| 00:01.19 | Vec | suma : take a look at chanspy |
| 00:03.29 | suma | Vec: let me look |
| 00:07.13 | *** join/#asterisk coppice (n=chatzill@46.166.17.210.dyn.pacific.net.hk) |
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| 00:30.49 | *** join/#asterisk ruben23 (n=AGENT@124.107.3.178) |
| 00:31.57 | ruben23 | hi anyone have ideas on this error http://pastebin.com/m1185c3c7 |
| 00:36.23 | *** join/#asterisk jaytee (n=jforde05@unaffiliated/jaytee) |
| 00:37.44 | *** part/#asterisk k-man (n=jason@unaffiliated/k-man) |
| 00:42.17 | tfrew | has your outbound route worked before? |
| 00:44.04 | ruben23 | <PROTECTED> |
| 00:44.33 | tfrew | who is your voip provider |
| 00:44.41 | ruben23 | is it a problem of voip..? |
| 00:47.11 | *** join/#asterisk hi365 (n=hi365@94.159.144.212) |
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| 01:01.35 | suma | how to limit the time in Meetme? |
| 01:03.31 | phix | hey |
| 01:03.42 | phix | not sure suma |
| 01:03.49 | phix | did you try googleing it? |
| 01:04.02 | suma | phix: i did, could not get a solution |
| 01:04.09 | phix | hmmmm |
| 01:09.36 | suma | phix, i got another problem too |
| 01:09.54 | suma | how to play a file into an existing call ? |
| 01:10.20 | suma | I tried to play with Dial, with the waning_sound |
| 01:10.41 | suma | but it plays only to either callee or caller at a time keeping the other channel silent |
| 01:13.45 | *** join/#asterisk [intra]lanman (n=lanman@freeswitch/developer/intralanman) |
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| 01:46.04 | phix | ? |
| 01:46.32 | phix | nfi |
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| 02:35.43 | kc8pxy | i have tested sip channel acceptance, and i've made a sip to sip call. I'm looking for something else i can do to challenge my config skills. all i have at the moment is sip softphones(2 of them, POSSIBLY 3).. recommendations? |
| 02:36.35 | jaytee | why not build an IVR menu? |
| 02:37.36 | *** join/#asterisk lanning (n=lanning@173.8.187.197) |
| 02:39.32 | tfrew | kc8pxy: how about this.. create a timegoup with timecondition, point that to either a ring group or an ivr based on the time |
| 02:40.34 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 02:40.42 | kc8pxy | tfrew: so make it "I'm sorry, our office is now closed" kinda thing. |
| 02:40.52 | *** join/#asterisk H3XiL3D (n=H3X@unaffiliated/hexiled) |
| 02:41.03 | tfrew | right |
| 02:41.11 | tfrew | are you using freepbx? |
| 02:41.39 | *** join/#asterisk HeXiLeD (n=H3X@unaffiliated/hexiled) |
| 02:41.55 | kc8pxy | tfrew: asterisk, 1.6.0.9. dahdi modules compuled, and avaiable, but no hardware yet. |
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| 02:47.45 | tfrew | kc8pxy: you should install freepbx |
| 02:47.52 | tfrew | makes asterisk managment alot eaiser |
| 02:48.09 | tfrew | if you compile from source, you can still modify raw asterisk without breaking anything |
| 02:48.18 | tfrew | unlike some certian distros |
| 02:51.02 | *** join/#asterisk jaytee (n=jforde05@unaffiliated/jaytee) |
| 02:55.34 | HeXiLeD | anything from source is always best |
| 02:55.43 | HeXiLeD | unlike some certian distros :) |
| 02:56.38 | kc8pxy | tfrew: freepbx is a distro? |
| 02:59.33 | tfrew | kc8pxy: no |
| 02:59.46 | tfrew | its a webserver frontend to manage asterisk's configuration |
| 03:03.11 | kc8pxy | tfrew: i don't see it in my distro's package tree. |
| 03:03.35 | tfrew | which distro are you using |
| 03:08.30 | *** join/#asterisk n3hxs (n=HAMming@65.214.117.3) |
| 03:13.01 | kc8pxy | tfrew-cigbrb: gentoo. the gentoo instructions describe asterisk 1.2-era install details. |
| 03:13.13 | kc8pxy | tfrew-cigbrb: it has to have LAMP? |
| 03:13.33 | *** join/#asterisk RypPn (i=TuMbL@rosscom.demon.co.uk) |
| 03:13.33 | HeXiLeD | gentoo :) |
| 03:15.25 | tfrew | kc8pxy: yes |
| 03:15.27 | *** join/#asterisk jgoo (n=eva@ppp101-241.adsl.forthnet.gr) |
| 03:15.35 | tfrew | you should move to asterisk 1.4 |
| 03:15.45 | tfrew | for daily use |
| 03:15.48 | tfrew | 1.2 is outdated |
| 03:16.11 | tfrew | 1.6 is techniaclly stable but i wouldn't touch it yet till the big distro's use it as the default |
| 03:16.25 | jgoo | Hi, I am using a PAPT2, several in fact - they register fine with asterisk, I see them registered, but when I dial the extension of one (121) from the other (122) it doesn't ring |
| 03:16.40 | jgoo | If they are both registered, what could be wrong? |
| 03:17.40 | jgoo | If I try 9 to outside, I get all circuits busy (correct, I have unplugged the ISDN from the card right now) |
| 03:17.55 | jgoo | so it gets the outside, but internal calls don't go through |
| 03:21.02 | rob327 | i am new to asterisk, could anyone point me in the direction of a good tutorial on setting up an asterisk+vicidial cluster/load balancing setup? |
| 03:28.37 | rob327 | or any kind of resource on the subject :P |
| 03:34.08 | jgoo | using the panel, if I drag an extension to another, the first phone rings, on pickup, it dials the second, but I can't just dial the extension, any ideas? |
| 03:41.27 | *** join/#asterisk triarius (n=vinnix@200-161-163-223.dsl.telesp.net.br) |
| 03:42.06 | triarius | hi people, I would to know if is possible configure an asterisk for test just with a single modem? |
| 03:43.36 | jblack | modem? |
| 03:43.41 | jblack | Check out voip. |
| 03:43.59 | tfrew | triarius: callcentric.com |
| 03:44.00 | jblack | for about $5 and a headset, you can fool around plenty. |
| 03:45.27 | tfrew | triaruis: you can call callcentric for free with your free callcentric 1-777-***-**** number |
| 03:45.30 | tfrew | if that is working |
| 03:45.44 | tfrew | you can get a pay per minute local number for like $5 |
| 03:46.43 | jblack | I'd go with ipkall and callwithus. ipkall will give you a washington state number for free, and callwithus has about $0.012/min calls in 1 second increments. |
| 03:47.06 | triarius | great, with this, I can mount a URA test? or test others features of asterisk? |
| 03:47.16 | jblack | yup. |
| 03:47.25 | kc8pxy | tfrew: why not 1.6? do you simply have a higher |
| 03:47.50 | kc8pxy | "tried and true" standard than the release team? |
| 03:48.05 | tfrew | only reason for skipping 1.6 right now |
| 03:48.27 | tfrew | is the distro's that build around asterisk aren't releasing 1.6 as the default setup |
| 03:48.36 | tfrew | they are staying on 1.4 for right now |
| 03:49.35 | jblack | Patience. 1.6 will roll out soon enough. The dahdi changes probably roiled things up. |
| 03:50.42 | jblack | checks jaunty |
| 03:51.26 | kc8pxy | tfrew: i had to do a few things to get dahdi installed. but i have dahdi modules, and asterisk with dahdi support.. just not any dahdi cards (yet) |
| 03:51.41 | jblack | Jaunty is at 1.4.21.2 |
| 03:52.29 | kc8pxy | only installs hardy right now. when it comes to ubuntu. |
| 03:52.40 | tfrew | lts for the win |
| 03:52.51 | tfrew | but centos > ubuntu |
| 03:53.17 | jblack | Are you serious? |
| 03:53.29 | tfrew | yes |
| 03:53.32 | kc8pxy | tfrew: in have a hate-hate relationship with anything based on "dead-rat".. we'll leave it at that. |
| 03:53.50 | *** join/#asterisk CunningPike (n=CunningP@S01060014bf81366b.vc.shawcable.net) |
| 03:53.54 | jblack | now knows not to seek tfrew's opinion. |
| 03:53.55 | tfrew | kc8pxy: the you have a hate-hate relationship with getting a job |
| 03:54.09 | *** join/#asterisk SkywaIker (n=pirch@58.147.17.166) |
| 03:55.10 | kc8pxy | tfrew: i don't plan to work for anyone running cent, fedora, or any of that band. i got a bad taste with rpm's early on, and yum has only partly cleared the ire. |
| 03:55.21 | tfrew | meh |
| 03:55.35 | tfrew | thats all the company i work for runs |
| 03:55.41 | jblack | I honestly don't know of any serious admins that intentionally deploy centos in production. |
| 03:55.59 | tfrew | the engineer i work under does just that |
| 03:56.09 | jblack | I come across it from time to time, as that troublesome box that has to be redeployed, typically with ubuntu. |
| 03:56.13 | tfrew | everything at the datacenter minus one windows 2003 server |
| 03:56.37 | jblack | Some admin starting of puts it up, doesn't maintain it well, ends up in a big mess. |
| 03:56.38 | rob327 | it is good to know rpm based distros |
| 03:57.11 | rob327 | i prefer dpkg to rpm, but it's served me well being able to maintain either type of system |
| 03:57.49 | tfrew | it's good to know the difference |
| 03:58.02 | tfrew | because " apt-get update " and " yum update " are completly different |
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| 03:58.12 | tfrew | one of those could break your system |
| 03:58.39 | kc8pxy | all but swears off (and occasionally at) binary-package-based distros. i use ubuntu @ the shop because it's simpler for new converts to linux. |
| 03:58.41 | tfrew | unfourtunatly its yum that can break your system |
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| 04:00.00 | kc8pxy | tfrew: yeah, if apt-get update breaks your system, you're in a whole heap of trouble. |
| 04:00.31 | tfrew | there is no way |
| 04:01.20 | tfrew | the only thing that command does is update apt-get's repository caches |
| 04:02.45 | kc8pxy | tfrew: no space left on device |
| 04:04.17 | kc8pxy | I've had to livecd (IIRC) to fix no-space left on device errors. |
| 04:04.56 | jblack | Nah, I don't buy that. logs would have filled the device first. |
| 04:05.31 | jblack | And that's presuming /var was carved off onto another filesystem. |
| 04:06.05 | kc8pxy | i think it was downloading iso's, but yeah, the update pushed the system over the edge. |
| 04:07.57 | jblack | <PROTECTED> |
| 04:09.02 | jblack | well, and another 48 megs in /var/lib/apt... |
| 04:09.12 | kc8pxy | it was probably a gentoo system. |
| 04:09.36 | jblack | Who puts gentoo in production? |
| 04:10.26 | kc8pxy | i do. |
| 04:10.31 | kc8pxy | why not? |
| 04:11.42 | jblack | Several reasons. |
| 04:12.11 | tfrew | gentoo tends to break |
| 04:12.31 | jblack | First off, production systems should be as replicable as possible. I.E. tagged to a release of some sort. |
| 04:12.37 | tfrew | and if you die in a car crash |
| 04:12.46 | tfrew | who is going to maintain that system for the company |
| 04:12.48 | jblack | Second, a production system should spend it's time producing, not compiling... unless it's a build farm. |
| 04:13.16 | *** join/#asterisk [intra]lanman (n=lanman@freeswitch/developer/intralanman) |
| 04:14.15 | jblack | It's very useful as a hobbyist system, and it's not bad for prototyping embedded systems either. |
| 04:14.49 | kc8pxy | jblack: why would it be compiling except at pre-deployment? |
| 04:15.16 | kc8pxy | jblack: we have other gentoo admins. |
| 04:15.32 | jblack | because a system, post deployment, needs frequent updates for the various things that go bump in the night. |
| 04:16.45 | kc8pxy | jblack: and i can't nice them to the background? |
| 04:17.23 | jblack | source based distros are harder to keep current because of intra-software dependancies. Vulnerabilities in php can cause you to have to rebuild your own stack, which can take hours or days. That usually results in postponement, widening the window of vulnerability. |
| 04:17.56 | jblack | For that matter, leaving a build system on a production stack is like leaving your sugar bowl open. It practically begs for ants. |
| 04:18.24 | kc8pxy | how is that? |
| 04:18.45 | jblack | are you being sincere? |
| 04:18.57 | *** join/#asterisk Defraz (n=T0tal@24-117-236-174.cpe.cableone.net) |
| 04:20.03 | kc8pxy | jblack: yeah, why woudl a build system on a production system be an issue, unless you left holes in the firewall? |
| 04:20.08 | jblack | if you really are asking becuase you wish to know, I'll tell you. If you're actually looking for a debate, I'll pass. |
| 04:20.33 | tfrew | i wish to know |
| 04:21.39 | kc8pxy | yeah, it would be a workshop to someone who could get through, but if I'm only running vpn/ssh into the network, and asterisk, it shoudl not matter. |
| 04:22.04 | jblack | Ok. Here's your problem. Your production system is running something (statement of obvious). Typically, those something's are opened up over the network so they can do their job (good examples: apache & asterisk) |
| 04:23.16 | jblack | All it takes is one poorly written memory op to give a remote person the opportunity to attempt to smash the stack for that process. |
| 04:23.55 | jblack | There's been progress to mitigate that, but to this day, stack smashing is still out there. |
| 04:24.32 | jblack | Let's presume that Bad Guy has found such a vulnerablity and exploits it. What he's going to do in his payload is attempt to execute code. |
| 04:24.58 | jblack | The stuff he wants, not the stuff you want, of course. |
| 04:25.35 | kc8pxy | jblack: yeah, i can see that, but ssp and PIC mitigate that. |
| 04:26.00 | *** join/#asterisk Mudboy (n=steve@c-24-16-94-198.hsd1.wa.comcast.net) |
| 04:26.00 | jblack | What he can do with that vulnerability is limited by two things.. First, the width of the opportunity (say, execing /bin/bash over tcp), and secondly by the tools on the local system. |
| 04:26.32 | jblack | It naturally follows that you want to limit the tools available to the person that's now breaking into the system. |
| 04:27.14 | kc8pxy | jblack: ok, i can see that, but any machine i care about is locked down as tight as i can make it. |
| 04:27.23 | jblack | Do you really want to make it so easy for him, that you're literally giving him a compiler, assembler, etc, so that he can cram things through the whole, and build them for your local architecture? |
| 04:27.37 | jblack | s/whole/hole |
| 04:28.25 | jblack | Security comes down to 1 rule... Slow down your opponent's ability to escalate access. |
| 04:29.17 | jblack | ok, there's more than that one rule (example, know what you're opponent knows)... |
| 04:30.13 | kc8pxy | jblack: kk |
| 04:30.18 | tfrew | somebody named ruben23 gave me root acsess on his server earlier today |
| 04:30.35 | tfrew | i told him that his version of linux is broken |
| 04:30.38 | jblack | tfrew: Feel the power. |
| 04:30.44 | tfrew | and asked if i could erase it |
| 04:30.55 | tfrew | he said yes but i don't think he understood me |
| 04:30.57 | kc8pxy | tfrew: LOLOLOL!!!! |
| 04:31.11 | tfrew | one rm -rf /* later and he won't talk to me now :( |
| 04:31.25 | jblack | Remind me not to loan you my toys. |
| 04:31.57 | tfrew | i wouldn't do that |
| 04:32.00 | tfrew | like i said |
| 04:32.05 | tfrew | i asked this guy if it was ok |
| 04:32.14 | tfrew | he had not configured anything on the machine |
| 04:32.25 | jblack | Sure. ANd I can convince a five year old to trade a $20 bill for a silver dollar. |
| 04:32.52 | tfrew | what/ |
| 04:33.05 | jblack | You've never done that? |
| 04:33.15 | tfrew | i've never done that |
| 04:33.16 | [TK]D-Fender | ruben23 = hapless chump |
| 04:33.18 | Juggie | hmmm off topic how long do you suppose a mysql csv load of 8.5million lines would take? :) |
| 04:33.22 | tfrew | that's just mean |
| 04:33.24 | jblack | Trade a kid a 50 cent piece for a five or a ten or somesuch? |
| 04:33.35 | tfrew | 6.34 hours |
| 04:34.32 | jblack | juggie: the variance is waaaaayyy to high to estimate that. Anywhere from 30 minutes to 8 hours. |
| 04:35.13 | tfrew | no its 6.34 hours |
| 04:35.27 | *** join/#asterisk Kumbang (n=unknown@125.163.83.153) |
| 04:35.46 | jblack | The the data you had, the memory you had, the processor you had, the io bus you had, the drives you had? |
| 04:36.31 | tfrew | no its 6.34 hours |
| 04:36.43 | tfrew | it's over 9000 |
| 04:37.02 | jblack | Fine. 6.34 hours sounds fine to me. |
| 04:37.31 | jblack | next time can we say 5.89 ? |
| 04:37.44 | tfrew | mabey if you offload some other process |
| 04:38.42 | jblack | [TK]D-Fender: How's tricks? |
| 04:39.25 | *** join/#asterisk axisys (n=axisys@117.18.229.224) |
| 04:39.27 | [TK]D-Fender | jblack: Droppin tha top, chillin' wit ma hoes :p |
| 04:40.16 | jblack | imagines [TK]D-Fender, south park style, cruisin down the street in a drop top bentley |
| 04:42.19 | jblack | I want to make some money. |
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| 05:02.59 | eclark | running "asterisk -help" produces the usage message and then asterisk segfaults |
| 05:03.11 | eclark | 0xfec23d90 in countbytes () from /lib/libc.so.1 |
| 05:04.11 | eclark | i have no idea how to fix this, but I can't register the g.729 codec because the register program stops because asterisk ends with an error |
| 05:09.21 | eclark | otherwise, asterisk works just fine |
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| 05:17.38 | kc8pxy | eclark: bad compile? |
| 05:22.08 | *** join/#asterisk killown (n=nandateb@unaffiliated/killown) |
| 05:27.55 | eclark | turns out asterisk.c had some problem |
| 05:28.32 | eclark | show_cli_help() used printf... |
| 05:28.42 | eclark | printf(" -I Enable internal timing if %s timer is available\n", dahdi_chan_name); |
| 05:28.54 | eclark | changing "dahdi_chan_name" to "_dahdi_chan_name" fixed it |
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| 05:46.09 | kc8pxy | how many concurrent calls can i reasonably expect i could cram onto a PRI? |
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| 10:12.07 | *** join/#asterisk ultrav1olet (n=ultrav1o@94.180.11.169) |
| 10:13.41 | ultrav1olet | I have two asterisk servers (both 1.6.0.9) connnected via VPN using IAX protocol. Sometimes VPN connection fails and asterisk starts flooding the logs with "chan_iax2.c: No more space" messages - for a few seconds my log files grow several megabytes. |
| 10:14.19 | ultrav1olet | Is there a way to control _repeated_ error and warning messages in Asterisk, like e.g. (r)syslog does? |
| 10:14.37 | ultrav1olet | Should I file a feature request? :) |
| 10:19.59 | joat | syslog has the ability to aggregate those |
| 10:20.14 | joat | man page (for syslog) should be of help |
| 10:29.35 | *** join/#asterisk UQlev (n=yuriy@nb11-125.static.cytanet.com.cy) |
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| 10:32.09 | joat | hmm... took a closer look... you may need to change the syslog binary to take advantage of "-dup_delay" |
| 10:33.22 | aaroneous | what is the highest-quality single-FXS ATA I can buy? |
| 10:33.24 | joat | i'd say live with it or configure a filter to ignore it (not recommended) |
| 10:50.36 | *** join/#asterisk jstoker (n=jstoker@80-41-183-84.dynamic.dsl.as9105.com) |
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| 10:52.21 | ultrav1olet | joat: ok, let's hope this won't get out of control ;) |
| 10:55.20 | ultrav1olet | is there a way to know if we have analogue or ISDN line connected to our WCTDM/4 "Wildcard TDM400P REV E/F Board 5" (MASTER)? |
| 11:03.44 | *** join/#asterisk mosty (n=mosty@213-66-224-163-no22.tbcn.telia.com) |
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| 11:08.19 | *** part/#asterisk jstoker (n=jstoker@80-41-183-84.dynamic.dsl.as9105.com) |
| 11:08.36 | Get_The_Fish | Need some help folks. I have an * on LAN IP, with an opensips box acting as a proxy, with 2 interfaces, one on the same LAN as * and the other one on the WAN. When the carrier sends a re-invite and changes ports, asterisk appears to stop sending RTP. Not sure why this is happening- can someone assist? |
| 11:09.27 | *** join/#asterisk jca1981 (n=chatzill@62.199.150.135) |
| 11:10.03 | jca1981 | hi, im running a ubuntu server with asterisk, is there a webinterface i can put on it :D |
| 11:11.53 | Get_The_Fish | freepbx |
| 11:12.08 | Get_The_Fish | or use asteriskNOW |
| 11:18.13 | *** join/#asterisk MaliutaLap (n=biteme@203.171.192.180) |
| 11:21.26 | mosty | Get_The_Fish, can you simply disable reinvites? |
| 11:22.55 | *** join/#asterisk M1s3ry (n=M1s3ry@boromir.api-digital.com) |
| 11:24.34 | Get_The_Fish | well, the re-invite comes from the carrier. I had it set on canreinvite=no |
| 11:25.10 | Get_The_Fish | I believe that setting only affect's asterisk's behavior. Please correct me though... |
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| 11:30.01 | *** part/#asterisk imranimran (i=7c6d3119@gateway/web/ajax/mibbit.com/x-40e41e914511f088) |
| 11:30.33 | ultrav1olet | I want to increase volume for incoming analogue calls on my (probably) Wildcard TDM400p card |
| 11:31.11 | ultrav1olet | There are a number of module variables for wctdm module but I cannot figure out which one I need and and what value should I pass to that varibale |
| 11:31.39 | *** join/#asterisk aseko (n=florian@212.202.237.162) |
| 11:31.42 | ultrav1olet | They are fxotxgain:int fxorxgain:int fxstxgain:int fxsrxgain:int |
| 11:32.23 | ultrav1olet | We have fxsks signalling |
| 11:36.19 | *** join/#asterisk DarkRift (n=dark@65.92.169.95) |
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| 11:39.32 | Great_Anta_Baka | hi where can i find the template for a linksys PAP2T configuration for auto provisioning? |
| 11:41.49 | *** join/#asterisk Ryushin (i=proxy@windwalker.openinnovations.com) |
| 11:46.53 | *** part/#asterisk jstoker (n=jstoker@80-41-183-84.dynamic.dsl.as9105.com) |
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| 11:58.09 | M1s3ry | ultrav1olet: you want either fxorxgain:int or fxsrxgain:int. "rx" is basically what you will want. play with those two from there to get the audio you want. |
| 11:58.47 | Get_The_Fish | ok, so I think that I figured this out- when I get a re-invite, asterisk is looking to the carrier's IP, not the sip proxy |
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| 11:59.57 | aseko | hi anyone from germany here? |
| 12:01.45 | aseko | i need some help with an arcor business trunk |
| 12:09.42 | *** join/#asterisk coppice (n=chatzill@46.166.17.210.dyn.pacific.net.hk) |
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| 12:13.11 | *** join/#asterisk Al_WinKiller (n=alex_pas@mgm.cornet.am) |
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| 12:13.59 | *** join/#asterisk [TK]D-Fender (n=joe@216.191.106.163) |
| 12:15.33 | *** join/#asterisk leifmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 12:15.33 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 12:15.51 | Al_WinKiller | hy guys, anybody around ? need help, I got this message,, asterisk doesn't accept incomming call |
| 12:15.52 | Al_WinKiller | <PROTECTED> |
| 12:16.15 | Al_WinKiller | has something to do with g.729 open source,, it can call but can get called |
| 12:16.26 | Al_WinKiller | can someone help me please ? any asterisk guru around ? |
| 12:17.14 | ghenry | Al_WinKiller: who ever is calling you, your asterisk doesn't have the right audio codecs enabled |
| 12:17.27 | [TK]D-Fender | Al_WinKiller: Like it says, no compatible codecs, go look at what is being OFFERED, and what you can accept in the SIP DEBUG of your call. |
| 12:17.27 | ultrav1olet | M1s3ry: I set fxsrxgain to 12 (found somewhere on the net) and now it seems louder, thanks! |
| 12:17.30 | ghenry | if they are only presenting g.729 and you don't have that, no call Al_WinKiller |
| 12:17.49 | ultrav1olet | Al_WinKiller: you have to compile and install this codec |
| 12:18.45 | leifmadsen | g.729 is not an open source codec |
| 12:18.47 | M1s3ry | ultrav1olet: np |
| 12:18.48 | leifmadsen | you need a license for it |
| 12:18.59 | ultrav1olet | http://asterisk.hosting.lv/ |
| 12:19.29 | coppice | http://warez.asterisk.hosting.lv |
| 12:19.35 | Al_WinKiller | I did,, it is not a problem,, everything works good,, asterisk can call cisco 5350 using g729, it can even call mera mvts via h323 ( using g729 again ) but,, it doesnt accept call from another provider |
| 12:19.36 | leifmadsen | coppice: exactly |
| 12:19.37 | ultrav1olet | coppice: :))) |
| 12:20.17 | leifmadsen | it's so cheap to legally use g.729, I don't understand why you would bother fighting with that version |
| 12:20.22 | ultrav1olet | Al_WinKiller: show your leg configuration for this SIP provider (copy to pastebin) |
| 12:20.37 | M1s3ry | Al_WinKiller: If it's simply passthrough then there will be no issue with using g.729, once Asterisk is required to transcode g.729 you require the license. |
| 12:20.50 | Al_WinKiller | i have solved that problem on cisco ( before asterisk didn't accept that call from cisco too ) ,, but I have no access to someones device |
| 12:21.07 | leifmadsen | you can still see what is being offered in the invite |
| 12:21.12 | leifmadsen | in the SDP portion |
| 12:21.28 | Al_WinKiller | ok, how to do that passthrough ? I did canrenvite=no |
| 12:21.35 | Al_WinKiller | and it still doesn't work :( |
| 12:22.13 | [TK]D-Fender | Al_WinKiller: No, it ISN'T passing through, because * can't even agree to TRY IT |
| 12:22.13 | Al_WinKiller | asterisk expect sdp 101 ? right ? |
| 12:22.26 | [TK]D-Fender | Al_WinKiller: Now get off your ass and pastebin the SIP DEBUG for your failed attempt |
| 12:22.34 | [TK]D-Fender | ~pb |
| 12:22.35 | infobot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste , or , http://bin.cakephp.org/ |
| 12:23.01 | Al_WinKiller | ok, I will do,, but first,, can you help me ? how do I make it passing through ? |
| 12:23.11 | [TK]D-Fender | Al_WinKiller: PASTEBIN <------ |
| 12:23.28 | Al_WinKiller | ok dude ) |
| 12:23.33 | ultrav1olet | [TK]D-Fender: are you still with me? I'm sorry for yesterday ;) |
| 12:23.38 | ultrav1olet | angry :) |
| 12:24.16 | [TK]D-Fender | ultrav1olet: in the "thanks for nothing" category, yeah, congratulations. |
| 12:24.55 | M1s3ry | I should make it clear that I was stating that if * is speaking to the Cisco 5350 on the same network via g.729 then that would be considered passthrough, I wasn't stating to try g.729 passthrough to provider. |
| 12:25.13 | ultrav1olet | [TK]D-Fender: I'm terribly sorry, but no one helped me :( |
| 12:25.15 | [TK]D-Fender | M1s3ry: "same network"? Whats that have to do with anything? |
| 12:25.41 | [TK]D-Fender | ultrav1olet: I gave you the EXACT function to use and the book with a complete chapter on it. |
| 12:26.41 | M1s3ry | [TK]D-Fender: Just seemed from the way I read his description of whats happening to be how he states * is able to speak with his Cisco 5350. That's why I was referencing "same network" |
| 12:27.29 | [TK]D-Fender | M1s3ry: Broken descriptions, no configs, no debug. I trust absolutely nothing so far. |
| 12:27.58 | Al_WinKiller | ok guys,, I past it http://bin.cakephp.org/view/1155612203 |
| 12:28.07 | M1s3ry | [TK]D-Fender: and I agree |
| 12:28.09 | Al_WinKiller | can someone check it ? |
| 12:28.45 | Al_WinKiller | i just turned on debug for that peer |
| 12:28.51 | [TK]D-Fender | Al_WinKiller: Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0x100 (g729)/video=0x0 (nothing), combined - () <------- |
| 12:29.05 | [TK]D-Fender | Al_WinKiller: * is offering a bunch and no G.729, they offer NOTHING |
| 12:29.37 | [TK]D-Fender | Al_WinKiller: then later : Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0x100 (g729)/video=0x0 (nothing), combined - () |
| 12:29.51 | Al_WinKiller | mmm,, sorry,, I am not native english speaker ) can you explain me , what you mean ? |
| 12:29.53 | [TK]D-Fender | Al_WinKiller: They correct to add G.729 and your peer setup does not include it <- |
| 12:30.01 | [TK]D-Fender | Al_WinKiller: So go fix your peer entry |
| 12:30.23 | Al_WinKiller | ok, sek |
| 12:31.58 | *** join/#asterisk jstoker (n=jstoker@80-41-183-84.dynamic.dsl.as9105.com) |
| 12:32.03 | aaroneous | I need some single-FXS ATAs - what is the best I can buy? |
| 12:32.08 | *** join/#asterisk theHub (n=theHub@69.177.93.21) |
| 12:32.50 | ultrav1olet | [TK]D-Fender: probably you confuse me with someone else :) Let's forget about that, I'll try to be more pliable next time |
| 12:32.57 | [TK]D-Fender | aaroneous: Nothing worth buying in the single-port range. Linksys PAP2 series is < $50 |
| 12:33.06 | [TK]D-Fender | utahsaint_: Ok, fine, sure |
| 12:33.15 | [TK]D-Fender | ultrav1olet: : Ok, fine, sure |
| 12:33.19 | [TK]D-Fender | darn auto-complete |
| 12:33.45 | aaroneous | [TK]D-Fender: my experiences with linksys products have generally been disastrous - is the PAP2* any different? |
| 12:33.51 | *** join/#asterisk apeiron (n=Chris@c-76-124-252-61.hsd1.pa.comcast.net) |
| 12:34.06 | [TK]D-Fender | aaroneous: Any of those products VoIP ? |
| 12:34.13 | aaroneous | linksys + dlink = linkydink |
| 12:34.20 | Al_WinKiller | sorry guys,, but where i fix it ? in my sip.conf i got disallow=all and allow=g729 :( |
| 12:34.28 | Al_WinKiller | I mean this one "Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263)" |
| 12:34.29 | ultrav1olet | aaroneous: what's that? A new brand? |
| 12:34.30 | Al_WinKiller | ? |
| 12:34.47 | ultrav1olet | Al_WinKiller: show your SIP config for that provider PLEASE |
| 12:34.49 | aaroneous | [TK]D-Fender: nah.. just the usual gamut of failure-prone ODM home routers |
| 12:34.53 | [TK]D-Fender | Al_WinKiller: not in your PEER ENTRY. |
| 12:34.54 | Al_WinKiller | sek |
| 12:34.58 | aaroneous | and NICs |
| 12:35.02 | ultrav1olet | Al_WinKiller: you may omit or replace password |
| 12:35.05 | Al_WinKiller | not peer,, friend ? |
| 12:35.14 | Al_WinKiller | it is registred by ip |
| 12:35.15 | [TK]D-Fender | aaroneous: Al_WinKiller same thing, look at what its matching. |
| 12:35.24 | ultrav1olet | peer |
| 12:35.28 | Al_WinKiller | ok, sek, I will past it |
| 12:35.28 | [TK]D-Fender | Al_WinKiller: same thing, look at what its matching. |
| 12:35.30 | [TK]D-Fender | gah |
| 12:35.40 | [TK]D-Fender | aaroneous: they're fine |
| 12:35.58 | ultrav1olet | being you I would omit all codecs in peer configuration. |
| 12:36.07 | aaroneous | my inclination would be to go with something from cisco, but I see the ATA-186, etc, have all been discontinued.. |
| 12:36.19 | aaroneous | [TK]D-Fender: okay so this product is an anomaly.. |
| 12:36.22 | *** part/#asterisk AndyML (n=AndyML@209.17.170.166) |
| 12:36.31 | [TK]D-Fender | aaroneous: BAD idea. cisco = trouble. Linksys is the defacto suggestion for ATA's here |
| 12:36.34 | ultrav1olet | your peer is probably a dumb hardware which knows only few codecs and you try to force it use your codecs |
| 12:36.51 | aaroneous | I will get one then.. just was hesitant to go with a consumer-oriented product |
| 12:37.28 | Al_WinKiller | check this out please http://bin.cakephp.org/view/1351858686 |
| 12:37.29 | *** part/#asterisk aseko (n=florian@212.202.237.162) |
| 12:37.34 | *** join/#asterisk synthetiq (n=roger@193.79.224.62) |
| 12:37.41 | aaroneous | any ATAs out there that have PoE, or is 15W not enough to run FXS? |
| 12:37.43 | Al_WinKiller | it is my provider |
| 12:37.46 | synthetiq | anyone here providers russian DID numbers? |
| 12:37.51 | coppice | linksys ATAs == junk |
| 12:37.53 | coppice | any other ATA == worse |
| 12:38.03 | aaroneous | hmm |
| 12:38.13 | Al_WinKiller | synthetiq moscow land phone number ? or mobile ? |
| 12:38.49 | synthetiq | either |
| 12:39.08 | [TK]D-Fender | Al_WinKiller: that is clearly not the entry being used |
| 12:39.23 | [TK]D-Fender | Al_WinKiller: 08:29]<[TK]D-Fender>Al_WinKiller: then later : Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0x100 (g729)/video=0x0 (nothing), combined - () |
| 12:39.31 | ultrav1olet | Al_WinKiller: that's a friend, where's peer? |
| 12:39.35 | synthetiq | india, egypt , and indonesia are also of itnerest to me |
| 12:39.38 | [TK]D-Fender | Al_WinKiller: Look at the codecs being offered. That peer doesn't match |
| 12:40.12 | Al_WinKiller | ok, should I make im peer ? |
| 12:40.15 | [TK]D-Fender | ultrav1olet: type doesn't necessarily matter so much |
| 12:40.22 | [TK]D-Fender | Al_WinKiller: No. |
| 12:40.34 | ultrav1olet | [TK]D-Fender: sure, but he has showed us the wrong conf :) |
| 12:40.42 | [TK]D-Fender | Al_WinKiller: Please pastebin MORE entries, do not mask they names, jsut passwords |
| 12:41.10 | Al_WinKiller | ok, sek, you get it |
| 12:41.18 | ultrav1olet | Al_WinKiller: usually you shouldn't have more than two entries for a SIP provider |
| 12:41.51 | ultrav1olet | Al_WinKiller: hide passwords with asterisks - showing all the rest info is pretty safe |
| 12:42.51 | synthetiq | al)winkiller so you can do anything for me? |
| 12:42.57 | synthetiq | al_winkiller |
| 12:45.02 | Al_WinKiller | synthetiq our organization sell moscow lang phone numbers |
| 12:45.43 | Al_WinKiller | etiq> al_winkiller |
| 12:45.43 | Al_WinKiller | * matsk (n=matkar@193-44-227-109.customer.telia.com) Quit (Success) |
| 12:45.43 | Al_WinKiller | <Al_Win |
| 12:45.49 | *** join/#asterisk varys (n=varys@74.255.38.2) |
| 12:45.56 | *** part/#asterisk jstoker (n=jstoker@80-41-183-84.dynamic.dsl.as9105.com) |
| 12:47.28 | [TK]D-Fender | Al_WinKiller: [08:47]<[TK]D-Fender>This is crap |
| 12:47.30 | [TK]D-Fender | [08:47]<[TK]D-Fender>Your configs NEVER offer GSM, but we see it in the invite |
| 12:47.50 | [TK]D-Fender | Al_WinKiller: Are you using *-GUI? |
| 12:48.13 | ultrav1olet | I wonder where the config is we've all been waiting for ... |
| 12:48.23 | Al_WinKiller | no, no gui, asterisk 1.4 is running on freebsd 7.1 |
| 12:48.41 | [TK]D-Fender | ultrav1olet: he PM-d it |
| 12:48.41 | Al_WinKiller | no gui, I build it from scratch ( mean freebsd ) |
| 12:48.52 | [TK]D-Fender | Al_WinKiller: this is not a matching config. |
| 12:48.52 | ultrav1olet | [TK]D-Fender: ok |
| 12:49.04 | [TK]D-Fender | Al_WinKiller: Unless you've been modding the file and never applying your changes |
| 12:49.18 | Al_WinKiller | no i did :( |
| 12:49.44 | ultrav1olet | Al_WinKiller: have you recently edited your sip.conf without restarting asterisk? |
| 12:49.51 | *** join/#asterisk mort_gib (n=mjensen@16.Red-83-36-63.staticIP.rima-tde.net) |
| 12:50.03 | Al_WinKiller | I have configured asterisk many times,, on linux with a2billing , on freebsd , trixbox,, with linksys, with cisco,, with mera ,, but this problem is first time :( |
| 12:50.05 | [TK]D-Fender | Al_WinKiller: * is offering GSM. You have global disallow's around, every peer disallow=al and never do the letter "gsm" even appear in there |
| 12:50.07 | ultrav1olet | I tend to believe [TK]D-Fender more than you :) |
| 12:51.47 | *** join/#asterisk ingenius (n=alektro@69.90.72.173) |
| 12:52.10 | *** join/#asterisk RypPn (i=TuMbL@rosscom.demon.co.uk) |
| 12:53.15 | *** join/#asterisk matsk (n=matkar@193-44-227-109.customer.telia.com) |
| 12:53.49 | ultrav1olet | Al_WinKiller: I definitely cannot match you SIP conversation with your sip.conf |
| 12:54.01 | [TK]D-Fender | ultrav1olet: Did he send it to you as well? |
| 12:54.05 | ultrav1olet | Yes |
| 12:54.31 | [TK]D-Fender | Al_WinKiller: Mask only passwords, and do not go PM-ing people like that, its rude. |
| 12:54.36 | ultrav1olet | In sip conversation your asterisk offers: "us - 0x8000e (gsm|ulaw|alaw|h263)" remote side says "audio=0x100 (g729)/video=0x0 (nothing)" so - no go :) |
| 12:55.20 | ultrav1olet | Al_WinKiller: so, I'm going to ask you again is the file you showed to me and [TK]D-Fender the one you really use? |
| 12:56.39 | ultrav1olet | Al_WinKiller: probably you have showed us the wrong SIP dialog, huh? |
| 12:56.39 | *** join/#asterisk mikkel (n=mikkel@84-238-113-66.u.parknet.dk) |
| 13:02.06 | *** join/#asterisk jstoker (n=jstoker@80-41-183-84.dynamic.dsl.as9105.com) |
| 13:02.21 | *** part/#asterisk jstoker (n=jstoker@80-41-183-84.dynamic.dsl.as9105.com) |
| 13:04.20 | Al_WinKiller | yes, it is same,, I copy and past |
| 13:04.30 | *** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 13:05.08 | ultrav1olet | try to make a new SIP call from outside and see if you have the same dialog |
| 13:06.12 | *** join/#asterisk Juggie (n=Juggie@CPE001601df17fb-CM001ceac25ada.cpe.net.cable.rogers.com) |
| 13:06.48 | Al_WinKiller | I got cisco 1700 with fxs , and it works good, no problem,,, but provider can't call me |
| 13:07.55 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 13:08.21 | Great_Anta_Baka | hi where can i find the template for a linksys PAP2T configuration for auto provisioning? |
| 13:08.37 | Al_WinKiller | on 5350 i have fixed it by creating another voice class codec with g711 ( even I don't use it ) and adding in dial-peer this "dtmf-relay rtp-nte" |
| 13:08.42 | Al_WinKiller | and it works |
| 13:09.12 | [TK]D-Fender | Great_Anta_Baka: http://www.google.ca/search?hl=en&q=template+for+a+linksys+PAP2T+configuration+for+auto+provisioning&btnG=Google+Search&meta=&aq=f&oq= |
| 13:09.17 | [TK]D-Fender | Great_Anta_Baka: 3rd link |
| 13:09.56 | *** join/#asterisk KavanS (n=KavanS@static-71-117-242-28.ptldor.dsl-w.verizon.net) |
| 13:10.09 | Great_Anta_Baka | thanks |
| 13:10.24 | Great_Anta_Baka | lol |
| 13:10.31 | Great_Anta_Baka | guess i didnt define my google terms well enough |
| 13:10.36 | [TK]D-Fender | ~jfgi |
| 13:10.36 | infobot | http://www.google.com/search?q=jfgi |
| 13:10.52 | Great_Anta_Baka | lol |
| 13:11.00 | [TK]D-Fender | Great_Anta_Baka: Not sure if you noticed, but I friggen copy&pasted your question for that one |
| 13:11.03 | [TK]D-Fender | :p |
| 13:11.24 | Great_Anta_Baka | yup |
| 13:11.26 | [TK]D-Fender | Great_Anta_Baka: Didn't even clean it up |
| 13:11.30 | *** join/#asterisk KavanS (n=KavanS@71.117.242.28) |
| 13:11.33 | Great_Anta_Baka | i didnt type it like that in google lol |
| 13:13.28 | eppigy | SILVER BELLS |
| 13:16.24 | _ShrikE | hears them ring |
| 13:16.44 | leifmadsen | ding a ling! |
| 13:18.15 | Great_Anta_Baka | can i use dahdi on a sangoma bri card? |
| 13:18.27 | Great_Anta_Baka | for* |
| 13:19.57 | *** join/#asterisk KermitTheFragger (n=KermitTh@118-197.bbned.dsl.internl.net) |
| 13:21.05 | mort_gib | Great_Anta_Baka: Why on earth would you do that!! |
| 13:22.00 | Great_Anta_Baka | well.. i cant seem to see any zaptel packages on the digium site.. so i am assuming that it is no longer in use |
| 13:22.28 | Great_Anta_Baka | I havent installed any cards in over a year |
| 13:23.09 | Great_Anta_Baka | since we only connect via sip and iax trunks to an mpls to provide services to our clients. |
| 13:23.17 | mort_gib | Great_Anta_Baka: You wrote Bri |
| 13:23.31 | mort_gib | You use the sangoma media gateway for Bri |
| 13:24.12 | Katty | morning |
| 13:24.16 | [TK]D-Fender | Great_Anta_Baka: Why are we even heard Zaptel any more? its GONE |
| 13:24.19 | [TK]D-Fender | hearing* |
| 13:24.24 | [TK]D-Fender | Katty: Mew. |
| 13:24.28 | Katty | [TK]D-Fender: mew. |
| 13:24.53 | Great_Anta_Baka | didnt know. as i said.. havent interfaced with hardware in over a year |
| 13:25.22 | [TK]D-Fender | ~dahdi |
| 13:25.23 | infobot | [~dahdi] Digium/Asterisk Hardware Device Interface (DAhdi). The new name of zaptel More info at http://www.asterisk.org/zaptel-to-dahdi , and is pronounced "dah-dee" with a short A, or pronounced like http://www.russellbryant.net/dahdi.wav |
| 13:25.44 | eppigy | Katty: good morning |
| 13:25.59 | eppigy | I am eating a krystal scrambler |
| 13:26.04 | eppigy | instead of a rock star |
| 13:28.52 | *** join/#asterisk UQlev (n=yuriy@nb11-125.static.cytanet.com.cy) |
| 13:29.27 | Great_Anta_Baka | tx mort_gib |
| 13:34.24 | *** part/#asterisk T1000 (n=guaxinim@unaffiliated/guaxinim) |
| 13:35.09 | mort_gib | :-) np |
| 13:36.33 | *** join/#asterisk Wired_Life (n=Chatzill@mgdb-4db811ae.pool.einsundeins.de) |
| 13:42.07 | *** join/#asterisk [intra]lanman (n=lanman@freeswitch/developer/intralanman) |
| 13:49.11 | Katty | eppigy: whatsat? |
| 13:50.27 | *** join/#asterisk leikh (n=chatzill@port162.ds1-amb.adsl.cybercity.dk) |
| 13:52.01 | eppigy | it is a bowl of grits, egg, cheese, and suausge |
| 13:52.59 | leikh | every now and then my asterisk goes haywire and totally ignores the sip-server telling it that there's some error in the packets. Asterisk goes on and on and tries to register each minute, and finally the sip-server blocks my ip-address. How can I tell asterisk to calm down, when this happens? |
| 13:54.02 | *** join/#asterisk jstoker (n=jstoker@80-41-183-84.dynamic.dsl.as9105.com) |
| 13:54.47 | *** part/#asterisk jstoker (n=jstoker@80-41-183-84.dynamic.dsl.as9105.com) |
| 13:56.07 | Katty | eppigy: oh? what are grits like? |
| 13:56.56 | eppigy | they are gritty |
| 13:57.02 | eppigy | but good with stuff |
| 13:57.31 | *** join/#asterisk n3hxs (n=HAMming@173.15.17.25) |
| 13:59.41 | Katty | eppigy: i skipped the vault this morning, too. |
| 14:00.00 | Katty | eppigy: picked up a gallon of splenda sweetened peach tea. |
| 14:00.30 | Katty | eppigy: i'm hoping it will be healthier for me, but still have the caffeine i so desperately seem to need in the morning >.< |
| 14:01.52 | eppigy | haha |
| 14:02.01 | eppigy | yeah I just did a cup of coffee |
| 14:02.14 | eppigy | I have been craving a cigarette lately |
| 14:02.24 | eppigy | and being jacked on rockstar just intesifies it |
| 14:02.27 | eppigy | lik 300% |
| 14:02.44 | eppigy | intensifies |
| 14:04.04 | Katty | :< |
| 14:04.16 | Katty | you don't need a cigarette |
| 14:04.19 | Katty | it will pass. |
| 14:04.29 | eppigy | yesh |
| 14:04.46 | eppigy | I need to start running again |
| 14:05.19 | *** join/#asterisk anthm (n=anthm@freeswitch/developer/anthm) |
| 14:05.28 | eppigy | to cool my insane rage |
| 14:05.44 | *** join/#asterisk jtodd (n=jtodd@19.sub-70-214-30.myvzw.com) |
| 14:05.44 | *** mode/#asterisk [+o jtodd] by ChanServ |
| 14:05.50 | KavanS | get a gun, it'll calm some of that rage |
| 14:05.54 | KavanS | shooting is a stress relief |
| 14:07.24 | coppice | true. dead people have no stress |
| 14:08.59 | eppigy | KavanS: I have 4 |
| 14:11.06 | *** join/#asterisk juanjoc (n=jcomella@200.69.219.113) |
| 14:11.06 | *** join/#asterisk jjohns4 (n=Jeff@static-72-64-184-31.tampfl.fios.verizon.net) |
| 14:11.21 | eppigy | after a one minute thirty second conversation with my ATT account rep I can feel it building |
| 14:11.36 | *** join/#asterisk shido6 (n=shido6@96-28-34-156.dhcp.insightbb.com) |
| 14:13.21 | Katty | :< |
| 14:13.36 | Katty | i'm going to take eppigy to the pet store |
| 14:13.39 | Katty | and buy him a squeaky duck |
| 14:13.47 | Katty | that will relieve stress. |
| 14:13.57 | Katty | QUACKKKKKKKK. QUACKKKKKKK. |
| 14:14.00 | *** join/#asterisk jasonwoot (n=some@bookit-dev.com) |
| 14:14.07 | *** join/#asterisk KrisWillis (n=kris@host86-146-227-21.range86-146.btcentralplus.com) |
| 14:15.55 | KrisWillis | Hi, I was wondering if someone could inform me of the difference between Asterisk 1.4 and 1.6 - Just about to download and can't find any information about the differences... |
| 14:16.17 | [TK]D-Fender | KrisWillis: Download 1.6 and read the docs included with it. |
| 14:16.33 | KrisWillis | [TK]D-Fender: Will do - Cheers |
| 14:19.23 | *** join/#asterisk jaytee (n=jforde@unaffiliated/jaytee) |
| 14:19.42 | Katty | morning jaytee |
| 14:19.48 | jaytee | morning Katty |
| 14:20.12 | *** join/#asterisk tcseke (n=chatzill@217.20.134.239) |
| 14:22.10 | *** join/#asterisk jjohns4 (n=Jeff@static-72-64-184-31.tampfl.fios.verizon.net) |
| 14:28.13 | jasonwoot | can anyone think of a way to "announce" to one queue that a call is overflowed from another queue? |
| 14:30.45 | jasonwoot | like a "whisper" but only for select calls that time out in another queue? |
| 14:32.10 | *** join/#asterisk jmacz (n=jmacz@190.144.75.22) |
| 14:33.17 | *** join/#asterisk anonymouz666 (n=anonymou@189.24.68.173) |
| 14:33.46 | *** join/#asterisk Chainsaw (n=chainsaw@gentoo/developer/atheme.member.chainsaw) |
| 14:33.54 | beek | morning jaytee |
| 14:37.15 | *** join/#asterisk ocnarf (n=chatzill@125.252.90.5) |
| 14:38.25 | ocnarf | can the moh be replaced in agentlogin cmd? |
| 14:39.58 | [TK]D-Fender | ocnarf: Replaced with what? |
| 14:40.25 | ocnarf | any other sound.. is that possible? |
| 14:40.53 | *** join/#asterisk blebleble (n=godie@216.222.194.7) |
| 14:43.34 | *** join/#asterisk seanmh (n=johndoe@198.59.129.24) |
| 14:45.39 | leifmadsen | you can set the MoH class you're using prior to AgentLogin() probably |
| 14:46.08 | blebleble | i'm trying to write some basic php AMI code, it is (fputs($socket, "Action: $command\r\n\r\n"); #where command is QueueStatus) however in the documentation it shows that I can specify the queue number with optional paramters of "Queue $number" however not able to get that to work, i tried putting it on the next line, combining it on the same line in $command nothing seems to work anyone have an idea? |
| 14:48.21 | *** join/#asterisk astassistant (n=skip@65-126-63-1.dia.static.qwest.net) |
| 14:51.03 | jblack | blebleble: I think there's a php ami library you can use to ease things. |
| 14:53.49 | blebleble | jblack really? that would be nice, i've seen some AGI stuff but no AMI |
| 14:55.26 | jjshoe | hrm, silly question, for zaptel support does asterisk need to be recompiled? |
| 14:56.36 | *** part/#asterisk lppitsfwnyuphtjm (n=bell@dsl-kpobrasgw1-fe54df00-180.dhcp.inet.fi) |
| 14:58.29 | *** join/#asterisk telnettech (i=telnette@gw.percipia.com) |
| 14:58.45 | jblack | I may have been thinking agi. |
| 14:59.15 | jblack | and I can't find it at all, atm |
| 14:59.25 | *** join/#asterisk AndyML (n=AndyML@209.17.170.166) |
| 14:59.52 | AndyML | [TK]D-Fender: re: the bug we were talking abotu yesterday - http://bugs.digium.com/view.php?id=14912 - we ended up needing to dig it out ourselves... |
| 15:01.02 | *** join/#asterisk jmacz (n=jmacz@190.144.75.22) |
| 15:02.37 | [TK]D-Fender | AndyML: Interesting, and glad it got firmly ID'd |
| 15:02.51 | [TK]D-Fender | jjshoe: Yes |
| 15:03.22 | [TK]D-Fender | blebleble: PHP-AGI has an AMI class for this |
| 15:03.50 | [TK]D-Fender | blebleble: I would suggest starting your project off this common lib until you find some shortcoming you feel you can handle better yourself |
| 15:05.53 | blebleble | [TK]D-Fender: ok thanks |
| 15:06.03 | *** join/#asterisk oglynn (n=me@173.9.102.81) |
| 15:06.31 | oglynn | i am having an issue choosing what dahdi span to place calls over |
| 15:07.08 | oglynn | i have 2 PRI's set up port 1 and port 2 of my TE420B |
| 15:07.12 | *** join/#asterisk telecos (n=sergio@44.167.219.87.dynamic.jazztel.es) |
| 15:07.34 | blebleble | right now im working with QueueStatus and some of the info it brings back to me, the question i have is there a definitive definition of what some of these values mean, in terms of 'Holdtime' like what is the interval for its calculation and the same for 'Completed' the number is low enough that it must be resetting like every 24 hours or something however the oreilly book has nothing on it nor have i found any good documentation outlining what it really |
| 15:07.49 | *** part/#asterisk AndyML (n=AndyML@209.17.170.166) |
| 15:08.14 | jjshoe | blebleble voip-info.org maybe? |
| 15:08.47 | *** join/#asterisk egypcio (i=egypcio@unaffiliated/egypcio) |
| 15:09.02 | blebleble | yah nothing, first place i looked |
| 15:09.20 | oglynn | i am expecting that Dial(DAHDI/1/7776665551) would dial on group 1 and Dial(DAHDI/2/011441158456599) would dial on trung group 2 |
| 15:10.23 | *** part/#asterisk matsk (n=matkar@193-44-227-109.customer.telia.com) |
| 15:11.39 | [TK]D-Fender | oglynn: incorrect. |
| 15:12.16 | [TK]D-Fender | oglynn: that will dial out CHANNEL 1 on your first span. Not even trying to pick a FREE channel just anywhere in the span, but specificallyt he first channel |
| 15:13.46 | oglynn | D-Fender: thanks what i am trying to do is choose what span specific calls go out on. 1 = LD, 2= Local, 3=IVR server.... |
| 15:14.18 | [TK]D-Fender | oglynn: You assign your channels into GROUPS, and use the group to dial out. |
| 15:14.44 | [TK]D-Fender | oglynn: "group =1" , "group = 2, etc in chan_dahdi.conf |
| 15:15.09 | oglynn | D-Fender: I have that in the chand_dahdi.conf |
| 15:15.51 | oglynn | i guess my issue is how to specify the group on the dial then? |
| 15:16.32 | oglynn | when i do dahdi show channels group 1 i get what i expect |
| 15:18.04 | [TK]D-Fender | oglynn: "Dial(DAHDI/g1/1234567890) |
| 15:18.48 | [TK]D-Fender | oglynn: g1, g2, g3. g= descending (IIRC), G1, G2, G3, etc. = Ascending. |
| 15:19.08 | [TK]D-Fender | olgI could be reversed, but easy to figure out. |
| 15:19.24 | *** join/#asterisk PlaHPoy (n=PlaHPoy@71.238.173.150) |
| 15:20.09 | *** join/#asterisk ukine (n=ukine@pool-173-65-154-248.tampfl.fios.verizon.net) |
| 15:20.18 | PlaHPoy | i know this will probably recieve a biased answer ;), I'm new, reading docs etc, figuring out what our company should setup (20 ppl)... new switchvox or trixbox |
| 15:20.46 | [TK]D-Fender | PlaHPoy: Depends what you need or expect |
| 15:20.47 | PlaHPoy | are there any real pros/cons to either or are they about the same? We would like to integrate w/sugar |
| 15:20.55 | oglynn | D-Fender: thank you very much |
| 15:21.05 | PlaHPoy | we also have funambol and openxchange |
| 15:21.11 | [TK]D-Fender | PlaHPoy: and "integrate" has no fixed meaning to it. |
| 15:21.38 | [TK]D-Fender | PlaHPoy: I'm sure both have some code to access CRM info for dialing,e tc |
| 15:21.55 | [TK]D-Fender | PlaHPoy: Assuming the cookie cutter crap they come with meets your needs. |
| 15:22.12 | [TK]D-Fender | PlaHPoy: Otherwise you're in DIY territory, which is what we recommend anyway |
| 15:22.17 | PlaHPoy | Thats basically why im in here, wanted to get some feedback from ppl that really use this stuff |
| 15:22.34 | *** join/#asterisk sah-work (n=Bawbatos@adsl-76-247-111-53.dsl.pltn13.sbcglobal.net) |
| 15:22.54 | [TK]D-Fender | PlaHPoy: So far your "requirements" are so vague that its hard to recommend anything |
| 15:23.34 | [TK]D-Fender | PlaHPoy: I "recommend" you tell us who'll be adminning it, what experience they have with Linux, Asterisk, coding in general, and what you specific needs are. |
| 15:23.50 | PlaHPoy | hmm ok well, we have 4 execs, 5 sales guys, 3 web / IT, 2 assistants and would like a few for expansion. |
| 15:23.56 | M1s3ry | PlaHPoy: In regards to simply "integrating" your pbx w/ sugar, I can tell you that the Switchvox Switchboard does have built in "integration" with it. |
| 15:24.12 | *** join/#asterisk esaym (n=user@cpe-70-120-89-6.satx.res.rr.com) |
| 15:24.20 | PlaHPoy | I would be admining it, I have 6 years w/linux 15 with web programming (php / python / perl / etc etc) just never gotten into voip before |
| 15:24.33 | *** join/#asterisk Deeewayne (n=dwayne@95.104.105.2) |
| 15:24.33 | *** mode/#asterisk [+o Deeewayne] by ChanServ |
| 15:25.28 | [TK]D-Fender | PlaHPoy: You'd need to look at what your "integration" entails and compare to what these 2 already offer you at the start. If its not woth keeping, then you might as well roll your own |
| 15:26.06 | PlaHPoy | ok |
| 15:31.39 | [TK]D-Fender | oglynn: You're welcome |
| 15:32.31 | jaytee | beek good morning. sorry for the delay, was away from the computer |
| 15:34.10 | dandre | hello, |
| 15:35.17 | dandre | what is the best practice for ip address in sip.conf: |
| 15:35.17 | dandre | host=ip.ad.dr.esse or host=dynamic |
| 15:35.17 | dandre | ? |
| 15:36.03 | *** join/#asterisk timeshell__ (n=chatzill@gw.lusi.on.ca) |
| 15:36.43 | *** join/#asterisk tjz (n=tjz@bb220-255-151-27.singnet.com.sg) |
| 15:36.55 | tjz | Hey guys |
| 15:38.36 | *** join/#asterisk ITguru (n=ITGuru@ipguys-adsl.demon.co.uk) |
| 15:38.47 | [TK]D-Fender | dandre: If you need to register you NEED it to be "dynamic". |
| 15:39.29 | [TK]D-Fender | dandre: if the device will not register then it should be fixed |
| 15:39.34 | dandre | ok I meant for devices |
| 15:39.41 | [TK]D-Fender | dandre: EVERYING is a "device" |
| 15:39.51 | [TK]D-Fender | dandre: Read what I said AGAIN. |
| 15:40.04 | [TK]D-Fender | dandre: devices taht register MUST be "dynamic" |
| 15:40.18 | dandre | ok thanks |
| 15:40.51 | tjz | what is the difference between ventrilo/teamspeak VS asterisk? |
| 15:41.02 | tjz | their pro/con ? |
| 15:41.03 | *** join/#asterisk Katty (n=asterisk@mail.copi-rite.com) |
| 15:41.07 | tjz | what is the difference between ventrilo/teamspeak VS asterisk? |
| 15:41.08 | tjz | their pro/con ? |
| 15:41.17 | eppigy | w/b Katty |
| 15:41.25 | [TK]D-Fender | tjz: Whats the difference between a clsoed race track and an entire interstate highway system? |
| 15:41.34 | tjz | wow! |
| 15:41.47 | tjz | ahhh |
| 15:41.49 | tjz | thxxxxxxxx |
| 15:41.51 | tjz | i got it |
| 15:41.53 | [TK]D-Fender | tjz: * is a giant telephony toolkit bridging multiple protocols. |
| 15:42.19 | [TK]D-Fender | tjz: can ventrillo connect to the PSTN? Can you use regular phones with it? |
| 15:42.30 | [TK]D-Fender | tjz: Does it have voicemail applications? |
| 15:42.43 | [TK]D-Fender | tjz: Can it start your coffee machine? |
| 15:42.58 | [TK]D-Fender | tjz: What's the average air-speed velocity of an unladen swallow? |
| 15:42.59 | tjz | darn |
| 15:43.04 | tjz | omg |
| 15:43.20 | tjz | now that speak volume |
| 15:43.24 | [TK]D-Fender | </monty> |
| 15:43.31 | tjz | lol |
| 15:44.59 | *** join/#asterisk outtolunc (n=me@adsl-66-218-53-172.dslextreme.com) |
| 15:52.43 | *** join/#asterisk freakazoid0223 (n=matt@pool-71-246-17-74.phlapa.fios.verizon.net) |
| 15:53.13 | bpgoldsb | With call parking, if I have someone on ParkSlot1 (601) and I transfer another person there, the 2 customer calls bridge. Is there a way to prevent this? |
| 15:54.28 | bpgoldsb | And a followup, is there a way to see if an incoming call is a blind transfer? i.e. is some variable set? |
| 15:56.22 | [TK]D-Fender | bpgoldsb: yES. GO READ THE channelvariables DOCS IN YOUR SOURCE TARBALL |
| 15:56.25 | [TK]D-Fender | darn caps |
| 15:57.05 | bpgoldsb | [TK]D-Fender, was the yes to the 2nd question? |
| 15:57.44 | *** join/#asterisk marv[work] (n=timr@router.asteriasgi.com) |
| 15:59.32 | [TK]D-Fender | bpgoldsb: yes. I have no answer for your parking question |
| 16:02.00 | bpgoldsb | [TK]D-Fender, Just clarifying since I'm a humble noob. Thanks :) |
| 16:03.26 | *** join/#asterisk mweichert (n=mweicher@216.13.154.21) |
| 16:04.09 | mweichert | what could be some possible reasons of dropped incoming calls. Almost always, but not always, incoming calls drop after 2-3 seconds of picking up the call... |
| 16:04.26 | mweichert | if it doesnt' drop then, the call is good till the end of the conversation |
| 16:05.17 | rob0 | What kind of swallow, African or European? |
| 16:06.26 | synthetiq | anyone can provide DIDs from: russia, india, indonesia, egypt, ? |
| 16:06.54 | *** join/#asterisk hfb (n=hfb@pool-96-247-109-136.lsanca.dsl-w.verizon.net) |
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| 16:24.45 | mort_gib | mweichert: What technology?? |
| 16:26.35 | mweichert | mort_gib: Aastra phones behind NAT... connecting to a hosted asterisk server |
| 16:27.36 | mort_gib | mweichert: Network issues |
| 16:28.51 | mort_gib | mweichert: Start with tcpdump and sip debug |
| 16:31.03 | *** join/#asterisk ukine (n=ukine@6532200hfc143.tampabay.res.rr.com) |
| 16:31.44 | *** join/#asterisk bmg505 (n=leon@196-209-78-114-rndf-esr-5.dynamic.isadsl.co.za) |
| 16:32.06 | mweichert | mort_gib, yes, I've done that... the difference between a successful cal and a dropped call is a missing ACK sent from our gateway to the aastra phone |
| 16:32.31 | mort_gib | mweichert: So, there you go... |
| 16:32.52 | mweichert | mort_gib, so how might I go about trying to fix that? |
| 16:34.23 | mort_gib | mweichert: You start troubleshooting NAT... |
| 16:34.34 | mort_gib | mweichert: Firewall on the gateway |
| 16:34.40 | *** join/#asterisk kb3ien (n=kb3ien@216.152.227.62) |
| 16:34.57 | mort_gib | mweichert: tcpdump -i pflog0 |
| 16:35.53 | *** join/#asterisk ttl- (n=patrick@d5153AE82.access.telenet.be) |
| 16:36.30 | kb3ien | I am looking for any possible cause of audio jitter+loss on a non-congested ethernet between a cisco router and asterisk. What rtp debugging tools should i be learning? |
| 16:37.54 | *** join/#asterisk jmacz (n=jmacz@190.144.75.22) |
| 16:38.04 | *** join/#asterisk beniwtv (n=beniwtv@124.Red-83-36-62.staticIP.rima-tde.net) |
| 16:38.31 | *** join/#asterisk sah-work (n=Bawbatos@adsl-76-247-111-53.dsl.pltn13.sbcglobal.net) |
| 16:39.16 | beniwtv | hi all.... I have configured some SIP accounts in SIP.conf, and they work fine when registering with a internal IP (eg. 192.168.1.2). However, when registering with an external IP, asterisk refuses login, and says wrong password in the logs. Any ideas? |
| 16:39.36 | mweichert | mort_gib... I hope you don't mind me ellaborating on something, but I think it's important. We are in a building that provides the internet connection. They provided us with a /29. The VoIP call appears to be initiated by the building's gateway, not ours. The missing acknowledge fails to be sent from the building's gateway to our Aastra phone. |
| 16:40.03 | mweichert | mort_gib: this would indicate that it's a problem at the building's isp level, correct - not our router/local NAT? |
| 16:42.44 | oglynn | benitwv: are you using the external ip address or a dnd name? |
| 16:43.05 | mweichert | mort_gib: sorry, I was very wrong about what I just said |
| 16:43.08 | beniwtv | oglynn: It's an external IP. |
| 16:44.03 | beniwtv | oglyn: But why would that matter? I mean, the authenticate fine from the inside, and they seem to have no connection problems. In fact, * responds wrong password to the SIP client, which displyas the message then. |
| 16:44.36 | mweichert | could someone have a look at the following please? http://pastie.org/448746 |
| 16:44.58 | mweichert | sometimes the ACK sent from 205.16.203.5 to 192.168.101.215 doesn't get sent and the call is dropped... |
| 16:45.04 | mweichert | but I don't know why or how to fix |
| 16:45.18 | oglynn | benitwv: i had issues when using the dns name from the sip client and the machine was not truely that name |
| 16:45.42 | *** join/#asterisk BlackRat (n=ulmik@balticom-17-8.balticom.lv) |
| 16:46.21 | BlackRat | ahoi maybe anyone got clue what fallowing errors means: |
| 16:46.21 | BlackRat | [Apr 16 19:42:34] WARNING[5636]: channel.c:2948 set_format: Unable to find a codec translation path from 0x400 (ilbc) to 0x8 (alaw) |
| 16:46.22 | BlackRat | [Apr 16 19:42:34] WARNING[5481]: chan_dahdi.c:5406 dahdi_write: Cannot handle frames in 1024 format |
| 16:47.03 | oglynn | benitvw: habe you got host=dynamic and nat=yes in sip.conf |
| 16:47.49 | oglynn | benitvw: when you are trying to connect to the external ip are you doing so from an external address? |
| 16:49.55 | *** join/#asterisk grEvenX (n=even@cB78D5AC1.dhcp.bluecom.no) |
| 16:50.50 | beniwtv | oglynn: yes, host is dynamic.... and I'm trying from an external IP. Maybe I need to verify the nat setting? |
| 16:52.16 | oglynn | beniwtv: are they truely on an external address or external and behind nat? |
| 16:53.10 | *** join/#asterisk grEvenX (n=even@cB78D5AC1.dhcp.bluecom.no) |
| 16:53.23 | beniwtv | oglynn: There is a pix in front of it, but it's allowed access. As I said, they seem to be communicating with * but it just refuses them. |
| 16:54.57 | oglynn | from the * cli if you sip set debug on |
| 16:55.09 | oglynn | and compare both registration attempts? |
| 16:59.03 | beniwtv | oglynn: 100% the same for the REGISTER event, but in case of the external IP I get SIP/2.0 403 Forbidden (Bad auth) as response. Maybe there's some restriction by default in Debian which prevents access from non-local subnet IP's? |
| 16:59.47 | *** part/#asterisk dandre (n=daniel@was59-3-82-236-48-30.fbx.proxad.net) |
| 17:01.14 | [TK]D-Fender | beniwtv: nope. |
| 17:01.28 | *** join/#asterisk toorima (n=bq@ip68-7-79-241.sd.sd.cox.net) |
| 17:01.29 | beniwtv | oglynn: oh, and both attempts are made with the same software, both the same configured, so I'd rule that out too |
| 17:02.31 | beniwtv | [TK]D-Fender: Maybe it's wanting MD5 for them? (I just have plaintext for now until I get this working) |
| 17:10.27 | hardwire | iaxy are sool cool.. when did they become unfashionable? |
| 17:11.20 | *** join/#asterisk ACiDV (n=acidv@97-170.dr.cgocable.ca) |
| 17:11.26 | [TK]D-Fender | hardwire: About 5 years ago |
| 17:11.32 | hardwire | what changed? |
| 17:11.36 | hardwire | cause I love these little buggers |
| 17:12.11 | [TK]D-Fender | the lack of all the common call feaures, jacked up price, fugly, no web admin, usable only with *, blah blah blah |
| 17:12.19 | [TK]D-Fender | hardwire: they mean well... |
| 17:12.22 | hardwire | sure |
| 17:12.30 | hardwire | looks like they still sell em |
| 17:12.40 | hardwire | I'm going through non-used inventory and I found.. 10 |
| 17:12.41 | hardwire | heh |
| 17:12.45 | hardwire | testing now.. ebay here I come. |
| 17:13.16 | hardwire | we used em for fax to iaxmodem for a while until we reorganized the office. |
| 17:14.05 | [TK]D-Fender | hardwire: I see no advantage to them for that |
| 17:14.13 | Katty | boingboingboing |
| 17:14.23 | hardwire | [TK]D-Fender: you don't work in an ass backwards place like I do. |
| 17:14.24 | hardwire | heh |
| 17:14.30 | hardwire | contains Katty |
| 17:14.40 | Katty | :< |
| 17:14.41 | [TK]D-Fender | hardwire: Correct... I don't buy IAXY's :) |
| 17:15.08 | Katty | why would you do that? :_< |
| 17:15.45 | imcdona | anyone know of a free tool that I can use to test the VoIP QOS on my network? |
| 17:17.49 | *** join/#asterisk pmhaddad-work (n=pmhaddad@141.219.87.43) |
| 17:18.21 | beniwtv | oglynn, [TK]D-Fender: So any more ideas? |
| 17:18.58 | [TK]D-Fender | beniwtv: Yes, start showing something useful like configs, identical reg attempts from in/out with SIP debug enabled, etc. |
| 17:19.13 | [TK]D-Fender | beniwtv: So far we see nothing and anything is a guess including "user error". |
| 17:20.34 | Katty | eppigy: i had wendy's for lunch. |
| 17:20.39 | Katty | eppigy: ultimate grill chicken sammich |
| 17:20.57 | Katty | eppigy: it was not worth 6 bucks :< |
| 17:21.19 | Katty | eppigy: do you know how much chicken i can buy for 6 bucks?! |
| 17:21.41 | Katty | eppigy: 3lbs :P |
| 17:21.56 | pmhaddad-work | wishes there was a Wendy's in Houghton |
| 17:23.18 | kc8pxy | is on his way to wnedy's |
| 17:24.35 | rob0 | is disgusted by fast food prices now ... WTF do they think they are? |
| 17:25.00 | pmhaddad-work | rob0, i know, its like $5.80 for a QPC and McD's |
| 17:25.03 | pmhaddad-work | *at |
| 17:25.20 | pmhaddad-work | i haven't had wendy's in like a year though |
| 17:25.42 | *** join/#asterisk jameswf (n=james@unaffiliated/jameswf-home) |
| 17:25.54 | *** join/#asterisk hi365 (n=hi365@94.159.144.212) |
| 17:26.27 | rob0 | McD and the Bell are the only ones that still have cheap choices, where you can get a bellyful of imitation food for $3 or so. |
| 17:26.29 | Nivex | kc8pxy de n8vnr/4 |
| 17:26.41 | seb- | [TK]D-Fender: got a few minutes right now? |
| 17:26.50 | kc8pxy | <PROTECTED> |
| 17:26.58 | kc8pxy | QSO |
| 17:27.16 | Nivex | I live in North Carolina now. Originally from Bowling Green, OH |
| 17:27.37 | pmhaddad-work | rob0, i eat at the bell a lot |
| 17:27.57 | kc8pxy | doh.. ok.. i've not been on the air for a bit. |
| 17:28.15 | beniwtv | [TK]D-Fender: Here's SIP debug. From external: http://pastebin.com/m74490505 and from internal: http://pastebin.com/m729e5ce9 |
| 17:29.33 | beniwtv | [TK]D-Fender: Note that I have X'd out the last two parts of the IP's, for security, and changed the USER and MD5 auth... (but from viewing it I think they seem ok) |
| 17:35.53 | kc8pxy | ok,, i need to go, but i have this logging, my softphone works when i just use the playback, but i added an attempted time condition, and it seems to blow through my GotoIfTime, and never go to voicemail. i get no sounds on the phone, and the softphone locks up(i have to force it to close). need my extensions? |
| 17:37.11 | *** join/#asterisk fun330 (n=jon@181.164.8.67.cfl.res.rr.com) |
| 17:37.42 | fun330 | has anyone experanced a large amount of external registration attempts on their asterisk server??? |
| 17:38.49 | kc8pxy | fun330: someone trying to brute-force into your asterisk? |
| 17:38.51 | [TK]D-Fender | seb-: nothing i can do testing with |
| 17:39.48 | seb- | [TK]D-Fender: you mean you aren't anywhere with an client app? |
| 17:39.55 | [TK]D-Fender | seb-: Correct |
| 17:40.08 | seb- | [TK]D-Fender: can we still chat Sat 12:15pm? |
| 17:41.39 | [TK]D-Fender | seb-: Unfortunately I've lost next Saturday to family staying in town. Sunday is open. |
| 17:41.51 | [TK]D-Fender | seb-: I'm hosting fri/sat |
| 17:42.00 | fun330 | i just had someone run a program that ran from ext # 100 all the way throuhg 999 |
| 17:42.05 | [TK]D-Fender | beniwtv: configs.. |
| 17:42.12 | [TK]D-Fender | (minus PW) |
| 17:42.22 | seb- | [TK]D-Fender: any time today? if not then how about Sun 11am? |
| 17:42.41 | seb- | [TK]D-Fender: PST |
| 17:42.47 | fun330 | yeah i think so |
| 17:42.57 | *** join/#asterisk jstoker (n=jstoker@80-41-183-84.dynamic.dsl.as9105.com) |
| 17:43.01 | [TK]D-Fender | seb-: Sun sounds workable |
| 17:43.19 | seb- | [TK]D-Fender: ok..i'll look for you here Sun 11am PST |
| 17:43.33 | *** part/#asterisk jstoker (n=jstoker@80-41-183-84.dynamic.dsl.as9105.com) |
| 17:43.43 | mweichert | hello - any Aastra users here? I noticed that in the configuration for my Aastra phones, the RTP port is set to 3000... and it's the same for every phone we have. Would that not cause conflicts? |
| 17:46.33 | beniwtv | [TK]D-Fender: Here's SIP config: http://pastebin.com/m1398df21. Anything else I should provide? |
| 17:46.37 | fun330 | is this a good solution for brute force? http://etel.wiki.oreilly.com/wiki/index.php/Asterisk_Brute_Force_Prevention |
| 17:49.29 | kc8pxy | fun330: looking for insecure voicemail to trash the greeting of??:) |
| 17:50.10 | [TK]D-Fender | beniwtv: Your remote phone should be listed as being behind NAT. Is your server as well? |
| 17:50.14 | *** join/#asterisk lanning (n=lanning@nat/yahoo/x-10dfcf14fb814c5e) |
| 17:51.24 | beniwtv | [TK]D-Fender: so nat=yes? And yes, the server is behind a pix, but it's allowed in the rules (sip/iax/rtp port range) |
| 17:51.30 | fun330 | kc8pxy: what are you asking? |
| 17:51.37 | [TK]D-Fender | beniwtv: PIX = severe PITA. |
| 17:51.53 | [TK]D-Fender | beniwtv: Go read the guide and make sure the PIX is not doing any transform on the SIP |
| 17:51.56 | [TK]D-Fender | ~sipnat |
| 17:51.57 | infobot | [~sipnat] Quick guide on configuring * + SIP behind NAT : http://www.aocomputing.net/?p=3 , otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
| 17:51.58 | [TK]D-Fender | ^^^^^^ |
| 17:52.35 | Katty | eyes [TK]D-Fender |
| 17:52.47 | beniwtv | [TK]D-Fender: yeah, agreed... it's a pita but I think they're communicating fine (so I'm still not sure about the nat) |
| 17:54.36 | [TK]D-Fender | beniwtv: Well you are running Zoiper on both sides, and I'd like to think I can trust the secret is identical on both, so recheck that another dozen times, and go fix your NAT setup |
| 17:55.51 | *** join/#asterisk sah-work (n=Bawbatos@65-119-47-100.dia.static.qwest.net) |
| 17:57.40 | *** join/#asterisk Superbartt (n=bart@ipd50a21c9.speed.planet.nl) |
| 17:58.10 | kc8pxy | fun330: sorry. i was thinking of what they would be trying, i had not scrolled down. |
| 17:58.39 | kc8pxy | <PROTECTED> |
| 17:58.40 | eppigy | Katty: haha i am sorry to hear that |
| 17:58.48 | eppigy | I am about to dip out and see a braves game |
| 17:59.44 | *** join/#asterisk beniwtv (n=beniwtv@124.Red-83-36-62.staticIP.rima-tde.net) |
| 18:00.03 | beniwtv | [TK]D-Fender: Sorry, lost connection. Any ideas? |
| 18:00.11 | mweichert | how to I specify the codec to use in a .call file? |
| 18:01.09 | [TK]D-Fender | beniwtv: http://pastebin.com/m17d1a79f |
| 18:01.13 | [TK]D-Fender | mweichert: You don't |
| 18:01.36 | [TK]D-Fender | mweichert: It will use whatever is conifgured by the devices you specify |
| 18:02.10 | beniwtv | [TK]D-Fender: Thanks, will try to read a bit more about NAT and asterisk |
| 18:02.55 | mweichert | [TK]D-Fender, in my SIP channel I have allow=ulaw, but then I try to place a call to that channel using a .call file, I get "no audio format found to offer. Cancelling call to XXXXXXXXXX" |
| 18:03.23 | mweichert | There has to be something else |
| 18:04.17 | [TK]D-Fender | mweichert: and you aren't showing anything |
| 18:05.45 | *** join/#asterisk jmacz (n=jmacz@190.144.75.22) |
| 18:10.09 | mweichert | [TK]D-Fender, here is the sip debug log: http://pastie.org/448853 |
| 18:12.05 | mweichert | [TK]D-Fender, this is my custom dialplan: |
| 18:12.05 | mweichert | http://pastie.org/448854 |
| 18:13.02 | mweichert | this is my bash script which creates the .call file: http://pastie.org/448856 |
| 18:13.05 | BlargMaN00 | asterisk -r |
| 18:13.09 | BlargMaN00 | oops |
| 18:13.28 | mweichert | [TK]D-Fender, I don't have the 1111 extension in sip.conf, is that a problem? |
| 18:14.09 | mweichert | [TK]D-Fender, I figured it was more like a phone feature, not an extension per say |
| 18:18.10 | *** join/#asterisk epaphus (n=unix3@190.10.68.228) |
| 18:18.45 | epaphus | Hi guys, quick question... is it true that aside from asterisk ... I need to install also an "interface" ...? whats the use of it? ... iam just planning to install asterisk for thr first time.. just needed to outline what i would need |
| 18:19.19 | *** join/#asterisk propellerhead (n=yogurt2u@host130.190-226-46.telecom.net.ar) |
| 18:20.05 | mweichert | [TK]D-Fender, sorry, not to pester - but any ideas? I'm at wits end |
| 18:20.47 | *** join/#asterisk voxter (n=voxter@76.77.91.250) |
| 18:21.11 | [TK]D-Fender | mweichert: And like always, something crucial missing : the PEER DATA |
| 18:21.12 | mweichert | epaphus, that would depend on what you're trying to do. I assume you're referring to a PSTN interface. What are you trying to do? |
| 18:21.29 | [TK]D-Fender | mweichert: And you should fix your chown BEFore you move the file over |
| 18:22.49 | mweichert | [TK]D-Fender, sorry http://pastie.org/448867 |
| 18:22.58 | mweichert | [TK]D-Fender, okay I'll do that now. |
| 18:23.34 | *** join/#asterisk apeiron (n=Chris@c-76-124-252-61.hsd1.pa.comcast.net) |
| 18:25.37 | [TK]D-Fender | epaphus: A computer to install * on. |
| 18:26.44 | *** join/#asterisk tfrew (n=tfrew@office.neteasyinc.com) |
| 18:27.00 | *** join/#asterisk pa (n=pa@unaffiliated/pa) |
| 18:27.07 | tfrew | does anyone know why asterisk wouldn't log a uniqueid in the cdr database |
| 18:27.48 | [TK]D-Fender | tfrew: Maybe because you didn't configure it to do so. IIRC you have to compile that option in. |
| 18:28.08 | tfrew | so diguims rpms does not have that set when they compile? |
| 18:28.13 | tfrew | that seems kind of pointless |
| 18:29.07 | [TK]D-Fender | tfrew: there are docs for this on the WIKI you might want to look up.. may be other config options related to it.. |
| 18:30.23 | *** join/#asterisk Deeewayne (n=dwayne@95.104.105.2) |
| 18:30.23 | *** mode/#asterisk [+o Deeewayne] by ChanServ |
| 18:31.47 | [TK]D-Fender | mweichert: "sip show peer link2voip-sw1" |
| 18:33.19 | *** join/#asterisk MaliutaLap (n=biteme@kiev.lusan.id.au) |
| 18:33.23 | mweichert | [TK]D-Fender, http://pastie.org/448880 |
| 18:34.18 | [TK]D-Fender | mweichert: I'd want to see a raw call file pre-move |
| 18:34.32 | [TK]D-Fender | mweichert: and please confirm your version of * |
| 18:35.19 | mweichert | [TK]D-Fender, Asterisk 1.4.22 |
| 18:35.58 | Merlin | If I am trying to setup and e-mail to fax gateway solution using T.38 and Asterisk 1.6, what part of the setup does the e-mail to fax, and fax to e-mail part? Do I need scripts? Is that included in Asterisk 1.6? |
| 18:37.22 | mweichert | [TK]D-Fender, http://pastie.org/448890 |
| 18:37.22 | *** join/#asterisk richard_ (n=richard@69.77.169.14) |
| 18:37.56 | mweichert | [TK]D-Fender, it wouldn't be the case that I need an additional newline character at the end of the .call file, no? |
| 18:38.49 | mweichert | [TK]D-Fender, I see I was using chmod rather than chown though! |
| 18:38.58 | [TK]D-Fender | ... |
| 18:39.04 | [TK]D-Fender | mweichert: FIX |
| 18:39.12 | [TK]D-Fender | mweichert: And remove that codec line fromt he file |
| 18:39.22 | [TK]D-Fender | (the var you're setting) |
| 18:40.16 | mweichert | [TK]D-Fender, done... unfortunately the same results |
| 18:40.44 | [TK]D-Fender | mweichert: Ok, no idea. |
| 18:41.09 | mweichert | [TK]D-Fender, thanks for your time and efforts. |
| 18:41.16 | mweichert | [TK]D-Fender, i appreciate your help. |
| 18:42.09 | [TK]D-Fender | mweichert: Wait.. 1 last thing : "core show modules like codec" |
| 18:42.39 | Juggie | anyone use queue metrics? have a quick q. |
| 18:44.39 | mweichert | [TK]D-Fender, that command doesn't exist. I can do "core show codecs" ? |
| 18:44.54 | bpgoldsb | Grrr, all this documentation is in latex files |
| 18:45.48 | [TK]D-Fender | mweichert: show modules like codec |
| 18:46.24 | mweichert | [TK]D-Fender, http://pastie.org/448904 |
| 18:47.05 | [TK]D-Fender | mweichert: Ok, I would recommend you make 1 large doc including all of this info to provide the dev's to try to debug this |
| 18:47.08 | mweichert | when I dial 1111, gsm and ulaw have a use count of 1 |
| 18:48.34 | mweichert | [TK]D-Fender, do I have to create a sip channel for each extension I create in the dialplan? |
| 18:49.49 | [TK]D-Fender | mweichert: you misunderstand what a call-file does. It calls the channel specified the same way Dial() would, and the when it answers it gets dumped into the applition or spot you specify in the dialplan. "extension" has hothing to do with the channel |
| 18:50.07 | [TK]D-Fender | application* |
| 18:51.14 | mweichert | no, I understand that. What I'm not 100% sure on, call file aside, is that if I specify an extension in a dialplan, such as play or 1111, do I have to create a sip channel in the sip.conf file? |
| 18:52.26 | mweichert | I guess what I don't get is - when do I NEED to define a channel? |
| 18:53.36 | *** join/#asterisk nullable_type (n=nullable@hq.verbx.net) |
| 18:53.57 | nullable_type | Has anyone used Sansay VSX Session Border Controller? |
| 18:54.14 | nullable_type | I keep on getting Reinvite Rejected |
| 18:54.44 | nullable_type | I am using Asterisk 1.4 and wondering it is an issue with Asterisk or SBC |
| 18:55.12 | [TK]D-Fender | mweichert: you don't create a "sip channel", it uses your PEER to call out. You're fine... |
| 18:55.54 | *** join/#asterisk mutilator (i=muti@65.111.212.94) |
| 18:56.18 | mutilator | my awesome has arrived |
| 18:56.21 | mutilator | you can now relax |
| 18:56.49 | mutilator | ok, now.. start talking |
| 18:56.57 | bpgoldsb | anyone know if the asterisk 1.6 doc/ directory is hosted somewhere in non-tex format? |
| 18:59.11 | [TK]D-Fender | bpgoldsb: Noone has converted and advertised publicy, but the doc I referred you to is perfectly readable as plain-text with what minimal TEX is really in there |
| 18:59.22 | [TK]D-Fender | bpgoldsb: I've done it so just go read |
| 19:00.00 | mutilator | anyone know who runs voip.com or if they run asterisk? |
| 19:01.22 | *** join/#asterisk jstoker (n=jstoker@80-41-183-84.dynamic.dsl.as9105.com) |
| 19:01.40 | *** join/#asterisk scardinal (n=supreme@0905ds1-rdo.0.fullrate.dk) |
| 19:02.45 | *** part/#asterisk jstoker (n=jstoker@80-41-183-84.dynamic.dsl.as9105.com) |
| 19:11.56 | pmhaddad-work | mutilator, actually i know the lead web designer that designed the backend for the site |
| 19:12.08 | pmhaddad-work | not 100% sure if they use asterisk or not though i can find out |
| 19:15.07 | pmhaddad-work | mutilator, nope, doesn't sound like it |
| 19:20.03 | nullable_type | hey guys anyone have tried SIP Reinvite with or without a Session Border Controller? I keep on getting Reinvite rejection |
| 19:20.07 | nullable_type | without a reason |
| 19:36.15 | bpgoldsb | Is there a way to check the value of a hint directly from the dialplan? i.e. see if a sip peer is in state 'InUse'? |
| 19:38.16 | Qwell | sip show peer? |
| 19:38.20 | *** join/#asterisk jmacz (n=jmacz@190.144.75.22) |
| 19:38.25 | Qwell | oh, dialplan, duh |
| 19:38.49 | Qwell | you can use func_devstate or whatever it's called, probably |
| 19:40.23 | denon | bpgoldsb: http://www.voip-info.org/wiki/view/Asterisk+func+Devstate |
| 19:40.32 | denon | you need 1.6, or the 1.4 backport |
| 19:40.40 | bpgoldsb | Using 1.6 |
| 19:40.43 | [TK]D-Fender | bpgoldsb: "core show application chanisavail" |
| 19:41.07 | bpgoldsb | [TK]D-Fender, The thing I'm wanting to check isn't actually a channel. |
| 19:41.23 | [TK]D-Fender | bpgoldsb: You said "sip peer". I have answered that |
| 19:49.16 | jjshoe | Macro exited with status 49 |
| 19:49.19 | jjshoe | what's status 49? :) |
| 19:50.04 | *** join/#asterisk degrade (n=degrade@unaffiliated/degrade) |
| 19:50.13 | degrade | Has anyone using Asterisk Over Solaris/SUN? |
| 19:52.37 | rob0 | 7 more than 42! |
| 19:56.40 | *** join/#asterisk watchy (n=watchy@76.196.98.139) |
| 19:59.15 | watchy | I have 2 phone systems connected together by IAX, I have a secretary at one end and with a 601 and 3 attendant modules, anyway to add the extensions of others on the other phone system and have presence? |
| 19:59.50 | mmlj4 | degrade: there's a sun voip server, probably it's an asterisk bundle for slowlaris |
| 20:00.07 | rob0 | ponders what it might mean to have a secretary at one end ... |
| 20:00.14 | *** join/#asterisk Chainsaw (n=chainsaw@gentoo/developer/atheme.member.chainsaw) |
| 20:01.25 | watchy | rob: hmm a secretary at 1 location? |
| 20:03.08 | degrade | mmlj4: slowlaris? I heard that Solaris have a good performance. I'm interested to run Asterisk in Solaris/SUN to see how they performe. |
| 20:03.09 | watchy | isn't sun dead pretty much? |
| 20:06.48 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 20:08.11 | *** part/#asterisk BlackRat (n=ulmik@balticom-17-8.balticom.lv) |
| 20:12.01 | jaytee | it looks like the talks for IBM to buy Sun pretty much died and that was about the only thing that would breath new life into Sun |
| 20:12.31 | watchy | well sun has sucked for years |
| 20:12.54 | watchy | they don't even use anything but x86/64bit anymore right? |
| 20:14.08 | *** join/#asterisk mweichert (n=mweicher@216.13.154.21) |
| 20:14.16 | Qwell | watchy: they still sell sparc systems |
| 20:14.28 | watchy | wow |
| 20:14.36 | watchy | i wonder why |
| 20:14.47 | Qwell | because for some applications is far better? |
| 20:15.11 | watchy | maybe, but apparently it doesn't do them well profit wise |
| 20:15.33 | watchy | well something isn't anyways |
| 20:27.22 | *** join/#asterisk raub (n=raub@mail2.rfengineers.com) |
| 20:28.58 | *** join/#asterisk Katty (n=asterisk@mail.copi-rite.com) |
| 20:30.00 | xheliox | random telecom question -- do area codes cross state lines? |
| 20:30.01 | C4colo | Random answer[75]: butt cheeks (from Phyre) |
| 20:34.14 | Corydon76-dig | xheliox: Yes, they do, but usually not by much |
| 20:34.48 | xheliox | Do you have any examples of that? Because I can't find any |
| 20:35.23 | Corydon76-dig | xheliox: Sure, Ardmore, TN, has 256 area code numbers |
| 20:35.43 | Corydon76-dig | 256 is mostly Alabama, though |
| 20:36.17 | Corydon76-dig | If Ardmore, TN was exclusively on the TN side, they'd be 931 area code, not 256 |
| 20:37.23 | rob0 | uh, really? I know Ardmore's on the state line, they are, legally speaking, two separate municipalities. |
| 20:37.25 | *** join/#asterisk mog (n=mog@c-68-62-169-246.hsd1.al.comcast.net) |
| 20:37.25 | *** mode/#asterisk [+o mog] by ChanServ |
| 20:37.46 | Corydon76-dig | rob0: correct, but both sides use 256 numbers |
| 20:38.02 | rob0 | I grew up a few blocks from a state line, and we had 816, the other side had 913. |
| 20:38.36 | Corydon76-dig | rob0: how rural/urban were you? |
| 20:38.37 | rob0 | it was and still is a local call across the line, however. |
| 20:38.46 | rob0 | that was Kansas City |
| 20:38.56 | Corydon76-dig | Most of it is a matter of convenience to the local telco |
| 20:38.59 | xheliox | Bristol, TN and Bristol, VA has different area codes. |
| 20:39.00 | rob0 | pretty much urban |
| 20:42.54 | mmlj4 | degrade: http://www.solarisvoip.com/ |
| 20:43.13 | watchy | anyone use devstate in 1.4? |
| 20:43.20 | mmlj4 | degrade: and no, slowlaris has always had a reputation of being, um, SLOW |
| 20:43.42 | xheliox | Well, thanks guys. |
| 20:43.54 | *** join/#asterisk putnopvut (n=putnopvu@asterisk/master-of-queues/mmichelson) |
| 20:43.54 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 20:43.57 | *** join/#asterisk jpeeler (n=jpeeler@asterisk/digium-software-dev/jpeeler) |
| 20:44.29 | *** join/#asterisk mnicholson (n=mnichols@nat/digium/x-6e0db4074b36f880) |
| 20:44.33 | *** join/#asterisk The_Boy_Wonder (n=davidvos@asterisk/batman-developer/dvossel) |
| 20:45.47 | *** join/#asterisk voxter (n=voxter@76.77.95.2) |
| 20:46.10 | *** join/#asterisk fish-bulb (n=cstewart@nat/digium/x-c4d087647585b86f) |
| 20:48.43 | *** part/#asterisk mutilator (i=muti@65.111.212.94) |
| 20:51.21 | *** join/#asterisk jeffgus (n=jeffgus@green.zimage.com) |
| 20:53.09 | *** join/#asterisk mattchis (n=mchisenh@c-98-198-204-178.hsd1.tx.comcast.net) |
| 20:53.09 | *** join/#asterisk ReDNeQ (n=ReDNeQ@70.114.226.70) |
| 20:55.57 | *** join/#asterisk unspin (n=unspin@96.49.129.159) |
| 20:56.14 | Katty | humm |
| 20:56.19 | Katty | too quiet |
| 20:56.34 | file | tackles Katty |
| 20:57.07 | Katty | meep. |
| 20:57.52 | watchy | tacos |
| 20:58.28 | *** join/#asterisk joako (n=joako@opensuse/member/joak0) |
| 20:58.30 | Katty | http://www.youtube.com/watch?v=UcWsTwvtyOI |
| 20:58.56 | Katty | i wonder if that's how republican think america will turn out like under obama's presidency |
| 20:59.29 | mattchis | Can some help me on an issue with Realtime? I am testing out 1.4.24.1 and all of my peers are showing up as Cached RT. I have set the server in full debug mode and do not see any issues with asterisk communicating with mysql. |
| 21:01.00 | voxter | any of you guys familiar with the asterisk.org/asterisknow centos repository? I cant find which package res_jabber might be in. |
| 21:01.02 | watchy | im republican |
| 21:01.09 | *** join/#asterisk BadHAL (n=nn@66.194.174.11) |
| 21:03.16 | slaney | heh |
| 21:03.44 | Katty | watchy: and? |
| 21:03.47 | Katty | watchy: what do you think? |
| 21:04.37 | watchy | i didnt watch it |
| 21:04.46 | watchy | but i hate obama, and people who want my money |
| 21:04.55 | *** join/#asterisk hi365 (n=hi365@94.159.144.212) |
| 21:05.00 | Katty | well thank you for your informational input. |
| 21:05.10 | watchy | but i'll watch the video |
| 21:05.17 | Katty | don't worry about it |
| 21:05.27 | watchy | but if its about poor people wanting my money it will depress me |
| 21:07.46 | watchy | haha that comercial is funny |
| 21:08.48 | *** join/#asterisk epaphus (n=unix3@190.10.68.228) |
| 21:09.29 | watchy | are you republican or democrat katty? |
| 21:13.07 | *** join/#asterisk vvuja (n=vvuja@79.175.115.174) |
| 21:13.09 | vvuja | hello |
| 21:13.12 | vvuja | anyone here |
| 21:13.36 | vvuja | i have some bussines proposal for you |
| 21:13.43 | slaney | heh |
| 21:13.43 | vvuja | i know that it will not work |
| 21:13.47 | vvuja | but i can try |
| 21:13.49 | vvuja | what else |
| 21:14.27 | vvuja | are you interested? |
| 21:14.46 | Katty | no |
| 21:14.52 | vvuja | ok sorry |
| 21:15.00 | vvuja | it concerns asterisk and SIP telephony |
| 21:15.01 | vvuja | over internet |
| 21:15.09 | vvuja | i'm in Serbia |
| 21:15.09 | watchy | will it make me rich? |
| 21:15.19 | vvuja | here is very hard to start your own bussines |
| 21:15.22 | Qwell | SIP works over the Internet? |
| 21:15.26 | vvuja | of course |
| 21:15.32 | Qwell | good to know |
| 21:15.43 | Katty | Qwell: he told you. |
| 21:15.54 | watchy | pwned. |
| 21:16.03 | vvuja | ok if you are not interested |
| 21:16.09 | Katty | nope |
| 21:16.10 | Qwell | vvuja: we aren't, no |
| 21:16.12 | *** join/#asterisk jmacz (n=jmacz@190.144.75.22) |
| 21:16.16 | watchy | well everyone here uses asterisk cause its free |
| 21:16.22 | watchy | they are to poor here to front a business |
| 21:16.36 | vvuja | yes but my service will cost maybe 5 euros a month |
| 21:16.36 | watchy | otherwise theyd run cisco |
| 21:16.57 | vvuja | of course I can not beat skype |
| 21:17.00 | Katty | Qwell: is there some policy on solicitation? |
| 21:17.01 | vvuja | because it is free |
| 21:17.17 | Qwell | vvuja: spit it out already |
| 21:17.22 | vvuja | ok idea is |
| 21:17.33 | vvuja | i want to buy a dedicated server in USA |
| 21:17.35 | vvuja | from Serbia |
| 21:17.43 | vvuja | on that server to install asterisk |
| 21:17.53 | vvuja | it will be on public IP address |
| 21:18.01 | watchy | hopefully you can beat skype, they don't make a profit or even break even |
| 21:18.08 | rob0 | I HAVE SUM OF FOR MILION DOLARS USA WOULD LIKE TO SHARE WITH YOU |
| 21:18.11 | vvuja | already tried people behind NAT can communicate |
| 21:18.20 | Qwell | vvuja: okay, and? |
| 21:18.21 | vvuja | so idea is to connect people on my SIP server |
| 21:18.26 | Katty | rob0: lol |
| 21:18.33 | Qwell | vvuja: uh huh? |
| 21:18.37 | vvuja | for 5 euros / month |
| 21:18.54 | Qwell | let me stop you right there. |
| 21:18.58 | vvuja | ok |
| 21:19.00 | vvuja | i'm reading |
| 21:19.04 | Qwell | Are you saying you want to be the next Vonage? |
| 21:19.10 | vvuja | don't know |
| 21:19.17 | vvuja | trying to make some buck |
| 21:19.20 | denon | Vonage is still around? |
| 21:19.24 | vvuja | and to connect people for little money |
| 21:19.24 | Qwell | denon: sadly |
| 21:19.24 | Katty | rob0: make some buck. |
| 21:19.31 | Qwell | vvuja: okay, and how is this novel? |
| 21:19.33 | vvuja | 80 euros is server in USA |
| 21:19.34 | denon | Qwell: how'd we let that one slide |
| 21:19.40 | vvuja | maybe 100 euros |
| 21:19.52 | vvuja | so if I connect 100 people for 5 euros |
| 21:19.54 | vvuja | it is 500 |
| 21:19.59 | vvuja | i can pay server for a month |
| 21:20.08 | vvuja | and that leaves me 400 euros |
| 21:20.09 | Katty | that seems like it would be a violation of some trade agreement |
| 21:20.15 | Qwell | 400 euros to pay for minutes? |
| 21:20.18 | vvuja | no no |
| 21:20.20 | Qwell | for 100 users? |
| 21:20.24 | vvuja | 400 euros profit |
| 21:20.37 | vvuja | 100 euro is server |
| 21:20.39 | vvuja | for a month |
| 21:20.39 | Qwell | and how are you paying for minutes? |
| 21:20.43 | vvuja | no minutes |
| 21:20.47 | vvuja | 5 euros / month |
| 21:20.50 | Katty | Qwell: oh well there you go. |
| 21:20.50 | Qwell | then what is the point of your service? |
| 21:20.52 | Katty | Qwell: no minutes. |
| 21:20.55 | vvuja | connect and speak whenever you want |
| 21:21.04 | Qwell | to...who? |
| 21:21.13 | vvuja | to people connected on that server |
| 21:21.25 | vvuja | in some future maybe connection to some service that can put calls on landlines |
| 21:21.27 | Qwell | I... |
| 21:21.29 | vvuja | don't know |
| 21:21.32 | vvuja | just thinking |
| 21:21.33 | vvuja | idea |
| 21:21.40 | vvuja | no one in Serbia supports |
| 21:21.47 | Qwell | ...Skype |
| 21:21.54 | *** join/#asterisk SomethingIsodd (n=dan@iphost-64-56-144-29.cgy.wiband.net) |
| 21:21.57 | vvuja | so I wanted to try with people abroad |
| 21:21.57 | SomethingIsodd | hello everyone. |
| 21:21.59 | vvuja | I know |
| 21:22.01 | Katty | SomethingIsodd: ohai |
| 21:22.01 | vvuja | Skype is free |
| 21:22.16 | Qwell | and about...you know...1000x more reliable |
| 21:22.23 | Katty | vvuja: why would someone pay 5 euros a month, when they can get skype for free, and already have tons of users and good reliability? |
| 21:22.25 | vvuja | but I think that connection with g711u codec can be better than skype |
| 21:22.28 | Qwell | with, let's see |
| 21:22.38 | SomethingIsodd | Question i was here a few days ago talking to yous about the issue i was having with the x100p. I have downgraded to Zaptel 1.2 and when i do ztcg -vvv i get ZT_CHANCONFIG failed on channel 1: Inappropriate ioctl for device (25) any ideas please |
| 21:22.38 | Qwell | 1.5 million times the users? |
| 21:22.41 | vvuja | Katty: I know |
| 21:22.44 | rob0 | So basically this is something like freeworlddialup ... |
| 21:22.51 | Qwell | vvuja: Skype uses a wideband codec. |
| 21:22.55 | Katty | rob0: i was going to say more like.. hogwash |
| 21:22.56 | rob0 | ... which is also free |
| 21:22.58 | Qwell | which is VASTLY superior to ulaw |
| 21:23.01 | Superbartt | Could someone please confirm my thoughts: You are unable to connect an ATA ("SIP to Analogue gateway") to an uplink (telco's) PSTN line, but only to an analogue phone device |
| 21:23.03 | vvuja | ok |
| 21:23.09 | vvuja | so that goes under water |
| 21:23.13 | vvuja | no bussines |
| 21:23.13 | rob0 | hogwash might fit better, indeed :) |
| 21:23.17 | vvuja | even for a little money |
| 21:23.18 | vvuja | :) |
| 21:23.26 | vvuja | just asked for opinion |
| 21:23.36 | rob0 | Sorry, but I think you're a bit late to the party. |
| 21:23.37 | *** join/#asterisk leifmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 21:23.37 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 21:23.41 | vvuja | how I will work in Serbia is my stuff |
| 21:23.43 | Katty | mister madsen |
| 21:23.48 | leifmadsen | ms. katty |
| 21:23.50 | vvuja | because I will not have company |
| 21:23.57 | Katty | get a job like a normal person? |
| 21:24.02 | vvuja | trying |
| 21:24.06 | vvuja | that is also a problem |
| 21:24.13 | Katty | i can see why. |
| 21:24.27 | vvuja | why? |
| 21:24.34 | Qwell | GUYS! I just had an idea! |
| 21:24.40 | vvuja | maybe you think i'm trying to rob someone |
| 21:24.43 | Katty | o/~ i didn't mean to start the flame war o/~ |
| 21:24.43 | vvuja | or to fool someone |
| 21:24.49 | vvuja | it is not a point |
| 21:24.52 | Qwell | We take computers, right... and we take "cables" and connect them all together, so they can "talk" to each other |
| 21:24.57 | vvuja | idea is how to start bussines from home |
| 21:25.07 | Qwell | I'm going to make a million dollars. |
| 21:25.11 | vvuja | ok now you are cynical |
| 21:25.19 | Qwell | a bit |
| 21:25.27 | vvuja | ok |
| 21:25.28 | Katty | yes. |
| 21:25.31 | Katty | that's why we love Qwell |
| 21:25.48 | vvuja | I didn't want to ask stupid question |
| 21:25.56 | vvuja | that was just an idea |
| 21:26.01 | Qwell | it wasn't a stupid question, but perhaps you should've thought it out a little more |
| 21:26.06 | vvuja | something needs to start from something |
| 21:26.18 | vvuja | problem is that I can not get dedicated server for free |
| 21:26.23 | Katty | So. |
| 21:26.26 | Katty | I have a business idea. |
| 21:26.29 | vvuja | and I can not get a dedicated server in Serbia |
| 21:26.32 | *** join/#asterisk [TK]D-Fender (n=joe@64.235.218.194) |
| 21:26.37 | Katty | I will setup a bar. |
| 21:26.41 | vvuja | with that much CPU and hard drive |
| 21:26.45 | Katty | With computers. and Wiis. |
| 21:26.48 | Katty | And couches. |
| 21:26.55 | vvuja | :) |
| 21:26.56 | vvuja | ok |
| 21:27.00 | vvuja | I understand |
| 21:27.07 | vvuja | can I stay here and talk with you |
| 21:27.10 | vvuja | ? |
| 21:27.11 | [TK]D-Fender | Katty: and... ALCOHOL.. right? Seem to have left that off :) |
| 21:27.23 | Katty | [TK]D-Fender: bar sorta means... alchohol.. and stuff. |
| 21:27.37 | rob0 | Wii like alcohol. |
| 21:27.45 | Qwell | [TK]D-Fender: no, you missed it |
| 21:27.56 | Katty | [TK]D-Fender: yes, horrible timing this afternoon. |
| 21:28.02 | vvuja | ok thank you for your time |
| 21:28.10 | vvuja | answer was the same as in Serbia |
| 21:28.12 | vvuja | ;) |
| 21:28.17 | SomethingIsodd | Can anyone help me with my issue |
| 21:28.20 | vvuja | just asked for opinion |
| 21:29.12 | *** part/#asterisk vvuja (n=vvuja@79.175.115.174) |
| 21:30.56 | *** join/#asterisk sah-work (n=Bawbatos@65-119-47-100.dia.static.qwest.net) |
| 21:31.32 | *** join/#asterisk popolinux (n=cyril@AToulouse-156-1-114-165.w90-30.abo.wanadoo.fr) |
| 21:31.42 | popolinux | Hello all |
| 21:32.29 | popolinux | I try to configure asterisk to send or recieve fax, but I have a error : channel.c:3059 set_format: Unable to find a codec translation path from g729 to unknown |
| 21:32.45 | popolinux | anybody can tel me what is the problem please ? |
| 21:32.50 | [TK]D-Fender | Fax over G.729... lol |
| 21:33.09 | SomethingIsodd | popolinux you can only Fax over G711 if i recall correctly |
| 21:33.32 | popolinux | ok ! I'm a newbie... |
| 21:33.51 | popolinux | I add allow=g711 on sip.conf to do that ? |
| 21:33.59 | Qwell | [TK]D-Fender: you missed the fun earlier. |
| 21:34.30 | [TK]D-Fender | Qwell: Did I? |
| 21:34.37 | SomethingIsodd | popolinux yes give it a try. but to be honest. I would not be a good source, as i have only been able to get ASterisk to fax 1 or two pages and they never turn out the best. |
| 21:34.39 | Qwell | should definitely look at IRC logs from about 15 minutes ago |
| 21:34.55 | [TK]D-Fender | Qwell: pastebin it.. no logs here |
| 21:35.04 | Qwell | you suck. hang on |
| 21:35.11 | popolinux | ok thanks SomethingIsodd I go to test that :) |
| 21:35.13 | *** join/#asterisk vvuja (n=vvuja@79.175.115.174) |
| 21:35.17 | vvuja | hello |
| 21:35.22 | rob0 | wb vvuja |
| 21:35.28 | vvuja | can I stay here and talk with you? |
| 21:35.33 | vvuja | no bussines proposal no anything |
| 21:35.38 | vvuja | it was just an idea |
| 21:35.58 | rob0 | Let's put it to a vote. All in favor? |
| 21:36.02 | rob0 | ;) |
| 21:36.05 | vvuja | :) |
| 21:36.07 | vvuja | no problem |
| 21:36.10 | vvuja | but calculate |
| 21:36.26 | vvuja | 0.17cents |
| 21:36.31 | vvuja | euro cents |
| 21:36.33 | vvuja | per day |
| 21:36.40 | SomethingIsodd | rob0 i am not voting i just want to fix my stupid x100p lol |
| 21:36.58 | Qwell | [TK]D-Fender: msg |
| 21:37.07 | rob0 | We've got the best Florida and Minnesota vote counters working on the tally. They should have the result in 6 years or so. |
| 21:37.10 | vvuja | so when you calculate it is 5e a month |
| 21:37.21 | Qwell | you get 1 minute per day of usage? |
| 21:37.25 | vvuja | no |
| 21:37.30 | vvuja | you get 24h of usage |
| 21:37.33 | vvuja | :) |
| 21:37.36 | vvuja | that is the point |
| 21:37.44 | nullable_type | hey guys anyone have tried SIP Reinvite with or without a Session Border Controller? I keep on getting Reinvite rejection |
| 21:37.47 | nullable_type | without a reason |
| 21:37.58 | vvuja | connect and talk 3h |
| 21:38.03 | rob0 | vvuja, if you can make the sales on your end, you might make it work. But like Microsoft, it's all about marketing, not technology. |
| 21:38.03 | popolinux | Asterisk seems to not find the g711 codec : [Apr 16 23:33:33] WARNING[20399] frame.c: Cannot allow unknown format 'g711' |
| 21:38.13 | Qwell | popolinux: allow=ulaw |
| 21:38.18 | SomethingIsodd | popolinux ulaw or alaw |
| 21:38.19 | SomethingIsodd | sorry |
| 21:38.29 | Qwell | (or alaw if you aren't in the US) |
| 21:38.35 | vvuja | Qwell: connect and talk 3h, then disonnect and then after 1h connect again and talk 3h |
| 21:38.39 | vvuja | Qwell: I don't care |
| 21:38.41 | *** join/#asterisk Pazzo (n=ugelt@sadsl-246059.rol.raiffeisen.net) |
| 21:38.44 | vvuja | Qwell: :) |
| 21:38.47 | popolinux | ok I'm french, thanks I test :) |
| 21:39.20 | SomethingIsodd | Qwell do youhave do you happen toknow anything about the digium x100p ya i know its been discontinued :-) |
| 21:39.25 | vvuja | rob0: problem is that in Serbia it is too much complicated to make a company |
| 21:39.34 | vvuja | rob0: so I will need to work underground |
| 21:39.46 | vvuja | rob0: visa electron to pay me |
| 21:39.47 | vvuja | rob0: visa internet for me to pay dedicated server |
| 21:39.55 | Qwell | so not only is it an extremely flawed plan...it's also illegal? |
| 21:39.56 | vvuja | rob0: I connect to my server and open you account |
| 21:39.58 | [TK]D-Fender | vvuja: There are MANY free services taht do what you're suggesting already. What makes your service something people will care about and you the person they would want to use? |
| 21:40.10 | vvuja | Qwell: No one can confirm that |
| 21:40.23 | vvuja | Qwell: Why will be someone interested why you paid me 5e |
| 21:40.27 | vvuja | ;) |
| 21:40.37 | Qwell | right, so it's illegal |
| 21:40.47 | vvuja | Qwell: Ok so it is illegal |
| 21:40.53 | [TK]D-Fender | Qwell: I've got this business plan involving selling air in a stand on my front lawn. Want to invest some VC in it with me? :) |
| 21:40.59 | vvuja | Qwell: no marketing, no site, close circle of people |
| 21:41.05 | popolinux | I've the same message with alaw : WARNING[24004]: channel.c:3059 set_format: Unable to find a codec translation path from alaw to unknown |
| 21:41.29 | rob0 | I'm selling instant DHMO packets, just add water. |
| 21:41.43 | vvuja | rob0: ACME :) |
| 21:41.44 | [TK]D-Fender | vvuja: You are looking to sell a service they can get for FREE better than they can through you. Why would anybody want to use your service then? |
| 21:41.52 | *** join/#asterisk SirThomas (n=tomc@mail.kendeco.com) |
| 21:42.01 | vvuja | maybe we can try |
| 21:42.08 | vvuja | Fender |
| 21:42.14 | Katty | oh. |
| 21:42.17 | Katty | look who came back. |
| 21:42.17 | [TK]D-Fender | rob0: I'm selling dehydrated water. Want some? ;) |
| 21:42.30 | vvuja | ACME I'm telling you |
| 21:42.34 | popolinux | SomethingIsodd > have you an idea ? |
| 21:42.36 | vvuja | and I'm trying to catch some bird |
| 21:42.42 | vvuja | my real name is Vuk Vujovic |
| 21:42.47 | vvuja | Vuk on Serbian means Wolf |
| 21:42.48 | vvuja | :) |
| 21:43.11 | [TK]D-Fender | vvuja: Your business model is certainly "endangered" |
| 21:43.16 | SomethingIsodd | popolinux no sorry never had that issue |
| 21:43.48 | vvuja | Fender: Thank you, Microsoft wasn't better when they stoled CP/M and then graphic OS from Macintosh |
| 21:43.55 | Qwell | using your real name on a publicly logged channel after admitting to wanted to take part in illegal activities...probably not the smartest thing in the world. |
| 21:44.00 | vvuja | so what they invented |
| 21:44.05 | popolinux | ok, thank you anyway |
| 21:44.12 | watchy | hey tk: you ever do presence over IAX for extensions? |
| 21:44.13 | vvuja | Qwell: Because I'm not doing anything |
| 21:44.22 | vvuja | Qwell: That is the point |
| 21:44.30 | vvuja | Qwell: Just talking |
| 21:44.36 | vvuja | Qwell: Is that forbbiden |
| 21:44.48 | Qwell | In your country? maybe. |
| 21:44.49 | popolinux | someone else would have an idea for my problem please? |
| 21:44.54 | [TK]D-Fender | vvuja: Stole OS from Macintosh? Harly. You also don't much about GUI history... |
| 21:45.02 | rob0 | Oh, the Serbian thugs^Wauthorities are probably not paying attention here anyway. |
| 21:45.16 | vvuja | :) |
| 21:45.40 | vvuja | Qwell: What I did? Nothing? What can you prove? Nothing. Just putted my name, Vuk Vujovic |
| 21:45.43 | [TK]D-Fender | watchy: 1.6 has something concerning distributed presence |
| 21:46.10 | vvuja | Fender: Microsoft didn't invented anything on their own, it was already made, they just stoled idea |
| 21:46.12 | watchy | tk: i see somone added devstate to 1.4, i'm going to try it, i just was curious if you had played with it |
| 21:46.28 | [TK]D-Fender | vvuja: And you have miscredited Apple for it. |
| 21:46.37 | [TK]D-Fender | vvuja: GUI's date much further back |
| 21:46.47 | watchy | in 1.4 you add a .c file some dude wrote, but apparently need a CLI based php script to sync it |
| 21:46.57 | *** join/#asterisk manxpower (n=Administ@router.asteriasgi.com) |
| 21:46.59 | vvuja | Fender: Do you know when Macintosh had GUI, when we worked in DOS which is Gary Kildall's CP/M |
| 21:47.00 | [TK]D-Fender | vvuja: And your business "plan" isn't exactly original either. |
| 21:47.09 | watchy | but if it works i won't complain |
| 21:47.16 | popolinux | [TK]D-Fender > have you an idea please? |
| 21:47.26 | Katty | popolinux: i have an idea. |
| 21:47.41 | [TK]D-Fender | vvuja: I know severeal older than Mac. Your point? |
| 21:47.41 | Katty | popolinux: take 7 shots of vodka, and you won't care anymore. |
| 21:48.04 | Katty | eppigy: what's for dinner. |
| 21:48.09 | vvuja | Fender: Ok, so what Microsoft invented? Tell me, I want to know, because I used computers since C64 |
| 21:48.29 | Katty | Microsoft invented one of hell of a marketing strategy. |
| 21:48.32 | mmlj4 | microsoft invented the forced upgrade |
| 21:48.43 | *** part/#asterisk manxpower (n=Administ@router.asteriasgi.com) |
| 21:48.52 | popolinux | Katty > we only have wine in France unfortunately:) |
| 21:48.57 | vvuja | ok, but original ideas was from people that didn't made to have million dollar account |
| 21:49.05 | nullable_type | Hey, didn't microsoft invent the service pack? |
| 21:49.07 | popolinux | I became resistant! |
| 21:49.13 | Katty | No I'm pretty sure Microsoft's marketing was invented in house. |
| 21:49.26 | Katty | It's truly amazing. |
| 21:49.27 | vvuja | yes yes |
| 21:49.27 | [TK]D-Fender | vvuja: only C-64? I remember UPGRADING from the Pet to the Vic-20. |
| 21:49.36 | vvuja | Bill Gates programmed in Basic |
| 21:49.43 | vvuja | know the whole story |
| 21:49.58 | Qwell | he...programmed basic...in basic? |
| 21:50.09 | vvuja | and had mark of yogurt on his shirt |
| 21:50.15 | vvuja | milk, whatever |
| 21:50.15 | [TK]D-Fender | vvuja: Hate to say it, but you're not impressing us very much..... |
| 21:50.28 | vvuja | Fender: Come to Serbia and you will be impressed |
| 21:50.40 | vvuja | Fender: Here you can not make a million buck even if you try |
| 21:50.52 | [TK]D-Fender | vvuja: that would be DEPRESSED |
| 21:50.57 | vvuja | Fender: So I can find here a job of 500e a month |
| 21:51.04 | Katty | I wonder if that has anything to do with the people that live there. |
| 21:51.19 | Katty | Or perhaps how much time and effort they put into education. |
| 21:51.33 | vvuja | Fender: House cleaning woman in Europe make more money than me |
| 21:51.38 | [TK]D-Fender | Katty: There is a reason eastern-block countries have perfected vodka ;) |
| 21:51.39 | vvuja | so now you know |
| 21:51.51 | [TK]D-Fender | vvuja: And knowing is half the battle... |
| 21:51.53 | Qwell | vvuja: why not go clean houses for a living? |
| 21:51.54 | vvuja | so we are still in eastern-block and western-block |
| 21:51.54 | Katty | What's wrong with house cleaning? |
| 21:52.17 | vvuja | i thought that was over with tearing down the Berlin wall |
| 21:52.18 | Katty | What's wrong with women? |
| 21:52.30 | watchy | i like women who clean houses |
| 21:52.34 | watchy | they are great |
| 21:52.40 | Katty | Really? |
| 21:52.54 | Katty | Sounded pretty negative earlier. |
| 21:52.58 | [TK]D-Fender | mail-orders another Russian "French-maid" |
| 21:53.16 | Katty | [TK]D-Fender: you would. |
| 21:53.19 | vvuja | :) |
| 21:53.31 | vvuja | so it is all about fu***ing other people |
| 21:53.31 | watchy | ill just pay some crackwhore $5 to dress up like one |
| 21:53.37 | *** join/#asterisk lasko (n=lasko@70.102.15.210) |
| 21:53.46 | vvuja | thank you for http://www.thehun.net |
| 21:53.53 | vvuja | and http://www.worldsex.com |
| 21:53.58 | vvuja | I enjoy every day |
| 21:54.02 | [TK]D-Fender | vvuja: No... you can go fuck yourself ;) |
| 21:54.03 | vvuja | just one thing |
| 21:54.08 | Katty | For the low pay that a european house cleaner makes... |
| 21:54.09 | vvuja | we do not do that in Serbia |
| 21:54.11 | Katty | I still can't afford one. |
| 21:54.14 | Katty | That has to say something. |
| 21:54.40 | vvuja | Fender: Oh, thank you, already did that, with million of porn sites made in western-block |
| 21:54.48 | vvuja | Fender: doing that every day |
| 21:54.55 | vvuja | Fender: and enjoying |
| 21:55.11 | Katty | goes home. |
| 21:55.13 | nullable_type | Sorry guys to disturb your "informative" discussion :) Does anyone have had a SIP reinvite work? |
| 21:55.30 | [TK]D-Fender | nullable_type: Pretty much everyone |
| 21:56.14 | vvuja | so I'm now watching www.thehun.net |
| 21:56.16 | vvuja | and laughing |
| 21:56.27 | vvuja | You think you can come here to do the same with Serbian women |
| 21:56.29 | vvuja | you are wrong |
| 21:56.38 | nullable_type | D-Fender >> I am getting a Reinvite Rejected from SBC. I am finding in some forums there is some order problems with Asterisk on Reinvite. Is that true? |
| 21:56.44 | watchy | maybe if you learned asterisk instead of jerking off allday instead of asking people here for help |
| 21:56.59 | vvuja | no brothers |
| 21:57.01 | [TK]D-Fender | "This is the first major spy/mlalware i have ever gotten after visiting TheHun.net. The IP " |
| 21:57.02 | watchy | you could help yourself |
| 21:57.05 | [TK]D-Fender | Do not visit <- |
| 21:57.12 | vvuja | you have the problem |
| 21:57.13 | vvuja | not us |
| 21:57.22 | watchy | you country is a poor shithole |
| 21:57.22 | vvuja | don't worry about us |
| 21:57.26 | watchy | i think you have a problem |
| 21:57.27 | vvuja | yes yes |
| 21:57.39 | [TK]D-Fender | 11 steps to go :) |
| 21:57.41 | vvuja | but in your countries only thing that you know to do is to fuck |
| 21:57.48 | watchy | fedex me some 16 year old serbian girls, and i'll help you |
| 21:58.02 | [TK]D-Fender | ... |
| 21:58.07 | Qwell | enough, right now |
| 21:58.13 | vvuja | ok ok |
| 21:58.24 | vvuja | tomorrow is Veliki Petak in Serbia |
| 21:58.33 | vvuja | orthodox |
| 21:58.35 | watchy | i don't think anyone knows who that is |
| 21:58.40 | watchy | or what that is |
| 21:58.44 | vvuja | and I'm wearing a cross on my chest |
| 21:59.01 | vvuja | so I think that my shithole country is laughing at you |
| 21:59.05 | watchy | is that a holiday? |
| 21:59.13 | vvuja | holiday and such a great holiday |
| 21:59.32 | [TK]D-Fender | Qwell: You were entirely right. |
| 21:59.45 | Qwell | [TK]D-Fender: about what? and am I ever wrong? |
| 21:59.53 | vvuja | so much story about your shitting in face style of living |
| 21:59.57 | vvuja | and pissing in mouth |
| 21:59.59 | watchy | wtf |
| 22:00.01 | *** mode/#asterisk [+b *!*n=vvuja@79.175.115.*] by Qwell |
| 22:00.01 | *** kick/#asterisk [vvuja!i=north@pdpc/sponsor/digium/Qwell] by Qwell (Qwell) |
| 22:00.04 | Qwell | moving on |
| 22:00.13 | [TK]D-Fender | Qwell: Mum's the word on that one, but on this last case... |
| 22:00.28 | rob0 | Do you have a phone number for the Serbian thugs? |
| 22:00.46 | rob0 | Any provider with good rates to Serbia? ;) |
| 22:00.46 | Qwell | [TK]D-Fender: I have no idea what you're talking about, heh |
| 22:01.05 | [TK]D-Fender | Qwell: :p |
| 22:01.31 | nullable_type | I am getting a Reinvite Rejected from SBC. I have canreinvite=yes and directrtpsetup=yes. Is there anything else i need to do? |
| 22:03.11 | *** join/#asterisk blebleble (n=godie@216.222.194.7) |
| 22:03.33 | blebleble | is there a way to monitor users DND status's? |
| 22:03.54 | lasko | yes <3 |
| 22:05.06 | lasko | There are a number of ways to do it depending upon your setup and the kind of phones being used. |
| 22:05.12 | blebleble | hmm guessing i would have to remap the key on my phone to something asterisk would understand for the dnd button on my phone im guessing is just local |
| 22:05.21 | blebleble | polycom 501's 601's |
| 22:06.00 | [TK]D-Fender | blebleble: "no" until * integrates with SIP 3.X's server-based DND |
| 22:06.40 | lasko | The issue is that DND is local to the phone really. I have our system setup with a GUI interface for phone control and managing DND via a webinterface (AGI scripts that check DND status in a database whenever a phone is called) |
| 22:06.55 | blebleble | gotcha |
| 22:07.46 | [TK]D-Fender | Off to exam time @ martial arts. back later. |
| 22:07.53 | lasko | good luck |
| 22:08.43 | *** part/#asterisk popolinux (n=cyril@AToulouse-156-1-114-165.w90-30.abo.wanadoo.fr) |
| 22:08.50 | *** join/#asterisk saftsack (n=saftsack@p57924F4F.dip.t-dialin.net) |
| 22:09.58 | saftsack | hi, i get acl.c errors "can not connect" in asterisk when ipsec tries to connect but fails because of no wan connection broken. so internal telephones on the same net are broken them to. is this problem known? |
| 22:11.44 | Corydon76-dig | Hang on. How is IPSEC connecting to Asterisk? |
| 22:11.49 | *** join/#asterisk vvuja (i=Neo@acc-ppp-5055.yubc.net) |
| 22:11.52 | vvuja | oh hello |
| 22:11.58 | vvuja | my country is a a shithole |
| 22:12.04 | vvuja | and I'm here again |
| 22:12.06 | vvuja | watching |
| 22:12.13 | vvuja | all-seeing eye |
| 22:12.19 | watchy | you were missed. |
| 22:12.55 | blebleble | vvuja: what shithole country is that |
| 22:13.09 | vvuja | they said Serbia |
| 22:13.13 | vvuja | i don't think so |
| 22:14.10 | vvuja | so Qwell unban my IP |
| 22:14.12 | saftsack | Corydon76-dig: what do you mean connecting to asterisk? |
| 22:14.19 | vvuja | because they insulted me |
| 22:14.22 | vvuja | not me them |
| 22:14.31 | *** mode/#asterisk [+b vvuja@*!*@*] by Qwell |
| 22:14.33 | *** kick/#asterisk [vvuja!i=north@pdpc/sponsor/digium/Qwell] by Qwell (Qwell) |
| 22:14.37 | saftsack | there is one iax2 connection which goes over the vpn the rest is sip to the phones and to the isdn gateway |
| 22:14.42 | *** join/#asterisk Chainsaw (n=chainsaw@gentoo/developer/atheme.member.chainsaw) |
| 22:14.42 | nullable_type | I am getting a Reinvite Rejected from SBC. I have canreinvite=yes and directrtpsetup=yes. Is there anything else i need to do? |
| 22:14.52 | *** mode/#asterisk [-b vvuja@*!*@*] by Qwell |
| 22:14.55 | *** mode/#asterisk [+b vvuja!*@*] by Qwell |
| 22:15.03 | saftsack | if i do a ipsec stop asterisk still runs fine. but if the vpn is in building up state it is broken |
| 22:15.11 | *** mode/#asterisk [-bb *!*n=vvuja@79.175.115.* Qw!*@*] by Qwell |
| 22:15.15 | *** mode/#asterisk [-b *!*n=vvuja@79.175.115.*] by Qwell |
| 22:16.14 | watchy | null: whats SBC? |
| 22:16.15 | Corydon76-dig | saftsack: then clearly, you have ipsec misconfigured. How that is Asterisk's problem, I don't understand |
| 22:16.48 | nullable_type | SBC = session border controller |
| 22:17.06 | saftsack | the whole sip part breaks in asterisk if vpn hangs because of a temporary wan failure |
| 22:17.10 | *** join/#asterisk stabler (n=seedbox@cpe-24-210-38-135.columbus.res.rr.com) |
| 22:17.21 | nullable_type | I am just stuck with this whole day now |
| 22:17.43 | Corydon76-dig | saftsack: that's not an Asterisk problem; it's a routing problem |
| 22:18.00 | saftsack | ping works fine to all hosts |
| 22:18.11 | saftsack | if the vpn is broken |
| 22:18.12 | Corydon76-dig | saftsack: udpping? |
| 22:18.25 | Corydon76-dig | or just icmp? |
| 22:18.32 | saftsack | aaahh ;) that would be worth a try |
| 22:18.34 | saftsack | just icmp |
| 22:18.41 | *** join/#asterisk war9407 (i=war@liquidswords.org) |
| 22:18.59 | war9407 | I have attached an SPA 3102 to my PSTN line and it works great for inbound calls, to voice mail etc, but I cannot dial out. |
| 22:19.13 | war9407 | [Apr 16 18:17:52] NOTICE[24390] chan_sip.c: Peer 'pstn' is now Reachable. (9ms / 2000ms) |
| 22:19.13 | war9407 | [Apr 16 18:17:52] NOTICE[24390] chan_sip.c: Peer 'line1' is now Reachable. (9ms / 2000ms) |
| 22:19.16 | war9407 | it says both are working |
| 22:19.23 | war9407 | but every time I dial out I get a busy signal |
| 22:20.06 | *** join/#asterisk ecret (n=ecret@CPE001e9002348e-CM001225d8ab30.cpe.net.cable.rogers.com) |
| 22:20.20 | war9407 | any clues? |
| 22:21.34 | Corydon76-dig | war9407: SIP trace? |
| 22:21.48 | lasko | war9407: What is your setup? Do you have Analog hardware? |
| 22:21.49 | saftsack | Corydon76-dig: but why should there be another route for udp as for tcp? because of an error in netkey which routes all trafic which is udp to the tunnel if it is broken? |
| 22:22.34 | war9407 | lasko: regular telephone ----> spa 3102 -------> asterisk |
| 22:22.41 | war9407 | regular=analog yes |
| 22:24.44 | war9407 | [Apr 16 18:24:30] NOTICE[24390] chan_sip.c: Call from 'line1' to extension '5551234555' rejected because extension not found. |
| 22:24.50 | war9407 | its treating my calls as extensions? |
| 22:25.11 | lasko | What context are you utilizing to go outbound? |
| 22:25.16 | watchy | your dialplan is haxored |
| 22:25.33 | Corydon76-dig | saftsack: I'm not an IPSEC expert |
| 22:26.02 | Corydon76-dig | saftsack: but routing is completely beyond what Asterisk is doing |
| 22:26.29 | war9407 | lasko: followed this guide |
| 22:26.30 | war9407 | http://blog.pathennessy.org/2009/01/01/configuring-linksys-spa-3102-for-asterisk/ |
| 22:27.16 | saftsack | Corydon76-dig: thats true but i have to find out whats going on. and why it affects asterisk and not any other applications like ssh |
| 22:27.24 | lasko | war9407: could you pastebin your [pstn] context from extensions.conf? |
| 22:28.18 | war9407 | [pstn] context is inbound only right? |
| 22:28.23 | war9407 | according to that doc |
| 22:28.31 | war9407 | http://pastebin.com/m79cf13c5 |
| 22:28.37 | hardwire | blah |
| 22:28.37 | lasko | sorry, and your [line1] context |
| 22:29.08 | war9407 | http://pastebin.com/m2510574 also line1 now |
| 22:29.55 | war9407 | from sip.conf |
| 22:29.59 | war9407 | line1 is from sip.conf |
| 22:30.04 | war9407 | pstn is from extensions.conf |
| 22:30.19 | lasko | war9407: The issue is that you have no context for dialing exten => NXXNXXXXXX |
| 22:30.58 | lasko | war9407: Asterisk saw you try to dial 5551234555 but could not find a way to dial it because all you have is exten => 123 |
| 22:31.01 | *** join/#asterisk jedaustin (n=chatzill@austin-j.its.dist.maricopa.edu) |
| 22:31.21 | war9407 | ah. |
| 22:31.46 | lasko | war9407: I suggest reading http://www.voip-info.org/tiki-index.php?page=Asterisk%20config%20extensions.conf |
| 22:31.58 | lasko | It will be a world of use to you |
| 22:32.44 | nullable_type | lasko >> Do you know much about SIP Reinvite? |
| 22:32.53 | jedaustin | Hi all... anyone here in the 619 area and have ATT on straight copper? I have a guy I'm assisting remotely and the FXO device hangs up immediately. When he takes it to another location it works. |
| 22:33.17 | lasko | nullable_type: I might be able to assist. Ask and we'll see lol |
| 22:33.49 | jedaustin | I'm wondering if I need to use loopstart signaling with ATT in 619 |
| 22:34.38 | nullable_type | ya so i have just one provider through SBC and i have canreinvite=yes and directrtpsetup=yes but after call is established and Askteisk tries native briding it is getting SIP Reinvite Rejected |
| 22:35.12 | nullable_type | i am wondering if its just one provider may be Asterisk is rejecting Asterisk (Both caller and callee going through same box) |
| 22:35.30 | nullable_type | or if its the SBC(Session Border Controller) |
| 22:35.33 | war9407 | lasko: hi |
| 22:35.41 | war9407 | lasko: so I need a [line1] context in extensions.conf, correct? |
| 22:36.19 | nullable_type | lasko >> let me know if you know anything about the above problem |
| 22:36.47 | lasko | nullable_type: Try "sip debug on" and see where the rejection takes place. That would be my first place to check, but I'm not really sure |
| 22:36.48 | war9407 | [Apr 16 18:36:39] WARNING[24577]: pbx_config.c:1551 pbx_load_config: ==!!== Unknown directive: extern at line 867 -- IGNORING!!! |
| 22:36.58 | nullable_type | i have sip debug on |
| 22:37.00 | war9407 | woops, typo |
| 22:37.15 | nullable_type | and yes it takes place after Asterisk sends Reinvite |
| 22:39.37 | war9407 | lasko: |
| 22:39.41 | war9407 | [line1] |
| 22:39.41 | war9407 | exten = s,1,Playback(hello-world) |
| 22:39.41 | war9407 | exten = 1,1,Dial(Zap/1) |
| 22:39.48 | war9407 | [Apr 16 18:38:59] NOTICE[24390]: chan_sip.c:17295 handle_request_invite: Call from 'line1' to extension 'XXXXXX' rejected because extension not found. |
| 22:40.21 | lasko | war9407: no no no. Okay, in your sip.conf you have [line1] pointing to context internal |
| 22:40.29 | lasko | your dialplan must reflect there |
| 22:42.21 | war9407 | how do I specify the context in extensions.conf so the dialplan is in-sync with the sip.conf? |
| 22:43.14 | war9407 | ah |
| 22:43.17 | war9407 | I see |
| 22:43.20 | lasko | war9407: example: I have an extension called 9001. In sip.conf I have [9001] and the context=from-internal. In my extensions.conf I have a context called [from-internal] |
| 22:43.40 | lasko | which specifies what to fo when something is dialed |
| 22:44.26 | war9407 | ok, so context=internal for sip.conf |
| 22:44.36 | war9407 | and then in extensions.conf I will have a context [internal] |
| 22:44.39 | lasko | *nod* |
| 22:45.01 | lasko | pastebin what [internal] is and lets have a look |
| 22:45.15 | war9407 | ok |
| 22:45.33 | war9407 | http://pastebin.com/m26d81812 |
| 22:47.56 | war9407 | hmm |
| 22:48.03 | war9407 | can I use a linksys 3102a to dial out? |
| 22:48.06 | lasko | Do you have Zaptel interface? |
| 22:48.09 | war9407 | nope |
| 22:48.20 | war9407 | am I screwed? |
| 22:48.22 | lasko | Okay so why are you trying to use Zap/ |
| 22:48.30 | war9407 | http://www.voip-info.org/wiki/view/Asterisk+Extension+Matching |
| 22:48.33 | war9407 | following the example |
| 22:48.36 | lasko | Do you have a SIP provider? |
| 22:48.46 | war9407 | nope, I just want to dial out the PSTN (regular analog phone line) |
| 22:48.57 | lasko | *nod* |
| 22:49.16 | war9407 | is that possible in my configuration? |
| 22:49.55 | lasko | http://forum.voxilla.com/linksys-sipura-voip-support-forum/starter-spa3102-asterisk-setup-18612.html |
| 22:50.02 | lasko | I think the instrcutions there are mroe useful for you |
| 22:50.24 | lasko | Note the [internal] context |
| 22:50.29 | war9407 | rechecking |
| 22:50.40 | lasko | specifically the exten => _XXXXXXXXXX,1,Dial(SIP/${EXTEN}@pstn,60,) |
| 22:51.48 | war9407 | that worked |
| 22:51.52 | war9407 | impressive |
| 22:51.58 | lasko | war9407: <3 |
| 22:52.26 | war9407 | lasko: thx much, btw I have to ask, I have went through the latest 190 page+ 1.6.0.x docs and some other asterisk docs and none of them are a really good start guide |
| 22:52.34 | war9407 | lasko: the docs seem to be strewed all over the place ? |
| 22:53.20 | lasko | war9407: voip-info.org is your god and also download the pdf at http://www.asteriskdocs.org/ |
| 22:53.35 | lasko | its 1.4 though for the pdf |
| 22:53.41 | *** join/#asterisk ruben23 (n=ITadmin@124.107.3.178) |
| 22:53.43 | war9407 | k, btw, my phone provider allows me to run options via '*' + some extensions |
| 22:53.47 | lasko | but the configuration portions will still be pretty relevant |
| 22:53.56 | war9407 | would I use _*XX,etc? |
| 22:53.59 | ruben23 | hi best gateway/firewall for asterisk.... |
| 22:54.01 | war9407 | lasko: checking doc now |
| 22:54.14 | ruben23 | what can you recommend..with VOIP.. |
| 22:54.37 | ruben23 | internet===gateway/firewall====>asterisk server |
| 22:54.45 | lasko | ruben23: Any gateway and any firewall. Depends on your design. Are you using SIP or IAX2? |
| 22:54.53 | war9407 | lasko: thx, thiswill keep me busy |
| 22:55.26 | war9407 | exten => _*99,1,Dial(SIP/${EXTEN}@pstn,60,) |
| 22:55.28 | lasko | war9407: No problem. Good luck and never hesitate to ask questions (even if you feel them to be stupid) |
| 22:55.34 | RypPn | ruben23 http://www.voip-info.org/wiki/view/Fail2Ban+(with+iptables)+And+Asterisk |
| 22:55.36 | war9407 | how do I make it so it sends *99 to the phone provider btw? |
| 22:55.38 | ruben23 | SIP |
| 22:55.52 | ruben23 | im using SIP..and local asterisk server setup |
| 22:55.56 | war9407 | lasko: I wish to send *99 to check voice mail etc to the phone company |
| 22:56.24 | lasko | war9407: exten => *99,1,Dial(SIP/${EXTEN}@pstn,60,) |
| 22:56.31 | *** join/#asterisk lesouvage (n=lesouvag@82.73.69.76) |
| 22:56.34 | *** join/#asterisk nullable_type (n=nullable@hq.verbx.net) |
| 22:57.12 | war9407 | lasko: I dial *99 and the dial tone still hums |
| 22:57.21 | ruben23 | lasko: im on SIP |
| 22:57.45 | *** join/#asterisk dominicl (n=chatzill@ri-sw-metro.ptgrey.com) |
| 22:58.09 | lasko | Does you spa3102 have a dialplan within it? |
| 22:58.10 | nullable_type | Hey guys, If i have one voip provider and i do an originate command from CLI to phoneNumber1@provider1, then in my extension.conf i DIAL to phoneNumber2@provider2.... Should the reinvite work with canreinvite=yes at sip.conf?! |
| 22:58.24 | nullable_type | phonenumber2@provider1 |
| 22:58.46 | war9407 | lasko: (*xx|[3469]11|0|00|[2-9]xxxxxx|1xxx[2-9]xxxxxxS0|xxxxxxxxxxxx.) |
| 22:58.55 | war9407 | lasko: on the spa 3102 page |
| 22:59.02 | lasko | war9407: What does asterisk say when you dial *99 from the phone? |
| 22:59.12 | dominicl | hey folks, I'm trying to setup an iaxmodem for hylafax, but it doesn't seem to be registering with the asterisk server. when I do "iax2 show peers", it always shows that extension offline / status UNKNOWN. any advice on how can I troubleshoot and fix this? |
| 22:59.30 | war9407 | lasko: sec |
| 22:59.48 | war9407 | lasko: nothing in the logs and core & debug is set to 99 |
| 22:59.56 | lasko | ruben23: Any firewall/gateway will be fine. You just have to make sure that you aren't blocking sip ports |
| 23:00.31 | nullable_type | lasko, If i have one voip provider and i do an originate command from CLI to phoneNumber1@provider, then in my extension.conf i DIAL to phoneNumber2@provider.... Should the reinvite work with canreinvite=yes at sip.conf?! |
| 23:00.33 | lasko | ruruber23: Like RypPn stated, check out http://www.voip-info.org/wiki/view/Fail2Ban+(with+iptables)+And+Asterisk |
| 23:00.33 | ruben23 | ok ill be installing untangle test server.. |
| 23:01.03 | lasko | war9407: Theres your first indication that its the 3102 |
| 23:01.13 | war9407 | k |
| 23:02.08 | lasko | war9407: Play around with the dialplan on the 3102 and reboot it and see what happens. I've never used one so I'm not sure what its design is. |
| 23:02.17 | war9407 | k |
| 23:02.55 | lasko | will be back shortly. |
| 23:09.34 | war9407 | FAX Line Toggle Code: is #99 |
| 23:09.36 | war9407 | lol |
| 23:09.40 | war9407 | same code as voicemail |
| 23:09.42 | war9407 | I will change that |
| 23:10.38 | dominicl | If I turn on iax2debug, it doesn't really seem to say much. a lot of these Retry frams and "iax.c line 2617 in iax_header_to_event: Cancelling transmission of packet 6" |
| 23:11.37 | war9407 | not fax, it was the modem one |
| 23:12.13 | war9407 | lasko: now it says call from pstn to extension '**' rejected because extension not found |
| 23:12.15 | war9407 | *99 rather |
| 23:12.23 | war9407 | lasko: I want that to send to the dline |
| 23:13.44 | lasko | war9407: is *99 in the context? |
| 23:13.48 | war9407 | yup |
| 23:13.53 | war9407 | [internal] |
| 23:13.53 | war9407 | exten => *99,1,Dial(SIP/${EXTEN}@pstn,60,) |
| 23:14.15 | lasko | *nod* hmm, hold on. Still on the phone |
| 23:14.48 | lasko | while you wait. Could you pastebin what asterisk says |
| 23:14.49 | dominicl | the first couple of lines in the debug output are a REGREQ which shows the correct value for USERNAME and then "Making new session" and then "iax.c line 2936 in iax_header_to_event: Don't know what to do with IAX command 13" |
| 23:14.59 | war9407 | lasko: its one line: [Apr 16 19:12:34] NOTICE[24390]: chan_sip.c:17295 handle_request_invite: Call from 'pstn' to extension '*99' rejected because extension not found. |
| 23:24.13 | *** join/#asterisk Mudboy (n=steve@c-24-16-94-198.hsd1.wa.comcast.net) |
| 23:24.17 | nullable_type | lasko >> Had you have a chance to use reinvite before, can u tell me how you used? |
| 23:25.21 | war9407 | lasko: at least I can dial out now, will look at the '*' more later, thx. |
| 23:26.52 | *** join/#asterisk egypcio (n=egypcio@unaffiliated/egypcio) |
| 23:33.52 | war9407 | hm btw local calls work XXX-XXX-XXXX |
| 23:33.57 | war9407 | but not long distance or 1800 #s |
| 23:34.00 | war9407 | with those on that page |
| 23:34.00 | war9407 | hmm |
| 23:41.53 | *** join/#asterisk jicksta (n=jicksta@c-67-169-165-162.hsd1.ca.comcast.net) |
| 23:41.55 | epaphus | hey guys, whats the best Asterisk administration GUI for a call center with 40 phones?... |
| 23:44.01 | jblack | console. |
| 23:45.58 | *** join/#asterisk edwin_quijada (n=macaruch@200.26.172.50) |
| 23:46.01 | edwin_quijada | Hi! |
| 23:46.19 | edwin_quijada | Iam trying to instal cdr_odbc in my debian with asterisk but I cant |
| 23:46.27 | nullable_type | Hey guys, Anyone have used canreinvite=yes succesfully? |
| 23:47.18 | edwin_quijada | I installed unixodbc and freetds but when I do make menuselect I dont get the cdr_odbc available |
| 23:47.28 | edwin_quijada | to compile |
| 23:47.46 | edwin_quijada | someboyd has instlled cdr for mssql |
| 23:52.37 | *** join/#asterisk f0ner00t (n=f0ner00t@c-67-187-154-111.hsd1.ca.comcast.net) |
| 23:52.53 | f0ner00t | What necessary files do I need to make an asterisk work properly? |
| 23:53.00 | f0ner00t | files I mean conf files |
| 23:53.01 | f0ner00t | SOrry. |
| 23:53.14 | drmessano | All of them |
| 23:56.36 | f0ner00t | drmessano: Bah humbug.. so the best way to install asterisk is via the sample fileS? |
| 23:57.32 | kc8pxy | f0ner00t: asterisk is designed to "work" in many different capacities, and for different tasks. what for of "work" do you mean? |
| 23:59.26 | kc8pxy | f0ner00t: i have a VERY simple testing setup running with just a modules.conf, sip.conf and extensions.conf you can do that with no special hardware. everything else is simply a level of complexity. |
| 23:59.39 | f0ner00t | kc8pxy: Meaning if I have the basic sip.conf extensions.conf will it run. |