| 00:00.38 | ruied | kervel: but than the phone need to speak IAX (don't know if it does...) |
| 00:00.48 | hal2385614 | could someone confirm what values echotraining can take? Is it 32,64,128 and 256? |
| 00:01.41 | hal2385614 | are the units of these values in "taps"? |
| 00:02.25 | *** join/#asterisk boch (n=fran@190.48.226.45) |
| 00:02.26 | ruied | what if I make sip=yes for all my sip accounts? should this be a problem if the phone is in my intranet ? |
| 00:02.40 | *** join/#asterisk russellb (n=russell@asterisk/developer-and-stable-maintainer/drumkilla) |
| 00:02.40 | *** mode/#asterisk [+o russellb] by ChanServ |
| 00:05.21 | osas | hi russellb :) |
| 00:05.29 | russellb | osas: greetings |
| 00:07.10 | *** join/#asterisk Gr1ncheux (n=devine@unaffiliated/gr1ncheux) |
| 00:08.00 | *** join/#asterisk gniretar (n=icechat5@fhd.wccnet.org) |
| 00:08.05 | gniretar | hi |
| 00:08.18 | gniretar | what kinda extension type will act as a catchall? |
| 00:08.25 | gniretar | i want to record EVERYTHING |
| 00:08.41 | gniretar | so i need a lint at the top of the default context that will be executed no matter what |
| 00:08.43 | gniretar | line** |
| 00:09.51 | elriah | If anyone else is using Cisco phones and have brokem Message Waiting Indicators, let me know I found an easy fix by modifying chan_sip.c (in all versions of asterisk from 1.09-1.4). I'm going to put it up on the wiki when I get a chance. |
| 00:10.01 | *** join/#asterisk Iamnacho (i=Iamnacho@ip68-103-153-140.ks.ok.cox.net) |
| 00:11.28 | hal2385614 | ok, by adding echotraining=400 to the FXO interface (telephone line) I have brought the echo down to about 3-5 seconds. Can I hope for any improvement on this, or does everyone who use asterisk suffer this quirk? |
| 00:12.29 | elriah | hal2385614: What FXO card? |
| 00:15.04 | hal2385614 | it's a TDM400P |
| 00:15.51 | hal2385614 | any ideas? |
| 00:16.45 | elriah | Hrm... |
| 00:16.58 | elriah | Does it only do it after you restart asterisk on the first call or *every* call? |
| 00:18.09 | hal2385614 | Elriah - you know, I am not sure, as I haven't fully tested the latest settings - as diclophis only just suggested them to me |
| 00:18.15 | hal2385614 | It is certainly for the first few calls |
| 00:18.31 | hal2385614 | let me try again |
| 00:18.42 | *** join/#asterisk litage (n=nick@203.220.55.70) |
| 00:18.51 | elriah | When I used that card, the first call (or first few calls) would do that, then everything was great. |
| 00:18.57 | kervel | anyone knows a viking electronics reseller in europe ? |
| 00:22.25 | *** join/#asterisk r2e (n=r2e@user-0ccengp.cable.mindspring.com) |
| 00:22.53 | *** join/#asterisk litage (n=nick@203.220.55.70) |
| 00:24.07 | hal2385614 | elriah - no, the problem lasts for 5 seconds on every call |
| 00:24.31 | JT | hal2385614: what happens when echo training is disabled? |
| 00:24.40 | ruied | hal2385614, I don't have a single problem about echo.... did you have a look at zapata.conf? I think there is several algorithms that you can choose when compiling.... |
| 00:24.41 | diclophis-work | damn ok so i have my other problems figured out |
| 00:24.57 | diclophis-work | why is my caller id being mutilated in transit from one asterisk machine to anther, over a SIP channel? |
| 00:25.36 | gniretar | how would i write a config to logg the voice data of a meetme conference room? |
| 00:25.40 | hal2385614 | diclophis - can't you use iax from asterisk to asterisk? |
| 00:25.41 | hal2385614 | it may be better |
| 00:26.04 | JT | iax and sip should work equally well in most scenarios |
| 00:26.08 | JT | to be fair |
| 00:26.23 | JT | it's only edge cases where one may shine |
| 00:26.45 | ruied | hal2385614, I meant zaptel not zapata.conf... |
| 00:27.17 | JT | hal2385614: so what happens with ec training disabled? |
| 00:28.38 | thekidrio | anyone have a url for a basic pstn set up? |
| 00:28.52 | thekidrio | i have one incoming line and i have no idea how to tell if anything is being read on that channel |
| 00:28.52 | gniretar | noone knows how to get logging going on a MeetMe conference? |
| 00:28.59 | gniretar | or turn it on inversally? |
| 00:28.59 | thekidrio | i can get sip and iax working fine |
| 00:29.02 | gniretar | universally** |
| 00:29.24 | diclophis-work | gniretar: what type of "logging" ? |
| 00:29.52 | gniretar | voice, i want to record the conversations. I understand how MixMonitor works i jsut cant figure out how to implement it |
| 00:31.40 | diclophis-work | what i have been doing is the "R" option of meetme |
| 00:31.46 | diclophis-work | which is a built in record |
| 00:32.01 | diclophis-work | it grabs the filename to record to from a variable |
| 00:32.10 | diclophis-work | been working like a charm |
| 00:32.25 | gniretar | OK, what about recording other calls |
| 00:32.33 | gniretar | here is my situation |
| 00:32.42 | diclophis-work | well, that you will need to use monitor or mixmonitor or something |
| 00:32.47 | gniretar | i am building a central server for a hub and spoke topology |
| 00:32.48 | diclophis-work | or make all calls route through a conference |
| 00:33.02 | gniretar | i ahve a total of 8 servers with this one at the center |
| 00:33.10 | gniretar | i wanna record EVERYTHING that goes through it |
| 00:33.25 | gniretar | is there anything i can put at the beginning the my default context that will always be executed on every call |
| 00:33.27 | diclophis-work | hmmm..., why not record everything at the endpoints, to a central file server? |
| 00:33.42 | gniretar | or put it right below meetme (which will be doing its own logging) |
| 00:33.42 | *** join/#asterisk sloth_ (n=josh@64.3.170.50.ptr.us.xo.net) |
| 00:33.58 | gniretar | because the endpoints arnt under my control |
| 00:34.19 | diclophis-work | thats pretty tricky |
| 00:35.33 | diclophis-work | i would probably just call mixmonitor right before every dial command |
| 00:35.45 | sloth_ | Weird thing, I just switched my inbound carrier and now my greetings are not heard. The console shows that the playback() is playing the file but rtp debug shows no rtp packets.......? |
| 00:35.47 | gniretar | OK, that will work |
| 00:36.27 | diclophis-work | you will end up with 2 recordings per call though |
| 00:36.36 | gniretar | why? |
| 00:36.37 | diclophis-work | so.. maybe only record all "out" calls |
| 00:36.47 | diclophis-work | one channel is used for each person in the call |
| 00:36.55 | diclophis-work | if your recording their channel, and they are talking to each other |
| 00:37.05 | diclophis-work | there will be 2 files, with very similar sounds |
| 00:37.33 | JT | diclophis-work: not with MixMonitor |
| 00:37.37 | JT | that's Monitor |
| 00:37.41 | diclophis-work | oh |
| 00:37.43 | diclophis-work | rocking then |
| 00:37.44 | JT | Monitor stores each leg of the call |
| 00:38.05 | JT | MixMonitor before Dial works fine, as long as sip endpoints are not allowed to reinvite |
| 00:38.13 | gniretar | exten => _10XX,1,Dial(IAX2/team1:1secret1@192.168.1.100/${EXTEN:0},30,r) |
| 00:38.18 | gniretar | i basically ahve a ton of those |
| 00:38.21 | gniretar | have** |
| 00:38.36 | gniretar | for routing over the trunks |
| 00:38.43 | JT | no offence, but that's a stupid way of dialling |
| 00:38.51 | diclophis-work | ahaha |
| 00:39.00 | diclophis-work | yes please share |
| 00:39.07 | diclophis-work | i have a bunch of Dial commands like that too |
| 00:39.08 | JT | you should have an entry for each different iax client in iax.conf |
| 00:39.12 | *** join/#asterisk russellb (n=russell@asterisk/developer-and-stable-maintainer/drumkilla) |
| 00:39.12 | *** mode/#asterisk [+o russellb] by ChanServ |
| 00:39.17 | elriah | Are the Cisco phones suppoes to have a Do Not Disturb button somewhere? |
| 00:39.37 | JT | and just Dial(IAX2/<entry>/${EXTEN:0}.... |
| 00:39.43 | gniretar | ah, so it like SIP would do it for outbound? |
| 00:39.44 | diclophis-work | elriah: yes... but on some models its hidden more than others ... |
| 00:39.51 | JT | it stops you from logging passwords |
| 00:39.58 | elriah | diclophis-work: On a 7941? I swear, I can't find it. |
| 00:40.01 | diclophis-work | haha |
| 00:40.06 | JT | and allows you to change settings for a peer in one place |
| 00:40.27 | diclophis-work | uh, actually i don;t know, i am using a shitcomm phone |
| 00:40.31 | elriah | diclophis-work: Is it supposed to be enabled in the xml config? |
| 00:40.36 | diclophis-work | it has an auto-answer button |
| 00:40.50 | diclophis-work | i pressed it once, and then i couldnt turn it off |
| 00:41.04 | JT | gniretar: i mean i'm sure it may work, it's just hard to manage |
| 00:41.11 | gniretar | wait, does that mean that this: (paste coming) |
| 00:41.21 | JT | don't paste more than a few lines here |
| 00:41.21 | gniretar | http://pastecode.com/14272 |
| 00:41.24 | JT | :) |
| 00:41.41 | gniretar | will work for both directiona assuming both sides share auth? |
| 00:41.52 | gniretar | directions** |
| 00:42.03 | JT | gniretar: yes, it think so |
| 00:42.15 | JT | i can't tell just from looking at it if you'll need more options |
| 00:42.18 | JT | but that's the gist of it |
| 00:42.59 | JT | and for recording and what not |
| 00:43.08 | JT | if you don't have a macro for dialling, i suggest you make one |
| 00:43.11 | perd | damnit i cant believe you cant modify softkeys on the damn cisco 7960/7912 |
| 00:43.15 | JT | have a general macro |
| 00:43.22 | JT | it will make life much easier |
| 00:43.28 | gniretar | yea, i figured i should do that |
| 00:43.30 | perd | i want to kill the world |
| 00:43.46 | perd | rawwwwr |
| 00:43.51 | JT | so if you want to change the standard extension behaviour, if you have 10 phones, instead of adding 10 MixMonitor command or whatever, you do it once |
| 00:43.52 | *** join/#asterisk qdk (n=qdk@90.184.3.249) |
| 00:45.41 | JT | gniretar: if you need some tips, i can show you what i've done in some setups |
| 00:45.59 | diclophis-work | arg, how do i reference the caller id number?? for use in a application? |
| 00:46.01 | gniretar | OK, let me wrench on it a sec and i'l let u know |
| 00:46.25 | gniretar | with macros i can jsut use the s extension? |
| 00:46.41 | gniretar | just** |
| 00:46.46 | JT | diclophis-work: ${CALLERID(num)} |
| 00:46.58 | diclophis-work | the docs say thats removed in 1.4 .... |
| 00:47.06 | JT | gniretar: yes |
| 00:47.16 | JT | diclophis-work: no, CALLERIDNUM is removed |
| 00:47.19 | JT | the variable |
| 00:47.25 | JT | the function is the forrect was to do it |
| 00:47.29 | JT | correct |
| 00:47.31 | diclophis-work | oh cool |
| 00:47.51 | *** join/#asterisk hal23456 (n=chatzill@host217-43-87-172.range217-43.btcentralplus.com) |
| 00:47.58 | JT | check out README.variables in your docs directory |
| 00:48.08 | diclophis-work | all righty, now how would i switch based on the length of the caller id ?? |
| 00:48.09 | hal23456 | sorry - I got cut off |
| 00:48.31 | hal23456 | JT - what were you saying? |
| 00:49.09 | JT | hal23456: what happens when echo training is disabled? |
| 00:49.21 | JT | diclophis-work: incoming call? |
| 00:49.38 | diclophis-work | sorta |
| 00:49.47 | JT | hmm? |
| 00:50.01 | hal23456 | JT - well that was what it was like before. i shall try again... |
| 00:50.15 | JT | hal23456: was it constanly echoing? |
| 00:50.50 | scarfy | hi, its me again. i now tried to ad a configuration for voicemail. But i seem to have set it up right but when i call the VoiceMailMain i always get " -- Incorrect password ' ' for user '101' (context = localhost)" no matter what i do.. is there a special way to give the password? i enterd it in the dailpad of my softphone.. :-/ |
| 00:51.28 | JT | scarfy: not very familar with voicemail, but does typing # at the end help? |
| 00:52.11 | scarfy | JT, no it doesn't |
| 00:52.16 | perd | how do you enter the voicemail application |
| 00:52.19 | JT | diclophis-work: so it's not an incoming call? |
| 00:52.30 | diclophis-work | well it is the caller id of an incoming call |
| 00:52.40 | diclophis-work | i need to do one thing if its 3 digits and another if its 10 |
| 00:52.41 | JT | diclophis-work: over what technology? |
| 00:52.46 | perd | Voicemail(extension@localhost) ? |
| 00:52.47 | scarfy | JT, most of all it's not like that he gets the wrong pasword he gets none at all it just says ' ' is wrong |
| 00:52.49 | diclophis-work | mmm well its SIP |
| 00:52.50 | *** join/#asterisk bkruse_home (n=kruz@69.73.127.92) |
| 00:52.55 | elriah | Anyone know where a list of Cisco Feature IDs might be for their phones? |
| 00:52.58 | diclophis-work | ... but i dont think that matters? |
| 00:53.03 | elriah | I've been searching for an hour and I can't find one... |
| 00:53.20 | JT | diclophis-work: it does a little, in theory, but here's the answer, i think: |
| 00:53.28 | scarfy | perd, exten => 8,1,VoiceMailMain(101@localhost) |
| 00:53.52 | JT | diclophis-work: whatever extension you have catching the incoming calls now, replace/modify it with 2 new extensions |
| 00:54.03 | perd | paste the relevant sections of extensions.conf and voicemail.conf to pastebin.ca |
| 00:54.13 | JT | with <whatevertheextensionwasbefore>/XXX,1,Blah..... |
| 00:54.15 | JT | and |
| 00:54.16 | perd | and paste logs of when you try to access the box |
| 00:54.34 | JT | <whatevertheextensionwasbefore>/XXXXXXXXXX,1,Blah..... |
| 00:54.41 | JT | oh |
| 00:54.48 | JT | i probably missed an underscore |
| 00:54.52 | JT | but you get the idea |
| 00:54.57 | JT | what's how you co callerid matching |
| 00:55.03 | JT | do |
| 00:56.24 | hal23456 | JT - when echocancel and echo training are turned off, echo lasts for 10 seconds |
| 00:56.31 | hal23456 | when echocancel is enabled, echo lasts for 5 seconds |
| 00:56.31 | JT | ah damn |
| 00:56.36 | JT | ok |
| 00:56.39 | diclophis-work | mmm, the thing is it should be switched on caller id |
| 00:56.43 | diclophis-work | it has nothing to do with the extension |
| 00:56.45 | JT | maybe your lines are just crap, hal23456 ? |
| 00:56.57 | JT | diclophis-work: sorry, not understanding what you mean |
| 00:57.09 | diclophis-work | ... i guess i can do a Goto(${CALLERID(num)}) then do a pattern match extension for 10 or 3 digits.... |
| 00:57.25 | JT | diclophis-work: what |
| 00:57.31 | diclophis-work | nvrmnd |
| 00:57.35 | diclophis-work | i am overcomplicating things |
| 00:57.39 | diclophis-work | and i just ran into another problem, |
| 00:57.41 | JT | diclophis-work: umm, what exactly are you trying to acheive? |
| 00:57.41 | scarfy | perd, http://pastebin.ca/354531 |
| 00:57.46 | hal23456 | I don't think so - I had the same problem when I was configuring this with the same equipment on a different line a couple of years ago |
| 00:57.51 | JT | asterisk has inbuilt callerid extension matching syntax |
| 00:57.52 | diclophis-work | JT dynamic logins to queues |
| 00:58.04 | diclophis-work | whereby we can have agents call in and join/leave queues |
| 00:58.08 | hal23456 | I think we need to assume that the lines are no better or worse than any other line |
| 00:58.19 | diclophis-work | is there a way to remove a partically member/agent from all queues? |
| 00:58.24 | diclophis-work | i cant seem to find that |
| 00:58.29 | JT | diclophis-work: ok, and why can't you do that using asterisk's inbuilt syntax? |
| 00:58.30 | JT | dunno |
| 00:59.00 | JT | hal23456: that's not true though, some lines ARE much worse |
| 00:59.08 | JT | hal23456: but there are different software EC algorithms |
| 00:59.14 | JT | perhaps try a different one |
| 00:59.37 | hal23456 | what determines which algorithm to use? Is it the country? |
| 01:01.29 | JT | nah, guessing, with lack of a better knowledge of them, some work better sometimes |
| 01:01.30 | scarfy | perd, did you see it? |
| 01:03.16 | hal23456 | ok - so after spending 100s of hours troubleshooting this configuration, I change the algorithm, then I presume i have to experiment with the 100s of permutations of possible settings? |
| 01:03.25 | elriah | You guys know what, there ain't SH*T documentation for Cisco IP phones. |
| 01:03.41 | JT | elriah: unless you're on a support contract ;) |
| 01:03.44 | *** join/#asterisk |dennis| (n=dennis@200.32.233.82) |
| 01:03.50 | mercestes | <PROTECTED> |
| 01:04.05 | elriah | I have a contract. There still ain't sh*t. I'm just trying to find a list of freakin' feature ids for their new XML config system... |
| 01:04.08 | hal23456 | elriah - which phones are you talking about - which 79xx? |
| 01:04.18 | elriah | 7941/7961 |
| 01:04.20 | JT | hal23456: perhaps.... what can i say, analogue lines suck, digital lines suck a bit less |
| 01:04.20 | scarfy | oh great as fast as perd came on the map he went again |
| 01:04.42 | *** join/#asterisk J4k3 (i=jsuter@dhcp-12-197-128-58.intrastar.net) |
| 01:04.42 | JT | mercestes: i don't know if getting certified alone gives you anything |
| 01:04.48 | diclophis-work | is it possible to define per an agent, multiple "channels/numbers/interfaces" to try ? |
| 01:05.04 | hal23456 | well, even though I a pretty pissed off, you have at least made me smile, JT! Thank you |
| 01:05.11 | mercestes | JT: I know a few CCIEs. Their website has a bunch of the firmware crap and "knowledge base" etc. basically all the things non-certified ppl need. |
| 01:05.40 | JT | right, CCIE i guess, wonder if they have to pay anual fees |
| 01:05.49 | mercestes | <PROTECTED> |
| 01:05.55 | scarfy | Doe anybody else know how the voicemail works? |
| 01:06.03 | mercestes | scarfy: yes. |
| 01:06.07 | gniretar | CCIEs are fun |
| 01:06.10 | gniretar | i know 2 of them |
| 01:06.14 | gniretar | i'm just a lowly CCNP |
| 01:06.15 | JT | hal23456: i don't deal with analogue lines to the telco myself, so hmm :/ |
| 01:06.42 | JT | i could've got my CCNA, but seeing as it expires every 3 years, i thought it was pointless |
| 01:06.50 | hal23456 | Elriah - did you say you were using firmware 7 or 8? |
| 01:07.12 | *** join/#asterisk BSDTech (n=RNeese@ppp-69-238-49-231.dsl.irvnca.pacbell.net) |
| 01:07.15 | gniretar | well, i was 16 when i got my CCNP so it was a serious accomplishment regardless of expiration |
| 01:07.21 | gniretar | i was 14 when i got the CCNA |
| 01:07.37 | scarfy | mercestes, i just configure an voicemail on my system http://pastebin.ca/354531 But everytime i call the VoiceMailMain i gon't get in " Incorrect password ' ' for user '101' (context = localhost)" no matter what i type on the softpohone when i call it |
| 01:07.45 | JT | heh, age isn't everything |
| 01:07.45 | BSDTech | MusicOnHold(default) with this option how do you set how long it should play moh |
| 01:07.46 | elriah | Firmware 8.4SR2 (java based). The phone works great, I'm just trying to figure out how to program the softkeys... And apparently, features are enabled by: <featureID>9</featureID> but a list of such ID's doesn't exist in the world... |
| 01:07.54 | JT | and to be honest, the ccna is super basic |
| 01:08.32 | mercestes | scarfy: This config makes no sense. |
| 01:08.33 | gniretar | CCNA basically means you know command line really well and you have a framework of knowledge that allows you to learn more advanced topics |
| 01:08.46 | BSDTech | ? |
| 01:08.47 | scarfy | mercestes, good then it would explain why it doesnt work :D |
| 01:09.01 | JT | i thought it means more that you can remember routing algorithms ;) |
| 01:09.02 | scarfy | mercestes, both of them? |
| 01:09.04 | hal23456 | elriah - are the configs different to firmware 7-04-00? |
| 01:09.08 | *** join/#asterisk naitek7954 (n=support@24.231.48.236) |
| 01:09.13 | naitek7954 | hello all |
| 01:09.17 | elriah | Not terribly... |
| 01:09.21 | gniretar | here is an interesting one : loader.c:325 __load_resource: /opt/asterisk-1.2.15/usr/lib/asterisk/modules/chan_zap.so: undefined symbol: ast_pickup_call |
| 01:09.21 | *** join/#asterisk hematitec (n=cratz@adsl-71-159-206-4.dsl.pltn13.sbcglobal.net) |
| 01:09.28 | JT | remember acronyms for routing algorithms, even |
| 01:09.30 | naitek7954 | I am trying to set up a simple pbx system for my startup company. |
| 01:09.31 | gniretar | anyone have any ideas? |
| 01:10.00 | naitek7954 | What do i need to connect the 2 telco lines to the system? i am not using T1/E1 but PSTN |
| 01:10.04 | mercestes | scarfy: All of it. You pay no regard to peer name, username, context, predictable call flow, legibility, etc. |
| 01:10.13 | JT | naitek7954: TDM400P |
| 01:10.17 | mercestes | scarfy: you really need to read the book again or hire a consultant. |
| 01:10.35 | naitek7954 | <JT> is that enough for 2 lines? |
| 01:10.54 | JT | it's enough for up to 4 |
| 01:10.56 | scarfy | mercestes, yeah its my first time i try to set up an asterisk server... |
| 01:11.00 | naitek7954 | Thank you so much.. |
| 01:11.03 | JT | it'd modular and you can get FXS or FXO modules |
| 01:11.21 | naitek7954 | WHat phones do you recommend? |
| 01:11.24 | mercestes | scarfy: You send a voicemail to {u or b}USERNAME not {u or b}SIP/username and it's username@voicemail_context your specifing vm contexts that don't exist (localhost) |
| 01:11.34 | elriah | Of course, who knows, since there isn't any freakin' docs anywhere... |
| 01:11.43 | hal23456 | elriah - what is the name of the xml config filename? |
| 01:11.53 | elriah | SEP<mac>.cnf.xml |
| 01:12.01 | naitek7954 | JT: What phones would be good to allow multiple lines |
| 01:12.11 | mercestes | scarfy: So you either need to create a [localhost] in your vm.conf or strip out the @localhost and let it go to default. |
| 01:12.54 | hal23456 | have you got these phones to load the configuration file yet, and you just need to find the options? |
| 01:13.15 | elriah | hal23456: I'm trying to figure out how to program the softkeys and find a list of feature IDs. |
| 01:13.22 | hematitec | <PROTECTED> |
| 01:13.53 | hal23456 | please confirm that you have confirmed that the phone is actually downloading the config from the tftp server? |
| 01:14.11 | scarfy | mercestes, ahhh that makes sense.. i added the localhost cause i thought i had to tell him where to put it didn't get that was a wm contexts .. must have interpreted it wrong |
| 01:14.14 | JT | naitek7954: SIP hard phones are the way to go |
| 01:14.40 | JT | naitek7954: "lines" is an antiquated concept when it comes to SIP |
| 01:14.50 | JT | sip can make and destroy connections at will |
| 01:15.03 | elriah | hal23456: Right, the phones work great, I have a sample config from voip-info.org. I'm making and receiving calls. I even have the phones branded with our company logo. Now I'm just trying to setup softkeys with features (ie, do not disturb, etc). |
| 01:15.17 | elriah | I even have ringtones... |
| 01:15.18 | elriah | lol |
| 01:15.19 | hal23456 | naitek - cisco phones are very good, however, polycom manufacture some of their phones, so polycom are also another good, but often cheaper option |
| 01:15.36 | JT | hal23456: cisco phones have poor sip support |
| 01:15.44 | hal23456 | ok - good jobĀ¬ |
| 01:15.45 | JT | and poor support if you don't pay them $$$ in general |
| 01:15.50 | perd | im going to have to give cisco phones a big thumbs down |
| 01:16.09 | elriah | JT: Really? It seems to be working great for me... We just deployed 25 of them... Hope we don't run into issues... |
| 01:16.18 | hal23456 | JT - you can get the firmware on P2P |
| 01:16.21 | perd | as jt says, cisco has horrible support, horrible restrictions on firmware AND their phones are completely unconfigurable past simple settings |
| 01:16.24 | JT | elriah: are you on a support contract? |
| 01:16.37 | perd | you cant even reprogram the softkeys on them.. which begs the question why the hell do you call them softkeys |
| 01:16.41 | elriah | We have smartnet on the phones so yea... |
| 01:16.45 | JT | hal23456: wtf, you're suggesting that people download illegal stolen firmware from P2P for business purposes? |
| 01:16.46 | hal23456 | I agree, that they are not easy to configure |
| 01:16.48 | elriah | It only cost $11.00 per phone. |
| 01:16.58 | hal23456 | no - I am not |
| 01:17.11 | perd | haha you werent? |
| 01:17.26 | hal23456 | I am saying it is an option, but the decision is the responsibility of the individual |
| 01:17.34 | hal23456 | I would not recommend this for business purposes |
| 01:17.43 | JT | the decision sounds simple: polycom for the best sip phones |
| 01:17.47 | scarfy | mercestes, yes now it accepts the mailbox number but still not the password.. is that still a problem with the configuration or could that be the softphone? cause he seems to get no numbers whatsoever |
| 01:17.56 | perd | just stay away from cisco phones |
| 01:17.59 | JT | cisco support SCCP for the phones first and foremost, SIP as an afterthought |
| 01:18.07 | perd | people who like cisco phones are insane. |
| 01:18.18 | hal23456 | Yes, that is true, JT |
| 01:18.31 | naitek7954 | polycom |
| 01:18.32 | naitek7954 | good |
| 01:18.40 | J4k3 | Crisco |
| 01:18.58 | hal23456 | What hard phone would you recommend, JT? |
| 01:19.00 | JT | mm, transfatty acids |
| 01:19.10 | J4k3 | fried right in crisco! |
| 01:19.12 | J4k3 | (and lightning....) |
| 01:19.21 | JT | hal23456: polycom is the name i hear time and time again :) |
| 01:19.28 | JT | then aastra |
| 01:19.43 | elriah | We have some polycom phones (601's) and they are good phones. The speakerphone sucks, though. |
| 01:19.50 | hal23456 | Yes, and it is the one that I mentioned also |
| 01:19.52 | J4k3 | haha |
| 01:20.10 | JT | to be honest, most sip phones are still pretty immature these days |
| 01:20.12 | hal23456 | Polycom makes (or used to make) the cisco's conference phones! |
| 01:20.24 | diclophis-work | how would i prompt for a number? from the caller? |
| 01:20.27 | sevard | JT: they're getting better |
| 01:20.31 | sevard | they were once, really, really bad |
| 01:20.34 | diclophis-work | that i then want to plug into an application |
| 01:20.37 | gniretar | i have cdr configured (with the mysql module in asterisk-addons) but its not logging and i cant seem to find any error messages |
| 01:20.40 | gniretar | where should i look? |
| 01:20.50 | JT | diclophis-work: there's a few commands like that |
| 01:20.59 | JT | DISA for a dialtone and dropping someone into a context |
| 01:21.00 | diclophis-work | Background perhaps? |
| 01:21.04 | J4k3 | I've had the displeasure of using an ISDN BRI desk phone. It was about equally as annoying/braindead/thoughtless as the typical sip phone. |
| 01:21.04 | JT | and there's others |
| 01:21.17 | diclophis-work | well i want to play a prompt/instructions |
| 01:21.29 | J4k3 | I think the SIP phone makers are all clueless |
| 01:21.32 | JT | i've never used an isdn bri phone, and isdn bri is fairly common in small businesses here |
| 01:21.42 | J4k3 | like, for example, I can't find a VoWIFI set that supports a loss-friendly protocol. |
| 01:21.50 | JT | the problem is SIP does not standardise enough pabx/key system functions |
| 01:21.52 | hal23456 | so the concensus is "polycom"?! Noone has good experiences with any others? |
| 01:21.58 | JT | so different manufacturers do it differently |
| 01:22.00 | J4k3 | ilbc? gsm? no. Youve got 729, 711, or completely broken 726. |
| 01:22.11 | elriah | I didn't have any issue setting up the Cisco 7941's with Asterisk. The only two things are: 1) Make sure you have the right firmware and 2) take two characters out of channel_sip.c in asterisk and recompile... |
| 01:22.11 | JT | why not get dect? |
| 01:22.39 | gniretar | i loaded cdr_addon_mysql.so thats all i should have to do right? besides configure the DB in the conf file? |
| 01:22.41 | scarfy | mercestes, it says "[Feb 14 02:14:59] WARNING[5919]: app_voicemail.c:6039 vm_authenticate: Unable to read password" |
| 01:22.43 | hal23456 | why, what does that do, elriah? |
| 01:22.46 | J4k3 | JT: hassle. I've already got solid 802.11g coverage around the place. |
| 01:22.52 | elriah | hal23456: Fixes MWI. |
| 01:23.01 | J4k3 | I believe the core of my asterisk issues is my box |
| 01:23.02 | hal23456 | please elaborate? |
| 01:23.09 | diclophis-work | JT would a combo of Background/Read do the trick? |
| 01:23.19 | elriah | hal23456: There's a string hardcoded in channel_sip.c "(0/0)" that messes up the sip packets. |
| 01:23.22 | JT | J4k3: i understand that voip over 2 fixed points wireless link that's running well should be reliable, but not moving about witha wifi phone |
| 01:23.23 | gniretar | <PROTECTED> |
| 01:23.26 | elriah | hal23456: When a message is waiting |
| 01:23.30 | JT | diclophis-work: check the book man |
| 01:23.31 | gniretar | == Unregistered 'mysql' CDR backend |
| 01:23.31 | J4k3 | I can't get the damned thing to show better than 99.92% in zttest :| |
| 01:23.34 | elriah | hal23456: 1.2 and 1.4 |
| 01:23.34 | gniretar | i get those lines |
| 01:23.37 | hal23456 | and what is the result? |
| 01:23.40 | diclophis-work | i am reading the book right now |
| 01:23.44 | diclophis-work | i have 10 windows open of the book |
| 01:23.54 | elriah | hal23456: ? |
| 01:24.06 | JT | J4k3: i think x100ps sometimes don't score very well |
| 01:24.09 | J4k3 | JT: yeah. I've carried one of my grandstream's to a very rural customer's house (actually a 3rd generation repeat site)... Perfect quality on 729 and 711. |
| 01:24.10 | JT | no matter what you do |
| 01:24.18 | hal23456 | what are the symptoms of the messing up sip packets? |
| 01:24.26 | J4k3 | the f1000g's sound great... then they start warbling |
| 01:24.36 | elriah | hal23456: Oh, no MWI on Cisco phones... |
| 01:24.50 | hal23456 | sorry, what is MWI? |
| 01:24.52 | *** join/#asterisk flenders (n=fserto@unaffiliated/flenders) |
| 01:24.52 | JT | elriah: submitted to mantis? |
| 01:24.54 | J4k3 | sometimes they start off warbling |
| 01:24.59 | elriah | Message Waiting Indicator |
| 01:25.07 | *** join/#asterisk PaulTech85 (i=PaulTech@72.29.76.254) |
| 01:25.13 | *** part/#asterisk naitek7954 (n=support@24.231.48.236) |
| 01:25.22 | elriah | JT: It was already there, for 1.4.x, I just backported the patch to 1.2, same function. |
| 01:25.23 | hal23456 | oh, really? my Message waiting work fine, i think |
| 01:25.27 | JT | ah ok |
| 01:25.38 | gniretar | JT: any ideas on cdr_addon_mysql? |
| 01:25.40 | PaulTech85 | Is there a trigger for when someone picksup a queue call? Say I want to send information to a persons screen when they answer a call in the queue. Can asterisk detect what extension answered a call and how do I get access to it? |
| 01:25.46 | elriah | hal23456: On 8.2SR1 or better its broke. |
| 01:25.52 | JT | gniretar: umm, do you want to use it, or not? |
| 01:26.10 | gniretar | i want to use it. When i reload the module it says its 'unregistering' it |
| 01:26.17 | gniretar | and its not logging |
| 01:26.24 | J4k3 | doesn't seem to matter if the AP is busy, using 11b or 11g, -40 or -80 signal, whatever.... it just works great sometimes and works like crap otherwise |
| 01:26.43 | *** join/#asterisk mtgll (n=mtg@static-71-125-10-2.nycmny.fios.verizon.net) |
| 01:26.46 | PaulTech85 | Any ideas? |
| 01:27.03 | hal23456 | I see |
| 01:27.06 | hal23456 | thanks for that |
| 01:27.57 | JT | gniretar: is it configured? |
| 01:28.06 | gniretar | i have a configuration for it in cdr_mysql.conf |
| 01:28.25 | JT | hmm ok |
| 01:28.39 | PaulTech85 | I see 'URL' |
| 01:28.44 | PaulTech85 | is a option for queue |
| 01:28.48 | PaulTech85 | Maybe I can work with that.. some how |
| 01:28.49 | gniretar | If i change the password in the config to something incorrect it gives me this: Feb 13 15:30:07 ERROR[6107]: cdr_addon_mysql.c:437 my_load_module: Failed to connect to mysql database asteriskcdrdb on localhost. |
| 01:28.53 | gniretar | so its trying to connect |
| 01:29.18 | hal23456 | elriah - do you know what the main advantages are of 8.2SR over 7.4? |
| 01:29.40 | gniretar | on the CLI when i do a CDR status |
| 01:29.47 | gniretar | mysql is a registered backend |
| 01:29.53 | elriah | hal23456: No clue really. I know that once I put the 8.2SR firmware in my TFTP directory, all my problems went away. |
| 01:30.08 | hal23456 | what kind of problems? |
| 01:30.48 | JT | gniretar: umm maybe it's working? |
| 01:31.09 | *** part/#asterisk mtgll (n=mtg@static-71-125-10-2.nycmny.fios.verizon.net) |
| 01:31.19 | gniretar | JT: yea.... i jsut looked at mysql directly (i was attempting to view through asterisk-stat) and i see the listings |
| 01:31.24 | gniretar | i must ahve misconfigured asterisk-stat |
| 01:31.27 | gniretar | have* |
| 01:31.31 | JT | heh |
| 01:31.36 | gniretar | sorry to be a bother ;-) |
| 01:32.02 | hal23456 | elriah - what kind of problems? |
| 01:32.04 | *** join/#asterisk wubba (n=kmurrey@cable-76-215.sssnet.com) |
| 01:32.07 | PaulTech85 | No one have any idea? ;) |
| 01:34.27 | hal23456 | guys - its late and I have to go. Thanks to everyone for all your help! |
| 01:34.37 | hal23456 | Byee |
| 01:36.23 | *** join/#asterisk umop3plsdn (n=da@cpe-76-179-77-186.maine.res.rr.com) |
| 01:38.21 | hematitec | PaulTech: I have not work in this area yet, but take a look at the manager interface |
| 01:38.21 | diclophis-work | is AEL the future of asterisk? |
| 01:38.32 | diclophis-work | like would it be worth the time to figure out how that works? |
| 01:41.39 | J4k3 | hrm... is this acceptable? |
| 01:41.39 | J4k3 | --- Results after 601 passes --- |
| 01:41.39 | J4k3 | Best: 99.987793 -- Worst: 99.877930 -- Average: 99.927926 |
| 01:41.47 | Qwell | J4k3: no |
| 01:41.50 | J4k3 | ok |
| 01:42.01 | J4k3 | thought so |
| 01:42.26 | JT | but it's an x100p, not sure if it would necessarily improve |
| 01:42.56 | *** join/#asterisk Snake-Eyes (n=blog@203.220.55.70) |
| 01:42.58 | diclophis-work | i have never seen better than 99.98 |
| 01:43.04 | J4k3 | noooo. |
| 01:43.08 | diclophis-work | and i have spent days tweaking the bios and shit |
| 01:43.10 | J4k3 | I'm not using the x100p anymore |
| 01:43.26 | JT | diclophis-work: 100% i've had, but not constant, other have had constant |
| 01:43.29 | JT | J4k3: what are you using? |
| 01:43.31 | J4k3 | the only ?100* product I currently use around here now is an f1000g. |
| 01:43.38 | *** join/#asterisk shodan- (n=shodan@ip043.96-113-216.pppoe1.joliette.intermonde.net) |
| 01:43.40 | J4k3 | vitelity |
| 01:43.47 | JT | err |
| 01:43.52 | JT | let me get this straight |
| 01:43.59 | J4k3 | and a ut starcom f1000g, and some grandstream phones |
| 01:44.01 | JT | you are doing zttest with no zaptel hardware? |
| 01:44.05 | PaulTech85 | hematitec I was trying to avoid doing a "constant" check |
| 01:44.06 | J4k3 | yes |
| 01:44.09 | flenders | hi, can anyone give me a hand with SIP accounts? I can receive calls via a sip channel, but I can't dial out. |
| 01:44.17 | JT | J4k3: do you use MeetMe or IAX2 Trunking? |
| 01:44.19 | PaulTech85 | I was hoping more for a trigger, Oh You mean it may pass back a 'Event' |
| 01:44.19 | PaulTech85 | ? |
| 01:44.39 | J4k3 | JT: no |
| 01:44.46 | JT | J4k3: zttest does not matter then |
| 01:44.50 | J4k3 | ok |
| 01:45.07 | JT | your zttest scores will be coming from ztdummy, which is not going to be as accurate as real hardware |
| 01:45.29 | J4k3 | that works, I was hoping it may be a clue to why the f1000g works well sometimes and not well others. |
| 01:45.45 | JT | it's friggen wifi :D |
| 01:45.46 | JT | jitter |
| 01:45.59 | BSDTech | ok when using MusicOnHold() what is the timeing set as seconds ? |
| 01:46.42 | Qwell | What timing? |
| 01:46.43 | Qwell | oh |
| 01:46.55 | Qwell | right, what timing? |
| 01:47.14 | J4k3 | well... xlite works great to my laptop |
| 01:47.42 | J4k3 | when using a usb headset (sound chip and wifi card sit on the same irq otherwise, it'll hiccup) |
| 01:47.51 | J4k3 | on the same wifi, throughout the house |
| 01:47.52 | J4k3 | so |
| 01:47.55 | J4k3 | I call it, the f1000g sucks. |
| 01:47.57 | J4k3 | ;) |
| 01:48.34 | JT | no shit ;) |
| 01:50.33 | *** join/#asterisk k-man (n=jason@unaffiliated/k-man) |
| 01:50.46 | BSDTech | () needs to know how long to ply moh |
| 01:50.48 | k-man | has anyone heard of the quadro2xi pbx? |
| 01:50.54 | k-man | with isdn port? |
| 01:51.00 | Qwell | BSDTech: show application MusicOnHold |
| 01:51.23 | k-man | is there anyone in australia providing supported asterisk instalations? |
| 01:51.26 | BSDTech | I am not currently infront of a box |
| 01:51.41 | BSDTech | I am just writing dial plan crap |
| 01:52.05 | BSDTech | and the webpage on voip-info does not have the answer |
| 01:52.26 | [TK]D-Fender | MoH does not need a timing source.... |
| 01:52.39 | BSDTech | I know this |
| 01:52.47 | BSDTech | I need to set how long it plays |
| 01:52.52 | Qwell | You can't |
| 01:53.04 | Qwell | [Synopsis] |
| 01:53.04 | Qwell | Play Music On Hold indefinitely |
| 01:53.17 | BSDTech | ? |
| 01:53.30 | Qwell | there is no timeout - it plays forever |
| 01:53.32 | BSDTech | why cant it be set to time out |
| 01:53.44 | sevard | why would you want it to time out? |
| 01:53.46 | wubba | can't you do a loop BSD? |
| 01:53.48 | BSDTech | like when you put a person on hold |
| 01:53.49 | Qwell | because there are plenty of other ways to do it |
| 01:53.58 | Qwell | when you put a person on hold...they're...on hold |
| 01:54.04 | [TK]D-Fender | What you do is dial Local with and Absolute timeout, and the when it expires the call will fall back <- |
| 01:54.05 | Qwell | they aren't sitting in dialplan |
| 01:54.05 | J4k3 | 2.4ghz antenna within a half inch of the human skull = stupid. |
| 01:54.12 | BSDTech | ok what I a\m doing is this |
| 01:54.27 | J4k3 | a couple inches of 'mast' would improve things. |
| 01:54.47 | BSDTech | a call comes in checks if a line is busy with a call and if chanstatus=busy it plays a menu |
| 01:54.56 | BSDTech | 1 to hold 2 to leave vm |
| 01:55.01 | BSDTech | 3 to hangup |
| 01:55.06 | Qwell | why would you put yourself on hold? |
| 01:55.13 | BSDTech | not myself |
| 01:55.18 | Qwell | why would I put yourself on hold? |
| 01:55.23 | Qwell | s/you/me |
| 01:55.25 | Qwell | whatever |
| 01:55.43 | BSDTech | I have a client that wants this |
| 01:55.45 | Qwell | what sane caller would hit 1? |
| 01:56.01 | Qwell | and your job is to say "No, you're an idiot. You should do it like this." |
| 01:56.13 | BSDTech | I have tried and tried |
| 01:56.17 | BSDTech | but they want it |
| 01:56.42 | wubba | I have heard some systems that do that BSD - so the request is not insane |
| 01:57.12 | Qwell | wubba: Just because more than 1 person does something, it doesn't mean that they're sane |
| 01:57.31 | BSDTech | so how can I put them on hold for a 30 seconds and then recheck the status and replay the menu |
| 01:57.53 | ping2921 | whenever I have queue annoucements, the music restarts, anyone knows why? |
| 01:58.16 | wubba | Qwell - When I was in charge of a Help Desk we would ask the user if they wanted to continue to hold for a technician or to leave a message. |
| 01:58.16 | *** join/#asterisk Snake-Eyes (n=blog@203.220.55.70) |
| 01:58.28 | wubba | By pressing 1 or 2 |
| 01:58.34 | wubba | so there is a real world need for that |
| 01:58.38 | Qwell | okay, that's a queue option |
| 01:58.48 | Moobius | wubba: i HATE that. |
| 01:58.49 | Qwell | completely different |
| 01:58.51 | JT | actually, functionally it seems reasonable to put yourself onhold |
| 01:58.59 | JT | i often would rather be on hold than leave a message |
| 01:59.04 | BSDTech | but I dont want a queue for every exten |
| 01:59.06 | JT | or call back random amount of time later |
| 01:59.25 | Moobius | how about "press 1 to wait on hold, otherwise" <get dumped to vm> |
| 01:59.28 | wubba | Moob.. I didn't create it |
| 01:59.33 | ping2921 | should the moh restart after a queue announcement? |
| 01:59.37 | BSDTech | thats the other thing is a request a call back when line is free |
| 01:59.49 | Moobius | wubba: but you must know that it is annoying and difficult for your callers. |
| 01:59.52 | BSDTech | but thats ltr |
| 02:00.12 | wubba | Moob - that was 10 years ago - I don't do it anymore - relax ok. |
| 02:00.17 | JT | Moobius: the better ones put you on hold, but give you the option of opting to leave a message |
| 02:00.32 | BSDTech | so how can I allow a user to put them selves on hold |
| 02:00.34 | Moobius | i've delt with it recently. in a very long queue. |
| 02:00.39 | JT | they even claim that the messages retains your position in queue |
| 02:00.44 | JT | but i'm skeptical ;) |
| 02:00.45 | Qwell | BSDTech: [TK]D-Fender gave you an answer already |
| 02:00.49 | wubba | Ours did |
| 02:01.12 | JT | wubba: retain position? |
| 02:01.16 | wubba | yes |
| 02:01.25 | JT | heh |
| 02:01.42 | wubba | But it was up to the tech to actually listen to the VM and call the EU |
| 02:02.09 | BSDTech | it seems it would be smart to have a timeout option in the MOH |
| 02:02.09 | Moobius | which, as we know, is different than a ringing phone.. |
| 02:02.18 | BSDTech | and I did not see a option |
| 02:03.13 | Qwell | BSDTech: Why? |
| 02:03.24 | Qwell | there are plenty of other ways to do it |
| 02:04.02 | gniretar | does anyone know what bitrate the wav files created by Asterisk are? |
| 02:04.05 | BSDTech | I am not finding any of them in the documentation |
| 02:04.16 | BSDTech | thats why I am asking for help with this |
| 02:04.17 | Qwell | BSDTech: [TK]D-Fender gave you an answer already |
| 02:04.33 | BSDTech | I did not see it |
| 02:04.54 | [TK]D-Fender | [20:54] <[TK]D-Fender> What you do is dial Local with and Absolute timeout, and the when it expires the call will fall back <- |
| 02:05.02 | JT | scroll up then |
| 02:05.06 | [TK]D-Fender | Nobody ever listens to me... *sigh* |
| 02:05.26 | BSDTech | but the idea is to have it play moh |
| 02:05.33 | BSDTech | while they wait |
| 02:05.38 | [TK]D-Fender | I swear... you can lead a horse to the wate... but you can't hold its head under long enough.... |
| 02:05.58 | diclophis-work | thanks all |
| 02:06.08 | PaulTech85 | Is there a way with queues, To not ring a agent thats already on a call |
| 02:06.17 | PaulTech85 | Its kind of annoying that it rings agents that are on calls |
| 02:06.28 | BSDTech | I have writen a sh*t load of dial stuff and this makes no sense why MusicOnHold() whould not allow a tiem out |
| 02:07.15 | [TK]D-Fender | BSDTech : Ok... fine... sure. We give you official permission to go write a patch for it :) |
| 02:07.20 | BSDTech | <PROTECTED> |
| 02:08.02 | BSDTech | if I new C I would |
| 02:08.13 | BSDTech | I am currently learning php |
| 02:08.17 | JT | lol |
| 02:08.30 | JT | i guess learning c is unlikely then |
| 02:10.41 | PaulTech85 | Cant you use 'WaitMusicOnHold' ? |
| 02:10.43 | PaulTech85 | It has a timeout.. |
| 02:10.48 | k-man | i'm still struggling with jittering audio in voicemail under vmware |
| 02:10.59 | k-man | i read it was an issue with some kernels in the past |
| 02:11.00 | JT | why vmware? |
| 02:11.10 | PaulTech85 | That was at BSDTech |
| 02:11.17 | k-man | jt... um... thats the best testing environment i can come up with right now |
| 02:11.29 | JT | yeah so don't expect it to work well |
| 02:11.30 | gniretar | noone knows the bitrate and sample rate of wav files generated by asterisk? |
| 02:11.36 | JT | try Xen if you want virtualisation |
| 02:11.37 | perd | k-man does it sound like this? http://spunknetwork.com/~bill/badaudio.wav |
| 02:11.40 | k-man | jt, i'm not doing a production system.. its a test system so i can learn how asterisk works before going live |
| 02:11.54 | perd | k-man and are you using wav or ulaw sound files? |
| 02:12.02 | PaulTech85 | let us know if that works BSDTech |
| 02:12.03 | JT | vmware is nowhere near as neat as xen |
| 02:12.16 | perd | can xen emulate ppc? |
| 02:12.17 | wubba | gniretar - 8k mono |
| 02:12.39 | k-man | perd, um kinda like that but about 50 times worse |
| 02:12.43 | BSDTech | that might work |
| 02:12.47 | PaulTech85 | RIFF (little-endian) data, WAVE audio, Microsoft PCM, 16 bit, mono 8000 Hz |
| 02:12.49 | perd | k-man oh :/ |
| 02:12.51 | k-man | perd, no idea what im using |
| 02:12.56 | PaulTech85 | at @perd |
| 02:13.20 | perd | im having a problem similar to yours, when i use wav or ulaw i get jittery sound but when i use the GSM files the audio is fine |
| 02:13.36 | *** join/#asterisk matiasds19 (n=matiasdo@host201.201-252-31.telecom.net.ar) |
| 02:13.49 | perd | kman try putting the gs msound files i nthere just for the hell of it and see if you still have the problem.. i'd be interested to know if someone else was experiencing the same issues i am :/ |
| 02:13.59 | perd | are you using centos btw? i didnt see if you said what os/flavor you were using |
| 02:14.15 | k-man | jt, windows host, linux guest os.. |
| 02:14.23 | k-man | jt, does xen work under win32? |
| 02:15.30 | Qwell | k-man: does anything work under win32? |
| 02:15.40 | BSDTech | WaitMusicOnHold(30) worked |
| 02:15.40 | matiasds19 | someone knows why my asterisk sends an OPTIONS query to my provider with username asterisk? (is not configured like that) |
| 02:15.41 | *** join/#asterisk rsalas_ (n=rsalas@65-122-15-162.dia.static.qwest.net) |
| 02:15.49 | k-man | look, some of us live in the real world where we have to use windows |
| 02:15.53 | k-man | its a fact of life |
| 02:15.57 | wubba | BSD - sweet |
| 02:16.01 | wubba | glad to hear that |
| 02:16.03 | Qwell | k-man: That wasn't my question :) |
| 02:16.17 | perd | k-man hehe as long as you arent going production with it i guess you're ok |
| 02:16.20 | k-man | i hate windows and microsoft as much as anyone else but i'm stuck with it - trolling like that doesn't help anyone Qwell |
| 02:17.42 | *** join/#asterisk etfonhomey (n=etfonhom@74-140-206-4.dhcp.insightbb.com) |
| 02:17.43 | rsalas_ | Hi guys. Question: SIP ATA to FWD dialing 613: negligible delay. Asterisk 1.2, very audible delay. No jitterbuffer. Any ideas? |
| 02:17.53 | *** join/#asterisk azfhasterisk (n=rickb@support.simplicitytechs.com) |
| 02:18.09 | Qwell | k-man: and no, xen is a paravirtualization layer |
| 02:19.24 | k-man | qwell, right... |
| 02:20.01 | christel | Qwell: you should be able to ban bhaal |
| 02:20.15 | christel | as i realised he doesnt have the privs to override them (very few have) |
| 02:20.16 | Qwell | christel: tried - didn't work |
| 02:20.27 | rsalas_ | leave |
| 02:20.28 | Qwell | LoRez did something, I think :p |
| 02:20.36 | christel | ah, perhaps lorez pulled it :) |
| 02:21.01 | Qwell | (s)he is currently banned now though, and (s)he answered and said (s)he would stop |
| 02:21.11 | Qwell | however, I think the other point I had still stands |
| 02:21.18 | christel | i was looking at it yesterday when i returned from scale, and got your email |
| 02:21.30 | Qwell | the whole opt-in thing |
| 02:21.31 | LoRez | christel: he does have the privs to. |
| 02:21.44 | k-man | perd, i'm just using the standard sounds that come with asterisk and I think they are all gsm from what i can tell |
| 02:22.05 | christel | LoRez: hrm, he didnt when i tried to remove them :) |
| 02:22.17 | LoRez | christel: I'd already removed them. |
| 02:22.23 | christel | Qwell: i very much agree, and i believe you raised a very good point wrt policy |
| 02:22.27 | LoRez | otherwise the ban was ineffective. |
| 02:22.39 | k-man | someone should put a fund together to make a set of sounds that are australian |
| 02:22.42 | christel | :) |
| 02:23.07 | christel | LoRez: well then, then i guess lorez pulled it was valid ;P |
| 02:23.45 | Qwell | and fyi, I wasn't trying to be a dick or anything - I totally tried going to "normal" route for several days |
| 02:23.53 | christel | i know you did :) |
| 02:24.05 | christel | and i appreciate that you raised your concerns with us |
| 02:26.27 | *** join/#asterisk IOscanner (n=IOscanne@cpe-76-187-205-211.tx.res.rr.com) |
| 02:26.45 | *** join/#asterisk Kumbang (n=kumbang@router-wmi.paume.ITB.ac.id) |
| 02:27.14 | IOscanner | Is there a way to bind asterisk to 2 different IP address in the sip.conf |
| 02:27.15 | christel | and your email was justified and to the point, made us wake up and smell the coffee :) |
| 02:27.25 | Qwell | christel: I'm a pro at that :P |
| 02:27.43 | IOscanner | I don't want to bind to all, but I need to bind to two. |
| 02:28.06 | christel | *grin*& |
| 02:28.13 | azfhasterisk | Can someone point me to some documentation on how to configure an Asterisk box to do Termination and Origination for the other Asterisk servers? |
| 02:28.48 | CJLinst | Is it possible to send messages to an snom or polycom? Something like "SLACKER LOGGED OFF BECAUSE HE WALKED AWAY FROM HIS PHONE WHILE LOGGED INTO THE SUPPORT QUEUE." Or something like that? Or is it possible to monitor queue login status with a subscription and hint? |
| 02:29.21 | Qwell | CJLinst: You mean like text? |
| 02:29.26 | CJLinst | Sure. |
| 02:29.34 | Qwell | not really |
| 02:30.36 | CJLinst | I'm kind of missing AgentCallbackLogin. Must've been pretty problematic. |
| 02:30.46 | Qwell | CJLinst: extremely - and it was wholely useless |
| 02:31.14 | CJLinst | It's taken me a while to duplicate it with a dialplan but it's getting there. |
| 02:31.27 | Qwell | well, there is the queues.txt in doc/ |
| 02:31.28 | CJLinst | Trouble is no status on the phone if a lumpy gets booted. |
| 02:31.32 | Qwell | it explains it pretty well |
| 02:31.37 | CJLinst | Yeah, I started there. |
| 02:32.02 | BSDTech | thanks its all workign |
| 02:32.08 | BSDTech | this rocks |
| 02:32.12 | BSDTech | much nicer |
| 02:32.13 | CJLinst | I want users to be able to roam so I had to insert their channel ID in a database and pullit back out in the call-agent context. |
| 02:32.30 | matiasds19 | someone from Argentina here? |
| 02:32.42 | *** part/#asterisk BSDTech (n=RNeese@ppp-69-238-49-231.dsl.irvnca.pacbell.net) |
| 02:32.49 | CJLinst | They're used to a Mitel where they get an indicator that illuminates if there's a call waiting and starts blinking if holding more than two minutes. |
| 02:32.50 | *** join/#asterisk putzz (n=me@CPE00155824933c-CM00111ae07ac2.cpe.net.cable.rogers.com) |
| 02:32.50 | *** join/#asterisk Swabby-- (n=dp@74-137-34-4.dhcp.insightbb.com) |
| 02:33.57 | CJLinst | But that only works on local Mitel phones and I'll be damned if I'm going to pay them what they want to go IP. |
| 02:35.10 | Swabby-- | Can you pimp up asterisk on voipyourlife? |
| 02:35.30 | CJLinst | Me? |
| 02:35.38 | perd | asterisk is not something to be pimped |
| 02:36.06 | perd | she's a classy lady who needs to be finessed. show some respect sir. |
| 02:36.11 | Swabby-- | LOL |
| 02:36.30 | Swabby-- | serious though..can you implement Asterisk with existing voip service? |
| 02:36.35 | perd | sure |
| 02:36.45 | perd | anything that allows a normal sip connection should work |
| 02:36.51 | Swabby-- | awesome.. i didn't realize that |
| 02:37.07 | perd | yep, it's pretty slick |
| 02:37.09 | *** join/#asterisk apardo (n=apardo@87.217.144.212) |
| 02:37.19 | Swabby-- | you ever done anything like that for playing with it home use/ |
| 02:37.29 | perd | yea a little |
| 02:37.36 | wubba | Can anyone tell me what conf file hold time condition info? |
| 02:37.46 | perd | voipjet.com is the only provider i can think of off the top of ym head |
| 02:38.55 | perd | they're good for testing stuff |
| 02:38.56 | Swabby-- | i use voipyourlife.com and have been really pleased with it |
| 02:38.56 | perd | kind of expensive though |
| 02:38.56 | Swabby-- | it's my home service |
| 02:38.56 | perd | ah |
| 02:38.56 | Swabby-- | i'm checking that website u sent now |
| 02:38.56 | *** join/#asterisk Dibbler_ (n=Dibbler@dsl-217-155-254-174.zen.co.uk) |
| 02:39.45 | [TK]D-Fender | wubba : clarify "time condition info" |
| 02:40.18 | wubba | My system is reading time conditions that were deleted - and I am not sure why - so I wanted to find the conf that holds that info |
| 02:41.07 | wubba | Goto (timeconditions,4,1) |
| 02:41.24 | wubba | It shouldn't be doing this |
| 02:41.36 | Qwell | I smell freepbx |
| 02:42.01 | perd | you're like a pig searching for truffles |
| 02:42.12 | wubba | Why is that? |
| 02:42.34 | Qwell | because asterisk has no concept of anything called "timeconditions" |
| 02:42.35 | [TK]D-Fender | wubba : Becaus thats clearly fromt he dialplan and you have no clue about the Goto command... |
| 02:43.10 | Swabby-- | What's the difference between freepbx and asterisk |
| 02:43.25 | Qwell | asterisk is a pbx - freepbx is not...ironically |
| 02:43.42 | [TK]D-Fender | wubba : And when you don't even know what your own system is doing its showing the confic came from somewhere else. And where would THAT happn to be... hmmm |
| 02:44.00 | Swabby-- | ah... |
| 02:44.08 | [TK]D-Fender | Qwell : Actually, Asterisk is a telephony toolkit that lets you do things like making a PBX ;) |
| 02:44.46 | perd | you guys are all bitches. just call it what it is: asterisk is a huge pain in the ass that pisses prett ymuch everyone off at some time or another but ends up being sweet once you figure out what the fuck you're doing.. |
| 02:45.11 | [TK]D-Fender | perd : No.... I got ti right the first time ;) |
| 02:45.14 | perd | you can copy/paste that for use on asterisk.org |
| 02:45.26 | perd | haha fender, i need your brain |
| 02:45.34 | wubba | TK - I know what my system is doing - I am trying to figure out why it is reading something that shouldn't be there. |
| 02:45.37 | perd | i broke mine with alcohol :P |
| 02:45.42 | [TK]D-Fender | perd : back off Frankenstein! |
| 02:46.03 | [TK]D-Fender | wubba : If its doing a goto its in your dialplan.. How can you NOT know whats in there? |
| 02:46.49 | perd | wubba one partial line from your dialplan is not enough info to help |
| 02:47.24 | [TK]D-Fender | wubba : Sure, why don't you paste the whole thing for us to see. It could be in multiple places... |
| 02:47.43 | [TK]D-Fender | wubba : And perhaps you can tell us what that context is doing while you're at it... |
| 02:47.45 | [TK]D-Fender | ~pb |
| 02:47.54 | jbot | it has been said that pb is a place to paste your stuff without flooding the channel - try http://pastebin.ca, or http://channels.debian.net/paste |
| 02:47.54 | Swabby-- | Does any website maintain a list of Asterisk friendly voip providers? |
| 02:47.56 | [TK]D-Fender | ^^^ |
| 02:48.00 | Qwell | Swabby--: on the wiki |
| 02:48.01 | Qwell | ~wikis |
| 02:48.03 | jbot | methinks wikis is http://www.voip-info.org |
| 02:48.03 | [TK]D-Fender | Swabby-- : www.voip-info.org |
| 02:48.04 | wubba | Cause I am loking at the CLI and that is where I seen it was being called form. |
| 02:48.38 | Swabby-- | tnx |
| 02:48.45 | [TK]D-Fender | wubba : what you see running through CLI is extensions.conf being executed. So why don't you go pastebin it up like we've asked and maybe we can tell you where to go.... |
| 02:50.57 | perd | dudes!! off topic but i know you're all beeer drinkers: http://www.boston.com/business/ticker/2007/02/sam_adams_unvei_1.html |
| 02:51.12 | perd | state of the art beer glasses, important stuff. |
| 02:52.43 | *** part/#asterisk azfhasterisk (n=rickb@support.simplicitytechs.com) |
| 02:53.20 | Swabby-- | i want one |
| 02:53.54 | J4k3 | I prefer 40 ouncers of malt liquor myself. |
| 02:54.01 | J4k3 | the bottle holds a nice head |
| 02:54.07 | perd | haha |
| 02:54.21 | perd | steel reserve, it's like liquid ecstacy |
| 02:54.47 | J4k3 | http://en.wikipedia.org/wiki/40_ounce |
| 02:54.51 | gniretar | when i'm in a macro am i allowed to do a regex (like exten => _20XX,1,.....) |
| 02:54.56 | *** join/#asterisk Snake-Eyes (n=blog@203.220.55.70) |
| 02:55.23 | gniretar | and, can i have a context call a macro, have that macro return then ahve the context continue? |
| 02:55.41 | ping2921 | whenever queue announcements are played, I have the moh start from beginning, anyway to fix this? |
| 02:55.47 | gniretar | exten =>100,1,marco(docrap) |
| 02:56.03 | Swabby-- | Is there an easy way to play with asterisk without having alot of expensive equipment |
| 02:56.06 | Swabby-- | like a simulated environment |
| 02:56.09 | Swabby-- | or do you HAVE to buy the cards |
| 02:56.12 | gniretar | exten =>100,2,dial(zap/5) |
| 02:56.23 | gniretar | can i get domething like that to work? |
| 02:56.29 | Qwell | Swabby--: no, you don't need to buy anything |
| 02:56.33 | [TK]D-Fender | gniretar : Yes, go read up on Macro's |
| 02:56.56 | [TK]D-Fender | Swabby-- : Got a reasonably computer wunnin *nix? DONE. |
| 02:57.08 | Swabby-- | TK: What about hte controller cards and such? |
| 02:57.11 | *** join/#asterisk topping (n=topping@204.152.96.238) |
| 02:57.15 | hematitec | ping: have you tried a 'random' or a streaming music on hold? |
| 02:57.20 | [TK]D-Fender | Swabby-- : You only need cards if you want to take in actual lines of some kind |
| 02:57.31 | perd | swabby you could set asterisk up in vmware or even on a pentium 2 |
| 02:57.32 | [TK]D-Fender | Swabby-- : What "controller" cards? |
| 02:57.41 | Swabby-- | TK: like for the phones |
| 02:57.47 | Swabby-- | so i can simulate voicemail and such |
| 02:58.17 | perd | you would need FXS to use an analog phone, and FXO to connect asterisk to a phone line in your house |
| 02:58.21 | [TK]D-Fender | Swabby-- : You can run * without any kind of special hardware at all. Now if you WANT to use a regular phone, you clearly can jsut plug it into a USB port now can you? THATS when you need hardware. |
| 02:58.39 | [TK]D-Fender | Swabby-- : Today's key-word is SOFT-PHONE <- |
| 02:58.55 | Swabby-- | TK: aahhhh..so basically i could setup like 3-4 softphones |
| 02:58.57 | [TK]D-Fender | Swabby-- : Go download one for whatever platfor you're on and get cracking.... |
| 02:58.58 | perd | not to be mistaken with soft-porn, which is not nearly as cool |
| 02:58.59 | Swabby-- | and simulate it |
| 02:59.07 | Swabby-- | that |
| 02:59.21 | JT | k-man: there are already a number of australian asterisk prompts collections |
| 02:59.21 | Swabby-- | That's awesome... |
| 02:59.26 | perd | x-lite is a good sip client. |
| 02:59.27 | bkruse_home | yep |
| 02:59.52 | JT | k-man: also, it's really not logical to expect asterisk under linux under vmware under windows to operate correctly |
| 03:00.00 | [TK]D-Fender | Swabby-- : Guess ti depends on what you think you're simulating. mean if you can use a soft-phone to place calls, receive calls and all that, how is that a SIMULATION? Its just not doing it through a NICE normally phone like piece of hardware. |
| 03:00.14 | JT | k-man: most people have at least a spare old pc to chuck linux on |
| 03:00.15 | Swabby-- | my goal is to setup an asterisk system for Big Brother Big sister in town here..i already do pro-bono computer work for them..so ifigured i could guenny pig a phone system setup |
| 03:00.25 | k-man | jt, yeah, i do have one... |
| 03:00.33 | k-man | jt, but its really old |
| 03:00.40 | k-man | jt, but works fine as my mailserver |
| 03:00.56 | perd | have any of you heard of people having problems with the playback of wav/ulaw sound files? i have sound issues with playback, especially in the voicemail menu, if i use any format other than GSM. all my sip clients are using ulaw... |
| 03:01.08 | Swabby-- | TK: if i had 3 pcs on the same network could i make one softphone one ext, and another one another ext..is this possible without additional equipment. When you say hooking up phone via usb do you just get a converter for that? |
| 03:01.14 | k-man | how much cpu would it need for managing 1-2 calls and voicemail? |
| 03:01.27 | perd | k-man 400mhz? heh |
| 03:01.28 | *** join/#asterisk Hmmhesays (n=Neg@24-117-135-28.cpe.cableone.net) |
| 03:01.31 | perd | you wouldnt need much at all |
| 03:01.33 | Swabby-- | k-man: I wouldn't think much based on my reading.like 450 mhz |
| 03:01.44 | [TK]D-Fender | Swabby-- : I was making a point by saying you can't jsut plug an extra boring old phone RIGHT INTO YOUR PC for use with * without special hardware.... |
| 03:01.53 | perd | in fact, my linksys router with openwrt can handle that, k-man |
| 03:01.56 | Swabby-- | TK: ahh..i understand :) |
| 03:02.02 | [TK]D-Fender | Swabby-- : You can just give the a bunch of cheap PC's and install softphone's on each. |
| 03:02.06 | k-man | its 467mhz |
| 03:02.13 | k-man | hmmm |
| 03:02.41 | [TK]D-Fender | k-man : taht would do if you stick to ULAW. |
| 03:02.55 | perd | k-man you could spend 150 bucks and get a WRAP board from pc engines |
| 03:02.55 | Swabby-- | TK: So..let's say I want to run 3 analog lines with 8 ext..i would need to buy the appropiate card that supports this for the PC (probably about $1300.00 based on my pricing)...wire the extensions to the pc, wire the analong lines to the pc and configure it |
| 03:03.47 | [TK]D-Fender | Swabby-- : You would need a card more in the $300 range to take in 3 lines, and the PC would be 500$ TOPS. |
| 03:03.59 | [TK]D-Fender | Swabby--: I'm sure you could do less... |
| 03:04.03 | perd | with a WRAP board you wouldnt have to have a big old pc sucking power up like a mofo |
| 03:04.05 | [TK]D-Fender | $300 should be easy |
| 03:04.15 | Swabby-- | T |
| 03:04.16 | Qwell | [TK]D-Fender: plus fxs, or SIP phones |
| 03:04.20 | Swabby-- | TK: Really? WOW |
| 03:04.44 | perd | analog phones are so 80s |
| 03:04.49 | [TK]D-Fender | Qwell : No we're still talking soft-phones here. And for FXS I'd recomment SPA-2002's @ 70$ for 2 ports each |
| 03:04.57 | perd | bluetooth headset + sip :P |
| 03:05.12 | Qwell | bluetooth headset + chan_whateverthehellweendupcallingit |
| 03:05.12 | [TK]D-Fender | perd : Funny think is few people really need more than an analog phone can offer. |
| 03:05.27 | Swabby-- | Two questions on softphone 1. Do you just install on the client PCs, and specify the name of the asterisk server and login info 2, do they talk via a microphone or? |
| 03:05.32 | perd | oh you guys have a channel for bluetooth devices? i figured they used sip.. |
| 03:05.40 | Qwell | perd: soon, hopefully |
| 03:05.44 | perd | man that's awesome. |
| 03:05.58 | Qwell | currently it only supports one end, but...meh |
| 03:05.58 | perd | very exciting |
| 03:05.58 | Qwell | I'd like to add headset support for it |
| 03:05.58 | [TK]D-Fender | Swabby-- : yes on both counts. You'd buy them a 7$ headset for sanitys sake |
| 03:06.06 | Qwell | "I'd like"....heh |
| 03:06.09 | perd | yeah if i could put bluetooth hotspots in my office here and have headsets that could roam around |
| 03:06.10 | perd | oh man |
| 03:06.31 | perd | i dont know what i would do, but it would be awesome. |
| 03:06.34 | Qwell | well, it wouldn't work quite like that, unless you had some sort of...I don't know...bluetooth repeater |
| 03:06.59 | Swabby-- | TK: nice.... ok..my understanding was I needed a Digium Wildcard with 8FXS / 8FXO to do 3 incoming ANALOG lines and 8 extensions |
| 03:07.15 | Swabby-- | with a breakout box.. |
| 03:07.25 | perd | oh, well that's not so cool anymore. maybe 802.11b headset would be better |
| 03:07.36 | Qwell | perd: no it wouldn't :p |
| 03:07.36 | perd | with built in sip client and scripting capabilities |
| 03:07.46 | Swabby-- | This is such a sweet channel..i've learned so much additional info about asterisk already.. |
| 03:07.51 | *** join/#asterisk fx0 (n=fx0@cypher.punk.net) |
| 03:07.53 | Qwell | but, you know |
| 03:07.59 | JT | Swabby--: you'd need 3FXO and 8 FXS to do it all analogue |
| 03:08.00 | Qwell | I guess you *could* have "hotspots" |
| 03:08.05 | Qwell | they'd be really small though |
| 03:08.25 | Swabby-- | JT: gotcha |
| 03:08.34 | perd | i need a headset that can script things like 'two taps dial this extension, three dial this, etc' |
| 03:08.47 | Swabby-- | perd: Vista does that |
| 03:08.47 | perd | instead of 'tap to answer, tap to hang up' boring. |
| 03:08.48 | Swabby-- | heh |
| 03:08.55 | Swabby-- | maybe |
| 03:08.59 | Swabby-- | it has voice recog. |
| 03:08.59 | JT | Swabby--: you don't need to use analogue extensions unless you want to |
| 03:09.00 | perd | swabby not with my plantronics headsets |
| 03:09.19 | umop3plsdn | speaking of headsets... i need some durable ones any suggestions? |
| 03:09.28 | perd | my plantronics headsets are great |
| 03:09.33 | Swabby-- | JT: so..if i really wanted to save money..give everyone soft phones...and then all i would need is 3FXO? |
| 03:09.43 | perd | CS351 with lifter |
| 03:09.54 | perd | good sound, comfortable |
| 03:09.56 | JT | Swabby--: depends on the application, if for an office, i'd suggest SIP phones for the users |
| 03:10.05 | perd | they do pick up a lot of background noise if you dont have the gains set well, though |
| 03:10.23 | Swabby-- | SIP phones are softphones also? |
| 03:11.02 | [TK]D-Fender | Swabby-- : SIP is a VoIP protocol. When we say "SIP Phones" we mean a "hardware" phone that talks SIP and plugs into your LAN |
| 03:11.22 | [TK]D-Fender | Swabby-- : Looks like a normal business phone. |
| 03:11.35 | perd | only more prone to failure! |
| 03:11.37 | perd | hehe |
| 03:11.49 | Swabby-- | TK: Gotcha |
| 03:11.51 | Swabby-- | makes snse |
| 03:12.00 | Swabby-- | SIP phone gets an IP and everything then |
| 03:12.09 | *** join/#asterisk fnordus (n=dnall@24.85.128.203) |
| 03:12.30 | russellb | SIP phones are more complicated to configure than you would have ever imagined a phone could be |
| 03:12.32 | JT | Swabby--: it's a hard phone technically |
| 03:12.38 | JT | while it may have software inside |
| 03:12.52 | JT | firmware really |
| 03:13.15 | Swabby-- | You folks ever buy anything from voipsupply.com? |
| 03:13.19 | Swabby-- | seems like they've got alot of good equipment |
| 03:13.35 | perd | russel i dunno, i have only dealt with cisco phones and they werent THAT bad... pain in the ass for a bit but once you figure out what needs to be in the 5 config files it's cake :P |
| 03:13.51 | russellb | "in the 5 config files" |
| 03:13.53 | perd | they suck though. |
| 03:21.30 | *** join/#asterisk KaiHanari (n=kaihanar@stjhnf0122w-142162043055.pppoe-dynamic.nl.aliant.net) |
| 03:22.07 | [TK]D-Fender | Swabby-- : Places to look for pricing : www.atacomm.com , www.telephonydepot.com , www.voipsupply.com . Best 3 places. |
| 03:25.04 | *** join/#asterisk etfonhomey (n=etfonhom@74-140-206-4.dhcp.insightbb.com) |
| 03:25.21 | *** join/#asterisk r0d3nt (i=nobody@foster.stonedcoder.org) |
| 03:25.55 | *** join/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net) |
| 03:25.55 | *** mode/#asterisk [+o mog] by ChanServ |
| 03:29.04 | *** join/#asterisk mtgh (n=chatzill@dsl093-001-038.det1.dsl.speakeasy.net) |
| 03:30.49 | *** join/#asterisk edison\ (n=edison@c-67-176-75-179.hsd1.co.comcast.net) |
| 03:31.16 | *** part/#asterisk mtgh (n=chatzill@dsl093-001-038.det1.dsl.speakeasy.net) |
| 03:31.20 | *** join/#asterisk mtgh (n=chatzill@dsl093-001-038.det1.dsl.speakeasy.net) |
| 03:32.17 | IOscanner | how can I bind two ip addresses to sip .conf |
| 03:33.10 | IOscanner | If I bind all asterisk wants to use the IP address of that interface |
| 03:33.20 | IOscanner | I just need to bind asterisk to both the internal IP and the extrnal IP |
| 03:33.39 | bkruse_home | just bind to 0.0.0.0 to listen to all no |
| 03:33.39 | bkruse_home | ? |
| 03:33.52 | bkruse_home | bind it to 0.0.0.0 |
| 03:33.52 | IOscanner | I don't want to listen to all |
| 03:34.08 | perd | is the external ip on a router of some sort? |
| 03:34.08 | IOscanner | I need it to listen to 192.168.47.253 and the public IP. |
| 03:34.14 | perd | are you natted? |
| 03:34.29 | IOscanner | no the external IP is on the obx |
| 03:34.31 | IOscanner | box |
| 03:34.38 | IOscanner | no natted for the main server. |
| 03:34.56 | bkruse_home | why not listen to all? |
| 03:34.56 | bkruse_home | is my question |
| 03:35.15 | IOscanner | but when I send traffic down an openvpn trunk asterisk uses the VPN ip |
| 03:36.29 | eald | which is your source of documentation for asterisk? I'm the only one who thinks that the wiki information is outdated? |
| 03:37.12 | JT | sometimes it is |
| 03:37.14 | JT | ~thebook |
| 03:37.16 | jbot | somebody said thebook was a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 03:37.21 | JT | sometimes the book is out of date |
| 03:37.39 | IOscanner | So no way to tell asterisk not to use an IP when binding to all |
| 03:38.24 | perd | ~telephony101 |
| 03:38.27 | JT | what is the ip it's binding to that you don't want it to? |
| 03:38.28 | perd | blarg |
| 03:38.42 | JT | what is the ip for? |
| 03:38.50 | perd | you could just filter the port with iptables |
| 03:39.04 | IOscanner | 10.0.0.1 |
| 03:39.08 | JT | i was thinking that |
| 03:39.11 | fx0 | learn more about linux and try to spend some minutes on policy based routing. |
| 03:39.13 | JT | yeah but what is it for, IOscanner |
| 03:39.39 | perd | it's the super horny porn SANS |
| 03:39.40 | eald | I'm reading the book, but knowing that is a book, I thunk that was outdated |
| 03:39.44 | JT | seems more like a firewalling than routing issue |
| 03:39.47 | *** join/#asterisk putzz (n=me@CPE00155824933c-CM00111ae07ac2.cpe.net.cable.rogers.com) |
| 03:39.55 | JT | eald: most is still relevan |
| 03:39.56 | IOscanner | asterisk keeps using 10.0.0.1 instead of its 192.168.47.253 address |
| 03:39.56 | JT | t |
| 03:40.06 | IOscanner | if I bind it to the internal IP it works fine |
| 03:40.21 | IOscanner | But then it breaks the external IP traffic |
| 03:40.31 | JT | err isn't this a sip.conf issue? you can tell it what ips to use |
| 03:40.41 | JT | what breaks external ip traffic? |
| 03:41.09 | IOscanner | when I bind sip.conf to the internal IP. I need sip to only bind to the internal ip and the external (public ip) |
| 03:41.15 | IOscanner | if I can do this it would work fine. |
| 03:41.57 | JT | what happens if you bind it to 0.0.0.0 and set externip to your public address? |
| 03:42.25 | IOscanner | if I bind to all it works, but asterisk keeps using the openvpn interface 10.0.0.1 |
| 03:42.50 | JT | it works, or does not work, that is the question |
| 03:43.05 | KaiHanari | is there a way to have an ivr play a dialtone instead of "press this to do this"? other than recording a dialtone and having it play back? |
| 03:43.23 | JT | KaiHanari: DISA |
| 03:43.35 | KaiHanari | whats DISA? |
| 03:43.45 | JT | the answer to your question, look it up |
| 03:43.52 | JT | it's in the book and wiki |
| 03:44.01 | KaiHanari | is there a site with manuals and examples? |
| 03:44.01 | IOscanner | no it does not work |
| 03:44.10 | JT | KaiHanari: |
| 03:44.13 | JT | ~thebook |
| 03:44.14 | jbot | methinks thebook is a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 03:44.16 | JT | ~thewiki |
| 03:44.19 | jbot | i guess thewiki is at http://www.voip-info.org/wiki-Asterisk |
| 03:44.19 | perd | Dont Indulge Selfish Assholes |
| 03:44.19 | perd | DISA. |
| 03:44.23 | JT | haha |
| 03:44.36 | JT | KaiHanari: also google |
| 03:45.00 | *** join/#asterisk andrew` (i=andrew@69-12-140-101.dsl.dynamic.sonic.net) |
| 03:45.05 | JT | IOscanner: ah ok, you said it works but uses 10.0.0.1 |
| 03:45.17 | KaiHanari | JT. You have no reason to be an ass about it. Ever hear of a thing called lag? Im on wireless right now so i get bad latency at times. So i didnt see what you said right away. |
| 03:45.52 | IOscanner | but it when it talks to the phones it uses the 10.0.0.1 |
| 03:46.04 | [TK]D-Fender | kaldemar : "show application disa" |
| 03:46.13 | JT | KaiHanari: umm, i'm hardly being an arse, yes i know what lag is, i'm getting som now, but the point is, you saw my answer but instead of searching for it, questioned it like i was trying to trick you |
| 03:46.21 | JT | it's a really simple application to be honest |
| 03:46.26 | JT | and probably does exactly what you want |
| 03:46.30 | IOscanner | I need the phone to use 192.168.47.253 when internal and the public IP (which is on the box) when public |
| 03:47.01 | [TK]D-Fender | IOscanner : Phones don't typicall gt to be smart about networking like that.... |
| 03:47.26 | edison\ | anyone have a clue as to why I'm getting: No user '800-555-1212' in SIP users list? |
| 03:47.27 | [TK]D-Fender | IOscanner : You can typically give it 1 server address and thats it. |
| 03:47.27 | IOscanner | Asterisk is the one using 10.0.0.1 the phones are using the correct ip |
| 03:47.34 | JT | IOscanner: so as a result, the phones cannot communicate? |
| 03:47.42 | *** join/#asterisk i3inary (i=i3inary@ip68-8-91-87.sd.sd.cox.net) |
| 03:47.48 | edison\ | the 8005551212 is the calling party's number |
| 03:48.00 | [TK]D-Fender | edison\ : becasue it means exactly what it says. go look in sip.conf and pastebin it for us if it exists.... |
| 03:48.01 | JT | IOscanner: i'm thinking iptables is the easiest way to do it, but i dunno |
| 03:48.15 | IOscanner | I did that, but it doesn't fix the SIP traffic |
| 03:48.16 | [TK]D-Fender | edison\ : Right now this soulds like you're calling Dial wrong. |
| 03:48.31 | JT | IOscanner: yeah it might be unsupported in asterisk, i dunno |
| 03:48.47 | edison\ | 'cuse my ignorace - pastebin? |
| 03:48.59 | perd | pastebin.ca |
| 03:49.05 | [TK]D-Fender | ~pb |
| 03:49.06 | jbot | [pb] a place to paste your stuff without flooding the channel - try http://pastebin.ca, or http://channels.debian.net/paste |
| 03:49.06 | KaiHanari | IOscanner, look into your router. Some can do static IP assigning with the DHCP. I use this with my d-link. DHCP has the MAC address of a device, and if that device connects, it gets the ip assigned to it every time. |
| 03:49.07 | perd | sho nuff, time to go home, later |
| 03:49.25 | IOscanner | No it is static on the server (with asterisk) |
| 03:49.42 | JT | KaiHanari: that won't actually help i think, he needs asterisk to bind to 2 ips, and use them as the relevant source addresses in the SIP SDPs |
| 03:49.49 | KaiHanari | Ah |
| 03:50.48 | IOscanner | JT: correct |
| 03:51.07 | IOscanner | the public IP receives anonymous traffic from the PSTN |
| 03:51.24 | IOscanner | if I bind to the private IP I can't get calls |
| 03:51.36 | IOscanner | If I bind to the public IP I can't use the system. |
| 03:52.02 | *** join/#asterisk bmg505 (n=leon@c1-154-16.rndf.isadsl.co.za) |
| 03:52.40 | edison\ | [TK]D-Fender: it was working on 1.4 release before nabbing the latest SVN |
| 03:52.41 | IOscanner | If I bind all I can't have remote at have openvpn links between the locations, because asterisk wants to use the IP address of the VPN tunnel to that location. |
| 03:52.50 | edison\ | that's what flummoxes me. |
| 03:53.16 | [TK]D-Fender | edison\ : hopefully you are realizing you are still just talking about your problem and not showing us anything that can let us help you... |
| 03:54.50 | edison\ | [TK]D-Fender: true, unless someone here was aware of some configs that have changed since 1.4.0 and SVN. |
| 03:55.25 | [TK]D-Fender | edison\ : And we have absolutulety no idea what you are doing, or how you are doing it. Please show us something useful. |
| 03:55.31 | edison\ | [TK]D-Fender: but what should exist in sip.conf that you're looking for? |
| 03:55.46 | edison\ | I can alreayd tell you the number doesn't exist. |
| 03:56.15 | eald | a month or so I had an asterisk with two cards trying to substitute two servers, so I think that was pretty easy, but after many troubles it wasn't, but in that time I found this: http://lists.digium.com/pipermail/asterisk-dev/2004-July/005211.html is old, but maybe it says something interesting... are your routes in the asterisk box perfectly clear? |
| 03:56.33 | [TK]D-Fender | edison\ : your description of the error is quite lacking. Please pastebing the entire CLI output of the call and show us your configs. |
| 03:56.38 | [TK]D-Fender | ~pb |
| 03:56.40 | jbot | extra, extra, read all about it, pb is a place to paste your stuff without flooding the channel - try http://pastebin.ca, or http://channels.debian.net/paste |
| 04:01.36 | *** join/#asterisk L|NUX (n=linux@unaffiliated/lnux/x-10290) |
| 04:03.32 | *** part/#asterisk loufoque (n=loufoque@smj33-1-82-233-64-48.fbx.proxad.net) |
| 04:03.55 | edison\ | http://pastebin.ca/354766 |
| 04:04.28 | eald | IOscanner, http://mailman.ds9a.nl/pipermail/lartc/2006q4/019958.html this seems the solution to your problem and is what JT said |
| 04:05.18 | [TK]D-Fender | edison\ : [20:54] <[TK]D-Fender> What you do is dial Local with and Absolute timeout, and the when it expires the call will fall back <- |
| 04:05.22 | [TK]D-Fender | # |
| 04:05.22 | [TK]D-Fender | No user '8015551212' in SIP users list |
| 04:05.22 | [TK]D-Fender | # |
| 04:05.22 | [TK]D-Fender | No matching peer for '8015551212' from '64.2.142.27:5060' |
| 04:05.24 | [TK]D-Fender | oops. |
| 04:05.39 | [TK]D-Fender | So evidently you need to have a user set up with the DID as the name of the context |
| 04:06.14 | edison\ | 8015551212 is the CALLING PARTY'S number, NOT the DID. |
| 04:06.29 | *** join/#asterisk CunningPike (n=CunningP@S010600095b33697f.vc.shawcable.net) |
| 04:07.54 | edison\ | obviously something's wrong if * expects me to have a sip entry for every number that calls. |
| 04:08.43 | edison\ | and it's something that has changed since 1.4.0 and yesterday's SVN |
| 04:08.59 | edison\ | this config worked perfectly on 1.4.0. |
| 04:09.12 | edison\ | unfortunately, BLFs didn't, hence the upgrade to SVN |
| 04:12.37 | [TK]D-Fender | Ah, ok, looking again |
| 04:14.48 | *** join/#asterisk gniretar_home (n=gniretar@24-247-81-221.dhcp.bycy.mi.charter.com) |
| 04:15.03 | gniretar_home | hey all |
| 04:15.24 | gniretar_home | what kinda config errors would cause one way communication over an IAX channel? |
| 04:15.50 | edison\ | [TK]D-Fender thanks. |
| 04:15.58 | [TK]D-Fender | edison\ : Sing into their site and take alook for config samples under "user portal" |
| 04:16.02 | [TK]D-Fender | sign* |
| 04:17.34 | gniretar_home | I dont understand this line (its taken from VOIP-info) |
| 04:17.36 | gniretar_home | If one side has trunk=yes and the other does not cannot validate the peer, you will get one-way audio. The easiest way to make this work is to use a register line to register to the systems you want to trunk with. |
| 04:17.54 | gniretar_home | 'the other does not cannot validate' |
| 04:17.55 | gniretar_home | ? |
| 04:18.11 | gniretar_home | can someone who undertands what this statement is trying to say please help me out |
| 04:18.41 | JT | do you have trunk=yes? |
| 04:19.23 | gniretar_home | yes |
| 04:19.29 | JT | why? |
| 04:19.30 | *** join/#asterisk andrew` (i=andrew@69-12-140-101.dsl.dynamic.sonic.net) |
| 04:19.31 | *** join/#asterisk mtgh (n=chatzill@dsl093-001-038.det1.dsl.speakeasy.net) |
| 04:19.32 | gniretar_home | this is a trunk between 2 servers |
| 04:19.33 | JT | do you need it? |
| 04:19.35 | JT | hmm |
| 04:19.41 | gniretar_home | i'm getting one way audio |
| 04:20.30 | JT | sounds like an auth problem or similar |
| 04:22.00 | edison\ | [TK]D-Fender: already did. That's what was working before this SVN |
| 04:22.28 | gniretar_home | JT: the entried in iax.conf for both servers are identical except for the host |
| 04:22.41 | gniretar_home | entries** |
| 04:22.47 | JT | have you done any iax debugging? |
| 04:23.01 | *** join/#asterisk andrew` (i=andrew@69-12-140-101.dsl.dynamic.sonic.net) |
| 04:23.02 | gniretar_home | you mean from the CLI? |
| 04:23.07 | JT | yes |
| 04:23.10 | gniretar_home | no, i got kicked ut of the lab :-/ |
| 04:23.20 | JT | huh? |
| 04:23.40 | JT | right |
| 04:23.51 | gniretar_home | http://www.voip-info.org/wiki-Asterisk+config+iax.conf |
| 04:23.53 | gniretar_home | in that page |
| 04:23.58 | gniretar_home | it has a register line |
| 04:24.02 | gniretar_home | should i have that? |
| 04:24.25 | JT | shouldn't be necessary, but you can put it in just to test |
| 04:25.28 | [TK]D-Fender | edison\ : Ok, if thre isn' anything more to see just to double check, then I'm fresh out of ideas, but at least now we've got something for someone else to pick up the ball on. |
| 04:26.12 | edison\ | that's it... |
| 04:26.16 | edison\ | I'm stumped. |
| 04:26.47 | *** join/#asterisk mtaht4 (n=m@c-71-198-23-124.hsd1.ca.comcast.net) |
| 04:26.48 | edison\ | of course the guys in -dev didn't care to help because I'm a user. |
| 04:27.06 | edison\ | I hate to submit a bug if it's a simple fix. |
| 04:28.01 | [TK]D-Fender | edison\ : Well it took a while to get this pastebin out, but now you've got something to show them. Your odds are much higher |
| 04:28.02 | gniretar_home | JT: http://pastebin.ca/354804 |
| 04:28.07 | gniretar_home | does anything jump out at you? |
| 04:28.34 | edison\ | [TK]D-Fender: thanks for taking a look. |
| 04:29.19 | JT | gniretar_home: insecure=very might help |
| 04:29.21 | JT | umm |
| 04:29.37 | JT | gniretar_home: do either machine have zaptel hardware? |
| 04:29.55 | gniretar_home | JT: Nope but i have ztdummy loaded on both |
| 04:30.05 | JT | not sure if ztdummy would be good enough |
| 04:30.08 | *** join/#asterisk pigpen (n=mark@fw.seamans.cc) |
| 04:30.26 | JT | what are your zttest scores like? |
| 04:30.27 | gniretar_home | the wierd part is that i had this working at one point |
| 04:30.30 | *** join/#asterisk [Latre] (n=chatzill@189.153.85.243) |
| 04:30.32 | gniretar_home | then it borke |
| 04:30.37 | JT | hmm |
| 04:30.49 | gniretar_home | how do i use zttest? |
| 04:30.57 | JT | you type in the command and hit enter |
| 04:31.48 | gniretar_home | 99.92xxxxxxx |
| 04:31.58 | gniretar_home | Best: 99.975586 -- Worst: 99.926758 -- Average: 99.955027 |
| 04:32.04 | JT | pretty crappy |
| 04:32.06 | gniretar_home | 19 passes |
| 04:32.24 | JT | 99.97 should be the minimum |
| 04:32.37 | J4k3 | do you have actual zaptel hardware? |
| 04:32.37 | JT | at least it is for zaptel hardware to not start playing up |
| 04:32.45 | JT | asume iax2 trunking would be similar |
| 04:32.51 | JT | J4k3: no, he only has ztdummy |
| 04:32.53 | J4k3 | ah |
| 04:33.01 | gniretar_home | Best: 99.963379 -- Worst: 99.938965 -- Average: 99.953050 |
| 04:33.04 | gniretar_home | thats on the other box |
| 04:33.13 | J4k3 | well those are about the same numbers I got on mine against ztdummy |
| 04:33.13 | JT | gniretar_home: yeah not god |
| 04:33.16 | JT | good |
| 04:33.22 | *** join/#asterisk DJS_2_6 (n=djstillm@cpe-066-057-115-255.nc.res.rr.com) |
| 04:33.32 | JT | gniretar_home: maybe you'll just have to disable trunking |
| 04:34.16 | gniretar_home | OK, well i should be able tog et away with that. the overhead is minimal |
| 04:34.16 | gniretar_home | hmmm, i'm trying to think about what changed |
| 04:34.20 | gniretar_home | when it was working chan_zap want loaded on either |
| 04:34.23 | JT | yeah, you're on a lawn :P |
| 04:34.25 | gniretar_home | now its loaded on both |
| 04:34.30 | JT | hmm |
| 04:34.38 | gniretar_home | or its loaded on one of them |
| 04:34.53 | JT | gniretar_home: are the versions all the same? |
| 04:35.02 | gniretar_home | no, one is 1.2.15 and the other is 1.4 |
| 04:35.36 | JT | is that the setup you had when it worked? |
| 04:35.40 | gniretar_home | yes |
| 04:36.42 | wunderkin | with ztdummy the scores are not going to come out the same, don't use zttest for it |
| 04:37.08 | gniretar_home | k |
| 04:37.14 | JT | wunderkin: that's because zaptel accuracy isn't good with ztdummy |
| 04:37.58 | gniretar_home | well, i'm using MeetMe on one of these |
| 04:38.02 | gniretar_home | so i hope it works |
| 04:38.08 | JT | arrgh |
| 04:38.10 | JT | meetme too |
| 04:38.14 | JT | you're insane :P |
| 04:38.19 | gniretar_home | well |
| 04:38.31 | wunderkin | no, its because of using 1024 instead of 1000, some stuff like that |
| 04:38.43 | gniretar_home | http://hosted.wccnet.edu/ccdc/ |
| 04:38.47 | gniretar_home | i'm part of that |
| 04:38.58 | gniretar_home | and if you look at the topology at the botton there is an asterisk server |
| 04:39.04 | JT | iax2 trunking and meetme are the two things outside of zap hardware that require accurate zap timing |
| 04:39.14 | *** join/#asterisk fatpelt (n=pat@mail.feltonline.com) |
| 04:39.23 | gniretar_home | i'm building the core asterisk server for a hub and spoke topology of 8 servers |
| 04:41.40 | fatpelt | hey all. i was reading somewhere a month or so ago (but i can't find it now) about long distance and restricting who can dial. what is the best way limit outbound long distance calls to one iax user ? |
| 04:41.40 | gniretar_home | hmmmm, perhaps i can get my hands on a TDM400P card |
| 04:41.40 | JT | fatpelt: different contexts in the dialplan |
| 04:41.41 | gniretar_home | just for the core server |
| 04:41.41 | JT | gniretar_home: there's an alternate conference app too |
| 04:41.41 | JT | that doesn't need zap timing |
| 04:41.41 | gniretar_home | which one? |
| 04:41.41 | JT | app_conference |
| 04:41.41 | gniretar_home | is it as good at meetme? |
| 04:42.06 | fatpelt | JT: so i change the iax user to default to the context that can dial out, but then do i have to duplicate all the other context's information ? |
| 04:42.23 | JT | gniretar_home: not sure |
| 04:42.24 | gniretar_home | eh, i'l see fi i can get the TDM card |
| 04:42.27 | gniretar_home | if* |
| 04:43.01 | [Latre] | hi people......in asterisk 1.2.15 must config something for better echo cancel ??? |
| 04:43.22 | JT | fatpelt: err what? |
| 04:43.52 | fatpelt | JT: basically the question is how do i nest contexts so that the one iax user that can dial out can access my other context ? |
| 04:44.43 | JT | fatpelt: you can include the other context with an include directive |
| 04:45.10 | [Latre] | there is some of configuration for the new release about echo cancell ? |
| 04:48.17 | *** join/#asterisk thoughtpolice (n=austin@ip70-185-140-61.lu.dl.cox.net) |
| 04:48.34 | JT | heh alright |
| 04:48.38 | JT | was it the book? |
| 04:49.38 | fatpelt | JT: http://www.voip-info.org/wiki/index.php?page=Asterisk+security+dialplan |
| 04:49.42 | *** join/#asterisk DaveCanoe (n=Dave@canoe404.dclg.ca) |
| 04:50.08 | JT | ah ok |
| 04:56.49 | *** join/#asterisk gerphimum (n=trekkie@207.190.58.85) |
| 05:01.46 | *** join/#asterisk aptura (n=jondoe@S010600a0c93f6f7e.vs.shawcable.net) |
| 05:03.04 | aptura | who has punched down with a 66 tool. I have done my share with the 110 but not a 66. Seems a nice hot job came up requiring this skill in addition to other requirments. |
| 05:03.11 | *** join/#asterisk cnbrk (n=asd@senolsun-2.user.msu.edu) |
| 05:03.35 | Strom_C | 66 is easy |
| 05:03.37 | cnbrk | hi guys |
| 05:03.41 | cnbrk | i have a question |
| 05:03.50 | cnbrk | how can i connect my asterisk server to another sip service provider |
| 05:04.10 | cnbrk | what lines should i add to my sip.conf |
| 05:04.11 | *** join/#asterisk ckck (n=ck@cm86.sigma59.maxonline.com.sg) |
| 05:04.37 | Strom_C | cnbrk: that will depend on your sip service provider's configuration |
| 05:04.56 | cnbrk | well, actually i can figure that out if you can just give me some brief info |
| 05:05.06 | cnbrk | i have a username and password from other service provider |
| 05:05.19 | cnbrk | and let's say numbers frtom 2000-3000 are on that service provider |
| 05:05.32 | aptura | Strom it probebly is. |
| 05:05.34 | cnbrk | i want asterisk to forward my calls to that provider when i call 2500 from my phone |
| 05:05.46 | aptura | What about extention t1 to a new demark point. |
| 05:05.50 | aptura | extending |
| 05:06.01 | cnbrk | how |
| 05:06.17 | aptura | Let me look up the discription cnbrk |
| 05:06.19 | Strom_C | aptura: well, that's more of a wiring issue than a straight 66 block issue |
| 05:06.28 | Strom_C | cnbrk: are you new to asterisk? |
| 05:06.59 | cnbrk | yes |
| 05:07.07 | Strom_C | I suggest you read the book |
| 05:07.10 | Strom_C | ~book |
| 05:07.23 | jbot | [book] a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 05:11.33 | *** join/#asterisk cnbrk (n=asd@senolsun-2.user.msu.edu) |
| 05:11.39 | cnbrk | hi again sorry lost connection |
| 05:11.45 | cnbrk | so what do u expect me to do fasrt |
| 05:11.46 | cnbrk | fasrt |
| 05:11.52 | cnbrk | fast, without reading a book |
| 05:12.35 | flenders | guys, sorry I asked a few times, but so far, haven't got any replies... can you please have a look at http://www.pastebin.ca/354865 in regards to my sip problems? |
| 05:14.32 | Strom_C | cnbrk: hire a consultant :) |
| 05:14.36 | pigpen | Could anyone point me in the direction to find a solution to save dtmf digits to a db value? |
| 05:15.06 | pigpen | I haven't done this one...and google is sucking on my queries... |
| 05:15.20 | CunningPike | pigpen: Read() followed by DBPut()? (or the modern equivalents thereof) |
| 05:15.33 | pigpen | ah....too easy..thanks. |
| 05:15.48 | pigpen | either I need sleep or coffeee.... |
| 05:15.52 | pigpen | yeah...coffee it is. |
| 05:16.03 | [TK]D-Fender | pigpen : You have chosen ... poorly |
| 05:16.10 | CunningPike | pigpen: That was totally off the top of my head, so double check to make sure I got the right ones |
| 05:16.25 | [TK]D-Fender | CunningPike : And DBPut is deprecated, shame on you! |
| 05:16.25 | pigpen | Tea? |
| 05:16.34 | CunningPike | [TK]D-Fender: Hence the caveat ;) |
| 05:16.57 | [TK]D-Fender | pigpen : no, SLEEP :) |
| 05:17.01 | CunningPike | Hey - at least I _knew_ it was deprecated :) |
| 05:17.19 | [TK]D-Fender | CunningPike : And it didn't slow down your advising it one bit! Impressive! ;) |
| 05:17.22 | pigpen | Yeah..that is what my wife says... |
| 05:17.35 | CunningPike | Just couldn't remember to what :) |
| 05:17.36 | [TK]D-Fender | pigpen : You should listen to her |
| 05:17.44 | [TK]D-Fender | pigpen : "show function DB" |
| 05:17.47 | CunningPike | [TK]D-Fender: Nothing slows me down |
| 05:18.31 | pigpen | [TK]D-Fender, yeah..I have been converting my db queries to the new style... |
| 05:18.41 | aptura | cnbrk, T1 data Demarc extensions. Must be able to extend the T1 from the LEC demark (pulling cable) into the retailers space |
| 05:18.44 | [TK]D-Fender | pigpen : Good work. |
| 05:18.53 | pigpen | Thank You Master. |
| 05:18.55 | pigpen | :) |
| 05:18.56 | *** join/#asterisk Ofca (i=ofca@akson.sgh.waw.pl) |
| 05:19.01 | aptura | extending means splicing or replacing the cable? |
| 05:20.00 | Ofca | hi guys, as I heard conflicting testimonies so far, i'll try here: does TE400P work under FreeBSD? |
| 05:20.07 | [TK]D-Fender | i'm off to soak for a bit.... causght some stupid bug thats got me completely wiped and cold. |
| 05:20.26 | [TK]D-Fender | Ofca : If that card existed.... maybe ;) |
| 05:20.58 | Ofca | or E400P, I'm not sure of anything anymore :) |
| 05:21.49 | [TK]D-Fender | Ofca : strike 2..... I suggest you "take one for the team" and walk ;) |
| 05:21.50 | Strom_C | aptura: what is the total distance, in copper, between the telco's equipment and your retailer? |
| 05:22.18 | flenders | fuck, I might be smelling or something... please guys, I really need help with the sip problem. |
| 05:22.26 | [TK]D-Fender | Ofca : for Zaptel BSD support I'd suggest you hit the message baords or call Digium direct... |
| 05:23.16 | Ofca | i'm not asking for support yet, i'm just asking if it's working. |
| 05:23.20 | pigpen | flenders, reading... |
| 05:23.44 | aptura | Strom_C this is for a job advertisment. |
| 05:24.01 | flenders | pigpen: thanks mate |
| 05:24.09 | aptura | Its a requirment. I have seen demarks in cisco training but extending one? what does that mean? |
| 05:24.39 | Strom_C | aptura: it means you run cable from the MPOE and install a connecting block to create a secondary demarc. |
| 05:24.41 | JT | [TK]D-Fender: digium does not support digium |
| 05:24.53 | aptura | Also, what is maximum distance for t-1 from the telcos equipment normally? |
| 05:24.57 | pigpen | flenders, double nat? |
| 05:25.05 | JT | err |
| 05:25.09 | JT | that made no sense heh |
| 05:25.14 | CunningPike | flenders: Are you sure the RTP ports are the right ones? |
| 05:25.17 | JT | [TK]D-Fender: digium does not support BSD |
| 05:25.17 | Strom_C | four-wire T1? 733 feet IIRC |
| 05:25.27 | CunningPike | flenders: Try a pcap |
| 05:25.41 | JT | pigpen: disregards the mate talk? |
| 05:25.43 | flenders | from rtp.conf: |
| 05:25.44 | flenders | rtpstart=10000 |
| 05:25.45 | flenders | rtpend=20000 |
| 05:26.10 | pigpen | JT, na..just too much Texan in me. |
| 05:26.17 | JT | ... |
| 05:26.27 | aptura | ahh okay |
| 05:26.39 | aptura | What is the impedence of 4 wire t1? |
| 05:26.45 | flenders | pigpen: sorry, but what do you mean by double nat? |
| 05:27.16 | Strom_C | aptura: beats me |
| 05:27.17 | pigpen | Well, you are having issues with audio to/from your sip provider correct? |
| 05:27.21 | aptura | so its 223 meters |
| 05:27.28 | Strom_C | aptura: you know how t1 works, right? |
| 05:27.34 | flenders | pigpen: yes, but only when dialing out. |
| 05:27.46 | aptura | the physical aspects of protocols? |
| 05:27.48 | flenders | pigpen: do you suggest I change to nat=no? |
| 05:27.59 | pigpen | I suggest you ditch the double nat. |
| 05:28.10 | Strom_C | aptura: the physical layer, the framing, the line coding, everything :) |
| 05:28.23 | pigpen | ie: eth0 is on a private ip block...so something somewhere is nat'ing for you. |
| 05:28.41 | pigpen | and sip does not do well with nat. |
| 05:28.42 | flenders | pigpen: yeah, my linux firewall/router |
| 05:28.55 | pigpen | using a sip proxy? |
| 05:29.40 | aptura | I took one quarter of telco couple years ago. leanred some of it previosly in cisco acadamy. 23 bearer c channels and one d channel for error correction framing? Im rusty on this. Every channel is 64k but 56 is usable with the rest overhead. |
| 05:29.48 | flenders | pigpen: do I set that up on sip.conf? |
| 05:30.05 | pigpen | well, it would be up to your provider if they provide it. |
| 05:30.05 | Strom_C | ugh no no no |
| 05:30.09 | pigpen | typically... |
| 05:30.09 | Strom_C | you're mixing up T1 and ISDN |
| 05:30.10 | aptura | :) |
| 05:30.14 | aptura | okay |
| 05:30.15 | aptura | hehe |
| 05:30.44 | aptura | Man I wished that I did work with this. It was not a job requirment in the past. |
| 05:31.06 | pigpen | flenders, my best results with * and sip over the internet is with the * box right on the internet (firewalled) or accessed via a VPN. |
| 05:31.31 | pigpen | Maybe your provider offers IAX? |
| 05:31.39 | pigpen | much better with NAT'ing. |
| 05:31.41 | flenders | pigpen: no IAX |
| 05:31.46 | aptura | I know my step moms brother and sister inlaw worked with it in the 70s. BTW DSL has been around since the mid 70s. |
| 05:31.46 | pigpen | bummer. |
| 05:32.16 | *** join/#asterisk Roach4 (n=Roach4@modemcable246.188-57-74.mc.videotron.ca) |
| 05:32.32 | pigpen | flenders, stick a box running ethereal between the segment of he firewall and the asterisk box....filtering the two involved IP addresses. |
| 05:32.33 | Strom_C | uh no |
| 05:32.35 | JT | aptura: were you born in north america? |
| 05:32.39 | Strom_C | ADSL has been around since 1988 |
| 05:32.42 | pigpen | if RDP is one way...you can start to narrow it down. |
| 05:32.55 | flenders | RDP? |
| 05:32.59 | Strom_C | HDSL is a late 80s / early 90s thing |
| 05:32.59 | flenders | isn't it RTP? |
| 05:33.34 | aptura | JT I am not going to reveal my age but do remember when Armstrong landed on the moon. |
| 05:33.51 | JT | aptura: that really doesn't answer my question at all |
| 05:33.51 | CunningPike | Someone landed on the moon???? |
| 05:33.52 | pigpen | flenders, sorry..typing too fast. |
| 05:34.07 | JT | aptura: i did not ask WHEN you were born |
| 05:34.08 | flenders | pigpen: so, using nat, it's common to have these problems? I mean, incoming calls are fine, but outgoing aren't |
| 05:34.12 | JT | i asked where |
| 05:34.13 | aptura | :) |
| 05:34.22 | aptura | PNW JT |
| 05:34.28 | pigpen | flenders, absolutely. |
| 05:34.29 | *** join/#asterisk kFuQ (n=somedude@67.185.123.34) |
| 05:34.32 | JT | pnw, i have no idea what that is |
| 05:34.37 | aptura | portland :) |
| 05:34.41 | pigpen | flenders, that is SIP's greatest downfall. |
| 05:34.42 | JT | oh ok |
| 05:34.52 | pigpen | and IAX's greatest feature. |
| 05:34.52 | JT | for the longest time, i thought you were from china, aptura |
| 05:34.59 | aptura | hahaha |
| 05:35.14 | aptura | Ever seen a scottish china man? |
| 05:35.16 | JT | sorry man, but that's what your english seems like :P |
| 05:35.21 | flenders | pigpen, well, I'll try the pcap thing |
| 05:35.31 | flenders | pigpen, thanks anyway |
| 05:35.37 | pigpen | good luck. |
| 05:36.07 | aptura | My spelling does suck at times :) |
| 05:36.19 | pigpen | my spelling sucks all the time. |
| 05:36.33 | JT | more the grammar |
| 05:37.25 | *** join/#asterisk joelsolanki (i=joelsola@202.160.161.94) |
| 05:37.29 | putzz | when we was young my motha said you was bugging me and I said no I Ain't |
| 05:37.59 | joelsolanki | Hi Evening |
| 05:38.34 | *** join/#asterisk PBX-Nick (i=Sktymelb@202.10.94.18) |
| 05:38.50 | joelsolanki | right now i m using asterisk 1.2.* version.what is the difference between in 1.2.x and 1.4.x version ? |
| 05:38.52 | KaiHanari | WHOA... finaly got moh streaming to play something.. but it sounds like a lion growling in my ear... playing too slow or something |
| 05:38.59 | joelsolanki | any major thing that i should use 1.4.x |
| 05:39.58 | CunningPike | joelsolanki: There's a good CHANGES document that lists them....... |
| 05:39.58 | JT | joelsolanki: no not usually |
| 05:39.58 | JT | and UPDATE |
| 05:39.58 | joelsolanki | Hmm |
| 05:39.58 | JT | or is it UPGRADE |
| 05:39.58 | CunningPike | Something like that :) |
| 05:39.58 | joelsolanki | :) |
| 05:39.59 | joelsolanki | hmm ok. |
| 05:39.59 | CunningPike | joelsolanki: Are you in production? |
| 05:39.59 | CunningPike | joelsolanki: With 1.2? |
| 05:40.00 | joelsolanki | but will the new version is heavy that it will use my cpu more ? |
| 05:40.09 | joelsolanki | yes i m using in production server. |
| 05:40.19 | CunningPike | joelsolanki: Stick with 1.2 for now..... |
| 05:40.20 | joelsolanki | 1.2.x version i m using in production version. |
| 05:40.24 | joelsolanki | hmm |
| 05:40.34 | flenders | pigpen: something just came to my mind... 192.168.100.0/24 dont have access to internet, could that be the problem? |
| 05:40.37 | joelsolanki | i think i will do that. :) Thanks. |
| 05:41.01 | flenders | the other interface 192.168.10.0/24 does |
| 05:41.22 | CunningPike | joelsolanki: I'm not dissing 1.4 - it's awesome. I just wouldn't run my production PBX on 1.4.0 |
| 05:41.48 | joelsolanki | hmm ok. |
| 05:41.51 | JT | awesome, apart from not being ;) |
| 05:41.54 | PBX-Nick | Hey guys. Need to write a "Server Status" application for a client that effectivly when a customer selects 1 from an IVR, if there are known issues with a particular server a message will be played stating which server is currently experincg issues. I plan to do this simply by pinging the servers. Im thinking though that pinging each server when the call is answered is a bad idea and will cause a massive delay (Talking 15 servers). So I plan to get my p |
| 05:42.37 | JT | PBX-Nick: ircds can only take so much text at a time |
| 05:42.56 | aptura | man I need a wiremap tester |
| 05:42.59 | PBX-Nick | Blargh I'll split it up now then. Sorry |
| 05:43.13 | PBX-Nick | <PROTECTED> |
| 05:43.18 | PBX-Nick | <PROTECTED> |
| 05:43.27 | aptura | other then fluke what is a good wiremap tester I can buy for 66/110 |
| 05:43.27 | PBX-Nick | <PROTECTED> |
| 05:43.32 | PBX-Nick | <PROTECTED> |
| 05:44.03 | JT | php is a horrible language to write sysadmin scripts in |
| 05:44.07 | JT | but i guess it'll work |
| 05:44.18 | CunningPike | PBX-Nick: Sounds reasonable to me - I was thinking along similar lines |
| 05:45.02 | PBX-Nick | Kewlioes. Now to work out how Asterisk IVR's work with DB's :-) Anything I do Ill upload somewhere and come back with a link when its working. |
| 05:46.02 | JT | you're killing those apostrophes :P |
| 05:46.04 | JT | but yeah |
| 05:46.09 | JT | asterisk IVR |
| 05:46.13 | JT | it's just dialplan really |
| 05:46.22 | JT | and that's up to however you write it |
| 05:47.04 | PBX-Nick | At least you didn't comment on my spelling :-) Thanks for the input though. Much appreciated |
| 05:49.43 | aptura | CunningPike how has been your * network as of late? reliable ? |
| 05:49.58 | CunningPike | aptura: You trying to jinx me? :D |
| 05:50.26 | CunningPike | aptura: We have 100% uptime - nothing less will do |
| 05:51.06 | aptura | good deal. |
| 05:51.34 | *** join/#asterisk gerphimum (n=trekkie@207.190.58.85) |
| 05:54.02 | *** join/#asterisk arcanine (n=arcanine@203.82.44.179) |
| 05:54.42 | wunderkin | 200% uptime |
| 05:54.53 | *** join/#asterisk blj (n=Balaji@59.92.65.128) |
| 05:55.35 | *** part/#asterisk blj (n=Balaji@59.92.65.128) |
| 05:55.36 | *** join/#asterisk BBHoss_ (n=hoss@adsl-249-64-222.hsv.bellsouth.net) |
| 05:56.08 | Strom_C | our phone system runs on milk crates and chewing gum |
| 05:56.30 | CunningPike | Strom_C: Nortel? |
| 05:56.51 | aptura | Has the new Ostastic Algorithim pretty much eliminated all echo on all Digiums cards? |
| 05:56.56 | Strom_C | DairyTel |
| 05:57.04 | CunningPike | Strom_C: Heh |
| 05:57.08 | aptura | funny |
| 05:57.17 | CunningPike | Strom_C: Did you enjoy your visit? |
| 05:57.22 | Strom_C | I had a great time |
| 05:57.35 | CunningPike | Excellent - where are you this week? |
| 05:57.36 | Strom_C | vancouver is a nice place :) |
| 05:57.42 | CunningPike | Yes, it is :) |
| 05:57.43 | Strom_C | i'm back home in los angeles |
| 05:58.06 | CunningPike | Cool - or, warm, rather |
| 05:58.14 | Strom_C | hehe |
| 05:58.31 | aptura | Vancouver is nice when its not raining all the time. |
| 05:58.44 | aptura | Strom_C why did you come here? |
| 05:58.45 | CunningPike | At least you don't have to shovel it |
| 05:59.39 | Strom_C | aptura: work :) |
| 06:00.10 | aptura | Did you come here when digium was teaching asterisk to the registered students here? |
| 06:01.03 | flenders | pigpen: alright!!! fixed! just had to do a 'iptables -t nat -A POSTROUTING -o eth0 -s 192.168.100.0/24 -j SNAT --to-source 192.168.10.x' and it worked. thanks dude (as you disregard mate). |
| 06:01.44 | joelsolanki | asterisk-ooh323c can this be used to translate h.323 to sip ? |
| 06:02.06 | joelsolanki | My provider support SIP only. and i have few client who wants h.323 support |
| 06:02.26 | joelsolanki | Can asterisk some serve h.323 and then forward sip to my provider ? |
| 06:03.08 | *** join/#asterisk BOLIVIAN (n=ewrtwert@201.222.93.229) |
| 06:03.19 | BOLIVIAN | is there asterisk for windows? |
| 06:04.39 | joelsolanki | Anybody have tried this ? |
| 06:05.03 | KaiHanari | im trying to use mpg123 to play back an MP3 stream, it works, but the playback is being slowed by at least 4x... any suggestions ? |
| 06:05.04 | CunningPike | BOLIVIAN: Unbelievably, yes - although I'm not sure why |
| 06:05.19 | *** join/#asterisk ShadowHntr (i=sentinel@wikipedia/Shadowhntr) |
| 06:05.42 | BOLIVIAN | cunningpike is this the page? http://www.asteriskwin32.com/ |
| 06:05.44 | joelsolanki | CunningPike: Any hints for converting h.323 to sip using asterisk |
| 06:05.54 | BOLIVIAN | the oficcial one? |
| 06:06.46 | CunningPike | BOLIVIAN: I believe so - but seriously, why? |
| 06:07.21 | *** join/#asterisk dlynes_laptop (n=dlynes@s209-121-68-66.bc.hsia.telus.net) |
| 06:07.26 | CunningPike | joelsolanki: I'm not sure if it translates between them - it should....... |
| 06:07.39 | joelsolanki | hmm i have to find |
| 06:07.47 | joelsolanki | Welcome Daniel :) |
| 06:08.04 | CunningPike | joelsolanki: I have no H.323 experience though - I think it can be tricky |
| 06:08.09 | joelsolanki | hmm |
| 06:08.16 | ShadowHntr | aserisk on win32? sounds dangerous. ;) |
| 06:09.54 | CunningPike | Everything on Windows is dangerous |
| 06:10.16 | BOLIVIAN | cunningpike i dont know how to install centos |
| 06:10.23 | BOLIVIAN | linux is dangerous too |
| 06:10.30 | BOLIVIAN | specially the installation |
| 06:10.45 | CunningPike | BOLIVIAN: It's easy - honest. Get the install CD, boot from it, accept the defaults, et voila |
| 06:10.47 | JT | dangerous wtf |
| 06:10.55 | JT | if you can't administer linux |
| 06:10.59 | JT | you cannot administer asterisk |
| 06:11.00 | JT | simple |
| 06:11.23 | BOLIVIAN | cunningpike i donwloaded 4 files |
| 06:11.23 | JT | CunningPike: yeah if you get h.323 to work in asterisk, like any other channel, it can translate |
| 06:11.27 | aptura | vi agree JT |
| 06:11.47 | BOLIVIAN | cunningpike i donwloaded 4 files do i have to burn this folder CentOS-4.3-x86_64-bin1of4 ? |
| 06:11.52 | CunningPike | JT: Figures - big 'if', though |
| 06:11.52 | BOLIVIAN | and then boot from it? |
| 06:12.16 | JT | BOLIVIAN: you've never burnt a cd image before? |
| 06:12.18 | CunningPike | BOLIVIAN: No - I would get the single installer CD and then install the packages from the Internet - hang on a sec |
| 06:12.26 | CunningPike | BOLIVIAN: Want to take this to #centos? |
| 06:12.41 | BOLIVIAN | k tanks |
| 06:15.44 | KaiHanari | Damn |
| 06:15.47 | *** join/#asterisk EyeCue (n=eyecue@miranda/admin/eyecue) |
| 06:16.06 | KaiHanari | cant get streams to stop sounding like its something out of a horror movie |
| 06:29.22 | putzz | whats does this mean: setformat: Unable to re-open DSP device /dev/dsp: No such file or directory |
| 06:30.14 | JT | means your soundcard is not setup |
| 06:31.38 | putzz | what do I need to install soundcard drivers or set it up? |
| 06:32.02 | *** join/#asterisk gerphimum (n=trekkie@207.190.58.85) |
| 06:32.46 | BOLIVIAN | does asterisk has a GUI? |
| 06:32.54 | BOLIVIAN | or all is console oriented |
| 06:33.14 | putzz | asteriskgui does |
| 06:33.35 | putzz | but that defeats the point of learning asterisk because its all automated |
| 06:33.48 | BOLIVIAN | so asteriskgui comes with asterik? |
| 06:34.27 | BOLIVIAN | i want a user friendly interface |
| 06:35.23 | putzz | asteriskgui is asterisk with a gui interface |
| 06:35.37 | putzz | and very friendly it will even spoon feed you |
| 06:35.38 | putzz | ;-) |
| 06:35.59 | BOLIVIAN | so i download asterisgui from the same pager?æ |
| 06:36.06 | BOLIVIAN | i mean web site |
| 06:36.08 | BOLIVIAN | ? |
| 06:36.17 | putzz | asteriskgui.com I belive |
| 06:36.29 | putzz | you know google is a good place for answers |
| 06:36.30 | putzz | ;-) |
| 06:39.00 | JT | BOLIVIAN: what does it need a gui? |
| 06:39.25 | CunningPike | BOLIVIAN: If you are planning on offering paid service to the public, you really need to know what you are doing..... |
| 06:40.08 | JT | or to any business really |
| 06:40.43 | BOLIVIAN | heh |
| 06:41.53 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 06:43.31 | putzz | use gui and give me ip I show how expensive I can be ;-) |
| 06:52.13 | *** join/#asterisk zeeesh (i=zeeesh@202.38.55.125) |
| 06:52.15 | zeeesh | hi |
| 06:53.42 | *** join/#asterisk scardinal (n=supreme@port816.ds1-suoe.adsl.cybercity.dk) |
| 07:05.17 | *** join/#asterisk nuonguy (n=john@c-24-6-175-26.hsd1.ca.comcast.net) |
| 07:14.39 | *** join/#asterisk jpe-nyc (n=jpe@p77-37.acedsl.com) |
| 07:14.56 | *** join/#asterisk vibi (n=Miranda@p54AE7DD9.dip.t-dialin.net) |
| 07:15.24 | vibi | good morning everyone |
| 07:15.31 | *** join/#asterisk thekidrio (n=thekidri@24-205-76-13.dhcp.psdn.ca.charter.com) |
| 07:15.38 | BOLIVIAN | cunningpike u awake? |
| 07:15.52 | BOLIVIAN | wouldnt be easier to install just asterisknow? |
| 07:16.19 | *** join/#asterisk s1gny|wrk (n=s1gny@p5491662F.dip.t-dialin.net) |
| 07:16.28 | vibi | i need some information about chan_sccp and snom phones, are there any "gurus" ? ;) |
| 07:16.30 | *** part/#asterisk s1gny|wrk (n=s1gny@p5491662F.dip.t-dialin.net) |
| 07:16.58 | thekidrio | Does anyone know of a site that reviews sip/iax providers? |
| 07:17.32 | vibi | after installing chan_sccp my snom phones do not register any moore, any idea why?? |
| 07:18.14 | putzz | BOLIVIAN: its easier but....I wouldnt |
| 07:18.18 | *** join/#asterisk adker_ (n=chatzill@74-33-198-79.br1.glv.ny.frontiernet.net) |
| 07:18.25 | *** join/#asterisk [Airwolf] (n=airwolf@attilla.nl) |
| 07:18.32 | BOLIVIAN | putzz why not |
| 07:18.42 | putzz | 1 u wont learn a thing |
| 07:18.46 | KaiHanari | How can i make a recording to be saved in gsm or wav for an IVR ? I know how you do this if your using freePBX but not otherwise |
| 07:19.07 | JT | Record |
| 07:19.25 | BOLIVIAN | i dont wan to learn i need a sofware to manage calls as friendly as can be with my x100p |
| 07:19.46 | JT | you don't want to learn |
| 07:19.50 | JT | that pretty much sums it up |
| 07:20.02 | BOLIVIAN | of course i will learn something |
| 07:20.08 | putzz | BOLIVIAN: nop u wont |
| 07:20.08 | hads | heh |
| 07:20.09 | BOLIVIAN | but i dont want to spend weeks in that |
| 07:20.11 | putzz | how to push 2 buttons? |
| 07:20.13 | putzz | heh |
| 07:20.13 | JT | better to pay someone else to fix it then |
| 07:20.21 | putzz | nothing on how it actually works |
| 07:21.00 | BOLIVIAN | the first step i have to take is to install centos with xp on the same machine |
| 07:21.05 | *** join/#asterisk h0 (i=fakhir@unaffiliated/fakhir) |
| 07:21.10 | BOLIVIAN | that would take me like 3 days |
| 07:21.21 | putzz | dont share machines |
| 07:21.23 | KaiHanari | JT, Heh. thats one of those "Durrr... " moments eh? how do i record... gee, i dunno, record? XD |
| 07:21.26 | putzz | get a crappy one to test it |
| 07:21.28 | BOLIVIAN | cause i bet it will come with an error |
| 07:21.45 | putzz | gui will most likelly give u 0 errors as it installs everything for u |
| 07:21.50 | *** join/#asterisk tzafrir_laptop (n=tzafrir@62.90.10.53) |
| 07:21.54 | *** join/#asterisk booray (n=booray@adsl-75-13-213-22.dsl.irvnca.sbcglobal.net) |
| 07:21.56 | Strom_C | and then give you 9999999999 headaches |
| 07:22.03 | putzz | yep |
| 07:22.11 | *** join/#asterisk Beirdo (n=gjhurlbu@unaffiliated/beirdo) |
| 07:22.33 | vibi | after installing chan_sccp my snom phones do not register any moore, any idea why?? |
| 07:22.34 | BOLIVIAN | i can use console but im planning to let use this program to a 18 y olg boy |
| 07:22.46 | BOLIVIAN | he wont be able to type console |
| 07:23.02 | *** join/#asterisk inspired (n=mikael@cl-330.sto-01.se.sixxs.net) |
| 07:23.05 | putzz | y not? I think my 10 year old can do it maybe my 8 can too |
| 07:23.06 | putzz | ;-) |
| 07:23.34 | BOLIVIAN | what im planning to do is rent my phone line with x100p help |
| 07:23.43 | BOLIVIAN | and asterisk |
| 07:23.51 | BOLIVIAN | so i can control the call duration |
| 07:24.10 | putzz | ok.... |
| 07:24.13 | BOLIVIAN | how asterisk show2 u that |
| 07:24.18 | BOLIVIAN | in a log text file? |
| 07:24.46 | hads | Yes |
| 07:24.49 | BOLIVIAN | by control i dont mean to cut the conversation but to see the time spent |
| 07:24.52 | BOLIVIAN | of my client |
| 07:24.58 | BOLIVIAN | and to see which number he dialed |
| 07:25.12 | putzz | ok |
| 07:25.51 | BOLIVIAN | and can i load this log text file to some excel aplication to calculate the price he must pay? |
| 07:26.05 | putzz | u can use some kind of billing software |
| 07:26.05 | BOLIVIAN | all this automatically? |
| 07:26.06 | putzz | with it |
| 07:26.20 | putzz | a2billing comes with asterisk@home |
| 07:26.27 | putzz | it will do that for u |
| 07:26.36 | putzz | but if u dont want to learn dont bother going into it |
| 07:26.37 | putzz | heh |
| 07:26.59 | BOLIVIAN | asterisk@home is the home version of asterisk? |
| 07:27.14 | tzafrir | no |
| 07:27.32 | BOLIVIAN | so what it is a tool |
| 07:27.33 | Strom_C | asterisk@home is the crap version of asterisk :) |
| 07:27.39 | BOLIVIAN | oh i c |
| 07:27.43 | tzafrir | The name stems from an original intent to make it an integration of Asterisk with X10 home integration. |
| 07:27.43 | *** join/#asterisk fwp (n=FWP@unaffiliated/fwp) |
| 07:28.04 | tzafrir | It has since been renamed to TrixBox |
| 07:28.26 | putzz | they r all junk. |
| 07:28.31 | BOLIVIAN | once i plug my x100p will asterisk detect it? |
| 07:28.31 | tzafrir | I haven't heard of anybody using it with X10, actually |
| 07:28.43 | putzz | BOLIVIAN: why dont u set it up try |
| 07:28.46 | putzz | then ask questions |
| 07:28.51 | tzafrir | BOLIVIAN, not exactly, but that's easy to configure |
| 07:29.06 | BOLIVIAN | i cant try cause i didnt receive the card yet |
| 07:29.14 | BOLIVIAN | its taking ages to come here from texas |
| 07:29.27 | BOLIVIAN | i bought it on ebay |
| 07:29.30 | tzafrir | with latest versions of Zaptel you have xpp/utils/genzaptelconf |
| 07:29.52 | BOLIVIAN | zaptel is another pbx similar to asterisk? |
| 07:29.55 | tzafrir | Though if you just have 1 x100p card, the sample config will work |
| 07:31.23 | BOLIVIAN | tzafrir so zaptel is another pbx similar to asterisk? |
| 07:31.46 | tzafrir | zaptel is one component related to Asterisk |
| 07:32.00 | BOLIVIAN | oh so is a tool inside of it |
| 07:32.01 | tzafrir | A sort of library Asterisk uses. |
| 07:32.05 | BOLIVIAN | oh ok |
| 07:32.59 | tzafrir | As for the name: http://zapatatelephony.org/ (that site is rather old and has no up-to-date information) |
| 07:33.39 | BOLIVIAN | i dont need to download zapter separated couse it comes with asterik right? |
| 07:33.44 | BOLIVIAN | zaptel i meant |
| 07:33.59 | tzafrir | This is a separate package |
| 07:34.26 | tzafrir | You'll find some places mmentioning libpri: you won't need it (you don't have a T1 card) |
| 07:36.04 | BOLIVIAN | sheesh that is like a ftp |
| 07:36.08 | BOLIVIAN | im on zaptel directory |
| 07:38.10 | *** join/#asterisk fwp (n=FWP@unaffiliated/fwp) |
| 07:48.36 | *** join/#asterisk dlynes_laptop (n=dlynes@S0106001346f7843f.vc.shawcable.net) |
| 07:50.21 | *** join/#asterisk FlatFoot (n=simon@80.88.192.113) |
| 07:55.34 | *** join/#asterisk SimoAmi (n=simoami@cpe-24-193-190-236.nyc.res.rr.com) |
| 07:55.43 | SimoAmi | hi there |
| 07:55.49 | BOLIVIAN | putz |
| 07:55.50 | BOLIVIAN | u stil there |
| 07:56.10 | SimoAmi | i have an AGI question |
| 07:56.34 | SimoAmi | anyone? |
| 07:56.49 | BOLIVIAN | tzafrir |
| 07:57.07 | Strom_C | ooh ooh i have a question ooh ooh ooh ooh ooh why is no one answering my question ooh ooh |
| 07:57.23 | BOLIVIAN | strom they must be sleepin |
| 07:57.23 | tzafrir | BOLIVIAN, there's a direct link from asterisk.org's downloads page |
| 07:57.35 | Strom_C | s/ooh ooh i have a question ooh ooh ooh ooh ooh why is no one answering my question ooh ooh/just ask the damned question/ |
| 07:57.50 | BOLIVIAN | tzafrir i know but i was wondering how to show |
| 07:57.57 | BOLIVIAN | the client the time he spent |
| 07:58.00 | SimoAmi | I just one to know if it's possible to use another voice for the SayNumber(), SayDigit() functions without affecting the general settings? |
| 07:58.04 | BOLIVIAN | in the call without turning my monitor |
| 07:58.21 | BOLIVIAN | they will be 2 meters from me in a small glass room |
| 07:58.55 | SimoAmi | Strom_C: calm down man |
| 07:59.21 | tzafrir | SimoAmi, You could use another language, but then you'd get another syntax, potentially |
| 07:59.49 | SimoAmi | I'm already using phpagi |
| 08:00.07 | tzafrir | SimoAmi, language: in Asterisk |
| 08:00.31 | SimoAmi | what do you suggest? |
| 08:00.32 | tzafrir | Set(LANGUAGE) . But those kind of hacks are more and more discourged |
| 08:00.56 | tzafrir | Why do you need to use different voices? |
| 08:01.30 | SimoAmi | I just don't want to reimplement the number to voice function "One Thousand Four Hundred Thirty Fire" |
| 08:01.31 | *** join/#asterisk Snake-Eyes (n=blog@203.220.55.70) |
| 08:01.43 | tzafrir | SayDigit is something that you could re-impement in phpagi to take a digit from a different directory |
| 08:02.29 | *** join/#asterisk CrashHD (n=crashhd@c-76-20-22-240.hsd1.ca.comcast.net) |
| 08:02.33 | tzafrir | something of the sort of: "play the file $customdir/$num" |
| 08:02.43 | SimoAmi | I want to use another voice for an application while leaving Allison's voice for the phone system |
| 08:02.53 | CrashHD | Hello |
| 08:02.55 | CrashHD | anyone know if multiple parking lots have been implemented in asterisk yet? |
| 08:03.40 | tzafrir | SayNumber is trickier and much more error-prone to re-implement |
| 08:03.48 | SimoAmi | tzafrir: I thought about that as a last option, but I would have to reimplement SayNumber |
| 08:04.05 | tzafrir | Look at say.c |
| 08:04.06 | SimoAmi | yep |
| 08:04.07 | tzafrir | :-( |
| 08:04.24 | SimoAmi | I'll look into the Set(LANGUAGE) feature |
| 08:05.04 | tzafrir | hmmm... also take a look at say.conf if you use 1.4, to make sure the "language" you choose will not do strange stuff |
| 08:06.43 | *** join/#asterisk stoffell (n=stoffell@d51A58215.access.telenet.be) |
| 08:09.21 | *** join/#asterisk UlbabraB (n=salama@host170-159-static.110-62-b.business.telecomitalia.it) |
| 08:12.04 | SimoAmi | thx tzafrir |
| 08:12.21 | *** join/#asterisk booray (n=booray@adsl-71-156-53-131.dsl.irvnca.sbcglobal.net) |
| 08:12.39 | KaiHanari | i have 1.4 installed, where do i go to make uninstall it so i can downgrade? Ive deleted the directory i compiled from. |
| 08:13.09 | CrashHD | anyone know the status of multiple parking lots in asterisk 1.4? |
| 08:13.15 | tzafrir | KaiHanari, there isn't really a relible uninstall method. I'd try to install 1.2 on top of it |
| 08:13.39 | KaiHanari | tzafrir, k, im gonna download the source and extract again to try make uninstall first. |
| 08:13.55 | tzafrir | be careful out there |
| 08:15.45 | KaiHanari | says its removed all but configs spools and logs, and offered a line to remove those aswell |
| 08:15.48 | KaiHanari | :) |
| 08:16.12 | *** join/#asterisk vgster (n=vgster@h14658.navonline.net) |
| 08:22.04 | FlatFoot | morning all |
| 08:23.34 | FlatFoot | is there a different cvs server other than cvs.digium.com ? cos i cannot resolve that |
| 08:23.49 | Strom_C | cvs has been deprecated forever now |
| 08:23.52 | Strom_C | use svn |
| 08:24.09 | FlatFoot | great just learning all this stuff and it dissapeeeeers |
| 08:24.34 | *** join/#asterisk af_ (n=getsmart@ip-179-53.sn1.eutelia.it) |
| 08:24.45 | FlatFoot | so what the heeeellll command do i use now to update on a freebsd 6.x box |
| 08:24.54 | *** join/#asterisk slima_ (i=slima@unaffiliated/slima) |
| 08:27.10 | e-ddie | FlatFoot: rtfm usually works |
| 08:28.39 | *** join/#asterisk KaiHanari (n=kaihanar@stjhnf0122w-142162043055.pppoe-dynamic.nl.aliant.net) |
| 08:28.43 | *** part/#asterisk mtaht4 (n=m@c-71-198-23-124.hsd1.ca.comcast.net) |
| 08:28.46 | FlatFoot | yeah just ranting am reading now# |
| 08:29.40 | *** join/#asterisk shinux__ (n=shinux@196.220.30.78) |
| 08:32.13 | dlynes_laptop | CrashHD: afaik, it's still something you have to check out of svn |
| 08:38.45 | *** join/#asterisk Simplix (n=loic@LSt-Amand-152-31-13-31.w82-127.abo.wanadoo.fr) |
| 08:39.50 | Simplix | hello, is there any solution to pickup a call in a macro ? |
| 08:40.10 | dlynes_laptop | Simplix: directed call pickup, group pickup? |
| 08:40.22 | Simplix | i have Executing [s@macro-foo:3] Dial("SIP/1234-081c2ad0", "SIP/1235|20|r") in new stack |
| 08:40.33 | Simplix | and i can't pickup 1235 |
| 08:41.36 | Simplix | directed call |
| 08:41.54 | Simplix | the s@blabla is not pickupable :) |
| 08:42.28 | Simplix | can i have [1235@macro-foo:3] instead of [s@macro-foo:3] |
| 08:45.22 | dlynes_laptop | no idea |
| 08:45.26 | dlynes_laptop | I've never seen your extensions.conf file |
| 08:49.05 | Simplix | my macro is like that : |
| 08:49.06 | Simplix | [macro-foo] |
| 08:49.06 | Simplix | exten => s,1,Set(CHANNEL(language)=fr) |
| 08:49.06 | Simplix | exten => s,2,Dial(SIP/${ARG1},20,r) |
| 08:50.15 | Simplix | so i can't pickup this call anymore |
| 08:50.38 | Simplix | because of the s |
| 08:52.32 | SheriF_SpacE | i have a small question i have 2 digiu card one with 4 fxo and one with 2 fxo the /etc/zaptel.conf be link fxsks=1-6 right? |
| 08:52.39 | *** join/#asterisk lorinc (n=ang@caracas-3536.adsl.interware.hu) |
| 08:53.55 | fetcher | SheriF_SpacE: sounds right |
| 08:56.03 | SheriF_SpacE | fetcher: not working |
| 08:56.07 | SheriF_SpacE | i want to understand something |
| 08:56.11 | SheriF_SpacE | how it will read teh cards ? |
| 08:56.14 | SheriF_SpacE | which one 1st? |
| 08:56.32 | *** join/#asterisk tsurko (n=tsurko@vpn-pppoe-213-240-242-237.megalan.bg) |
| 08:58.59 | dlynes_laptop | SheriF_SpacE: probably the one in the lowest pci slot number will get counted first |
| 08:59.09 | dlynes_laptop | SheriF_SpacE: erm enumerated |
| 08:59.21 | dlynes_laptop | Simplix: pastebin your extensions.conf file |
| 08:59.42 | fetcher | SheriF_SpacE: 1st slot is usually to the right (closer to AGP slot, or ATX backplate) |
| 08:59.53 | Simplix | what is pastebin ... ? i paste here ? |
| 09:00.21 | SheriF_SpacE | if the cards only fxo enough to modprob wctfxo ? |
| 09:02.07 | SheriF_SpacE | hmm still not working :-s |
| 09:02.30 | SheriF_SpacE | may be the modules not in the right number |
| 09:02.34 | *** join/#asterisk stkn_ (n=stkn@gentoo/developer/pdpc.active.stkn) |
| 09:02.39 | *** join/#asterisk Dibbler_XP (n=Dibbler_@host217-45-198-229.in-addr.btopenworld.com) |
| 09:02.47 | SheriF_SpacE | and i don't need external power right ? |
| 09:03.46 | *** join/#asterisk Dibbler_XP (n=Dibbler_@host217-45-198-229.in-addr.btopenworld.com) |
| 09:03.47 | fetcher | SheriF_SpacE: I thought wcfxo was only for the old single-port X100P cards |
| 09:04.14 | *** join/#asterisk Dibbler_XP (n=Dibbler_@host217-45-198-229.in-addr.btopenworld.com) |
| 09:04.42 | Simplix | http://www.pasteserver.net/73 for my extension.conf |
| 09:05.34 | dlynes_laptop | ~pb |
| 09:05.42 | jbot | well, pb is a place to paste your stuff without flooding the channel - try http://pastebin.ca, or http://channels.debian.net/paste |
| 09:06.06 | dlynes_laptop | Simplix: the advantage of pastebin.ca is that you can tell it to syntax highlight asterisk configuration files |
| 09:06.29 | fetcher | SheriF_SpacE: try loading wctdm instead (or 'wcfxs' for older zaptel versions) |
| 09:06.33 | Simplix | dlynes_laptop, nice i will use this one next :) |
| 09:07.09 | dlynes_laptop | SheriF_SpacE: wcfxo is for x100p cards, not tdm400p cards |
| 09:07.13 | fetcher | SheriF_SpacE: former name was wcfxs even for all-FXO boards... they changed it to wctdm to be less confusing |
| 09:07.39 | dlynes_laptop | fetcher: i don't remember there ever being a wcfxs driver |
| 09:07.46 | SheriF_SpacE | dyiccc |
| 09:07.47 | SheriF_SpacE | thx |
| 09:07.49 | dlynes_laptop | fetcher: i've seen it referred to, but never actually seen it |
| 09:08.20 | dlynes_laptop | fetcher: was it in old 1.0 zaptels? |
| 09:08.34 | *** join/#asterisk FreezeS (n=bla@82.208.157.125) |
| 09:08.41 | FreezeS | hey guys |
| 09:08.53 | fetcher | dlynes: yup, and old machine with zaptel-1.0.8 still has it. Not sure when they made the change |
| 09:08.55 | FreezeS | do you have any idea what "m-a" is and how to install it on debian ? |
| 09:08.58 | fetcher | s/and/an/ |
| 09:09.10 | Simplix | http://pastebin.ca/355086 for my extension.conf, highlight is good :) |
| 09:09.24 | dlynes_laptop | Simplix: I don't see a macro-foo in there |
| 09:09.33 | Simplix | macro-abbsil |
| 09:09.44 | *** join/#asterisk diclophis-work (n=jbardin@comet.cisdata.net) |
| 09:09.45 | diclophis-work | hello al |
| 09:09.47 | dlynes_laptop | Simplix: ok |
| 09:09.53 | dlynes_laptop | morning, diclophis-work |
| 09:10.04 | diclophis-work | do all of the asterisk manager commands accept the Async paramter? |
| 09:10.17 | diclophis-work | for some reason, the QueueAdd method is taking forever to complete |
| 09:10.20 | *** join/#asterisk zapp-branigan (n=zapp-bra@81-202-140-56.user.ono.com) |
| 09:10.39 | Simplix | dlynes_laptop, have i to insert the macro in the contexte to pick it up ? |
| 09:10.45 | fetcher | am I likely to have trouble feeding an old Panasonic PBX using PAP2 ATA's? |
| 09:11.09 | *** join/#asterisk Makenshi (n=makenshi@kenshi.fox.furry.be) |
| 09:11.15 | dlynes_laptop | fetcher: nope |
| 09:11.18 | fetcher | I think it's a KX-TD1232. Defintely loop-start on the CO lines |
| 09:11.25 | dlynes_laptop | fetcher: that's a KSU, not a PBX |
| 09:11.50 | dlynes_laptop | fetcher: and yes you can feed it using PAP2's, no problem |
| 09:12.13 | dlynes_laptop | fetcher: i've got at least two customers with that setup |
| 09:12.29 | dlynes_laptop | fetcher: KX-TD1232 and KX-TD816 or something like that as well |
| 09:12.40 | fetcher | dlynes_laptop: thanks. I expected it to work OK, but was a little worried about audio levels, echo, ring voltage etc. issues |
| 09:13.05 | dlynes_laptop | fetcher: yeah...depending on your panasonic system, you might have to adjust the voltage |
| 09:13.13 | dlynes_laptop | fetcher: we had to adjust that for one customer |
| 09:13.15 | FreezeS | does anybody know what "m-a" is ? |
| 09:13.18 | dlynes_laptop | fetcher: but the others were just fine |
| 09:13.29 | fetcher | dlynes_laptop: adjusted it on the PAP2's? |
| 09:13.43 | dlynes_laptop | fetcher: in our case, it was on sipura 2000's |
| 09:13.54 | dlynes_laptop | fetcher: but pap2 is just the newer generation of the sipura 2000's |
| 09:14.05 | dlynes_laptop | fetcher: it's on your regional settings page |
| 09:14.29 | fetcher | yeah, I was surprised to see that some of them default to 25 Hz ring frequency, instead of 20 Hz, and a trapezoidal waveform |
| 09:14.46 | fetcher | (phones don't seem to mind) |
| 09:15.42 | dlynes_laptop | fetcher: yeah...just leave the voltage alone unless you have issues |
| 09:16.05 | dlynes_laptop | Simplix: your extensions.conf isn't really making a lot of sense |
| 09:16.18 | dlynes_laptop | Simplix: you're not handling a number of conditions |
| 09:16.49 | Simplix | i want to learn a lot of things :) |
| 09:16.50 | fetcher | dlynes_laptop: what makes the KX-TD1232 a KSU? I thought KSU referred to the old 1A2-style systems with mechanical line keys |
| 09:17.23 | dlynes_laptop | fetcher: nah...KSU is a small-scale pbx |
| 09:17.37 | dlynes_laptop | fetcher: most KSU's cannot handle DDI's for instance |
| 09:17.38 | fetcher | or newer units operating the same way... having to yell "Pick up line 4!" across the room :) |
| 09:17.57 | *** join/#asterisk xermesx (n=ermsewrk@217.220.121.62) |
| 09:18.02 | dlynes_laptop | fetcher: also KSU's cannot handle PRI's |
| 09:18.19 | dlynes_laptop | fetcher: they can only handle very limited number of handsets and analog phone lines |
| 09:18.37 | dlynes_laptop | fetcher: PBXs typically use blades |
| 09:18.43 | dlynes_laptop | fetcher: KSUs do not |
| 09:18.50 | *** join/#asterisk lukketto (n=lukketto@host86-8-dynamic.7-87-r.retail.telecomitalia.it) |
| 09:19.08 | dlynes_laptop | fetcher: it's more a matter of scale than anything |
| 09:19.45 | fetcher | hmm. I guess Panasonic labeling those small units "PBXes" is more marketing than substance |
| 09:19.53 | *** join/#asterisk Ahrimanes (n=ma@81.7.159.2) |
| 09:26.13 | Simplix | dlynes_laptop, for example when the 4420 is ringing and i call *7*4420, i got this message : |
| 09:26.14 | Simplix | <PROTECTED> |
| 09:27.49 | dlynes_laptop | Simplix: of course not |
| 09:27.56 | dlynes_laptop | Simplix: there's no extension called '4420' |
| 09:28.25 | Simplix | in default ... no ? |
| 09:28.38 | dlynes_laptop | Simplix: erm nvm |
| 09:28.46 | dlynes_laptop | Simplix: Your first attempt is to SIP/4420 |
| 09:28.54 | dlynes_laptop | Simplix: do you have a SIP node called '4420'? |
| 09:29.22 | Simplix | yes i have |
| 09:29.34 | dlynes_laptop | Simplix: Ok, how about an IAX2 node called '4420'? |
| 09:29.42 | Simplix | there is not |
| 09:30.16 | putzz | context issue? |
| 09:30.25 | dlynes_laptop | Simplix: and how about a call forward registry of '4420'? |
| 09:30.42 | Simplix | not forward neither |
| 09:30.53 | dlynes_laptop | Simplix: can you show the three lines before that error and the three lines after it in the log? |
| 09:30.55 | Simplix | s/not/no |
| 09:31.37 | Simplix | <PROTECTED> |
| 09:31.37 | Simplix | <PROTECTED> |
| 09:31.37 | Simplix | <PROTECTED> |
| 09:31.37 | Simplix | [Feb 14 10:18:21] NOTICE[18010]: app_directed_pickup.c:159 pickup_exec: No target channel found for 4420. |
| 09:32.07 | Simplix | <PROTECTED> |
| 09:32.07 | Simplix | <PROTECTED> |
| 09:32.07 | Simplix | <PROTECTED> |
| 09:34.14 | dlynes_laptop | Simplix: add the following before 'exten => s,n(cfim),Dial(Local/${temp}@interne/n)' in [macro-stdexten] |
| 09:34.49 | dlynes_laptop | Simplix: exten => s,n(cfim),Noop(temp=${temp}) |
| 09:35.10 | dlynes_laptop | Simplix: and then change your dial command so that it's exten => s,n(cfim+1),... |
| 09:35.27 | dlynes_laptop | Simplix: then reload and show the new error context in the log |
| 09:38.28 | Simplix | i have no contect error :_/ |
| 09:38.43 | Simplix | context error |
| 09:46.21 | *** join/#asterisk A-Tuin|work (n=A-Tuin@212.41.185.81) |
| 09:47.10 | dlynes_laptop | ? |
| 09:51.54 | vibi | where can i get the latest sccp firmware for cisco 7910G? any idea?? |
| 09:52.07 | xermesx | is there a changelog for dialplan in asterisk 1.4 ? |
| 09:53.00 | *** join/#asterisk hematitec (n=cratz@adsl-71-159-206-4.dsl.pltn13.sbcglobal.net) |
| 09:53.50 | *** join/#asterisk giasai68 (n=giasai@ip-240-130.sn2.eutelia.it) |
| 09:55.09 | Hmmhesays | what changes are you looking for? |
| 09:58.19 | *** join/#asterisk ghenry (n=ghenry@suretecsystems.plus.com) |
| 10:00.46 | *** join/#asterisk Dibbler_XP (n=Dibbler_@host217-45-198-229.in-addr.btopenworld.com) |
| 10:01.25 | giasai68 | hello, i'm getting this worning on CLI asterisk: Don't know how to indicate condition -1 on ooh323c_286 |
| 10:01.51 | giasai68 | can you help me to urderstand whats mean? |
| 10:07.04 | dlynes_laptop | giasai68: it means you're probably getting an invalid indication |
| 10:07.18 | dlynes_laptop | giasai68: and ooh323c_286 isn't written to handle buggy asterisk parameters |
| 10:10.08 | giasai68 | can you help me to how i can i found this i valid indication? i'm checking ooh323.conf extension.conf zaptel.conf zapata.conf and seems all ok |
| 10:13.56 | dlynes_laptop | giasai68: use the source, luke! |
| 10:14.12 | Ahrimanes | mmmm source |
| 10:14.15 | dlynes_laptop | giasai68: I don't know what ooh323c_286 is |
| 10:14.25 | dlynes_laptop | giasai68: is it a branch of ooh323c? |
| 10:14.30 | Ahrimanes | <homer>mmm source</homer> would be more indicative i guess |
| 10:15.24 | dlynes_laptop | Ahrimanes: nah...that's a 'homer' indicator, not a -1 indicator |
| 10:15.42 | Ahrimanes | ah |
| 10:18.27 | giasai68 | this is complete warnig : src/chan_h323.c:977 ooh323_indicate: Don't know how to indicate condition -1 on ooh323c_374 |
| 10:18.40 | giasai68 | as you can see the number _xxx change |
| 10:19.48 | giasai68 | the calls are active but some time i have this warning |
| 10:20.30 | dlynes_laptop | giasai68: ah..that's your channel # |
| 10:20.47 | dlynes_laptop | giasai68: or rather call leg # |
| 10:21.22 | dlynes_laptop | giasai68: which version of asterisk-addons? |
| 10:21.45 | dlynes_laptop | giasai68: 1.2.5? 1.4.0? |
| 10:21.59 | dlynes_laptop | Ahrimanes: tea is better for you, anyways |
| 10:22.05 | dlynes_laptop | Ahrimanes: try a nice tall mug of matcha |
| 10:22.41 | Ahrimanes | dlynes_laptop: hm not enough caffeine i'm afraid |
| 10:23.50 | dlynes_laptop | giasai68: i'm guessing 1.4.0? |
| 10:24.13 | giasai68 | i'm using 1.4 |
| 10:24.30 | *** join/#asterisk osiris (n=osiris@c-71-205-27-131.hsd1.mi.comcast.net) |
| 10:24.38 | Ahrimanes | hm I should be submitting a bugreport just about now.. but am feeling lazy |
| 10:26.09 | *** join/#asterisk lukketto (n=lukketto@host86-8-dynamic.7-87-r.retail.telecomitalia.it) |
| 10:26.53 | *** part/#asterisk zapp-branigan (n=zapp-bra@81-202-140-56.user.ono.com) |
| 10:30.59 | giasai68 | sorry my internet go down for some minutes i0m back now |
| 10:31.31 | *** join/#asterisk hematitec (n=cratz@adsl-71-159-206-4.dsl.pltn13.sbcglobal.net) |
| 10:31.37 | dlynes_laptop | giasai68: anyways...what's happening is that something's creating an illegal indication on your h323 channel |
| 10:31.58 | dlynes_laptop | giasai68: i woudl spend some time in debug mode to determine where that indication is coming from |
| 10:32.10 | dlynes_laptop | giasai68: so you can file a proper bug report |
| 10:38.02 | *** join/#asterisk tcseke (n=chatzill@22-36.adsl.etel.hu) |
| 10:45.05 | *** join/#asterisk angryuser (n=Miranda@i03v-213-44-169-43.d4.club-internet.fr) |
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| 10:50.20 | *** join/#asterisk oej (n=olle@apollo.webway.se) |
| 10:51.41 | angryuser | goog day i am trying to make a really simple script au auto forward, but i have some problems, when i use DB_EXISTS function, DB_RESULT is not set, or meybe i did some error on saving the var in asdb, script is here http://www.pastebin.ca/355211 |
| 10:52.05 | angryuser | brrrrr "Good" need more coffee |
| 10:52.38 | angryuser | If someone can help me out;) |
| 10:54.28 | canapa | is it difficult to run a asterisk inside a LAN (inside a NAT) |
| 10:54.56 | *** join/#asterisk Bruno_Volvo (i=Bruno@81-174-39-150.f5.ngi.it) |
| 10:54.56 | angryuser | canapa: my asterisk is behind nat, have no problems |
| 10:55.00 | Bruno_Volvo | Hi all |
| 10:55.02 | Bruno_Volvo | I need help with dynamic agents, please! I added some agents to agents.conf, I can make agent login on queue, but if I type "show agents" I see alla agents off! How i can configure agents? Very Thanks! |
| 10:55.44 | angryuser | Bruno_Volvo: sorry dont understand a thing about agents;) |
| 10:56.05 | Bruno_Volvo | :) thanks anyway! |
| 10:56.21 | *** join/#asterisk |dennis| (n=dennis@200.32.233.82) |
| 10:57.39 | SheriF_SpacE | i have a small questiin if i have 2 digium card with 6 fxo modules " 2 on TDM400p and the 4 on other it should work using fxsks=1-6 right? |
| 10:57.44 | SheriF_SpacE | in zaptel.conf |
| 10:58.23 | angryuser | SheriF_SpacE: fxs_ks |
| 10:58.52 | *** join/#asterisk blj (n=Balaji@59.92.40.104) |
| 10:59.25 | *** join/#asterisk giasai68 (n=giasai@ip-240-130.sn2.eutelia.it) |
| 10:59.45 | giasai68 | sorry, my connection was down |
| 10:59.48 | SheriF_SpacE | angryuser: zaptel not zapata.conf |
| 11:00.19 | angryuser | SheriF_SpacE: ah sorry, yes;) drinking coffee |
| 11:01.06 | SheriF_SpacE | angryuser: hehe i hope u don't get drunk :P |
| 11:01.11 | SheriF_SpacE | it's okay ;-) thx for help |
| 11:01.22 | SheriF_SpacE | i love genzaptelconf |
| 11:02.57 | giasai68 | how I can know asterisk addons version? |
| 11:04.05 | angryuser | SheriF_SpacE: no risk;) |
| 11:06.26 | *** join/#asterisk phearless (i=[U2FsdGV@host81-148-13-17.in-addr.btopenworld.com) |
| 11:06.40 | phearless | hello ! |
| 11:09.07 | *** part/#asterisk blj (n=Balaji@59.92.40.104) |
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| 11:12.57 | giasai68 | hello |
| 11:13.08 | giasai68 | I have this warning: WARNING[13497]: src/chan_h323.c:977 ooh323_indicate: Don't know how to indicate condition -1 on ooh323c_616 |
| 11:13.17 | giasai68 | any sugget for fix it? |
| 11:13.29 | giasai68 | i'm using asterisk 1.4.x and zaptel 1.4.x |
| 11:14.41 | *** join/#asterisk bmg505 (n=leon@196.209.250.233) |
| 11:16.26 | phearless | anybody know a voip service in UK for asterisk ? with for example 8 phone channels |
| 11:16.27 | angryuser | giasai68: check your zone settings in zaptel.conf loadzone=xx and defaulzone=xx also check that in indications.conf such a country exists |
| 11:17.03 | tzafrir | surelyy unrelated to zaptel |
| 11:17.26 | tzafrir | and I don't think this is related to "indication" as in indications.conf |
| 11:17.53 | tzafrir | giasai68, any idea what triggers that message? |
| 11:18.08 | tzafrir | What version of asterisk-addons is it? |
| 11:18.14 | angryuser | tzafrir: i have the same proble with flash-hook button being pressed, if you can help me out too;) |
| 11:18.33 | giasai68 | asterisk-addons are last version |
| 11:18.34 | tzafrir | What problem? |
| 11:19.03 | giasai68 | angryuser: which zone is better use? |
| 11:19.28 | giasai68 | when user start conversation appear this warning: WARNING[13497]: src/chan_h323.c:977 ooh323_indicate: Don't know how to indicate condition -1 on ooh323c_616 |
| 11:19.41 | angryuser | http://forums.digium.com/viewtopic.php?p=43919 tzafrir my topic on forum |
| 11:21.20 | tzafrir | angryuser, is this an astribank you use to connect to the PSTN? |
| 11:21.23 | angryuser | giasai68: i think tzafrir is right you pb in not related to country, mine, yes |
| 11:21.35 | angryuser | tzafrir: yes |
| 11:21.55 | tzafrir | two things that may be of great help with the FXO module |
| 11:22.00 | tzafrir | 1.: set opermode |
| 11:22.05 | tzafrir | 2 fxotune |
| 11:22.09 | *** join/#asterisk reber (n=reber@cl-158.dub-01.ie.sixxs.net) |
| 11:22.44 | angryuser | tzafrir: not sure whre this option is set, zapata.conf? |
| 11:22.54 | tzafrir | opermode is set by the module init script. In earlier versions it had to be edited in the script itself (/usr/share/zaptel/init_card_4_*) |
| 11:23.20 | tzafrir | In latest version it reads the variable opermode from /etc/sysconfig/zaptel or /etc/default/zaptel |
| 11:23.32 | angryuser | tzafrir: it will give me more info? |
| 11:24.01 | tzafrir | try setting there opermode=FRANCE |
| 11:24.08 | phearless | anybody know a voip service in UK for asterisk ? with for example 8 phone channels |
| 11:24.14 | tzafrir | (you're in France, right?) |
| 11:24.15 | phearless | do I have to search IAX? |
| 11:24.23 | angryuser | tzafrir: ok setting options, yes i am in france |
| 11:24.25 | tzafrir | Next thing is fxotune |
| 11:24.47 | tzafrir | Works with our latest drivers (e.g: zaptel 1.2.13) |
| 11:25.03 | tzafrir | angryuser, sorry, I have to got to lunch now, be back soon |
| 11:25.16 | angryuser | tzafrir: ok thx for help |
| 11:30.17 | giasai68 | I have in indications.conf and in zaptel.conf same zone |
| 11:31.58 | phearless | hello |
| 11:31.59 | phearless | 1 |
| 11:33.50 | angryuser | giasai68: well let's try to do what tzafrir said, you have this problems with fxs clients? |
| 11:41.11 | giasai68 | I have this problem with PRI clients |
| 11:41.24 | giasai68 | warning do not appear for each call |
| 11:42.26 | giasai68 | tzafrir: can let me know any suggest? |
| 11:43.37 | phearless | i found http://www.gradwell.com/voip/asterisk.php but i do not know if we can get more than 1 channel |
| 11:43.38 | *** join/#asterisk WhizzKid (n=apryer@cpc2-lutn6-0-0-cust243.lutn.cable.ntl.com) |
| 11:45.39 | *** join/#asterisk apardo (n=apardo@87.217.144.119) |
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| 11:54.19 | *** join/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 11:56.33 | tzafrir | angryuser, still here? What version of zaptel do you use? |
| 12:01.48 | *** join/#asterisk zotz (n=zotz@24.244.163.157) |
| 12:06.28 | *** join/#asterisk zapp-branigan (n=zapp-bra@81-202-140-56.user.ono.com) |
| 12:07.15 | *** join/#asterisk oej (n=olle@apollo.webway.se) |
| 12:07.16 | zapp-branigan | |
| 12:07.16 | zapp-branigan | hello, somebody has installed |
| 12:07.28 | zapp-branigan | the asterisk in gumstick |
| 12:07.29 | zapp-branigan | ? |
| 12:07.34 | zapp-branigan | gumstix |
| 12:12.22 | giasai68 | tzafrir: can I paste ooh323 debug? |
| 12:12.43 | giasai68 | so u can see whats happned when error appear |
| 12:13.25 | *** join/#asterisk psk (n=psk@golia.caltanet.it) |
| 12:13.36 | tzafrir | giasai68, sorry, I don't have time to delve into ooh323c |
| 12:14.19 | tzafrir | zapp-branigan, also consider asking in #astlinux |
| 12:28.47 | *** join/#asterisk corruptor (n=corrupto@styx.mcn.ru) |
| 12:30.16 | *** join/#asterisk xheliox (n=jeff@pdpc/supporter/active/xheliox) |
| 12:33.58 | xF|Nakaori | how can i tell asterisk to use the voicemailbox instead of telling the caller that i am not available at the moment? |
| 12:35.10 | xF|Nakaori | i mean, if the client is not logged onto asterisk.. |
| 12:36.30 | creativx | extensions.conf |
| 12:36.36 | creativx | check ${dialstatus} |
| 12:37.11 | vibi | has anybody ever connected an old ccm3.3 to asterisk? |
| 12:38.19 | xF|Nakaori | and then? |
| 12:38.24 | xF|Nakaori | thats all? |
| 12:39.06 | tzafrir | run Voicemail(${EXTEN}) or whatever |
| 12:39.23 | xF|Nakaori | what about exten => s-CHANUNAVAIL,1,Voicemail ? |
| 12:39.33 | xF|Nakaori | why doesnt that work? |
| 12:41.46 | phearless | *CLI> pri show span 1 |
| 12:41.46 | phearless | No PRI running on span 1 |
| 12:41.53 | phearless | how can I get the PRI working? |
| 12:42.00 | phearless | I RTFM since 3 days |
| 12:42.01 | x86 | ztcfg/ |
| 12:42.01 | x86 | ? |
| 12:42.41 | phearless | SPAN 1: CCS/HDB3 Build-out: 0 db (CSU)/0-133 feet (DSX-1) |
| 12:42.42 | phearless | and many lines |
| 12:42.42 | phearless | Channel 01: Clear channel (Default) (Slaves: 01) |
| 12:42.42 | phearless | Channel 02: Clear channel (Default) (Slaves: 02) |
| 12:42.42 | phearless | C |
| 12:42.42 | phearless | etc |
| 12:45.09 | xF|Nakaori | ah i got it :> |
| 12:46.45 | phearless | ? |
| 12:49.48 | zamba | what do i need to set up to accept calls coming from the zap channel? |
| 12:50.05 | zamba | i've set up context and defined some extensions for the context in the dialplan.. what more do i need to do? |
| 12:50.16 | zamba | like context=pbx in zapata.conf |
| 12:54.30 | *** join/#asterisk grEvenX (n=even@pc107-130.ktv.no) |
| 12:57.26 | creativx | aha |
| 12:57.59 | creativx | so Queue() with r param (no MOH) will make announce queue pos not work |
| 12:58.01 | creativx | whyyyyyy! |
| 13:00.03 | mafkees | indeed |
| 13:00.13 | mafkees | I found out the hard way too |
| 13:00.21 | creativx | but i dont wanna torture callers with moh |
| 13:00.27 | creativx | but i still want them to know what position they are in |
| 13:00.44 | mafkees | play silence |
| 13:00.53 | mafkees | or record progress sound |
| 13:00.56 | creativx | yea i tried that now |
| 13:00.58 | mafkees | and make that the moh |
| 13:01.02 | creativx | i should get a moh that has a beep |
| 13:01.07 | creativx | silence is just odd |
| 13:01.11 | mafkees | yeah |
| 13:01.17 | creativx | makes the caller wonder |
| 13:01.36 | mafkees | I had to take my moh out |
| 13:01.38 | creativx | you happen to have any good idea how i can make a wav of the ringing() ? |
| 13:01.42 | creativx | why? |
| 13:01.51 | mafkees | my boss didn't like it that I put Cradle of Filth as default moh |
| 13:01.54 | creativx | hahaha |
| 13:02.00 | creativx | well this isnt a production system yet |
| 13:02.07 | mafkees | this was |
| 13:02.07 | creativx | but all my mp3s on the ipod were VBR |
| 13:02.10 | creativx | so asterisk went nuts |
| 13:02.15 | mafkees | yeah |
| 13:02.24 | creativx | and i didnt manage to sox it into something else |
| 13:02.55 | mafkees | use mpg123 to dump it to wav |
| 13:03.00 | mafkees | and lame to remake mp3 |
| 13:04.00 | creativx | gg idea |
| 13:06.52 | creativx | 100 mb drum&bass set must be good moh |
| 13:07.42 | x86 | 8000Hz doesn't do much justice for D&B |
| 13:09.48 | creativx | haha.. good point |
| 13:09.56 | creativx | i cant use mpg123 anyways.. no /dev/dsp |
| 13:11.02 | angryuser | tzafrir: dinner passed well?:) |
| 13:11.36 | tzafrir | yeah |
| 13:11.43 | tzafrir | Some soup |
| 13:12.05 | angryuser | tzafrir: well i did what you told me to do;) |
| 13:12.21 | angryuser | so is set opermode=Mycountry in /etc/sysconfig/zaptel(had to create that file, and executed fxotune -i 4, i got no more messages on zaptel fxs channels "Don't know how to indicate condition 9" and when i press R(flash hook) i got fxs led on astribank blinking like line is went off/on |
| 13:12.39 | tzafrir | it should be in capital letters |
| 13:12.56 | angryuser | tzafrir:it is in capital |
| 13:13.13 | tzafrir | see the names in /usr/share/zaptel/init_fxo_modes |
| 13:15.36 | angryuser | tzafrir:ok names checked, no more errors but, still R flash-hook does not have any effect(on sip it is working) |
| 13:16.34 | tzafrir | What is your value of rxwink? |
| 13:17.35 | tzafrir | You could also enable debugging in the xpd_fxs module, to see if the button indeed sends a on-hook and off-hook signals up |
| 13:17.49 | angryuser | tzafrir:on sip (pap2) R opens another line, rxwink is not set under [channels] |
| 13:18.12 | tzafrir | sorry, I thought you meant zaptel |
| 13:18.21 | zamba | what's the different between trunkgroup and group? |
| 13:19.22 | angryuser | tzafrir:rxwink and flash option should e set under [channels] or [general]? |
| 13:19.32 | tzafrir | channels |
| 13:19.54 | angryuser | tzafrir: il will set it, and test |
| 13:21.17 | *** join/#asterisk oej (n=olle@apollo.webway.se) |
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| 13:29.52 | creativx | hehe.. indeed dnb at 8khz was really bad |
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| 13:34.14 | angryuser | tzafrir: result of tests are: from fxs to fxs(astribank) R does nothing, no effect, from fxs to Sip i got same error message http://www.pastebin.ca/355386 from sip(pap2) feature working |
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| 13:41.23 | juans | I have a conexion h323 whith g729 .. its runs OK... But, in one side of the conexion i dont hear anithing |
| 13:41.31 | juans | Help me please |
| 13:42.36 | *** join/#asterisk putzz (n=me@CPE00155824933c-CM00111ae07ac2.cpe.net.cable.rogers.com) |
| 13:43.00 | *** join/#asterisk wiizard2 (n=nick@87.127.85.43) |
| 13:43.02 | wiizard2 | hi |
| 13:43.08 | wiizard2 | need help with zaptel on fbsd |
| 13:43.15 | *** join/#asterisk Corydon-w (n=tilghman@pdpc/supporter/sustaining/Corydon76-home) |
| 13:43.15 | *** mode/#asterisk [+o Corydon-w] by ChanServ |
| 13:43.22 | wiizard2 | ie. installing the foooker |
| 13:44.54 | wiizard2 | its failing out on me with ztcfg.c:865: error: `ZT_GET_PARAMS_RETURN_MASTER' undeclared (first use in this function) |
| 13:45.36 | *** join/#asterisk appelza (n=pieter@196.40.106.74) |
| 13:46.18 | appelza | hi guys, Im trying to emerge asterisk with the BRI useflag and I recieve : error: 'pri_event' has no member named 'facility' |
| 13:46.21 | appelza | any ideas? |
| 13:47.55 | mafkees | appelza: get the sources from http://www.asterisk.org |
| 13:47.59 | mafkees | compile those |
| 13:48.02 | mafkees | it's not that hard |
| 13:51.47 | creativx | mafkees: did you ever try streaming moh from some mp3 shoutcast source? |
| 13:52.35 | mafkees | nope |
| 13:53.02 | *** part/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net) |
| 13:53.18 | creativx | aight |
| 13:53.37 | creativx | oh well, people just gotta call in and be patient.. hehe |
| 13:54.45 | wiizard2 | is asterisk ok on fbsd 6.1 |
| 13:55.26 | wiizard2 | ah |
| 13:55.33 | wiizard2 | gr8 zaptel is only testing on 5.1-5.2 |
| 13:56.30 | x86 | why can't people just understand that asterisk was designed for linux |
| 13:56.46 | x86 | and that's the best implementation path |
| 13:57.33 | x86 | juans: upgrade to SIP or IAX2 |
| 13:58.01 | juans | x86, i cant upgrade, this is for my job |
| 13:58.18 | wiizard2 | x86 what are u dribbling about |
| 13:58.20 | x86 | juans: the other side wont do SIP? |
| 13:58.35 | juans | x86, the other side have h323 |
| 13:58.47 | x86 | juans: then it's crap and you should get rid of it |
| 13:59.23 | juans | x86, works fine... the only problem is.. in my side i dont hear anything |
| 13:59.59 | x86 | juans: the fundamental problem with h323 is that it's not really a standard, and every vendor has their own implementation... it's almost impossible to properly troubleshoot if two sides "speak h323", but don't speak the same protocol... |
| 14:00.11 | x86 | juans: then it sounds like it doesnt "work fine" ;-) |
| 14:00.44 | x86 | if one of the legs can't hear the other, that's a major problem... not even close to "works fine" |
| 14:01.34 | wiizard2 | wasnt h323 standardised by cisco |
| 14:01.38 | *** join/#asterisk matiasds19 (n=matiasdo@host201.201-252-31.telecom.net.ar) |
| 14:01.43 | wiizard2 | isnt SIP the unstandardised one |
| 14:02.02 | juans | x86, please help me :P |
| 14:02.32 | x86 | i told you how to fix it |
| 14:02.40 | x86 | that's about the only support you'll get here with h323 |
| 14:02.58 | juans | x86, :S |
| 14:03.19 | x86 | wiizard2: google it |
| 14:03.28 | wiizard2 | nah |
| 14:03.34 | juans | x86, Ok.. IE: I have a problem with Linux, Answer X86: Install MS windows, works fine |
| 14:03.49 | wiizard2 | its all a mess in my eyes |
| 14:04.17 | x86 | juans: every vendor has a different implementation of h323... it's called vendor lock-in |
| 14:04.18 | matiasds19 | someone knows how to make asterisk refresh the externalip? beacuase i have the asterisk behind nat, and sometimes my internet provider change my ip, and then i dont have audio |
| 14:04.33 | x86 | juans: solution: don't use it |
| 14:04.42 | x86 | how can it be any more clear cut? |
| 14:04.45 | juans | x86, THIS IS NOT A SOLUTION. |
| 14:04.55 | wiizard2 | lol |
| 14:04.59 | matiasds19 | i have to reset asterisk to refresh it |
| 14:05.05 | x86 | THEN DEAL WITH HALF CONVERSATIONS |
| 14:05.12 | juans | x86, My grandma, have cancer... Solution: Kill them? |
| 14:05.21 | Dr-Linux | cisco cco login |
| 14:05.26 | wiizard2 | give me |
| 14:05.28 | Dr-Linux | CCO stands for? |
| 14:05.32 | x86 | ahh... the beauty of /ignore :) |
| 14:05.37 | wiizard2 | Comminuty Care Office |
| 14:05.43 | Dr-Linux | :S |
| 14:05.43 | wiizard2 | *Officer |
| 14:05.43 | *** part/#asterisk juans (n=jsacco@200.69.219.113) |
| 14:06.20 | Dr-Linux | ok |
| 14:06.21 | x86 | it's sad how people just implement shit without researching it first |
| 14:06.30 | x86 | especially interoperability |
| 14:06.33 | Dr-Linux | i'm going to download cisco sip firmware 5 |
| 14:06.40 | *** join/#asterisk blitzrage (n=blitzrag@asterisk/documenteur-extraordinaire/blitzrage) |
| 14:06.49 | wiizard2 | nah nah nah |
| 14:06.52 | wiizard2 | is this for a 79xx ? |
| 14:07.31 | wiizard2 | Dr-Linux |
| 14:07.59 | Dr-Linux | wiizard2: yes, currently i'm using 7.4 for all of my 79XX |
| 14:08.14 | wiizard2 | yeh thats the shit |
| 14:08.20 | wiizard2 | either 7.4 or 7 |
| 14:08.21 | creativx | intershitoperability, x86 |
| 14:08.26 | wiizard2 | 7.5 is pants so is 8 |
| 14:08.28 | *** join/#asterisk apardo (n=apardo@87.217.144.62) |
| 14:08.41 | Dr-Linux | opss |
| 14:08.47 | Dr-Linux | wiizard2: yes, i meant 8 |
| 14:08.54 | Dr-Linux | wiizard2: did you try 8? |
| 14:09.04 | Dr-Linux | is it out? |
| 14:09.58 | wiizard2 | yes |
| 14:10.01 | wiizard2 | 8 is shite |
| 14:10.03 | wiizard2 | shite shite shite |
| 14:10.21 | Dr-Linux | wiizard2: then which version you recommend? |
| 14:10.42 | Dr-Linux | currently i'm using 7.4 far an year with no issues |
| 14:10.53 | wiizard2 | yeah thats the one |
| 14:11.08 | Dr-Linux | thanks |
| 14:11.09 | wiizard2 | im running that on most of mine and it seems to be the best |
| 14:11.16 | Dr-Linux | wiizard2: great |
| 14:11.19 | wiizard2 | :) |
| 14:11.48 | Dr-Linux | wiizard2: i also want xml phonebook or directory for the 7960 phones |
| 14:11.51 | wiizard2 | heh |
| 14:11.54 | wiizard2 | dont go there |
| 14:12.01 | Dr-Linux | i see |
| 14:12.09 | wiizard2 | unless you wish to lose about 5 months of your life |
| 14:12.11 | Dr-Linux | ok, but i'd ask the reason |
| 14:12.19 | Dr-Linux | aww |
| 14:12.43 | wiizard2 | Tbh, ive never spent the time to look into it.. It was hard enough trying to get an Image on the phone |
| 14:13.00 | wiizard2 | like a .bmp image |
| 14:13.00 | Dr-Linux | actually i don't want add manually on each phone, i'm looking for someway, so i can grab the phonebook or directory from the tftp |
| 14:13.09 | wiizard2 | well |
| 14:13.12 | wiizard2 | its all done via xml |
| 14:13.26 | Dr-Linux | wiizard2: yeah, that's what i'm asking |
| 14:13.27 | wiizard2 | you would specifiy the xml directory in the SIPxxxxxxxxxxxxxxxx.cnf |
| 14:13.33 | wiizard2 | BUT |
| 14:13.34 | Ahrimanes | dr-Linux i have a php xml phonebook for 7940 |
| 14:13.42 | wiizard2 | i think that is a cisco call manager trick |
| 14:13.58 | Dr-Linux | Ahrimanes: can i see it? :$ |
| 14:14.00 | wiizard2 | Ahrimanes you do?!?!? |
| 14:14.03 | wiizard2 | :) |
| 14:14.10 | Ahrimanes | yup |
| 14:14.12 | Ahrimanes | hm |
| 14:14.13 | wiizard2 | can i see it aswell? :$ |
| 14:14.22 | Ahrimanes | gimme a sec to dig it out from the ceiling, hehe |
| 14:14.23 | Dr-Linux | Ahrimanes: thanks friend |
| 14:14.30 | wiizard2 | nice |
| 14:14.34 | wiizard2 | might be worth looking into |
| 14:14.38 | Dr-Linux | sure |
| 14:14.40 | Ahrimanes | dr-Linux directories and then press 5 thingy, right? |
| 14:15.02 | Ahrimanes | external directory |
| 14:15.04 | Dr-Linux | Ahrimanes: i never use it, but i want to use it for every phone |
| 14:15.12 | Ahrimanes | yup |
| 14:15.14 | wiizard2 | i think u need to setup an http proxy to use it also |
| 14:15.39 | Ahrimanes | this one runs from an apache with php and mysql to present the database as xml |
| 14:15.41 | *** join/#asterisk dasenjo (n=dasenjo@190.24.176.242) |
| 14:15.45 | Ahrimanes | brb |
| 14:15.55 | Dr-Linux | ok |
| 14:16.10 | Dr-Linux | will it place it on some linux box? |
| 14:16.42 | nas_lslsa | when I am in CLI console , and I do module show |
| 14:17.19 | nas_lslsa | is it normal that I don't see any module loaded ? I am trying to connect an sjphone ( voip software ) with asterisk , without any luck yet .. |
| 14:19.29 | Ahrimanes | dr-Linux you can place it anywhere the phone can reach via http |
| 14:20.45 | Dr-Linux | Ahrimanes: ok great, then i'll place it at local Linux server which has PHP, web etc |
| 14:20.55 | Ahrimanes | directory_url: "http://webserver/path/ |
| 14:21.01 | Ahrimanes | directory_url: "http://webserver/path/" |
| 14:21.08 | Ahrimanes | in the cisco tftp config file |
| 14:21.59 | Dr-Linux | Ahrimanes: in which file at cisco config? SIPXXXXXXXXXXXX.cnf ? |
| 14:22.07 | Ahrimanes | yup |
| 14:22.12 | Dr-Linux | ok |
| 14:23.01 | Ahrimanes | oh |
| 14:23.05 | Ahrimanes | this is clever actually |
| 14:23.06 | Ahrimanes | hehe |
| 14:23.10 | Ahrimanes | also works for snom 360 |
| 14:23.20 | Dr-Linux | ok |
| 14:23.22 | Ahrimanes | although the t9 xml phonebook for snom 360 i nicer |
| 14:23.27 | Dr-Linux | i have all 7960's |
| 14:24.16 | Ahrimanes | where do you save your contact info? mysql db? |
| 14:24.56 | Dr-Linux | Ahrimanes: where should i? in xml file or mysql DB? |
| 14:25.18 | Dr-Linux | i can do wherever it works good |
| 14:25.23 | Ahrimanes | dr-Linux well this php script reads a mysql db and outputs xml for the cisco |
| 14:25.36 | *** join/#asterisk cpm (n=Chip@pdpc/supporter/active/cpm) |
| 14:26.19 | Dr-Linux | Ahrimanes: ok i'll do that |
| 14:26.34 | Dr-Linux | but maybe i'd need your guideness :) |
| 14:26.51 | Dr-Linux | Ahrimanes: can i /msg you ? |
| 14:29.22 | *** join/#asterisk coppice (n=chatzill@13.168.17.210.dyn.pacific.net.hk) |
| 14:30.11 | *** join/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net) |
| 14:30.13 | *** mode/#asterisk [+o mog] by ChanServ |
| 14:30.52 | *** join/#asterisk [TK]D-Fender (n=joe@MTRLPQ02-1177745839.sdsl.bell.ca) |
| 14:31.02 | Ahrimanes | http://www.bsd-dk.dk/~aron/phonebook.txt <- save this as phonebook.php, correct database info and point directoryurl: "http://webserver/phonebook.php/cisco" |
| 14:31.06 | Ahrimanes | dr-Linux sure |
| 14:31.23 | Dr-Linux | ok thanks |
| 14:31.30 | Ahrimanes | wiizard2: try that out |
| 14:31.36 | Ahrimanes | dr-Linux feel free to /msg me |
| 14:33.49 | *** join/#asterisk parag_ast (n=Administ@dxb-b110166.alshamil.net.ae) |
| 14:33.50 | parag_ast | ls |
| 14:34.00 | parag_ast | hiii |
| 14:34.05 | Ahrimanes | wiizard2: and please do provide feedback |
| 14:35.46 | vibi | infos about oh323 and ccm 3.3 connect? |
| 14:36.22 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 14:38.11 | *** join/#asterisk etfonhomey (n=etfonhom@74-140-206-4.dhcp.insightbb.com) |
| 14:38.46 | yansolo90 | hey, anybody knows what login/password are for ssh Cisco 79XX (other than "debug/debug" or "log/log") ? |
| 14:38.48 | *** join/#asterisk orkid (n=orkid@dataq2.utias.utoronto.ca) |
| 14:38.56 | parag_ast | admin admin |
| 14:39.07 | parag_ast | or user/admin |
| 14:40.20 | xermesx | every time i login to asterisk-gui, asterisk 1.4 goes in segmentation fault |
| 14:43.16 | *** join/#asterisk |dennis| (n=dennis@200.32.233.82) |
| 14:44.35 | *** join/#asterisk anthm (n=anthm@CPE-72-131-113-50.wi.res.rr.com) |
| 14:44.35 | *** mode/#asterisk [+o anthm] by ChanServ |
| 14:45.59 | *** join/#asterisk docelmo (n=vircuser@c-68-80-121-53.hsd1.de.comcast.net) |
| 14:46.02 | docelmo | Anyone know if FastAGI is broken in 1.4? |
| 14:46.14 | sweeper | docelmo: I highly doubt it |
| 14:46.27 | *** join/#asterisk heison (n=heison@ns.somanetworks.com) |
| 14:46.32 | sweeper | seems it would be one of those critical things... |
| 14:46.34 | docelmo | Im trying to EXEC DIAL SIP/XXX@YYY and sip keeps bitching it cant find YYY |
| 14:46.59 | sweeper | well, try doing it from the cli |
| 14:47.03 | docelmo | what? |
| 14:47.10 | docelmo | exec command? |
| 14:47.13 | docelmo | is that possible? |
| 14:47.19 | sweeper | just do dial |
| 14:47.24 | sweeper | but yea |
| 14:47.38 | docelmo | I need to read up on 1.4.. |
| 14:47.45 | sweeper | anything you can run in the dialplan, (and thus, through AGI) can be done from the cli |
| 14:47.59 | [TK]D-Fender | docelmo: as a sanity check how about you post the actual attempt & message... |
| 14:48.16 | Ahrimanes | dialplan functions can be run from cli ? |
| 14:48.37 | [TK]D-Fender | Ahrimanes: No, I think he meant AGI |
| 14:48.55 | Ahrimanes | [TK]D-Fender: ok, but he did say cli.. |
| 14:49.09 | docelmo | [TK]D-Fender droping it on pastebin now |
| 14:49.11 | [TK]D-Fender | Ahrimanes: Thats why I corrected for what he MEANT :) |
| 14:49.19 | Ahrimanes | :) |
| 14:49.22 | *** join/#asterisk drako (n=ljd@unaffiliated/luisjose) |
| 14:50.44 | docelmo | http://www.pastebin.ca/355491 |
| 14:51.11 | *** join/#asterisk Skaag (n=hintza@212.199.180.157.static.012.net.il) |
| 14:51.30 | *** join/#asterisk jeebusroxors (n=jeebusro@cpe-75-80-248-142.dc.res.rr.com) |
| 14:51.39 | docelmo | [TK]D-Fender any ideas? |
| 14:51.48 | Skaag | Ok, I have ztdummy, and i've defined two IAX2 peers/trunks, but when I show peers, the status is unmonitored - what does that mean? |
| 14:52.08 | [TK]D-Fender | docelmo: And what exactly is YYY? |
| 14:52.09 | Skaag | [TK]D-Fender: by the way I found out that creating the ztdummy kernel module is a 20 second process |
| 14:52.15 | Skaag | at least on ubuntu |
| 14:52.24 | [TK]D-Fender | Skaag: Good, now stop being such a pessimist ;) |
| 14:52.41 | [TK]D-Fender | Skaag: And its a 20 second process on any sane system |
| 14:53.24 | Skaag | cool :) |
| 14:53.37 | *** join/#asterisk angryuser (n=Miranda@i03v-213-44-169-43.d4.club-internet.fr) |
| 14:53.49 | Skaag | now just to understand how to hook up this trunk i bought |
| 14:54.04 | Skaag | I should be able to dial it, and it would invoke a specific extension on my asterisk |
| 14:54.08 | Skaag | then I can start doing the IVR |
| 14:54.09 | Skaag | :) |
| 14:54.16 | Skaag | exciting times for me. |
| 14:54.37 | angryuser | can somebody help me with my call forwarding mini script? http://www.pastebin.ca/355211 i am unable to register values in astdb |
| 14:56.17 | [TK]D-Fender | angryuser: exten => _**21X.,1,Set(DB(users/${CALLERID(num))=${EXTEN:4}) ;Set the value to be forwarded to, user dials **21+ number to be ;forwarded to |
| 14:56.26 | [TK]D-Fender | angryuser: Bad brackets. Pay attention. |
| 14:56.34 | [TK]D-Fender | angryuser: And braces. |
| 14:56.41 | docelmo | [TK]D-Fender any ideas? |
| 14:56.53 | [TK]D-Fender | docelmo: I asked you a question..... |
| 14:57.16 | [TK]D-Fender | docelmo: And I have a pretty good idea I have no reason to trust what I see.... |
| 14:57.17 | docelmo | sorry.. YYY is a peer in sip.conf which I did sip show peers and its there |
| 14:57.20 | angryuser | [TK]D-Fender:hm ok correcting thank you |
| 14:58.04 | [TK]D-Fender | docelmo: Show me the WHOLE thing, and show that the peer is valid. |
| 14:58.17 | [TK]D-Fender | docelmo: Heck, enable SIP debug while you're at it. |
| 14:58.30 | docelmo | will do.. |
| 14:58.40 | [TK]D-Fender | docelmo: Trust is an inverse coefficient to what I see :) |
| 14:59.06 | [TK]D-Fender | (I think I just got that a little backwards, but you get the picture) |
| 15:00.07 | angryuser | [TK]D-Fender:i have used example from wiki, whick brackets shoul i change? |
| 15:00.13 | angryuser | which? |
| 15:00.25 | FreezeS | anybody has any idea if it's possible to combine DUNDi with queues ? |
| 15:00.39 | [TK]D-Fender | angryuser: You should already know how to call functions... just re-read it till your eyes bleed. |
| 15:01.16 | angryuser | [TK]D-Fender:i cant see anything;) i am blind fender you made me blind:) |
| 15:01.30 | angryuser | seriosly gone to read |
| 15:01.49 | [TK]D-Fender | angryuser: Good, and you were already deaf, now if only you were MUTE, we'd all sleep easier ;) |
| 15:02.13 | angryuser | [TK]D-Fender: oh sorry to woke your hibernation;) |
| 15:02.22 | sweeper | [TK]D-Fender: errr, you can run Dial commands from the cli, I thought... |
| 15:02.28 | docelmo | [TK]D-Fender there is no sip debug |
| 15:02.38 | docelmo | it fails before it even tries to dial |
| 15:02.44 | [TK]D-Fender | docelmo: *huh* ? |
| 15:02.53 | docelmo | the call from FastAGI |
| 15:02.55 | nas_lslsa | how can I see if my Asterisk supports SIP or not ?! |
| 15:02.55 | sweeper | he doesn't get any sip output |
| 15:03.01 | [TK]D-Fender | docelmo: Ok, well show me the rest exactly as it is. |
| 15:03.23 | docelmo | basicly call comes in goes to fastagi works asks for DTMF then does a DIAL command.. it fails there |
| 15:03.34 | Defend | anyone heard of any issues with snom and getting this error? 481 Call Leg/Transaction Does Not Exist |
| 15:03.36 | FreezeS | nas_lslsa: do you have /etc/sip.conf ? |
| 15:03.40 | docelmo | k hold on |
| 15:03.42 | nas_lslsa | yes |
| 15:03.43 | [TK]D-Fender | nas_lslsa: Unless you've done something remarkably silly, SIP is always included in all compiles. But I think TRYING it would be a good test. |
| 15:03.56 | Bobthehunter | is e164 have a 1 or not in front |
| 15:04.20 | FreezeS | [TK]D-Fender: if you don't run make samples, the sip module isn't loaded |
| 15:04.47 | nas_lslsa | I tried a lot without any luck .. and I am trying commands at CLI> like sip show users and says No such command :.. |
| 15:05.03 | docelmo | Bobthehunter huh? |
| 15:05.07 | FreezeS | nas_lslsa: did you run "make samples" after "make" ? |
| 15:05.15 | nas_lslsa | I made the default configure / make |
| 15:05.16 | FreezeS | sorry, "make install" |
| 15:05.23 | nas_lslsa | yes , I did that .. |
| 15:05.27 | *** join/#asterisk codazoda (n=chatzill@mail.hurdmanivr.com) |
| 15:05.30 | *** join/#asterisk wherethebandsr (n=warren@75.112.128.194) |
| 15:05.33 | FreezeS | so you do have sip.conf ? |
| 15:05.51 | nas_lslsa | yes , .. and I vi it many times .. |
| 15:05.58 | [TK]D-Fender | FreezeS: That has no impact on whether SIP is installed and usable... |
| 15:06.08 | nas_lslsa | and I tried with sjphone to connect to asterisk .. |
| 15:06.21 | *** join/#asterisk crich1999 (n=crich@pd956852e.dip0.t-ipconnect.de) |
| 15:06.25 | FreezeS | [TK]D-Fender: actually, for me it had. Without it, when trying "sip show peers", I got an error |
| 15:06.32 | codazoda | My system has a lot of noise on my analog lines. I replaced an older phone system that didn't have the noise, so I think it's coming from the Asterisk system itself. The noises are a sligh "hiss" in the background, clicks, and volume drops. Any idea what it might be? |
| 15:06.52 | [TK]D-Fender | FreezeS: Perhaps you could pastebin the error so we can all see.... |
| 15:06.54 | [TK]D-Fender | ~pb |
| 15:07.04 | jbot | rumour has it, pb is a place to paste your stuff without flooding the channel - try http://pastebin.ca, or http://channels.debian.net/paste |
| 15:07.12 | [TK]D-Fender | codazoda: First suspect : your card |
| 15:07.32 | FreezeS | [TK]D-Fender: my problem now is that I don't know if it's possible to use a context or macro instead of member => in queues |
| 15:07.35 | *** join/#asterisk zotz (n=zotz@24.244.163.157) |
| 15:07.44 | codazoda | I originally had a PIKA card. I thought the same thing, so I purchased a TDM2400E and replaced the PIKA. The problem is actually a little worse with the Digicom card. |
| 15:08.24 | codazoda | Err. Digum. :-| |
| 15:08.35 | codazoda | Okay, so it's early and I can't type either. |
| 15:09.25 | angryuser | [TK]D-Fender: yessss it is working;) i was 1 bracked away from the solution |
| 15:09.30 | docelmo | codazoda as least your not dealing with a Ice Storm and 1/2 inch of ice |
| 15:09.41 | *** join/#asterisk brussel_ (n=brussel@cpe-24-165-1-241.san.res.rr.com) |
| 15:09.54 | [TK]D-Fender | FreezeS: How about showing use what you did? |
| 15:09.56 | codazoda | Docelmo: No, no I'm not. We've got nice weather out this way. Should be close to 50 today. |
| 15:10.00 | *** join/#asterisk wunderkin (i=kev@ip72-208-3-221.ph.ph.cox.net) |
| 15:10.13 | *** join/#asterisk awannabe (n=gti@ip24-251-135-202.ph.ph.cox.net) |
| 15:10.17 | docelmo | codazoda where are you? |
| 15:10.28 | codazoda | Docelmo: I'm in Salt Lake City, Utah. |
| 15:11.28 | docelmo | lucky bastard.. Im in delaware.. trade ya at this point |
| 15:11.35 | FreezeS | [TK]D-Fender: I didn't do anything right now. I just want to know if it's possible to use a script or context instead of a normal member in a queue |
| 15:11.38 | docelmo | Also thats where my school is |
| 15:11.57 | awannabe | [TK]D-Fender: thanks for help yesterday, i apperciate it :) |
| 15:12.17 | *** join/#asterisk [[blah]asfd (n=ckwall@63.149.122.93) |
| 15:12.18 | [TK]D-Fender | FreezeS: Use a local channel as a member. |
| 15:12.36 | [TK]D-Fender | FreezeS: And what would you intend to DO in there? |
| 15:12.46 | FreezeS | [TK]D-Fender: DUNDi lookups :) |
| 15:12.47 | [TK]D-Fender | awannabe: Npp.... what did I help you with again? :) |
| 15:12.58 | [TK]D-Fender | FreezeS: Thats all fine. |
| 15:13.15 | awannabe | [TK]D-Fender: the hotdesking/agentcallbacklogin() i think i almost got it, almost! heh |
| 15:13.16 | codazoda | Any other thoughts on my noise? What should I look for? System load is good. Sound is good if I plug into the lines with a typical analog phone. |
| 15:13.27 | [TK]D-Fender | awannabe: Oh yeah.... |
| 15:13.30 | FreezeS | [TK]D-Fender: how can I do it ? |
| 15:13.47 | [TK]D-Fender | codazoda: pastebin "cat /proc/interrupts" |
| 15:14.01 | FreezeS | normally I use member sip/user or zap/channel# |
| 15:14.11 | [TK]D-Fender | FreezeS: add "member => Local/123@context" and do what you want in there |
| 15:14.23 | FreezeS | ok, that's sweet :) |
| 15:14.24 | FreezeS | thanks |
| 15:14.52 | codazoda | [TK]D-Fender: my /proc/interrupts is here: http://www.pastebin.ca/355517 |
| 15:15.23 | [[blah]asfd | with queue() I guess you can SetVar(QUEUE_PRIO=10) to change the priority of the call before it enters the queue... Am i understanding that correctly? So if I have a call that comes in to a queue without the QUEUE_PRIO and one that I set a QUEUE_PRIO value, I can have that call go before the call without a priority in the queue? |
| 15:15.26 | [[blah]asfd | is that right? |
| 15:15.30 | [TK]D-Fender | codazoda: Ok, looks clean... |
| 15:16.05 | giasai68 | hello |
| 15:16.21 | giasai68 | how I can use g729r8 codec in asterisk 1.4.x? |
| 15:16.29 | Qwell | r8? |
| 15:19.01 | *** join/#asterisk drako (n=ljd@unaffiliated/luisjose) |
| 15:19.03 | xermesx | every time i login to asterisk-gui, asterisk 1.4 goes in segmentation fault |
| 15:20.37 | giasai68 | yes... do u know it? |
| 15:20.54 | Qwell | giasai68: never heard of it. how is it different from regular g729? |
| 15:22.46 | *** join/#asterisk mercestes (n=merceste@rrcs-71-41-157-70.sw.biz.rr.com) |
| 15:26.54 | nas_lslsa | sould I change linux ? I have Suse now ... |
| 15:29.05 | nas_lslsa | when I do CLI> module reload I can't see the sip.conf to the list files it parses .. |
| 15:30.01 | [TK]D-Fender | nas_lslsa: try "sip reload" |
| 15:30.15 | nas_lslsa | No such command 'sip reload' (type 'help' for help) |
| 15:30.17 | nas_lslsa | :( |
| 15:30.26 | codazoda | If anyone is using an analog (FXO) card, what type and how does it work for you? |
| 15:30.39 | cpm | reload sip |
| 15:31.03 | nas_lslsa | No such module 'sip' .... hmmmm ... |
| 15:31.19 | nas_lslsa | so I have to recompile it ? / install and so ? |
| 15:33.16 | *** join/#asterisk marv[work] (n=timr@24.214.206.254) |
| 15:33.20 | *** join/#asterisk putzz (n=me@CPE00155824933c-CM00111ae07ac2.cpe.net.cable.rogers.com) |
| 15:33.31 | x86 | reload chan_sip.so |
| 15:33.56 | nas_lslsa | same : No such module 'chan_sip.so' |
| 15:34.04 | x86 | load chan_sip.so then |
| 15:34.07 | putzz | im trying to call local numbers without having to dial the 1 but the carrier requires 1 to be sent out in front of are code, and I cant remember how to do it anyone wanna refresh my memory? |
| 15:34.31 | x86 | putzz: voip-info.org, search for dialplans |
| 15:35.31 | docelmo | dial(SIP/1${EXTEN}@peer) |
| 15:35.37 | docelmo | putzz there ya go |
| 15:35.46 | putzz | thanx boss |
| 15:36.00 | putzz | that was easy but for some reason I could not remember |
| 15:36.14 | Skaag | I setup an IAX peer but it is unmonitored, how do I get it to connect? |
| 15:37.11 | blitzrage | qualify=yes if you want to see monitoring |
| 15:37.25 | blitzrage | unmonitored does not mean disconnected |
| 15:38.23 | nas_lslsa | seems it worked ! ! ! ! ! ! !:)! :!) !:!) !:) !:)! !:) !:)! !:!) !!!L: |
| 15:38.47 | nas_lslsa | the loading .. now I have to check if it works with clients , thanks :)* see ya ! |
| 15:42.06 | *** join/#asterisk jcmoore (n=jcmoore@unaffiliated/tgrman) |
| 15:42.07 | *** join/#asterisk JoNate (n=noone@mail.wmelec.com) |
| 15:45.23 | *** join/#asterisk giasai68 (n=giasai@ip-240-130.sn2.eutelia.it) |
| 15:49.40 | *** join/#asterisk websae (n=websae@adsl-75-48-246-49.dsl.milwwi.sbcglobal.net) |
| 15:50.59 | *** join/#asterisk angler (i=angler@pdpc/sponsor/digium/angler) |
| 15:50.59 | *** mode/#asterisk [+o angler] by ChanServ |
| 15:52.18 | *** join/#asterisk ChicagoBud (n=Bud@adsl-70-228-35-78.dsl.chcgil.ameritech.net) |
| 15:53.56 | websae | does anyone have a recommendation for DID provider that doesn't charge per minute? |
| 15:56.11 | putzz | there r a few |
| 15:56.24 | putzz | but the ones that dont charge u per minute usualy limits u to 1-2 channels |
| 15:56.34 | putzz | per min usually has unlimited channels |
| 15:56.45 | websae | i don't mind 1-2 channels |
| 15:56.52 | websae | telasip use to be really inexpensive |
| 15:57.02 | websae | but now it's like $10 |
| 15:57.07 | websae | when it used to be like $4 |
| 15:57.21 | putzz | well how many are you looking for just 1? |
| 15:57.42 | websae | looking for 3-4 DIDs/ 3-4 channels |
| 15:58.33 | putzz | les.net has good prices for dids |
| 15:58.35 | putzz | and they r also good |
| 15:58.42 | websae | hehe, is that you? |
| 15:58.50 | putzz | nop |
| 15:59.03 | ChicagoBud | I need some advice on how to handle a situation. I have one phone that is shared by 4 people (mostly in shifts but sometimes overlapping) - this phone also serves as "operator" since someone is usually there. Should I have the users setup as agents and have them login? |
| 15:59.29 | *** join/#asterisk PakiPenguin (n=uppal@linuxpakistan/admin/pakipenguin) |
| 15:59.32 | PakiPenguin | hello everyone |
| 15:59.56 | Skaag | hi PakiPenguin |
| 16:00.02 | ChicagoBud | the phone can have 4 line appearances so I could set up a line for each person. but then how do I handle the operator? |
| 16:00.18 | xermesx | [Feb 14 14:05:14] WARNING[12252]: pbx.c:1755 pbx_extension_helper: No application 'Cut' for extension (from-pstn-reghours, s, 5) |
| 16:01.28 | websae | putzz: I wish I could see their rate centers |
| 16:01.47 | Bobthehunter | whos |
| 16:01.57 | Bobthehunter | putzz, where you need your did ? |
| 16:01.59 | websae | les.net |
| 16:02.08 | Bobthehunter | lol ok |
| 16:02.39 | Bobthehunter | i work with www.voicemeup.com and could arange a flat monthly phone number i guess where you need the did from ? |
| 16:03.57 | wunderkin | xermesx, check UPGRADE.txt... it is function cut now |
| 16:05.12 | *** join/#asterisk ledoktre (n=x@209.152.67.6) |
| 16:05.14 | phearless | CLI> pri show span 1 |
| 16:05.14 | phearless | No PRI running on span 1 |
| 16:05.17 | Bobthehunter | seems what you need is a monthly did subsription ;) |
| 16:05.17 | ledoktre | hello |
| 16:05.19 | phearless | how can I activate it ? |
| 16:05.23 | phearless | please :) :) |
| 16:06.06 | *** part/#asterisk [[blah]asfd (n=ckwall@63.149.122.93) |
| 16:06.54 | ledoktre | does anyone know here why I can dial but not hear asterisk -> sip -> asterisk, and udp5060 and rtp10000:20000 are open on the firewall? |
| 16:07.18 | Bobthehunter | ledoktre, sip debug , check the sdp packets |
| 16:08.04 | matiasds19 | maybe you have to set externalip in sip.conf |
| 16:08.10 | aydiosmio | anyone know if the new Digium TC400B can hirpin to do g729 <-> g723 transcoding? |
| 16:08.49 | Qwell[] | aydiosmio: I want to say no, but your best bet would be to call and ask sales |
| 16:09.03 | aydiosmio | excellent idea |
| 16:09.21 | *** join/#asterisk elriah (i=elriah@adsl-072-149-159-016.sip.bhm.bellsouth.net) |
| 16:09.25 | aydiosmio | excuse the op/ed, but digium kind of dropped the ball there (plus no gsm) |
| 16:09.55 | elriah | Hi all. Anyone know how to get Cisco phones working behind NAT with an Asterisk server on a public IP (without doing port forwarding, we have a number of phones so that won't work)? |
| 16:10.27 | *** join/#asterisk eald (n=eald@189.157.105.134) |
| 16:12.04 | aydiosmio | why would oyu port forward? |
| 16:12.27 | aydiosmio | if the asterisk server is at the edge of the network, you'd have two call legs |
| 16:12.37 | aydiosmio | and the asterisk server would manage the translations |
| 16:12.56 | *** join/#asterisk russellb (n=russell@asterisk/developer-and-stable-maintainer/drumkilla) |
| 16:12.56 | *** mode/#asterisk [+o russellb] by ChanServ |
| 16:13.03 | blitzrage | russellb: oh snap |
| 16:13.24 | russellb | oh? |
| 16:13.34 | russellb | what'd i do now? |
| 16:13.47 | blitzrage | russellb: you created a wicked function -- nice moves |
| 16:13.59 | russellb | oh :-D |
| 16:14.42 | russellb | took about half a day from start to commit :) |
| 16:15.08 | awannabe | anyone use asterfax in here? |
| 16:15.28 | elriah | awannabe: Use IAXmodem and a bash script, works great, never fails. |
| 16:15.51 | [TK]D-Fender | phearless: You might want to start by correctly configuring zaptel.conf and zapata.conf. Also helps if you compiled libpri before Zaptel & Asterisk. |
| 16:15.59 | awannabe | well, let me rephrase, I want fax2email :) |
| 16:16.25 | elriah | Anyone know how to enable DSCP on polycom phones from the xml config files? |
| 16:16.31 | aydiosmio | anyone have digium's sales number? |
| 16:16.42 | Defend | our calls seem to randomly drop once in a while and i keep seeing Got SIP response 481 "Call Leg/Transaction Does Not Exist do you think this might have somethign to do with it and if so do you think it is a sip bug or cfg issue or soemthing else? |
| 16:16.46 | aydiosmio | n/m found it |
| 16:16.51 | elriah | awannabe: Same deal, just use IAXmodem and a bash script.. easy.. just hit google, there are a bunch of examples... |
| 16:17.06 | awannabe | does it work well? |
| 16:17.24 | [TK]D-Fender | elriah: DSCP? |
| 16:17.29 | awannabe | im looking to install in a office, and have each user have their own fax number that points to their personal email |
| 16:17.38 | phearless | [TK]D-Fender, thank you, I compiled libpri before zaptel/asterisk, or I would not be able to use the pri command |
| 16:17.40 | awannabe | [TK]D-Fender: diff serv code priorities |
| 16:17.59 | [TK]D-Fender | phearless: Ok, then move on to reviewing your config files... |
| 16:18.02 | phearless | [TK]D-Fender, I will paste my zaptel and zapata config |
| 16:18.09 | phearless | 1s |
| 16:18.11 | awannabe | its for QoS, but if you have good switches just let them do all the QoS for you :) |
| 16:18.14 | [TK]D-Fender | phearless: And verifying things like "is it plugged in?" |
| 16:21.18 | ledoktre | bobthehunter, matiasds19: I tried the externalip, did not seem to do it. tried enabling debugging, but im just not sure what im looking for. On the first side, I am connecting to the asterisk server with a aastra phone or xlite, and canreinvite=no. this is behind a router but the udp ports are forwarded to the * box. the problem I have is when dialling another * box, which is not behind a nat. ? |
| 16:21.53 | [TK]D-Fender | ledoktre: pastebin the [general] section of your sip.conf |
| 16:22.21 | phearless | [TK]D-Fender, http://paste.lisp.org/display/36862 |
| 16:22.31 | angryuser | when i receive the call from anywhere sip/Rnis/fxs/fxo, the variable of the person being called is ${DIALEDPEERNUMBER} ? |
| 16:22.39 | phearless | [TK]D-Fender, yes the things are plugged in, I checked 10 times |
| 16:22.45 | aydiosmio | it's kind of funny calling places I know use * pbxes |
| 16:22.56 | aydiosmio | like it's an old friend directing my call. |
| 16:23.09 | angryuser | in other way to say i need the $var of the person being called |
| 16:23.37 | elriah | I found the Polycom attributes, just can't figure out where they go in the XML files... For QoS... |
| 16:24.10 | Bobthehunter | anyway to decide to record a channels AFTER its been bridged ? |
| 16:24.12 | [TK]D-Fender | phearless: "cat /proc/interrupts" |
| 16:24.25 | [TK]D-Fender | Bobthehunter: features.conf |
| 16:24.30 | aydiosmio | sales guy didn't know if g729 <> g711 <> g723 would work-- he's off asking the engineers |
| 16:24.43 | Bobthehunter | if im not in the leg |
| 16:24.50 | Bobthehunter | automon cant apply |
| 16:25.15 | [TK]D-Fender | Bobthehunter: You mean you want to start recording someone elses call in mid-stream? |
| 16:25.21 | Bobthehunter | yes lol |
| 16:25.32 | [TK]D-Fender | Bobthehunter: Look at the AMI. there is an option for this |
| 16:25.35 | Bobthehunter | basically |
| 16:25.38 | Bobthehunter | AMI ? oh |
| 16:25.47 | Qwell[] | aydiosmio: g729 <> g711 <> g723 will absolutely work |
| 16:25.56 | Qwell[] | it's g729 <> g723 that I'm unsure of |
| 16:26.12 | aydiosmio | digium engineer says you can do g723 <> g729 with two channels |
| 16:26.20 | Qwell[] | direct? |
| 16:26.20 | aydiosmio | yeah |
| 16:26.21 | elriah | Does anyone know where the QOS settings go in Polycom xml config files? |
| 16:26.24 | Qwell[] | I mean, all through the card? |
| 16:26.24 | aydiosmio | no not directly |
| 16:26.35 | *** join/#asterisk Creathir (n=Creathir@mobile-166-214-199-213.mycingular.net) |
| 16:26.44 | Qwell[] | okay, so it'll have to come out twice - that's what I would've assumed |
| 16:26.50 | ledoktre | TDK-Fender : http://paste.uni.cc/13172 |
| 16:27.00 | Creathir | Hey everyone, quick question: |
| 16:27.22 | [TK]D-Fender | elriah: Perhaps you could actually look in sip.cfg.... |
| 16:27.42 | Creathir | I have a server I put together with a Digium POTS card, and when I make a call out, it rings the recieving party |
| 16:27.43 | Creathir | but |
| 16:27.49 | phearless | <[TK]D-Fender> phearless: "cat /proc/interrupts" --> http://paste.lisp.org/display/36862#1 |
| 16:27.54 | Creathir | when they pick up, the card never picks up |
| 16:28.22 | [TK]D-Fender | ledoktre: First of all its "externip", not "externalip", and you are missing "nat=yes", "canreinvite=no", and your "localnetnet=" statements. |
| 16:28.53 | [TK]D-Fender | phearless: Card is sharing IRQ... not good. What do you get on ztcfg -vvvv? |
| 16:29.00 | elriah | Oh geez. I was looking for QOS and it's qos .. lol .. vi is so picky |
| 16:29.33 | phearless | [TK]D-Fender, http://paste.lisp.org/display/36862#2 |
| 16:29.41 | phearless | for ztcfg |
| 16:30.02 | [TK]D-Fender | phearless: and "zap show channels" followed by "pri show span 1"? |
| 16:30.12 | sevard | elriah: i hink you can do :ic to ignore case, but i'm not a vi guy |
| 16:30.46 | phearless | [TK]D-Fender, http://paste.lisp.org/display/36862#3 |
| 16:33.19 | [TK]D-Fender | phearless: IIRC its "channel" not "channels" in zapata.conf... |
| 16:33.49 | joe | is there a command at the cli to output what it things the voicemail confs are? |
| 16:34.16 | ledoktre | TKD-Fender : thank you for the info. I had tried several of those options one at a time, but not all of them (the one that seemed to do it was localnet). thanks, works good!! |
| 16:34.21 | Creathir | any ideas anyone? |
| 16:34.30 | phearless | <[TK]D-Fender> phearless: IIRC its "channel" not "channels" in zapata.conf... > right |
| 16:34.33 | phearless | I correct it... |
| 16:34.37 | joe | nm |
| 16:34.57 | phearless | Feb 14 16:34:45 ERROR[5004]: chan_zap.c:10460 setup_zap: Unable to reconfigure channel '1-8' |
| 16:34.58 | phearless | oh. |
| 16:35.08 | phearless | (when I reload asterisk) |
| 16:35.19 | [TK]D-Fender | :/ |
| 16:35.46 | xermesx | exten => s,5,Set(intype=$CUT(intype|-|1)) |
| 16:35.56 | xermesx | is the sintax right ? |
| 16:36.09 | Qwell[] | xermesx: I think the last two args are backwards |
| 16:36.13 | [TK]D-Fender | xermesx: No, go read up on how to call functions and variables properly again. |
| 16:36.16 | Qwell[] | I'm not sure though.. show function CUT |
| 16:36.26 | matiasds19 | im getting this message when pressing any dtmf in inbound calls from my sip provider rtp.c:331 process_rfc3389: Comfort noise support incomplete in Asterisk (RFC 3389). Please turn off on client if possible. Client IP: 200.69.206.7 |
| 16:36.39 | Corydon-w | xermesx: the syntax is right, but you're missing the {}'s |
| 16:36.58 | [TK]D-Fender | Corydon-w: Hence it is wrong :) |
| 16:36.59 | elriah | Hi all. Anyone know how to get Cisco phones working behind NAT with an Asterisk server on a public IP (without doing port forwarding, we have a number of phones so that won't work)? |
| 16:37.03 | Corydon-w | Set(intype=${CUT(intype|-|1)}) |
| 16:37.54 | [TK]D-Fender | Corydon-w: "Light a fire for a man and he's warm for a night. Light a man on fire and he's warm for the rest of his life." |
| 16:38.17 | Qwell[] | teach a man to light himself on fire, and he's fed/sheltered for the rest of his life |
| 16:38.43 | angryuser | [TK]D-Fender: yes that's what Bender said in futurama;) |
| 16:38.53 | phearless | <[TK]D-Fender> :/ <-- and how can I fix this ?> |
| 16:38.56 | [TK]D-Fender | angryuser: I never knew the origin. |
| 16:39.10 | phearless | I do not see the problem.. |
| 16:39.17 | phearless | (I googled it etc) |
| 16:39.21 | [TK]D-Fender | phearless: try redoing ztcfg, and then start up * again |
| 16:39.32 | *** join/#asterisk apardo (n=apardo@87.217.144.62) |
| 16:39.36 | *** join/#asterisk PupenoR (n=pupeno@200.123.183.91) |
| 16:41.20 | phearless | Feb 14 16:40:49 ERROR[5287] chan_zap.c: Unable to open D-channel 24 (No such device or address) |
| 16:41.20 | phearless | Feb 14 16:40:49 ERROR[5287] chan_zap.c: Unable to start D-channel on span 1 |
| 16:42.04 | sweeper | yarg |
| 16:42.04 | phearless | waw.... :-S |
| 16:42.04 | sweeper | having issues with call parking |
| 16:42.04 | sweeper | * doesnn't seem to be detecting # :/ |
| 16:42.04 | [TK]D-Fender | phearless: 24? it was 16 a sece ago.... |
| 16:42.04 | phearless | i know, it is strange |
| 16:42.19 | phearless | i still got dchan=16 in zaptel.conf |
| 16:42.19 | [TK]D-Fender | phearless: Stop tunning that stupid script :) |
| 16:42.24 | [TK]D-Fender | ZOMG |
| 16:42.29 | [TK]D-Fender | *ugh* |
| 16:42.33 | sevard | ZOMFLGY |
| 16:42.36 | phearless | bchan=1-8 |
| 16:42.36 | phearless | dchan=16 |
| 16:42.39 | endre | omfglol |
| 16:42.45 | sevard | LOLWTFBBQ |
| 16:42.52 | endre | aybabtu |
| 16:43.02 | [TK]D-Fender | stfukplzbibi |
| 16:43.08 | phearless | which script? |
| 16:43.13 | sweeper | bbq :3 |
| 16:43.16 | [TK]D-Fender | phearless: Genzaptel |
| 16:43.31 | sweeper | anyways |
| 16:43.49 | *** join/#asterisk ghecken (n=info@pd95b1af6.dip0.t-ipconnect.de) |
| 16:43.52 | sweeper | both softphone and 186 ata don't seem to be playing nice with # |
| 16:44.16 | *** part/#asterisk ghecken (n=info@pd95b1af6.dip0.t-ipconnect.de) |
| 16:46.46 | aydiosmio | I'm not seeing an econonmical way to do our transcoding |
| 16:47.00 | phearless | [TK]D-Fender, I started /usr/src/zaptel-1.2.13/xpp/utils/genzaptelconf -c uk |
| 16:47.18 | *** join/#asterisk J4k3- (i=jsuter@dhcp-12-197-128-58.intrastar.net) |
| 16:47.28 | phearless | but... it creates : |
| 16:47.30 | phearless | span=1,1,1,esf,b8zs |
| 16:47.30 | phearless | bchan=1-23 |
| 16:47.30 | phearless | dchan=24 |
| 16:47.44 | *** join/#asterisk doolph (n=doolph@200.46.148.61) |
| 16:47.49 | [TK]D-Fender | phearless: Which is clearly not E1 PRI standard... |
| 16:47.50 | phearless | it is wrong, I should use hdb3,crc4 |
| 16:47.57 | doolph | hi |
| 16:48.02 | doolph | anyone know how to use mastershaper? |
| 16:48.05 | [TK]D-Fender | phearless: So like I said stop using that silly script! |
| 16:48.13 | phearless | BT (british telecom) said it is hdb3,crc4 E1 PRI |
| 16:48.35 | phearless | ok [TK]D-Fender |
| 16:49.12 | [TK]D-Fender | "Doctor, doctor, it hurts when I raise my arm like this!" |
| 16:49.25 | phearless | so what is wrong in : span=1,1,0,ccs,hdb3,crc4 bchan=1-8 dchan=16 ? |
| 16:49.46 | *** join/#asterisk |dennis| (n=dennis@200.32.233.82) |
| 16:50.05 | joe | why would a context in voicemail.conf not show up in show voicemail users? |
| 16:50.10 | Creathir | anyone, could I get some help if possible? |
| 16:50.21 | *** join/#asterisk L|NUX (n=linux@unaffiliated/lnux/x-10290) |
| 16:50.22 | joe | and yes I restarted * completely |
| 16:50.37 | [TK]D-Fender | Creathir: Try asking a specific question |
| 16:51.09 | [TK]D-Fender | joe: try and make sure you have a [default] even if its empty |
| 16:51.09 | Creathir | okay, I am trying to find out what would be wrong with my digium/* setup |
| 16:51.27 | [TK]D-Fender | Creathir: .... |
| 16:51.27 | joe | [TK]D-Fender: yup, it's there |
| 16:51.29 | Creathir | I have a digium pots card that cannot make outbound calls |
| 16:51.30 | Creathir | it rings out |
| 16:51.53 | Creathir | but when the recieving party answers |
| 16:51.59 | Creathir | it continues ringing |
| 16:52.14 | Creathir | and their is dead air on the recieving party's end |
| 16:52.17 | [TK]D-Fender | Creathir: Pastebin the CLI output of your call from beginning to end |
| 16:52.19 | [TK]D-Fender | ~pb |
| 16:52.28 | jbot | [pb] a place to paste your stuff without flooding the channel - try http://pastebin.ca, or http://channels.debian.net/paste |
| 16:52.29 | Creathir | k |
| 16:52.30 | aydiosmio | http://www.signalogic.com/index.pl?page=asterisk_ip_pbx |
| 16:52.34 | aydiosmio | can't wait to get pricing on this |
| 16:52.37 | joe | [TK]D-Fender: even copied over the example and config and the default or other context doesn't show up? wtf? |
| 16:52.45 | *** join/#asterisk Cresl1n (i=matt@nat/digium/x-736218cfdc6e9aa3) |
| 16:52.45 | *** mode/#asterisk [+o Cresl1n] by ChanServ |
| 16:53.10 | [TK]D-Fender | joe: Pastebin the whole thing |
| 16:53.16 | phearless | [TK]D-Fender, do you see the wrong thing in my zaptel.conf ? I understand that I must not use genzaptel |
| 16:53.53 | [TK]D-Fender | phearless: Well the fact it switched to a North American config and killed your framing, etc should be a tip-off. Go fix it. |
| 16:54.18 | *** join/#asterisk autobus (n=autobus@80.172.149.126) |
| 16:54.23 | phearless | my zaptel is back like before |
| 16:54.42 | phearless | [TK]D-Fender: like in http://paste.lisp.org/display/36862 |
| 16:55.12 | xermesx | Qwell |
| 16:55.17 | *** join/#asterisk Xen^ (n=linux@unaffiliated/lnux/x-10290) |
| 16:55.18 | Qwell[] | ? |
| 16:55.27 | xermesx | you re also in #asterisk-gui |
| 16:55.33 | xermesx | yuo can help me perhaps |
| 16:55.35 | autobus | |
| 16:55.36 | autobus | that dailplan I must use to leave called with the prefix: 5029# will be example 5029#351933366009 |
| 16:55.44 | xermesx | every time i login to asterisk-gui, asterisk 1.4 goes in segmentation fault |
| 16:55.47 | *** join/#asterisk bkruse (i=bkruse@nat/digium/x-a3217ede26e1f4c0) |
| 16:56.29 | [TK]D-Fender | phearless: Yes, that look better |
| 16:56.33 | *** join/#asterisk anthm (n=anthm@adsl-68-74-96-61.dsl.milwwi.ameritech.net) |
| 16:56.33 | *** mode/#asterisk [+o anthm] by ChanServ |
| 16:56.47 | autobus | ? |
| 16:57.11 | Creathir | http://channels.debian.net/paste/5385 |
| 16:57.12 | autobus | exten => 5029#|X. is correct? |
| 16:57.22 | Creathir | TK: http://channels.debian.net/paste/5385 |
| 16:57.36 | [TK]D-Fender | autobus: <AlexTrebek> I'm sorry, you did not phrase that in the form of a question. |
| 16:57.40 | Creathir | I grabbed all that was in the buffer |
| 16:57.48 | Creathir | I have my -v set to 9 |
| 16:57.54 | Creathir | I can lower it if you would like me to |
| 16:58.05 | phearless | [TK]D-Fender, I correcte the "channel" instead of "channels" and now I've got : http://paste.lisp.org/display/36862#4 |
| 16:58.16 | dlynes_laptop | For anyone that's interested |
| 16:58.30 | dlynes_laptop | The yet unreleased wanpipe 2.3.4-8 will have a patch for zaptel 1.2.13 |
| 16:59.03 | [TK]D-Fender | phearless: It evidently still thinks you set d-chan for 24... double check everything, rerun ztcfg -vvvv and restart *> |
| 16:59.35 | autobus | that dailplan I must use to leave called with the prefix: 5029# will be example 5029#351933366009 ?? |
| 16:59.37 | autobus | exten => 5029#|X. is correct? |
| 17:00.19 | [TK]D-Fender | autobus: No. exten => _5029#.,1,DoWhatever |
| 17:00.33 | autobus | ok thanks ! |
| 17:02.04 | Creathir | TK: okay, starting at line 103 |
| 17:02.06 | Creathir | that is the call |
| 17:02.18 | Creathir | 106* |
| 17:02.32 | *** join/#asterisk CunningPike (n=CunningP@dhcp-10-234.district.north-van.bc.ca) |
| 17:02.40 | dlynes_laptop | Good morning, CP |
| 17:02.53 | CunningPike | Morning, dlynes_laptop |
| 17:02.59 | CunningPike | How's it going? |
| 17:03.25 | [TK]D-Fender | Creathir: Get rid of the "r" option. |
| 17:03.26 | autobus | [TK]D-Fender, my idea is only use 5029# for prefix not complete number. |
| 17:03.34 | autobus | its only for access |
| 17:03.43 | autobus | [TK]D-Fender you understand? |
| 17:03.59 | [TK]D-Fender | autobus: Yes, and what I showed you will allow you to dial exactly that. |
| 17:04.03 | autobus | in this case its possible use _5029#|. |
| 17:04.04 | autobus | ? |
| 17:04.12 | xermesx | gdb /usr/sbin/asterisk /path/to/core.12345 |
| 17:04.17 | [TK]D-Fender | autobus: NO. I told you how you have to enter it. |
| 17:04.24 | Creathir | TK: the r option? on my asterisk CLI login? |
| 17:04.29 | xermesx | what shoul i write instead of /path/to/core.12345 ? |
| 17:04.41 | [TK]D-Fender | Creathir: In your dial statement. |
| 17:04.59 | Creathir | which file is that in? |
| 17:05.34 | [TK]D-Fender | Creathir: Where dfo you THINK "exten" goes? |
| 17:05.48 | Creathir | lol exten.conf;) |
| 17:05.53 | [TK]D-Fender | Creathir: This is the point where I remind you that : |
| 17:05.56 | [TK]D-Fender | ~freepbx |
| 17:06.07 | jbot | freepbx is, like, unable to be supported here. It's a complex piece of dialplan and GUI, and can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking there |
| 17:06.24 | *** join/#asterisk mitcheloc (n=mitchelo@titaniumsoft.net) |
| 17:06.25 | [TK]D-Fender | Creathir: Try and tell your interface to not send "forced rining" on your channels |
| 17:07.01 | Creathir | k |
| 17:07.24 | Creathir | I did not realize this was a FreePBX issue |
| 17:07.45 | Creathir | I would have thought it was the zapata.conf file or something |
| 17:08.01 | [TK]D-Fender | Creathir: And what do you think BUILT all your configs? |
| 17:08.05 | Creathir | so the extensions.conf has a forced ringing option set |
| 17:08.30 | [TK]D-Fender | Creathir: There is a reason why FreePBX isn't supported here. This is one of them. |
| 17:08.33 | *** join/#asterisk Skarmeth (n=Skarmeth@201009076197.user.veloxzone.com.br) |
| 17:08.45 | Creathir | gotcha |
| 17:08.49 | Creathir | thanks for your help anyways |
| 17:10.22 | *** join/#asterisk genz (n=erdo@im.jobdig.com) |
| 17:11.55 | *** join/#asterisk Crescendo_ (n=martinda@adsl-072-151-080-148.sip.rmo.bellsouth.net) |
| 17:12.33 | genz | Could someone take a look at something in the source before I submit a bug report? |
| 17:13.36 | drako | why a TDM card is hanging the system |
| 17:13.45 | drako | it wont response even to the keyboard |
| 17:14.11 | genz | I don't think cdr_addon_mysql.c is reading in the loguniqueid flag |
| 17:15.09 | Skarmeth | good afternoon all |
| 17:15.13 | wiizard2 | what goes around |
| 17:15.14 | wiizard2 | goes around |
| 17:15.34 | wiizard2 | dont wanna think about dont wanna talk about it just so sick about it |
| 17:16.02 | *** join/#asterisk gork (n=gork@gw.amarillo.energynet.com) |
| 17:17.01 | Skarmeth | I have a Asterisk system that I want to create a special exten that will receive a information from user (Read application) and I wanto to process this input by a external program... which kind of AGI do best fits my needs? |
| 17:18.03 | joe | wow, logs really need to say oh I found voicemail.conf but can't read it, not just found it! grumble |
| 17:18.06 | gork | Hello all -- I asked yesterday but nobody at the time knew -- Is there a tool I can use to convert an audio file for use as an asterisk-compatible .g722 prompt? Sox (as far as I know) can't do it and the only thing I can find via google are companies trying to sell DSP vocoder code... |
| 17:18.16 | *** join/#asterisk Qwell[] (i=qwell@pdpc/sponsor/digium/Qwell) |
| 17:18.16 | *** mode/#asterisk [+o Qwell[]] by ChanServ |
| 17:18.35 | wiizard2 | g722 |
| 17:18.38 | wiizard2 | wtf is that |
| 17:18.51 | wiizard2 | is that some hybrid codec |
| 17:18.58 | endre | g723 or g711 |
| 17:19.02 | endre | typomix |
| 17:19.05 | Skarmeth | I am reading voip-info now and I found AGI, EAGI, FastAGI and DeadAGI kinds of AGI |
| 17:19.26 | gork | G.722 is a 16kHz sample rate (wideband) codec |
| 17:19.31 | endre | agi is agi |
| 17:19.31 | wiizard2 | urgh |
| 17:19.32 | drako | do I need the module wcfxo ? |
| 17:19.36 | wiizard2 | 16khz |
| 17:19.44 | wiizard2 | You hosting this system on a dialup connection? |
| 17:19.45 | Skarmeth | wiizard2, did your looked up at digium web site? there is a transcoding tool online |
| 17:20.00 | wiizard2 | Skarmeth --> gork |
| 17:20.13 | Skarmeth | wiizard2, sorry |
| 17:20.18 | gork | Skarmeth: they dont support g722 with that tool (yet?) |
| 17:20.33 | *** join/#asterisk argus (n=ryan@c-24-147-44-55.hsd1.nh.comcast.net) |
| 17:20.47 | Cresl1n | gork: You could use app_record on a G722 compatible phone |
| 17:21.09 | wiizard2 | ive never seen anything support that codec |
| 17:21.19 | Cresl1n | A lot of newer phones do |
| 17:21.33 | Cresl1n | newer polycoms, I think SNOMs as well, and a couple of others |
| 17:21.35 | gork | yeah thats what im doing now but rerecording a lot of prompts isnt really fun when i already have CD quality wav's sitting around |
| 17:21.43 | gork | I am using polycom 650's yeah; they sound really great |
| 17:21.44 | Cresl1n | I gotcha |
| 17:21.48 | wiizard2 | ah |
| 17:21.51 | wiizard2 | polycom and snow |
| 17:21.53 | Cresl1n | well, I ended up writing a tool |
| 17:21.53 | wiizard2 | *snom |
| 17:21.58 | Cresl1n | for doing conversions |
| 17:21.59 | wiizard2 | both of which are sitting in my draw |
| 17:22.02 | wiizard2 | *drawer |
| 17:22.16 | *** join/#asterisk apardo (n=apardo@87.217.145.246) |
| 17:22.21 | wiizard2 | is it a "Real" codec |
| 17:22.25 | Cresl1n | yes |
| 17:22.27 | Cresl1n | very much so |
| 17:22.32 | *** join/#asterisk h0 (i=fakhir@unaffiliated/fakhir) |
| 17:22.36 | wiizard2 | odd. |
| 17:22.45 | Cresl1n | it was supposed to replace g.722 at some time or another in the PSTN world |
| 17:22.45 | gork | lol everytime i have talked to anyone about g722 there is always this disbelief about it |
| 17:22.54 | Cresl1n | since it fits into a 64kbps channel |
| 17:22.58 | wiizard2 | ah i see |
| 17:23.11 | Cresl1n | but it's seeing a lot more use in IP telephony |
| 17:23.22 | wiizard2 | surely the quality is low |
| 17:23.29 | gork | the quality is superb |
| 17:23.29 | Cresl1n | sorry "supposed to replace g.711 in the pstn world" |
| 17:23.34 | Cresl1n | that was what I meant |
| 17:23.41 | wiizard2 | hehe thought as much |
| 17:23.51 | Cresl1n | yeah, it's much better than a lot of the other codecs |
| 17:23.55 | angryuser | bye everybody;) |
| 17:24.05 | wiizard2 | whats the bandwidth with it? |
| 17:24.08 | argus | anyone have luck getting a 7940/60 with fw>=SCCP 8.4 to switch to SIP, it seems to be ignoring the usual tags in SEP[mac].cnf.xml |
| 17:24.09 | Cresl1n | And it's a wideband codec, so it sounds amazing |
| 17:24.16 | angryuser | *gone |
| 17:24.16 | Cresl1n | 64kbps, at least in asterisk |
| 17:24.24 | wiizard2 | ouch |
| 17:24.35 | wiizard2 | well actually thats not too bad |
| 17:24.43 | Cresl1n | it's the same as g.711 |
| 17:24.51 | wiizard2 | yes exactly what i just realised |
| 17:25.51 | wiizard2 | i really need to get an asterisk box up |
| 17:26.25 | *** join/#asterisk yassine (n=yassine@dsl.voicint.com) |
| 17:27.00 | Creathir | TK: you were a lot of help |
| 17:27.10 | Creathir | I dug around in the config files |
| 17:27.17 | Creathir | and you were right about the r option |
| 17:27.21 | Creathir | what does that do exactly? |
| 17:27.58 | [TK]D-Fender | Creathir: Forces ringing and causes all sort of nasty problems. If you think you need it, think again. Its only useful on bad SIP connections with improper signalling. |
| 17:28.20 | Creathir | why on earth does FreePBX enable this? |
| 17:28.27 | *** join/#asterisk lowlevel (n=Stuart@74.108.229.254) |
| 17:28.29 | *** join/#asterisk kanelbullar (n=kanelbul@83.240.200.92) |
| 17:28.32 | Creathir | or, is that the host of problems associated with FPBX |
| 17:28.34 | Creathir | ? |
| 17:28.35 | [TK]D-Fender | wiizard2: Funny.... why are you here if you're not running *? |
| 17:28.58 | [TK]D-Fender | Creathir: I am relatively sure this is an option YOU set sopmewhere. |
| 17:29.11 | Creathir | that I set? really? |
| 17:29.11 | [TK]D-Fender | Creathir: And who know why they do ANYTHING the way they do. |
| 17:29.17 | Creathir | lol k |
| 17:29.47 | *** join/#asterisk anonymouz666 (n=anonymou@re366.compuland.com.br) |
| 17:30.24 | ChicagoBud | question on the directory(0 application. how do you match a name with only 2 characters? |
| 17:31.06 | *** join/#asterisk Cyon (n=cyon@216.179.31.170) |
| 17:31.22 | *** join/#asterisk reber (n=reber@cl-158.dub-01.ie.sixxs.net) |
| 17:32.19 | kanelbullar | Hi guys, does anyone have any experience with blocking collect calls in Brazil, using digital lines? |
| 17:33.00 | sweeper | ok, so my signalling is out of band. why does * persist in not allowing call parking? |
| 17:33.03 | *** join/#asterisk IronHelixz (n=irc@ool-45785cfe.dyn.optonline.net) |
| 17:33.11 | sweeper | I hit #, to no avail! |
| 17:33.43 | *** join/#asterisk penguinFunk (n=penguin@87.224.86.46) |
| 17:34.43 | phearless | what is the difference between span=1,1,0,ccs,hdb3,crc4 and span=1,1,1,ccs,hdb3,crc4 |
| 17:34.55 | *** join/#asterisk DaPrivateer (i=Privatee@crimson.66fruit.com) |
| 17:37.10 | Creathir | okay TK, one last question |
| 17:37.17 | Creathir | and I know I'm using FPBX |
| 17:37.20 | Creathir | and you can't support me |
| 17:37.26 | Creathir | but you really did point me in the right direction |
| 17:37.43 | [TK]D-Fender | Creathir: Then get it over with and just ask :) |
| 17:38.05 | Creathir | on inbound calls, when a station answers, I get this strange pulse like sound on the line, and it continues to ring |
| 17:39.08 | Creathir | want to see the CLI? |
| 17:39.34 | phearless | what is the difference between span=1,1,0,ccs,hdb3,crc4 and span=1,1,1,ccs,hdb3,crc4 ? I am sure that [TK]D-Fender knows ;) |
| 17:40.15 | [TK]D-Fender | phearless: The extra "1" is for LBO (Line Build-out), which is for amping up really long cable runs from your demarc. You "shouldn't" need this |
| 17:41.19 | phearless | ok thank you [TK]D-Fender |
| 17:41.32 | Creathir | tk: http://channels.debian.net/paste/5386 |
| 17:42.12 | phearless | Feb 14 17:41:54 ERROR[4455]: chan_zap.c:9241 start_pri: Unable to open D-channel 24 (No such device or address) |
| 17:42.12 | phearless | Feb 14 17:41:54 ERROR[4455]: chan_zap.c:11056 setup_zap: Unable to start D-channel on span 1 |
| 17:42.27 | phearless | I still got this , even the fact that I got no 24 or 23 anywhere! |
| 17:43.29 | Corydon-w | phearless: it thinks it's a T1 and not an E1. Have you set the jumpers correctly? |
| 17:43.43 | *** join/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 17:45.08 | [TK]D-Fender | phearless: Abort * altogether, and redo "ztcfg -vvvv", if that looks ok, then verify your files, then restart * |
| 17:45.08 | [TK]D-Fender | Creathir: No idea... |
| 17:45.08 | [TK]D-Fender | phearless: Yes... that WOULD be a potential problem.... |
| 17:45.09 | [TK]D-Fender | phearless: There is a jumper in the middle fo the card IIRC to set the mode... |
| 17:45.58 | phearless | I did check the jumpers |
| 17:46.03 | phearless | I will re-check :) |
| 17:46.08 | *** join/#asterisk Dr-Linux (n=Nothing@202.125.139.198) |
| 17:46.10 | Dr-Linux | g723? |
| 17:46.24 | Dr-Linux | is there any softphone that supports g723? i mean any sip client? |
| 17:46.43 | Creathir | TK: k, thanks for your help anyways |
| 17:46.43 | L|NUX | eyeBeam |
| 17:46.45 | L|NUX | ekiga |
| 17:46.55 | gork | yeah xten eyebeam supports it |
| 17:47.02 | L|NUX | but not for sale :) |
| 17:47.23 | L|NUX | only available for bluck purchase |
| 17:47.24 | L|NUX | ;) |
| 17:47.28 | Dr-Linux | L|NUX: i've eyebeam, but it doesn't support |
| 17:47.44 | L|NUX | well it does not have g723 codec |
| 17:47.45 | L|NUX | :) |
| 17:47.48 | *** join/#asterisk rene- (n=rene-@200.34.66.137) |
| 17:48.05 | L|NUX | you can use TerraCall X-PRO its also have G.723 ;) |
| 17:48.07 | Dr-Linux | then how does it support it? |
| 17:48.10 | L|NUX | or use Ekiga |
| 17:48.19 | Dr-Linux | they are free? |
| 17:48.20 | phearless | FUCKING HELL |
| 17:48.24 | L|NUX | nah |
| 17:48.25 | phearless | THE JUMPER WAS WRONG |
| 17:48.26 | L|NUX | they are not free |
| 17:48.35 | L|NUX | ekiga is free though |
| 17:48.48 | phearless | Corydon-w let me kiss you via IRC |
| 17:48.54 | phearless | *smack* |
| 17:48.56 | Dr-Linux | lol |
| 17:48.59 | rene- | does anybody knows a POE switch whose ports can be 'resetted' or some how connect/disconnect from a web page / api ? |
| 17:49.08 | phearless | and thanks to [TK]D-Fender too ;) |
| 17:49.15 | phearless | now the LED is green |
| 17:49.34 | rene- | i dont want to go upstairs to unplug phones |
| 17:49.36 | gork | rene-: DLINK DES1526 |
| 17:49.44 | rene- | thanks |
| 17:49.47 | Skaag | Why do I get this: 070214-194927 WARNING[30961]: manager.c:1793 init_manager: Unable to bind socket: Address already in use |
| 17:49.49 | Dr-Linux | L|NUX: which one is best you mentioned two above? |
| 17:49.56 | Skaag | it happens when asterisk is parsing manager.conf |
| 17:50.17 | CJLinst | is there any way to make a SIP channel return BUSY instead of CHANUNAVAIL when sip.conf:call-limit is exceeded? A user calling such an extension should get a busy signal instead of a fast busy, IMHO. |
| 17:50.17 | L|NUX | Dr-Linux : eyeBeam but you can not afford to buy it :) |
| 17:50.37 | Dr-Linux | L|NUX: i have it already, as i told you |
| 17:50.45 | L|NUX | well brother |
| 17:50.52 | L|NUX | you do not have the one which support g723 |
| 17:50.53 | L|NUX | i told you |
| 17:50.56 | L|NUX | understand |
| 17:50.57 | Dr-Linux | L|NUX: but i can't see g723 codec support there |
| 17:51.04 | Dr-Linux | i see |
| 17:51.07 | L|NUX | see your Audio Codecs |
| 17:51.13 | L|NUX | in eyeBeam settings |
| 17:51.13 | rene- | gork: cool and can be had for under 420 |
| 17:51.16 | L|NUX | then you will understand |
| 17:51.17 | *** join/#asterisk gniretar (n=icechat5@fhd.wccnet.org) |
| 17:51.22 | gniretar | hiya |
| 17:51.34 | Dr-Linux | yeah i'm there |
| 17:51.38 | gniretar | what CLI commands should i use to debug a misbehaving IAX channel? |
| 17:51.52 | gork | rene-: yes its inexpensive and fairly capable. It doesnt support CDP or cisco power though so keep that in mind; what phones are you using? |
| 17:51.56 | [TK]D-Fender | D-Line DES-1526 = nice simple PoE switch..... great bang/buck. |
| 17:52.09 | rene- | mitel but soon i should get polycoms |
| 17:52.23 | *** join/#asterisk jimbo- (n=dsgsdfga@modemcable066.82-200-24.mc.videotron.ca) |
| 17:52.47 | jimbo- | how do I retrieve the source IP of an inbound call from AGI? |
| 17:53.20 | gork | rene-: I use it with polycoms; works great. Also its worth noting that the PoE adaptor polycom ships for the 301 phones will power up cisco stuff that is not 802.11af compliant.. |
| 17:53.31 | jimbo- | this works fine with SIP -> SIPCHANINFO(recvip) |
| 17:53.41 | jimbo- | but it doesn't work with H.323 inbound calls |
| 17:54.00 | *** join/#asterisk jmls (n=asterisk@host81-159-198-100.range81-159.btcentralplus.com) |
| 17:54.03 | jmls | hey |
| 17:54.14 | jmls | what does |
| 17:54.15 | jmls | WARNING[4298]: app_dial.c:734 wait_for_answer: Unable to forward voice |
| 17:54.26 | elriah | Anybody here use SCCP with Cisco phones? How does it work with NAT? |
| 17:54.28 | jmls | mean in real terms ? Is it a bad thing ? What could cause it ? |
| 17:54.35 | L|NUX | Dr-Linux : you from ? |
| 17:54.41 | Skaag | ok fixed that |
| 17:54.46 | *** join/#asterisk friedrich| (n=friedric@e177247098.adsl.alicedsl.de) |
| 17:54.50 | Dr-Linux | L|NUX: pakistan |
| 17:55.00 | jimbo- | anyone? |
| 17:55.08 | [TK]D-Fender | gork: For the price you pay for the cable, I expect it to make me coffee too ;) |
| 17:55.13 | jimbo- | how do I retrieve the source IP of an inbound call? |
| 17:55.47 | bcnl | could anyone explain to me why $TIMESTAMP doesn't work in 1.4? |
| 17:55.48 | gork | TK: haha yeah they are really proud of it; still cheaper than buying a cisco switch if you just need to run a couple of AP's on it |
| 17:56.14 | [TK]D-Fender | gork: Then again, if you wanted PoE, you should get the IP 430 over the IP 301 anyways... |
| 17:56.20 | bcnl | I wound up having to do this "Set(TIMESTAMP=${STRFTIME(||%Y%m%d-%H:%M:%S)})" to get the same functionality as 1.2.x |
| 17:56.39 | bcnl | well, without the :'s (but I think they look nicer) |
| 17:56.57 | rene- | the 301 requires an additional cable for POE |
| 17:56.59 | rene- | that isnt cool |
| 17:57.45 | *** join/#asterisk topping (n=topping@204.152.96.238) |
| 17:57.53 | jimbo- | how do I retrieve the IP address of an inbound call? |
| 17:58.03 | *** join/#asterisk eltech (i=G00Ds@ool-457c93b6.dyn.optonline.net) |
| 17:58.04 | rene- | one question tho, that is the country of origin of Polycom? my distributor still doesnt have the 430 but i dont want to end paying up 300% in taxes like i did with uniden phones just because i didnt make sure the phones i was buying were tax free in mexico |
| 17:58.14 | [TK]D-Fender | rene-: For 15$ more you get a phone with it integrated, Speakerphone, lighted indicators, a pixel based display, and MicroBrowser. Fair deal if you ask me... |
| 17:58.38 | bcnl | I can't believe that $TIMESTAMP is not being used by anyone running 1.4 |
| 17:58.48 | rene- | [TK]D-Fender: yes it sounds like a better deal |
| 17:59.56 | rene- | D-Fender: do you know the country of origin of the Poly 430? |
| 18:00.03 | gork | yeah the 301s are complete garbage compared to 430's |
| 18:01.19 | [TK]D-Fender | rene-: Not offhand. Malaysia I think |
| 18:01.38 | [TK]D-Fender | gork: Not so much garbage as "totally wrong idea for PoE environments" |
| 18:01.53 | [TK]D-Fender | gork: Nifty saving is you only need basic use in non-PoE |
| 18:03.40 | blitzrage | bcnl: $TIMESTAMP is wrong -- use ${TIMESTAMP} |
| 18:04.21 | gniretar | gah! I turned off zap timing and i'm still getting one way audio on this IAX channel |
| 18:04.28 | *** join/#asterisk HockeyInJune (n=HockeyIn@pool-151-204-139-13.ny325.east.verizon.net) |
| 18:04.35 | gniretar | what sorts of things could i do wrong that would result in me getting one way audio? |
| 18:05.07 | gniretar | on IAX of all things |
| 18:06.31 | aydiosmio | NAT/addressing |
| 18:06.39 | aydiosmio | routing |
| 18:06.39 | gniretar | no NAT in use |
| 18:06.44 | gniretar | no routing |
| 18:06.54 | gniretar | 2 boxes together on a switch |
| 18:07.23 | aydiosmio | have you checked the packet stream with a tcpdump or wirshark capture? |
| 18:07.27 | aydiosmio | wireshark |
| 18:07.38 | gniretar | NO, i shall do that next i think |
| 18:07.56 | aydiosmio | good way to separate network problems from audio problems |
| 18:07.57 | gniretar | however, i have tried many hosts and i have universally the same results |
| 18:11.55 | jimbo- | how do I retrieve the IP address of an inbound call? |
| 18:12.09 | bcnl | <PROTECTED> |
| 18:12.10 | bcnl | 10:14 < blitzrage> bcnl: $TIMESTAMP is wrong -- use ${TIMESTAMP} |
| 18:12.19 | bcnl | yea I just was lazy typing in the channel and made a error |
| 18:12.26 | jimbo- | I can do it under SIP using SIPCHANINFO(recvip) but how can I get the IP under H323 and IAX? |
| 18:17.25 | *** join/#asterisk J4k3 (i=jsuter@178.sub-70-216-172.myvzw.com) |
| 18:21.02 | *** part/#asterisk Janos (n=Janos@nat1.inalambrica.net) |
| 18:21.48 | gniretar | OK, i ahve this working |
| 18:21.50 | gniretar | have** |
| 18:21.53 | gniretar | here is what was wrong: |
| 18:22.25 | *** join/#asterisk brussel (n=brussel@cpe-24-165-1-241.san.res.rr.com) |
| 18:22.30 | gniretar | both sides had the identical IAX config except for the host= part |
| 18:22.40 | gniretar | it didnt like that and it ended up getting very confused |
| 18:22.56 | gniretar | as soon as i changed them to be more like this: http://astrecipes.net/index.php?n=204 |
| 18:23.00 | gniretar | it started working |
| 18:27.18 | *** join/#asterisk Aurs (n=Aurs@81.191.112.104) |
| 18:29.16 | blitzrage | bcnl: ok cool, I'll be lazy too then :) |
| 18:30.27 | *** join/#asterisk ping2921 (n=marc3234@206-248-153-114.dsl.teksavvy.com) |
| 18:30.52 | ping2921 | hi, I'm looking for a softphone that would allow 5+ hold lines. |
| 18:30.54 | *** join/#asterisk [[blah]asfd (n=ckwall@63.149.122.93) |
| 18:31.13 | [[blah]asfd | has anyone seen this error before? Feb 14 11:28:59 WARNING[23176]: chan_sip.c:2552 sip_write: Asked to transmit frame type 64, while native formats is 2 (read/write = 2/2) |
| 18:34.23 | jimbo- | can anyone answer the question? |
| 18:34.27 | jimbo- | how do I retrieve the IP address of an inbound call? |
| 18:34.33 | jimbo- | on 1.4 |
| 18:34.35 | jimbo- | or 1.2 |
| 18:34.40 | *** join/#asterisk zeedo (n=zeedo@reboot-robot.net) |
| 18:36.32 | *** join/#asterisk Shmattie (n=Shmattie@cpe-75-179-182-202.woh.res.rr.com) |
| 18:37.30 | *** join/#asterisk J4k3 (n=jsuter@mhset-growth7.memorialhealth.org) |
| 18:37.59 | CunningPike | [[blah]asfd: Looks like a transcoding error to me - what codecs are you using? |
| 18:38.32 | *** join/#asterisk RoyK (n=roy@ti211310a080-5551.bb.online.no) |
| 18:38.35 | [[blah]asfd | ulaw |
| 18:38.37 | *** join/#asterisk ghecken (n=info@pd95b1af6.dip0.t-ipconnect.de) |
| 18:38.45 | [[blah]asfd | disallow=all |
| 18:38.45 | [[blah]asfd | allow=ulaw |
| 18:46.21 | CunningPike | [[blah]asfd: And at the other end of the call? |
| 18:47.47 | CunningPike | [[blah]asfd: Isn't frame type 64 G.729? |
| 18:49.56 | *** join/#asterisk tsurko (n=tsurko@vpn-pppoe-213-240-242-237.megalan.bg) |
| 18:50.53 | jimbo- | hello |
| 18:50.59 | jimbo- | can anyone help? |
| 18:52.41 | *** join/#asterisk JoNate (n=noone@mail.wmelec.com) |
| 18:53.14 | CunningPike | ~ask |
| 18:53.21 | jbot | rumour has it, ask is Questions in the channel should be specific, informative, complete, concise, and on-topic. Don't ask if you can ask a question first. Don't ask if a person is there, just ask what you intended to ask them. Better questions more frequently yield better answers. We are all here voluntarily. See also ... |
| 18:53.21 | tzafrir | jimbo-, I expect it to be part of the channel name |
| 18:53.29 | tzafrir | (he did ask) |
| 18:54.08 | tzafrir | Though there is probably a more direct way. Isn't there a PEER function or something similar? |
| 18:54.51 | jimbo- | jbot_: rumor has it that I've been asking that question for about 45 minutes now and nobody has responded other that tzafrir |
| 18:55.03 | jimbo- | other than, i mean |
| 18:55.34 | jimbo- | there is this function: SIPCHANINFO(recvip) but it only works for SIP |
| 18:55.56 | tzafrir | jimbo-, jbot is just a bot. Blame CunningPike for giving the command ;-) |
| 18:56.05 | CunningPike | jimbo-: I apologize - I didn't look back far enough |
| 18:56.34 | jimbo- | ok |
| 18:56.37 | jimbo- | no problem |
| 18:56.41 | tzafrir | BTW: when is jbot going to get its original name back? |
| 18:57.54 | eald | jimbo-: and what other kind of call you have? |
| 18:58.01 | jimbo- | H.323 and IAX |
| 18:58.28 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 18:58.33 | eald | for IAX there is IAX peer as I can see wiki |
| 18:58.41 | jimbo- | link? |
| 18:58.59 | eald | http://www.voip-info.org/wiki/view/Asterisk+functions |
| 18:59.01 | jimbo- | why isn't there a more general function, like PEER |
| 18:59.24 | tzafrir | jimbo-, in the asterisk CLI: show function <tab><tab> |
| 18:59.33 | *** join/#asterisk toxap (n=toxap@213.227.193.74) |
| 18:59.39 | eald | dunno, maybe * is waiting for somene to code it |
| 18:59.41 | jimbo- | got it |
| 18:59.47 | jimbo- | ok |
| 18:59.59 | jimbo- | i can see that being the case |
| 19:00.18 | jimbo- | so there's actually no way of knowing the inbound peer address of an H.323 connection |
| 19:00.30 | eald | I don't know |
| 19:01.22 | eald | maybe if you look in the wiki there is something about it, maybe writing it based from IAXPEER or whatever is trivial |
| 19:01.45 | eald | I just looked for the function that you said in the wiki and found that |
| 19:02.17 | jimbo- | thx |
| 19:04.40 | drako | ok here is a big problem, when i put on zapata.conf echocancel=yes |
| 19:04.56 | drako | the computer hangs up when I try to call through the TDM |
| 19:07.37 | tzafrir | And if you disable the echocanceller, it won't freeze? |
| 19:07.48 | drako | tzafrir, no. |
| 19:07.59 | drako | tzafrir, thats in debian btw. |
| 19:08.05 | *** join/#asterisk Crescendo_ (n=martinda@adsl-072-151-080-148.sip.rmo.bellsouth.net) |
| 19:12.48 | *** join/#asterisk philippel (n=p_lindhe@c-24-17-254-189.hsd1.mn.comcast.net) |
| 19:13.52 | philippel | pardon me if I missed it (I have looked) but, is there a patch to 1.2 to provide multiple parking lots? (to accomodate a multi-tenant implementation)? I've only seen the meter-maid patch for hints? |
| 19:14.33 | philippel | mitcheloc ping are you around (I have a unrelated question for you)? |
| 19:20.52 | *** join/#asterisk sahafeez (n=sahafeez@67.109.14.227.ptr.us.xo.net) |
| 19:25.58 | *** join/#asterisk Gamercjm (n=chris@pool-71-254-183-243.lsanca.fios.verizon.net) |
| 19:26.13 | Gamercjm | Anyone here have lumenvox? |
| 19:26.26 | tg | - |
| 19:28.48 | *** join/#asterisk zeeesh (n=aadilism@202.125.143.65) |
| 19:28.50 | zeeesh | hi |
| 19:29.42 | *** join/#asterisk tsurko (n=tsurko@vpn-pppoe-213-240-242-237.megalan.bg) |
| 19:32.10 | *** join/#asterisk topping (n=topping@h-67-100-91-18.snfccasy.covad.net) |
| 19:33.16 | *** join/#asterisk topping (n=topping@h-67-100-91-18.snfccasy.covad.net) |
| 19:33.26 | zeeesh | <PROTECTED> |
| 19:34.11 | mog | you want to develop your own voice codec? |
| 19:34.24 | aemoth | c with asm |
| 19:34.30 | aemoth | for performance |
| 19:34.36 | mog | or just c |
| 19:34.40 | aemoth | or just asm |
| 19:34.43 | mog | depending on how much performance is needed |
| 19:34.57 | mog | hell there is even an implementation of gsm in java |
| 19:37.57 | elriah | java voice codecs? yuck |
| 19:38.12 | *** join/#asterisk dasenjo (n=dasenjo@190.24.176.242) |
| 19:38.37 | jmls | Hmm. would a module reload make cdr reread cdr.conf, or is it a full reload or a restart ? |
| 19:38.54 | *** join/#asterisk putzz (n=me@CPE00155824933c-CM00111ae07ac2.cpe.net.cable.rogers.com) |
| 19:39.50 | elriah | jmls: I'm pretty sure you have to restart... |
| 19:47.26 | autobus | for autorized 2 ips autentication: its possible use: HOST=64.192.196.12,64.192.196.10 |
| 19:47.33 | autobus | or , its not correct? |
| 19:48.47 | blitzrage | think you need two separate definitions |
| 19:49.14 | autobus | hot=64.192.196.12 |
| 19:49.18 | autobus | host=64.192.196.12 |
| 19:49.25 | autobus | and host=64.192.196.10 |
| 19:49.28 | autobus | right? |
| 19:49.43 | blitzrage | no |
| 19:49.46 | blitzrage | [peer1] |
| 19:49.49 | blitzrage | [peer2] |
| 19:50.19 | autobus | ok i understand ! |
| 19:50.26 | autobus | thanks very much |
| 19:53.01 | *** join/#asterisk Crescendo_ (n=martinda@adsl-072-151-080-148.sip.rmo.bellsouth.net) |
| 19:58.45 | *** join/#asterisk DougSmith (n=dsmith@12.43.48.194) |
| 20:06.19 | *** join/#asterisk redondos (n=redondos@190.48.35.71) |
| 20:21.14 | *** join/#asterisk pythos (i=lanebob@unaffiliated/pythos) |
| 20:21.20 | pythos | Good day! |
| 20:21.56 | pythos | Im wondering where to find example of simple settings for doing two FXS where one can dial the other -- dialplan I think its called? |
| 20:29.47 | *** join/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 20:30.39 | *** join/#asterisk alephant (n=cmd@c-24-3-52-93.hsd1.pa.comcast.net) |
| 20:30.40 | alephant | Hi all! |
| 20:31.00 | *** join/#asterisk anthony] (n=anthony@175.21.188.72.cfl.res.rr.com) |
| 20:31.03 | alephant | A customer has expressed interest in having his VoIP phones act like TDMA phones: |
| 20:31.14 | *** join/#asterisk anthm (n=anthm@adsl-68-74-96-61.dsl.milwwi.ameritech.net) |
| 20:31.14 | *** mode/#asterisk [+o anthm] by ChanServ |
| 20:31.17 | alephant | When trunk1 is in use, line1 on all phones lights up |
| 20:31.26 | alephant | When trunk2 is in use, line2 on all phones lights up |
| 20:31.26 | alephant | etc. |
| 20:31.34 | alephant | I have explained that this is kinda silly |
| 20:31.54 | alephant | and that the Flash Operator Panel or Hud or whatever else can provide the same "what trunks are available" |
| 20:31.56 | alephant | but |
| 20:31.58 | *** join/#asterisk drako (n=ljd@unaffiliated/luisjose) |
| 20:31.58 | *** join/#asterisk foobar778 (n=bryan@ip68-100-41-120.dc.dc.cox.net) |
| 20:31.59 | alephant | academically |
| 20:32.01 | *** join/#asterisk ghecken (n=info@pd95b1af6.dip0.t-ipconnect.de) |
| 20:32.16 | alephant | is there any way to bind an extension to a trunk? |
| 20:32.27 | *** part/#asterisk ghecken (n=info@pd95b1af6.dip0.t-ipconnect.de) |
| 20:32.49 | *** join/#asterisk yassine (n=yassine@dsl.voicint.com) |
| 20:32.55 | Cresl1n | yuck |
| 20:33.03 | Cresl1n | you want shared line appearance |
| 20:33.05 | *** join/#asterisk ghecken (n=info@pd95b1af6.dip0.t-ipconnect.de) |
| 20:33.10 | alephant | is that the term? |
| 20:33.14 | *** part/#asterisk ghecken (n=info@pd95b1af6.dip0.t-ipconnect.de) |
| 20:33.15 | Cresl1n | yep |
| 20:33.16 | alephant | I sympathize with your "yuck" |
| 20:33.17 | Cresl1n | it's in 1.4 |
| 20:33.44 | alephant | BTW, is there a quick link for diff 1.x 2.x? |
| 20:33.46 | *** join/#asterisk ghecken (n=info@pd95b1af6.dip0.t-ipconnect.de) |
| 20:33.53 | foobar778 | no reply to our critical packet anyone see thaty error? |
| 20:33.53 | *** join/#asterisk tzafrir_laptop (n=tzafrir@bzq-88-154-6-99.red.bezeqint.net) |
| 20:34.13 | [TK]D-Fender | alephant: This can be done with simple Presence. What phone models do you have? |
| 20:34.34 | alephant | Grandstream GXP?2000 |
| 20:34.36 | CunningPike | foobar778: Usually caused by bad latency or dropped packets |
| 20:34.57 | [TK]D-Fender | alephant: Go ready on how to set up BLF on them, and setup hints for your various channels |
| 20:35.20 | foobar778 | Can someone test by calling my PBX??? |
| 20:35.25 | alephant | BLF? |
| 20:35.34 | Qwell[] | ~blf |
| 20:35.37 | jbot | i guess blf is Busy Lamp Field, aka little lights next to speed dials that light up when the person is on the phone and blink when that line is ringing. hint extensions are static mapped to SIP or other channels. |
| 20:35.43 | alephant | oooh! |
| 20:36.01 | foobar778 | Cunning Pak thanks how to fix that |
| 20:36.19 | foobar778 | CunningPike sorry |
| 20:36.38 | CunningPike | foobar778: Fix your network :_) |
| 20:36.50 | foobar778 | vague isnt that |
| 20:37.30 | foobar778 | does that mean u dont know? |
| 20:38.14 | foobar778 | CunningPike can u call my PBX to test |
| 20:39.04 | foobar778 | or anyone for that matter |
| 20:39.13 | *** part/#asterisk redondos (n=redondos@190.48.35.71) |
| 20:39.38 | foobar778 | 68.100.41.120 user 6000 secret boy125 than dial 6001 |
| 20:42.51 | *** join/#asterisk rene- (n=rene-@200.34.66.137) |
| 20:45.19 | *** join/#asterisk DukeOfURL (n=chatzill@72.18.136.240) |
| 20:45.58 | *** join/#asterisk Ryushin (i=proxy@windwalker.openinnovations.com) |
| 20:46.04 | *** join/#asterisk s1gny|wrk (n=s1gny@p549174E5.dip.t-dialin.net) |
| 20:46.30 | *** part/#asterisk s1gny|krank (n=s1gny@p549174E5.dip.t-dialin.net) |
| 20:47.35 | foobar778 | Can anyone test by calling my PBX???? |
| 20:51.01 | *** join/#asterisk anthm][ (n=anthm@adsl-68-74-96-61.dsl.milwwi.ameritech.net) |
| 20:52.26 | *** join/#asterisk anthm (n=anthm@adsl-68-74-96-61.dsl.milwwi.ameritech.net) |
| 20:52.26 | *** mode/#asterisk [+o anthm] by ChanServ |
| 20:59.24 | *** join/#asterisk KaiHanari (n=kaihanar@stjhnf0122w-142162043055.pppoe-dynamic.nl.aliant.net) |
| 21:00.14 | *** join/#asterisk BOLIVIAN (n=ewrtwert@201.222.93.229) |
| 21:00.54 | BOLIVIAN | my HDD has 3 partitions. In one i have Winxp pro. How would i install asterisk now? |
| 21:01.01 | BOLIVIAN | asterisknow |
| 21:06.21 | *** join/#asterisk [TK]D-Fender (n=joe@MTRLPQ02-1177745839.sdsl.bell.ca) |
| 21:06.24 | *** join/#asterisk sjwilliamson (i=swilliam@209.42.110.46) |
| 21:06.41 | sjwilliamson | hey is the business edition reg server down, any digium folks here? |
| 21:07.15 | *** part/#asterisk [[blah]asfd (n=ckwall@63.149.122.93) |
| 21:08.21 | tzafrir_laptop | BOLIVIAN, not sure |
| 21:08.40 | tzafrir_laptop | Is there a partition you can wipe out? |
| 21:08.47 | BOLIVIAN | of course |
| 21:08.51 | BOLIVIAN | there are 2 |
| 21:09.07 | BOLIVIAN | what i want to know if its not gonna erase my winxp |
| 21:09.16 | BOLIVIAN | and will able to double boot |
| 21:09.19 | Qwell[] | sjwilliamson: works here...of course, I'm internal, so... I would recommend calling support |
| 21:09.25 | tzafrir_laptop | I don't really remember if the astrisknow installer asked me about partitioning or not |
| 21:09.37 | Qwell[] | if needed, I believe they can do a manual registration |
| 21:09.59 | Qwell[] | BOLIVIAN: use vmware |
| 21:10.09 | BOLIVIAN | qwell whast that |
| 21:10.25 | sjwilliamson | Qwell: I am on that |
| 21:10.31 | sjwilliamson | or on with them |
| 21:12.15 | BOLIVIAN | qwell whats vmware? where can i find that on the same asteriskpage? |
| 21:14.21 | *** join/#asterisk Dr-Linux|home (n=Dreamer@DSL-202-59-73-131.nexlinx.net.pk) |
| 21:15.13 | tzafrir_laptop | BOLIVIAN, there should be a live cd version there |
| 21:19.40 | BOLIVIAN | tzafrir but what is vmware. A program that allows me to run asterisknow on windows? |
| 21:20.22 | *** join/#asterisk Assid (n=assid@202.88.132.238) |
| 21:20.24 | Assid | heya |
| 21:21.01 | Assid | i am facing alot of DTMF issues, anyone got any suggestions on how to go about fixing it ? i am using ULAW codec, doesnt seem to help |
| 21:24.46 | tzafrir_laptop | a machine emulator. See vmware.com |
| 21:25.02 | tzafrir_laptop | BOLIVIAN, qemu can also run vmware images. See qemu.org |
| 21:25.55 | tzafrir_laptop | Assid, are the DTMFs sent inline? |
| 21:26.22 | *** join/#asterisk matt_ (n=matt@82-46-58-45.cable.ubr01.trow.blueyonder.co.uk) |
| 21:26.57 | Assid | not sure |
| 21:27.47 | Assid | tzafrir_laptop: how do i check |
| 21:28.07 | Assid | working on sip.. |
| 21:28.16 | tzafrir_laptop | not sure as well |
| 21:29.22 | Assid | should i just add dtmfmode=rfc2833 |
| 21:31.18 | alephant | Hey now... |
| 21:31.31 | alephant | ...I've got a Voipstreet SIP trunk that I thought was setup correctly... |
| 21:31.49 | alephant | ...how can I see whether the trunk has in fact registered correctly? |
| 21:32.06 | Assid | sip show registry |
| 21:32.36 | alephant | I see in /var/log/asterisk/full that there are registrations successful, but there's no indication of *which* registration was successful (I have multiple trunks) |
| 21:32.59 | alephant | Ok, I did that before, and they're both "Registered" |
| 21:33.08 | alephant | I just didn't trust that answer :-) |
| 21:33.18 | alephant | So let's try some more outgoing calls |
| 21:33.22 | alephant | Assid: thanks |
| 21:34.55 | *** join/#asterisk tsurko (n=tsurko@vpn-pppoe-213-240-242-237.megalan.bg) |
| 21:39.54 | alephant | Outgoing call via Voipstreet just failed. |
| 21:39.59 | alephant | /var/log/asterisk/full: |
| 21:40.37 | alephant | "No path to translate from SIP/555-... to SIP/Voipstreet |
| 21:40.43 | alephant | Googling it... |
| 21:40.46 | alephant | any clues? |
| 21:41.18 | Assid | do you have a context Voipstreet in sip.conf? |
| 21:41.40 | alephant | Yep |
| 21:41.46 | bkruse | alephant: what codecs are on both people? |
| 21:41.56 | alephant | Just saw a post to that effect :-) |
| 21:41.58 | bkruse | it means your trying to do something that you cannot support transcoding for |
| 21:41.58 | alephant | looking at it now... |
| 21:42.14 | bkruse | pretty easy to diagnose with sip debug |
| 21:42.39 | alephant | sip debug... run asterisk -r and try the call again? |
| 21:43.02 | *** join/#asterisk russellb (n=russell@asterisk/developer-and-stable-maintainer/drumkilla) |
| 21:43.02 | *** mode/#asterisk [+o russellb] by ChanServ |
| 21:43.14 | bkruse | nah |
| 21:43.19 | bkruse | just type sip debug and then call |
| 21:43.28 | bkruse | you should see what codecs its trying to transcode, and why you cannot |
| 21:43.38 | alephant | oO isn't that what I said? |
| 21:43.52 | bkruse | sure |
| 21:43.57 | bkruse | i thought you said restart, my bad. |
| 21:44.13 | alephant | I can just look at the phone's config and the trunk's config and compare codecs, right? |
| 21:44.47 | bkruse | i thought you would have already done that though :P |
| 21:45.03 | alephant | Not yet -- this codec mismatch is new to 10 minutes ago :-) |
| 21:45.07 | bkruse | :P |
| 21:46.20 | alephant | ``allow=g729'' means to only accept that codec along the trunk? |
| 21:47.01 | *** join/#asterisk Defraz (n=t0tal@208.71.137.40) |
| 21:47.17 | Defraz | Do they make a DS3 card for Asterisk? |
| 21:49.42 | funkmaster | sorry for the n00b question, what's the difference between trixbox and freepbx? |
| 21:49.55 | alephant | trixbox is a Linux distribution, like Red Hat |
| 21:50.03 | alephant | FreePBX is a web application |
| 21:50.05 | endre | a form of centos |
| 21:50.07 | endre | afaik |
| 21:50.13 | alephant | FreePBX can be installed on any Linux |
| 21:50.13 | endre | fork* |
| 21:50.32 | alephant | Trixbox is an OS in itself. |
| 21:51.01 | alephant | Like the difference between iTunes (which can be installed on any OS) and OSX, which includes iTunes. |
| 21:51.50 | *** join/#asterisk Cyon (n=cyon@216.179.31.170) |
| 21:52.06 | alephant | Trixbox is a superset of CentOS, which is a "respin" of Red Hat Enterprise Linux. It's 100% compatible (minus the non-free RH stuff like up2date, support contracts and red hats) |
| 21:52.25 | alephant | funkmaster: dig it? |
| 21:52.58 | *** join/#asterisk nosbig (n=nosbig@rrcs-70-62-223-6.central.biz.rr.com) |
| 21:53.07 | funkmaster | yep thx got it |
| 21:53.17 | funkmaster | :) |
| 21:57.53 | tzafrir_laptop | not exactly 100% compatible. They still stick with older kernels and you should pray you don't actually need the recent kernel of CentOS |
| 21:58.20 | *** join/#asterisk booray (n=booray@64.70.85.100) |
| 21:58.28 | JT | and the important thing to note is trixbox includes freepbx |
| 21:58.50 | *** join/#asterisk gerphimum (n=trekkie@207.190.58.85) |
| 21:58.56 | JT | it's the main "drawcard" |
| 21:59.20 | JT | freepbx is mostly a web administration customised asterisk which i don't recommend anyone uses |
| 21:59.26 | booray | is there something else I need to do to disable callwaiting on my sip phones other than having "callwaiting = no" in my users.conf file? |
| 22:00.50 | alephant | JT: why don't you recommend anybody use freepbx? |
| 22:01.22 | JT | it makes disgusting dialplans and prevents you from understanding asterisk correctly and taking full advantage of its capabilities |
| 22:01.30 | JT | ~freepbx |
| 22:01.31 | jbot | somebody said freepbx was unable to be supported here. It's a complex piece of dialplan and GUI, and can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking there |
| 22:01.33 | alephant | fair enough |
| 22:03.29 | alephant | bkruse: what should I be looking for in the very verbose output of sip debug? |
| 22:03.39 | alephant | Should I see the destination phone#? |
| 22:04.37 | bkruse | yes, youll see the codec mismatch |
| 22:04.43 | bkruse | what are the 2 codecs on both ends |
| 22:04.51 | bkruse | and can you have both phones do a playback on the local asterisk machine |
| 22:05.25 | alephant | It's an outgoing call from Asterisk --> PSTN |
| 22:05.29 | alephant | so, no :-) |
| 22:05.49 | alephant | well... from IP phone --> * --> PSTN |
| 22:05.54 | aydiosmio | how do I find out what's causing * to send back a 403 forbidden? I don't see the call in the CDR |
| 22:06.14 | aydiosmio | and asterisk is running -dgvvvv with no output |
| 22:06.25 | aydiosmio | not likely asterisk's fault? |
| 22:06.45 | alephant | bkruse: I think I see the call in my debug output... |
| 22:06.49 | *** join/#asterisk zmef420 (n=zmef420@metarb3-pool3-13.mtco.com) |
| 22:06.56 | alephant | bkruse: ...if I pastebin it, can you give it a look? |
| 22:09.12 | JT | aydiosmio: CDRs aren't usually for stuff that SIP has rejected |
| 22:09.25 | JT | aydiosmio: you may need to do a sip debug |
| 22:09.38 | *** join/#asterisk [TK]D-Fender (n=joe@64.235.216.2) |
| 22:10.18 | aydiosmio | oh duh |
| 22:10.19 | aydiosmio | thanks |
| 22:12.44 | aydiosmio | asterisk was matching the peer on the outbound context instead of the inbound |
| 22:13.05 | JT | ah ok |
| 22:13.27 | *** join/#asterisk flenders (n=fserto@unaffiliated/flenders) |
| 22:13.45 | bkruse | yes |
| 22:13.47 | bkruse | aydiosmio: sup |
| 22:13.56 | alephant | ok... so it looks like [to the untutored eye] * is sending ulaw, alaw to Voipstreet: |
| 22:14.02 | aydiosmio | aloha |
| 22:14.03 | alephant | "Added codec... to SDP" |
| 22:14.29 | *** join/#asterisk dasenjo (n=dasenjo@190.24.178.10) |
| 22:14.32 | *** join/#asterisk shanky (n=shanky@85.137.127.142) |
| 22:17.37 | *** join/#asterisk hematitec (n=cratz@adsl-71-159-206-4.dsl.pltn13.sbcglobal.net) |
| 22:18.41 | alephant | Ok: |
| 22:18.46 | alephant | so on the subject of codecs: |
| 22:19.03 | alephant | The provider's config lists "disallow=all; allow=g729" |
| 22:19.13 | alephant | so we have to use g729 to talk to them |
| 22:19.30 | JT | must be a cheaparse provider |
| 22:19.46 | alephant | the phone is configured to prefer codecs PCMU, PCMA, G.723.1, G.729A/B |
| 22:20.11 | alephant | but in the "sip debug" I see what looks like an INVITE with codecs ulaw, alaw |
| 22:20.20 | alephant | is this *'s own codec choice? |
| 22:20.22 | JT | invite from who |
| 22:20.51 | alephant | invite from * on-behalf-of-phone to sip:4125551212@sip.broadvoice.com (the number dialed) |
| 22:21.10 | alephant | PCMU != ulaw, right? |
| 22:21.19 | JT | PCMU == ulaw |
| 22:21.23 | JT | of course |
| 22:21.34 | alephant | really? |
| 22:21.41 | JT | = g.711u |
| 22:21.43 | JT | Mu law |
| 22:21.44 | JT | etc |
| 22:21.48 | JT | all the same thing |
| 22:21.55 | alephant | ok, so that at least makes sens |
| 22:22.08 | shanky | I know this could be a bit off-topic: Any recommendation for a softphone that works in linux and supports h264? |
| 22:22.37 | alephant | so since the phone is configured to prefer g729 as its fourth-choice, and I only see its first and second choice in the SIP invite |
| 22:22.52 | alephant | is there an * setting which controls how many codecs are offered? |
| 22:23.06 | *** join/#asterisk orkid (n=orkid@bas1-barrie18-1242378267.dsl.bell.ca) |
| 22:23.26 | JT | allow= and disallow= |
| 22:23.48 | alephant | where, in my peer's context? |
| 22:24.19 | JT | yes |
| 22:24.26 | JT | have you looked at the book? |
| 22:24.31 | alephant | so it seems wrong -- the trunk is configured to only send g729, but the SIP invite offers other codecs |
| 22:24.41 | alephant | Not for this issue, no. |
| 22:25.08 | JT | are you using the configuration files? |
| 22:25.18 | *** join/#asterisk nortex (n=breeves@snapper.titanspecialties.com) |
| 22:25.29 | alephant | yep |
| 22:25.52 | Hmmhesays | bah |
| 22:26.48 | JT | ah ok, because i've only ever heard the term "trunk" applied to sip, by freepbx |
| 22:27.23 | alephant | I'm using freepbx as a key to how config files are built, but I always look at what it's done to see how it looks under the hood. |
| 22:27.46 | alephant | ("trunk") is also used by FOP |
| 22:28.26 | alephant | JT: by "book" you mean O'Reilly's, right? |
| 22:29.16 | *** join/#asterisk CrashHD (n=crashhd@c-76-20-22-240.hsd1.ca.comcast.net) |
| 22:30.39 | *** join/#asterisk diclophis-work (n=jbardin@65.203.37.58) |
| 22:30.42 | diclophis-work | hello all |
| 22:30.52 | sahafeez | hum, just got another stupid telemarket call at work. if i have the caller id can i make a list of inbound numbers to not answer/ or play discontect msg to |
| 22:31.00 | diclophis-work | anyone have any expeirence with the build in asterisk-gui (the beta thing) |
| 22:31.30 | Hmmhesays | vbscript is pissing me off today |
| 22:31.56 | *** join/#asterisk [Latre] (n=chatzill@dsl-148-233-19-133.prod-empresarial.com.mx) |
| 22:32.25 | russellb | sahafeez: there is a blacklist application of some sort |
| 22:32.29 | russellb | haven't looked at that in a long time ... |
| 22:32.50 | sahafeez | hum, i will have to look. i just want it to say, oh, that number, and play the disconenct msg |
| 22:33.43 | JT | alephant: yes |
| 22:33.45 | JT | ~thebook |
| 22:33.46 | jbot | somebody said thebook was a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 22:33.51 | alephant | Got it |
| 22:33.52 | alephant | reading it |
| 22:34.00 | alephant | unfortunately it's now time for dinner :-/ |
| 22:34.01 | JT | alephant: yeah but it's sort of like how not to build a dialplan |
| 22:34.13 | JT | 40+lines to ring a simple phone number is ridiculous |
| 22:34.28 | alephant | JT: I can appreciate that, and ssh+vi is way faster than freepbx |
| 22:34.31 | JT | trunk is a term that makes no sense on SIP |
| 22:34.42 | JT | sip can establish connections ad-hoc |
| 22:34.43 | alephant | so anyway |
| 22:35.01 | alephant | I have two avenues for investigation: codec mismatch and sip invite debugging |
| 22:35.09 | alephant | JT: thanks for the help |
| 22:36.06 | eald | 5~ |
| 22:38.01 | *** join/#asterisk topping (n=topping@h-67-100-91-18.snfccasy.covad.net) |
| 22:39.27 | *** join/#asterisk topping (n=topping@h-67-100-91-18.snfccasy.covad.net) |
| 22:39.41 | *** join/#asterisk riddlebox (n=james@75-132-205-166.dhcp.stls.mo.charter.com) |
| 22:40.08 | *** join/#asterisk vlt (n=daniel@dslb-088-073-187-196.pools.arcor-ip.net) |
| 22:40.59 | riddlebox | wohoo, my company is looking at becoming digium resellers |
| 22:41.17 | russellb | yay |
| 22:41.32 | riddlebox | I did a demo of asterisknow for them yesterday and they were impressed |
| 22:41.49 | twisted[asteria] | russellb |
| 22:41.56 | russellb | twisted[asteria]: |
| 22:42.08 | russellb | this is not the russellb you are looking for |
| 22:43.02 | vlt | Hello. I just installed a 4xBRI card (HFC chip) to my asterisk server. The kernel seems to recognize it: I get every call listed in dmesg (and to which of the ports 0-3 the call goes to). How can I access this card from asterisk? What to do now? Do I need bristuff or ...? |
| 22:43.43 | *** join/#asterisk r0d3nt (i=nobody@foster.stonedcoder.org) |
| 22:44.12 | vlt | zaptel ... Can anyone point me to the right direction? |
| 22:44.17 | *** join/#asterisk Defraz (n=t0tal@208.71.137.40) |
| 22:44.27 | Defraz | Hey all! |
| 22:45.29 | JT | vlt: i recommend bristuff |
| 22:45.59 | JT | vlt: which card in particular? |
| 22:48.59 | vlt | JT: lspci says: "0000:00:0b.0 ISDN controller: Cologne Chip Designs GmbH: Unknown device 08b4 (rev 01)", dmesg output follows ... |
| 22:49.22 | *** join/#asterisk cthorner (n=cthorner@209-234-185-130.static.twtelecom.net) |
| 22:49.26 | Cresl1n | mISDN works as well |
| 22:49.29 | Cresl1n | pick your poison |
| 22:49.45 | JT | vlt: you don't know who made your card? |
| 22:49.55 | JT | misdn works a little bit, for some basic stuff |
| 22:50.34 | nibbler_de | JT: it's rather unimportant who made the card, isn't it? |
| 22:50.54 | vlt | JT: "HFC 08b4" isn't enough? |
| 22:51.02 | JT | nibbler_de: it's important, in sofar as bristuff checks for the junghanns vendor id if you don't patch it |
| 22:51.19 | JT | vlt: well where did it come from? i assume it didn't come from outer space |
| 22:51.29 | JT | vlt: they're all pretty much the same except for vendor id |
| 22:51.43 | vlt | JT: Ok, it's not a genuine Junghanns card |
| 22:52.06 | vlt | JT: I don't know the exact vendor |
| 22:52.06 | JT | it's either junghanns, beronet or digium, i would think |
| 22:54.10 | *** join/#asterisk bhrobinson (n=brobinso@northtx1-static.telwestonline.com) |
| 22:54.26 | JT | vlt: no stickers or anything on the card? |
| 22:55.10 | bhrobinson | does anyone know much about the iaxy? |
| 22:55.58 | vlt | JT: Sorry, the card is about 80 miles away from me ... |
| 22:56.24 | vlt | JT: The kernel module used now is "hfc4s8s_l1" |
| 22:56.46 | vlt | JT: (and hisax). |
| 22:57.53 | *** join/#asterisk booray (n=booray@64.70.85.100) |
| 22:58.13 | JT | doesn't sound like misdn or bristuff |
| 22:59.42 | booray | can someone please help me with my subscription status? a sip_show_subscriptions either says Idle or <none>, and core_show_hints is either Idle or Unavailable. I'd think I would at least be able to see in use or ringing. Does anyone have an idea of what I can look at or try? Thanks... |
| 23:00.11 | cthorner | it's broken in 1.4 |
| 23:00.26 | cthorner | presence awareness |
| 23:00.27 | booray | really... |
| 23:00.36 | cthorner | patches are available from svn |
| 23:00.59 | *** join/#asterisk ToyMan (n=Stuart@user-12lcqu6.cable.mindspring.com) |
| 23:01.01 | booray | cthorner: do you know offchance if a svn update would break anything else? |
| 23:01.02 | cthorner | 1.4 branches not sure if it is in trunk yet |
| 23:01.12 | *** part/#asterisk jmls (n=asterisk@host81-159-198-100.range81-159.btcentralplus.com) |
| 23:01.33 | diclophis-work | why would a SIP INVITE try to use username = "s" ?? |
| 23:01.35 | cthorner | not sure |
| 23:01.46 | booray | cthorner: thanks for the info. After hours I'll compile from fresh svn |
| 23:01.55 | cthorner | ok, good luck |
| 23:02.18 | booray | thanks |
| 23:02.29 | cthorner | np |
| 23:02.52 | bkruse | you ever pastbin that config? |
| 23:02.57 | *** join/#asterisk foobar778 (n=bryan@ip68-100-41-120.dc.dc.cox.net) |
| 23:03.18 | bkruse | alephant left, darn |
| 23:03.40 | jpe-nyc | hello all, I dont know where to start, so I will ask here. Whre can I find info relating to having asterisk playing back sound files in a folder in succession, with play last, skip and mark functionality.. |
| 23:04.48 | foobar778 | Hi friends got asterisk working have a few issues can someone call me on my PBX People have connectefd finde sometimes just one way audio if anyone can calll me to test would be appreciayive |
| 23:06.15 | *** join/#asterisk nighty- (n=nighty-@66-163-28-100.ip.tor.radiant.net) |
| 23:07.10 | vlt | JT: I found the dmesg output from boot process: http://paste.ubuntu-nl.org/5841/ |
| 23:07.20 | JT | jpe-nyc: ControlPlayback allows fast forwarding and rewinding etc, not sure about marking, you could probably mark a file as a whole using dialplan logic, but not a specific position within the file |
| 23:07.36 | *** part/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net) |
| 23:08.13 | *** join/#asterisk defi (n=defi@209-234-185-130.static.twtelecom.net) |
| 23:08.20 | jpe-nyc | JT thanks... |
| 23:10.07 | JT | vlt: hisax is a really old driver |
| 23:10.12 | JT | back from the isdn4linux days |
| 23:10.38 | bkruse | <3 |
| 23:12.10 | vlt | JT: So what to do now? |
| 23:14.02 | JT | remove that stuff an install bristuff :) |
| 23:14.31 | JT | i have no idea if it's a junghanns card, do you'll probably need to patch qozap before compiling to remove vendorid checking |
| 23:15.12 | bkruse | </3 bristuff |
| 23:15.27 | JT | bristuff is awesome, don't knock it |
| 23:15.44 | JT | all bri software has a level of betaness it seems, bristuff has the lowest level |
| 23:16.04 | bkruse | JT: well worded. |
| 23:16.11 | vlt | JT: remove the hisax driver? I didn't install anything before. I just plugged in the card and rebooted. I think I need some more help ... |
| 23:16.30 | JT | misdn is piss fart useless if you want to connect a bri from asterisk to an isdn pabx |
| 23:16.35 | JT | was that as well worded ;) |
| 23:16.48 | JT | vlt: hrm |
| 23:16.56 | JT | vlt: what kernel are you running? |
| 23:17.26 | JT | vlt: i still can't understand how you get a quad port bri card without knowing where it came from, those things aren't cheap |
| 23:18.09 | *** join/#asterisk doolph (n=doolph@200.46.148.61) |
| 23:18.23 | doolph | hello, anyone know how to match rtp packets with iptables? |
| 23:18.25 | vlt | JT: 2.6.15 (Ubuntu Dapper) |
| 23:18.48 | vlt | JT: The card was about 400 EUR, I think ... |
| 23:18.58 | JT | sounds about right |
| 23:20.44 | JT | vlt: i have no idea why it'd be loading hisax, apart from maybe ubuntu's default kernel being on crack |
| 23:21.25 | vlt | :) |
| 23:22.39 | vlt | Can anyone tell me what exactly mISDN or bristuff is? |
| 23:23.01 | vlt | For BRI I don't need zaptel, is that right? |
| 23:23.13 | JT | misdn is a channel driver for isdn |
| 23:23.25 | Hmmhesays | bah I can't get apache set up to use mod_perl |
| 23:23.32 | JT | bristuff is a patchset for asterisk that patches asterisk, zaptel, and libpri to add bri functionality |
| 23:23.48 | JT | so you can access a bri channel just as if it were a zaptel pri channel |
| 23:23.56 | JT | and use all zap features |
| 23:25.24 | vlt | What would be an alternative way for asterisk accessing the card (if any) and what features are missing there? |
| 23:25.55 | JT | misdn, bristuff, and tzafrir's zaphfc patch are the current ways to access it |
| 23:26.18 | *** join/#asterisk anthonyl (n=fbfff@c-67-167-73-64.hsd1.il.comcast.net) |
| 23:26.24 | anthonyl | #asterisk-bugs |
| 23:26.35 | foobar778 | friend call i get han_sip.c:1244 retrans_pkt: Hanging up call NmQxMDMwODcyY2RmNWMzNDBjODAyODZkNjlkYThiMWI. - no reply to our critical packet. |
| 23:26.35 | foobar778 | <PROTECTED> |
| 23:26.35 | foobar778 | <PROTECTED> |
| 23:26.36 | anthonyl | oops, sorry about that |
| 23:27.30 | JT | misdn does not support group dial on NT ports, and crashed a hell of a lot on my 8 port hfc card |
| 23:29.40 | vlt | JT: I'm just reading a chan_misdn tutorial ... should I rather stop it? |
| 23:30.15 | *** join/#asterisk ManxPower (n=manxpowe@stirprop-s4-0-0-21.ndcr2.datasync.net) |
| 23:30.16 | JT | i'd recommend bristuff to be honest, there will always be people to try and sway you either way |
| 23:30.46 | JT | but i'd say that since you have a multiport card, you should definately not go misdn, as its multiport driver is extremely buggy |
| 23:31.58 | foobar778 | Can anyone help with this problem : chan_sip.c:1244 retrans_pkt: Hanging up call NmQxMDMwODcyY2RmNWMzNDBjODAyODZkNjlkYThiMWI. - no reply to our critical packet. |
| 23:32.03 | vlt | JT: I want to understand it: The replacement for chan_misdn you'd reccomend is a bristuff patched chan_zap, right? |
| 23:33.07 | JT | vlt: yes, bristuff is very easy to install, you download the latest bristuff from junghanns.net/downloads and extract it, and there are some scripts that download and patch all parts for you |
| 23:33.10 | tzafrir_laptop | for starters, there is the zaptel patch and there is the libpri/asterisk patch |
| 23:33.33 | JT | there's an article on voip info you'll need to find on how to manually patch qozap so it does not look for junghanns vendorid |
| 23:33.56 | tzafrir_laptop | The zaptel patch is small and understandable. The asterisk/libpri is some sort of a rewrite |
| 23:35.08 | vlt | I found this one that comes really close (dapper and 3rd party vendor here): http://www.voip-info.org/wiki/view/Asterisk+BRI-stuffed+Ubuntu+Edgy+Junghanns |
| 23:35.32 | foobar778 | <PROTECTED> |
| 23:35.37 | tzafrir_laptop | vlt, Debian basically has bristuffed packages |
| 23:35.45 | JT | your distro shouldn't matter too much as long as you have 2.6 and the kernel headers are installed |
| 23:36.00 | JT | they're probably out of date |
| 23:36.41 | JT | i recommend bristuff-0.3.0-PRE-1w at a MINIMUM |
| 23:36.42 | Cresl1n | tzafrir_laptop: actually, the zaptel patch is unnecessary now |
| 23:37.00 | tzafrir_laptop | Cresl1n, huh? |
| 23:37.01 | JT | Cresl1n: why's that? |
| 23:37.02 | Cresl1n | he just needs to update to use the new interface in zaptel for accessing HDLC buffers directly |
| 23:37.13 | tzafrir_laptop | interesting |
| 23:37.22 | vlt | JT: From that page: "The card I used is a Junghanns QuadBRI ISDN card, which uses a Cologne HFC4S chipset. If you install a default OS with the card, the kernel tries to load the wrong module (hfc4s8s_l1) " |
| 23:37.25 | JT | what is this new interface |
| 23:37.31 | Cresl1n | I added it when I did hardware hdlc support for zaptel |
| 23:37.35 | foobar778 | <PROTECTED> |
| 23:37.45 | Cresl1n | JT: it's a software API interface |
| 23:37.51 | vlt | JT: That's the module mentioned earlier. |
| 23:37.58 | JT | vlt: right |
| 23:38.01 | ManxPower | foobar778: if you repeat the same question too many times the bot will kickban you. |
| 23:38.07 | ManxPower | If someone could help, they will help. |
| 23:38.12 | JT | Cresl1n: what uses this interface? |
| 23:38.14 | foobar778 | ok |
| 23:38.25 | Cresl1n | right now, just the wct4xxp driver |
| 23:38.34 | ManxPower | foobar778: and if you cannot get help here, ask on the mailing lists |
| 23:38.38 | ManxPower | ~mailinglist |
| 23:38.48 | jbot | Search Asterisk mailing lists by prepending site:lists.digium.com to your Google search. Browse the mailing list archive at http://lists.digium.com/ or search through it at http://www.asteriskguru.com/archives, or and there is also the Macintosh Asterisk mailing list at http://www.astmasters.net/maml.htmm |
| 23:38.49 | JT | Cresl1n: so this is useful to us a bri card, how? :) |
| 23:38.49 | Cresl1n | but the hfc based cards could use them as well |
| 23:38.49 | JT | could... |
| 23:39.08 | Cresl1n | JT: baseically it's an interface for the BRI cards to use so they don't have to patch zaptel anymore |
| 23:39.47 | JT | Cresl1n: there are a number of extra functions that bristuff adds however, in any case |
| 23:40.02 | Cresl1n | JT: I'm just talking about the zaptel patch |
| 23:40.07 | Cresl1n | not everything else bristuff does |
| 23:40.12 | JT | right |
| 23:41.02 | JT | Cresl1n: i don't suppose you have an authorative answer to the mystical question of if it is possible to use bri cards under asterisk on any USA BRIs? :) |
| 23:41.19 | tzafrir_laptop | Cresl1n, so with a bit of work, someone could actually use qozap with a b410p card? |
| 23:41.21 | Cresl1n | with an S/T to U interface converter you can |
| 23:41.56 | JT | tzafrir_laptop: how different is the b410p to the junghanns and beronet cards? |
| 23:42.05 | JT | Cresl1n: do US BRIs not have an NT1? |
| 23:42.15 | JT | Cresl1n: i was talking from a switchtype perspective |
| 23:42.29 | JT | as it seems no-one has reported physically testing it and being successful |
| 23:42.52 | ManxPower | JT: In the USa you have to buy your own NT1, the telco does not provide it. |
| 23:42.55 | Cresl1n | JT: probably, with a little bit of work |
| 23:42.58 | tzafrir_laptop | JT, I haven't seen any mention of hardware EC on the Junghanns or the ber cards |
| 23:43.49 | JT | ManxPower: right |
| 23:43.57 | ManxPower | However, in the USA the telco normally provides you with S/T Interface, I believe. It might be U, whatever it is, it's not what you get in the rest of the worlld. |
| 23:44.05 | bkw_ | U |
| 23:44.09 | bkw_ | you get a U interface in the US |
| 23:44.11 | Cresl1n | yeah, it's U |
| 23:44.13 | JT | Cresl1n: that's the main issue, the switchtype, no-one is convinced there's actually any support |
| 23:44.19 | JT | U is what goes in the street |
| 23:44.25 | Cresl1n | JT: that probably won't be hard to do |
| 23:44.40 | JT | Cresl1n: it also sounds like no-one has tested it |
| 23:44.44 | ManxPower | Thanlks. I've not dealt with BRI in many years |
| 23:44.45 | bkw_ | every BRI I have had was given to me in a U interface |
| 23:44.57 | JT | Cresl1n: hrm, there's a few people in this channel who would be interested in that |
| 23:46.37 | ManxPower | It's nice that Digium has a good EC available, it is just several years too late for us. |
| 23:48.14 | JT | for what interface? |
| 23:48.55 | ManxPower | JT: Hmm? |
| 23:49.06 | JT | your EC? |
| 23:49.14 | ManxPower | T-1 |
| 23:49.25 | JT | right |
| 23:49.53 | ManxPower | About $150 for chassis and power supply for up to 12 T-1s, not that we ever need more than 2 T-1s at an office |
| 23:52.22 | *** join/#asterisk MooingLemur (n=troy@unaffiliated/mooinglemur) |
| 23:53.18 | JT | ManxPower: is it modules on top of that, or it does up to 12 as is? |
| 23:55.56 | vlt | JT: There's a bristuffed asterisk package in ubuntu with zaptel which I installed: http://paste.ubuntu-nl.org/5846/ |
| 23:56.04 | ManxPower | The chassis has 12 slots IIRC |
| 23:56.26 | ManxPower | Many of the used Tellabs comes with 24 modules (1 modules = 1 T-1) |
| 23:56.35 | vlt | JT: I didn't patch it before. What does bristuff do to non-Junghanns cards? |
| 23:57.18 | JT | for multiport bri cards, it doesn't work, usually, if they're not junghanns |
| 23:57.35 | JT | why can't you just compile it, forget all this prepackaged stuff |
| 23:57.39 | JT | i mean you can try it |
| 23:57.48 | JT | but if it doesn't work, just compile it |
| 23:58.29 | diclophis-work | is there a manager command that will "pause" a queue ? |
| 23:59.46 | vlt | JT: I'll try. I have to. I get "qozap: not re-activating layer1 span 2" every 10 seconds in dmesg |