00:04.12 | *** join/#asterisk |dennis| (i=dennis@200.32.215.84) |
00:07.50 | mog_work | it builds here MRH2 |
00:07.53 | mog_work | trunk does not though |
00:07.55 | mog_work | at the moment |
00:16.11 | *** join/#asterisk IceManRISK (n=kart@200.138.147.142) |
00:16.23 | MRH2 | ok this time it worked how wierd |
00:16.47 | MRH2 | (weird) |
00:19.42 | *** join/#asterisk kamileon (n=kamileon@68.62.190.253) |
00:23.21 | *** part/#asterisk CoffeeIV_ (n=CoffeeIV@64.149.168.97) |
00:25.45 | websae | heloo, is this thing on? |
00:25.47 | websae | *hello |
00:26.01 | websae | anyone getting these RTP packets... |
00:26.01 | mog_work | no |
00:26.05 | websae | didn't think so |
00:27.03 | mog_work | this being irc |
00:27.06 | mog_work | its difficult |
00:27.28 | websae | chuckles |
00:27.42 | websae | mog_work where are you from? |
00:28.03 | mog_work | huntsville alabama |
00:28.19 | websae | that's right |
00:28.33 | orlok | ok, i want to kill the fucker esponsible for sorbs |
00:29.55 | MRH2 | ok definately all working must have been something from the prev install it didn't like |
00:30.02 | MRH2 | thanx mog |
00:30.46 | *** join/#asterisk alephcom (n=alephcom@host75.net14.mcsnet.ca) |
00:32.07 | xachen | Heya there aleph :) |
00:32.31 | alephcom | greetings xachen |
00:32.32 | xachen | orlok: What did SORBS do to you? |
00:33.44 | xachen | :) |
00:34.29 | xachen | Anybody know if Rackmountetc boxes are any good? |
00:36.06 | *** join/#asterisk demigod2k (n=joey@cpe-24-210-97-162.twmi.res.rr.com) |
00:36.16 | demigod2k | hi |
00:36.27 | *** part/#asterisk willt (i=wt@wifi-napanet-static-206-81-99-68.napanet.net) |
00:37.24 | *** join/#asterisk Ariel_ (n=Ariel@dsl-20-177.cofs.net) |
00:38.14 | orlok | xachen: Hmm. Listed our static ranges |
00:38.25 | orlok | xachen: then, they list our ISP's SMTP server |
00:38.55 | xachen | heh |
00:39.05 | xachen | I do my part and refuse to use BL's |
00:39.09 | orlok | yeah |
00:39.15 | orlok | one of the big isp's here uses sorbs though |
00:39.27 | *** join/#asterisk vexorg (n=vexorg@CPE000021ded913-CM001225419164.cpe.net.cable.rogers.com) |
00:39.29 | orlok | its funny, the guy that started sorbs actuall works for somebody we do business with |
00:39.31 | xachen | instead I just use Spamassassin/Dspam |
00:39.33 | xachen | works good ;) |
00:39.35 | orlok | yeah |
00:39.38 | orlok | and greylisting |
00:39.53 | orlok | do some smart/funky stuff on the secondary mx's too |
00:39.53 | xachen | I block a few fraud countires |
00:39.53 | xachen | but thats it |
00:39.55 | alephcom | xachen: Do you still like Telus? :-) |
00:39.58 | xachen | Malaysia, Romania |
00:39.58 | demigod2k | those BLs cause me nothing but trouble. I publish SPF records and that's about it |
00:40.01 | xachen | alephcom: I hate it |
00:40.21 | demigod2k | the store-and-forward thing looks like it'd be worthwhile, but other than that just too many rejections |
00:40.27 | alephcom | They changed a bunch of "static" addresses last week with absolutely no warning. |
00:40.33 | xachen | oh nice |
00:40.47 | xachen | the ISP i'm getting NATs its customers onto like 3 public IPs |
00:40.49 | demigod2k | those turds. they should use a DNS record |
00:41.07 | *** join/#asterisk e-milio (n=emilio@pmr.pmrtechnologies.com) |
00:42.02 | e-milio | hello everybody |
00:42.04 | *** join/#asterisk websae (n=websae@CPE-24-167-206-22.wi.res.rr.com) |
00:42.28 | e-milio | have a quick question with eagi perl sayunixtime |
00:42.51 | xachen | anyways I'm going for dinner |
00:42.52 | xachen | bbiab |
00:43.41 | e-milio | this is not working: $AGI->exec('SayUnixTime', sprintf("%s||ABdY \'digits/at\' HM", UnixDate($data, "%s"))); |
00:44.03 | alephcom | xachen: They check sites you visit too? |
00:45.03 | e-milio | if i use default format works |
00:45.34 | e-milio | but if i tried to use dif from default dont work |
00:45.48 | e-milio | anbody can elighten ?? |
00:49.23 | *** join/#asterisk riddlebox (n=james@24-171-10-102.dhcp.stls.mo.charter.com) |
00:49.26 | *** join/#asterisk terrapen (n=cjs@166.70.183.109) |
00:50.54 | terrapen | anybody seen this before? |
00:50.56 | terrapen | http://pastebin.com/693403 |
00:51.12 | terrapen | As soon as I dial an extension on the PBX at the other side of this PRI, it hangs up |
00:51.27 | terrapen | funny thing is, I had it working this morning. Not sure how I broke it |
00:53.50 | terrapen | sigh |
00:54.29 | *** join/#asterisk zagaya971 (n=almeli@APointe-a-Pitre-102-1-3-9.w81-248.abo.wanadoo.fr) |
01:01.57 | *** join/#asterisk surfdue (n=tyler@unaffiliated/surfdue) |
01:01.58 | surfdue | hey |
01:02.10 | surfdue | why would asterisk be playing a no service announcement ? |
01:02.14 | surfdue | whats a possiblity |
01:02.22 | *** join/#asterisk moprilo (n=jjohn@201.198.78.23) |
01:02.55 | Strom_C | surfdue: I'm going to guess there's no serice |
01:02.58 | Strom_C | er, serice |
01:03.02 | Strom_C | service |
01:03.03 | moprilo | hi, i've been using asterisk for a while now, i have xorcom on my main servers, but asterisk@home on the small offices, I'm looking for an alternative to asterisk@home |
01:03.04 | surfdue | Strom_C, how? |
01:03.05 | moprilo | any ideas? |
01:03.14 | surfdue | Strom_C, i can see in asterisk -r that it connects and all |
01:03.16 | Strom_C | surfdue: what is the text of the announcement? |
01:03.22 | surfdue | Strom_C, asterisk is just playing a message lol. |
01:03.29 | surfdue | there is no service. |
01:03.39 | surfdue | <PROTECTED> |
01:03.40 | Strom_C | surfdue: yes, please transcribe exactly what the recording is saying |
01:03.41 | surfdue | :P |
01:03.55 | Strom_C | ah ok |
01:03.57 | Strom_C | *shrug* |
01:05.09 | *** join/#asterisk harryvv (n=none@S010600a0c93f6f7e.vs.shawcable.net) |
01:09.29 | *** join/#asterisk trbldwine (i=trbldwin@c-71-194-161-170.hsd1.il.comcast.net) |
01:11.48 | orlok | nemith: wireless isp? |
01:11.55 | orlok | man i've dealt with some shit wisp's |
01:12.09 | *** join/#asterisk MacWeenie (n=chatzill@82-35-73-28.cable.ubr02.dals.blueyonder.co.uk) |
01:13.47 | e-milio | Hello |
01:14.13 | e-milio | I and trying to sayunixtime |
01:14.26 | e-milio | but it always says one hourearlier |
01:14.34 | e-milio | than the time that i give to it |
01:14.54 | e-milio | please, anyboyd has anyidea ? |
01:22.17 | dlynes | tainted-, qwell: seems like the problem might be the machine for that error it's spitting up, not the card |
01:23.25 | dlynes | e-milio: what's your time zone? |
01:24.38 | dlynes | e-milio: EDT, right? |
01:26.36 | e-milio | yes |
01:26.54 | e-milio | dlynes: I have just changed it to ESt |
01:27.05 | e-milio | dlynes: but same problem |
01:27.55 | dlynes | e-milio: um....you should be EDT |
01:28.04 | dlynes | e-milio: EST was about a month ago |
01:28.35 | dlynes | e-milio: what does it say when you type 'date' at a linux prompt? |
01:29.57 | e-milio | Mon May 1 21:33:45 EDT 2006 |
01:30.22 | e-milio | so it is right now |
01:31.24 | e-milio | but still |
01:31.36 | e-milio | dlynes: $AGI->exec('SayUnixTime', sprintf("%s", UnixDate($data, "%s"))); |
01:31.53 | IceManRISK | boa garoto |
01:32.33 | dlynes | ummm |
01:32.46 | dlynes | nvm |
01:33.04 | e-milio | ? |
01:34.54 | dlynes | e-milio: in agi, can you specify parameters to 'SayUnixTime'? |
01:35.15 | e-milio | dlynes: I tried but doesn't seem so |
01:35.44 | e-milio | dlynes: i just rejects the date and 'says' today |
01:36.11 | dlynes | e-milio: like maybe $AGI->exec('SayUnixTime(,EDT)', sprintf("%s", UnixDate($date, "%s"))) ; |
01:36.22 | dlynes | e-milio: btw...what's "UnixDate"? |
01:36.53 | e-milio | it returns the num of mins from epoch |
01:37.00 | *** join/#asterisk bjohnson (n=bjohnson@i216-58-62-76.cybersurf.com) |
01:37.06 | e-milio | seems standard |
01:37.34 | dlynes | e-milio: in perl? |
01:37.38 | dlynes | e-milio: or C? |
01:37.47 | dlynes | e-milio: or? |
01:37.50 | e-milio | at least in AGI, on all examples |
01:38.05 | dlynes | e-milio: what language are you writing AGI in? |
01:38.10 | e-milio | perl |
01:38.14 | dlynes | ah |
01:41.42 | e-milio | dlynes: didnt work |
01:41.59 | e-milio | dlynes: sure about syntax ? |
01:42.02 | gaupe | e-milio: number of seconds |
01:42.24 | e-milio | gaupe: yes, your are right |
01:43.37 | dlynes | e-milio: Nope, I am not |
01:43.42 | demigod2k | opinion. buy all polycom 301s or all gxp2000s ? |
01:43.58 | Strom_C | why in god's name would you even consider grandstream sets? :P |
01:44.00 | dlynes | e-milio: but I've never used the UnixDate function in perl, either |
01:44.11 | demigod2k | the gxp has tons of buttons to assign to features |
01:44.19 | dlynes | If it was a choice between grandstreams and polycoms |
01:44.19 | demigod2k | also the 301 has 2 line appearances while the gxp has 4 |
01:44.28 | dlynes | I wouldn't even think about the grandstreams |
01:44.38 | Strom_C | also the grandstream will fall apart if you sneeze on it |
01:44.53 | demigod2k | anything else in that pricerange is fine too but I just own those two to compare so far |
01:44.56 | dlynes | Unfortunately, for a lot of my customers, polycoms aren't even an option; they're too cheap to pay for them |
01:45.09 | demigod2k | ya the gxp does sort of feel like crap |
01:45.16 | demigod2k | I also really, really hate the binary configuration file |
01:45.30 | dlynes | demigod2k: sipura's same crap |
01:45.49 | e-milio | I have had very good experiences with polycom |
01:45.58 | e-milio | many callcenters working for many time |
01:46.19 | demigod2k | I have one polycom for testing. it seems ok so far, nice provisioning files, nice ringtones, just the fewest buttons and line appearances |
01:46.33 | e-milio | i agree |
01:46.35 | demigod2k | and the manual claims no paging |
01:46.39 | demigod2k | I havent tried though |
01:48.14 | harryvv | dam door bell hold on |
01:48.17 | *** part/#asterisk surfdue (n=tyler@unaffiliated/surfdue) |
01:51.21 | e-milio | dlynes:thanks anyway |
01:51.56 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
01:54.22 | SplasPood | demigod2k: define "paging" ... You can set the polycom up to autoanswer on a given ring type sip header |
01:54.37 | demigod2k | that would be my definition |
01:54.54 | SplasPood | yea works fine.. howto on voip-info.org |
01:55.17 | demigod2k | said in the manual book that it wouldnt do it, but I had heard it in here once before |
01:55.33 | demigod2k | the manual may be phone-to-phone without a server, or something |
02:03.02 | *** join/#asterisk surfdue (n=tyler@unaffiliated/surfdue) |
02:03.04 | surfdue | hi |
02:08.27 | *** join/#asterisk Eggplant (i=No@dsl-332.cascadeaccess.com) |
02:09.59 | mog_work | w |
02:11.49 | *** join/#asterisk MacDome (n=eseidel@c-71-198-177-144.hsd1.ca.comcast.net) |
02:16.54 | DoktorGreg | ok, any suggestions on which mobo i need to get 100% in zttest? |
02:18.32 | DoktorGreg | pri + musiconhold = dropped call at 98.75% zttest results.... |
02:18.34 | surfdue | having problem with linksys pap2 im getting the signal from asterisk its gooing in and out for some reason ? I have a oruter but the ports are unblocked :| |
02:19.09 | demigod2k | bummer. I get around 98% or so |
02:19.21 | demigod2k | I figured that was natural |
02:19.24 | DoktorGreg | docs say i should shoot for 100 |
02:19.34 | DoktorGreg | pri is bridging just fine |
02:19.47 | DoktorGreg | sip phones hit all the features just fine |
02:19.57 | demigod2k | makes sense. I bought an off-the-shelf box, no clue what mobo. mine gets that 98% or so |
02:20.01 | DoktorGreg | pri cant hit the features |
02:20.06 | surfdue | oh disabled firewall and it works :| |
02:20.11 | surfdue | not safe though lol oh well. |
02:20.59 | DoktorGreg | Do i just need to burn the cash and get smp for that last 1.25%? |
02:21.11 | DoktorGreg | and |
02:21.22 | DoktorGreg | do i need actual smp mobo? |
02:21.33 | DoktorGreg | or can i get one of the fancy new multi core cpus? |
02:21.49 | DoktorGreg | oh |
02:21.53 | DoktorGreg | and there blows my budger |
02:21.58 | DoktorGreg | budget |
02:23.13 | demigod2k | no idea. mine is only about 4 lines |
02:23.16 | demigod2k | and a 1.0 ghz machine |
02:23.24 | *** part/#asterisk alephcom (n=alephcom@host75.net14.mcsnet.ca) |
02:23.24 | xachen | alephcom: As far as I know they are known to do that |
02:23.26 | xachen | blah |
02:23.27 | xachen | haha |
02:23.34 | xachen | talk about timing |
02:24.07 | DoktorGreg | lol, i must need different pci chipset |
02:24.24 | DoktorGreg | cpu is amd is 2200 thingie |
02:24.47 | demigod2k | or different PCI cards on your bus or something |
02:24.50 | demigod2k | maybe something doesn't play fairly |
02:24.58 | DoktorGreg | and i have shut everything off, except for on mobo lan |
02:24.59 | *** join/#asterisk Strom_C (n=strom@gateway.digium.com) |
02:25.12 | DoktorGreg | i even took out the graphics card and ran it headless |
02:25.26 | demigod2k | theres a possibility too, check if you have a shit chipset |
02:25.52 | DoktorGreg | iirc nforce2 |
02:26.06 | DoktorGreg | msi motherboard |
02:26.39 | DoktorGreg | i wonder if i should just build a 4 way server |
02:27.00 | DoktorGreg | naw |
02:27.09 | demigod2k | I doubt it's necessary, the processor isnt that huge a deal really |
02:27.30 | demigod2k | something with clean interrupts seems most important from what I've read. although my system is tiny compared to others |
02:27.31 | DoktorGreg | well digium does suggest smp may be needed |
02:28.05 | demigod2k | could be I've never had that large a configuration. I just imagine you hit the point of diminishing returns |
02:28.27 | *** join/#asterisk brockj49464 (n=brockj49@63.87.56.236) |
02:28.39 | DoktorGreg | well generally on big servers the idea is |
02:28.53 | DoktorGreg | you can run more threads |
02:29.09 | DoktorGreg | and server software is generally multi threaded |
02:29.10 | demigod2k | assuming the linux multiprocessor code is good which is a big question |
02:29.27 | demigod2k | I'd just make damn sure that every single card on the bus has the worlds best driver first |
02:29.41 | demigod2k | the fastest processor in the world won't make up for something eating up that 33 mhz bus |
02:29.42 | DoktorGreg | I have had better results with linux on servers than i have had with windows... |
02:30.00 | demigod2k | I wouldn't say windows does any better with multiprocessor |
02:30.07 | demigod2k | nothing does well enough to justify the cost if you ask me |
02:30.50 | DoktorGreg | well i have a dual 1 ghz p3 laying around |
02:30.58 | DoktorGreg | I could try it on that |
02:31.20 | demigod2k | eh maybe. I bet replacing that nforce2 based NIC with something decent would do more |
02:31.38 | DoktorGreg | yah... |
02:31.43 | demigod2k | I wouldnt be surprised if you ran into more problems than ever with a dual processor |
02:32.11 | DoktorGreg | i consider this a bump in the road to be solved |
02:33.10 | DoktorGreg | so tomorrow i try different nic |
02:33.21 | DoktorGreg | at least i figured out my problem with my ata today |
02:33.23 | demigod2k | ya its a start |
02:33.46 | DoktorGreg | lol, problem with my ata was me assuming i knew better than default settings |
02:34.46 | DoktorGreg | i had messed with a bunch of things in the nat config on the ata |
02:35.02 | DoktorGreg | when all i had to do was say, nat=yes |
02:35.14 | DoktorGreg | stun server = my stun server |
02:35.21 | DoktorGreg | use stun = yes |
02:35.36 | dlynes | Anyone know where the Makefile.patch file is for spandsp 0.0.3? |
02:36.25 | DoktorGreg | can i get a mobo with multiple pci busses? |
02:36.28 | *** join/#asterisk Alystair (n=bob@CPE001109c15241-CM00407b8794db.cpe.net.cable.rogers.com) |
02:36.38 | Alystair | Anyone here heard of Thirdlane? |
02:36.39 | dlynes | DoktorGreg: you mean like pci-x and pci? |
02:37.37 | DoktorGreg | um kinda |
02:37.56 | *** join/#asterisk linlin (n=linlin@c-67-184-230-198.hsd1.il.comcast.net) |
02:38.07 | DoktorGreg | can i get 2x+ pci busses on the same mobo? |
02:38.28 | *** join/#asterisk linlin (n=linlin@c-67-184-230-198.hsd1.il.comcast.net) |
02:40.58 | DoktorGreg | ok here i go, found some more stuff to tru |
02:41.01 | DoktorGreg | try |
02:41.07 | DoktorGreg | scarry |
02:41.13 | DoktorGreg | im gonna build a kernel remotely |
02:41.49 | xachen | I do that all the time |
02:42.02 | xachen | nothing scary about it.... except when you get it perfect except build the wrong NIC module |
02:43.30 | DoktorGreg | well im gonna use old config file |
02:43.56 | DoktorGreg | official docs call for 2.4.20 kernel |
02:44.13 | *** join/#asterisk JunK-Y (n=junky@modemcable205.175-81-70.mc.videotron.ca) |
02:44.13 | DoktorGreg | and i have a 2.4.27 kernel |
02:44.24 | *** join/#asterisk angom_h (n=angom@red-corp-200.79.134.173.telnor.net) |
02:44.39 | DoktorGreg | so my first thing to do is an apt get |
02:45.11 | DoktorGreg | er rather wget |
02:49.35 | *** join/#asterisk speedracer (n=collin@24.96.142.189) |
02:50.16 | *** join/#asterisk fami (n=fami@unaffiliated/pmai) |
02:51.11 | tainted- | anyone need colocation |
02:51.21 | speedracer | free? |
02:51.30 | speedracer | lol |
02:51.32 | speedracer | hey file |
02:51.36 | file | hi! |
02:52.03 | xachen | tainted-: TEll ya what, I'll talk colocation if you pay me $50/mo |
02:52.26 | speedracer | lol |
02:52.34 | xachen | I think thats fair |
02:52.48 | *** join/#asterisk trig_hm (i=jason@home.monkeypr0n.org) |
02:53.58 | Netgeeks | anyone ever seen a SIP/2.0 603 Declined message before from asterisk? |
02:54.04 | speedracer | Okay, so I'm a FC nub...as well as an Asterisk nub I guess, and I'm trying to compile Asterisk. I'm starting with Zaptel, and `make clean` works fine, but `make` exits with an error about the sources for my kernel not being installed...but at the same time, yum tells me they are... |
02:54.24 | speedracer | any ideas? |
02:54.47 | Netgeeks | speedracer there is a text file in the zaptel distribution that tells you what you need to do to fix that |
02:55.15 | Netgeeks | don't remember off the top of my head what it's called but it's there |
02:55.27 | file | speedracer: you should call support tomorrow ^_^ |
03:00.10 | speedracer | file: when? I'll be at work at the same time they are... |
03:00.14 | speedracer | no good |
03:00.47 | file | grab someone from support and offer to buy them lunch if they get your stuff working :P |
03:00.57 | speedracer | lol |
03:01.00 | speedracer | if only it were that easy |
03:01.01 | speedracer | :-\ |
03:01.13 | speedracer | i figured I could get some help in here |
03:01.16 | file | dooooooooooo it |
03:04.38 | russellb | file: shush you |
03:05.21 | file | russellb: do I have to? :( |
03:05.27 | russellb | file: YES |
03:05.38 | russellb | and go get me something to drink! |
03:05.47 | file | you're not my boss! :P |
03:06.05 | russellb | gooood! |
03:06.08 | russellb | that would be weird |
03:06.17 | file | or what if I was YOUR boss... |
03:06.23 | russellb | NEVAR |
03:06.39 | *** join/#asterisk doughecka (n=Miranda@unaffiliated/doughecka) |
03:06.43 | orlok | bloody. sorbs. |
03:06.48 | *** join/#asterisk tessier_ (n=treed@adsl-75-5-99-178.dsl.sndg02.sbcglobal.net) |
03:06.49 | dlynes | Does anyone have spandsp 0.0.3 working with asterisk 1.2.7.1? |
03:07.33 | dlynes | I've decided to try giving it another go, but the Makefile.patch doesn't seem to exist in either |
03:07.58 | dlynes | Or the app_txfax.so and app_rxfax.so, for that matter |
03:08.03 | dlynes | erm .c i mean |
03:09.50 | *** join/#asterisk dahunter3 (n=dahunter@pool-71-110-89-49.lsanca.dsl-w.verizon.net) |
03:15.08 | *** join/#asterisk OloBola (n=not@netblock-68-183-67-158.dslextreme.com) |
03:15.52 | Alystair | When a phone is single line does that mean you cannot do a call waiting kinda setup? |
03:16.14 | Alystair | (like pressing flash to switch conversations on normal phone when you get another call) |
03:18.50 | dlynes | Alystair: sure you can, if you have call waiting subscriber service |
03:19.08 | dlynes | Alystair: it all depends on what line subscription services you have on that line |
03:20.15 | Alystair | you mean from the DID provider |
03:20.34 | Alystair | or an internal asterisk setup kinda thing |
03:23.45 | *** join/#asterisk CunningPike (n=arodgers@S010600095b33697f.vc.shawcable.net) |
03:31.39 | *** join/#asterisk Iaxy (n=Iaxy@modemcable236.55-131-66.mc.videotron.ca) |
03:31.52 | Iaxy | Hi all |
03:32.37 | Iaxy | I just installed asterisk for buisness and there is no /etc/zaptel |
03:32.47 | Iaxy | Did they omit that? |
03:33.00 | CunningPike | Iaxy: I have to ask - did you install zaptel? |
03:34.43 | Iaxy | it is a binary version of Asterisk, and there is one install script. I will check the book again. but thats what I read |
03:38.02 | *** join/#asterisk MacDome (n=eseidel@c-71-198-177-144.hsd1.ca.comcast.net) |
03:38.05 | CunningPike | Iaxy: I'm afraid I've never installed ABE |
03:38.29 | CunningPike | But I would check to see if zaptel is included |
03:38.50 | CunningPike | It's /etc/zaptel.conf you're looking for, right? |
03:39.02 | Iaxy | CunningPike: yup. its not there |
03:39.28 | CunningPike | Hmm |
03:39.48 | *** join/#asterisk r0d3nt (n=RatMan@foster.stonedcoder.org) |
03:39.54 | CunningPike | Double-check the docs - I don't want to advise you install the OSS version of zaptel if it will break your suppoirt |
03:40.36 | Iaxy | hehe what good is ABE if there is no zaptel and I got a quad T card in it...:-) |
03:41.56 | *** part/#asterisk speedracer (n=collin@24.96.142.189) |
03:42.04 | Iaxy | it says to uninstall Asterisk run rpm -e asterisk libtonezone libpri |
03:42.20 | Iaxy | where the hell is zaptel? |
03:42.59 | *** join/#asterisk junbug (i=junya@67.191.62.53) |
03:43.01 | *** join/#asterisk _x0r (n=dusty@12-219-148-217.client.mchsi.com) |
03:43.31 | *** join/#asterisk salviadud (n=dude@dsl-201-129-86-188.prod-infinitum.com.mx) |
03:43.47 | salviadud | dammit, nobody's from brazil |
03:44.35 | CunningPike | Iaxy: This is ABE, right? |
03:45.11 | Iaxy | CunningPike: yes Asterisk Buisness Edition |
03:45.31 | CunningPike | Well, I would contact Digium - you've paid for it :) |
03:45.47 | salviadud | yeah man, who pays for asterisk anyways? |
03:45.51 | salviadud | i thought it was free... |
03:46.03 | salviadud | i'm using it... it tastes like it's free |
03:46.07 | Iaxy | Companies that feel better when they pay. |
03:46.26 | salviadud | well, you get what you pay for |
03:46.32 | salviadud | call the dudes at digium |
03:46.49 | salviadud | they outta be nicer than the guys at broadvoice |
03:46.52 | salviadud | or so i've heard |
03:47.21 | Iaxy | This is a bitch.... |
03:47.42 | salviadud | somegeek, what's wrong eh? |
03:47.53 | salviadud | oh.. damn xchat |
03:48.01 | russellb | Iaxy: zaptel isn't installed as an RPM on ABE |
03:48.02 | salviadud | i meant to say, so, whats wrong eh? |
03:48.19 | russellb | since it's very specific to the kernel you have, it has to be built on the system |
03:48.44 | russellb | Iaxy: however, the source is included on the CD, and the install script will install it for you automatically. |
03:48.53 | *** part/#asterisk _x0r (n=dusty@12-219-148-217.client.mchsi.com) |
03:49.30 | russellb | or at least it is supposed to. |
03:49.48 | russellb | Iaxy: if you would like, I could log in to your system and fix it up for you ... |
03:49.51 | Iaxy | the install script did not install it. |
03:49.51 | *** part/#asterisk justinu|laptop (n=Justin@dsl081-083-093.lax1.dsl.speakeasy.net) |
03:50.15 | Iaxy | I don't think I put the kernel sources in the dist install |
03:50.18 | Iaxy | hmmm... |
03:50.35 | russellb | so maybe the script tried to install it and failed |
03:50.49 | russellb | it's supposed to handle dependency checking, but i haven't touched that script in about a year |
03:50.58 | Iaxy | It should have told me. |
03:51.08 | *** join/#asterisk bmg505 (n=leon@c1-175-1.rndf.isadsl.co.za) |
03:51.19 | Iaxy | Let me look at the script |
03:51.28 | *** join/#asterisk TheCops (i=nobody@got.securebinary.com) |
03:51.40 | russellb | Iaxy: just untar the source that's on the cd and install it manually |
03:55.02 | Iaxy | it supposed to have installed it but it didn't. |
03:55.45 | CunningPike | Iaxy: If you do what russellb says, you might get an error you can work with |
03:56.57 | Iaxy | I just installed bear minimum. Thats alot of stuff I am going to have to do to get it to compile. |
03:57.47 | russellb | well, you can contact Digium support if you would like help doing it. |
03:57.47 | CunningPike | Probably why the install didn't work off the CD then |
03:57.57 | CunningPike | What he saisd |
03:58.33 | *** join/#asterisk Johnnie (n=jdlewis@dynamic-acs-24-154-91-195.zoominternet.net) |
03:58.42 | Iaxy | yup.... |
04:01.30 | Iaxy | I added all the requirements. |
04:03.09 | Iaxy | tar: zaptel-be/zconfig.h: time stamp 2005-09-13 16:34:50 is 7380521 s in the future |
04:03.09 | Iaxy | zaptel-be/zonedata.c |
04:03.10 | Iaxy | tar: zaptel-be/zconfig.h: time stamp 2005-09-13 16:34:50 is 7380521 s in the future |
04:03.10 | Iaxy | zaptel-be/zonedata.c |
04:03.11 | Iaxy | tar: zaptel-be/zconfig.h: time stamp 2005-09-13 16:34:50 is 7380521 s in the future |
04:03.11 | Iaxy | zaptel-be/zonedata.c |
04:03.12 | Iaxy | hahaha, it didn't compile because blabla timestamp is in the future.... hmmm |
04:03.46 | russellb | well fix your clock |
04:03.51 | russellb | 2005 is obviously not the future |
04:03.54 | Iaxy | now thats funny |
04:04.38 | CunningPike | What does 'date' say |
04:05.05 | Iaxy | jun 20 2005....:-) |
04:08.38 | *** join/#asterisk kmilitzer (n=km@office-gw.westend.com) |
04:09.00 | *** join/#asterisk xtr (n=94752345@S0106000c41ed11e1.vf.shawcable.net) |
04:16.53 | Iaxy | oo those are ugly errors |
04:20.05 | CunningPike | Iaxy: ntp is your friend ;) |
04:20.39 | *** join/#asterisk MacDome (n=eseidel@c-71-198-177-144.hsd1.ca.comcast.net) |
04:21.07 | Iaxy | it has ugly compile errors |
04:22.30 | Iaxy | It says required dists. Redhat enterprise 3 or fedora Core 3. |
04:23.07 | Iaxy | I used redhat enterprise 4 |
04:23.25 | russellb | should still work |
04:24.22 | Iaxy | it shows RHEL 4 in the install script. so yeah it should work, anything other than normal is it is smp, dual cpu's. |
04:25.10 | asterboy | @russellb, whats it take to setup a jbot in a channel? |
04:25.11 | *** join/#asterisk carrar (i=tim@osburn.com) |
04:25.14 | carrar | hi |
04:25.17 | *** join/#asterisk sternn (n=sternn@user-0c938ku.cable.mindspring.com) |
04:25.19 | asterboy | high |
04:25.23 | carrar | Anyone use the unistim driver? |
04:25.38 | asterboy | ~unistim |
04:26.13 | russellb | asterboy: i have no idea |
04:26.28 | asterboy | who setup jbot here so I can poll them? |
04:26.38 | russellb | don't know |
04:26.42 | russellb | it has been here longer than i have |
04:26.50 | carrar | I can't get a unistim phone to call a cisco sip phone |
04:26.52 | asterboy | interesting |
04:27.01 | carrar | cisco can call unistim though |
04:27.12 | asterboy | ~jbot |
04:27.14 | jbot | from memory, jbot is only marginally useful at best, He got a C- on his Turing Test |
04:27.18 | carrar | I've tried forcing ulaw and alaw |
04:27.25 | russellb | jbot: who is your daddy |
04:27.26 | jbot | YOU are, Mr. Sexy Pants |
04:27.29 | asterboy | lol |
04:27.32 | russellb | :) |
04:27.51 | asterboy | jbot: who is your real daddy |
04:27.54 | jbot | asterboy: what are you talking about? |
04:28.03 | russellb | jbot: who is your owner |
04:28.05 | jbot | I think you lost me on that one, russellb |
04:28.09 | russellb | oh well :) |
04:28.16 | asterboy | huh |
04:28.27 | Iaxy | make -C /lib/modules/2.6.9-34.ELsmp/build SUBDIRS=/tmp/zaptel/zaptel-be modules |
04:28.27 | Iaxy | make[1]: Entering directory `/usr/src/kernels/2.6.9-34.EL-smp-i686' |
04:28.28 | Iaxy | <PROTECTED> |
04:28.28 | Iaxy | make -C /lib/modules/2.6.9-34.ELsmp/build SUBDIRS=/tmp/zaptel/zaptel-be modules |
04:28.29 | Iaxy | make[1]: Entering directory `/usr/src/kernels/2.6.9-34.EL-smp-i686' |
04:28.29 | Iaxy | <PROTECTED> |
04:28.30 | Iaxy | . |
04:28.30 | asterboy | I want to get jbot in #mythtv |
04:28.54 | russellb | Iaxy: those aren't errors |
04:28.58 | russellb | and please use pastebin ... |
04:29.15 | distortion | ./k Iaxy spammah! |
04:29.23 | asterboy | lol, not only is it not an error, but its spam |
04:29.27 | Iaxy | yeah sure.... sorry |
04:30.15 | *** join/#asterisk L|NUX (n=linux@202.5.145.58) |
04:30.20 | Iaxy | the errors didn't make the paste |
04:30.26 | asterboy | ah |
04:30.50 | asterboy | that sucks...maybe a 2>&1 at the end of the command |
04:30.59 | russellb | asterboy: mythtv rocks :) |
04:31.19 | asterboy | I'll say, I'm can't wait to get my box up and running. |
04:32.03 | Iaxy | http://pastebin.com/693589 |
04:32.10 | Iaxy | there are the errors... |
04:33.08 | asterboy | Just picked up a P4 1.8GHz, 40Gb, 256Mb box for $140 CDN. |
04:33.25 | asterboy | That will be my Myth box with 3 PVR-250s |
04:34.03 | asterboy | Frontend will be a MediaMVP with mvpmc |
04:34.15 | russellb | eep, you need more drive space than that |
04:34.23 | russellb | but i guess this is the wrong channel for that discussion :) |
04:34.50 | *** join/#asterisk MikeJ__ (n=vircuser@d14-69-8-30.try.wideopenwest.com) |
04:34.53 | asterboy | oh ya...I'm adding a 120Gb Sata but would be nice to have much more. |
04:34.55 | Iaxy | russellb: did you take a look at the compile errors? |
04:35.19 | russellb | ~centosbug |
04:35.20 | jbot | hmm... centosbug is a problem with the latest Centos kernel (4.2 and 4.3). To fix it, edit the file /usr/src/kernels/2.6.9-34.EL-i686/include/linux/spinlock.h and change 'rw_lock_t' on line 407 to 'rwlock_t'. This is part of the 'kernel-devel' package. |
04:35.41 | russellb | sooo ... those errors look just like that bug |
04:36.34 | russellb | yes, that's what it is |
04:36.36 | Iaxy | hmmm... |
04:36.53 | russellb | I would be happy to fix that for you if you would like. |
04:37.53 | distortion | hmm, i think im going to grab the dell 2405 monitor |
04:38.02 | distortion | its so big and sexy |
04:39.27 | *** join/#asterisk gursikh (n=guriskh1@adsl-68-95-82-50.dsl.hstntx.swbell.net) |
04:40.53 | *** part/#asterisk ghost99 (n=neville@222-152-219-77.jetstream.xtra.co.nz) |
04:41.06 | Iaxy | russellb: you da man... |
04:41.20 | blitzrage | hail! |
04:41.32 | *** join/#asterisk yxa (n=diablo@58.185.90.101) |
04:41.33 | russellb | Iaxy: that will be $50. |
04:41.45 | blitzrage | russellb: you're cheap! |
04:41.50 | russellb | i know! |
04:41.57 | Iaxy | $50 for typing ~centosbug |
04:41.58 | blitzrage | I could afford you at that rate :) |
04:42.20 | Iaxy | how about a supper |
04:42.28 | russellb | Iaxy: I'm just kidding ... consider that your Digium support |
04:42.38 | blitzrage | Iaxy: what is your time worth, and how long would you have spent searching for the answer? :) |
04:42.45 | Qwell | russellb: "That'll be $125". |
04:43.16 | JunK-Y | blitzrage: your mother is 10$ :P |
04:43.22 | Iaxy | hehehe.. I know.... |
04:43.23 | russellb | but since I helped you in the middle of the night ... you owe me $50. :D |
04:43.56 | blitzrage | JunK-Y: your mother owes *me* $10 |
04:46.07 | russellb | I'm trying to update my resume, and it's very hard to explain what I do in words |
04:46.33 | blitzrage | russellb: it should just say, "What *don't* I do?" |
04:46.39 | Iaxy | russellb: need a reference? :-) |
04:46.50 | russellb | Iaxy: ha, no thanks :) |
04:46.54 | Qwell | heh, I think russellb has plenty of references |
04:46.59 | blitzrage | aye |
04:47.17 | russellb | Iaxy: you can email my boss and tell him how grateful you are that I helped you :-p |
04:47.22 | blitzrage | ok, I think I need to go read for 30 mins, then finally go to bed :) |
04:47.47 | Iaxy | will do. |
04:48.00 | russellb | blitzrage: g'night! |
04:48.04 | Iaxy | shopuld I tell him you charged me? ....hehe |
04:48.15 | russellb | but I didn't! |
04:48.21 | blitzrage | russellb: night! |
04:48.22 | Iaxy | $50 |
04:48.28 | russellb | i was kidding! |
04:48.31 | blitzrage | lol |
04:48.47 | *** join/#asterisk alephcom (n=alephcom@host75.net14.mcsnet.ca) |
04:48.48 | Iaxy | and so was I. |
04:48.50 | russellb | hehe |
04:49.02 | MrDigital | russellb: what do you do? |
04:49.23 | russellb | honestly, I saw that you were using ABE, so I wanted to make sure you got everything resolved ASAP |
04:49.29 | russellb | otherwise, i probably would have been working on code :) |
04:49.47 | blitzrage | what the heck... my asterisk seems like its ignoring modules.conf noload statements |
04:49.48 | russellb | MrDigital: I'm a full time student, but I also work for Digium |
04:50.18 | Iaxy | I appreciate that. Its very noble of you.... thanks you |
04:50.27 | CunningPike | russellb is who the users of 1.0 owe a tremendous debt of gratitude |
04:50.36 | Qwell | CunningPike: and 1.2 |
04:50.37 | russellb | CunningPike: :D |
04:50.55 | CunningPike | ;) I've not forgotten, russellb |
04:51.49 | CunningPike | :) |
04:51.52 | russellb | it's cool, a lot of people still run the 1.0 series |
04:52.12 | CunningPike | We're not using it any more, but we did right up to 1.2.1 |
04:52.26 | russellb | I don't touch it anymore, either |
04:52.28 | CunningPike | And we are eternally grateful to you |
04:52.37 | russellb | thank you, very much. I appreciate that |
04:52.54 | Qwell | CunningPike: he likes free beer |
04:53.03 | russellb | Qwell: :) |
04:53.09 | CunningPike | Now we're on 1.2, you can esad ;) |
04:53.13 | CunningPike | j/k |
04:53.14 | Qwell | of course...he and blitzrage turned MINE down, but.. :P |
04:54.46 | *** join/#asterisk alephcom (n=alephcom@host75.net14.mcsnet.ca) |
04:54.46 | russellb | CunningPike: when we started 1.2, we had a discussion and decided to make it a policy that bugs are fixed in 1.2 first, before the trunk |
04:55.00 | russellb | that way, it's not just me coming behind and pulling them over by hand |
04:55.19 | russellb | that has worked out really well, and now the release branch is a group maintained thing for the most part |
04:55.27 | CunningPike | That's great - and I think the quality shows through - we're loving 1.2, even more than 1.0 |
04:56.12 | russellb | yeah, and i didn't have to stay up all night to catch up on commits at any point :/ |
04:56.20 | russellb | I had a few of those nights last year ... |
04:56.58 | dlynes | russellb: there's one bug fixed in trunk that's not fixed in release, afaik :) |
04:57.04 | russellb | dlynes: d'oh |
04:57.14 | dlynes | russellb: but not asterisk....zaptel and libpri |
04:57.25 | russellb | well that's not really my fault, heh |
04:57.26 | russellb | but which one |
04:57.28 | dlynes | asterisk 1.2.7.1 for the most part is pretty stable |
04:57.37 | dlynes | I don't know who all it affects |
04:57.45 | dlynes | I would imagine not too many people |
04:57.57 | russellb | you like that super-minor release number? :) |
04:58.21 | russellb | that means ... really refined ... or something |
04:58.42 | dlynes | But for me, when I've got incoming calls on the pri, sometimes, the pri will hang, giving all incoming calls a busy signal, and giving all outgoing calls a busy signal |
04:58.57 | dlynes | I just figured why bother upgrading to 1.2.7, if 1.2.7.1 was otu |
04:58.58 | russellb | weird |
04:59.11 | dlynes | libpri-trunk and zaptel-trunk from apr 24 fixed that problem for me |
04:59.13 | *** join/#asterisk scardinal (n=supreme@port816.ds1-suoe.adsl.cybercity.dk) |
04:59.16 | russellb | and this doesn't occur running trunk? |
04:59.30 | russellb | libpri and zaptel trunk and 1.2 don't have many differences. |
04:59.34 | russellb | i actually thought they were still the same |
04:59.47 | dlynes | i'm too scared to try a newer trunk, for fear it might blow up my system even more than zaptel 1.2.5 did |
04:59.59 | dlynes | well zaptel-1.2.5 with libpri 1.2.2 |
05:00.24 | russellb | ah, well there was a fix that went in very recently, yes |
05:00.30 | russellb | that went into both the trunk and 1.2 branch |
05:00.35 | russellb | but hasn't made it into a release tarball yet |
05:01.07 | dlynes | Well, i don't know if the problem was with zaptel, libpri, or both |
05:01.23 | dlynes | I just know at first the problem would hit me about once a week |
05:01.28 | dlynes | then it was every couple of days |
05:01.33 | dlynes | and then it got to be once a day |
05:01.49 | dlynes | I couldn't deal with it anymore, so someone suggested grabbing trunk to me |
05:05.14 | *** join/#asterisk LasaK (n=mypain@202.158.79.151) |
05:05.23 | LasaK | hi all |
05:11.13 | *** part/#asterisk gursikh (n=guriskh1@adsl-68-95-82-50.dsl.hstntx.swbell.net) |
05:11.54 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
05:14.47 | Iaxy | gotta run, Thanks again russellb . |
05:16.01 | *** join/#asterisk kristalino (n=kristali@LNeuilly-152-21-126-73.w193-253.abo.wanadoo.fr) |
05:25.31 | *** join/#asterisk parag7732 (n=root@de2-b1835.alshamil.net.ae) |
05:25.57 | parag7732 | I am a new bie to Free PBX...Can anybody let me know that Free PBX supports all the features of Digium Asterisk |
05:28.29 | dlynes | parag7732: FreePBX afaik, is Asterisk, with some management tools included |
05:29.06 | dlynes | parag7732: The most significant of which is AMP (which is now called freepbx) |
05:29.44 | *** join/#asterisk Pageus (n=FreePBX2@ip70-190-19-6.ph.ph.cox.net) |
05:30.14 | Pageus | Howdy all |
05:30.47 | Qwell | No, FreePBX *IS* AMP. |
05:30.59 | Qwell | nothing more, nothing less |
05:31.17 | parag7732 | There are two features are given in the digium site Call Recording and Call snooping. What is the differnece ? |
05:31.40 | Qwell | parag7732: recording records it to disk, snooping, or ChanSpy, allows you to listen in on live calls |
05:36.00 | parag7732 | what does " Blind Transfer " means |
05:36.00 | parag7732 | ?? |
05:36.16 | parag7732 | This is again a feature of asterisk |
05:36.18 | Qwell | when you transfer...blindly |
05:36.21 | jql | it means you transfer the call without first talking to the person you're transfering to |
05:36.31 | jql | "attended" transfer would be when you do |
05:36.40 | Qwell | I would have called it a deaf transfer, but that's just me |
05:36.52 | *** join/#asterisk MikeJ[Laptop] (n=vircuser@d14-69-8-30.try.wideopenwest.com) |
05:38.08 | MikeJ[Laptop] | the dumb transfer is transfering tothe guy sitting next to you :P |
05:38.08 | *** join/#asterisk sergeus (n=s@195.112.98.13) |
05:38.17 | dlynes | blind transfer |
05:38.33 | dlynes | that's what it's called in most pbxes and keysystems |
05:38.35 | Qwell | no, you know what would be cool... |
05:38.37 | Qwell | mute transfer |
05:38.46 | Qwell | transfer a call, but force the callee to be muted. :P |
05:38.53 | MikeJ[Laptop] | hehe |
05:39.21 | jql | sometimes, you need a pimp-slap transfer |
05:39.45 | MikeJ[Laptop] | I like when people call wrong number into our conference bridge, you ask the number, they tell you, you tell them, that's this number, but I am not who you were trying to reach... and they CALL BACK |
05:41.20 | *** join/#asterisk MacDome (n=eseidel@c-71-198-177-144.hsd1.ca.comcast.net) |
05:42.09 | parag7732 | what does Call Queuing means??? |
05:42.21 | parag7732 | i mean in asterisk how it works?? |
05:42.39 | Qwell | same as any other PBX |
05:43.41 | *** part/#asterisk parag7732 (n=root@de2-b1835.alshamil.net.ae) |
05:43.56 | *** join/#asterisk parag7732 (n=root@de2-b1835.alshamil.net.ae) |
05:44.14 | Qwell | ~root |
05:44.15 | jbot | i guess root is not a Good Thing to use when using IRC. Please use a different account. You will probably not be able to speak until change your user account. |
05:44.32 | MikeJ[Laptop] | ?? |
05:44.51 | *** join/#asterisk chapeaurouge (n=chap@user-85-201-82-146.tvcablenet.be) |
05:45.07 | LasaK | i had problem to get asterisk works with NAT issue |
05:45.18 | dlynes | MikeJ[Laptop]: parag's ircing as root |
05:45.35 | MikeJ[Laptop] | ahh |
05:45.39 | MikeJ[Laptop] | now I see it |
05:46.13 | MikeJ[Laptop] | wow.. state your question, then immdeiately leave |
05:46.31 | root | hmmm |
05:46.57 | russellb | /kick root |
05:47.00 | russellb | dang, just missed |
05:47.03 | russellb | :-p |
05:47.17 | MikeJ[Laptop] | heh |
05:47.26 | MikeJ[Laptop] | what, no one likes root? |
05:47.35 | MikeJ[Laptop] | I thought everyone like root |
05:47.36 | Qwell | real men use... |
05:48.16 | jql | sudo bash |
05:48.24 | Qwell | sudo su - |
05:48.55 | dlynes | Now that's just sudo cool |
05:49.04 | *** join/#asterisk austinnichols101 (i=austinni@dsl-10-169.cofs.net) |
05:49.11 | luke-jr_ | sudo ./x --login |
05:49.12 | russellb | sudo -s |
05:49.36 | Qwell | russellb: yeah, that works too |
05:49.48 | Qwell | and you keep cwd |
05:49.56 | russellb | :) |
05:50.02 | Qwell | except, no root $PATH, right? |
05:50.04 | luke-jr_ | sudo visudo |
05:50.11 | *** join/#asterisk downunder33 (n=robert@219.95.158.235) |
05:50.15 | luke-jr_ | Qwell: erm, bash --login shouldn't change cwd ... |
05:50.17 | Qwell | hmm, I guess you do get path |
05:50.36 | Qwell | russellb: neat |
05:50.37 | luke-jr_ | tsurukikun postfix # bash --login |
05:50.37 | luke-jr_ | tsurukikun postfix # |
05:51.12 | Qwell | /bin/ls: /bin/ls: cannot execute binary file |
05:51.13 | Qwell | heh |
05:51.15 | Qwell | sudo -s ls |
05:52.33 | downunder33 | hi all. Is there an announce list for asterisk security related bulletins? thx |
05:52.56 | Qwell | downunder33: closest thing, is the svn-commits or asterisk-dev list |
05:53.16 | *** part/#asterisk Alystair (n=bob@CPE001109c15241-CM00407b8794db.cpe.net.cable.rogers.com) |
05:54.12 | parag7732 | I have just installed asterisk@home 2.8 with free pbx......But it dosn't support GUI. So can i install all the rpms of Cent OS 4.3 now. So that i can get the GUI |
05:54.27 | downunder33 | thx qwell. So, there is no standard protocol for announcing security related issues or patches? |
05:54.49 | Qwell | downunder33: pretty much just the -dev list, or they're sent straight (privately) to Digium |
05:55.15 | dlynes | downunder33: http://search.securityfocus.com/swsearch?sbm=%2F&metaname=alldoc&query=asterisk&x=0&y=0 |
05:55.25 | russellb | we'd prefer private communication first ... |
05:57.17 | CunningPike | russellb: Have you ever seen really high size-32 usage with * running? |
05:57.45 | russellb | i believe i'm too tired to think right now |
05:58.08 | Qwell | russellb: sneak off to bed - I won't tell |
05:58.59 | *** join/#asterisk [Airwolf] (n=airwolf@83.98.235.219) |
05:59.25 | Pageus | Hey guys, I'm having some echo trouble. It seems to echo a bit when a call comes through on my T1.. but my other connections are fine.. |
05:59.25 | dlynes | Has anyone been able to get spandsp 0.0.3 to work with asterisk 1.2.x? |
05:59.42 | Pageus | i have echo canceling turned on.. and the echotraining set to 400.. |
06:05.18 | CunningPike | Pageus - which EC are you using? |
06:05.32 | Pageus | digium t1000p |
06:05.49 | Pageus | software i think.. since it's not built into the card |
06:06.01 | Pageus | couldn't get the additional funds for the better card |
06:07.05 | *** join/#asterisk chapeaurouge (n=chap@user-85-201-82-146.tvcablenet.be) |
06:07.29 | Pageus | t100p |
06:07.30 | Pageus | sorry |
06:07.34 | Pageus | long weekend. |
06:10.16 | CunningPike | Pageus: Which software EC - look in zconfig.h |
06:12.09 | Pageus | sec |
06:14.21 | dlynes | CunningPike: cdr pivottable reports, as in Excel? |
06:14.26 | CunningPike | Yes |
06:14.31 | CunningPike | My head hurts :) |
06:14.36 | dlynes | ah |
06:14.56 | CunningPike | We import our CDR into MSSQL and join it with our employee database |
06:15.13 | CunningPike | Works out OK, actually |
06:15.17 | dlynes | ah...and then spit out the data from mssql into excel for a report? |
06:15.49 | CunningPike | Yes - I create a PivotTable with the MSSQL view as an external data source |
06:16.24 | CunningPike | Lets people slice and dice themselves without me having to create upmteen reports |
06:16.50 | CunningPike | I got your email, thanks - reply will be forthcoming shortly :) |
06:16.58 | dlynes | nod |
06:17.36 | CunningPike | You still there, Pag |
06:17.39 | CunningPike | Pageus: |
06:17.58 | *** join/#asterisk MacDome (n=eseidel@c-71-198-177-144.hsd1.ca.comcast.net) |
06:18.40 | Pageus | yeah was looking through the file |
06:18.53 | Pageus | it looks like all the echo canceling is commented out |
06:18.55 | Pageus | how odd |
06:19.06 | *** join/#asterisk lorinc (n=ang@caracas-2758.adsl.interware.hu) |
06:19.23 | Pageus | oh wait |
06:19.34 | Pageus | #define ECHO_CAN_KB1 |
06:19.37 | Pageus | was burried |
06:19.38 | Pageus | lol |
06:20.27 | CunningPike | Pageus: OK, try MG2 - we had better results with it |
06:20.37 | Pageus | ok |
06:20.41 | CunningPike | If that doesn't work, consider an external hardware EC |
06:20.46 | Pageus | so uncomment that one comment the other one.. |
06:20.59 | CunningPike | Yes - make clean; make; make install |
06:21.06 | Pageus | yeah when i use my usr USB skype phone it seems to work fine |
06:21.15 | *** join/#asterisk rdgzt (n=joakim@201.137.86.15) |
06:21.22 | CunningPike | Is that on your PRI? |
06:21.24 | Pageus | but that isn't an option for the rest of my office |
06:21.26 | Pageus | no |
06:21.28 | Pageus | no pri |
06:21.28 | CunningPike | OK |
06:21.31 | Pageus | T1 voice.. |
06:21.37 | Pageus | E&m Wink |
06:21.40 | CunningPike | Oh |
06:21.45 | dlynes | So did you ever get it to work properly, Pageus? |
06:21.55 | rdgzt | Loading the wctdm and wcfxo modules don't really seem to find my TDM400P card. |
06:22.02 | rdgzt | Even though it's showing up in lspci. |
06:22.12 | dlynes | rdgzt: wctdm would be the one you need, not wcfxo |
06:22.34 | dlynes | rdgzt: make sure you've got your card defined properly in zaptel.conf, too |
06:22.59 | dlynes | rdgzt: how many fxo ports, and how many fxs ports do you have? |
06:23.20 | Pageus | yeah.. the system came online this morning with no issues.. as long as i was running hard phones i had no echo issues |
06:23.23 | rdgzt | dlynes: I have 4 FXOs. |
06:23.30 | _Vile | HI |
06:23.44 | dlynes | rdgzt: make sure you have a line like: fxsks=4 in your zaptel.conf file then |
06:24.00 | dlynes | rdgzt: fxo ports use fxs signalling, not fxo signalling |
06:24.04 | Pageus | cept they sent my fax did to my system and i didn't have it setup for it.. so it caused my * to loop uncontrolled which kept anything else from coming in |
06:24.17 | CunningPike | Good night all |
06:24.19 | Pageus | ok now for the stupid question.. |
06:24.19 | dlynes | cool |
06:24.22 | dlynes | Night, cp |
06:24.23 | _Vile | going to get some food finally ttyl |
06:24.32 | Pageus | how do i recompile that part of the system |
06:24.33 | Pageus | lol |
06:24.35 | CunningPike | Night, dlynes |
06:24.57 | dlynes | Pageus: recompile? what for? |
06:25.04 | Pageus | ok so i don't have to? |
06:25.08 | dlynes | Pageus: That sounds like an extensions.conf misconfiguration to me |
06:25.41 | dlynes | Pageus: Check your log to see if it's dropping an incoming call into a non-existent extension |
06:25.45 | Pageus | no.. the system wasn't setup for fax in the extensions.. but it was setup in the main system.. it was never supposed to take faxes.. we have an alternative efax service for it |
06:25.49 | Pageus | yeah |
06:26.16 | rdgzt | dlynes: Now I get ZT_CHANCONFIG failed on channel 1: No such device or address (6) |
06:26.24 | rdgzt | When I modprobe wctdm. |
06:26.27 | luke-jr_ | Pageus: so rm the fax module? |
06:26.41 | dlynes | So, if you have a fax come in on that number, just do a Dial(Zap/g1/xxxxxxxx) where xxxxxxxx is the number for your efax server |
06:26.43 | Pageus | never installed it |
06:26.44 | Pageus | lol |
06:27.15 | dlynes | Pageus: Asterisk doesn't have any problems bridging two zaptel channels for delivering faxes |
06:27.27 | Pageus | considering the PC handles it all.. i wasn;t worried.. took them all of 2 mins to change it |
06:27.29 | rdgzt | dlynes: That means it doesn't find the card, I assume. |
06:27.33 | dlynes | It's not the best way to deliver faxes, tying up two lines |
06:27.40 | dlynes | but, at least your faxes get through |
06:27.48 | *** join/#asterisk [Airwolf] (n=airwolf@83.98.235.219) |
06:27.55 | luke-jr_ | are PRI T1s the same price range as data T1s? |
06:28.01 | Pageus | ok so when i made the change in the header file it should automaticly take |
06:28.58 | dlynes | rdgzt: Yeah...I'm getting weird ass sh*t like that, too, with an x100p card |
06:29.23 | dlynes | I'm going to be trying it in a different machine |
06:29.25 | rdgzt | dlynes: It's worth noting that this used to work with older drivers. Maybe I should downgrade? |
06:29.39 | dlynes | rdgzt: which version are you using? |
06:29.46 | rdgzt | This is 1.2.5. |
06:30.26 | *** join/#asterisk chapeaurouge (n=chap@user-85-201-82-146.tvcablenet.be) |
06:30.33 | *** join/#asterisk TonyM (n=TonyM@adsl-solo-80-168-224-92.claranet.co.uk) |
06:30.57 | dlynes | yeah...that's the one i'm having problems with, too |
06:31.00 | rdgzt | dlynes: I tried with the drivers that come packaged with Ubuntu, which are 1.2.1, a few days ago, and those work. |
06:31.04 | rdgzt | Well, worked. |
06:31.10 | dlynes | same crap happens with zaptel-trunk |
06:31.18 | rdgzt | I'm considering going to the svn version, seeing if that works better. |
06:31.24 | dlynes | nope :) |
06:31.29 | dlynes | like i said...i was trying that :) |
06:31.31 | dlynes | same problem |
06:31.37 | rdgzt | Heh, ok, I'm going to try downgrading, then. |
06:31.57 | rdgzt | I'll go one version at a time back until it works, let's see... |
06:33.51 | *** join/#asterisk MacDome (n=eseidel@c-71-198-177-144.hsd1.ca.comcast.net) |
06:34.16 | rdgzt | Ok, 1.2.4 still fails the same way, it seems. |
06:34.24 | dlynes | Maybe 1.2.1? :) |
06:34.37 | *** join/#asterisk MGSsancho (n=user@adsl-67-125-156-130.dsl.irvnca.pacbell.net) |
06:35.05 | rdgzt | As I said, I'll go one version at a time backwards. |
06:35.34 | rdgzt | And we'll see if I don't end up with 1.2.1 anyway. :) |
06:35.39 | *** join/#asterisk Zeeek (n=icechat5@pdpc/supporter/active/Zeeek) |
06:35.49 | dlynes | Lemme know how you make out :) |
06:35.57 | dlynes | This problem is pissing me off, too |
06:36.10 | dlynes | I was just going to reformat and reinstall on a different machine to see if that fixed it |
06:36.23 | dlynes | The driver was working in that machine for about 2 days |
06:36.24 | rdgzt | Yeah, it's pretty quick testing this, I'll keep you updated. :) |
06:36.29 | dlynes | and then i shut it off for about 12 hours |
06:36.31 | dlynes | turned it back on |
06:36.36 | dlynes | and all hell broke loose |
06:37.01 | rdgzt | Whoa. |
06:37.06 | rdgzt | That sounds like fun. |
06:37.33 | dlynes | ah...i take it you have the driver versions all downloaded and compiled already? |
06:37.48 | rdgzt | I'm downloading and compiling as I test. |
06:37.52 | dlynes | ah |
06:37.53 | rdgzt | 1.2.3 was no good either. |
06:37.53 | dlynes | hehe |
06:38.15 | *** join/#asterisk Givur (i=anwi73@p54BC8EFB.dip0.t-ipconnect.de) |
06:38.34 | rdgzt | Do specific versions of Asterisk require specific versions of Zaptel? |
06:38.42 | Zeeek | ya |
06:38.52 | dlynes | rdgzt: sometimes |
06:38.56 | dlynes | rdgzt: but not always |
06:38.57 | rdgzt | Because I'd like to run the most recent asterisk version, to solve a problem I had, will I be able to do that even if I run, say, zaptel 1.2.1? |
06:39.12 | dlynes | rdgzt: try it and see |
06:39.15 | rdgzt | I guess we'll see. ;0 |
06:39.18 | rdgzt | Um, :) |
06:39.25 | dlynes | rdgzt: that's the only way you'll know |
06:39.34 | Zeeek | rdgzt the only caveat is that if you have problems, it's one more possible hidden cause |
06:39.45 | dlynes | rdgzt: certain versions of asterisk will only work with certain zaptel versions |
06:39.52 | dlynes | but not every asterisk version is like that |
06:40.03 | Zeeek | but certain others will work with certain others ;) |
06:40.42 | rdgzt | Zeeek: Well, given that newer versions of zaptel don't seem to detect my hardware at all, I don't have many options. |
06:40.50 | Zeeek | what hw ? |
06:41.10 | rdgzt | TDM400P |
06:41.14 | rdgzt | With 4 FXOs. |
06:41.29 | Zeeek | that's extremely odd! |
06:41.38 | Zeeek | what does Digium say? |
06:41.39 | rdgzt | I agree. |
06:41.42 | rdgzt | I have no idea. |
06:41.55 | Zeeek | you didn't contact them? |
06:42.03 | dlynes | Zeeek: I've got the same problem with an x100p card |
06:42.15 | dlynes | rdgzt: actually |
06:42.17 | Zeeek | there must be thousands of people that have it then |
06:42.22 | rdgzt | Zeeek: No, I don't really have the time to go through customer support stuff right now. |
06:42.29 | dlynes | rdgzt: do you get the same problem when you try to load the ztdummy driver? |
06:42.31 | rdgzt | I just need this to work. |
06:42.36 | rdgzt | dlynes: Good question, I haven't tried. |
06:42.40 | Zeeek | how can anyone fix the problem if they don't get input? |
06:42.46 | dlynes | If you do, you've got the same damned problem I do |
06:43.02 | dlynes | Zeeek: I havne't reported the problem yet, because I don't know what the problem is yet |
06:43.29 | rdgzt | Ok, confirmed. |
06:43.35 | rdgzt | 1.2.1 works, later versions don't. |
06:43.44 | dlynes | And besides...I'm running a clone card which digium probably isn't too keen on making run with asterisk |
06:43.54 | Zeeek | dlynes you should call Dig or email with the symptoms. THey may have a workaround |
06:44.00 | rdgzt | Or, wait, maybe not, let me see... |
06:44.09 | Zeeek | oh, well then you can't call or write them |
06:44.28 | dlynes | Zeeek: well, i've got another problem, too |
06:44.29 | Zeeek | I'm running 1.2 so I can't help you |
06:44.38 | Zeeek | I have two X100P and a TDM400P |
06:44.39 | dlynes | Zeeek: and it's with a genuine digium x100p card |
06:44.43 | rdgzt | Nope, 1.2.1 doesn't work either. |
06:44.50 | dlynes | piece of crap doesn't get the caller id some fo the time |
06:45.03 | rdgzt | This is very weird. |
06:45.24 | Zeeek | callerid can vary accoring to a lot of stuff both inside and outside the hw |
06:45.34 | dlynes | Zeeek: how so? |
06:45.38 | Zeeek | but I assume this problem JUST started |
06:45.48 | russellb | rdgzt: would you like me to take a look? |
06:45.51 | dlynes | nah...caller id problem was there all along |
06:46.10 | dlynes | the problem with the driver not loading is something that started recently |
06:46.13 | Zeeek | well then it can depend on your telco, the dialplan and you connections |
06:46.19 | rdgzt | russellb: Possibly, who are you? :) |
06:46.22 | russellb | rdgzt: i work for digium, by the way, not in support, though. |
06:46.31 | Zeeek | russellb heh he doesn't have time for support ;) |
06:46.39 | dlynes | lol |
06:46.42 | Zeeek | bugmonger |
06:46.44 | russellb | rdgzt: i'm an asterisk developer ... |
06:46.45 | rdgzt | russellb: If you could, that'd be great. |
06:46.47 | russellb | that should be sleeping |
06:46.53 | russellb | but i'm not, so i might as well be useful |
06:47.05 | russellb | rdgzt: sure, just msg me the login info |
06:47.05 | dlynes | Yeah...maybe if he's able to solve rdgzt's problem |
06:47.10 | dlynes | It's probably the same as my problem |
06:47.12 | rdgzt | I just need to figure out how to make you able to log in through the fw. |
06:47.18 | rdgzt | Hold on a sec. |
06:47.21 | russellb | k |
06:47.48 | russellb | i'm really not all that experienced with zaptel :) |
06:48.00 | Zeeek | hardware sucks anyway |
06:48.01 | russellb | but ... i should be able to figure out the common stuff, heh |
06:49.26 | *** join/#asterisk chapeaurouge (n=chap@user-85-201-82-146.tvcablenet.be) |
06:50.16 | Zeeek | how's it going in general russellb? Life treating you ok? |
06:50.20 | dlynes | Heh...looks like JerJer's awake now |
06:50.33 | Zeeek | I won't ask him than question ;) |
06:50.39 | russellb | Zeeek: just fine, thanks for asking. :) I'm in the middle of final exams, actually |
06:50.42 | *** join/#asterisk Johnnie (n=jdlewis@pdpc/supporter/active/Johnnie) |
06:50.59 | *** join/#asterisk joelsolanki (n=joelsola@202.160.163.144) |
06:51.28 | dlynes | Evening, solanki |
06:51.48 | joelsolanki | Good evening daniel |
06:51.49 | joelsolanki | :) |
06:53.43 | *** part/#asterisk joelsolanki (n=joelsola@202.160.163.144) |
06:53.48 | dlynes | fine! |
06:53.49 | *** join/#asterisk joelsolanki (n=joelsola@202.160.163.144) |
06:53.52 | dlynes | be that way, joel |
06:54.12 | joelsolanki | hehe. i m on asteris-unregistered :( |
06:54.26 | dlynes | and so am i |
06:54.35 | dlynes | I didn't realize #asterisk existed? |
06:54.41 | joelsolanki | but y ..i m already registed on #asterisk |
06:55.32 | joelsolanki | No probs. atleast i m able to talk |
06:55.43 | *** join/#asterisk bzbw (n=wlwzhang@68-190-223-129.dhcp.mtpk.ca.charter.com) |
06:55.53 | dlynes | i dunno |
06:56.06 | dlynes | i join #asterisk and it always throws me into #asterisk-unregistered |
06:56.32 | Zeeek | ur in asterisk now |
06:56.37 | dlynes | it seems #asterisk and #asterisk-unregistered are the same thing |
06:56.47 | Zeeek | funny |
06:57.10 | joelsolanki | oh ok. |
06:57.12 | dlynes | X-Chat [2.0.9]: dlynes @ herbert.freenode.net / #asterisk (+tncrf #asterisk-unregistered) |
06:57.43 | joelsolanki | is this due to xchat ? |
06:57.49 | Zeeek | reminds of a chat I once ran. I announced that every member had their own chat room. When they started inviting each other in to their private rooms, they figurered out it was all the same room, only the name displayed was fifferent :) |
06:58.20 | Zeeek | Are you recognized before you join? If not that's why |
06:58.32 | dlynes | Yeah, i'm recognized before i join |
06:58.50 | dlynes | My nick is registered, and i autoregister on login |
06:58.52 | Zeeek | oh well |
06:58.57 | dlynes | after i log in, i join asterisk |
06:58.59 | *** join/#asterisk iceyp (n=icepick@firewall.unix.co.nz) |
06:59.13 | joelsolanki | assword accepted - you are now recognized |
06:59.14 | joelsolanki | * services. sets mode +e joelsolanki |
06:59.14 | joelsolanki | -MemoServ- You have no new memos |
06:59.20 | joelsolanki | me too same. |
06:59.31 | dlynes | i dunno |
06:59.39 | dlynes | if i was the server, i wouldn't recognize you |
06:59.43 | joelsolanki | i guess me and dlynes both are using xchat in linux |
06:59.52 | joelsolanki | heheh :) |
06:59.54 | dlynes | i'm sure most people are, joel |
07:00.34 | iceyp | hey guys, i'm trying to connect 2 asterisk systems together, all appears fine in show iax2 peers, but when trying to call between them i get an error like: chan_iax2.c:6786 socket_read: Rejected connect attempt from 60.234.x.x, who was trying to reach '99705580@' |
07:00.40 | dlynes | Well, Zeeek's using icechat |
07:01.11 | dlynes | iceyp: the context that you're trying to go into doesn't exist |
07:01.27 | dlynes | iceyp: I suspect it's an error in your dial command |
07:01.28 | iceyp | but it does ;/, do i need to add an exten => s |
07:01.28 | iceyp | ? |
07:01.29 | Zeeek | OMG everyone will know I use Windows now!!! |
07:01.44 | dlynes | icechat's a windows client? |
07:01.46 | dlynes | never knew |
07:01.49 | dlynes | didn't care, either :) |
07:02.06 | dlynes | only two windows clients i know are mirc and pirch |
07:02.08 | iceyp | [sip_incoming] |
07:02.08 | iceyp | exten => 99705580,1,Dial(IAX2/dhodd@voip.unix.co.nz/0508888802,60,r) |
07:02.18 | Zeeek | Well it might run on Mac in Windoze mode |
07:02.34 | dlynes | you mean in windows on a mac? |
07:02.39 | Zeeek | yeah |
07:02.42 | dlynes | don't need to emulate windows anymore :) |
07:03.13 | iceyp | dlynes can both sides be peer? |
07:03.22 | *** join/#asterisk nagl (n=nagl@rih.zid-nw.wu-wien.ac.at) |
07:03.23 | Strom_C | iceyp: no |
07:03.29 | Strom_C | iceyp: peer is for ourbound only |
07:03.31 | Strom_C | er |
07:03.34 | Strom_C | outbound |
07:03.42 | Strom_C | user == inbound |
07:03.47 | Strom_C | friend == both |
07:04.04 | iceyp | ahhh ok |
07:04.09 | iceyp | thats probably my issue |
07:04.55 | Strom_C | change them both to friend and see what happens |
07:05.00 | iceyp | May 2 19:01:13 NOTICE[63559]: chan_iax2.c:6786 socket_read: Rejected connect attempt from 60.234.68.100, who was trying to reach '99705580@' |
07:05.03 | iceyp | same thing |
07:05.47 | iceyp | <PROTECTED> |
07:05.52 | iceyp | from iax debug |
07:05.57 | *** join/#asterisk chapeaurouge (n=chap@user-85-201-82-146.tvcablenet.be) |
07:07.54 | joelsolanki | I need some information regarding the pstn card. my requirement is i need 4 outgoing lines and 16 to 20 internal extensions ..so can any body recommend any card or solution ? |
07:08.21 | Strom_C | joelsolanki: if you want analog sets internally, use a tdm2400p |
07:08.37 | iceyp | any ideas? |
07:09.00 | Strom_C | iceyp: show me the relevant sections of iax.conf and extensions.conf |
07:09.13 | Strom_C | use pastebin |
07:09.16 | Strom_C | ~pb |
07:09.17 | jbot | extra, extra, read all about it, pb is a place to paste your stuff without flooding the channel - try http://pastebin.com/, or http://pastebin.ca |
07:09.18 | iceyp | yep |
07:10.33 | joelsolanki | Strom_C: yes means i have 4 pstn lines from my telephone company and i want to setup 16 to 20 analog extensions. so is the tdm2400p accurate for it ? |
07:10.52 | Strom_C | yes |
07:10.57 | Strom_C | it's perfect |
07:11.12 | iceyp | http://pastebin.ca/index.php |
07:11.12 | Strom_C | you can do a quad-FXO module and five quad-FXS modules |
07:11.24 | Strom_C | iceyp: um, thats not the link to your code ;) |
07:11.32 | iceyp | the http://pastebin.ca/52797 |
07:11.34 | iceyp | sorry |
07:12.24 | *** join/#asterisk CpuID2 (n=nathan@gentoo/contributor/cpuid) |
07:12.32 | Strom_C | iceyp: and what's the dial line you're using on the second system? |
07:12.50 | CpuID2 | anyone aware of if theres anything in zaptel-trunk that would make it incompatible with asterisk-1.2.x in any way? |
07:13.12 | CpuID2 | its working fine atm, just wanted some feedback from others that have tried the same zap/ast combination |
07:13.36 | iceyp | Strom_C that part appears to be working ... exten => 99705580,1,Dial(IAX2/dhodd@littledan/${EXTEN},60,r) |
07:13.37 | Strom_C | iceyp: so you're dialing from the first system to the second system? |
07:13.46 | iceyp | yes |
07:13.46 | joelsolanki | Ok let me go through this tdm2400p card |
07:13.50 | carrar | joelsoanki |
07:13.55 | carrar | TDM2451B PCI Card |
07:14.03 | carrar | 20 fsx, 4 fxo |
07:14.21 | Strom_C | iceyp: the string in brackets is what defines the username |
07:14.43 | Strom_C | so for calls to the second box, you need to call littledan@(second_box_url) |
07:14.50 | dlynes | CpuID2: nah...i've been running the two together for about 2 months now |
07:14.55 | CpuID2 | cool, np |
07:14.58 | CpuID2 | no weird crashes or anything? |
07:15.02 | CpuID2 | pretty stable? |
07:15.05 | joelsolanki | carrar: oh is that digium card ? |
07:15.10 | dlynes | CpuID2: on the contrary...for me, it was more stable than zaptel 1.2.5 |
07:15.14 | CpuID2 | ive only been using it maybe 3 days or so, with 2 tdm400p's |
07:15.16 | iceyp | Strom_C littledan@HOSTNAME ? |
07:15.19 | CpuID2 | nice :) |
07:15.22 | CpuID2 | cool, good to hear that |
07:15.36 | dlynes | CpuID2: but that's just me....zaptel 1.2.5 was extremely unstable for me |
07:15.40 | CpuID2 | i think im gonna switch another installation to it actually |
07:15.40 | Strom_C | iceyp: yes, because your second box's IAX2 config is called [littledan] |
07:15.50 | CpuID2 | zaptel-1.2.5 was usable here, but i think ive found trunk better |
07:16.08 | dlynes | CpuID2: yeah, for me, zaptel-1.2.5 was putting my pri card in an unusable state |
07:16.14 | CpuID2 | from what i heard, a lot better echo cancellation in trunk atm (with the default cancellation) |
07:16.15 | iceyp | umm so ... exten => 99705580,1,Dial(IAX2/littledan@60.234.x.x/${EXTEN},60,r) |
07:16.16 | CpuID2 | ick |
07:16.17 | dlynes | CpuID2: the only way i was able to fix it was to reboot asterisk |
07:16.25 | CpuID2 | thats a PITA |
07:16.34 | CpuID2 | hmm i really should test out my iaxY again |
07:16.40 | CpuID2 | see if i did actually kill it last time lol |
07:16.47 | dlynes | CpuID2: the default cancellation for trunk is still available in zaptel 1.2.5 as well...it's just not the default in 1.2.5 |
07:16.47 | Strom_C | iceyp: well, you have an entry in the firsy box's iax.conf already |
07:16.54 | iceyp | but its actually looking at iax config to get the host currently cuz it is getting to the system |
07:17.01 | Strom_C | so it would be littledan@voip.unix.co.nz |
07:17.08 | CpuID2 | dlynes, ah np, i wasnt sure if it was a new canceller or not |
07:17.09 | Strom_C | or even better |
07:17.13 | Strom_C | just change user= to littledan |
07:17.23 | Strom_C | and then remove the part before the @ in extensions.conf |
07:17.42 | Strom_C | so you just dial IAX2/voip.unix.co.nz/whatever |
07:17.50 | iceyp | Strom_C exten => 99705580,1,Dial(IAX2/littledan/${EXTEN},60,r) |
07:17.50 | Strom_C | and have the username in the iax.conf |
07:17.51 | dlynes | CpuID2: nah...don't know if it was in zaptel 1.2.4 or not...never checked |
07:17.55 | dlynes | but it's definitely in 1.2.5 |
07:17.59 | Strom_C | iceyp: you're not listening to me |
07:18.02 | CpuID2 | np |
07:18.24 | iceyp | it's littledan not accepting the call not voip.unix.co.nz |
07:18.34 | iceyp | voip.unix.co.nz is passing the cll to littledan |
07:18.43 | Strom_C | iceyp: yes I know |
07:19.28 | Strom_C | when you have a string such as Dial(IAX2/dogballs/555), what happens is that your PBX looks up a section called "dogballs" in your iax.conf and dials 555 into whatever box that section specifies |
07:21.52 | iceyp | Strom_C thanks, all working :) |
07:22.09 | *** join/#asterisk shiznatix (n=shiznati@213-35-237-38-dsl.end.estpak.ee) |
07:22.23 | dlynes | What happens when it encounters monkeyballs? |
07:22.27 | Strom_C | and with that I think I'm going for a bike ride |
07:22.32 | *** join/#asterisk chapeaurouge (n=chap@user-85-201-82-146.tvcablenet.be) |
07:22.42 | Strom_C | dlynes: well then it plays "The monkeys! THE MONKEYS!" |
07:22.52 | dlynes | heh |
07:29.30 | dlynes | ~seen coppice |
07:29.35 | jbot | coppice <n=chatzill@153.192.17.210.dyn.pacific.net.hk> was last seen on IRC in channel #asterisk, 16h 35m 59s ago, saying: 'the real downside is you can't keep doing it. the occassional dead meter is one thing, but failing between every reading it quite another'. |
07:30.26 | rdgzt | Is there a flag somewhere in the Makefile to optimize asterisk for AMD K8? |
07:30.38 | dlynes | doubtful |
07:31.07 | dlynes | but you came back rather late |
07:31.26 | dlynes | I think russellb got tired of waiting for you :) |
07:31.32 | rdgzt | No, he helped me out. |
07:31.35 | dlynes | ah |
07:31.40 | dlynes | so did you get the driver loading, then? |
07:31.40 | rdgzt | My problem was pretty simple, I just had to power-cycle the box. |
07:31.45 | dlynes | oh |
07:31.52 | dlynes | yeah...my problem is bigger than that |
07:31.58 | dlynes | power cycle didn't help |
07:32.00 | rdgzt | The card didn't like having the drivers changed from underneath it several times without power cycling, it seemed. |
07:32.04 | dlynes | i've power cycled like 5 or 6 times |
07:32.15 | rdgzt | Ok, probably not the same thing, then. |
07:32.38 | rdgzt | I'm wondering if I can use the PPro optimization flag on K8, maybe. |
07:40.02 | joelsolanki | does echo cancellation required while buying tdm2400 ? |
07:41.47 | joelsolanki | any idea |
07:41.54 | *** join/#asterisk chapeaurouge (n=chap@user-85-201-82-146.tvcablenet.be) |
07:42.20 | dlynes | you can get hardware cancellation for it? |
07:42.39 | joelsolanki | uhh whats that ? |
07:43.35 | dlynes | Well, you asked if echo cancellation was required when buying a tdm2400 |
07:43.49 | dlynes | You didn't ask about after you buy it...you asked when you buy it |
07:46.06 | joelsolanki | ok :( |
07:47.48 | dlynes | I've never heard of a hardware canceller for the tdm2400, but htat doesn't mean there isn't one |
07:47.52 | dlynes | it's a pretty new card |
07:48.20 | dlynes | but if there is a hardware canceller for it, that'd be a better bet than using hte software one built into zaptel |
07:50.16 | Strom_C | dlynes: there is a cancellation module for the 2400 |
07:50.28 | dlynes | joelsolanki: there ya go |
07:50.35 | Strom_C | joelsolanki: how far away are you from the CO? |
07:53.59 | joelsolanki | CO ? |
07:54.03 | dlynes | Central Office |
07:54.03 | joelsolanki | what is CO ? |
07:54.06 | dlynes | switching station |
07:54.28 | joelsolanki | I dont understand u. |
07:54.33 | Strom_C | joelsolanki: the CO is the telephone company's switch |
07:54.52 | Strom_C | joelsolanki: it's where your dial tone comes from |
07:54.55 | dlynes | Where all the telco company's switches are centralized for your phone number prefeix |
07:54.58 | joelsolanki | we have been provided 4 analog pstn lines. |
07:55.09 | Strom_C | joelsolanki: how long are those lines? |
07:55.36 | joelsolanki | i can extend it anywhere. |
07:55.41 | Strom_C | no no no |
07:55.59 | joelsolanki | long means ? |
07:56.00 | Strom_C | how many feet of copper wire are between your premises and the telephone company's switching equipment? |
07:56.28 | dlynes | joelsolanki: long means what length? |
07:56.35 | joelsolanki | oh it is in my building only. |
07:56.40 | joelsolanki | means its very near. |
07:56.46 | dlynes | Strom_C: English isn't his native language; he's in India |
07:56.50 | Strom_C | joelsolanki: the telephone company's switch is downstairs? |
07:57.11 | joelsolanki | yes it in downstairs |
07:57.31 | Strom_C | you live in the telephome company building? :O |
07:57.47 | joelsolanki | no |
07:58.01 | Strom_C | joelsolanki: where is the telephone company building? |
07:58.26 | joelsolanki | its far way appx 9 kms |
07:58.36 | joelsolanki | y are u asking this questions ? |
07:58.43 | *** join/#asterisk chapeaurouge (n=chap@user-85-201-82-146.tvcablenet.be) |
07:58.49 | x86 | 9km?! |
07:58.55 | Strom_C | joelsolanki: if you have a long telephone line, you will have a greater need for hardware echo cancellation |
07:58.56 | joelsolanki | yes |
07:59.11 | x86 | 9km is very far ;) |
07:59.11 | Strom_C | joelsolanki: is that the building where the telephone company's equipment is located? |
07:59.15 | joelsolanki | oh ok got it. |
07:59.19 | x86 | i bet your LBO is crazy ;) |
07:59.23 | joelsolanki | yes |
07:59.33 | Strom_C | joelsolanki: so the switch is not downstairs |
07:59.36 | joelsolanki | i got your point Strom_C |
07:59.38 | Strom_C | joelsolanki: the switch is 9km away |
07:59.51 | joelsolanki | they have keep some switch in the downstair of our building. |
08:00.07 | joelsolanki | from there we have got our 4 pstn lines |
08:00.11 | Strom_C | what kind of switch? |
08:00.37 | joelsolanki | oh not familiar with it. it has lot of wires and thing in it :) |
08:00.47 | Strom_C | joelsolanki: is it just wire? |
08:00.53 | Strom_C | is it just a room with lots of wire in it? |
08:01.05 | joelsolanki | no wires connect to some switch. |
08:01.27 | dlynes | by switch, do you mean a white board with a lot of terminals sticking out of it? |
08:01.34 | *** join/#asterisk MstlyHrmls (n=mh@melbourne.mostly-harmless.ca) |
08:01.51 | joelsolanki | hang on. let me ask someone and give u clear information. |
08:01.53 | rdgzt | Any ideas why, when registering a softphone via SIP, I get May 2 03:01:48 NOTICE[14256]: chan_sip.c:10886 handle_request_register: Registration from 'john <sip:john@10.0.254.35>' failed for '10.0.2.2' - Username/auth name mismatch |
08:02.08 | dlynes | because your username and password don't match |
08:02.26 | Strom_C | rdgzt: it helps if you read the error message |
08:03.00 | rdgzt | I'm reading it, but I've seen before that if the password is actually wrong, I'll get a more specific message. |
08:03.19 | x86 | uh |
08:03.19 | x86 | no |
08:03.20 | x86 | ;) |
08:03.40 | dlynes | nah...that's the message you'll get |
08:03.47 | dlynes | it's pretty clear |
08:04.05 | joelsolanki | In india we called it DP. junction box. |
08:04.16 | joelsolanki | where all pysical cables gets connected. |
08:04.17 | rdgzt | Ok, so what if I know that the password is correct? |
08:04.20 | dlynes | Yeah...that's not a switch, joel |
08:04.28 | joelsolanki | oh |
08:04.29 | Strom_C | joelsolanki: junction box is just a splice point |
08:04.32 | dlynes | That's just called demarc |
08:04.38 | joelsolanki | ok |
08:04.40 | Strom_C | a switch is an actual piece of electronic equipment |
08:04.46 | *** join/#asterisk darkskiez (n=darkskie@194.247.78.146) |
08:04.52 | Strom_C | a fairly large one at that |
08:05.02 | joelsolanki | hmm ok. got your point |
08:05.13 | dlynes | Usually a specialized computer specifically for handling tens of thousands of conversations at a time |
08:05.16 | joelsolanki | so i assume i need echo cancellation. |
08:05.18 | rdgzt | Because I've checked and double-checked the password on both ends. |
08:05.24 | rdgzt | So I know I'm entering it correctly. |
08:05.34 | rdgzt | Any other reasons I might be getting that error message? |
08:05.54 | dlynes | Your username? |
08:06.21 | dlynes | You're trying to authenticate using an auth? |
08:06.57 | dlynes | rdgzt: What kind of sip device is this that you're trying to connect? |
08:07.05 | rdgzt | I don't know what "trying to authenticate using an auth" means. |
08:07.07 | rdgzt | It's a softphone. |
08:07.12 | rdgzt | Well, I've tried two, both do the same thing. |
08:07.26 | joelsolanki | ok going from lunch. will be back in 20 mins. |
08:07.50 | dlynes | Some softphones and/or sip phones ask you for a username, password, authorization name |
08:08.02 | dlynes | You don't need to specify the authorization name for asterisk |
08:08.06 | dlynes | It just confuses things |
08:08.10 | rdgzt | Ok... |
08:08.33 | x86 | i always specify both with asterisk |
08:08.40 | x86 | and asterisk does handle both of them |
08:08.40 | rdgzt | Ok, I removed that, I had it set, doesn't seem to make a difference, though. |
08:08.48 | dlynes | Does your sip context look like [john]? |
08:09.02 | rdgzt | Yeah. |
08:09.13 | dlynes | rdgzt: do you have a username= set? |
08:09.19 | rdgzt | It's a very basic one, I'm following the Asterisk: Future of Telephony book. |
08:09.24 | rdgzt | I did, but removing it makes no difference either. |
08:09.47 | dlynes | rdgzt: and do you have a secret= set? |
08:09.48 | rdgzt | It just has the secret now, no diff. |
08:09.49 | rdgzt | Yes. |
08:10.04 | rdgzt | And I've confirmed that I typed the secret correctly in the client. |
08:10.22 | dlynes | and do you have auth=md5? |
08:10.57 | rdgzt | In the sip context? |
08:11.02 | dlynes | correct |
08:11.03 | rdgzt | No? |
08:11.06 | dlynes | good |
08:11.06 | rdgzt | Do I need that? |
08:11.09 | rdgzt | Ok. |
08:11.10 | dlynes | no |
08:11.20 | dlynes | but that could mess things up |
08:11.33 | rdgzt | It contains just: |
08:11.34 | rdgzt | type=friend |
08:11.34 | rdgzt | secret=welcome |
08:11.34 | rdgzt | qualify=yes ; Qualify peer is no more than 2000 ms away |
08:11.34 | rdgzt | nat=no ; This phone is not natted |
08:11.34 | rdgzt | host=dynamic ; This device registers with us |
08:11.38 | rdgzt | canreinvite=no ; Asterisk by default tries to redirect |
08:11.40 | rdgzt | context=internal ; the internal context controls what we can do |
08:11.49 | dlynes | ~pb |
08:11.50 | jbot | hmm... pb is a place to paste your stuff without flooding the channel - try http://pastebin.com/, or http://pastebin.ca |
08:12.12 | rdgzt | Yeah, I know, figured it was short enough that it wouldn't matter much. |
08:12.26 | rdgzt | Not like there's a ton of activity right now, but sorry if that was out of line. |
08:12.40 | dlynes | well, this shouldn't affect the authentication |
08:12.58 | dlynes | but qualify is usually set to a numeric value |
08:13.01 | dlynes | such as 2000 |
08:13.25 | rdgzt | Yeah, according to the book, yes means the same as 2000, as it's the default. |
08:13.29 | rdgzt | But I can set it to 2000. |
08:13.40 | dlynes | but, as for authentication, it's probably a username or password mismatch |
08:13.51 | dlynes | if you turn on sip debug |
08:13.58 | dlynes | you can at least see if your username is correct |
08:14.07 | *** join/#asterisk saftsack (n=saftsack@p54A7FC22.dip.t-dialin.net) |
08:14.09 | dlynes | the password will be encrypted, though |
08:14.39 | rdgzt | From: <sip:john@10.0.254.35>;tag=1032e54d-21d8-da11-8234-0011098d4c7d |
08:14.42 | rdgzt | Etc.? |
08:14.48 | dlynes | correct |
08:14.52 | dlynes | you'll get an error there somewhere |
08:14.57 | dlynes | take a look at the error |
08:15.04 | dlynes | and look at the context that it's within |
08:15.11 | dlynes | i.e. look at the surrounding text |
08:15.19 | dlynes | You'll probably see the answer to your problems |
08:15.32 | rdgzt | Well, it seems it gets that request, and sends a response. |
08:15.38 | dlynes | exactly |
08:15.41 | *** join/#asterisk chapeaurouge (n=chap@user-85-201-82-146.tvcablenet.be) |
08:15.43 | dlynes | look at your request |
08:15.48 | dlynes | and then look at the response |
08:15.50 | rdgzt | SIP/2.0 404 Not found |
08:15.53 | rdgzt | What's up with that? |
08:15.55 | dlynes | there ya go |
08:16.01 | dlynes | you probably forgot to do sip reload |
08:16.24 | rdgzt | Nope, I've actually restarted asterisk completely several times. |
08:16.38 | dlynes | use pastebin, and paste your sip log, then |
08:16.40 | rdgzt | However, can I list the sip contexts on the asterisk console with settings, to see that it's loaded? |
08:16.55 | dlynes | sip show peer john |
08:17.10 | *** join/#asterisk Shoragan (n=shoragan@d072.apm.etc.tu-bs.de) |
08:17.10 | rdgzt | Peer john not found. |
08:17.11 | rdgzt | Interesting. |
08:17.13 | dlynes | there ya go |
08:17.15 | dlynes | not loaded |
08:17.25 | dlynes | imagine that :) |
08:17.46 | rdgzt | So, it's in my sip.conf, and I've restarted asterisk, is asterisk loading its config from somewhere different than I think, I wonder? |
08:17.58 | dlynes | /etc/asterisk/sip.conf? |
08:18.17 | rdgzt | Yeah, although I think I just figured out what's going on. |
08:18.32 | rdgzt | Asterisk doesn't handle install prefixes like autoconf programs do. |
08:18.38 | *** join/#asterisk janekm (n=janek@host81-157-239-71.range81-157.btcentralplus.com) |
08:18.49 | x86 | err |
08:18.52 | dlynes | look at /etc/asterisk/asterisk.conf |
08:18.53 | x86 | asterisk uses autoconf... |
08:18.58 | x86 | wtf are you talking about? :P |
08:19.05 | dlynes | x86: asterisk-trunk uses autoconf |
08:19.06 | rdgzt | If you give it /opt/asterisk as a install prefix, it puts its config in /opt/asterisk/etc/ |
08:19.10 | dlynes | and it's a bastardized autoconf |
08:19.27 | rdgzt | Unlike autoconf programs that use config_prefix (I think) for that. |
08:19.28 | dlynes | rdgzt: correct |
08:19.30 | rdgzt | Anyway, let me see. |
08:19.38 | x86 | rdgzt: "prefix" and "sysconfdir" are completely different though ;) |
08:19.39 | rdgzt | That probably just fixed my problem. |
08:19.41 | rdgzt | Right, sysconfdir. |
08:19.58 | dlynes | x86: Yeah, prefix sets INSTALL_PREFIX atm |
08:19.59 | rdgzt | x86: Non-autoconf asterisk thinks the install prefix is for everything. |
08:20.18 | dlynes | x86: russellb's aware of the problem, and he's working on a solution |
08:20.50 | *** join/#asterisk SheriF_WorK (n=sherif@212.103.170.135) |
08:20.53 | x86 | dlynes: right, and SYSCONFDIR is appended to INSTALL_PREFIX... standard way of life with autoconf |
08:21.03 | dlynes | x86: Yeah, but that's not standard |
08:21.12 | x86 | sure it is |
08:21.26 | x86 | the standard behaviour of autoconf, anyway |
08:21.29 | dlynes | because with standard autoconf, i can do make install INSTALL_PREFIX=/usr/local/src/staging, and it will actually stage |
08:21.41 | dlynes | with asterisk, it totally fubars everything |
08:21.43 | janekm | Hi everyone, I've been playing with originating calls from asterisk (using the call file interface). |
08:21.52 | x86 | dlynes: i see what you're saying |
08:22.07 | janekm | For my experiments I've been connecting the calls to the demo extension... |
08:22.10 | dlynes | If you're not doing staging, the asterisk autoconf is fine, though |
08:22.17 | x86 | dlynes: i thought you were talking about an autoconf issue, not automake ;) |
08:22.33 | janekm | But I haven't figured out a way to have asterisk drop its side of the call if the called party hangs up? |
08:22.50 | janekm | Haven't had much luck googling it but then perhaps I don't know the right terms yet... |
08:22.51 | dlynes | Yeah, but that's just it...asterisk doesn't do a standard autoconf...you have to use ./bootloader.sh or whatever it was called again |
08:23.12 | x86 | there is no configure.in and configure.am ? |
08:23.34 | dlynes | x86: You try generating configure from those, and the install fubars, unless russellb's fixed it |
08:23.44 | *** join/#asterisk A-Tuin|work (n=A-Tuin@212.41.185.81) |
08:24.03 | dlynes | it goes into the never-ending configure loop of hell |
08:24.07 | x86 | dlynes: works fine with gentoo's patches ;) |
08:24.22 | dlynes | x86: are you talking about asterisk-trunk, or asterisk-1.2.7.1? |
08:24.39 | x86 | asterisk 1.0.8 - 1.2.7.1 |
08:24.42 | dlynes | ah |
08:24.47 | dlynes | then that's a gentoo thing |
08:24.53 | dlynes | asterisk doesn't come with autoconf |
08:25.08 | x86 | hmm |
08:25.08 | dlynes | 1.2.8 or 1.3 will be the first to have it |
08:25.25 | dlynes | and it's not gnu autoconf compliant, either |
08:26.09 | dlynes | janekm: it should do it automatically |
08:26.24 | x86 | you're right... it is a gentoo thing |
08:26.25 | x86 | heh |
08:26.30 | x86 | never knew that :P |
08:26.47 | dlynes | I use slackware, so I always compile the original sources |
08:27.04 | dlynes | i just compile them once, and then deploy the binary packages everywhere |
08:27.05 | x86 | gentoo compiles the original sources too, but adds some cool patches ;) |
08:27.23 | x86 | same here... compile once, deploy binary |
08:27.48 | dlynes | gentoo doesn't use some horrid packaging method like rpm does it? |
08:27.55 | snitt | asterisk on gentoo? comeon, 1.0.8 is the stable afaik |
08:27.56 | *** join/#asterisk [Airwolf] (n=airwolf@83.98.235.219) |
08:28.02 | x86 | no way, it's source-based, like BSD's ports |
08:28.13 | dlynes | you said you deployed binaries |
08:28.16 | x86 | snitt: they have 1.2.7.1 in overlay... that's what i'm running |
08:28.18 | dlynes | binaries aren't source |
08:28.19 | janekm | dlynes: it hasn't done so far... It would keep the connection open (the called party was able to pick the phone off the hook again and still had it running) |
08:28.24 | x86 | dlynes: uh, right ;) |
08:28.34 | janekm | perhaps some of the changes I made to the demo extension could have screwed it up? |
08:28.35 | x86 | dlynes: you compile the source and get a binary ;) |
08:28.36 | dlynes | so what format are the deployed binaries? |
08:28.43 | x86 | dlynes: ah... tar/bz2 |
08:28.51 | dlynes | ah...so gentoo doesn't support packages, then, per se |
08:28.52 | snitt | x86: i've compiled 1.2.7.1 for gentoo by hand |
08:29.02 | snitt | for overlay.. |
08:29.05 | snitt | i dunno |
08:29.10 | snitt | i dont trust it simply |
08:29.24 | dlynes | janekm: no idea...but then again, i've only used call files for faxing |
08:29.25 | x86 | dlynes: sure it does |
08:29.26 | *** join/#asterisk mikl (n=mikl@pdpc/supporter/active/mikl) |
08:29.28 | *** join/#asterisk [Airwolf] (n=airwolf@83.98.235.219) |
08:29.42 | x86 | dlynes: all an RPM is, is a tar/gz with a preinstall and postinstall script ;) |
08:30.01 | mikl | err, when connected to an asterisk server in ekiga, how do I call out to "real" numbers? |
08:30.03 | dlynes | x86: then why is it I can never find any documentation that tells me how to make one? only how to install one? |
08:30.20 | x86 | dlynes: because you've never dug into the man page of 'rpm' ? |
08:30.50 | rdgzt | Ok, so now I have the sip context loaded and whatnot. Now, asterisk says: May 2 03:30:10 NOTICE[14353]: chan_sip.c:10886 handle_request_register: Registration from '<sip:john@10.0.254.35>' failed for '10.0.2.2' - Wrong password |
08:30.50 | dlynes | ummm |
08:30.51 | dlynes | yeah i have |
08:30.56 | dlynes | it's not terribly useful |
08:31.07 | x86 | dlynes: it explains how to create an RPM heh |
08:31.10 | dlynes | it mentions rpmbuild |
08:31.13 | dlynes | and that's about it |
08:31.14 | x86 | (check out -b, for example) |
08:31.19 | x86 | --build |
08:31.28 | dlynes | pattern not found |
08:31.43 | rdgzt | Which is even more weird, since I'm again pretty sure the password is correct on both ends. |
08:32.00 | dlynes | Yeah...no --build or -b switch |
08:32.16 | dlynes | rdgzt: Your password is incorrect |
08:32.21 | janekm | dlynes: I guess with faxes asterisk does know for sure when the transmission finished though... I guess I'll have to try it with sip debug sometime to see if asterisk actually gets any notification that the called party hung up |
08:32.30 | rdgzt | Now, in the SIP request, there's a mention of algorithm=md5, is that what's messing me up? |
08:32.32 | Zeeek | you gotta love a SIP phone that reboots in 5 seconds! |
08:32.37 | rdgzt | dlynes: No, it's really not. |
08:32.57 | dlynes | rdgzt: No, algorithm=md5 is normal for sip...you just don't specify it in your sip.conf file |
08:33.06 | *** join/#asterisk opus_ (n=opus@68.216.187.60) |
08:33.08 | *** part/#asterisk opus_ (n=opus@68.216.187.60) |
08:33.19 | dlynes | opus must be getting bored |
08:33.24 | dlynes | nobody talking to him in asterisk-dev |
08:33.47 | rdgzt | I'm 100% sure the password in the client matches the secret in sip.conf. |
08:33.51 | rdgzt | So what else could be tripping me up? |
08:33.59 | dlynes | rdgzt: sip debug |
08:34.14 | x86 | dlynes: dont you love repeating yourself? :P |
08:34.26 | rdgzt | Not that useful, SIP/2.0 403 Forbidden (Bad auth) |
08:34.29 | rdgzt | Is the response. |
08:34.31 | dlynes | x86: Well, it's pretty obvious what the problem is |
08:34.43 | dlynes | rdgzt: Your password is incorrect |
08:34.55 | dlynes | rdgzt: Try retyping your password into your sip client |
08:34.55 | rdgzt | I've retyped the secret in sip.conf, and in the client, restarted asterisk, repeat 3 times. |
08:35.02 | rdgzt | So no, the password is *not* incorrect. |
08:35.04 | x86 | dlynes: hehe, you told him like 50 times and he still asks ;) |
08:35.06 | rdgzt | I'm not an idiot. |
08:35.07 | x86 | dlynes: it's kinda funny ;) |
08:35.18 | dlynes | Well, that's the only explanation for that error message |
08:35.34 | dlynes | If it's not the truth, there's something fubared with your install |
08:35.59 | dlynes | And one can only guess what's wrong then |
08:36.11 | rdgzt | That's interesting, but doesn't really get me that far. |
08:36.15 | x86 | rdgzt: look at your extconfig.conf... it's not pointing to mysql for anything is it? |
08:36.47 | rdgzt | It's the standard example file, looks like everything's commented out. |
08:36.49 | dlynes | x86: i doubt it...he's gotten this far...he's not getting a userid/password mismatch anymore...it's just a password mismatch now |
08:37.11 | x86 | true |
08:37.35 | dlynes | i suspect he's probably including one context into another |
08:37.42 | rdgzt | Thing is, this exact problem happened to me two days ago, and was why I went from the prebuilt binaries on Ubuntu to installing from source, since I wasn't at the latest version. |
08:37.47 | rdgzt | dlynes: How do I tell? |
08:37.50 | dlynes | and not realizing it because a include => is covered up by a #include |
08:37.55 | dlynes | or something similar |
08:38.07 | rdgzt | Are we talking extensions.conf here? |
08:38.15 | dlynes | i was giving an example |
08:38.24 | dlynes | for sip.conf, the only thing you could be doing is doing a #include |
08:38.41 | dlynes | and in that case, you probably have a sip context defined where you're not expecting one |
08:39.23 | dlynes | If all of your sip clients are defined in one big sip.conf file, there's no guessing |
08:39.36 | dlynes | but otoh, it makes it harder to manage then |
08:40.02 | rdgzt | Let me try to cut down my sip.conf, it's basically just the sample file with this one context added at the end. |
08:40.16 | x86 | realtime > sip.conf, imho |
08:40.18 | dlynes | yeah...get rid of all the sample cruft |
08:40.38 | dlynes | x86: yeah, but if you're having troubles with sip.conf, you're going to have real troubles with realtime, i woudl imagine |
08:41.14 | dlynes | better to get the simple case working first |
08:41.20 | dlynes | and then gradually go more advanced |
08:43.02 | rdgzt | Ok, so with a clean sip.conf, the same thing happens. |
08:43.06 | *** join/#asterisk oej (n=oej@apollo.webway.se) |
08:43.19 | rdgzt | Clean defined as a general section and then my context. |
08:44.40 | dlynes | sip show peers |
08:44.43 | dlynes | What do you get? |
08:44.53 | rdgzt | Hm, wait, I think I solved it. |
08:45.19 | rdgzt | Realm defaults to "asterisk", while the client was trying to use the server's IP as the realm. |
08:45.31 | rdgzt | Not the most obvious thing to get a "wrong password" error in that case, perhaps. |
08:46.09 | dlynes | heh |
08:46.15 | dlynes | first time i've heard of that problem |
08:46.25 | dlynes | every client i've tried using just ignores the realm |
08:46.38 | rdgzt | I'd argue that that error message is plain wrong. But well, I got it working, which is the important thing. |
08:46.39 | mikl | argh, I can't dial out :( |
08:46.49 | *** join/#asterisk |cleric| (n=dacleric@p54821028.dip0.t-ipconnect.de) |
08:46.53 | mikl | how do you configure Ekiga to work with asterisk? |
08:46.56 | *** join/#asterisk Tangent (n=Tangent@connerdata.plus.com) |
08:47.13 | dlynes | would help if you could explain what ekiga is |
08:47.20 | saftsack | new gnome meeting |
08:47.28 | dlynes | ah yeah...i forgot |
08:47.35 | dlynes | why'd they change the name, anyways? |
08:47.39 | dlynes | Just to confuse us? |
08:47.44 | saftsack | i think so ^^ |
08:48.02 | mikl | dlynes: no idea - guess gnomemeeting wasn't sexy enough :) |
08:48.11 | dlynes | mikl: don't you just configure it as a sip client? |
08:48.28 | saftsack | yes i would guess too |
08:49.15 | mikl | dlynes: yes, I can connect to the server, but I can't dial out - If i try to enter a normal phone number in the field, I just get "Abnormal call termination" |
08:49.32 | dlynes | I would check the asterisk logs |
08:49.39 | dlynes | the client isn't gonna tell you why |
08:52.10 | *** join/#asterisk gmaruzz (n=gmaruzz@217-133-80-112.b2b.tiscali.it) |
08:52.13 | mikl | meh :( |
08:52.28 | mikl | the server isn't mine, so i can't really check the logs :( |
08:53.12 | dlynes | tell them to fix their server then |
08:53.28 | dlynes | the only other problem it could be probably is a codec mismatch |
08:53.42 | dlynes | they might be trying to get you to use a codec you don't have |
08:53.46 | dlynes | such as g729 |
08:53.51 | mikl | ah :/ |
08:53.52 | *** join/#asterisk bintut (n=bintut@202.164.162.222) |
08:54.05 | dlynes | which in that case |
08:54.12 | dlynes | is not a very good error message |
08:54.29 | mikl | true |
08:56.02 | janekm | mikl: just to check the obvious, do you know if that server can dial out to the same number using a different client? |
08:56.21 | mikl | janekm: I've tried in twinkle, that works fine |
08:56.50 | *** join/#asterisk Gunnar (n=gunnar@bkkb-gw.bitcon.no) |
08:57.12 | dlynes | mikl: yeah, so it's probably either your username/password, or your codec settings then |
08:58.11 | dlynes | anyways...back to coding |
09:05.27 | *** join/#asterisk littlejohn (n=little@host146-255.pool8289.interbusiness.it) |
09:07.50 | *** join/#asterisk KPax (n=ask@droid.chaosmedia.org) |
09:08.47 | *** join/#asterisk SheriF_WorK (n=sherif@212.103.170.135) |
09:09.43 | *** join/#asterisk Assaf (n=Drake@62.90.49.96) |
09:11.40 | *** join/#asterisk Assaf (n=Drake@62.90.49.96) |
09:13.11 | dlynes | man...just when i want to get something done on the server |
09:13.22 | dlynes | some clown logs in wanting to read three weeks worth of voicemail |
09:13.26 | janekm | mikl: I imagine that the folks in the ekiga channel might know a bit more about those sorts of things, have you tried there? |
09:13.34 | janekm | dlynes: heh ;) |
09:14.34 | shiznatix | when a phone call comes in on a zapata card how can you tell what number that phone line has...like the number that the caller dialed. ${CALLERID(num)} does not work |
09:14.41 | dlynes | what's he doing in at his office at 2am, anyways? |
09:15.06 | dlynes | ${CALLERIDNUM} |
09:15.16 | dlynes | if neither one of those work, |
09:15.26 | dlynes | a) that caller is 'unknown caller' |
09:15.38 | dlynes | b) your line doesn't have caller id subscription service |
09:15.46 | shiznatix | no i dont want the caller's number, i want my number |
09:15.56 | dlynes | oh yeah...nvm |
09:16.00 | dlynes | ${DNID} |
09:16.06 | dlynes | or ${EXTEN} |
09:16.09 | dlynes | heh |
09:16.13 | dlynes | too late |
09:16.24 | shiznatix | i have 4 lines coming in to a zapata card all on the same context. they are all fax lines. i want to organize my faxes based off number which is easy but i just need the number of the line to do that |
09:17.21 | *** join/#asterisk L|NUX (n=linux@202.5.145.56) |
09:17.22 | *** join/#asterisk RoyK (n=roy@80.239.107.70) |
09:17.23 | dlynes | exten => 6042663001,1,... exten => 6042663002,1,... exten => 6042663003,1,... exten => 6042663004,1,.... |
09:17.40 | dlynes | shiznatix: is that not how you're handling them? |
09:17.55 | jql | you want the actual channel number? ${CHANNEL}? |
09:18.11 | dlynes | yeah, or what jql suggested |
09:18.18 | RoyK | tag |
09:18.46 | shiznatix | dlynes, no im doing it like this |
09:18.57 | shiznatix | dlynes, s,1,Dial(... |
09:19.33 | shiznatix | and it has to stay like that because people are idiots and this is being shipped to many different places so the numbers will change |
09:19.47 | dlynes | shiznatix: try jql's suggestion then |
09:20.11 | jql | yeah, people do tend to be idiots |
09:20.16 | dlynes | shiznatix: but i think it might actually be more like ${DESTCHANNEL} or something |
09:20.50 | OloBola | why is callerID not always supported? |
09:20.56 | dlynes | ok, or not |
09:21.14 | jql | because callerid isn't required to complete a call. sadly |
09:21.16 | dlynes | seems kinda silly destchannel, but it's not specifying whether it's the called channel, or the calling channel |
09:21.26 | KPax | lo all, is it okay to ask a few basic questions ? i've read some docs and installation manuals but there's a few things i'd like to clear up |
09:21.30 | janekm | OloBola: Some phone companies like to charge extra to send that info to your line... |
09:21.39 | dlynes | KPax: don't ask if it's ok to ask, just ask |
09:21.56 | OloBola | sucks butt |
09:22.29 | janekm | (or, of course, some people just don't want you to know their number so you can't complain about their agressive sales tactics...) |
09:23.12 | dlynes | or their aggressive collections tactics |
09:24.39 | KPax | k, well i'd like to know if asterisk config can be stored in a database (mysql would be better), i've read that mysql support was removed for license reasons but it was still available thru some addons, unfortunately i can't really find a clear tutorial on how to enable mysql support.. asterisk install from source worked nice on my debian sarge. |
09:24.59 | *** join/#asterisk salviadud (n=noyb@dsl-201-129-86-188.prod-infinitum.com.mx) |
09:25.12 | salviadud | !seen _paulo_ |
09:25.28 | bintut | what are the basic features for an asterisk voip and pbx that are easy to configure to beginners? :) |
09:25.42 | salviadud | sip and iax chans |
09:25.53 | salviadud | and um... a default context that dials out |
09:26.01 | salviadud | to FWD for example |
09:26.07 | salviadud | that's what i did as a newbie |
09:26.10 | dlynes | ~seen _paulo_ |
09:26.16 | jbot | _paulo_ <n=pirch@201-13-16-73.dsl.telesp.net.br> was last seen on IRC in channel #asterisk, 13d 13h 40m 48s ago, saying: 'Why dont Borland made Delphi with C++?'. |
09:27.25 | salviadud | dlynes, you know paulo? |
09:27.25 | dlynes | nope |
09:27.26 | *** join/#asterisk v3rmap (n=puser@unaffiliated/v3rmap) |
09:27.26 | dlynes | i was just showing you it's '~seen', not '!seen' |
09:27.26 | KPax | when i look for mysql stuff i usually end up on some AMP tutorials, should i start from there ? |
09:27.26 | salviadud | i got a private message with !seen |
09:27.26 | salviadud | look |
09:27.26 | salviadud | [04:25] -GerbilWrk- [LAST SEEN] "_paulo_" - n=pirch@201-13-16-73.dsl.telesp.net.br Quit 1wk 6days 11hrs 54mins 54secs ago Last Note: private/secret channel Quit: (freenode) Remote closed the connection |
09:27.30 | dlynes | ah |
09:27.40 | salviadud | lots of ways i guess |
09:28.10 | salviadud | i need to find that dude. |
09:28.26 | salviadud | my prank calls depend on him |
09:28.48 | salviadud | i feel so proud, i do my pranks with asterisk |
09:28.58 | v3rmap | My asterisk installation on Ubuntu is not listening on port 5060, so sip phones can't login. Any suggestions on what could be wrong with my installation?? |
09:29.12 | salviadud | v3rmap, iptables? |
09:29.19 | salviadud | btw, i hate ubuntu... |
09:29.23 | dlynes | KPax: http://www.google.com/custom?tk=d86a5ba922fc092a368c&domains=www.voip-info.org&q=mysql&sitesearch=www.voip-info.org&sa=Google+Search&client=pub-6210650267389726&forid=1&ie=ISO-8859-1&oe=ISO-8859-1&cof=GALT%3A%23008000%3BGL%3A1%3BDIV%3A%23336699%3BVLC%3A663399%3BAH%3Acenter%3BBGC%3AE9ECEF%3BLBGC%3AFFFFFF%3BALC%3A0000FF%3BLC%3A0000FF%3BT%3A000000%3BGFNT%3A0000FF%3BGIMP%3A0000FF%3BLH%3A20%3BLW%3A100%3BL%3Ahttp%3A%2F%2Fwww.voip-info.org%2Fimages%2FVOI |
09:29.24 | dlynes | P-info.jpg%3BS%3Ahttp%3A%2F%2Fwww.voip-info.org%3BLP%3A1%3BFORID%3A1%3B&hl=en |
09:29.27 | dlynes | ack |
09:30.02 | dlynes | anyways...append the 'P-info.jpg' to the '2FVOI', KPax |
09:30.52 | v3rmap | salviadud: iptables --list shows this: http://pastebin.com/693802 I hope this means I have no active firewall? |
09:30.58 | dlynes | v3rmap: edit your /etc/asterisk/modules.conf file so that you have a line: load => chan_sip.so |
09:31.13 | janekm | dlynes: the first part of the URL seems to work too |
09:31.24 | dlynes | v3rmap: it has nothing to do with your firewall |
09:31.40 | v3rmap | dlynes I'll check that ...just a min. |
09:31.43 | dlynes | cool |
09:31.44 | salviadud | yeah, looks like you're open like a pornstar |
09:32.07 | dlynes | nah...even pornstars aren't that open |
09:32.16 | v3rmap | hehe |
09:32.53 | salviadud | you know, back in the day... i used to trade pr0n |
09:33.01 | salviadud | but NOOOOW |
09:33.12 | dlynes | now you keep it all to yourself |
09:33.19 | salviadud | they got their fingle fangled kazaa's and emules |
09:33.26 | janekm | now you spend your time prank calling people with asterisk instead? ;) |
09:33.45 | salviadud | and here on irc, no more trades, just leech |
09:33.53 | salviadud | i dunno if that's good or bad |
09:33.57 | dlynes | you suck |
09:34.19 | salviadud | ugh? elaborate |
09:34.25 | dlynes | you said you were a leech |
09:34.27 | dlynes | leeches suck |
09:34.44 | salviadud | of course, i suck the pr0n off the fserves |
09:35.02 | dlynes | i'd hate to see what else you suck \;) |
09:35.07 | salviadud | i miss the trades |
09:35.29 | salviadud | i suck suculent women, which i also prank call |
09:35.54 | janekm | salviadud: have you tried the ratio-based bittorrent sites? or directconnect? Might satisfy some of the nostalgic feelings... |
09:36.28 | salviadud | it's not the same... i liked the social engineering bit |
09:36.34 | salviadud | asking around |
09:37.18 | janekm | that's true, the personal connection to the pusher is really lacking... |
09:37.26 | *** join/#asterisk gmaruz1 (n=gmaruzz@217-133-80-112.b2b.tiscali.it) |
09:37.57 | salviadud | yeah, you see, if i found out some dude had lots of... fat girl vids |
09:38.06 | salviadud | then, i would have this really big conversation |
09:38.20 | salviadud | like "dude, they're really big... why would you like 'em big?" |
09:38.40 | salviadud | and he would go "cause i'm a skinny ass dude, i need some meat" |
09:38.40 | janekm | perhaps #empornium might amuse? |
09:38.45 | salviadud | you know, funny stuff |
09:38.49 | janekm | true |
09:39.07 | salviadud | empornium? is that here on freenode? |
09:39.24 | janekm | irc.whatnet.org apparently |
09:39.41 | salviadud | i consider freenode to be a very subtle network, full of smart people and devs |
09:39.57 | janekm | haven't been there actually, just imagine it might be amusing since that's a fairly community-oriented torrent site... |
09:40.15 | janekm | hence you probably won't find as much pr0n amusement on freenode ;) |
09:40.32 | v3rmap | dlynes, I added "load => chan_sip.so" in modules.conf and restarted asterisk. The login from the sip phone still timeouts and "netstat -a | grep 5060" returns nothing. Anything else I could try? |
09:40.40 | Johnnie | You can always start. |
09:40.49 | Johnnie | A little T&A never hurt. |
09:41.02 | salviadud | you see, it's not about the pr0n, broadband has made leeching cheap |
09:41.04 | Johnnie | You can liberate FreeNode. |
09:41.33 | salviadud | T&A? |
09:41.40 | salviadud | you mean R&R? |
09:41.45 | janekm | http://www.urbandictionary.com/define.php?term=t+%26+a |
09:41.59 | janekm | had to look it up too ;) |
09:42.40 | Johnnie | hahaha |
09:43.31 | salviadud | anybody else like to prankcall? |
09:43.42 | *** join/#asterisk faljse (n=martin@213.235.245.210) |
09:43.43 | salviadud | i speak 3 different languages |
09:43.51 | salviadud | and 2 kinds of spanish |
09:43.52 | janekm | looks like they're playing trivia on #empornium so maybe it's not that amusing after all ;) |
09:44.24 | faljse | is there a way i can limit the call duration(just the billsec time...)? |
09:44.27 | janekm | mexican and, eh, spanish spanish? |
09:44.43 | salviadud | yeah, castellano |
09:44.46 | salviadud | maybe i can do more |
09:44.51 | salviadud | argentinian spanish |
09:44.53 | salviadud | dominican spanish |
09:45.03 | salviadud | it's freakin easy |
09:45.09 | salviadud | i used to be an international operator |
09:45.13 | janekm | ah, right |
09:45.24 | salviadud | i used to be a monkey, taking calls all day |
09:45.43 | janekm | salviadud: I was sort of prank calling people yesterday trying to get originate to work ;) |
09:45.59 | salviadud | yeah, that's the spirit |
09:46.09 | salviadud | you prank while you test, win/win |
09:46.43 | janekm | was at a friend's place while I'm waiting for BT to fix my landline... |
09:47.28 | *** part/#asterisk bintut (n=bintut@202.164.162.222) |
09:47.28 | janekm | they were meant to enable ADSL on the line and instead cross connected my incoming number to some little old scottish lady... |
09:47.53 | salviadud | where are you from janekm? |
09:47.57 | janekm | well I suppose I don't know if she's little that's just my imagination... |
09:48.03 | janekm | In Scotland |
09:48.16 | salviadud | i can do british english too |
09:48.22 | salviadud | i find it impossible to do pikey |
09:48.23 | janekm | oh, sorry I'm from Germany but have lived in Scotland for 10 years so I guess I get confused ;) |
09:48.35 | salviadud | germany, very interesting |
09:49.02 | salviadud | i suppose i'll have to speak german sooner o later |
09:49.29 | salviadud | i can say basic stuff |
09:49.43 | salviadud | ich bin studentin |
09:49.57 | salviadud | ich heisse Salviadude |
09:50.08 | janekm | it's a horrible language to try and learn... I always get really lost when friends ask me about something grammatical and I can't coherently explain how it works ;) |
09:50.16 | salviadud | heir ist der schlangemann |
09:50.36 | salviadud | well, i got this book, learn german in 40 lessons |
09:50.43 | *** join/#asterisk psk (n=psk@golia.caltanet.it) |
09:50.45 | salviadud | i found it very english-like |
09:50.54 | salviadud | lots of similarities |
09:51.14 | salviadud | yet, i was only reading it, i would need to prank call more germans to get the hang of it |
09:51.40 | salviadud | well, i'm gonna go to sleep |
09:51.45 | salviadud | got work in the mornin' |
09:51.45 | janekm | there are lots of similarities, but those are what trips people up with more complex things where it's different... |
09:51.53 | salviadud | nice talking to you, janekm |
09:51.57 | janekm | you too |
09:52.11 | *** part/#asterisk salviadud (n=noyb@dsl-201-129-86-188.prod-infinitum.com.mx) |
09:54.30 | *** join/#asterisk apardo (n=apardo@87.217.145.111) |
09:55.50 | *** join/#asterisk Greek-Boy (n=grb@193.220.93.162) |
09:55.58 | *** join/#asterisk X-Gen (n=X-Gen@dsl-145-220-183.telkomadsl.co.za) |
09:56.17 | X-Gen | hey freaks |
09:56.44 | Greek-Boy | any voip software for Nokia 6680? |
10:04.09 | *** join/#asterisk niZon (n=ilt@S010600080db4ab60.wp.shawcable.net) |
10:06.52 | Zeeek | vas ist das? |
10:09.13 | *** join/#asterisk MrEntropy (n=entropy@ppp142-239.lns2.adl2.internode.on.net) |
10:09.13 | MrEntropy | yo |
10:10.20 | MrEntropy | can i configure a hunt group in asterisk? i dont want to call all phones at once and whichever one picks up first wins, i want the first not-busy phone to ring exclusively from a list of phones. |
10:10.58 | Ahrimanes | MrEntropy: hm queue with round robin strategy? |
10:11.45 | Greek-Boy | doesn't anybody know voip software that will work with symbian OS on 6680? |
10:12.15 | Ahrimanes | Greek-Boy: hm i havent found any yet.. not for qtek 8310 either.. |
10:12.18 | MrEntropy | Ahrimanes: what config file is that configured in? |
10:12.31 | X-Gen | Greek-Boy, write some, i dont think there is any |
10:12.52 | starlein | Greek-Boy: http://www.forum.nokia.com/main/0,6566,034-561,00.html |
10:12.59 | Ahrimanes | MrEntropy: queues.conf |
10:13.02 | starlein | go and develope your own |
10:14.03 | Greek-Boy | thanks |
10:14.06 | Greek-Boy | lol |
10:15.08 | dlynes | X-Gen: there is some...they just suck |
10:15.33 | Greek-Boy | dlynes how u doing |
10:15.43 | dlynes | tired |
10:15.45 | Ahrimanes | but nokia are launching their own voip clients on some of the new phones |
10:15.45 | Greek-Boy | do they work, even though they suck? |
10:16.05 | dlynes | well, i woudl guess if they charge money for them, they must work |
10:16.47 | Ahrimanes | dont bet too much on that |
10:17.01 | Greek-Boy | lol |
10:17.03 | X-Gen | Ahrimanes, SUCKER !!! |
10:17.09 | Ahrimanes | X-Gen: :P |
10:17.35 | Ahrimanes | if only my homebanking software and such would work under freebsd.. |
10:17.58 | X-Gen | i suppose writing a SIP client in Java will really suck on a mobile device |
10:19.32 | janekm | I would expect so... |
10:19.56 | janekm | but on the S60 mobiles you should be able to write it using the Psion API too I guess |
10:19.58 | Ahrimanes | s/on a mobile device// |
10:20.35 | janekm | so you should be able to write it in C++ |
10:22.28 | janekm | ah I just read the link starlein posted, looks like Nokia already released a SIP plugin for the SDK |
10:22.35 | v3rmap | ppl, I added "load => chan_sip.so" in modules.conf and restarted asterisk. The login from the sip phone still timeouts and "netstat -a | grep 5060" returns nothing. Anything else I could try? |
10:22.57 | janekm | so that should make it really easy, you'd have thought someone would have released a SIP phone for S60 based on that by now? |
10:23.10 | v3rmap | ^^ That means asterisk is not listening on port 5060, any ideas whay that could be so? |
10:24.23 | *** join/#asterisk apardo (n=apardo@87.217.145.111) |
10:25.24 | *** join/#asterisk syle (n=blah@unaffiliated/syle) |
10:25.26 | *** join/#asterisk backblue (n=igor@82.102.1.42) |
10:25.33 | backblue | morning all |
10:26.06 | Skid__ | question: is there a way to "ping" a sip phone, using some sip command? |
10:26.28 | Skid__ | sip notify? |
10:26.38 | Ahrimanes | sip options ? |
10:26.39 | *** join/#asterisk r0d3nt|m (i=r0d3nt@tinfoilhat.net) |
10:26.59 | Skid__ | I want to see if the phone is alive |
10:27.39 | Ahrimanes | Skid__: hm for what purpose? are you using qualify=yes in asterisk? |
10:29.17 | Skid__ | Ahrimanes: I want to make a call-forward if a phone is nologer reachable. |
10:29.41 | *** join/#asterisk [Airwolf] (n=airwolf@83.98.235.219) |
10:29.45 | Skid__ | the phone is a member of a queue |
10:30.19 | Skid__ | and I dont want to use call agents |
10:30.58 | Ahrimanes | Skid__: http://www.voip-info.org/wiki-Asterisk+call+forwarding <- *61* should be useful for you |
10:34.32 | backblue | ppl, anyonw know something about dialing in block, or digit by digit, some convencional pbx (alcatel,siemens) do that... they dial digit by digit, how can i put asterisk receiving digit by digit? |
10:34.33 | Skid__ | Ahrimanes: not that simple.. |
10:34.53 | Ahrimanes | Skid__: could be done with that |
10:35.37 | Skid__ | Ahrimanes: I dont think you have the whole picture.. |
10:36.19 | Ahrimanes | Skid__: phone is in the queue, if it's not available, as in unreachable or offline, you want to forward the call somewhere else? |
10:37.07 | syle | if you use canreinvite, do you loose cdr record for talk time on that call? |
10:37.43 | Skid__ | Ahrimanes: If both phones in the queue ar not available or not reachable.. I want to forward all calls.. and not use the queue anymore |
10:39.05 | Ahrimanes | Skid__: then add exten => 1234,N,Dial(SIP/otherphone) after exten => 1234,N,Queue()... if noone in the queue can answer it will jump to the next priority |
10:39.32 | Skid__ | Ahrimanes: It dont work. |
10:39.51 | Skid__ | becase I dont use agents |
10:40.08 | Ahrimanes | Skid__: i use dynamic members in the queues and it works for me |
10:41.30 | Skid__ | Ahrimanes: well.. Is it possible to autamatic logon an sip client to the queue? |
10:42.31 | *** join/#asterisk ivanfm (n=ivanfm@201-27-67-81.dsl.telesp.net.br) |
10:42.34 | Ahrimanes | Skid__: yes i have setup a button on a snom phone that dials an agi that uses cmd add queue member SIP/1234 to queue queue1 fx |
10:43.50 | Skid__ | Skid__: that is not automatic enough :( |
10:43.58 | Skid__ | oups |
10:44.16 | Ahrimanes | Skid__: you want to add the sip client as soon as it registers? |
10:44.25 | shiznatix | im trying to get the phone number of the zapata line that the call is coming in on. i have tried these ${DNID}-${CHANNEL}-${DIALEDPEERNUMBER} but none of them are the number of the line. how can i get the number of the line? |
10:44.40 | Skid__ | Ahrimanes: rigtht. |
10:45.02 | Ahrimanes | Skid__: i dont believe this is currently possible.. but i have some ideas on how to do it.. will require some code though |
10:45.45 | Zeeek | Beer is currently possible though |
10:46.33 | Skid__ | Ahrimanes: for the moment I have a static entry for every phene in the queue. |
10:46.39 | Ahrimanes | Zeeek: hehe |
10:46.42 | *** join/#asterisk chapeaurouge (n=chap@user-85-201-82-146.tvcablenet.be) |
10:47.10 | *** join/#asterisk stoffell (n=stoffell@fw.catsanddogs.com) |
10:47.10 | Ahrimanes | Skid__: hm ok, did you look at levewhenempty and related options in queues.conf ? |
10:47.21 | Skid__ | Ahrimanes: yeah. |
10:47.45 | Skid__ | Ahrimanes: "If you have any static queue members defined as "Member => Agent/XXXX" in your Queue definiation, Asterisk will considered the Queue in-use" |
10:48.38 | Ahrimanes | Skid__: ah yes |
10:48.59 | *** join/#asterisk rkr245 (n=ravi@office.callsat-telecom.com) |
10:49.19 | Ahrimanes | Skid__: my idea on the automatic way: listen on the manager interface for register, unreachable and logoff events and have a script add and remove queue members based on those events |
10:49.49 | Ahrimanes | using add queue member and remove queue member |
10:50.43 | Skid__ | Ahrimanes: Is there possible to run a script extensions.conf ? |
10:51.07 | Ahrimanes | Skid__: hm no, but you can run commands via the manager interface |
10:51.08 | Skid__ | that check status for any phone in the queue |
10:51.39 | Ahrimanes | Skid__: i think reading up on the manager interface would help you |
10:51.53 | Skid__ | Ahrimanes: looks like so.:( |
10:52.29 | Ahrimanes | Skid__: it should be doable in something like 100 lines of perl |
10:52.52 | backblue | how can i put asterisk waiting for X number of digits, when some zap channel it's dialing? |
10:53.06 | Skid__ | Ahrimanes: but.. to many ways of things going wrong:( |
10:53.25 | Ahrimanes | Skid__: hm how? |
10:56.07 | Ahrimanes | Skid__: i've been doing perl interaction with the manager interface for a long time now, only problems i had were hardware related |
10:57.11 | Skid__ | Ahrimanes: well, I havent.. and It must work at 100% today. |
10:58.55 | Ahrimanes | Skid__: hm, then i'd contact digium and have a large amount of money at hand |
10:59.54 | Skid__ | Ahrimanes: well I dont:) |
11:00.00 | *** join/#asterisk AsteriskAlbania (n=info@217.24.244.130) |
11:00.19 | AsteriskAlbania | asterisk <-> Radius ? any help please |
11:00.31 | Skid__ | Ahrimanes: tanx anyway. |
11:00.42 | Ahrimanes | np |
11:00.59 | Ahrimanes | AsteriskAlbania: what info do you want to get from/save to radius? |
11:02.38 | AsteriskAlbania | I need to nake asterisk talk to radius |
11:02.44 | AsteriskAlbania | make sorry |
11:02.58 | Ahrimanes | AsteriskAlbania: something like http://www.voip-info.org/tiki-index.php?page=PortaOne+Radius+auth ? |
11:03.19 | AsteriskAlbania | Ahrimanes: I have tried it but there is a patch for 1.0.9 |
11:03.42 | syle | radius unreliable, why use it |
11:04.04 | Ahrimanes | radius is quite reliable if setup correctly like most things |
11:04.07 | AsteriskAlbania | I have a billing system with alepo |
11:04.09 | syle | funny |
11:04.25 | AsteriskAlbania | what do you think is the best way for billing |
11:04.36 | syle | i won;t even begin listing all the problems using radius |
11:04.45 | syle | miss calls if you want |
11:05.17 | syle | best is to embed directly into asterisk |
11:05.32 | AsteriskAlbania | syle: which is the best way for doing billing with asterisk |
11:05.49 | Ahrimanes | hm, so the current asterisk radius implementations are faulty.. but that's not radius' fault |
11:05.50 | syle | use the existing c source and code c functions to do what you need |
11:06.18 | AsteriskAlbania | syle: I am not good at programming myself |
11:07.12 | syle | well that sucks, well use what you can i guess |
11:07.17 | janekm | yikes the nokia n80 is one expensive phone... |
11:07.27 | Ahrimanes | AsteriskAlbania: http://www.paskambink.lt/mcc/index.php?option=com_content&task=view&id=77&Itemid=1 works for me |
11:07.41 | AsteriskAlbania | let me check it |
11:14.39 | *** join/#asterisk chapeaurouge (n=chap@user-85-201-82-146.tvcablenet.be) |
11:15.09 | shiznatix | im trying to get the phone number of the zapata line that the call is coming in on. i have tried these ${DNID}-${CHANNEL}-${DIALEDPEERNUMBER} but none of them are the number of the line. how can i get the number of the line? |
11:16.10 | Ahrimanes | shiznatix: you're matching the call on an exten => XX ? |
11:16.42 | Skid__ | Ahrimanes: Hey.. It will work.. I connected to the manager interface and removed the static members of a queue..:) |
11:17.09 | shiznatix | Ahrimanes, no, its all on the s extension. 4 lines share the same context and I need to split them up by the number of that phone line |
11:17.22 | Ahrimanes | Skid__: cool :) |
11:17.30 | AsteriskAlbania | Ahrimanes: Have you tested MCC |
11:17.51 | Ahrimanes | AsteriskAlbania: yes, am currently running it on a test server and doing code changes to fit it to my needs |
11:19.12 | Ahrimanes | shiznatix: hm ok, usually that's what you use exten => for.. maybe ${EXTEN} will tell you the number though |
11:19.15 | *** join/#asterisk razu_ (n=razu@tln-kontor.norby.ee) |
11:20.01 | I-MOD | shiznatix: you should probably just make a macro that gets the number from the channel |
11:30.46 | joelsolanki | Hi all. planning to buy tdm244E from digium. but i want to know which cable they will provide which i have to join on tdm244E card ? are they going to ship the cable too ? |
11:32.36 | joelsolanki | syle: u there / |
11:32.56 | *** join/#asterisk chapeaurouge (n=chap@user-85-201-82-146.tvcablenet.be) |
11:33.03 | *** join/#asterisk apardo (n=apardo@87.217.144.132) |
11:33.23 | Ahrimanes | joelsolanki: call up digium and ask :) |
11:33.47 | joelsolanki | hehe :) i m poor with cards so dont know much about it. so thought to ask on list. |
11:34.02 | Ahrimanes | hehe |
11:34.16 | Ahrimanes | well i'm sure digium can answer any and all questions about their products :) |
11:35.10 | joelsolanki | Anybody on list can answer my small question :( |
11:37.22 | *** join/#asterisk coppice (n=chatzill@153.192.17.210.dyn.pacific.net.hk) |
11:38.16 | MrChimpy | joel: if your question made sense you might get an answer |
11:39.00 | Zeeek | the question is: what is small? |
11:39.18 | Ahrimanes | Zeeek: dont get me started |
11:40.48 | joelsolanki | i m planning to buy tdm244E which has 4 FXS module and 2 FXO module. means i can have 16 FXS ports and 8 FXO ports. |
11:41.31 | MrChimpy | can't find 244E on digium's site |
11:41.32 | joelsolanki | now i want to know where do i plug the pstn lines. ? |
11:42.20 | MrChimpy | RTFM, dude |
11:42.21 | Ahrimanes | joelsolanki: TDM2400P ? |
11:42.30 | Zeeek | best to just call them |
11:42.32 | joelsolanki | yes |
11:42.33 | joelsolanki | http://www.voipsupply.com/product_info.php?products_id=1152 |
11:42.39 | joelsolanki | above is the link. |
11:42.41 | MrChimpy | FXO/FXS stuff is all explained in the docs |
11:42.45 | Zeeek | call voipsupply - or email Corey |
11:42.55 | Zeeek | he's good about giving the answers |
11:43.20 | joelsolanki | who is the email id of corey ? |
11:43.35 | MrChimpy | i'm sure corey will be pleased |
11:44.02 | MrChimpy | probably want to get the product code right before asking |
11:44.21 | joelsolanki | can anybody give me the email id of corey ? |
11:45.44 | Zeeek | there's the interent - that's what it's for! |
11:46.22 | MrChimpy | http://www.voipsupply.com/product_info.php?&products_id=1164 |
11:46.27 | MrChimpy | i suspect that is your answer |
11:46.58 | Zeeek | be sure to buy the tech support option |
11:47.00 | tzanger | why on earth would you use one of those things |
11:47.03 | MrChimpy | horrid idea using those things though |
11:47.09 | tzanger | terminate to BIX and be happy |
11:47.25 | tzanger | any office with more than a half dozen phones will have terminated to BIX anyway |
11:47.26 | MrChimpy | horrid idea using fat analogue cards for starters |
11:47.33 | joelsolanki | Yes i founded that :) ;) |
11:47.43 | joelsolanki | MrChimpy: thanks :) |
11:48.10 | tzanger | I mean hell they make pre-punched D50->BIX cables |
11:48.18 | tzanger | like $3 |
11:52.18 | AsteriskAlbania | have any installed asterisk on FC5 |
11:52.28 | AsteriskAlbania | I mean DIGIUM CARDS |
11:52.34 | Greek-Boy | what is BIX? |
11:52.40 | AsteriskAlbania | any problem ? |
11:52.52 | Zeeek | BIX was the name of one of my dad's dogs |
11:53.05 | Greek-Boy | seriously |
11:53.10 | Greek-Boy | for us newbies |
11:53.12 | Greek-Boy | what is BIX |
11:55.25 | *** join/#asterisk myiagy (n=myiagy@mail.voffice.com.br) |
11:56.49 | Greek-Boy | so whats the alternative to horid analogue cards? |
11:58.05 | tzanger | analogue cards aren'tbad |
11:58.21 | tzanger | if you're going high density though take alook at direct digital (i.e. CAS T1 or PRI) |
11:58.49 | mut | man we used to be 'lax' in our adding late fees to customers |
11:58.57 | mut | ever since we started doing it, it's like a cash machine |
11:59.16 | mut | i can't believe how many ppl are past due all the time |
11:59.46 | *** join/#asterisk UlbabraB (n=caplaz@host241-43.pool8172.interbusiness.it) |
11:59.50 | Greek-Boy | what if the premises only has analogue lines available? what good will a digital card be? |
12:00.14 | *** join/#asterisk pigpen2 (n=mark@fw.seamans.cc) |
12:00.24 | coppice | 2 analogue pair == 1 E1/T1 :-) |
12:02.11 | *** join/#asterisk littlejohn (n=little@host146-255.pool8289.interbusiness.it) |
12:03.13 | Greek-Boy | lol |
12:03.17 | Greek-Boy | thats not efficient |
12:11.15 | MrChimpy | i'd sooner go voip or use something else to do analogue->t1 than have buttloads of analogue lines coming into my asterisk box |
12:11.57 | MrChimpy | at least a T1/E1 card is more use later |
12:13.31 | *** join/#asterisk [TK]D-Fender (n=joe@toronto-HSE-ppp4122655.sympatico.ca) |
12:14.22 | *** join/#asterisk apardo (n=apardo@87.217.145.29) |
12:18.35 | tzanger | Greek-Boy: I'd still pull in everything into a good channel bank and go to the Asterisk box over a T1 |
12:18.49 | tzanger | i.e. POTS -> Adit600 FXO -> T1 -> TE110P |
12:19.18 | tzanger | I like using the channel banks because they're a proven technoplogy. Just don't use CarrierAccess Access Bank I or II for FXO, they do not have CPD |
12:19.24 | Greek-Boy | hmmm |
12:19.28 | Greek-Boy | even for a few lines? |
12:19.53 | *** join/#asterisk bjohnson (n=bjohnson@jecinc.tor.istop.com) |
12:20.00 | tzanger | define 'few' |
12:20.01 | *** join/#asterisk bjohnson_ (n=bjohnson@jecinc.tor.istop.com) |
12:20.48 | Greek-Boy | 4 lines |
12:20.55 | tzanger | Greek-Boy: use a TDM400 then |
12:21.08 | tzanger | 4 lines is fine, you don't need to spend a pile on a TDM2400 for 4 lines |
12:21.21 | *** join/#asterisk Ariel_ (n=Ariel@70.46.87.158) |
12:23.44 | [TK]D-Fender | Greek-Boy : How many lines can you forsee upgrading to? |
12:23.51 | Greek-Boy | 16 |
12:24.20 | coppice | some bought a MiG21 for $24,730 on E-Bay. I guess that is a slight discount from the new price :-) |
12:24.43 | *** join/#asterisk alib80 (n=chatzill@196.31.11.194) |
12:24.43 | tzanger | yeah see once you getabove about 8 lines I start recommending a channel bank |
12:25.08 | [TK]D-Fender | Greek-Boy : if you think of 8+ I'd suggest trying to get a partial PRI. |
12:25.19 | alib80 | hi all does anyone know how to get multiple instances of asterisk on different servers to write to the same cdr |
12:25.43 | tzanger | alib80: use a database |
12:25.48 | tzanger | I have 6 asterisk boxes dumping into one DB |
12:25.57 | [TK]D-Fender | coppice : I'm Looking for an AH-64 for "recreational" use, keep an eye out for me, ok? ;) |
12:26.01 | Greek-Boy | ok |
12:26.09 | tzanger | # select count(*) as "Total Calls",host as "Host" from cdr group by host; |
12:26.09 | tzanger | <PROTECTED> |
12:26.09 | tzanger | -------------+---------- |
12:26.09 | tzanger | <PROTECTED> |
12:26.09 | tzanger | <PROTECTED> |
12:26.10 | [TK]D-Fender | Greek-Boy : Where are you located again? |
12:26.11 | tzanger | <PROTECTED> |
12:26.13 | tzanger | <PROTECTED> |
12:26.16 | tzanger | <PROTECTED> |
12:26.18 | tzanger | (5 rows) |
12:26.21 | tzanger | sorry 5 boxes not 6 |
12:26.22 | [TK]D-Fender | SPAM!!!!! |
12:26.29 | alib80 | tzanger: ia m running mysql but2nd instance won't write to it |
12:26.31 | [TK]D-Fender | uNF! |
12:26.35 | *** part/#asterisk pigpen2 (n=mark@fw.seamans.cc) |
12:26.37 | tzanger | [TK]D-Fender: that's flood, not spam :-) |
12:26.49 | [TK]D-Fender | Same shit, different marketing! |
12:26.50 | alib80 | no error messages either |
12:27.00 | coppice | [TK]D-Fender I used to fly a radio controlled model of one :-) |
12:27.01 | tzanger | alib80: I use postgres,and have a trigger on the cdr table that injects the hostname |
12:27.20 | Greek-Boy | another question; If i have 4 analogue lines but I want one number to always be available how would i do that? Ie, one number is for four lines and if first one is busy it goes to second line. if second is busy it goes to third line, etc, etc. |
12:27.31 | Greek-Boy | [TK]D-Fender i'm from east africa :( |
12:27.32 | Greek-Boy | lol |
12:28.18 | alib80 | tzanger: would one be able to do the same with mysql. I must admit my db understanding is quite simple |
12:28.18 | [TK]D-Fender | Greek-Boy : What you are describing is called "line-hunting" and is offered by your telco as a base service across the lines you have with them. |
12:28.19 | *** join/#asterisk [Airwolf] (n=airwolf@83.98.235.219) |
12:28.25 | tzanger | alib80: I don't know, I refuse to support mysql |
12:28.48 | coppice | line-hunting in east africa might have a different meaning :-) |
12:28.57 | tzanger | that's *terrible*, coppice |
12:29.20 | Greek-Boy | [TK]D-Fender i suppose its never done on the pbx side? |
12:29.25 | alib80 | tzanger: does the trigger basically mimic the same server or distinguish each server entry? |
12:29.33 | coppice | it wasn't intended as a joke. have you tried to get lines in places liek that? |
12:29.37 | tzanger | yep |
12:29.42 | tzanger | alib80: it distinguishes the server |
12:29.44 | tzanger | let me show you what I have |
12:29.54 | alib80 | thanks |
12:30.10 | [TK]D-Fender | Greek-Boy : You can't MAKE line 1 fall over to line 2 when busy :) |
12:30.20 | [TK]D-Fender | Greek-Boy : Its a telco feature |
12:30.53 | [TK]D-Fender | Greek-Boy : And another advantage to PRI. Another alternative would be to find a VoIP carrier that services your area. |
12:31.23 | coppice | you are more likely to be able to get MFC/R2 than PRI in east africa |
12:32.46 | shiznatix | are there any SIP fax machines? |
12:33.44 | [TK]D-Fender | coppice : E1 varient? I've seen that acronym before but never followed up on non-North American tech... |
12:35.08 | [TK]D-Fender | coppice : What is the identifying factors & advantages to MFC/R2 over "standard" E1? |
12:35.46 | coppice | what is standard E1? |
12:36.36 | [TK]D-Fender | coppice : Hmmm I suppose that would eb grey... ok, what is the TYPICALY signalling put over E1 (As PRI has become for T1) |
12:37.21 | coppice | shiznatix: while T.38 is designed to talk directly to FAX machines, most machines with an RJ-45 do strange proprietary things |
12:37.50 | coppice | typical signalling over T1 would be a list of about 20 things. same with E1 |
12:38.53 | *** join/#asterisk parag7732 (n=root@de2-b15868.alshamil.net.ae) |
12:38.56 | parag7732 | Linux people don't get a good job...then why they strugle too much |
12:39.39 | coppice | the commonest things over E1 these days are EuroISDN and MFC/R2. In many countries ISDN is almost impossible to get, and you have to use MFC/R2 |
12:39.54 | Greek-Boy | what is MFC/R2? |
12:40.00 | Greek-Boy | analogue line hunting? |
12:40.20 | coppice | one nice thing about E1s is apart from a couple of weird places, like the UK, pretty much everywhere uses the same variant of ISDN. |
12:40.33 | coppice | MFC/R2 is a signaling system for E1s |
12:42.03 | Ariel_ | so why is there so many different signalling. MFR/R2 SS7 etc.. |
12:42.27 | tzanger | that's the great thing about standards... there's just so many to choose from! |
12:42.40 | coppice | why are there so many different OSes, linux, unix, window os/400, etc.? |
12:43.09 | Ariel_ | I know it's just would have been allot easyer if we had only 1 or 2. |
12:43.58 | coppice | well, things change over time. MFC/R2 dates back to the mid 50s, when ISDN was not viable |
12:44.21 | *** join/#asterisk linville (n=linville@nat-pool-rdu.redhat.com) |
12:45.23 | Greek-Boy | so if the telco providers line hunting and I phone out on any of those lines will it show the main number on caller ID? |
12:45.38 | tzanger | ok that's it... I'm registering the german pastebin.. pastenderbin.de |
12:45.38 | [TK]D-Fender | Greek-Boy : THAT is another feature |
12:45.46 | coppice | caller ID varies from telco to telco |
12:46.07 | tzanger | alib80: http://pastebin.ca/52862 that's the trigger and function |
12:46.22 | alib80 | thanks |
12:46.53 | parag7732 | Okk i have one query |
12:47.13 | parag7732 | Our company is running a traditional PBX system |
12:47.34 | parag7732 | they want Voice mail and auto attender facility |
12:47.40 | parag7732 | I suggested to go for asterisk |
12:47.47 | alib80 | tzanger: did u have to setup postgres specifically to allow writing from different servers at the same time or does it do this by default. |
12:47.48 | parag7732 | but they don't want to change |
12:47.50 | [TK]D-Fender | parag7732 : How big is your current setup? |
12:48.06 | tzanger | alib80: they just access the DB as regular users with insert access on the table |
12:48.08 | parag7732 | arround 300 |
12:48.10 | tzanger | nohting fancy at all |
12:48.10 | parag7732 | users |
12:48.28 | tzanger | 300 users is a fairly hefty investment in terms of hardware |
12:48.36 | parag7732 | yaa thats why |
12:48.42 | parag7732 | can i do one thing |
12:48.54 | tzanger | what are you suggesting they replace it with? |
12:48.58 | [TK]D-Fender | parag7732 : You can do it in many case, but transferring calls back out of there can be challenging. You'd want to like them by wat of a T1/E1 trunk |
12:49.05 | alib80 | this is what stumps me is that i get no error msg's but when my sencond server tries to write nothing happens even though i tested it on another db |
12:49.05 | parag7732 | No i suggested that I can integrate it |
12:49.18 | parag7732 | with asterisk |
12:49.18 | tzanger | integration is tricky |
12:49.18 | tzanger | most PBXes have no desire to do so |
12:50.01 | tzanger | I can get calls IN to my norstar very easily (assign a DID for each extension, then call that DID), and calls out are trivial, howeverhaving the norstar treat that call out as an extension call is next to impossible. |
12:50.20 | tzanger | there is a protocol called MCDN which is used to tie together meridian systems but of course it's proprietary |
12:50.29 | tzanger | and q.sig does not work on the norstars in north america |
12:50.54 | tzanger | basically if you can get a PRI connection into your PBX and use Q.Sig you're a great ways ahead ofthe game |
12:51.26 | coppice | 1st law of telecoms: if the telcos didn't spec it, its proprietary :-) |
12:52.56 | *** join/#asterisk pointer (i=pointer@aj.catt.com) |
12:53.13 | parag7732 | in TDM400P with 1 fxo and 1 fxs.....Can i integrate 1 fxo with one incomming no. and out put 1 fxs to traditional pbx |
12:53.28 | pointer | parag7732: yes |
12:53.56 | pointer | parag7732: we do something similar to our pbx, but we use an FXO between * and the definity |
12:55.24 | pointer | parag7732: we're changing our config a bit now though... pstn pri-> * -> analog ports + pri -> definity -> another pri -> pstn |
12:55.46 | *** join/#asterisk MikeJ[Laptop] (n=vircuser@d14-69-8-30.try.wideopenwest.com) |
12:56.37 | pointer | after poking around a bit, I have been unable to find a provider for free 800 termination that doesn't state that they have call time limits, known reliability issues, or service "holes" as it were...does anyone else know of a SIP/IAX provider that does free 800 termination? |
12:57.50 | Hmmhesays | 3 days down, 2 to go |
12:58.15 | Greek-Boy | so with a PRI/E1 the call is digital all the way? |
12:58.20 | Greek-Boy | or still goes over analogue |
12:58.31 | [TK]D-Fender | Greek-Boy : E1 = digital |
12:59.11 | Greek-Boy | and PRI T1 |
12:59.12 | Greek-Boy | ? |
13:00.00 | starlein | the same! |
13:00.57 | pointer | no, 8 extra ports! |
13:01.05 | pointer | 24/32 |
13:01.12 | starlein | yes but digital too |
13:01.16 | pointer | Greek-Boy: they're both digital |
13:01.51 | Greek-Boy | E1 is mostly used in europe, right? |
13:01.58 | pointer | PRI/T1 == 24;E1 = 32 digital channels, but in both cases you have a control channel, so it's actually 23 and 31 lines |
13:02.07 | pointer | yup, E1 -> .eu |
13:02.42 | pointer | they've also got some pretty cool calling features that .us doesn't have, like overlapdial |
13:03.00 | *** join/#asterisk Sonderblade (n=muh@host-213.131.147.169.addr.tdcsong.se) |
13:06.57 | *** join/#asterisk LoRez (i=lorez@freenode/staff/lorez) |
13:07.00 | Hmmhesays | oh I miss my guitar |
13:07.18 | Hmmhesays | i woke up last night and I was playing air guitar |
13:08.40 | MrChimpy | sure you weren't just playing your banjo string? |
13:09.07 | Hmmhesays | no because I was also saying the words to backwater by the meat puppets |
13:09.17 | Katty | hi lads. |
13:09.41 | Hmmhesays | Katty |
13:09.56 | coppice | pointer only 30 voice lines on an ISDN E1 |
13:10.57 | Hmmhesays | Oh hell yeah! check this out http://today.reuters.com/news/articlenews.aspx?type=filmNews&storyid=2006-05-02T083541Z_01_N02260638_RTRIDST_0_FILM-NIGHTMARE-DC.XML |
13:13.00 | coppice | Hmmhesays: Chicken Little could only possibly do good business because of the novelty of 3D. it really really sucked |
13:13.16 | Hmmhesays | I was talking about nightmare before christmas, that was a fantastic movie |
13:13.59 | coppice | I was referring to the article. it says nothing about the potential for 3D audiences, because they tried such as lousy movie |
13:14.26 | *** join/#asterisk SheriF_WorK (n=sherif@212.103.170.135) |
13:14.38 | starlein | exactly there are 30 B channels + 2 D channels |
13:15.07 | coppice | starlein: I wish people wouldn't make this stuff up as they go along. its 30 B + 1 D |
13:15.12 | Hmmhesays | Are there any decent hardware video phones out there? |
13:15.13 | pointer | coppice oh, is it 31 and not 32? |
13:15.16 | starlein | sorry thats true |
13:15.23 | pointer | coppice I stand corrected |
13:15.35 | pointer | that's interesting |
13:15.41 | pointer | why 2 control channels? |
13:16.00 | pointer | I should finish reading before I type |
13:16.22 | coppice | there aren't 2 control channels. there's one time slot used to create framing,; one is signalling; the other 30 are voice |
13:16.45 | pointer | gotcha |
13:16.46 | MikeJ[Laptop] | coppice. or 31 if you are not using isdn |
13:16.52 | MikeJ[Laptop] | ;) |
13:17.07 | pointer | I've only worked with PRI here in the us, personally |
13:17.10 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
13:17.33 | [TK]D-Fender | coppice : Entirely agree on your movie critique |
13:17.37 | coppice | 31 is rarely available. only things like SS7 ever use it, and then only rarely. when the channel isn't used as a D channel it is used to provide 30 CAS channels |
13:17.38 | pointer | and ds3/oc3,12,48 |
13:18.11 | Hmmhesays | has the gxv-3000 come out yet? |
13:18.20 | MikeJ[Laptop] | coppice, yah.. I was thinking of wasim's setup |
13:18.42 | coppice | Disney got out of 2D animation, because their audiences sucked. this is entirely because their stories sucked, and now they are making 3D animation that sucks as badly |
13:18.52 | *** join/#asterisk Aurs (n=Aurs@host-81-191-123-189.bluecom.no) |
13:19.41 | pointer | coppice: do you expect anything good out of disney? |
13:19.55 | pointer | coppice: other than theme parks that kids like, that is |
13:20.07 | coppice | Hayao Miyazaki still gets *huge* audiences for 2D animation. maybe it because he's a story teller |
13:20.20 | Hmmhesays | nightmware before christmas was a good movie |
13:20.28 | Hmmhesays | but it was all because of tim burton |
13:20.43 | coppice | without successful movies the theme parks are doomed. they feed entirely for the movies |
13:20.48 | [TK]D-Fender | Tim Burton = GOLD |
13:21.03 | coppice | Tim Burton == quirky |
13:21.03 | MrChimpy | yeah. until you see corpse bride. |
13:21.17 | coppice | he's made more stinkers than successes |
13:21.23 | [TK]D-Fender | coppice : A sorely need trait in this homogenized world.. |
13:21.25 | Hmmhesays | Him and quinten tarrention should make a movie |
13:21.44 | [TK]D-Fender | Hmmhesays : Hard to picture that... |
13:21.47 | pointer | Hmmhesays: ROTFL |
13:21.50 | coppice | corpse bride isn't too bad. seen charlie and the chocolate factory? :-) |
13:21.54 | Hmmhesays | It would be dark and weird |
13:21.58 | MrChimpy | charlie was ok! |
13:21.59 | pointer | Hmmhesays: QT is my hero |
13:22.12 | Hmmhesays | a vampires with scissor hands |
13:22.14 | [TK]D-Fender | coppice : I loved that one personally, then again, its Depp & Burton together that made his successes |
13:22.26 | coppice | charlie was terrible, compared the gene wilder one |
13:22.46 | Hmmhesays | he would never surpass gene wilder |
13:22.52 | Hmmhesays | no one would |
13:23.07 | coppice | that guy looks sooooo old now |
13:24.01 | MrChimpy | is it normal to see occasional loads of the following on a E1 card (TE411 in this case) |
13:24.07 | MrChimpy | <PROTECTED> |
13:24.07 | MrChimpy | <PROTECTED> |
13:24.07 | MrChimpy | <PROTECTED> |
13:24.09 | MrChimpy | etc etc |
13:24.10 | MrChimpy | ?# |
13:24.11 | coppice | (I have small kids - I only see kids movies there days :-) ) |
13:24.30 | MrChimpy | looks like somethings renegociating. work though... |
13:24.32 | *** join/#asterisk iulius (n=iulius@mail1.technologieshq.com) |
13:24.32 | [TK]D-Fender | MrChimpy : Yeah sometimes telcose do a forced bchan reset on interval/idle |
13:24.38 | *** part/#asterisk pointer (i=pointer@aj.catt.com) |
13:24.52 | MrChimpy | ah, cool. that'd be our switch then :) |
13:25.43 | tzanger | MrChimpy: asterisk is doing that |
13:25.45 | *** join/#asterisk littlejohn (n=little@host146-255.pool8289.interbusiness.it) |
13:25.47 | tzanger | MrChimpy: it's totally normal |
13:26.02 | tzanger | MrChimpy: if you don't want it or if it's causing issues, play with resetinterval in zapata.conf |
13:26.02 | MrChimpy | good, good. I thought something was potentially broken :) |
13:26.25 | MrChimpy | not bothered as long as nothing is borken. thanks tz/tk |
13:26.34 | [TK]D-Fender | MrChimpy : First few times I saw it here scared the shit outta me given the problems we'd encountered earlier |
13:27.19 | *** join/#asterisk mercestes (n=merceste@69.15.174.114) |
13:31.15 | *** join/#asterisk suma (n=suma@cm69.gamma29.maxonline.com.sg) |
13:31.32 | suma | Is there is any IAX Wifi Phone Commercially available in the market? |
13:31.59 | suma | or even firmware upgradable ? |
13:32.57 | tzanger | suma: no. |
13:33.01 | [TK]D-Fender | suma : I don't think I've heard of an IAX2 Wi-Fi phone ANYWHERE.... |
13:33.19 | tzanger | I want a wifi phone with bluetooth |
13:33.24 | tzanger | haven't found one of those yet, either |
13:33.34 | [TK]D-Fender | tzafrir : UTStarcom PPC6700 |
13:33.40 | [TK]D-Fender | tzanger rather... |
13:34.04 | tzanger | [TK]D-Fender: any better than their F1000Gs in terms of feel and flimsiness? |
13:34.13 | [TK]D-Fender | tzafrir : Wifi / CDMA / BT / PPC (WinCE 2003) |
13:34.45 | tzanger | [TK]D-Fender: I don't think those are actually shipping yet either |
13:35.02 | tzanger | and CDMA is *useless* if the carriers don't support it |
13:35.02 | file | the 6700? yes it is |
13:35.24 | tzanger | I don't need a damn smartphone, just a damn wifi phone with BT. no PPC, no fucking camera.. ugh (general rant) |
13:35.32 | bkw__ | haha |
13:36.18 | blitzrage | tzanger: haven't seen you here in a while (was just thinking that last night :)) |
13:37.21 | suma | Will any wifi SIP Phone works behind NAT and the asterisk in public IP ? |
13:37.49 | tzanger | blitzrage: :-) |
13:37.59 | tzanger | suma: depends |
13:38.07 | tzanger | symmetric RTP is a glorious thing |
13:38.33 | suma | without symmetric RTP, i guess |
13:38.51 | suma | i can use IAX without any configuration or port forwarding |
13:39.01 | [TK]D-Fender | tzanger : Yeah, they're shipping, Telus is pimping them to me. |
13:39.14 | tzanger | [TK]D-Fender: telus is, really |
13:39.33 | [TK]D-Fender | tzanger : yup. I was going to get one to eval here |
13:42.11 | *** join/#asterisk pif (n=ldm@zenon.apartia.fr) |
13:44.19 | blitzrage | [TK]D-Fender: send me one too :) |
13:45.52 | [TK]D-Fender | blitzrage : ! ! ! |
13:46.07 | blitzrage | heheheheheheheheheheheheehehe |
13:46.34 | [TK]D-Fender | blitzrage : Sorry, I'm putting them off for now since Bell may be able to offer me the same and we're with them already and there is a factor in having to change salesmen's phone #'s. This project is "on hold" for a while/ |
13:47.18 | file | yeah... I wish we had number porting |
13:48.50 | tzanger | yeah |
13:49.00 | tzanger | april 2007 is when LNP comes to mobile networks in Canada |
13:49.59 | [TK]D-Fender | Montreal and the surrounding region hits forced 10-digit dialing next month.... |
13:50.01 | blitzrage | w00t |
13:50.13 | blitzrage | [TK]D-Fender: welcome to Toronto :) |
13:50.34 | [TK]D-Fender | Following all of Toronto's biggest mistakes including forced mega-mergers |
13:50.38 | *** join/#asterisk BadPacket (n=root@unaffiliated/badpacket) |
13:51.25 | [TK]D-Fender | I should move to BC.... little snow, and still far enough north :) |
13:51.46 | [TK]D-Fender | Unfortunately I still love this place for other reasons... |
13:53.38 | file | it's the chinese food |
13:53.41 | *** join/#asterisk froguz (n=alvaro@pc-95-155-104-200.cm.vtr.net) |
13:53.45 | file | it's poisoned your mind |
13:53.52 | *** join/#asterisk gr0mit (n=w10277@dhcp4.zuk40.mot-tools.co.uk) |
13:55.33 | [TK]D-Fender | file : Then again, BC has a huge asian population and is better for imports also being closer to CA.... |
13:56.37 | *** join/#asterisk anthm (n=anthmct@CPE-69-76-83-52.wi.res.rr.com) |
13:56.37 | *** mode/#asterisk [+o anthm] by ChanServ |
13:57.22 | vader-- | have any of you guys had to provision a bulk of phones at one time? |
13:57.40 | vader-- | i was looking to program a script to write the conf files and stuff but im a vb programmer not perl |
13:57.52 | vader-- | wondering if there is any already writen utilities |
13:58.56 | tzanger | vader--: polycom makes it nice |
14:00.23 | *** part/#asterisk kmilitzer (n=km@office-gw.westend.com) |
14:01.18 | [TK]D-Fender | vader-- : Easily scriptable. Make a template and parse and extensions list and you can generate sip.conf & the provisioning files for them based on MAC |
14:02.13 | [TK]D-Fender | vader-- : What kind if "bulk" are we talking about anyways? |
14:04.35 | Hmmhesays | snap into a slim jim |
14:05.31 | *** join/#asterisk Lino` (n=Lino@i577BC236.versanet.de) |
14:06.26 | pif | does a jitterbuffer make a big quality difference with iax? |
14:07.18 | Hmmhesays | if you have a lot of jitter it sure does |
14:07.48 | coppice | and if you have packet loss, PLC makes a big difference too |
14:08.25 | pif | "jitter" meaning variations in latency ? |
14:09.21 | coppice | well, if doesn't mean a caller with a stutter |
14:09.41 | Katty | unless it's natural stutter. |
14:09.47 | Katty | in which case, they just need a speech therapist. |
14:10.08 | Ahrimanes | w-w-w-w-w-ha-a-t-s t-t-t-tha-t-t ? |
14:10.23 | *** join/#asterisk mhnoyes (n=mhnoyes@dialup-4.246.6.63.Dial1.SanJose1.Level3.net) |
14:10.26 | Hmmhesays | great a huge thorn in my side is back |
14:10.37 | *** part/#asterisk mhnoyes (n=mhnoyes@dialup-4.246.6.63.Dial1.SanJose1.Level3.net) |
14:10.40 | pif | hi coppice, and what is the tradeoff? increased latency? |
14:12.13 | coppice | well, a really good jitterbuffer balances packet loss and latency to maximise perceived quality. the one in *'s IAX is not as good as it could be. however, if you have jitter it is still a huge improvement |
14:12.52 | *** join/#asterisk brodiem (i=1000@cpe-66-69-222-36.austin.res.rr.com) |
14:13.25 | *** part/#asterisk parag7732 (n=root@de2-b15868.alshamil.net.ae) |
14:13.39 | pif | for instance a DSL connection going from ping 12ms to 100ms all the time, would be a good candidate? |
14:13.54 | coppice | sounds like it |
14:14.15 | pif | thanks, and google says you wrote the PLC code |
14:14.28 | pif | is that included in the jitterbuffer=yes ? |
14:14.43 | coppice | yes |
14:14.54 | pif | thanks thanks thanks |
14:16.15 | Katty | i could use a new car. |
14:16.20 | sevard | <PROTECTED> |
14:16.40 | Ahrimanes | i've had a 1001 that did it |
14:16.51 | sevard | what the hell is up with that |
14:16.58 | Katty | sevard: it needs a hug. |
14:17.04 | sevard | i'm just sitting here and it scares the crap out of me by giving me one ring |
14:17.09 | Katty | sevard: possibly a firmware update. |
14:17.09 | MrChimpy | dunno, but it's a good way to pretend to be popular |
14:17.13 | sevard | Katty: i've tried hugging/raping it... doesn't work. |
14:17.25 | Katty | sevard: your hugs must suck then. |
14:17.41 | sevard | they can't, i won a contest. |
14:17.49 | Katty | for strangling to death? |
14:17.50 | MrChimpy | i'm not sure if a rapey hug counts |
14:18.06 | sevard | Katty: for hugs, i haven't been entered into a raping contest although if you're up for it |
14:18.06 | coppice | isn't sucking while hugging pretty normal? |
14:18.35 | Katty | sevard: you shouldn't joke about things like that. |
14:18.39 | Katty | sevard: you have no idea what people go through |
14:18.43 | sevard | Katty: the world needs jokes. |
14:18.54 | MrChimpy | why not? 8 mile was a whole movie about raping contests |
14:19.00 | MrChimpy | :p |
14:19.01 | sevard | :) |
14:19.05 | Katty | sevard: but it's not a joke if it isn't funny. |
14:19.19 | Katty | sevard: if i were you, i'd never joke about raping a female. |
14:19.20 | sevard | Katty: only to you dear, I found it funny. |
14:19.29 | Katty | it's just not polite. |
14:20.00 | sevard | Katty: I might have a hissy fit about irish catholic jokes but whenever somebody lays one down it gets me in stiches |
14:20.19 | Katty | we're not all you, sevard |
14:20.22 | MrChimpy | and if it's ok to joke about male rape, well, that's just SEXIST AND WRONG |
14:20.28 | sevard | how is that different than rape or black jokes? it's not. |
14:20.41 | sevard | Katty: you can't please them all honey. |
14:20.45 | Katty | why are we arguging about this? |
14:20.49 | Katty | we have better things to argue about |
14:21.07 | sevard | there's always somebody somewhere who is going to be offended by something. it's unavoidable. |
14:21.13 | sevard | Katty: not sure why you brought it up then. |
14:21.20 | MrChimpy | sve: how dare you say that! |
14:21.34 | Katty | sevard: let's just move on. |
14:21.35 | Nivex | sevard: yeah, but there are some doozies that are pretty well known that can generally be avoided. |
14:21.38 | Katty | sevard: if you want the last word, then have it. |
14:21.43 | sevard | Heh |
14:21.59 | *** join/#asterisk chapeaurouge (n=chap@vilhost1.vision.lu) |
14:22.03 | sevard | Last words: cooooooooooooooooooooooooooooooooookie chrisp! |
14:23.06 | Katty | right. |
14:23.12 | Katty | now, i'd like to know something. |
14:23.17 | *** join/#asterisk psk (n=psk@golia.caltanet.it) |
14:23.27 | Katty | why do lamborghinis always break down after a month if you're not driving them over 70mph all the time? |
14:23.40 | Katty | there's something terribly wrong with this. |
14:23.47 | sevard | I'm not sure if that's true |
14:23.47 | MrChimpy | because they're made by italians |
14:23.56 | MrChimpy | they go vroom, or not at all |
14:24.05 | Katty | sevard: it's true all right. |
14:24.05 | MrChimpy | it's the same with ducattis |
14:24.12 | Katty | and they lag on the shirt |
14:24.14 | Katty | i mean shift |
14:24.31 | syzygybsd | Katty: it sounds like there is an easy fix for that problem |
14:24.34 | Katty | torque converter, i think. |
14:24.37 | coppice | its better than BMWs. they break down every month whatever you do with them |
14:24.51 | sevard | I think sports cars run just fine at normal speeds people just want to drive them fast. |
14:24.52 | Katty | syzygybsd: yeah, get soemthing else. |
14:25.04 | syzygybsd | I was thinking drive them faster |
14:25.09 | Katty | not that i have a lamborghini. |
14:25.18 | coppice | wankel engines only work properly when driven fast |
14:25.34 | Katty | actually, i'm trying to decide what my next car will be. |
14:26.05 | Katty | a coupe of some sort. |
14:26.05 | sevard | Katty: get a cadillac 16 |
14:26.09 | coppice | and you've discounted the lambourgini, due to reliability? |
14:26.10 | Katty | hell no |
14:26.17 | Katty | coppice: that and price |
14:26.23 | sevard | the Cadillac Sixteen is the best car ever made |
14:26.36 | Katty | i disagree. |
14:26.44 | sevard | Have you ever driven in one? |
14:26.45 | coppice | GM and good don't seem like a matching combination |
14:26.51 | Katty | there are too many factors to say Best Car |
14:26.53 | Katty | best car for what? |
14:26.54 | MrChimpy | forget the ticket price. then look at the service costs. then say goodbye to comfortable retirement. |
14:27.01 | sevard | Katty: it's simplistic math |
14:27.04 | sevard | it _is_ the best car |
14:27.06 | Katty | coppice: my dad works for GM/chevrolet |
14:27.16 | Katty | coppice: they seem to be ok vehicles. |
14:27.22 | Katty | coppice: tho, i'm kinda partial to pontiac myself. |
14:27.25 | Hmmhesays | anyway ever heard of "virtual line"? |
14:27.34 | sevard | s/way/one |
14:27.41 | sevard | oh, you can't sed eachother's lines. |
14:27.42 | MrChimpy | they sell some chevrolet thing over here. it looks bloody horrible |
14:27.48 | coppice | the european GMs really are the worst crap made in europe |
14:27.52 | *** join/#asterisk [Airwolf] (n=airwolf@83.98.235.219) |
14:28.07 | Katty | coppice: european cars don't lag as much on the gear shift. |
14:28.11 | coppice | and some of the US GMs are like death traps |
14:28.22 | Katty | i think i'm going to stick with pontiac |
14:28.24 | MrChimpy | oh, maybe not |
14:28.40 | sevard | i had a pontiac for a couple years, i hated it. |
14:28.44 | Katty | coppice: i've had a cavalier (first car), a sunfire, and a trans am. |
14:28.46 | MrChimpy | the chevys on .co.uk look like everything else |
14:29.01 | MrChimpy | chevys are sold as vauxhall over here, or were. |
14:29.11 | MrChimpy | now they sell under chevy brand |
14:29.15 | MrChimpy | still look dull |
14:29.18 | Katty | coppice: this G6 coupe is starting to look real nice |
14:29.20 | [TK]D-Fender | Katty : I'm just waiting to ditch my 1987 Camaro and get something about a decade newer.... |
14:29.58 | coppice | the nastiest thing I ever drove was a Chevy. Can't remember the name, but it had plenty of power to go, and no brakes to stop. |
14:30.03 | Katty | [TK]D-Fender: that's a cute little car i bet |
14:30.31 | Katty | [TK]D-Fender: they kinda remind me of corvettes for some reason |
14:30.59 | chapeaurouge | how can i start * to have more debug messages in /var/log/asterisk/messages ? |
14:31.09 | Katty | curvy little bubble corvette |
14:31.56 | *** join/#asterisk ibob63 (n=hp@bb-87-82-7-89.ukonline.co.uk) |
14:32.07 | sevard | I'm telling you man, the Cadillac Sixteen |
14:32.07 | vader-- | do you guys know off the top of your heads what color the FXS modules are suppose to be on the TDM2400P cards? |
14:32.28 | sevard | FXO = red FSX= green |
14:32.35 | Katty | sevard: they're not pretty! |
14:32.35 | [TK]D-Fender | Katty : I'm looking at getting likely either a Cavalier or Sunfire, 1995-1997 |
14:32.44 | vader-- | thanks |
14:32.45 | sevard | s/FSX/FXS/g |
14:32.46 | Hmmhesays | camaro what |
14:32.46 | Katty | [TK]D-Fender: they're reliable little things, i'll say that. |
14:32.58 | Katty | [TK]D-Fender: i think my cavalier handled a little better than the sunfire. |
14:33.07 | ibob63 | Occasionally the my office asterisk server mysteriously can't register with the gateway. Is there a way I can get asterisk to email me when it fails to register with the gateway? |
14:33.08 | [TK]D-Fender | Katty : Maybe more where you are... lets say that Quebec winter is remarkably UNKIND to cars.... |
14:33.09 | Katty | [TK]D-Fender: but that's probably cause the cavalier was my first. and you always remember your first |
14:33.16 | Katty | [TK]D-Fender: yeah. |
14:33.25 | Katty | [TK]D-Fender: if it was snowing, that trans am wasn't going /anywhere/ |
14:33.37 | Ahrimanes | ibob63: monitor the log files and have a script email you when it's logged? |
14:33.39 | Katty | [TK]D-Fender: which is why i'm back to driving a sunfire again |
14:33.47 | Katty | [TK]D-Fender: i loved that trans am... |
14:33.51 | coppice | i've just looked up the cadillac 16. I see a lot of those in china. probably the world's ugliest car (which is saying something, considering the current BMWs) |
14:33.55 | [TK]D-Fender | Katty : I'm going to miss my 0 degree turn radius :D |
14:34.04 | Katty | [TK]D-Fender: i miss driving a manual. |
14:34.09 | Katty | [TK]D-Fender: and i miss the purr :< |
14:34.14 | sevard | coppice: 16 cyl man. best car. |
14:34.22 | [TK]D-Fender | coppice : Doubt it beats Renaut, Hugo, or Lada for that... |
14:34.28 | *** join/#asterisk kristalino (n=kristali@LNeuilly-152-21-126-73.w193-253.abo.wanadoo.fr) |
14:34.32 | Katty | [TK]D-Fender: *shift*, *exhaust*, *swoon* |
14:34.34 | [TK]D-Fender | I'm regretably stuck on automatics. |
14:34.41 | ibob63 | Ahrimanes: so there isn't a built in function for emailing when an error is logged? |
14:34.46 | Katty | [TK]D-Fender: don't take corners at 60mph, your ears will pop :P |
14:35.16 | *** join/#asterisk apardo (n=apardo@87.217.145.29) |
14:35.17 | Ahrimanes | ibob63: no, really isnt a job for asterisk.. more a job for a monitoring system |
14:35.21 | coppice | a lada looks elegant compared to that cadillac |
14:36.17 | coppice | though whole car industry seems to be going through a butt ugly phase |
14:36.23 | Katty | yeah, it is. |
14:36.28 | Katty | especially the suvs |
14:37.06 | [TK]D-Fender | Katty : Yeah, the Honda Element is FUGLY. |
14:37.15 | [TK]D-Fender | Like a brick with wheels... |
14:37.17 | coppice | dunno. the only reasonable looking thing BMW make now is their SUV, and the porsche one isn't too bad |
14:37.31 | [TK]D-Fender | And the ever gay Aztec... |
14:37.41 | sevard | honda element is one ugly POS but not quite as ugly as the PT Cruiser |
14:37.44 | Hmmhesays | I miss driving |
14:37.45 | Katty | the pontiac solistice......i dunno what they were thinking. |
14:37.58 | Hmmhesays | jamie, I wish you a happy life with your fiance |
14:38.01 | Hmmhesays | bwhahaha |
14:38.11 | Katty | it looks like a very seriously deformed coupe of some sort. |
14:39.00 | Katty | sevard: the pt curisers remind me of old lady cars. |
14:39.13 | Katty | sevard: i saw one with a little old lady and wood paneling on it ;) |
14:39.20 | Katty | sevard: after that, i was ruined. |
14:39.35 | sevard | I used to work with this jesus freak who had a gold PT cruiser |
14:39.42 | sevard | apparently it made him closer to god |
14:40.08 | Hmmhesays | and I always that f@#$#@ you like an animal brought you closer to god |
14:40.31 | [TK]D-Fender | Katty : Even worse was Chevy's attempt to market a 3-speed automatic veriosn of the Nova to South America. Keeping in mind that in Spanish "Nova" means "Doesn't Go" :D |
14:40.37 | sevard | Hmmhesays: combine that with a heavy dose of acid |
14:41.24 | Katty | [TK]D-Fender: 3 speed automatic? |
14:41.33 | Katty | [TK]D-Fender: 1,2, and ohgodoverload? |
14:41.35 | [TK]D-Fender | Katty : Special transmission... |
14:41.46 | Katty | 3 speed automatic is confusing |
14:41.52 | Katty | i can't even picture that |
14:41.54 | [TK]D-Fender | Hmmhesays : Occasionally I like NIN too... |
14:42.01 | Katty | 1,2 and reverse..... |
14:42.04 | Katty | 1,2 and 'splode |
14:42.05 | coppice | all automatics were 3 speed just a few years ago |
14:42.14 | Hmmhesays | a local band here used to a do a kickass rock cover of "head like a hole" |
14:42.17 | Katty | coppice: i've never /seen/ a 3 speed automatic before |
14:42.28 | Katty | coppice: how many is a 'few years' |
14:42.33 | sevard | 40ish |
14:42.54 | *** join/#asterisk MacDome (n=eseidel@c-71-198-177-144.hsd1.ca.comcast.net) |
14:43.24 | coppice | 20 |
14:43.42 | coppice | in the mid 80s only the high end had more than 3 speeds |
14:44.15 | sevard | coppice: 20 years != few years |
14:44.27 | *** join/#asterisk MacDome (n=eseidel@c-71-198-177-144.hsd1.ca.comcast.net) |
14:44.31 | coppice | you must be rather young |
14:45.09 | coppice | the last time I bought a car was 1985. that might colour my thinking |
14:45.14 | Katty | coppice: i'm only one year over 20 :P |
14:45.44 | coppice | I haven't owned a car for 15 years |
14:46.33 | tzanger | inconcievable! |
14:47.04 | Nugget | I own a car but I rarely drive it. |
14:47.27 | Nugget | I've put less than 5,000 miles on it in the past year. |
14:47.46 | tzanger | wow..> I typically put about 20-25k on mine a year |
14:47.49 | tzanger | (km not miles) |
14:47.55 | BadPacket | has anyone seen JerJer? |
14:48.01 | tzanger | I don't consider thata lot of driving though |
14:48.17 | Nugget | My mileage was 23,619 on 6-Apr-2005 and 27,514 on 3-Apr-2006. |
14:48.26 | *** join/#asterisk [hC] (n=hardcore@mail.metrobridge.com) |
14:48.41 | Nugget | dunno what it is today, but it can't be more than 27,800 since I haven't filled it up since 3 Apr. |
14:48.52 | Katty | Nugget: what do you drive? |
14:49.00 | [TK]D-Fender | I do around 8-10k on mine... |
14:49.05 | Nugget | a bmw m roadster. |
14:49.09 | Katty | ah, k |
14:49.12 | Katty | don't want a bmw :) |
14:49.16 | Nugget | http://slacker.com/~nugget/mroadster/ |
14:49.23 | Nugget | (car stats. I love stats) |
14:49.33 | [TK]D-Fender | Break My Window? Bimbette Motor Weapon? Bus Metro Walk? |
14:49.42 | Nugget | broke my wallet. |
14:50.03 | sevard | buggered my mom |
14:50.14 | *** join/#asterisk jhava (n=icechat5@200.58.26.21) |
14:50.18 | [hC] | This might sound weird, but has anyone experienced a problem on a PRI where 9 times out of 10, you can place a call, but occasionally you get channel unavailable for a number that you can normally dial (and since i dial it via another provider right away,and it goes thru, theres nothing wrong with the number itself)? Is it possible that theres something wrong on my end, or is this likely my provider's fault? |
14:52.33 | Hmmhesays | what was the website where you can share a whiteboard |
14:53.46 | noname32 | netmeeting |
14:54.01 | *** join/#asterisk klasstek (n=nunyobiz@ip67-154-143-190.z143-154-67.customer.algx.net) |
14:55.19 | jhava | hello, has anyone configured two step dialing when connecting Asterisk to an ISDN PABX (PABX dialing to Asterisk)? |
14:55.43 | sevard | hahah |
14:56.15 | sevard | Hmmhesays: was it a website recently bought by google? |
14:56.43 | Hmmhesays | aren't they all? |
14:57.01 | sevard | I forgot Internets == GOOG |
14:57.02 | syzygybsd | can someone direct me somewhere to learn about call routing by callerid? |
14:57.16 | Hmmhesays | I put up an example on the digium forums |
14:57.17 | coppice | if google buys something released, does it immediately go back to being beta? :-) |
14:57.28 | MrChimpy | cop: heheh |
14:57.29 | mut | yes |
14:57.39 | syzygybsd | thanks |
14:57.47 | Hmmhesays | http://forums.digium.com/viewtopic.php?t=6171&highlight= |
14:58.17 | *** join/#asterisk gmaruzz (n=gmaruzz@217-133-80-112.b2b.tiscali.it) |
14:58.24 | Hmmhesays | should be pretty straightforward |
14:59.19 | sevard | Hmmhesays: www.writeboard.com ? |
14:59.53 | [TK]D-Fender | syzygybsd : Read up on "gotoif" and "asterisk functions" on the WIKI. Everything you should need is in those 2 references |
15:00.20 | syzygybsd | [TK]D-Fender: thanks now that you mention that function it all makes sense... |
15:00.31 | Hmmhesays | syzygybsd: did you even click that link? |
15:00.45 | Hmmhesays | [TK]D-Fender hogging all the credit |
15:00.46 | sevard | Hmmhesays: the one google acquired was Writely |
15:00.51 | syzygybsd | oh.. not yet, was off looking for what you said |
15:00.57 | Hmmhesays | http://forums.digium.com/viewtopic.php?t=6171&highlight= |
15:01.00 | Hmmhesays | haha |
15:01.07 | Hmmhesays | google ownz joo |
15:01.07 | syzygybsd | thanks |
15:01.14 | syzygybsd | jah |
15:01.34 | file | sweet |
15:01.38 | sevard | I forgot google bought sketchup, i haven't tried it yet |
15:01.40 | *** join/#asterisk gursikh (n=gursikh@158.135.0.125) |
15:01.40 | noname32 | hey guys is there any good howtos for upgrading asterisk on a centos box? |
15:01.46 | file | high speed data service is deployed here :D |
15:01.49 | file | from Rogers |
15:01.53 | Hmmhesays | same way you would do it on a none centos box |
15:02.00 | MrChimpy | 1) back up |
15:02.02 | MrChimpy | 2) build |
15:02.08 | *** join/#asterisk SplasPood (n=jwb@206.252.198.101) |
15:02.08 | Hmmhesays | 3) sammichs |
15:02.12 | MrChimpy | 3) sacrifice chicken |
15:02.15 | sevard | 3) profit! |
15:02.28 | noname32 | lol i dont have a chicken will a dog do? |
15:02.29 | MrChimpy | 4) ? |
15:02.30 | Hmmhesays | no 2) is always ??? if 3 is profit |
15:02.37 | sevard | right, you did 2 wrong |
15:02.44 | syzygybsd | I am going to build it anyway, but is there a script that will call a number, and return whether the call was sucessfull or not (read answered, possible digits returned) |
15:02.50 | sevard | 1) download new copy |
15:02.52 | sevard | 2) ???? |
15:02.54 | sevard | 3) PROFIT! |
15:03.28 | sevard | syzygybsd: what you're thinking of is a 'phone' |
15:03.44 | syzygybsd | how many people know where that is from? It has been used so much I don't think lots of people do |
15:03.59 | sevard | syzygybsd: the underpants gnomes in a south park episode |
15:04.12 | MrChimpy | sev: seen the new ones? |
15:04.14 | syzygybsd | sevard: I want an automated job so it can alert someone if there is a problem |
15:04.27 | MrChimpy | oprah winfrey's lower regions going psychotic? |
15:04.27 | sevard | MrChimpy: I haven't had cable television in over a year so I'm a bit behind ;/ |
15:04.28 | syzygybsd | new one tomorrow right? |
15:04.38 | syzygybsd | oprah's minge |
15:04.45 | MrChimpy | sev: but you have t'internet. www.mrtwig.net |
15:04.53 | [TK]D-Fender | sevard : Been 2 for me, and thats immeditaely following a purchase of a 52" HDTV :D |
15:05.02 | Sonderblade | anyone know from where i can get a backport of asterisk 1.2.6 or greater for debian sarge? |
15:05.22 | codebreaker | Sonderblade: which one is at bpo? |
15:05.23 | syzygybsd | Sonderblade: just download adn compile the source |
15:05.24 | sevard | MrChimpy: I know. I don't have any stable HDDs at the moment. I needs moola to get disks so I can keep content. |
15:05.45 | codebreaker | okay 1.2.1 |
15:05.48 | syzygybsd | that is what I am running on sarge |
15:06.03 | MrChimpy | wow. you must be poor. i'd throw you a dime if my throwing arm could manage the atlantic :) |
15:06.25 | *** join/#asterisk nahirean (n=nahirean@unaffiliated/nahirean) |
15:06.26 | sevard | MrChimpy: poor atm :) i havn't found any nice 10 gbs in the trash yet |
15:06.28 | MrChimpy | i keep choking at how cheap HDs are :) |
15:06.46 | Sonderblade | codebreaker: 1:1.2.1.dfsg-2bpo1 which is to old i think |
15:06.49 | gursikh | I love how cheap they have gotten, and hope they go cheaper |
15:06.52 | MrChimpy | i have a bunch of 40s and stuff doing nowt. |
15:06.56 | syzygybsd | heh, I remember having a 10MB hd when I was younger |
15:06.59 | sevard | it's okay being poor though, you wait till you have money then what you wanted back then is 100x cheaper now |
15:07.04 | syzygybsd | then before that a computer without a HD |
15:07.12 | *** join/#asterisk SuperLag (n=aaron@gentoo/developer/SuperLag) |
15:07.14 | codebreaker | Sonderblade: it depends on waht features you need |
15:07.17 | sevard | syzygybsd: I paid $150 for 10 gb back in the day |
15:07.26 | Sonderblade | syzygybsd: cool, it would be nice of you to upload it to the backports site |
15:07.30 | syzygybsd | sevard: that wasn't that long ago |
15:07.41 | sevard | about 9-10 years ago |
15:07.47 | Sonderblade | codebreaker: i need >= 1.2.6 |
15:07.52 | MrChimpy | yes, we all had big HDs with tiny capacity. 10meg full height SASI. etc etc :) |
15:07.59 | sevard | hahaha |
15:08.15 | *** join/#asterisk wunderkin (n=kev@mmds-216-19-40-108.mm.az.commspeed.net) |
15:08.16 | sevard | Back when I was your age we had these huge fucking cabnits that stored a character a piece |
15:08.28 | MrChimpy | ST506, now that was a interface |
15:08.35 | syzygybsd | Sonderblade: it is just the source... no packages |
15:08.39 | sevard | it only took 3 square blocks of network equiptments to map out a paragraph! |
15:08.50 | coppice | SMD. now that was an interface |
15:08.53 | sevard | that was back when computars were fast |
15:09.00 | sevard | computers* |
15:09.13 | coppice | 1kW per drive. worse than a pentium 4 :-) |
15:09.19 | MrChimpy | pppft. computers are MODERN |
15:09.25 | Sonderblade | syzygybsd: oh, for my purpouses i need to have a sarge compatible binary package |
15:09.29 | MrChimpy | I used a DIFFERENCE ENGINE and LIKED IT. |
15:09.56 | codebreaker | Sonderblade: backport it yourselve? |
15:10.21 | Sonderblade | codebreaker: am trying but its hard |
15:10.47 | [TK]D-Fender | Sonderblade : Why binary? |
15:10.56 | [TK]D-Fender | Sonderblade : Use the Source Luke! |
15:11.10 | codebreaker | Sonderblade: now its normaly not hard. fetch source do uupdate and dpkg-buildpackage |
15:11.27 | sevard | I have an uncle who sells electronics, one of his clients were buying all this old shit, tubes from the 70's, switches from the 60's |
15:11.51 | sevard | he got curious and made a visit to the client, found out it was a bank under another name with 40 year old machines |
15:11.56 | *** join/#asterisk MRH2 (n=Mr_happy@fcirc-adsl.demon.co.uk) |
15:12.07 | codebreaker | Sonderblade: i have no time now. but if you can wait a day or two. ask me again |
15:12.51 | MRH2 | hi can someone point me in the direction of what the "new sip tranfer code" is all about? |
15:13.22 | pif | what is the default value when using jitterbuffer=yes ? |
15:13.22 | *** join/#asterisk Zeeek (n=IceChat7@pdpc/supporter/active/Zeeek) |
15:14.25 | *** join/#asterisk lzhang (n=rjrae@67.95.13.46) |
15:14.25 | MRH2 | can't u save to 1 db and replicate? |
15:15.35 | Sonderblade | [TK]D-Fender: to be able to distribute it |
15:16.08 | *** join/#asterisk adker (n=adker@67-136-210-63.dsl1.glv.ny.frontiernet.net) |
15:16.43 | *** join/#asterisk inv_arp[work] (i=junya@c-67-191-62-53.hsd1.fl.comcast.net) |
15:18.42 | *** join/#asterisk zaf (n=zaf@wsip-68-228-9-79.br.br.cox.net) |
15:18.49 | codebreaker | MRH2: yes/no i think about storing to one host and then replicating to many(more then 2) and then accessing the mbox directly from there. but this is know to be sometimes a little bit tricky when doing deletes |
15:18.50 | [TK]D-Fender | Sonderblade : If there is already a binary package, wouldn't it ALREADY be in distribution? |
15:19.07 | chapeaurouge | anyone using Junghanns QuadBRI card? |
15:19.28 | codebreaker | MRH2: and one db is a SPOF |
15:20.41 | stoffell | yes chapeaurouge |
15:20.48 | MRH2 | i mean replication and the ''traditional' failover type stuff |
15:21.25 | chapeaurouge | stoffell, im gonna paste my config on some site... will give you a link... if you could take a look quickly and tell me if you see anything wrong, it'd be great |
15:21.40 | *** join/#asterisk Cresl1n (n=matt@gateway.digium.com) |
15:21.42 | stoffell | it's not working then |
15:21.43 | stoffell | ? |
15:22.28 | chapeaurouge | i guess it is. i need to plug some isdn lines in :) but i want to see if everything is normal so far. i cant unplug the isdn lines in the middle of the day :) |
15:23.20 | MRH2 | anyone know what new stuff the the new sip transfer code does? (/team/oej/siptransfer) |
15:23.25 | Zeeek | some can... replicant |
15:23.53 | oej | MRH2: No, haven't got a clue :-) |
15:24.02 | oej | Sorry |
15:24.03 | MRH2 | lol |
15:24.12 | oej | It implements REFER in a better way. |
15:24.14 | Ahrimanes | hehe |
15:24.22 | oej | It's gradually moving into SVN trunk |
15:24.27 | chapeaurouge | stoffell, http://pastebin.ca/52888 |
15:24.36 | Zeeek | oej you should change the method to REEFER |
15:24.39 | oej | Will continue as soon as all my branches are back |
15:24.44 | oej | zeek: :-) |
15:24.47 | Cresl1n | oej!!!!!! |
15:24.53 | oej | Cresl1n: Hi!!! |
15:25.00 | Zeeek | oej Paris soon? |
15:25.16 | chapeaurouge | stoffell, note i dont know if i need the span stuff in zaptel.conf... kinda copied-pasted ;) |
15:25.23 | oej | zeeek: Paris was cancelled |
15:25.29 | Zeeek | ah. sorry |
15:25.31 | oej | zeeek: Brussels and London this week |
15:25.38 | oej | Yes, I would have loved going to Paris now |
15:25.52 | Zeeek | well, break a leg as we say. A call leg obviously in this case! |
15:26.06 | MrChimpy | london/paris as in astricon? |
15:26.09 | coppice | paris often gets cancelled. they keep closing down the country |
15:26.10 | MRH2 | thanx |
15:26.11 | Zeeek | Weather finally got decent and I was all set to buy you a glass of fine wine |
15:26.13 | *** join/#asterisk spatulamaan (n=ggilmore@ip66-107-33-196.z33-107-66.customer.algx.net) |
15:26.39 | oej | MrChimpy: No, meetasterisk.com. I am no part of Astricon and won't be there |
15:26.41 | Zeeek | coppice nah, that's just video we put out to the press to keep the tourists away |
15:26.56 | MrChimpy | paris is a craphole. I went to biarritz in south of france last week. much better. gorgeous girls. |
15:27.11 | Zeeek | biarritz is really great |
15:27.12 | coppice | the only time I tried to go to paris the whole bloody country shut down |
15:27.17 | MRH2 | is it to make it more compatible or are there changes from the POV of us end users? |
15:27.18 | Hmmhesays | paris almost got destroyed in the last book i read |
15:27.23 | Zeeek | they knew of you in advance coppice |
15:27.37 | MrChimpy | so is astricon some unholy thing? |
15:27.45 | Ahrimanes | no astricon is good |
15:27.52 | Ahrimanes | well was last year anyways |
15:27.57 | MrChimpy | goodo. i shall attend the london one |
15:28.23 | oej | This year's Astricon has nothing todo with previous Astricons |
15:28.23 | MrChimpy | I was trying to blag the stockholm one, then kind of undermined myself by implementing what we needed from the book :) |
15:28.31 | *** join/#asterisk kristalino (n=kristali@LNeuilly-152-21-126-73.w193-253.abo.wanadoo.fr) |
15:29.20 | MrChimpy | not got much reason to go as yet, other than skiving for a day. everything i've done in asterisk has just "worked". |
15:29.27 | *** join/#asterisk DoktorGreg (n=Greg@70.91.121.89) |
15:29.38 | MrChimpy | past a bit of trickiness getting the E1s talking |
15:29.58 | Ahrimanes | oej: oh? |
15:30.29 | *** join/#asterisk angler (n=angler@pdpc/sponsor/digium/angler) |
15:30.31 | stoffell | chapeaurouge, looks good |
15:31.07 | Zeeek | Ahrimanes oej ins't doing this one. Just for info |
15:31.22 | Ahrimanes | Zeeek: ok |
15:31.26 | Zeeek | MrChimpy yes, but there's the beer and girls, not nec. in that order |
15:31.42 | Zeeek | and if your thing is boys, there's even more of that :) |
15:31.48 | MrChimpy | pppfftft. |
15:31.48 | *** join/#asterisk salviadud (n=ralfalfa@dsl-200-78-64-10.prod-infinitum.com.mx) |
15:32.01 | Zeeek | meeting other geeks in meat space is a trip! |
15:32.12 | salviadud | meat space? |
15:32.16 | MrChimpy | i'm sure it'll be chock full of sexy women. just like linux today. |
15:32.21 | Zeeek | hahahaha |
15:32.23 | vader-- | do you guys know the difference between the s400M revision B and revision B2? |
15:32.32 | Zeeek | well, there were a few handing out t-shirts |
15:33.00 | MrChimpy | i spent about 30 mins in that show. it started getting depressing very quickly :) |
15:33.39 | rpm | is there something im missing if i connect to port 5038 (astmanager) and type 'Action: SIPpeers' i want to get a list of SIPpeers |
15:33.59 | MrChimpy | best trade show I ever went to was CabSat. there were porno channels trying to sell content with these hookers trotting round in next to nothing, then an irish tv station had a free guinness bar. |
15:35.10 | Zeeek | I went to a linux one in L.A. once years ago and the little LED mouse keychain light they gave out *still* works! |
15:35.33 | MrChimpy | yeah, crap freebies at linux today too. |
15:35.44 | Zeeek | I think it was worth the time and money to meet some of the majors last year (including oej) |
15:36.09 | Zeeek | since Mark is in Paris several time a year we don't need Astricon for that |
15:36.23 | SplasPood | hrm.. I have a context which does a Background() (IVR menu) and lets people either dial a single digit extension from the local context, or a direct extension which is included ... seems sometimes people miss the first digit of the exten .. 2002 and instead dial 002... problem is that gets matched the _0, exten rather than going to invalid... any thoughts? |
15:36.44 | SplasPood | or rather my exten is simply '0' not _0 |
15:37.39 | MrChimpy | yeah, i'm not about to bat around with the big-wigs :) at the moment this is just my job, so unless work want something super thrilling... |
15:37.54 | SplasPood | seems as soon as it gets that first 0 it dials rather than waiting for the timeout... |
15:38.22 | sevard | MrChimpy: DUDE, cabsat sounds awesome |
15:38.26 | MrChimpy | things will probably get hairy when i get some investment in the project and we start scaling up |
15:38.57 | MrChimpy | sev: yeah, it was. but then that was 1998 I think. I was drunk when I got there too thanks to a pub lunch. |
15:39.10 | sevard | that sounds like one awesome show. |
15:39.14 | DoktorGreg | Any digium guys around? |
15:39.47 | noname32 | hey when compiling asterisk-addons do you do that before or after the core ? |
15:40.40 | *** join/#asterisk trelane_ (n=trelane@pdpc/supporter/sustaining/trelane) |
15:40.59 | chapeaurouge | stoffell, cool thx. What about these May 2 18:12:56 WARNING[11306] chan_zap.c: Detected alarm on channel 10: No Alarm |
15:40.59 | chapeaurouge | May 2 18:12:56 WARNING[11306] chan_zap.c: Unable to disable echo cancellation on channel 10 |
15:41.07 | chapeaurouge | normal behavior? |
15:41.23 | asterboy | yep |
15:41.29 | file | DoktorGreg: maybe, wazzup? |
15:41.30 | SplasPood | Hrm.. why does background abort immediately on a single digit 0, but on 165 (for example) it waits for all digits? |
15:41.58 | vader-- | voipsupply.com sent me a TDM2400P board with 5 s400M Rev B2 modules and 1 s400M Rev B module |
15:42.04 | vader-- | wonder if i should bitch |
15:42.15 | Zeeek | SplasPood have you looked at the extension evaluation order on the wiki? |
15:42.50 | vader-- | the board they sent me didn't work so i RMAed it and they sent out another board with FXO modules on it |
15:43.07 | DoktorGreg | I need to know what kind of motherboard to buy to get zttest score closer to 100% |
15:43.19 | DoktorGreg | file |
15:43.45 | file | what is it at now? |
15:43.48 | DoktorGreg | right now i am at 98.75% |
15:43.59 | file | using what, actual zaptel hardware? |
15:44.08 | DoktorGreg | 405 |
15:44.12 | SplasPood | Zeeek: Yes I'm looking at it now, although I'm not sure how it applies to my issue |
15:44.13 | DoktorGreg | er 205 |
15:44.20 | DoktorGreg | sip phones work fine |
15:44.26 | jhava | Hello all, quick question: is there a configuration in ISDN PRI to accept digit by digit dialing from a PABX using a TE110P? |
15:44.28 | DoktorGreg | however when i hit music on hold with |
15:44.31 | SplasPood | if the exten => 0,1,NoOp() |
15:44.34 | DoktorGreg | one of the pri lines |
15:44.36 | SplasPood | and you dial 011 |
15:44.36 | *** join/#asterisk mog_work (n=mogorman@gateway.digium.com) |
15:44.37 | SplasPood | for example |
15:44.39 | DoktorGreg | the call drops |
15:44.42 | SplasPood | shouldn't that NOT match? |
15:45.26 | file | DoktorGreg: only when it goes through the 205? |
15:45.29 | *** join/#asterisk bkw_ (n=brian@adsl-70-143-63-171.dsl.tul2ok.sbcglobal.net) |
15:45.32 | DoktorGreg | yah... |
15:45.41 | DoktorGreg | the 205 is bridging calls just fine |
15:45.50 | file | interesting |
15:46.18 | DoktorGreg | and zttest reports that i drop to 97.5% when the call drops |
15:46.42 | eric_hill | DoktorGreg: What motherboard/system type do you have now? |
15:47.02 | DoktorGreg | its a MSI mobo, AMD nforce2 thingie |
15:47.09 | Zeeek | SplasPood your issue is that the evaluation is "premature" is it not? |
15:47.11 | *** join/#asterisk Xacau (i=TCHE@201-35-189-247.smace701.dsl.brasiltelecom.net.br) |
15:47.15 | *** join/#asterisk hrhrhr (n=c1@87.127.7.210) |
15:47.16 | eric_hill | Can you pastebin the output of lspci? |
15:47.21 | DoktorGreg | I am gonna try a different notwork |
15:47.22 | SplasPood | Zeeek: Correct |
15:47.23 | DoktorGreg | sure |
15:47.28 | DoktorGreg | one sec |
15:47.34 | hrhrhr | hello :) |
15:47.49 | eric_hill | Also, an "lspci -tv" please. |
15:47.58 | Zeeek | SplasPood and that page is about the order in which the exten are evaluated, is it not? |
15:48.29 | SplasPood | Zeeek: Yes, but based upon the page I don't see what's wrong with my setup. I'm not trying to argue here.. maybe you could show me where I'm going wrong.. |
15:48.43 | DoktorGreg | http://pastebin.ca/52890 |
15:48.48 | Zeeek | What URL are you looking at? |
15:49.27 | Xacau | can i disc to anyone using asterik? |
15:49.28 | DoktorGreg | http://pastebin.ca/52891 |
15:49.31 | Xacau | *call |
15:49.42 | Xacau | including cellphone? |
15:49.54 | Zeeek | SplasPood as far as the wrong first digit, all you can do is prevent it by limiting what is evaluated |
15:50.15 | SplasPood | Zeeek: http://www.voip-info.org/wiki/index.php?page=Asterisk+config+extensions.conf+sorting |
15:50.25 | Zeeek | yeah that's the one |
15:50.29 | DoktorGreg | afk getting hot coffee |
15:50.32 | DoktorGreg | brb |
15:50.47 | Zeeek | don't let any extension be dialed wihout checking it first would be my suggestion |
15:51.38 | SplasPood | Zeeek: What do you mean checking it first? |
15:51.53 | eric_hill | DoktorGreg: any idea what the "unknown device" is? Also, are you using a Wildcard? |
15:52.12 | *** join/#asterisk marl (n=matt@albacom.plus.com) |
15:52.12 | DoktorGreg | the unknown device is the 205 |
15:52.48 | Zeeek | Splaspod, I reread what you wrote higher up. Background is falling thru after a single zero, is that it? |
15:53.19 | hrhrhr | could you point me in the direction of a guide to getting a generic wildcard fxo card working with asterisk (built from src)? :) |
15:53.22 | SplasPood | Zeeek: Correct |
15:54.03 | Zeeek | and beneath Background() there is a check for a few digits and then you dial anything that doesn't meet the few single digits you allow? |
15:54.07 | marl | hi, can anyone point me in the rite direction for this, ive search google etc. and not found anything, i have an TDM400 card with an 2 x fxo and 1 x fxs boards on it, i find that any calls through the fxs phone tend to be very quite at my end, hoiw can i turn the gain up on this board? ive used fxotune to tune the fxo cards, but cant find the equiv for fxs, any one any pointers? |
15:54.32 | Zeeek | marl look at zaptel.conf |
15:54.46 | Zeeek | and serach google for zaptel gain or something like that |
15:55.06 | SplasPood | Zeeek: No, I define 4 extens... 0, 1, 2, 3 and have an 'i' exten that plays 'pbx-invalid' then goto's back to the background() |
15:55.21 | SplasPood | Zeeek: I also include another context which handles my direct dial extens... _20XX and _21XX |
15:55.24 | Zeeek | SplasPood maybe you want to pb that |
15:55.34 | SplasPood | ~pb |
15:55.35 | jbot | i heard pb is a place to paste your stuff without flooding the channel - try http://pastebin.com/, or http://pastebin.ca |
15:55.51 | Zeeek | heh, I'm getting lazy in my advanced age |
15:56.39 | marl | thanks Zeeek ill have a look there |
15:56.52 | Zeeek | yeah I don't recall the gain lines in my head |
15:57.06 | Zeeek | txgain? rxgain? |
15:57.42 | Zeeek | zapata.conf, not zaptel, sorry |
15:57.58 | Zeeek | and it is rxgain= and txgain= |
15:59.12 | hrhrhr | i've followed this so far http://forums.digium.com/viewtopic.php?t=6151&highlight=wildcard+x100p |
15:59.26 | hrhrhr | but looking at the asterisk console whilst dialling that pstn number doesn't give any output |
15:59.45 | hrhrhr | i guess i've missed something huge out somewhere... |
16:00.48 | *** join/#asterisk spunz_ (n=spunz@h081217096096.dyn.cm.kabsi.at) |
16:02.02 | vader-- | heh dude at voipsupply.com screwed up my RMA |
16:02.16 | vader-- | sent me a 2406E instead of a 2460E which is what he sent me first |
16:02.28 | salviadud | voipsupply suck |
16:02.36 | salviadud | they wouldn't even sell to me |
16:02.41 | vader-- | really why? |
16:02.42 | salviadud | cause i'm international |
16:02.47 | vader-- | gotcha |
16:02.50 | salviadud | i used voxilla |
16:02.54 | salviadud | they rock |
16:02.55 | vader-- | they probably can't or something |
16:03.06 | salviadud | voipsupply is based in NY |
16:03.13 | salviadud | they're a bunch of sissies |
16:03.15 | vader-- | some US companies won't send international |
16:03.19 | Zeeek | where are you salviadud? |
16:03.22 | vader-- | because of US laws or what not |
16:03.23 | salviadud | mexico |
16:03.29 | salviadud | because they are idiots |
16:03.31 | Zeeek | funny they sold to me |
16:03.45 | salviadud | i have bought some good stuff from the USA, me being mexican |
16:03.47 | vader-- | heh tomorrow i will have 3 TDM2400P cards on my desk |
16:03.51 | salviadud | the smart companies, those are the ones that sell |
16:03.52 | vader-- | 1 which doesn't work |
16:03.57 | vader-- | 1 which has the wrong modules |
16:04.07 | vader-- | and 1 which hopefully will have the right modules and works |
16:04.08 | *** part/#asterisk downunder33 (n=robert@219.95.158.235) |
16:04.20 | salviadud | so, you'll have to return 2 |
16:04.23 | vader-- | ya |
16:04.23 | vader-- | hehe |
16:04.25 | salviadud | that's a bummer |
16:04.52 | vader-- | ya it sucks because the first one they sent has the right modules but the card doesn't work |
16:05.00 | vader-- | and the second one works but doens't have the right modules |
16:05.02 | brookshire | what's wrong with the tdm2400p? |
16:05.10 | vader-- | and i asked if i can just switch the modules he said no |
16:05.15 | vader-- | they will send a whole new card |
16:05.24 | brookshire | that's efficient, lol |
16:05.36 | vader-- | brookshire no matter what computer i put the card into it doesn't recognize |
16:05.41 | SplasPood | brookshire: any idea if asterisk treats '0' during a Background() differently than any other digits? |
16:05.43 | eric_hill | DoktorGreg: Based on the lspci, try moving the 205 to a different slot and see what happens. |
16:06.01 | Hmmhesays | god there is a lot of shiat broken in the latest a@h |
16:06.02 | SplasPood | brookshire: it seems to be taking the 0 and immediately executing rather than waiting for more digits |
16:06.02 | brookshire | spals: is this during voicemail? |
16:06.03 | DoktorGreg | eric_hill, thats where i was headed next |
16:06.10 | SplasPood | brookshire: nope, during a Background() |
16:06.13 | Zeeek | SplasPood it doesn't in 1.2 because I use it there with 0 |
16:06.39 | salviadud | splashpood, do you have an i extension? |
16:06.45 | *** join/#asterisk myiagy (n=myiagy@mail.voffice.com.br) |
16:06.45 | brookshire | splas: you mean like 0XXXX |
16:06.48 | SplasPood | salviadud: yes |
16:06.58 | salviadud | i was playing with background just last night |
16:07.00 | SplasPood | I have an i exten and an exten => 0,1,NoOp() |
16:07.06 | SplasPood | if someone dials 0123 |
16:07.11 | SplasPood | I expect it to hit the 'i' exten |
16:07.14 | SplasPood | instead it goes to 0 |
16:07.16 | SplasPood | immediately |
16:07.37 | Zeeek | normal |
16:07.39 | brookshire | make sure the 0123 pattern is before the 0,1,NoOp() |
16:07.56 | brookshire | i don't know why that would matter, but! |
16:07.57 | salviadud | you see, that pattern does not exist |
16:08.03 | salviadud | hehe |
16:08.12 | SplasPood | right |
16:08.14 | SplasPood | its invalid |
16:08.25 | SplasPood | there is no 0123 pattern |
16:08.25 | brookshire | 1.2.7.1 |
16:08.28 | SplasPood | yes |
16:08.43 | salviadud | mmmm |
16:08.48 | salviadud | take out that extension? |
16:08.49 | Zeeek | SplasPood show us the context in pb |
16:08.54 | salviadud | it doesn't seem to be useful |
16:09.11 | salviadud | or just change the 0 for something else |
16:09.26 | SplasPood | Zeeek: I did, I msg'd it to you when you asked the first time :P |
16:09.43 | salviadud | splaspood |
16:09.43 | Zeeek | oh. I don't get messages, sorry |
16:09.45 | salviadud | i want to see it |
16:09.46 | SplasPood | salviadud: 0? Well 0 rings the operator... so I'd say thats useful :) |
16:09.52 | *** join/#asterisk fu3 (n=kaa@234-200-29-134.hcc.mnscu.edu) |
16:09.59 | fu3 | hello lads |
16:10.06 | brookshire | splaspod: works for me |
16:10.20 | Zeeek | SplasPood what's the big secret, show it to the world |
16:10.27 | fu3 | Is there any way to expand the line number on a Polycom 301? It only shows "...900" as line1 and i'd like it to show the entire number. |
16:10.33 | brookshire | exten => 0123,1,Answer |
16:10.33 | brookshire | exten => 0123,2,MusiconHold(native-random) |
16:11.14 | Zeeek | SplasPood by the way if you hit '*' or '7' does it go to i ? |
16:11.45 | brettnem | fu3: no |
16:12.12 | fu3 | no? really? theres all that space next to it! |
16:12.26 | fu3 | so i'll only ever be able to see the last three digits of an extension? |
16:12.43 | SplasPood | Zeeek: 7 does |
16:12.50 | SplasPood | Zeeek: as does * |
16:12.59 | SplasPood | immediately too |
16:13.02 | Zeeek | ok |
16:13.02 | brettnem | fu3: You can't change the display on the polycom |
16:13.14 | salviadud | that's gotta be the weirdest ivr i've ever seen |
16:13.19 | fu3 | weak.. why would they design it so as to only show the last three digits.. argh |
16:13.33 | SplasPood | salviadud: what's weird about it? |
16:13.40 | SplasPood | seems pretty straight forward to me... |
16:14.02 | *** part/#asterisk ibob63 (n=hp@bb-87-82-7-89.ukonline.co.uk) |
16:14.21 | salviadud | well, it's not that it's weird, it's a different style |
16:14.31 | *** join/#asterisk dos000 (n=dos000@wsp05974758wss.cr.net.cable.rogers.com) |
16:14.51 | SplasPood | salviadud: What specifically? |
16:14.58 | dos000 | is there a command to read from arbitrary fields in the mysql db ? |
16:15.49 | salviadud | they way you start the backgroudn app |
16:16.12 | salviadud | i usually just do exten => s,1,Bacground... |
16:16.26 | noname32 | any ideas on what i cant record with *1? i have it enabled in features and the dial and when i press it this is what shows in the log res_features.c: Feature interpret: chan=SIP/200-3650, peer=SIP/201-42b5, sense=1, features=16 res_features.c: Set time limit to 500 res_features.c: Timed out for feature! |
16:16.43 | noname32 | i just cant get it to work for the life of me |
16:17.11 | *** join/#asterisk Gamercjm (n=chris@pool-71-254-177-36.lsanca.fios.verizon.net) |
16:17.16 | salviadud | well, there's no real error here... |
16:17.26 | salviadud | if they dial an initial 0 |
16:17.30 | SplasPood | salviadud: Thats exactly how I'm doing it... |
16:17.32 | salviadud | it will always go to the operator |
16:17.42 | SplasPood | yes |
16:17.44 | SplasPood | I don't want that |
16:17.49 | SplasPood | only if they dial 0 on it's own |
16:17.53 | SplasPood | since it's not _0. |
16:17.56 | SplasPood | its just 0 |
16:18.07 | coppice | the weirdest ivr i've ever seen says "press 1 for english" in cantonese and "press 3 for cantonese" in english :-) |
16:18.31 | *** join/#asterisk Assid (n=assid@203.115.64.12) |
16:18.36 | Assid | yoza |
16:18.40 | salviadud | the guy who made that. must be a prankster |
16:18.49 | salviadud | gotta admire the ingenuity |
16:19.49 | salviadud | splaspood, you are not giving * any reason to wait for another digit |
16:20.00 | salviadud | all your extensions are single numbers |
16:20.13 | salviadud | you would need to have a 02 extension |
16:20.16 | salviadud | for example |
16:20.41 | SplasPood | salviadud: what's TIMEOUT(digit) for then |
16:20.49 | SplasPood | yea I setup exten => _0X.,1,Background(invalid) and that fixes it |
16:21.14 | salviadud | that variable is the time between digits |
16:21.23 | [TK]D-Fender | SplasPood : Just pastebin the whole thing so we can steer you right in 1 shot.... |
16:21.25 | salviadud | but since, they are all single digits... |
16:21.30 | salviadud | it's kinda pointless |
16:21.32 | Zeeek | <PROTECTED> |
16:22.02 | SplasPood | [TK]D-Fender: I did pastebin it and gave it to the 3 people I was talking to :P |
16:22.17 | *** join/#asterisk mtaht3 (n=m@c-71-198-23-124.hsd1.ca.comcast.net) |
16:22.18 | Zeeek | if you did it publicly you'd have 240 possible helpers |
16:22.26 | [TK]D-Fender | SplasPood : If its still being discussed maybe you might want to ask someone else :) |
16:22.31 | Zeeek | heh |
16:22.37 | Assid | sup Zeeek |
16:22.45 | Zeeek | nada Assid |
16:23.09 | SplasPood | [TK]D-Fender: Nah I think i've "fixed it" ... |
16:23.28 | [TK]D-Fender | SplasPood : "think"? :) |
16:23.53 | salviadud | well, it seems like it could work splaspood, give it a try |
16:23.56 | salviadud | debug the thing |
16:24.15 | SplasPood | it does work |
16:24.17 | SplasPood | I have tried it |
16:24.21 | SplasPood | I just don't like the solution :) |
16:24.32 | salviadud | why, the code is ugly? |
16:24.34 | [TK]D-Fender | SplasPood : C'mon show me.... |
16:25.01 | salviadud | d-fender, when is your b-day? |
16:25.15 | [TK]D-Fender | salviadud : March 7th |
16:25.59 | [TK]D-Fender | salviadud : Why do you ask? |
16:26.14 | *** join/#asterisk lzhang (n=rjrae@67.95.13.46) |
16:26.39 | salviadud | i was wondering about your astrological sign |
16:26.52 | [TK]D-Fender | "Octagon... as in STOP" :D |
16:26.53 | lzhang | I'm trying to make a call using the asterisk manager to have the system dial a number and just play a wav file |
16:26.59 | salviadud | you are a water element, that's nice |
16:27.08 | LostFrog | Damn, [TK]D-Fender, you stole my joke. |
16:27.09 | lzhang | I'm looking at originate, but I'm not sure what I put for the channel |
16:27.33 | [TK]D-Fender | LostFrog : Depends, how old are you? |
16:27.34 | *** join/#asterisk MacDome (n=eseidel@A17-255-100-181.apple.com) |
16:27.55 | LostFrog | [TK]D-Fender: Old enough to know better, but young enough not to care. |
16:28.12 | SplasPood | salviadud: Just kludgy |
16:28.28 | [TK]D-Fender | LostFrog : "Better than whom" :) |
16:28.32 | *** join/#asterisk [Airwolf] (n=airwolf@83.98.235.219) |
16:28.41 | salviadud | well, i would worry if it works, rather than if it's kludgy |
16:28.59 | LostFrog | I would worry if it *didn't* work. |
16:29.03 | [TK]D-Fender | SplasPood : Just pastbin it... I'm sure theres something we can do to clean it up. |
16:29.37 | SplasPood | [TK]D-Fender: Forget it man.. I'm convinced this is by design and kludging it with dummy patterns is the only fix |
16:30.25 | *** join/#asterisk [Airwolf] (n=airwolf@83.98.235.219) |
16:30.43 | salviadud | right, if it didn't |
16:30.50 | salviadud | O_o |
16:30.53 | [TK]D-Fender | SplasPood : Your call... though I doubt it.... |
16:31.30 | salviadud | SplasPood, i think you fixed it. |
16:32.31 | *** join/#asterisk [Airwolf] (n=airwolf@83.98.235.219) |
16:32.46 | [TK]D-Fender | [Airwolf] : *boing* |
16:33.00 | *** join/#asterisk MacDome (n=eseidel@A17-255-100-181.apple.com) |
16:33.01 | [TK]D-Fender | Thats the best part of my God complex.... no peer pressure :) |
16:33.14 | *** join/#asterisk Lino` (n=Lino@i577BC81E.versanet.de) |
16:34.17 | coppice | I have a superiority complex without equal :-) |
16:35.37 | noname32 | any ideas why one touch recording throughs this error Timed out for feature! and never works :/ |
16:35.53 | [TK]D-Fender | noname32 : Taking too long between the * and 1 maybe? |
16:36.09 | SplasPood | [TK]D-Fender: You're the only dissenting voice of 4 |
16:36.10 | *** join/#asterisk oej (n=oej@apollo.webway.se) |
16:36.30 | Zeeek | what 4? I didn't vote yet |
16:36.39 | noname32 | i dont think so cause soon as the call is connect i press it and soon as i do it it outputs that to the log |
16:36.44 | *** join/#asterisk Moradinn (n=H0t-5auc@213.122.19.209.transedge.com) |
16:36.57 | noname32 | but if i change the command in features to ** it will work |
16:37.14 | Zeeek | SplasPood would say, "change it to **" |
16:37.25 | SplasPood | ** ? |
16:37.25 | noname32 | the features cmds dont work for me as long as they are * followed by a number |
16:37.41 | Zeeek | check your phone |
16:37.53 | froguz | i'm trying to get realtime to work (sipusers and sipeers in realtime mode) but i'm getting : res_config_mysql.c: MySQL RealTime: Failed to connect database server realtime on localhost. |
16:38.10 | CunningPike | Good morning |
16:38.30 | noname32 | well i can use the other stuff like *61 *43 *98 ect |
16:38.50 | froguz | my res_mysql.conf file : |
16:38.52 | froguz | [general] |
16:38.52 | froguz | dbhost = localhost |
16:38.52 | froguz | dbname = realtime |
16:38.52 | froguz | dbuser = root |
16:38.52 | froguz | dbpass = pass |
16:38.53 | froguz | dbport = 3306 |
16:38.55 | froguz | dbsock = /tmp/mysql.sock |
16:38.59 | noname32 | it seems obnly when the call is active i cant do like *0 to hangup or *2 att transfer |
16:39.07 | froguz | sorry about the flooding |
16:39.14 | noname32 | i have tryed with x-lite and a cisco ata 186 |
16:40.19 | *** join/#asterisk pythos (i=pythos@unaffiliated/pythos) |
16:41.08 | noname32 | it says i need to turn on dynamic features for this right so in extensions.conf i put [globals] |
16:41.08 | noname32 | Set(DYNAMIC_FEATURES=automon) |
16:41.08 | noname32 | <PROTECTED> |
16:44.38 | *** join/#asterisk mko-025 (n=korpim@p54989A8C.dip0.t-ipconnect.de) |
16:44.39 | froguz | i've created static_table, sip_table, user 200 and 300 in sip_table, i've edited extconfig.conf to the database settings, but it still can't connect |
16:45.28 | froguz | i've edited the res_mysql.conf too |
16:47.08 | froguz | am i missing something |
16:47.10 | froguz | ? |
16:48.45 | *** join/#asterisk esculapio_ (i=elvyn@200.88.44.66) |
16:48.56 | *** join/#asterisk Abydos313 (i=abydos31@adsl-71-129-52-80.dsl.irvnca.pacbell.net) |
16:48.57 | esculapio_ | hola quien habla espanol? |
16:49.30 | *** join/#asterisk suma (n=suma@cm69.gamma29.maxonline.com.sg) |
16:49.41 | suma | which is the best calling card app for asterisk? |
16:49.54 | suma | with web user interface |
16:50.00 | *** part/#asterisk Moradinn (n=H0t-5auc@213.122.19.209.transedge.com) |
16:50.06 | esculapio_ | tecnico, Hi, Hola |
16:50.28 | froguz | esculapio_, yo hablo español ¿qué necesitas? |
16:50.53 | *** join/#asterisk razu (n=razu@dhcp-84-52-1-207.cable.infonet.ee) |
16:50.55 | esculapio_ | froguz, me puedes ayudar con la coneccion de dos asterisk |
16:51.22 | esculapio_ | s/s/x |
16:51.22 | suma | am i in the wrong channel !, something other than english is going on here |
16:51.45 | mog_work | no your in the irght channel |
16:51.58 | suma | mog_work: thanks |
16:52.08 | [TK]D-Fender | <mog_work> no your in the irght channel <- no we have American in here too ;) |
16:52.22 | tainted- | lol |
16:52.23 | mog_work | heh |
16:52.32 | tainted- | I speak well Aelmican |
16:52.49 | [TK]D-Fender | tainted- : I read that like "pellican" on first clance :D |
16:52.49 | Nivex | "Lady, I only speak two languages: English and bad English." |
16:52.53 | suma | me to, i speak well almecidnassan |
16:53.17 | suma | which is the best calling card app for asterisk with web user interface? |
16:53.37 | [hC] | a2billing seems popular |
16:53.42 | tainted- | u speak delicatessen? |
16:53.49 | [TK]D-Fender | tainted- : YUM! |
16:54.03 | [TK]D-Fender | I speak in tongues... just ask my ex :) |
16:56.56 | pythos | <PROTECTED> |
16:57.37 | esculapio_ | froguz, como es mejor por que solo estoy empezando |
16:58.02 | esculapio_ | froguz, para empezar el que tu me digas\ |
16:58.20 | froguz | esculapio_, read the private message window, ppl don't like any other lang. tha english here |
17:00.28 | esculapio_ | froguz, no tengo problemas con leer el ingles muchas gracias en esto momento me encuento en el link que me diste |
17:01.51 | froguz | ok |
17:02.30 | Hmmhesays | is froguz from france? |
17:02.56 | coppice | waak je kui dei m'se ying man |
17:04.47 | *** join/#asterisk zagaya971 (n=almeli@APointe-a-Pitre-102-1-3-9.w81-248.abo.wanadoo.fr) |
17:05.41 | *** join/#asterisk saftsack (n=saftsack@p54A7F975.dip.t-dialin.net) |
17:05.49 | pythos | who is psychic here? What do I need now :-) |
17:06.59 | *** join/#asterisk markus99 (n=markus@165.154.121.219) |
17:07.30 | pythos | ok... that was sillyness, now then: I got the hardware al figured out for my tdm400p, but I am not sure what to use for a web based management product, suggestions? |
17:08.10 | Hmmhesays | don't you wish your girlfriend was hot like me |
17:08.27 | pythos | are you a female? |
17:08.37 | Hmmhesays | no but I like them |
17:09.39 | pythos | hmm, well then, I don't wish my girlfriend was hot like you |
17:10.31 | [TK]D-Fender | pythos : Sorry, we don't have any spare clues to give you :) |
17:11.03 | pythos | [TK]D-Fender: hmm. well Im looking at sourceforge... there are a TON, just wanted to know where to start |
17:11.22 | froguz | Hmmhesays, nop. is froguz from chile |
17:11.30 | [TK]D-Fender | pythos : All web interfaces for * suck, just some less than others. Why do you think you want/need it, and describe the size and type of system its expected to help you administer. |
17:11.38 | Hmmhesays | Do they hold carnivale in chile? |
17:11.52 | tainted- | froguz chilean women are HOT!@#!@$ |
17:12.08 | coppice | what do you expect in that climate |
17:12.13 | froguz | hahaha tainted- yes they are |
17:12.14 | Hmmhesays | there are beautiful women everywhere |
17:12.32 | froguz | that's true too |
17:12.39 | coppice | I have two in this house :-) |
17:13.52 | froguz | i can't get asterosk to connect to the realtime database... i think there's something wrong with my res_mysql.conf but i can't find what is wrong |
17:14.02 | froguz | maybe the socket name |
17:14.21 | froguz | is /tmp/mysql.sock an standard to all distros?? |
17:15.22 | Nugget | no. |
17:15.43 | froguz | mmmm... that may be the problem |
17:16.58 | tainted- | froguz cafe w/ legs :D |
17:18.04 | froguz | tainted-, hahaha are you chilean? |
17:18.21 | tainted- | i wish |
17:18.24 | tainted- | hahaha |
17:18.30 | froguz | you surely have been here |
17:18.39 | froguz | café con piernas rocks |
17:18.57 | tainted- | are u there now? |
17:19.46 | froguz | in a cafe??? i wish i could, but i'm at the office, here in santiago |
17:22.20 | sevard | can anyone help me with a zap option I can't find |
17:22.47 | sevard | I know how to direct all calls from ZAP to my IVR, but I can't figure out how to direct individual ports to different sip lines |
17:25.58 | froguz | tainted-, were are you from? |
17:26.08 | froguz | where* |
17:26.15 | *** join/#asterisk MrDigital (n=VBDIGITA@pool-72-81-113-227.phlapa.east.verizon.net) |
17:26.26 | tainted- | froguz i'm in Los Angeles currently |
17:26.34 | tainted- | (not by choice) |
17:26.36 | tainted- | haha |
17:27.16 | vader-- | is it normal for there to be no ringing when dialing an sip extension that has the phone off or not connected? |
17:27.53 | syzygybsd | on an incomming call on a registered sip connection, should it be asking me to reregister? |
17:28.15 | vader-- | when there is no phone connected and you try to dial that sip extension the phone dialing just sits silent |
17:29.52 | froguz | not by choice!!! wow! don't you like to be there? |
17:31.07 | Hmmhesays | geebus it's like eleventy billion degrees in here |
17:33.08 | Hmmhesays | I love it when clients of clients try to contact me |
17:33.34 | Hmmhesays | they think they can just bypass the chain of command |
17:34.08 | *** join/#asterisk Astinus- (n=abba@213.167.111.138) |
17:34.12 | syzygybsd | lol @ Hmmhesays what about providers of your clients? |
17:34.15 | salviadud | so, what do you do? |
17:34.31 | salviadud | send em' to your clients? |
17:34.34 | Hmmhesays | bingo |
17:35.00 | salviadud | yeah, that's lovely |
17:35.25 | salviadud | ask Bob, cause it's not my problem yet |
17:35.36 | froguz | wiki says "You can keep any sip users in the flatfile AND use RealTime. How cool is that?" but in other page of the wiki says : "If you store sip.conf in the RealTime database, you need to rename/remove the text file otherwise the text file will superceed RealTime." wich one is right?? |
17:35.57 | Hmmhesays | i also dislike cheap bastards |
17:36.11 | salviadud | what kinda cheap are we talking about here? |
17:37.54 | *** join/#asterisk insomni (n=insomni@x1-6-00-e0-18-6f-34-ad.k455.webspeed.dk) |
17:38.18 | Hmmhesays | the kind of cheap that thinks they should get something for nothing |
17:38.58 | Zeeek | entropy is against something for nothing. It won't allow it, ever. |
17:39.14 | Zeeek | You can only *think* it's for nothing. |
17:39.30 | sevard | anyone? zap help? |
17:39.42 | *** join/#asterisk myiagy (n=myiagy@mail.voffice.com.br) |
17:39.48 | Zeeek | no thanks |
17:39.57 | sevard | Hmmhesays: btw thanks for something for next to nothing. |
17:40.06 | Hmmhesays | you paid |
17:40.12 | Hmmhesays | therefore your legs aren't broken |
17:40.24 | *** join/#asterisk tzafrir_laptop (n=tzafrir@bzq-179-75-202.cust.bezeqint.net) |
17:40.58 | Zeeek | sevard have you asked your question yet? |
17:41.12 | sevard | Zeeek: yeah |
17:41.19 | sevard | I don't like reposting, but if you'd like |
17:41.21 | Zeeek | and still no answer? |
17:41.29 | sevard | It was about a half hour ago, but no. |
17:41.32 | Zeeek | how about re-stating? |
17:41.34 | sevard | I know how to direct all calls from ZAP to my IVR, but I can't figure out how to direct individual ports to different sip lines |
17:41.46 | Zeeek | what do you mean by ports? |
17:41.56 | sevard | a TDM400P has four ports |
17:42.02 | Zeeek | channels |
17:42.15 | Zeeek | you have 4 FXS? |
17:42.15 | sevard | I'm pretty sure that 'port' isn't the techincal term but that's what I've been using |
17:42.17 | sevard | okay, channels. |
17:42.27 | sevard | I have four FXO i believe, four red ones :) |
17:42.38 | Zeeek | well otherwise we'd be forwarding 5060 to one place, 5061 to another |
17:42.52 | Zeeek | ok you have four incoming ZAP channels |
17:43.06 | sevard | Yup, outgoing works great |
17:43.23 | sevard | I have round robin on one and two and long distance on 3, four isn't hooked up yet |
17:43.37 | *** join/#asterisk drfoomod2 (i=DrFooMod@ool-43501d9f.dyn.optonline.net) |
17:43.44 | drfoomod2 | can you get info in the sip.conf file from the management API? |
17:43.47 | *** join/#asterisk markus99 (n=markus@165.154.121.219) |
17:43.53 | sevard | I have all zap incoming directed to my ivr and the question arose how do I direct channel 1 to a sip line directly and the rest to the ivr |
17:44.14 | sevard | I already know how to direct a IAX trunk directly to a SIP line |
17:44.19 | Zeeek | wait I now have to deal with a totally incompetent person who is crossing my emails with incredibly silly questions about an order |
17:44.24 | Katty | to all of you guys that have offered to get someone a drink when they're having a really crappy, i'd like to thank you obsessively. |
17:44.25 | myiagy | using different contexts? |
17:44.32 | Katty | s/crappy/crappy day/ |
17:44.43 | Katty | thank you thank you thank you |
17:44.45 | *** join/#asterisk gammacoder (n=chatzill@64-132-192-33.gen.twtelecom.net) |
17:44.48 | sevard | <PROTECTED> |
17:44.51 | myiagy | sevard if i'm not wrong, you can use a different context for each channel.. you define them in the zapata.conf file |
17:44.55 | Katty | no, i'm being serious |
17:45.06 | Katty | i was /just/ at the gas station, and the guy offered to get my tea for me |
17:45.13 | Katty | he must have been able to tell i was having a really shitty day |
17:45.14 | *** join/#asterisk _-Jon-_ (n=jon@CPE000d8861e8f7-CM00080d290642.cpe.net.cable.rogers.com) |
17:45.21 | Katty | just remember that boys! |
17:45.22 | sevard | Katty: or thought you were pretty. |
17:45.27 | Katty | if she looks down, offer to get her tea. |
17:45.34 | _-Jon-_ | hey everyone |
17:45.38 | Katty | even if she doesn't take you up on it, she'll feel better. |
17:45.41 | sevard | Katty: in kung foo you hit when they look away |
17:46.00 | Katty | sevard: would you shut up and take me seriously, you goofball. |
17:46.12 | Zeeek | Katty I am so resisting answering the above statement about "looking down" |
17:46.29 | Katty | oh dear. |
17:46.40 | _-Jon-_ | I'm having some slight issues with my Asterisk setup and wondering if there's any way to pinpoint the problem. Basically when a call comes in, 9/10 times it works, but sometimes it doesn't. And I'm not sure if it's Asterisk, BroadVoice, or what |
17:46.40 | Katty | for /once/ don't twist what i say :P |
17:46.46 | Zeeek | and you weren't speaking in the first person so I could have |
17:46.54 | *** join/#asterisk Seggy (i=rbutler@tsss.org) |
17:46.59 | Zeeek | but as a gentlement... didn't |
17:47.02 | Katty | good job. |
17:47.07 | _-Jon-_ | I can sit here all day and place test calls and they'll all work, but sometimes other people call and it won't come through |
17:47.08 | Zeeek | again! |
17:47.14 | Katty | the Good Girls like gentleman. |
17:47.28 | Katty | if you're not a gentleman, we're not going to waste our time |
17:47.33 | Katty | or at least i wouldn't. |
17:47.50 | Katty | but then again i'm insanely picky :P |
17:47.54 | Zeeek | I can't believe how stupid some people are! This woman says "just email me your credit card info" |
17:48.06 | Zeeek | yeah right, let me send my carys with that too |
17:48.17 | Zeeek | <PROTECTED> |
17:48.26 | *** join/#asterisk schirpich (n=kvirc@ip21.farheap.net) |
17:48.28 | Katty | if she's asking you for the credit card number, and she's not your wife or a sales rep.... |
17:48.32 | Katty | then don't waste your time |
17:48.49 | Zeeek | no it's legit, just a stupid person with the title "Vice Presifent" |
17:49.10 | Katty | there are still good people out there :P |
17:49.11 | Zeeek | proving that women *can* be as stupid as men sometimes :) |
17:49.30 | Katty | yeah, we can all be stupid sometimes though |
17:49.34 | Katty | whether intentionally or not. |
17:49.40 | *** join/#asterisk Lord_Drachenblut (n=Lord@12-210-117-62.client.insightBB.com) |
17:49.42 | Zeeek | plus all my tollfree are down and I can't reach her for what would take 1 min on the phone |
17:50.07 | Katty | Hmmhesays: you around? |
17:50.25 | schirpich | Im kinda green when it comes too asterisk, but i was wondering if anyone could point me in the right direction. I wanna have a SIP polycom 301 phone connect to my work asterisk server through our vpn, how would this be done? |
17:50.52 | schirpich | do i use something like a sipproxy? |
17:51.00 | _-Jon-_ | Is there any way to have the SIP debugging information saved to a file? |
17:51.58 | _-Jon-_ | schirpich, just tell your phone to connect to IP of the Asterisk server that goes through the tunnel |
17:53.12 | schirpich | i get what your sayin, but the phone would still have to be connected to the vpn some way. i wanna make it so one of these phones can be plugged into an internet connection anywhere and connect up to my asterisk server |
17:53.22 | Hmmhesays | Katty: aye |
17:53.25 | Hmmhesays | i'm avoiding sevard |
17:53.51 | Katty | Hmmhesays: how often would you say you get a drink for a girl? |
17:54.01 | Hmmhesays | a girl that i'm trying to woooo? |
17:54.08 | _-Jon-_ | schirpich, hmm, i don't know if that's possible. I have mine connected through a vpn, but the VPN tunnel is done with my router |
17:54.09 | Katty | Hmmhesays: a girl in general. |
17:54.15 | Katty | Hmmhesays: whether you're trying to woo or not |
17:54.19 | Katty | Hmmhesays: your mom does not count |
17:54.20 | schirpich | what kind of router? |
17:54.21 | Hmmhesays | it's different in each case |
17:54.31 | Katty | Hmmhesays: 3 or 4 times a week? |
17:54.36 | _-Jon-_ | schirpich, Linksys WRT54G running openwrt |
17:54.55 | Hmmhesays | I don't buy girls drinks unless they are 1. a good friend or 2. a girlfriend |
17:55.38 | schirpich | -jon- that a modded router right? |
17:55.54 | _-Jon-_ | schirpich, yup. flashed with openwrt firmware |
17:55.59 | Katty | Hmmhesays: not even if they look really sad? |
17:56.14 | schirpich | -jon- are there any home/residental routers out there that have support for connecting to a vpn? |
17:56.18 | Hmmhesays | why I need to buy them a drink? |
17:56.18 | Katty | Hmmhesays: you mean to tell me if i was moping in the corner somewhere, you wouldn't offer to get me a drink? |
17:56.30 | Hmmhesays | I don't see that at the bars I go to |
17:57.01 | Katty | k |
17:57.13 | _-Jon-_ | schirpich, i think so. they might be a little bit more expensive. but you can buy a ton of different routers and run openwrt on them, not just Linksys |
17:57.14 | Hmmhesays | why do you ask? |
17:57.47 | Katty | Hmmhesays: if you ever /do/ see some mopey chick at a bar. |
17:57.52 | Katty | Hmmhesays: get her a drink, k? |
17:57.55 | Hmmhesays | why? |
17:57.57 | schirpich | -jon- i'll have to look into that |
17:58.04 | Katty | Hmmhesays: because it will turn her entire day around |
17:58.07 | Hmmhesays | ok this guy is really getting on my nerves |
17:58.12 | Hmmhesays | unless she wants to be alone |
17:58.23 | Katty | Hmmhesays: she can be alone. |
17:58.27 | _-Jon-_ | brb |
17:58.36 | Katty | Hmmhesays: but the offer is still a really great gesture. |
17:58.54 | Katty | Hmmhesays: even if she won't take you up on it :P |
17:59.14 | *** join/#asterisk C4T3l (n=rcall01@216.54.143.2) |
17:59.56 | C4T3l | howdy asterisk buddies! |
18:00.13 | Zeeek | are we still on that? After the call to the Vice President, I can now relax |
18:00.30 | Katty | Zeeek: yes, we are. |
18:00.31 | Hmmhesays | C4T3l: hi Dr. Nick? |
18:00.32 | Katty | Zeeek: it's important. |
18:00.38 | Katty | Zeeek: also, coupes are too |
18:00.39 | *** join/#asterisk MacDome (n=eseidel@A17-255-100-181.apple.com) |
18:00.46 | Katty | Hmmhesays: are you driving? |
18:00.56 | Zeeek | Well, if I offer something to a woman to make her feel better, she should take it! |
18:01.08 | *** join/#asterisk MikeJ__ (n=vircuser@d14-69-8-30.try.wideopenwest.com) |
18:01.11 | Hmmhesays | Katty: I got thrown in jail |
18:01.11 | Katty | Zeeek: sometimes it not polite to accept |
18:01.13 | Zeeek | I'm talking about herbal tea, obviously |
18:01.18 | Katty | Hmmhesays: still no driving then :< |
18:01.24 | Hmmhesays | 1 year |
18:01.25 | Katty | Hmmhesays: you can still help me pick out a car! |
18:01.46 | Katty | Hmmhesays: are you goign to be alright? |
18:02.01 | Hmmhesays | Katty: do I have a choice? |
18:02.03 | Katty | Hmmhesays: getting attached to cars can be emotional :< |
18:02.11 | Katty | Hmmhesays: sarcasm, dear ;) |
18:02.19 | Zeeek | Question to the world: now that most companies use voIP, is it possible that some calls will not be properly routed and won't work between some carriers? |
18:02.22 | syzygybsd | I am having trouble with my sip.conf file. I would like to be able to make incomming and outgoing calls on the same sip connection, however don't you need a register => and a context? |
18:02.26 | Hmmhesays | I'll be alright when I get the fuck outta this town |
18:02.39 | Katty | yes, you're doomed there. |
18:02.47 | justinu | LA? |
18:02.52 | Hmmhesays | syzygybsd: what an impossibly vague question |
18:02.55 | C4T3l | syzygy |
18:03.05 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
18:03.06 | C4T3l | more info plz |
18:03.08 | Katty | Hmmhesays: i've decided i want a manual coupe of some sort. |
18:03.12 | Zeeek | no, vague is "Newbie-pls help" |
18:03.22 | Hmmhesays | spelled poorly |
18:03.28 | Hmmhesays | better make sure you can drive a manual |
18:03.35 | justinu | anyone can drive a manual |
18:03.39 | Hmmhesays | not true |
18:03.43 | Katty | mom can't |
18:03.45 | [TK]D-Fender | syzygybsd : No, you don't necessarily need to register |
18:03.48 | Hmmhesays | this is the USofA buddy |
18:03.49 | Katty | i can sorta drive a manual |
18:03.51 | justinu | they just didn't try hard enough |
18:03.55 | Katty | but i've not had a lot of 'sperience with it |
18:04.03 | justinu | if my wife can drive a manual, anyone can |
18:04.05 | Hmmhesays | Automatics are the transmission of choice up here |
18:04.18 | syzygybsd | funny, I tried to give enough. Ok, simple sip connection to another server. if I want to accept incomming calls, I need a register => |
18:04.26 | Hmmhesays | how many clutches you go through a year |
18:04.31 | *** join/#asterisk Danett (i=none@a144029.upc-a.chello.nl) |
18:04.31 | Katty | Hmmhesays: i think it really comes down to I Love Exhaust. |
18:04.31 | justinu | none |
18:04.32 | Zeeek | makes talking on your cell while driving so much easier |
18:04.32 | Danett | heya |
18:04.47 | syzygybsd | if I want to make outgoing calls, I need a context in sip.conf with all the info so I can include that in my dial() statement |
18:04.48 | [TK]D-Fender | syzygybsd : You only need to register if the other side doesn't know your IP |
18:05.06 | [TK]D-Fender | syzygybsd : Thats the reason for its existance. |
18:05.08 | Danett | When you run a asterisk box, wich is attached to a domain, is it possible to dial to sip://user@domain.tld? |
18:05.23 | syzygybsd | [TK]D-Fender: I don't think they do... |
18:05.37 | [TK]D-Fender | syzygybsd : Thats why then. |
18:05.46 | justinu | i live in LA, i know all about people who buy automatics so they can put on makeup/eat/read/bucketmouth/beat their children/etc |
18:06.23 | [TK]D-Fender | justinu : Seen Pink's video for "Stupid Girls"? :) |
18:07.05 | justinu | no |
18:07.11 | C4T3l | Is the X-lite soft phone inherently choppy when it comes to the audio stream? |
18:07.19 | justinu | pop culture usually eludes me somehow |
18:07.42 | *** join/#asterisk dwildes2 (n=dwildes@209.164.237.195) |
18:07.44 | justinu | i watched some guy crash his vette into a center divider because he was too busy smaking his kid |
18:08.16 | *** join/#asterisk cytrak (n=kvirc@adelphi.geofocus.com) |
18:08.21 | [TK]D-Fender | justinu : Go watch! |
18:08.49 | Danett | How can i setup asterisk so it can relay sip calls made to user@domain.tld? |
18:09.20 | [TK]D-Fender | Danett : * is not a SSIP Proxy. Go look at SER |
18:09.42 | C4T3l | Danett: I think SER is better for that. |
18:09.50 | Danett | I am registere with x-lite to my asterisk box which sits on a certain domain. I want other agents to be able to call to user@domain.tld and reach my softphone |
18:09.54 | Danett | hmm ser is an option |
18:09.56 | justinu | you can terminate them pretty easily, by defining an exten => user |
18:10.57 | C4T3l | Danett: Have them register with yer * |
18:11.05 | Danett | well |
18:11.09 | Danett | i don't want that really |
18:11.17 | cytrak | is it ok to have two TE110P cards in one box ? |
18:11.17 | Danett | i want them to be registered at *some* provider |
18:11.44 | [TK]D-Fender | Danett : Does domain.tld point to your * box direct? |
18:11.55 | Danett | yes it does |
18:12.04 | Danett | on standard ports |
18:12.05 | [TK]D-Fender | cytrak : Cure, though You're better off with a single multi-port card |
18:12.13 | Danett | it points to voice.pbxservices.nl |
18:12.16 | [TK]D-Fender | Danett : Then you can already do that. |
18:12.27 | Danett | I am not able to reach it... |
18:12.43 | C4T3l | Danett: is sip reg failing? |
18:12.50 | Danett | doesn't show any debug |
18:12.55 | [TK]D-Fender | Danett : pastebin your sip.conf |
18:12.57 | Danett | it think x-lite cannot local loop |
18:13.33 | Danett | http://pastebin.com/694602 |
18:13.42 | cytrak | I've been trying to setup my * to talk to a siemens PBX using a single span card and it works fine for internal calls between soft<->hard phones but the soft phone can't get a way out to the world |
18:14.29 | cytrak | week configuration wouldn't work with me |
18:14.35 | cytrak | i mean wink |
18:15.36 | [TK]D-Fender | Danett : that is not SIP.CONF. |
18:15.44 | Danett | sorry |
18:15.46 | Danett | pasted the wrong one |
18:15.49 | Danett | ;) |
18:16.03 | Danett | which part do you want? |
18:16.29 | [TK]D-Fender | Danett : In you domain.tld server you'll need to set allowguest=yes, and set up a context. in extensions.conf you'll create that context containing the names you with to allow and then tell * what to do with it. |
18:16.42 | [TK]D-Fender | Danett : Just follow that and let us know after. |
18:17.06 | Danett | ok |
18:17.08 | Danett | thanks |
18:17.52 | *** join/#asterisk trelane_ (n=trelane@pdpc/supporter/sustaining/trelane) |
18:18.06 | Hmmhesays | 2 more days in the hole |
18:18.07 | Hmmhesays | ugh |
18:18.26 | C4T3l | 2 days?? |
18:18.33 | Danett | ; By default, all domains are accepted and sent to the default context or the |
18:18.33 | Danett | ; context associated with the user/peer placing the call. |
18:18.36 | C4T3l | what you mean? |
18:18.51 | *** part/#asterisk mtaht3 (n=m@c-71-198-23-124.hsd1.ca.comcast.net) |
18:23.26 | *** join/#asterisk stoffell_h (n=stoffell@81.83.249.224) |
18:24.02 | Danett | [TK]D-Fender |
18:24.05 | Danett | done all that |
18:24.10 | Danett | can you try it for me? |
18:24.18 | syzygybsd | how can I just accept all calls over sip? It is complaining that the authentication is bad because of how dial is different then register (ie it thinks the phonenumber being called is the user not extension) |
18:25.27 | Danett | yeah! |
18:25.30 | Danett | working baby |
18:25.32 | Danett | that's nice |
18:27.51 | _-Jon-_ | is it possible to log debug messages to a file? |
18:28.44 | *** join/#asterisk [Airwolf] (n=airwolf@83.98.235.219) |
18:31.07 | *** join/#asterisk Johnnie (n=jdlewis@pdpc/supporter/active/Johnnie) |
18:31.19 | *** join/#asterisk Sixes (n=Cliff@73.Red-217-125-3.staticIP.rima-tde.net) |
18:31.29 | Danett | is there a way to set a maximum amount of minutes/ per month / per peer? |
18:32.16 | Hmmhesays | there is always a way |
18:32.28 | Danett | ;) |
18:32.34 | Hmmhesays | whether you want to write it, or pay to have it done is another question |
18:32.47 | Danett | making it is not the problem |
18:32.48 | stoffell_h | hm, is there no #asterisk irc log any more? |
18:32.57 | Danett | I just thought there was a native way |
18:33.01 | Danett | have to go |
18:33.02 | Danett | bbl |
18:33.52 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
18:34.39 | C4T3l | anybody know what the cause of jittery audio streams through a voip connection. I'll post my sip.conf if needbe. |
18:34.54 | C4T3l | two x-lite phones |
18:35.04 | rpm | ls |
18:35.08 | rpm | erm |
18:35.29 | rpm | are they both over a wan connection? |
18:35.40 | rpm | or on a local network? |
18:35.43 | Lord_Drachenblut | C4T3l, what does the ping time between them look like |
18:36.23 | C4T3l | Ooh, hold on... they are con over a WAN |
18:37.17 | C4T3l | i beleive that icmp-echos are dropped on the other end!! |
18:37.32 | rpm | C4T3l: from what i remember the X-ten X-Lite client does not support much besides g723, alaw, and ulaw.. it will take at least a constant 64kbps on both ends. you can use the jitterbuffer in the sip.conf though and tweak it. |
18:39.35 | [TK]D-Fender | Danett : Sorry, no subdomain routing. if it hits your * it all goes to the same place. Not context control. |
18:40.37 | *** join/#asterisk MacDome (n=eseidel@A17-255-100-181.apple.com) |
18:40.55 | [TK]D-Fender | Danett : You want something more, get a SIP proxy and multiple other * servers. |
18:42.16 | C4T3l | rpm: I cant find jitterbuffer in my sip.conf is there a standard value that should be set??? |
18:42.32 | vader-- | are any of you guys pulling your conf files from mysql? |
18:42.40 | [TK]D-Fender | C4T3l : there IS no jitterbuffer for SIP in * |
18:43.00 | lunk | C4T3l: is the machine slower? |
18:43.02 | C4T3l | I thought that jitterbuffer was an IAX thing |
18:43.08 | [TK]D-Fender | I believe there is a flakey experimental one in the works, but nothing worth trying. |
18:43.10 | tzanger | C4T3l: nope |
18:43.15 | lunk | C4T3l: i have a p2-300 i do testing on, and it's sound quality is soo crappy |
18:43.21 | tzanger | the sip jitter buffer is the same as the IAX2 one |
18:44.12 | justinu | 64kbps is just the g711 payload, you need more like 80kbps to account for ip/udp/rtp overhead |
18:44.38 | C4T3l | the machine is a celeron 2.5 GHz with a gig of ram just sitting at my house |
18:44.47 | C4T3l | nothing really runs on it |
18:45.29 | C4T3l | i'm using a high-speed cable connection and so is the far end |
18:46.02 | *** join/#asterisk VxJasonxV (n=jason@unaffiliated/VxJasonxV) |
18:46.45 | C4T3l | vader--: at work we pull our sip.conf from mysql |
18:47.07 | [TK]D-Fender | tzanger : * needs to add support for preferred codec settings on a per-peer basis like "allow=gsm:40" for 40 MS frams, ets |
18:47.07 | tzanger | yes |
18:47.23 | justinu | wouldn't that be something |
18:47.29 | [TK]D-Fender | tzanger : Major savings ther... |
18:47.34 | [TK]D-Fender | justinu : it would... |
18:48.14 | [TK]D-Fender | justinu : Also nifty for IAX2 as well.... if possible |
18:48.24 | starlein | is that a correct sql structure of the cdr table: ALTER TABLE `cdr` ADD INDEX ( `uniqueid` ) ? |
18:48.28 | [TK]D-Fender | justinu : Depending who gets to negociate. |
18:48.33 | starlein | because on voip-info i saw an INDEX on dst,calldate,accountcode ? |
18:49.17 | *** join/#asterisk Gamercjm (n=chris@pool-71-254-177-36.lsanca.fios.verizon.net) |
18:49.21 | justinu | d-fender: i won't hold my breath |
18:49.32 | starlein | ah i think it should be FULLTEXT |
18:50.57 | [TK]D-Fender | justinu : Whats scary is I don't actually see this is that big a problem, maybe its just me.... but that doesn't necessarily change my odds. |
18:51.43 | justinu | it seems like a lot of "little things" get held up like that |
18:52.04 | *** join/#asterisk caloi (n=caloi@nat-66-218-1-139.usadatanet.com) |
18:53.36 | [TK]D-Fender | justinu : Besides I have other priorities like some Queue basics (bug fixes), SIP-B, and a prayer for Polycom ACD support. |
18:54.17 | starlein | 720 rows in set (2 min 17.79 sec) |
18:54.17 | [TK]D-Fender | justinu : PRI 2BT is up there for a lot of people. The sort of stuff we should concentrate on. |
18:54.19 | starlein | oh damn |
18:54.32 | justinu | i thought it did support 2bct? |
18:54.36 | [TK]D-Fender | Where database performance is anything but! Whe! |
18:54.52 | tzanger | I'd LOVE to use 2BCT |
18:54.56 | [TK]D-Fender | justinu : Maybe I'm just a little out of the loop, or its on different tech |
18:55.22 | justinu | i remember hearing that at astricon, but i have no experience trying to make it happen |
18:55.46 | tzanger | our telco doesn't support it |
18:55.50 | tzanger | at least not to my knowledge |
18:56.03 | justinu | do they support NI2? |
18:56.13 | tzanger | yes |
18:56.19 | tzanger | that's what my switchtype emulation is set for |
18:56.19 | justinu | 2bct is part of NI2, but has to be "provisioned" |
18:56.30 | [TK]D-Fender | What else does NI2 offer over NI1? |
18:56.45 | justinu | um, stuff like ANI-II delivery, iirc |
18:56.50 | Lord_Drachenblut | any good suggestions of a distro of linux to run asterisk on |
18:57.02 | justinu | i can't remember all the differences anymore |
18:57.16 | justinu | i sorta turned my back on legacy telco stuff :) |
18:57.29 | *** join/#asterisk brif8 (n=Techno@lazyjtrainingcenter.com) |
18:57.35 | tzanger | justinu: hmm... does Asterisk provide proper CDR with 2BCT? Or is it similar to the IAX2 media path minimization where there is no callback or even attempt to callback to update CDR? |
18:58.19 | CunningPike | Lord_Drachenblut: Take your pick, really - I think it runs on almost anything. My advice is go with the distro you are most comfortable with |
18:58.38 | justinu | tzanger: proper CDR? lol |
18:58.39 | brif8 | besides the book "Asterisk the future of telephony" what other good source material is there? I have played around with * but when I read the book it made a lot of puzzle pieces fall in place |
18:59.09 | justinu | tzanger: actually, i have no idea... but I have little faith in CDRs WRT asterisk |
18:59.19 | CunningPike | brif8: The Book, The Wiki and The List and This Channel |
18:59.23 | justinu | ~docs |
18:59.24 | jbot | i guess docs is probably Documentation can be found at http://www.digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or http://www.voip-info.org/wiki-Asterisk or http://www.asteriskdocs.org or http://www.oreilly.com/catalog/asterisk or http://www.asteriskguru.com, or http://www.astmasters.net/howtos.html |
18:59.54 | Lord_Drachenblut | CunningPike, bout the answer i was expecting.... was just wanting to make sure that there was one to avoid completly |
19:00.29 | CunningPike | Lord_Drachenblut: Others may chip in here, but I believe SuSE can be a pain, but I have no direct evidence |
19:00.34 | [TK]D-Fender | brif8 : What specific aspects? |
19:02.44 | Hmmhesays | house is such a freaking good show |
19:03.10 | drfoomod2 | is anyone using Asterisk RealTime? |
19:03.43 | Lord_Drachenblut | what is asterisk realtime |
19:04.06 | drfoomod2 | http://www.voip-info.org/wiki/view/Asterisk+RealTime |
19:04.19 | Lord_Drachenblut | there now |
19:04.19 | Lord_Drachenblut | tell me your version |
19:04.23 | Nugget | Lord_Drachenblut: asterisk with a database backend instead of flatfile config files, and a really confusing name. |
19:05.15 | C4T3l | Doesn't Asterisk Realtime allow you to change exensions.conf on-the-fly? |
19:05.31 | *** join/#asterisk pif (n=ldm@zenon.apartia.fr) |
19:05.33 | Nugget | plain asterisk lets you do that too, just less efficiently. |
19:05.53 | pif | I need some guidance on the new jitterbuffer ... |
19:06.02 | tzanger | pif: hunt down oej |
19:06.16 | g__ | Question for anthm, if you're around.. the wiki suggests we avoid using the 'r' flag on Dial().. is that still true? |
19:06.22 | oej | Hunt down zoa instead |
19:06.25 | oej | It's his code, not mine |
19:06.26 | pif | oki |
19:06.28 | tzanger | g__: you NEVER use the 'r' option unless you know WHY you need it |
19:06.38 | anthm | r to generate ring ? |
19:06.39 | tzanger | oej: :-) |
19:06.41 | tzanger | but you run faster |
19:06.44 | g__ | tzanger: because it doesn't work unless I use it? |
19:06.45 | oej | :-) |
19:06.48 | g__ | <PROTECTED> |
19:06.52 | pif | is jitterbuffer=yes enough or must I adjust other paramerters? |
19:06.53 | tzanger | g__: explain "doesn't work" |
19:07.02 | tzanger | anthm: yes. basically replace early audio with ring |
19:07.10 | tzanger | which is only infrequently what you actually want to do |
19:07.10 | file | tzanger: is it working... |
19:07.24 | tzanger | file: not a crash yet... <knocks on file's head> |
19:07.28 | file | sweet |
19:07.34 | anthm | if you are calling a circuit for instance |
19:07.37 | tzanger | file: so what precisely did you do? |
19:07.37 | g__ | I'm dail()ing a sip channel and a ZAP channel with a 30 second timeout. There's no ringing sound unless I use 'r'. |
19:07.41 | anthm | real early media is better |
19:07.46 | anthm | cos you have realtime ring |
19:07.53 | tzanger | anthm: exactly |
19:07.57 | anthm | you can trick ppl with r if it ends up being busy |
19:08.06 | anthm | so you get RRRING RRING BUSY SIGNAL |
19:08.13 | Sixes | Can someone help me with a DTMF problem please? On the Wiki page for the ZyXEL P200W, smeone's written "DMTF relay: outband did the trick" .... I have no clue what he means. :( |
19:08.15 | tzanger | anthm: well your PBX just starts to sound like ass if you hear RING RIBEEP BEEP BEEP |
19:08.18 | tzanger | oh you beat me |
19:08.20 | tzanger | heh |
19:08.27 | *** join/#asterisk insomni (n=insomni@x1-6-00-e0-18-6f-34-ad.k455.webspeed.dk) |
19:08.42 | websae | sixes: i have that phone |
19:09.02 | Sixes | websae: can you make DTMF work? |
19:09.11 | websae | mine works |
19:09.12 | tzanger | g__: what's your Dial command? |
19:09.13 | g__ | Ok, I hear you both tzanger and anthm. so the next question is how do I figure out how why it isn't ringing by itself. |
19:09.28 | tzanger | dialing zap should always give you early audio, even if it's just the zaptel driver giving it to you |
19:09.29 | anthm | what is the path of the call |
19:09.32 | marcus2 | has anyone heard any updates about nufones 888 service? |
19:09.47 | tzanger | marcus2: not I, no |
19:09.53 | marcus2 | sigh |
19:10.08 | tzanger | I've been using my asterlink termination since this happened, haven't turned up nufone's yet |
19:10.19 | marcus2 | now my 888 is in limbo :( |
19:10.25 | tzanger | marcus2: that sucks :-( |
19:10.37 | tzanger | I've always been leery of origination for that very reason |
19:10.43 | Sixes | websae: what setting do you use for DTMF in sip.conf? rfc2833 ? |
19:10.46 | marcus2 | i see that voicepulse connect is offering inbound 888 now |
19:10.49 | tzanger | termination I can flip to whomever by dialing a few digits |
19:10.56 | marcus2 | maybe i should see if i can port it to them |
19:11.16 | g__ | It's basically Dial("SIP/itd371hw&Local/6135554848@itd01-out,30,r").. |
19:11.16 | brad_mssw | i've had the best luck with junctionnetworks ... |
19:11.25 | g__ | Err, without the 'r'. |
19:11.26 | brad_mssw | it takes about 2 days to port an 800/866/888/877 number |
19:11.46 | tzanger | ok so it's SIP and Local/ not Zap/ |
19:11.56 | brad_mssw | glad we didn't go with nufone here :) |
19:11.57 | anthm | what is the path like phone->sip->ast->zap etc |
19:12.01 | g__ | Local/ does eventually call Zap/ though. |
19:12.03 | *** join/#asterisk ringhals (i=fwuser@firewall.drgutah.com) |
19:13.19 | g__ | The path is ast->sip and ast->Local/6135554848@itd01-out->Zap/g1/6135554848 |
19:13.35 | ringhals | Ok .. new to the irc thing.. is this a channel that I can ask questions about configuration etc in? |
19:13.37 | anthm | what about into ast |
19:13.39 | anthm | what is that |
19:14.12 | g__ | anthm: you mean the channel that placed the call? It's a SIP client. so sip->ast. |
19:14.17 | lzhang | anybody ever use app_machinedetect? |
19:14.18 | justinu | ~suggestions |
19:14.20 | jbot | it has been said that suggestions is 1) Don't ask to ask. Just say your problem, 2) Don't repeat until 5 mins after, 3) Read and re-read the docs first, then admit it if you REALLY don't understand. You're wasting your time and ours if you haven't at least tried. 4) If your problem ain't solved, come back in 12 hrs or 24 hrs later. We're very international. 5) Be ... |
19:15.01 | anthm | if you make an exten that you can call with the sip phone that just goes to zap as a test does that work? |
19:15.45 | g__ | Yes. Actually, the scenario I described above works for some people but not for others. |
19:17.05 | anthm | it may be because of the &'ed together channels |
19:17.10 | *** join/#asterisk Johnnie (n=jdlewis@pdpc/supporter/active/Johnnie) |
19:17.15 | anthm | cos in that case you are not one on one with 1 channel |
19:17.23 | anthm | you are waiting for one of them to answer first |
19:17.32 | syzygybsd | SOB! |
19:17.45 | g__ | So what is the expected behaviour for multiple channels? |
19:17.52 | syzygybsd | May 2 13:08:31 NOTICE[24932]: chan_sip.c:8727 reload_config: Unable to load config sip.conf, SIP disabled |
19:17.55 | CunningPike | ringhals: Shoot |
19:18.07 | syzygybsd | when I get that line, it works, when I fix the configuration it doesnt' |
19:18.08 | anthm | sit there in silence till one or the other answers |
19:18.45 | g__ | And the call progress information--just don't pass it on at all? |
19:18.46 | anthm | that would for sure be a case where you have no choice but to provide ringback or something or other |
19:18.48 | *** join/#asterisk gursikh (n=guriskh1@adsl-68-93-88-61.dsl.hstntx.swbell.net) |
19:18.52 | g__ | (Because sometimes it does.) |
19:19.51 | anthm | what is the reason for calling sip and zap at the same time just for optional answering in 2 locations |
19:20.01 | *** join/#asterisk jjwx (n=samuel@levinux.UQAR.UQUEBEC.CA) |
19:20.03 | jjwx | hi |
19:20.26 | ringhals | Ok.. I have been trying to get chanspy to work for me with little success. I have a number of devices set up and working. For example I have an iax device and a sip device set up. I can place and receive calls from both. I also have the extension exten => 8102,1,Chanspy(SIP|g) however I get no audio and the Chanspy commands do not seem to work. So to clarify I place a call from the sip device. then from the iax device dial the spy extension, if I |
19:20.42 | g__ | anthm: basically. It's a personal extension with a hard phone and a cell phone. |
19:20.51 | anthm | maybe you can use the "please hold while i try that extension" file |
19:20.55 | g__ | (err, mobile phone--depending on which English you speak.) |
19:21.00 | anthm | or the m opt for hold music |
19:21.18 | anthm | doing r will seem ok but it will make no sense if you end up with a busy tone |
19:21.23 | jjwx | is it possible to do a regex in asterisk? I mean, something like Set(BLEH = /SIP:(.?)@.*/) |
19:21.32 | *** join/#asterisk gr0mit_home (n=wendolen@extrt.txrx.org.uk) |
19:21.58 | g__ | Ok, I'll be sure not to send them a busy tone then. I've been opting for a "no one is available to take you call at the moment.. please try your call again later" for extensions without voicemail. |
19:22.02 | anthm | i recall making the regex diaplan function dont know if it's still there |
19:22.43 | jjwx | shit man, regex is a must... |
19:22.45 | g__ | Thanks anthm. I'll add a couple of notes to the voip-wiki documentation about this. |
19:23.48 | anthm | yah i agree I made a perl compatable regex mod for another project i am working on |
19:23.59 | schirpich | what would the correct terminology be for the recording you hear on the IVR when you call into a queue |
19:24.15 | Zodiacal | anyone know how i can create a .call file to use for testing? i tried to make one, and it works, but it hangs up after like 20 seconds.. anyone know why it doesn't stay connected longer? http://pastebin.com/694770 |
19:24.24 | justinu | g__: what are you gonna do when your cell phone voicemail picks up? |
19:24.58 | anthm | yah listen to justin i think he is well aware of this snafu =p |
19:25.03 | justinu | ;) |
19:25.22 | justinu | i had to write a patch to AMI, and a manager app to make it do what I want |
19:25.36 | dpryo | Is it possible to detect the "referer" from a remotely transfered call? (Say, detect that the call has been transfered from my cellphone-number) |
19:25.46 | *** join/#asterisk generalhan (i=general_@ip67-90-64-2.z64-90-67.customer.algx.net) |
19:25.49 | generalhan | whats up all ! |
19:26.16 | justinu | dpyro: if you have PRI, it should be available |
19:26.36 | dpryo | justinu: Do you know which parameter? I have an E1. |
19:26.36 | anthm | and if it's within the pbx you can save it in a variable |
19:26.40 | justinu | RDNIS? |
19:27.01 | justinu | http://www.voip-info.org/wiki/view/RDNIS |
19:27.01 | dpryo | Okay, thanks. I'll look into it :) |
19:27.06 | dpryo | Yeah, looking at it :) |
19:27.20 | vader-- | can you pass an extension into the voicemailmain app? |
19:27.21 | Zodiacal | any ideas? |
19:27.22 | dpryo | Would be cool to use it for a voicemail solution. |
19:27.27 | vader-- | so you don't have to enter your extension |
19:27.33 | justinu | of course, vader |
19:27.43 | noname32 | anyone here use the Grandstream 2000? |
19:27.45 | justinu | VoiceMailMain(${CALLERIDNUM}) |
19:28.24 | drfoomod2 | has anyone suggested using SQLite> |
19:28.24 | drfoomod2 | ? |
19:29.05 | drfoomod2 | oh?> |
19:29.14 | vader-- | justinu thanks |
19:29.18 | justinu | np |
19:29.20 | Hmmhesays | What is more annoying that people that don't follow your precise instructions? |
19:29.23 | pif | _must_ I use trunktimestamps for jitterbuffer to work? |
19:29.44 | Hmmhesays | People that fail to follow your precise instructions repeatedly |
19:29.51 | drfoomod2 | anthm: what is being stored in/with sqlite>? |
19:30.14 | *** join/#asterisk PakiPenguin (n=Junaid@linuxpakistan/admin/pakipenguin) |
19:30.24 | PakiPenguin | ~seen areski |
19:30.33 | jbot | areski <n=areski@polar.es6.egwn.net> was last seen on IRC in channel #asterisk, 40d 9h 24m 8s ago, saying: 'the real billable seconds'. |
19:30.36 | g__ | justinu: we don't have voicemail on the company cell phones |
19:30.48 | justinu | so they just ring forever? |
19:31.05 | anthm | it can store config, be a dialplan switch, umm i forget what else |
19:31.10 | anthm | an app i think |
19:31.18 | anthm | to do sql st |
19:31.21 | g__ | Yup. For values of "forever" that are more than 30 seconds, anyways. |
19:31.25 | anthm | another one to select a row in to vars |
19:31.35 | justinu | g__: lucky you then |
19:31.50 | PakiPenguin | justinu, there's a "No Answer" tone after a certain number of secods |
19:31.54 | PakiPenguin | which actually shows the |
19:32.03 | PakiPenguin | "no answer" text on your cell screen |
19:32.05 | g__ | But on my personal voicemail, I've been using the zap "push-# after picking up the phone" feature. |
19:32.06 | justinu | here the crooks answer the call regardless |
19:32.15 | ringhals | ~seen jsmith |
19:32.17 | jbot | jsmith is currently on #asterisk-doc #utah. Has said a total of 48 messages. Is idling for 8m 51s, last said: 'Wow... looks like all this talk about documentation has more people in here!'. |
19:32.23 | justinu | g__: what's that? |
19:33.09 | anthm | and cdr i think |
19:33.10 | g__ | justinu, look up option 'c' in http://www.voip-info.org/wiki/index.php?page=Asterisk+Zap+channels |
19:33.52 | justinu | g__: interesting... didn't know about that... doesn't help me tho, since I don't use zap channels |
19:33.59 | g__ | PakiPenguin: is that only when you place the call from your cell phone? |
19:34.24 | PakiPenguin | nope , from the pots line , i get a no answer tone |
19:34.30 | PakiPenguin | which is different from everything else |
19:34.37 | PakiPenguin | but i think its a exchange dependent thingy |
19:34.38 | g__ | justinu: yeah, I'm going to run into trouble if we switch from zap channels. It'd be nice to build it into a Local/ or Dial() instead. |
19:34.51 | justinu | g__: won't be easy |
19:35.00 | g__ | I'm not volunteering.. |
19:35.06 | g__ | not now, anyways. |
19:35.18 | justinu | i'm just warning you |
19:36.05 | g__ | But if you want to do it for me.. I'd be more than happy to help you test it. |
19:36.21 | [TK]D-Fender | semi-lazy question : Can the AMI be used to place calls in a similar manner to .call files? |
19:36.30 | justinu | i've already done it in a technology agnostic way, but it's not very elegent |
19:36.33 | *** part/#asterisk Sixes (n=Cliff@73.Red-217-125-3.staticIP.rima-tde.net) |
19:36.41 | justinu | [TK]D-Fender: yes |
19:37.07 | *** join/#asterisk ph|ber (i=phiber@slackwaresupport.com) |
19:37.12 | [TK]D-Fender | justinu : Neato.... going to have to start playing around with this stuff... |
19:37.34 | ph|ber | anyone have an example of multiple DID's that route to different extensions?? i cant seem to get this one working. |
19:37.48 | justinu | the originate action is somewhat flawed, imo, but I fixed that too |
19:38.44 | [TK]D-Fender | ph|ber : What are the calls coming in on? (interface) |
19:38.47 | anthm | along with the concept of actions and the entire manager interface itself. |
19:38.54 | justinu | heh |
19:39.27 | ph|ber | zap |
19:39.37 | justinu | pots line? |
19:39.48 | justinu | anthm: i can only imagine |
19:39.58 | *** join/#asterisk [hC] (n=hardcore@mail.metrobridge.com) |
19:40.06 | anthm | i did manage to add some patches to it eventually |
19:40.25 | justinu | why isn't the simplest of all apps "app_bridge" in the standard asterisk disto? |
19:40.26 | anthm | like the one that lets you turn events off so you can have a command only pipe |
19:40.35 | justinu | isn't that the first thing you write when you write a switch? |
19:41.24 | dwildes2 | ph|ber for Zap lines, you'll have to create a context for each incoming channel |
19:42.09 | ringhals | does anyone know of a good write up on the chanspy app? I have been over http://www.voip-info.org/wiki-Asterisk+cmd+ChanSpy and the info from show application chanspy but am still a bit lost.. not as to setting up the extension but as to how to debug the fact I get no sound when using it. |
19:42.10 | anthm | i can only guess but i'd say the first thing would be a solid core |
19:42.19 | [TK]D-Fender | ph|ber : Can you be a little more precise... |
19:42.20 | ph|ber | wait, this is a iax connection |
19:42.20 | ph|ber | sorry |
19:42.43 | [TK]D-Fender | ph|ber : Depends on how your provider works. |
19:43.01 | anthm | ringhals at one point I could have since I invented it but i have no idea what they have done to it by now |
19:43.02 | [TK]D-Fender | ph|ber : Its possible you could use a single registration for multiple DID's or that each could be its own. |
19:43.20 | *** join/#asterisk Renacor (n=kvirc@ip21.farheap.net) |
19:43.23 | ph|ber | atm it is dialing all extensions with exten => s,1,dial(SIP/0960&SIP/0961&SIP/0962&SIP/0963&SIP/0978|20|tr) |
19:43.27 | *** join/#asterisk R3DB0x (i=nobody@66.142.28.36) |
19:43.37 | ringhals | bummer ... it is kicking my butt |
19:43.41 | dwildes2 | ph|ber, are they passing the DID? |
19:43.55 | ph|ber | the extensions are relative to the last 4 of the #.. ie 555-555-1234 should go to extension 1234 |
19:43.58 | *** join/#asterisk fjean (n=fjean@201.29.130.118) |
19:44.11 | ph|ber | yes. |
19:44.16 | justinu | how, pri? |
19:44.20 | ph|ber | and i tried exten => 18043430978,2,Dial(SIP/0978,20,m) |
19:44.23 | ph|ber | doesnt work. |
19:44.40 | justinu | they might be sending you 4 digit dnis only |
19:44.43 | justinu | or 10 digit |
19:44.43 | justinu | or 7 |
19:45.00 | *** join/#asterisk MacDome (n=eseidel@A17-255-100-181.apple.com) |
19:45.04 | ph|ber | <PROTECTED> |
19:45.08 | ph|ber | ahh |
19:45.11 | dwildes2 | try putting a underscore: |
19:45.16 | dwildes2 | _18043430978 |
19:45.46 | justinu | so they're sending you 7 digit dnis |
19:45.49 | justinu | there's your answer |
19:45.50 | dwildes2 | you can remove the 's' extension and see if you get a reject. the reject should have the DID as they are passing it to you |
19:46.08 | justinu | exten => 3430978 ... |
19:46.41 | *** join/#asterisk MacDome (n=eseidel@A17-255-100-181.apple.com) |
19:46.42 | dwildes2 | justinu, yeah - looks that way :) |
19:46.49 | ph|ber | this is what happens. |
19:47.04 | ph|ber | http://pastebin.com/694842 |
19:47.59 | justinu | fucking amp |
19:48.05 | justinu | ~amp |
19:48.07 | jbot | i guess amp is "amp is, like, NOT supported here! people using it should join #freepbx (the new name of amp)" |
19:48.07 | anthm | ringhals, if they still preserved the basic functionality you shuld be able to make exten => 1000,Chanspy and when you call 1000 amd every time you press * you will cycle active channels |
19:48.55 | [TK]D-Fender | justinu : Hello Mr. Bile! |
19:48.59 | justinu | ;) |
19:49.05 | justinu | just following the trend |
19:49.21 | justinu | ph|ber: the reason it doesn't work is because of the goto(menu,s,1) |
19:49.43 | ph|ber | ? |
19:49.56 | fjean | hi everybody , anyone can give me a hint on troubleshooting zaptel / ztdummy installation.... |
19:50.19 | ringhals | anthm thats basically what I have done. however for whatever reason there is no audio. I have open sip and iax channels but unless I ommit the |g I get nothing at all |
19:50.58 | anthm | what do you have in the g() |
19:51.27 | ph|ber | ;exten => s,1,Goto(menu|s|1) |
19:51.28 | ph|ber | that ? |
19:51.39 | ringhals | looking |
19:51.41 | justinu | yeah, it's throwing out the DID info |
19:51.47 | ph|ber | it is commented out. |
19:51.55 | ph|ber | ; |
19:52.00 | justinu | so then why do lines 13 and 14 print out in your pastebin? |
19:52.19 | [TK]D-Fender | justinu : I try to leave off the colourful expletives... its poorly received news as it is... |
19:52.20 | ph|ber | after it does not answer it rings all extensions |
19:52.40 | anthm | if you add g(wazzup) to it then you need Set(SPYGROUP=wazzup) at the top of every extension you want to spy on |
19:52.50 | *** join/#asterisk insomni (n=insomni@x1-6-00-e0-18-6f-34-ad.k455.webspeed.dk) |
19:53.38 | ph|ber | exten => _X.,7,SetVar(FROMEXTEN=${DID${EXTEN}}) |
19:53.38 | ph|ber | exten => _X.,8,Goto(extensions|${DID${EXTEN}}|1) |
19:53.38 | ph|ber | exten => _X.,10,Goto(menu|s|1) |
19:53.54 | *** join/#asterisk IceManRISK (n=kart@201-40-207-74.mganm702.dsl.brasiltelecom.net.br) |
19:53.56 | ringhals | I think I missed that actually..I did the g(29) but not the SPYGROUP |
19:54.08 | brookshire | _X. is bad! |
19:54.11 | ringhals | I missunderstood what I read LOL |
19:54.18 | anthm | yah you need to set the var |
19:54.36 | ph|ber | im walking into a previously install |
19:54.51 | anthm | and you can make it like _88XX to match g(${EXTEN:2}) so you can dial 8829 to spy on group 29 |
19:54.52 | justinu | ph|ber: ask on #freepbx, they're more familiar with that stuff |
19:55.32 | ringhals | I will try that |
19:55.36 | ph|ber | hrm this is 1.2.4 how old is that |
19:55.38 | *** join/#asterisk ToTo (n=ToTo@host128-207.pool879.interbusiness.it) |
19:55.42 | justinu | recent enough |
19:55.48 | ph|ber | nm. found it.. 2/06 |
19:56.02 | [TK]D-Fender | anthm : We need multiple spy/ring/other groups pe entry! |
19:56.23 | justinu | ph|ber: problem is that AMP/FreePBX created that dialplan |
19:56.35 | ph|ber | ahh. |
19:56.44 | ph|ber | i think it has AMP installed. |
19:56.45 | anthm | would be not too hard |
19:56.48 | *** join/#asterisk klasstek (n=nunyobiz@ip67-154-143-190.z143-154-67.customer.algx.net) |
19:57.00 | ph|ber | whats the default uname and pass to AMP?? |
19:57.03 | ph|ber | admin and amp111 |
19:57.17 | justinu | ~amp |
19:57.18 | jbot | rumour has it, amp is "amp is, like, NOT supported here! people using it should join #freepbx (the new name of amp)" |
19:57.32 | ph|ber | yea yea |
19:59.26 | *** join/#asterisk mtaht3 (n=m@reserve-64-79-114-30.wiline.com) |
19:59.26 | fjean | any reasons why udev would not create the zap devices in /dev.... ? with no hardware or such |
19:59.37 | *** join/#asterisk Gunnar (n=gunnar@34.80-203-209.nextgentel.com) |
20:00.36 | ringhals | so I tried to make it more simple.. I took out the |g etc and set my extension to Chanspy(SIP) placed the call from my sipphone and then dialed the extension for Chanspy from the iax device.. still no audio and a show channels does not show me bridging to anything.. any suggestions? |
20:01.43 | anthm | (SIP) means only spy on sip channels |
20:01.55 | anthm | just put nothing in the () |
20:02.01 | anthm | and you will get everything |
20:02.11 | ringhals | kk I will try that next |
20:02.41 | *** join/#asterisk brockj49464 (n=brockj49@41.105.dhcp.hope.edu) |
20:02.44 | *** join/#asterisk techie (n=gus@antibala.com) |
20:03.04 | justinu | anthm: you taking a break or something? not used to seeing you in these parts ;) |
20:03.26 | Aurs | fjean: checked README.udev? |
20:04.10 | fjean | Aurs: yes...I did it the same way for the two files, permissions.d and rules.d |
20:05.03 | Aurs | fjean: ok |
20:05.05 | fjean | Aurs: I am using mandrake 10.1, I did the installation already four times on different computers using the same distro |
20:05.32 | fjean | Aurs: but this time, it does not want to :- ) |
20:06.37 | Aurs | sounds nice ;) |
20:06.57 | *** join/#asterisk xunil (n=wkurdzio@office1.visionpointsystems.com) |
20:07.15 | fjean | Aurs: hey I noticed my computer is a 686...dual CPU but actually just one |
20:07.27 | *** join/#asterisk ToTo (n=ToTo@host128-207.pool879.interbusiness.it) |
20:08.05 | justinu | 1337 |
20:08.43 | brif8 | ~docs |
20:08.44 | jbot | hmm... docs is probably Documentation can be found at http://www.digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or http://www.voip-info.org/wiki-Asterisk or http://www.asteriskdocs.org or http://www.oreilly.com/catalog/asterisk or http://www.asteriskguru.com, or http://www.astmasters.net/howtos.html |
20:10.38 | stoffell_h | is it allowed to use irclog to log the channel to a webpage? |
20:10.40 | *** join/#asterisk Tangent (n=Tangent@connerdata.plus.com) |
20:11.03 | dlynes_ | zaptel seems to autconfigure udev under zaptel 2.6.13 and higher, anyways...2.6.12 and earlier it seems you have to manually configure it |
20:11.22 | dlynes_ | or, something to that effect |
20:11.39 | dlynes_ | I've seen a number of people with 2.6.9 running into troubles with udev and zaptel |
20:11.59 | *** join/#asterisk Eight (n=blake@12-227-169-99.client.mchsi.com) |
20:12.06 | fjean | dlynes: ok, cool |
20:13.11 | sevard | Riddle me this * gurus. I have an Aastra 480i CT. I could make calls out but no audio. I restarted *, no go, I restarted *, no go -- desperation, I restarted *, everything magically works. |
20:13.29 | fjean | dlynes: i'll take a look |
20:13.34 | sevard | Am I safe to assume * reliability == windows or would this be a configuration issue |
20:13.50 | dlynes_ | fjean: which version of asterisk are you using? |
20:13.59 | dlynes_ | erm |
20:14.04 | dlynes_ | sevard, i mean |
20:14.18 | sevard | dlynes_ Asterisk 1.2.5 |
20:14.26 | gursikh | Just wondering, what distro do you guys recomend to run * on? |
20:14.30 | dlynes_ | sevard: upgrade to 1.2.7 or higher |
20:14.40 | websae | fedora core here :) |
20:14.40 | dlynes_ | sevard: You'll find the sip subscriptions work infinitely better |
20:14.41 | sevard | gursikh: I run on slackware. I recommend it for everything. |
20:14.50 | dlynes_ | Slackware here, too |
20:14.51 | websae | debian works nice as well |
20:15.07 | gursikh | oh websae, is that what you would recomend? |
20:15.11 | sevard | dlynes_: When I upgrade is there any sort of configuration changes between these versions? |
20:15.19 | dlynes_ | From what I hear though, Gentoo has some special tweaks for Asterisk, too |
20:15.35 | dlynes_ | but I'm not into running source code based distros for a telephony server |
20:15.46 | *** join/#asterisk clive- (n=pirch@dsl-145-52-204.telkomadsl.co.za) |
20:15.50 | *** join/#asterisk Ariel_ (n=Ariel@70.46.87.158) |
20:15.52 | dlynes_ | don't want to have to do compiles on my main telephony server, degrading call quality for the users |
20:15.56 | websae | dlynes: I had heard that as well |
20:16.03 | dlynes_ | sevard: no |
20:16.17 | tzanger | dlynes_: ever hear of 'nice' :-) |
20:16.17 | dlynes_ | websae: yeah...gentoo also has a gnu autoconfigure script for asterisk |
20:16.18 | Ikarus | dlynes_: "special tweaks" ofcourse, the same ones like -funroll-loops ? |
20:16.26 | Hmmhesays | who can we get on the case, we need peerrrrryyyyy maaaaasssssssson |
20:16.26 | sevard | Am I safe to assume upgrades are fairly painless as long as I put my builds into packages? |
20:16.36 | DoktorGreg | I recently switched from slack to debian |
20:16.42 | Hmmhesays | for you sevard? no |
20:16.44 | *** join/#asterisk [hC] (n=hardcore@mail.metrobridge.com) |
20:16.45 | dlynes_ | tzanger: still...i don't like the possibility :) |
20:16.48 | *** join/#asterisk klasstek (n=nunyobiz@ip67-154-143-190.z143-154-67.customer.algx.net) |
20:16.57 | dlynes_ | sevard: extremely painless then |
20:17.19 | dlynes_ | Hmmhesays: eh? I use slackware, put everything for asterisk into packages...it's super easy |
20:18.13 | dlynes_ | sevard: You might want to try installing a 2.6 kernel too, if you haven't done so already |
20:18.47 | dlynes_ | sevard: 2.6 has low latency optimizations and ztdummy is automatically compiled for it |
20:19.19 | Hmmhesays | dylnes i was using something not common in the geek wordl |
20:19.21 | Hmmhesays | humour |
20:20.04 | tzanger | Hmmhesays: ? |
20:20.09 | Hmmhesays | haha |
20:20.26 | fjean | could msec have an impact on the installation ? |
20:20.41 | Danett | Does anyone know a provider which sells sip up? |
20:20.44 | clive- | fjean hardly imho |
20:20.49 | fjean | k |
20:20.56 | Danett | like a local telefone number which endpoint is a sip user agent |
20:21.05 | sevard | dlynes_: I'm alreadying running 2.6 and I hav ea zaptel device |
20:21.40 | blitzrage | hrmmm... still can't seem to get qualify to work as expected -- peers keep going REACHABLE then UNREACHABLE |
20:21.51 | dlynes_ | Hmmhesays: Well, it usually helps if it's obvious |
20:22.07 | dlynes_ | blitzrage: that's normal |
20:22.14 | dlynes_ | blitzrage: as long as they become reachable again |
20:22.33 | blitzrage | dlynes_: but when it goes unreachable, asterisk doesn't even attempt to place a call to that end point |
20:22.50 | blitzrage | I'd hardly say that is ideal |
20:23.04 | dlynes_ | blitzrage: so your sip devices don't become reachable again? |
20:23.04 | blitzrage | and I realize its "normal" its been doing it since I started using Asterisk 3 years ago :) |
20:23.19 | file | what are you babbling about now blitzrage |
20:23.19 | blitzrage | dlynes_: yes, they do -- but during that interum you can't send calls to that device |
20:23.25 | blitzrage | file: qualify! :) |
20:23.32 | file | what about it1 |
20:23.33 | dlynes_ | blitzrage: how long are they unreachable for? |
20:23.33 | file | ! |
20:23.45 | sevard | dlynes_: Do you ever have any issues with DTMF and cellphones? |
20:23.48 | syzygybsd | does anyone have a working configuration for connecting 2 asterisk boxes? Anytime I try to send calls between them I get a "Failed to authenticate user ${phonenumberdialed}" |
20:24.10 | syzygybsd | *sip configuration |
20:24.17 | blitzrage | file: peers go REACHABLE then UNREACHABLE and back again -- causing some calls to be sent to voicemail when the peer is actualyl reachable (although asterisk doesn't see it that way :)) |
20:24.17 | dlynes_ | sevard: don't know offhand |
20:24.25 | dlynes_ | sevard: define 'issues' |
20:24.43 | file | blitzrage: ah, stupid SIP device that gives lesser priority to OPTIONS so it doesn't respond in time and thus Asterisk thinks it isn't there? |
20:24.54 | sevard | * thinks I'm entering 501 or 01 when I try entering 5001 |
20:24.55 | blitzrage | file: seems that way --- polycom's mostly |
20:25.10 | blitzrage | file: although it happens with other Astersk boxes too :) |
20:25.13 | file | can't say my Polycom does that option |
20:25.15 | file | er often |
20:25.37 | dlynes_ | sevard: Do you answer, then wait a bit, then do the ivr? |
20:25.37 | file | therefore you must be a witch |
20:25.49 | sevard | dlynes_: of course. |
20:26.04 | blitzrage | file: :D |
20:26.28 | sevard | dlynes_: answer, wait 2, ivr asks for input, you enter in 5001, it says Sorry, that's an invalid extension. |
20:27.11 | dlynes_ | sevard: weird...never had that problem |
20:27.33 | dlynes_ | sevard: I think I might have had someone complain about that happening once or twice |
20:27.39 | dlynes_ | sevard: but it's very rare |
20:27.56 | sevard | dlynes_: I have sipura 2002 ATAs going into a TDM400P that might have something to do with it |
20:28.02 | dlynes_ | sevard: is this on a zaptel channel, or a sip channel? |
20:28.09 | *** join/#asterisk [Airwolf] (n=airwolf@83.98.235.219) |
20:28.42 | sevard | dlynes_: a zaptel channel |
20:29.45 | *** part/#asterisk Lord_Drachenblut (n=Lord@12-210-117-62.client.insightBB.com) |
20:31.02 | *** join/#asterisk dsfr (n=dsfr@pdpc/sponsor/digium/dsfr) |
20:31.57 | dlynes_ | sevard: try adjusting your gain |
20:33.57 | *** join/#asterisk Skarmeth (n=Skarmeth@201009033182.user.veloxzone.com.br) |
20:34.01 | Skarmeth | hi all |
20:34.10 | *** join/#asterisk flop110 (n=ozverenm@162.27.103-84.rev.gaoland.net) |
20:34.14 | flop110 | hello all |
20:34.31 | flop110 | i have question about DACS modes on zaptel cards |
20:35.21 | Skarmeth | I'm in trouble with a TE110P and a TDM04B. When I had only the TDM04B, everything worked fine, I was able to receive and make calls. But after installing a TE110P, I cannot receive calls or make calls anymore |
20:36.12 | Skarmeth | I've a span configured for TE110P as 1-10 for b channels and 16 for d channel, and TDM04B card configured from 32-35 as fxsks |
20:36.30 | OloBola | what do you think a good per minute price is for 800 origination? 3ish? |
20:36.52 | Skarmeth | at zapata.conf, all needed parameters, but when I make a call to a line pluged at TDM04B, I can't see no action on the console screen |
20:37.36 | Skarmeth | any idea? /proc/zaptel/1 tell me it is a TE110P (as I put 1-10 for it) and 2 as a TDM04B (as I put 32-35 for it) |
20:39.17 | syzygybsd | wow, that took me way too long to figure out |
20:39.20 | *** join/#asterisk heison (n=heison@ns.somanetworks.com) |
20:40.00 | syzygybsd | if you have a register => in sip.conf, don't have type = friend for that same connection, they will conflict and not allow incomming calls |
20:40.28 | *** join/#asterisk Druken (n=Druken@CPE00121716da99-CM00159a090acc.cpe.net.cable.rogers.com) |
20:40.45 | Hmmhesays | why is brian setzer such a damn good guitarist |
20:41.11 | *** join/#asterisk StevenL (n=steve@216.62.85.65) |
20:41.15 | clive- | prolly because he works hard ata it:) |
20:42.47 | *** join/#asterisk RoyK (n=roy@ti211310a080-13754.bb.online.no) |
20:42.49 | sevard | dlynes_: the gain on my zaptel card? |
20:43.07 | dlynes_ | correct |
20:43.11 | sevard | clive-: you've apparently typed ata too many times |
20:43.21 | sevard | dlynes_: is that in zaptel.conf or in zapata.conf |
20:43.33 | dlynes_ | sevard: zapata.conf |
20:43.46 | dlynes_ | sevard: rxgain=, txgain= |
20:43.56 | sevard | should I play with my gain or do you have suggested values |
20:43.58 | dlynes_ | sevard: the values can be between -7 (minimum) and +7 maximimum |
20:44.08 | dlynes_ | sevard: I'd suggest starting with 0, and adjust it from there |
20:44.18 | sevard | really |
20:44.24 | sevard | Hmmhesays: a maximum of 7 |
20:44.31 | dlynes_ | s/maximimum/maximum |
20:44.32 | sevard | Hmmhesays: not 12.0 |
20:44.32 | *** join/#asterisk Dr-Linux (n=Linux@202.59.73.131) |
20:44.40 | Hmmhesays | wtf are you talking about |
20:44.50 | sevard | rxgain=12.0 |
20:44.50 | sevard | txgain=3.0 |
20:44.55 | Danett | Anyone knows a international carrier which has sip connectivity? |
20:45.01 | Hmmhesays | haha thats right, quintums gains are max 12 |
20:45.02 | Hmmhesays | my bad |
20:45.06 | Danett | like a reliable one for professional use |
20:45.12 | Hmmhesays | sevard I found you a russian bride |
20:45.16 | Hmmhesays | on www.bride.ru |
20:45.17 | sevard | Hmmhesays: sweet! |
20:45.24 | Hmmhesays | she kind of has a mustache |
20:45.31 | sevard | sweet! |
20:45.49 | dlynes_ | sevard: that's why your dtmf tones aren't working |
20:45.57 | dlynes_ | sevard: the dtmf is all distorted like shit :) |
20:46.20 | Hmmhesays | and her head is disproportionate to her body |
20:46.26 | Hmmhesays | http://www.bride.ru/ph/htcgi/ladies/461/461229P1.html |
20:46.27 | Dr-Linux | hi folks |
20:46.50 | Dr-Linux | anybody expert with "call by name directory" ? |
20:47.15 | Nugget | I turned it on and it worked. Does that make me an expert? |
20:47.30 | dlynes_ | Nugget: no doubt...never had any problems myself, either :) |
20:47.40 | Nugget | how about if I'm not an expert but I do know the answer to the question you haven't asked yet. Would that help? |
20:47.45 | Nugget | or would you rather wait for an expert? |
20:47.46 | Hmmhesays | Dr-Linux |
20:47.58 | Hmmhesays | cmd directory() |
20:48.09 | Dr-Linux | opss |
20:48.32 | Dr-Linux | well, im thinking how can i ask my question, bcoz of damn english |
20:48.33 | Dr-Linux | :P |
20:48.35 | Dr-Linux | well, |
20:48.57 | Hmmhesays | I don't think i'm going to find a sugar momma on this site |
20:49.18 | Dr-Linux | i want, when call routes to the caller, system should say " call is going to abc at extension xyz" ? |
20:49.19 | Druken | Hmmhesays: http://www.bride.ru/ph/htcgi/ladies/461/461655P1.html |
20:49.22 | Hmmhesays | http://bride.ru/ph/phcgi/ladies/457/457037P1.jpg this one kinda looks like she shiat herself |
20:49.22 | Dr-Linux | something like that |
20:49.37 | Hmmhesays | Dr-Linux: check out the directory app |
20:49.39 | noname32 | anyone here use the polycom phones? |
20:49.40 | Hmmhesays | does exactly that |
20:50.06 | Dr-Linux | Hmmhesays: what's the best way to have this feature? |
20:50.19 | Hmmhesays | go to the wiki and read about "cmd directory" |
20:50.30 | Hmmhesays | Druken: she's got kind of a chubby face for being 106lbs |
20:50.39 | Druken | Dr-Linux: when your people record their name, it should be <name> <extension> |
20:50.43 | sevard | Awesome |
20:51.05 | Druken | Hmmhesays: very true, but i'd still give her something to put in it :) |
20:51.13 | Hmmhesays | yeah a freaking sammich |
20:51.14 | znoG | chan_sip.c:1210 retrans_pkt: Maximum retries exceeded on transmission 50f2fa9234600011222e38ec220cad6f@192.168.136.67 for seqno 102 (Critical Request) |
20:51.16 | sevard | DTMF works flawlessly |
20:51.21 | flop110 | on a PRI link, if i place one double E1 between my PBX and my telco and if i read audio buffer in userland and |
20:51.22 | znoG | anyone know what would cause that? |
20:51.24 | sevard | I love all of you. |
20:51.29 | znoG | packets being sent to 192.168.136.67 but not getting back? |
20:51.39 | *** part/#asterisk StevenL (n=steve@216.62.85.65) |
20:51.49 | *** join/#asterisk riksta (n=rick@62.6.163.87) |
20:52.06 | flop110 | write these audio buffers on HDLC drivers, its will have a repercution on performances ? |
20:53.27 | Hmmhesays | geebus i swear these girls don't eat |
20:54.17 | dlynes_ | znoG: probably a conversation that's long since ended, but for whatever reason asterisk still insists on sending sip packets to ti |
20:54.22 | *** join/#asterisk papo (n=mathias@adsl-177-161-fixip.tiscali.ch) |
20:54.39 | dlynes_ | znoG: I see that regularly in the debug logs |
20:55.38 | znoG | dlynes_: whats happening is that when I call in via Zap, it should dial SIP/3000 and on the calling end, i hear a busy tone. On the asterisk side, i see that maximum transmission message that I just pasted. |
20:56.08 | dlynes_ | znoG: ah...there's probably other errors as well |
20:56.26 | dlynes_ | znoG: but you're probably getting spammed so much with the retrans_pkt messages that you don't see them |
20:56.50 | znoG | thats the thing, I debugged the peer IP where the SIP extensions sits and there's no clear error message or stuff like that |
20:57.11 | znoG | and if i call SIP->SIP, its fine... only when coming in via Zap doesn't work |
20:57.15 | dlynes_ | znoG: what's your logger.conf setting for that log file? |
20:57.32 | dlynes_ | znoG: I get the feeling it's your dialplan for zap |
20:57.52 | znoG | dlynes_: but i can see in the asterisk console its trying to dial the SIP extension just fine |
20:58.02 | dlynes_ | znoG: Oh |
20:58.07 | znoG | and as soon as it starts to ring, the SIP extension keeps ringing and the Zap channel drops |
20:58.14 | *** join/#asterisk Dr-Linux (n=Linux@202.59.73.131) |
20:58.31 | Dr-Linux | sorry, i got d/c |
20:58.39 | Dr-Linux | sorry my question was about Dirctory |
20:58.47 | dlynes_ | znoG: so you're not getting a SIP 40x error message? |
20:58.55 | Dr-Linux | Hmmhesays: any clue, maybe i missed your answer |
20:59.17 | Hmmhesays | You want people to dial by name right? |
20:59.29 | vader-- | do you guys know what this line will do |
20:59.30 | vader-- | exten => s,1,Playback(transfer,skip) |
20:59.36 | papo | Hi. I set up my asterisk-server with sip. It's not behind NAT. When I call in with a client which is behind nat using a sip provider, I don't hear anything. Though I see in the log that the files I indicated in the Playback() application are being played |
21:00.03 | *** join/#asterisk |dennis| (i=dennis@200.32.215.84) |
21:00.11 | Dr-Linux | Hmmhesays: that's already working fine, but i want as the call routes to the callee, system should announce extension |
21:00.24 | Dr-Linux | so the caller should know at what extension he called. |
21:00.41 | Dr-Linux | i don't know why asterisk doesn't provide this feature. |
21:00.44 | Hmmhesays | So you want to complicate things |
21:00.51 | papo | I have to mention that the client is behind the NAT of which the server asterisk is running on is the gateway, but that shouldn't matter theoretically |
21:01.01 | Dr-Linux | Hmmhesays: yes maybe |
21:01.31 | Dr-Linux | Hmmhesays: by default call by name app is very easy |
21:01.46 | Dr-Linux | but i want that option which i described |
21:02.06 | Hmmhesays | I could probably write you a patch, if motivated by monitary compensation |
21:02.26 | dlynes_ | Dr-Linux: The feature you're asking for is 'Call Announce' |
21:02.35 | znoG | dlynes_: nope, just a temporarily not available |
21:02.54 | Dr-Linux | Hmmhesays: i didn't understand your last words? |
21:02.55 | dlynes_ | znoG: Yeah...that phone might have do not disturb enabled, then |
21:03.10 | Dr-Linux | motivated by monitary compensation? :S |
21:03.14 | dlynes_ | Dr-Linux: he's saying he can add that feature to asterisk if you pay him |
21:03.22 | dlynes_ | s/monitary/monetary |
21:03.31 | Hmmhesays | haha my typing skillz are bad today |
21:03.52 | Hmmhesays | good call dlynes_ |
21:04.01 | dlynes_ | Dr-Linux: The grandstream 102's have that feature to announce the number coming in |
21:04.02 | znoG | dlynes_: nope, it's not that as i can ring that extension directly |
21:04.09 | Dr-Linux | i'd love to pay, if i was in US or CA, |
21:04.10 | *** join/#asterisk nagl (n=nagl@86.59.54.237) |
21:04.11 | dlynes_ | Dr-Linux: However, they don't announce the name of the person |
21:04.21 | Hmmhesays | pakistan doesn't have paypal do they |
21:04.36 | dlynes_ | Dr-Linux: OTOH, the Dial application(?) has a call screening feature |
21:04.40 | Dr-Linux | Hmmhesays: yes, but not that much money i have |
21:04.52 | Hmmhesays | yeah no one wants to pay |
21:05.01 | dlynes_ | znoG: perhaps the zaptel channel is trying to ring a different account on the sip device than the other sip devices are |
21:05.10 | Dr-Linux | dlynes_: do you understand what i want? |
21:05.24 | dlynes_ | Dr-Linux: Yeah...and Asterisk already has that feature |
21:05.41 | znoG | ahh i fixed it! |
21:05.43 | Hmmhesays | Does directory announce the extension its calling? |
21:05.46 | dlynes_ | I think it's called the privacy manager |
21:05.46 | Dr-Linux | dlynes_: can you guide me how can i do that? |
21:05.53 | znoG | dlynes_: the whole problem is me trying to use the latest version... asterisk 1.2.4 works fine |
21:06.01 | dlynes_ | Dr-Linux: Use the force, use the search engine on voip-info.org |
21:06.15 | dlynes_ | znoG: which phones are you using? |
21:06.30 | Dr-Linux | dlynes_: yes, but what should i search in voip-info.org? |
21:06.44 | dlynes_ | Dr-Linux: or try the applications section under asterisk |
21:06.50 | dlynes_ | Dr-Linux: Call Privacy Manager |
21:07.13 | dlynes_ | Dr-Linux: also try call screening, or call announce |
21:07.18 | Dr-Linux | dlynes_: i tried much but still no luck |
21:07.28 | dlynes_ | Dr-Linux: watch...i'll get it first try :) |
21:07.47 | Dr-Linux | dlynes_: that's fine, but i need this option in "call by name directory" |
21:07.58 | Dr-Linux | dlynes_: ok |
21:08.19 | dlynes_ | I have no idea about call by name directory, but if it works, it works |
21:08.46 | Dr-Linux | dlynes_: it's not a feature .. that's why i'm asking |
21:08.51 | sevard | Hmmhesays: sorry for humping you |
21:08.57 | Dr-Linux | but asterisk should have this feature. |
21:09.04 | sevard | actually sorry for the delayed response, i couldn't see and had to get some visine |
21:09.24 | dlynes_ | http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+PrivacyManager |
21:09.25 | dlynes_ | first try |
21:09.53 | anthm | forgot who asked but here you go http://bugs.digium.com/view.php?id=7072 |
21:10.05 | anthm | chanspy with mutiple groups |
21:10.19 | Dr-Linux | smacks? :P |
21:10.36 | dlynes_ | anthm: heh...nice plug for cluecon :) |
21:11.40 | *** join/#asterisk hans (n=fugalh@dhcp21.cs.nmsu.edu) |
21:12.05 | dlynes_ | Dr-Linux: Have you read that link yet? |
21:12.09 | dlynes_ | Dr-Linux: read it! |
21:12.23 | *** join/#asterisk CunningPike (n=arodgers@S010600095b33697f.vc.shawcable.net) |
21:12.31 | *** part/#asterisk gmaruzz (n=gmaruzz@217-133-80-112.b2b.tiscali.it) |
21:13.02 | Dr-Linux | dlynes_: wait my friend |
21:13.05 | Dr-Linux | reading |
21:13.25 | dlynes_ | yeah...you got another answer in asterisk-dev, too |
21:13.56 | *** join/#asterisk viperdude (n=viperdud@84-45-168-60.no-dns-yet.enta.net) |
21:14.12 | *** join/#asterisk TonyM (n=TonyM@adsl-solo-80-168-225-147.claranet.co.uk) |
21:15.19 | znoG | dlynes_: all PAP2-NA |
21:15.25 | *** join/#asterisk MacDome (n=eseidel@A17-255-100-181.apple.com) |
21:16.36 | hans | trying to get realtime to look up voicemail users in the db (still storing vm on the filesystem). it's making the odbc connection but never querying it |
21:16.47 | hans | I followed the instructions on the wiki very carefully, but nada |
21:17.26 | dlynes_ | znoG: damnit |
21:17.50 | dlynes_ | znoG: Do all your pap2-na's have a nat between them and asterisk? |
21:21.42 | *** join/#asterisk LBJ- (n=no@0x5551a675.adsl.cybercity.dk) |
21:22.15 | Dr-Linux | dlynes_: heh |
21:23.56 | *** part/#asterisk fjean (n=fjean@201.29.130.118) |
21:28.55 | *** join/#asterisk [TK]D-Fender (n=joe@66.11.164.239) |
21:30.55 | *** join/#asterisk RoyK (n=roy@10.80-203-106.nextgentel.com) |
21:32.17 | Dr-Linux | what's the best way to apply a patch into the asterisk? |
21:32.47 | Damin | cd /usr/src/asterisk ; patch < patchfile.txt |
21:34.03 | *** join/#asterisk Marquis42 (i=HydraIRC@c-24-56-217-74.chrlmi.cablespeed.com) |
21:34.27 | Dr-Linux | Damin: i my case it will be " cd /usr/src/asterisk ; app_dicrectory.c < patchfile.txt ? |
21:34.39 | Damin | No. |
21:34.40 | Damin | cd /usr/src/asterisk ; patch < patchfile.txt |
21:35.49 | Damin | Dr-Linux: And with that, I must leave. If you can't figure it out, try "man patch" |
21:35.50 | *** part/#asterisk TonyM (n=TonyM@adsl-solo-80-168-225-147.claranet.co.uk) |
21:35.50 | *** join/#asterisk gmaruzz (n=gmaruzz@217-133-80-112.b2b.tiscali.it) |
21:35.50 | Dr-Linux | Damin: okey thanks, so after doing that, do i need to recompile? |
21:35.58 | ravenpi | Dr-Linux: Yes. |
21:36.03 | russellb | Dr-Linux: there is a patch howto on http://www.asterisk.org/developers/ |
21:36.22 | Qwell[] | howtos are for people who can't RTFM :P |
21:36.23 | Dr-Linux | russellb: that's would be helpful thanks |
21:36.27 | ravenpi | Dr-Linux: And re-install. Note that you probably do NOT want to do a "make samples", 'cause your config files will get overwritten. |
21:36.48 | russellb | Qwell[]: it's only a few commands :-p ... svn diff and patch |
21:36.50 | russellb | ok, 2 commands |
21:37.07 | Qwell[] | "make oops-I-screwed-up-and-overwrote-my-configs" |
21:37.12 | Dr-Linux | ravenpi: yeah that i know, but what about modules? they will not bother? |
21:37.15 | russellb | make uninstall-all |
21:37.25 | LBJ- | Anyone know if there is a problem with callstatus returning busy when the called phone i disconnected? It should return unavail. asterisk 1.2.5 |
21:37.33 | Qwell[] | well, I was thinking a little less malicious :) |
21:37.37 | ravenpi | Dr-Linux: They'll install fine with "make install". |
21:37.42 | Qwell[] | like...replacing the samples with the backups |
21:38.03 | Dr-Linux | i have backup |
21:38.21 | Dr-Linux | heh, i have very good expereince with such thing |
21:38.35 | *** join/#asterisk kaz0358 (n=kaz@kazg5.telecom.ksu.edu) |
21:38.39 | ravenpi | Speaking of overwriting files, what's The Right version control system for stuff like /etc files? CVS? Subversion? |
21:38.56 | Dr-Linux | you know, once mistakenly i did rm -rf /etc/asterisk |
21:39.05 | Dr-Linux | i had no back for all |
21:39.10 | Dr-Linux | it was peak time |
21:39.19 | Dr-Linux | but i checked, everything was working fine :P |
21:39.24 | CunningPike | ravenpi: Whatever works for you - I'm using Visual Sourcesafe |
21:39.26 | Marquis42 | I personally use a central Subversion repository for all of my configs, both /etc/asterisk and others. |
21:39.35 | Dr-Linux | so i hold my head and was thinking what to do now :P |
21:39.38 | Marquis42 | VSS!?! Ewww.... :P |
21:39.46 | CunningPike | ravenpi: But I'm planning to switch to Subversion |
21:40.07 | CunningPike | Marquis42: Actually, VSS is not that bad - I like it |
21:40.14 | ravenpi | CunningPike: <grin> I guess so, based on Marquis' reaction. Never heard of VSS, m'self. |
21:40.37 | *** join/#asterisk RoyK (n=roy@10.80-203-106.nextgentel.com) |
21:40.46 | CunningPike | ravenpi: It works for me :) |
21:41.11 | ravenpi | Dr-Linux: Yeah, deleting Asterisk's /etc hierarchy is a Bad Thing, but, so long as you don't terminate the program, or do a reload, it'll keep chugging. Just hope you can get your restores back quickly. |
21:41.25 | Marquis42 | I'm just razzing you. It's not entirely bad, though there are certainly issues that exist. My biggest one is any attempt at remote repository access over a WAN that isn't capacious |
21:41.44 | CunningPike | Marquis42: True enough |
21:42.28 | *** join/#asterisk hads (n=hads@203.109.245.87) |
21:42.53 | BadPacket | does this syntax work for anyone: Dial(SIP/user:pass@${EXTEN}@gateway.itsp.com) ?? |
21:42.54 | CunningPike | And the other issue, and the reason I'm considering a move to SVN is that it deploys DOS formatted files, regardless of the format of the original. This has no effect for .conf files, but buggers up init scripts and the liek |
21:43.26 | ravenpi | CunningPike: Oh, wow. Yeah, that would suck. |
21:43.26 | Marquis42 | CunningPike: I didn't know that, I've never used VSS outside a Windows environment. Ouch! |
21:43.37 | CunningPike | Yes - it's real dumb |
21:43.43 | Marquis42 | I agree. |
21:43.44 | [TK]D-Fender | BadPacket : that be better done as : Dial(SIP/user:pass@gateway.itsp.com/${EXTEN}) |
21:43.56 | ravenpi | CunningPoke: You'd think, I dunno... that it'd give a binary-complete restore of the file. That's just silly. |
21:44.02 | BadPacket | [TK]D-Fender: thanks |
21:44.05 | CunningPike | It only matters when transferring files between servers - it doesn't mess up the original |
21:44.24 | CunningPike | ravenpi: You'd think. MS doesn't |
21:44.31 | BadPacket | [TK]D-Fender: ...the example sip.conf in the /configs directory is wrong then... |
21:44.43 | [TK]D-Fender | BadPacket : Although typically bad idea. Better to make a peer entry in sip.conf so you don't have to filter out password in pastebin'ing your diaplan for help, or repeating that long string in many places. |
21:44.53 | ravenpi | CunningPike: Oooooohhh. Didn't realize it was an MS product. *nods* All clear, now. |
21:45.01 | [TK]D-Fender | BadPacket : Never ask me about "whats wrong", I might never stop.... |
21:45.21 | BadPacket | [TK]D-Fender: heh |
21:45.30 | CunningPike | ravenpi: Not all MS products are bad |
21:46.11 | BadPacket | [TK]D-Fender: hmm... still not working |
21:46.19 | [TK]D-Fender | CunningPike : yeah MS had a money market simulator game back around 1980 in BASIC that I loved! |
21:46.24 | *** join/#asterisk eric_hill (i=EricHill@204.94.175.11) |
21:46.24 | *** part/#asterisk gr0mit_home (n=wendolen@extrt.txrx.org.uk) |
21:46.35 | CunningPike | [TK]D-Fender: Heh heh |
21:46.50 | *** join/#asterisk jsaunders (n=root@216.86.121.58) |
21:46.51 | Hmmhesays | yep i was -2 in 1980 |
21:46.55 | tainted- | QBASIC? |
21:47.00 | [TK]D-Fender | Actually I think it was stock market, not specifically curreny... |
21:47.09 | jsaunders | I still have an ol RPG I wrote in QBasic years ago. Heh. It rules. |
21:47.12 | [TK]D-Fender | tainted- : No, GW at best... |
21:47.15 | tainted- | nibbles was great |
21:47.19 | jsaunders | Nibbles was fun. |
21:47.19 | tainted- | jsaunders send send! |
21:47.21 | [TK]D-Fender | QBASIC is for NEWBS! |
21:47.23 | tainted- | haha |
21:47.23 | nahirean | gorilla owned nibbles |
21:47.28 | jsaunders | gorilla was kew too |
21:47.36 | CunningPike | OMG - what have I done......... |
21:47.37 | tainted- | was that the velocity, trajectory game |
21:47.49 | nahirean | ya.. two apes chucking fruit at one another |
21:48.03 | jsaunders | tainted: Will do... check www.northworld.ca for "what it became" :) Ported it to Pascal, but send me an email @ jsaunders@devarus.ca and I'll send you QBasic code. |
21:48.26 | CunningPike | jsaunders: Another Canuck? |
21:48.32 | jsaunders | yessir, vancouver bc |
21:48.36 | *** join/#asterisk squinky86 (n=squinky8@gentoo/developer/squinky86) |
21:48.39 | CunningPike | Me too :D |
21:48.45 | [TK]D-Fender | I remember working on a real-time 3-D (calculated, not drawn) version of Trade Wars 2002 I was working on in BASIC back in the day.... |
21:48.55 | *** join/#asterisk Zodiacal (i=hehehe@bdsl.66.14.242.199.gte.net) |
21:48.57 | jsaunders | Well, technically I'm in Chilliwack although I'm at our company office in Langley atm. |
21:48.57 | tainted- | jsaunders that's awesome! |
21:49.01 | CunningPike | Maybe we should set up #asterisk-canada |
21:49.05 | jsaunders | Heheh |
21:49.12 | jsaunders | Perhaps ya'll could help me w/ a problem. |
21:49.17 | CunningPike | Leave this channel for those who speak only Americanish ;) |
21:49.19 | jsaunders | [ext-did] has following line.... |
21:49.34 | Zodiacal | anyone know of a good sound file to play that represents break times? e.g. over a loud speaker. |
21:49.42 | jsaunders | exten => s,1,Set(FROM-DID=s) |
21:49.52 | *** join/#asterisk techman97_andy (n=me@70-98-31-249.dsl1.rsm.mn.frontiernet.net) |
21:49.59 | jsaunders | Which is an amp thing yes... but it's still valid * so I figure I can ask in here. |
21:49.59 | [TK]D-Fender | Zodiacal : Watch a hockey game... plenty of inspiration there.... |
21:50.23 | jsaunders | I'm assuming FROM-DID=s should replace s w/ DID dialed. |
21:50.26 | [TK]D-Fender | jsaunders : So far yet... and your point is...?? :) |
21:50.37 | [TK]D-Fender | (keep going so we have something to roast you for) |
21:50.46 | jsaunders | Heheh, k. :) |
21:51.03 | jsaunders | But when it gets to inbound routing... it's looking for extension 's', not populating w/ DID dialed. |
21:51.06 | jsaunders | ugh |
21:51.24 | [TK]D-Fender | jsaunders : What is the incoming interface? |
21:51.29 | jsaunders | zap or sip |
21:51.44 | [TK]D-Fender | jsaunders : And DID's don't LAND on s. they hit an EXTEN in the target context. |
21:52.02 | *** join/#asterisk shido6 (n=shido6@d38-45-81.commercial1.cgocable.net) |
21:52.13 | jsaunders | Well, I can get it to catch S if I put a "catch all" route in there. |
21:52.31 | jsaunders | Ugh, I'm confused. Heh. Excuse my newbness. |
21:52.34 | CunningPike | jsaunders: Wouldn't it be exten => s,1,Set(FROM-DID=${s}), or am I am idiot - warning, both may be true |
21:52.41 | jsaunders | That's what I was thinking.... |
21:52.48 | [TK]D-Fender | jsaunders : "s" is when the target is UNKNOWN, which is exactly the opposite of that a DID is. |
21:52.50 | Dr-Linux | i just wanna make sure will this patch work on my 1.2.0 version? http://www.delink.net/software.php |
21:53.02 | [TK]D-Fender | jsaunders : as a Catch-all it'd be more like _X. |
21:53.12 | jsaunders | Ah, very useful info. |
21:53.16 | *** part/#asterisk wrmem (n=monnin@monnin-win.ci.uiuc.edu) |
21:53.17 | jsaunders | I wonder why ZAP doesn't know number dialed. |
21:53.19 | jsaunders | weird |
21:53.30 | Dr-Linux | the Directory() patch. |
21:53.46 | [TK]D-Fender | jsaunders : What kind of Zap? |
21:53.50 | jsaunders | tdm2432e |
21:53.57 | CunningPike | jsaunders: What I do with DIDs is this: exten => _604990XXXX,1,Set(DID-FROM=${EXTEN:-4}) |
21:54.08 | [TK]D-Fender | jsaunders : Analog doesn't support the concept of DID/ |
21:54.13 | [TK]D-Fender | jsaunders : Thats why |
21:54.23 | [TK]D-Fender | jsaunders : Analog lines "just ring". |
21:54.41 | jsaunders | hmm |
21:54.55 | [TK]D-Fender | jsaunders : only digital interfaces support DID signalling typically. |
21:54.56 | CunningPike | jsaunders: What type of service do you have? |
21:55.10 | [TK]D-Fender | CunningPike : Looking like POTS to me.... |
21:55.22 | [TK]D-Fender | CunningPike : do look at the card he referenced |
21:55.39 | jsaunders | I'm trying to do following... have fax come in on zap chan 1 and get routed to zap chan 13. |
21:55.45 | jsaunders | Using freepbx / amp. |
21:56.01 | CunningPike | Ah, but there are flavors of POTS, no? More than one phone line a single POTS line can have |
21:56.21 | [TK]D-Fender | jsaunders : Can't help you with the AMP part of that, but * can do taht easily enough. |
21:56.29 | jsaunders | Indeed. :) Hmm. |
21:56.40 | LBJ- | Hi can someone help me with a vm problem ? |
21:56.56 | [TK]D-Fender | CunningPike : If you're referring to distictive ring, perhaps, but thats only questionably reliable. |
21:57.12 | justinu | pots is pots |
21:57.13 | Dr-Linux | does recompile asterisk effect of /var/lib/asterisk/agi-bin/ dir? |
21:57.19 | brodiem | maybe you're thinking of centrex |
21:57.19 | justinu | if you had distinctive ring, it wouldn't be pots anymore, would it |
21:57.36 | [TK]D-Fender | justinu : Um, yeah, thats part of POTS in the boring analog world... |
21:57.51 | jsaunders | Sure is a nice card though... thanks Digium. |
21:57.58 | [TK]D-Fender | justinu : Simple cadence to the ring pulse. |
21:58.18 | [TK]D-Fender | jsaunders : Not to fit it in my MiniITX EPIA case.... hmmm! |
21:58.18 | justinu | but it's not plain old, anymore |
21:58.27 | jsaunders | Heheh |
21:58.32 | jsaunders | Yeah, she's a beast. |
21:58.44 | *** join/#asterisk lzhang (n=rjrae@67.95.13.46) |
21:58.55 | jsaunders | Atleast it doesn't take up multiple pci slots like Sangoma. |
21:59.22 | [TK]D-Fender | jsaunders : That IS a minus in certain cases. Each have their benifits |
21:59.26 | lzhang | If I make a call from the system to an external number, is it possible to pass that call to an extension locally? (through the asterisk manager) |
21:59.41 | jsaunders | Although I heard the sangoma architecture is supposed to be very good. |
21:59.53 | [TK]D-Fender | lzhang : take a look at the "m" parameter of app_dial.... |
22:00.26 | [TK]D-Fender | jsaunders : it is. MUCH kinder on IRQ calls, and no know incompatabilities with any standard hardware. |
22:00.27 | kaz0358 | is anyone getting involved with ISN? anyone have thoughts on the idea? http://www.freenum.org/ descibes ISN |
22:01.07 | [TK]D-Fender | jsaunders : And can support a serious EC DSP. |
22:01.12 | websae | anyone done any T.38 faxing? |
22:01.28 | jsaunders | Glad to hear Fender... I heard talk about the irq thing. Dunno the full deal though. |
22:01.34 | jsaunders | Soo much to learn. |
22:01.56 | jsaunders | I kinda like the backplane idea... kewl. |
22:03.06 | *** join/#asterisk zippp (n=zip@63.99.9.4) |
22:03.11 | CunningPike | LBJ-: What was your vm question? |
22:04.02 | zippp | can anyone tell me how to set the caller id to xxx-xxx-xxxx if the current channel CID number isn't in that format? |
22:04.15 | kaz0358 | there are already about 100 univerities and companies involved in ISN.. it seems like a better solution than having a centralized location like sipbroker.com provide all the 3-5 digit provider gateway numbers.. and there is a better seperation between the subscriber number and provider number |
22:05.16 | [TK]D-Fender | jsaunders : I can explain better later as I own 2 of their card, and support several with the Analog flavour. |
22:05.26 | [TK]D-Fender | But for now, I'm off to class, later all. |
22:05.54 | CunningPike | zippp: Set(CallerID())? |
22:06.20 | *** join/#asterisk ReD-MaN (i=redman@dhcp-0-2-b3-9a-4a-5b.cpe.quickclic.net) |
22:06.26 | zippp | I know that part, I need to logic to say if callerid != fmt(xxx-xxx-xxxx) then setcallerid(xxx-xxx-xxxx) |
22:06.33 | [TK]D-Fender|AFK | zippp : CallerID is plain numberic and doesn't include non-digits like "-". |
22:06.35 | CunningPike | Dr-Linux: What's the question |
22:06.39 | jsaunders | Later Fender. :D |
22:06.45 | [TK]D-Fender|AFK | anyways, I'm off |
22:07.01 | [TK]D-Fender|AFK | CunningPike : CALLERID() function is case sensitive and all-caps BTW |
22:07.21 | CunningPike | [TK]D-Fender|AFK: Thanks :) |
22:07.21 | Dr-Linux | CunningPike: 2 questions |
22:07.27 | Dr-Linux | i just wanna make sure will this patch work on my 1.2.0 version? http://www.delink.net/software.php |
22:07.38 | *** join/#asterisk Tall-guy (i=tall-guy@207-195-103-110.regn.hssx.sasknet.sk.ca) |
22:07.41 | Dr-Linux | does recompile asterisk effect of /var/lib/asterisk/agi-bin/ dir? |
22:08.09 | Tall-guy | Ok, before I rant...anyone using Eyebeam 1.5 with asterisk? (just released last week) |
22:08.14 | CunningPike | "If you need to add it to a production version of Asterisk, here is a patch to Asterisk 1.2.6" - I would take that to mean that you need 1.2.6 for it |
22:09.07 | Dr-Linux | CunningPike: but i have asterisk 1.2.0 |
22:09.16 | Dr-Linux | but i need that feature |
22:09.21 | CunningPike | Dr-Linux: Then you'll likely need to upgrade to 1.2.6 |
22:09.28 | CunningPike | You probably should anyway |
22:09.53 | CunningPike | Neat little patch, btw |
22:10.06 | Dr-Linux | yeah :S |
22:12.27 | lzhang | Anybody know how to use the manager command Redirect? documentation is a little sparse... |
22:12.38 | Tall-guy | eyebeam...anyone?....Beuhler??? Beuhler?? :) |
22:12.58 | lzhang | I'm trying to pass a call that the system made to an external number to a sip phone locally |
22:13.07 | *** join/#asterisk SuperLag (n=aaron@gentoo/developer/SuperLag) |
22:13.07 | Dr-Linux | Tall-guy: i use eyeBeam but don't know what version |
22:13.10 | *** join/#asterisk kainam (n=Jake@202.137.160.110) |
22:13.24 | Tall-guy | dr-linux: problay 1.1 if you don't know (cause they just released a new version that is all f*KED up in my opinion) |
22:13.39 | *** join/#asterisk tdmoc (n=jghj@VA1-1B-u-0044.mc.onolab.com) |
22:14.02 | tdmoc | hola |
22:14.58 | *** part/#asterisk ph|ber (i=phiber@slackwaresupport.com) |
22:14.59 | Dr-Linux | Tall-guy: lolz i have eyeBeam for unlimited users, but still i don't use it |
22:15.05 | Dr-Linux | i love SJphone |
22:15.18 | *** part/#asterisk gmaruzz (n=gmaruzz@217-133-80-112.b2b.tiscali.it) |
22:17.28 | *** join/#asterisk brif8 (n=Techno@lazyjtrainingcenter.com) |
22:18.18 | brif8 | anyone using astertext? I've compiled and now trying to restart * is failes "app_securax_serverload.so: undefined symbol: ast_load |
22:18.18 | brif8 | " |
22:18.22 | brif8 | any ideas ? |
22:19.07 | *** part/#asterisk Marquis42 (i=HydraIRC@c-24-56-217-74.chrlmi.cablespeed.com) |
22:19.40 | *** part/#asterisk Tall-guy (i=tall-guy@207-195-103-110.regn.hssx.sasknet.sk.ca) |
22:22.21 | *** join/#asterisk brockj49464 (n=brockj49@63.87.56.236) |
22:26.53 | jsaunders | I'm diggin' idefisk these days. Works decent. No probs *that I know of*. |
22:27.38 | jsaunders | SJPhone is a sip classic... but I'm enjoying ease of cutting thru nat w/ IAX. |
22:29.16 | syle | iax hmmm |
22:29.28 | syle | there a iax proxy you guys use? |
22:29.42 | jsaunders | sure, it's called asterisk. Heh. |
22:29.48 | justinu | iax proxy? uh |
22:30.00 | syle | something like ser for iax maybe |
22:30.10 | jsaunders | For? |
22:30.21 | syle | voip provider |
22:31.18 | justinu | it would be asterisk, like jsaunders said |
22:31.28 | jsaunders | asterisk & yate are only switches I know of that deal w/ IAX. Yate apparently is not doing a very good job of it, although that's just hearsay. |
22:31.37 | jsaunders | I'm sure there are others but I am not aware of 'em. |
22:31.59 | *** join/#asterisk r0d3nt|m (i=r0d3nt@tinfoilhat.net) |
22:32.09 | syle | i guess the biggest question is does the iax protocal support redirects |
22:32.37 | jsaunders | It facilities something similar, yes. google iax. |
22:32.40 | [hC] | it supports native transfer, which eliminates middle-man hops that are unnecessary if thats what you want. |
22:32.58 | *** join/#asterisk kFuQ (n=somedude@c-67-185-114-199.hsd1.wa.comcast.net) |
22:32.59 | syle | but if that happens do you loose accounting info |
22:34.54 | MikeJ[Laptop] | syle, iax is media and signaling all in one |
22:35.05 | MikeJ[Laptop] | so if you do a native transfer your out of the loop |
22:35.33 | MikeJ[Laptop] | to do what I think you want is tricky, I think file is playing with ideas on how to do it |
22:35.44 | jsaunders | Go file. |
22:35.46 | syle | i have been wondering say with iax native transfer and sip's canreinvites if you loose cdr for not handling SDP stream |
22:35.59 | jsaunders | rtp |
22:37.02 | *** join/#asterisk docelm0 (n=docelmo@55-65.126-70.tampabay.res.rr.com) |
22:37.07 | jsaunders | To stay in signalling path when forwarding rtp you need a proper b2bua... only one I know of that handles *questionably* properly is Yate. |
22:37.47 | syle | i see what mikej is saying, if your signalling is lost then your foobarred then |
22:37.52 | syle | what about sip |
22:38.40 | jsaunders | Last comment was regarding sip. |
22:38.50 | jsaunders | iax does not use rtp |
22:38.53 | *** join/#asterisk r0d3nt|m (i=r0d3nt@tinfoilhat.net) |
22:39.27 | syle | is it possible for a provider to avoid being the man in the middle but still bill his customers for a call |
22:39.40 | jsaunders | w/ sip & h323, yes. |
22:39.43 | dlynes_ | ~seen coppice |
22:39.45 | jbot | coppice <n=chatzill@153.192.17.210.dyn.pacific.net.hk> was last seen on IRC in channel #asterisk, 5h 27m 6s ago, saying: 'I have two in this house :-)'. |
22:40.09 | dlynes_ | damnit...keep missing him :) |
22:40.34 | syle | why h323 |
22:41.08 | jsaunders | We have a commercial yate switch that handles just signalling for tens of thousands of calls daily.... forwards audio between endpoints w/out first having to start rtp and the "reinvite" like asterisk does. |
22:41.26 | jsaunders | the = then |
22:41.28 | *** join/#asterisk danilom (n=danilom@201.209.43.241) |
22:41.50 | *** join/#asterisk MstlyHrmls (n=mh@melbourne.mostly-harmless.ca) |
22:41.55 | *** join/#asterisk ptiggerdine (n=ptiggerd@router-bne.action-international.com) |
22:42.05 | syle | do you have to pull billing info off the yate switch then? |
22:42.17 | justinu | jsaunders: for a n00b, you sure seem to understand this stuff |
22:42.23 | jsaunders | Can be annoying when trying to make sure endpoints have firewalls configured properly. Ugh. |
22:42.34 | jsaunders | justinu :D |
22:42.38 | jsaunders | I try. |
22:42.44 | jsaunders | I'm weak at * scripting though. :( |
22:43.32 | syle | i'm trying to think of a way for asterisk to handle signalling to bill for call but toss the RTP stream |
22:43.35 | jsaunders | syle: Yate provides all call related info and passes via sql to database. |
22:43.47 | jsaunders | Asterisk will always start audio (rtp) stream. |
22:43.57 | jsaunders | canrinvite will simple move audio after 8 or so seconds. |
22:44.01 | jsaunders | simple = simply |
22:44.08 | dlynes_ | jsaunders: freeswitch does iax as well, but it's far from being a finished product |
22:44.13 | jsaunders | freeswitch pwns |
22:44.18 | jsaunders | I can't wait to hop onboard. |
22:44.40 | jsaunders | anthm & MikeJ and the crew are doing a great job from what I can see. |
22:44.43 | jsaunders | Love the community. |
22:44.49 | dlynes_ | Yeah, they're doing a great job |
22:44.56 | dlynes_ | I'm just saying it's far from being finished :) |
22:44.57 | justinu | 8 seconds? |
22:45.05 | justinu | my asterisk systems reinvite the call as soon as it answers |
22:45.22 | jsaunders | Cannot verify that justinu... I read somewhere it was something like 6-8 seconds. |
22:45.35 | jsaunders | And testing seemed to confirm. |
22:45.40 | mog_work | heh i got you beat justinu mine does is it before the cal is answered !!!!! ^_^ |
22:45.47 | jsaunders | Heheh |
22:45.50 | justinu | well, i'm running early media |
22:45.54 | justinu | for call progress tones, etc. |
22:45.58 | justinu | so that isn't possible. |
22:46.05 | mog_work | mmmhmm |
22:46.18 | mog_work | man telephony would be so much simpler |
22:46.24 | jsaunders | word mog |
22:46.25 | mog_work | if everyone just charged when you went off hook |
22:46.25 | syle | i guess i;ll test later this week if i can;t get a direct answer with tcpdump lol, thx anyways |
22:46.38 | mog_work | earlymedia is just retarded in my opinion |
22:47.04 | justinu | lol, that statement is kinda retarded in my opinion |
22:47.13 | jsaunders | heheh |
22:47.18 | mog_work | sure |
22:47.22 | justinu | some of us need to generate call progress tones |
22:47.27 | justinu | or messages |
22:47.34 | justinu | and we don't want people to pay for that call |
22:47.35 | mog_work | just think how much simpler all the protocols would be if when you dialed it was considered answered |
22:47.41 | mog_work | and you just got charged for busy tone |
22:47.44 | mog_work | or whatever |
22:47.48 | mog_work | it would all even out in the end |
22:47.50 | justinu | say what? |
22:48.09 | mog_work | when you called to hear that busy tone it costs the provider to connect |
22:48.12 | *** join/#asterisk kainam (n=Jake@202.137.160.110) |
22:48.14 | mog_work | and everything |
22:48.19 | mog_work | i mean the cost is still their |
22:48.22 | mog_work | err there |
22:48.31 | syle | yeah and think how kewl it woudl be if everytime you had sex you never orgasmed! |
22:48.38 | justinu | lol |
22:48.40 | justinu | you guys are whack |
22:48.54 | jsaunders | ahhh.... syle... lets not get carried away here. |
22:49.09 | dlynes_ | early media has its uses |
22:49.14 | jsaunders | Heh |
22:49.18 | dlynes_ | such as announcing the caller |
22:49.25 | jsaunders | Isn't that the whole point of sex? Heh |
22:49.32 | dlynes_ | then the other end can refuse the call if it's that bastard collection agency |
22:49.38 | justinu | i use it to announce number changes, etc. |
22:49.40 | justinu | typical telco crap |
22:49.44 | *** part/#asterisk websae (n=websae@CPE-24-167-206-22.wi.res.rr.com) |
22:50.19 | mog_work | syle what the hell does that have to do with anything? |
22:50.41 | mog_work | but why shouldnt person x pay for that |
22:50.47 | mog_work | you still used telco services etc |
22:50.55 | justinu | because they didn't actually talk to the person they called |
22:51.09 | dlynes_ | people don't want to have to pay for dialing the wrong number |
22:51.09 | mog_work | but you were connected |
22:51.15 | mog_work | you arent paying for the conversation |
22:51.18 | mog_work | but the connection] |
22:51.19 | justinu | connected to a media server |
22:51.25 | justinu | not the destination |
22:51.36 | dlynes_ | even when they dial the wrong number on a typical telco, and they realize that, a lot of people will call up the telco to refute the charges |
22:51.44 | mog_work | yeah i know |
22:51.47 | justinu | back in the way old days, all calls were person-to-person, i think |
22:51.55 | mog_work | as you have used their services |
22:52.05 | mog_work | it shouldnt matter that you didnt get result x |
22:52.11 | jsaunders | thru the ol switchboard operators? Heh |
22:52.11 | mog_work | services were still rendered |
22:52.12 | dlynes_ | so work that into your cost of doing business |
22:52.18 | dlynes_ | deal with it |
22:52.21 | mog_work | why not just have it simple |
22:52.24 | mog_work | and average it out |
22:52.25 | dlynes_ | people are people...you can't change the way they think |
22:52.31 | dlynes_ | they don't think like computer programs |
22:52.32 | mog_work | and just charge people for using services? |
22:52.36 | mog_work | it makes it so much simpler |
22:52.40 | jsaunders | I like mog's thinkin' outside the box though... can't knock him for that. |
22:52.55 | justinu | i dunno, i'm too programmed |
22:53.01 | justinu | i'm always concerned about answer sup |
22:53.09 | dlynes_ | jsaunders: sounds more like a lazy programmer that doesn't want to have to do the extra programming to allow for that :) |
22:53.09 | justinu | too many years in telco, i guess |
22:53.16 | mog_work | see in my system you wouldnt have to deal with it |
22:53.19 | mog_work | lol |
22:53.22 | CunningPike | How about making it even easier - flat rate, like internet ;) |
22:53.40 | mog_work | as i make maybe 3 dollars worth of calls a month |
22:53.47 | jsaunders | A lazy programmer @ Digium? Heh |
22:53.54 | DoktorGreg | the best model i have found for the small to no cap startup is a ma-pa enterprise |
22:54.02 | justinu | jsaunders: lol |
22:54.09 | jsaunders | I think they're doin' a good job over there. But, that's just my opinion. |
22:54.24 | dlynes_ | yeah...russellb rocks |
22:54.26 | jsaunders | That wasn't a burn mog. Heh. Was actually a question. |
22:54.41 | mog_work | ahh okies, feelings less hurt |
22:54.51 | dlynes_ | He's a full time student, and still finds time to manage trunk |
22:54.57 | jsaunders | How could I burn ya... you helped me w/ my tdm2400. |
22:55.07 | jsaunders | You pwn in my books mang. |
22:55.07 | dlynes_ | plus help everyone out on #asterisk/#asterisk-dev |
22:55.16 | mog_work | really /me doesnt remember i help lot of people out |
22:55.19 | mog_work | sorries |
22:55.36 | jsaunders | Yep, just under 2 weeks ago. I've chatted ya up a few times... no worries, I'm sure your busy. |
22:56.23 | mog_work | oh i know i have talked with you few times |
22:56.31 | mog_work | but dont always remember what i fix |
22:56.37 | jsaunders | heheh, I hear ya. |
22:56.43 | jsaunders | I can't remember what I fixed 5 minutes ago. |
22:56.58 | dlynes_ | btw |
22:57.00 | jsaunders | James, fix this.... James, fix that.... ugh. |
22:57.22 | dlynes_ | Anyone know why Caller ID wouldn't work on either x100p.com cards, or a genuine Digium X100P card? |
22:57.33 | dlynes_ | It works on one out of every ten calls or so |
22:57.40 | dlynes_ | but that's just a statistic, not a hard and fast rule |
22:58.03 | mog_work | where you located dlynes |
22:58.06 | dlynes_ | The rest of the time, the caller id shows 'asterisk' |
22:58.06 | jsaunders | a guesstimate? |
22:58.08 | jsaunders | :D |
22:58.13 | dlynes_ | mog_work: Canada (Vancouver) |
22:58.16 | CunningPike | dlynes: I'm sure you're doing the obvious Wait(2) |
22:58.34 | dlynes_ | probably |
22:58.37 | mog_work | should be fine |
22:58.49 | mog_work | you have the caller id configged correctly in zapata.conf? |
22:59.06 | CunningPike | dlynes: We have very inconsistent incoming CID, too - most of our calls have number but no name |
22:59.28 | justinu | bitch at your telco about that |
22:59.30 | dlynes_ | Yeah, I've done Wait(1), Wait(2), Wait(3) |
22:59.35 | dlynes_ | it doesn't seem to make a difference |
22:59.36 | *** join/#asterisk RoyK (n=roy@10.80-203-106.nextgentel.com) |
22:59.51 | dlynes_ | It works fine with a Nortel CICS keysystem, but not with asterisk |
22:59.52 | CunningPike | Telus and Allstream just mumble about CO and network transfer hand-offs |
23:00.16 | justinu | dlynes: rxgain too high? |
23:00.23 | CunningPike | And yet, every call I get at home has something in the name |
23:00.37 | dlynes_ | justinu: rxgain/txgain are both set at 1.5 |
23:00.50 | justinu | is that tuned, or arbitrary? |
23:00.52 | jsaunders | You should be able to get a doctorate in telco... people who hax0r this all day deserve a flashy title. |
23:00.55 | *** join/#asterisk RoyK (n=roy@10.80-203-106.nextgentel.com) |
23:01.01 | dlynes_ | justinu: does rxgain/txgain affect callerid? I thought it only affected dtmf and faxing? |
23:01.10 | justinu | callerid is delivered via FSK |
23:01.15 | justinu | modem tones |
23:01.17 | dlynes_ | You've lost me |
23:01.19 | dlynes_ | ah |
23:01.23 | justinu | so if gain is too high, it'll be distorted |
23:01.29 | justinu | and the decoder says "wtf is this?" |
23:01.32 | dlynes_ | Yeah, it was just arbitrary |
23:01.52 | dlynes_ | So the only way you can check it is by playing with it continuously until the error rate settles down? |
23:02.02 | justinu | ok, so start going down in increments of 4 until you can't hear people well enough |
23:02.03 | *** join/#asterisk Eggplant (i=No@dsl-332.cascadeaccess.com) |
23:02.14 | justinu | make it as low as you can without being too soft |
23:02.22 | dlynes_ | increments of 4? I thought the range was only -7 to +7? |
23:02.32 | dlynes_ | or you mean increments of .4? |
23:02.57 | justinu | no, it's +/- 100 |
23:03.00 | dlynes_ | oh |
23:03.01 | dlynes_ | ok |
23:03.06 | justinu | and those increments aren't dB |
23:03.24 | dlynes_ | what are they? just some arbitrary logarithmic values? |
23:03.27 | justinu | they're something like "100%" of what the card is capable of attenuating/boosting |
23:03.40 | justinu | i can't find any better spec than that |
23:03.44 | dlynes_ | ah |
23:07.09 | *** join/#asterisk shido6 (n=shido6@d38-45-81.commercial1.cgocable.net) |
23:09.27 | dlynes_ | Yeah, that definitely seems to be affecting it |
23:09.30 | dlynes_ | Thanks, justinu |
23:09.35 | justinu | np |
23:09.41 | *** join/#asterisk Abydos313 (i=abydos31@adsl-71-129-52-80.dsl.irvnca.pacbell.net) |
23:10.04 | jsaunders | quit |
23:10.07 | jsaunders | oops :) |
23:10.36 | dlynes_ | Hrm |
23:10.44 | dlynes_ | But now it doesn't seem to be recognizing hangups |
23:10.52 | dlynes_ | When it kicks into Background() |
23:11.12 | dlynes_ | Or maybe it is...just extremely delayed |
23:11.58 | justinu | using kewlstart? |
23:12.02 | dlynes_ | correct |
23:12.11 | dlynes_ | fxsks |
23:12.17 | justinu | can't imagine how gains would affect that |
23:12.19 | dlynes_ | both zaptel.conf and zapata.conf |
23:12.30 | dlynes_ | Maybe it's just the damned cellphone then |
23:12.40 | justinu | since kewlstart is looking for a loop current drop as a disconnect supervisory |
23:12.57 | dlynes_ | Nortel CICS also has problems on line 1 when you call into it with a cell phone |
23:13.36 | Dr-Linux | when my rxgain is set to 2.0 , caller hear very low, when rxgain is 3.0 then caller gets echo |
23:13.44 | justinu | ouch |
23:13.45 | Dr-Linux | not sure where to go :S |
23:13.47 | justinu | 2.5? |
23:15.21 | dlynes_ | It only happens on line 1, and it only happens when you hang up; the Nortel cics often doesn't recognize the hangup |
23:15.25 | *** part/#asterisk BadPacket (n=root@unaffiliated/badpacket) |
23:15.34 | dlynes_ | So, I'm wondering if maybe asterisk has the same problem |
23:15.58 | justinu | and it's only when you call with a cell phone? |
23:16.04 | dlynes_ | I've noticed Nortel CICS and Asterisk also both suffer from the same problem if the caller is a voip caller |
23:16.15 | dlynes_ | It doesn't happen every time though...just most times |
23:16.36 | dlynes_ | justinu: cell phone or voip phone |
23:16.55 | dlynes_ | justinu: in my case, it would be ACT/Azatel/GVC SIP phones |
23:17.08 | justinu | that phone line originates on what piece of equipment? |
23:17.24 | dlynes_ | justinu: it doesn't seem to happen for asterisk originated calls |
23:17.59 | justinu | i'd have to know more topology to understand it |
23:18.08 | dlynes_ | justinu: The cell phone problem always happens on Fido (Microcell Solutions) |
23:18.16 | file | Rogers. |
23:18.18 | dlynes_ | justinu: the voip phone problem doesn't happen every time though |
23:18.33 | dlynes_ | file: Don't know about Rogers, but fido had that issue long before Rogers bought them |
23:18.51 | justinu | so you have a phone line coming from an ILEC? |
23:18.52 | dlynes_ | it could affect other cell phones too, for all i know |
23:18.56 | dlynes_ | but most people i know are using fido |
23:19.10 | dlynes_ | justinu: yes...Telus |
23:19.27 | dlynes_ | justinu: the ones it seems to work on for cell phone hangups are allstream lines |
23:19.37 | dlynes_ | justinu: the ones it doesn't are telus lines |
23:19.38 | justinu | and when you call that PSTN number from certain types of phones, asterisk doesn't detect the hangup? |
23:19.59 | dlynes_ | justinu: correct, but the problem manifests itself on nortel cics keysystems too...not just asterisk |
23:20.00 | justinu | or call it from certain networks? |
23:20.24 | justinu | that's bizzare |
23:20.24 | dlynes_ | justinu: i don't know if it's all networks or not...the only networks i can test it from are fido |
23:20.42 | dlynes_ | but the problem only seems to exist on telus lines |
23:20.42 | justinu | does an IVR answer the line? |
23:20.50 | dlynes_ | For the asterisk box, yes |
23:20.53 | dlynes_ | for the nortel box, no |
23:20.56 | justinu | k |
23:21.24 | justinu | i dunno what fido or telus are... canadian ilecs? |
23:21.38 | dlynes_ | doesn't seem to matter whether i'm using an x100p card, or a sipura 3000, either |
23:21.48 | dlynes_ | fido is a clec, telus is an ilec, allstream is a clec |
23:22.01 | *** join/#asterisk bzbw (i=bwz@ip67-153-142-109.z142-153-67.customer.algx.net) |
23:22.05 | dlynes_ | all canadian, yes |
23:22.19 | dlynes_ | Fido is also known as Rogers |
23:22.24 | dlynes_ | Telus is part of Stentor |
23:22.30 | dlynes_ | Allstream is owned by AT&T |
23:23.04 | justinu | it almost sounds like your ilec doesn't have disco supervision with some of the networks, but that seems really hard to believe |
23:23.55 | dlynes_ | stranger things have happened |
23:24.26 | dlynes_ | one night I couldn't place any calls to certain co's on the fido network because fido's peering with group telecom was down |
23:24.38 | dlynes_ | that was totally fubar |
23:24.44 | justinu | wow |
23:25.09 | dlynes_ | 604628xxxx was working, but not 604484xxxx |
23:25.14 | dlynes_ | it was really strange |
23:25.25 | dlynes_ | I'm a customer of both companies, so I was able to bitch at both of them :) |
23:25.35 | *** join/#asterisk phonic (i=phonic@antisocial.nu) |
23:26.25 | justinu | so bitch at them about their shitty disco sup |
23:26.58 | *** join/#asterisk shido6 (n=shido6@d38-45-81.commercial1.cgocable.net) |
23:27.54 | phonic | how do i disable or enable a port in my tele-pci card? |
23:28.01 | dlynes_ | yeah..never had a problem with group telecom in taht respect...their equipment works great |
23:28.17 | dlynes_ | but, yeah...i'll talk to telus about it i guess then |
23:28.41 | phonic | i will make sure asterisk is using port 1 |
23:29.53 | *** join/#asterisk zariok (n=zariok@24-217-9-202.dhcp.stls.mo.charter.com) |
23:30.10 | *** join/#asterisk shido6 (n=shido6@d38-45-81.commercial1.cgocable.net) |
23:31.11 | *** join/#asterisk SplasPood (n=jwb@brooklyn.paravolve.net) |
23:31.22 | *** join/#asterisk b4ka (i=WinNT@200-127-198-118.cab.prima.net.ar) |
23:31.47 | sevard | OH SHIT OH SHIT ON SHIT OH SHOT |
23:31.50 | sevard | OH FIUCK |
23:32.23 | sevard | i just upgraded asterisk and I didn't make make samples but all of my configs are gone |
23:32.25 | sevard | my configs were massive |
23:32.43 | sevard | why would it do that |
23:32.44 | sevard | why |
23:32.45 | sevard | why |
23:32.59 | dlynes_ | why would sevard not make backups? |
23:33.10 | sevard | I thought I did |
23:33.12 | sevard | oi thought i did |
23:33.15 | sevard | i thought i did |
23:33.35 | dlynes_ | and why would you blindly type 'make samples'? |
23:33.44 | dlynes_ | without knowing what it would do to a live install? |
23:33.45 | CunningPike | sevard: For next time, 'OVERWRITE=no' in your Makefile |
23:33.52 | sevard | i NOT type make samples |
23:33.54 | dlynes_ | oh |
23:34.02 | dlynes_ | you installed from a binary package? |
23:34.04 | sevard | yup |
23:34.12 | dlynes_ | damn |
23:34.13 | sevard | FUCK |
23:34.23 | dlynes_ | for slackware? |
23:34.28 | sevard | yes |
23:34.29 | dlynes_ | was it your own binary package? |
23:34.31 | sevard | yes |
23:34.44 | sevard | i removepkg the old one and installpkg the new one |
23:35.01 | sevard | i thought i made a fucking back up :( |
23:35.01 | dlynes_ | sevard: a hint for next time |
23:35.07 | dlynes_ | sevard: when you make the staging directory |
23:35.38 | dlynes_ | sevard: rename all the /staging/etc/asterisk/*.* to *.*-sample |
23:37.07 | dlynes_ | sevard: also, make a backup copy of your /var/spool/asterisk and /var/lib/asterisk/keys, /var/lib/asterisk/licenses/ directories |
23:37.24 | sevard | there is no way i can recover this can i |
23:37.26 | sevard | i'm so fucked |
23:37.29 | sevard | i'm fucked more than ever |
23:38.14 | dlynes_ | do you have any old backups? |
23:38.52 | *** join/#asterisk Gamercjm (n=chris@pool-71-254-177-36.lsanca.fios.verizon.net) |
23:38.52 | dlynes_ | guess not |
23:38.53 | *** join/#asterisk elg (n=fugalh@falcon.fugal.net) |
23:39.14 | justinu | lol |
23:39.35 | dlynes_ | I'm sure he doesn't think it's so funny |
23:39.37 | *** join/#asterisk ghost99 (n=neville@222-152-219-77.jetstream.xtra.co.nz) |
23:40.04 | justinu | i'm allowed to laugh, having been in similar situations |
23:40.07 | dlynes_ | CunningPike: besides...he did a removepkg |
23:40.08 | VoicePulse | Did you pastebin any of your configs? |
23:40.19 | dlynes_ | CunningPike: OVERWRITE=no won't solve that problem |
23:40.25 | *** join/#asterisk bjohnson (n=bjohnson@i216-58-62-76.cybersurf.com) |
23:40.40 | dlynes_ | CunningPike: because he had all the /etc/asterisk/*.* with the original filenames |
23:40.47 | CunningPike | dlynes_: I guess not - at the outset, I thought he was installing from source |
23:41.01 | dlynes_ | CunningPike: He staged from source, and installed from binary package |
23:41.04 | sevard | wish reiserfs has an undelete |
23:41.06 | dlynes_ | CunningPike: It's the same way I do it, too |
23:41.23 | dlynes_ | CunningPike: But, I do a test on the package before I actually try deploying it any where |
23:41.55 | CunningPike | dlynes_: I install from source - on a test server first |
23:42.16 | dlynes_ | CunningPike: exactly...staging is almost the same, but there's one more step |
23:42.45 | dlynes_ | CunningPike: I just build everything into /usr/local/src/staging first |
23:42.48 | sevard | looks like reiserfs 'undelete' only works when the directory you wish to undelete is a partition |
23:43.04 | dlynes_ | CunningPike: then edit /usr/local/src/staging/etc/asterisk/asterisk.conf |
23:43.17 | dlynes_ | CunningPike: then edit /usr/local/src/staging/usr/sbin/safe_asterisk |
23:43.26 | dlynes_ | CunningPike: check all symbolic links to make sure they're relative |
23:43.40 | dlynes_ | CunningPike: and then make a package |
23:43.58 | dlynes_ | after i've renamed the files in the /etc/asterisk directory in the staging directory, that is |
23:44.29 | dlynes_ | It's a bit more work than compiling from scratch |
23:44.41 | dlynes_ | but i don't need to put a build load on the target machine then |
23:45.06 | dlynes_ | and i can easily advance/revert version numbers |
23:45.49 | CunningPike | dlynes_: Makes sense - especially from a loading perspective as our production server gets busier |
23:46.02 | dlynes_ | Yeah...it doesn't make any difference to me now |
23:46.09 | dlynes_ | but when the server is at capacity |
23:46.12 | dlynes_ | it makes a huge difference |
23:46.17 | dlynes_ | I'd rather be prepared ahead of time |
23:46.27 | CunningPike | dlynes_: Good thinking |
23:46.48 | *** join/#asterisk Weezey (n=ohno@206.210.109.235) |
23:47.03 | dlynes_ | George went home already, Weezey. |
23:47.12 | Weezey | that ass |
23:47.48 | Weezey | what's the cheapest way to get a zaptel timer in a box? |
23:48.36 | dlynes_ | modprobe ztdummy |
23:48.52 | *** join/#asterisk terrapen (n=cjs@166.70.183.109) |
23:48.57 | dlynes_ | Assuming you're using a 2.6 kernel |
23:49.02 | *** join/#asterisk xcoyote (n=farfan@201.135.194.118) |
23:49.03 | terrapen | does anybody have b2llc's new ftp server login? |
23:49.35 | xcoyote | question: does exists api documentation for the asterisk source code ? |
23:49.53 | dlynes_ | xcoyote: make docs |
23:50.02 | terrapen | (voipsupply) |
23:51.15 | znoG | dlynes_: nope, direct to Asterisk |
23:51.22 | Weezey | terrapen: they changed it? |
23:51.56 | dlynes_ | znoG: huh? that response was so delayed, i don't even remember what i was talking to you about |
23:52.08 | Weezey | terrapen: damn, they changed it. call them. |
23:52.18 | znoG | dlynes_: you asked me if I had to NAT between the ATAs and Asterisk |
23:52.33 | dlynes_ | ah |
23:52.49 | dlynes_ | can't remember what the problem was, either :) |
23:53.06 | dlynes_ | just that it was something to do with sip and pap2-na's |
23:54.06 | dlynes_ | sounds like a lot of the digium servers are going down now :) |
23:54.10 | terrapen | weezey, i just tried |
23:54.13 | terrapen | nobody answering |
23:54.59 | terrapen | i really need some 501 firmw4rez |
23:55.18 | terrapen | why the hell doesn't polycom just provide it? |
23:55.26 | terrapen | it's not like i didn't buy an assload of their phones |
23:55.42 | dlynes_ | terrapen: they do, but to their channel partners, not end consumers |
23:56.08 | dlynes_ | terrapen: if you buy that many phones, why not just become a channel partner? |
23:56.23 | terrapen | maybe i should. |
23:56.35 | terrapen | do you have the latest IP501 fw/br? |
23:56.47 | dlynes_ | nope...i'm not a channel partner though, either |
23:57.00 | terrapen | certainly someone here must have some |
23:57.01 | dlynes_ | We've filled out all the necessary paperwork already though |
23:57.05 | CunningPike | Bootrom: http://support.gafana.com/polycom/bootrom_313.zip |
23:57.05 | CunningPike | SIP: http://support.gafana.com/polycom/SoundPoint_IP_SIP_1_6_5.zip |
23:57.08 | terrapen | how many do you have to buy? |
23:57.11 | dlynes_ | now we need to write all the online tests |
23:57.20 | dlynes_ | not really a limit |
23:57.24 | terrapen | is that fairly recent, cunning? |
23:57.28 | CunningPike | From a recent posting to asterisk-users |
23:57.32 | terrapen | sweet |
23:57.48 | CunningPike | 3.1.3 is the current bootrom - Sip 1.6.6 just came out |
23:57.58 | CunningPike | But 1.6.5 is current |
23:58.15 | terrapen | i have BootROM 2.6.2 and sip 1.6.3 |
23:58.29 | terrapen | hrmm...shouldn't the IP601 bootrom work for the IP501, too? |
23:59.17 | *** join/#asterisk ringhals (i=fwuser@firewall.drgutah.com) |
23:59.26 | CunningPike | terrapen: Yes - we have it on both |
23:59.33 | CunningPike | It's the same |
23:59.46 | terrapen | ok, maybe my configs are just wrong...for some reason this new 501 was saying invalid bootrom |
23:59.53 | terrapen | and the bootrom is there and works just fine on the 601 |
23:59.59 | CunningPike | What is your current bootrom |