00:04.36 | *** join/#asterisk ehsjoar (n=ehsjoar@c-24-9-91-203.hsd1.co.comcast.net) |
00:10.28 | Xetrov` | is there a good free softphone app for windows mobile? |
00:12.39 | *** join/#asterisk viq (n=viq@unaffiliated/viq) |
00:14.36 | Orbixx | Seems I have to compile a kernel as my provider uses a custom kernel on their servers. |
00:16.29 | Snoogan | [TK]D-Fender: what settings do i need for freepbx to register the ekiga sip trunk? |
00:20.24 | *** join/#asterisk ariel_ (n=ariel_@c-71-196-99-26.hsd1.fl.comcast.net) |
00:22.13 | *** join/#asterisk propellerhead (n=yogurt2u@host66.190-31-153.telecom.net.ar) |
00:31.16 | *** join/#asterisk elite (n=elite@c-71-197-242-157.hsd1.wa.comcast.net) |
00:31.46 | Guest441 | Is it possible to call a number and get the charger number and hangup so it never rings? |
00:31.47 | *** join/#asterisk Alfio (n=Amunoz@adsl-54-26.tricom.net) |
00:35.19 | ariel_ | the charger number? what is that? |
00:35.45 | Guest441 | the charge number |
00:38.11 | ariel_ | I still don't understand what you mean by the charge number. |
00:38.46 | Guest441 | The carge number in the SS7 ISUP IAM |
00:38.49 | Guest441 | charge* |
00:39.35 | ariel_ | not unless you have access to the ss7 switch |
00:41.38 | Guest441 | Is it possible to forward an incoming call to asterisk and then set the charge #? |
00:42.53 | ariel_ | what you are trying to do is spoofing and actually it's doable but I can't and will not go into this |
00:44.04 | Guest441 | Really wahts the point of idling in a channel and not going full out with your information, stop being a pussy |
00:44.24 | Guest441 | Everyones security is already fked, WEP and WPA is cracked like butter anyways |
00:46.10 | Guest441 | Can someone who is not a pussy please inform me |
00:46.39 | Guest441 | http://www.binrev.com/forums/index.php/topic/14010-ss7-isup-number-delivery-fields/ |
00:47.34 | ariel_ | Like being called that, but I can tell you that I will not due to many issue's but mainly my own in being on the level and having respect for others as well. |
00:48.01 | Guest441 | are you a christian dude? |
00:48.29 | Orbixx | Nobody is obliged to help you. |
00:48.41 | Guest441 | Is it in your hands as a christian to save the world from spoofing? |
00:48.47 | ariel_ | don't have to be to be on the level |
00:59.50 | *** part/#asterisk ehsjoar (n=ehsjoar@c-24-9-91-203.hsd1.co.comcast.net) |
01:09.05 | leifmadsen | Zup yo |
01:09.36 | leifmadsen | Guest441: you're odd |
01:10.26 | Guest441 | is that an insult these days? |
01:10.33 | leifmadsen | no, I'm just saying you're odd |
01:10.47 | leifmadsen | you've got a unique approach to asking questions is all |
01:11.39 | Guest441 | leifmadsen: well thank you, the odd people make the world go around |
01:12.24 | Alfio | Guest441 like in my country say you are a macagrano |
01:12.54 | *** join/#asterisk Iamnacho (n=Iamnacho@wsip-98-191-129-196.ks.ks.cox.net) |
01:13.01 | Guest441 | does your country have clean water? |
01:13.50 | Alfio | yes and lambetallos like you |
01:14.31 | Guest441 | i will glady be your king |
01:14.49 | Guest441 | now go get your sister so i can fuck her |
01:14.59 | Alfio | thx |
01:15.07 | [TK]D-Fender | Guest441: Cool it. |
01:15.38 | Guest441 | See you guys should follow fenders example |
01:15.50 | Guest441 | hes dealt with my kind before |
01:16.04 | *** mode/#asterisk [+b *!*@*.hsd1.wa.comcast.net] by leifmadsen |
01:16.04 | *** kick/#asterisk [Guest441!n=Leif@asterisk/documenteur-extraordinaire/blitzrage] by leifmadsen (leifmadsen) |
01:16.12 | Alfio | :) |
01:16.25 | leifmadsen | who's the king? |
01:16.32 | [TK]D-Fender | leifmadsen: Kill-joy :) |
01:16.35 | Alfio | hehehehhehehehhee |
01:16.39 | leifmadsen | he'll be back in a minute telling me I'm ghey or something |
01:16.55 | leifmadsen | [TK]D-Fender: ya, I almost left he be just for fun, but i'm leaving soon |
01:17.24 | [TK]D-Fender | leifmadsen: I'd have dealt with him... |
01:17.26 | *** mode/#asterisk [-b *!*@*.hsd1.wa.comcast.net] by leifmadsen |
01:17.34 | leifmadsen | time out is over ;) |
01:18.12 | leifmadsen | sweet, just found out I've had a record year for consulting |
01:18.15 | leifmadsen | recession, what?! |
01:18.26 | Alfio | hehehheehhe |
01:18.28 | jaytee | your patience is admirable, i'd have reduced him to a bowl of snot several minutes before you kicked him |
01:18.41 | leifmadsen | jaytee: he was kind of entertaining me :) |
01:18.47 | Alfio | the recession its so good for asterisk |
01:18.54 | leifmadsen | it does seem to be |
01:19.12 | leifmadsen | "what? we can't spend $1m on Cisco or Avaya? I guess we'll look at this Asterisk thing" |
01:19.15 | Alfio | many companys are lookint to cut their budgets |
01:19.24 | jaytee | yeah, it was fun to listen to. I was tempted to engage him but I usually have an effect on people like that that escalates into all out war :-) |
01:20.03 | [TK]D-Fender | jaytee: I'd have just taken him apart one little piece at a time and then casually discarded the carcass... |
01:20.12 | leifmadsen | with acid? |
01:20.37 | [TK]D-Fender | leifmadsen: Mr. Pointy :p |
01:21.03 | leifmadsen | heh |
01:21.17 | [TK]D-Fender | should be 2nd kyu now... |
01:21.55 | leifmadsen | weak |
01:22.45 | [TK]D-Fender | leifmadsen: Sensei has only shown up once this month and I've actually been teaching the newbs with almost all the other senior students on vacation, etc. |
01:22.59 | Alfio | one question why itsnt an spanish channel like this? |
01:23.36 | jaytee | so was I wrong in getting the impression that he was pumping people for info on SS7 headers in order to commit toll fraud? |
01:25.43 | *** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
01:31.14 | *** join/#asterisk Faiz (n=otakucon@c-98-221-51-177.hsd1.nj.comcast.net) |
01:33.10 | drmessano | She's like a rainbow |
01:53.16 | *** join/#asterisk Rob3Rt (i=R0b3Rt@181.45.96.58.static.exetel.com.au) |
02:04.10 | *** join/#asterisk mbrevda (n=mbrevda@unaffiliated/mbrevda) |
02:34.54 | Faiz | is there a particular reason why the asterisk (1.6) CLI doesn't reload modules properly? |
02:35.18 | Faiz | it hangs after i issue 'module unload chan_dahdi.co' |
02:35.34 | Faiz | since it does not allow me to run dahdi commands |
02:37.21 | Snoogan | can someone offer assistance and help me to diagnose why I can't register sip trunks on my Asterisk box? |
02:42.30 | [TK]D-Fender | Snoogan: What was the result of the last test I told you to do? |
02:44.25 | *** join/#asterisk RypPn (i=TuMbL@rosscom.demon.co.uk) |
02:44.28 | Faiz | hey Fender, would it be alright if I private message you about a few issues? |
02:44.34 | Snoogan | i couldn't work out the settings for freepbx for the free sip account. I have signed up for a gotalk account, and put in settings according to what ppl have listed on whirlpool. |
02:44.49 | Snoogan | the go talk trunk won't register either |
02:50.50 | [TK]D-Fender | FaiAsk in channel; |
02:51.23 | [TK]D-Fender | Snoogan: Doesn't maater that it fails, it matter HOW it fails |
02:53.19 | Snoogan | should i do a sip set debug? |
02:54.22 | [TK]D-Fender | Snoogan: Do you think looking at the problem might help you see what it is? |
02:54.41 | [TK]D-Fender | Snoogan: Its a really novel idea I'll admit... |
02:54.54 | Snoogan | yeah, of course, but i don't understand the output ... that's why i'm asking for assistance |
02:55.49 | Snoogan | you've been helpful so far, and i appreciate that. But i need guidance. |
02:56.19 | [TK]D-Fender | Snoogan: Yes, and I looked at it for you last time... |
02:56.31 | [TK]D-Fender | Snoogan: Do you really have to ask if I would again? |
02:56.38 | *** join/#asterisk leifmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
02:56.38 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
02:58.06 | *** join/#asterisk ingenius (n=alektro@host95.190-229-175.telecom.net.ar) |
02:58.34 | Snoogan | i wasn't sure, that's why i asked. http://pastebin.ca/1506545 shows the output |
03:00.18 | [TK]D-Fender | Snoogan: Ok, packets simply aren't making it back. Either your router is bad and screwing up, or your internet connection is being filtered. |
03:01.21 | Snoogan | the mynetfone trunk registered yesterday |
03:01.46 | Snoogan | and i've registered before on this connection with other boxes. So i rule out the ISP connection |
03:01.58 | Snoogan | so i'll try a different modem |
03:02.01 | [TK]D-Fender | Snoogan: well you confirmed your IP, I suppose I'm not going to see your router settings to trust that what needs to be forwarded is... we're jsut about out of options here |
03:02.32 | Snoogan | i can send you a screen shot of my port forwarding |
03:05.35 | *** join/#asterisk tflgen2 (n=clay@fl-67-233-27-60.dhcp.embarqhsd.net) |
03:05.51 | [TK]D-Fender | Snoogan: imagebin.ca |
03:07.24 | Snoogan | http://imagebin.ca/view/1oW4Aq4.html |
03:09.41 | [TK]D-Fender | Snoogan: Ranges look right. Does your router know anything about SIP aside from those settings you made? AKA is it "SIP aware"? |
03:10.07 | Snoogan | i'll check through the settings now |
03:10.40 | *** join/#asterisk tflgen2 (n=clay@fl-67-233-27-60.dhcp.embarqhsd.net) |
03:11.48 | *** join/#asterisk darkmadda (n=none@c-76-27-95-83.hsd1.ut.comcast.net) |
03:13.12 | *** join/#asterisk Katty (n=Katty@mail.copi-rite.com) |
03:13.22 | Katty | mew? |
03:13.36 | jaytee | mew |
03:13.50 | Katty | jmirc has odd colors |
03:13.57 | Katty | Herroes |
03:14.27 | jaytee | rowr |
03:14.49 | Katty | whatcha doin |
03:14.53 | [TK]D-Fender | Katty: Mew. |
03:15.08 | Katty | Hi fender |
03:15.18 | jaytee | watchin tv, playin Mafia Wars and watching [TK]D-Fender beat sense into newbs |
03:15.37 | Katty | fun times |
03:15.43 | [TK]D-Fender | jaytee: This one is getting it easy... |
03:16.01 | jaytee | [TK]D-Fender, yeah it's a case of [TK]D-Fender Lite |
03:16.09 | Katty | I'm in bed, avoiding sleep |
03:16.26 | [TK]D-Fender | jaytee: Diet-D-Fender : just like the real thing, only half the ClueBat |
03:16.30 | [TK]D-Fender | ....(tm) |
03:16.41 | Katty | hehe |
03:16.49 | jaytee | hahaha |
03:17.01 | jaytee | Katty, why are you avoiding sleeping? |
03:17.08 | Katty | Fun size! |
03:17.29 | Katty | no particular reason |
03:17.58 | Katty | just tinkering with the blackberry |
03:18.12 | Snoogan | [TK]D-Fenderk: can't find anything relating to sip |
03:18.30 | Snoogan | i will try a firmware upgrade and get back to you shortly |
03:18.44 | Katty | I'm going to invent a sip fax |
03:18.44 | *** join/#asterisk wonderworld (n=w@ip-62-143-16-28.unitymediagroup.de) |
03:19.20 | Katty | and sell it to kyocera |
03:19.36 | jaytee | awesome! |
03:20.06 | Katty | yes. It will be grand |
03:20.49 | Katty | I will make billions, retire, and adopt jaytee |
03:20.54 | jaytee | had a 3 minute network outage today at work. all network traffic. |
03:21.10 | eppigy | hi Katty |
03:21.12 | Katty | from what |
03:21.19 | eppigy | DONDE |
03:21.24 | *** part/#asterisk Snoogan (n=asdf@115.69.179.5) |
03:21.29 | Katty | herro |
03:22.06 | jaytee | error logs on the Cisco Catalyst 4507 that is our main switch and includes our backbone registered about 3 minutes worth of HOSTFLAPPING errors on various gigabit fiber ports. |
03:22.22 | jaytee | the network engineer still hasn't figured it out |
03:22.30 | Katty | eeeesha |
03:22.37 | eppigy | that is a terrible position to be in |
03:22.41 | eppigy | WE NEED A ROOT CAUSE |
03:22.48 | tflgen2 | ouch |
03:23.11 | Katty | How about a rootbeer float instead |
03:23.16 | eppigy | :D |
03:23.24 | jaytee | well, shouldn't get HOSTFLAPPING with spanning tree so odds are the Catalyst had some kind of "brainfart" and healed itself |
03:23.55 | tflgen2 | had some issues with a 2811 earlier this month |
03:23.58 | Katty | I'd call cisco )= |
03:24.00 | tflgen2 | voip related |
03:24.33 | eppigy | http://www.cisco.com/en/US/products/hw/switches/ps4324/products_tech_note09186a008063c36f.shtml |
03:24.34 | eppigy | lol |
03:25.09 | jaytee | the errors were on multiple VLANS and ports and none of the connect points are close enough to each other that someone could have plugged in a cable looping any of them. |
03:26.00 | tflgen2 | jaytee: hmm, sure that someone didn't misconfigure forwarding on a box that had access to multiple vlans? |
03:26.40 | jaytee | tflgen2, we're a 5 person shop. Of the 3 of us that might make a net change all of us were at lunch |
03:27.10 | eppigy | i like the rootbeer float idea |
03:27.29 | jaytee | eppigy, this message is the one we were getting http://www.cisco.com/en/US/products/hw/switches/ps4324/products_tech_note09186a008063c36f.shtml#cg1 |
03:27.47 | jaytee | but across multiple ports and from multiple addresses. |
03:28.11 | eppigy | i would question the otters |
03:28.36 | jaytee | hahhahaa |
03:29.03 | jaytee | my first on the list of "Usual Suspects" is the lemurs and then Keyser Sose |
03:29.48 | eppigy | true |
03:36.32 | darkmadda | so i'm new to asterisk so i could use some help. Here is what i want to do. I'd like to make a dial plan/outgoing route. When a call is placed and matches a pattern i'd like it to place the call on hold. execute a script(bash/perl/php doesn't matter), then it changes an incoming route so that it connects to the line placed on hold instead of ringing it's normal extension. |
03:36.53 | darkmadda | after that the outgoing route should revert back to it's initial state. |
03:37.37 | darkmadda | playing a message to the call on hold would also be nice. |
03:38.33 | darkmadda | i have no clue how to do any of that. :-) |
03:51.09 | *** join/#asterisk DrAk0 (n=luisjose@nelug/coreteam/luisjose) |
04:16.53 | Faiz | is there a particular reason why the asterisk (1.6) CLI doesn't reload modules properly? |
04:16.55 | Faiz | it hangs after i issue 'module unload chan_dahdi.co' |
04:16.56 | Faiz | since it does not allow me to run dahdi commands |
04:28.46 | *** join/#asterisk Rob3Rt (i=R0b3Rt@181.45.96.58.static.exetel.com.au) |
04:36.34 | *** join/#asterisk voxter (n=voxter@190.241.15.56) |
05:32.04 | *** join/#asterisk Snoogan (n=asdf@115.69.179.5) |
05:32.54 | Snoogan | [TK]D-Fender: my problem is in my modem. Tried a different dsl modem and it registered immediately and both calls in and out are working |
05:40.42 | kn0x | Snoogan: what does it have some kind of builtin router? |
05:46.53 | Faiz | I'm sorry for sounding like a noob, but I still can't seem to understand why I can't load DAHDI modules from my asterisk CLI (v. 1.6) ? |
05:47.21 | Faiz | is there a particular version of DAHDI linux+tools I need to install for ast.1.6.11 ? |
05:54.57 | *** join/#asterisk Rob3Rt (i=R0b3Rt@181.45.96.58.static.exetel.com.au) |
05:57.11 | drmessano | 1.6.11 is out???? |
05:57.19 | Faiz | yes |
05:57.20 | drmessano | ZOMG TORRENT PLZ!!! |
05:57.49 | Faiz | wget http://downloads.asterisk.org/pub/telephony/asterisk/releases/asterisk-1.6.1.1.tar.gz |
05:58.27 | drmessano | Thats 1.6.1.1 not 1.6.11 |
05:58.32 | drmessano | :( |
05:58.50 | Faiz | my apologies for the typo |
05:59.02 | drmessano | :``( |
05:59.04 | Faiz | but perchance, do you have any idea regarding my situation? |
05:59.05 | [TK]D-Fender | Snoogan: Glad you found it |
05:59.39 | Faiz | i have installed asterisk after making sure that dahdi was started |
06:00.01 | Faiz | configured the extensions, chan_dahdi and system.config files and double-checked thoroughly |
06:00.01 | drmessano | hmmm |
06:00.53 | drmessano | Show us some CLI output |
06:00.56 | drmessano | ~pb |
06:00.57 | infobot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste , or , http://bin.cakephp.org/, or apt-get install pastebinit |
06:00.58 | Faiz | sure |
06:03.33 | Faiz | also, when I try to issue command: "module unload chan_dahdi.so", i do not get an output, and the CLI hangs |
06:04.13 | *** join/#asterisk SkywaIker (n=pirch@58.147.17.166) |
06:04.34 | leifmadsen | y0 |
06:05.09 | leifmadsen | there will never be a 1.6.11 |
06:05.14 | leifmadsen | 1.6.11.0 maybe |
06:06.36 | Faiz | again, my apologies as it was a typo |
06:06.40 | Faiz | my pastebin: http://pastebin.com/m1c5fc08f |
06:07.29 | Faiz | you can see at line 117 that when i issue chan_dahdi.co to stop, all succeeding commands are ignored as the CLI hangs |
06:07.53 | Faiz | i CTRL+Z'd out of the CLI, and included the DAHDI configs |
06:10.00 | leifmadsen | did you compile asterisk after installing dahdi |
06:10.02 | leifmadsen | ? |
06:10.15 | leifmadsen | order should be: libpri, dahdi, asterisk |
06:10.27 | leifmadsen | (order of compilation and installation |
06:10.35 | Faiz | yes, libpri first, dahdi, restarted and ran as: /etc/init.d/dahdi start |
06:10.42 | Faiz | compiled and installed asterisk |
06:10.44 | *** join/#asterisk _mwoodj_ (n=MWoodJ@pdpc/sponsor/digium/hyper-eye) |
06:10.50 | Faiz | saw the chan_dahdi.so file being parsed |
06:11.10 | leifmadsen | well, that *should* work fine |
06:11.13 | leifmadsen | what version of dahdi? |
06:11.23 | leifmadsen | what OS? |
06:11.36 | Faiz | version 2.2.0.2, tools: 2.2.0 |
06:11.39 | Faiz | CentOS v 5.3 |
06:11.50 | leifmadsen | never had a problem with that configuration |
06:12.00 | leifmadsen | although 1.6.1.1 is getting pretty old |
06:12.05 | leifmadsen | you should try 1.6.1 branch |
06:12.15 | Faiz | hm |
06:12.17 | leifmadsen | svn co http://svn.asterisk.org/svn/asterisk/branches/1.6.1 |
06:12.21 | Faiz | thank you |
06:12.31 | leifmadsen | 1.6.1.1 is a security release after 1.6.1.0 |
06:12.45 | Faiz | ah |
06:12.48 | leifmadsen | which was like..... June 5th |
06:12.49 | Faiz | i'm updating as we speak |
06:12.56 | leifmadsen | i.e. 1.6.1.1 is 1.6.1.0 |
06:13.06 | Faiz | ah, i see |
06:13.10 | leifmadsen | and 1.6.1.0 is even older, and we all know what .0 releases are |
06:13.13 | leifmadsen | (useless) |
06:13.17 | Faiz | heh |
06:13.24 | Faiz | should i restart the machine? |
06:13.26 | Faiz | or just asterisk |
06:13.27 | leifmadsen | I say that, and I'm even the one who makes the releases |
06:13.31 | leifmadsen | just asterisk |
06:13.38 | leifmadsen | no point in restarting the machine |
06:13.39 | Faiz | very cool |
06:13.49 | leifmadsen | check the ChangeLog :) |
06:14.07 | leifmadsen | anyways, you shouldn't have those locking issues |
06:14.15 | leifmadsen | it looks like you're getting deadlocks |
06:14.38 | leifmadsen | add the DONT_OPTIMIZE and MALLOC_DEBUG flags under the Compiler Flags option of menuselect |
06:14.47 | leifmadsen | then you can do 'core show locks' to see if you're getting deadlocks |
06:14.54 | leifmadsen | (which shouldn't happen) |
06:15.03 | Faiz | alright |
06:15.08 | leifmadsen | suggest you try 1.6.1 branch with DAHDI verision you're running |
06:16.18 | Faiz | i'll re-run the menuselect now |
06:16.26 | Faiz | but dahdi is still not being recognized by the asterisk CLI |
06:16.57 | leifmadsen | huh? |
06:17.05 | leifmadsen | is chan_dahdi.so being loaded? |
06:17.29 | Faiz | nope |
06:17.33 | Faiz | nor unloaded |
06:17.34 | leifmadsen | then it isn't being compiled |
06:17.39 | leifmadsen | ./configure |
06:17.52 | leifmadsen | it will pick up dahdi if it was installed |
06:17.56 | Faiz | should i pastebin my ./configure output? |
06:17.59 | leifmadsen | no |
06:18.04 | carrar | heh |
06:18.07 | leifmadsen | it'll be obvious if it found it |
06:18.43 | leifmadsen | install dahdi, run ./configure in asterisk source dir, install asterisk -- you will see chan_dahdi in the Channels section of menuselect if it found it |
06:18.45 | [TK]D-Fender | leifmadsen: Just because its obvious... doesn't mean he's not BLIND :p |
06:19.03 | leifmadsen | otherwise, run "asterisk -cvvvg" to see the modules load, then look for "WARNING" and "ERROR" |
06:19.20 | leifmadsen | you may have a couple WARNING, but never should have an ERROR |
06:19.51 | leifmadsen | [TK]D-Fender: your a first class dick... |
06:19.54 | leifmadsen | you're* |
06:20.50 | [TK]D-Fender | *cough* that's what she said... |
06:20.58 | [TK]D-Fender | zing! |
06:21.19 | Faiz | when i run the CLI and issue: module show |
06:21.24 | Faiz | chan_dahdi shows up on the list |
06:21.52 | Faiz | (i ran with asterisk -rvvvvg and it did not show me a list of modules loaded |
06:22.25 | carrar | would 3rd class be better then 1st class in this case? |
06:22.29 | leifmadsen | you need to stop asterisk, and run it in the foreground |
06:22.37 | [TK]D-Fender | falz: "r" will not show you on start because you're connecting to a runnign process |
06:23.28 | Faiz | ah, misread, sorry |
06:25.24 | Faiz | "[Jul 25 02:24:06] ERROR[11647]: codec_dahdi.c:626 find_transcoders: Failed to open /dev/dahdi/transcode: No such file or directory" |
06:25.33 | leifmadsen | no biggie |
06:25.48 | Faiz | as you said, few warnings, |
06:26.27 | Faiz | but it isnt responding now |
06:29.18 | drmessano | You broke it |
06:29.24 | drmessano | Bad cop, no donut |
06:30.08 | carrar | Bad cop, you're on video |
06:30.42 | Faiz | sigh.. well I can't get a proper readout, since it hangs and I can't exit out |
06:31.12 | carrar | SCREWED |
06:31.55 | drmessano | taskmgr > select asterisk.exe > end task |
06:32.03 | carrar | hahah |
06:33.41 | drmessano | I spent more time getting ready to go geocaching than I do actually caching |
06:33.43 | drmessano | This is nuts |
06:34.04 | carrar | yeah |
06:34.27 | carrar | Where is the thrill in the hunt when you know right where to go |
06:34.43 | drmessano | If only that were the case |
06:35.54 | carrar | Once there, look for the funny looking rock, 6 paces to the left and you will find a box with a eraser in it |
06:36.00 | drmessano | 20ft accuracy of your GPS combined with 20ft accuracy of the person setting the cache + muggles + sneaky bastards |
06:36.23 | drmessano | Eraser? Damn.. thats more than I normally get.. normally its all microcaches |
06:36.31 | carrar | heh |
06:36.41 | carrar | I once got a MicroSoft optical mouse from one |
06:36.58 | carrar | granted I do live in the capitcal of MicroSoft |
06:37.05 | carrar | capital |
06:37.24 | drmessano | 45 mins looking for a film can shoved in a fake log under a dead dog |
06:37.30 | drmessano | THAT is caching |
06:37.35 | carrar | hahah |
06:37.45 | drmessano | and the hints |
06:37.51 | Faiz | just to make sure, install DAHDI as so: make clean, ./configure, make, make install |
06:37.53 | Faiz | correct? |
06:38.08 | carrar | and dahdi tools |
06:38.41 | Faiz | and then proceed to asterisk as such: make clean, ./configure, make menuselect, make install |
06:38.41 | Faiz | yes, tools as well |
06:38.41 | drmessano | "Old mother hubbard, went to her cupboard, and a tree wouldnt leave her alone"... Which means look under the dead dog for the fake log |
06:38.41 | drmessano | ZOMG how didnt I see that? |
06:38.55 | carrar | no fake dog under a dead log? |
06:39.14 | drmessano | carrar: That was the second one in the multi-cache |
06:39.58 | drmessano | carrar: The third one was a 125lb pit bull with a pink ribbon on its neck holding a wet tree limb looking dog toy and the cache was hidden in his food dish |
06:40.42 | drmessano | That cache was called "Enuff with the Dogs, call 911" |
06:41.19 | Faiz | is there a way i can tell that dahdi has been installed correctly, before i proceed to compiling asterisk? |
06:41.38 | drmessano | Question.. |
06:41.51 | drmessano | Have you upgraded your kernel recently and not yet rebooted? |
06:42.01 | Faiz | no |
06:42.13 | Faiz | is my /usr/src/kernel folder supposed to have 1 or 2 directories? |
06:42.56 | drmessano | Do you have two for two different kernels? |
06:42.58 | carrar | speaking of kernels I need to upgrade my 2.6.30 box |
06:43.23 | carrar | .3 |
06:43.39 | Faiz | i'm not sure, i havent touched the OS settings |
06:43.49 | drmessano | Fuckin A |
06:43.50 | Faiz | besides configuring the network adapter |
06:44.04 | drmessano | What are the DIRECTORY NAMES? |
06:44.10 | carrar | wait you really are running in windows? |
06:44.12 | drmessano | Do they represent KERNELS? |
06:44.34 | Faiz | issuing ls under /usr/src/kernels the directory names are: |
06:44.46 | Faiz | 2.6.18-128.2.1.el5-i686 2.6.18-128.el5-i686 |
06:45.01 | drmessano | Do a uname -r |
06:45.17 | Faiz | i get: 2.6.18-128.2.1.el5 |
06:45.26 | Faiz | is there an issue with symbolic links? |
06:45.41 | drmessano | Only if you make it an issue |
06:46.08 | Faiz | i've been stressing about this for some time now |
06:46.15 | Faiz | i just want to hear a dialtone on my phone :( |
06:46.43 | Faiz | nonetheless, i'll recompile dahdi kernel + tools, restart, start the service, compile asterisk |
06:50.17 | carrar | You don't need * to hear a dialtone!! |
06:51.39 | drmessano | No, just a Yaesu FT-2400m and a lid with the repeater cheat sheet |
06:51.49 | drmessano | Dialtone ALL DAY |
06:53.12 | drmessano | Was chatzilla not updated for 3.5? |
06:54.34 | Faiz | drmessano, would you like to see my ./configure output? |
06:54.48 | drmessano | sure |
06:57.45 | Faiz | http://pastebin.com/m671aa4a1 |
06:58.57 | drmessano | Ok |
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06:59.06 | drmessano | Looks good to me |
06:59.20 | Faiz | alright, proceeding to compile |
06:59.30 | Faiz | dahdi service is running |
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07:15.18 | Faiz | heh, holy hell |
07:15.47 | Faiz | i think it was the USB_Radio module that was causing asterisk to hang |
07:16.14 | Faiz | ran asterisk -cvvv and got a clean scan; dahdi show channels has a proper output |
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07:28.33 | Faiz | wow, let the fun begin |
07:28.37 | Faiz | thanks to all that helped me |
07:29.03 | Faiz | i'll be sure to document this as a guide for estranged folks in the future |
07:29.08 | Faiz | good night |
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07:53.05 | pif | how does one set a minimum ping time for sip peers again? |
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08:07.15 | pif | how do I test a function at the console? for example SIPPEER() |
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09:02.13 | VooDooNOFX | Hey guys. I'm going through the astbook, and i've configured the base sip.conf. In the CLI, I get a register, then an immediate unregister for my xlite client in *1.4.26 |
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09:49.18 | ramindia | VooDooNOFX: check over cli what happends when it unregisters.. enable debug |
09:49.58 | VooDooNOFX | i've got -vvvr enabled |
09:50.14 | VooDooNOFX | All I get is "Registered SIP '1000' at 71.104.35.56 port 52184" |
09:50.29 | VooDooNOFX | then on the next line "Unregistered SIP '1000'" immediately |
09:50.50 | ramindia | is the xlite behind nad ? |
09:50.52 | ramindia | NAT ? |
09:51.01 | VooDooNOFX | yes |
09:51.12 | ramindia | did u enable nat=yes |
09:51.30 | VooDooNOFX | host-dynamic |
09:52.46 | VooDooNOFX | yeah, it was the NAT issue. |
09:52.53 | VooDooNOFX | ty :D |
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09:55.19 | VooDooNOFX | Other than in the astbook, is there a single site to get phone configs for asterisk? |
09:56.20 | ramindia | voip-info.org |
09:56.55 | ramindia | get 2 books of *..both are well enough if u playing with * |
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09:58.28 | VooDooNOFX | yeah, the * book is in the mail, but it won't be here till Aug 4th or something |
09:58.33 | VooDooNOFX | stupid Borders bookstores |
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10:49.14 | VooDooNOFX | I've got some Aastra 55i's from an old packet8 system. Has anyone sucessfully reflashed these back to astra defaults so I can remove the Packet8 branding logos and such? |
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11:00.14 | VooDooNOFX | Is there a seperate loader firmware I need to load on this thing? Seems packet8's loader code fetches the newest firmware on every reboot from their servers. |
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11:10.04 | VooDooNOFX | Oh, I think I found it. Perhaps someone can tell me how to make the aastra.xml Config file for this phone's auto-provisioning |
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11:48.46 | tokozedg | hi, i have two asterisk connected with iax2, and on one side i make a outgoing call to through another *, and for outgoing application i use txfax, so it begins to send fax, and on another side i have rxfax at incoming extension |
11:49.13 | tokozedg | but i cant receive fax? codec is g711 |
11:49.36 | tokozedg | sr, i cant receive fax, and whats is the problem? |
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13:15.07 | marl_scot | hi folks, anyone know if there is a way to show what commands are being sent through the manager interface? i am trying to debug a program that is connecting to the manager interface, but need to know what is being sent and recived by the manager interface something like : manager set debug ? |
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13:34.59 | mwalling | marl_scot: i couldnt find anything, and ended up to watching packet captures |
13:35.22 | marl_scot | ok, thanks, will go down that rout, just thought id ask first |
13:35.55 | mwalling | theres a good chance i'm wrong |
13:36.05 | marl_scot | think i might see if i can do something that will act as a passthrough proxy type thing, cus its a problem ive had in the past |
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14:43.52 | artemmakhutov | Hello, is it possible to use wideband MOH with asterisk 1.6.2-beta3 ? |
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14:51.43 | Orbixx | Any idea why Record() is not responding to #? |
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15:03.33 | Orbixx | This is so wierd. |
15:04.37 | Orbixx | http://pastebin.com/d451e52e6 |
15:04.44 | Orbixx | Anything look wierd with that? |
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15:38.48 | drmessano | Wideband MOH? |
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15:48.40 | Orbixx | drmessano: wav and alaw |
15:49.19 | Orbixx | [TK]D-Fender: Know of any issues that would cause Record() to ignore # to terminate the recording? (other than the option to ignore it) |
15:49.33 | artemmakhutov | 16 KHz MOH with G722 |
15:49.56 | [TK]D-Fender | Orbixx: Yeah, * not getting DTMF at all because its mode doesn't match |
15:50.31 | Orbixx | [TK]D-Fender: To get to the bit that does Record(), I have to traverse an IVR. |
15:50.40 | Orbixx | Which would suggest DTMF is working, would it not? |
15:51.07 | [TK]D-Fender | Orbixx: I suggest your show me something useful |
15:51.27 | Orbixx | [TK]D-Fender: http://sprunge.us/IiQW |
15:51.49 | Orbixx | The problem is at ivr,4 |
15:52.46 | [TK]D-Fender | That looks like my timeout code... |
15:55.13 | Orbixx | Possibly, nabbed it off voip-info. |
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15:56.59 | [TK]D-Fender | Show a failed call with SIP debug as appropriate |
15:58.29 | drmessano | Orbixx: [10:44] <artemmakhutov> Hello, is it possible to use wideband MOH with asterisk 1.6.2-beta3 ? <-- I doubt thats what he's asking about.. which is why I wanted a clairification |
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16:00.56 | Orbixx | [TK]D-Fender: Not sure how to do that. Sorry. |
16:01.19 | Orbixx | drmessano: Ah my bad, didn't see that. |
16:02.44 | Orbixx | ah |
16:02.48 | Orbixx | found it [TK]D-Fender |
16:03.52 | [TK]D-Fender | Orbixx: and the problem? |
16:04.28 | Jumpie | anybody ever heard of a real time invoice summary off a call? i'm trying to see if i can do this (i know itll take work, just wanna be sure its possible), i wanna make it when a call comes in or is made, it checks incoming/destination number against a customer database, if so, in addition to the regular CDR addition, i want it to be able to generate a summary of the call, time/date/ to/from, length, in a pdf and emailed |
16:05.24 | Orbixx | [TK]D-Fender: Don't see it. |
16:06.29 | drmessano | Jumpie: Of course you can |
16:06.44 | Jumpie | cool, i figured...it'd do a similar check like it does against blacklist database |
16:06.51 | Jumpie | just create another one and add/delete accordingly |
16:07.01 | Jumpie | one of the first things thast processed in a call if ir ecall |
16:07.06 | Orbixx | If it makes any difference, when I call with a SIP phone to the IVR and then select 4 for record - when I hit hash, it hangs up, when I dial from PSTN, it does nothing. |
16:07.22 | Jumpie | hit hash? maybe thast your problem, you're too stoned |
16:07.33 | drmessano | The only relevant asterisk bit is a SQL connection to a DB to parse data, and writing the CDR info.. thats very basic DB stuff.. all else you mention has 0% to do with Asterisk |
16:07.34 | Jumpie | oh....pound sign ;) |
16:07.38 | Orbixx | yeah |
16:07.41 | Orbixx | I'm a Brit :> |
16:07.45 | Jumpie | rofl im messin |
16:07.50 | Orbixx | i know |
16:08.04 | Jumpie | drmessano well i thought maybe the conversion to pdf/mail was also part of asterisk engine |
16:08.13 | drmessano | No |
16:08.21 | Jumpie | ok |
16:08.47 | drmessano | Asterisk wont write the app for you.. it will handle the call, parse your db and write your CDR.. the rest is up to your magic app |
16:10.27 | artemmakhutov | Jumpie, maybe you should try out a2billing, its a pretty nice billing solution for asterisk or check out this page: http://www.voip-info.org/wiki/view/Asterisk+billing |
16:10.31 | drmessano | As far as mail goes, Asterisk has a very limited/basic/but useful function to generate some ascii text that it pushes to a spool folder your completely configured external SMTP MTA will send off for you |
16:10.59 | drmessano | But thats a voicemail function and as close as asterisk gets to caring about mail |
16:11.14 | falz | pwd |
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18:52.42 | intel352 | hey all, I've gotten Asterisk 1.4 latest and FreePBX 2.5 latest installed on my Centos 5.3 server |
18:52.52 | intel352 | but i'm unable to connect with a sip device |
18:53.09 | intel352 | nmap shows the ports open (5060 specifically) |
18:53.17 | Dovid | what does the asterisk cli show ? |
18:53.22 | Dovid | try sip debug |
18:53.34 | intel352 | k, checking |
18:54.13 | intel352 | now that's handy |
18:55.38 | intel352 | okay, Dovid, I'm seeing where it's connecting to proxy01.sipphone.com (Gizmo5 for forwarding) |
18:55.47 | intel352 | but it doesn't show any connection from me (using X-Lite) |
18:56.26 | intel352 | ah, just saw one |
18:56.26 | intel352 | 1s |
18:56.26 | ddickenson | can anyone help me understand what this error is trying to tell me? http://pastebin.com/d5ca04f6a |
18:57.42 | Orbixx | If I have a group of people in [this] context using extensions 200-299 who have voicemail boxes 300-399, what would be the best way to construct a dial plan to let them access their voicemail boxes? |
18:58.07 | ariel_ | ddickenson, you have something configured incorrectly |
18:58.15 | intel352 | Dovid, here's the sip debug output regarding my conn: http://pastebin.org/4195 |
18:59.12 | ddickenson | ariel_: I figured this much, but this was a running production system that I needed to update dahdi and asterisk code and no changes have been made, also I have an identical "backup" server that updated with no problem. Any idea how to figure out what it is that is configured incorrectly? |
19:00.43 | ariel_ | ddickenson do dahdi_cfg -vvv to see what it's error's are from the cli |
19:01.20 | ddickenson | I was about to say I'd already done that but I guess I haven't had a cli running while I did it... hang on |
19:01.22 | ariel_ | intel352, 401 not authorized pass word or type of security would be the first to look at. |
19:02.57 | *** join/#asterisk fn0rd0 (n=fnord0@unaffiliated/fnord0) |
19:03.34 | intel352 | thx ariel, I'm checking. turns out sipdroid on my g1 was still trying to connect (which apparently has problems with authentication), while X-Lite on my computer isn't generating any response |
19:04.36 | ddickenson | ariel: yeah, no output to CLI, and no errors in the regular output from the command... |
19:05.13 | ddickenson | just this: http://pastebin.com/d7a01ac6a |
19:06.22 | ariel_ | then restart asterisk |
19:06.38 | ddickenson | tried it.. |
19:06.44 | ddickenson | just now |
19:08.44 | ddickenson | side note: intel352: How did you get sipdroid working? I can get it to connect, but I have some sort of codec error or something and I think I have everything installed |
19:09.46 | intel352 | ddickenson: not working, it's not auth'ing or something. same with x-lite. once i can find out the issue in my server conf, i'll be able to troubleshoot sipdroid :-) |
19:11.31 | ddickenson | intel352: gotcha, I had to add in a couple of extra things in my sip.conf like nat=yes and a few others that I didn't need for other phones/softphones on the same system to make it register, now I can make it ring but can't answer and if I call out I get "no compatible codecs, not accepting this offer!" |
19:11.49 | intel352 | ddickenson: interesting, i'll check that out too ;-) |
19:11.57 | intel352 | what codecs do you have setup? |
19:12.24 | ddickenson | pretty much everything but alaw, g729, and ilibc I think |
19:12.33 | *** join/#asterisk tflgen2 (n=clay@fl-67-233-27-60.dhcp.embarqhsd.net) |
19:12.38 | ddickenson | i think I got those right |
19:13.16 | ddickenson | on the business critical subject that has me working on a saturday... ariel, any other ides? |
19:15.03 | tflgen2 | what is a minimum ram req for asterisk (1-2 active calls)? also is it a .deb already? |
19:21.15 | [TK]D-Fender | tflgen2: * can run on a dumb Linksys router. |
19:21.50 | [TK]D-Fender | tflgen2: And Debian has * in the repos |
19:23.53 | tflgen2 | [TK]D-Fender: really? openwrt can support it? |
19:26.45 | ddickenson | [TK]D-Fender: Do you have any ideas on where I can go to at least figure out what that error means that I was talking to you about yesterday? I've been searching and can't come up with anything. Here's a refresher in case you don't remember http://pastebin.com/d1b1e7835 |
19:28.11 | [TK]D-Fender | ddickenson: no idea |
19:28.40 | ddickenson | dang.. |
19:29.03 | *** join/#asterisk BadHAL (n=nn@173-112-149-32.pools.spcsdns.net) |
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19:57.06 | *** join/#asterisk ManxPower (n=manxpowe@69.73.94.162) |
19:57.50 | ddickenson | how about this, can anyone tell me ALL the files chan_dahdi.so parses when it loads? |
20:01.10 | ManxPower | ddickenson: I believe it is chan_dahdi.conf and any files #include'd from that file. If you're using some sort of "Asterisk GUI" then I doubt anyone knows other than people that use that GUI. (We don't use GUIs here, those people are on a different channel) |
20:03.37 | ddickenson | ManxPower: no gui involved. I'm getting an error after recompiling new asterisk and dahdi codes. this was a working system (pure asterisk system) and I have some config error that has come up on one of the two servers I updated and am trying to narrow down where it is |
20:04.11 | ManxPower | What is the actual error message? |
20:04.40 | ddickenson | hang on a sec. lemme get on some irc client other than my phone. I just got back to office |
20:05.48 | *** join/#asterisk ddickenson (n=ddickens@67-198-0-5.static.grandenetworks.net) |
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20:06.58 | ddickenson | ManxPower: http://pastebin.com/m52603664 |
20:07.39 | ManxPower | looks to me like you might now have the DAHDI kernel modules loaded. |
20:07.48 | ManxPower | s/now/not/ |
20:07.49 | ddickenson | as you can see it's getting through chan_dahdi.conf and then to users.conf and I'm pretty sure it should get to /etc/dahdi/system.conf somewhere but that file is identical to my working server |
20:08.08 | ManxPower | no, /etc/system/dahdi.conf is read by the KERNEL drivers, not Asterisk |
20:08.18 | ddickenson | ah |
20:08.57 | ManxPower | does lsmod show any dahdi drivers loaded? |
20:09.07 | ddickenson | I didn't know that file existed... I've only written the /etc/dahdi/system.conf |
20:09.30 | ddickenson | yeah, it does |
20:09.44 | ddickenson | looks pretty normal I think |
20:09.53 | ManxPower | First you get the DAHDI kernel drivers loaded and configured correctly. Then you tell asterisk (chan_dahdi.conf) about the dahdi config. |
20:10.19 | ManxPower | what does (I think this is the correct filename) dahdi_cfg -vvv give you? (might be dahdi_config -vvv" |
20:10.27 | ddickenson | where does that happen? In the compile process? |
20:10.32 | ManxPower | no. |
20:10.36 | ManxPower | where does what happen? |
20:10.40 | *** join/#asterisk RypPn (i=TuMbL@rosscom.demon.co.uk) |
20:11.38 | ManxPower | yup, "dahdi_cfg -vvv" |
20:11.49 | ddickenson | output of dahdi_cfg -vv looks normal http://pastebin.com/m223a0e0c |
20:12.06 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
20:12.54 | ManxPower | now what is the chan_dahdi.conf contain? |
20:14.19 | ddickenson | http://pastebin.com/m1e9f99b |
20:15.11 | ManxPower | you sort of need a switchtype if you have PRI |
20:16.17 | ddickenson | I thought so too, those were commented out by digium support when I was first bringing the system online several months ago... Has been working good though.. |
20:16.39 | ddickenson | It's a nortel at CO, I've seen the stupid thing |
20:18.53 | ManxPower | try it and see |
20:19.00 | ddickenson | just uncomment? |
20:20.23 | ddickenson | actually, doesn't matter cuz I still have that other problem that won't let dahdi load (just tried anyway) |
20:21.33 | ManxPower | are you running Asterisk as root |
20:21.39 | ddickenson | yeah |
20:22.14 | ManxPower | what card do you have? |
20:22.52 | ddickenson | te410p |
20:23.58 | ManxPower | what specific kernel drivers are loaded? |
20:24.22 | ddickenson | noob questions, how do I check that? |
20:25.03 | grandpapadot | lsmod |
20:25.15 | ddickenson | I lsmod | grep dahdi but I don't know what I'm looking at |
20:25.29 | ManxPower | no, pastebing the whole output, don't filter it |
20:26.19 | ddickenson | http://pastebin.com/pastebin.php?erase=m1e9f99b |
20:26.34 | ddickenson | crap... http://pastebin.com/d4d529328 |
20:27.18 | *** join/#asterisk minsa (n=minsa@71.202.98.193) |
20:27.45 | ManxPower | I don't see anything obviously wrong |
20:28.17 | ManxPower | you recompiled asterisk after you upgraded/installed DAHDI? |
20:28.30 | ddickenson | do you see something that should correspond to my card, or is it possible that I somehow didn't install the one for my card |
20:28.41 | ManxPower | (3:27:44 PM) ManxPower: I don't see anything obviously wrong |
20:29.05 | ddickenson | yeah, I actually recompiled new libpri, dahdi, then asterisk. when that didn't work I tried recompiling old code in that order again and no go there either |
20:29.26 | ManxPower | what is the output of "grep -i "pri" /var/lib/asterisk/modules/chan_dahdo.so"? |
20:31.23 | ddickenson | http://pastebin.com/d9657dd6 |
20:32.32 | *** join/#asterisk grantm (n=grant@68.142.138.4) |
20:33.03 | ManxPower | I don't have an asterisk install handy to find the modules directoty. Maybe /usr/lib/asterisk/modules |
20:33.55 | ddickenson | Binary file /usr/lib/asterisk/modules/chan_dahdi.so matces |
20:34.09 | ddickenson | s/matces/matches |
20:34.25 | grandpapadot | omg, we just pulled off our first automated asterisk update to 46 servers using a script that was less than 25 lines, nice ... (pats self on back) |
20:34.44 | grandpapadot | Total per server downtime: < 15 seconds |
20:34.50 | ddickenson | and I can't even manage one server... ouch! |
20:34.58 | ManxPower | ddickenson: I have no more suggestions |
20:35.18 | ddickenson | ManxPower: thanks for the help anyway |
20:35.23 | grandpapadot | wget; configure; make; asterisk -rx "stop now"; make install; /etc/init.d/asterisk start |
20:35.51 | grandpapadot | oh, and a tar |
20:40.49 | *** join/#asterisk DrAk0 (n=luisjose@nelug/coreteam/luisjose) |
20:41.31 | Strogg | Yar! |
20:41.42 | Strogg | hacks on his extensions.conf |
20:46.34 | Orbixx | Is it possible to do the following... |
20:46.51 | Orbixx | ${EXTEN} should be equal to "200" for example. |
20:46.58 | Orbixx | I want to dial 300 programmatically. |
20:47.18 | Orbixx | I need to do something similar to Dial(${EXTEN}+100) |
20:47.27 | Orbixx | But that obviously doesn't work, else I wouldn't be asking. |
20:52.58 | Strogg | Can I use wav files with my asterisk dialplan or do I have to convert to ulaw? |
20:56.31 | *** join/#asterisk bbryant (n=brett@c-68-59-22-114.hsd1.sc.comcast.net) |
20:56.54 | WindowsUser | Strogg: they need to be mono and 8khz sample cycle |
20:58.02 | *** join/#asterisk lesouvage (n=lesouvag@82.73.69.76) |
20:58.07 | Strogg | how do I tell asterisk where my sound files are? |
20:58.23 | Strogg | my custom sound files.. I had someone record a bunch for me |
20:58.44 | [TK]D-Fender | Strogg: absolut path when you play them, or relative to the folder in asterisk.conf |
20:58.52 | [TK]D-Fender | astlib |
20:58.57 | Strogg | ahh ok. thanks! |
20:59.22 | ManxPower | Orbixx: If you want EXTEN to be "300" thhen use a Goto |
20:59.28 | ManxPower | Goto(300,1) |
20:59.56 | ManxPower | That is just about the ONLY way to do that other than dialing 300 from a phone. |
21:00.08 | ManxPower | EXTEN = the currently executing extension. |
21:00.55 | ManxPower | why not exten => 200,1,Goto(300,1)? |
21:02.25 | [TK]D-Fender | Orbixx: how do you Dial() "300"? 300 is not a tech or device name |
21:05.17 | ManxPower | [TK]D-Fender: It's the new chan_300! |
21:06.42 | carrar | yeah |
21:06.45 | carrar | get with it |
21:07.08 | carrar | I'm always on the phone with 300 |
21:09.34 | *** join/#asterisk L|NUX (n=linux@unaffiliated/lnux/x-10290) |
21:09.36 | L|NUX | hello |
21:10.17 | carrar | HARRO |
21:11.09 | L|NUX | i have simple but compilcated query :) |
21:12.20 | L|NUX | http://www.pastie.org/private/kabdocstsp82saf7xyxq |
21:12.26 | L|NUX | i have these rules |
21:12.38 | L|NUX | what i want to do is to access them with single DID |
21:13.05 | L|NUX | means if user call on number it will see if the number is not listed in db it will ask for PIN |
21:13.12 | L|NUX | and list their CallerID into db |
21:13.18 | L|NUX | and next time give them call back |
21:14.40 | Orbixx | Any idea why when I call internally, and enter pins and stuff via DTMF they go through fine, yet when I call via PSTN, only the first digit of the pin gets caught and the authentication fails...? |
21:16.02 | ManxPower | Orbixx: "PSTN" = ??? |
21:16.13 | ManxPower | SIP, IAX, Analog Zap, PRI, CAS T-1? |
21:16.15 | grandpapadot | Orbixx: Are your calls coming in via sip or direct through a card in your asterisk server? |
21:16.42 | Orbixx | grandpapadot: They're coming in via IAX, which eventually leads to a PSTN and a landline phone number. |
21:16.58 | Orbixx | If you see what I mean. |
21:17.35 | ManxPower | Orbixx: So calls come in via ??? -> IAX2 -> Asterisk -> ?? -> analog standalone phone |
21:18.11 | *** join/#asterisk SlicerDicer (n=SlicerDi@24.138.244.145) |
21:18.30 | Orbixx | Caller (PSTN) => Switch => Provider with number allocations => IAX2 => My Asterisk Server => IVR => SIP client |
21:18.40 | ManxPower | I guess I could list every possible problem/fix for every possible combination of "PSTN" technologies, but I don't think you have the time for that. |
21:18.48 | Orbixx | That ^ does not let anything but the first DTMF digit through. |
21:19.05 | Orbixx | SIP client => My Asterisk Server => IVR => SIP client |
21:19.06 | ManxPower | Orbixx: Caller = analog phone or cellular phone? |
21:19.11 | Orbixx | Lets all DMTF buttons through. |
21:19.15 | Orbixx | ManxPower: Either. |
21:19.25 | ManxPower | Orbixx: So the IVR is not detecting your DTMF? |
21:19.49 | Orbixx | ManxPower: I'm not so sure, because it detects the first digit correctly without flaw. |
21:20.03 | Orbixx | It just refuses to take into consideration any further digits pushed. |
21:20.12 | ManxPower | If the problem was just with Cellular I'd suggest turning on long DTMF codes, but since it also does not work from analog, then the only thing I can think of is your carrier has something screwed up. |
21:20.46 | Orbixx | And if I said I had my analog/cellular phones working with multiple DTMF digits before? |
21:20.50 | Orbixx | Config issue :> |
21:20.51 | ManxPower | Orbixx: pastebin the CLI output of a failed call? |
21:20.53 | Orbixx | I just don't know what. |
21:21.06 | Orbixx | Sure. |
21:23.14 | ManxPower | Most of the time IAX2 DTMF issues are caused by the carrier using SIP from their DID provider to their Asterisk box, then IAX2 to your Asterisk box. |
21:24.10 | Orbixx | Hmm. |
21:24.20 | Orbixx | I have a Background() and a WaitExten() |
21:24.31 | Orbixx | If I try to dial whilst Background() is executing, I get 1 digit through. |
21:24.35 | ManxPower | waits for the pastebin |
21:24.46 | Orbixx | If I wait until WaitExten() executes, nothing gets through and it times out. |
21:25.01 | Orbixx | ManxPower: I'm just saying, a pastebin doesn't say everything with Asterisk. |
21:25.46 | *** join/#asterisk swaj (n=scott@unaffiliated/swaj) |
21:25.59 | ManxPower | Orbixx: let go of my ears |
21:26.12 | Orbixx | http://pastebin.com/d6c0d5a8b |
21:26.58 | swaj | We just implemented a new Asterisk 1.4.18 PBX here, and the AMI and web interfaces are enabled. I'm try to Originate calls programatically through AMI. I'd like to ring the local extension first, and then dial an outside number. I cannot get it to work. It will ring my extension, but then it will give a busy signal when I pick up. |
21:27.10 | swaj | I can get it to call other extensions, just not outside lines |
21:27.35 | Orbixx | ManxPower: Here's the context the problem lies in: http://pastebin.com/d6741fbde |
21:27.46 | Orbixx | CLI output further up |
21:28.11 | swaj | When calling with originate, I use context=default, exten=<numbertocall>, channel=Local/5002, priority=1, action=originate |
21:28.43 | swaj | if I reverse things, I.E. channel=Zap/g2/<number>, exten=5002 -- it will work, but it calls out first |
21:28.45 | *** join/#asterisk jmacz (n=mcorb@201.244.169.20) |
21:29.01 | swaj | what I want it to do is ring my extension first and then let me listen while it calls out |
21:29.02 | ManxPower | Orbixx: Now pastebin your [dialextension] context |
21:29.10 | Orbixx | I just did. |
21:29.17 | Orbixx | http://pastebin.com/d6741fbde |
21:30.02 | swaj | any help would be greatly appreciated |
21:30.36 | ManxPower | Orbixx: that is the entire context? |
21:30.40 | Orbixx | Yes. |
21:31.17 | ManxPower | swaj: chan_local requires Local/extension@CONTEXT |
21:31.31 | swaj | ManxPower: well I've tried using SIP too |
21:32.04 | [TK]D-Fender | swaj: Channel: should point to your phone, and Context, Exten, Priority to an exten that will dial out |
21:32.10 | ManxPower | Orbixx: your dialplan looks correct. Iax has virtually no options for DTMF since IAX2 doesn't have DTMF issues. Have you tried using SIP? |
21:33.06 | Orbixx | No, I'll give that a shot. |
21:33.14 | swaj | [TK]D-Fender: so if I want to dial out from a local phone, I should do Action=Originate, Channel=Local/5002@DID_span_3, Priority=1, exten=8005551212 ? |
21:33.56 | ManxPower | swaj: have it dial your phone, then handle everything else in the dialplan |
21:34.26 | ManxPower | Orbixx: I suspect if you do an iax2 debug you'll see that the carrier is not sending all the digits. |
21:35.44 | ManxPower | swaj: Define "dial out from the local phone"? The only you can "dialout from a phone" is to pick up the phone and dial. |
21:36.05 | swaj | ManxPower: I guess I'm not understanding. All I really want to do is call AMI or the HTTP interface (rawman) and have it ring my extension, then dial an external number |
21:36.41 | ManxPower | swaj: I believe the term is "click-to-call" |
21:37.12 | ManxPower | swaj: look into .call files rather than using AMI. I don't know which way is better for you. |
21:37.28 | swaj | ManxPower: yes, basically I'm trying to write a one-touch-dial program where a user can click a button on their screen and it will dial their phone for them. |
21:37.50 | ManxPower | sounds like you want to use AMI to generate a call with 2 legs. the first leg is your phone, the second leg is a PSTN number. |
21:38.03 | Orbixx | ManxPower: iax2 debug output: http://sprunge.us/gHCC |
21:38.33 | swaj | ManxPower: yes basically. Right now I can make it call other extensions... for example, if I set Channel to Local/5002 and Exten to 5001, it will ring my extension at 5002, and then ring the person at 5001 |
21:39.05 | swaj | ManxPower: and I can make it work in reverse... setting exten to 5002 and channel to Zap/g2/8005551212 will call 8005551212 first, wait for them to answer, and then ring my extension |
21:39.32 | ManxPower | Orbixx: looks to me like you may be getting a max of 2 DTMFs from your carrier |
21:39.47 | ManxPower | swaj: how are you connecting to the PSTN? |
21:39.49 | [TK]D-Fender | Sw* calls the CHANNEL first |
21:39.53 | [TK]D-Fender | swaj: * calls the CHANNEL first |
21:40.00 | [TK]D-Fender | swaj: THAT should point to your phone |
21:40.34 | *** join/#asterisk serph (n=serph@70.50.133.67) |
21:40.45 | swaj | [TK]D-Fender: right! so what I want to do is Channel = Local/5002, Exten = outside number, but when that happens, I get a busy signal after answering the phone, like it never completes the call |
21:41.29 | ManxPower | Orbixx: Looks to me like you are not getting ANY DTMF from your carrier. |
21:41.30 | swaj | ManxPower: we have I think 2 PRI's coming from the phone company with a T1. We're using SIP. |
21:41.35 | [TK]D-Fender | swaj: Show me the full originate and your dialplan. |
21:41.42 | ManxPower | swaj: as long as it's not analog |
21:42.13 | Orbixx | ManxPower: That cannot be correct, as I have to choose an extension to get to the context I am having a problem with. |
21:42.35 | Orbixx | (and choosing said extension works with DTMF) |
21:43.33 | [TK]D-Fender | Orbixx: I never saw the failed call |
21:43.57 | Orbixx | [TK]D-Fender: Hmm? |
21:45.15 | swaj | [TK]D-Fender: the originate is this-> http://10.7.30.21/rawman?action=originate&channel=Local/5002&exten=6363573061&priority=1 |
21:45.16 | swaj | [TK]D-Fender: unfortunately I don't know what you mean by dialplan |
21:45.16 | Orbixx | Good job, give him a local IP why don't you. |
21:45.16 | ManxPower | Orbixx: Looks to me like the carrier is sending the DTMF begind and END frames but no actual dtmf frames. |
21:45.37 | [TK]D-Fender | swaj: You are not specifying CONTEXTs anywhere! |
21:45.43 | [TK]D-Fender | SwDialplan = EXTENSIONS.CONF |
21:45.54 | *** part/#asterisk ManxPower (n=manxpowe@69.73.94.162) |
21:46.02 | swaj | [TK]D-Fender: I use &context=default |
21:47.12 | [TK]D-Fender | swaj: In that URI you specified nothing. |
21:47.25 | [TK]D-Fender | swaj: pastebin a failed attempt an your dialplan. |
21:47.26 | [TK]D-Fender | ~pb |
21:47.27 | infobot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste , or , http://bin.cakephp.org/, or apt-get install pastebinit |
21:47.32 | swaj | [TK]D-Fender: http://pastie.org/558944 |
21:48.30 | [TK]D-Fender | swaj: [default] has no match for the # you are dialing <- |
21:51.16 | swaj | [TK]D-Fender: I think I'm starting to understand |
21:52.04 | swaj | [TK]D-Fender: I noticed my phone is set to use DialPlan1 -- is there a way to originate using that Dial Plan through AMI? |
21:52.46 | [TK]D-Fender | swaj: your local channel should specify the context, as should your context field. |
21:53.09 | [TK]D-Fender | [17:31]<ManxPower>swaj: chan_local requires Local/extension@CONTEXT <--- |
21:53.37 | [TK]D-Fender | swaj: and ManxPower already told you how to properly format that line |
21:56.47 | swaj | sigh |
21:57.26 | swaj | http://10.7.30.21/rawman?action=originate&channel=Local/5002@numberplan-custom-1&exten=6363573061&priority=1&context=numberplan-custom-1 |
21:57.40 | swaj | I used the plan in extensions.conf |
21:58.49 | swaj | wait I got it |
21:58.49 | swaj | need the 91 in front of the exten |
22:03.46 | *** join/#asterisk lesouvage (n=lesouvag@82.73.69.76) |
22:06.40 | *** join/#asterisk MatBoy (n=MatBoy@wiljewelwetenhe.xs4all.nl) |
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22:38.56 | Dovid | ~pb |
22:38.57 | infobot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste , or , http://bin.cakephp.org/, or apt-get install pastebinit |
22:39.00 | Dovid | ~google |
22:39.01 | infobot | hmm... google is http://lmgtfy.com/?q=google |
22:39.56 | Dovid | ~trfm |
22:40.02 | Dovid | ~rtfm |
22:40.02 | infobot | hmm... rtfm is Read The F*cking Manual (TM). It is a suggestion to do your homework before posting a question. Sometimes used as RTFM $SPECIFIC_MANUAL to refer to a specific source of information. See also http://en.wikipedia.org/wiki/RTFM |
22:55.53 | *** join/#asterisk sjobeck (n=Adium@208-151-246-203.dq1sn.easystreet.com) |
23:06.51 | WindowsUser | ~spam |
23:06.52 | infobot | it has been said that spam is a preferred environment. SPAM; Shut up, You damn Vikings! SPAM SPAM SPAM SPAM SPAM SPAM SPAM SPAM SPAM SPAM SPAM SPAM SPAM SPAM SPAM SPAM SPAM SPAM, or to destroy it, try spamassassin (spamd+spamc) and razor. |
23:24.27 | *** join/#asterisk errotan (n=errotan@5403E42C.catv.pool.telekom.hu) |
23:25.33 | *** join/#asterisk Alfio (n=Amunoz@adsl-54-26.tricom.net) |
23:29.29 | *** join/#asterisk ecret (n=ecret@CPE001e9002348e-CM001225d8ab30.cpe.net.cable.rogers.com) |
23:38.31 | *** join/#asterisk shinao1 (n=shinao1@41.219.241.80) |
23:51.03 | *** join/#asterisk alrs (n=lars@170.206.224.54) |