ZaVoid | can ya gimme a sample command line for it to do that? |
| 16:10.35 | lirakis | ZaVoid: sox your.wav -r 8000 /var/lib/asterisk/sounds/your.gsm |
| 16:10.47 | ZaVoid | ok |
| 16:10.54 | ZaVoid | what if i wanted to do all the files in a dir? |
| 16:11.00 | ManxPower | SOME versions of sox do not generate good GSM files, BTW. |
| 16:11.21 | ZaVoid | ok |
| 16:11.27 | ZaVoid | if the are in wav i thin that would or g729 |
| 16:11.39 | ManxPower | ZaVoid: sox does not support G729 or G723.1 |
| 16:11.56 | ZaVoid | oh ok |
| 16:11.57 | ManxPower | .WAV files are usually ulaw |
| 16:12.05 | lirakis | ZaVoid: bash script... for i in `ls *.wav`; do sox $i -r 8000 /var/lib/asterisk/sounds/$i.gsm; done |
| 16:12.13 | ZaVoid | thanks man |
| 16:12.18 | Corydon76-work | ManxPower: uh, they are? |
| 16:12.46 | ManxPower | Corydon76-work: perhaps "ulaw" is not the correct term, as it is a codec. |
| 16:12.49 | *** join/#asterisk ToyMan (n=Stuart@host10.chelsmoreip.c.subnet.rcn.com) |
| 16:12.50 | Corydon76-work | ManxPower: ulaw is an 8-bit encoding. WAV files are usually encoded as 16-bit |
| 16:13.20 | ManxPower | [root@pbx-1 asterisk]# file hurricane.wav |
| 16:13.20 | ManxPower | hurricane.wav: RIFF (little-endian) data, WAVE audio |
| 16:13.26 | ManxPower | Corydon76-work: Ah. |
| 16:13.35 | Corydon76-work | WAV tends to be signed linear |
| 16:15.24 | ZaVoid | hmm reduced the file size to 44 |
| 16:15.32 | *** join/#asterisk Qwell[] (i=qwell@pdpc/sponsor/digium/Qwell) |
| 16:15.32 | *** mode/#asterisk [+o Qwell[]] by ChanServ |
| 16:15.51 | ZaVoid | thats small |
| 16:16.11 | Corydon76-work | Uh, 44 is too small |
| 16:16.20 | ManxPower | ZaVoid: download and install sox from source. |
| 16:16.22 | Corydon76-work | 44k maybe? |
| 16:16.36 | ManxPower | I vaguely recall 44 bytes meant "sox screwed up" |
| 16:16.50 | coppice | 44 == basic wave file header |
| 16:17.00 | *** join/#asterisk ToyMan (n=Stuart@host10.chelsmoreip.c.subnet.rcn.com) |
| 16:17.14 | Corydon76-work | gsm files are all a multiple of 66-bytes |
| 16:17.22 | ManxPower | ZaVoid: "sox -h" is gsm listed? |
| 16:17.49 | ZaVoid | yeah it is |
| 16:17.53 | ZaVoid | Supported file formats: aiff al au auto avr cdr cvs dat vms gsm hcom la lu maud nul ossdsp prc raw sb sf sl smp sndt sph 8svx sw txw ub ul uw voc vorbis vox wav wve |
| 16:18.01 | ManxPower | I still say download and install from source. |
| 16:18.23 | Corydon76-work | The other possibility is that you're using an unrecognized codec inside WAV |
| 16:18.31 | ZaVoid | yeah |
| 16:18.34 | ZaVoid | let me try a few things |
| 16:18.36 | ZaVoid | thanks guysx |
| 16:19.13 | Corydon76-work | After all, wav simply specifies a file encapsulation method. There are multiple codecs that can be encoded within |
| 16:20.03 | ZaVoid | right |
| 16:21.14 | *** join/#asterisk guillote_GNU (n=guillote@190.7.27.17) |
| 16:24.05 | *** join/#asterisk qdk (n=qdk@0x50c627be.bynxx11.adsl-dhcp.tele.dk) |
| 16:28.48 | *** join/#asterisk tsurko (n=tsurko@77.70.15.52) |
| 16:29.01 | *** join/#asterisk iBuMp- (n=ibump@cpe-66-68-37-190.austin.res.rr.com) |
| 16:29.15 | *** join/#asterisk Qwell[] (i=qwell@pdpc/sponsor/digium/Qwell) |
| 16:29.15 | *** mode/#asterisk [+o Qwell[]] by ChanServ |
| 16:31.21 | *** join/#asterisk Voicemeup (n=VoiceMeU@modemcable159.131-56-74.mc.videotron.ca) |
| 16:31.45 | Voicemeup | any reason i get dupes in chanel name. ? its sip/user-RANDOM but i get more then 2.. i get like 10 of those same for one day |
| 16:31.53 | Voicemeup | shoudnt it be random like can we add bits ? |
| 16:32.43 | *** join/#asterisk doolph (n=doolph@200.115.147.74) |
| 16:32.46 | doolph | hello |
| 16:33.04 | Mercestes | Voicemeup, it's only unique in that no two identifiers should be up at the same time. |
| 16:33.04 | doolph | I just installed asterisk 1.4 and I lost all sip* commands in the cli any idea? |
| 16:33.16 | Voicemeup | ah |
| 16:33.20 | Mercestes | Voicemeup, it is not unique forever, once that call ends that identifier may be used again |
| 16:33.48 | Mercestes | doolph, some retard completely redid the syntax on everything. Try core sip whatever and see if that works. |
| 16:33.54 | Voicemeup | ok |
| 16:34.06 | Voicemeup | ok then why only 8 bit ? |
| 16:34.07 | doolph | well it doesnt work |
| 16:34.10 | Voicemeup | its hec right ? |
| 16:34.14 | Voicemeup | hex |
| 16:34.22 | Mercestes | Voicemeup, I dunno. |
| 16:34.25 | Voicemeup | hmm ok |
| 16:34.30 | Mercestes | Voicemeup, you see any letters higher than F? |
| 16:34.33 | Voicemeup | any other option i have to uniq each Call ( all legs) |
| 16:34.58 | Voicemeup | i need a way to find all legs of a call.. and right now that was only thing |
| 16:35.06 | Voicemeup | unless i geenrate a callid myself on each call |
| 16:35.09 | *** join/#asterisk |dennis| (n=dennis@200.32.236.10) |
| 16:35.17 | Mercestes | Voicemeup, if you want it unique for each call I would say Rand() is not the way to go with that. |
| 16:35.29 | doolph | omg |
| 16:35.32 | Mercestes | and....There is only one cdr per call anyways as far as I know. |
| 16:35.37 | doolph | why is upgrade so hard to do |
| 16:35.39 | Voicemeup | hmm no |
| 16:35.44 | Voicemeup | we have 2-3 boxes per call |
| 16:35.51 | Voicemeup | EX: Auth box -> ser -> pri box |
| 16:35.51 | Mercestes | then your screwed |
| 16:35.54 | Voicemeup | that one example |
| 16:35.56 | Mercestes | set an account code |
| 16:35.58 | Voicemeup | so ill had a header |
| 16:36.05 | Voicemeup | ahah we have acocuntcode |
| 16:36.11 | Mercestes | there you go |
| 16:36.15 | Mercestes | I still wouldn't use Rand() |
| 16:36.16 | Mercestes | I'd increment |
| 16:36.17 | Voicemeup | but you could have an office with 10 people calling same number from same callerid in same time |
| 16:36.23 | Mercestes | ok... |
| 16:36.24 | Voicemeup | ah |
| 16:36.27 | Voicemeup | true |
| 16:36.29 | Mercestes | Still use account code |
| 16:37.22 | pifiu | wow so provisioning polycoms is super fuckign easy now?! |
| 16:37.29 | Mercestes | pifiu, yes. |
| 16:38.53 | doolph | so upgrading from 1.2 to 1.4 |
| 16:39.08 | pifiu | i RAELLY need to redo my setup |
| 16:39.17 | pifiu | one thing i remember doing ages ago was the dial plan? |
| 16:39.46 | pifiu | to make it autodial after 10 digits without hitting send |
| 16:39.46 | doolph | guys |
| 16:39.51 | pifiu | is that standard now in the new versions? |
| 16:41.11 | doolph | can I use the same extensions.conf from asterisk 1.2 to asterisk 1.4 ??? |
| 16:45.08 | doolph | cp: cannot stat `.libs/libchan_h323.so.1.0.1': No such file or directory |
| 16:45.09 | doolph | make[1]: *** [install] Error 1 |
| 16:45.10 | doolph | wow |
| 16:45.20 | doolph | is that bug still there |
| 16:45.55 | *** join/#asterisk msetim (n=marcos@200.195.161.164) |
| 16:47.38 | [TK]D-Fender | pifiu: Depends on your phones... |
| 16:48.17 | _Sam-- | [TK]D-Fender : what level of "set verbose" will make it so i cant see the presence changes on console, but can still see incoming calls and stuff? |
| 16:48.22 | _Sam-- | i tried 1 and that doesnt show enough |
| 16:48.26 | _Sam-- | and 2 shows too much |
| 16:48.38 | [TK]D-Fender | _Sam--: Not sure. Might not be possible. |
| 16:49.07 | _Sam-- | ok, thanks. |
| 16:49.09 | [TK]D-Fender | _Sam--: "core set verbose 1.5" ? ;) |
| 16:49.14 | _Sam-- | lol |
| 16:51.50 | *** join/#asterisk fakhir (n=Fakhir@unaffiliated/fakhir) |
| 16:52.09 | *** join/#asterisk yebo (i=private@mongoose.hypa.net) |
| 16:57.04 | *** join/#asterisk grantm (n=grantm@kolob.wingateservices.com) |
| 16:57.21 | *** part/#asterisk msetim (n=marcos@200.195.161.164) |
| 16:58.34 | *** join/#asterisk ToyMan (n=Stuart@host10.chelsmoreip.c.subnet.rcn.com) |
| 17:02.34 | LakeSolon | Anyone played with the Grandstream 503? |
| 17:03.04 | LakeSolon | Know if it propagates CID info from PSTN via SIP to Asterisk? |
| 17:04.59 | *** join/#asterisk tsurko (n=tsurko@77.70.15.52) |
| 17:06.13 | Dr-Linux | _Sam--: hey |
| 17:06.21 | Dr-Linux | http://www.topix.com/forum/city/welch-wv/TIHLQ9J46LPHP3714 |
| 17:06.28 | Dr-Linux | :( |
| 17:06.29 | _Sam-- | heya Doc...i thought i saw ya recently someplace else too! |
| 17:06.44 | _Sam-- | maybe #rhel? |
| 17:06.52 | Dr-Linux | _Sam--: yes, you are right :P |
| 17:06.55 | Dr-Linux | i was there |
| 17:07.00 | _Sam-- | how you been bud? |
| 17:07.13 | Dr-Linux | i'm good thanks |
| 17:07.17 | Dr-Linux | yourself? :) |
| 17:08.25 | _Sam-- | Dr-Linux : really good thanks, just too busy as usual. |
| 17:08.30 | *** part/#asterisk Defraz (n=t0tal@24-116-152-177.cpe.cableone.net) |
| 17:08.39 | _Sam-- | what are you doing with rhel? |
| 17:09.44 | ZaVoid | hey so sounds fils 8khz mono |
| 17:09.46 | ZaVoid | any other setting for it? |
| 17:12.33 | *** part/#asterisk Voicemeup (n=VoiceMeU@modemcable159.131-56-74.mc.videotron.ca) |
| 17:15.40 | LakeSolon | I don't suppose anyone has a link off the top of their head that's got a good overview of what a 1.2.24 -> 1.4.10 migration would be like? |
| 17:16.08 | mvanbaak | upgrade.txt |
| 17:16.51 | Corydon76-work | If you're been paying attention to all of the deprecation notices in 1.2 and aren't using any of those items, then an upgrade should be seamless |
| 17:17.18 | Corydon76-work | However, if you've been ignoring the deprecation notices, then stuff may not work |
| 17:17.32 | LakeSolon | kk, ty |
| 17:18.20 | *** join/#asterisk gerphimum (n=trekkie@cpe-70-125-148-108.satx.res.rr.com) |
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| 17:32.34 | generalhan | hey all ! |
| 17:32.49 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 17:34.25 | generalhan | i have a couple of remote phones connecting via SIP, and when they dial out or get called from here there is no ringing, thought the CLI shows that extension being called... i have port 5060 being forwarded at my local router, but do i also need to forward the RTP range to the * box ? |
| 17:37.01 | Zhad | Y |
| 17:37.56 | Juggie | generalhan, yes. |
| 17:38.06 | Juggie | you need to forward whatever is defined within rtp.conf to your * box |
| 17:38.18 | generalhan | Juggie: ok thanks ill give that a shot ! |
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| 17:38.44 | generalhan | Juggie: did you go by a different name in here a couple years ago ? |
| 17:38.51 | Juggie | no |
| 17:39.13 | generalhan | Juggie: ok just wondering ... your responses sound familiar to me like ive seen them before |
| 17:39.28 | Juggie | i think i helped you with something a few weeks ago |
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| 17:39.59 | generalhan | Juggie: yes and that is what i was referring to |
| 17:40.38 | *** join/#asterisk Op3r (n=Op3r@121.97.147.190) |
| 17:42.03 | Juggie | you said years though :() |
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| 17:42.53 | generalhan | Juggie: i meant that in talking to you a short while back and just now, your responses sound like some one that i remember talking to years ago in here ! |
| 17:43.16 | generalhan | Juggie: and thank you, because changing the RTP ports that were being forwarded made everything fall back into place !! |
| 17:43.40 | Juggie | oh, nope, i've allways been me. |
| 17:43.42 | Juggie | your welcome. |
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| 17:44.30 | doolph | do i need to enable dummy with zaptel 1.4.4 ??? |
| 17:45.06 | Juggie | if you dont have a zaptel card and you want to use meetme or iax trunking, yes. |
| 17:45.27 | doolph | how can I enable it? |
| 17:45.59 | Juggie | did you compile and install zaptel? |
| 17:46.21 | doolph | im doit right now |
| 17:46.38 | Juggie | what linux distribution? |
| 17:46.45 | doolph | centos5 |
| 17:47.04 | Juggie | after you do a make install, try '/sbin/service zaptel start' |
| 17:47.42 | doolph | done |
| 17:48.02 | doolph | there's no answer |
| 17:48.09 | Juggie | no answer? |
| 17:48.09 | Op3r | if you dont have a digium card compile ztdummy |
| 17:48.16 | doolph | nope |
| 17:48.16 | Op3r | for meetme |
| 17:48.22 | doolph | Op3r yes... how |
| 17:48.25 | Juggie | doolph, explain no answer. |
| 17:48.29 | Juggie | i dont know what you mean |
| 17:48.36 | doolph | [root@cl-t064-160cl zaptel-1.4.4]# /sbin/service zaptel start |
| 17:48.40 | Op3r | doolph: before u compile |
| 17:48.43 | doolph | enter... and nothing coming up |
| 17:48.51 | doolph | what file should i edit |
| 17:48.53 | Op3r | doolph: before u compile zaptel you must enable ztdummy |
| 17:49.10 | Op3r | !ztdummy |
| 17:49.13 | Op3r | errr |
| 17:49.14 | Op3r | wait |
| 17:49.14 | doolph | i think its enabled by default |
| 17:49.16 | generalhan | All i had to do without hardware with CentOS was 'modprobe ztdummy' |
| 17:49.19 | doolph | hwo can i test it |
| 17:49.25 | generalhan | after modprobe zaptel |
| 17:49.32 | generalhan | but this was MANY versions ago |
| 17:49.32 | doolph | ah ok |
| 17:49.33 | Op3r | modprobe zaptel then modprobe ztdummy |
| 17:49.46 | doolph | did that |
| 17:49.47 | doolph | no errors |
| 17:49.53 | Op3r | ok |
| 17:50.02 | Op3r | whats in your zapata.conf? |
| 17:50.14 | Juggie | Op3r, ztdummy is enabled by default |
| 17:50.16 | doolph | i dont have file yet |
| 17:50.24 | generalhan | lol |
| 17:50.29 | Op3r | lol |
| 17:50.41 | Op3r | Juggie: for 1.4 its enabled by default I think |
| 17:53.33 | _Sam-- | [TK]D-Fender : you dont know any fix for all the presence changes on the console, and you have them too? |
| 17:53.48 | _Sam-- | its crazy, if one phone changes status, it generates like 15 lines on my console since i have a lot of watchers of that phone |
| 17:54.13 | [TK]D-Fender | _Sam--: that the thing... there nothing to FIX, its not BROKEN. It gives you detail. |
| 17:54.14 | _Sam-- | so all i can see are the damn presence changes...and if i lower the verbosity, i dont even see any incoming calls happening |
| 17:54.30 | [TK]D-Fender | _Sam--: www.drphil.com <--------------- |
| 17:55.20 | _Sam-- | the feature request would be to be able to configure the presence changes |
| 17:55.32 | _Sam-- | right now i dont know how you can say you like how it works and what is displayed |
| 17:56.00 | _Sam-- | if 1 phone changes state, it generates 1 line on the console for any sip clients that may be watching its state |
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| 17:56.06 | *** mode/#asterisk [+o mog] by ChanServ |
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| 18:05.25 | [TK]D-Fender | _Sam--: so you ignore a few lines... big deal.... |
| 18:05.33 | [TK]D-Fender | _Sam--: tahts what scroll-back is for. |
| 18:05.39 | _Sam-- | how do i ignore a few lines if all my screen is full of presence change? |
| 18:05.47 | _Sam-- | it should be ONE line for each presence change, not for each watcher. |
| 18:05.57 | Yourname` | [TK]D-Fender: Features.conf seems to have the attended transfer function!! :) |
| 18:05.59 | *** join/#asterisk Mercestes (n=Merceste@rrcs-71-41-157-70.sw.biz.rr.com) |
| 18:06.04 | _Sam-- | the fact that Ext111 changed to ringing from idle i dont need to see for each watcher |
| 18:06.19 | [TK]D-Fender | Yourname`: yes it does... not DTMF based crap. Go for it. |
| 18:06.27 | [TK]D-Fender | s/not/more/ |
| 18:07.19 | _Sam-- | do you at least acknowledge what im saying? |
| 18:07.39 | _Sam-- | if you are watching the console -- and extXXX changes state -- there is no reason i need to see that state change listed 15 times if i have 15 watchers |
| 18:07.46 | [TK]D-Fender | _Sam--: Yes it displays the info just like its supposed to and you don't like it. I fully acknowledge this :) |
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| 18:08.26 | [TK]D-Fender | _Sam--: Feel free to place a bounty on having it recoded, or do it yourself. |
| 18:08.29 | _Sam-- | ok fair enough -- as long as you comprehend what im trying to convey. i cant see how any busy system with more than 2 phones changing state, and someone watching the console, i dont know how that person isnt offended |
| 18:08.51 | _Sam-- | over 95% of my console output is now related to presence |
| 18:09.17 | _Sam-- | i will see if i can pay zoa to fix mine |
| 18:09.23 | _Sam-- | !seen zoa |
| 18:09.34 | Yourname` | [TK]D-Fender: More DTMF crap? A lot of it depends on the softphone's DTMF, right? |
| 18:09.39 | [TK]D-Fender | _Sam--: Then nobody is placing calls and EVERYBODY is spying on everybody else.... now tell me... who's getting any work done there? ;) |
| 18:09.54 | _Sam-- | not spying on anyone -- but everyone's phone shows the state of everyone else. |
| 18:10.10 | _Sam-- | so an employee can look at their phone, and see if their co-worker is already on the line |
| 18:10.29 | [TK]D-Fender | Yourname`: Only the shit ones that don't have standard functionality built in and * is being asked to compensate for. Your dial strings are going to look like alphabet soup before long... |
| 18:10.34 | _Sam-- | in the old days it was called "BLF" |
| 18:10.38 | _Sam-- | i dont know what it is anymore |
| 18:10.50 | [TK]D-Fender | _Sam--: I didn't know this... </sarcasm> ;) |
| 18:11.05 | Yourname` | [TK]D-Fender: What if we use Express Talk? |
| 18:11.08 | [TK]D-Fender | _Sam--: BLF, Presence, LIU, etc, take your pick |
| 18:11.25 | _Sam-- | well you were implying that nobody is doing any real phone work if all the presence changes were all over my console...i was only trying to ellaborate upon our setup. |
| 18:11.40 | [TK]D-Fender | Yourname`: I avoid soft-phones, and only use X-Lite, Ekiga, and Zoiper myself, and only VERY infrequently at best. |
| 18:12.15 | [TK]D-Fender | _Sam--: I understand, just that if presence is 95% of your CLI output, the you have more watchers than processing. |
| 18:12.40 | _Sam-- | this is true -- currently, there are about 15 watchers, and at the moment, 4 people on calls. |
| 18:12.55 | [TK]D-Fender | _Sam--: Do you just sit and stare at it roll by constantly? I can see it as a bit of a nuisance during debugging (minor really), but whatever... |
| 18:13.17 | *** join/#asterisk r0d3nt (i=nobody@foster.stonedcoder.org) |
| 18:13.33 | _Sam-- | i admin a bunch of servers. normally, i run 'screen' from my main host, and ssh to each each of my boxes from within screen... |
| 18:13.46 | _Sam-- | then i just flip through screens alot of day looking at logs, loads, top, mrtg, etc |
| 18:13.50 | [TK]D-Fender | _Sam--: imagine a "dial-all" :) 15 * 15 notifications! Whee! |
| 18:13.51 | _Sam-- | (not mrt in screen, sorry) |
| 18:14.21 | *** join/#asterisk tsurko (n=tsurko@77.70.15.51) |
| 18:14.24 | _Sam-- | so alot of times when i get to the asterisk box, in my screen, its all just presence output |
| 18:14.36 | _Sam-- | i have 9 different screen 'windows' right now |
| 18:14.39 | red9012 | I like to generate ring tones... is there a command for that? |
| 18:14.45 | _Sam-- | and i ctrl-a n through them all |
| 18:15.29 | [TK]D-Fender | red9012: "show application ringing" |
| 18:15.37 | ZaVoid | cool that sound file format changed worked well |
| 18:15.38 | ZaVoid | thanks guys |
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| 18:16.47 | *** join/#asterisk Cyon (n=cyon@216.179.31.170) |
| 18:18.21 | red9012 | the ringing command requires an additional wait() command. I need a one line command |
| 18:18.50 | *** join/#asterisk caio1982 (i=caio1982@CAcert-br/caio1982) |
| 18:20.22 | *** join/#asterisk kv0s (n=kv0s@dslb-088-065-228-038.pools.arcor-ip.net) |
| 18:20.25 | kv0s | Hi! |
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| 18:20.45 | [TK]D-Fender | red9012: No, it does not require a "wait" |
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| 18:24.33 | ToTo | hi all |
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| 18:25.00 | tomcontr3 | hi, does anyone knows how to configure and FXO-02 device with asterisk? |
| 18:25.18 | ToTo | does asterisk support h.281? |
| 18:25.24 | caio1982 | Qwell[]: ping |
| 18:25.47 | _Sam-- | how can you implement a content delivery network on a small scale -- ie, i dont know how to do the secret sauce -- like how do the CDN providers know which host is closest / fastest to a given http client connection? |
| 18:26.16 | _Sam-- | er wrong win -- sorry |
| 18:26.47 | ZaVoid | hey fender |
| 18:26.52 | ZaVoid | can you translate this for me? exten => s,n,SayDigits($[${BALANCE} : "([0-9]+\)\.\[0-9]+\"]) |
| 18:26.56 | ZaVoid | the \.\ junk |
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| 18:28.58 | *** join/#asterisk florenz (n=fl@p54978BEA.dip0.t-ipconnect.de) |
| 18:29.18 | florenz | hello everyone |
| 18:29.37 | [TK]D-Fender | ZaVoid: Didn't know you could do regex like that in an eval... and I have no idea on the interpretation. |
| 18:29.50 | ZaVoid | lol |
| 18:30.49 | *** join/#asterisk rene- (n=rene@200.34.66.137) |
| 18:31.04 | florenz | is someone here running Asterisk on OpenBSD? |
| 18:31.46 | rene- | hey, is anybody willing to relocate to cancun and do asterisk work here? the pay is not incredible but well you get to live here... msg me for details, must speak english or spanish or both |
| 18:33.06 | ZaVoid | no hablo espanol |
| 18:33.41 | rene- | zavoid: english is ok |
| 18:33.44 | rene- | for the most part |
| 18:33.51 | ZaVoid | fender any debug i could turn on to see which directory its actually playing from? |
| 18:34.00 | generalhan | lol ... what is "not incredible" ? |
| 18:34.18 | [TK]D-Fender | ZaVoid: from the default w/ lang considerations. |
| 18:34.48 | rene- | around 1000 - 1500 month, you have to realize that this is not a senior position and that well mexican wages are a lot lower than ocde countries |
| 18:34.48 | ZaVoid | .. /var/lib/asterisk/sounds |
| 18:34.53 | brettnem | are there any known issues with Asterisk RTP and netfilter ipconntrack? |
| 18:35.02 | florenz | hmm, so, no OpenBSD installations? What platform are you running it on if I may ask? |
| 18:35.05 | ZaVoid | no way to have it print to console that you know of? |
| 18:35.39 | [TK]D-Fender | florenz: Oh I don't know... maybe LINUX?! |
| 18:35.42 | caio1982 | seems that Qwell[] is away (he alrerady knows the subject), so could another op from #asterisk help me to solve a request from a Freenode' staff? |
| 18:36.02 | *** join/#asterisk NightMonkey (n=NightMon@pdpc/supporter/sustaining/NightMonkey) |
| 18:36.04 | Qwell[] | caio1982: I'm here... the ircops just have to like...answer me |
| 18:36.08 | [TK]D-Fender | florenz: There is no Zaptel for OpenBSD last I checked, and FreeBSD was possible though troublesome. |
| 18:36.15 | caio1982 | Qwell[]: hey! |
| 18:36.31 | caio1982 | Qwell[]: do you have some time to solve that issue with UdontKnow, the staff? |
| 18:36.41 | Qwell[] | if he'll actually answer me today :p |
| 18:36.43 | caio1982 | Qwell[]: he told me you didnt reply back |
| 18:36.46 | ZaVoid | hey Qwell you know any way to set a debug to print that out about which dir i'm playing a sound file from? |
| 18:36.48 | caio1982 | Qwell[]: oh :( |
| 18:37.05 | florenz | lol, fender, man of the generic answer :-) - yes, lack of ISDN support in OpenBSD makes me consider it's not a good idea right now... |
| 18:37.52 | florenz | [TK]D-Fender, reason I'm asking is that I just like OpenBSD on small, embedded things |
| 18:38.16 | florenz | it relatively painless, compared to a lot of Linux distributions |
| 18:38.19 | caio1982 | Qwell[]: would it be okay for you to send an e-mail instead? I have his personal address for stuff like that |
| 18:38.42 | [TK]D-Fender | florenz: How about your generic QUESTION? ;) Yes there are people runninf Zaptel-less * installs on OpenBSD. |
| 18:38.45 | Qwell[] | I'd rather do any IRC related stuff on IRC. How about I just ask another op? O.o |
| 18:39.15 | caio1982 | Qwell[]: that's fine for me, it's just that I know him, but it solves the probleme the same away, yeah |
| 18:39.58 | florenz | [TK]D-Fender, I said "generic" because of "maybe LINUX" - no offense, but that's a funny answer. Like, what tires do you have on your car - round ones... |
| 18:40.05 | florenz | hehe |
| 18:40.16 | florenz | like my own jokes best, never mind :-) |
| 18:40.38 | florenz | [TK]D-Fender, so what distribution are you using? |
| 18:41.07 | caio1982 | Qwell[]: would you mind ping me after talking to them? :) |
| 18:41.09 | *** join/#asterisk nDuff (n=ccd@fw2.isgenesis.com) |
| 18:41.41 | [TK]D-Fender | florenz: Well OpenBSD is a very unique kernel. We can beat to death the concept of "distro" over "kernel" if you like... but I'll win (not like it isn't inevitable ANYWAYS ;)) |
| 18:41.43 | nDuff | Is there a reasonable way to load a newly built/installed module without interrupting ongoing calls? |
| 18:42.20 | nDuff | n/m |
| 18:42.25 | [TK]D-Fender | florenz: Most people run * on one of the top-10 Linux distros. Centos, Debian, Slackware, etc... |
| 18:42.38 | [TK]D-Fender | nDuff: Thats it. |
| 18:43.26 | florenz | [TK]D-Fender, doubt that you win, have your asbestos pants ready? But let's save that for a day when we both are terminally bored. What distribution is it for you, then? |
| 18:43.57 | Mercestes | caio1982, Are you ever going to "slip" and tell us what it's about or are you just going to tease us with obscurity by having a private conversation in public? |
| 18:43.59 | florenz | [TK]D-Fender, I'm leaning most to either slack or deb stable, what do you think? |
| 18:44.11 | lirakis | can i hangup a channel from cli? |
| 18:44.12 | Mercestes | caio1982, *stares* |
| 18:44.14 | *** join/#asterisk RypPn (i=TuMbL@rosscom.demon.co.uk) |
| 18:44.23 | lirakis | .. rather.. how do i hangup a channel from cli |
| 18:44.36 | caio1982 | Mercestes: haha that's not private actually, it's about a channel redirect from #asteriskbrasil.org to #asterisk-br (which needs to be allowed by a digium guy to happen) |
| 18:44.40 | caio1982 | Mercestes: just that :-) |
| 18:44.41 | [TK]D-Fender | florenz: I'm persistent and certified non-flammable ;) Personally I use CentOS / Slackware for my installs. |
| 18:44.53 | [TK]D-Fender | lirakis: "soft hangup [channel]" |
| 18:45.13 | Mercestes | caio1982, oh..I thought it involved drugs, or the exportation of random underaged foreigner sex-slaves or something interesting like that. |
| 18:45.30 | Mercestes | caio1982, next time...make something up. :P |
| 18:45.45 | lirakis | [TK]D-Fender: thank you |
| 18:45.46 | Yourname` | Hello again... Aug 8 10:44:49 NOTICE[19303]: res_musiconhold.c:533 monmp3thread: Request to schedule in the past?!?! -> what am I doing wrong? |
| 18:45.47 | [TK]D-Fender | florenz: Either of your two choices would do just fine. Install *'s pre-req's and away you go. |
| 18:46.15 | [TK]D-Fender | Yourname`: Thats just an mpg123 warning you safe pretty much safely ignore. |
| 18:46.22 | caio1982 | Mercestes: you should know that brazil is famous because underaged sex-slaves anyway... it's ajust a smoke curtain, sshhhh |
| 18:46.35 | Yourname` | [TK]D-Fender: But it's not playing the MOH file tho. |
| 18:46.37 | Mercestes | caio1982, ooooOoo...that's more like it. :D |
| 18:47.10 | [TK]D-Fender | Yourname`: Well I guess you'd better show us something USELF in helping debug your problem. |
| 18:47.17 | [TK]D-Fender | USEFUL* |
| 18:47.30 | Mercestes | [TK]D-Fender, you don't use capslock to do that, do you? |
| 18:47.47 | [TK]D-Fender | Mercestes: nope. |
| 18:47.52 | Mercestes | Didn't think so. |
| 18:47.57 | florenz | [TK]D-Fender, thx. deb is nice on the comfort level, but I saw that they have a patched 1.2.13 in stable... |
| 18:47.59 | Mercestes | your pinky fingers must look like another set of thumbs |
| 18:48.14 | [TK]D-Fender | florenz: Forget about packaged * though, always compile from source. |
| 18:48.26 | florenz | nah, I bet he's mapped SHIFT to left-Alt |
| 18:48.35 | [TK]D-Fender | florenz: Besides, tahts an old version with many nasty security wholes. |
| 18:48.53 | [TK]D-Fender | holes |
| 18:48.54 | *** join/#asterisk mtgll (n=mtg@static-71-125-10-2.nycmny.fios.verizon.net) |
| 18:49.05 | [TK]D-Fender | damn.... I am completely cross-wired today... |
| 18:49.14 | Yourname` | [TK]D-Fender: lol, ok.. one min |
| 18:49.45 | florenz | [TK]D-Fender, you, that's what I read in the asterisk release notes, my point beeing I can just go ahead and choose slack, as deb would be nice for the confort level because of apt-get, but moot, because of a too-old version. |
| 18:49.55 | florenz | hmm |
| 18:50.55 | yebo | weird |
| 18:51.02 | [TK]D-Fender | florenz: You can still choose Debian just fine, but do * from source on it. |
| 18:51.14 | [TK]D-Fender | florenz: Doesn't have to be an all-or-nothing you know... |
| 18:51.28 | Yourname` | [TK]D-Fender: extensions.conf? sip debug? |
| 18:51.57 | florenz | [TK]D-Fender :-O I thought it was mutually exclusive... duh |
| 18:52.07 | [TK]D-Fender | Yourname`: proof of the location and state of your MoH files, your MoH config, CLI output where you see these errors/message/activity, etc |
| 18:52.18 | florenz | [TK]D-Fender, thx for the input |
| 18:52.31 | [TK]D-Fender | florenz: But Coming from BSD you might appreciate Slack more. |
| 18:52.58 | [TK]D-Fender | florenz: glad to help. Sarcasm is just part of the package deal ;) |
| 18:53.20 | Mercestes | at no extra charge! |
| 18:54.05 | florenz | [TK]D-Fender, I'm even more evil. I started with Real Unix(tm). got my first Linux fix, and was terminally annoyed with it when everyone installed Linux in droves, which messed up the documentation thing quite good, thank you very much. |
| 18:54.17 | florenz | way back when :-) |
| 18:54.44 | florenz | keeps me on my toes to paly with all nioce OS.. |
| 18:54.45 | Mercestes | Everyone installing it screwed up the documentation? |
| 18:55.06 | Yourname` | [TK]D-Fender: ok |
| 18:55.22 | florenz | well, evryone changed it increased the signal/noise ration to unbeareable leveles |
| 18:55.33 | Mercestes | was that english? |
| 18:55.53 | Mercestes | I mean, I understood most of it, but, spanish is tricky like that. |
| 18:56.32 | Mercestes | Well, as much as I appreciate a bit of distrowars..... |
| 18:56.56 | Mercestes | Yea, I just can't finish it. |
| 18:57.04 | Mercestes | someone else make up the rest |
| 18:57.29 | florenz | everyone installing it lead to a lot more questions about it, which increased the uninformed answers flying around, because the idiots distribution in the general population follows the bell-curve |
| 18:58.00 | *** part/#asterisk mtgll (n=mtg@static-71-125-10-2.nycmny.fios.verizon.net) |
| 18:58.21 | florenz | the more people play with something, the more bullshit circulates about that something, in other words. Makes it harder to pick out the gems. |
| 18:58.25 | Mercestes | florenz, no it doesn't. |
| 18:58.44 | florenz | Mercestes, oh? How so? |
| 18:58.50 | kv0s | Hi! I've several problems with my local installation of asterisk. But i think it isn't a asterisk problem! ;-) I've some echo on my lines, it is possible, that the echo produced thrugh my bluetooth headset? |
| 18:58.52 | Mercestes | I would say there is a direct correlation between # of users and idiots so it's absolutely not a bell curve. |
| 18:59.21 | hmmhesays | heh nice |
| 18:59.27 | Mercestes | But, again, it's distrowars at this point. You like OpenBSD....that's good enough for us... |
| 18:59.49 | Mercestes | You don't have to make up crap after that to defend it. we accept your preference and do not judge you for it. |
| 19:00.09 | Mercestes | kv0s, describe your calling environment. |
| 19:00.21 | florenz | Mercestes, ugh, you are trying to nail me for something I did not say. Careful. |
| 19:00.30 | *** part/#asterisk RealBorg (n=tom@38.pool85-48-226.static.orange.es) |
| 19:00.31 | Mercestes | I have witnesses |
| 19:00.52 | Mercestes | 279 total, as a matter of fact. |
| 19:01.18 | Mercestes | ... |
| 19:01.20 | Mercestes | ok, 278 |
| 19:01.44 | Mercestes | But, be that as it may..... |
| 19:01.54 | florenz | I do like Linux, a lot of them, and *BSD, and Macs. Even Windows. What I said is that the popularity of Linux was detrimental to the signal to noise ratio of all available information. |
| 19:02.12 | Mercestes | Again, I disagree.... |
| 19:02.16 | Mercestes | but...still, Distrowars. |
| 19:02.28 | florenz | ok, nevermind. We have different opinions. |
| 19:02.36 | Mercestes | I mean, if you want to see the effect of a small community on a project, go join #callweaver sometime |
| 19:02.40 | Mercestes | or #plan9 |
| 19:03.15 | kv0s | Mercestes: Outgoing SIP oder Zap(bristuffed isdn), internal i've sip (x-lite on my notebook) - it's makes no difference between calling out with the isdn or sip trunk ... |
| 19:03.24 | florenz | I don't think my statement is invalidated by fringe OS examples :-) |
| 19:04.42 | florenz | Mercestes, are you from Houston (like it says in the blog?) |
| 19:04.56 | *** join/#asterisk tzafrir_laptop (n=tzafrir@bzq-82-81-101-116.red.bezeqint.net) |
| 19:05.57 | Yourname` | [TK]D-Fender: Figured it out, when you said "the location".. I looked around, and it was bad. :) |
| 19:06.01 | Yourname` | [TK]D-Fender: Thanks! |
| 19:06.26 | [TK]D-Fender | Yourname`: You should apply that kind of thinking to ALL of your problems... |
| 19:07.28 | Yourname` | [TK]D-Fender: If the errors were more in english than eblish.. it'd help in pinpointing the fault. :) |
| 19:07.46 | Yourname` | Still, your commitment to the asterisk community, IMHO, is HUGE. |
| 19:09.40 | Yourname` | Meanwhile, here's another question. When the agent calls the external number to do an "attended transfer", and if that party doesn't pick up quick enough.. the call is aborted and the agent is dropped back to the customer. Again, what value did I set too less? |
| 19:10.46 | [TK]D-Fender | Yourname`: I'm not sure on how *'s fatures.conf transfer work exactly..... can't help you there unfortunately |
| 19:11.06 | Yourname` | [TK]D-Fender: That's ok.. :) |
| 19:14.33 | Yourname` | Hmm, I thought parkedtime needs to be increased, and I did.. and it didn't work |
| 19:17.50 | *** join/#asterisk galeras (n=galeras@200.31.204.42) |
| 19:19.08 | *** join/#asterisk [hC] (n=hardcore@66.119.167.162) |
| 19:20.14 | galeras | Where is the best source to know changes between ami 1.2 and 1.4? |
| 19:20.50 | Strom_M | galeras: UPGRADE.txt |
| 19:21.07 | galeras | Thanks Strom_M |
| 19:22.04 | *** join/#asterisk tracinet (n=tracinet@216.242.235.2) |
| 19:25.49 | brodiem | mog - hey could you PM me your e-mail address? The one from doc/jabber.txt is rejected |
| 19:26.17 | mog | really, what email brodiem |
| 19:26.35 | brodiem | mogorman@digum.com |
| 19:26.39 | ToTo | can i move remote camera with asterisk? |
| 19:26.51 | brodiem | ToTo sure |
| 19:27.02 | *** join/#asterisk merkurie (n=merkurie@192.153.163.44) |
| 19:27.04 | mog | is thats whats in there? |
| 19:27.11 | mog | should be mogorman@digium.com |
| 19:27.21 | ToTo | brodiem: how? |
| 19:27.27 | ZaVoid | hey brodiem |
| 19:27.36 | brodiem | The maintainer of res_jabber is Matthew O'Gorman <mogorman@digum.com> |
| 19:27.42 | brodiem | hey |
| 19:27.56 | ToTo | brodiem: is there a protocol? |
| 19:27.58 | mog | well i change that |
| 19:28.04 | Qwell[] | mog: fixing now :p |
| 19:28.13 | r0d3nt | <SecNews> Title: Vuln: Asterisk Skinny Channel Driver Remote Denial of Service Vulnerability |
| 19:28.13 | r0d3nt | <SecNews> Link: http://www.securityfocus.com/bid/25228 |
| 19:28.29 | mog | wasnt that forever ago |
| 19:28.34 | Qwell[] | mog: yesterday |
| 19:28.38 | Qwell[] | but it only affects like 8 people |
| 19:28.55 | brodiem | mog - ok maybe I'm blind but I'm not seeing a difference in what you said it should be and what it actually is :) |
| 19:29.13 | mog | you said it didnt have an i in digium |
| 19:29.22 | ToTo | brodiem: ? |
| 19:29.26 | brodiem | ahh |
| 19:29.27 | brodiem | lol |
| 19:29.30 | brodiem | right |
| 19:29.45 | brodiem | should have caught that... |
| 19:30.09 | Qwell[] | mog: fixed :D |
| 19:33.44 | ToTo | brodiem: do you have an idea about how i can move a remote camera with ast? |
| 19:34.03 | brodiem | ToTo ast itself will not move anything |
| 19:34.22 | brodiem | but you can have ast launch whatever script that interfaces with your cameras |
| 19:34.43 | brodiem | I'm working on an X10 camera interface.. just uses an open source command line linux script to move them |
| 19:35.12 | merkurie | whats a good asterisk book for beginners? I was looking at getting 'asterisk cookbook' but it looks like the release date is getting pushed to october |
| 19:35.35 | ToTo | brodiem: ..and, i wold to move a polycom vsx 7000s cam.. |
| 19:35.41 | *** join/#asterisk anonymouz666 (n=anonymou@189.25.123.107) |
| 19:36.05 | ToTo | brodiem: polycom use h.281 to send remote command to move it |
| 19:36.21 | *** join/#asterisk RypPn (i=TuMbL@rosscom.demon.co.uk) |
| 19:38.42 | [TK]D-Fender | ToTo: * cannot by itelf move the camera. Go see what other libraries & apps can and see if * can be made to control THEM. |
| 19:40.38 | *** join/#asterisk guillote_GNU (n=bancaria@host191.190-31-26.telecom.net.ar) |
| 19:42.36 | tracinet | hey guys - got a weird one for you.... |
| 19:42.43 | tracinet | i have an asterisk server hosting virtual pbx's for about 20 clients and it has been running for over a year with no problems at all. |
| 19:42.53 | tracinet | the other day - ext. 107 from "company A" dials extension 111, but instead of ext. 111 from "company A" ringing, ext. 111 from "company B" rang. I can not duplicate the problem, however, I do see in the CDR log that it did happen. |
| 19:43.07 | tracinet | Configs were checked and checked again and all seems ok (like i said - can't duplicate it again). any thoughts? |
| 19:43.14 | tracinet | asteirks 1.2.10 |
| 19:43.18 | tracinet | asterisk* |
| 19:43.30 | Sweeper | if you can't duplicate it, it didn't happen :v |
| 19:43.35 | tracinet | lol |
| 19:43.41 | tracinet | well my logs say otherwise |
| 19:43.49 | Sweeper | they lie! |
| 19:43.51 | tracinet | not to mention the owner of company B who answered the phone |
| 19:43.52 | tracinet | LOL |
| 19:44.00 | tracinet | while one of my sales guys happened to be there |
| 19:44.04 | tracinet | how embarrassing |
| 19:44.27 | tracinet | i checked my call logs and it looks like it happened a total of 13 times since customer has come online |
| 19:45.13 | tracinet | tried to find a bug in bugzilla talking about incorrect context routing |
| 19:45.19 | tracinet | but have not found anything |
| 19:45.21 | *** join/#asterisk zydrunas (n=Miranda@216.49.228-ip-184.ckt.net) |
| 19:45.25 | pepesz76 | Hi folks, I'm dialing phone_1 -> asterisk_1 -> asterisk_2 ->phone_2 , but got "call failed: 503 Service Unavailable". The log file from asterisk_2: http://pastebin.ca/650266. What am I doing wrong? |
| 19:45.55 | *** join/#asterisk Igbothom_3rd (n=Hilton@office.quarkit.com.au) |
| 19:46.13 | *** join/#asterisk thx2000 (n=the@netblock-208-127-94-59.dslextreme.com) |
| 19:46.33 | tracinet | would need to see your sip.conf and ext.conf files |
| 19:46.41 | [TK]D-Fender | pepesz76: Looking for 55 in default (domain 84.41.234.196) |
| 19:46.47 | [TK]D-Fender | pepesz76: SIP/2.0 404 Not Found |
| 19:47.11 | tracinet | well that is a problem too |
| 19:47.12 | tracinet | LOL |
| 19:47.23 | [TK]D-Fender | pepesz76: I see a **404** not a 503, and the location of the error is pretty blatant |
| 19:48.55 | pepesz76 | just a sec |
| 19:49.37 | *** join/#asterisk Stromthipper (n=unnamedf@sonicwall.mercyships.org) |
| 19:50.49 | Stromthipper | greetings eh? |
| 19:51.02 | Stromthipper | I'm in trouble deep here eh? |
| 19:52.29 | tracinet | i am |
| 19:52.36 | tracinet | and have an issue if you are interested |
| 19:52.45 | tracinet | let me paste what i just posted... hold |
| 19:52.46 | Stromthipper | I'm here for help as well my friend |
| 19:52.56 | Stromthipper | if you paste I'll be clueless =( |
| 19:53.00 | tracinet | lol |
| 19:53.09 | tracinet | the blind leading the blind i see |
| 19:53.09 | Stromthipper | My clue bag is...empty =| |
| 19:53.13 | Stromthipper | hehee |
| 19:53.15 | Stromthipper | indeed |
| 19:53.27 | Stromthipper | sorry, I wish I could see for you sake eh |
| 19:53.29 | Stromthipper | ? |
| 19:53.44 | Stromthipper | Tracinet? |
| 19:53.54 | Stromthipper | Asterisknow seems to have some activity |
| 19:53.59 | Stromthipper | I'm checking there too |
| 19:54.05 | Stromthipper | even though I don't know what it is |
| 19:54.55 | *** join/#asterisk cellphone (i=lysol@monoperative.net) |
| 19:55.09 | tracinet | sorry - got a call... |
| 19:55.16 | Stromthipper | sure thing... |
| 19:55.43 | ManxPower | Stromthipper: #asterisknow is really the place to talk about...AsteriskNOW |
| 19:56.01 | *** join/#asterisk styelz (n=yoohoo@2001:388:f000:0:0:0:0:20b) |
| 19:56.21 | tracinet | ok back |
| 19:56.31 | Stromthipper | ah |
| 19:56.40 | Stromthipper | I'm asking in there right now Manxpower |
| 19:56.55 | Stromthipper | they appear to be software dev's |
| 19:57.06 | styelz | hey if i have a zap channel, and i set echocancel=yes .. do i need to set echotraining ? |
| 19:57.17 | tracinet | i believe it has a default |
| 19:57.17 | *** join/#asterisk Igbothom_3rd (n=Hilton@office.quarkit.com.au) |
| 19:57.34 | tracinet | which is enabled |
| 19:57.36 | styelz | if I do set echotraining my fxo card stops working |
| 19:57.38 | styelz | correctly |
| 19:57.46 | tracinet | weird |
| 19:57.47 | styelz | ok |
| 19:58.02 | tracinet | have you tried setting it to "no" |
| 19:58.13 | styelz | no i havent |
| 19:58.15 | pepesz76 | here are mine sip and extensions files: pastebin.ca/650281 pastebin.ca/650283 |
| 19:58.22 | styelz | i just commented it out in zapata.conmf |
| 19:58.32 | styelz | it was =800 |
| 19:59.16 | styelz | it only just started to hapen since i upgraded |
| 19:59.58 | *** join/#asterisk techie (n=techie@c-67-181-184-170.hsd1.ca.comcast.net) |
| 20:00.06 | [TK]D-Fender | [TK]D-Fender>pepesz76: Looking for 55 in default (domain 84.41.234.196) |
| 20:00.07 | [TK]D-Fender | [15:46]<[TK]D-Fender>pepesz76: SIP/2.0 404 Not Found |
| 20:00.08 | *** join/#asterisk tomcontr3 (n=tomcontr@37-161-28.dial.terra.cl) |
| 20:00.14 | [TK]D-Fender | pepesz76: What part of this was not clear? |
| 20:02.01 | [TK]D-Fender | pepesz76: This should be a clue as to an AUTH problem too : Found no matching peer or user for '82.210.120.77:5060' <---- |
| 20:03.25 | mog | brodiem: simple fix |
| 20:03.28 | mog | ill work on it tonight |
| 20:03.35 | Stromthipper | found problem with voip not working. phones weren't getting an IP address but found it to be that the DHCP bombed. reset that and bingo! |
| 20:03.35 | pepesz76 | tracinet asked for conf files, that's why. I'm kind of lost. 55 is the phone behind nat with internal address 10.0.0.20 (external 84.....) |
| 20:03.49 | Stromthipper | gold luck all eh? |
| 20:03.51 | *** part/#asterisk Stromthipper (n=unnamedf@sonicwall.mercyships.org) |
| 20:03.56 | *** part/#asterisk ricky (i=ricky@fedora/ricky) |
| 20:05.07 | [TK]D-Fender | pepesz76: You have no user for [51] and its falling back to [default] which doesn't even exist in the dialplan. |
| 20:05.38 | tracinet | pepesz - i asked for the files bfore looking at your pastebin |
| 20:05.49 | pepesz76 | <PROTECTED> |
| 20:05.50 | *** join/#asterisk guillote_GNU (n=bancaria@host136.200-117-227.telecom.net.ar) |
| 20:05.55 | tracinet | [TK]D-Fender is rigth - 404 is the problem |
| 20:05.59 | *** join/#asterisk anthm (n=anthm@adsl-69-216-26-86.dsl.milwwi.ameritech.net) |
| 20:05.59 | *** mode/#asterisk [+o anthm] by ChanServ |
| 20:06.05 | [TK]D-Fender | tracinet: PART of it,. |
| 20:06.35 | [TK]D-Fender | pepesz76: You are getting a call from someone who isn't a user on yoursystem, and when treated generically there is no context for the calls to land on. |
| 20:07.07 | styelz | here is a pastebin of what "log/full" did when i did both echotraining and not |
| 20:07.07 | styelz | http://pastebin.ca/650291 |
| 20:08.01 | ManxPower | put exten => 55,1,Noop(Call worked! Lets go drinking!) in the [default] part of extensions.conf |
| 20:08.12 | pepesz76 | so to receive a call I have to put him in my config ? |
| 20:08.35 | ManxPower | pepesz76: if you don't, then the call will land in whatever context is in [general] in sip.conf |
| 20:09.03 | [TK]D-Fender | pepesz76: Is he SUPPOSED to be an SIP device registered to your system? |
| 20:09.05 | ManxPower | and if you don't have anything matching the destination number, the call will fail. |
| 20:09.06 | tomcontr3 | hi, is anyone here using a FXO Gateway? |
| 20:09.16 | ManxPower | tomcontr3: I'd rather have a root canal |
| 20:09.36 | *** part/#asterisk florenz (n=fl@p54978BEA.dip0.t-ipconnect.de) |
| 20:09.37 | [TK]D-Fender | tomcontr3: Yes, many of us, now just go and ask a SPECIFIC question. |
| 20:09.42 | *** part/#asterisk tracinet (n=tracinet@216.242.235.2) |
| 20:10.11 | tomcontr3 | I have an FXO-02 Gateway, but I can figure it out how to configure it so it can work with Asterisl |
| 20:11.06 | tomcontr3 | here is the manual : http://www.netkrom.com/support/NetGate_FXO_SIP_manual.pdf |
| 20:13.23 | tomcontr3 | if anyone could give me a little hand, I will really apreciated |
| 20:14.57 | Deeewayne | tomcontr3: did you read the manual ? |
| 20:15.24 | pepesz76 | Lost again 51 @ 82.210... (internal 192.168.1.10) (registered to asterisk1) is calling 55 @ 84.41... (internal 10.0.0.20) registered in asterisk 2 (which logs and config files I presented). So where to put calling party (conf?) to make this call working. Sorry to annoy you with questions. Still learning. |
| 20:16.50 | jcolp | Deeewayne: can you help me become the next Vonage? |
| 20:17.35 | [TK]D-Fender | jcolp: No, I'm sure you can reach Chapter 11 status without our "assistance" ;) |
| 20:18.30 | tomcontr3 | yes I did |
| 20:18.31 | *** join/#asterisk |dennis| (n=dennis@200.32.236.20) |
| 20:18.47 | tomcontr3 | but I can figure out how to configure it well |
| 20:20.15 | tomcontr3 | how does this FXO Gateway interact with my asterisk server, does the AS conect to this FXO box, or the FXO box conects to the AS? |
| 20:20.33 | *** join/#asterisk easimon_ (n=easimon@baghira.kawo2.RWTH-Aachen.DE) |
| 20:21.43 | *** join/#asterisk el_critter (n=chatzill@190.74.100.35) |
| 20:22.51 | Deeewayne | analog and the transitive property of equality say, "yes" |
| 20:24.55 | *** join/#asterisk zotz (n=zotz@24.244.163.157) |
| 20:25.49 | tomcontr3 | how does this FXO Gateway interact with my asterisk server, does the AS conect to this FXO box, or the FXO box conects to the AS?? |
| 20:25.54 | tomcontr3 | sorry about that |
| 20:28.46 | *** join/#asterisk vgster (n=vgster@cpc2-ledn1-0-0-cust826.leed.cable.ntl.com) |
| 20:29.14 | Deeewayne | tomcontr3: are you asking which side to connect first ? |
| 20:29.47 | *** join/#asterisk flujan_ (n=flujan@200-160-115-020.static.spo.ctbc.com.br) |
| 20:30.35 | tomcontr3 | right, well im not sure what exactly I have to do |
| 20:31.01 | tomcontr3 | I 've been trying all day |
| 20:31.03 | *** join/#asterisk zcionn_ (n=a@58.69.243.203) |
| 20:31.10 | *** join/#asterisk generalhan (n=asd@ip67-90-64-2.z64-90-67.customer.algx.net) |
| 20:31.17 | lmadsen | FYI: Just found out TFoT2 goes to the printers next Wednesday, then printed copies will be shipping 2 weeks after that |
| 20:31.51 | tomcontr3 | but for example in the SIP configuration Y have to options, once says Peer2Peer and the other Proxy |
| 20:31.55 | *** join/#asterisk tsurko (n=tsurko@77.70.15.52) |
| 20:32.24 | flujan_ | guys, I need to monitor all calls placed and answered by a peer... This peer is also a agent from a queue and I want to record the calls it receives from the peer... |
| 20:32.31 | flujan_ | how can I set a channel to be monitored? |
| 20:46.28 | *** join/#asterisk [TK]D-Fender (n=Joe@64.235.216.2) |
| 20:47.44 | *** join/#asterisk rmayorga (i=rmyorg@unaffiliated/rmayorga) |
| 20:51.44 | lirakis | whew... working from home sux |
| 20:51.49 | lirakis | .. chair is a lot less comfortable |
| 20:53.21 | *** join/#asterisk zeeesh (n=crosslim@202.125.143.66) |
| 20:55.15 | *** join/#asterisk timholum (n=tim@66-191-97-163.static.eucl.wi.charter.com) |
| 20:58.44 | *** join/#asterisk sakic (n=sakic@cpe-071-075-118-121.carolina.res.rr.com) |
| 20:58.51 | *** join/#asterisk rmayorga (i=rmyorg@unaffiliated/rmayorga) |
| 20:59.28 | lirakis | [TK]D-Fender: still at work??? |
| 20:59.37 | lirakis | [TK]D-Fender: late day for you |
| 20:59.45 | easimon | has anyone in here achieved to run a recent bristuffed asterisk with NT mode? i've got problems since upgrading to 1.4 |
| 21:00.17 | *** join/#asterisk jkiff (n=jkiffmey@unaffiliated/vorondil) |
| 21:01.03 | timholum | hello, i am wondering if there is a variable in asterisk of the users extention? |
| 21:01.20 | mvanbaak | timholum: explain |
| 21:01.20 | anonymouz666 | jcolp: how many RTP ports are necessary for.. let's say 10 active calls? |
| 21:01.28 | Qwell[] | anonymouz666: 20 |
| 21:01.35 | jcolp | gold star for Qwell |
| 21:01.40 | mvanbaak | anonymouz666: 20 |
| 21:01.44 | mvanbaak | oh wait |
| 21:01.44 | anonymouz666 | RTCP? |
| 21:01.45 | mvanbaak | lol |
| 21:01.56 | timholum | i am trying to write a gotoif() statment, i need if phone 403 calls it does somthing different then if 404 |
| 21:02.01 | mvanbaak | for RTCP you need something that's not asterisk |
| 21:02.20 | timholum | the phone that is making the call |
| 21:02.21 | easimon | timholum: : $EXTEN |
| 21:02.35 | timholum | $EXTEN is the phone you are calling |
| 21:02.37 | mvanbaak | timholum: CALLERID(num) |
| 21:02.43 | easimon | $CALLERID(num) |
| 21:02.59 | timholum | i will give that a shot, thanks a lot :) |
| 21:03.13 | *** join/#asterisk clandmeter (n=Carlo@81.175.82.2) |
| 21:03.28 | Netgeeks | hay Qwell, what would you say if I told you there is an instance where a bye recieved by asterisk causes it to reply back with an invite with no media descritors |
| 21:03.39 | Netgeeks | :s/hay/hey/ |
| 21:04.04 | anonymouz666 | very very strange. I got a server running about 2800 complete calls/day, But never more than 16 active calls... and at the some point I got that beautiful no RTP ports remaining |
| 21:04.07 | styelz | its faaaat albert |
| 21:04.11 | Qwell[] | I'd tell you that if it isn't right, to report a bug. |
| 21:04.21 | anonymouz666 | the range is from 20000 to 21000 |
| 21:04.51 | anonymouz666 | I increase to 25000 and the problem stops. |
| 21:04.59 | anonymouz666 | but I didn't understand the math |
| 21:05.08 | Netgeeks | darn, I was hoping for a 'thats cool, or 'holy bat turds, that doesn't sound right!'.... ;) |
| 21:05.32 | [hC] | anyone know why when a cisco 7970 asks for DHCP, a regular old DHCP server (linux) that answers everyone else wouldnt answer? |
| 21:05.40 | [hC] | this phone refuses to get dhcp, even though its asking for it. |
| 21:06.04 | anonymouz666 | nevermind |
| 21:06.18 | Qwell[] | [hC]: bug, factory reset it ;/ |
| 21:06.39 | [hC] | Qwell: thats what caused it. I factory reset it and now its sitting at 'Upgrading' and asking for DHCP repeatedly and getting nowhere. |
| 21:06.51 | Qwell[] | it wants option-150 |
| 21:07.00 | [hC] | which is a tftp server right? |
| 21:07.03 | Qwell[] | yeah |
| 21:08.41 | *** join/#asterisk kannan (n=kannan@121.246.27.179) |
| 21:08.51 | [hC] | hmmm. I do have option 150 in there. |
| 21:08.57 | [hC] | unless i specified it wrong. |
| 21:09.00 | Qwell[] | is it hitting the tftpd? |
| 21:09.16 | [hC] | I dont see it get an answer back from the dhcp server, so far |
| 21:09.17 | *** join/#asterisk saftsack (n=saftsack@pD9E05769.dip.t-dialin.net) |
| 21:10.03 | [hC] | gonna try tcpdumping from one other spot |
| 21:11.12 | [hC] | ah it does reply. further testing shall ensue. :P |
| 21:13.03 | fujin | hey, if I go to Voicemail with bEXT, and they haven't recorded a busy message, does it fall back to unavailable? |
| 21:13.03 | [hC] | my tcpdump shows a dhcp request packed come out from the phone, then the packet back contains my dhcp options, i dont see an ip in there, but i could just be looking at the wrong part of the packet. |
| 21:13.08 | [hC] | time to dig further. |
| 21:13.59 | *** join/#asterisk denon (n=denon@tooth.decay.org) |
| 21:14.00 | *** mode/#asterisk [+o denon] by ChanServ |
| 21:14.37 | lirakis | is there a listing anywhere of each of the sound files included in asterisk and a short description?? |
| 21:14.54 | lirakis | or do we get to guess what is in a sound file? |
| 21:16.02 | x86 | look at the contents of the asterisk-sounds package |
| 21:21.01 | *** join/#asterisk [T]ank (n=ckwall@206.71.78.172) |
| 21:21.24 | [T]ank | i am using asterisk 1.4 and music on hold just sounds horrible... is this normal? |
| 21:21.31 | [T]ank | very choppy |
| 21:21.48 | *** join/#asterisk [hC] (n=hardcore@66.119.167.162) |
| 21:23.05 | lirakis | x86: sounds-extra.txt ;) |
| 21:23.26 | x86 | [T]ank: what kind of timing device are you using? |
| 21:25.06 | *** join/#asterisk frenzy (n=frenzy@unaffiliated/frenzy) |
| 21:25.40 | [T]ank | checking to see if ztdummy is running. |
| 21:25.59 | [T]ank | nope |
| 21:26.00 | [T]ank | loaded it. |
| 21:26.09 | [T]ank | think that is the case? |
| 21:26.43 | pepesz76 | Thanks ! After some struggling it's finally working :) |
| 21:27.04 | brodiem | [T]ank yes |
| 21:28.04 | *** join/#asterisk [hC] (n=hardcore@66.119.167.162) |
| 21:28.12 | x86 | [T]ank: you need some timing source, yes |
| 21:28.24 | x86 | [T]ank: and if you dont have real TDM hardware, then you need to use ztdummy |
| 21:28.34 | x86 | anyone know where I can find some free MoH music? |
| 21:28.54 | [T]ank | restarted asterisk after loading ztdummy... no change. |
| 21:28.59 | *** join/#asterisk easimon_ (n=easimon@baghira.kawo2.RWTH-Aachen.DE) |
| 21:29.15 | brodiem | x86 I found some by googling for royalty free music, lot of crap to sort through though.. |
| 21:29.24 | x86 | no kidding |
| 21:29.36 | brodiem | any search string with "free" is never good... |
| 21:29.37 | brodiem | lol |
| 21:29.42 | *** join/#asterisk flujan (n=flujan@200-160-115-020.static.spo.ctbc.com.br) |
| 21:29.45 | levi | Try searching for 'creative commons' |
| 21:30.14 | brodiem | levi where were you when I was looking for this stuff? |
| 21:30.34 | [T]ank | i have also loaded the music and am running it from ramdisk to help speed up the read of the file. but that did not help either |
| 21:31.08 | brodiem | [T]ank make sure your /dev/zap files are usable by the asterisk user |
| 21:31.29 | brodiem | [T]ank if you aren't running ast as root then there's a good chance that's your problem |
| 21:31.50 | [hC] | Qwell: so it was option 150, i had ""'s around it in the config file. it didnt like that. thank you for helping me save the day. |
| 21:31.57 | levi | brodiem: Not here yet, apparently. :) |
| 21:32.13 | flujan | guys, I am having this error: sent into invalid extension '23008' in context 'hints', but no invalid handler |
| 21:33.02 | [hC] | Qwell: I wonder if this will solve the other problem i was having. For some reason when dialing out using sccp, i get one way audio. UNLESS I call into my *'s IVR first, and dial out from it. Then i get two way audio... ?!?! |
| 21:33.46 | *** join/#asterisk denon (n=denon@tooth.decay.org) |
| 21:33.46 | *** mode/#asterisk [+o denon] by ChanServ |
| 21:33.59 | flujan | any ideas? |
| 21:34.04 | flujan | it was working sometime ago? |
| 21:34.08 | x86 | where does fpm-sunshine come from? |
| 21:34.52 | generalhan | x86: www.freeplaymusic.com i believe |
| 21:35.18 | *** join/#asterisk Igbothom_III (n=Hilton@office.quarkit.com.au) |
| 21:35.30 | generalhan | x86: the license file in the moh says "Music Provided By www.freeplaymusic.com. These sound files are provided by Digium under license from Freeplay Music Corporation for use in conjunction with the Asterisk software only. |
| 21:35.58 | brodiem | sounds logical |
| 21:36.39 | [T]ank | brodiem: asterisk is running as root |
| 21:36.48 | brodiem | ugh |
| 21:37.27 | brodiem | rules out the /dev/zap prob though.. |
| 21:37.44 | brodiem | make sure /dev/zap/* exists |
| 21:39.58 | flujan | guys, I have the extension 23008 in the context hints... |
| 21:40.54 | putnopvut | Out of curiosity, is anyone here using IMAP storage for voicemail? |
| 21:41.01 | brodiem | what gpl'd webgui's do you guys use for cdr analyzing/recordings? |
| 21:41.25 | mvanbaak | brodiem: areski |
| 21:41.40 | fujin | anyone familiar with agentcallbacklogin? I need to make it *not* ask for a new extension |
| 21:41.44 | brodiem | mvanbaak will it locate recordings? |
| 21:41.55 | Qwell[] | fujin: Don't use agentcallbacklogin |
| 21:41.57 | mvanbaak | recording != cdr |
| 21:42.04 | Qwell[] | fujin: You'll regret it in the immediate future. |
| 21:42.08 | fujin | How so? |
| 21:42.17 | Qwell[] | because it's horribly broken, and now deprecated |
| 21:42.18 | mvanbaak | fujin: agentcallbacklogin is deprecated, use addqueuemember |
| 21:42.26 | fujin | heh |
| 21:42.34 | flujan | guys, could you please check my extensions .conf? |
| 21:42.36 | flujan | http://pastie.caboo.se/86024 |
| 21:42.36 | fujin | addqueuemember doesn't actually register an agent, though |
| 21:42.43 | Qwell[] | fujin: You don't need agents |
| 21:42.54 | mvanbaak | Qwell[]: WRONG |
| 21:43.01 | Qwell[] | agents are a silly concept :p |
| 21:43.01 | brodiem | Qwell[] I been meaning to ask... what is the problem with agentcallbacklogin? I was in for a shock when I had to recreate the entire agent structure in 1.4 using chan_local instead of chan_agent lol |
| 21:43.02 | mvanbaak | we use agents for freeseeting |
| 21:43.04 | fujin | I know *I* don't, but the development people want me to have developments. |
| 21:43.09 | fujin | and we do use hotdesking |
| 21:43.14 | fujin | want me to have it** |
| 21:43.18 | fujin | uhh, haven't had a cofee. |
| 21:43.23 | mvanbaak | we dont use agents for queues |
| 21:43.39 | mvanbaak | but we use agents to make ppl login on a workplace |
| 21:43.59 | fujin | the software we're running unfortunately requires Agents for tracking |
| 21:44.02 | fujin | part of the scope |
| 21:44.03 | mvanbaak | and in our dialplan we use: Dial(Agent/<whatever> |
| 21:44.05 | *** join/#asterisk jarrod (i=anon@theos.org) |
| 21:44.09 | fujin | I'm just trying to deal with the tools that I have. |
| 21:44.12 | brodiem | I feel my new setup using chan_local is much better, but at the same time chan_agent works great for me on 1.2 and curious why this is being changed? For a lot of people chan_agent is useless without agentcallbacklogin |
| 21:44.13 | jarrod | what version of rtpproxy do i need to run with openser 1.2 ? |
| 21:44.15 | Qwell[] | brodiem: very major locking issues |
| 21:44.31 | brodiem | ah |
| 21:44.32 | jarrod | the latest is giving me errors, about openser not being able to communicate with rtpproxy.sock |
| 21:44.40 | lmadsen | jarrod: sounds kinda like you're in the wrong channel... |
| 21:44.42 | Qwell[] | brodiem: agentcallbacklogin is useless :) |
| 21:44.55 | mvanbaak | agentcallbacklogin is horrible |
| 21:45.02 | brodiem | lol |
| 21:45.03 | mvanbaak | nothing but trouble with it |
| 21:45.09 | brodiem | how so? |
| 21:45.29 | brodiem | I do about 300 calls a day all going to an agent channel (on 1.2/agentcallbacklogin) |
| 21:45.33 | mvanbaak | missed calls, wrong agent status, asterisk lockups |
| 21:45.36 | brodiem | I have over 500 days of uptime on that box |
| 21:45.49 | *** join/#asterisk el_critter (n=chatzill@190.74.100.35) |
| 21:45.53 | el_critter | hi |
| 21:46.25 | mvanbaak | we switched to use local channels long before the agentcallbacklogin was deprecated |
| 21:46.34 | mvanbaak | it just turned out to be way more stable |
| 21:46.44 | Qwell[] | stable and more configurable |
| 21:46.53 | lmadsen | and awesome |
| 21:46.55 | brodiem | my new setup with chan_local+func_devstate is way better it was just a shock in the first place that basically said sotp using agentcallbacklogin because it will be gone and no other way of hotdesking with chan_agent lol |
| 21:46.57 | mvanbaak | agentcallbacklogin was bringing our asterisk to it's knees several times a day |
| 21:47.51 | *** join/#asterisk jarg (n=jarg@200.56.225.61) |
| 21:48.10 | *** join/#asterisk Tommy2 (n=Tommy@66.0.46.210) |
| 21:48.40 | flujan | guys, any idea why the error: ent into invalid extension '23008' in context 'hints', but no invalid handler |
| 21:49.02 | brodiem | does qwell = kevin? |
| 21:49.06 | Qwell[] | no |
| 21:49.41 | el_critter | can you please recomend me another softphone for linux besides x-lite (having problems with usb headset) |
| 21:50.13 | brodiem | el_critter i haven't found anything better than xlite for *nix |
| 21:50.26 | Yourname` | Hello, what could this mean? It happens when a call is coming in. WARNING[28102]: chan_sip.c:2585 sip_write: Asked to transmit frame type 4, while native formats is 256 (read/write = 4/4) |
| 21:50.32 | Yourname` | And there's dead air on both sides. |
| 21:50.43 | brodiem | Yourname` codec mismatch |
| 21:50.56 | Yourname` | Hmm. |
| 21:51.01 | [hC] | Well.. My Cisco came back up, but now I still get one way audio unless i dial to an IVR, then use an option off the IVR to call out. THEN I get two way audio. Ever seen something like that qwell? |
| 21:51.04 | Yourname` | brodiem: Let me look, thanks. |
| 21:51.10 | el_critter | brodiem: :( thanks |
| 21:51.37 | brodiem | el_critter i found it best to just not use a softphone.. |
| 21:52.56 | Yourname` | brodiem: Codec mismatch? |
| 21:53.07 | Yourname` | brodiem: Because it worked earlier.. :S |
| 21:53.17 | *** join/#asterisk AC_Jay (n=Jay@ns1.accu-com.com) |
| 21:53.27 | AC_Jay | howdy folks |
| 21:53.40 | AC_Jay | having some moh probs, was hoping someone could maybe help? :) |
| 21:54.32 | AC_Jay | *tap tap* this thing on? ;) |
| 21:55.31 | Yourname` | brodiem: Also, outbound works.. it's just happening on when the calls come IN. |
| 21:55.49 | AC_Jay | well anyway, my moh mysteriously stopped working. all signs point to asterisk no longer being able to read the .gsm files I have setup for moh, and I don't know why. it never had a problem before and I haven't made changes to my asterisk box in weeks. |
| 21:56.07 | brodiem | Yourname` while the call is active sip show channels to see the codecs |
| 21:56.25 | Yourname` | ok |
| 21:57.30 | Yourname` | brodiem: Where would it show the codec? It's not showing currently |
| 21:57.59 | Yourname` | Nvmd. |
| 21:58.00 | Yourname` | Got it.. |
| 21:58.05 | Yourname` | Hmm, now I gotta figure this out. |
| 21:58.24 | *** join/#asterisk riddlebox (n=james@75-132-225-75.dhcp.stls.mo.charter.com) |
| 21:59.05 | Yourname` | brodiem: How do I make it so all these warnings don't happen? I set verbose 10, and it still fills up the window.. and I can't see sip channels |
| 22:00.03 | brodiem | set verbose 0 |
| 22:00.14 | AC_Jay | anyone? :( |
| 22:00.35 | Yourname` | err, sorry.. I meant 0 |
| 22:01.30 | brodiem | guess you'd havve to turn it off from logger.conf...just do a sip show channels, then hangup, then scroll up your buffer |
| 22:01.51 | *** join/#asterisk BSD_Tech (n=BSDTech@adsl-69-234-234-206.dsl.irvnca.pacbell.net) |
| 22:01.53 | BSD_Tech | hey guys |
| 22:01.57 | BSD_Tech | need info |
| 22:02.02 | BSD_Tech | why this wont woork |
| 22:02.06 | BSD_Tech | conf => 7[1-9][1-9] |
| 22:02.17 | BSD_Tech | or conf => _7[1-9][1-9] |
| 22:02.32 | BSD_Tech | in meetme.conf |
| 22:02.45 | Qwell[] | because...it's not supposed to |
| 22:02.49 | brodiem | heh |
| 22:02.54 | russellb | where did you see that file supported patterns? |
| 22:02.56 | Qwell[] | use dynamic conferences |
| 22:03.19 | Qwell[] | (or add 88 lines...either way) |
| 22:03.30 | Qwell[] | 88? 81 |
| 22:04.01 | BSD_Tech | I want it to create conf rooms on the fly |
| 22:04.08 | Qwell[] | so use dynamic conferences |
| 22:04.50 | AC_Jay | any ideas on my moh problem? |
| 22:05.46 | *** join/#asterisk blackmousepad (n=blackmou@71-13-69-254.static.aldl.mi.charter.com) |
| 22:05.46 | BSD_Tech | ok I will have to mape it |
| 22:05.46 | BSD_Tech | grrr |
| 22:05.57 | BSD_Tech | now to find the page |
| 22:06.34 | *** join/#asterisk |dennis| (n=dennis@200.32.236.10) |
| 22:08.18 | *** join/#asterisk mindCrime (n=chatzill@cpe-065-190-188-124.nc.res.rr.com) |
| 22:08.40 | BSD_Tech | ok is there a way to have it read the user passwords and admin passwords from a file |
| 22:08.59 | BSD_Tech | for meet me |
| 22:09.44 | AC_Jay | my moh mysteriously stopped working. all signs point to asterisk no longer being able to read the .gsm files I have setup for moh, and I don't know why. it never had a problem before and I haven't made changes to my asterisk box in weeks. i have autoload enabled in my modules.conf file. I can't figure this out. Any thoughts? |
| 22:10.33 | brodiem | BSD_Tech yes, the meetme.conf :) |
| 22:10.52 | BSD_Tech | I ment for dynamic conf rooms |
| 22:10.58 | BSD_Tech | not statis |
| 22:11.02 | BSD_Tech | static |
| 22:11.48 | *** join/#asterisk mirco (n=mirco@p54B27673.dip.t-dialin.net) |
| 22:12.00 | AC_Jay | hello? |
| 22:12.22 | BSD_Tech | time to play around and test new ideas |
| 22:12.57 | AC_Jay | anyone out there? can anyone even see my cries for help? :P |
| 22:13.25 | brodiem | BSD_Tech meetme.conf is ready each time MeetMe() is executed |
| 22:13.28 | De_Mon | AC_Jay what errors do you receive? |
| 22:13.29 | brodiem | ready=read |
| 22:13.56 | Qwell[] | BSD_Tech: If you were trying to use pattern matching, it means that all of the pins would've been the same - so why not just use a pattern match in the dialplan, and set the pin there? |
| 22:14.00 | AC_Jay | De_Mon: "Found no files in <directory here>" |
| 22:14.24 | brodiem | so have meetme.conf include your room identifiers in a sep file that you generate |
| 22:14.28 | De_Mon | AC_Jay have you made sure that directly really exists? maybe the name changed |
| 22:14.29 | lmadsen | do files exist? does the path exist? does it match to what is in 'moh show classes' ? |
| 22:14.33 | AC_Jay | De_Mon: "Unable to spawn mp3player" |
| 22:14.51 | AC_Jay | Yes, De_Mon and the files are there. They're .gsm files however. Never had a problem playing them before |
| 22:14.52 | De_Mon | AC_Jay that would cause a problem for sure |
| 22:15.01 | AC_Jay | Up until about 2 hours ago |
| 22:15.06 | De_Mon | if the mp3player doesn't start, you have a big problem |
| 22:15.09 | AC_Jay | I have made no changes to Asterisk |
| 22:15.17 | BSD_Tech | ok |
| 22:15.18 | De_Mon | AC_Jay maybe you made changes to the mp3player |
| 22:15.26 | AC_Jay | I haven't touched the system in weeks |
| 22:15.58 | AC_Jay | MoH was working this morning. A client of mine told me there was dead air when I placed him on hold. |
| 22:16.03 | De_Mon | AC_Jay find out what mp3player its trying to spawn and find out why it woln't spawn |
| 22:16.05 | AC_Jay | I checked the CLI and sure enough he was right |
| 22:16.21 | *** join/#asterisk ShadowHntr (i=sentinel@wikipedia/Shadowhntr) |
| 22:16.40 | AC_Jay | I'm somewhat of an asterisk newbie. How can I find out what mp3player it's trying to spawn? |
| 22:17.10 | De_Mon | look at /etc/asterisk/musiconhold.conf |
| 22:17.26 | De_Mon | how did you setup music on hold and not know that? |
| 22:18.55 | AC_Jay | I know about musiconhold.conf. There's nothing listed in there that shows what application it's trying to use. My "application =" line is commented out and always has been. |
| 22:19.07 | AC_Jay | To my knowledge. |
| 22:23.23 | *** join/#asterisk riddlebox (n=james@75-132-225-75.dhcp.stls.mo.charter.com) |
| 22:27.54 | *** join/#asterisk VoIPMasta (n=bsd@dial-148-240-58-177.zone-2.dial.net.mx) |
| 22:27.57 | VoIPMasta | Hi there |
| 22:28.15 | VoIPMasta | I'm stuck with something here and about to kill myself, as it's an error that I hadn't seen in a while |
| 22:28.44 | VoIPMasta | retrans_pkt: Maximum retries exceeded on transmission. asterisk and SIP device on the same LAN... |
| 22:28.46 | VoIPMasta | any ideas? |
| 22:28.54 | *** part/#asterisk brodiem (n=brodiem@67.110.68.66.ptr.us.xo.net) |
| 22:29.14 | JT | yes, try to send rtp packets to the correct address :) |
| 22:29.21 | VoIPMasta | hi JT |
| 22:29.26 | JT | hi |
| 22:29.45 | VoIPMasta | apparently it has the correct address |
| 22:29.54 | VoIPMasta | or at least that seems when doing a show peer |
| 22:30.16 | JT | doing a packet dump shows all the rtp flowing to the correct places? |
| 22:30.22 | AC_Jay | any other ideas, de_mon? |
| 22:30.49 | VoIPMasta | JT: I don't have any packet monitoring software in that box :( |
| 22:31.25 | JT | it's not hard to install tdpdump |
| 22:31.37 | VoIPMasta | but the rtp debug shows the right IP |
| 22:32.16 | JT | is it actually making it to the destination? |
| 22:32.39 | VoIPMasta | JT do you mind if I open a query (PM) window to show you the output? |
| 22:32.51 | fujin | ~pb |
| 22:32.54 | 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 |
| 22:32.54 | VoIPMasta | I'm about to reformat that box lol |
| 22:33.00 | *** join/#asterisk mmlj4 (n=jkelly@ip70-171-92-106.no.no.cox.net) |
| 22:33.23 | fujin | Use the pastebin |
| 22:33.26 | fujin | so we can ridicule you |
| 22:33.27 | fujin | ;| |
| 22:33.32 | VoIPMasta | fujin: I know what pb is... but I'm sure the solution to this problem is far too simple, so simple that it's being overlooked |
| 22:33.43 | JT | use pastebin |
| 22:33.54 | _bobweever_ | Does asterisk have any particular problems with fragmented invites? |
| 22:34.34 | fujin | It's udp :P |
| 22:35.18 | VoIPMasta | http://www.pastebin.ca/650428 |
| 22:35.20 | VoIPMasta | ok there ya go |
| 22:37.10 | VoIPMasta | I can dial ATA=>Asterisk but not the other way |
| 22:37.55 | JT | VoIPMasta: what's the bottom bit, from the linksys? |
| 22:38.11 | VoIPMasta | that's the output of a sip show peer |
| 22:38.52 | JT | ok, so how about you turn on sip debug and try that again? |
| 22:39.28 | *** join/#asterisk anthm (n=anthm@mba0736d0.tmodns.net) |
| 22:39.28 | *** mode/#asterisk [+o anthm] by ChanServ |
| 22:41.09 | *** join/#asterisk Igbothom_3rd (n=Hilton@office.quarkit.com.au) |
| 22:41.19 | *** part/#asterisk frenzy (n=frenzy@unaffiliated/frenzy) |
| 22:41.27 | *** join/#asterisk CtRiX (n=CtRiX@aretha.navynet.it) |
| 22:44.39 | x86 | re |
| 22:45.19 | VoIPMasta | http://www.pastebin.ca/650438 |
| 22:45.54 | *** join/#asterisk zerohalo (n=zeroHalo@h-74-2-90-66.cmbrmaor.covad.net) |
| 22:47.58 | JT | VoIPMasta: sure there's no firewall? |
| 22:48.19 | VoIPMasta | nope |
| 22:48.31 | VoIPMasta | never mind, just found the solution |
| 22:48.41 | VoIPMasta | as I said... so simple I was overlooking it |
| 22:48.43 | VoIPMasta | :) |
| 22:50.03 | drwelby | VoIPMasta: What was it? |
| 22:50.20 | x86 | quick poll: digium, sangoma, or rhino for best quad-span T1 card? |
| 22:50.27 | VoIPMasta | the freaking callerid setting |
| 22:51.08 | x86 | any opinions? |
| 22:51.11 | VoIPMasta | the extension should be 101@ip, but since there was a caller id set, asterisk was changing the extension to the one in the caller id |
| 22:51.34 | x86 | eh, shouldn't have |
| 22:51.35 | VoIPMasta | x86, I would go or sangoma but would feel guilty about not supporting digium :) |
| 22:52.10 | JT | why would you feel guilty |
| 22:52.15 | JT | it's an open marketplace |
| 22:52.21 | VoIPMasta | well because I'm making a profit out of asterisk |
| 22:52.34 | VoIPMasta | and not paying a single cent for it, so at least I try to buy their hardware and licenses |
| 22:52.48 | JT | yawn |
| 22:54.29 | x86 | heh |
| 22:55.21 | VoIPMasta | well, I gotta go. bbl. thank you guys |
| 22:56.22 | russellb | but what you want, but the more cards we sell, the more people we can pay to work on asterisk :) |
| 22:56.29 | russellb | s/but/buy/ |
| 22:56.56 | russellb | as they are directly proportional |
| 22:57.29 | russellb | the only other hardware vendor that contributes anything is Xorcom |
| 22:58.12 | AC_Jay | so my asterisk box decided to stop playing my directory full of .gsm files for callers on hold. replacing them with their mp3 counterparts fixed the problem, but I have no idea why * would just stop playing the .gsm files out of the blue. Any ideas? |
| 23:00.25 | x86 | russellb: ever play with those Xorcom Astribanks ? |
| 23:00.34 | russellb | nope, sure haven't |
| 23:00.36 | x86 | USB channel banks... i'm kind of weary about trying them |
| 23:00.44 | x86 | supposedly supported natively by zaptel |
| 23:00.45 | russellb | i bet they're pretty cool, actually |
| 23:00.48 | russellb | yes, they are |
| 23:00.57 | russellb | they contribute a lot of patches for zaptel stuff |
| 23:01.00 | russellb | and their driver is in the source |
| 23:01.04 | x86 | they'd save me money on T1 cards, that's for sure ;) |
| 23:01.12 | russellb | as i said, they are the only other vendor that contributes *anything* ... |
| 23:01.18 | JT | usb is never cool for telecomms ;) |
| 23:01.22 | russellb | heh |
| 23:01.26 | x86 | JT: ever try it? |
| 23:01.30 | russellb | well, like i said, never had one .. |
| 23:01.34 | JT | nope, not planning to |
| 23:02.32 | x86 | I think if I actually got to play with one and test it out, i might feel more comfortable laying down $1,000 USD or more on one ;) |
| 23:02.43 | *** join/#asterisk mercestes (n=Merceste@rrcs-71-41-157-70.sw.biz.rr.com) |
| 23:02.56 | JT | i'd feel more comfortable knowing usb was nowhere to be seen |
| 23:03.01 | x86 | as it is now, I buy Rhino channel banks, sangoma T1 cards, and digium analog cards |
| 23:03.29 | x86 | the rhino channel banks are actually quite nice (note, i've not played with any other channel banks) |
| 23:03.44 | x86 | auto-detects signalling and framing... how cool is that? |
| 23:04.00 | JT | i dunno |
| 23:04.05 | x86 | like 20 second initial setup right out of the box... |
| 23:04.13 | x86 | JT: what channel banks do you use? |
| 23:04.19 | mercestes | Are quintum sip gateways any good? |
| 23:04.35 | JT | autodetection doesn't sound like something very telco grade |
| 23:04.40 | JT | i generally don't use channel banks |
| 23:04.47 | JT | but the best brands are Adtran and CAC |
| 23:05.16 | red9012 | I can I generate ring tones using only one command ( ringing cmd requires a wait() ) |
| 23:05.38 | x86 | yeah i've heard nothing but good about adtran too |
| 23:05.44 | x86 | been thinking about trying them |
| 23:06.07 | *** join/#asterisk jm|home (n=jm|home@zen.jamiem.com) |
| 23:06.11 | x86 | i've used adtran dax's before |
| 23:06.29 | x86 | and adtran netvanta routers, and (of course) adtran CSU/DSU's ;) |
| 23:07.04 | russellb | i can practically see adtran out my window |
| 23:08.59 | russellb | my first "real job" was testing netvanta routers :) |
| 23:10.04 | nDuff | (funny thing is that the only Polycoms we have are analog). |
| 23:11.19 | nDuff | nice little family-owned sandwich shop downstairs in their building (the Austin location). |
| 23:11.52 | russellb | their phones rock |
| 23:12.27 | Dan0maN_Work | austin polycom location? |
| 23:12.29 | nDuff | hrm, actually looks like they've moved too. |
| 23:12.38 | nDuff | ...they're off of 620 now, not right on 360. |
| 23:12.50 | nDuff | Dan0maN_Work: *nod*. |
| 23:12.53 | Dan0maN_Work | gotcha |
| 23:13.04 | Dan0maN_Work | they used to have the dream work building there on the lake |
| 23:13.13 | nDuff | Dan0maN_Work: yeah, we were right next door. |
| 23:13.22 | nDuff | Dan0maN_Work: ...subletting from trilogy, until they moved out. |
| 23:13.22 | Dan0maN_Work | i applied to work in the data center of some company at one point |
| 23:13.29 | Dan0maN_Work | coo |
| 23:13.36 | Dan0maN_Work | i'm at duval n mopac |
| 23:13.49 | Dan0maN_Work | small world this internet is ;) |
| 23:14.26 | Dan0maN_Work | heh. my company used to be there. moved in '02 |
| 23:14.29 | russellb | w00t! |
| 23:14.43 | mog | how did you do it rus |
| 23:14.46 | russellb | magic. |
| 23:14.47 | mog | er russellb |
| 23:14.49 | mog | nice |
| 23:15.16 | AC_Jay | so my asterisk box decided to stop playing my directory full of .gsm files for callers on hold. replacing them with their mp3 counterparts fixed the problem, but I have no idea why * would just stop playing the .gsm files out of the blue. Any ideas? |
| 23:15.16 | nDuff | Dan0maN_Work: good for 'yall. bloody hate this building. Granted, it's big and cheap... but there's a reason for the latter part. |
| 23:15.17 | mog | iax2 element? |
| 23:15.26 | russellb | truthfully ... a combination of an publish/subscribe event API written for inside asterisk ... and a module to tie into a clustering framework that has an eventing service (openais) |
| 23:16.37 | mog | is it in a public branch? |
| 23:16.45 | russellb | mog: yeah |
| 23:16.52 | russellb | well, the event API is already in trunk |
| 23:16.59 | mog | nice |
| 23:17.02 | russellb | the clustering framework glue is in ... asterisk/team/russell/ais |
| 23:17.06 | *** part/#asterisk AC_Jay (n=Jay@ns1.accu-com.com) |
| 23:18.48 | *** join/#asterisk p0lar69 (i=p0lar69@155.101.179.29) |
| 23:18.51 | p0lar69 | Hey all |
| 23:19.28 | p0lar69 | I gots a quick questions |
| 23:19.32 | p0lar69 | any takers? |
| 23:20.11 | generalhan | p0lar69: no one is going to offer their assistance before hearing the question(s) |
| 23:20.17 | p0lar69 | heh |
| 23:20.18 | p0lar69 | ok |
| 23:20.39 | p0lar69 | how do I force a Cisco 7960 to use 5060 port |
| 23:20.46 | p0lar69 | I have port=5060 in the sip.conf |
| 23:21.01 | p0lar69 | but the * server says its registered at 1365 |
| 23:21.06 | p0lar69 | in sip show peers |
| 23:21.14 | generalhan | p0lar69: configuration files (tftp) or on the actual phone itself in SIP Settings |
| 23:21.19 | p0lar69 | i have defined it as dynamic and static |
| 23:21.24 | p0lar69 | both |
| 23:22.02 | *** join/#asterisk wishes (n=wishes@60.234.20.178) |
| 23:22.13 | wishes | coor blimey, thats a good sized channel |
| 23:22.35 | wishes | :D |
| 23:22.48 | mercestes | 98% of us are lurkers |
| 23:22.56 | wishes | ahh such is irc :) |
| 23:23.00 | generalhan | p0lar69: i am no expert, in any way, shape, or form, all i can tell you is what i did ... and that was configure the SIPDefault, and SIP<MAC> files to use port 5060 and it worked |
| 23:23.13 | p0lar69 | ok ill check again |
| 23:23.24 | wishes | well, im here to be picking your brains and reading docs. I have an asterisk server i have to mangle into doing my bidding and ive never touched one before :) |
| 23:23.30 | p0lar69 | any way that you know of to do tftp on a different port? |
| 23:23.34 | wishes | but for now, i gotta read some docs :) |
| 23:23.52 | generalhan | p0lar69: what do you mean ? |
| 23:24.56 | *** join/#asterisk Ryushin (i=proxy@windwalker.openinnovations.com) |
| 23:27.13 | x86 | p0lar69: the phone IS connected to Asterisk on port 5060 |
| 23:27.30 | x86 | p0lar69: 5060 is the remote port (asterisk side), 1365 is the local port (phone side) |
| 23:27.42 | x86 | p0lar69: phone port is completely random |
| 23:29.05 | *** join/#asterisk l2trace9999 (n=l2trace@75.112.133.254) |
| 23:30.15 | *** join/#asterisk angom_h (n=Angel@dsl-200-67-220-63.prod-empresarial.com.mx) |
| 23:31.30 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 23:31.34 | el_critter | you have been able to use x-lite with a bluetooth headset? |
| 23:32.20 | p0lar69 | ok |
| 23:32.37 | p0lar69 | is there a way to force the phone side port? |
| 23:32.43 | p0lar69 | for firewall forwarding? |
| 23:32.45 | JT | p0lar69: no, why would you? |
| 23:32.58 | el_critter | JT: sorry, wrong channel :), but anyway... anyone in here is able to use it? |
| 23:32.59 | p0lar69 | NAT Firewall |
| 23:33.05 | JT | err, you only need to forward in one direction |
| 23:33.16 | JT | p0lar69: there is no need to play silly buggers to use nat |
| 23:33.43 | p0lar69 | well I have my firewall setup to forward port 5060 to my phone |
| 23:34.06 | p0lar69 | the phone registers but in the sip show peers its 1365 or something random |
| 23:34.16 | JT | that's silly |
| 23:34.17 | l2trace9999 | anyone know of a way to implement 3 way calling as a feature ? |
| 23:34.22 | JT | you don't need to forward to phones |
| 23:34.24 | p0lar69 | 1316 now |
| 23:34.33 | p0lar69 | well my firewall does |
| 23:34.37 | JT | ... |
| 23:34.39 | JT | ~sipnat |
| 23:34.40 | jbot | [~sipnat] Quick guide on configuring * + SIP behind NAT : http://www.aocomputing.net/?p=3 otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
| 23:34.52 | JT | allow is not the same as forward |
| 23:35.10 | p0lar69 | ok lemme read. its a netscreen firewall in nat mode... |
| 23:35.13 | p0lar69 | reading now |
| 23:35.15 | p0lar69 | thx |
| 23:36.08 | *** join/#asterisk vaughany (n=vaughanc@vcahill.globalcenter.net.au) |
| 23:36.24 | p0lar69 | also |
| 23:36.28 | p0lar69 | while your here. |
| 23:36.36 | p0lar69 | can you use a port number in the tftp line |
| 23:36.37 | p0lar69 | ? |
| 23:36.52 | p0lar69 | to get configs? on a cisco 7960? |
| 23:37.07 | p0lar69 | like alt tftp server 1.1.1.1:6969 |
| 23:37.12 | JT | no idea |
| 23:37.20 | JT | i avoid cisco |
| 23:37.32 | p0lar69 | heh |
| 23:37.36 | p0lar69 | ok thanks anyways |
| 23:38.26 | x86 | cisco phones sucks, tbh |
| 23:38.31 | x86 | polycom FTW! |
| 23:39.49 | generalhan | bah ! |
| 23:39.54 | wishes | softphones :D |
| 23:39.56 | generalhan | LOVE my 7960s |
| 23:39.59 | wishes | cheaper! |
| 23:40.12 | generalhan | wishes: not if you have to buy a computer for it to sit on ! lol |
| 23:40.12 | p0lar69 | can you define alt tftp ports on the Cisccos? |
| 23:40.23 | generalhan | p0lar69: not sure ... never had a reason to try that |
| 23:40.24 | wishes | generalhan: my cellphone does voip :D |
| 23:40.29 | JT | wishes: softphones are ratshit |
| 23:40.30 | wishes | via wifi even |
| 23:40.52 | wishes | why are softphones ratshit anyway? |
| 23:40.57 | vaughany | <PROTECTED> |
| 23:41.04 | vaughany | doesn't look like you can specify port |
| 23:41.06 | wishes | just quality etc? bad software? |
| 23:41.21 | p0lar69 | ok |
| 23:41.23 | JT | because they've never put the effort in to making them sound as good as a polycom or similar |
| 23:41.23 | p0lar69 | thx |
| 23:41.26 | JT | and also |
| 23:41.28 | JT | support nightmare |
| 23:41.39 | JT | you need to get all your speaker and mic levels right for it to sound ok |
| 23:41.46 | generalhan | p0lar69: im sure there are hackish ways to do that |
| 23:41.51 | JT | which is bad on an end users' desk |
| 23:41.54 | wishes | well i dunno, ive had a crapload of problems with the bunch of voip phones we've had , but few problems with softphones |
| 23:42.30 | JT | wishes: what phones? |
| 23:42.38 | generalhan | p0lar69: like port forwarding at a router from port 69 to port whatever and configuring your tftp server to listen on that port ... then just direct your ciscos to the router for the tftp if you REALLY need to get it away from the default tftp port |
| 23:42.39 | wishes | umm hang on ill go check |
| 23:42.42 | wishes | shit ones anyway :) |
| 23:42.58 | wishes | "grandstream" |
| 23:43.00 | JT | haha |
| 23:43.02 | JT | ~gs |
| 23:43.02 | jbot | GrandSuck phones & gateways are cheap junk which should be avoided with extreme prejudice. |
| 23:43.15 | generalhan | ~grandstream |
| 23:43.16 | jbot | well, grandstream is the Yugo of VoIP hardware. Run. Run away now. |
| 23:43.29 | JT | ~phones |
| 23:43.29 | jbot | phones is probably http://bani.anime.net/phones/. While personal preference will dictate which phone works best for you, general consensus on a rough order of quality and suggestibility is as follows: Polycom (any), Aastra 480i, Aastra 5i Series, Cisco 7940+, Linksys SPA-9XX, Snom, and finally everything else. Do not consider Grandstream phones. Ever. |
| 23:43.30 | generalhan | i know there is a funnier one ... where is it |
| 23:43.44 | wishes | im just taking over this job, i gotta deal with the previous persons poor choices (which he made a lot of) |
| 23:43.55 | JT | fair enough |
| 23:44.04 | JT | but i'd hate to deploy softphones to end users |
| 23:44.17 | vaughany | Guys, i keep getting registration timeout requests on my trunks, they will work for a few hours then just working |
| 23:44.28 | wishes | as a company, softphones = cheap |
| 23:44.29 | generalhan | i never hear anyone talk about the Aastra 9133i phones ... i have had VERY good luck with those ! |
| 23:44.37 | wishes | and people can work from home and externally |
| 23:44.39 | vaughany | show sip, shows request sent. |
| 23:44.43 | wishes | (mostly the latter) |
| 23:44.51 | generalhan | very easy setup and i have had some for 3+ years now that are still running great |
| 23:45.17 | vaughany | can i paste small debug here? |
| 23:45.18 | x86 | i've had very good luck with polycom 601's :) |
| 23:45.29 | wishes | luck? or skill? |
| 23:45.36 | generalhan | vaughany: pastebin |
| 23:45.40 | generalhan | ~pb |
| 23:45.41 | 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 |
| 23:46.34 | JT | wishes: cheap, not to support. i'd give home telecommuters ip phones or ATAs |
| 23:46.39 | vaughany | is there anything in a debug, that could gain passwords from before i pass my debugs |
| 23:46.45 | vaughany | paste |
| 23:47.07 | generalhan | vaughany: lol depends on what youre pasting ... why not look through it and search for your passwords |
| 23:47.22 | *** join/#asterisk Yourname` (n=chatzill@76-10-128-178.dsl.teksavvy.com) |
| 23:47.28 | wishes | JT: support isnt a major problem, mostly developers or geeks using the softphones working from home or technically capable people. The customer support will probably still stay here using the ugly grandstream |
| 23:47.54 | JT | wishes: the audio quality is massively different, too |
| 23:48.01 | JT | as well as ease of use and what not |
| 23:48.09 | Yourname` | Hello, I have two DIDs and two extensions. I've successfully routed both the DIDs to go to separate extensions. Now, I'm trying to make it so that each extension shows its own callerid when it dials out. How can I do that? |
| 23:48.14 | x86 | JT: i've always set the phone up to point to a hostname (ex: voip.domain.com), then have internal DNS on my LAN pointing that hostname to an internal IP (ex: 10.1.2.3), and external DNS pointing to an external IP that's 1:1 NAT'd to the Asterisk box (ex: 12.3.4.5) |
| 23:48.40 | wishes | Im not the money spender or decider of such things. im just setting it up how its been asked so ill do that to the best of my abilities |
| 23:48.44 | *** join/#asterisk Cyllene (n=cy@unaffiliated/cyllene) |
| 23:48.52 | JT | x86: okay? |
| 23:48.56 | x86 | so people can take their phones home if they want with no configuration changes needed at all |
| 23:49.01 | JT | ah |
| 23:49.06 | JT | nice |
| 23:49.10 | x86 | *nod* |
| 23:49.15 | x86 | works very well |
| 23:49.30 | Cyllene | Hi. I am using asterisk 1.4.10 and am having problems using one touch monitoring. My extansions.conf file looks like this: |
| 23:49.43 | vaughany | here is my /var/log/asterisk/full log http://pastebin.com/d3b898eaa <- 24lines |
| 23:49.52 | x86 | Cyllene: DO NOT PASTE HERE |
| 23:50.02 | x86 | Cyllene: use a pastebin, like http://pastebin.ca/ |
| 23:50.23 | generalhan | x86: haha he just pastes his entire extensions.conf here ! lol ... how many lines before flood boots ? |
| 23:50.29 | Cyllene | 2,1,Dial(IAX2/tom,30,HWT) |
| 23:50.34 | Cyllene | It's just one line, x86. |
| 23:50.47 | jgoddess | where is the extension? |
| 23:51.10 | jgoddess | shouldn't be IAX1/tom/$EXTEN,30,HWT? |
| 23:51.19 | jgoddess | user variable for real exten |
| 23:51.21 | Cyllene | I am using IAX2. |
| 23:51.30 | jgoddess | whatever |
| 23:51.33 | jgoddess | that isn't the questino |
| 23:51.33 | Cyllene | And I am calling "tom" directly. |
| 23:51.50 | Cyllene | I do not believe I need an extension. "tom" is a softphone |
| 23:52.12 | x86 | jgoddess: looks fine if he's registering to a context at "tom", and that context at "tom" has an "s" exten setup |
| 23:52.26 | x86 | Cyllene: that wont work then |
| 23:52.33 | *** join/#asterisk JackEStorm (n=no@ip68-225-77-136.no.no.cox.net) |
| 23:52.35 | Cyllene | How come? |
| 23:52.39 | x86 | Cyllene: is your softphone registered to the same asterisk server? |
| 23:52.44 | Cyllene | Yes. |
| 23:52.57 | x86 | hmm, then yeah it should work |
| 23:53.05 | Cyllene | Right |
| 23:53.13 | Cyllene | It rings fine |
| 23:53.19 | Cyllene | I can establish a connection. |
| 23:53.23 | Cyllene | But OTR doesn't work |
| 23:53.43 | Yourname` | Heylo? |
| 23:53.53 | x86 | you sure you have the correct flags in the dial command? |
| 23:53.59 | x86 | Yourname`: hey |
| 23:54.04 | Cyllene | As you can see, "HWT" |
| 23:54.09 | Yourname` | x86: I have two DIDs and two extensions. I've successfully routed both the DIDs to go to separate extensions. Now, I'm trying to make it so that each extension shows its own callerid when it dials out. How can I do that? |
| 23:54.33 | De_Mon | Yourname` how do you set the callerid now? |
| 23:54.37 | vaughany | can anyone help with my sip timeout issues? I know when i do a sip reload, that all the registrys return to normal. When i checked the logs yesterday it was a dns issue, so i added the sip server to /etc/hosts, but today i get .. registration timeout, trying again, and it looks like it just keeps trying. |
| 23:54.45 | x86 | Cyllene: is that correct though? |
| 23:54.59 | generalhan | Cyllene: you have features.conf setup correctly for that W feature ? |
| 23:55.30 | Yourname` | De_Mon: Using the callerid= on the provider's context. |
| 23:55.42 | Cyllene | I do believe so. |
| 23:55.46 | generalhan | Cyllene: Dial option "W: Allow the calling user to start recording after pressing *1 or what defined in features.conf (Asterisk v1.2.x); requires Set(DYNAMIC_FEATURES=automon)" |
| 23:55.54 | Yourname` | De_Mon: Tried doing the same with the extensions as well, doesn't work. |
| 23:55.59 | Cyllene | generalhan: Right |
| 23:56.04 | Cyllene | features.conf: |
| 23:56.09 | Cyllene | [featuremap] |
| 23:56.16 | Cyllene | automon => *1 |
| 23:56.19 | De_Mon | Yourname` sip.conf allows you to set callerID per peer ya know |
| 23:57.23 | generalhan | Cyllene: and exten => 123,1,Set(DYNAMIC_FEATURES=automon) in your dialplan ?> |
| 23:57.34 | Yourname` | De_Mon: [gafachi] is my outbound context that has callerid= .. [100] or [200] even though they have their own 'callerid=' .. it still defaults to the callerid set in [gafachi] |
| 23:58.00 | *** part/#asterisk BSD_Tech (n=BSDTech@adsl-69-234-234-206.dsl.irvnca.pacbell.net) |
| 23:58.05 | generalhan | Cyllene: http://www.voip-info.org/wiki/index.php?page=Asterisk+config+features.conf read that ESPECIALLY THE "EXAMPLES" |
| 23:58.07 | Cyllene | generalhan: I have it globally defined, yes. |
| 23:58.25 | x86 | Cyllene: it's a channel variable, can't be globally defined ;) |
| 23:58.52 | jgoddess | what I was thinking x86 ;) |
| 23:59.01 | generalhan | x86: it says it can in the config page on voip-info |
| 23:59.13 | De_Mon | Yourname` yes.. forcing a callerid change before the call goes out will replace any existing callerid |
| 23:59.16 | *** join/#asterisk angom_w (n=Angel@189.140.23.110) |
| 23:59.32 | Yourname` | De_Mon: Meaning, I remove the callerid from gafachi, and then try? |
| 23:59.51 | De_Mon | well, if its not changing it, that would make sense wouldn't it? |