| 00:06.11 | *** join/#asterisk rossand (~aross@CPE485b390978ce-CM00222ddf42dd.cpe.net.cable.rogers.com) |
| 00:07.42 | gunnarar | anyone here want to help with echo problems? |
| 00:15.40 | *** join/#asterisk vinhdizzo (~vinh@dhcp-v021-062.mobile.uci.edu) |
| 00:22.37 | davidcsi | chris, yes nat=yes |
| 00:23.04 | davidcsi | there's no alg (i think) everything else works fine |
| 00:24.07 | ChrisInSydney | davidcsi: Have you tried another ITSP ? like an ideasip or some freebie ? |
| 00:24.43 | ChrisInSydney | I'm just suggesting things I would try |
| 00:24.55 | ChrisInSydney | like use a bigger hammer |
| 00:25.01 | ChrisInSydney | take up a new hobby |
| 00:25.29 | ChrisInSydney | take some time off on "stress" leave |
| 00:26.22 | ChrisInSydney | Not that I am suggesting them as a solution, for your problem..... |
| 00:26.34 | dym | Heyo ChrisInSydney |
| 00:26.38 | ChrisInSydney | ...asterisk problem... |
| 00:26.40 | ChrisInSydney | hey dym |
| 00:26.50 | dym | All good bruv? |
| 00:27.29 | ChrisInSydney | yup, sorta |
| 00:27.53 | ChrisInSydney | damn SPA525G2s are locking up on a client site, randomly. Not on an Asterisk system. IPFX |
| 00:28.18 | dym | Could you do me a quick favor?urgh |
| 00:28.31 | ChrisInSydney | but only handsets with a MAC address of 4055, not the 0007 MACs |
| 00:28.47 | dym | odd |
| 00:28.49 | dym | google says? |
| 00:29.53 | dym | Do you have a softphone handy and could possibly do me a quick favour? |
| 00:31.22 | ChrisInSydney | got a couple of hard phones with free accounts |
| 00:31.28 | ChrisInSydney | what do you want me to log into >> |
| 00:31.30 | ChrisInSydney | ?? |
| 00:31.33 | dym | astconf@sip.openroot.de |
| 00:31.38 | dym | just plain call |
| 00:31.46 | ChrisInSydney | Done |
| 00:31.49 | dym | Just see if you get through |
| 00:32.04 | ChrisInSydney | give me a couple to hack the dial plan of my dev system |
| 00:32.12 | dym | np |
| 00:32.56 | *** join/#asterisk waschtl (~waschtl@216.129.239.149) |
| 00:39.30 | ChrisInSydney | Im in ! |
| 00:39.39 | ChrisInSydney | dym: im in |
| 00:39.44 | dym | Great - All I needed (: |
| 00:39.45 | dym | Thanks |
| 00:39.54 | ChrisInSydney | cool. Audio was OK |
| 00:39.57 | *** join/#asterisk mjordan (~mjordan@user-69-1-6-49.knology.net) |
| 00:39.57 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 00:40.01 | ChrisInSydney | a little scratchy. gsm ?? |
| 00:40.13 | ChrisInSydney | I'm dailling in using 722 |
| 00:41.36 | dym | yeah its gsm |
| 00:41.43 | dym | i could crank it up tho |
| 00:43.37 | dym | done |
| 00:44.02 | *** join/#asterisk LostyJai (~blah@202.171.190.130) |
| 00:53.38 | ChrisInSydney | dym: Sounds like 711 for the voice prompts, but the moh is 722 |
| 00:53.49 | dym | mhh |
| 00:53.55 | dym | could be |
| 00:57.11 | *** part/#asterisk LostyJai (~blah@202.171.190.130) |
| 01:00.40 | *** join/#asterisk rossand (~aross@CPE485b390978ce-CM00222ddf42dd.cpe.net.cable.rogers.com) |
| 01:03.36 | WIMPy | Is there something broken about IAX in TRUNK? |
| 01:04.14 | WIMPy | It looks like all IAX calls go to the h extensions when they match something else. |
| 01:04.55 | phix | _s.,1,catchallcommand() |
| 01:05.37 | phix | ooops, remove that s |
| 01:05.57 | WIMPy | It only happens on calls that match an extension. |
| 01:06.10 | WIMPy | Calls that don't match are rejected. |
| 01:06.13 | phix | hmmmm |
| 01:07.03 | phix | would i use iax over sip? |
| 01:07.15 | WIMPy | There are switches involved, but the same context works for sip and lcr. |
| 01:07.33 | WIMPy | Can you rephrase that? |
| 01:07.49 | *** join/#asterisk kessius (~cassio@201.21.173.58) |
| 01:15.05 | *** part/#asterisk vinhdizzo (~vinh@dhcp-v021-062.mobile.uci.edu) |
| 01:18.37 | dym | ChrisInSydney: This is up and running 24/7 - if anyone would like to may some use of it. ill be on there too |
| 01:19.13 | *** part/#asterisk mjordan (~mjordan@user-69-1-6-49.knology.net) |
| 01:23.08 | ChrisInSydney | dym: cool |
| 01:24.58 | dym | Theres landline access too |
| 01:25.05 | dym | 0049 541 50799943 |
| 01:25.39 | WIMPy | Oskar Drück? |
| 01:26.49 | dym | jepp |
| 01:28.23 | *** join/#asterisk oli004 (2edf109e@gateway/web/freenode/ip.46.223.16.158) |
| 01:28.37 | *** part/#asterisk oli004 (2edf109e@gateway/web/freenode/ip.46.223.16.158) |
| 01:29.31 | *** join/#asterisk Bullmoose (~Bullmoose@71-37-175-157.bois.qwest.net) |
| 01:29.54 | *** join/#asterisk oli004 (2edf109e@gateway/web/freenode/ip.46.223.16.158) |
| 01:31.57 | kessius | hi good friends |
| 01:32.34 | dym | Hey |
| 01:32.41 | dym | Good friends? (: |
| 01:32.42 | kessius | I need help, I have problem in voicemail.conf |
| 01:35.53 | kessius | how do I pass voicemail.conf for realtime |
| 01:37.06 | *** join/#asterisk pdtpatr1ck_ (~pdtpatric@12.249.4.226) |
| 01:43.15 | kessius | anyone knows, pass voicemail.conf for real-time ? |
| 01:45.00 | *** join/#asterisk puzzled (~patrick@2001:980:5e31:1:6ef0:49ff:fe50:659d) |
| 01:49.56 | kessius | voicemail facility, Anyone know if need realtime static or dynamic ? |
| 01:53.36 | *** join/#asterisk oli004 (~IceChat77@ex.mediothek.eu) |
| 01:53.43 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.218.194) |
| 01:56.13 | *** join/#asterisk mjordan (~mjordan@user-69-1-6-49.knology.net) |
| 01:56.13 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 01:56.28 | oli004 | spricht jemand deutsch ? |
| 01:56.44 | WIMPy | #asterisk-de |
| 01:56.58 | *** join/#asterisk oli_004 (2edf109e@gateway/web/freenode/ip.46.223.16.158) |
| 01:57.02 | kessius | My boss wants to voicemail, easy voicemail - how to automate to real-time ? |
| 01:57.19 | dym | kessius: wait for someone to reply instead of re-posting every 10 minutes. |
| 01:57.51 | WIMPy | oli004: /join #asterisk-de |
| 01:57.58 | oli004 | thanks |
| 01:58.31 | oli004 | hi oli004, cool nick !! |
| 01:59.45 | *** join/#asterisk oli_004 (~oli_004@HSI-KBW-46-223-16-158.hsi.kabel-badenwuerttemberg.de) |
| 02:00.38 | *** join/#asterisk oli_004 (~oli_004@HSI-KBW-46-223-16-158.hsi.kabel-badenwuerttemberg.de) |
| 02:01.50 | dym | kicks [TK]D-Fender |
| 02:01.52 | dym | unbelievable! |
| 02:02.44 | WIMPy | o.O |
| 02:04.54 | Bullmoose | kessius. I'm new to Asterisk myself. Without knowing your situation, "simple" could be an answering machine from WalMart |
| 02:05.58 | Bullmoose | I'm going through Asterisk Essentials class and also have the new Asterisk Book. Asterisk Book is free online. It shows how. |
| 02:06.27 | Bullmoose | ...and one person's easy is another person's impossible |
| 02:10.04 | *** join/#asterisk GGD (~deberle@pool-173-72-204-39.clppva.fios.verizon.net) |
| 02:10.24 | WIMPy | ~pb |
| 02:10.24 | infobot | [~pb] A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few: http://www.pastebin.com , http://pastebin.us , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://bin.cakephp.org/ , http://asterisk.pastey.net/ ; or install pastebinit with yum or aptitude. |
| 02:11.25 | ChrisInSydney | dym: Its lonely in here, the music is cool, but gets repetative the 4th or 5th time you hear it ;-) |
| 02:11.35 | ChrisInSydney | Hey WIMPy |
| 02:12.06 | dym | ChrisInSydney: Im off to bed. 3 am - 3 hrs sleep left :/ feel free to pull others in/spread it and ill be there tomorrow. |
| 02:12.11 | ChrisInSydney | kessius: There is a gotcha with MWI and Real TIme Voicemail |
| 02:12.46 | ChrisInSydney | dym: you're a stronger man than I, hence I am not on the 4am Saturday VUC calls |
| 02:13.17 | ChrisInSydney | get some zzzs |
| 02:13.21 | ChrisInSydney | catch ya |
| 02:13.49 | *** join/#asterisk Neptu (~Neptu@213-67-244-97-no100.tbcn.telia.com) |
| 02:14.31 | dym | ayy |
| 02:17.35 | kessius | Você quis dizer: podes explicar me ? |
| 02:17.35 | kessius | MWI can you explain? |
| 02:19.03 | kessius | <PROTECTED> |
| 02:19.04 | kessius | MWI can you explain? |
| 02:20.13 | ChrisInSydney | Message Waiting Indicator: The little voicemail light on your SIP handset |
| 02:20.55 | ChrisInSydney | there are issues with a "RealTime" database storage for voicemail and MWI |
| 02:21.24 | ChrisInSydney | kessius: It depends on what you are trying to achieve |
| 02:21.40 | ChrisInSydney | if you need to use Asterisk real time components |
| 02:22.27 | *** join/#asterisk serafie (~erin@75.76.38.159) |
| 02:30.44 | *** part/#asterisk oli004 (~IceChat77@ex.mediothek.eu) |
| 02:31.40 | kessius | already some files in real-time, - I ask you guys - the file voicemail.conf how to proceed to go to real-time |
| 02:37.04 | *** join/#asterisk postconf (~postconf@msfree.com) |
| 02:41.15 | kessius | ChrisInSydney, never heard of real-time components! |
| 02:41.38 | ChrisInSydney | you can have real time SIP but your dial plan in extension.conf |
| 02:41.52 | ChrisInSydney | that was what I was refering to |
| 02:42.02 | ChrisInSydney | for example |
| 02:43.58 | *** part/#asterisk rue_house (~rue@h24-207-19-104.cst.dccnet.com) |
| 02:45.22 | kessius | ChrisInSydney, my SIP is already realtime, my extensions is already realtime |
| 02:48.20 | volga629 | How I can debug /usr/sbin/safe_asterisk: line 145: 5904 Segmentation fault (core dumped) nice -n $PRIORITY ${ASTSBINDIR}/asterisk -f ${CLIARGS} ${ASTARGS} > /dev/${TTY} 2>&1 < /dev/${TTY} |
| 02:48.20 | volga629 | Asterisk ended with exit status 139 |
| 02:50.36 | mjordan | volga629: https://wiki.asterisk.org/wiki/display/AST/Getting+a+Backtrace |
| 02:51.33 | *** join/#asterisk Maliuta (~nobusines@kiev.lusan.id.au) |
| 02:52.57 | kessius | ChrisInSydney after referencing the voicemail.conf - where else change? |
| 02:53.04 | volga629 | thank mjordan |
| 02:53.23 | volga629 | Just can't see why is crash it |
| 02:56.05 | ChrisInSydney | volga629: Its not a hardware issue, is it ?? Had a faulty mainboard that used to do that to me. Also had OS crashes too though |
| 02:57.02 | volga629 | I am think some module I rebooted server and ran diagnostic all ok |
| 02:57.28 | ChrisInSydney | kessius: MAybe start here:http://www.voip-info.org/wiki/view/Asterisk+RealTime+Voicemail |
| 02:58.22 | volga629 | Before reboot I so this crash segfault at 3 ip 00000003 sp bfac14bc error 4 in libodbcinst.so.1.0.0[101000+12000] |
| 02:58.32 | ChrisInSydney | volga629: what version of ast are you running ?? |
| 02:58.47 | volga629 | 10 |
| 02:58.52 | ChrisInSydney | ahh |
| 02:59.02 | ChrisInSydney | have fun ;-) |
| 02:59.46 | ChrisInSydney | volga629: its the blood of people like you that make it possible for people like me to enjoy open source software that works |
| 03:00.07 | volga629 | where usually stored dump file ? |
| 03:00.38 | volga629 | this test box in production all ok ;-) |
| 03:00.41 | ChrisInSydney | Mate, I am not too sure. find / -name core ? |
| 03:01.51 | lanning | core files are usually in the programs current working directory |
| 03:01.52 | kessius | ChrisInSydney had already seen this link voip-info, but does not explain everything, that is so fraqmento |
| 03:02.39 | ChrisInSydney | volga629: found your previous post. looks like some odbc issues, are you using odbc at all ? |
| 03:04.03 | volga629 | yes connector to MYSQL |
| 03:04.10 | ChrisInSydney | kessius: Asterisk RT info is pretty fragmented. I personally have not done much, (anything worthwhile speaking about) with real time otehr than using the ODBC drivers |
| 03:04.35 | ChrisInSydney | I can not really help you other than to point you at places I vaguely remember stuff on Asterisk real time. |
| 03:05.51 | volga629 | I found core going what crushed |
| 03:05.56 | volga629 | see |
| 03:06.23 | ChrisInSydney | kessius: All I can suggest is that google is your friend, doble check your settings, and change one thing at a a time and test. When you work it out, document what you have done and post it in the forums or on a blog so when you need to do it again in 18 months time, and you have forgotten, you get to remind yourself of how clever you once were ;-) |
| 03:07.07 | *** join/#asterisk blizzow (~jburns@67.50.165.58) |
| 03:07.11 | ChrisInSydney | volga629: have you tried against a 1.8 compile ?? As a reference ? |
| 03:07.24 | ChrisInSydney | eliminates the ODBC drivers |
| 03:07.53 | volga629 | Yes I tried 1.8.9.2 and was working no issue |
| 03:08.07 | ChrisInSydney | so what about an earler 10 release ? |
| 03:08.10 | ChrisInSydney | same ?? |
| 03:10.30 | kessius | ChrisInSydney I will check these issues |
| 03:11.11 | ChrisInSydney | kessius: Good luck. When you work it out, it will become obvious. It always does. |
| 03:11.16 | ChrisInSydney | :] |
| 03:11.24 | volga629 | all point for 10 because of video support app conference |
| 03:16.07 | *** join/#asterisk Juggie (~Juggie@unaffiliated/juggie) |
| 03:17.42 | ChrisInSydney | volga629: Maybe you need some #asterisk-dev help to get them some debug info. Something feels broken in the code |
| 03:19.54 | volga629 | This the who need to blame codec_g729a.so.broken |
| 03:20.06 | volga629 | debug always work |
| 03:20.22 | volga629 | some reason broken need reinstall |
| 03:21.19 | *** part/#asterisk postconf (~postconf@msfree.com) |
| 03:21.25 | volga629 | Thank you everybody |
| 03:34.46 | *** join/#asterisk pdtpatr1ck_ (~pdtpatric@12.249.4.226) |
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| 04:29.25 | *** join/#asterisk reenigne_esrever (~reenignee@static-50-53-151-158.bvtn.or.frontiernet.net) |
| 04:29.43 | reenigne_esrever | Hello |
| 04:30.23 | reenigne_esrever | I'm looking to see if anyone has any feedback on the digium training or is thinking of attending the upcoming training in Vegas. |
| 04:30.39 | *** join/#asterisk gajini (~root@61.12.17.171) |
| 04:36.13 | reenigne_esrever | quiet room again tonight, I'll have to try again tomorrow to see if anyone has any feedback |
| 05:01.14 | *** join/#asterisk ferdna (~ferdna@cpe-67-10-220-35.elp.res.rr.com) |
| 05:17.50 | phix | wb ChrisInSydney! |
| 05:18.01 | ChrisInSydney | hey |
| 05:18.08 | ChrisInSydney | hows the shire ?? |
| 05:18.14 | phix | How was the madigras? |
| 05:18.23 | phix | Shire is still kicking ass :) |
| 05:18.39 | ChrisInSydney | cant remember, woke up witha sore arse and $100 in my pocket ;-) |
| 05:18.45 | phix | heh |
| 05:18.57 | ChrisInSydney | just kidding |
| 05:19.12 | ChrisInSydney | did a gig out at Mulgoa oin Sunday |
| 05:19.46 | ChrisInSydney | They were waiting for the Warragamba dam to spill over |
| 05:21.10 | ChrisInSydney | gotta make a call. Pissed off client. Not my doing, the vendor did it and I have to take it, and I won't have $100 in my pocket at the end of it either |
| 05:21.49 | ChrisInSydney | X| |
| 05:21.58 | ChrisInSydney | back in a bit |
| 05:36.23 | *** join/#asterisk mchou (~quassel@unaffiliated/mchou) |
| 05:46.50 | ChrisInSydney | phix: ouch |
| 05:47.48 | ChrisInSydney | I'm about to rip it all out and throw in an Asterisk in the interum. Keep them happy until they fix this hardware |
| 06:32.48 | phix | nice |
| 06:33.10 | phix | asterisk <3 |
| 06:34.43 | *** join/#asterisk Faustov (~fst@gentoo/user/faustov) |
| 06:38.10 | *** join/#asterisk gerhard7 (~gerhard7@82-168-115-40.ip.telfort.nl) |
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| 06:55.39 | *** join/#asterisk v0lZy (~chatzilla@mail.silk-group.net) |
| 06:55.45 | v0lZy | hi |
| 07:03.06 | dym | hi v0lZy |
| 07:04.07 | v0lZy | hello |
| 07:08.26 | *** join/#asterisk roham (~ali@31.184.187.2) |
| 07:12.13 | *** join/#asterisk jkroon (~jkroon@dsl-244-38-199.telkomadsl.co.za) |
| 07:15.12 | v0lZy | how to mute the caller's line? |
| 07:15.44 | v0lZy | I'm writting an intercom application and I want to mute the caller, page the phones, play a beep to the caller, unmute the caller. |
| 07:21.34 | v0lZy | i see theres a muteaudio function |
| 07:21.44 | v0lZy | but I dont exactly understand how to use it in the dialplan application |
| 07:23.07 | v0lZy | using set? |
| 07:24.07 | *** join/#asterisk schmidts (~schmidts@Lo20-rt01.lm33.sil.at) |
| 07:24.09 | schmidts | good morning |
| 07:27.10 | v0lZy | morning |
| 07:31.41 | *** join/#asterisk fisted (~fisted@unaffiliated/fisted) |
| 07:32.14 | *** join/#asterisk qakhan (~qakhan@203.130.22.202) |
| 07:32.17 | qakhan | hi all |
| 07:33.35 | qakhan | i have 4 agents in a queue. every thing is working fine. here i have a problem, if agent 1001 login it should not login on any other station |
| 07:34.27 | qakhan | how can i restict agent to be loggedin twice |
| 07:51.15 | *** join/#asterisk BarthezZ (~bart@2001:41d0:2:9d0c::2) |
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| 08:20.26 | schmidts | does anyone use a Cisco 5400 media gateway? |
| 08:20.42 | *** join/#asterisk timahvo1 (~rogue@41.80.32.166) |
| 08:25.13 | *** join/#asterisk qakhan (~qakhan@203.130.22.202) |
| 08:25.16 | qakhan | hi all |
| 08:26.54 | qakhan | i have 4 agents in a queue. all agents can multiplelogin in a queue. i set multiplelogin=no in agents.conf but agents still can login multiple times |
| 08:28.53 | kaldemar | multiplelogin is to prevent a single device from logging in as more than one agent. it's the opposite of what you want. |
| 08:31.23 | qakhan | kaldemar i want to setup if agent 1001 is login then no one can login using 1001. they should use other agent like 1002, 1003, 1004 |
| 08:32.07 | kaldemar | you can use func AGENT to see if the agent is already logged in in your dialplan. $["${AGENT(status)}" = "LOGGEDIN"]?hangup:login |
| 08:32.11 | kaldemar | core show function AGENT |
| 08:33.21 | kaldemar | oops. ${AGENT(status)} should be ${AGENT(agentid:status)}. |
| 08:33.54 | qakhan | where i put this? |
| 08:34.43 | qakhan | in which context? in queue context? |
| 08:34.44 | kaldemar | in the extension that your agents use to log in. |
| 08:34.58 | kaldemar | it's your dialplan, how could i know? |
| 08:35.27 | ChannelZ | c'mon, Miss Cleo |
| 08:35.39 | qakhan | [agent] |
| 08:35.39 | qakhan | exten => _1XXX,hint,SIP/${EXTEN} |
| 08:35.39 | qakhan | exten => _1XXX,n,Dial(SIP/${EXTEN},,t) |
| 08:35.39 | qakhan | exten => _1XXX,n,Hangup |
| 08:36.16 | qakhan | it is my agent context. now where i put ${AGENT(agentid:status)} |
| 08:36.51 | kaldemar | agents log in using app AgentLogin. |
| 08:37.02 | kaldemar | so that is not the right place. |
| 08:38.20 | qakhan | sir plz help me how i can do this |
| 08:38.21 | kaldemar | also ${AGENT(agentid:status)} only shows the status of an agent by id "agentid". you need to check the status with GotoIf or IF and make a decision on what to do. |
| 08:38.34 | kaldemar | also, use a pastebin. do not paste configs on the channel. |
| 08:38.43 | qakhan | ok |
| 08:38.48 | qakhan | help me |
| 08:38.57 | kaldemar | i have given you all the information you need. |
| 08:39.39 | qakhan | i am new in asterisk that y i am asking you the exact location and syntex |
| 08:41.32 | kaldemar | i already told you which place to look for. |
| 08:42.27 | qakhan | i m asking in which context i put it in |
| 08:42.33 | qakhan | queue or agent? |
| 08:42.47 | kaldemar | i don't know what your contexts look like. |
| 08:45.00 | *** join/#asterisk Azrael808 (~peter@212.161.9.162) |
| 08:46.42 | qakhan | kaldemar here is my agent and queue context http://pastebin.com/ApB7Bs1B |
| 08:49.03 | kaldemar | i told you to look for app AgentLogin. neither of those have it. |
| 08:49.42 | qakhan | can i modify agnetlogin app? |
| 08:49.56 | kaldemar | i give up. |
| 08:50.03 | kaldemar | ~book |
| 08:50.03 | infobot | Asterisk: The Definitive Guide, 3rd Edition (ISBN 0-596-51734-3) available at http://oreilly.com/catalog/9780596517342 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and is available for reading online at http://www.asteriskdocs.org/ or see ~buybook |
| 08:51.01 | kaldemar | read that until you know what a dialplan is, what a dialplan aplication is and what a dialplan function is. i can't hold your hand through all of that. |
| 08:52.08 | *** join/#asterisk Nasga (~Nasga@82.113.117.78.rev.sfr.net) |
| 09:02.59 | *** join/#asterisk felimwhiteley (~quassel@089-101-203026.ntlworld.ie) |
| 09:22.14 | *** join/#asterisk jzaw (~jzaw@macbook.dzki.co.uk) |
| 09:22.23 | aurs | gives kaldemar a pat on the shoulder |
| 09:22.26 | aurs | :P |
| 09:28.44 | *** join/#asterisk gain_ (~gain@mail.ufficyo.com) |
| 09:31.32 | *** join/#asterisk screenn (~screenn@178.151.86.196) |
| 09:34.51 | *** join/#asterisk tzafrir_laptop (~tzafrir@local.xorcom.com) |
| 09:40.14 | *** join/#asterisk oej (~olle@static-213-115-251-100.sme.bredbandsbolaget.se) |
| 09:43.58 | *** join/#asterisk sekil (~sekil@78.24.104.73) |
| 09:51.44 | kaldemar | aurs: i'll survive. :) |
| 09:57.25 | v0lZy | Hi kaldemar |
| 09:57.28 | v0lZy | quick question |
| 09:58.24 | v0lZy | this is my page part of the application |
| 09:58.48 | v0lZy | 1,Answer(100) |
| 09:58.50 | v0lZy | n,Playback(beep) |
| 09:58.51 | v0lZy | n,SIPAddHeader(Call-Info: sip:askoziapbx.silknet.local\;answer-after=0) |
| 09:58.53 | v0lZy | n,Page(SIP/00&SIP/10&SIP/12&SIP/14&SIP/16&SIP/18&SIP/19&SIP/20&SIP/21&SIP/22&SIP/23&SIP/25&SIP/31&SIP/44&SIP/45&SIP/47&SIP/48&SIP/50&SIP/61&SIP/62&SIP/64&SIP/65&SIP/68&SIP/69,qis) |
| 09:59.20 | v0lZy | is there any better way to do this with page? (something cleaner?) |
| 09:59.56 | cr_ellis | thanks for the help yesterday.. i did it by taking epoch before and after to determine if recordings were silent |
| 10:00.09 | cr_ellis | seems to work much better |
| 10:11.51 | *** join/#asterisk jacc0 (~jacc0@D522448D.static.ziggozakelijk.nl) |
| 10:12.47 | *** join/#asterisk bjornts (~BTS@it010226.klientdrift.uib.no) |
| 10:12.52 | ChrisInSydney | v0lZy: Nope. Maybe use a dial plan variable |
| 10:13.32 | ChrisInSydney | exten => Pageall,n,Page($PAGELIST}) |
| 10:26.53 | v0lZy | and i do |
| 10:27.03 | v0lZy | Set($PAGELIST=whathat?) |
| 10:27.56 | ChrisInSydney | no $ Set(PAGELIST="stuff") |
| 10:28.13 | ChrisInSydney | no $ |
| 10:28.20 | ChrisInSydney | Set(PAGELIST="stuff") |
| 10:28.39 | ChrisInSydney | should be a fullstop there. |
| 10:28.48 | ChrisInSydney | :-/ |
| 10:31.20 | ChrisInSydney | Hey, Anyone seen one of these ?? [Mar 8 21:31:19] WARNING[16724]: res_odbc.c:1355 _ast_odbc_request_obj2: SetConnectAttr (Txn isolation) returned an error: HY000: [MySQL][ODBC 3.51 Driver]MySQL server has gone away |
| 10:31.47 | jacc0 | what asterisk version? |
| 10:31.58 | ChrisInSydney | 1.8.9ish |
| 10:32.19 | jacc0 | there has been a fix in the latest release involving odbc |
| 10:32.35 | jacc0 | btw, I'm using 1.8.9 with odbc |
| 10:32.45 | jacc0 | I have had no problems so far |
| 10:32.56 | ChrisInSydney | just got it then. Reconnected and continued on, but it did crap out on two lines of code before it reconnected |
| 10:33.15 | ChrisInSydney | never seen ot before myself. |
| 10:33.43 | ChrisInSydney | maybe time to bump up as its not in production....yet |
| 10:33.46 | *** join/#asterisk pwnfactory (~adam@247.Red-88-26-194.staticIP.rima-tde.net) |
| 10:34.10 | ChrisInSydney | thanks. I'll look into it |
| 10:35.36 | jacc0 | the ODBC fix is in 1.8.11.0-rc2 . You might want to wait for 1.8.11.0 |
| 10:35.56 | jacc0 | Remove possible segfaults from res_odbc by adding |
| 10:35.56 | jacc0 | <PROTECTED> |
| 10:36.55 | jacc0 | doesn't look like the issue you are talking about |
| 10:37.33 | ChrisInSydney | Thanks. This just looked like a plain old timeout on a MySQL handle. The system has sat idle for a day or three |
| 10:38.55 | v0lZy | fullstop where ChrisInSydney ? |
| 10:39.13 | ChrisInSydney | after the $ and before the Page |
| 10:39.25 | ChrisInSydney | s/Page/Set/ |
| 10:39.49 | v0lZy | $.Set(? |
| 10:40.09 | *** join/#asterisk ollii (~risker@2001:470:1f15:1384:62eb:69ff:fe31:b4) |
| 10:40.50 | ChrisInSydney | not asterisk syntax, English syntax |
| 10:40.59 | ChrisInSydney | forget it |
| 10:41.46 | *** join/#asterisk TimeRider (~steve@host81-136-216-215.in-addr.btopenworld.com) |
| 10:41.48 | ChrisInSydney | v0lZy: What I was saying was no $ in the variable name. Youv'e probably been PHP coding |
| 10:42.41 | v0lZy | started with mIRC, moved to bash |
| 10:42.50 | v0lZy | some php, but dont like it |
| 10:43.27 | ChrisInSydney | its alright, its a bit like losely typed C |
| 10:47.36 | pwnfactory | Would appreciate some help with a Asterisk 1.4.43 problem -> http://pastebin.com/n2GwqBDp |
| 10:49.24 | pwnfactory | Googleing has not be successful... |
| 10:50.29 | *** join/#asterisk FiReSTaRT (~dlyh@unaffiliated/firestart) |
| 11:00.02 | ChrisInSydney | pwnfactory: Looks codec"ish" |
| 11:00.51 | pwnfactory | Could be.. we are recording the call |
| 11:01.12 | pwnfactory | but i'm not sure, because it happens sporadically |
| 11:05.15 | pwnfactory | I've seen this bug report, but it yields little information https://issues.asterisk.org/view.php?id=18744 |
| 11:08.56 | *** join/#asterisk ChrisInSydney (~Chris@60-242-81-231.tpgi.com.au) |
| 11:10.02 | ChrisInSydney | pwnfactory: When you get those messages, does it still record ?? |
| 11:10.09 | ChrisInSydney | ans is he recording OK ?? |
| 11:10.21 | ChrisInSydney | s/ans/and |
| 11:10.43 | ChrisInSydney | s/ans/and/ |
| 11:10.47 | pwnfactory | yes, the calls are recorded. The problem is that the 2 parties cannot hear each other |
| 11:11.22 | ChrisInSydney | but you can hear them on the recording ? |
| 11:11.27 | pwnfactory | yes |
| 11:11.32 | v0lZy | hm |
| 11:11.36 | *** join/#asterisk jsjc (~Adium@226.Red-80-33-236.staticIP.rima-tde.net) |
| 11:11.44 | v0lZy | any idea why it takes a long while before all the phones are paged? |
| 11:11.50 | ChrisInSydney | interesting |
| 11:12.05 | ChrisInSydney | s/interesting/buggered if I know/ |
| 11:12.40 | v0lZy | ChrisInSydney: is there a way to mute and unmute the caller? |
| 11:12.45 | ChrisInSydney | v0lZy; Beacause the page is a sort of conference call |
| 11:13.01 | v0lZy | what I want to do is mute the caller, page all phones, beep the caller, unmute him |
| 11:13.13 | ChrisInSydney | ahh |
| 11:13.20 | pwnfactory | ChrisInSydney: you and i both |
| 11:13.23 | v0lZy | so that he knows when he can speak |
| 11:13.52 | *** join/#asterisk justdave (~dave@unaffiliated/justdave) |
| 11:14.06 | v0lZy | 1,Answer(100) |
| 11:14.08 | v0lZy | n,Playback(beep) |
| 11:14.09 | v0lZy | n,SIPAddHeader(Call-Info: sip:askoziapbx.silknet.local\;answer-after=0) |
| 11:14.11 | v0lZy | n,Page(SIP/00&SIP/10&SIP/12&SIP/14&SIP/16&SIP/18&SIP/19&SIP/20&SIP/21&SIP/22&SIP/23&SIP/25&SIP/31&SIP/44&SIP/45&SIP/47&SIP/48&SIP/50&SIP/61&SIP/62&SIP/64&SIP/65&SIP/68&SIP/69,qis) |
| 11:14.12 | v0lZy | this is what I have so far |
| 11:14.17 | v0lZy | and of course ,the beep is too fast |
| 11:14.35 | ChrisInSydney | v0lZy: how long does it take for the page to work ? If you prune the page list, does it get better ? |
| 11:14.48 | v0lZy | what do u mean prune the page list? |
| 11:15.04 | ChrisInSydney | less SIP clients in the Page ? |
| 11:15.35 | v0lZy | yeah, that seems to be better |
| 11:15.37 | ChrisInSydney | say start with 5, is that much faster ? |
| 11:15.40 | *** join/#asterisk fisted (~fisted@unaffiliated/fisted) |
| 11:15.52 | v0lZy | yeah,it seems to be better if its less |
| 11:16.03 | v0lZy | thats why, i want to do the beep thing with mute |
| 11:16.15 | v0lZy | i dont want the user to start speaking before the intercom has reached everyone |
| 11:16.20 | ChrisInSydney | pwnfactory: are you using g729 ? |
| 11:16.46 | ChrisInSydney | v0lZy; How big is the machine you are running on ? |
| 11:16.50 | v0lZy | so... answer, mute, page, beep, unmute |
| 11:17.06 | v0lZy | pentium 4 or something like that |
| 11:17.16 | v0lZy | lots of ram |
| 11:17.45 | v0lZy | intel celeron 3.2 ghz |
| 11:17.58 | v0lZy | 256 cache |
| 11:18.39 | v0lZy | 3gb of ram. |
| 11:18.52 | v0lZy | supposed to be overkill |
| 11:19.53 | ChrisInSydney | should cope |
| 11:21.03 | v0lZy | theres a noticable delay |
| 11:21.06 | ChrisInSydney | v0lZy: You could try Multicast. |
| 11:21.07 | v0lZy | between the time the caller gets the beep |
| 11:21.11 | v0lZy | and between the intercom is activated. |
| 11:21.26 | ChrisInSydney | ahh |
| 11:21.45 | v0lZy | so what happens is that the user starts early |
| 11:21.50 | v0lZy | and then they dont get the beginning of the intercom |
| 11:21.59 | v0lZy | they being other phoens |
| 11:24.26 | ChrisInSydney | try answer(500) or Answer(1000) |
| 11:24.46 | ChrisInSydney | it allows the audio to negoatiate |
| 11:26.32 | ChrisInSydney | a little better |
| 11:26.57 | v0lZy | is this stuff procedural |
| 11:26.58 | ChrisInSydney | I find that the Snoms take a bit longer for the audio to work |
| 11:26.59 | v0lZy | like |
| 11:27.04 | v0lZy | 1, after 1 2 etc |
| 11:27.09 | v0lZy | or does it all run at the same time? |
| 11:27.11 | ChrisInSydney | yup |
| 11:27.12 | v0lZy | if i have |
| 11:27.15 | v0lZy | 1 anser |
| 11:27.16 | v0lZy | 2beep |
| 11:27.19 | v0lZy | 3 page |
| 11:27.25 | v0lZy | its gonna *start* paging after 2 |
| 11:27.30 | v0lZy | and since paging takes long... |
| 11:27.43 | v0lZy | its gonna beep sooner than itnercoms will be established |
| 11:27.47 | v0lZy | and the user will start speaking too early |
| 11:30.46 | kaldemar | v0lZy: any particular reason for you to use both Playback(beep) and option q for Page? |
| 11:31.59 | v0lZy | that q thing only beeps on the receiver end |
| 11:32.07 | v0lZy | or its not really audiable |
| 11:32.09 | v0lZy | i can tyr, hold on |
| 11:32.20 | kaldemar | no, it beeps the caller. |
| 11:32.27 | v0lZy | hm... it does |
| 11:32.33 | v0lZy | lets see what happens if i remove the beep |
| 11:33.38 | v0lZy | ok |
| 11:33.42 | v0lZy | that seems to work better at the moment |
| 11:33.51 | v0lZy | thanks kaldemar |
| 11:33.52 | v0lZy | orz |
| 11:46.38 | *** join/#asterisk doolittlework (~doolittle@41-134-22-14.dsl.mweb.co.za) |
| 11:46.51 | doolittlework | i just got tagged by hacker |
| 11:47.31 | ChrisInSydney | doolittlework: How so ? |
| 11:47.54 | doolittlework | he is running the application playback(hello1) but i checked all the dialplans and i can not see hello1 anywhere |
| 11:48.12 | doolittlework | he is not registering to my box |
| 11:48.44 | doolittlework | but he makes calls automaticaly, where is that folder where you dump files into to automaticaly dial |
| 11:50.33 | ChrisInSydney | doolittlework: iptables -I INPUT -s ipaddress/bits -j DROP |
| 11:51.05 | ChrisInSydney | eg iptables -I INPUT -s 1.2.3.4/32 -j DROP will stop 1.2.3.4 |
| 11:51.15 | ChrisInSydney | eg iptables -I INPUT -s 1.2.3.4/23 -j DROP will stop 1.2.3.0-255 |
| 11:51.51 | v0lZy | kaldemar: is the beep in Page sent out only after all phones are connected? |
| 11:53.11 | kaldemar | v0lZy: no, but leaving the Playback saves you from |
| 11:54.27 | v0lZy | i just read a sequence of numbers |
| 11:54.32 | v0lZy | and most people got me at 2 |
| 11:54.37 | v0lZy | somone reported hearing me at 4... |
| 11:55.52 | v0lZy | kaldemar, ChrisInSydney : is there a way to code it like answer, mute caller, page phones, after all connections are established, play beep to caller, unmute caller |
| 11:56.11 | kaldemar | saves you from some extra messages and processing |
| 11:57.05 | ChrisInSydney | yes. Have a look at the code in ./apps/app_page.c and go for it ;-) |
| 11:59.07 | v0lZy | uh.. .really, theres no other way |
| 11:59.19 | v0lZy | i mean .... put the user on mute, wait 2 seconds, then release? |
| 11:59.31 | ChrisInSydney | unless you originate a call, and call yourself back |
| 11:59.49 | v0lZy | what do u mean? |
| 12:00.18 | v0lZy | i want tis function to be global for all users.. so i put all my phones into the page call. |
| 12:00.23 | v0lZy | even the one the call originates from |
| 12:00.46 | ChrisInSydney | http://www.voip-info.org/wiki/view/Asterisk+cli+originate |
| 12:01.17 | ChrisInSydney | call the page group, then call the caller, then make him wait, then beep, then he talks |
| 12:01.19 | ChrisInSydney | messy |
| 12:01.48 | ChrisInSydney | but it might work |
| 12:02.07 | ChrisInSydney | otherwise, check the c code, may be some hints in there |
| 12:03.20 | ChrisInSydney | hints = clues as opposed to dialplan subscriptions |
| 16:02.57 | *** join/#asterisk infobot (~infobot@rikers.org) |
| 16:02.57 | *** topic/#asterisk is Asterisk: The Open Source PBX and Telephony Platform (asterisk.org) -=- Asterisk 10.2.0 (2012/03/05), 1.8.10.0 (2012/03/05), dahdi-linux 2.6.0 (2012/01/04), dahdi-tools 2.6.0 (2012/01/04), libpri 1.4.12 (2011/07/06) -=- Visit the official Asterisk wiki: wiki.asterisk.org |
| 16:09.08 | *** join/#asterisk vinhdizzo (~vinh@dhcp-v019-092.mobile.uci.edu) |
| 16:22.12 | leifmadsen | mjordan: you just dropped the maintainers page? |
| 16:23.24 | Qwell | leifmadsen: que? |
| 16:24.04 | leifmadsen | Asterisk Open Source Maintainers Page removed by Matt Jordan |
| 16:24.06 | leifmadsen | email I got |
| 16:24.15 | mjordan | leifmadsen: that was my temporary copy |
| 16:24.28 | leifmadsen | guess it's not very good about showing that you "published" it |
| 16:24.32 | leifmadsen | probably need a cleaner work flow :( |
| 16:24.39 | mjordan | https://wiki.asterisk.org/wiki/display/AST/Asterisk+Open+Source+Maintainers |
| 16:24.49 | mjordan | Yeah, I'm surprised it did since that was on my personal workspace |
| 16:25.02 | leifmadsen | must have been subscribed to it |
| 16:25.04 | leifmadsen | my bad |
| 16:25.18 | mjordan | leifmadsen: I'm just glad I have a subscriber |
| 16:25.26 | leifmadsen | had :) |
| 16:25.29 | leifmadsen | burn! |
| 16:25.31 | mjordan | :-( |
| 16:26.01 | leifmadsen | mjordan: I kid |
| 16:26.06 | leifmadsen | mjordan: I have a serious suggestion though |
| 16:26.41 | leifmadsen | this link: https://wiki.asterisk.org/wiki/display/AST/Asterisk+11+Projects <-- if there are issue numbers/links, it would be ideal that each of the sections have a link back to where the work is being completed (whether in jira, reviewboard, or svn branch) |
| 16:27.15 | mjordan | yes - right now, each task has associated with it some time to put in depth material on each project on this wiki page |
| 16:27.56 | mjordan | so... yeah. People *should* be doing that. For a few of those, the work was already getting pushed forward by the time I had the wiki article up |
| 16:28.21 | leifmadsen | mjordan: no worries, would just be nice to have the ability to easily track the progress from that page of anything in particular someone was interested in |
| 16:37.24 | *** join/#asterisk eelcob (~eelco@2001:67c:26c0:21:7aac:c0ff:fe97:da58) |
| 16:39.54 | eelcob | Is there any chance of asterisk 1.8.10.0 being packaged for debian squeeze? I only see the ubuntu variants... Seems like 1.8.8.1 was the last version packaged for debian. |
| 16:40.11 | *** join/#asterisk jkroon (~jkroon@dsl-244-32-73.telkomadsl.co.za) |
| 16:40.40 | *** join/#asterisk jkroon (~jkroon@dsl-244-32-73.telkomadsl.co.za) |
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| 16:48.06 | *** join/#asterisk waschtl (~waschtl@216.129.239.149) |
| 16:56.26 | *** join/#asterisk salz212 (~chatzilla@182.178.148.198) |
| 16:59.27 | salz212 | I am finding it very difficult to understand the limitation of Asterisk servers.....I am using a server class machine having 8 processors and 32 GB RAM for Asterisk IVR setup..... but the call capacity and aterisk process utilization is almots same as a P3 or dual core machine..... Can some one tell me if there is possibility of enhancing the pbx using a quality server or not. |
| 17:00.32 | Qwell | salz212: more hardware == better |
| 17:00.52 | [TK]D-Fender | salz212, Capacity depends on what you're doing and if you are comparing a P3 to a dual core or higher end system then you are doing something very wrong |
| 17:01.09 | [TK]D-Fender | salz212, and you need to look at the procise circumstances of your calls |
| 17:02.53 | salz212 | actually I have noticed that when the asterisk server on 8 processors machine the system utilization remian very nominal where as asterisk utilization seems to be raised. by asterisk utilization i am refering to the asterisk process |
| 17:03.12 | Qwell | What, specifically, is being "raised"? |
| 17:03.44 | salz212 | well, the Asterisk Process CPU and something the QoS |
| 17:03.57 | salz212 | sometimes* |
| 17:04.59 | salz212 | I am decided to go for virtualizationfor the utilization of such heavy servers... but it does not satisfy my concision. |
| 17:06.05 | Qwell | QoS goes up? O.o That doesn't really even make sense. |
| 17:07.11 | salz212 | no it does not go up.. definitely it is compromised. |
| 17:07.26 | Qwell | spell it out - what is happening, exactly? |
| 17:08.03 | salz212 | <PROTECTED> |
| 17:08.50 | salz212 | now I can not make nay changes at the AGI level..... the call volume at one asterisk server at max provides not more than 60 calls. |
| 17:10.11 | salz212 | my concern is to raise the call volume.. but what I want to know is.... is there any limitation of asterisk in terms of number of calls? or not. |
| 17:10.34 | salz212 | is it making any sense? |
| 17:10.47 | Qwell | there are no artificial limits in Asterisk |
| 17:11.38 | salz212 | ok so it is purely dependent on the server quality/specs, right? |
| 17:12.08 | [TK]D-Fender | salz212, "now I can not make nay changes at the AGI level...." <-- you have handcuffed yourself and we have no information of what this is doing. AGI = load an you hav guaranteed we will remain blind. |
| 17:13.20 | salz212 | ah I am a terrible at explaining..things :( |
| 17:13.52 | salz212 | its Java Asterisk which is being used for AGI... |
| 17:14.24 | *** join/#asterisk darkbasic (~quassel@niko.linuxsystems.it) |
| 17:14.44 | *** join/#asterisk SeriousMatters (~Sirius@53.66.113.87.dyn.plus.net) |
| 17:14.47 | Qwell | salz212: What makes you think you're running out of resources? |
| 17:15.24 | salz212 | Asterisk CPU utilization... |
| 17:15.29 | Qwell | Show me. |
| 17:15.48 | salz212 | what? |
| 17:15.54 | Qwell | Show me how you've determined that. |
| 17:16.23 | salz212 | 'top' or just check the asterisk process |
| 17:16.33 | Qwell | okay - show me |
| 17:16.51 | salz212 | what the output of top? |
| 17:16.55 | Qwell | sure |
| 17:18.39 | salz212 | can't at the moment, do not have remove access,, and currently I am not in office.. but what exactly do you want to see in it...that may help me troubleshoot. |
| 17:18.57 | Qwell | The number that makes you think you've hit a resource limit. |
| 17:19.30 | *** join/#asterisk oej (~olle@h87-96-134-129.dynamic.se.alltele.net) |
| 17:20.40 | salz212 | Asterisk Process Percentage goes up to 60-80% where as the system CPU utilization do not even reach 5%... this is the main thing that is making me unable to understand... |
| 17:21.05 | Qwell | what is 100 / 8? |
| 17:21.12 | Qwell | (hint: 12.5) |
| 17:21.16 | Qwell | Now what is 60% of that? |
| 17:22.05 | salz212 | doesen't it show the combined utilization of all CPUs? |
| 17:22.07 | sp00kz | the % being shown, salz212, is % cpu use, not of potential cpu use |
| 17:22.24 | sp00kz | so 1 app using a tiny bit of cpu will show up as 100% |
| 17:22.31 | sp00kz | and 2 apps using a tiny bit will show 2 @ 50% |
| 17:22.55 | Qwell | CPU % is a useless metric. |
| 17:23.17 | salz212 | oh .. okay.. |
| 17:23.31 | salz212 | so how do I check the load on asterisk server other than number of calls? |
| 17:23.43 | Qwell | The load is a better number. |
| 17:23.45 | sp00kz | the load average |
| 17:24.15 | salz212 | load on asterisk process or load on the entire server? |
| 17:24.23 | Qwell | `w` |
| 17:25.27 | salz212 | are you talking about average load on the server? |
| 17:25.35 | sp00kz | yes |
| 17:26.22 | Qwell | If you aren't experiencing actual issues, none of this is even relevant. |
| 17:26.29 | salz212 | but I am interested in Asterisk only.. how much does it takes ... |
| 17:26.30 | Qwell | You're making up ghosts |
| 17:27.09 | salz212 | the issue is call distortion.. which we face when the Asterisk Process Utilization raises up to 80% |
| 17:28.45 | [TK]D-Fender | salz212, You've said nothing about how your calls are arriving in the first place |
| 17:28.58 | *** join/#asterisk r33dtard (~r33dtard@gateway/tor-sasl/r33dtard) |
| 17:29.25 | salz212 | I said QoS is being compromised... |
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| 17:33.47 | Qwell | compromised how? |
| 17:34.05 | Qwell | If your network can't keep up with the traffic, that has nothing to do with Asterisk.. |
| 17:34.10 | salz212 | distortion what else? |
| 17:34.27 | Qwell | what kind of connectivity do you have, to where ever those 60 calls are going? |
| 17:34.42 | *** part/#asterisk salz212 (~chatzilla@182.178.148.198) |
| 17:35.17 | Qwell | alright then |
| 17:35.45 | sp00kz | your network classification settings have been compromised. |
| 17:35.48 | sp00kz | o.O |
| 17:35.50 | *** join/#asterisk salz212 (~chatzilla@182.178.148.198) |
| 17:35.55 | *** join/#asterisk vinhdizzo (~vinh@dhcp-v019-092.mobile.uci.edu) |
| 17:36.00 | Qwell | sp00kz: yeah that's what I'm trying to figure out |
| 17:36.03 | *** join/#asterisk gonewage (~gonewage@72.2.130.205) |
| 17:36.09 | Qwell | salz212: <Qwell> what kind of connectivity do you have, to where ever those 60 calls are going? |
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| 17:37.31 | salz212 | <PROTECTED> |
| 17:37.56 | sp00kz | salz212: what is your upload/download on your primary voip internet link |
| 17:37.57 | Qwell | and what is the size of the pipe from Verizon? |
| 17:38.04 | sp00kz | sorry Qwell |
| 17:38.05 | [TK]D-Fender | salz212, Codecs? Conversion? Recording? conferencing? |
| 17:38.31 | salz212 | 12 MB .. codes g711 and g729 .. currently we are using g711 for simplicity. |
| 17:39.04 | salz212 | recording yes.. through Mix Monitor in Java AGI .. conferencing NO |
| 17:39.20 | Qwell | 12 up and down? |
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| 17:40.13 | salz212 | 12up 8 down .. dedicated.. but we also have IM running through this.. but never seen any abruptness in bandwidth uilization graphs |
| 17:40.30 | [TK]D-Fender | salz212, mixmonitor + G.729 = LOAD |
| 17:40.37 | sp00kz | if you have sysstat installed can you pastebin the output of the linux command: sar |
| 17:40.46 | [TK]D-Fender | salz212, megaBYTE, or megaBIT? |
| 17:40.54 | salz212 | MB |
| 17:41.16 | salz212 | MBytes |
| 17:41.53 | salz212 | mixmonitor + G.729 = LOAD .. didnt know that .. but currently we have switched to 64kHz g711 |
| 17:41.57 | Qwell | 60 * 80KB = 4800KB |
| 17:42.23 | Qwell | 4.8MB on an 8MB pipe is not an unreasonable number. |
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| 17:43.42 | salz212 | hmm.. |
| 17:44.06 | Qwell | my math isn't mathing. |
| 17:44.12 | Qwell | You can safely ignore me. |
| 17:44.13 | leifmadsen | heh |
| 17:44.30 | salz212 | okay leave this... please assist me in just one thing that would help a lot... |
| 17:44.32 | Qwell | I doubt you have an 8MB upstream. |
| 17:44.40 | leifmadsen | 8Mbit |
| 17:44.43 | leifmadsen | Mb, not MB |
| 17:44.49 | Qwell | leifmadsen: yes |
| 17:44.56 | salz212 | MB Comcast |
| 17:45.02 | leifmadsen | 8MB would be like 64Mb |
| 17:45.06 | leifmadsen | approx |
| 17:45.38 | salz212 | there are other VPNs also with different Banks on it.. |
| 17:45.48 | Qwell | first it's Verizon, now Comcast? O.o |
| 17:46.01 | sp00kz | salz212: please run a speedtest.net and show us the results for your voip link |
| 17:46.19 | Qwell | sp00kz: You and your logic. |
| 17:46.26 | [TK]D-Fender | other VPN's? Who does this relate to actual usable bandwidth? #VPN's != # of internet connections or bandwadth. |
| 17:46.39 | salz212 | no from Verizon we only have TF and DID setuyp and SIP gateway.. fro IP address and Internet we have comcats,.. and IPLS another vendor. |
| 17:47.51 | salz212 | it will take atleast and hour to give you just an over view of the current of this company I am working with.. |
| 17:48.21 | Qwell | Figure out where the bottleneck is. It's not Asterisk. |
| 17:48.30 | leifmadsen | that ^^ |
| 17:48.40 | leifmadsen | it's a networking/management issue, not layer 7 |
| 17:48.48 | salz212 | dont tell me its me :P |
| 17:48.52 | [TK]D-Fender | LAYER 0 |
| 17:48.57 | leifmadsen | it's a layer 8 problem |
| 17:49.17 | leifmadsen | layer 8 isn't configuring layers 1 through 4 correctly |
| 17:50.21 | salz212 | how to troubleshoot ?..apparently there is nothing wrong other than voice quality. |
| 17:51.02 | [TK]D-Fender | salz212, First.. you are virtualizing * and complaining about quality. This is typically a "shoot on sight offense" |
| 17:51.19 | [TK]D-Fender | salz212, Please stand on that plastic sheet over there in the corner first... |
| 17:51.22 | *** join/#asterisk gonewage (~gonewage@72.2.130.205) |
| 17:51.33 | salz212 | OMG.. I am not virtualizing ..:( |
| 17:51.35 | Qwell | salz212: If the network is as complicated as you say it is - a competent network engineer would need to troubleshoot. |
| 17:51.52 | [TK]D-Fender | " I am decided to go for virtualizationfor the utilization of such heavy servers... but it does not satisfy my concision." |
| 17:52.09 | [TK]D-Fender | Please clarify then as this appears to be in conflict |
| 17:53.07 | salz212 | yes I said by using such pecs servers we are unable to generate the required call capacity.. so we are thinking of virtualization so use the max out of such server calss machines. |
| 17:53.42 | salz212 | specs* |
| 17:53.43 | Qwell | The servers are not the problem. |
| 17:55.52 | salz212 | thanks guys .. |
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| 19:24.32 | mogra | looking into setting up a home pbx virtual appliance and can't seem to find where to get a public phone number for non SIP users to dial in to. |
| 19:31.23 | Qwell | ~itsplist-us |
| 19:31.24 | 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 |
| 19:31.34 | Qwell | mogra: pick one |
| 19:32.15 | mogra | thank you |
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| 19:49.53 | *** join/#asterisk enoch (~enoch@unaffiliated/enoch) |
| 19:49.56 | enoch | hi all |
| 19:50.27 | enoch | i have a pbx (freepbx) with isd INPUT and a tdm410p with 2fxs |
| 19:50.37 | enoch | 1 of the fxs is in group with 3 sip client |
| 19:51.08 | enoch | when someone of the sip answer the call the fxs device continues ringing for 3-4 times |
| 19:51.13 | enoch | how can i fix it? |
| 19:51.43 | Qwell | enoch: You should ask in #freepbx |
| 19:51.57 | enoch | freepbx runs asterisk |
| 19:52.06 | enoch | and freepbx is only a web guy |
| 19:52.18 | Qwell | ~freepbx |
| 19:52.18 | 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 |
| 19:52.30 | enoch | this is a more "technical" question |
| 19:52.33 | enoch | ok |
| 19:53.23 | *** join/#asterisk fisted (~fisted@unaffiliated/fisted) |
| 19:53.30 | thehar | whoa |
| 19:54.18 | enoch | yep but what controls the ring time in asterisk/dahdi? |
| 19:56.15 | Qwell | enoch: Your configuration. |
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| 20:28.19 | cmendes0101 | When creating meetme in marked A mode. Does that mean that user is the marked user and if another user joins with x they close when the marked user leaves? |
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| 20:36.56 | WindBack | Anybody has experience making work Cisco 3905 phone in Asterisk. I can't find any documentation in the web |
| 20:37.01 | WindBack | ? |
| 20:47.33 | [TK]D-Fender | WindBack, Crappy looking phone... You're already stuck with them? |
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| 21:04.16 | p3nguin | I'm almost stumped on a problem... |
| 21:04.30 | p3nguin | I have an extension which dials two devices simultaneously... |
| 21:05.13 | WindBack | [TK]D-Fender: yes :( |
| 21:05.17 | p3nguin | I need to use GROUP and GROUP_COUNT to ensure that only one call can be active on the two phones at any given time. |
| 21:05.25 | p3nguin | Not one call per phone, but one call total. |
| 21:05.38 | WindBack | [TK]D-Fender: I have to make it work with asterisk. My boss want this |
| 21:06.03 | [TK]D-Fender | WindBack, Probably provisions just like the rest. Go hit their docs |
| 21:06.20 | p3nguin | If the person picks up phone A and dials outbound, and then someone calls the extension which Dials() both phone A and phone B, I need GROUP_COUNT to reflect the call and I'll use GotoIf() to send the calls to vm. |
| 21:06.24 | WindBack | [TK]D-Fender: thanks |
| 21:06.53 | p3nguin | But what "item" do I need to use to check for calls both out of either phone AND calls into the extension (which dials both phones)? |
| 21:07.21 | [TK]D-Fender | p3nguin, name the group. the count should be trackable by that. |
| 21:08.12 | p3nguin | I don't have a problem with that part. My problem is how to track both phones having calls in both directions. |
| 21:08.20 | p3nguin | On the extension, I can check for calls to it. |
| 21:08.44 | p3nguin | On the calls out, I don't know what to check for. |
| 21:09.17 | p3nguin | If I check for CHANNEL, how do I track that for calls going *to* the extension or the phones? |
| 21:09.18 | [TK]D-Fender | the same counter |
| 21:09.53 | p3nguin | The extensions that the phones can dial can be dialed by all of the other phones as well. |
| 21:10.05 | p3nguin | I wasn't wanting to create a context just for these two phones to dial out. |
| 21:10.10 | [TK]D-Fender | lots of code to split around... |
| 21:10.18 | [TK]D-Fender | what you have to do... |
| 21:11.25 | p3nguin | If I created a special context just for those two phones, that would solve it, but I would have to duplicate the code or use local channels which will dial the other extensions in the regular outgoing context. I don't really want to do either of those -- I don't want a special context for a single pair of phones which share an extension. |
| 21:14.23 | [TK]D-Fender | p3nguin, Because it's a pair you've got some work to do. if it was just one you could limit at the SIP level in that case. |
| 21:14.28 | [TK]D-Fender | (if SIP) |
| 21:14.40 | [TK]D-Fender | But separate = yup, unfortunate pain |
| 21:14.44 | p3nguin | It's SIP, but the problem is the "two phone" configuration. |
| 21:15.00 | p3nguin | I have an idea. It might be clunky, but it might work. |
| 21:16.44 | p3nguin | I hate having to hard-code channel names, but I think I can make it work if I do it. |
| 21:17.00 | *** part/#asterisk matach (~matach@84-75-0-151.dclient.hispeed.ch) |
| 21:21.03 | p3nguin | Let's say it's just one phone and I need to check for the the number of active calls either to or from before trying to dial it. |
| 21:21.30 | p3nguin | I have no trouble doing it for calls in one direction only. But how do I check for both to and from? |
| 21:21.50 | *** join/#asterisk matach (~matach@84-75-0-151.dclient.hispeed.ch) |
| 21:22.08 | p3nguin | For to, I'd use extension. For from, I would use the channel name. |
| 21:23.32 | [TK]D-Fender | checkout time, BBL |
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| 21:40.46 | gunnarar | I'm getting /dev/dahdi/1 absent: Device or resource busy when I run fxotune, I have a digium card with echo cancelling module that I'm trying to activate |
| 21:41.00 | gunnarar | do you know what gives? |
| 21:42.40 | gunnarar | apparently you need to stop asterisk before running it... |
| 21:42.56 | gunnarar | have a good day :) |
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| 22:33.01 | fprior | hi, how debug a call for comprehend if there is a codec translation ? |
| 22:48.25 | p3nguin | fprior: Don't debug it. Check with "sip show channels" and see if the two phones involved in the call are using the same codec or different codecs. |
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| 23:06.35 | fprior | p3nguin: thanks |
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| 23:20.56 | volga629 | srtp continue crash on asterisk 10 |
| 23:20.59 | *** join/#asterisk citywok (~kvirc@67-134-194-33.dia.static.qwest.net) |
| 23:21.25 | volga629 | Is really so unstable ? |
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| 23:38.41 | kessius | hi, need help - how to install skype in asterisk 1.8 - ___ skype worked as a gateway ___ |
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| 23:50.01 | kessius | will skype trunk inbound and outbound , __ skype does not use sip __ how to overcome this problem |