| 00:03.37 | *** join/#asterisk Sponge_bob (n=None@cpe-66-27-162-13.socal.res.rr.com) |
| 00:10.19 | *** join/#asterisk carl0s- (n=carl0s@compsup.demon.co.uk) |
| 00:14.58 | *** join/#asterisk trelane (i=trelane@unaffiliated/trelane) |
| 00:23.45 | *** join/#asterisk SplasPood (n=jwb@brooklyn.paravolve.net) |
| 00:33.40 | *** part/#asterisk TommyTheKid (n=tommythe@mpk-edge.cto.sunit.net) |
| 00:37.25 | *** join/#asterisk Skarmeth (n=Skarmeth@201008231125.user.veloxzone.com.br) |
| 00:51.14 | *** join/#asterisk Samoied (n=Samoied@201-25-23-59.paemt705.dsl.brasiltelecom.net.br) |
| 00:52.45 | docelmo | OI! |
| 00:53.37 | [TK]D-Fender | VEY! |
| 00:53.43 | russellb | hi? |
| 00:53.58 | file | moo |
| 00:56.28 | russellb | file: status report |
| 00:56.40 | *** join/#asterisk Mercestes (n=Merceste@216.54.143.2) |
| 00:57.33 | file | russellb: going to... blow this popsicle stand |
| 00:58.39 | *** join/#asterisk QbY (n=Kelvin@cm-64-221-172-88.dhcp.southerncoastalcable.net) |
| 00:58.53 | QbY | What is the command that will accept the dtmf entry and store it into a variable? |
| 00:59.21 | russellb | Read |
| 01:02.03 | *** join/#asterisk ToyMan (n=stuq@user-12lcqia.cable.mindspring.com) |
| 01:02.38 | Bullseye_Network | <PROTECTED> |
| 01:03.05 | Bullseye_Network | Right now I have 14 available agents and 2 calls on hold. |
| 01:03.08 | Bullseye_Network | any ideas? |
| 01:04.22 | Bullseye_Network | Im Using Sip to a voip proveder so its not an IRQ with a T1 card problem. |
| 01:09.03 | *** join/#asterisk wunderkin (n=wunderki@216-19-202-7.getnet.net) |
| 01:10.10 | *** join/#asterisk RoyK (n=roy@122.80-203-78.nextgentel.com) |
| 01:14.28 | *** join/#asterisk ariel_ (n=Ariel@dsl-20-177.cofs.net) |
| 01:15.15 | *** join/#asterisk THX2000 (i=AgentFLY@adsl-66-51-192-221.dslextreme.com) |
| 01:15.52 | THX2000 | Anyone know if its possible to get the hold button on a cisco 79xx to play MOH? |
| 01:16.19 | Bullseye_Network | If you have a default setup asterisk should detect the hold and play music. |
| 01:16.22 | Bullseye_Network | It does here |
| 01:16.42 | THX2000 | alright, i must have a bad setting in there somewhere then |
| 01:16.44 | THX2000 | thanx |
| 01:16.54 | Bullseye_Network | np |
| 01:19.18 | *** join/#asterisk ToyMan (n=stuq@user-12lcqia.cable.mindspring.com) |
| 01:21.59 | Dovid | what is the variable for the channel ID ? |
| 01:23.29 | Bullseye_Network | ${CHANNEL} |
| 01:23.30 | Bullseye_Network | ? |
| 01:24.16 | Bullseye_Network | ${SIPCALLID} |
| 01:24.16 | Bullseye_Network | ? |
| 01:24.21 | Bullseye_Network | what channel |
| 01:24.44 | russellb | UNIQUEID ... |
| 01:25.25 | *** join/#asterisk Qwell_ (n=north@unaffiliated/qwell) |
| 01:25.25 | *** mode/#asterisk [+o Qwell_] by ChanServ |
| 01:26.00 | Qwell | That was..interesting |
| 01:28.03 | russellb | Qwell: ? |
| 01:28.19 | Qwell | xchat was already open, but I couldn't find it |
| 01:28.36 | russellb | ha |
| 01:31.51 | russellb | so Qwell. |
| 01:31.57 | Qwell | nope |
| 01:32.00 | Qwell | not yet |
| 01:32.07 | russellb | lol |
| 01:32.11 | Qwell | sooooon |
| 01:32.13 | russellb | i didn't even ask a question |
| 01:32.33 | russellb | but I gues you knew :-p |
| 01:32.37 | Qwell | indeed :p |
| 01:32.40 | *** part/#asterisk foo (n=foo@unaffiliated/foo) |
| 01:33.42 | *** join/#asterisk tengulre (n=tengulre@61.185.224.66) |
| 01:34.04 | tengulre | Hi,all |
| 01:34.10 | russellb | Qwell: did the goods arrive and your waiting to deliver a message? or still waiting on goods? |
| 01:34.42 | Qwell | russellb: both |
| 01:34.59 | file | codewords! |
| 01:35.09 | Qwell | l33tsp34k |
| 01:35.13 | russellb | i ... don't get it |
| 01:35.16 | russellb | i'm lost in my own code |
| 01:35.19 | *** join/#asterisk kiong (n=root@bb219-74-251-84.singnet.com.sg) |
| 01:35.35 | file | russellb: the goods are held up, but the message is clear... just needs to be delivered |
| 01:35.38 | clyrrad1 | whats that command that allows you to create a voicemail box? |
| 01:35.51 | clyrrad1 | from the command line |
| 01:36.10 | russellb | there was one, but that has been deprecated .... for many years |
| 01:36.20 | russellb | you just edit voicemail.conf ... |
| 01:36.24 | clyrrad1 | really....??? ..... ewww... didnt know bout it |
| 01:36.43 | clyrrad1 | yea but it does not create the directories untill a message is left |
| 01:37.10 | russellb | i suppose so, yes |
| 01:37.26 | *** part/#asterisk kiong (n=root@bb219-74-251-84.singnet.com.sg) |
| 01:39.55 | Qwell | Anybody have any idea what the equiv of /proc/cpuinfo is on Solaris? |
| 01:40.44 | *** join/#asterisk Asterisk_Newbie (n=a_ti_tu_@bl7-129-123.dsl.telepac.pt) |
| 01:41.35 | Asterisk_Newbie | Hi all from Portugal :) |
| 01:45.23 | *** join/#asterisk gopherspidey (n=spidey@12.179.8.2) |
| 01:46.44 | russellb | Hi f |
| 01:46.46 | russellb | errr |
| 01:55.51 | *** join/#asterisk mtaht4 (n=m@c-71-198-23-124.hsd1.ca.comcast.net) |
| 01:56.48 | pdthome | what soft phone do people suggest on OS X 1.4 Intel |
| 01:56.52 | pdthome | 10.4 even |
| 02:08.30 | RoyK | xlite? |
| 02:08.41 | RoyK | os x on intel runs ppc code after all |
| 02:11.40 | *** join/#asterisk kcortez (n=kevin@208.49.103.100) |
| 02:11.50 | *** join/#asterisk wunderkin (n=kev@ip68-226-113-228.ph.ph.cox.net) |
| 02:14.30 | pdthome | it crashes constantly on me |
| 02:15.06 | pdthome | i have run across two or three sip phones that run like ass on my intel mac but run fine on my ppc mac |
| 02:15.10 | pdthome | rosetta isn't 100% |
| 02:16.31 | rob0 | If there's an OS X equivalent of chan_alsa or chan_oss, run * console as your softphone. :) |
| 02:16.34 | Luke-Jr | PPC is better anyhow |
| 02:17.33 | *** join/#asterisk tenlet (n=tenlet@pool-138-89-84-128.mad.east.verizon.net) |
| 02:18.34 | russellb | rob0: there isn't |
| 02:18.36 | *** join/#asterisk blaylock (n=sfv100@68-69-102-120.chvlva.adelphia.net) |
| 02:18.39 | blaylock | hello |
| 02:18.54 | russellb | i started working on one last year, i should pick it back up ... |
| 02:19.12 | blaylock | can anyone tell me what --Unregistered SIP '<context name>' means? |
| 02:19.24 | blaylock | im getting this message over and over when using asterisk -r |
| 02:20.17 | file | it means the SIP device unregistered itself |
| 02:20.32 | blaylock | so why does it keep doing it? |
| 02:20.43 | *** join/#asterisk knarfly (n=root@c-69-180-98-189.hsd1.fl.comcast.net) |
| 02:20.44 | clyrrad1 | does anyone know if there is a way to force MOH to start from the beginning of a song each time instead of having a memory where it left off last time? |
| 02:20.45 | russellb | sounds like you have a sip client going crazy |
| 02:20.53 | blaylock | hmm |
| 02:20.54 | russellb | sending REGISTER messages with an expiry of 0 seconds |
| 02:20.54 | file | well, what SIP device is it? what type... |
| 02:21.16 | blaylock | either a grandstream or Aastra 480 series phone |
| 02:21.17 | knarfly | any fwdnet users out there who can help me with a couple of tests calls? |
| 02:21.27 | blaylock | not exactly sure, its one of our customers |
| 02:21.32 | russellb | clyrrad1: no, you can't do that ... it just doesn't work that way |
| 02:21.42 | clyrrad1 | really? Too bad.... |
| 02:21.46 | clyrrad1 | there is not hack for it? |
| 02:21.48 | russellb | well ... unless ... |
| 02:22.00 | russellb | i'm really resistant to encourage users to use "hacks" |
| 02:22.07 | clyrrad1 | yea..... |
| 02:22.11 | russellb | i can usually think of one, though :) |
| 02:22.11 | clyrrad1 | but is there some way to do it? |
| 02:22.18 | clyrrad1 | I am using MOH as ring tones |
| 02:22.25 | clyrrad1 | is there a better way? |
| 02:22.29 | russellb | well, if you define a seperate music class for every channel ... |
| 02:22.35 | knarfly | clyrradi: I thought they did...my system always starts a new song at the beginning |
| 02:22.47 | clyrrad1 | nah it has a memory |
| 02:22.54 | clyrrad1 | it picks up where it left off |
| 02:22.54 | russellb | yeah, that still isn't going to work |
| 02:23.03 | russellb | yeah, so nevermind. |
| 02:23.05 | knarfly | clyrrad1: when I don't use random it always starts with the same song |
| 02:23.11 | clyrrad1 | is there a way to Backgroun() play while ringing a phone is happening? |
| 02:23.31 | clyrrad1 | knarfly does it start from the start of the song when you dont use random? |
| 02:23.47 | knarfly | clyrrad1: yes |
| 02:23.52 | clyrrad1 | what are you using? |
| 02:23.53 | clyrrad1 | what mode? |
| 02:23.58 | Qwell | russellb: Was that a giggletackle(TM)? |
| 02:24.03 | knarfly | clyrrad1: and always the same song starts first |
| 02:24.07 | russellb | Qwell: why, yes, it was |
| 02:24.11 | knarfly | clyrrad1: mode = files |
| 02:24.36 | clyrrad1 | i have mode=mp3 |
| 02:24.37 | knarfly | any fwdnet users out there who can help me with a couple of tests calls? |
| 02:24.45 | clyrrad1 | maybe just set random=no? |
| 02:25.31 | clyrrad1 | I thought if you wanted mp3s to play you needed mode=mp3 |
| 02:25.35 | knarfly | clyrrad1: I'm using the native player...mpg123 has some bugs, especially with FreeBSD but it works...the native player works better IMHO |
| 02:26.00 | clyrrad1 | knarfly can you PM me a context that you have that does not contiue |
| 02:26.02 | clyrrad1 | continue* |
| 02:26.11 | clyrrad1 | I would like to compaire it to my config |
| 02:26.15 | rob0 | knarfly: call me @fwd 783889 if you want |
| 02:27.04 | knarfly | rob0: calling now |
| 02:27.34 | clyrrad1 | knarfly ... can you please paste me one of your MOH contexts? |
| 02:27.53 | *** join/#asterisk eBody (n=ehernand@207.71.51.162) |
| 02:28.48 | eBody | please, what ports ports need to be nat'd for asterisk to asterisk connections, IAX and SIP? |
| 02:29.23 | blaylock | eBody are you running behind a router? |
| 02:29.32 | eBody | yup |
| 02:29.37 | blaylock | why? |
| 02:29.48 | knarfly | clyrrad1: give me just a moment I'm testing with rob0 |
| 02:30.06 | blaylock | put it in front of the router and use ip_proxy in the vlan package |
| 02:30.13 | eBody | because i'm trying to test dual asterisk servers, which will eventually be across the net |
| 02:30.19 | blaylock | or make the asterisk box a DMZ |
| 02:30.32 | eBody | there we go, that sounds good. |
| 02:30.42 | blaylock | heh |
| 02:33.25 | *** part/#asterisk Samoied (n=Samoied@201-25-23-59.paemt705.dsl.brasiltelecom.net.br) |
| 02:36.51 | Dovid | if i switch a call between diffrent contexts does the channel change ? |
| 02:38.49 | *** join/#asterisk TheCops (i=nobody@got.securebinary.com) |
| 02:47.47 | knarfly | clyrrad1: http://pastebin.ca/94240 |
| 02:52.45 | JunK-Y | Dovid: no |
| 02:53.39 | blaylock | does it affect anything if an extension gets hung up on immediately when dialed? Like screw something up down the line? |
| 02:53.51 | knarfly | clyrrad1: did you see the paste? |
| 03:05.22 | *** join/#asterisk cheeze (n=coolchee@mt-sterling-bsr1-68-65-162-83.ironoh.adelphia.net) |
| 03:10.59 | *** join/#asterisk babyju (n=babyju@h-67-102-255-186.nycmny83.covad.net) |
| 03:22.40 | *** join/#asterisk Gamercjm (n=chris@pool-71-254-164-47.lsanca.fios.verizon.net) |
| 03:25.28 | *** join/#asterisk Wi_Fi (n=OUT@c-24-127-12-85.hsd1.ca.comcast.net) |
| 03:26.52 | *** join/#asterisk jeebusmobile (n=jeebusmo@29palms-cuda1-68-170-42-234.losaca.adelphia.net) |
| 03:27.03 | *** join/#asterisk jeebusmobile (n=jeebusmo@29palms-cuda1-68-170-42-234.losaca.adelphia.net) |
| 03:40.23 | *** join/#asterisk ben_d (n=ben@cpe-66-67-129-4.rochester.res.rr.com) |
| 03:50.32 | Asterisk_Newbie | bye all, from Portugal |
| 03:51.37 | *** part/#asterisk Asterisk_Newbie (n=a_ti_tu_@bl7-129-123.dsl.telepac.pt) |
| 03:52.09 | *** join/#asterisk bmg505 (n=leon@c1-208-8.rndf.isadsl.co.za) |
| 03:54.57 | *** join/#asterisk tempest1 (n=asf@adsl-144-36-60.chs.bellsouth.net) |
| 04:03.08 | *** join/#asterisk [pyro] (i=pyro@tor/regular/bracketed-pyro) |
| 04:06.06 | *** join/#asterisk andrew` (i=andrew@69-12-136-56.dsl.static.sonic.net) |
| 04:13.25 | *** join/#asterisk [pyro] (i=pyro@tor/regular/bracketed-pyro) |
| 04:13.30 | *** join/#asterisk kiong (n=kiong@bb58-185-172-59.singnet.com.sg) |
| 04:14.57 | *** join/#asterisk dprevite (n=dprevite@cpe-66-61-129-18.insight.res.rr.com) |
| 04:27.17 | kiong | hi all, i got 1 way voice, asterisk behind NAT but under DMZ, other one using public ip, he can hear but can't speak. any idea what to do ? |
| 04:28.58 | Juggie | set externip in sip.conf |
| 04:29.11 | kiong | in [general] section ? |
| 04:29.15 | Juggie | yep |
| 04:29.19 | Juggie | its in the example config |
| 04:31.02 | kiong | can i put like myserver.dyndns.org as externip ? |
| 04:31.11 | Qwell | kiong: externhost |
| 04:31.19 | *** join/#asterisk AJaymn (n=Ya@70.59.126.206) |
| 04:32.12 | kiong | so i put externhost=myserver.dyndns.org, and i don't need to put externip |
| 04:32.17 | Qwell | correct |
| 04:32.21 | Qwell | but you'll want externrefresh |
| 04:33.10 | *** join/#asterisk Netgeeks (n=chris@68-185-24-8.static.mdfd.or.charter.com) |
| 04:33.12 | kiong | mmmm what is the normal value for externrefresh ? |
| 04:33.21 | Qwell | dunno |
| 04:33.42 | kiong | is it in minute ? |
| 04:33.47 | Qwell | seconds? |
| 04:34.36 | kiong | oh oke |
| 04:46.20 | kiong | Qwell, still cannot, which port do i need to do forwarding ? |
| 04:49.22 | Juggie | you need to forward the range of ports you define in rtp.conf |
| 04:49.26 | Juggie | to the * box. |
| 04:50.40 | clyrrad1 | yep and besure to choose UDP not TCP |
| 04:54.48 | *** join/#asterisk lilo (i=levin@freenode/staff/pdpc.levin) |
| 04:57.32 | *** join/#asterisk lunaphyte (n=lunaphyt@overlord.bloodline.org) |
| 05:00.25 | clyrrad1 | good night guys have a good one |
| 05:03.18 | *** part/#asterisk ipfw (i=family@host-64-72-46-149.classicnet.net) |
| 05:12.53 | *** join/#asterisk kiong (n=kiong@bb58-185-167-101.singnet.com.sg) |
| 05:12.58 | kiong | sorry DC |
| 05:13.06 | kiong | i forward both udp and tcp, but still |
| 05:25.11 | *** join/#asterisk EyeCue (n=eyecue@unaffiliated/eyecue) |
| 05:25.24 | EyeCue | oh man, did i have 'fun' last night with some sip clients or what. |
| 05:28.31 | andrew` | which providers can port a number to some sort of low usage plan, preferably pay as you go that i can access with asterisk? |
| 05:28.42 | andrew` | (if any) |
| 05:38.07 | *** join/#asterisk FaithX (n=FaithX@ns.linuxterminal.com) |
| 05:38.31 | *** join/#asterisk Faithful (n=Faithful@ns.linuxterminal.com) |
| 05:41.05 | *** join/#asterisk daysmen3 (n=primus@host81-158-207-131.range81-158.btcentralplus.com) |
| 05:42.59 | *** join/#asterisk xxoxx (n=xxoxx@tor/regular/xxoxx) |
| 05:44.02 | dlynes_laptop | Is Flash operator panel predominantly useful as a status monitor? |
| 05:44.12 | dlynes_laptop | Or can it be used as a switchboard as well? |
| 05:44.20 | docelmo | ask FOP questions in #freepbx |
| 05:44.27 | dlynes_laptop | What? |
| 05:44.32 | dlynes_laptop | I'm not using freepbx |
| 05:44.49 | dlynes_laptop | Nor am I using trixbox, a@h, amp or any of that other garbage |
| 05:44.51 | docelmo | they are the guys who designed it. Just like if you said you had a AMP question |
| 05:44.57 | dlynes_laptop | Really? |
| 05:45.05 | docelmo | from my knowledge |
| 05:45.07 | docelmo | yes |
| 05:45.12 | dlynes_laptop | I thought the guy that developed fop was somebody altogether different |
| 05:45.20 | docelmo | but good to hear your not using amp etc.. |
| 05:45.31 | dlynes_laptop | docelmo, ummm...I thought you knew that :) |
| 05:45.44 | Mercestes | dlynes_laptop: FOP works as a switchboard because you can transfer and initiate calls through it, but the flash version only supports so many phones. |
| 05:45.44 | dlynes_laptop | I wouldnt' dream of trying to run a business off of that crap |
| 05:45.46 | docelmo | I was told they are the ones who wrote it. I dont use it so never paid attention |
| 05:46.07 | *** join/#asterisk scardinal (n=supreme@port816.ds1-suoe.adsl.cybercity.dk) |
| 05:46.14 | hads | I think FOP is a seperate thing |
| 05:46.17 | Mercestes | dlynes_laptop: Of the 6 times our server crashed, however, 5 of those were due to FoP |
| 05:46.29 | docelmo | I am building a 100% custom foundation right now on asterisk and freeswitch |
| 05:46.29 | dlynes_laptop | Mercestes, damn |
| 05:46.41 | Mercestes | dlynes_laptop: Though, to their credit, all were faulty config options well documented and warned about in the documentation. |
| 05:46.50 | dlynes_laptop | Mercestes, i'm just looking for something in the meantime until I've got enough time to write a Java applet to talk to the asterisk manager |
| 05:47.32 | Mercestes | dlynes_laptop: Yea, FoP should work with up to like 30 phones I believe...there is a non flash version that supports more phones but has none ofthe swichboard support youw ant. |
| 05:47.57 | dlynes_laptop | the fop is far from ideal for me....damned fop only works for very simple consoles under linux...anything more complicated and the flash plugin under linux doesn't display it properly |
| 05:48.06 | Mercestes | dlynes_laptop: If you follow the instructions though and configure it correctly, it does neat things like transfer calls, initiates calls between peers, call barges/monitors, etc. |
| 05:48.36 | dlynes_laptop | Mercestes, so I can tell it to start a call to a certain number, and to a call a sip client, and then tie the two togehter? |
| 05:48.53 | dlynes_laptop | Mercestes, or can i only transfer a ringing incoming call to a sip extension? |
| 05:48.53 | Mercestes | dlynes_laptop: THe sixth crash was some dumbass trying to mount an NFS on a dev box to the production server.... |
| 05:49.08 | dlynes_laptop | cool |
| 05:49.15 | Mercestes | dlynes_laptop: You can drag one remote phone to another and force that remote phone to call that other remote phone. |
| 05:49.17 | dlynes_laptop | Only time I ever use nfs is for installing linux |
| 05:49.35 | Mercestes | dlynes_laptop: Or drag a call into a conference....or drag a remote phone into an initiated call, or monitor a call via your phone... |
| 05:49.40 | dlynes_laptop | ok, so for an external call, i have to have a predefined number that it calls, then |
| 05:49.47 | dlynes_laptop | it won't prompt me for a phone number to call |
| 05:49.55 | *** join/#asterisk YoYo (n=troy@asterisk.office.psknet.com) |
| 05:50.07 | Mercestes | dlynes_laptop: nah, but you can drag a ringing phone to yours and answer it *i believe*... |
| 05:50.18 | dlynes_laptop | ok, cool |
| 05:50.18 | dlynes_laptop | thanks |
| 05:50.18 | YoYo | does zaptel still require newt? |
| 05:50.23 | dlynes_laptop | YoYo, never did |
| 05:50.43 | *** join/#asterisk DHuang (n=DHuang@mail.medec.com.au) |
| 05:50.44 | YoYo | weird |
| 05:50.45 | dlynes_laptop | YoYo, only one utility in the zaptel directory requires newt |
| 05:50.46 | EyeCue | it does in freebsd |
| 05:50.48 | EyeCue | :D |
| 05:50.49 | dlynes_laptop | YoYo, ztmonitor I think |
| 05:50.52 | Mercestes | dlynes_laptop: np. |
| 05:51.09 | DHuang | Afternoon.. :-) |
| 05:51.09 | YoYo | hrrm... I never used that |
| 05:51.14 | dlynes_laptop | EyeCue, freebsd must build ztmonitor by default then |
| 05:51.35 | YoYo | wait... freebsd? is zaptel finally working on freebsd? |
| 05:51.39 | EyeCue | well id really like the maintainer to include a few more knobs for removing dependencies |
| 05:51.40 | dlynes_laptop | EyeCue, in the default build from source, ztmonitor and all the other utilities aren't built |
| 05:51.48 | dlynes_laptop | EyeCue, he does |
| 05:51.50 | EyeCue | asterisk is working on my freebsd 6.1-stable box |
| 05:51.53 | EyeCue | with the latest port |
| 05:52.01 | YoYo | tor2 drivers? |
| 05:52.03 | dlynes_laptop | EyeCue, you need to define or undefine certain definitions, and they're not compiled |
| 05:52.07 | EyeCue | dlynes_laptop, 'a few more' |
| 05:52.11 | dlynes_laptop | EyeCue, vi your makefile |
| 05:52.13 | EyeCue | :) |
| 05:52.19 | EyeCue | already done, i want a few more. |
| 05:52.20 | dlynes_laptop | EyeCue, ok, so you know about the existing ones, then |
| 05:52.23 | EyeCue | *nods* |
| 05:52.28 | EyeCue | i bitched about it last night |
| 05:52.33 | DHuang | anyone know if it possible to setup 1 voicemail box for multiple SIP account? and when use VoicemailMain(${CALLERID}) to access that shared box? |
| 05:52.35 | EyeCue | apparently bison isnt needed anymore either |
| 05:52.57 | dlynes_laptop | DHuang, yes...voicemail boxes are just numbers...they're not tied to a certain phone |
| 05:53.20 | YoYo | eyecue: nods to me about tor2? so I can use freebsd to terminate my PRI now? |
| 05:53.24 | dlynes_laptop | EyeCue, was bison ever needed? |
| 05:53.26 | EyeCue | zaptel requires libpri and newt |
| 05:53.35 | EyeCue | s'all i know |
| 05:53.35 | Mercestes | DHuang: You can't use ${CALLERID} unless all the phones have the same callerID though. |
| 05:53.37 | DHuang | dlynes: :-) Yeah, so how to set the VoicemailMain to access the mbox number speicifed in the SIP? |
| 05:53.48 | dlynes_laptop | YoYo, yeah...I wouldn't suggest it though, unless Sobol(?)'s improved the driver since the last time I used it |
| 05:53.50 | DHuang | Mercestes: that's right..... |
| 05:54.00 | Mercestes | Dhuang: Unless you wanna set(CallerID(number)) on a specific dial in. |
| 05:54.08 | EyeCue | WITHOUT_MOH or WITHOUT_MPG123 would be nice too |
| 05:54.19 | YoYo | maybe I give up for the night and try again with freebsd... maybe then I won't have issues with whatever this CRC_CCIT_SHIT_ON_ME driver is in the linux kernel |
| 05:54.23 | dlynes_laptop | YoYo, I was running freebsd 6.0 with zaptel 1.2.5, and I was only able to get about a two week uptime |
| 05:54.24 | DHuang | Mercestes: No.. don't want to do that... any easier way apart from writing my own query? |
| 05:54.38 | hads | EyeCue: zaptel doesn't require libpri |
| 05:54.40 | YoYo | dlynes: hrrm... oh well |
| 05:54.49 | Mercestes | Dhuang: If you just call Voicemailmain it will ask for a mailbox that the user can enter. |
| 05:54.51 | EyeCue | well the port needs some haxing then. |
| 05:54.54 | YoYo | needs it in production... |
| 05:55.08 | Mercestes | Dhuang: You can teach the sheeple to dial a specific shared mailbox... |
| 05:55.08 | dlynes_laptop | EyeCue, asterisk under freebsd has a dependency on libpri, not zaptel |
| 05:55.15 | DHuang | Mercestes: I see... so not auto going into user's vbox.. |
| 05:55.19 | EyeCue | correct. |
| 05:55.24 | EyeCue | i 'was' talking freebsd specific |
| 05:55.31 | Mercestes | Dhuang: Hmm....polycom has a vm extension you can set.... |
| 05:55.34 | EyeCue | funny though, it installed libogg too. |
| 05:55.35 | DHuang | Mercestes: that's kewl... I guess there is no build-in function... |
| 05:55.49 | EyeCue | YoYo, for your reference, the port installed zaptel-1.0 |
| 05:55.50 | dlynes_laptop | EyeCue, it installed libogg for the format_ogg.so support |
| 05:55.54 | Mercestes | Dhuang: So you can set multiple phoens to go to the same vm extension and instead of $calllerID just specify the mailbox under that extension. |
| 05:55.58 | *** join/#asterisk CrashHD (n=crashhd@c-67-182-167-222.hsd1.ca.comcast.net) |
| 05:55.58 | CrashHD | hello |
| 05:56.12 | CrashHD | has anyone experienced subtle poping with their pri cards? |
| 05:56.23 | Mercestes | Dhuang: VM_ext = 380 exten => 380,1,Voicemailmain(380) |
| 05:56.31 | CrashHD | with no line codes, slips or bit errors? |
| 05:56.31 | DHuang | Mercestes: :-) just want to use a generic vm extensions...... |
| 05:56.56 | dlynes_laptop | EyeCue, erm format_ogg_vorbis.so i mean |
| 05:57.00 | Mercestes | DHuang: So you want to associate several mailbox numbers to a singular mailbox account then? |
| 05:57.13 | DHuang | Mercestes: that's right... |
| 05:57.25 | EyeCue | i 'kinda' got sip working last night |
| 05:57.26 | EyeCue | :| |
| 05:57.33 | Mercestes | DHuang: ln -s :) Just create the vm accounts then delete the extraneous ones and create links to the "shared box." |
| 05:57.37 | EyeCue | but i could never be sure it wasnt a client issue |
| 05:57.48 | Mercestes | Dhuang: Asterisk will see it as an extisting directory and treat it as such. |
| 05:57.49 | dlynes_laptop | EyeCue, sip works fine on freebsd |
| 05:57.52 | dlynes_laptop | EyeCue, no issues there |
| 05:58.02 | EyeCue | i could get calling/receiving working |
| 05:58.06 | EyeCue | but my friend couldnt hear audio |
| 05:58.08 | DHuang | Mercestes: yeah, let me explain. 10 ppl, 5 sales and they access 1 vmbox, the other 5 has their own. |
| 05:58.18 | dlynes_laptop | EyeCue, that's an issue of whether rtp is passing or not |
| 05:58.23 | Mercestes | Dhuang: For vmbox 1, 2, 3, 1 being the shared, just create the mailbox accounts, and ln -s 2 /var/spool/asterisk/voicemail/default/1 |
| 05:58.32 | EyeCue | i forwarded sip ports to my asterisk box |
| 05:58.35 | dlynes_laptop | EyeCue, you might want to look on the wiki for troubleshooting sip and sip/nat stuff |
| 05:58.40 | dlynes_laptop | EyeCue, rtp, not sip |
| 05:58.45 | EyeCue | :| |
| 05:58.47 | YoYo | like 2 years? 3? |
| 05:58.48 | DHuang | Mercestes: ${VoiceMailExt},1,VoicemailMain(s${Mailbox}@${CONTEXT}) |
| 05:58.49 | EyeCue | ftp = port which!? |
| 05:58.50 | EyeCue | :D |
| 05:58.50 | Mercestes | Dhuang: A symbolic link should do what you want then with separate mailbox numbers. |
| 05:58.51 | dlynes_laptop | EyeCue, are you defining nat=yes? |
| 05:58.57 | EyeCue | hang 2 |
| 05:59.03 | EyeCue | in what conf |
| 05:59.07 | dlynes_laptop | EyeCue, sip.conf |
| 05:59.17 | Mercestes | Dhuang: Yea, use that code, and just create symbolic links in your file system... |
| 05:59.20 | DHuang | Mercestes: Looking for something simpler... without symb link |
| 05:59.26 | denon | YoYo: I'm a pretty big fan of freebsd myself, but I still dont understand why a PBX needs to be so multi-platform |
| 05:59.33 | YoYo | ok, what's this CRC_CCITT_SHIT_ON_ME module that zaptel needs? |
| 05:59.38 | denon | I mean, it's an appliance, stripped down to only run the essentials |
| 05:59.40 | Mercestes | Dhuang: Damn man..;) Umm.....that's about all I got then. |
| 05:59.43 | denon | and do it very robustly |
| 05:59.46 | EyeCue | nope. |
| 05:59.47 | dlynes_laptop | YoYo, crc_ccitt.ko |
| 05:59.53 | YoYo | denon: because an appliance shouldn't depend on a hacked OS like linux |
| 05:59.59 | DHuang | Mercestes: that's kewl, I can write my own SQL command to retrieve the mailbox and pass into asterisk |
| 06:00.02 | denon | true enough |
| 06:00.04 | Mercestes | Dhuang: I don't think there is an * option to link mailboxes....you can specify any mailbox you want |
| 06:00.04 | dlynes_laptop | YoYo, it's a linux module, and freebsd might not have an equivalent |
| 06:00.05 | YoYo | dlynes: yeah, but what is it? |
| 06:00.11 | dlynes_laptop | YoYo, freebsd might ahve support for that natively |
| 06:00.13 | Mercestes | Dhuang: Might be the best plan. |
| 06:00.16 | DHuang | Mercestes: Yeah... :-( |
| 06:00.23 | dlynes_laptop | YoYo, it's just an implementation of hte ccitt crc algorithm |
| 06:00.24 | DHuang | Mercestes: thanks for the help...... |
| 06:00.29 | denon | but unless you can actually shift dev efforts away from linux to freebsd .. |
| 06:00.33 | Mercestes | YoYo: Just type it into your .config and recompile. |
| 06:00.43 | YoYo | yeah... that's what I'm doing now |
| 06:00.44 | denon | seems like it'd be hard to standardize on fbsd |
| 06:00.50 | Mercestes | YoYo: That will fix it....:) |
| 06:00.56 | dlynes_laptop | YoYo, i thought you were using freebsd, not linux? |
| 06:01.17 | Mercestes | YoYo: emerge gentoo-sources make menuconfig then vi .config (because I don't think make menuconfig puts it there) and then just type in exactly what it's bitching about then do a make. |
| 06:01.19 | YoYo | standardize on freebsd? easier than standardizing on redhate, suse, ubunto, gentoo, slackware, debian, and joes-garage-linux |
| 06:01.34 | *** part/#asterisk DHuang (n=DHuang@mail.medec.com.au) |
| 06:01.37 | dlynes_laptop | denon, especially when there's a lot more people using linux than freebsd |
| 06:02.03 | YoYo | dlynes: heh... I'd like to use freebsd :) |
| 06:02.07 | denon | well, I dont think that's a good argument |
| 06:02.11 | Mercestes | YoYo: /join #gentoo-voip ;) |
| 06:02.19 | denon | FreeBSD is by far a superior OS .. |
| 06:02.27 | dlynes_laptop | denon, in what way? |
| 06:02.30 | Mercestes | Best of both worlds, man..:P |
| 06:02.32 | YoYo | Mercestes: did I mention gentoo, or are you reading my mind? |
| 06:02.33 | denon | I just don't know if asterisk will ever be 110% on fbsd, unless dev shifts that way |
| 06:02.47 | Mercestes | YoYo: I saw you in the gentoo channel suffering the same pain..;) |
| 06:02.50 | dlynes_laptop | denon, please don't say because it's more secure |
| 06:02.51 | YoYo | ah... yeah |
| 06:02.59 | denon | dlynes_laptop: nah, that's openbsd |
| 06:03.00 | dlynes_laptop | denon, because that's opinionated |
| 06:03.13 | denon | besides, who cares if the OS is secure, some wanker usually leaves an old version of apache running anyway |
| 06:03.31 | dlynes_laptop | denon, i just don't think freebsd is any more secure than linux |
| 06:03.40 | denon | I didnt say it was, you did |
| 06:03.51 | dlynes_laptop | denon, the admins running freebsd are probably more capable than the ones running linux...that's why freebsd is usually more secure :p |
| 06:04.01 | hads | Yay #distrowars :) |
| 06:04.06 | YoYo | my preference for FreeBSD has nothing to do with security |
| 06:04.12 | YoYo | it has to do with ease of management |
| 06:04.15 | dlynes_laptop | nah...didn't say it was, but usually first argument people have for freebsd is security |
| 06:04.15 | denon | I just like the way FreeBSD is developed |
| 06:04.22 | denon | like a commercial product, not like an opensource fiasco |
| 06:04.36 | dlynes_laptop | denon, that i have to agree with |
| 06:04.40 | YoYo | for security, I dun care if it's linux, BSD, or windows... none are terribly hard to lock down |
| 06:04.44 | dlynes_laptop | denon, glibc is a freaking nightmare |
| 06:04.59 | denon | linux feels a lot like public SVN users have commit privs |
| 06:05.03 | dlynes_laptop | denon, even winnt\system32 is better |
| 06:06.15 | dlynes_laptop | denon, otoh, linux's kernel seems to have more driver support...it might not be a better kernel, but it has more hardware support, which is kinda crucial to most people |
| 06:06.37 | dlynes_laptop | I'd rather make the tradeoff for functionality, personally |
| 06:06.40 | *** join/#asterisk NDT (n=nunya@cpe-24-195-66-214.nycap.res.rr.com) |
| 06:06.40 | denon | dunno, I think most good servers are built with support in mind .. |
| 06:06.53 | denon | Linux has more driver support, because most of it's users are too cheap to buy good stuff |
| 06:06.55 | dlynes_laptop | YoYo, Use Sangoma! |
| 06:06.56 | YoYo | either that, or maybe I should just stick with my 2 year old system... |
| 06:07.09 | denon | FreeBSD only has hardware support, when they take the time to do it really well |
| 06:07.10 | dlynes_laptop | YoYo, Sangoma has excellent FreeBSD support |
| 06:07.11 | YoYo | oh wait... 1 year old |
| 06:07.16 | benjk | anybody knows if there is a channel for the sipura 3000? my IRC clients channel listsing feature seems broken |
| 06:07.34 | dlynes_laptop | benjk, try the sipura user's group forum on voxilla.org |
| 06:07.36 | YoYo | slynes: can't afford another line card right now |
| 06:07.45 | dlynes_laptop | benjk, or maybe it was voxilla.com |
| 06:07.52 | dlynes_laptop | benjk, anyways..you get the idea |
| 06:08.05 | benjk | that's too slow, I want to talk to somebody in real-time |
| 06:08.06 | dlynes_laptop | benjk, you can also try the linksys user's group forum on the same website |
| 06:08.10 | YoYo | hell, if I had the balls, I'd reconfigure my TNT to handle the PSTN gateway functions and ship everything to asterisk via SIP |
| 06:08.18 | benjk | forums are a waste of time |
| 06:08.26 | dlynes_laptop | benjk, ah...I just find there's a lot of good answers there for sipura's that are already answered |
| 06:08.41 | benjk | especially if you are not a Windows user, they are so likely to get back with an answer |
| 06:08.45 | dlynes_laptop | benjk, i find it's more useful reading the forums there, than asking the question here |
| 06:08.58 | NDT | ...anyone feel like helping me with a formula for this calculation? http://pastebin.ca/94358 |
| 06:09.14 | dlynes_laptop | benjk, why don't you just ask your question in here? |
| 06:09.19 | benjk | I am almost 100% certain that the forums will not cover how to upload firmware without Windows |
| 06:09.29 | dlynes_laptop | benjk, umm....tftp |
| 06:09.55 | benjk | yeah, so how to you get the Sipura to start looking for the file? |
| 06:09.57 | dlynes_laptop | benjk, http://ip.address.of.phone/upgrade?tftp://ip.address.of.tftp/path/to/firmware.bin |
| 06:10.03 | YoYo | timeout = (balance - connect_fee) / per_minute_charge |
| 06:10.26 | dlynes_laptop | benjk, it's all covered in yoru sipura administrator's manual |
| 06:10.30 | benjk | the Sipura firmware upload is a pull mechanism, not push |
| 06:10.41 | dlynes_laptop | benjk, yeah...it pulls it from your tftp server |
| 06:10.42 | benjk | you have to get the adapter to start the process |
| 06:10.45 | YoYo | if timeout < 1, do not connect |
| 06:10.51 | dlynes_laptop | benjk, did you read what i just typed? |
| 06:10.57 | YoYo | NDT, did you ever take 7th grade math? |
| 06:11.06 | benjk | but the web interface of the adapter does not have anything to INITIATE the pull |
| 06:11.20 | dlynes_laptop | benjk, READ THE URL I JUST GAVE YOU |
| 06:11.58 | benjk | well, the sipura doesn't pull |
| 06:12.21 | NDT | YoYo: I been up 28 hours...Having a hard enough time remembering my name...lol |
| 06:12.24 | dlynes_laptop | benjk, let's split it out, because it seems you're not seeing it |
| 06:12.37 | benjk | the sipura does not start the download |
| 06:12.47 | dlynes_laptop | benjk, yes it does |
| 06:12.51 | YoYo | NDT: ah... well, anyways that works... I just tested it in excell. but WTF are you doing charging $7 connect fee? |
| 06:12.54 | benjk | it doesn't |
| 06:12.57 | dlynes_laptop | benjk, trust me...I've upgraded firmware on like 40 of them |
| 06:12.58 | YoYo | and, where you get customers? I want in on that game |
| 06:13.07 | NDT | LOL these are jails |
| 06:13.11 | YoYo | AH |
| 06:13.13 | benjk | yeah, well I have just been trying all day long, this one doesn't |
| 06:13.14 | dlynes_laptop | benjk, everything from sipura 2000's up to Linksys PAP2-NA's |
| 06:13.24 | dlynes_laptop | benjk, then it's a broken POS |
| 06:13.27 | dlynes_laptop | benjk, return it |
| 06:13.34 | NDT | Actually more like $5.99 |
| 06:13.37 | NDT | hehe |
| 06:13.38 | YoYo | so... ($10 - $7) / $0.10 = 30 minutes |
| 06:13.39 | benjk | Sipura adapters are by definition POS |
| 06:13.45 | dlynes_laptop | benjk, one other thing yhou might try to get it to work |
| 06:14.05 | dlynes_laptop | benjk, factory default it, give it a hard reboot after the factory default, and then try your upgrade again |
| 06:14.06 | benjk | I was even trying with that Windows utility from the sipura site |
| 06:14.18 | dlynes_laptop | benjk, i've only ever had one sipura unit that wouldn't upgrade |
| 06:14.44 | benjk | but I only have a virtual Windows, which is sharing the NIC of my host machine and thus internally NATed, which will break the process |
| 06:14.51 | dlynes_laptop | I've had a couple that decided they wouldn't take it, but after a factory default and a reboot, they worked fine |
| 06:15.01 | dlynes_laptop | benjk, you don't need windows |
| 06:15.12 | dlynes_laptop | benjk, i've never used windows to upgrade a sipura unit |
| 06:15.18 | YoYo | ok scrwe this linux stuff... and this asterisk stuff... I'm going home to my nice comfy bed |
| 06:15.28 | benjk | I so f***** hate those Sipuras |
| 06:15.30 | Gamercjm | Any one seen VoipMasta latley? |
| 06:15.33 | EyeCue | grr @ confs. |
| 06:15.41 | NDT | YoYo: Yeah then just changing to milliseconds...but got that done LOL...thanks had brain lock |
| 06:15.45 | dlynes_laptop | ~seen VoipMasta |
| 06:15.51 | jbot | voipmasta <n=John@201.160.17.205.cableonline.com.mx> was last seen on IRC in channel #asterisk, 2d 7h 38m 1s ago, saying: 'hehehe'. |
| 06:15.57 | benjk | its the biggest pile of crap every released on humankind, even worse than Windoze |
| 06:16.00 | Gamercjm | kk |
| 06:16.04 | benjk | unleashed on |
| 06:16.22 | dlynes_laptop | benjk, have you tried a factory reset on it before trying a firmware upgrade? |
| 06:17.22 | dlynes_laptop | benjk, and have you tried a tftp upgrade, or only that stupid windows upgrade utility? |
| 06:17.23 | benjk | I dont know how to do a factory reset on that thing |
| 06:17.34 | dlynes_laptop | benjk, it's a dtmf code |
| 06:17.42 | Gamercjm | Anyone decent at flash and have some free time? working on a project thats voip related ... its another project like the ones on astertoys.com |
| 06:17.43 | benjk | I have tried a whole bunch of things, the Windows utility was my last shot |
| 06:17.47 | dlynes_laptop | benjk, do you not have a user's or administrator's manual for it? |
| 06:18.08 | benjk | I have a PDF somewhere, yes |
| 06:18.16 | dlynes_laptop | benjk, the dtmf code for factory reset is covered in both of those manuals |
| 06:18.20 | YoYo | ok, stupid question time... do I need something special in order for "shutdown -r now" to work in 2.6.16 kernel? |
| 06:18.28 | dlynes_laptop | benjk, basically **** on a phone connected to it |
| 06:18.37 | dlynes_laptop | benjk, then you'll enter the menu |
| 06:18.39 | benjk | haha |
| 06:18.47 | dlynes_laptop | benjk, then factory reset code# |
| 06:18.51 | Gamercjm | YoYo: i thought it was -t.. but i dunno |
| 06:18.51 | dlynes_laptop | benjk, then 1 to confirm |
| 06:18.58 | benjk | I am not physically at the same location though |
| 06:19.00 | dlynes_laptop | benjk, then it reboots |
| 06:19.13 | dlynes_laptop | benjk, then you're screwed for doing a factory reset |
| 06:19.21 | benjk | I guess so |
| 06:19.23 | dlynes_laptop | benjk, and even then, that sequence doesn't work for PAP2-NA's |
| 06:19.30 | *** join/#asterisk Assid (i=assid@203.115.83.215) |
| 06:19.36 | Assid | heya |
| 06:19.38 | benjk | its a Sipura |
| 06:19.40 | dlynes_laptop | benjk, i have no idea how to get into those yet |
| 06:19.42 | YoYo | shutdown -r is stopping everything, then it says "restarting system" |
| 06:19.47 | benjk | vanilla |
| 06:20.01 | dlynes_laptop | benjk, ok...that works for sipura 2000 and sipura 2002 |
| 06:20.01 | benjk | without any provider locks or anything |
| 06:20.02 | YoYo | but then it just sits there... like a windows machine saying "it is now safe to turn off your computer" |
| 06:20.09 | benjk | no its a 3000 |
| 06:20.20 | dlynes_laptop | benjk, and sipura 3000 |
| 06:20.23 | dlynes_laptop | :) |
| 06:20.24 | Assid | mornin dlynes_laptop |
| 06:20.36 | dlynes_laptop | good morning, Assid |
| 06:21.17 | EyeCue | dlynes_laptop, any nat related stuff to change for iax2 ? |
| 06:21.41 | benjk | looks like I have to get somebody there to do the firmware upgrade from a local Windows machine |
| 06:21.44 | dlynes_laptop | EyeCue, for iax2, you should be able to just port forward udp 4569, and set nat=no |
| 06:21.51 | benjk | what a piece of dog poo those Sipuras are |
| 06:22.14 | EyeCue | hmm 4569 |
| 06:22.15 | dlynes_laptop | benjk, the easiest way to do it, is get a sipura 3000 at your office all tested and ready to go |
| 06:22.21 | dlynes_laptop | benjk, and then just do a drop in replacement |
| 06:22.36 | Assid | iax2 doesnt need any special nat configuration except for port forwarding.. thats pretty much it |
| 06:22.52 | EyeCue | i take it 4569 is the registered port for iax ? |
| 06:22.58 | dlynes_laptop | EyeCue, for iax2, not iax |
| 06:22.59 | CrashHD | has anyone experienced subtle poping with their pri cards? |
| 06:23.00 | CrashHD | with no line codes, slips or bit errors? |
| 06:23.00 | EyeCue | so i can scrap sip right? |
| 06:23.03 | dlynes_laptop | EyeCue, iax isn't used anymore |
| 06:23.05 | Assid | and that too, only if you have incoming nat connections. if you plan to have outgoing calls, you dont even need that |
| 06:23.08 | EyeCue | ok. |
| 06:23.21 | dlynes_laptop | EyeCue, if the only outbound protocol is iax2, yeah, you don't need to port forward sip or anything |
| 06:23.23 | YoYo | YAY!!! I have modules! |
| 06:23.29 | *** join/#asterisk jeffik (n=Jeff@kns226.NetSurf.Net) |
| 06:23.36 | EyeCue | i just wanna get people being able to go user@myip |
| 06:23.39 | EyeCue | with an iax2 client |
| 06:23.41 | dlynes_laptop | EyeCue, even if you are using sip, i wouldn't port forward it...just set nat=yes, canreinvite=no |
| 06:23.52 | EyeCue | router has wan ip, with nat for the internal machine |
| 06:24.16 | EyeCue | asterisk = bsd server, internal ip, workstation is xp with iax client internal ip as well |
| 06:24.24 | EyeCue | and apprent iax2 > sip. |
| 06:24.27 | Assid | dlynes_laptop: actually.. nat=yes can have an issue.. i would use nat=route |
| 06:24.31 | EyeCue | far as nat is concerned, among other things |
| 06:24.37 | dlynes_laptop | EyeCue, once you start port forwarding sip, you're going to start running into all kinds of shitty problems |
| 06:24.47 | EyeCue | well ill remove those forwards :) |
| 06:24.54 | EyeCue | but question |
| 06:24.58 | EyeCue | for inbound calls |
| 06:25.07 | EyeCue | they need to be 'forwarded' to the asterisk server no ? |
| 06:25.12 | Assid | inbound iax2 calls or sip calls? |
| 06:25.13 | EyeCue | for sip that is |
| 06:25.17 | dlynes_laptop | Assid, what's nat=route do? |
| 06:25.23 | dlynes_laptop | Assid, i've never had to use that option |
| 06:25.24 | EyeCue | understand with iax2 i need to forward 4569 |
| 06:25.36 | *** join/#asterisk kmilitzer (n=km@office-gw.westend.com) |
| 06:25.40 | Assid | dlynes_laptop: it fixed the 1 way audio which nat=yes couldnt |
| 06:25.43 | dlynes_laptop | EyeCue, yes, forward 4569 for iax2, to keep everything simple |
| 06:25.52 | EyeCue | <3 long time |
| 06:25.53 | dlynes_laptop | Assid, then that was a problem with your router interfering |
| 06:25.54 | EyeCue | and just udp ? |
| 06:26.02 | dlynes_laptop | Assid, I get two way audio with nat=yes |
| 06:26.17 | Assid | dlynes_laptop: some people do, some dont.. |
| 06:26.21 | dlynes_laptop | Assid, the only time it doesn't work, is when it's an old router that's acting flaky |
| 06:26.29 | dlynes_laptop | Assid, replace the router, and the problem goes away |
| 06:26.52 | dlynes_laptop | Assid, even then, i've only had that problem with Linksys WRT54G's |
| 06:34.10 | *** join/#asterisk EyeCue (n=eyecue@unaffiliated/eyecue) |
| 06:34.26 | EyeCue | gah, gay client. |
| 06:34.37 | EyeCue | recommendations for a not gay win32 iax2 client? |
| 06:34.39 | EyeCue | :| |
| 06:34.52 | Dovid | is there any way to play a sound in the channel once a call is estableshed ? |
| 06:37.48 | Assid | dlynes_laptop: i dont remember if they gave me an issue with the 54G's or the dlink 724 |
| 06:38.04 | Assid | sorry.. had a call |
| 06:38.19 | Assid | but either which way .. nat=route seems to give the best compatibility for me |
| 06:38.24 | dlynes_laptop | yeah...actually...come to think of it, I think i had one issue with a dlink 4 port wireless router, too |
| 06:38.32 | dlynes_laptop | replacing it fixed it |
| 06:39.02 | dlynes_laptop | funny how it's only wireless routers i've had issues with, though |
| 06:39.05 | docelmo | idefisk is about the only good iax win32 client I have found |
| 06:39.19 | Assid | yeah |
| 06:39.41 | *** join/#asterisk nailbags|work (n=neilbags@149.171.94.134) |
| 06:39.47 | Assid | sip hardware /software phones should support uPnP ! |
| 06:39.54 | Assid | best way to fix a nat issue |
| 06:40.22 | EyeCue | amen. |
| 06:40.49 | EyeCue | im havin a prick of a time find a softphone that doesnt break, or isnt gey. |
| 06:41.06 | Assid | EyeCue: sip softphones dont really break |
| 06:41.13 | Assid | use xten's stuff.. |
| 06:41.22 | EyeCue | diax just hung and 99% cpu. |
| 06:41.34 | EyeCue | xten? |
| 06:41.41 | Assid | www.xten.com |
| 06:41.49 | EyeCue | ta |
| 06:43.38 | Assid | one day im gonna try and play with SER |
| 06:48.38 | Assid | man i wish those stupid sipdiscount and stuff would let me set my callerid |
| 06:49.13 | Mercestes | http://pastebin.ca/94377 Can I get help with this?? |
| 06:49.31 | Mercestes | oops..wrong chan..>.> |
| 06:50.38 | *** join/#asterisk UlbabraB (n=UlbabraB@host241-43-static.72-81-b.business.telecomitalia.it) |
| 06:52.59 | *** join/#asterisk MatsK (n=mats@141.221.181.62.in-addr.dgcsystems.net) |
| 06:54.26 | h3x0r | UPnP would break having multiple SIP phones behind the same router |
| 06:54.47 | Assid | well.. thats where we could use signalling port. and actual port |
| 06:55.18 | h3x0r | you mean the RTP port? |
| 06:55.25 | Assid | yes |
| 06:56.10 | h3x0r | i guess its a good idea but it would be even more of a hack than the way its done now |
| 06:56.18 | *** join/#asterisk r0d3nt (n=RatMan@foster.stonedcoder.org) |
| 07:01.09 | *** join/#asterisk nicox (n=nicox@83-64-42-210.prater.xdsl-line.inode.at) |
| 07:07.38 | nicox | is there any problem with the digium mailing list servers? i posted a message, but its not getting into the mailing list... |
| 07:09.21 | Faithful | When my internet link drops momentarily the pbx needs to be rebooted because it no longer passes calls out to providers. Is there a way to avoid rebooting? |
| 07:09.46 | stoffell | nicox: be patient, they sometimes tend to be slow.. |
| 07:10.00 | stoffell | Faithful: use proper dns settings on your lan? |
| 07:10.57 | Snake-Eyes | is there a variable or app i can use in a agi script to step the call length? |
| 07:11.18 | nicox | slow? the mail is from yesterday |
| 07:13.21 | stoffell | nicox: ouch :) did you find it on http://www.mail-archive.com/asterisk-users@lists.digium.com/ ? |
| 07:13.38 | docelmo | Snake-Eyes what do you mean? |
| 07:14.02 | Snake-Eyes | step = set |
| 07:14.39 | Snake-Eyes | docelmo, i wish to set the call length of a current call/channel |
| 07:14.55 | docelmo | ohh you mean so it hangs them up in say 60 seconds or something? |
| 07:15.02 | Snake-Eyes | yes |
| 07:15.11 | docelmo | absolutetimeout |
| 07:15.18 | docelmo | I belive is the app |
| 07:15.23 | Snake-Eyes | yea, ive tried that one |
| 07:15.31 | Snake-Eyes | i was hoping there was another |
| 07:15.43 | docelmo | nope |
| 07:16.45 | Snake-Eyes | absolutetimeout starts soon as its used, eg 60 seconds from when the person picks up not from when the phone rings |
| 07:17.07 | docelmo | ohh then take into consideration for PDD |
| 07:17.26 | Snake-Eyes | PDD? |
| 07:17.31 | docelmo | post dial delay |
| 07:17.49 | Dovid | i have IP phones and an FXS zap channel |
| 07:18.16 | Snake-Eyes | docelmo, how does PDD work? |
| 07:18.29 | Dovid | all IP phones can call each other and hear each other. for some reason when I call from ZAP to an IP phone it rings and you can pick up the call however poth parties can not hear each other |
| 07:18.42 | Dovid | anyone know what the issue may be ? |
| 07:18.46 | docelmo | it doesnt.. its a term. PDD refers to the amount of time between the last number dialed and the first ring |
| 07:19.00 | Snake-Eyes | ah |
| 07:19.29 | Assid | actually.. i think you can set it in the dial app |
| 07:20.08 | docelmo | normally PDD is around 2sec but can take upto 23 sec before a call is answered.. 23 seconds is approx 4 rings on the PSTN |
| 07:20.23 | Snake-Eyes | ok |
| 07:20.31 | Dovid | anyone ??? |
| 07:20.32 | Snake-Eyes | Assid, ill check now |
| 07:20.32 | docelmo | Your new to telecom? |
| 07:20.54 | docelmo | Dovid yes check your codecs |
| 07:20.59 | docelmo | and make sure canreinvite=no |
| 07:21.08 | Dovid | under what the SIP phone ? |
| 07:21.16 | docelmo | sip.conf |
| 07:21.31 | *** join/#asterisk psk (n=psk@golia.caltanet.it) |
| 07:22.41 | Snake-Eyes | cool Dial does it: S(n): Hangup the call n seconds AFTER called party picks up. |
| 07:22.51 | Assid | yep |
| 07:23.17 | Snake-Eyes | thanks |
| 07:23.29 | Assid | np |
| 07:23.30 | EyeCue | snake :D |
| 07:24.00 | Snake-Eyes | hey EyeCue |
| 07:24.15 | EyeCue | im still overwhelmed, but meh |
| 07:24.16 | EyeCue | :) |
| 07:24.21 | Snake-Eyes | lol |
| 07:24.22 | EyeCue | dlyness is iax'ing me. |
| 07:24.24 | docelmo | ohh thats something new |
| 07:24.35 | EyeCue | ive found that most clients blow |
| 07:24.35 | EyeCue | brb |
| 07:24.49 | Assid | someone in here had made an iax client |
| 07:24.54 | Assid | cant remember who |
| 07:25.03 | Dovid | doclemo: i have canreinvite=no and it still wont put the audio thru |
| 07:25.04 | Assid | it was in beta last time i checked |
| 07:25.29 | *** join/#asterisk tlow (n=tlowe@bgp.terrorist.net) |
| 07:25.43 | Assid | seems to be TDM.. |
| 07:25.50 | Assid | is your asterisk box local to your network? |
| 07:25.52 | Snake-Eyes | stay away from virbiage iax ata |
| 07:25.58 | EyeCue | HEHEHE |
| 07:26.06 | EyeCue | *talks with dlynes |
| 07:26.07 | EyeCue | * |
| 07:26.08 | EyeCue | :D |
| 07:26.15 | Dovid | doclemo: ? |
| 07:26.17 | docelmo | uhh sip or iax debug? |
| 07:26.35 | docelmo | are you transcoding? or ULAW pass thru? |
| 07:26.46 | Dovid | docelemo: what do the codecs have to be when calling from ZAP to IP ? |
| 07:27.15 | docelmo | TDM channels are always ulaw |
| 07:27.24 | docelmo | or alaw depending on where you are in the world |
| 07:27.29 | docelmo | but they are 64k |
| 07:27.39 | docelmo | so you will need to use ulaw if you dont plan to transcode |
| 07:27.42 | Dovid | yes. ia m using ulaw |
| 07:27.59 | Dovid | but i cant hear the audio |
| 07:28.06 | docelmo | I would do a sip debug and see whats up |
| 07:28.09 | Assid | Dovid: you cant hear incomng or outgoing? |
| 07:28.13 | Dovid | npe |
| 07:28.16 | docelmo | I dont know without looking at the issue |
| 07:28.17 | Assid | you may wanna play with txgain and rxgain ? |
| 07:28.20 | Dovid | ok |
| 07:28.35 | docelmo | your gains should be fine on the default |
| 07:28.46 | docelmo | Im telling you this sounds like a codec problem |
| 07:28.53 | docelmo | do a sip debug and check.. |
| 07:28.54 | Dovid | at what level ? |
| 07:28.56 | Dovid | ok |
| 07:29.10 | docelmo | it could be an issue with your config of the zap to your channelbank |
| 07:30.48 | Dovid | here is the debug |
| 07:30.49 | Dovid | http://pastebin.ca/94404 |
| 07:32.27 | h3x0r | its probably because of NAT |
| 07:32.58 | hads|home | Snake-Eyes: Have you had trouble with the Virbiage ATA? |
| 07:38.10 | *** join/#asterisk kilobit (n=seth@210.193.57.155) |
| 07:38.58 | *** join/#asterisk kristalino (n=kristali@84-50-84-146-dsl.trt.estpak.ee) |
| 07:46.27 | Dovid | Assid: Didnt test incoming. Dont have a pots line yet |
| 07:55.36 | *** join/#asterisk [Airwolf] (n=airwolf@dsl5402BD52.pool.t-online.hu) |
| 07:56.04 | EyeCue | man this is confusing :) |
| 07:58.53 | *** join/#asterisk schurzi (n=schurzi@www.verdammte-seelen.de) |
| 08:02.32 | Snake-Eyes | hads|home, i got it to use with sip to find that sip firmware promised on the website(i think) is no where done (after alot of phone calls..) and saw alot of posts of ppl using the iax side of it having quite a few problems. Overall it doesn't come across as a well supported product (even the website is out of date, last time i checked) |
| 08:03.36 | *** join/#asterisk syle (n=blah@unaffiliated/syle) |
| 08:04.22 | Snake-Eyes | hads|home, i was using it as a paper weight for month or two until i put it away |
| 08:04.39 | *** join/#asterisk kiong (n=kiong@bb58-185-167-101.singnet.com.sg) |
| 08:05.46 | hads|home | Snake-Eyes: Interesting. I noticed that the SIP firmware wasn't done, but I'd rather use a Linksys for that anyway. |
| 08:05.48 | kiong | hi how do i make call number that is start with * ? |
| 08:06.43 | hads|home | Snake-Eyes: I sell VoIP gear so I was interested when it came out and emailed them about supplying it but I never got an email back from them |
| 08:08.49 | docelmo | put a * in front of your number being dialed.. |
| 08:09.22 | Snake-Eyes | hads|home, im not surprised you didnt get a repsonse |
| 08:10.19 | kiong | mmm i mean, i want to dial *99 from my x100p, in my dialplan i put _999,Dial(Zap/g0/*99) but then it just pass the 99 without * |
| 08:10.32 | hads|home | I decided not to bother with them after that. It makes you think bad things when you don't get a response from a company when you want to give them money. |
| 08:12.09 | Snake-Eyes | hads|home, they use there parent company's (freshtel) helpdesk ppl, who know hardly anything about it. I got fed all sorts of stories about the firmware when i got throught! Eventually I got some who went down to development, and couldn't find any one and told me that it was a way off, after that I wrote it off |
| 08:13.14 | hads|home | Fair enough. Did you ever use it with the IAX firmware? |
| 08:13.32 | Snake-Eyes | hads|home I think I got mine through some voip shop, not the website |
| 08:14.31 | Snake-Eyes | hads|home, nope, but after starting a forum thread asking about firmware and seeing ppl go on about the iax being crap... |
| 08:14.31 | *** join/#asterisk darkskiez (n=mbryars@194.247.78.146) |
| 08:15.39 | Snake-Eyes | hads|home, we use sip mostly, didnt really wanted to spend more time on this product, had enough on my plate :) |
| 08:16.00 | hads|home | Fair enough, just interested. I heard one person say it was great and you are now the third or forth saying it's bad so I think I have enough opinions now. |
| 08:17.53 | Snake-Eyes | hads|home, http://forums.whirlpool.net.au/forum-replies.cfm?t=482954 |
| 08:19.02 | hads|home | Cheers. Ah, another aussie :) |
| 08:19.18 | hads|home | Quite a few around here. |
| 08:19.38 | Snake-Eyes | hads|home, yea i wonder why *cough* telstra *cough* |
| 08:20.18 | hads|home | :) |
| 08:21.39 | Snake-Eyes | hads|home, btw i heard rumor that it might get locked in the future which wouldnt be surprising :) |
| 08:21.40 | EyeCue | can someone confirm this logic for me |
| 08:22.14 | EyeCue | create users in iax.conf, with context 'whatever', add [whatever] context into extensions, and give each user an extension right? |
| 08:22.22 | hads|home | Snake-Eyes: Yeah, wouldn't be surprising. It was originally a freshtel only thing wasn't it. |
| 08:23.34 | hads|home | EyeCue: Pretty much. Have you seen "the book"? |
| 08:23.52 | EyeCue | nope. |
| 08:23.54 | Snake-Eyes | hads|home, no idea if it started out as only frestel thing |
| 08:24.11 | EyeCue | just this whole context thing and Dial([SIP/IAX2/user]) blah is confusing me |
| 08:24.29 | kiong | exit |
| 08:24.30 | hads|home | Snake-Eyes: I think it was originally. |
| 08:24.32 | Snake-Eyes | ah context |
| 08:24.35 | EyeCue | now, im on the same local net as the pbx. but the other client is on the outside |
| 08:24.42 | EyeCue | both have peer definitions in iax.conf |
| 08:24.48 | EyeCue | both with the same context |
| 08:24.59 | EyeCue | should the dial args be IAX2, or SIP ? |
| 08:25.03 | *** part/#asterisk kilobit (n=seth@210.193.57.155) |
| 08:25.11 | Snake-Eyes | IAX2 |
| 08:25.29 | Snake-Eyes | if all the phones are using IAX |
| 08:25.45 | EyeCue | softphones |
| 08:25.45 | EyeCue | yeh |
| 08:25.48 | EyeCue | iaxComm |
| 08:25.55 | EyeCue | one can only assume they are. |
| 08:26.04 | EyeCue | im getting ring, but i cant freakin answer. |
| 08:26.09 | EyeCue | Jul 21 18:26:02 NOTICE[88337]: app_dial.c:1040 dial_exec_full: Unable to create channel of type 'SIP' (cause 3 - No route to destination) |
| 08:26.21 | *** join/#asterisk Gr1ncheux_ (n=devine@AStDenis-105-1-57-12.w80-8.abo.wanadoo.fr) |
| 08:26.33 | Snake-Eyes | whats in your extension.conf |
| 08:26.37 | hads|home | EyeCue: You are trying to dial SIP/somthing? |
| 08:26.40 | EyeCue | [iax] |
| 08:26.40 | EyeCue | exten => 1000,1,Dial(IAX2/koobs) |
| 08:26.40 | EyeCue | exten => 3000,1,Dial(IAX2/ryan) |
| 08:27.23 | hads|home | EyeCue: Again, have you seen the book? |
| 08:27.35 | EyeCue | 'the' book ? |
| 08:27.37 | EyeCue | the handbook ? |
| 08:27.42 | hads|home | ~thebook |
| 08:27.44 | jbot | thebook is probably a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 08:28.00 | hads|home | It's a pretty good starting point. |
| 08:29.53 | EyeCue | yeh but i dont get it, we're talkin super simplicity here :) |
| 08:30.00 | EyeCue | or perhaps is a conceptual misunderstanding. |
| 08:30.57 | hads|home | Well if you read the book then it will give you an overview of the concepts ;) |
| 08:31.19 | EyeCue | http://pastebin.ca/94443 |
| 08:31.22 | EyeCue | for the record |
| 08:32.08 | *** join/#asterisk ghenry (n=ghenry@80.229.93.1.plusnet.pte-ag2.dyn.plus.net) |
| 08:32.21 | *** join/#asterisk Modcuts (n=bob@lan.proporta.com) |
| 08:32.50 | *** join/#asterisk pa (n=Paolo@unaffiliated/pa) |
| 08:35.08 | NDT | Do any of the SAY cmds say currency right? Like 7.54 for 7 dollars 54 cents? |
| 08:35.53 | *** join/#asterisk r0d3nt (n=RatMan@foster.stonedcoder.org) |
| 08:38.30 | Dovid | can i have one macro call another macro ? |
| 08:40.07 | Snake-Eyes | EyeCue, are both registered with Asterisk? (iax2 show peers) |
| 08:40.36 | EyeCue | yup |
| 08:42.18 | EyeCue | see privmsg for output |
| 08:43.26 | hads|home | NDT: There is something like that on the bug tracker from memory. |
| 08:44.27 | *** join/#asterisk Gunnar (n=gunnar@62.97.242.6) |
| 08:45.36 | *** join/#asterisk stephane_ (n=stephane@merlin.cabale.net) |
| 08:45.41 | stephane_ | e |
| 08:46.36 | EyeCue | i swear its the client. |
| 08:46.59 | EyeCue | i get the ring, but i click to answer and i get > == Everyone is busy/congested at this time (1:0/0/1) in log |
| 08:52.46 | *** join/#asterisk Vec (n=Vector@dsl-146-119-118.telkomadsl.co.za) |
| 08:53.26 | Vec | I would like to playback(ringing) but can't find the sound file, does asterisk have a built in ringing sound ? |
| 08:54.54 | Vec | I tried ringing but it does not seem to work, with a Wait(3) |
| 08:59.34 | Vec | got it working :P |
| 09:00.56 | Vec | It does not work when I do the ringing after a dial command ? |
| 09:01.01 | Vec | only if I do it on its own |
| 09:01.13 | Vec | I need to get it to do the ringing during the Dial command ? |
| 09:01.27 | Vec | Do I set the priorities the same ? |
| 09:02.04 | h3x0r | you cant ever have the same priority number |
| 09:02.10 | h3x0r | what you want is to add the r option to dial |
| 09:02.13 | *** join/#asterisk nailbags (n=nailbags@c220-237-12-224.randw1.nsw.optusnet.com.au) |
| 09:02.38 | h3x0r | also there is a Ringing() command but its not useful if you want it to ring during Dial |
| 09:03.02 | *** join/#asterisk abatista (n=Ariel@dsl-20-177.cofs.net) |
| 09:03.07 | Vec | h3x0r : oh k thanks |
| 09:03.10 | nailbags | hi, what do i use in extensions.conf to notify when an extension is invalid? i thought exten => i,1,Playback(pbx-invalid), but thats not it |
| 09:03.46 | Vec | h3x0r : what is the n priority for ? (or is it for nothing and was just used in an example) ? |
| 09:04.00 | EyeCue | previous priotity plus 1 |
| 09:04.11 | EyeCue | previous priority + 1 |
| 09:04.12 | EyeCue | rather |
| 09:05.57 | EyeCue | dial 1000 for me |
| 09:06.41 | Vec | EyeCue : cool, it makes things easer |
| 09:06.50 | EyeCue | oops wrong window :D~ |
| 09:07.25 | h3x0r | ael2 is better |
| 09:07.26 | h3x0r | no line numbers |
| 09:07.29 | h3x0r | it uses labels |
| 09:07.59 | nailbags | noone knows how to match an invalid extension in extension.conf? |
| 09:11.24 | Vec | The r option in the dial command does not work with my hardware. |
| 09:11.45 | *** join/#asterisk h3x0r4t0r (i=hex@ip70-189-236-254.lv.lv.cox.net) |
| 09:12.24 | *** join/#asterisk speedwagon (n=Ariel@dsl-20-177.cofs.net) |
| 09:14.01 | stoffell | nailbags: option i, you can find this in the book and on voip-info.org.. |
| 09:14.39 | *** join/#asterisk l-fy (n=pchitesc@yate/developer/l-fy) |
| 09:15.10 | nailbags | stoffell: i'm trying option i, is the line i posted wrong? |
| 09:15.13 | *** join/#asterisk tparcina (n=tparcina@83-131-133-206.adsl.net.t-com.hr) |
| 09:15.23 | tparcina | hi channel! |
| 09:15.24 | nailbags | stoffell: i'm looking at voip-info.org |
| 09:15.39 | nailbags | hi tparcina |
| 09:15.50 | tparcina | grandstream gxp2000, by default what is password for regular user? |
| 09:15.53 | EyeCue | Anyone have recommendations for a decent win32 iax client? |
| 09:15.59 | tparcina | hi nailbags! |
| 09:16.36 | nailbags | tparcina: do you know what to use in extensions.conf to notify when an extension is invalid? i thought exten => i,1,Playback(pbx-invalid), but it doesn't work |
| 09:17.21 | tparcina | it's if someone dials unexisting extension |
| 09:17.33 | tparcina | then "i" tells asterisk what to do |
| 09:18.12 | zoa | eyecue, try idefisk |
| 09:18.19 | zoa | shameless plug : we make that |
| 09:18.26 | nailbags | yeah i want it to play 'pbx-invalid' when anyone dials an extension that doesn't exist. does that look right? |
| 09:18.30 | EyeCue | i dont mind, at least ill have someone to bitch to |
| 09:18.31 | EyeCue | :) |
| 09:18.35 | zoa | http://www.asteriskguru.com/idefisk/ |
| 09:18.37 | EyeCue | when shit breaks :D |
| 09:18.38 | EyeCue | yeh im there |
| 09:18.39 | EyeCue | ta |
| 09:18.48 | jalsot | hi |
| 09:19.23 | EyeCue | *installs* |
| 09:19.35 | stoffell | zoa: ah, you are from belgium too :) |
| 09:19.41 | EyeCue | nice logo |
| 09:20.02 | EyeCue | nice audio options ui, mostly :) |
| 09:20.39 | *** part/#asterisk kmilitzer (n=km@office-gw.westend.com) |
| 09:20.50 | EyeCue | nice small ui, cant find config to setup registration though :| |
| 09:21.13 | l-fy | morning |
| 09:21.28 | l-fy | hey jalsot |
| 09:21.29 | jalsot | does anybody know what might be the reason of too much 'Resyncing the jb'? |
| 09:22.16 | jalsot | the network seems to be ok, LAN and others on 2M Leased line... |
| 09:22.45 | jalsot | is there a tool which can analyze pure iax2 packets? plugint to ethereal? |
| 09:23.13 | jalsot | hi l-fy |
| 09:23.19 | l-fy | jalsot > ethereal supports iax2 |
| 09:23.26 | *** join/#asterisk creativx (n=creadure@196.82-134-19.bkkb.no) |
| 09:23.33 | l-fy | i can tell you for sure because we used when we've developed the yiax stac |
| 09:23.34 | l-fy | k |
| 09:23.41 | h3x0r4t0r | unleash networks makes a iax2 analyzer |
| 09:24.15 | jalsot | l-fy: yes, I know, but I didn't find any iax analyze option, like for RTP |
| 09:24.17 | h3x0r4t0r | just turn off jitterbuffer |
| 09:24.18 | h3x0r4t0r | its broke |
| 09:24.34 | jalsot | what I mean, to get a general info about lost frames, jitter, etc. |
| 09:24.54 | jalsot | h3x0r4t0r: yep, we turned off JB, however quality is still bad :( |
| 09:25.15 | jalsot | also tried resynctreshold=-1 |
| 09:25.31 | jalsot | I have a guess, that the problem is with monitor application |
| 09:25.49 | jalsot | every call [iax2<->zap] is monitored into file... |
| 09:26.04 | MikeJ[Laptop] | try mixmonitor? |
| 09:26.10 | jalsot | over about 25-30 calls, quality starts to be bad, as customer says |
| 09:26.30 | *** join/#asterisk kmilitzer (n=km@office-gw.westend.com) |
| 09:26.30 | jalsot | MikeJ[Laptop]: yes, that would be good, but mixmonitor had a bug and I should upgrade asterisk for that |
| 09:26.33 | EyeCue | zoa, very very nice. |
| 09:26.48 | EyeCue | zoa, you're just missing an option button in the main window. |
| 09:27.03 | jalsot | and while it seems asterisk has general issues with queues/agentcallbackligin over 1.2.6, I fear to upgrade |
| 09:27.06 | zoa | i dont want one :p |
| 09:27.19 | jalsot | what I fear is: http://bugs.digium.com/view.php?id=6626 |
| 09:27.32 | h3x0r4t0r | as i was saying |
| 09:27.39 | h3x0r4t0r | unleash networks has a iax2 analyzer |
| 09:28.17 | *** join/#asterisk af_ (n=af@ip-164-6.sn2.eutelia.it) |
| 09:28.26 | h3x0r4t0r | a company |
| 09:28.36 | *** join/#asterisk viperdude (n=jon@195.74.96.114) |
| 09:28.37 | MikeJ[Laptop] | jalsot, I have talked to several people using callback login with that same issue as of 1.2.9 and later in a very bad way. |
| 09:29.24 | jalsot | MikeJ[Laptop]: so you see, why I fear to upgrade |
| 09:29.58 | jalsot | older mixmonitor stopped storing data in some conditions, which was fixed in 1.2.10 |
| 09:30.50 | nailbags | stoffell: i'm reading the book and my dialplan entry is exactly the same: 'exten => i,1,Playback(pbx-invalid)' do you know why it wouldn't be working? |
| 09:32.00 | *** join/#asterisk fenlander (n=fenlande@82.152.81.57) |
| 09:32.27 | jalsot | MikeJ[Laptop]: do you know wether anybody is working on that agentcallback issue? |
| 09:33.03 | MikeJ[Laptop] | no.. I was told that it was a race condition and that people were having a hard time catching it to find it. |
| 09:33.16 | *** join/#asterisk bofh42 (n=bofh42@p54828ED5.dip0.t-ipconnect.de) |
| 09:33.22 | l-fy | MikeJ[Laptop] > thank you |
| 09:41.12 | jalsot | MikeJ[Laptop]: do you think the quality issues might be introduced because of Monitor? I have a doubt if mixmonitor will help |
| 09:43.04 | EyeCue | Anyone mind testing my iax with a client, whatever? :) |
| 09:43.58 | l-fy | EyeCue > shoot |
| 09:44.07 | EyeCue | <3 long time |
| 09:45.34 | MikeJ[Laptop] | jalsot, with monitor, are you using sox to mix the audio after the call? |
| 09:46.30 | *** join/#asterisk xbit` (n=xbit@frugalware.elte.hu) |
| 09:46.33 | xbit` | hi |
| 09:47.24 | stephane_ | re |
| 09:48.01 | jalsot | MikeJ[Laptop]: yes, asterisk runs with -p option |
| 09:48.18 | jalsot | right now I'm not mixing right after the end of call |
| 09:49.47 | *** join/#asterisk carl0s- (n=carl0s@compsup.demon.co.uk) |
| 09:51.15 | MikeJ[Laptop] | well.. my guess out the window then... heh |
| 09:54.01 | nailbags | can someone please give me a hand with my extensions.conf (http://pastebin.ca/94553) everything works except the 'i' extension. i'm trying to play a message if an invalid extension is dialed |
| 09:56.10 | xbit` | i have a voip provider whos incoming calls looks like come in twice, and then the connection hangs up. |
| 09:56.13 | xbit` | http://pastebin.com/753917 |
| 09:56.30 | zoa | jalsot: yes |
| 09:56.38 | zoa | if you do a lot of monitoring yes |
| 09:56.42 | zoa | quality will go down |
| 09:56.45 | zoa | because of the IO |
| 09:56.54 | zoa | mixmonitor will only get things worse |
| 09:57.36 | jalsot | zoa: oh, what can be the solution? |
| 09:58.20 | jalsot | I read some comments about IAX2 and its single thread processing [in 1.2] |
| 09:58.31 | jalsot | maybe multithreaded IAX might help? |
| 09:58.50 | jalsot | [unfortunately upgrading to trunk is not a way to go right now] |
| 09:59.45 | *** join/#asterisk _omer (n=_omer@202.38.51.2) |
| 09:59.48 | *** join/#asterisk dlynes_laptop (n=dlynes@zz212094.cipherkey.net) |
| 09:59.52 | _omer | hi |
| 10:00.12 | _omer | when I do make clean in zaptel ... i get following errors.. |
| 10:00.14 | _omer | make -C /lib/modules/2.6.11-1.1369_FC4/build SUBDIRS=/usr/src/zaptel-1.2.7 clean |
| 10:00.14 | _omer | make: *** /lib/modules/2.6.11-1.1369_FC4/build: No such file or directory. Stop. |
| 10:00.14 | _omer | make: *** [clean] Error 2 |
| 10:00.18 | _omer | anyone plz? |
| 10:00.41 | h3x0r4t0r | why are you using 2.6.11 |
| 10:00.48 | *** join/#asterisk bionoid (n=root@5.81-166-175.customer.lyse.net) |
| 10:00.51 | dlynes_laptop | _omer, you don't have kern-dev installed |
| 10:00.53 | nailbags | _omer: is that module directory correct? (does it exist?) |
| 10:01.06 | _omer | not sure.... |
| 10:01.16 | dlynes_laptop | h3x0r4t0r, because he's using fedora core 4, obviously |
| 10:01.18 | _omer | dlynes_laptop : where do I get it from ? |
| 10:01.26 | h3x0r4t0r | yum update |
| 10:01.27 | _omer | yes FC4 |
| 10:01.30 | dlynes_laptop | _omer, beats the hell out of me...I don't use fedora |
| 10:01.44 | dlynes_laptop | try /join #fedora |
| 10:01.49 | _omer | ohhh great! |
| 10:02.14 | dlynes_laptop | or maybe what h3x0r4t0r said...but i think he only told you a vague answer |
| 10:02.46 | hads|home | nailbags: http://www.voip-info.org/wiki/index.php?page=Asterisk+i+extension |
| 10:03.08 | dlynes_laptop | EyeCue, btw...i'm back now |
| 10:03.15 | EyeCue | hehe wb :) |
| 10:03.15 | hads|home | lo dlynes_laptop |
| 10:03.18 | EyeCue | more issues ;) |
| 10:03.19 | dlynes_laptop | EyeCue, but can't help you with testing..I'm at a coffee shop now |
| 10:03.26 | EyeCue | thats ok :D |
| 10:03.26 | dlynes_laptop | EyeCue, so no access to a sip phone |
| 10:03.31 | EyeCue | ill be makin an espresso soon |
| 10:03.32 | EyeCue | hehe |
| 10:03.33 | EyeCue | well |
| 10:03.34 | dlynes_laptop | heya hds |
| 10:03.36 | EyeCue | ive gotten rid of sip users |
| 10:03.40 | EyeCue | and tried to get iax workin |
| 10:03.45 | dlynes_laptop | EyeCue, yeah...turns out that fire alarm was my house after all |
| 10:03.46 | EyeCue | now im getting weird codec errors |
| 10:04.09 | dlynes_laptop | EyeCue, one of the sprinkler heads burst out of the end of the pipe |
| 10:04.12 | EyeCue | ack |
| 10:04.13 | EyeCue | :) |
| 10:04.16 | dlynes_laptop | EyeCue, so we had water everywhere |
| 10:04.26 | dlynes_laptop | had to get the water shut off at the street |
| 10:04.42 | hads|home | Doesn't sound like fun |
| 10:04.43 | EyeCue | not good :) |
| 10:04.47 | dlynes_laptop | not especially |
| 10:04.50 | l-fy | dlynes_laptop > i'm testing yate with eyecue and are small problems |
| 10:04.53 | l-fy | but will be fixed |
| 10:05.08 | nailbags | hads |
| 10:05.09 | dlynes_laptop | thankfully only the odd receipt that I left on the floor got wet, and most of my clothes |
| 10:05.19 | hads|home | nailbags |
| 10:05.21 | dlynes_laptop | l-fy, cool |
| 10:05.36 | EyeCue | <PROTECTED> |
| 10:05.37 | EyeCue | gah |
| 10:05.40 | EyeCue | new problems |
| 10:05.48 | dlynes_laptop | l-fy, yeah...I think most of the problems are on his end |
| 10:05.53 | l-fy | EyeCue > please call on sip or disable iax bridging that's normal |
| 10:05.54 | EyeCue | id say so :) |
| 10:06.03 | l-fy | dlynes_laptop > you're right but can be fixed |
| 10:06.05 | nailbags | hads|home: dude, i've read it. and my 'i' in extensions.conf is identical to the book, but it doesn't work and i get nothing in the asterisk console |
| 10:06.06 | l-fy | so is not an issue |
| 10:06.19 | dlynes_laptop | l-fy, and it looks like he's having yate problems, not asterisk problems :) |
| 10:06.28 | l-fy | yate problems? |
| 10:06.29 | EyeCue | ;) |
| 10:06.31 | l-fy | what do you mean? |
| 10:06.33 | EyeCue | well hang on, clients have a codec config, and so does the server |
| 10:06.38 | dlynes_laptop | l-fy, he's using yate, right? |
| 10:06.40 | nailbags | hads|home: and if i change it to something else like '106' (an extension that doesn't exist) then it plays the invalid message |
| 10:06.41 | l-fy | no |
| 10:06.41 | EyeCue | do i HAVE TO specify codec in the peer config? |
| 10:06.44 | l-fy | him is using asterisk |
| 10:06.46 | l-fy | i'm using yate |
| 10:06.50 | l-fy | this is why i don't have problems |
| 10:06.51 | EyeCue | dlynes, idefisk |
| 10:06.57 | dlynes_laptop | l-fy, well, the error message he got isn't an asterisk error |
| 10:06.58 | dlynes_laptop | ah |
| 10:07.01 | l-fy | it is |
| 10:07.03 | dlynes_laptop | it's an idefisk error |
| 10:07.04 | zoa | what is eyecue ? |
| 10:07.12 | EyeCue | well apparently these guys know |
| 10:07.13 | l-fy | yate dosen't even supports iax bridging |
| 10:07.13 | EyeCue | :) |
| 10:07.27 | l-fy | EyeCue > ok, ready to do one more test? |
| 10:07.31 | EyeCue | so its idefisk |
| 10:07.33 | EyeCue | sure |
| 10:07.39 | EyeCue | suggest config changes if youre up for it :) |
| 10:07.44 | l-fy | did you connected yourself over sip? |
| 10:07.49 | nailbags | hads|home: can you see anything wrong with my extensions.conf? |
| 10:07.57 | hads|home | nailbags: Did you actually read that wiki page? |
| 10:08.08 | EyeCue | no. |
| 10:08.10 | nailbags | hads|home: yeah |
| 10:08.17 | EyeCue | dlynes got me off sip :) |
| 10:08.37 | nailbags | hads|home: oh sorry, maybe not |
| 10:08.42 | l-fy | is there anyway bridging of native channels on iax can be disabled? |
| 10:08.46 | dlynes_laptop | EyeCue, i merely suggested you use iax for the end points, not for the phones |
| 10:08.52 | EyeCue | :| |
| 10:08.55 | EyeCue | copy that. |
| 10:08.59 | l-fy | dlynes_laptop > let's fix that first |
| 10:09.07 | jalsot | zoa: storing 60 channels with 2x16bit/8kHz is about 16Mbit/s load on HDD, is that a problem on modern system? [we have 3ware RAID2 with SATA discs] |
| 10:09.13 | l-fy | and btw for the phones is good since is the only way sometimes to pass nat |
| 10:09.20 | zoa | jalsot: yes |
| 10:09.28 | zoa | its not ok |
| 10:09.30 | EyeCue | so do i want to take my user defs out of iax.conf? |
| 10:09.33 | dlynes_laptop | l-fy, he's got an asterisk box going through the nat though, right? |
| 10:09.35 | EyeCue | and whack em in sip? |
| 10:09.39 | EyeCue | correct. |
| 10:09.40 | dlynes_laptop | l-fy, i think his phones are on the same network |
| 10:09.48 | EyeCue | well my softphone is |
| 10:09.51 | EyeCue | outside peoples arent |
| 10:09.55 | zoa | even if you have fast scsi disks it would still be a problem |
| 10:10.03 | dlynes_laptop | EyeCue, ah, and outside peeps are using sip? |
| 10:10.06 | nailbags | hads|home: so is the book incorrect? |
| 10:10.08 | EyeCue | well |
| 10:10.10 | zoa | i wrote about it on the mailinglist before |
| 10:10.12 | EyeCue | ive had them try iaxcomm |
| 10:10.13 | jalsot | [sorry, I meant RAID5] |
| 10:10.14 | EyeCue | and idefisk too |
| 10:10.20 | dlynes_laptop | EyeCue, ah |
| 10:10.22 | *** join/#asterisk Sonderblade (n=mah@static-213.131.147.169.addr.tdcsong.se) |
| 10:10.31 | *** join/#asterisk qdk (n=qdk@213.237.44.34) |
| 10:10.37 | EyeCue | xtens is sip, and makes pc go all haywire. |
| 10:10.42 | jalsot | what is the problem actually? storing continouse small packets? |
| 10:10.46 | EyeCue | iaxcomm im assuming is aix |
| 10:10.54 | dlynes_laptop | EyeCue, iax, not aix |
| 10:10.56 | jalsot | buffering writes can help? |
| 10:10.58 | EyeCue | yeh yeh :) |
| 10:11.00 | dlynes_laptop | EyeCue, remember...aix is ibm's unix |
| 10:11.04 | EyeCue | iax. |
| 10:11.12 | nailbags | hads|home: wow, that works. ty so much |
| 10:11.12 | hads|home | nailbags: I'm not sure what part of the book you are referring to so I'm unsure. Also I don't use the i extensions myself, but I know that it doesn't work in the way that some people think it should. |
| 10:11.24 | jalsot | I mean, to collect frames for e.g. 5 seconds and write only these chunks together |
| 10:11.24 | hads|home | nailbags: np |
| 10:11.36 | EyeCue | soi move users from iax.conf into sip.conf, yes? |
| 10:11.49 | dlynes_laptop | hads|home, 'i' invalid extension happens when you enter an invalid extension for the given context |
| 10:12.16 | EyeCue | except for your definition for iax |
| 10:12.28 | *** part/#asterisk Sonderblade (n=mah@static-213.131.147.169.addr.tdcsong.se) |
| 10:12.31 | dlynes_laptop | hads|home, so say you dial 300 in the incoming context, and you've got every extension covered except extension 300, if you have 'i' defined, it'll go to that |
| 10:12.45 | hads|home | dlynes_laptop: Yeah, but aparently not all the time. I know I have seen a number of people have trouble with it not working as they expect. |
| 10:12.54 | hads|home | Like on that wiki page. |
| 10:13.00 | zoa | jalsot: http://www.asteriskguru.com/archives/asterisk-users-success-512-simultaneous-calls-with-digit-vt40206.html?highlight=zoa |
| 10:13.10 | dlynes_laptop | hads|home, because they're using that new option in 1.2, and they don't understand the implications of it |
| 10:13.11 | EyeCue | zoa, does idefisk do sip? |
| 10:13.13 | EyeCue | 1.37 that is |
| 10:13.21 | dlynes_laptop | hads|home, autofallthrough=yes |
| 10:13.25 | zoa | eyecue, not yet |
| 10:13.25 | jalsot | zoa: thx, reading |
| 10:13.29 | EyeCue | ahhhhh |
| 10:13.30 | zoa | v.2.0 |
| 10:13.31 | nailbags | dlynes_laptop: hads|home: yeah it doesn't for my configuration. the wiki page seems accurate to me, but the book seems to be wrong |
| 10:13.33 | EyeCue | *changes client* |
| 10:13.49 | dlynes_laptop | nailbags, are you using autofallthrough=yes? |
| 10:13.52 | nailbags | dlynes_laptop: isn't that set by default? or are you saying we should unset it? |
| 10:13.59 | knarfly | I have X-Lite working fine from a remote location with my * box |
| 10:14.10 | knarfly | I teid to setup a friend last night and it would not work |
| 10:14.11 | dlynes_laptop | nailbags, i'm saying you should set autofallthrough=no, unless you have a specific reason to set it |
| 10:14.29 | knarfly | is it poosible his ISP is blocking ports to discourage VOIP? |
| 10:14.30 | dlynes_laptop | nailbags, 99 times out of 100, you don't want that behaviour |
| 10:14.44 | nailbags | dlynes_laptop: ok, i see. so why the hell is it enabled by default? |
| 10:14.51 | EyeCue | ok |
| 10:14.57 | dlynes_laptop | nailbags, i don't think it is |
| 10:15.04 | nailbags | dlynes_laptop: it is |
| 10:15.11 | dlynes_laptop | nailbags, ah, ok...disable it then |
| 10:15.22 | hads|home | dlynes_laptop: But autofallthrough shouldn't affect calls that haven't matched an extension yet. |
| 10:15.22 | dlynes_laptop | nailbags, it's a huge source of frustration |
| 10:15.42 | nailbags | dlynes_laptop: well if you have no autofallthough line at all it defaults to on. the documentation says that too |
| 10:15.56 | nailbags | dlynes_laptop: but it makes no mention of that stopping 'i' from working :( |
| 10:16.08 | knarfly | is it possible an ISP is blocking ports to discourage VOIP? |
| 10:16.12 | *** join/#asterisk DarKnesS_WolF (n=wolf@62.114.187.139) |
| 10:16.23 | dlynes_laptop | nailbags, also, are you using anything like Set(TIMEOUT(response)=10) ? |
| 10:16.26 | EyeCue | zoa, wanna send me 2.0? :) |
| 10:16.45 | zoa | nopez |
| 10:16.52 | zoa | its not available yet |
| 10:17.03 | dlynes_laptop | nailbags, it's not 'i' specifically it affects |
| 10:17.11 | dlynes_laptop | nailbags, it's all jumping, in general, I think |
| 10:17.29 | dlynes_laptop | nailbags, also, there's another option in 1.2 something like priorityjumping=no |
| 10:17.37 | *** join/#asterisk RoyK (n=roy@122.80-203-78.nextgentel.com) |
| 10:17.39 | EyeCue | well now just need someone to test dial, now that i switches users back to sip |
| 10:17.46 | nailbags | dlynes_laptop: no. i don't think so. i pastebinned my extensions.conf if you scroll up. gotta run right now. but feel free to grab me later if you want more info |
| 10:17.48 | dlynes_laptop | nailbags, set that option to yes; in 1.2, the new default is no, but in 1.0 it was yes |
| 10:18.03 | dlynes_laptop | nailbags, you pb'd it before i logged back in |
| 10:18.28 | hads|home | dlynes_laptop: autofallthrough shouldn't affect jumping. |
| 10:18.45 | EyeCue | wow, xten does blow, fear the resource usage and everything else hanging. |
| 10:18.58 | dlynes_laptop | nailbags, but priority jumping is say before if there was no answer, it would jump to 101 priority, it wasn't anything to do with predefined extensions such as 'i' |
| 10:19.44 | dlynes_laptop | hads|home, no, they're two different things, but if he was reading documentation that was out of date, he might have mistakenly assumed priority jumping was still the norm in 1.2, also |
| 10:19.46 | carl0s- | I'm a bit stuck here. I have my voip GSM gateway, but in the SIP configuration it only allows me to set: "user name" "register name" "register password" "domain server" "proxy server" and "outbound proxy". This suggests it is to be configured as an Extension, rather than a Trunk. What do you think? |
| 10:19.58 | dlynes_laptop | hads|home, a lot of the documentation on voip-info.org is out of date |
| 10:20.26 | hads|home | dlynes_laptop: Yes, right you are. |
| 10:20.35 | dlynes_laptop | of course i am |
| 10:20.38 | dlynes_laptop | i'm always right |
| 10:20.43 | hads|home | :) |
| 10:26.53 | *** join/#asterisk Faithful (n=Faithful@ns.linuxterminal.com) |
| 10:27.01 | *** join/#asterisk FaithX (n=FaithX@ns.linuxterminal.com) |
| 10:28.16 | nailbags | dlynes_laptop: well here's the pastebin anyway: http://pastebin.ca/94553 |
| 10:28.43 | dlynes_laptop | ok, checking |
| 10:29.02 | nailbags | dlynes_laptop: trying with autofallthrough=no now |
| 10:29.38 | dlynes_laptop | nailbags, yeah...i don't see anything there that would prevent 'i' from getting called |
| 10:29.48 | dlynes_laptop | nailbags, nor do i see any reason why it would get called |
| 10:31.24 | nailbags | dlynes_laptop: so are you saying i did something wrong? autofallthough=no does work ... |
| 10:32.49 | nailbags | dlynes_laptop: oops i mean it _doesn't_ work |
| 10:33.01 | dlynes_laptop | heh |
| 10:33.20 | dlynes_laptop | nailbags, can you paste a snapshot of your full log when the problem occurs? |
| 10:33.29 | dlynes_laptop | nailbags, i.e. /var/log/asterisk/full? |
| 10:33.54 | nailbags | dlynes_laptop: i don't have that log file |
| 10:34.03 | dlynes_laptop | nailbags, so enable it then |
| 10:34.08 | nailbags | how? |
| 10:34.12 | dlynes_laptop | nailbags, /etc/asterisk/logger.conf is where you configure it |
| 10:34.12 | *** join/#asterisk creativx (n=creadure@196.82-134-19.bkkb.no) |
| 10:34.27 | dlynes_laptop | nailbags, after you've changed that file, just do a 'logger reload' from the cli |
| 10:36.03 | nailbags | dlynes_laptop: ok got full loggin turned on. nothing appears in that log when i dial the invalid extension |
| 10:37.30 | dlynes_laptop | can you enable debug for the full log as well? |
| 10:37.39 | dlynes_laptop | also, set verbose to 9 before calling |
| 10:38.26 | bionoid | Hello sirs; I'm trying to install a 'Tiger3XX' (complete newbie, sorry), as far as I can tell I need the Zaptel driver. I've loaded the zaptel and wctdm modules without errors, however I can't find any indicators that it actually _detects_ the card. Any pointers on how to verify this? Thanks |
| 10:38.49 | dlynes_laptop | bionoid, after you do the modprobe wctdm |
| 10:38.58 | dlynes_laptop | bionoid, then wait about 5 seconds |
| 10:39.03 | dlynes_laptop | bionoid, and then do a ztcfg -vvvvvvvvvvvv |
| 10:39.21 | dlynes_laptop | bionoid, you should get a status message back letting you know if the channels are enabled or if there was an error |
| 10:39.49 | bionoid | It says 0 channels, but no errors as such |
| 10:40.01 | dlynes_laptop | bionoid, btw...is your tiger3xx an x100p/x101p clone, or is it a tdm400p clone? |
| 10:40.18 | bionoid | I was hoping noone would ask those kinds of questions. :P |
| 10:40.28 | dlynes_laptop | bionoid, so you don't know? |
| 10:40.38 | bionoid | I have no idea, sorry, I've been working with Asterisk and telephone technology for ~2 hours of my life |
| 10:41.07 | bionoid | What I do know, is that it's very cheap and has only one analog port |
| 10:41.13 | dlynes_laptop | bionoid, does it have little mini cards that plug into the pci card, or does it have two hard-wired ports in the middle of the card at the back? |
| 10:41.30 | bionoid | ah, two, yes sorry |
| 10:41.31 | dlynes_laptop | bionoid, one of the ports would be labelled line, and the other labelled phone |
| 10:41.47 | bionoid | Yup that's what I have |
| 10:41.55 | dlynes_laptop | bionoid, ok, you're loading hte wrong driver then |
| 10:41.57 | *** join/#asterisk xbmodder_newlapp (i=nobody@atarack/staff/xbmodder) |
| 10:41.58 | xbmodder_newlapp | hey |
| 10:42.06 | dlynes_laptop | bionoid, you should be loading wcfxo, not wctdm |
| 10:42.11 | carl0s- | is SIP like a peer-to-peer protocol? If I have a GSM VoIP gateway which I am able to get running as an 'extension' within Asterisk, should I therefore be able to configure it as a Trunk also? |
| 10:42.11 | bionoid | aha |
| 10:42.26 | dlynes_laptop | carl0s-, correct |
| 10:42.39 | bionoid | my uneducated driver guess was wrong, then, thank you very much sir, I'm confident I will be bothering you again shortly ;-) |
| 10:43.02 | carl0s- | dlynes_laptop: hmm. OK. So I should be able to do what I want with my VoIP gateway then. I just need to understand the Asterisk trunk configuration a bit more. |
| 10:43.38 | dlynes_laptop | carl0s-, yeah...just think of your gsm gateway as another sip upstream provider |
| 10:43.52 | hads|home | I've just been doing some testing with an i extension and Asterisk returns a 404 instead of executing the i extension. It won't show anything on the console unless you enable sip debug. |
| 10:44.30 | *** join/#asterisk [Airwolf] (n=airwolf@dsl5402BD52.pool.t-online.hu) |
| 10:44.43 | nailbags | dlynes_laptop: ok sorry, though i was verbose but i must've restarted asterisk. |
| 10:44.45 | carl0s- | dlynes_laptop: I'm trying to. But the only options the gateway gives me are "username, password, proxy server, domain server, and outbound proxy". Is there some stuff missing? |
| 10:44.45 | nailbags | Jul 21 20:42:35 DEBUG[11550] chan_sip.c: Setting NAT on RTP to 0 |
| 10:44.45 | nailbags | Jul 21 20:42:35 DEBUG[11550] chan_sip.c: Stopping retransmission on '140a926-d6181ea4@10.10.10.99' of Response 101: Match Found |
| 10:44.46 | nailbags | Jul 21 20:42:35 DEBUG[11550] chan_sip.c: Setting NAT on RTP to 0 |
| 10:44.46 | nailbags | Jul 21 20:42:35 DEBUG[11550] chan_sip.c: Checking SIP call limits for device ext100 |
| 10:44.46 | dlynes_laptop | hads|home, so, what's happening then is it's dialing a valid extension...that's why i is never reached |
| 10:45.19 | dlynes_laptop | nailbags, ext100 is at ip address 10.10.10.99, right? |
| 10:45.28 | nailbags | dlynes_laptop: yep |
| 10:45.42 | dlynes_laptop | nailbags, yeah..you need to show the log after this |
| 10:45.49 | dlynes_laptop | nailbags, you haven't showed me the failure yet |
| 10:45.57 | dlynes_laptop | nailbags, that's just the call setup |
| 10:46.08 | nailbags | oops there's another line: |
| 10:46.10 | nailbags | Jul 21 20:42:35 DEBUG[11550] chan_sip.c: Stopping retransmission on '140a926-d6181ea4@10.10.10.99' of Response 102: Match Found |
| 10:46.31 | nailbags | then later i get this, but i don't know if its related: |
| 10:46.33 | nailbags | Jul 21 20:42:41 DEBUG[11550] chan_sip.c: Auto destroying call '2b5a710845c5aa5608ae902831e52b51@iinetphone.iinet.net.au' |
| 10:46.33 | nailbags | Jul 21 20:42:53 DEBUG[11550] chan_sip.c: Auto destroying call '2b4f27b367b9447a637f006b6b7fc462@sip01.mynetfone.com.au' |
| 10:46.35 | dlynes_laptop | carl0s-, username, password, proxy server, domain server, and forget outbound proxy |
| 10:46.53 | dlynes_laptop | nailbags, nope |
| 10:46.54 | carl0s- | dlynes_laptop: and I just set corresponding in the trunk configuration? type=friend or peer? |
| 10:47.05 | dlynes_laptop | nailbags, can you pastebin about 300 lines of log files or so? |
| 10:47.09 | nailbags | dlynes_laptop: thats all thats in the log |
| 10:47.10 | dlynes_laptop | nailbags, i.e. do a logger rotate |
| 10:47.14 | dlynes_laptop | nailbags, then do a call |
| 10:47.15 | nailbags | dlynes_laptop: there are no more lines |
| 10:47.18 | dlynes_laptop | nailbags, then logger rotate |
| 10:47.24 | dlynes_laptop | nailbags, then pastebin the entire full log file |
| 10:47.55 | dlynes_laptop | carl0s-, type=friend |
| 10:48.07 | carl0s- | dlynes_laptop: what about the register-string which is normally needed with SIP providers? just skip it? |
| 10:48.07 | dlynes_laptop | carl0s-, forget user and peer |
| 10:48.18 | dlynes_laptop | carl0s-, does the gsm gateway have a dynamic ip address? |
| 10:48.38 | nailbags | dlynes_laptop: i pasted you all the lines. thats all there is |
| 10:48.47 | hads|home | dlynes_laptop: I just setup a context with only a couple of SIP devices and an i extension and dialed a known non-existant extension. The i extension doesn't get executed, Asterisk returns a 404 instead. |
| 10:48.59 | carl0s- | dlynes_laptop: no. fixed private ip. |
| 10:49.02 | nailbags | hads|home: same here |
| 10:49.10 | dlynes_laptop | nailbags, you couldn't have, unless all you're logging is debug priority |
| 10:49.33 | dlynes_laptop | nailbags, you should have something like full => error,warning,notice,debug,verbose,dtmf |
| 10:49.36 | nailbags | dlynes_laptop: in logger.conf: full => notice,warning,error,debug,verbose |
| 10:49.53 | dlynes_laptop | or without the dtmf is fine, too |
| 10:50.10 | dlynes_laptop | ok, all i'm seeing from you is debug priority logging messages |
| 10:50.25 | dlynes_laptop | i'm not seeing any notices, warnings, or errors |
| 10:50.36 | Assid | hey dlynes_laptop: do you sleep? |
| 10:50.37 | dlynes_laptop | you sure there's none of those types of messages? |
| 10:50.51 | dlynes_laptop | Assid, yeah, i do but our fire sprinkler broke tonight |
| 10:51.01 | dlynes_laptop | Assid, so i'm at a 24 hour coffeeshop with a hotspot right now |
| 10:51.05 | rob0 | And besides, it's not daylight yet! |
| 10:51.08 | nailbags | dlynes_laptop: no. if i do a 'reload' i get them. but not after trying to call an invalid extension |
| 10:52.02 | Assid | okay when you mean it broke ? as in its blasting away water? or its considered unsafe to be in the building without it ? |
| 10:52.14 | dlynes_laptop | Assid, was blasting away water |
| 10:52.21 | dlynes_laptop | Assid, the fire chief shut the water off at the street |
| 10:52.33 | Assid | wow |
| 10:52.49 | l-fy | back |
| 10:53.27 | l-fy | hey RoyK |
| 10:53.30 | l-fy | where are you now? |
| 10:53.39 | RoyK | oslo |
| 10:53.43 | RoyK | home |
| 10:53.50 | Assid | dlynes_laptop: you planning on going to work 'today' ? |
| 10:54.00 | dlynes_laptop | yeah |
| 10:54.07 | dlynes_laptop | Assid, i have a meeting with Telus at 8:30am |
| 10:54.18 | dlynes_laptop | so another 4.5 hours |
| 10:55.02 | l-fy | cool rob0 |
| 10:55.04 | l-fy | cool RoyK |
| 10:55.16 | l-fy | i got your message than but i was unable to answer for 2 days |
| 10:56.42 | Assid | cool |
| 10:56.47 | Assid | good determination.. |
| 10:58.36 | *** join/#asterisk arcy (n=arcanum@ppp139-238.adsl.forthnet.gr) |
| 11:02.43 | hads|home | dlynes_laptop, nailbags: see bug 4038 on mantis for some disscussion of the i extension. The summary (which I have come accross before come to think of it) is that the i extension is only for handling an invalid extension dialed from WaitExten / Background etc. |
| 11:03.15 | nailbags | hads|home: can you link me? (sorry) |
| 11:03.35 | hads|home | nailbags: http://bugs.digium.com/view.php?id=4038 it's from a while back |
| 11:04.14 | carl0s- | dlynes_laptop: what about "fromuser=" and "fromdomain=". These are configured on my (working) upstream SIP provider trunk. Should I enter them for my local gsm voip gateway too? |
| 11:04.26 | _omer | make clean in zaptel gives this error |
| 11:04.27 | _omer | make -C SUBDIRS=/usr/src/zaptel-1.2.7 clean |
| 11:04.27 | _omer | make: *** SUBDIRS=/usr/src/zaptel-1.2.7: No such file or directory. Stop. |
| 11:04.27 | _omer | make: *** [clean] Error 2 |
| 11:04.31 | _omer | any help ? |
| 11:04.40 | *** join/#asterisk BugKham (i=BugKham@202.8.86.164) |
| 11:04.42 | _omer | I am already in the same folder |
| 11:05.07 | BugKham | hello, how can we reset the ${DNID} in the dialplan |
| 11:05.17 | dlynes_laptop | _omer, you don't have zaptel installed in /usr/src/zaptel-1.2.7 |
| 11:05.35 | dlynes_laptop | carl0s-, probably no need |
| 11:05.37 | BugKham | SetVar(DNIS=${EXTEN}) ? |
| 11:05.59 | BugKham | does it effect the ${DNID} value? |
| 11:06.08 | _omer | dlynes_laptop : /usr/src/zaptel-1.2.7 <--- thats a source directory ..where I do "make clean" |
| 11:06.46 | carl0s- | dlynes_laptop: thanks |
| 11:06.47 | nailbags | hads|home: dlynes_laptop: ok, but i don't really understand. is this bug affecting everyone, or am i doing something straing? |
| 11:07.22 | dlynes_laptop | nailbags, it's not affecting me, but i've relied on 'i' as an invalid dialed extension in a given context, either |
| 11:07.32 | dlynes_laptop | s/i've/i've never/g |
| 11:07.33 | nailbags | dlynes_laptop: i see |
| 11:08.01 | hads|home | nailbags: I'm in the same boat as dlynes_laptop |
| 11:08.18 | nailbags | dlynes_laptop: but voip-info's extension.conf page and the book are both wrong, right? |
| 11:08.47 | nailbags | well i can use _. instead, but i've read that its discouraged as well |
| 11:09.10 | hads|home | The solution of including a context with _X. in it at the bottom of your context seems reasonable. |
| 11:09.29 | _omer | dlynes_laptop : /usr/src/zaptel-1.2.7 <--- thats a source directory ..where I do "make clean" |
| 11:09.58 | nailbags | hads|home: well it definitely works. and doesn't seem to have any side-effects at least in my dialplan |
| 11:10.52 | carl0s- | hmm SIP2.0/404 Not found: http://pastebin.ca/94612 |
| 11:11.44 | nailbags | hads|home: oops i spoke to soon. that seems to break my outgoing calls, cause they're in the same context |
| 11:12.07 | dlynes_laptop | nailbags, yeah...your outgoing calls and incoming calls going into the same context is just asking for trouble |
| 11:12.10 | nailbags | hads|home: but i think i can hack around that |
| 11:12.45 | *** join/#asterisk SplasPood (n=jwb@brooklyn.paravolve.net) |
| 11:13.05 | hads|home | So did you include a context with the _X. in it? Or just bung the _X. in your context? Cos it does make a difference. |
| 11:13.16 | dlynes_laptop | hads, bung |
| 11:13.36 | _omer | http://pastebin.ca/94615 .... any help ? |
| 11:13.53 | hads|home | sorry, put :) |
| 11:16.36 | dlynes_laptop | hads, put, bung, bunghole, what's the difference? :) |
| 11:17.06 | hads|home | It almost looks like you included my nick in that list :) |
| 11:18.00 | dlynes_laptop | heh |
| 11:18.02 | *** join/#asterisk knarfly (n=bmorris@c-69-180-98-189.hsd1.fl.comcast.net) |
| 11:18.09 | dlynes_laptop | that was on purpose, didntcha know? |
| 11:18.17 | hads|home | :) |
| 11:19.43 | _omer | http://pastebin.ca/94615 .... any help ? |
| 11:20.39 | dlynes_laptop | _omer, looks like you've got a buggered up makefile |
| 11:20.42 | *** join/#asterisk saftsack (n=saftsack@p54A7F36D.dip.t-dialin.net) |
| 11:20.44 | saftsack | hi |
| 11:20.47 | dlynes_laptop | _omer, however, it cleaned just fine |
| 11:21.01 | saftsack | do i have to set opermode with my tdm400p if i use it as a host for my phones? |
| 11:21.28 | _omer | make install |
| 11:21.29 | _omer | make: cc: Command not found |
| 11:21.29 | _omer | make: *** [gendigits.o] Error 127 |
| 11:21.58 | hads|home | saftsack: It would probably pay because the phones you buy will be setup for your countrys impedance. |
| 11:21.59 | dlynes_laptop | _omer, you don't have the GNU development tools installed |
| 11:22.48 | nailbags | dlynes_laptop: hads|home: ok, all sorted. using _., just had to do some includes to sort out the order |
| 11:22.50 | nettie | Hey guys, anyone running trunk with chan_sip jb enable please? |
| 11:22.55 | _omer | any name? |
| 11:23.05 | saftsack | hads, ok can this be the reason why all telephones are working but the line is hanging up sometimes? |
| 11:23.11 | hads|home | nailbags: You should use _X. |
| 11:23.21 | dlynes_laptop | _omer, it helps to know what distro and version of that distro you're using |
| 11:23.31 | saftsack | for example i can't dial out with my telephone because the line is dead. then i have to call the line from the other side and then all works fine |
| 11:23.51 | nailbags | hads|home: ok, thats working too |
| 11:24.02 | nailbags | hads|home: dlynes_laptop: thanks heaps guys |
| 11:24.09 | dlynes_laptop | no problem |
| 11:24.12 | hads|home | nailbags: np |
| 11:24.35 | hads|home | saftsack: So you have a TDM400 with ? modules |
| 11:24.49 | dlynes_laptop | nailbags, one thing that'll help you in the future |
| 11:24.59 | dlynes_laptop | nailbags, try to learn how to read your sip debug logs |
| 11:25.00 | _omer | whats going on with my yum ..... http://pastebin.ca/94619 |
| 11:25.09 | *** part/#asterisk BugKham (i=BugKham@202.8.86.164) |
| 11:25.09 | dlynes_laptop | nailbags, i.e. sip debug peer exten100 |
| 11:25.11 | carl0s- | OK I'm getting this far. Through the SIP debug, I can see that on an incoming GSM call, we get as far as "SIP/2.0 407 Proxy Authentication Required" when the asterisk box tries to accept an INVITE. Prior to that, there are some REGISTER request things which fail due to "404 not Found" and "403 Forbidden". |
| 11:26.12 | dlynes_laptop | carl0s-, you might have a username/password that didn't match |
| 11:26.53 | saftsack | hads|home, yes |
| 11:27.02 | saftsack | i have 4 green modules |
| 11:27.14 | carl0s- | dlynes_laptop: I can see that the incoming call goes through to asterisk as being to: "useip@192.168.253.15" (<-- my asterisk box). Is that acceptable (the 'useip') bit? |
| 11:27.26 | hads|home | saftsack: And what are you using to dial out on? |
| 11:28.10 | saftsack | TELCO -> ISDN with Bristuff -> * -> TDM400P -> Telephones |
| 11:29.00 | dlynes_laptop | carl0s-, i can't remember, but i think the 'useip' might be the username |
| 11:29.20 | hads|home | saftsack: I don't know anything about BRIstuff sorry. |
| 11:29.33 | carl0s- | ok. what does srvlookup=yes mean in sip.conf? can't find anything on that from google. |
| 11:29.43 | saftsack | yes but bristuff isnt an issue because internal i have other telephones too for example SIP phones and with them i can dialout everytime |
| 11:29.55 | saftsack | so it IS an issue with my tdm400p card and with nothing else |
| 11:30.02 | dlynes_laptop | carl0s-, it means determine the sip server and sip registrar ip addresses from the dns records |
| 11:30.13 | hads|home | saftsack: Interesting, what does the console say when you try and dial out and it doesn't work. |
| 11:30.37 | saftsack | starting simple switch and then hangup |
| 11:30.55 | saftsack | something like that. im not there now but i can get the infos later |
| 11:31.10 | dlynes_laptop | saftsack, you can't plug BRI into TDM400P...it only handles analog, not BRI |
| 11:31.14 | jalsot | zoa: I read the whole topic, but didn't find if an internal buffering in asterisk would be a good solution or not [what I mean is to collect frames for e.g. 1 second, that will save 49 small writes [for 20ms] to the disk] |
| 11:31.16 | saftsack | but the problem is, that theres not a scheme for the problem. it comes sometimes and it goes sometimes |
| 11:31.30 | hads|home | saftsack: Come back in when you have access to the console. |
| 11:31.47 | saftsack | dlynes_laptop, i ve never had a BRI telephone plugged into my tdm400p |
| 11:32.00 | saftsack | i have two cards. one tdm400p and one bn4s0 |
| 11:32.22 | dlynes_laptop | saftsack, looking above: TELCO -> ISDN with Bristuff -> * -> TDM400P -> Telephones |
| 11:33.11 | hads|home | saftsack: I don't think you'll get very far debugging without having access to the console. |
| 11:33.24 | dlynes_laptop | oh nvm |
| 11:33.31 | dlynes_laptop | i didn't see the asterisk in the middle there |
| 11:33.38 | saftsack | TELCO -> ISDN with Bristuff(bn4s0) -> * (-> TDM400P -> analog Telephones) AND (bn4s0 -> BRI Phones) AND (SIP-Phones) |
| 11:33.38 | dlynes_laptop | obviously i need to get some sleep :) |
| 11:33.51 | saftsack | * = asterisk :-P |
| 11:33.57 | saftsack | you need some sleep *G* |
| 11:34.11 | hads|home | It's getting late for dlynes_laptop :) |
| 11:34.17 | dlynes_laptop | saftsack, kinda difficult when i can't sleep in my house |
| 11:34.23 | saftsack | hads|home, yes thats true but i thought that there are some known issues |
| 11:34.30 | saftsack | dlynes_laptop, why that? |
| 11:34.37 | dlynes_laptop | saftsack, broken sprinkler system |
| 11:34.37 | saftsack | is it broken? |
| 11:34.42 | saftsack | oh :( |
| 11:34.47 | dlynes_laptop | pretty stupid, too |
| 11:34.52 | dlynes_laptop | the house is only 4yrs old |
| 11:34.55 | saftsack | think so |
| 11:35.01 | hads|home | dlynes_laptop: At least there's a bright side i.e not everything got trashed. |
| 11:35.06 | saftsack | yes but this must be an expensive house ;) |
| 11:35.14 | saftsack | i mean not every house has a sprinkler system *G* |
| 11:35.18 | dlynes_laptop | hads, well, the other bright side |
| 11:35.34 | dlynes_laptop | hads|home, it's summer, getting wet from the sprinkler helped cool me down |
| 11:35.38 | dlynes_laptop | :0 |
| 11:35.45 | dlynes_laptop | saftsack, every new house does |
| 11:35.51 | saftsack | oh, ok |
| 11:35.58 | dlynes_laptop | saftsack, building code requires it |
| 11:36.09 | saftsack | ok didnt know that because i dont live iun the susa |
| 11:36.12 | saftsack | USA |
| 11:36.15 | dlynes_laptop | neither do i |
| 11:36.22 | hads|home | heh, it's probably about 5 degrees C over here :) |
| 11:36.39 | saftsack | hads|home, here it is 33° Oo |
| 11:36.43 | dlynes_laptop | It's 4:30am here, and it's still about 70F |
| 11:36.46 | saftsack | celsius |
| 11:36.49 | dlynes_laptop | freaking crazy |
| 11:37.04 | dlynes_laptop | and the americans think it snows all year up here |
| 11:37.14 | saftsack | 32% humidity @ 33°C celsius |
| 11:37.21 | saftsack | thats a deathbringer :( |
| 11:37.22 | dlynes_laptop | nice |
| 11:37.30 | dlynes_laptop | 32% humidity is pretty dry, dood |
| 11:37.35 | hads|home | Yeah |
| 11:37.43 | dlynes_laptop | how is that a deathbringer? |
| 11:37.52 | saftsack | what? ^^ it isnt really dry |
| 11:38.01 | dlynes_laptop | 40 degrees centigrade at 120% humidity is a deathbringer |
| 11:38.13 | dlynes_laptop | 32% humidity is quite dry |
| 11:38.18 | dlynes_laptop | not super dry |
| 11:38.25 | dlynes_laptop | but it's a far cry from being humid |
| 11:38.33 | saftsack | hehe |
| 11:38.47 | hads|home | 120% is definitly humid ;) |
| 11:39.00 | dlynes_laptop | during the summer |
| 11:39.14 | dlynes_laptop | they get 90-120% humidity in southern ontario regularly |
| 11:39.40 | dlynes_laptop | 120% is when the water is just rolling off your back, constantly |
| 11:39.58 | saftsack | oh :( thats not good |
| 11:40.03 | dlynes_laptop | i don't miss the weather there, one iota |
| 11:40.14 | dlynes_laptop | you couldn't pay me enough to move back there |
| 11:40.22 | dlynes_laptop | even China's better weather |
| 11:40.24 | hads|home | Yeah, that's not so good. We don't get that sort of humidity over in my end of the world. |
| 11:40.40 | dlynes_laptop | HK and the phillipines do, don't they? |
| 11:41.06 | dlynes_laptop | i thought they were even worse |
| 11:41.13 | hads|home | For sure, but .nz is quite a bit south. |
| 11:41.25 | dlynes_laptop | yeah, but nobody cares about nz |
| 11:41.26 | dlynes_laptop | :) |
| 11:41.34 | hads|home | Oi! :) |
| 11:41.41 | knarfly | dlynes_laptop: ever heard of an ISP blocking sip or rtp ports |
| 11:41.45 | dlynes_laptop | spoken like a true aussie :) |
| 11:41.51 | dlynes_laptop | knarfly, yep, all the time |
| 11:41.52 | hads|home | heh |
| 11:42.21 | dlynes_laptop | hads|home, just havin' fun with ya |
| 11:42.31 | knarfly | dlynes_laptop: no fooling...I tried to set a friend up with X-Lite last night and nothing would work |
| 11:42.35 | dlynes_laptop | hads|home, i know the kiwis and aussies are rivals :) |
| 11:42.57 | knarfly | dlynes_laptop: my firewall did show him even getting to me |
| 11:42.59 | hads|home | Yep, it's pretty funny really. I lived over there for a few years. |
| 11:43.13 | dlynes_laptop | knarfly, did or didn't? |
| 11:43.33 | knarfly | dlynes_laptop: I can only assume blockage somewhere because I made this work from my remote office |
| 11:44.00 | knarfly | dlynes_laptop: did not...no packets were logged denied |
| 11:44.15 | dlynes_laptop | knarfly, he couldn't get through to you? |
| 11:44.26 | dlynes_laptop | knarfly, and you're suspecting his isp is blocking the packets? |
| 11:44.55 | knarfly | dlynes_laptop: nope...and we tried setting him up with FWDNET...he could call me and I could hear him but he could not hear me |
| 11:45.14 | dlynes_laptop | knarfly, that sounds more like an rtp issue |
| 11:45.30 | dlynes_laptop | knarfly, not an isp issue |
| 11:45.40 | knarfly | dlynes_laptop: but I have this same X-Lite setup working from other places |
| 11:45.55 | hads|home | What NAT is involved? |
| 11:45.57 | dlynes_laptop | knarfly, i think you should look more at what router he's using |
| 11:46.08 | dlynes_laptop | knarfly, i think you'll find that's what's different |
| 11:46.13 | dlynes_laptop | knarfly, not the isp |
| 11:46.54 | dlynes_laptop | knarfly, make sure he's not doing any port forwarding for SIP traffic, too |
| 11:46.54 | knarfly | dlynes_laptop: hope you're right...he has only a cable modem...no router per se as he only uses one Windows XP box |
| 11:47.12 | hads|home | Righto, bed for me. Later guys. |
| 11:47.27 | dlynes_laptop | knarfly, windows xp firewalling is probably enabled, or norton internet security is fucking with things |
| 11:47.35 | knarfly | dlynes_laptop: he can ftp and http into my LAN |
| 11:47.46 | dlynes_laptop | knarfly, that doesn't mean squat |
| 11:47.53 | dlynes_laptop | knarfly, he's initiating the dataflow |
| 11:48.00 | knarfly | dlynes_laptop: we turned off his McAfee Firewall but I guess WinXP firewall was still doing something |
| 11:48.02 | dlynes_laptop | knarfly, not you |
| 11:48.20 | *** join/#asterisk tuxd00d (n=tuxinato@netblock-68-183-136-97.dslextreme.com) |
| 11:48.55 | knarfly | dlynes_laptop: I follow...I did a test call with another FWDNET user last night...worked fine so it has to be in his setup |
| 11:48.57 | dlynes_laptop | knarfly, anyways...the problem you're experiencing is almost definitely an rtp sisue |
| 11:49.28 | dlynes_laptop | knarfly, if it was an isp issue, you wouldn't even be able to make the call, much less get one way audio |
| 11:49.41 | knarfly | dlynes_laptop: this guy has Vonage working so we're going to try that |
| 11:49.58 | dlynes_laptop | knarfly, if vonage is working, it's definitely not his isp |
| 11:50.08 | dlynes_laptop | knarfly, it's definitely knarfly then |
| 11:50.13 | knarfly | dlynes_laptop: he's very computer ILLITERATE and has little patience so debugging is not an option |
| 11:50.44 | EyeCue | *curseS* |
| 11:50.55 | EyeCue | turkish tea anyone? before i explode. |
| 11:50.58 | dlynes_laptop | knarfly, I have a test facility where I simulate the client's environment, get all of their phones set up and tested before bringing them to the client's site |
| 11:50.59 | knarfly | dlynes_laptop: that's what I think too. Does Vonage use sip...i thought there's was proprietary |
| 11:51.11 | dlynes_laptop | knarfly, vonage uses sip |
| 11:51.21 | knarfly | EyeCue: Irish Tea with Vanilla and lemon |
| 11:51.26 | dlynes_laptop | knarfly, they just lock their ata's so you can't change the sip registrar/proxy |
| 11:51.58 | carl0s- | Sending to 192.168.253.3 : 5060 (NAT) |
| 11:51.58 | carl0s- | Transmitting (NAT) to 192.168.253.3:5060: |
| 11:52.25 | carl0s- | why does it say NAT? I've got "nat=no" in the sip.conf , and the device has the LAN option of "Bridged" or "NAT". So I have set it to Bridged. |
| 11:52.28 | knarfly | dlynes_laptop: I think that narrows it down to his WinXP machine then |
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| 11:52.45 | dlynes_laptop | knarfly, or to that router he's forgotten to mention |
| 11:53.17 | knarfly | dlynes_laptop: if you want to call his cable modem a router because that's all he has. |
| 11:53.30 | dlynes_laptop | heh |
| 11:54.06 | knarfly | dlynes_laptop: his Vonage goes through this cable modem so that's probably not it...u think? |
| 11:54.39 | dlynes_laptop | knarfly, ummm...vonage goes into the cable modem, and his computer goes into vonage, right? |
| 11:55.22 | knarfly | dlynes_laptop: no, Vonage gave him some kind of ATA that he plugs into the cable modem and then plugs his analog phone into this device |
| 11:55.38 | dlynes_laptop | knarfly, so he's using a switch then? |
| 11:57.32 | knarfly | dlynes_laptop: I'd have to ask him more...but he really has no clue...just plugs something in and hopes it works...his one hell of a carpenter though. But with computers he's not into knowing anything other than turn it on and click a mouse. |
| 11:58.44 | knarfly | dlynes_laptop: we're going to try direct dialing between Vonage and FWDNET later tonight...if that works it gets us almost there |
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| 11:59.07 | knarfly | dlynes_laptop: I really wanted to get him setup on my * box for some tests. |
| 11:59.12 | dlynes_laptop | knarfly, anyways...there's gotta be another piece of hardware there that he's not telling you about |
| 11:59.20 | dlynes_laptop | it's going to be either a router or a switch |
| 11:59.42 | dlynes_laptop | otherwise he's not able to have his voip phone working at the same time as when he's browsing the internet |
| 12:00.05 | dlynes_laptop | knarfly, or his ata from vonage is both an ata and a router |
| 12:00.12 | dlynes_laptop | and the computer's plugged into that |
| 12:00.21 | knarfly | dlynes_laptop: You's think but I've seen this guys place. He has a cable modem plugged into the wall and that's plugged directly into his WinXP. He doesn't have a LAN |
| 12:00.52 | dlynes_laptop | so he's got two cable modems then? |
| 12:00.54 | knarfly | dlynes_laptop: I should mention we tested this without the Vonage equipment... |
| 12:01.03 | dlynes_laptop | cable modems only have one cat 5e jack |
| 12:01.45 | knarfly | dlynes_laptop: right and he plugs his COAX connection into the wall port and the CAT5 into his WinXP |
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| 12:02.12 | carl0s- | arrgh. please help. "NOTICE[31924] chan_sip.c: Registration from '<sip:103@192.168.253.15>' failed for '192.168.253.3' - Username/auth name mismatch". What does this mean? It couldn't be any simpler, I have set "username=103", "secret=103" and the same on the gsm box. |
| 12:02.37 | knarfly | dlynes_laptop: just like mine here but CAT5 plugs into a dual homed FreeBSD server that's connected to a switch. This guy doesn't have a LAN |
| 12:02.56 | EyeCue | natd on the bsd box? |
| 12:03.33 | knarfly | wow |
| 12:03.33 | carl0s- | oh well. I'm still here :) |
| 12:03.38 | EyeCue | yay |
| 12:03.50 | knarfly | EyeCue: yes Natd on the FreeBSD box |
| 12:04.48 | EyeCue | all this asterisk just to find a replacement for ventrilo. |
| 12:04.49 | EyeCue | :| |
| 12:05.00 | knarfly | dlynes_laptop: thanks for the advice...you're always a great help...gotta run and start my real job now. |
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| 12:10.03 | dlynes_laptop | argh |
| 12:10.23 | dlynes_laptop | plugged into a dual honed freebsd box which is your NAT!!!!!!!! NOT WINXP WITH MACAFEE FIREWALL |
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| 12:20.58 | nailbags | how do i play a .gsm file? |
| 12:21.03 | nailbags | (not in asterisk) in linux |
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| 12:24.02 | RoyK | nailbags: sox |
| 12:31.18 | nailbags | RoyK: can it play directly or only convert? |
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| 12:35.04 | RoyK | nailbags: 'play' command should work |
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| 12:42.12 | carl0s- | I'm now getting "NOTICE[27132] chan_sip.c: Failed to authenticate user "07766087677" <sip:1001@192.168.253.3:5060>;tag=7d16d9d1". |
| 12:42.24 | carl0s- | It looks like the VoIP GSM gateway is sending the incoming Caller-ID as the username. |
| 12:44.05 | jalsot | does anybody use the MONITOR_CONSTANT_DELAY 'feature' in channel.c? |
| 12:44.38 | carl0s- | ooh. I just had it work. sort of. |
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| 12:46.06 | bionoid | Hello everyone, after some fiddling I got my Tiger3XX operational on one channel (as far as I can tell). Is there a quickie way to have asterisk dial a specified number from CLI to test whether or not it actually has a line? |
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| 13:02.25 | [TK]D-Fender | Katty : Mew. |
| 13:03.19 | nortex | What actually causes "WARNING[3473] chan_zap.c: Ring requested on channel 0/1 already in use on span 1. Hanging up owner." I got 15-20 in a row last night in my logs. |
| 13:05.02 | *** join/#asterisk nextime (n=nextime@213-140-6-103.ip.fastwebnet.it) |
| 13:05.11 | Katty | [TK]D-Fender: mew. |
| 13:05.50 | carl0s- | Hmm. "Peer '103' is trying to register, but not configured as host=dynamic". I don't want it configured as host-dynamic do I? I have set host=192.168.253.3 |
| 13:06.22 | RoyK | carl0s-: then the client has no need of registering |
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| 13:06.46 | nortex | That span is a PRI, so I don't know why there would be attempted calls on channels that are in use. |
| 13:07.17 | carl0s- | RoyK: hmm. OK. but I don't see an option in the client (GSM VoIP gateway) to tell it not to register. Should I remove the "SIP Proxy: 192.168.253.15" entry in the client? Is that what causes it to try to register? |
| 13:08.37 | carl0s- | roy: here is the only SIP configuration I have on the GSM box: http://www2.css-networks.com/cfg.JPG |
| 13:10.28 | carl0s- | anyone? |
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| 13:18.50 | SynUK | Anyone know where i can get the SIP firmware for a Cisco 7970 IP Phone ? |
| 13:23.35 | zoa | SynUK: cisco.com |
| 13:24.58 | carl0s- | what's the name of that ncurses-based packet capture/analysis tool for Linux? |
| 13:25.55 | carl0s- | or is there an easier way of debugging the Username/auth name mismatch problem I'm having? |
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| 13:36.16 | Katty | ariel_: (= |
| 13:36.47 | ariel_ | morning Katty |
| 13:37.00 | ariel_ | how have you been? |
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| 13:37.31 | Pj_ | Heya people ! |
| 13:39.02 | carl0s- | this is bloody annoying. First time I start the box up, I can call in on GSM, although I still get a second dialtone and have to press some shit. Then after that it doesn't work and I get the "auth failed" shit. |
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| 13:47.14 | *** join/#asterisk visiperl (n=visiperl@h185.29.29.71.ip.alltel.net) |
| 13:48.02 | visiperl | Hello all, I've got a question regarding the TDM2400 series. Is this the correct place to ask? |
| 13:48.20 | carl0s- | I guess so, but everyone's asleep or at work. |
| 13:48.42 | visiperl | Thank you I will try now.... is there a better time? |
| 13:48.43 | ariel_ | visiperl, post the question someone might be up. |
| 13:48.49 | visiperl | ok great. :) |
| 13:49.01 | ariel_ | there is never an wrong time or good time. |
| 13:49.59 | visiperl | I've got a tdm2400 with 2 (4) port modules installed. When a person is already in a call speaking with someone outside the office an another call comes in, the ringing comes through on the call already in progress an blanks out the sound. |
| 13:50.10 | visiperl | anybody heard of this problem? |
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| 13:50.44 | *** mode/#asterisk [+o anthm] by ChanServ |
| 13:50.47 | visiperl | I'm leaning toward either a callwaiting issue or possibly voltage on the card issue. |
| 13:51.39 | ariel_ | visiperl, check to make sure you don't have the wires punched down incorrectly. |
| 13:52.03 | visiperl | ? as in R/T backwards? |
| 13:53.19 | ariel_ | visiperl, if your lines seem sound like you said you should make sure your wires are not crossed. Are they punched down on a AMP 66 block? |
| 13:53.33 | visiperl | yes AMP66 |
| 13:53.53 | *** join/#asterisk tbright (n=tbright@h-68-165-94-218.nycmny83.covad.net) |
| 13:53.56 | tbright | hello all |
| 13:54.10 | visiperl | hi tbright |
| 13:55.23 | tbright | .. i have my pbx set up at home, it is behind a nat. I have setup an ip phone at work. My phone registers, and i can make calls to my extensions, information regarding those calls is shown on my console, but i can hear no sound on my phone. |
| 13:56.20 | tbright | I have dialed all these extensions from within my home network .. and they work fine. i have setup some nat stuff in my sip.conf .. but i dont really know what im doing. .any help is appreciated |
| 13:56.25 | trelane` | visiperl, sounds like a short in one of the modules? |
| 13:56.54 | trelane` | visiperl, check your 66 block, if that works check your wctdm24xxp |
| 13:56.59 | [TK]D-Fender | tbright : pastebin your sip.conf (edit out passwords). |
| 13:57.01 | [TK]D-Fender | ~pb |
| 13:57.02 | jbot | methinks pb is a place to paste your stuff without flooding the channel - try http://pastebin.com/, or http://pastebin.ca/, or http://channels.debian.net/paste, or for #oe use http://oe.pastebin.com/, or http://bzflag.pastebin.ca/, or for images use http://imageshack.us/ |
| 13:57.11 | trelane` | visiperl, that is also a HEAVY card, ensure it's not contacting the case or any metal within the case |
| 13:58.21 | visiperl | thanks ariel_ and trelane I will go check / test those items now. Really appreciate the suggestions. |
| 13:59.08 | tbright | http://pastebin.ca/94769 |
| 13:59.11 | tbright | thats my sip.conf |
| 13:59.29 | tbright | .. thinkbright gxp-2000 is the line that is in my office |
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| 14:00.25 | ariel_ | trelane, most one way sound issues are due to nat. and your firewall. Make sure you have your extenip= and localnet= setting in the sip.conf |
| 14:00.41 | ariel_ | and make sure you have the rtp ports (UDP) open on the firewall. |
| 14:01.23 | [TK]D-Fender | tbright : You should be using "externhost" and "externrefresh" for that domain, not "externip". Also if your work phone is behind NAT your phone entries should have NAT=YES as well. Your L"localnet" clause should be in the format of IP/MASK and not in 2 lines. |
| 14:01.34 | *** part/#asterisk visiperl (n=visiperl@h185.29.29.71.ip.alltel.net) |
| 14:01.47 | [TK]D-Fender | tbright : And of course I'm assuing you forwarded the ports for SIP & RTP.....\ |
| 14:02.11 | tbright | [TK]D-Fender: i have fowarded 5060 for sip.. but i didnt do anything for rtp |
| 14:03.04 | tbright | so i have externhost=tamnet.linuxhome.org in general .. what do i use externrefresh for?? |
| 14:03.19 | Dr-Linux|work | tbright, also forward a range 10k to 20k udp |
| 14:04.12 | tbright | I also have nat=yes for [2005] in sip.conf .. i do i need to set nat for all my other extensions too?? |
| 14:04.25 | tbright | sorry .. i really dont know what im doing.. ive just been playing with * for a week now |
| 14:04.36 | nortex | What actually causes "WARNING[3473] chan_zap.c: Ring requested on channel 0/1 already in use on span 1. Hanging up owner." I got 15-20 in a row last night in my logs. |
| 14:04.47 | [TK]D-Fender | tbright : not a bad idea to have NAT=YES globel. your localnet range will let everything "play nice" |
| 14:05.19 | tbright | [TK]D-Fender so wtih nat=yes in [general] then i dont need to set nat for each extension |
| 14:05.56 | [TK]D-Fender | tbright : I believe it inherits like the rest... |
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| 14:07.02 | tbright | hrmm .. i dont know if i can login to my router to foward rtp from here.. is that absolutely needed all the time?? .. i mean.. if i cant do it now .. icant test till i get home and foward rtp? |
| 14:07.28 | tbright | .. and im still not sure where to put or what to set for "externrefresh" |
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| 14:08.21 | nighty_ | anyone using asterisk with freebsd ? |
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| 14:12.19 | tbright | im emerging links so i can try to access my routers config through ssh'ing into my home pbx |
| 14:12.34 | tbright | .. hopefully links can handle that.. we will see |
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| 14:24.09 | carl0s- | hmm :( |
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| 14:24.33 | saftsack | hi |
| 14:24.50 | carl0s- | can anybody see any 'reason why' in this sip debug output? : http://pastebin.ca/94782 |
| 14:25.00 | saftsack | where to set opermode? in modprobe.d/zaptel is written that i dont have to edit this file |
| 14:25.13 | carl0s- | saftsack: /etc/modules.conf is where I did it. |
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| 14:25.29 | carl0s- | sorry, modprobe.conf |
| 14:25.29 | saftsack | carl0s-, can you show me a line? |
| 14:25.38 | carl0s- | options wctdm opermode=UK |
| 14:25.49 | saftsack | thanks :) |
| 14:25.52 | file | carl0s-: Looking for useip in from-trunk, do you have an extension useip in the context from-trunk? |
| 14:25.55 | carl0s- | no prob :) |
| 14:26.26 | carl0s- | file: no. I have been asking about 'useip'. this seems to be the sip-name that the VoIP GSM gives itself, I think |
| 14:26.46 | file | sure. |
| 14:27.12 | carl0s- | What do you think I should do? |
| 14:27.49 | file | well what do you want to do? because it's the SIP device sending that, not Asterisk so you should probably look at the manual if you want to change... |
| 14:28.20 | carl0s- | there isn't anything in the device really other than: http://www2.css-networks.com/cfg.JPG |
| 14:28.36 | nortex | What actually causes "WARNING[3473] chan_zap.c: Ring requested on channel 0/1 already in use on span 1. Hanging up owner." I got 15-20 in a row last night in my logs. |
| 14:28.36 | file | okay. |
| 14:28.40 | Toerkeium | guys, I am going to try installing asterisk on a VPS, does anyone know which devnodes are needed for zaptel ? |
| 14:28.53 | carl0s- | however, I suspect it sends "useip" because in the GSM -> LAN routing table I have put the IP address of the asterisk box. I'll see if it's possible to put other things in there. |
| 14:29.23 | carl0s- | file: what would a gateway device normally send in the "to:" line? if it's meant to be working as a trunk? |
| 14:29.42 | carl0s- | perhaps I could put "useip" as the CID or DID for the trunk line in my trunk configuration? |
| 14:30.52 | file | it's up to the SIP device... |
| 14:31.13 | carl0s- | but there would always be something there? |
| 14:31.38 | file | usually. |
| 14:31.49 | carl0s- | for example, you can't just have a device pipe the sip stuff right to the IP address.. has to be 'some-extension'@ipaddr. |
| 14:31.51 | carl0s- | hmm |
| 14:32.06 | file | no, you could... |
| 14:32.19 | file | it would be a request URI without user portion |
| 14:32.52 | carl0s- | right. I tried "fromuser=useip" in the trunk config but that hasn't helped. |
| 14:32.58 | *** join/#asterisk UlbabraB (n=UlbabraB@host241-43-static.72-81-b.business.telecomitalia.it) |
| 14:33.06 | file | in Asterisk? |
| 14:33.34 | carl0s- | yes |
| 14:33.44 | *** join/#asterisk eKo1 (n=eKo1@190.4.7.90) |
| 14:33.45 | file | which problem are you having? |
| 14:34.22 | carl0s- | well when I call the gsm gateway from my mobile number (07766087677), the call is just hung up on right away. That's what the pastebin sip debug was relating to. |
| 14:34.45 | file | I told you... the extension does not exist in the context, so Asterisk can not route the call and sends back a 404 Not Found |
| 14:34.53 | nighty_ | anyone knows why chan_sccp would not compile on FreeBSD against the latest port of asterisk 1.2.9.1 ? |
| 14:35.08 | nighty_ | http://pastebin.ca/94781 |
| 14:35.16 | file | the GSM gateway interpetes that and probably drops the call or rejects it |
| 14:35.44 | carl0s- | file: OK. I'm just confused where to go from here. I shouldn't have to create a specific extension just to use this gateway should I? |
| 14:35.53 | saftsack | howto see all flags of opermode? |
| 14:36.14 | file | carl0s-: how else is it going to know how to route things? it's not sending it to any number, it's sending it to the extension "useip" |
| 14:37.07 | carl0s- | hmm. I'm new to all this. be gentle with me :) |
| 14:37.27 | file | okay, how is Asterisk supposed to know how you want to handle calls from the GSM gateway? |
| 14:37.38 | carl0s- | Ideally then, I'd want the gateway to not pass the user part at all. |
| 14:37.53 | carl0s- | well, I was wanting it to be matched with the "any cid / any did" incoming rule that I have. |
| 14:38.00 | file | Asterisk will just send calls to the 's' extension then |
| 14:38.04 | carl0s- | and then ring all my extensions |
| 14:38.27 | *** join/#asterisk c4t3l (n=c4t3l@69.15.174.114) |
| 14:38.32 | file | see, all you have to do is add a few lines to extensions.conf to have an extension useip and your problem will be over :) |
| 14:39.10 | carl0s- | and I can then just make this "useip" extension route directly to my normal user extensions e.g. 201, 202 (ring all) ? |
| 14:39.20 | file | sure |
| 14:39.33 | carl0s- | OK. I'm going to have a try now :D |
| 14:39.35 | file | it just seems like you're overcomplicating it |
| 14:40.01 | carl0s- | well, I just wanted it to work like my upstream SIP provider, and my tdm400 ZAP trunks do - haven't had to give them any funny extensions :) |
| 14:40.04 | *** join/#asterisk unixgeek (n=unixgeek@216-220-234-197.exploremaine.com) |
| 14:40.13 | *** join/#asterisk EyeCue (n=eyecue@unaffiliated/eyecue) |
| 14:40.24 | file | well you can just make useip a Goto to your regular stuff... |
| 14:40.41 | carl0s- | fair enough |
| 14:40.45 | EyeCue | egads. |
| 14:41.14 | carl0s- | and it's OK to use letters for the extension though? it doesn't have to be a number? |
| 14:41.18 | file | sure |
| 14:41.29 | carl0s- | h'okay! |
| 14:42.09 | *** join/#asterisk TripleFFFF (n=TripleFF@147-102.mc.cite.net) |
| 14:42.15 | TripleFFFF | guys.. got emergency |
| 14:42.32 | TripleFFFF | is there a way to make sure a manager call out is done only once ? it seems to retry retry retry |
| 14:42.33 | System010 | anyone here have experience with an inter-tel system |
| 14:42.55 | file | TripleFFFF: an originate using Manager you mean? |
| 14:44.29 | TripleFFFF | yes |
| 14:44.31 | TripleFFFF | file ;) |
| 14:44.45 | TripleFFFF | <PROTECTED> |
| 14:44.49 | TripleFFFF | added this |
| 14:44.51 | TripleFFFF | but no luck |
| 14:45.25 | file | it uh shouldn't retry... |
| 14:45.51 | *** join/#asterisk System010 (n=jgargano@hide247.cybergnostic.com) |
| 14:46.20 | TripleFFFF | guess its not asnwering so its retrying |
| 14:46.41 | TripleFFFF | i need waittime ? |
| 14:47.17 | file | have you turned up verbosity and debug to see exactly what's going on and made sure that manager is retrying? |
| 14:47.31 | TripleFFFF | yes |
| 14:47.35 | TripleFFFF | i see multiple cdr's |
| 14:47.48 | file | I didn't ask that... |
| 14:48.05 | jbalcomb | HOSTNAME="myamotomusashi" |
| 14:48.18 | file | can you put the verbose/debug from console up on pastebin? |
| 14:48.47 | *** join/#asterisk salviadud (n=ralfalfa@201.153.40.45) |
| 14:49.40 | unixgeek | Are there multiple flavors of PRI D channel protocols? |
| 14:49.54 | [TK]D-Fender | jbalcomb : Book of Five Rings... a good read... |
| 14:50.09 | TripleFFFF | hmm |
| 14:50.10 | TripleFFFF | well |
| 14:50.13 | jbalcomb | [TK]D-Fender: indeed. =) |
| 14:50.45 | jbalcomb | [TK]D-Fender: have you also read, hagakure? |
| 14:50.50 | [TK]D-Fender | jbalcomb : I own a Gorin Iaito (go-rin) which is styled after Musashi's diato. |
| 14:50.50 | *** join/#asterisk jcmoore (n=jcmoore@unaffiliated/tgrman) |
| 14:51.11 | [TK]D-Fender | jbalcomb : I study Katori Shinto currently, not into ninutsu |
| 14:52.05 | jbalcomb | [TK]D-Fender: Very interesting and impressive. |
| 14:52.12 | [TK]D-Fender | jbalcomb : My baby : http://forums.swordforum.com/showthread.php?s=&threadid=67812&highlight=bushi+pics |
| 14:53.29 | [TK]D-Fender | jbalcomb : Thats my Oni Forge Bushi... real nice piece of work.. never took pics of my iaito. |
| 14:53.37 | TripleFFFF | can i send retry ? and waittime ? |
| 14:53.39 | salviadud | that's a nice katana |
| 14:53.47 | [TK]D-Fender | jbalcomb : Should get around to, as well for my new MIDI controller |
| 14:54.07 | TripleFFFF | fputs($handle, "RetryTime: 60\n"); |
| 14:54.12 | salviadud | is anyone here any good at ninja gaiden from NES? |
| 14:54.12 | TripleFFFF | this valid ? |
| 14:54.12 | [TK]D-Fender | salviadud : I'm almost to nervous to wield it :) |
| 14:54.36 | file | TripleFFFF: no, the only valid options are - Channel, Exten, Context, Priority, Timeout, CallerID, Account, Application, Data, Async, and ActionID |
| 14:54.37 | salviadud | [TK]D-Fender, I didn't know you were interested in martial arts |
| 14:55.04 | Assid | hey [TK]D-Fender |
| 14:55.22 | [TK]D-Fender | salviadud : I do a lot of things. Musician (guitar & piano), 9-ball billiards God, VoIP, general tech, HTPC stuff. |
| 14:55.26 | TripleFFFF | timout does what |
| 14:55.27 | TripleFFFF | hmmm |
| 14:55.28 | salviadud | very impressive indeed |
| 14:55.35 | [TK]D-Fender | Play volleyball weekly. |
| 14:55.37 | TripleFFFF | well it seemd that if you just hangup it redials |
| 14:55.50 | file | how long to wait to give up |
| 14:55.50 | [TK]D-Fender | salviadud : I try to keep busy. Keeps me sane following my breakup. |
| 14:55.54 | *** part/#asterisk kmilitzer (n=km@office-gw.westend.com) |
| 14:55.55 | *** join/#asterisk mkrufky (n=mk@68.160.103.77) |
| 14:56.00 | jbalcomb | [TK]D-Fender: that is quite a nice piece. |
| 14:56.08 | salviadud | you'll get another lady... don't worry about it |
| 14:56.23 | [TK]D-Fender | jbalcomb : I'll take some shots of my iaito over the weekend. |
| 14:56.39 | [TK]D-Fender | salviadud : Not worried, just a little empty. |
| 14:57.04 | eKo1 | [TK]D-Fender: you practice kendo or some kenjustu? |
| 14:57.15 | jbalcomb | [TK]D-Fender: bushido to shinto to kendo to aikido ga omoshiroii |
| 14:57.34 | mkrufky | a co-worker just asked me: is it possible for asterisk to answer data calls ( like a modem would ) and assign an ip address to the caller? |
| 14:58.09 | eKo1 | mkrufky: you need a modem server |
| 14:58.15 | eKo1 | not a pbx |
| 14:58.27 | mkrufky | so the answer is, NO . correct? |
| 14:58.41 | eKo1 | correct |
| 14:58.50 | file | trick question, if you have zaptel hardware... PRI... you can use zapras |
| 14:59.11 | mkrufky | i have heard that there is an application that runs within asterisk that can answer fax calls and dump a tiff to the disk, so i think it would be a reasonable question |
| 14:59.21 | mkrufky | anyway, thanx :-) |
| 14:59.29 | file | faxing and modem usage are two different things |
| 14:59.40 | mkrufky | understood |
| 14:59.41 | [TK]D-Fender | eKo1 : Katori covers kenjutsu & iaido, as well as bow, naginata, and two-handed (diasho). |
| 14:59.45 | eKo1 | mkrufky: faxing can be done as you've just described. |
| 14:59.51 | [TK]D-Fender | eKo1 : Kendo is a SPORT. |
| 15:00.23 | eKo1 | what is iaido? |
| 15:00.29 | carl0s- | ooh. we have progress. I'm now getting "SIP/2.0 603 Declined". |
| 15:00.56 | mkrufky | slightly related..... if the right application were available, do you think that the digium zaptel hardware would be able to answer data calls? |
| 15:00.57 | [TK]D-Fender | jbalcomb : Aikido has a natural extension in aikibudo which is a compatible art. We have one new student who is relearning things the Katori way :) |
| 15:01.05 | file | carl0s-: sip debug and dialplan logic please |
| 15:01.30 | [TK]D-Fender | eKo1 : The art of drawing the sword for an attck and following through resheathing. |
| 15:01.39 | [TK]D-Fender | eKo1 : Go google it. |
| 15:01.47 | EyeCue | im so over this. |
| 15:01.51 | [TK]D-Fender | jbalcomb : my other new baby : http://www.m-audio.ca/products/en_ca/KeystationPro88-main.html |
| 15:01.55 | eKo1 | [TK]D-Fender: ah, ok |
| 15:02.06 | [TK]D-Fender | jbalcomb : Gonna take some pics of my studio once I mount my guitars up. |
| 15:02.24 | [TK]D-Fender | eKo1 : Heck, check out google vids on it,. |
| 15:02.38 | jbalcomb | [TK]D-Fender you seem like a busy man |
| 15:02.55 | EyeCue | question, why is there a context=default in sip.conf ? |
| 15:03.02 | file | carl0s-: there's an application called ZapRAS that will do it on some zaptel channels... what types I don't know... otherwise you'd probably have to loop it through a modem/modems, or emulate a modem... I suppose |
| 15:03.17 | jbalcomb | EyeCue: because there is a context in extensions.conf named [default] |
| 15:03.39 | EyeCue | and what is the relationship of general(sip) to default(ext) ? |
| 15:03.40 | [TK]D-Fender | jbalcomb : Like I said... keeps me sane (and now that I'm armed, even better ;)) |
| 15:03.55 | carl0s- | file: you got the wrong person there.. i'm stuggling with gsm gateways, not modems :D I'll giv eyou my dialplan shortly |
| 15:04.00 | jbalcomb | [TK]D-Fender: yeah, being armed keeps a lot of people sane... |
| 15:04.02 | file | oh, right |
| 15:04.08 | file | mkrufky: read above |
| 15:04.28 | [TK]D-Fender | jbalcomb : Referring to my being sane as a good thing NOW that I'm armed ;) |
| 15:04.44 | file | [TK]D-Fender: pfft... you sane... right |
| 15:05.00 | [TK]D-Fender | jbalcomb : Otherwise it'd be like hearing on the news of a cubicle-to-cubicle killing spree here... |
| 15:05.01 | mkrufky | file: awesome... i will google zapras |
| 15:05.02 | eKo1 | hehehe |
| 15:05.04 | mkrufky | thanks a lot |
| 15:06.00 | [TK]D-Fender | Funny thing is that MIDI controller I bought is one of the nicest things I've done for myself, yet strangely one of the CHEAPEST. It only set me back $375 CAD. |
| 15:06.19 | tzanger | [TK]D-Fender: I want to get back into midi |
| 15:06.25 | jbalcomb | [TK]D-Fender ah, i understand better now,. |
| 15:06.46 | [TK]D-Fender | I'm looking at this : http://www.m-audio.ca/products/en_ca/KeystationPro88-main.html next for a more synth feel. |
| 15:06.52 | EyeCue | uh, question, for a sip client outside of my natted network (with my asterisk server behind the nat) |
| 15:06.54 | [TK]D-Fender | Quoted at $300 CAD |
| 15:06.58 | EyeCue | i do need to port forward 5060 right ? |
| 15:07.20 | EyeCue | to the asterisk box. |
| 15:07.21 | jbalcomb | [TK]D-Fender i took myself to japan for three weeks instead.. ;) |
| 15:07.29 | [TK]D-Fender | EyeCue : that and the ports used by RTP, as well as a pile of settings in sip.conf |
| 15:07.35 | EyeCue | :| |
| 15:07.40 | EyeCue | ports used by rtp ? |
| 15:07.41 | tzanger | I want to go get a "dumb" keyboard |
| 15:07.57 | [TK]D-Fender | jbalcomb : instead of what? |
| 15:07.57 | file | EyeCue: audio stream... |
| 15:07.57 | jbalcomb | [TK]D-Fender i do hope all the memories are worth not having anything to /show/ for it |
| 15:07.57 | EyeCue | yeh i know what it is |
| 15:07.57 | [TK]D-Fender | tzanger : look at those 2... they ROCK. |
| 15:07.57 | EyeCue | what ports |
| 15:07.59 | file | EyeCue: you'll also need localnet and either externip or externhost set |
| 15:08.04 | tzanger | [TK]D-Fender: those aren't dumb |
| 15:08.04 | EyeCue | cant asterisk take the initial connection and do the rest? |
| 15:08.05 | file | EyeCue: default is 10000 to 20000 for RTP |
| 15:08.11 | EyeCue | yeh theyre set in rtp |
| 15:08.16 | saftsack | is it possible to install a nfs server on a asterisk server without getting robbed much irqs? |
| 15:08.16 | EyeCue | .conf |
| 15:08.17 | [TK]D-Fender | tzanger : yes they are.. they are bare MIDI controllers. NOTHING in them./ |
| 15:08.17 | jbalcomb | [TK]D-Fender: i have a dresser and a laptop |
| 15:08.22 | TripleFFFF | hey got a fast one then.. i know why it redialed.. its coz number not exists.. |
| 15:08.26 | tzanger | ahh |
| 15:08.28 | TripleFFFF | anyway to know about that ? |
| 15:08.40 | [TK]D-Fender | tzanger : I'm using my Audigy with sSoundFonts as a soft-synth. |
| 15:08.48 | tzanger | right on |
| 15:09.00 | [TK]D-Fender | tzanger : the 88-key was only $375 CAD, and the 61 @ $300 |
| 15:09.05 | file | EyeCue: doesn't quite work like that when going through NAT... when the packet goes out for RTP it'll probably have a different source port, but the end device probably won't send back to that... it'll send back to the negotiated port - which if not forwarded, will just discard the packet... and you'll get one way audio |
| 15:09.16 | System010 | inter-tel axxess, anyone? |
| 15:09.18 | EyeCue | ok |
| 15:09.21 | EyeCue | so let me rephrase |
| 15:09.46 | EyeCue | what is the simplest way to allow an outside internet sip client to talk to me on my workstation, that is sipp'd into asterisk |
| 15:09.46 | System010 | need to connect it to asterisk, mgcp or sip. |
| 15:09.49 | EyeCue | :| |
| 15:09.51 | carl0s- | file: just prior to the "603 Declined", I'm now seeing "Looking for useip in from-sip-external". I'm afraid I am using AMP for my configuration so my understanding of contexts etc. isn't great, although I'm getting there. slowly. |
| 15:10.00 | file | eep AMP |
| 15:10.05 | carl0s- | ssh |
| 15:10.07 | carl0s- | ;) |
| 15:10.09 | [TK]D-Fender | tzanger : So cheap its kinda scary. I really like the feel of the KSP-88, and I have an idea how the semi-weighted Axiom 66 will be. |
| 15:10.09 | EyeCue | rofl |
| 15:10.10 | EyeCue | :) |
| 15:10.11 | carl0s- | just for now |
| 15:10.15 | file | EyeCue: simplest way? this is an all or nothing thing with your setup lol |
| 15:10.40 | tzanger | yeah it's amazing how cheap some stuf fis |
| 15:10.50 | EyeCue | sip/iax -> internut -> router (nat) -> asterisk -> sip -> workstation |
| 15:10.51 | file | EyeCue: port forward 5060, 10000-20000 to your Asterisk machine, set localnet to your local network information, set externip to your external IP address, set canreinvite=no and nat=yes for the SIP friend/peer, and voila |
| 15:11.00 | file | then SIP will work externally |
| 15:11.14 | EyeCue | question, will it be easier if he uses an iax client? |
| 15:11.30 | EyeCue | or must i still initiate. |
| 15:11.30 | file | you'll still need to port forward 4569 UDP, but that would be it |
| 15:11.30 | eKo1 | yes it will |
| 15:11.39 | EyeCue | thank fuck for that. |
| 15:11.50 | [TK]D-Fender | tzanger : My goal is probably to get a cheap Compaq MATX desktop system used, bring my audigy over so as to regain my main PC straight up. Another benifit of the M-Audio gear is it's USB powered too.... |
| 15:12.01 | EyeCue | theyre already forwarded |
| 15:12.15 | carl0s- | I'm beginning to realise that Extensions and Trunks are kind of quite similar aren't they? They're both configured in the same way, in the same configuration files, but they have their context set differently. Is that right? |
| 15:12.37 | file | carl0s-: there's no difference internally... it's all the same... |
| 15:13.01 | EyeCue | file, is it ok if 'i' sip to asterisk, since im internal, and add the outside client to iax.conf ? |
| 15:13.09 | file | EyeCue: sure |
| 15:13.25 | carl0s- | file: yeah. That's why I was confused when you were saying to add an 'useip' to my extensions.conf . I was thinking 'an extension?' but instead I named the Trunk (AMP) to 'useip' and that acheives the same but makes sense to me. |
| 15:14.03 | file | carl0s-: oh you thought extension as in phone? |
| 15:14.07 | carl0s- | yes |
| 15:14.10 | file | ah |
| 15:14.22 | carl0s- | then I was thinking about making that phone 'follow-me' to the actual phones.. etc. |
| 15:14.41 | *** join/#asterisk mog (i=ejabberd@68.62.237.103) |
| 15:14.41 | *** mode/#asterisk [+o mog] by ChanServ |
| 15:14.45 | carl0s- | hence thinking it was a bit wierd to have to do that. |
| 15:14.50 | EyeCue | at what verbosity are registrations shown at ? |
| 15:14.52 | trelane` | carl0s-, actually you want all the phones to follow file ;) |
| 15:15.04 | carl0s- | :) |
| 15:15.11 | *** part/#asterisk mitcheloc (n=mitchelo@70-32-189-246.lmdaca.adelphia.net) |
| 15:15.14 | file | my desk phone hardly ever rings... it's beautiful |
| 15:15.42 | file | and with that I go to make lunch |
| 15:15.45 | TripleFFFF | i got s,1,goto(telemarketer-hell) |
| 15:15.58 | TripleFFFF | if they make it trought s,2,dial(mycisco) |
| 15:16.04 | Assid | how doy ou know if its a telemarketer? |
| 15:16.19 | TripleFFFF | i dont.. SET(ASSUMETELEMARKETER=YES) |
| 15:16.25 | TripleFFFF | ;) |
| 15:16.32 | TripleFFFF | guilty till proven innocent |
| 15:18.11 | tzanger | haha |
| 15:19.10 | carl0s- | It's a shame I can't just see everything relating to a particular context. e.g. each context having it's own configuration file. Do I need to grep the whole of /etc/asterisk for "from-sip-external" to see what's in that context? |
| 15:21.45 | unixgeek | Are there multiple flavors of PRI D channel protocols? |
| 15:21.52 | *** join/#asterisk brad6254 (n=brad6254@pool-71-162-32-182.altnpa.east.verizon.net) |
| 15:22.06 | EyeCue | ok werd |
| 15:22.10 | EyeCue | got internal iax -> sip working |
| 15:22.56 | eKo1 | carl0s-: what's wrong with grepping? |
| 15:23.08 | carl0s- | nothin I suppose :). |
| 15:23.16 | eKo1 | unixgeek: that question doesn't make sense |
| 15:23.36 | EyeCue | sip - iax crashed the sip client. |
| 15:23.38 | EyeCue | awesome. |
| 15:23.42 | *** join/#asterisk trbldwine (i=trbldwin@adam.ur.northwestern.edu) |
| 15:24.55 | *** join/#asterisk hfb (n=hfb@pool-71-106-220-165.lsanca.dsl-w.verizon.net) |
| 15:25.27 | file | eep |
| 15:25.46 | *** part/#asterisk trbldwine (i=trbldwin@adam.ur.northwestern.edu) |
| 15:25.59 | EyeCue | gah |
| 15:26.04 | EyeCue | and the aix client. |
| 15:26.11 | EyeCue | im thinking idefisk is causing dramas :D |
| 15:27.10 | *** join/#asterisk wunderkin (n=kev@ip68-226-113-228.ph.ph.cox.net) |
| 15:27.19 | brad6254 | I am using asterisk 1.2.9 and zaptel 1.2.6 and tdm400 with 3 fxo modules I can dial from sip to zap ok, but incoming call from zap to sip are answered, but have no voice. can anyone help? |
| 15:27.29 | eKo1 | aix client? |
| 15:27.38 | EyeCue | iax. |
| 15:27.39 | *** join/#asterisk DarKnesS_WolF (n=wolf@62.114.197.95) |
| 15:27.43 | EyeCue | i keep fscking it up. |
| 15:27.47 | file | brad6254: is the SIP client behind NAT? |
| 15:28.06 | brad6254 | yes, but not going out |
| 15:28.18 | brad6254 | it's all on our lan |
| 15:28.21 | *** join/#asterisk visba (n=dca[lapt@sta-208-139-193-162.rockynet.com) |
| 15:28.30 | nortex | I need some help handling dialing 2 sip devices when an extension is called. I got the call to go, but I get a deadlock when one of the two answers |
| 15:28.34 | EyeCue | then it aind behind nat? :) |
| 15:28.49 | *** join/#asterisk af_ (n=af@ip-164-6.sn2.eutelia.it) |
| 15:28.59 | clyrrad1 | We have been using mpg123 and have just deleted it - and trying to use format_mp3 - is there any config file changes that need to be made to have Asterisk use format_mp3? |
| 15:29.35 | brad6254 | it is all on our local network |
| 15:30.05 | EyeCue | clie log any help? |
| 15:30.20 | EyeCue | i swear im just gonna give up, s/clie/cli |
| 15:30.27 | EyeCue | my typing blows tonight. |
| 15:31.53 | clyrrad1 | anyone on the format_mp3 question? |
| 15:32.19 | [TK]D-Fender | clyrrad1 : "mode=files" |
| 15:32.33 | clyrrad1 | thats the only file? |
| 15:32.56 | [TK]D-Fender | clyrrad1 : thats it. You'll then be on Native MoH. |
| 15:33.07 | clyrrad1 | Great - thank you sir :) |
| 15:33.15 | [TK]D-Fender | clyrrad1 : and clearly I'm reffering to modding musiconhold.conf |
| 15:33.19 | brad6254 | the only error in the logs is missing caller id and a features error |
| 15:33.22 | clyrrad1 | tired of seeing all those "Request to schedule in the past errors" |
| 15:33.24 | *** join/#asterisk paolob (n=donpaolo@pri-214-b7.codetel.net.do) |
| 15:33.46 | clyrrad1 | TKD - yes I knew you were talking about musiconhold.conf :p |
| 15:34.07 | paolob | Hi guys! My asterisk server gives me the following error: "WARNING[9309]: chan_sip.c:1066 __sip_xmit: sip_xmit of 0x81c31f8 (len 471) to 83.138.130.145:5060 returned -1: Bad file descriptor". What is it? |
| 15:34.26 | paolob | It gives it repeatedly, and not only for that IP |
| 15:35.20 | *** join/#asterisk anonymouz666 (n=anonymou@200.218.193.6) |
| 15:35.43 | carl0s- | piece of shit. it's gone back to :Looking for useip in from-trunk (domain 192.168.253.15) |
| 15:35.43 | carl0s- | Reliably Transmitting (no NAT) to 192.168.253.3:5060: |
| 15:35.43 | carl0s- | SIP/2.0 404 Not Found |
| 15:35.50 | *** join/#asterisk SplasPood (n=jwb@206.252.198.101) |
| 15:35.54 | anonymouz666 | russellb, Asterisk 1.4 is ready? I wanna use :) |
| 15:35.57 | carl0s- | the trunk is called useip. it's bloody there. |
| 15:36.50 | russellb | no, it is not |
| 15:37.08 | carl0s- | i'll double-check.. |
| 15:37.17 | anonymouz666 | russellb: tomorrow? |
| 15:37.18 | anonymouz666 | :D |
| 15:37.56 | carl0s- | russellb: it's there for sure. |
| 15:38.03 | russellb | it's not that what is there isn't bad |
| 15:38.08 | *** join/#asterisk boch (n=root@201.216.241.97) |
| 15:38.15 | russellb | it's just that we have some more big features we want to finish, first |
| 15:38.24 | file | carl0s-: ignore russellb, he's talking to anonymouz666 |
| 15:38.31 | carl0s- | yeah, I just realised that! |
| 15:38.34 | file | carl0s-: are you talking about in AMP? |
| 15:39.04 | *** join/#asterisk Vorondil (n=jkiffmey@miro.voltaiccommerce.com) |
| 15:39.14 | carl0s- | file: well, it's set in AMP, but I've got the file (sip_additional.conf - which is #included in sip.conf) open in vi and it's all there. |
| 15:39.26 | boch | i've set my sip peer dtmfmode=auto but when i do 'sip show peer' in CLI it says rfc2833, is it right? |
| 15:39.40 | file | carl0s-: okay so it's looking for exten => useip in from-trunk, or in an included section (context) |
| 15:41.08 | carl0s- | file: well I have this in my sip.conf, is it not enough? http://pastebin.ca/94836 |
| 15:41.13 | file | no |
| 15:41.33 | file | sip.conf controls SIP configuration, ie: phones, trunks, etc... |
| 15:41.33 | carl0s- | hrm |
| 15:41.47 | file | extensions.conf controls how each dialed number is handled... what instructions get executed |
| 15:41.59 | file | note this is REALLY boiled down and not using the right terms, since you don't know them yet |
| 15:42.29 | mutilator | yes, it also controls global climate changes |
| 15:42.35 | mutilator | but you'll find that out later |
| 15:43.03 | mutilator | +2 Troll |
| 15:44.57 | carl0s- | file: I was hoping that the call would be handled by the wide-open 'match any did/cid' rule which I have set in amp. |
| 15:45.18 | brad6254 | Maybe i'll try again. asterisk 1.2.9 zaptel 1.2.6 sip to zap work fine, incoming calls on zap ring at sip, asterisk shows them answered, but no voice. |
| 15:45.23 | file | well, it isn't :) |
| 15:46.12 | carl0s- | I've added an incoming route, with the DID as useip. I wonder if this will help. |
| 15:46.21 | brad6254 | rtp.conf shows rtpstart=10000 rtpend=20000 |
| 15:47.18 | *** join/#asterisk trbldwine (n=trbldwin@adam.ur.northwestern.edu) |
| 15:48.14 | brad6254 | sip using 11160 |
| 15:48.26 | carl0s- | HEY |
| 15:48.28 | carl0s- | it worked. |
| 15:48.33 | TripleFFFF | t = timeout |
| 15:48.36 | TripleFFFF | h=hangup |
| 15:48.39 | TripleFFFF | o = what ? |
| 15:48.54 | carl0s- | I added a new Inbound route with did=useip, and whaddaya know! my Cisco 7960 phone rang.. with my Caller ID showing :D |
| 15:50.19 | *** part/#asterisk brad6254 (n=brad6254@pool-71-162-32-182.altnpa.east.verizon.net) |
| 15:50.24 | file | yay |
| 15:51.02 | carl0s- | This 'useip' thing obviously isn't the ideal thing to have. If the GSM VoIP gateway just passed the URI with no user-part, then none of it would have been necessary would it? This also means you couldn't use more than one of these GSM boxes together, 'cause the 'useip' userpart seems to be hard coded. |
| 15:51.11 | Toerkeium | guys, trying to "make linux26" zaptel, I get : You do not appear to have the sources for the 2.6.8-022stab078.10 kernel installed. |
| 15:51.24 | Toerkeium | and sources are there.. any idea why it doesn't find it ? |
| 15:51.50 | stoffell | Toerkeium: ls -l /usr/src/ |
| 15:52.03 | carl0s- | Thanks for your help file!. I still have 1 or 2 problems, but I suspect they're just bad firmware in the GSM box. If I restart Asterisk, the GSM box isn't able to do it's thing until I reboot that. Could this be solved by altering who is the initiator of the connection (asterisk -> gsm, or vice versa?) or not? |
| 15:52.13 | Toerkeium | stoffell: drwxr-xr-x 4 root root 4096 Jul 21 12:53 kernels |
| 15:52.16 | *** join/#asterisk brad6254 (n=brad6254@pool-71-162-32-182.altnpa.east.verizon.net) |
| 15:52.19 | *** join/#asterisk Winkie (n=urmom@cpc3-stre1-0-0-cust656.bagu.cable.ntl.com) |
| 15:52.22 | file | trial/error is great |
| 15:52.23 | Winkie | damnit freenode |
| 15:52.27 | Toerkeium | stoffell: drwxr-xr-x 7 root root 4096 May 29 12:43 redhat |
| 15:52.39 | [TK]D-Fender | file : trial/execution is much more fun.... |
| 15:52.40 | Qwell | file: trial/success is better :D |
| 15:52.44 | stoffell | Toerkeium: try pastebin.ca instead of pasting in channel |
| 15:53.02 | carl0s- | file: it's bloody not. I was about to say before "I've fscked right off with this trial and error crap". but then i'd have to read books for months to get anything done at all :( |
| 15:53.06 | Toerkeium | stoffell: it's only that 2 lines |
| 15:53.31 | stoffell | Toerkeium: okay, then there's no kernel on the default location.. (and I'm not into readhat, so i don't know how to get them there:) ) |
| 15:53.47 | carl0s- | So what's next.. hmm. I need to have my incoming upstream SIP calls (0845-number) coming through to my mobile, via Asterisk and the GSM gateway :D |
| 15:54.14 | Toerkeium | stoffell: I will try to find which is the default location, thank you |
| 15:54.56 | brad6254 | When setting up sip phones, is it better to use different rtp ports for each phone or the same rtp port for all phones? |
| 15:56.43 | clyrrad1 | brad6254 - I belve thats handled internally by Asterisk but I could be wrong |
| 15:57.14 | brad6254 | clyrrad1, I can dial from sip to zap, but from zap to sip is answered but not sound |
| 15:57.26 | brad6254 | I mean no sound |
| 15:57.33 | *** join/#asterisk backblue (n=igor@82.102.1.42) |
| 15:57.39 | backblue | hi, ppl |
| 15:57.44 | clyrrad1 | do you have your firewall ports open and matched to rdp.conf? |
| 15:57.50 | clyrrad1 | rpt* |
| 15:57.53 | backblue | it's there any way we can do progress to a call, without it allready haves? |
| 15:58.12 | brad6254 | it is all on our local network, so no nat |
| 15:58.12 | clyrrad1 | errrr rtp* |
| 15:58.14 | clyrrad1 | there we go |
| 15:58.56 | clyrrad1 | when you call from Zap to sip i assume the phone rings? |
| 15:58.59 | brad6254 | rtp.conf has rtpstart=10000 rtpend=20000 |
| 15:59.08 | brad6254 | yes it rings |
| 15:59.12 | carl0s- | dial pattern: "907|07. " that should do the trick for having all "9 07xxx" numbers dialed out as just "07xxx" over a specified trunk shouldn't it? |
| 15:59.20 | clyrrad1 | and if you hang up the ZAP does the SIP stop ringing? |
| 15:59.27 | brad6254 | yes |
| 15:59.34 | clyrrad1 | so its getting the singals okay |
| 15:59.43 | brad6254 | again, yes |
| 16:00.03 | clyrrad1 | it looks like it is being FW somewhere |
| 16:00.15 | clyrrad1 | every time I had this issue it was FW related |
| 16:00.33 | brad6254 | by fw do you mean forwarding |
| 16:00.38 | clyrrad1 | Firewall |
| 16:00.45 | EyeCue | hmm ok |
| 16:00.54 | brad6254 | it isn't passing a firewall |
| 16:00.57 | EyeCue | outside iax client to my * to my workstation sip is working |
| 16:00.58 | EyeCue | :D |
| 16:00.58 | clyrrad1 | blocking LAN packets or mis-directing them |
| 16:01.07 | EyeCue | but iax client to * to iax client is |
| 16:01.07 | brad6254 | that what has me so confused |
| 16:01.12 | clyrrad1 | there gotta be something blocking it |
| 16:01.12 | EyeCue | -- Attempting native bridge of IAX2/knarfly-7 and IAX2/koobs-8 |
| 16:01.12 | EyeCue | -- Operating with different codecs 4[(ulaw)] 524[(ulaw|alaw|speex)] , can't native bridge... |
| 16:01.16 | EyeCue | any clues? :) |
| 16:01.20 | backblue | boobs? :o |
| 16:01.37 | clyrrad1 | EyeCue - set your phones and dial plans to use the same codecs |
| 16:01.38 | backblue | EyeCue: not a problem. |
| 16:01.39 | *** join/#asterisk stoffell (n=stoffell@pot.catsanddogs.com) |
| 16:01.44 | Toerkeium | stoffell: the default location would be: /usr/src/kernels/ and I have it in that location |
| 16:01.59 | EyeCue | hmm, * cant override that? |
| 16:02.02 | *** join/#asterisk ToTo (n=ToTo@host212-207.pool872.interbusiness.it) |
| 16:02.14 | backblue | EyeCue: yes, it do transcoding. |
| 16:02.15 | clyrrad1 | brad6254 - how does the ZAP connect to your * box? |
| 16:02.22 | backblue | if it suports any of the both ends protocols |
| 16:02.24 | clyrrad1 | directly by card? |
| 16:02.31 | stoffell | Toerkeium: most config scripts expect it to be in /usr/src/linux or /usr/src/linux-2.6 etc.. read up on the docs at voip-info.. it'll explain it better |
| 16:02.37 | brad6254 | tdm400 |
| 16:02.45 | EyeCue | so do i need to override disallow all allow ulaw in my iax.conf ? |
| 16:02.50 | EyeCue | or is that the wrong place |
| 16:02.52 | clyrrad1 | you sure you have the card configed properly? |
| 16:03.06 | clyrrad1 | EyeCue set disallow=all - then allow only what you want |
| 16:03.14 | EyeCue | copy that |
| 16:03.25 | EyeCue | in general i assume? |
| 16:03.30 | clyrrad1 | yes |
| 16:03.34 | clyrrad1 | or in each context if you like |
| 16:03.37 | EyeCue | sure |
| 16:03.48 | TripleFFFF | is manager timout in MS ? |
| 16:03.49 | TripleFFFF | ms |
| 16:03.51 | EyeCue | once i get it working iax > * > iax, that is, ill fine tune |
| 16:03.52 | *** join/#asterisk jhiver (n=jhiver@LReunion-151-20-4.w193-253.abo.wanadoo.fr) |
| 16:03.53 | TripleFFFF | 45000 is 45 sec |
| 16:04.00 | clyrrad1 | TDK - you still here? |
| 16:04.03 | brad6254 | I did have it working at one time, again that's what puzzles me, but i have something i want to try it'll take a bit |
| 16:04.11 | Toerkeium | thank you |
| 16:04.54 | clyrrad1 | Anyone here ever made a *XX feature that users can press that streams MOH music to the phones? |
| 16:05.22 | nortex | What actually causes "WARNING[3473] chan_zap.c: Ring requested on channel 0/1 already in use on span 1. Hanging up owner." I got 15-20 in a row in a second. |
| 16:05.27 | nortex | clyrrad1, Yes |
| 16:05.36 | clyrrad1 | nortex - can you explain how you set it up |
| 16:05.46 | clyrrad1 | I get issues with it just stopping the music in mid play |
| 16:05.48 | clyrrad1 | and hanging up |
| 16:06.03 | eKo1 | nortex: someone is trying to call a zap channel that's already in use. |
| 16:06.18 | paolob | Guyy, what should I put in sip.conf as bindaddress? If I put 0.0.0.0 I get a "chan_sip.c:1066 __sip_xmit: sip_xmit of 0x81c31f8 (len 471) to 83.138.130.145:5060 returned -1: Bad file descriptor" error, if I put 10.152.58.0 (my LAN network) I doesn't work. Any hint? |
| 16:06.27 | nortex | eKo1, This is on a PRI, is that possible? |
| 16:06.50 | clyrrad1 | nortex - can you tell me how you set it up? |
| 16:06.56 | *** join/#asterisk brif8 (n=Administ@ns1.ttienterprises.org) |
| 16:07.17 | eKo1 | nortex: seems like it from the message. |
| 16:08.31 | brif8 | I first have SetVar(Counter=1) and then I have SetVar(Counter = ${MATH(${Counter})+1)}) yet every I repeat to the second SetVar (Counter) is always 1 thus I get 1+1 = 2 and it doesn't become 2+1 = 3 it remains 1+1 = 2 for each loop Why and how do I fix Thanks |
| 16:09.00 | nortex | eKo1, Could that be from the telco side? I don't have any outgoing calls at the time. |
| 16:09.24 | eKo1 | nortex: most likely |
| 16:09.40 | eKo1 | or it could just be * playing with your head. |
| 16:10.17 | nortex | eKo1, Well it seems to be related to people not being able to get through and dropped calls. |
| 16:10.31 | EyeCue | HEHAHEAH |
| 16:10.42 | EyeCue | iax2*2iax is workies :D |
| 16:12.49 | russellb | brif8: take those spaces out of your Counter assignment |
| 16:12.56 | russellb | you are literally setting a variable called "Counter " |
| 16:13.40 | brif8 | yes a variable called counter |
| 16:13.53 | russellb | note the space inside the quotes in my message |
| 16:14.09 | russellb | and no need for MATH, either |
| 16:14.31 | russellb | change that whole thing to ... Set(Counter=$[${Counter} + 1]) |
| 16:16.55 | brif8 | thanks the removing of the ' ' between ' = ' did the job thanks |
| 16:17.58 | *** join/#asterisk manopulus (n=manopulu@cable-10-68.cgates.lt) |
| 16:19.00 | russellb | and use Set instead of SetVar ... |
| 16:19.02 | manopulus | hello, please someone help me with PHP agi. first i need to send command 'Ringing' to the context (long resolving, 4 seconds, ringing to avoid disconnects). second i need to use command SET or SETVAR, but: AGI Rx << Fatal error: Call to undefined method AGI::set_var() in /var/lib/asterisk/agi-bin/lnp-lookup.php on line 58 |
| 16:19.15 | russellb | SetVar is deprecated and will not be present in future releases |
| 16:19.42 | manopulus | ok, but what i can use instead in PHP AGI |
| 16:19.55 | brif8 | russellb: in a gotoif do you need spaces between < like gotoif($[${Counter}] < 5?10:13) and do I need 5 in "5" ? |
| 16:21.00 | russellb | the only thing wrong with that is that you misplaced your '] |
| 16:21.14 | russellb | move the ']' to be after the 5 and it's fine |
| 16:21.24 | *** join/#asterisk dahunter3 (n=dahunter@pool-71-110-103-209.lsanca.dsl-w.verizon.net) |
| 16:22.20 | brif8 | so GotoIf($[${Counter} < 5]?10:13) correct ? |
| 16:22.28 | russellb | yes |
| 16:22.32 | brif8 | thanks very much |
| 16:24.08 | rpm | is it possible to include contexts with realtime in a pgsql database? or does it have to all be in one big context? |
| 16:24.31 | russellb | not currently, no. you have to set up the contexts in extensions.conf |
| 16:24.46 | russellb | contexts and includes ... only the extensions themselves can be in realtime |
| 16:24.50 | *** part/#asterisk brif8 (n=Administ@ns1.ttienterprises.org) |
| 16:25.18 | rpm | russellb: thanks. |
| 16:25.20 | *** join/#asterisk Qwell[] (i=north@unaffiliated/qwell) |
| 16:25.20 | *** mode/#asterisk [+o Qwell[]] by ChanServ |
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| 16:30.07 | nestar | anyone here using a channel bank to take POTS lines and present them to Asterisk? |
| 16:30.31 | mog | sure |
| 16:30.32 | eKo1 | many |
| 16:30.35 | eKo1 | but not me |
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| 16:31.42 | *** join/#asterisk parag_ast (n=root@dxb-b18160.alshamil.net.ae) |
| 16:32.03 | parag_ast | Anybody can provide me SIP/IAX SDK... |
| 16:32.13 | nestar | seems like i might be better off just using a 4 port FXO card, just concerned with flakeyness |
| 16:32.20 | nestar | used to the simplicity of a PRI |
| 16:32.53 | [TK]D-Fender | russellb : why is it that it was done that way anyhow? |
| 16:33.06 | russellb | why is what which way |
| 16:33.06 | *** part/#asterisk System010 (n=jgargano@hide247.cybergnostic.com) |
| 16:33.08 | carl0s- | hmm. I'm getting close. When I answer my mobile phone, Asterisk doesn't seem to know that the call has been answered. It still shows it as 'Ringing'. This is when the call is placed out via the GSM box. |
| 16:33.21 | parag_ast | <PROTECTED> |
| 16:33.36 | russellb | parag_ast: iaxclient, google it |
| 16:33.48 | [TK]D-Fender | russellb : realtime not including context info. |
| 16:33.51 | parag_ast | ahh i searched a lot |
| 16:34.05 | nestar | anyone used the Rhino FXO card? |
| 16:34.48 | knarfly | parag_ast: just curious...whats SDK |
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| 16:34.48 | knarfly | oops |
| 16:34.48 | Bullseye_Network | hate it when that happends :) |
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| 16:34.58 | knarfly | I gott quit hitting that escape key |
| 16:34.58 | russellb | damnit |
| 16:34.59 | russellb | i was trying to have a conversation |
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| 16:34.59 | *** mode/#asterisk [+o anthm] by ChanServ |
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| 16:35.07 | Bullseye_Network | Tell them to quit flooding the channel |
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| 16:35.12 | [TK]D-Fender | eek |
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| 16:35.17 | Bullseye_Network | ~pb |
| 16:35.18 | jbot | it has been said that pb is a place to paste your stuff without flooding the channel - try http://pastebin.com/, or http://pastebin.ca/, or http://channels.debian.net/paste, or for #oe use http://oe.pastebin.com/, or http://bzflag.pastebin.ca/, or for images use http://imageshack.us/ |
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| 16:36.39 | EyeCue | wb filo |
| 16:36.41 | nestar | what's the best phone for the money right now? i'm using polycoms, currently.. but deploying a new office, so considering different options |
| 16:36.58 | mutilator | i like the polycom phones myself |
| 16:37.03 | knarfly | how much money you got? |
| 16:37.07 | mutilator | theres always cisco too |
| 16:37.14 | nestar | deff not going cisco |
| 16:37.19 | nestar | too much $$$ for a name |
| 16:37.32 | mutilator | then stick with that.. |
| 16:42.30 | *** join/#asterisk jbot (i=ibot@pdpc/supporter/active/TimRiker/bot/apt) |
| 16:42.30 | *** topic/#asterisk is Asterisk: The Open Source PBX -=- http://www.asterisk.org -=- Asterisk 1.2.10 and Zaptel 1.2.7 released! (July 14, 2006) -=- FreePBX/AMP/Asterisk@Home/Trixbox support in #freepbx |
| 16:42.34 | *** join/#asterisk AJaymn (n=Ya@70.59.126.206) [NETSPLIT VICTIM] |
| 16:42.34 | Winkie | freenode wins again |
| 16:42.36 | *** join/#asterisk Corydon76-home (i=mauve@pdpc/supporter/sustaining/Corydon76-home) |
| 16:42.36 | *** mode/#Asterisk [+o Corydon76-home] by ChanServ |
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| 16:42.40 | *** part/#asterisk Sonderblade (n=mah@213.131.147.169) |
| 16:42.40 | nestar | RE: polycom; the only thing i don't like about the polycom's is the call waiting thing.. was that ever resolved? |
| 16:42.40 | *** join/#asterisk devel (n=devel@wiggum.digitalcoven.com) |
| 16:42.43 | *** join/#asterisk ixx (i=foobar@cpe-70-112-73-77.austin.res.rr.com) |
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| 16:42.59 | [TK]D-Fender | nestar : Which?> |
| 16:43.11 | nestar | where you couldn't disable call waiting |
| 16:43.11 | *** join/#asterisk anderiv (n=anderiv@207.67.87.34) |
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| 16:43.14 | *** join/#asterisk scardinal (n=supreme@port816.ds1-suoe.adsl.cybercity.dk) |
| 16:43.17 | [TK]D-Fender | chan_nuke strikes again! |
| 16:43.17 | *** join/#asterisk test34 (n=test34@unaffiliated/test34) |
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| 16:43.28 | Toerkeium | guys, I now when I try to "make linux26" zaptel, it says: make[1]: *** No rule to make target `modules'. Stop. |
| 16:43.30 | *** join/#asterisk parag_ast (n=root@dxb-b18160.alshamil.net.ae) |
| 16:43.36 | Toerkeium | any idea what it means? |
| 16:43.41 | *** join/#asterisk Wi_Fi (n=OUT@c-24-127-12-85.hsd1.ca.comcast.net) |
| 16:43.43 | parag_ast | anybody is using vaxvoip SDK |
| 16:43.56 | *** join/#asterisk UlbabraB (n=UlbabraB@81.72.43.241) |
| 16:43.57 | Qwell[] | Toerkeium: got the kernel source or headers installed for your kernel? You need them |
| 16:44.02 | nestar | you'll have to excuse me, i'm still running a old version of *, but i was having an issue with a queue sending calls to a agent already on the phone, thus them getting call waiting beeps. |
| 16:44.24 | *** join/#asterisk rob0 (n=rob0@cardinal.lizella.net) |
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| 16:44.32 | nestar | i am using SetGroup/CheckGroup to get around that, but just having a call waiting disable in the polycom would have made it a non-issue |
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| 16:44.42 | *** join/#asterisk ben_d (n=ben@cpe-66-67-129-4.rochester.res.rr.com) |
| 16:44.46 | [TK]D-Fender | nestar : not a polycom issue.. its an * queue issue |
| 16:44.48 | *** join/#asterisk ptblank (n=MURDER1@ca-yorbalnd-cuda2-c3c-88.anhmca.adelphia.net) |
| 16:44.56 | Toerkeium | Qwell[]: yes I did.. before it was complaining about a the kernel no bein installed, and then I just copied kernels to linux and now it seems to find it |
| 16:44.58 | [TK]D-Fender | nestar : I csolved that by limiting the calls on mine. |
| 16:45.14 | *** join/#asterisk file (n=file@216.237.114.82) |
| 16:45.14 | *** mode/#Asterisk [+o file] by ChanServ |
| 16:45.15 | nestar | [TK]D-Fender: I agree to a point, but even my $59 Budgetone lets you disable call waiting |
| 16:45.26 | [TK]D-Fender | nestar : you CAN disable it.... |
| 16:45.43 | [TK]D-Fender | nestar : numlinekeys=1 callsperlinekey=1 |
| 16:45.56 | nestar | is that something they added to a later firmware? seems like i tried that to no avail. |
| 16:46.10 | *** join/#asterisk MatsK (i=MatsK@83.233.97.229) |
| 16:46.36 | [TK]D-Fender | nestar : Been there since the dawn of time. I'm only running 1.5.3 here myself. |
| 16:46.43 | nestar | i'm only running 1.3.1 |
| 16:46.44 | nestar | ;) |
| 16:46.45 | [TK]D-Fender | nestar : I run 2.0 Beta at home. |
| 16:46.57 | *** join/#asterisk TripleFFFF (n=TripleFF@147-102.mc.cite.net) [NETSPLIT VICTIM] |
| 16:47.01 | *** join/#asterisk Bobcat_1966 (n=chatzill@cpe-069-132-139-254.carolina.res.rr.com) |
| 16:47.03 | TripleFFFF | <PROTECTED> |
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| 16:47.03 | *** mode/#Asterisk [+o anthm] by ChanServ |
| 16:47.06 | TripleFFFF | any idea ? |
| 16:47.08 | *** join/#asterisk Beirdo (n=gjhurlbu@unaffiliated/beirdo) |
| 16:47.09 | [TK]D-Fender | nestar : You deperately need to get your ass in gear and get SOMEWHAT up to date... |
| 16:47.10 | *** join/#asterisk Nugget (i=nugget@dazed.slacker.com) [NETSPLIT VICTIM] |
| 16:47.24 | nestar | [TK]D-Fender: just trying not to break something that works |
| 16:47.24 | nestar | :) |
| 16:47.27 | [TK]D-Fender | TripleFFFF : Try turning off CNG support in your client ;) |
| 16:47.39 | [TK]D-Fender | nestar : Apparently it DOESN'T ;) |
| 16:48.02 | Qwell[] | [TK]D-Fender: Go for the grin smiley hattrick |
| 16:48.09 | TripleFFFF | well |
| 16:48.12 | nestar | hehe |
| 16:48.16 | [TK]D-Fender | nestar : I don't ahve a 1.3 series firmware pack so i can't confirm where in ipmid it appears... that is a NASTY out of date combo. |
| 16:48.17 | TripleFFFF | its both outbounf |
| 16:48.17 | *** join/#asterisk nighty_ (n=nighty@66-163-28-100.ip.tor.radiant.net) |
| 16:48.19 | Qwell[] | lame |
| 16:48.35 | nestar | heheh, i rule. |
| 16:48.44 | [TK]D-Fender | Qwell : X > X |
| 16:48.56 | nestar | good thing i'm giving my 2 weeks today |
| 16:49.02 | nestar | let them figure it out when i'm gone |
| 16:49.03 | nestar | :) |
| 16:49.05 | *** join/#asterisk Ironhand (i=arjen@meek.xs4all.nl) |
| 16:49.26 | nestar | actually, i know that's not true. i bet i'll be doing consulting on this system for a dacade |
| 16:49.31 | nestar | i should fix it before i leave |
| 16:50.46 | *** join/#asterisk Samoied (n=Samoied@ip47092.static.poa.virtua.com.br) |
| 16:50.49 | nestar | holy crap, a PRI one county over is $1100 a month |
| 16:50.54 | nestar | i'm used to paying 350 |
| 16:50.58 | *** join/#asterisk Sponge_bob (n=None@cpe-66-27-162-13.socal.res.rr.com) |
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| 16:51.15 | nestar | telecommunications blackhole! |
| 16:52.50 | LoneShadow | so asterisk 1.2.10 still dosnt have googletalk support ? |
| 16:52.53 | Hmmhesays | so don't use a pri |
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| 16:53.27 | *** join/#asterisk backblue (n=igor@82.102.1.42) |
| 16:53.29 | nestar | Hmmhesays: but i <3 the pri |
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| 16:54.07 | Hmmhesays | spendy though |
| 16:54.22 | *** join/#asterisk tzanger (n=tzanger@mixdown.ca) |
| 16:54.44 | tzanger | fucking freenode and irssi |
| 16:55.03 | mutilator | and tuna fish |
| 16:55.07 | mutilator | fuck all tuna fish |
| 16:55.20 | nestar | yeah, we're not getting a pri. i wish though |
| 16:55.24 | *** join/#asterisk infinity1 (i=foobar@208.184.76.100) |
| 16:55.37 | *** join/#asterisk skraelings001 (n=skraelin@201.230.149.248) |
| 16:55.39 | infinity1 | anyone know why my music on hold sounds like darth vader? |
| 16:55.40 | *** join/#asterisk salviadud (n=ralfalfa@201.153.40.45) [NETSPLIT VICTIM] |
| 16:57.17 | skraelings001 | hello, i'm outside northamerica i want to buy some equipment, i know abptech and voipsupply , any other good voip equipment supplier? |
| 16:57.18 | eKo1 | maybe because you have a darth vader moh theme in your setup? |
| 16:57.38 | eKo1 | skraelings001: in na? |
| 16:57.57 | Hmmhesays | infinity1: could be many a reason |
| 16:58.10 | skraelings001 | eko1: yes in na |
| 16:58.25 | Hmmhesays | bad source? |
| 16:58.30 | Hmmhesays | using two lossy codecs? |
| 16:59.18 | nestar | so the polycom 430 has a speakerphone? |
| 16:59.43 | *** join/#asterisk masonf (n=masonf@dungle.vineyard.net) |
| 17:00.02 | Qwell[] | mutilator: I'm pretty sure that's illegal in at least 42 states |
| 17:00.10 | *** join/#asterisk heliosj (n=jeff@pdpc/supporter/active/xheliox) |
| 17:00.28 | masonf | why was the appliucation prefix deprecated? |
| 17:00.32 | mutilator | only if it's alive |
| 17:00.33 | mutilator | sheesh |
| 17:00.33 | Hmmhesays | tuna fish condom? |
| 17:00.53 | mutilator | gotta get the old rotten ones so it's atleat warm |
| 17:00.58 | *** join/#asterisk russellb (n=russell@asterisk/developer-and-stable-maintainer/drumkilla) |
| 17:00.59 | *** mode/#Asterisk [+o russellb] by ChanServ |
| 17:01.08 | Qwell[] | masonf: Because it's useless |
| 17:01.19 | Qwell[] | masonf: Set(MYVAR=abc${BLAH}) |
| 17:01.35 | *** join/#asterisk brad6254 (n=brad6254@pool-71-162-32-182.altnpa.east.verizon.net) |
| 17:01.42 | Qwell[] | instead of Prefix(MYVAR=abc,${BLAH}) or whatever crazy syntax it used |
| 17:02.04 | parag_ast | Qwell[],do u know any SIP/IAX sdk, I want to hide proxy settings.. |
| 17:02.17 | *** join/#asterisk carl0s- (n=carl0s@compsup.demon.co.uk) |
| 17:02.30 | masonf | would this work? exten => _NXXXXXX,Set(EXTEN=508${EXTEN}) |
| 17:02.36 | MatsK | ~debug |
| 17:02.38 | jbot | ACTION DeBuggers $1 |
| 17:02.41 | Qwell[] | masonf: sure |
| 17:02.45 | russellb | masonf: no, you would use Goto |
| 17:02.52 | russellb | you can't set EXTEN |
| 17:02.56 | Qwell[] | russellb: it would technically work :p |
| 17:03.03 | russellb | no it wouldn't! |
| 17:03.06 | Qwell[] | oh ;/ |
| 17:03.07 | russellb | you wouldn't be able to read it back |
| 17:03.15 | Qwell[] | details |
| 17:03.19 | parag_ast | Qwell[],do u know any SIP/IAX sdk, I want to hide proxy settings.. |
| 17:03.24 | russellb | it justs wastes some memory, that's about it |
| 17:03.37 | russellb | parag_ast: stop asking the same question over and over |
| 17:04.03 | parag_ast | russellb, I m in big problem thats why i m asking |
| 17:04.21 | russellb | there are many IAX and SIP client libraries out there |
| 17:04.25 | russellb | go search for them |
| 17:04.30 | *** join/#asterisk stkn (n=stkn@gentoo/developer/pdpc.active.stkn) |
| 17:04.30 | *** join/#asterisk stephane_ (n=stephane@merlin.cabale.net) |
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| 17:04.55 | *** join/#asterisk blebleble (i=godie@caesar.godie.net) |
| 17:05.21 | blebleble | what is the best sollution for setting up a scalable asterisk server? cluster? some other method? any good stable utilities out there? |
| 17:05.30 | infinity1 | anyone know why my music on hold sounds like darth vader? there aren't any errors on the console. very strang |
| 17:06.08 | *** join/#asterisk nortex (n=breeves@69.6.154.70) |
| 17:06.55 | Bullseye_Network | infinity1: make sure you are using mpg123 NOT mpg321 |
| 17:07.04 | *** join/#asterisk niZon (n=ilt@S010600080db4ab60.wp.shawcable.net) |
| 17:07.07 | Bullseye_Network | and the correct version |
| 17:07.09 | Qwell[] | infinity1: and more specifically, mpg 0.59r |
| 17:07.29 | *** join/#asterisk key2 (n=key2@sd-420.dedibox.fr) |
| 17:07.33 | infinity1 | Bullseye_Network: check. |
| 17:07.37 | infinity1 | hm |
| 17:07.43 | skraelings001 | any good voip equipment supplier in NA besides abptech and voipsupply?? |
| 17:08.06 | nestar | i've had good dealings with Atacomm |
| 17:08.11 | Bullseye_Network | skraelings001: What are you looking for? |
| 17:09.34 | infinity1 | ah ha! |
| 17:09.46 | skraelings001 | Bullseye_Network: TDM13B && SPA2100 && SPA3000 && SPA2002 |
| 17:09.48 | infinity1 | mpg123 doesn't exist as a package for amd64 in debian |
| 17:10.03 | nortex | Qwell, Have you used the pause and unpause queue member applications? |
| 17:10.14 | Qwell[] | no |
| 17:10.52 | *** join/#asterisk trbldwine (n=trbldwin@adam.ur.northwestern.edu) |
| 17:10.55 | Bullseye_Network | infinity1: I had the same problem. I could NOT find a work around |
| 17:11.11 | infinity1 | Bullseye_Network: uhhh |
| 17:11.12 | Qwell[] | infinity1: in the asterisk source, type `make mpg123` |
| 17:11.14 | [TK]D-Fender | Screw MPG123, use Native |
| 17:11.21 | Qwell[] | or use native :p |
| 17:11.25 | *** join/#asterisk manopulus (n=manopulu@cable-10-68.cgates.lt) |
| 17:11.28 | infinity1 | Qwell[]: native? |
| 17:11.31 | Bullseye_Network | BUT that was awile ago. And now use the builtin mp3 player in the new version |
| 17:11.35 | infinity1 | convert to gsm? |
| 17:11.36 | [TK]D-Fender | infinity1 : *'s built in MoH |
| 17:11.41 | nortex | Is there a way to check a queue members status without trying to change it, ie check if they are paused or a member or not? |
| 17:11.43 | manopulus | hello. question. Jul 21 18:09:02 NOTICE[9949]: frame.c:179 __ast_smoother_feed: Dropping extra frame of G.729 since we already have a VAD frame at the end. any way how to fight with this ? |
| 17:12.01 | Qwell[] | nortex: I was looking...I don't recall if I found an answer or not |
| 17:12.03 | manopulus | it is in sip |
| 17:12.44 | Bullseye_Network | skraelings001: I have some cards I can sell TDM400P, what modules do you need? |
| 17:12.55 | *** join/#asterisk saftsack (n=saftsack@p54A7D810.dip.t-dialin.net) |
| 17:13.12 | *** join/#asterisk GerjanT (n=gerjan@frontgate.watchthe.net) |
| 17:13.15 | GerjanT | thomson 2030 is also very nice |
| 17:13.22 | GerjanT | end of statement ;) |
| 17:13.45 | skraelings001 | Bullseye_Network: hmm.. i require those equipment |
| 17:14.28 | Bullseye_Network | SO: 3 FXO and 1 FXS? |
| 17:14.43 | nortex | Qwell, the macro I did yesterday works great for add and remove since there is a status of already a member or not an member, but the pause status is either paused or not. I will have to add a astdb key I guess and check it. |
| 17:14.47 | russellb | manopulus: you can ignore it ... |
| 17:15.00 | brad6254 | zap to sip doesn't work, sip to zap works fine. The call is answered, but there is no audio. rx monitor on the channel shows rx nothing, tx is transfering. don't understand why |
| 17:15.11 | manopulus | russellb: but i get less quality |
| 17:15.17 | manopulus | russellb: tested just.. |
| 17:15.20 | russellb | well we don't fully support VAD |
| 17:15.22 | Qwell[] | nortex: If you pause an already paused member, it should be fine, right? |
| 17:15.26 | russellb | so just turn it off, then |
| 17:15.28 | Qwell[] | Or are you trying to have one macro that does both? |
| 17:15.29 | skraelings001 | Bullseye_Network: yes |
| 17:16.04 | nortex | Do both, I like making it on feature code and let the system sort it out. Asterisk is smarter then users you know :) |
| 17:16.13 | Qwell[] | yeah |
| 17:16.16 | Bullseye_Network | skraelings001: I mostly have the other way.... |
| 17:16.25 | Qwell[] | So...that won't be a very large patch to add that |
| 17:16.45 | Qwell[] | It *should* IMO return ALREADYPAUSED or something |
| 17:16.47 | skraelings001 | Bullseye_Network: thanks man, i'll keep googling |
| 17:17.21 | *** join/#asterisk MooingLemur (n=troy@shells200.pinchaser.com) |
| 17:17.59 | nortex | Qwell, Agreed but I am now programmer so I will use the status of paused to the write to the db and then read that into a gotoif |
| 17:18.09 | nortex | s/now/no |
| 17:18.35 | infinity1 | hm. tried setting mode=files, but that doesn't work. seems like a lib is missing |
| 17:19.01 | nortex | Qwell, are you a programmer? |
| 17:19.05 | Qwell[] | I am |
| 17:19.16 | russellb | Qwell[] is a l33t cod3 h4x0r |
| 17:19.21 | Qwell[] | russellb: :D |
| 17:19.55 | russellb | and the reason chan_skinny will rock in 1.4 |
| 17:20.01 | Qwell[] | Just have to convince a person or two of that... |
| 17:20.06 | file | LIES |
| 17:20.09 | Qwell[] | so "things" can proceed :p |
| 17:20.11 | salviadud | when will 1.4 be released? |
| 17:20.16 | Qwell[] | salviadud: "soon" |
| 17:20.19 | salviadud | it's torture! |
| 17:20.20 | file | when it's ready |
| 17:20.24 | russellb | Qwell[]: yes, things indeed |
| 17:20.29 | salviadud | slackware 11... asterisk 1.4 |
| 17:20.34 | infinity1 | is 1.4 going to have better ael stuff? |
| 17:20.38 | salviadud | the new pansat fix |
| 17:20.39 | Qwell[] | infinity1: MUCH better |
| 17:20.41 | *** join/#asterisk jbalcomb (n=jbalcomb@216.28.180.158) |
| 17:20.43 | russellb | infinity1: a *new* implementation |
| 17:20.47 | nortex | Cool :D So what would I need to do to get a patch for the pause/unpause to return an already is status? |
| 17:20.49 | russellb | from scratch |
| 17:20.50 | Qwell[] | murf did a GREAT job on AEL |
| 17:21.13 | Qwell[] | nortex: a programmer who is currently accepting new work.. |
| 17:21.14 | russellb | AEL2 is the new hotness |
| 17:21.19 | salviadud | will 1.4 be backwards compatible? |
| 17:21.28 | russellb | salviadud: with what |
| 17:21.29 | jbalcomb | [TK]D-Fender: are you waiting on me or am i waiting on you? I have a meeting monday morning about the queues. |
| 17:21.34 | russellb | salviadud: yes, it will be :) |
| 17:21.52 | russellb | that has always been a priority for us |
| 17:22.00 | *** join/#asterisk arcy (n=arcanum@ppp139-238.adsl.forthnet.gr) |
| 17:22.04 | salviadud | yeah :) |
| 17:22.13 | nortex | Qwell, How about where do I create a request for it to be added in future versions? |
| 17:22.24 | salviadud | russellb, i can't wait man |
| 17:22.28 | Qwell[] | nortex: bugs.digium.com, but if there is no patch...don't hold your breath... |
| 17:22.28 | russellb | now, if we could get IMAP storage support for voicemail merged ... |
| 17:22.39 | russellb | just a few more big things before we can start the beta process ... |
| 17:23.21 | russellb | asterisk 1.4 will be hot |
| 17:23.31 | nortex | Qwell, Maybe I should go take a c++ class :) |
| 17:23.37 | Qwell[] | just c |
| 17:23.40 | *** join/#asterisk knarfly (n=bmorris@c-69-180-98-189.hsd1.fl.comcast.net) |
| 17:23.45 | *** join/#asterisk samrobb_ (n=samrobb_@65.117.135.105) |
| 17:23.45 | russellb | C will do, indeed |
| 17:23.50 | russellb | though I wish we'd switch to C++ |
| 17:23.53 | Qwell[] | eww |
| 17:23.57 | *** part/#asterisk parag_ast (n=root@dxb-b18160.alshamil.net.ae) |
| 17:24.02 | russellb | Qwell[]: yep, i said it. |
| 17:24.03 | *** part/#asterisk samrobb_ (n=samrobb_@65.117.135.105) |
| 17:24.08 | sponix | russellb: so, how does asterisk work anyway ? |
| 17:24.09 | nortex | Not much for me, basic and cobol are my limits :) |
| 17:24.14 | Qwell[] | might as well switch to java :P |
| 17:24.17 | russellb | sponix: um ... magic? |
| 17:24.19 | sponix | russellb: its like voip, riiight ? |
| 17:24.23 | russellb | Qwell[]: oh shush |
| 17:24.27 | Qwell[] | heh |
| 17:24.35 | russellb | that's just ignorant talk :) |
| 17:24.41 | Qwell[] | it is :) |
| 17:24.50 | *** join/#asterisk marv0997 (i=marv0997@190.4.2.86) |
| 17:25.02 | jbalcomb | russellb: is that a quote? |
| 17:25.10 | russellb | jalsot: no, it's not |
| 17:25.18 | russellb | :) |
| 17:26.15 | russellb | you guys are going to get me in trouble or something, heh |
| 17:26.16 | jbalcomb | watch the other SysEng. run a cable is like watching my dad try to hook up the entertainment system |
| 17:26.28 | *** join/#asterisk Nivex (i=kjotte@user-0ce2nsu.cable.mindspring.com) |
| 17:26.59 | *** join/#asterisk dahunter3 (n=dahunter@pool-71-110-103-209.lsanca.dsl-w.verizon.net) |
| 17:27.08 | *** join/#asterisk marv0997 (i=marv0997@190.4.2.86) [NETSPLIT VICTIM] |
| 17:28.10 | jbalcomb | can someone do a comprehensive write up on what normal phone functions are, what keys codes they are bound too, and a somewhat standard flow of call queues for a call center? |
| 17:28.25 | carl0s- | Where can I go to learn about SIP registation? Peers, friends, and registration strings? I suspect my GSM VoIP gateway isn't registering properly, which could explain why I have to reboot it if I restart asterisk. When I do "sip show registry" it doesn't show, so it's obviously registering in the same way Handsets do, rather than Gateways/Providers. |
| 17:28.25 | jbalcomb | I need it by Monday at 10 AM. |
| 17:28.31 | brad6254 | what are the best setting for tdm400 fxo modules in zapata.conf. |
| 17:28.43 | mutilator | jbalcomb i got a flow chart right here |
| 17:28.48 | russellb | brad6254: the best setting of all is signalling=fxs_ks |
| 17:28.53 | jbalcomb | mutilator: jbalcomb@imtco.com |
| 17:28.54 | russellb | the others are optional, really :) |
| 17:28.56 | mutilator | pstn -> pbx -> queue -> phone -> person |
| 17:28.59 | *** join/#asterisk paolob (n=donpaolo@pri-214-b7.codetel.net.do) |
| 17:28.59 | mutilator | ; |
| 17:29.03 | mutilator | :P |
| 17:29.10 | jbalcomb | haha.. thanks. that should work. |
| 17:29.14 | mutilator | coo coo |
| 17:29.14 | carl0s- | brad6254: make sure to set correct opermode= in /etc/modules.conf if you're outside of North America also. |
| 17:29.17 | mutilator | glad i could help |
| 17:29.22 | jbalcomb | "It's all very simple really..." |
| 17:29.30 | brad6254 | have that. the problem is i can call sip to zap, but zap to sip has no voice |
| 17:29.36 | mutilator | heh |
| 17:29.56 | Qwell[] | brad6254: Your problem is very likely with SIP |
| 17:30.04 | *** join/#asterisk mkrufky (n=mk@68.160.103.77) |
| 17:30.16 | Qwell[] | specifically related to nat |
| 17:30.35 | brad6254 | i can also call sip to sip, and it works |
| 17:30.42 | brad6254 | over the same lan |
| 17:31.15 | brad6254 | i can dial from sip to zap, and then voice comes through fine. |
| 17:34.29 | brad6254 | qwell[] - is it best to set all rtp to the same port, or use different ports for each phone? |
| 17:34.34 | brad6254 | sip phone that is |
| 17:34.53 | Toerkeium | anyone know what means this message while "make linux26" zaptel? Warning: "zt_hooksig" [/usr/src/zaptel-1.2.7/pciradio.ko] has no CRC |
| 17:35.04 | Toerkeium | and that for all modules |
| 17:36.03 | *** join/#asterisk Assid (i=assid@203.115.83.215) |
| 17:37.06 | *** join/#asterisk stoffell (n=stoffell@pot.catsanddogs.com) |
| 17:37.34 | carl0s- | I thought the proper syntax was "register => blah" but AMP has done a "register=blah" entry in my sip.conf and it is working. Are both accepted or is there a difference? |
| 17:37.51 | *** join/#asterisk X-Gen (n=X-Gen@dsl-145-192-128.telkomadsl.co.za) |
| 17:38.32 | *** join/#asterisk Bert- (n=bert@i05v-87-90-132-119.d4.club-internet.fr) |
| 17:38.35 | Bert- | hello there |
| 17:38.59 | carl0s- | hi |
| 17:40.06 | Bert- | I have something strange : asterisk is running fine since few days, but now some sip phones can't register anymore. I'm not sure, but it appears on remote phones, not those on the lan. Does someone has any idee about that please ? |
| 17:40.57 | Bert- | for example, I was registered to * yesterday at same hour |
| 17:41.15 | Bert- | I changed nothing in conf, but now I can't register |
| 17:41.33 | *** join/#asterisk Dr-Linux|work (n=Linux@202.125.139.198) |
| 17:41.36 | nestar | firewall rules? |
| 17:41.37 | Bert- | That's why I'm a bit surprised |
| 17:42.13 | Bert- | there is no firewall rules on the host, and corrects ports have been set on the router since one month |
| 17:42.30 | Bert- | I mean |
| 17:43.07 | Bert- | I can reach asterisk, if I activate sip debug I can see my phone trying to register, but asterisk return 401 Unauthorized |
| 17:43.23 | *** join/#asterisk innatech (n=daf@netblock-72-25-97-119.dslextreme.com) |
| 17:43.42 | masonf | is there a way to dynmicly edit the extension that was dailed? I want to add an area code on to my a phone number and then I want it it to be picked up by the regexp _NXXNXXXXXX. |
| 17:43.59 | nestar | you try a sip reload, see if that changes the behaviour? |
| 17:44.06 | Qwell[] | masonf: goto |
| 17:44.29 | masonf | ty |
| 17:44.51 | *** join/#asterisk marv[work] (n=timr@64.89.118.139) |
| 17:44.56 | Bert- | I restarted asterisk with restart now, same issue |
| 17:45.00 | *** join/#asterisk noname32 (n=noname@38.113.5.165) |
| 17:45.40 | innatech | can anyone point me towards docs discussing how to designate when different MOH categories should be used? Haven't had much luck searching forums/listservs. |
| 17:46.16 | nestar | SetMOH |
| 17:46.16 | Bert- | So I wonder if is a kind of debug for the REGISTER process in asterisk, to try to understand why asterisk returns 401 |
| 17:46.35 | noname32 | hey guys i am trying to setup parking using hints with snom phones and i followed the istrustions on the wiki and the lines dont see to be lighting up |
| 17:47.03 | nestar | http://voip-info.org/tiki-index.php?page=Asterisk+cmd+Musiconhold |
| 17:48.46 | Bert- | no way ? |
| 17:49.17 | nestar | Bert-: i'm not sure. something changed, and if asterisk is saying the client is unauthorized, i would check the phone config |
| 17:50.28 | paolob | guys, I have * running on a ubuntu pc, and I'd like to upgrade it. Any idea if installing debian testing packet with all its deps. could work? |
| 17:52.17 | *** join/#asterisk mhnoyes (n=mhnoyes@dialup-4.246.6.10.Dial1.SanJose1.Level3.net) |
| 17:52.30 | TripleFFFF | 54 WARNING[32136]: cdr.c:550 ast_cdr_disposition: Cause not handled |
| 17:52.33 | TripleFFFF | whats this ? |
| 17:53.23 | *** part/#asterisk mhnoyes (n=mhnoyes@dialup-4.246.6.10.Dial1.SanJose1.Level3.net) |
| 17:54.52 | *** join/#asterisk bofh42 (n=bofh42@p5482C70B.dip0.t-ipconnect.de) |
| 17:55.27 | carl0s- | When I 'busy-out' an incoming call on my cellphone, my Asterisk box continues to ring out. Where should I be looking to fix this? |
| 17:55.33 | Bert- | nestar I tried with other phone and it is the same thing |
| 17:56.03 | Bert- | on what criterias asterisk auth or not a user ? |
| 17:56.10 | Bert- | I know login/pass and IP |
| 17:56.16 | Bert- | but what else ? |
| 17:57.34 | TripleFFFF | <PROTECTED> |
| 17:57.37 | TripleFFFF | on 200 calls |
| 17:57.39 | TripleFFFF | i get this |
| 17:57.40 | innatech | thanks for the URL, nestar. |
| 17:59.44 | TripleFFFF | anyone ? |
| 18:00.38 | TripleFFFF | and 3:12 WARNING[32136]: rtp.c:460 ast_rtp_read: RTP Read too short |
| 18:00.38 | TripleFFFF | also |
| 18:00.50 | TripleFFFF | does that mean box loaded ? |
| 18:00.53 | Bullseye_Network | trippleFFFF:I've seen that too: Cause not handled |
| 18:01.00 | Bullseye_Network | but havnt figured it out |
| 18:02.16 | *** join/#asterisk TrixVox (n=Spy007@c-69-248-121-104.hsd1.nj.comcast.net) |
| 18:02.18 | *** join/#asterisk ApEtc (i=apetc@ip70-162-197-214.ph.ph.cox.net) |
| 18:02.43 | TripleFFFF | asterisk -vrx 'show channels' | grep 'channels' |
| 18:02.44 | TripleFFFF | 101 active channels |
| 18:02.46 | TripleFFFF | shoudnt be |
| 18:02.52 | sponix | So, what is this asterisk good for .. you saying I can take some bw, and turn it into a phone service ? |
| 18:03.04 | Qwell[] | sponix: sure |
| 18:03.20 | sponix | Qwell[]: don't play with my emotions ! |
| 18:03.24 | *** join/#asterisk jets (i=jetsnoc@72.22.224.81) |
| 18:03.35 | *** join/#asterisk TypMic (n=TypMic@134.207.12.239) |
| 18:03.41 | Qwell[] | sponix: Do whatever you like with it |
| 18:03.45 | rob0 | Throw in some hardware, some bandwidth, and a lot of RTFM, and you have something useful! |
| 18:04.02 | TypMic | Strom_C, Tall-guy you around |
| 18:04.05 | TripleFFFF | lol |
| 18:04.11 | TripleFFFF | Jul 21 13:06:19 WARNING[32136]: cdr.c:550 ast_cdr_disposition: Cause not handled |
| 18:04.21 | *** join/#asterisk gandhijee (n=gandhije@mail.win-ent.com) |
| 18:04.33 | sponix | Qwell[]: what do you typically use it for ? |
| 18:04.43 | Qwell[] | sponix: pr0n mostly |
| 18:04.53 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 18:04.55 | sponix | rob0: eww, sounds like work |
| 18:05.21 | sponix | Qwell[]: its a pr0n transfer protocol in hiding ? |
| 18:05.27 | Qwell[] | it can be |
| 18:05.42 | Qwell[] | or you can use it for traditional telecom stuff |
| 18:06.02 | rob0 | It's *fun* work. |
| 18:06.15 | *** join/#asterisk effectiveape (n=nick@82.153.22.16) |
| 18:06.44 | effectiveape | Anyone any experience in configuring a cisco 7940 for sip? |
| 18:06.46 | sponix | but, most home users would just it computer-to-computer for free calling, similar to skype riiight ? |
| 18:07.34 | TripleFFFF | oh well |
| 18:07.53 | Assid | arghh... im looking for a good router.. can someone suggest me some names? |
| 18:07.56 | sponix | rob0: hmm, so I should rtfm or stfu ? :) |
| 18:08.09 | sponix | Assid: what exactly are you routing ? |
| 18:08.14 | Assid | internet? |
| 18:08.22 | effectiveape | heh |
| 18:08.27 | [TK]D-Fender | Assid : iptables |
| 18:08.34 | sponix | Assid: two NICS, in a p120 Obsd box works pretty darn good :) |
| 18:09.07 | Assid | no no.. need one of those linksys and crap.. |
| 18:09.16 | sponix | can do highly advanced routing (more features that $2,000 cisco's) for $200 or so |
| 18:09.23 | effectiveape | any special interfaces? (adsl?) |
| 18:09.29 | Assid | verizon |
| 18:11.39 | *** part/#asterisk TypMic (n=TypMic@134.207.12.239) |
| 18:13.15 | Nugget | pf+altq is dreamy. |
| 18:16.44 | Assid | effectiveape: dsl |
| 18:19.35 | *** join/#asterisk Samoied (n=Samoied@ip47092.static.poa.virtua.com.br) |
| 18:24.25 | *** join/#asterisk nentis (n=nentis@hotblack.opensourcery.com) |
| 18:24.48 | [TK]D-Fender | sponix : I can get a nice P3 for less than that ;) |
| 18:25.08 | nentis | odd issue. Even after a reboot everything audio related that comes out of asterisk is really, rrreeeaallly, slowed down. Voice prompts, voicemail, everything. |
| 18:26.22 | *** join/#asterisk cluetrain (n=bionoid@148.80-202-39.nextgentel.com) |
| 18:27.07 | effectiveape | anyone know anyone who might be into cisco phones? |
| 18:27.54 | effectiveape | yeah i've found it a squeeze myself |
| 18:29.12 | carl0s- | Anyone fancy debugging my 'sip debug' output? Something is Not Found, but I don't know what! http://pastebin.ca/95005 |
| 18:30.30 | *** join/#asterisk s0lid (n=s0lid@202.71.179.140) |
| 18:32.36 | carl0s- | chan_sip.c: Registration from '<sip:useip@192.168.253.15>' failed for '192.168.253.3' - Username/auth name mismatch |
| 18:33.17 | carl0s- | Where can I find out what username/auth name the box is sending to Asterisk? I was presuming that "useip" was the user/auth name? as in the uri there? or is 'sip' the username and 'useip' the password? |
| 18:33.27 | TripleFFFF | TIMEOUT(digit)=2") |
| 18:33.34 | TripleFFFF | will make it 2 sec or 2 digits ? |
| 18:33.43 | TripleFFFF | TIMEOUT(response)=3" |
| 18:33.44 | Dovid | carl0s-: on your phone u are using useip for the user name, either thats not right or the pass is wrong or both |
| 18:33.44 | TripleFFFF | and |
| 18:33.50 | cluetrain | Hello. I've been looking around at the Asterisk online docs, in early planning stage. My goal is to extend a single analog line to a remote user via Internet. Set up Asterisk as a SIP server, add a Zaptel device that connects to analog line, and perform some black magic. What I'm a bit confused about, however, is which of the 50 configuration files I should look into ;-) Is the dialplan used for such "1:1 forwarding", or is there some ot |
| 18:35.00 | carl0s- | Dovid: ok. so 'useip' is the username. I thought so. This is a GSM VoIP box and it seems to be hard-coded. I'm sure I've set the pass to 'useip' too at both ends. Is it not possible to see what password is being sent? |
| 18:35.18 | Dovid | nope |
| 18:35.43 | Dovid | sip is the protocol being used and useip is the userid being sent |
| 18:35.50 | Dovid | something is not set properly |
| 18:35.55 | carl0s- | fair enough. |
| 18:36.01 | Dovid | did u add a sip account to asterisk ? |
| 18:36.10 | Dovid | make sure u dont have a typo ? |
| 18:36.14 | Dovid | did u reload asterisk ? |
| 18:36.30 | carl0s- | I added a sip extension, with context from-trunk, with username=useip and secret=useip |
| 18:36.55 | Dovid | this error will come up with what ever context u put in |
| 18:36.59 | Dovid | post ur sip.conf |
| 18:37.00 | Assid | err.. wouldnt i need lke a dsl modem? |
| 18:37.08 | carl0s- | 1 sec |
| 18:38.20 | *** part/#asterisk nentis (n=nentis@hotblack.opensourcery.com) |
| 18:38.25 | carl0s- | Dovid: http://pastebin.ca/95026 is the relavent part. |
| 18:39.39 | innatech | anyone know why a registering a broadvoice SIP trunk might hang on "request sent"? The registration string appears to be correct. |
| 18:39.49 | Dovid | ok. thats correct |
| 18:40.08 | Dovid | it must be the way u set it up in the device |
| 18:40.11 | *** join/#asterisk sevard (n=sev@adsl-71-129-115-244.dsl.irvnca.pacbell.net) |
| 18:40.43 | carl0s- | i'm going to look at it again now. It's very frustrating because after a few minutes, you can't get into HTTP configurator anymore on the device and have to power it off. Bad firmware I think. |
| 18:42.09 | *** join/#asterisk mitcheloc (n=mitchelo@70-32-189-246.lmdaca.adelphia.net) |
| 18:42.29 | *** join/#asterisk a1fa (n=a1fa@207.210.210.202) |
| 18:42.35 | a1fa | hey |
| 18:42.42 | a1fa | can * send sms directly to a gsm cellphone? |
| 18:43.19 | a1fa | i found an addon, but routing is expensive.. $0.16/message |
| 18:43.38 | a1fa | http://www.bayhamsystems.com/ |
| 18:45.45 | Dovid | a1fa: where is the number ? |
| 18:45.47 | nortex | Where does * store who been added dynamicly to the queues? |
| 18:47.29 | a1fa | Dovid : anywhere in the world |
| 18:47.46 | *** join/#asterisk dahunter3 (n=dahunter@64.239.166.5) |
| 18:47.52 | Dovid | oh, thought u wanted some one to send a sms |
| 18:47.54 | Qwell[] | nortex: memory, and during a reload, it stores it in astdb |
| 18:48.11 | a1fa | yes |
| 18:48.15 | a1fa | I want to send and recieve messages |
| 18:49.21 | a1fa | i guess this is not possible? |
| 18:49.34 | Qwell[] | a1fa: There is an ap_sms |
| 18:49.36 | Qwell[] | app_sms |
| 18:50.03 | a1fa | but dont you need some kind of access numbers |
| 18:50.07 | a1fa | to upload the sms too |
| 18:50.13 | *** join/#asterisk kc5cqm (i=mwilliam@2002:a55f:d1d:0:0:0:0:1) |
| 18:50.16 | a1fa | so you really cant deliver it directly to the client |
| 18:50.24 | *** part/#asterisk kc5cqm (i=mwilliam@2002:a55f:d1d:0:0:0:0:1) |
| 18:50.48 | *** join/#asterisk momelod (n=momelod@bas5-toronto12-1128748195.dsl.bell.ca) |
| 18:50.55 | momelod | greetings room |
| 18:51.08 | Assid | isnt there gnokii or something.. which you can connect your cellphone to ? |
| 18:51.10 | sevard | I wish there was an option in screen to monitor activity or no activity in a portion of a window. |
| 18:51.42 | Qwell[] | momelod: AOL has rooms...IRC has channels. :p |
| 18:52.20 | Assid | sevard: open a console and run asterisk in there.. and resize it to smalll and move it to a section of your screen |
| 18:52.52 | momelod | im trying to setup a menu system. Ive setup * to play a recording presenting the caller with some options.. if the caller enters a digit during the playback of the recording it works, but after the recording and during the Wait(3) stage, the caller cannot enter his option?? |
| 18:53.08 | sevard | Assid: wtf are you talking about. |
| 18:53.47 | momelod | does Wait() not accept dtmf? should i be using something else? |
| 18:54.03 | russellb | momelod: use WaitExten |
| 18:54.10 | momelod | :D thanks |
| 18:54.58 | *** join/#asterisk wunderkin (n=wunderki@216-19-202-11.getnet.net) |
| 18:55.12 | *** join/#asterisk Mercestes (n=Merceste@216.54.143.2) |
| 18:55.24 | *** join/#asterisk c4t3l (n=c4t3l@69.15.174.114) |
| 18:59.44 | carl0s- | 10hrs now I've been sat in front of this screen today# |
| 19:03.04 | Assid | man.. i wish i could access my directory from the p301 in 1 button or 2 max.. instead of searching |
| 19:03.23 | *** join/#asterisk Zodiacal (i=hehehe@bdsl.66.14.242.199.gte.net) |
| 19:03.25 | Assid | and going through the whole menu |
| 19:04.04 | [TK]D-Fender | Assid : reprogram a button for it. |
| 19:04.16 | [TK]D-Fender | Assid : take over the DND button for it |
| 19:06.33 | *** join/#asterisk frenzy (n=frenzy@196.46.104.184) |
| 19:06.43 | frenzy | anyone here owning a grandstream device? |
| 19:06.53 | [TK]D-Fender | Polycom should include a small pile of lre-labeled key-caps with their phones... |
| 19:07.02 | Cresl1n | I sell granstream insurance for a fair price |
| 19:07.12 | nestar | lol |
| 19:07.16 | nestar | ghettotone |
| 19:07.19 | nestar | i have 3 of them |
| 19:07.20 | Qwell[] | Cresl1n: I thought of a good insurance you can sell... |
| 19:07.24 | [TK]D-Fender | Barbietone! |
| 19:07.25 | Cresl1n | and random gravity well insurance as well |
| 19:07.26 | Qwell[] | but I forgot it :( |
| 19:07.27 | Cresl1n | :-D |
| 19:07.29 | nestar | they are "meh" to "feh" |
| 19:07.33 | Cresl1n | cheaper if it's bundled |
| 19:07.39 | frenzy | I can get flash to work |
| 19:07.44 | frenzy | cant** |
| 19:07.54 | [TK]D-Fender | frenzy : Tried a trench-coat? |
| 19:08.11 | Cresl1n | Qwell[]: !!!! |
| 19:08.17 | Qwell[] | Cresl1n: !!!!? |
| 19:08.19 | frenzy | I've got send send flash even enabled |
| 19:08.25 | Qwell[] | You get a "?", because you're just THAT cool |
| 19:08.42 | frenzy | event** |
| 19:09.07 | *** join/#asterisk stkn_ (i=nobody@gentoo/developer/pdpc.active.stkn) |
| 19:10.05 | frenzy | anyone? |
| 19:10.49 | *** join/#asterisk h3x0r4t0r (i=hex@ip70-189-236-254.lv.lv.cox.net) |
| 19:11.59 | *** join/#asterisk JohnJacob (n=JohnJaco@pool-71-127-102-43.aubnin.fios.verizon.net) |
| 19:13.15 | Assid | [TK]D-Fender: i can reprogram a button ? |
| 19:13.28 | Assid | i'd like to get rid of the my buddies |
| 19:14.08 | bjohnson | wouldn't we all |
| 19:14.14 | [TK]D-Fender | Assid : not the soft-keys (AFAIK), but I mean reprogram a hard-button. |
| 19:14.49 | [TK]D-Fender | Assid : My professional "removal" services are requested on another channel : #eastriver |
| 19:16.35 | Assid | i cant get rid of the soft buttons? |
| 19:17.49 | [TK]D-Fender | Assid : not any way that I know of. Buddies can be removed by specifically provisioning your phone against a folder whose sip.cfg disables presence. |
| 19:18.23 | Assid | well.. i just wanna put directory in there.. somewhere.. |
| 19:18.30 | *** join/#asterisk mog (i=ejabberd@68.62.237.103) |
| 19:18.30 | *** mode/#Asterisk [+o mog] by ChanServ |
| 19:19.09 | *** join/#asterisk amdtech (n=amd011@ss-5-100.shsu.edu) |
| 19:19.23 | *** join/#asterisk tempest1 (n=asf@adsl-144-36-60.chs.bellsouth.net) |
| 19:19.59 | [TK]D-Fender | Assid : So like I said, overide the "DND" hard-button and change the key-cap |
| 19:20.28 | [TK]D-Fender | Assid : on a PBX I just did consulting work for I changed it to VM access so they could have 1-touch VM on them |
| 19:21.41 | Assid | [TK]D-Fender: how do i override? |
| 19:21.50 | Assid | as in what values to change? |
| 19:22.59 | Assid | for a smart phone.. they really should let us program the softkeys instead.. that would be more versatile |
| 19:23.13 | Qwell[] | *cough*skinny*cough* |
| 19:24.41 | [TK]D-Fender | *cough*nociscohalfbakedproprietarycrapthatforcesmetorunchanneldriverspimpedonbyqwellforhisownselfimageandfindmyselfstrandedwhenichoosetouseanotherpbxlikefreeswitchperhaps*cough* |
| 19:24.44 | *** join/#asterisk amdtech (n=amd011@ss-5-100.shsu.edu) |
| 19:25.02 | Qwell[] | ;) |
| 19:25.27 | [TK]D-Fender | </whitespacenullify> |
| 19:25.27 | Assid | hehe |
| 19:25.31 | Assid | man.. you guys are so lucky |
| 19:25.54 | [TK]D-Fender | <- teh comic |
| 19:25.58 | Assid | fios!! .. and here.. im spending about $30 anwaysy.. and getting a shitty 256kbps.. which dies out 1/2 the time |
| 19:26.11 | *** join/#asterisk Synyn_ (n=Synyn@cpe-72-181-72-81.houston.res.rr.com) |
| 19:26.22 | Synyn_ | afternoon folx |
| 19:27.02 | Synyn_ | going to try to move to asterisk to solaris, wish me lock ) |
| 19:27.09 | [TK]D-Fender | Assid : I spend $45 CAD at home for 5mbps/800kbps (solid synch and transfer), fixed IP, unlimited bandwidth, and running on bare copper (no phone line required) |
| 19:27.17 | Qwell[] | Synyn_: x86? |
| 19:27.25 | Synyn_ | Qwell: aye |
| 19:27.26 | [TK]D-Fender | Synyn_ : I"m sure it'll lock all over the place ;) |
| 19:27.34 | Qwell[] | might be a little better, but... |
| 19:27.47 | Assid | [TK]D-Fender: you guys are lucky |
| 19:27.53 | Synyn_ | I'm better on solaris then linux... |
| 19:28.05 | Qwell[] | Synyn_: good, feel like helping me tonight? :P |
| 19:28.14 | eKo1 | [TK]D-Fender: bare copper? |
| 19:28.15 | Synyn_ | qwell: sure |
| 19:28.28 | [TK]D-Fender | eKo1 : Yup. Dry-line (naked) DSL |
| 19:28.49 | eKo1 | You mean the cable has no shielding? |
| 19:28.50 | [TK]D-Fender | eKo1 : thats my only bill asides from my rent for my apt :) |
| 19:29.01 | Synyn_ | I need a new setup, I got a 5mb cable, but I really need 5 statics |
| 19:29.02 | [TK]D-Fender | eKo1 : No, nmeans I have no DIALTONE, and no telco bill. |
| 19:29.17 | Qwell[] | [TK]D-Fender: How does that work? Do you just connect to somebodys DSLAM? |
| 19:29.37 | nestar | same as regular DSL |
| 19:29.41 | eKo1 | I have Dry-line DSL here as well. |
| 19:29.41 | nestar | just no phone service |
| 19:29.48 | eKo1 | yeah... |
| 19:29.49 | Synyn_ | its unsplit |
| 19:30.04 | nestar | phone service is not required for dsl, it's just a company policy with most lecs |
| 19:30.04 | [TK]D-Fender | Qwell : just like any other ADSL. difference is the CRTC told Bell to fuck off and allow users to get bare line w/o Bell service on top of it. EVERYONE uses Bell's DSLAM here. |
| 19:30.19 | Assid | [TK]D-Fender : as i said im spending that much anyways.. and getting shit.. |
| 19:30.22 | Qwell[] | right, but who do you actually get IP from? |
| 19:30.37 | Assid | iana! |
| 19:30.59 | nestar | wish my boss would get out of this meeting so i can give my 2 weeks |
| 19:31.05 | nestar | lalalala |
| 19:31.07 | Qwell[] | nestar: heh |
| 19:31.21 | nestar | wish i could get naked dsl |
| 19:31.22 | Synyn_ | the provider will have circuits in bell's dslam |
| 19:31.26 | nestar | i would be all over that |
| 19:31.29 | [TK]D-Fender | Qwell : that comes from my ISP which is a small UNIX house. uber-techie guys. |
| 19:31.36 | Qwell[] | [TK]D-Fender: fun |
| 19:31.38 | *** join/#asterisk Maxxed (i=foobar@65.59.245.122) |
| 19:31.48 | Assid | anyone wanna give my isp bandwith ?!?? |
| 19:31.54 | nestar | bellsouth is the scum of the earth |
| 19:32.02 | nestar | they put the cum in scum |
| 19:32.09 | Qwell[] | s/bellsouth/bell/ |
| 19:32.18 | nestar | meh |
| 19:32.21 | Synyn_ | cat *bell* > /dev/null |
| 19:32.24 | [TK]D-Fender | Qwell : indeed. dry-line DSL only became available to Bell's wholesale customers at the start of this year which coincided with my move to a new apt. |
| 19:32.47 | *** join/#asterisk klasstek (n=nunyobiz@ip67-154-143-190.z143-154-67.customer.algx.net) |
| 19:33.11 | [TK]D-Fender | Qwell : And naturally I'm running my * box on a Sangoma S518 ADSL card to boot :) |
| 19:33.11 | Synyn_ | I"m lucky, where I live is one of the few places I can only cable internet without paying for cable tv |
| 19:33.30 | nestar | we can get cable modem without cable tv |
| 19:33.49 | nestar | but dsl you have to have a phone line.. which is why 60% of the broadband in the city is with the cable company |
| 19:33.50 | [TK]D-Fender | Synyn_ : you can get that here to, but that company are a TOTAL bunch of retards, and there is like 2 wholesalers. Largely get screwed by them. |
| 19:33.51 | Maxxed | hey i upgraded my sip firmware on mt 7960's |
| 19:33.56 | nestar | and they are true scumbags as well. |
| 19:33.57 | Maxxed | now when i get a phone call in via the pstn |
| 19:34.02 | Maxxed | it shows like |
| 19:34.06 | Qwell[] | it shows the IP? |
| 19:34.12 | Maxxed | the phonenumber@ip of the pbx |
| 19:34.13 | Qwell[] | yeah, tell cisco about it |
| 19:34.17 | Nugget | Maxxed: yeah, that's how the new firmware behaves |
| 19:34.17 | Synyn_ | [TK]D-Fender: Hehe, thats was I was thinking of doing for my * box, but I'm moving it to solars (driver blackhole) |
| 19:34.20 | [TK]D-Fender | Synyn_ : SO we can get cable internet w/o TV, but they remove the "incentive" *rebate* if you don't which jacks the price up |
| 19:34.23 | Maxxed | known problem? |
| 19:34.25 | Maxxed | thats lame |
| 19:34.29 | Maxxed | no way to switch it off? |
| 19:34.33 | effectiveape | ooh someone mention cisco? |
| 19:34.45 | effectiveape | i can't get my 7940 to update to a sip image |
| 19:34.47 | [TK]D-Fender | Synyn_ : Solaris?! why the heck would you want to do that? |
| 19:35.07 | Synyn_ | how the heck do I pm in xchat...nothing works |
| 19:35.14 | Maxxed | effectiveape: you prob need to start with an older image, and work your way up to the new stuff |
| 19:35.19 | nestar | Synyn_: /msg ? |
| 19:35.29 | Synyn_ | ah, nm, I'm tardo |
| 19:35.32 | nestar | lol |
| 19:35.35 | Maxxed | hah |
| 19:35.36 | nestar | me too |
| 19:35.37 | effectiveape | Maxxed: Any idea how i can get hold of the older images? |
| 19:35.39 | Maxxed | dub click the nick? |
| 19:35.41 | Maxxed | um, well |
| 19:35.47 | Maxxed | see if u can ask one of these guys in here |
| 19:35.51 | effectiveape | Cisco have the 8.2 avail for download but i can't find any other |
| 19:35.58 | Maxxed | i have them but it would be a roal bitch to post it right now |
| 19:36.04 | x86 | re |
| 19:36.13 | effectiveape | anyone else? |
| 19:36.38 | nestar | sorry, i don't ball on cisco's level |
| 19:36.38 | nestar | maybe next year |
| 19:36.46 | x86 | effectiveape: you have a CCO login? |
| 19:36.58 | effectiveape | Not sure how to get one |
| 19:37.19 | nestar | have to have cco to download from their page |
| 19:37.21 | *** join/#asterisk innatech (n=daf@netblock-72-25-97-119.dslextreme.com) |
| 19:37.39 | effectiveape | for the 8.2 image you can just download it with anon-email |
| 19:37.43 | Maxxed | you gota pay for it |
| 19:37.49 | effectiveape | they have just put that on |
| 19:37.52 | Maxxed | the old images i think |
| 19:38.18 | effectiveape | I just need to get these phones working with * |
| 19:38.23 | Maxxed | im thinkin about downgrading if i cant fix this XXXXXXX@1.2.3.4 crap |
| 19:38.36 | Qwell[] | effectiveape: link? |
| 19:38.39 | effectiveape | it's a right pain |
| 19:38.44 | Maxxed | nah |
| 19:38.48 | Maxxed | its not that bad |
| 19:38.52 | Maxxed | once u get the swing of it |
| 19:38.53 | nestar | <3 cisco |
| 19:38.58 | nortex | effectiveape, Why not run 8.2? |
| 19:39.11 | Maxxed | he needs to load an older version |
| 19:39.12 | effectiveape | i'd love to but i can't get it to use the image |
| 19:39.19 | Maxxed | do the stagard upgrade if im not mistaken |
| 19:39.38 | Maxxed | like start at ver a then upgrade to c then d and then i think u can use the lastest shit then |
| 19:39.39 | Qwell[] | What's on it now? skinny 3.x? |
| 19:39.41 | effectiveape | and by the sounds i'd have to pay to get the older ones |
| 19:39.48 | Maxxed | or its like 2 images u have to load before u can load the new chit |
| 19:40.02 | Maxxed | sombody should beable to hook u up with the old images |
| 19:40.25 | Synyn_ | anyone used connect.voicepulse.com? they claim one line can use 4 channels outbound and 4 channels in bound concurrently, sounds nice |
| 19:40.34 | effectiveape | P0030301MFG2 |
| 19:40.40 | nortex | Check with the Vendor who sold it to you. |
| 19:40.43 | effectiveape | Which is guess is something like that |
| 19:40.44 | Qwell[] | effectiveape: upgrade to latest skinny first |
| 19:40.51 | Qwell[] | then go right over to sip |
| 19:40.57 | Maxxed | hey effective |
| 19:41.00 | Maxxed | your in luck |
| 19:41.02 | effectiveape | where do i get the image for that from? |
| 19:41.07 | tzanger | exten => 2915112,n,GotoIfTime(14:30-17:00|*|21|Jun?call-bvines,2915112,1) |
| 19:41.12 | Qwell[] | same place you get the image for the others |
| 19:41.13 | Maxxed | i got a bunch of my old images tar.gz'd up and on a webserver |
| 19:41.13 | tzanger | ok why is that not executing today |
| 19:41.17 | Maxxed | hang on il get u the link |
| 19:41.22 | effectiveape | excellent. |
| 19:41.30 | effectiveape | Yeah but the only image avail is the sip8.2 |
| 19:41.47 | Qwell[] | effectiveape: get a cisco contract |
| 19:41.52 | Qwell[] | it's cheap, and it makes you legal |
| 19:41.58 | effectiveape | how much is cheap |
| 19:42.03 | Qwell[] | $8? |
| 19:42.12 | effectiveape | and why (having spent on buying 8 phones) isn't this legal |
| 19:42.30 | nortex | tzanger, It is July here not Jun |
| 19:42.33 | Maxxed | hey effectiveape u want the hook up? |
| 19:42.38 | effectiveape | defo |
| 19:42.41 | Maxxed | i got the old images tar'd up and ready to wget ;) |
| 19:42.42 | tzanger | oh for fuck sakes |
| 19:42.44 | tzanger | thank you nortex |
| 19:42.50 | effectiveape | excellent |
| 19:42.52 | nortex | no prob |
| 19:43.12 | nestar | lol |
| 19:43.30 | x86 | Maxxed: URL? |
| 19:44.12 | sevard | muhahahahaha |
| 19:45.10 | nortex | Bad idea having a cat in your lap. |
| 19:45.42 | sevard | only when you're naked - sicko. |
| 19:45.44 | Skarmeth | hi all |
| 19:47.28 | nortex | claws are claws, no matter the layers. |
| 19:48.03 | Skarmeth | how I can set a variable for a certain context? like, I have 5 contexts [morning] [launch] [afternoon] [evening] [menu] and inside of each time context I want set the Audio File Name that I needs to be played... |
| 19:48.46 | *** join/#asterisk Juggie (n=agony@CPE00c049d9f271-CM00137186c8d8.cpe.net.cable.rogers.com) |
| 19:49.03 | Skarmeth | this way, [menu] include's another contexts and inherits it's variables |
| 19:50.19 | [TK]D-Fender | Skarmeth : Variables are kept between contexts. |
| 19:50.23 | Assid | are the pap2 u get there unlocked by default? |
| 19:50.36 | *** join/#asterisk quid2478 (n=quid24@CPE00131078ba5d-CM000f9f7eff1e.cpe.net.cable.rogers.com) |
| 19:52.06 | Skarmeth | [TK]D-Fender] ok, but how I can set it, since I am not using a extension... ? like in [globals]... |
| 19:52.50 | [TK]D-Fender | Skarmeth : Variables are CREATED in the dialplan. CONSTANTS are created in [globals] |
| 19:52.52 | Skarmeth | my actual definition was [morning] \n MENUFILE=filename |
| 19:53.01 | *** join/#asterisk tsurk0 (n=tsurko@85.187.160.157) |
| 19:53.20 | Skarmeth | this way I need to create varios menu's |
| 19:53.28 | [TK]D-Fender | Skarmeth : thats not the way.... |
| 19:53.34 | a1fa | anybody doing peering with SPA-3000? |
| 19:53.54 | [TK]D-Fender | a1fa : Peering? I am clearly SUPERIOR! |
| 19:53.56 | Skarmeth | I need to use SetVar apps right? it only can be used in exten => ... instructions... |
| 19:54.04 | a1fa | [TK]D-Fender :) |
| 19:54.15 | a1fa | [TK]D-Fender : using SPA-3000 as a peer? |
| 19:54.16 | [TK]D-Fender | Skarmeth : that'd be Set in 1.2+ and yes... |
| 19:54.27 | a1fa | [TK]D-Fender : i want to do GSM RELAYING from one country to another |
| 19:54.28 | [TK]D-Fender | a1fa : I use as "firend" |
| 19:54.30 | Assid | okay whats the different between spa-2002 and pap2 ? |
| 19:54.51 | a1fa | [TK]D-Fender : heh. i want to try it as a peer |
| 19:54.58 | eKo1 | Assid: packaging |
| 19:54.59 | [TK]D-Fender | Assid : PAP2 doesn't have all the CLASS optiosn and a few other things I think. Also branding .... |
| 19:55.06 | a1fa | so i can route calls to it |
| 19:55.13 | [TK]D-Fender | a1fa : it isn't jsut a peer.... it places AND receives calls. |
| 19:55.17 | Assid | class options [TK]D-Fender ? |
| 19:55.27 | [TK]D-Fender | a1fa : thats why it should be as a "friend" |
| 19:55.40 | a1fa | [TK]D-Fender : can you send calls to it via 093499@devic? |
| 19:55.41 | [TK]D-Fender | Assid : Vertical services... (* codes) |
| 19:55.42 | a1fa | device? |
| 19:55.48 | Assid | aah okay |
| 19:55.55 | Assid | so spa2002 better? |
| 19:56.04 | [TK]D-Fender | a1fa : I did like Dial(SIP/SPA3000FXO/5551212) |
| 19:56.11 | [TK]D-Fender | Assid : Safer choice. |
| 19:56.26 | Skarmeth | [TK]D-Fender] Set(Var(VARNAME=VARVALUE)) right? |
| 19:56.27 | Assid | know a good place he can get it from? for ny ? |
| 19:56.36 | a1fa | hm? and that works |
| 19:57.17 | [TK]D-Fender | Skarmeth : nope.... |
| 19:57.33 | a1fa | [TK]D-Fender : is that because it has incoming/outgoing extension? |
| 19:57.35 | [TK]D-Fender | Assid : Voipsupply. and get the SPA-2002. |
| 19:57.58 | a1fa | one stage dialing yes |
| 19:58.00 | [TK]D-Fender | a1fa : friend because you PLACE calls to the PSTN, but INCOMING calls from the PSTN go to *. get it? |
| 19:58.08 | a1fa | yeah |
| 19:58.12 | a1fa | i get it |
| 19:58.22 | Assid | [TK]D-Fender: http://www.linksys.com/servlet/Satellite?c=L_Product_C2&childpagename=US%2FLayout&cid=1152745270028&pagename=Linksys%2FCommon%2FVisitorWrapper <-- thats the one |
| 19:58.23 | [TK]D-Fender | a1fa : Good. I owned one and used it as my primary FXO. |
| 19:58.33 | a1fa | [TK]D-Fender : its cheap as f0ck |
| 19:59.05 | *** join/#asterisk Bobcat_1966 (n=chatzill@cpe-069-132-139-254.carolina.res.rr.com) |
| 19:59.15 | carl0s- | Right. What's the friggin' deal with 'REGISTER'ing? My GSM VoIP gateway *seems* to be mostly working, most-ish of the time. But I'm still seeing "REGISTER sip:[gwbox-addr]" following by a reponse of "SIP/2.0 404 Not Found". |
| 19:59.20 | [TK]D-Fender | a1fa : Gets the job done. Not glorious, but simple and no messing with your box for it. |
| 19:59.31 | a1fa | [TK]D-Fender : http://voipspeak.net/index.php?option=com_content&task=view&id=24&Itemid=27&limit=1&limitstart=3 |
| 20:00.03 | Assid | i dont get it.. why would someone wanna buy the spa3000 when you can get the 2002 cheaper |
| 20:00.10 | a1fa | here is the guide to set it up as a trunk |
| 20:00.12 | noname32 | hi there any recomandations that is reliable and affordable for a company that does 7k outbound mins 12k toll free, 1k inbound on did with about 120 did numbers and 30 channels |
| 20:00.17 | a1fa | i guess, there is no difference between the two setupd |
| 20:01.59 | [TK]D-Fender | a1fa : Thats an AMP sample and as you can see they are using PEER & user. hence duplication for which was the very reason of the compuond "friend" class. |
| 20:02.01 | Skarmeth | [TK]D-Fender] thanks I got it (/usr/share/doc/asterisk-1.2.10/README.variables) |
| 20:02.09 | [TK]D-Fender | a1fa : thiers is DOUBLE. |
| 20:02.10 | quid2478 | hmm.. what's the purpose of appending an extension to a SIP registration string ie. username:password@sipprovider.com/NXXNXXXXXX? |
| 20:02.12 | Assid | [TK]D-Fender: im guessing the ones from voip-supply are unlocked? |
| 20:02.15 | Skarmeth | Set(Var=value) |
| 20:02.30 | [TK]D-Fender | Assid : SPA-2002's are typically unlocked. |
| 20:02.40 | a1fa | [TK]D-Fender : yup i get it |
| 20:03.17 | a1fa | [TK]D-Fender : so you set it as a friend with SPA300FXO |
| 20:03.34 | a1fa | so when you SIP/SPA300FXO/ you can pass more things to it |
| 20:03.37 | carl0s- | quid2478: the sip.conf developers docs says that doing this means incoming calls will go to that extension, rather than the default 's' extension. I think. |
| 20:04.18 | [TK]D-Fender | a1fa : Yup, just like any SIP provider. SPA register's to * and thats that. |
| 20:04.45 | a1fa | [TK]D-Fender : lol.. broadvoice is set as peer in my settings |
| 20:05.05 | quid2478 | carlos: Okay, rings a bell now.... trying to remmeber everything from when I briefly used * back in '04 |
| 20:05.10 | [TK]D-Fender | a1fa : thats because their INBOUND stuff comes in under a different authorization. Kinda screwed up.... |
| 20:05.25 | a1fa | [TK]D-Fender : they are idiots |
| 20:05.29 | [TK]D-Fender | a1fa : Proxy BS |
| 20:05.33 | a1fa | [TK]D-Fender : yeah |
| 20:05.35 | *** part/#asterisk arcy (n=arcanum@ppp139-238.adsl.forthnet.gr) |
| 20:05.38 | a1fa | cool |
| 20:05.44 | a1fa | i am going to order this today |
| 20:05.47 | a1fa | and play with it |
| 20:05.48 | [TK]D-Fender | a1fa : BV is so flakey Tony The Tiger should do their ads.... |
| 20:06.01 | a1fa | i think thats the reason why they dont have them |
| 20:06.08 | sevard | your worst joke yet, TK. |
| 20:06.20 | [TK]D-Fender | sevard : My depravety knows no bounds! ;) |
| 20:06.24 | sevard | :\ |
| 20:06.41 | [TK]D-Fender | sevard : Direct proporions to my creativity ;) |
| 20:06.58 | carl0s- | I've got a packet-sniffer running on the Asterisk box now but it isn't showing me anything more than what 'sip debug ip' was showing me. I hoped to be able to see the password which was being sent. :( |
| 20:07.01 | sevard | or use of capitalization. |
| 20:07.22 | *** join/#asterisk tenlet (n=tenlet@pool-141-153-216-29.mad.east.verizon.net) |
| 20:07.29 | [TK]D-Fender | sevard : WhAt ArE yOu TaLkInG AbOuT?!?! |
| 20:07.36 | sevard | exactly :) |
| 20:07.47 | Damin | carl0s-: That is because they are sent as MD5 encrypted hashes.. |
| 20:08.15 | a1fa | [TK]D-Fender : what is the difference between SPA-3000 and SPA-3100? |
| 20:08.21 | [TK]D-Fender | sevard : my use of capitalization is typically appropriate and substitutes use of bold where emphasis is deserved... |
| 20:08.22 | sevard | <[TK]D-Fender> here's something that should be SAID with a TINY bit of inflection, BUT I'M GOING to RUN IT IN ALL CAPS if that's cool with you GUYS. |
| 20:08.53 | [TK]D-Fender | a1fa : Don't know the 3100... the 3102 is just and upgraded version with a bigger CPU and more RAM. Should provide a better upgrade path./ |
| 20:08.56 | sevard | [TK]D-Fender: you know that there is a bold bit you can throw infront of your text in your client, it's just as annoying as caps :) |
| 20:09.10 | carl0s- | Damin: ah. |
| 20:09.20 | masonf | when you include a context is priority reset? |
| 20:09.44 | a1fa | [TK]D-Fender : SPA-3000 comes with 2 FXS ports and 1 FXO? |
| 20:10.06 | [TK]D-Fender | sevard : All caps either means "yes I'm yelling at you in order to penetrate that thick skull of yours" or "I work in this dumb AS/400 all day and forget sometimes that I'm in caps-lock so if I slip up on occasion, just STFU" |
| 20:10.08 | [TK]D-Fender | ... ;) |
| 20:10.09 | a1fa | anyway there is a two port FXO? |
| 20:10.19 | [TK]D-Fender | a1fa : not by them |
| 20:10.26 | a1fa | anybody else |
| 20:10.30 | a1fa | that can work with * |
| 20:10.36 | file | yeekz, what did I miss... |
| 20:10.44 | sevard | [TK]D-Fender: i was going ot ask what pos you work on, -- why do you work on that? ddoesn't it kill your brian cells if you use caps all day? |
| 20:10.56 | carl0s- | The register string says "REGISTER sip:192.168.253.15 SIP/2.0.". Does this mean it's trying to register as username=192.168.253.15. The From: and To: lines say "sip:useip@192.168.253.15". I had been taking this to assume the username should be 'useip' |
| 20:11.16 | [TK]D-Fender | sevard : AND YES i KNOW HOW TO GET SOMEONES ATTENTION colourfully as well... |
| 20:11.32 | sevard | I swear. If I was next to you I'd punch you. |
| 20:11.41 | Qwell[] | colourfully? |
| 20:11.44 | file | lol |
| 20:12.08 | [TK]D-Fender | sevard : My company runs JD Edwards in 80x25 on an AS/400 using 5250 emulation. It sucks, but gets its job done so i could care less. |
| 20:12.26 | sevard | i don't mind 80x25, but all caps :| |
| 20:12.28 | [TK]D-Fender | sevard : With my frame I dare you to try :D |
| 20:12.50 | sevard | Your fat will swallow my fist? good defense |
| 20:12.56 | carl0s- | [cartman]: "I'm not fat. I'm big boned" |
| 20:13.24 | sevard | [kyle]: "Then you have a big bone in your ass, fatass" |
| 20:13.26 | [TK]D-Fender | sevard : lol... nope. 6'2" 195lbs and now training with swords :) |
| 20:13.31 | eKo1 | lol |
| 20:13.34 | carl0s- | sevard: LOL. :D |
| 20:13.35 | Qwell[] | swords...pfft |
| 20:13.41 | sevard | swords? |
| 20:13.53 | sevard | "When you die can I give your knife to me sister?" |
| 20:14.32 | sevard | come on people, nobody catches that reference? |
| 20:14.32 | sevard | sad. |
| 20:15.15 | [TK]D-Fender | Qwell : Seen my gallery of it? |
| 20:15.20 | Qwell[] | no |
| 20:15.27 | Qwell[] | /msg Qwell |
| 20:15.27 | eKo1 | sevard: eh no |
| 20:15.33 | sevard | sweet! my gf just scored a butt load of potery from the university off the termination cart. |
| 20:15.43 | a1fa | sevard : pathetic |
| 20:15.46 | sevard | eKo1: 13th Warrior, it's an awesome movie |
| 20:16.05 | sevard | how is free potery pathetic? |
| 20:16.11 | eKo1 | sevard: is that with antonio banderas? |
| 20:16.17 | carl0s- | i thought you said poetry |
| 20:16.21 | sevard | eKo1: hells yeah |
| 20:16.24 | sevard | carl0s-: haha |
| 20:16.32 | sevard | free poetry, just go to livejournal |
| 20:16.33 | eKo1 | sevard: that movie sucks bro |
| 20:16.37 | sevard | bring your razor blades |
| 20:16.48 | sevard | eKo1: dude, that's one of the best movies ever made |
| 20:17.15 | [TK]D-Fender | Qwell :http://forums.swordforum.com/showthread.php?s=&threadid=67812&highlight=bushi+pics |
| 20:17.21 | sevard | eKo1: you come to my house, i'll get a couple kegs and bring the guys and we'll all watch 13th warrior, then after we're all liqoured up go out in the woods and hunt animals |
| 20:17.22 | *** join/#asterisk chorlick (n=chorlick@gateway.digium.com) |
| 20:17.24 | eKo1 | sevard: hahahaah |
| 20:17.45 | sevard | with sharpened sticks |
| 20:17.51 | eKo1 | I'd rather hunt chicks personally. |
| 20:18.10 | sevard | that's what you're doing when you're not watching 13th warrior! |
| 20:18.13 | sevard | duh! |
| 20:19.24 | sevard | stupid script |
| 20:19.37 | x86 | sevard: check your messages ;) |
| 20:19.49 | sevard | w/indow size 4 |
| 20:19.54 | sevard | i'm having issues today. |
| 20:20.05 | [TK]D-Fender | sevard : www.drphil.com |
| 20:20.28 | sevard | [TK]D-Fender: if i ever met dr. phil i don't think i could restrain my fingers from interlocking around his neck. |
| 20:20.35 | sevard | ^personal opinion people. |
| 20:21.06 | [TK]D-Fender | sevard : He's HUGE you know... I'd bet he'd total you... |
| 20:21.38 | sevard | wait dude, you have no idea what size i am / how strong i am, don't make judgements now. |
| 20:21.42 | sevard | :) |
| 20:21.54 | x86 | sevard: ? |
| 20:22.13 | sevard | in any case i think most of that mass he has under the clothes are vaginas, it's a theory |
| 20:22.17 | x86 | [TK]D-Fender: eh, you're saying you know him like THAT ? |
| 20:22.26 | x86 | [TK]D-Fender: i didnt think dr. phil was your type |
| 20:22.27 | sevard | x86: check YOUR (in the imortal words of [TK]D-Fender) messages. |
| 20:22.40 | sevard | hahaha |
| 20:22.54 | x86 | sevard: eh? |
| 20:23.01 | x86 | sevard: no messages from you ;) |
| 20:23.07 | x86 | sevard: are you identified to nickserv? |
| 20:23.19 | sevard | i _was_ |
| 20:23.39 | [TK]D-Fender | sevard : My words huh.... definately not my capitalization you CHEAP KNOCKOFF! ;) |
| 20:24.10 | sevard | ONE FO TWO DOLLA OR TWO FO FOUR! |
| 20:24.14 | [TK]D-Fender | x86 : Not a bad jab there... |
| 20:25.02 | [TK]D-Fender | sevard : I'm sure that be more like "FO-A". The U is silent like the P in swimming :) |
| 20:25.31 | sevard | has anyone seen Jesus is Magic |
| 20:26.04 | sevard | the movie kind of sucks except for a few songs. the best scene in the movie is with the tear drop. |
| 20:26.54 | frenzy | What does the Send Flash event in grandstream do ? |
| 20:26.57 | *** join/#asterisk Gunnar (n=gunnar@34.80-203-209.nextgentel.com) |
| 20:27.01 | sevard | sends a flash? |
| 20:27.09 | sevard | ^ guess. |
| 20:27.21 | frenzy | I tried to turn it on... but then flash no longer works.. |
| 20:28.14 | frenzy | the problem I face is that asterisk is far and many time I hit the hang up for a second but instead of completing hangup it simply gives me a new line... |
| 20:29.02 | eKo1 | frenzy: that happens to many folks. turn off the feature |
| 20:29.05 | sevard | if you hit the hangup for a second that's a flash. |
| 20:29.24 | *** join/#asterisk trelane_ (n=trelane@pdpc/supporter/sustaining/trelane) |
| 20:29.25 | frenzy | it gets really annoying |
| 20:29.36 | frenzy | the other line goes to MOH |
| 20:29.50 | frenzy | and I get billed for hours :( |
| 20:30.10 | eKo1 | turn off the feature |
| 20:30.26 | [TK]D-Fender | BBIAB |
| 20:30.35 | frenzy | If I turn on the Feature Flash doesnt work :) |
| 20:30.43 | frenzy | if I turn it off Flash works |
| 20:31.04 | frenzy | what I need is ONLY to disable flash for the hang up |
| 20:32.08 | eKo1 | that doesn't exist |
| 20:33.00 | frenzy | what does Onhook Threshold: mean? |
| 20:34.38 | Beirdo | WTF? |
| 20:34.51 | Beirdo | I think my crappy ISP just started filtering IAX traffic |
| 20:35.13 | frenzy | where do you reside? |
| 20:35.14 | quid2478 | What 3rd world ocuntry do you live in? |
| 20:35.16 | frenzy | UAE |
| 20:35.18 | *** join/#asterisk paolob (n=donpaolo@pri-214-b7.codetel.net.do) |
| 20:35.21 | Beirdo | Puerto Rico |
| 20:35.28 | frenzy | LOL |
| 20:35.29 | *** join/#asterisk brimstone (n=brimston@pdpc/sponsor/digium/brimstone) |
| 20:35.34 | quid2478 | Hmm... yup, 3rd world. |
| 20:35.35 | paolob | Guys, where do I change the log level in asterisk? thank you! |
| 20:35.36 | quid2478 | haha |
| 20:35.40 | Beirdo | this cable company is frigging incompetent |
| 20:35.55 | Beirdo | quid2478, Rogers ain't much better. |
| 20:36.01 | Beirdo | I used to live in Toronto. |
| 20:36.02 | quid2478 | Beirdo: Touche there. |
| 20:36.24 | quid2478 | They are big time filtering BT traffic... well uh.. "shaping it". |
| 20:36.31 | Beirdo | heh |
| 20:36.34 | Beirdo | can't blame them for that |
| 20:36.44 | Beirdo | but JEEZ. |
| 20:36.48 | Beirdo | I get no response to UDP/4569 |
| 20:36.49 | quid2478 | Yeah, but if they give you 60 Gigs a month... aren't you free to do withit as you like |
| 20:37.13 | TrixVox | obviously not |
| 20:37.52 | eKo1 | that is typical because the ISP believes that people's bandwidth is spent on web browsing and not downloading |
| 20:38.05 | eKo1 | which is wrong but anywho... |
| 20:38.14 | nestar | most people only surf the web |
| 20:38.23 | nestar | people who download a ton cut into the margains |
| 20:38.34 | nestar | it's better to just ask them to leave, and make the numbers go up. |
| 20:38.35 | a1fa | anybody know on laws regarding voip in other countries |
| 20:38.46 | eKo1 | nestar: I doubt that. Most people who surf download as well. |
| 20:38.46 | a1fa | using DIDs outside the originating country? |
| 20:39.22 | Beirdo | and this HAS to be recent as I called out two days ago |
| 20:39.33 | nestar | eKo1: our average DSL customer transfers less than 1gig per month |
| 20:39.41 | nestar | sure, there's some downloading there |
| 20:39.43 | nestar | but not much |
| 20:40.01 | Beirdo | let's see if they changed the DHCP AGAIN |
| 20:40.03 | nestar | i can watch 500mb worth of youtube a day, if i have nothing better to do. |
| 20:40.07 | Synyn_ | nestar: wow, low usage, I'm about 4-5gb / day |
| 20:40.07 | eKo1 | nestar: and what is the typical bandwidth bought by these customers? |
| 20:40.18 | Beirdo | oh WTF? |
| 20:40.26 | nestar | it's unmetered.. either 1.5/256k or 3.5/384 |
| 20:40.41 | Beirdo | my NAT is the problem, it's still using old IP |
| 20:40.42 | Beirdo | hehe |
| 20:40.53 | eKo1 | wow, your customers must by old retired folks that just check their e-mail |
| 20:40.56 | nestar | my usage for my personal dsl last month |
| 20:40.57 | nestar | usage -p loudsl01.4.0.1.185 |
| 20:40.57 | nestar | Total dsl transfers = 2146.25 megabytes |
| 20:41.12 | *** part/#asterisk paolob (n=donpaolo@pri-214-b7.codetel.net.do) |
| 20:41.16 | nestar | my colo box moved 250gb though. ;) |
| 20:41.38 | *** join/#asterisk pa (n=Paolo@unaffiliated/pa) |
| 20:41.41 | Beirdo | OK, it wasn't the ISP's fault for filtering |
| 20:41.46 | Beirdo | it was my firewall not flushing the states when the IP changed |
| 20:42.27 | *** join/#asterisk gbodemantv (n=gbodeman@216.142.38.154) |
| 20:43.00 | Beirdo | did a pfctl -F state, and poof, there it goes again :) |
| 20:43.02 | *** part/#asterisk mog (i=ejabberd@68.62.237.103) |
| 20:43.14 | *** join/#asterisk websae (n=websae@adsl-64-149-206-121.dsl.milwwi.sbcglobal.net) |
| 20:43.25 | nestar | eKo1: for what it's worth, back in the day before we had a DS3 to bell's ATM cloud... we had t1's.. we could oversell 75-90 1.5mb dsl customers on one t1 before we got any speed complaints. |
| 20:43.42 | nestar | we were up to 6 t1's, and then moved everyone to a single DS3 |
| 20:43.47 | Beirdo | nestar: obviously I wasn't one of em |
| 20:44.10 | *** join/#asterisk topping (n=topping@adsl-68-122-71-30.dsl.pltn13.pacbell.net) |
| 20:44.26 | Beirdo | eek, my NFS connection didn't like that flush much |
| 20:45.11 | nestar | example of real-world usage... we have ~1300 dsl customers on a single ATM DS3 to Bell... |
| 20:45.25 | nestar | our average transfer is 18mbit |
| 20:45.29 | nestar | max is 32 |
| 20:45.54 | clyrrad1 | is there an asterisk command that will let you get the extension numbers for members of a given queue? |
| 20:46.02 | nestar | take 18 megabits and divide by 1300 customers |
| 20:46.44 | eKo1 | My ISP here as about 15 Megabits divided by 600 or so customers. |
| 20:46.49 | nestar | clyrrad1: show queue queue-name |
| 20:46.56 | *** join/#asterisk paolob (n=donpaolo@pri-214-b7.codetel.net.do) |
| 20:46.59 | eKo1 | Of course, we sell dedicated connections. |
| 20:47.02 | clyrrad1 | I mean in the dialplan |
| 20:47.04 | clyrrad1 | not on the CLI |
| 20:47.17 | nestar | clyrrad1: what's the difference |
| 20:47.23 | nestar | oh |
| 20:47.24 | nestar | i see |
| 20:47.27 | clyrrad1 | the difference is in the dial plan I can code and the CLI i cant |
| 20:47.59 | nestar | you're wanting to be able to have the extensions read to you over the phone? |
| 20:48.16 | nestar | you'd have to extract the date from the manager, i guess, and then use a AGI to read it out |
| 20:48.23 | clyrrad1 | nope - It is for call forwarding |
| 20:48.49 | clyrrad1 | so if a queue member has their phone forwarded - it will forward the call out the PSTN instead of ringing thier SIP line |
| 20:48.58 | clyrrad1 | but first I need to know what members are part of the queue |
| 20:49.19 | nestar | shouldn't asterisk handle that directly? IE, the phone sends the redirect, asterisk makes the call |
| 20:49.35 | *** join/#asterisk s0lid (n=s0lid@124.6.176.99) |
| 20:49.37 | nestar | just like dialing a phone directly, and it redirecting to a FWD'ed number |
| 20:49.43 | clyrrad1 | it does if you dial their extension directly - but not if you reach them via a queue |
| 20:50.53 | nestar | not sure then |
| 20:51.01 | clyrrad1 | so before i call Queue() - I would like to be able to detect if the member has forwarded or not |
| 20:51.17 | clyrrad1 | ideally you could code it in queues.conf - but as we know that just wont work |
| 20:51.28 | nestar | dialplan has no concept of who's in the queue before it gets to queue() |
| 20:51.29 | *** join/#asterisk l-fy (n=pchitesc@yate/developer/l-fy) |
| 20:51.46 | nestar | i'm not sure. i told my call people that they'd die a fiery death if they forwarded their phones |
| 20:51.48 | clyrrad1 | yea - thats what I was wondering if there was an appliation or something that did that |
| 20:51.51 | *** join/#asterisk frenzy (n=frenzy@196.46.104.184) |
| 20:51.59 | clyrrad1 | lol |
| 20:52.06 | nestar | besides, none of my employees want to forward their calls, they all hate the calls. :D |
| 20:52.10 | clyrrad1 | these people want that functionality |
| 20:52.11 | nestar | sorry, i'm rambling |
| 20:52.14 | clyrrad1 | so i need to come to some solution |
| 20:52.27 | frenzy | this sounds really stupid... i cant get three way calling to work... |
| 20:52.31 | clyrrad1 | just not sure how to go about it |
| 20:52.59 | frenzy | Clicked flash dailed the other party clicked flash again but the calls didnt get bridged... |
| 20:53.45 | masonf | exten => s,1,Set(GROUP=${IF($[ ${ARG1} = line1]?${GROUP1}); sets group to GROUP=¿p+(") any ideas? |
| 20:54.22 | frenzy | ? |
| 20:55.27 | *** join/#asterisk moon06 (n=michael@cim06-1-82-228-240-97.fbx.proxad.net) |
| 20:55.31 | frenzy | do I need canreinvite enabled to do 3 way calling ? |
| 20:56.05 | nestar | don't think so.. |
| 20:56.06 | *** join/#asterisk Alric (n=nbowyer@masq.hyperusa.com) |
| 20:56.17 | nestar | come to think of it, i don't think i've ever done 3way |
| 20:56.23 | nestar | my wife is not a fan. |
| 20:56.39 | Cresl1n | ugh |
| 20:56.45 | Cresl1n | nestar: you had to go there |
| 20:56.50 | Corydon-w | Would your wife be a fan if it was another guy? |
| 20:56.57 | frenzy | hahahhaha |
| 20:57.13 | frenzy | record that call nestar :P |
| 20:58.04 | frenzy | is there any specific setting required for 3 way calling ? |
| 20:58.15 | *** part/#asterisk murf (n=steve_mu@216.166.159.235) |
| 20:59.17 | *** join/#asterisk Dr-Linux (n=Linux@202.59.73.131) |
| 20:59.22 | *** join/#asterisk murf (n=steve_mu@216.166.159.235) |
| 21:01.17 | carl0s- | HMMMMmmmmm. |
| 21:01.21 | carl0s- | I think it's all working now. |
| 21:01.57 | carl0s- | No more 404 Not Found messages, and the GSM VoIP device says "Registered" instead of "Not registered". Strange that it would still 'mostly' work even when it said Not registered. |
| 21:02.34 | c4t3l | anyone here ever use hylafax? |
| 21:02.35 | *** join/#asterisk JackEStorm (n=thinkthi@68.225.72.125) |
| 21:02.40 | nestar | Cresl1n: sorry. |
| 21:02.41 | nestar | :) |
| 21:02.56 | c4t3l | hylafax and iaxmodem |
| 21:03.53 | paolob | Guys, what is the meaning of the skinny.conf file? do I need it if I only use ATAs to communicate with telcos and phones? |
| 21:04.38 | carl0s- | Skinny is the Cisco SCCP protocol, so the answer is probably NO. |
| 21:05.19 | *** join/#asterisk anonymouz666 (n=anonymou@200.218.193.6) |
| 21:07.15 | *** part/#asterisk a1fa (n=a1fa@207.210.210.202) |
| 21:07.46 | carl0s- | I notice a good half-second perhaps longer delay on my calls which are coming : caller -> sipgate.co.uk -> asterisk -> gsm-gateway -> cellphone. Is this acceptable? |
| 21:09.11 | nortex | Has anyone heard a possible date for the new Polycom firmware? |
| 21:09.40 | *** part/#asterisk Samoied (n=Samoied@ip47092.static.poa.virtua.com.br) |
| 21:14.35 | *** join/#asterisk pa (n=Paolo@unaffiliated/pa) |
| 21:17.17 | *** part/#asterisk mkrufky (n=mk@68.160.103.77) |
| 21:19.23 | *** join/#asterisk xenoterracide (n=xenoterr@69.89.98.120) |
| 21:20.05 | *** join/#asterisk jake1932 (n=Administ@pool-68-163-61-240.phil.east.verizon.net) |
| 21:21.12 | jake1932 | any way to get a value from a specified channel (not the current one) using a function or app? |
| 21:21.43 | *** part/#asterisk xenoterracide (n=xenoterr@69.89.98.120) |
| 21:22.13 | eKo1 | jake1932: use the manager api |
| 21:22.47 | jake1932 | eKo1: tnx - i know i could do that - but wanted to know if there was a function or app |
| 21:23.08 | carl0s- | That's so cool. Just been chatting to the girlfriend. She called me on a local-rate number (0845), and it came though Asterisk from Sipgate, then out via Orange over to my cellphone. Volume was a bit quiet but other than that it was great! |
| 21:23.32 | eKo1 | jake1932: what info are you looking for? |
| 21:23.43 | jake1932 | caller id |
| 21:24.29 | *** part/#asterisk TripleFFFF (n=TripleFF@147-102.mc.cite.net) |
| 21:24.43 | jake1932 | "features" spawns a new channel which has BRIDGEDPEER populated, so I have the chan name |
| 21:25.07 | jake1932 | just need to get a value from it |
| 21:25.39 | *** join/#asterisk MatsK (i=MatsK@83.233.97.229) |
| 21:26.08 | eKo1 | jake1932: you can get that info from your dialplan with the appropriate variables. |
| 21:26.34 | jake1932 | eKo1: nope |
| 21:27.01 | jake1932 | eKo1: when you use "features", it makes a new chan |
| 21:27.27 | eKo1 | jake1932: use the manager api then |
| 21:27.37 | eKo1 | there is no app that will tell you as far as I know. |
| 21:28.28 | carl0s- | Is it a bad idea for me to forbat this Trixbox machine and try the unreleased Asterisk-1.4? Or should I wait 'til it's released and more importantly until I know what I'm doing? |
| 21:29.04 | jake1932 | carl0s-: are you relying on said box? |
| 21:29.54 | eKo1 | carl0s-: Do it because ti is the only way you're going to learn. |
| 21:30.01 | carl0s- | jake1932: no but it would be detrimental to my health if I couldn't get anything at all to work. (I'd end up with a permanent frown worse than that which I developed today). |
| 21:30.52 | carl0s- | eKo1: Do it, as in... ditch Trixbox and install from source. What about using the 1.4 code? |
| 21:31.37 | jake1932 | carl0s-: if you can, get another crap box to play with. this way you have the best of both worlds |
| 21:32.20 | jake1932 | i run one of my installs on an old P3 (can't be worth more than $50 |
| 21:32.28 | carl0s- | I have one computer to the left of this desk (small-ish) and two big-ish towers to the right. My mum wouldn't be happy about a third. I'm expanding outwardly-right. |
| 21:32.56 | *** join/#asterisk marv0997 (i=marv0997@190.4.2.86) |
| 21:33.01 | carl0s- | (it's her living room :D ) |
| 21:33.06 | jake1932 | or play with a hosted one |
| 21:33.14 | eKo1 | carl0s-: install * with trixbox |
| 21:33.50 | carl0s- | eKo1: what, install a second * installation on my existing trixbox installation? ? |
| 21:33.59 | eKo1 | carl0s-: yes |
| 21:34.27 | carl0s- | eKo1: I'd have to ditch the existing * on there though wouldn't I.. then it's no better than starting with FC5 or something is it? |
| 21:34.37 | *** part/#asterisk jake1932 (n=Administ@pool-68-163-61-240.phil.east.verizon.net) |
| 21:35.18 | eKo1 | just have two * installs |
| 21:35.20 | eKo1 | no big deal |
| 21:35.59 | l-fy | hi eKo1 |
| 21:36.04 | *** join/#asterisk paolob (n=donpaolo@pri-214-b7.codetel.net.do) |
| 21:36.11 | carl0s- | hmm. I'll have to read up on the ./configure options or whatever. It's going to need to use a different /etc2 or at least and /etc/asterisk2 and /var/log/asterisk2 isn't it? and /var/spool/asterisk2 |
| 21:36.12 | brad6254 | What is the best way to find out why the rtp packets aren't getting from a zap channel to a sip phone on our lan |
| 21:37.36 | *** join/#asterisk malcolmd (n=malcolmd@pdpc/sponsor/digium/malcolmd) |
| 21:38.02 | *** part/#asterisk brad6254 (n=brad6254@pool-71-162-32-182.altnpa.east.verizon.net) |
| 21:38.09 | eKo1 | carl0s-: no, just change the install prefix in the make file |
| 21:38.23 | eKo1 | pretty simple. I have asterisk 1.0 and 1.2 on my dev box |
| 21:40.38 | *** join/#asterisk cybergypsy (n=mark@90.5.62.66) |
| 21:41.25 | carl0s- | eKo1: great. thanks |
| 21:43.08 | paolob | guys, anyone knows anything about dlynes? |
| 21:43.46 | carl0s- | grr. having just had that sucessful call. I'm now getting "All circuits are busy" on outbound GSM calls. The last entry in the log relating to this trunk says: Device 'SIP/103' changed to state '5' (Unavailable). Why might that happen? |
| 21:45.32 | *** join/#asterisk juanjoc (n=juanjoc@200.73.189.82) |
| 21:45.40 | *** join/#asterisk Azrael (n=Azrael@orion.negativeblue.com) |
| 21:46.24 | *** join/#asterisk denon (i=denon@synapse.subneural.net) |
| 21:46.24 | *** mode/#Asterisk [+o denon] by ChanServ |
| 21:56.01 | *** join/#asterisk s0lid (n=s0lid@124.6.176.100) |
| 22:05.33 | carl0s- | is there a parameter I can give to 'svn' to make it a little bit more verbose. It doesn't say jack for the first five minutes. |
| 22:07.09 | eKo1 | which svn command? |
| 22:07.33 | carl0s- | svn checkout http://svn.digium.com/svn/asterisk/trunk asterisk |
| 22:07.54 | carl0s- | just subversion.. not asterisk related at all really |
| 22:08.15 | carl0s- | man svn is equally quiet. |
| 22:08.19 | eKo1 | do svn help checkout and look at the options |
| 22:08.24 | carl0s- | ah |
| 22:08.28 | carl0s- | thx |
| 22:09.47 | quid2478 | anyone using axvoice and able to get * to direct the call from that particular DID to a particular destination? |
| 22:10.06 | carl0s- | If I'm builing asterisk-1.4 to run side-by-side with 1.2.9, do I need to build the newer zaptel as well? |
| 22:10.32 | quid2478 | while I can register no problem, it seems incoming calls have a slightly different IP |
| 22:10.57 | *** join/#asterisk tempest1 (n=asf@adsl-144-36-60.chs.bellsouth.net) |
| 22:12.59 | *** part/#asterisk c4t3l (n=c4t3l@69.15.174.114) |
| 22:25.33 | carl0s- | runbning configure for latest svn of asterisk 1.4: configure:4904: error: C++ preprocessor "/lib/cpp" |
| 22:25.40 | carl0s- | fails sanity check |
| 22:26.49 | *** join/#asterisk SarahEmm (n=sarahemm@MTL-HSE-ppp159791.qc.sympatico.ca) |
| 22:31.57 | Synyn_ | how can I make my solaris * connect to another comp to use the fox cards for pstn dialing? Just install another * and use iax2? |
| 22:32.09 | Synyn_ | fxo* |
| 22:34.16 | SarahEmm | yep, iax2 is good stuff :) |
| 22:39.50 | l-fy | hi SarahEmm |
| 22:41.49 | SarahEmm | hihi l-fy |
| 22:45.42 | Synyn_ | carl0s: does gcc -v match cpp --version? |
| 22:47.26 | *** join/#asterisk bjohnson (n=bjohnson@i216-58-50-1.cybersurf.com) |
| 22:49.53 | *** part/#asterisk Mother (n=mother@93.Red-80-32-127.staticIP.rima-tde.net) |
| 22:50.25 | *** join/#asterisk mishkiz (n=Janus@zeus.corsidian.com.br) |
| 22:51.34 | *** join/#asterisk mhnoyes (n=mhnoyes@dialup-4.246.232.6.Dial1.SanJose1.Level3.net) |
| 22:51.55 | mishkiz | hello all.....when I do a "zap show status"on CLI I get IRQ=0....In lspci I get IRQ=15...that is normal ? |
| 22:55.27 | carl0s- | mishkiz: yep. irq always shows 0 for me in zap show status. |
| 22:55.40 | *** join/#asterisk DrkShdw (n=DrkShdw@unaffiliated/drkshdw) |
| 22:55.47 | carl0s- | Synyn_: sorry, missed that. It was missing package gcc-c++ |
| 22:56.15 | Synyn_ | ah |
| 22:56.28 | *** join/#asterisk DrkShdw (n=DrkShdw@unaffiliated/drkshdw) |
| 22:58.05 | *** join/#asterisk ariel_ (n=Ariel@dsl-20-177.cofs.net) |
| 22:58.33 | mishkiz | carl0s-, thanks... |
| 22:58.37 | *** join/#asterisk xbmodder_newlapp (i=nobody@atarack/staff/xbmodder) |
| 22:59.59 | mishkiz | just only one more thing.....since monday im tring to put my asterisk box to work...I'm getting the message: "Unable to create channel of type 'Zap' (cause 0 - Unknown)".....anybosy have any idea of where is the problem ? |
| 23:00.43 | mishkiz | sorry anybosy = anybody :-) |
| 23:00.50 | carl0s- | I'm afraid I don't know the answer to that one. |
| 23:01.16 | *** join/#asterisk oomph (n=oomph@69-175-194-51.chvlva.adelphia.net) |
| 23:02.13 | oomph | anyone know of an adapter similiar to the sipura 3000 that supports skype and multiple sip providers? |
| 23:04.03 | Synyn_ | oomph: isn't skype a closed system? |
| 23:04.29 | oomph | yes it is but i saw some type of USB phone adapter once |
| 23:04.35 | oomph | can't recall its name |
| 23:04.44 | oomph | that had some sort of a client |
| 23:04.48 | oomph | that logged into skype |
| 23:04.58 | carl0s- | Portech have a skype gateway device. Probably a bit ott for what you need. http://www.portech.com.tw |
| 23:05.31 | oomph | yeah that sounds right |
| 23:05.40 | oomph | http://www.cuphone.com/products/index.htm found a link to another |
| 23:06.04 | carl0s- | http://www.portech.com.tw/eweb/skytrunk/skytrunk.htm |
| 23:06.11 | carl0s- | http://www.portech.com.tw/eweb/skytrunk/st1004.htm |
| 23:06.34 | carl0s- | the 1004 is a rackmount gateway device |
| 23:06.49 | oomph | wow |
| 23:07.48 | carl0s- | it looks like it's FXO/FXS though, not SIP. |
| 23:08.17 | oomph | yeah, i wanna be able to use SIP/FXO/FXS and skype all in one |
| 23:09.17 | Synyn_ | that looks pretty cool |
| 23:09.36 | Synyn_ | but kinda sad you need a hw fix for something that should be a sw fix |
| 23:09.57 | oomph | yeah |
| 23:10.15 | carl0s- | yeah. just picked up one of their GSM VoIP gateways. The firmware is problematic though so only time will tell if they fix it and therefore whether their products should be recommended |
| 23:10.35 | Synyn_ | I see no price for it, must mean something |
| 23:10.53 | carl0s- | well... the GSM gateway was only $170 + delivery. (MV-370) |
| 23:10.57 | oomph | this cuphone seems linux friendly |
| 23:11.53 | Bobcat_1966 | ha anybody tried to upgrade to asterisk trunk and then put it back to 1. stable? |
| 23:12.14 | Bobcat_1966 | 1.2 stable |
| 23:12.47 | carl0s- | Is Asterisk 'trunk' another name for 1.4? If so then I'm just trying it now. Trying a side-by-side install on my trixbox1.1 machine, but when I try to start * it's using the /etc/asterisk/modules.conf and trying to laod the 1.2 modules and segfaulting. |
| 23:12.56 | Bobcat_1966 | I think so |
| 23:13.19 | carl0s- | I think it is because the svn url was /trunk |
| 23:13.42 | Bobcat_1966 | ya I have the same issue but when I reinstall 1.2 stable it wont start now |
| 23:13.57 | file | wipe your modules directory |
| 23:14.10 | file | rm -rf /usr/lib/asterisk/modules |
| 23:14.13 | Bobcat_1966 | its giving me alot of errors with a module called res_convert.so |
| 23:14.28 | Bobcat_1966 | got it, that will probubly do it let me try |
| 23:14.33 | Bobcat_1966 | thanks file |
| 23:14.53 | carl0s- | file: are the modules not needed? |
| 23:14.54 | file | at the end of make install it tells you about modules that shouldn't be there too ;) |
| 23:15.09 | carl0s- | ah |
| 23:15.25 | file | carl0s-: code changes, modules go away |
| 23:15.30 | carl0s- | cool |
| 23:16.43 | Bobcat_1966 | So file have you got 1.4 working? |
| 23:16.49 | carl0s- | How can I make my 'side-by-side' asterisk-1.4 *not* look for /etc/asterisk/modules.conf ? I've created a /etc/asterisk14 directory |
| 23:16.55 | Qwell[] | 1.4? |
| 23:16.57 | hads | 1.4 doesn't exist |
| 23:17.11 | Bobcat_1966 | sorry asterisk trunk |
| 23:17.21 | carl0s- | I thought the not-quite-released SVN stuff was 1.4? |
| 23:17.27 | Bobcat_1966 | some call it 1.4 |
| 23:17.34 | file | the definition of "working" can vary between person to person, depending on what they use it for |
| 23:17.34 | Qwell[] | carl0s-: no, the not-quite-release SVN stuff is SVN trunk |
| 23:17.41 | hads | I would imagine that file has got it going :) |
| 23:17.58 | Qwell[] | file IS like 4 hours in the future though |
| 23:17.58 | Bobcat_1966 | kinda sounds like it |
| 23:18.02 | Qwell[] | so he may very well have 1.4 already |
| 23:18.07 | hads | heh |
| 23:18.09 | carl0s- | :D |
| 23:18.10 | Bobcat_1966 | lol |
| 23:18.14 | file | it's true! |
| 23:18.35 | Bobcat_1966 | where you located file? |
| 23:18.41 | file | Atlantic Canada |
| 23:18.46 | SarahEmm | wooo canada |
| 23:18.49 | Qwell[] | He's a newbie |
| 23:18.58 | Bobcat_1966 | thats eastern to me right |
| 23:18.59 | *** join/#asterisk saftsack (n=saftsack@p54A7D810.dip.t-dialin.net) |
| 23:19.02 | *** join/#asterisk shido6 (n=shido6@d221-68-200.commercial.cgocable.net) |
| 23:19.03 | Qwell[] | Bobcat_1966: no |
| 23:19.06 | mishkiz | there is anyway to make a loopback cable to test a ISDN board ? |
| 23:19.09 | Qwell[] | Bobcat_1966: It's atlantic canada |
| 23:19.12 | hads | I win the time war. 2006-07-22 11:18 |
| 23:19.24 | Bobcat_1966 | ahhh so you are 1 hour ahead? |
| 23:19.32 | file | it is 8:19PM here |
| 23:19.44 | Bobcat_1966 | yep one hour ahead |
| 23:20.17 | Bobcat_1966 | no good with maps :) |
| 23:21.05 | Bobcat_1966 | so once you out the modules director do you have to rebuild just asterisk or libpri and zaptel? |
| 23:21.16 | file | make install |
| 23:21.23 | Bobcat_1966 | ok |
| 23:21.26 | carl0s- | 'trunk' will shorltly become 1.4 I presume? |
| 23:21.33 | Qwell[] | carl0s-: eventually |
| 23:21.38 | carl0s- | ok |
| 23:21.42 | Qwell[] | it will be branches to 1.4 |
| 23:21.45 | Qwell[] | branched |
| 23:21.47 | carl0s- | right |
| 23:21.55 | file | eventually! |
| 23:21.59 | Qwell[] | indeed |
| 23:22.02 | Qwell[] | when it's good and ready |
| 23:22.25 | Toerkeium | guys, how can I make asterisk detect my ethernet device which is called venet0 ? |
| 23:22.25 | carl0s- | I need to stop it looking at /etc/asterisk/modules.conf . Is this a compile time option? I gave --prefix= to ./configure. Perhaps I need to look for a --etc option and recompile again. |
| 23:22.41 | Qwell[] | Toerkeium: asterisk doesn't need to know about the interface |
| 23:23.08 | Synyn_ | do I need libpri? |
| 23:23.13 | Synyn_ | if I don't plan to use it |
| 23:23.16 | Qwell[] | Synyn_: no |
| 23:23.26 | Toerkeium | I Qwell[]: i get No ethernet interface found for seeding global EID You will have to set it manually. |
| 23:23.38 | file | oh, DUNDi |
| 23:23.40 | Qwell[] | Toerkeium: for dundi..if you don't use dundi, you can ignore it |
| 23:23.50 | Qwell[] | I get the same on my sparc |
| 23:24.01 | Toerkeium | what's DUNDi ? |
| 23:24.18 | Qwell[] | ~dundi |
| 23:24.19 | jbot | hmm... dundi is http://www.dundi.com |
| 23:24.19 | carl0s- | it's like some cool thing where phone numbers can be resolved to IP addresses |
| 23:24.36 | Toerkeium | oh, ok.. for a next stage then :) |
| 23:24.38 | Synyn_ | Qwell[]: you know what package crvs is in? gmake is complaining |
| 23:24.45 | carl0s- | ^^^ that's not valid any more.. goes to the wrong page. |
| 23:25.02 | Qwell[] | crvs? |
| 23:25.07 | Toerkeium | what abou this ? :Unable to op en pseudo channel for timing... Sound may be choppy. ? |
| 23:25.10 | file | carl0s-: what is not valid? |
| 23:25.16 | carl0s- | www.dundi.com |
| 23:25.19 | file | sure it is. |
| 23:25.20 | carl0s- | 'ave a look |
| 23:25.25 | Qwell[] | heh |
| 23:25.31 | file | already there |
| 23:25.32 | Qwell[] | yeah, that's b0rked |
| 23:25.42 | Qwell[] | file: You're internal, so it's probably different |
| 23:25.45 | carl0s- | "Purchasing Voice Prompts" |
| 23:25.46 | Qwell[] | it goes to thevoice |
| 23:25.48 | Synyn_ | Qwell[]: on solaris, trying to make * |
| 23:25.48 | file | my laptop isn't internal |
| 23:25.53 | Qwell[] | oh |
| 23:25.56 | Toerkeium | am I messing the pseudo devo ? |
| 23:25.57 | Qwell[] | well, it's b0rked here :p |
| 23:26.00 | Toerkeium | mi* |
| 23:26.03 | file | how odd |
| 23:26.04 | Qwell[] | Synyn_: What's it say exactly? |
| 23:26.14 | carl0s- | it was b0rked here a few days ago too. Someone else mentioned it. |
| 23:26.15 | saftsack | has asterisk TAPI support? |
| 23:26.21 | Bobcat_1966 | Ok file I deleted the modules directory. recompiled asterisk and its getting further but is failing on a Loading module format_mp3.so failed!....any ideas? |
| 23:26.39 | Qwell[] | Bobcat_1966: You did a load => format_mp3.so in modules.conf |
| 23:26.44 | Qwell[] | either take that out, or install asterisk-addons |
| 23:26.57 | Bobcat_1966 | never had to do that before, let me check |
| 23:27.22 | file | format_mp3 is not part of the normal asterisk checkout |
| 23:28.44 | Bobcat_1966 | hmmm when I built the box oridginally I did not do anything special for mp3...but I did look in my modules.conf file and it has this (load => format_mp3.so) |
| 23:29.19 | Qwell[] | Bobcat_1966: If you need it, install -addons |
| 23:29.43 | Bobcat_1966 | I was just going to ask that question about asterisk addons...let me try that. thaks |
| 23:29.45 | *** join/#asterisk pdtwork (n=ptinsley@209.12.249.243) |
| 23:30.04 | carl0s- | so how do I point asterisk at a different /etc/asterisk directory? I want to be able to run both versions by starting/stopping one or the other |
| 23:30.18 | rob0 | carl0s-: -C |
| 23:30.29 | rob0 | asterisk --help |
| 23:30.47 | carl0s- | -C = configfile. hmm. ok |
| 23:30.56 | Bobcat_1966 | file....that did it. Thanks Very Much |
| 23:31.04 | Qwell[] | pfft |
| 23:31.10 | file | and what did we learn today class? |
| 23:31.13 | Bullseye_Network | anyone having problems with 1.2.10 crashing and not giving errors? Last thing mine says is : Jul 21 16:26:20 VERBOSE[18731] logger.c: -- Attempting call on SIP/14789577289@cvcdial for 880590242675410006003960@rca:1 |
| 23:31.13 | Bullseye_Network | <PROTECTED> |
| 23:31.13 | Bullseye_Network | Jul 21 16:26:20 VERBOSE[18411] logger.c: -- Started music on hold, class 'default', on channel 'SIP/tsr007-b5902c08' |
| 23:31.13 | Bullseye_Network | Jul 21 16:26:21 VERBOSE[17614] logger.c: == Spawn extension (npn, 1, 6) exited non-zero on 'SIP/66.235.234.150-b7179450' |
| 23:31.14 | Bullseye_Network | Jul 21 16:26:21 VERBOSE[17695] logger.c: == End MixMonitor Recording SIP/66.235.234.150-b7179450 |
| 23:31.16 | Bullseye_Network | Jul 21 16:28:42 NOTICE[18826] cdr.c: CDR simple logging enabled. |
| 23:31.18 | Bullseye_Network | ops sorry |
| 23:31.22 | Bobcat_1966 | That Im stupid |
| 23:31.27 | rob0 | oops -h not --help |
| 23:31.27 | Bullseye_Network | didnt man to put that much |
| 23:31.29 | Bobcat_1966 | but willing to learn |
| 23:31.38 | Qwell[] | file: To thank somebody for something that somebody else answered :P |
| 23:31.38 | carl0s- | file: I learned.. lots! |
| 23:31.51 | file | I was going for, "when going between major versions wipe out the modules directory" but whatever |
| 23:32.09 | carl0s- | file: :) |
| 23:32.30 | Bobcat_1966 | so if I wanted to try trunk again, I would wipe the modules directory first? |
| 23:32.32 | carl0s- | file: I haven't forgotten that, but then when I fire up 1.2 again it'll not have its modules. can't I just tell each one to use it's own set of modules. |
| 23:32.51 | carl0s- | file: I want to be switching back and to between versions like every half hour. |
| 23:32.53 | file | Bobcat_1966: or else it'll freak. |
| 23:32.55 | hads | carl0s-: mv |
| 23:33.10 | Bobcat_1966 | cool then that is what I learned. |
| 23:33.14 | carl0s- | OK :) |
| 23:33.27 | file | carl0s-: you can tell Asterisk where to look... |
| 23:33.39 | pdtwork | I learned that queues can massively hose your PBX |
| 23:33.39 | file | it's in the asterisk.conf file |
| 23:33.45 | file | which you specify using asterisk -C |
| 23:33.57 | hads | I learned not to play with fire |
| 23:34.04 | carl0s- | thx |
| 23:34.19 | file | hads: you can play with fire... just don't expect someone here to call 911 |
| 23:34.40 | rob0 | RIGHT because E911 service is NOT PROVIDED. |
| 23:34.53 | Bobcat_1966 | I learned not to throw a cigerett between your legs while using the john. |
| 23:34.59 | *** join/#asterisk malverian (n=malveria@gentoo/developer/malverian) |
| 23:35.10 | Qwell[] | We're sorry. Your call cannot be completed as dialed. Please stop the bleeding, and try your call again. |
| 23:35.25 | Synyn_ | anyone know what the program "crvs" is that make is looking for when building asterisk ? |
| 23:35.34 | Qwell[] | Synyn_: What error are you getting? |
| 23:35.36 | hads | It would take the US emergency services a long time to get over here. |
| 23:35.53 | Qwell[] | hads: probably faster than here... |
| 23:36.09 | hads | heh |
| 23:37.02 | hads | The emergency number is 111 over here, but after all the american shows on TV etc they frwarded 911 to emergency too. |
| 23:37.09 | Qwell[] | heh |
| 23:38.09 | *** join/#asterisk Lyfe (n=lyfe@69.8.146.78) |
| 23:38.55 | carl0s- | Our cellphones accept 911 too. I'm not sure if it actually goes through to the Police (999), but they let you dial it with the keylock on. |
| 23:38.59 | rob0 | I lived a couple years (early 80's) in a small town where I got the old police dispatcher's phone number. Occasionally I got calls ... not real emergencies. It was fun taunting them. |
| 23:39.26 | file | lol |
| 23:39.42 | Lyfe | anyone familiar with a warning like this, which keeps my softphone connecting to a gxp2000: Jul 21 18:38:57 WARNING[9712]: channel.c:2341 set_format: Unable to find a codec translation path from gsm to g729 |
| 23:39.43 | rob0 | Thar's a feller parkt out front ma house, whut I orter due 'bout it? |
| 23:40.00 | Qwell[] | Lyfe: You need a license if you want to use g729 |
| 23:40.07 | Qwell[] | either tell the phones not to use it, or...get a license |
| 23:40.25 | file | please note the g729 licensing model does NOT allow you to take over the world |
| 23:40.32 | Qwell[] | file: g723?! |
| 23:40.35 | Lyfe | hrmm.. well, i don't care to use it, but ok.. i'll look into trying to get the gxp2000 to stop usin git :\ |
| 23:40.48 | file | Qwell[]: then you can take over a small coutnry |
| 23:40.48 | Qwell[] | Lyfe: easiest is to probably set disallow=g729 in sip.conf |
| 23:40.51 | file | er country |
| 23:40.53 | Qwell[] | file: excellent |
| 23:40.57 | Qwell[] | how small? Cuba? |
| 23:40.58 | Lyfe | well: thanks.. that would probably solve it. |
| 23:41.01 | rob0 | Poetic license? |
| 23:41.20 | Lyfe | sorry, qwell. (can't type today) |
| 23:41.33 | file | russellb: sandwich?!? |
| 23:41.40 | russellb | steak, actually |
| 23:41.42 | Qwell[] | mmm |
| 23:41.48 | file | I was close |
| 23:42.00 | Qwell[] | is it a steak sandwich maybe? |
| 23:42.28 | russellb | nope. |
| 23:42.32 | russellb | slab of meat on a plate. |
| 23:42.35 | Qwell[] | mmm |
| 23:43.22 | file | russellb: with... ice cream?!? |
| 23:43.32 | carl0s- | cool. trunk is working now. I like the colourful output. I notice "quit" doesn't work in the CLI though. |
| 23:43.43 | Qwell[] | carl0s-: You used -c, not -C |
| 23:43.51 | carl0s- | Qwell: I used both :) |
| 23:43.52 | Qwell[] | try again.. |
| 23:44.03 | Qwell[] | well, that's why you get the colors and the no quit |
| 23:44.05 | carl0s- | Qwell: that's where the colour came from then! I used -vvvc -C |
| 23:44.14 | carl0s- | ah |
| 23:44.16 | *** join/#asterisk tempest1 (n=asf@adsl-144-36-60.chs.bellsouth.net) |
| 23:44.39 | russellb | file: nope, but that sounds like a good idea |
| 23:44.41 | russellb | tempest1: ! |
| 23:45.07 | file | it's a trick! |
| 23:46.53 | Lyfe | thanks Qwell, problem solved. |
| 23:47.59 | file | Qwell r0x0rz |
| 23:48.06 | Qwell[] | He so does |
| 23:48.33 | rob0 | Just don't tell anyone, 'specially him. |
| 23:49.42 | Qwell[] | it goes straight to his head |
| 23:51.08 | *** join/#asterisk trbldwine (n=trbldwin@71.194.161.170) |
| 23:51.59 | anonymouz666 | Run down ghost trail, no chance of love no sign of life, just wild dogs howlin in the night! |
| 23:51.59 | *** join/#asterisk benjk (n=benjamin@f8a01-0357.din.or.jp) |
| 23:56.31 | *** join/#asterisk empiric (n=empiric@203.130.1.42) |
| 23:57.04 | empiric | Guys I have a weird problem in my dial plan? |
| 23:57.18 | empiric | everything is working fine except outbound dialing |
| 23:57.51 | empiric | anytime I dial to a pstn no. it routes my call to the a ZAP local extension |
| 23:57.55 | empiric | any ideas?? |
| 23:58.22 | SarahEmm | pastebin your dialplan :) |
| 23:58.49 | empiric | hold on a sevc |
| 23:59.05 | SarahEmm | kk |
| 23:59.06 | ariel_ | I don't understand pstn line should be connected to the Zap card |
| 23:59.13 | *** join/#asterisk MikeJ__ (n=vircuser@d14-69-8-30.try.wideopenwest.com) |
