| 00:03.28 | raden | Naikrovek, what u doing on so late ? |
| 00:04.04 | Naikrovek | hell i don't know |
| 00:08.20 | SeRi | Naikrovek, you know how I can rset the device? |
| 00:08.42 | Naikrovek | what do you mean reset it? |
| 00:08.55 | SeRi | to factory |
| 00:08.56 | Naikrovek | power cycle it? format it? reset something else? |
| 00:09.15 | Naikrovek | best you can do is format it, as far as I know |
| 00:09.30 | Naikrovek | in the admin menu, you can do that |
| 00:09.54 | *** join/#asterisk eject_ck (~eject_ck@62.205.134.210) |
| 00:09.56 | SeRi | mhhh ok ill try that... If I can get in :) |
| 00:19.11 | pdtpatrick1 | Question ... anyone here have experience with asterisk + exchange server? unified communications .. any idea? |
| 00:28.59 | treborsux | <treborsux> if one phone boots from tftp no issue |
| 00:29.00 | treborsux | <treborsux> but another doesnt what do i change? |
| 00:34.49 | *** join/#asterisk JonasHB (~chatzilla@hb.lcaig.com) |
| 00:34.56 | JonasHB | hey guys, any help here appreciated, im playing with Marcus Brown's awesome Google Voice module for FreePBX, ive got it all working, but incoming calls are not getting caught by DID incoming routes and are going through to my default inbound route... Any ideas how to catch these, I'd like send incoming goovle voice calls to a diffrent destination |
| 00:38.37 | p3nguin | ~freepbx |
| 00:38.38 | infobot | [~freepbx] FreePBX is unable to be supported here. It is made up of complex dialplans and scripts which can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking there, or fpbx |
| 00:39.04 | p3nguin | If you were using Asterisk, I'd probably tell you how. |
| 00:39.19 | *** join/#asterisk leftist (~dizzy@50-10-91-159.gar.clearwire-wmx.net) |
| 00:39.49 | JonasHB | p3nguin: no disrespect and understood |
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| 06:35.08 | SteveWilliams | Hi! I need some guidance on Setting up my asterisk server to make outbound ivr calls. Please help. I am a newcomer. |
| 06:36.46 | jkroon | hi guys, having a problem on ast 1.8.5.0 server and 1.8.5.0 client with realtime sip on server side. can register, ut can't place calls. |
| 06:37.08 | jkroon | client-side spits out Failed to authenticate on INVITE to '"asterisk" <sip:0860100001@c3po.local.uls.co.za>;tag=as0831596b' |
| 06:37.50 | irroot | jkroon hi there pb a sip debug please |
| 06:38.08 | irroot | SteveWilliams hi outbound ivr calls ?? please elaborate |
| 06:38.17 | jkroon | INVITE, response, 401, ACK, error message. |
| 06:38.34 | irroot | you get a 407 ?? |
| 06:38.52 | jkroon | 401 |
| 06:39.55 | irroot | you should get a 407 [request auth] |
| 06:40.03 | irroot | before the 401 |
| 06:40.05 | jkroon | 407 is proxy auth |
| 06:40.20 | jkroon | no proxies involved, so straight 401 seems reasonable to me. |
| 06:41.33 | SteveWilliams | @irroot. I want my customers to be called up by our autodialer and listen to a voice recording. I was able to setup outbound campaign through VICIDial. |
| 06:42.18 | SteveWilliams | Then they would press 1 for a specific recording to play |
| 06:42.28 | SteveWilliams | and press 2 for another one |
| 06:42.37 | jkroon | irroot, http://pastebin.com/w7twpAeu |
| 06:42.44 | kaldemar | jkroon: have you set fromuser and/or defaultuser for the peer you use to dial out? |
| 06:42.57 | irroot | there will be a request for auth first |
| 06:43.09 | jkroon | i have authuser .. but no, and *facepalm* doh |
| 06:43.35 | jkroon | actually, no, already had both fromuser and defaultuser |
| 06:44.09 | irroot | got a secret or md5 bits on one side but not other ?? |
| 06:44.19 | jkroon | secret= on both sides. |
| 06:44.30 | jkroon | remote secret? |
| 06:44.45 | jkroon | server-side sip show peer on pb. |
| 06:44.53 | jkroon | also pasted the sip.conf from the client. |
| 06:45.00 | irroot | c3po nice :P |
| 06:45.22 | jkroon | :D |
| 06:45.35 | SteveWilliams | Is there a way i can do that by configuring the dialplan? |
| 06:45.35 | jkroon | also have r2d2 in production. |
| 06:45.56 | jkroon | SteveWilliams, sounds like a simple IVR type setup. |
| 06:46.13 | SteveWilliams | @jkroon, yep |
| 06:46.19 | jkroon | IIRC the application you're looking for is Background and WaitForExten |
| 06:46.24 | SteveWilliams | but i am a fresher |
| 06:46.34 | SteveWilliams | ok |
| 06:46.47 | *** join/#asterisk NourSs (~gholzinge@LAubervilliers-151-13-22-64.w217-128.abo.wanadoo.fr) |
| 06:46.54 | SteveWilliams | lemme give them a try |
| 06:47.06 | SteveWilliams | @jkroon, thank you for your help |
| 06:47.22 | NourSs | Hi, i sell two Digium cards ( TE121P and TDM411B ), private me for more details :-) |
| 06:47.27 | jkroon | SteveWilliams, Background(recording) followed by WaitExten |
| 06:47.38 | jkroon | irroot, any ideas for me? |
| 06:47.56 | irroot | jkroon put a defaultuser in |
| 06:48.16 | jkroon | wtf?!? |
| 06:48.26 | jkroon | why does that make a difference? |
| 06:48.36 | irroot | suspect it might |
| 06:48.56 | jkroon | it does! from my reading of the code defaultuser only ever gets used for authuser and fromuser if those options aren't set. |
| 06:49.43 | irroot | yeah thats what it should be agreed |
| 06:50.53 | irroot | as long as it works dude ill put a beer on my account :P |
| 06:52.55 | jkroon | deal |
| 06:54.13 | jkroon | and thanks. |
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| 07:00.45 | irroot | pleasure |
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| 07:11.08 | jkroon | irroot, in a generic setup, fromuser and authuser may differ, to which of those should I set defaultuser? |
| 07:12.02 | irroot | ill need to poke the code defaultuser seems to be one used in auth |
| 07:12.51 | kaldemar | defaultuser is used in digest unless authuser is defined. |
| 07:14.01 | irroot | jkroon kaldemar problem is that authuser seems to be ignored here |
| 07:14.11 | *** join/#asterisk shtoom (~shtoom@59.93.113.184) |
| 07:18.10 | jkroon | kaldemar, so set to authuser. kaldemar isn't that a bug ? |
| 07:18.28 | *** join/#asterisk mintos (~mvaliyav@115.241.53.127) |
| 07:21.31 | kaldemar | based on a brief glance, looks like authuser is used in subscriptions, registers and mwi. |
| 07:23.15 | irroot | but not invites ?? |
| 07:23.21 | irroot | thats where the problem is |
| 07:23.33 | kaldemar | yep, but i may be wrong. |
| 07:25.01 | *** join/#asterisk nunne (~nunne@static-213-115-116-75.sme.bredbandsbolaget.se) |
| 07:25.32 | nunne | Does anyone know where i can get "on hold tone" instead of moh? I guess i need a music file that plays "on hold tone"? :) |
| 07:26.52 | irroot | nunne intresting question not sure there should be a simple playtone on hold not music option i guess but dont recall seen it so may not exist |
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| 07:31.46 | jkroon | nunne, that would be the only solution I can think of yes. |
| 07:34.22 | jkroon | irroot, ever had to interface with MWeb's SIP Proxy thing? |
| 07:34.36 | jkroon | they are killing my auths with SIP/2.0 500 |
| 07:34.43 | jkroon | worked yesterday, today it fails ... |
| 07:34.54 | irroot | jkroon i avoid mweb as a rule they complete moronic muppets |
| 07:35.31 | jkroon | agreed. |
| 07:38.23 | *** join/#asterisk mintos (~mvaliyav@115.184.0.93) |
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| 07:45.43 | *** join/#asterisk mirko_brankovic (~mirko_bra@212.200.146.253) |
| 07:46.33 | mirko_brankovic | does anyone know how to trim a string in Asterisk from right side to some delimiter, for example /? |
| 07:51.22 | *** join/#asterisk ocx (4d2adb52@gateway/web/freenode/ip.77.42.219.82) |
| 07:52.01 | kaldemar | mirko_brankovic: see function CUT |
| 07:52.52 | *** join/#asterisk hehol (~hehol@2001:1438:1009:200:3098:bda:ca21:4da6) |
| 07:52.52 | mirko_brankovic | i look at it, but nothing about cut-ing from right side |
| 07:54.33 | kaldemar | mirko_brankovic: func FIELDQTY will help with that. |
| 07:55.42 | kaldemar | ${CUT(varname,/,${FIELDQTY(varname,/)})} |
| 07:55.45 | *** join/#asterisk tamiel (~tamiel@213.30.183.226) |
| 07:55.54 | mirko_brankovic | aha thx, i have to use both :) |
| 07:57.21 | mirko_brankovic | kaldemar: thank you :) |
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| 08:13.09 | NourSs | Hi, i sell two Digium cards ( TE121P and TDM411B ), private me for more details :-) |
| 08:25.16 | *** join/#asterisk sekil (~sekil@78.24.104.73) |
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| 08:51.09 | irroot | jkroon ASTERISK-18223 |
| 08:53.39 | *** join/#asterisk jmls (~Julian@host217-36-208-155.in-addr.btopenworld.com) |
| 08:53.45 | jmls | morning all |
| 08:53.49 | jmls | is there a way of playing a sound file as a participant in a conference room ? |
| 08:54.16 | jmls | so I dial in to the conference, and some dialplan magic or ami command plays a file to me |
| 08:55.32 | kaldemar | jmls: originate a call, other end to the conference and other and to playback application. |
| 08:56.54 | *** join/#asterisk jkroon (~jkroon@dsl-241-232-137.telkomadsl.co.za) |
| 08:57.52 | jmls | oh, I see - don't use the dialplan |
| 08:58.02 | jmls | must get out of that habit |
| 08:58.15 | jmls | kaldemar: many thanks |
| 09:02.38 | catphish | is it possible to configure how asterisk responds to SIP redirect responses |
| 09:02.54 | *** join/#asterisk cstachris_ (~chrismylo@202.182.147.82) |
| 09:07.33 | catphish | http://www.adambotbyl.com/2010/07/03/moved-temporarily-302-call-forward-in-asterisk-for-cdr-billing/ |
| 09:07.33 | catphish | eww |
| 09:08.46 | *** part/#asterisk mirela666 (~mirko_bra@212.200.146.253) |
| 09:08.52 | *** join/#asterisk mirela666 (~mirko_bra@212.200.146.253) |
| 09:09.42 | NourSs | Hi, i sell two Digium cards ( TE121P and TDM411B ), private me for more details :-) |
| 09:13.36 | cstachris | catphish, yeah - that billing thing is a problem!!! |
| 09:14.39 | cstachris | on that note, can anyone give me some details about a Notify message that is (queued) ?? |
| 09:14.51 | catphish | the problem is that when asterisk received a 302 it dials using a local channel, which is ideal, but it doesn't set "norelease" |
| 09:16.22 | *** join/#asterisk WiretapSeven (~Wiretap@unaffiliated/wiretap) |
| 09:17.03 | SteveWilliams | Hi all, could you help me with my dial plan, please. I want to call up people and want them to listen to a .wav file. This is my incorrect dialplan: ( I am a newbie, please help ) |
| 09:17.11 | SteveWilliams | exten => _8X.,1,AGI(agi://127.0.0.1:4577/call_log) |
| 09:17.23 | SteveWilliams | exten => _8X.,n,Dial(${SIPD}/${EXTEN:2},,tToR) |
| 09:17.30 | SteveWilliams | exten => _8X.,n,Playback(welcome) |
| 09:17.36 | SteveWilliams | exten => _8X.,n,Hangup |
| 09:23.54 | irroot | SteveWilliams rather use the originate app |
| 09:24.07 | jkroon | on SIP, if the INITIATING end is behind nat, does it affect anything if I set nat=yes vs nat=no vs nat=auto? |
| 09:24.58 | SteveWilliams | @rroot, okay, sir. lemme check that out on google. Thanks |
| 09:25.04 | jkroon | SteveWilliams, one of the Dial() options also requests the answered channel be Gosub()ed into another context first before being bridged, you can probably abuse that too./ |
| 09:25.35 | irroot | jkroon dont use nat and then question is answered :P |
| 09:25.59 | irroot | the nat behaviour i dont understand properly but it seems inconsistant |
| 09:26.24 | SteveWilliams | @jkroon thanks! checking that too |
| 09:27.16 | jkroon | irroot, i am dealing with an IEC(N)S license holder here. If you ever hear the names Talkworld or Rick Brits, take my advise: find your closest illegal arms dealer . |
| 09:27.28 | jkroon | the guy is an amateur. |
| 09:27.47 | irroot | jkroon or give them uls number :P |
| 09:28.16 | *** join/#asterisk e7e5 (~rudenko@188.134.2.33) |
| 09:28.21 | kaldemar | jkroon: if someone sends an invite with a private address in the SDP, asterisk will try to use it without nat=yes. |
| 09:28.24 | jkroon | irroot, i gave the guy's client a ULS number as proof that my stuff is working correctly. now he's saying i'm purposefully sabotaging his crap. |
| 09:28.24 | irroot | most them licence holders are clueless |
| 09:29.04 | jkroon | kaldemar, the other end is a "Server: Sip EXpress router (0.9.6 (i386/freebsd))", I'm the poor ass stuck behind NAT. |
| 09:29.15 | irroot | jkroon they using lots of mikrotik with masq on every router ?? if so you screwed |
| 09:29.39 | jkroon | irroot, there is a fix for that conntrack bug. it applies to D-Link and Linux gateways too btw. |
| 09:29.49 | irroot | jkroon sounds like ECN's setup |
| 09:30.16 | jkroon | that was my first reaction too. |
| 09:30.34 | irroot | jkroon but the muppets dont have upgrades or any idea how to do it |
| 09:30.39 | jkroon | interesting capatilization on the Sip EXpress :p |
| 09:30.51 | irroot | indeed |
| 09:31.07 | irroot | thats what got me thinking also bsd |
| 09:31.33 | jkroon | sip trace gets me a 500 error after sending back the auth details. |
| 09:32.01 | kaldemar | jkroon: if your asterisk is behind a NAT, you need to define externaddr and localnet in addition to having nat=yes under [general]. |
| 09:32.02 | jkroon | so sequence is now C:INVITE, S:401, C:ACK, C:INVITE (with WWW-Auth), S:500 |
| 09:32.22 | irroot | 500 eish dis no n poes klap wat eimand soek |
| 09:32.32 | jkroon | kaldemar, ok, let me try that. but how do I deal with a dynamic externaddr? |
| 09:32.48 | irroot | host = dynamic |
| 09:32.54 | irroot | nat = force-rport |
| 09:33.23 | irroot | force_rport |
| 09:33.40 | irroot | or nat = yes |
| 09:33.45 | jkroon | and also, can I have multiple subnets in localnet? specifically, localnet=10.0.0.0/8, 172.16.0.0/12 and 192.168.0.0/16 ? |
| 09:34.53 | kaldemar | jkroon: you define it as externhost and define lookup interval with externrefresh. |
| 09:35.26 | *** join/#asterisk WiretapSeven (~Wiretap@unaffiliated/wiretap) |
| 09:35.28 | jkroon | goes off convincing the client he has to set up a dyndns or get me direct access to the pppoe connection to get rid of the NAT. |
| 09:35.33 | kaldemar | jkroon: yes, you can have multiple localnet definitions. |
| 09:36.11 | *** join/#asterisk gmaruzz (~gmaruzz@2-225-249-20.ip178.fastwebnet.it) |
| 09:36.57 | jkroon | ok, localnet on all local subnets, nat=yes globally, no joy. on that peer, nat=force_rport ? |
| 09:38.33 | jkroon | externrefresh in minutes or seconds? |
| 09:39.22 | *** join/#asterisk aglenday (~Impatient@59.167.161.74) |
| 09:40.42 | irroot | jkroon stun ?? |
| 09:40.42 | kaldemar | jkroon: seconds. the sample config is very helpful with the options. |
| 09:40.58 | irroot | res_stun |
| 09:41.27 | jkroon | irroot, would help if someone in ZA actually had a working stun server. |
| 09:42.04 | jkroon | i've just set up another hack, updating to /etc/hosts with a whatismyip.com hack |
| 09:42.11 | jkroon | http://pastebin.com/4qFkfiEE |
| 09:42.42 | jkroon | irroot, if 500 was a slap in the face, then putting the SIP trace up public must be a nuke up the rear-end. |
| 09:43.02 | jkroon | or enable someone to point out what I'm doing wrong. |
| 09:43.28 | irroot | jkroon mine ecn.dnstelecom.co.za |
| 09:43.36 | jkroon | awesome. |
| 09:43.48 | jkroon | i'll have to figure out how to use res_stun though :p |
| 09:43.53 | *** join/#asterisk mintos (~mvaliyav@117.206.19.90) |
| 09:45.05 | irroot | hint it has a config file with one option |
| 09:45.24 | irroot | when you think mweb will catch a clue ?? |
| 09:46.10 | irroot | there marketing gurus want to meet with us to discuss my rants re mweb on twitter want to join :P |
| 09:46.13 | *** join/#asterisk godmachine-x6 (~godmachin@unaffiliated/godmachine-x6) |
| 09:53.21 | *** join/#asterisk p0r0h (~p0r0h@mail.mtel.su) |
| 09:53.29 | p0r0h | hi all |
| 09:53.52 | jkroon | irroot, sure, then we can meet as well. |
| 09:54.01 | jkroon | i need your services as well anyway. |
| 09:54.17 | jkroon | and the entire community might benifit from me throwing you some $$$ |
| 09:54.23 | p0r0h | What is the best monitoring system for Asterisk |
| 09:54.25 | p0r0h | ? |
| 09:54.30 | irroot | no prob we can do something i lean toward a braai |
| 09:54.38 | *** join/#asterisk Dovid (~Dovid@office.mypbxmanager.net) |
| 09:54.42 | catphish | i assume nagios has plugins |
| 09:54.47 | nunne | i have a problem with asterisk 1.4.39.2.. i get chan_sip.c:1896 __sip_xmit: sip_xmit of 0x1daea28 (len 814) to 192.168.5.125:5060 returned -1: Operation not permitted when trying to send a call to 14-15+ SIP devices.. and sometimes asterisk crashes when doing this. |
| 09:54.54 | jkroon | irroot, always sounds good. |
| 09:55.13 | catphish | nunne: maybe a ulimit |
| 09:55.52 | p0r0h | difficult to set up nagios to work with Asterisk? |
| 09:56.34 | Dovid | p0r0h: I tried and failed. i failed with Nagios in general |
| 09:56.46 | irroot | jkroon you see the bug i buzzed you with |
| 09:56.54 | jkroon | no i didn't. |
| 09:56.55 | catphish | nagios seeme unnecessarily complicated at first |
| 09:57.08 | catphish | but its pretty powerful once you get used to it |
| 09:57.20 | catphish | still a PITA to maintain though |
| 09:57.26 | Dovid | catphish: Hard to learn. there are docs but for people like me.... |
| 09:57.31 | Dovid | how about Cacti ? |
| 09:57.52 | *** join/#asterisk cerberus_za (~coert@41-134-110-162.dsl.mweb.co.za) |
| 09:57.59 | catphish | cacti is a different kind of monitoring really |
| 09:58.09 | catphish | depends what you want to achieve |
| 09:58.47 | jkroon | is looking for a solution to try and determine if SIP and/or IAX/2 has locked up (ie, can calls be made out via an asterisk system yes or no) |
| 09:59.05 | jkroon | simple script is good enough, can cron it to run every minute. |
| 09:59.06 | Dovid | catphish: Meaning ? |
| 09:59.15 | jkroon | cacti monitors snmp. |
| 09:59.33 | catphish | i'd use nagios |
| 09:59.36 | Dovid | jkroon: You want to see if you can make a call out? If the peer is available? is the remote side your systems or some one else ? |
| 09:59.57 | jkroon | Dovid, i want a way to ensure that I know before my clients do when an asterisk system goes down. |
| 10:00.03 | Dovid | jkroon: So cacti would be good for say Routers/Switches while Cacti would be better for Asterisk? |
| 10:00.15 | Dovid | i want to see Bandwdith graphs/ call amounts/ cpu usage etc. |
| 10:00.22 | jkroon | doing a pidof asterisk is not good enough, even doing asterisk -rx "core show uptime" isn't good enough. |
| 10:00.31 | jkroon | Dovid, those I've already got. |
| 10:00.34 | catphish | Dovid: err, is this 2 separate questions?? |
| 10:00.45 | jkroon | catphish, a sidetrack notion :) |
| 10:00.55 | p0r0h | For example, if there was a problem in the system may indicate the status of "Work" and stop an alert about the problem at a time |
| 10:01.27 | Dovid | jkroon: So what are you looking for? |
| 10:01.32 | catphish | i'd use nagios to monitor for errors |
| 10:01.39 | irroot | jkroon a nagios script that does sip options will be usefull |
| 10:01.49 | catphish | then i'd use rrdtool to log call volume etc |
| 10:01.56 | catphish | with some kind of wrapper |
| 10:01.56 | Dovid | i want to make sure that everything is in order. also to see graphs of usage over time |
| 10:02.01 | irroot | started doing one |
| 10:02.15 | catphish | if you want a system that does both, custom to asterisk, someone will need to write one |
| 10:02.23 | catphish | wouldn't really be hard |
| 10:02.36 | irroot | i use a script + rrdtool in php to log calls via sql query to CDR |
| 10:02.39 | catphish | making a sip call is easy, as is piping the call volume to rrdtool |
| 10:03.36 | p0r0h | I want to have a monitoring system that processed data from the team that gets the "sip show peers" |
| 10:03.49 | nunne | catphish: i was thinking something like that.. buut it's an embedded platform running uClinux.. and i dont have the ulimit command.. nor a ulimit config file anywhere :( does anyone know where to set it? against /proc somewhere maybe? |
| 10:04.06 | jkroon | Dovid, the usage over time i've got working on cpu, bandwidth etc .. (no call volumes yet, but relatively trivial to add) |
| 10:04.11 | catphish | nunne: not sure, sorry |
| 10:04.39 | jkroon | i'm just interested in knowing if/when things break. notifications to my phone for critical infrastructure, email for client-perimiter equipment. |
| 10:04.55 | catphish | jkroon: probably want nagios |
| 10:08.14 | *** join/#asterisk Faustov (~fst@gentoo/user/faustov) |
| 10:10.20 | *** join/#asterisk schmidts (~schmidts@Lo20-rt01.lm33.sil.at) |
| 10:10.49 | schmidts | Hello |
| 10:13.05 | p0r0h | bloody asterisk, he fell again))) |
| 10:13.50 | jkroon | p0r0h, version? |
| 10:15.46 | *** join/#asterisk BuenGenio (~Gene@4.Red-83-44-75.dynamicIP.rima-tde.net) |
| 10:15.50 | p0r0h | how to see the version? |
| 10:16.17 | jkroon | core show version |
| 10:17.06 | p0r0h | 1.8.4.2 |
| 10:19.00 | jkroon | uprade to 1.8.5.0 at least, i recommend 1.8.6.0, i'll quickly check what stability patches is currently in our gentoo build so that you can pull them too. |
| 10:19.07 | nunne | does anyone know if changing /proc/sys/fs/file-max will change my ulimit "on the fly".. or am i needed to reboot etc? |
| 10:19.28 | jkroon | nunne, at a max you might need to restart the asterisk daemon. |
| 10:20.14 | nunne | jkroon: thanks :) |
| 10:21.06 | p0r0h | He falls out of the ugly hardware |
| 10:25.06 | jkroon | p0r0h, the patch set we're using at the moment is available at ftp://ftp.is.co.za/mirror/ftp.gentoo.org/distfiles/gentoo-asterisk-patchset-1.2.tar.bz2 |
| 10:25.19 | jkroon | you're going to have to go through them manually, their names are mostly explanatory. |
| 10:25.56 | jkroon | all of those afaik has been submitted to issues.asterisk.org, some has even been merged into trunk and will be in 1.8.7.0, some has not. |
| 10:27.11 | eject_ck | Hi all |
| 10:28.36 | eject_ck | I'm trying to make call from my sip phone via asterisk to my new sip provider. SIP PHONE -> ASTERISK -> VoIP PROVIDER. I'm getting message: 2011-09-22 13:23:01] NOTICE[930]: chan_sip.c:21253 handle_request_invite: Sending fake auth rejection for device <1xxxxxxxx><sip:10000000@212.58.xx.xxx> |
| 10:28.48 | eject_ck | what does it mean ? |
| 10:30.57 | *** join/#asterisk m_tadeu (~quassel@segredosdavida.com) |
| 10:40.11 | catphish | does anyone know what variables are set after a 302 redirect? i was under the impression that CALLERID(rdnis) was set, but this doesn't seem to be the case any more |
| 10:50.12 | *** join/#asterisk blackcat73 (~Blackcat@adonis.iportalmais.pt) |
| 10:50.18 | catphish | i fixed it my setting my own variable before running the dial(sip/ |
| 10:50.36 | blackcat73 | Hi, need some help to use sipp-tester to stress asterisk |
| 10:50.49 | blackcat73 | need 500+ registered phones |
| 10:51.14 | blackcat73 | how can I config sipp to register all this without starting 500+ instances? |
| 10:51.26 | catphish | i didn't know sipp did registrations |
| 10:51.35 | catphish | i use it to test call volume |
| 10:51.51 | blackcat73 | I believe it can register phones also |
| 10:52.02 | blackcat73 | and then you can feed it to generate calls |
| 10:52.09 | blackcat73 | but I might be mistaken |
| 10:53.29 | kaldemar | eject_ck: it means that you have alwaysauthreject=yes (default value was changed from no to yes in 1.8.0). see sample config for an explanation. |
| 10:54.51 | catphish | i use: ./sipp -sn uac [asterisk ip] -s testtone -d 10000 -r 3 -l 4096 -i [local ip] -rtp_echo |
| 10:55.08 | catphish | to generate 10 second calls with rtp data |
| 10:55.15 | catphish | but i don't know about registration |
| 10:55.42 | catphish | since i use mysql realtime, registrations don't really cost me anything |
| 10:55.49 | *** join/#asterisk grEvenX (~even@apb9hb.ip.ssc.net) |
| 10:56.04 | catphish | except the initial query to register them |
| 10:56.40 | *** join/#asterisk BuenGenio (~Gene@4.Red-83-44-75.dynamicIP.rima-tde.net) |
| 10:58.12 | blackcat73 | catphish, ok, but I believe there's a way to register N phones from a file and them I can generate calls from a csv file |
| 10:58.57 | catphish | that makes a lot of sense |
| 10:59.04 | catphish | afraid my usage has never been that advanced |
| 11:03.10 | blackcat73 | catphish, that's my problem also |
| 11:03.11 | blackcat73 | :) |
| 11:03.16 | *** part/#asterisk p0r0h (~p0r0h@mail.mtel.su) |
| 11:06.58 | cstachris | cacti = pretty graphs but no event based alerts |
| 11:07.42 | cstachris | oops, i was scrolled up |
| 11:08.48 | cstachris | blackcat73, sipp with registrations - i did 20 with this script http://wiki.opencsta.org/index.php/SIPp_3.1_Registering_on_Asterisk_1.6.1_beta |
| 11:10.22 | blackcat73 | cstachris, thx |
| 11:10.29 | blackcat73 | gonnja give it a try |
| 11:10.59 | cstachris | np |
| 11:11.56 | eject_ck | kaldemar: reason was that I was calling myself :) |
| 11:13.09 | kaldemar | eject_ck: and the call didn't match any defined device. |
| 11:14.28 | SteveWilliams | Hi all, is there an option within asterisk that Dials a number and when the phone is picked up, it starts playing a message stored in a .wav file. Please help me with an example. |
| 11:15.09 | cstachris | SteveWilliams, who does it play the wav file to - the caller or the callee |
| 11:15.12 | SteveWilliams | Can I implement that in my dialplan |
| 11:15.29 | blackcat73 | cstachris, this only does the registration part, right? |
| 11:15.29 | SteveWilliams | the callee |
| 11:15.45 | blackcat73 | cstachris, we can then feed more .csv files to generate calls, right? |
| 11:15.54 | cstachris | blackcat73, yes - i didn't read your whole requirement |
| 11:16.11 | cstachris | blackcat73, yeah you can script call stuff too - just need that pcap file |
| 11:16.25 | SteveWilliams | @ cstachris the callee |
| 11:16.27 | kaldemar | SteveWilliams: see option A() for app dial. |
| 11:16.37 | blackcat73 | cstachris, I have to read more about sipp then |
| 11:16.38 | SteveWilliams | ok, thanks |
| 11:17.12 | cstachris | blackcat73, use some of what catphish put above as well - in half a day you'll have a kickass load tester :) |
| 11:17.42 | cstachris | blackcat73, there is a program called "sip inspector" for doing similar stuff - it's on code.google.com - just google it |
| 11:18.54 | cstachris | caller and callee should be calling and called party for next time - it's not a cheque! |
| 11:19.24 | blackcat73 | cstachris, thx so much |
| 11:19.36 | cstachris | np |
| 11:24.25 | eject_ck | kaldemar: yes |
| 11:24.30 | eject_ck | thank you very much |
| 11:25.03 | eject_ck | Can I ask personal recommendations for best office SIP phone :) ~ 80$ |
| 11:25.37 | eject_ck | I have number of dlink dph-150s |
| 11:26.14 | eject_ck | voice quality is good but usability and comfort is not |
| 11:28.00 | cstachris | eject_ck, polycom 320 - no network port in the back for your PC though |
| 11:34.36 | kaldemar | 320 is a discontinued model replaced by 321. 321 has more internal memory. |
| 11:35.10 | eject_ck | kaldemar: let me see,m how much it cost? |
| 11:35.52 | *** join/#asterisk godmachine-x6 (~godmachin@unaffiliated/godmachine-x6) |
| 11:42.40 | *** join/#asterisk rgsteele (~rgsteele@apo.aweber.com) |
| 11:48.04 | atan | I have a device that shows as registered within sip show peer, can call out, but cannot rx a call. I switched devices out from an ATA to a polycom phone and the phone as the same issue with not ringing. It doesn't even see the inbound call. Did something change in * 10 with nat= and qualify= in sip.conf? |
| 11:50.32 | kaldemar | atan: is the phone behind a NAT? what address does sip show peers show for it? |
| 11:51.13 | kaldemar | atan: what does Status column say? |
| 11:53.39 | atan | kaldemar, it is behind a dlink router on a cable modem |
| 11:53.42 | atan | Status says OK (xxxms) |
| 11:53.53 | atan | I have qualify=yes nat=yes |
| 11:55.31 | NourSs | Hi, i sell two Digium cards ( TE121P and TDM411B ), private me for more details :-) |
| 11:57.21 | *** join/#asterisk Naikrovek (~mbluth@unaffiliated/naikrovek) |
| 11:58.33 | *** join/#asterisk Hanumaan (~Hanumaan@dslb-092-074-250-063.pools.arcor-ip.net) |
| 12:04.06 | *** join/#asterisk wesphillips (~wphill04@137-237-233-124.harris.com) |
| 12:04.19 | leifmadsen | NourSs: this is not #asterisk-biz |
| 12:04.50 | kaldemar | atan: what do you see in sip debug when making a call to the device? |
| 12:06.02 | *** part/#asterisk asterisk-Tester (~RAMYT@210.5.215.40) |
| 12:07.57 | atan | I didn't place a call just yet but right now I see this, http://pastebin.com/zwQgG5yY |
| 12:16.40 | *** join/#asterisk AviMarcus (~avi@bzq-79-180-184-200.red.bezeqint.net) |
| 12:16.47 | AviMarcus | What does " retail traffic only (above 45 seg) " mean? or is that a typo? |
| 12:22.56 | *** join/#asterisk gmaruzz (~gmaruzz@2-225-249-20.ip178.fastwebnet.it) |
| 12:23.04 | *** part/#asterisk gmaruzz (~gmaruzz@2-225-249-20.ip178.fastwebnet.it) |
| 12:31.56 | *** join/#asterisk Buklov (~Buklov@mail.sapsun.su) |
| 12:34.53 | leifmadsen | AviMarcus: huh? |
| 12:39.14 | AviMarcus | on the asterisk-biz mailing list, leifmadsen |
| 12:40.04 | *** join/#asterisk bchia (~Adium@nat/digium/x-otjfleethrkjcdxh) |
| 12:41.01 | AviMarcus | <PROTECTED> |
| 12:41.17 | leifmadsen | AviMarcus: could be a typo -- not everyone uses spell check :) |
| 12:42.29 | *** join/#asterisk asterisk-Tester (~RAMYT@210.5.215.40) |
| 12:43.55 | *** join/#asterisk WiretapSeven (~Wiretap@unaffiliated/wiretap) |
| 12:47.06 | *** join/#asterisk TimeRider (~steve@188-220-33-123.zone11.bethere.co.uk) |
| 12:49.11 | atan | If you see Retransmitting #6 (NAT) to 71.7.x.x:60050: in your SIP debug when you attempt to call an extension what could cause it? |
| 12:49.32 | atan | I can only assume this means it is on try #6 to contact the device but not getting an answer |
| 12:49.41 | WIMPy | No answer. |
| 12:49.51 | WIMPy | indeed |
| 12:50.06 | atan | WIMPy, I have nat=yes in my sip.conf and qualify=yes in there also. Should I replace the router? |
| 12:50.17 | kaldemar | atan: it means you tried to contact 1-6 times without an answer. do you feel like answering the previous questions? |
| 12:50.35 | atan | kaldemar: I'm sorry I went right past that |
| 12:50.41 | WIMPy | If the device exists and is working, that's at networking issue, yes. |
| 12:50.52 | atan | kaldemar: let me get something together for you, but sorry about skipping right past that |
| 12:51.12 | atan | kaldemar, wait I did pastebin something for you http://pastebin.com/zwQgG5yY |
| 12:51.25 | atan | Is that not what you were seeking? |
| 12:51.30 | *** join/#asterisk WiretapSeven (~Wiretap@unaffiliated/wiretap) |
| 12:51.34 | leifmadsen | if asterisk keeps retransmitting, it's because the other end isn't responding |
| 12:51.42 | leifmadsen | (or asterisk isn't getting the response) |
| 12:51.48 | *** join/#asterisk irroot (~irroot@pbx.distrotech.co.za) |
| 12:51.59 | leifmadsen | so as WIMPy said, it's a networking issue |
| 12:52.03 | WIMPy | Or the other end didn't receive the request. |
| 12:52.23 | atan | Okay... time to grab a new router for these people then. Anyone have a suggestion as to which cheaper brand router could work for this? :-) |
| 12:52.51 | atan | Doesn't need to be wireless. I'm thinking like a cheap little linkys or something. |
| 12:53.00 | kaldemar | atan: no. |
| 12:53.43 | WIMPy | If the router has any features to do with SIP, disable them. |
| 12:55.24 | atan | The router that is there now which isn't letting these requests through is a dlink wireless g thinger. Older style... black with silder around the sides, and I think one black antenna on the back side. WIMPy, I don't think it had any features within it to do with SIP. |
| 12:57.05 | *** join/#asterisk WiretapSeven (~Wiretap@unaffiliated/wiretap) |
| 12:57.42 | atan | Could I turn off the NAT setting and forward ports to the device? |
| 12:58.22 | WIMPy | Is it only a single device behind that router? |
| 12:58.28 | atan | WIMPy, yep! :-) |
| 12:58.54 | WIMPy | Sp no need for a router. |
| 12:59.18 | atan | Well, there is a computer and a wireless client on there also... |
| 12:59.33 | atan | Sorry, I was thinking phone. Only one phone. |
| 13:00.15 | WIMPy | Son not just a single device. |
| 13:00.29 | atan | No, sorry. I was thinking device == phone, but my bad ;) |
| 13:00.32 | WIMPy | s/n/,/ |
| 13:00.38 | atan | One phone. More than one computer. |
| 13:01.13 | irroot | <PROTECTED> |
| 13:01.25 | WIMPy | You could forward the SIP port. Maybe you'd need to forward RTP ports as well. But you need to find out which ports the phone uses. |
| 13:01.57 | irroot | WIMPy maybe set the phone to only use 10 ports and forward only those |
| 13:02.13 | WIMPy | If the phone can do that. |
| 13:02.16 | atan | irroot, it's a cable connection |
| 13:02.34 | atan | irroot, as far as I know there is no PPPoE going on. |
| 13:03.01 | atan | How could I go about finding out which ports a phone uses for this? :-) |
| 13:03.22 | WIMPy | The phones manual or it's configuration. |
| 13:03.26 | atan | I see Retransmitting #6 (NAT) to 71.7.157.102:60050:, does this mean it uses 60000- something? |
| 13:04.33 | WIMPy | Yes, but as SIP port. |
| 13:05.21 | *** join/#asterisk serafie (~erin@nat/digium/x-alicfoyfgsqupsqf) |
| 13:05.31 | WIMPy | And that's the router, not the phone. |
| 13:06.03 | *** join/#asterisk LittleFool (~LittleFoo@95.129.212.120) |
| 13:06.16 | atan | Well color me confused then. So the phone always uses port 5060 then? It's the router which uses those funky high numbered ports? |
| 13:06.52 | *** part/#asterisk mirela666 (~mirko_bra@212.200.146.253) |
| 13:06.59 | WIMPy | It might be the phone as well. |
| 13:07.02 | wdoekes2 | atan: what does the Via line in the request from your phone say? |
| 13:07.04 | WIMPy | Some use random ports. |
| 13:07.10 | irroot | atan phone uses sip [5060] and rtp [assigned from a pool of ports or random] |
| 13:07.19 | atan | Via: SIP/2.0/UDP 66.228.34.248:5060;branch=z9hG4bK56b0397e;rport |
| 13:07.51 | irroot | atan need to look at the sdp attachment at bottom of invite |
| 13:07.53 | wdoekes2 | that IP is a bit odd if this phone is behind nat |
| 13:09.30 | WIMPy | Yes |
| 13:09.45 | wdoekes2 | if your phone has special-nat-magic capabilities disabled, it should list its real IP (rfc1918) in there |
| 13:09.47 | WIMPy | Do you have NAT support enabled on the phone? Switch that off. |
| 13:10.51 | irroot | stun / ice settings |
| 13:11.16 | atan | I will need to look in the settings on the device but I don't recall seeing such an option on the IP300 menus |
| 13:12.22 | *** part/#asterisk AviMarcus (~avi@bzq-79-180-184-200.red.bezeqint.net) |
| 13:12.26 | atan | Well I'll be there is a whole section with that stuff. Oh boy. |
| 13:17.39 | atan | Okay I am going to go take a better look at what settings are enabled on there and will come back if I run into trouble :-) Thank you guys again for pointing me in the right direction ;) |
| 13:20.36 | *** join/#asterisk squig (~bendeluca@soho-94-143-249-50.sohonet.co.uk) |
| 13:22.41 | *** join/#asterisk WiretapSeven (~Wiretap@unaffiliated/wiretap) |
| 13:22.55 | *** part/#asterisk sekil (~sekil@78.24.104.73) |
| 13:24.36 | Katty | i had an awful dream last night :< dreamt i got bit by a snake, and had to go to ER for 3 shots. |
| 13:26.37 | *** join/#asterisk SteveWilliams (~SteveWill@220.224.235.78) |
| 13:26.56 | *** part/#asterisk SteveWilliams (~SteveWill@220.224.235.78) |
| 13:27.46 | squig | what kind of shots? vodka or tequila ? |
| 13:30.29 | *** join/#asterisk dym (~patrick@netsplit.me) |
| 13:31.31 | leifmadsen | squig: yes |
| 13:31.38 | Katty | the kind with a big needle |
| 13:31.54 | irroot | caffine that way is my dream |
| 13:33.45 | *** join/#asterisk urishk (~chatzilla@bzq-218-189-229.red.bezeqint.net) |
| 13:34.56 | urishk | hi.... help with asterisk/freepbx & Dahdi is needed.... anyone? |
| 13:35.07 | sunfone | Bueler? |
| 13:35.36 | WIMPy | ~ask |
| 13:35.36 | infobot | Questions in the channel should be specific, informative, complete, concise, and on-topic. Don't ask if you can ask a question first. Don't ask if a person is there; just ask what you intended to ask them. Better questions more frequently yield better answers. We are all here voluntarily or against our will. |
| 13:35.43 | WIMPy | LOL |
| 13:35.48 | urishk | :-) |
| 13:35.48 | sunfone | :) |
| 13:35.52 | Katty | hugs irroot |
| 13:36.32 | irroot | {{{{}}} katty thx |
| 13:36.52 | *** join/#asterisk FiReSTaRT (~dlyh@unaffiliated/firestart) |
| 13:38.11 | urishk | freepbx / asterisk 1.6.2.12, single trunk (sip), vast majority of extension are SIPs. system works fine for the last two years. Single FXS card (Single PCI/e card, 4 FXS ports - Wildcard TDM400P REV E/F Board 5 , DAHDI Version: 2.4.0 Echo Canceller: MG2) |
| 13:40.05 | urishk | When (outging) dialing from FXS which is a new fax machine, I get busy tone (both internal and external dialing). Incoming calls to these FXS extension are OK. Door keypad works ok (same PCI/e TDM400P) |
| 13:40.20 | irroot | ~freepbx |
| 13:40.20 | infobot | [~freepbx] FreePBX is unable to be supported here. It is made up of complex dialplans and scripts which can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking there, or fpbx |
| 13:40.53 | urishk | 10x I'll do that |
| 13:41.24 | irroot | urishk try turning off faxdetect on the fxs |
| 13:41.53 | *** join/#asterisk _Corey_ (~chatzilla@64.121.4.75) |
| 13:45.01 | Faustov | fkn quotes do not match! |
| 13:45.13 | Katty | hi Faustov |
| 13:45.22 | Faustov | hi Katty |
| 13:46.10 | Katty | hugs Faustov |
| 13:46.12 | Katty | how're you dear |
| 13:46.42 | Faustov | hmm... I guess I'm high on tanine? |
| 13:46.47 | Faustov | was that the thing in tea? |
| 13:47.54 | Faustov | add that to being usually pedantic - makes me react to braces not matching as irroot wrote, and even name them quotes for some reason |
| 13:47.57 | Faustov | thanks for asking, you? ;) |
| 13:48.41 | *** join/#asterisk file (~file@asterisk/developer-and-muffin-lover/file) |
| 13:48.41 | *** mode/#asterisk [+o file] by ChanServ |
| 13:50.21 | Katty | what's tanine supposed to do? |
| 13:50.51 | Katty | i'm doing ok, not the best of mornings...had a bad evening last night. |
| 13:50.57 | leifmadsen | I think I'm having a brain malfunction (more so than usual!) -- what is the function/application/setting for setting ring indications on a channel? |
| 13:51.01 | Katty | but the day is young, and there is plenty of time for improvement |
| 13:51.11 | Katty | the function is hugs. |
| 13:51.15 | Katty | hugs leifmadsen |
| 13:51.36 | Katty | how're you dear, besides suffering from memory lapse |
| 13:51.49 | leifmadsen | Katty: I'm totally alive, so I'm going to count that :) |
| 13:52.12 | kaldemar | leifmadsen: PlayTones? |
| 13:52.32 | leifmadsen | kaldemar: probably not, as I want to set it for when calling another channel |
| 13:52.47 | leifmadsen | kaldemar: oh nevermind -- that might do it actually |
| 13:52.56 | leifmadsen | I assumed it worked differently ;) |
| 13:53.30 | singler | r option on Dial() does not suit? |
| 13:56.26 | irroot | leifmadsen progress ?? |
| 13:56.40 | *** join/#asterisk LittleFool (~LittleFoo@95.129.212.120) |
| 13:56.50 | irroot | Ringing |
| 13:56.59 | leifmadsen | singler: no, because that doesn't change the ringing indication |
| 13:57.15 | leifmadsen | I want different ringing indications in certain situations |
| 13:59.33 | leifmadsen | gonna play with PlayTones() and see if that does what I want |
| 13:59.37 | leifmadsen | kaldemar: thanks for the suggestions |
| 14:03.12 | kaldemar | leifmadsen: i assume that you alreary remembered also StopPlayTones. |
| 14:03.36 | leifmadsen | kaldemar: not yet :) |
| 14:03.51 | leifmadsen | this probably isn't going to do exactly what I want |
| 14:03.58 | irroot | ~thebook leifmadsen |
| 14:04.00 | leifmadsen | it might work as a stop gap solution |
| 14:04.07 | leifmadsen | irroot: don't start with me! |
| 14:06.23 | Katty | leifmadsen: horay for alive, that is always a lovely start to the morning |
| 14:06.29 | *** join/#asterisk master_of_master (~master_of@p57B54B69.dip.t-dialin.net) |
| 14:06.34 | leifmadsen | Katty: sometimes ;) |
| 14:07.03 | *** join/#asterisk urishk (~chatzilla@bzq-218-189-229.red.bezeqint.net) |
| 14:07.21 | Qwell | being alive in the morning would suck if you were a zombie or vampire or something |
| 14:07.58 | WIMPy | wonders if he qualifies as zombie. |
| 14:08.30 | Katty | WIMPy: i often do pre-caffeine |
| 14:08.34 | Katty | hugs Qwell |
| 14:12.47 | *** join/#asterisk aberrios (~aberrios@195.171.4.82) |
| 14:14.39 | catphish | are there any apps that dial an extension then ask the user if they want to take the call? |
| 14:14.46 | catphish | *dial a channe |
| 14:15.29 | Qwell | You don't need an app for that. It's simple dialplan. |
| 14:15.37 | *** join/#asterisk Fritz09 (~Adium@pop1-2551.catv.wtnet.de) |
| 14:15.42 | catphish | how would you do it? |
| 14:16.38 | catphish | ParkAndAnnounce? |
| 14:16.51 | _Corey_ | catphish: Look at Dial argument M |
| 14:17.10 | _Corey_ | You can deal with that using MACRO_RESULT |
| 14:17.12 | Kobaz | should use U, which uses GoSub |
| 14:17.26 | Kobaz | Macro is oldschool |
| 14:17.42 | _Corey_ | lol, I guess I'm oldschool :) |
| 14:18.24 | Katty | hugs Kobaz |
| 14:18.26 | Katty | hugs _Corey_ |
| 14:18.40 | Kobaz | howdy howdy |
| 14:18.49 | _Corey_ | good morning Katty |
| 14:20.23 | catphish | when using dial with U(), is it possible to connect the original channel with the dialed channel after the macro completes? |
| 14:20.46 | Katty | how goes? |
| 14:20.59 | Kobaz | catphish: dial does that for you |
| 14:21.23 | catphish | in which case? when no GOSUB_RESULT is specified |
| 14:21.28 | Kobaz | catphish: U/M are 'pickup handlers'. they run when the call gets picked up, and depending how they exit, you can either have the call accepted or hung up |
| 14:21.59 | catphish | is it possible to use U/M with multiple dialed channels? |
| 14:22.09 | catphish | and connect to the first that completed the macro |
| 14:22.16 | Kobaz | yeap |
| 14:22.21 | catphish | excellent |
| 14:22.23 | Kobaz | it runs on which ever channel picked up the call |
| 14:22.40 | catphish | i hoped it would run on all of them |
| 14:22.50 | WIMPy | Doesn't it release all other calls as soon as one is connected? |
| 14:22.59 | catphish | what i'm actually looking to do it call several people and ask them all to accept the call |
| 14:23.06 | Kobaz | oh |
| 14:23.09 | catphish | then connect the first person to press yes |
| 14:23.12 | Kobaz | you can't do that with M/U |
| 14:23.19 | WIMPy | You cold use a set of local channels, I guess. |
| 14:23.24 | Kobaz | you'll have to yeah... local channels |
| 14:23.43 | catphish | how would that work? |
| 14:23.48 | catphish | excuse my ignorance here |
| 14:24.34 | WIMPy | Define a bunch of extensions that dial one peer, each with taht macro, then dial all those extensions via local channels. |
| 14:24.47 | Kobaz | you would have to get the call in asterisk, and then do an originate with a parameter of the calling channel |
| 14:24.56 | Kobaz | and then who ever picks up the call. Bridge() to the calling channel |
| 14:25.48 | Kobaz | WIMPy: you just need one exten, you can do a _X. to match everything |
| 14:26.12 | WIMPy | Right. You could use the extension as parameter. |
| 14:28.09 | catphish | hmm |
| 14:28.19 | catphish | going to have to get my head around that |
| 14:29.38 | leifmadsen | darn, PlayTones() doesn't do what I want -- but the r([tone]) indication does (I'm running Asterisk 10) -- looks like I may have to backport that feature to 1.8 |
| 14:30.07 | *** join/#asterisk The_Boy_Wonder (~vossel@asterisk/batman-developer/dvossel) |
| 14:30.14 | catphish | WIMPy: Kobaz: do either of you have time to actually walk me through some dialplan for that? |
| 14:30.50 | Kobaz | if you read up on using Originate() and local channels, you can figure it out |
| 14:30.58 | Kobaz | otherwise I would wind up writing it for you |
| 14:31.02 | WIMPy | catphish: It really isn't more that what I wrote and Kobaz corrected. |
| 14:31.06 | Kobaz | and I don't really have the time |
| 14:31.22 | kaldemar | leifmadsen: you don't, it is in 1.8. |
| 14:31.27 | WIMPy | Is it all sip peers or numbers to the same acount? |
| 14:32.44 | catphish | i want to dial several numbers at varying sip peers |
| 14:33.36 | catphish | the purpose here is that if an emergency call comes in our of hours, we need to dial everyone's cell, and route the call to the first person who answers, with the caveat that answering services shouldn't count as an answer |
| 14:33.43 | catphish | *out of |
| 14:33.43 | *** part/#asterisk dym (~patrick@netsplit.me) |
| 14:33.48 | leifmadsen | kaldemar: oh you're right -- I was looking at the xml docs in app_dial.c and I didn't read far enough :) |
| 14:33.51 | WIMPy | Ok, so you can to an exten => _.,1,Dial(SIP/${EXTEN},,U(whatever)) in an extra context. |
| 14:33.52 | leifmadsen | excellent! |
| 14:34.27 | *** join/#asterisk corretico (~luis@200.12.40.18) |
| 14:34.59 | WIMPy | Then you can use Dial(local/peer@extracontext1&local/peer2~localcontext&local/itsp2/numerb@extracontext...) |
| 14:35.04 | *** join/#asterisk corretico (~luis@200.12.40.18) |
| 14:35.22 | catphish | oh yeah, of course |
| 14:35.26 | *** join/#asterisk bintut (~bintut@cm7.sigma15.maxonline.com.sg) |
| 14:35.34 | catphish | that's simple enough |
| 14:35.46 | catphish | sorry for not getting that sooner |
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| 14:40.57 | *** join/#asterisk kdmessano (~nonya@unaffiliated/kdmessano) |
| 14:44.40 | Qwell | kdmessano: Did you lose your M.D.? |
| 14:45.26 | *** join/#asterisk aidinb (~aidin@unaffiliated/aidinb) |
| 14:47.57 | *** part/#asterisk asterisk-Tester (~RAMYT@210.5.215.40) |
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| 15:12.27 | Katty | 3 rows of knitting and my hands hurt. |
| 15:12.30 | Katty | seriously? |
| 15:12.35 | Katty | this yarn is trolling me. |
| 15:13.25 | irroot | Katty know a knitting club here "bitches who stich" |
| 15:13.27 | p3nguin | In Soviet Russia, yarn... wait, you're in Soviet Russia? |
| 15:15.10 | *** join/#asterisk kdmessano (~nonya@unaffiliated/kdmessano) |
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| 15:15.48 | Katty | irroot: yeah but i'm not a bitch. |
| 15:15.52 | Katty | ..most of the time. |
| 15:15.54 | p3nguin | Bitchin' and Stitchin' |
| 15:16.03 | Katty | now that i might fit into |
| 15:16.14 | irroot | Katty lol |
| 15:16.15 | Katty | maybe knit-wit is more me tho |
| 15:17.45 | Katty | wonders if thats danny |
| 15:18.21 | p3nguin | I figured it wasn't, but I never bothered to ask. |
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| 15:22.41 | *** join/#asterisk devil_evoxxx (~d3v1l@87.13.67.31) |
| 15:22.47 | devil_evoxxx | hi all :) |
| 15:22.52 | Katty | hi devil_evoxxx |
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| 15:24.23 | devil_evoxxx | i've got a 1.4.36 ast box wich have some "difficult" in t.38 passtrought |
| 15:25.11 | devil_evoxxx | i think is because is not compiled with decommenting |
| 15:25.17 | devil_evoxxx | t.38 directive.. |
| 15:25.18 | devil_evoxxx | :( |
| 15:25.19 | p3nguin | As far as I know, 1.4 does not do t.38 pass-through. |
| 15:25.28 | leifmadsen | that'd do it :) |
| 15:25.45 | devil_evoxxx | 1.4 have pass-trough |
| 15:25.46 | devil_evoxxx | righ? |
| 15:26.39 | leifmadsen | p3nguin is saying it doesn't have pass-through for T.38 in 1.4 |
| 15:26.45 | leifmadsen | hasn't used 1.4 in... years really |
| 15:27.00 | devil_evoxxx | is there a way to check if its compile with ? |
| 15:27.27 | p3nguin | As far as I know, 1.4 does not even support the option of t.38 pass-through. |
| 15:28.02 | p3nguin | But I've never tried to obtain or activate it, so at this point I'm only giving my opinion. |
| 15:28.06 | *** join/#asterisk Fritz09 (~Adium@2002:542e:5a02:e472::3) |
| 15:29.14 | irroot | p3nguin devil_evoxxx it has pass through its ugly but there 1.6 had T38 support iorned out |
| 15:30.03 | p3nguin | Grr. Where do people get the "d" in "fridge" when they write it? |
| 15:30.12 | p3nguin | There's no d in refrigerator. |
| 15:31.36 | irroot | went to attic t38pt_udptl is in 1.4 |
| 15:32.03 | irroot | but cannot be used with agent/local must be SIP<->SIP |
| 15:32.05 | p3nguin | Does it require a patch, or is it something that needs to be enabled? |
| 15:33.14 | irroot | p3nguin to fax from S/R fax is impossible |
| 15:33.25 | irroot | its pure pass through |
| 15:33.35 | irroot | 1.6 added endpoint support |
| 15:33.39 | p3nguin | He was looking for pass-through support. |
| 15:35.47 | devil_evoxxx | hi irroot !! it's all ok |
| 15:36.02 | devil_evoxxx | ? |
| 15:36.15 | irroot | devil_evoxxx you got the problem solved on the quescom ?? and its working for you |
| 15:36.39 | devil_evoxxx | yes, i've upgraded from 5.00 to 5.22 (still on windows system) but now, it works!! |
| 15:36.56 | devil_evoxxx | there is a quescom firmware version 6.20 based on linux..in the few days i provide to upgrade again |
| 15:38.03 | devil_evoxxx | now faxes trought quescom work like a sharm! But i've got an old 1.4 boxes connected to sip providere ( sip2sip connection) |
| 15:38.40 | devil_evoxxx | that provide fax in t.38..but 1.4 does not support it.. |
| 15:38.53 | devil_evoxxx | and upgrading this box..at this time is not possible :( |
| 15:40.20 | p3nguin | irroot: How would he go about enabling t38pt? |
| 15:40.35 | p3nguin | I never knew 1.4 even recognized t38. |
| 15:40.49 | irroot | check the config file for t38 passthrough support |
| 15:41.04 | irroot | t38pt_udptl = yes |
| 15:41.19 | devil_evoxxx | i've found this http://www.voip-info.org/wiki/view/Asterisk+T.38 |
| 15:41.40 | devil_evoxxx | i think i've to recompile 1.4 with t38_version=1 and not 0.. |
| 15:43.32 | devil_evoxxx | irroot: setting t38pt_udptl=yes i got this on cli rtp.c:1377 ast_rtp_read: Unknown RTP codec 90 received from .. |
| 15:44.49 | *** join/#asterisk brad_mssw (~brad@shop.monetra.com) |
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| 16:01.42 | brad_mssw | any plans of providing binary packages for libpri-1.4.12 for ubuntu lucid/10.04 amd64? looks like 1.4.11.2 is currently what asterisk is distributing : http://packages.asterisk.org/deb/pool/main/libp/libpri/ |
| 16:02.28 | luke-jr | are the people handling Asterisk bugs volunteers? |
| 16:02.40 | p3nguin | Sometimes. |
| 16:02.43 | Qwell | luke-jr: Asterisk doesn't pay anyone. |
| 16:02.50 | Qwell | Even Digium employees should be considered volunteers. |
| 16:03.15 | Faustov | i hope you do pay for allthe redbull you mention in articles |
| 16:03.16 | p3nguin | Digium doesn't pay employees for work on Asterisk? |
| 16:03.28 | Qwell | p3nguin: Asterisk != Digium :) |
| 16:04.23 | p3nguin | brad_mssw: If you have the source available, it shouldn't be too hard to build your own package and distribute it to all of your systems which require it. |
| 16:05.02 | luke-jr | Qwell: Asterisk is a Digium product |
| 16:05.10 | Qwell | no, no, no, no, no |
| 16:05.10 | p3nguin | project |
| 16:05.23 | brad_mssw | p3nguin: that's not a very helpful reply ... we use the asterisk-provided package repo explicitly so we don't have to build from source ourselves |
| 16:05.47 | p3nguin | I felt like it was a very helpful answer. |
| 16:05.48 | brad_mssw | p3nguin: kind of the reason for its existence, no? |
| 16:06.10 | luke-jr | Qwell: if it isn't, then Digium has no business asking for special treatment in licensing |
| 16:06.51 | *** join/#asterisk vinhdizzo (~vinh@pool-173-60-112-55.lsanca.fios.verizon.net) |
| 16:07.14 | Qwell | luke-jr: You are extremely confused on the relationship. |
| 16:07.43 | Qwell | Digium sponsors and maintains the Asterisk project. It's as simple as that. |
| 16:08.18 | luke-jr | then they shouldn't be getting special license treatment to produce proprietary versions |
| 16:09.04 | luke-jr | anyhow, I'm being asked to provide a test case for a bug where I pointed out exactly where in the code the bug is, and the nature of it, before it will get fixed.. |
| 16:09.13 | luke-jr | I'll get around to it eventually I guess |
| 16:10.01 | Qwell | rolls his eyes |
| 16:12.00 | *** join/#asterisk adyn (~adyn@pdpc/supporter/active/adyn) |
| 16:12.41 | p3nguin | I fail to see the problem. If a company produces a proprietary version of ANYTHING, the company can do whatever they choose with the product. |
| 16:13.44 | p3nguin | Including, but not limited to, sell it to you, refuse to sell it to you, or give it away to you. |
| 16:13.51 | *** join/#asterisk beek (~klinebl@pdpc/supporter/bronze/beek) |
| 16:14.26 | adyn | fully working virtual FreePBX server in 5 minutes... yay for templates! |
| 16:14.32 | catphish | asterisk is dual licence right? |
| 16:14.42 | adyn | oops wrong channel |
| 16:14.42 | Qwell | adyn: But you're stuck with that awful distro. |
| 16:14.43 | p3nguin | And if you want free support on a free product, you'd better be willing to do your part to get that support. |
| 16:15.00 | catphish | i assume digium own the base copyright and can therefore make it dual licence |
| 16:15.07 | SunTsu | p3nguin: luke-jr is asking what's so special about Digium that asterisk licensing allows them to sell a proprietary version of asterisk if it's not their product in the first place |
| 16:15.12 | Qwell | catphish: It's not about copyright |
| 16:15.28 | catphish | which harms nobody as long as the gpl aspects remain gpl compliant |
| 16:15.40 | p3nguin | If Digium owns all the rights to the project, they can do with it anything they wish to do. |
| 16:15.59 | catphish | correct, and since it's LGPL basically anyone else can do the same |
| 16:16.10 | catphish | but digium own the copyright so they are special |
| 16:16.18 | Qwell | catphish: again - it's not about copyright |
| 16:16.22 | catphish | how so? |
| 16:16.25 | p3nguin | As long as they don't call it Asterisk, they can. |
| 16:16.28 | Qwell | It's a licensing agreement. |
| 16:16.34 | catphish | with who? |
| 16:16.42 | luke-jr | p3nguin: I see it more of, I'm doing Digium a favour by not only reporting the bug, but finding exactly what the cause of the bug is. ;) |
| 16:16.43 | Qwell | With individual contributors. |
| 16:16.45 | catphish | you can't licence a gpl product, except from the copyright holder |
| 16:17.11 | Qwell | Individual contributors retain the copyright on their code. |
| 16:17.23 | catphish | i didn't know that |
| 16:17.31 | catphish | i assumed that had to hand it over for the dual licence to work |
| 16:17.39 | luke-jr | Digium *effectively* gets copyright |
| 16:17.51 | Qwell | nope. in fact, we're less restrictive than GNU in that regard |
| 16:18.02 | luke-jr | techncially speaking, the license Digium *insists on* is the *same* thing as "copyright assignment" under German law, at least |
| 16:18.05 | Qwell | You have to assign copyright to them to get patches into gcc and other GNU projects. |
| 16:18.18 | p3nguin | luke-jr: They appreciate your efforts. If, however, you don't want to do such work, you are not required to do it. |
| 16:18.22 | luke-jr | (Germany doesn't have *actual* "copyright assignment" as other jurisdictions) |
| 16:18.24 | catphish | but... can't anyone make a proprietary closed source app from an LGPL project? |
| 16:18.33 | Qwell | catphish: It isn't LGPL |
| 16:18.42 | catphish | oh yeah sorry |
| 16:18.44 | catphish | misread |
| 16:18.53 | luke-jr | Qwell: GCC and GNU projects *are* products of GNU |
| 16:19.07 | catphish | so how is it not based on digium's copyright ownership? |
| 16:19.17 | Qwell | catphish: how is what? |
| 16:19.34 | catphish | if it were purely gpl they couldn't distribute a closed proprietary copy could they? |
| 16:19.51 | Qwell | Individual contributors allow us to relicense it. |
| 16:20.03 | catphish | and that's my point |
| 16:20.13 | Qwell | copyright != license |
| 16:20.29 | catphish | it's all based on digium holding the copyright, or asking those who do to licence it |
| 16:20.38 | luke-jr | Qwell: individual contributors have no choice but to allow it |
| 16:20.49 | catphish | i assume patches are not accepted without turning over those rights |
| 16:20.51 | Qwell | catphish: when you include the latter half of your sentence, it is true, yes |
| 16:20.54 | Qwell | correct |
| 16:20.58 | luke-jr | at least under German law, there is *no* distinction between what Digium is doing and what GNU is doing |
| 16:21.03 | catphish | that makes perfect sense |
| 16:21.05 | luke-jr | anyhow, this isn't worth arguing over |
| 16:21.13 | luke-jr | I'll make a test case when I get time |
| 16:21.13 | Qwell | catphish: That doesn't mean you can't distribute them on your own though. |
| 16:21.14 | catphish | nobody's arguing are they? |
| 16:21.26 | catphish | i was just confirming my understanding |
| 16:21.36 | Qwell | and fwiw, http://gcc.gnu.org/contribute.html |
| 16:21.48 | catphish | personally i don't like contributors having to give up some rights to digium |
| 16:21.55 | catphish | but the project has good results |
| 16:21.58 | catphish | so i wouldn't argue |
| 16:22.02 | Qwell | they don't give up any rights whatsoever |
| 16:22.07 | catphish | if it were a problem it would have been forked, and nobody wants that |
| 16:22.12 | Qwell | They retain all rights to the code that they've written. |
| 16:22.54 | Qwell | Somebody that has contributed code to Asterisk can *also* license their patch (separately) under whatever they'd like. |
| 16:23.20 | catphish | so it must be licenced under gpl and under digium's licence |
| 16:23.27 | WIMPy | Or revoke their licence. That would be fun, I guess. |
| 16:23.30 | catphish | but no rights are given up to publish in other ways |
| 16:23.43 | Qwell | catphish: https://issues.asterisk.org/jira/secure/DigiumLicense.jspa |
| 16:24.09 | catphish | i've never fully understood how the GPL is compatible with dual licencing |
| 16:24.15 | catphish | but obviously it works |
| 16:24.18 | Qwell | catphish: it's a separate issue |
| 16:24.23 | *** join/#asterisk imox (~imox@p4FC5C751.dip0.t-ipconnect.de) |
| 16:24.41 | *** join/#asterisk ph8 (~ph8@unaffiliated/ph8) |
| 16:24.42 | SunTsu | catphish: as a copyright holder you are able to distribute your stuff under different licenses |
| 16:24.45 | malcolmd | WIMPy: irrevocable. that kind of fun wouldn't be good for asterisk. :D |
| 16:25.02 | Qwell | malcolmd: fortunately that's included in the license :p |
| 16:25.07 | malcolmd | indeed |
| 16:25.20 | WIMPy | malcolmd: It is alwyas revokable. At least under german law, possibly the whole EU. |
| 16:25.35 | WIMPy | Which would make the licence void, legally. |
| 16:25.38 | catphish | so the copyright holder of a work is allowed to sell it under both the gpl and a restrictive licence, but users who obtain it under the GPL are not? |
| 16:25.41 | luke-jr | WIMPy: srsly? |
| 16:25.46 | Qwell | catphish: correct. |
| 16:25.48 | SunTsu | catphish: you can give one person license a and anotherone license b - both need to adhere to the license they accepted, they can't switch to the other one on their own |
| 16:25.58 | Qwell | SunTsu: also correct |
| 16:26.06 | catphish | well that all seems sensible :) |
| 16:29.46 | Qwell | WIMPy: I can't see that being correct. It would be easy to force people to give you tons of money by revoking a license to use something. |
| 16:30.18 | Qwell | "Here you go. Have some free software to control your $100,000,000,000 hardware." "Oh, nevermind. Give us money." |
| 16:30.32 | *** join/#asterisk Assuero (~Assuero@187.114.181.21) |
| 16:31.27 | WIMPy | Usually you can revoke anything that doesn't include limits. |
| 16:31.49 | WIMPy | If you give a licence for a certain amount for a certain time, that's it. |
| 16:32.04 | WIMPy | But unlimited stuff is usually revokable. |
| 16:32.41 | malcolmd | is there a practical qualifier on that? "i give you a license for 1 billion years" vs. "i give you a license for an unlimited time period." |
| 16:33.15 | WIMPy | You need to discuss that with a lawyer. |
| 16:33.28 | malcolmd | "i give you a license for one billion years, after there you, and only you, not your descendants, nor heirs, nor estate holders might revoke it." ;) |
| 16:33.35 | WIMPy | Legal stuff isn't always logical. |
| 16:33.42 | malcolmd | are you a lawyer? |
| 16:33.51 | WIMPy | nope |
| 16:34.06 | WIMPy | Actually I think it usually isn't logical. |
| 16:40.36 | WIMPy | Maybe I should point out that we don't have such a thing as copyright. We have an authors right, which is inalienable. |
| 16:41.47 | WIMPy | What the media industry is about is utilization rights. |
| 16:49.07 | *** join/#asterisk Fritz09 (~Adium@pop1-2551.catv.wtnet.de) |
| 16:50.11 | JonathanRose | I would imagine that an unlimited free use license would only be revocable if it was stated in the original license that it was revocable. I'm no lawyer though. It just seems to me that once you've made an agreement, you can't undo it unless you specifically said you can undo it and under what (if any) terms. |
| 16:51.28 | mtbf | Hey, how can i enable failed login attempts to appear in the console? verb is 15 but they still don't appear. |
| 16:52.05 | JonathanRose | Just now in Asterisk 10, you can make failed SIP login attempts appear by adding 'security' to your console logs in logger.conf |
| 16:52.12 | JonathanRose | I think there are other ways too. |
| 16:52.27 | mtbf | Thanks. What about 1.8? |
| 16:52.44 | JonathanRose | I think they might display with 'notice', but I'm not 100% sure on that at the moment. |
| 16:52.52 | JonathanRose | pastebin your logger.conf and I'll compare. |
| 16:52.59 | mtbf | Ok. |
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| 16:54.55 | mtbf | http://pastie.org/2574767 here, it's rather default. |
| 16:55.43 | mtbf | I was just curious if threre's a way to start seeing them by calling some CLI command. |
| 16:55.47 | p3nguin | When you say failed logins, do you mean failed registration or failed to authenticate an invite? |
| 16:55.56 | mtbf | Failed registration attempts. |
| 16:56.24 | JonathanRose | Alright, yeah. You just need to add a console logging profile thing. |
| 16:56.45 | p3nguin | console => notice,warning,error |
| 16:56.45 | mtbf | In the logger.conf? |
| 16:56.45 | JonathanRose | console => notice |
| 16:56.47 | JonathanRose | add that |
| 16:56.48 | p3nguin | That hsould be enough. |
| 16:56.52 | JonathanRose | and you'll get failed logins |
| 16:57.00 | JonathanRose | warnings and errors are nice too. |
| 16:57.00 | p3nguin | Yes, logger.conf. |
| 16:57.01 | mtbf | Thanks guys . |
| 16:57.07 | JonathanRose | No problem. |
| 16:58.10 | p3nguin | Don't forget to run "logger reload" after you save the changes to logger.conf. |
| 16:58.36 | mtbf | Yup, i figured it out :) |
| 16:58.41 | JonathanRose | Just a small warning though, you will be getting a lot of unrelated messages with notice, even more with warnings/errors. The new logging level in 10 pertains strictly to security warnings like failed logins and such. |
| 16:58.51 | JonathanRose | Should be fine though. |
| 16:59.14 | JonathanRose | Besides, that feature hasn't even gone into a release version yet :P |
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| 17:00.29 | mtbf | Thanks for the hint, I'll keep it in mind, now I'm just using my local 1.8 for learning purposes ;) |
| 17:13.54 | azv4 | Any Panasonic Digital Hybrid system PROs out there?!?! |
| 17:17.15 | *** join/#asterisk irroot (~irroot@41.49.152.149) |
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| 17:21.53 | rampage73 | ok have an intermittent issue that i can not pin down , sometimes incoming calls are getting "the number you have dialed is not in service check the number and try again" it is like one out of 5 and it is on a production box. |
| 17:22.06 | rampage73 | i notice when the caller gets the above message that where it normally says "SIP/trunk-name/extension,60" that instead it says "SIP/trunk-name/,60" leaving out the extension but why? if I call from the same # 10 times in a row at least once i get the number i dialed is not in service. |
| 17:22.21 | rampage73 | the version of asterisk is "Asterisk 1.6.0.26-FONCORE-r78" |
| 17:23.15 | *** join/#asterisk jits (b7521711@gateway/web/freenode/ip.183.82.23.17) |
| 17:23.39 | jits | Hi .. looking for a one to many asterisk video calling solution. Can someone please help me.. thanks. |
| 17:25.37 | rampage73 | jits, sorry no experience with that here. |
| 17:26.12 | jits | rampage73: any idea where i can look for help on this ? |
| 17:27.00 | rampage73 | jits, sorry i was unclear there i mean I have no idea how to help you but you are in the right place, i just did not want you to feel ignored |
| 17:27.03 | navaismo | jits asterisk 10 beta support videoconference |
| 17:27.15 | navaismo | check it in the wiki |
| 17:28.11 | navaismo | rampage73 maybe the sip line is not registered in that time, when the incoming call arrives |
| 17:28.38 | p3nguin | That wouldn't really make sense. |
| 17:29.14 | navaismo | my sip lines when isnt registered the telco sayme that |
| 17:30.00 | *** join/#asterisk tick (tick@80.54.23.253) |
| 17:30.03 | p3nguin | But how would that affect the extension being called? |
| 17:30.04 | rampage73 | navaismo, i thought something like that also but not the case |
| 17:30.13 | navaismo | maybe in your country not happen like this but keep in mind that asterisk is used inmany other countries than EU |
| 17:30.20 | navaismo | ok |
| 17:30.53 | p3nguin | What does the Dial command look like? |
| 17:31.54 | pabelanger | https://wiki.asterisk.org/wiki/display/AST/ConfBridge+10 |
| 17:32.02 | pabelanger | jits: Better info in here ^ |
| 17:32.15 | rampage73 | p3nguin, exten => s,1,Dial(SIP/trunkname/${DID},60,r) |
| 17:32.38 | p3nguin | Where is the DID variable being set? |
| 17:32.45 | p3nguin | Where is the call coming FROM? |
| 17:32.47 | rampage73 | p3nguin, i put trunkname there it is not actually called that |
| 17:32.52 | p3nguin | That's fine. |
| 17:33.52 | rampage73 | p3nguin, call is from our provider broadvox I am not sure where the DID variable is set myself as i am not the sole person working on the phone system and the other person is unavailable to ask at the moment |
| 17:34.03 | jits | pabelanger: thanks .. looking into it.. |
| 17:34.47 | p3nguin | So a call is coming in from broadvox and then going back out another provider? |
| 17:35.49 | *** join/#asterisk fisted (~fisted@unaffiliated/fisted) |
| 17:35.50 | rampage73 | p3nguin, sorry not familiar with all the terms call comes from pstn -> broadvox -> main asterisk/trixbox -> secondary trixbox at our office |
| 17:38.12 | Katty | guess who got a free shirt from charlotte rousse!!! |
| 17:38.20 | Katty | not that you guys are shoppers |
| 17:38.27 | WIMPy | Who's that? |
| 17:38.33 | Katty | but hey, free stuff right? |
| 17:38.42 | irroot | katty who is charlotte rousse |
| 17:39.11 | Katty | irroot: it's a store in the mall. |
| 17:39.15 | Katty | irroot: lots of sexy women's stuff |
| 17:39.19 | Katty | irroot: like clubbing tops and what not |
| 17:39.24 | irroot | im now intrested .... |
| 17:39.28 | WIMPy | <AOL>send pix!</AOL> |
| 17:39.29 | jits | pabelanger: it appears to be transmitting only one way video. from the description. Is it so ? |
| 17:40.10 | Katty | the shirt might be on the website |
| 17:40.11 | Katty | checks |
| 17:40.41 | pabelanger | jits: depending on how you configure the bridge, yes. EG: the video broadcast can change to who ever is talking, or focus on 1 specific talker |
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| 17:41.45 | jits | pabelanger: humm.. that won't do. We would like the trainer to see all the other participants. Though the participants may see only the trainer |
| 17:41.57 | jkroon | hi guys, does anybody know of a way to execute an external command (ala System()) and grab the output in your dialplan? |
| 17:42.11 | Katty | nope not on the website |
| 17:43.08 | irroot | jkroon agi not good for you ? |
| 17:44.03 | pabelanger | jits: Right, asterisk does not support that |
| 17:44.21 | pabelanger | so, Sales was correct that coding would need to be done |
| 17:44.50 | jits | pabelanger: :-( .. but i think it does support multiple one-to-one video calling.. right ? |
| 17:45.02 | pabelanger | yes |
| 17:45.09 | rampage73 | p3nguin, are you rofl at me or did i lose you? : |
| 17:45.10 | pabelanger | you can have multiple rooms |
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| 17:45.28 | jits | pabelanger: then can we have a client do multiple one-to-one video calls ? |
| 17:45.40 | p3nguin | rampage73: We don't support trixbox here. |
| 17:46.21 | p3nguin | But if I knew where DID variable was being set, I might be able to understand why it is sometimes null. |
| 17:46.23 | rampage73 | p3nguin, k sorry and thank you |
| 17:46.47 | rampage73 | p3nguin, I will look and see if I can find it |
| 17:46.52 | pabelanger | At the same time, no. They would need to switch from room to room. Remember, asterisk does not do any transcoding of the video, so if you phone supported multiple video interfaces it _might_ be possible |
| 17:47.01 | rampage73 | p3nguin, again thank you for helping |
| 17:47.27 | pabelanger | But most phone screens are only big enough for a single video source |
| 17:47.41 | jits | pabelanger: it is going to be a softphone, so it should be possible to tweak a client to do that .. isn't it ? |
| 17:47.55 | pabelanger | If you have source, sure |
| 17:48.17 | jits | pabelanger: can you recommend someone here who can help me .. ? |
| 17:49.14 | jkroon | irroot, no. hate agi. |
| 17:49.18 | pabelanger | I don't think may people have done work with video support and asterisk, aside from the people inside Digium. |
| 17:49.27 | jkroon | anyway, SHELL() is what I was looking for. |
| 17:49.42 | pabelanger | You could ask on asterisk-biz mailing list |
| 17:49.55 | jits | pabelanger: okay .. let me give it a shot.. |
| 17:50.23 | *** join/#asterisk libryder (~david@stg.maculon.com) |
| 17:50.28 | libryder | just learning asterisk :D |
| 17:50.37 | pabelanger | jits: You are basically looking to do the same thing as google hangouts, but viewers only see the talker, not other viewers |
| 17:50.40 | jkroon | irroot, when would suit you for that braai? you can pick almost any weekend in oct, just not the 1st or 15th. |
| 17:51.01 | jits | pabelanger: yes. exactly. |
| 17:51.09 | irroot | will get back to you on that one |
| 17:51.28 | jits | pabelanger: actually, viewers only see one person, not switch based on who is talking. |
| 17:51.39 | pabelanger | jits: Describe it like that in your post and see if you get any hits |
| 17:52.00 | jits | yeah .. sent in add request to mailing list |
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| 17:55.01 | *** part/#asterisk libryder (~david@stg.maculon.com) |
| 17:55.18 | jits | pabelanger: just did.. lets see .. :-) |
| 17:55.25 | jits | pabelanger: thanks for your help |
| 17:55.32 | pabelanger | np |
| 18:01.22 | jits | pabelanger: it seems to be putting me under wrong thread, hope thats not a problem. |
| 18:02.30 | brad_mssw | any know if the asterisk binary maintainers (ubuntu packages) plan on updating the binary libpri packages to 1.4.12 for ubuntu lucid/10.04 amd64? looks like 1.4.11.2 is currently what asterisk is distributing : http://packages.asterisk.org/deb/pool/main/libp/libpri/ |
| 18:02.56 | pabelanger | brad_mssw: no |
| 18:03.10 | pabelanger | what the issue? |
| 18:04.15 | brad_mssw | pabelanger: having an odd issue where my PRI starts reporting [Sep 19 19:17:56] ERROR[14624]: chan_dahdi.c:13941 dahdi_pri_error: PRI Span: 1 PTP MDL can't handle error of type F and [Sep 19 19:17:56] ERROR[14624]: chan_dahdi.c:13941 dahdi_pri_error: PRI Span: 1 MDL-ERROR (F), SABME in state 7 |
| 18:05.14 | brad_mssw | pabelanger: seems to be happening daily ... calling out doesn't work, but as soon as an incoming call is made ... it fixes the issue ... googling, the only thing I found was a reference to this : http://wiki.sangoma.com/Asterisk-FAQ#mdl-error-libpri and thought it could be related |
| 18:05.26 | pabelanger | brad_mssw: And upgrading libpri fixes? |
| 18:05.47 | brad_mssw | pabelanger: oh, I'm in the process of testing that ... I'll know in a couple of days ;) |
| 18:07.31 | pabelanger | We only backported libpri for ubuntu lucid because asterisk 1.8 requires 1.4.11.2 as a minimum version. So, if this is a bug in libpri, we should have Ubuntu Lucid backports team actually backport it, rather then us |
| 18:07.52 | pabelanger | However, that may take some time to do |
| 18:08.42 | brad_mssw | pabelanger: we're just making our own private libpri-1.4.12 deb and upgrading the 1.4.11.2 ... you don't see any issue with that, right? Should be ABI compatible without needing to recompile asterisk-dahdi, right? |
| 18:09.00 | pabelanger | I don't know |
| 18:09.18 | pabelanger | I'd have to check out it out and see |
| 18:09.23 | pabelanger | hopefully not |
| 18:10.32 | brad_mssw | guess we'll find out tonight when we do the package upgrade and restart asterisk |
| 18:13.02 | brad_mssw | pabelanger: I'm assuming it is probably related to this too https://issues.asterisk.org/view.php?id=17845 |
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| 18:22.55 | brad_mssw | and/or https://issues.asterisk.org/view.php?id=17360 |
| 18:23.00 | brad_mssw | both fixed in later libpri releases |
| 18:27.04 | pabelanger | brad_mssw: Ya, I'm trying to figure out with release fixed them |
| 18:27.23 | pabelanger | talking with rmudgett now |
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| 18:36.30 | pabelanger | brad_mssw: Yes, there will be ABI changes between 1.4.11 and 1.4.12. You should be safe with 1.4.11.5 |
| 18:38.41 | brad_mssw | pabelanger: ok, thanks, we'll try 1.4.11.5 then |
| 18:46.11 | jkroon | irroot, how easy will it be to implement a Monitor() for faxing? |
| 18:46.28 | jkroon | i've got a client that would be more than happy to hand you some $$$ for that. |
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| 18:47.12 | irroot | jkroon its not so simple really thats the problem can look into it |
| 18:48.08 | jkroon | you've got my email - send me a quote please? |
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| 19:04.30 | rampage73 | p3nguin, i found it in extensions_additional.conf here is the first "exten => _X.,1,Set(DID=${EXTEN})" excluding quotes and the second "exten => s,n,Set(DID=${IF($["${DID}"= ""]?s:${DID})})" does that help? |
| 19:07.06 | p3nguin | Maybe. Set DID to the extension called... then, if ${DID} is null, Set DID to s, else leave it as what it currently is. |
| 19:07.37 | p3nguin | So if a call comes in from the provider, it has to have at least two characters to even match that extension. |
| 19:08.15 | p3nguin | So ${EXTEN} would _always_ be at least one number 0-9 and at least one more character. |
| 19:08.26 | jkroon | rampage73, in my experience you probably actually want _X! not _X. |
| 19:08.54 | p3nguin | Therefore, DID will always have a value, and the second check would always be false. |
| 19:09.40 | p3nguin | Futhermore, ${DID} would always have a value when dialing out in that other Dial() you showed me. |
| 19:10.09 | p3nguin | Which means I need to see some verbosity/debug to know what went wrong. |
| 19:11.48 | rampage73 | p3nguin, in case no one has told you lately you are worth Gold! thank you for the explanation |
| 19:12.16 | rampage73 | jkroon, sorry I am a newb to asterisk what is the difference? |
| 19:13.08 | p3nguin | . requires one or more characters in order to match; ! matches zero or more characters. |
| 19:13.51 | rampage73 | p3nguin, in case you are willing to take a look at it how much verbosity are we talking? our default is on 3 but I do know how to set it higher if needed |
| 19:13.54 | p3nguin | Meaning _X! would accept 1 number, or 1 number and additional characters. |
| 19:13.56 | rampage73 | p3nguin, thanks again |
| 19:14.19 | p3nguin | 3 is high enough -- it won't increase usefulness above that. |
| 19:15.21 | p3nguin | Also, since you're never going to need to match only a single digit number from the provider. _X! isn't going to really be beneficial, and could actually cause a problem. |
| 19:15.36 | p3nguin | _X. will be fine. |
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| 19:28.56 | _abc_ | Hello. Is someone here who knows about unistim support? There seems to be a button on Nortel/Avaya 1150e which does not work on unistim. The green right bottom one, which switches between handset and speakerphone. Does anyone know if this is supported in newer unistims? |
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| 19:42.18 | dym | Hey http://pastebin.com/ckzuDEbk - Any hints? The file is in place. |
| 19:43.00 | p3nguin | Show me that it exists. |
| 19:43.04 | dym | sec |
| 19:44.42 | dym | mhh |
| 19:44.48 | dym | my locate db was old |
| 19:44.50 | dym | its actually gone |
| 19:45.13 | WIMPy | Then not only your locate db, but your locate is out of date. |
| 19:47.29 | dym | locate uses the locatedb |
| 19:47.49 | p3nguin | I didn't understand the statement, either. |
| 19:48.07 | p3nguin | Maybe he's suggesting that you should be using something like mlocate. |
| 19:48.16 | dym | locate |
| 19:48.24 | dym | is what i used to locate the file |
| 19:48.35 | dym | but the index of the filedb that the tool uses was outdated |
| 19:48.38 | WIMPy | or slocate. |
| 19:48.39 | p3nguin | No sense in going over it time after time. |
| 19:48.41 | dym | so it showed files where there were none |
| 19:48.44 | dym | wtf |
| 19:48.59 | WIMPy | It won't show files that have gone. |
| 19:49.27 | p3nguin | Really? I didn't know that. |
| 19:50.00 | WIMPy | That's because it checks permissions for all hits before displaying them. |
| 19:50.10 | p3nguin | I had no idea. |
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| 20:07.23 | Dovid | hello Y'all |
| 20:12.37 | azv4 | Anyone here very familiar with Panasonic Digital Hybrid phone systems? I am having a terrible time getting support for ours, and I am hoping to find someone who can offer some suggestions or point me in the right direction on getting support |
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| 20:34.38 | atan | Well I just had the strangest thing happen to me. Upgraded to the newer * 10 and when you leave a voicemail it detects it as being under 3 seconds and deletes the voicemail. I had to set the voicemail min time limit thing to 0 before it will accept voicemails. |
| 20:35.39 | dym | Does anyone have a complete set of german voiceprompts, including all call screening options? |
| 20:35.47 | *** join/#asterisk Wiretap7 (~Wiretap@unaffiliated/wiretap) |
| 20:35.51 | atan | The voicemails it does save are indeed longer than 3 seconds but it's reading them wrong. Watching the console thinger (asterisk -r) shows it as saving it but then checks length and throws it in the trash :) |
| 20:36.04 | *** join/#asterisk tamiel (~tamiel@ip-183.net-81-220-86.toulouse.rev.numericable.fr) |
| 20:37.34 | *** join/#asterisk kc8pxy (~gecko@99-182-113-98.lightspeed.clmboh.sbcglobal.net) |
| 20:37.37 | *** join/#asterisk dlynes_laptop (~dlynes@bas8-hamilton14-3096452086.dsl.bell.ca) |
| 20:47.57 | *** join/#asterisk jeffgus (~jeffgus@2001:470:f2eb:1::4) |
| 20:54.31 | *** join/#asterisk ferdna (~yup@cpe-67-10-220-35.elp.res.rr.com) |
| 20:54.42 | eject_ck | Hi all |
| 20:55.56 | dym | hi |
| 20:56.48 | citywok | marco |
| 20:56.48 | eject_ck | I'm using connection to internetcalls voip provider and it works fine for a long time, today I've noticed strange Unauthorized headers during sip debug. I have registration via register => and I'm able to make calls without any issues. |
| 20:57.02 | eject_ck | How can I understand this log http://pastebin.com/TDDewXac |
| 20:57.02 | *** join/#asterisk mocker (~mocker@206.55.118.84) |
| 20:57.27 | mocker | Dooes 1.8.7-rc2 contain the following patch: https://reviewboard.asterisk.org/r/1402/ ? |
| 20:57.32 | eject_ck | many thanks in advance |
| 20:58.13 | mocker | Or is that something I'll have to use SVN to get? |
| 20:58.24 | Qwell | mocker: It does not. |
| 20:58.55 | eject_ck | I have active registration: |
| 20:58.55 | eject_ck | sip.internetcalls.com:5060 N eject 3585 Registered Thu, 22 Sep 2011 23:47:09 |
| 21:00.31 | mocker | Qwell: Think I'm hitting that bug, would you suggest just doing a trunk svn checkout and running on that? |
| 21:04.42 | anonymouz666 | mocker: get the diff and apply the patch |
| 21:04.54 | anonymouz666 | then you don't have to wait |
| 21:05.51 | *** join/#asterisk Dovid (~Dovid@213.8.121.90) |
| 21:07.34 | mocker | anonymouz666: But the trunk version would also have that patch and potentially other patches, so I'm considering running that. |
| 21:07.48 | mocker | Althought I've never run trunk in production before. :( |
| 21:07.56 | mocker | although |
| 21:09.38 | *** join/#asterisk evan458 (~evan@c-71-59-154-33.hsd1.or.comcast.net) |
| 21:10.21 | evan458 | I have questions about setting up 5 VOIP lines for a new company/office, can someone PM me to help? |
| 21:10.47 | Qwell | ~ask |
| 21:10.47 | infobot | Questions in the channel should be specific, informative, complete, concise, and on-topic. Don't ask if you can ask a question first. Don't ask if a person is there; just ask what you intended to ask them. Better questions more frequently yield better answers. We are all here voluntarily or against our will. |
| 21:11.02 | evan458 | okay okay |
| 21:12.10 | evan458 | I want to setup an office with 5 VOIP lines. I want each line to connect through a computer. Each line will be used to continually call out all day. What is the required hardware and software, and what service do I have to pay for? |
| 21:12.12 | p3nguin | Your questions can be answered here, unless you're wanting to hire a consultant to do it for you. |
| 21:12.23 | Qwell | stop |
| 21:12.31 | p3nguin | VoIP lines... this is a contradiction. |
| 21:12.35 | Qwell | don't ever use the word "lines" when referring to VoIP |
| 21:12.40 | evan458 | ok |
| 21:12.45 | p3nguin | Do you want lines, or do you want to use VoIP? |
| 21:13.10 | evan458 | I want to be able to call people from a computer with an assigned phone number, which i think is VOIP. |
| 21:13.21 | Qwell | It doesn't have to be. |
| 21:13.22 | evan458 | I want to use a high speed internet connection to support the traffic |
| 21:13.25 | p3nguin | So you're going to have an ITSP? |
| 21:13.28 | p3nguin | ~itsp |
| 21:13.28 | infobot | [~itsp] An ITSP is an Internet Telephony Service Provider (or VoIP telephone company). They allow you to either SEND calls to the PSTN (this is called termination), RECEIVE calls from the PSTN (called origination), or both. Some offer fixed rates, others $/min. Enter ~itsplist-us (USA) or ~itsplist-ca (Canada) for a listing of popular ITSPs. |
| 21:13.30 | kc8pxy | p3nguin: voIP == channels? |
| 21:13.51 | Qwell | kc8pxy: lines carry channels. channels is a valid term with VoIP. |
| 21:14.09 | kc8pxy | kk.. though so |
| 21:14.37 | evan458 | okay, so i think i need ITSP with 5 'channels' that have unlimited minutes |
| 21:14.45 | p3nguin | Using a computer to make phone calls via ITSP is an extremely common concept. |
| 21:15.08 | dym | Which is the call screening option that always queries for the name, regardless if a callee has already been recorded? |
| 21:15.24 | evan458 | yea, i'm trying to make sense of these sites that offer services and they have so many acronyms it's hard to grasp off the bat for me |
| 21:16.01 | p3nguin | Some ITSPs will happily give you more than five channels, where others will limit you to two channels unless you beg them and pay them a ridiculous fee for each additional channel. |
| 21:16.20 | p3nguin | What site are you talking about? |
| 21:16.41 | mocker | anonymouz666: trunk will contain that patch, right? |
| 21:16.58 | evan458 | lemme find it |
| 21:17.03 | *** join/#asterisk DanFromUK (DanFromUK@2.27.10.123) |
| 21:17.10 | Qwell | mocker: You don't want to run trunk. |
| 21:17.34 | mocker | Qwell: Ok, so 1.8.7rc2 + diff? |
| 21:17.38 | DanFromUK | Hello. Does anyone have access to Cyprus DIDs that can support fax2email? or T38? |
| 21:17.58 | anonymouz666 | mocker: you could do that or wait the 1.8.8.0-rc1 |
| 21:18.52 | evan458 | okay, so i was just trying to find the cheapest way, and i found http://www.consumer-rankings.com/voip/pricing and it looked like I would want to go with 'nextiva' which offered 4 unlimited lines for 100$ |
| 21:19.14 | Qwell | ugh |
| 21:19.21 | Qwell | ~cheap |
| 21:19.21 | infobot | well, cheap is a bad idea. If you're setting up an Asterisk system, one of the wisest pieces of advice you can take is this: DON'T BE A CHEAPSKATE. |
| 21:19.31 | p3nguin | If they say lines, they probably aren't worth it anyway. |
| 21:19.40 | *** join/#asterisk heffer (~felix@fedora/heffer) |
| 21:19.40 | *** join/#asterisk dre (dre@gateway/shell/xzibition.com/x-aagulhyylwfnqgjb) |
| 21:19.40 | *** join/#asterisk madduck (~madduck@debian/developer/madduck) |
| 21:19.42 | p3nguin | ~savemoney |
| 21:19.42 | infobot | <Gremlin> It's okay, but I was hoping to save some money with a soldering iron and a bunch of sound cards. |
| 21:20.12 | anonymouz666 | looooooooool |
| 21:20.21 | p3nguin | How many minutes do you project will be used in a month? |
| 21:20.27 | p3nguin | in total, of course. |
| 21:20.27 | Qwell | he left |
| 21:20.29 | _Corey_ | did someone actually say that once? |
| 21:20.30 | p3nguin | oh? |
| 21:20.48 | Qwell | _Corey_: surely you aren't surprised? |
| 21:20.50 | p3nguin | Well hell. Talking to myself again. |
| 21:21.00 | _Corey_ | no, not really just really amused |
| 21:21.29 | *** join/#asterisk evan574 (~evan@c-71-59-154-33.hsd1.or.comcast.net) |
| 21:21.38 | p3nguin | How many minutes do you project will be used in a month in total? |
| 21:21.38 | evan574 | woops, using web client, tried to drag tab out of window |
| 21:22.22 | evan574 | the lines will be used by people calling businesses constantly to sell them on web services (like a call center) |
| 21:22.33 | p3nguin | So you're back to line again. |
| 21:22.34 | _Corey_ | Never underestimate the power of the word "unlimited" to prompt someone who makes almost no phone calls to use a different provider |
| 21:22.44 | evan574 | so i figure i will want unlimited minutes |
| 21:23.08 | evan574 | lol yea, i think i actually qualify as needing it |
| 21:23.10 | _Corey_ | I've got a client who has an analog line from Verizon on his desk next to his Polycom because it is "unlimited" |
| 21:23.24 | p3nguin | I realize you think unlimited means unlimited, but there's always a limit. How many minutes do YOU think YOUR staff will use in a month? |
| 21:23.38 | evan574 | well,, lemme calculate |
| 21:23.42 | p3nguin | Perfect. |
| 21:24.37 | eject_ck | Is that register => related ? |
| 21:24.51 | evan574 | 6160 minutes a month per person, and so at least 3 people right off the bat are going to be using that much, so 18000 minutes a month for them |
| 21:25.52 | p3nguin | With that much volume, you'll probably want unmetered service. |
| 21:26.32 | mocker | Ok, downloaded the diff from https://reviewboard.asterisk.org/r/1402/diff/#index_header and applied it with patch <diff_file to asterisk-1.8.7-rc2 |
| 21:26.35 | p3nguin | Did Nextiva give you a quote for five unmetered channels, or was that number a guess? |
| 21:26.45 | mocker | That sound correct? |
| 21:27.33 | evan574 | i haven't talked to sales reps from any company because i figure they just want to sell me their products. |
| 21:28.27 | evan574 | according to http://www.nextiva.com/products/pbx-sip-trunking.html it says 29.95/mo. per 'seat' (whatever seat means) |
| 21:28.42 | p3nguin | I'd guess per seat probably means per channel. |
| 21:30.32 | evan574 | and i've never setup things like this before, so am i right to assume that if i paid for 3 seats/channels, then I could download a softphone like ekiga and input some form of user/pass or credentials that i got from nextiva and it would be operational? or is there more equipment i'm missing |
| 21:30.42 | p3nguin | That would assume they aren't allowing a single person to make two calls (one on hold, make a second call). |
| 21:31.05 | p3nguin | I don't know how they would make that limitation, though, since Asterisk would be the only peer they would ever see. |
| 21:31.28 | p3nguin | So... I'd have to ask them what the heck they mean by per seat. |
| 21:32.34 | evan574 | and i should specify, i don't necessarily know that I need asterisk, but it was the only place i found that had an IRC channel and i figured people here could answer questions about VOIP setups. |
| 21:33.23 | p3nguin | I doubt you'd want to make that kind of call volume without something such as Asterisk, even if it's not necessarily Asterisk that you decide to use. |
| 21:33.55 | evan574 | somehow i imagined that VOIP could be cheap even with unlimited calling, (like 10$ a line), since i've always used googlevoice until now. |
| 21:34.06 | p3nguin | Without something, you won't have the option of recording calls for quality assurance, nor have the ability to spy on the agents' calls, etc. |
| 21:34.22 | p3nguin | But since VoIP doesn't have lines, I guess you were wrong. |
| 21:34.24 | evan574 | ooh, good point, i'm def going to want those features |
| 21:35.43 | p3nguin | If you will have incoming calls, you'd want a queue system as well. |
| 21:35.46 | *** join/#asterisk tyman (~tyman@173-12-219-189-Fresno.hfc.comcastbusiness.net) |
| 21:37.55 | evan574 | yea, we won't really have incoming calls except directly back to the person who made an outgoing call. For example, Joe is calling all realtors in Cali to offer them advertising, he really only needs one line to call out on, and sometimes people who missed his call will call back, but they will need to get back to Joe (not another agent) |
| 21:38.12 | brad_mssw | pabelanger: just deployed libpri 1.4.11.5 to see if it corrects the issue, should know in a couple of days. So far tested out ok. |
| 21:38.18 | *** part/#asterisk wesphillips (~wphill04@137-237-233-124.harris.com) |
| 21:38.31 | p3nguin | Will you have a DID for every agent who is calling out? |
| 21:39.03 | evan574 | DID? |
| 21:39.16 | p3nguin | phone number for someone to call inward |
| 21:39.20 | evan574 | yes. |
| 21:42.37 | evan574 | (i'm calling nextiva to see what they say) |
| 21:43.09 | atan | thinks he might be right behind evan574 in nextivia queue |
| 21:43.17 | p3nguin | Make sure you tell them that you are doing telemarketing. |
| 21:43.24 | atan | evan574, while you have them on the line ask if the $29.95 unmetered includes calls to Aussie land for me :P |
| 21:43.29 | p3nguin | Many providers do not allow telemarketing. |
| 21:43.35 | evan574 | lol, i am caller number 3 |
| 21:43.37 | atan | evan574: ask my question before asking that question |
| 21:43.46 | evan574 | rofl, ok |
| 21:43.51 | atan | evan574, I'm #2 so neener neener neener :D but I'll be quick I swear :-) |
| 21:44.00 | p3nguin | haha |
| 21:44.12 | atan | Follow up the telemarketing thing by "and I was told to come here from IRC" |
| 21:44.18 | _Corey_ | haha |
| 21:44.28 | atan | Tell then p3nguin sent you. |
| 21:44.45 | p3nguin | That might get you hung up on. |
| 21:44.54 | evan574 | rofl |
| 21:44.58 | atan | Well actually I just got hung up on. |
| 21:45.03 | devil_evoxxx | \quit |
| 21:45.03 | p3nguin | See?! |
| 21:45.10 | evan574 | OOOH ME TOO |
| 21:45.17 | atan | Fsck. I had a good line about 2600hz to tell them |
| 21:45.19 | evan574 | but i called back and i'm caller number 3 again |
| 21:45.40 | atan | evan574: okay you hold on there and ask about Aussie land I need to go rotate stuff I have on the stove |
| 21:45.44 | *** join/#asterisk Defraz (~Defraz@corp.fuzecore.com) |
| 21:45.54 | evan574 | does their robot glitch out on you too, "You are currently caller number /glitch/ you are currently/ glitch/ 3." |
| 21:46.28 | p3nguin | I'd be skeptical of a VoIP company that can't even provide reasonable services to callers. |
| 21:46.31 | atan | Yes. It does. |
| 21:47.09 | KavanS | evan574, what does the 574 stand for may I ask? |
| 21:48.12 | evan574 | brb |
| 21:50.57 | Qwell | KavanS: You're on to him |
| 21:51.04 | KavanS | lol |
| 21:52.03 | atan | http://www.alcazarnetworks.com/wholesaleterm seems interesting... 0.00143... hmm. |
| 21:52.19 | _Corey_ | I know those guys, would recommend |
| 21:52.29 | Qwell | ~itsplist-us |
| 21:52.30 | infobot | Here are some popular ITSPs (USA) starting with the more respected ones: http://www.teliax.com , http://www.voicepulse.com/connect/ , http://www.broadvoice.com , http://www.jnctn.com , http://www.bandwidth.com , http://vitelity.net , http://voip.ms and http://flowroute.com |
| 21:52.32 | Qwell | fwiw |
| 21:53.45 | atan | _Corey_ do they do international term? |
| 21:53.48 | dym | Im trying to send a MixMonitor recorded file after its been recorded in the dialplan priority. Any idea why this fails? http://pastebin.com/7qRw8cKw |
| 21:54.47 | _Corey_ | atan: Yeah, I think so. They actually sell bulk termination to a few of the guys on the itsplist-us list, surprisingly... |
| 21:55.21 | atan | _Corey_, well at those prices they beat voip.ms. |
| 21:55.46 | _Corey_ | They had a booth at ITEXPO last week in Austin... seem to be getting bigger |
| 21:56.02 | _Corey_ | they were new to the wholesale market last year when I first ran into them |
| 21:56.37 | p3nguin | dym: What's the error? |
| 21:56.56 | p3nguin | dym: Does asterisk run as user asterisk rather than root? |
| 21:57.21 | dym | p3nguin: runs as root |
| 21:57.27 | p3nguin | Carry on. |
| 21:57.29 | p3nguin | Bad. |
| 21:57.37 | atan | _Corey_ interesting. |
| 21:57.54 | atan | _Corey_ they don't have a form to register on their website. Is it prepaid or invoiced at the end of the month? |
| 21:58.03 | dym | p3nguin: how was that helpful? |
| 21:58.32 | _Corey_ | atan: I think they offer both. |
| 21:59.18 | p3nguin | dym: How was what I said helpful? It wasn't. You didn't tell me the information I asked for to try to help you, but then you told me you were running as root, which eliminated the concern I had anyway. |
| 21:59.38 | dym | well |
| 21:59.41 | dym | there is no error at all |
| 21:59.49 | dym | on CLI in debug mode |
| 21:59.50 | dym | no err |
| 21:59.50 | p3nguin | I'm sure there's something. |
| 22:00.02 | _Corey_ | atan: I think you need to e-mail them or call and they'll set you up with a trunk |
| 22:00.29 | dym | p3nguin: where would that be? |
| 22:00.31 | p3nguin | Either in the sent file or stdout/stderr, which you won't be seeing by looking at the asterisk CLI. |
| 22:00.49 | p3nguin | Redirect the output into a file, then read the file. |
| 22:01.07 | p3nguin | mutt &> mutt.log |
| 22:04.52 | p3nguin | On another note, I'd consider using ${MIXMONITOR_FILENAME} as the attachment. |
| 22:05.12 | dym | Oh |
| 22:05.19 | dym | does that grab the current recording filenameà |
| 22:05.41 | dym | including path, or just filename? |
| 22:05.43 | p3nguin | The variable contains the name of the file that MixMonitor() has recorded on the current channel. |
| 22:06.18 | dym | With its path, or just the filename? |
| 22:06.24 | evan574 | i didn't get a chance to ask about aussie |
| 22:06.36 | p3nguin | I just use mutt -a ${MIXMONITOR_FILENAME}, so I guess it's the full path. |
| 22:06.45 | p3nguin | I never expanded it to see the data. |
| 22:07.12 | evan574 | they provide 4-7 unlimited calling lines/channels at 24.97$/ea. + 6$ in taxes, so 5 lines came to 153$ a month |
| 22:07.22 | evan574 | thank you guys for the help! |
| 22:07.30 | evan574 | i'm off to buy usb headsets on newegg |
| 22:07.40 | evan574 | and btw, 574 was randomly generated by my webIRC client |
| 22:07.49 | evan574 | irc2go.com |
| 22:07.52 | evan574 | ciao! |
| 22:08.31 | dym | p3nguin: could i see your dialplan line for comparison? |
| 22:11.13 | p3nguin | exten => h,n,System(/bin/echo "Please see attachment."|/usr/bin/sudo -u asterisk /usr/bin/mutt -a ${MIXMONITOR_FILENAME} -s "Recording" -- me@email); |
| 22:11.30 | dym | cheers |
| 22:12.33 | dym | whats with the trailing ; = |
| 22:12.34 | dym | ? |
| 22:12.38 | p3nguin | end of line |
| 22:12.42 | dym | oh |
| 22:12.53 | p3nguin | It's not required in .conf, but I have them anyway. |
| 22:14.01 | *** join/#asterisk TimeRider (~steve@188-220-33-123.zone11.bethere.co.uk) |
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| 22:16.02 | dym | odd |
| 22:16.18 | dym | i checked my line and seems similar - still no errors |
| 22:16.47 | p3nguin | When you redirected the output to a file, the file was empty? |
| 22:17.01 | dym | indeed |
| 22:17.14 | p3nguin | That sucks. Do you have an MTA configured and running? |
| 22:17.19 | dym | yupp |
| 22:17.19 | *** join/#asterisk CrossWired (~CrossWire@pool-71-101-94-85.tampfl.dsl-w.verizon.net) |
| 22:17.30 | p3nguin | Do you have mutt compiled with debug support? |
| 22:17.46 | dym | debian package |
| 22:17.50 | dym | suppose so |
| 22:17.51 | p3nguin | *shrug* |
| 22:18.09 | dym | ah -d |
| 22:18.10 | p3nguin | Throw in a -d 5 and see what you get in the file. |
| 22:18.12 | dym | yes :D |
| 22:21.10 | atan | Well I'll be. With these prices one could give voip.ms a run for their money and still maintain a nice profit margin... |
| 22:21.15 | atan | puts on a big grin |
| 22:21.28 | *** join/#asterisk bchia (~Adium@user-24-236-94-155.knology.net) |
| 22:21.31 | p3nguin | Which company are you considering? |
| 22:21.41 | atan | p3nguin, alkazara |
| 22:21.49 | atan | Alcazara even. |
| 22:24.41 | dym | ffs |
| 22:24.43 | dym | this is a pain |
| 22:24.48 | dym | now it refuses to write debug |
| 22:25.02 | p3nguin | Still redirecting output? |
| 22:25.11 | dym | no |
| 22:25.15 | dym | as of -d not anymore |
| 22:25.25 | p3nguin | But the debug file isn't created? |
| 22:25.30 | dym | indeed |
| 22:25.41 | dym | no ~/.muttdebug0 |
| 22:25.49 | p3nguin | Did you redirect with the -d 5 even once? |
| 22:26.03 | dym | nah |
| 22:26.04 | dym | trying now |
| 22:26.25 | p3nguin | When I use -d, it doesn't help. When I redirect, I see why: mutt is not compiled with debug support. |
| 22:27.52 | dym | this is so annoying |
| 22:28.00 | *** join/#asterisk BuenGenio (~Gene@176.Red-79-156-96.staticIP.rima-tde.net) |
| 22:29.41 | dym | Kinda getting the feeling my System() is not even executed. |
| 22:30.09 | p3nguin | System(echo hi > testfile) |
| 22:31.33 | dym | nope |
| 22:31.35 | dym | no testfile |
| 22:31.37 | dym | thought so |
| 22:31.45 | p3nguin | Where are you looking for it? |
| 22:31.54 | dym | systemwide |
| 22:32.06 | p3nguin | Running as asterisk, it should end up in /var/lib/asterisk. |
| 22:32.17 | p3nguin | As root, it probably ends up in /root. |
| 22:33.20 | dym | retrying with limited access now |
| 22:34.37 | dym | okay |
| 22:34.38 | dym | well |
| 22:34.42 | dym | sticking with root for now. |
| 22:34.45 | dym | but still - no file |
| 22:35.08 | dym | hah |
| 22:35.10 | dym | this is odd |
| 22:35.20 | dym | mutt does create a "sent" file, containing all the mails and wav attachments |
| 22:36.06 | p3nguin | What's your reason for running it as root? |
| 22:36.21 | *** join/#asterisk neurosys (~neurosys@70-7-90-136.pools.spcsdns.net) |
| 22:36.27 | dym | no particular reason for now. its a non-productive env |
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| 22:44.37 | dym | Cant think straight - probably easy reason its not working |
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