00:00.27 | bulle | CrashHD: sweet, then perhaps anyone in here that gets on just fine can explain the pretty basic info, on how to setup asterisk to place certain sip calls over a certain outbound proxy ? that should be a fairly common setup |
00:00.51 | mekong | what is the best way to strip ${BLINDTRANSFER} so that I am let with technology/exten as opposed to technology/exten-3145 ? |
00:01.04 | bulle | JT: the book is great, i have been reading it, but sadly the sip chapter doesnt talk about how to use different oubound sip proxies |
00:01.08 | CrashHD | your proxy has to be setup for it |
00:01.20 | CrashHD | you would setup the proxy info in your sip.conf |
00:01.32 | CrashHD | then SIP/${EXTEN}@proxy |
00:01.39 | mekong | thats left with not let with |
00:02.13 | bulle | CrashHD: SIP/extension@ekiga.net@sipproxy.com ? |
00:02.37 | bulle | CrashHD: give that i have a sipproxy.com in my sip.conf |
00:02.52 | CrashHD | I don't believe asterisk is setup that way |
00:03.29 | CrashHD | unless your other proxy parsed the sip headers |
00:03.38 | CrashHD | to understand that |
00:03.43 | CrashHD | sip debug it |
00:03.47 | CrashHD | see what is actually being sent |
00:04.07 | bulle | CrashHD: well, the sip.conf docs has a keyword "outboundproxy" so i think its supported |
00:05.40 | bulle | just cant find any documentation about how to use it, and how it relates to what input i feed to Dial |
00:06.32 | *** join/#asterisk tzafrir (n=tzafrir@62.90.10.53) |
00:07.11 | CrashHD | http://www.sineapps.com/news.php?rssid=1677 |
00:07.34 | CrashHD | that took me all of 15 seconds to type asterisk sip.conf outboundproxy in google |
00:08.28 | bulle | CrashHD: oh yes, as i said, i alreay know the keyword is supported, but how do i specify the given outbound proxy for my Dial ? |
00:08.31 | bulle | CrashHD: that is the question |
00:09.16 | CrashHD | from what I gathered just glancing, asterisk will use it on the backend when outbound sip calls are made |
00:09.34 | CrashHD | so you would specify SIP/extension@ekiga.net |
00:09.36 | bulle | CrashHD: yes, but i only want to use the outbound proxy on certain calls, not all calls |
00:09.51 | bulle | as i said earlier |
00:09.55 | CrashHD | so you setup the outbound proxy for [system1] in your sip.conf |
00:10.07 | CrashHD | hmm |
00:10.14 | CrashHD | not sure it will be that dynamic |
00:10.23 | CrashHD | you can use it on a per sip.conf entry |
00:10.32 | bulle | SIP/extension@ekiga.net@system1 ? |
00:10.49 | bulle | SIP/system1/extension@ekiga.net |
00:10.58 | CrashHD | I don't think that is how it was written |
00:11.03 | CrashHD | write oje on the mailing list |
00:11.05 | CrashHD | and ask him |
00:11.24 | bulle | ye, i think i will actualy mail olle and ask |
00:11.37 | bulle | olle is the man, most likely |
00:14.33 | bulle | i suspect the last syntax is the correct one |
00:22.44 | ManxPower | generally it is SIP/destinationumber@sipconfentry |
00:22.59 | ManxPower | if you need to specify a username you would use fromuser= in the sipconfentry |
00:23.13 | ManxPower | but since the outbound proxy stuff is so new.... |
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00:28.29 | jeffik | All: need help setting up remote access for SPA-942 thorugh linksys router |
00:29.02 | CrashHD | ~jbot crickets |
00:30.59 | *** join/#asterisk voiper1 (n=luke@ozvoip.dsl.onthenet.net) |
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00:46.35 | lokkju_wrk | any of you use vonage with a motorola VT1005v? |
00:46.54 | CrashHD | uh oh...better not say "vonage"...verizon might come get you |
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00:46.59 | bkruse_home | ! |
00:47.01 | bkruse_home | lol |
00:47.10 | bkruse_home | whats a good voip provider that does flat rate and not pay per minute |
00:47.12 | CrashHD | with their big enum patents |
00:47.58 | ManxPower | bkruse_home: they all suck |
00:47.59 | lokkju_wrk | heh, I need a valid MAC number so I can pull down a firmware image to play with |
00:48.18 | bkruse_home | ManxPower: really? should i just pay per minute? |
00:48.20 | bkruse_home | for home use, that is |
00:48.23 | lokkju_wrk | so I can whip up some instructions on unlocking it - permenently |
00:48.37 | ManxPower | bkruse_home: ALL ITSPs suck, but per min is actually usually the better deal |
00:48.45 | bkruse_home | really? |
00:48.47 | CrashHD | bkruse_home: I find most the time you over pay with flat rate |
00:48.49 | bkruse_home | i dont want to have to worry about it, though |
00:49.00 | aptura | ManxPower is it the tisp or the carrier thay use? |
00:49.03 | lokkju_wrk | bkruse_home, forget flat rate - it is actually usually cheaper to go with ppm - try genericvoip, voipjet, and voipstream connect (sp?) |
00:49.08 | bkruse_home | i might juts use my pots line to see how many minutes i use now |
00:49.20 | bkruse_home | thanks guys :] i might give it a shot |
00:49.30 | ManxPower | I'm a fan of Teliax, but I've not used them since I had to go to satellite internet |
00:49.32 | bkruse_home | sometimes the minutes dont seem like much, but im not sure how many realistic mintues i use per month anyways |
00:49.33 | lokkju_wrk | voipjet == dynamically set CID :) |
00:49.34 | CrashHD | bkruse_home, I like vitelity personally |
00:49.35 | bkruse_home | yay iax! |
00:49.48 | bkruse_home | lokkju_wrk: i likey that :] |
00:49.58 | bkruse_home | ima have to rip my phone number away from good ole bell ;[ |
00:50.02 | aptura | crashev I use them. ocationally there is a issue with vitelity |
00:50.12 | CrashHD | aptura: what issues have you had? |
00:50.14 | bkruse_home | i dont think itll be that hard, though, since that last law was passed having to do with #'s i thought..... |
00:50.28 | CrashHD | we have 60k minutes a month running over them with very few issues |
00:50.43 | aptura | mmm disconect or the occational delay in the media stream. but its uncommon now. |
00:50.54 | *** part/#asterisk wwalker (n=wwalker@pdpc/supporter/sustaining/wwalker) |
00:51.01 | bkruse_home | im thinking more like........400 |
00:51.12 | aptura | CrashHD is that for your own company or you a reseller |
00:51.24 | bkruse_home | im getting 1 meg up and down from this random provider in monrovia |
00:51.27 | asteriskguy | Bladerunner, I would go into the console itself and check to see if you have zap channels loaded |
00:51.38 | *** join/#asterisk jovannotti (n=jovannot@190.84.87.64) |
00:51.43 | jovannotti | hi to all, |
00:51.45 | CrashHD | we have lots of clients |
00:51.51 | CrashHD | but directly used |
00:51.55 | CrashHD | brb |
00:52.01 | CrashHD | vitelity works for 400 as well |
00:52.05 | jovannotti | I am looking for a tool to generate h323 calls , someone knows sb about ? |
00:52.08 | aptura | 400 what? |
00:52.38 | bulle | jovannotti: ekiga supports h323 |
00:53.06 | jovannotti | I can generate simultaneous calls from ekiga ? |
00:53.30 | bulle | jovannotti: simultaneous calls ? |
00:53.48 | jovannotti | the purpose is to generate at least 50 simultaneous calls in h323, to test my asterisk server |
00:54.13 | bulle | jovannotti: he, guess you should have said that then |
00:54.21 | bulle | jovannotti: then ekiga wont fit your needs afaik |
00:54.21 | bkruse_home | 400 minutes |
00:54.25 | bkruse_home | how many calls per account? |
00:54.30 | bkruse_home | ima rent some out to my aunt with her DID |
00:55.10 | jovannotti | sorry bullet, do you know some another tool ? If I could generate them via linux comands, I can create a perl scrip to send 50 calls simulataneously |
00:55.19 | bkruse_home | anyone ever transfered their DID from the nasty ole bell south? |
00:58.19 | fetcher | you mean the Nasty New AT&T :) |
00:58.27 | *** join/#asterisk Strom_M (n=pocketir@m040e36d0.tmodns.net) |
00:58.34 | bkruse_home | fetcher: yes |
00:58.37 | bkruse_home | anyone/ |
00:58.49 | jovannotti | something could help me ? with the tool to generate at least 50 simultaneous call in h323 |
00:59.03 | fetcher | I've transferred business numbers (ISP dialup pools) to CLECs from them, without any problems. Nothing involving VoIP so far... |
00:59.07 | bkruse_home | call files? |
00:59.07 | bkruse_home | lol |
00:59.19 | bkruse_home | fetcher: interesting, i dont think itll be a problem |
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01:02.56 | Strom_M | hello |
01:03.23 | JT | hi Strom_M |
01:03.56 | Strom_M | how goes it |
01:04.05 | JT | not too bad |
01:04.07 | JT | yourself? |
01:04.18 | Strom_M | doing well |
01:04.40 | JT | good to hear |
01:05.32 | Strom_M | just sitting down for dinner :) |
01:06.29 | *** join/#asterisk drumkilla (n=russell@asterisk/developer-and-stable-maintainer/drumkilla) |
01:06.29 | *** mode/#asterisk [+o drumkilla] by ChanServ |
01:08.26 | JT | Strom_M: irc dinners ;) |
01:08.40 | jovannotti | something could help me ? with the tool to generate at least 50 simultaneous call in h323 |
01:08.49 | Qwell | jovannotti: use asterisk to generate the calls |
01:09.53 | jovannotti | I have 1.4 with most of its tools, can you guide me with which command I could generate this calls ? |
01:10.15 | bkruse_home | jovannotti: cd /usr/src/asterisk ; vi sample.call |
01:10.44 | jovannotti | thanks a lot bkruse |
01:10.57 | jovannotti | I'll try right now |
01:11.33 | bkruse_home | kk |
01:13.39 | bkruse_home | then after you have a good one, woot=0 ; while [ "$woot" -le "50" ] ; do cp newfile.call /var/spool/asterisk/outgoing/call$woot.call ; done |
01:13.42 | bkruse_home | i think.... |
01:14.40 | jovannotti | yes, thanks ! I am lookin the sintaxis to create the files |
01:17.22 | jovannotti | I think I'll can test my (*) server. It works fine, but if I receive more than 50 calls ->> seg fault :( |
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01:35.08 | flenders | JT: on a PRI, can I create a group of channels? for example sip/01, sip/02 and sip/03 can only access 3 lines? if those lines are busy, they won't try to use other 'lines', even though the other (let's say) 20 are available |
01:35.19 | *** join/#asterisk kupsi (n=kupsi@210.213.101.34) |
01:36.14 | JT | yes |
01:36.16 | JT | zap groups |
01:36.52 | flenders | JT: so it works the same way as channels on a TDM400P? |
01:37.17 | JT | i guess |
01:37.33 | JT | except works better, you'll notice it over time |
01:37.49 | flenders | and on extensions.conf I would just use g1, g2, etc? |
01:38.00 | JT | yes |
01:38.06 | flenders | too easy |
01:38.08 | JT | either g or G or r or R |
01:38.19 | flenders | g == r? |
01:38.24 | JT | no |
01:38.30 | flenders | what does r stand for? |
01:38.36 | JT | round robin != normal group ringing |
01:38.43 | mmartinn | arrrrrr |
01:38.48 | JT | there's a wiki article that explains it real well |
01:38.56 | mmartinn | ~extensions.conf |
01:39.12 | jbot | hmm... extensions.conf is at http://voip-info.org/wiki-Asterisk+config+extensions.conf, or know as dialplan, or known as extensions, or known as exten |
01:39.12 | kupsi | hello guys, i have successfuly built an asterisk box. I can configure 2 softphones to talk to each other. Now, my question is: What are the things that I need in order for me to make outgoing calls to our PSTN, also what devices are needed in order for my asterisk box to receive multiple calls from the PSTN? Sorry I'm a n00b. |
01:39.14 | Strom_C | flenders: it would be a better idea to restrict a group of phones to using only a certain quantity of channels rather than dedicating those actual channels to those sets |
01:39.46 | JT | well i think he was suggesting the use of groups |
01:39.55 | flenders | Strom_C: that was the plan, actually, just didn't know it was possible |
01:40.11 | Strom_C | flenders: yes, it's totally doable with dialplan logic |
01:40.37 | flenders | JT: I thought about groups as that was I thought was possible |
01:42.40 | flenders | Strom_C: any idea how I should search for that? |
01:42.59 | Strom_C | *shrug* |
01:43.10 | Strom_C | it's something you'd have to hand-cruft |
01:44.02 | flenders | ah ok, so, I could, based on context, check how many channels are already in use? |
01:44.03 | kupsi | T_T |
01:44.10 | *** part/#asterisk mog (n=mog@c-71-207-200-130.hsd1.al.comcast.net) |
01:44.42 | JT | flenders: why would you need to do that? |
01:44.53 | JT | flenders: why are zap groups insufficient? |
01:45.20 | flenders | JT: dunno |
01:45.45 | flenders | JT: they're probably the same |
01:45.49 | Strom_C | JT: he shouldnt use zap groups in this situation because the telco won't segment inbound calls on their end |
01:46.09 | Strom_C | and also, the telco will always dictate which channel you are to use on PRI, even if you request a different one |
01:46.15 | Strom_C | so zap groups in this case reek of kludgery |
01:47.01 | flenders | so, inbound calls could come in on a channel that is "assigned" to another group? |
01:47.07 | *** join/#asterisk mog (n=mog@c-71-207-200-130.hsd1.al.comcast.net) |
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01:47.07 | Strom_C | yes |
01:47.11 | JT | flenders: oh definately |
01:47.17 | flenders | makes a lot of sense then |
01:47.34 | JT | i didn't know telcos dictated channels for outbound calls |
01:48.00 | Strom_C | JT: when you place a PRI call, you request a channel, but the telco can force you onto a different channel |
01:48.04 | *** join/#asterisk Noodleman (n=tuckerm@ip68-0-112-170.tu.ok.cox.net) |
01:48.19 | JT | hmm ok |
01:50.35 | lokkju_wrk | I'm looking for someone that has an active vonage motorola vt1005v ATA - I just need you to either run a tftp command, or to give me the MAC address of your unit - I am attempting to provide some unlock methods for it, and to do so I need to get a new formware image. someone want to help me? |
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02:14.58 | kupsi | hello guys, i have successfuly built an asterisk box. I can configure 2 softphones to talk to each other. Now, my question is: What are the things that I need in order for me to make outgoing calls to our PSTN, also what devices are needed in order for my asterisk box to receive multiple calls from the PSTN? Sorry I'm a n00b. |
02:20.26 | gambolputty | fxo card |
02:21.03 | gambolputty | http://www.digium.com/en/products/hardware/tdm800p.php |
02:23.23 | Strom_C | or an itsp |
02:24.06 | bkruse_home | successfully built an asterisk box??? sh configure ; make && make install |
02:24.07 | bkruse_home | lol |
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02:26.17 | CunningPike | kupsi: Your choices are an FXO port (something you can plug a regular 1B line into), an ITSP (a company that has an existing connection to the PSTN and to which you connect via the Internet), or a PRI (a multi-channel digital connection to the PSTN) |
02:27.01 | *** join/#asterisk Cybertoy (n=cybertoy@dsl254-123-112.nyc1.dsl.speakeasy.net) |
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02:43.49 | *** join/#asterisk voltagex (n=voltagex@124-254-123-181-dsl.ispone.net.au) |
02:45.46 | voltagex | All incoming IAX2 calls are being dropped. |
02:45.54 | voltagex | with a message something like NO AUTH |
02:46.07 | Strom_C | voltagex: dude, I said I'd help you |
02:46.13 | Strom_C | will you hang on a few minutes? |
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02:47.43 | Strom_C | ok, this is weird....make install on zaptel 1.4 svn branch is dying with "build_tools/genudevrules: line 1: udevinfo: command not found" on a fairly ordinary debian sarge install |
02:48.43 | *** join/#asterisk PakiPenguin (n=uppal@linuxpakistan/admin/pakipenguin) |
02:48.51 | Corydon76-home | udevtools not installed? |
02:49.21 | Strom_C | no such package |
02:49.42 | voltagex | Strom_C: I was just coming in here because this is the official asterisk channel |
02:49.47 | Strom_C | how do I tell if I'm running udev or not? |
02:49.56 | buggsy | Could someone point me to some documentation that would help me setup SLA? The doc/sla.txt is too vauge for my brain at this time of night |
02:50.03 | voltagex | Strom_C: udev is for >=kernel 2.6 |
02:50.12 | Strom_C | yes, I'm running 2.6 |
02:50.25 | voltagex | well then it's likely that you have udev |
02:50.50 | Strom_C | well, "likely" my ass - how do I tell definitively? :) |
02:51.04 | voltagex | ps aux | grep udevd |
02:51.12 | Strom_C | nope, it isnt running |
02:51.25 | HockeyInJune | Anybody here mix their own music? My buddy needs someone for a video project. If your good with stuff like that, please gimme a PM, for some details. |
02:51.25 | voltagex | well afaik it doesn't exist then |
02:51.50 | *** join/#asterisk isamar (i=1000@202.95.221.156) |
02:51.54 | isamar | hi folks |
02:52.46 | isamar | n1 here ? |
02:52.52 | Strom_C | no, we're all dead |
02:52.54 | Strom_C | sorry |
02:53.03 | Strom_C | funeral begins tomorrow at 11 |
02:53.04 | isamar | :-) I see... |
02:53.06 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
02:53.16 | voltagex | Strom_C: also, my machine with udev has /sbin/udevd |
02:53.22 | isamar | I felt the smell here... |
02:53.29 | isamar | :-) |
02:53.36 | voltagex | isamar: sorry, I farted. |
02:53.40 | isamar | heheh |
02:53.56 | Strom_C | ok, now I have it running |
02:53.57 | isamar | just wondering with anybody is playing with Yate+Sangoma to gimme a hand... |
02:54.01 | Strom_C | let's see if it works yet |
02:54.03 | isamar | with=if |
02:54.19 | isamar | here |
02:55.39 | *** part/#asterisk isamar (i=1000@202.95.221.156) |
02:57.20 | voltagex | ok, normality resumes |
02:58.26 | Strom_C | hey, what do you know, it works |
02:58.29 | Strom_C | thanks voltagex :) |
02:59.52 | voltagex | I don't feel like I helped much, but hey, can't turn down thanks from a man named after a phone :P |
02:59.57 | voltagex | np |
03:00.00 | Strom_C | hehe |
03:00.59 | voltagex | now, to teh debug logs |
03:01.56 | voltagex | Strom_C: are you able to call me on my extension/ip directly? |
03:02.25 | Strom_C | i can if you have a guest iax account configured correctly... |
03:02.51 | voltagex | err, I am an asterisk noob, please explain |
03:03.07 | Strom_C | well, by default, you generally do |
03:03.25 | Strom_C | have a look in iax.conf at the [guest] section |
03:03.30 | Strom_C | which context does that have set? |
03:03.57 | voltagex | sorry, not asterisk, trixbox |
03:04.01 | voltagex | shite. |
03:04.04 | Strom_C | ... |
03:04.07 | Strom_C | no |
03:04.15 | Strom_C | get rid of that shit and install asterisk |
03:04.28 | voltagex | oh come on, everything else works |
03:04.38 | Strom_C | it's impossible to debuf |
03:04.40 | Strom_C | er, debug |
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03:05.21 | Strom_C | ~trixbox |
03:05.32 | jbot | i heard trixbox is junk - avoid. It is also unable to be supported here. Try joining #freepbx and asking there, or use the trixbox forums at http://www.trixbox.org/modules/newbb/ |
03:06.22 | voltagex | ok, try calling me at 0@124.254.123.181 |
03:06.30 | Strom_C | SIP, or IAX? |
03:06.34 | Qwell | h323 |
03:06.35 | voltagex | IAX |
03:06.51 | Strom_C | ok, i'll do so once asterisk finishes compiling |
03:06.57 | Qwell | So, I have a logistics question for you guys |
03:07.05 | Qwell | let's pretend I'm talking about a server room |
03:07.05 | Strom_C | qwell: the answer is cheese |
03:07.13 | Qwell | Assume the A/C is broken |
03:07.14 | JT | i use chan_modem, Qwell |
03:07.17 | JT | is this ok?? |
03:07.18 | Qwell | How do you get hot stale air out? |
03:07.25 | Strom_C | qwell: fans |
03:07.31 | Strom_C | i ran into this problem at ticketmaster |
03:07.32 | Qwell | inward or outward? |
03:07.38 | Strom_C | we had an AC unit that blew up |
03:07.57 | JT | fans, or portable AC |
03:07.57 | Qwell | for some reason, my office is *hot* |
03:08.07 | JT | hire a portable AC |
03:08.11 | Strom_C | we ended up blowing cool air into the room from outside and had every fan on the floor moved into the server room to circulate it |
03:08.12 | Qwell | currently I've got a fan blowing out the window, but it isn't helping a whole lot |
03:08.20 | Strom_C | blow cool air into the room |
03:08.22 | Qwell | it |
03:08.23 | Qwell | erm |
03:08.26 | Qwell | it's helping some, I guess |
03:08.32 | Qwell | there is no cool air :P |
03:08.38 | _Vile | figure out an exit for the ac, figure out how to create a negative pressure..... fans, yes... |
03:08.40 | mmartinn | Do you have a hallway? |
03:08.56 | Strom_C | blow air onto yourself then |
03:08.56 | mmartinn | Open the door to the hallway too |
03:08.56 | _Vile | open windows |
03:08.56 | Qwell | but no, I tried having air coming in, but it wasn't helping |
03:08.56 | voltagex | Quell: run away? |
03:08.57 | Qwell | Strom_C: that worked, but everywhere but my chair |
03:09.08 | Qwell | erm |
03:09.16 | Qwell | but everywhere but my chair was still really hot |
03:09.33 | voltagex | Qwell: get a new chair? |
03:09.38 | Qwell | chair was fine :D |
03:09.58 | Strom_C | qwell: is this at home, or at digigraph |
03:10.01 | mmartinn | If this is mission critical, you rent a portable AC unit until the regular stuff is fixed |
03:10.02 | Qwell | home :p |
03:10.19 | Qwell | A/C works fine for the rest of the house, but sucks in my office |
03:10.26 | mmartinn | So this is long termZ? |
03:10.28 | mmartinn | err term? |
03:10.32 | Qwell | mmartinn: pretty much |
03:10.40 | mmartinn | Get a second A/C unit or put more ducts in |
03:10.52 | mmartinn | I did the fan thing for a bit; Im' in FL and my office had crappy ventilation in the summer time |
03:11.05 | Qwell | the main duct is right below me, with an outlet right here |
03:11.34 | mmartinn | Get the main duct fixed then, or get a seperate fan for outside to pump hot air out, maybe? |
03:13.03 | Strom_C | qwell: I went to homo depot and got one of those giant metal fans |
03:13.27 | mmartinn | Qwell: You could punch a whole in the door and mount a box fan |
03:13.27 | Qwell | I think I'm just gonna get a window fan |
03:13.34 | Qwell | mmartinn: door is open |
03:13.37 | mmartinn | Window AC unit could help |
03:13.49 | voltagex | Strom_C: I'm immature enough to find homo dept funny. |
03:14.00 | Strom_C | :) |
03:14.11 | Qwell | offtopic, but somebody could easily hack a home depot sign to say that |
03:14.18 | Strom_C | oh, totally |
03:14.27 | Strom_C | or, even more amusingly, "The Homo Despot" |
03:14.46 | Qwell | where you gonna find an s? |
03:14.53 | voltagex | wait, physical signs as in wood and metal and plastic or something like an LED sign |
03:15.12 | Strom_C | voltagex: there's a home improvement warehouse chain in the states called "The Home Depot" |
03:15.15 | Strom_C | www.homedepot.com |
03:15.18 | JT | or neon, if you're a particularly skilled home craftsperson |
03:15.23 | voltagex | I know |
03:15.49 | voltagex | unlike you USians, I know what the rest of the world is (j/k) |
03:15.56 | *** join/#asterisk DaveCanoe (n=Dave@m815f36d0.tmodns.net) |
03:17.23 | Strom_C | ok, finally, it's done building |
03:18.31 | mmartinn | sleep! |
03:19.44 | *** join/#asterisk nuonguy (n=john@c-24-6-175-26.hsd1.ca.comcast.net) |
03:20.43 | Strom_C | gah, wtf did you do to asterisk, qwell |
03:20.50 | Qwell | broke it |
03:21.13 | voltagex | oh nice, now my provider tell me that IAX calling is broken. |
03:21.28 | Strom_C | voltagex: that's what "free" buys you |
03:21.49 | voltagex | Strom_C: no, see this is the trial account to make sure things work before I pay. |
03:22.03 | Strom_C | i thought this was FWD you were trying to debug |
03:22.18 | voltagex | Strom_C: gave up on FWD |
03:22.54 | Strom_C | none of my damned channel drivers will load |
03:23.11 | Qwell | make distclean |
03:23.14 | Strom_C | and the verbose console output is completely useless |
03:23.15 | Strom_C | ah ok |
03:23.30 | voltagex | well I am going now for a bit |
03:23.50 | Strom_C | voltagex: still want me to call you when i get my system up? |
03:23.55 | *** join/#asterisk Kumbang (n=kumbang@router-wmi.paume.itb.ac.id) |
03:24.11 | voltagex | Strom_C: not just yet, sorry, gotta run an errand |
03:24.15 | Strom_C | ok |
03:24.47 | voltagex | faktortel.com.au is my provider |
03:25.04 | *** part/#asterisk jeffik (n=Jeff@c-24-7-242-120.hsd1.in.comcast.net) |
03:25.04 | voltagex | read that as f..kertel.com.au when I first read it |
03:25.23 | JT | you read right ;) |
03:25.32 | JT | they get angry if you "make too many calls" |
03:26.12 | voltagex | JT: I'm limiting myself to VOIP providers that are on the Aussie PSTN gateway |
03:26.43 | voltagex | because the DIDs I can get are all STD rates. |
03:27.22 | JT | voltagex: "on the aussie pstn gateway"? |
03:27.23 | voltagex | I'll be back later, JT if you have any suggestions I'd love to hear them |
03:27.24 | Qwell | std rates? |
03:27.31 | voltagex | as in timed calls to that number |
03:27.32 | JT | subscriber trunk dialling |
03:27.38 | voltagex | local calls are 25c |
03:27.44 | JT | std is a term used in .au to describe national calls |
03:27.52 | JT | national/non local landline |
03:28.00 | voltagex | JT: 1300 558 592 |
03:28.15 | JT | voltagex: what's that number for? |
03:28.24 | voltagex | JT calling voip phones |
03:28.34 | JT | oh right |
03:28.35 | voltagex | although I think it's under heavy load today |
03:28.40 | JT | why don't you just get a DID? |
03:28.42 | voltagex | bbl |
03:28.51 | voltagex | as I said, STD rates |
03:29.46 | JT | not if you get a voip did |
03:29.54 | JT | you can get a did in whatever state you wanr |
03:29.55 | JT | want |
03:30.02 | voltagex | no the point is calling my computer from a landline. |
03:30.58 | *** join/#asterisk aptura (n=jondoe@S010600a0c93f6f7e.vs.shawcable.net) |
03:31.10 | Strom_C | oh, hehehehehe, i made an uber nub mistake |
03:31.22 | Strom_C | i turned off "loadable module support" in menuselect |
03:31.28 | voltagex | Strom_C: but you *are* an uber nub |
03:31.31 | Qwell | nice :p |
03:31.32 | Strom_C | yes yes |
03:31.40 | voltagex | oh, ok I shouldn't say anything, I've done that before |
03:31.46 | JT | voltagex: so you call the DID number? |
03:31.47 | voltagex | + turning off filesystem support |
03:31.55 | Strom_C | heheh |
03:32.10 | voltagex | JT: it needs to be a local call... |
03:32.36 | voltagex | hence the voip gateway, 1300 number is 25c...at least I hope it is. |
03:32.45 | JT | voltagex: ok, maybe i'm missing something, why won't getting a DID in an area where it will be a local call work? |
03:33.01 | voltagex | JT: I live in a hole? |
03:33.06 | voltagex | JT: Narooma, NSW |
03:33.12 | JT | voltagex: hrm |
03:33.21 | JT | voltagex: have you checked if engin can do it? |
03:33.23 | Strom_C | voltagex: also, for testing things, calling direct from your IP phone will be cheaper than using your DID |
03:33.31 | JT | they have some of the most PSTN gateways in .au |
03:33.33 | voltagex | what IP phone? |
03:33.34 | Strom_C | and by cheaper I mean free |
03:33.55 | Strom_C | a softphone maybe? |
03:34.20 | voltagex | JT: can I use asterisk with Engin though? I thought I had to use their proprietry stuff |
03:34.39 | JT | voltagex: yes, as long as you are on a Voiper plan |
03:34.51 | voltagex | JT: I'll look at the rates later |
03:35.07 | JT | alright |
03:35.10 | JT | if not |
03:35.16 | JT | there may be other options |
03:35.22 | voltagex | I'm not rich :/ |
03:35.30 | JT | voltagex: let me know your scenario when you get back |
03:35.32 | JT | neither am i |
03:35.43 | JT | i like saving money with toll bypass where possible :) |
03:48.59 | *** join/#asterisk stkn_ (n=stkn@gentoo/developer/pdpc.active.stkn) |
03:49.47 | *** join/#asterisk dc3aes (n=matt@S01060001023fe8ca.no.shawcable.net) |
03:50.22 | *** join/#asterisk bigred (n=ian@75-1-208-180.lightspeed.snantx.sbcglobal.net) |
03:50.56 | bigred | is there a way to tell how many active outbound connections i currently have? |
03:51.05 | Strom_C | "show channels" |
03:51.08 | Strom_C | or, if on 1.4 |
03:51.11 | Strom_C | "core show channels" |
03:51.49 | bigred | Strom_C: ok, what if they are all going out a single sip provider channel? |
03:52.00 | Strom_C | each call is a separate channel |
03:52.04 | Strom_C | even if it's the same sip provider |
03:52.14 | bigred | awesome. |
03:52.17 | *** join/#asterisk bmg505 (n=leon@c1-50-2.rndf.isadsl.co.za) |
03:52.55 | bigred | is this call available in the manager api. i am not seeing it |
03:53.09 | JT | bigred: the "channels" philosphy doesn't quite mesh with sip |
03:53.14 | JT | it's more connection oriented |
03:53.31 | *** join/#asterisk kgx (n=kgx@60.234.20.178) |
03:53.31 | aydiosmio | concurrency is only limited by the software |
03:53.46 | aydiosmio | by default the switch will accpet any call regardless of its source |
03:54.09 | bigred | so i am going to be doing a lot of outbound calling, and i want to make sure that my server doesnt get bogged down, so i want to rate limit the outgoing calls |
03:55.29 | aydiosmio | you can limit eh number of calls per trunk |
03:55.32 | aydiosmio | the |
03:55.50 | aydiosmio | concurrent calls per trunk |
03:56.00 | aydiosmio | JT: I mean asterisk trunk |
03:56.12 | JT | with sip? |
03:56.14 | aydiosmio | you know, what trixbox calls contexts |
03:56.27 | aydiosmio | SORRY |
03:56.30 | JT | i know trixbox sometimes uses the term trunks :) |
03:56.48 | aydiosmio | I use freebpx |
03:56.56 | JT | haha |
03:57.03 | JT | there there, we have top notch counsellors |
03:57.16 | bigred | hrm. so any suggestions on how to rate limit? |
03:57.21 | bigred | i guess i could use call files |
03:57.27 | bigred | and count the number of files in the queue :-) |
03:58.22 | aydiosmio | you could write a script to monitor the asterisk API for calls, no? |
03:58.55 | aydiosmio | or use the CDR (seriously hackey) |
03:59.13 | aydiosmio | bigred: why would you rate limit vs. limiting concurrency? |
03:59.36 | bigred | i guess i am using the terms interchangably |
03:59.53 | bigred | i just want to say "no more than 200 concurrent outbound calls at a time" |
04:00.06 | aydiosmio | JT: PEER! That's what asterisk calls them |
04:00.17 | aydiosmio | bigred: configure your sip peer with call-limit = number : Number of simultaneous calls through this user/peer. |
04:00.36 | ManxPower | bigred: there are several features for handling limiting stuff in asterisk see GROUPCOUNT |
04:01.23 | aydiosmio | bigred: also incominglimit and outgoinglimit = Number : Limits for number of simultaneous active calls for a SIP client. Valid only for type=peer. |
04:02.33 | bigred | aydiosmio: the problem that i have seen with the call-limit feature is that if i am already at my limit, then the waiting calls will get a busy signal, so i will have to somehow guess the number of retrys |
04:03.02 | ManxPower | bigred: that is why GROUP and GROUCOUNT is handy |
04:03.26 | bigred | ManxPower: i am looking at that right now |
04:03.32 | ManxPower | bigred: also Dial will give you a HANGUPCAUSE or DIALSTATUS for help you with retries |
04:16.42 | *** join/#asterisk xo8ox (n=pride_32@wsip-66-210-250-2.ph.ph.cox.net) |
04:17.04 | xo8ox | anyone up ? |
04:19.36 | *** join/#asterisk Iamnach0 (i=Iamnacho@ip68-103-153-140.ks.ok.cox.net) |
04:19.56 | wunderkin | yes.. |
04:22.27 | *** join/#asterisk L|NUX (n=linux@unaffiliated/lnux/x-10290) |
04:39.48 | *** join/#asterisk wundaboy (n=hixscrip@c-67-171-137-48.hsd1.or.comcast.net) |
04:41.10 | vader-- | hey any of oyu uy familiar with wiring 66 blocks? |
04:41.10 | *** join/#asterisk reber (n=reber@cl-158.dub-01.ie.sixxs.net) |
04:41.25 | vader-- | i have a place that has 4 pots lines |
04:41.28 | aptura | just a punch down tool |
04:41.41 | vader-- | they want to put in 10 locations that have access to all 4 lines |
04:41.50 | vader-- | i am going to run cat 5 to each location |
04:41.58 | vader-- | and use a pair per connector |
04:42.22 | vader-- | a 66 punch down block can be setup to handle 2 phone lines right? |
04:42.31 | vader-- | so i would need two 66 punch down blocks |
04:42.57 | aptura | I am upgrading and seems zaptel src is missing autoconfig.h when using make or is there another reason. |
04:43.54 | aptura | vader you have to look at one but I think the blocks are grouped so more then one phone on the same circuit can ring. |
04:44.32 | aptura | I dont think it would be to hard to figure out. |
04:44.39 | sevard | vader--: you might want to google or wikipedia a 66 block if you are questioning its ability to handle two lines. |
04:45.40 | vader-- | this is pots lines |
04:45.47 | sevard | uh huh |
04:46.36 | aptura | sevard ever compile 1.2 and got some zaptel error of missing autoconfig.h |
04:46.37 | aptura | ? |
04:46.44 | *** part/#asterisk Noodleman (n=tuckerm@ip68-0-112-170.tu.ok.cox.net) |
04:47.07 | bkruse_home | now to sleep ;[ |
04:47.12 | sevard | http://www.wikipedia.org |
04:47.25 | bkruse_home | lata all |
04:47.33 | sevard | aptura: no |
04:47.38 | *** part/#asterisk bkruse_home (n=kruz@69.73.127.92) |
04:47.45 | sevard | are you using a hardware device? |
04:48.00 | aptura | you mean a fxo card yes. |
04:48.45 | Strom_C | woot...I'm totally managing to nub up this skinny firmware upgrade on my 7960 |
04:48.59 | *** join/#asterisk brian (i=brian@unaffiliated/brian) |
04:49.22 | sevard | aptura: do you have the modules for your card and zaptel compiled before you attempted to compile asterisk? |
04:50.04 | aptura | I downloaded the zaptel source from digiums site. Seems the xorcom download failed misserably so downloading the needed source code and do it from scratch :) |
04:50.13 | *** join/#asterisk Kumbang (n=kumbang@router-wmi.paume.itb.ac.id) |
04:50.35 | sevard | start from square one. |
04:50.37 | aptura | I have 1.2.16 zap src |
04:50.39 | aptura | hehe |
04:50.59 | aptura | great |
04:51.01 | aptura | :) |
04:51.08 | sevard | Strom_C: you using skinny now? |
04:51.12 | aptura | well this is a file that is missing :) |
04:51.15 | Strom_C | well, trying to |
04:51.30 | sevard | why? |
04:51.52 | JT | asterisk was treating him too well |
04:51.54 | Strom_C | why not? |
04:51.57 | *** join/#asterisk noli_r (n=chatzill@destiny-mis.d-one.net) |
04:51.58 | JT | he needed a challenge |
04:52.28 | sevard | I'm just curious, i never hear of anyone using skinny anymore |
04:53.26 | noli_r | I need help in configuring E1 DID to asterisk |
04:53.28 | lokkju_wrk | I'm looking for someone that has an active vonage motorola vt1005v ATA - I just need you to either run a tftp command, or to give me the MAC address of your unit - I am attempting to provide some unlock methods for it, and to do so I need to get a new formware image. someone want to help me? |
04:53.33 | noli_r | any tutorial available? |
04:54.16 | aptura | good luck! |
04:55.34 | sevard | skinny sounds pretty cool, central administration, faster for the cisco phones, etc |
04:55.43 | *** join/#asterisk asteriskguy (n=learnast@cpe-75-80-111-113.socal.res.rr.com) |
04:55.52 | Strom_C | sevard: yeah, i've been promising qwell i'd use it for some time now |
04:55.57 | JT | noli_r: do you mean an E1 PRI? |
04:56.21 | noli_r | EI, as Ive seen on Definity's config its CAS |
04:56.43 | noli_r | sorry but a have very little knowlege about E1 |
04:56.52 | JT | i see |
04:56.53 | Strom_C | bleh, "application upgrade failed" |
04:57.01 | voltagex | I'm baaaaaaaaaaaaaaaaaaaaack |
04:57.03 | JT | noli_r: what will asterisk be connecting to over E1? |
04:57.11 | Strom_C | hi voltagex |
04:57.41 | voltagex | JT: can I /msg you? |
04:57.47 | JT | voltagex: i suppose |
05:01.18 | *** join/#asterisk noli_r (n=chatzill@destiny-mis.d-one.net) |
05:01.38 | noli_r | :( |
05:01.47 | noli_r | got disconnected |
05:03.02 | noli_r | JT: can you please point me to some good tutorial? |
05:04.21 | *** part/#asterisk mog (n=mog@c-71-207-200-130.hsd1.al.comcast.net) |
05:05.43 | *** join/#asterisk neoalex (n=neoalex@user-0ccenic.cable.mindspring.com) |
05:05.48 | *** join/#asterisk b0on (n=b0on@cpe-66-61-165-172.indy.res.rr.com) |
05:06.14 | neoalex | hey guys... why is it that my asterisk is running two identical mpg123 processes for my MOH |
05:06.33 | neoalex | I'm using an online radio as my MOH |
05:07.53 | [TK]D-Fender | neoalex, Perhaps you left a MOH definition with an mpg123 mode still in existance yeet not attached to a device... |
05:08.45 | neoalex | I didn't I checked, plus the processes are identical... meaning they're playing the same URL |
05:09.30 | b0on | could you maybe not have commented out a previous MOH? |
05:09.57 | neoalex | nope... I checked |
05:10.03 | vader-- | tkd work much with 66 punch down blocks? |
05:10.15 | Strom_C | vader--: i punch those down all the time |
05:10.32 | b0on | a whole main frames worth |
05:10.51 | vader-- | i have a place that has 4 pots lines |
05:10.52 | b0on | :) |
05:11.01 | vader-- | and they want to have 10 jacks that have all 4 lines available to them |
05:11.15 | Strom_C | vader--: simple |
05:11.16 | vader-- | if get a 66M block i can put two pots lines per right? |
05:11.25 | Strom_C | vader--: get one pair of blocks |
05:11.31 | Strom_C | on one block, have the telco lines terminate |
05:11.33 | vader-- | so i would need to 66 blocks |
05:11.39 | Strom_C | on the other block, have the station wire terminate |
05:11.43 | vader-- | to =two |
05:12.14 | Strom_C | then run jumpers from the telco block to the station block, using the non-cutting side of the 66 blade to have the line show up on multiple station cables |
05:12.25 | vader-- | ya |
05:12.34 | vader-- | like a zig zag down the blocks |
05:12.41 | Strom_C | well, down the ONE block |
05:12.42 | vader-- | on the innter channels |
05:12.57 | Strom_C | you should only have one jumper per line |
05:13.09 | *** join/#asterisk Noodleman (n=tuckerm@ip68-0-112-170.tu.ok.cox.net) |
05:13.24 | neoalex | b0on: just to be on the safe side, I just copied only the class I wanted in musiconhold.conf, and reloaded and it still started two mpg123s |
05:13.41 | b0on | thats crazy |
05:13.47 | b0on | i've never seen that happen b4 |
05:14.00 | neoalex | le'me just restart asterisk alltogeather |
05:14.03 | b0on | (not saying its not tho) |
05:14.20 | Noodleman | question: do payphones still use dtmf to signal coins, and could asterisk/zap device/whatever pick those up? |
05:14.32 | b0on | payphones don't no |
05:14.37 | Strom_C | Noodleman: the phones use 1700+2200 |
05:14.42 | Strom_C | not "DTMF" |
05:14.52 | Strom_C | and yes, in theory, one could hack the zap driver to recognize it |
05:15.06 | b0on | setting up some SIP payphones? :) |
05:15.10 | Noodleman | hrmzz |
05:15.18 | Noodleman | b0on: i'm thinking about it :-) |
05:15.21 | voltagex | more like an asterisk redbox |
05:15.28 | *** join/#asterisk Beirdo (n=gjhurlbu@unaffiliated/beirdo) |
05:16.10 | Noodleman | *gears turn upstairs" |
05:19.29 | neoalex | b0on: http://paste.biz/paste-1239.html see if you think there's anything wrong with my musiconhold.conf |
05:20.22 | neoalex | really weird |
05:20.34 | b0on | you restarted * |
05:20.37 | b0on | ? |
05:20.43 | neoalex | yeah |
05:20.48 | neoalex | stop gracefully |
05:21.01 | neoalex | killall mpg123 (none left, but just to make sure) |
05:21.09 | neoalex | then asterisk again |
05:21.17 | neoalex | started two mpg123s again |
05:25.27 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
05:26.05 | *** join/#asterisk voiper1 (n=luke@ozvoip.dsl.onthenet.net) |
05:26.24 | b0on | wow |
05:26.30 | b0on | i'm sorry man.. its beyond me |
05:27.33 | andrew` | i once had something like 75 mpg123s |
05:28.05 | neoalex | ha... so I guess I'm a lightweight |
05:28.10 | JT | to correspond with a load average of 75? ;) |
05:28.24 | b0on | mine only opened up one with that conf |
05:28.27 | b0on | so... i know its not that |
05:29.00 | neoalex | weirder |
05:29.08 | vader-- | what the price of cat5e plenum 1000ft going for? |
05:29.22 | neoalex | forget about it... it's not that big of a deal anyway |
05:29.27 | neoalex | le'me ask you something else |
05:29.40 | neoalex | just noticed AsteriskNow exists |
05:29.53 | neoalex | can I install the interface over my existing asterisk |
05:30.08 | JT | asterisknow is a distro that wipes everything |
05:30.10 | *** join/#asterisk fx0 (n=ariel@cypher.punk.net) |
05:30.15 | JT | asterisk-gui is the interface |
05:30.21 | b0on | what distro u running? |
05:30.45 | b0on | oh. asterisknow? never used it |
05:31.02 | b0on | i have used freepbx and i've really really liked that |
05:31.12 | vader-- | how is asterisk-gui? |
05:31.54 | *** join/#asterisk `p4r14h (n=j0sh@69.92.145.178) |
05:32.02 | neoalex | I'm using it, and I wanted to put freepbx but the guys supporting it said, migrating the current configs I have would be a royal pain |
05:32.18 | neoalex | I'm using slackware I meant |
05:32.55 | neoalex | ok... so JT, is asterisk-gui downloadable and instalable sepparately |
05:33.52 | JT | it is, but i don't use it |
05:34.24 | b0on | hey neoalex.. look in your usr/bin .. check out the symlinking for mpg123 |
05:35.29 | *** join/#asterisk ShadowHntr (i=sentinel@wikipedia/Shadowhntr) |
05:35.32 | neoalex | no symlinks, only the executable itself |
05:36.02 | neoalex | JT would I be able to use the current configs? |
05:36.11 | JT | no idea |
05:36.33 | b0on | have u tried executing the program itself outside of asterisk, to see if it opens up the dual processes |
05:39.14 | neoalex | no... just a sec |
05:40.24 | neoalex | yes it does... ok then it's not an asterisk issue |
05:41.22 | vader-- | what is a good 4 pots line phone? |
05:41.32 | JT | what |
05:41.37 | JT | 4 pots line phone? |
05:41.48 | vader-- | ya |
05:41.53 | vader-- | analog phone lines |
05:42.01 | vader-- | pots = plain old telephone systems |
05:42.20 | JT | i know what it stands for |
05:42.28 | JT | what do you mean 4 pots line phone? |
05:42.46 | vader-- | a phone that can handle 4 lines |
05:43.17 | JT | those sort of things are relics of the past |
05:43.23 | JT | and pretty rare these days |
05:44.17 | voltagex | ebay it! |
05:45.02 | JT | yeah that's probably where he'll need to go |
05:45.21 | vader-- | ya im not sure what to recommend to this company |
05:45.29 | vader-- | they want to have all 4 lines on every desk |
05:45.30 | b0on | might get lucky at a Salvation Army or goodwill |
05:45.37 | vader-- | and they have to be pots |
05:45.50 | b0on | why not setup some channel banks |
05:46.01 | vader-- | they don't want pbx or voip |
05:46.03 | b0on | and have them connect to said four lines thorugh each computer at the desk through a softphone |
05:46.09 | b0on | ah |
05:46.09 | JT | vader--: can't you just say to them "you guys are crackheads!" |
05:46.10 | b0on | well |
05:46.35 | aptura | odd 4 lines to every desk and no pbx? |
05:46.42 | vader-- | it's only 4 desks |
05:46.46 | aptura | okay |
05:46.48 | JT | sometimes a client needs to be told when they're dead wrong |
05:46.56 | JT | that's just a stupid setup |
05:47.06 | b0on | heh.. is the answering machine micro-cassette based too? |
05:47.06 | vader-- | and they can't use voi because one of the line is used for programming some sort of hardware panel |
05:47.11 | vader-- | and voip doesn't workwell with it |
05:47.12 | aptura | i could have seen this as a possible wiring nightmare if it was big :) |
05:47.23 | vader-- | and one of the other lines is a fax line |
05:47.23 | JT | what hardware panel? |
05:47.32 | vader-- | security alarm |
05:47.37 | JT | ah |
05:47.48 | JT | people should not be making calls on those lines anyway |
05:47.51 | aptura | what kind of biz is this? |
05:48.13 | aptura | does the alarm have a panic button? |
05:48.19 | neoalex | vader--: if they still want to be in the dark ages, then here: http://www.nextag.com/4_-_line-phone/search-html |
05:48.22 | aptura | probebly not :) |
05:48.27 | b0on | ya.. it just says .. "That was easy!" |
05:48.31 | vader-- | monitoring of the alarms is done by another company |
05:48.34 | JT | you can get 5 line analogue phones |
05:48.35 | vader-- | they just do some sort of programing |
05:48.37 | JT | i have one at home |
05:48.42 | JT | mediatrix or something |
05:48.43 | *** join/#asterisk flip123 (n=flip@ip70-162-47-138.ph.ph.cox.net) |
05:48.50 | JT | sitting in its box |
05:49.05 | aptura | I can see the fire alarm going off and sending a signal to the ununicator panel then trying to dial out when the line is in use. big problem. |
05:49.07 | flip123 | sigh. Some days |
05:49.27 | flip123 | upgraded to 1.2.17, because of the skinny/chan_sip vulns |
05:49.38 | flip123 | now the audio goes all choppy after about 5 seconds |
05:50.34 | flip123 | though, I did a kernel upgrade at the same time, I suppose I could blame that |
05:50.40 | vader-- | ya i need 8 rj 45 connectors and 32 6 pin connectors |
05:50.45 | vader-- | with wall plates |
05:51.42 | flip123 | hmm |
05:51.53 | flip123 | looks like there are some reports of choppy sound on 2.6.9-42 |
05:51.54 | *** join/#asterisk oej (n=olle@apollo.webway.se) |
05:52.13 | flip123 | but its in reference to vmware, which I don't use |
05:52.14 | vader-- | fucking cat5e is exspensive as shit |
05:52.32 | b0on | copper in general |
05:52.38 | vader-- | ya |
05:52.55 | vader-- | 1000ft plenum is close to 250-300$ |
05:53.00 | b0on | holy crap |
05:53.05 | JT | lol, plenum |
05:53.11 | JT | that's why |
05:53.14 | vader-- | ya |
05:53.18 | b0on | reminds me of college |
05:53.20 | vader-- | well it's in a ceiling |
05:53.25 | JT | world copper prices are high though |
05:53.46 | neoalex | vader--: http://www.warehousecables.com/cgi-bin/shopper.cgi?keywords=%22cat%205e%22%20and%20solid%20and%20utp%20not%20plenum&search=action |
05:53.55 | neoalex | $108 for 1000 ft |
05:54.03 | JT | neoalex: he needs plenum |
05:54.28 | neoalex | oh... yeah that's 210 there too |
05:55.18 | b0on | ok heres what you do.. you find a new some construction going on for new office buildings.. |
05:55.36 | b0on | sneak in at night.. swipe a couple boxes of plenum |
05:55.42 | flip123 | yarg, anyone else reported issues with 2.6.9-42 and asterisk? |
05:56.48 | JT | flip123: why not just upgrade? |
05:57.14 | neoalex | b0on: great idea |
05:57.23 | flip123 | JT: asterisk or the kernel? |
05:57.27 | JT | kernel |
05:57.27 | voltagex | what's plenum? |
05:57.31 | JT | that's a very old kernel |
05:57.32 | flip123 | I'm trying to stick with the centos packaged kernels |
05:57.36 | JT | pfft |
05:57.46 | voltagex | hang on, trixbox uses centos |
05:57.55 | flip123 | voltagex, lots of people use centos |
05:58.05 | JT | i don't :) |
05:58.09 | b0on | ya.. not that great of an idea.. my mind is gone, and i'm up writing design doco's |
05:58.12 | JT | and screw pre-packaged kernels |
05:58.14 | *** join/#asterisk MooingLemur (n=troy@unaffiliated/mooinglemur) |
05:58.16 | flip123 | and lots of people dont :-) |
05:58.17 | b0on | and not watning to |
05:58.17 | neoalex | <PROTECTED> |
05:58.18 | neoalex | <PROTECTED> |
05:58.29 | JT | plenum refers to airspace |
05:58.49 | voltagex | oh, it's a brand? |
05:58.51 | JT | no |
05:58.56 | JT | it's a specification of cable |
05:59.03 | *** join/#asterisk tengulre (n=tengulre@222.90.66.10) |
05:59.07 | JT | that has a fire-resistant protective jacket |
05:59.13 | JT | required by a lot of building codes |
05:59.31 | neoalex | youp |
05:59.34 | voltagex | I want some |
05:59.37 | neoalex | plenum space is the airspace |
05:59.51 | neoalex | plenum cable goes into the plenum space and must have that jacket |
05:59.56 | voltagex | then if I have a pyromania outburst, at least my cable's ok |
06:00.12 | Strom_C | it's not that the jacketing is fire-resistant |
06:00.22 | neoalex | it's not ok... it's just slightly better off then regular cable |
06:00.25 | Strom_C | it's that the jacketing doesn't produce toxic smoke |
06:00.36 | Strom_C | or at least, smoke that's as toxic as non-plenum cable |
06:00.36 | rudholm | a plenum in a building is anything that has positive air pressure with respect to the rest of the building, in this context "plenum cable" means cable rated for use in building plenums. |
06:00.41 | neoalex | it's both Strom_C |
06:00.53 | voltagex | ok, if I go pyromaniac, I won't breathe toxic smoke from my cables. |
06:01.12 | rudholm | it's not really for you, it's to protect firefighters mostly. |
06:01.17 | neoalex | yeah... that's why I used it throughout my apartment too |
06:01.28 | neoalex | :)) |
06:01.35 | voltagex | oh ok, I don't matter, only the firefighters do. |
06:01.41 | *** join/#asterisk zeeesh (i=zeeesh@202.38.55.125) |
06:01.42 | zeeesh | hi |
06:01.53 | neoalex | they hav oxygen anyway |
06:02.00 | rudholm | well, the idea being that in a commercial building, the occupants evacuate quickly, but the firefighters do not |
06:02.13 | voltagex | yeah. |
06:02.13 | b0on | heh .. start free-basing your plenum... |
06:02.15 | rudholm | but yes, you don't count as much as a fireman :) |
06:02.20 | voltagex | :P |
06:02.31 | tengulre | hi,all |
06:02.36 | neoalex | aaaaw.... that's gotta hurt voltagex :D |
06:02.43 | voltagex | Firefighting outfit: +10 of worthiness |
06:02.56 | tengulre | what is different between codec g723 and g723A? |
06:02.57 | voltagex | neoalex: I've been in here for a while, I'm getting use to it |
06:03.08 | neoalex | =)) |
06:03.20 | flip123 | JT: yeah, generally I agree, prepackaged kernels suck. |
06:03.46 | flip123 | but with remote boxes, sometimes its just the way to be |
06:04.09 | flip123 | not that it helps any tonight |
06:04.22 | voltagex | flip123: I don't actually know what hardware is in this box so the latest ubuntu kernel suits me fine |
06:05.00 | flenders | root@pbx:/etc/asterisk# uname -a |
06:05.01 | flenders | Linux pbx 2.6.8-3-686 #1 Tue Dec 5 21:26:38 UTC 2006 i686 GNU/Linux |
06:05.05 | flenders | :D |
06:05.10 | flenders | and it works just fine |
06:05.29 | neoalex | Linux pbx... what's that |
06:05.38 | voltagex | neoalex: hostname |
06:05.43 | aptura | I once worked at a In route ATC center and the fire/hvac system is interesting. The main ATC control room is supplied with positive air pressure. So the entire building except that room can be on fire and the controlers should still do there job. Only in a last second case do the controlers leave but then all inbound air traffic comming into Vancouver would be in limbo. 50 miles of wires in that one building ;) |
06:05.44 | flenders | neoalex: that's my hostname |
06:05.55 | neoalex | oh... sorry... /me dumb |
06:06.00 | flip123 | flenders, except it has multiple locate root vulnerabilities |
06:06.06 | flip123 | s/locate/local |
06:06.19 | flip123 | trival ones |
06:06.28 | flenders | flip123: no one has access to it |
06:06.38 | voltagex | flip123: why are you using something that old? |
06:07.01 | flip123 | voltagex, I'm using 2.6.9-42 |
06:07.18 | rudholm | flip123: that's like, so 10 minuts ago |
06:07.28 | voltagex | ah, it was flenders with the really old one |
06:07.30 | flip123 | because its the latest kernel available for centos/rhel |
06:07.33 | voltagex | still, yours is old too |
06:07.35 | neoalex | voltagex: I use 2.4.33.3 if you think that's old |
06:07.38 | neoalex | :D |
06:07.44 | flenders | voltagex: debian stable |
06:07.45 | flip123 | yes, i know its old |
06:07.59 | flip123 | 2.6.9 is, supposedly, stable |
06:07.59 | voltagex | neoalex: isn't that the latest 2.4 stable? |
06:08.01 | *** join/#asterisk scardinal (n=supreme@port816.ds1-suoe.adsl.cybercity.dk) |
06:08.10 | voltagex | flenders: aaah. makes sense now. |
06:08.12 | *** join/#asterisk wundaboy (n=hixscrip@c-67-171-137-48.hsd1.or.comcast.net) |
06:08.22 | voltagex | flenders: that's cutting edge for debian. |
06:08.23 | neoalex | yes it is, but it's 2.4 not 2.6, therefore it's old |
06:08.23 | flip123 | flenders, wow, debian stable has moved onto 2.6? |
06:08.35 | flip123 | hasn't 2.6 only been out for 3 years or so? |
06:08.41 | flenders | :D |
06:08.43 | voltagex | meh, 2.4 works better in some cases |
06:09.03 | neoalex | 2.4.34.2 is actually the latest |
06:09.26 | neoalex | mine is whateva came with Slack 11.0 |
06:09.48 | flip123 | so I assume there's no work around for the alleged 2.6.9-42 issues? |
06:09.51 | flenders | it doesn't bother me... as I said, no one access it |
06:10.43 | flip123 | flenders, nod, as long as you have no services exposed to the network, you should be fine |
06:11.32 | flenders | flip123: exactly... all you can reach on this box is SIP and ssh from my box |
06:12.01 | vader-- | yo are you guys familiar with number porting? |
06:12.02 | voltagex | do I need a DNS server on an Asterisk box? |
06:12.29 | flip123 | voltagex, no |
06:12.43 | *** part/#asterisk b0on (n=b0on@cpe-66-61-165-172.indy.res.rr.com) |
06:12.44 | vader-- | there is a number this one person owns and this person wants to move it to another person's account |
06:12.52 | vader-- | verizon says they can't port the number |
06:13.26 | Strom_C | vader--: is it across rate center boundaries? |
06:13.38 | vader-- | same city |
06:13.48 | Strom_C | doesnt matter |
06:13.52 | Strom_C | is it across rate center boundaries? |
06:13.58 | vader-- | i dunno |
06:14.31 | flip123 | guess I'll try downgrading to 2.6.9-34 |
06:14.36 | flenders | aren't numbers tied to exchanges? |
06:14.40 | Strom_C | vader--: which city |
06:14.51 | Strom_C | flenders: numbers are tied to rate centers |
06:14.56 | vader-- | philadelphia, pa |
06:15.33 | voltagex | doh, just went to make a call and realised asterisk isn't installed yet |
06:15.38 | Strom_C | PM me the area code and prefix of the number you're trying to port, and the number on the account that you're trying to port it to |
06:16.01 | flenders | Strom_C: had no idea what a rate center was |
06:16.32 | voltagex | question...what's involved in connecting a mobile phone up to asterisk? |
06:17.14 | *** join/#asterisk Avochelm (n=damien__@gw-morphett.koalatelecom.com.au) |
06:18.29 | neoalex | voltagex: a SIP GSM gateway |
06:18.30 | neoalex | like this one: |
06:18.30 | neoalex | http://www.gsmsave.co.uk/VOIP_GSM_SIP_Gateway.htm |
06:18.30 | voltagex | neoalex: I was told it was possible using bluetooth |
06:18.39 | JT | flenders: rate center is an american concept |
06:18.42 | *** join/#asterisk [Airwolf] (n=airwolf@voip.ymav.nl) |
06:18.51 | flenders | JT: how do we call it here? |
06:18.54 | neoalex | hmmm... I'm not sure voltagex |
06:18.58 | JT | no idea |
06:19.04 | JT | voltagex: chan_cellphone |
06:19.07 | *** join/#asterisk k-man_ (n=jason@unaffiliated/k-man) |
06:19.08 | JT | and a bluetooth adapter |
06:19.18 | k-man_ | anyone heard of the epygi voip pabx system? |
06:19.24 | neoalex | now I am... here: http://www.voip-info.org/wiki-Asterisk+Bluetooth+channels |
06:19.30 | JT | heard of it |
06:19.31 | JT | it costs a bomb |
06:19.41 | k-man_ | jt, does it? |
06:19.49 | JT | yes |
06:19.55 | JT | i don't see the point in it |
06:19.59 | k-man_ | my wifes work is thinking of getting it - and im trying to convince them to go asterisk |
06:20.09 | JT | shrug |
06:20.13 | JT | up to them i guess |
06:20.19 | voltagex | JT: will that kind of thing work over USB1? |
06:20.21 | k-man_ | the vendor says asterisk and epygi will have similar cost of install, but asterisk has a higher cost to maintain |
06:20.28 | voltagex | as always |
06:20.34 | JT | the question i'd ask is who would be maintaining the asterisk |
06:20.42 | k-man_ | jt, they would |
06:20.45 | neoalex | voltagex: http://www.voip-info.org/wiki-Asterisk+Bluetooth+channels |
06:20.53 | k-man_ | jt, either way, they would be maintaining it |
06:21.08 | JT | voltagex: i don't see why not, bluetooth isn't high bandwidth |
06:21.44 | voltagex | ok |
06:21.54 | voltagex | unfortunately only usb1 ports on this old box |
06:22.31 | neoalex | the problem with bluetooth is you obviously have to be close to the box |
06:22.31 | *** join/#asterisk Mahmoud (n=fake@unaffiliated/mahmoud) |
06:22.32 | voltagex | <PROTECTED> |
06:22.51 | voltagex | I'll just buy the cheapest bluetooth enabled phone I can find on ebay. |
06:23.21 | neoalex | oh... you want to use your cellphone service as a peer? |
06:23.47 | voltagex | neoalex: if you mean to receive SMS to asterisk, yes |
06:23.52 | k-man_ | is it possible to run asterisk on an embeded machine with no hdd and no fan? |
06:24.04 | k-man_ | or no moving parts hdd at any rate, flash is fine |
06:24.24 | neoalex | k-man_: yes |
06:24.38 | Mahmoud | any one tried this out? http://www.digium.com/en/products/hardware/asteriskappliance.php |
06:24.45 | Mahmoud | Asterisk APpliance |
06:24.45 | k-man_ | neoalex, is there any hardware you can point me to? |
06:25.04 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
06:25.29 | neoalex | k-man_: Mahmoud just did, that one is a pretty good choice |
06:25.44 | neoalex | if you want to install it on one yourself then... |
06:25.47 | neoalex | wait a sec |
06:26.15 | neoalex | http://www.embeddedarm.com/epc/ts7400-spec-h.htm |
06:26.15 | Mahmoud | i'm actually looking at Sipura 3000 |
06:26.23 | *** join/#asterisk ftexcom (n=ftexcom@14.Red-80-26-4.staticIP.rima-tde.net) |
06:26.33 | neoalex | I meant the asterisk appliance |
06:26.34 | k-man_ | thanks |
06:26.37 | Mahmoud | sipura is the best for my choice, one fxs/fxo port |
06:27.08 | neoalex | k-man_: openwrt also has an asterisk package I believe |
06:27.19 | k-man_ | neoalex, ok, thanks |
06:27.25 | Mahmoud | neoalex, what's this? |
06:27.37 | neoalex | what's what? |
06:27.48 | Mahmoud | the chip you posted earlier |
06:28.24 | flip123 | oh thank goodness |
06:28.38 | flip123 | a downgrade to 2.6.9-34.0.2 solved the problem |
06:28.43 | flip123 | (choppy audio) |
06:28.46 | neoalex | it's an embedded device |
06:28.56 | flip123 | which means tomorrow, when this customer goes to show me the issue |
06:29.14 | flip123 | I can continue to blame his internet service provider |
06:29.15 | neoalex | has an ethernet, 2 USB 2.0 and an SD reader |
06:29.18 | k-man_ | flip123, please tell me about your choppy audio problem , i have been having the same problem |
06:29.29 | k-man_ | flip123, was it the voice prompts that were choppy? |
06:29.31 | Mahmoud | neoalex, how is it related to asterisk appliance? |
06:29.33 | flip123 | k-man_, i upgrade to kernel 2.6.9-42 |
06:29.36 | flip123 | k-man_, yes |
06:29.50 | flip123 | k-man_, anyway, i had upgraded to kernel 2.6.9-42 |
06:30.14 | flip123 | and the audio got super choppy |
06:30.19 | flip123 | and would degrade over time |
06:30.19 | neoalex | I was responding to k-man which was looking for a no fan no moving parts device for asterisk |
06:30.19 | k-man_ | flip123, which distro? |
06:30.19 | flip123 | k-man_, centos |
06:30.21 | flip123 | I downgraded to 2.6.9-34.0.2 |
06:30.27 | flip123 | and the choppy audio is fixed |
06:30.31 | *** join/#asterisk voltagex (n=voltagex@124-254-123-181-dsl.ispone.net.au) |
06:30.35 | Mahmoud | neoalex, do you use this embeded computer? |
06:30.36 | k-man_ | neoalex, how would you hook the ts700 up to pstn lines? |
06:31.18 | neoalex | you wouldn't, you'd have to go SIP only |
06:31.39 | flip123 | now to think of anything else I should fix before I have to talk to this customer tomorrow |
06:31.54 | k-man_ | neoalex, ah, ok |
06:31.56 | Mahmoud | k-man_, buy Sipura 3000, FXS/FXO PCI cards, or asterisk appliance |
06:32.13 | Mahmoud | k-man_, if you want one fxs and one fxo port, then Sipura 3000 is the best choice |
06:32.31 | neoalex | Mahomoud the TS7400 deosn't have a PCI bus |
06:32.51 | k-man_ | ok |
06:33.03 | Mahmoud | neoalex, i'm not talking about your embeded computer |
06:33.21 | Mahmoud | neoalex, TDM400 has pci bus |
06:33.29 | flip123 | i could be a real bastard and drop all these test calls out of the billing system |
06:33.31 | neoalex | <k-man_>neoalex, how would you hook the ts700 up to pstn lines? |
06:33.33 | neoalex | he was |
06:33.36 | voltagex | Strom_C: you still around? |
06:33.42 | Strom_C | yes |
06:33.47 | neoalex | nevermind... |
06:33.47 | flip123 | In fact, I am a real bastard, and I am going to do that |
06:34.03 | voltagex | any compile/configure options I should use for asterisk? |
06:34.20 | voltagex | shiny new ubuntu server install is ready |
06:35.20 | Strom_C | voltagex: go with defaults for now |
06:35.27 | voltagex | ok |
06:36.25 | voltagex | 17mb...I am going over my download quota this month |
06:36.46 | flenders | voltagex: who are you with? |
06:36.47 | voltagex | just installing build-essential then I'll compile. |
06:36.59 | voltagex | flenders: australian isp, Southern Phone |
06:37.18 | flenders | voltagex: i'm in .au too |
06:37.38 | voltagex | flenders: ah, so we can bitch and whine about broadband here together! |
06:37.46 | flenders | :D |
06:37.53 | JT | oh no |
06:37.57 | JT | not another whirlpool |
06:38.14 | voltagex | JT: oh come on! we're justified in whinging |
06:38.21 | flenders | voltagex: well, I don't have much to complain about... I guess adsl2 plans are good for what you pay for them |
06:38.27 | JT | my broadband is fine :) |
06:38.37 | *** join/#asterisk Kizmet (n=Kizmet@AeriaSolutionsPtyLtd.fe0-1.aes-brd-0.agl.cbr.as-ip.net.au) |
06:38.43 | flenders | JT: which one? |
06:38.44 | voltagex | yes, you lucky people who can get ADSL2 |
06:38.57 | flenders | voltagex: :D |
06:39.22 | flenders | I have 2 x internode and 1 x iinet here |
06:39.27 | flenders | and with iinet at home too |
06:39.55 | tzafrir | aptura, here? |
06:40.42 | Kizmet | 2x Internode Syncing at 23821/2282 bound using a Cisco 3620 |
06:40.47 | Kizmet | *grin* |
06:41.29 | voltagex | Kizmet: how the f.... did you get that high sync |
06:41.40 | voltagex | are you sleeping in internode's dslam? |
06:41.54 | Kizmet | voltagex, If I stick my head out my windo I can hear the exchange humming |
06:41.59 | voltagex | even APC didn't get that high (I don't think) in a test lab! |
06:42.06 | voltagex | Kizmet: you lucky bastard |
06:42.14 | Kizmet | its over the back fence |
06:42.35 | JT | Kizmet: you have 2 connections at home? |
06:42.41 | Kizmet | JT, Yes. |
06:42.51 | JT | nice |
06:42.55 | Kizmet | I also have a PRI/10 for fun |
06:42.59 | JT | cool |
06:43.04 | *** join/#asterisk Kumbang (n=kumbang@router-wmi.paume.itb.ac.id) |
06:43.04 | JT | does it actually bind properly? |
06:43.12 | JT | like it's supported by internode? |
06:43.12 | voltagex | I'll be sleeping outside of your house, stealing your wifi, Kizmet |
06:43.19 | Kizmet | yes, Its bound at the internode side of things too |
06:43.30 | JT | did you need to pay them big bucks to do that? |
06:43.39 | Kizmet | Its connected to my companys 'Internode Business Connect' network |
06:43.55 | JT | so not to the Internet? |
06:44.00 | Kizmet | yep approx $800 p/m for both the tails |
06:44.04 | JT | so i have to ask, did *that* cost a lot? |
06:44.05 | zeeesh | want to configure mailbox .. for my .. peers .. if i have ... 2 .. 3 or 10 peers .. i want .. everybody .. shud press the .. same digit for listening voicemail.. could not found how to do it ... ?? |
06:44.06 | JT | right |
06:44.21 | JT | $800 just on your house end? |
06:44.29 | Kizmet | It routes back through my companys Rack in the Datacenter |
06:44.31 | Kizmet | yep |
06:44.46 | voltagex | you're rich too. |
06:44.48 | JT | i assume the PRI does something useful |
06:44.53 | JT | or you just like spending money ;) |
06:45.12 | Kizmet | Not really. Just serves as a backup PRI for the company |
06:45.25 | JT | fair enough |
06:45.32 | JT | through hellstra? |
06:45.33 | voltagex | ah, he may not have a house, there might be all kinds of cabling coming into a cardboard box in the ghetto. |
06:45.39 | Kizmet | hehe, TransACT |
06:45.43 | JT | ah |
06:45.46 | Mahmoud | wow found sipura 3000 |
06:45.51 | Mahmoud | they'll tell me the price later |
06:45.57 | Kizmet | lol |
06:46.01 | JT | spa-3102 is current model |
06:48.04 | Mahmoud | JT, didn't find it in sipura.com :/ |
06:48.07 | Mahmoud | nor linksys.com |
06:48.20 | Mahmoud | JT, any advantages over 3000? |
06:48.23 | JT | yeah anyway, it exists |
06:48.30 | JT | well the 3000 is end of line |
06:49.43 | Kizmet | bbl |
06:50.32 | flip123 | welp, looks like I can sleep ok tonight |
06:50.52 | flip123 | g'night all |
06:51.27 | Mahmoud | i got sipura at 300AED |
06:51.38 | Mahmoud | means 81USD |
06:51.50 | Mahmoud | horribly cheap? |
06:52.04 | Mahmoud | people here look at it as a normal adapter :P |
06:53.27 | Mahmoud | sipura 3000 |
06:53.34 | Mahmoud | they didn't have sipura 3102 :/ |
06:57.45 | voltagex | <PROTECTED> |
06:57.45 | voltagex | <PROTECTED> |
06:57.53 | voltagex | Strom_C: help now! |
06:58.49 | Strom_C | um |
06:58.55 | Strom_C | you must ask me specific questions |
06:59.05 | Mahmoud | man Strom_C |
06:59.12 | voltagex | ok, which config do I edit now |
06:59.20 | Mahmoud | Strom_C is the man page |
06:59.24 | voltagex | yep |
06:59.28 | Strom_C | voltagex: did you "make samples"? |
06:59.36 | voltagex | he was the one who made me remove trixbox |
06:59.46 | voltagex | so now he gets to help me! |
06:59.50 | voltagex | Strom_C: yes |
06:59.53 | Strom_C | ok |
07:00.07 | Strom_C | if you're setting up your iax provider, give iax.conf and extensions.conf a go |
07:00.30 | ParaNoir | hey, anybody experienced with SwyxWare? |
07:00.48 | Mahmoud | i hate "make samples" |
07:00.50 | voltagex | Strom_C: well apparently IAX isn't working properly with my provider at the moment, but I'm not tethered to any one provider yet |
07:00.55 | voltagex | :/ |
07:01.06 | Strom_C | how about this |
07:01.08 | Strom_C | set up a softphone |
07:01.24 | Strom_C | then I won't have to hear you complaining about how it costs 25c a call to use your box |
07:01.26 | voltagex | done |
07:01.30 | voltagex | lol |
07:01.42 | voltagex | set up a softphone with my current provider |
07:01.47 | voltagex | ? |
07:01.58 | *** join/#asterisk FlatFoot (n=simon@80.88.192.83) |
07:01.59 | Strom_C | no |
07:02.00 | Strom_C | with your box |
07:02.06 | voltagex | done |
07:02.10 | voltagex | already |
07:02.44 | Strom_C | that was quick |
07:03.15 | *** join/#asterisk nuonguy (n=john@c-24-6-175-26.hsd1.ca.comcast.net) |
07:03.22 | voltagex | same IP from the trixbox install |
07:03.28 | voltagex | so the softphone is set up |
07:03.37 | Strom_C | and what about on the asterisk box side of things? |
07:03.44 | Strom_C | did you configure sip.conf? |
07:04.07 | voltagex | err, just looking at iax.conf now because I will probably go with another provider on what I'm hearing about Faktortel - not good |
07:05.08 | voltagex | woot, I have a free $10 mynetfone credit here |
07:06.05 | voltagex | even better, they provide the asterisk configuration for me :P |
07:06.12 | voltagex | I win, Strom_C |
07:06.32 | Strom_C | god, you're irritating |
07:06.34 | voltagex | although, are the configurations listed up the top ok? http://www.mynetfone.com.au/support/ |
07:07.01 | Strom_C | yeah, but you need to pick the relevant bits out of the config |
07:07.28 | Strom_C | and their extensions.conf example uses way deprecated syntax |
07:07.52 | voltagex | I'm not intentionally trying to be irritating |
07:08.03 | voltagex | oh, and extensions.conf doesn't look that scary. |
07:09.34 | voltagex | whoa, mynetfone have lower prices than the others I've looked at |
07:09.37 | ParaNoir | hey, is it possible to use a bri card with asterisk 1.4? |
07:10.59 | Strom_C | ParaNoir: yes |
07:11.02 | Strom_C | b410p |
07:12.24 | *** join/#asterisk tzafrir_laptop (n=tzafrir@62.90.10.53) |
07:12.51 | ftexcom | ParaNoir never use a bri card. It's hell on earth |
07:12.51 | rudholm | Strom_C: what about with National ISDN BRI? :-p |
07:13.21 | Strom_C | rudholm: please hold |
07:13.28 | Strom_C | your estimated wait time is |
07:13.30 | Strom_C | six |
07:13.31 | ParaNoir | why? |
07:13.31 | Strom_C | months |
07:13.34 | ParaNoir | works like a charm ;) |
07:13.55 | ftexcom | ParaNoir I always have issues configuring it...but maybe it's because I suck |
07:14.01 | ParaNoir | ;) |
07:14.16 | *** join/#asterisk pif (n=ldm@zenon.apartia.fr) |
07:14.22 | ParaNoir | could get quadbri and single bri working, not bad for a newbie right? ;) |
07:14.36 | *** join/#asterisk plasmid (n=noway@c-24-127-166-193.hsd1.pa.comcast.net) |
07:14.40 | rudholm | Strom_C: six months?!? damn, I thought I was done with that kind of service when I cancelled Charter. |
07:14.41 | ftexcom | I also had a b410p working...once |
07:14.55 | rudholm | ftexcom: with what kind of BRI? |
07:15.04 | *** join/#asterisk nasls_lsa (n=chatzill@athedsl-145799.otenet.gr) |
07:15.06 | rudholm | ftexcom: Euro ISDN? |
07:15.17 | ftexcom | rudholm yeap |
07:22.26 | *** join/#asterisk yonahw-work (n=yonahw-w@genie03-173-74.inter.net.il) |
07:29.02 | voltagex | ok Strom, now that I have proper asterisk, would you be able to help me get FWD working properly? |
07:29.33 | *** join/#asterisk putzz (n=me@CPE00155824933c-CM00111ae07ac2.cpe.net.cable.rogers.com) |
07:30.10 | Strom_C | maybe |
07:30.17 | Strom_C | it's been literally forever since I screwed with FWD |
07:30.51 | JT | voltagex: fwd's online help, and the book should be enough to get you up and running with FWD |
07:31.12 | voltagex | JT: I also found a nice tutorial |
07:31.23 | voltagex | will Asterisk tell me if I'm using deprecated syntax? |
07:37.16 | *** join/#asterisk Pilosopas (n=103392DD@clt-84-32-39-71.ktv.lt) |
07:38.01 | sevard | voltagex: asterisk will tell you if you're using a depressing syntax. |
07:38.11 | voltagex | :/ |
07:38.27 | *** join/#asterisk tsurko (n=tsurko@77.70.24.142) |
07:38.55 | Mahmoud | 81 USD for sipura 3000, what do you say? heap? normal? expensive? |
07:39.09 | JT | ok |
07:39.32 | JT | not brilliant, but it's not that awful i guess |
07:39.42 | Mahmoud | JT, talking to me? |
07:39.46 | JT | yes |
07:39.58 | Mahmoud | so, 81 USD is normal price? |
07:40.03 | flenders | I bought mine for 120AUD |
07:40.19 | JT | they probably cost less than that in the US, considering they're end of line |
07:40.29 | Mahmoud | flenders, i see |
07:40.41 | flenders | I thought they would be like 50 bucks over there. |
07:40.57 | Mahmoud | FlatFoot, aus? |
07:41.10 | JT | Mahmoud: who cares, you've bought it already |
07:41.16 | Mahmoud | i didn't |
07:41.18 | JT | why the need to validate your purchase so badly? |
07:41.22 | Mahmoud | i just found a shop selling it at 81USD |
07:41.26 | JT | i see |
07:41.42 | voltagex | flenders: whoa, best price I can see is $120AUD |
07:41.44 | voltagex | oops |
07:41.45 | voltagex | 130 |
07:41.54 | JT | either buy it or don't, i'm about sick of seeing "81USD" now :P |
07:42.11 | Mahmoud | i'm sick of seeing your nick |
07:42.14 | flenders | hahaha |
07:42.27 | Pilosopas | can anyone help me with zaptel problem? |
07:42.31 | Mahmoud | 81USD o/~ |
07:42.36 | JT | Mahmoud: your inane questions are what make people decide to take a break from #asterisk |
07:42.46 | JT | ever notice everyone shuts up when you say stuff now? |
07:42.58 | Mahmoud | what question? |
07:43.12 | Mahmoud | bring one inane questoin that i asked here dude |
07:43.13 | JT | your continuous repetative questions, beating dead horses :) |
07:43.23 | Mahmoud | bring _one_ example |
07:43.30 | voltagex | Sipura SPA-3000 FXO/FXS VoIP Gateway SPA-3000 AU $1.00, $6.00 postage on eBay |
07:43.39 | JT | IS 81USD GOOD PRICE |
07:43.42 | Mahmoud | lol 6 USD |
07:43.48 | JT | I THINK SIP CAN WORK THROUGH HTTP PROXY |
07:43.54 | voltagex | Mahmoud: is 81US okis 81US okis 81US okis 81US okis 81US okis 81US okis 81US okis 81US okis 81US okis 81US okis 81US okis 81US okis 81US okis 81US okis 81US okis 81US okis 81US okis 81US okis 81US okis 81US okis 81US okis 81US okis 81US okis 81US okis 81US okis 81US okis 81US okis 81US okis 81US okis 81US okis 81US okis 81US okis 81US okis 81US okis 81US okis 81US okis 81US okis 81US ok |
07:44.00 | Mahmoud | JT, yeah dude, it can work |
07:44.17 | JT | Mahmoud: yes, ok, prove it, whatever, it's been discussed plenty already |
07:44.19 | Mahmoud | it needs some heavy modification to the client + server |
07:44.20 | JT | :) |
07:44.20 | voltagex | no, really, is 81US ok? |
07:44.31 | JT | voltagex: depends where you are |
07:44.34 | Mahmoud | JT, study how TCP works, and how to feed it with RTP data |
07:44.38 | voltagex | really really really really really |
07:44.44 | JT | voltagex: i'd buy the SPA-3102 if possible :) |
07:44.58 | voltagex | *Mahmoud bounces up and down like a fucking jack russel* |
07:45.01 | flenders | but not for 81USD |
07:45.23 | Strom_C | hey guys, you'll never believe this, but |
07:45.25 | Strom_C | 81USD |
07:45.37 | voltagex | /kick Strom_C |
07:45.49 | pif | is 81 USD too cheap ? |
07:45.50 | voltagex | /obliterate Mahmoud |
07:46.01 | Pilosopas | can anyone help me with zaptel problem? I get phantom rings every 2-5 minutes on my CO line (crapy line noise) :( Can zaptel deiver after registering incoming call doublecheck if it is ringing after a period of time and only then pass it to asterisk? |
07:46.02 | pif | please please please tell me |
07:46.08 | voltagex | /youtoo pif |
07:46.24 | voltagex | there needs to be a /obliterate command |
07:46.35 | Mahmoud | idiot |
07:46.47 | voltagex | and proud of it |
07:46.53 | Mahmoud | that's the issue |
07:46.54 | pif | terrorist |
07:46.56 | voltagex | I can play the troll game too. |
07:46.58 | Mahmoud | you are also a praud tax payer |
07:47.07 | voltagex | whoa, nasty Pif |
07:47.28 | Pilosopas | :/ |
07:47.53 | voltagex | Pilosopas: I'm sure someone can answer after this flamewar is done. |
07:48.07 | Pilosopas | the world's idiocy has no boundries :( |
07:48.17 | Pilosopas | I know |
07:48.39 | Pilosopas | got my head full with this problem for 3 days |
07:49.09 | Mahmoud | channel operators need to be more strik against off-topic discussions, just like ##c =] |
07:49.39 | voltagex | Mahmoud: 80US is offtopic |
07:49.54 | Strom_C | voltagex: shut up already plzkthx |
07:50.01 | Mahmoud | not really, its aobut voip equipments anyway |
07:50.10 | *** part/#asterisk voltagex (n=voltagex@124-254-123-181-dsl.ispone.net.au) |
07:50.57 | dc3aes | anyone having a problem with nufone, 2 of us here cant dial out |
07:51.51 | Pilosopas | voltagex: who can answer my question? |
07:52.20 | Pilosopas | I believe I'm newer here then you.... |
07:52.46 | Strom_C | Pilosopas: hmm |
07:53.05 | Strom_C | you have bad enough line noise that it causes the phones to ring? |
07:53.11 | Pilosopas | yes |
07:53.30 | Pilosopas | I get random line spikes every 2-5 minutes |
07:53.36 | Strom_C | ... |
07:53.38 | Strom_C | ouch |
07:54.02 | Pilosopas | shielding is bad enough somewhere... |
07:54.29 | Strom_C | do you currently have usecallerid=yes? |
07:54.36 | Pilosopas | yes |
07:54.39 | Strom_C | hm |
07:55.01 | tengulre | how to building the asterisk cluster? |
07:55.45 | Pilosopas | :/ what does RING_DEBOUNCE_TIME do? |
07:56.39 | *** join/#asterisk s-ndh-c (n=michi@85.93.11.18) |
07:57.47 | *** join/#asterisk awk (i=nobody@dsl-242-92-152.telkomadsl.co.za) |
07:57.51 | awk | ~pb |
07:58.02 | jbot | rumour has it, pb is a place to paste your stuff without flooding the channel - try http://pastebin.ca, or http://channels.debian.net/paste |
07:58.02 | awk | hmm, what can I use for pastebin |
07:59.03 | awk | http://channels.debian.net/paste/5820 |
07:59.19 | awk | please can somebody look at that paste, and on a default install of version 1.4 I get all those warning messages |
07:59.49 | Pilosopas | so.... No ideas? anyone? |
07:59.59 | Strom_C | awk: don't worry about it |
08:00.07 | awk | Strom_C: I dont want to see those warnings |
08:00.18 | Strom_C | Pilosopas: try again in eight to twelve hours |
08:00.23 | Strom_C | awk: boo hoo |
08:00.26 | awk | I want a clean startup |
08:00.37 | Strom_C | is this an upgrade from a previous install? |
08:00.40 | Pilosopas | yup, just like windows |
08:00.43 | awk | Strom_C: what are they related to |
08:00.44 | Strom_C | or is this on a totally fresh machine? |
08:00.45 | Pilosopas | no |
08:00.47 | Pilosopas | not upgrade |
08:00.53 | Pilosopas | clean gentoo install |
08:00.53 | Strom_C | i was asking awk |
08:01.05 | Pilosopas | ups |
08:01.08 | Strom_C | so let me restate |
08:01.12 | Strom_C | awk: is this a clean install? |
08:01.15 | awk | well it was an upgrade, but I removed the modules directory and /etc/asterisk/* |
08:01.24 | awk | so its kind of clean |
08:01.43 | Strom_C | /usr/lib/asterisk/modules? |
08:01.50 | awk | yes |
08:03.03 | Strom_C | well, it's not going to cause you any trouble |
08:03.10 | Strom_C | so live with it, or don't watch asterisk start |
08:03.53 | awk | so you not sure how to get rid of it |
08:04.20 | Strom_C | no, I'm not, but it's not a warning which is going to actually affect anything |
08:04.32 | awk | i still dont like seeing it |
08:04.42 | awk | and its a warning because things are not working 100^% |
08:04.54 | awk | nevr mind, ill fix it, just thought somebody had come across it |
08:05.37 | Strom_C | if you do find the cause, please let us know what it is |
08:06.59 | awk | sure |
08:07.17 | s-ndh-c | hey guys |
08:07.45 | awk | busy going through /usr/src/asterisk-1.4.2/main/translate.c |
08:09.55 | s-ndh-c | i have a my asterisk working with one softphone and a voip provider and would now like to connect asterisk to out existing pbx, i would like to be able to make outgoing calls from the softphone on asterisk via out existing pbx`s t1 interface and have the phones connected to the existing pbx make voip calls via asterisk. what are my options to connect both? |
08:10.18 | *** join/#asterisk rocket007 (n=saim@ner-as14402.alshamil.net.ae) |
08:10.41 | Strom_C | s-ndh-c: do you have an extra T1 port on the PBX? |
08:11.04 | s-ndh-c | Strom_C: no idea, will have to look up that |
08:11.57 | Strom_C | i'm not completely sure I understand your situation though |
08:12.03 | *** part/#asterisk rocket007 (n=saim@ner-as14402.alshamil.net.ae) |
08:12.04 | *** join/#asterisk jubei_ (n=jubei_@adsl46-40static.access.altecnet.gr) |
08:12.12 | Strom_C | you want asterisk and the existing pbx to both be able to place outbound calls over the T1? |
08:12.40 | s-ndh-c | yeah and have both make outbound calls via voip if thats possible |
08:12.48 | Strom_C | sure |
08:12.50 | Strom_C | easy enough |
08:13.01 | *** join/#asterisk jm|laptop (n=jm@sentry.flags.co.uk) |
08:13.07 | Strom_C | get a dual-span T1 card, and put asterisk on the T1 between the telco and the existing pbx |
08:13.57 | s-ndh-c | hehe, but if the asterisk box dies the existing pbx is dead too right? |
08:14.29 | Strom_C | yes |
08:14.40 | Strom_C | but usually that's not a problem if you get a decent server-grade machine |
08:15.05 | Strom_C | you'd be an idiot to run this on some frankenbox you cobbled together out of parts from the basement |
08:16.30 | jubei_ | guys what's the best kernel to build for a 1.4.2 system? |
08:16.30 | s-ndh-c | Strom_C: i see, its just some p4 desktop machine atm i cant do something like that i guess |
08:16.34 | s-ndh-c | :) |
08:17.00 | brian | HAY GUISE |
08:17.16 | Strom_C | s-ndh-c: well, i run my own personal box on a piece of shit machine i got for free, but i wouldnt run a business on it :) |
08:17.22 | s-ndh-c | so easiest way is have the asterisk conencted to the telco t1 line and connect the exeting pbx to the second port on the dualspan card |
08:17.33 | Strom_C | yeah |
08:17.46 | s-ndh-c | ok will talk to my boss about that later |
08:22.04 | kupsi | hi |
08:27.22 | *** join/#asterisk psk (n=psk@golia.caltanet.it) |
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08:30.28 | *** join/#asterisk oej (n=olle@66.152.241.83.in-addr.dgcsystems.net) |
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08:42.48 | *** join/#asterisk voltagex (n=voltagex@124-254-123-181-dsl.ispone.net.au) |
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08:43.04 | voltagex | I need some help with something that doesn't seem to be in TFM - Auto fallthrough, channel 'SIP/phone1-081ddeb8' status is 'CONGESTION' |
08:44.24 | *** join/#asterisk Dibbler_XP (n=Dibbler_@host217-45-198-229.in-addr.btopenworld.com) |
08:45.58 | *** join/#asterisk svenna_ (n=svenna@p548d1ac0.dip0.t-ipconnect.de) |
08:47.06 | *** join/#asterisk zeedo (n=zeedo@reboot-robot.net) |
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09:00.34 | *** join/#asterisk zeedo (n=zeedo@reboot-robot.net) |
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09:03.55 | kupsi | hello guys, i'm currently choosing between Sangoma's a-104 and a-104d... the a-104d has echo cancellation but it is $1000 more expensive... is it really worth to have this echo cancellation? afaik, most IP phones comes with "echo cancellation"... please help... tia |
09:05.18 | JT | if you want to be sure, then yes, it is worth it |
09:05.30 | JT | phones don't echo cancel far end echo as a rulee |
09:05.33 | JT | -e |
09:06.31 | voltagex | JT: having problems here without my shiny GUI ;) |
09:07.12 | kupsi | JT: should i go for it? |
09:09.16 | JT | kupsi: yes, if you can afford a quad port card, you should go EC |
09:10.30 | kupsi | thanks... i'm now doing the project costing |
09:15.12 | voltagex | JT: getting circuit-busy, what should I check |
09:15.25 | voltagex | NOTICE[22240]: chan_sip.c:11831 handle_response_invite: Failed to authenticate on INVITE to '"Adam" <sip:0@fwd.pulver.com>;tag=as6219ab19' |
09:21.10 | *** join/#asterisk Turt|e (n=danny@0x55532532.adsl.cybercity.dk) |
09:22.31 | *** join/#asterisk gr1ncheux (n=devine@unaffiliated/gr1ncheux) |
09:29.58 | demlak | hmm.. kannt find login and password for the trixbox 2.0 vmware image.. anyone got them? =) |
09:30.16 | Turt|e | Hi, I have an linksys spa941 behind nat, and i cant get i to connect to asterisk, in asterisk i have out in the nat=yes for the client, the asterisk server is on a global ip and not behind a nat, but there is an pix in between. My phone is setup with nat enabled and my router is setup with port forwared to my phone: 5060-5065, 10000-20000 and 5461. What could posible be wrong here ? |
09:35.28 | voltagex | I'm getting DNS errors, this host resolves on the asterisk box though |
09:35.36 | voltagex | e.g. [Mar 26 19:35:23] WARNING[22400]: chan_sip.c:7242 transmit_register: Probably a DNS error for registration to 762697@fwd.pulver.com, trying REGISTER again (after 20 seconds) |
09:40.42 | *** join/#asterisk [Airwolf] (n=airwolf@voip.ymav.nl) |
09:46.54 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
09:50.17 | *** join/#asterisk frenzy (n=frenzy@unaffiliated/frenzy) |
09:50.38 | frenzy | is there a way to fake DNID on incoming IAX calls? |
09:51.15 | awk | ofcourse, you can ever spoof you outgoing calls to say its coming from any 10 digit number u want |
09:51.20 | awk | ever/even |
09:51.41 | frenzy | I know I can set caller id |
09:51.58 | frenzy | but i'm looking at setting dnid for inboud |
09:52.17 | frenzy | what is the command/code for it? |
09:52.51 | awk | why would you want to spoof the dnid ? |
09:53.15 | frenzy | for filtering |
09:53.34 | awk | so you want to filter certain did's to not route? |
09:53.38 | awk | and route certain did's ? |
09:53.42 | frenzy | i am able to do it via SIP |
09:53.48 | frenzy | actually more like billing filtering |
09:53.58 | voltagex | [Mar 26 19:53:18] NOTICE[22501]: chan_sip.c:11831 handle_response_invite: Failed to authenticate on INVITE to '"Adam" <sip:0@fwd.pulver.com>;tag=as2152124a' << why is this? I'm getting registered ok. |
09:54.08 | frenzy | between toll-frees and local dids |
09:54.50 | *** join/#asterisk tr2x (n=alvar@80-218-185-55.dclient.hispeed.ch) |
09:55.59 | awk | frenzy: sorry things have just gone pete tong here |
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09:56.05 | awk | *gone* |
09:56.40 | giasai68 | hello i get this error can you help me to understand howi can fix it? |
09:56.55 | giasai68 | <PROTECTED> |
09:56.55 | giasai68 | [Mar 26 11:54:18] WARNING[2911]: app_macro.c:174 _macro_exec: Context 'macro-stdexten' for macro 'stdexten' lacks 's' extension, priority 1 |
09:56.55 | giasai68 | <PROTECTED> |
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10:01.36 | nothinman | Hello (-: |
10:04.58 | nothinman | guys, how can I get just the first character from a variable? (let's say ${ABC}) |
10:05.13 | nothinman | so when it's "blah" I will get "b" |
10:06.57 | Strom_C | nothinman: ${ABC:0:1} |
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10:11.43 | awk | Strom_C: 1.4 is it ready for production ? |
10:11.59 | Strom_C | sorta |
10:12.10 | Strom_C | i'd still use 1.2 if you aren't feeling adventurous |
10:12.28 | awk | any idea when comfort noise will be included in 1.4 ? |
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10:12.50 | awk | errr silence suppression |
10:14.49 | giasai68 | hello i get this error can you help me to understand howi can fix it? |
10:15.03 | giasai68 | Executing [832113137@default:1] Macro("SIP/832113137-08208838", "stdexten|832113137|SIP/832113137&Zap/1") in new stack |
10:15.03 | giasai68 | [Mar 26 12:12:12] WARNING[3402]: app_macro.c:174 _macro_exec: Context 'macro-stdexten' for macro 'stdexten' lacks 's' extension, priority 1 |
10:15.03 | giasai68 | <PROTECTED> |
10:16.57 | tzafrir | giasai68, Context 'macro-stdexten' for macro 'stdexten' lacks 's' extension, priority 1 |
10:17.12 | Strom_C | no, reading the error message would be too easy |
10:17.13 | tzafrir | show dialplan macro-stdexten |
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10:17.50 | zeeesh | hi |
10:17.56 | tzafrir | (and please don't paste it here in the channel) |
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10:21.22 | giasai68 | ok thank you |
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10:28.19 | awk | somebody should stick pasting in the topic |
10:28.19 | awk | or have a services join message to state to use a pastebin |
10:28.31 | Strom_C | awk: that wouldnt help |
10:28.56 | Strom_C | trixbox is in the topic, yet every day we get people in here that say "HI OMG I ARE TEH NEETING HALP WITH TRIXBOX" |
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10:31.07 | sbingner | omg nobody said anything that wasn't in the error msg.... lol |
10:34.16 | awk | Strom_C: heh |
10:34.57 | Ch0Hag | Is anyone here who is familiar with the Asterisk source code? |
10:35.08 | Ch0Hag | Especially chan_sip.c |
10:38.06 | awk | Ch0Hag: hmf, i've actually only started really looking at it today |
10:38.25 | awk | busy playing with translate.c |
10:38.49 | Ch0Hag | Oh. |
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10:39.27 | Ch0Hag | Well I want some way to put a string in a variable in the dialplan and then have that string used, if set, in inviteprep() |
10:39.33 | Ch0Hag | I guess I will keep poking around. |
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10:54.27 | Vec | How do I view all zap channels, if I do a zap show channels, it seems to only show me incoming calls on zap ? |
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10:55.20 | Ch0Hag | 'zap show channel <n>'? |
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10:57.19 | awk | hmm |
10:57.21 | awk | 5004:5082 |
10:57.28 | awk | could SIP use this entire range? |
10:58.01 | Ch0Hag | Surely it can use any port you like? |
10:58.27 | awk | i'm talking about default |
10:58.30 | Ch0Hag | I can immediately see peers with ports as low as 1024 or as high as 5060 |
10:58.32 | awk | i see some f/w rules on voip-info |
10:58.48 | awk | hmf! |
10:58.56 | awk | that doesn't make sense |
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11:03.31 | Ch0Hag | What doesn't? |
11:03.47 | awk | well, it cant just decide to use what it wants |
11:03.51 | Ch0Hag | Phone registers with *, says it's listening on port X. |
11:03.51 | awk | you open a set of rules |
11:03.58 | awk | most poeple just have 5060 open |
11:04.12 | awk | so how could it use port 1024 - 5060 |
11:04.22 | Ch0Hag | There's an option to make * listen to a port (or ports?) other than 5060. |
11:04.37 | Ch0Hag | Not sure if it can do a range or not. |
11:04.43 | Ch0Hag | Unlikely. |
11:04.57 | awk | 12:58 < Ch0Hag> I can immediately see peers with ports as low as 1024 or as high as 5060 |
11:05.20 | Ch0Hag | That's the phone. |
11:05.26 | Ch0Hag | Not *. |
11:05.49 | Vec | I just had my asterisk box get stuck, on -- Executing [s@macro-dialBasicExten:3] Dial("Zap/0-2", "SIP/2128|20") in new stack |
11:05.49 | Vec | <PROTECTED> |
11:05.55 | Vec | oops |
11:06.25 | Vec | After a number was dialed it would say Called 2128, but then it did not ring and then after a few seconds it would go engaged |
11:06.26 | awk | Ch0Hag: so for your phones to work what range did you need open |
11:06.39 | Ch0Hag | The phones only conect to port 5060. |
11:07.10 | awk | then? |
11:07.23 | Ch0Hag | Then they tell * what port they are listening on. |
11:07.43 | Ch0Hag | Whether that's explicit or part of IP I don't know. |
11:07.45 | Ch0Hag | Probably the latter. |
11:07.46 | awk | ok, but what happens if everything is closed, except your rtp range |
11:07.57 | awk | and sip for registration |
11:08.07 | awk | how would it comunicate on port 1024 as you said earlier |
11:08.17 | Ch0Hag | If the firewall was blocking it, it wouldn't. |
11:08.46 | awk | so thats what im getting at |
11:08.58 | awk | what solution would you do, to not allow such a large range open |
11:09.14 | Ch0Hag | Tell the phone somehow what port range it can listen on. |
11:09.26 | Assid | awk: set your rtp range on your own |
11:09.29 | awk | eg: 5060 only |
11:09.36 | awk | Assid: hey ? |
11:09.41 | awk | i have my rtp range set |
11:09.48 | Assid | on the server? |
11:09.53 | awk | yes |
11:09.59 | awk | 10000 10100 |
11:10.02 | Assid | okay so.. set the same on yuor firewall |
11:10.13 | awk | i dont have f/w issues |
11:10.19 | awk | im trying to understand what Ch0Hag was saying |
11:10.28 | Assid | oh ok |
11:10.34 | Assid | sorry.. me goes away again |
11:11.33 | Ch0Hag | I don't know whether the phones are listening on (eg.) 1024 for SIP *and* RTP. |
11:11.56 | Ch0Hag | I have no need to restrict outgoing traffic through the firewall at all so it has yet to be a problem. |
11:11.58 | awk | oh, I see |
11:12.31 | Ch0Hag | I think that 10000-10100 setting is for * listening for RTP. |
11:12.39 | Ch0Hag | Not for a phone listening. |
11:13.04 | Ch0Hag | On this Snom that is set with 'Dynamic RTP port start/stop:' to 49152-65534 |
11:13.07 | Ch0Hag | For whatever reason. |
11:13.23 | awk | oh, intresting |
11:13.40 | Ch0Hag | Ah 49152 == 0xC000 |
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11:15.19 | awk | maybe 1 of you can help. snmp |
11:15.48 | awk | does mrtg use udp 161 or tcp 199 |
11:15.56 | awk | to query your snmp daemon ? |
11:16.01 | Ch0Hag | Dont know snmp. |
11:22.21 | nothinman | Strom_C: thank you. it does seem to work :-) |
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11:33.24 | AdamCrowe | Hello |
11:33.43 | AdamCrowe | Does anybody have any experience with ISDN cards? |
11:34.38 | tzafrir | some people have some experince some of the time |
11:35.40 | AdamCrowe | I have a B410P which I managed to install and I can do incoming and outgoing calls with... |
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11:35.48 | mkl1525 | Hi, does anybody know how to write sms/text on snom displays? tried sipsak but get a "destination unreachable" although I don't have problems to access the web interface. |
11:36.29 | AdamCrowe | BUT - the L2Link keeps going down, and unless an incoming calls "waked it up" or I manually type "misdn port up 1", I cannot make any outgoing calls on that port. |
11:36.36 | Ch0Hag | mkl1525: http://www.snom.com/wiki/index.php/Xmlobjects |
11:37.53 | mkl1525 | Ch0Hag, thanks for the hint, but we've got snom300 too that don't have the xml browser so I think I'll need some simple text message |
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11:38.24 | Ch0Hag | I don't know about that. |
11:38.34 | Ch0Hag | Probably just replace it with a newer one :) |
11:39.01 | Ch0Hag | What are ANI, DNID and RDNIS? |
11:39.40 | zeeesh | i registered 2 peers 100 and 200 .. at my asterisk server .. 100 can make call at 200 but 200 unable to make call 100 ... extensions are (from 100 exten => 200,1,Dial(SIP/200) ) (from 200 exten => 100,1,Dial(SIP/100) ERROR IS .. "5 NOTICE[3633]: app_dial.c:1056 dial_exec_full: Unable to create channel of type 'SIP' (cause 3 - No route to destination) |
11:39.41 | zeeesh | <PROTECTED> |
11:39.41 | zeeesh | <PROTECTED> |
11:40.20 | Ch0Hag | 100 may need to reregister. |
11:40.37 | zeeesh | 100 is registered .. |
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11:43.14 | AdamCrowe | Anybody??? |
11:44.37 | Ch0Hag | Not me. I know next to nothing about ISDN. |
11:46.50 | zeeesh | NEHLEEEOOOOOOO ... |
11:47.55 | Ch0Hag | I think nobody is really here. |
11:48.20 | Ch0Hag | Except me, and I'm by no means an Asterisk guru. |
12:41.30 | *** join/#asterisk jbot (i=ibot@pdpc/supporter/active/TimRiker/bot/apt) |
12:41.30 | *** topic/#asterisk is Asterisk: The Open Source PBX -=- Asterisk 1.4.2 (Mar. 19, 2007), Asterisk 1.2.17 (Mar. 19, 2007), Zaptel 1.2.16 (Mar. 19, 2007), Zaptel 1.4.1 (Mar. 23, 2007) -=- Other fun channels: #asterisk-gui, #asterisknow, #asterisk-commits -=- Join #freepbx for freepbx/trixbox support. |
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12:44.58 | m-00kie | hello. |
12:45.22 | m-00kie | does asterisk's oh323 support conferencing? |
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12:56.37 | puzzled | m-00kie: why not? just send the call coming in via h323 to a meetme |
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13:02.06 | zaffa | Anyone had problems with zaptel 1.2.14? I Have strange problem with te4xxp - once or twice a month PRI line 'locks' that no calls can be made nor received. |
13:02.37 | zaffa | 'No D-channels available! Using...' and then 'Unable to create channel of type 'ZAP' (cause 34 - Circuit/channel congestion)'. zap show status shows no alarms. |
13:02.44 | necromcr | hi, i'm having some "goto" problems (i cant seem to make it work to dial 0 and then match following numbers in other context using misdn line). can please anyone take a moment? |
13:02.57 | zaffa | reloading zaptel modules and asterisk solves the problem. Any ideas what can be the reason? |
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13:11.59 | unice | hi all |
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13:12.34 | angryuser | i have some problems with my isdn ports, any idea what is it? http://www.pastebin.ca/403464 asterisk 1.41 |
13:12.47 | angryuser | i am using B410P misdn |
13:13.48 | angryuser | they are configured in TE mode now, but still same problems |
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13:27.38 | mkl1525 | Hi, it's possible to change the idle screen of snoms but is there also a way to change the screen when I'm talking to somebody? |
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13:33.40 | f_akmal | hi all, i have problems compiling app_voicechangedial.c, can anyone help? |
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13:36.58 | Katty | mew. |
13:37.54 | Ch0Hag | Where is struct varshead defined? |
13:38.21 | Ch0Hag | I can't find it anywhere, except in use. |
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13:48.01 | Ch0Hag | Well that doesn't seem to be what I want anyway, so if I Set() a variable in the dialplan, how can I access it in the source? |
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13:51.47 | sashion | Ch0Hag, I believe it might be under pbx.c |
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14:00.21 | achu | can we redirect asterisk's /var/log/full to stdout ? |
14:00.38 | Gido-E | why not? |
14:01.08 | achu | can you pls tell me which file I have edit ? |
14:04.03 | necromcr | achu: logger.conf? |
14:05.32 | achu | its onlys shows full => notice,warning,error,debug,verbose , can I remove all that option and add /dev/stdout ? |
14:05.49 | achu | sorry if I am wrong |
14:05.50 | necromcr | stdout? |
14:05.56 | necromcr | why would you want to do that? |
14:05.57 | e-ddie | achu: can you please read the manual? |
14:06.07 | necromcr | log to a file and do tail -F <that file> |
14:07.37 | e-ddie | sounds like a better idea would be for him to reinstallwindows |
14:08.50 | achu | necromcr : with my small knowledge I understand that the full message specified in logger.conf goes to /var/log/asterisk/full , I would like to use it for daemontools for multilog |
14:09.06 | achu | when I configured it is not rotating logs |
14:09.32 | achu | so I want to redirect the asterisk full log to /dev/stdout |
14:10.50 | necromcr | achu: i'm not quite familliar with multilog but.. cant you use metalog? it's got filtering options as well |
14:11.06 | achu | k |
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14:11.33 | necromcr | awsome.. i just made infinite loop in my context :P |
14:12.36 | anonymouz666 | infinite loop? |
14:12.39 | anonymouz666 | welcome to the club |
14:12.41 | anonymouz666 | lol |
14:12.54 | Dr-Linux | cisco 7920 is a WIFI phone? |
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14:13.40 | tdonahue-laptop | anyone have a link to the changelog for 1.4.2 |
14:13.44 | tdonahue-laptop | ? |
14:14.57 | achu | necromcr : When I enabled syslog.local0 in logger.conf it starts showing the messages in /var/log/messages |
14:14.58 | heison | i'd like to get Asterisk 1.4.2 via svn, and i'm unsure if this will actually get me 1.4.2... svn checkout http://svn.digium.com/svn/asterisk/tags/1.4.2 asterisk how is that different from http://svn.digium.com/svn/asterisk/trunk ? if i'm getting the latest release, should they be the same? |
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14:18.19 | necromcr | achu: you need to log it into some other file? |
14:18.27 | russellb | heison: that is correct for getting 1.4.2 from svn |
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14:18.38 | russellb | heart: trunk is the development tree, where new things are getting added and broken |
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14:19.08 | AdamCrowe | Anybody here with ISDN experience? |
14:19.44 | heison | russellb: thx! |
14:19.58 | coppice | I developed the world's first ISDN mux. does that count? :-) |
14:20.04 | heison | does anyone know if rx_fax and tx_fax are working with 1.4.2? |
14:20.33 | sashion | what would cause asterisk to send a CANCEL to a phone after 1 ring? |
14:20.54 | necromcr | damn.. one would really need a documentation in a style of chm or javadoc :-| |
14:21.18 | AdamCrowe | I'm from the UK... Layer 2 keeps going down (as it suppose to do in most Europe), but while it's down, I cannot make outgoing calls unless I manually take it up via "misdn port up 1" or through an incoming call. |
14:21.37 | russellb | coppice: oh yeah? well I had the de-mux before you had the mux. |
14:21.50 | coppice | probably not |
14:21.58 | achu | necromcr : I don't want asterisk to right it's logs to file |
14:22.34 | AdamCrowe | How can I keep layer 2 up or at least make it go up when making an outgoing call? |
14:22.47 | Ch0Hag | Does anyone know the * source? |
14:22.55 | necromcr | achu: eee.. |
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14:23.05 | coppice | russellb: i suspect you might have been a bit young at that time |
14:23.10 | Ch0Hag | Because either I don't get something or a huge chunk is broken and I'd like to work out which it is. |
14:23.17 | russellb | coppice: probably :) |
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14:25.11 | AdamCrowe | coppice: any ideas? |
14:25.11 | coppice | I think that must have been about 1986 |
14:25.12 | coppice | AdamCrowe: I said I *developed* the first mux. I didn't say I ever used one :-\ |
14:25.12 | russellb | ha .. |
14:25.12 | russellb | coppice: yeah, i was only 2 years old. |
14:25.12 | necromcr | achu: sorry, too much for me |
14:25.13 | AdamCrowe | O_o |
14:25.46 | achu | necromcr : ok , I am sorry |
14:26.51 | necromcr | what's the idle policy on this channel? |
14:27.00 | russellb | idling is encouraged |
14:27.08 | russellb | is that a policy? |
14:27.27 | adam_vollrath | There probably isn't a policy, per se. |
14:27.47 | file | anyone who fails to idle may be randomly kicked ... |
14:27.50 | file | (not really) |
14:28.12 | russellb | well, talking does encourage your chances of getting kicked |
14:28.47 | russellb | nooo |
14:28.49 | adam_vollrath | Anyone who idles will not be kicked. I think that's certain. |
14:28.59 | russellb | unless your nick alone is trolling |
14:29.10 | russellb | like ... /nick asterisk_is_so_laME |
14:29.11 | adam_vollrath | I think the bar is pretty high for that. |
14:29.35 | russellb | pwned by the length limit |
14:30.16 | *** kick/#asterisk [asterisk_is_lame!i=russell@asterisk/developer-and-stable-maintainer/drumkilla] by russellb (russellb) |
14:30.38 | russellb | i have teh power! |
14:31.00 | necromcr | ok.. idle it is |
14:31.25 | *** part/#asterisk achu (n=achu@124.125.38.238) |
14:31.55 | *** kick/#asterisk [russellb!n=file@asterisk/developer-and-muffin-lover/file] by file (file) |
14:31.57 | file | !!! |
14:32.05 | *** join/#asterisk russellb (i=russell@asterisk/developer-and-stable-maintainer/drumkilla) |
14:32.05 | *** mode/#asterisk [+o russellb] by ChanServ |
14:32.16 | russellb | er, not Fieldy ! |
14:32.17 | russellb | file. |
14:32.25 | russellb | stupid tab completion :( |
14:33.42 | Ch0Hag | voip-info says 'the variable ${VXML_URL} can be used to add additional items to the To: header'. This is done bi setting p->options->vxml_url in a AST_LIST_TRAVERSE loop, however that loop is never entered because there are no variables to loop over. |
14:36.33 | *** join/#asterisk infernix (i=nix@unaffiliated/infernix) |
14:39.51 | putzz | 9000000000000=[-p999990-\=-[][ |
14:39.52 | putzz | =;='] |
14:39.57 | putzz | \--0p-;[' |
14:54.04 | *** join/#asterisk adam_vollrath (n=adam@pc152.BISonline.com) |
14:56.27 | *** join/#asterisk Mercestes (n=Merceste@rrcs-71-41-157-70.sw.biz.rr.com) |
14:59.18 | Bladerunner05 | using gui in the installation procedure (browser based) in the Local Extension Settings the next button is disabled, could be a permission problem in /etc/asterisk directory ? |
14:59.53 | Mercestes | Bladerunner05, Could definately be a #freepbx or #asterisknow question. |
15:00.11 | Mercestes | we don't do GUI here. We're text only. |
15:01.29 | *** join/#asterisk qdk (n=qdk@0x50c627be.bynxx11.adsl-dhcp.tele.dk) |
15:02.17 | necromcr | yup, text only (CTCP VERSION reply from necromcr: irssi v0.8.10 - running on Linux i686) |
15:04.59 | Mercestes | necromcr, are you a bot? |
15:05.34 | Mercestes | We're thinking necromcr is a bot. |
15:07.44 | Mercestes | necromcr, what is your prime directive? |
15:09.16 | adam_vollrath | About that GUI thing. We're using a 3Com NBX100 system now, and we like its web-based administration (even if it is ancient), because we change our dial plan frequently. Is there a GUI to ease Asterisk dial plan maintenance? |
15:09.30 | russellb | adam_vollrath: check out asterisknow.org |
15:09.40 | adam_vollrath | ressellb, kthx |
15:10.14 | *** join/#asterisk `p4r14h`work (n=josh@24-119-48-78.cpe.cableone.net) |
15:10.34 | Ch0Hag | Is SipAddHeader() supposed to do absolutely nothing? |
15:10.38 | adam_vollrath | Can I use the AsteriskGUI without using this Asterkisknow Linux distro? We'd planned on using Fedora. |
15:10.48 | Qwell[] | Ch0Hag: clearly it isn't |
15:10.52 | russellb | adam_vollrath: yeah, you can |
15:10.58 | Ch0Hag | Well it does. |
15:10.58 | russellb | adam_vollrath: it's super easy to install, too |
15:11.01 | Ch0Hag | Or doesn't... |
15:11.07 | russellb | adam_vollrath: the instructions are in the topic of #asterisk-gui |
15:11.18 | Qwell[] | what header are you setting, and do you know that your provider/device supports said header? |
15:12.08 | *** join/#asterisk aptura (n=jondoe@S010600a0c93f6f7e.vs.shawcable.net) |
15:12.18 | russellb | Qwell[]: /me drools at the weather report ... high of 84 |
15:12.47 | russellb | bah |
15:12.51 | Qwell[] | heh |
15:12.53 | Ch0Hag | That is neither here nor there. |
15:12.53 | Ch0Hag | I have tried both SipAddHeader(Bro=ken) and SipAddHeader(Bro: ken), and the headers remain unaffected. |
15:12.57 | Qwell[] | How was SJC? |
15:12.58 | aptura | sounds like LA |
15:12.59 | Qwell[] | weatherwise |
15:13.02 | Ch0Hag | Unless for whatever random reason the headers 'sip debug' shows are not the headers that are actually sent. |
15:13.16 | aptura | Our area has been beaten with tons of snow and rian this year. |
15:13.18 | russellb | Qwell[]: nice ... 60's 70's ... |
15:13.33 | Qwell[] | russellb: we should move Digium there :P |
15:13.49 | Qwell[] | AL summer is gonna suck - I know it |
15:14.43 | aptura | Qwell, the city managers and planners did a poor job of building the city in the first place. Turned it into a huge oven with not enough trees or white roofs to absorb or reflect the heat. |
15:14.55 | Qwell[] | what city? |
15:15.08 | aptura | you are in Alabama right? |
15:15.11 | Qwell[] | yeah |
15:15.14 | russellb | AL summer is going to r0x0r |
15:16.12 | aptura | I read a report it was Birmingham |
15:16.32 | Qwell[] | good thing we aren't in Birmingham then ;) |
15:16.42 | aptura | where are you then |
15:16.46 | Qwell[] | Huntsville |
15:16.55 | aptura | Okay wait it was Atlanata |
15:17.06 | Dr-Linux | cisco 7920 is a WIFI phone? |
15:17.11 | Dr-Linux | Qwell[] :) |
15:17.13 | Qwell[] | Dr-Linux: skinny, yes |
15:17.16 | *** join/#asterisk jcmoore (n=jcmoore@unaffiliated/tgrman) |
15:17.17 | Qwell[] | it isn't great |
15:17.17 | aptura | Here in the summer it is cool |
15:17.43 | aptura | we get the fresh pacific air but on the downside the tons of moisture it brings in the winter. |
15:17.55 | Qwell[] | Seattle? :P |
15:18.04 | Dr-Linux | Qwell[]: will 7920 will work for me, or still i'll have issues like other phone? |
15:18.10 | Qwell[] | Dr-Linux: I don't really recommend it |
15:18.12 | Dr-Linux | Qwell[]: and what protocol should i use? |
15:18.13 | *** join/#asterisk Damin (n=damin@nucleus.nacs.net) |
15:18.19 | Qwell[] | You can only use skinny wth the 7920 |
15:18.36 | Qwell[] | there are better wifi phones |
15:18.44 | Dr-Linux | Qwell[]: you mean the asterisk skinny will work for me? |
15:18.58 | Dr-Linux | Qwell[]: i'd agree but my boss has one, and he asked me to configure it |
15:19.03 | aptura | Qwell, wait till the ice caps melt it regultes the tempature of the atlantic ocean. And with it the eastern seaboard. |
15:19.10 | Dr-Linux | even the phone is in USA and i'm here |
15:19.27 | Qwell[] | Dr-Linux: it'll work with chan_skinny in 1.4 |
15:19.58 | *** join/#asterisk codefreeze (n=steve_mu@216.166.159.235) |
15:20.09 | Dr-Linux | Qwell: what about 1.2.0? |
15:20.12 | Qwell[] | no |
15:20.23 | mafkees | chan_sccp |
15:20.54 | Dr-Linux | mafkees: chan_sccp sucks :@ |
15:21.05 | mafkees | no kidding |
15:21.18 | *** join/#asterisk dasenjo (n=dasenjo@190.24.178.69) |
15:21.30 | Dr-Linux | Qwell[]: the thing is that, i can't use 7935 on production system |
15:21.39 | Dr-Linux | that patch is only for svn version |
15:21.53 | *** join/#asterisk sahafeez (n=sahafeez@67.109.14.227.ptr.us.xo.net) |
15:22.22 | Dr-Linux | Qwell[]: can you please give me the link for that patch again? the for cisco 7935 |
15:22.36 | Qwell[] | I don't know the bug number, you'll have to search for it |
15:22.42 | Qwell[] | look for bugs by "slimey" |
15:22.57 | Dr-Linux | ok |
15:24.03 | *** join/#asterisk __freedom__lover (n=eduardo@clipper.provale.com.br) |
15:24.06 | __freedom__lover | hi all |
15:24.22 | *** join/#asterisk `p4r14h`work (n=josh@24-119-48-78.cpe.cableone.net) |
15:24.40 | __freedom__lover | can anyone tell me what mean blue/yellow/red alarm in zap status? |
15:24.59 | *** join/#asterisk ManxPower (n=manxpowe@71-8-56-64.dhcp.leds.al.charter.com) |
15:31.55 | __freedom__lover | alow?! |
15:32.12 | tzafrir | red: "nothing connected" (or: no layer 1 connection) |
15:32.34 | *** join/#asterisk tuan_modulis (n=chatzill@3-82-252-216-static.enter-net.com) |
15:33.00 | tzafrir | yellow: IIRC it means bad CRC. Not really sure |
15:33.15 | __freedom__lover | hum, thanks |
15:33.16 | tzafrir | As for blue: I don't know. |
15:33.24 | tuan_modulis | hello everyone, im looking for a recommendation for an IAX2 softphone |
15:33.34 | tzafrir | kiax |
15:33.38 | tuan_modulis | gotcha! |
15:33.52 | __freedom__lover | tzafrir: i've looked for that information, but nothing good explanation |
15:35.07 | *** join/#asterisk Slingky (n=na@modemcable199.182-200-24.mc.videotron.ca) |
15:35.18 | ManxPower | __freedom__lover: What issue are you having? |
15:35.30 | Slingky | hi! could somebody tell me how to change sounds ? |
15:35.36 | Slingky | i download french-gsm |
15:35.45 | Slingky | but do i just need to replace a folder or what ? |
15:35.53 | tuan_modulis | make a french folder |
15:35.58 | tuan_modulis | put them all in there |
15:36.01 | tzafrir | Slingky, it is done automatically when you set the language |
15:36.14 | tuan_modulis | when you set the language, set to name of folder |
15:37.14 | Slingky | tzafrir, i think it point currently to /var/lib/asterisk/sounds , no ? |
15:37.29 | tuan_modulis | see thewiki for SetLanguage |
15:37.34 | tuan_modulis | ~thewiki |
15:37.36 | jbot | somebody said thewiki was at http://www.voip-info.org/wiki-Asterisk |
15:38.07 | Slingky | thanks a lot guys! |
15:38.55 | *** join/#asterisk sahafeez (n=sahafeez@67.109.14.227.ptr.us.xo.net) |
15:38.58 | Ch0Hag | I think I have a bug here. |
15:39.24 | Ch0Hag | sip_addheader() is called with struct ast_channel set to one value, then sip_call is called with it set to another. |
15:40.08 | Ch0Hag | Either another ast_channel is erroneously created or one is mis-copied to the other. |
15:40.54 | __freedom__lover | ManxPower: hi, i've configured my te110p card, but it is changing its status, among blue, yellow and red |
15:41.22 | ManxPower | __freedom__lover: That is usually a telco issue. |
15:41.27 | ManxPower | Assuming you have your card configured |
15:41.35 | ManxPower | __freedom__lover: what country are you in? |
15:41.41 | ManxPower | and what is your span= line? |
15:42.59 | *** join/#asterisk _VoicePulse (n=contact@unaffiliated/voicepulse) |
15:44.37 | __freedom__lover | ManxPower: i'm from brazil |
15:44.53 | ManxPower | (10:41:35) ManxPower: and what is your span= line? |
15:45.36 | __freedom__lover | ManxPower: my span is 1 |
15:46.39 | ManxPower | That is wrong. It should be something similar to span=1,1,0,ccs,hdb3,crc4 |
15:46.56 | __freedom__lover | oh yeah |
15:47.09 | __freedom__lover | my span is 'span=1,0,0,cas,hdb |
15:47.14 | ManxPower | What is your EXACT span= line. copy and paste it. |
15:47.14 | __freedom__lover | sorry |
15:47.27 | __freedom__lover | span=1,0,0,cas,hdb3,crc4 |
15:47.36 | *** join/#asterisk bkuhn (n=bkuhn@fsf/member/bkuhn/bkuhn) |
15:47.58 | ManxPower | __freedom__lover: Do you know for sure that your line is cas, hdb3 and crc3? Only your telco can tell you. |
15:48.05 | ManxPower | ..er..crc4 |
15:48.37 | ManxPower | you should have a 1 instead of a 0 as the 2nd field or your timing will be off. |
15:48.59 | *** join/#asterisk Ebola (n=Ebola@host86-136-190-11.range86-136.btcentralplus.com) |
15:49.38 | *** join/#asterisk oQPa (n=uawename@84.Red-83-40-182.dynamicIP.rima-tde.net) |
15:54.30 | Dr-Linux | ManxPower: what you would recommend upgrade to latest 1.2.x? or 1.4.x? |
15:56.40 | aydiosmio | I had no idea the original telephone systems had on clock source (like a cluster of cesium clocks) |
15:56.45 | aydiosmio | one* |
15:57.46 | *** join/#asterisk stkn|work (n=stkn|wor@gentoo/developer/pdpc.active.stkn) |
15:59.03 | Dr-Linux | Qwell[]: what you recommend? |
15:59.42 | *** join/#asterisk Hmmhesays (n=Neg@24-117-131-41.cpe.cableone.net) |
16:01.02 | ManxPower | Dr-Linux: I have never used 1.4 because it has not been out long enough and there seems to be too many outstanding issues. |
16:01.35 | Dr-Linux | i see |
16:01.37 | Mercestes | Dr-Linux: CCM. *runs and hides* |
16:01.50 | ManxPower | aydiosmio: T-1 timing has NOTHING TO DO WITH TIME |
16:02.01 | Ch0Hag | Hence the name. |
16:02.21 | Hmmhesays | ack i'm out of cdrs |
16:02.23 | Dr-Linux | Cisco Call Manger? |
16:02.35 | Mercestes | Dr-Linux, yea, but I'm kidding. I've never used it. |
16:02.41 | aydiosmio | ManxPower: not for tick tock time |
16:02.44 | aydiosmio | but bit timing |
16:03.01 | aydiosmio | http://www.oreilly.com/catalog/t1survival/chapter/ch05.html |
16:03.04 | ManxPower | A more correct term would be "sync" not timing |
16:03.12 | aydiosmio | what I'm reading |
16:03.12 | Dr-Linux | Mercestes: if i'd have it handly i would do something, but i never seen |
16:03.28 | Dr-Linux | handy* |
16:03.32 | Mercestes | Dr-Linux, It's like the opposite of free. |
16:03.58 | *** join/#asterisk deeperror (n=deeperro@mail.banctel.com) |
16:04.00 | Dr-Linux | sorry , but i didn't understand :S |
16:04.21 | Mercestes | Dr-Linux, I was just teasing/trolling. |
16:04.30 | Dr-Linux | lol |
16:04.32 | Dr-Linux | good good |
16:04.33 | Dr-Linux | do it |
16:04.38 | Dr-Linux | i wont mind :D |
16:06.06 | *** join/#asterisk nasls_lsa (n=chatzill@87.203.116.128) |
16:06.12 | deeperror | does anyone know of some good quality cordless sip phones? |
16:07.00 | eald | fun stuff, I have an asterisk box that grows in cup usage by their own, each week has 10% more CPU usage than the previous one, but the machine is rebooted every night and numbers of calls is around the same |
16:07.08 | *** join/#asterisk SwK[Work] (n=SwK@24.214.206.254) |
16:07.52 | *** part/#asterisk oQPa (n=uawename@84.Red-83-40-182.dynamicIP.rima-tde.net) |
16:08.54 | k31th | bit off topic i know but can anyone recomend a decent UK VoIP provider? |
16:09.07 | Ch0Hag | Define 'decent'. |
16:11.17 | *** join/#asterisk musse- (n=kallekas@static-212.214.40.123.addr.tdcsong.se) |
16:12.07 | *** join/#asterisk Gunnar (n=gunnar@nat.sigmasoft.com) |
16:17.15 | ManxPower | All VoIP providers suck |
16:17.38 | mog | heh |
16:18.02 | *** join/#asterisk fakhir (i=fakhir@unaffiliated/fakhir) |
16:20.44 | ManxPower | For some bizarre reason when I added a 4th site to my network OSPF started losing routes. |
16:21.47 | *** join/#asterisk harleya (n=harleya@207.108.166.2) |
16:22.07 | __freedom__lover | ManxPower: thanks |
16:22.13 | *** join/#asterisk koel (n=ParaNoir@84-53-96-51.fiber.unet.nl) |
16:22.33 | koel | Hey, does anybody know which SIP client displays contact status information? (online/offline) |
16:22.45 | *** part/#asterisk harleya (n=harleya@207.108.166.2) |
16:23.19 | ManxPower | koel: what makes you think SIP even supports that feature? |
16:25.58 | koel | :) the wiki says the latest dev of asterisk has a presence feature ;) |
16:26.13 | koel | http://www.voip-info.org/wiki/view/Asterisk+presence |
16:26.39 | Gido-E | koel, asterisk doesn't only support sip. |
16:27.57 | ManxPower | most of those features are for call status. hint is specifically for call status (in use, not in use) |
16:28.06 | ManxPower | not "ping able, not pingable) |
16:29.24 | ManxPower | the only way to really know of a device is reachable is to send data to it. |
16:31.12 | *** join/#asterisk syzygyBSD (n=chatzill@66.226.228.204.cpe.speedyquick.net) |
16:31.22 | necromcr | is there any way to push an extensio on to stack manually? |
16:31.26 | syzygyBSD | hi |
16:31.36 | *** join/#asterisk russellb (i=russellb@asterisk/developer-and-stable-maintainer/drumkilla) |
16:31.36 | *** mode/#asterisk [+o russellb] by ChanServ |
16:31.40 | necromcr | say, i have an extension number in a variable and want to dial it |
16:31.42 | syzygyBSD | necromcr: do you mean through the console? |
16:31.47 | syzygyBSD | oh... |
16:31.56 | syzygyBSD | goto(context,extension,priority) |
16:32.13 | aydiosmio | stack? hehe |
16:32.38 | necromcr | syzygyBSD: no.. um.. |
16:33.01 | syzygyBSD | umm explain yourself better |
16:33.27 | necromcr | exten => _80!,1,Set(ORIG_EXTEN=${EXTEN:2}) exten => _80!,n,Goto(notranji-klic,s,1) |
16:33.33 | necromcr | (trying to :) |
16:34.02 | necromcr | i pick up isdn phone, dial 80 and get notranji-klic. there i handle the next 3 digit extension |
16:34.11 | necromcr | (so full number i dial is say 80123) |
16:34.52 | necromcr | :-\ |
16:35.12 | necromcr | well, if i enter 80123 and then dial.. 80 get's handlede and 123 doesnt |
16:35.16 | necromcr | handled |
16:35.20 | syzygyBSD | oh.. so in the end you want either a exten=>_XXX,1,goto(newcontext,${ORIG_EXTEN}${EXTEN},1)? |
16:35.22 | aydiosmio | why not goto(notranji-klic,${EXTEN},1)? |
16:35.49 | syzygyBSD | another good solution... |
16:36.00 | aydiosmio | otherwise ${EXTEN} in the new context is "s" not "80123" |
16:36.37 | ManxPower | uh, why not just Goto(newcontext,${EXTEN},1) |
16:36.51 | ManxPower | the only time you would have "s" is in macros or if you Goto(s,1) |
16:37.22 | aydiosmio | anyone else want to give him the solution? |
16:37.24 | ManxPower | and in a macro you can do an exten => s,1,Goto(${MACRO_EXTEN},1) and having a matching exten line in the macro |
16:38.45 | necromcr | darn |
16:38.50 | necromcr | i bow to you all |
16:39.47 | necromcr | i must have been really sleepy when i was testing that option as i was sure, goto(..,EXTENSION...) wouldnt work if i had pattern matching extensions.. |
16:39.58 | syzygyBSD | it is hard when you think something is a lot harder than it is... |
16:40.08 | necromcr | syzygyBSD: yea :) |
16:40.11 | penguinFunk | it's easy when you know how |
16:40.25 | penguinFunk | same with anything |
16:40.41 | syzygyBSD | na, some things are never easy |
16:41.24 | penguinFunk | like dealing with girlfriend on pmt |
16:41.27 | penguinFunk | ? |
16:42.16 | syzygyBSD | I was thinking comming up with an answer to "does this make me look (fat|bloated|like a cow)" |
16:42.39 | aydiosmio | the answer is "honey, you look amazing" |
16:42.41 | *** join/#asterisk mmartinn (n=martins@128.227.123.22) |
16:43.02 | penguinFunk | ;) |
16:43.04 | *** join/#asterisk L|NUX (n=linux@unaffiliated/lnux/x-10290) |
16:43.04 | penguinFunk | easy |
16:43.14 | syzygyBSD | hmm, guess I didn't know how... |
16:44.01 | mmartinn | hey hey... question for the gurus... Gentoo wants to give me a new /etc/hotplug/usb/xpp_fxloader.usermap... will this affect Zaptel? |
16:44.59 | syzygyBSD | maybe.. just recompile afterwards... |
16:45.08 | bulle | ManxPower: just save the old file just in case ? |
16:45.09 | ManxPower | mmartinn: do you have an XPP channel bank? |
16:45.15 | mmartinn | ManxPower: Negative |
16:45.25 | ManxPower | mmartinn: then it should not be a problem |
16:45.28 | mmartinn | I suppose I'll save the old one and see what happens :) |
16:45.34 | mmartinn | thanks guys :) |
16:45.46 | necromcr | hotplug? |
16:45.56 | necromcr | isnt' hotplug out of the date with udev? |
16:46.04 | bulle | necromcr: yes and no afaik |
16:46.07 | necromcr | (remove the) |
16:46.15 | bulle | necromcr: the hotplug functinality is now mainly handled by udev |
16:46.26 | mmartinn | I believe it was built in, but the names of some scripts and files are still hotplug |
16:46.34 | bulle | necromcr: but some distros ( dont know about gentoo ) still keep the name as hotplug |
16:46.37 | bulle | ManxPower: ye |
16:46.53 | mmartinn | Yeah... Gentoo does AFAIK; udev now handles it but there's still stuff called it. |
16:46.59 | *** join/#asterisk nextime (n=nextime@unaffiliated/nextime) |
16:47.22 | mmartinn | Thanks guys... I'll be back in later, when not at work 8-) |
16:47.28 | necromcr | good luck mmartinn |
16:47.36 | nextime | hello. In * 1.4.x ther's a "stun" command in cli. Is it a stun support for sip channels? how to can i use it? |
16:47.44 | *** part/#asterisk mmartinn (n=martins@128.227.123.22) |
16:47.46 | *** join/#asterisk Fieldy (i=eKsWJ8KX@gentoo/contributor/Fieldy) |
16:50.01 | *** join/#asterisk shido6 (n=shido6@d221-68-200.commercial.cgocable.net) |
16:50.19 | Mercestes | nextime, yes. You take an FXS module and run two leads off of it, wet someone' skin a little, place the leads on their wet skin, and then call a "ringing" to that interface. viola, stunned |
16:50.47 | aydiosmio | Ladies and gentlemen, set your faces to "Stunned" |
16:51.04 | nextime | "ah. ah. ah." |
16:51.31 | necromcr | rofl :) |
16:51.55 | Mercestes | nextime, If you mean a "stun server" I've only really seen that used in ATAs and I'm not sure how much Asterisk is involved in that functionality. |
16:52.08 | Mercestes | maybe google aterisk wiki stun |
16:52.20 | ManxPower | I've never actually seen a situation where you need STUN |
16:52.22 | Mercestes | s/aterisk/asterisk/ |
16:52.35 | Mercestes | ManxPower, Might explain why I've never used it. lol |
16:52.56 | *** join/#asterisk jm|laptop (n=jm|home@dilbert.jamiem.com) |
16:53.02 | ManxPower | Mercestes: STUN is one of MANY ways to deal with NAT + SIP |
16:53.21 | syzygyBSD | another way is not to use NAT |
16:53.46 | necromcr | syzygyBSD: some are forced to use nat |
16:54.05 | necromcr | syzygyBSD: here in .si umts lines are all nat-ed |
16:54.07 | ManxPower | or use the Asterisk SIP+NAT features |
16:54.14 | nextime | ManxPower : for example where you are using asterisk as a "client", you are inside a nat, and you can't do a port forward, and you are a dummy user that want to set stun 'cause your voip isp tell to use it? |
16:54.36 | syzygyBSD | where is .si |
16:54.47 | ManxPower | I guess the only time STUN might be useful is if you are a cheap bastard and have your Asterisk server behind a dynamic IP address that is natted, but honestly, if that is the case SIP+NAT is the least of your problems |
16:55.13 | ManxPower | nextime: in that case you also have much more serious issues than SIP+NAT |
16:55.27 | nextime | ManxPower : i don't need stun for myself. I'm working on a http manager based gui (something like the asterisk-gui), so, if the stun cli command exists, i want to know why exists and eventually how can i use it |
16:55.28 | syzygyBSD | We have used stun for phones behind nat |
16:55.30 | bulle | ManxPower: its not a choice for everyone, to get a public static ip adress |
16:55.45 | penguinFunk | slovenia |
16:55.54 | ManxPower | bulle: if you want to run a server you should have a public static IP address |
16:56.14 | bulle | ManxPower: yeah, but what do you do, when you cant get one ? |
16:56.25 | ManxPower | bulle: you don't run a server. |
16:56.42 | ManxPower | "How can I drive to work if I don't have a car?" "You get a car." |
16:56.44 | bulle | ManxPower: sounds like a pretty crappy solution to the problem imho |
16:56.59 | penguinFunk | ManxPower: or cycle ? |
16:57.07 | bulle | ManxPower: or just take the buss, or train, or walk ? |
16:57.23 | nextime | ManxPower : what about running a "local" server for clients in lan, and use it as a "concentrator" to connect all clients on a voip isp in a small office where you can't have a pubblic ip? ( just to say an example ) |
16:57.33 | ManxPower | bulle: correct. So co-locate a server somewhere with a static IP. |
16:57.43 | nextime | servers != public servers |
16:57.48 | ManxPower | nextime: you don't need a static ip for that |
16:57.53 | bulle | nextime: indeed |
16:58.15 | *** join/#asterisk raisendman (n=masterra@203.87.200.78) |
16:58.20 | ManxPower | you need a static IP if you want clients to connect to the asterisk server from outside the NAT domain/network |
16:58.32 | penguinFunk | dyndns ? |
16:58.38 | ManxPower | you do not need it if asterisk is acting as a sip client |
16:58.59 | ManxPower | penguinFunk: that would be a solution if asterisk did not stop working after any DNS failure. |
16:59.09 | Mercestes | ManxPower, don't you love how people come in here askign questions and then tell you your answers are wrong? lol |
16:59.22 | ManxPower | Mercestes: you get used to it after a while. |
16:59.24 | cpm | Mercestes, you R rong! |
16:59.33 | Mercestes | cpm: I'm sorry! I tried. |
16:59.51 | ManxPower | Mercestes: I've been using asterisk for a long time. |
16:59.53 | nextime | ManxPower : right. so, if asterisk is running, from the "public" view, only as a client, and your voip isp want that you use stun, and "maybe" stun is now supported, why don't use it? |
16:59.55 | Ch0Hag | It's not quite as much fun as coming in to ask a question to be told that your question is wrong. |
17:00.10 | ManxPower | nextime: because STUN IS NOT NEEDED in that situation |
17:00.10 | Qwell[] | Ch0Hag: questions are very often wrong |
17:00.11 | adam_vollrath | Fun? |
17:00.27 | cpm | would be nice if there was a bar one could set, that forced folks to actually have a basic * system working before coming here with all their wierd stuff |
17:00.29 | Qwell[] | usually because the person doesn't understand asterisk enough to ask what they want |
17:00.42 | ManxPower | heck you don't even need to port forward if asterisk is acting as a client in that case |
17:00.49 | syzygyBSD | It takes a true genius like Qwell to know what people mean without them saying it |
17:00.52 | *** join/#asterisk mattchis (n=mattchis@216.54.143.246) |
17:00.58 | *** join/#asterisk outlier (n=tom@70.141.147.180) |
17:00.59 | Qwell[] | syzygyBSD: indeed |
17:01.00 | Ch0Hag | Quite. |
17:01.01 | *** join/#asterisk anthony] (n=anthony@175.21.188.72.cfl.res.rr.com) |
17:01.08 | nextime | ManxPower : you know. I know. but not all people know. Expecially where voip isp explicitally say "if you are under nat, use stun", and expecially when you can't put your hand on the router to do a port forward. |
17:01.27 | ManxPower | nextime: then you should read the asterisk docs and not your isps docs. |
17:01.43 | Ch0Hag | Asterisk docs? |
17:01.44 | ManxPower | nextime: you don't need to port forward in that situation either |
17:01.48 | Ch0Hag | Which ones would those be then? |
17:01.58 | Ch0Hag | Incomplete set A, or contradictory set B? |
17:01.58 | ManxPower | Ch0Hag: The Book |
17:02.05 | nextime | ManxPower : I know. You know. not all people know. |
17:02.06 | ManxPower | ~osmosis |
17:02.17 | jbot | osmosis is probably the act of beating yourself on the head repeatedly with THE BOOK, until some measure of absorption has occured ... or at least until your unconsciousness restores peace to the channel ... |
17:02.17 | nextime | ManxPower : anyway |
17:02.17 | syzygyBSD | ~book |
17:02.20 | jbot | methinks book is a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
17:02.20 | nextime | my question was simple |
17:02.20 | syzygyBSD | ~docs |
17:02.22 | jbot | it has been said that docs is Documentation can be found at http://www.digium.com/index.php?menu=documentation or http://www.voip-info.org/wiki-Asterisk or http://www.asteriskdocs.org or http://www.oreilly.com/catalog/asterisk or http://www.asteriskguru.com |
17:02.24 | nextime | with asterisk 1.4 a "stun" cli command appear. |
17:02.29 | Ch0Hag | Or the incomplete *and* contradictory source, maybe? |
17:02.29 | mattchis | Can anyone point me to where I can set long distance authorization codes in asterisk? |
17:02.42 | nextime | so, what's this fucking stun command? |
17:02.42 | ManxPower | mattchis: you don't. you write your dialplan for that feature. |
17:02.54 | ManxPower | nextime: "help stun" does not help? |
17:02.59 | nextime | ManxPower : no. |
17:03.01 | *** join/#asterisk ToyMan (n=Stuart@72.168.167.241) |
17:03.05 | ManxPower | odd |
17:03.19 | nextime | <PROTECTED> |
17:03.19 | nextime | <PROTECTED> |
17:03.29 | nextime | but stun debug on what? |
17:03.32 | ManxPower | nextime: that sounds pretty simple. |
17:03.42 | ManxPower | um, debug information for stun stuff. |
17:03.52 | ManxPower | like sip debug is debug info for sip stuff |
17:04.02 | nextime | ManxPower : yes, it's simple to debug. But how can i use it? ( if i don't use it, what sense as debugging it? ) |
17:04.16 | Ch0Hag | And iax debug is debug info for the bits of iax you don't need to debug. |
17:04.22 | ManxPower | nextime: if you don't use it then I would expect you would not get any debug output |
17:04.32 | nextime | ManxPower : ta-da! |
17:04.32 | *** join/#asterisk inv_arp[work] (n=junya@c-67-191-12-203.hsd1.fl.comcast.net) |
17:04.39 | nextime | ManxPower : so, how can i use it? |
17:04.56 | aydiosmio | just give up |
17:05.01 | ManxPower | aydiosmio: me too |
17:05.24 | Mercestes | Ch0Mag: might I suggest Cisco Call Manager then so you can berid yourself of asterisk's presence? |
17:05.55 | Ch0Hag | Um. |
17:05.56 | Ch0Hag | OK? |
17:05.57 | ManxPower | great. Now I have people /msg'ing me asking questions. |
17:06.07 | ManxPower | Like I'm their own personal tech support bitch |
17:06.13 | Mercestes | lol |
17:06.16 | Qwell[] | ManxPower: aren't you? |
17:06.17 | Mercestes | your popular, Manx. :) |
17:06.19 | aydiosmio | <PROTECTED> |
17:06.20 | Qwell[] | ;) |
17:06.35 | *** join/#asterisk oej (n=olle@apollo.webway.se) |
17:06.53 | Mercestes | see. If you were a complete ass like me, you wouldn't have ppl messaging you (like me). |
17:06.56 | ManxPower | If you want personal private help from me the first thing we need to do is decide how I am going to bill you for it. |
17:07.09 | nextime | ManxPower : setup an * server connected to telephony world by a PRI, take a premium number and redirect people to your phone numbers, so, you get some money :) |
17:07.12 | mattchis | Hows it going Mercestes!! |
17:07.24 | Qwell[] | m/sg ManxPower hi, can you help me be the next vonage? I can paypal you $20 |
17:07.26 | Mercestes | mattchis! Sup man? What are you doing on this side of freenode? |
17:07.37 | Mercestes | you already have private support. :P |
17:07.44 | ManxPower | m/sg My rate is $78/hr |
17:07.53 | Qwell[] | ManxPower: 78? |
17:08.03 | ManxPower | Qwell: Yes, I am a cheap whore. |
17:08.21 | ManxPower | Qwell: the rate does go up if you are an idiot. |
17:08.32 | Qwell[] | to 79.42? |
17:08.40 | ManxPower | $2,000/day |
17:08.42 | Mercestes | ManxPower: I was wondering why you charged me twice that. |
17:09.02 | lokkju_wrk | wtf is this vonage patent on VOIP stuff all about? |
17:09.09 | Qwell[] | Ch0Hag: and btw, if you don't like the current documentation situation... feel free to contribute |
17:09.10 | ManxPower | Mercestes: you didn't use any lube, that's why |
17:09.18 | *** join/#asterisk canapa (n=canapa@83-64-148-98.wolfsberg.xdsl-line.inode.at) |
17:09.28 | Ch0Hag | Well yes, but unfortunately I need to actually *understand* the spaghetti code first. |
17:09.34 | *** join/#asterisk CunningPike (n=CunningP@204.239.8.149) |
17:09.40 | Qwell[] | such as? |
17:09.47 | Mercestes | ManxPower, Oh. I thought there was an *extra* charge for lube, not a storage and handling fee for nto using it. =/ |
17:09.56 | Qwell[] | because a lot of people understand it just fine |
17:10.02 | Ch0Hag | The 13 thousand lines in chan_sip.c, for example. |
17:10.21 | Mercestes | Ch0Mag: Might I suggest Plan9 then? YOu can code the entire thing in 13 lines. |
17:10.23 | canapa | what kind of port forewarding would i have to make in my firewall for asterisk to work proper ? |
17:10.24 | aydiosmio | lokkju_wrk: I'm sure Google News can help you there |
17:10.25 | Qwell[] | well, ask Cisco for their Call Manager code then so you can modify that |
17:10.27 | *** join/#asterisk Y0da^ (n=jwilson@70.159.118.70) |
17:10.29 | Qwell[] | ...oh, right |
17:10.45 | Ch0Hag | Mercestes: gcc has this wonderful feature called 'linking'. |
17:10.53 | Ch0Hag | Perhaps you've heard of it? |
17:11.06 | Mercestes | Ch0Hag: Sorry? I'm not entirely fluent in troll. |
17:11.10 | *** join/#asterisk caio1982 (i=caio1982@CAcert-br/caio1982) |
17:11.33 | Qwell[] | Mercestes: he's saying we should break up chan_sip.c so that in order to change things, you need to edit 10 files |
17:11.46 | Mercestes | Qwell[]: Oh!!! brilliant! |
17:11.57 | Ch0Hag | I must say, this Digium-recommended discussion forum hasn't exactly got me enamoured of any paid-for support they may be offering. |
17:12.17 | Mercestes | Qwell[]: And instead of doing it himself (since we gave him the source and all) he wishes to incessantly bitch in here (us being responsible for the source and all) until it magically fixes itself, right? |
17:12.36 | Mercestes | Ch0hag: There is a major poitn your missing. This isn't paid-for support. |
17:13.01 | Mercestes | Ch0Hag: If you wish to *pay* us to bitch then I will be more than happy to personaly listen to you and reciprocate for as long as you feel it' snecessary. |
17:13.13 | Mercestes | Ch0Hag: I am certain Digium feels the same way. |
17:14.48 | *** join/#asterisk sevard (i=chuck-th@adsl-71-129-115-242.dsl.irvnca.pacbell.net) |
17:15.12 | aydiosmio | is it too early to tell Ch0Hag to GFY? |
17:15.20 | Ch0Hag | Perhaps you should look up the definition of 'enamoured'. |
17:15.24 | heison | anyone here using blacklist() in Asterisk 1.4? |
17:15.34 | Mercestes | Ch0Hag: Perhaps you should look up the definition of troll. |
17:15.44 | Qwell[] | "No definitions were found for enamoured." |
17:16.10 | Ch0Hag | Americans... |
17:16.12 | Ch0Hag | Enamored thes. |
17:16.13 | *** join/#asterisk Dovid (n=Dovid@l192-117-114-1.broadband.actcom.net.il) |
17:16.14 | Ch0Hag | then |
17:16.23 | heison | i used to have exten => s,n,LookupBlacklist(j) and i wonder how the new function should perform the jump |
17:16.25 | giasai68 | hello i have some problem with fax t.38 on my aterisk 1.4.1 : Unsupported SDP media type in offer: |
17:16.26 | Mercestes | This isn't digium. This is #freenode. Digium merely suggested #asterisk was helpful for being free. If you want digium...go buy something. |
17:16.30 | giasai68 | can you help me? |
17:16.40 | Dovid | hi guys. i am trying to install g729 on my asterisk. does asterisk only have g729a or can i get vanilla g729 ? |
17:16.47 | Qwell[] | Dovid: only g729a |
17:16.49 | Mercestes | We already gave you the book and voip-info.org . Everyone else figured it out with that. |
17:16.50 | Dovid | :( |
17:16.53 | Dovid | thanx |
17:16.59 | Qwell[] | but isn't it supposed to be compat or something? |
17:17.19 | Qwell[] | Mercestes: the people who wrote the book figured it out with far less.. |
17:17.24 | giasai68 | chan_sip.c:4586 process_sdp: Unsupported SDP media type in offer: image 10912 udptl t38 |
17:17.26 | Ch0Hag | As Digium explicitely recommend this channel, it must be considered an advert for them, and by my experience, and watching the experience of others, it is not a very good advert at all. |
17:17.46 | denon | this channel isn't an advertisement for anyone |
17:17.51 | Gido-E | Ch0Hag go wank your self. |
17:17.51 | denon | we're just here to talk about asterisk |
17:17.55 | Mercestes | Ch0Hag: Sorry to loose you as a customer then. please feel free to report me specifically. bye. |
17:18.08 | Ch0Hag | I am not, and have never been, expecting free support, but some manners wouldn't go amiss. |
17:18.17 | ManxPower | Dovid: almost nothing out there supports plain g729 |
17:18.32 | adam_vollrath | Manners? Welcome to IRC. |
17:18.33 | aydiosmio | amiss? |
17:18.40 | tuan_modulis | there are lots of unix users here.... we're all egoists by nature |
17:18.49 | Gido-E | I am VERRY VERRY VERRY happy with Asterisk and Digium ofcourse. |
17:18.54 | Ch0Hag | But then I guess I can't really expect very much on that score from Americans. |
17:19.07 | denon | Ch0Hag: stop trolling. now. |
17:19.14 | aydiosmio | ya rly |
17:19.27 | denon | if you have a legitimate question - someone may try to help. Not everyone knows everything, and not everyone has time to help. |
17:19.41 | Ch0Hag | I don't expect them to. |
17:19.49 | Ch0Hag | Hell, I don't even *expect* them to be polite. |
17:19.53 | denon | this is peer to peer - and a very international committee |
17:20.01 | denon | er community |
17:20.13 | Ch0Hag | But as the public face of a growing company, it's at least a bloody good idea. |
17:20.34 | aydiosmio | pretty sure this channel predates Digium. |
17:20.40 | aydiosmio | wink wink |
17:20.48 | denon | Ch0Hag: are the people you're talking about even involved with digium? |
17:20.53 | Mercestes | #asterisk is not digium, public face or otherwise. We're users of the product, asterisk with little or no affiliation to digium. |
17:20.58 | denon | any random troll can walk into the channel and start beging rude |
17:21.06 | Ch0Hag | That is irrelevant. |
17:21.08 | Mercestes | yea, look at me. |
17:21.13 | denon | point :) |
17:21.48 | denon | Ch0Hag: it's a community, not a digium support channel |
17:22.35 | Ch0Hag | Digium recommend #Asterisk, there is even a chance that some of its staff are present. In that case any company with half a clue would do its level best to ensure the company and product is always looked upon in a good light. |
17:22.38 | tuan_modulis | I join the channel mostly to leech off circumstantial advice |
17:22.56 | Qwell[] | tuan_modulis: that's basically how I learned.. |
17:22.58 | denon | Ch0Hag: that would involve us silencing everyone but digium staff |
17:23.04 | denon | which is impossible in a peer to peer community |
17:23.10 | denon | you take the good with the bad on irc |
17:23.13 | Qwell[] | at that point, it's no longer a community |
17:23.18 | denon | exactly |
17:23.30 | denon | Ch0Hag: you may find the mailing list to be more to your liking, if you don't like the casual atmosophere of irc |
17:23.32 | Ch0Hag | My point, from way back when, is that either Digium don't care about their public appearance or would probably be wise to no longer recommend this channel in any form. |
17:23.39 | Mercestes | Ch0Hag: Is there a specific issue/problem you have? now would be a good time to bring it up. You have our attention now. |
17:23.56 | denon | Ch0Hag: I believe it's not a matter of them not caring about their appearance, just that they were trying to give you some community options |
17:24.06 | aydiosmio | Ch0Hag: please, just leave |
17:24.21 | Mercestes | yea, judging Digium based on an IRC channel is like judging microsoft based on the Dixie Chicks. |
17:24.45 | denon | Ch0Hag: you'll have to excuse me, I'm late for a meeting - but I do think you'll find a mix of good and bad on irc, but sift through it all and you'll find an extensive amount of knowledge |
17:24.56 | Mercestes | I mean, all that awful noise did cmoe through media player, right? If Microsoft cared about their image they'd censure some of that instead of allowing their trademark to be defamed. |
17:24.59 | Ch0Hag | "Many of these people join *our live IRC Asterisk chat channel*..." |
17:25.14 | aydiosmio | Ch0Hag: just. go. |
17:25.18 | adam_vollrath | They should probably change that. |
17:25.26 | Gido-E | Ch0Hag bye! |
17:25.42 | Ch0Hag | That would be a good idea. It demonstrates a sense of ownership/control which is not there. |
17:25.45 | aydiosmio | if we're wasting so much of your precious time not answering your query, perhaps it's not wise to waste it arguing with the wind |
17:26.01 | Mercestes | he doesn't even have a query... |
17:26.42 | Ch0Hag | All the egotism and bad manners I've seen could easily be ignored if Digium made it explicit that they only affiliated with the channel in an anciliary manner. |
17:26.50 | aydiosmio | thanks, noted |
17:26.53 | aydiosmio | see ya later |
17:27.01 | adam_vollrath | Send them an email. Seriously. |
17:27.12 | Ch0Hag | As they state the opposite, I assumed that they had at the very least some presence in here. |
17:27.22 | Qwell[] | of course they have a presence in here |
17:28.06 | Ch0Hag | Personally, I couldn't give a toss, I'm in a large family and I've been a geek all my life, I can give and take internet crap with the best of them. |
17:28.23 | CunningPike | Ch0Hag: If you want the complete and undivided attention of Digium staff, an ABE license may be right for you |
17:28.24 | aydiosmio | ]okay see ya then |
17:28.25 | wunderkin | ... |
17:28.28 | aydiosmio | buh bye |
17:28.33 | Ch0Hag | But Digium's public image is almost certainly suffering. |
17:28.45 | techie | nah |
17:28.49 | Gido-E | Ch0Hag no, you let us suffering. |
17:28.51 | aydiosmio | no, but thanks for your concern |
17:29.01 | Qwell[] | Ch0Hag: Do you have a specific complaint? |
17:29.04 | Corydon-w | Ch0Hag: that's enough. You've berated the channel for the past hour. Ask a question now or leave. |
17:29.12 | Ch0Hag | Well, no, Digium's public image *has* suffered. |
17:29.15 | Dovid | Qwell: can you have a look at this ? |
17:29.16 | Dovid | http://www.pastebin.ca/410774 |
17:29.38 | Qwell[] | Dovid: You're going to have to call Digium support |
17:29.48 | Dovid | is it a server issue or an issue with me ? |
17:29.53 | Gido-E | :-) |
17:29.58 | Dovid | ur guess |
17:29.58 | Qwell[] | well, it looks like a proxy problem on your end perhaps |
17:29.59 | Dovid | ? |
17:30.06 | Dovid | thanks |
17:30.28 | Ch0Hag | Qwell[]: No. A suggestion. That Digium don't condone the channel or give explicit, clear indication that it has nothing (formally) to do with them. |
17:30.38 | *** mode/#asterisk [+b %Ch0Hag!*@*] by Corydon-w |
17:30.52 | Gido-E | thanx Corydon-w |
17:31.22 | Dovid | Thanks Corydon |
17:31.32 | Dovid | Qwell: do u guys work on UDP or TCP for the server ? |
17:31.34 | CunningPike | Ch0Hag: You're absolutely correct - sales@digium.com |
17:31.56 | Corydon-w | CunningPike: please don't feed the troll. |
17:31.56 | Dovid | i tried to get to it from a diffrent server in a diffrent data center thru telent and conneciton refused |
17:32.05 | *** join/#asterisk rudholm (i=rudholmm@nat/yahoo/x-29f6201b3a2f8b78) |
17:32.24 | Corydon-w | CunningPike: he can't respond in-channel anyway |
17:32.42 | Qwell[] | Dovid: unfortunately, nobody in here is going to be able to debug that issue at all.. about the best I can recommend is checking your firewall settings |
17:33.00 | Dovid | Qwell: even from 2 diffrent server in 2 seperate locations ? |
17:33.03 | Dovid | ok. i will call em |
17:33.11 | Qwell[] | Dovid: anything's possible :) |
17:33.13 | Qwell[] | and that would be best |
17:33.38 | Dovid | thx. in Israel ATM - on evdo so i cant use voip @ 15cents a minute.... |
17:33.58 | aydiosmio | talk faster. |
17:34.03 | Dovid | lol |
17:34.06 | *** join/#asterisk dhill (i=dhill@fog.mindcry.org) |
17:34.11 | Dovid | like the cell commercials |
17:34.18 | dhill | hello |
17:34.26 | dhill | any chan_sip developers here? |
17:34.43 | aydiosmio | dhill: tried #asterisk-dev? |
17:34.47 | dhill | ahh, ok |
17:34.49 | *** part/#asterisk dhill (i=dhill@fog.mindcry.org) |
17:35.13 | russellb | there are no chan_sip developers |
17:35.40 | aydiosmio | in soviet russia, chan_sip develops you! |
17:36.24 | russellb | i was looking at some code in there last week |
17:36.27 | russellb | it did something to me ... |
17:36.40 | Mercestes | nothing....unnatural I hope. |
17:37.05 | russellb | mommy, chan_sip touched me :( |
17:37.07 | Dovid | Qwell: may I PM ? |
17:37.52 | *** part/#asterisk nextime (n=nextime@unaffiliated/nextime) |
17:37.57 | Dovid | or russelb: may I PM ? |
17:38.20 | russellb | no |
17:38.30 | Dovid | ok. something that i dont wana say in channel ;) |
17:39.06 | Mercestes | So! I'm installing asterisk onto a 256k flash memory card using Soentoo with a Tei410P mounted in an embedded Soekris box. Anyone familiar? |
17:39.23 | Mercestes | Dovid: We already know. You have a crush on russelb. |
17:39.25 | Qwell[] | 256k? |
17:39.38 | Qwell[] | seriously? |
17:39.41 | Mercestes | Qwell[]: yea, my boss is cheap. Plz! help! I'll lose my job. |
17:39.48 | Qwell[] | ahh, smartass |
17:39.50 | Qwell[] | :p |
17:39.51 | Mercestes | Qwell[]: 256mb actually |
17:39.55 | Qwell[] | You had me for a second there ;) |
17:40.01 | Mercestes | lol |
17:40.04 | Mercestes | actually, I really am.... |
17:40.11 | Mercestes | but I'm not asking for help (yet.) |
17:40.30 | Mercestes | actually, th ecard says 256(sigma symbol) no clue what that means. |
17:40.40 | Mercestes | oh. *pulls card out* it's mb sideways. |
17:40.59 | Dovid | hehe |
17:41.29 | Dovid | Qwell: turns out my register program is from the last time i bought codecs which was 3 years ago |
17:41.38 | Qwell[] | nice |
17:41.41 | *** join/#asterisk Fieldy (i=ntes4WZj@gentoo/contributor/Fieldy) |
17:42.56 | Mercestes | now I just need to know if I can fit a kernel/asterisk/mysql/apache/Fop all on a 256mb card. =/ |
17:43.07 | Qwell[] | Mercestes: 256mb is *huge* |
17:43.23 | Mercestes | yea, for linux I suppose. |
17:43.29 | Mercestes | I plan to remap all the /var stuff onto a lappy Hdd. |
17:43.39 | Mercestes | For CDrs and recordings. |
17:43.50 | Dovid | i like the new one much better ;) |
17:44.37 | Mercestes | I haven't even installed an OS yet. I dragged out a PC, put a gentoo boot cd in, downloaded stage 3 and source, chrooted into gentoo, downloaded soentoo source, chrooted into that... |
17:46.20 | Mercestes | Someone wanted to knwo how to setup phones through 2 nats and a firewall. I told them OpenVPN. |
17:50.27 | *** join/#asterisk oej (n=olle@apollo.webway.se) |
17:50.50 | *** join/#asterisk r0d3nt (n=RatMan@punk.valuetel.net) |
17:53.58 | Mercestes | Hrm. I should write a wiki on this Soentoo build. |
17:55.17 | *** join/#asterisk DarKnesS_WolF (n=sherif@unaffiliated/sherif) |
18:02.11 | ManxPower | Mercestes: I've done Asterisk <-> NAT Router <-> Internet <-> NAT Router <-> NAT Router before without any issues |
18:02.49 | ManxPower | ASTERISK and the NAT Router for Asterisk had to be set up correctly, but that was all |
18:03.04 | syzygyBSD | how do I view the debug log.. stupid question I know |
18:03.44 | ManxPower | syzygyBSD: do you have logging of debug mesages enabled? |
18:03.56 | Mercestes | ManxPower, Your not even from this solar system tho. We're talking about humans setting it up here. |
18:04.07 | syzygyBSD | ManxPower: yes |
18:04.18 | ManxPower | syzygyBSD: /var/log/asterisk |
18:04.46 | Mercestes | But they are gonig gateway -> some unnamed firewall -> asterisk (with a public ip of 192.169?? huh?) with their own iptables magic setup. so, yea, not setup right. |
18:05.15 | Mercestes | syzygyBSD, You can also add debug to the console in logger.conf if you wish' |
18:05.36 | syzygyBSD | hmm, I checked that and I dind't think it had the debug messages... |
18:05.36 | syzygyBSD | ahh, thanks |
18:05.38 | toombaloomba | quick question, when I have something like exten => 1111,1,Agi(agi://<server_ip>/cnam.agi?number=${CALLERID}) what kind of connection is being made from my asterisk box to this server at the IP, is it HTTP? |
18:05.39 | ManxPower | Mercestes: that is like watching a retarded kid take the SATs. It would be funny if it was not so sad. |
18:05.40 | syzygyBSD | that is what I was looking for Mercestes |
18:05.52 | Mercestes | syzygyBSD, Your welcome. |
18:05.56 | Mercestes | ManxPower, I agree. Lol |
18:06.03 | ManxPower | toombaloomba: IP |
18:06.28 | *** join/#asterisk lucifr (n=chatzill@66.6.221.64) |
18:07.00 | Mercestes | syzygyBSD, don't forget logger reload or core reload logger or whatever it is . |
18:07.13 | lucifr | Hi everybody.. |
18:07.15 | toombaloomba | ManxPower im asking because this is for a CNAM service but I dont want asterisk to call directly, I want to make a HTTP call to it from another system, and then my asterisk will get the CNAM from here |
18:07.29 | toombaloomba | ManxPower so im wonder if I do http://IP/cnam.agi?number=12354 what will I get |
18:07.31 | ManxPower | toombaloomba: it sucks to be you |
18:07.53 | Mercestes | hi, luci. |
18:07.54 | ManxPower | toombaloomba: What in the world makes you think asterisk will do an HTTP connection |
18:07.59 | ManxPower | or even support it. |
18:08.02 | toombaloomba | not from asterisk |
18:08.06 | lucifr | Hi there |
18:08.13 | toombaloomba | ManxPower not from asterisk itself |
18:08.19 | toombaloomba | ManxPower and yes it can do HTTP, u heard of CURL? |
18:08.22 | ManxPower | toombaloomba: if it's not from Asterisk then why are you running exten => 111,1,AGI(whatever) |
18:08.29 | *** join/#asterisk Dovid (n=Dovid@l192-117-114-1.broadband.actcom.net.il) |
18:08.30 | lucifr | I'm new to Asterisk (AsteriskNOW) and I've been trying to get it to work with my VoIP provider but no luck. Now I'm trying it for the second time, but I think I'll need some help. |
18:08.32 | ManxPower | toombaloomba: Curl != AGI |
18:08.34 | giasai68 | i have this problem:chan_sip.c:4586 process_sdp: Unsupported SDP media type in offer: image 10912 udptl t38 |
18:08.35 | Dovid | hi guys |
18:08.46 | giasai68 | can you help me? |
18:08.50 | *** join/#asterisk logan|w (n=sadfasdf@station441.dallasix.net) |
18:08.55 | Dovid | Mar 26 14:07:54 WARNING[2416]: channel.c:2603 ast_request: No channel type registered for 'h323' |
18:08.55 | Dovid | Mar 26 14:07:54 NOTICE[2416]: app_dial.c:1059 dial_exec_full: Unable to create channel of type 'h323' (cause 66 - Channel not implemented) |
18:08.58 | ManxPower | giasai68: tell your SIP client to stop sending unsupported stuff. |
18:09.11 | ManxPower | Dovid: That is a pretty obvious error message. |
18:09.17 | Dovid | this means that i didnt install the h323 driver for asterisk ? |
18:09.26 | ManxPower | Dovid: Yup! |
18:09.36 | Dovid | wierd. cause i did it |
18:09.37 | lucifr | If I install AsteriskNOW can I avoid the GUI for now and configure the configuration file manually? |
18:09.38 | Dovid | let me try again |
18:09.50 | ManxPower | actually it means "none of the channel drivers told asterisk they support Dial(h323/whatever) |
18:09.50 | giasai68 | how can i fix? i'm tring to send a fax trought atserisk but i'm not able to do this |
18:10.10 | ManxPower | giasai68: Since asterisk does not support fax over ip.... |
18:10.16 | Dovid | ah |
18:10.23 | Dovid | the readme says to do make opt |
18:10.24 | ManxPower | giasai68: what verison of Asterisk |
18:10.40 | giasai68 | version 1.4.1 |
18:10.42 | Dovid | but when doing make opt i get |
18:10.42 | Dovid | make: *** No rule to make target `opt'. Stop. |
18:10.54 | ManxPower | Dovid: I can't help yoiu with H323, very few people can. |
18:11.32 | ManxPower | giasai68: you are trying to do T.38. Check the Wiki, as I don't know anyone that has gotten T.38 to work with Asterisk, but rumor is that 1.4 is supposed to support it in a limited way. |
18:11.40 | Mercestes | lucifr, #asterisknow can help you with asterisknow. I woul dsuggest just using the config files tho. |
18:11.46 | Mercestes | lucifr, I can't answer your question tho. |
18:12.02 | ManxPower | giasai68: by "limited way" I mean asterisk cannot terminate a T.38 call, it can only pass it thru to the final destination |
18:13.12 | Dovid | anyone here know h323 ? |
18:13.42 | ManxPower | Dovid: there are at least FOUR different H323 channel drivers for asterisk. you'll need to ask if people are using the one you are using |
18:14.18 | Dovid | vanilla h323 |
18:14.18 | ManxPower | Dovid: no such thing |
18:14.19 | Dovid | trying to compile /asterisk/channels/h323 |
18:14.19 | *** join/#asterisk jcmoore (n=jcmoore@unaffiliated/tgrman) |
18:14.19 | ManxPower | Dovid: that would be the "Nufone H323" |
18:14.22 | Dovid | ah. |
18:14.27 | Dovid | that is made for nufone ? |
18:14.51 | *** join/#asterisk mark_coastal (n=chatzill@rrcs-67-78-216-114.se.biz.rr.com) |
18:14.51 | ManxPower | as opposed to the Objective Systems H323, Woomera H323, and the GPL one in asterisk-addons |
18:15.00 | Dovid | ManxPower: where are the other ones ? and i want a list of other ones |
18:15.01 | giasai68 | yes now i'm doing this i'm using asterisk as pass it thru... i'm snding fat with fax machine on adapter phone via ip, this equipment sipport t.38, i'm using asterisk as sip proxy and asterisk machine send this to a gateway connected to pri |
18:15.04 | ManxPower | Dovid: no it is made BY nufone. |
18:15.08 | Dovid | ok |
18:15.53 | mark_coastal | hello, looking for a manageable way to set oubound callerid for extensions (or groups of extensions) - preferably using realtimedb. Using vanilla asterisk. |
18:16.25 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
18:17.00 | ManxPower | Since the GPL one in asterisk-addons was comissioned by Digium, I would look at that one. |
18:18.02 | *** part/#asterisk deeperror (n=deeperro@mail.banctel.com) |
18:18.02 | *** join/#asterisk xo8ox (n=pride_32@wsip-66-210-250-2.ph.ph.cox.net) |
18:18.07 | xo8ox | guys when dialing an extention I get this warning: |
18:18.09 | xo8ox | WARNING[16593]: app_dial.c:1081 dial_exec_full: Unable to create channel of type 'SIP' (cause 3 - No route to destination) |
18:18.26 | Mercestes | xo8ox: That means the peer is offline. |
18:18.37 | xo8ox | the phone u mean ? |
18:18.44 | Mercestes | whatever it is. |
18:18.50 | Mercestes | could be a sip oxen for all I care. |
18:18.56 | xo8ox | its a polycom phone I'm trying to dial to |
18:18.57 | xo8ox | lol |
18:19.01 | Dovid | ManxPower: whats the diffrence between h323, oh323 and ooh323 ? |
18:19.03 | Mercestes | It is offline then. |
18:19.04 | xo8ox | I hate these polycom phones |
18:19.07 | Dovid | diffrent protocols ? |
18:19.09 | Mercestes | polycom ftw! |
18:19.10 | ManxPower | it could also mean that the device never registered or that you screwed up your sip.conf entry |
18:19.15 | Mercestes | polycom > world |
18:19.22 | ManxPower | Dovid: they are all written by different people |
18:19.35 | *** mode/#asterisk [-b %Ch0Hag!*@*] by Corydon-w |
18:19.41 | ManxPower | xo8ox: "sip show peers" will tell you if it is registered or not |
18:19.44 | lucifr | Mercestes, do yo know where I can find the configuration files for AsteriskNOW> |
18:19.46 | lucifr | ? |
18:19.48 | Dovid | but they call act the same ? |
18:19.55 | xo8ox | aha ok thanks |
18:19.56 | Mercestes | lucifr, in #asterisknow |
18:20.00 | ManxPower | xo8ox: we drown people on this channel that dis polycoms |
18:20.05 | lucifr | ok, thanks |
18:20.07 | Qwell[] | polycom sucks |
18:20.07 | ManxPower | Dovid: I highly doubyt it. |
18:20.08 | Dovid | just compiled it from asterisk add-ons and still getting errors - this is fun |
18:20.10 | Qwell[] | kidding! |
18:20.17 | Qwell[] | don't tar/feather me please :p |
18:20.18 | *** join/#asterisk techie (n=gus@voip.routedsystems.com) |
18:20.26 | *** join/#asterisk jm|laptop (n=jm|home@dilbert.jamiem.com) |
18:20.31 | *** join/#asterisk billytwowilly (n=chris@S01060016b649355d.ed.shawcable.net) |
18:20.41 | ManxPower | Dovid: h323 is one of the hardest, most poorly documented, most complicated things you can do with asterisk. NOBODY uses h323 if they have any way around it. |
18:21.05 | Dovid | ManxPower: Yup. I agree. But I must use it. carrier only supports it :( |
18:21.12 | Mercestes | Dovid: get a new carrier |
18:21.17 | Dovid | i may have to get a cisco box - which i dont wana so |
18:21.23 | ManxPower | Dovid: expect to spend a week getting it to work the way you want. |
18:21.26 | Qwell[] | mog: :( |
18:21.28 | Dovid | Mercestes: no one can beat thier rates :( |
18:21.32 | billytwowilly | speaking of new carriers, which one should I use in Canada for the cheapest/best service? The voip wiki is.. confusing. |
18:21.45 | mog | Qwell, ? |
18:21.53 | Qwell[] | <--- |
18:21.55 | Mercestes | Dovid: yea, your bill will continue to be $0. Now if yo ucan get it to actually *work* every now and then, they may begin charging you. |
18:22.02 | Qwell[] | relative to your current location |
18:22.14 | Qwell[] | or maybe --->? whichever |
18:22.31 | Ch0Hag | What's the state of h323 generally? |
18:22.40 | Mercestes | *sighs* Any chance for inherent SS7 support in Asterisk anytime in the near future? in time for 1.6 maybe? |
18:22.54 | Ch0Hag | If this carrier supports only it, that implies that [some] people aren't doing everything they can to avoid it. |
18:23.10 | Mercestes | Ch0Hag: I think manxpower covered it earlier with: h323 is one of the hardest, most poorly documented, most complicated things you can do with asterisk. NOBODY uses h323 if they have any way around it. |
18:23.22 | Ch0Hag | Mercestes: Except Dovid's carrier. |
18:23.33 | mog | Mercestes, it already works |
18:23.35 | mog | libss7 |
18:23.37 | Dovid | hehe |
18:23.38 | Mercestes | Ch0Hag, didn't you just get unbanned? |
18:23.43 | Ch0Hag | Is it something it's worth adding to [the bottom of] my list of things to learn? |
18:24.53 | mog | but i dont know anyone that would reccomend it for production |
18:24.53 | Mercestes | mog: zomg! what ver? 1.4? |
18:24.53 | Mercestes | Ch0Hag: no. Honestly, no. Unless you wanna maintain/fix it. |
18:24.53 | mog | its a seperate library i think it only works with trunk |
18:24.54 | Mercestes | mog: hrm. Yea, I avoid trunk. |
18:24.54 | ManxPower | Mercestes: where are you located? |
18:24.54 | Ch0Hag | Right. |
18:24.54 | mog | beggers cant be choosers Mercestes |
18:25.06 | Mercestes | mog: Aye. Maybe I can patch it into 1.2.17 or something. =/ |
18:25.07 | Dovid | mog: if i get ABE will i get h323 support ? |
18:25.14 | Mercestes | ManxPower, H-town. |
18:25.23 | xo8ox | guys in linux how do I add a perm route ? |
18:25.27 | ManxPower | Mercestes: you must be a carrier. |
18:25.42 | ManxPower | since no telco that I know will provide an SS7 line to a non-carrier |
18:25.49 | Mercestes | ManxPower, Used to be. this is to plug into sprint and make my PBX part of their network for free cell phone calls. |
18:26.08 | Dovid | Qwell: if i get ABE will i get h323 support ? |
18:26.19 | Qwell[] | Dovid: You'll need to call sales and ask :) |
18:26.23 | Dovid | thanx |
18:26.35 | Mercestes | lol |
18:27.13 | Mercestes | mog: ....maybe I can come up with a trunk box and iax it on over as a ss7 gateway. =/ hrm. *ponders evil* |
18:27.31 | *** join/#asterisk `p4r14h`work (n=josh@24-119-48-78.cpe.cableone.net) |
18:28.31 | ManxPower | Mercestes: you'll give me a guest account on the box, right? |
18:28.45 | *** join/#asterisk spatulamaan (n=ggilmore@207.188.8.252) |
18:28.52 | Mercestes | ManxPower, I unno, maybe I can hook something up. |
18:29.31 | Mercestes | ManxPower, Or I can document the setup and point you at the (rather cheap) sprint plan that allows you to do it. :) |
18:29.55 | *** join/#asterisk jm|laptop (n=jm|home@dilbert.jamiem.com) |
18:30.17 | ManxPower | Mercestes: how are you linking to spring? |
18:30.20 | ManxPower | sprint, even |
18:30.24 | jm|laptop | :( |
18:30.58 | jm|laptop | I'm having problems with my zap channel - when I try to place a call through the zap chan 1 or if I pickup an incoming call there is just loud hum and a high pitched squeak |
18:31.09 | jm|laptop | the POTS line is fine, I checked with an analogue phone |
18:31.27 | ManxPower | jm|laptop: could it be ECFO? |
18:31.30 | ManxPower | ~ecfo |
18:31.41 | jbot | Echo Canceler Freak Out, this happens when the rxgain is too high and the echo canceler freaks out. Some users describe it as "screeching", "feedback", "static", or other useless terms. If users report "static" on a system where there cannot be static (all digital, PRI, SIP, etc), you might be experiencing ECFO. what happens when the echo canceller suddenly ... |
18:31.54 | jm|laptop | hmm |
18:32.00 | jm|laptop | how might I check? |
18:32.03 | ManxPower | usually it happens in the middle of a call. |
18:32.11 | ManxPower | jm|laptop: set your rxgain to -6 |
18:32.18 | *** join/#asterisk Deeewayne (i=dwayne@nat/digium/x-a35a3dbd23c050d7) |
18:32.27 | ManxPower | or better yet, lower it in increments of 2 |
18:32.48 | jm|laptop | this is way before a call |
18:32.50 | Mercestes | ManxPower: http://www.sprint.com/business/products/products/pcsIntegratedOffice_small_tabA.html |
18:33.56 | ManxPower | Mercestes: Uh, yeah. Let me know how it works out for you. |
18:34.09 | Qwell[] | eww, sprint |
18:34.13 | Strom_C | hi |
18:34.17 | jm|laptop | maybe it's the card |
18:34.19 | jm|laptop | brb |
18:34.23 | ManxPower | jm|laptop: it can't hurt to try it. also confirm you do not have irq issues |
18:34.58 | Mercestes | ManxPower, Sure. I'll document for you. Could be a good sale. |
18:35.09 | ManxPower | Mercestes: Did they forget to put in the part where it cures baldness, is a solution to world hunger and gives great backrubs? |
18:35.34 | ManxPower | Mercestes: I'm skeptical, but if it works for you, let me know |
18:35.53 | Mercestes | ManxPower, they forgot to put in the part where it requirse SS7 signalling and actually getting it setup requires you to jump through about 90 hoops just to talk to the guy who knows what it is |
18:36.02 | *** join/#asterisk TAMIKAMI01 (n=radamant@200.34.113.90) |
18:36.11 | ManxPower | Mercestes: that is typical |
18:36.18 | ManxPower | Mercestes: how much does it cost? |
18:36.20 | Mercestes | yea. ss7 is my only hangup righ tnow. |
18:36.23 | Mercestes | about 10 bucks a phone. |
18:37.54 | *** join/#asterisk groogs[h] (n=gregm@cbl-66-102-80-229.wtccommunications.ca) |
18:38.03 | ManxPower | It would he a hard sell to my users |
18:38.23 | Mercestes | gives you 4 digit extension dialing into the office. |
18:38.33 | ManxPower | that is what would confuse them |
18:38.40 | Mercestes | true. |
18:38.41 | Qwell[] | ss7 phones? |
18:38.54 | Mercestes | Qwell[] Nah, I just need ss7 in *. |
18:39.00 | Qwell[] | oh |
18:39.02 | Mercestes | Mog says it's in trunk as libss7 but...it's trunk only |
18:39.18 | Mercestes | stupid zaptel blowing up on stupid embedded gentoo system on my stupid dell. |
18:39.25 | ManxPower | I'll have to wait for 1.6 to be released themn |
18:39.28 | Mercestes | stuipd ebuilds. |
18:39.28 | mog | just take trunk chan_zap and run it in 1.4 |
18:39.57 | Qwell[] | frankenversion? |
18:40.18 | ManxPower | Any time there is a phone glitch the head of accounting goes to the president of the company demanding he fire the entire IT department |
18:40.20 | TAMIKAMI01 | hi |
18:40.42 | Strom_C | Qwell[]: i got my 7960 working with skinny firmware |
18:40.46 | Qwell[] | Strom_C: w00t |
18:40.49 | Qwell[] | 1.4? |
18:40.56 | Strom_C | is it possible to do transfers? the key doesnt seem to do anything |
18:40.59 | Qwell[] | no :D |
18:41.00 | Strom_C | yeah, 1.4 svn branch |
18:41.06 | Strom_C | that's...disappointing |
18:41.19 | Qwell[] | eventually it will |
18:41.52 | Mercestes | yay, frakenversion. |
18:41.59 | Mercestes | mog: svn checkout zaptel? |
18:42.01 | Mercestes | hrm. |
18:42.09 | Mercestes | so half ebuild, half source, half trunk. ...I like it |
18:42.10 | tzafrir | where can I find some asterisk-based games? (dialplan) |
18:42.12 | TAMIKAMI01 | i have and TDM2400P with (S400M)X4 AND (X400M)X1 and my cuestion i how i be sure i have this modules presents for zaptel? |
18:42.25 | Qwell[] | tzafrir: menuselect :p |
18:42.28 | TAMIKAMI01 | exist any command to checkout this? |
18:42.32 | tzafrir | anybody with useful keywords? |
18:42.49 | tzafrir | Qwell[], have you integrated tetris into it yet? or minesweeper? |
18:42.56 | Qwell[] | something like that |
18:43.02 | Qwell[] | tzafrir: hit 'i' ;) |
18:43.20 | Mercestes | tzafrir: I have some instructions on some dialplans using system() to destroy linux distros |
18:43.23 | Qwell[] | russellb: gmenuselect ^^ |
18:43.43 | tzafrir | no, somethis that can be done through the phone |
18:43.53 | *** join/#asterisk jm|laptop (n=jm|home@dilbert.jamiem.com) |
18:43.58 | Qwell[] | yeah, twisted and russellb wrote something in dialplan once |
18:44.01 | jm|laptop | ffs |
18:44.01 | Mercestes | This can be done through the phone. |
18:44.03 | Qwell[] | (I think) |
18:44.07 | adam_vollrath | I guess all user input should be validated, even in a Dialplan. |
18:44.09 | TAMIKAMI01 | i have and TDM2400P with (S400M)X4 AND (X400M)X1 how i configure them? |
18:44.14 | Mercestes | twisted doesn't like me. :( |
18:44.20 | tzafrir | TAMIKAMI01, at the load of wctdm2400xxp |
18:44.25 | Qwell[] | Mercestes: s/me/a lot of people/ :p |
18:44.30 | Mercestes | lol |
18:44.33 | Qwell[] | twisted rocks though :D |
18:44.35 | tzafrir | or using either zapscan (.bin) or genzaptelconf |
18:44.43 | TAMIKAMI01 | zaptel 211364 12 wcusb,wctdm,wcfxo,wcte11xp,wct1xxp,tor2,wctdm24xxp,wct4xxp |
18:44.45 | Mercestes | eh, he's never been of any real use to me. |
18:44.56 | Qwell[] | Mercestes: get him drunk |
18:44.59 | tzafrir | TAMIKAMI01, no need to re-ask the same question over and over again |
18:45.17 | tzafrir | TAMIKAMI01, which distro do you use? |
18:45.27 | TAMIKAMI01 | i'm using fedora core 6 |
18:45.41 | TAMIKAMI01 | with linux kernel 2.6.18-1.2798.fc6xen |
18:45.48 | Mercestes | eh. |
18:45.49 | jm|laptop | when I invoke ztmonitor 1 I get a loud tone from my soundcard, too |
18:46.01 | Qwell[] | jm|laptop: that's totally a shared inq |
18:46.02 | tzafrir | look at /var/log/messages . Look for the part where the module has loaded. |
18:46.03 | Qwell[] | irq* |
18:46.11 | Mercestes | I'd rather get him roofied and drop him off at a few bars I know of in the more festive side of town. |
18:46.12 | jm|laptop | erk |
18:46.15 | tzafrir | so: games, anybody |
18:46.21 | jm|laptop | it sort of changes when I dial out via zaptel |
18:46.35 | russellb | you weren't supposed to just tell people. |
18:46.35 | russellb | nub |
18:46.37 | jm|laptop | but my tones chirrup rather than ... well ... tone |
18:46.38 | Qwell[] | :( |
18:46.40 | TAMIKAMI01 | tzafrir, kernel: Found a Wildcard TDM: Wildcard TDM2400P (24 modules |
18:46.42 | Qwell[] | my point is still valid! |
18:46.49 | russellb | heh, yeah |
18:47.00 | russellb | but i'm too lazy |
18:47.03 | Qwell[] | heh |
18:47.29 | jm|laptop | hmm |
18:47.47 | jm|laptop | so sip to sip works and when someone dials IN they can here me but I just get a zuzzing noise |
18:48.56 | jm|laptop | s/here/hear/ |
18:49.10 | jm|laptop | er: yeah. Thanks jbot |
18:49.21 | *** join/#asterisk ToyMan (n=Stuart@user-12lcqol.cable.mindspring.com) |
18:50.37 | Mercestes | jm|laptop, something is severely wrong. :P |
18:50.52 | Mercestes | jm|laptop: what card is it? |
18:51.09 | jm|laptop | OK Wildcard X100P Board 1 |
18:51.35 | *** join/#asterisk bkuhn (n=bkuhn@fsf/member/bkuhn/bkuhn) |
18:51.49 | TAMIKAMI01 | why i when i use make linux26 for zaptel ztdummy, zttool doesn't compiled? |
18:51.57 | jm|laptop | Zaptel Configuration ====================== Channel map: Channel 01: FXS Kewlstart (Default) (Slaves: 01) 1 channels configured. |
18:52.29 | TAMIKAMI01 | why i when i use make linux26 for zaptel ztdummy, zttool doesn't compiled... make: *** [zttool.o] Error 1 ?? |
18:52.37 | jm|laptop | it was working until a few days ago, incidentally |
18:53.53 | *** join/#asterisk sysreq (n=sysreq@41-198-0-72-ppp.3menatwork.com) |
18:54.17 | TAMIKAMI01 | why i when i use make linux26 for zaptel ztdummy, zttool doesn't compiled... make: *** [zttool.o] Error 1 ?? |
18:54.28 | wunderkin | heh z0mg |
18:55.15 | aydiosmio | TAMIKAMI01: google PEBKAC |
18:55.16 | Corydon-w | TAMIKAMI01: that doesn't tell us the error, only that there was one. Please use http://pastebin.ca to paste the entire set of error messages |
18:56.25 | TAMIKAMI01 | ERROR: Module wcfxs does not exist in /proc/modules ???? |
18:56.30 | TAMIKAMI01 | ERROR: Module wcfxs does not exist in /proc/modules ???? |
18:56.34 | wunderkin | heh z0mg |
18:56.41 | wunderkin | :D |
18:57.15 | TAMIKAMI01 | CAS signalling on span 3 conflicts with Clear channel on channel 64??? |
18:57.17 | TAMIKAMI01 | ERROR: Module wcfxs does not exist in /proc/modules ???? |
18:57.26 | Mercestes | what part of "pastebin.ca to paste the entire set of error messages" did you not understand??? |
18:57.53 | wunderkin | all of it apparantly |
18:58.06 | Mercestes | Just a random guess here, but I think your problem is that you didn't follow directions. |
18:58.13 | wunderkin | ha |
19:02.12 | syzygyBSD | should asterisk auto create the tables in mysql it needs? |
19:02.26 | syzygyBSD | assuming it has permission |
19:02.29 | Qwell[] | syzygyBSD: no |
19:02.42 | gambolputty | I don't think * can auto create tables |
19:02.58 | *** join/#asterisk pfn (n=pfnguyen@netblock-66-245-252-239.dslextreme.com) |
19:03.04 | syzygyBSD | ahh, well, does anyone have a dump of the create statments I need? |
19:03.19 | Qwell[] | syzygyBSD: they should be in the doc/ directory |
19:03.26 | syzygyBSD | thanks |
19:03.32 | Mercestes | syzygyBSD, They also show up under asterisk wiki RTA google searches |
19:03.53 | syzygyBSD | Mercestes: doing one now... |
19:03.57 | Mercestes | syzygyBSD, Possibly not 100% correct but pretty close, and core debug on with debug to console will reveal the SQl statements that are failing if any are failing. |
19:04.09 | syzygyBSD | sadly no... |
19:04.11 | Mercestes | . |
19:04.25 | syzygyBSD | but I am using an old version trying to debug someone elses |
19:05.35 | *** join/#asterisk bkuhn (n=bkuhn@fsf/member/bkuhn/bkuhn) |
19:10.20 | *** join/#asterisk markit (n=konversa@host119-245-static.72-81-b.business.telecomitalia.it) |
19:11.05 | markit | hi, how can I "reject" an incoming call with mISDN? without answering it, I mean (I think is possible) |
19:11.28 | *** join/#asterisk CrashHD (n=crashhd@c-67-166-155-233.hsd1.ca.comcast.net) |
19:12.37 | necromcr | markit: hangup doesnt work? |
19:13.15 | markit | necromcr: I suppose I have to answer first... I want the caller find "busy" without waste a call |
19:13.44 | markit | necromcr: I think ISDN is like GSM, that you can "reject" incoming call without the need of answering it first |
19:15.20 | necromcr | markit: well.. work for me .. |
19:15.41 | necromcr | markit: do you want a busy signal or just a simple rejection? |
19:18.22 | anonymouz666 | r TOK_LP or TOKEN; Input: |
19:18.23 | anonymouz666 | <PROTECTED> |
19:18.32 | anonymouz666 | WTF? |
19:19.27 | *** join/#asterisk tzafrir_laptop (n=tzafrir@bzq-88-155-97-204.red.bezeqint.net) |
19:19.59 | aydiosmio | oh my lord. verizon biz just sent me 150 pages of test procedures for certification |
19:21.16 | *** join/#asterisk Gunnar (n=gunnar@nat.sigmasoft.com) |
19:22.03 | *** join/#asterisk |dennis| (n=dennis@200.32.233.82) |
19:22.13 | sahafeez | question, and yes i googled but it is a common term. if i have a CRM like in this case Act! and Asterisk with Polycom IP phones, is there anyway to get the phones to dial via the Act! applicaiton? |
19:23.00 | sahafeez | i see the TAPI stuff but i am trying to figure how it dials the phone on the desk |
19:24.17 | Mercestes | sahafeez, google asttabi |
19:24.23 | Mercestes | s/asttabi/asttapi/ |
19:24.45 | aydiosmio | you have the option of dropping a call file into asterisk via some api or netowrk file system and putting the two lines into a conference or connect them in a sepearate context |
19:25.04 | markit | necromcr: mmm what is the difference? I would like the caller to think is busy |
19:25.04 | TAMIKAMI01 | somebody speak spanish???? |
19:25.16 | Mercestes | markit, match the callerID and feed them Congestion() with no Answer() |
19:25.30 | aydiosmio | Mercestes: this looks neat, I may have to integrate this into our SRM |
19:25.31 | aydiosmio | CRM |
19:25.44 | necromcr | markit: http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+Busy ? |
19:25.51 | MACscr | can someone remind me of the name of the package that includes debian, asterisk, and freepbx? |
19:26.04 | sahafeez | thanks. reading now |
19:26.29 | *** join/#asterisk Moobius (i=Moobius@www2.techcavalry.com) |
19:26.46 | Mercestes | TAMIKAMI01, Hablo un poco de spainish pero el pequeño muchacho de I habla el asno roto muy pequeño que viola el sintaxis bueno con la lengua no. |
19:26.48 | aydiosmio | MACscr: you mean CentOS, asterisk and freepbx, trixbox? |
19:27.04 | markit | Mercestes:, necromcr thanks a lot, I will try/have a look |
19:27.06 | Mercestes | aydiosmio, pretty sweet. |
19:27.13 | MACscr | no, its actually debian and asterisk, it might include freepbx |
19:27.17 | MACscr | i think it starts with an x |
19:27.38 | aydiosmio | too bad our office phones are POTS with some propreitary PBX, or I could use it to ring our extentions here |
19:28.24 | Mercestes | TAMIKAMI01, EstarÃa alegre comer lo más humildemente posible a tus niños asados en un mirador con las flores en asterisco. |
19:28.29 | markit | I'm trying to replicate the behaviour of my old isdn pbx... when someone calls, 3 phone ring, but if the "main" phone is busy, then no other phones rings and the caller gets a "busy" signal |
19:28.38 | aydiosmio | Xorcom Rapid is a Debian/Asterisk distribution program that features an auto-install for |
19:28.45 | aydiosmio | MACscr: that it? |
19:28.45 | *** join/#asterisk fbffff (n=fbffff@r74-193-79-81.pfvlcmta01.grtntx.tl.dh.suddenlink.net) |
19:29.00 | Mercestes | markit: but.. Asterisk can do so much better than that. |
19:29.03 | MACscr | aydiosmio: yep, taht was it, thanks |
19:29.09 | markit | Mercestes: like? |
19:29.30 | Mercestes | markit: voicemail. Queues. Messages that go "please continue to hold, or press * to leave a message and we will return your call." |
19:29.43 | *** join/#asterisk frenzy (n=frenzy@unaffiliated/frenzy) |
19:29.43 | Mercestes | markit: Call forwarding. Telepathy. etc. |
19:29.55 | *** join/#asterisk brodiem (n=brodiem@67.110.68.66.ptr.us.xo.net) |
19:29.56 | frenzy | Hi |
19:29.56 | aydiosmio | lolIVR |
19:30.00 | markit | Mercestes: LOL :) queques don't need "agents" and registration? |
19:30.22 | frenzy | for somereason i cannot register from one * to another * |
19:30.23 | frenzy | using IAX |
19:30.23 | frenzy | <PROTECTED> |
19:30.23 | Mercestes | markit: only if you want them to |
19:30.23 | frenzy | <PROTECTED> |
19:30.23 | frenzy | <PROTECTED> |
19:30.29 | frenzy | Tx-Frame Retry[001] -- OSeqno: 000 ISeqno: 001 Type: IAX Subclass: REGAUTH |
19:31.00 | tzafrir_laptop | MACscr, looking for something? |
19:31.03 | drako | hey |
19:31.12 | drako | can I use zaptel 1.4 with asterisk 1.2 ? |
19:31.22 | tzafrir_laptop | drako, yes |
19:31.37 | markit | Mercestes: in any case, I think that the simple "busy" works very good for my business, since seems hard to replicate... I've a isdn bri, 1 port, so a "line" is available even if the "main phone" is used, and the caller does not find busy |
19:31.52 | MACscr | tzafrir_laptop: it was xorcom, i couldnt think of the name, but aydiosmio helped me |
19:31.58 | drako | tzafrir, i still have the problem with freezing the linux with the echocancelation |
19:32.23 | tzafrir_laptop | anybody here actually managed to get a working gtalk connection? |
19:32.24 | markit | Mercestes: in short, I would like to have the possibility of 2 calls outgoing, or 2 coming, but have to be answered only if the "main" phone (my secretary) is free |
19:32.25 | MACscr | i really dont like rpath at all, so im looking into new options |
19:32.29 | Mercestes | markit: congestion() should work then. I think we also have a busy(). |
19:32.40 | MACscr | man, i cant stand rpath |
19:32.48 | Mercestes | markit: And a chanisavail( as well |
19:33.00 | TAMIKAMI01 | jajajaj |
19:33.28 | TAMIKAMI01 | verdaderamente no se que es lo quieresw decir Mercestes!!!! |
19:33.43 | *** join/#asterisk jm|laptop (n=jm|home@dilbert.jamiem.com) |
19:33.49 | markit | Mercestes: yes, think is the way to go, thanks a lot. Wondering if will be possible to configure a similar setup with asterisk-gui in the future |
19:34.04 | Mercestes | markit: I hope not. |
19:34.33 | tzafrir_laptop | MACscr, Debian Unstable has pretty up-to-date asterisk 1.2 packages . 1.4 is now in experimental |
19:34.58 | tzafrir_laptop | And Etch backport will follow soon |
19:35.01 | Mercestes | TAMIKAMI01, Mercestes es grande y bueno y todo que es santo. La genuflexión abajo y adora lo y supplicate alegre, porque él es bueno y bueno y tu salvación. darte tus vÃrgenes en sacrificio de tu ayuda libre del asterisco. |
19:35.05 | markit | Mercestes: "hope not"? |
19:35.11 | Mercestes | markit: I hate GUIs |
19:35.22 | frenzy | ? |
19:35.32 | TAMIKAMI01 | JAJAJAJA |
19:35.46 | TAMIKAMI01 | man u r mad!!!! |
19:35.46 | TAMIKAMI01 | jajajaja |
19:35.51 | markit | Mercestes: lol, often are good, depends from the gui design... some are terrible in usage even if easy to "click" |
19:35.52 | tzafrir_laptop | TAMIKAMI01, Mercestes , English, please |
19:35.53 | cpm | s s settle down, beavis |
19:36.08 | Mercestes | tzafrir_laptop, hehe, yessir. |
19:36.40 | Mercestes | markit: They stifle creativitiy and problem solving ability\ |
19:37.59 | markit | Mercestes: sometime it saves a lot of timewaste too |
19:38.14 | drako | TAMIKAMI01, yes |
19:38.18 | drako | TAMIKAMI01, I do. |
19:38.46 | markit | btw, all-circuits-busy-now seems not part of the "basic" sound package... isn't strange? |
19:38.53 | TAMIKAMI01 | i have a question why when i compile zaptel somes commands like zttool aren't compiled? i'm also try with make zttool but appear only the error " make: *** [zttool.o] Error 1 " |
19:38.54 | drako | TAMIKAMI01, #asterisk-es for spanish. |
19:39.07 | *** join/#asterisk Assid (n=assid@59.183.18.232) |
19:39.22 | TAMIKAMI01 | tnks drako! |
19:39.24 | *** part/#asterisk TAMIKAMI01 (n=radamant@200.34.113.90) |
19:39.37 | Mercestes | markit: congestion() |
19:39.37 | drako | too bad i don't have the answer |
19:39.58 | Mercestes | asterisk-es doesn't even exist does it? |
19:40.37 | drako | Mercestes, it does, seem like im the only active tho... |
19:41.12 | *** join/#asterisk pigpen (n=mark@fw.seamans.cc) |
19:44.56 | frenzy | IAX2 registration failing http://www.pastebin.ca/410964 |
19:46.48 | Mercestes | drako: lol. Sorry about the strange spainish then.. ;)\ |
19:46.53 | *** join/#asterisk Waverly360 (n=irc@209.12.249.243) |
19:47.13 | tzafrir_laptop | drako, hint: with such a make error, look at the real error a bit above the error from make |
19:47.15 | drako | Mercestes, don't worry , it was fun anyway |
19:48.08 | Mercestes | drako: yea, english to spanish translators are a blast. |
19:48.40 | tzafrir_laptop | ERROR[16033]: chan_gtalk.c:1649 gtalk_create_member: No Connection or Username! |
19:49.28 | tzafrir_laptop | After looking at chan_gtalk.c and res_jabber.c I still fail to understand what I need to put in th config file(s?) . |
19:51.39 | *** join/#asterisk Ch0Hag (n=mking@87.127.170.250) |
19:53.22 | *** join/#asterisk klasstek (n=nunyobiz@ip67-154-143-190.z143-154-67.customer.algx.net) |
19:53.42 | frenzy | ? |
19:54.13 | klasstek | anyone used the tc400b yet? |
19:54.16 | markit | Mercestes: do you use chanisavail yourself? Seems I'm not able to make it work, but maybe is my foult |
19:54.23 | *** join/#asterisk groogs (n=greg@d38-54-164.commercial1.cgocable.net) |
19:54.47 | JunK-Y | tzafrir_laptop: want an example? |
19:54.57 | tzafrir_laptop | yes |
19:55.23 | Mercestes | markit: It failed miserably for me. But I hear it's supposed to work. |
19:55.31 | Mercestes | I'musing asterisk 1.2.13 tho so you may have much better luck |
19:56.21 | tzafrir_laptop | JunK-Y, I see all sorts of configurations. Including a non-working one in the sample configus |
19:57.24 | JunK-Y | mine is working. |
19:57.46 | JunK-Y | ya will see ur jabber user log in after. |
19:58.01 | *** join/#asterisk Katty (n=Katty@hera.copi-rite.com) |
19:58.13 | Katty | who's in charge of the person making calls for cluecon |
19:58.25 | Katty | i'm about to wring their neck for interupting my warcrafting :P |
19:58.41 | JunK-Y | tzafrir_laptop: sent |
19:58.43 | Katty | well, not really. but it sounded like a good excuse. |
19:58.47 | JunK-Y | Katty: dunno, ive got a call too. |
19:59.27 | Mercestes | hey katty.\ |
19:59.48 | *** join/#asterisk Peri (n=redanti@xtreme-44-7.dyn.aci.on.ca) |
19:59.49 | Katty | JunK-Y: did they speak french? |
19:59.52 | JunK-Y | tzafrir_laptop: ya can add this user if u want to see. |
19:59.56 | JunK-Y | Katty: hell no. |
19:59.59 | Katty | JunK-Y: or did they have a heck of a time with your accent? ;) |
20:00.06 | JunK-Y | i had to speak like a frog in english :) |
20:00.15 | Katty | file: why didn't you call me instead of random stranger?! |
20:00.25 | file | Katty: ummm I still can! |
20:00.25 | Katty | file: that's bad marketing. |
20:00.27 | JunK-Y | apparently he understood what i said, so i guess im not too bad :) |
20:00.36 | Katty | JunK-Y: yeah, you're pretty good junky |
20:00.36 | JunK-Y | even file understands my english! |
20:00.50 | Katty | JunK-Y: and if i can't understand you, i'll just make funny faces at you :P |
20:01.01 | Katty | JunK-Y: or drag out emacs. |
20:02.38 | file | that's sad |
20:02.43 | JunK-Y | virtual accent are much better! |
20:02.45 | Katty | file: did you hang up on me?! |
20:02.53 | file | Katty: no ;( |
20:02.57 | file | my internet is sad |
20:03.14 | file | I talked to Katty on the phone! she's a real person! |
20:03.23 | anonymouz666 | haha |
20:03.26 | Katty | well of course i'm real |
20:03.28 | Mercestes | her voice is hot |
20:03.30 | Katty | you've met me before, you silly rabbit |
20:03.37 | file | Katty: it could have been a robot... |
20:03.38 | anonymouz666 | first time file is talking with a woman :D:D:D |
20:03.40 | Strom_C | file: is she really? or does she sound suspiciously like Allison Smith? |
20:03.48 | file | she sounded like... Katty |
20:03.57 | Katty | i have a unique voice. |
20:04.00 | Katty | i sound like i'm 10 heh |
20:04.08 | Mercestes | it's hot |
20:04.37 | file | Katty: put down the chainsaw and listen to me... it's time for us to join in on the fight |
20:04.50 | Katty | mew? |
20:04.50 | Katty | you do not parse |
20:05.02 | file | stick your head in the microwave and give yourself a tan! |
20:05.15 | tzafrir_laptop | JunK-Y, thanks. But that is jabber.conf. What about gtalk.conf? |
20:05.43 | tzafrir_laptop | Katty, rumour has it that you managed to configure gtalk with Asterisk |
20:06.20 | *** join/#asterisk ToyMan (n=Stuart@user-12lcqol.cable.mindspring.com) |
20:07.16 | tzafrir_laptop | ERROR[21924]: chan_gtalk.c:1649 gtalk_create_member: No Connection or Username! |
20:08.00 | tzafrir_laptop | Do I need to point from an entry in gtalk.conf to that "connection" in jabber.conf? |
20:08.16 | JunK-Y | tzafrir_laptop: i just sent ya my gtalk.conf |
20:08.25 | Katty | tzafrir_laptop: rumor has it? |
20:08.38 | Katty | tzafrir_laptop: i dunno where you get your rumers...but you need to put my name back in the bag and draw another one. |
20:08.54 | robin_sz | eek ... asterisk, gtalk, jabber .. all we need to do knwo is link it up to Skype and the whole thng willl begin to hummmmmm |
20:09.06 | Katty | tzafrir_laptop: more like i setup wildfire, a jabber server, and i'm still struggling with centericq connection problems. |
20:09.48 | robin_sz | Skype to gtalk bridges ... now that would be scary |
20:10.52 | adam_vollrath | <PROTECTED> |
20:13.31 | vader-- | do you guys know if asterisk 1.2.7.1 is affected by the Daylight savings time changes in the US for 2007? |
20:14.01 | Cybertoy | that's controlled by the operating system and not asterisk |
20:14.07 | Mercestes | vader--, Shouldn't be. It's a kernel-time / device time thing.\ |
20:14.22 | Igbothom_III | vader--, Linux is not Windows! |
20:16.07 | Corydon-w | Asterisk uses whatever timezone file exists in the OS |
20:16.19 | Corydon-w | specifically, in /usr/share/zoneinfo |
20:16.47 | JunK-Y | tzafrir_laptop: are ya okay with these configs now? |
20:17.01 | JunK-Y | ya should see it as connected after this. |
20:17.02 | tzafrir_laptop | yeah, working, thanks |
20:17.15 | Mercestes | wiki it. |
20:17.17 | JunK-Y | if ya can get audio workings behind 2 nats, let me know. |
20:17.21 | Mercestes | please. :)\ |
20:17.49 | JunK-Y | tzafrir_laptop: feel free to wiki it, but hide few of my stuff please :) |
20:18.46 | *** join/#asterisk |dennis| (n=dennis@200.32.233.82) |
20:19.04 | *** join/#asterisk tsurko (n=tsurko@77.70.24.142) |
20:22.05 | *** join/#asterisk PupenoR (n=pupeno@200.123.183.91) |
20:23.16 | *** join/#asterisk zotz (n=zotz@24.244.163.157) |
20:23.35 | *** join/#asterisk rdb_ (n=rdb@gw.avila.edu) |
20:23.53 | PupenoR | Is there any dialplan function/application that would check if a variable exists or not? |
20:24.01 | JunK-Y | PupenoR: EXISTS |
20:25.55 | Corydon-w | except that EXISTS really doesn't do anything more than just LEN() |
20:26.29 | *** join/#asterisk justdave (n=dave@unaffiliated/justdave) |
20:30.37 | ManxPower | Can a varliable exist with a 0 length? |
20:31.21 | JunK-Y | i guess, "" is 0 |
20:31.22 | adam_vollrath | What is the sound of one hand clapping? |
20:31.25 | Juggie | could god microwave a burrito so hot, he himself could not eat it. |
20:31.43 | JunK-Y | Juggie: no more ski this winter i think :( |
20:31.55 | Juggie | JunK-Y, i know, skiing is over, next year we go to lemassif! |
20:32.20 | JunK-Y | sure, let me know when, im in! |
20:32.34 | JunK-Y | i bring julie last week-end, she did snowboarding, god, this was terrible! |
20:32.35 | JunK-Y | heheh |
20:32.48 | Juggie | haha. |
20:32.56 | Juggie | i hate snowboarders |
20:32.58 | *** join/#asterisk Dr-Linux|work (n=asfdf@DSL-202-59-73-131.nexlinx.net.pk) |
20:33.03 | ManxPower | It seems like playing in traffic would be just as effective and less work than skiing |
20:33.03 | Juggie | allways have to wait for them |
20:33.14 | JunK-Y | thats why im doing mini-skiing ! |
20:33.21 | *** join/#asterisk _Vile (n=vile@bc182112.bendcable.com) |
20:33.23 | Juggie | skiing with snowboarders is a constant waiting game |
20:33.40 | JunK-Y | Juggie: have ya ever been to bromont? |
20:33.49 | Juggie | no |
20:34.02 | Juggie | i've only been to ski hills around ottawa, one in newfoundland, and tremblant. |
20:34.10 | Juggie | i dont really like tremblant. |
20:34.16 | Juggie | nice skiing but too many people. |
20:34.26 | JunK-Y | we have to go to bromont, it rocks |
20:34.35 | JunK-Y | and btw, we plans to go to ottawa in like 1month |
20:34.47 | Juggie | what for? |
20:34.57 | JunK-Y | visit unlimitel. |
20:35.03 | Juggie | who is we? |
20:35.29 | JunK-Y | me, julie, my friend and his gf. |
20:36.07 | Juggie | the guy i met @ astricon05? |
20:36.23 | JunK-Y | no, hes my ex-coworker. |
20:36.30 | *** join/#asterisk reber (n=reber@cl-158.dub-01.ie.sixxs.net) |
20:36.45 | *** join/#asterisk brea (n=rollergr@c-71-195-248-169.hsd1.ma.comcast.net) |
20:36.58 | Juggie | cool, let me know when you have firm plans. |
20:37.21 | JunK-Y | email me ur cell phone. |
20:37.41 | e-milio | hello all |
20:37.50 | *** join/#asterisk bkuhn (n=bkuhn@fsf/member/bkuhn/bkuhn) |
20:37.57 | *** join/#asterisk aaronr (n=arussell@87.127.234.100) |
20:38.04 | e-milio | I have this situacion: channel.c:2379 set_format: Unable to find a codec translation path from g729 to gsm |
20:38.24 | *** part/#asterisk frenzy (n=frenzy@unaffiliated/frenzy) |
20:38.35 | justdave | we're having problems with people with noisy phone lines on conference calls in meetme... 20 or 30 people diaed in, and it's an ineractive meeting, so administratively muting people at random would be disruptive. Is there any way to get a running "vu-meter" type thing for each person dialed into a conference to see who's got the noise? |
20:38.58 | Juggie | e-milio, in the configuration for your end device in asterisk you must of did allow=all or allow=g729 yet you do not have a g729 license. |
20:39.14 | Juggie | asterisk only supports g729 passthrough unless you buy a license. |
20:39.27 | ManxPower | justdave: you can use ztmonitor on each of the zap ports that people are dialed into then manually mute them. No, there is nothing to automatically do this |
20:39.46 | ManxPower | and g729 passthru is almost useless for most people |
20:39.52 | ManxPower | em see this: |
20:39.54 | ManxPower | ~codec |
20:40.05 | ManxPower | ~codecs |
20:40.07 | jbot | from memory, codecs is http://snipurl.com/wiki_codecs. If you have audio/codec problems, first try to 'disallow=all' and 'allow=ulaw' and see if that works. Anyone that tells you to use 'allow=all' is an idiot as it usually causes audio problems, or Number/Name: 1/g723, 2/gsm, 4/ulaw, 8/alaw, 16/g726, 32/adpcm, 64/slin, 128/lpc10, 256/g729, 512/speex, 1024/ilibc |
20:40.08 | Juggie | ManxPower, jbot is lagged. |
20:40.11 | Juggie | there we go |
20:41.03 | justdave | my idea for a dirty hack at this point is to set up the conference system to turn on monitoring as each user enters the conference, and have something tail the generated wav files |
20:41.12 | e-milio | ManxPower: after g729 which is the next most recommended codec ? |
20:41.13 | justdave | most of the people dialed in are on SIP or IAX |
20:41.31 | e-milio | I am doing IAX trunking between 2 servers |
20:41.36 | ManxPower | e-milio: what in the WORLD makes yo think G729 is a recommended codec? |
20:41.43 | ManxPower | justdave: there are no options for sip/iax |
20:42.13 | ManxPower | e-milio: what codec is "best for you" depends on many things, bandwidth, CPU, and what your end points support. |
20:42.31 | e-milio | ManxPower: I was under the impression that is was good to save bandwidth ? |
20:42.37 | ManxPower | The "best" codecs are ulaw and alaw, but they take up a lot of bandwidth |
20:43.05 | ManxPower | ilbc and Speex and G729 sound pretty good, don't take up a lot of bandwidth, but require a fair amount of CPI |
20:43.12 | ManxPower | CPU too |
20:44.46 | e-milio | ManxPower: The will be a lot conferencing going on, and recording (due to vicidial) |
20:45.03 | e-milio | In that sense ilibc will be still recommended ? |
20:47.08 | Hmmhesays | this chanskype pos sucks |
20:47.48 | *** join/#asterisk dj-fu (n=ajc@unaffiliated/dj-fu) |
20:47.56 | aydiosmio | Hmmhesays: I could have told you that |
20:49.34 | ManxPower | chanskype does Point of Sale?? |
20:50.32 | Nugget | heh |
20:51.26 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
20:54.02 | syzygyBSD | I am playing around with realtime asterisk... I like it |
20:54.31 | syzygyBSD | how well does it scale? |
20:58.02 | *** join/#asterisk Fieldy (i=5rohni9q@gentoo/contributor/Fieldy) |
20:58.54 | *** join/#asterisk tkowal (n=nospamto@74.93.82.14) |
20:59.01 | *** join/#asterisk Vec (n=Vec@dsl-241-206-133.telkomadsl.co.za) |
21:02.07 | anonymouz666 | exten => s,n,While($[${j} <= ${ARG1}]) and J++ in a macro... how could I set ${j} to be another extension in another context... [another] ${j},n,blah() |
21:02.21 | anonymouz666 | while running the loop i am building another extensions in another context...based on the j result |
21:02.37 | *** join/#asterisk |dennis| (n=dennis@200.32.233.82) |
21:04.38 | syzygyBSD | whats the program to create a script of all the commands you type? |
21:05.14 | bulle | Hmmhesays: why does it suck ? |
21:05.52 | anonymouz666 | I am stuck :) |
21:05.57 | ManxPower | bulle: other than requieing you to have skype installed on a PC running windows? |
21:12.43 | *** join/#asterisk Tclp (n=tcalp@S01060014bf0ffd47.ed.shawcable.net) |
21:12.56 | bulle | ManxPower: he, thats enough |
21:13.34 | Tclp | hey all, what would you guys recomend for multi-line voip phones ? (am using Axon for windows as my virtual PBX) |
21:13.36 | *** join/#asterisk Mercestes (n=Merceste@rrcs-71-41-157-70.sw.biz.rr.com) |
21:14.12 | Tclp | I don't want something horribly expensive, but good build quality |
21:15.11 | Vec | Does anyone know what could possibly cause asterisk to get stuck on the line -- Called 2101, the line -- SIP/2101-00832680 is ringing does not appear? I restarted asterisk and it sorted it out. |
21:17.31 | *** join/#asterisk zogulus (n=zogulus@58.98.adsl.brightview.com) |
21:17.43 | koel | already tried set verbose 10? |
21:17.55 | e-milio | ManxPower: I switched to ilbc, ulaw and asterisk seems to be taking 66% cpu time on a p4/512. Seems reasonable ? |
21:20.01 | syzygyBSD | are there known issues with queues and realtime_mysql |
21:22.46 | *** join/#asterisk voltagex (n=voltagex@124-254-100-201-dsl.ispone.net.au) |
21:23.08 | *** join/#asterisk jubei_ (n=Stormtro@147.27.47.114) |
21:23.50 | voltagex | hi, I'm not receiving any audio from my calls. Connecting directly Softphone->FWD works, but Asterisk->Softphone->FWD doesn't |
21:23.59 | jubei_ | anybody know of a *** Warning: "zt_register" compile error with bristuff ? it seems to be all over the internet, only in German :) |
21:24.03 | *** join/#asterisk dwmw2_gone (i=ctrlprox@81.187.2.161) |
21:24.15 | voltagex | of course that should be, Softphone->Asterisk->FWD |
21:24.19 | *** join/#asterisk MinotaurUK (n=minotaur@89-145-201-162.xdsl.murphx.net) |
21:24.35 | MinotaurUK | greetings all |
21:24.57 | MinotaurUK | would some kind soul mind giving me a hand with a chan_sip issue for a few minutes? |
21:25.40 | bulle | voltagex: best bet is that it has to do with routing and rtp |
21:26.23 | voltagex | bulle: If the softphone is in the DMZ connecting directly to FWD, it works, if the Asterisk box is in the DMZ it doesn't |
21:26.54 | *** join/#asterisk bmd (n=bmd@72.54.252.34) |
21:27.20 | bulle | voltagex: dmz doesnt mean that the asterisk box gets a public ip, so you will have to setup the nat part of sip.conf properly ( i assume yoru softphone uses sip ) |
21:27.42 | Mercestes | MinotaurUK, only if you tell us what the chan_sip issue is. |
21:27.43 | bulle | voltagex: then the best bet is to force both sip and rtp trough the asterisk box |
21:27.57 | *** join/#asterisk [TK]D-Fender (n=Joe@64.235.216.2) |
21:28.03 | voltagex | bulle: how do I do that? |
21:28.27 | bulle | voltagex: canreinvite=no is the option i think, check sip.conf |
21:29.02 | *** join/#asterisk lunaphyte (n=lunaphyt@static-71-120-128-10.gdrpmi.dsl-w.verizon.net) |
21:29.04 | MinotaurUK | Mercestes: thanks. About 6 hours ago I had to restart an asterisk box that's been running for a good few months without any issues. Upon restarting, asterisk segfaults. I've done the usual verbose startup to the console, and the last line before the segfault is "parsing sip.conf" |
21:29.14 | voltagex | bulle: in the FreeWorldDialup section or the phone section? |
21:29.29 | bulle | voltagex: phone section |
21:29.35 | voltagex | bulle: because canreinvite=no is already set in FWD |
21:29.42 | MinotaurUK | Mercestes: If I remove "host=dynamic" from all entries in sip.conf, asterisk loads correctly, but of course no sip peers can register |
21:31.06 | Mercestes | MinotaurUK, polycom endpoints? |
21:31.12 | MinotaurUK | Snoms |
21:31.29 | Mercestes | onlytime I've seen that is when a polycom phone freaked out on me with corrupted firmware. |
21:31.32 | MinotaurUK | all local, on same subnet as asterisk box, no nat involved |
21:31.49 | Mercestes | try loading up asterisk isolated from the endpoints with dynamic=yes and then start bringing the peers online one at a time and see what happens. |
21:32.37 | MinotaurUK | will login to the switches and disable the ports, back in a couple mins |
21:34.22 | MinotaurUK | as a test, I've just tried it with one phone in sip.conf and it's still doing it, so even with only 1 host=dynamic entry in sip.conf, asterisk is still segfaulting |
21:34.35 | Mercestes | what ver of asterisk? |
21:34.40 | MinotaurUK | 1.2.13 |
21:34.45 | Mercestes | that[s the ver I run |
21:35.33 | Mercestes | have yo urestarted the box yet? |
21:35.48 | [TK]D-Fender | MinotaurUK, Take 1 entry, change the type from "friend" to "user", then reboot the phone fter restarting * |
21:35.58 | MinotaurUK | bringing the sip.conf over to another box and reloading asterisk works fine |
21:36.05 | [TK]D-Fender | MinotaurUK, See if you can PLACE a call seperate from registering |
21:36.06 | MinotaurUK | D-Fender: will go try that, thanks |
21:36.17 | voltagex | where are contexts defined again? :/ |
21:36.30 | koel | extensions.conf... |
21:36.30 | [TK]D-Fender | voltagex, ..... |
21:36.31 | [TK]D-Fender | ~book |
21:36.44 | jbot | rumour has it, book is a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
21:36.45 | koel | find /etc/asterisk | xargs grep content |
21:36.50 | koel | find /etc/asterisk | xargs grep context |
21:36.58 | voltagex | I must have spelt my context wrong then |
21:37.45 | MinotaurUK | D-Fender: that seems to be okay |
21:38.08 | *** join/#asterisk l3jj (n=chatzill@dsl093-001-038.det1.dsl.speakeasy.net) |
21:38.26 | *** join/#asterisk PakiPenguin (n=uppal@linuxpakistan/admin/pakipenguin) |
21:38.27 | [TK]D-Fender | MinotaurUK, Do soft-ophones crash * as well? |
21:38.31 | PakiPenguin | hi |
21:38.38 | Tclp | are the Grandstream 2020's going to be much better then the 2000's ? ... it seems like a great time to pick-up some 2000 series models -- I guess they are on clearance ? seeing them for like $75us ? |
21:38.54 | Mercestes | Tclp: there is a reason for that. |
21:39.39 | *** join/#asterisk kgx0 (n=karuna@60.234.196.160) |
21:39.40 | [TK]D-Fender | Mercestes, Lol... looks like a Polycom 601 rip-off :) |
21:39.47 | [TK]D-Fender | http://www.grandstream.com/gxp2020.html |
21:39.59 | [TK]D-Fender | Tclp, ... |
21:40.00 | [TK]D-Fender | ~gs |
21:40.01 | jbot | rumour has it, gs is South Georgia and the South Sandwich islands, or ghostscript. GrandSuck phones are cheap junk which should be avoided with extreme prejudice |
21:40.01 | MinotaurUK | D-Fender: yes, seems any host=dynamic entry (if "firiend") is killing asterisk - have tried sip.conf with entries for cisco, snom and x-lite (individually of course) |
21:40.40 | [TK]D-Fender | MinotaurUK, I'm betting the register itself is nuking *. |
21:40.45 | Tclp | D-Fender ... I'm not big on telephony technology ? .. why is the 2020 so much better ? |
21:40.48 | Vec | I have a problem where the Dial cmd is (sometimes) still indicating rining to the person who dialed the call even after the call has been answered, it seems it only happens when dialing from a SIP phone to an IAX phone/trunk, any ideas ? |
21:40.54 | voltagex | bulle: canreinvite=no didn't change anything |
21:40.57 | [TK]D-Fender | MinotaurUK, Hate to say it but upgrade you * to the latest and try again |
21:41.50 | Mercestes | or reinstall your current version if yoru really in love with it. |
21:41.56 | Tclp | I mean aside from having support for some extra lines and a nicer display ? ... is there some core feature that the 2020 has over the 2000 ? |
21:41.58 | voltagex | people convinced me to dump Trixbox but now I can't get anything working :/ |
21:42.43 | MinotaurUK | D-Fender: any way to manually clear the registry? |
21:42.52 | bulle | voltagex: and you have put your public ip on the asterisk box, and forwarded sip and rtp ports to the asterisk box from the router ? |
21:43.02 | [TK]D-Fender | Tclp, How about you go to GS's website and grab the datasheet & compaison charts and judge for yourself? |
21:43.38 | [TK]D-Fender | MinotaurUK, with * closed, rename /var/lib/asterisk/astdb to something else and restart. HEY! Maybe thats it |
21:43.46 | [TK]D-Fender | are you running * as root? |
21:44.00 | *** join/#asterisk techie (n=gus@voip.routedsystems.com) |
21:44.07 | sahafeez | what is the state of the zaptel support in freebsd? is it production level yet? |
21:44.12 | voltagex | bulle: asterisk box is in the dmz, externip is set in sip.conf |
21:44.12 | Tclp | D-Fender .. was more curious in the sense that .. does the 2020 have some new feature that makes the 2000 outdated ... eg. if I buy 2000's for a small office will I regret it for a lack of some new technology support ? |
21:44.12 | MinotaurUK | I am at the moment whilst testing, usually runs as asterisk/asterisk |
21:44.46 | bulle | voltagex: then i would use wireshark or similar, running on the asterisk box, to record and analyze the trafic |
21:45.02 | *** join/#asterisk CrazyTux[m] (n=CrazyTux@ppp-70-255-140-128.dsl.hstntx.swbell.net) |
21:45.19 | bulle | Tclp: how should [TK]D-Fender know if you will be missing some feature ? |
21:45.21 | Mercestes | MinotaurUK, Some jerk-wad could have run it as root and writeprotected all your files |
21:45.24 | *** join/#asterisk |dennis| (n=dennis@200.32.233.82) |
21:45.46 | *** join/#asterisk mmartinn (n=martin@adsl-065-005-200-225.sip.gnv.bellsouth.net) |
21:45.57 | [TK]D-Fender | MinotaurUK, I can imaging that perhaps when the phone reg's it tries to write the values to astdb and without authority crashes completely... |
21:46.15 | *** part/#asterisk mog (n=mog@c-71-207-200-130.hsd1.al.comcast.net) |
21:46.28 | voltagex | bulle: for 1.4, is the command externip or externalip? |
21:46.41 | [TK]D-Fender | Tclp, Maybe I wasn't clear : GO COMPARE YOURSELF ON THEIR SITE. http://www.grandstream.com |
21:47.09 | bulle | voltagex: the sip.conf has all the examples in it |
21:47.11 | bulle | voltagex: just check there |
21:47.20 | [TK]D-Fender | </bile> |
21:47.54 | Mercestes | Tclp: If it's a grandstream, it will absolutely lack some technology that you will miss later. Namely voip functionality. |
21:48.11 | Mercestes | Tclp: Actuallly, now that I think of it, it will lack overall telephony functionality. |
21:48.18 | brea | Does anyone have any recommendations for an inexpensive SBC? |
21:48.28 | Mercestes | brea: AT&T |
21:49.02 | brea | err :p |
21:49.15 | brea | inexpensive Session Border Controller |
21:49.59 | [TK]D-Fender | brea, Broadsoft :D |
21:50.04 | brea | something that works as a b2bua |
21:50.18 | [TK]D-Fender | brea, Asterisk? :) |
21:50.48 | brea | and supports g729ab with cRTP ;) |
21:51.00 | Mercestes | brea: openser? |
21:51.11 | [TK]D-Fender | brea, Would you like fries with that, sir? |
21:51.29 | brea | har har |
21:51.50 | brea | Mercestes: Didn't know openser supported cRTP |
21:52.00 | Mercestes | brea: Ask clona |
21:52.09 | Tclp | Mercestes .. what would you recomend in that case ? |
21:52.19 | Mercestes | Tclp: Polycom. |
21:52.31 | Tclp | k thnx |
21:55.50 | *** join/#asterisk |dennis| (n=dennis@200.32.233.82) |
21:57.04 | *** join/#asterisk sashion (n=djbdsf@dsl-241-213-43.telkomadsl.co.za) |
21:57.45 | sashion | Anyone run into the following issue? RTCP SR transmission error |
21:58.58 | Vec | I have a problem where the Dial cmd is (sometimes) still indicating rining to the person who dialed the call even after the call has been answered, it seems it only happens when dialing from a SIP phone to an IAX phone/trunk, any ideas ? |
22:01.29 | *** part/#asterisk MarkWD (n=MarkWD@rrcs-67-78-88-186.sw.biz.rr.com) |
22:01.48 | *** join/#asterisk tsurko (n=tsurko@77.70.24.142) |
22:02.37 | *** part/#asterisk brodiem (n=brodiem@67.110.68.66.ptr.us.xo.net) |
22:06.11 | *** join/#asterisk jaxxan (n=jaxxan@202.70.125.109) |
22:06.29 | jaxxan | ~video |
22:06.30 | jbot | methinks video is interrupt 10h for the video BIOS |
22:06.39 | jaxxan | does asterisk support video ? |
22:06.45 | koel | jups |
22:06.46 | koel | ;) |
22:06.51 | jaxxan | sweet |
22:07.04 | jaxxan | gotta preferred link? |
22:07.21 | koel | voip wiki |
22:07.32 | jaxxan | yeah that's where i was heading (= |
22:08.18 | koel | http://revision3.com/systm/asterisk |
22:08.31 | koel | nice video |
22:08.33 | koel | ;) |
22:11.08 | *** join/#asterisk _DAW (n=chatzill@adsl-156-109-78.msy.bellsouth.net) |
22:13.52 | *** join/#asterisk PakiPenguin (n=uppal@linuxpakistan/admin/pakipenguin) |
22:16.12 | jaxxan | i'm running asterisk 1.2.16. do i still need to patch in order to use h.263p ? |
22:17.52 | ManxPower | jaxxan: 1.2 does not get new features, only bug fixes. |
22:18.21 | jaxxan | ummm |
22:18.27 | jaxxan | is that a no ? |
22:18.53 | aydiosmio | that's a yes |
22:18.57 | aydiosmio | go patch |
22:21.03 | *** join/#asterisk MooingLemur (n=troy@unaffiliated/mooinglemur) |
22:21.09 | jaxxan | i'm soooo not sure how to patch |
22:21.10 | *** join/#asterisk |dennis| (n=dennis@200.32.233.82) |
22:21.22 | *** join/#asterisk sysreq (n=sysreq@modemcable171.134-81-70.mc.videotron.ca) |
22:22.09 | *** join/#asterisk hedge77 (n=netwalke@209.42.192.202) |
22:22.35 | jaxxan | ~patch |
22:22.37 | jbot | rumour has it, patch is just a tool to handle text file mutations according to a common format known as 'diff'. For kernel patching use (depending on type of patch): 'patch -p1 < patch.diff' or 'zcat patch.gz | patch -p1' or 'bzcat patch.bz2 | patch -p1', or `patch -ruN old_dir new_dir > tree.diff` |
22:22.42 | koel | http://bibliotecnica.upc.es/PFC/arxius/migrats/40377-2.pdf |
22:24.14 | *** join/#asterisk [hC] (n=hardcore@66.119.172.82) |
22:24.19 | *** join/#asterisk PupenoR (n=pupeno@200.123.183.91) |
22:24.46 | [hC] | Anyone tried setting option 66 on a cisco IOS based router that's doing dhcp, and have a polycom ignore the setting? I cant figure out why my polycom phones arent picking it up and discovering the ftp server. |
22:26.01 | koel | hc |
22:26.23 | koel | can;t you try a dhcp debug client to ensure option 66 gets to the client? |
22:26.28 | *** join/#asterisk CVirus (n=GoD@196.205.193.189) |
22:26.30 | koel | don't you need 67 too? |
22:27.29 | koel | 67 is BootFile Name |
22:27.39 | CVirus | Is the default asterisk GUI better than FreePBX ? |
22:27.47 | koel | gui? |
22:27.50 | [TK]D-Fender | [hC], Typically that'd be because you hard-coded an IP into the BootROM |
22:28.19 | CVirus | koel: yes |
22:28.26 | [TK]D-Fender | [hC], Make sure it says 0.0.0.0 |
22:28.30 | koel | cli? |
22:29.20 | CVirus | koel: I'm talking about the official GUI |
22:29.35 | koel | didn't know there was one.. |
22:29.36 | *** join/#asterisk Discard (n=Discard@lev92-1-82-67-255-74.fbx.proxad.net) |
22:29.44 | Discard | hi |
22:30.12 | CVirus | koel: #asterisk-gui |
22:30.26 | koel | don't need one ;) |
22:30.30 | koel | but thanks.. |
22:30.37 | mcab | [hC]: make sure the polycom is set to use Option 66 and not 'Custom' or 'Static' or somesuch |
22:30.49 | *** join/#asterisk Waverly360 (n=irc@209.12.249.243) |
22:30.49 | Discard | what is the extention channel for a queue or a ring group ? |
22:30.58 | Discard | extension |
22:31.32 | *** join/#asterisk netlouis (n=netlouis@2001:b18:400d:0:280:c8ff:fe17:276) |
22:33.03 | *** join/#asterisk kiko69 (n=keith@adsl-75-16-91-106.dsl.irvnca.sbcglobal.net) |
22:33.21 | [TK]D-Fender | Discard, ... HUH!? |
22:33.32 | jaxxan | how do i patch asterisk? |
22:33.40 | jaxxan | i've never done it before |
22:33.46 | Vec | Is there more details release notes availible besides http://www.asterisk.org/node/48338 ? |
22:33.52 | koel | , patch is just a tool to handle text file mutations according to a common format known as 'diff'. For kernel patching use (depending on type of patch): 'patch -p1 < patch.diff' or 'zcat patch.gz | patch -p1' or 'bzcat patch.bz2 | patch -p1', or `patch -ruN old_dir new_dir > tree.diff` |
22:33.53 | koel | ;) |
22:34.38 | [TK]D-Fender | Vec, Go to the FTP site and check the changelog. |
22:34.39 | Discard | ok |
22:34.50 | koel | Vec: maybe changes in cvs tree |
22:34.51 | Vec | [TK]D-Fender : ta |
22:35.03 | Discard | for a SIP account I can use : ZIP/<extension> |
22:35.03 | Vec | no one has any ieas for my question earlier ? |
22:35.08 | Discard | for a zap channel |
22:35.09 | koel | SIP/ |
22:35.18 | Discard | ZAP/g0 |
22:35.35 | koel | only if you have defined a trunk... |
22:35.53 | Discard | ok and for a Queue is there anything or ring group ? |
22:36.10 | koel | can't tell you... |
22:36.19 | koel | can't you define a context? |
22:37.02 | [TK]D-Fender | Discard, Queue is just an APPLICATION, and "reig group" is not a valid * term. What is generally reffered to as such is a Dial command that jsut target multiple devices. |
22:37.02 | Discard | i'm working on a click2call script |
22:37.11 | Discard | ring |
22:37.11 | [TK]D-Fender | Discard, "show application dial" |
22:37.20 | Discard | ok |
22:38.13 | koel | discard application dial ;) |
22:38.24 | Discard | thank you |
22:38.45 | Discard | where could I find AGI reference ? |
22:39.08 | sashion | Discard: www.voip-info.org |
22:39.14 | sashion | do a seach of AGI |
22:40.03 | *** join/#asterisk techie (n=gus@voip.routedsystems.com) |
22:40.09 | [TK]D-Fender | Discard, ... |
22:40.12 | [TK]D-Fender | ~osmosis |
22:40.24 | jbot | i heard osmosis is the act of beating yourself on the head repeatedly with THE BOOK, until some measure of absorption has occured ... or at least until your unconsciousness restores peace to the channel ... |
22:40.32 | [TK]D-Fender | ~book |
22:40.35 | jbot | somebody said book was a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
22:40.49 | [TK]D-Fender | Discard, make sure to order your NOW (in hardcover) |
22:41.57 | hedge77 | i'm trying to set up a queue with dynamic members like the docs suggest, but when the first agent phone rings it goes to the voicemail for that extension instead of staying in the queue and rolling to the next one. BONK |
22:42.34 | [TK]D-Fender | hedge77, Nevere EVER let it ring an extension where VM can come into play at all. that will ANSWER the channel |
22:42.34 | *** join/#asterisk P4C0 (n=ash@200.124.22.34) |
22:43.28 | P4C0 | hello I'm having a problem: I have a asterisk server and local clients connected to it (alaw codec), my asterisk server connects to the voip provider using one license g729 codec, but I can't make calles |
22:44.14 | sashion | P4C0: what error do you get when trying to dial the voip trunk? |
22:44.30 | [TK]D-Fender | P4C0, *PASTEBIN* |
22:44.39 | P4C0 | dial_exec_full: Had to drop call because I couldn't make SIP/3-081f14f8 compatible with SIP/mysipprovider-out-081eb598 |
22:44.43 | sashion | ~pastebin |
22:44.44 | jbot | pastebin is probably a place to paste your stuff without flooding the channel - try http://pastebin.ca, or http://channels.debian.net/paste, or http://rafb.net/paste/, or http://pastebin.com is usually painfully too slow and unresponsive to use, use one of the other pastebin sites, or dpaste.com is a very nice pastebin as well |
22:45.30 | hedge77 | it happens even using a target ext with no vm defined (just errors out "no entry in voicemail config" and hangs up) |
22:45.40 | *** join/#asterisk borisyaltsin (n=chris@dhcp-43-41.arts.ualberta.ca) |
22:45.50 | *** join/#asterisk bkruse_home (n=kruz@69.73.127.92) |
22:45.56 | *** part/#asterisk mattchis (n=mattchis@216.54.143.246) |
22:46.35 | *** join/#asterisk ZX81 (n=ZX81@60-234-238-188.bitstream.orcon.net.nz) |
22:46.43 | *** join/#asterisk |dennis| (n=dennis@200.32.233.82) |
22:47.07 | [TK]D-Fender | Dear God Aastra's admin guide is HUGE |
22:47.08 | *** join/#asterisk kiwoneka (n=kiwoneka@KTNRON06-1168103823.sdsl.bell.ca) |
22:47.16 | sashion | hedge77: i imagine you are using a Local channel to call your agents back on? |
22:47.36 | ZX81 | file: #9281 patch didn't work - had a half duplex call on call 9/15 - [Mar 26 18:11:18] Sent packets: 0 |
22:47.42 | sashion | P4C0: you sure you have a licensed version of g729? |
22:47.43 | hedge77 | yes local |
22:47.48 | [TK]D-Fender | sashion, Rather obviously, yes |
22:47.53 | P4C0 | sashion, yes |
22:48.08 | borisyaltsin | so, what's the cheapest way to get some phones hooked up to my asterisk box? I'm not having any luck finding a usb-fxs product or something cheap to connect a phonet o the computer. Is it cheaper to buy some sip phones? |
22:48.19 | [TK]D-Fender | P4C0, You know you can show us debug for that broken call at any time now... |
22:48.26 | ZX81 | borisyaltsin - cheapest is a headset and free softphone |
22:48.35 | [TK]D-Fender | borisyaltsin, Linksys SPA-2102 |
22:48.37 | borisyaltsin | ok, besides that option;) |
22:48.39 | ZX81 | after that I'd say a barbie tone (BT101) |
22:48.43 | ZX81 | but they're crap |
22:48.43 | P4C0 | this is my console log: http://rafb.net/p/YYab4b39.html |
22:48.51 | hedge77 | oh god bt101's suck |
22:48.54 | ZX81 | yep |
22:49.03 | sashion | hedge77: pastebin your callback context |
22:49.11 | [TK]D-Fender | P4C0, c'mon... with SIP DEBUG ENABLED! |
22:49.24 | ZX81 | I have an SPA921, and a few weird IAX2 chinese phones on my desk, a channel bank beside it and a tdm400 card also |
22:49.36 | P4C0 | [TK]D-Fender, all pers? |
22:49.37 | [TK]D-Fender | ~gs |
22:49.38 | jbot | hmm... gs is South Georgia and the South Sandwich islands, or ghostscript. GrandSuck phones are cheap junk which should be avoided with extreme prejudice |
22:49.41 | ZX81 | the phone I use the most is actually a cordless plugged into the channel bank |
22:49.47 | *** join/#asterisk tsurko (n=tsurko@77.70.24.142) |
22:49.50 | sashion | P4C0: Just one of the peers you dialling from |
22:49.52 | [TK]D-Fender | P4C0, Enough to debug that call... |
22:50.13 | P4C0 | ok moment |
22:50.34 | [TK]D-Fender | ZX81, What CB are you using? |
22:50.43 | ZX81 | eh - access bank II |
22:50.44 | ZX81 | :) |
22:50.47 | ZX81 | crappy and cheap |
22:50.50 | ZX81 | but works fine |
22:50.51 | ZX81 | :) |
22:50.55 | P4C0 | show g729 |
22:50.55 | P4C0 | 0/0 encoders/decoders of 1 licensed channels are currently in use |
22:50.57 | ZX81 | CAC :) |
22:51.04 | [TK]D-Fender | ZX81, Any of the CID flakeyness that's been reported for others? |
22:51.12 | P4C0 | that means my g729 is installed ok, right? |
22:51.12 | ZX81 | nah working fine here |
22:51.22 | ZX81 | P4C0 yep |
22:51.30 | sashion | P4C0: yes |
22:51.31 | [TK]D-Fender | ZX81, How many phones, and what kind of working envirnmoent? |
22:51.41 | borisyaltsin | hmm. with that linksys product I wouldn't necessarily even need an asterisk box if I had a sip voip termination provider like les.net would I? |
22:51.42 | ZX81 | 5 phones - my house |
22:51.43 | ZX81 | :) |
22:52.02 | ZX81 | borisyaltsin correct but then you wouldn't be able to do leet stuff |
22:52.06 | sevard | borisyaltsin: that's right |
22:52.08 | [TK]D-Fender | ZX81, neato... pricey... (assuming you had to actually PAY for it & the digital card). |
22:52.08 | ZX81 | like ivr, conferences etc |
22:52.22 | ZX81 | the CAC was like $200 off ebay |
22:52.24 | [TK]D-Fender | borisyaltsin, Naturally no. |
22:52.41 | ZX81 | the card was from a customer who ran out of money and gave it to me :) |
22:52.53 | [TK]D-Fender | ZX81, My favourite price :) |
22:52.56 | borisyaltsin | hmm. that's interesting. I might buy one to just play around with then;) I have a one bedroom apartment right now, so maybe I'll save asterisk for a few years from now when I have a house and need to do .. leet stuff;) |
22:52.57 | ZX81 | yep :) |
22:53.18 | [TK]D-Fender | borisyaltsin, Its worth it for more than that. |
22:53.29 | hedge77 | sashion: http://pastebin.ca/411301 (if that isn't what you're looking for let me know) |
22:53.40 | [TK]D-Fender | borisyaltsin, With * you can run your own VM, have MULTIPLE providers, gain access to analog lines, etc... |
22:54.08 | borisyaltsin | hmm. I think I have a box like that from when I was using vonage a couple years ago. I wonder if I can flash the vonage off it and update it with some stock firmware.. |
22:54.57 | [TK]D-Fender | hedge77, You clearly aren't thinking straight showing us that and not the [staff] context which is used to DIAL THE PHONES. |
22:54.59 | P4C0 | http://rafb.net/p/lx4wAS33.html |
22:55.08 | hedge77 | agh hurr |
22:55.17 | [TK]D-Fender | :) |
22:55.20 | *** join/#asterisk welby (i=welby@gateway/tor/x-48f13712927ec4bb) |
22:55.33 | P4C0 | is there a way to set priority in the codecs? or to force codecs by peers? |
22:58.07 | mmartinn | P4C0: sip.conf has statements that do that, I believe |
22:59.10 | mmartinn | P4C0: Something like "disallow=all \ allow=gsm \ allow=ulaw \ allow=alaw" etc... |
22:59.56 | P4C0 | mmartinn, yes but inside the peer scope? |
23:01.19 | mmartinn | P4C0: Yes, inside each one |
23:01.42 | [TK]D-Fender | P4C0, Restrict both to alaw only and attempt again. |
23:02.00 | [TK]D-Fender | P4C0, Thogh mind you your debug tells me nothing about the reason for incompatability |
23:02.00 | P4C0 | [TK]D-Fender, with alaw only works fine... |
23:02.04 | hedge77 | ok context support { |
23:02.04 | hedge77 | <PROTECTED> |
23:02.07 | hedge77 | <PROTECTED> |
23:02.09 | hedge77 | woops |
23:02.12 | mmartinn | ~pastebin |
23:02.13 | jbot | rumour has it, pastebin is a place to paste your stuff without flooding the channel - try http://pastebin.ca, or http://channels.debian.net/paste, or http://rafb.net/paste/, or http://pastebin.com is usually painfully too slow and unresponsive to use, use one of the other pastebin sites, or dpaste.com is a very nice pastebin as well |
23:02.18 | [TK]D-Fender | unless its a G729 vs G729A thing |
23:02.37 | hedge77 | http://pastebin.ca/411309 <-just about everything |
23:04.40 | [TK]D-Fender | hedge77, Well you can clearly see that using [staff] as your context leads you to dialing extens with Voicemail. This jsut doesn't cut it. make a new context that only DIALS the device |
23:05.24 | *** join/#asterisk zmef420 (n=zmef420@metarb3-pool4-153.mtco.com) |
23:06.53 | hedge77 | another context with _1XX => Dial() that doesn't use the normal macro then? |
23:06.55 | *** join/#asterisk [hC] (n=hardcore@66.119.172.82) |
23:07.02 | [TK]D-Fender | hedge77, As I said... |
23:07.30 | *** join/#asterisk `Sauron (i=sauron@dsl001-130-033.aus1.dsl.speakeasy.net) |
23:07.49 | hedge77 | i'll give that a try tomorrow. Do I even need to fill out the members in agents.conf anymore? |
23:07.56 | *** join/#asterisk mog (n=mog@c-71-207-200-130.hsd1.al.comcast.net) |
23:07.57 | *** mode/#asterisk [+o mog] by ChanServ |
23:07.58 | *** join/#asterisk anthm (n=anthm@m815f36d0.tmodns.net) |
23:07.58 | *** mode/#asterisk [+o anthm] by ChanServ |
23:08.16 | P4C0 | [TK]D-Fender, what can I do? |
23:08.39 | [TK]D-Fender | P4C0, I gave you 1 suggestion. Follow it and see. Also ensure that reinvites are disabled |
23:08.58 | hedge77 | woops i meant members in queues.conf and agents in members.conf |
23:09.15 | P4C0 | [TK]D-Fender, you mean restrict it to alaw only? that the way it was working before... |
23:09.16 | hedge77 | ^^^ agents.conf gak |
23:10.18 | Braxus | anyone here try any of the new polycom phones that were recently released? |
23:10.53 | *** join/#asterisk infinity1 (i=foobar@modena.netcal.com) |
23:11.11 | *** join/#asterisk ltdwk (n=z@203-173-10-9.perm.iinet.net.au) |
23:11.13 | infinity1 | how can i make an agi script continue to run even if the person hangs up? |
23:11.59 | [TK]D-Fender | Braxus, IIRC they are not in fact released yet |
23:12.25 | JT | morning [TK]D-Fender |
23:13.18 | [TK]D-Fender | JT : GMT -5 says "hello" to you! |
23:13.29 | JT | UTC+10 hi! |
23:14.18 | Braxus | wondering if the polycom 320/330 series will give the linksys spa-92x/94x line a run for their money. |
23:14.22 | Juggie | infinity1, ignore the SIGHUP from asterisk |
23:14.38 | infinity1 | Juggie: umm ...how do I do that? |
23:15.13 | infinity1 | Juggie: is it possible to trigger a function when receiving a sighup? hmm |
23:18.08 | *** join/#asterisk flenders (n=fserto@unaffiliated/flenders) |
23:18.32 | [TK]D-Fender | Braxus, Of course they will. Why wouldn't they? |
23:18.32 | hedge77 | hey that worked thanks |
23:18.43 | *** join/#asterisk mrbnet (n=mrbnet@corpmail1.mrbnetworks.com) |
23:19.10 | [TK]D-Fender | Braxus, the only downside of the 320/330 is that they only support 2 calls per line-key |
23:19.10 | Juggie | infinity1, well, depends on how your agi script is coded. |
23:19.15 | Juggie | but the answer would be yes. |
23:19.15 | JT | [TK]D-Fender: my Avaya 4621SW SIP phone has arrived |
23:19.20 | JT | now all i need is PoE ;) |
23:19.25 | Juggie | depending on your programming language, etc. |
23:19.26 | [TK]D-Fender | JT : same with my Aastra 57i CT :D |
23:19.30 | JT | it doesn't have DC in |
23:19.44 | JT | nice |
23:19.50 | JT | CT = cordless? |
23:20.00 | [TK]D-Fender | JT yup.. the reason I got it |
23:20.19 | jaxxan | bah i didn't need to patch |
23:20.25 | P4C0 | how can I know what codec is a call using? |
23:20.30 | [TK]D-Fender | JT : I set it up via the Web interface for the "quick launch". Am about to go throught the 600+pg admind guid ~>~ |
23:20.39 | JT | the avaya has quite a big screen, 12 line appearances i'm guessing |
23:20.52 | JT | [TK]D-Fender: nice |
23:21.12 | [TK]D-Fender | JT Absurd QTY of calls.... |
23:21.15 | Braxus | calls per line-key shouldn't be a major issue. What I'm wondering is if the 320/330 line will have around the same speakerphone quality of the 501s. |
23:21.31 | [TK]D-Fender | Braxus, I'd imagine more like the IP 430 |
23:21.35 | JT | [TK]D-Fender: heh, the aastra or the avaya? |
23:21.42 | [TK]D-Fender | Braxus, There is a small diff betweent he 430 & 501 |
23:22.04 | [TK]D-Fender | JT Avaya is it wupports that many |
23:22.12 | [TK]D-Fender | if it supports* |
23:22.14 | [TK]D-Fender | ak;sljdkjhlgfd |
23:22.25 | Braxus | the 501s I have deployed here seem to be awesome for the most part, especially in small offices. Was considering the 430 for cubicles... though the new 320s look interesting as well. |
23:22.27 | JT | doesn't the 601 have something like 12 line keys? |
23:22.39 | Braxus | for the cheaper phones, most people here are on linksys 941s |
23:22.43 | [TK]D-Fender | JT 6 on the unit, more if you add a console |
23:22.55 | JT | hrm, what about the 650? only 6? |
23:22.58 | [TK]D-Fender | Braxus, I'd take any polycom over any Linksys, hands down |
23:23.06 | [TK]D-Fender | JT same thing for the 650 |
23:23.10 | JT | ah ok |
23:23.20 | JT | well i haven't got it powered up yet |
23:23.24 | JT | but time will tell |
23:23.29 | [TK]D-Fender | JT : I might like to get my hands on a 550 as well... am awaiting final pricing |
23:23.44 | P4C0 | how can I know what codec is a call using? |
23:23.47 | JT | it has 6 keys on left and 6 on right of screen, and 4 on the bottom of screen, and a couple of dozen other buttons |
23:23.50 | JT | buttons are nice ;) |
23:23.51 | Braxus | is the HD Voice thing done via a different codec? |
23:23.51 | Qwell[] | P4C0: show channels |
23:23.57 | Qwell[] | Braxus: g722 |
23:24.10 | [TK]D-Fender | Braxus, G722 its a standard |
23:24.28 | P4C0 | Unable to find a codec translation path from alaw to g729 |
23:24.36 | JT | [TK]D-Fender: oh and the vaya cost AUD$130, about USD$110 :D |
23:24.39 | JT | avaya |
23:24.44 | P4C0 | but some times it works |
23:24.51 | voltagex | JT: Have you had any experience with KoalaVOIP? |
23:24.53 | *** join/#asterisk coppice (n=chatzill@90.203.17.210.dyn.pacific.net.hk) |
23:24.55 | flenders | JT: where did you get it? |
23:25.00 | JT | flenders: ebay :) |
23:25.02 | [TK]D-Fender | JT : excellent sounding deal... if it works, and works well :) |
23:25.04 | flenders | JT: us? |
23:25.07 | JT | au |
23:25.20 | P4C0 | show channels doesn't give me the codecs |
23:25.29 | JT | voltagex: not first hand, but everything i've heard says "avoid" |
23:25.34 | P4C0 | show channels |
23:25.34 | P4C0 | Channel Location State Application(Data) |
23:25.34 | P4C0 | SIP/3-081f1a98 (None) Up Bridged Call(SIP/3800735-081ec |
23:25.34 | P4C0 | SIP/3800735-081ecd28 s@commercial:1 Up Dial(SIP/2&SIP/3&SIP/1|30) |
23:25.34 | P4C0 | 2 active channels |
23:25.35 | P4C0 | 1 active call |
23:25.38 | [TK]D-Fender | P4C0, "sip show channels" |
23:25.52 | voltagex | JT: yeah, but I have $10 free credit |
23:26.02 | JT | voltagex: i guess you could give it a go |
23:26.03 | voltagex | it's just not registering :/ |
23:26.11 | [TK]D-Fender | voltagex, I'm suspecting it'll cost you :) |
23:26.20 | Qwell[] | sorry, show channel <channel> |
23:26.25 | JT | voltagex: check ozvoipstatus to see if it's up |
23:26.33 | voltagex | [TK]D-Fender: in time only, they don't have any details |
23:26.40 | P4C0 | does this makes any sense: 192.168.6.5 (None) 0F7C823D-ED 00101/00074 unkn No Rx: OPTIONS ? (in sip show channels? i have that line... aditional to the call from my server to phone and server to voip provider) |
23:27.11 | *** join/#asterisk tzafrir_laptop (n=tzafrir@bzq-88-155-97-204.red.bezeqint.net) |
23:27.24 | JT | voltagex: also check the whirlpool voip wiki, it will have details on how to setup asterisk for it |
23:28.39 | voltagex | koala's up, my config matches whirlpool's |
23:29.01 | voltagex | iax2 show registry is empty |
23:29.49 | *** join/#asterisk lerat (n=dnormand@70.55.203.236) |
23:29.50 | ZX81 | P4C0: will be a registration or SIP Options packet |
23:29.56 | fall0ut | Has anybody done anything with ss7box? |
23:29.57 | flenders | JT: do you have a poe switch? |
23:30.02 | JT | flenders: no |
23:30.11 | lerat | Hi all |
23:30.23 | JT | i should see if PoE modules are available for my ProCurve |
23:30.26 | JT | i doubt it |
23:30.32 | P4C0 | maybe it's because i'm not registered to my voip provider... |
23:31.04 | flenders | you have a procurve at home? |
23:31.33 | ZX81 | P4C0: I doubt your provider will be 192.168.x.x :) |
23:31.42 | ZX81 | unless you are your provider |
23:31.51 | lerat | I have already an asterisk ver. 1.4.1 with druid and i want to install an other GUI... how can i remove my old GUI without having to reinstall asterisk ? |
23:31.55 | P4C0 | ZX81, no, it's not, I mean for the codec problem |
23:31.57 | JT | flenders: yeah, only 48 ports fitted i think... |
23:32.16 | ZX81 | P4C0: when you do sip show channels, what codec does it say you are using? |
23:33.08 | P4C0 | ZX81, the correct ones... alaw to the local phone, g729 to the provider... but that's when ringing... |
23:33.28 | [TK]D-Fender | P4C0, what phone? |
23:33.47 | P4C0 | [TK]D-Fender, local sip phone... sjphone |
23:34.14 | [TK]D-Fender | P4C0, You have both with a #1 priority of G729. IIRC it doesn't SUPPORT G729 |
23:34.24 | lerat | Does anybody can help me ????? |
23:34.27 | [TK]D-Fender | P4C0, If you set the soft-phone for ALAW only, does it work? |
23:35.05 | [TK]D-Fender | lerat, This is not a GUI support channel. Check the one appropriate to the one you're using |
23:35.18 | P4C0 | [TK]D-Fender, disallow=g729 inside the sip.conf local phone? |
23:35.18 | lerat | sorry |
23:35.25 | lerat | my mistake |
23:35.31 | [TK]D-Fender | P4C0, disallow=all, allow=alaw |
23:35.39 | *** part/#asterisk lerat (n=dnormand@70.55.203.236) |
23:35.49 | P4C0 | [TK]D-Fender, let me check... i did only with disallow=g729 moment |
23:35.49 | *** join/#asterisk Fieldy (i=sK5kdpjm@gentoo/contributor/Fieldy) |
23:35.50 | *** join/#asterisk tcastleman (n=chatzill@pool-81-187-17-249.b3-it.com) |
23:35.51 | *** join/#asterisk ToyMan (n=Stuart@user-12lcqol.cable.mindspring.com) |
23:36.08 | P4C0 | [TK]D-Fender, what about in global scope? |
23:36.24 | Fieldy | hm.. is there a certain type of MP3 that works best for music on hold? i put some up and some come over solid, but some come over extremely broken up. set verbose 10 isn't complaining |
23:38.32 | P4C0 | http://rafb.net/p/7Q3FPS41.html |
23:39.17 | [TK]D-Fender | Fieldy, 128kbit non VBR |
23:39.19 | P4C0 | this is fustrating... |
23:39.44 | [TK]D-Fender | P4C0, make sure canreinvite=no for both. |
23:39.51 | [TK]D-Fender | ok, gotta go for now, back in a few hours |
23:39.53 | P4C0 | [TK]D-Fender, it is |
23:40.16 | Fieldy | [TK]D-Fender: ok thanks |
23:40.26 | P4C0 | I'll go as well |
23:42.05 | tcastleman | Hi there. We have a strange problem with asterisk's sounds suddenly not being audible. It is intermittent. Once it works it works. But intermittently on starting zaptel, misdn and amportal it occurs. Calls between SIP extensions are unaffected. It's just the sounds played by asterisk itself. |
23:42.12 | tcastleman | On the console all seems normal and asterisk 'says' its playing the sound, the nothing is audible on the handset |
23:42.16 | tcastleman | we are using asterisk 1.2.17, zaptel 1.2.17 with misdn for a TDM880P and a B410P |
23:42.27 | tcastleman | on a Dell Poweredge 840 |
23:42.52 | *** join/#asterisk Stridernzl (n=neville@125-239-175-26.jetstream.xtra.co.nz) |
23:44.13 | tcastleman | once this problem is being experienced asterisk sounds are inaudible whether it be on a SIP extension, ZAP extension or via an misdn or SIP trunk. Nothing. |
23:44.22 | tcastleman | a series of starts and stops can fix it |
23:44.31 | tcastleman | but it seems to be completely random |
23:45.02 | tcastleman | and the asterisk console shows no errors, it looks like all is operating normally.. |
23:45.08 | Dr-Linux|work | any perl expert around? :P |
23:45.21 | tcastleman | permissions on the asterisk sounds is correct |
23:45.31 | flenders | JT: just about to buy that sangoma 101 |
23:45.45 | JT | flenders: cool |
23:45.47 | eald | does the order of extra keys in AMI Actions, matters? for example in Action: Login if I send secret before login? |
23:45.59 | JT | tcastleman: ok, try this tesT: |
23:46.23 | JT | tcastleman: add a Wait(2) line before the audio is played back |
23:46.49 | JT | tcastleman: tell me if it does anything |
23:47.04 | tene | Dr-Linux|work: a little bit. what's up? |
23:47.39 | Dr-Linux|work | don't worry, it's my 2nd day looking perl |
23:47.45 | Dr-Linux|work | tene: can i /msg you? :) |
23:47.56 | tene | Dr-Linux|work: sure. |
23:47.59 | tcastleman | JT, it's working now.. let me try to break it and test |
23:48.05 | Dr-Linux|work | thanks |
23:48.09 | JT | tcastleman: ok |
23:48.53 | tcastleman | JT, I'm using free PBX. Where did you want me to put the Wait(2)? |
23:49.23 | JT | tcastleman: eh, well i expect you to know what you're doing with the dialplan if you come here with a freepbx problem |
23:50.58 | tcastleman | JT, I do know what I am doing with the dialplan to a point. I have several custom contexts I could text with. Did you want me to test with one of those? |
23:51.19 | JT | well yeah, somewhere that you are having a problem |
23:51.31 | tcastleman | JT, ok cool no problem. give me a sec |
23:53.41 | *** join/#asterisk Dovid (n=Dovid@l192-117-114-1.broadband.actcom.net.il) |
23:58.05 | tcastleman | JT, typical I can't break it now. |
23:58.22 | tcastleman | JT, the annoying think is that it seems to be completely random. |
23:58.49 | JT | hmm |