00:00.08 | obsidian-studios | jskcr: in ~x86 tree? |
00:00.19 | marc324 | how to delete full lines in vi? |
00:00.26 | Ariel_ | dd |
00:00.27 | Mother_ | AAAARGH |
00:00.28 | Icemaann | Ariel_: I can use pastebin if you want. Basically, i have extensions 1000, and 2000 in home. they dial VoiceMailMain. The context line for the sip user is home. When he dials 1000 it doesnt work. Ill figure it out. Thanks! |
00:01.01 | Ariel_ | marc324, use nano instead |
00:01.18 | hardwire | marc324: you are gonna cry when I say this |
00:01.21 | Ariel_ | Mother_, are you ok, do you need some water? |
00:01.25 | hardwire | but learn how to run a linux workstation first |
00:01.33 | hardwire | just do it |
00:01.38 | Mother_ | Ariel_: thanks, I just fell off the chair, I'm OK now... |
00:02.06 | *** join/#asterisk joat (n=joat@laketaylor.org) |
00:02.18 | SkramX | localhost / # |
00:02.27 | Mother_ | wrong window |
00:02.33 | SkramX | wooops sorry. |
00:03.35 | jskcr | <obsidian-studios> yes the beta in the x86 tree |
00:04.03 | obsidian-studios | jskcr: I sync on Sundays and did not see it? |
00:04.22 | jskcr | the beta-1.2.0 is in the cvs tree |
00:04.30 | jskcr | err not cvs in the portage |
00:05.07 | harryvv | rain rain rain |
00:05.14 | harryvv | we are in the rainy season |
00:05.18 | Ariel_ | yes we are |
00:05.28 | harryvv | your in florida |
00:05.29 | harryvv | :) |
00:05.29 | marc324 | vi crashed. swap file created. where is .swp file |
00:05.30 | Ariel_ | looks like it's going to rain till the morning. |
00:05.35 | harryvv | liquid sunshine there |
00:05.36 | Ariel_ | harryvv, yes so |
00:05.36 | harryvv | :) |
00:05.51 | Ariel_ | well at least it cools it down to 80 degrees |
00:05.56 | obsidian-studios | harryvv: opposite of normal weather, normally more sun than rain, this year? |
00:06.08 | harryvv | my backyard is looking like a mud pit ;) yanked a large tree stump out of the ground now filling in the hole. |
00:06.15 | Ariel_ | Our sunny weather comes after this month |
00:06.17 | SarahEmm | marc324: this really isn't the place for unix help. |
00:06.26 | SarahEmm | marc324: if you need * help, okay, but... unix help this isn't the place. |
00:06.36 | harryvv | obsidian-studios most days of the week are rain in the winter for washington and bc canada. |
00:06.42 | Icemaann | Ariel_: thanks for your help. i was doing it correctly, just had a typo. |
00:06.45 | Ariel_ | marc324, hahaha look for .name.swp |
00:06.50 | harryvv | once you head into the interior of bc is mostly snow. |
00:06.52 | *** join/#asterisk Veto_laptop (i=mdkuser@cpe-66-69-38-192.satx.res.rr.com) |
00:07.00 | sleepy_one | msg marc324 ls -al |more |
00:07.06 | sleepy_one | whoops |
00:07.06 | obsidian-studios | harryvv: yes, why I left Northern CA, I could not take 3-4 months of overcast rainy weather, much more north, no thank you |
00:07.09 | sleepy_one | ls -alt |more |
00:07.12 | Mother_ | locate .filename.swp |
00:07.26 | sleepy_one | find . -iname "*.swp" |
00:07.38 | Mother_ | so many ways to do the same thing :) |
00:07.45 | *** join/#asterisk Softpac2000 (i=user@rrcs-24-227-32-203.se.biz.rr.com) |
00:08.10 | harryvv | Wwe pacific northwesterners used to say we have webbed feet :) |
00:08.18 | sleepy_one | ahhhh yes indeed, Mother is absolutely correct :-) |
00:08.21 | justinu | dry as a bone here |
00:08.46 | obsidian-studios | harryvv: FL is known for rain but we have been getting allot this year, more than normal |
00:08.54 | Ariel_ | I used to live in the northwest. It's nice. But wet |
00:08.59 | harryvv | Ariel did you read the nifty news on cnn.com technoligy page how a wisp owner headed to new orleans and hooked up a voip asterisk service? |
00:09.14 | harryvv | summers here are great |
00:09.16 | Ariel_ | obsidian-studios, it's ok it's going to be sunny soon |
00:09.37 | harryvv | its the soggy wet days of winter where it can rain almost consistantly for days. |
00:09.42 | *** part/#asterisk Softpac2000 (i=user@rrcs-24-227-32-203.se.biz.rr.com) |
00:09.44 | Ariel_ | harryvv, yes saw a something on it. |
00:09.56 | obsidian-studios | Ariel_: I am holding out hope, running out of beach time though |
00:10.11 | Ariel_ | well it's suppose to be sunny this weekend |
00:10.24 | Ariel_ | too bad water is too cold in winter for me. |
00:10.26 | *** join/#asterisk wundaboy (n=asdf@67.189.30.47) |
00:10.35 | harryvv | I rmemeber the winter of 98, pinapple express one after the next slamming our state for 3 weeks non stop. Put a world seasonal record of 100 feet of snow on mount baker. |
00:10.59 | obsidian-studios | Ariel_: nothing compare to Pacific temp though ;) |
00:11.09 | fifer | harryw: where exactly are you? |
00:11.13 | harryvv | That was the time when I worked in the cozy warm buildings of microsoft. |
00:11.25 | harryvv | fifer, near Vancouver BC |
00:11.26 | obsidian-studios | Ariel_: what part? Im in Jax |
00:11.33 | Ariel_ | Vancover is nice. But I like Victoria Island better. |
00:11.35 | fifer | Ah |
00:11.37 | Juggie | harryvv, how about 1meter of snow in 2 days |
00:11.41 | Ariel_ | obsidian-studios, Miami |
00:11.51 | harryvv | ohh yea there are much dryer parts of vancouver island then here. |
00:12.07 | *** part/#asterisk blkbearnh (i=turner@c-24-147-155-3.hsd1.nh.comcast.net) |
00:12.07 | obsidian-studios | Ariel_: almost tropical there, I would be heading to Bahamas every chance I got ;) |
00:12.16 | fifer | Victoria is my favorite city on the continent |
00:12.17 | Ariel_ | no need |
00:12.21 | Ariel_ | we have the keys |
00:12.46 | harryvv | a tanker fire on the bridge to the keys the other day. |
00:12.48 | obsidian-studios | Ariel_: it's ok, but not the same mentality, or kettle drum music |
00:12.51 | Ariel_ | victoria has great gardens |
00:13.10 | Ariel_ | obsidian-studios, you have not been to the tiki hut then. |
00:13.20 | Juggie | harryvv, http://www.stuffintheair.com/images/Snowhouses.jpg |
00:13.24 | pc2 | how bad is iax/g711u over 65 ms? |
00:13.25 | Ariel_ | harryvv, yes it happens every other week |
00:13.29 | obsidian-studios | Ariel_: love to fish in the keys though, but hardly any beaches when I was there long ago hotels made their own beaches ? |
00:13.44 | harryvv | btw, want to goto a soaker city and live? goto Forks washington. Thay get 80 inches of rain a year. On the oposite side of the olympics resides Squim Washington with 18 inches of rain a year. |
00:13.46 | harryvv | :) |
00:13.49 | Ariel_ | beaches you need to head south to the dry tortugus |
00:14.29 | harryvv | juggie, did u take that pic? |
00:14.39 | Ariel_ | wow 80 inch. This month along we are going to get 17 inches |
00:14.42 | obsidian-studios | Ariel_: tortugus? |
00:14.49 | Juggie | harryvv, no. |
00:14.55 | justinu | i think we got 18 inches of rain this year... it was a record |
00:15.22 | Ariel_ | 18 inch is nothing |
00:15.22 | obsidian-studios | justinu: end of June we got like 10 in less than 5 days in Jax |
00:15.30 | justinu | yeah, this is a desert |
00:15.49 | harryvv | http://classic.mountainzone.com/news/99/bakerrecord.html |
00:15.57 | *** join/#asterisk huslage (n=huslage@c-67-169-200-122.hsd1.or.comcast.net) |
00:15.59 | harryvv | a offical record by noaa |
00:16.00 | harryvv | :) |
00:16.26 | Ariel_ | last month we had a low for Sept only 16 inches of rain. |
00:16.32 | harryvv | <PROTECTED> |
00:16.32 | *** join/#asterisk opus__ (n=opus@dahphish.org) |
00:16.35 | justinu | i like it dry |
00:16.43 | justinu | i can keep the top down year round on my s2000 |
00:16.45 | opus__ | is there some kind of dynamic universal way to detect the external ipaddress? |
00:16.53 | justinu | STUN works ok |
00:17.09 | harryvv | The snow on mount baker was so deep thay had to dig out the chair lifts and dig a ravine for them to go though. |
00:17.12 | opus__ | are there public STUN servers that do this? is there any example script/code that can pull this from STUN? |
00:17.16 | pc2 | justinu - rain isn't such a problem here. But it's damn cold. |
00:17.19 | justinu | yeah |
00:17.22 | obsidian-studios | it was nice in CA to wash your car 1 or twice a month in the summer :( |
00:17.24 | Ariel_ | opus__, fwd has one |
00:17.26 | justinu | there's some stun implementations |
00:17.29 | justinu | i found one in java |
00:17.36 | justinu | and stun.xten.net works |
00:17.41 | opus__ | ok |
00:17.45 | justinu | there's an RFC for it too |
00:19.58 | harryvv | Thats funny, mount baker closed for the season because of to much snow |
00:19.59 | harryvv | :) |
00:20.09 | pc2 | lol |
00:20.12 | harryvv | why need a stun server? |
00:20.16 | pc2 | Not many skiers think of having that problem. |
00:20.16 | pc2 | hehe. |
00:20.23 | lancey | anyone here familiar with chan_sip? |
00:20.35 | justinu | the code? |
00:20.39 | harryvv | pc2, read that link |
00:20.40 | harryvv | ? |
00:20.47 | pc2 | harryvv - I did. |
00:20.50 | harryvv | yea |
00:20.57 | harryvv | that was a sucky 3 weeks. |
00:20.59 | pc2 | harryvv - pic says it all |
00:21.00 | pc2 | hehe |
00:21.09 | pc2 | take snowmobiles around town =). |
00:21.13 | harryvv | yea |
00:21.16 | harryvv | no kidding. |
00:21.53 | harryvv | :) |
00:21.56 | Mother_ | grrr I can't seem to be able to read a variable I pass to an AGI script |
00:22.27 | opus__ | I think my WRT54g is stripping my ToS bit |
00:22.28 | *** join/#asterisk huslage_ (n=huslage@c-67-169-200-122.hsd1.or.comcast.net) |
00:22.36 | opus__ | is there a way I can turn this on with IPTABLES? |
00:22.38 | Ariel_ | use NoOp(${NAMEOFVARIABLE}) |
00:22.55 | *** join/#asterisk Pete_Largo (n=PeteLarg@225-196.35-65.tampabay.res.rr.com) |
00:23.00 | Ariel_ | stripping. no don't say it's so. Get Freeman on it |
00:23.00 | Pete_Largo | hello all |
00:23.09 | Mother_ | Ariel_: thanks, what will that do? (out of curiousity) |
00:23.28 | harryvv | opus, that is a router with asterisk on it? |
00:23.28 | lancey | opus__ : have a look at openwrt |
00:23.32 | Ariel_ | NoOP will display via the extension on the CLI what the variable has |
00:23.33 | pc2 | opus__ - Is it possible to use iptables for good packet shaping? |
00:23.40 | opus__ | harryvv , no |
00:23.46 | Ariel_ | Freeman/freeman |
00:23.46 | opus__ | pc2 I think so |
00:23.53 | opus__ | http://edseek.com/~jasonb/articles/traffic_shaping/scenarios.html |
00:23.58 | opus__ | i need to run sveasoft |
00:24.02 | opus__ | it has a lower TCO |
00:24.10 | Ariel_ | opus__, use freeman |
00:24.15 | pc2 | freeman? |
00:24.15 | opus__ | i can't afford to waste an entire day on it |
00:24.19 | Ariel_ | it's the gpl of sveasoft |
00:24.25 | Mother_ | ah OK - what I'm doing is a DBGet(HD_ID=whatever), then the next line is AGI(script.agi,${HD_ID}) |
00:24.26 | opus__ | freewan is kind of buggy |
00:24.32 | jskcr | pc2 yes |
00:24.52 | Mother_ | the variable is correctly set to the DB contents, I can see that in the CLI |
00:25.03 | jskcr | or you can set tos with iptables |
00:25.21 | pc2 | jskcr - tos = pita |
00:25.26 | Mother_ | but no matter how I try to access it from the agi script it's always empty |
00:25.32 | pc2 | tc isn't standard linux though :P |
00:25.34 | jskcr | pita? |
00:25.36 | Juggie | mother, what are you programming agi in? |
00:25.37 | pc2 | pain in the ass |
00:25.42 | opus__ | 'tc' is on the sveasoft firmware, hmm. |
00:25.45 | jskcr | not really |
00:25.55 | opus__ | jskcr can you explain how i can get it working? |
00:26.08 | pc2 | Someone needs to make a web front end to tc :) |
00:26.16 | jskcr | iptables -A OUTPUT -t mangle -p udp -m udp --dport 5060 -j DSCP --set-dscp 0x28 |
00:26.16 | jskcr | iptables -A OUTPUT -t mangle -p udp -m udp --sport 10000:20000 -j DSCP --set-dscp 0x28 |
00:26.20 | Mother_ | Juggie: bash |
00:26.42 | jskcr | http://www.netfilter.org/ |
00:26.53 | Juggie | mother, it should be like arg1,arg2, etc |
00:27.06 | pc2 | jskcr - Your upstream router has to acknolowege tos and act accordingly. I don't know any that do. |
00:27.16 | opus__ | pc2 - most do |
00:27.19 | Juggie | however bash receives parameters from the command line |
00:27.19 | Mother_ | yep, I can read fine arguments like agi_callerid etc. but this one is a nono |
00:27.20 | lancey | [03:22] <opus__> i need to run sveasoft |
00:27.27 | lancey | have a look at openwrt! |
00:27.32 | lancey | it's worth |
00:27.43 | opus__ | lancey Linux is only free if your time is worthless |
00:27.44 | Mother_ | Juggie: you mean the variable's name will also be ARGx? |
00:27.50 | pc2 | opus__ - Most upstream that I know of will just ignore your tos bit. |
00:27.58 | jskcr | get a cheap dlink and throw openwrt on it and it will work fine with tos |
00:27.59 | pc2 | opus__ - Otherwise, I'd set all my traffic to high priority. |
00:28.01 | opus__ | pc2 - comcast doesn't |
00:28.12 | *** join/#asterisk tehdely (n=delysiid@home.teambarry.org) |
00:28.16 | tehdely | boy howdy |
00:28.23 | lancey | opus__ but it works... |
00:28.23 | opus__ | pc2 - i hope you don't run p2p software :) |
00:28.36 | Juggie | mother, well, how does a bash script receive command line parameters |
00:28.36 | *** join/#asterisk jeffgus (n=jeffgus@2002:d856:c704:0:0:0:0:1) |
00:28.37 | pc2 | opus__ - why? |
00:28.40 | Juggie | i dont know how it does it |
00:28.44 | Ariel_ | can the openwrt use shorewall? |
00:28.46 | Juggie | it should be in there. |
00:28.49 | jskcr | pc2: check out http://peertech.org/IpQosTricks?pollresults%5B108%5D=1 too |
00:28.51 | Juggie | php receives arg1 arg2 and so on |
00:29.04 | Mother_ | OK thanks...I'll investigate that way |
00:29.08 | tehdely | quick dialplan question... let's say i have a specific context which maps to one trunk, and i want all numbers dialed in that context to dial to the equivalent number on the iax peer |
00:29.11 | jskcr | pc2: use google cache for that one |
00:29.14 | tehdely | how do i describe the extensions |
00:29.17 | tehdely | _ ? |
00:29.23 | *** join/#asterisk rigid (n=The@port-212-202-73-207.dynamic.qsc.de) |
00:29.24 | tehdely | i don't want to use a prefix |
00:29.25 | rigid | re |
00:29.38 | Ariel_ | you don't have to use prefixes |
00:29.47 | tehdely | wonderful. what is the syntax if i don't? |
00:30.17 | opus__ | hmmm |
00:30.30 | Ariel_ | 1NXXNXXXXXX,1,Dial(IAX2/Blah@ipaddress/${EXTEN}) |
00:30.33 | rigid | what do i have to take into special account when setting up asterisk for use with a hardware sip-router? i read http://gentoo-wiki.com/HOWTO_Asterisk and wanna use asterisk to disallow certain prefixes... |
00:30.44 | tehdely | isn't the '1' a prefix? |
00:30.54 | tehdely | also what if i want the number to be of arbitrary length |
00:30.55 | Ariel_ | ~docs |
00:30.56 | jbot | somebody said docs was Documentation can be found at http://www.digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or http://www.voip-info.org/wiki-Asterisk or http://www.asteriskdocs.org or http://www.oreilly.com/catalog/asterisk |
00:31.06 | tehdely | the peer in question is FWD |
00:31.07 | Ariel_ | X. |
00:31.17 | rigid | it all sounds straightforward, but i didn't find a doc concerning proxying |
00:31.58 | Druken | ok... does anyone know how the god damn auth work for sip ? |
00:32.25 | pc2 | jskcr - I still don't think any upstream providers would honor the qos bit. |
00:32.56 | Ariel_ | Druken, what do you mean? |
00:33.24 | Ariel_ | rigid, sip proxy, stun servers? SER what are you trying to do? |
00:33.30 | jskcr | pc2 it does not matter |
00:33.45 | jskcr | as long as the router thats connected to the upstream does |
00:34.31 | opus__ | god damn it. |
00:34.40 | rigid | Ariel_, now my sip-router logs into my providers sip.server.com ... i want to login at my server which then forwards everything to my provider. i want to have control over the numbers dialed (forbidding expensive dial-prefixes) |
00:34.42 | Ariel_ | tos bits are ok for your own network but once out in on the internet it's a waste of time. |
00:34.43 | opus__ | If I install openwrt, i need to setup dhcp tftp |
00:35.01 | rigid | Ariel_, +it |
00:35.13 | rigid | Ariel_, i want it to login |
00:35.21 | opus__ | Ariel, then what would work? |
00:35.21 | *** join/#asterisk jdv79 (n=jdv79@u1057064.ul.warwick.net) |
00:35.31 | jdv79 | can i gen a call from the cmdline? |
00:35.34 | jdv79 | just for kicks:) |
00:35.46 | jdv79 | the console is what i meant |
00:35.47 | Druken | Ariel_: i mean i have a sip provider, and they are telling me my shit's broken... (personally i think they are wrong) but anyways... they say incoming calls require authentication |
00:36.02 | tzanger | did that blkbearnh guy get his stuff figured out? |
00:36.21 | lancey | anyone any ideas how chan_sip works out the "fromuser" when none specified in sip.conf ? |
00:36.22 | Ariel_ | Druken, what is the error your getting |
00:36.27 | lancey | it should use the callerid, right? |
00:36.28 | Ariel_ | use sip debug |
00:36.35 | file | lancey: correct |
00:36.44 | lancey | file: well it doesn't all the time |
00:36.49 | Druken | i did use sip debug... he's telling me that my server has to auth to his, to receive calls... |
00:36.51 | *** join/#asterisk bjohnson (n=bjohnson@i216-58-58-237.cybersurf.com) |
00:36.56 | Druken | don't exactly make sence in my mind... |
00:36.56 | Ariel_ | fromuser fromdomain is used more for broadworks setups |
00:37.00 | sleepy_one | Druken, sip debug are Ariel said and are you registering with them ? for your incoming |
00:37.01 | lancey | any hints? |
00:37.07 | file | Druken: not usually... |
00:37.07 | rigid | Ariel_, is asterisk not the right choice for this? |
00:37.11 | lancey | or i should file a bug for this? |
00:37.18 | tzanger | I love the pig named bacon hahaha |
00:37.23 | Ariel_ | rigid, I kinda don't know whatyou want to do? |
00:37.29 | tzanger | <-- wathcing varsity blues |
00:37.34 | lancey | when receiving call from another * box, it gets set to "Unknown" |
00:37.41 | Ariel_ | rigid, but asterisk can setup accounts and do routes for dialing |
00:37.47 | Druken | sleepy_one: don't get anything on sip debug when i call the number, and yes, i'm registered with them |
00:38.00 | sleepy_one | Druken, have you tried ethereal and filtering for your provider's IP ? |
00:38.21 | sleepy_one | to see what packets are getting exchanged |
00:38.24 | Druken | no... i can dial out them, i just don't get the calls |
00:38.38 | Pete_Largo | you mean incoming calls Drunken? |
00:38.42 | Ariel_ | is the router port forwarding port 5060 to your box Druken |
00:38.42 | Druken | yeah |
00:38.58 | Pete_Largo | in sip.conf do you have them set up as peer/user/friend? |
00:39.00 | rigid | Ariel_, using asterisk without a phone card... a hardware-sip router (with phone connected) should be proxied... so that illegal prefixes won't be dialed |
00:39.01 | Druken | Ariel_: you think i'd run a server behind a router? |
00:39.12 | Ariel_ | Druken, I do |
00:39.20 | sleepy_one | tried sip show registry? iax2 show registry? |
00:39.22 | lancey | file: -- Executing Set("IAX2/csvarna-3", "CALLERID(number)=086510510") in new stack |
00:39.25 | Mother_ | I think it's definitely not passing the variable to the bash script |
00:39.32 | Druken | sip show registry shows it as registered |
00:39.37 | lancey | and then : From: "Unknown" <sip:Unknown@x.x.x.x>;tag=as098e3719 |
00:39.50 | FuriousGeorge | if i have exten=> s,1,answer(2) in my inbound calls context for zap, what do i gotta do to make it check for fax? just have a 2nd priority context for the fax? |
00:39.57 | Pete_Largo | Drunken, what is the 'type' in sip.conf for that provider? |
00:40.05 | Ariel_ | wait(3) |
00:40.17 | FuriousGeorge | exten=> fax,2,blah? |
00:40.17 | Druken | Ariel_: no... not behind a common firewall/router... 5060 is wide open |
00:40.34 | Ariel_ | Druken, what service |
00:40.42 | Ariel_ | FuriousGeorge, no |
00:40.43 | Druken | Pete_Largo: friend, and i have autocreatepeer=yes |
00:40.54 | Ariel_ | all fax extension must start on 1 |
00:41.03 | Druken | Ariel_: define service |
00:41.04 | file | lancey: it should set the user part of the SIP URI to that number, funky |
00:41.16 | lancey | file: funky, indeed |
00:41.25 | lancey | and when using IAX2 phone to originate the call, it does |
00:41.33 | Ariel_ | Druken, who is sending you the call? |
00:41.35 | lancey | when originating the call with another * box, it doesn't |
00:41.36 | FuriousGeorge | Ariel_: so it will automagically check for fax after an answer even if i dont tell it to beyond putting fax in the dialplan? |
00:41.46 | lancey | that's why i assume some sort of bug |
00:41.50 | Druken | Ariel_: netfone.ca |
00:41.50 | SarahEmm | gaah |
00:41.54 | SarahEmm | anyone using polycoms with HEAD? |
00:42.00 | sleepy_one | Druken: do you have the username and pass defined in sip.conf? or their public key if they use keys? |
00:42.04 | Ariel_ | FuriousGeorge, fax is detected via zap and it goes the fax,1 extension |
00:42.23 | Druken | sleepy_one: yes of course |
00:42.37 | sleepy_one | I know voicepulse and others use their public key for auth |
00:42.40 | rigid | Ariel_, i have a hardware sip-router that registers with a sip.server.com ... |
00:42.40 | rigid | Ariel_, the hardware router has not the option to control prefixes... it allows every number and dials it. |
00:42.43 | rigid | Ariel_, maybe i can use asterisk to "control" prefixes and send some "cmd failed" to the hardware-sip router so that it gives a busy-tone... |
00:42.48 | rigid | understand? |
00:42.50 | rigid | just simple |
00:43.19 | Ariel_ | rigid, if the router does it ahead of your asterisk box and you have not control over it your stuck. |
00:43.35 | Ariel_ | Remove vonage router exchange for real one |
00:43.40 | lancey | :) |
00:44.05 | Ariel_ | not/no |
00:44.07 | rigid | Ariel_, before my box? i don't have asterisk, yet... |
00:44.08 | lancey | anyone knows something about this SIP URI weirdness, or i'm posting a bug? |
00:44.21 | rigid | Ariel_, what do you suggest? |
00:44.40 | Ariel_ | rigid, ebay |
00:44.58 | rigid | Ariel_, so i can't control prefixes using linux? |
00:45.03 | rigid | doubt that... |
00:45.13 | Ariel_ | linux has nothing to do with a dial plan |
00:45.47 | rigid | Ariel_, my provider seems to use sipX ... i could set it up (in theory) to fake the server, and forward legal numbers to the real sipX server... |
00:45.57 | rigid | i thought there was an easier solution |
00:46.15 | rigid | i also came along "partysip" but rarely documented |
00:46.23 | Ariel_ | rigid, your trying to change your callerID |
00:46.56 | rigid | Ariel_, no... i'm trying to prevent everyone from dialing expensive prefixes... 0900-xxx |
00:47.01 | rigid | Ariel_, for example |
00:47.28 | Ariel_ | rigid, your unit does not allow for a dial plan get rid of it get an asterisk box instead. |
00:48.11 | rigid | Ariel_, so asterisk can not forward? i don't wanna buy isdn equipment... i wanna use 2 analog normal type of phones |
00:48.22 | pc2 | I have 7 spa-2000 vonage boxes here that I got free after mail in rebate at staples :) |
00:48.28 | justinu | nice |
00:48.31 | justinu | can you unlock them? |
00:48.38 | Ariel_ | rigid, yes but it we are not on the same page |
00:48.40 | pc2 | justinu - yp :) |
00:48.43 | pc2 | justinu - yup :) |
00:48.44 | rigid | rigid, i wouldn't be here if i could simply change the equipment |
00:48.44 | justinu | nice |
00:48.45 | pc2 | justinu - did, hehe. |
00:48.54 | pc2 | Don't know what I'm going to do with them (ebaY?) |
00:48.56 | pc2 | I only need a few. |
00:48.59 | pc2 | hook up the family I guess. |
00:49.04 | justinu | is there anyway to unlock my rtp300? |
00:49.22 | pc2 | google :) |
00:49.23 | pc2 | I don't know. |
00:49.27 | pc2 | probably not |
00:49.28 | pc2 | it's rare |
00:49.32 | Ariel_ | rigid, I don't actually understand what your asking. |
00:49.47 | Mother_ | well...good night |
00:49.48 | rigid | Ariel_, you're fooling me... |
00:50.10 | rigid | what else can i tell? |
00:50.15 | justinu | off for the night |
00:50.15 | *** part/#asterisk Mother_ (n=Mother@93.Red-80-32-127.staticIP.rima-tde.net) |
00:50.16 | justinu | later |
00:50.21 | *** join/#asterisk robertwaters (i=H2O@ppp-69-223-114-159.dsl.kntpin.ameritech.net) |
00:50.28 | Ariel_ | rigid, you have a box that sends calls via the internet but you want to have your people not dial numbers then put the asterisk box between them |
00:51.01 | rigid | Ariel_, i want asterisk to forward/manipulate/generate sip-packets... manipulate packets coming from a sip-router |
00:51.07 | *** join/#asterisk jeffgus (n=jeffgus@2002:d856:c704:0:0:0:0:1) |
00:51.10 | Ariel_ | rigid, your phones (Sip) ----- Asterisk ----- Your funnyrouter box ---- internet |
00:51.25 | *** join/#asterisk wundaboy (n=asdf@67.189.30.47) |
00:51.40 | rigid | Ariel_, hmm... can asterisk handle analog equipment and generate a dialtone and stuff? |
00:51.46 | Ariel_ | yes |
00:51.54 | Ariel_ | tdm400p |
00:51.59 | Ariel_ | t1/pri cards |
00:52.07 | rigid | Ariel_, hmm... how much are they? |
00:52.19 | Ariel_ | that depends |
00:52.28 | Ariel_ | how many ports |
00:52.33 | rigid | Ariel_, at least 2 |
00:52.58 | rigid | Ariel_, can you tell me some brands or what is needed to distinguish them from cards that don't work? |
00:53.11 | jskcr | If im using a 64 bit intel proccessor should I use k8 |
00:53.16 | Ariel_ | rigid, 2 analog ports get a sipura 2002 it's cheaper connect it to the asterisk box then an account form the asterisk to the box. |
00:53.18 | jskcr | in the makefile |
00:53.49 | rigid | Ariel_, i have to buy new equipment... is it available? how much round about? |
00:53.56 | rigid | Ariel_, the router is 80 euro |
00:54.05 | rigid | Ariel_, more or less? |
00:54.23 | Ariel_ | rigid, I am in the states but a sipura 2002 is about 70 dollars |
00:54.42 | *** join/#asterisk robertwaters (i=H2O@ppp-66-72-73-200.dsl.kntpin.ameritech.net) |
00:56.04 | *** part/#asterisk tehdely (n=delysiid@home.teambarry.org) |
00:56.27 | robertwaters | can anyone help me find information about routing calls by incomeing trunk? |
00:57.10 | Ariel_ | robertwaters, what type of trunk |
00:57.20 | rigid | Ariel_, that's worth a try... |
00:57.56 | lancey | ok, here it is : http://bugs.digium.com/view.php?id=5405 |
00:58.04 | lancey | if someone has any ideas... |
00:58.27 | robertwaters | I am useing 2 x100p modems |
00:59.19 | robertwaters | what I am trying to do is have a diffrent menu based on what line they call |
01:01.28 | jdv79 | how do you pickup a line, play some file in the background, and handle DTMF at the same time? |
01:01.50 | Druken | to make a call i need a user or peer entry ? |
01:02.07 | jdv79 | http://sial.org/pbot/13502 |
01:02.11 | jdv79 | that should work, right? |
01:02.40 | Pete_Largo | Drunken, user |
01:02.42 | Pete_Largo | no |
01:02.43 | Pete_Largo | peer |
01:02.44 | Pete_Largo | sorry |
01:02.48 | Pete_Largo | user is to receive call |
01:02.56 | Druken | hmm... |
01:03.17 | Druken | seems when i put in the peer entry block, i loose my incoming calls |
01:03.26 | Druken | when i remove it... i get my incoming calls |
01:03.34 | Pete_Largo | robertwaters, you can send the call to a different context in zaptel.conf |
01:03.53 | Pete_Largo | Drunken, you can have two entries for one provider, make on a peer and one a user |
01:03.54 | jdv79 | it answers and plys the file fine but it doesn't catch my DTMF and switch |
01:04.50 | lancey | isn't the "friend" like a peer + user, or i'm wrong? |
01:04.53 | Druken | Pete_Largo: i realize this... your missing what i'm saying... if i don't have a block for it... my incoming works... |
01:04.54 | Pete_Largo | or is it zapata.conf |
01:05.05 | lancey | Druken: try type=friend |
01:05.26 | *** join/#asterisk lters (n=lters@mrtcdsl-034.mis.net) |
01:05.56 | jdv79 | Set(TIMEOUT(response)=seconds) ? |
01:06.04 | jdv79 | what would a real world version of that look like? |
01:06.26 | jdv79 | SetTimeout(5) ? |
01:06.28 | Pete_Largo | robertwaters, you can send the call to a different context in zapata.conf - sorry for the mixup |
01:06.53 | *** join/#asterisk robertwaters (i=H2O@ppp-66-72-73-200.dsl.kntpin.ameritech.net) |
01:07.10 | Pete_Largo | robertwaters, you can send the call to a different context in zapata.conf - sorry for the mixup |
01:07.55 | Pete_Largo | lancey, you are right |
01:08.02 | *** part/#asterisk lters (n=lters@mrtcdsl-034.mis.net) |
01:08.09 | robertwaters | that is what I thought you could do but I must not be reading my zapata.conf right becouse I thought it only shows the one trunk |
01:08.20 | *** join/#asterisk glm2k (n=GLM@rrcs-24-199-11-41.west.biz.rr.com) |
01:08.28 | Ariel_ | robertwaters, just asign a different context to each cards setup |
01:08.31 | Pete_Largo | you have 2 cards? |
01:08.35 | robertwaters | yeah |
01:08.39 | Pete_Largo | and they both work? |
01:08.41 | robertwaters | both are answering the phone |
01:08.48 | Ariel_ | card one channel=1 context pstn1 |
01:08.57 | Ariel_ | channel=2 context=pstn2 |
01:09.04 | robertwaters | ok |
01:09.21 | Druken | Pete_Largo: is that amazing or something? 2 cards? |
01:09.24 | Ariel_ | Then put your rules to do what you want in the extensions.conf [pstn1] blah |
01:09.29 | Ariel_ | [pstn2] blah |
01:09.30 | Pete_Largo | no, just asking the question |
01:09.32 | Pete_Largo | ass |
01:10.06 | Ariel_ | Pete_Largo, it's hard to get to x100p to work but it's possible |
01:10.19 | Ariel_ | to/two |
01:10.23 | robertwaters | I did not have a problem getting them to answer |
01:10.25 | Pete_Largo | It was hard when I got my single one working |
01:10.42 | robertwaters | but I think my config is not right as the way I look at it they show up as one card |
01:10.51 | Ariel_ | robertwaters, yes but next comes echo problems and other things |
01:10.53 | robertwaters | but then again I am an amagure |
01:11.06 | robertwaters | sorry amature |
01:11.28 | robertwaters | really? I had great quality out of them today |
01:11.56 | Ariel_ | robertwaters, nice |
01:12.56 | robertwaters | I keep hearing they are bad but I am just playing for a house system |
01:13.42 | Pete_Largo | robertwaters, I'm not 100% sure, because I haven't done it, but I think if you just put something in zapata.conf like: channel => 1 \ context = first-line \ channel => 2 \ context=second-line that should do the trick... I think |
01:13.50 | jskcr | x100p make sure your use aggressive supression |
01:14.02 | Pete_Largo | where \ means a new line |
01:14.44 | Ariel_ | robertwaters, pastebin.ca your zapata.conf and let us look at it. Then we can give you better pointers. |
01:15.07 | Ariel_ | context goes above the channel=1 |
01:15.15 | Pete_Largo | good thinking Ariel_ |
01:15.39 | Pete_Largo | whoops, I new it was something like that though |
01:16.04 | Ariel_ | Pete_Largo, is it raining on your side of the state |
01:16.22 | Pete_Largo | not that I'm aware of |
01:16.44 | Pete_Largo | but then again, I haven't moved from my perch for a few hours |
01:16.47 | jdv79 | what is autofall through? |
01:17.01 | jdv79 | i thought the t extension would get hit first |
01:17.09 | *** join/#asterisk alrs (n=lars@dsl092-033-090.lax1.dsl.speakeasy.net) |
01:17.13 | Ariel_ | t is for timeout |
01:17.31 | jdv79 | wouldn't that get hit before a "auto fallthrough"? |
01:17.41 | jdv79 | if set timeout response was set |
01:17.46 | *** part/#asterisk joat (n=joat@laketaylor.org) |
01:17.51 | Ariel_ | yes if it's in the context |
01:18.10 | jdv79 | i see it execute the set timeout but not the t |
01:18.14 | jdv79 | how do i debug that? |
01:18.33 | Ariel_ | what does the cli say |
01:18.36 | robertwaters | hey I have a question for you ariel |
01:19.07 | Ariel_ | ok |
01:19.09 | robertwaters | would the info your looking for in the zapata.conf be in the zapta-auto.conf |
01:19.25 | Ariel_ | robertwaters, are you using amp |
01:19.31 | robertwaters | yes |
01:19.45 | Ariel_ | ahh that is a different thing |
01:19.59 | robertwaters | is it possible to do what I am trying to do without killing amp? |
01:20.40 | Ariel_ | yes but your going to have to do some editing in your extensions_custom.conf |
01:21.00 | jdv79 | Ariel_, http://sial.org/pbot/13503 |
01:21.03 | Ariel_ | and also in the zapata-auto.conf |
01:21.53 | robertwaters | ok any ideas of what to search for to find the information as to what I need to change? |
01:22.08 | jdv79 | and the context is: http://sial.org/pbot/13504 |
01:22.14 | Ariel_ | jdv79, it's not finding the exten => t |
01:22.36 | Ariel_ | robertwaters, #amportal |
01:22.37 | robertwaters | I dont mind looking it up myself and learning just been looking for a week and can not figure out what I am looking for |
01:22.38 | jdv79 | it looks like its there? |
01:23.19 | robertwaters | ok thank you very much for the information |
01:23.28 | Ariel_ | exten => t, 1, Goto(vm-end) |
01:23.33 | Ariel_ | is your problem |
01:23.36 | jdv79 | i was just wondering about that:) |
01:23.38 | jdv79 | thanks |
01:23.42 | Ariel_ | Goto(vm-end,s,1) |
01:24.53 | lancey | bye guys |
01:24.57 | jdv79 | i think i meant playback |
01:25.03 | jdv79 | i dont knpw |
01:30.43 | *** join/#asterisk groogs (n=greg@d221-73-237.commercial.cgocable.net) |
01:31.27 | groogs | hey is the person that runs asteriskgui.com (flobi?) here? I'm working on a very similar problem to that dialplan inserter |
01:31.33 | Ariel_ | jdv79, your goto are not right |
01:31.48 | jdv79 | why not? |
01:31.58 | Ariel_ | your going to have to do more reading look at the one I put up for you goto(contex,s,1) |
01:32.00 | jdv79 | the docs on voip-info seems to say they are |
01:32.20 | jdv79 | aren't context and extension optional? |
01:32.22 | FuriousGeorge | Ariel_: i asked this question ans missed if you answered or not. do u recall when we were talking about fax detection? were you saying all i have to do is have the "fax" priority, and * will check calls to see if its a fax? |
01:32.41 | jdv79 | http://voip-info.org/wiki/index.php?page=Asterisk+cmd+Goto |
01:33.05 | FuriousGeorge | its just that obviously i have to answer it fo it to listen to the tone |
01:33.20 | Ariel_ | FuriousGeorge, if your system detects a fax in the default context for that zap channel it needs to have an exten => fax,1,BLAH |
01:33.21 | FuriousGeorge | so i figured i'd have to start with an exten=> s,1,answer |
01:33.49 | Qwell | how good is the zap fax detection? |
01:33.53 | Ariel_ | FuriousGeorge, yes you need to answer it but then 2,wait(3) on 2nd line |
01:34.05 | Ariel_ | Qwell, seems to work fine for me. |
01:34.16 | FuriousGeorge | so what do i start with? exten=> fax,1,answer |
01:34.30 | FuriousGeorge | exten => fax,2,wait(2) |
01:34.32 | Ariel_ | FuriousGeorge, no fax exten is a fall through |
01:35.26 | FuriousGeorge | so i answer on s, wait 2, and if it detects a fax it goes to exten=> fax,1 |
01:35.34 | groogs | dial() has a failure goto priority (whatever its called) n+101... are there any other applications that do that? |
01:35.36 | Ariel_ | FuriousGeorge, your channel context is it just for a fax or does it do other things |
01:35.46 | FuriousGeorge | no its for incoming zap calls |
01:36.04 | FuriousGeorge | rings extensions then goes to vm, the ususal |
01:37.02 | Ariel_ | FuriousGeorge, if you have fax detection set to incoming then you just need to have in your context a line for the exten => fax,1,Blah but also ahead of it exten => s,1,answer exten => s,2,wait(3) exten=> s,3,Goto(what you want) |
01:37.51 | FuriousGeorge | ok i get it |
01:38.01 | FuriousGeorge | i dont necesarily need a goto with s,3 i assume |
01:38.20 | Ariel_ | FuriousGeorge, well you need what you want it to do next |
01:38.26 | jskcr | with a xeon should I still use PROC=k8 |
01:38.38 | FuriousGeorge | and answer(2) makes it wait two seconds to answer, not answer and wait right? |
01:38.53 | Ariel_ | I have not used answer(2) |
01:39.07 | Ariel_ | I always use the next line with wait(3) |
01:39.57 | FuriousGeorge | k |
01:40.00 | FuriousGeorge | thanks |
01:40.20 | Ariel_ | FuriousGeorge, np |
01:40.38 | jdv79 | is it possible that 2way audio could be off? |
01:42.14 | jdv79 | wow, on the CLI it sets the timeout and kills the call in about the same second |
01:42.19 | jdv79 | does that sound right? |
01:42.38 | *** join/#asterisk xlogik (n=xlogik@pool-141-154-127-152.bos.east.verizon.net) |
01:42.45 | hardwire | mutilator: not to sound odd.. but can I go bowling too? |
01:43.05 | jdv79 | maybe i shouldn't be using HEAD:) |
01:48.15 | *** join/#asterisk vtsherwood (n=vtsherwo@cpe-24-210-51-17.columbus.res.rr.com) |
01:48.30 | jdv79 | Ariel_, i added the context like you said and still same thing |
01:48.41 | jskcr | chan_zap.so: undefined symbol: ast_pickup_call anyone know why im getting that with cvs head |
01:49.03 | vtsherwood | hello all |
01:52.11 | *** part/#asterisk robertwaters (i=H2O@ppp-66-72-73-200.dsl.kntpin.ameritech.net) |
01:52.38 | JerJer | jskcr: have you been noload'ing modules in modules.conf ? |
01:52.52 | *** join/#asterisk littleball (n=littleba@bb219-75-114-103.singnet.com.sg) |
01:53.20 | jskcr | JerJer this is on a xeon |
01:53.30 | jskcr | It seems to be using the k8 proc |
01:53.34 | littleball | hi, who is using TE411P card? |
01:53.34 | JerJer | res_features.so is required :( |
01:54.00 | jskcr | WARNING[13979]: loader.c:314 __load_resource: /usr/lib/asterisk/modules/res_features.so: undefined symbol: adsi_available |
01:54.05 | jskcr | this is 1.2.0 beta |
01:54.11 | vtsherwood | I was wondering if anyone might have information on a rather weird problem our *boxes are having. It's rather long so I won't write the whole thing to the group, but anyone who's interested in trying to help, I'll send you the story. |
01:54.12 | JerJer | jskcr: because res_features.so is not loading |
01:54.16 | littleball | i got the following error during run ztcfg command :ZT_SPANCONFIG failed on span 1: No such device or address (6) |
01:54.28 | jskcr | JerJer I know that :) I dont know why :( |
01:54.39 | JerJer | have you mucked with modules.conf ? |
01:54.52 | jskcr | its a sample install |
01:55.05 | jskcr | so not at all |
01:56.10 | jskcr | its a freshly emerged gentoo system |
01:56.28 | Qwell | 866-378-1477 - anybody recognize this number? |
01:56.35 | jskcr | with a clean compile of 1.2.0 beta |
01:56.59 | jskcr | Hmm ill try killing the -m64 and proc=k8 |
01:57.12 | jskcr | since its not recognizing the xeon for being a intel |
01:57.48 | jskcr | but it is 64 bit because of the em64t processor |
01:57.50 | *** join/#asterisk ManxPower (n=eric@adsl-70-247-221-174.dsl.lgvwtx.swbell.net) |
01:58.52 | jdv79 | JerJer, any idea? - http://sial.org/pbot/13507 |
01:59.52 | vtsherwood | in case you want an advance idea of the problem, it involves incoming/outgoing sip channel limits, 480 "Temporarily Unavailable (Call Limit)" and being unable to make or receive calls for a long time even after the channels in use have returned to an acceptable number |
02:01.34 | marc324 | unrelated--where is swap directory in linux |
02:01.49 | *** join/#asterisk r0d3nt (i=r0d3nt@tinfoilhat.net) |
02:02.54 | *** join/#asterisk protien (i=jjmtrev@203-173-26-187.dyn.iinet.net.au) |
02:03.06 | protien | is skype supported under asterisk the same way fwd is? |
02:04.29 | denon | no, skype is a hack |
02:04.37 | denon | </unofficial_response> |
02:04.39 | ManxPower | protien, If Sykpe used the same protocol as FWD (SIP) then it would be |
02:04.55 | protien | what protocol does skype use |
02:04.59 | denon | skype |
02:05.02 | protien | oh |
02:05.10 | ManxPower | protien, Their own protocol that they don't document. |
02:05.15 | protien | are there any other free sip providers other than fwd? |
02:05.18 | marc324 | can someone tell where the .swp are located? |
02:05.23 | *** join/#asterisk ronaldl79 (n=chatzill@c-24-8-54-203.hsd1.co.comcast.net) |
02:05.23 | mmlj4 | hey ManxPower |
02:05.25 | ManxPower | (Hmm...sounds like IAX, but at least there's souce for IAX stacks) |
02:05.29 | ronaldl79 | Hey folks |
02:05.39 | ronaldl79 | Just found out about OpenPBX.org. WTF? |
02:05.40 | ManxPower | Hello mmlj4 |
02:06.14 | ronaldl79 | Is anyone following this OpenPBX development? |
02:06.19 | mmlj4 | what's wrong with sticking a wildcard in your box and plugging that into your skype ATA? |
02:06.36 | ronaldl79 | It's a branch of Asterisk -- but is this really necessary? |
02:06.44 | mmlj4 | ManxPower: using wireless? |
02:07.09 | ManxPower | ronaldl79, Remember by GMOME got started? They were unhappy with the QT licensing? |
02:07.31 | Lathos42 | Evening everyone |
02:07.46 | ManxPower | ronaldl79, same thing here. OpenPBX.org is forking Asterisk because they don't like the idea of having to license to Digium any code they want put in the main Asterisk source tree |
02:07.48 | ronaldl79 | Hmmm, I actually don't recall, Manx. I didn't take Linux seriously until about two years ago. |
02:08.28 | ManxPower | They seem to want to have Digium include the code in the main Asterisk source tree, but not license it to Digium. |
02:08.50 | ManxPower | mmlj4, If I didn't hate wireless before, I do now. |
02:08.58 | mmlj4 | hee |
02:08.59 | *** join/#asterisk AlexCTI (n=alex@weston-69.65.86.197.myacc.net) |
02:09.10 | AlexCTI | Hi.. |
02:09.20 | ManxPower | ronaldl79, You would have to ask one of the OpenPBX people to get the "official story" |
02:09.54 | ManxPower | mmlj4, I had to replace all three of my 2.4Ghz cordless phones too |
02:10.01 | mmlj4 | heh, nice |
02:10.03 | AlexCTI | Anyone knows how can I know if my Asterisk box is using "Silence Suppresion" |
02:10.07 | mmlj4 | switching channels didn't help? |
02:10.21 | ManxPower | AlexCTI, Asterisk NEVER EVER uses silence supression |
02:10.21 | Qwell | AlexCTI: * doesn't do VAD |
02:10.31 | ManxPower | mmlj4, I have the higher end DSS phones. |
02:10.39 | mmlj4 | channels 1, 6 and 11 seem to be immune to cordless interference |
02:10.47 | mmlj4 | oh, hmm... |
02:10.57 | ManxPower | I'm on channel 1 for my uplink and 6 for my local wirelss |
02:11.13 | ManxPower | I got the 5.6Ghz phones |
02:11.16 | vtsherwood | can anyone help with a sip response code 480? |
02:11.42 | mmlj4 | vtsherwood: i can tell you it's a server error |
02:12.00 | vtsherwood | I understand, I know what it is, I have a rather complicated issue with it |
02:12.01 | ManxPower | The the guy that runs the "wireless ISP" is a radio guy with a mission to beat the telcos. As far as I can tell, he can't route himself out of a paper bag |
02:12.02 | AlexCTI | Manxpower, but the RTP Silence Suppression is not used on asterisk? |
02:12.08 | vtsherwood | I have two servers running * |
02:12.15 | mmlj4 | heh |
02:12.18 | vtsherwood | and one gets calls from the other |
02:12.26 | ManxPower | AlexCTI, RTP silence supression is not supported with Asterisk |
02:12.45 | AlexCTI | ok, thanks. |
02:12.51 | vtsherwood | but when it's limit is reached on the SIP channels my main server sends a 480 back (temporaril unavailable) |
02:12.54 | ManxPower | My wireless link has been up and down more times today then Linta Lovelance |
02:13.07 | ManxPower | Linda, that is |
02:13.10 | Qwell | my god, worst IVR ever |
02:13.13 | vtsherwood | and then the slaved server cannot make calls even after it's channel usage goes under the limit amount |
02:13.18 | vtsherwood | any ideas? |
02:13.43 | Qwell | "Your call will be placed in queue for the next availablone moe reprmentesentativent please. |
02:13.48 | groogs | ok with this "n priority" stuff, do you do eg: exten => 1,MainDial+101,... exten => 1,n,.... ? or Maildial+102 ? |
02:14.45 | vtsherwood | I'd give more info, but I don't think the _whole_ room wants to hear about it |
02:14.50 | Ariel_ | groogs, in head you can just do n for next then n for the next statement |
02:14.58 | ManxPower | vtsherwood, what version of Asterisk? |
02:15.05 | wunderkin | groogs: you don't see the examples in extensions.conf? |
02:15.16 | vtsherwood | CVS-HEAD from about two weeks ago |
02:15.28 | wunderkin | if you are running off an old one, check /usr/src/asterisk/configs/extensions.conf.sample |
02:15.29 | groogs | wunderkin: hm guess i could look at that :p |
02:15.34 | vtsherwood | I don't know about the slave server, it's a customers |
02:15.36 | ManxPower | vtsherwood, update, if the problem persists, report it on bugs.digium.com |
02:15.40 | jskcr | hya Qwll |
02:15.47 | vtsherwood | hrm... |
02:15.50 | vtsherwood | ok, I'll do that |
02:15.58 | *** part/#asterisk vtsherwood (n=vtsherwo@cpe-24-210-51-17.columbus.res.rr.com) |
02:16.00 | ManxPower | vtsherwood, you ARE on the asterisk-cvs mailing list, right? |
02:16.04 | Qwell | anybody happen to have an escallation number at sprint (pcs if possible)? |
02:16.20 | Qwell | -l |
02:16.23 | protien | is there any other totally free sip services like fwd? |
02:16.47 | ManxPower | protien, If it's free, it's not worth much |
02:17.01 | protien | i know manx, but im playing learning asterisk |
02:17.05 | protien | im just trying to play arounds |
02:18.11 | Ariel_ | Proteque, sipphone, stanaphone are sip service out there |
02:18.23 | Qwell | oh shit, $68.88 additional usage on my sprint account...thats awesome |
02:18.28 | Ariel_ | protien, not Proteque |
02:18.47 | protien | thanks ariel ill check them out |
02:18.54 | Qwell | Anytime Minutes - $0.50/min43.0 $21.50 |
02:19.28 | *** join/#asterisk AJ-Mpls (i=DJAJay@63.231.252.9) |
02:19.56 | AJ-Mpls | what config file hold the PORT info that asterisk listens to? |
02:20.09 | *** join/#asterisk mtgh (n=chatzill@dsl093-001-038.det1.dsl.speakeasy.net) |
02:22.12 | jdv79 | is head not working for anyone? |
02:22.20 | *** part/#asterisk AJ-Mpls (i=DJAJay@63.231.252.9) |
02:22.33 | Ariel_ | jdv79, I don't use head I use stable |
02:23.11 | mmlj4 | what could cause voicemailmain to not see keypresses? -- No username but # key pressed. Using CID '2076' / -- Playing 'vm-password' (language 'en') / -- Incorrect password '' for user '2076' (context = <any>) |
02:23.32 | jdv79 | ok |
02:23.39 | Damin | mmlj4: Because you have the wrong DTMF mode setup for your device? |
02:23.45 | jdv79 | 1.0.9? |
02:23.49 | mmlj4 | hmm... |
02:23.59 | mmlj4 | but the thing dials out |
02:24.28 | ManxPower | mmlj4, SIP devices collect the digits, then send them all at once in a SIP message. |
02:24.32 | mmlj4 | actually, no device is able to send DTMF to voicemail anymore, dunno what I broke |
02:24.47 | ManxPower | mmlj4, dtmfmode=rfc2833 |
02:24.53 | mmlj4 | ManxPower: so how to actually sent them? |
02:24.59 | mmlj4 | ah, well, lemme try that |
02:25.04 | ManxPower | mmlj4, it's sent as a call setup message. |
02:25.04 | *** join/#asterisk mcf3782 (n=mcf3782@adsl-19-83-37.asm.bellsouth.net) |
02:26.30 | Ariel_ | jdv79, yes I only use stable for my production systems. |
02:28.13 | jdv79 | what's the rev for that? i'm only familiar with svn:( |
02:29.08 | jdv79 | cvs update -r v1-0-9zaptel libpri asterisk ? |
02:29.35 | ManxPower | jdv79, cvs update -r v1-0 blah blah blah |
02:29.44 | jdv79 | thanks |
02:32.51 | mmlj4 | ok, that fixed it, thanks |
02:33.57 | Qwell | anybody wanna do me a huge favor? |
02:34.07 | Qwell | I need like 5-6 8xx numbers called, and tell me what they are... |
02:36.10 | denon | hmm .. I would, except I know when people want that, there's a reason they dont want to do it themselves :) |
02:38.14 | *** part/#asterisk obsidian-studios (n=obsidian@c-66-177-188-197.hsd1.fl.comcast.net) |
02:42.19 | Qwell | nevermind |
02:45.02 | *** join/#asterisk Veto_laptop (i=mdkuser@cpe-66-69-38-192.satx.res.rr.com) |
02:45.46 | Damin | So.. |
02:45.52 | Damin | Who is going to Astricon? |
02:47.04 | protien | hm |
02:47.18 | protien | im trying to connect to this stanaphone service, i can register okay |
02:47.21 | protien | but when i try to make a call |
02:47.22 | protien | Oct 7 12:02:14 NOTICE[20804]: chan_sip.c:9385 handle_response_invite: Failed to authenticate on INVITE to '"pro" <sip:1001@203.173.26.187>;tag=as03301ca3' |
02:48.21 | *** part/#asterisk msw (n=msw@rdu-nat.rpath.com) |
02:48.29 | *** join/#asterisk msw (n=msw@rdu-nat.rpath.com) |
02:56.03 | jdv79 | how does one record a session? |
03:08.19 | *** join/#asterisk n0where (n=kc@d254204.ppp.asahi-net.or.jp) |
03:09.57 | *** join/#asterisk n3u7 (n=neutrin0@69.197.165.118) |
03:10.55 | jskcr | yay I found my gentoo asterisk problem :) |
03:11.22 | ronaldl79 | Has anyone tried GNU's PBX? (the name slips my memory currently) |
03:12.32 | hypa7ia | ronaldl79: yate? |
03:13.04 | ronaldl79 | Is that it? I'm not sure, I thought it was something else. |
03:14.40 | *** join/#asterisk AlexCTI (n=alex@weston-69.65.86.197.myacc.net) |
03:16.08 | marc324 | <PROTECTED> |
03:16.08 | marc324 | <PROTECTED> |
03:16.08 | marc324 | <PROTECTED> |
03:19.36 | *** join/#asterisk Corydon76-home (i=mauve@pdpc/supporter/sustaining/Corydon76-home) |
03:20.52 | *** join/#asterisk luke-jr_ (n=luke-jr@CPE-65-26-133-171.kc.res.rr.com) |
03:22.57 | hypa7ia | ronaldl79: there are a couple |
03:23.32 | ronaldl79 | Which ones? I think there's one which starts with a B, but damn, the name again escapes me. |
03:24.02 | ronaldl79 | And, hypa, have you tried them? If so, how do they compare to *? I've considered testing other open source PBXs, but why bother? * works just fine, and it's extremely powerful. |
03:25.38 | hypa7ia | ronaldl79: SER is used a lot and is powerful in some ways * is not, from what i hear. but i've only used asterisk |
03:26.32 | JamesDotCom | ser is not a pbx |
03:26.42 | ronaldl79 | SER is definitely something I want to checkout for SIP messaging |
03:26.54 | JamesDotCom | ser rules for a sip proxy |
03:27.16 | file[laptop] | yes, yes it does |
03:27.27 | jskcr | ser 250,000+ on a dual xeon |
03:27.31 | jskcr | that rules |
03:27.39 | hypa7ia | whoa :-) |
03:28.20 | jskcr | Ive personally benchmarked 60,000 on a ser system with only a 20% load |
03:28.26 | jskcr | that was a p4 |
03:28.29 | ronaldl79 | Cool |
03:28.35 | file[laptop] | straight proxying is fine |
03:28.38 | ronaldl79 | I need to learn more about SIP proxies. |
03:28.39 | file[laptop] | as it's not that CPU intensive |
03:28.55 | *** join/#asterisk Ariel_ (n=Ariel@dsl-20-177.cofs.net) |
03:29.19 | hypa7ia | ronaldl79: http://www.voip-info.org/wiki-Open+Source+VOIP+Software |
03:29.48 | hypa7ia | do any of you know about the borderware sipproxy / "firewall" thing? |
03:29.51 | marc324 | <jskcr> that was a p4 |
03:29.51 | marc324 | <ronaldl79> Cool |
03:29.51 | marc324 | <file[laptop]> straight proxying is fine |
03:29.51 | marc324 | <ronaldl79> I need to learn more about SIP proxies. |
03:29.51 | marc324 | <file[laptop]> as it's not that CPU intensive |
03:29.52 | marc324 | * Ariel_ has quit IRC (Read error: 104 (Connection reset by peer)) |
03:29.54 | marc324 | * Ariel_ has joined #asterisk |
03:30.19 | file[laptop] | ...yeah |
03:30.26 | *** join/#asterisk jero (n=sflphone@savoirfairelinux.net) |
03:31.19 | jdv79 | anyone know if legacy carriers block DTMF? |
03:31.26 | jdv79 | i'm getting some weird stuff here |
03:31.32 | jskcr | noooo |
03:31.36 | jskcr | never |
03:31.41 | jdv79 | huh |
03:32.05 | jskcr | inband signaling, is never blocked |
03:32.24 | jdv79 | tht's what i though but my buddy is trying to tell me different |
03:32.36 | jskcr | well hes wrong |
03:33.09 | jskcr | legacy systems inheritly encode everything at twice the bandwidth of human speech around 3500hz |
03:33.22 | jskcr | dtmf is within that range |
03:34.55 | jskcr | 697hz,1209hz for 1 |
03:35.32 | jskcr | dtmf is in the middle land of human speech |
03:36.35 | jdv79 | i think his CLASS4 swtich is being a little more intelligent than it ought to be:) |
03:38.02 | jskcr | in that case the dtmf-mute-encoder may be active |
03:38.21 | jskcr | its a picnic problem in that case |
03:38.28 | jdv79 | i'm getting really short blips and inconsistent timings when i monitor |
03:38.43 | jdv79 | what is the time it takes for asterisk to recognize a tone? |
03:39.03 | jdv79 | 1/4 second? |
03:39.04 | *** join/#asterisk ManxPower (n=eric@adsl-70-247-221-174.dsl.lgvwtx.swbell.net) |
03:39.07 | jskcr | at least 40ms |
03:39.21 | jdv79 | i think that much is getting through... |
03:39.22 | jdv79 | weird |
03:41.05 | marc324 | why do I get this: Starting Zap/1-1 at test,s,1 failed so falling back to exten 's' |
03:41.44 | jdv79 | i think its getting through but its definitely being mucked with |
03:44.47 | *** join/#asterisk santiago (n=santiago@208.195.215.231) |
03:48.38 | Sedorox | anyone read/try the howto on unlocking the Linksys PAP2? |
03:49.41 | jdv79 | asterisk do h323? |
03:49.51 | jdv79 | cause that CLASS4 swtich is just gwing for me to SIP... |
03:51.56 | *** join/#asterisk bmg505 (n=leon@rndf-146-32-157.telkomadsl.co.za) |
03:55.05 | protien | exten => 2002,1,SetCallerId("" <3477274087@sip.stanaphone.com>|a) |
03:55.10 | protien | is this syntax incorrect |
03:55.19 | file[laptop] | uh no |
03:55.41 | protien | hm |
03:55.41 | protien | Oct 7 13:21:53 NOTICE[20804]: chan_sip.c:9385 handle_response_invite: Failed to authenticate on INVITE to '"asterisk" <sip:asterisk@203.173.26.187>;tag=as55e58ffe' |
03:55.48 | protien | still showing as asterisk@ |
03:55.55 | file[laptop] | you create an entry in sip.conf, a peer entry |
03:55.58 | file[laptop] | with username and fromuser set |
03:56.17 | protien | done that |
03:56.38 | file[laptop] | and you're using that to dial out? |
03:56.42 | protien | yeah |
03:56.46 | *** part/#asterisk jdv79 (n=jdv79@u1057064.ul.warwick.net) |
03:56.50 | file[laptop] | funky |
03:57.30 | file[laptop] | I'd look into it further except I'm literally in bed typing sideways |
03:57.36 | file[laptop] | because that's how big a geek I am |
03:58.10 | marc324 | i try to use asterisk to answer a call using x100p. i get this: Starting Zap/1-1 at test,s,1 failed so falling back to exten 's' |
03:58.31 | protien | hm |
03:58.40 | protien | am i setting the caller id right |
03:58.49 | protien | because when i dont have the @sip.stanaphone.com |
03:58.58 | file[laptop] | you shouldn't have the @sip.stanaphone.com part... |
03:59.01 | protien | it identifies me as 3477274087@ip |
03:59.06 | protien | well how can i set |
03:59.09 | protien | sip.stanaphone.com |
03:59.12 | file[laptop] | you can't set callerid if you're using fromuser |
03:59.13 | jskcr | Sedorox: you can unlock th pap2's |
03:59.15 | protien | because if i dont have that |
03:59.27 | protien | i get errors |
04:00.05 | Sedorox | I'm reading the article on telephreak.com (I think) |
04:00.05 | file[laptop] | protien: have you tried searching for existing working configs? |
04:00.06 | Sedorox | about how to do |
04:00.07 | Sedorox | it |
04:00.13 | protien | yeah |
04:00.25 | protien | im using one thats ment o be working |
04:00.30 | protien | this seems to be a regular problem |
04:00.44 | protien | Oct 7 13:27:41 NOTICE[20804]: chan_sip.c:9385 handle_response_invite: Failed to authenticate on INVITE to '"3477274087" <sip:3477274087@203.173.26.187>;tag=as159a8c3c' |
04:01.19 | file[laptop] | marc324: it's trying to start in the context test, on extension s, first priority |
04:06.13 | digime | anyone know how to set the ToS bit in Windows for a softphone? |
04:06.47 | marc324 | the x100p is a fxsks module? |
04:07.05 | marc324 | should i use modprobe fxsks or modprobe fxoks? |
04:07.35 | *** join/#asterisk mog_home (n=mogorman@user-24-236-84-48.knology.net) |
04:07.39 | *** join/#asterisk jeanmi__ (n=besnard@front.tekkno.net) |
04:08.17 | jeanmi__ | I would like to test a SIP phone (soft). Is there any echo server I could use for that purpose ? |
04:08.31 | Sedorox | link2voip doesn't do tollfree anymore huh? |
04:09.43 | jeanmi__ | and also, could someone recommend me SIP provider with a kind of "pay as you go" offer ? |
04:10.19 | Sedorox | jeanmi__, there is a ton.... check out voip-info.org... they have good info on providers |
04:10.51 | *** join/#asterisk krisguy (n=krisguy@h216-170-039-057.adsl.navix.net) |
04:17.32 | protien | Oct 7 13:40:09 NOTICE[20804]: chan_sip.c:9385 handle_response_invite: Failed to authenticate on INVITE to '"3477274087" <sip:3477274087@203.173.26.187>;tag=as24ac16ba' |
04:17.42 | protien | how can i cahnge that to 3477274087@sip.stanafone.com |
04:17.53 | protien | instead of sip@ip |
04:23.24 | jlewis | anyone here going to astricon/been to that area before? |
04:23.36 | Qwell | jlewis: I live there |
04:23.47 | Qwell | well...practically |
04:24.02 | Qwell | on Harbor? probably a bit |
04:24.37 | jlewis | whats weather out there been like? |
04:24.50 | Qwell | hot |
04:24.58 | Qwell | it should cool down by then though |
04:26.06 | jlewis | trying to figure out if I should rent a car, or just take the airport shuttle from/to the airport and be stuck at the hotel for the week |
04:28.39 | jlewis | is it safe to assume there'll be 802.11b internet at Astricon? |
04:29.14 | mog_home | well |
04:29.16 | mog_home | last year |
04:29.20 | mog_home | there was no bandwith |
04:29.26 | mog_home | as we only had a t1 for everyone |
04:29.32 | mog_home | this year i imagine it will be better |
04:29.52 | twisted | no bandwidth was an understatement |
04:30.20 | mog_home | hey i was able to co asterisk 1.0 from hotel |
04:30.25 | mog_home | so there was bandwith |
04:30.51 | jlewis | IIRC, the hotel itself provides internet to guests |
04:30.52 | twisted | you said there wasn't, ninny |
04:31.24 | mog_home | i mean there was bandwith |
04:31.27 | mog_home | but the latency |
04:31.30 | mog_home | oh the latency |
04:31.38 | twisted | nah |
04:31.39 | twisted | latency was fine |
04:31.44 | twisted | it was congested |
04:32.07 | jlewis | yeah...they claim highspeed wired and wireless...is this the same hotel as last year? |
04:32.18 | twisted | jlewis, no, wrong side of the US |
04:32.22 | jlewis | ah |
04:32.33 | jlewis | so last year it was on my side? |
04:32.37 | mog_home | yeah |
04:32.40 | mog_home | atlanta |
04:32.50 | twisted | driving distance for me :) |
04:32.53 | *** join/#asterisk KaBewM (n=kabewm@66-215-7-106.dhcp.psdn.ca.charter.com) |
04:32.55 | jlewis | geez...that would have been so much more convenient |
04:33.08 | twisted | this year it's a 5 your plane ride |
04:33.11 | twisted | s/your/hour |
04:33.15 | jlewis | a few hours drive rather than a day of flying |
04:33.21 | mog_home | as all digium could come |
04:33.53 | twisted | i'm glad it is where it is |
04:34.02 | twisted | another city i haven't visited yet that i've wanted to |
04:34.25 | twisted | all the flights? |
04:34.30 | jlewis | from FL to CA |
04:34.49 | twisted | ah, which end of the phallus are you on? |
04:35.02 | jlewis | the layovers, the missed flights, the getting stuck places |
04:35.14 | marc324 | I have a x100p... do i use modprobe wcfxo or modprobe wcfxs? |
04:35.18 | jlewis | neither...central FL |
04:35.23 | mog_home | wcfxo |
04:35.28 | twisted | ah |
04:35.51 | twisted | jlews == jerry lewis? |
04:35.55 | twisted | as in jerry lewis kids? |
04:35.58 | jlewis | and then run fxsks signaling |
04:36.02 | jlewis | twisted: no |
04:36.05 | twisted | :P |
04:36.09 | twisted | sorry, couldn't resist |
04:36.43 | twisted | so lessee |
04:36.56 | twisted | tomorrow i work, get home, relax. |
04:37.10 | twisted | saturday - clean apt, do laundry, start adding new tracks to ipod |
04:37.28 | twisted | sunday - finish cleaning - pack up for trip, sleep lots. |
04:37.38 | marc324 | I get this with x100p Channel 'Zap/1-1' sent into invalid extension 's' in context 'default', but no invalid handler |
04:37.51 | twisted | monday - work from 9 till 2, go to airport, leave for LA |
04:37.53 | twisted | yay |
04:38.35 | marc324 | I have this in extensions.conf --------[incoming] |
04:38.35 | marc324 | exten => s,1,Answer() |
04:38.35 | marc324 | exten => s,2,Playback(hello-world) |
04:38.35 | marc324 | exten => s,3,Hangup() |
04:38.49 | twisted | marc324, your problem is quite obvious |
04:39.11 | twisted | i imagine if you look a little closer you'll see it |
04:39.29 | mog_home | ent into invalid extension 's' in context 'DEFAULT', |
04:39.35 | mog_home | there is a hint ^_^ |
04:40.17 | twisted | <PROTECTED> |
04:40.39 | twisted | irc needs a control character for italics |
04:41.37 | twisted | we have bold, inverse, mIRC colors (*choke*gag*), and actions |
04:41.52 | twisted | irc is quite civilized. |
04:42.00 | mog_home | its stone age |
04:42.02 | mog_home | heh |
04:42.05 | marc324 | alright. that worked. thx |
04:42.14 | mog_home | i would like jabber or something but people are here |
04:42.18 | mog_home | and thus im here |
04:42.44 | twisted | mog_home, irc is the grandfather of protocols like jabber, aim, and the like. |
04:42.52 | mog_home | yeah |
04:43.02 | mog_home | but i dont use minix |
04:43.03 | twisted | show respect to your elders. |
04:43.04 | mog_home | i use linux |
04:43.23 | mog_home | i dont use a mac classic i use my mini |
04:43.24 | mog_home | etc |
04:43.25 | mog_home | etc |
04:43.27 | mog_home | the old |
04:43.29 | mog_home | they die |
04:43.32 | mog_home | it happens |
04:43.36 | mog_home | except irc |
04:43.44 | twisted | irc is in a nursing home on life support |
04:44.29 | mog_home | irc is opensource life support |
04:44.33 | twisted | and will probably remain that way until the powers that be decide to let the plug be pulled |
04:44.34 | mog_home | or the freenode |
04:44.48 | twisted | oh wait, that was schaivo.. |
04:44.54 | mog_home | heh |
04:48.15 | marc324 | are there additional gsm files that can be downloaded? |
04:48.48 | *** join/#asterisk mut (i=WebChat@i.think.napoleon.dynamiteblows.com) |
04:48.52 | mut | i need to buy a laptop right now.. where to do.. needs to be financed |
04:50.19 | mog_home | asterisk-sounds |
04:50.24 | mog_home | has more |
04:53.16 | *** join/#asterisk viLeR (i=1000@66.128.47.232) |
04:56.40 | *** join/#asterisk SplasPood (n=jwb@brooklyn.paravolve.net) |
05:01.49 | *** join/#asterisk sleepy_one (n=chatzill@cpe-24-166-32-45.neo.res.rr.com) |
05:02.09 | sleepy_one | hello everyone, what's up? |
05:02.34 | sleepy_one | beta1 is out already???? |
05:07.09 | wunderkin | already? |
05:07.30 | wunderkin | where have you been sleepyhead |
05:08.21 | sleepy_one | been sleeping |
05:08.38 | wunderkin | ah haha didnt even look at your nick |
05:08.51 | wunderkin | wow i wonder where you got your nick from |
05:09.15 | sleepy_one | lol |
05:09.18 | marc324 | is it possible to listen to the list of availablbe gsm over the net? |
05:09.29 | sleepy_one | what do u mean? |
05:09.34 | mog_home | just co it marc |
05:09.37 | mog_home | its not that big |
05:10.00 | sleepy_one | you can Playback any gsm |
05:10.14 | sleepy_one | with Playback(filename) |
05:10.54 | sleepy_one | or cp them to the vm directory, pretend they are voicemails and use the voicemail system to navigate |
05:11.01 | marc324 | i need: "enter extension" sound file |
05:11.09 | sleepy_one | ohhhhh |
05:11.23 | sleepy_one | cvs co asterisk-sounds # I believe |
05:11.49 | sleepy_one | wget http://ftp.digium.com/pub/asterisk/asterisk-sounds-1.0.9.tar.gz # here's the direct link |
05:12.59 | sleepy_one | the default sounds reside in /var/lib/asterisk/sounds IIRC |
05:13.06 | mog_home | there are more sounds in cvs |
05:13.12 | mog_home | than in web |
05:13.13 | *** join/#asterisk tartar (n=tartar@CPE0004e27b716e-CM014370001917.cpe.net.cable.rogers.com) |
05:18.00 | marc324 | how do youplay gsm file in xp? |
05:18.26 | sleepy_one | winamp ? broken player |
05:18.33 | sleepy_one | errrr I mean media player |
05:18.37 | denon | marc324: download the j2 viewer or WavePad |
05:18.43 | *** join/#asterisk scardinal (n=supreme@port816.ds1-suoe.adsl.cybercity.dk) |
05:20.01 | sleepy_one | winamp and or broken player should play them if you have the right libraries / codecs / etc |
05:21.35 | Dr_Ray | if you search google for winamp gsm, you'll find a solution |
05:23.58 | marc324 | how can you playback digits, like 905 |
05:26.17 | sleepy_one | Playback(nine) Playback(zero) Playback(five) |
05:26.36 | sleepy_one | IIRC they are in the digits subdirectory |
05:27.53 | *** join/#asterisk BhaalWK (i=bhaal@freenode/staff/bhaal) |
05:29.22 | sleepy_one | my bad Playback(9) Playback(0) Playback(5) |
05:29.37 | sleepy_one | ls -al /var/lib/asterisk/sounds/digits |
05:31.58 | sleepy_one | exten => 666,1,Playback(digits/9) |
05:31.59 | sleepy_one | exten => 666,2,Playback(digits/0) |
05:32.01 | sleepy_one | exten => 666,3,Playback(digits/5) |
05:32.05 | sleepy_one | here's a real working example |
05:32.16 | sleepy_one | very basic but it works |
05:34.35 | sleepy_one | for your convenience here's the relevant wiki page http://www.voip-info.org/wiki-Asterisk+cmd+Playback |
05:35.48 | sleepy_one | I would also urge you to read up on Background() and ControlPlayback() |
05:36.36 | sleepy_one | Background will allow you to play a sound file and still be able to process DTMF etc |
05:38.21 | *** join/#asterisk tsp (n=tyler@S01060013102699ac.vc.shawcable.net) |
05:38.30 | tsp | how can I quickly and easilly set asterisk up as a sip server so I can use it to talk to my friends? |
05:38.50 | shimi | tsp: about 5 minutes if you use Asterisk@Home |
05:39.25 | *** join/#asterisk shido6 (n=curtis@d221-68-210.commercial.cgocable.net) |
05:39.33 | shimi | tsp: note, however, that the installation CD formats whatever you have on the computer you run it on ;) |
05:39.43 | tsp | ouch |
05:39.52 | marc324 | i found the function SayDigits() |
05:39.57 | tsp | I just need a chat client to talk to my firends but i'm stuck with console only |
05:40.07 | marc324 | it works. |
05:40.17 | shimi | I think there's an installer there for existing living machines, however I did not try it. The full OS (running CentOS3), works great. |
05:40.36 | sleepy_one | marc324, yes that too :-) |
05:40.38 | tsp | what's a good console only chat client for voip? |
05:41.10 | shimi | I'm afraid i am not sure ehow is asterisk related to your client being console or not? :) |
05:41.40 | sleepy_one | tsp, are you looking for a text messanging program over SIP???? |
05:41.58 | tsp | nope, just voice |
05:42.03 | tsp | I"m blind so use console with speakup |
05:42.15 | sleepy_one | you mean a command line SIP softphone? |
05:43.35 | sleepy_one | it sounds like you may be looking for a commandline capable SIP / VoIP softphone program |
05:43.39 | marc324 | what directory would be appropriate for the downloaded gsm files? |
05:43.50 | marc324 | where should it go? |
05:43.59 | sleepy_one | <PROTECTED> |
05:44.11 | tsp | yeah |
05:44.15 | tsp | minisip won't work |
05:44.21 | tsp | only one i've found |
05:44.35 | tsp | I can use speexenc | ssh speexdec but that lags like hell |
05:44.36 | *** join/#asterisk implicit (n=implicit@ip70-181-114-97.oc.oc.cox.net) |
05:46.03 | sleepy_one | tsp, hmmmm well gnophone linphone and kphone are open source - I don' sure if they have a console mode |
05:46.03 | marc324 | there is a 1.0.9 sound file and a 1.2.0 -- what is the difference? |
05:46.15 | sleepy_one | marc324, not sure |
05:47.00 | sleepy_one | tsp, hmmmm well gnophone linphone and kphone are open source - I'm not sure if they have a console mode but I'm sure one can be added or maybe run them in debug mode |
05:47.53 | sleepy_one | marc324, like mog_home said there are more sounds in CVS |
05:48.06 | marc324 | cvs? |
05:48.28 | marc324 | how can i download that to xp? |
05:48.35 | sleepy_one | <PROTECTED> |
05:48.37 | sleepy_one | # export CVSROOT=:pserver:anoncvs@cvs.digium.com:/usr/cvsroot |
05:48.38 | sleepy_one | # cvs login - the password is anoncvs. |
05:48.40 | sleepy_one | # cvs checkout |
05:48.43 | sleepy_one | cvs checkout asterisk-sounds |
05:48.55 | shimi | I think "Cornfed" supports CLI mode |
05:49.23 | sleepy_one | you can use CVS for win32 or native CVS under cygwin running under XP |
05:49.43 | sleepy_one | http://cygwin.com/setup.exe |
05:51.10 | sleepy_one | marc324, if you install cygwin under XP you will be able to use CVS, make, gcc/g++, etc etc |
05:51.27 | sleepy_one | and many unix/ GNU/Linux *BSD apps |
05:53.05 | sleepy_one | tsp, http://freshmeat.net/projects/cornfedsipua/ http://freshmeat.net/redir/cornfedsipua/48563/url_rpm/cornfedsipua-0.9.5-1.i386.rpm |
05:55.40 | sleepy_one | asterisk itself allows making phone calls from the asterisk console using your sound card but it can be tricky to setup sometimes |
06:04.01 | *** join/#asterisk _Thor (i=Christia@user-vc8fl7l.biz.mindspring.com) |
06:05.45 | *** join/#asterisk Tili (i=Tili@202-133-67-57-dialup.sat.net.pk) |
06:06.25 | *** join/#asterisk viLeR (i=1000@66.128.47.232) |
06:09.01 | *** join/#asterisk djin_ib (n=djin_ib@gridfox.xs4all.nl) |
06:09.18 | tsp | goodnight |
06:09.20 | tsp | thanks all |
06:09.42 | marc324 | how do you disable the overwrite prompt when using cp |
06:10.02 | djin_ib | doing a simple 'make install' on asterisk sound v1.2b1: |
06:10.05 | djin_ib | No description for sounds/access-code.gsm |
06:10.10 | djin_ib | What's wrong ?!? |
06:11.02 | _Thor | hi everybody... Anyone using the PrepaidCall application? |
06:11.17 | sleepy_one | there shouldn't be anything wrong, I think that might be a warning not a fatal error |
06:12.16 | djin_ib | sleepy_one, I think it's a fatal. The /var/lib/asterisk/sounds is quite empty |
06:12.53 | djin_ib | or can I just copy the sounds/ to /var/lib/asterisk ? |
06:13.01 | sleepy_one | sure that would work |
06:13.13 | sleepy_one | there's isn't anything that needs installed really IIRC |
06:13.46 | djin_ib | That's what I thought. I looks just like a copy process |
06:13.52 | _Thor | <PROTECTED> |
06:14.07 | sleepy_one | brb |
06:14.46 | *** join/#asterisk MrB0B0 (n=bobo@203.94.141.98) |
06:14.59 | Dr_Ray | n3u7 - it's a good book |
06:15.18 | MrB0B0 | anyone feel like lending a hand re: dundi |
06:15.43 | n3u7 | Dr_Ray:ya I'm pretty excited about this |
06:16.08 | Dr_Ray | I read it cover to cover, the day I got it |
06:16.17 | Dr_Ray | except for the isp stuff.. zzz. ") |
06:16.21 | Dr_Ray | er, sip |
06:16.49 | n3u7 | well I just poured myself another coffee and I'm planning on putting a pretty big dent in it tonight |
06:17.03 | n3u7 | i havean x100p card so far |
06:17.05 | djin_ib | DR-Ray, n3u7, what book is that? |
06:17.10 | Dr_Ray | ~docs |
06:17.12 | jbot | docs is, like, Documentation can be found at http://www.digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or http://www.voip-info.org/wiki-Asterisk or http://www.asteriskdocs.org or http://www.oreilly.com/catalog/asterisk |
06:17.22 | Dr_Ray | the last link |
06:17.22 | *** join/#asterisk websae (i=websae@207-118-134-96.dyn.centurytel.net) |
06:17.26 | n3u7 | djin_ib:its the Oreilly Asterisk book |
06:17.46 | djin_ib | Ok, I've seen that one. |
06:17.57 | djin_ib | Does it cover stuff supported in 1.2? |
06:18.02 | Dr_Ray | some |
06:18.08 | websae | i am wondering...does anyone here know of company that builds voip phones that if you're a VoIP company, you can provide your users with these phones and update them remotely from a server? |
06:18.25 | Dr_Ray | it makes mention of add n, instead of 1,2,3,4 in dialplans |
06:18.30 | djin_ib | websae, there're many |
06:18.34 | denon | websae: that'd be .. any voip phone, almost .. |
06:18.43 | denon | websae: they almost all support tftp provisioning |
06:19.01 | websae | im talking about software package that helps deploy those settings |
06:19.04 | djin_ib | websae, most support tftp, but SNOM has a http porvisioning as well. |
06:19.14 | websae | SNOM? |
06:19.22 | djin_ib | yes. |
06:19.34 | websae | djin_ib: what is that? |
06:19.37 | Dr_Ray | I bet a cisco provisioning tool would not be that hard to write |
06:19.43 | djin_ib | http://www.snom.com |
06:19.50 | djin_ib | it's a brand |
06:20.32 | websae | ahh okay |
06:20.53 | djin_ib | Dr_Ray, isn't that a bit limited (add n). v1.2 seems to introduce a lot more. |
06:21.34 | MrB0B0 | has anyone got the dundi enterprise example from voip-info.org working? I have two test boxes running and peering okay but all lookups return empty :$ |
06:21.52 | Dr_Ray | djin_ib - perhaps, I'm a bit behind on asterisk.. I mainly use it for zaptel.. so I've not kept up.. for example I did not know that you could spy sip/iax channels now |
06:22.01 | *** join/#asterisk Rowter (n=SilverDr@201.135.26.195) |
06:22.11 | Rowter | TE205P has E1/T1 jumpers? |
06:22.30 | Rowter | asterisk is detecting TE205P as T1, and I need it as E1! |
06:22.47 | djin_ib | Dr-Ray, it's never wrong to learn the basics (again). I'll order the book as well. |
06:23.38 | Dr_Ray | That's what it was for me, a review of the current state of it, Like I had never heard of call files, or the Directory() app.. I bought two copies, one for me to highlight.. and one for me to lend out. |
06:24.05 | djin_ib | The TE210P supports E1, T1, and J1 environments and is selectable on a per-card or per-port basis. |
06:24.15 | djin_ib | It doens't mention jumpers :) |
06:24.17 | Dr_Ray | When I started with asterisk a year or so ago, I learned exactly enough to replace our mitel sx50 |
06:24.29 | marc324 | how do you create a delay between playbacks? |
06:24.41 | djin_ib | uuuu, Wait(1) |
06:25.07 | Dr_Ray | the person who made call files gets a gold star in my book |
06:25.41 | djin_ib | Dr-Ray, same problem here. Learning by experience tends to limit your knowledge :) |
06:26.05 | Rowter | djin_ib, yeah but its getting T1, cause Channel 24 is reserved for D-channel when I start * |
06:26.51 | *** join/#asterisk argos73 (n=mike@65-85-207-101.client.dsl.net) |
06:26.57 | djin_ib | Rowter, I have no TE2XX experiece, aren't there any jumpers on the board? |
06:27.24 | Rowter | djin_ib, I didn't saw any, but I think I'll get it out again >) |
06:27.47 | *** join/#asterisk sleepy_one (n=chatzill@cpe-24-166-32-45.neo.res.rr.com) |
06:28.21 | argos73 | thought there was a t1/e1 jumper on it |
06:28.27 | argos73 | (err, two of them) |
06:28.38 | djin_ib | Rowster, there are jumpers on the TE100P, TE110P, TE410P, TE405P, so my guess TE2xxP has them as well ;) |
06:29.58 | Rowter | djin_ib, ok let me see.. |
06:30.22 | websae | what's a good billing software for asterisk |
06:30.29 | websae | anyone have any ideas or experience? |
06:30.56 | argos73 | dammit... gonna need to upgrade our merlin legend to make it do what I want with asterisk... |
06:33.07 | sleepy_one | why not replace it with * instead ? |
06:33.52 | sleepy_one | websae, not sure but since * keeps great CDR records it's a piece of cake to sum up usage and invoice clients |
06:34.38 | argos73 | would like to, but the cost of new phones (150+ of them) is too high right now... |
06:35.19 | argos73 | easier to integrate the two, and switch out phones a few at a time |
06:35.42 | sleepy_one | yes, I see |
06:36.19 | sleepy_one | will you need a T1 card for your merlin ? |
06:36.30 | argos73 | heh - just received it today |
06:36.54 | sleepy_one | must have cost you a small fortune |
06:37.05 | argos73 | nah - $675 or so |
06:37.10 | argos73 | not too bad |
06:37.17 | sleepy_one | still a good chunk of change |
06:37.22 | sleepy_one | new or ebay? |
06:37.30 | Rowter | djin_ib, yep it had a jumper >) |
06:37.35 | Rowter | hehe |
06:37.36 | argos73 | refurb from our local dealer |
06:37.59 | argos73 | might go ebay to get one for my legend system at home... |
06:37.59 | sleepy_one | ahh I see |
06:38.05 | djin_ib | Rowter, ran into the same reminder with my TE410P yesterday ;) |
06:38.32 | *** join/#asterisk ozant (n=ozan@85.96.199.40) |
06:39.11 | sleepy_one | I'm sure a lot of vendors would love to offer you a discount on 150+ phones but I can understand that's going to be a huge expenditure :-( |
06:40.02 | argos73 | to not lose any functionality, would prob go with cisco phones... around $50K for all of them |
06:40.27 | sleepy_one | plus the hidden costs of adding enough ethernet ports and POE unless you already have enough |
06:41.05 | sleepy_one | most IP phones have a built-in switch but I wouldn't recommend it |
06:41.14 | Rowter | djin_ib, got a diferent jumper for each.. excelent.. lets test now hehe.. |
06:41.34 | argos73 | plenty of ethernet ports (i went "wiring crazy" last summer), but poe isn't there yet |
06:42.05 | sleepy_one | http://www.voipsupply.com/product_info.php?products_id=505&ref=froogle |
06:42.19 | argos73 | have six 24-port HP switches just waiting for traffic.. :) |
06:42.25 | sleepy_one | they have 7960 refurbs for 294 each |
06:42.47 | sleepy_one | I wouldn't recommend refurbs but I would recommend the 7960 it's a decent phone |
06:42.47 | argos73 | hmm - not bad |
06:42.59 | argos73 | have a 7960 here at home - love it |
06:43.05 | Dr_Ray | I love my 7960 |
06:43.10 | Dr_Ray | the smartnet contract blew |
06:43.27 | *** join/#asterisk mmmToop (n=chatzill@196.31.11.194) |
06:43.35 | sleepy_one | I wish I had one too, I have an Iaxy with an analog phone instead |
06:43.53 | sleepy_one | not bad for 100 bux |
06:44.28 | argos73 | now the big question is "do I go ebay for the legend upgrade ($100, assuming it works) and risk alienating our hardware support contract, or do I get the upgrade from them for $1750?" |
06:44.32 | Dr_Ray | I could probably get by with a 7940 |
06:44.34 | argos73 | decisions, decisions.... |
06:44.52 | sleepy_one | 1750???? geeeeez |
06:45.10 | sleepy_one | 1750 = 10 snom phones IIRC |
06:45.25 | argos73 | it's the old "cheap bastard vs responsible corporate guy" battle |
06:45.41 | Dr_Ray | well, if it's works money, I'd say spend it |
06:45.55 | *** join/#asterisk KaBewM (n=kabewm@66-215-7-106.dhcp.psdn.ca.charter.com) |
06:46.21 | sleepy_one | the snom 190 is NOT a cisco but it's a very good entry-level phone for the $$$ (about 180 each) |
06:46.28 | argos73 | i'll probably blow the $100 myself for my legend system - if it works, i'll have work spend the major bucks for theirs |
06:47.07 | websae | im curious...if someone has vonage...how can they make it so they can use all their existing phones in their house with the ATA adapter that comes with vonage service? |
06:47.12 | argos73 | still trying to get my wife to appreciate the beauty of a 7960 in the living room.. :) |
06:48.03 | argos73 | if you don't exceed the REN max of the ATA, you should be able to just split the line |
06:48.31 | sleepy_one | ya just plug your analog line into it and disconect it at the telco box |
06:48.52 | sleepy_one | all the phones should work that way |
06:48.58 | websae | hmm |
06:49.10 | websae | so at the telco box... |
06:49.15 | websae | i would just wire that in? |
06:49.19 | websae | the ata adapter |
06:49.25 | argos73 | yup |
06:49.27 | sleepy_one | yup |
06:49.32 | websae | that should work out alright? |
06:49.35 | argos73 | just make sure the telco line is disconnected first... |
06:49.43 | sleepy_one | just disconnect from the telco so you don't fry anything |
06:49.50 | argos73 | didn't do that with my tdm400 fxs card - POOF! |
06:49.52 | websae | is there a site with a tutorial on this? |
06:49.54 | dersteer | I use the old telco lines to charge batteries |
06:49.55 | dersteer | :) |
06:50.19 | *** join/#asterisk Bonzai070 (n=pirch@wbs-146-129-162.telkomadsl.co.za) |
06:50.29 | MrB0B0 | hi guys, I hate to be a pain in the *ss but... theres always a but isn't there... I'm so close to getting enterprise dundi going its driving me crazy |
06:50.53 | MrB0B0 | any general pointers on failed lookups between otherwise happy peers? |
06:51.07 | argos73 | websae - any site that describes how house phone wiring should be done will apply... just replace "line coming from telco" with "line coming from ata" |
06:51.08 | sleepy_one | websae, label the wires in your telco terminal box and then disconnect them from the telco terminals |
06:51.19 | *** join/#asterisk nesys (n=nesys@2001:1418:1a6:0:20d:93ff:fe28:3ef8) |
06:51.31 | sleepy_one | make sure there's no power on the line |
06:51.51 | sleepy_one | then put an RJ11 on the wires and plug that into your ATA from vonage |
06:52.17 | argos73 | easiest way I can think of - disconnect telco line, then plug ata into a normal phone jack somewhere in your house |
06:52.35 | dersteer | thats how I did it |
06:52.36 | sleepy_one | aye that would work too :-) |
06:53.08 | argos73 | i firmly believe in the "stupid simple" way of doing things. :) |
06:53.08 | sleepy_one | however some phones may prevent other phones from getting caller ID info and such |
06:54.14 | argos73 | if your telco demarc is one of those big gray boxes they attach to the back of your house, just open the cover and unplug the rj11 inside it |
06:54.15 | sleepy_one | I have tons of tools and crimpers so I can put an RJ11, RJ45, RJ48 or RG6 connectors on just about anything ;-) |
06:54.50 | argos73 | yes, but do you have a crimper for DB9/25 pins? :) |
06:54.56 | *** join/#asterisk jdiskywlkr (n=kvirc@ip68-0-83-251.tu.ok.cox.net) |
06:55.06 | sleepy_one | yes actually |
06:55.17 | argos73 | heh - one of my prized posessions |
06:55.19 | sleepy_one | have an IDC crimper too |
06:55.29 | sleepy_one | can crimp BNC too |
06:55.40 | sleepy_one | although I haven't done that in 10 years |
06:55.50 | *** join/#asterisk Inv_arp (i=junya@adsl-156-145-65.mia.bellsouth.net) |
06:55.53 | argos73 | pull mine out occasionally for my radio work |
06:56.51 | *** join/#asterisk MuppetMaster (n=MuppetMa@62.57.152.193) |
06:56.58 | *** part/#asterisk MuppetMaster (n=MuppetMa@62.57.152.193) |
06:59.19 | sleepy_one | I misplaced the remote terminator for one of my LAN testers once so I made one with some CAT5e a breadboard and 8 LEDs |
07:00.17 | argos73 | heh - cool |
07:00.45 | denon | sure, but will it tell you how far down the wire your break is? |
07:00.57 | sleepy_one | lol |
07:01.04 | sleepy_one | no, of course not |
07:01.19 | denon | sounds to me like you wasted that breadboard .. |
07:01.29 | sleepy_one | it wasn't permanent |
07:01.29 | denon | a couple ICs and a power source, and you could have somethin |
07:01.36 | argos73 | sure it will... you just need to be really good at measuring the time between when you flip the switch and when the light comes on |
07:01.36 | sleepy_one | aye |
07:01.45 | sleepy_one | lol |
07:01.50 | denon | argos73: if there's a break, the light wont light :) |
07:02.24 | denon | now, you could slowly increase the voltage, until there's enough for it to arc over the gap .. and that might tell you something ;) |
07:02.28 | sleepy_one | that's what my inductive amp and probe is for |
07:02.32 | argos73 | hence the reason people have built "power off" lights into stuff. :) |
07:02.37 | denon | course 480vac may not traverse cat5 too well |
07:03.26 | argos73 | cya |
07:03.31 | sleepy_one | IIRC CAT5 will only survive to about 120VDC MAX before it blows |
07:04.05 | sleepy_one | at low amps |
07:04.22 | sleepy_one | nite denon |
07:05.06 | sleepy_one | IIRC T1s use 100VDC and I run that over CAT5 all the time |
07:05.09 | argos73 | sigh... one more day, and I'm outta that place for two weeks.... |
07:05.31 | sleepy_one | vacation? |
07:05.36 | argos73 | yup |
07:05.50 | sleepy_one | oh good :-) |
07:06.13 | argos73 | i'm sure they'll be calling me every two hours, though... they always do. |
07:06.26 | sleepy_one | the question is how will they survive without you? |
07:06.35 | argos73 | usually, not well. |
07:07.00 | sleepy_one | be happy 2 cover 4 u :-) |
07:07.11 | *** join/#asterisk samyl (n=samyl@194.167.18.244) |
07:07.51 | argos73 | absolutely LOVED it when they called my cell phone when I was in Florida about 2 years ago... "Help! We're too dumb to follow your simple instructions!" "Well, I'm on a boat in the middle of the Gulf of Mexico, and can't do a damn thing about it right now." |
07:08.23 | samyl | Hi |
07:09.00 | sleepy_one | that's why I always have a laptop and one or more internet-capable phones with me when I leave the batcave (at least when I was able to afford the cell phones) |
07:09.02 | *** join/#asterisk Uberbot (n=Uberbot@69.252.219.76) |
07:09.37 | sleepy_one | naturally reception wouldn't be too good in the gulf |
07:09.58 | argos73 | funny thing about it was that my laptop was back at the place I was staying - condo owned by my boss |
07:10.07 | *** part/#asterisk Uberbot (n=Uberbot@69.252.219.76) |
07:10.20 | argos73 | surprisingly, we were about 6 miles out, and the reception was still pretty good |
07:10.35 | *** join/#asterisk Mw3 (i=mw3@national.t-error.hu) |
07:10.39 | samyl | I've a problem with meetme |
07:10.47 | samyl | when i try to use it |
07:10.56 | samyl | my asterisk server tell me |
07:11.09 | samyl | <PROTECTED> |
07:11.14 | timecop | heh |
07:11.21 | samyl | Unable to open pseudo channel - trying device |
07:11.24 | timecop | because meetme needs a zap channel. |
07:11.43 | sleepy_one | my old cell phone service sucks for voice but data coverage is pretty good coast to coast worked cross-country |
07:11.48 | samyl | I know but i've zaptel module loaded |
07:11.52 | samyl | with ztdummy |
07:12.03 | samyl | but i don't have a digium card |
07:12.18 | samyl | i use ztdummy as timer for meetme |
07:12.35 | sleepy_one | try running ztcfg -vvvvvvvvvvvvvvvvvv |
07:12.46 | sleepy_one | modprobe zaptel |
07:12.52 | sleepy_one | modprobe ztdummy |
07:12.57 | sleepy_one | ztcfg -vvvvvvvvvvvvvv |
07:13.00 | sleepy_one | safe_asterisk |
07:13.15 | samyl | Zaptel Configuration |
07:13.15 | samyl | ====================== |
07:13.15 | samyl | Channel map: |
07:13.15 | samyl | Channel 01: FXS Kewlstart (Default) (Slaves: 01) |
07:13.15 | samyl | 1 channels configured. |
07:13.16 | samyl | ZT_CHANCONFIG failed on channel 1: No such device or address (6) |
07:13.32 | sleepy_one | you might have to run modprove ztdummy before zaptel I forget |
07:13.55 | sleepy_one | is zaptel there when you lsmod ? |
07:13.59 | samyl | yes |
07:14.04 | sleepy_one | hmmm |
07:14.15 | sleepy_one | did you configure udev for zap? |
07:14.16 | samyl | wcfxo 12192 0 |
07:14.17 | samyl | ztdummy 3620 0 |
07:14.17 | samyl | zaptel 186756 2 wcfxo,ztdummy |
07:14.22 | samyl | yes |
07:14.30 | sleepy_one | ls -al /dev/zap |
07:14.49 | samyl | crw-rw---- 1 root asterisk 196, 254 2005-10-06 12:59 channel |
07:14.49 | samyl | crw-rw---- 1 root asterisk 196, 0 2005-10-06 12:59 ctl |
07:14.49 | samyl | crw-rw---- 1 root asterisk 196, 255 2005-10-06 12:59 pseudo |
07:14.49 | samyl | crw-rw---- 1 root asterisk 196, 253 2005-10-06 12:59 timer |
07:15.28 | samyl | i must loaded zaptel before ztdummy? |
07:15.40 | sleepy_one | maybe |
07:15.45 | samyl | i can try |
07:15.46 | sleepy_one | I don't remember exactly |
07:15.51 | sleepy_one | rmmod and try |
07:15.57 | sleepy_one | shouldn't hurt anything |
07:16.02 | sleepy_one | try it both ways |
07:16.31 | sleepy_one | oh wait a min |
07:16.53 | sleepy_one | channel 01 doesn't exist because you don't have an FXS card |
07:16.57 | sleepy_one | or FXO car |
07:16.59 | sleepy_one | d |
07:17.00 | sleepy_one | right? |
07:17.39 | samyl | yes, i don't have a digium card |
07:17.51 | sleepy_one | comment that out in zaptel.conf and zapata.conf |
07:18.00 | samyl | i use the zaptel driver for ztdummy |
07:18.05 | samyl | for meetme |
07:18.09 | sleepy_one | then ztcfg -vvvvvvvvvvvv |
07:18.12 | sleepy_one | safe_asterisk |
07:18.19 | djin_ib | but i don't have a digium card <-> wcfxo ?? |
07:18.45 | samyl | I've tried all |
07:18.47 | sleepy_one | just comment out the channels in zaptel.conf and zapata.conf |
07:18.50 | samyl | ok |
07:18.55 | sleepy_one | rmmod |
07:19.05 | sleepy_one | modprobe zaptel and ztdummy again |
07:19.08 | sleepy_one | ztcfg -vvvvvvvvvvvvvvvv |
07:19.11 | sleepy_one | safe_asterisk |
07:19.11 | *** join/#asterisk Gunnar (n=gunnar@bkkb-gw.bitcon.no) |
07:19.15 | sleepy_one | see what it does |
07:20.10 | samyl | I'll try |
07:20.15 | samyl | thank you |
07:20.19 | sleepy_one | yw |
07:20.23 | argos73 | hmmm.. this zyxel wifi sip phone isn't too bad, but the battery life sucks... |
07:20.54 | argos73 | leave it turned on (idle) for 24 hours, and it's dead. |
07:21.03 | sleepy_one | dang :-( |
07:22.01 | *** join/#asterisk gvag11 (n=g@84.254.12.236) |
07:22.04 | sleepy_one | I guess you need to carry more than 1 battery or a charger with you |
07:22.12 | sleepy_one | or both |
07:22.45 | sleepy_one | have you seen those solar chargers? |
07:23.08 | sleepy_one | or those hand-crank emergency radios that can charge cell phones ipods, etc ? |
07:23.15 | argos73 | unfortunately, the designers didn't seem to think of that possibility.. the battery is a real pain to take out - battery cover is difficult to remove, and getting the battery out requires a backhoe. |
07:23.42 | *** join/#asterisk Cresl1n (n=matt@user-24-236-124-147.knology.net) |
07:23.52 | sleepy_one | does it have charging contacts underneath ? |
07:23.53 | argos73 | but it looks like the charger jack is one of those mini-usb connectors |
07:24.14 | argos73 | 5vdc charger |
07:24.34 | argos73 | maybe.. there's two rubber plugs on the bottom |
07:24.38 | sleepy_one | I saw a hand-crank emergency radio / cell phone / battery / ipod charger / usb power thingy |
07:24.53 | argos73 | heh - emergency ipod charger. that's funny |
07:26.10 | gvag11 | hi, does anybody have used Asterisk + Spandsp + TE110P (or any E1 board) for faxing? It works fine? |
07:26.28 | Cresl1n | gvag11: what does it do when you try it? |
07:27.43 | gvag11 | Cresl1n : I am thinking to buy the equipment to implement such a solution and i wanted to know if it works and if yes how much reliable it is. |
07:28.03 | Cresl1n | just make sure you don't get your timing mixed up :-0 |
07:28.17 | argos73 | gvag11: ask me in a week, and i'll let you know how it went |
07:28.56 | sleepy_one | http://www.lillianvernon.com/catalog/product_display.jsp?pdId=6063 |
07:29.08 | sleepy_one | something like this ^ |
07:29.09 | gvag11 | Cresl1n: when you say timing you mean to sychronise the E1 board correct? |
07:29.28 | Cresl1n | gvag11: yeppers |
07:29.39 | *** join/#asterisk struct2 (n=struct@81-17-62-133.dsl.uwadslprovider.nl) |
07:29.45 | gvag11 | argos73: so you are about to implement such a solution with E1/T1? |
07:30.18 | struct2 | I have chicage Micro 330 ISDN Dect phones with a Chicaco Vox 390 Access point, i connected those dects with my octoBri cards |
07:30.18 | gvag11 | cresl1n: so there are no slip frames, right? |
07:30.26 | struct2 | i want to use 3 way call transfers |
07:30.51 | argos73 | gvag11: going to try... want to use PRI with DID to do per-department fax receiving |
07:31.02 | struct2 | when i let the PHONE handle the 3 way call transfer, then it works nicely , i can recieve the call, connect to the 3th party, take the call back or hangup to transfer |
07:31.16 | Cresl1n | gvag11: exactly |
07:31.36 | struct2 | Only thing when the PHONE handles the 3 wall call transfer it uses 2 channels, while these 2 channels are shared with 2 Phones, so one is out of service during transfer |
07:32.11 | struct2 | When i let Asterisk handle the 3 way transfer, then asterisk places the call nicely on hold , and i can call the 3th party |
07:32.16 | struct2 | only i cannot return the call |
07:32.28 | argos73 | speaking of such, I know that fax over voip is flaky.. how about PRI -> Asterisk -> Channel bank -> Fax machine? |
07:32.30 | gvag11 | cresl1n: i was thinking to have two PC with one TE110P each one connected back-to-beck (cpe-network config), do you think this is gone be a problem with sync? |
07:32.33 | struct2 | i alway transfer the call regardless if i try to take it back or hang it up |
07:32.48 | struct2 | Anyone an idea? |
07:33.01 | Cresl1n | gvag11: no, that should be ok |
07:33.18 | gvag11 | cresl1n: Cool, thanks |
07:33.30 | Cresl1n | np :-D |
07:34.30 | gvag11 | cresl1n: Any idea by the way about the traffic load that TE110P can handle? Can i have 30 concurent channels in use? |
07:34.51 | sleepy_one | yes if your CPU can handle the load |
07:34.56 | argos73 | gvag11: if the host machine is up to it, shouldn't be a problem |
07:35.48 | Cresl1n | gvag11: sure, that's usually not too bad |
07:35.55 | gvag11 | argos73: in other words if the CPU has power enough to handle these calls so there is no problem, right? |
07:35.57 | sleepy_one | a really nice server should be able to handle 4 Quad Span T1/E1 cards |
07:36.03 | argos73 | gvag11: yup |
07:36.08 | sleepy_one | yes correct |
07:36.33 | sleepy_one | if you do not have to do serious transcoding you can support hundreds of channels on one server |
07:36.51 | argos73 | gvag11: i have a Pentium 4 with two T1 channels - haven't seen any problems yet |
07:37.52 | gvag11 | what about having more than one wildcard boards (lets say 2 TE110P) in one server, is this a good practice? Is it better to stick with one board in each server? |
07:38.03 | sleepy_one | for example if you do ULAW across the board you should be able to handle a large number of channels, however if you have to go from ULAW or ALAW on the PRI to GSM or something that will be a lot more CPU intensive |
07:39.01 | gvag11 | sleepy_one: Ok i got it... |
07:40.02 | gvag11 | argos73: what about the traffic load, i mean you could have two T1 channels but there might be low traffic load. What happens when both of the T1 are full, did you ever test it? |
07:40.03 | sleepy_one | as long as you don't run out of interrupts and such you should be fine I've been told you can put up to 4 quad cards in a compatible box |
07:40.35 | sleepy_one | however some motherboards cannot support that many cards |
07:41.10 | argos73 | gvag11: number of cards usually doesn't matter as long as you're careful |
07:41.14 | gvag11 | and for me to take care of interrupts i should take care the of the CPU and the motherboard, right? |
07:41.21 | sleepy_one | and you have to make sure you buy the right version of the TExxx cards there's 5v and 3.3v versions |
07:41.39 | sleepy_one | IIRC |
07:41.39 | argos73 | gvag11: tested it at 75% busy without any complaints |
07:41.41 | *** join/#asterisk \PsyKo\ (n=xxxxxxxx@golia.caltanet.it) |
07:42.16 | sleepy_one | yes that's right |
07:43.03 | sleepy_one | found a few boards that will not work with the T100p single span T1/E1 card if it is in the wrong PCI slot |
07:43.09 | shimi | TE110P supports both 3v and 5v |
07:43.14 | argos73 | from my experience, intel motherboards (even the cheap ones) are pretty well-behaved... some other brands can cause weird problems |
07:43.34 | *** join/#asterisk Snake-Eyes (n=blog@203.220.55.70) |
07:44.19 | argos73 | (plus, intel boards usually have more PCI slots than most of the other brands) |
07:44.20 | gvag11 | what about asus motherboards? Did someone knows any problem that can be caused by PCI express (since most of the motherboards comes with it now) |
07:44.34 | sleepy_one | even on asus boards which are very well respected I had PCI slots that didn't like the digium cards so the card hard to be moved |
07:44.58 | sleepy_one | hard = had |
07:45.11 | sleepy_one | and the PCI slot couldn't be used |
07:45.36 | gvag11 | sleeply_one: while on the Intel motherboard everything was fine? |
07:45.47 | sleepy_one | haven't tried intel |
07:46.16 | sleepy_one | I used several AMD boards some didn't have that problem |
07:46.30 | argos73 | gvag11: have an intel board with two x100p cards, one tdm400 card, and one te100p card - no problems |
07:46.36 | sleepy_one | the Via K8T800 based board had a problem |
07:46.43 | sleepy_one | the nForce didn't IIRC |
07:47.18 | *** join/#asterisk nfi|ermes (n=ermsewrk@217.220.121.62) |
07:47.30 | sleepy_one | it sometimes varies from board to board even if the chipset is the same |
07:47.30 | nfi|ermes | hi all |
07:47.49 | argos73 | did have to disable a couple things build onto the board in the bios to free up some interrupts, but who cares about the dumb built-in sound card on an ast server... :) |
07:47.50 | sleepy_one | sometimes the BIOS makes a huge difference |
07:48.03 | gvag11 | i see... So most of the time you have to gamble and pray that it will work, right? |
07:48.07 | sleepy_one | aye I did too |
07:48.21 | sleepy_one | motherboards come with too much crap nowadays |
07:48.55 | sleepy_one | there is no guarantee until you try it for yourself |
07:49.32 | sleepy_one | unless you buy a system guaranteed to work by the vendor but that's expensive |
07:49.35 | argos73 | that's another reason I like intel boards - tons of different versions to select from... even a couple models with no extra crap built in. (geez, I'm starting to sound like an Intel sales rep!!) |
07:50.07 | shimi | the only real problem with intel boards is that they suck in performance. :) |
07:50.35 | sleepy_one | Intel is usually pretty well behaved and pretty stable but slower as shimi said |
07:50.37 | argos73 | some of them do... |
07:50.43 | argos73 | not all |
07:50.46 | *** join/#asterisk tobiasWolf (n=konversa@195.162.255.10) |
07:50.59 | sleepy_one | but now Intel is out of the motherboard biz from what I heard |
07:51.08 | sleepy_one | I personally prefer AMD |
07:51.36 | shimi | amd doesn't make motherboards, methinks |
07:51.38 | sleepy_one | AMD64 K8 family preferably with nForce chipsets |
07:51.57 | shimi | i got an amd64 with nf4, and I am very happy. amazing machine |
07:52.19 | sleepy_one | shimi, yes indeed I meant AMD64 K8 compatible motherboard |
07:52.44 | gvag11 | there is should be a forum or something that user can report the hardware compatibility issues since the compatibility notes from Digium is really nothing.... |
07:52.48 | sleepy_one | the nF4 is a great chipset |
07:52.58 | sleepy_one | I take that back! |
07:53.08 | sleepy_one | it's an AWESOME chipset!!!! :-D |
07:53.25 | shimi | gigabyte recently made a motherboard that supports SLI, and placing the nf4 chipset in it too, resulting in the possibility to put 4 (!!!) graphic cards in one motherboar |
07:53.27 | shimi | d |
07:53.47 | sleepy_one | only thing is the new nF4 and K8T890 chipsets have PCI express so they reduce the number of normal PCI slots |
07:54.00 | shimi | here's a picture of the beast: http://www.tomshardware.com/motherboard/20051004/images/platform-intro.jpg |
07:54.23 | *** join/#asterisk dmg123 (n=mechanix@mechanix.riscom.net) |
07:54.25 | argos73 | off to snooze... one more day of work.... later |
07:54.29 | sleepy_one | nice :-) |
07:54.32 | sleepy_one | gnite argos |
07:55.04 | gvag11 | bye bye guys... Thanks for the usefull things... |
07:55.09 | *** part/#asterisk gvag11 (n=g@84.254.12.236) |
07:55.18 | sleepy_one | yw |
07:56.25 | *** join/#asterisk [Airwolf] (n=airwolf@attilla.nl) |
07:56.26 | *** join/#asterisk zoa (n=zoa@pirus.securax.be) |
07:56.29 | zoa | hey ho |
07:56.35 | sleepy_one | greetings |
08:01.27 | *** join/#asterisk montag___ (n=montag@195.223.103.50) |
08:02.59 | montag___ | hi, my asterisk have a strange beahviour, when i try to connect with a remote natted sip client i receive the register request (i can see it via tcpdump), but asterisk, with sip debug on, show nothing.....my asterisk box have a public ip, i've set bindaddr=0.0.0.0 in my sip.conf, what's the problem ??? |
08:06.53 | kaldemar | set bindaddr=yourpublicip |
08:07.37 | *** join/#asterisk Snake-Eyes (n=blog@203.220.55.70) |
08:07.56 | montag___ | but if i want to bing my public ip and my ethernet ip too ? |
08:08.03 | sleepy_one | gnite all |
08:08.50 | *** join/#asterisk flenders (n=fserto@61.8.29.101) |
08:09.58 | flenders | hey, does anyone know a VoIP provider in the US that doesn't have a monthly fee? |
08:10.36 | flenders | a sort of prepaid account |
08:15.14 | flenders | MGSsancho, thanks mate |
08:15.21 | MGSsancho | np |
08:16.49 | montag___ | my netstat show a thousands of Recv packets in queue for port 5060, seems that asterisk don't read the buffer..... |
08:18.18 | *** join/#asterisk T0aD (n=toad@epsylon.org) |
08:18.23 | T0aD | asterisk sucks |
08:18.31 | T0aD | im sorry but sip auth is a PAIN IN THE ASS |
08:18.38 | JamesDotCom | oh |
08:18.41 | JamesDotCom | so you're retarded |
08:18.58 | T0aD | do you know what im talking about ? |
08:19.07 | shimi | T0aD, so buy a better solution at $2000 :) |
08:19.19 | T0aD | shimi, they received the patch to fix that |
08:19.20 | shimi | if you can find one, that is. |
08:19.35 | T0aD | but apparently they prefer to make some conf |
08:20.07 | T0aD | cya, dumb protectors of nothing |
08:20.11 | *** part/#asterisk T0aD (n=toad@epsylon.org) |
08:20.15 | MGSsancho | what? |
08:20.18 | shimi | heh |
08:20.22 | *** join/#asterisk Tili (i=Tili@202-133-67-153-dialup.sat.net.pk) |
08:20.25 | MGSsancho | hahahah |
08:20.32 | JamesDotCom | hahaha |
08:20.39 | JamesDotCom | what a weird unit |
08:20.47 | *** join/#asterisk lehel (n=asd@82.79.20.17) |
08:20.52 | MGSsancho | hes just really pissed of |
08:20.59 | MGSsancho | didnt een ask for help |
08:21.09 | lehel | hello |
08:21.32 | shimi | maybe he wants a better price... after all, * is very costy. ;) |
08:26.34 | iDunno | damned expensive it is ;) |
08:27.37 | shimi | The CD I burned the software on costed about 50 cents... I think I'm going to bankrupt because of * |
08:32.36 | protien | can anyone help me to troubleshoot this error |
08:32.37 | protien | Oct 7 17:59:28 NOTICE[20804]: chan_sip.c:9385 handle_response_invite: Failed to authenticate on INVITE to '"3477274087" <sip:3477274087@203.173.26.187>;tag=as752065d4' |
08:32.43 | protien | it happens when i make an outgoing call |
08:33.46 | Delvar | well it look sliek it Failed to authenticate on INVITE |
08:33.58 | Delvar | try using the right username/fromuser/password |
08:34.13 | protien | im using the right username/fromuser/password |
08:34.16 | protien | because its registering juse |
08:34.28 | Delvar | check the from user is corect |
08:34.34 | protien | its correct |
08:34.37 | Delvar | hmmm |
08:34.49 | Delvar | get siptrace and pastbin it |
08:34.50 | Delvar | ~pb |
08:34.53 | jbot | extra, extra, read all about it, pastebin is a place to paste your stuff without flooding the channel - try http://pastebin.ca/ |
08:34.58 | protien | siptrace? |
08:35.04 | Delvar | o_0 |
08:35.40 | Delvar | at asterisk cli `sip debug` |
08:35.42 | Delvar | then make teh call |
08:35.48 | Delvar | copy the results to pb |
08:36.55 | fishboy1669 | morning |
08:38.02 | protien | okay delvar |
08:38.16 | protien | http://pastebin.com/385981 |
08:40.57 | Delvar | hmm interesting |
08:41.05 | Delvar | whats your sip.conf look like? |
08:41.43 | protien | http://pastebin.com/385983 |
08:41.50 | protien | thats my [stanaphone] entry |
08:41.56 | lancey | hi guys |
08:42.42 | montag___ | there's some problem with asterisk and kernel 2.6 network ? My receive queue for port 5060 it's full, but asterisk (with sip debug on) shows nothing.... |
08:43.28 | Delvar | protien: type=friend so you can recive incoming calls.. but that besides the point |
08:43.39 | protien | yeah |
08:43.44 | Delvar | protien: config looks ok... |
08:43.57 | protien | thats what i cant work out delvar |
08:44.00 | protien | the only thing i can think of is |
08:44.03 | protien | Oct 7 17:59:28 NOTICE[20804]: chan_sip.c:9385 handle_response_invite: Failed to authenticate on INVITE to '"3477274087" <sip:3477274087@203.173.26.187>;tag=as752065d4' |
08:44.09 | protien | instead of sending 3477274087@203.173.26.187 |
08:44.13 | Delvar | protien: is that number in your dialplan on providers side? |
08:44.23 | protien | i should be sending 3477274087@sip.stanaphone.com |
08:44.29 | Delvar | fromdomain |
08:44.59 | protien | oh there is a fromdomain feature, for the sip.conf? |
08:45.08 | Delvar | yeha |
08:45.12 | protien | lemme try that |
08:45.43 | shimi | anyone here has a GXP-2000 phone ? |
08:45.50 | Delvar | me! |
08:45.56 | shimi | how is it? |
08:46.01 | Delvar | ugly |
08:46.12 | Delvar | works ok though |
08:46.13 | shimi | functionality wise :) |
08:46.31 | Delvar | not realy used it much, im using a snom mainly |
08:46.53 | Delvar | does prety much everything you need form a phone |
08:47.02 | shimi | did you encounter a problem where you hear nothing coming from asterisk ? |
08:47.06 | Delvar | you can like make calls.. recive calls.. |
08:47.24 | Delvar | yeha but that was a hardware fault and got replacement |
08:47.40 | shimi | only from asterisk, I mean. other sounds sound well |
08:47.42 | protien | well that didnt work delvar |
08:48.00 | Delvar | well id say go to there support |
08:49.01 | *** join/#asterisk CleanerX (n=nix@nat-ph3-wh.rz.uni-karlsruhe.de) |
08:52.44 | *** join/#asterisk darkskiez (n=darkskie@194.247.78.146) |
09:01.35 | *** part/#asterisk samyl (n=samyl@194.167.18.244) |
09:03.06 | *** join/#asterisk szer (n=Miranda@217.116.36.22) |
09:03.09 | szer | hi |
09:04.11 | *** join/#asterisk kamuix (n=kamuix@195.78.4.174) |
09:05.21 | *** join/#asterisk Hupe (n=Hupe@iD4CC1376.versanet.de) |
09:09.00 | *** join/#asterisk jonathh (n=asd@host81-154-159-222.range81-154.btcentralplus.com) |
09:09.10 | jonathh | morning! |
09:09.18 | *** join/#asterisk malabar (n=mala@bkkb-gw.bitcon.no) |
09:11.04 | lancey | morning, jonathh |
09:15.17 | jonathh | i like today. |
09:15.19 | jonathh | it is Friday |
09:15.50 | *** join/#asterisk asco (n=just_me@193.173.119.247) |
09:16.02 | lancey | :) |
09:16.16 | lancey | i like tomorrow more :) |
09:16.23 | lancey | AND the day after tomorrow |
09:23.38 | *** part/#asterisk Hupe (n=Hupe@iD4CC1376.versanet.de) |
09:23.41 | Delvar | i lik eeating |
09:24.11 | *** join/#asterisk e3g (i=ee@u15157627.onlinehome-server.com) |
09:24.12 | *** join/#asterisk FABRIZIOxxx (n=FABRIZIO@81-208-26-86.ip.fastwebnet.it) |
09:24.15 | e3g | hi |
09:24.41 | asco | is there some one that will help me (complete newbie) with some questions i have got regarding forwarding a group to a queue, and logon/logoff status indication of agents on cisco 79xx phones |
09:25.12 | FABRIZIOxxx | hello all .. how do i transmit silence when using pcma codec?? i'm having problems with users that are complaining that the phiones do not work but they actually are .. this because on boith sides they hear silence until one of them says something .. |
09:25.24 | e3g | I want to allow _44., in my extensions and want to hangup() at _447., but everynumber goes to _44 even it is 4477 .. |
09:25.46 | Delvar | <FABRIZIOxxx>: disable silance supprestion on the phones |
09:26.00 | lancey | *silence suppresion |
09:26.06 | lancey | or, actually, VAD |
09:26.06 | Delvar | that too |
09:26.09 | lancey | on most phones |
09:26.19 | lancey | Delvar it's the same, i just corrected you |
09:26.25 | Delvar | i know :) |
09:26.25 | lancey | VAD stands for Voice Activity Detection |
09:26.49 | Delvar | i realy should learn to type |
09:26.55 | lancey | :) |
09:26.59 | FABRIZIOxxx | i tried .. but on GXP2000 it seems to work only for g729 protocol |
09:27.00 | lancey | *really :) |
09:27.03 | Delvar | you think after 15 years id be good at it |
09:27.11 | Delvar | :P |
09:27.14 | *** join/#asterisk oej (n=Olle@83.210.106.9) |
09:27.50 | e3g | any help ??? |
09:28.15 | e3g | I want to allow _44., in my extensions and want to hangup() at _447., but everynumber goes to _44 even if number is 4477123345 .. |
09:28.20 | Delvar | e3g: |
09:28.44 | *** join/#asterisk Guus_ (n=g@139.63.241.162) |
09:28.55 | Delvar | e3g: its a problem with asterisk dial plan, it matches on the _44 before it matches teh _447.. you need to add teh _44 to a context and include it in the main contact below the _447 |
09:29.29 | lancey | hmz |
09:29.39 | lancey | i've had similar setups with latest CVS |
09:29.47 | e3g | ok.... |
09:29.49 | lancey | overriding just one of the priorities |
09:29.53 | lancey | and it works like a charm |
09:29.54 | e3g | I check it out |
09:29.54 | e3g | thanks |
09:30.30 | Guus_ | Hiya. We're using an asterisk with zaptel setup. Does anyone has an idea why the time between phone-pickups changes everytime someone calls in? (sometimes, asterisk kicks in after one ring, sometimes it takes forever before the line gets picked up) |
09:31.16 | oej | Up in the air above Norway on my way to Astricon. On the plane! |
09:32.37 | lancey | :) |
09:32.52 | iDunno | e3g: erm, is the . representing a bunch of numbers, or just 1 in your case? because you could use _44X and _447X |
09:33.09 | Delvar | iDunno: he left already |
09:33.14 | iDunno | oh, must remember to read the rest of the sentence too. |
09:33.19 | iDunno | Delvar: oh yes. |
09:33.39 | iDunno | I'm hoping that the afternoon is better ;) |
09:33.45 | Delvar | :) |
09:35.39 | *** join/#asterisk coppice (n=chatzill@213.168.17.210.dyn.pacific.net.hk) |
09:36.21 | *** join/#asterisk Attila_Kovacs (n=kovacsat@dsl51B790CC.pool.t-online.hu) |
09:36.31 | *** join/#asterisk Szolke (n=Szolke@217.116.36.22) |
09:37.44 | *** part/#asterisk Szolke (n=Szolke@217.116.36.22) |
09:39.57 | *** join/#asterisk samyl (n=samyl@194.167.18.244) |
09:39.59 | samyl | hi |
09:40.16 | samyl | i've a little question about the timer for meetme |
09:40.54 | samyl | i'd like to know if it is possible to use a card FXS Digium for the timer |
09:41.14 | samyl | or if i must use FXO card |
09:42.20 | *** join/#asterisk Szolke (n=Szolke@217.116.36.22) |
09:42.42 | Szolke | hi |
09:45.20 | *** join/#asterisk Hupe (n=Hupe@iD4CC1C7C.versanet.de) |
09:52.06 | *** join/#asterisk ful|work (n=fulgas@213.58.130.46) |
09:52.14 | Delvar | samyl: just a card on its own should do it |
09:54.32 | *** join/#asterisk Tangent (n=Arc_Tang@connerdata.plus.com) |
09:58.02 | samyl | Delvar : so, I can use FXS for the timer? |
09:58.32 | Delvar | samyl: i think so... as i understand it it takes timing off teh card not eh module |
09:58.45 | Delvar | but i oculd be wrong :) |
09:58.54 | Delvar | you could always use ztdummy :) |
10:07.16 | szer | hi everyone |
10:08.48 | szer | I've got some problem with chanspy. When we try to listen a conversation between A and B we can hear only inaudible noises at the end connected to the chan-spy. This is only a problem if either A or B, or the spy is softphone (IAX2 based). If I switch on the jitterbuffer it's solve the problem, but the spying party got a 500ms delay |
10:19.51 | *** join/#asterisk asco (n=just_me@193.173.119.247) |
10:20.49 | *** join/#asterisk montag___ (n=montag@195.223.103.50) |
10:21.29 | montag___ | my asterisk have bindaddr=0.0.0.0 in sip.conf, the system receive upd sip packets (checked via tcpdump..) but asterisk wint sip debug enable show no output....any tips ? |
10:21.58 | fishboy1669 | hi guys, anyone know if there is a command in extentions.conf to detect if a phone has a call on it. |
10:22.25 | lancey`away | fishboy1669: ChanAvail |
10:22.52 | *** join/#asterisk Simon- (i=byte@2001:4bd0:1000:1:2e0:4cff:feed:1cfb) |
10:24.09 | asco | can someone tell me why i can't use queue's is a ring group? i am new to asterisk and thi is one of the functions it has got to have |
10:25.07 | Szolke | Hi all, I would like to ask the following questions of Aastra 9133i phones. We would like to use these phone under Asterisk... is it possible when there is incoming call to a group and it is not answered, then a led can be blinking showing that there is an incoming call in that group. So anyone could answer that call even from a different group. |
10:25.17 | lancey | asco: what are you trying to do? |
10:25.21 | lancey | ring several lines at once? |
10:25.30 | kaldemar | montag___: what does 'netstat -nl' say? are your interfaces listening to port 5060? |
10:26.12 | montag___ | udp 39292 0 0.0.0.0:5060 0.0.0.0:* |
10:26.25 | montag___ | why this value in Recv queue ? |
10:27.50 | fishboy1669 | lancey i have looked at ChanAvail it seems to tell me there is a phone plugged in but not wethere there is a call in process to it |
10:28.57 | lancey | If the option 's' is specified (state), will consider channel unavailable |
10:28.57 | lancey | when the channel is in use at all, even if it can take another call. |
10:29.10 | asco | sorry lancey i was held up biy someone |
10:29.31 | lancey | fishboy1669, actually, it is ChanIsAvail, my mistake |
10:29.43 | lancey | when used with the "s" option, it does exactly what you need |
10:29.55 | lancey | though i have not tested or used that... but it should work :) |
10:30.23 | asco | i am trying to call a queue via en ring group so that i can add an caller id to identify the alled number |
10:30.48 | *** join/#asterisk queuetue (n=queuetue@toronto-HSE-ppp4122670.sympatico.ca) |
10:31.01 | queuetue | Hi. I am trying to create a digital receptionist with a stock AAH 1.5 ... I Click the menu entry, it asks for my extension, I give it, and then it tells me to hit *77 to record. I do, and as soon as I hit the second 7, I get back to dialtone. Nothing shows in the asterisk console .. is my sipura swallowing the *77? |
10:31.07 | lancey | i hardly understood anything, sorry, asco |
10:31.44 | asco | the idea is: caller -> group-> queue |
10:32.43 | asco | i want to use groups to channel in to one queue |
10:33.57 | Delvar | queuetue: check the dialplan on teh sipura default has a *XX, you should remove it or change it to *XXX |
10:34.20 | Delvar | i think its an X anyway... dont take my word for it :) |
10:34.28 | fishboy1669 | cheers lancey i will have a play and let u know how i get on |
10:34.46 | lancey | don't beat me if it doesn't work :") |
10:36.06 | queuetue | Delvar: grep does not find anything in /etc/asterisk that has *X in it ... |
10:36.27 | Delvar | queuetue: not in asterisk in the sipura |
10:36.45 | queuetue | Delvar: Ahhh. |
10:38.15 | *** join/#asterisk without (n=dean_dav@CPE-60-226-183-127.qld.bigpond.net.au) |
10:39.10 | nfi|ermes | anyone knows a good printable ansd updated documentation for asterisk ? |
10:39.44 | queuetue | Delvar: I changed *XX to *XXX and now, I get a 2-second pause ... and a return to dialtone. What should the dialplan setting be? |
10:40.10 | Delvar | in that case there is a problem with a@h |
10:40.28 | Delvar | iv not used it so i cant help :( |
10:40.33 | asco | lancey: i hope i can make istclear by showing the route a call has to take. |
10:40.43 | queuetue | Delvar: I'm not positive. since asterisk still does not put any SI debug information out in the console... |
10:40.56 | queuetue | s/SI/SIP/ |
10:40.58 | lancey | asco, maybe, i'm also not very familiar with queues, this could be the problem too |
10:41.01 | lancey | :) |
10:41.15 | Delvar | queuetue: can you dial other numbers? |
10:41.21 | asco | ok sorry |
10:41.29 | queuetue | Delvar: Yes. |
10:41.40 | asco | do you know how a caller is is passed trough |
10:41.41 | Delvar | can you dial other *XX numbers? |
10:41.47 | lancey | asco: you read about queues at voip-info.org ? |
10:41.49 | asco | caller ID i mean |
10:42.03 | queuetue | Delvar: *98 wrks for vm, I can dial other extensions and outgoing numbers. |
10:42.43 | CleanerX | anyone capable of calling enum numbers? |
10:42.54 | asco | i'llread it rightaway |
10:42.56 | Delvar | Hmmhesays: look in your sipura config for a *77 code.. i cant remember one but there might be, you can do it quicly by 'view source' on teh page and doing a find |
10:43.01 | CleanerX | would like to test if everything works as expected |
10:43.03 | Delvar | eh! |
10:43.06 | Delvar | sorry |
10:43.25 | Delvar | i meant Hmm queuetue: |
10:43.33 | *** join/#asterisk mmmToop (n=chatzill@196.31.11.194) |
10:43.35 | lancey | asco, each channel has its CallerID attached to it |
10:43.38 | lancey | so it's there |
10:43.44 | lancey | routed wherever you route the call |
10:43.53 | lancey | if the equipment is capable, of course |
10:44.15 | Delvar | exept when using SIP and fromuser.. as it overwrites callerid number with your sip username :) |
10:45.07 | asco | i know but a group can adb some text/number to the caller ID but strips is as soon as it leaves the group an goes to the no answer destination |
10:45.15 | *** join/#asterisk _omer (i=p@203.215.180.250) |
10:45.18 | _omer | hi |
10:45.27 | queuetue | Delvar: Yes, "Block ANC code" is set to *77... Gah. :) |
10:45.28 | lancey | Delvar, sometimes SIP gets fromuser to show "unknown" |
10:45.41 | lancey | i'm now playing with that |
10:45.53 | lancey | http://bugs.digium.com/view.php?id=0005405 |
10:45.58 | Delvar | queuetue: whos your daddy :) |
10:46.09 | queuetue | Delvar's my daddy. |
10:46.14 | Delvar | heheh |
10:46.25 | _omer | How to do Proxy authentication in asterisk ???? |
10:46.33 | _omer | SIP Proxy |
10:47.12 | Szolke | I would like to ask the following questions of Aastra 9133i phones. We would like to use these phone under Asterisk... is it possible when there is incoming call to a group and it is not answered, then a led can be blinking showing that there is an incoming call in that group. So anyone could answer that call even from a different group. |
10:47.18 | queuetue | Delvar: Can I clear these all out? |
10:47.34 | Delvar | queuetue: yeah unless you need them :) |
10:47.48 | *** join/#asterisk Spacebar (n=stingray@stingr.net) |
10:50.28 | *** join/#asterisk jeffik (n=Jeff@CPE0020ed8494b8-CM0012c999ca4e.cpe.net.cable.rogers.com) |
10:54.30 | *** join/#asterisk A-Tuin|work (n=A-Tuin@nat.office.legend.net.uk) |
10:55.12 | fishboy1669 | yay lancey u are a genius |
10:55.53 | lancey | no, i'm not |
10:56.00 | lancey | show application is my friend :) |
10:56.03 | fishboy1669 | well u solved my problem |
10:56.15 | queuetue | Ok, I can now see the *77 call coming into the console, but the call is terminating with a "403 Forbidden" error... |
10:56.23 | fishboy1669 | was battling with that all yest |
10:56.46 | fishboy1669 | had done some school boy errors though which meant that it wouldnt work no matter what |
10:56.48 | fishboy1669 | he he |
10:56.56 | fishboy1669 | but not its fine |
10:57.30 | fishboy1669 | sweeeet |
10:57.35 | fishboy1669 | so chuffed here |
10:57.39 | fishboy1669 | big jump forward |
10:58.01 | lancey | :) |
11:03.02 | *** join/#asterisk ful|work (n=fulgas@213.58.130.46) |
11:05.15 | CleanerX | anyone capable of calling enum numbers? |
11:05.41 | *** part/#asterisk Proteque (n=gjorans@213.184.199.245) |
11:08.30 | asco | lancey: thank you for your help, ill try to find a work around. |
11:08.53 | *** part/#asterisk asco (n=just_me@193.173.119.247) |
11:17.29 | *** join/#asterisk cpm (n=Chip@pdpc/supporter/sustaining/cpm) |
11:17.58 | *** join/#asterisk w14 (n=asterisk@62.140.193.212) |
11:30.56 | *** join/#asterisk hadi57 (n=al_moghr@83.136.8.206) |
11:31.41 | *** join/#asterisk christo (n=chris@195.82.114.14) |
11:32.16 | *** join/#asterisk Zeeek (n=icechat5@pdpc/supporter/active/Zeeek) |
11:32.33 | Zeeek | ~seen kram |
11:32.36 | jbot | kram <n=mark@pdpc/sponsor/digium/kram> was last seen on IRC in channel #asterisk, 4d 18h 36m 46s ago, saying: 'cool'. |
11:33.06 | christo | mark spencer was out getting sloshed with my buddies last night :) |
11:33.21 | christo | too bad I missed the sesh |
11:33.54 | christo | anyway, I'm starving... biab |
11:34.40 | Zeeek | where? |
11:35.10 | *** part/#asterisk cpm (n=Chip@pdpc/supporter/sustaining/cpm) |
11:37.17 | *** join/#asterisk coppice (n=chatzill@175.203.17.210.dyn.pacific.net.hk) |
11:37.55 | johnm | christo: heh, ditto funilly enough. |
11:38.14 | johnm | christo: I was gonna go to the LWE, but never got the chance in the end. |
11:39.51 | *** join/#asterisk oej (n=Olle@83.210.106.1) |
11:43.19 | *** join/#asterisk _omer (i=p@203.215.180.250) |
11:43.45 | _omer | Hi,,,,,is there any Asterisk Based VoIP Service Provider?? Pc 2 Phone, Phone 2 Phone etc????? |
11:48.19 | *** join/#asterisk Szolke (n=Szolke@217.116.36.22) |
11:48.45 | Zeeek | this was London last night? |
11:49.01 | Zeeek | _omer there are several in France |
11:50.07 | _omer | any url ? |
11:51.23 | lancey`away | byez all |
11:53.36 | *** join/#asterisk kahuna_ (n=booger@209-254-56-194.ip.mcleodusa.net) |
11:54.44 | johnm | Zeeek: yeah |
11:54.51 | queuetue | When I dial 411 or #, I get a fast busy - does that indicate the AGI directory script has failed? How would I debug this? |
11:56.14 | Zeeek | Tonight it's Paris |
11:58.04 | queuetue | Gah, spirura was eating it again. |
11:58.18 | *** join/#asterisk Tili (i=Tili@202-133-67-71-dialup.sat.net.pk) |
11:58.55 | kahuna_ | Hi. I want to increase the time between digits that a user dials. I have this: exten => _9.,1,DigitTimeout(10) exten => _9.,2,Dial(Zap/g1/${EXTEN:1}) exten => _9.,3,Congestion |
11:59.16 | kahuna_ | but the inter digit timeout is not 10 like I would like it to be. |
12:00.25 | *** join/#asterisk JunK-Y (n=junky@Toronto-HSE-ppp3781375.sympatico.ca) |
12:04.45 | *** part/#asterisk _omer (i=p@203.215.180.250) |
12:11.31 | Tili | kahuna_: use responsetimeout |
12:12.48 | Tili | but digit timeout should work |
12:14.08 | *** join/#asterisk RoyK (n=roy@80.239.107.70) |
12:17.25 | *** join/#asterisk bjohnson (n=bjohnson@jecinc.tor.istop.com) |
12:23.49 | kahuna_ | The only thing I can think of is that * thinks it has a complete number. |
12:25.05 | kahuna_ | Does responsetimeout and digittimeout work for outbound when a user picks up the hook to dial out? All of the examples I've seen are for inbound contexts after Answer() or something similar |
12:34.37 | *** join/#asterisk apardo (n=apardo@200.Red-83-50-234.dynamicIP.rima-tde.net) |
12:37.07 | christo | Zeek - yeah in London |
12:37.30 | Zeeek | having dinner with him tonight in Paris |
12:37.45 | Zeeek | so he must be on his way |
12:37.49 | Zeeek | hangover time |
12:37.52 | christo | yes he is :) |
12:39.14 | christo | Zeeek - ask Mark if he'd like a double Jameson Whiskey and if he likes we can sponge one up off the pavement in High Street Kensington (where he was drinking last night) |
12:39.32 | christo | and where he manage to vomit it back up :) rofl |
12:40.18 | christo | that's if he managed to get the train.. |
12:41.43 | Zeeek | hahaha |
12:41.54 | *** join/#asterisk Ariel_ (n=Ariel@adsl-068-157-125-248.sip.mia.bellsouth.net) |
12:41.56 | Zeeek | amateau drinkers |
12:42.13 | christo | aye |
12:42.14 | Ariel_ | Morning everyone |
12:42.26 | christo | morning Ariel_ |
12:42.30 | Zeeek | I can't get the BT102 to hear me calling! |
12:42.31 | Zeeek | <PROTECTED> |
12:42.31 | Zeeek | Oct 7 14:41:26 NOTICE[21579]: chan_sip.c:8117 sip_poke_noanswer: Peer '2000' is now UNREACHABLE! |
12:42.51 | Zeeek | drops out right away |
12:42.52 | Ariel_ | Zeeek, network issue |
12:43.02 | Zeeek | both on same side of NAT roter |
12:43.07 | christo | well it's registering at least |
12:43.20 | Zeeek | although I moved it to an outside network and it still didn't work |
12:43.40 | Ariel_ | Zeeek, have you upgraded the firmware of the BT102? |
12:43.52 | Zeeek | actually I tried without registering and a fixed ip but it never beocmes reachable. It worked for months before |
12:43.59 | Zeeek | Yes and I suspect the upgrade! |
12:44.12 | Zeeek | I'm trying to revert but I don'thave the older files here |
12:44.39 | christo | Granstreams are poop anyway afaics |
12:44.53 | christo | altho mine are about 18 mths old now.. |
12:45.08 | Zeeek | mine have all worked fine for many months |
12:45.26 | Ariel_ | I only have 1.0.6.2 |
12:45.35 | *** join/#asterisk |dennis| (i=dennis@200.32.215.84) |
12:45.37 | Zeeek | I even kinda like them, although my newer polycom takes things to a higher level |
12:45.48 | Zeeek | I was on 1.0.5.12! that was working |
12:46.03 | Zeeek | yeah they're great |
12:46.14 | *** join/#asterisk Simon- (i=byte@proxima.arlott.org.uk) |
12:47.00 | Zeeek | I do that with the IAXy |
12:47.06 | christo | aye |
12:47.11 | Zeeek | but the cordless are old and don't ring, no CID |
12:48.15 | Ariel_ | I have a Sipura 2002 with a panasonic 2.4gh 2 line phone in my office. Other offices here use the Polycoms and some sipura 841's. |
12:49.14 | Ariel_ | I even have a head set for the Panasonic.....(love to walk around the office and still get my calls). |
12:52.36 | *** join/#asterisk |cleric| (n=dacleric@p548293C8.dip0.t-ipconnect.de) |
12:57.07 | Katty | mew. |
12:57.23 | iDunno | mew2! |
12:57.25 | christo | ggrrrr |
12:57.50 | *** join/#asterisk popvoxdave (i=user@dave2.toad.net) |
12:58.01 | Katty | so, does anyone besides me know what today is? |
12:58.06 | christo | I keep getting 'no channel type registered for IAX, but I have a good IAX.conf.. |
12:58.10 | christo | friday |
12:58.12 | christo | poets day |
12:58.20 | Katty | because it's a very important day in katland. |
12:58.22 | Katty | twisted[asteria]: i know you know. |
12:58.31 | Lathos42 | Hmm.. Talk like a Pirate day was September 19th, so that can't be it |
12:58.40 | tzanger | Katty: friday? |
12:58.42 | Katty | Lathos42: no, it's October 7th. |
12:58.45 | iDunno | it's the 7th Oct... |
12:58.46 | christo | Fire christo from his job day - according to my boss |
12:58.53 | Lathos42 | Katty: The day you first installed *? |
12:58.54 | Katty | tzanger: yes it's a friday, but it's much more important than just friday. |
12:58.59 | iDunno | is it Katty's birthday? aniversary? first install of *? |
12:59.10 | tzanger | Katty: it's Friday the 7th? |
12:59.11 | Katty | I'm all grown up today. |
12:59.18 | tzanger | it's your bday? |
12:59.19 | iDunno | issa birthday @:) |
12:59.20 | Katty | yes. |
12:59.23 | Katty | It's my 21st. |
12:59.23 | tzanger | how old now? |
12:59.28 | tzanger | nice |
12:59.32 | Lathos42 | Well Happy Birthday to you |
12:59.32 | tzanger | happy birthday, katy |
12:59.33 | iDunno | ahh - youngster :) |
12:59.34 | tzanger | er katty |
12:59.36 | christo | happy birthday Katty |
12:59.40 | Katty | thanks :> |
12:59.42 | Ariel_ | happy Birthday to Katty |
12:59.45 | christo | 21?? |
12:59.55 | Zeeek | happy happy kappy |
12:59.59 | protien | wow happy bday katty |
12:59.59 | iDunno | # happy birthday to you, happy birthday to you, happy birthday deaaaaaaar Katty, happy birthday to you. |
13:00.01 | protien | it was mine 2 days ago |
13:00.04 | protien | it must be 21st month |
13:00.09 | *** join/#asterisk SpaceBass (n=sp@c-24-125-184-203.hsd1.va.comcast.net) |
13:00.10 | SpaceBass | hey folks |
13:00.12 | tzanger | heh |
13:00.13 | Zeeek | sip show peers |
13:00.16 | tzanger | I turn 30 next may :-) |
13:00.20 | Zeeek | sip show beers |
13:00.23 | protien | the big 3-0 |
13:00.23 | mmlj4 | dip show peers |
13:00.24 | tzanger | Zeeek: hahaha |
13:00.45 | Katty | Normally work orders a cake for people's birthdays. Since I don't eat cake (cause of milk, eggs, etc) I thought they'd have something else. ..but they didn't. And no one has said a word.......i'm a little sad :< |
13:00.48 | protien | hah if its your 30th maybe it should be sip slow beers |
13:01.14 | iDunno | Katty: go stab them all with sharpened pencils! |
13:01.24 | Katty | iDunno: shan't |
13:01.25 | protien | im recieving a call but when it comes in im getting |
13:01.26 | protien | Oct 7 22:25:51 NOTICE[24120]: chan_iax2.c:6690 socket_read: Rejected connect attempt from 204.13.233.114, who was trying to reach 's@' |
13:01.33 | protien | how can i recieve that call? |
13:01.44 | tzanger | protien: :-) nah I don't drink much beer. rye and coke, gin and ginger ale... oh yeah |
13:01.55 | Zeeek | no rum? |
13:01.57 | protien | yeah rye and coke ive got one in my hand right now ;) |
13:02.02 | tzanger | Zeeek: no I'm not a fan of rum at all |
13:02.04 | *** part/#asterisk popvoxdave (i=user@dave2.toad.net) |
13:02.10 | SpaceBass | someone got a second to look at a config for me... I'm racking my brain here |
13:02.11 | SpaceBass | http://pastebin.ca/24843 |
13:02.15 | Zeeek | rum+pineapple mmmmmmm |
13:02.16 | iDunno | hmm. dark rum)++ |
13:02.21 | protien | rum and pineapple is nice? |
13:02.26 | Zeeek | great |
13:02.29 | SpaceBass | http://pastebin.ca/24844 |
13:02.31 | tzanger | gin, rye, tequila, vodka... no problem |
13:02.38 | tzanger | beer is alright |
13:02.39 | Zeeek | especially rum, sparkling pineapple juice |
13:02.45 | tzanger | I prefer the ales though |
13:02.52 | SpaceBass | basically, randomlly, calls on zap/1 are getting into the zap/2 context |
13:02.59 | Zeeek | I'll have what everyone else is having |
13:03.04 | iDunno | good real ales are the way forwards :) |
13:03.06 | tzanger | SpaceBass: wow that I find VERY hard to believe |
13:03.46 | tzanger | SpaceBass: I've been using * for close to 2.5 years now and it's never "jumped context" on me |
13:03.56 | SpaceBass | tzanger i'm sure its something in my configs |
13:04.02 | cpatry | sup? |
13:04.05 | SpaceBass | tzafrir doubt its a bug or anything... |
13:04.05 | Katty | my age! |
13:04.13 | tzanger | Katty: that sucks ass. we have to bring donuts in on our birthday |
13:04.13 | cpatry | you're 21 !!!!!!! |
13:04.15 | cpatry | :) |
13:04.18 | Katty | cpatry: yes :> |
13:04.18 | SpaceBass | tzafrir but I've been looking at these for weeks and cannot find it |
13:04.25 | cpatry | so now we wont have to drink dasani :) |
13:04.35 | Katty | cpatry: but i like dasani |
13:04.38 | Katty | cpatry: you silly rabbit. |
13:04.38 | tzanger | SpaceBass: :-) That is good for the soul now and again |
13:04.48 | Katty | cpatry: but not you can bring your alochol in ;) |
13:04.54 | iDunno | hmmm. rabbit in red wine sauce. |
13:05.01 | Katty | cpatry: but not drink it...cause you might be a french canadian terrorist. |
13:05.02 | tzanger | nice |
13:05.02 | SpaceBass | arrruuugggggggg I wish zapata.conf was a person so I could hit it in the face |
13:05.16 | cpatry | Katty: they sucks! |
13:05.38 | Katty | cpatry: indeed. you poor thing. |
13:05.48 | Katty | cpatry: should have given it to matt ;) |
13:05.53 | tzanger | wow there's 625k files on my hard drive and it's still going |
13:06.02 | Ariel_ | SpaceBass, could you might have rollover on those lines from the phone co? |
13:06.23 | tzanger | 700k |
13:06.25 | cpatry | i already paid him vodka, thats enuf :) |
13:06.30 | SpaceBass | Ariel_ no chance... |
13:06.34 | Katty | cpatry: k (= |
13:06.35 | tzanger | mind you I have a few development environments on here so that makes some sense |
13:06.53 | Ariel_ | SpaceBass, IRQ sharing? |
13:07.02 | *** join/#asterisk IPmonger (n=ipmonger@pcp0010577106pcs.coatsv01.pa.comcast.net) |
13:07.32 | Katty | cpatry: what do /normal/ people do on their birthdays? |
13:07.34 | SpaceBass | Ariel_ maybe, but don't think so... cli looks right, zap/2 picks up and throws the call into the zap/2 context |
13:07.37 | Hmmhesays | oh the roosters rocked last night, what a grand time |
13:07.38 | *** join/#asterisk firestorm-voip (n=firestor@62-181-86-226.skbbip.com) |
13:07.38 | tzanger | 800k |
13:07.43 | Katty | Hmmhesays: come visit for my birthday. |
13:07.47 | SpaceBass | Ariel_ and it worked fine until I upgraded to aah 2.0b or what ever |
13:08.01 | Ariel_ | ahh |
13:08.03 | tzanger | Katty: where do you live again? |
13:08.12 | Katty | tzanger: moo ssouri |
13:08.26 | Hmmhesays | Katty: probably not going to happen, but if you give me your addy i'll send a card |
13:08.38 | tzanger | Katty: that's right... Manxpower could have stayed with you :-) |
13:08.49 | Katty | Hmmhesays: i don't want a card you goofball :P |
13:09.01 | Hmmhesays | fine |
13:09.14 | Katty | i'd rather have a hug anyway (= |
13:09.25 | shimi | 00:0a.0 Network controller: Unknown device e159:0001 < can this be the digium card ? |
13:10.12 | Simon- | possibly |
13:10.16 | *** part/#asterisk firestorm-voip (n=firestor@62-181-86-226.skbbip.com) |
13:10.27 | Simon- | google says: Class 0780: e159:0001 Communication controller: Tiger Jet Network Inc. Tiger3XX Modem/ISDN interface |
13:10.34 | Hmmhesays | oh wow, foxnews had a wonderfully spot on article against MADD yesterday |
13:10.36 | SpaceBass | Is there a way to have a zap channel ignore a 1/2 ring? |
13:10.40 | shimi | <PROTECTED> |
13:10.45 | Hmmhesays | s/against/about |
13:10.49 | shimi | is from /proc/pci |
13:11.07 | tzanger | SpaceBass: callwaiting=asreceived? |
13:11.26 | shimi | how can I "communicate" with the card to check if it works before trying to mess with asterisk on it? :) |
13:11.43 | SpaceBass | tzafrir i'll try that... whats happening is that I've forward my zap/1 to my BV account (temp fix) but I still get a 1/2 ring on incoming calls which triggers zap/1 to pick up |
13:11.45 | tzanger | shimi: asterisk is what communicates with it. |
13:12.00 | shimi | but anything in /proc, etc ? |
13:12.05 | tzanger | 1.25mil files so far |
13:12.11 | tzanger | shimi: not to communicate with it, no |
13:12.21 | shimi | to see status, etc ? |
13:13.55 | tzanger | /proc/zap/ |
13:14.01 | tzanger | but you need the drivers loaded obviously |
13:14.24 | shimi | I have "zaptel" and "wcte11xp" and "wcusb" loaded. is that enough ? |
13:14.30 | *** part/#asterisk samyl (n=samyl@194.167.18.244) |
13:14.58 | Katty | cold :< |
13:15.03 | shimi | oh, I can see /proc/zaptel/1. Guess it's OK :) |
13:15.26 | *** join/#asterisk musical_Duck (n=kvirc@wblv-146-236-254.telkomadsl.co.za) |
13:15.31 | musical_Duck | lo |
13:15.44 | Katty | quack. |
13:15.52 | tzanger | 1572333 files on my hard drive :-) |
13:15.54 | tzanger | filelight is cool |
13:15.57 | musical_Duck | you should go meow :) |
13:16.02 | shimi | I see there what appears to be 31 "WCT1/0" channels |
13:16.09 | tzanger | yup |
13:16.12 | Katty | musical_Duck: actually, i mew. |
13:16.48 | Katty | file: :< |
13:16.53 | Katty | Hmmhesays: :> |
13:16.53 | file[laptop] | :( |
13:16.55 | musical_Duck | say anyone here you tx/rx fax? |
13:17.01 | musical_Duck | use even |
13:17.05 | file[laptop] | this woman says her number isn't working, yet I can't get it to not work |
13:17.36 | musical_Duck | You have the right number? :) |
13:18.03 | Katty | mister file always has the right number |
13:18.11 | file[laptop] | I pulled it up by the account |
13:18.13 | Katty | he practically is the number. |
13:18.21 | *** join/#asterisk jeffik (n=Jeff@CPE0020ed8494b8-CM0012c999ca4e.cpe.net.cable.rogers.com) |
13:18.33 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
13:18.36 | file[laptop] | I've called directly, through a third party toll-free termination provider, and through our own toll-free termination provider |
13:18.40 | file[laptop] | works in every case |
13:18.45 | musical_Duck | My faxes over plain pstn break up for some reason |
13:19.13 | Katty | file: i don't suppose she told you exactly why she thought it wasn't working, did she? |
13:19.25 | file[laptop] | Katty: she said it just did nothing |
13:19.53 | file[laptop] | "That DID is not working, we get nothing when we call it" |
13:19.59 | Katty | file[laptop]: did you talk to her on that line when you called? |
13:20.07 | file[laptop] | no it's an automated system |
13:20.14 | file[laptop] | it's been intermittent since yesterday |
13:20.18 | Connor- | What the frack is going on with asterlink switches today.. they're bouncing up and down like mad. |
13:20.19 | Katty | hmmm. |
13:20.34 | Katty | file[laptop]: it clearly requires hugs. |
13:20.35 | SpaceBass | Ok... profress |
13:20.35 | bjohnson | I see your frack and raise you a frick |
13:20.35 | file[laptop] | bloody hell |
13:21.42 | Katty | file[laptop]: mew? |
13:21.59 | musical_Duck | They stole some of my telco's copper the other day, network was dodgy all day |
13:22.23 | Katty | maybe they needed it for something more important |
13:22.30 | protien | [macro-stdexten]; |
13:22.33 | protien | all these macros, are they needed |
13:22.38 | protien | or can i remove them? |
13:22.45 | file[laptop] | if you use them, they're needed |
13:22.51 | protien | what are they for mate |
13:22.55 | musical_Duck | Nah, was probably just being shiny ;) |
13:23.47 | musical_Duck | So bout them faxs programs, anyone use rx/tx fax? |
13:24.19 | musical_Duck | silent s BTW :) |
13:26.24 | musical_Duck | mmm Friday evening slump heh? |
13:29.09 | Katty | musical_Duck: ((= |
13:29.16 | Hmmhesays | and I never, what to do this again, heartbreaker |
13:29.21 | *** part/#asterisk Zeeek (n=icechat5@pdpc/supporter/active/Zeeek) |
13:32.09 | *** join/#asterisk MikeJ[Laptop] (n=ircatjer@mi.origenfinancial.com) |
13:39.14 | *** join/#asterisk bon (n=bon@fux.wnet.sk) |
13:39.41 | protien | im having problems with incoming calls, outgoing work fine but when i recieve a call on this iax interface i get |
13:39.42 | protien | Oct 7 22:50:58 NOTICE[24120]: chan_iax2.c:6690 socket_read: Rejected connect attempt from 204.13.233.114, who was trying to reach 's@' |
13:39.46 | protien | how can i recieve calls at s@ |
13:40.30 | *** join/#asterisk ret28 (i=rt@82-71-120-246.dsl.in-addr.zen.co.uk) |
13:40.34 | bon | what kind of billing solution wouuld you suggest? |
13:40.39 | bon | or managing solution.. |
13:40.45 | bon | (adding users/extensions/cdr..) |
13:41.45 | bjohnson | protien: I don't think you can use '@' as an exten name |
13:42.06 | musical_Duck | chan_zap.c:4543 zt_write: Cannot handle frames in 2 format <- what does this mean? Was recving fax when this happend |
13:42.10 | protien | hm i dont understand how to recieve these calls then |
13:42.28 | ret28 | This a SIP problem ooi? |
13:42.30 | bjohnson | try getting them to not specify an exten |
13:42.43 | ret28 | (just been dealing with '@'s myself ... ) |
13:42.46 | Katty | Hmmhesays: i have to hand it to work.......they tried. |
13:42.50 | protien | its a public system |
13:42.54 | protien | i cant get them to do anything bjohnson |
13:42.55 | bjohnson | protien: if no exten is specified, * will default to use the 's' exten |
13:43.09 | Katty | Hmmhesays: halfway expected a cake this morning...with milk and eggs in it...but instead, they brought in fruit salad and a brownie recipe of mine. |
13:43.11 | protien | hm |
13:43.18 | protien | how can i forward the s exten to a number |
13:43.18 | bjohnson | ret28: his error said iax |
13:43.25 | bjohnson | goto() |
13:43.27 | Katty | Hmmhesays: sadly, they didn't follow the recipe in such a way that would make it vegan (wrong margarine and the wrong chocolate chips) but they did try. |
13:43.31 | ret28 | Oh ok, this is what I get for leaping in halfway through :) |
13:44.18 | file[laptop] | are you feeling old now? |
13:44.23 | Katty | file[laptop]: not yet. |
13:44.23 | kpettit | Anybody had any luck using t.38 or t.38 for faxing? I haven't seen alot of information out there on how to get it working with asterisk |
13:44.28 | Katty | file[laptop]: sorta depressed actually. |
13:44.35 | *** join/#asterisk marc324 (n=marc3234@206-248-159-4.dsl.teksavvy.com) |
13:44.51 | ret28 | Though for anyone clued up about SIP ... how to get asterisk to stop demanding authentication from guest calls where the From username just happens to match one of my own SIP users? (even though the from domain is entirely different, and not one of my own) |
13:44.58 | file[laptop] | kpettit: coppice did something to make passthrough work, but I dunno if it really works... besides that you should have read tons of posts about how it doesn't work and isn't supported |
13:45.11 | Katty | file[laptop]: mamma called me around 6:30 this morning, but didn't say a thing about me being 21 |
13:45.13 | marc324 | for sending incoming calls to xlite running on a pc... where do I set the ip address of xlite? |
13:45.26 | file[laptop] | Katty: scary |
13:45.34 | kpettit | file[laptop], ugggh sounds wonderfull |
13:45.48 | file[laptop] | marc324: it either registers to your Asterisk, or you specify it's IP in the host entry for the peer |
13:45.48 | bjohnson | how does margerine and chocolate chips (both from plants) get to be non-vegan? |
13:45.49 | Katty | file[laptop]: i didn't expect her to say Happy birthday...she's a JW... but she usually says something along the lines of how does it feel to be $age |
13:45.50 | tzanger | Katty: your mom calls you at 6:30am regularly? |
13:46.03 | ret28 | marc324: In a host=<ip> for a user in sip.conf |
13:46.06 | file[laptop] | it was my birthday yesterday, and today it is Katty's birthday! |
13:46.06 | tzanger | bjohnson: milk chocolate? |
13:46.07 | Katty | tzanger: she was returning my call from yesterday aftenroon |
13:46.09 | kpettit | The impressing I've been getting through readying about problems with fax is t.37 and t.38 will be about the only thing that can work reliably |
13:46.14 | tzanger | Katty: ahh. |
13:46.19 | Katty | bjohnson: milk |
13:46.39 | kpettit | I'm using ulaw right now with a upstream sip provider and it seems to work in most cases, but of course there are some that we just can't get to work |
13:46.58 | file[laptop] | oh beautiful |
13:47.04 | file[laptop] | my copy of the O'Reilly book is out for delivery! |
13:47.26 | bjohnson | (non-vegan) birthday brownie rejects |
13:47.30 | file[laptop] | yay brownies |
13:47.31 | Katty | file[laptop]: hippo birdie two ewe. |
13:47.34 | *** join/#asterisk jimmy_deanPB (n=jhodapp@adsl-70-228-242-126.dsl.ipltin.ameritech.net) |
13:47.44 | Katty | file[laptop]: hippo birdie two ewe. hippo birdie deer ewe. |
13:47.52 | Katty | file[laptop]: hippo birdie two ewe. |
13:49.34 | *** join/#asterisk Godsey (i=lanny@pdpc/supporter/sustaining/Godsey) |
13:50.11 | *** join/#asterisk queuetue (n=queuetue@toronto-HSE-ppp4122670.sympatico.ca) |
13:50.59 | marc324 | in [xlite] section in sip.conf , the line "host=" is what? |
13:51.23 | bjohnson | probably the ip of the host you want to connect to |
13:51.23 | file[laptop] | dynamic if X-Lite registers to your asterisk box, or the ip/host of where the X-Lite is |
13:51.40 | ret28 | Yeah, you could just set host=dynamic, and have X-Lite register |
13:51.52 | ret28 | (then asterisk will do magic to send the call to the SIP/xlite channel) |
13:51.54 | *** join/#asterisk dalabera (n=Dalabera@pmr.pmrtechnologies.com) |
13:51.57 | queuetue | What causes 503, service unavailable errors? The user is on a rt31p2-na, picks up the line, hears the stuttered dialtone (message waiting), hits *98 and gets a fast busy - the logs show "503 service unavailable". Same happens if he tries to dial an extension or dial out. The rt31p2 *is* the firewall, so there should not be a NAT issue, and my firewall here is forwading 5060-5069 and 10000-20000 to asterisk... What could be cau |
13:52.28 | queuetue | (Connections from inside my firewall work just fine...) |
13:52.54 | Katty | anyone hear about that python that ate a 6ft alligator? |
13:53.01 | *** join/#asterisk funxion (n=nunya@mtnuser.icgws.com) |
13:53.23 | ret28 | The one that blew up? Yeah. |
13:53.33 | ret28 | Would make a good motivational poster, "Ambition" |
13:54.24 | Katty | ret28: yes, it would |
13:54.57 | ret28 | Or whatever the opposite of motivational is anyway, considering how it turned out ... |
13:55.17 | Katty | ambition fits nicely |
13:55.19 | *** join/#asterisk FABRIZIOxxx (n=FABRIZIO@81-208-26-86.ip.fastwebnet.it) |
13:56.22 | ret28 | http://despair.com/ambition.html ... hahah, already done ... sort of |
13:57.04 | SpaceBass | the CLI is showing that i have a failure in a context and then it starts over, how can I get more detail? |
13:57.16 | SpaceBass | i've increased to vvvvv but it doesnt tell me more |
13:57.21 | FABRIZIOxxx | hello all .. is it possible to send the called party ID on the display of a SIP phone? |
13:57.22 | SpaceBass | like what line is failing |
13:58.14 | *** join/#asterisk n0where (n=kc@q041140.ppp.asahi-net.or.jp) |
13:59.05 | funxion | what is the way to read the results of a system() to a variable? |
13:59.06 | protien | how do i forward any incoming calls that dont have a defined extension to a certain number? |
13:59.21 | tzanger | Donald Rumsfeld rushes into Bush's office. "Mr. President, I have terrible news. I've just received a report that three Brasillian solders were killed!" |
13:59.26 | tzanger | Bush replied "That's a tragedy!... wait... How many's in a brasillian again?" |
13:59.31 | SpaceBass | lol |
13:59.39 | SpaceBass | heard this the other day |
13:59.45 | tzanger | yeah it's old but funny |
13:59.46 | SpaceBass | s/this/that |
14:00.13 | Ariel_ | So any word on the L3/Cogent problems??? |
14:00.49 | FABRIZIOxxx | protien, you could try with the gotoif() command .. maybe |
14:01.05 | ret28 | protien: Something like "exten => _.,1,Dial(<whatever>)" , make sure it's the last one in the context of valid numbers though. (. is a wildcard). Disclaimer; I think I might be wrong, try it and see. |
14:01.11 | SpaceBass | I'm getting Spawn extension (from-pstn, s, 2) exited non-zero on 'Zap/1-1' |
14:01.24 | SpaceBass | how can I pinpoint which line in the dialplan is causing it? |
14:01.45 | SpaceBass | unfortunatly there are some includes, etc in the context... there is no exten=> s,2... |
14:01.52 | protien | lemme try that ret28 |
14:02.46 | protien | nop |
14:02.47 | protien | Oct 7 23:29:37 NOTICE[24116]: chan_sip.c:10154 handle_request_invite: Failed to authenticate user "2025560000" <sip:2025560000@66.54.140.46>;tag=as12b501b7 |
14:02.53 | protien | still get these |
14:03.02 | ret28 | That's not a dial plan problem ... |
14:03.18 | tzanger | SpaceBass: unfortunately that's not able to be done from my experience |
14:03.33 | tzanger | you need to sprinkle NoOps throughout and follow the breadcrumbs |
14:03.34 | SpaceBass | tzafrir mack to the manual parsing :) |
14:03.37 | ret28 | Is 2025560000 one of your users? |
14:03.46 | protien | its an incoming call |
14:03.54 | protien | its a remote user |
14:03.57 | SpaceBass | tzafrir thanks... always forget about noops |
14:04.08 | tzanger | noops ist WUNDERBAR! |
14:04.18 | *** part/#asterisk Ariel_ (n=Ariel@adsl-068-157-125-248.sip.mia.bellsouth.net) |
14:04.38 | Katty | anyone have a recipe for rum cake? |
14:04.50 | SpaceBass | would a noops in zapata.conf show up when a call comes in on a zap channel |
14:05.00 | SpaceBass | or are they all parsed at startup only? |
14:05.01 | tzanger | Katty: take cake and soak in rum |
14:05.03 | iDunno | take cake mix, add run, cook. |
14:05.08 | tzanger | SpaceBass: no |
14:05.14 | tzanger | SpaceBass: noop is a dialplan command |
14:05.16 | iDunno | or tzanger's way :) |
14:05.22 | Katty | hmm |
14:05.29 | Katty | cake mix is not vegan |
14:05.30 | tzanger | although I don't think soy cake is all that tasty |
14:05.34 | Katty | nor is most cake you buy on the market. |
14:05.52 | Katty | tzanger: soy cake? the non vegan parts of cake are milk and eggs |
14:06.04 | Katty | tzanger: which are easily replaced with corstarch, water, and soy milk |
14:06.04 | tzanger | soy milk and soy eggs, of course |
14:06.09 | Katty | haha soy eggs |
14:06.17 | tzanger | from those soy chickens you were talking about earlier |
14:06.19 | Katty | cornstarch, silly |
14:06.27 | iDunno | heh |
14:06.35 | ret28 | protien: Hmm, does that user have an entry in sip.conf ? Wondering why asterisk is insisting on authenticating it. |
14:06.36 | tzanger | dammit now I have to watch chicken run again |
14:06.38 | tzanger | I love that movie |
14:06.48 | ret28 | (entry as peer, user or friend) |
14:06.48 | SpaceBass | i haven't seen it yet.... good? |
14:06.50 | iDunno | chicken run is good :) |
14:06.59 | tzanger | NOOOOOOOOOOOOOOOO CHICKEN ESCAAAAAAAAAAAPES FROM TWEEDY'S FAAAAAAAAAARM!!! |
14:07.01 | iDunno | SpaceBass: how on earth have you not seen it?! |
14:07.07 | tzanger | I say that to my kids when I put them to bed sometimes :-) |
14:07.27 | SpaceBass | duno... I usually like animated stuff like that |
14:07.32 | tzanger | wallace and gromit rocks |
14:08.44 | tzanger | just don't ask where the soy nuts come from |
14:10.29 | SpaceBass | can I check a caller ID and drop a certian caller right into DISA with out a password? |
14:10.46 | file[laptop] | yes. |
14:10.50 | tzanger | although I also had the kids going around the supermarket saying in low growley voices "mmm.... beef is goooooood..." this summer |
14:11.06 | tzanger | which is rather funny to hear a 4 year old boy in as low a voice as he can muster |
14:11.08 | file[laptop] | exten => 8777811208/5068780147,1,Playback(muffins) |
14:11.18 | tzanger | SpaceBass: yes, but be careful, that's security through obscurity |
14:11.19 | file[laptop] | 8777811208 is the DID, 5068780147 is the callerid number |
14:11.34 | SpaceBass | tzafrir yeah, not toooo worried... home setup |
14:12.06 | SpaceBass | going to Ireland, got an irish sim card and cell, want to call my irish voip DID and have it drop me into DISA so I can get dialtone and make us calls |
14:12.21 | iDunno | file[laptop]: you'd probably want to Answer() it before trying to play back stuff at them. |
14:12.38 | file[laptop] | iDunno: it'll automatically Answer |
14:12.45 | mutilator | whats a good way to get cron to run on the last day of the month every month? |
14:13.00 | file[laptop] | there's an option to disable that too... *G* so you can play back audio as inband session progress |
14:13.11 | marc324 | how do I restrict xlite users from registering? |
14:13.17 | Katty | tzanger: save a chicken, eat an oreo. |
14:13.25 | iDunno | mutilator: can't you just run it on the first day of the month instead? |
14:13.31 | ret28 | marc324: With a username= and a secret= for the [xlite] entry |
14:13.36 | protien | sorry ret28 |
14:13.38 | protien | i pasted you the wrong line |
14:13.39 | protien | Oct 7 23:40:22 NOTICE[24120]: chan_iax2.c:6690 socket_read: Rejected connect attempt from 204.13.233.114, who was trying to reach 's@' |
14:13.40 | file[laptop] | marc324: if they're peer/friend entry in sip.conf is not dynamic, ie not host=dynamic, they can't register.... |
14:13.46 | protien | this is the reject attempt, that im trying to filter |
14:13.50 | mutilator | iDunno: no it has to log the AR at the end of the month |
14:13.56 | protien | this is an incoming call from goiax network |
14:14.04 | tzanger | Katty: I actually really dislike oreos, I never ever understood why people liked them so |
14:14.10 | tzanger | pirate cookies though... now that's good |
14:14.18 | iDunno | mutilator: hmm, that's a bit of a pain then. |
14:14.29 | mutilator | nope found it |
14:14.31 | Katty | http://us.news3.yimg.com/us.i2.yimg.com/p/ap/20051005/capt.mh10310051654.gator_python__mh103.jpg <- why you shouldn't have an alligator for breakfast. |
14:14.33 | ret28 | Hmmm ... I think if that dialplan was wrong though, there'd still be an error unrelated to the incoming channel |
14:14.36 | mutilator | there is a L character |
14:14.52 | marc324 | right now, any xlite can register.... where do I specify the password in sip.conf? |
14:14.52 | mutilator | * * L * * |
14:14.54 | SpaceBass | arrruuuggg why are these calls in the wrong context! I hate zapata.conf and all its friends |
14:14.57 | ret28 | Still, I'm not certain, I've only been playing with asterisk for a few days :) |
14:14.58 | mutilator | last day of the month |
14:15.29 | ret28 | And wondering how to stop it being dumb with insisting that remote SIP calls authenticate just because the username matches one of my own |
14:15.41 | ret28 | (regardless of the from address domain) |
14:16.43 | *** join/#asterisk DarthClue (i=user76@wsip-68-99-73-32.tu.ok.cox.net) |
14:17.11 | iDunno | mutilator: the only way I can think of getting cron to do that would be to write a script that checked wether it was the last day of the month and call the script instead... and make the cronjob run between the 28-31 |
14:17.28 | mutilator | i jus said.. |
14:17.28 | *** join/#asterisk dalfry (n=dalfry@gateway.ishisystems.com) |
14:17.28 | mutilator | * * L * * |
14:17.31 | iDunno | mutilator: coo, when did that get there? |
14:17.39 | mutilator | runs on the "last" of that type |
14:17.44 | mutilator | http://wiki.opensymphony.com/display/QRTZ1/CronTriggers+Tutorial?decorator=printable |
14:17.54 | *** join/#asterisk PMantis (n=Miranda@66.251.89.34) |
14:18.07 | mutilator | i just have to get that |
14:18.22 | iDunno | doesn't appear to be documented in my crontab page :( |
14:18.51 | PMantis | My IAXy's recently started giving me one-way audio. What should I check? |
14:18.54 | DarthClue | is there a reference on how to set channel variables within c code? i need to hack the originate command to let me set some channel variables when i originate a call? |
14:19.27 | file | DarthClue: I'll give you a hint. |
14:19.47 | Katty | neat! a blue whale is bigger than a tyrannosaurus |
14:19.48 | file | in a sec |
14:20.03 | file | pbx_builtin_setvar_helper(channel, "name", "value"); |
14:20.15 | file | where channel is a pointer to the channel structure |
14:20.21 | iDunno | Katty: surely only larger than the largest set of bones they've found of a tyrannosaurus? |
14:20.28 | Katty | http://www.enchantedlearning.com/sgifs/Sizecomparisons.GIF |
14:20.43 | file | Katty: don't you have works-like stuff? |
14:20.54 | *** join/#asterisk Abbas (n=Abbas@203.81.225.91) |
14:20.55 | Katty | iDunno: uhh....much bigger. i dont' think it would matter. |
14:21.01 | Katty | file: nah. i'm on call though! |
14:21.08 | file | Katty: ooooooh |
14:21.22 | Katty | mew :< |
14:21.46 | Abbas | hello |
14:21.48 | Abbas | Oct 7 10:24:43 WARNING[990]: rtp.c:1425 ast_rtp_bridge: codec0 = 13 is not codec1 = 256, cannot native bridge. |
14:21.51 | iDunno | Katty: coo - yes, hmm. interesting. |
14:21.56 | Abbas | y this warning comes |
14:22.11 | iDunno | Katty: then again, I think I'd be more worried by a tyrannasaurus chasing me than a blue whale... |
14:22.13 | DarthClue | size doesn't matter. at least not when you are a bowl of petunias materializing in mid-air |
14:22.37 | iDunno | for one, if it was a blue whale I'd probably more be worried about the fact that I was in the middle of a cold sea type thing. |
14:22.46 | Katty | iDunno: you silly humans and your fears of being eaten by extinct creatures. |
14:23.02 | iDunno | heh |
14:23.09 | Katty | iDunno: i'd be worried about a sperm whale |
14:23.18 | Katty | iDunno: they're the largest carnivore |
14:23.46 | Katty | http://www.oceanwanderers.com/SpermWhale.7342.JPG |
14:24.08 | iDunno | Katty: I'd still be more worried by the whole being in the cold sea thing, probably ;) |
14:24.19 | Katty | http://www.oceanwanderers.com/NYSpermWhale.html <- poor thing :< |
14:24.33 | file | I'm hungry :\ |
14:25.28 | Katty | file: muffins. |
14:25.29 | iDunno | it was a bit of a careless spermwhale... |
14:25.39 | Katty | iDunno: it got beached by a storm |
14:25.42 | iDunno | dya reckon that it was out getting drunk with it's mates before that? |
14:25.42 | *** join/#asterisk hotgrits (n=hotgrits@192.160.238.156) |
14:25.47 | iDunno | ahh. that'd do it. |
14:25.53 | Katty | iDunno: they live in deep water, obviously |
14:25.58 | file | Katty: the downstairs freezer is inaccessible right now |
14:26.07 | Katty | file: :< |
14:26.14 | file | it's sad :( |
14:26.29 | *** join/#asterisk jonathh (n=asd@host81-154-159-222.range81-154.btcentralplus.com) |
14:26.42 | christo | I wonder if they removed it |
14:26.51 | christo | or if it's still there in some advanced stage of rot |
14:28.03 | Katty | iDunno: that whale was a calf too |
14:28.14 | Katty | iDunno: just a little baby :< |
14:28.22 | iDunno | Katty: indeed :( |
14:28.49 | Katty | file: don't breathe that. eat it. |
14:28.57 | file | ohhhh |
14:29.05 | iDunno | heh |
14:29.15 | Katty | you son of a silly person. |
14:29.55 | christo | I'm having the same problem as this bloke: http://asterisk.linkx.net/asteriskusers/200401/msg00627.html |
14:29.59 | christo | damned if I can fix it :( |
14:30.39 | file | Katty: I force you to listen to the soundtrack for The Phantom of the Opera |
14:30.39 | Katty | yay! |
14:30.40 | file | christo: no chan_iax2.so |
14:31.38 | christo | file - chan_iax2.so is there |
14:31.44 | file | is it loaded? |
14:31.55 | christo | oh poop |
14:32.00 | christo | no |
14:32.01 | znoG | i think the guys at openpbx should put something on their page stating whether they do or do not have anything to do with Asterisk, cause it has created confusion. |
14:32.42 | christo | can I reload modules from the CLI? |
14:32.49 | FuriousGeorge | whats the range for the rxgain setting in zapata.conf? 10 - -10 gb? |
14:32.51 | christo | ie without dropping * |
14:32.55 | FuriousGeorge | db*** |
14:32.59 | file | christo: yes, load <module name> reload <module name> |
14:33.03 | file | ie: load chan_iax2.so |
14:33.06 | christo | thanks file :) |
14:33.14 | Dr_Ray | 100%-100% |
14:33.36 | FuriousGeorge | Dr_Ray: u sure thats a percentage? i heard it was db |
14:34.02 | Dr_Ray | my asterisk book says it's a percentage.. but it could be wrong |
14:34.03 | Hmmhesays | so strange |
14:34.41 | Katty | Hmmhesays: a dolphin's iq is only slightly lower than the average human's iq |
14:35.30 | FuriousGeorge | Katty: i find that difficult to believe, you'd think they'd be able to teach'em morris code or something to communicate if that were the case |
14:35.33 | ian_k | Katty: did it take the supervised WISC-R test? |
14:35.47 | Hmmhesays | Hmmm yes very strange |
14:35.54 | Hmmhesays | my meetme is acting up |
14:36.15 | FuriousGeorge | Katty: not that they arent intelligent, they recognize themselves in mirrors. which is more than you can say for almost any animal |
14:36.17 | Dr_Ray | rxgain: Adjusts receive gain. This can be used to raise or lower the incoming volume to compensate for hardware differences. Takes a percentage of capacity, from -100% to +100%rxgain: Adjusts receive gain. This can be used to raise or lower the incoming volume to compensate for hardware differences. Takes a percentage of capacity, from -100% to +100% |
14:36.24 | Dr_Ray | oops |
14:36.26 | Katty | FuriousGeorge: google it (= |
14:36.33 | Hmmhesays | i initiate a call via the manager, it calls out my vonage account, I answer my cell and it should dump me into meetme |
14:36.38 | Hmmhesays | it does, but then I hear nothing |
14:36.48 | FuriousGeorge | Dr_Ray: thanks i believe you |
14:37.02 | FuriousGeorge | did you guys here about the parrot who understands the conzep of zero |
14:37.04 | *** join/#asterisk SuPrSluG (n=SuPrSluG@pool-141-149-242-183.buff.east.verizon.net) |
14:37.06 | FuriousGeorge | *concept |
14:37.07 | ian_k | Dr_Ray: gain should not be adjusted beyond 10-15 in either direction, or hardware damage may result |
14:37.18 | file | Katty: masquerade! |
14:37.29 | Dr_Ray | well, I double checked for my memory too.. |
14:37.46 | file | mřřse attack!!! |
14:37.58 | RoyK | møø |
14:38.47 | protien | is there a way i can set a callback script, so if someone dials an extension, it gets their number and calls them back? |
14:38.56 | Hmmhesays | yes |
14:39.00 | file | protien: you can do anything |
14:39.08 | Hmmhesays | file, fix it |
14:39.09 | file | you just have to research how, learn, and find examples |
14:39.13 | DarthClue | file: does originate use a channel structure? |
14:39.17 | protien | yeah im trying to find an example |
14:39.27 | file | Hmmhesays: mmm don't remember |
14:39.28 | FuriousGeorge | so how come meetme keeps prompting me for a conference pin when i specified none in metme.conf? |
14:39.31 | file | er that was for DarthClue |
14:39.43 | file | DarthClue: show me source code. |
14:39.51 | *** join/#asterisk mkrufky (n=mk@68.160.103.77) |
14:39.51 | Hmmhesays | protien: check out the wakeup call script |
14:40.05 | Hmmhesays | on the wiki wiki |
14:40.20 | Hmmhesays | as for my problem, its an odd one |
14:40.34 | file | I hear the cops! they're coming for me |
14:40.37 | DarthClue | http://www.pbxfreeware.org/app_changrab.c ... looking at the originate_cli section |
14:40.41 | FuriousGeorge | Hmmhesays: apply ointment, it should subside |
14:40.45 | file | kk |
14:40.48 | *** join/#asterisk Cresl1n (n=matt@gateway.digium.com) |
14:40.59 | christo | is the iax.conf now called iaxprov.conf, or is that something else? |
14:41.05 | christo | I've never needed that before on older versions of * |
14:41.13 | Dr_Ray | iaxprov is for the iaxy, i think |
14:41.14 | *** part/#asterisk ret28 (i=rt@82-71-120-246.dsl.in-addr.zen.co.uk) |
14:41.22 | Hmmhesays | FuriousGeorge my wonderfully delightful nick'ed friend... i don't think so |
14:41.24 | christo | hmm |
14:41.29 | file | DarthClue: I see what it does |
14:41.32 | christo | I don't care about the IAXy, I just wanna trunk :S |
14:41.37 | FuriousGeorge | Hmmhesays: lol |
14:41.39 | Dr_Ray | iax.conf |
14:41.46 | Katty | Hmmhesays: i think the dolphins are plotting |
14:41.53 | SuPrSluG | DarthClue:they use it in the callme.php script |
14:41.53 | file | DarthClue: look at the function static void *originate(void *arg) |
14:42.09 | file | DarthClue: you can probably put in the setvar stuff in that if statement |
14:42.34 | *** join/#asterisk rudi_ (n=rudi@karfire.megabit.net) |
14:42.40 | rudi_ | hey there |
14:42.43 | Hmmhesays | -- Executing MeetMe("SIP/17327905389-2694", "1337|dM") in new stack |
14:42.43 | Hmmhesays | <PROTECTED> |
14:42.47 | Hmmhesays | seriously wtf |
14:43.39 | christo | Dr_Ray - the odd thing is that I have an iax.conf, but when I reload chan_iax2.so it complains that there's no iaxprov.conf |
14:43.47 | *** join/#asterisk shido6 (n=shido6@d221-68-210.commercial.cgocable.net) |
14:44.10 | Dr_Ray | that is odd, I thought iaxprov.conf was just for firmware upgrades |
14:44.28 | christo | well I copied one over from the samples directory |
14:44.54 | christo | this is a PITA actually.. I'm still getting this error http://asterisk.linkx.net/asteriskusers/200401/msg00627.html |
14:44.57 | malverian[work] | Hmmm... |
14:45.20 | Dr_Ray | I Was wrong, it's not for the iaxy |
14:45.39 | christo | I have checked the module is there, reloaded it at the CLI, created the iax.conf, copied over an iaxprov.conf just to keep it happy, but still it doesn't seem to know what IAX2 is |
14:45.43 | christo | crazy |
14:45.55 | malverian[work] | SNOM doesn't resend it's SUBSCRIPTIONs when you reload Asterisk... |
14:45.58 | malverian[work] | That kinda sucks. |
14:46.21 | *** join/#asterisk Blazint (n=blazin@cm225.epsilon203.maxonline.com.sg) |
14:46.27 | FuriousGeorge | does anyone care to speculate as to whymeetme prompts me for a PIN when there is none and complains my pin is wrong when i specify one? |
14:46.40 | FuriousGeorge | DTMF is fine |
14:46.41 | malverian[work] | FuriousGeorge, There's a flag to not prompt for pin. |
14:47.01 | rudi_ | does anyone know how a good web-interface to the asterisk-config? |
14:47.05 | FuriousGeorge | malverian[work]: ILL LOOK INTO THAT, THINKS |
14:47.12 | FuriousGeorge | *thanks |
14:47.22 | FuriousGeorge | sorry 4 caps |
14:49.06 | FuriousGeorge | *of |
14:49.11 | rudi_ | anyone? :> |
14:49.13 | *** part/#asterisk Guus_ (n=g@139.63.241.162) |
14:49.36 | rudi_ | (i'm talking about one that supports using existing config files without any kind of databases, templates, profiles etc.) |
14:50.03 | christo | poopah |
14:50.30 | FuriousGeorge | rudi_: by gui for the asterisk config you mean "a way of getting around manually editing the configs"? |
14:51.08 | Hmmhesays | meetme hates me today, as soon as I enter it stops the music on hold |
14:51.10 | rudi_ | yeah...i actually prefer doing it the 'manual way' but not everbudy else does.. |
14:51.23 | rudi_ | *everybody |
14:51.54 | iDunno | people like guis?! woah. |
14:52.51 | iDunno | is good for graphical manipulation, and looking at pretty rather than functional websites... |
14:53.03 | iDunno | but what else do you actually want or need a GUI for? |
14:53.26 | rudi_ | apparantly my company wants to offer asterisk-based solutions to their customers |
14:53.38 | PMantis | Anyone know why all my IAXy's would just start to receive audio, but not send? DTMF works, but not voice! |
14:53.48 | *** join/#asterisk MicC_ (n=sum1@CPE000c419ce901-CM000a7363f92c.cpe.net.cable.rogers.com) |
14:53.50 | MicC_ | hey guys |
14:53.55 | *** join/#asterisk AlivesWrk (n=email@static-70-19-114-50.ny325.east.verizon.net) |
14:54.20 | AlivesWrk | what is the most used windows popup call manager software? |
14:54.21 | rudi_ | and they're asking for something that is also managable by not-extremly-experienced admins |
14:54.35 | AlivesWrk | i need something with caller id and possibly transfer/on hold for calls |
14:54.40 | *** join/#asterisk gandhijee (n=gandhije@host-66-202-34-162.spr.choiceone.net) |
14:54.52 | Hmmhesays | ok this is the strangest thing evar |
14:54.58 | Hmmhesays | some of the conferences work, some don't |
14:55.00 | Katty | AlivesWrk: mew? |
14:55.08 | Katty | AlivesWrk: popup call manager software? |
14:55.13 | Beirdo | morning once again, it seems |
14:55.17 | Katty | Beirdo: rehi |
14:55.27 | Beirdo | Heya, Katty. Good morning |
14:56.15 | rudi_ | hm no ideas at all? :> |
14:56.29 | AlivesWrk | i am new to asterisk, yes |
14:56.45 | Katty | Hmmhesays: mew? what is this popup call manager software stuff? |
14:56.48 | rudi_ | i checked some of the links from the voip-info.org wiki |
14:56.50 | AlivesWrk | katty: yes, just somethign that would interface with calls on a windows machine |
14:57.07 | Hmmhesays | can I kill a conference from the clie? |
14:57.09 | Hmmhesays | *cli even |
14:57.25 | MicC_ | hmmmhesays: do yu have al ot of pseudo channels not disappearing ? |
14:57.32 | Hmmhesays | just one |
14:57.34 | MicC_ | yah..you can soft hangup it |
14:57.43 | MicC_ | if it will let you |
14:58.15 | MicC_ | is there a way to specify and external IP on the polycom IP501s? for Nat'ing ...etc? |
14:58.44 | Hmmhesays | hey file |
14:58.51 | Hmmhesays | i've come back to the world of voip for awhile |
14:58.57 | Katty | MicC_: why would you want a phone on an external ip? |
14:59.16 | file | Hmmhesays: yay! |
14:59.44 | MicC_ | Katty: umm....I wanna bring a polycom home with me :P |
14:59.51 | *** join/#asterisk shido6 (n=shido6@d221-68-210.commercial.cgocable.net) |
14:59.55 | MicC_ | I have firewall at home. |
15:00.01 | *** join/#asterisk gvag11 (n=g@ppp30-adsl-61.ath.forthnet.gr) |
15:00.03 | file | nat=yes is your friend |
15:00.08 | MicC_ | I got it working with Xlite...just don't see the option in the Polycomms. |
15:00.22 | MicC_ | file that simple eh? |
15:00.26 | file | yup |
15:00.34 | Katty | MicC_: try it. you may see. you may like it in a tree. |
15:00.46 | file | Katty: excellent! gold star for you! |
15:00.46 | Katty | MicC_: or in a house with a mouse. |
15:00.54 | Katty | file: mew? |
15:01.01 | MicC_ | I don't it all, you will surely have a mysterious "fall" |
15:01.20 | MicC_ | err I screwed that up |
15:01.46 | MicC_ | file: here is the fun part, my * is nat'd as well :P |
15:01.58 | file | externip and localnet |
15:02.11 | MikeJ[Laptop] | yes! |
15:02.16 | Katty | is that something like a parsnip? |
15:02.19 | MikeJ[Laptop] | good morning mr. file. |
15:02.24 | MikeJ[Laptop] | Katty, yes... errr.. no |
15:02.26 | file | hi MikeJ!!! |
15:02.36 | MikeJ[Laptop] | FUN DAY! |
15:02.38 | Katty | MikeJ[Laptop]: you sure are chipper. |
15:02.40 | MikeJ[Laptop] | not really :( |
15:02.44 | Katty | oh :< |
15:02.50 | file | MikeJ[Laptop] is death! |
15:02.50 | Katty | i'm mostly chipper! |
15:02.55 | Katty | MikeJ[Laptop]: i'm all grown up today. |
15:02.59 | malverian[work] | Is "Event: -" for a NOTIFY actually RFC compliant? |
15:03.14 | malverian[work] | My SNOM phone seems to see it as a bad event. |
15:03.45 | malverian[work] | The server sends this when doing a "reload" to tell the phone to resend SUBSCRIBEs. |
15:04.01 | MicC_ | file: yah...it works great...just haven't done the polycom yet |
15:04.34 | file | MicC_: nat=yes is pretty simple, it ignores the IP information in the SIP messages and uses the received IP and port |
15:04.55 | MicC_ | kewlio...thanks file. |
15:05.07 | MicC_ | so basically I would have no problem when I go home this weekend :P |
15:05.13 | Katty | hmm. |
15:05.17 | file | ...yeah |
15:05.41 | Hmmhesays | so dynamic conferences work like shiat in an old version of aah |
15:06.34 | MikeJ[Laptop] | Katty, all grown up? |
15:06.44 | MikeJ[Laptop] | file, umm.. I'm a bastard arn't I? |
15:06.50 | MikeJ[Laptop] | it was yesterday wasn't it? |
15:06.52 | Beirdo | I don't ever wanna grow up |
15:07.07 | Hmmhesays | i'm a toys r mine brat, I go running through the store yelling give me this and that |
15:07.07 | mutilator | i hope i'm never born |
15:07.19 | Hmmhesays | i hope i'm never birthed again |
15:07.26 | file | MikeJ[Laptop]: nah you're all peachy! |
15:07.34 | MikeJ[Laptop] | but the 6th right? |
15:07.37 | Beirdo | Hmmhesays: I'm sure your mother would appreciate that |
15:07.48 | file | MikeJ[Laptop]: the 6th what |
15:08.38 | *** join/#asterisk e3g (i=ee@u15157627.onlinehome-server.com) |
15:09.29 | e3g | anyone who could help me in Linux commands???? :( learning Asterisk is not enough.... |
15:09.49 | AlivesWrk | what do you need? |
15:09.51 | AlivesWrk | ##linux |
15:09.55 | PMantis | I thought one-way audio was a SIP problem! Now, my IAXy's suddenly have this problem. Ideas? |
15:11.14 | e3g | need to change the Primary IP address of my machine through SSH |
15:11.17 | mishehu | PMantis: an iax2 bug. I had the same problem when I was running an earlier version of head (prior to 1.2.0b1) |
15:11.40 | Beirdo | mishehu: sure it's not just missing firewall rules? |
15:11.59 | marc324 | xlite --> asterisk --> pstn -- is this outgoing? |
15:12.15 | *** join/#asterisk Alives (i=loot@cpe-68-173-215-206.nyc.res.rr.com) |
15:12.24 | e3g | marc324: yep |
15:12.34 | PMantis | mishehu: Weird, I was running this version of Asterisk for some time... (1.0.9) |
15:12.35 | Katty | MikeJ[Laptop]: today's my 21st. |
15:12.35 | DarthClue | oh file... |
15:12.36 | file | TECHNICALLY it's both ;) |
15:12.45 | MikeJ[Laptop] | file can drink legally now too |
15:12.47 | file | DarthClue: did you break it? |
15:12.54 | Hmmhesays | in canadia |
15:13.06 | MikeJ[Laptop] | which just happens to be where he is |
15:13.13 | file | crazyness! |
15:13.22 | MikeJ[Laptop] | yes |
15:13.37 | DarthClue | file: remind me to buy you dinner next time i see ya, k? |
15:13.39 | MikeJ[Laptop] | carefull they'll stick |
15:13.48 | bkw_ | file |
15:13.50 | bkw_ | oh file |
15:13.51 | bkw_ | call 42 |
15:13.56 | file | bkw_: there you are! |
15:14.04 | Katty | file: bkw_ was hiding. |
15:14.08 | Katty | file: if by hiding i mean having breakfast. |
15:14.18 | marc324 | I get "Cannot find extension context default" when attempting to make calls from xlite. |
15:14.22 | PMantis | mishehu: And... it was working on the 3rd... I havne't even restarted * since then! |
15:14.23 | Katty | file: in his better homes and gardens kitchen. |
15:15.47 | *** join/#asterisk olivier_ (n=olivier_@obs92-4-82-239-116-113.fbx.proxad.net) |
15:16.04 | file | it's networking day |
15:16.17 | Katty | http://kat.mandriva.com/ <- kat goes public. |
15:16.29 | protien | hm im trying to run this yet another wakepu script |
15:16.31 | protien | ive got it all setup and |
15:16.32 | malverian[work] | I'm concerned about this NOTIFY stuff in chan_sip. |
15:16.32 | protien | <PROTECTED> |
15:16.32 | protien | <PROTECTED> |
15:16.37 | protien | is all i get when i dial it |
15:16.42 | protien | it just finishes before it starts |
15:16.46 | malverian[work] | protien, "agi debug" in the cli. |
15:17.13 | DarthClue | new question...where are the c code functions documented? and...how do i set the cdr field values from c code |
15:17.19 | gandhijee | i know this is a retarded question |
15:17.44 | gandhijee | but how come when i include my local extentsion under default |
15:17.49 | gandhijee | i can't ring them |
15:17.53 | *** join/#asterisk ManxPower (n=eric@adsl-70-247-221-174.dsl.lgvwtx.swbell.net) |
15:18.08 | *** join/#asterisk fiber0pti (n=johndoe@pcp01876618pcs.sandia01.nm.comcast.net) |
15:18.14 | protien | malverian[work], http://pastebin.com/386239 |
15:18.46 | ManxPower | If I ever find the cisco programmer that decided to default async interfaces to 8 data bits, no parity, and TWO stop bits, I will kill them slowly. |
15:18.47 | malverian[work] | protien, It's apparently exiting immediately upon invocation. |
15:18.55 | protien | yeah man |
15:18.56 | malverian[work] | Open the script and add some debugging. |
15:19.08 | protien | thats above me |
15:20.42 | marc324 | how do you make outbound call? xlite -> asterisk --what should I put in extensions.conf ? |
15:20.54 | ManxPower | ~docs |
15:20.56 | jbot | from memory, docs is Documentation can be found at http://www.digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or http://www.voip-info.org/wiki-Asterisk or http://www.asteriskdocs.org or http://www.oreilly.com/catalog/asterisk |
15:20.58 | Katty | DarthClue: how's the bus? |
15:21.02 | PMantis | mishehu: I just figured it out! I have my IAXy provisioned with "codec: adpcm" as well as pcm. Created tons of CLI errors! Ugh. |
15:21.49 | DarthClue | hell, living hell. but i'm managing almost 90 hours every 2 weeks which makes for a nice check and a really easy job. kids suck. well teenagers do. |
15:21.59 | *** join/#asterisk groogs (n=greg@d221-73-237.commercial.cgocable.net) |
15:22.08 | Katty | DarthClue: whyfor? |
15:22.16 | *** part/#asterisk PMantis (n=Miranda@66.251.89.34) |
15:22.24 | groogs | anyone here done the linksys PAP2 unlock thing? |
15:22.37 | gandhijee | marc: you make your sips context in the extensions |
15:22.42 | gandhijee | ie [sip_peers] |
15:22.43 | Katty | groogs: no |
15:23.01 | Katty | Hmmhesays: have you even said hi to me this morning yet? :P |
15:23.07 | protien | im still having this annoy as problem, when i recieve calls via IAX from goiax.com i get this |
15:23.10 | protien | Oct 8 00:49:42 NOTICE[24120]: chan_iax2.c:6690 socket_read: Rejected connect attempt from 204.13.233.114, who was trying to reach 's@' |
15:23.13 | DarthClue | they are just bad kids. the parents don't care, and the kids are just stupid. they have no respect for authority (spitting on cops, throwing trash at the principal) and i can't do anything but drive em around. |
15:23.14 | groogs | doesnt look very hard. $65cdn for 2 fxs... |
15:23.19 | *** join/#asterisk pa (n=Paolo@unaffiliated/pa) |
15:23.20 | protien | how can i forward those calls to my extension? i cant work that out |
15:23.25 | gandhijee | the do exten => exten#,priorty,Dial(SIP/<user> |
15:23.28 | Katty | DarthClue: ewww |
15:23.44 | ManxPower | DarthClue, You are now a school bus driver? |
15:23.50 | DarthClue | katty, it's better than flipping burgers, and pays better too. |
15:23.50 | marc324 | gandhi -- yes, what do i put in it... I have Dial(zap/1) |
15:24.03 | Katty | DarthClue: oh, right....i never did the burger flipping thing |
15:24.14 | ManxPower | Isn't being a school bus driver the 3rd level of Dante's hell? |
15:24.15 | Beirdo | DarthClue: take corners extra hard? |
15:24.17 | DarthClue | ManxPower: it's a part time gig that allows me to do consulting on the side. |
15:24.32 | Beirdo | oh look, half the brats are on the floor... oooops |
15:24.39 | DarthClue | Beirdo: as hard as possible, but have to limit it to about 20 mph so the bus doesn't flip/roll. |
15:24.42 | Katty | Beirdo: now, now |
15:24.49 | Beirdo | true enough, don't want to crash |
15:24.53 | Katty | Beirdo: someone's little girl is on that bus..and she's a good kid. |
15:25.05 | Beirdo | Katty: true. |
15:25.07 | DarthClue | ManxPower: could be, but if that is true, then I must be king. |
15:25.11 | Katty | Beirdo: a quiet little 6 year old with pigtails. |
15:25.25 | Beirdo | won't somebody think of the children |
15:25.27 | Beirdo | :) |
15:25.33 | Katty | exactly |
15:25.39 | Katty | file: mew? |
15:25.40 | DarthClue | Katty: no girls, just stupid little boys. that's how bad these kids are, they have 2 buses, one for the girls (who are worse) and one for the boys. |
15:25.45 | Beirdo | you are right of course, Katty, but that won't stop us from thinking of punishing the bratty ones |
15:26.03 | Katty | :< |
15:26.06 | *** join/#asterisk gabb0 (n=gabb0@131.202.90.23) |
15:26.10 | gabb0 | hello all |
15:26.18 | Katty | if i ever adopt, maybe i'll send them to a private school |
15:26.35 | DarthClue | file: how do i set the cdr user field? any quick ideas on that? |
15:26.42 | gabb0 | are the DS3000P out at all |
15:26.43 | Beirdo | not all public shools are that bad |
15:26.51 | Beirdo | but there seem to be a fair number that are |
15:26.51 | *** join/#asterisk tzafrir_laptop (n=tzafrir@bzq-179-75-202.cust.bezeqint.net) |
15:26.57 | Katty | Beirdo: but how would you know the difference? |
15:27.05 | Beirdo | true |
15:27.07 | Katty | Beirdo: in my first day of school... |
15:27.13 | Katty | Beirdo: i thought my mother had abandoned me |
15:27.16 | Beirdo | and some private schools are hellish too |
15:27.19 | Katty | Beirdo: to a much of hideous monsters |
15:27.27 | Beirdo | awww. Poor Katty. |
15:27.31 | Katty | s/much/bunch/ |
15:27.35 | tzanger | yeah my kids get a good healthy dose of discipline. I won't tolerate that kind of thing from my kids |
15:27.38 | tzanger | and they know it |
15:27.40 | Beirdo | I had my cousin there to help me not feel that way |
15:27.42 | DarthClue | that is one advantage of driving, i can see which schools are bad/good. |
15:27.45 | file | DarthClue: not particularly |
15:28.09 | Katty | tzanger: there's a fine like between discipline and overboard. i hope you're not crossing it. |
15:28.11 | DarthClue | file: if you want dessert with that lunch, you need to think harder. |
15:28.23 | Katty | goshdangitanyhow i can't type today. |
15:28.29 | tzanger | Katty: discipline means teaching self respect and others-respect. overboard usually only happens when you do it angry |
15:28.35 | tzanger | I don't discipline my kids angry, ever |
15:28.41 | Katty | good |
15:28.46 | Beirdo | good choice |
15:28.47 | Katty | hopefully they won't turn out like me then :P |
15:28.54 | ManxPower | I'll never have kids |
15:28.54 | tzanger | and it's very very very rare that I ever strike them, and if so it's usually on the hand (when they're little) |
15:29.12 | Katty | i would never strike a child |
15:29.13 | Katty | ever |
15:29.18 | Katty | for /any/ reason. |
15:29.19 | tzanger | children are NOT little adults, that was one of the biggest fallacies in recent parenting doctrine |
15:29.22 | tzanger | Katty: that's wrong |
15:29.30 | Katty | tzanger: what's wrong? |
15:29.35 | ManxPower | tzafrir, exactly! |
15:29.44 | ManxPower | Children are unsocialized animals. |
15:29.46 | tzanger | when a child does something that puts them in immediate danger they need immediate and very clear correction |
15:29.54 | Abbas | hello ManxPower |
15:29.59 | Katty | tzanger: hitting a child is not the answer. |
15:30.07 | Katty | tzanger: i am living proof of that statement. |
15:30.09 | ManxPower | One of the major jobs of a parent is to socialize them. |
15:30.11 | tzanger | a smack on the hand gets their attention, does NOT hurt and snaps them out of whatever spinlock they have |
15:30.21 | Abbas | can u pls guide me about this warning |
15:30.31 | Abbas | Oct 7 10:24:43 WARNING[990]: rtp.c:1425 ast_rtp_bridge: codec0 = 13 is not codec1 = 256, |
15:30.33 | Beirdo | and if they are about to grab a pot of boiling water on the stove... |
15:30.41 | Katty | tzanger: if it doesn't hurt, then it's not a strike. |
15:30.46 | tzanger | Abbas: it can't transcode between codec 13 and codec 256 |
15:31.01 | tzanger | Katty: they ACT like it's the most painful thing ever but that's normal :-) |
15:31.14 | Abbas | tzanger i am testing intel's 723 |
15:31.31 | ManxPower | You don't talk to a dog when it does something bad. Before a child learns language skills physical things are the only way to communicate |
15:31.34 | protien | is there any examples of how to forward all unused/bad numbers to a recoding |
15:31.41 | Katty | tzanger: i wouldn't recommend ignoring a child if they think it hurts. |
15:31.53 | tzanger | sometimes a child's mind is in such a tight little loop that the only thing you can do to get them out of it is to do something physical. (screaming their head off, inconsolable) -- you give 'em a little smack and their brain snaps out of it because it has to respond to the stimulus. THEN you can begin to talk them down or reason with them if they're older |
15:32.00 | Katty | tzanger: but do what you want. it's their psychology you're playing with. |
15:32.05 | tzanger | Katty: you're overexaggerating |
15:32.10 | Katty | tzanger: i'm not. |
15:32.14 | tzanger | you are. |
15:32.16 | Katty | tzanger: i lived it |
15:32.20 | *** join/#asterisk zpn (n=xpn@gateway.digium.com) |
15:32.22 | Katty | tzanger: i'm not exaggerating |
15:32.23 | tzanger | smacking them on the hand is not "playing with their psychology" |
15:32.28 | nfi|ermes | i have some problems with voicebox |
15:32.35 | tzanger | Katty: if you were physically abused that is *much* different than what I'm describing |
15:32.35 | Abbas | tzanger: how can i avoid this warning |
15:32.38 | Katty | tzanger: striking them, then ignoring that they're in pain... |
15:32.41 | protien | manxpower, is your example callback script, included in that tar of all your website? |
15:32.44 | Beirdo | I got spanked may times (deservedly) as a child, and I turned out just fine |
15:32.53 | Katty | tzanger: if it hurts the child, why do you do it? |
15:32.54 | nfi|ermes | -- Playing 'vm-review' (language 'it') <---- but i listen in english |
15:33.07 | tzanger | Katty: you give them a swat on the hand and they recoil as if you branded them -- that's not hurting them. |
15:33.07 | Katty | tzanger: i don't understand that. |
15:33.25 | Katty | tzanger: did you bother to ask them if it hurt? |
15:33.31 | ManxPower | protien, Yes, but I don't provide ANY help for thost scripts |
15:33.41 | protien | thats fine, i just want to know which one it is |
15:33.44 | protien | so i can try it out |
15:33.58 | tzanger | I agree with corporal punishment for small children. They do not understand words or reason because they themselves are not capable of the level of reason that is required. Corporal punishment is NOT beatin gthe shit out of them because you're drunk and they're annoying |
15:34.15 | *** join/#asterisk paryl (n=paryl@209.236.78.59) |
15:34.19 | tzanger | Katty: I know what hurts and what doesn't, and I am very much aware of the level of force I use, which is NOT a lot |
15:34.24 | Katty | tzanger: well if you don't think they can comprehend reason, then do you think they can comprehend why you're smacking them? |
15:34.32 | tzanger | there is no mark left, no redness, no swelling, nothing. |
15:34.43 | paryl | i can't find any working sites that have drivers for the tdm400p |
15:34.44 | Katty | tzanger: or just that you hit them, for no reason they could understand? |
15:34.46 | ManxPower | Katty, They can't. Neither can a dog, but it works for dogs. |
15:35.01 | tzanger | Katty: because animals (people included) respond to it. It's not a way to teach them, it's a way to train them. |
15:35.01 | ManxPower | My cats know that if they get on the counter top they will get smacked. |
15:35.12 | Katty | tzanger: children are NOT dogs. |
15:35.13 | tzanger | Katty: you're again exaggerating. i'm not beating them |
15:35.16 | tzanger | Katty: you're absolutely right |
15:35.18 | protien | manxpower, which script should i be looking at? |
15:35.19 | Katty | tzanger: dogs should not be trained by hitting them |
15:35.25 | Katty | tzanger: and neither should children. |
15:35.36 | tzanger | but before they understand reason they must understand basic rules and the most basic rule is cause and effect |
15:35.37 | ManxPower | protien, I don't recall. It's been a year since I even looked at the script. |
15:35.40 | Katty | tzanger: dogs are trained by positive reinforcement. |
15:35.40 | mutilator | Katty: sorry but yes, they do |
15:35.50 | tzanger | you do something wrong, I will be upset. |
15:35.59 | Katty | you know what, we'll just agree to disagree |
15:36.09 | Katty | because there's nothing that you can say that will change what i believe |
15:36.15 | Katty | and i will NEVER strike a child for any reason. |
15:36.15 | Katty | none |
15:36.17 | Katty | ever |
15:36.19 | Katty | end of story |
15:36.25 | tzanger | but as I said, now that they understand reason I very very very rarely ever have to resort to corporal punishment |
15:36.33 | Katty | NEXT |
15:36.35 | mutilator | heh |
15:36.46 | tzanger | because the rules are enforced and showing them how upset you are is FAR more painful to them than any smack |
15:37.04 | Hmmhesays | anyone ever heard of "nailing" in the cisco voip world? |
15:37.15 | mutilator | nope |
15:37.51 | tzanger | Katty: you look at my children: very well behaved, very sociable, funny, friendly, happy but more importnatly: respectful to themselves and respectful to others. This is a 5 and 4 year old too. I think that my ex and I have done very very well so far |
15:37.53 | Hmmhesays | yeah some guy just called me "cisco has a new technology called nailing, it doesn't require any digital signal processing, it just reroutes calls IP to IP" |
15:38.06 | Hmmhesays | i'm like uhhhhhh dude, that sure sounds like a proxy |
15:38.18 | tzanger | these children are not social misfits, they don't cower when someone raises their hand, they are not afraid of expressing their opinions or disagreeing with their parents |
15:38.30 | Beirdo | sounds like what asterisk does if there's no codec change |
15:38.31 | tzanger | Katty: what you're describing is *very* different from what I am trying to describe |
15:38.39 | paryl | oxymoron? "I think that my ex and I have done very very well so far" |
15:38.44 | mutilator | beating a child senseless is dumb |
15:38.45 | Hmmhesays | stateful or stateless, he described a proxy |
15:39.23 | paryl | where can i find tdm400p drivers? |
15:39.38 | gandhijee | compile them... when you compile zaptel |
15:39.45 | protien | where are Asterisk::AGI and Asterisk::Outgoing available, i cant find them on google |
15:39.52 | tzanger | never striking your children is a laudable goal but I don't agree with it being the best solution ever |
15:40.10 | *** join/#asterisk wunderkin (i=kev@12-219-162-233.client.mchsi.com) |
15:40.18 | *** join/#asterisk Veto_laptop (i=mdkuser@cpe-66-69-38-192.satx.res.rr.com) |
15:41.00 | iDunno | tzanger: the best solution is to just not have children. works quite well ;) |
15:41.03 | tzanger | I will not argue with my children, I will not bargain with them. They know the rules and they know that their parents have the final say, especially at these ages. As they get older (it's already happened, but it'll happen much more as they age) they will be given more freedoms and more opportunities ot execercise their minds and their decision making abilities and ability to listen ot their inner voices |
15:41.13 | paryl | gandhijee: i can't find the source to compile... is it packaged with the asterisk source? |
15:41.26 | tzanger | that's the big one right onw, especially with my daughter. She KNOWS she's doing things that aren't good but she still does them, I'm trying to teach her to listen to that inner voice because I know she has it |
15:41.34 | gandhijee | http://ftp.digium.com/pub/zaptel/zaptel-1.0.9.2.tar.gz |
15:41.37 | gandhijee | download that |
15:41.45 | gandhijee | then once its compiled and installed |
15:41.50 | gandhijee | modprobe wcfxs |
15:41.54 | mutilator | ya |
15:42.00 | mutilator | else your child will end up like my gf |
15:42.07 | paryl | awesome, thats what i was looking for ;) |
15:42.11 | nfi|ermes | -- Playing 'vm-review' (language 'it') <---- but i listen in english |
15:42.19 | mutilator | pissed off at the world, doesn't do anything for herself, and doesn't care about much else but her |
15:42.25 | mutilator | and not social at all |
15:43.42 | *** join/#asterisk apardo (n=apardo@200.Red-83-50-234.dynamicIP.rima-tde.net) |
15:43.49 | *** join/#asterisk florz (n=florz@2001:1a50:503c:0:0:0:0:1) |
15:43.59 | tzanger | my daughter is far more hurt when her daddy's upset with her than if her daddy ever hit her. Guilt is a much bigger punishment, and wielded incorrectly it would be far worse than any kind of physical abuse I could give to her, but katty I don't think you agree |
15:44.59 | Beirdo | for sure. the mental pain is way harder to judge too |
15:45.12 | mutilator | i don't think it is.. |
15:45.40 | mutilator | you have to know a person for a bit, but everyone feels the same to an extent |
15:45.54 | mutilator | less they're just plain nuts |
15:46.39 | Beirdo | heh |
15:46.45 | Beirdo | have a good sleep |
15:46.45 | gandhijee | yep |
15:47.21 | mutilator | actaully i have to get back to work |
15:47.28 | mutilator | my fsckin other engineer quit |
15:47.40 | file | mutilator: well that bites |
15:47.41 | mutilator | so i have like 2x the work and i was already too busy as it was |
15:48.11 | mutilator | this weekend is going to be heaven |
15:48.22 | mutilator | my gf promised me bowling and sex tomorrow |
15:48.24 | mutilator | :P |
15:48.45 | *** join/#asterisk n0where (n=kc@q041140.ppp.asahi-net.or.jp) |
15:48.56 | mutilator | think i'll turn the pager off and dissapear |
15:48.57 | Beirdo | way to go, mutilator |
15:49.41 | mutilator | i still havn't figured out my callerid problem |
15:49.56 | mutilator | fristrates me |
15:50.00 | mutilator | *u |
15:50.54 | tzanger | bowling and sex? |
15:50.58 | tzanger | that is a VERY odd combination |
15:51.00 | iDunno | at the same time? |
15:51.04 | iDunno | that's going to hurt. |
15:51.19 | mutilator | will be interesting tho no? |
15:51.36 | mmmToop | ...has anyone here tried to install gnudialer? |
15:51.52 | nfi|ermes | anyone can help me with some problems with voicemailbox ? |
15:54.51 | *** join/#asterisk miztic (n=gerard@rarcoa.com) |
15:57.14 | *** join/#asterisk kaushal (i=kaushal@202.159.244.45) |
15:57.40 | *** part/#asterisk kaushal (i=kaushal@202.159.244.45) |
16:01.07 | *** join/#asterisk AsterNov (i=richard_@82-45-239-141.cable.ubr01.enfi.blueyonder.co.uk) |
16:03.05 | *** join/#asterisk jdv79 (n=jdv79@u1057064.ul.warwick.net) |
16:03.47 | jdv79 | if Monitor() a call and i hold down a DTMF tone constant, should i hear that or should it be reduced to a blip instead? |
16:05.32 | *** join/#asterisk Qwell (n=north@24-50-66-194.vnnyca.adelphia.net) |
16:11.44 | marc324 | i get Unable to create channel of type 'Zap' (cause 0) |
16:12.40 | Hmmhesays | okrut? |
16:12.47 | Katty | orkut |
16:12.53 | Katty | i'm feeling dyslexic apparently |
16:13.11 | iDunno | orkut, the fad of 2 years ago... |
16:13.12 | Katty | and hungry. bye now. |
16:13.20 | Katty | iDunno: heh, i still have icq |
16:13.24 | iDunno | google sponsored, wasn't it? kinda sucked arse ;) |
16:13.33 | Katty | iDunno: my 5 digit number FROM THE 5TH GRADE |
16:13.35 | Beirdo | it's still around |
16:13.41 | iDunno | 3442444 |
16:13.42 | Beirdo | linked closer to gmail now |
16:13.49 | iDunno | nope, I was in the 6 digit group. |
16:13.54 | Katty | bye now. |
16:13.58 | iDunno | 5 digits is just scary ;) |
16:13.59 | file | Katty: noooooooooo |
16:14.01 | file | :( |
16:14.03 | iDunno | bye bye Katty ;) |
16:14.11 | iDunno | have a good rest-of-ya birthday |
16:15.00 | *** join/#asterisk DeeJayTwo (i=deejay2@215-238.sh.cgocable.ca) |
16:15.27 | jdv79 | anyone else have trouble passing audio like DTMF and stuff |
16:15.31 | AsterNov | I cant start asterisk@home manually using asterisk -vvvg, it scrolls some stuff and doesn't start up. |
16:15.32 | Hmmhesays | beverly hills |
16:15.35 | Hmmhesays | that's where I want to be |
16:15.42 | Hmmhesays | ASterNov that is right |
16:15.45 | Hmmhesays | don't do that |
16:15.53 | Hmmhesays | reboot your machine |
16:15.53 | AsterNov | why? |
16:16.04 | Hmmhesays | stuff in aah has to be started in order |
16:16.04 | *** join/#asterisk queuetue (n=queuetue@toronto-HSE-ppp4122670.sympatico.ca) |
16:16.25 | AsterNov | I dont want to use the asterisk_safe script |
16:17.01 | AsterNov | I want asterisk to fork on the 2.6 Kernel |
16:18.04 | paryl | i'm trying to follow http://www.asteriskguru.com/tutorials/wildcard_tdm400p.html, but the setup in zaptel.conf is confusing me, and only refers to 1 card (i have 2)... is there a better tutorial out there? |
16:18.07 | *** join/#asterisk iCEBrkr (i=icebrkr@rrcs-24-129-130-158.se.biz.rr.com) |
16:20.55 | *** join/#asterisk Simon-_ (i=byte@proxima.arlott.org.uk) |
16:23.40 | opus__ | why does DTMF fail with SIP? |
16:23.53 | file | have you specified a dtmfmode? |
16:23.57 | opus__ | yes |
16:24.05 | file | do they match on both sides? |
16:24.09 | opus__ | yes |
16:24.12 | *** join/#asterisk zetabug (n=zetabug@207.44.170.12) |
16:24.12 | file | have you checked to see if DTMF is being received? |
16:24.22 | rayvd | Have you placed a middle-aged goat near a box of feathers? |
16:24.42 | opus__ | yes its being received. |
16:24.46 | opus__ | according to RTP debug |
16:24.58 | opus__ | and Oct 7 09:24:11 DEBUG[32725]: rtp.c:251 process_rfc2833: - RTP 2833 Event: 00000002 (len = 4) |
16:25.02 | file | so what are you trying to do that uses DTMF? |
16:25.08 | *** join/#asterisk mhnoyes (n=mhnoyes@user-2ivfl1c.dialup.mindspring.com) |
16:25.13 | opus__ | Dial an extension. |
16:25.19 | opus__ | Works sometimes.. |
16:25.19 | file | using what? |
16:25.22 | file | DISA? |
16:25.37 | opus__ | Nope, just WaitExten after my * box picks up from my provider |
16:25.58 | opus__ | what do you use? |
16:26.02 | file | Background |
16:26.29 | opus__ | it works sometimes, but not all the time |
16:26.45 | opus__ | do you use 'ztdummy' ? |
16:27.12 | file | why would a timing device matter? |
16:27.28 | opus__ | I'm thinking maybe |
16:27.34 | opus__ | I've been trying to trace this problem for months |
16:27.36 | nfi|ermes | anyone can help me with some problems with voicemailbox ? |
16:28.10 | *** join/#asterisk mcreedjr (n=mcreedjr@72.240.172.15) |
16:28.15 | opus__ | For example, I just dialed out. |
16:28.20 | opus__ | It dialed out correctly. |
16:28.24 | opus__ | Now DTMF works fine |
16:29.07 | mcreedjr | Does anyone have any ideas on troubleshooting choppy VoIP using a cable modem on one end and a T1 on the asterisk side of things. I'm not sure where to start with identifying where the congestion(?) lies. |
16:29.20 | opus__ | mcreedjr - you need QoS |
16:29.28 | *** join/#asterisk leszq (n=leszq@82.177.97.254) |
16:29.34 | leszq | hiiiiii allll |
16:29.38 | nfi|ermes | hi |
16:29.42 | mcreedjr | I have it setup on both ends.. |
16:29.47 | iCEBrkr | opus__: I don't use QoS.. But QoS would help :) |
16:30.21 | mcreedjr | Opus_: The T1 side prioritizes traffic by IP address in the edge router and the Cable side of things does layer 2 QoS by MAC address. |
16:30.45 | opus__ | mcreedjr, double check everything. then tripple check it. |
16:31.27 | file | silly people who don't answer the phone |
16:32.22 | InfraRed | phones are so 19,99 |
16:33.36 | jdv79 | where's iaxy prov stuff again? |
16:34.45 | iDunno | this one was 50,00 (ish) |
16:35.18 | *** join/#asterisk myiagy (n=myiagy@200.138.215.78) |
16:35.21 | *** join/#asterisk viLeR (i=1000@66.128.47.232) |
16:37.24 | *** join/#asterisk ke4qqq (n=chatzill@68-115-212-158.static.spbg.sc.charter.com) |
16:42.13 | nfi|ermes | anyone knows how to set language for voicemail registered messages ? |
16:42.31 | christo | nunnite all |
16:42.38 | nfi|ermes | -- Playing 'vm-login' (language 'en') |
16:42.48 | nfi|ermes | en--->it |
16:43.06 | InfraRed | did you check the wiki? |
16:43.11 | InfraRed | ~docs |
16:43.15 | jbot | [docs] Documentation can be found at http://www.digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or http://www.voip-info.org/wiki-Asterisk or http://www.asteriskdocs.org or http://www.oreilly.com/catalog/asterisk |
16:44.41 | InfraRed | ~google prompt language site:www.voip-info.org |
16:44.49 | nfi|ermes | thx |
16:44.50 | nfi|ermes | :| |
16:45.00 | InfraRed | check that |
16:45.06 | InfraRed | it's 3rd result |
16:45.10 | InfraRed | +on |
16:46.46 | *** join/#asterisk dalfry (n=dalfry@gateway.ishisystems.com) |
16:48.52 | *** join/#asterisk tsetane (n=tsetane@87.252.68.0) |
16:50.30 | marc324 | a context [ccc] in extensions.conf has to match a context in zapata.conf ? |
16:51.08 | iCEBrkr | So where can I find a changelog or whatsup file for 1.2.0 beta1? |
16:51.18 | iCEBrkr | It'd be nice if there were a link to a changelog on asterisk.org |
16:54.09 | *** join/#asterisk fordvoice (n=chrisf0r@rrcs-70-61-133-91.central.biz.rr.com) |
17:00.56 | jdv79 | Oct 7 12:59:31 WARNING[20788]: chan_iax2.c:9434 load_module: Unable to open IAX timing interface: No such file or directory |
17:01.07 | jdv79 | is that a real issue? |
17:01.53 | jdv79 | Oct 7 13:00:22 WARNING[20808]: res_musiconhold.c:827 moh_register: Unable to open pseudo channel for timing... Sound may be choppy. |
17:01.56 | jdv79 | or that one:) |
17:01.59 | opus__ | yes |
17:02.01 | opus__ | it can be |
17:02.14 | jdv79 | how can i fix it - there is no digium hw in the box |
17:02.43 | bkw_ | ztdummy |
17:03.07 | jdv79 | thanks |
17:03.22 | *** join/#asterisk justinu (n=j2@72.18.13.40) |
17:03.27 | *** join/#asterisk Rav1974 (n=r@static-70-19-119-112.ny325.east.verizon.net) |
17:05.07 | *** join/#asterisk harryvv (n=none@S010600a0c93f6f7e.vs.shawcable.net) |
17:07.22 | protien | can an extension, call a number from another context |
17:08.10 | ender | protien: depends on what context you list in sip.conf |
17:08.22 | ender | protien: they'll only be able to dial contexts reachable by that context. |
17:08.26 | protien | ive got a sip on default context |
17:08.34 | protien | and i want it to call another context |
17:08.35 | protien | possible? |
17:08.43 | protien | via an extension? |
17:08.43 | ender | protien: so then they'll be able to reach all contexts that are included in default |
17:08.51 | justinu | include => othercontext |
17:08.55 | protien | oh |
17:08.56 | justinu | in default |
17:08.58 | pifiu | how do you set music on hold up in extensions.conf? |
17:08.58 | ender | yep |
17:09.10 | ender | protien: it's on by default, you'd have to turn it off IIRC |
17:09.10 | protien | but how do i then |
17:09.19 | ender | protien: you just dial it. |
17:09.30 | protien | if ive got two sets of s, |
17:09.33 | ender | protien: if the digits dialed match any ext that is in a context you i nclude, i twill dial it. |
17:09.34 | protien | which one is used |
17:09.40 | ender | protien: none. |
17:09.47 | ender | protien: s is special, for entering that context. |
17:10.04 | SwK[Work] | is opencall.org b0rked? |
17:10.06 | protien | let me tell you what i want to do |
17:10.08 | *** join/#asterisk sahafeez (n=sahafeez@67.109.14.227.ptr.us.xo.net) |
17:10.10 | protien | and maybe you can give me the best solution |
17:10.15 | protien | im building a basic menu system |
17:10.19 | ender | protien: you could have an extension, which has 'Goto(Context,s,1) |
17:10.21 | protien | i want to know the best way to call it up |
17:10.32 | protien | oh thats possible is it |
17:10.41 | ender | protien: exten => 5555,1,Goto(mainmenu,s,1) |
17:11.15 | protien | do i still need to include => mainmenuu |
17:11.19 | protien | in default context then |
17:11.23 | ender | no |
17:11.31 | ender | Gotos can jump contexts. |
17:11.40 | protien | ah then thats a good solution |
17:11.41 | protien | lemme try it |
17:12.10 | protien | :> |
17:12.12 | protien | very nice ender |
17:12.25 | ender | your welcome |
17:12.25 | protien | i can call park into that aswell right? |
17:12.34 | ender | hrm. |
17:12.37 | ender | park is different. |
17:12.47 | ender | park will take an unused park ext in a range. |
17:12.56 | ender | protien: you could blind transfer somebody to that ext. |
17:13.01 | protien | ah okay cool |
17:13.12 | protien | another thing i wanted to know is |
17:13.19 | protien | if i have a list of phones numbers |
17:13.25 | shido6 | I recommend, dentibone! |
17:13.29 | protien | i want banned, like sent to a certain extension |
17:13.35 | protien | how would i go about that, for say 10 numbers |
17:14.05 | ender | ar ethey numbers you own, or are they numbers being called from? |
17:14.18 | protien | well numbers called from |
17:14.23 | protien | that are calling into my pbx |
17:14.24 | marc324 | the context=aaaa in zapata.conf has to match a corresponding [aaaa] in extensions.conf ?? |
17:14.26 | protien | weither it be from voip or pstn |
17:14.54 | justinu | exten => did/callerid,1,Goto(banned,s,1) |
17:14.56 | ender | marc324: the context in zapata is where incoming calls will start in your extensions.conf |
17:15.14 | protien | i have to enter that for each number? |
17:15.30 | justinu | you could create a macro that checks all inbound calls |
17:15.37 | marc324 | ender-- therefore a [context] has to be present in extensions.conf ? |
17:15.47 | ender | marc324: yes, it should be. |
17:15.57 | ender | protien: 10 numbers isn't that much. |
17:15.57 | marc324 | how about for outgoing calls? |
17:16.03 | protien | tru ender |
17:16.38 | ender | marc324: outgoing calls care not about the context listed in zapata.conf |
17:16.51 | ender | protien: plus if htey are in a specific range, you can do wildcard matching. |
17:17.09 | ender | _206295426X |
17:17.23 | protien | exten => did/_8270XXXX,1,goto(banned,s,1) ? |
17:17.33 | ender | that about does it. |
17:17.35 | justinu | yep |
17:18.00 | ender | although thats a really odd phone number |
17:18.18 | protien | thats australian |
17:18.22 | ender | oh ok. |
17:18.29 | protien | im just doing a quick test |
17:18.37 | protien | my local sip is 1001, i tried banning that |
17:18.40 | *** join/#asterisk mcreedjr (n=mcreedjr@72.240.172.15) |
17:18.44 | protien | or doesnt it work local? |
17:18.51 | ender | it works local |
17:19.20 | ender | the CallerID if your Sip is most often the [foo] part of the sip.conf entry |
17:19.57 | mcreedjr | Can anyone point me on where to change the SIP jitter buffer in asterisk? |
17:19.57 | protien | yeah |
17:19.59 | protien | 1001 is mine |
17:20.14 | ender | mcreedjr: searched the wiki? |
17:20.15 | protien | exten => did/1001,1,goto(menusys,87,1) |
17:20.16 | protien | is my entry |
17:20.28 | ender | hrm, the 'did' is the target extension. |
17:20.33 | ender | or hte Direct Dial |
17:20.44 | justinu | direct INWARD dial ;) |
17:20.55 | mcreedjr | ender: Searched for "jitter buffer" in the Wiki and found articles about what the jitter buffer is, but not where to change it. |
17:20.56 | ender | so if you wanted to block 1001 from ever calling 1005, you would put 1005/1001,1,goto( |
17:20.57 | protien | oh |
17:21.07 | protien | can i go |
17:21.25 | protien | _./1001,1,goto |
17:21.29 | protien | for * |
17:21.39 | ender | yep |
17:21.46 | Qwell | You probably want _X. |
17:22.16 | ender | mcreedjr: hrm, isn't sip jitter buffer a phone level thing? I know you can adjust IAX2 jitter buffer... |
17:22.35 | mcreedjr | Ender: I don't know, thats why I'm asking :) |
17:22.46 | mcreedjr | Ender: It's beginning to look that way, just trying to verify that for sure. |
17:22.54 | ender | ah. |
17:26.27 | *** part/#asterisk joelsolanki (i=joelsola@202.160.161.93) |
17:28.08 | *** join/#asterisk redder86 (n=lee@gateway.howardsilvan.com) |
17:28.30 | redder86 | ~seen coppice |
17:28.37 | jbot | coppice <n=chatzill@127.202.17.210.dyn.pacific.net.hk> was last seen on IRC in channel #asterisk, 4d 17h 2m 59s ago, saying: 'dudes: ghostscript is the usual suspect'. |
17:28.45 | redder86 | still in India |
17:29.14 | *** part/#asterisk redder86 (n=lee@gateway.howardsilvan.com) |
17:31.19 | harryvv | in india for what |
17:32.38 | justinu | no kitty, that's mah pot pie! |
17:33.03 | Katty | kats do not eat average pot pies. |
17:33.08 | Katty | they have meat and cheeses and things |
17:33.09 | *** join/#asterisk paulc (n=Paul@216.187.75.190.novuscom.net) |
17:33.13 | justinu | no kitty, that's a bad kitty! |
17:34.19 | *** join/#asterisk Hupe (n=Hupe@iD4CC1003.versanet.de) |
17:34.48 | Katty | actually, i have an interesting little issue. |
17:35.01 | Beirdo | justinu: no need to say the next line :) |
17:35.04 | Katty | after x ammount of calls, these polycom 500s cannot hear anything. |
17:35.06 | *** join/#asterisk [Jedi] (n=hhgds4@154.Red-217-127-168.staticIP.rima-tde.net) |
17:35.08 | [Jedi] | Hello |
17:35.12 | justinu | beirdo: damn... |
17:35.23 | Katty | they can call in....the person on the other end can hear us.....and after rebooting the phone......it's fine. |
17:35.25 | [Jedi] | Anyone can help me with SpanDSP and TxFax please? |
17:35.30 | harryvv | katty, since when do 500s hear? |
17:35.40 | Katty | harryvv: since katty was born, obviously. |
17:35.41 | Qwell | harryvv: latest firmware |
17:35.49 | gandhijee | hey ingorepat => number is supposta give a tone |
17:36.00 | gandhijee | after you hit the number right? |
17:36.01 | Katty | maybe a routing issue? |
17:36.02 | Qwell | gandhijee: no, its supposed to not remove the tone, iirc |
17:36.05 | Katty | a network hiccup? |
17:36.06 | justinu | RTP send on out out of range udp port? |
17:36.16 | gandhijee | Qwell: thats what i meant |
17:36.24 | gandhijee | so is it broken in 1.0.9?? |
17:36.28 | Katty | Hmmhesays: i bet you would know (= |
17:36.32 | Qwell | gandhijee: don't think so |
17:36.36 | [Jedi] | anyone with TxFax ? |
17:36.56 | harryvv | so has the fax issue with voip been corrected? |
17:36.56 | justinu | can you capture an invite/200OK exchange with SDP on a call that doesn't get RX audio? |
17:37.10 | paulc | JerJer: You around? |
17:37.27 | Katty | Hmmhesays: right. when you come back. these polycom 500s are ruining my 21st birthday. |
17:37.35 | Katty | Hmmhesays: i demand you stabity them with your hair. |
17:37.51 | Beirdo | Katty: happy birthday, BTW |
17:37.56 | Katty | Beirdo: thanks (= |
17:38.17 | Beirdo | hehe |
17:38.25 | Katty | mew :< |
17:39.05 | Beirdo | use the claws on it, maybe it will smarten up |
17:40.20 | justinu | is it possible that the polycom's have an upper RTP port limit, and that asterisk eventually starts sending the RTP to the polycom above that limit? |
17:40.26 | ender | gandhijee: your phone is more likely to handle the tone. |
17:40.29 | gandhijee | then is there something wrong with my inore pat |
17:40.35 | ender | gandhijee: the ignorepat is more for analog phones. |
17:40.39 | gandhijee | http://pastebin.com/386351 |
17:40.54 | gandhijee | ender: i know, i have it setup and it seems right to me |
17:40.56 | ender | gandhijee: are you using a sip phone? |
17:41.10 | gandhijee | show dialplan shows that its there |
17:41.17 | gandhijee | a buncha dif phones |
17:41.27 | [Jedi] | anyone with TxFax ? |
17:41.35 | gandhijee | i have 3 analogs off a TDM400, an IAX softclient and SIPs softclient |
17:41.50 | ender | gandhijee: well, w/ sip phones, the digitmap on the phone itself will take care of the tone generation. |
17:41.55 | gandhijee | ok |
17:42.08 | gandhijee | when i hit 9 or 8 on my analogs |
17:42.15 | gandhijee | i lose the tone, but i can still dial |
17:42.23 | ender | gandhijee: and your analog phones, do you have the right context= in zapata.conf so that they are by default sent to the context that has the ignorepat ? |
17:44.08 | Katty | Beirdo: nothing in the cli about my errrr :< |
17:44.19 | Beirdo | hmmm, that sucks |
17:44.31 | Katty | i really think it's a routing issue |
17:44.33 | Beirdo | don't let it ruin your day :) |
17:44.40 | Katty | pfft |
17:44.53 | Katty | i have to reboot my phone 8 times a day |
17:44.55 | justinu | i gave you some suggestions, but you don't listen |
17:45.23 | Katty | justinu: if you're talking to me, you need to let me know. |
17:45.37 | Katty | justinu: with as many conversations as go on in here...heh, it's hard telling who you're talking to. |
17:45.38 | gandhijee | ender: i thikn so |
17:45.46 | justinu | who else was asking about polycoms not receiving audio? |
17:45.47 | justinu | der |
17:46.08 | Katty | justinu: i don't read every line in this channel. |
17:46.13 | [Jedi] | I'm the only one in the world trying to use txfax in asterisk 1.1 or 1.2? |
17:46.20 | Katty | justinu: in fact, i'm not even in here most of the time (= |
17:46.23 | justinu | oh well |
17:46.25 | Beirdo | :) |
17:46.34 | gandhijee | ender: i have a dial_pstn context |
17:46.35 | Katty | justinu: i'll just wait for Hmmhesays or anthm |
17:46.41 | Katty | justinu: they speak kat well. |
17:46.43 | justinu | lol |
17:46.45 | Katty | justinu: you do not. |
17:46.52 | justinu | whatever you say |
17:46.56 | Katty | kthx |
17:46.59 | *** join/#asterisk jeffgus (n=jeffgus@2002:d856:c704:0:0:0:0:1) |
17:47.02 | gandhijee | ender: that i include in my analog_handsets context, which is what is defined in my zapata.conf |
17:47.05 | ender | gandhijee: and in zapta.conf for the lines that you have analongs plugged into, they have 'context=dial_pstn' ? |
17:47.22 | *** join/#asterisk danalien (n=danalien@unaffiliated/danalien) |
17:47.33 | ender | gandhijee: try shortcutting the include and go straight to the dial_pstn |
17:48.11 | *** join/#asterisk bsd3 (n=bsd@203.134.193.168) |
17:48.32 | gandhijee | ender: no in zapata.conf the handset lines are def a zap_lines |
17:49.02 | gandhijee | ender: then there is a dial_pstn context to access theat |
17:49.19 | gandhijee | which is included in under a section called dialables |
17:49.28 | gandhijee | then dialables is included in zaplines |
17:50.24 | *** part/#asterisk bsd3 (n=bsd@203.134.193.168) |
17:50.37 | *** join/#asterisk FABRIZIOxxx (n=FABRIZIO@81-208-26-86.ip.fastwebnet.it) |
17:50.47 | mutilator | anyone ever admined a teamspeak server? |
17:51.08 | marc324 | ~docs |
17:51.09 | jbot | docs is, like, Documentation can be found at http://www.digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or http://www.voip-info.org/wiki-Asterisk or http://www.asteriskdocs.org or http://www.oreilly.com/catalog/asterisk |
17:51.42 | ender | gandhijee: so again, try shortcutting all that including mishmach and go straight to the context that has the ignorepat |
17:51.51 | FABRIZIOxxx | hi guys .. when i call through a zap channel i'm getting a very soft sizzle on the background .. everything else is ok .. could this be caused by bad and/or long cables ..?? I couldn't find anything on voip-info.org regarding this problem .. |
17:52.03 | ender | anybody smell bacon? |
17:52.07 | justinu | lol |
17:52.07 | [Jedi] | anyone with TxFax ? |
17:52.11 | [Jedi] | :((((( |
17:52.12 | *** join/#asterisk Godsey (n=lanny@pdpc/supporter/sustaining/Godsey) |
17:52.18 | justinu | ender: it's sizzlean |
17:52.26 | Katty | [Jedi]: sorry, i don't know anything about that. |
17:52.27 | bjohnson | FABRIZIOxxx: maybe try adjusting gains to turn down volume |
17:52.34 | bjohnson | [Jedi]: yes. but not me |
17:52.35 | gandhijee | ender: ok |
17:52.44 | [Jedi] | I'm going crazy |
17:52.46 | [Jedi] | with it |
17:52.54 | bjohnson | sounds very appealing |
17:52.56 | gandhijee | ender: but like i said, under the show dialplan they ingnore pat shows up |
17:53.13 | FABRIZIOxxx | bjohnson, already tried .. but i'm getting complaints about low voice volume .. and at the moment the tx and rx are set to 3.5 .. i don't think its very hig .. |
17:53.18 | [Jedi] | TxFax gets executed but does nothing |
17:53.38 | bjohnson | low voice at which end? |
17:53.50 | bjohnson | the rx end? |
17:53.57 | gandhijee | anyone also know where i can get a hold of some decent sips phones for kinda cheap |
17:54.05 | gandhijee | none of those grandstream craps |
17:54.09 | bjohnson | sip? never heard of it |
17:54.10 | FABRIZIOxxx | yes .. people from outside my * server |
17:54.23 | gandhijee | well how about an IP phone then |
17:54.29 | bjohnson | FABRIZIOxxx: increase tx and decrease rx |
17:54.41 | bjohnson | gandhijee: you pee? |
17:54.46 | FABRIZIOxxx | ok .. i'll try straight away ... thks |
17:54.50 | ender | gandhijee: voipsupply/cdw Polycom phones. |
17:55.41 | gandhijee | so i guess voipsupply is the cheapest, nuts |
17:55.45 | bjohnson | gandhijee: all the cheap variants are reported to feel cheap .. what is "decent phone" to you |
17:56.10 | justinu | um, atacomm sells polycom 501s for cheaper than voipsupply |
17:56.11 | justinu | 169 |
17:56.47 | gandhijee | i have one of those phones from iareaphone. doesn't feel cheap |
17:56.52 | paulc | Has anyone had problems with failed call attempts through NuFone in the past couple of hours? I've had a customer complaining about congestion + audio quality problems but have been unable to replicate myself. |
17:56.55 | *** join/#asterisk halogen8 (n=halogen8@66-146-190-146.skyriver.net) |
17:57.53 | halogen8 | I'm looking at the OEM X100P Digium cards on ebay, and wanted to know what people thought of them? |
17:58.05 | halogen8 | are they absolute garbage? do they work ok, or great? |
17:58.16 | gandhijee | i found a place on the net that sold them for 10 bux |
17:58.20 | gandhijee | like 15 with shipping |
17:58.30 | gandhijee | worked fine for me |
17:58.41 | halogen8 | gandhijee: you have a link to that place? |
17:58.45 | *** part/#asterisk n0where (n=kc@q041140.ppp.asahi-net.or.jp) |
17:58.50 | gandhijee | i can try and google it back up |
17:58.58 | gandhijee | i bought it about a year ago |
17:59.08 | gandhijee | give me a min |
17:59.19 | gandhijee | i gotta order some ip phones and ata' |
17:59.21 | gandhijee | s first |
18:01.29 | FABRIZIOxxx | bjohnson, actually its quite better .. but i can hear still a little bit ... i'll check the cables and buy some new ones since the ones i have now are ahdn made .. and about 25 meters long.. |
18:02.17 | Katty | http://groups.google.com/group/Asterisk-users/browse_thread/thread/91e6002e0ed66504/704f829272d6d75f?hl=en#704f829272d6d75f <- mine issue. |
18:02.24 | halogen8 | anyone know where i can find the diff between asterisk@home 1.5 and the 2.0 betas? |
18:02.43 | [Jedi] | bkw_ are you there? got a min? |
18:03.12 | gandhijee | FABRIZIZOxxx: make sure your not running your phone line parallel to any power lines |
18:03.30 | gandhijee | and a ferite core to wrap your line around wouldn't hurt either |
18:03.43 | Qwell | halogen8: They are absolute garbage - and they ARE NOT Digium cards. |
18:03.55 | Katty | justinu: looks like the port is 2226 on rtp debug |
18:04.07 | FABRIZIOxxx | gandhijee, thks for the info .. i'll definately check that |
18:04.27 | znoG | Qwell: not sure about "absolute garbage", I have one and it works fine. |
18:04.32 | justinu | katty: is that what port the pbx is sending audio to you on? |
18:04.44 | Qwell | znoG: Would you use one in a production system? |
18:04.49 | halogen8 | Qwell: why do you say they are garbage? what problems have you had with them? |
18:04.51 | Qwell | I most certainly would not. |
18:04.52 | Katty | justinu: both directions |
18:05.14 | justinu | katty: is this a call you didn't receive audio on? |
18:05.19 | Qwell | halogen8: They're unsupported, for one. |
18:05.30 | Katty | justinu: no |
18:05.31 | Qwell | and they're just...cheap |
18:05.35 | Katty | justinu: i enabled rtp debug |
18:05.35 | [Jedi] | Zaphfc bri's are unsupported too |
18:05.39 | Qwell | "you get what you pay for" |
18:05.40 | [Jedi] | and work perfect |
18:05.44 | [Jedi] | and cost 15 EUR |
18:05.48 | justinu | katty: my guess is that subsequent calls will incrment the port number |
18:05.55 | Katty | now it's 2228 |
18:06.01 | Katty | justinu: mew? |
18:06.13 | halogen8 | Qwell: yeah, but right now, i can't really justify spending alot on a FXO card......unless those OEM cards just won't work |
18:06.23 | gandhijee | Qwell: from the hardware standpoint they are the exact same thing |
18:06.25 | Qwell | halogen8: it's hit or miss |
18:06.32 | halogen8 | Qwell: I've heard they shouldn't be used for production, but I just wanna know what problems people have had with them |
18:06.36 | gandhijee | Qwll: on production, yeah don't use them |
18:06.40 | znoG | Qwell: no, hell no, this is for my home system. You never asked halogen8 what kind of system he was putting it in, did you? |
18:06.41 | *** join/#asterisk MnDBnDr (n=MnDBnDr@198.234.224.6) |
18:06.50 | gandhijee | but if you are just tryin to get your hands wet, its the way to go |
18:07.09 | gandhijee | 10 bux v 100 bux.... |
18:07.16 | halogen8 | Yeah, I'm just using this as a learning experiment.....can't justify spending alot on it yet |
18:07.18 | Katty | too confusing. not sinking in. |
18:07.24 | gandhijee | yeap |
18:07.30 | znoG | halogen8: call quality, echo problems, volume issues.. mine seems to work fine apart from maybe a slight volume problem and occasional caller ID issues, but I can live with that. |
18:07.31 | gandhijee | i did the same thing halogen |
18:07.38 | justinu | katty: my guess is that eventually when the port eventually gets above a certain number, you will stop getting audio |
18:07.50 | MnDBnDr | can someone point me in the right direction for a good doc to configure oh323 on AAH? |
18:07.54 | gandhijee | the asterisk card just has a resistor moved to change its ID number. thats all |
18:07.56 | halogen8 | znoG: thanks for the examples...... |
18:08.09 | iCEBrkr | znoG: Yea, same here, I don't get consistant volume with calls.. but that's my only problem. |
18:08.21 | justinu | katty: when you reset your phone, you're causing the phone to start at the lower part of the port range again, thus making it work |
18:08.22 | gandhijee | halogen8: i don't seem to have a problem with mine |
18:08.49 | halogen8 | it just seems, it would be a good card to learn on, then when it goes to production, use the $100 real digium card |
18:08.51 | gandhijee | none of the issues these guys are talkin about, then again my line might be "cleaner" than theres |
18:09.14 | halogen8 | gandhijee: thanks.....i think i'll try it.....cus its just so cheap |
18:09.24 | gandhijee | halogen8: if you go in to production, get the TDM400P, that way you can expand if u just use POTS lines |
18:09.27 | iCEBrkr | gandhijee: I used mine for years. It was fine.. But I did have an few volume problems. Just had to tweak my gain |
18:09.40 | Katty | justinu: i see. |
18:09.51 | Katty | justinu: there's no port range specifications on the phone's settings though |
18:09.51 | gandhijee | iCEBrkr: i've only had mine for about a year |
18:10.08 | iCEBrkr | It worked well enough that ALL my calls went through my asterisk box. |
18:10.09 | justinu | katty: hmm... you can change the rtp ports asterisk uses in rtp.conf, but I'm not sure if that'll help you. |
18:10.14 | iCEBrkr | So I really guess I can't complain |
18:10.14 | halogen8 | gandhijee: yes, that is the card I would like......but its just too costly right now......looks like a great card though |
18:10.25 | Katty | justinu: i don't even know it's rtp causing it. |
18:10.31 | Katty | justinu: i'm not going to go poking about my conf files. |
18:10.42 | justinu | well, one way audio is usually an RTP problem |
18:10.52 | Katty | justinu: i'll ask elsewhere first. |
18:10.57 | justinu | if you can't hear, you're not receiving the RTP stream correctly. |
18:10.58 | Katty | justinu: second opinions good. |
18:11.16 | gandhijee | halogen8: it is =) i just got mine yesterday |
18:12.09 | halogen8 | gandhijee: how much did you pay? I found this: http://voipstore.atacomm.com/shops/ViewItem.aspx/27934028032-45624523776.htm |
18:12.18 | halogen8 | looks pretty cheap without the adapter cards |
18:12.30 | MnDBnDr | can someone point me in the right direction for a good doc to configure oh323. It is compiled and installed. |
18:12.39 | gandhijee | yeah it is, add bout 70 bux i think for each module |
18:12.52 | gandhijee | my card cost 274, 1 FXO 3 FXS |
18:13.10 | MnDBnDr | just got my hands on 3 symbol h323 phones. 802.11b |
18:13.45 | halogen8 | yup, thats looks like a good price |
18:13.52 | halogen8 | i think i just want one with 2 FXO ports |
18:13.57 | gandhijee | anyone know of a decent ATA too? |
18:14.01 | halogen8 | with a way to add 2 more FXO cards |
18:14.22 | gandhijee | halogen8: if you are just playing, get the 2 clones |
18:14.38 | gandhijee | halogen8: and just use softphones for now. till you are comfortable |
18:14.54 | halogen8 | gandhijee: yes.....I will get the clones to learn, then if they ever go to production, I will get the good one |
18:15.16 | gandhijee | halogen8: good idea |
18:15.37 | halogen8 | gandhijee: I have 5 Linksys PAP2 adapters that I can use with it.....so don't need softphones |
18:16.00 | halogen8 | gandhijee: staples just had a great deal on the PAP2's....they were $50/ea with a $50 rebate |
18:16.14 | *** join/#asterisk DagMoller (n=DagMolle@2001:5c0:8fff:ffff:0:0:0:2b) |
18:16.23 | Qwell | halogen8: locked for vonage |
18:16.27 | gandhijee | halogen8: aren't they tied to like vonage though |
18:16.34 | halogen8 | gandhijee: easily unlocked |
18:16.41 | gandhijee | sipath?? |
18:16.42 | halogen8 | all of mine are unlocked now |
18:16.46 | justinu | i wish I could get the rtp300 unlocked |
18:16.47 | Qwell | somebody needs to toss me one of those pap2's, heh |
18:16.49 | gandhijee | what you do to unlock them? |
18:16.49 | DagMoller | asterisk -r without colors in CLI, why? |
18:17.16 | gandhijee | cuz if thats the case, i'll run to worstbuy right now and buy 2 |
18:17.21 | halogen8 | gandhijee: lemme find you the forum that talks about it......its actually quite easy.....if you get it before VONAGE updated the firmware |
18:18.06 | halogen8 | http://www.broadbandreports.com/forum/remark,14450684 |
18:18.11 | syle | dagmoller: cause your most likely not starting asterisk with -vvvgc options |
18:18.51 | gandhijee | sweeet |
18:18.52 | gandhijee | thanx man |
18:18.55 | halogen8 | gandhijee: I dunno if they still have the rebates or what, but they are only about $50 without the rebate....... |
18:19.06 | gandhijee | yeah, that better than the other prices. |
18:19.08 | halogen8 | gandhijee: sure, enjoy.....thanks for the info......gotta get back to work....ttyl |
18:19.14 | gandhijee | later |
18:20.41 | DagMoller | syle, -c if for console, and i dont want console at start time!! |
18:20.51 | syle | why not |
18:20.54 | DagMoller | syle, sorrry my english! |
18:20.56 | syle | i just start it in a screen session |
18:22.16 | DagMoller | because asterisk can be run as a daemon. |
18:22.37 | syle | up to you man |
18:23.49 | *** join/#asterisk syle2 (n=blag@unaffiliated/syle) |
18:24.17 | DagMoller | syle, its not make sense! |
18:26.48 | *** join/#asterisk stkn (n=stkn@gentoo/developer/pdpc.active.stkn) |
18:27.33 | syle2 | dagmoller: /usr/bin/screen -L -d -m -S asterisk /bin/nice -n -19 /usr/sbin/asterisk -U asterisk -vvvgc ...that make more sense to you? |
18:28.09 | DagMoller | lol |
18:28.11 | *** join/#asterisk h4mm3r` (n=h4mm3r@85-18-14-10.fastres.net) |
18:30.30 | MnDBnDr | can someone point me in the right direction for a good doc to configure oh323 |
18:37.39 | *** join/#asterisk azzie (n=az@azzie.net) |
18:38.17 | ender | um. |
18:38.23 | ender | why put asterisk into screen? |
18:38.28 | marc324 | what port does sip use? |
18:38.41 | ender | asterisk has a console you can attach to w/ -r |
18:39.16 | ender | marc324: a bunch of them. |
18:39.25 | marc324 | 5560? |
18:39.45 | ender | 5060 is the main one. |
18:39.46 | ender | udp |
18:40.41 | MicC_ | anyone try backing up and restoring voicemails to a new system? |
18:40.50 | MicC_ | does some strange stuff. |
18:41.13 | MicC_ | won't let you record a voice mail on the new system. |
18:41.32 | jontow | micc; ? |
18:41.46 | MicC_ | its hard to explain. |
18:41.56 | MicC_ | I have 2 * boxes...1 primary and 1 backup |
18:42.00 | jontow | shouldn't be.. the format is pretty..reasonable |
18:42.01 | groogs | MicC_: permissions |
18:42.10 | groogs | MicC_: vm files are just that .. files |
18:42.22 | DagMoller | ender, why in -r mode dont have colors? |
18:42.23 | ender | perms on the directories for VMs |
18:42.24 | MicC_ | groogs: that was my first thoguht....I tar''d them and verified perms |
18:42.30 | jontow | the files won't differ from box to box, there isn't anything unique about 'em.. |
18:42.30 | ender | DagMoller: hrm, dunno. |
18:42.52 | jontow | when you ran tar, did you give it the 'p' option? (both when creating the archive and when extracting it?) |
18:43.11 | groogs | MicC_: well, there's nothing else to it. if the permissions are right and you can't record vm, it has nothing to do with backing up / restoring .. you probably can't record ANY voicemails, for whatever reason |
18:43.33 | MicC_ | jontow: to keep the permissions. No not really. I have never had a problem having perms changed on an extract |
18:43.44 | jontow | double check it, then |
18:43.47 | groogs | well, what ARE the permissions? |
18:43.48 | jontow | all files, all directories |
18:43.55 | ender | MicC_: specifically user and group ownership. |
18:44.01 | MicC_ | groogs: if I blow away all the voicemails it works. |
18:44.09 | groogs | (ownership, and mode) .. and what user/group is asterisk running as? |
18:44.16 | MicC_ | ender: boxes are exact mirrors of each other UID and GUID :P |
18:44.45 | groogs | i guarantee its a permission problem because well, it can't be anything else |
18:44.57 | *** join/#asterisk brookshire (n=matt@gateway.digium.com) |
18:44.59 | groogs | check permisions on the voicemail directories and their parents |
18:45.07 | jontow | just isn't much to fuck up with the voicemail structure.. its solid |
18:45.10 | MicC_ | I came to the conclusion it must keep a counter (msg.001) somewhere in the databas(mysql backend) |
18:45.37 | MicC_ | jontow: it seems like a no brainer...backup...then restore. |
18:46.20 | MicC_ | my goal is to eventually store the voicemail from both the primary and backup on the same SAN device. |
18:46.37 | MicC_ | that will help me make a seamless failover from one box to the other. |
18:51.44 | MicC_ | I will find the culprit and expose it...If its DFU (me) you might now hear anything :P |
18:51.53 | MicC_ | now=not |
18:54.17 | *** join/#asterisk cio (n=na@adsl-072-149-159-016.sip.bhm.bellsouth.net) |
18:54.57 | cio | Hi all. When a remote sip phone registers on say port 63686, is that an RTP port? |
18:55.12 | *** join/#asterisk arguile (i=user224@66.38.201.234) |
18:56.15 | zetabug | so I got my extensions working, to get behind a NAT. Now how do I make it so I hear a ring ring, until they pick up the line on their end? |
18:56.17 | *** join/#asterisk kue (n=Administ@CPE-70-94-56-196.wi.res.rr.com) |
18:56.22 | zetabug | (sip) |
18:56.56 | zetabug | this is what I have: 'exten => 6666,1,Dial(SIP/bob,,rm)' |
18:57.10 | zetabug | and it works. but no ringing :( |
18:59.15 | *** join/#asterisk pc2 (n=pc@209.151.52.81) |
18:59.36 | *** part/#asterisk kue (n=Administ@CPE-70-94-56-196.wi.res.rr.com) |
18:59.49 | pc2 | Ok -- what does an "IXC interconnect" mean as far as telephone lines go? |
19:03.28 | *** join/#asterisk cio_flood (n=na@adsl-072-149-159-016.sip.bhm.bellsouth.net) |
19:03.54 | cio_flood | When a sip phone registers on a high port, like 62823 or something, is that a Dynamic Nat port or a RTP port? |
19:05.06 | cio_flood | Here's what I'm getting: NOTICE[1524]: chan_sip.c:8034 sip_poke_noanswer: Peer '804' is now UNREACHABLE! |
19:05.17 | *** join/#asterisk harryvv (n=none@S010600a0c93f6f7e.vs.shawcable.net) |
19:05.18 | cio_flood | I get this immediately after it registers. |
19:06.49 | cio_flood | Anyone even here? |
19:09.27 | marc324 | what port does xlite use? |
19:09.31 | *** join/#asterisk wolfson (n=hehe@usr-kdh-208-6-58-26.beachlink.com) |
19:10.03 | *** join/#asterisk kue (n=Administ@CPE-70-94-56-196.wi.res.rr.com) |
19:10.13 | paulc | Has anyone had problems with failed call attempts through NuFone in the past couple of hours? I've had a customer complaining about congestion + audio quality problems but have been unable to replicate myself. |
19:10.47 | hardwire | do you qualify your customers? |
19:11.25 | *** join/#asterisk Gunnar (n=gunnar@34.80-203-209.nextgentel.com) |
19:12.56 | *** join/#asterisk stkn_ (n=stkn@gentoo/developer/pdpc.active.stkn) |
19:13.18 | *** join/#asterisk razu_ (n=razu@213-35-173-39-dsl.prn.estpak.ee) |
20:44.09 | *** join/#asterisk jbot (i=ibot@pdpc/supporter/active/TimRiker/bot/apt) |
20:44.09 | *** topic/#asterisk is Asterisk 1.2.0 Beta1 - http://www.asterisk.org || Astricon 2005 - Anaheim, CA - Oct 12-14 - http://www.astricon.net/2005 - Sign Up Now! |
20:44.10 | Qwell | Katty: I was mostly just joking, until you asked. :p |
20:44.17 | Katty | what a weirdo |
20:44.36 | Qwell | psuedo-meat is like...that fake sausage they put on pizza |
20:44.44 | Qwell | soy protein or whatever |
20:44.49 | Katty | it's real sausage |
20:44.51 | Katty | i checked. |
20:44.58 | Qwell | nah, veggie pizza |
20:44.59 | Katty | just not the grade you're used to |
20:45.05 | Qwell | my friend bought some one day |
20:45.14 | Katty | mew? |
20:45.20 | Katty | veggie pizza shouldn't have meat on it |
20:45.22 | Qwell | tasted just like sausage and pepperoni...but it wasn't |
20:45.25 | Katty | most pizzas still aren't vegan. |
20:45.48 | file[laptop] | TastyKatty(tm) |
20:45.59 | Katty | file[laptop]: mew? ^_- |
20:46.05 | Qwell | Katty: So, what don't you eat, exactly? |
20:46.17 | Katty | Qwell: don't eat meat, dairy, honey, etc. |
20:46.22 | Qwell | honey? wow |
20:46.29 | Katty | Qwell: no animals or animal biproducts. |
20:46.37 | Qwell | do you wear cloth? |
20:46.39 | Katty | Qwell: some animal biproducts, if free range, i will eat |
20:46.51 | Alives | how do you place a call on hold in asterisk? |
20:46.54 | Katty | Qwell: no wool or angora fur or leather, etc. |
20:47.37 | Katty | Qwell: and no, my shoes are not made of leather. |
20:47.58 | Katty | Qwell: i do not wear makeup, and my conditioners are not tested on animals, nor do they contain animal. |
20:47.58 | Qwell | croc? |
20:48.18 | Qwell | Katty: examples of free range animal biproducts? |
20:48.20 | shido6 | the hold button, Alives, or offhook quickly |
20:48.26 | Katty | Qwell: free range honey |
20:48.36 | Katty | Qwell: it is illegal to buy free range milk |
20:49.43 | Qwell | Katty: Do they actually market and sell free range honey? |
20:49.59 | *** join/#asterisk marc324 (n=marc3234@206-248-159-253.dsl.teksavvy.com) |
20:50.12 | nestAr | PIZZA! |
20:50.17 | nestAr | i could never be vegan |
20:50.21 | Qwell | hmm...what about mushrooms? Do you eat mushrooms? |
20:50.24 | Qwell | nestAr: yeah, totally |
20:50.31 | marc324 | is the te411p worth an upgrade from te410p? |
20:50.35 | nestAr | i did the whole ovo lacto veggie thing for a while. |
20:50.39 | nestAr | but i <3 cheese |
20:51.03 | Qwell | I know vegetarians who eat cheese and eggs...kinda weird |
20:51.06 | scsponger | hey does anyone what the headset command is to put someone on hold? |
20:51.22 | nestAr | Qwell: that's ovo lacto |
20:51.30 | Qwell | oh |
20:51.35 | Alives | shido6: its not working |
20:51.39 | Qwell | nestAr: and fish? |
20:51.42 | Alives | do i have to dial a * command? |
20:51.46 | Qwell | co-worker eats fish too |
20:52.11 | nestAr | i ate fish occasionally |
20:52.28 | nestAr | this was when i was a teenager |
20:52.44 | Qwell | nestAr: glad to hear you grew out of it :P |
20:52.55 | nestAr | I LOVE LAMP |
20:52.57 | nestAr | err steak. |
20:54.51 | Katty | Qwell: of course. |
20:54.59 | Qwell | Katty: to which? |
20:55.00 | Katty | Qwell: are mushrooms animals? |
20:55.08 | Qwell | no, but the way some of them are grown... |
20:55.14 | Qwell | not sure that would constitute a biproduct |
20:55.19 | Katty | Qwell: mew? |
20:55.32 | Qwell | you know...the fun kind? |
20:55.36 | Katty | mrow? |
20:55.39 | Qwell | nevermind |
20:55.42 | Katty | k |
20:56.52 | Qwell | I bet fungi have souls though |
20:58.00 | iDunno | mouldy souls. |
20:59.42 | Qwell | alright, I'll shut up now |
21:02.01 | *** part/#asterisk colinm_ (n=colol@VDSL-130-13-9-157.PHNX.QWEST.NET) |
21:03.24 | *** join/#asterisk mcf3782 (n=mcf3782@adsl-19-83-37.asm.bellsouth.net) |
21:05.33 | mcf3782 | Anyone in here had any luck getting kphone to work with Asterisk? I could use a little debugging help, as I've done everything I can think of. |
21:06.53 | *** join/#asterisk abcbooze (i=abcbooze@70.153.216.92) |
21:06.55 | abcbooze | sup |
21:07.10 | abcbooze | anyone have a cisco 12SP+ phone? |
21:09.34 | mcf3782 | Asterisk seems to think that my kphone client is connecting and registering, as it prints messages to the console like: "-- Registered SIP 'kphone' at 200.200.200.123 port 5060 expires 900" |
21:09.43 | gandhijee | umm so can my IAX local phones not call my SIP local phones? |
21:11.24 | mcf3782 | When I try and call an extension on my Asterisk server from kphone, I see messages in the Asterisk console like: "Oct 7 16:14:49 WARNING[4222]: chan_sip.c:695 retrans_pkt: Maximum retries exceeded on call 1116616469@200.200.200.123 for seqno 3362 (Non-critical Response)" |
21:12.27 | ManxPower | Can anyone help me debug a jitterbuffer issue? |
21:12.46 | *** join/#asterisk Gibster (n=hnd@paradise.baysider.com) |
21:13.07 | *** part/#asterisk mkrufky (n=mk@68.160.103.77) |
21:14.27 | harryvv | sixtel sound quality seems to have really improved since thay made some changes. |
21:15.06 | harryvv | Does anyone know if netzero is selling voip services? I dont see it on there page yet my perents say thay pay 16 dollars a month for long distance though netzero. |
21:15.14 | Gibster | Hi. I'm battling to get asterisk to listen on port 5060. I cant find anything in the logs and starting asterisk with -vvv shows starting sip and listening on interface fine. I have set listenaddress and port in sip.conf and I'm finally running out of ideas. Could someone please point me in the right direction. |
21:15.45 | ManxPower | Gibster, what makes you think Asterisk is NOT listening on port 5060? |
21:16.32 | gandhijee | hey can one not dial IAX to SIP and SIP to IAX? |
21:16.33 | Gibster | I get a timeout when trying to use a sip client and netstat doesnt show anything on port 5060 |
21:16.51 | ManxPower | Gibster, sounds like a NAT problem. |
21:17.12 | ian_k | iptables -F |
21:17.33 | *** join/#asterisk Rav1974 (n=r@static-70-19-119-112.ny325.east.verizon.net) |
21:17.41 | ian_k | gandhijee: no, unless you have asterisk between them |
21:17.58 | Rav1974 | does amp support asterisk 1.2 |
21:18.04 | Primer | ian_k: heh, that's not A Good Thing™ |
21:18.05 | Gibster | hmmm. I'm not doing any NAT on this box. should I try setting the nat address anyway? |
21:18.13 | ManxPower | gandhijee, you can call between protocols just fine. |
21:18.18 | Primer | ian_k: especially if you're remote, and your default policy is DROP |
21:18.19 | ian_k | Primer: what isn't? |
21:18.25 | Primer | <ian_k> iptables -F |
21:18.51 | ManxPower | Gibster, in sip.conf [general] set port=5060 and do NOT have a bindaddr= line |
21:18.54 | syle2 | gibster: it should be a udp port |
21:18.55 | Primer | default INPUT policy, that is |
21:19.05 | Primer | ManxPower: why not? |
21:19.13 | ian_k | Primer: true.. most people that can't determine why a port isn't open won't have an overly complex firewall though. Most distros have some lame default that is easily disposed of. |
21:19.15 | Primer | why not have a bindaddr? I use it...works fine |
21:19.18 | ManxPower | Primer, older versions od Asterisk had problems with it. |
21:19.28 | ManxPower | and really, most people don't need it. |
21:19.33 | Primer | ManxPower: ahhh |
21:19.52 | ManxPower | without a bindaddr, asterisk should bind to all IPs. |
21:20.05 | ian_k | does it bind to yours? |
21:20.09 | Primer | ian_k: all I'm saying is I've been bitten by that once when a dude with sudo on my box ran iptables -F when I was on vacation |
21:20.16 | Primer | and he was supposed to know what he was doing |
21:20.36 | Primer | asterisk 27133 root 15u IPv4 4037522 UDP 216.239.132.121:5060 |
21:21.03 | Primer | works for me |
21:21.03 | ian_k | Primer: I agree. I should have probably not put that in command-line-ready form. |
21:21.28 | Primer | ian_k: this dude got said command from a friend on IRC ;) |
21:22.38 | ian_k | Primer: Ah.. |
21:24.13 | *** join/#asterisk florz (i=nobody@2001:1a50:503c:0:0:0:0:1) [NETSPLIT VICTIM] |
21:24.27 | gandhijee | *CLI> -- Registered SIP '2003' at 201.1.1.116 port 5060 expires 1800 |
21:24.27 | gandhijee | <PROTECTED> |
21:24.27 | gandhijee | <PROTECTED> |
21:24.27 | gandhijee | Oct 7 21:07:03 WARNING[682]: channel.c:1891 ast_request: No translator path exists for channel type IAX2 (native 63502) to 512 |
21:24.27 | gandhijee | Oct 7 21:07:03 NOTICE[682]: app_dial.c:764 dial_exec: Unable to create channel of type 'IAX2' |
21:24.52 | Rav1974 | does amp support asterisk 1.2 |
21:25.43 | ManxPower | gandhijee, you have a CODEC issue. |
21:26.09 | gandhijee | ok |
21:26.10 | Gibster | ManxPower: thanks. seems to be binding now :) |
21:26.12 | ian_k | gandhijee - Does your iax phone work at all? ever? |
21:26.15 | gandhijee | yeah |
21:26.26 | gandhijee | i had speex as the first on |
21:26.29 | gandhijee | i just changed it |
21:26.32 | gandhijee | gonna try it now |
21:26.41 | ManxPower | gandhijee, One side is trying to use SpeeX and one side is trying to use something else. |
21:26.42 | ian_k | gandhijee - use ulaw on both and see what happens |
21:28.19 | gandhijee | just did |
21:28.19 | gandhijee | works |
21:28.23 | gandhijee | gotta go |
21:28.24 | gandhijee | thanx |
21:32.58 | *** join/#asterisk Chotaire (i=chotaire@chotaire.net) [NETSPLIT VICTIM] |
21:36.12 | *** join/#asterisk viLeR (i=1000@66.128.47.232) |
21:36.44 | *** join/#asterisk Simon- (i=byte@proxima.arlott.org.uk) [NETSPLIT VICTIM] |
21:41.06 | *** join/#asterisk Corydon-w (i=purple@pdpc/supporter/sustaining/Corydon76-home) [NETSPLIT VICTIM] |
21:41.12 | *** join/#asterisk A-Tuin|work (n=A-Tuin@nat.office.legend.net.uk) [NETSPLIT VICTIM] |
21:43.03 | *** join/#asterisk Hupe (n=Hupe@iD4CC1003.versanet.de) |
21:43.39 | Katty | hmm |
21:44.50 | *** join/#asterisk zeedo (n=zeedo@obsidis.org) |
21:45.25 | Katty | file[laptop]: are you in montreal? |
21:46.05 | abcbooze | anyone have a cisco 12SP+ phone? |
21:46.12 | abcbooze | and have config'd it for use with * |
21:47.21 | Hupe | Hi, is it a known Bug that Asterisk-System-Load grows to 100% when sip register fails for a longer time? |
21:48.04 | *** part/#asterisk halogen8 (n=halogen8@66-146-190-146.skyriver.net) |
21:48.36 | ManxPower | Hupe, no. |
21:49.37 | Nyvar | abc, yuk no. you look at the tiki page for 12SP+ phones? |
21:49.51 | lancey | hi all |
21:49.56 | Katty | wikiwikiwikiwiki mushroom mushroom |
21:51.05 | *** join/#asterisk flynux (i=hmj4hun@cl-8.bru-01.be.sixxs.net) [NETSPLIT VICTIM] |
21:51.35 | uuuppz | anyone here done much using BRI cards in NT mode? |
21:51.44 | uuuppz | I'd really appreciate some advice :) |
21:53.16 | *** join/#asterisk SkramX (n=mark@mark-s.net) |
21:53.22 | Katty | skram. |
21:53.31 | SkramX | Hello, Katty ...? |
21:53.43 | Katty | SkramX: that's what oscar says. |
21:53.48 | Katty | SkramX: skram. |
21:54.03 | SkramX | Oh. Sorry. |
21:54.35 | SkramX | Whatever. |
21:54.55 | Katty | SkramX: are you having a bad day? |
21:55.22 | SkramX | Katty: No. |
21:55.41 | Katty | SkramX: k |
21:56.53 | SkramX | I am alright.. you do know I am not Mark Spencer or anything, though I have been using the alias MarkS and Skram for a while now. |
21:58.13 | abcbooze | hook me up skram ;) |
21:59.00 | *** join/#asterisk Corydon-w (i=cinnamon@pdpc/supporter/sustaining/Corydon76-home) |
21:59.43 | Katty | byebye |
21:59.56 | Katty | SkramX: heh |
22:02.38 | file[laptop] | paulc: !!! |
22:02.51 | *** join/#asterisk brc_ (n=Brian@pdpc/supporter/basic/brc) |
22:03.08 | paulc | I AM I AM.. kinda.. |
22:03.19 | angler | paulc, whats up! |
22:03.28 | paulc | Hey handsome :) |
22:03.41 | paulc | Not much.. Friday afternoon.. testing RMA devices.. packing and shipping others.. waiting for the weekend to begin! And you? |
22:03.59 | jake1932 | paulc - how come when you leave and go you have this message come up |
22:04.02 | angler | waiting for 6pm myself |
22:04.13 | jake1932 | leave and return |
22:04.31 | jake1932 | now if you were elvis we'd have a tired joke |
22:04.42 | pc2 | What's the command to automagically update asterisk from cvs head? Isn't it like make update? |
22:05.17 | paulc | angler: can't come soon enough eh... weekend plans? |
22:07.21 | file[laptop] | feisty |
22:08.54 | brc_ | FILE!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!11 |
22:08.56 | brc_ | WHAT |
22:08.58 | brc_ | IS |
22:08.59 | brc_ | UP! |
22:09.05 | brc_ | mishehu, hey man |
22:09.17 | mishehu | brc_: hey |
22:09.30 | mishehu | been what... a few months already |
22:09.33 | brc_ | ha |
22:09.35 | brc_ | been busy |
22:09.51 | brc_ | nat suckorz |
22:09.57 | mishehu | same here. 3 classes at college, a company to run, a parrot to train, and a world to conquer... |
22:10.40 | *** join/#asterisk FuriousGeorge (n=brian@ool-44c5a9b8.dyn.optonline.net) |
22:10.43 | FuriousGeorge | hi all |
22:11.24 | marc324 | is it possible to create a web interface that would update the extensions.conf file (without shutting down asterisk) |
22:11.40 | brc_ | hahahaha |
22:11.45 | brc_ | you kill me man |
22:11.50 | mishehu | marc324: if you build it, they will come. |
22:11.57 | mishehu | if you want them to build it, dial 1 |
22:12.30 | pc2 | make update says this. hmm? |
22:12.30 | pc2 | The following files have conflicts: |
22:12.30 | pc2 | apps/Makefile |
22:13.16 | FuriousGeorge | vote: usb vs. soundcard headset for softphone |
22:13.21 | FuriousGeorge | ? |
22:13.51 | Nyvar | usb |
22:14.03 | FuriousGeorge | Nyvar: its unanimous |
22:14.06 | Nyvar | lol |
22:14.58 | Nyvar | although there is that cool hack of only plugging in a headset halfway :) |
22:15.15 | marc324 | is answer yes or no? |
22:16.00 | file[laptop] | the answer will always be yes |
22:16.13 | file[laptop] | you could make Asterisk make toast for you if you wanted |
22:16.59 | pc2 | FuriousGeorge - ata adapter :) |
22:17.15 | mishehu | file[laptop]: that's when it crashes and burns |
22:17.19 | mishehu | then you get toast. |
22:19.29 | FuriousGeorge | file[laptop]: toasts sounds good. do you expect that to make it into 1.2? i forgot if you were working on sip messaging or breakfast... |
22:19.43 | FuriousGeorge | it was breakfast, right? |
22:20.02 | file[laptop] | app_breakfast |
22:20.02 | file[laptop] | duh |
22:20.07 | lancey | :)))) |
22:20.10 | FuriousGeorge | of course |
22:20.14 | lancey | app_antiheadache |
22:20.18 | lancey | is what i need now :) |
22:20.22 | Nyvar | app_chicken_n_waffles |
22:20.33 | file[laptop] | lancey: that's deprecated, it's app_advil now |
22:20.35 | FuriousGeorge | Nyvar: you beat me to it |
22:22.51 | *** join/#asterisk nagl (n=nagl@213.235.241.6) |
22:23.10 | lancey | :>>> |
22:23.33 | FuriousGeorge | m@rch81 |
22:23.38 | FuriousGeorge | there goes my aim pw |
22:23.57 | ender | *snicker* |
22:24.22 | pauldy | anyone know an elegant way to give permissions to a user for the audio device when using devfs |
22:24.24 | FuriousGeorge | damn sloppy focus |
22:24.38 | pauldy | trying to get console audio working with asterisk and this part has me a bit stumpet |
22:24.45 | pauldy | stumped |
22:25.02 | ender | pauldy: I know a good way when using udev.. |
22:25.17 | pauldy | how than ender? |
22:25.22 | shido6 | :( |
22:25.29 | pauldy | hows that ender? |
22:25.46 | ender | pauldy: there are permissions.d files for udev that you can define who gets ownership and what permission levels they have. |
22:26.01 | pauldy | hrm wonder if there is something similar for devfs |
22:26.08 | ender | pauldy: but devfs is deprecated and inherrently broken, so I don't know how to do it there. |
22:27.34 | *** join/#asterisk buddah (n=djbrianc@67.110.253.129) |
22:27.49 | buddah | can you upload .wav's to polycom phones to use as ring tones for distinctive ring? |
22:28.18 | *** join/#asterisk pifiu (n=myassisb@208.205.181.170) |
22:28.30 | pifiu | hey what are some good sites for royalty free music? |
22:28.42 | Nyvar | bittorrent |
22:28.48 | buddah | yup |
22:28.59 | pifiu | which bit-torrent |
22:29.07 | Nyvar | j/k |
22:31.36 | pauldy | ender, if I upgrade to udev where would I find the permission files? |
22:32.39 | ender | pauldy: /etc/udev/rules.d/ |
22:32.56 | ender | I think the main one is 50-udev.rules |
22:33.48 | pauldy | kewl thanks |
22:34.10 | ender | np |
22:34.17 | pifiu | so no one knows? |
22:35.45 | FuriousGeorge | pifiu: aussievoip-org |
22:35.53 | mcf3782 | pifiu: I just did a google search for "royalty free music" and found quite a few sites. Have you tried that? |
22:36.04 | FuriousGeorge | whats google? |
22:36.19 | FuriousGeorge | i come here whenever i need to know anything |
22:36.23 | Nyvar | the key is finding 'free' royalty free music |
22:36.34 | marc324 | can the extensions.conf be from a database? |
22:36.42 | marc324 | built from db. |
22:37.20 | marc324 | for large extensions, it becomes impossible to edit the file |
22:37.38 | ender | marc324: it can be done from include files as well |
22:37.57 | ender | use many small files to create your big dial plan |
22:37.59 | lancey | and from db as well |
22:38.05 | ender | yep |
22:38.16 | lancey | res_config or something like that, it was, i recall |
22:44.10 | pifiu | yeah i have tried google |
22:44.14 | pifiu | but i was wondering if there were any free ones? |
22:45.23 | Nyvar | marc, why not use 'realtime' asterisk and just everything from a DB? |
22:45.24 | pauldy | pifiu, googleis your best bet they are out there but they come and go so there is no defacto site |
22:47.00 | paulc | JerJer: You around? I've got a quick question on nufone/remote party ID for presentation CLI |
22:47.32 | mcf3782 | I just downloaded several mp3 music files from http://www.incompetech.com/m/c/royalty-free/ |
22:47.58 | mcf3782 | Their FAQ seems to indicate that all their stuff is royalty free and licensed under CCL |
22:48.20 | Nyvar | ohhh.. space opera... my fav |
22:48.32 | syle2 | PRI's are free inbound and outbound dialing right |
22:48.48 | Nyvar | all free, all the time |
22:48.54 | Qwell | syle2: not always |
22:49.02 | Nyvar | assuming that someone else is paying the LD bill |
22:49.09 | Qwell | my work pays like 2c/min for both local and long distance |
22:49.11 | syle2 | assuming no long distance |
22:49.19 | Qwell | still no |
22:49.45 | Nyvar | depends on the carrier, in the US its easy to find flat-rate-per-month PRI's, around the world some providers still charge per-minute fees |
22:49.59 | syle2 | on inbound or outbound? |
22:50.06 | Nyvar | how's that workin out for you aussies? |
22:50.28 | syle2 | i;m not from US so just wondering |
22:51.25 | opus__ | does anyone here use T38-bits? |
22:52.00 | Nyvar | is that a device? |
22:52.21 | mcf3782 | Well, I have to say that thus far; I am very unimpressed by any of the softphones I've tried. They either don't work at all, or I can only get audio to go in one direction. |
22:52.35 | Nyvar | i use t.38, but not familiar with 't38-bits' |
22:53.10 | Nyvar | mcf, sounds like nat to me |
22:53.29 | marc324 | how do you make an extension automatically execute once the context is called? |
22:53.55 | lancey`busy | chan_sip rulz |
22:53.56 | lancey`busy | <PROTECTED> |
22:53.58 | lancey`busy | :))) |
22:54.01 | syle2 | mcf3782: lots of sites like that, i just want a site where i can download gigs of mp3 files in some compressed format |
22:54.19 | mcf3782 | If I'm talking across the same class-c lan from one machine to another; I wouldn't think nat would be involved. But maybe I don't yet know enough to know what it is that I don't yet know. |
22:54.28 | syle2 | i got about 15 thousand songs , always willing to trade |
22:59.19 | mcf3782 | syle2: don't know what to tell you there. I was replying to pifiu's question with what I found. I'm not yet at the point of needing music for MoH. I'm still trying to get some more basic stuff working like 2-way audio from a client (windows or Linux) and a softphone to my Asterisk server. |
23:00.26 | mcf3782 | Best I've been able to do thus far is get the 4.1.0 version of kphone to register, call, connect, and receive audio from the server; but it can't/won't send anything back to it. |
23:01.18 | ender | mcf3782: sounds like you don't have your dtmf settings right. |
23:01.36 | ender | wait, n/m |
23:01.40 | mcf3782 | dtmf? |
23:01.46 | mcf3782 | :) |
23:01.48 | ender | dtmf is for sending digits down the line. |
23:02.05 | mcf3782 | yea.. I was wondering how that was related to sending audio. :) |
23:02.07 | ender | like passcode for voicemail or something. My co-worker had dtmf issues w/ kphone. You're talking about audio though. |
23:02.26 | ender | He had to use a specifc audio codec too.... |
23:02.32 | ender | sure your mic is working and all that rot? |
23:04.12 | mcf3782 | I can hear the audio from my mic (tried the built-in one and one plugged into the mic jack) in the speakers on the client. I can start up a sound recorder on the client and record my voice from both the built-in and the external mic; I just don't seem to be having any luck getting kphone to send it to the server. |
23:04.48 | mcf3782 | Watching the lan with tcpdump, there's no indication that there's any audio traffic leaving the client, headed toward the server... or anywhere else for that matter. |
23:05.10 | ender | ah |
23:06.41 | mcf3782 | I've tried several versions of kphone, starting with the most recent version posted on their web site, and walking down through older versions. the 4.1.0 version is the first one that will even connect to the point that it receives audio being sent from the asterisk server to the client. |
23:08.00 | mcf3782 | Maybe I just haven't had enough caffeine yet for it to make sense.. |
23:08.17 | Nyvar | maybe kphone has a mic volume contorl |
23:08.28 | opus__ | mcf3782 check externip, make sure you define localnet. check nat=yes |
23:08.52 | *** join/#asterisk nvrs (i=RUR@Toronto-HSE-ppp3866734.sympatico.ca) |
23:09.05 | mcf3782 | opus: are those kphone settings or asterisk settings? |
23:09.22 | opus__ | asterisk settings |
23:10.17 | Nyvar | http://www.linuxjournal.com/node/8165/print -- search for Turn on the microphone |
23:10.21 | *** join/#asterisk ZeMMad (n=blah@72.252.15.246) |
23:10.23 | mcf3782 | Ahh.. ok. I will look at both of those. |
23:10.38 | *** join/#asterisk zotz (n=zotz@24.231.36.100) |
23:11.19 | mcf3782 | Nyvar - good thought. The mic is definitely turned on though according to alsamixer. |
23:12.10 | mcf3782 | I'll check the externip and localnet settings that opus suggested, but first...more caffeine. ;) |
23:16.11 | opus__ | finally I have some dvds to burn all my shit off to! |
23:27.02 | *** join/#asterisk lters (n=lters@mrtcdsl-034.mis.net) |
23:29.18 | *** join/#asterisk asterisk99 (n=dunoowhy@modemcable169.194-130-66.mc.videotron.ca) |
23:29.44 | asterisk99 | anyone here using PRIs? I have a really dumb question!!!!! ;) |
23:30.11 | denon | just ask it |
23:33.01 | asterisk99 | denon: I configed a context called pri in my zapata.conf... In extensions.conf I created a siple Dial(zapata/1/4165551234) to try an outbound call... I get 'no channel type registered for pri' ideas? |
23:33.17 | asterisk99 | denon: I configed a context called pri in my zapata.conf... In extensions.conf I created a simple Dial(zapata/1/4165551234) to try an outbound call... I get 'no channel type registered for pri' ideas? |
23:34.00 | lters | Zap/1/12345 |
23:34.01 | asterisk99 | denon: I don;t see a type= entry in zapata.conf |
23:34.09 | ManxPower | asterisk99, contexts are normally only use for INCOMING calls. |
23:34.19 | ManxPower | And it would be Dial(Zap/1/12345) |
23:34.31 | asterisk99 | ManxPower: Zap??? OK I'll try |
23:34.55 | ManxPower | context=pri in zapata.conf would cause incoming calls from the PSTN into your PRI to land in the [pri] section of extensions.conf |
23:35.52 | asterisk99 | ManxPower: OK tried Dial(Zap/1/4165551234) I get no channel type registered for 'Zap' |
23:36.35 | asterisk99 | ManxPower: It's a PRI |
23:36.45 | ManxPower | asterisk99, then 1) you don't have the zap drivers installed, 2) you have noload => chan_zap.so in /etc/asterisk/zapata.conf, or 3) you didn't rebuild Asterisk after installing zapata |
23:37.04 | ManxPower | All digium cards are Zap, regardless of the interface |
23:38.58 | *** join/#asterisk stkn (i=nobody@gentoo/developer/pdpc.active.stkn) |
23:39.29 | asterisk99 | ManxPower: Oh oh... ok Zaptel is loaded (lsmod sees it)... no Noload in zapata.conf... I dodn;t rebuild Asterisk (that's definately required??) |
23:39.40 | asterisk99 | ManxPower: Oh oh... ok Zaptel is loaded (lsmod sees it)... no Noload in zapata.conf... I didn't rebuild Asterisk (that's definately required??) |
23:39.59 | ManxPower | asterisk99, If asterisk doesn't see zaptel installed when you build Asterisk, it won't build zap support |
23:40.13 | ManxPower | asterisk99, and if you keep repeating every line you type nobody will help you |
23:40.50 | asterisk99 | ManxPower: Hmmmmmm. (Sorry. I repeated the lines to correct typos. I'll let the typos alnoe from now on.) |
23:41.10 | mcf3782 | the 'gotta rebuild asterisk after installing zap' part got me the first time around. |
23:41.17 | ManxPower | I hate IPSec |
23:42.05 | asterisk99 | ManxPower: Do I need set idledial, idleext, minunused for PRIs or is that optional? |
23:42.21 | ManxPower | asterisk99, I've never used them |
23:44.08 | opus__ | damn it |
23:44.16 | opus__ | Anyone here use a T.38 provider? |
23:44.55 | *** join/#asterisk generalhan (i=general_@ip67-90-64-2.z64-90-67.customer.algx.net) |
23:45.05 | generalhan | whats up everyone ?> |
23:45.36 | generalhan | who in here is pretty good with the FOP ? rather is anyone in here using the FOP ? |
23:47.50 | *** join/#asterisk _zemmad (n=blah@72.252.15.246) |
23:48.09 | _zemmad | hi is there an asterisk software for windows?? |
23:48.57 | *** join/#asterisk hypnox (n=dan@cornelyn.force9.co.uk) |
23:49.13 | mmlj4 | _zemmad: you might be able to compile it under cygwin |
23:49.33 | *** join/#asterisk PakiPenguin (n=uppal@linuxpakistan/admin/pakipenguin) |
23:49.36 | mmlj4 | but no, there is not and will not ever be an official asterisk release for windows |
23:50.16 | _zemmad | o |
23:50.24 | _zemmad | good stuff |
23:53.42 | mmlj4 | well, hrm, i was sorta wrong: http://www.voip-info.org/tiki-index.php?page=AsteriskWin32 |
23:53.51 | ManxPower | mmlj4, it's not supported |
23:54.30 | ManxPower | mmlj4, and it's really a linux emulator running under windows, which Asterisk runs inside of, and it only supports VoIP, and people will point and laugh at you if you try to use it. |
23:54.57 | mmlj4 | notice i said "sorta"? |
23:55.07 | ManxPower | opus__, Since Asterisk does not support T.38, not many people use it in this channe |
23:55.41 | ManxPower | mmlj4, my "isp" replaces one of their boxes and my internet connection seems more stable. |
23:56.08 | mmlj4 | hmm... they couldn't take anymore complaining from you? heh |
23:56.25 | ManxPower | Of course, jitter is still pretty horrible |
23:56.28 | hypnox | how popular is call routing by srv record? |
23:57.12 | FuriousGeorge | what do i need besides this http://pastebin.ca/24897 to detect fax? |
23:57.21 | ManxPower | I hate IPSec |
23:57.23 | FuriousGeorge | i cant find anything on the wiki besides that, but that doesnt seem to do it |
23:57.42 | ManxPower | I'm starting to think that IPSec doesn't actually encrypt anything, it's just so complicated nobody can figure out how to read the data. |
23:57.54 | hardwire | boob |
23:58.01 | *** join/#asterisk toddf (n=toddf@adsl-67-65-251-106.dsl.okcyok.swbell.net) |
23:58.16 | mmlj4 | ok, so why do you hate it? |
23:58.29 | ManxPower | mmlj4, because it's so complicated |
23:59.34 | mmlj4 | heh, well, have you looked at openswan? it's tons easier than freeswan was, i'm told |
23:59.44 | ManxPower | Huh? I use Cisco |
23:59.54 | mmlj4 | mmm, ok |