| 00:01.17 | ruben23 | hi |
| 00:02.58 | *** join/#asterisk jksM (jks@193.189.93.254) |
| 00:10.42 | hackeron | [TK]D-Fender: Kobaz: OK, I'm an idiot :) -- The BT phone cables use pin 1 and 4 instead of pin 2 and 3 to make a call - I just tried the second line with a different cable and it works! |
| 00:11.57 | *** join/#asterisk coppice (~chatzilla@94.201.17.210.dyn.pacific.net.hk) |
| 00:20.35 | *** join/#asterisk hesco (~hesco@c-24-99-160-121.hsd1.ga.comcast.net) |
| 00:23.42 | [TK]D-Fender | hackeron: SMRT |
| 00:23.55 | hesco | Where would I find Comedian Mail documented? I do not seem to be able to check voicemail by dialing the extension provided for that purpose. It answers "Comedian Mail, mailbox", I enter the extension, it answers "Password", I enter that and it tells me "Password incorrect", whether I terminate the entries with '#' signs or not. |
| 00:24.53 | hackeron | [TK]D-Fender: SMRT? |
| 00:26.56 | jaytee | obviously not a Simpsons fan |
| 00:27.53 | *** join/#asterisk Akiraa (~Akiraaaa@79.112.12.6) |
| 00:32.22 | *** part/#asterisk Tech_Travis (~Travis@mail.techglia.com) |
| 00:32.46 | *** join/#asterisk dzup (~alex@unaffiliated/dzup) |
| 00:34.49 | *** join/#asterisk Micc (~quassel@c-98-225-57-96.hsd1.wa.comcast.net) |
| 00:35.46 | *** join/#asterisk ecrane (~ecrane@o1-69-19-166-10.static.o1.com) |
| 00:35.57 | *** join/#asterisk jetlag (jetlag@pool-70-18-186-205.pskn.east.verizon.net) |
| 00:36.27 | *** join/#asterisk jasonwert (~jasonwert@97-83-97-13.dhcp.trcy.mi.charter.com) |
| 00:39.12 | *** join/#asterisk leifmadsen (~Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 00:39.12 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 00:45.08 | Katty | HELLO THAR MADSEN |
| 00:45.24 | spenguin[work] | HALLO Katty |
| 00:45.26 | leifmadsen | KATTY! |
| 00:45.29 | Katty | ohai mister penguin |
| 00:45.35 | Katty | have you been enjoying monteray bay? |
| 00:45.39 | Katty | hugs leifmadsen |
| 00:46.14 | leifmadsen | hugs back |
| 00:48.46 | spenguin[work] | leifmadsen: isnt this good for asterisk - https://developer.skype.com/silk |
| 00:49.11 | leifmadsen | spenguin[work]: could be I suppose |
| 00:49.31 | spenguin[work] | should be a while untill its implemented every where I guesss |
| 00:50.19 | coppice | spenguin[work]: they made a big splash about SILK a year ago, but its still in limbo |
| 00:50.25 | spenguin[work] | Katty: havent gotten much time to watch em penguins today :p |
| 00:50.53 | spenguin[work] | tunes to splash cam |
| 00:50.54 | *** join/#asterisk StarFyter (~ken@ip68-97-4-47.ok.ok.cox.net) |
| 00:57.29 | *** join/#asterisk leifmadsen (~Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 00:57.29 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 00:58.27 | StarFyter | Too Quiet... |
| 00:58.37 | leifmadsen | shhhh! |
| 00:58.48 | leifmadsen | now we have to start again |
| 00:58.54 | StarFyter | LOL |
| 01:00.02 | *** join/#asterisk mbranca (~matteo@host139-217-static.224-95-b.business.telecomitalia.it) |
| 01:01.38 | StarFyter | Will asterisk work with my SPA3102? |
| 01:01.53 | *** join/#asterisk caim (~user@unaffiliated/caim) |
| 01:02.06 | coppice | only on days with a y in them |
| 01:02.08 | spenguin[work] | it should, I just got one too :p |
| 01:02.10 | *** join/#asterisk jShaf (~draco@mail.nymble.net) |
| 01:02.26 | StarFyter | Only the Y days eh... |
| 01:03.12 | StarFyter | Guess not Tomorrow. Better get on it Today. |
| 01:05.39 | *** join/#asterisk Tech_Travis (~Administr@cpe-76-168-191-127.socal.res.rr.com) |
| 01:08.26 | *** join/#asterisk lmadsen (~Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 01:08.26 | *** mode/#asterisk [+o lmadsen] by ChanServ |
| 01:11.22 | hackeron | Hmm, new problem, if I have answeronpolarityswitch=yes then asterisk thinks my outgoing calls are never answered and if I set it to no then I don't hear anything when I pick up incoming calls - any ideas? |
| 01:12.39 | Kobaz | hackeron: did you downgrade your kernel? |
| 01:12.59 | *** join/#asterisk neurosys (~neurosys@c-71-196-20-208.hsd1.fl.comcast.net) |
| 01:13.06 | hackeron | Kobaz: nope |
| 01:14.16 | hackeron | Kobaz: ah, looks like callprogress = yes fixed it |
| 01:14.57 | Kobaz | callprogress is buggy |
| 01:21.09 | jShaf | does asterisk support VVOIP (Video & Voice) ? |
| 01:21.30 | Kobaz | it depends |
| 01:21.37 | jShaf | meaning? |
| 01:21.55 | Kobaz | some stuff is supported, some stuff isn't... i'm not too familiar with video... you cah poke around on the wiki |
| 01:22.01 | jShaf | okay |
| 01:22.18 | Kobaz | ~wiki |
| 01:22.30 | jShaf | one more thing, does Asterisk support SIP <--> H323 signaling? |
| 01:22.30 | Kobaz | er |
| 01:22.39 | Kobaz | it's voip-info |
| 01:22.44 | Kobaz | the bot should have a link |
| 01:22.55 | jShaf | ah, ok |
| 01:22.59 | Kobaz | yeah there's h323, i haven't gotten it to work |
| 01:23.03 | Kobaz | i tried three different stacks |
| 01:23.07 | Kobaz | but this was in the 1.4 days |
| 01:23.18 | jShaf | okay |
| 01:23.22 | jShaf | thanks |
| 01:23.31 | Kobaz | i have two nice polycom 400 series phones that do only h323 |
| 01:23.49 | Kobaz | and a company we work with has a basement full of them |
| 01:23.56 | Kobaz | it would be nice to get h323 working |
| 01:51.07 | *** part/#asterisk korihor (~korihor@190.205.251.97) |
| 01:54.31 | *** join/#asterisk Jhirley (~Jhirley@adsl-161-99-28.mia.bellsouth.net) |
| 01:55.05 | Jhirley | o/ |
| 01:57.44 | *** join/#asterisk cvnet (~cvnet@dsl-69-172-67-161.acanac.net) |
| 01:57.46 | cvnet | hi all |
| 02:00.37 | cvnet | you can hang up a channel by using "soft hangup" from CLI correct? I tried its not working soft hangup 4748abac56f -- > 4748abac56f is not a known channel |
| 02:00.44 | cvnet | xx.xx.x.xx 141688898989 4748abac56f 00102/00000 0x100 (g729) No Tx: ACK |
| 02:01.59 | Kobaz | what the heck output is that |
| 02:02.07 | Kobaz | do: core show channels |
| 02:02.13 | Kobaz | and get the channel name |
| 02:02.26 | Kobaz | it should be something like SIP/foo or IAX2/bar or whatever you're using |
| 02:03.27 | cvnet | i tried that as well, i got the above versin from sip show channels |
| 02:03.34 | Jhirley | hey anyone ever tried the roll your own asterisk at http://www.asterisk.org/applications/pbx with cent os 5.4 vs the older 5.3 in this links ? |
| 02:03.39 | cvnet | SIP/xx.xx.x.xx-0000 (None) Up AppDial((Outgoing Line)) <- -- core show channels |
| 02:04.20 | Kobaz | okay so you do soft hangup SIP/xx.xx.x.xx-0000 |
| 02:04.27 | cvnet | localhost*CLI> soft hangup SIP/xx.xx.x.xx-0000 |
| 02:04.27 | cvnet | SIP/xx.xx.x.xx-0000 is not a known channel |
| 02:04.33 | cvnet | doesnt work |
| 02:04.42 | Kobaz | do soft hangup <tab> |
| 02:04.44 | cvnet | where xx.xx.x.xx is the ip of the trunk |
| 02:04.44 | Kobaz | what does it say |
| 02:05.04 | cvnet | soft hangup <tab> SIP/xx.xx.x.xx-0000 ? |
| 02:05.14 | Kobaz | no |
| 02:05.16 | Kobaz | hit the tab key |
| 02:05.23 | Kobaz | it should expand out possible matches |
| 02:07.00 | cvnet | woow |
| 02:07.02 | cvnet | that did it |
| 02:07.06 | cvnet | thanks Kobaz |
| 02:07.22 | Kobaz | core show channels will cut off the end of the name if it's too long |
| 02:07.25 | cvnet | 100-000001b2 <-- did not show the whole digits when i did core show channels |
| 02:07.30 | cvnet | ya |
| 02:07.30 | Kobaz | core show channels concise will show you the whole thing |
| 02:10.41 | cvnet | thanks |
| 02:11.06 | cvnet | when i do core show channels, there is not active channels, but when i do sip show channels, there is about 5 dead channels, how do i kill them? |
| 02:11.42 | cvnet | xx.xx.x.xx 14162758 753a1d7b5f0 00102/00000 0x0 (nothing) No Init: INVITE <-- like this one |
| 02:12.04 | Kobaz | that's fine |
| 02:12.06 | Kobaz | they will go away |
| 02:12.17 | cvnet | it has been there for hours |
| 02:12.26 | cvnet | the only way to get ride of them is to restart asterisk |
| 02:12.32 | Kobaz | the sip driver needs to store metadata in case there's a continueation of the dialogue |
| 02:12.50 | Kobaz | that particular invite was hours ago? |
| 02:12.57 | cvnet | so its not using cpu system resources? |
| 02:13.13 | cvnet | i didnt check this one exactly how long, but usually they stay for while |
| 02:13.21 | cvnet | i usually restart the system |
| 02:15.22 | Kobaz | it's just data |
| 02:15.26 | Kobaz | it should be gone by now though |
| 02:15.42 | Kobaz | try to see if you can reproduce it, and file a bug |
| 02:16.13 | cvnet | it takes a long time for it to go away, you c we have vpn with our provider, sometimes when the vpn is down, we get those channles staying almost for hours |
| 02:18.52 | *** join/#asterisk QubeZ (~qube@97.102.53.250) |
| 02:18.54 | QubeZ | hello all |
| 02:18.58 | cvnet | anyway Kobaz thanks |
| 02:19.10 | Kobaz | asl |
| 02:19.18 | Kobaz | cvnet: no |
| 02:19.22 | Kobaz | cvnet: np |
| 02:19.41 | *** part/#asterisk StarFyter (~ken@ip68-97-4-47.ok.ok.cox.net) |
| 02:20.59 | QubeZ | I am setting a variable in my extensions.conf in one context [default-test] like this: exten => Set(__vFASTAGI_IP=10.11.12.1), should it be available in my other context [default] in the same extensions.conf file? I set the variable in [default-test] then call the default context to handle the reset but the variable is missing when I retrieve it via: $vFASTAGI_IP |
| 02:21.10 | QubeZ | any ideas? |
| 02:24.56 | manxpower | I hope it's actually a valid exten => line. variables are not constrained by contexts. They are constrained by the CHANNEL on which the variable is set. |
| 02:26.29 | manxpower | if you run Set(VAR=BOB) as part of your dialplan for a call, then that variable will be set for the duration of the channel, with leading _ and double __ you also set it for any child channels created by the channel. |
| 02:26.32 | *** join/#asterisk Gary_B (~gary@85.211.141.97) |
| 02:26.41 | manxpower | QubeZ: pastebin a minimal example of the problem you are having and I'll look at it. |
| 02:26.56 | Gary_B | anyone familiar with vicidial here? |
| 02:27.15 | manxpower | show the extensions.conf stuff as well as the CLI output of a call that uses that dialplan |
| 02:27.53 | QubeZ | Gary_B: i use it |
| 02:28.21 | Gary_B | Any ideas how they impement the lead hopper, do they have developer documentation on it? |
| 02:28.24 | *** join/#asterisk Blackthorn (~blackthor@76-77-161-220.smyth.net) |
| 02:29.29 | cvnet | can u do wild card when using soft hangup ? like sip/* ? |
| 02:29.50 | *** join/#asterisk Torrieri (~Torrieri@nelug/crew/torrieri) |
| 02:30.50 | Blackthorn | For some reason my disa stoped working and i've been messing with it for a few days now. it will work for the first cid match but not for the rest. any suggestions? http://pastebin.com/Q3rfEXrz |
| 02:33.45 | *** join/#asterisk OrNix (~ornix@l151-249-47.static.cn.ru) |
| 02:33.59 | *** join/#asterisk QubeZ (~qube@97.102.53.250) |
| 02:34.10 | manxpower | Blackthorn: http://pastebin.com/devCdaM2 |
| 02:34.44 | QubeZ | here's my pastebin of my variable setting issue: http://pastebin.com/Vv6ERQYT |
| 02:34.53 | manxpower | just remember DISA is DANGEROUS |
| 02:35.09 | QubeZ | again, very minimal posting... I left the rest of the dialplan out because its working.. only the variable setting is not |
| 02:35.24 | manxpower | your CLI output is horrible |
| 02:35.58 | QubeZ | its sufficient to troubleshoot this issue |
| 02:36.32 | manxpower | It looks like it should work. You have confirmed THIS diaplan exhibits the problem? |
| 02:37.27 | QubeZ | yes, the error is from this dialplan. The error "unable to locate host ''" is because i am trying to set the variables. When I take it all out and set the host back to 127.0.0.1 it works or if I manually set to 10.11.12.1, also works |
| 02:37.49 | manxpower | Actually, I just confirmed that you have not confirmed that. it would be unable to work because there is no priority 1 for exten 8368 |
| 02:38.29 | QubeZ | ahh that might be it.. copy/paste error. Let me check. |
| 02:38.34 | manxpower | you can't Goto(default,${EXTEN},1) when there is no priority 1 |
| 02:38.42 | Blackthorn | thank you manx, that worked. I guess the syntax changed from the past? |
| 02:39.21 | manxpower | Blackthorn: I could not even hazard a guess. Your entire DESIGN was bad. |
| 02:41.16 | *** join/#asterisk QubeZ (~qube@97.102.53.250) |
| 02:41.27 | QubeZ | manxpower: I forgot to paste the first line: exten => 8368,1,Playback(sip-silence) |
| 02:41.47 | QubeZ | and the CLI shows the dialplan executing, its just not setting variables |
| 02:42.10 | manxpower | QubeZ: Normally you do NOT want to add extra quotes, but in your instance, you might want to try quoting the IP addresses |
| 02:42.29 | QubeZ | will give it a shot |
| 02:42.43 | QubeZ | man i need to get split tunneling work, i keep vpn'ing then dropping my internet access |
| 02:43.06 | manxpower | QubeZ: I don't see the SET happening on the CLI output |
| 02:43.27 | manxpower | QubeZ: you may be experiencing TPF. |
| 02:43.36 | Blackthorn | welp thanks anyway, g'night |
| 02:43.49 | QubeZ | TPF? |
| 02:44.01 | manxpower | Technical Problem Fatigue. It's a problem where you've been working on a problem for so long that you stop thinking and just keep reacting. Happens to the best of us. |
| 02:44.44 | QubeZ | manxpower: ya man, i've been at the real issue (Vicidial instance in VM) for 3 days now. This is part of the progress towards resolving the main issue. Almost there if I can just get this working. |
| 02:45.07 | QubeZ | because when I set things statically, it works so now I'm trying to make into variables so I can have multiple call centers |
| 02:45.17 | manxpower | QubeZ: the fact I don't see the Sets in the CLI output is a redflag. your call is not hitting those Sets |
| 02:45.41 | QubeZ | i can't see why based on what i paste bin'd |
| 02:46.04 | *** join/#asterisk fofware (~chatzilla@190.229.137.89) |
| 02:46.37 | manxpower | QubeZ: look at the cli output. |
| 02:47.40 | manxpower | it looks to me that the call is coming into [default] and not [default-test] |
| 02:56.44 | *** join/#asterisk LemensTS (~LemensTS@adsl-70-238-175-27.dsl.stlsmo.sbcglobal.net) |
| 03:08.23 | *** join/#asterisk boch (~boch@190.174.10.205) |
| 03:10.01 | boch | good night, one question, how is the proper way to send SMS using asterisk? |
| 03:11.38 | LemensTS | . |
| 03:15.47 | *** join/#asterisk JAMMAN2110 (~james@unaffiliated/jamman2110) |
| 03:23.51 | *** join/#asterisk Dittohead (~Dittohead@blender/artist/dittohead) |
| 03:27.46 | TJNII | grumbles |
| 03:29.03 | TJNII | Why'd I have to go poking around the craigslist classifieds? Now I wanna buy a boat. |
| 03:32.07 | Gary_B | anyone know how astGUIclient implement the hopper to choose the records to call? |
| 03:32.24 | Dittohead | Is there a way (outside of Asterisk) to convert sln (slinear) files to wav? |
| 03:33.53 | ChannelZ | Is 'boat' code for 'male prostitute'? |
| 03:34.21 | TJNII | No. |
| 03:34.37 | ChannelZ | Oh. I didn't know craigslist had anything else. |
| 03:34.44 | TJNII | It has spambots. |
| 03:35.03 | TJNII | Tousands and thousands of spambots. |
| 03:35.19 | TJNII | And the fortcollins version has an asshole who flags everything, but I think that feature is only local. |
| 03:36.02 | ChannelZ | isn't slin wav? (or raw?) |
| 03:37.08 | Dittohead | ChannelZ: I don't know... |
| 03:37.11 | *** join/#asterisk tkrn (~tkrn@WS1-DSL-208-102-253-13.fuse.net) |
| 03:38.18 | Dittohead | ChannelZ: I tried renaming it and playing it via vlc, but to no avail |
| 03:38.49 | Dittohead | Also I get this when I run sox on Mac OS X, "sox FAIL formats: no handler for file extension `sln' |
| 03:38.58 | TJNII | I'd look into sox first. |
| 03:44.10 | ChannelZ | I think sln is just raw (8khz 16-bit mono) |
| 03:44.42 | ChannelZ | signed, little-endian |
| 03:45.08 | *** join/#asterisk timeshell (~chatzilla@206.248.136.108) |
| 03:45.16 | timeshell | Hey... |
| 03:45.26 | timeshell | Did anyone else just have this happen to them? http://www.thecybersecurity.com/2009/citibank-uk-number-442075005000-under-a-dos-telephone-calling-attack/ |
| 03:46.02 | Dittohead | Hmm, changing the extension to .sl works. Interpreting it at 8khz it plays back very, very quickly. |
| 03:46.12 | timeshell | Someone just tried to send like a 100 calls through my server to DOS Citibank UK |
| 03:46.31 | timeshell | IP's were from China |
| 03:47.05 | ChannelZ | Did you accidentally do it as stereo? |
| 03:47.36 | ChannelZ | (it'll play twice as fast) |
| 03:48.43 | ChannelZ | We should just do ourselves a favor and chop the cable connecting us to China |
| 03:48.50 | ChannelZ | 90% of spam gone, just like that. |
| 03:51.32 | timeshell | Agreed |
| 03:51.58 | Dittohead | ChannelZ: no, it's in mono and I specified it. |
| 03:51.59 | jaytee | 80% of the trojans and viruses gone, just like that. |
| 03:52.05 | timeshell | I've been considering blocking entire networks from China. |
| 03:52.09 | timeshell | On my router |
| 03:52.20 | timeshell | Not like anyone from there has any business connecting to my servers anyway |
| 03:52.25 | jaytee | cut off Eastern Europe and you eliminate another 14%, mostly those damn Romanians |
| 03:53.57 | ChannelZ | timeshell: I already do.. and russia.. etc |
| 03:54.13 | timeshell | http://www.parkansky.com/china.htm |
| 03:54.55 | timeshell | http://www.wizcrafts.net/iptables-blocklists.html |
| 04:01.45 | doneir | i've moved a working asterisk configuration on a test server, to another server for production with the E1 card. I can't seem to get the span working correctly, although the config is exactly the same. The light is green on the card, dahdi_tool shows everything as 'OK' (no alarms). However, when trying to dial in there is no output from asterisk console and an engaged signal is given, when trying to dial out asterisk says call failed (Congestion) |
| 04:02.14 | doneir | pri intensive output shows 'SABME (set asynchronous balanced mode extended)' over and over |
| 04:02.54 | doneir | anyone have an ideas? |
| 04:02.57 | doneir | *any |
| 04:11.19 | *** join/#asterisk Cain` (~Geek@unaffiliated/cain) |
| 04:14.23 | doneir | i'm sure the switchtype is correct as it was working with this exact config and card in a test server |
| 04:30.56 | *** join/#asterisk Linuturk (~linuturk@unaffiliated/linuturk) |
| 04:31.15 | *** join/#asterisk timeshell (~chatzilla@206.248.136.108) |
| 04:31.42 | *** join/#asterisk sysreq (~sysreq@unaffiliated/sysreq) |
| 04:32.50 | TelTek | I block all of APNIC on my firewall. |
| 04:36.50 | *** join/#asterisk mintos (~mvaliyav@nat/redhat/x-hmtzxjdtqrbqmnqv) |
| 04:39.38 | *** join/#asterisk githogori (~githogori@adsl-66-123-22-146.dsl.snfc21.pacbell.net) |
| 04:39.57 | *** join/#asterisk ltd_wk (z@patwk.transact.net.au) |
| 04:43.07 | carrar | and leaft russia wide open to attack you? |
| 04:43.13 | carrar | left |
| 04:43.47 | carrar | Better block all the RIR's IP's! |
| 04:44.16 | carrar | LacNIC, RIPE, AftiNIC also! |
| 04:45.06 | carrar | APNIC is what, 80 routes |
| 04:45.54 | manxpower | doneir: is this a PRI? |
| 04:49.16 | TelTek | I use www.countryipblocks.net to block other troublemakers. |
| 04:51.22 | *** join/#asterisk SSJGotenks (~ae0377be@gateway/web/freenode/x-yxgesoktswrlsngr) |
| 04:51.48 | SSJGotenks | hi, what would you guys say is the most solid asterisk distro out there? |
| 04:51.58 | TelTek | unfortunately banning APNIC also bans Australia. |
| 04:53.41 | SSJGotenks | nevermind I googled it, just craved the human touch for a sec |
| 04:55.25 | TelTek | We were going to Google it for you, just to be touchy feely. |
| 04:56.43 | manxpower | We use the source here. |
| 04:57.08 | doneir | manxpower: yep |
| 04:57.21 | doneir | E1 PRI 10 channel |
| 04:57.59 | doneir | damn, phone conference, bbl |
| 04:59.14 | manxpower | doneir: Asterisk is not receiving PRI information from the telco |
| 05:10.54 | *** join/#asterisk uqlev (~yuriy@91.184.221.31) |
| 05:15.19 | carrar | TelTek, you can just grep APNIC, from this list: http://www.iana.org/assignments/ipv4-address-space/ |
| 05:17.10 | carrar | or if you just want to block China ;) |
| 05:17.11 | carrar | curl ftp://ftp.apnic.net/apnic/dbase/data/country-ipv4.lst; grep ": cn :" country-ipv4.lst | awk -F: '{print $2}' | sed "s/ //g" | sort -n |
| 05:20.19 | Jhirley | night |
| 05:31.15 | p3nguin | "curl ftp://ftp.apnic.net/apnic/dbase/data/country-ipv4.lst" will just output the file to stdout, so there will be no country-ipv4.lst file in which to grep. |
| 05:32.00 | p3nguin | unless I'm missing something, of course. |
| 05:33.16 | TelTek | I use a smoothwall, so I have to add the blocks manually. They are huge blocks though. most are /8 CIDR. |
| 05:33.59 | p3nguin | curl -O ftp://ftp.apnic.net/apnic/dbase/data/country-ipv4.lst would output to a file by the name of country-ipv4.lst, though. You could then grep it. |
| 05:34.02 | *** join/#asterisk SSJGotenks (~41ffe7e3@gateway/web/freenode/x-kfgqzauscbvcqmkv) |
| 05:35.51 | TelTek | 58.0.0.0/8 - 61.0.0.0/8, 112.0.0.0/8 - 126.0.0.0/8 that will block all of APNIC. |
| 05:36.37 | *** join/#asterisk Sargun (~Sargun@atarack/Staff/Sargun) |
| 05:36.57 | TelTek | 210.0.0.0/8 and 211.0.0.0/8 and 218.0.0.0/8 - 222.0.0.0/8 as well |
| 05:40.06 | TelTek | You may also want to block major cable internet providers, since they have a lot of compromised machines on them. most of my attacks come from those ISPs. |
| 05:51.01 | dddh | what is "X-Asterisk-HangupCause: Bearer capability not available"? :) |
| 05:51.20 | ChannelZ | Sounds grizzly |
| 05:52.18 | *** join/#asterisk Caplain (shayne@86-138.35-65.tampabay.res.rr.com) |
| 05:57.04 | ChannelZ | Interesting. Just went past my SMTPD log: n081022008062.avoid-mail-from-theese-hosts.sprintnet.ru:81.22.8.62 |
| 06:01.41 | dddh | what is bearer capability? |
| 06:04.10 | *** part/#asterisk benngard (~benngard@90-230-92-67-no148.tbcn.telia.com) |
| 06:09.46 | dddh | "X-Asterisk-HangupCauseCode: 58"? |
| 06:12.05 | manxpower | dddh: Asterisk uses Q.931 (ISDN) cause codes. There's an RFC that shows what Q.931 cause codes map to which SIP response codes. I don't remember the number, |
| 06:12.49 | manxpower | http://www.quintum.com/support/xplatform/network/Q931_Disconnect_Cause_Code_List.pdf for more information on Q.931 cause codes |
| 06:13.30 | manxpower | bearer cap not avail sounds like a codec issue. |
| 06:16.00 | dddh | manxpower: thank you |
| 06:16.01 | *** join/#asterisk b14ck (~comradeb1@cpe-24-24-136-239.socal.res.rr.com) |
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| 06:27.29 | pyite | i don't know if this is the right place to ask this, but not sure where else to ask, so here goes... |
| 06:27.46 | pyite | does anyone use these virtual PRI services that provide "unlimited" $19.95/month inbound calls (per channel) + cost of a DID? |
| 06:28.21 | pyite | how is this price point even possible? is there a catch? does the quality stink? seems like service providers would just pool all their DIDs to these services and spread them across as many clients as they have, adding capacity as needed... |
| 06:29.06 | Micc | pyite, Yes, but if you need 100% call quality, you might want to find a local one. We used one of those for a while, but found it had issues once or twice a month due to weird stuff happening on the internet that is out of their control. |
| 06:29.21 | *** join/#asterisk JT (~j@unaffiliated/jt) |
| 06:29.56 | pyite | Micc: but is it legit? I mean a PRI w/ 23 channels (not including minutes) is $400 minimum... or $17.39/month per channel. how can you buy them for $19/month? are people really just making $2/month per channel or something else? |
| 06:30.08 | pyite | or are these people buying cheap OC3s and reselling the channels off those? |
| 06:30.34 | Micc | I think you miss read or missunderstand that. |
| 06:30.39 | Micc | Its 19.95/month per channel. |
| 06:30.43 | pyite | right |
| 06:30.49 | pyite | so how is the provider making any money? |
| 06:30.55 | pyite | there's like no margin... |
| 06:31.23 | Micc | So for 458.85 you get 23 channels inbound |
| 06:31.28 | Micc | how is that a deal? |
| 06:31.41 | pyite | nono |
| 06:32.08 | pyite | i'm making the presumption that the guy selling it to ME (based on the sites who offer this) are just people who buy a PRI, convert it to VoIP, split it up amongst their customers and keep the difference |
| 06:32.28 | pyite | so if THAT GUY is paying $400/month for 23 channels and selling it for $458.85 for 23 channels, yeah, i'm suspecting $58.85 doesn't cover much for him :) |
| 06:32.54 | pyite | so is there some phone service higher then a PRI that brings the cost down sub-$400/month for 23 channels? like, can you buy a 198 channel "PRI" or something? |
| 06:32.56 | pyite | I guess I'm missing the term |
| 06:33.00 | pyite | what's above a PRI? maybe that's the question |
| 06:33.14 | Micc | pyite, hes hoping they don't all use it and he can sell that same thing x4 or x8 |
| 06:33.30 | pyite | Micc: ahhh... ok. so it's a utilization game strictly? |
| 06:33.50 | Micc | yeah, plus he probably gets cheap PRI if he is a CLEC. |
| 06:33.58 | pyite | interesting |
| 06:34.36 | florz | plus, buying more of it probably makes it cheaper, plus placing the equipment in some well-connected data center probably makes it cheaper, too |
| 06:34.46 | pyite | ok |
| 06:35.00 | pyite | so then this goes back to my original question, are the people who offer $19.95/month inbound "unlimited" legit... the answer appears to be "probably" |
| 06:35.02 | pyite | :) |
| 06:35.11 | pyite | just trying to gain some comfort in this pricing model based on facts about how it operates |
| 06:35.58 | florz | after all, you still have to pay for the wires to your house in addition to that |
| 06:37.26 | Micc | pyite, I think vitelity has unlimited inbound for 7.50$ a month for a single DID. |
| 06:37.32 | *** join/#asterisk dkirker-openmobl (~dkirker@openmobl/ceo/dkirker) |
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| 06:38.40 | *** join/#asterisk sourcode (~code@ppp-58-8-236-217.revip2.asianet.co.th) |
| 06:39.10 | pyite | cool |
| 06:39.11 | pyite | thanks guys |
| 06:40.59 | Micc | I think we got hit by those attackers trying to make calls to that bank. Only tried about 15-20 times though. |
| 06:45.19 | *** join/#asterisk benngard (~benngard@213.88.138.230) |
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| 07:21.18 | _saya | it seems asterisk is using glib database system .. I am having seg fault in btree .. |
| 07:23.25 | *** join/#asterisk jetlag (jetlag@pool-70-18-186-205.pskn.east.verizon.net) |
| 07:30.07 | _saya | is there any chance of conflict in registration database when I am using multiple asterisk in one server ? |
| 07:30.51 | *** join/#asterisk chilicuil (~chilicuil@unaffiliated/chilicuil) |
| 07:33.24 | *** join/#asterisk maxagaz (~maxagaz@soho2.i-xanadu.com) |
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| 07:47.36 | Terminus | hello. question, are there any database specific functions like func_odbc? |
| 07:49.33 | hardwire | ? |
| 07:51.51 | Terminus | i mean, you'd use func_odbc if you were connecting via odbc, but what if you're connecting straight to postgresql? |
| 07:58.44 | *** part/#asterisk _saya (~kamanashi@202.79.19.96) |
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| 08:03.57 | carrar | p3nguin, you are missing something of course, my .curlrc |
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| 08:34.09 | *** join/#asterisk E-bola (~bola@smtp.techbiz.dk) |
| 08:35.08 | E-bola | My newly compiled asterisk 1.6.2.5 uses 100% cpu on 1 core when running with the sample config |
| 08:35.21 | E-bola | any idea to how i can debut that? I mean just see what precisely is causing it |
| 08:35.27 | E-bola | debut=debug |
| 08:36.13 | E-bola | This is on debian squeeze 2.6.32 64bit btw |
| 08:38.06 | c0rnoTa | as i know, there is debug option in menuselect named like "SHOW LOCKS" or something like that |
| 08:38.49 | c0rnoTa | so, you can try to compile with 'do not optimize' option ON and whit 'show locks' option |
| 08:39.12 | c0rnoTa | then try to use command in CLI 'core show locks' |
| 08:39.32 | E-bola | a |
| 08:39.33 | E-bola | I see |
| 08:39.36 | E-bola | thanks i will try that |
| 08:39.40 | c0rnoTa | i don't know will there output of locks or not, but it's nice try |
| 08:40.14 | c0rnoTa | you have 100% cpu usage without calls? |
| 08:40.17 | E-bola | yep |
| 08:40.20 | E-bola | no activity at all |
| 08:40.29 | E-bola | just started asterisk with sample config and it spikes imidiately |
| 08:40.50 | c0rnoTa | ok, then try to look in full.log |
| 08:40.57 | E-bola | I did |
| 08:40.59 | E-bola | its empty |
| 08:41.03 | E-bola | besides the startup text |
| 08:41.05 | kaldemar | did you start it with -c? |
| 08:41.16 | E-bola | mmm i used the debian init script, lemme check |
| 08:41.18 | c0rnoTa | but it's uncommented in logger.conf? |
| 08:41.31 | *** join/#asterisk gr0mit (~tim@extrt.txrx.org.uk) |
| 08:41.32 | E-bola | root 28701 147 0.9 455264 14212 ? Rl 09:42 0:01 /usr/sbin/asterisk -c |
| 08:41.33 | E-bola | yep |
| 08:41.41 | E-bola | c0rnoTa: yep i uncommented full in logger.conf |
| 08:41.42 | c0rnoTa | hm.. |
| 08:41.56 | kaldemar | there's no -c in the init script. |
| 08:42.13 | E-bola | [Mar 11 09:42:29] WARNING[28701] utils.c: trying to reset empty pool |
| 08:42.16 | kaldemar | well, remove that option. |
| 08:42.21 | E-bola | thats the onyl semi weird thing i see on startyup in full log |
| 08:42.36 | E-bola | kaldemar: there is a -c in the init script, since thats how im starting it... |
| 08:42.57 | E-bola | hmmm |
| 08:43.00 | E-bola | if i just type asterisk |
| 08:43.07 | E-bola | no parameters it starts and doesnt use 100% cpu ??? |
| 08:43.28 | *** join/#asterisk creativx (~creadurex@197.82-134-19.bkkb.no) |
| 08:43.38 | c0rnoTa | wow. why they put -c option to init script? |
| 08:43.53 | E-bola | This is a virtual server |
| 08:44.00 | E-bola | can that be relevant since -c is something to due with the terminal |
| 08:44.11 | E-bola | and i guess a terminal on a virtual server mgiht act a bit different |
| 08:44.55 | c0rnoTa | it is strange |
| 08:45.14 | E-bola | this is the init script fromt he tarball, lemme check what the .deb init script looks like |
| 08:45.15 | kaldemar | is this a debian-packaged init script or the one in the source? |
| 08:45.27 | kaldemar | ah. the vanilla one from source doesn't have -c. |
| 08:45.35 | E-bola | ionteresting |
| 08:45.41 | E-bola | kaldemar: mmm wrong it does |
| 08:45.52 | kaldemar | show me |
| 08:45.52 | E-bola | i just checked. The tarball one uses -c, the debian .deb package version does not |
| 08:45.58 | E-bola | normally i use the .deb, but not for this project |
| 08:46.10 | *** join/#asterisk bn-7bc (bjarne@mac.lan.noare-1.holmedal.net) |
| 08:47.09 | *** join/#asterisk sulex (~sulex@88-149-154-95.static.ngi.it) |
| 08:47.39 | kaldemar | uh, comment out the COLOR=yes line in the init script. |
| 08:47.39 | E-bola | mmm well search the rc.asterisk.init script for "-c" |
| 08:47.41 | E-bola | third hit |
| 08:48.26 | E-bola | haha colors did the trick |
| 08:48.40 | E-bola | beats me why it causes 100% cpu though |
| 08:48.47 | *** join/#asterisk Slashman (~Slash@ariane.fimasys.com) |
| 08:49.08 | E-bola | thats another thing the tarball init script does differently |
| 08:49.09 | c0rnoTa | :) |
| 08:49.23 | E-bola | wonders why a hyper-v console doesnt support colors |
| 08:49.36 | E-bola | but ohh well dont need it at all so thankfully it was an easy fix :) |
| 08:49.37 | E-bola | Thanks guys |
| 08:50.01 | c0rnoTa | but -c is not for colors |
| 08:50.33 | kaldemar | it's for other things too, but it does enable colors in CLI. |
| 08:50.47 | *** join/#asterisk lftsy (~lftsy@leonhart.leurent.eu) |
| 08:51.03 | E-bola | if test "x$COLOR" = "xyes" ; then |
| 08:51.12 | E-bola | if colors are enabled the init script add's -c |
| 08:51.32 | kaldemar | it makes asterisk not detach from the controlling terminal or fork. |
| 08:51.36 | *** join/#asterisk Da-Geek (~Da-Geek@nat/redhat/x-kocovjqdiprelehv) |
| 08:51.55 | lftsy | Sorry, I cannot find the answer on google for the time being, regarding app_voicemail, how it is possible to set the default timezone in voicemail.conf, not on a per user bases? Thanks |
| 08:53.03 | lftsy | Oups, it's written on the default file, sorry for the inconvenience |
| 09:01.41 | *** join/#asterisk ChannelZ (channelz@burner.com) |
| 09:01.43 | *** join/#asterisk ttwhy (~tekkno@p4FECF5BD.dip.t-dialin.net) |
| 09:04.57 | E-bola | Did anybody try running several asterisk installations on 1 physical server but in seperate vservers? |
| 09:05.09 | c0rnoTa | nop |
| 09:06.22 | E-bola | Seems like a smart way to seperate multiplie companies to me |
| 09:06.25 | *** join/#asterisk angryuser (~angryuser@LPuteaux-151-42-27-99.w193-251.abo.wanadoo.fr) |
| 09:06.40 | E-bola | since having multi tenancy in asterisk isnt optimal as far as i can see |
| 09:09.08 | *** join/#asterisk krion (~seb@unaffiliated/krion) |
| 09:12.55 | *** join/#asterisk fofware (~chatzilla@190.229.137.89) |
| 09:13.09 | kaldemar | you can separate them in dialplans and peer definitions |
| 09:13.21 | *** join/#asterisk sgimeno (~chatzilla@226.Red-80-33-64.staticIP.rima-tde.net) |
| 09:14.34 | *** join/#asterisk TommyBotten (tommy@broken.pipe.no) |
| 09:14.44 | E-bola | ya i started doing that |
| 09:14.48 | E-bola | so far i got 4 companies on 1 box |
| 09:14.54 | E-bola | but sip show peers is getting very messy |
| 09:15.06 | E-bola | and the amount of activity will make it a pain to debug if it keeps growing |
| 09:23.04 | *** join/#asterisk chilicuil (~chilicuil@unaffiliated/chilicuil) |
| 09:24.43 | *** join/#asterisk CoolCat2012 (~IceChat7@189.100.217.254) |
| 09:25.12 | CoolCat2012 | hi all! |
| 09:25.29 | c0rnoTa | hello |
| 09:26.01 | CoolCat2012 | i need someone to help me... i would like someone to place a call to 200@pbx.sisdipe.com.br |
| 09:26.23 | CoolCat2012 | (i would like to know if im receiving foreign network sip calls) |
| 09:27.38 | c0rnoTa | <PROTECTED> |
| 09:27.38 | c0rnoTa | <PROTECTED> |
| 09:28.19 | c0rnoTa | nano sip.conf > allowguest=no |
| 09:28.19 | CoolCat2012 | c0rnoTa it didnt ringed here! =o/ |
| 09:28.36 | CoolCat2012 | i want to guest yes! |
| 09:29.00 | c0rnoTa | so, you will receive calls from different networx |
| 09:29.06 | CoolCat2012 | i would like that anyone can reach me whatever the network he is |
| 09:29.15 | CoolCat2012 | c0rnoTa thats what i would like |
| 09:29.48 | CoolCat2012 | c0rnoTa i saw you sucess on plcace the call, but didnt ringed here, i must have something wrong somewhere |
| 09:30.49 | c0rnoTa | CoolCat2012: http://www.pastebin.org/109514 - if it will help you |
| 09:31.09 | ChannelZ | I get 'unavailable' and dumped into your voicemail |
| 09:31.14 | *** join/#asterisk TimeRider (~steve@109.224.131.68) |
| 09:31.39 | CoolCat2012 | ChanServ very thanks. that confirm my suspect that something is wrong! |
| 09:32.24 | ChannelZ | probably whatever device your Dial()ing is wrong |
| 09:33.31 | CoolCat2012 | sorry... |
| 09:33.36 | CoolCat2012 | my fring was off... |
| 09:33.46 | CoolCat2012 | could try 201@pbx.sisdipe.com.br ? |
| 09:33.48 | CoolCat2012 | plz |
| 09:35.08 | CoolCat2012 | thankkkkkkkkkk you! =o) |
| 09:35.10 | ChannelZ | yeah your audio had some bzzzzz |
| 09:35.12 | CoolCat2012 | \o/ worked! |
| 09:35.27 | CoolCat2012 | ChanServ its a dirt and quick install |
| 09:35.35 | CoolCat2012 | (proof of concept) |
| 09:36.14 | ChannelZ | well have fun, I'm off to bed |
| 09:36.29 | CoolCat2012 | gnight! |
| 09:36.30 | c0rnoTa | ChannelZ: good night |
| 09:37.48 | redax | what is the common way to handle a call in some kind of context, and after that either Goto(Internal,${EXTEN},1) or Dial(Local/${EXTEN}@Internal) ? |
| 09:38.20 | redax | the Dial() method results 2 CDR entry |
| 09:39.23 | *** join/#asterisk maxagaz (~maxagaz@soho2.i-xanadu.com) |
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| 09:54.12 | *** join/#asterisk frea2 (~frea@dslb-088-077-194-161.pools.arcor-ip.net) |
| 09:54.17 | frea2 | hello all |
| 09:55.17 | frea2 | I'm having an issue with call transfer using ldap in realtime. Could anyone help me, please? |
| 10:00.17 | Znuff | Hi. |
| 10:00.47 | Znuff | Is there any application will sit between my client and pbx and add the privacy headers to a sip request? |
| 10:03.40 | *** join/#asterisk Mark22 (~mark@unaffiliated/mark21) |
| 10:06.25 | frea2 | hi Znuff, I joined a few mins ago and I guess people are a bit awayish. |
| 10:06.40 | Znuff | I'm guessing the same :) |
| 10:06.54 | frea2 | what do you mean by adding privacy headers to a sip request? |
| 10:07.25 | frea2 | do you want to encrypt the conversations? |
| 10:07.34 | Znuff | well, my client doesn't add them, and I'd like my number not to show up to the person I'm calling |
| 10:08.05 | Znuff | no, not encryption, just want the clid not to show up |
| 10:08.31 | kaldemar | Znuff: core show function CALLERPRES |
| 10:08.52 | Znuff | ...considering that I don't own the asterisk box. |
| 10:09.54 | *** join/#asterisk [netman] (~netman@80.31.164.76) |
| 10:10.20 | frea2 | ah I see what you mean |
| 10:10.58 | Znuff | or, if someone could suggest a client that adds those headers :-) |
| 10:11.49 | frea2 | i believe that has to be configured in the pbx |
| 10:11.49 | kaldemar | asterisk :P |
| 10:12.06 | *** join/#asterisk fofware (~chatzilla@190.229.137.89) |
| 10:12.16 | *** join/#asterisk pyite (~dschreibe@unaffiliated/pyite) |
| 10:12.35 | Znuff | frea2, it *might* me configured in the pbx, but I'm not sure it supports or not |
| 10:12.55 | Znuff | it's kind of an overkill to installa asterisk just for that : |
| 10:12.57 | Znuff | :> |
| 10:13.18 | Znuff | can't type in the morning, clearly |
| 10:13.43 | Mark22 | Hello, what could be a reason that when there is a call (SIP) over Asterisk (1.6.2) the sounds (at both ends) stop at the same time for around 10 seonds every minute? |
| 10:14.13 | frea2 | hm, so you want to supress callerid but don't have asterisk installed? what are you using then? |
| 10:15.02 | Znuff | frea2, I'm just a voip client :P |
| 10:17.30 | Znuff | I wonder why someone thought I actually needed dahdi when I just want to install asterisk |
| 10:17.33 | Znuff | *kicks ubuntu* |
| 10:17.58 | *** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/atheme.member.chainsaw) |
| 10:18.38 | *** join/#asterisk michael-i (~michael-i@141.41.40.185) |
| 10:18.42 | c0rnoTa | Znuff: for MeetMe timing |
| 10:18.52 | Znuff | do I want that? |
| 10:19.40 | Znuff | ofc, dahdi doesn't want to install |
| 10:20.01 | Znuff | Error: missing /dev/dahdi ?! |
| 10:20.26 | Chainsaw | Znuff: That generally suggests that you did not (yet?) load the kernel modules it created for you. |
| 10:20.43 | Znuff | Chainsaw, I just did apt-get install asterisk :-/ |
| 10:21.01 | Chainsaw | Znuff: So it looks like you're not done then. |
| 10:21.14 | Znuff | how can I be done if it's not even installing? |
| 10:21.34 | Chainsaw | Znuff: If you want help, I shall require the actual error messages, in context, on pastebin.ca |
| 10:21.34 | Znuff | that's what I get during the install |
| 10:22.02 | *** join/#asterisk cnu (cnu@the.ultimate.lamer.la) |
| 10:22.44 | Znuff | http://pastebin.ca/1833429 |
| 10:23.07 | Znuff | OH! |
| 10:23.15 | Znuff | it needs kernel headers |
| 10:23.24 | Znuff | Would have been nice to actually install it, if it depends on it.. |
| 10:23.37 | Chainsaw | How can you even run a system without kernel headers anyway? Silly Debian. |
| 10:23.58 | Znuff | ubuntu actually =) |
| 10:24.05 | Chainsaw | They're just as daft. |
| 10:25.35 | Znuff | compiling this takes forever |
| 10:27.04 | kaldemar | that's an old release candidate of asterisk you're installing. be warned. |
| 10:27.16 | franck34 | is there some TAPI Win32 hackers here ? :) |
| 10:27.28 | Znuff | curses at tapi win32 |
| 10:27.56 | Mark22 | debian has a policy for it kaldemar, it is often an old release with security patches done without updating the version number |
| 10:29.42 | kaldemar | Mark22: there are quite a few non-security patches that have been applied to 1.6.2 after rc2. |
| 10:30.20 | franck34 | Znuff: wow :) got a question :) i'm using TAPIEx ActiveX solution. Everyting works fine (ie: all callbacks are called : OnRing, onNewCall, ...) I'm trying to display the phone number of the caller. The only thing i succeed to fetch is the callerID, which is in fact the ID of the card which take the call on the siemens pabx (using TAPI 120 drivers). Is there any tips to get the phone number of the caller ? |
| 10:30.40 | Znuff | franck34, no, seriously - I have no idea how that bloody thing works. |
| 10:30.46 | Znuff | but I hate its guts |
| 10:31.07 | franck34 | arf |
| 10:31.08 | franck34 | :) |
| 10:31.51 | franck34 | if you know an better irc channel to ask this fu....g question, i'll take it ;) |
| 10:32.13 | franck34 | <PROTECTED> |
| 10:32.41 | tzafrir_laptop | Znuff, what version of Asterisk? What Debian? |
| 10:33.07 | Znuff | tzafrir_laptop, ubuntu-server |
| 10:33.11 | Znuff | got it working anyway |
| 10:33.40 | tzafrir_laptop | to build dahdi, try installing dahdi-source |
| 10:33.49 | tzafrir_laptop | and then run m-a a-i dahdi |
| 10:34.25 | tzafrir_laptop | Besides, 1.6.2 has app_confbridge, which does not require dahdi |
| 10:36.22 | Znuff | tell that to the package mantainer :-/ |
| 10:36.33 | Znuff | anyway, can I install freepbx easily for this? |
| 10:37.52 | frea2 | does anybody have any experience with ldap integration? call transfer used to work as my dialplan was configured in the extensions.conf. After I switched to ldap in realtime it doesn't work. when I press # I get the message I should enter the extension. But as soon as I enter a number it says there is no such extension. This happens because asterisk begins to search with the number followed by underscore e.g 2_* Ist htere anyway I |
| 10:37.52 | frea2 | <PROTECTED> |
| 10:39.54 | Znuff | how the hell do you configure this thing |
| 10:40.23 | tzafrir_laptop | Some help from hell might do the trick, I guess |
| 10:40.24 | frea2 | I recommend you read the o'reilly book |
| 10:40.39 | tzafrir_laptop | (if you meant FreePBX) |
| 10:41.25 | Znuff | frea2, I'd recommend myself not to wait 2 weeks for a book to arrive |
| 10:42.08 | Znuff | great |
| 10:42.16 | Znuff | "install instructions for asterisk coming soon" |
| 10:42.29 | b14ck | Znuff, what are you trying to do? |
| 10:42.43 | Znuff | install freepbx |
| 10:42.51 | b14ck | Which version? |
| 10:42.58 | Znuff | *any* version? |
| 10:43.05 | frea2 | if you go to the asterisk.org website you will find a link to it. You can download it for free |
| 10:43.15 | b14ck | Just download it, and install... |
| 10:43.40 | Znuff | b14ck, "installation instructions for asterisk coming soon" |
| 10:43.51 | b14ck | Znuff, I found this on their main website: |
| 10:43.54 | b14ck | http://www.freepbx.org/support/documentation/installation/install-process-for-centos-5-1 |
| 10:44.04 | Znuff | which doesn't help me at all |
| 10:44.07 | b14ck | Why not? |
| 10:44.35 | b14ck | It lists the dependencies, goes through configuring permissions, and setting up required programs. |
| 10:44.36 | b14ck | :x |
| 10:44.44 | tzafrir_laptop | Znuff, you have asterisk installed |
| 10:44.49 | Znuff | tzafrir_laptop, yes |
| 10:44.50 | tzafrir_laptop | What's the problem? |
| 10:45.06 | Znuff | trying to install freepbx, but their install instructions are... well, a mess |
| 10:45.12 | kaldemar | Znuff: #freepbx |
| 10:45.38 | kaldemar | and there's a free pdf version of the o'reilly book. |
| 10:46.14 | kaldemar | http://asteriskdocs.org/ |
| 10:46.42 | tzafrir_laptop | (which does not deal with installing freepbx) |
| 10:52.48 | frea2 | bye everyone |
| 10:52.52 | *** part/#asterisk frea2 (~frea@dslb-088-077-194-161.pools.arcor-ip.net) |
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| 11:42.39 | chiller76 | hello, i have a problem with asterisk. its not closing unused udp-ports for rtp anymore |
| 11:43.27 | chiller76 | i have 25 calls going on (25 in, 25 out) and more then 5000 udp ports are opened |
| 11:44.14 | chiller76 | about the same amount "files" are opened |
| 11:44.35 | chiller76 | i am using 1.6.2.5 compiled on a debian lenny |
| 11:45.46 | chiller76 | then after 2-3 days all rtp ports, which are assigned in rtp.conf are used.. and asterisk is not accepting calls anymore |
| 11:46.11 | *** join/#asterisk hehol (~hehol@buero-gw.dortmund.loca.net) |
| 11:47.10 | chiller76 | i am running asterisk with asterisk:asterisk (user:group) |
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| 12:15.42 | TommyBotten | Using asterisk 1.6.2.x and attended transfer, is it possible to update the end callees display with the actual callers callerID? As of now, only the secretarys number shows. |
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| 12:43.29 | benngard | hmmm, [Mar 11 13:41:50] ERROR[4742]: chan_ooh323.c:4035 close_udptl_connection: Channel has no owner <- that doesnt look to good :( |
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| 13:30.04 | ManxPower-work | ~answers |
| 13:30.05 | infobot | rumour has it, answers is Asterisk Book: http://astbook.asteriskdocs.org & SIP w/ NAT: http://www.aocomputing.net/?p=3 & Wiki: http://www.voip-info.org & docs: /path/to/src/asterisk/doc & 1.4 .vs. 1.6.0 .vs. 1.6.1: : /path/to/src/asterisk/UPGRADE*.txt |
| 13:31.26 | *** join/#asterisk maxagaz (~maxagaz@soho2.i-xanadu.com) |
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| 13:41.56 | ManxPower-work | Looks like Polycom will be releasing their SDK on Mar 1 |
| 13:42.48 | tzafrir_laptop | ManxPower-work, Mar 1 2011 or Mar 1 2012? |
| 13:44.44 | ManxPower-work | tzafrir_laptop, Exactly! |
| 13:48.09 | *** join/#asterisk coppice (~chatzilla@94.201.17.210.dyn.pacific.net.hk) |
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| 13:49.30 | *** join/#asterisk contrabanda (~contr@188.123.133.165) |
| 13:50.06 | contrabanda | Hi. Please i need help with cdr_mysql. I have installed it but it doesnot work for me. |
| 13:50.38 | *** join/#asterisk TheDavidFactor (~chatzilla@c-68-34-116-180.hsd1.md.comcast.net) |
| 13:51.42 | zamba | i'm looking for a way of sending high-quality (bitwise) audio with the lowest possible latency.. any ideas? |
| 13:52.11 | ManxPower-work | zamb Use G722 |
| 13:54.21 | *** join/#asterisk Da-Geek (~Da-Geek@nat/redhat/x-xnipzjyjlconjhzx) |
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| 13:56.31 | [TK]D-Fender | zamba: between what and what? |
| 13:56.56 | zamba | just two endpoints.. it's for radio distribution.. so i basically need an encoder at one end and a decoder at the other end |
| 13:57.11 | zamba | the stream's just going to go one way |
| 13:57.50 | *** join/#asterisk petern_ (~petern@lachesis.fuzzle.org) |
| 13:58.15 | *** join/#asterisk anonymouz666 (~anonymouz@187-28-37-118.poolip.RJO.embratel.net.br) |
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| 13:59.50 | *** part/#asterisk muiro (~muiro@unaffiliated/muiro) |
| 13:59.54 | zamba | ManxPower-work: that's a general codec good for both voice and audio? |
| 14:00.12 | zamba | 64 kbit/s is a bit too less for this, i think |
| 14:00.35 | ManxPower-work | zamba, no voice codec is good for music. |
| 14:01.11 | zamba | yeah, so i need a general codec that works for both |
| 14:01.56 | petern_ | hi hi, is it possible to do "delayed" enum lookups? generally it takes a few seconds to do the look up and people calling out don't like it. i wondered if we could do the lookup behind the scenes and use it next time the number was called? |
| 14:04.50 | hesco | Where would I find Comedian Mail documented? I do not seem to be able to check voicemail by dialing the extension provided for that purpose. It answers "Comedian Mail, mailbox", I enter the extension, it answers "Password", I enter that and it tells me "Password incorrect", whether I terminate the entries with '#' signs or not. |
| 14:05.13 | contrabanda | Need help with SIP - Realtime. I have added following line to my extconfig file : sipusers => mysql,asterisk,sipusers in res_mysql.conf i have configured all parameters. What else should i do toy use realtime sip users from mysql? |
| 14:05.37 | hesco | On article I read suggested that I may not be sending DTFM, although my ATA was configured to work with another application which does need DTFM. |
| 14:07.08 | TommyBotten | zamba: hold your horses ans wait for CELT then |
| 14:07.30 | [TK]D-Fender | zamba: then why are you asking in here? *'s core is 8khz, and G.722 is only pass-through anyway |
| 14:07.55 | TommyBotten | zamba: It is an extremely good codec, but it is not bit-stream static yet |
| 14:08.03 | zamba | [TK]D-Fender: because i've found no other place to ask.. #icecast's got too high-latency and you have too low quality :) |
| 14:08.12 | [TK]D-Fender | hesco: "extension provided"? PARDON? This is your dialplan, it does what YOU code it to do. There is nothing "provided" |
| 14:08.12 | zamba | TommyBotten: what does bit-stream static mean? |
| 14:08.34 | TommyBotten | zamba: The bit streams alters with each version |
| 14:08.49 | zamba | TommyBotten: what kind of implication does that have? |
| 14:08.54 | TommyBotten | zamba: You can se an implementation of it done in mumble |
| 14:08.55 | zamba | TommyBotten: you can't use different versions together? |
| 14:08.59 | TommyBotten | zamba: exactly |
| 14:09.06 | zamba | ah, ok |
| 14:09.23 | TommyBotten | zamba: They are constantly changing it... thus, it has not been adopted by most software yet |
| 14:09.38 | zamba | but i want to implement a encoder side where you take input from a sound card and then a decoder side where you play it back over a sound card.. |
| 14:09.46 | zamba | daemon thingy, basically |
| 14:10.06 | zamba | TommyBotten: it's a replacement for something like this: http://www.remotesoverip.com/compared.html |
| 14:10.29 | Chainsaw | hesco: If you get from the "mailbox?" prompt to the "password?" prompt, you are sending DTMF just fine. |
| 14:10.45 | TommyBotten | zamba: Ah..then you definetly want to take a look at celt |
| 14:10.57 | TommyBotten | zamba: http://www.celt-codec.org/docs/ |
| 14:11.12 | TommyBotten | zamba: See the developer podcast aswell |
| 14:11.13 | zamba | TommyBotten: but no software available for it yet? |
| 14:11.27 | zamba | TommyBotten: thanks a bunch! finally a lead in my project.. this looks very promising |
| 14:11.31 | *** join/#asterisk binbash_ (~peter@ip4da53781.direct-adsl.nl) |
| 14:11.50 | TommyBotten | zamba: There is some software... but since you write the encoder/decoder, that should not be an issue? |
| 14:12.36 | *** join/#asterisk voipmonk (~shido6@dsl-67-204-1-83.acanac.net) |
| 14:12.37 | *** join/#asterisk neurosys (~neurosys@173-9-159-177-miami.txt.hfc.comcastbusiness.net) |
| 14:12.40 | contrabanda | Need help with SIP - Realtime. I have added following line to my extconfig file : sipusers => mysql,asterisk,sipusers in res_mysql.conf i have configured all parameters. What else should i do toy use realtime sip users from mysql? |
| 14:12.43 | zamba | TommyBotten: not sure i'm capable of actually writing an encoder, but :) |
| 14:12.44 | Katty | plops down |
| 14:13.21 | Katty | infobot: squirrel |
| 14:13.22 | infobot | squirrel is probably something you can have fun with if you've got some duct tape handy... |
| 14:13.26 | *** join/#asterisk sysreq (~sysreq@unaffiliated/sysreq) |
| 14:13.30 | Katty | infobot: common squirrel |
| 14:13.40 | Katty | infobot: common squirrel is http://twitter.com/common_squirrel |
| 14:13.40 | infobot | Katty: okay |
| 14:14.02 | Katty | infobot: common squirrel |
| 14:14.03 | infobot | i guess common squirrel is http://twitter.com/common_squirrel |
| 14:14.04 | *** join/#asterisk claudeHH (~claudius@h-213.61.101.146.host.de.colt.net) |
| 14:14.06 | Katty | infobot: <3 |
| 14:14.07 | infobot | I love you. <3 |
| 14:14.10 | claudeHH | hi |
| 14:14.25 | Katty | hi |
| 14:14.32 | Katty | did you bring me cookies? |
| 14:14.35 | Katty | breakfast burrito? |
| 14:14.38 | Katty | pretty flowers? |
| 14:14.54 | claudeHH | just cookies *handover* |
| 14:14.59 | TommyBotten | zamba: The API is pretty decent |
| 14:15.00 | Katty | :>>>>>>>>>>>>>>>>>> |
| 14:15.06 | claudeHH | i think flowers aren't that good for your stomach ;) |
| 14:15.44 | Katty | who said i was going to eat them. |
| 14:16.19 | claudeHH | nobody. its just my experience, not all of them are healthy :D |
| 14:16.53 | TommyBotten | zamba: And it should be pretty stable by now ... all except the bit stream |
| 14:16.56 | claudeHH | katty would you answer me a question regarding asterisk? |
| 14:17.39 | claudeHH | is it possible to process the audio stream in realtime during a conversation? |
| 14:17.56 | [TK]D-Fender | claudeHH: In what way? |
| 14:18.13 | claudeHH | well like changing the pitch |
| 14:18.21 | claudeHH | i want to module the voice |
| 14:18.48 | [TK]D-Fender | claudeHH: http://lobstertech.com/2009/jun/07/asterisk_voice_changer/ |
| 14:19.35 | *** join/#asterisk zerohalo (~zerohalo@173-13-92-17-NewEngland.hfc.comcastbusiness.net) |
| 14:19.38 | [TK]D-Fender | claudeHH: Probably a few others you can JFGI |
| 14:19.50 | claudeHH | [TK]D-Fender: THX! i will take a look at it... |
| 14:19.55 | *** join/#asterisk brad_mssw (~brad@shop.monetra.com) |
| 14:19.56 | claudeHH | jfgi? |
| 14:20.46 | *** join/#asterisk Naikrovek (~jjohnson@unaffiliated/naikrovek) |
| 14:21.09 | [TK]D-Fender | ~jfgi |
| 14:21.09 | infobot | http://www.google.com/search?q=jfgi |
| 14:21.13 | [TK]D-Fender | :) |
| 14:21.15 | contrabanda | Need help with SIP - Realtime. I have added following line to my extconfig file : sipusers => mysql,asterisk,sipusers in res_mysql.conf i have configured all parameters. What else should i do toy use realtime sip users from mysql? |
| 14:21.37 | claudeHH | LOL |
| 14:21.41 | claudeHH | sry fender :D |
| 14:22.12 | *** join/#asterisk UQlev (~yuriy@212.50.99.8) |
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| 14:34.00 | contrabanda | hello |
| 14:34.24 | Katty | i read that as Contra Bandana |
| 14:35.09 | *** join/#asterisk Corydon76-dig (two@c-69-137-80-31.hsd1.tn.comcast.net) |
| 14:35.09 | *** mode/#asterisk [+o Corydon76-dig] by ChanServ |
| 14:36.13 | Katty | hi Corydon76-dig |
| 14:39.30 | ManxPower-work | contrabanda, not many people use Realtime |
| 14:39.42 | *** join/#asterisk JanSch (~Jan@p5B2411D5.dip0.t-ipconnect.de) |
| 14:41.01 | JanSch | hi. i know it is off topic here, but i think that this is the best existing channel to ask. can anyone help me with international dialing? |
| 14:41.54 | JanSch | microsoft posted a hotline number, 2388 9600, on their web site. usually, US telephone numbers have more digits. can that number be called from other countries? |
| 14:42.50 | LemensTS | . |
| 14:43.30 | contrabanda | is it possible to separate signaling and median in asterisk? To use asterisk only as mediagateway? |
| 14:43.35 | LemensTS | us numbers are 10 digits |
| 14:43.49 | LemensTS | im not sure what that number is |
| 14:43.54 | [TK]D-Fender | contrabanda: No. * is not a proxy, it is a B2BUA |
| 14:44.27 | *** join/#asterisk leifmadsen (~Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 14:44.27 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 14:45.52 | contrabanda | ok thanks |
| 14:46.35 | *** part/#asterisk franck34 (~franck34@sd-10138.dedibox.fr) |
| 14:48.27 | JanSch | LemensTS: ok, found out. just because it is in english and the first google result for "microsoft security hotline", it does not mean it is not in hongkong. |
| 14:48.46 | JanSch | sorry. |
| 14:49.43 | *** join/#asterisk Faithful (~Faithful@202.6.145.116) |
| 14:50.43 | coppice | all the best things are in hong kong |
| 14:50.49 | Katty | like your mom |
| 14:51.01 | coppice | no. she's in london |
| 14:51.06 | Katty | k |
| 14:51.13 | Akiraaa | What is a backplate used for in modular telephony PCI/PCIex cards? |
| 14:51.18 | *** join/#asterisk Jhirley (~Jhirley@mail.mmdlaw.com) |
| 14:51.45 | Akiraaa | I thought all it takes is the main board and the individual modules |
| 14:52.42 | [TK]D-Fender | Akiraahuh? |
| 14:54.19 | Akiraaa | Browsing some price lists and among the items for say, Sangoma A200, there are items listed as "backplate 1", "backplate 2" etc |
| 14:54.38 | *** join/#asterisk Rienzilla (rien@sinas.rename-it.nl) |
| 14:55.07 | Rienzilla | hi |
| 14:55.25 | *** join/#asterisk hetii (~lew@static-87-105-185-199.ssp.dialog.net.pl) |
| 14:55.26 | [TK]D-Fender | Akiraaa: Well there are also 2U and full-height backplates |
| 14:55.29 | Katty | are you related to godzilla |
| 14:55.31 | hetii | Hello :) |
| 14:55.34 | Akiraaa | ah wait, "backplane 1/2/3/4" |
| 14:55.45 | [TK]D-Fender | Akiraaa: AH... |
| 14:55.56 | [TK]D-Fender | Akiraaa: that is for the REMORA bridgeing cards |
| 14:56.19 | [TK]D-Fender | Akiraaa: A200's are expandable across a single PCI() slot |
| 14:56.30 | Akiraaa | the remoras are separate PCI cards that connect into the main sangoma card? |
| 14:56.37 | [TK]D-Fender | Akiraathey take up slot space but don't conenct to the MB, only to the master card <- |
| 14:56.49 | Akiraaa | ah, k |
| 14:57.33 | hetii | i use Asterisk 1.6.2.0~rc2-0ubuntu1.2, phone and asterisk are on diferent locationa and both of them are behind NAT, when i call to echotest or to other SIP user i hear voice but after 6 of this meessage SIP: Retransmitting #6 (NAT) to ... |
| 14:57.35 | Rienzilla | Hi. I'm sorry if this is somewhat off topic. I'm looking for someone with experience in provisioning snom voipphones. I have a dozen of them in use. I just bought a numer of extra phones (which have newer firmware) and it seems that the automatic provisioning doesnt work for the new phones (They get their dhcp lease but do not try to pull the configuration file from my webserver). Have the options to provision the phone somehow changed? |
| 14:57.51 | [TK]D-Fender | hetii: Go follow the guide <- |
| 14:57.52 | [TK]D-Fender | ~sipnat |
| 14:57.53 | infobot | [~sipnat] Quick guide on configuring * + SIP behind NAT : http://www.aocomputing.net/?p=3 , otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
| 14:57.56 | [TK]D-Fender | ^^^^^ |
| 14:58.26 | hetii | Spawn extension (from-internal, *43, 3) exited non-zero on 'SIP/100-b1ec7038' |
| 14:58.34 | leifmadsen | cringes every time he sees someone using asterisk 1.6.2.0~rc2-0ubuntu1.2 |
| 14:58.41 | leifmadsen | 1.6.2.0-rc2 is soooooooooooooooooooooo old |
| 14:58.57 | Akiraaa | [TK]D-Fender: thanks |
| 14:59.07 | hetii | what is the reason of this behavior |
| 14:59.13 | hetii | ? |
| 14:59.14 | leifmadsen | other end is not responding |
| 14:59.27 | [TK]D-Fender | hetii: the extension is finished. That is not an error |
| 14:59.36 | [TK]D-Fender | hetii: And it is meaningless |
| 14:59.36 | leifmadsen | [TK]D-Fender: look at the first thing he said |
| 14:59.59 | [TK]D-Fender | leifmadsen: you mean the part of where I told him to follow the NAT guide? |
| 15:00.11 | [TK]D-Fender | leifmadsen: or the next part where I tell him to head to #freepbx ? ;) |
| 15:00.13 | leifmadsen | hetii: [TK]D-Fender is right that the 2nd part just means the extension hung up, because the other end didn't respond |
| 15:00.27 | Katty | scowls |
| 15:00.38 | hetii | ok but why ? i test this on grandstrim phone also on many softphone and differen configuration and still dont know why ? |
| 15:00.38 | Katty | if people don't leave me alone i'm going to start hacking |
| 15:00.44 | [TK]D-Fender | cowls ..... then looks for a cape |
| 15:00.50 | Katty | borrows [TK]D-Fender's katana |
| 15:01.03 | Katty | locks her office door and puts phone on dnd |
| 15:01.07 | [TK]D-Fender | hetii: because * needs to be configured to work from behind NAT |
| 15:01.08 | coppice | how about cape cod? |
| 15:01.11 | Katty | mmmm cod. |
| 15:01.26 | coppice | cape of good hope |
| 15:01.35 | coppice | cape of so so indifference |
| 15:01.36 | hetii | so you say that is an NAT problem |
| 15:01.54 | Katty | cape girardeau |
| 15:02.42 | Katty | heh. BB's passed out with his head hanging off the side of the second story |
| 15:03.19 | Katty | and it looks like 5 squirrels in the yard this morning |
| 15:03.32 | Akiraaa | cape of good phone and no echo |
| 15:05.51 | *** join/#asterisk grEvenX (~even@apb9hb.ip.ssc.net) |
| 15:06.40 | *** join/#asterisk KavanS (~KavanS@static-173-50-141-22.ptldor.fios.verizon.net) |
| 15:06.58 | *** join/#asterisk ezeevoice (~rajesh@114.143.218.11) |
| 15:07.51 | *** join/#asterisk ReDNeQ (~ReDNeQ@70.114.229.58) |
| 15:07.51 | eliyahud | I have calls dropping randomly after about a minute or 2.. any ideas how to track it down? |
| 15:08.22 | *** join/#asterisk af_ (~getsmart@88-149-240-251.dynamic.ngi.it) |
| 15:09.16 | Katty | scowls. |
| 15:09.30 | Katty | so another person walks into my office and then the symantec people pick up. |
| 15:09.47 | Katty | and the boss is still talking to me so i can't answer the symantec person. he says hello once and then hangs up on me )=< |
| 15:10.14 | Katty | so now i get to sit on the phone for another hour waiting to talk to someone |
| 15:10.20 | Katty | fillets people |
| 15:10.43 | ezeevoice | How many maximum channels can an Asterisk server support? |
| 15:10.56 | afo0l | ezeevoice: depends on your hardware |
| 15:10.59 | coppice | politicians start almost filleted - no backbone |
| 15:11.26 | Jhirley | yes they are the "backbone" of our government. |
| 15:11.46 | Jhirley | which is why we are in the shitter .. ;) |
| 15:11.51 | leifmadsen | ezeevoice: asterisk has no limit -- depends on many factors such as hardware, bandwidth, features being used, etc... |
| 15:11.55 | Katty | politicians serve useful purposes |
| 15:11.57 | [TK]D-Fender | ezeevoice: The record has passed 10000 |
| 15:11.59 | ezeevoice | I have a Quad core system with 4GB RAM |
| 15:12.07 | [TK]D-Fender | ezeevoice: As leifmadsen said... MANY factors |
| 15:12.22 | Katty | course they could be better if people actually cared about politics |
| 15:12.34 | Katty | but that's not the standard generally ;) |
| 15:12.41 | [TK]D-Fender | [10:11]<ezeevoice>I have a Quad core system with 4GB RAM <- quad-core repersents a LARGE range of systems... no stats on HD performance or even what load each call may represent |
| 15:12.54 | [TK]D-Fender | ezeevoice: How about you tell us what you have in mind and see what we suggest? |
| 15:13.01 | leifmadsen | ezeevoice: http://astbook.asteriskdocs.org/en/2nd_Edition/asterisk-book-html/asterisk-book.html#asterisk-CHP-2-TABLE-1 |
| 15:13.07 | Katty | i want my phone system to make me coffee |
| 15:13.07 | *** join/#asterisk Z_God (~julius@wlan229239.mobiel.utwente.nl) |
| 15:13.12 | Katty | well, maybe not coffee. |
| 15:13.18 | Katty | i want it to make me redbull |
| 15:13.19 | [TK]D-Fender | Katty: Mine did that 5 years ago |
| 15:14.46 | *** join/#asterisk neurosys (~neurosys@173-9-159-177-miami.txt.hfc.comcastbusiness.net) |
| 15:15.54 | *** join/#asterisk hluesea (~hulusikah@88.247.127.66) |
| 15:16.07 | *** join/#asterisk putnopvut (~putnopvut@asterisk/master-of-queues/mmichelson) |
| 15:16.07 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 15:16.56 | Jhirley | If you get one from Ronco, it will cook chicken for you. |
| 15:16.58 | ezeevoice | ok, I want to design a system that can handel atleast 720 channels for making pre-recorded outbound calls |
| 15:18.01 | Jhirley | your want 720 trunks to go offhook all at once , do you know the call volume that will produce. Ouch . |
| 15:18.11 | leifmadsen | ezeevoice: you will likely be better off with multiple machines |
| 15:18.23 | [TK]D-Fender | ezeevoice: using what for PSTN connectivity? Only playing back audio? |
| 15:18.34 | leifmadsen | ezeevoice: make all systems the same, then use an external script to connect to multiple Manager interfaces (AMI) to trigger the calls |
| 15:18.42 | [TK]D-Fender | ezeevoice: And yes... I'd suggest 2-3 servers for this |
| 15:19.36 | ezeevoice | only playing back audio |
| 15:19.43 | *** join/#asterisk TheDavidFactor (~chatzilla@c-68-34-116-180.hsd1.md.comcast.net) |
| 15:20.17 | Jhirley | lets start with how many calls do you need to make , average all durraction, and the time limit for making the calls ? |
| 15:22.16 | *** join/#asterisk corretico (~laguilar@201.201.46.106) |
| 15:22.37 | ezeevoice | We will try to fill all the chanells.. average call duration 30 sec.. simultanous calla |
| 15:23.07 | [TK]D-Fender | ezeevoice: Same answer. 2-3 servers recommended |
| 15:24.03 | *** part/#asterisk benngard (~benngard@213.88.138.230) |
| 15:25.35 | *** join/#asterisk ehsjoar (~ehsjoar@c-98-245-155-132.hsd1.co.comcast.net) |
| 15:26.15 | Jhirley | so your saying you would like to make on average 30 million outbound calls a month, ( if you run the system for 12 hours, not account for time zone ). |
| 15:27.27 | contrabanda | Need help with SIP - Realtime. I have added following line to my extconfig file : sipusers => mysql,asterisk,sipusers in res_mysql.conf i have configured all parameters. What else should i do toy use realtime sip users from mysql? |
| 15:27.51 | ezeevoice | yes |
| 15:31.28 | Jhirley | good for you, be sure to make nice with your carrier. And fight for a good rate on so much outbound. |
| 15:32.04 | ezeevoice | :-) |
| 15:33.36 | ezeevoice | actually we need to have a system of 500 RRI's.. need to reduce the number of servers |
| 15:35.11 | ezeevoice | actually we need to have a system of 500 PRI's.. need to reduce the number of servers |
| 15:35.19 | Jhirley | <PROTECTED> |
| 15:35.23 | *** join/#asterisk andres833 (~andres833@190.144.75.22) |
| 15:35.45 | Jhirley | wait what ccountry is this going to be in ? |
| 15:36.02 | ezeevoice | India |
| 15:36.11 | *** join/#asterisk gr0mit (~tim@extrt.txrx.org.uk) |
| 15:37.06 | [TK]D-Fender | ezeevoice: 500 PRI? 15000 channels? |
| 15:37.16 | ezeevoice | ya |
| 15:37.30 | [TK]D-Fender | ezeevoice: You'll need a LOT more servers for that |
| 15:37.45 | Jhirley | yeah i am shaking my head too. This is massive |
| 15:38.01 | Akiraaa | If you want to build a 30 terminal voip network, would you use analog phones with ATAs (like Linksys PAP2T or SPA8000), analog phones with PCI cards (Sangoma or Digium-clone) or just plain IP phones? The cheapest option would be to use IP phones (50 EUR per), while the most expensive option seems to be the modular PCI card setup (Sangoma); there is also the option of using Digium-clone cards, but those don't have a great reputation |
| 15:38.33 | [TK]D-Fender | Akiraaa: PCI FXS = ass |
| 15:38.41 | Akiraaa | Digium original cards appear to be somewhat overpriced, unless there is something special about them |
| 15:38.45 | Jhirley | Akiraaa: get simple ship phones , save time and head ache |
| 15:38.48 | [TK]D-Fender | Akiraaa: SIP phones if you can. Analog gatways if you must |
| 15:38.53 | Nugget | any chance you have to eliminate analog phones is a win on cost, reliability, and function. |
| 15:38.56 | ezeevoice | Thats why we need to reduce the Number of servers.. and thats why looking to get 720 from one |
| 15:39.22 | [TK]D-Fender | ezeevoice: What will you connect to this server for PSTN access? |
| 15:39.46 | *** join/#asterisk titter` (~titter@c-98-208-158-125.hsd1.fl.comcast.net) |
| 15:40.45 | ezeevoice | TDM Cards - Sangoma A108 |
| 15:41.19 | [TK]D-Fender | ezeevoice: You know that you should never put more than 2 cards per server, right? |
| 15:41.26 | [TK]D-Fender | ezeevoice: that is a LOT of servers <- |
| 15:41.33 | *** part/#asterisk ehsjoar (~ehsjoar@c-98-245-155-132.hsd1.co.comcast.net) |
| 15:41.40 | *** join/#asterisk titter (~titter@c-98-208-158-125.hsd1.fl.comcast.net) |
| 15:41.41 | [TK]D-Fender | ezeevoice: And without EC you are increasing system load. |
| 15:41.41 | Akiraaa | I am inclined to use a non-modular Sangoma B600 PCI (4FXO + 1FXS) for the PSTN trunk lines |
| 15:41.54 | [TK]D-Fender | Akiraaa: Good choice |
| 15:42.41 | *** join/#asterisk codefreeze-lap (~murf@mail.parsetree.com) |
| 15:44.05 | ezeevoice | We have tested 16 PRI's (with 2 Sangoma A108 Cards) in a system for pre recorded outbound calls and it works fine.. |
| 15:44.33 | [TK]D-Fender | ezeevoice: 500 / 16 = 32 servers |
| 15:44.46 | [TK]D-Fender | ezeevoice: PRI direct into * is going to cost you on the server side |
| 15:44.55 | *** join/#asterisk jpeeler (~jpeeler@asterisk/digium-software-dev/jpeeler) |
| 15:45.00 | [TK]D-Fender | ezeevoice: lot of space, power, and systems to manage |
| 15:45.11 | [TK]D-Fender | ezeevoice: I'd recommend scaling up to something considerably larger |
| 15:45.27 | ezeevoice | we need to have 500/ 24 = 21 servers appx |
| 15:45.39 | [TK]D-Fender | ezeevoice: PCI solutions like Sangoma, Digium, etc weren't made for this scale |
| 15:45.56 | *** join/#asterisk Deeewayne (~dwayne@75.76.254.162) |
| 15:45.57 | *** mode/#asterisk [+o Deeewayne] by ChanServ |
| 15:46.02 | [TK]D-Fender | ezeevoice: 24? how do you come up with 24? |
| 15:46.41 | [TK]D-Fender | ezeevoice: 500 PRI. 2 cards per server max. thats 16 PRI per server |
| 15:46.58 | [TK]D-Fender | @ 2 x 8-port cards |
| 15:47.35 | ezeevoice | We need to have atleast 3 cards in a system |
| 15:47.59 | jaytee | can you say "interrupt hell"? sure you can |
| 15:48.25 | Naikrovek | why don't you use a SIP provider and some T3s |
| 15:49.05 | Akiraaa | Modular PCI: mother card + individual modules + backplane + remora cards = an expensive package |
| 15:49.21 | *** join/#asterisk ehsjoar (~ehsjoar@c-98-245-155-132.hsd1.co.comcast.net) |
| 15:50.36 | *** join/#asterisk Defraz (~Defraz@c72co-edge-router.fuzecore.com) |
| 15:51.29 | [TK]D-Fender | ezeevoice: Get something like a Cisco channelized DS3 gateway and do the rest SIP to your * servers |
| 15:53.03 | *** join/#asterisk Faithful (~Faithful@202.6.145.116) |
| 15:53.34 | *** join/#asterisk Geminizer (~whoami@cpe-76-180-27-4.buffalo.res.rr.com) |
| 15:56.18 | Geminizer | Hello all. Does anyone know how to drop "zombied" channels? e.g. I have a channel that has been active for at least 22 hours, and it is not used by either the intended caller or recipient. I have tried running soft hangup and hangup via AMI, yet the channel persists. I would prefer not to have to restart the actual server hardware. |
| 15:56.53 | [TK]D-Fender | gemeUse AMI to redirect the channel to a hangup exten |
| 15:58.40 | *** join/#asterisk anthm (~anthm@freeswitch/developer/anthm) |
| 16:05.27 | *** join/#asterisk nightrid3r (kvirc@41.214.209.177) |
| 16:10.31 | *** join/#asterisk rgsteele (~rgsteele@207.106.239.81) |
| 16:16.27 | p3nguin | carrar: I knew there was always that possibility, so I tried to cover all bases. |
| 16:16.45 | *** join/#asterisk nbash (~NickBenne@wsip-70-167-227-83.om.om.cox.net) |
| 16:17.25 | *** join/#asterisk atis_work (~atis_work@193.238.212.171) |
| 16:18.04 | Naikrovek | how would you do that - say you want to set up a phone system with 15000 channels |
| 16:18.15 | Naikrovek | to the outside world. what kind of hardware would you need |
| 16:19.48 | *** join/#asterisk timeshell_atwork (~timeshell@gw.lusi.on.ca) |
| 16:20.46 | nbash | Hey all...complete noob here to voip...I have a linode vps and just got asterisk and freepbx installed...everything seems to be working...but what do I need to do now? I have been searching for a tut on it but I keep finding 'how to install x' as opposed to what I need now to be able to place calls. Can anyone refer me to a doc or fill me in? |
| 16:20.59 | leifmadsen | Naikrovek: you wouldn't do it on a single system |
| 16:22.01 | mmlj4 | ~book |
| 16:22.01 | infobot | [~book] Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://astbook.asteriskdocs.org or see ~buybook |
| 16:22.06 | [TK]D-Fender | nbash: Docs on setting up FreePBX are available on their site and via any of the dozens of imminently Google-able guides out there |
| 16:22.08 | mmlj4 | nbash: read that |
| 16:22.14 | [TK]D-Fender | nbash: And FreePBX is NOT supoprted here |
| 16:22.22 | [TK]D-Fender | mmlj4: BIG PRINT <- |
| 16:22.29 | [TK]D-Fender | mmlj4: Book will do him no good |
| 16:22.36 | mmlj4 | oh, I agree with that |
| 16:23.14 | p3nguin | nbash: Got a bunch of IP phones already installed? Got some analog phones and ATAs? Want to make calls to the PSTN? You haven't really told us what you have nor what your goal is. |
| 16:24.43 | [TK]D-Fender | nbash: Add Extensions. Add outbound routes (being on a VPS that prettu much leaves ITSPs for PSTN access) |
| 16:24.45 | bmoraca_work | p3nguin: you're just supposed to know |
| 16:24.52 | [TK]D-Fender | ~nowwhat |
| 16:24.53 | infobot | So you just installed Asterisk now what? http://www.youtube.com/watch?v=FJ3oHpup-pk |
| 16:24.55 | [TK]D-Fender | :D |
| 16:25.19 | [TK]D-Fender | drmessano: Props to mah homies! |
| 16:25.26 | [TK]D-Fender | boyzzzzz |
| 16:25.40 | *** join/#asterisk MAbbas (~abbas4u@203.215.177.194) |
| 16:25.45 | wpbrown | I have a general question: Have you guys ever had a issue with Asterisk/Sangoma PRI card where cell phones can't dial extentions very well? The IVR says "Invalid number" If so where should I begin looking to tweak this issue? |
| 16:25.46 | nbash | lol my goal is just to learn more about voip...I learn better by tinkering. Since I'm on a vps I dont have any cards I can use for direct pstn connections so I'm guessing I would need to setup a sip account with someone to route the pstn calls to. I'm not looking for spoon fed instructions just a good guide. It's hard to google when you dont have any idea what to google for. The terminology is all new to me |
| 16:25.55 | drmessano | HA! |
| 16:26.01 | wpbrown | Asterisk 1.6.0.9 |
| 16:26.33 | [TK]D-Fender | wpbrown: echo, gain issues, general line noise can all be factors |
| 16:26.36 | leifmadsen | nbash: http://astbook.asteriskdocs.org |
| 16:26.37 | nbash | All I'm looking to do right now is be able to grab a voip phone and connect to my vps and make/recieve calls |
| 16:26.40 | nbash | thanks |
| 16:26.40 | p3nguin | nbash: You'll learn more about Asterisk by NOT using FreePBX. |
| 16:26.47 | [TK]D-Fender | leifmadsen: he's on FreePBX <- |
| 16:26.52 | nbash | k |
| 16:26.52 | leifmadsen | yuck |
| 16:26.53 | [TK]D-Fender | leifmadsen: Boko won't do much for him |
| 16:26.57 | [TK]D-Fender | book |
| 16:26.59 | *** join/#asterisk vettehead (~mvarner@205.127.233.238) |
| 16:27.01 | nbash | is freepbx a bad idea? |
| 16:27.03 | leifmadsen | [TK]D-Fender: it can with terms and overall information |
| 16:27.10 | [TK]D-Fender | nbash: Go Google up some "how to set up FreePBX" guides |
| 16:27.11 | leifmadsen | [TK]D-Fender: there are chapters that are information only |
| 16:27.17 | nbash | I have no gui so I though for learning it would be better |
| 16:27.26 | wpbrown | The card doesn't have a echo canceler on it.. i wonder if a sangoma card with it built on would help some? |
| 16:27.32 | leifmadsen | nbash: asterisk vanilla requires nothing but SSH access |
| 16:27.38 | [TK]D-Fender | nbash: You said you set up FreePBX. Sue sounds like you set up a GUI to me... |
| 16:27.52 | bmoraca_work | wpbrown: never had those problems on a PRI. DTMF is transmitted out of band on PRIs, from what I've experienced |
| 16:28.17 | [TK]D-Fender | wpbrown: Test with a varienty of different PSTN targets. Verify what EC you're using. Try differnt ones, etc. |
| 16:28.19 | nbash | yeah I have a gui now becuase of freepbx...I |
| 16:28.34 | [TK]D-Fender | wpbrown: Go check how many digits you miss in a long test sample |
| 16:28.43 | p3nguin | nbash: Why do you feel that you need some type of GUI? |
| 16:28.47 | [TK]D-Fender | nbash: It now owns your configs. Go lern it. |
| 16:28.49 | [TK]D-Fender | learn* |
| 16:29.26 | [TK]D-Fender | nbash: FreePBX isn't for learning. Its for people who want to follow its cookie cutter thought-process without any actual thinking |
| 16:30.07 | p3nguin | nbash: Read the book, don't use FreePBX, and start the tinkering in which you previously mentioned. |
| 16:30.12 | pmhaddad | well to be fair freepbx is also a good admin tool for end users who just need to modify extension settings and whatnot. that's why we use it. |
| 16:30.53 | [TK]D-Fender | pmhaddad: There is no "just" it owns yuor entire config, not just a little part |
| 16:31.14 | [TK]D-Fender | pmhaddad: "just need to modify extension settings and whatnot" = controls everything |
| 16:31.31 | pmhaddad | yeah, but we still run our own routing and dialplans |
| 16:31.36 | pmhaddad | just takes some php |
| 16:31.40 | nbash | because as of now I have 0% knowledge of working with a voip system. Now I can copy/paste my inital configs to get something setup but to me thats the same thing as letting a gui "cookie cut" a config for me. This is just a test box and nothing is going into production. Once I learn more about voip configurations I prolly wouldnt use a front end such as freepbx |
| 16:32.12 | p3nguin | nbash: You won't learn about them if you do use FreePBX. |
| 16:32.21 | *** join/#asterisk mvarner_ (~mvarner@205.127.233.238) |
| 16:32.28 | Kobaz | wow |
| 16:32.32 | [TK]D-Fender | bn-7bc: "VoIP configurations" is actually NOTHING in the big scheme of things. |
| 16:32.37 | Kobaz | my 0 button just stopped working on my desk polycom |
| 16:32.39 | [TK]D-Fender | <PROTECTED> |
| 16:32.46 | Kobaz | that's the second button failure i've seen this year |
| 16:32.58 | Kobaz | the 1 button stopped working on a customer phone a few months ago |
| 16:33.01 | [TK]D-Fender | nbash: Setting up a phone device is about 10 lines max in sip.conf. Any ITSP is the same. |
| 16:33.03 | p3nguin | nbash: Read the book. Learn about things. Configure it the old fashioned way. |
| 16:33.06 | MAbbas | There are difference b/w AMI event messages for asterisk 1.4 and 1.6 versions. My question, Is AMI event messages same across all version of Asterisk 1.6.X?? |
| 16:33.08 | AlanBell | nbash: so do you have a SIP or IAX2 softphone registering as an extension to it? |
| 16:33.22 | [TK]D-Fender | nbash: the DIALPLAN is 95% of * |
| 16:33.22 | leifmadsen | MAbbas: that's what the UPGRADE.txt file is for |
| 16:33.31 | [TK]D-Fender | nbash: And that has nothing to do with "VoIP". |
| 16:33.41 | p3nguin | alanbell: extensions don't register. Don't try to confuse the guy on his first day. |
| 16:33.46 | [TK]D-Fender | nbash: VoIP is just a means of getting a call into *. What you DO with it is another matter. |
| 16:33.55 | leifmadsen | MAbbas: it should answer those types of questions. Note that 1.6.0 vs. 1.6.1 vs. 1.6.2 are all MAJOR version changes, much like between 1.2 vs 1.4 |
| 16:35.21 | atis_work | russellb: a theoretical question on JACK_HOOK |
| 16:35.43 | Kobaz | yeap, the zero key is definitly dead |
| 16:36.51 | Naikrovek | leifmadsen: yeah i know you couldn't do it on a single system, but you wouldn't have numlines/16 of servers |
| 16:36.55 | atis_work | russellb: if i add JACK_HOOK before Dial(), i.e. normal call, and then JACK (the software) crashes.. would that interrupt audio streams? |
| 16:37.06 | *** join/#asterisk TheDavidFactor (~chatzilla@c-68-34-116-180.hsd1.md.comcast.net) |
| 16:37.27 | nbash | lol, 0 knowledge guys...everyone has to start somewhere. My first car was a 90 accord...doesn't mean I knew how to work on the engine. I get that to you using a cookie cutter app is "yuck" but I can either spend the next month reading trying to figure out what the hell is what or get a system working and learn about dialplans (no idea what the hell those are). lol, it was my mistake to come to irc and ask for guidance when I dont eve |
| 16:37.27 | timeshell_atwork | Can someone please point to any good documentation on fax detection with dahdi? |
| 16:37.43 | Tim_Toady | whats going on with this error when i try to compile * from svn? ‘struct dahdi_spaninfo’ has no member named ‘bpvcount’ |
| 16:37.48 | *** join/#asterisk maxagaz (~maxagaz@soho2.i-xanadu.com) |
| 16:37.55 | mmlj4 | Kobaz: "if you need help, don't other trying to dial the operator" |
| 16:37.56 | *** join/#asterisk murdock_ut (~chatzilla@mail.kimballequipment.com) |
| 16:38.08 | Kobaz | totally |
| 16:38.18 | p3nguin | nbash: It should take you a day to read "the book." |
| 16:38.20 | mmlj4 | what brand of phone? |
| 16:38.23 | Kobaz | mmlj4: polycom |
| 16:38.35 | Kobaz | this is the second polycom that a number key has died on |
| 16:38.43 | Kobaz | well, second one i've seen |
| 16:38.44 | mmlj4 | i haven't heard of buttonitis on a polycom before |
| 16:38.45 | AlanBell | nbash: so start by getting a couple of softphones making calls to each other, then move on to the outbound trunks later |
| 16:38.56 | *** join/#asterisk cusco_ (~trilili@2001:0:53aa:64c:1c52:32f4:2ac0:762d) |
| 16:39.03 | cusco | hi |
| 16:39.27 | Naikrovek | guaranteed it's fixable |
| 16:39.35 | [TK]D-Fender | bn-7bc: No, coming here is a good thing. Lets you know what to expect from your approach and what others are out there. |
| 16:39.39 | [TK]D-Fender | nbash: No, coming here is a good thing. Lets you know what to expect from your approach and what others are out there. |
| 16:39.50 | [TK]D-Fender | bn-7bc: Sorry for the contant bad autocompletes BTW |
| 16:40.05 | MAbbas | <@leifmadsen>: well, actullay I am writing a parser for AMI events for * 1.6.X version.. So it would help if you can tell me |
| 16:40.18 | [TK]D-Fender | nbash: You haven't stated what any of your longer term goals are. |
| 16:40.38 | raden_work | <PROTECTED> |
| 16:40.51 | russellb | atis_work: yes |
| 16:41.13 | [TK]D-Fender | nbash: Learning * isn't a big deal... and using * to "learn VoIP" is pretty weak. * implementation of any given protol isn't the greatest. *'s value comes from the sum of its parts, and VoIP protocols are only a part. |
| 16:41.29 | raden_work | enter it really fast i get -- Incorrect password '33003300' for user '110033' (context = default) |
| 16:41.37 | *** join/#asterisk mintos (~mvaliyav@nat/redhat/x-bchzfjkuhjxhlqvz) |
| 16:41.39 | raden_work | anyone have any idea A???? |
| 16:41.46 | p3nguin | nbash: Dialplan consists of contexts and extensions. Extensions consist of priorities and commands. Extensions are rules that say what asterisk is supposed to do when a call matches the phone number that is being called. Extensions can be letters as well as numbers. Phone should NEVER be referred to as extensions, because extensions do not always route calls to phones. |
| 16:41.58 | [TK]D-Fender | raden_work: Only reason to get double is allowing auto with inband and OOB active simultaneously |
| 16:42.06 | [TK]D-Fender | raden_work: Pick an OOB mode and stick with it |
| 16:42.13 | raden_work | oob ? |
| 16:42.13 | cusco | I am looking to MixMonitor a conference room |
| 16:42.15 | cusco | a meetme |
| 16:42.25 | p3nguin | oob = out of band |
| 16:42.35 | [TK]D-Fender | cusco: "core show application meetme" <--------- |
| 16:42.35 | ManxPower-work | cusco, you don't do that. You mixmonitor a channel |
| 16:43.10 | raden_work | Force RFC2833 Out-of-Band DTMF ??? |
| 16:43.12 | ManxPower-work | cusco, actually [TK]D-Fender is correct (he usually is) |
| 16:43.22 | raden_work | that enabled |
| 16:43.26 | raden_work | DTMF Method SIP INFO |
| 16:43.41 | ManxPower-work | raden_work, SIP INFO != RFC2833! |
| 16:44.01 | cusco | :( |
| 16:44.02 | murdock_ut | [TK]D-Fender: Got a dialplan question regarding the Dial application. If A calls B and B transfers A why does the dialplan continue to execute for B? I thought once dial application gets an answer the dialplan ends there. |
| 16:44.02 | cusco | but then I will have lots of recordings that have the same audio... |
| 16:44.20 | ManxPower-work | raden_work, sounds like you have relaxdtmf=yes |
| 16:44.22 | cusco | ahhh record conference |
| 16:44.41 | raden_work | ManxPower-work, on my phone |
| 16:44.58 | ManxPower-work | raden_work, It sounds like you have relaxdtmf=yes in Asterisk. |
| 16:45.47 | *** join/#asterisk ocbill (~none@static-63-238-185-39.t1.cavtel.net) |
| 16:45.53 | ManxPower-work | raden_work, most people never have to mess with their phone settings except to set the username/password and server IP. |
| 16:46.00 | [TK]D-Fender | murdock_ut: Shouldn't in general. SHOW ME |
| 16:47.05 | murdock_ut | [TK]D-Fender: Let me put some stuff together for you. |
| 16:47.45 | *** join/#asterisk Tech_Travis (~tech_trav@208.179.137.131) |
| 16:50.05 | cusco | [TK]D-Fender: and ho can I change the default recording name and format? or where is ${MEETME_RECORDINGFORMAT} set? |
| 16:50.44 | Katty | ARJLJGLKDJSFLKJSDLFKJ |
| 16:50.47 | Katty | my NOSE KEEPS ITCHING |
| 16:50.50 | Kobaz | Qwell: okay, bug still exists in trunk |
| 16:50.52 | Kobaz | er |
| 16:51.45 | *** join/#asterisk aruntomar (~aruntomar@61.17.193.163) |
| 16:52.36 | aruntomar | is anyone using redfone's fonebridge with asterisk? |
| 16:52.54 | Katty | never heard of it before |
| 16:53.17 | ManxPower-work | aruntomar, very few people use them, but from what I've seen they are very reliable and work well with Asterisk. |
| 16:53.20 | aruntomar | Katty: it's a pri appliance, rather than using pri card from digium or sangoma |
| 16:53.49 | ManxPower-work | It uses TDMoE for low latency "wireline" emulation for the connection to Asterisk. |
| 16:54.40 | Katty | aruntomar: k |
| 16:55.06 | aruntomar | ManxPower-work: right, it works well. we've been using it for quite sometime. |
| 16:55.30 | *** join/#asterisk michael-i (~michael-i@141.41.40.185) |
| 16:55.34 | *** join/#asterisk smanek (~smanek@rrcs-24-213-164-136.nyc.biz.rr.com) |
| 16:56.03 | coppice | I wonder how many of those redfone boxes they have sold? it seems very niche |
| 16:56.20 | aruntomar | ManxPower-work: the problem is that asterisk shows that pri is down and the indicator in redfone is also red, and when i tried to check the fonulator status, it shows this : /etc/init.d/fonulator status |
| 16:56.20 | aruntomar | fonulator dead but subsys locked |
| 16:56.29 | aruntomar | is this normal? |
| 16:56.38 | Katty | i wouldn't know. i haven't worked with it. |
| 16:56.52 | murdock_ut | [TK]D-Fender: http://pastebin.com/hqJSvgF0 |
| 16:57.01 | *** join/#asterisk rttrey (~trey@209.208.18.121) |
| 16:57.21 | ManxPower-work | aruntomar, I've seen that error message on other systems. I bet you are not running the "service" command as root, are you? |
| 16:57.47 | aruntomar | ManxPower-work: the telephone line is not working at our end and the telecom company says that everything is fine from their end |
| 16:57.48 | Katty | but it would be very interesting to find out, coppice |
| 16:58.09 | *** join/#asterisk Tim_Toady (~moi@77.49.114.31.dsl.dyn.forthnet.gr) |
| 16:58.09 | aruntomar | ManxPower-work: i'm running it as root |
| 16:58.11 | murdock_ut | [TK]D-Fender: 1124 calls 1134. 1134 answers 1123 does a blind transfer to 700 by dialing *3 700. 1123 hears the parking spot and then hears weasles. |
| 16:58.31 | *** join/#asterisk luckyaba (~lucky@ip72-194-218-169.sb.sd.cox.net) |
| 16:58.37 | ManxPower-work | murdock_ut, you should not hear the parking spot when doing a blind transfer |
| 16:59.05 | ManxPower-work | that's why you should not blind transfer to part a call. |
| 16:59.31 | murdock_ut | ManxPower-work: Well, I don't know what to tell you. I do. Let me try an attended transfer then..... |
| 16:59.39 | *** join/#asterisk Maliuta (~scooby@kiev.lusan.id.au) |
| 17:00.06 | aruntomar | <PROTECTED> |
| 17:00.08 | p3nguin | Perhaps your blind transfer key is configured to do an attended transfer. |
| 17:00.41 | [TK]D-Fender | aruntomar: Virtually noone uses TDMoE here. |
| 17:00.47 | [TK]D-Fender | arethose devices are not recommended |
| 17:00.52 | ManxPower-work | aruntomar, I don't have any ideas. My biggest issue with the idea of using a Redfone unit is that so few people use them, if I have a problem then support could be a problem. |
| 17:00.53 | murdock_ut | ManxPower-work: I get an error when trying to do an attended transfer... *2 700 |
| 17:01.22 | ManxPower-work | murdock_ut, fix your attended transfers, then come back |
| 17:01.27 | *** join/#asterisk Alagar (~Administr@122.164.43.155) |
| 17:01.30 | aruntomar | ManxPower-work: looks like we made a bad choice of getting the device then :( |
| 17:01.36 | murdock_ut | ManxPower-work: I'll look into it. |
| 17:01.49 | ManxPower-work | aruntomar, I personally don't think they are bad devices. Have you contacted Redfone support. |
| 17:02.12 | aruntomar | ManxPower-work: they are offline as of now, i'll mail them |
| 17:03.53 | [TK]D-Fender | murdock_ut: Interesting... seems like a features.conf quirk. What phones are you using? |
| 17:04.09 | coppice | ManxPower-work: I object to anything that red. Red on the outside of the box. Red on the inside of the box. Red PCBs |
| 17:04.35 | ocbill | quick question - i installed asterisk via synaptic on ubuntu 9.10 |
| 17:04.42 | ManxPower-work | They must be made by Communists. |
| 17:04.45 | murdock_ut | [TK]D-Fender: Snom 370, 320, and bria softphone. |
| 17:04.53 | murdock_ut | [TK]D-Fender: For testing. |
| 17:04.55 | p3nguin | ocbill: yes. |
| 17:05.02 | ocbill | for some reason, all vm boxes play the default greeting |
| 17:05.04 | drmessano | Yet people don't object to SCCP being so close to CCCP |
| 17:05.10 | ocbill | even after i record a new greeting |
| 17:05.13 | ocbill | meaning |
| 17:05.24 | [TK]D-Fender | murdock_ut: FFS stop using DTMF transfers. Use the native features on your phone |
| 17:05.34 | ocbill | when I dial the extension, it says "please leave your message after the tone" |
| 17:05.43 | ocbill | it never plays the users' greeting |
| 17:05.56 | p3nguin | ocbill: Did the user record a greeting? |
| 17:05.58 | *** part/#asterisk ManxPower-work (~manxpower@216.186.151.147) |
| 17:06.05 | p3nguin | ocbill: Successfully, I mean. |
| 17:06.18 | murdock_ut | [TK]D-Fender: Running 1.6.1.17. I do not see this issue with a 1.2 build. |
| 17:06.18 | murdock_ut | [TK]D-Fender: Haven't tried the 1.60 series. |
| 17:06.18 | murdock_ut | [TK]D-Fender: 1.6.0.x series. |
| 17:06.23 | ocbill | I've verified that the recorded greeting does exist in var/spool... when i d/l it, it plays correctly, etc - but asterisk never tries to "use" it |
| 17:06.26 | p3nguin | ocbill: Can you find the sound file where you recorded the greeting? |
| 17:06.34 | p3nguin | ocbill: okay. |
| 17:06.48 | ocbill | figured it might be a simple setting i'm missing |
| 17:06.51 | [TK]D-Fender | murdock_ut: it is an * issue... but one you could avoid by using phone features instead |
| 17:07.13 | Akiraa | Anyone using Beronet cards for GSM-IP connectivity? |
| 17:07.14 | ocbill | i also verified ownership and permissions on those /var/spool files |
| 17:08.08 | [TK]D-Fender | Obeliks: because you never told * which to play when calling VoiceMail() <- |
| 17:09.01 | murdock_ut | [TK]D-Fender: Well, if it is an * issue then I would like to help get it fixed. |
| 17:09.11 | murdock_ut | [TK]D-Fender: I was just making sure my understanding was correct about the Dial application. |
| 17:09.16 | p3nguin | ~bugs |
| 17:09.23 | p3nguin | ~issues |
| 17:09.24 | infobot | Yep, got 'em. |
| 17:11.14 | ocbill | [TK]D-Fender: I'm assuming you meant ocbill when you sent the vm message. if so, how do i tell asterisk to play the message they recorded, and not the default * greeting every time? |
| 17:11.32 | *** join/#asterisk Talirk81 (~tt@rrcs-67-78-39-22.sw.biz.rr.com) |
| 17:12.29 | Talirk81 | If I plan to use extconfig.conf to load my extensions with RealTime, do i have to load all extensions via real time, or will it simply append to the ones defined in extensions.conf. I dont see anything in the docs on voip-info.org that says one way or another. |
| 17:13.27 | murdock_ut | manxpower: FYI, I got attended transfer to work and it does the same thing. :) |
| 17:15.42 | *** join/#asterisk rare1980_ (~rare1980@115.186.30.152) |
| 17:16.01 | rare1980_ | hi all... i am using nat for virtualBox.. and i m using SIP on virtual machine. for sip i have forward port 5060 and port 10000 to 10010 ... sip use port 5060 to establishing call and sip use port 10000 to 10010 after establishing call... i can get connected with sip using port 5060 which is forward to 3121... but after call is established i can't hear other end which mean port 10000 to 10010 is not getting forward.... |
| 17:16.01 | rare1980_ | <rare1980_> this is how i am fowrding port 10000 to 10010 |
| 17:16.18 | rare1980_ | > VBoxManage setextradata "elastix" "VBoxInternal/Devices/e1000/0/LUN#0/Config/sip1/HostPort" 10001 |
| 17:16.26 | rare1980_ | VBoxManage setextradata "elastix" "VBoxInternal/Devices/e1000/0/LUN#0/Config/sip1/GuestPort" 10001 |
| 17:16.34 | rare1980_ | VBoxManage setextradata "elastix" "VBoxInternal/Devices/e1000/0/LUN#0/Config/sip1/Protocol" UDP |
| 17:16.39 | rare1980_ | and soo on till 10010 |
| 17:16.45 | rare1980_ | plz help |
| 17:17.16 | p3nguin | port 3121? Why a non-standard port? |
| 17:17.39 | rare1980_ | i have forward port 5060 to 3121 and it is working fine |
| 17:17.39 | p3nguin | Also, configure rtp.conf to the same port range that you have forwarded. |
| 17:17.50 | rare1980_ | they are same |
| 17:18.07 | p3nguin | Note that I didn't ask if it worked, I asked why you chose a non-standard port. |
| 17:18.14 | *** join/#asterisk smanek (~smanek@static-71-249-221-129.nycmny.east.verizon.net) |
| 17:18.45 | rare1980_ | coz i have fward 5060 to 3121 |
| 17:18.47 | rare1980_ | thts y |
| 17:19.09 | Talirk81 | why not just foward 5060 to 5060 on the internal nat'd ip? |
| 17:19.14 | p3nguin | Alright, smart ass. Good luck with your issues. |
| 17:19.24 | rare1980_ | humm |
| 17:19.28 | rare1980_ | let me try like this |
| 17:19.34 | Jhirley | lol |
| 17:19.41 | Jhirley | ;) |
| 17:20.58 | p3nguin | talirk81: You're kidding, right? That might actually make sense! |
| 17:21.36 | Talirk81 | Any happen to know if when using RealTime in extconfig.conf if all extensions must by in the DB, or if astersik load the extensions.conf and then merges the data from the DB into it |
| 17:22.11 | Talirk81 | yeah i know stupid me p3nguin why do something that would make it helpful to other to create a mental image of :P |
| 17:22.16 | drmessano | which mean port 10000 to 10010 is not getting forward <--- or improper externhost/localnet settings, or or or or |
| 17:23.33 | p3nguin | I think I would rather configure vbox with host interface networking and skip all the nat crap through the vbox engine. |
| 17:24.12 | Talirk81 | The simplest solution is ussally the bes afterall |
| 17:24.19 | Talirk81 | *best |
| 17:27.04 | *** join/#asterisk voipmonk (~shido6@dsl-67-204-1-83.acanac.net) |
| 17:28.27 | ocbill | well.. since I may have stumped you guys (as I clearly am), I may try simply installing from source to see if I get different results |
| 17:28.51 | [TK]D-Fender | ocbill: "core show application voicemail" <---- |
| 17:29.11 | p3nguin | ocbill: voicemail won't be compiled any differently in the package than you will compile it outside of a package. |
| 17:29.16 | [TK]D-Fender | ocbill: Doesn't matter if you installed from packages. What matters is what you put in your dialplan |
| 17:29.40 | *** join/#asterisk anonymouz666 (~anonymouz@189.24.141.138) |
| 17:29.51 | *** join/#asterisk smanek (~smanek@static-71-249-221-129.nycmny.east.verizon.net) |
| 17:31.57 | ocbill | sweet... [TK}D-Fender, you rock! (you too p3nguin) |
| 17:35.30 | *** join/#asterisk DrGeek (~geek@c-24-21-244-173.hsd1.or.comcast.net) |
| 17:35.32 | DrGeek | g'day all. |
| 17:36.05 | DrGeek | is this the right channel to ask about digium T1 PPP configuration? |
| 17:37.50 | redax | do you have any idea, how to solve the next problem: selected callers can call a DID, and if the authorized calls the DID, another Call should be done _without_ answering the incoming call, and play some DTMF to the 2nd call |
| 17:38.42 | redax | DrGeek: last time I have the versy same problem with E1 and HDLC, the sethdlc utility not worked for me, where you could set the protocol for the given interface to ppp/hdlc |
| 17:39.04 | DrGeek | redax, did you get it working eventually? |
| 17:39.55 | redax | no. had to use an old cisco router |
| 17:40.01 | redax | check https://issues.asterisk.org/view.php?id=16910 |
| 17:44.34 | *** join/#asterisk Alagar (~Administr@122.164.43.155) |
| 18:08.40 | *** join/#asterisk Skeeter- (skeeter@190-141.cgocable.ca) |
| 18:10.33 | Katty | WHY SO QUIET |
| 18:10.49 | Naikrovek | good question |
| 18:10.56 | Naikrovek | it's nice out |
| 18:10.57 | Naikrovek | finally |
| 18:10.58 | *** join/#asterisk VaGoNeTaS (~cchavez@200.111.138.170) |
| 18:11.01 | VaGoNeTaS | hello everyone |
| 18:11.04 | Naikrovek | 6 months of crap now it's nice out |
| 18:11.12 | Naikrovek | i've been hanging out in the fresh air myself |
| 18:11.19 | VaGoNeTaS | i have a little doubt |
| 18:11.31 | VaGoNeTaS | im using the blacklist feature on asterisk which is working just perfect |
| 18:11.45 | VaGoNeTaS | but my question is, im able to blacklist an entire prefix? |
| 18:11.58 | VaGoNeTaS | ex, i've blocked 9374400 |
| 18:12.07 | VaGoNeTaS | i would like to block everything after 937 |
| 18:12.14 | VaGoNeTaS | it is possible? |
| 18:12.35 | p3nguin | http://www.bored.com/photos/thefamilyguyyearbookphoto.html |
| 18:13.26 | *** join/#asterisk garymc (~chatzilla@host86-135-19-162.range86-135.btcentralplus.com) |
| 18:13.28 | redax | it's snowing there |
| 18:13.30 | Naikrovek | p3nguin: "UH OH!!" |
| 18:13.41 | redax | nice spring days :-o |
| 18:14.20 | *** join/#asterisk jaskew (~jdaskew@netblock-75-79-182-110.dslextreme.com) |
| 18:15.02 | p3nguin | It was so nice over here last evening, I actually got out the motorcycle and rode around for a while. 75F yesterday, but only 62F today. |
| 18:15.38 | VaGoNeTaS | omg |
| 18:15.42 | VaGoNeTaS | another earthquake |
| 18:15.46 | VaGoNeTaS | im in Chile |
| 18:15.49 | VaGoNeTaS | crap |
| 18:15.52 | p3nguin | right now?~ |
| 18:15.58 | VaGoNeTaS | yes, just right now |
| 18:16.01 | p3nguin | ack! |
| 18:16.03 | p3nguin | take cover! |
| 18:16.18 | VaGoNeTaS | 2 hours ago, there was 3 earthquakes |
| 18:16.22 | VaGoNeTaS | 7.2 |
| 18:16.24 | VaGoNeTaS | 6.9 |
| 18:16.37 | redax | p3nguin: 75F? where are you living? |
| 18:16.37 | VaGoNeTaS | normal in Chile |
| 18:16.43 | p3nguin | Those are pretty strong quakes. |
| 18:16.47 | *** join/#asterisk atis_work (~atis_work@193.238.212.171) |
| 18:16.51 | VaGoNeTaS | those are normal here |
| 18:16.53 | p3nguin | redax: southern Illinois USA |
| 18:16.56 | VaGoNeTaS | 8.8 arent |
| 18:17.03 | VaGoNeTaS | like feb 27th |
| 18:17.04 | redax | you can't say 7.2mag is normal, anywhere in the world |
| 18:17.17 | VaGoNeTaS | redax : read about Chile |
| 18:17.35 | VaGoNeTaS | that is very normal for us |
| 18:17.42 | VaGoNeTaS | but 8.8 like Feb 27th are not |
| 18:17.45 | redax | did a few weeks before |
| 18:18.03 | VaGoNeTaS | that was STRONG |
| 18:18.20 | VaGoNeTaS | also the tsunami that came after |
| 18:18.30 | VaGoNeTaS | 1 thousand ppl died coz of that |
| 18:18.37 | redax | but the 27th of feb earthquake was unique because it was long as well, not just strong |
| 18:18.42 | VaGoNeTaS | instead of that, 1 thousand ppl is a little |
| 18:18.48 | VaGoNeTaS | compared to HAITI |
| 18:18.50 | VaGoNeTaS | 250k |
| 18:19.02 | Katty | redax: actually, you can |
| 18:19.11 | VaGoNeTaS | yes, looong |
| 18:19.12 | VaGoNeTaS | 3 mins |
| 18:19.21 | Katty | redax: but it's not above the water level |
| 18:19.55 | p3nguin | 3 minutes worth of 8.8 ... no wonder it tore up a bunch of stuff and created a tsunami. |
| 18:20.08 | Katty | hawaii didn't get a tsunami |
| 18:20.14 | Katty | if that's what you're talking about |
| 18:20.21 | Katty | which probably isn't |
| 18:20.24 | redax | yeah. |
| 18:20.36 | leifmadsen | at least we can't blame asterisk |
| 18:21.10 | p3nguin | lol http://www.bored.com/photos/didnotwinmillion.html |
| 18:21.11 | VaGoNeTaS | we are prepared |
| 18:21.12 | drmessano | We can't? |
| 18:21.25 | drmessano | I already filed a bug report |
| 18:21.26 | Katty | yeah looks like Libertador general bernardo o'higgins got a few |
| 18:21.30 | VaGoNeTaS | did u saw the teleton |
| 18:21.35 | Katty | 6.0, 6.7, 6.9, 3 hours ago |
| 18:21.39 | redax | but haiti got tsunami as well? |
| 18:21.48 | *** join/#asterisk adnc (~numer@unaffiliated/adnc) |
| 18:21.54 | VaGoNeTaS | Katty where u from? |
| 18:22.02 | drmessano | "Kernel panic during tsunami under 1.6" |
| 18:22.17 | drmessano | Gah |
| 18:22.23 | Katty | looks lik eastern turkey got one at 6.1 on the 8th |
| 18:22.35 | Katty | and on the 5th southern sumatara, indonesida got a 6.8 |
| 18:22.35 | VaGoNeTaS | 6.1 its nothing babe |
| 18:22.37 | VaGoNeTaS | nothing at all |
| 18:22.44 | Katty | don't call me babe. |
| 18:22.45 | VaGoNeTaS | the mayor quake was on the 60 |
| 18:22.47 | Katty | that is highly inappropriate |
| 18:22.49 | drmessano | "System unresponsive during tsunami after upgrade to 1.6.1.17" |
| 18:22.53 | VaGoNeTaS | ok babe |
| 18:22.58 | drmessano | ^^ Asterisk, blamed |
| 18:23.04 | VaGoNeTaS | the mayor quake was on the 60 |
| 18:23.09 | VaGoNeTaS | 9.5 , for 10 minutes |
| 18:23.20 | VaGoNeTaS | that is reaaaaaaaally long |
| 18:24.34 | Katty | the one from chili isn't up on the list yet |
| 18:24.42 | Katty | if it was very recent |
| 18:24.47 | p3nguin | mmmmmmmm... chili. |
| 18:24.49 | VaGoNeTaS | im from chili |
| 18:24.52 | VaGoNeTaS | and im IN chili |
| 18:24.53 | VaGoNeTaS | right now |
| 18:24.56 | VaGoNeTaS | babe |
| 18:24.57 | VaGoNeTaS | =) |
| 18:25.27 | Katty | you clearly think i'm female. |
| 18:25.42 | Katty | also, there was one in Southern Alaska this morning |
| 18:26.31 | *** join/#asterisk bsaxon (~bsaxon@12.68.234.174) |
| 18:26.56 | Naikrovek | when i lived in Alaska, 7.2 earthquakes were common there |
| 18:27.09 | Naikrovek | happened a lot more than in california |
| 18:27.14 | Naikrovek | and they were much larger, on average |
| 18:27.16 | Naikrovek | but no one lives there |
| 18:27.17 | Jhirley | is it day time there or night time ? |
| 18:27.20 | Naikrovek | and they do zero damage |
| 18:27.22 | Naikrovek | usually at night |
| 18:27.41 | drmessano | Igloo's resist vibration |
| 18:27.46 | Naikrovek | chile is central time zone i think, maybe mountain or pacific |
| 18:27.47 | Katty | Naikrovek: same here on the new madrid fault |
| 18:27.48 | drmessano | and velociraptor attacks |
| 18:28.00 | Katty | Naikrovek: there's just not a lot on it, for the most part |
| 18:28.06 | Naikrovek | Katty: i remember feeling one on new madrid fault when a was a young lad of 10 or so |
| 18:28.15 | p3nguin | naikrovek: Should be GMT-4. |
| 18:28.17 | Naikrovek | i felt it when i was standing up |
| 18:28.22 | Katty | Naikrovek: yeah, that one made the river run backwards ;) |
| 18:28.23 | Naikrovek | -4? |
| 18:28.33 | p3nguin | 14:28 |
| 18:28.42 | *** join/#asterisk ehsjoar (~ehsjoar@c-98-245-155-132.hsd1.co.comcast.net) |
| 18:28.48 | Naikrovek | Katty: i'm not that old. you're thinking of the one in the 1800s i think |
| 18:28.55 | Katty | no. |
| 18:28.57 | VaGoNeTaS | katty is a female name babe |
| 18:28.57 | VaGoNeTaS | =) |
| 18:28.59 | Katty | but! there was one at that time too |
| 18:29.04 | VaGoNeTaS | and your real name says Angela |
| 18:29.05 | VaGoNeTaS | =) |
| 18:29.09 | Znuff | Hi. |
| 18:29.10 | p3nguin | lol |
| 18:29.12 | Katty | that one perminently altered the path of mississippi |
| 18:29.15 | Naikrovek | Katty: the one in the 80s/90s made the river run backwards? |
| 18:29.18 | *** join/#asterisk lanning (~lanning@208.87.235.224) |
| 18:29.34 | drmessano | http://www.xkcd.com/711/ <-- Good to know |
| 18:29.35 | Katty | Naikrovek: everytime there is an earthquake on the new madrid over about 5 or 6, the river runs backwards |
| 18:29.42 | Naikrovek | Katty: didn't know that |
| 18:29.44 | Katty | Naikrovek: and then it floods |
| 18:29.47 | Naikrovek | heh |
| 18:29.55 | Katty | i'm sure p3nguin loves the flooding ;) |
| 18:29.58 | Naikrovek | i was in stl over the weekend |
| 18:29.59 | *** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/atheme.member.chainsaw) |
| 18:30.02 | Znuff | I just installed AsteriskNOW under virtualbox, I created an extension but apparently I can't register to the box. Any tips? I don't see it listening on port 5060 |:-/ |
| 18:30.05 | Katty | we've not had an earthquake of any magnitude in awhile tho |
| 18:30.06 | Naikrovek | probably was pretty close to you, katty |
| 18:30.17 | Katty | Znuff: please visit #asterisknow |
| 18:30.23 | Katty | Znuff: you are not in the correct channel |
| 18:30.27 | Znuff | pfft |
| 18:30.30 | Naikrovek | heh |
| 18:30.36 | Naikrovek | nothign like a well placed 'pfft' |
| 18:30.37 | Katty | hi Chainsaw |
| 18:30.38 | Naikrovek | afk a few |
| 18:30.45 | Znuff | Branching these things sucks. |
| 18:31.08 | Chainsaw | Hello there Katty :) |
| 18:31.24 | Katty | Naikrovek: also, where abouts st. louis were you? |
| 18:32.24 | [TK]D-Fender | Znuff: Does your VB install have a routed IP or is it NAT'd through your host? |
| 18:32.26 | p3nguin | Oh, timeanddate.com seems to think Chile is currently in DST, so the time is actually 15:32, not 14:32. |
| 18:32.38 | Znuff | [TK]D-Fender, it's bridged, I can ssh to it just fine |
| 18:33.09 | [TK]D-Fender | Znuff: may not be enough |
| 18:33.19 | Znuff | how come? |
| 18:33.22 | [TK]D-Fender | Znuff: SIP uses UDP which may get screwed by it |
| 18:33.35 | Znuff | IAX2 uses udp, too? |
| 18:33.42 | p3nguin | screwed by what? |
| 18:33.53 | Znuff | (screwed by bridgin?!) |
| 18:34.00 | Znuff | bridging, I mean |
| 18:34.04 | redax | so don't you have any good idea how to make a new call and play DTMF, without answering the orignal call? |
| 18:35.17 | p3nguin | If it's bridged, the networking is protocol agnostic. |
| 18:35.22 | [TK]D-Fender | redax: dial with M() and don't accept the call. |
| 18:35.30 | [TK]D-Fender | redax: or originate another. |
| 18:36.23 | Znuff | I don't seem to see any "ports" definition in the config files |
| 18:36.46 | p3nguin | znuff: sip.conf? |
| 18:36.53 | Znuff | and I see asterisk is only listening on port 5038 |
| 18:36.55 | [TK]D-Fender | Znuff: rtp.conf begs to differ, and SIP's ports are well documented |
| 18:37.06 | [TK]D-Fender | Znuff: and I see you showing us nothing at all |
| 18:37.15 | p3nguin | znuff: bindport |
| 18:37.20 | Znuff | [TK]D-Fender, I'm showing you exactly what you asked for. |
| 18:37.21 | redax | [TK]D-Fender: exten => s,1,Dial(SIP/509,,D(ww55)) |
| 18:37.37 | redax | now I doing this.. but that's pick up the call. |
| 18:37.44 | *** join/#asterisk RobH (~robh@rob.tech.wikimedia.org) |
| 18:37.45 | Znuff | p3nguin, shouldn't asterisknow have set that up automatically? just saying... |
| 18:37.49 | [TK]D-Fender | Znuff: You say it isn't listening. I don't see you showing us how you're looking at it |
| 18:38.07 | p3nguin | znuff: I woul have expected it to be configured, yes. |
| 18:38.09 | [TK]D-Fender | [13:35]<[TK]D-Fender>redax: dial with M() and don't accept the call. |
| 18:38.13 | Jhirley | folks is there a resource for asteriskNOW documentation ? |
| 18:38.15 | *** join/#asterisk hfb (~hfb@pool-98-112-219-90.lsanca.dsl-w.verizon.net) |
| 18:38.16 | redax | [TK]D-Fender: how can I `don't accept the call' ? |
| 18:38.19 | Naikrovek | Katty: stl zoo |
| 18:38.23 | Znuff | there: http://pastebin.ca/1834165 |
| 18:38.31 | [TK]D-Fender | redax: "core show application dial" |
| 18:39.06 | Znuff | here: http://pastebin.ca/1834166 |
| 18:39.32 | redax | [TK]D-Fender: Dial(M(answermakro)) normally calls the answermakro when the called party picks up . |
| 18:39.34 | Katty | Naikrovek: ahhh the zoo |
| 18:39.44 | *** join/#asterisk Defraz (~Defraz@corp.fuzecore.com) |
| 18:40.32 | Katty | Naikrovek: how did you like the life-size decorations at the entrance? |
| 18:40.35 | redax | [TK]D-Fender: what should I do in the macro, to avoid answering the call ? |
| 18:41.55 | redax | [TK]D-Fender: ok. got it . "..before connecting |
| 18:43.03 | redax | cool, thanks |
| 18:43.21 | Znuff | So... any ideas? |
| 18:46.45 | rare1980_ | hi experts |
| 18:46.59 | rare1980_ | if i need to change rtp port 10000 |
| 18:47.02 | *** join/#asterisk grEvenX (~even@cC0FD00C3.dhcp.bluecom.no) |
| 18:47.07 | rare1980_ | too 30000 |
| 18:47.19 | rare1980_ | then i will only do this on /etc/asterisk/rtp.com |
| 18:47.23 | rare1980_ | rite? |
| 18:47.32 | rare1980_ | or is there any other conf file i need to change? |
| 18:47.33 | p3nguin | You want to have starting port of 30000 and the end port of 20000? |
| 18:48.01 | rare1980_ | coz i am using running asterisk on virtual server and i m using nat |
| 18:48.08 | rare1980_ | on host machine |
| 18:48.34 | rare1980_ | now i m tryig to froward rtp port |
| 18:48.46 | rare1980_ | soo is it i m doing rite or wrong? |
| 18:48.51 | Znuff | Shouldn't this thing be MUCH easier? and easy to set up "in 15 minutes"? :-/ |
| 18:48.57 | *** part/#asterisk aruntomar (~aruntomar@61.17.193.163) |
| 18:49.18 | p3nguin | znuff: UDP doesn't LISTEN, so your grep would filter out the necessary lines. Try "lsof -i udp" instead. |
| 18:49.21 | rare1980_ | Znuff: are you talking to me? |
| 18:49.29 | rare1980_ | ooh ok |
| 18:49.43 | Znuff | p3nguin, still not listening |
| 18:49.54 | p3nguin | exactly |
| 18:50.01 | drmessano | 15 minutes? |
| 18:50.09 | Znuff | I mean, lsof -i udp doesn't show anythign asterisk related |
| 18:50.13 | rare1980_ | p3nguin: any idea? |
| 18:50.33 | p3nguin | rare1980_: Three words... |
| 18:50.41 | p3nguin | rare1980_: Host Interface Networking |
| 18:51.54 | rare1980_ | yes |
| 18:51.57 | rare1980_ | but i using nat on host interface |
| 18:52.11 | p3nguin | If you are using NAT, you aren't using Host Interface networking. |
| 18:52.17 | drmessano | ~sipnat |
| 18:52.18 | infobot | [~sipnat] Quick guide on configuring * + SIP behind NAT : http://www.aocomputing.net/?p=3 , otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
| 18:52.21 | drmessano | Thats your issue then |
| 18:52.26 | drmessano | Not ports |
| 18:52.49 | rare1980_ | let me check this link |
| 18:52.50 | p3nguin | Host Interface is when you have bridged the vbox devices to the physical device and your virtual machine is now on your LAN. |
| 18:52.51 | Naikrovek | Katty: the metal sculptures? awesome |
| 18:54.01 | Znuff | I seriously don't understand why this damn thing works and I have no idea how to debug it. I see the packets arriving at the asterisk box, but I get a "recovery on timer expiry" in my client |
| 18:54.27 | Znuff | doesn't work* |
| 18:56.49 | Naikrovek | Katty: we stayed at the mariott near the airport. that place is epic |
| 18:56.51 | *** join/#asterisk ecrane (~ecrane@o1-69-19-166-10.static.o1.com) |
| 18:59.13 | Katty | you go see the penguins? |
| 18:59.51 | Jhirley | The penguins are my cat, they are ebil ! |
| 18:59.58 | p3nguin | http://www.bored.com/photos/dieselfriedchicken.html |
| 19:05.43 | *** join/#asterisk ruben23 (~ITadmin@122.55.48.243) |
| 19:06.40 | Znuff | Finally :-/ |
| 19:07.55 | Katty | humm. |
| 19:08.24 | Katty | what do you guys do with sip phones. do you dial xxx or whatever, do an #include /file/ or include => elsewhere |
| 19:08.51 | p3nguin | For what purposes? |
| 19:09.00 | Katty | cleanliness |
| 19:10.25 | leifmadsen | I tend to #include sip-phones.inc |
| 19:10.34 | leifmadsen | (in sip.conf) |
| 19:10.43 | leifmadsen | I never use extension numbers for devices |
| 19:11.16 | leifmadsen | devices are always uniquely identified only by device, and always abstract people and extension numbers from them (i.e. use MAC address to uniquely identify a device) |
| 19:11.33 | p3nguin | How does Joe place a call to Sally? Most desk phones don't have letters and symbols. |
| 19:11.38 | leifmadsen | dialplan is the part that determines how to add the logical layer of people and extensions to the device |
| 19:12.03 | leifmadsen | p3nguin: that's what the dialplan finds out -- I never said what the person dials is the unique identifier of the phone |
| 19:12.20 | leifmadsen | that information could be static, or dynamic like from AstDB or via func_odbc |
| 19:13.02 | leifmadsen | What happens when you have a device that you want shared amongst many people? (sales persons) -- all those people have different extensions, and maybe you want them to be able to login and out of the device |
| 19:13.21 | leifmadsen | naming it [305] in sip.conf does you no good |
| 19:14.16 | leifmadsen | Set(DEVICE=${DB(phones/${EXTEN})}) |
| 19:14.28 | leifmadsen | ${DEVICE} could then return SIP/0004f2040001 |
| 19:14.46 | leifmadsen | or something like that -- I use func_odbc for that kind of stuff, but the same logic can be applied to the AstDB |
| 19:14.55 | Katty | cute. |
| 19:15.09 | Jhirley | Folks, I am having a issue, I installed *NOW. It did not see my ethernet card, so I upgraded the Kernel and dependencies. But I am now getting dahdi error on boot, ethernet is working but I may have a larger issue. Is there an *NOW documentation source ? Look at the website, there is no *NOW docs, it the same asterisk.org refrence for 1.6.1.6 . Any input welcome, and TIA .. |
| 19:15.22 | Qwell | Jhirley: yum update |
| 19:15.42 | p3nguin | Dahdi has to be built with the same version of kernel that you are running. |
| 19:16.07 | Jhirley | So should I update my Dahai as well ? |
| 19:16.13 | Qwell | yes.. |
| 19:16.17 | p3nguin | Do what qwell said. |
| 19:16.48 | p3nguin | If you updated the kernel, dahdi is now built with the wrong kernel. The new packages should resolve it. |
| 19:16.54 | Jhirley | I did yum update but that does a large update and I did not see errors on boot but the freepbx interface was way borken. |
| 19:17.24 | *** join/#asterisk atis_work (~atis_work@193.238.212.171) |
| 19:18.23 | Jhirley | Are you guys saying I would I be better off with the freepbx issues ? |
| 19:18.36 | p3nguin | Did you already update everything? |
| 19:19.40 | Jhirley | I did that first, and after I saw the interface was borken, I save the image and rolled it back to a starting point. AKA I LOVE Clonezilla |
| 19:20.43 | Jhirley | Right now I a stock *NOW with an yum -y kernel.i686 and dependanceis. |
| 19:21.11 | Qwell | yum update |
| 19:21.11 | p3nguin | If everything is working except dahdi, update that next. |
| 19:23.15 | p3nguin | [tk]d-fender: http://www.bored.com/photos/cluepon_4.html |
| 19:23.41 | Jhirley | Well everything else is less borken, there are still errors on freebpx on the box. I guess I will go all in and yum -y upgrade then deal with the freepbx issues. |
| 19:23.48 | drmessano | Ugh.. Sound on Ubuntu is killing me.. I reboot more than effing Windows |
| 19:24.53 | p3nguin | jhirley: I installed AsteriskNOW 1.5.0 on a 32-bit system, and everything seemed to be working perfectly. |
| 19:25.01 | Jhirley | that does not sound like Ubuntu, there must be something seriously wrong. |
| 19:25.17 | drmessano | That statement doesnt make any sense |
| 19:25.22 | drmessano | Yes, there is something wrong |
| 19:25.46 | [TK]D-Fender | p3nguin: Comedy GOLD |
| 19:26.47 | Jhirley | p3nguin: this all started with *NOW not seeing my ethernet card, so I updated the kernel to see the card and so forth and how we are on the road to hell. |
| 19:26.53 | Znuff | Hmm. So. I've added an extension on my box. I can register it. I added a sip trunk. It registers. I set up everything to go trough that sip trunk, but when I dial something, I get number invalid |
| 19:27.47 | *** join/#asterisk Netgeeks (~chris@173-11-68-157-SFBA.hfc.comcastbusiness.net) |
| 19:27.57 | [TK]D-Fender | Znuff: FreePBX is NOT supported here... |
| 19:27.59 | [TK]D-Fender | ~freepbx |
| 19:28.00 | infobot | [~freepbx] FreePBX is unable to be supported here. It is made up of complex dialplans and scripts which can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking there, or fpbx |
| 19:28.01 | [TK]D-Fender | ^^^^^^^^ |
| 19:28.17 | Znuff | Then may I ask why is the default asterisknow gui? |
| 19:28.51 | [TK]D-Fender | Znuff: You are in the car radio department. Not the "Complete care for my car" department |
| 19:28.58 | [TK]D-Fender | Znuff: We support a piece. |
| 19:29.05 | Micc | I was so happy to find the new cli command sip show channelstats |
| 19:29.10 | Znuff | Fine! |
| 19:29.20 | Znuff | removes the asterisknow virtual machine |
| 19:29.25 | Micc | Then I found out it had wrong output for jitter. but there is a patch. now I'm happy again. |
| 19:30.15 | Znuff | downloads ubuntu-server and installs asterisk |
| 19:30.41 | Micc | Znuff, had znuff of asterisk runnning on a virtual machine? |
| 19:31.07 | Znuff | uh? |
| 19:31.19 | bmoraca_work | just put together an HP DL380 G6 server...that is one NICE piece of equipment |
| 19:31.50 | Micc | I haven't used ubuntu server in production yet, but I was thinking about it for the next server I build. |
| 19:31.58 | Znuff | meh, ubuntu's .torrents are so slow |
| 19:32.12 | Micc | setting up raid mirroring is super easy on ubuntu server 9.10 |
| 19:32.44 | Micc | Znuff, where are you downloading from? I got 1.5mb/s the other day. |
| 19:32.52 | *** join/#asterisk rare1980_ (~rare1980@115.186.0.54) |
| 19:32.56 | Znuff | tried .torrent |
| 19:32.57 | Micc | I downloaded the iso in about 8 minutes. |
| 19:33.03 | Znuff | switched to http, 9MB/sec :P |
| 19:33.10 | Micc | nice. |
| 19:34.44 | Micc | Is there any reason why I wouldn't want to change the sip and rtp COS to 6? I can understand not going to 7, but 6 seems like the right setting to me. |
| 19:34.59 | *** join/#asterisk ajohnson (~aaron@65-122-4-130.dia.static.qwest.net) |
| 19:35.02 | *** part/#asterisk ajohnson (~aaron@65-122-4-130.dia.static.qwest.net) |
| 19:39.49 | Katty | infobot: asterisk standard extensions |
| 19:39.49 | infobot | methinks asterisk standard extensions is http://www.voip-info.org/wiki/view/Asterisk+standard+extensions |
| 19:40.40 | Naikrovek | asterisk runs fine on modern virtual machines, fyi |
| 19:40.47 | Naikrovek | i think he was tired of running asterisknow |
| 19:41.46 | adnc | after calling a 0800xxx number i saw in the CDR that this calls billing-time was 0 and the duration was regular duration. how does asterisk know that this call did not cost anything? |
| 19:44.03 | *** join/#asterisk gego (~gego@dyndsl-091-096-092-071.ewe-ip-backbone.de) |
| 19:46.10 | *** join/#asterisk drmessano (~nonya@pdpc/supporter/active/drmessano) |
| 19:46.21 | drmessano | &%$%#$ |
| 19:46.32 | Katty | orly |
| 19:46.53 | drmessano | Had to WINBOOT, I mean reboot |
| 19:47.47 | Katty | is the syntax for timeout (t) still Set(TIMEOUT(digit)=9999999999999999999999999999999999999999999999999999999999999999999etc) |
| 19:48.04 | Katty | well i guess it's {} |
| 19:48.08 | Katty | perhaps |
| 19:48.11 | Katty | i'll figure that out later |
| 19:48.21 | adnc | Katty, how is your mom? |
| 19:48.43 | Katty | she's superduper, thanks. |
| 19:49.48 | Katty | tests |
| 19:50.12 | Jhirley | cape with a big red S and everything , AWESOME ! |
| 19:50.28 | adnc | Katty, do you know where to set on asterisk billing informations? i wonder how my asterisk installation knows that 0800 numbers do not cost anything |
| 19:51.17 | *** join/#asterisk ttwhy (~tekkno@p4FECED79.dip.t-dialin.net) |
| 19:52.31 | Katty | adnc: i have no idea what you're talking about |
| 19:52.47 | Katty | Naikrovek: ping |
| 19:53.22 | *** join/#asterisk dddh (~b23104cc@pdpc/supporter/active/dddh) |
| 19:53.29 | Naikrovek | pong |
| 19:53.31 | dddh | is back |
| 19:53.53 | Naikrovek | oh lemme rephrase: |
| 19:53.55 | Naikrovek | Katty: pong |
| 19:53.56 | Katty | Naikrovek: do you have a sample snippet for a functional timeout(digit) |
| 19:54.13 | Naikrovek | Katty: don't, sorry. stuck on trixbox here for the moment |
| 19:54.16 | adnc | i do CDR with sqlite and there is an entity that is called duration and also billsec, when i do a call billsec field is filled with the billable time, if i do a call to a toll-free number like 0800xxx than the billsec time is 0 |
| 19:54.18 | Naikrovek | which is why i dont offer a lot of help |
| 19:54.29 | Katty | Naikrovek: oooh, bummer :< |
| 19:54.46 | Naikrovek | Katty: my dialplans are rediculous, and system generated |
| 19:54.52 | Katty | sounds icky. |
| 19:54.55 | Katty | tugs on p3nguin's sleeve |
| 19:54.57 | Naikrovek | but if you need syntax help maybe i can offer you some assistance |
| 19:54.59 | SuPrSluG | anyone know the default registration time for polycom phones? and how to change? |
| 19:55.11 | Katty | Naikrovek: k, -> |
| 19:55.17 | Naikrovek | default registration time? |
| 19:55.35 | Naikrovek | SuPrSluG: do you mean how often asterisk checks to see if the phone is there? |
| 19:56.24 | SuPrSluG | Naikrovek: no how long the phone waits to try and register |
| 19:56.51 | SuPrSluG | think it's like 3600. not sure if that's seconds or minutes |
| 19:57.57 | drmessano | seconds |
| 19:58.22 | [TK]D-Fender | Katty: Set(TIMEOUT(digit)=3) |
| 19:58.50 | Katty | [TK]D-Fender: i was trying to figure out what timeout type i wanted |
| 19:59.09 | Katty | [TK]D-Fender: how does 'digit' compare with absolute or response |
| 19:59.29 | [TK]D-Fender | Katty: Not sure how we're supposed to know what you want. Perhaps how to DO what you want, but not to figure it out for you |
| 19:59.46 | Katty | [TK]D-Fender: you should know what i want. |
| 19:59.48 | Katty | [TK]D-Fender: i want chocolate. |
| 20:00.04 | [TK]D-Fender | Katty: Response is how long till the 1st digit. "digit" is how long between subsequent dgitis befor acepting that input is finiished |
| 20:00.11 | Katty | [TK]D-Fender: and you shouldn't get your panties in a wad if i didn't include you in my conversation |
| 20:00.25 | Carlos_PHX | Well, here's the craziest thing I've ever seen from an IP phone: http://www.speedextreme.com/temp/mar/upsidedown.jpg |
| 20:00.55 | Carlos_PHX | Glad the customer sent a picture, because otherwise I'd tell him he needed to stop smoking his lunch. |
| 20:01.14 | *** part/#asterisk lynxsys (~lynxsys@82-71-19-61.dsl.in-addr.zen.co.uk) |
| 20:01.28 | Qwell | Carlos_PHX: wtf |
| 20:01.42 | *** join/#asterisk RobH_ (~robh@cpe-173-169-30-118.tampabay.res.rr.com) |
| 20:01.50 | Carlos_PHX | Indeed. |
| 20:01.53 | Katty | looks |
| 20:01.57 | *** join/#asterisk lynxsys (~lynxsys@82-71-19-61.dsl.in-addr.zen.co.uk) |
| 20:02.01 | Qwell | that isn't even a "flip the screen" fix |
| 20:02.05 | Katty | that's hot |
| 20:02.15 | Carlos_PHX | searches for <invert_screen_enable> in configs. |
| 20:03.07 | Carlos_PHX | And it's not a new phone which could have been assembled wrong. In production, just woke up that way. |
| 20:04.12 | Katty | so are there any super cute sound files i can use for timeout to play before it hangs up |
| 20:04.23 | Katty | like KBAI! |
| 20:04.49 | Carlos_PHX | I like the monkeys, our customers do too. Not sure if that's all that cute. |
| 20:06.29 | Naikrovek | Katty: search video game sample sites. there are tons |
| 20:07.16 | Katty | :> |
| 20:07.52 | rare1980_ | sip listen on port 5060 and 10000 to 20000 please could you tell on which port IAX listen? |
| 20:08.09 | [TK]D-Fender | rare1980_: 4569 |
| 20:08.12 | [TK]D-Fender | rare1980_: iax.conf <- |
| 20:08.20 | rare1980_ | only one port ? |
| 20:08.39 | rare1980_ | not like sip 10000 to 20000 |
| 20:08.48 | rare1980_ | like range of ports? |
| 20:09.07 | Jhirley | Katty: random and totally off topic , I emailed the link to the suiette birdfeeder thing from like 2 months back. She loved it. !. |
| 20:09.21 | Jhirley | Thanks |
| 20:09.38 | rare1980_ | [TK]D-Fender: ?? |
| 20:09.41 | rare1980_ | plz can u tell |
| 20:09.42 | rare1980_ | ? |
| 20:10.01 | [TK]D-Fender | [15:08]<rare1980_>not like sip 10000 to 20000 <- No, NOT like SIP |
| 20:10.20 | [TK]D-Fender | rare1980_: 1 port. Media & signaling |
| 20:10.28 | Carlos_PHX | rare1980_: That's why IAX can solve some NAT issues and why it was created. |
| 20:10.29 | Katty | Jhirley: oh the hanging melon one? |
| 20:10.50 | *** part/#asterisk gego (~gego@dyndsl-091-096-092-071.ewe-ip-backbone.de) |
| 20:11.14 | [TK]D-Fender | Katty: Hanging on a melon here with you.... |
| 20:11.45 | Jhirley | wasn't it a pomegranate or something ? I had a skin and the red berries. |
| 20:12.41 | Katty | yeah you can use pomegranate |
| 20:13.05 | Katty | depending on how much of a demand there is for suet, you might do better using a smaller fruit |
| 20:13.14 | Katty | also depends on the weather ;) |
| 20:13.18 | afo0l | can anyone recommend a good tutorial to run asterisk with dahdi and HFC-S based ISDN cards? |
| 20:13.19 | Jhirley | [TK]D-Fender: you missed it . |
| 20:13.25 | afo0l | documentation is rare |
| 20:14.10 | Jhirley | no only good documentation, bad documentation is all over . |
| 20:14.37 | afo0l | yeah, it is |
| 20:14.41 | *** join/#asterisk TheDavidFactor (~chatzilla@c-68-34-116-180.hsd1.md.comcast.net) |
| 20:14.59 | afo0l | and it |
| 20:15.08 | afo0l | 's horribly decentralized |
| 20:15.11 | Znuff | yeah, someone should really take down voip-info.org |
| 20:15.16 | afo0l | isn't there an official wiki or something? |
| 20:15.32 | Jhirley | there is , just no documentation on it. |
| 20:15.32 | *** join/#asterisk stix (~stix@x1-6-00-1e-2a-87-8d-aa.k1.webspeed.dk) |
| 20:16.00 | Jhirley | hence i am bald from pulling my hair out. |
| 20:16.01 | Znuff | For a project of asterisk's size... I wonder why there's no good docs on it, yet :-/ |
| 20:16.04 | [TK]D-Fender | [15:15]<Jhirley>there is , just no documentation on it. <- there isn't |
| 20:16.17 | Qwell | ~book |
| 20:16.17 | infobot | [~book] Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://astbook.asteriskdocs.org or see ~buybook |
| 20:16.21 | Qwell | Znuff: There are plenty of good docs... |
| 20:16.23 | afo0l | if i get this s**t to work, i'll create a howto |
| 20:17.27 | afo0l | zaphfc and ubuntu kernel 2.6.31, using module-assistant to install results in kernel panic, as soon as i plug the cable into the card |
| 20:17.29 | drmessano | ~boschlawnmower |
| 20:17.30 | Jhirley | [TK]D-Fender: http://www.asterisk.org/docs |
| 20:17.32 | drmessano | ~boschlawnmowers |
| 20:17.33 | infobot | http://www.voip-info.org/wiki/view/Bosch+lawnmowers .. "Same to you, buddy!" |
| 20:17.38 | drmessano | There you go |
| 20:18.19 | Jhirley | rofl |
| 20:18.34 | p3nguin | katty: hmm? |
| 20:18.45 | drmessano | Look, im not saying thats more proof the wiki is mostly ignored and useless |
| 20:18.51 | drmessano | Wait, yes I am |
| 20:19.23 | afo0l | well, i'm looking for exotic stuff, kind of, so i can't really complain |
| 20:20.13 | *** join/#asterisk moy (~chatzilla@74.12.129.100) |
| 20:23.22 | Katty | p3nguin: baroo? |
| 20:23.32 | Katty | p3nguin: idk. that was like forever ago |
| 20:23.42 | Geminizer | ~dropinfobot |
| 20:24.02 | *** join/#asterisk Skeeter- (skeeter@c216.218.2-65.clta.globetrotter.net) |
| 20:25.22 | p3nguin | katty: forgot? |
| 20:26.04 | *** join/#asterisk chazzm (~chazz@173-24-238-25.client.mchsi.com) |
| 20:27.52 | *** join/#asterisk lupino3 (~lupino3@static-217-133-45-108.clienti.tiscali.it) |
| 20:28.41 | lupino3 | Hello everybody. I am having some DTMF talk-off issues in conversations between SIP (local) users. I'm using * 1.4.21.2 and can't upgrade... any suggestions? TIA |
| 20:31.29 | Qwell | upgrade |
| 20:32.27 | leifmadsen | Qwell: oh snap |
| 20:32.55 | [TK]D-Fender | First I was like ... :| |
| 20:32.57 | [TK]D-Fender | Then I was like ... :D |
| 20:33.15 | lupino3 | Qwell, I can't, due to some CDR changes in 1.4.22 that break my billing.. |
| 20:33.26 | *** join/#asterisk atis_work (~atis_work@193.238.212.171) |
| 20:33.28 | Qwell | You're 9 months behind. |
| 20:33.40 | lupino3 | Qwell, you're right :'( |
| 20:33.44 | Qwell | Install 1.4.30 on your test system, and...test it. |
| 20:34.12 | lupino3 | actually I'm testing 1.4.25, because it's the last 1.4.* that works with junghann's bristuff |
| 20:34.24 | lupino3 | the problem is that I can't reproduce their issue on my test system |
| 20:34.30 | Qwell | then don't use bristuff |
| 20:35.00 | lupino3 | I need it for astribank :( |
| 20:35.30 | bmoraca_work | see, this is what you get for basing your phone system off of other peoples' work, :P |
| 20:36.00 | [TK]D-Fender | Qwell: ...1.4.29.1, no? |
| 20:36.11 | Qwell | looks at leifmadsen |
| 20:36.15 | Qwell | cracks the whip |
| 20:36.21 | [TK]D-Fender | Qwell: thats the ver on tyour site |
| 20:36.35 | leifmadsen | Qwell: I'm just writing the release announcement(s) now :) |
| 20:36.41 | tzafrir_laptop | lupino3, bristuff can be ported to later versions |
| 20:36.49 | leifmadsen | 1.4.30 will be released today or tomorrow |
| 20:36.51 | tzafrir_laptop | of 1.4, though |
| 20:37.03 | lupino3 | right now it's OK for me 1.4. |
| 20:37.15 | lupino3 | is there any doc on how to port it? |
| 20:38.00 | tzafrir_laptop | well, I just try to apply the patch, and fix where it breaks |
| 20:38.23 | lupino3 | I'm not so comfortable with Asterisk and Zaptel internals |
| 20:38.51 | lupino3 | and even if it compiles, I think that it doesn't mean that it will work in production environments.. |
| 20:38.54 | tzafrir_laptop | As I already reduced the patch to only chan_dahdi.c bits, and chan_dahdi.c doesn't change that much, it normally works |
| 20:39.28 | lupino3 | if it's so easy, why aren't there more recent bristuff version? |
| 20:39.44 | lupino3 | I mean, bristuff versions with more recent * versions |
| 20:39.48 | lupino3 | > 1.4.25 |
| 20:40.21 | bmoraca_work | cause there's very little reason to use a bri anymore? hell, you can't even get them in the US |
| 20:40.33 | lupino3 | in Italy it's full of BRI |
| 20:40.43 | lupino3 | (and I'm from Italy :) ) |
| 20:41.00 | Naikrovek | guy named lupino from italy? no WAY! |
| 20:41.01 | Naikrovek | :P |
| 20:41.10 | lupino3 | :) |
| 20:41.21 | lupino3 | actually, from Sicily |
| 20:41.34 | Naikrovek | neat |
| 20:41.49 | Naikrovek | sometimes the internet amazes me for more or less mundane reasons |
| 20:41.51 | Naikrovek | like just now |
| 20:43.06 | tzafrir_laptop | bmoraca_work, silly US folks use analog |
| 20:43.22 | bmoraca_work | or SIP or PRI |
| 20:44.14 | lupino3 | however, no hints for me? :( I tried relaxdtmf=no, but still I get some pretty long spurious dtmf.. |
| 20:45.00 | bmoraca_work | sure it's not someone being a dick and holding a button down? |
| 20:45.16 | Znuff | So, ok. |
| 20:45.24 | bmoraca_work | you did say it was SIP to SIP calls...DTMF should be out of band...hard to imagine that would happen on accident |
| 20:45.31 | Znuff | I've added an extension, I added a sip trunk, I can call PSTN numbers. |
| 20:45.34 | Znuff | Now, how do I hide my clid? |
| 20:45.51 | lupino3 | bmoraca_work, dtmf was configured as rfc2833 |
| 20:45.53 | Znuff | (in the case my provider supports this) |
| 20:46.06 | bmoraca_work | Znuff: Set(CALLERPRES()=prohib) |
| 20:46.27 | bmoraca_work | lupino3: which is out-of-band...which is why i say it's pretty hard to imagine it happening by accident |
| 20:46.53 | lupino3 | bmoraca_work, I know it's oob, but still * detects dtmf!!! |
| 20:46.53 | Znuff | bmoraca_work, mkay, where do I add that? :) |
| 20:47.08 | *** join/#asterisk anthm (~anthm@freeswitch/developer/anthm) |
| 20:47.10 | bmoraca_work | Znuff: in your dialplan...where else? |
| 20:47.18 | lupino3 | bmoraca_work, I mean, it seems like it doesn't rely only on rfc2833 |
| 20:47.28 | Znuff | Where would that be in a default asterisk install? |
| 20:47.43 | bmoraca_work | Znuff: erm. extensions.conf... |
| 20:47.50 | bmoraca_work | Znuff: .... |
| 20:47.51 | bmoraca_work | ~book |
| 20:47.52 | infobot | [~book] Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://astbook.asteriskdocs.org or see ~buybook |
| 20:48.02 | *** join/#asterisk jmls (~jmls@host217-36-208-155.in-addr.btopenworld.com) |
| 20:48.09 | jmls | evening all |
| 20:48.44 | lupino3 | hi |
| 20:49.11 | jmls | got an issue with inbound SIP calls from a voip provider |
| 20:49.18 | jmls | I suspect it's me |
| 20:49.20 | jmls | :) |
| 20:49.44 | lupino3 | gotta go, thanks for the pointers bmoraca_work, Qwell and tzafrir_laptop |
| 20:49.48 | lupino3 | bye :) |
| 20:49.51 | jmls | does register => 2345:password@mysipprovider.com/1234 |
| 20:49.58 | jmls | (: P) |
| 20:50.15 | *** join/#asterisk garymc (~chatzilla@host86-135-19-162.range86-135.btcentralplus.com) |
| 20:50.18 | Znuff | bmoraca_work, can you spoon feed me? |
| 20:50.26 | jmls | go to the 1234 extension in the context defined under [general] in sip.conf ? |
| 20:50.36 | bmoraca_work | Znuff: probably, but that's not how i roll |
| 20:50.47 | Znuff | ^^ |
| 20:50.58 | *** join/#asterisk Talirk81 (~tt@rrcs-67-78-39-22.sw.biz.rr.com) |
| 20:51.21 | bmoraca_work | jmls: no, it'll attempt to match a peer based on source IP address/auth name/peer name depending on how your sip.conf is set up and jump to 1234 in whatever context is defined under that peer |
| 20:51.24 | Znuff | Not really sure where I should add that :-/ |
| 20:51.27 | Talirk81 | What is the best IAX softphone that supports SendURL, zoiper free doesnt seem to be opening urls for me |
| 20:51.41 | bmoraca_work | jmls: if it doesn't match a peer, it'll go to 1234 in whatever context is defined in [general] |
| 20:52.26 | bmoraca_work | Znuff: "Set()" is a dialplan application just like Dial(). you add it the same way you would add "Dial()" and it modifies variables for the current channel. |
| 20:53.07 | bmoraca_work | Znuff: CALLERPRES() is a special function that controlls how ANI information is displayed/allowed over digital channels (T1/E1/SIP) |
| 20:53.44 | Znuff | bmoraca_work, how would I know in what [dialplan] I should add that? Considering I have't changed absolutely anything in a default install, except adding a sip peer and a sip friend |
| 20:54.01 | spenguin[work] | hello |
| 20:54.17 | bmoraca_work | Znuff: bad idea. but, it goes in whatever context dials out that sip trunk. |
| 20:54.28 | bmoraca_work | Znuff: or whatever context your sip phone is dialing in on. |
| 20:55.06 | bmoraca_work | Znuff: remember that there are two legs to every SIP transaction (in and out)...this is where the book comes in handy. |
| 20:55.30 | bmoraca_work | Znuff: i can't really dumb it down any more than that. if you want to know more, you're going to have to work for it by reading the book and doing some experimenting on your own |
| 20:56.28 | Micc | Is there any simple command line app that can be used for testing an sip/rtp from a customers location? |
| 20:56.44 | bmoraca_work | sipp? |
| 20:56.45 | manxpower | Znuff: there is no such thing as a "default" set of working config files. |
| 20:56.47 | Micc | To determine if their network is sufficient for unjittered communications. |
| 20:57.01 | Micc | hmm.. let me look at sipp |
| 20:57.31 | manxpower | Micc: um, it would measure the jitter and latency at that moment in time. It means nothing if someone starts uploading a big file. |
| 20:57.33 | bmoraca_work | Micc: also, there are some other applications (ookla has one) that will test jitter and latency |
| 20:57.34 | *** join/#asterisk norrec (~norrec@76-201-85-140.lightspeed.frokca.sbcglobal.net) |
| 20:57.37 | manxpower | Welcome to voice over INTERNET |
| 20:58.17 | Micc | manxpower, I would want to leave it running for a week or so to watch all that. |
| 20:58.31 | norrec | how do i add the cdr_pgsql module into asterisk after i've installed already? |
| 20:58.36 | Micc | manxpower, most customers need some type of bandwidth management for that scenario. |
| 20:58.55 | bmoraca_work | norrec: add it to your make menuselect and compile it. |
| 20:58.57 | [TK]D-Fender | Znuff: Trash your sip.conf, extensions.conf, and give this a read for some inspiration i starting from scratch <- |
| 20:59.00 | [TK]D-Fender | ~jerjerguide |
| 20:59.01 | infobot | [~jerjerguide ] Jeremy McNamara's quick sample guide to setting up * : http://www.jeremy-mcnamara.com/2007/02/26/how-to-configure-asterisk-your-first-installation/ |
| 20:59.05 | [TK]D-Fender | ^^^^ |
| 20:59.47 | tzafrir_laptop | hmm... asterisk patch almost applied. I just needed to remove a hunk from it that became pointless |
| 20:59.57 | Micc | manxpower, I want to find those rare cases where their internet provider sucks ass and they need a private dsl. |
| 21:00.22 | *** join/#asterisk Slugs_ (Slugs_@c-76-97-205-31.hsd1.ga.comcast.net) |
| 21:00.24 | bmoraca_work | Micc: you mean "all of them"? :P |
| 21:01.02 | Micc | manxpower, we spent months with one customer trying bandwidth management devices and qos and replacing routers and finally we found out even if nothing else was on their dsl it was still delaying audio for a few seconds. |
| 21:01.20 | norrec | bmoraca_work: so just make menuselect then make install from the src dir? |
| 21:01.26 | Micc | bmoraca_work, actually so far its only been a small percentage of customers that really need a private dsl. |
| 21:01.29 | Corydon76-dig | Comcast was sucking wind this morning in Nashville |
| 21:02.12 | Corydon76-dig | It all depends on whether "best effort" internet service is good enough or not |
| 21:02.18 | Micc | Most customers are ok with the minor jitter now and then as long as they can still hear and understand the person. |
| 21:02.22 | bmoraca_work | norrec: yep. if the compiled modules still exist, it won't recompile them, so you'll only be compiling new modules |
| 21:02.33 | Micc | But when its happening every other day, thats not going to work. |
| 21:02.41 | [TK]D-Fender | Corydon76-dig: http://despair.com/fail24x30pri.html |
| 21:02.42 | bmoraca_work | Micc: jitter isn't my biggest problem. my biggest problems are lost packets due to congestion. |
| 21:02.44 | norrec | bmoraca_work: so i dont have to worry about overwriting confgs? |
| 21:02.57 | *** join/#asterisk KavanS (~KavanS@static-173-50-141-22.ptldor.fios.verizon.net) |
| 21:02.59 | bmoraca_work | norrec: not unless you do a "make samples" |
| 21:03.05 | Corydon76-dig | [TK]D-Fender: quite |
| 21:03.14 | bmoraca_work | or is it "make configs"...i don't remember |
| 21:03.36 | *** join/#asterisk sdake (~sdake@cm-24-121-126-215.flagstaff.az.npgco.com) |
| 21:04.04 | norrec | bmoraca_work: oh right, its make samples i believe |
| 21:04.26 | DrGeek | is this the right channel to ask about digium T1 bonded (ifenslave) hdlc configuration? |
| 21:04.27 | Micc | bmoraca, we rarely have lost packets except on the customers that need a private dsl. We have a bunch of customers on comcast and qwest dsl that have no problems. |
| 21:04.55 | Micc | But we have a couple that have the worst time on comcast, and some that can't get a good route with qwest dsl. |
| 21:05.26 | *** join/#asterisk tkrn (~tkrn@WS1-DSL-208-102-253-13.fuse.net) |
| 21:06.24 | Micc | We've had a 24 hour call center for about a year that has been on comcast with only a problem when comcast did maintenance a few months ago. |
| 21:07.16 | *** join/#asterisk luckyaba (~lucky@ip72-194-218-169.sb.sd.cox.net) |
| 21:07.24 | *** join/#asterisk murdock_ut (~chatzilla@mail.kimballequipment.com) |
| 21:07.35 | Micc | bmoraca_work, are you still running virtualized asterisk? |
| 21:08.26 | Znuff | [TK]D-Fender, that's helpful, thanks |
| 21:08.33 | Znuff | at least I'm getting somewhere |
| 21:08.44 | Micc | I don't know if it was the virtualization itself or the network the virtual machine was on, but we had a horrible time with asterisk on ubiquity. worked fine most of the time, but every few days it would get funky. |
| 21:08.46 | *** join/#asterisk uqlev (~yuriy@91.184.221.31) |
| 21:12.19 | Slugs_ | after installing asterisk how can i tell it's working |
| 21:12.20 | stix | Hi guys. What is the official website for the asterisk .NET framework project? |
| 21:15.00 | [TK]D-Fender | Slugs_: Use it |
| 21:15.33 | [TK]D-Fender | stix: http://sourceforge.net/projects/asterisk-dotnet/ |
| 21:16.19 | [TK]D-Fender | Slugs_: Also take not that you never provided the SIP DEBUG I asked for hours ago, and we cannot help with your FreePBX config parts. |
| 21:16.21 | stix | [TK]D-Fender: thanks - it looks a bit old and I cannot find any documentation on it |
| 21:16.24 | Slugs_ | [TK]D-Fender, after i insall asterisk and freepbx i should be all set? |
| 21:16.32 | [TK]D-Fender | stix: Because virtually noone uses it |
| 21:16.46 | stix | oh okay, too bad |
| 21:16.47 | [TK]D-Fender | Slugs_: Set for what? You have to CONFIGURE things for it to do anything |
| 21:17.18 | Slugs_ | [TK]D-Fender, can you point me to the docs |
| 21:17.31 | [TK]D-Fender | Slugs_: freepbx.org |
| 21:17.42 | tkrn | I'm having some trouble with my iptables and getting asterix to work out. Eth0 = To my external network/router (192.168.100.x), Eth1 = VoIP Network (10.10.20.x), http://pastebin.ca/1834510 any recommendations? |
| 21:17.57 | [TK]D-Fender | ~sipnat |
| 21:17.58 | infobot | [~sipnat] Quick guide on configuring * + SIP behind NAT : http://www.aocomputing.net/?p=3 , otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
| 21:18.02 | [TK]D-Fender | tkrn: ^^^^^^^^^ |
| 21:18.09 | [TK]D-Fender | tkrn: go follow the guide |
| 21:18.12 | tkrn | nice ty |
| 21:20.13 | tkrn | My setup is slightly different, I made the voice network seperate with trixbox having two network cards |
| 21:21.56 | [TK]D-Fender | tkrn: Nothing special. I run 2 LANs here and NAT'd out one of them to the internet |
| 21:22.10 | *** join/#asterisk ebroad (~EB@72.11.213.195) |
| 21:22.22 | [TK]D-Fender | tkrn: Follow these more specifically |
| 21:22.29 | [TK]D-Fender | [16:21]<coopbot>nat: If behind a NAT, create an /etc/asterisk/sip_nat.conf file with AT LEAST these three lines: 1) nat=yes 2) externip=your.external.IPaddess (or externhost=your.external.hostname) 3) localnet=192.168.0.0/24 (assuming your network uses 192.168.0.x addresses). Then "asterisk -rx sip reload" at the CLI. See ?? ports and ?? rtp for port forwarding information see also ??... |
| 21:22.31 | [TK]D-Fender | ...natexplain Freepbx2.6.x has a tool for sip and iax2 nat |
| 21:22.31 | tkrn | k ty |
| 21:22.32 | [TK]D-Fender | [16:21]<coopbot>ports: If you are behind a NAT firewall, you may need to open the following ports to your FreePBX box: For IAX protocol, UDP (NOT TCP) port 4569 (IAX). For SIP protocol, UDP (NOT TCP) port 5060 (SIP) AND ports 10000-20000 (RTP, must also be defined in /etc/asterisk/rtp.conf, see ?? rtp). All these ports are *UDP*, opening the *TCP* ports will NOT help anything and may... |
| 21:22.34 | [TK]D-Fender | ...expose your system needlessly. Also see ?? rtp and ?? nat |
| 21:23.35 | Znuff | [TK]D-Fender, so, I setup fresh sip.conf, extensions.conf |
| 21:23.42 | Znuff | I can register, my sip trunk registers |
| 21:23.45 | ebroad | mnicholson, ping |
| 21:23.45 | p3nguin | The bot has one error. |
| 21:23.59 | Znuff | I created a simple dial-plan |
| 21:24.12 | Znuff | exten => _.,1,Dial(SIP/VoIPProvider/1${EXTEN}) |
| 21:24.20 | tkrn | with localnet= should this be the 192.168.100.x network which is directly behind the router, or the 10.10.20.x of the phones which is through a 2nd trixbox nic?? |
| 21:24.32 | Znuff | but when I dial-out, I get line busy? :-/ |
| 21:24.36 | *** join/#asterisk ehsjoar (~ehsjoar@c-98-245-155-132.hsd1.co.comcast.net) |
| 21:24.43 | p3nguin | IAX2 port is 4569, IAX port is 5036. |
| 21:25.18 | [TK]D-Fender | tkrn: BOTH |
| 21:25.23 | [TK]D-Fender | ok, checkout time, BBIAB |
| 21:26.18 | tkrn | D-Fender: so two entries such as localnet=192.168.0.0/24 |
| 21:26.18 | tkrn | localnet=10.10.20.0/24 |
| 21:26.25 | *** join/#asterisk Cain (~Geek@unaffiliated/cain) |
| 21:27.20 | Znuff | /cry |
| 21:28.12 | Slugs_ | ive installed asterisk/freepbx and looking at the docs it dosent tell you what to do from a 'blank slate' |
| 21:28.28 | Slugs_ | ive got nothing configured |
| 21:28.47 | Znuff | Is this thing corectly done? http://pastebin.com/0zvigKKf |
| 21:28.47 | Slugs_ | freepbx is missing docs |
| 21:29.18 | Slugs_ | can anybody point me in the right direction? |
| 21:30.03 | p3nguin | exten => _.,1,Dial(SIP/VoIPProvider/1${EXTEN}) <-- Very Bad (tm) |
| 21:30.23 | Znuff | p3nguin, I don't need inbound calls, I just need outbound calls |
| 21:30.35 | p3nguin | Never pattern match EVERYTHING. _. means EVERYTHING. |
| 21:30.45 | Znuff | that's what I want, everything |
| 21:30.49 | manxpower | and everything includes extensions "s", "h", "t" |
| 21:30.51 | p3nguin | No it isn't. |
| 21:30.55 | manxpower | Znuff: no you don't |
| 21:31.01 | Znuff | yes I do |
| 21:31.03 | p3nguin | You may think you want that, but you really do not. |
| 21:31.16 | manxpower | otherwise your asterisk will trying dialing the same number again when you hangup since _. will match the hangup extension |
| 21:31.22 | p3nguin | You probably want _X. |
| 21:31.26 | Znuff | _X didn't work |
| 21:31.37 | manxpower | Znuff: it wouldn't, you fogot the . |
| 21:31.41 | p3nguin | <PROTECTED> |
| 21:31.56 | manxpower | Znuff: maybe if you spend less time arguing and more time listening you might learn something. |
| 21:32.03 | p3nguin | _X matches exactly one number. |
| 21:32.08 | Znuff | ah, ok, I see |
| 21:32.13 | Znuff | works now, with _X. |
| 21:32.21 | Znuff | indeed, I didn't add the . after X when I tried it |
| 21:32.22 | manxpower | Znuff: you would KNOW this if you had read the Asterisk book |
| 21:32.33 | Znuff | I'm here: http://astbook.asteriskdocs.org/en/2nd_Edition/asterisk-book-html-chunk/asterisk-CHP-5-SECT-3.html |
| 21:32.41 | manxpower | Slugs_: FreePBx is not supported here. |
| 21:32.44 | Znuff | didn't really see the dot :-/ |
| 21:33.09 | p3nguin | Other than that one little problem, the rest of that paste is not unreasonable. |
| 21:33.37 | *** join/#asterisk defsdoor (~andy@defsdoor.gotadsl.co.uk) |
| 21:33.46 | manxpower | people forget that there can be non-numeric extensions and the special extensions are all non-numeric |
| 21:33.48 | p3nguin | I'm not a huge fan of the macros, but it's not done incorrectly. |
| 21:34.39 | Znuff | This ok? |
| 21:34.40 | Znuff | exten => _X.,1,Set(CALLERPRES()=prohib) |
| 21:34.40 | Znuff | exten => _X.,n,Dial(SIP/VoIPProvider/${EXTEN}) |
| 21:35.08 | p3nguin | Isn't it SetCallerPres()? |
| 21:35.28 | p3nguin | an application rather than a function |
| 21:35.29 | Znuff | No idea |
| 21:35.59 | p3nguin | "core show application SetCallerPres" |
| 21:36.05 | p3nguin | This is on 1.4 |
| 21:36.39 | Znuff | so, exten => _X.,1,SetCallerPres(prohib) |
| 21:36.41 | Znuff | right? |
| 21:37.10 | Znuff | bah, |
| 21:37.11 | Znuff | [Mar 11 23:36:50] WARNING[30865]: app_setcallerid.c:94 setcallerid_pres_exec: SetCallerPres is deprecated. Please use Set(CALLERPRES()=prohib) instead. |
| 21:37.46 | p3nguin | lol |
| 21:37.50 | p3nguin | You have 1.6? |
| 21:37.52 | Znuff | Yes |
| 21:37.56 | Znuff | apparently |
| 21:38.01 | p3nguin | I guess they changed it, then. |
| 21:38.20 | p3nguin | I wouldn't know, but I did let you know I was basing my statement on 1.4. |
| 21:39.48 | manxpower | Znuff: you need to read UPGRADE*.txt files so you know how to change the old commands you are reading about into the new commands of the version you are using |
| 21:40.00 | Znuff | meh, and apparently my provider doesn't support that, still can see the CLID |
| 21:40.20 | manxpower | you want callING pres, I think |
| 21:40.35 | manxpower | Otherwise you just hid the number of the person you are calling from yourself |
| 21:41.03 | Znuff | oh |
| 21:41.15 | *** join/#asterisk Cain (~Geek@unaffiliated/cain) |
| 21:41.24 | manxpower | Znuff: one of these days your lack of attention to detail is going to open up a security hole on your Asterisk. |
| 21:41.52 | *** join/#asterisk fifer (~fifer@67.208.108.228) |
| 21:41.55 | manxpower | Hope you have allowguest=no already, that's a good start |
| 21:42.50 | Znuff | I do |
| 21:43.04 | p3nguin | I use SetCallerPres(prohib) to block my number on outgoing calls. There is no CallingPres in 1.4. |
| 21:43.07 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.218.194) |
| 21:43.25 | *** join/#asterisk mspiceland (~mike@nat/digium/x-lxwmbdolxszkorpy) |
| 21:44.25 | manxpower | p3nguin: I don't need to hide my identity so I only know about what I see in "core show applications" and UPGRADE*.txt |
| 21:44.41 | p3nguin | I wish there were an intuitive document that descibed each of the Presentiation types. doc/callingpres.txt is totally useless. |
| 21:45.23 | p3nguin | I typically do not hide my number, but sometimes it is nice to have the option. |
| 21:46.00 | *** join/#asterisk Cosmin (~Ionut@208.93.208.120) |
| 21:46.16 | Cosmin | how can i block an ip from registering extensions? |
| 21:46.26 | p3nguin | extensions do not register. |
| 21:46.34 | p3nguin | So it's a non-issue. |
| 21:46.39 | manxpower | Cosmin: permit/deny |
| 21:46.46 | [TK]D-Fender | Cosmin: "deny=" <- |
| 21:46.55 | p3nguin | Oh, maybe you meant devices rather than extensions. |
| 21:47.17 | *** join/#asterisk mweichert_ (~mweichert@216.13.154.21) |
| 21:47.21 | manxpower | p3nguin: if he didn't, then he's far beyond of any of our help. |
| 21:48.19 | Cosmin | im getting these |
| 21:48.21 | Cosmin | Registration from 'sip:ext@0.0.0.0' failed for 'x.x.x.x' - Username/auth name mismatch |
| 21:48.24 | Cosmin | trying to avoid them |
| 21:48.38 | Cosmin | they are all comming from same ip |
| 21:49.00 | manxpower | Cosmin: that would be tough to block |
| 21:49.19 | Cosmin | there is no iptables :( |
| 21:49.21 | manxpower | maybe deny=0.0.0.0/32 permit=0.0.0.0/0 |
| 21:49.31 | manxpower | Cosmin: why are you talking about uptables? |
| 21:49.35 | Cosmin | thats just an example of ip ... :) |
| 21:49.37 | manxpower | sip.conf, dude, sip.conf |
| 21:49.42 | Cosmin | its not really 0.0.0.0 |
| 21:49.51 | Znuff | deny=0.0.0.0 |
| 21:49.59 | manxpower | Cosmin: when you are prepared to provide us with ACCURATE information then we can help you. |
| 21:50.10 | manxpower | or I can help you for another 4 mins |
| 21:50.20 | Znuff | deny=192.168.1.1/32 or whatever :P |
| 21:50.38 | Znuff | even I know that |
| 21:50.42 | p3nguin | I would rather block it before it ever hits asterisk. |
| 21:51.03 | Znuff | I'm guessing that's what he means by "no iptables" |
| 21:51.25 | p3nguin | I can't even imagine why there isn't some type of filtering mechanism for the network. |
| 21:51.34 | Cosmin | i have an asa |
| 21:51.44 | p3nguin | It is capable of blocking IP addresses. |
| 21:51.53 | Cosmin | but for some reason the ip in the asterisk shows like its comming from the router ip |
| 21:51.55 | p3nguin | Someone that owns an ASA would typically know that. |
| 21:52.06 | Cosmin | and its trying to register some bogus extensions |
| 21:52.39 | p3nguin | realizes this is a lost cause after the second mention of registering extensions |
| 21:52.44 | manxpower | why don't you block everything and only allow the ip block of your device? |
| 21:52.54 | manxpower | p3nguin: I think you are right. |
| 21:53.13 | manxpower | Cosmin: you CANNOT REGISTER EXTENSIONS. |
| 21:53.17 | manxpower | You register DEVICES |
| 21:53.28 | manxpower | a device is not an extension no matter how much you want it to be. |
| 21:53.57 | Cosmin | right i got that |
| 21:54.11 | p3nguin | He said 4 minutes, and he meant it. |
| 21:54.15 | Cosmin | im trying to prevent an ip from registering |
| 21:54.29 | Cosmin | with the system |
| 21:54.33 | Cosmin | or accessing the system |
| 21:54.53 | doneir | manxpower: you responded to me yesterday regarding a problem I had. The issue was moving an asterisk setup to another server, now there is no way to call out without congestion errors, and dialing in gets line busy without output from asterisk. You asked if it was a PRI setup, it is E1 ISDN pri_cpe euroisdn (australia). Still haven't worked out what the issue is this side, running out of ideas. Asterisk 1.6.1.17 libpri 1.4.10.2 dahdi 2.2.1 |
| 21:54.53 | Corydon76-dig | Cosmin: what version? |
| 21:55.03 | p3nguin | Now on to something more important: what happened to iptables? |
| 21:56.42 | doneir | pri debug span shows "Sending Set Asynchronous Balanced Mode Extended", dahdi restart gives "Failed to read gains: Invalid argument" on channels, though the config is fine |
| 21:56.44 | Cosmin | asterisk -V |
| 21:56.44 | Cosmin | Asterisk |
| 21:58.10 | wpbrown | Quick Question: I am running some tests to look at dtmf issues with cell phones. Specifically dialing in with a iPhone. I have the asterisk -r console up with "core set verbose 10" running. Will this fill up a log file someplace that I don't know about lol? |
| 21:58.11 | p3nguin | http://www.bored.com/photos/chickenhead.html |
| 21:58.48 | *** part/#asterisk rttrey (~trey@209.208.18.121) |
| 21:59.29 | Qwell | How can a mcnugget have a chicken head in it, when it isn't even chicken? |
| 21:59.29 | doneir | http://pastebin.ca/1834569 <--- config data |
| 21:59.38 | mweichert_ | Hi! I'm preparing to build a system to support 200 extensions and 20 simultaneous callsm, with 1 T1 and two FXS ports... I was originally intending on using a Pentium dual-core 1.6GHZ processor and 2Gb of RAM. However, I started to get concerned as I read some issues with Asterisk and dual-core processors |
| 22:00.03 | mweichert_ | does Asterisk work well on dual-core CPUs or would I best to use a single-core but higher clock CPU? |
| 22:00.04 | Corydon76-dig | Qwell: sure it is chicken... just "miscellaneous parts" |
| 22:00.15 | Corydon76-dig | which includes the head |
| 22:00.21 | Qwell | through a grinder |
| 22:00.28 | wpbrown | I am using Xeon on mine and it works nicely. dual core |
| 22:00.51 | Corydon76-dig | True enough. |
| 22:01.19 | Corydon76-dig | Nothing says the batter can't deform to look like a chicken head, though |
| 22:01.44 | p3nguin | http://www.bored.com/photos/stupidmouse_5.html |
| 22:01.56 | Corydon76-dig | We find the virgin Mary on pastries... why can't a mcnugget look like a chicken head? |
| 22:02.40 | *** part/#asterisk ebroad (~EB@72.11.213.195) |
| 22:02.41 | p3nguin | That's not even a McNuggets box -- it appears to be Chicken Selects. |
| 22:03.17 | Corydon76-dig | p3nguin: I wouldn't know... Not part of my regular diet |
| 22:04.40 | wpbrown | McShizzle is what I call it. |
| 22:06.33 | [TK]D-Fender | Corydon76-dig: http://www.whatwouldthembido.com/2009/04/what-the-eff-wednesday-down-with-fast-food.html |
| 22:08.29 | mweichert_ | wpbrown, how many extensions or simultaneous calls can you support? |
| 22:12.42 | LemensTS | . |
| 22:13.47 | *** join/#asterisk werdan7 (~w7@freenode/staff/wikimedia.werdan7) |
| 22:16.47 | wpbrown | mwceichert currently I have 25 extentions with a pri. |
| 22:17.20 | wpbrown | I probably handle between 700-1000 incoming calls and faxes a day |
| 22:18.57 | p3nguin | 25 extensions... crappy. I have 214 extensions (783 priorities) in 55 contexts. |
| 22:19.08 | *** join/#asterisk bn-7bc (bjarne@mac.lan.noare-1.holmedal.net) |
| 22:19.22 | wpbrown | p3nguin, I don't have that many people. LOL |
| 22:19.33 | p3nguin | I have less than a dozen phones. |
| 22:19.39 | wpbrown | lmao |
| 22:19.52 | *** join/#asterisk bn-7bc (bjarne@mac.lan.noare-1.holmedal.net) |
| 22:20.11 | wpbrown | Has anyone every used digium support to look at a Asterisk box to see if it is running at peak efficiency? |
| 22:20.19 | leifmadsen | -= 471 extensions (1821 priorities) in 56 contexts. =- |
| 22:20.30 | wpbrown | I would be willing to pay someone to look at mine to help explain a few things.. |
| 22:21.00 | leifmadsen | p3nguin: I have ONE MORE context than you :) |
| 22:21.10 | leifmadsen | dinner! |
| 22:21.11 | p3nguin | and lots more extens. |
| 22:21.20 | leifmadsen | p3nguin: that's because I use a lot of named extensions |
| 22:22.25 | wpbrown | A programmer friend of mine set my box up but some of the configs are in a SQL data base. Has asterisk moved to that or is this something he did on his own. I used to run asterisk years ago and lived in /etc/asterisk. |
| 22:22.46 | p3nguin | He did it on his own accord. |
| 22:23.00 | wpbrown | Is it good/bad? |
| 22:23.08 | wpbrown | I don't like the SQL set up. |
| 22:23.08 | p3nguin | It's subjective. |
| 22:23.14 | [TK]D-Fender | [17:20]<wpbrown>Has anyone every used digium support to look at a Asterisk box to see if it is running at peak efficiency? <- this implies nothing of value |
| 22:23.26 | [TK]D-Fender | wpbrown: No basis of comparison. |
| 22:23.36 | p3nguin | Some people prefer to use DBs for things, where others don't. |
| 22:24.15 | Talirk81 | it also depends on your change frequency |
| 22:24.34 | Talirk81 | if you have a very dynamic system reloading isnt very efficent when realtime queries dont need it. |
| 22:24.42 | Talirk81 | but there is negatives to that also |
| 22:24.44 | wpbrown | I noticed its easier to make a change on the fly using SQL. |
| 22:25.05 | p3nguin | That seems to be the biggest reason people use realtime. |
| 22:25.34 | wpbrown | however, I had issues and looked in queues in /etc/asterisk nada |
| 22:25.54 | wpbrown | so I dug around on the ubuntu box and found phpmyadmin |
| 22:26.07 | wpbrown | and saw he had the files in sql |
| 22:26.17 | *** join/#asterisk rare1980_ (~rare1980@115.186.2.19) |
| 22:26.43 | rare1980_ | in rtp.conf........ port 10000 to 20000 is tcp port or udp |
| 22:26.44 | rare1980_ | ? |
| 22:26.53 | p3nguin | udp, of course. |
| 22:27.23 | rare1980_ | why it it not working over nat |
| 22:27.31 | rare1980_ | i m getting Packet2Packet bridging SIP/1-04f08d80 and SIP |
| 22:27.33 | rare1980_ | msg |
| 22:27.40 | rare1980_ | i m using nat |
| 22:27.58 | jaytee | ~sipnat |
| 22:27.59 | infobot | [~sipnat] Quick guide on configuring * + SIP behind NAT : http://www.aocomputing.net/?p=3 , otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
| 22:29.10 | rare1980_ | thanks |
| 22:30.58 | *** join/#asterisk rossand (~aross@dhcp-233-179.tb-classrooms.carleton.ca) |
| 22:33.30 | wpbrown | [TK]D-Fender are you saying my statement is bad mojo for this chat room or that their support isn't very good? |
| 22:34.37 | [TK]D-Fender | wpbrown: I'm saying that you asked an open question with no target on what constitutes "efficiency". Like every dumb-ass manager getting sold a snake-oil "evaluation" to fix problems that really aren't problems. |
| 22:35.26 | [TK]D-Fender | wpbrown: How much of an * system would count as having some significant inefficiency that is blatantly obvious? |
| 22:35.43 | [TK]D-Fender | wpbrown: and of anything short of a "complex" system... whats left? |
| 22:35.46 | cusco | I remember a while ago somebody was telling me a function where I could set my number private |
| 22:35.47 | wpbrown | Ahh, I should have elaborated. |
| 22:35.58 | cusco | there was several options . besides private... |
| 22:36.01 | [TK]D-Fender | wpbrown: Yeah... Vagueland is not a nice place. |
| 22:36.57 | *** join/#asterisk ruben23 (~ITadmin@122.55.48.243) |
| 22:37.03 | wpbrown | I am decent using unix and I wasn't bad using Asterisk in 2005. However, my box has a few quirks that I would like to have a professional (IE someone with more experience than my self) look at to see if it is properly configured. |
| 22:37.05 | cusco | I am looking to hide our outbound number trough dahdi |
| 22:37.16 | cusco | can you point me in the right direction? |
| 22:37.42 | p3nguin | cusco: doc/callingpres.txt |
| 22:37.46 | cusco | thanks |
| 22:38.28 | [TK]D-Fender | wpbrown: Wow... you elaborated on "vauge" with .... "I have quirks". |
| 22:38.28 | cusco | thats it thanks p3nguin |
| 22:38.44 | [TK]D-Fender | wpbrown: What makes it "not proper"? |
| 22:38.46 | p3nguin | On 1.4, I use SetCallerPres(prohib) to make the number "private." |
| 22:39.25 | wpbrown | I have DTMF issues with cell phones, I have software that is ancient for call reporting, would like to move from round robin based set up to agents.. |
| 22:39.38 | wpbrown | Things like that. |
| 22:40.04 | wpbrown | Problem is the PBX is in use and is the back bone of our company. Can't really take it down to test on my own. |
| 22:40.21 | wpbrown | I don't have a spare PRI. |
| 22:40.24 | wpbrown | things like that |
| 22:40.36 | [TK]D-Fender | wpbrown: What of your setup requires a PRI for testing? |
| 22:41.18 | *** join/#asterisk bjhaid (~herbayjha@41.206.15.4) |
| 22:41.30 | Talirk81 | is it possible to have IAX softphone -> Asterisk -> SIP trunk to Telco -> Outside number or would i need to have IAX softphone -> Asterisk -> IAX trunk to telco -> Outside number |
| 22:41.40 | p3nguin | wpbrown: Often, increasing toneduration in dahdi will help dtmf issues. |
| 22:41.54 | p3nguin | talirk81: either way |
| 22:42.45 | p3nguin | talirk81: As long as you have the channel drivers installed/loaded, you can use any channel technology on either side of the call. |
| 22:42.52 | [TK]D-Fender | wpbrown: I'm getting the impression you aren't going to give anyone enough information to come up with anything meaningful for you at all. That in mind... if the only way you'll find yourself able to really look at what you're doing is to pay someone to look at it and then force yourself to follow through if olny to justify said exprense.... then more power to you |
| 22:43.01 | *** join/#asterisk kotp (~vgoff@96.2.187.66) |
| 22:44.18 | *** join/#asterisk Pan3D (~Pan3D@streaming.sensoryresearch.net) |
| 22:45.30 | wpbrown | im on a call one second [TK] |
| 22:48.00 | *** join/#asterisk dennisG (~dennisG@2002:541e:88d0:0:213:2ff:fe56:e380) |
| 22:48.46 | dennisG | hello all, who's using fax over voip in combo with asterisk ? |
| 22:49.55 | cusco | p3nguin: thanks. That doesn't work with me yet. Telco said they could make it so it would depend on out PBX |
| 22:49.58 | *** join/#asterisk [netman] (~netman@115.Red-79-156-153.staticIP.rima-tde.net) |
| 22:50.02 | cusco | looks like they haven't done it yet |
| 22:50.11 | Qwell | ~fovoip |
| 22:50.28 | cusco | dennisG: I am using spandsp with asterisk |
| 22:51.00 | cusco | another question.. |
| 22:51.31 | Qwell | dennisG: http://www.soft-switch.org/foip.html |
| 22:51.46 | cusco | how can I chose wich number from telco to use? we have 100 DDIs on each telco... but outbounds are always with the number ending in 00 |
| 22:51.58 | cusco | I would like to use other numbers... |
| 22:52.07 | wpbrown | [TK]D-Fender, lets start with the first issue. DTMF issues with cell phones. We established it can be caused with noise on the line, echo,. P3nguin says to increase tone duration to fix it. I have no clue how to increase tone duration. |
| 22:53.10 | p3nguin | dahdi.conf, perhaps? |
| 22:53.26 | [TK]D-Fender | wpbrown: You can't. |
| 22:53.47 | cusco | in chan_dahdi.conf I defined a group that would use a specific channel, and dialing DAHDI/g60/number would go trough that channel, therefore use a different number than the default |
| 22:53.50 | [TK]D-Fender | wpbrown: What interface are you using? What software versions? |
| 22:53.56 | [TK]D-Fender | wpbrown: you know... DETAILS. |
| 22:54.22 | wpbrown | Asterisk 1.6.0.9 |
| 22:54.53 | jaytee | http://akrobotics.com/images/comics/akrobotics/last-unicorn.jpg |
| 22:55.12 | [TK]D-Fender | wpbrown: Keep going... |
| 22:55.23 | Carlos_PHX | Hey, would anyone that has a couple minutes look at my update to this page and comment/critique the section on the 's' priority? http://www.voip-info.org/wiki/view/Asterisk+priorities |
| 22:55.37 | Carlos_PHX | It's confusing so I want to make sure it makes sense. |
| 22:55.56 | *** join/#asterisk [netman] (~netman@115.Red-79-156-153.staticIP.rima-tde.net) |
| 22:56.33 | p3nguin | carlos_phx: There is no 's' priority. |
| 22:56.43 | Carlos_PHX | Hahahaha |
| 22:56.49 | Carlos_PHX | Well, yeah, there is. |
| 22:56.54 | p3nguin | Show me. |
| 22:56.58 | Carlos_PHX | I know, nobody knows. |
| 22:57.01 | Carlos_PHX | See link above. |
| 22:57.06 | Carlos_PHX | That's WORKING code. |
| 22:57.12 | Carlos_PHX | kpfleming wrote it. |
| 22:57.14 | p3nguin | I'm on the page. Show me where the s priority is. |
| 22:57.25 | Carlos_PHX | Near the bottom, after the n section. |
| 22:57.56 | Carlos_PHX | So anyway, nobody knows, and that's why I'm documenting it. |
| 22:58.16 | Carlos_PHX | revision 3885, 2004-10-02 |
| 22:58.18 | [TK]D-Fender | p3nguin: s=same (across differnt extens. I heard about it recently. Retarded and worthless.. but... whatever |
| 22:58.25 | p3nguin | heh |
| 22:58.40 | Carlos_PHX | It may be worthless in most cases, as an ITSP it's been critical. |
| 22:58.59 | jaytee | and you all heard it here first! |
| 22:59.11 | *** join/#asterisk xpot-mobile (~xpot@66.60.101.91) |
| 22:59.26 | Carlos_PHX | So basically I was looking for people to see if my explanation makes any sense. |
| 22:59.38 | Carlos_PHX | Because I wrote it ten times and never liked it. |
| 22:59.56 | Carlos_PHX | Kevin helped me flesh it out but since he wrote the code... |
| 23:02.24 | [TK]D-Fender | Carlos_PHX: Critical? Hardly. It only parses back to a hard number. It offers nothing you couldn't do yourself alreeady |
| 23:03.44 | cusco | so ... what would be the right way to choose the telco number to outbound with? |
| 23:03.45 | Nugget | yay McNuggets |
| 23:04.05 | jaytee | yay McTelnet |
| 23:04.05 | Nugget | telnet is eeeeeeevil! |
| 23:04.15 | *** join/#asterisk Gary_B (~gary@85.211.141.97) |
| 23:04.15 | jaytee | haha |
| 23:05.38 | *** join/#asterisk riddlebox (~james@75-132-225-75.dhcp.stls.mo.charter.com) |
| 23:06.29 | Nugget | hee |
| 23:07.46 | Carlos_PHX | [TK]D-Fender: Sure, but when you have to manage thousands of lines of customer DID numbers, that becomes critical. |
| 23:10.29 | [TK]D-Fender | Carlos_PHX: Sorry... but no. Better abstraction and scripting options out there |
| 23:15.19 | rare1980_ | finally sip is working fine behind nat.. but now after 20 second voice get disconnected .. |
| 23:15.28 | rare1980_ | any help |
| 23:16.32 | Carlos_PHX | rare1980_: What router? |
| 23:16.42 | Carlos_PHX | And are both ends behind NAT? |
| 23:17.10 | rare1980_ | linux nat on virutalbox |
| 23:17.21 | rare1980_ | on virtual box i m running asterisk |
| 23:19.42 | rare1980_ | ? |
| 23:22.32 | dennisG | rare1980, can you check your firewall settings? |
| 23:22.45 | dennisG | it looks like there is a problem with the RTP ports |
| 23:22.55 | rare1980_ | humm ok |
| 23:23.02 | dennisG | asterisk push 2 RTP ports open with the signaling |
| 23:23.02 | rare1980_ | let me try that |
| 23:23.16 | dennisG | but that 20 sec is an timeout of your router/firewall |
| 23:23.35 | dennisG | so your router/firewall closes that open RTP port |
| 23:24.10 | dennisG | just kick your router/firewall and tell him/her to keep the RTP ports open |
| 23:24.11 | rare1980_ | let me check thanks |
| 23:24.26 | *** part/#asterisk jmls (~jmls@host217-36-208-155.in-addr.btopenworld.com) |
| 23:24.30 | dennisG | i had that problem too (more then one time :P) |
| 23:25.02 | dennisG | btw.. do you use a cisco router/firewall? |
| 23:25.24 | rare1980_ | no i m using linux firewall |
| 23:25.34 | dennisG | with iptables? |
| 23:25.55 | rare1980_ | i have to foward 100000:20000 port to 10000:20000 |
| 23:25.57 | rare1980_ | rite |
| 23:25.57 | rare1980_ | ? |
| 23:26.00 | dennisG | yes! |
| 23:26.03 | dennisG | very good =) |
| 23:26.12 | dennisG | and 5060 -> 5060 :) |
| 23:26.16 | dennisG | for the sip signal |
| 23:26.29 | Qwell | udp.. |
| 23:26.37 | dennisG | yes, everything is udp |
| 23:26.49 | rare1980_ | dennisG: yes thanks |
| 23:26.56 | rare1980_ | let me test |
| 23:26.59 | dennisG | rtp (for the voice) is based on udp |
| 23:27.06 | dennisG | oke give it a try |
| 23:27.08 | *** join/#asterisk Caplain (shayne@86-138.35-65.tampabay.res.rr.com) |
| 23:27.15 | dennisG | i'll wait:P |
| 23:27.20 | rare1980_ | :) |
| 23:29.09 | dennisG | and problem solved? (A) |
| 23:33.31 | dennisG | i hope so! |
| 23:34.30 | *** join/#asterisk Caplain (shayne@86-138.35-65.tampabay.res.rr.com) |
| 23:40.07 | *** join/#asterisk neurosys (~neurosys@c-71-196-20-208.hsd1.fl.comcast.net) |
| 23:44.19 | *** join/#asterisk adnc (~numer@unaffiliated/adnc) |
| 23:47.24 | drmessano | Maybe its his crappy connection |
| 23:48.06 | *** join/#asterisk [netman] (~netman@193.153.154.98) |
| 23:50.44 | *** join/#asterisk telnettech (~telnettec@cpe-24-33-229-14.insight.res.rr.com) |
| 23:51.14 | *** join/#asterisk bn-7bc (bjarne@mac.lan.noare-1.holmedal.net) |
| 23:51.33 | jaytee | telnettech, hi! |
| 23:52.20 | telnettech | hey jaytee....hows everything |
| 23:52.31 | jaytee | ok, how about you? |
| 23:52.41 | *** join/#asterisk kazoo (~WHY@pool-71-104-120-200.lsanca.dsl-w.verizon.net) |
| 23:53.12 | *** join/#asterisk ken^ (~ken@roam-71.roam.intelenet.net) |
| 23:53.15 | ken^ | yo |
| 23:53.20 | ken^ | hey, i got a quick question |
| 23:53.26 | jaytee | 42 |
| 23:53.39 | ken^ | i just read a whitepaper from intel about the performance of asterisk under intel's compiler vs gcc |
| 23:53.54 | Qwell | ken^: link? |
| 23:53.55 | ken^ | anyone out there think it's better to use their compiler ? |
| 23:54.00 | ken^ | Qwell: just a sec |
| 23:54.23 | ken^ | http://download.intel.com/design/intarch/papers/318862.pdf |
| 23:54.40 | ken^ | i believe them, because, hey, they're intel |
| 23:55.03 | ken^ | i was just trying to find out though, if basically every library that asterisk uses would need to be recompiled as well |
| 23:55.13 | ken^ | ie, if i wanted to do it, am i looking at an entire os/package rebuild ? |
| 23:55.36 | adnc | how can i get a list of missed calls with asterisk? do i need to use cdr for this? |
| 23:56.56 | p3nguin | That might be easiest. |
| 23:57.13 | telnettech | pretty good....working for jpm chase |
| 23:57.24 | telnettech | working on Cisco CM :( |
| 23:57.40 | telnettech | but still keeping up with asterisk doing things on the side and reading |
| 23:57.56 | jaytee | cool! |
| 23:59.37 | Qwell | ken^: Page 24 invalidates the entire document. |
| 23:59.53 | Qwell | ## DEBUG=-g3 (removed for Intel compiler) |
