| 00:00.20 | *** join/#asterisk moy (~moy@bas1-unionville55-1177733627.dsl.bell.ca) |
| 00:01.58 | [TK]D-Fender | TedNJ40: #trixbox <-------- |
| 00:05.24 | seanjohn | hey fender, i'm sick of having freepbx my basic dial plan |
| 00:05.28 | seanjohn | <PROTECTED> |
| 00:05.28 | seanjohn | <PROTECTED> |
| 00:06.04 | seanjohn | its on the same damn device, just another FXS that DOES work just fine so is it really trixbox or the device or asterisk? |
| 00:12.53 | seanjohn | i don't really use trixbox, i just call freepbx trixbox as its about as buggy as trixbox now |
| 00:14.54 | [TK]D-Fender | seanjohn: And what you've shown vs what you ask makes no sense |
| 00:15.37 | *** join/#asterisk Znuff (~ibm86@89.47.35.9) |
| 00:19.18 | *** join/#asterisk Carlos_Tico (~carlos@c-98-201-56-25.hsd1.tx.comcast.net) |
| 00:20.00 | thansen | [TK]D-Fender: how about my sip client question :) does that make sense? |
| 00:20.59 | Carlos_Tico | I need some help with my incoming calls anyone ? |
| 00:21.11 | wcselby | Carlos_Tico - what kind of help? |
| 00:21.26 | ChannelZ | Pick up phone, dial number |
| 00:21.56 | wcselby | thansen - you need to set the context in your peer definition that you're using |
| 00:22.04 | wcselby | to a context that exists in extensions.conf, etc.... |
| 00:22.14 | thansen | wcselby: that's the thing, I do |
| 00:22.36 | wcselby | thansen - well, I wouldn't know that, you've provided very little information. pastebin is your friend.... |
| 00:22.37 | wcselby | ~pb |
| 00:22.38 | infobot | [~pb] A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://bin.cakephp.org/ , http://asterisk.pastey.net/ , or install pastebinit with yum or aptitude. |
| 00:22.57 | *** join/#asterisk sponger (~sponger@udp261429uds.hawaiiantel.net) |
| 00:23.15 | Carlos_Tico | <wcselby> Carlos_Tico - what kind of help? ---> In my sip trunks the incoming are not working |
| 00:23.30 | Carlos_Tico | all the fxo , gsm, and one sip trunk ipkall works the rest its not working |
| 00:24.59 | ChannelZ | Can you be more vague? |
| 00:25.08 | Carlos_Tico | :) |
| 00:25.27 | thansen | wcselby: yeah, hold on and I'll clean it up |
| 00:26.02 | Carlos_Tico | http://pastebin.com/85U5DS3W |
| 00:27.02 | *** join/#asterisk [intra]lanman (~lanman@freeswitch/developer/intralanman) |
| 00:31.12 | thansen | wcselby: http://pastebin.ca/1873180 |
| 00:31.46 | *** join/#asterisk TJNII (~TJNII@207.189.199.62) |
| 00:32.31 | [TK]D-Fender | thansen: Where is the FAILED CALL with SIP DEBUG to examine? |
| 00:33.44 | kyosh | are there any windows apps that uses the manager api to provide real time montoring? |
| 00:33.48 | thansen | [TK]D-Fender: that results in [May 27 18:33:23] NOTICE[4743]: chan_sip.c:20059 handle_request_invite: Call from 'foo-105' to extension '800' rejected because extension not found. |
| 00:34.01 | [TK]D-Fender | thansen: Where is the FAILED CALL with SIP DEBUG to examine? <------- |
| 00:34.01 | seanjohn | well my question isn't my client as I just switched around fxs ports and its still the same scenario |
| 00:34.14 | seanjohn | switching back to 1.4 |
| 00:34.22 | [TK]D-Fender | seanjohn: And your description is still weak |
| 00:34.37 | [TK]D-Fender | seanjohn: I doubt * has anything to dow ith this issue |
| 00:34.39 | *** join/#asterisk romb (~romb@unaffiliated/romb-work/x-7222485) |
| 00:34.50 | seanjohn | two sip configs setup same way one works and one doesn't |
| 00:34.52 | seanjohn | circuit busy |
| 00:35.05 | [TK]D-Fender | seanI see nothing of value to debug... |
| 00:35.16 | [TK]D-Fender | "circuit-busy" = meaninglyess |
| 00:36.00 | *** join/#asterisk neurosys (~neurosys@adsl-64-99-196.mia.bellsouth.net) |
| 00:36.10 | thansen | [TK]D-Fender: well does that config look legit? |
| 00:36.31 | *** join/#asterisk ming_zym (~ming_zym@114.251.86.0) |
| 00:36.33 | seanjohn | Connecting to Asterisk manager interface..Thu, 27 May 2010 20:36:00 -0400 - Unable to connect to manager 127.0.0.1:5038 (111): Connection refused |
| 00:36.45 | seanjohn | manager.conf has the same values its had for a year |
| 00:38.07 | *** join/#asterisk g_r_eek (~g_r_eek@dslb-092-075-173-012.pools.arcor-ip.net) |
| 00:38.28 | [TK]D-Fender | thansen: It means NOTHING without the debug to match |
| 00:38.29 | wcselby | thansen - you've got 800 declared twice in your extensions.conf |
| 00:38.42 | wcselby | thansen - also, we need to see what happens when a call comes in |
| 00:38.59 | [TK]D-Fender | thansen: Its no different than showing me a picture of your car before you bought it and then asking me after 10 years of driving it (does it look ok?" |
| 00:39.04 | wcselby | thansen - that means we need to see cli output of a failed call, along with a sip debug |
| 00:39.49 | thansen | wcselby: 800 isn't defined twice |
| 00:40.01 | [TK]D-Fender | thansen: PASTEBIN <----------- |
| 00:40.07 | thansen | [TK]D-Fender: well if it's configured wrong it doesn't make any sense to debug |
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| 00:40.11 | wcselby | thansen - missed the ;. sorry |
| 00:40.18 | thansen | np |
| 00:40.26 | wcselby | thansen - by seeing the debug, we can help narrow down the config issue |
| 00:42.06 | thansen | wcselby: how can I turn on debug for a sip registered account? |
| 00:42.18 | thansen | done with register => |
| 00:42.22 | seanjohn | [root@mail asterisk-1.6.0.28]# telnet localhost 5038 |
| 00:42.22 | seanjohn | Trying 127.0.0.1... |
| 00:42.22 | seanjohn | telnet: connect to address 127.0.0.1: Connection refused |
| 00:42.23 | seanjohn | telnet: Unable to connect to remote host: Connection refused |
| 00:42.23 | seanjohn | [root@mail asterisk-1.6.0.28]# |
| 00:42.23 | Nugget | telnet is eeeeeeevil! |
| 00:42.28 | wcselby | thansen - sip set debug on; or sip set debug ip xxx.xxx.xxx.xxx on; |
| 00:42.33 | wcselby | i'm gonna go watch a movie with the wife |
| 00:42.41 | wcselby | i'm frustrated with this stupid compilation issue |
| 00:42.49 | thansen | wcselby: there is a ton of traffic on this machine right now |
| 00:42.51 | wcselby | i can compile dahdi fine on one box, but not on another... |
| 00:43.11 | [TK]D-Fender | seanjohn: pastebin "netstat -an|grep 5028" |
| 00:43.18 | [TK]D-Fender | seanjohn: pastebin "netstat -an|grep 5038" |
| 00:43.47 | wcselby | thansen - well, grab what you can, and filter it into a text file, then pastebin it. i've done it before. it's a pain, but it does the trick. |
| 00:43.56 | wcselby | you could also try "sip set debug peer foo-800" |
| 00:44.01 | wcselby | you could also try "sip set debug peer foo-800 on" |
| 00:44.08 | thansen | yeah |
| 00:44.09 | *** join/#asterisk Cain` (~Geek@unaffiliated/cain) |
| 00:44.22 | wcselby | but anyways, gonna watch a movie, f' this stupid compile stuff |
| 00:44.22 | *** join/#asterisk jnicola (~shisha@51-53-231-201.fibertel.com.ar) |
| 00:44.23 | seanjohn | its not listening fender but asterisk IS running |
| 00:44.28 | jnicola | hello all |
| 00:45.28 | [TK]D-Fender | seanjohn: then the manager isn't even LOADED |
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| 00:46.53 | seanjohn | [May 27 20:46:00] NOTICE[27383]: manager.c:3895 __init_manager: Unable to open AMI configuration manager.conf. Asterisk management interface (AMI) disabled. |
| 00:47.02 | [TK]D-Fender | WOW!!!!! |
| 00:47.02 | seanjohn | manager.conf is there and has user rights |
| 00:47.50 | seanjohn | i didn't have so many problems until march when i switched to 1.6 |
| 00:47.52 | [TK]D-Fender | seanjohn: You're making assertions and showing us JACK SHIT. Its like squeezing blood from a stone. |
| 00:48.26 | seanjohn | -rw-rw-r-- 1 asterisk asterisk 418 May 27 20:31 cdr_manager.conf |
| 00:48.26 | seanjohn | -rw-rw-r-- 1 asterisk asterisk 873 May 27 20:32 manager.conf |
| 00:48.26 | seanjohn | -rw-rw-r-- 1 asterisk asterisk 873 May 27 20:26 manager.conf.old |
| 00:48.42 | [TK]D-Fender | seanjohn: and the full CONTENTS |
| 00:49.24 | seanjohn | http://pastebin.ca/1873185 |
| 00:49.29 | *** join/#asterisk CrashSys (~kumba@azrael.crashsys.com) |
| 00:49.36 | [TK]D-Fender | seanjohn: Also no prrof as to the folders it's in, where * is really running out of, etc |
| 00:49.37 | CrashSys | Anyone ever seen this error before: sip_write: Asked to |
| 00:49.37 | CrashSys | transmit frame type 4, while native formats is 0x100 |
| 00:49.51 | CrashSys | Hmm, weirds paste... |
| 00:50.05 | seanjohn | you didn't ask for asterisk.conf |
| 00:50.21 | CrashSys | That is just a convoluted way of saying i'm out of G729 licenses right? |
| 00:50.56 | seanjohn | check your pm's crashsys |
| 00:51.03 | [TK]D-Fender | seanjohn: Like I said... squuezing blood from a stone. |
| 00:51.27 | [TK]D-Fender | CrashSys: No, 256 = G729 last I checked |
| 00:52.10 | CrashSys | well 4 = ulaw... wonder what 100 is... |
| 00:53.19 | thansen | [TK]D-Fender: can I PM you link? |
| 00:53.24 | CrashSys | 0x100 hex = 256 int |
| 00:53.36 | CrashSys | So, yeah, looks like he's got a license or bad codec issue... |
| 00:53.48 | [TK]D-Fender | thansen: Yes |
| 00:54.29 | [TK]D-Fender | thansen: Found peer 'alarmguard-outbound' for '801' from 67.223.99.12:5060 <--- matched ANOTHER DAMN PEER |
| 00:54.45 | [TK]D-Fender | thansen: Looking for 800 in alarmguard-outbound (domain 192.41.11.103) <--- looking in ANOTHER CONTEXT |
| 00:56.11 | *** join/#asterisk [intra]lanman (~lanman@freeswitch/developer/intralanman) |
| 00:56.31 | [TK]D-Fender | Ok... checkout time.... |
| 00:56.41 | thansen | [TK]D-Fender: so what does that mean for me? |
| 00:57.10 | *** join/#asterisk RobH_ (~robh@wikimedia/RobH) |
| 00:57.17 | thansen | can I solve the issue on my end? |
| 00:57.43 | [TK]D-Fender | thansen: It measn that the SIP peer you showed us is NOT the one your caller is matching, and because of that it is looking in ANOTHER context for your exten and it does NOT exist |
| 00:57.56 | [TK]D-Fender | thansen: Start by correctly matching your CALLER |
| 00:58.30 | [TK]D-Fender | 'alarmguard-outbound' <--- sure doesn't say 'foo-800' to me |
| 00:58.37 | [TK]D-Fender | heads out |
| 00:58.42 | thansen | :( |
| 00:59.03 | thansen | that went just a tad over my head as to the correctional steps |
| 00:59.14 | thansen | [TK]D-Fender: thanks for the help |
| 01:00.53 | Raden | evening |
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| 01:11.27 | Knightfal | When using MixMonitor how do I make it write a new folder if it doen not currently exist |
| 01:11.45 | Knightfal | For example storing the calls by day |
| 01:12.00 | Knightfal | I have done it before but it doesnt seem to be working out for me |
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| 01:56.57 | kyosh | are there any windows apps that uses the manager api to provide real time montoring? |
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| 02:35.51 | kyosh | bah there's nothing good |
| 02:36.20 | kyosh | nothing that works for realtime |
| 02:36.30 | kyosh | i fporgot i said that |
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| 02:39.02 | wcselby | why will dahdi compile for me on one machine but not another....as far as I can tell everything is the same. It's a compile issue, not a dependancy issue, as far as I can tell.... :( |
| 02:43.44 | *** join/#asterisk Middleman (~Dystopia@dsl-173-206-241-120.tor.primus.ca) |
| 02:44.13 | Middleman | Hey, looking for clarification. If i buy a DID, that can be routed to IP and i dont need any PSTN hardware right. |
| 02:44.53 | Middleman | I was going to buy a digium analog card, but read i just need to buy DIDs instead |
| 02:44.56 | *** join/#asterisk brut- (~brut-@h66-173-4-254.mntimn.dedicated.static.tds.net) |
| 02:46.00 | wcselby | you can get a SIP trunk (incorrect term, but it gets the point across) from many SIP providers. Then all you need is an internet connection, and DAHDI_DUMMY for timing (unless you use one of the new timers, but then you'll need to use ConfBridge instead of MeetMe, etc...) |
| 02:46.02 | TJNII | Middleman: Two things: If you buy service through an ITSP then you do not need any analog hardware, and DID doesn't mean what you think it means. |
| 02:46.50 | Middleman | ok, thanks for the clarification |
| 02:47.09 | Middleman | i will look into this |
| 02:49.16 | Middleman | so if i want to be my own itsp, i still need a T1 or T3 to connect to the POTS? |
| 02:50.19 | TJNII | ~itsplist-us |
| 02:50.20 | infobot | Here are some popular ITSPs (USA) starting with the more respected ones: http://www.teliax.com , http://www.voicepulse.com/connect/ , http://www.broadvoice.com , http://www.jnctn.com , http://www.bandwidth.com , http://vitelity.net , http://voip.ms and http://flowroute.com |
| 02:50.30 | Middleman | thanks tj |
| 02:50.34 | TJNII | Why not look as some of the services they offer? |
| 02:51.05 | Middleman | ~itsplist-ca |
| 02:51.05 | infobot | [~itsplist-ca] Here are some popular ITSPs (Canada) starting with the more respected ones : http://www.unlimitel.ca , http://www.les.net , http://www.babytel.ca |
| 02:51.54 | Middleman | see, i signed up with les.net, and they are sellin Dids , i think i use that did and refer it to domain name for sip phones |
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| 03:00.35 | kyosh | wow les.net 's prices arent really that good are they |
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| 03:00.59 | kyosh | i mean they're ok |
| 03:01.37 | kyosh | well actually |
| 03:01.44 | kyosh | unmetered did's are pretty good |
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| 03:17.06 | ruben23 | hi guys |
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| 03:24.29 | jrthomerson | is there any plan to support speex wideband in 1.6.x? i saw a patch that was declined for 1.4, but couldn't find much about 1.6 |
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| 03:49.34 | Knightfal | Is there a way to change the number that is dialed when a parked call times out. I tried using [parkinglot] context but it doesnt seem to be working correctly |
| 03:49.47 | Knightfal | Someone give me a bit of direction id appreciate it. |
| 03:50.58 | ChannelZ | when you Park() you can specify the context, exten and priority to return to, can you not? |
| 03:52.19 | Knightfal | I thought so but i dont any documentation |
| 03:52.27 | ChannelZ | oh but you're probably transferring to the park extension from features.conf |
| 03:52.34 | Knightfal | well |
| 03:52.49 | Knightfal | was going to do 7XX |
| 03:52.53 | Knightfal | then Park() |
| 03:53.11 | Knightfal | I want to park it and then have it fail to the XX vm box |
| 03:53.58 | ChannelZ | well if you use the Park application you can specify the timeout exten and such.. core show application park |
| 03:55.07 | Knightfal | Im using 1.4.31 |
| 03:55.46 | ChannelZ | oh, those args aren't in 1.4? drat |
| 03:55.56 | Knightfal | So its not showing me much |
| 03:56.37 | Knightfal | ya I see this http://pastebin.com/mcRkrgM4 |
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| 03:59.24 | ChannelZ | 1.4 might not support timeouts then. Upgrade? |
| 03:59.56 | Knightfal | blah |
| 04:00.10 | ChannelZ | unless it just pays attention to the values in features.conf, but that might only be through the features.conf-style parking and not the Park() app |
| 04:00.27 | ChannelZ | And I"m not sure what it even does.. probably sends you to the 't' extension of the park context |
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| 04:00.55 | Knightfal | hmmm |
| 04:04.35 | Coolrazor | If my inbound route points to a Misc Destination and that Misc Destination has a bogus "dial", what happens? |
| 04:05.49 | ChannelZ | The machine will wipe out its hard drive and catch fire to its self. |
| 04:06.24 | ChannelZ | I also have no idea what a 'Misc Destination' is |
| 04:09.45 | Knightfal | ChannelZ http://www.voip-info.org/wiki/view/Asterisk+call+parking |
| 04:09.54 | Knightfal | Can you make sense of theat |
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| 04:14.16 | ChannelZ | Of what part |
| 04:15.01 | *** join/#asterisk magicblaze007 (~ravindra@117.205.147.159) |
| 04:15.52 | magicblaze007 | is anyone here? I need some help. I've a sip client that can make calls flawlessly. When I try to receive calls, it wont ring. At the asterisk CLI I see "Congestion". Any ideas on how to fix this? |
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| 04:20.53 | ChannelZ | Let's see more of the actual console output |
| 04:22.53 | magicblaze007 | here is something else. I've xlite on the same router...that can receive calls. |
| 04:22.57 | magicblaze007 | the spa wont receive calls |
| 04:23.11 | magicblaze007 | the spa can only make calls. |
| 04:23.36 | magicblaze007 | and i also tried the same extension on the x-lite client, that worked and could receive calls. |
| 04:23.55 | magicblaze007 | that seems to tell me that some setting on the spa needs changing...cant pin point what... |
| 04:23.57 | *** join/#asterisk romb (~romb@unaffiliated/romb-work/x-7222485) |
| 04:24.37 | ChannelZ | Is DND turned on or something? |
| 04:24.49 | magicblaze007 | lemme check... |
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| 04:25.41 | magicblaze007 | ChannelZ: By any chance do you know where on SPA 3102 menu is DND? |
| 04:26.32 | magicblaze007 | found it... |
| 04:26.37 | magicblaze007 | dnd serv = yes |
| 04:26.46 | ChannelZ | no. Voice tab, User 1 |
| 04:27.12 | ChannelZ | there is also a service code for turning it on and off, maybe you accidentally dialed it |
| 04:27.24 | magicblaze007 | dnd setting = no |
| 04:27.59 | magicblaze007 | somehow it also takes a long time to register and show up when i do "sip show peers" |
| 04:28.04 | magicblaze007 | but it can still make calls |
| 04:28.51 | magicblaze007 | Its still showing --> 13/13 (Unspecified) D N A 0 UNKNOWN , but the client does have dial tone and asterisk does see the client |
| 04:30.59 | magicblaze007 | any other ideas? |
| 04:31.09 | ChannelZ | Well if the thing is misconfigured and doesn't know how to reach your *..... |
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| 04:32.20 | ChannelZ | The 3102 appears as two SIP devices; One represents the FXO port (for making/reciving POTS calls) and the other represents the FXS port (an analog phone hooked up to the device) |
| 04:32.32 | magicblaze007 | yes indeed |
| 04:32.39 | magicblaze007 | I'm just using it as a dumb client right now |
| 04:32.49 | magicblaze007 | sip client - extension 13. |
| 04:33.01 | magicblaze007 | x-lite works behing this router |
| 04:33.13 | ChannelZ | ok neither 'sip client' nor 'extension 13' tells me anything |
| 04:33.43 | ChannelZ | Are you trying to use it as an FXO device or are you trying to use an analog phone hooked up to it? |
| 04:33.43 | magicblaze007 | I agree. |
| 04:34.51 | magicblaze007 | FXO - uniden dect 6 phone |
| 04:35.14 | ChannelZ | uhm |
| 04:35.44 | ChannelZ | you mean fxs then (the uniden is an analog phone) |
| 04:36.23 | magicblaze007 | the phone is plugged into the "phone" line of the spa-3102. |
| 04:37.06 | ChannelZ | So you can pick up the analog phone, dial a number, and it enters your dialplan correctly and goes someplace. |
| 04:37.12 | magicblaze007 | yes |
| 04:37.24 | ChannelZ | What are you trying to Dial() to reach the phone connected to the SPA? |
| 04:37.59 | magicblaze007 | the extension 13. That's what is set in Line 1 page of the SPA |
| 04:38.34 | p3nguin | uh no |
| 04:38.41 | p3nguin | That's not how it works. |
| 04:38.50 | carrar | Is extension 13 in hanger 13? |
| 04:38.51 | magicblaze007 | ? |
| 04:38.57 | ChannelZ | and he's running freepbx |
| 04:39.08 | p3nguin | oh brother. |
| 04:39.14 | magicblaze007 | when i use extension 13 on my x-lite it works |
| 04:39.20 | magicblaze007 | when i use it in my spa it does not |
| 04:39.42 | ChannelZ | I'm not even sure what you're talking about on the Line 1 page to boot |
| 04:40.10 | p3nguin | There's no place to enter "extension 13" on that page. |
| 04:40.44 | magicblaze007 | voice -> line 1 -> Subscriber information -> User id = 13 |
| 04:40.49 | magicblaze007 | auth id is empty |
| 04:40.57 | magicblaze007 | use auth id is no |
| 04:40.57 | p3nguin | alright |
| 04:41.08 | p3nguin | Did you put in the proxy address? |
| 04:41.18 | p3nguin | That's where the Asterisk address goes. |
| 04:41.19 | magicblaze007 | yes |
| 04:41.20 | ChannelZ | that's totally different than an extension number that you dial, which is a manifestation of the dialplan in Asterisk and has nothing to do with the device |
| 04:41.31 | p3nguin | And you probably don't need to use an outbound proxy. |
| 04:41.34 | magicblaze007 | no hold on-checking proxy |
| 04:41.40 | *** join/#asterisk n3hxs (~HAMming@72-255-41-166.client.stsn.net) |
| 04:41.45 | *** join/#asterisk moy (~moy@bas1-unionville55-1177733627.dsl.bell.ca) |
| 04:41.48 | magicblaze007 | yes proxy = my asterisk server = freepbx |
| 04:42.35 | magicblaze007 | register expires = 3600. register = yes. Info tab shows => Registration State = Registered |
| 04:42.46 | p3nguin | There you go, then. All done. |
| 04:42.53 | magicblaze007 | i cant receive calls |
| 04:43.06 | p3nguin | Does asterisk have a dialplan to send calls to your device? |
| 04:43.09 | magicblaze007 | it says "Congestion" and hangs up when i try to receive the call on an spa |
| 04:43.21 | p3nguin | Where's the sip debug? |
| 04:43.25 | *** join/#asterisk igorg (~igorg@net182.255.92-116.dynamic.omsk.ertelecom.ru) |
| 04:43.29 | magicblaze007 | it must. otherwise why can x-lite with the same extension receive calls |
| 04:43.53 | p3nguin | I see what you mean. |
| 04:45.06 | *** join/#asterisk mr_ian (~mr_ian@S0106001b63f49383.du.shawcable.net) |
| 04:46.40 | ChannelZ | fuck FreePBX and its shititude |
| 04:46.44 | magicblaze007 | the other wierd thing is that the spa shows "Registered". whereas "sip show peers" shows -> 13/13 (Unspecified) D N A 0 UNKNOWN |
| 04:47.53 | p3nguin | lol |
| 04:47.57 | p3nguin | shititude |
| 04:49.00 | magicblaze007 | dont you think its not freepbx/asterisk but my spa/router thats the problem? |
| 04:50.39 | ChannelZ | It could be anything but freepbx just gets in the way in figuring it out |
| 04:51.05 | ChannelZ | what does the entry in sip.conf look like for that device? |
| 04:53.26 | TedNJ40 | Cam someone help me please? |
| 04:53.40 | TedNJ40 | I need to re-install tbm-guicore from the command line. How can I do it? |
| 04:54.52 | ChannelZ | yum? apt? configure/make/make install? We don't know what you're running.... |
| 04:55.14 | carrar | Running ATARI OS |
| 04:55.19 | carrar | HOT++ |
| 04:57.14 | *** part/#asterisk TedNJ40 (~HungLad@ool-43527869.dyn.optonline.net) |
| 05:00.26 | ChannelZ | Oh that's easy then. Take out the cartridge, blow on it, and then put it back in. |
| 05:11.17 | *** join/#asterisk ruben23 (~ITadmin@125.212.40.2) |
| 05:17.23 | *** join/#asterisk ianweller (~ianweller@fedora/ianweller) |
| 05:22.41 | *** join/#asterisk githogori (~githogori@adsl-66-123-22-146.dsl.snfc21.pacbell.net) |
| 05:28.13 | *** join/#asterisk kruemeltee (~Maddin@port-92-198-62-82.static.qsc.de) |
| 05:31.17 | *** part/#asterisk magicblaze007 (~ravindra@117.205.147.159) |
| 05:57.13 | *** join/#asterisk Znuff (~ibm86@89.47.35.9) |
| 06:03.40 | *** join/#asterisk spenguin[work] (~penguin@59.162.86.164) |
| 06:03.44 | spenguin[work] | TEST |
| 06:03.53 | *** part/#asterisk ianweller (~ianweller@fedora/ianweller) |
| 06:23.37 | *** join/#asterisk romb (~romb@unaffiliated/romb-work/x-7222485) |
| 06:27.02 | *** join/#asterisk kaldemar (~kaldemar@unaffiliated/kaldemar) |
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| 06:40.32 | drmessano | oh god |
| 06:40.39 | drmessano | tbm-guicore :( |
| 06:47.57 | ChannelZ | core dumps |
| 06:52.23 | *** join/#asterisk SingerWang (~wang@203-206-207-156.perm.iinet.net.au) |
| 06:52.36 | SingerWang | hey does anyone know about the g729 licensing well enough to answer a question |
| 06:52.42 | ChannelZ | That's a talented wang. |
| 06:52.53 | SingerWang | if I have 2 asterisk boxes and I need to call from one to the other |
| 06:53.00 | SingerWang | 1 call, do I need one license on each side? |
| 06:53.04 | SingerWang | 1 total or 2 on each side? |
| 06:53.15 | ChannelZ | It depends |
| 06:53.44 | ChannelZ | You need one on each side |
| 06:54.04 | SingerWang | phone1->uLaw->Asterisk1->G729->Asterisk2->ulaw->Phoen2 |
| 06:54.10 | ChannelZ | Though if * doesn't need to be in the media stream at all, it can pass-thru g729 if the end points have licenses I think (IE your phones) |
| 06:54.12 | SingerWang | in this scenarion, phone1 calling phone2 |
| 06:55.14 | *** join/#asterisk lowlevel (~Stuart@lowlevel.ca) |
| 06:55.43 | ChannelZ | yeah.. it'd be more like phone1 (ulaw) -> Asterisk 1 (ulaw-to-g729) -> Asterisk 2 (g729-to-ulaw) -> phone2 |
| 06:56.13 | SingerWang | true.. I more labelled the codec in the channel in use.. for that scenario I need 2 (1 each side right?) |
| 06:56.21 | ChannelZ | yes |
| 06:56.49 | SingerWang | okay.. I'll have to order some up.. |
| 06:57.12 | SingerWang | also, how is the gsm codec in terms of voice quality? |
| 06:57.30 | ChannelZ | it's a little scratchy but not terrible |
| 06:57.55 | SingerWang | so compare that to g729 g729 is better? |
| 06:58.41 | WIMPy | I'm not sure, but usually only the old GSM codec is implemented. |
| 06:58.48 | ChannelZ | I dunno, I've never really done any objective side-by-side's since I don't really use either. All my devices are internal and thus all ulaw |
| 06:59.05 | SingerWang | I have all that in canada |
| 06:59.18 | SingerWang | but I'm setting up a AA60 in Sydney here |
| 06:59.25 | SingerWang | and bandwidth is a premium :( |
| 07:00.31 | *** join/#asterisk tamiel (~tamiel@213.30.183.226) |
| 07:02.17 | spenguin[work] | SingerWang: just curious do a lot of businesses run opensource in AU? |
| 07:02.30 | spenguin[work] | or NSW |
| 07:03.38 | SingerWang | spenguin[work]: I realluy don't know.. I'm only down here for 4 months to setup more infrasture for the new branch office and I just arrived 3 days ago |
| 07:04.21 | SingerWang | but there is oen thing I don't like about austraila |
| 07:04.39 | SingerWang | and that's asterisk stuff is overpriced |
| 07:04.50 | *** join/#asterisk Tim_Toady (~moi@178.128.28.90.dsl.dyn.forthnet.gr) |
| 07:04.58 | SingerWang | even thought its closer to China |
| 07:06.17 | spenguin[work] | SingerWang: you mean digium hardware? |
| 07:06.30 | SingerWang | and the other stuff |
| 07:07.02 | SingerWang | for example: in our main office we have Netgear FS728TP for switches.. they retail around $300 CAD... I can't source them here for any less then $700 |
| 07:07.17 | spenguin[work] | man |
| 07:07.21 | spenguin[work] | thats expensive |
| 07:07.36 | spenguin[work] | buy them online |
| 07:07.39 | spenguin[work] | or from India |
| 07:07.39 | SingerWang | PolyCOM IP450 (we use the 430 at the main office but they are EOL'ed).. can be gotten for $200 in Canada, $307 is the cheapest I can find there here |
| 07:07.42 | spenguin[work] | its all cheap here |
| 07:07.56 | SingerWang | (1CAD = 1.04 AUD) |
| 07:08.02 | spenguin[work] | shipping might be expensive though |
| 07:08.17 | SingerWang | problem is we want local support.. the office here isn't technical enough to do it themselves |
| 07:08.29 | spenguin[work] | afaik, bandwidth is pretty expensive too |
| 07:08.33 | spenguin[work] | in AU |
| 07:08.55 | SingerWang | yeah.. the hardware I dont' get its MUCH closer to China then USA |
| 07:10.09 | spenguin[work] | I guess the vendors are taking a huge scoop |
| 07:10.12 | spenguin[work] | in AU |
| 07:10.22 | SingerWang | gotta also upgrade the mail server |
| 07:10.31 | SingerWang | that was the first issue we had after we upgraded to asterisk (GRR) |
| 07:16.58 | *** join/#asterisk sawgood (~sawgood@173-13-158-25-sfba.hfc.comcastbusiness.net) |
| 07:21.38 | spenguin[work] | Hey, need to add another pri card for another PRI line into the current asterisk box, instead was thinking of having a bridge (redphone) or such |
| 07:21.46 | *** join/#asterisk sgimeno (~chatzilla@226.Red-80-33-64.staticIP.rima-tde.net) |
| 07:21.52 | spenguin[work] | is it a good idea, or any recommendations |
| 07:26.18 | *** join/#asterisk renshen (~renshen@93-63-217-144.ip29.fastwebnet.it) |
| 07:26.34 | *** join/#asterisk Da-Geek (~Da-Geek@a88-112-255-212.elisa-laajakaista.fi) |
| 07:28.11 | *** join/#asterisk krion (~seb@unaffiliated/krion) |
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| 07:41.29 | *** join/#asterisk gr0mit (~tim@extrt.txrx.org.uk) |
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| 07:45.58 | *** join/#asterisk gego (~quassel@b238085.customer.hansenet.de) |
| 07:48.34 | *** join/#asterisk Seb^ (~sebspiers@188.39.20.226) |
| 08:02.16 | *** join/#asterisk werdan7 (~w7@freenode/staff/wikimedia.werdan7) |
| 08:08.53 | *** join/#asterisk Coolrazor (~Tim@76.91.184.133) |
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| 08:28.52 | *** join/#asterisk Z_God (~julius@wlan232002.mobiel.utwente.nl) |
| 08:30.01 | *** join/#asterisk IsUp (~nocturne@unaffiliated/isup) |
| 08:30.03 | IsUp | hello |
| 08:30.22 | IsUp | i am always getting this warning messages, "Maximum retries exceeded on transmission SBCac9383f24fa7eeeb246b7bd701a4ab61@10.18.5.64 for seqno 1 (Critical Response) -- See doc/sip-retransmit.txt." |
| 08:31.43 | *** join/#asterisk QN (~mIRC@cable-dyn.116-51-217-2.ntti.net.sg) |
| 08:32.12 | QN | hi all. anyone has any idea on how could i get ipkall to work with elastix properly without allowing annonymous sip? |
| 08:34.31 | IsUp | QN, ipkall using 1 or 2 IPs for sending calls to you |
| 08:34.39 | IsUp | define them to your Elastix |
| 08:35.41 | IsUp | as an alternative, you can check http://www.ukddi.com/ |
| 08:35.46 | IsUp | it's free too |
| 08:35.49 | *** join/#asterisk chilicuil (~chilicuil@unaffiliated/chilicuil) |
| 08:37.32 | IsUp | "Calls will come from 66.54.140.46 (voiper.ipkall.com) or 66.54.140.47 (voiper2.ipkall.com)" |
| 08:37.33 | *** join/#asterisk giany (~giany@shifu.x83.org) |
| 08:37.52 | giany | hello |
| 08:37.58 | giany | is a line like this : SIPAddHeader(Alert-Info: \;info=Ring Answer) valid? |
| 08:38.08 | giany | isn't the ; supposed to be a comment? |
| 08:38.52 | IsUp | it's correct, i think |
| 08:39.19 | QN | IsUp: do you happen to know how do i defind them in elastix? i tried adding the DID number into the incoming route, it dosen't work. =( |
| 08:39.42 | giany | IsUp: i'm asking because in the SIP header I see this : Alert-Info: ;info=Ring Answer |
| 08:39.51 | giany | and some phones doesn't like that as far as I can see |
| 08:40.08 | IsUp | So use it as "SIPAddHeader(Alert-Info: ;info=Ring Answer)"? |
| 08:40.18 | IsUp | QN: PM me |
| 08:40.50 | *** join/#asterisk ltd (~z@pat.transact.net.au) |
| 08:40.59 | IsUp | giany: you are using Aastra phones? |
| 08:41.26 | giany | yes |
| 08:41.56 | IsUp | check here: http://www.voip-info.org/wiki/view/Asterisk+and+Aastra+Phones |
| 08:43.16 | giany | ah..I've missed that..I'll test it |
| 08:43.16 | giany | thx |
| 08:46.51 | *** join/#asterisk blop (~blop@217.112.179.12) |
| 08:46.58 | blop | hello |
| 08:47.27 | IsUp | hello |
| 08:48.29 | blop | i have some trouble getting FXS modules of one of my TDM400P to work properly, i upgraded to dahdi-linux-2.3, can it be related ? |
| 08:49.00 | IsUp | probably |
| 08:49.03 | giany | blop: dahdi_tool |
| 08:49.07 | giany | does it show anything? |
| 08:49.15 | IsUp | and do: dahdi_scan |
| 08:49.35 | *** join/#asterisk joobie (~joobie@CPE-121-220-3-162.lnse1.win.bigpond.net.au) |
| 08:49.36 | giany | 2.3.0 has some compiling issues |
| 08:50.03 | joobie | man what a week |
| 08:50.11 | joobie | cracks open a bottle of scotch |
| 08:50.11 | giany | IsUp: is it possible to add in SIP header something like : INVITE sip:204@192.168.1.142:5060;transport=udp;intercom=true SIP/2.0 |
| 08:50.16 | blop | giany: i get my 2 TDM400P listed |
| 08:50.29 | giany | IsUp: notice the intercom=true |
| 08:50.42 | giany | IsUp: seems that newer Astra phones need that too.. |
| 08:51.10 | giany | blop: ok, define work properly then..what is wrong with them? |
| 08:51.11 | IsUp | giany: i dont know so much about Aastra phones, i never did any tricks to make phones work :P |
| 08:51.55 | giany | IsUp: this kinda sucks big time..every time a new device is out they modify something then i need to adapt the code.. |
| 08:52.01 | blop | well, so I have 2 TDM400P, one of them with 4 FXS, one with 1 FXO + 3 FXS. |
| 08:52.16 | blop | when i load the wctdm module, I only get 4 FXS + 1 FXO |
| 08:52.22 | blop | and i get things like : |
| 08:52.24 | blop | <PROTECTED> |
| 08:52.24 | blop | Indirect Registers failed verification. |
| 08:52.24 | blop | Module 1: FAILED FXS (FCC) |
| 08:52.24 | blop | ProSLIC on module 2 insane (1) 18 should be 2 |
| 08:52.24 | blop | Module 2: Not installed |
| 08:52.24 | blop | ProSLIC on module 3 insane (3) |
| 08:52.25 | blop | Module 3: Not installed |
| 08:52.27 | blop | :) |
| 08:52.28 | IsUp | ooops |
| 08:52.29 | *** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/atheme.member.chainsaw) |
| 08:52.33 | IsUp | use pastebin |
| 08:53.49 | blop | http://pastebin.com/3YK92nnk :) |
| 08:54.35 | blop | so i tried removing the tdm400p that was working, and put only the 1 FXO + 3 FXS card, not better |
| 08:54.54 | blop | i tried removing/swapping the FXS modules |
| 08:54.57 | blop | no luck :) |
| 08:55.17 | IsUp | did you removed all old drivers? |
| 08:55.23 | IsUp | try a fresh DAHDI install |
| 08:56.07 | blop | i have a fresh new kernel 2.6.34 |
| 08:56.22 | blop | and i recompiled all the dahdi tools/modules |
| 08:56.31 | Chainsaw | 2.6.34 should be okay, I haven't seen compile failures there with 2.3.0 |
| 08:56.41 | blop | it compiles fine |
| 08:56.48 | Chainsaw | That's a good start :) |
| 08:56.53 | IsUp | i am using 2.6.32, works great |
| 08:57.02 | russellb | if you don't get it going, please contact digium tech support so that they can get the dahdi team to fix it - http://www.digium.com/en/supportcenter/ |
| 08:57.21 | blop | uhm ok |
| 08:58.29 | russellb | digium hw, right? |
| 08:58.37 | blop | yup |
| 08:58.40 | russellb | nods |
| 09:03.12 | Chainsaw | blop: I see PCI master aborts. Just a hunch... can I see your /proc/interrupts please? |
| 09:04.35 | *** join/#asterisk Dovid (~annon@213.8.121.90) |
| 09:05.25 | Chainsaw | blop: Because unless you don't have *any* legacy serial ports... IRQ 3 is unlikely to be free for use. |
| 09:05.47 | blop | Chainsaw : http://pastebin.com/90M66fuP |
| 09:06.21 | Chainsaw | blop: I stand corrected. Your IRQs are not shared. |
| 09:06.31 | blop | :) |
| 09:06.39 | blop | which is good |
| 09:06.51 | WIMPy | Chainsaw: How would you know? |
| 09:06.53 | Chainsaw | blop: Which is good, because you have the old-fashioned dual-PIC setup for interrupts. |
| 09:07.11 | Chainsaw | WIMPy: In case of a shared IRQ, you will see more then one driver name listed. |
| 09:07.14 | *** join/#asterisk hmny (~Frans@89.146.33.179) |
| 09:07.21 | Chainsaw | WIMPy: Something along the lines of: ehci_hcd, ohci_hcd, wctdm |
| 09:07.42 | Chainsaw | WIMPy: 11 is shared for example. Ethernet & USB. |
| 09:07.44 | hmny | hi everybody! |
| 09:07.51 | Chainsaw | WIMPy: And even sound. |
| 09:07.54 | Chainsaw | Morning hmny. |
| 09:08.00 | hmny | I'm new here, just started with asterisk and voip and what not :) |
| 09:08.07 | Chainsaw | Any questions so far? |
| 09:08.45 | hmny | nah not as of yet. I am still struggeling to understand all these protocols |
| 09:08.50 | blop | the tdm on irq 3 is working fine, its the one on the irq 5 which is not, and there is a very low number of interrups :/ |
| 09:08.54 | WIMPy | But you can't know what's good. |
| 09:09.18 | WIMPy | I've already seen disabled hardware making IRQs unusable. |
| 09:10.12 | Chainsaw | blop: Yes, I tend to expect an APIC setup these days, which will make more IRQs available for use. (And Message Signalled Interrupts can improve the situation further) |
| 09:10.50 | Chainsaw | blop: If you don't mind me asking... how old is this system? |
| 09:11.08 | Chainsaw | blop: Because I haven't seen non-APIC level-triggered interrupts since the Pentium II days. |
| 09:11.20 | blop | its a lenovo thinkcenter m55 core2 duo |
| 09:11.37 | Chainsaw | blop: Are there any BIOS updates available that you know of? |
| 09:11.48 | Chainsaw | blop: Failing that, see if your BIOS has an "APIC" setting. That'll need to be on. |
| 09:12.06 | Chainsaw | blop: Anything involving ACPI (particularly MADT) needs to be enabled as well. |
| 09:12.42 | *** join/#asterisk m_tadeu (~quassel@89.180.4.126) |
| 09:12.42 | Chainsaw | blop: And as a last resort... if you see an MPTABLE option, set it to 1.4 |
| 09:13.45 | blop | i upgraded the bios yesterday :) |
| 09:14.03 | Chainsaw | blop: Okay, please check the options I mentioned. Make a note of any changes you make. |
| 09:14.21 | *** join/#asterisk romb_work (~romb@unaffiliated/romb-work/x-7222485) |
| 09:14.27 | Chainsaw | blop: If you look in /proc/interrupts again you should see a different system then XT-PIC in use. |
| 09:14.45 | blop | ok, but i dont have physical access to the machine right now |
| 09:15.00 | Chainsaw | *nod* When you get the opportunity. |
| 09:15.21 | blop | :) |
| 09:15.47 | Chainsaw | hmny: Well, it's important to remember that SIP does call setup. And call setup only. |
| 09:16.20 | Chainsaw | hmny: If I call you, SIP packets will flow between our machines to set up a session. And then the actual speech will go over RDP packets (two streams, one me->you and one you->me). |
| 09:17.05 | Chainsaw | hmny: RTP even, sorry. |
| 09:17.44 | hmny | Ah ok interesting. |
| 09:17.50 | *** join/#asterisk sourcode (~code@ppp-58-8-122-10.revip2.asianet.co.th) |
| 09:18.09 | hmny | my first project will be to setup an asterisk server, my ip phone will go to that server, and the asterisk server will connect to our mitel. |
| 09:18.26 | hmny | now I see the mitel only has license for sip trunks, so I will setup a sip trunk between them. |
| 09:18.41 | hmny | if thats working I am gonna try to get mutliple phones to ring when my number is called. |
| 09:18.58 | Chainsaw | hmny: Okay, so you're mostly looking at writing dial plans for answering machine/interactive menu/call filtering situations? |
| 09:19.11 | hmny | well. I am getting an android with internet |
| 09:19.23 | hmny | zo I want my phone to ring everywhere, at work,. on my pc, on my android, on my ipphone ;) |
| 09:19.24 | *** join/#asterisk gr0mit (~tim@extrt.txrx.org.uk) |
| 09:19.44 | Chainsaw | hmny: *nod* Asterisk can do that for you. How do you use that Mitel now? |
| 09:19.46 | hmny | I am assuming its possible, but like I said, very fresh at this. |
| 09:19.49 | *** join/#asterisk af_ (~getsmart@88-149-241-99.dynamic.ngi.it) |
| 09:19.57 | Chainsaw | hmny: The IP phone registers straight to it? |
| 09:19.59 | hmny | Chainsaw: our mitel ip phones connect to them in mitel mode |
| 09:20.04 | hmny | so not in sip mode. |
| 09:20.19 | hmny | and I see on the mitel box it has no sip user licenses, so I think I can only use the trunk for now. |
| 09:20.26 | *** join/#asterisk Tim_Toady (~moi@77.49.18.121) |
| 09:20.43 | Chainsaw | hmny: I hope it'll work at all. It may just decide "no SIP for you!" if there are no licenses for it. |
| 09:20.53 | hmny | hehe it has a trunk license |
| 09:20.55 | hmny | 2 I think. |
| 09:21.13 | hmny | I am hoping over the trunk I can route infinite calls. but like I said I am sitll learning about what I can do with all this :) |
| 09:21.30 | hmny | luckly I know bgp and ospf and all that routing business so it should be doable for me. |
| 09:21.42 | Chainsaw | hmny: It's worth having a strategic google with the words Mitel, the type number of the box you have and the word Asterisk. |
| 09:21.55 | Chainsaw | hmny: It's unlikely that you're the first to try this. |
| 09:21.55 | hmny | nods |
| 09:22.03 | hmny | I found many similar setups already yes. |
| 09:22.54 | Chainsaw | If you use your routing expertise to avoid NAT... you will save yourself quite some headaches. |
| 09:23.26 | hmny | hmmm ok. yes thats doable. |
| 09:24.22 | hmny | Chainsaw: any books you would recommend ? |
| 09:25.14 | Chainsaw | (Because with NAT, and those two independent RTP streams being set up... there is one-way audio in your future) |
| 09:25.34 | Chainsaw | hmny: There's word of an Asterisk book, but it seems written for 1.4 while we are at 1.6.2 these days. |
| 09:26.05 | Chainsaw | hmny: What I will warn you about is voip-info.org; it is a wiki site with some good information, some very outdated articles and some that are factually incorrect. |
| 09:26.11 | hmny | how about Practical Asterisk 1.4 and 1.6: From Beginner to Expert (Paperback) Stefan Wintermeyer (Author), Stephen Bosch (Author) |
| 09:26.20 | Chainsaw | hmny: I'm not saying don't use it at all, but check against other sites. |
| 09:26.34 | hmny | Chainsaw: lol thanks for that. I have many bookmarks to that site already ;) |
| 09:26.35 | Chainsaw | hmny: i've not read that, so I can't vouch for it. But yes, make sure it is written with 1.6 in mind. |
| 09:26.41 | hmny | I will not trust it blindly now hehe. |
| 09:26.49 | hmny | Ok |
| 09:26.52 | hmny | thanks for your time :) |
| 09:26.56 | Chainsaw | You're welcome :) |
| 09:27.02 | hmny | I will lurk in here and report back my adventures ;) |
| 09:27.13 | Chainsaw | Alright, I'll be around :) |
| 09:28.03 | drmessano | ~boschlawnmowers |
| 09:28.04 | infobot | http://www.voip-info.org/wiki/view/Bosch+lawnmowers .. "Same to you, buddy!" |
| 09:32.00 | *** join/#asterisk stephank (~urk@212.178.158.35) |
| 09:35.50 | stephank | Hi! We're using TE412Ps, with the VPM450 echo canceller. My understanding is that the echo canceller disables itself automatically when it detects a fax or modem, or is there some configuration involved? (asterisk 1.4.31, dahdi 2.3.0) |
| 09:40.49 | *** join/#asterisk Martinblr (~Miranda@61.12.17.170) |
| 09:49.37 | *** join/#asterisk viraptor (~viraptor@212.11.65.66) |
| 09:50.47 | viraptor | hi, is there a way to make asterisk switch the rtp destination if it starts receiving the stream from an unexpected source? (client behind nat with port randomisation situation) |
| 09:55.46 | russellb | nat=yes :-) |
| 09:58.34 | *** join/#asterisk huye (~huye@soho2.i-xanadu.com) |
| 09:59.15 | viraptor | russellb: is there a way to not touch any of the sip nat options and still affect rtp? i.e. I want to deal with address mapping, routes, rport, etc. in some other way... - I need only the symmetric rtp enabled really |
| 10:03.01 | *** join/#asterisk romb (~romb@unaffiliated/romb-work/x-7222485) |
| 10:03.16 | russellb | I don't think there is a way to enable that specifically |
| 10:03.23 | russellb | it's just bundled into the big fat "nat" option I think |
| 10:03.25 | russellb | i'll look though ... |
| 10:04.13 | IsUp | how can i edit voicemail completely? |
| 10:04.22 | IsUp | it's not supporting Turkish sound files |
| 10:04.27 | viraptor | russellb: ok, thanks |
| 10:04.28 | IsUp | so i need make some changes |
| 10:04.42 | russellb | viraptor: doesn't look like it |
| 10:05.16 | russellb | IsUp: open up app_voicemail.c ... there is a ton of language specific code in there |
| 10:06.39 | viraptor | russellb: where would be the likely place to patch it: chan_sip or rtp? |
| 10:06.54 | russellb | chan_sip I think |
| 10:07.01 | russellb | it shouldn't be _too_ hard to do |
| 10:07.14 | russellb | it should just be a tweak to the existing nat= option handling |
| 10:07.21 | russellb | to say symmetric RTP _only_ plzkthx |
| 10:07.21 | IsUp | russellb: how can i compile app_voicemail.so without compiling whole Asterisk? |
| 10:07.31 | IsUp | i mean app_voicemail.c |
| 10:07.32 | russellb | IsUp: not easily |
| 10:07.43 | IsUp | any gcc command to make it? |
| 10:07.46 | russellb | you can run menuselect and disable most modules i guess ... |
| 10:07.50 | viraptor | cool, thanks... |
| 10:07.56 | IsUp | oh |
| 10:07.58 | IsUp | got it |
| 10:10.22 | Martinblr | did video is supported in SIP? |
| 10:12.39 | IsUp | yes |
| 10:12.50 | IsUp | h263, h264 |
| 10:13.22 | Martinblr | IsUp: is it related to H323...? |
| 10:15.22 | kaldemar | Martinblr: ITU-T standards, hence the naming. |
| 10:18.08 | Martinblr | kaldemar: is it only the naming standard between H323 and H263,H264? |
| 10:18.18 | *** join/#asterisk Tim_Toady (~moi@62.1.148.103.dsl.dyn.forthnet.gr) |
| 10:19.03 | kaldemar | Martinblr: well, H.323 may also use those codecs for video, but they're just names that don't indicate any requirements. |
| 10:20.57 | Martinblr | kaldemar: as like SIP and IAX.. Am i correct? |
| 10:22.16 | *** join/#asterisk TimeRider (~steve@109.224.131.68) |
| 10:22.26 | *** join/#asterisk m_tadeu (~quassel@89-180-4-126.net.novis.pt) |
| 10:24.08 | m_tadeu | hi everyone...what's the difference from asterisk to switchboard? |
| 10:26.01 | kaldemar | Martinblr: well... SIP is on IETF standards track, while IAX is only in the informational category, and they are both signaling protocols (although IAX also defines a payload type). H.323 is a signaling protocol and H.263 and H.264 are video codecs. |
| 10:26.14 | kaldemar | Martinblr: depends on what you mean. |
| 10:27.02 | Martinblr | kaldemar: ok I got it, it seems to like G729 in SIP, simliar like H264 in H323 |
| 10:27.15 | *** join/#asterisk centoslinux (~centoslin@1x-193-157-205-236.uio.no) |
| 10:29.32 | *** join/#asterisk kartik (~koolkarti@117.207.83.242) |
| 10:36.06 | *** join/#asterisk Tim_Toady (~moi@62.1.148.103.dsl.dyn.forthnet.gr) |
| 10:36.31 | blop | going back to 2.6.28 and my old dahdi-drivers to test :) |
| 10:42.39 | QN | gtg.. bye all |
| 10:45.25 | *** join/#asterisk Infin1ty (~Infin1ty@pdpc/supporter/active/infin1ty) |
| 10:48.27 | *** join/#asterisk ming_zym (~ming_zym@114.251.86.0) |
| 10:52.27 | *** join/#asterisk Trixboxer (~Trixboxer@115.113.145.85) |
| 10:53.00 | *** join/#asterisk romb (~romb@unaffiliated/romb-work/x-7222485) |
| 10:56.11 | *** part/#asterisk ming_zym (~ming_zym@114.251.86.0) |
| 10:58.01 | *** join/#asterisk kchehab (~ksdc@212.98.141.202) |
| 10:58.15 | kchehab | hi ppl |
| 10:58.57 | kchehab | i am trying to build a qualiy indicator ,so i am in need to record the tx channel |
| 10:59.11 | kchehab | how can i record only the tx channel only |
| 10:59.28 | IsUp | are you using DAHDI or SIP, IAX? |
| 10:59.32 | kchehab | sip |
| 10:59.39 | IsUp | ummm |
| 10:59.41 | IsUp | i dunno |
| 11:00.13 | kchehab | i used Macro(record-enable but it record both channels |
| 11:00.23 | *** join/#asterisk sergey (~sergey@ua0zeh.iks.ru) |
| 11:00.31 | kchehab | any usefull idea's |
| 11:02.12 | kchehab | something like recording chanspy |
| 11:02.20 | *** join/#asterisk Katoen (70baa4b29b@xs8.xs4all.nl) |
| 11:02.36 | *** join/#asterisk Trixboxer (~Trixboxer@datacenter3.supportdepartment.net) |
| 11:03.56 | kaldemar | kchehab: core show application Monitor |
| 11:04.59 | kaldemar | kchehab: then again, if you're using freepbx, ask in #freepbx |
| 11:07.36 | kchehab | kaldemar thanks |
| 11:11.24 | *** join/#asterisk Chinorro (~Chino@244.213.117.91.dynamic.mundo-r.com) |
| 11:13.29 | *** join/#asterisk Infin1ty (~Infin1ty@pdpc/supporter/active/infin1ty) |
| 11:21.29 | *** join/#asterisk Naikrovek (~jjohnson@unaffiliated/naikrovek) |
| 11:22.16 | *** join/#asterisk leifmadsen (~Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 11:22.16 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 11:26.34 | m_tadeu | what is the admin login/pass for asterisknow? |
| 11:31.24 | Tim_Toady | m_tadeu admin/admin or freepbx/fpbx |
| 11:31.39 | *** join/#asterisk puzzled (~foobar@puzzled.xs4all.nl) |
| 11:32.02 | m_tadeu | Tim_Toady: cool :) thx |
| 11:32.22 | Tim_Toady | make sure u change them both for obvious reasons |
| 11:32.44 | *** join/#asterisk manopulus (~manopulus@217.118.212.45) |
| 11:34.43 | manopulus | hello. question. for some reason i want to mark sip calls with custome header. why this code won't work? SIPAddHeader(X-blah: a=123\;b=111); NoOp(${SIP_HEADER(X-blah)}/${SIP_HEADER(blah)}); i receiving nothing on asterisk. but after that, when call goes out to kamailio, at kamailio i have X-blah, can read it. |
| 11:35.28 | Naikrovek | where the heck is Katty lately |
| 11:36.31 | *** join/#asterisk Chinorro (~Chino@244.213.117.91.dynamic.mundo-r.com) |
| 11:40.20 | *** join/#asterisk romb (~romb@unaffiliated/romb-work/x-7222485) |
| 11:57.26 | *** join/#asterisk [TK]D-Fender (~chatzilla@216.191.106.163) |
| 11:58.34 | m_tadeu | Tim_Toady: just trying asterisk in a VM for now...latter, for sure :) |
| 11:59.49 | *** join/#asterisk arpu (~arpu@chello062178159144.10.14.univie.teleweb.at) |
| 12:03.00 | *** join/#asterisk rttrey (~trey@andc-office-fw.atlantic.net) |
| 12:03.06 | m_tadeu | on the FOP screen, what sould I insert as security code? |
| 12:05.30 | Tim_Toady | m_tadeu passwords for fop database web intergace ami etc are set in /etc/amportal.conf |
| 12:05.50 | Tim_Toady | for fop the default is passw0rd IIRC |
| 12:06.13 | IsUp | use #freepbx |
| 12:06.27 | IsUp | if you have questions regarding FreePBX |
| 12:07.22 | *** join/#asterisk cusco (~trilili@213.63.137.210) |
| 12:07.24 | cusco | hi |
| 12:07.29 | cusco | Extension '0' in context 'incoming-pri' from '935296916' does not exist. Rejecting call on channel 0/1, span 7 |
| 12:08.23 | cusco | http://paste.debian.net/75172/ thats the AEL |
| 12:08.26 | cusco | it should work |
| 12:08.30 | m_tadeu | thanx again |
| 12:08.36 | *** part/#asterisk rttrey (~trey@andc-office-fw.atlantic.net) |
| 12:09.21 | *** join/#asterisk coppice (~chatzilla@m121-202-107-14.smartone-vodafone.com) |
| 12:13.42 | cusco | why extension 0? |
| 12:14.58 | *** join/#asterisk Poincare (~jefffnode@213.219.184.23.wls.msr03bkc3.adsl.static.edpnet.net) |
| 12:15.04 | *** join/#asterisk saftsack (~oliver@p5DDCF9FA.dip.t-dialin.net) |
| 12:15.18 | saftsack | hey, what is rtp codec 100? [May 28 11:55:29] NOTICE[3501]: res_rtp_asterisk.c:2076 ast_rtp_read: Unknown RTP codec 100 received from '10.10.10.179' |
| 12:15.46 | saftsack | i've heard of something related to faxing ... (t.38?) |
| 12:19.36 | *** join/#asterisk chuckf (~chuckf@ubuntu/member/chuckf) |
| 12:20.06 | *** join/#asterisk Yon (~Yon@212.247.19.244) |
| 12:21.22 | *** join/#asterisk verywiseman (~khaled@unaffiliated/verywiseman) |
| 12:23.04 | *** join/#asterisk n3hxs (~HAMming@173-162-253-153-NewEngland.hfc.comcastbusiness.net) |
| 12:26.04 | kaldemar | cusco: someone is dialing 0. |
| 12:27.00 | *** join/#asterisk romb (~romb@unaffiliated/romb-work/x-7222485) |
| 12:30.58 | *** join/#asterisk arpu (~arpu@chello062178159144.10.14.univie.teleweb.at) |
| 12:33.19 | saftsack | [May 28 12:21:06] NOTICE[3703]: res_fax.c:1131 receivefax_t38_init: Negotiated T.38 for receive on SIP/fax-0000001e |
| 12:33.20 | saftsack | [May 28 12:21:06] ERROR[3703]: res_fax.c:575 fax_session_new: Could not locate a FAX technology module with capabilities (0xA) |
| 12:33.26 | saftsack | what is technology 0xA? |
| 12:36.21 | Gnarfy | are the rtpkeepalive shown in rtp debug as: "Sent Comfort Noise RTP packet to xxx" ? |
| 12:36.32 | *** join/#asterisk wubba (~kmurrey@24.140.169.52) |
| 12:36.47 | wubba | is there a way to dial directly from the cli for testing? |
| 12:37.02 | cusco | kaldemar: ok.. telco sends only the last digit (as we only have 10 DDIs) |
| 12:37.07 | cusco | grrr |
| 12:37.32 | saftsack | Gnarfy, are you talking to me? |
| 12:37.33 | xheliox | well that's not at all short sighted. |
| 12:37.41 | m_tadeu | how can I experiment a voice call with a fresh installation of asterisknow? |
| 12:37.48 | Gnarfy | Ah no saftsack, just a general question... |
| 12:38.16 | cusco | kaldemar: can I have that? http://paste.debian.net/75174/ |
| 12:38.50 | cusco | having 0 => { bla} goto the same context,210332344,1 |
| 12:39.00 | cusco | that should be caught by _NXXXXXXXX |
| 12:39.06 | cusco | in that context, right? |
| 12:39.25 | [TK]D-Fender | cusco: What do you think? Does yuor Goto look valid? |
| 12:39.36 | cusco | it looks like, but doesnt' work |
| 12:39.38 | cusco | :( |
| 12:40.24 | [TK]D-Fender | cusco: Of course not. You can't seem to spell "incoming" properly consistently |
| 12:41.06 | cusco | argh!!!!!!!!!!!!! |
| 12:41.10 | manopulus | cusco: in AEL use jump statement, not goto |
| 12:41.11 | cusco | im rushing trough it |
| 12:41.21 | cusco | manopulus: really? why? |
| 12:41.28 | cusco | thanks a bunch [TK]D-Fender |
| 12:41.43 | manopulus | jump ${EXTEN} = goto ${EXTEN},1 |
| 12:42.20 | manopulus | cusco: i had troubles with some 1.6, ael and goto. jump fiexd them |
| 12:43.14 | *** join/#asterisk Sipster (~Sipster@modemcable045.5-200-24.mc.videotron.ca) |
| 12:43.36 | cusco | ok.. well its working now, I rather not fiddeling with it no more... |
| 12:45.00 | *** part/#asterisk wubba (~kmurrey@24.140.169.52) |
| 12:46.54 | *** join/#asterisk Sipster_ (~Sipster@modemcable045.5-200-24.mc.videotron.ca) |
| 12:47.01 | *** join/#asterisk E-bola (~bola@smtp.techbiz.dk) |
| 12:47.23 | E-bola | Have anybody seen that sometimes when u reload, it takes longer for asterisk to pickup all the settings? I've seen it on both 1.4 and 1.6 |
| 12:47.42 | E-bola | It doesnt show all the peers in sip show peers for example, but then after 10-30 secs they suddenly all get "registered" |
| 12:48.37 | manopulus | question. 2nd time. but now are some activity here. for some reason i want to mark sip calls with custome header. why this code won't work? SIPAddHeader(X-blah: a=123\;b=111); NoOp(${SIP_HEADER(X-blah)}/${SIP_HEADER(blah)}); i receiving nothing on asterisk. but after that, when call goes out to kamailio, at kamailio i have X-blah, can read it. |
| 12:48.41 | *** join/#asterisk wonderworld (~ww@ip-62-143-22-226.unitymediagroup.de) |
| 12:48.45 | Gido-E | E-bola ,/me not |
| 12:49.29 | E-bola | Its anoying and weird :( |
| 12:49.51 | Gido-E | E-bola normal CENTOS installations? |
| 12:50.03 | E-bola | Gido-E: No, all Debian |
| 12:50.14 | Gido-E | ok, i dont have anny server on Debian |
| 12:55.02 | saftsack | no t.38 experts here? |
| 12:55.16 | *** join/#asterisk jnicola (~jnicola@186.124.123.2) |
| 12:55.20 | jnicola | hello all |
| 12:55.30 | jnicola | i have incoming calls from "asterisk" |
| 12:55.37 | [TK]D-Fender | manopulus: Because it doesn't add it to the CURRENT call, it WILL add it to the next call you PLACE off of that active channel |
| 12:55.43 | jnicola | when answer.... any speaker is there.. |
| 12:56.00 | [TK]D-Fender | jnicola: Means no caller ID came over |
| 12:57.35 | jnicola | how can i fix that.. |
| 12:57.36 | jnicola | ? |
| 12:58.22 | manopulus | [TK]D-Fender: i see. so, for curent call it is better to set variable prefixed with __ (then such variable goes to all dauther calls) ? (i want to send call source and DB details over cluster of asterisks with single kamailio) |
| 12:59.06 | [TK]D-Fender | manopulus: That's one way to track. |
| 13:04.53 | *** join/#asterisk MhaddogM1 (~MhaddogM1@adsl-32-172-182.mia.bellsouth.net) |
| 13:07.36 | *** join/#asterisk The-Bat (~The-Bat@59.162.86.164) |
| 13:08.30 | *** join/#asterisk ming_zym (~ming_zym@124.14.85.6) |
| 13:08.46 | *** join/#asterisk brad_mssw (~brad@shop.monetra.com) |
| 13:10.19 | *** join/#asterisk muiro (~muiro@unaffiliated/muiro) |
| 13:11.37 | *** join/#asterisk MiserySoft (~LND@94.197.120.72.threembb.co.uk) |
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| 13:14.10 | *** part/#asterisk ming_zym (~ming_zym@124.14.85.6) |
| 13:14.41 | *** join/#asterisk Sipster (~Sipster@modemcable045.5-200-24.mc.videotron.ca) |
| 13:14.55 | *** join/#asterisk ming_zym (~ming_zym@124.14.85.6) |
| 13:15.09 | E-bola | its just odd, the first minute or so after asterisk is started, it doesnt respond to reload |
| 13:15.23 | *** join/#asterisk coppice (~chatzilla@153.166.232.220.dyn.pacific.net.hk) |
| 13:17.11 | *** join/#asterisk kruemeltee (~Maddin@port-92-198-62-82.static.qsc.de) |
| 13:17.40 | Gnarfy | This might be a silly question... but why does Playback(beep) in a 40 sec loop keep an otherwise silent call open, but rtpkeepalive=15 not? |
| 13:18.05 | *** join/#asterisk eliel (~eliels@200.43.6.244) |
| 13:18.21 | Gido-E | silent cal open? |
| 13:18.33 | Gido-E | there is always a rtp stream |
| 13:19.03 | Gnarfy | well, there is no audio going over the call... i see all rtp traffic going, but the provider delivering the calls to us, cut off the call after 60 secs of silence |
| 13:21.31 | *** join/#asterisk sysreq (~sysreq@unaffiliated/sysreq) |
| 13:21.46 | *** join/#asterisk pabelanger (~pabelange@nat/digium/x-hkrujuaqislfatro) |
| 13:22.16 | *** part/#asterisk muiro (~muiro@unaffiliated/muiro) |
| 13:22.24 | *** join/#asterisk vbman2 (~JZell2@24.229.51.108.res-cmts.sm.ptd.net) |
| 13:22.33 | vbman2 | how does POE work? |
| 13:23.22 | Katty | usually...you have a POE switch |
| 13:23.31 | Katty | each port in the switch, pipes power down the cat5 |
| 13:23.38 | Katty | the phone then plugs into the port in the wall, and gets power |
| 13:23.44 | Katty | much like the ole pots line phones |
| 13:23.55 | vbman2 | a poe switch? |
| 13:24.05 | Katty | yep. looks just like a regular switch. |
| 13:24.09 | Katty | cept it's a lottttt more expensive |
| 13:24.16 | vbman2 | so i plug the switch into the wall and thats it |
| 13:24.17 | Katty | some polycom phones will do POE without needing a poe switch |
| 13:24.24 | vbman2 | it distrib power to the phones? |
| 13:24.29 | Katty | there's a little ac adaptor which pipes power directly into the cat5 cable |
| 13:24.29 | [TK]D-Fender | vbman2: yes |
| 13:24.44 | vbman2 | im doing a office install and i never used poe before but i installed asterisk many times |
| 13:24.49 | Katty | without running the power through the entire network |
| 13:24.49 | vbman2 | and he dosnt care how much |
| 13:24.50 | vbman2 | lol |
| 13:24.56 | vbman2 | i got a good deal |
| 13:25.00 | vbman2 | Cisco phones for $50 |
| 13:25.00 | [TK]D-Fender | Katty: that isn't poe |
| 13:25.06 | Katty | it's like poe. |
| 13:25.08 | Katty | so simmer down |
| 13:25.12 | *** join/#asterisk bn-7bc (bjarne@lan-gw.noare-1.holmedal.net) |
| 13:25.16 | vbman2 | thanks |
| 13:25.28 | [TK]D-Fender | Katty: the IP 50X/30X "PoE" was faked by the dongle wire |
| 13:25.28 | Katty | poe switches are expensive. |
| 13:25.30 | Katty | very. expensive. |
| 13:25.32 | Katty | [TK]D-Fender: hush. |
| 13:25.35 | *** join/#asterisk blaines (~blaines@ip70-190-67-126.ph.ph.cox.net) |
| 13:26.05 | [TK]D-Fender | vbman2: Be VERY careful. Older cisco phones used CISCO stand PoE, and NOT the 802.3af Standard that just about everyone else uses |
| 13:26.06 | Katty | crittercam is on today |
| 13:26.49 | vbman2 | its model Cisco CP-7912G amn i safe? |
| 13:27.08 | Katty | jeebus. i think my windows need washing |
| 13:28.08 | Katty | http://www.ustream.tv/channel/squirrel-critter-cam <- shameless promotion |
| 13:28.47 | vbman2 | [TK]D-Fender: is that phone ok? |
| 13:28.56 | xheliox | lol - you still have the critter cam? |
| 13:29.13 | Katty | of course i do. |
| 13:29.17 | Katty | why wouldn't i? |
| 13:29.33 | Katty | you think my boys stop eating in the summer or somethin? :P |
| 13:29.48 | *** join/#asterisk pa (~pa@unaffiliated/pa) |
| 13:30.01 | *** join/#asterisk verywiseman (~khaled@unaffiliated/verywiseman) |
| 13:30.02 | xheliox | nah, I just hadn't been around in awhile |
| 13:30.06 | xheliox | forgot all about it |
| 13:30.25 | vbman2 | can i plug in the asterisk box into a poe switch/ |
| 13:30.33 | vbman2 | it wont give power to the nic right? |
| 13:30.44 | vbman2 | only if the nic requests power |
| 13:30.46 | Katty | xheliox: ahh. |
| 13:31.01 | Katty | xheliox: well the front yard does look a bit greener these days. no more dead looking trees and snow on the ground. |
| 13:32.02 | *** join/#asterisk jpeeler (~jpeeler@asterisk/digium-software-dev/jpeeler) |
| 13:32.23 | [TK]D-Fender | vbman2: http://www.velocityreviews.com/forums/t363262-cisco-7912-ip-phone-poe-woes.html |
| 13:32.25 | [TK]D-Fender | vbman2: NO |
| 13:33.13 | [TK]D-Fender | vbman2: There is a reason the phone is cheap... because the only switches that will power it are nasty expensive |
| 13:39.24 | *** join/#asterisk Poincare (~jefffnode@213.219.184.23.wls.msr03bkc3.adsl.static.edpnet.net) |
| 13:43.29 | *** join/#asterisk tempspace (~tempspace@cloudtrainer.com) |
| 13:46.52 | *** join/#asterisk tempspace (~tempspace@173-203-94-204.static.cloud-ips.com) |
| 13:48.31 | *** join/#asterisk mikkel (~mikkel@84-238-113-66.u.parknet.dk) |
| 13:52.40 | *** join/#asterisk sysop1 (~gerrym@gw.futuremetals.com) |
| 13:53.16 | *** join/#asterisk arturio_ (~c313ccda@gateway/web/freenode/x-chzrlgzzoaothfyz) |
| 13:54.46 | arturio_ | php-api. The clling is started. How to check does the second user hangup phone? |
| 13:55.18 | *** join/#asterisk mikkel (~mikkel@84-238-113-66.u.parknet.dk) |
| 13:55.30 | arturio_ | I want to send notfication e-mail to the second person if the phone was buisy |
| 13:56.21 | *** join/#asterisk romb (~romb@unaffiliated/romb-work/x-7222485) |
| 14:03.14 | *** join/#asterisk RobH (~robh@wikimedia/RobH) |
| 14:03.52 | Wimme | can someone please take a look at: http://pastebin.com/XA4J4fkT im having problems with a bri card |
| 14:04.23 | Wimme | i have a 4port bri card with 3 idsn lines connected to it, but i cant make more than 2 simultanious calls... |
| 14:08.07 | *** join/#asterisk nighty^ (~nighty@x122091.ppp.asahi-net.or.jp) |
| 14:08.29 | *** join/#asterisk jmacz (~jmacz@190.144.75.22) |
| 14:08.49 | *** join/#asterisk ariel_ (~chatzilla@63.214.236.169) |
| 14:09.43 | *** join/#asterisk mv2 (~maverick@83.240.229.38) |
| 14:10.34 | mv2 | i'm trying to play a sound in asterisk but i don't hear nothing ... i can see in log asterisk saying "playing sound X" |
| 14:10.55 | mv2 | with tcpdump i only see to rtp packets from asterisk to phone |
| 14:11.06 | mv2 | already checked the sound and it's ok |
| 14:11.10 | mv2 | any ideias ? |
| 14:11.33 | mv2 | asterisk 1.2.29 |
| 14:13.30 | *** join/#asterisk UQlev (~yuriy@212.50.99.8) |
| 14:13.48 | pabelanger | ~book |
| 14:13.48 | 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 |
| 14:13.57 | pabelanger | mv2: ^^^ start there. |
| 14:14.12 | pabelanger | mv2: We need way for information about your network setup. |
| 14:14.27 | pabelanger | s/for/more/g |
| 14:14.36 | pabelanger | hehe |
| 14:14.52 | *** join/#asterisk eliel_ (~eliels@201.234.94.226) |
| 14:16.47 | *** join/#asterisk wubba (~chatzilla@cable-74-53.sssnet.com) |
| 14:16.58 | mv2 | my network setup ? |
| 14:17.05 | mv2 | for what ? |
| 14:17.12 | mv2 | it's not a network issue |
| 14:17.26 | mv2 | i can hear sound in normal calls |
| 14:17.28 | wubba | I need some help with a call file. I dropped one into the spool directory and it's just sitting there. made sure it was owned by asterisk, and made sure the spool module was loaded |
| 14:17.45 | mv2 | so it's not a fw/ip/omq issue |
| 14:18.42 | mv2 | if i'm saying that I only see two rtp packet exiting asterisk box ... |
| 14:18.50 | mv2 | do you think it's a network issue ? |
| 14:19.06 | mv2 | if i can hear sound in regular calls |
| 14:19.12 | mv2 | do you still******** think |
| 14:19.14 | mv2 | that ? |
| 14:19.45 | Gido-E | wubba you copy it, or touched it? |
| 14:19.47 | pabelanger | mv2: yes: it is mostly a network issue. Are you doing NAT for example? |
| 14:19.53 | mv2 | no |
| 14:20.00 | wubba | gido-e did a mv |
| 14:20.14 | pabelanger | mv2: Are you answering the channel or early media? |
| 14:20.17 | Gido-E | ok, than touch the file |
| 14:20.23 | Gido-E | cd /var/spool/bla |
| 14:20.31 | pabelanger | ~collectdebug |
| 14:20.31 | infobot | collectdebug is, like, a method of collecting logs allowing others help troubleshoot an issue. Refer to doc/HOWTO_collect_debug_information.txt or http://svn.digium.com/svn/asterisk/trunk/doc/HOWTO_collect_debug_information.txt |
| 14:20.31 | Gido-E | touch callfile.call |
| 14:20.42 | pabelanger | mv2: ^^^ |
| 14:20.53 | mv2 | answering |
| 14:21.01 | *** join/#asterisk moy (~moy@bas1-unionville55-1177733627.dsl.bell.ca) |
| 14:21.42 | pabelanger | mv2: pastebin your output |
| 14:22.41 | wubba | Gido-E: ok touched the file - still sitting there |
| 14:23.55 | pabelanger | wubba: *CLI> module show like spool |
| 14:24.07 | pabelanger | is it loaded? |
| 14:24.30 | wubba | yep |
| 14:24.33 | wubba | it's loaded |
| 14:24.35 | *** join/#asterisk Defraz (~Defraz@c72co-edge-router.fuzecore.com) |
| 14:24.51 | m_tadeu | to use an adsl line I need a T1/E1 interface? |
| 14:26.17 | [TK]D-Fender | m_tadeu: To use a car, do you need a bicycle? |
| 14:26.45 | Gido-E | wubba ok, touch the file outside the outgoing dir, and then mv it to the outgoing dir |
| 14:26.48 | [TK]D-Fender | m_tadeu: Do I need a VCR to play a DVD disc? |
| 14:28.01 | Gido-E | [TK]D-Fender yes |
| 14:29.26 | wubba | OK - I touched it out side the directory mv it and it just sitts there |
| 14:29.58 | Gido-E | wubba than something impossible is happening :-) |
| 14:30.06 | m_tadeu | [TK]D-Fender: sorry I'm so ignorant on telecom stuff...just not my area |
| 14:30.27 | [TK]D-Fender | m_tadeu: clarify what you mean by "use an ADSL line" |
| 14:30.30 | Gido-E | <PROTECTED> |
| 14:30.35 | wubba | Gido-E: Yes |
| 14:31.00 | *** join/#asterisk [intra]lanman (~lanman@freeswitch/developer/intralanman) |
| 14:31.06 | wubba | -rw-r--r-- 1 asterisk asterisk 20 May 28 10:27 try.call |
| 14:31.42 | Gido-E | wubba, you can check on the shell: Change owner to Asterisk, and read a file from that location on the shell |
| 14:31.58 | m_tadeu | [TK]D-Fender: to use the voice from my home adsl in asterisk...I'd like to do some exepriments ;) |
| 14:32.18 | [TK]D-Fender | m_tadeu: How does "voice" comes your ADSL link? |
| 14:33.17 | wubba | Gido-E: not sure what you mean by that |
| 14:33.32 | m_tadeu | [TK]D-Fender: have a filter that separates the voice frequencies from the data frequencies |
| 14:33.49 | Gido-E | wubba on the shell: su - asterisk and read that file from /var/spool/asterisk/outgoing |
| 14:33.56 | [TK]D-Fender | m_tadeu: then that would be "use my ANALOG LINE which just happens to also have ADSL running over it" |
| 14:34.08 | [TK]D-Fender | m_tadeu: ADSL is not a pertinent factor in what you are doing with it |
| 14:34.17 | [TK]D-Fender | m_tadeu: You want to use your Joe Blow home line. |
| 14:34.28 | [TK]D-Fender | m_tadeu: that will take an analog FXO interface. |
| 14:34.50 | [TK]D-Fender | m_tadeu: So... before you go and spend actual money on this... what are your short & longer-term goals with this? |
| 14:35.24 | m_tadeu | [TK]D-Fender: researching for a small call center |
| 14:35.47 | [TK]D-Fender | m_tadeu: How would that call center be set up when it comes time? |
| 14:36.00 | [TK]D-Fender | m_tadeu: What exactly would they be using for PSTN access? |
| 14:37.40 | *** part/#asterisk sysop1 (~gerrym@gw.futuremetals.com) |
| 14:37.43 | m_tadeu | [TK]D-Fender: I'm still trying to figure that out with the operators...some details like the number of lines, etc are still not decided |
| 14:38.05 | tempspace | Does anybody know of an issue with Caller ID HTTP queries not parsing [NUMBER] correctly? |
| 14:38.09 | *** part/#asterisk elende (~LND@94.197.120.72.threembb.co.uk) |
| 14:38.09 | m_tadeu | [TK]D-Fender: I'll need to have 1 number with several lines associates |
| 14:38.21 | [TK]D-Fender | m_tadeu: And a question of what KIND of link. Analog lines, digital lines, ITSP, etc. |
| 14:38.36 | [TK]D-Fender | m_tadeu: Analog isn't a really nice option... |
| 14:39.33 | m_tadeu | [TK]D-Fender I'm having a meeting with an operators' salesman...which kind of link you advise? |
| 14:39.44 | *** join/#asterisk romb (~romb@unaffiliated/romb-work/x-7222485) |
| 14:39.47 | wubba | Gido-E: Well it won't let me copy into that directory when I go to asterisk |
| 14:41.05 | *** join/#asterisk methodvon (~methodvon@pool-71-191-177-146.washdc.fios.verizon.net) |
| 14:42.32 | *** join/#asterisk flapjacks (~flapjacks@wsip-72-214-208-206.ph.ph.cox.net) |
| 14:44.41 | Gido-E | wubba ok, if you cant do it as that user, you got to set the rights as needed |
| 14:44.55 | flapjacks | is there a way to limit the number of calls on a group of extensions? For example say I have 20 extensions I only want no more then 16 on the phone at one time. Is there a way to do that |
| 14:45.30 | Gido-E | flapjacks groupcount |
| 14:45.41 | flapjacks | thx |
| 14:47.44 | *** join/#asterisk freckle (~Jon@195.74.96.118) |
| 14:48.47 | [TK]D-Fender | m_tadeu: depends where you are, what the costs are, what your exact needs are, what your security requirements are. |
| 14:48.52 | wubba | Gido-E: I would chown the file and then move it - and it still just sits there. |
| 14:48.57 | wubba | even as root |
| 14:48.59 | wubba | just sits |
| 14:49.01 | wubba | there |
| 14:50.04 | Gido-E | ok, asterisk runs under? |
| 14:50.15 | Gido-E | that user reads and removes the callfile |
| 14:50.40 | wubba | Gido-E: if the context of the call file is wrong - it would still read it correct? |
| 14:51.34 | m_tadeu | [TK]D-Fender: still don't know about the costs...I'll only know that on monday...but I'd like to ask about viable links...the project is a small call center with 10 to 20 people |
| 14:52.00 | m_tadeu | [TK]D-Fender: I'm in portugal, btw |
| 14:52.04 | [TK]D-Fender | m_tadeu: people does not imply how many PSTN channels you'll need/want |
| 14:52.16 | *** join/#asterisk rocksfrow (~kyle@74-92-146-97-WashingtonDC.hfc.comcastbusiness.net) |
| 14:52.21 | [TK]D-Fender | m_tadeu: I've seen acll centers with 100 channel for 10 people. |
| 14:52.31 | [TK]D-Fender | m_tadeu: and 3 channels for the same for others |
| 14:53.21 | m_tadeu | [TK]D-Fender: I'm figuring 70% of the colaborators busy in peak hours |
| 14:53.44 | [TK]D-Fender | m_tadeu: Better research the costs in your area then. |
| 14:54.20 | *** join/#asterisk KavanS (~KavanS@static-173-50-141-22.ptldor.fios.verizon.net) |
| 14:54.45 | *** join/#asterisk CrimsonX (~ccc1490a@gateway/web/freenode/x-asezpugdyzoxvhkt) |
| 14:54.48 | m_tadeu | [TK]D-Fender: ok...I'll have more info on monday, for sure |
| 14:56.36 | *** join/#asterisk anthm (~anthm@freeswitch/developer/anthm) |
| 14:56.55 | wubba | hey is there anyway using a softphone that I could send *73 to a zap line? |
| 14:58.03 | [TK]D-Fender | wubba: ... its called "placing a call" |
| 14:58.14 | [TK]D-Fender | wubba: Why can't THEY do it themselves then? |
| 14:58.33 | [TK]D-Fender | wubba: What the problem with the actual phones they have on-site? |
| 14:58.50 | wubba | cause they are freaking retarded and I am an hour away. The analog phone we put in for them to do this doesn't have any dial tone now. All of their phones are IP |
| 14:59.20 | [TK]D-Fender | wubba: who needs an analog phone? Its connected to the PBX. Have them dial the *#&$ing number |
| 14:59.21 | wubba | So last week they took the analog phone and forwarded their lines |
| 15:00.02 | KavanS | how would one determine why asterisk is taking 100% cpu with no calls? |
| 15:00.03 | wubba | we put the analog phone so they could forward 'before' the pbx |
| 15:00.34 | wubba | what is that? Have them dial the *#&$ing number |
| 15:00.46 | *** part/#asterisk moy (~moy@bas1-unionville55-1177733627.dsl.bell.ca) |
| 15:00.47 | wubba | I need to dial *73 |
| 15:00.55 | wubba | but that sends it to asterisk |
| 15:00.58 | KavanS | wubba, just put it in the dial command then |
| 15:01.00 | *** join/#asterisk CunningPike (~CunningPi@S01060014bf81366b.vc.shawcable.net) |
| 15:01.11 | [TK]D-Fender | wubba: They GET A CLUE and set up a pattern to dial that OUT |
| 15:01.40 | *** join/#asterisk moy (~moy@bas1-unionville55-1177733627.dsl.bell.ca) |
| 15:02.06 | [TK]D-Fender | wubba: **73 <- have this send *73 to your Zap channel |
| 15:02.11 | [TK]D-Fender | wubba: or WHATEVER |
| 15:03.04 | Katty | or |
| 15:03.06 | Katty | your mom. |
| 15:03.24 | KavanS | wubba, if you don't add that to your dial command right now - the god's of asterisk will be unhappy! |
| 15:03.41 | [TK]D-Fender | KavanS: FreePBX <- |
| 15:03.55 | KavanS | lol just kidding |
| 15:04.03 | KavanS | needs more coffee |
| 15:05.49 | *** join/#asterisk emora (~emora@213.236.9.114) |
| 15:08.19 | *** join/#asterisk Linuturk (~linuturk@unaffiliated/linuturk) |
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| 15:14.05 | *** join/#asterisk Entulho (~foo@sol.sj.ifsc.edu.br) |
| 15:20.51 | Infin1ty | there's a bug in the asterisk centos/rh repo, asterisk14 conflict with 16 |
| 15:22.16 | Qwell | Infin1ty: that isn't a bug |
| 15:22.52 | Infin1ty | Qwell, it's a "bug" , it should detect that if version 14 is not installed than 16 should be, in the spec file |
| 15:23.10 | Qwell | Package managers are dumb. It can't "detect" anything. |
| 15:24.01 | Qwell | Having said that.. you get me a spec diff for asterisk and asterisk-addons that makes that work, and I'll add it |
| 15:24.44 | Infin1ty | oh well, it looks like it has conflict with the epel repo, yum --disablerepo=epel solve it |
| 15:25.02 | Infin1ty | perhaps it will be nice if it could be added to the manual http://www.asterisk.org/downloads/yum |
| 15:25.03 | Qwell | yes, they are stepping all over our toes. don't use epel |
| 15:25.06 | *** join/#asterisk romb (~romb@unaffiliated/romb-work/x-7222485) |
| 15:25.15 | Qwell | EPEL != CentOS. We support CentOS. |
| 15:25.20 | Infin1ty | Qwell, I know :) |
| 15:29.05 | *** join/#asterisk lirakis (~lirakis@static-173-210-1-18.ngn.onecommunications.net) |
| 15:30.52 | ChannelZ | TGIF |
| 15:31.11 | p3nguin | Whoa! It's Friday? Already? |
| 15:31.37 | p3nguin | That's messed up; I seriously thought it was only Thursday. |
| 15:31.47 | KavanS | heck yeah |
| 15:31.51 | ChannelZ | Blessing and a curse really. I'm glad it's the end of the week but I really need a few more days. Working through the weekend again for me. Bleh. |
| 15:33.01 | *** join/#asterisk Zeeek (~anonymous@pdpc/supporter/active/zeeek) |
| 15:33.38 | Zeeek | here we are, Friday and it's time to talk VoIP on the live conference starting in about 15 minutes on #vuc |
| 15:33.39 | p3nguin | Good day, Zeeek. |
| 15:34.05 | Zeeek | Today's guest is Ward Mundy of Nerd Vittles talking about the Asterisk-based Incredible PBX |
| 15:34.06 | p3nguin | What's on the agenda for the conf today? |
| 15:34.10 | Zeeek | hi p3nguin |
| 15:34.13 | Zeeek | Today's guest is Ward Mundy of Nerd Vittles talking about the Asterisk-based Incredible PBX |
| 15:34.17 | KavanS | oh shit Zeeek |
| 15:34.24 | Zeeek | no that was this morning |
| 15:34.29 | p3nguin | haha |
| 15:34.46 | p3nguin | Right after a second cup of coffee, I assume. |
| 15:35.00 | Zeeek | jump on #vuc and phone it in at 200901@login.zipdx.com in about 15 min |
| 15:35.12 | Zeeek | p3nguin: you are a mind reader! |
| 15:35.41 | Zeeek | so now that we have that out of the way |
| 15:35.44 | KavanS | nice |
| 15:35.45 | p3nguin | Aww, I need to go cut the grass before it heats up or rains. I probably will miss most of the important part. |
| 15:35.49 | Zeeek | (until tomorrow morning) |
| 15:35.55 | KavanS | p3nguin, what do you do for a living? |
| 15:36.04 | Zeeek | but we hang out on the conferenc efor hours after |
| 15:36.11 | Zeeek | come on by later |
| 15:36.15 | KavanS | do you work from remote? |
| 15:36.30 | p3nguin | kavans: I'm a SysEng for a small company in a rural area. |
| 15:36.33 | Zeeek | if you click on my remote, you can change my volume |
| 15:36.42 | KavanS | oh cool |
| 15:36.55 | *** join/#asterisk hugorebelo (~hugo@200-171-132-124.completo.com.br) |
| 15:36.59 | jnicola | hello... . i have a problem.... i call from 200 to 201... phones are registred to the same asterisk... the scenary is 200 --> asterisk -->billing proxy --> asterisk ---> 201 |
| 15:37.00 | Zeeek | http://vuc.me It's Friday ! |
| 15:37.12 | KavanS | Zeeek, I'm going to call in a bit |
| 15:37.14 | jnicola | call is ok.. |
| 15:37.21 | KavanS | and it had better sound crystal clear! |
| 15:37.21 | Zeeek | Testing Skype for SIP beta today, works well. As does Skype for Asterisk |
| 15:37.37 | Zeeek | KavanS: if you use g722 it will |
| 15:37.41 | p3nguin | I work remotely, on-site, or sometimes not at all. |
| 15:38.38 | *** join/#asterisk smooth_penguin (~smoove@59.95.27.135) |
| 15:39.00 | *** join/#asterisk sat-man (~jlupresto@c-174-52-20-94.hsd1.ut.comcast.net) |
| 15:39.30 | KavanS | Zeeek, I dialed in and hear no sound |
| 15:39.39 | KavanS | Zeeek, using the phone number listed on the website |
| 15:39.46 | p3nguin | 200901@login.zipdx.com |
| 15:39.54 | Zeeek | Everyone join us for the VoIP Users Confernece in about 15 minutes. ON the hour, Incredible PBX: http://nerdvittles.com/?p=677 |
| 15:40.02 | Zeeek | p3the conferenc ebridge is not up yet |
| 15:40.10 | p3nguin | :/ |
| 15:40.48 | jnicola | i cant get the 200 ok, neither bye message in the proxy... |
| 15:41.13 | KavanS | cool, sip is working |
| 15:41.32 | KavanS | p3nguin, do you have this sip address stored in asterisk somewhere, or some other method of dialing? |
| 15:42.00 | p3nguin | Yes, I have an exten for it. |
| 15:42.59 | *** join/#asterisk Yon (Yon@c80-217-244-16.bredband.comhem.se) |
| 15:43.06 | p3nguin | exten 882 (vuc) |
| 15:43.32 | p3nguin | I wonder if I can get zipdx to call me this time. |
| 15:44.12 | KavanS | nice |
| 15:46.47 | Zeeek | SIP: 7463#22622#1@proxy.ideasip.com |
| 15:47.37 | Zeeek | THat will get you to the Talkshoe g711 bridge |
| 15:48.38 | Zeeek | see you on #vuc |
| 15:48.39 | *** part/#asterisk Zeeek (~anonymous@pdpc/supporter/active/zeeek) |
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| 16:03.16 | *** mode/#asterisk [+o jtodd] by ChanServ |
| 16:10.49 | *** join/#asterisk cusco (~trilili@213.63.137.210) |
| 16:10.53 | cusco | hi |
| 16:11.20 | cusco | a digium cart te120p should have incomming pri with 120ohm or 75ohm ?? |
| 16:12.51 | coppice | 120 ohms. 75 ohms is normally something you only meet with coaxial cables |
| 16:13.44 | cusco | would the card couple with it tho? |
| 16:14.14 | coppice | couple with what? |
| 16:14.51 | cusco | with 120ohms |
| 16:15.06 | cusco | I guess using a converter from those "bnc" tips to rj45 |
| 16:15.12 | cusco | not sure |
| 16:16.15 | coppice | if you have coaxial cables coming in, you will need one of those little gadgets to convert to 120 ohm twisted pairs to attach to the card |
| 16:17.41 | cusco | yea ok... |
| 16:20.49 | *** join/#asterisk cesar_CR (~cesar@201.192.86.30) |
| 16:25.09 | *** join/#asterisk verywiseman (~khaled@unaffiliated/verywiseman) |
| 16:26.07 | cusco | another non related question... |
| 16:26.17 | cusco | is there soemthing wrong with: Read(Contacto,audio/Sapo/naofoipossivel,9,,3,8); |
| 16:26.56 | cusco | Contacto is the var, audio/Sapo/naofoipossivel is the audio file, 9 digits (mobile number), no options, 3 attempts, 8secs |
| 16:33.10 | *** join/#asterisk smooth_penguin (~smoove@59.95.17.9) |
| 16:40.29 | pabelanger | cusco: core show application Read |
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| 16:43.22 | *** join/#asterisk |amadeus| (~amadeus@75.130.147.152) |
| 16:48.04 | *** join/#asterisk wcselby (~wcselby@208.180.112.123) |
| 16:51.45 | wcselby | oh yeah |
| 16:51.47 | wcselby | o/ |
| 16:54.14 | *** join/#asterisk x303 (~X303@187.159.121.70.cfl.res.rr.com) |
| 17:00.02 | pabelanger | infobot hi |
| 17:00.03 | infobot | hello, pabelanger |
| 17:00.08 | cusco | what is this t10i piece that comes with the Digium card? |
| 17:00.50 | pabelanger | cusco: loopback adapter |
| 17:00.57 | cusco | err.. what do I do with it? |
| 17:01.01 | cusco | where do I plug it? |
| 17:01.14 | pabelanger | cusco: if you want to loopback your interface |
| 17:01.21 | pabelanger | cusco: usually for testing |
| 17:01.46 | cusco | can I plug it on the PRI card? or where? |
| 17:01.52 | cusco | or nowhere? |
| 17:01.59 | cusco | just plug the cable in? |
| 17:02.11 | cusco | then I have no interface to perform test |
| 17:02.28 | pabelanger | cusco: no, you plug one end of your RJ-45 cable into your card, and the other end of the RJ-45 to the t10i |
| 17:02.40 | pabelanger | cusco: it then loops the circuit |
| 17:05.52 | cusco | ahh |
| 17:06.02 | cusco | loopback for the pri card, not for the pri line |
| 17:06.23 | cusco | ok |
| 17:06.46 | cusco | not needed, we have more pri lines than pri slots |
| 17:06.47 | cusco | :P |
| 17:06.54 | cusco | (cards are expensive as hell) |
| 17:17.53 | *** join/#asterisk jaytee (~jforde051@unaffiliated/jaytee) |
| 17:30.31 | *** part/#asterisk jaytee (~jforde051@unaffiliated/jaytee) |
| 17:33.08 | cusco | hio |
| 17:33.21 | cusco | one more question... |
| 17:33.25 | cusco | http://paste.debian.net/75211/ dahdi errors |
| 17:33.41 | cusco | like PRI got event: HDLC Abort (6) on Primary D-channel of span 1 |
| 17:33.48 | cusco | or PRI got event: HDLC Bad FCS (8) on Primary D-channel of span 1 |
| 17:33.51 | cusco | what to these mean? |
| 17:34.24 | *** join/#asterisk smooth_penguin (~smoove@59.95.24.162) |
| 17:39.07 | *** join/#asterisk Jinxed- (~93b12a95@gateway/web/freenode/x-xcnsjtvcobsrerjc) |
| 17:39.43 | Jinxed- | Does anyone know why when you plug in a polycom 335 when you start dialing it says "sip: " |
| 17:39.53 | Jinxed- | enter url |
| 17:40.25 | [TK]D-Fender | jinkBecause you left URI dial enabled, or it failed to register so URI is its only option |
| 17:42.01 | Naikrovek | you don't have it configured properly |
| 17:42.25 | Jinxed- | [TK]D-Fender: where is the url dial enabled option at |
| 17:42.41 | Naikrovek | you're in over your head a bit if you don't know where to look |
| 17:42.50 | Naikrovek | no offense intended |
| 17:42.57 | Naikrovek | take some time to read the Polycom Admin Guide |
| 17:43.12 | Naikrovek | or at least skim it over while you use the toilet nex |
| 17:43.13 | Naikrovek | t |
| 17:43.31 | Jinxed- | Naikrovek: just waiting for the 5 min restart |
| 17:43.36 | Jinxed- | at the moment |
| 17:43.41 | Naikrovek | what 5 min restart |
| 17:43.46 | wcselby | Naikrovek - haha |
| 17:43.55 | wcselby | @ the toilet comment |
| 17:44.15 | Jinxed- | Naikrovek: the polycom phone takes forever to restart |
| 17:44.25 | wcselby | Jinxed- - i get that sometimes as well |
| 17:44.26 | Naikrovek | hmm |
| 17:44.32 | Jinxed- | grandstream and cisco phones take a 1/8th the time |
| 17:44.41 | Naikrovek | my polycoms take about 30 seconds |
| 17:44.43 | wcselby | Jinxed- - it sits at the screen where it's loading the new config or something, right? |
| 17:44.52 | Nugget | can you at least remotely reboot a polycom? that's still not possible with a cisco |
| 17:45.01 | Naikrovek | yes you can remote boot polycom |
| 17:45.47 | *** join/#asterisk vivekp (~vivekp@117.99.73.86) |
| 17:45.49 | wcselby | i just completely reformated, loaded new firmware and provisioned a new 650 in less than 5 minutes |
| 17:46.02 | Naikrovek | yeah |
| 17:46.13 | Naikrovek | the first one is always a bit rough though |
| 17:46.24 | Naikrovek | took me like a day to configure my first |
| 17:46.29 | [TK]D-Fender | wcselby: From box to desk in 3 minutes flat on my side. No scripts or anything on my side |
| 17:46.34 | wcselby | but i worked with some 550's and 330's (or 335, not sure) last week that took five minutes a pop, at least. and then the time was still off... |
| 17:46.47 | Naikrovek | have configured several dozen since then and i let the user unpack them and plug them in |
| 17:47.03 | Naikrovek | They Just Work (tm) |
| 17:47.14 | *** join/#asterisk lhz (~shrekz@c-dba672d5.021-158-73746f34.cust.bredbandsbolaget.se) |
| 17:47.51 | wcselby | Naikrovek - i'm not sure why this was happening on these particular phones, the only thing I did was change a setting in sip.cfg |
| 17:48.05 | Jinxed- | yeah it seems like you can do some nice auto config after awhile |
| 17:48.19 | Jinxed- | but right now im just trying to get one to register with my call manager |
| 17:48.56 | *** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/atheme.member.chainsaw) |
| 17:49.05 | [TK]D-Fender | Jinxed-: Call manager? Pardon? |
| 17:49.27 | Naikrovek | uh oh |
| 17:49.42 | drmessano | CCCP Call Manager? |
| 17:50.05 | [TK]D-Fender | drmessano: Foxconn produces those... right? ;) |
| 17:50.16 | drmessano | lol |
| 17:50.21 | [TK]D-Fender | drmessano: I know... I shouldn't "jump" to conclusions ;) |
| 17:50.38 | [TK]D-Fender | drmessano: And trust me... its concluded at the end |
| 17:50.41 | *** join/#asterisk dgd_ (~user@195.230.115.10) |
| 17:50.51 | *** join/#asterisk t_dot_zilla (~chatzilla@firewall-a.buf.ny.i-evolve.net) |
| 17:50.58 | t_dot_zilla | does anybody know how to configure the mp118 fxs ? |
| 17:51.29 | Jinxed- | restarting phone again |
| 17:51.38 | drmessano | Only PBX that plays "I must break you" from the internal speaker on boot |
| 17:52.29 | [TK]D-Fender | drmessano: If the phone dies.. it dies |
| 17:52.46 | drmessano | What it calls, it destroys |
| 17:52.52 | drmessano | Whatever it calls, it destroys |
| 17:53.23 | *** join/#asterisk jaskew (~jdaskew@netblock-75-79-182-110.dslextreme.com) |
| 17:57.38 | *** part/#asterisk jaskew (~jdaskew@netblock-75-79-182-110.dslextreme.com) |
| 17:58.12 | Jinxed- | I thought this phone would be the most straightforward |
| 17:58.16 | Jinxed- | to get working |
| 17:58.19 | Naikrovek | well |
| 17:58.26 | Naikrovek | once you understand how to do it, it is |
| 17:58.34 | Naikrovek | but there's a small learning curve right at the beginning |
| 17:58.39 | Naikrovek | (that you're about to overcome) |
| 17:58.48 | Naikrovek | so, how many phones, just this one? |
| 17:59.10 | Jinxed- | 26 of this type |
| 17:59.13 | Naikrovek | perfect |
| 17:59.14 | Jinxed- | well give or take |
| 17:59.17 | [TK]D-Fender | people think all sorts of things... with no background.. |
| 17:59.26 | Naikrovek | setting one up is exactly as difficult as setting 26 of them up |
| 17:59.37 | [TK]D-Fender | Jinxed-: Harsher starting curve, big payoff in the end |
| 17:59.37 | Naikrovek | so setting one up alone will seem like a lot of work. |
| 17:59.50 | Jinxed- | yeah, it's been my experience with other types |
| 17:59.54 | Jinxed- | of phones |
| 17:59.58 | Naikrovek | set up an FTP server somewhere, set up a login just for the phones |
| 18:00.19 | Naikrovek | the phone server (if you're using asterisk) is probably a good spot for an ftp server |
| 18:00.42 | Naikrovek | set up the ftp server only to listen on the LAN interface, not the WAN, if you have it configured that way |
| 18:01.56 | Naikrovek | step 2: http://www.polycom.com/support/voice/index.html select your phone then download the latest SIP firmware (SPLIT version) and bootrom for that phone model |
| 18:02.18 | Naikrovek | if they're all soundpoint IP phones, then one bootrom and one sip firmware download will cover you for all phones |
| 18:02.41 | Naikrovek | step 3: (anyone writing this down? this needs to go on a wiki or something) |
| 18:03.33 | Naikrovek | step 4: unzip the .zips you downloaded from polycom in to the home folder of the phone user you set up for FTP. the / directory when connecting via FTP should be where these files get unzipped to, so that when the phone user logs in, all the files are right there |
| 18:03.48 | Naikrovek | leave any directory structure in the zips alone, let it create folders if there are any (i forget) |
| 18:04.20 | Naikrovek | do you have a dhcp server? a real one, like on windows or linux? |
| 18:04.25 | Naikrovek | not linksys router dhcp |
| 18:05.07 | Jinxed- | yeah |
| 18:05.19 | Naikrovek | okay. is it windows or linux |
| 18:05.23 | Jinxed- | 2008 |
| 18:05.26 | Jinxed- | doz |
| 18:05.59 | Naikrovek | okay, in the dhcp scope, set up option 66 and give it the FTP url for the user you created on the FTP server. will look like ftp://user:pass@ip.or.host.name/ |
| 18:06.17 | Naikrovek | this will let the phones speak to the FTP server without any intervention from you |
| 18:06.19 | Naikrovek | or anyone else |
| 18:06.23 | Naikrovek | that was step 5 |
| 18:06.37 | Naikrovek | now the per-phone configuration begins |
| 18:06.47 | Naikrovek | i need to pastebin these files so i can refer to them later |
| 18:06.49 | Naikrovek | stand by |
| 18:07.50 | *** join/#asterisk githogori (~githogori@SJC-Office-DHCP-135.mail-abuse.org) |
| 18:10.06 | *** join/#asterisk mrgabu (~gbdurante@187.38.158.209) |
| 18:12.34 | *** join/#asterisk KavanS (~KavanS@static-173-50-141-22.ptldor.fios.verizon.net) |
| 18:13.37 | *** join/#asterisk retentiveboy (~pdugas@74-95-28-37-Atlanta.hfc.comcastbusiness.net) |
| 18:14.53 | Naikrovek | i have some scripts that will generate per-phone configs for you, based on a text file |
| 18:15.11 | *** join/#asterisk Coolrazor (~Tim@71.95.209.242) |
| 18:15.11 | Naikrovek | it's the most rudimentary provisioning system known, other than typing the configs out manually. |
| 18:15.13 | Naikrovek | http://pastie.org/982177 |
| 18:16.00 | *** join/#asterisk Sargun (~Sargun@atarack/Staff/Sargun) |
| 18:16.52 | Naikrovek | step 6a: edit phonelist.txt to reflect your actual phone MAC addresses, user extensions, names and SIP passwords. |
| 18:17.10 | Naikrovek | step 6b: run ./provisioner.pl to generate the phone configs |
| 18:17.25 | Naikrovek | step 7: boot a phone up and see what happens |
| 18:18.52 | Naikrovek | correction: http://pastie.org/982183 |
| 18:21.11 | *** join/#asterisk tripps (~sean@66.60.235.100) |
| 18:22.53 | *** join/#asterisk wonderworld (~ww@ip-62-143-22-226.unitymediagroup.de) |
| 18:23.27 | tripps | question - setting sip configs for 7960 phones using directives like dst_stop_week_of_month, dst_start_week_of_month, etc. In the US, DST begins the second Sunday of March and ends the first Sunday in November. Does dst_start_week_of_month mean the first week or the first week of the month that day falls on? |
| 18:25.08 | *** join/#asterisk mr_ian (~mr_ian@S0106001b63f49383.du.shawcable.net) |
| 18:29.41 | *** join/#asterisk Z_God (~julius@schwartzenberg.xs4all.nl) |
| 18:29.43 | *** join/#asterisk chilicuil (~chilicuil@unaffiliated/chilicuil) |
| 18:30.09 | *** join/#asterisk giesen (giesen@snickers.org) |
| 18:30.18 | giesen | What's the best way to debug musiconhold problems |
| 18:30.35 | giesen | I see nothing in the debug except that musiconhold started and then immediately stopped |
| 18:38.11 | *** join/#asterisk Woody2143 (~Woody2143@machine76.Level3.com) |
| 18:40.07 | *** join/#asterisk pabelanger (~pabelange@nat/digium/x-ncowicvjqqhdhovw) |
| 18:53.36 | *** join/#asterisk Beta2K (~Beta2K@d24-36-97-38.home1.cgocable.net) |
| 18:53.39 | Beta2K | Hello all |
| 18:54.04 | *** join/#asterisk Failrar (~Failrar@5ED66E6D.cable.ziggo.nl) |
| 18:54.32 | Beta2K | Will Asterisk terminate a T.38 fax call from a SIP device to a FXO port installed in the server? |
| 18:55.31 | *** join/#asterisk VultureZ (~chuck@74-93-168-241-Jacksonville.hfc.comcastbusiness.net) |
| 18:59.50 | *** join/#asterisk F_Madison (~Madison@c-71-226-162-205.hsd1.sc.comcast.net) |
| 19:01.38 | *** join/#asterisk Micc (~quassel@c-98-225-57-96.hsd1.wa.comcast.net) |
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| 19:09.47 | *** join/#asterisk Gnarfy (~ik@213.144.244.115) |
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| 19:21.07 | *** join/#asterisk t_dot_zilla (~chatzilla@firewall-a.buf.ny.i-evolve.net) |
| 19:21.12 | t_dot_zilla | hello |
| 19:21.22 | t_dot_zilla | has anybody set up the mp118 with asterisk ? |
| 19:21.32 | t_dot_zilla | i can make a call but no audio each way |
| 19:21.34 | t_dot_zilla | any ideas? |
| 19:22.07 | t_dot_zilla | http://pastebin.com/L5LFQxL1 |
| 19:23.01 | Beta2K | t_dot_zilla: behind a NAT? |
| 19:23.07 | t_dot_zilla | no |
| 19:23.24 | Beta2K | t_dot_zilla: my experience says when you have no audio, you have no RDP packets passing somewhere :) |
| 19:24.43 | t_dot_zilla | could you look at the PB |
| 19:28.42 | Qwell | looks at Deeewayne |
| 19:30.38 | Deeewayne | hey Qwell ! nice storm, huh? |
| 19:30.44 | Qwell | huh? |
| 19:31.11 | Deeewayne | my side of the mountain is rockin |
| 19:31.11 | Qwell | Deeewayne: I was going to leave a random Simpsons quote for you. Not related to any current event. |
| 19:31.24 | Qwell | What'chu talkin' 'bout, everyone! |
| 19:31.43 | Qwell | there are clouds over here... that's about it |
| 19:31.58 | t_dot_zilla | rtp is working |
| 19:33.24 | Deeewayne | Qwell, I like --> Bob: "your children are no more than a pair of ill-bred troublemakers", Homer: "Lisa too?", Bob: "especially Lisa, but especially bart" |
| 19:33.37 | Qwell | Deeewayne: heh, yes |
| 19:33.41 | *** join/#asterisk giesen (giesen@snickers.org) |
| 19:33.58 | Qwell | it's great when you know the quote well, and can hear it in their voice, with the pauses he makes |
| 19:34.22 | Deeewayne | yeah :-) I did have to look that up though to get the exact verbage |
| 19:35.28 | Deeewayne | Qwell, do you hear of people customizing and re-spinning asterisknow? |
| 19:35.47 | *** join/#asterisk KNERD (~KNERD@129.113.130.121) |
| 19:35.48 | Qwell | not really. there are a few distros using my RPMs though |
| 19:36.00 | Qwell | either the binaries we provide, or rebuilt from SRPM |
| 19:36.05 | Deeewayne | it wouldn't be too hard to do, though, right? |
| 19:36.18 | Qwell | not really |
| 19:36.57 | KNERD | Qwell: speaking of binaries...How can I prevent this from happening? "Error: asterisk16-core conflicts with asterisk14-core" I am trying only to install extra sounds |
| 19:37.01 | Deeewayne | I was thinking: do an install on a vm, grab the anaconda.cfg, maybe setup pxe, and do some customizations to the anaconda.cfg |
| 19:37.06 | Qwell | KNERD: disable EPEL |
| 19:37.09 | VultureZ | .... |
| 19:37.22 | KNERD | Qwell: groovy..then plet me try that now..thanks |
| 19:40.52 | Deeewayne | not sure if that's a good way to do it, just thinking out loud |
| 19:50.26 | *** join/#asterisk romb (~romb@unaffiliated/romb-work/x-7222485) |
| 19:51.05 | Qwell | Deeewayne: now it's'a brewin' |
| 19:53.46 | *** join/#asterisk seanjohn (~shanebrig@173.50.101.14) |
| 19:53.48 | seanjohn | I have 2.7.0, I have been writing dial plans for asterisk for years but trying to do Dial(Local/XXXXXXXXX@from-internal), it IGNORES the callerid(name) and callerid(number) I SET in the contexts immediately before using DIAL(). I have even tried setting BOTH USEROUTCID and TRUNKOUTCID in the context before Dial(). How do I override the default macro-user-callerid dialplans when trying to use from-internal? |
| 19:53.53 | seanjohn | freepbx 2.7.0 |
| 19:54.20 | seanjohn | USEROUTCID and TRUNKOUTCID are variables from-internal, of freepbx, uses |
| 19:54.26 | Qwell | seanjohn: #freepbx |
| 19:54.38 | seanjohn | I know but this is over their head, qwell |
| 19:54.45 | seanjohn | they are use to ONLY freepbx |
| 19:54.49 | Qwell | we can't help you with their configs |
| 19:55.14 | seanjohn | its not a config of freepbx, its OVERRIDING their configs with an asterisk dial plan |
| 19:55.42 | Qwell | "How do I override the default macro-user-callerid dialplans when trying to use from-internal?" |
| 19:55.45 | Qwell | we don't know. ask them. |
| 19:59.13 | *** join/#asterisk plundra (404@article.se) |
| 20:01.08 | *** join/#asterisk kisom (~kisom@c-f9dde155.648-1-64736c11.cust.bredbandsbolaget.se) |
| 20:03.21 | Deeewayne | gary coleman died today |
| 20:04.05 | Qwell | Deeewayne: see Simpsons quote above ;) |
| 20:04.09 | KNERD | what you talkin' 'bout Willis? |
| 20:04.38 | Deeewayne | Qwell, did you know when you said that? |
| 20:04.42 | Qwell | I did |
| 20:04.45 | Deeewayne | oh |
| 20:04.55 | Qwell | that was the first thing I thought of when I read about it |
| 20:05.10 | Deeewayne | first thing I thought too when I heard |
| 20:06.00 | Deeewayne | KNERD too, I guess |
| 20:10.36 | KNERD | yes |
| 20:10.40 | KNERD | whatchoo |
| 20:14.46 | *** join/#asterisk verywiseman (~khaled@unaffiliated/verywiseman) |
| 20:16.08 | *** join/#asterisk bsaxon (~bsaxon@12.68.234.174) |
| 20:17.23 | Beta2K | Will Asterisk terminate a T.38 fax call from a SIP device to a FXO port installed in the server? |
| 20:17.45 | *** join/#asterisk Baylink1 (~jra@cerberus.vicimarketing.com) |
| 20:18.07 | Qwell | that wouldn't be terminating it. that would be gatewaying it |
| 20:18.30 | Qwell | Asterisk doesn't currently support gatewaying like that. You can do store and forward though. |
| 20:18.49 | Beta2K | That's fine |
| 20:20.21 | Beta2K | Would I have to use something like hylafax to gateway the call? and if I'm using hylafax do I have to use a modem, or will a standard voice card work? |
| 20:20.33 | Qwell | ~faxforasterisk |
| 20:20.34 | infobot | somebody said faxforasterisk was Digium's commercial Fax For Asterisk module is available at http://www.digium.com/en/products/software/faxforasterisk.php |
| 20:20.35 | Qwell | or that |
| 20:21.28 | Beta2K | As much as I would love a plugin module I'm having a hard enough time convincing the boss that paying for g729 is a good idea :) |
| 20:23.20 | Qwell | and how much is your time worth? surely it's cheaper to buy a license than spend a bunch of time setting something else up.. |
| 20:24.59 | Beta2K | One would think that :) |
| 20:25.21 | *** join/#asterisk verywiseman (~khaled@unaffiliated/verywiseman) |
| 20:25.56 | Beta2K | Looks like fax for asterisk does a store and forward approach as well? |
| 20:26.19 | Qwell | that is what I said |
| 20:27.18 | Deeewayne | the one free channel license for FFA is pretty cheap too |
| 20:27.36 | Beta2K | Ok, so Asterisk currently can't gateway a fax on the fly at all? (just for my own knowledge) |
| 20:28.01 | Beta2K | Deeewayne: YEah I think I'm going to install it then when they need more channels say NP, fork out the $40 :) |
| 20:29.38 | *** join/#asterisk t_dot_zilla (~chatzilla@firewall-a.buf.ny.i-evolve.net) |
| 20:30.42 | *** join/#asterisk hfb (~hfb@96.247.64.187) |
| 20:31.59 | *** join/#asterisk waa (~waa@balrog.credipar.com.br) |
| 20:32.16 | Beta2K | Hrm, I'm on 1.6.0.9, there's downloads for 1.4.22 and later, 1.6.0.14 and later, which do I pick? |
| 20:32.26 | Qwell | Neither. upgrade. |
| 20:32.48 | Beta2K | Crap... |
| 20:34.08 | Beta2K | It's Trixbox and that's the latest version they seem to be offering |
| 20:34.19 | *** join/#asterisk _mwoodj_ (~mwoodj@pdpc/sponsor/digium/hyper-eye) |
| 20:34.25 | Qwell | use something else |
| 20:35.20 | waa | I'm running Asterisk SVN-trunk-r266240 and testing free fax for asterisk when I try to load res_fax.so asterisk crash with no error, someone have success running free fax in trunk? |
| 20:36.01 | Qwell | waa: not supported on trunk. use a released branch |
| 20:36.21 | waa | Qwell, thanks |
| 20:36.37 | Beta2K | Qwell: Thought you might say that :) I wonder if I could upgrade Asterisk in the TB machine without killing it... |
| 20:36.45 | Qwell | Beta2K: unlikely |
| 20:38.10 | *** join/#asterisk romb (~romb@unaffiliated/romb-work/x-7222485) |
| 20:40.13 | *** join/#asterisk x303 (~X303@187.159.121.70.cfl.res.rr.com) |
| 20:45.59 | *** join/#asterisk datarecal (~data@2607:fb98:4::2) |
| 20:46.22 | datarecal | how would i go about getting asterisk to do text messaging. Is it possible ? |
| 20:46.57 | Deeewayne | Qwell, I have to restart dahdi every time there is a lightning storm |
| 20:47.16 | Deeewayne | can you fix that please? |
| 20:47.27 | Deeewayne | :-) |
| 20:48.24 | hardwire | datarecal: text messaging could mean lots of things. which type? |
| 20:48.34 | jasonjjohnsonjr | Is there anyway to have asterisk set the callerid on an attended transfer to the number of the call being transferred? |
| 20:48.35 | datarecal | cell phone text messaging |
| 20:48.58 | hardwire | datarecal: there are SMS gateways that use GSM modems |
| 20:49.05 | hardwire | asterisk can execute external commands |
| 20:49.07 | [TK]D-Fender | jasonjjohnsonjr: No. That is the entire nature of an attended transfer |
| 20:49.29 | hardwire | datarecal: alternatively you just know the sms email ID for the phone and email it |
| 20:49.33 | jasonjjohnsonjr | [TK]D-Fender: Thanks |
| 20:50.00 | datarecal | yes basically what I would like to do is sms the asterisk phone number with a query and it text back information |
| 20:51.06 | *** join/#asterisk notjohn (~john@106.165.61.69.DED-DSL.fuse.net) |
| 20:51.12 | hardwire | datarecal: then use a GSM gateway |
| 20:51.16 | hardwire | or use smstools |
| 20:51.30 | hardwire | smstools operates with email-sms gateways as well as serially attached cellular phones |
| 20:51.56 | hardwire | it's easier if your cell phone or whoevers has email capability |
| 20:51.57 | hardwire | and cheaper |
| 20:59.10 | *** join/#asterisk verywiseman (~khaled@unaffiliated/verywiseman) |
| 21:13.11 | *** join/#asterisk Baylink (~jra@cerberus.vicimarketing.com) |
| 21:20.30 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.218.194) |
| 21:25.49 | *** join/#asterisk romb (~romb@unaffiliated/romb-work/x-7222485) |
| 21:26.18 | *** join/#asterisk Deeewayne (~dwayne@75-150-14-49-Atlanta.hfc.comcastbusiness.net) |
| 21:26.18 | *** mode/#asterisk [+o Deeewayne] by ChanServ |
| 21:29.07 | hardwire | is this thing on? |
| 21:29.12 | Jumpie | check chekc |
| 21:36.04 | Katty | hello my asterisk does not work at all how to fix pls |
| 21:36.13 | hardwire | Katty: READ THE BOOK YOU NEWB |
| 21:36.38 | Katty | what is book pls |
| 21:37.06 | hardwire | ~book |
| 21:37.07 | 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 |
| 21:37.09 | hardwire | newb! |
| 21:37.09 | [TK]D-Fender | hotm~osmosis |
| 21:37.14 | [TK]D-Fender | ~osmosis |
| 21:37.15 | infobot | [~osmosis] Osmosis is the act of beating yourself on the head repeatedly with THE BOOK, until some measure of absorption has occured ... or at least until your unconsciousness restores peace to the channel ... |
| 21:37.21 | hardwire | hotm... ? |
| 21:37.25 | Qwell | ail.com |
| 21:37.25 | hardwire | ama? |
| 21:37.28 | hardwire | onkeylove? |
| 21:37.29 | [TK]D-Fender | hardwire: over-type leftovers |
| 21:37.42 | hardwire | thinks Qwell is right. |
| 21:37.45 | hardwire | MSN lover. |
| 21:37.46 | Katty | Qwell: hello my asterisk does not work at all how to fix pls |
| 21:37.53 | Qwell | Katty: remove system32 |
| 21:38.04 | Katty | *hee* |
| 21:38.07 | Qwell | BALEET IT |
| 21:38.28 | Katty | i love gettign responses to that. |
| 21:38.32 | Katty | sometimes they are /really/ good |
| 21:38.35 | Qwell | troll |
| 21:38.39 | Katty | i know |
| 21:38.45 | Katty | but it's soooo fun |
| 21:39.05 | Beta2K | Katty: http://workaround.org/getting-help-on-irc |
| 21:39.45 | Katty | Beta2K: sorry, already got that one |
| 21:39.52 | Katty | Beta2K: how about something more original |
| 21:40.09 | Beta2K | Meh, it works most of the time :) |
| 21:40.30 | Katty | s'all good. i'll disappear for a couple hours and ask again ;P |
| 21:47.47 | Baylink-work | I particular liked my reply yesterday, but I got nothin. :-) |
| 21:54.12 | *** part/#asterisk bn-7bc (bjarne@lan-gw.noare-1.holmedal.net) |
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| 22:15.52 | *** part/#asterisk jasonjjohnsonjr (~jasonjjoh@asa1.jsmc.org) |
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| 22:43.20 | *** join/#asterisk mchou (~quassel@unaffiliated/mchou) |
| 22:44.06 | mchou | anyone use asterisk1.6 on openwrt/tomato? |
| 22:54.29 | *** join/#asterisk romb (~romb@unaffiliated/romb-work/x-7222485) |
| 22:59.06 | Beta2K | mchou: I can't imagine it being able to handle very many calls.... |
| 22:59.27 | mchou | Beta2K: that's not a requirement |
| 22:59.32 | mchou | residential use |
| 22:59.37 | Beta2K | Aye |
| 23:00.00 | Beta2K | Can't be too hard, I seem to remember there being packages for openwrt |
| 23:00.14 | mchou | it's not about being hard |
| 23:00.31 | mchou | it's about what HW is most stable and hassle free |
| 23:00.56 | mchou | most HW have too little flash or ram |
| 23:00.58 | Beta2K | Aye |
| 23:01.18 | Beta2K | Use a old PC |
| 23:01.30 | mchou | please |
| 23:01.55 | *** join/#asterisk fatman1683 (~IceChat7@207.34.9.120) |
| 23:02.05 | fatman1683 | Hi, anyone have any experience with the OpenVox analog cards? |
| 23:02.20 | mchou | you gonna use 90W/h on an old pc to save $30 on phone bill per mon? |
| 23:02.34 | mchou | makes no sense whatsoever |
| 23:03.19 | *** join/#asterisk jksM (~jks@193.189.93.254) |
| 23:03.40 | mchou | Beta2K: you've clearly not given the matter much consideration |
| 23:04.48 | mchou | I'd run it on a sheevaplug but that has issues too |
| 23:05.09 | *** part/#asterisk giany (~giany@shifu.x83.org) |
| 23:05.27 | mchou | not to mention up front expense for the "dev kit" |
| 23:05.45 | *** join/#asterisk jks (jks@193.189.93.254) |
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| 23:45.32 | KavanS | is there a way to get information on agents in asterisk? |
| 23:45.41 | KavanS | like if they are idle/busy/on a call, etc? |
| 23:46.17 | *** join/#asterisk neurosys (~neurosys@c-71-196-20-208.hsd1.fl.comcast.net) |
| 23:49.51 | carrar | like show agents or something? |