| 00:08.59 | ChannelZ | Skype is just for masturbating on camera anyway. Or is that MSN? I forget. |
| 00:10.09 | cusco | wasn't that netmeeting? |
| 00:10.41 | *** join/#asterisk TimeRider (~steve@timerider.plus.com) |
| 00:14.28 | [TK]D-Fender | ChatRoulette <- |
| 00:26.37 | *** join/#asterisk pabelanger (~pabelange@asterisk/contributor-and-bug-marshal/pabelanger) |
| 00:26.37 | *** mode/#asterisk [+o pabelanger] by ChanServ |
| 00:43.25 | leifmadsen | themrrobert: where did you see "reload acl" ? |
| 00:43.55 | leifmadsen | dlynes: unknown at this point -- we're still finishing up the draft, then when we deliver that, probably another 2-3 months by the time it is shipping |
| 00:44.01 | leifmadsen | so I expect... May-ish |
| 00:50.02 | jeffspeff | is there another method to specify ami users other than in the flat file? like db, odbc, ldap? |
| 00:55.37 | *** join/#asterisk bebe-lala (~big_ben@ip216-239-66-30.vif.net) |
| 01:12.53 | leifmadsen | jeffspeff: sure, you could use static realtime |
| 01:21.29 | *** join/#asterisk youjelly (~bwahahaha@39.47.224.83) |
| 01:21.59 | youjelly | hey guys, I wanted to know if I can load share on a SIp trunk that is defined on 2 asterisk servers |
| 01:22.27 | youjelly | or would I need a dedicated SIP proxy |
| 01:23.02 | WIMPy | As long as those two servers can reach each other. |
| 01:26.22 | youjelly | yeah WIMPy they are going to be part of the same set-up |
| 01:26.42 | youjelly | running the same IVR |
| 01:27.00 | youjelly | I just need load sharing implemented on SIP, over TDM its simple |
| 01:27.31 | youjelly | you just need to connect PRIs to each server and the Switch balances it for you |
| 01:28.08 | WIMPy | So you want it for incomming calls? |
| 01:28.16 | youjelly | yeah |
| 01:28.35 | WIMPy | Then you need to talk to your ITSP. |
| 01:28.47 | WIMPy | Ob build something in between. |
| 01:29.23 | youjelly | That was my first guess too... |
| 01:29.35 | youjelly | Thought I'd ask anyway |
| 01:30.44 | youjelly | ldirector might help |
| 01:34.06 | *** join/#asterisk youjelly (~bwahahaha@39.47.224.83) |
| 01:34.10 | youjelly | meh |
| 01:34.25 | youjelly | so either ldirectord or HAProxy |
| 01:35.09 | WIMPy | I'm not sure how much sense that makes on your end. |
| 01:35.14 | youjelly | will have to give it a try |
| 01:35.21 | WIMPy | Or do you have more load than one box can handle? |
| 01:35.41 | youjelly | can I PM? |
| 01:35.56 | WIMPy | You shouldn't. |
| 01:36.20 | youjelly | ok |
| 01:36.20 | WIMPy | I'm not always fast to respond and others can help as well. |
| 01:36.53 | youjelly | naa I wasn't asking for help |
| 01:37.04 | youjelly | just telling you about what we're planning on deploying |
| 01:37.44 | WIMPy | And you don;t want imput fromothers, how might have more to say about it? |
| 01:38.06 | *** join/#asterisk kresp0 (~kresp0@81.61.24.25.dyn.user.ono.com) |
| 01:38.12 | youjelly | Its irrelevant to this conversation |
| 01:38.18 | WIMPy | (Bad typing day today) |
| 01:38.37 | youjelly | What I need help on |
| 01:41.09 | *** join/#asterisk deo_ (~deo@58.71.19.178) |
| 01:57.18 | dlynes | leifmadsen, I guess it's already available on safaribooksonline.com, though? |
| 01:57.55 | mnathani | Does Asterisk provide addons that can generate Call Detail Records |
| 01:59.44 | WIMPy | Yes the CDR modules. |
| 02:00.19 | *** join/#asterisk bmg505 (~leon@196-209-120-100.dynamic.isadsl.co.za) |
| 02:00.21 | leifmadsen | dlynes: it's available as a rough edits probably |
| 02:00.28 | leifmadsen | dlynes: and ofps.oreilly.com has it up for review right now |
| 02:00.41 | leifmadsen | we try to thank everyone who contributes useful reviews |
| 02:02.29 | *** join/#asterisk rue_house (~rue@24-207-103-226.eastlink.ca) |
| 02:02.33 | rue_house | popularity votes? tftpd vs atftpd vs tftpd-hpa ? |
| 02:14.16 | adeel | is it possible to have multiple voicemail backends configured and or used simultaneously? |
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| 02:18.53 | *** join/#asterisk rue_house (~rue@24-207-103-226.eastlink.ca) |
| 02:18.55 | rue_house | erp |
| 02:19.06 | rue_house | anyone installed asterisk on a wrt54g? |
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| 03:02.24 | jpsharp | I use tftpd-hpa, it deals with NAT better. |
| 03:03.11 | WIMPy | ? |
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| 03:54.14 | [TK]D-Fender | adeelis it possible to have multiple voicemail backends configured and or used simultaneously? <- nope |
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| 05:08.32 | kakain | So im trying to set up a home system, I went out and got a DiD provider, all incoming call are going to context 'default', but I dont see where iv told asterisk to send calls to that context |
| 05:08.52 | kakain | can anyone advise? |
| 05:11.30 | *** join/#asterisk Nugget (~nugget@rennsport.macnugget.org) |
| 05:11.34 | sruffell | kakain: DiD provider is sending calls to you via sip or iax? |
| 05:14.36 | kakain | sip |
| 05:14.49 | kakain | i think i just figured it out |
| 05:14.55 | sruffell | look in /etc/ …. cool... |
| 05:15.11 | kakain | i had it pointed at the wrong ip |
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| 05:17.23 | dlynes | leifmadsen, cool...thanks |
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| 08:05.06 | *** join/#asterisk BorjaGVO (d51beb92@gateway/web/freenode/ip.213.27.235.146) |
| 08:06.34 | BorjaGVO | Hi everyone. I want to get rid of the content in queue_log under /var/log/asterisk. I don't know if deleting the file and creating a new one with same permissions is sufficient...should I just do that? |
| 08:07.24 | kaldemar | what if you just delete the file? |
| 08:08.53 | BorjaGVO | kaldemar: yeah, but just deleting it would Asterisk create a new one? |
| 08:11.41 | kaldemar | did you try it? |
| 08:12.53 | ChannelZ | try? there is no try! |
| 08:13.11 | *** join/#asterisk creativx (~creadurex@226.62-97-205.bkkb.no) |
| 08:14.44 | ChannelZ | lsof |grep aster |grep log |
| 08:15.19 | ChannelZ | Might show you if it's keeping a file handle open. In which case you probably need to 'logger reload' or something to get it to behave if you mess with them behind it's back. |
| 08:17.57 | BorjaGVO | logger reload Reopens the log files |
| 08:18.38 | BorjaGVO | if I do lsof |grep aster |grep log: asterisk 25501 root 20u REG 202,65 8408257 279692 /var/log/asterisk/queue_log |
| 08:18.38 | ChannelZ | yes |
| 08:19.27 | ChannelZ | so if you delete the log something unexpected will probably happen since the file handle Asterisk has is not exactly valid. |
| 08:19.58 | BorjaGVO | So, if I want to left blank the queue_log? 1. Delete queue_log and make "logger reload"? |
| 08:20.51 | BorjaGVO | 1. Delete queue_log 2. execute "logger reload".. |
| 08:20.56 | ChannelZ | If you want to empty it and have it continue writing log entries to a new one, yes |
| 08:22.12 | ChannelZ | If you have exceptionally busy queues it's also entirely possible you might lose a line or two in the interum but I'm guessing it's not terribly important if you're deleting the old one in the first place. |
| 08:23.26 | BorjaGVO | yeah, you're right |
| 08:23.29 | BorjaGVO | ok, thanks very much |
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| 10:56.37 | ghost75 | when i am using pap2t, is the busy signal (the audio tone) coming from asterisk or pap2t ? |
| 10:58.20 | *** join/#asterisk hebber (~hebber@node-14pe.pool-125-25.dynamic.totbb.net) |
| 11:02.15 | kaldemar | ghost75: depends on what is really happening, but most likely the pap2t. |
| 11:07.10 | *** join/#asterisk youjelly (~bwahahaha@39.47.99.132) |
| 11:07.18 | youjelly | WIMPy: I found this http://stackoverflow.com/questions/1112191/asterisk-load-balancing-using-openser-opensips |
| 11:07.27 | youjelly | \o/ hope is restored |
| 11:27.28 | *** join/#asterisk italorossi (~italoross@187.60.66.11) |
| 11:35.34 | ghost75 | if the dialed person is busy |
| 11:35.57 | ghost75 | the tone sounds very ugly |
| 11:46.35 | hebber | exit |
| 11:49.52 | kaldemar | ghost75: is the call answered somewhere before the busy tone? |
| 11:58.54 | *** join/#asterisk vlad_starkov (~vlad_star@213.79.102.250) |
| 12:03.08 | ghost75 | in cli is written that everbody is busy |
| 12:04.39 | kaldemar | which means pretty much nothing at all by itself. |
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| 12:32.01 | *** join/#asterisk FiReSTaRT (~dlyh@unaffiliated/firestart) |
| 12:40.24 | *** join/#asterisk felipealmeida (~user@mvx-187-16-79-187.mundivox.com) |
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| 12:46.39 | youjelly | ghost75: can you share the log |
| 12:54.20 | *** join/#asterisk nantou (~phonetic@gateway/tor-sasl/martinphone) |
| 13:08.35 | ghost75 | is only: [Dec 28 13:09:15] VERBOSE[26318] app_dial.c: == Everyone is busy/congested at this time (1:0/1/0) |
| 13:10.07 | kaldemar | there is plenty of other output related to that call. |
| 13:12.04 | ghost75 | not relevant, its right after dial and before: [Dec 28 13:09:15] VERBOSE[26318] pbx.c: == Spawn extension (phones998780, 081932125336, 2) exited non-zero on 'SIP/10-00000156' |
| 13:12.05 | *** join/#asterisk bebe-lala (~big_ben@ip216-239-83-253.vif.net) |
| 13:13.19 | kaldemar | ghost75: oh it is relevant. |
| 13:13.27 | ghost75 | why |
| 13:14.03 | ghost75 | or you want to see sip log? |
| 13:15.07 | kaldemar | i already asked you a question which you did not answer. the logs would provide an answer to the question and tell where the tone comes from. |
| 13:15.41 | ghost75 | but which log, verbose or sip |
| 13:16.15 | kaldemar | verbosity might be enough. |
| 13:16.24 | ghost75 | might not but i can show |
| 13:18.03 | ghost75 | http://pastebin.com/HaRdte7t |
| 13:19.08 | kaldemar | it's the pap2t. |
| 13:19.20 | kaldemar | check it's tone zone setting if it has one. |
| 13:21.06 | ghost75 | do you know whats also strange |
| 13:21.15 | ghost75 | why its hanging up immediately |
| 13:21.41 | kaldemar | for that, see sip debug. |
| 13:22.38 | ghost75 | this gets also written into messages file? |
| 13:22.47 | kaldemar | no. |
| 13:22.57 | kaldemar | in CLI, "sip set debug on" |
| 13:26.43 | *** join/#asterisk Dovid (~Dovid@static-173-63-105-210.nwrknj.fios.verizon.net) |
| 13:27.22 | *** join/#asterisk Dovid (~Dovid@static-173-63-105-210.nwrknj.fios.verizon.net) |
| 13:30.47 | ghost75 | http://pastebin.com/CW72mCxJ |
| 13:30.52 | ghost75 | problem with audio codec? |
| 13:31.22 | *** join/#asterisk kresp0 (~kresp0@81.61.24.25.dyn.user.ono.com) |
| 13:31.56 | ghost75 | SIP/2.0 488 Not Acceptable here |
| 13:32.01 | ghost75 | Warning: 305 arcor.de " Incompatible media format" |
| 13:33.01 | kaldemar | ghost75: seems so. the only codec you offer is G.729a and they don't like it. fix that and move on to the next issue. |
| 13:33.31 | ghost75 | strange though that i can receive calls with g.729 but i will try alaw/ulaw also |
| 13:34.56 | *** join/#asterisk felipealmeida (~user@mvx-187-16-79-187.mundivox.com) |
| 13:37.38 | ghost75 | yes is working now |
| 13:38.38 | ghost75 | i thought my itsp will convert from g.729 to the callers needed codec |
| 13:39.49 | ghost75 | so this means the called person has also voip but no g.729 |
| 13:53.00 | kaldemar | it means that your provider does not accept calls that use G.729a. |
| 14:07.47 | *** join/#asterisk shadar (~eugene@37.113.133.194) |
| 14:27.51 | WIMPy | Hmm. These new SIP scanners are a little annoying. |
| 14:29.05 | igcewieling | We block all off-net traffic with iptables so we never see those scans. 8-) |
| 14:32.58 | Katty | hello my asterisk does not work at all how to fix plz?? is urgent plz answer thx. |
| 14:35.20 | leifmadsen | Katty: step one -- 3 shots of vodka |
| 14:37.02 | Katty | leifmadsen: is that before or after the 3 shots of espresso |
| 14:37.17 | leifmadsen | Katty: use the espresso as a chaser |
| 14:37.23 | Katty | i love you. |
| 14:37.29 | leifmadsen | >3 |
| 14:37.33 | leifmadsen | huh... |
| 14:37.36 | leifmadsen | that didn't work out at all |
| 14:37.37 | leifmadsen | <3 |
| 14:37.51 | Katty | looks like a bubbly eyed goldfish. |
| 14:38.24 | leifmadsen | then we all win |
| 14:38.36 | Katty | yay |
| 14:38.46 | leifmadsen | RJD2 on the decks! |
| 14:39.03 | *** join/#asterisk santa0536 (~santa@cn-bgp-nat.portaone.com) |
| 14:44.29 | *** join/#asterisk santa0536 (~santa@cn-bgp-nat.portaone.com) |
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| 15:17.39 | *** join/#asterisk moy (~moy@UNVLON55-1176057127.sdsl.bell.ca) |
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| 15:31.03 | *** join/#asterisk falz (~falz@rainbowdivider.com) |
| 15:31.24 | falz | hi. is there a 'sip clear peer' type of command? I have two devices fighting over one thing (vpn user vs office user phone) |
| 15:32.01 | kaldemar | sip unregister |
| 15:32.31 | falz | don't have that command |
| 15:32.39 | falz | 1.4.11 |
| 15:33.03 | *** join/#asterisk Defraz (~Defraz@mail.pocatellochildren.com) |
| 15:33.29 | ghost75 | (14:53:29) kaldemar: it means that your provider does not accept calls that use G.729a. <- for some it works |
| 15:33.38 | kaldemar | consider upgrading. you're using ancient code. |
| 15:33.58 | ghost75 | provider just doesnt translate codecs |
| 15:35.02 | kaldemar | maybe, maybe not. i have no idea about your complete scenario. |
| 15:35.16 | ghost75 | if i call my number with mobile then g729 works |
| 15:35.32 | ghost75 | if i dial a analog or isdn number then g729 also works |
| 15:35.48 | leifmadsen | falz: sip prune perhaps |
| 15:35.56 | leifmadsen | but that's more for pruning from realtime I think |
| 15:36.26 | leifmadsen | falz: also, you won't be able to stop the peers fighting unless you block one of them |
| 15:36.30 | igcewieling | falz: won't do any good unless you stop the devices from fighting with each other |
| 15:36.35 | igcewieling | looks at leifmadsen |
| 15:36.39 | leifmadsen | igcewieling: I win! |
| 15:38.14 | kaldemar | prune is for realtime only |
| 15:38.20 | falz | yeah doesnt let you specify one |
| 15:38.21 | falz | blegh |
| 15:38.30 | falz | will null route/firewall his home vpn ip off or something |
| 15:38.41 | leifmadsen | ya, pruning wouldn't have helped anyways |
| 15:38.51 | leifmadsen | the peers would still attempt to request |
| 15:39.04 | leifmadsen | the alternative is to use permit and deny on the peer |
| 15:39.40 | kaldemar | i prefer sledgehammer |
| 15:41.31 | *** part/#asterisk falz (~falz@rainbowdivider.com) |
| 15:44.15 | Katty | plops |
| 15:44.21 | Katty | i just moved furniture. ugg. |
| 15:44.39 | Katty | how dare i have to do work on a friday. |
| 15:44.42 | Katty | grumps. |
| 15:44.47 | Katty | k, over it! |
| 15:44.54 | Katty | returns to knitting and watching birdies |
| 15:47.39 | *** join/#asterisk WIMPy (~wimpy@e183095026.adsl.alicedsl.de) |
| 15:50.17 | *** join/#asterisk Rico29 (~rico@oceanet-telecom-fttb-129-2.olm.fr) |
| 15:50.20 | Rico29 | hi |
| 15:50.39 | Rico29 | are panasonic KX-NT phones compatibles with asterisk |
| 15:50.40 | Rico29 | ? |
| 15:52.09 | *** join/#asterisk vinhdizzo (~vinh@cpe-66-74-182-202.socal.res.rr.com) |
| 15:53.34 | leifmadsen | Rico29: are they SIP? |
| 15:53.39 | Rico29 | yes |
| 15:53.44 | leifmadsen | then yes |
| 15:54.42 | Rico29 | don't you think it can be "panasonic-made SIP" ? |
| 15:54.49 | leifmadsen | no |
| 15:54.52 | Rico29 | ok |
| 15:55.06 | leifmadsen | this is ISDN BRI |
| 15:55.18 | leifmadsen | s/is/isn't/ |
| 15:55.31 | leifmadsen | facepalms |
| 15:55.43 | Rico29 | héhé |
| 15:56.19 | Katty | yarn bombs leifmadsen's keyboard. |
| 15:56.29 | Katty | fixed! |
| 16:08.10 | *** join/#asterisk acidfu (~inetrio.c@75-119-230-242.dsl.teksavvy.com) |
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| 16:21.21 | *** join/#asterisk saint_ (~saint@c-68-38-56-184.hsd1.nj.comcast.net) |
| 16:21.58 | saint_ | hi all |
| 16:22.20 | saint_ | what's the big reason to switch from 1.8 to 10.x or 11.x ..? |
| 16:22.54 | *** join/#asterisk pabelanger (~pabelange@asterisk/contributor-and-bug-marshal/pabelanger) |
| 16:22.54 | *** mode/#asterisk [+o pabelanger] by ChanServ |
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| 16:52.56 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 16:59.52 | saint_ | i installed mysql and all the os related binaries , what do i need to do in order for asterisk to take in account mysql , and install odbc modules ? |
| 17:00.02 | saint_ | i did a ./configure && make && make install , but it still does not work |
| 17:00.20 | saint_ | i do not see any *odbc* modules in /usr/lib/asterisk/modules |
| 17:00.36 | igcewieling | saint_: how about ./configure && make menuconfig [select the correct modules and save] make && make install , but it still does not work |
| 17:00.59 | saint_ | i ll try |
| 17:04.09 | igcewieling | saint_: I see no reason to switch away from 1.8 right now. Asterisk 11 is too new for me to trust it in production. |
| 17:04.49 | [TK]D-Fender | saint_i installed mysql and all the os related binaries <- that doesn't tell us which ones. |
| 17:05.19 | youjelly | does asterisk like more cores or higher clocks |
| 17:06.51 | saint_ | [TK]D-Fender: on CentOS, unixODBC unixODBC-devel libtool-ltdl libtool-ltdl-devel mysql-connector-odbc |
| 17:07.13 | saint_ | [TK]D-Fender: when I run odbcinst -q -d , I see the driver MySQL |
| 17:07.41 | saint_ | [TK]D-Fender: finally when I run echo "select 1" | isql -v asterisk-connector i see the ODBC connector |
| 17:07.49 | [TK]D-Fender | saint_: ./configue && make menuconfig |
| 17:08.02 | [TK]D-Fender | saint_: the curses install should tell you what's missing... |
| 17:08.10 | [TK]D-Fender | +n |
| 17:08.23 | youjelly | so cores or clock? |
| 17:08.41 | *** join/#asterisk nantou (~phonetic@gateway/tor-sasl/martinphone) |
| 17:10.19 | saint_ | [TK]D-Fender: it ends with no error message |
| 17:10.35 | saint_ | [TK]D-Fender: is there anywhere in menuconfig a place to select mysql ? if yes, I can t find it |
| 17:10.38 | [TK]D-Fender | make menuconfg is an onscreen interface... not an "end status" |
| 17:12.03 | igcewieling | youjelly: servers |
| 17:12.31 | youjelly | igcewieling: ... |
| 17:12.45 | igcewieling | in order of preference, more servers, more cores, more CPU speed. |
| 17:12.45 | saint_ | [TK]D-Fender: i understand that, but when i have the interface, is there anything to select within it ? |
| 17:13.36 | [TK]D-Fender | It lets you hand-pick the ones to compile and TELLS you what dependencies you are missing. |
| 17:13.45 | youjelly | so instead of having 1 8 core machine I should have 4 2 core machines? |
| 17:14.01 | igcewieling | saint_: you mean like some option like Dialplan Functions / func_odbc |
| 17:14.30 | igcewieling | youjelly: or two 4 core machines |
| 17:14.44 | igcewieling | Personally I think 8 cores is a good number. |
| 17:14.59 | youjelly | I second that |
| 17:15.29 | igcewieling | But yes, four 2 core machines is likely to perform better than one 8 core machine, but I don't think you'll notice that much difference. |
| 17:15.50 | youjelly | going with an e5-2680 |
| 17:15.50 | igcewieling | keep mind this is my OPINION. I'm not aware of any benchmarking for recent Asterisk versions |
| 17:15.56 | saint_ | [TK]D-Fender: ha... : XXX func_odbc |
| 17:16.15 | igcewieling | We use three 16 core machines, but the machines also run lots of AGIs and use MySQL |
| 17:16.15 | [TK]D-Fender | youjelly: You also haven't stated what your actual needs are... |
| 17:16.19 | [TK]D-Fender | saint_: LOOK DOWN |
| 17:16.39 | saint_ | [TK]D-Fender: yeah, depends on res_odbc ... and res_odbc has XXX too |
| 17:16.57 | [TK]D-Fender | and what do ITS dependencies say? |
| 17:17.05 | saint_ | generic.. looking for it |
| 17:17.19 | igcewieling | saint_: this means asterisk did not think the required libs for that function are installed in the system. |
| 17:17.35 | saint_ | ok.. let me double chck |
| 17:17.45 | *** join/#asterisk timahvo1 (~rogue@41.212.120.45) |
| 17:18.09 | igcewieling | you MAY have to do a "make distclean" before the ./configure for Asterisk to pick up any libs installed since the first invocation of ./configure. |
| 17:18.51 | youjelly | needs as in? |
| 17:19.16 | youjelly | I need to accommodate as many users as I can on 1 server |
| 17:20.23 | [TK]D-Fender | youjelly: "as many users as I can" is not a need. |
| 17:20.51 | saint_ | igcewieling: make distclean did it. i can now see the func_odbc with a * in front of it. thanks ! |
| 17:21.19 | [TK]D-Fender | youjelly: I'm quite sure if I just through 10 million out there it will be far more than your need. Care to shave that down to a REALISTIC number and describe what the server will actually be doing with them? |
| 17:21.36 | igcewieling | saint_: Asterisk pretends to be your friend and then hits you on the head with a 2x4. You learn to live with it. I, like you, assumed that ./configure would actually, you know, CONFIGURE |
| 17:22.03 | saint_ | igcewieling: lol.. yeah.. lesson learned and will be remembered.. |
| 17:22.39 | youjelly | The dialplan itself isn't doing much really, just some agi calls, playbacks, dtmf (its an IVR) |
| 17:22.58 | igcewieling | we have a custom ISO to install the OS, packages, Asterisk, etc so we don't run into those sorts of issues anymore. |
| 17:23.00 | youjelly | most of the heavy lifting is done on a separate AGI server |
| 17:23.20 | saint_ | igcewieling: i m making myself a template.. |
| 17:24.09 | youjelly | and umm, there's some AMI actions being handled |
| 17:24.12 | youjelly | that's about it |
| 17:25.58 | youjelly | no trans-coding etc, and using SIP, just incoming calls. |
| 17:28.23 | igcewieling | youjelly: if you want lots of users/callers then put a SIP Proxy in front of Asterisk |
| 17:28.45 | youjelly | doing that |
| 17:28.54 | youjelly | but there's lots and lots of users |
| 17:29.00 | youjelly | and I can only put so many servers |
| 17:29.18 | *** join/#asterisk vlad_starkov (~vlad_star@178.177.240.108) |
| 17:29.29 | youjelly | kamilio/openSER right? |
| 17:30.38 | [TK]D-Fender | users != callers |
| 17:30.44 | [TK]D-Fender | there is a distinction in the load. |
| 17:30.58 | [TK]D-Fender | And I haven't seen any numbers yet |
| 17:32.10 | youjelly | callers |
| 17:32.53 | youjelly | ~5000 channels |
| 17:35.57 | *** join/#asterisk chris_n (~Chris@184.7.21.42) |
| 17:36.28 | [TK]D-Fender | Definitely multiple servers. |
| 17:36.58 | [TK]D-Fender | Given your AGI's are off-loaded it should be that bad though. |
| 17:37.12 | [TK]D-Fender | (comparatively) |
| 17:37.33 | youjelly | it should be? |
| 17:38.12 | youjelly | I know but I'm planning around 1000-1500 calls per server |
| 17:40.03 | [TK]D-Fender | shouldn't* |
| 17:44.08 | *** join/#asterisk evil_gordita (robert@ip70-188-50-186.rn.hr.cox.net) |
| 17:49.43 | youjelly | so do I need that much clock speed with 8 cores |
| 17:56.51 | *** join/#asterisk fisted (~fisted@unaffiliated/fisted) |
| 18:00.04 | *** join/#asterisk sruffell (~sruffell@asterisk/the-kernel-guy/sruffell) |
| 18:00.04 | *** mode/#asterisk [+o sruffell] by ChanServ |
| 18:06.13 | *** join/#asterisk navaismo (~navaismo@189.144.207.195) |
| 18:17.24 | ghost75 | a 1000 calls on one server? which company is this? |
| 18:20.34 | *** join/#asterisk timahvo1 (~rogue@41.212.120.45) |
| 18:26.45 | *** join/#asterisk Nugget (nugget@rennsport.macnugget.org) |
| 18:33.19 | *** join/#asterisk appleboy (~appleboy@about/cooking/nakedchef/apple/tarts) |
| 18:43.10 | *** join/#asterisk blee (~blee@68.204.217.123) |
| 19:04.01 | adeel | youjelly: clarification, when you say 1 server, do you mean 1 IP or 1 actual box? |
| 19:05.06 | [TK]D-Fender | Actual box |
| 19:06.33 | *** join/#asterisk Nugget (nugget@rennsport.macnugget.org) |
| 19:06.51 | *** join/#asterisk vlad_starkov (~vlad_star@178.177.240.108) |
| 19:07.14 | adeel | i was able to hit ~2200 concurrent calls on an 8 core 3.4 ghz box with full dialplan, and also had mysql & kamailio in front to handle the registrations/etc |
| 19:07.44 | adeel | but that was in a VM, so that might be higher |
| 19:12.03 | *** join/#asterisk mjordan (~mjordan@198.2.4.225) |
| 19:12.03 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 19:14.37 | igcewieling | adeel: Was Asterisk handling RTP or was that reinvited off Asterisk? Also how many calls per second or calls per min? |
| 19:14.56 | igcewieling | adeel: also what asterisk version? |
| 19:16.15 | [TK]D-Fender | adeel: What is the average airspeed velocity of an unladen swallow? |
| 19:16.43 | ghost75 | binladen? |
| 19:17.15 | Penguin | /bin/laden |
| 19:17.22 | *** join/#asterisk sruffell (~sruffell@asterisk/the-kernel-guy/sruffell) |
| 19:17.22 | *** mode/#asterisk [+o sruffell] by ChanServ |
| 19:17.53 | [TK]D-Fender | high-5's Penguin |
| 19:17.55 | adeel | igcewieling: without RTP traffic, with rtp traffic that dropped to about half that value. i sustained about 35-40 cps. 50 cps would result in too many failed calls (i considered any failed call to be too many) and this was asterisk 1.6.2 |
| 19:18.23 | adeel | err, 1.6.20 |
| 19:18.36 | adeel | and i think my real bottleneck was the fact it was in a VM rather than on baremetal |
| 19:19.20 | adeel | in all honesty, i'd rather implement it in kamailio and just proxy the relevant calls to asterisk only when necessary |
| 19:21.07 | [TK]D-Fender | adeel: however that does not match his call requirements at all. |
| 19:32.06 | *** join/#asterisk mjordan (~mjordan@198.2.4.225) |
| 19:32.06 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 19:37.57 | *** join/#asterisk vlad_sta_ (~vlad_star@178.177.240.108) |
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| 19:42.16 | *** join/#asterisk greenwolf (42570426@gateway/web/freenode/ip.66.87.4.38) |
| 19:42.35 | n2tech | hello |
| 19:46.34 | SuperNull | is there a variable that would hold a registration username ? |
| 19:47.02 | *** join/#asterisk mjordan (~mjordan@198.2.4.225) |
| 19:47.02 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 19:47.56 | Penguin | What would you use it for if there is one? |
| 19:48.49 | n2tech | why dont u use func_odbc.conf for mysql database ? |
| 19:49.04 | n2tech | create a database for usernames for asterisk to lookup and retrieve those varaibles |
| 19:49.13 | Penguin | Not everyone uses mysql. |
| 19:49.33 | Penguin | If you're only going to put the names into a db, why not use the asterisk db? |
| 19:50.03 | n2tech | tru penguin true |
| 19:50.24 | n2tech | but at least you can keep track of things alot better using mysql database |
| 19:50.45 | n2tech | can use phpmyadmin to actually see what going on and what info is actually in the db |
| 19:50.48 | Penguin | Regardless, if we know what he wants to do with such data, we can recommend ways to make it happen. |
| 19:51.07 | n2tech | so you recommend to use asterisk db for things like this |
| 19:51.19 | Penguin | Usually, yes. |
| 19:51.31 | Penguin | Depends on what you're trying to store. |
| 19:51.38 | n2tech | like if i wanted a caller to login using a username and password via dtmf i can use the internal asterisk db to lookup that info? |
| 19:51.46 | Penguin | Yes. |
| 19:52.10 | n2tech | ok nice i have been setting up external mysql databases for this |
| 19:52.20 | Penguin | If you need a lot of fields, the astdb might not be the best db for it. |
| 19:52.24 | SuperNull | Penguin im trying to track down some 'leaking' calls through our call detail records.. |
| 19:52.28 | n2tech | like there was a banking system i created for customers to login using their account number and password |
| 19:52.49 | n2tech | but i kept creating mysql databases for this not realizing that i could have actually used the internal asterisk db for this type of operation |
| 19:53.15 | n2tech | awesome im glad u told me this penguin...this could save me alot of time and programming instead of using external databases |
| 19:57.57 | SuperNull | alright.. im running 1.4.20 on 2 servers. one updates regserver in ast realtime registration table.. one doesn't .. asterisk.conf and sip.conf both appear to be .. 100% correct. |
| 19:58.37 | SuperNull | waits for the 'omg its old as shit' |
| 19:59.30 | n2tech | olmg its old as shit |
| 19:59.43 | n2tech | actually i think 1.4 and 1.6 are the best running version of asterisk |
| 19:59.50 | n2tech | seem to be alot more stable than any other versions |
| 20:00.11 | SuperNull | 1.6 is a rock. |
| 20:00.17 | WIMPy | thinks that 1.6 is the worst. |
| 20:00.29 | Penguin | 1.6 isn't even a branch, much less a version. |
| 20:00.35 | kaldemar | there is no 1.6 |
| 20:00.43 | SuperNull | but butttt |
| 20:00.50 | SuperNull | oh. |
| 20:00.51 | SuperNull | wait. |
| 20:00.55 | SuperNull | i ment 1.8 ;) |
| 20:01.05 | SuperNull | 1.8 is a rock .. |
| 20:01.06 | SuperNull | LOL |
| 20:01.20 | Penguin | Anyway... CDR already stores the peer name for each call. |
| 20:01.45 | n2tech | what version does everyone think is the best version so far? |
| 20:01.56 | n2tech | ill say 1.4 or 1.8 |
| 20:02.08 | Penguin | 1.8.11.0 seems pretty good. |
| 20:02.12 | WIMPy | Or 10 or 11. |
| 20:02.13 | lanning | "the next version" :) |
| 20:02.20 | Penguin | Neither 1.4 nor 1.8 is a version. |
| 20:02.30 | n2tech | why do you say that penguin? |
| 20:02.46 | *** join/#asterisk moos3 (~moos3@pool-72-73-84-223.ptldme.east.myfairpoint.net) |
| 20:02.51 | Penguin | It's stable. |
| 20:03.27 | *** join/#asterisk infobot (~infobot@rikers.org) |
| 20:03.28 | *** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- Asterisk 11.1.0 (2012/12/10), 10.11.0 (2012/12/10), 1.8.19.0 (2012/12/10), DAHDI-linux 2.6.1 (2012/04/20), DAHDI-tools 2.6.1 (2012/04/20), libpri 1.4.14 (2012/12/20) -=- Visit the official Asterisk wiki: wiki.asterisk.org |
| 20:03.45 | Penguin | With the exception of 1.6, you've listed a bunch of branches, not versions. 1.6 isn't even a branch. |
| 20:03.58 | Penguin | ~asterisk10 |
| 20:03.58 | infobot | Asterisk 10 -- http://blogs.digium.com/2011/07/21/the-evolution-of-asterisk-or-how-we-arrived-at-asterisk-10/, or a Standard Release. It was released on 2011-12-15, with maintenance until 2012-12-15. Asterisk 10 will be end of life on 2013-12-15. |
| 20:04.14 | SuperNull | i have not even started using it yet and its EOLed. |
| 20:04.17 | SuperNull | lol |
| 20:04.22 | Penguin | What is? |
| 20:04.30 | SuperNull | Ast 10. |
| 20:04.36 | SuperNull | obviously gonna skip it. |
| 20:04.45 | Penguin | EOL already? I doubt that. |
| 20:04.47 | SuperNull | still contemplating about freeswitch |
| 20:04.48 | [TK]D-Fender | n2tech: Instead of going from 1.8 to 1.10 they decided to just call it "10" |
| 20:04.56 | n2tech | i see |
| 20:05.01 | Penguin | But go for LTS branches instead. |
| 20:05.04 | n2tech | so is version 10 or 11 better to use? |
| 20:05.05 | n2tech | :) |
| 20:05.16 | SuperNull | 11 is higher numerically ? |
| 20:05.22 | Penguin | They aren't versions, they're branches. |
| 20:05.33 | n2tech | ok so which is better in your opinion? |
| 20:05.38 | [TK]D-Fender | Asterisk 11 = LTS and includes all the goodies from 10 and quite a bit more. So far no horror stories on it so not a lot of reason to think about anything lower |
| 20:05.40 | Penguin | Any LTS branch. |
| 20:06.57 | Penguin | Even the "best branch" can still have a really crappy version or two in it. |
| 20:09.24 | n2tech | yea it happens |
| 20:09.26 | n2tech | nothings perfect |
| 20:09.54 | n2tech | i just wish they would fix the core to handle sip better and handle more sip concurrent calls |
| 20:09.55 | n2tech | dead locks suck' |
| 20:10.20 | *** join/#asterisk elguero (~miguel323@2001:470:1f06:12c4::2) |
| 20:10.21 | n2tech | i can never seem to get an asterisk system to handle anymore than 150 concurrent calls at one time before it totally crashes |
| 20:13.55 | [TK]D-Fender | n2tech: Digium was selling 4-port E1 cards a decade ago (120 channel just there). You must be doing something wrong.... |
| 20:14.18 | n2tech | nope its a complete voip sip machine |
| 20:14.31 | n2tech | never got it to hold any more than 150 calls at 1 time |
| 20:14.38 | n2tech | dead locks would occur |
| 20:15.09 | n2tech | yea your talking about analogue cards to PSTN |
| 20:15.09 | n2tech | im talking about SIP/VoIP |
| 20:15.20 | [TK]D-Fender | n2tech: Do let us know when that is something you are preparing to actually debug. |
| 20:15.35 | n2tech | yes i will |
| 20:15.52 | n2tech | maybe i can help develop to fix this |
| 20:16.03 | n2tech | get asterisk to hold more sip calls then it currently can |
| 20:16.20 | n2tech | i would be willing to dedicate time in helping fix it to allow this no problem |
| 20:16.28 | [TK]D-Fender | n2tech: Since you are the only one I've heard with issues like that I'm not sure that Asterisk at a whole is at faul but rather something rather tragic in your scenario |
| 20:16.52 | [TK]D-Fender | n2tech: There are user pushing thousands of simultaneous calls through their servers.... |
| 20:17.00 | n2tech | how many current SIP calls have you been able to handle at one time in an asterisk machine? |
| 20:17.29 | n2tech | i have never heard anyone running 500 sip calls concurrently accept in freeswitch |
| 20:17.39 | [TK]D-Fender | n2tech: I'm not the best comparative sample for that... |
| 20:17.48 | [TK]D-Fender | n2tech: I jsut know what others have come through with. |
| 20:17.57 | n2tech | well then how could u comment on the subject |
| 20:18.19 | kaldemar | http://www.voip-info.org/wiki/view/Asterisk+dimensioning |
| 20:18.22 | [TK]D-Fender | n2tech: Countless stats from OTHERS who have told us what they've done |
| 20:18.56 | [TK]D-Fender | n2tech: I don't have to duplicate the experiments of others to tell you the RESULTS. |
| 20:19.05 | n2tech | yea creating a cluster of asterisk machines |
| 20:19.27 | n2tech | but 1 asterisk server isnt capable of handling that many sip calls at one time...thats y asterisk created IAX |
| 20:19.29 | [TK]D-Fender | n2tech: And Digium was selling 4-port E1 cards (120 channels) a decade ago. And that's just the TDM side.... |
| 20:19.34 | n2tech | to handle VoIP calls better than sip does |
| 20:19.48 | n2tech | exactly TDM its analogue and digtal lines |
| 20:19.54 | n2tech | they go thru dedicated lines to the POTS |
| 20:20.06 | [TK]D-Fender | IAX doesn't "handle" better than SIP does. IAX2 has one thing on its side : IAX2 Trunk Mode. |
| 20:20.08 | n2tech | totally different than handling VoIP calls using SIP |
| 20:20.13 | [TK]D-Fender | And that's just bandwidth sasvings. |
| 20:20.24 | [TK]D-Fender | Changes nothing about what the machine spec is capable of handling. |
| 20:20.58 | Penguin | supernull: I just looked it up... The 10.x branch doesn't EOL until 2013-12-15, so you've got a full year to play with it if you prefer to use non-LTS branches. |
| 20:21.08 | [TK]D-Fender | It has virtually no impact on CPU/memory etc vs SIP |
| 20:21.51 | n2tech | listen fender i have deployed these machines in many conditions many times i know what im talking about |
| 20:22.00 | n2tech | your going on info from others and have not actually done it yourself |
| 20:22.11 | WIMPy | You mean: You know how not to do it? |
| 20:22.23 | n2tech | im not going to argue with you about this event the developers will tell you sip calls are limited |
| 20:22.36 | n2tech | its the nature of VoIP |
| 20:23.06 | [TK]D-Fender | n2tech: No-one else is coming in here stating issues like yours. I've only been here for 8 YEARS. What do I know? |
| 20:23.49 | n2tech | umm..the whole freeswitch team will tell you. thats y they branced off and left asterisk because its limited ability in SIp calls it can handle |
| 20:23.52 | [TK]D-Fender | n2tech: You have offered us no details and speak generically about "the nature of VoIP" |
| 20:23.53 | n2tech | duh |
| 20:24.02 | n2tech | they created a better core to handle sip calls without dead locks |
| 20:24.17 | n2tech | thats y an entire project is dedicated to handle this |
| 20:24.34 | n2tech | and this is y they create opensips or sip proxy's because a pbx is limited in this nature |
| 20:24.35 | WIMPy | At which Asterisk Version was that? |
| 20:24.40 | n2tech | everyone know this |
| 20:24.58 | n2tech | volume* |
| 20:24.59 | [TK]D-Fender | Yup, we've entered full troll-mode.... |
| 20:24.59 | n2tech | asterisk is not a switch and cant handle this amount of call voume |
| 20:25.14 | [TK]D-Fender | n2tech: Incorrect. |
| 20:26.04 | WIMPy | n2tech: There is one thing I don;t understand about the whole thing. If Asterisk can't do what you want, but freeswitch can, then why are you using Asterisk? |
| 20:26.57 | [TK]D-Fender | http://www.voip-info.org/wiki/view/Asterisk+dimensioning |
| 20:27.09 | [TK]D-Fender | LOts of samples with 5-year old specs list over 300 calls. |
| 20:27.15 | n2tech | because asterisk is a pbx |
| 20:27.18 | n2tech | i love asterisk |
| 20:27.25 | n2tech | its great for what it does its just not a sip proxy |
| 20:27.28 | [TK]D-Fender | 3.2 Gig Pentium 4, HyperThreading (HT) turned off, Memory 1 Gig, no hard drive, running in Ramdrive, Memory usage never goes over 256Meg. I'm using asterisk with looping call test configs to play audio and using 3 of the same spec servers to pound calls through 1 server. I managed to get 350 concurrent calls through (g711, no transcoding) with perfect audio consistently with ~20% idle... |
| 20:27.30 | [TK]D-Fender | ...processor load. Anything above that and things start breaking up. Using Asterisk 1.2.6 I'm running into a limit of 276 SIP calls and no more. IAX calls can go 400+, so I test with combination 200+ SIP calls and the rest IAX and a combination of more and less SIP and IAX calls. With HT turned on, SMB loaded in the kernel gave ~20% performance increase, BUT, using 425+ channels gave very... |
| 20:27.31 | [TK]D-Fender | ...inconsistent results, choppy audio, calls dropped, no audio, and call setup time slowed. |
| 20:27.37 | n2tech | whoa dude us pb |
| 20:27.43 | WIMPy | I wouldn't call Asterisk a PBX, but that's another matter. |
| 20:27.46 | n2tech | i love asterisk it is a great PBX |
| 20:27.48 | [TK]D-Fender | that was one line, it jsut happened to span. |
| 20:27.50 | [TK]D-Fender | Take a pill. |
| 20:27.51 | n2tech | awesome system for office and businesses |
| 20:27.53 | Penguin | three lines |
| 20:27.59 | Penguin | Still within tolerance. |
| 20:28.00 | [TK]D-Fender | At least I'm showing details while you fling FUD. |
| 20:28.05 | n2tech | but its not a sip proxy thats all im saying |
| 20:28.14 | [TK]D-Fender | DUH |
| 20:28.19 | [TK]D-Fender | We all know this |
| 20:28.24 | WIMPy | Finally a correct statement. |
| 20:28.36 | n2tech | ok i was just saying i wasnt able to get over 150 calls thats all |
| 20:28.43 | n2tech | your making a big thing about thing fender |
| 20:28.57 | n2tech | thats all i said was i wasnt able to get to handle over 150 SIP calls |
| 20:29.04 | [TK]D-Fender | and that's just YOU |
| 20:29.10 | n2tech | ok fine |
| 20:29.18 | n2tech | i never said "everyone" couldn;t |
| 20:29.19 | [TK]D-Fender | 5 year old benchmarks have done more than twice that. |
| 20:29.24 | n2tech | i was just making a statment |
| 20:29.50 | WIMPy | So do you want to find out why you can't do it or do you want to senselessly try yo convince others that it's the way it has to be? |
| 20:29.53 | [TK]D-Fender | that Asterisk can't handle it. |
| 20:30.07 | kaldemar | i've put more than 150 calls through asterisk on a 1.0.x version. |
| 20:30.30 | n2tech | im trying to figure out how to imporve wimpy |
| 20:30.39 | n2tech | can someone assist me in helping improve these numbers |
| 20:30.40 | n2tech | ?? |
| 20:30.58 | kaldemar | ~ask |
| 20:30.58 | 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. |
| 20:31.00 | [TK]D-Fender | n2tech: You aren't providing any details or debug. You are not actually doing anything to help your situation yet. We might be able to help when that changes... |
| 20:31.14 | WIMPy | That's certainly not the impression you made so far. |
| 20:31.17 | n2tech | ok so what should i post to help me in this matter |
| 20:31.49 | [TK]D-Fender | DETAILS |
| 20:32.30 | n2tech | alright let me login to this server ssh |
| 20:32.46 | n2tech | should i post sip.conf or extensions.conf ? |
| 20:32.53 | [TK]D-Fender | CONFIGS aren't to blame. |
| 20:32.54 | n2tech | what and where do you think is causing this limitation? |
| 20:32.59 | [TK]D-Fender | Machine spec & environment are |
| 20:33.04 | n2tech | machine |
| 20:33.04 | n2tech | ok |
| 20:33.11 | [TK]D-Fender | .... |
| 20:33.14 | n2tech | do u think iptables can cause this issue |
| 20:33.32 | [TK]D-Fender | you mean something that can throttle the shit out of your packets? Nah... couldn't be... |
| 20:33.35 | n2tech | or maybe fail2ban? |
| 20:33.42 | [TK]D-Fender | (see above) |
| 20:34.17 | n2tech | i mean the machine is a powerful machine. xean processor 4 cores 4 processors. 8GB ram |
| 20:34.36 | WIMPy | Have you looked ate what fail2ban did when you had issues? |
| 20:34.41 | [TK]D-Fender | n2tech: And if you read what the dimensioning reporting page said your machine kills those spec. |
| 20:34.52 | [TK]D-Fender | n2tech: Did you knock out fail2ban & iptables to test? |
| 20:35.08 | n2tech | im going to try and do that now fender see if i get different results |
| 20:35.19 | [TK]D-Fender | n2tech: Are you at some point going to tell us what OS you're running it on? any other services on that server? What version of Asterisk? |
| 20:35.37 | n2tech | ubuntu server 11.04LTS |
| 20:35.40 | [TK]D-Fender | n2tech: What call processing is actually being done to those calls? |
| 20:35.44 | n2tech | asterisk 1.8 |
| 20:35.51 | n2tech | g.711 ulaw for codec |
| 20:36.03 | [TK]D-Fender | Straing install? Virtualized? What kind of connection are those 150 calls going over? |
| 20:36.08 | [TK]D-Fender | What codec(s)? |
| 20:36.10 | n2tech | im doing alot of databses lookups and retrieval of data |
| 20:36.17 | n2tech | it is virtualized |
| 20:36.19 | n2tech | maybe thats y |
| 20:36.24 | [TK]D-Fender | No, that couldn't possibly be a load issue.... |
| 20:36.28 | n2tech | 100 mbps up/down |
| 20:36.42 | n2tech | deidcated bandwidth |
| 20:36.47 | n2tech | umm.. |
| 20:36.53 | [TK]D-Fender | Virtualized and lots of DB lookups... you ar already starting crippled.... |
| 20:37.04 | [TK]D-Fender | And placing the load issue on Asterisk |
| 20:37.16 | n2tech | ok with fail2ban i was able to sip test and performance the max amount of calls into the system..i was able to get 212 now |
| 20:37.17 | n2tech | yaya |
| 20:37.19 | [TK]D-Fender | Start removing roadblocks |
| 20:37.43 | n2tech | ok so how could i lower the db lookups? |
| 20:37.56 | [TK]D-Fender | Wow 41% increase right off the bat, imagine that... |
| 20:37.56 | n2tech | i need them for customers to login and check their account info |
| 20:38.14 | [TK]D-Fender | Maybe look at the efficiency in the method you use. |
| 20:38.14 | n2tech | yes very nice! IM VERY HAPPY |
| 20:38.25 | n2tech | i also have the mysql sever on same server |
| 20:38.34 | n2tech | which could be limiting |
| 20:38.38 | WIMPy | What dos top give you when you run those test? |
| 20:38.42 | [TK]D-Fender | Eggs in a basket |
| 20:38.56 | n2tech | do u think i should use asterisk db over the external mysql database? |
| 20:39.17 | WIMPy | Thate entirely depends on your application. |
| 20:39.31 | [TK]D-Fender | n2tech: Probably not applicable to your app. |
| 20:39.47 | n2tech | ok |
| 20:39.53 | [TK]D-Fender | n2tech: You should probably move it to another VM to keep the CPU for the * low. |
| 20:40.16 | [TK]D-Fender | If you're virtualizing and putting all your load on one VM then you aren't saving yourself much |
| 20:40.19 | n2tech | ok what about using asterisk db instead of external db mysql for lookups and info keeping |
| 20:40.19 | n2tech | good idea |
| 20:40.22 | WIMPy | What's the database performance like? |
| 20:40.22 | [TK]D-Fender | Just common sense... |
| 20:40.38 | WIMPy | How much time does the DB take and how much Asterisk? |
| 20:41.44 | WIMPy | What about disk I/O? |
| 20:43.45 | *** join/#asterisk gusto (~gusto@adsl-dyn-75.95-102-92.t-com.sk) |
| 20:44.31 | n2tech | ok let me do something different here' |
| 20:44.31 | n2tech | see if i get better results |
| 20:44.39 | n2tech | im going to put the db on a different virtual server |
| 20:44.45 | n2tech | or should i just use asterisk db |
| 20:45.19 | [TK]D-Fender | n2tech: You haven't told us what's that DB, what else access it, etc. |
| 20:45.21 | WIMPy | A different physical server would be a good idea. |
| 20:45.37 | n2tech | mysql |
| 20:45.42 | n2tech | i said that already |
| 20:45.47 | [TK]D-Fender | n2tech: what's IN* |
| 20:45.51 | n2tech | sorry i got my girlfriend yelling in my hear right now |
| 20:46.03 | WIMPy | And taking a very close look at your database queries might also be a good idea. |
| 20:46.13 | n2tech | asterisk and a website where users can register their account |
| 20:46.24 | n2tech | so that the phone system is up-to-date with the web site |
| 20:46.44 | n2tech | like i said tho should i use asterisk db over mysql |
| 20:46.47 | [TK]D-Fender | n2tech: Your "website" will likely have no realistic way to use the Asterisk DB (BDB which is not meant for this really), so MySQL is it. |
| 20:46.48 | n2tech | is there better handling |
| 20:46.58 | [TK]D-Fender | n2tech: It's a question of tuning where it is and how you use it |
| 20:47.03 | n2tech | ok |
| 20:47.05 | [TK]D-Fender | n2tech: HOW are you making your DB calls? |
| 20:47.34 | n2tech | the databse is mysql...people can register an account online |
| 20:47.36 | n2tech | then they call use those creds to login to the phone system |
| 20:47.44 | [TK]D-Fender | kay_: Why do you have multiple peers with the same IP? <- |
| 20:47.53 | n2tech | its just a simple payment gateway to accept and make payments on the phone rather than paying a bill online |
| 20:48.00 | [TK]D-Fender | n2tech: Precicesly HOW are you issuing the command to query the DB during your calls? |
| 20:48.07 | [TK]D-Fender | n2tech: I wasn't asking "why" |
| 20:48.23 | n2tech | im using func_odbc.conf |
| 20:48.40 | n2tech | then add that query into the existing dialplan |
| 20:48.43 | [TK]D-Fender | n2tech: Ok, that;s the most internal at least. Offload your DB as your first step |
| 20:48.44 | n2tech | i create the query in func |
| 20:49.11 | n2tech | then i found out about func |
| 20:49.13 | n2tech | yea i used to use php scripts and call those via agi to do that before |
| 20:49.21 | n2tech | then started using that...i noticed a dramatic increase in savings |
| 20:49.24 | n2tech | and fast load times |
| 20:49.30 | n2tech | i hate agi |
| 20:49.49 | n2tech | so i use func |
| 20:49.58 | n2tech | no need to call a php script to do it when theres a built in func_odbc.conf |
| 20:50.03 | *** join/#asterisk nantou (~phonetic@gateway/tor-sasl/martinphone) |
| 20:50.15 | n2tech | it actually made things so much easier once i found out about it lol |
| 20:50.24 | [TK]D-Fender | n2tech: If the work you need to do is simply enough for it, yes |
| 20:50.34 | n2tech | great application |
| 20:50.41 | n2tech | yup |
| 20:51.02 | n2tech | hey do u have and recommend reading material that talks more about func_odbc.conf and how to use it? |
| 20:51.14 | n2tech | i would like to touch up my skills and be able to use more advanced logic with that applciation |
| 20:51.30 | [TK]D-Fender | ~book |
| 20:51.31 | 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 |
| 20:51.32 | [TK]D-Fender | ^^ |
| 20:51.34 | n2tech | also could i use func_odbc.conf with the internal asterisk db? |
| 20:51.44 | [TK]D-Fender | There really isn't that much to say about it... it is just what it appears to be |
| 20:51.50 | n2tech | or would i just use the dial plan and call DB application |
| 20:52.18 | n2tech | like if i wanted to use the internal DB for asterisk do i use the dialplan or can i use func_odbc.conf? |
| 20:52.36 | [TK]D-Fender | n2tech: Should be more or less the same I might figure. Start by taking the DB location out of the mix. |
| 20:53.03 | n2tech | yea ill put the db on a different server |
| 20:53.08 | n2tech | so the load isn't so much |
| 20:53.35 | n2tech | but im wondering if i use the internal asterisk db would i still use the func_odbc or just the dialplan to access and retrieve values? |
| 20:53.50 | WIMPy | func_db |
| 20:54.20 | n2tech | ok so wimpy i could still use func_odbc and have it use the internal asterisk DB? |
| 20:54.34 | [TK]D-Fender | n2tech: Forget the internal Asterisk DB. It is BDB and not suitably for outside integration. |
| 20:54.35 | WIMPy | No. That doesn;t work. |
| 20:55.00 | n2tech | oh ok so if i wanted to use it in conjunction with a website the internal db wont work then |
| 20:55.02 | WIMPy | It can be used. But I don't think that's a good idea. |
| 20:55.03 | n2tech | so keep the mysql then? |
| 20:55.21 | n2tech | ok makes sense fender |
| 20:55.33 | n2tech | ok |
| 20:55.40 | n2tech | once i move the db over to a new server i will run another test and see what type of performance i get |
| 20:55.44 | n2tech | ill let you guys kno |
| 20:56.01 | *** join/#asterisk nantou (~phonetic@gateway/tor-sasl/martinphone) |
| 20:56.14 | n2tech | ok ill be back guys i gotta log off real quick to do something with this server |
| 20:56.20 | n2tech | thanks wimpy, fender |
| 20:56.22 | n2tech | for everything |
| 20:56.36 | *** join/#asterisk cisco (~cisco@201.20.110.36) |
| 20:56.52 | n2tech | im sorry if i came off rude or ignorant...i have been in really bad mood lately ...girl problems if you kno what i mean :) |
| 20:56.54 | n2tech | didn't mean to take it out on you guys or anyone if i offend |
| 20:57.49 | n2tech | later guys |
| 20:58.30 | Penguin | It's that time of the month, I guess. |
| 20:58.40 | n2tech | lol shes PMSing |
| 20:58.42 | n2tech | later penguin |
| 20:59.01 | n2tech | catch u guys in a little bit...probably bout an hr or sp |
| 20:59.02 | n2tech | so* |
| 21:00.50 | *** join/#asterisk jeffspeff (~jeffspeff@12.49.160.131) |
| 21:01.46 | jeffspeff | using the ami, how can i show who called who and the caller cid(num) ? |
| 21:02.03 | *** join/#asterisk elico (~Thunderbi@bzq-79-181-228-197.red.bezeqint.net) |
| 21:02.10 | WIMPy | "Show"? |
| 21:02.17 | [TK]D-Fender | When? |
| 21:02.20 | WIMPy | Just listen for the events you get. |
| 21:02.20 | [TK]D-Fender | How? |
| 21:04.09 | jeffspeff | thanks wimpy, i'd looked over the 'dial' event status |
| 21:05.22 | WIMPy | newchannel would also be an usual suspect. |
| 21:20.48 | ageis | am trying to send SMS with smsq. 'smsq --motx-channel=voicepulse-primary/<my VP username> 1<seven digit numbr> test' here's the error I'm getting: http://pastebin.com/Mccgqvfe |
| 21:21.56 | WIMPy | Looks like you're missing the channeltype. |
| 21:22.14 | ageis | type is set to peer in sip.conf |
| 21:22.30 | ageis | is that different? |
| 21:23.03 | ageis | ok so this would be SIP i assume |
| 21:23.08 | WIMPy | You didn't say anything about sip. |
| 21:23.22 | WIMPy | That's the point. |
| 21:23.50 | ageis | how would I set the channel type? simply append SIP/ to the beginning of my motx-channel? |
| 21:24.17 | WIMPy | Probably. |
| 21:24.26 | ageis | k |
| 21:24.41 | WIMPy | I don't have SMS anywhere anymore. |
| 21:24.53 | ageis | oh great, now we're getting somewhere |
| 21:33.03 | rrittgarn | Kamailio or OpenSIPS, anyone have a preference? Looking to move towards one of the two with heavy DB integration. I keep hearing they are interchangeable, and am just looking for any insight as far as deployability and ease of management. |
| 21:33.25 | *** join/#asterisk Russ (~russ@206.29.182.246) |
| 21:42.12 | *** join/#asterisk Invader (~Invader@unaffiliated/invader) |
| 21:43.48 | *** join/#asterisk vlad_starkov (~vlad_star@178.177.240.108) |
| 21:49.26 | ageis | now I keep getting Congestion (circuits busy). I'm thinking at this point that maybe my SIP provider doesn't support SMS.. here's the debug: http://pastebin.com/R75RP8k6 |
| 21:51.11 | WIMPy | likes the "500 Nice try" |
| 21:51.26 | *** join/#asterisk appleboy (~appleboy@about/cooking/nakedchef/apple/tarts) |
| 21:51.50 | ageis | yeh |
| 22:02.13 | ageis | just tried SMS over sip2sip... same congestion (circuits busy) message but no 500 Nice try |
| 22:02.21 | *** join/#asterisk elico (~Thunderbi@bzq-79-181-228-197.red.bezeqint.net) |
| 22:03.20 | ageis | erm SIP/2.0 404 User not found I don't think it likes my "From" address |
| 22:03.53 | WIMPy | That's usually about the TO. |
| 22:20.54 | leifmadsen | rrittgarn: basically, from a friend of mine, use Kamailio -- OpenSIPS and them are essentially the same thing, but Kamailio is more community based, vs OpenSIPS being "some guy" based |
| 22:24.55 | rrittgarn | Leif: Thanks... Is there more documentation on one than the other by chance? Right now I'm trying to get my head around the whole routing thing to apply it to how I want to implement it (Like we discussed @ Astricon) |
| 22:32.55 | leifmadsen | rrittgarn: unfortunately I'm unsure -- I've not really used a lot of opensips/kamailio yet |
| 22:32.58 | leifmadsen | just started |
| 22:33.08 | leifmadsen | although the docs on the kamailio site seems to have a fair amount of docs |
| 22:35.53 | *** join/#asterisk GINO_SSA_BR (~GINO-SSA-@189-104-216-169.user.veloxzone.com.br) |
| 22:36.32 | GINO_SSA_BR | Hi folks . |
| 22:40.52 | *** join/#asterisk GINO_SSA_BR (~GINO-SSA-@189-104-216-169.user.veloxzone.com.br) |
| 22:42.42 | *** join/#asterisk GINO_SSA_BR (~GINO-SSA-@189-104-216-169.user.veloxzone.com.br) |
| 22:59.54 | *** join/#asterisk teloniusz (goldie@inferno.hell.pl) |
| 23:00.31 | teloniusz | hi. Is there a way to check after Dial whether there was PROGRESS before CONNECT? |
| 23:01.18 | teloniusz | (other than looking into logs, that is; I need to use that information in dialplan or at least in AGI app) |
| 23:01.34 | *** join/#asterisk fisted (~fisted@unaffiliated/fisted) |
| 23:01.40 | WIMPy | AMI might tell you via newstate. |
| 23:06.16 | *** join/#asterisk GINO_SSA_BR (~GINO-SSA-@189-104-216-169.user.veloxzone.com.br) |
| 23:07.12 | teloniusz | hm. To be more precise: I need to dial somewhere and when hangupcause was 16, but call time was less than 5 seconds and there was no progress, I need to hangup with cause 1 |
| 23:07.43 | teloniusz | because that's the way my provider signals there is no such number |
| 23:08.10 | WIMPy | Tell them to fix their shit. |
| 23:08.13 | teloniusz | do I need to write dual AMI/AGI script for this then? |
| 23:08.29 | teloniusz | WIMPy: they're much bigger than me ;> |
| 23:08.33 | Penguin | And then you need to fix your dial plan so that you aren't sending bogus numbers to the provider. |
| 23:09.12 | teloniusz | Penguin, I'm routing, there always will be some invalid numbers. |
| 23:09.17 | WIMPy | Penguin: Do you validate the numbers your users dial before trying to call them? |
| 23:09.49 | Penguin | I have a sane dial plan so that numbers dialed are valid numbers. |
| 23:10.14 | WIMPy | Anyway: Relating the information you get via AMI to simeting you can use in your dialplan could be interesting. |
| 23:10.22 | Penguin | There is no need for me to validate numbers that are valid. |
| 23:10.34 | teloniusz | Penguin, a number can be valid (as in: in proper format), but unallocated; there is no way to know this without trying to dial it. |
| 23:10.40 | WIMPy | How do you know they are valid? |
| 23:10.54 | Penguin | NANP defines validity. |
| 23:11.14 | WIMPy | Only for very few numbers. |
| 23:11.39 | Penguin | For the numbers I send to my provider, it covers virtually all of them. |
| 23:11.57 | *** join/#asterisk lvlinux (~n1gg@c-50-147-64-9.hsd1.tn.comcast.net) |
| 23:12.32 | teloniusz | Penguin, great and lucky for you; it would be imprudent for me to assume the same while all of my destinations are in Europe. |
| 23:13.11 | WIMPy | And you wouldn;t know if they are allocated anyway. |
| 23:13.14 | Penguin | I'm glad I didn't try to guess how to make your dial plan better. |
| 23:13.55 | WIMPy | You could set a variable via AMI as soon as you receive a PROGRESS message. |
| 23:13.57 | Penguin | I have no way to know if every number is allocated before dialing, but I also don't care if it is allocated or not. |
| 23:14.31 | WIMPy | The caller might want to know why his call failed. |
| 23:14.46 | teloniusz | WIMPy: on the channel my connection is on? So I'd need to implement some kind of AMI monitor... looks like it can work |
| 23:14.59 | WIMPy | But maybe that's only Europeans who want such strange things. |
| 23:15.19 | WIMPy | Just make sure you get the event you want. |
| 23:15.31 | WIMPy | But I thing progreass gives one. |
| 23:15.32 | Penguin | I suppose variable length phone numbers would pose a problem. I'm fortunate enough to not have that as a factor. |
| 23:15.57 | teloniusz | WIMPy: before this I was seriously considering implementing my own Dial application... |
| 23:16.31 | WIMPy | I'd consider using a provider that isn't broken. |
| 23:17.34 | teloniusz | WIMPy: pity that there lies some money ;) |
| 23:17.43 | WIMPy | Penguin: That has nothing to do with variable length numbers. |
| 23:17.55 | teloniusz | still, now I can estimate how much work it demands |
| 23:18.11 | teloniusz | WIMPy: thx |
| 23:18.46 | WIMPy | If a call fails, I'd like to know why. But that's a concept that seems strangely new to Asterisk. |
| 23:19.39 | Penguin | Most people don't care why. They dial a number and it either works or it doesn't. If it doesn't, they dial a different number. |
| 23:22.42 | teloniusz | Penguin, most modern phones show different status messages for "unallocated number", "busy", "network busy" and "subscriber temporary unavailable", at least. Properly established routing should not lose that information. |
| 23:26.54 | *** join/#asterisk vinhdizzo (~vinh@cpe-66-74-182-202.socal.res.rr.com) |
| 23:33.06 | *** join/#asterisk acidfu (~inetrio.c@108.161.125.187) |
| 23:35.51 | gusto | yes |
| 23:35.58 | gusto | i have today a different problem |
| 23:36.39 | gusto | i have a noise on the line, but only on one phone, i screwed two cables together on it so it seems to be something related to that |
| 23:36.52 | gusto | i ll check it tomorrow |