| 00:00.46 | The-Kernel | register=122059:*******@losangeles.voip.ms:5060 |
| 00:01.19 | Tozz_ | thats not correct |
| 00:01.22 | Tozz_ | it should be |
| 00:01.34 | Tozz_ | register => 122059:**@losangelas.voip.ms |
| 00:01.40 | Tozz_ | not 'register=' |
| 00:04.25 | The-Kernel | no change |
| 00:04.49 | Tozz_ | what do the logs say? |
| 00:05.10 | Tozz_ | u did sip reload? |
| 00:05.15 | The-Kernel | yeah |
| 00:05.48 | Tozz_ | then look at the logs / turn on some debugging / verbose logging |
| 00:06.02 | *** join/#asterisk puzzled (~patrick@puzzled.xs4all.nl) |
| 00:06.24 | puzzled | hi |
| 00:10.48 | *** join/#asterisk CoderForLife (~Miranda@cpe-174-103-152-61.cinci.res.rr.com) |
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| 00:16.05 | The-Kernel | Tozz_, is there a way to force it to register |
| 00:16.22 | The-Kernel | I don't see anything in the debug after a reload about it trying to register with the SIP Provider |
| 00:16.50 | Tozz_ | it should register as soon as you hit sip reload, so there is something wrong with your config |
| 00:17.01 | Tozz_ | where did u place the register => line? |
| 00:17.10 | The-Kernel | sip.conf |
| 00:17.11 | Tozz_ | is it within the [global] context? |
| 00:17.19 | The-Kernel | hrm let me check |
| 00:17.37 | The-Kernel | [general] |
| 00:17.52 | Tozz_ | yes, thats ok |
| 00:17.53 | Tozz_ | my bad |
| 00:18.11 | luckman212 | er, can MOH play back anything besides 16bit 8000hz PCM? I ask because I was wanting to make a high-quality MOH stream for G722 callers |
| 00:18.48 | Tozz_ | paste your sip.conf somewhere |
| 00:18.54 | Tozz_ | on pb |
| 00:23.52 | The-Kernel | http://pastebin.com/6Y186CtG |
| 00:25.06 | *** part/#asterisk qhartman (~qhartman@quentin.telecom.uoregon.edu) |
| 00:25.24 | Tozz_ | strange, it should try to register |
| 00:26.10 | The-Kernel | this is all that pops up http://pastebin.com/LpWcZtXW |
| 00:28.16 | *** join/#asterisk lanning (~lanning@173-8-187-197-SFBA.hfc.comcastbusiness.net) |
| 00:30.15 | The-Kernel | Tozz_, ever seen this before? |
| 00:35.06 | luckman212 | anybody know the answer to my q? |
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| 01:20.47 | FLeiXiuS | IS there a protocol that asterisk supports that I would enable me to make calls from an embedded piece of code in a website |
| 01:20.51 | leifmadsen | luckman212: MOH can play any format that Asterisk has the capability to play. If you want to play back G722 quality audio, then create G722 files. |
| 01:21.32 | leifmadsen | FLeiXiuS: that would be the capability of the web application to connect to asterisk and place the call. The short answer is no. The longer answer is yes, with the right web interface. |
| 01:22.01 | leifmadsen | and now I go get into my robe and watch TV |
| 01:22.38 | FLeiXiuS | <leifmadsen> Sure, but there has to be a protocol that asterisks speaks which is easily incorporated into modern browsers. |
| 01:23.41 | ChannelZ | Not really |
| 01:24.10 | ChannelZ | But there are a couple of (probably Java?) web-based SIP clients |
| 01:24.10 | FLeiXiuS | Flash has RTMP support, but asterisk really doesnt handle rtmp that well |
| 01:24.32 | FLeiXiuS | I'll take a SIP Java based client, if I could find a free OSS one |
| 01:24.58 | ChannelZ | good luck |
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| 04:04.38 | fauxalliance | chokes on a fax |
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| 05:21.53 | ledoktre | good evening |
| 05:22.47 | ledoktre | i just installed Asterisk 1.6.x for the first time (I used to use Asterisk pre 1.4 but I am rusty !!) |
| 05:23.22 | ledoktre | and I can't seem to get my Sipura SPA-1001 boxes to register. THey either say, they need to be set host=dynamic, or even when they are set as "friend", they say that a peer cannot be registered |
| 05:23.27 | ledoktre | any idea where to start debugging ? |
| 05:30.38 | *** join/#asterisk aidinb (~Aidin@unaffiliated/aidinb) |
| 05:35.49 | ChannelZ | sip.conf |
| 05:37.32 | ChannelZ | if your types are peers and set to be dynamic, then the username the phone is registering with must match the peer name (the bit between [brackets] in sip.conf) |
| 05:39.32 | ledoktre | I have the type set to friend, and host=dynamic |
| 05:39.49 | ledoktre | I am actually getting a different error now (i am messing with the settings, trying to figure it out) |
| 05:40.21 | ledoktre | handle_request_register: Registration from '<sip:100@domain.com>' failed for 'xxx.xxx.xxx.xxx' - No matching peer found |
| 05:41.02 | ledoktre | 100 is the extension i assigned in extensions.conf. the user was actually called "shop" |
| 06:15.59 | *** join/#asterisk aidinb (~Aidin@unaffiliated/aidinb) |
| 06:30.58 | *** join/#asterisk aidinb (~Aidin@unaffiliated/aidinb) |
| 06:36.39 | *** join/#asterisk aidinb (~Aidin@unaffiliated/aidinb) |
| 06:36.44 | ChannelZ | extensions and SIP devices have nothing to do with one another |
| 06:36.58 | ChannelZ | nothing you do in extensions.conf will make your phone register |
| 06:39.32 | *** join/#asterisk aidinb (~Aidin@unaffiliated/aidinb) |
| 06:47.05 | *** join/#asterisk raden (~J@66-168-15-100.dhcp.stpt.wi.charter.com) |
| 06:47.26 | raden | anyone have a script to move call files one by one into the spool dir ? |
| 06:48.10 | WIMPy | for? |
| 06:48.32 | WIMPy | Or foreach if you want to use a csh. |
| 06:50.09 | raden | ? |
| 06:50.44 | raden | <PROTECTED> |
| 06:51.19 | kaldemar | for i in /path/to/*.callfile ; do mv "$i" /your/spool/dir/ ; done |
| 06:51.21 | kaldemar | for example. |
| 06:51.47 | WIMPy | So per call from cron only one file? |
| 06:52.43 | *** join/#asterisk Tim_Toady (~moi@178.128.24.211.dsl.dyn.forthnet.gr) |
| 06:53.01 | raden | WIMPy, yes |
| 06:53.41 | raden | for i ? |
| 06:54.18 | WIMPy | f=`ls;head -n 1`;mv $f /spool/dir/ |
| 06:54.38 | kaldemar | raden: looping, i gets the filename for each file in the specified subset. |
| 06:54.49 | WIMPy | Apply sorting options to ls as you like. |
| 06:55.31 | kaldemar | ; -> | between ls and head. |
| 06:55.50 | raden | kaldemar, how does it only know to do 1 ? |
| 06:55.54 | WIMPy | oops. yes |
| 06:56.23 | kaldemar | raden: it doesn't, i wrote that before reading your requirement to move only 1 file. that would move *.callfile. |
| 06:56.27 | raden | kaldemar, not very good with bash i must admit :( |
| 06:56.36 | raden | kaldemar, gotcha |
| 06:56.50 | raden | so what WIMPy has would be correct ? |
| 06:57.18 | WIMPy | Almost |
| 06:57.23 | raden | lmao |
| 06:57.25 | WIMPy | f=`ls|head -n 1`;mv $f /spool/dir/ |
| 06:57.32 | WIMPy | As kaldemar pointed out. |
| 06:57.38 | raden | tht i caught :) |
| 06:57.49 | kaldemar | one way of doing it. "mv `ls -1 | head -n 1` /spool/" would be my choice. :) |
| 06:58.15 | *** join/#asterisk drmessano (~nonya@pdpc/supporter/active/drmessano) |
| 06:58.25 | WIMPy | Yes, That's because I often rename files as well when doing such things. |
| 06:58.29 | raden | kaldemar, I did not know you could run a command in a command like that that would make most sense |
| 06:58.47 | WIMPy | Thate apre probably at leas tens of obvious ways :-) |
| 06:58.56 | WIMPy | argh |
| 06:59.09 | raden | lol they are both good : ) |
| 06:59.13 | WIMPy | There are probably at least tens of obvious ways :-) |
| 06:59.18 | kaldemar | you can use bacticks or $(), depends on what syntax you like and want to conform to. |
| 06:59.34 | raden | i need to read bash book one of these days :P |
| 06:59.59 | WIMPy | $() is bash only, I guess? |
| 07:00.08 | raden | everyone says not to parse ls ? is there a more correct way to do it ? |
| 07:00.59 | WIMPy | If you have files in that dir with unknown names it might be better not to. |
| 07:02.32 | raden | there all *.call files |
| 07:02.48 | WIMPy | It's only about the names. |
| 07:02.54 | raden | there all ten digit phone numbers 8005551234.call |
| 07:03.22 | WIMPy | If there's apossibility someone put files with interesting names there, you could get in trouble. |
| 07:03.47 | raden | php script that writes to that directory and thats it ! |
| 07:03.57 | *** join/#asterisk puzzled (~patrick@puzzled.xs4all.nl) |
| 07:04.03 | WIMPy | ls -b might help. |
| 07:04.04 | raden | can find be used somehow ? |
| 07:04.26 | kaldemar | certainly. |
| 07:04.30 | WIMPy | There's an exec to find, but no limit. |
| 07:04.38 | raden | ? |
| 07:04.50 | raden | the files will always be XXXxxxXXXX.call |
| 07:05.27 | WIMPy | If you're sure the names don't contain evil characters, you're ok. |
| 07:06.03 | WIMPy | I'm not sure -b catches everything, but it certainlty won't hurt. |
| 07:06.29 | raden | ok |
| 07:06.31 | raden | thanks |
| 07:07.48 | WIMPy | And you can quote the inlined command. |
| 07:09.39 | kaldemar | $() is also defined in the POSIX spec, had to check. :) |
| 07:09.41 | WIMPy | "`ls`" |
| 07:10.01 | raden | good deal |
| 07:10.09 | raden | learned alot tonight : ) |
| 07:10.37 | kaldemar | http://pubs.opengroup.org/onlinepubs/9699919799/utilities/V3_chap02.html#tag_18_06_03 |
| 07:10.38 | raden | works nicely |
| 07:11.19 | kaldemar | that whole chapter is a good read if you're new to shell syntax. |
| 07:11.46 | raden | ls did not work with option -l |
| 07:11.59 | raden | thanks :) |
| 07:12.10 | raden | removed that to remove total and all is well :) |
| 07:12.14 | *** join/#asterisk matt4d617474 (~mjb@ks-138-210-221-24.dhcp.embarqhsd.net) |
| 07:12.17 | WIMPy | Sure. You only want the name. |
| 07:14.33 | *** join/#asterisk ChannelZ (channelz@burner.com) |
| 07:17.58 | raden | if there is nothing in that directory how would I keep the script from trying to copy ? |
| 07:18.33 | *** join/#asterisk werdan7 (~w7@freenode/staff/wikimedia.werdan7) |
| 07:19.22 | WIMPy | test -e *&& |
| 07:19.31 | raden | ? |
| 07:19.41 | WIMPy | And the rest thereafter. |
| 07:19.51 | WIMPy | Again only one of many possibilities. |
| 07:21.34 | *** join/#asterisk gerhard7 (~gerhard7@212-123-146-122.ip.telfort.nl) |
| 07:22.36 | raden | so test checks to see if the file exists and what does *&& do ? |
| 07:23.05 | WIMPy | Hmm. No that was not good. |
| 07:23.15 | raden | ??? |
| 07:23.20 | WIMPy | test -n "*"&& |
| 07:23.21 | *** join/#asterisk Juggie (~Juggie@unaffiliated/juggie) |
| 07:23.51 | WIMPy | test -e checks the existance of a file. But * would be expanded to nothing. |
| 07:24.10 | WIMPy | test -e \*&& should work however. |
| 07:24.36 | WIMPy | * is the name to search for and && mens to continue on success only. |
| 07:25.03 | WIMPy | like if [ -e \* ];then |
| 07:25.08 | WIMPy | just shorter |
| 07:25.50 | raden | lil confussed |
| 07:26.32 | WIMPy | test -n "*" tests if "*" expands to more than nothing. |
| 07:28.27 | *** join/#asterisk DJClean (~djclean@unaffiliated/djclean) |
| 07:28.44 | raden | interesting |
| 07:31.08 | WIMPy | I know. Lots of interesting stuff in that adventure :-) |
| 07:31.44 | WIMPy | BTW: Do you call it in regular intercals or only on certain times of day? |
| 07:33.01 | WIMPy | An infinite loop with sleep might be an alternative to cron. |
| 07:33.30 | raden | between 10 am - 5 pm |
| 07:33.36 | raden | mon - fri |
| 07:34.31 | raden | i need to sleep thanks for everything ill catchya tommorow when i have more questions :) |
| 07:42.29 | WIMPy | kaldemar: Thanks for that link and raden for asking :-) I didn't know you can use ${x%y}. Should be useful. |
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| 07:55.55 | *** join/#asterisk black187 (54ffea5e@gateway/web/freenode/ip.84.255.234.94) |
| 07:56.17 | black187 | hello, anybody here using Digium skype solution? |
| 07:57.30 | black187 | I have a question - is there a two way calling from Asterisk to Skype and back - Asterisk user calls pre-defined skype user. Skype user calls via skype manager to Asterisk user? |
| 08:02.23 | ChannelZ | You can call Skype-to-Asterisk or Asterisk-to-Skype, yes, if that's what you're asking |
| 08:04.20 | *** join/#asterisk awclin (~alinford@pha75-1-87-91-238-195.dsl.club-internet.fr) |
| 08:04.47 | black187 | OK. So no need for skype credit - and can you manipulate rules from skype network to Asterisk (iVR, or calling from Skype via Asterisk to external numbers)? |
| 08:07.40 | *** join/#asterisk manji (~manjiki@penicuik.physics.upatras.gr) |
| 08:12.08 | ChannelZ | Skype for Asterisk is a channel driver just like SIP or IAX or DAHDI... so you can do anything you want within the dialplan |
| 08:12.55 | ChannelZ | The only difficulty is calling Skype usernames, since it's a bit difficult to dial alphanumeric names from most phones |
| 08:13.14 | ChannelZ | but you can program the dialplan to call specific users via some arbitrary extension |
| 08:16.57 | *** join/#asterisk tamiel (~tamiel@213.30.183.226) |
| 08:17.12 | *** join/#asterisk Corydon76-home (mauve@c-69-137-80-31.hsd1.tn.comcast.net) |
| 08:17.12 | *** mode/#asterisk [+o Corydon76-home] by ChanServ |
| 08:19.23 | WIMPy | Aren't IP phones considered standard in here? |
| 08:20.19 | ChannelZ | with qwerty keyboards? |
| 08:20.47 | WIMPy | You don't need that. But they certainly make it easier. |
| 08:21.41 | ChannelZ | thus I said "it's a bit difficult", not impossible |
| 10:56.30 | *** join/#asterisk infobot (ibot@rikers.org) |
| 10:56.30 | *** topic/#asterisk is Asterisk: The Open Source PBX and Telephony Platform (asterisk.org) -=- Asterisk 1.8.2.3 (2010/01/26), 1.6.2.16.1 (2010/01/18), 1.4.39.1 (2010/01/18), *-Addons 1.6.2.3, 1.4.13 (2010/01/14), dahdi-linux 2.4.0 (2010/09/01), dahdi-tools 2.4.0 (2010/09/01), libpri 1.4.11.5 (2010/11/22) -=- Visit the new official Asterisk wiki: wiki.asterisk.org -=- http://xkcd.com/806/ |
| 10:56.31 | wdoekes2 | ok.. I wasn't sure about what it would do.. too bad |
| 10:56.50 | asterisk-learner | :-( |
| 10:57.39 | schmidts | asterisk-learner from what version to now have you noticed this problem? |
| 10:58.08 | asterisk-learner | schimdts: i used an old one , 1.4.10 and today upgraded to 1.4.36 |
| 10:58.42 | schmidts | asterisk-learner oh thats only 3 or 4 years in between :( |
| 10:59.22 | asterisk-learner | yeah exactly |
| 11:17.50 | *** part/#asterisk JAMMAN2110 (~JAMMAN211@unaffiliated/jamman2110) |
| 11:29.02 | asterisk-learner | So no one has used Agentlogin() in asterisk 1.4.36 without moh on log in ? |
| 11:38.35 | *** join/#asterisk Shazaum (~1@unaffiliated/shazaum) |
| 11:40.42 | *** join/#asterisk xheliox (jeff@pdpc/supporter/student/xheliox) |
| 11:43.00 | Shazaum | Hi |
| 11:43.08 | Shazaum | I have a problem |
| 11:43.10 | Shazaum | http://pastebin.com/yvrfdjC8 |
| 11:43.12 | Shazaum | :( |
| 11:44.52 | *** join/#asterisk matt4d617474 (~mjb@ks-138-210-221-24.dhcp.embarqhsd.net) |
| 11:45.34 | *** join/#asterisk hrhrhr (~c1@unaffiliated/hrhrhr) |
| 11:47.27 | *** join/#asterisk codebanshee (~chris@81.171.245.107) |
| 11:48.02 | *** part/#asterisk codebanshee (~chris@81.171.245.107) |
| 11:49.53 | schmidts | shazaum looks like a deadlocked channel, you only can try to restart asterisk but i think this will not help, you have to kill it |
| 11:50.55 | Shazaum | schmidts, yes, if i restart it happens again |
| 11:51.46 | Shazaum | otherwise i can not kill, unless a restart |
| 11:52.28 | schmidts | shazaum no i meant a restart will not work, cause if this channel is really deadlocked asterisk could not shut down so you have to kill the asterisk process |
| 11:56.06 | asterisk-learner | schimdts: how did u know it is a deadlocked channel ? bcz it is in the h extension and its state is up or bcz he tried to softhang up it and it didnt hang up |
| 11:56.07 | asterisk-learner | ? |
| 11:57.13 | Shazaum | schmidts, hm ok |
| 11:57.14 | Shazaum | thks |
| 11:58.59 | *** join/#asterisk timahvo1 (~rogue@41.72.215.94) |
| 12:01.03 | schmidts | asterisk-learner if you do a softhangup and nothing happens then the channel is mostly deadlocked cause to hang up a channel * has to get the lock |
| 12:07.04 | asterisk-learner | schimdts: ok , thx for the info |
| 12:09.57 | joachim_- | schmidts: Had to go on a support call and lunch.. Wanna see something strange? This is the messages I get when connecting to the call manager: |
| 12:10.03 | joachim_- | INFO - Connected via Asterisk Call Manager/1.1 |
| 12:10.03 | joachim_- | INFO - Successfully logged in |
| 12:10.03 | joachim_- | INFO - Determined Asterisk version: Asterisk 1.6 |
| 12:10.49 | schmidts | joachim_- then this comes from astman, cause if try it with telnet i see this: |
| 12:10.49 | Nugget | telnet is eeeeeeevil! |
| 12:11.02 | schmidts | Connected to 127.0.0.1. |
| 12:11.02 | schmidts | Asterisk Call Manager/1.1 |
| 12:11.26 | joachim_- | Says the same her when using telnet |
| 12:11.27 | *** join/#asterisk niekniek (~niekniek@92.70.112.34) |
| 12:11.40 | niekniek | ringback on iax is working! needs playtones(ring) in dialplan |
| 12:11.42 | joachim_- | schmidts: And in this case we dont use astman |
| 12:11.46 | schmidts | joachim_- middleware is evil :D |
| 12:11.48 | *** join/#asterisk joobie (~joobie@CPE-121-214-141-146.lnse3.lon.bigpond.net.au) |
| 12:11.51 | niekniek | should not be required |
| 12:11.51 | niekniek | s |
| 12:11.59 | niekniek | should i report this as a bug? |
| 12:12.06 | *** join/#asterisk FiReSTaRT (~dlyh@unaffiliated/firestart) |
| 12:12.13 | joachim_- | schmidts: Its an applet my collegue has coded, not using astman here |
| 12:12.16 | *** join/#asterisk anonymouz666 (~anonymouz@187-28-37-118.poolip.RJO.embratel.net.br) |
| 12:12.17 | andylockran | guys, this is dead strange. Got a phone setup and registering. hits the server on udp port 5060 but get nothing in the logs.. |
| 12:13.05 | joobie | guys |
| 12:13.17 | joobie | if i ping a US site from AU |
| 12:13.19 | joobie | i got 300ms ping |
| 12:13.24 | schmidts | joachim_- this determined asterisk verison doesnt comes from asterisk itself |
| 12:13.30 | joobie | i have an asterisk box at each site nad wanted to run calls between the two |
| 12:13.32 | joobie | but 300ms sucks |
| 12:13.36 | joobie | ADSL at both sites |
| 12:13.36 | schmidts | joachim_- AFAIK |
| 12:13.46 | joobie | is the only wa to do this to pay for a premium typ elink? |
| 12:13.55 | joobie | i mean even my ping to google is 250ms |
| 12:13.56 | joobie | from AU |
| 12:14.02 | schmidts | joobie try TCP echo instead of ICMP, this is slowed down on most core routers so you can not trust this values |
| 12:14.11 | joobie | ahh nice |
| 12:14.17 | joobie | is that a linux / mac tool? |
| 12:14.24 | joachim_- | schmidts: Hmmm... Then it has to be the asterisk java server |
| 12:14.28 | schmidts | joobie hping is what you are looking for ;) |
| 12:14.38 | joobie | thanks schmidts |
| 12:14.44 | joobie | im about to play dota |
| 12:14.50 | joobie | but ill check it out after i own these punks |
| 12:15.34 | schmidts | joachim_- asterisk java? is this something of asterisk directly cause i have search for this "Determined Asterisk version" in the 1.8 source and cant find it |
| 12:16.42 | joachim_- | schmidts: http://asterisk-java.org/ |
| 12:17.16 | andylockran | sorted, need to understand udp port range.. 5060+ |
| 12:17.46 | niekniek | zoiper with iax to asterisk requires playtones(ring) in the dialplan, otherwise zoiper will not play ringback, is this a bug or a feature? |
| 12:17.57 | schmidts | andylockran sip over udp normaly use port 5060 up to 5069, but this depends on your clients |
| 12:19.21 | schmidts | joachim_- but it really looks like this determined asterisk version comes from this asterisk-java thing |
| 12:21.21 | andylockran | schmidts: I set multiple clients to all use 5060, but that seemed to fail. |
| 12:21.33 | andylockran | I switched one client phone to 5067 and it works great |
| 12:22.57 | schmidts | andylockran it depends on your network structure, differnt ports are only usefull if all of them are behind the same nat (and if this nat doesnt work well) |
| 12:23.08 | asterisk-learner | schimdts: on my mahcine with the old * i have "preload => res_musiconhold.so" in modules.conf, while on the new asterisk i have "load => res_musiconhold.so" |
| 12:23.15 | asterisk-learner | do u think that might cause any issue ? |
| 12:23.15 | schmidts | normally a peer is matched by ip + port |
| 12:24.34 | schmidts | asterisk-learner IMHO it depends on the order how modules are load, if res_moh is loaded after the queue then it maybe will work |
| 12:25.03 | asterisk-learner | ok |
| 12:28.18 | *** join/#asterisk leifmadsen (~Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 12:28.18 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 12:29.35 | schmidts | morning leif |
| 12:32.24 | leifmadsen | morn! |
| 12:38.22 | *** join/#asterisk FiReSTaRT (~dlyh@unaffiliated/firestart) |
| 12:40.44 | *** join/#asterisk gerhard7 (~gerhard7@212-123-146-122.ip.telfort.nl) |
| 12:47.37 | *** join/#asterisk sekil (~sekil@80.93.247.26) |
| 12:47.53 | *** join/#asterisk fauxalliance (~fauxallia@142.163.155.36) |
| 12:51.19 | *** join/#asterisk skrusty (~ben@93-97-20-22.zone5.bethere.co.uk) |
| 12:53.35 | *** join/#asterisk angryuser_laptop (~angryuser@90-156-167-83.reverse.alphalink.fr) |
| 12:55.28 | schmidts | you have killed the Waffles leifmadsen!!! |
| 12:55.58 | skrusty | afternoon all |
| 12:56.13 | devmod | I am inviting participants from within a conference, how can I make it so that all participants in the conference hear the ringing? |
| 12:56.35 | skrusty | i dont think you can... |
| 12:56.49 | schmidts | devmod how do you invite these participants? |
| 12:56.59 | devmod | AMI's Originate |
| 12:57.01 | skrusty | i think the channel has to be bridged first before audio is sent into the conf |
| 12:57.27 | skrusty | yeah deffo if you are using originate (not that i know of any other way) |
| 12:57.31 | schmidts | devmod you can try a callfile instead, first leg goes into conference and second dials out |
| 12:58.06 | devmod | Agents might or not bring ppl in at any point during the duration of the call |
| 12:58.24 | skrusty | why would you want to hear the rining though, out of interest? |
| 12:58.33 | skrusty | g |
| 12:58.57 | devmod | Agents want a audible feedback as well as their visual (web interface) feedback |
| 12:59.39 | skrusty | ok |
| 13:00.16 | schmidts | devmod do you originate directly to the dial app or to an extension? |
| 13:00.55 | devmod | To an application (app_konference) |
| 13:01.23 | schmidts | devmod please show me an example how you do this |
| 13:01.32 | *** join/#asterisk philfine (~philfine@56.4.166.178.rev.vodafone.pt) |
| 13:01.35 | devmod | Sure hd |
| 13:05.36 | devmod | schmidts, http://pastebin.com/W95HPCnR |
| 13:07.55 | schmidts | devmod ok i see |
| 13:08.27 | *** join/#asterisk pinoyskull (~pinoyskul@112.198.64.80) |
| 13:09.24 | *** join/#asterisk avruser (~user@94.158.60.4) |
| 13:10.01 | schmidts | devmod not so easy then ;) my idea is that you try to turn it around, using a local channel as originating target, wchich just enters the conference and an extension to dialing out to the target |
| 13:10.03 | *** join/#asterisk Olivier_54 (~Olivier_5@vaillant.famille-fontes.net) |
| 13:11.18 | devmod | schmidts, ohhh ok I see.. I had something similar before but the pain was the originate1 had diff uniqueid from originate2/dial so keeping track from the beginning was a little messy |
| 13:12.08 | schmidts | devmod why not using extensions logic for this, than doing it via ami? |
| 13:13.59 | devmod | What I described is just a little piece of a much bigger functionality, so I need to have external control over what is going on |
| 13:14.21 | devmod | Can I redirect a channel with a state that is not Up? |
| 13:14.27 | avruser | Hello, people! Does exist a method to install h323 (not ooh323) package on redhat (centos) distributions without compiling source every time updated version? Thanks in advance. |
| 13:14.47 | schmidts | devmod imho you can redirect a channel only if its not in down state ;) |
| 13:18.07 | devmod | hehe |
| 13:21.42 | schmidts | devmod this works, i am just testing it to get the ringtone into the conference ;) |
| 13:22.49 | devmod | does it work? I was about to test it |
| 13:23.02 | schmidts | devmod not with a ringtone but with moh ;) |
| 13:23.14 | devmod | ohhh |
| 13:24.24 | schmidts | ringtone doesnt work, IMHO this couldnt work cause the ring tone generator will not work for meetme but moh would be better than nothing ;) |
| 13:25.25 | schmidts | http://pastebin.com/grXuCzwn |
| 13:25.29 | schmidts | thats how i have done this |
| 13:26.07 | schmidts | you can also use the app dial for the originate command directly but it changes nothing on the ringtone :( |
| 13:26.28 | *** join/#asterisk Andrew_Majosi (62712e9c@gateway/web/freenode/ip.98.113.46.156) |
| 13:27.04 | schmidts | i use the wait to make sure the other side of the call leg enters the conference before dialing out |
| 13:29.51 | devmod | schmidts, isnt there an option to generate a ringing to the calling party or something |
| 13:30.42 | schmidts | devmod the "r" option for dial, but it doesnt work |
| 13:30.45 | schmidts | i have tried it |
| 13:30.54 | devmod | schmidts, ohhh ok :/ |
| 13:31.08 | devmod | Maybe just play a ringing.wav to the channel lol |
| 13:34.12 | schmidts | devmod i dont want to say it but why not :D |
| 13:38.02 | Tozz_ | does anyone know if X-Lite softphones can be 'provisioned' ? We use provisioning to change the textlabel on Cisco phones, and would like to do the same with X-Lite |
| 13:38.26 | *** join/#asterisk tuxxie (~Ryan@rrcs-70-63-90-226.midsouth.biz.rr.com) |
| 13:39.08 | tuxx- | afaik, no. |
| 13:39.21 | Tozz_ | k, thnx tuxie ;) |
| 13:39.26 | tuxx- | =] |
| 13:41.10 | schmidts | Tozz_ maybe you can find where xlite saves the userdata on disk and use the os provisoning :D |
| 13:45.41 | *** join/#asterisk rrb3942 (~rbullock@208.34.105.161) |
| 13:51.21 | Tozz_ | hehe, yeah I could try that ;) |
| 13:51.25 | Tozz_ | and reboot windows on any change |
| 13:51.31 | Tozz_ | sounds like a great idea ;) |
| 13:51.31 | *** join/#asterisk Cain (~Geek@unaffiliated/cain) |
| 13:52.31 | devmod | what would be the implications of using progressinband=yes on channels that do render ringing ? |
| 13:54.05 | joachim_- | I wanna drown my Linksys SPA 942 in the snow! :P |
| 13:54.09 | schmidts | tozz_ thats why you should use a hard phone ;) |
| 13:54.15 | schmidts | joachim_- LOL :D |
| 13:55.07 | schmidts | devmod AFAIK progressinband has to do with dtmf not with ringing |
| 13:55.35 | Tozz_ | schmidts: Yes, well. Customer works from home, doesnt have a router, doesnt want to lay extra cables, etc, etc |
| 13:55.43 | Tozz_ | and she likes X-Lite alot |
| 13:55.49 | Tozz_ | I dont, but ah well ;) |
| 15:51.21 | *** join/#asterisk infobot (ibot@rikers.org) |
| 15:51.21 | *** topic/#asterisk is Asterisk: The Open Source PBX and Telephony Platform (asterisk.org) -=- Asterisk 1.8.2.3 (2010/01/26), 1.6.2.16.1 (2010/01/18), 1.4.39.1 (2010/01/18), *-Addons 1.6.2.3, 1.4.13 (2010/01/14), dahdi-linux 2.4.0 (2010/09/01), dahdi-tools 2.4.0 (2010/09/01), libpri 1.4.11.5 (2010/11/22) -=- Visit the new official Asterisk wiki: wiki.asterisk.org -=- http://xkcd.com/806/ |
| 15:51.58 | *** join/#asterisk awclin (~alinford@80.169.133.251) |
| 15:54.40 | ketas | i love ipv6, every client and every service can have it's own dedicated ip, no need to split the ip anymore like name-based vhosting does |
| 15:55.00 | leifmadsen | although people will still put it behind NAT to obfuscate network topology |
| 15:55.55 | Kobaz | but the nat wouldn't be required |
| 15:56.16 | leifmadsen | aye |
| 15:56.28 | Kobaz | although isps would probably only give you one ipv6 address for your home connection |
| 15:56.30 | ketas | to obfuscate topology, yes, maybe |
| 15:56.39 | ketas | why one? |
| 15:56.44 | Kobaz | $$ |
| 15:56.47 | leifmadsen | because ISPs are stupid :) |
| 15:56.52 | Kobaz | yeah, exactly |
| 15:56.57 | ketas | a cpe would have /64 |
| 15:57.01 | Kobaz | pay up, for a /32 |
| 15:57.09 | mzb | no, because they can make $ from it |
| 15:57.14 | leifmadsen | I have to get rid of this ISP home router with built in everything -- it's awful |
| 15:57.16 | Kobaz | er, right /64 |
| 15:57.36 | ketas | i have a pc router at home |
| 15:57.45 | Kobaz | i <3 my linux router |
| 15:57.47 | mzb | #openwrt |
| 15:57.55 | ketas | i run fbsd there |
| 15:58.09 | Kobaz | any cheap boxes that come preinstalled with openwrt? |
| 15:58.16 | ketas | adsl2+ device runs as bridge only |
| 15:58.32 | mzb | nothing with adsl in it |
| 15:58.37 | ketas | this is what it can do best |
| 15:58.39 | mzb | but have a look at: |
| 15:58.55 | mzb | wndr3700, dir814, tp??1034 |
| 15:59.05 | mzb | (off the top of my head) |
| 15:59.14 | mzb | ask in #openwrt |
| 15:59.28 | *** join/#asterisk Scorpio2007 (~Scorpio20@jose-tc.ctc.biz) |
| 15:59.37 | ketas | i'm sure my isp gives me large network when they start offering v6 |
| 16:00.41 | Scorpio2007 | I ring a phone # to a group of extension .. but sometimes i get this error message though all phones rings . .what does it mean? -- Incoming call: Got SIP response 500 "CSeq Number Out of order" back from XXX.XXX.XXX.XXX |
| 16:01.00 | *** join/#asterisk Tech_Travis (~Travis_El@cpe-76-168-191-127.socal.res.rr.com) |
| 16:01.46 | ketas | i really don't know what the isp's are smoking in other countries |
| 16:03.18 | *** join/#asterisk The_Boy_Wonder (~vossel@asterisk/batman-developer/dvossel) |
| 16:03.35 | ketas | one ipv6 per connection makes it all useless |
| 16:03.57 | *** part/#asterisk mickecarlsson (~Micke@h238n1c1o1101.bredband.skanova.com) |
| 16:04.16 | TSM | when using playback and a extension is keyed in how does it jump around to dial the extension, is this a built in action that cant be overriden? |
| 16:05.15 | *** part/#asterisk awclin (~alinford@80.169.133.251) |
| 16:05.16 | *** join/#asterisk drift- (ae3001a4@gateway/web/freenode/ip.174.48.1.164) |
| 16:05.54 | *** part/#asterisk tomsh (~tomsh@host3-85-static.89-82-b.business.telecomitalia.it) |
| 16:06.20 | *** join/#asterisk rgsteele (~rgsteele@apo.aweber.com) |
| 16:08.56 | *** join/#asterisk [Outcast] (~anonymous@64.202.62.5) |
| 16:10.39 | *** join/#asterisk aidinb (~Aidin@unaffiliated/aidinb) |
| 16:17.01 | leifmadsen | TSM: using Playback() doesn't do anything when you enter DTMF |
| 16:17.18 | leifmadsen | TSM: Only Playback(), Read(), etc... act like that |
| 16:19.54 | *** join/#asterisk JamesHarrison (~jharr@hometree.mmmetrics.co.uk) |
| 16:24.00 | nestAr | <PROTECTED> |
| 16:24.03 | nestAr | oops |
| 16:25.23 | TSM | leifmadsen: i dont have any Read() in my script, Playback then while message is playing i press dial a ext number and it starts calling that phone |
| 16:28.49 | *** join/#asterisk ChannelZ (channelz@burner.com) |
| 16:29.04 | nestAr | anyone know what the alsa module depends on? it doesn't seem to love me. |
| 16:29.33 | leifmadsen | TSM: that makes no sense because Playback() doesn't listen for DTMF |
| 16:31.10 | *** join/#asterisk Maliuta (~nobusines@kiev.lusan.id.au) |
| 16:34.40 | *** join/#asterisk tamiel (~tamiel@213.30.183.226) |
| 16:34.40 | *** join/#asterisk codebanshee (~chris@81.171.245.107) |
| 16:34.44 | TSM | weird |
| 16:40.46 | *** join/#asterisk RypPn (~TuMbL@unaffiliated/ryppn) |
| 16:42.34 | *** join/#asterisk GameGamer43 (~GameGamer@li119-60.members.linode.com) |
| 16:44.50 | *** join/#asterisk blee__ (~blee@99-117-188-231.lightspeed.dybhfl.sbcglobal.net) |
| 16:44.56 | sbrath | Any sugestions on eliminating "hum" on a TDM400 series card.. Sounds like computer noise on the Phone Line? |
| 16:46.55 | *** join/#asterisk LemensTS (~matthew@adsl-70-238-156-107.dsl.stlsmo.sbcglobal.net) |
| 16:47.10 | LemensTS | What Asterisk DECT system do you prefer? |
| 16:47.16 | TSM | sbrath: noisy psu, incorrectly wired lines, etc |
| 16:52.38 | sbrath | well, I re-pulled the line from the server to the station and ran Cat5e, and avoided any power cables. But even removing the cords, and just hooking a pots phone to the back of the server I get noise. PS. The server is a PhoneBochs Enterprise box, so it's huge, and has dual PS but the PS are on the complete other side of the case. |
| 17:01.53 | ketas | he maybe meant noise via wires |
| 17:01.54 | ketas | from psu |
| 17:02.41 | ketas | but that's probably not hum |
| 17:04.25 | sbrath | Maybe it's more of a Buzzing.... Sometimes it's there , sometimes not. |
| 17:07.02 | Katty | heeeeellllllllllllooooooooooooooooo nurse! |
| 17:08.02 | nestAr | libasound2-dev is what i was missing to get chan_alsa compiled. |
| 17:09.10 | carrar | Who's the nurse? |
| 17:15.55 | nestAr | lol @ hello nurse |
| 17:18.24 | drmessano | You have found 1 Nurse. You earn 10 gold and 3 xp. |
| 17:20.49 | drmessano | What was the unlock code again? |
| 17:21.12 | p3nguin | 18436572 |
| 17:21.29 | p3nguin | No, wait... |
| 17:21.30 | p3nguin | That's the firing order of a small block Chevy. |
| 17:21.40 | p3nguin | My mistake. |
| 17:22.35 | drmessano | Up, Up, Down, Down, Left, Right, Left, Right, B, A, Start |
| 17:22.52 | p3nguin | ;) |
| 17:23.07 | p3nguin | iddqd |
| 17:23.14 | drmessano | You earn 1e00000000 gold and 1e00000000 xp. |
| 17:23.18 | drmessano | YAY |
| 17:23.42 | drmessano | I got a free license for Asterisk too |
| 17:24.21 | drmessano | Better than the one I got off eBay |
| 17:24.49 | drmessano | LQQK!!! ASTERISK PBX 100% CRACKED UNLOCKED AND KEYLESS LQQK~ |
| 17:25.03 | drmessano | :( |
| 17:30.02 | *** join/#asterisk manji (~manjiki@ppp-94-65-208-18.home.otenet.gr) |
| 17:37.48 | nestAr | ok, i'm stumped on this alsa thing. i got the module compiled. and it's loaded into asterisk, but i'm getting "alsa_card_init: snd_pcm_open failed: No such file or directory" |
| 17:37.52 | nestAr | from asterisk |
| 17:38.13 | nestAr | i can play an mp3 via mpg321 though |
| 17:40.30 | *** join/#asterisk jaskew (~jdaskew@netblock-75-79-182-110.dslextreme.com) |
| 17:53.31 | *** part/#asterisk LemensTS (~matthew@adsl-70-238-156-107.dsl.stlsmo.sbcglobal.net) |
| 17:56.14 | *** join/#asterisk angryuser_laptop (~angryuser@90-156-167-83.reverse.alphalink.fr) |
| 17:56.17 | luckman212 | arggg I am wrestling with madplay |
| 17:56.59 | luckman212 | mpg123 goes zombie on my eating 99% cpu. so I switched to madplay but the sound is kind of lousy and I get gaps of silence every few seconds |
| 17:57.58 | luckman212 | i tried piping wget into buffer first... e.g wget http://mystream:port | buffer | /usr/local/bin/madplay |
| 17:58.38 | luckman212 | how about mplayer? anyone using that for streaming moh |
| 18:02.18 | *** join/#asterisk IsUp (IsUp@unaffiliated/isup) |
| 18:02.43 | IsUp | hello |
| 18:03.20 | *** join/#asterisk omani (~hasan@33.37.69.80.in-addr.net-lab.net) |
| 18:03.37 | omani | is the registration of sips on ther * server in cleartext? |
| 18:03.52 | omani | cant find any entries in ngrep output on server while registration of sip |
| 18:03.57 | IsUp | I have a context for outgoing cals: http://pastebin.com/yULWgsQq - i want to record outgoing calls, should i use Monitor or MixMonitor? |
| 18:04.09 | omani | would be nice to know whether the registration is encrypted or not. |
| 18:05.31 | p3nguin | I prefer MixMonitor. |
| 18:05.40 | *** join/#asterisk drmessano-lt (~nonya@pdpc/supporter/active/drmessano) |
| 18:07.40 | *** join/#asterisk alphawave (~aw@unaffiliated/alphawave) |
| 18:08.41 | IsUp | p3nguin: how can i use it? |
| 18:09.14 | p3nguin | MixMonitor() in the dial plan will do it. "core show application MixMonitor" will show you the details. |
| 18:13.22 | IsUp | p3nguin: how can i get date like d-m-Y H:i:s with STRFTIME? |
| 18:13.23 | *** join/#asterisk rrb3942 (~rbullock@208.34.105.161) |
| 18:16.19 | p3nguin | isup: Something such as: MixMonitor(${STRFTIME(${EPOCH},,%d-%m-%Y %H:%M:%S)}.WAV,a) |
| 18:16.26 | IsUp | p3nguin: ${STRFTIME(${EPOCH},GMT+2,%C%y-%m-%d-%H-%M)} |
| 18:17.04 | p3nguin | I personally like to use the year, month, day, and callerid number: MixMonitor(${STRFTIME(${EPOCH},,%Y%m%d)}-${CALLERID(num)}.WAV,a) |
| 18:17.21 | IsUp | i did MixMonitor(${EXTEN}_${STRFTIME(${EPOCH},,%d-%m-%Y %H:%M:%S)}.WAV,a) |
| 18:17.48 | p3nguin | Since you are using the seconds in the file name, you may want to forget the ,a there on the end. |
| 18:18.41 | p3nguin | Also, that will leave a space in the file name, so take that into consideration. |
| 18:19.09 | IsUp | MixMonitor(${EXTEN}_${STRFTIME(${EPOCH},,%d-%m-%Y_%H_%M_%S)}.WAV) |
| 18:19.23 | IsUp | file name -> 58363_04-02-2011_201840.WAV |
| 18:19.33 | IsUp | it seems ok |
| 18:20.03 | IsUp | replaced spaces with _ and - |
| 18:22.09 | IsUp | p3nguin: works great, thank you |
| 18:22.28 | p3nguin | Cool. |
| 18:57.05 | *** join/#asterisk lanning (~lanning@208.87.233.137) |
| 19:08.31 | *** join/#asterisk n3hxs (~ed@63.68.135.4) |
| 19:09.51 | *** part/#asterisk omani (~hasan@33.37.69.80.in-addr.net-lab.net) |
| 19:13.25 | n1x0n | hmm I'm running out of ideas - someone is somhow using my asterisk to callout , it happened like 4 times now - last time I did a full reinstall using debian original packages , reset all passwords to something stupidly long etc.. and bang - same thing today - any clues where should I look for a problem ? |
| 19:14.21 | *** join/#asterisk vinhdizzo (~vinh@dhcp-v000-157.mobile.uci.edu) |
| 19:16.41 | *** join/#asterisk raden_work (~jon@69-179-99-17.stat.centurytel.net) |
| 19:17.09 | raden_work | when we transfer a call is there a way to play a beep or something on the recieve end before the call is connected ? its hard to tell sometimes if someone is there |
| 19:21.34 | *** join/#asterisk doolittlework (~doolittle@41-134-22-10.dsl.mweb.co.za) |
| 19:21.39 | doolittlework | evening |
| 19:21.55 | pabelanger | raden_work: Yes, do an attended transfer |
| 19:22.05 | raden_work | ???? |
| 19:22.11 | doolittlework | does anyone know if we can search a network by mac address |
| 19:22.36 | raden_work | what im saying is if we hit transfer and then speak to the person and hit transfer sometimes people dont realize it already transfered |
| 19:22.45 | raden_work | doolittlework, yes you can |
| 19:22.58 | doolittlework | tell you why i have a patton gateway anf can not remember the ip address, yes i can flush and reset it but i want me settings of the device |
| 19:23.23 | doolittlework | the mac address is on the device how can i determine the ip address? |
| 19:23.45 | pabelanger | raden_work: how do you 'hit transfer' ? The phone or are you using asterisk? |
| 19:24.10 | raden_work | phone |
| 19:24.27 | pabelanger | then no, Asterisk is not in the picture |
| 19:24.34 | doolittlework | raden_work: do you know of a app i can use to search the network to find the ip address? |
| 19:24.36 | raden_work | Shit : ( |
| 19:24.41 | n1x0n | doolittlework: arp -na (if mac / ip is in the arp), if not - arping |
| 19:24.57 | doolittlework | ta thx |
| 19:25.46 | doolittlework | n1x0n: would this wotk if the device is on a different network address range? |
| 19:26.08 | doolittlework | excuse my spelling |
| 19:26.38 | n1x0n | doolittlework: if there is a router between them - no |
| 19:26.58 | doolittlework | n1x0n: thank so much |
| 19:27.35 | n1x0n | np, now tell me how some assh**** are using my tiny asterisk setup and waste my money ;-) |
| 19:28.24 | doolittlework | are you talking to me n1x0n? |
| 19:29.11 | p3nguin | I can tell you how and I just got here. You didn't set it up properly. |
| 19:29.12 | n1x0n | doolittlework: nah, need some help in figuring out how some *guys* are using my asterisk gw to callout :-/ I've reset all passwords and stuff - and they just did it again - running out of ideas |
| 19:29.57 | p3nguin | You probably allow anonymous calls to come in and you probably allow outbound calling from the context where the anonymous calls come in. Common mistake. |
| 19:30.03 | doolittlework | the basterds |
| 19:30.11 | n1x0n | p3nguin: yeah I think I narrowed it down to that already by reinstalling asterisk and resetting all passwords |
| 19:30.12 | Qwell | common mistake outlined in SECURITY.txt, which you read, I hope |
| 19:30.14 | n1x0n | aha ! |
| 19:30.27 | p3nguin | In your sip.conf, look under the [general] section. Tell me what the context= line says. |
| 19:30.36 | Qwell | $20 is on default |
| 19:30.41 | p3nguin | :D |
| 19:30.44 | doolittlework | set your sip allow=xxx.xxx.xxx.xxx and deny=XXX.XXX.XXX.XXX |
| 19:30.58 | n1x0n | checks |
| 19:31.02 | p3nguin | Then I'm going to ask to see that context from extensions.conf to see what it does. |
| 19:31.21 | Qwell | $20 says there is _1NXXNXXXXXX,1,Dial(SIP/...) |
| 19:31.27 | n1x0n | context=default |
| 19:31.30 | n1x0n | yes |
| 19:31.30 | Qwell | p3nguin: that's $40 if you're keeping track |
| 19:31.31 | n1x0n | there is |
| 19:31.42 | doolittlework | lol |
| 19:31.42 | Qwell | yeah, don't do that. |
| 19:31.46 | p3nguin | There's yer problem. |
| 19:31.46 | n1x0n | and then I have exten => _00.,1,Dial(SIP/${EXTEN:0}@internetcalls,30,r) |
| 19:31.52 | n1x0n | right |
| 19:31.55 | p3nguin | qwell: Is there anything wrong with "default" if [default] is empty? |
| 19:31.59 | n1x0n | shame on me |
| 19:32.03 | Qwell | p3nguin: no |
| 19:32.52 | p3nguin | I know I have a misc-calls context which is technically a dead end, but I think I still have default as the general context. It's empty, regardless. |
| 19:33.24 | p3nguin | I should probably get around to changing it to my misc-calls dead end route one of these days just so I can record what is going on. |
| 19:33.40 | n1x0n | so I have default then in extensions.conf I just havfe extensions setup and the Dial line I pasted - I understand from Qwell's previous comment that this is covered in that .txt |
| 19:33.50 | n1x0n | s/havfe/have/g |
| 19:34.19 | p3nguin | #1 - don't put useful stuff in the default context. Leave it blank until you KNOW what you are doing. |
| 19:34.41 | p3nguin | #2 - consider NOT allowing anonymous calls inbound. |
| 19:35.16 | p3nguin | #3 - consider an ACL for every SIP device where an ACL is possible and practical. |
| 19:35.35 | p3nguin | Some dynamic hosts just don't work ou with ACLs. |
| 19:36.03 | n1x0n | ok I don't understand that consepct of context here *but* I'm not going to waste your time anymore - very useful and helpful - thanks a lot ! |
| 19:36.13 | n1x0n | hides and goes to do some rtfm |
| 19:36.22 | p3nguin | It's in The Book. |
| 19:36.25 | n1x0n | p3nguin: Qwell thx |
| 19:36.26 | p3nguin | ~newbook |
| 19:36.26 | infobot | Help review the 3rd edition of the Asterisk book published by O'Reilly at http://ofps.oreilly.com/titles/9780596517342/ Released under a Creative Commons license: http://creativecommons.org/licenses/by-nc-nd/3.0/us/ |
| 19:36.40 | Qwell | also README-SERIOUSLY.bestpractices.txt |
| 19:38.12 | p3nguin | So... during the wonderful ice storms I blew out an UPS battery. There's $179.99 I could have used on something else. |
| 19:40.59 | *** join/#asterisk drmessano-lt (~nonya@pdpc/supporter/active/drmessano) |
| 19:43.53 | *** join/#asterisk luckman212 (~quassel@pool-173-77-253-145.nycmny.fios.verizon.net) |
| 19:45.06 | n3hxs | When we replace UPS batteries, we do all 20 at one time. |
| 19:45.28 | n3hxs | Lucky it only happens every 3 to 5 years. |
| 19:50.23 | doolittlework | n1x0n: can figure out the arp command :arp -Na MAC ADDRESS |
| 19:50.54 | doolittlework | nudge me in the right direction or push me in front of a bus |
| 19:50.59 | doolittlework | please |
| 19:51.45 | *** join/#asterisk Micc (~quassel@c-24-18-20-54.hsd1.wa.comcast.net) |
| 19:52.15 | Micc | Anyone know of any good office routers that have dual WAN and can failover if one goes down? |
| 19:53.21 | Micc | I suspect it would drop the call still if that happened. But at least they would reregister with the new ip and could make calls right away. |
| 19:56.29 | *** join/#asterisk Olivier_54 (~Olivier_5@vaillant.famille-fontes.net) |
| 19:59.38 | *** join/#asterisk luckman212_ (~quassel@pool-173-77-253-145.nycmny.fios.verizon.net) |
| 20:13.06 | *** join/#asterisk x1nux (~x1nux@unaffiliated/x1nux) |
| 20:13.08 | x1nux | hi |
| 20:15.21 | x1nux | i have problem with asterisk; the problem is: "i have a virtual machine OpenVZ ; i installed asterisk but i want a web manager applications for asterisk" FreePBX reload and reload asterisk and No Work; Asterisk GUI, said Page No found ! so i dont knos what i do.... can you Helpme ??? excuseme my english i from colombia and try to make me Understand !! thks |
| 20:18.39 | wdoekes2 | x1nux: if you have problems with freepbx, please go to #freepbx |
| 20:19.01 | x1nux | auch ! |
| 20:19.02 | x1nux | Ok |
| 20:19.04 | x1nux | thkks |
| 20:20.08 | *** join/#asterisk nix8n82 (~nate@63.162.28.112) |
| 20:20.11 | x1nux | and about asterisk gui ??? |
| 20:20.25 | x1nux | asterisk-gui is a add for asterisk no ? |
| 20:21.21 | wdoekes2 | ok, that might be in here.. although I've never heard anyone about that |
| 20:21.22 | nestAr | odd, /dev/dsp crw-rw---- 1 root audio 14, 3 Feb 4 15:04 dsp |
| 20:21.25 | Kobaz | gooey |
| 20:21.28 | nestAr | asterisk is in the audio group |
| 20:21.37 | nestAr | yet it gets a permission denied.. |
| 20:21.45 | nestAr | works find if i chmod 666 dsp |
| 20:22.05 | wdoekes2 | did you restart asterisk after putting it in the audio group? |
| 20:22.22 | nestAr | yeah, i just rebooted to make sure everything worked.. |
| 20:22.46 | nestAr | and i had had it 666'd, it reverted back to 660 |
| 20:22.52 | nestAr | asterisk mad.. :) |
| 20:23.03 | wdoekes2 | are you sure it's running as the asterisk user? |
| 20:23.22 | nestAr | i just noticed that asterisk doesn't seem to have a primary group |
| 20:23.28 | nestAr | i wonder if that has anything to do with it. |
| 20:23.29 | nestAr | mecca:/etc/asterisk# id asterisk |
| 20:23.29 | nestAr | uid=104(asterisk) gid=65534(nogroup) groups=65534(nogroup),29(audio) |
| 20:23.48 | nestAr | mecca:/etc/asterisk# ps aux | grep asterisk |
| 20:23.48 | nestAr | asterisk 1917 0.0 6.4 132904 116440 ? Ssl 15:05 0:00 /usr/sbin/asterisk -U asterisk -G asterisk |
| 20:23.54 | nestAr | i should go fix that now. |
| 20:24.06 | wdoekes2 | strange |
| 20:24.58 | wdoekes2 | su asterisk -c 'yes | /dev/dsp' |
| 20:26.52 | nestAr | that seems to work.. |
| 20:27.03 | nestAr | wth |
| 20:27.48 | *** join/#asterisk b14ck (~b14ck@cpe-72-129-70-245.socal.res.rr.com) |
| 20:28.15 | nestAr | something is not right there.. my user is in the audio group |
| 20:28.24 | nestAr | i can't access dsp either |
| 20:29.20 | wdoekes2 | perhaps something is locking it? |
| 20:29.44 | nestAr | perhaps, but if it was locked, would 666 help? |
| 20:29.48 | wdoekes2 | no |
| 20:29.49 | nestAr | i wouldn't think so |
| 20:29.51 | nestAr | yeah |
| 20:29.57 | wdoekes2 | strange :) |
| 20:30.52 | wdoekes2 | wait.. that should be > /dev/dsp, not | |
| 20:31.04 | wdoekes2 | my bad :) |
| 20:34.35 | *** join/#asterisk werdan7 (~w7@freenode/staff/wikimedia.werdan7) |
| 20:34.43 | *** join/#asterisk drmessano-lt (~nonya@pdpc/supporter/active/drmessano) |
| 20:35.25 | x1nux | no have help |
| 20:35.27 | x1nux | :s |
| 20:38.18 | wdoekes2 | x1nux: users of this channel assume you're able to install asterisk, edit the config files in /etc/asterisk and run commands from the asterisk console (asterisk -r). feel free to use gui's, but you should probably be looking elsewhere for help with those |
| 20:39.17 | nestAr | i wouldn't run the yes command as asterisk until i set the account with a real shell (not /bin/false) |
| 20:39.27 | nestAr | it worked fine as me. |
| 20:39.34 | nestAr | i set the shell to /bin/bash for asterisk |
| 20:39.53 | nestAr | and it works, but yet the asterisk process can't get to dev/dep until i 666 it. |
| 20:40.00 | nestAr | the mind boggles. |
| 20:40.40 | wdoekes2 | and if you change the primary group to audio> |
| 20:41.02 | wdoekes2 | grasping for straws here |
| 20:42.09 | wdoekes2 | (you don't need a real shell btw, but you would've needed -s /bin/sh as su option) |
| 20:43.32 | n1x0n | doolittlework: arp -na will show you all macs in your arp table. so if you have zilions of these you can do arp -na | grep mac ... |
| 20:45.51 | *** part/#asterisk tuxxie (~Ryan@rrcs-70-63-90-226.midsouth.biz.rr.com) |
| 20:46.50 | drift- | can you have more then 1 digium g729 card |
| 20:46.52 | drift- | work in same server? |
| 20:47.00 | *** join/#asterisk werdan7 (~w7@freenode/staff/wikimedia.werdan7) |
| 20:48.05 | Qwell | drift-: I would think so. Probably aught to call sales or support though (not really sure which..) |
| 20:50.53 | wdoekes2 | nestAr: to trace the open(2) calls: strace -eopen -fp`pidof asterisk` |
| 20:51.25 | *** join/#asterisk TimeRider (~steve@188-220-33-123.zone11.bethere.co.uk) |
| 20:53.51 | nestAr | [pid 2193] open("/dev/dsp", O_RDWR|O_NONBLOCK|O_LARGEFILE) = -1 EACCES (Permission denied) |
| 20:54.03 | nestAr | i haven't tried to switch the main group to audio |
| 20:57.09 | nestAr | http://pastebin.com/wS0hv4n1 |
| 21:04.14 | *** join/#asterisk NightMonkey (debian-tor@pdpc/supporter/professional/nightmonkey) |
| 21:06.04 | *** join/#asterisk snarvaez (~seba@190.179.18.222) |
| 21:06.05 | *** join/#asterisk lost_soul (shawn@cpe-74-78-191-114.twcny.res.rr.com) |
| 21:06.08 | *** join/#asterisk aidinb (~Aidin@unaffiliated/aidinb) |
| 21:13.48 | *** join/#asterisk philfine (~philfine@a213-22-45-215.cpe.netcabo.pt) |
| 21:14.50 | *** join/#asterisk CaseSensitive (~pop@bzq-79-179-30-89.red.bezeqint.net) |
| 21:15.19 | CaseSensitive | Wow. didnt expect so many people on the asterisk channel |
| 21:17.35 | NightMonkey | CaseSensitive: We knew you were coming. |
| 21:22.07 | CaseSensitive | last time ive been on an IRC channel was 10 years ago. |
| 21:22.21 | CaseSensitive | so now its also an IRC channel, and also asterisk related |
| 21:22.50 | CaseSensitive | thats it, i finished being amazed. anybody know of a source for agi scripts ? |
| 21:25.49 | *** join/#asterisk TimeRider (~steve@5ac318b6.bb.sky.com) |
| 21:30.58 | *** join/#asterisk lost_soul (shawn@cpe-74-78-191-114.twcny.res.rr.com) |
| 21:39.18 | *** join/#asterisk IsUp (IsUp@unaffiliated/isup) |
| 22:10.46 | matt4d617474 | hi all... trying my hand at asterisk |
| 22:11.04 | matt4d617474 | i've got a weird situation with ekiga |
| 22:11.13 | matt4d617474 | softphone client |
| 22:11.18 | IsUp | whats up |
| 22:11.24 | matt4d617474 | i'm trying to use sip |
| 22:11.29 | matt4d617474 | it registers successfully |
| 22:11.33 | matt4d617474 | but when i place a call |
| 22:11.54 | matt4d617474 | i hear nothing and don't see the usual verbose logs scroll by on the asterisk console |
| 22:12.11 | matt4d617474 | just "Using SIP RTP CoS mark 5" |
| 22:12.36 | matt4d617474 | so... does that mean it's not happy in the rtp communication? |
| 22:13.26 | matt4d617474 | anybody willing to point me in the right direction? |
| 22:13.27 | matt4d617474 | tia |
| 22:14.36 | IsUp | matt4d617474: this output normal and "happy meal" |
| 22:14.55 | IsUp | type 'core set verbose 10' on your Asterisk console |
| 22:14.59 | IsUp | and place a call again |
| 22:15.31 | matt4d617474 | same results |
| 22:15.44 | matt4d617474 | so it seems like it just goes into the ether |
| 22:17.06 | matt4d617474 | whereas when i connect via x-lite i get a follow-up log: Executing [42@users:1] Answer("SIP/chris-00000002", "") in new stack |
| 22:17.22 | matt4d617474 | that never happens with ekiga |
| 22:17.27 | matt4d617474 | i'll keep digging |
| 22:17.33 | *** join/#asterisk CaseSensitive (~pop@IGLD-84-229-204-177.inter.net.il) |
| 22:17.40 | CaseSensitive | mmm. |
| 22:17.55 | IsUp | matt4d617474: ive never used Ekiga before, but if it works with x-lite, probably something wrong with your Ekiga settings |
| 22:19.12 | matt4d617474 | right... i didn't know if other folks used that client or not |
| 22:19.36 | matt4d617474 | or if there are more highly recommended softphone recommendations for a linux workstation |
| 22:24.17 | *** join/#asterisk philfine (~philfine@a213-22-45-215.cpe.netcabo.pt) |
| 22:32.29 | Kobaz | question on deadlock avoidance |
| 22:32.33 | Kobaz | er wrong window |
| 23:32.37 | *** join/#asterisk infobot (ibot@rikers.org) |
| 23:32.37 | *** topic/#asterisk is Asterisk: The Open Source PBX and Telephony Platform (asterisk.org) -=- Asterisk 1.8.2.3 (2010/01/26), 1.6.2.16.1 (2010/01/18), 1.4.39.1 (2010/01/18), *-Addons 1.6.2.3, 1.4.13 (2010/01/14), dahdi-linux 2.4.0 (2010/09/01), dahdi-tools 2.4.0 (2010/09/01), libpri 1.4.11.5 (2010/11/22) -=- Visit the new official Asterisk wiki: wiki.asterisk.org -=- http://xkcd.com/806/ |
| 23:32.55 | whtsup | if u doo tht my kernel devel and kernel header will be different ? |
| 23:33.13 | erinspice | not if you're running 2.6.28.7 as you said. |
| 23:33.25 | mzb | dahdi thinks you DON'T have the right headers ... get them! |
| 23:33.27 | whtsup | but i m not running this one |
| 23:33.42 | erinspice | the headers wont' affect your kernel. IT isn't using them. |
| 23:35.29 | mzb | if redhat package management dependencies actually worked you *should* just be able to install "kernel headers" (as a generic/metapackage) and have it automatically update the kernel headers when the kernel is updated |
| 23:35.31 | whtsup | shall i remove my old kernel header 1st ? |
| 23:35.45 | mzb | but then I live in a Debianised world, so what would I know? ;)) |
| 23:35.53 | mzb | NO |
| 23:35.59 | mzb | don't bother |
| 23:36.12 | mzb | unless you've done some *tinkering* |
| 23:36.23 | whtsup | ah |
| 23:36.30 | whtsup | really confusing |
| 23:36.36 | mzb | ie: broken it |
| 23:37.16 | mzb | <whtsup> kernel-headers.i386 2.6.18-194.32.1.el5 installed |
| 23:37.25 | whtsup | yah |
| 23:37.34 | whtsup | this one is installed |
| 23:37.34 | mzb | ^^ obviously not the same as the kernel you say you're running |
| 23:38.01 | whtsup | soo wht to do |
| 23:38.02 | mzb | have you just tried doing: |
| 23:38.17 | mzb | yum install kernel-headers |
| 23:38.18 | mzb | ? |
| 23:38.22 | mzb | and see if it updates? |
| 23:38.34 | whtsup | no i have just updated to this kernel |
| 23:38.37 | whtsup | right now |
| 23:38.44 | mzb | doh! |
| 23:38.58 | mzb | no ... yesterday ALREADY! ;) |
| 23:39.08 | mzb | bbl |
| 23:39.38 | whtsup | u tell me |
| 23:39.41 | whtsup | wht shud i do |
| 23:49.30 | mzb | I just have |
| 23:49.54 | mzb | if that doesn't work I wouldn't have a clue |
| 23:50.08 | whtsup | hm |
| 23:50.10 | whtsup | listen |
| 23:50.13 | whtsup | uname -r |
| 23:50.36 | whtsup | 2.6.28.7 |
| 23:50.43 | whtsup | showing this :s |
| 23:50.49 | whtsup | but i have updated the kernel |
| 23:51.12 | whtsup | yum list kernel it is showing some different kernel installed |
| 23:51.28 | mzb | you're not telling me anything new ... sorry, I can't help you any further as I have to go out |