| 00:00.09 | }btorch{ | I'm changing nothing on zaptel and libpri and I'm just recompiling them because I think some guy from digium messed them up |
| 00:00.58 | dlynes_office | justinu: yeah...vancouver is definitely a friendly city |
| 00:01.20 | dlynes_office | justinu: lots of peeps end their workday on friday at 2 or 3pm |
| 00:01.30 | justinu | nice |
| 00:01.37 | jake1932 | I'd be surprised - the digium people know Asterisk pretty well |
| 00:01.46 | Flauto | i will pass by vancouver by the end of this month |
| 00:01.47 | dlynes_office | CunningPike and I aren't two of the lucky ones though :( |
| 00:02.14 | [hC] | Me either |
| 00:02.16 | [hC] | I work late :) |
| 00:02.21 | [hC] | then again |
| 00:02.27 | [hC] | i start my day at about 10:30 or 11 |
| 00:02.32 | dlynes_office | then again, you're your own boss, too |
| 00:02.39 | [hC] | There's also that... |
| 00:03.08 | Flauto | i dont' see full under /var/log/asterisk |
| 00:03.10 | dlynes_office | Flauto: where you headed that's bringing you through vancouver? |
| 00:03.19 | dlynes_office | Flauto: did you do a logger reload? |
| 00:03.26 | dlynes_office | Flauto: or a reload? |
| 00:03.28 | file | libpri doesn't depend on zaptel unless you use the test utilities |
| 00:03.28 | *** join/#asterisk cpowers (n=cpowers@ip68-3-58-247.ph.ph.cox.net) |
| 00:03.32 | Flauto | i am going to beijing china |
| 00:03.33 | Flauto | yes |
| 00:03.34 | Flauto | i did |
| 00:03.39 | dlynes_office | Flauto: ah |
| 00:03.43 | dlynes_office | Flauto: You'll love it there |
| 00:03.56 | dlynes_office | Flauto: I went there a few years back...it was one of the best times of my life |
| 00:04.03 | Flauto | i will stay a whole afternoon and over night |
| 00:04.06 | dlynes_office | Flauto: but you're going to cook your ass off this time of the year |
| 00:04.07 | Flauto | leaving the next day |
| 00:04.32 | dlynes_office | Flauto: it'll be around 90F every day, unless it's overcast or raining |
| 00:04.39 | Flauto | hehe |
| 00:04.41 | dlynes_office | Flauto: then it might drop down to about 75 or 80F |
| 00:04.45 | justinu | heh |
| 00:05.06 | dlynes_office | Flauto: and it doesn't cool down at night, either |
| 00:05.40 | dlynes_office | Flauto: if you're going to cross a street, make sure nobody's coming, and then run like the dickens |
| 00:05.45 | justinu | lol |
| 00:05.47 | dlynes_office | Flauto: automobiles don't stop for pedestrians |
| 00:05.48 | justinu | sounds like LA |
| 00:06.06 | dlynes_office | Flauto: nor for cyclists, for that matter |
| 00:06.08 | justinu | all my NYC friends say: "we can always tell you LA people... you won't step off the curb until the sign says "walk" |
| 00:06.12 | justinu | and it's true! |
| 00:06.28 | dlynes_office | I seen this one cyclist get sandwiched between a car and a delivery van on the freeway exit |
| 00:06.40 | *** part/#asterisk cpowers (n=cpowers@ip68-3-58-247.ph.ph.cox.net) |
| 00:06.44 | dlynes_office | The cyclist was slightly injured, but his bike was a twisted mess |
| 00:07.15 | dlynes_office | I even seen a public bus hop over the meridian on the freeway and do a u-turn |
| 00:07.16 | *** join/#asterisk wunderkin (i=kev@ip68-226-113-228.ph.ph.cox.net) |
| 00:07.19 | dlynes_office | that was funny as hell |
| 00:07.51 | dlynes_office | Flauto: btw...make sure you get to see the summer palace |
| 00:07.57 | dlynes_office | Flauto: it's definitely a must see |
| 00:08.08 | dlynes_office | Flauto: and the forbidden city |
| 00:08.42 | CunningPike | dlynes_office: Sure explains some of the driving I see around here...... |
| 00:08.49 | Flauto | now, i got full there |
| 00:08.52 | dlynes_office | I must've seen something like 25 temples, forbidden city, great wall, summer palace, min tombs, ... when i was there |
| 00:09.15 | dlynes_office | I was on the go for about 16-18 hours a day |
| 00:09.16 | CunningPike | dlynes_office: I'm buying a scooter next weekend for the commute |
| 00:09.30 | dlynes_office | CunningPike: lol...to north van from richmond? |
| 00:09.41 | justinu | you ride motorcycles before? |
| 00:09.43 | CunningPike | dlynes_office: Yup :D |
| 00:10.06 | dlynes_office | cool...record the sound of the gears grinding trying to get up those north van hills for me :) |
| 00:10.30 | dlynes_office | justinu: yeah...I used to write a Honda Nighthawk 450 |
| 00:10.30 | CunningPike | justinu: Not really - I used to bicycle the route quite often - but it's too far to cycle every day |
| 00:10.38 | justinu | be careful man |
| 00:10.52 | justinu | i quit riding here... too many psychos |
| 00:10.57 | CunningPike | justinu: I intend to - I'll be sticking to the quieter streets where I can |
| 00:11.01 | dlynes_office | justinu: it's not so bad here...it's bad, but most drivers are aware of motorcyclists here |
| 00:11.11 | justinu | cool |
| 00:11.26 | dlynes_office | justinu: lots of motorcycles on the road in the summer here |
| 00:11.40 | CunningPike | It's one of the things I noticed when I first moved here - traffic is relatively benign |
| 00:11.53 | CunningPike | It's getting worse, but it's still heaps better than most places |
| 00:11.54 | dlynes_office | CunningPike: benign? vancouver? hehehhehehee |
| 00:11.59 | Flauto | i am looking at the file now |
| 00:12.08 | Flauto | what should i look for |
| 00:12.16 | dlynes_office | Flauto: the full file? |
| 00:12.17 | CunningPike | dlynes_office: Oh, yes - I'm from Dublin. |
| 00:12.36 | dlynes_office | CunningPike: well...hehehe...i was comparing vancouver to other Canadian cities though :) |
| 00:12.52 | dlynes_office | Flauto: try searching for the phrase 'invalid context' |
| 00:13.10 | CunningPike | dlynes_office: It's better than TO.... |
| 00:13.23 | dlynes_office | CunningPike: that's debatable :) |
| 00:13.34 | dlynes_office | CunningPike: they're both bad...it's pretty hard to pick which one's worse :0 |
| 00:13.42 | dlynes_office | CunningPike: Montreal's pretty good, though |
| 00:13.46 | Flauto | that is a lot to go through |
| 00:13.47 | Flauto | hehe |
| 00:13.49 | Flauto | okay |
| 00:14.13 | dlynes_office | Flauto: grep "invalid context" full is your friend |
| 00:14.26 | dlynes_office | Flauto: then no need to view the log file |
| 00:15.21 | dlynes_office | cool |
| 00:15.33 | dlynes_office | 1.2.8 is going to be released shortly |
| 00:15.42 | CunningPike | dlynes_office: Reminds me - I have to phone ICBC - someone backed into our van at the mall on the weekend |
| 00:15.49 | dlynes_office | doh |
| 00:15.57 | Flauto | nothing came up |
| 00:15.58 | dlynes_office | i hate dealing with icbc |
| 00:16.00 | dlynes_office | buncha crooks |
| 00:16.16 | dlynes_office | Flauto: try 'grep "sent into" full' |
| 00:16.29 | *** join/#asterisk Johnnie (n=jdlewis@pdpc/supporter/active/Johnnie) |
| 00:16.43 | dlynes_office | Flauto: also are you logging at full logging level? |
| 00:17.29 | Flauto | yes |
| 00:17.58 | dlynes_office | not getting a hit on 'sent into', either? |
| 00:19.16 | Flauto | right |
| 00:19.35 | dlynes_office | The remote end that's dialing into your box via iax2 |
| 00:19.44 | dlynes_office | Do you control that box as well? |
| 00:21.17 | *** join/#asterisk syle (n=blah@unaffiliated/syle) |
| 00:25.49 | CunningPike | dlynes_office: I prefer dealing with them than the private compaines |
| 00:27.04 | dlynes_office | CunningPike: you mean like hsbc's insurance company (Canadian Direct)? |
| 00:32.15 | CunningPike | dlynes_office: Our home insurance is with them - it's just in Ireland, where there is commercial motor insurance, it's impossible to get insurance at all if your risk is even a scintilla above zero |
| 00:32.38 | dlynes_office | CunningPike: so, basically i'd be screwed |
| 00:32.48 | dlynes_office | CunningPike: icbc bastards doubled my insurance this year |
| 00:32.53 | CunningPike | All the companies cherry-pick the low-risk drivers, and quote 5-figure sums to everyone else |
| 00:32.57 | CunningPike | What did you do? |
| 00:33.39 | CunningPike | It's kinda like what was happening in the Maritmes |
| 00:34.16 | CunningPike | OK - dinner calls - bbl |
| 00:35.21 | *** join/#asterisk chaoscon (n=ph33r@smartserv/ceo/chaoscon) |
| 00:36.32 | *** join/#asterisk coppice (n=chatzill@32.193.0.218.broad.hz.zj.dynamic.cndata.com) |
| 00:37.28 | *** join/#asterisk |dennis| (i=dennis@200.32.215.84) |
| 00:44.12 | dlynes_office | I got a slight fender bender...did "$2500 in damages to their minivan" and about $800 in damages to my car |
| 00:45.51 | *** join/#asterisk schirpich (n=dvs@ip21.farheap.net) |
| 00:46.45 | *** part/#asterisk diclophis (n=diclophi@65.203.37.58) |
| 00:47.35 | jsaunders | Beauty |
| 00:47.43 | *** join/#asterisk |dennis| (i=dennis@200.32.215.84) |
| 00:48.22 | dlynes_office | Well, probably woulda cost me about $1200-1500 if I didn't do the repairs myself :) |
| 00:48.38 | jsaunders | If I can, I prefer to do them myself. |
| 00:48.51 | *** join/#asterisk justinu|laptop (n=Justin@dsl081-083-093.lax1.dsl.speakeasy.net) |
| 00:49.23 | *** join/#asterisk nassy (n=nassy@207-38-197-201.c3-0.wsd-ubr1.qens-wsd.ny.cable.rcn.com) |
| 00:50.28 | *** part/#asterisk nassy (n=nassy@207-38-197-201.c3-0.wsd-ubr1.qens-wsd.ny.cable.rcn.com) |
| 00:51.37 | [TK]D-Fender | jsaunders: Start a shell company on auto-repair, bill yourself excessively and claim it on insurance :) |
| 00:52.23 | jsaunders | You know.... that's not such a bad idea. |
| 00:53.21 | *** join/#asterisk Freman (n=twitsrus@jaguar.wbs.net.au) |
| 00:53.30 | *** join/#asterisk jayk- (i=jayk@lasziv.reprehensible.net) |
| 00:53.39 | jayk- | anybody know where i can get the replacement asterisk sounds? |
| 00:53.41 | Freman | I fixed all my problems from last night (RE: dial a group of extensions so long as one of them isn't busy) |
| 00:54.03 | Freman | I give you the solution in case I'm not the only person in the world that needed it: http://fremnet.net/article/194 |
| 00:55.04 | *** join/#asterisk adker (n=adker@67-136-210-63.dsl1.glv.ny.frontiernet.net) |
| 00:56.42 | [TK]D-Fender | Freman : Umm... I'm not sure I see the point or goal in there... |
| 00:58.12 | *** join/#asterisk Exstatica (i=exstatic@redline.mednor.net) |
| 00:58.19 | [TK]D-Fender | Freman : and that out of context dialstatus jumping.. ICK |
| 01:01.10 | Freman | the goal is I have a 2 line pap system with two phones off it... I'm only one person and don't want to have to answer the second line while I'm on the first one. |
| 01:01.33 | Freman | if I just dial(line1&line2) if I'm on line1, line2 will ring. |
| 01:01.51 | Freman | using that code I can check to see if I'm on one of the extensions before dialing |
| 01:02.02 | Freman | and the jumping thing is only for 'example' |
| 01:02.05 | *** join/#asterisk paolob (n=donpaolo@pri-214-b7.codetel.net.do) |
| 01:03.11 | [TK]D-Fender | Freman : for 2 extens that would take at most 4 lines of dial-plan to do..... |
| 01:03.24 | pjchilds | neat... now if you could 'pip' a call waiting tone into your active extension, and allow switching and conferencing that would just be way over the top... :) |
| 01:03.43 | [TK]D-Fender | Freman : Perhaps you should take a look at "ChanIsAvail" |
| 01:04.01 | Freman | I don't like call waiting... |
| 01:04.16 | Freman | [TK]D-Fender: I spent an hour reading documentation and didn't find that... )c; |
| 01:04.46 | pjchilds | who does like call waiting --- but perhaps how about 'call waiting for people I think are important...' |
| 01:04.48 | [TK]D-Fender | ChanIsAvail(SIP/1,sj) <- 1 down, 1 to go... |
| 01:05.28 | Freman | pjchilds: the full code actualy has "emergency code" in place so that if I am on the phone and it is an emergency then the other line will ring |
| 01:05.37 | [TK]D-Fender | pjchilds : callerID based routing to an exten thats already busy. Easy enough, but augly to hardcode if you have more than a few people to search for. |
| 01:05.40 | cytrak | ok this blows ... I got my cards on their own IRQs and I disabled serial, usb, sound, parallel port ... |
| 01:06.23 | cytrak | now my sound is pretty low for some reason and my calls have some zzzzz noise very low in the background |
| 01:06.39 | [TK]D-Fender | cytrak : What cards? |
| 01:06.52 | cytrak | TE205P and TE110P |
| 01:07.10 | [TK]D-Fender | cytrak : Ummm... why not a single 4 port card? |
| 01:07.30 | *** join/#asterisk MacDome (n=eseidel@c-71-198-177-144.hsd1.ca.comcast.net) |
| 01:07.35 | cytrak | because I already had a 110 |
| 01:08.01 | [TK]D-Fender | cytrak : Well hum really doesnt' make sense.. its a digital circuit.... |
| 01:08.20 | cytrak | what doesn't make sense ? having both cards ? |
| 01:08.42 | [TK]D-Fender | cytrak : The "zzzzz" hum you describe. Missing interrupts? |
| 01:08.58 | [TK]D-Fender | cytrak : What does ZTTOOL tell you? |
| 01:09.04 | *** join/#asterisk eeya (n=eeya@221.221.27.140) |
| 01:09.27 | cytrak | don't seem like a have that , zttool is normal, interrupts look good and zztest is 100% |
| 01:09.57 | cytrak | just the sound quality is not great |
| 01:09.59 | [TK]D-Fender | cytrak : CPU load? |
| 01:10.12 | cytrak | it coul dbe better if I new what else to tweak |
| 01:10.16 | [TK]D-Fender | cytrak : All ports, or just 1 card? |
| 01:10.46 | cytrak | cpu load is fine too 0.3% |
| 01:11.14 | Freman | ChanIsAvail(sip/1&sip/2) check ${AVAILCHAN}well now that I've re-invented the wheel I'll go home and try chanisavail... |
| 01:11.18 | Freman | doh |
| 01:11.20 | [TK]D-Fender | cytrak : Let me clarify, is it only 1 card that does this? |
| 01:11.24 | Freman | well now that I've re-invented the wheel I'll go home and try chanisavail... |
| 01:11.28 | cytrak | all ports ... when I call outside I use span 1 which I have rxgain txgain to 0 |
| 01:11.39 | [TK]D-Fender | Freman : But yours MUST be rounder! ;) |
| 01:12.01 | *** join/#asterisk MacDome (n=eseidel@c-71-198-177-144.hsd1.ca.comcast.net) |
| 01:12.06 | [TK]D-Fender | cytrak : Tested on another system? |
| 01:12.12 | Freman | well, the get_channels.pl can be used for other things (c: |
| 01:12.24 | cytrak | span 2 &3 I have rxgain = -6 and txgain =-16 since those worked better for me |
| 01:12.31 | [TK]D-Fender | Freman : There's the spirit! Recycle! |
| 01:12.39 | [TK]D-Fender | cytrak : OUCH.... |
| 01:12.43 | cytrak | [TK]D-Fender: no don't really have one right now |
| 01:12.47 | Smi|k | anyone here work with ecommerce a lot? |
| 01:12.54 | *** join/#asterisk Stereo (n=stereo@hertz.stereo.lu) |
| 01:13.05 | cytrak | [TK]D-Fender: why what you use ? |
| 01:13.10 | Stereo | hi everyone |
| 01:13.11 | *** join/#asterisk supjigatr (n=syslod@152.53.16.10) |
| 01:13.35 | [TK]D-Fender | cytrak : Well my work server is a Supermicro, but I'm not on Digium cards. |
| 01:13.35 | supjigatr | Hi. |
| 01:14.14 | supjigatr | Anyone point me into a debug track for Spandsp rxfax? Sometimes it works then it just quits. |
| 01:14.33 | cytrak | I have a regular optiplex P4 2.8ghz 1.5gb ram |
| 01:15.41 | [TK]D-Fender | cytrak : Don't know about their workstations but several Dell servers are known to be DOA choices for Digium boards. |
| 01:16.00 | [TK]D-Fender | cytrak : pastebin a "cat /proc/interrupts" if you can. |
| 01:16.05 | cytrak | yeah that's what I heard |
| 01:16.24 | supjigatr | Sangoma fixes digum wierdness? |
| 01:16.34 | cytrak | i wanted to get a penquim computer server but .... |
| 01:16.39 | [TK]D-Fender | Supaplex : replaces it at the very least ;) |
| 01:16.39 | pjchilds | cytrak: why the different tx/rx gains on different spans? have you milliwatt tested and balanced them all? |
| 01:18.22 | cytrak | pjchilds: well when I first had only the te110p card I only had it connected to my PBX so only local calls could be done between iax2 client <-> PBX clients and those gains were the best ones that eliminated my echo problems and sound wierdness |
| 01:18.36 | cytrak | I used ztmonitor for that |
| 01:18.51 | cytrak | I haven't done it again since I reconfigured the setup |
| 01:19.21 | pjchilds | cytrak: might be worth having a bash ... |
| 01:20.03 | cytrak | on me ? |
| 01:20.47 | cytrak | i should gonna go home watch gray's anatomy |
| 01:21.00 | pjchilds | cytrak: hmm... perhaps thats a localized expression 'having a bash..' == 'having a go' == 'might be worth trying...' |
| 01:22.05 | [TK]D-Fender | cytrak : Any chance for that pastebin? |
| 01:22.14 | cytrak | [TK]D-Fender: opps sorry |
| 01:22.28 | cytrak | on the way |
| 01:23.38 | cytrak | http://pastebin.com/719863 |
| 01:23.56 | cytrak | yeah I got regulate the TX gain for span 1 . its pretty low |
| 01:24.34 | [TK]D-Fender | cytrak : care to slap up "dmesg" as well? |
| 01:24.59 | cytrak | sorry I'm heading out now but thanks, maybe tomorrow |
| 01:26.38 | [TK]D-Fender | cytrak : Ok, take it easy |
| 01:26.44 | cytrak | later |
| 01:27.08 | Freman | hey [TK]D-Fender I found an example that does more or less exactly what I want to do (c: http://tinyurl.com/afysd (example 2 revised)... Still I couldn't find that last night after an hour of looking |
| 01:29.04 | [TK]D-Fender | Freman : what are your 2 extens? |
| 01:29.18 | Freman | sip/9100 and sip/9101 |
| 01:29.36 | Freman | from the internal context: 9100 and 9101 |
| 01:32.19 | *** join/#asterisk danzig (n=chatzill@130.226.169.177) |
| 01:33.55 | [TK]D-Fender | Freman : Small sample http://pastebin.com/719869 |
| 01:34.37 | *** join/#asterisk lzhang (n=rjrae@67.95.13.46) |
| 01:34.47 | lzhang | has anybody used app_amd? |
| 01:34.55 | Freman | oh ok, that'll work also |
| 01:34.56 | Freman | thanks |
| 01:35.38 | Freman | perhaps I should give you my entire extensions file to revise (c: |
| 01:35.59 | *** join/#asterisk tardisx (n=justin@ppp167-251-29.static.internode.on.net) |
| 01:36.23 | [TK]D-Fender | Freman : Perhaps :) |
| 01:37.13 | Freman | I've basicly re-invented the privacy function - but better (c: |
| 01:37.31 | supjigatr | would this cause calls to drop and rxfax to not work? HDLC Bad FCS |
| 01:37.38 | [TK]D-Fender | Freman : Dunno... thats a heavy system call ;) |
| 01:37.41 | Agrajag- | gday - does anyone happen to know that if i have a sipura spa-3000 which is registered to my asterisk box, if it's possible, when an incoming call comes in from the pstn line plugged into the spa-3000, to make the call go to the asterisk box so that it can make any number of SIP phones in the house ring? (including the phone connected to the spa-3000) |
| 01:38.04 | [TK]D-Fender | supjigatr : Um yeah, failing to synch will kill all sorts of stuff... |
| 01:38.18 | tardisx | Agrajag-: sure |
| 01:38.25 | [TK]D-Fender | Agrajag- : yup |
| 01:38.41 | supjigatr | How do I debug/fix it? |
| 01:38.43 | [TK]D-Fender | Agrajag- : thats the point. Its not just a PSTN failover, its a Gateway |
| 01:39.19 | Agrajag- | [TK]D-Fender/tardisx: excellent, thanks. |
| 01:39.37 | [TK]D-Fender | supjigatr : Make sure your car is on its own IRQ, that you aren't running any of the incompatible kernal modules (e1000, i7105 ChipSet, etc) |
| 01:41.10 | dlynes_office | Agrajag-: comopletely possible |
| 01:41.18 | supjigatr | IRQ looks ok http://pastebin.ca/56095 |
| 01:41.34 | supjigatr | e1000 I think is built into the MB. Its a MPC server. |
| 01:42.00 | dlynes_office | Agrajag-: there's a number of good tutorials for getting the sipura 3000 set up as a gateway on Voxilla's Sipura Users' Group Forum |
| 01:42.04 | [TK]D-Fender | supjigatr : Pastebin "dmesg" |
| 01:42.42 | dlynes_office | supjigatr: chances are your e1000 is sharing an interrupt with your zaptel card |
| 01:43.21 | [TK]D-Fender | dlynes_office : Not according to his interrupts list, but it doesn't mean its not interfering |
| 01:43.38 | supjigatr | http://pastebin.ca/56096 |
| 01:43.56 | dlynes_office | [TK]D-Fender: ah...yeah...I had that problem, myself...the e1000 wasn't showing as sharing on cat /proc/interrupts, but it sure as hell was on lspci -v |
| 01:44.07 | supjigatr | proc/pci indicates it on another IRQ. |
| 01:44.24 | supjigatr | zttool reports no problems. |
| 01:45.10 | dlynes_office | supjigatr: do you have dma enabled on your hard drives? |
| 01:45.36 | supjigatr | Its a LSI raid controller. |
| 01:45.40 | supjigatr | How do I check? |
| 01:45.44 | [TK]D-Fender | supjigatr : Yup.. the evil E1000 strikes again |
| 01:45.47 | supjigatr | haha |
| 01:45.56 | supjigatr | Is there any way to use it? |
| 01:46.00 | dlynes_office | [TK]D-Fender: yeah...that thing is pure evil |
| 01:46.08 | [TK]D-Fender | supjigatr : With another T1 card :) |
| 01:46.11 | supjigatr | Haha. |
| 01:46.22 | supjigatr | Got a sangoma u'll drive to my office? |
| 01:46.33 | dlynes_office | [TK]D-Fender: I've got it disabled, my pri card on a different slot, no driver loaded, and it's still mucking with crap |
| 01:46.33 | [TK]D-Fender | dlynes_office : No it isn't... only 1 thing I've seen doesn't like them! |
| 01:46.56 | [TK]D-Fender | supjigatr : Sorry, you'll have to order one yourself :) |
| 01:47.09 | dlynes_office | [TK]D-Fender: well, the only thing I've had a problem with, with the e1000 is the digium card |
| 01:47.20 | supjigatr | IS there anyway to cluge this till sangoma and NDA me a card? |
| 01:47.20 | [TK]D-Fender | dlynes_office : Exactly! |
| 01:47.47 | [TK]D-Fender | supjigatr : Sorry, I'm just a consumer :) |
| 01:48.10 | supjigatr | So disabling and putting a realtek or something won't fix it? |
| 01:48.15 | [TK]D-Fender | supjigatr : Let me guess, you're running Slackware 10.1 on that box? |
| 01:48.21 | supjigatr | 10.2 I think |
| 01:48.35 | [TK]D-Fender | supjigatr : You can TRY disabling it, but I went through 2 revisions of TE405P on mine. |
| 01:48.36 | dlynes_office | what does slackware have to do with anything? |
| 01:48.55 | [TK]D-Fender | supjigatr : 10.1 use 2.4.29, 10.2 used 2.4.31 :) |
| 01:49.06 | dlynes_office | ah...both work fine for me though |
| 01:49.12 | [TK]D-Fender | dlynes_office : Absolutely nothing, just an observation from a fellow Slacker :) |
| 01:49.14 | dlynes_office | but I use my own 2.6 kernel on both of them |
| 01:49.16 | supjigatr | Yea I've been experimenting with kernels to try apci. |
| 01:49.29 | dlynes_office | i don't bother with the 2.4 kernels |
| 01:49.40 | supjigatr | I have 2.4.29 - 2.6.xx playing trying to fix. |
| 01:49.46 | [TK]D-Fender | dlynes_office : Once I learn how to switch to 2.6 I will. |
| 01:49.58 | supjigatr | I bought the MPC cause everyone said digium and it were great together. |
| 01:50.10 | supjigatr | slack 10.2 has 2.6 |
| 01:50.14 | dlynes_office | [TK]D-Fender: learn how to switch? |
| 01:50.26 | supjigatr | and its really easy to do 2.6 on 10.1 |
| 01:50.39 | dlynes_office | [TK]D-Fender: just install slackware 10.2, download 2.6.15.5 kernel source, compile, install, reboot |
| 01:50.44 | supjigatr | So I guess digium has screwed me? |
| 01:51.05 | dlynes_office | supjigatr: could just be a faulty card, too |
| 01:51.09 | [TK]D-Fender | dlynes_office : Switch kernels, configure GRUB, etc, install that tools set you need for boot up, etc. |
| 01:51.15 | dlynes_office | grub? |
| 01:51.23 | dlynes_office | i've never wasted my time with grub :) |
| 01:51.25 | [TK]D-Fender | dlynes_office : Err, LILO :) |
| 01:51.26 | supjigatr | I've got two of them. |
| 01:51.32 | supjigatr | One with echo can one without. |
| 01:51.35 | supjigatr | Both seem to suck. |
| 01:51.51 | supjigatr | I have like $2000 invested and its a paperweight I guess. |
| 01:54.40 | *** join/#asterisk Pegger (n=peg@pool-68-163-217-147.bos.east.verizon.net) |
| 01:55.35 | Pegger | I am trying to get asterisk working with unixodbc, but I am having trouble getting it to connect I keep on getting res_odbc: Error SQLConnect=-1 errno=0 [unixODBC][Driver Manager]Data source name not found, and no default driver specified i have a ssytem DNS set up do I need a user DNS set up also? |
| 01:55.49 | Qwell | Pegger: dns? dsn? |
| 01:56.01 | Pegger | the are a part of unixodbc |
| 01:56.12 | Qwell | you mean dsn, not dns |
| 01:56.37 | Pegger | yes dns I got it right the second time |
| 01:56.46 | *** join/#asterisk Johnnie (n=jdlewis@pdpc/supporter/active/Johnnie) |
| 01:57.05 | Qwell | Then, no, you don't need to setup DNS |
| 01:57.07 | dlynes_office | Pegger: dsn (data source name); dns (domain name service) |
| 01:57.38 | Pegger | yes I know I was one letter off let it rest already |
| 01:57.56 | Qwell | It's a very important distinction |
| 01:58.13 | Qwell | Did you setup the dsn in asterisk? |
| 01:58.30 | dlynes_office | Pegger: we weren't trying to correct your spelling; we were just making sure we were on the same page |
| 01:58.33 | Qwell | res_odbc.conf |
| 01:58.36 | *** join/#asterisk shido6 (n=shido6@d221-68-200.commercial.cgocable.net) |
| 01:59.25 | Pegger | i set it up in /etc/asterisk/res_odbc.conf and /etc/asterisk/extconfig.conf |
| 02:00.18 | Pegger | and I put switch => Realtime/mycontext@ext_switch in extensions.conf |
| 02:07.33 | Pegger | anyone |
| 02:11.14 | dlynes_office | Pegger: qwell's the expert here on odbc |
| 02:11.41 | Qwell | No he's no |
| 02:11.42 | Qwell | t |
| 02:11.46 | dlynes_office | lol |
| 02:12.49 | Pegger | Qwell, yes i have res_odbc.conf set up |
| 02:13.14 | *** part/#asterisk esandeen (n=sandeen@sandeen.net) |
| 02:13.16 | Qwell | setup properly? |
| 02:13.47 | *** join/#asterisk Evanrude (n=david@ip68-107-162-212.lu.dl.cox.net) |
| 02:14.18 | Pegger | Qwell, looks like it |
| 02:14.22 | *** join/#asterisk iq|mobile (n=iq@71-38-73-211.omah.qwest.net) |
| 02:14.36 | sevard | Qwell: have you ever had the problem where you have an extenion on boxA that rings an extension on boxB via iax, the call comes in through boxA, boxB says it's dialing the extension but the phone doesn't ring, it says Called 5555, but the phone doesn't ring ;\ |
| 02:15.58 | *** join/#asterisk raidenz (i=raiden@205-200-66-136.static.mts.net) |
| 02:16.15 | Qwell | firewall? |
| 02:16.36 | sevard | no, when the phone is registered to boxA or boxB and the extension is dialed without IAX coming into play it rings just fine |
| 02:16.40 | raidenz | Hi guys |
| 02:16.48 | Qwell | right |
| 02:16.49 | Qwell | firewall? |
| 02:16.52 | sevard | no. |
| 02:17.05 | sevard | there is no firewall between the two servers. |
| 02:18.04 | raidenz | Does anyone know where to get an updated test g729 ipp codec patch that works with the latest svn? |
| 02:19.07 | Qwell | raidenz: You don't |
| 02:19.51 | raidenz | no? |
| 02:39.29 | *** join/#asterisk Ixthod (n=Ixthod@198.174.206.41) |
| 02:43.14 | Faithful | anyone with experience with Aastra phones? |
| 02:43.40 | litage | hey guys, i just compiled and installed 1.2.7.1 and noticed that there aren't any "sip ..." commands within the asterisk CLI. how can i fix this? |
| 02:45.04 | litage | there any iax commands either... |
| 02:45.18 | justinu|laptop | your modules aren't loaded right then |
| 02:46.52 | [TK]D-Fender | litage : What do you have in modules.conf? |
| 02:47.11 | litage | justinu|laptop: i just checked /etc/asterisk/modules.conf and noticed that chan_sip, chan_iax2, etc are missing |
| 02:49.01 | [TK]D-Fender | litage : What about auto-load? |
| 02:49.13 | jake1932 | Faithful: your aastra phone doesn't work? |
| 02:49.17 | *** join/#asterisk Eggplant (n=none@dsl-216-155-212-221.cascadeaccess.com) |
| 02:52.41 | Snake-Eyes | fixed litage problem, recompiled and install asterisk :) |
| 02:52.52 | *** join/#asterisk eeya_1128 (n=eeya@221.221.27.140) |
| 02:53.33 | *** part/#asterisk eeya_1128 (n=eeya@221.221.27.140) |
| 02:53.54 | *** join/#asterisk eeya_1128 (n=eeya@221.221.27.140) |
| 02:53.58 | *** join/#asterisk Johnnie (n=jdlewis@pdpc/supporter/active/Johnnie) |
| 02:54.41 | eeya_1128 | hello |
| 02:55.23 | [TK]D-Fender | justinu : Sorry, I've only used one once :) |
| 02:56.05 | eeya_1128 | ? |
| 03:02.24 | *** join/#asterisk imperfect- (n=tbw@c-68-58-148-186.hsd1.in.comcast.net) |
| 03:02.26 | imperfect- | Evening. |
| 03:02.44 | imperfect- | I'm attempting to dialout with broadvoice using asterisk, and I keep getting Got SIP response 604 "Does Not Exist Anywhere" back from 147.135.12.128 |
| 03:03.07 | imperfect- | Anyone know what I could be doing wrong? When I dial the number it attempts to dial bv at SIP/number@sip.broadvoice.com |
| 03:03.11 | *** join/#asterisk Olobola (n=casper_s@216.218.221.166) |
| 03:03.12 | imperfect- | seems right but again, I don't know |
| 03:04.20 | *** join/#asterisk wwalker (n=wwalker@pdpc/supporter/sustaining/wwalker) |
| 03:05.25 | wwalker | does anyone know how to make sjphone listen on a particular IP? I want to run sjphone and asterisk on the same machine for demoing and testing. |
| 03:05.55 | CunningPike | wwalker: Make SJPhone listen? |
| 03:06.42 | wwalker | it listens on 0.0.0.0:5060 ucp and tcp I need to narrow it to one IP |
| 03:06.58 | wwalker | so that |
| 03:07.13 | wwalker | asterisk can listen on 5060 on the other address |
| 03:08.19 | *** join/#asterisk viLeR (i=1000@200.114.70.228) |
| 03:10.21 | *** join/#asterisk eeya (n=eeya@221.221.27.140) |
| 03:12.08 | CunningPike | Hmm - I don't see any settings for that - does it actually tie up a port when it's running? |
| 03:15.43 | *** join/#asterisk Telamon (i=telamon@blk-222-22-126.eastlink.ca) |
| 03:15.51 | *** part/#asterisk Telamon (i=telamon@blk-222-22-126.eastlink.ca) |
| 03:20.36 | *** join/#asterisk supjigatr (n=syslod@152.53.17.26) |
| 03:22.42 | *** join/#asterisk catch23 (n=catch23@c-67-191-252-3.hsd1.ga.comcast.net) |
| 03:37.59 | *** join/#asterisk jeffgus (n=jeffgus@alpha.zimage.com) |
| 03:49.42 | *** join/#asterisk subdolus (n=subby@subby.afraid.org) |
| 03:58.12 | *** join/#asterisk bmg505 (n=leon@c1-242-7.rndf.isadsl.co.za) |
| 04:22.36 | asterboy | so what is the deal with SKype? |
| 04:22.50 | CunningPike | ~skype |
| 04:22.53 | jbot | it has been said that skype is evil. see gizmoproject.com |
| 04:22.59 | CunningPike | There ya go ;) |
| 04:23.17 | asterboy | well, that would explain why they are giving away FREE phone service now. |
| 04:23.43 | orlok | Hmm.. |
| 04:24.08 | CunningPike | Presumably only because it isn't costing them anything either, or they're prepared to eat the cost in order to grow their international businesss |
| 04:25.20 | *** join/#asterisk Faithful (n=Faithful@202-6-145-116.ip.adam.com.au) |
| 04:27.03 | asterboy | http://share.skype.com/sites/en/2006/05/free_calls_to_all_landlines_an.html |
| 04:27.19 | orlok | Can somebody have a look at some sip debugs for me? |
| 04:27.30 | orlok | Incoming calls are not working, and it seems to all be correct to me. |
| 04:27.43 | asterboy | jbot, skype is also offering free landlines: http://share.skype.com/sites/en/2006/05/free_calls_to_all_landlines_an.html |
| 04:27.44 | jbot | asterboy: okay |
| 04:28.09 | asterboy | sip debug peer orlok |
| 04:28.26 | orlok | asterboy: http://pastebin.com/719980 |
| 04:29.06 | *** join/#asterisk scardinal (n=supreme@port816.ds1-suoe.adsl.cybercity.dk) |
| 04:29.45 | *** join/#asterisk chaoscon (n=ph33r@smartserv/ceo/chaoscon) |
| 04:29.46 | asterboy | ok, first...are you behind a firewall/NAT? |
| 04:29.50 | orlok | yup |
| 04:30.01 | orlok | asterisk box is also the nat gateway for the phones |
| 04:30.02 | asterboy | have you tried it outside the NAT? |
| 04:30.11 | asterboy | oh |
| 04:30.26 | asterboy | can you turn off the firewall portion? |
| 04:30.31 | orlok | i can get them working correctly if i configure the phones to register with the sip provider |
| 04:30.41 | orlok | going through this same gateway |
| 04:30.49 | orlok | inbound and outbound calls |
| 04:31.03 | orlok | outbound calls via asterisk work correctly, however inbound does not |
| 04:31.08 | asterboy | ok so something is going on with * acting as the NAT |
| 04:31.25 | orlok | i have no public IP's free either |
| 04:31.28 | asterboy | tcpdump? |
| 04:31.53 | orlok | asterboy: when i dial in without sip debugging, i do not see it reach a context |
| 04:31.53 | dlynes_office | wunderkin: You could always make Asterisk listen on a different port...that's easily done |
| 04:32.06 | dlynes_office | erm |
| 04:32.22 | dlynes_office | wwalker: You could always make Asterisk listen on a different port...that's easily done |
| 04:32.22 | asterboy | ya, it sure sounds like something is blocking |
| 04:32.53 | dlynes_office | orlok: check your firewall log |
| 04:33.03 | dlynes_office | orlok: see if one of your deny or drop rules is getting matched |
| 04:33.36 | orlok | so that sip debug looks correct? |
| 04:33.47 | dlynes_office | nothing to do with sip debug |
| 04:34.12 | dlynes_office | Just check to see if deny or drop rules are getting matched with udp 5060 |
| 04:34.14 | dlynes_office | if they are |
| 04:34.19 | dlynes_office | then that's why nothing's working |
| 04:34.26 | dlynes_office | orlok: or another way to know for sure |
| 04:34.32 | dlynes_office | orlok: turn off your firewall |
| 04:34.44 | dlynes_office | orlok: then get all sip traffic to pass through asterisk |
| 04:35.29 | orlok | well, sip traffic is definetly getting to asterisk |
| 04:35.37 | *** join/#asterisk sternn (n=sternn@user-0c938ku.cable.mindspring.com) |
| 04:36.43 | orlok | i can see the port 5060 traffic both ways on the external interface |
| 04:36.56 | dlynes_office | orlok: what about rtp traffic? |
| 04:37.45 | dlynes_office | orlok: check for 10000-30000 range, udp |
| 04:37.54 | orlok | nope, i can only see 5060 |
| 04:37.57 | asterboy | nmap |
| 04:38.33 | orlok | ahh |
| 04:40.15 | asterboy | that article on skype got >5000 diggs...that is the most I've seen |
| 04:40.47 | Sedorox | hmmm |
| 04:41.44 | dlynes_office | diggs.be.com? |
| 04:42.43 | asterboy | no, there is just a puppy there |
| 04:43.21 | dlynes_office | that's sad |
| 04:43.32 | dlynes_office | some domain squatter snatched up be.com |
| 04:45.07 | asterboy | seen benkjk |
| 04:45.46 | asterboy | benjk, hates skype with a passion...this news will only furl his feathers. |
| 04:47.16 | orlok | when should i see rtp packets? |
| 04:48.20 | asterboy | Isn't that at the beginning of a call? |
| 04:48.33 | orlok | before or after i see port 5060 traffic/ |
| 04:49.04 | asterboy | and isn't there instances where rtp is used throughout the entire session as the signalling protocal? |
| 04:49.23 | Freman | any REAL reason to upgrade from 1.0.9 to 1.2.7? (c: |
| 04:49.50 | asterboy | the satisfaction that your current? |
| 04:49.57 | *** join/#asterisk tuxd00d (n=tuxinato@netblock-68-183-112-100.dslextreme.com) |
| 04:50.02 | Freman | I can get current from a power point... |
| 04:50.09 | Freman | with much less effort I might add (c: |
| 04:50.45 | asterboy | ok, forget the amps...go for the higher volts |
| 04:51.14 | asterboy | if it isn't broke...don't fix it. |
| 04:51.34 | Freman | I'm installing 1.2.7 on the work asterisk... so we'll see (c: |
| 04:52.17 | dlynes_office | Freman: i can think of plenty of reasons to go to 1.2.7.1; i can think of only one to go back to 1.0.9 |
| 04:53.34 | Freman | tell me more |
| 04:53.46 | asterboy | wait, arn't you suppose to screw with your home setup first? |
| 04:53.53 | orlok | asterboy: heh |
| 04:53.59 | Freman | blow that, my home setup is working |
| 04:54.08 | Freman | work setup is a fresh install |
| 04:56.03 | CunningPike | Freman: The 'n' priority, native MOH, voicemail gain setting, hints - need I go on? |
| 04:56.09 | dlynes_office | 1.2.7.1 has much better hinting support, better sip subscription support, native music on hold, asterisk extension language, ... |
| 04:56.28 | dlynes_office | oh yeah...and labelled priorities |
| 04:56.36 | Freman | oooh... native moh would be nice... |
| 04:56.50 | Freman | ael? just another language I have to learn |
| 04:57.23 | CunningPike | Freman: No more memory leaks |
| 04:57.29 | CunningPike | That we know of anyway :) |
| 04:57.52 | *** join/#asterisk carrar (i=tim@osburn.com) |
| 04:58.08 | Freman | I havn't seen any in my home box )c: |
| 04:58.24 | Freman | spoze the real question is... how much is my extensions file gonna break if I upgrade... |
| 04:58.25 | dlynes_office | Freman: And I'm rewriting app_voicemail.c, and adding native smtp support to it |
| 04:58.31 | Freman | res_perl still works? |
| 04:58.45 | Freman | that'll be nice dlynes_home (c: |
| 04:58.53 | dlynes_office | Freman: Almost everything you currently have configured will not need to be reconfigured on the upgrade |
| 04:59.10 | dlynes_office | Freman: however, some of your configuration file directives might be deprecated |
| 04:59.25 | CunningPike | Freman: But they will still work |
| 05:00.07 | Freman | okey... might consider it after this weeken |
| 05:00.19 | dlynes_office | Freman: you might want to hold off about a week or so |
| 05:00.28 | dlynes_office | Freman: 1.2.8 has already been marked for release |
| 05:00.30 | CunningPike | Freman: 1.2.8 is imminent....... |
| 05:00.44 | Freman | gentoo is about 6 weeks behind you (c; |
| 05:00.54 | dlynes_office | But I would still wait for 1.2.8.1 :) |
| 05:00.57 | Freman | what's in 1.2.8 (c: |
| 05:01.45 | dlynes_office | Freman: i believe a configure script, and some other things |
| 05:02.11 | dlynes_office | The configure script even has a cool looking asterisk ascii art logo |
| 05:02.49 | Freman | hehe |
| 05:03.07 | Freman | I like the zaptel complaint about crc_citt - "zaptel is not happy" |
| 05:03.27 | dlynes_office | Freman: oh yeah...a very strong argument to upgrade |
| 05:03.37 | dlynes_office | Freman: You can use linux 2.6 now |
| 05:03.52 | Freman | umm... I have been using 2.6 with asterisk for like... 8 months |
| 05:04.09 | dlynes_office | oh...didn't think 1.0.9 was compatible with 2.6 |
| 05:04.45 | Freman | asterisk ~ # uname -r |
| 05:04.45 | Freman | 2.6.13-gentoo-r3 |
| 05:04.50 | Freman | nope, defidently is |
| 05:06.16 | *** join/#asterisk terrapen_ (n=cjs@166.70.206.242) |
| 05:06.55 | *** join/#asterisk tuxd00d (n=tuxinato@netblock-68-183-112-100.dslextreme.com) |
| 05:07.15 | *** join/#asterisk Altair256 (n=Altair25@tn-greenback1a-30.rhmdky.adelphia.net) |
| 05:08.30 | dlynes_office | ~seen ciber311 |
| 05:08.42 | jbot | ciber311 <n=Ciber@user-1087e94.cable.mindspring.com> was last seen on IRC in channel #asterisk, 9d 3h 7m 49s ago, saying: 'well afk a bit'. |
| 05:08.56 | dlynes_office | afk for a bit is 9 days? |
| 05:09.28 | dlynes_office | could've sworn i seen him in here a day or two ago |
| 05:14.13 | asterboy | ~seen benjik |
| 05:14.15 | jbot | asterboy: i haven't seen 'benjik' |
| 05:14.22 | asterboy | ~seen benjk |
| 05:14.24 | jbot | benjk <n=benjamin@f8a01-0357.din.or.jp> was last seen on IRC in channel #asterisk, 83d 5h 12m 33s ago, saying: 'for example'. |
| 05:14.51 | terrapen_ | i need a new * project |
| 05:15.00 | terrapen_ | thinking about maybe setting up asterfax |
| 05:15.04 | terrapen_ | i need to "wow |
| 05:15.10 | terrapen_ | err "wow" the bosses |
| 05:15.22 | dlynes_office | How about rewriting app_dial.c? |
| 05:15.29 | terrapen_ | heh |
| 05:15.37 | terrapen_ | it works just fine for me :) |
| 05:15.49 | dlynes_office | so you're not that masochistic, eh? |
| 05:15.59 | terrapen_ | we're corporate :) |
| 05:16.31 | dlynes_office | so rewrite it in cobol, then |
| 05:17.10 | terrapen_ | hahaha |
| 05:17.16 | asterboy | lol, cobol. |
| 05:17.19 | terrapen_ | not THAT corporate :P |
| 05:17.24 | dlynes_office | lol |
| 05:17.36 | terrapen_ | i must say, xmission is the best ISP i've ever had |
| 05:17.41 | asterboy | cobol was written by a commitee |
| 05:17.59 | terrapen_ | never had an ISP give access to their internal nagios server |
| 05:18.01 | asterboy | tells you how great a language it must be to program in |
| 05:18.09 | terrapen_ | or full traffic stats for my connection |
| 05:18.09 | blitzrage | hey! |
| 05:18.18 | dlynes_office | dood |
| 05:18.26 | dlynes_office | blitzrage wants to write in cobol...I know that for sure! |
| 05:18.41 | blitzrage | fuck yah -- cobol is awesome |
| 05:18.57 | dlynes_office | Cobol is the OTHER C |
| 05:19.00 | Freman | cool... I've got my asterisk installed I just can't remmember where to go from here... been a long time |
| 05:19.00 | blitzrage | ok, maybe not so much |
| 05:19.08 | blitzrage | Freman: what do you want to do? |
| 05:19.50 | blitzrage | I just got back from a 20km bike ride at midnight in the rain... |
| 05:19.57 | blitzrage | I'm feeling kinda generous -- ask me a question! |
| 05:20.01 | dlynes_office | Yeah...I noticed |
| 05:20.10 | dlynes_office | But everyone ignored you in #asterisk-dev :) |
| 05:20.14 | blitzrage | I kmnow :) |
| 05:20.18 | blitzrage | I think everyhone went to bed |
| 05:20.36 | terrapen_ | blitz, i just got back from a 20km bike ride...but it was hot and dusty |
| 05:20.38 | Freman | Play a message to zap/1 and zap/2 when they call, record their caller id, give them the option to change the number left, then have the number forwarded to a web interface... but first I'd better configure the zaptel card (c: |
| 05:20.39 | terrapen_ | and rocky |
| 05:20.52 | terrapen_ | hahah |
| 05:20.57 | dlynes_office | pothead :) |
| 05:21.07 | blitzrage | dlynes_office: I meant the donut |
| 05:21.09 | terrapen_ | i could use some dutchy |
| 05:21.15 | *** join/#asterisk marcoprechel (n=marcopre@69.34.76.200) |
| 05:21.16 | dlynes_office | sure ya did :) |
| 05:21.21 | dlynes_office | the song's about hashish :) |
| 05:21.22 | asterboy | that's what they call it"? |
| 05:21.28 | blitzrage | Freman: yah :) |
| 05:21.37 | blitzrage | Freman: thats all dialplan logic though -- lucky for you! |
| 05:21.49 | blitzrage | Just got back from volleyball -- won 6 games in a row |
| 05:22.00 | terrapen_ | Musical Youth |
| 05:22.07 | dlynes_office | yeah |
| 05:22.09 | terrapen_ | back in the day |
| 05:22.11 | terrapen_ | i remember the video |
| 05:22.12 | dlynes_office | old song |
| 05:22.18 | dlynes_office | late 70's/early 80's |
| 05:22.27 | dlynes_office | I've got it on mp3 in my car :0 |
| 05:22.28 | blitzrage | Musical Youth is a band? |
| 05:22.30 | terrapen_ | yep, early days of MTV |
| 05:22.31 | terrapen_ | yes |
| 05:22.33 | dlynes_office | blitzrage: yeah |
| 05:22.39 | blitzrage | really? wow... I thought I knew a lot about music.... |
| 05:22.42 | blitzrage | never even heard of them |
| 05:22.46 | terrapen_ | man, i think im gonna have to sign up for flickr's pay service |
| 05:22.49 | dlynes_office | blitzrage: they were a one hit wonder |
| 05:22.53 | terrapen_ | i have the MP3 right here |
| 05:22.54 | blitzrage | ahhhhhhhh |
| 05:22.57 | blitzrage | that's why then |
| 05:22.58 | dlynes_office | their only hit was the dutchy song |
| 05:23.05 | blitzrage | anyone got a copy of it? :) |
| 05:23.11 | dlynes_office | likke i said |
| 05:23.15 | dlynes_office | it's on mp3 in my car :0 |
| 05:23.29 | dlynes_office | along with a shitload of reggae and calypso :0 |
| 05:23.33 | blitzrage | niiiiiiice |
| 05:23.33 | asterboy | break on through |
| 05:23.44 | terrapen_ | i will upload for you |
| 05:23.45 | terrapen_ | hang on |
| 05:24.10 | blitzrage | Not To Touch The Earth (The Celebration of the Lizard) |
| 05:24.38 | dlynes_office | anyways, lataz |
| 05:24.43 | blitzrage | dlynes_office: peas |
| 05:25.05 | blitzrage | Freman: you check out the book for how to configure zaptel? |
| 05:25.30 | terrapen_ | uploading now |
| 05:25.40 | *** join/#asterisk tuxd00d (n=tuxinato@netblock-68-183-112-100.dslextreme.com) |
| 05:25.41 | blitzrage | terrapen_: where and what are you uploading? :) |
| 05:26.31 | terrapen_ | ok, it's in http://saba.island.nu/tunes/ |
| 05:26.33 | terrapen_ | almost uplaoaded |
| 05:26.36 | terrapen_ | 30 more sec |
| 05:27.05 | blitzrage | nice |
| 05:27.12 | terrapen_ | ok, its done |
| 05:27.29 | blitzrage | too bad you don't have a whole bunch of music in there |
| 05:27.32 | terrapen_ | i will not listen, for it will be in my head all night |
| 05:27.38 | blitzrage | lol |
| 05:27.41 | terrapen_ | that's just the stuff i share with friends |
| 05:28.46 | terrapen_ | uploading my mountain biking photos to flicr |
| 05:30.06 | terrapen_ | DOH FLICKR |
| 05:30.12 | terrapen_ | you can pay them with Paypal |
| 05:30.19 | terrapen_ | and it shows how many transactions they've completed |
| 05:30.20 | enkrypted | what do you get for paying them? |
| 05:30.23 | terrapen_ | 32308 |
| 05:30.29 | terrapen_ | now I know how much money they took in :) |
| 05:30.44 | Freman | blitzrage: book? (c: I'll just work off my existing configs from my home system |
| 05:31.03 | blitzrage | ~thebook |
| 05:31.07 | jbot | hmm... thebook is somebody said thebook was Asterisk: The Future of Telephony, released under a Creative Commons license and available at http://www.asteriskdocs.org << Meet Jim Van Meggelen at Cluecon http://www.cluecon.com |
| 05:31.23 | blitzrage | apparently Jim is going to Cluecon |
| 05:31.26 | terrapen_ | enkrypted, unlimited upload capacity, unlimited bandwidth, and unlimited photosets |
| 05:31.26 | blitzrage | but that makes sense |
| 05:31.50 | blitzrage | terrapen_: 20km on a mountain bike? you're insane |
| 05:31.59 | terrapen_ | well, err 2GB upload |
| 05:32.05 | terrapen_ | blitz, i'm trying to get in shape |
| 05:32.33 | blitzrage | terrapen_: me too -- but jeezus :) that'll do it alright -- I have a mountain bike as well -- is yours hardtail or shock system? |
| 05:33.00 | blitzrage | I've alredy done like 147km in the first week of my new bike |
| 05:34.04 | terrapen_ | blitz: check it out: |
| 05:34.25 | terrapen_ | http://justlivehere.blogspot.com/ |
| 05:34.28 | terrapen_ | pics of the bike on there |
| 05:34.37 | terrapen_ | about 2/3 way down |
| 05:34.48 | terrapen_ | it's full suspension, full carbon fiber, disc brakes |
| 05:35.07 | blitzrage | terrapen_: http://leifmadsen.com/gallery/culhamtrail |
| 05:35.18 | blitzrage | disk brakes -- nice :) |
| 05:35.44 | terrapen_ | niice... lots of wateR! |
| 05:35.54 | terrapen_ | almost done uploading my ride photos |
| 05:36.03 | blitzrage | terrapen_: yah -- that trail is awesome -- I've been doing a lot of road biking lately though |
| 05:36.15 | terrapen_ | where its that, blitzrage |
| 05:36.25 | terrapen_ | my road bike sits unused in my storage unit |
| 05:36.39 | blitzrage | terrapen_: what kinda bike? |
| 05:36.47 | blitzrage | that trail is in Mississauga, ON |
| 05:36.50 | blitzrage | (outside Toronto) |
| 05:37.02 | terrapen_ | blitz, Trek Project One - Fuel EX 9.5, custom paint |
| 05:37.13 | blitzrage | nice -- I just got a Trek 7.5 FX |
| 05:37.21 | blitzrage | any pictures of it? |
| 05:37.23 | *** join/#asterisk hayburn (i=chiaborg@concorde.hayburn.net) |
| 05:37.33 | terrapen_ | http://www.flickr.com/photos/defender90/sets/72057594136286804/ |
| 05:37.38 | terrapen_ | sweet, blitz |
| 05:37.40 | blitzrage | oooo... OCLV eh? |
| 05:37.50 | terrapen_ | http://photos1.blogger.com/blogger/3576/2135/1600/DSC01902op.jpg |
| 05:37.52 | terrapen_ | thats my bike |
| 05:37.53 | terrapen_ | yeah |
| 05:38.04 | blitzrage | thats a nice bike -- still looks too heavy for me :) |
| 05:38.14 | terrapen_ | that photo was taken by my bike shop back in texas when they were building it for me |
| 05:38.16 | blitzrage | looks damn tough |
| 05:38.26 | terrapen_ | 27.3 pounds, IIRC |
| 05:38.32 | blitzrage | I like the flames |
| 05:38.37 | terrapen_ | thanks :) |
| 05:38.40 | terrapen_ | check out the ride photos |
| 05:39.11 | blitzrage | damnit -- where's the next button? |
| 05:39.23 | blitzrage | oh there's the more => button |
| 05:39.44 | blitzrage | terrapen_: where is this? |
| 05:40.02 | terrapen_ | Round Valley Trail, Park City, Utah |
| 05:40.18 | blitzrage | nice! |
| 05:40.22 | blitzrage | you're near Jared Smith |
| 05:40.26 | terrapen_ | yeah, I like it :) |
| 05:40.32 | terrapen_ | jared smith? |
| 05:40.40 | terrapen_ | there are lots of jared smiths in utah :) |
| 05:40.41 | hayburn | subway man? |
| 05:40.47 | terrapen_ | probably the two most common names here |
| 05:40.47 | blitzrage | third author of the O'Reilly book :) |
| 05:40.55 | terrapen_ | ahhh |
| 05:40.56 | blitzrage | heh -- yah, I bet it is |
| 05:40.59 | terrapen_ | asterisk dev? |
| 05:41.21 | blitzrage | ummmm... he does some development work |
| 05:42.34 | enkrypted | so you would be the first author? |
| 05:45.56 | terrapen_ | i really like flickr |
| 05:46.00 | blitzrage | I'm just one of the authors... but I had already talked about Jim earlier... I suppose you called me out on it -- although I wasn't meaning it conciously :) |
| 05:46.00 | terrapen_ | its a good deal |
| 05:46.09 | blitzrage | yah -- it looks pretty good actually |
| 05:46.18 | terrapen_ | sure, i could write something similar...but for $25/year? |
| 05:46.25 | blitzrage | I like having the full 5.1MP up there though :) |
| 05:46.27 | justinu | blitzrage: what are you up to these days? |
| 05:46.35 | blitzrage | running an ITSP |
| 05:46.38 | blitzrage | mostly -- and training |
| 05:46.38 | terrapen_ | i think i have the full size photos there |
| 05:46.45 | justinu | are you allowed to say which one? |
| 05:46.53 | blitzrage | sure, I'm allowed :) |
| 05:47.04 | justinu | no more books? |
| 05:47.20 | blitzrage | well, I'm supposed to be working on another one... but writing books pays like selling CDs |
| 05:47.24 | justinu | heh |
| 05:47.30 | justinu | that's a shame |
| 05:47.33 | blitzrage | yah... really is |
| 05:47.41 | justinu | the world needs better documentation! |
| 05:47.44 | blitzrage | I really like doing it too |
| 05:47.47 | terrapen_ | how does selling CDs pay? |
| 05:47.54 | blitzrage | pays like crap :) |
| 05:48.12 | blitzrage | I would have made more money working at McDonalds |
| 05:48.33 | justinu | well, a lot of n00bs appreciate it... whether they recognize it or not. |
| 05:49.17 | blitzrage | well I hope so... the really reason was really to help out the community... do my part you know? |
| 05:49.32 | terrapen_ | i guess youmake your money back from your reputation though, no? |
| 05:49.37 | blitzrage | I can't really programming significantly in Asterisk by any stretch of the imagination... so I wanted to give back |
| 05:49.42 | justinu | and a big thanks and mad propz from me and my crew :) |
| 05:49.42 | terrapen_ | money is overrated |
| 05:49.43 | *** join/#asterisk shido6 (n=shido6@d221-68-200.commercial.cgocable.net) |
| 05:49.46 | blitzrage | yah... I got a lot of rep, lol |
| 05:49.55 | blitzrage | money *IS* overrrated |
| 05:50.06 | terrapen_ | i could make a hell of a lot more than i make here but would I be able to ride my mountain bike on kick-ass trails every day after work? |
| 05:50.11 | blitzrage | although it's damn nice when you have school loans to pay off |
| 05:50.15 | terrapen_ | or take mornings off to ski on powder days? |
| 05:50.25 | terrapen_ | blitz, i hear ya |
| 05:50.49 | justinu | theres supposedly some good trails out here too |
| 05:50.54 | terrapen_ | i'm hugely into the outdoors and since i work for an outdoor gear supply company, i get all my shit at cost which really helps |
| 05:51.01 | terrapen_ | justin, where is here |
| 05:51.02 | blitzrage | oh well... coming up on the first year of the business... 1.5 years since out of school |
| 05:51.07 | terrapen_ | oh, LA |
| 05:51.07 | justinu | i dunno tho, i'm not coordinated enough to ride that kind of bike |
| 05:51.07 | enkrypted | you could speak at conferences |
| 05:51.10 | blitzrage | that's awesome |
| 05:51.17 | justinu | terrapen: santa monica mountains |
| 05:51.22 | terrapen_ | those are nice |
| 05:51.23 | blitzrage | speaking at conferences pays nothing usually |
| 05:51.35 | blitzrage | I'd bed 99% of the speakers are volunteers |
| 05:51.39 | justinu | cottonwood canyon is.... awesome |
| 05:51.39 | terrapen_ | i'm HOPING to land a speaking thing at VON |
| 05:51.44 | terrapen_ | hoping it works out |
| 05:51.45 | blitzrage | I've spoken at conferences :) |
| 05:51.52 | terrapen_ | justin, my cottonwood canyon or some other? |
| 05:51.55 | justinu | yours |
| 05:52.00 | terrapen_ | ah, yeah :) |
| 05:52.02 | justinu | i skiied there once, and it was kick ass |
| 05:52.12 | terrapen_ | justin, been here? |
| 05:52.12 | terrapen_ | http://flickr.com/photos/defender90/sets/72057594136286804/ |
| 05:52.23 | blitzrage | I want to try skiing so badly... but I'm afraid I'd destroy my knees |
| 05:52.36 | justinu | don't think so |
| 05:52.43 | justinu | where at? |
| 05:52.43 | terrapen_ | justin: http://flickr.com/photos/defender90/sets/72057594096475919/ |
| 05:52.47 | blitzrage | I'd say honestly skiing is the only sport that scares me |
| 05:52.53 | terrapen_ | justin, that's just outside of park city |
| 05:53.11 | justinu | odd ball :P |
| 05:53.20 | terrapen_ | yeah, that's me :) |
| 05:53.27 | justinu | most of the telemark skiiers i've seen are so good |
| 05:53.33 | justinu | never seen a n00b before |
| 05:53.40 | terrapen_ | i am a tele n00b |
| 05:53.42 | justinu | they just blow by you |
| 05:53.51 | terrapen_ | but yeah, it seems wierd how good they all are |
| 05:54.00 | terrapen_ | i feel like i really suck...and i do |
| 05:54.20 | justinu | heh i suck at downhill... it takes a lot of core strength that i just don't have |
| 05:54.44 | justinu | at least the technical stuff |
| 05:54.49 | terrapen_ | the hardest thing about tele is the balance |
| 05:54.58 | *** join/#asterisk mackss (n=mack@c-68-41-75-221.hsd1.mi.comcast.net) |
| 05:55.08 | justinu | the fun thing for me is just being in the mountains |
| 05:55.10 | terrapen_ | you have to weight your dropped side, which is really counter-intuitive |
| 05:55.19 | terrapen_ | justin, what do you do out therE? |
| 05:55.28 | justinu | once I skiied in vermont, and it was right at closing time on an otherwise empty day |
| 05:55.36 | blitzrage | terrapen_: so you drop the weight to the outside? |
| 05:55.36 | justinu | we got the last run, after the ski patrol |
| 05:55.52 | justinu | they came by and said if we don't see you at the bottom in 10 minutes, we'll come back for you |
| 05:56.01 | terrapen_ | blitz, you put at least 50% of your weight on your dropped knee, which is always on the inside of the teurn |
| 05:56.18 | justinu | it was an amazing feeling being up on that peak by ourselves |
| 05:56.20 | terrapen_ | hahah justin, i always try to make those last chairs |
| 05:56.24 | *** join/#asterisk tuxd00d (n=tuxinato@netblock-68-183-112-100.dslextreme.com) |
| 05:56.39 | blitzrage | terrapen_: oh... ok, that kinda makes sense... I guess that really is a lot of power of the leg though |
| 05:56.43 | Freman | I think I gonna have to read about this subscripe stuff../ |
| 05:56.52 | blitzrage | at least how I picture it in my mind it feels like a lot of weight on my leg :) |
| 05:57.07 | justinu | bed time for me, take it easy :) |
| 05:57.10 | blitzrage | peas man |
| 05:57.17 | blitzrage | bowl! |
| 05:57.19 | terrapen_ | cool |
| 05:57.22 | terrapen_ | take care justin |
| 06:02.58 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 06:06.39 | *** join/#asterisk tuxd00d (n=tuxinato@netblock-68-183-112-100.dslextreme.com) |
| 06:08.40 | *** join/#asterisk De_Mon (n=de_mon@69.69.138.85) |
| 06:08.49 | *** join/#asterisk Johnnie (n=jdlewis@pdpc/supporter/active/Johnnie) |
| 06:11.42 | mackss | i am lost can someone help me |
| 06:12.03 | CunningPike | Where are you? :) |
| 06:12.06 | mackss | i can't figure out if i need to run my own asterisk server to use nufone.net |
| 06:12.14 | mackss | CunningPike, i wish i knew ;-) |
| 06:12.32 | CunningPike | Hmm - can't help you there...... |
| 06:12.55 | mackss | that and i am not sure if i should even run asterisk |
| 06:13.26 | mackss | i would like to make SIP calls and also to make REAL calls using it from computers on my network |
| 06:13.31 | *** join/#asterisk saftsack (n=saftsack@p54A7CCE4.dip.t-dialin.net) |
| 06:13.41 | mackss | real = landline calls |
| 06:13.51 | CunningPike | mackss: Doesn't look like you need Asterisk... |
| 06:14.08 | CunningPike | They have client software |
| 06:14.31 | mackss | then who would need asterisk? |
| 06:14.45 | CunningPike | Anyone who needs a software PBX |
| 06:15.03 | mackss | but wouldn't i need one for what i would like to do? |
| 06:15.37 | CunningPike | If all you want is to place calls to regular phone numbers from a networked PC, no |
| 06:16.05 | mackss | well i would like my caller ID to be the same from whereever i am |
| 06:16.17 | mackss | if i were in the UK or w/e..it would route a local call |
| 06:16.26 | CunningPike | That's up to your provider |
| 06:16.34 | *** join/#asterisk Assid (n=assid@203.115.83.214) |
| 06:16.35 | mackss | so asterisk + provider? |
| 06:16.51 | CunningPike | Just provider and their client |
| 06:16.55 | mackss | hrm |
| 06:17.03 | mackss | do the providers run asterisk? |
| 06:17.05 | mackss | lol |
| 06:17.10 | CunningPike | Some do |
| 06:17.13 | *** join/#asterisk CrummyGummy (n=wayne@dsl-145-117-03.telkomadsl.co.za) |
| 06:17.25 | mackss | could you refer a good provider for me that could fufill those needs? |
| 06:17.37 | Assid | heya |
| 06:17.49 | mackss | i looked into a few but never really understood...i guess that's why i am here |
| 06:17.53 | CunningPike | There are any amount of VOIP providers - your ISP or TV cable company may even do it |
| 06:18.02 | CunningPike | Where are you located? |
| 06:18.23 | mackss | yes comcast does however they don't do sip |
| 06:18.43 | CunningPike | Why is it important for you to do SIP? |
| 06:19.09 | terrapen_ | its great to have designed your company's phone system |
| 06:19.12 | *** join/#asterisk tardisx (n=justin@ppp167-251-29.static.internode.on.net) |
| 06:19.18 | terrapen_ | i have a extension on the work PBX at home |
| 06:19.33 | terrapen_ | very nice for making calls on, saving cellphone minutes :) |
| 06:19.33 | CunningPike | terrapen: Me too :) |
| 06:19.37 | mackss | well with SIP i can call from my computer anywhere in the world right? |
| 06:19.47 | mackss | cause i can login from anywhere |
| 06:19.55 | CunningPike | mackss: SIP is just one of the protocols that can do that, yes |
| 06:19.56 | Assid | mackss: no * box? |
| 06:20.00 | terrapen_ | cunningpike, i brought the silly SPA-942 phone home, since we went Polycom at work |
| 06:20.12 | CunningPike | terrapen: I have an IAXy :D |
| 06:20.19 | CunningPike | We're Polycom at work too |
| 06:20.30 | mackss | Assid, i have asterisk installed |
| 06:20.41 | *** join/#asterisk neilbags (n=neil@149.171.94.137) |
| 06:20.49 | Assid | mackss: so * can support iax/sip |
| 06:21.04 | Assid | hell.. i acutally have all my providers on iax |
| 06:21.06 | Assid | and clients on sip |
| 06:21.12 | terrapen_ | i have an iaxy but i think its a piece of crap |
| 06:21.18 | terrapen_ | no offense, digium folks |
| 06:21.25 | terrapen_ | it just does not work well for me |
| 06:21.34 | CunningPike | mackss: But if all you want to do is place calls from your PC, the nuphone client software is all you need |
| 06:21.40 | Assid | terrapen: shouldnt be any difference from any other pap2 |
| 06:21.50 | hayburn | does an iaxy only connect one rj11 port? |
| 06:21.50 | Assid | nuphone client ? |
| 06:21.54 | CunningPike | terrapen: Works OK for me - but you can sure tell that I'm not on my polycom |
| 06:21.58 | Assid | hell.. you can use any sip client |
| 06:22.02 | terrapen_ | assid, there is a delay getting a dialtone when you pick up the receiver |
| 06:22.08 | terrapen_ | and often, you don't get a dialtone |
| 06:22.12 | Assid | xlite/eyebeam.. even the old messenger |
| 06:22.23 | terrapen_ | cunningpike, the SPA-942 phone is pretty decent for home |
| 06:22.30 | terrapen_ | not as nice as the IP501 but... |
| 06:22.35 | CunningPike | terrapen: Ya - we're 100% Polycom |
| 06:22.44 | Assid | i wanted a 501 ! |
| 06:22.48 | Assid | got a 301 instead :( |
| 06:22.58 | CunningPike | I have a 600 |
| 06:23.00 | CunningPike | :) |
| 06:23.01 | terrapen_ | i just got an IP4000 in today |
| 06:23.08 | CunningPike | terrapen: Nice |
| 06:23.16 | terrapen_ | cp: i have an IP601 on my desk at work |
| 06:23.20 | Assid | IP4000 ? arent those conference phones? |
| 06:23.22 | terrapen_ | but everyone else got 501s |
| 06:23.25 | terrapen_ | assid, yep |
| 06:23.28 | Assid | hrmm |
| 06:23.30 | Assid | nice |
| 06:23.40 | Assid | > $700 right ? |
| 06:23.51 | CunningPike | terrapen: Yes - we had a 600 for test early on, so it ended up on my desk |
| 06:23.58 | mackss | I would like to be able to make calls from where ever i am in the world as long as i have an internet connection...I would also like to be able to change my caller id to a call back number at the destination that i would be at. Could someone advise me as to the best solution for this? |
| 06:24.02 | Assid | mackss: please dont pm me.. got wayyy too many windows open to find yours |
| 06:24.21 | CunningPike | All the others are 501s so far, but we'll probably deploy some 601s |
| 06:24.22 | *** join/#asterisk kmilitzer (n=km@office-gw.westend.com) |
| 06:24.22 | mackss | Assid, sorry..that's why i justed asked again here ;-) |
| 06:24.38 | hayburn | does an iaxy only connect one rj11 port? |
| 06:24.58 | Assid | mackss: use a any regular sip phone .. software/hardware.. doesnt matter.. software are easier to configure |
| 06:25.04 | Assid | and sign up with any provider |
| 06:25.19 | CunningPike | mackss: If you are using a VOIP provider, your access number and therefore caller ID will always be the same, regardless of where you are |
| 06:25.25 | CunningPike | That's the beauty of it |
| 06:25.26 | mackss | so like x-chat or sjphone with a provider? |
| 06:25.32 | CunningPike | mackss: Exactly |
| 06:25.34 | Assid | most allow you to sset your own callerid |
| 06:26.00 | hayburn | is this thin on? |
| 06:26.05 | mackss | Assid, could you tell me of a good provider that allows the caller id to be changed? |
| 06:26.06 | Assid | nope |
| 06:26.10 | terrapen_ | the quality of this SPA-942 at home, across the internet, is amazing |
| 06:26.17 | CunningPike | hayburn: You're 'g' key isn't working..... ;) |
| 06:26.19 | terrapen_ | sounds like im sitting in the office |
| 06:26.21 | Assid | mackss: nufone/voicepulse/voipjet |
| 06:26.29 | mackss | thank you Assid |
| 06:26.29 | terrapen_ | mackss, the number or name? |
| 06:26.33 | terrapen_ | try teliax |
| 06:26.33 | mackss | right |
| 06:26.35 | mackss | both |
| 06:26.36 | terrapen_ | i've tried those other |
| 06:26.37 | terrapen_ | s |
| 06:26.40 | Assid | teliax too |
| 06:26.47 | CunningPike | mackss: <cough> Vonage <cough> |
| 06:26.50 | Assid | tons of providers |
| 06:26.52 | CunningPike | :D |
| 06:26.52 | mackss | name and number would prolly be best |
| 06:26.52 | Assid | hahaha |
| 06:26.53 | terrapen_ | i did not have good luck with those others...but that's just me |
| 06:26.55 | Assid | vonage |
| 06:27.03 | hayburn | terrapen_: does an iaxy only connect to one rj11 port? |
| 06:27.10 | terrapen_ | you can't change the name on most providers, AFAIK |
| 06:27.14 | Assid | terrapen: u sure you configured it right ? |
| 06:27.17 | terrapen_ | hay: yes |
| 06:27.21 | mackss | vonage allows it to be changed easily and works with sip clients? |
| 06:27.25 | CunningPike | hayburn: Yes, it's a straight single port FXO |
| 06:27.31 | terrapen_ | assid, read the callerid section on the wiki, it explains more |
| 06:27.32 | Assid | mackss: stay away from them |
| 06:27.37 | mackss | lol |
| 06:27.38 | terrapen_ | i dont remember the reasons |
| 06:27.43 | CunningPike | Phone in one side, ethernet in the other |
| 06:27.43 | mackss | ya i always thought they were evil ;-) |
| 06:27.57 | Assid | terrapen_: what did you have a problem with ? |
| 06:28.06 | terrapen_ | assid, setting the name |
| 06:28.20 | hayburn | CunningPike/terrapen_ : thanks |
| 06:28.21 | Assid | calleridname isnt a universally accepted feature |
| 06:28.32 | CunningPike | hayburn: You're totally welcome |
| 06:28.33 | Assid | works fine when you do *-* connections |
| 06:28.46 | Assid | BUT, providers do not support.. since local pstnsdont support |
| 06:28.51 | Assid | therefore they mostly drop it |
| 06:29.15 | CunningPike | Assid: Tell me about it - we can't even get exchanges from the same bloody telco to pass it consistently |
| 06:29.19 | Assid | i am not sure tho.. but i think if you do calls within the same carrier.. they MIGHT just go through |
| 06:29.40 | Assid | i.e if you call from voicepulse to voicepulse |
| 06:29.52 | Assid | CunningPike: thats why we use the sipbroker project |
| 06:30.16 | Assid | free calling galore |
| 06:30.24 | CunningPike | Assid: Sorry - I was muddying the waters by referring to PSTN CID |
| 06:30.52 | Assid | CunningPike: yeah.. as i mentioned above.. very very poor / non existent support for that |
| 06:31.02 | Assid | pstn cid wasnt designed for name .. |
| 06:31.14 | *** join/#asterisk Shoragan (n=shoragan@d072.apm.etc.tu-bs.de) |
| 06:31.27 | CunningPike | What's frustrating is that our outbound CID is solid - it's the inbound name that's hardly ever there |
| 06:31.31 | CunningPike | Very annoying |
| 06:31.47 | hayburn | Assid: can you elaborate on pstn cid wasnt designed for name? |
| 06:32.35 | dlynes_office | CunningPike: Your outbound cid name is actually working? |
| 06:32.45 | CunningPike | dlynes_office: Yes |
| 06:32.59 | hayburn | is cid number different than cid name? |
| 06:33.00 | dlynes_office | CunningPike: did you have to get telus to do something on their end for it to work? |
| 06:33.01 | Assid | dlynes_office: hes sending it.. doesnt mean anyone is receiving it |
| 06:33.15 | Assid | hayburn: CID has 2 parts.. name and number |
| 06:33.21 | Assid | refer to the CID wiki |
| 06:33.27 | terrapen_ | ok bedtime for me |
| 06:33.31 | CunningPike | dlynes_office: It's just a regular PRI - and yes, it's received at the other end |
| 06:33.41 | dlynes_office | CunningPike: ah..weird |
| 06:33.46 | CunningPike | But not the other way around |
| 06:33.46 | *** join/#asterisk oej (n=oej@apollo.webway.se) |
| 06:33.49 | CunningPike | Go fiogure |
| 06:33.53 | dlynes_office | CunningPike: for us, it's almost like group telecom's stripping it off |
| 06:34.11 | Assid | CunningPike: you sure its received? |
| 06:34.13 | dlynes_office | CunningPike: our inbound caller id name and number are showing up, but the remote end only receives our caller id number |
| 06:34.20 | Assid | CunningPike: which provider you using |
| 06:34.32 | CunningPike | dlynes_office: That's what happens when we receive calls |
| 06:34.37 | CunningPike | Assid: Telus |
| 06:34.43 | CunningPike | Assid: And Allstream |
| 06:34.46 | dlynes_office | CunningPike: your received calls have no caller id name? |
| 06:34.55 | CunningPike | dlynes_office: Correct |
| 06:34.56 | dlynes_office | CunningPike: but your outbound calls do? |
| 06:35.05 | dlynes_office | CunningPike: damn...you're backwards to us |
| 06:35.07 | Assid | CunningPike: you wanna call me ? |
| 06:35.21 | Assid | i'd like to try them |
| 06:35.33 | CunningPike | Yes - but now I've given my head a shake, our outbound calls go out the Allstream PRI with name and number |
| 06:35.55 | dlynes_office | CunningPike: ah...so it works on allstream, but not on telus? |
| 06:36.10 | CunningPike | Inbound calls come in the Telus PRI usually (but not always) without name - Allstream is the same inbound |
| 06:36.14 | dlynes_office | CunningPike: fwiw, we're using group telecom |
| 06:36.37 | CunningPike | dlynes_office: It's baffling to me, and none of the telcos can explain it |
| 06:36.58 | dlynes_office | CunningPike: they can probably explain it...it's probably on purpose :) |
| 06:37.02 | Assid | im thinking of signing up with wengo for outgoing carrier |
| 06:37.09 | Assid | around 1.1 c/min or so |
| 06:37.14 | CunningPike | When I phone home from the office, we get name and number. When I phone the office from home, number only |
| 06:38.02 | hayburn | CunningPike: does that Telus PRI support Called Number? |
| 06:38.06 | x86 | my company is offering completely unlimited US domestic calling (including Hawaii, Alaska, Guam, and Canada) for $24.50/mo; no contracts; no softcap |
| 06:38.11 | x86 | http://www.shellshark.net/voip/ |
| 06:38.17 | x86 | check it out :) |
| 06:38.21 | SheriF_WorK | x86: why ur not in bed !? |
| 06:38.27 | Assid | is he for real? |
| 06:38.29 | CunningPike | hayburn: How do you mean? |
| 06:38.30 | tessier_ | x86: So I could just call Guam, leave the phone off the hook for months, and you would be fine with it? |
| 06:38.49 | SheriF_WorK | Assid: u mean x86 ? yes he is |
| 06:38.50 | x86 | tessier_: as long as you paid your bill every month, sure |
| 06:39.05 | tessier_ | And the bill would be $24.95? |
| 06:39.10 | dlynes_office | CunningPike: I think everyone thinks we're both Americans and that they can get pris from Telus or Allstream :) |
| 06:39.11 | x86 | $24.50 |
| 06:39.17 | hayburn | CunningPike : I use to have PRI for PPP dialin and could do special things with RADIUS based upon the number the user Called |
| 06:39.18 | tessier_ | oops..right. |
| 06:39.30 | x86 | tessier_: however, we dont offer E-911 |
| 06:39.34 | hayburn | CunningPike : perhaps its called ANI? |
| 06:39.37 | x86 | just to pre-warn you on that |
| 06:39.38 | Assid | x86: how many simultanous minutes? |
| 06:39.39 | tessier_ | Nobody does really. I think E911 is a crock anyhow |
| 06:39.40 | Assid | err |
| 06:39.41 | websae | we do e911 :) |
| 06:39.43 | Assid | sim. calls? |
| 06:39.50 | dlynes_office | We offer e911 |
| 06:39.51 | x86 | Assid: 1 |
| 06:39.58 | Assid | okay 1 sim. call |
| 06:40.03 | Assid | wont do |
| 06:40.06 | hayburn | or maybe DNIS |
| 06:40.15 | x86 | Assid: how many do you need? |
| 06:40.15 | CunningPike | hayburn: I'm thinking DIDs? We can tell which of a couple of hundred numbers was called? |
| 06:40.36 | CunningPike | dlynes_office: How are you doing e911? |
| 06:40.46 | dlynes_office | CunningPike: probably the same way you're doing it |
| 06:40.49 | CunningPike | OK |
| 06:40.52 | Assid | around 4-5 ? |
| 06:40.56 | dlynes_office | CunningPike: we're required by law to be able to offer it |
| 06:40.56 | *** join/#asterisk boddy (n=e@212.58.24.138) |
| 06:40.57 | CunningPike | Not really e911 then ;) |
| 06:41.01 | dlynes_office | CunningPike: i would assume you are too |
| 06:41.03 | CunningPike | Just ANI/ALI |
| 06:41.06 | x86 | Assid: well we offer business plans also :) |
| 06:41.14 | *** part/#asterisk tardisx (n=justin@ppp167-251-29.static.internode.on.net) |
| 06:41.35 | Assid | not spending $125 |
| 06:41.50 | boddy | hi all I bought te110p which channels I will use on rj45 connector |
| 06:41.51 | dlynes_office | CunningPike: We provide GT with the customer's full address, suite number, floor number, and cubicle number..t.hat kinda thing, and they create both an e911 and a white pages listing for us |
| 06:41.52 | x86 | good luck in your search :) |
| 06:41.54 | boddy | 12 45 ? |
| 06:42.17 | CunningPike | dlynes_office: Well, legally, we're not a provider, but we are a workplace and WCB would require us to provide 911 service to our employees. We've basically taken the view that we are a VOIP provider to our employees |
| 06:42.23 | dlynes_office | CunningPike: so any time a customer calls 911, all that information for the customer is forwarded along to the 911 call center |
| 06:42.39 | CunningPike | dlynes_office: OK, real e911 then - using the Telus service, right? |
| 06:42.43 | CunningPike | The database thing?> |
| 06:42.44 | dlynes_office | CunningPike: GT |
| 06:42.57 | CunningPike | Oh, OK - probably a similar product |
| 06:43.06 | dlynes_office | CunningPike: We don't do any of the dispatching...group telecom takes care of all that |
| 06:43.13 | CunningPike | We're just doing ANI/ALI with 1B lines at each location |
| 06:43.42 | CunningPike | Good enough for us |
| 06:43.45 | dlynes_office | CunningPike: the main requirement is that if a customer requests e911 service, they have to be in a geographical location that's covered by one of our co's |
| 06:43.53 | CunningPike | Right - makes sense |
| 06:44.10 | CunningPike | But you're basically reselling a GT service, right? |
| 06:44.16 | dlynes_office | CunningPike: no |
| 06:44.24 | boddy | ?? |
| 06:44.33 | dlynes_office | CunningPike: We're selling predominantly voip service, with pri failover |
| 06:44.42 | dlynes_office | CunningPike: so all local calls go through the pri |
| 06:44.53 | dlynes_office | CunningPike: and any long distance failovers go through the pri |
| 06:44.54 | CunningPike | I meant for e911 - you maintain a database at the CO? |
| 06:45.10 | dlynes_office | CunningPike: No...GT maintains a database at the CO on our behalf |
| 06:45.26 | CunningPike | Right |
| 06:45.31 | dlynes_office | CunningPike: so when we call out to 911, and specify the did, they match that against their database, and then send the 911 info to the call center |
| 06:45.43 | CunningPike | Yup - so it's the same service that Telus offers then |
| 06:45.58 | CunningPike | We elected not to pay for that :D |
| 06:46.15 | CunningPike | Seeing as we had a 1B at each location anyway |
| 06:47.09 | CunningPike | OK - bedtime |
| 06:47.13 | boddy | anybody help me ? |
| 06:47.14 | boddy | :) |
| 06:47.28 | CunningPike | boddy: What's the problem? |
| 06:47.37 | boddy | I bought te110p which channels I will use on rj45 connector |
| 06:47.42 | boddy | 12 45 ? |
| 06:48.14 | CunningPike | boddy: Not sure I understand, but it's a PRI card - you plug your PRI into it |
| 06:48.31 | Assid | anyone else here subscribing to sipbroker? |
| 06:49.09 | *** join/#asterisk tuxd00d (n=tuxinato@netblock-68-183-112-100.dslextreme.com) |
| 06:50.07 | boddy | yes but I dont know pri connection so which cable colour which rj45 channel |
| 06:50.45 | CunningPike | boddy: You should talk to your provider - some PRI terminations require a crossover, some don't |
| 06:51.20 | CunningPike | boddy: If no crossover is required, then a straight CAT5 cable will work |
| 06:51.47 | CunningPike | boddy: If you do require a crossover, google for 'PRI crossover' for the pinouts |
| 06:52.52 | *** join/#asterisk rKR245 (n=ravi@office.callsat-telecom.com) |
| 06:52.57 | boddy | I will use cat5 but I dont know which channels I will use on rj45 example 1. and 2. channel blue/bluewhite |
| 06:53.13 | CunningPike | dlynes_office: BTW - I'm gauging interest in a Vancouver or BC AUG |
| 06:53.26 | *** join/#asterisk lorinc (n=ang@caracas-0450.adsl.interware.hu) |
| 06:53.42 | CunningPike | boddy: Just use regular ethernet cabling - the colors don't matter |
| 06:53.51 | dlynes_office | CunningPike: cool |
| 06:54.04 | CunningPike | I assume you would be interested? |
| 06:54.11 | dlynes_office | CunningPike: I know there's at least six of us in this chat channel |
| 06:54.17 | dlynes_office | CunningPike: yeah, definitely |
| 06:54.19 | CunningPike | I see that - amazing, really |
| 06:54.27 | dlynes_office | CunningPike: Gurpreet and Alex I think both would be, too |
| 06:54.31 | CunningPike | When I was at Astricon I kept meeting Canucks, too |
| 06:54.38 | dlynes_office | CunningPike: not to mention Alex's friend Slobodan |
| 06:54.46 | CunningPike | dlynes_office: Great |
| 06:54.55 | CunningPike | I posted to the list |
| 06:55.02 | dlynes_office | CunningPike: asterisk-users? |
| 06:55.04 | CunningPike | Half a dozen replies already |
| 06:55.05 | CunningPike | Yes |
| 06:55.06 | boddy | ok colors dosent matter but which channels example ethernet uses 123 6 channels on rj45 connector |
| 06:55.38 | CunningPike | boddy: I'm sorry - I have no idea what you need - I just plugged it in and it worked |
| 06:55.51 | hayburn | 1236 is t1? |
| 06:55.54 | websae | anyone here going to Cluecon? |
| 06:55.57 | *** join/#asterisk MGSsancho (n=user@adsl-67-126-128-145.dsl.irvnca.pacbell.net) |
| 06:56.15 | dlynes_office | websae: bkw and anthm both are |
| 06:56.24 | *** join/#asterisk darkskiez (n=darkskie@bb-87-81-62-203.ukonline.co.uk) |
| 06:56.35 | CunningPike | boddy: Sorry - I see what you are talking about now - which pins? |
| 06:56.45 | boddy | yes |
| 06:56.46 | websae | yeah i was reading up on it, sounds like a worthwhile endeavor..... |
| 06:56.55 | websae | dlynes: are you gonna go? |
| 06:57.16 | CunningPike | boddy: Sorry about that - 1,2,4,5 |
| 06:57.18 | dlynes_office | CunningPike: oh yeah...what's his facer at netvoice.ca might be interested, too |
| 06:57.32 | CunningPike | Already heard from him - he was the first - George |
| 06:57.35 | boddy | 12 45 ok |
| 06:57.43 | dlynes_office | CunningPike: George Pajari...yeah |
| 06:57.49 | CunningPike | Yup |
| 06:57.58 | dlynes_office | CunningPike: he did a presentation on Asterisk two VanLUG meetings ago |
| 06:58.17 | CunningPike | Really? Any good? |
| 06:58.25 | dlynes_office | CunningPike: i don't know...wasn't able to make it |
| 06:58.29 | CunningPike | Ah - OK |
| 06:58.30 | *** join/#asterisk rushowr (n=rushowr@cpe-24-26-133-52.columbus.res.rr.com) |
| 06:58.38 | CunningPike | Well, gents - it's been a slice |
| 06:58.42 | dlynes_office | CunningPike: but, he seems to be in a fairytale land |
| 06:58.46 | *** join/#asterisk qdk (n=qdk@213.237.44.34) |
| 06:58.46 | CunningPike | lol |
| 06:58.48 | rushowr | greets all, howzit goin'? |
| 06:58.55 | rushowr | hope everyone's been well |
| 06:59.01 | dlynes_office | CunningPike: you know that land where asterisk doesn't have any bugs, and you don't have any voip hardware problems? |
| 06:59.10 | CunningPike | We dealt with him briefly at the outset of our projecct |
| 06:59.29 | dlynes_office | Yeah...Gurpreet's had a runin with him too |
| 06:59.34 | dlynes_office | He's not fond of George :) |
| 06:59.54 | rushowr | If anyone might have the time, I'm wrestling with AGI and php scripts. "Getting Exec format error" returned |
| 07:00.05 | rushowr | first real crack at AGI scripts |
| 07:00.10 | CunningPike | We just outgrew him fairly quickly - found that we could find what we needed here, on the list and the wiki |
| 07:00.21 | dlynes_office | CunningPike: ah you mean for a supplier? |
| 07:00.40 | CunningPike | Consultant - also we bought our Digium cards from hium |
| 07:00.56 | CunningPike | He did our initial install |
| 07:01.04 | dlynes_office | CunningPike: yeah...we outgrew him as soon as we realized how badly he was ripping us off on the price for sipura units |
| 07:01.17 | CunningPike | I've rebuilt it since |
| 07:01.17 | rushowr | cheers, figured it out, sorryto hve jumped the gun |
| 07:01.18 | *** part/#asterisk rushowr (n=rushowr@cpe-24-26-133-52.columbus.res.rr.com) |
| 07:01.33 | CunningPike | We got better pricing elsewhere too |
| 07:01.48 | dlynes_office | CunningPike: Yeha...store.voxilla.ca has decent pricing |
| 07:01.52 | CunningPike | OK - I really must go to bed |
| 07:01.55 | dlynes_office | CunningPike: so does voipdepot.ca |
| 07:01.59 | CunningPike | Later, gentlemen |
| 07:02.02 | dlynes_office | seeya |
| 07:04.23 | websae | we have terrific pricing :) |
| 07:04.33 | websae | on all VoIP equipment and PBX hardware :) |
| 07:04.59 | Assid | hrmm.. id love to play with agi;s |
| 07:05.07 | *** join/#asterisk af_ (n=af@ip-143-220.sn1.eutelia.it) |
| 07:06.54 | *** join/#asterisk tuxd00d (n=tuxinato@netblock-68-183-112-100.dslextreme.com) |
| 07:12.13 | *** join/#asterisk ghenry (n=ghenry@mail.suretecsystems.com) |
| 07:14.26 | *** join/#asterisk Gunnar (n=gunnar@bkkb-gw.bitcon.no) |
| 07:14.51 | *** join/#asterisk MGSsancho (n=user@adsl-67-126-128-145.dsl.irvnca.pacbell.net) |
| 07:20.23 | mackss | can someone explain to me what dlns is? |
| 07:20.58 | mackss | dnls |
| 07:21.10 | *** join/#asterisk UlbabraB (n=caplaz@host-84-222-46-79.cust-adsl.tiscali.it) |
| 07:22.16 | *** join/#asterisk lilo_ (i=levin@freenode/staff/pdpc.levin) |
| 07:22.36 | *** join/#asterisk somegeek_ (i=levin@unaffiliated/somegeek) |
| 07:23.24 | pjchilds | mackss, with respect to what? |
| 07:23.51 | *** join/#asterisk nagl (n=nagl@rih.zid-nw.wu-wien.ac.at) |
| 07:26.54 | *** join/#asterisk tuxd00d (n=tuxinato@netblock-68-183-112-100.dslextreme.com) |
| 07:28.55 | mackss | teliax |
| 07:29.14 | Assid | dnls ? |
| 07:29.19 | Assid | stf is that |
| 07:30.33 | mackss | it's under features in teliax |
| 07:31.00 | mackss | "Check if you want your primary device to be SIP for this number. |
| 07:31.00 | mackss | With dnis will add the did # to the string passed." |
| 07:31.10 | mackss | i guess i just don't understand that |
| 07:32.41 | mackss | oh and Assid, teliax caller id setting isn't working properly :( |
| 07:43.01 | *** join/#asterisk VxJasonxV (n=jason@unaffiliated/VxJasonxV) |
| 07:45.24 | *** join/#asterisk tuxd00d (n=tuxinato@netblock-68-183-112-100.dslextreme.com) |
| 07:57.10 | *** join/#asterisk X-Rob (n=rob-x@dsl-202-173-151-24.qld.westnet.com.au) |
| 07:57.24 | *** join/#asterisk somegeek (i=levin@unaffiliated/somegeek) |
| 07:58.13 | *** join/#asterisk saftsack (n=saftsack@p54A7CCE4.dip.t-dialin.net) |
| 08:00.40 | *** join/#asterisk lilo (i=levin@freenode/staff/pdpc.levin) |
| 08:02.16 | stoffell | is it possible to reprogram the "services" button on a polycom 501 to dial a specific number in the dialplan? (wich would then execute an ivr for example) |
| 08:05.22 | *** join/#asterisk tuxd00d (n=tuxinato@netblock-68-183-112-100.dslextreme.com) |
| 08:07.16 | *** join/#asterisk darkskiez (n=darkskie@194.247.78.146) |
| 08:08.47 | *** join/#asterisk [Airwolf] (n=airwolf@83.98.235.219) |
| 08:19.26 | *** join/#asterisk Sonderblade (n=muh@host-213.131.147.169.addr.tdcsong.se) |
| 08:19.45 | *** join/#asterisk littleball (n=littleba@26.203-123-30.leased.qala.com.sg) |
| 08:20.09 | littleball | hi, how to show registered sip users through CLI? |
| 08:20.24 | stoffell | littleball: sip show peers |
| 08:21.59 | Sonderblade | if you are dialing from a SIP phone through asterisk to a PSTN phone, is there a way to timeout the connection attempt if the receiver hasn't answered within X seconds? |
| 08:22.28 | macTijn | dial with the "g" flag |
| 08:22.34 | macTijn | show application dial |
| 08:27.10 | *** join/#asterisk IronHelixz (n=irc@ool-45785cfe.dyn.optonline.net) |
| 08:27.33 | *** join/#asterisk somegeek_ (i=levin@unaffiliated/somegeek) |
| 08:27.53 | *** join/#asterisk lilo_ (i=levin@freenode/staff/pdpc.levin) |
| 08:27.54 | *** join/#asterisk tuxd00d (n=tuxinato@netblock-68-183-112-100.dslextreme.com) |
| 08:29.23 | *** join/#asterisk Zeeek (n=icechat5@pdpc/supporter/active/Zeeek) |
| 08:30.13 | Sonderblade | macTijn: what does that do? |
| 08:30.47 | pjchilds | ~docs |
| 08:30.50 | jbot | rumour has it, 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 |
| 08:31.11 | nettie | Hey guys, I'm having some issues with one of my voip carrier. It always worked but this morning I'm experiencing problem. I place a call when the called party picks up I only hear the first few words, then I dont get anything more. If I put him on HOLD and I take him back a couple of times something happens and I can talk with him again. Anyone know what could be wrong, I'm using asterisk 1.2.6? Thanx in advance. |
| 08:33.22 | *** join/#asterisk gmaruzz (n=gmaruzz@217-133-80-112.b2b.tiscali.it) |
| 08:33.46 | pjchilds | nettie: you could try turning on sip debug and rtp debug, run your test case and see what is happening... but if you have multiple voip providers, and one stopped working right, perhaps give them a call... |
| 08:36.44 | nettie | pjchilds thanx, I'll give rtp debug a try |
| 08:41.58 | *** join/#asterisk RoyK (n=roy@213.160.242.134) |
| 08:43.17 | *** join/#asterisk uwe (n=uwe@dogbert.palnet.com) |
| 08:43.41 | nettie | pjchilds looks pretty incredible I can see my asterisk server receiving and sending the packet, the problem is I dont hear the voice on the phone. I'm also checking phone call statistics and I see 0 as RX packets |
| 08:44.01 | SheriF_WorK | i have GRANDSTREAM BT100 and i can't see the caller ID on the screen in the incoming calls any idea ? |
| 08:44.33 | nettie | asterisk is actually exchanging RTP packets |
| 08:44.58 | nettie | incredible |
| 08:45.34 | Zeeek | SheriF_WorK you know it only shows numerics right? |
| 08:46.32 | SheriF_WorK | Zeeek: nop i don't :-) it can't show user NAME ?? and there is something else even the numerics it's show it own number not the Caller ID |
| 08:46.49 | Zeeek | BT100 shows only numbers |
| 08:47.00 | SheriF_WorK | Zeeek: BT100 extn number is 107 when i call it from 200 i see on the ID 107 |
| 08:47.13 | Zeeek | if there is no callerid you should set the cid to 000 for the BT 100 |
| 08:47.37 | macTijn | Sonderblade: oh, sorry, I made a mistake |
| 08:47.44 | Zeeek | shit happens |
| 08:47.54 | *** join/#asterisk tuxd00d (n=tuxinato@netblock-68-183-112-100.dslextreme.com) |
| 08:47.54 | macTijn | Sonderblade: type "show application dial" in the * console |
| 08:48.14 | macTijn | Sonderblade: it'll explain how to set a timeout on pickup |
| 08:48.49 | Sonderblade | macTijn: so it is not the g flag? |
| 08:50.17 | nettie | anyone know a way to be sure asterisk is transcoding calls? |
| 08:51.06 | nettie | careinvite=no |
| 08:51.12 | macTijn | Sonderblade: no, sorry, I made a mistake there |
| 08:52.08 | *** join/#asterisk IronHelix (n=irc@ool-45785cfe.dyn.optonline.net) |
| 08:52.10 | RoyK | <PROTECTED> |
| 08:52.52 | *** join/#asterisk rodolfo (n=rodolfo@213.215.176.97) |
| 08:52.54 | *** join/#asterisk [Airwolf] (n=airwolf@83.98.235.219) |
| 08:57.26 | *** join/#asterisk PoWeRKiLL (n=PoWeRKiL@84.205.154.179) |
| 08:57.36 | PoWeRKiLL | hi |
| 08:57.54 | PoWeRKiLL | I got this error May 16 10:47:53 WARNING[13409]: chan_zap.c:8392 pri_dchannel: Ring requested on channel 0/1 already in use on span 1. Hanging up owner. |
| 08:58.10 | PoWeRKiLL | It's happens after some minute using queue |
| 09:01.23 | Zeeek | hey PoWeRKiLL |
| 09:01.27 | *** join/#asterisk groogs (n=greg@d221-73-237.commercial.cgocable.net) |
| 09:01.38 | PoWeRKiLL | Hi Zeeek |
| 09:01.52 | Zeeek | working hard? |
| 09:03.40 | *** join/#asterisk chapeaurouge (n=chapeaur@80.92.83.2) |
| 09:04.20 | PoWeRKiLL | Yes very |
| 09:04.38 | Zeeek | J'ai oublié qui t'es.... Olivier? |
| 09:04.51 | PoWeRKiLL | non Lirone |
| 09:04.52 | *** join/#asterisk chapeaurouge (n=chapeaur@80.92.83.2) |
| 09:04.58 | Zeeek | ah yes |
| 09:05.03 | PoWeRKiLL | :) |
| 09:05.13 | Zeeek | I never did get an acocunt at your guys' |
| 09:05.46 | Zeeek | how's it doing? |
| 09:05.48 | PoWeRKiLL | I know I got back the email I sent to you |
| 09:05.58 | *** join/#asterisk tuxd00d (n=tuxinato@netblock-68-183-112-100.dslextreme.com) |
| 09:06.23 | Zeeek | I got the first one but then talking to service commercial I think it never happened |
| 09:06.34 | Zeeek | I should go look at the offer |
| 09:06.42 | Zeeek | anyway you're still there? |
| 09:06.44 | PoWeRKiLL | yes |
| 09:06.51 | Zeeek | so it's cool... |
| 09:08.35 | *** join/#asterisk subdolus (n=subby@subby.afraid.org) |
| 09:09.00 | *** join/#asterisk rhowe (i=rhowe@87.127.14.179) |
| 09:11.08 | *** join/#asterisk Doctor (n=mage@S01060007950ade7b.vc.shawcable.net) |
| 09:13.57 | Doctor | I'd wanna set up asterisk with a phoneline in card and have it give an error when I try to dial out through it, and the phone is off the hook, but also be able to just join in as well. that doable? |
| 09:14.01 | *** join/#asterisk MGSsancho (n=user@adsl-67-126-128-145.dsl.irvnca.pacbell.net) |
| 09:18.19 | fourcheeze | why would I get a message "Stopped Sounds" back from a channel? |
| 09:21.40 | *** join/#asterisk tparcina (n=tparcina@wr-lama.iskon.hr) |
| 09:21.47 | tparcina | hi group |
| 09:22.04 | tparcina | i have missed you guys for last 10 days |
| 09:22.31 | tparcina | doo, It was beautifull in budapest... |
| 09:24.55 | *** join/#asterisk zobia (n=laura_sh@222.209.225.26) |
| 09:25.01 | zobia | Hello everyone |
| 09:25.09 | tparcina | hi zobia |
| 09:25.15 | zobia | does any one has experience to config realtime of sip peers? |
| 09:25.32 | zobia | hi tparcina |
| 09:25.36 | tparcina | question - where to download Grandstream gpx-2000 firmware |
| 09:25.49 | tparcina | don't use realtime... |
| 09:25.55 | zobia | sorry tparcina, i don't know that |
| 09:26.01 | tparcina | I don't use realtime... |
| 09:26.02 | zobia | why don't use realtime? |
| 09:26.12 | stoffell | tparcina, check voip-info.org, there's the GXP-2000 info and firmware |
| 09:26.32 | tparcina | stoffell, thanks, I will |
| 09:26.52 | *** join/#asterisk gmaruz1 (n=gmaruzz@217-133-80-112.b2b.tiscali.it) |
| 09:26.54 | tparcina | zobia, I'm not so good with sql and I didn't have time to try it |
| 09:26.58 | zobia | i follow the steps to config sip realtime. but could not work |
| 09:27.08 | *** join/#asterisk sternn (n=sternn@user-0c938ku.cable.mindspring.com) |
| 09:28.54 | *** join/#asterisk tuxd00d (n=tuxinato@netblock-68-183-112-100.dslextreme.com) |
| 09:29.38 | tparcina | zobia, that's the reason I haven't try it :)) I don't have that much knowledge to use it in production, and I don't have time to playwith it |
| 09:30.18 | zobia | ok. thanks tparcina. |
| 09:30.40 | zobia | anyone else has the experience of realtime sip? |
| 09:30.52 | *** join/#asterisk coppice (n=chatzill@32.193.0.218.broad.hz.zj.dynamic.cndata.com) |
| 09:31.26 | tparcina | stoffell, is Unofficial Internal Build of 1.1.0.1 good enough? can I download some official Grandstream firmware? |
| 09:33.45 | fourcheeze | zobia what do you mean by "doesn't work" ? |
| 09:34.10 | tparcina | I have downloaded http://www.grandstream.com/DOWNLOAD/FIRMWARE/GXP2000/GXP2000_Release_1.0.1.12.zip Can someone confirm that is working fine? |
| 09:34.51 | zobia | ok fourcheeze, i create change extconfig.conf. and create sip tables. and through 1 sip users inside, then restart asterisk. try to registered this user. it showes user regiter failed |
| 09:35.22 | zobia | and if i use sip show peers could not see the on in the database(while i think the realtime could not be seen by sip show peers at all). |
| 09:36.17 | zobia | and i don't know how to debug with this sip users. |
| 09:36.39 | zobia | could not find any error in /var/log/asterisk/debug as it mentioned on the page |
| 09:36.39 | fourcheeze | zobia what do you have in your extconfig? |
| 09:36.54 | fourcheeze | zobia use pastebin.ca |
| 09:37.01 | zobia | i put two lines in extconfig |
| 09:37.02 | zobia | sipusers => mysql,scriptmaker,sip_users |
| 09:37.03 | zobia | sippeers => mysql,scriptmaker,sip_users |
| 09:37.23 | fourcheeze | ok, I use odbc for my realtime |
| 09:37.27 | fourcheeze | but mysql is supposed to work |
| 09:37.59 | zobia | yes. my voicemail and extension realtim is working at the same time |
| 09:37.59 | fourcheeze | zobia one idea is to turn on full logging in mysql |
| 09:38.07 | fourcheeze | ok |
| 09:38.45 | fourcheeze | you need to see what's being logged in mysql |
| 09:38.47 | zobia | ok. fourcheeze, is there any way to debug this in asterisk CLI? |
| 09:38.59 | zobia | ok. let me do the log |
| 09:39.27 | fourcheeze | zobia it's normal for sip show peers to be empty when you're using realtime and no peers are registered |
| 09:40.06 | zobia | yes. i think so. by sip show peers is not the right way. |
| 09:40.26 | fourcheeze | sip show peers will work once you have some that register |
| 09:41.05 | nettie | guys, anyone know if there's an opensource g729 codec for asterisk please? |
| 09:41.22 | fourcheeze | nettie not in the true sense of the word |
| 09:41.23 | zobia | ok. thanks fourcheeze. let me try that log |
| 09:41.38 | fourcheeze | nettie all the g729 codecs come with some restrictions |
| 09:42.48 | fourcheeze | zobia do you have the dbname/dbhost etc stuff in the [general] section of sip.conf ? |
| 09:44.05 | coppice | nettie: true open source fo G.729 isn't really possible, because of patents |
| 09:45.29 | fourcheeze | coppice funny how that didn't stop anyone writing their own pdf viewer |
| 09:46.25 | coppice | what has that to do with it? PDF is an open spec, which people are encouraged to implement. There are no patent issues |
| 09:46.30 | kay2 | is it possible to do an Hangup on AGI using the uniqueID instead of the channel ?? |
| 09:47.03 | fourcheeze | coppice ok, maybe I meant ebooks |
| 09:47.17 | zobia | fourcheeze. i dont have dbname/dbhost in sip.conf |
| 09:47.29 | *** join/#asterisk tuxd00d (n=tuxinato@netblock-68-183-112-100.dslextreme.com) |
| 09:47.45 | zobia | but i have them in res_mysql.conf |
| 09:47.49 | dlynes_office | fourcheeze: yeah...there's issues with ebooks, but not pdf |
| 09:48.57 | fourcheeze | zobia interesting - I have both |
| 09:49.03 | nettie | fourcheeze, coppice I understnad.. in fact I bought 1 channel license from digium :) |
| 09:49.12 | fourcheeze | I'm not sure if I really need the ones in sip.conf |
| 09:49.34 | tparcina | any grandstream gpx-2000 user? |
| 09:49.49 | tparcina | is this firmware good enough? - Software Version: Program-- 1.0.1.9 Bootloader-- 1.0.1.2 |
| 09:50.08 | fourcheeze | nettie it depends where you live - in some places software patents are a grey area or not supported |
| 09:50.19 | nettie | fourcheeze, if I use g729 to connect my voip carrier and g729 on the phones I should be fine with just 1 license? |
| 09:50.49 | fourcheeze | nettie it depends if your users need to do things like voicemail, meetme etc |
| 09:50.53 | nettie | even if the phones are configured with canreinvite=no which means there's an actual transcoding on the server? |
| 09:51.18 | *** join/#asterisk voipaster (i=25x8supp@203.192.191.36) |
| 09:51.30 | fourcheeze | if you're just doing passthrough then you don't need the codec AFAIK |
| 09:51.32 | *** join/#asterisk A-Tuin|work (n=A-Tuin@212.41.185.81) |
| 09:51.54 | dlynes_office | nettie: canreinvite=no means asterisk stays in the media path |
| 09:51.54 | nettie | fourcheeze the mainly need to make outbound calls.. |
| 09:52.14 | fourcheeze | then you probably don't need a codec |
| 09:52.33 | fourcheeze | also calls between clients shouldn't need a codec |
| 09:52.46 | nettie | dlynes_office yes, I use that option because my phones are behind NAT and the asterisk server is at the colo |
| 09:52.49 | fourcheeze | just don't expect to be able to play them sounds |
| 09:53.00 | nettie | uhmm |
| 09:53.04 | nettie | what about MOH? |
| 09:53.30 | dlynes_office | nettie: correct. however, because it's still in the media path, you need to rely on asterisk's codec resolution which is less than perfect |
| 09:54.19 | dlynes_office | nettie: asterisk won't always get the codec translation paths correct as needed for passthrough codecs |
| 09:55.13 | dlynes_office | nettie: however, if you're going to run into that problem, you'll notice that problem during the first ten tries or so |
| 09:55.40 | dlynes_office | nettie: if you don't notice codec resolution problems after the first ten tries or so, it probably won't be an issue for you |
| 10:00.09 | *** join/#asterisk gmaruzz (n=gmaruzz@217-133-80-112.b2b.tiscali.it) |
| 10:00.41 | nettie | got it.. |
| 10:01.09 | fourcheeze | nettie but you won't get MOH |
| 10:01.27 | fourcheeze | unless you buy codecs |
| 10:02.17 | fourcheeze | i.e. convert mp3 to g729 |
| 10:02.35 | fourcheeze | and then just play it back to a client that wants it |
| 10:04.04 | nettie | I'm actuall yusing ulaw, the problem is that I'm having problems prioritizing it using Cisco LLQ QoS.. so I thought that a codec with a smaller "footprint" could make things a bit better. |
| 10:04.09 | *** join/#asterisk MacDome (n=eseidel@c-71-198-177-144.hsd1.ca.comcast.net) |
| 10:04.22 | *** part/#asterisk rodolfo (n=rodolfo@213.215.176.97) |
| 10:05.47 | *** join/#asterisk kio (n=kio@ool-4577ae5e.dyn.optonline.net) |
| 10:05.56 | nettie | my upstream is 256Kbit, (320Kbit synched on the ATM int) if I make calls and I upload the voice gets very choppy.. I already opened a case with Cisco but in the meantime I was looking for a workaround.. |
| 10:06.00 | fourcheeze | nettie how many concurrent calls? |
| 10:06.10 | nettie | this is the idea to switch codec.. |
| 10:06.16 | nettie | fourcheeze 1 call is nice |
| 10:06.20 | nettie | 2 calls are nice |
| 10:06.20 | *** join/#asterisk mut (n=animenod@65.111.222.120) |
| 10:06.29 | *** join/#asterisk iceyp (n=icepick@firewall.unix.co.nz) |
| 10:06.33 | *** join/#asterisk mrtwister|mobile (n=andrius@cable-10-68.cgates.lt) |
| 10:06.36 | iceyp | hey guys, how can i solve this issue: May 16 22:04:34 NOTICE[59198]: rtp.c:570 ast_rtp_read: Unknown RTP codec 101 received |
| 10:06.37 | nettie | the problem appear with even 1 call and an upload |
| 10:06.48 | fourcheeze | yeah, qos should solve that |
| 10:06.51 | nettie | only when Internet is used for data |
| 10:06.53 | *** join/#asterisk [bsd] (n=bsd@59.144.253.40) |
| 10:06.54 | *** join/#asterisk tuxd00d (n=tuxinato@netblock-68-183-112-100.dslextreme.com) |
| 10:07.13 | nettie | fourcheeze I'm astonished because I have experience with tc on linux, on the wrt54g as well and so on |
| 10:07.16 | nettie | never had issues |
| 10:07.17 | fourcheeze | however, you're right that reducing the bandwidth of your voice helps |
| 10:07.34 | nettie | I configured it on this cisco router |
| 10:07.37 | fourcheeze | nettie do you have any other codec options, like gsm for instance? |
| 10:07.38 | [bsd] | hi, friends! |
| 10:08.09 | fourcheeze | nettie in any case you should be able to test your theory without needing to buy a codec |
| 10:08.20 | PoWeRKiLL | someone can help me with this message May 16 11:58:10 WARNING[6019]: chan_zap.c:8392 pri_dchannel: Ring requested on channel 0/3 already in use on span 1. Hanging up owner. |
| 10:08.26 | nettie | I quadchecked the policies, they both matches, the acl match traffic gets out in the right queue but it still soudns choppy |
| 10:08.39 | nettie | yeah |
| 10:08.42 | nettie | I could use GSM |
| 10:08.46 | fourcheeze | gsm is cool |
| 10:08.48 | *** join/#asterisk voipaster (i=25x8supp@203.192.191.36) |
| 10:08.50 | fourcheeze | I really like it |
| 10:08.50 | nettie | is it? |
| 10:09.11 | [bsd] | Can someone help me to compile chan_h323 for * svn on knoppix |
| 10:09.14 | Zeeek | GetSomeMoney |
| 10:09.18 | mut | anyone know how to block pings on a SPA ata other than putting a firewall infront of it? |
| 10:09.23 | nettie | I read around that the MOS wasnt big deal and I never spent time on it.. |
| 10:09.36 | mut | if ya send it a 65000 byte ping you can kill it for a good 20 seconds with 1 ping |
| 10:10.30 | *** join/#asterisk A-Tuin|work (n=A-Tuin@212.41.185.81) |
| 10:10.34 | coppice | GSM is not a bad balance - modest bit rate, reasonable quality, unencumbered |
| 10:10.45 | mut | well all of ours are natt'd to the net |
| 10:10.48 | fourcheeze | gsm would be better if more providers supported it |
| 10:10.49 | mut | but on the wan there are a lot |
| 10:10.49 | coppice | modest compute load |
| 10:10.50 | nettie | fourcheeze I'll give it a shot in the meantime the Cisco engineer figure out what's wrong with my router |
| 10:11.12 | *** join/#asterisk puzzled (n=patrick@puzzled.xs4all.nl) |
| 10:11.22 | fourcheeze | all the big providers seem to want ulaw/alaw or g729 |
| 10:11.25 | fourcheeze | or else g723 |
| 10:11.37 | nettie | yeah.. anyway I can definitely hear the difference |
| 10:11.48 | fourcheeze | you'll hear the difference with g729 as well |
| 10:11.53 | nettie | from ulaw and g729 |
| 10:11.57 | nettie | yeah |
| 10:11.57 | fourcheeze | yeah |
| 10:11.59 | nettie | definitely |
| 10:12.02 | coppice | g.723.1 is fading away now |
| 10:12.04 | nettie | I have 1 channel license |
| 10:12.06 | nettie | I tried it.. |
| 10:12.09 | puzzled | afternoon all |
| 10:12.12 | nettie | incredible difference |
| 10:12.26 | nettie | not that much when you talk but with MOH is just crazy |
| 10:12.29 | fourcheeze | nettie in that case have you tried g726? |
| 10:12.35 | nettie | like tapes vs CDs :) |
| 10:12.50 | fourcheeze | l |
| 10:13.05 | nettie | fourcheeze nope.. all I tried was g729 and ulaw |
| 10:13.05 | puzzled | anyone know if "freshmaker failed register test" on a TDM400P indicates a hardware failure? |
| 10:13.19 | coppice | nettie: low bit rate codecs are tuned for human speech. the less like a single person talking, the worse it codes |
| 10:13.19 | nettie | eheh fourcheeze u're 0ldsk000l |
| 10:13.20 | nettie | ehehe |
| 10:13.21 | nettie | :) |
| 10:13.21 | fourcheeze | g726 is about the quality of ulaw but about 3/5 of the bandwidth |
| 10:13.52 | nettie | g726 looks interesting then.. |
| 10:13.52 | fourcheeze | never understood anyone that likes tapes though |
| 10:13.58 | nettie | yeah |
| 10:14.01 | nettie | tapes are crazy |
| 10:14.04 | fourcheeze | they keep going wobbly |
| 10:14.26 | fourcheeze | yeah, try g726 |
| 10:14.34 | fourcheeze | see if you can find someone that can terminate it for you too |
| 10:14.44 | nettie | my carrier support it |
| 10:14.45 | nettie | uhmm |
| 10:14.52 | nettie | not sure my phones does anyway |
| 10:14.56 | nettie | supports it |
| 10:14.57 | fourcheeze | which phone? |
| 10:14.59 | nettie | poly |
| 10:15.05 | fourcheeze | can't remember |
| 10:15.11 | nettie | I think they're strictly u/alaw and g729 |
| 10:15.18 | nettie | checking now.. |
| 10:15.20 | *** part/#asterisk [bsd] (n=bsd@59.144.253.40) |
| 10:15.24 | fourcheeze | well you can do ulaw -> asterisk -> g726 -> provider |
| 10:15.32 | nettie | this wont solve |
| 10:15.33 | nettie | the issue |
| 10:15.38 | uwe | wasim, hello, yesterday i asked about how to know what file name is being recorded to, you pointed out MONFILE, this is for asterisk 1.2.x, im still stuck with 1.0.x, is there a way to do it too? |
| 10:15.48 | nettie | the problem is phones->asterisk |
| 10:15.49 | fourcheeze | nettie where's your asterisk? |
| 10:15.52 | nettie | colo |
| 10:15.57 | fourcheeze | ahh |
| 10:16.41 | fourcheeze | then you are stuck |
| 10:17.34 | nettie | eheh |
| 10:17.36 | nettie | I know :) |
| 10:17.38 | uwe | wasim, i found ${CALLFILENAME} and SetVar(MONITOR_FILENAME ... , im unable to make use of them, can someoe give any hints |
| 10:17.52 | nettie | gigastucked |
| 10:17.54 | ghenry | can blf be done over a IAX2 channel? Looks like only SIP and ZAP in the docs |
| 10:18.15 | dlynes_office | ghenry: there's a number of channels that support blf |
| 10:18.25 | dlynes_office | ghenry: skinny supports it too |
| 10:18.36 | dlynes_office | ghenry: i don't know if iax supports it, offhand...one second |
| 10:19.34 | fourcheeze | is it possible to have a separate server handle the blf stuff? |
| 10:19.42 | nettie | fourcheeze yeah.. the phones supports only 711u A and 729AB |
| 10:19.42 | fourcheeze | I'm looking to scale up from a single server |
| 10:19.54 | fourcheeze | and that seems to be the sticking point if I go for SER |
| 10:20.00 | wasim | uwe: should be simple |
| 10:20.00 | fourcheeze | as a front end |
| 10:20.35 | fourcheeze | what I'd like is to be able to subscribe to a differnt server for the blf/presence |
| 10:20.35 | ghenry | one of our 9133i is dead |
| 10:20.50 | ghenry | doesn't take a DHCP IP, although I can see it asking for one |
| 10:21.28 | dlynes_office | ghenry: looks like iax supports it too |
| 10:21.40 | dlynes_office | ghenry: but iax and mgcp don't call it blf |
| 10:21.47 | dlynes_office | ghenry: they refer to it as devstate |
| 10:21.47 | uwe | wasim, well, the problem is that SetVar(MONITOR_FILENAME is only used once, for those files which i have here starting with "q" |
| 10:21.48 | ghenry | cool |
| 10:21.57 | coppice | one day the phone makers will see a good surgeon and get their heads removed from the other end of their anatomy. Then they might start supporting some wideband codecs :-) |
| 10:21.59 | nettie | fourcheeze SER is fine only to route SIP calls right? If you need VM, IVR and stuff like that you need * |
| 10:22.20 | dlynes_office | ghenry: try this, and scroll down to the section about the hint extension: http://www.voip-info.org/wiki/view/Asterisk+standard+extensions |
| 10:22.29 | fourcheeze | dlynes_office ghenry : which iax clients support blf? |
| 10:22.44 | dlynes_office | fourcheeze: i have no idea, but it's not blf, per se |
| 10:22.48 | dlynes_office | fourcheeze: it's devstate |
| 10:24.34 | file | IAX2 does not support having a device get the status of another device like SIP does |
| 10:25.42 | dlynes_office | file: voip-info says otherwise |
| 10:25.55 | file | URL? |
| 10:26.08 | dlynes_office | http://www.voip-info.org/wiki/view/Asterisk+standard+extensions |
| 10:26.36 | *** join/#asterisk tuxd00d (n=tuxinato@netblock-68-183-112-100.dslextreme.com) |
| 10:26.57 | file | I don't see it saying IAX2 supports it |
| 10:27.00 | ghenry | dlynes_home: It's just I have 2 * boxes at different sites, connected via IAX2, you _5XXX,1,Dial blah. You think I could get the status over that channel? |
| 10:27.04 | dlynes_office | although according to that url, zap supports it, and zap doesn't support it |
| 10:27.13 | dlynes_office | file: type show channeltypes at your cli |
| 10:27.14 | file | you're misunderstanding it |
| 10:27.33 | dlynes_office | file: ? |
| 10:27.38 | file | it means that chan_iax2 has the capability to give the state of a peer to the core |
| 10:27.52 | dlynes_office | ah...not to other peers? |
| 10:27.58 | file | correct |
| 10:28.13 | dlynes_office | So why is it grouped under the heading of 'hint', then? |
| 10:28.35 | file | on voip-info? because that's the way someone wrote it... |
| 10:28.41 | file | if you don't like it, edit it |
| 10:28.51 | dlynes_office | Yeah....it's misleading |
| 10:28.55 | *** join/#asterisk Dr-Linux (n=Nothing@202.125.141.6) |
| 10:29.07 | dlynes_office | ghenry: i guess not :0 |
| 10:29.20 | Dr-Linux | dlynes_office: oo i got you |
| 10:29.27 | ghenry | thanks |
| 10:29.35 | Dr-Linux | dlynes_home: i'm facing some problem, wanna ask you |
| 10:30.32 | dlynes_office | Dr-Linux: just ask on here...i'm going to go grab a coffee shortly |
| 10:30.40 | Dr-Linux | I am getting not echo, but duplicate packets on the voice for the PBX. Its not as if I’m getting echo coming back, but I actually can hear duplicate frames. I say this because of the tone. It does not seem to be an echo as much as getting a frame repeatedly. |
| 10:31.13 | dlynes_office | Dr-Linux: iax, sip, or pri? |
| 10:31.23 | Dr-Linux | dlynes_office: SIP |
| 10:31.33 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 10:32.54 | dlynes_office | hrm...could've sworn i heard someone mention something about a jitter buffer on sip earlier today |
| 10:33.08 | dlynes_office | Dr-Linux: Is this on an ata, or a sip phone? |
| 10:33.29 | *** part/#asterisk Poincare (n=jefffnod@amp89.ampersant.be) |
| 10:33.45 | Dr-Linux | it's all |
| 10:33.52 | Dr-Linux | it's something wrong with server |
| 10:33.55 | dlynes_office | if it's a sipura ata |
| 10:34.04 | dlynes_office | you can try adjusting the jitter buffer |
| 10:34.11 | dlynes_office | I would suggest upgrading to the latest firmware |
| 10:34.24 | dlynes_office | the latest firmware has more flexibility with respect to jitter buffers |
| 10:34.58 | dlynes_office | also, is this for the american phones, or for the pakistani phones? |
| 10:37.56 | *** join/#asterisk iceyp (n=icepick@firewall.unix.co.nz) |
| 10:38.32 | iceyp | hey guys, is it possible to do a Dial() then a wait,5 then send a dtmf ... not sending it myself, but getting the pabx to dial the options for me? |
| 10:39.09 | dlynes_office | iceyp: is this for an autoattendant you wan tto dial through after the remote end answers the call? |
| 10:39.10 | ghenry | see with agents (new to this ;-) ), i've logged in, my phone rings when the exten with the queue is dialed, but do I need to acknowledge the call to get them transferred to my phone? |
| 10:39.17 | *** join/#asterisk Tili (n=Tili@cm33.gamma249.maxonline.com.sg) |
| 10:39.32 | iceyp | dlynes_office having issues sending dtmf myself, so want the pabx to do it for me |
| 10:39.38 | ghenry | brb |
| 10:39.49 | dlynes_office | iceyp: but after the remote end answers the call, right? |
| 10:39.55 | dlynes_office | iceyp: not before? |
| 10:39.58 | iceyp | yes |
| 10:40.07 | dlynes_office | iceyp: Use the D() parameter to dial then |
| 10:40.11 | iceyp | so remote answers (thats why wait,5) |
| 10:40.57 | iceyp | ahh ok |
| 10:41.02 | Dr-Linux | dlynes_home: this problem is for american phone, and for Cisco 7960 |
| 10:41.17 | iceyp | so dial.. then enxt line d(5) |
| 10:41.18 | dlynes_office | Dr-Linux: i know nothing about cisco anything |
| 10:41.30 | iceyp | or within the same line as dial, where r for ring usualy goes |
| 10:41.30 | Dr-Linux | dlynes_office: but i'm not sure, if it's server end problem or client end issue, what you say? |
| 10:41.38 | dlynes_office | Dr-Linux: But try adjusting the jitter buffer on your iax trunk |
| 10:41.59 | dlynes_office | Dr-Linux: also check iax2 show netstats on your iax trunk to see if you're getting any jitter |
| 10:42.10 | iceyp | would this work ,rwd(2)wd(1) |
| 10:42.11 | Dr-Linux | dlynes_office: i'm not using any iax trunk, |
| 10:42.28 | dlynes_office | Dr-Linux: you're not doing iax2 between the two asterisk boxes? |
| 10:42.45 | Dr-Linux | dlynes_office: never |
| 10:42.52 | dlynes_office | Dr-Linux: ah...just sip? |
| 10:43.16 | Dr-Linux | dlynes_office: and forget any any trunk, it's only server and SIP extensions are registered with it, and facing such problems |
| 10:43.17 | iceyp | dlynes_office didnt work .... the reason i want to do it is because i get the following error when inserting dtmf myself... May 16 22:42:44 NOTICE[29450]: rtp.c:570 ast_rtp_read: Unknown RTP codec 101 received |
| 10:43.40 | iceyp | dlynes_office would this be correct dial SIP/0018774678758@sip.voipdiscount.com|60|rwd(2)wd(1) |
| 10:43.53 | dlynes_office | iceyp: what dtmfmode are you using in your sip.conf file? |
| 10:43.57 | iceyp | r for ring, w for wait 5 seconds, d(2) for option 2 and d(1) for option 1 |
| 10:44.40 | iceyp | dlynes_office works fine when i use my sip phone but for some reason my dsl is broken... i'm calling into my pabx via pstn -> sip inbound provider -> asterisk -> voipdiscount |
| 10:44.58 | iceyp | so i think it's when it merges the 2 channels |
| 10:45.17 | dlynes_office | Your sip inbound provider configuration file on asterisk |
| 10:45.23 | dlynes_office | How do you have your dtmfmode set on it? |
| 10:45.38 | *** join/#asterisk tuxd00d (n=tuxinato@netblock-68-183-112-100.dslextreme.com) |
| 10:45.49 | *** join/#asterisk stoffell (n=stoffell@fw.catsanddogs.com) |
| 10:45.55 | Dr-Linux | dlynes_office: anything, that should at server end? |
| 10:46.07 | Dr-Linux | dlynes_office: do you think it's a server side issue? |
| 10:46.08 | iceyp | dtmfmode=rfc2833 |
| 10:46.18 | Dr-Linux | dlynes_office: even i didn't change anything. |
| 10:46.18 | iceyp | auto on voipdiscount |
| 10:46.24 | iceyp | i've tried hard setting it too |
| 10:46.24 | dlynes_office | iceyp: now, your inbound provider...are they using rfc2833, or info? |
| 10:46.44 | iceyp | dlynes_office rfc2833 works fine when i call my pbx as i have an ivr |
| 10:46.58 | iceyp | and i have to input a number to dial via the voipdiscount |
| 10:47.12 | iceyp | and if i call direct from my cisco phone through voipdiscount dtmf works fine |
| 10:47.43 | dlynes_office | iceyp: have you tried setting your inbound provider to info, instead of rfc2833? |
| 10:47.50 | iceyp | let me try |
| 10:48.06 | dlynes_office | iceyp: also, what codec are you using on your inbound? ulaw, g729, ...? |
| 10:48.25 | iceyp | gsm |
| 10:48.31 | iceyp | and gsm to the voipdiscount |
| 10:49.19 | iceyp | if i set inbound provider to info, i cant select any options in my ivr, so i cant even call via voipdiscount |
| 10:50.36 | dlynes_office | iceyp: ah..ok...set it back to rfc2833 |
| 10:50.56 | dlynes_office | is it just a particular phone number you're calling where you're having problems with ivr? |
| 10:51.05 | dlynes_office | or is it all numbers through voipdiscount? |
| 10:51.20 | dlynes_office | Dr-Linux: i have no idea |
| 10:51.30 | dlynes_office | Dr-Linux: but it sounds like possibly a jitter issue |
| 10:52.00 | iceyp | if i call into my inbound number (ivr) its fine... if i call from my ivr to a USA number (via voipdiscount) i cant enter dtmf.... If i call direct (from my sip cisco phone - when it works) via voipdiscount i can send dtfm |
| 10:52.03 | *** join/#asterisk chapeaurouge (n=chapeaur@vilhost1.vision.lu) |
| 10:52.13 | dlynes_office | Dr-Linux: how many simultaneous calls do you usually have between pakistan and the us? |
| 10:52.26 | iceyp | so right now i need to make an urgent call and want the pabx to insert the dtmf options for me so i can go to bed then ill troubleshoot tommorow |
| 10:52.34 | *** join/#asterisk [Airwolf] (n=airwolf@83.98.235.219) |
| 10:52.41 | dlynes_office | iceyp: like i said...the D() parameter |
| 10:53.17 | iceyp | dlynes_office exten => 222,1,Dial(SIP/0018774678758@sip.voipdiscount.com,60,rwd(2w1)) |
| 10:53.21 | iceyp | doesnt seem to work |
| 10:54.18 | iceyp | ahh D |
| 10:54.20 | iceyp | woops |
| 10:54.21 | *** join/#asterisk Winkie (n=urmom@cpc3-stre1-0-0-cust656.bagu.cable.ntl.com) |
| 10:54.24 | dlynes_office | exten => 222,1,Dial(SIP/0018774678758@sip.voipdiscount.com|60|D(ww1)) ; |
| 10:54.44 | dlynes_office | ww will give you a 1 second pause |
| 10:54.56 | dlynes_office | after the remote party answers, of course |
| 10:56.27 | Dr-Linux | <dlynes_office> Dr-Linux: but it sounds like possibly a jitter issue: |
| 10:57.05 | Dr-Linux | dlynes_office: what jitter issue, even same problem is gettting the peoples who are on same LAN , where pbx box |
| 10:57.40 | dlynes_office | Dr-Linux: are you using qos? |
| 10:57.53 | iceyp | thnx it worked |
| 10:59.21 | Dr-Linux | dlynes_office: nope |
| 10:59.54 | dlynes_office | Dr-Linux: then don't discount it being a jitter issue just because it's all on the same lan |
| 11:00.35 | Dr-Linux | dlynes_office: i'm using softphone here in pakistan, and we have no issues |
| 11:00.50 | dlynes_office | yet |
| 11:00.59 | Dr-Linux | dlynes_office: not sure, why 2 guys reported this problem, both are using Cisco phones :S |
| 11:01.14 | Dr-Linux | my CEO and president |
| 11:01.17 | dlynes_office | Tell them they're lying |
| 11:01.24 | dlynes_office | Then you don't have to deal with it |
| 11:01.27 | dlynes_office | Just ignore the problem |
| 11:01.31 | dlynes_office | Hope it'll go away |
| 11:01.59 | Dr-Linux | dlynes_office: He is my CEO and other one is our operational director VP |
| 11:02.37 | dlynes_office | You liars!!!! You're not getting stuttery talk! You couldn't possibly be! You're both using Cisco phones!!! Cisco's immune!!! :) |
| 11:03.00 | Dr-Linux | lolzzz |
| 11:03.48 | Dr-Linux | dlynes_office: however i just restart my asterisk |
| 11:04.06 | Assid | stupid sipdiscount.. its charging me for calls to hk |
| 11:04.18 | dlynes_office | The nerve of them |
| 11:04.26 | dlynes_office | charging for international toll calls |
| 11:05.01 | Assid | err.. sipdiscount not supposed to charge for calls to HK |
| 11:05.09 | Dr-Linux | dlynes_office: how can i change it |
| 11:05.11 | Dr-Linux | May 16 03:42:52 WARNING[23098]: pbx.c:5818 pbx_builtin_dtimeout: DigitTimeout is deprecated, please use Set(TIMEOUT(digit)=timeout) instead. |
| 11:05.19 | fourcheeze | anyone got any advice on scaling up and maintaining working blfs ? |
| 11:05.20 | dlynes_office | Dr-Linux: ummm |
| 11:05.24 | dlynes_office | Dr-Linux: are you retarded? |
| 11:05.30 | Dr-Linux | after restarting asterisk, i got this warning |
| 11:05.34 | Dr-Linux | retarded? :S |
| 11:05.36 | dlynes_office | Dr-Linux: the warning message is telling you point blank how to fix it |
| 11:06.00 | dlynes_office | fourcheeze: i've got it working fine for 6 extensions |
| 11:06.16 | fourcheeze | dlynes_office I want it working for N |
| 11:06.24 | fourcheeze | and over several servers |
| 11:06.38 | Dr-Linux | dlynes_office: currenlty i'm using like this >> DigitTimeout,2 |
| 11:06.46 | Dr-Linux | ResponseTimeout,6 |
| 11:06.57 | dlynes_office | fourcheeze: i would think that would be very dependent on how you're doing your distributed setup |
| 11:07.09 | fourcheeze | dlynes_office well that's all up for grabs ATM |
| 11:07.17 | fourcheeze | right now we have a single asterisk server with about 100 users |
| 11:07.35 | dlynes_office | fourcheeze: and you want to be able to do hinting on 100 extensions? |
| 11:07.44 | fourcheeze | no, I want it to scale up |
| 11:07.46 | Dr-Linux | Set(TIMEOUT(digit)=6) << it will be like this? |
| 11:07.57 | dlynes_office | fourcheeze: ok...let's restate it |
| 11:08.07 | dlynes_office | fourcheeze: you want to be able to do hinting on 100 extensions, or more? |
| 11:08.18 | dlynes_office | Dr-Linux: very good |
| 11:08.34 | Dr-Linux | dlynes_office: :S |
| 11:08.37 | Dr-Linux | dlynes_office: kidding? :S |
| 11:08.48 | fourcheeze | dlynes_office ok let's make some assumptions |
| 11:09.06 | dlynes_office | fourcheeze: have you got it working with any extensions yet? |
| 11:09.09 | zobia | thanks fourcheeze. i made it |
| 11:09.25 | zobia | the realtime sip needs host to be "dynamic |
| 11:09.33 | Dr-Linux | dlynes_office: is that fine as i tried? if you are not kidding? |
| 11:09.39 | fourcheeze | dlynes_office I have hints working fine ATM |
| 11:09.40 | dlynes_office | Dr-Linux: yes |
| 11:09.45 | Dr-Linux | dlynes_office: thanks |
| 11:09.55 | dlynes_office | fourcheeze: ah |
| 11:09.58 | fourcheeze | dlynes_office what I want to do is scale up to several thousand users |
| 11:10.03 | fourcheeze | which means more than 1 asterisk |
| 11:10.12 | dlynes_office | fourcheeze: yeah, personally I think you're asking for trouble |
| 11:10.20 | fourcheeze | I want users to be able to log into any asterisk and still get blf |
| 11:10.26 | dlynes_office | fourcheeze: the hinting code from what i understand is not terribly well written |
| 11:10.44 | fourcheeze | yes, that might well be a problem |
| 11:10.51 | fourcheeze | in which case I need "something else" |
| 11:10.55 | fourcheeze | :-) |
| 11:11.17 | dlynes_office | fourcheeze: you could always patch it up, and store the blf states in your realtime configuration |
| 11:11.29 | fourcheeze | well, that's ideally what I'd like |
| 11:11.37 | dlynes_office | fourcheeze: and then when you need to push the states, just grab the states from the database |
| 11:11.50 | fourcheeze | realtime doesn't seem to work like that though |
| 11:12.01 | fourcheeze | the realtime database seems to be read fairly infrequently |
| 11:12.04 | dlynes_office | fourcheeze: i'm guessing certain extensions will only care about the extensions in their own group |
| 11:12.08 | fourcheeze | wiht a lot of caching |
| 11:12.14 | fourcheeze | which is good for performance |
| 11:12.34 | fourcheeze | but not good for a blf |
| 11:13.02 | fourcheeze | so, I've looked into proxies, like SER |
| 11:13.09 | Ahrimanes | blf ? |
| 11:13.10 | fourcheeze | which seems to have a presence module |
| 11:13.18 | dlynes_office | Ahrimanes: busy lamp field |
| 11:13.24 | Ahrimanes | oh |
| 11:13.27 | dlynes_office | Ahrimanes: extensions statii |
| 11:13.39 | dlynes_office | Ahrimanes: it'll let you know if it's busy, ringing, or idle |
| 11:13.50 | Ahrimanes | dlynes_office: ie hints? |
| 11:13.55 | fourcheeze | yep, that's it |
| 11:13.56 | dlynes_office | Ahrimanes: correct |
| 11:14.09 | fourcheeze | Ahrimanes any idea how to scale it past a single * ? |
| 11:15.07 | stoffell | I'm using a SIP softphone (x-lite) to register to an * behind a nat, but the * server replies not on 5060, but tries send to port 10043 |
| 11:15.15 | Ahrimanes | fourcheeze: i would suspect that's not really currently possible unless you do your own code for it, i think that there may be something on the way to help use blf anyway you want which could be used with agi or other external code to sync between servers |
| 11:15.18 | stoffell | anything i can try to troubleshoot? (i found that out through tcpdump) |
| 11:15.36 | Ahrimanes | stoffell: * behind nat is not good |
| 11:15.37 | Assid | argh.. stupid nuts are charging me for that call |
| 11:15.58 | fourcheeze | anyone used SER on the front of asterisk with blfs? |
| 11:15.58 | Dr-Linux | May 16 04:03:42 WARNING[23954]: ast_expr2.fl:176 ast_yyerror: ast_yyerror(): syntax error: syntax error, unexpected TOK_PLUS, expecting TOK_MINUS or TOK_COMPL or TOK_LP or TOKEN; Input: |
| 11:15.58 | Dr-Linux | <PROTECTED> |
| 11:15.58 | Dr-Linux | <PROTECTED> |
| 11:15.58 | Dr-Linux | May 16 04:03:42 WARNING[23954]: ast_expr2.fl:180 ast_yyerror: If you have questions, please refer to doc/README.variables in the asterisk source. |
| 11:16.08 | Dr-Linux | what the hell is this :S |
| 11:16.13 | stoffell | Ahrimanes, yeah, sure isn't :( when building my own vpn's, i use openvpn and don't do nat, this is a different thing :( |
| 11:16.15 | dlynes_office | Dr-Linux: you're asking us? |
| 11:16.21 | fourcheeze | Dr-Linux "If you have questions, please refer to doc/README.variables in |
| 11:16.22 | fourcheeze | <PROTECTED> |
| 11:16.24 | fourcheeze | " |
| 11:16.27 | dlynes_office | Dr-Linux: it's a syntax error in your dialplan, obviously |
| 11:16.56 | Ahrimanes | stoffell: having the softphone behind nat is pretty easy, but having * behind nat really is shooting yourself in the foot |
| 11:18.02 | stoffell | Ahrimanes, i'm not even sure. I have a client, connecting to checkpoint-vpn server (with token), and the * server is in that network. so the checkpoint-server is doing the NAT I suppose. is that possible? |
| 11:18.20 | Assid | anyone know any providers for free callsto HK ? |
| 11:18.30 | dlynes_office | Assid: skype |
| 11:18.52 | Assid | dlynes_office: not free to HK .. and i need something on sip/iax to put into my box |
| 11:19.17 | *** part/#asterisk zobia (n=laura_sh@222.209.225.26) |
| 11:19.18 | dlynes_office | man |
| 11:19.24 | dlynes_office | wtf is with these 9133i's? |
| 11:19.56 | Ahrimanes | stoffell: hm it's not really needed to do nat in all vpn situations, if you do routing instead of nat a lot of problems are avoided.. i would contect the checkpoint vpn-1 admin to ask whether nat is happening or not |
| 11:21.37 | fourcheeze | dlynes_office are blf and presence interchangeable terms? |
| 11:21.49 | dlynes_office | fourcheeze: no |
| 11:22.00 | dlynes_office | presence is soemthing else, altogether |
| 11:22.18 | fourcheeze | "Presence is something else" |
| 11:22.26 | fourcheeze | sorry, bad joke |
| 11:22.27 | mut | anyone keep any call stats at all? |
| 11:22.31 | mut | like an mrtg style? |
| 11:22.33 | stoffell | Ahrimanes, ok, will try.. |
| 11:22.35 | stoffell | thanks |
| 11:22.38 | Ahrimanes | stoffell: np |
| 11:22.49 | mut | or like |
| 11:22.51 | fourcheeze | dlynes_office so presence doesn't work with things like snom phones |
| 11:23.05 | mut | times of day, days of week, avg duration |
| 11:23.07 | mut | anythin |
| 11:23.46 | dlynes_office | hrm |
| 11:23.48 | dlynes_office | nvm |
| 11:23.55 | dlynes_office | seems like presence might be the same thing |
| 11:24.07 | fourcheeze | it would be nice if it was |
| 11:24.13 | dlynes_office | Just it used to be called presence when the feature was first being developed, or soemthing |
| 11:24.38 | dlynes_office | but there's another presence supported by asterisk too |
| 11:24.42 | dlynes_office | bluetooth presence |
| 11:24.53 | dlynes_office | and that's completely unrelated |
| 11:25.35 | ghenry | is it possible to only give the dail tone when 9 is pressed? |
| 11:25.54 | *** join/#asterisk denon (i=denon@synapse.subneural.net) |
| 11:25.54 | *** mode/#asterisk [+o denon] by ChanServ |
| 11:26.41 | dlynes_office | ghenry: look up the info for the application, 'DISA' |
| 11:27.06 | dlynes_office | is it just me, or are 9133i's kinda flaky? |
| 11:27.09 | ghenry | you do plain E&M with a hardware card and 2 ports, when properly configured in zaptel.conf right? So the other just rings once the first is picked up? |
| 11:27.30 | dlynes_office | no idea |
| 11:27.54 | ghenry | just thinking back to my graduate days as a telecoms eng. |
| 11:28.10 | *** join/#asterisk tuxd00d (n=tuxinato@netblock-68-183-112-100.dslextreme.com) |
| 11:29.19 | *** join/#asterisk gmaruz1 (n=gmaruzz@217-133-80-112.b2b.tiscali.it) |
| 11:29.47 | *** join/#asterisk michael-i (n=michael-@141.41.38.58) |
| 11:30.38 | vgster | whats up with 9133is |
| 11:30.38 | dlynes_office | Yeah...they seem to be flaky as hell |
| 11:30.44 | dlynes_office | They never seem to want to stay registered |
| 11:31.03 | dlynes_office | Adn i'm continually getting internal server error 500's on them |
| 11:31.33 | ghenry | my 1 of 2 is fine here |
| 11:31.41 | ghenry | 2 one dead. won't get an IP |
| 11:31.48 | ghenry | or let me in options on phone |
| 11:31.52 | dlynes_office | ghenry: i've got 8 |
| 11:32.00 | ghenry | even default 192.168.0.50 won't work |
| 11:32.10 | dlynes_office | ghenry: 6 of them seem to have a local echo problem on the handset |
| 11:32.11 | ghenry | dlynes_home: Ok |
| 11:32.28 | dlynes_office | ghenry: the web interface for them is quite buggy |
| 11:32.34 | ghenry | dlynes_home: My one has that, but figured it was due to the X100p not setup right |
| 11:32.40 | dlynes_office | ghenry: apparently the best way to go on them is through the configuration file |
| 11:33.08 | dlynes_office | ghenry: is it where the speaker on the aastra phone hears his/her own voice, and doesn't hear any other echo? |
| 11:33.09 | ghenry | ok |
| 11:33.18 | dlynes_office | ghenry: and the remote caller hears no echo? |
| 11:33.34 | dlynes_office | ghenry: I get that on six of them, but it only happens on voip calls, not on analog calls |
| 11:33.44 | ghenry | yeah |
| 11:33.57 | ghenry | only via sip. Don't have any FXS ports |
| 11:34.19 | dlynes_office | Yeah, for me it's SIP->IAX->SIP, or SIP->IAX->PRI |
| 11:34.46 | dlynes_office | or PRI->IAX->SIP |
| 11:35.06 | ghenry | We do SIP<->IAX<->SIP and SIP<->ZAP |
| 11:35.16 | ghenry | or SIP<->SIP internally |
| 11:35.20 | dlynes_office | yeah..i do both of those, two |
| 11:35.24 | dlynes_office | s/two/too/ |
| 11:35.44 | ghenry | nice bot ;-) |
| 11:35.58 | ghenry | The other person at end of line could hear their voice a wee bit too |
| 11:36.15 | dlynes_office | Yeah...that's not the case with my problem |
| 11:36.16 | ghenry | where's the echo settings on a X100P? zapata.conf? |
| 11:36.22 | dlynes_office | ghenry: correct |
| 11:36.32 | ghenry | ta |
| 11:36.49 | ghenry | dlynes_home: I do like the phone though |
| 11:36.51 | dlynes_office | ghenry: rxgain, txgain, echocancel=yes/no, echocancelwhenbridged=yes/no, echotraining=yes/0-infinity |
| 11:37.07 | dlynes_office | ghenry: yeah...it's a nice looking phone, with a nice, heavy, durable handset |
| 11:37.24 | dlynes_office | and the 480i's have that nice big display |
| 11:37.38 | ghenry | Although, is it standard to have to program a viocemail exten no. on the phone, even if the phone tells you you have a voicemail? |
| 11:37.41 | ghenry | I think so... |
| 11:37.51 | dlynes_office | but now, with gwb screwing the us economy over royally |
| 11:38.01 | dlynes_office | polycoms might be cheaper than aastras now, for us :) |
| 11:38.11 | ghenry | you in states? |
| 11:38.14 | dlynes_office | Canada |
| 11:38.16 | vgster | i havent had any probs with my 9133is |
| 11:38.27 | vgster | other than restarting the asterisk server does confuse the blf |
| 11:38.37 | dlynes_office | vgster: which firmware/hardware revisions are you using? |
| 11:38.43 | vgster | 1.3.1 |
| 11:38.47 | vgster | firmware |
| 11:39.13 | dlynes_office | vgster: i'm using 1.3.1.1095 firmware, and 1.1.0.11 boot rom |
| 11:39.27 | vgster | ill check for exact numbers |
| 11:39.43 | ghenry | me too |
| 11:39.49 | vgster | im on the same vers as you |
| 11:39.53 | ghenry | can you dl a new bootrom? |
| 11:39.54 | vgster | whats flaky about them? |
| 11:39.58 | ghenry | couldn't see one |
| 11:40.08 | vgster | i thougth aastra were working on a new firmware anyways |
| 11:40.36 | ghenry | also, timezone can set London, but doesn't have +1 to Daylight Saving |
| 11:40.43 | Assid | err.. how do i make something like random? where it randomly chooses which provider to make a call through |
| 11:41.03 | vgster | my BST conversion works |
| 11:41.11 | dlynes_office | I get internal server error 500's on some phones, flaky hinting on some phones, settings that seem to stick, ... |
| 11:41.39 | *** join/#asterisk psk (n=psk@golia.caltanet.it) |
| 11:41.39 | vgster | the blf i get on a few phones |
| 11:41.49 | ghenry | DISA isn't right dlynes_home |
| 11:41.51 | vgster | think thats firmware but i dont get any other errors |
| 11:41.57 | dlynes_office | ghenry: sure it is |
| 11:42.16 | dlynes_office | ghenry: disa drops you into a new dial tone |
| 11:42.23 | dlynes_office | ghenry: and then you can dial whatever number you want |
| 11:43.00 | dlynes_office | ghenry: you can optionally specify an authentication code to prompt the user with, and you can also specify the dialplan context to dial out on |
| 11:43.25 | ghenry | Direct Inward System Access? |
| 11:43.36 | ghenry | Allows someone from outside the telephone switch (PBX) to obtain an "internal" system dialtone |
| 11:43.36 | dlynes_office | ghenry: like exten => 9,1,DISA(mycontext) ; or something like that |
| 11:43.54 | ghenry | or is this just what someone wrote on yhe wiki? |
| 11:44.02 | dlynes_office | ghenry: you can describe it however you want... |
| 11:44.09 | dlynes_office | ghenry: however, it does what you want |
| 11:44.31 | ghenry | ok, thanks |
| 11:44.41 | dlynes_office | ghenry: it's the same thing as on a normal pbx, where you dial 9 to get an outside line |
| 11:44.46 | ghenry | yeah |
| 11:44.59 | dlynes_office | ghenry: so just set up extension 9 to go into disa |
| 11:45.08 | ghenry | will try |
| 11:45.19 | dlynes_office | ghenry: make sure you set an authentication code on it so that not just anybody can use it |
| 11:45.35 | dlynes_office | ghenry: also make sure that it's not accessible from your main incoming context |
| 11:45.35 | Assid | if i use a random for goto.. and if it doesnt go to that label.. does it just jump to the next priority ? |
| 11:45.47 | dlynes_office | Assid: should, yeah |
| 11:45.48 | ghenry | well, dlynes_home users need to dial out. |
| 11:46.01 | *** join/#asterisk smurf (n=smurf@debian/developer/smurf) |
| 11:46.01 | vgster | aastra should stick a gsm codec in too |
| 11:46.08 | dlynes_office | ghenry: just make sure it's not accessible from the incoming context |
| 11:46.21 | ghenry | dlynes_home: I have this at the mo: exten => _9XXXXXXXXXXX,1,Dial(${OUTBOUNDTRUNK}/${EXTEN:1}) |
| 11:46.22 | dlynes_office | ghenry: or you'll have outsiders making free long distance calls on your tab |
| 11:46.45 | ghenry | dlynes_home: We don't use an incoming just now, but thanks for the tip |
| 11:47.08 | many | hai smurf. |
| 11:47.40 | *** join/#asterisk tuxd00d (n=tuxinato@netblock-68-183-112-100.dslextreme.com) |
| 11:48.17 | dlynes_office | exten => s,8,DISA,no-password|outbound-context |
| 11:48.22 | dlynes_office | something like that, ghenry |
| 11:48.38 | dlynes_office | inside the DISA, they'll dial another number |
| 11:48.55 | dlynes_office | and then it'll go to the outbound-context dialplan context to determine how to dial that number |
| 11:49.17 | dlynes_office | so if you want, you can additionally prevent people from calling long distance, too |
| 11:50.04 | *** join/#asterisk RoyK (n=roy@80.239.107.70) |
| 11:56.45 | *** join/#asterisk kiania (n=kiania@196.201.229.36) |
| 11:58.51 | kiania | boy this chat group is lively |
| 12:00.39 | ghenry | do I have do reload the zaptel module is I change zapata.conf ? |
| 12:00.41 | ghenry | is jsut *? |
| 12:00.47 | ghenry | or just * |
| 12:01.09 | kiania | Need some help I have a channel bank connecting to a TE401P card on the E1 interface |
| 12:01.36 | kiania | the connection comes up for a while then goes down when I do PRI Show Span 1 on the asterisk box |
| 12:01.57 | kiania | the Channel bank shows TE1 on the same interface ... what am I doing wrong |
| 12:02.42 | kiania | just the * by doing > reload |
| 12:03.00 | stoffell | if I use sip on local lan, the conversations between pbx<->softphone go on port UDP5060, but on doing this through a remote firewall, the return from the pbx is not on port sip but on pord 100XX, why is that? |
| 12:07.10 | *** join/#asterisk tuxd00d (n=tuxinato@netblock-68-183-112-100.dslextreme.com) |
| 12:09.00 | *** part/#asterisk tparcina (n=tparcina@wr-lama.iskon.hr) |
| 12:12.58 | dlynes_office | stoffell: that's because your client is going outbound, and hte port number you see is the arbitrary port number that it created on |
| 12:13.19 | dlynes_office | stoffell: 5060 is the sip listening port |
| 12:13.30 | dlynes_office | stoffell: outbound client sockets can be on any port |
| 12:13.47 | stoffell | dlynes_office, yes, but on my lan the outbound (on the client) is also 5060, not an arbitrary port.. |
| 12:14.15 | dlynes_office | yeah..it could be 5060, it could be 10020, it could be anything |
| 12:14.57 | stoffell | hm, but it can be forced to always use 5060, not? |
| 12:15.18 | dlynes_office | sure, assuming you have a phone that allows you to do that |
| 12:15.26 | dlynes_office | but it doesn't matter |
| 12:15.31 | dlynes_office | it'll work, regardless |
| 12:15.43 | stoffell | it'll work, unless there's a firewall in between.. |
| 12:15.56 | stoffell | so to ease up on fw administration, forcing it to 5060 could be a good thing |
| 12:16.04 | dlynes_office | stoffell: the port number it's on has nothing to do with your firewall |
| 12:16.19 | dlynes_office | stoffell: why do you need to administer anything, period? |
| 12:16.47 | dlynes_office | stoffell: the whole point of nat=yes, canreinvite=no, qualify=xxxx is so that you don't have to do any firewall administration |
| 12:17.15 | *** join/#asterisk ghenry (n=ghenry@mail.suretecsystems.com) |
| 12:17.28 | dlynes_office | ghenry: reload chan_zap.so |
| 12:17.45 | stoffell | hm... okay.. so connecting 'to' the * works, because i see incoming packets (tcpdump), but they don't get to the client, that's a firewall problem, right? |
| 12:18.11 | dlynes_office | stoffell: it's a firewall problem, but it's not solved by opening up ports on the firewall |
| 12:18.20 | dlynes_office | stoffell: that only exacerbates the problem |
| 12:18.27 | dlynes_office | stoffell: are you using nat=yes? |
| 12:18.42 | stoffell | well, yes. nat=yes, qualify=yes and canreinvite=no |
| 12:18.47 | ghenry | dlynes_home: I just hung the server by messing with modprobe and had to reboot ;-) |
| 12:18.53 | dlynes_office | stoffell: set qualify=300 |
| 12:19.00 | dlynes_office | stoffell: see if that makes a difference |
| 12:19.10 | dlynes_office | ghenry: good job :) |
| 12:19.12 | stoffell | ok, trying |
| 12:19.18 | ghenry | Ha! |
| 12:19.33 | dlynes_office | stoffell: you might need a reboot of your sip phones |
| 12:19.43 | dlynes_office | stoffell: shouldn't need to reboot asterisk though...just sip reload |
| 12:19.56 | *** join/#asterisk oej (n=oej@apollo.webway.se) |
| 12:20.05 | dlynes_office | good morning, olle |
| 12:20.09 | stoffell | yeah, i'm using x-lite, first stopped |
| 12:20.10 | stoffell | it |
| 12:20.17 | oej | Morning |
| 12:20.20 | oej | Afternoon |
| 12:20.21 | *** join/#asterisk [TK]D-Fender (n=joe@toronto-HSE-ppp4122655.sympatico.ca) |
| 12:20.24 | oej | Or whatever :-) |
| 12:20.35 | dlynes_office | stoffell: is xlite behind the firewall? |
| 12:20.37 | stoffell | hm, the sip request gets to the *, and the * tries sending back to port 10100 ... but still not logged in |
| 12:21.07 | stoffell | dlynes, yes. the * server is on the remote end. the vpn between my laptop and the firewall is built (checkpoint firewall/vpn) |
| 12:21.23 | dlynes_office | stoffell: is asterisk behind a different firewall? |
| 12:21.26 | [TK]D-Fender | oej : Happy Birthday ! |
| 12:21.40 | stoffell | dlynes_office, no, that same checkpoint |
| 12:22.03 | dlynes_office | stoffell: ok, so they're both on the same vpn? |
| 12:22.22 | dlynes_office | stoffell: therefore, effectively on the same subnet? |
| 12:22.44 | stoffell | dlynes_office, not the same subnet, but the routing is ok, there's no firewall between |
| 12:23.03 | stoffell | (too bad we don't have a drawing board here) |
| 12:23.04 | dlynes_office | ok, i have no idea then |
| 12:23.12 | dlynes_office | never tried to work with sip on a vpn before |
| 12:23.24 | stoffell | okay, my best bet is to consult the firewall/vpn admin to see if he sees anything |
| 12:23.35 | stoffell | i do it all the times, but with openvpn... that works like a breeze.. |
| 12:24.01 | *** join/#asterisk UlbabraB (n=caplaz@host-84-222-47-43.cust-adsl.tiscali.it) |
| 12:24.49 | dlynes_office | man |
| 12:24.52 | dlynes_office | almost 5am already |
| 12:25.00 | dlynes_office | erm 5:30am i mean |
| 12:25.46 | *** join/#asterisk tuxd00d (n=tuxinato@netblock-68-183-112-100.dslextreme.com) |
| 12:27.58 | *** join/#asterisk gmaruzz (n=gmaruzz@217-133-80-112.b2b.tiscali.it) |
| 12:28.32 | *** join/#asterisk fourcheeze (n=rich@westbury.doilywood.org.uk) |
| 12:29.11 | hwt | what does this mean? |
| 12:29.12 | hwt | May 16 14:29:02 WARNING[4886]: channel.c:506 ast_best_codec: Don't know any of 0xf800 formats |
| 12:29.36 | [TK]D-Fender | hwt : Means the client is offering something outside of the range of anything known to * |
| 12:30.17 | oej | Thanks D-Fender! |
| 12:30.51 | zoa | hey ho mr olle! |
| 12:30.55 | ghenry | vgster: You still there? |
| 12:30.58 | zoa | it your birthday again today ?! |
| 12:31.11 | oej | Every day |
| 12:31.20 | *** join/#asterisk Red5 (n=Red5@CPE-203-56-119-133.ethertech.com.au) |
| 12:31.21 | oej | No, it was yesterday but I can't remove the tag line :-) |
| 12:31.22 | ghenry | vgster: Does aastra.cfg overwrite everything on the phone? |
| 12:31.23 | michael-i | a bit OT, but i just got a box here that someone else setup and it keeps looking for a proxy that they were using... |
| 12:31.26 | hwt | [TK]D-Fender: codec? |
| 12:31.33 | hwt | [TK]D-Fender: this is astertest |
| 12:31.34 | Ahrimanes | oej: thanks for the mailinglist :) |
| 12:31.42 | michael-i | i'm a bsd guy and don't know where that'd be setup....it's knoppix.. |
| 12:31.44 | michael-i | any ideas? |
| 12:31.48 | oej | Ahrimanes: You're welcome |
| 12:32.21 | Ahrimanes | oej: good to see people allready coordinating on it :) |
| 12:33.04 | oej | :-) |
| 12:33.19 | Red5 | hey all.. a quickie.. i want to be able to call into asterisk as normal, but enter a secret exten/password to enable me to dialout again? is this poss ? whats it called ? |
| 12:33.47 | [TK]D-Fender | oej : Any chance you can add the AMP/FreePBX/A@H referral to #freepbx warning back into the channel topic? :) |
| 12:34.05 | oej | I can't change the topic, sorry |
| 12:34.13 | [TK]D-Fender | NNot op? wow |
| 12:34.23 | [TK]D-Fender | I just sortof assumed... |
| 12:34.30 | [TK]D-Fender | That'll learn me ;) |
| 12:34.59 | [TK]D-Fender | Red5 : DISA <- Look it up on the Wiki |
| 12:35.01 | *** join/#asterisk Winkie (n=urmom@cpc3-stre1-0-0-cust656.bagu.cable.ntl.com) |
| 12:35.05 | Ahrimanes | Red5: you can set up the default context to allow incomming calls to sip devices but require outgoing calls to come from other contexts |
| 12:35.45 | dlynes_office | michael-i: what kinda proxy? |
| 12:35.46 | Red5 | thank you.. D-Fender.. i'll check... |
| 12:36.08 | dlynes_office | ghenry: it's supposed to |
| 12:36.39 | dlynes_office | ghenry: but you need macaddress.cfg also |
| 12:36.42 | michael-i | dlynes_office, not sure...links keeps redirecting web traffic to a proxy, nothing shows in ps, nothing shows in netstat... |
| 12:36.58 | ghenry | thanks dlynes_home |
| 12:37.06 | michael-i | the beauty of recieving no docs... |
| 12:37.15 | dlynes_office | michael-i: man links? man elinks? |
| 12:37.17 | *** join/#asterisk dpryo (i=hn@donatello.nesland.net) |
| 12:37.43 | dlynes_office | ghenry: i haven't gone home yet |
| 12:38.02 | dlynes_office | ghenry: had to provision some hardware and aastra's been driving me up the wall all night |
| 12:38.08 | ghenry | dlynes_office: I am letting you go home now ;-) |
| 12:38.23 | michael-i | dlynes_office, links isn't my problem...just a side-effect thereof. Perhaps just a few places to poke around? |
| 12:38.24 | dlynes_office | ghenry: it's almost 6am...i'll be staying up |
| 12:38.24 | Ahrimanes | michael-i: is the HTTP_PROXY enviroment variable set? |
| 12:38.34 | ghenry | ;-) |
| 12:38.57 | dlynes_office | Ahrimanes: exactly why i was suggesting the man page :)...i figured it probably used an environment variable :) |
| 12:39.00 | michael-i | pinging doesn't work either, nor dns resolv... even though the network's setup correctly as far as ifconfig goes |
| 12:39.03 | ghenry | anyone done it so users number can be picked of via LDAP? |
| 12:39.19 | ghenry | For part of a FindMe DialPlan |
| 12:39.55 | dlynes_office | michael-i: no go on the http_proxy environment variable? |
| 12:40.12 | Ahrimanes | hm that sounds like firewalling |
| 12:40.31 | dlynes_office | Ahrimanes: well, if the environment variable doesn't solve it |
| 12:40.40 | dlynes_office | Ahrimanes: i'm guessing it might be a transparent proxy server |
| 12:40.47 | michael-i | dlynes_office, nothing there... |
| 12:41.03 | dlynes_office | michael-i: iptables -nL | grep 3128 |
| 12:42.41 | dlynes_office | anything there? |
| 12:42.48 | michael-i | dlynes_office, complains about missing/outdated kern mods |
| 12:43.03 | dlynes_office | any hits, though? |
| 12:43.45 | *** join/#asterisk bkw_ (n=brian@adsl-70-142-39-36.dsl.tul2ok.sbcglobal.net) |
| 12:43.54 | dlynes_office | morning, bkw |
| 12:45.05 | michael-i | iptables can't initialize .... blah blah |
| 12:45.25 | michael-i | missing module ip_tables |
| 12:45.37 | dlynes_office | ah...ok so you're not running a firewall then |
| 12:45.39 | dlynes_office | erm wait |
| 12:45.55 | dlynes_office | uname -a |
| 12:46.36 | michael-i | Linux 2.6.16.14-nosmp #7 PREEMPT |
| 12:46.51 | dlynes_office | ok, so no chance of you using ipchains then |
| 12:46.58 | dlynes_office | ipchains isn't even supported on 2.6 |
| 12:47.19 | *** join/#asterisk sremington (n=sremingt@rrcs-24-123-247-29.central.biz.rr.com) |
| 12:47.30 | michael-i | it boggles me because I can't find any evidence of anything running that'd be doing this... |
| 12:47.44 | dlynes_office | michael-i: have you tried a different browser? |
| 12:47.54 | dlynes_office | michael-i: like maybe telnet hostname 80? |
| 12:47.55 | Nugget | telnet is eeeeeeevil! |
| 12:48.18 | *** join/#asterisk tuxd00d (n=tuxinato@netblock-68-183-112-100.dslextreme.com) |
| 12:48.32 | michael-i | dlynes_office, if i could even get my dns/ping to work, i'd think about browsing. I'll see what it says |
| 12:48.36 | dlynes_office | if telnet doesn't go through all that crap, it's gotta be an environment variable or a configuration file |
| 12:48.48 | michael-i | it's HTTP 1.0 \n\n right? |
| 12:48.53 | Ahrimanes | yeah |
| 12:48.59 | dlynes_office | GET / HTTP/1.0\n\n |
| 12:49.01 | Ahrimanes | GET / HTTP/1.0\n\n |
| 12:49.02 | Ahrimanes | hehe |
| 12:49.09 | *** join/#asterisk stkn_ (n=stkn@gentoo/developer/pdpc.active.stkn) |
| 12:49.11 | michael-i | whoa :) thanks |
| 12:49.13 | Ahrimanes | telnet doesnt ready *_PROXY vars |
| 12:49.53 | dlynes_office | michael-i: did that work? |
| 12:50.13 | michael-i | nope |
| 12:50.22 | mut | anyone wanna give me a gift of $280? |
| 12:50.26 | dlynes_office | michael-i: you're going through a transparent proxy server then |
| 12:50.28 | Ahrimanes | mut: what? |
| 12:50.29 | michael-i | destination net unreachable |
| 12:50.31 | mut | i need to get another monitor |
| 12:50.38 | dlynes_office | michael-i: oh...then maybe not |
| 12:50.39 | mut | so i can learn to RE better |
| 12:50.41 | Ahrimanes | michael-i: network even working in _any_ way?? |
| 12:50.53 | Ahrimanes | mut: Regular Expressions? |
| 12:51.01 | mut | and 2 monitors are a necessity |
| 12:51.04 | dlynes_office | michael-i: traceroute www.microsoft.com? |
| 12:51.07 | mut | reverse engineer |
| 12:51.10 | Ahrimanes | ah |
| 12:51.13 | michael-i | Ahrimanes, anything within my subnet is working. Even though my gateway is setup correctly (route) nothing works outside of my /24 |
| 12:51.24 | michael-i | dlynes_office no dns... |
| 12:51.25 | Ahrimanes | mut: RE your current monitor and rebuilt another one from that ;) |
| 12:51.32 | mut | heh |
| 12:51.34 | dlynes_office | michael-i: ah |
| 12:51.48 | Ahrimanes | michael-i: can you ping your gateway? |
| 12:51.50 | *** join/#asterisk [Airwolf] (n=airwolf@83.98.235.219) |
| 12:52.01 | michael-i | Ahrimanes, yup...but the next hop nope. |
| 12:52.11 | dlynes_office | michael-i: try this DNS server: 216.251.149.66 |
| 12:52.26 | Ahrimanes | michael-i: check your gateway? |
| 12:52.32 | SplasPood | michael-i: you have a full /24? |
| 12:53.00 | *** join/#asterisk bjohnson (n=bjohnson@jecinc.tor.istop.com) |
| 12:55.36 | *** join/#asterisk arcy (n=arcanum@ppp78-37.adsl.forthnet.gr) |
| 12:55.53 | SplasPood | michael-i: whats your subnet mask? |
| 12:56.14 | michael-i | ok...some of my fooling around had lost the default route (checked like 5 commands ago...= |
| 12:56.22 | michael-i | so...now I'm still working on resolution |
| 12:57.20 | michael-i | i need to keep rolling across the room to work on this thing so I'll be back in a bit....my cavvy's are gettin a burn :) |
| 12:57.40 | SplasPood | double check the ip of the machine and it's netmask while yer at it |
| 12:59.34 | michael-i | SplasPood, /24 as it should be |
| 13:00.11 | michael-i | but yes, the thievery begins....wget's looking for a 192.168.x.x:3128 !! |
| 13:00.12 | SplasPood | Just checking... |
| 13:00.22 | SplasPood | heh |
| 13:00.24 | michael-i | no problem, it's always the small stuff |
| 13:01.02 | SplasPood | michael: so wait, you wget some IP /w port 80, and it redirs you to some 192.168 port 3128? |
| 13:01.56 | Ahrimanes | env var or login.conf entry sort of thing? |
| 13:02.16 | michael-i | ok, http_proxy and ftp_proxy are set (lowercase) now to remove that permanently |
| 13:02.32 | michael-i | Ahrimanes, looking now... |
| 13:03.21 | *** join/#asterisk fjean (n=fjean@201.29.140.206) |
| 13:03.32 | fjean | 'morning.. |
| 13:03.42 | *** join/#asterisk imperfect- (n=tbw@207.210.85.90) |
| 13:04.11 | imperfect- | Is anyone here using asterisk w/ broadvoice? |
| 13:04.26 | *** join/#asterisk jmacz (n=jmacz@201.244.241.55) |
| 13:04.28 | *** join/#asterisk satlan32 (n=pargit@212.150.142.211) |
| 13:04.32 | *** join/#asterisk znoG (n=gs@109-130-89-200.fibertel.com.ar) |
| 13:04.38 | [TK]D-Fender | imperfect- : 2 of my clients do. |
| 13:05.21 | imperfect- | Well I'm about to pull my hair out. ;) |
| 13:05.36 | imperfect- | I can recieve calls just peachy, alas, when i try to make them I keep getting 604 errors. |
| 13:05.51 | imperfect- | (doesn't exist anywhere) from broadvoice |
| 13:06.28 | [TK]D-Fender | pastebin your dial-plan that calls out, and your sip.conf entries related to your account (edit out PW's) |
| 13:06.29 | [TK]D-Fender | ~pb |
| 13:06.31 | jbot | methinks pb is a place to paste your stuff without flooding the channel - try http://pastebin.com/, or http://pastebin.ca |
| 13:06.48 | imperfect- | okay. |
| 13:07.24 | fjean | hey guys - I am having a lot of SIP channels that would hang there hours after the call is terminated, and it happens a lot, is this "normal" behaviour ? |
| 13:07.33 | *** join/#asterisk mercestes (n=merceste@69.15.174.114) |
| 13:07.42 | fjean | I mean, is there a way to control this |
| 13:07.55 | *** join/#asterisk tuxd00d (n=tuxinato@netblock-68-183-112-100.dslextreme.com) |
| 13:09.12 | brettnem | imperfect-: maybe you need a "1" in front of your PSTN number |
| 13:09.26 | brettnem | fjean: what asterisk version? |
| 13:09.30 | imperfect- | yeah I've done that |
| 13:09.46 | fjean | imperfect: 1.2.6 |
| 13:09.54 | imperfect- | http://pastebin.ca/56198 |
| 13:10.13 | *** join/#asterisk jimmy_deanPB (n=jhodapp@indianalifesciences.com) |
| 13:10.14 | fjean | I got like 70 calls like that just yesterday |
| 13:10.26 | brettnem | fjean: I get a lot of this kind of behavior with asterisk.. however, that kind of behavior could be caused from a network problem or a faulty UA |
| 13:10.35 | brettnem | fjean: perhaps the RTP timeout stuff could help you out |
| 13:10.57 | *** join/#asterisk Eggplant (n=none@dsl-216-155-212-221.cascadeaccess.com) |
| 13:11.11 | [TK]D-Fender | imperfect- : Ok, pastebin a call attempt in CLI. in sip.conf you mis-spelled "canreinvite", and it should be set to NO. |
| 13:11.23 | imperfect- | Ah! |
| 13:11.25 | imperfect- | Lesse |
| 13:11.34 | michael-i | ok, now I can't seem to find where these things are set upon boot.... no rc's at all in /root/ |
| 13:11.43 | imperfect- | can reinvite? |
| 13:11.48 | imperfect- | not careinvite ? |
| 13:11.50 | Ahrimanes | michael-i: in /etc/ somewhere? |
| 13:11.51 | fjean | brettnem: I notice it happens a lot on certain days, otherwise it's only one or two which I consider normal |
| 13:11.54 | imperfect- | no space, though, right |
| 13:11.57 | [TK]D-Fender | imperfect- : You alsomispelled the realm entry which I'm sure is your REAL problem. |
| 13:12.14 | brettnem | rc's in /root/ ? |
| 13:12.14 | imperfect- | hehe |
| 13:12.16 | imperfect- | You are so right! |
| 13:12.29 | imperfect- | I knew an extra set of eyes would help. |
| 13:12.36 | [TK]D-Fender | imperfect- : You're typing worse tham ME now.. not a good sign... |
| 13:12.42 | Ahrimanes | brettnem: nevermind :P |
| 13:12.45 | imperfect- | Ever just look at something until you glaze over and don't notice little stuff? |
| 13:12.56 | brettnem | like this channel? |
| 13:12.58 | brettnem | ;) |
| 13:13.22 | Ahrimanes | hehe |
| 13:13.45 | fjean | brettnem: could this be caused by miscomunication between the asterisk and the termination ? |
| 13:13.51 | brettnem | yes |
| 13:14.12 | brettnem | fjean: what do you think would happen if CANCELs or BYEs never made it to asterisk? |
| 13:14.16 | *** join/#asterisk zotz (n=zotz@24.231.36.155) |
| 13:14.19 | brettnem | and say... |
| 13:14.26 | brettnem | the UA was something like.. asterisk |
| 13:14.30 | imperfect- | [TK]: Dude. |
| 13:14.34 | fjean | bretnem, well yes |
| 13:14.38 | imperfect- | [TK]: Will you be my new daddy. Thank you so much. |
| 13:14.39 | *** join/#asterisk hgaillac (n=Harry@224.15.119-80.rev.gaoland.net) |
| 13:14.50 | brettnem | imperfect-: haha |
| 13:14.51 | hgaillac | Hello, |
| 13:14.56 | imperfect- | brettnem: ;) |
| 13:15.03 | imperfect- | Man I was looking at it until like 1am |
| 13:15.06 | imperfect- | just boggled |
| 13:15.07 | imperfect- | I'm like |
| 13:15.08 | imperfect- | it looks right |
| 13:15.12 | imperfect- | flippin mispelling! |
| 13:15.16 | michael-i | ok...nothing there except /etc/login.defs |
| 13:15.25 | imperfect- | I actually had someone else look at it too and they didnt see it either |
| 13:15.27 | znoG | hah, gotta love google and forums |
| 13:15.36 | michael-i | nothing meaningful in /etc/skel either |
| 13:15.40 | znoG | just found the hidden page on the RT31P2 to edit the dial plan and all the other settings |
| 13:16.14 | fjean | brettnem: the termination happens to be SER, anything special I should tell them , or check with them ? |
| 13:16.15 | hgaillac | I need somebody to test gnugk=oh323/asterisk is their somebody with a h323 terminal not behind a nat |
| 13:16.45 | brettnem | fjean: what does sip show channels look like? |
| 13:16.51 | brettnem | hgaillac: barf |
| 13:17.15 | brettnem | RT31P2? Is that a star wars character? |
| 13:17.18 | fjean | brettnem: when this happens, last night I had 80 channels but only two active calls |
| 13:17.34 | brettnem | fjean: what kind of call volume do you "typically" have? |
| 13:18.00 | brettnem | you should check a typical call and make sure all the right ACKs and BYEs and such are transmitted and received |
| 13:18.29 | brettnem | it could be a SER routing problem (I've messed up SER code and made this happen before) |
| 13:19.22 | imperfect- | Thanks again. |
| 13:19.38 | *** topic/#asterisk by russellb -> Asterisk: The Open Source PBX -=- http://www.asterisk.org/ |
| 13:19.57 | [TK]D-Fender | russellb : ! |
| 13:20.01 | fjean | brettnem: around 10k calls |
| 13:20.16 | [TK]D-Fender | russellb : Care to add the usual FreePBX referral notice back in? ;) |
| 13:20.50 | brettnem | fjean: 10k a day? |
| 13:20.52 | hgaillac | brettem: what's barf |
| 13:21.00 | fjean | bretnem - yes.. |
| 13:21.39 | brettnem | fjean: hmm.. then that doesn't sound like a configuration problem.. otherwise the number would be a lot higher |
| 13:22.11 | fjean | brettnem: so it could be a network problem , hey, I mean it's all over the internet so.. |
| 13:22.13 | *** join/#asterisk Ariel_ (n=Ariel@70.46.87.158) |
| 13:22.24 | brettnem | fjean: using any unusual equipment or applications? |
| 13:22.37 | fjean | brettnem: DeadAGI only |
| 13:23.17 | fjean | brettnem: i know it's suppose to be still working on a dead channel, so that might be part of the answer...or... |
| 13:23.34 | *** join/#asterisk lorinc (n=ang@caracas-0541.adsl.interware.hu) |
| 13:24.27 | fjean | brettnem: it's like the script is never returning, when I do ps -ef | grep asterisk I see them all |
| 13:24.53 | fjean | bretnem: plus the channels are still there doing sip show channels |
| 13:26.12 | *** join/#asterisk motu (n=motu@192.165.166.183) |
| 13:27.25 | *** join/#asterisk tuxd00d (n=tuxinato@netblock-68-183-112-100.dslextreme.com) |
| 13:29.12 | *** join/#asterisk wunderkin (i=kev@ip68-226-113-228.ph.ph.cox.net) |
| 13:31.43 | dlynes_office | anyways |
| 13:32.08 | dlynes_office | i think i'm gonna head home and get about a 2 hour power nap before i need to go downtown |
| 13:32.18 | dlynes_office | good night, peeps |
| 13:32.25 | hgaillac | Nobody use netmeeting or gnomeeting !! |
| 13:32.51 | Nivex | H.323 is so last century |
| 13:33.51 | tzanger | ok maybe a dumb question, but how do I get the "other side" of the call in the dialplan? I.e. Zap/5 or IAX2/user@peer/9003 ? |
| 13:34.09 | tzanger | CHANNEL(channeltype) will give me Zap or IAX2 |
| 13:34.21 | tzanger | CDR(src) maybe but that seems wrong |
| 13:34.25 | xachen | I still use H.323 :( |
| 13:35.35 | sremington | Anybody having call parking problems? Other side hears no MOH... but when I pick them up they *START* to hear MOH! |
| 13:35.43 | *** join/#asterisk trelane_ (n=trelane@pdpc/supporter/sustaining/trelane) |
| 13:36.46 | tzanger | just ${CHANNEL} gives me the unique callid too, which is too much |
| 13:37.17 | kay2 | is there a way when I'm into a AGI and I queue() to know when the call has been answered ? |
| 13:38.37 | X-Gen | *yawn* |
| 13:39.42 | mercestes | *volunteers* |
| 13:39.44 | [TK]D-Fender | kay2 : It all in the queue log file |
| 13:40.01 | hgaillac | xachen can you help me to check my conf i need a sip/H323 gateway |
| 13:41.14 | *** join/#asterisk Cresl1n (n=matt@gateway.digium.com) |
| 13:41.42 | *** join/#asterisk zparta_ (i=zparta@ipv6.hispan.se) |
| 13:43.51 | *** join/#asterisk buleriando (n=chatzill@anda.demon.nl) |
| 13:44.46 | sevard | Where would one get the default factory config file in XML format that ships with the Sipura 2002 |
| 13:45.17 | *** join/#asterisk ad3 (n=ad3@walden.acns.colostate.edu) |
| 13:45.28 | *** join/#asterisk iulius (n=iulius@mail1.technologieshq.com) |
| 13:45.56 | *** join/#asterisk tuxd00d (n=tuxinato@netblock-68-183-112-100.dslextreme.com) |
| 13:46.38 | *** join/#asterisk obiwanmikenolte (n=obiwanmi@mail.efc-intl.com) |
| 13:47.51 | Hmmhesays | hook me up with some blow |
| 13:48.17 | Hmmhesays | tzanger so chop the unique id off |
| 13:48.26 | MrChimpy | will asterisk for pot? |
| 13:48.54 | lunk | only one of those is legal in the states :( |
| 13:48.59 | lunk | stupid puritan settlers |
| 13:49.02 | Hmmhesays | i don't know blow was marijuana |
| 13:49.26 | buleriando | hi, anyone have any tips about provisioning a gxp-2000 via HTTP? The same config file that works via TFTP is not being accepted via HTTP |
| 13:50.06 | tzanger | Hmmhesays: yeah but the problem is that it is messy |
| 13:50.07 | *** join/#asterisk anthm (n=anthmct@CPE-69-76-83-52.wi.res.rr.com) |
| 13:50.07 | *** mode/#asterisk [+o anthm] by ChanServ |
| 13:50.15 | MrChimpy | lunk: like bill hicks said, they're making out like god made a mistake, leaving all this weed around |
| 13:50.19 | Hmmhesays | tzanger: why? |
| 13:50.25 | tzanger | I think I'll make a patch for the function that gives you the uniqueid and extension |
| 13:50.28 | tzanger | er not extension |
| 13:50.33 | tzanger | Hmmhesays: because what if the sip peer has '-' in it? |
| 13:50.50 | Hmmhesays | why would you do something so silly |
| 13:50.53 | tzanger | i.e. ${CHANNEL}=SIP/foo-bar/580-14 |
| 13:50.57 | *** join/#asterisk dwclarkNU (n=dwclark@129.10.215.24) |
| 13:51.00 | tzanger | Hmmhesays: because Asterisk allows such nuttiness |
| 13:51.18 | tzanger | blow is marijuana?? |
| 13:51.19 | Hmmhesays | the guberment allows you to own a gun, doesn't mean you go around shooting people |
| 13:51.25 | tzanger | which poor neighbourhood do you live in? |
| 13:51.25 | Hmmhesays | no its not |
| 13:51.32 | tzanger | yes I know it's not |
| 13:51.49 | Hmmhesays | blow is a fantastic movie starring that loveable johnny depp |
| 13:52.13 | obiwanmikenolte | And Pee Wee Herman |
| 13:52.55 | Hmmhesays | woohoo |
| 13:53.08 | fjean | just by curiosity, mission impossible just came out at the movie theaters in brazil, like a week or os, it out since a long while already in the states ? |
| 13:53.26 | MrChimpy | blow is cocaine in the US. |
| 13:53.32 | fjean | we get them late |
| 13:53.44 | Hmmhesays | look at imdb |
| 13:53.52 | MrChimpy | not so much in UK, generally means weed. depends who you speak to. |
| 13:53.57 | Hmmhesays | why you looking for a crappy telesync copy? |
| 13:53.59 | obiwanmikenolte | Yeah, dunno, but you can keep MI:III |
| 13:54.03 | obiwanmikenolte | IIIIIIIIIIIIIIIIIIIIIIII |
| 13:55.59 | *** join/#asterisk C4T3l (n=rcall01@216.54.143.2) |
| 13:56.18 | tzanger | yeah it wasn't all that great |
| 13:56.35 | tzanger | although the exploding capsules were pretty cool |
| 13:57.17 | C4T3l | sup mercestes! |
| 13:57.23 | Hmmhesays | haven't seen, i usually drink instead of going to movies |
| 13:59.19 | mut | none of the mission impossibles were really that good |
| 13:59.52 | mut | shoulda been a B movie |
| 14:00.44 | sevard | Does anyone provision sipura ATAs via TFTP? |
| 14:01.00 | mut | i thought about it once |
| 14:01.04 | mut | then decided not to |
| 14:01.07 | fjean | hehe |
| 14:01.11 | *** join/#asterisk frawd (n=francois@62-15-215-21.inversas.jazztel.es) |
| 14:01.21 | frawd | hello all |
| 14:01.23 | sevard | mut: .. |
| 14:01.51 | fjean | but that last one wasn't that bad, hehe |
| 14:02.25 | *** join/#asterisk miguel3239 (n=chatzill@ns1.nashuacs.com) |
| 14:02.30 | *** join/#asterisk jeebusroxors (n=jeebusro@29palms-cuda1-68-170-36-65.losaca.adelphia.net) |
| 14:04.00 | dwclarkNU | is there a way to use asterisk to fax a pdf file located on the server? |
| 14:04.05 | frawd | hi, i just spotted a weird problem: new sip hardphone (ALLNET) when connected to asterisk gave real choppy robotized sound when calling a GXP2000, and it got all perfect when I added the "tT" functionality in the Dial command (many many tests)... any clues?? :S:S |
| 14:04.37 | *** join/#asterisk mhnoyes (n=mhnoyes@dialup-4.246.236.206.Dial1.SanJose1.Level3.net) |
| 14:04.38 | frawd | (it also worked perfectly with direct IP calling) |
| 14:05.12 | frawd | does anyone know if that "tT" option modifies or buffers the RTP stream or something? |
| 14:05.22 | *** part/#asterisk mhnoyes (n=mhnoyes@dialup-4.246.236.206.Dial1.SanJose1.Level3.net) |
| 14:06.10 | *** join/#asterisk jeebusmobile (n=jeebusmo@29palms-cuda1-68-170-36-65.losaca.adelphia.net) |
| 14:07.01 | *** join/#asterisk saftsack (n=saftsack@p54A7CCE4.dip.t-dialin.net) |
| 14:07.57 | hgaillac | No H323 user here !! |
| 14:08.56 | *** join/#asterisk tuxd00d (n=tuxinato@netblock-68-183-112-100.dslextreme.com) |
| 14:09.04 | sevard | holy shit, why is it so hard to get information on sipuras |
| 14:09.12 | buleriando | anyone provisioning grandstream gxp-2000 via HTTP? |
| 14:09.14 | websae | sevard: what do you mean? |
| 14:09.30 | robtro | sevard: ? |
| 14:10.11 | sevard | I'm trying to find example configs on the net and have googled all morning coming up with nothing |
| 14:10.42 | sevard | I found a provider's cfg file but apparently there's a 'compiler tool' which takes the plaintext xml and sisterfucks it into a strange binary |
| 14:11.05 | Nivex | sisterfucks? That's a new one on me. |
| 14:11.30 | sevard | I'm full of `em :) |
| 14:13.03 | Hmmhesays | british telecom needs to die |
| 14:13.21 | sevard | DIE |
| 14:13.41 | obiwanmikenolte | Yeah, just who do they think they are, differing from the US? |
| 14:14.08 | obiwanmikenolte | With their variable-numbered schemes and their ISDNonsense |
| 14:14.33 | *** join/#asterisk darkskiez (n=darkskie@194.247.78.146) |
| 14:14.44 | Hmmhesays | obiwanmikenolte: um.. BT varies from the entire world |
| 14:14.47 | C4T3l | sevard: didnt the sisterfuck hack originate in texas? |
| 14:14.52 | Hmmhesays | dass2 is uniquely british |
| 14:15.51 | sevard | C4T3l: it orginated at Hmmhesays's place |
| 14:16.07 | C4T3l | sevard: ohh! |
| 14:16.13 | Hmmhesays | yeah sevards sister |
| 14:16.24 | sevard | That's what we told Hmmhesays anyway |
| 14:16.35 | Hmmhesays | hot little thing, but she was into whips n sh1@t |
| 14:16.45 | obiwanmikenolte | No wonder he's taken to drinking |
| 14:16.56 | C4T3l | lol |
| 14:16.59 | sevard | heh |
| 14:17.03 | obiwanmikenolte | And making me look dumb. Pffft. Like I need help |
| 14:17.12 | Hmmhesays | that and when we got undressed it was a big ol' mess, sheena was a man |
| 14:17.18 | sevard | Hmmhesays: give me the sipura 'provider only' tools |
| 14:17.33 | Hmmhesays | i'll give you a handful of stfu |
| 14:17.53 | sevard | :'( baby don'tcha luv me no more |
| 14:18.06 | *** join/#asterisk BadPacket (n=root@unaffiliated/badpacket) |
| 14:18.27 | Hmmhesays | i was using you for cotton candy and episodes of stargate |
| 14:18.42 | Hmmhesays | get over it |
| 14:18.43 | sevard | i don't like either of those |
| 14:19.14 | Hmmhesays | i'm par2'ing an episode of stargate as we speak |
| 14:19.24 | Ahrimanes | mmm par2 |
| 14:19.53 | hgaillac | bye |
| 14:20.09 | fjean | can I kill a channel cleanly using kill ? |
| 14:20.32 | sevard | par2? oolllld school shit |
| 14:21.28 | Hmmhesays | par2 is not oldschool |
| 14:21.29 | Hmmhesays | par is |
| 14:22.02 | *** join/#asterisk paolob (n=donpaolo@pri-214-b7.codetel.net.do) |
| 14:22.55 | Hmmhesays | god you're such a nub |
| 14:24.13 | sevard | anything usenet related is oldsk00l |
| 14:26.48 | C4T3l | the best way to kill a channel is to exit the CLI and type "rm -rf /" |
| 14:27.05 | SplasPood | Whats the current best way of testing what will become 1.4 ? |
| 14:27.10 | C4T3l | OUCH! |
| 14:27.20 | *** join/#asterisk ManxPower (n=ewieling@dpc67142183150.direcpc.com) |
| 14:27.32 | fjean | c4t3i...mm, they are still there..remember it's in memory ;-) |
| 14:27.55 | *** join/#asterisk tuxd00d (n=tuxinato@netblock-68-183-112-100.dslextreme.com) |
| 14:29.03 | C4T3l | ahh crap! |
| 14:31.35 | boddy | hii all when should I use fxo_ks signaling or pri_cpe signaling |
| 14:31.44 | boddy | what is the differences |
| 14:33.28 | *** join/#asterisk trbldwine (n=trbldwin@adam.ur.northwestern.edu) |
| 14:33.56 | *** join/#asterisk unixgeek (n=unixgeek@216-220-234-197.exploremaine.com) |
| 14:34.38 | *** join/#asterisk UlbabraB (n=salama@host-84-222-47-43.cust-adsl.tiscali.it) |
| 14:34.46 | ManxPower | ~fxofxs |
| 14:34.48 | jbot | fxofxs is probably An FXO port expects to receive dialtone and receive ring voltage, you plug a phone LINE into this. An FXS port expects to provide dialtone and provide ring voltage, you plug a PHONE into this. |
| 14:34.49 | *** join/#asterisk visba (n=dca[lapt@sta-208-139-193-163.rockynet.com) |
| 14:35.06 | *** join/#asterisk Money5ack (i=moneysac@wer.will.spontanficken.de) |
| 14:35.08 | *** part/#asterisk VoicePulse (n=contact@unaffiliated/voicepulse) |
| 14:35.41 | brettnem | yay, that again |
| 14:36.05 | *** join/#asterisk VoicePulse (n=contact@unaffiliated/voicepulse) |
| 14:36.14 | *** part/#asterisk UlbabraB (n=salama@host-84-222-47-43.cust-adsl.tiscali.it) |
| 14:38.33 | *** join/#asterisk PoWeRKiLL (n=PoWeRKiL@195.167.202.197) |
| 14:38.54 | websae | nothing like a good polycom :) |
| 14:40.07 | brettnem | grr polycom |
| 14:40.31 | websae | brettnem: what do you like? |
| 14:40.40 | brettnem | polycom |
| 14:40.46 | brettnem | it's a love hate relationship |
| 14:41.03 | brettnem | I just hate those conf files.. |
| 14:41.29 | brettnem | I actually kinda like the SNOM phones.. easy to configure.. predictable |
| 14:44.11 | PoWeRKiLL | how can I dial a group of sip user ? |
| 14:44.18 | *** join/#asterisk chaoscon (n=ph33r@smartserv/ceo/chaoscon) |
| 14:45.51 | frawd | hi, anyone knows a bit how channels are handled in asterisk? |
| 14:46.09 | file | what a generic question... try asking a more detailed one |
| 14:46.10 | frawd | (especialy chan_features) |
| 14:46.18 | file | that channel driver is not used |
| 14:46.23 | frawd | oh |
| 14:46.36 | brettnem | heh |
| 14:46.43 | frawd | i just spotted a weird problem: new sip hardphone (ALLNET) when connected to asterisk gave real choppy robotized sound when calling a GXP2000, and it got all perfect when I added the "tT" functionality in the Dial command (many many tests)... any clues?? :S:S |
| 14:47.01 | frawd | isn't "tT" in the chan_features stuff? |
| 14:47.02 | *** join/#asterisk tuxd00d (n=tuxinato@netblock-68-183-112-100.dslextreme.com) |
| 14:47.12 | file | no. |
| 14:47.25 | brettnem | frawd: tT will require audio to pass through asterisk.. maybe there is a network problem passing audio direct.. <shrug> |
| 14:47.41 | frawd | nop, direct IP calling works very well |
| 14:48.24 | frawd | i'm on a LAN, but it appears that bridging with "tT" is different than simple bridging (passing through asterisk) |
| 14:48.52 | frawd | i just want to know the difference in order to track the problem |
| 14:48.54 | brettnem | you got it backwards |
| 14:49.23 | brettnem | tT passes it through asterisk... without it and canreinvite=yes it BYpasses asterisk |
| 14:49.32 | *** join/#asterisk UlbabraB (n=caplaz@host-84-222-47-43.cust-adsl.tiscali.it) |
| 14:49.37 | frawd | canreinvite is set to no |
| 14:49.56 | brettnem | hm |
| 14:50.06 | frawd | the audio passes through asterisk in both cases, but with "tT" it eliminates my choppy-sound problem |
| 14:50.08 | brettnem | well you did say "grandstream"... anything is possible. |
| 14:50.23 | brettnem | btw, when is t AND T ever appropriate?! |
| 14:50.39 | frawd | it's not the point, i don't need t NOR T |
| 14:50.42 | jake1932 | if you have a remote wanting to xfer |
| 14:50.46 | file | fight fight fight |
| 14:50.48 | *** part/#asterisk kmilitzer (n=km@office-gw.westend.com) |
| 14:50.53 | brettnem | I'm going to call your call center and then transfer myself to china! |
| 14:50.58 | brettnem | haha |
| 14:51.08 | frawd | but the fact is that if the stream passes through asterisk, it gives choppy sound unless tT option is here |
| 14:51.17 | frawd | :) |
| 14:51.20 | jake1932 | ie - i'm out on a cell and get a call - want to xfer to the desk phone |
| 14:52.08 | frawd | so noone could give me a hint on where the problem could be? |
| 14:52.13 | frawd | :'( |
| 14:52.35 | *** join/#asterisk [Airwolf] (n=airwolf@83.98.235.219) |
| 14:52.37 | brettnem | maybe in your grandstream phone? |
| 14:53.11 | frawd | i don't think, does the grandstream make a difference if the audio stream passes by asterisk or not? |
| 14:53.18 | *** join/#asterisk holaaa (n=holaaa@85.137.82.173) |
| 14:53.21 | holaaa | reload chan_zap.o and I get a lot of "ignoring signalling".. "ignoring hangonpolarityswithch".. in /var/log/asterisk/full... its like it just can't recognize half ot the directives in zapata.conf |
| 14:53.52 | frawd | holaaa: do a restart instead of a reload |
| 14:54.21 | dwclarkNU | frawd: configure the grandstream to use something besides *, that'll determine whether its the adapter or the server |
| 14:54.22 | frawd | holaaa: reload ignores some directives |
| 14:54.28 | holaaa | frawd... so it is recognizing the directives but after a reboot? |
| 14:55.40 | frawd | holaaa: for some type of directives it seems to be able to change them only after a reboot |
| 14:55.51 | holaaa | frawd you mean a restar of the service zaptel or the whole machine? |
| 14:56.12 | frawd | holaaa: restart gracefully on asterisk |
| 14:56.18 | brettnem | reboot your power grid |
| 14:56.20 | brettnem | ;) |
| 14:56.37 | *** join/#asterisk miguel3239 (n=chatzill@ns1.nashuacs.com) |
| 14:57.08 | holaaa | ok frawd... so there is nothing wrong with my *. Thank you. |
| 14:57.46 | frawd | dwclarkNU, what do you mean by "use something besides *"? I tried direct IP calling between the 2 phones and it works perfectly |
| 14:58.43 | *** part/#asterisk jake1932 (n=Administ@68.236.22.143) |
| 14:58.45 | dwclarkNU | frawd: register the phones to another VoIP provider and call through them |
| 14:58.47 | frawd | passing through asterisk still gives me choppy sound in the allnet phone and only in reception (from gxp2000 to allnet phone)... unless i add tT to the Dial command...... |
| 14:59.39 | frawd | i tried and it also works (just like direct IP), so i really think it might come from asterisk |
| 14:59.52 | Juggie | frawd, turn on silence supression on the phones. |
| 14:59.55 | Juggie | er, turn off |
| 14:59.56 | Juggie | i mean. |
| 14:59.59 | frawd | can it be a jitter problem? (sorry, still a bit newb) |
| 15:00.06 | frawd | that's the first thing i did |
| 15:00.08 | frawd | :) |
| 15:00.23 | dwclarkNU | frawd: what codec's do you have turned on/off? |
| 15:00.28 | dwclarkNU | and the order |
| 15:00.30 | frawd | (i also tried turning it on, just in case, but doesn't work) |
| 15:00.58 | frawd | i tried ALAW and ULAW (disabling the others) in any order |
| 15:01.24 | frawd | i tried with transcoding (one in alaw and the other in ulaw) and without (everything in A or ULAW) |
| 15:01.46 | frawd | :).... |
| 15:01.48 | *** join/#asterisk coppice (n=chatzill@74.192.0.218.broad.hz.zj.dynamic.cndata.com) |
| 15:01.51 | frawd | i'm in a LAN |
| 15:02.18 | dwclarkNU | frawd: tT is used to transfer calls, i dont see why or how it would effect the audio |
| 15:02.19 | frawd | same IP range and everything... |
| 15:02.22 | brettnem | I wonder if it's alaw talking to ulaw.. broken device.. |
| 15:02.58 | brettnem | I'd set the call up without your "tT" and do a sip show channel on both sides and see what's going on. compare that to using your "tT" |
| 15:03.01 | frawd | neither do I !!!!! but the fact is, it works only like that... and having to dig that far to make it work... |
| 15:03.01 | VoicePulse | Notice to VoicePulse Connect for Asterisk users -- If you are using gwiax-in-01, gwiaxt01, or gwiaxt02 for origination/termination via IAX2, those servers are now scheduled to be decommissioned on June 5th. Please begin moving your traffic to the newer connect01 and connect02 servers which will handle incoming and outgoing for both SIP and IAX2. |
| 15:03.29 | brettnem | VoicePulse: fabulous.. how convienent! |
| 15:03.41 | brettnem | I'm so glad you are making US do YOUR migration!! |
| 15:03.43 | frawd | brettnem: i did that also and it spitts out exactly the same on sip show channel |
| 15:03.44 | file | VoicePulse: you just know people are just not going to do that |
| 15:03.54 | fjean | hey guys, anybody using rtpkeepalive or rtptimeout in a production server ? |
| 15:04.00 | frawd | brettnem: which is quite logical... |
| 15:04.25 | *** join/#asterisk af_ (n=af@ip-143-220.sn1.eutelia.it) |
| 15:04.29 | ghenry | what's the * voice saying when you dial VoiceMailMain? Medium Mail? |
| 15:04.42 | ghenry | I can' tmake out what she's saying |
| 15:04.46 | r_evolution | i always thought it said Comedian Mail |
| 15:04.50 | file | Comedian Mail |
| 15:04.51 | r_evolution | like haha funny joke |
| 15:04.54 | VoicePulse | file: I'm sure they'll have no problem doing it considering most calls through the new servers cost 1¢/min while the old ones are 2.4¢/min. |
| 15:04.59 | *** join/#asterisk eKo1 (n=bernd@190.4.7.90) |
| 15:05.04 | file | it's a play... |
| 15:05.15 | eKo1 | Is there a way to set the ${HANGUPCAUSE} variable? |
| 15:05.21 | frawd | i was wondering if the "tT" options activated some other parallel channel for asterisk to get DTMF and thus it could have modified/buffered/corrected the RTP stream or something, and improving sound quality |
| 15:05.25 | brettnem | ahh.. bribes! |
| 15:05.28 | brettnem | ;) |
| 15:05.41 | frawd | but i really don't have a clue... maybe i'm speaking nonsense |
| 15:05.57 | file | Meridian Mail, Comedian Mail, get it? |
| 15:06.06 | websae | VoicePulse: I liked you keyword...."most" calls through the new servers cost 1cent |
| 15:06.26 | brettnem | wabsae: Like, I "mostly" call cheap places |
| 15:06.42 | dwclarkNU | frawd: configure your network on a hub or if you have a manged switch set it so you cna monitor the traffic on a port, use ethereal to compare the SIP packets between direct IP dialing and going through asterisk |
| 15:06.55 | ghenry | I thought it said Comedian Mail |
| 15:07.01 | ghenry | I dont get it :-( |
| 15:07.06 | ghenry | PBX in-joke? |
| 15:07.14 | websae | are you kidding me, I call bumble*uck all the time....what are you talking about |
| 15:07.15 | dwclarkNU | the SIP packets will configure the session and tell each device what codecs to use, etc... |
| 15:07.21 | VoicePulse | websae: Considering VoicePulse now offers an Asterisk macro to look up our rates in realtime, users will have no problem sending us calls where we're cheaping and sending the rest through you. :) |
| 15:07.29 | VoicePulse | *cheaper |
| 15:07.32 | coppice | Its a play on how nortel's systems say "meridian mail. mailbox?" |
| 15:08.07 | websae | I don't even think they have a CO where I call, I think they just have some switch that converts the old can on one end -- string --- another can into some sort of PSTN gateway |
| 15:08.21 | hwt | i think voicemailmain is way too complicated. |
| 15:08.32 | hwt | it least it should be possible to scale it down to something simpler. |
| 15:08.37 | hwt | who needs folders? |
| 15:08.40 | frawd | dwclarkNU, i've tried with sip debug ip 192.168.10.X (without anymore hint), isn't that enough to get SIP configuration? |
| 15:08.59 | websae | VoicePulse: who is your backbone? |
| 15:09.00 | motu | hw do i uninstall zaptel? there doesnt seem to be a uninstall target in the makefile |
| 15:09.33 | websae | *primary* backbone I mean .... L3? Gx? PW? PT? |
| 15:09.33 | [TK]D-Fender | coppice : Yeah well we really should drop that... sounds too "kiddy" and hurts credibility when its treated as such |
| 15:09.33 | hwt | motu: look at the install target and do the excact opposite? |
| 15:09.37 | frawd | dwclarkNU, in fact, the sip debug output is exactly the same... i thing the problem come from the RTP stream, but i don't know where to look |
| 15:09.41 | *** join/#asterisk ravenpi (n=chatzill@host-64-65-199-187.man.choiceone.net) |
| 15:09.41 | BadPacket | VoicePulse: what do I have to run on my side to look up the rates in real time? |
| 15:09.48 | dwclarkNU | frawd: i'm not sure, i dont use sip debug for debugging, i usually use ethereal....what does it output? |
| 15:10.11 | motu | hwt: im a bit too newbie for that |
| 15:10.14 | frawd | dwclarkNU, one second |
| 15:10.29 | Nivex | coppice: Is there an alternate sound file available, ie something like "Welcome to the voice mail system." |
| 15:10.30 | dwclarkNU | frawd: as far as i know, the rtp stream is setup during the sip authentication process |
| 15:10.41 | hwt | Nivex: make your own |
| 15:10.56 | dwclarkNU | frawd: and from there, the devices just blast packets using that configuration |
| 15:10.58 | hwt | we recorded the whole soundeset (in Norwegian) in a studio today. |
| 15:11.01 | Nivex | hwt: but then the voice is inconsistent. I am asking if there is an Allison recording to that effect. |
| 15:11.13 | coppice | [TK]D-Fender: "we" don't have it. digium does. any reasonable person would replace it with a custom message. its surprising how many meridians still use the default message, though. |
| 15:11.17 | frawd | dwclarkNU, but it doesn't seem to be setup any differently, it's just during the call that one way is choppy and the other perfect |
| 15:11.21 | hwt | Nivex: she makes custom prompts. |
| 15:11.36 | [TK]D-Fender | coppice : YeahI suppose... maybe we should draft Allison for a nice generic one :) |
| 15:11.37 | *** join/#asterisk denon (i=denon@synapse.subneural.net) |
| 15:11.37 | *** mode/#asterisk [+o denon] by ChanServ |
| 15:11.46 | Nivex | So I can infer the answer to my question is "At present, no." |
| 15:11.56 | hwt | Nivex: |
| 15:12.04 | hwt | Nivex: www.thevoice.com |
| 15:12.07 | dwclarkNU | frawd: how can you monitor the packets using sip debug in asterisk when you do direct ip dialing? |
| 15:12.07 | VoicePulse | BadPacket: You can cut and paste the macro which is found in the Account Center. It is completely open and uses built-in Asterisk functionality. To implement it in your extensions.conf, you add a few simple lines: http://pastebin.ca/56240 |
| 15:12.15 | Nivex | hwt: yes, I am familiar with the fact that she does custom prompts. |
| 15:12.16 | hwt | Nivex: www.theirvoice.com* |
| 15:12.27 | hwt | Nivex: *sigh* IVR. |
| 15:12.33 | *** join/#asterisk duckz (n=duckz@193.192.47.26) |
| 15:12.34 | BadPacket | hmm |
| 15:13.00 | hwt | Nivex: not to my knowledge, there is. |
| 15:13.41 | *** join/#asterisk tuxd00d (n=tuxinato@netblock-68-183-112-99.dslextreme.com) |
| 15:13.44 | frawd | dwclarkNU, i was just saying that i compared with or without the "tT" option and the output is the same |
| 15:13.56 | motu | i installed asterisk from svn, but many applications are lacking, why is that? |
| 15:14.08 | motu | e.g. cut, setvar |
| 15:14.16 | frawd | dwclarkNU, but without the option it doesn't work well, and just adding it makes all sound perfect in both ways |
| 15:14.55 | dwclarkNU | frawd: ohh, ok, mis understanding....can you show me a sample dump from sip debug so i can see what it looks like? |
| 15:14.58 | *** join/#asterisk trelane_ (n=trelane@pdpc/supporter/sustaining/trelane) |
| 15:15.10 | dwclarkNU | frawd: i'd do it on mine, but its at the office and i'm not there right now |
| 15:17.02 | BadPacket | holy crap, it works |
| 15:17.25 | dwclarkNU | frawd: nvm, i was able to vpn in and get into it |
| 15:19.12 | BadPacket | VoicePulse: is this just for US? does it give international rates in realtime too? |
| 15:19.12 | frawd | dwclarkNU, did you get it? |
| 15:19.12 | frawd | dwclarkNU, i just sent you the output, basicly it's showing what the negociation looks like, or the inside of SIP packets |
| 15:20.04 | dwclarkNU | frawd: yea, i got it |
| 15:20.11 | dwclarkNU | frawd: let me look over it |
| 15:20.29 | BadPacket | never mind, I see it does |
| 15:21.11 | dwclarkNU | frawd: did you see the "Capabilites:" line? |
| 15:21.19 | *** join/#asterisk klasstek (n=nunyobiz@ip67-154-143-188.z143-154-67.customer.algx.net) |
| 15:21.29 | frawd | dwclarkNU, i didn't send you the complete whole output but anyways... |
| 15:21.32 | frawd | yes i did |
| 15:21.42 | dwclarkNU | frawd: you support gsm, ulaw, and alaw while the peer only support alaw |
| 15:21.59 | frawd | dwclarkNU, that's right |
| 15:22.08 | dwclarkNU | frawd: if i'm not mistaking, ulaw is used the us |
| 15:22.35 | frawd | dwclarkNU, i'm in europe, but still... they negociate in Alaw and it works |
| 15:23.08 | frawd | dwclarkNU, i had added ulaw to do some more tests doing transcoding (setting up the phone in ulaw) to see if it wasn't a codec problem |
| 15:23.21 | frawd | dwclarkNU, but same problem disabling ulaw in asterisk |
| 15:23.22 | dwclarkNU | frawd: i would set a disable all in teh sip.conf and then only enable alaw |
| 15:23.32 | *** join/#asterisk Ironhand (i=arjen@meek.xs4all.nl) |
| 15:23.44 | frawd | dwclarkNU, i tried that also |
| 15:23.56 | dwclarkNU | frawd: dont just do a reload, actually stop and start the server |
| 15:24.12 | frawd | dwclarkNU, i always did that (restart gracefully) |
| 15:24.22 | frawd | dwclarkNU, i don't trust reload :) |
| 15:24.38 | dwclarkNU | frawd: and this only occurs with one grandstream adapter? |
| 15:25.10 | frawd | this occurs in the ALLNET phone, calling a GXP2000 (the ALLNET gets the choppy sound) |
| 15:25.29 | frawd | oh, also when calling a GStream HandyTone |
| 15:25.53 | dwclarkNU | frawd: does the gxp2000 and gstream work if you call them from each other? |
| 15:25.58 | frawd | but works well with softphones and BT100 phones |
| 15:26.07 | frawd | i don't have anything else to test with |
| 15:26.24 | frawd | dwclarkNU, yes they do, perfectly |
| 15:26.56 | frawd | dwclarkNU, it's the first time i had this kind of sound-quality problem... in fact i cannot even understand the other person speaking |
| 15:27.37 | dwclarkNU | frawd: i have to run in a few minutes, but i would try and look at the packets using ethereal and look for something there, the sip debug did a good job of displaying the information, but ethereal will shop the sip and the rtp |
| 15:27.38 | *** join/#asterisk yxa (i=lonari@cm121.gamma228.maxonline.com.sg) |
| 15:27.47 | dwclarkNU | frawd: i'll check back in on your issue when i get back |
| 15:28.10 | frawd | dwclarkNU, okay, i'll try that thanks |
| 15:28.27 | dwclarkNU | frawd: focus your search on the packets the allnet phone sends out during its authentication process |
| 15:28.28 | *** join/#asterisk dsfr (n=dsfr@pdpc/sponsor/digium/dsfr) |
| 15:28.45 | ManxPower | ~thebook |
| 15:28.46 | jbot | thebook is probably somebody said thebook was Asterisk: The Future of Telephony, released under a Creative Commons license and available at http://www.asteriskdocs.org << Meet Jim Van Meggelen at Cluecon http://www.cluecon.com |
| 15:29.43 | websae | yay for Cluecon :)! |
| 15:29.43 | *** join/#asterisk moprilo (n=jjohn@201.192.107.57) |
| 15:29.44 | VoicePulse | BadPacket: Yes, it gives all of our rates in realtime, US and international. |
| 15:30.33 | moprilo | hi, i have a _9. in my context, and when i try to put 9113 (cause i'm calling 113, 9 is for outside), my asterisk calls 911 ! :S, how can i change that?, there's no 911 in my context |
| 15:31.14 | *** join/#asterisk tuxinator_linux (n=tuxinato@netblock-68-183-112-99.dslextreme.com) |
| 15:31.17 | Nivex | moprilo: sounds like you need ${EXTEN:1} where the :1 strips the beginning 9 |
| 15:32.57 | moprilo | but the thing is, it works for all numbers that don't start with 1, like if try 92244444, it works, but 9113 will call 911 |
| 15:32.57 | moprilo | :S |
| 15:33.30 | moprilo | i had the 911 thing before, but i had taken it out already and restarted asterisk |
| 15:33.38 | moprilo | is like it does reload the context :S |
| 15:33.41 | ManxPower | moprilo, What device are you dialing from? |
| 15:34.32 | BadPacket | VoicePulse: yes, it works. thanks. I might actually start using voicepulse for outbound calls now ;) |
| 15:34.57 | moprilo | i'm calling from sip, into zap |
| 15:35.14 | moprilo | i did restart gracefully + reload on the asterisk console |
| 15:39.20 | *** join/#asterisk SplasPood (n=jwb@206.252.198.101) |
| 15:39.52 | PoWeRKiLL | I got this error May 16 10:47:53 WARNING[13409]: chan_zap.c:8392 pri_dchannel: Ring requested on channel 0/1 already in use on span 1. Hanging up owner. |
| 15:39.54 | rpm | has anyone had any good experiences with asterisk on freebsd with a te410p card? |
| 15:39.56 | mercestes | ${EXTEN:1} instead of ${EXTEN} maybe, moprilo?? |
| 15:40.27 | PoWeRKiLL | it's happens after some agent answer to the queue |
| 15:40.58 | moprilo | i have ${EXTEN:1}.. :S. i think is something wrong with asterisk restart |
| 15:41.30 | VoicePulse | BadPacket: Glad to hear you got it working so quickly! |
| 15:42.10 | Hmmhesays | omg i missed last nights episode of prison break |
| 15:42.17 | BadPacket | websae: you do it too? |
| 15:42.42 | moprilo | is there a way to extend the asterisk timeout for numbers.. in _9. |
| 15:43.30 | [TK]D-Fender | moprilo : Its not an Asrterisk timeout, but rather dependant on the kind of equipment you are using |
| 15:43.54 | r_evolution | why are you requiring a 9 to be dialed anyway? why not just make it dependant on the number of digits sent? |
| 15:44.02 | r_evolution | coming in late on this one :) |
| 15:44.06 | jsaunders | I have a odd problem. In zapata.conf I have "#include zapata_additional.conf" which holds all my 'station' contexts, followed by [channels] which is group=1 and context=from-trunk. How can I have the 'from-trunk' context use the caller id inbound from my tdm2400, instead of using the last 'station' which has caller id implicitly set (callerid=308), which is apparently 'bleeding over' from the last entry in '#include zapata_add |
| 15:44.06 | jsaunders | itional.conf'?? |
| 15:44.38 | [TK]D-Fender | jsaunders : that would be an AMP/FreePBX problem... |
| 15:44.40 | [TK]D-Fender | ~amp |
| 15:44.42 | jbot | hmm... amp is "amp is, like, NOT supported here! people using it should join #freepbx (the new name of amp)" |
| 15:44.50 | jsaunders | Why is it an amp problem? |
| 15:44.55 | jsaunders | I'm interested in hearing your reasoning. |
| 15:45.29 | jsaunders | Anyone can do an #include, it's not specific to AMP. |
| 15:46.05 | *** join/#asterisk nortex (n=nortex@ama-wldhcp.696130103.amaonline.com) |
| 15:46.15 | jsaunders | I know your ~amp trigger happy and all but... this is perfectly relevant to #asterisk. |
| 15:46.18 | *** join/#asterisk dwclarkNU (i=dwclarkN@155.33.196.139) |
| 15:46.29 | dwclarkNU | frawd: any luck? |
| 15:46.32 | jsaunders | jsaunders != n00b; |
| 15:46.33 | mercestes | But it's not asterisk...it's AMP. |
| 15:46.41 | jsaunders | wtf are you talking about. |
| 15:46.58 | jsaunders | I'm using asterisk, w/ zap chans, and zapata.conf and I have a line in there that says #include. Whoop di doo. |
| 15:47.03 | jsaunders | Anyone can do that, with or without amp. |
| 15:47.11 | moprilo | r_evolution: because, it could 7 or it could more if its outside. |
| 15:47.22 | jsaunders | It's not amp that makes the settings global all the way down in zapata.conf, it's asterisk. |
| 15:47.27 | [TK]D-Fender | jsaunders : AMP makes your config files every time you commit changes and would OBLITERATE anything we try to suggest for you anyways. Thats why. |
| 15:47.52 | jsaunders | Not w/ zapata.conf it doesn't. |
| 15:48.15 | nortex | AMP allows changes to the _custom files. |
| 15:48.51 | *** join/#asterisk Assid (n=assid@203.115.83.214) |
| 15:50.00 | nortex | I have a server dimensioning question, should I base the number of servers I need on the # of users or on the # of concurrent calls? |
| 15:50.14 | Assid | nortex: concurrent calls |
| 15:51.05 | [TK]D-Fender | nortex : POssibly a little of each, but more on concurrency. |
| 15:51.25 | tamp4x | ~amp |
| 15:51.27 | jbot | it has been said that amp is "amp is, like, NOT supported here! people using it should join #freepbx (the new name of amp)" |
| 15:51.37 | tamp4x | ~jbot |
| 15:51.39 | jbot | jbot is, like, only marginally useful at best, He got a C- on his Turing Test |
| 15:51.47 | tamp4x | ~? |
| 15:51.49 | nortex | So one server would be fine handling a single PRI and 30 max calls if it were say a dual 2.8Ghz and 2 GB ram |
| 15:52.03 | wunderkin | omg |
| 15:52.05 | tamp4x | yes nortex |
| 15:52.11 | wunderkin | and what? |
| 15:52.51 | *** join/#asterisk Greek-Boy (n=grb@193.220.93.162) |
| 15:53.00 | nortex | thanks, I was panicing a little after reading in TFOT about 100 users needing multiple boxes. |
| 15:53.07 | Greek-Boy | I'm buying cisco phones and I've just heard that i need a lience for them? |
| 15:53.16 | [TK]D-Fender | nortex : you could get away with a single processor system with half the ram just eassy. Depends on codec use, etc.... |
| 15:53.18 | tamp4x | tfot? |
| 15:53.32 | Hmmhesays | users is really not relevant, number of concurrent calls and what tech they are using is of importance |
| 15:53.59 | [TK]D-Fender | Hmmhesays : Not even necessarily.... if its all SIP with no transcoding you can pass a massive # of calls... |
| 15:54.10 | nortex | The future of Telophany, TOFT, my own version :) |
| 15:54.20 | Hmmhesays | like I said, number of concurrent calls and what tech they are using |
| 15:54.37 | Hmmhesays | ex: sip calls v. zap calls |
| 15:55.19 | Hmmhesays | anyone got a giganews account they want to give me? |
| 15:55.27 | [TK]D-Fender | Hmmhesays : Even SIP > ZAP is no big deal if the codec matches the PSTN format ULAW/T1, ALAW/E1 |
| 15:55.39 | Greek-Boy | so do i need a lience for cisco phones or not? |
| 15:56.06 | nortex | According to Cisco, yes |
| 15:56.17 | Hmmhesays | that doesn't invalidate my statement |
| 15:56.22 | sevard | Hmmhesays: lemme access your news server |
| 15:56.31 | Greek-Boy | nortex, is it enforced? |
| 15:56.36 | Hmmhesays | stfu sevard i pay $14.95 a month for this |
| 15:56.45 | sevard | you pay for warez |
| 15:56.48 | sevard | noob. |
| 15:57.18 | Hmmhesays | i'd rather pay $14.95 a month for all the tv I want with no commercials, and the $45 a month for shi@tty cable |
| 15:57.19 | *** join/#asterisk Katty (n=angela@64.82.232.54) |
| 15:57.22 | *** join/#asterisk salviadud (n=ralfalfa@201.138.132.204) |
| 15:57.23 | sevard | my tummy box feels broken |
| 15:57.25 | jsaunders | Here's a simple one that gives you no chance of saying '~amp'. :P Shall I use 'callerid=asreceived'? This line in 'http://www.voip-info.org/wiki-Asterisk+config+zapata.conf' concerns me... "Important Note: Caller ID can only be transmitted to the public phone network with supported hardware, such as a PRI. It is not possible to set external caller ID on analog lines.". The reason I am not testing myself is system is live an |
| 15:57.25 | jsaunders | d zap changes require a restart as far as I can tell. |
| 15:57.42 | jsaunders | oops, 'transmitted'. |
| 15:57.44 | jsaunders | Just noticed that. |
| 15:57.46 | jsaunders | My bad. |
| 15:57.55 | sevard | YOUR BAD, DOG. |
| 15:57.59 | jsaunders | Regardless.... anyone ever used callerid=asreceived? :) |
| 15:58.01 | jsaunders | heheh |
| 15:58.07 | nortex | Greek-boy, I ask that same question of our Cisco rep, and pointed out the free download off Cisco's site of the current SIP firmware. I could not get a straight answer. |
| 15:58.14 | trelane_ | jsaunders, not unless I was using AMP |
| 15:58.35 | jsaunders | Heheheh |
| 15:58.37 | jsaunders | Touche' |
| 15:59.13 | *** join/#asterisk mercestes (n=merceste@69.15.174.114) |
| 15:59.30 | Greek-Boy | nortex but have u actually setup cisco phones without a licence? |
| 15:59.47 | *** join/#asterisk awe6 (n=lba@user-12lml5g.cable.mindspring.com) |
| 16:00.18 | ravenpi | I'm trying to set up a transfer-direct-to-voicemail extension; I have to assume it's been done before -- anyone have an example? |
| 16:00.40 | docelmo | ya |
| 16:01.06 | docelmo | exten => _*XXX,1,Voicemail(${EXTEN:1}) |
| 16:01.09 | docelmo | something like that |
| 16:01.27 | ravenpi | *laughs* That'll work. I was trying to make it far more difficult than it needs to be, in hindsight... |
| 16:02.09 | jsaunders | Or he/she could Dial() the actual voicemail extension. |
| 16:02.09 | nortex | Greek-Boy, I have 1 that was a test phone. There is no license file, so to answer you it is possible if you get the current firmware to run the phone, but you are probably in violation of something. |
| 16:02.42 | dwclarkNU | is there a way in * to call the voicemail and be prompted for the mailbox (being able to hit # to enter the default)? |
| 16:02.55 | ravenpi | Well, I'm actually trying to make it be such that a call to the switchboard can be transferred from the switchboard directly to someone's VM. But I think a variation on docelmo's line might do the trick... |
| 16:05.56 | sevard | ravenpi: K.I.S.S. |
| 16:06.39 | nortex | ? |
| 16:06.43 | Hmmhesays | i love it when customers call and ask about some obscure software they found on google |
| 16:07.05 | C4T3l | customers suck! |
| 16:07.22 | salviadud | what do you tell em hmmm? |
| 16:07.25 | Hmmhesays | end user support sucks |
| 16:07.31 | nortex | dwclarckNU, Ive used the application VoiceMailMain() to force a prompt of mailbox. |
| 16:07.46 | Hmmhesays | salviadud: that they should research the software further because I am not familiar with it |
| 16:07.58 | C4T3l | good one ;) |
| 16:08.15 | sevard | Hmmhesays: did you say "... did you google furthure and say... research" |
| 16:08.17 | salviadud | Hmmhesays, good answer |
| 16:08.26 | *** join/#asterisk tuxinator_linux (n=tuxinato@netblock-68-183-112-99.dslextreme.com) |
| 16:08.40 | *** part/#asterisk dwclarkNU (i=dwclarkN@155.33.196.139) |
| 16:09.11 | salviadud | i got a question |
| 16:09.17 | salviadud | if i got an x100p card |
| 16:09.23 | *** join/#asterisk chaoscon (n=ph33r@smartserv/ceo/chaoscon) |
| 16:09.27 | salviadud | could i send and receive? |
| 16:09.37 | salviadud | even if it's only an fxo port |
| 16:09.39 | sevard | i heard those cards suck balls |
| 16:09.46 | salviadud | yes they do |
| 16:09.49 | salviadud | x100penis |
| 16:09.59 | salviadud | that's how we call em |
| 16:10.04 | Hmmhesays | send and receive from a pstn line or equivalent |
| 16:10.07 | *** join/#asterisk IMG-SD (n=IMG-SD@ismex.imperialgroup.ca) |
| 16:10.20 | Hmmhesays | i don't know what i'd do without man parts |
| 16:10.38 | salviadud | we're all men here right? |
| 16:10.49 | salviadud | why don't women bother with asterisk? |
| 16:10.57 | jsaunders | oooh, oooh, good setup... *waits to see the response* |
| 16:10.59 | Hmmhesays | the occasional female wanders in here |
| 16:11.37 | salviadud | in my job, my bosses are like a unix shell, i do things right, they wont say a thing |
| 16:12.02 | salviadud | but if something is not right Command not found |
| 16:12.08 | jsaunders | amen |
| 16:15.40 | Hmmhesays | yeah but they can suck it |
| 16:16.18 | salviadud | Hmmhesays. who sucks who? |
| 16:16.26 | Hmmhesays | whiny bosses |
| 16:16.36 | salviadud | yeah. hehe |
| 16:16.39 | Hmmhesays | ooh someone uploaded season 3 of stargate |
| 16:16.40 | Hmmhesays | rock |
| 16:16.56 | salviadud | i got pansat, don't need to download... |
| 16:17.13 | salviadud | i like this crazy prog on the SciFi chan, called, Ghost Hunters |
| 16:17.25 | Hmmhesays | I got a bunch of those episodes |
| 16:17.34 | salviadud | they got all this high tech equipment, infrared, emp thingies |
| 16:17.37 | [TK]D-Fender | Hmmhesays : I've gota friend with way toomuch bandwidth and even more time to kill. He has everything. |
| 16:17.46 | moprilo | i have 2 asterisk boxes.. same conf. one with TDM24xx and another with the tdmxx, the one with the simple tdmxx (4 fxos), is passing through the tones, (like when i'm in a menu of a company digital recepcionist), but the one with the TDM24xx is not passing the tones, anyone have any idea why this is ? |
| 16:17.54 | salviadud | what do you mean everything? |
| 16:18.16 | Hmmhesays | its nice to have a massive library of tv shows to veg for an hour here and there |
| 16:20.05 | jsaunders | Stargate you say. I'm an avid Trekkie... Should I give Stargate a go? |
| 16:20.50 | Hmmhesays | yeah |
| 16:20.54 | Hmmhesays | completely different they are |
| 16:21.01 | *** join/#asterisk surfdue (n=tyler@unaffiliated/surfdue) |
| 16:21.03 | surfdue | hey |
| 16:21.09 | Hmmhesays | i'm up to season 3 episode 12 right now |
| 16:21.14 | surfdue | can youmake asterisk ring a skype contact? |
| 16:21.15 | surfdue | :P |
| 16:21.32 | Assid | surfdue: if the person has skype in |
| 16:21.33 | Assid | yes |
| 16:21.41 | surfdue | oh |
| 16:21.44 | surfdue | without skype in |
| 16:21.44 | surfdue | :P |
| 16:21.47 | jsaunders | There is some discussion on trickery regarding that very question. |
| 16:22.01 | Hmmhesays | yes you can |
| 16:22.07 | iDunno | surfdue: if skype used anything but a proprietary shitty in house evil protocol... |
| 16:22.08 | surfdue | how? |
| 16:22.15 | Hmmhesays | if you have a windoze machine handy |
| 16:22.19 | surfdue | i do |
| 16:22.20 | surfdue | lol |
| 16:22.26 | surfdue | sadly :( |
| 16:22.40 | Hmmhesays | paypaly me a $20 |
| 16:22.44 | jsaunders | heheh. |
| 16:22.47 | Hmmhesays | and I'll tell you how |
| 16:22.48 | surfdue | oh my gosh, i have to put taht online called micrsoft with skype out for free and they were helping me with ubuntu! lol |
| 16:22.48 | jsaunders | Go Hmmhesays. :) |
| 16:22.57 | surfdue | i only have cc :( |
| 16:23.15 | Hmmhesays | liar |
| 16:23.19 | websae | paypa takes cc |
| 16:23.23 | Assid | surfdue: you can make a payment with cc |
| 16:23.28 | surfdue | how? |
| 16:23.35 | surfdue | with registering an acocunt yes.. |
| 16:23.37 | surfdue | lol |
| 16:23.50 | moprilo | nevermind me,.. jej, it was the dtmfmode |
| 16:23.51 | moprilo | :P |
| 16:23.53 | Hmmhesays | if you skype hmmhesays it rings my asterisk box |
| 16:24.02 | *** join/#asterisk dsfr (n=dsfr@pdpc/sponsor/digium/dsfr) |
| 16:24.13 | surfdue | thats not what i mean |
| 16:24.18 | surfdue | i mean asterisk to skype |
| 16:24.18 | surfdue | :P |
| 16:24.28 | Hmmhesays | yes I can skype you from my asterisk box |
| 16:24.28 | Katty | Hmmhesays: ooh. |
| 16:24.30 | Katty | Hmmhesays: it does? |
| 16:24.38 | surfdue | oh well |
| 16:24.42 | surfdue | tell me when you release for free |
| 16:24.42 | surfdue | :P |
| 16:24.46 | Katty | Hmmhesays: i wanna know how to do that. |
| 16:24.49 | Hmmhesays | yeah last time I checked, i've futzed with it bit |
| 16:25.03 | Hmmhesays | someone skype hmmhesays |
| 16:25.14 | Hmmhesays | i might have played with the routing |
| 16:25.30 | Hmmhesays | hmm yes yes i have |
| 16:25.34 | Hmmhesays | hang up surfdue |
| 16:25.41 | surfdue | lol |
| 16:25.45 | Katty | surfdue: you're calling accounting. |
| 16:25.53 | surfdue | hmm? |
| 16:26.04 | Katty | surfdue: instead of Hmmhesays |
| 16:26.13 | surfdue | what |
| 16:26.15 | Katty | or so i claim. |
| 16:26.16 | surfdue | i just called hmmhesays.. |
| 16:26.20 | Hmmhesays | hehe |
| 16:26.32 | Katty | surfdue: that's what you think. |
| 16:26.44 | surfdue | nah |
| 16:26.48 | surfdue | it was skypes simulated ring |
| 16:27.12 | Hmmhesays | i'll fix the routing on that |
| 16:27.13 | Katty | file: you put that back down. |
| 16:27.20 | file | Katty: noes! do I have to? :( |
| 16:27.20 | Katty | file: also, come have lunch with me. |
| 16:27.31 | surfdue | lol |
| 16:27.31 | file | I can't :( |
| 16:27.34 | Katty | file: i 'spose not. |
| 16:27.34 | surfdue | i would |
| 16:27.36 | Katty | file: :< |
| 16:27.40 | Katty | file: well /fine/ hotshot. |
| 16:27.41 | *** join/#asterisk tuxinator_linux (n=tuxinato@netblock-68-183-112-99.dslextreme.com) |
| 16:28.00 | Katty | no one will go to lunch with me. sniffle. |
| 16:28.19 | *** part/#asterisk surfdue (n=tyler@unaffiliated/surfdue) |
| 16:29.44 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 16:31.08 | *** join/#asterisk froguz (n=alvaro@pc-95-155-104-200.cm.vtr.net) |
| 16:31.10 | Katty | twisted[asteria]: i demand you drive out here and go to lunch with me. |
| 16:31.38 | froguz | what is exactly overlapdial option in zapata.conf for? |
| 16:31.45 | Katty | froguz: hi. |
| 16:31.50 | justinu | for overlap dialing |
| 16:31.57 | Katty | hey justinu (= |
| 16:31.58 | froguz | hi Katty |
| 16:31.58 | salviadud | hide your pr0n, Katty's here! |
| 16:32.14 | Katty | salviadud: that's right! |
| 16:32.34 | Dr-Linux | ~Katty |
| 16:32.37 | jbot | extra, extra, read all about it, katty is the only girl in the channel, so be nice to her |
| 16:32.41 | justinu | froguz, if you're not sure what it is, it's probably irrelevant to you |
| 16:32.53 | froguz | hey! i didn't know we have a lady here |
| 16:32.57 | justinu | yeah, so be nice |
| 16:33.01 | Katty | what a weirdo. |
| 16:33.53 | froguz | justinu, i have to do a lot of testing in a PRI line, and i don't understand what is that parameter (and others) for |
| 16:34.32 | justinu | overlap dialing is digit at a time |
| 16:34.49 | justinu | opposite of enbloc, which is sending all the digits in the q931 SETUP message |
| 16:35.25 | ManxPower | People in countries with fixed length dialing don't need overlap dialing |
| 16:35.36 | ManxPower | i.e. USA, Canada |
| 16:35.47 | froguz | thanks justinu, that should be in the documentation |
| 16:36.23 | *** join/#asterisk FlyboySR22 (n=rsears@gateway.americanis.net) |
| 16:36.28 | *** join/#asterisk wunderkin (i=kev@69.26.192.234) |
| 16:36.41 | justinu | a lot of stuff isn't in the docs |
| 16:37.18 | justinu | assumed knowledge, i guess |
| 16:37.32 | twisted[asteria] | Katty, fyi, IRC is usually in the background on my screen ;) |
| 16:38.03 | _Sam-- | must get annoying when you have real work to do to have IRC scrolling as your background |
| 16:38.33 | justinu | sam! |
| 16:38.49 | _Sam-- | afternoon! how goes it? i been in hiding...busy season aroiund here. |
| 16:38.50 | justinu | ltns, how's it going? |
| 16:38.57 | justinu | yeah - summer is upon us |
| 16:39.06 | justinu | all the squids are out in force |
| 16:39.18 | _Sam-- | things here are great...busier than ever, and i had a minority shareholder of my business that i bought out...things couldnt be too much better! |
| 16:39.24 | _Sam-- | hows things out there? |
| 16:39.24 | justinu | nice |
| 16:39.35 | *** join/#asterisk Lino` (n=Lino@i577BD937.versanet.de) |
| 16:39.40 | justinu | slower than I'd hope |
| 16:39.43 | *** join/#asterisk Ironhand (i=arjen@meek.xs4all.nl) |
| 16:39.45 | justinu | but that was expected |
| 16:40.05 | justinu | what's the latest word on GXP2000 firmwares? :) |
| 16:40.21 | _Sam-- | you would know alot better than me, but there is some new alpha version out that that is working nice. |
| 16:40.35 | _Sam-- | im still 100% gxp at my cycle business -- maybe 22 phones there now |
| 16:40.45 | justinu | what are you running? |
| 16:40.58 | SheriF_WorK | [TK]D-Fender: u don't sleep ? |
| 16:41.08 | *** join/#asterisk MattB2 (n=MattB2@mail.tricycleinc.com) |
| 16:41.13 | _Sam-- | 1.0.2.8 on most user phones, and the new alpha stuff on the computer geeks |
| 16:41.13 | Hmmhesays | sleep is for small children |
| 16:41.17 | justinu | ahh |
| 16:41.22 | MattB2 | hi all |
| 16:41.31 | nortex | MattB2, hi |
| 16:41.32 | [TK]D-Fender | SheriF_WorK : I'm at work... they frown on my sleeping here :( |
| 16:41.42 | _Sam-- | for me, having my ISP be my CLEC is what really turned around the phone performance.... |
| 16:41.59 | justinu | so you got all that worked out then |
| 16:42.05 | MattB2 | not strictly an asterisk question, but hoping someone could help - i'm looking for a provider in the US to provide a single DID that can take man y simulatneous calls (ie up to 100). VoxBone are great but their 2 call limit is prohibitive. Any recommendations? |
| 16:42.15 | [TK]D-Fender | Having my work become my ITSP worked great for me :) |
| 16:42.19 | _Sam-- | as worked out as it will get...they want to charge too much for all the minutes i use...so i use them for certain calls. |
| 16:42.23 | MattB2 | DID -> SIP/IAX I mean |
| 16:42.23 | r_evolution | sleep is for people who don't do drugs ;x |
| 16:42.34 | SheriF_WorK | [TK]D-Fender: i'm at work too 10 hours :D and sleepy and working in things not my work :( was just telling x86 about it. |
| 16:42.35 | _Sam-- | i route nearly all of our outgoing calls through them |
| 16:42.47 | _Sam-- | and most incoming (all actually) comes from teliax, and has been great for a few months now |
| 16:42.52 | ManxPower | r_evolution, I thought that was reality is for people that can't handle drugs |
| 16:43.06 | r_evolution | that too! |
| 16:43.10 | [TK]D-Fender | ManxPower : NO, reality is for people who can't handle Star Trek. |
| 16:43.53 | _Sam-- | [TK]D-Fender : if your work is your ITSP, then your calls are only as good as their connection to PSTN....that was my whole problem...getting calls to PSTN. |
| 16:44.16 | _Sam-- | and now my ISP who is on the other side of my 5ms latency T1 does the calls to PSTN right at their side (their side is in the CO) |
| 16:44.43 | [TK]D-Fender | _Sam-- : Well my work is running * on PRI with good EC and everything... |
| 16:45.06 | _Sam-- | fair enough! (my work is my ITSP too, when im at home) :) |
| 16:45.36 | [TK]D-Fender | Broungh my phone bill from $40-50/month down to 5$ (dry-DSL base fee) |
| 16:45.48 | IMG-SD | There's mention on bugs.digium.com that Asterisk 1.2.8 is out... any idea where I can find it? |
| 16:46.30 | _Sam-- | justinu: why do you think things slowed down for you? |
| 16:46.39 | *** join/#asterisk dwclarkNU (i=dwclarkN@155.33.196.139) |
| 16:46.44 | *** join/#asterisk Nugget (i=nugget@dazed.slacker.com) |
| 16:46.47 | justinu | just the powers that be |
| 16:46.51 | justinu | out of my control |
| 16:47.05 | justinu | i only do technology, other people run the business... :/ |
| 16:47.18 | justinu | actually it's picking up, so the trend is good |
| 16:47.37 | _Sam-- | how many people are in your business there? |
| 16:47.55 | nortex | IMG-SD, which bug? |
| 16:48.10 | justinu | still pretty small... under 50 |
| 16:49.25 | *** join/#asterisk bmg505 (n=leon@c1-128-8.rndf.isadsl.co.za) |
| 16:49.28 | _Sam-- | and that business focuses mainly on pbx setups and installs? |
| 16:49.33 | unixgeek | Any PRI gurus out there today? |
| 16:49.42 | justinu | no, no |
| 16:49.58 | justinu | pbx setups is a hobby/sidejob for me |
| 16:50.01 | unixgeek | Well, I have a TE110P that is driving me crazy. |
| 16:50.12 | [TK]D-Fender | unixgeek : just ask |
| 16:50.16 | _Sam-- | i see...got it. how are the polycoms working out at the real estate office? |
| 16:50.19 | r_evolution | funny... I have a reality that's driving me crazy |
| 16:50.23 | justinu | polycom == good |
| 16:50.25 | justinu | they're happy |
| 16:50.25 | [TK]D-Fender | unixgeek : and try #therapy as well |
| 16:50.29 | unixgeek | I can not get it to switch to externally clocked source for the life of me. |
| 16:50.37 | _Sam-- | what type of tech work do you do for your real day job? phone related? |
| 16:50.50 | justinu | yeah, pure VoIP |
| 16:50.56 | justinu | all SIP |
| 16:50.57 | [TK]D-Fender | unixgeek : pastebin a "cat /proc/interrupts", your zaptel & zapata as well |
| 16:50.58 | [TK]D-Fender | ~pb |
| 16:51.20 | jbot | pb is, like, a place to paste your stuff without flooding the channel - try http://pastebin.com/, or http://pastebin.ca |
| 16:51.20 | unixgeek | Anyone see any of the digium card ignore the timing parameter in an span definition before? |
| 16:51.25 | *** join/#asterisk florz (n=florz@2001:1a50:503c:0:0:0:0:1) |
| 16:51.46 | dwclarkNU | is there anyway for asterisk to log the number of minutes used on each sip line? |
| 16:52.00 | r_evolution | hey dw... isn't that called CDR :) |
| 16:52.21 | dwclarkNU | CDR? |
| 16:52.43 | _Sam-- | call detail records |
| 16:52.47 | unixgeek | http://pastebin.com/720879 (/proc/interrupts) |
| 16:52.52 | *** join/#asterisk [Airwolf] (n=airwolf@83.98.235.219) |
| 16:53.06 | _Sam-- | if you log them to an SQL database, you can use a program like this to report on your CDRs: http://www.areski.net/asterisk-stat-v2/about.php |
| 16:53.11 | r_evolution | try using mysql to store the CDR on a database that's accessible from a web-page |
| 16:53.37 | justinu | unixgeek: are you using zttool to check the timing source? |
| 16:54.03 | unixgeek | justinu: yes. each time I go into zttool, it says "internally clocked" |
| 16:54.04 | dwclarkNU | _Sam--: thanx |
| 16:54.25 | _Sam-- | dwclarkNU : see /etc/asterisk/cdr_* |
| 16:54.33 | justinu | unixgeek: are there any messages in syslog that confirm or contradict that? |
| 16:55.21 | unixgeek | http://pastebin.com/720885 (/etc/zaptel.conf) |
| 16:55.58 | unixgeek | justinu: I have not really seen any message in the messages file that confirms or contradicts the timing setting. |
| 16:57.29 | justinu | T1 carrier is up? (no alarms?) |
| 16:57.55 | unixgeek | justinu: No it is in constant yellow alarm. |
| 16:57.58 | [TK]D-Fender | unixgeek : Ok, nothing I see in there tells me anything as to why.... |
| 16:58.15 | justinu | ok, then that's why you have no ext clock ref |
| 16:58.21 | MattB2 | anyone used GAFACHI before? |
| 16:58.29 | justinu | yellow alarm means that the far end cannot see your TX signal. |
| 16:58.30 | tzanger | MattB2: gezundheit! |
| 16:58.37 | MattB2 | thankyou ;) |
| 16:58.38 | ManxPower | huh? the span=1,1 tells Zap to get it's timing from the T-1. |
| 16:58.46 | ManxPower | I assume the T-1 is to/from the telco and not a PBX? |
| 16:58.54 | *** join/#asterisk MRH2 (n=Mr_happy@fcirc-adsl.demon.co.uk) |
| 16:59.13 | ManxPower | unixgeek, it's a KNOWN bug that zttool does not correctly show the timing source. |
| 16:59.18 | froguz | i'm a little confused about international, national, local, private and unknow prefix parameters (zapata.conf) |
| 16:59.34 | ManxPower | froguz, leave them all out unless you have problems dialing |
| 16:59.39 | MRH2 | anyone else getting modprobe.conf being overwritten on install? |
| 16:59.55 | ManxPower | MRH2, you mean on Fedora Core? |
| 17:00.36 | froguz | ManxPower, are those for sending the correct ANI to the called party? |
| 17:00.47 | MRH2 | yep fc3, from an update maybe a week ago |
| 17:00.48 | froguz | correct=complete |
| 17:00.55 | ManxPower | froguz, no. |
| 17:01.19 | ManxPower | if you leave them out then they will be "unknown" |
| 17:01.47 | *** join/#asterisk tuxinator_linuxM (n=tuxinato@netblock-68-183-112-99.dslextreme.com) |
| 17:02.55 | ManxPower | and "unknown" means "telco, don't fuck with my dialed digits" |
| 17:03.12 | unixgeek | Sorry, damn interruptions! |
| 17:03.22 | ManxPower | MRH2, /join #asterisk-dev and type "M7150" |
| 17:03.50 | unixgeek | ManxPower: so how does one insure that the card is accepting external clocking? |
| 17:04.11 | justinu | fix the yellow alarm first. |
| 17:04.20 | ManxPower | unixgeek, the second parameter to the span= line will force it, even if it can't get timing. |
| 17:04.42 | froguz | ManxPower, it's ok, it's just i would like to know what am i doing rather than just leave it untouched just because i don't know what are they for |
| 17:04.48 | unixgeek | justinu: I have been focusing on the yellow alarm as being a symptom of a timing mismatch. is this wrong? |
| 17:04.57 | justinu | yes, wrong. |
| 17:05.09 | froguz | besides, i'm connecting directly to a telco E1 switch |
| 17:05.14 | froguz | if for a centrex test |
| 17:05.16 | ManxPower | froguz, pridialplan=national means you will not be able to dial local or international calls. |
| 17:05.17 | justinu | (09:58:33) justinu: yellow alarm means that the far end cannot see your TX signal. |
| 17:05.40 | *** join/#asterisk syle (n=blah@unaffiliated/syle) |
| 17:05.47 | *** join/#asterisk dwclarkNU2 (i=dwclarkN@155.33.196.236) |
| 17:06.09 | unixgeek | justinu: OK. I will be working with the telco guys again tomorrow on this circuit. BTW it is a tie line to a Nortel switch. |
| 17:06.23 | unixgeek | justinu: any thoughts as to anything to be warry of? |
| 17:06.37 | ManxPower | unixgeek, nortel PBX? |
| 17:06.59 | MRH2 | thanx manx |
| 17:07.06 | ManxPower | or a telco switch |
| 17:07.07 | unixgeek | ManxPower: Thanks for the info on zttool and that span forces it. |
| 17:07.10 | froguz | ManxPower, i'm not talking about pridialplan, i'm talking about nationalprefix, international prefix, localprefix, and so on.. |
| 17:07.20 | justinu | unixgeek: knowing the signalling method would help... PRI, or CAS? |
| 17:07.21 | unixgeek | ManxPower: nortel PBX. |
| 17:07.51 | unixgeek | justinu: we have been setting up as a PRI. Not familure with what a CAS is. |
| 17:07.54 | ManxPower | unixgeek, PBXs normally expect to get their timing from the switch. i.e. Asterisk in this case. |
| 17:08.08 | ManxPower | or the pri_net device |
| 17:08.12 | justinu | PRI is better, so don't worry |
| 17:08.20 | ManxPower | froguz, those options do not apply in the USA/Canada |
| 17:08.35 | froguz | but i'm in Chile |
| 17:08.43 | ManxPower | froguz, those options tell asterisk to force a call to national if the leading digits are whatever that prefix is. |
| 17:08.52 | ManxPower | usually "0" for E-1 |
| 17:09.19 | ManxPower | unixgeek, CAS is usually Channelized Voice T-1 (NOT PRI) |
| 17:09.36 | *** join/#asterisk keyhack (n=keyhack@68.236.93.254) |
| 17:10.05 | froguz | thanks |
| 17:10.26 | ManxPower | most of this info should be in zapata.conf.sample and the Wiki and The Book |
| 17:11.06 | froguz | yep, but is not too clear as you explained to me |
| 17:11.18 | tzanger | ManxPower: usually? |
| 17:11.26 | tzanger | when is a PRI able to be CAS T1? |
| 17:11.39 | ManxPower | tzanger, There are some REALLY fucked up telcos out there. |
| 17:11.59 | justinu | yeah, you can have some CAS timeslots, and some timeslots slaved to a D channel |
| 17:12.02 | *** join/#asterisk supjigatr (n=syslod@152.53.16.10) |
| 17:12.03 | tzanger | how do you get common channel signaling in a per-channel signalled environment? |
| 17:12.04 | froguz | also my english is not too good, so i sometimes need to read the same thing written in differents ways |
| 17:12.05 | ManxPower | tzanger, I can imagine a T-1 without CAS |
| 17:12.06 | justinu | on the same T1 |
| 17:12.13 | tzanger | yes I can too |
| 17:12.15 | brettnem | well the technologies have widely acceptable ways of being used, but don't assume that's the only way |
| 17:12.17 | tzanger | raw T1 |
| 17:12.24 | unixgeek | ManxPower: we have been setting the * box up to use pri_cpe signalling. (still waiting for the zapata.conf file to post to pastebin) |
| 17:12.36 | brettnem | how about a NFAS PRI ground with a Dchan in a different T1 |
| 17:12.42 | brettnem | s/ground/group/ |
| 17:12.48 | ManxPower | unixgeek, You have to use whatever the other end is not using, but STILL, that would not cause a yellow alarm |
| 17:13.33 | brettnem | i like yellow alarms |
| 17:13.47 | ManxPower | A PBX usually assumes it will be PRI CPE. |
| 17:13.59 | ManxPower | But that is not always the case and depends on how you configure the port on the PBX |
| 17:14.15 | tzanger | brettnem: that's still not a CAS T1 then |
| 17:14.19 | brettnem | just because the pbx manufacturers assume that you arn't; the phone company |
| 17:14.19 | *** join/#asterisk MacDome (n=eseidel@c-71-198-177-144.hsd1.ca.comcast.net) |
| 17:14.29 | tzanger | that particular T1 has no signaling internal to itself (common channel or per-channel) |
| 17:14.30 | justinu | right, that would be a clear channel T1 |
| 17:14.34 | *** join/#asterisk ToyMan (n=stuq@74-32-76-147.dsl1.mdl.ny.frontiernet.net) |
| 17:14.35 | brettnem | tzanger: right |
| 17:14.50 | tzanger | but that's still not a channel-associated signalled PRI |
| 17:15.05 | brettnem | tzanger: in re-reading the above messages, I'm not sure what my point was.. haha |
| 17:15.12 | *** join/#asterisk MacDome (n=eseidel@c-71-198-177-144.hsd1.ca.comcast.net) |
| 17:15.40 | brettnem | anyway, pri is just a way to use a T1 |
| 17:15.49 | unixgeek | ManxPower: OK. I have to ask then what are the typical causes of a yellow alarm? To my knowledge, it seems that it is either timing or lots of errors on the line. |
| 17:15.56 | brettnem | it doesn't have to be 23+1 |
| 17:16.20 | brettnem | unixgeek: yellow alarm on side A means that Side B is telling you it isn't getting anything from side A |
| 17:16.22 | supjigatr | unixgeek: You are receving a yellow alarm? |
| 17:16.40 | *** join/#asterisk spatulamaan (n=ggilmore@ip66-107-33-196.z33-107-66.customer.algx.net) |
| 17:16.42 | justinu | man... how many times does it have to be explained? |
| 17:16.48 | unixgeek | ManxPower: The telco guys have the circuit disabled on the switch so that it does not generate service tickets. Could this be causing the yellow alarm? |
| 17:16.52 | brettnem | T1 interfaces typicaly spit out yellow alarms when they arn't receivng a proper signal |
| 17:16.56 | *** part/#asterisk spatulamaan (n=ggilmore@ip66-107-33-196.z33-107-66.customer.algx.net) |
| 17:17.04 | ManxPower | unixgeek, Well DUG! |
| 17:17.10 | *** join/#asterisk abatista (n=Ariel@70.46.87.158) |
| 17:17.13 | ManxPower | DUH! too |
| 17:17.22 | robl^ | I want an OC12 hooked up in my apartment. the "new at&t" laughed at me. |
| 17:17.24 | brettnem | heh, well that should be AIS. I'd think.. but it could generate any number of alarm |
| 17:17.37 | ManxPower | unixgeek, do you have it set up as ESF/B8ZS or AMI/D4 ? |
| 17:17.40 | brettnem | robl^: why did they laugh? |
| 17:17.49 | supjigatr | robl^: They are running a special right now. $3856/month and $248 per mile. |
| 17:17.52 | *** join/#asterisk Ironhand (i=arjen@meek.xs4all.nl) |
| 17:17.57 | brettnem | haha.. ami/d4.. :) |
| 17:17.59 | unixgeek | brettnem: yes, to me that is more at the physical layer. Where as if the line was enabled on the switch, that would be the D channel signalling. |
| 17:18.17 | unixgeek | ManxPower: ESF/B8ZS |
| 17:18.22 | brettnem | unixgeek: yellow alarmshave nothing to do with d-channels |
| 17:18.38 | ManxPower | unixgeek, just making sure. |
| 17:18.39 | brettnem | why don't people call ami/d4 sf/d4? |
| 17:18.45 | unixgeek | brettnem: that is what I sort of thought. |
| 17:18.51 | justinu | sf == d4 |
| 17:18.53 | supjigatr | super freak / df |
| 17:18.56 | justinu | ami is a line coding, not a framing |
| 17:19.22 | ManxPower | brettnem, because all the manuals I have seen refer to it that way |
| 17:19.26 | brettnem | right.. .. so why do people say "b8zs/esf....d4/ami" it's silly |
| 17:19.33 | brettnem | ManxPower: right, they do.. but it's just plain silly |
| 17:19.38 | unixgeek | So even with the PRI disabled on the nortel switch I should not be getting a yellow alarm. |
| 17:19.39 | brettnem | like as if b8zs isn't AMI? |
| 17:19.40 | supjigatr | Cause Adtran prints it on the HTU-R cards. |
| 17:19.51 | justinu | b8zs isn't AMI |
| 17:19.51 | ManxPower | BTW, any ISPs or CLECs here that offer CHEAP T-1 internet service? /msg me. |
| 17:19.57 | brettnem | unixgeek: it's hard to tell what you get.. |
| 17:20.14 | supjigatr | unixgeek: If you have a DS1 signal you should not be getting alarm. regardless of the PRI status. |
| 17:20.23 | brettnem | justinu: B8ZS is most definately Alternate mark inversion.. except for the substitutions |
| 17:20.48 | justinu | generally if the span is administratively down you'd get something other than a yellow alarm... RED, or possibly blue |
| 17:20.51 | justinu | brettnem: touche |
| 17:20.57 | brettnem | ;) |
| 17:21.14 | *** part/#asterisk IMG-SD (n=IMG-SD@ismex.imperialgroup.ca) |
| 17:21.37 | brettnem | it's just always been a peeve.. b8zs/esf...d4/ami.. ;) but that's how they do it.. oh well whatever.. :P |
| 17:21.47 | ManxPower | BLUE alarm means "Call telco and say 'I have a blue alarm'" |
| 17:21.51 | unixgeek | supigator: OK. good that confirms what I thought should be the case. |
| 17:21.59 | justinu | blue means an all 1's condition |
| 17:22.11 | ManxPower | RED means "check your cables, if everything is good call the telco and say 'I have a red alarm'" |
| 17:22.12 | justinu | why all ones? back in the day of wet T1s, they wanted to keep the repeaters powered up. |
| 17:22.25 | brettnem | I thought blue was like a yellow ds3 |
| 17:22.47 | unixgeek | OK. finally my /etc/asterisk/zapata.conf got posted to pastebin. http://pastebin.com/720919 |
| 17:23.13 | brettnem | justinu: you can power a repeater with improper T1 timing! |
| 17:23.53 | brettnem | all ones wouldn't have anything to do with power |
| 17:24.38 | justinu | i thought the repeaters needed the 1s to stay powered up |
| 17:24.48 | supjigatr | unixgeek: do you have access to the smartjack? |
| 17:24.57 | justinu | but i've never worked on that style T1... only got them delivered via HDSL or DS3, or OC3/12 |
| 17:24.58 | brettnem | justinu: no the T1 line power is DC |
| 17:25.20 | brettnem | justinu: you mean, you've recevied DS1s ;) not T1 |
| 17:25.25 | unixgeek | supigator: This is actually a rollover cable into the nortel switch. |
| 17:25.29 | justinu | yeah |
| 17:25.38 | supjigatr | Do you have a loopback plug? |
| 17:25.57 | supjigatr | Hmm. Nortel what? Option 11? Meridian? |
| 17:26.00 | brettnem | justinu: T1, being an AMI signal is AC in nature (btw, if anyone wants to correct me here, I'm a bit weak here..) |
| 17:26.15 | unixgeek | supjigator: although the cable does go into a some sort of txport box that is hooked to the switch. |
| 17:26.17 | justinu | that makes sense |
| 17:26.36 | unixgeek | Nortel meridian. |
| 17:26.40 | brettnem | justinu: so the repeater power is DC.. which runs on the same cable, but does not "interfere" with the signal |
| 17:26.56 | supjigatr | unixgeek: try to loop both pieces of equipment and make sure you can get it to work first then try hooking them together. |
| 17:27.01 | supjigatr | U need a loopback coard. |
| 17:27.02 | unixgeek | supjigator: I did try a loop back plug the other day. Everything worked perfectly. |
| 17:27.02 | supjigatr | cord |
| 17:27.13 | supjigatr | On the nortel too? |
| 17:27.14 | *** part/#asterisk dwclarkNU2 (i=dwclarkN@155.33.196.236) |
| 17:27.15 | justinu | i thought the whole reason behind AMI was to keep the power flowing |
| 17:27.27 | unixgeek | NO, I did not try the nortel end. |
| 17:27.27 | Hmmhesays | i love it how "techs" can't figure out how to forward a port on a linksys |
| 17:27.28 | brettnem | justinu: there are lots of reasons for AMI |
| 17:27.29 | justinu | since theoretically you could have a long string of 0's on the circuit, which wouldn't provide power |
| 17:27.34 | supjigatr | I would try it. |
| 17:27.43 | brettnem | justinu: probably the most important is spectrum management |
| 17:27.52 | supjigatr | If you are getting a yellow it means the nortel is sending you an alarm. |
| 17:28.14 | brettnem | justinu: no, that has nothing to do with it... however all 0's (or all 1s for that matter) would cause a repeater to lose timing sync.. which is why b8zs exists |
| 17:28.26 | unixgeek | supjigator: OK. that will be something we will try tomorrow when the telco guys are around. |
| 17:28.31 | justinu | ok, so it's a sync issue, not a powe rissue |
| 17:28.42 | brettnem | ami, and b8zs, yes |
| 17:28.58 | brettnem | but it IS a "power spectrum" issue |
| 17:29.06 | *** join/#asterisk Ironhand (i=arjen@meek.xs4all.nl) |
| 17:29.17 | unixgeek | OK. I have to get off on a few other issues. I appreciate everyone's help with this. |
| 17:29.35 | *** join/#asterisk pjchilds (n=pjchilds@pdpc/supporter/student/pjchilds) |
| 17:29.36 | brettnem | justinu: I'm thinking that with AMI you half the required spectrum and you can travel twice the distance |
| 17:29.53 | brettnem | but, It's been years since I did transport engineering, I might be off |
| 17:30.34 | justinu | it's not gonna matter for much longer anyways |
| 17:30.49 | brettnem | what does that means? awfully foreboding |
| 17:30.57 | justinu | even tho transmission theory is interesting to me |
| 17:31.16 | justinu | T1 technology is over 50 years old, I think that its time is past. |
| 17:31.31 | brettnem | that's an interesting assessment |
| 17:31.33 | supjigatr | We still use garden hoses. |
| 17:31.39 | [TK]D-Fender | justinu : POTS is over 100 and still kicking... past doesn't stop much.... |
| 17:31.41 | *** join/#asterisk jpablo (n=jpablo@200.94.130.194) |
| 17:31.48 | brettnem | supjigatr: hold on to your hats, that's about to come to an end |
| 17:32.01 | ManxPower | justinu, Yeah, that's why all the telcos use SDSL now for the physical layer |
| 17:32.07 | brettnem | justinu: name a technology that can actually replace T1 |
| 17:32.31 | brettnem | the DSLs are just a spectrum management game, they can't do what T1 can do in distance. |
| 17:32.39 | brettnem | although some of the newer ones are interesting |
| 17:32.43 | jpablo | hey people. anyway i can identify incoming calls in the cdr but in a uniq way. right now i have a lot of records in the same calls, since people can make a lot of dials in my dialplan |
| 17:32.54 | ManxPower | brettnem, um, real T-1s need a repeater every 1,500 ft. |
| 17:33.08 | ManxPower | Hence the SDSL stuff the telcos have been using for a few years now |
| 17:33.44 | brettnem | ManxPower: but you CAN repeat the T1 signals |
| 17:34.02 | *** join/#asterisk gursikh (n=guriskh1@dsl254-123-245.nyc1.dsl.speakeasy.net) |
| 17:34.07 | ManxPower | shine a flashlight thru the smoky windows in your local smart jack. |
| 17:34.09 | brettnem | ManxPower: it's a good technology for probably 60% of a city maybe more |
| 17:34.12 | ManxPower | Then read the labels carefully |
| 17:34.15 | gursikh | hey, can anyone tell me if the polycom se-220 is a decent phone to put in a small conference room? |
| 17:34.42 | brettnem | ManxPower: right, a lot of that has moved to NGDLCs anyway.. but still.. t1s can be repeated |
| 17:36.06 | *** join/#asterisk shido6 (n=shido6@d221-68-200.commercial.cgocable.net) |
| 17:36.36 | *** join/#asterisk bjohnson (n=bjohnson@jecinc.tor.istop.com) |
| 17:36.57 | *** join/#asterisk meshuga (i=meshuga@71-212-107-44.tukw.qwest.net) |
| 17:37.05 | *** join/#asterisk tld (n=tld@343006.ds.nac.net) |
| 17:37.42 | tld | Interesting problem: I have a FastAGI script, that picks up a call, plays a prompt etc. Based on interaction with the user, I'd like the script to forward the call to another (possibly external) extension. |
| 17:37.50 | tld | Any advice on how to do that? |
| 17:37.52 | justinu | so a true T1 repeater has how many wires going into it? |
| 17:38.11 | brettnem | I believe.. 8 4 in, 4 out |
| 17:38.47 | justinu | so if there's DC power on the pairs, how do you encode a digital signal on that? |
| 17:39.48 | *** join/#asterisk xp_prg (n=wgheath@67-102-228-17.adsl.lbdsl.net) |
| 17:39.57 | xp_prg | is there a windows command line sip client? |
| 17:40.40 | xp_prg | I need to tell asterisk to call a phone # and play a pre-recorded message, anybody know the best way to do that? |
| 17:42.03 | justinu | http://www.voip-info.org/tiki-index.php?page=Asterisk+auto-dial+out |
| 17:43.18 | r_evolution | noooooooooooooooooooooooooooooooooooooooooooooooooooooooooooooooooooooooooooooooooooooooooooooooooo |
| 17:43.19 | r_evolution | etc |
| 17:45.33 | *** join/#asterisk oej (n=oej@apollo.webway.se) |
| 17:47.08 | *** join/#asterisk nortex (n=nortex@ama-wldhcp.696130103.amaonline.com) |
| 17:47.42 | brettnem | playback(tt-monkeys) |
| 17:52.01 | PoWeRKiLL | when trying to compile asterisk-addons from svn I got this error In file included from /usr/include/asterisk/logger.h:28, |
| 17:52.02 | PoWeRKiLL | <PROTECTED> |
| 17:52.02 | PoWeRKiLL | <PROTECTED> |
| 17:52.51 | PoWeRKiLL | then /usr/include/asterisk/compat.h:20: error: syntax error before "__extension__" |
| 17:54.46 | nortex | Anybody worked on Nortel BCM to Asterisk connections? |
| 17:57.37 | *** join/#asterisk feld (n=feld@66-190-56-100.dhcp.mdsn.wi.charter.com) |
| 17:58.05 | feld | can anyone give me a quick answer -- is it possible to have multiple asterisk servers on a WAN and have them forward to one another? just wondering... |
| 17:58.43 | nortex | feld, do you me route calls between them by forward? Is so yes. |
| 17:58.47 | *** join/#asterisk malverian (n=malveria@gentoo/developer/malverian) |
| 17:58.52 | *** part/#asterisk file (i=jcolp@216.237.114.82) |
| 17:58.56 | *** join/#asterisk file (i=jcolp@216.237.114.82) |
| 17:59.13 | mercestes | *squeals* I got the new polycom Firmware! *plays with polycoms* |
| 17:59.22 | brettnem | what firmware is that? |
| 17:59.30 | feld | nortex: yeah like i have my asterisk station with phones #6000-6001 and the rest of the network is pointing to a different asterisk server |
| 17:59.33 | MikeJ[Laptop] | brettnem, the new one ;) |
| 18:00.16 | mercestes | 1.6.6 |
| 18:00.21 | mercestes | The one with the budywatch. |
| 18:00.29 | mercestes | 48 buddy watches baby. |
| 18:01.25 | *** join/#asterisk carlos-the-man (n=carlos@201.155.235.25) |
| 18:01.57 | nortex | feld, Yes you can build trunks between them, I used IAX and gsm mainly over WAN links. |
| 18:02.03 | *** join/#asterisk papo (n=mathias@vpn-global-dhcp3-094.ethz.ch) |
| 18:02.05 | papo | Hi |
| 18:02.05 | arcy | i just upgrade from 1.0.x to 1.2.7 and asterisk failes to start. Log entry reads: "loader.c: /usr/lib/asterisk/modules/chan_sip.so: undefined symbol: ast_osp_terminate" |
| 18:02.30 | arcy | i confess to being a Gentoo user so... :( |
| 18:02.58 | papo | If I want to dial ${EXTEN}@ethz.ch using the peer I defined in [ekiga.net], how should my Dial() application look like? |
| 18:03.03 | *** part/#asterisk MattB2 (n=MattB2@mail.tricycleinc.com) |
| 18:03.13 | RaYmAn-Bx | There's nothing wrong with being a gentoo users. There's just something wrong with the majority of gentoo users |
| 18:03.24 | Qwell[] | mercestes: It isn't stuck at 7 anymore? |
| 18:03.37 | *** join/#asterisk ghento (n=ghento@CPE0014bfaf7d7a-CM00111ae4684c.cpe.net.cable.rogers.com) |
| 18:03.50 | arcy | indeed. But i was refering mostly to compiler issues common with Gentoo users. I don't have anything outrageous though |
| 18:04.09 | arcy | no --omit-instructions and such :p |
| 18:04.10 | Qwell[] | arcy: Did you use the package? |
| 18:04.28 | arcy | yes, i upgraded using the official portage package |
| 18:04.34 | Qwell[] | Don't do that |
| 18:04.51 | Qwell[] | unmerge it, and grab the source from asterisk.org |
| 18:05.10 | ghento | Hi folks. I'm trying to get festival working with asterisk, and it is always hanging at "Parsing '/etc/asterisk/festival.conf': Found" |
| 18:05.19 | *** join/#asterisk tainted_ (n=point@ppp-71-134-51-75.dsl.irvnca.pacbell.net) |
| 18:05.50 | arcy | Qwell, tnx. |
| 18:06.13 | *** join/#asterisk harryvv (n=none@S010600a0c93f6f7e.vs.shawcable.net) |
| 18:06.14 | justinu|laptop | anyone running res_snmp? |
| 18:06.50 | many | wtf |
| 18:06.55 | many | now its getting weird :P |
| 18:07.06 | brettnem | "it" |
| 18:07.32 | brettnem | wahoo buddy watch is fixed?! |
| 18:07.54 | many | Yea. |
| 18:07.56 | many | it. |
| 18:08.00 | xp_prg | Asterisk::AGI anyone understand how this perl module works? |
| 18:08.03 | mercestes | Yup |
| 18:08.08 | justinu|laptop | that's what fender says, brettnem |
| 18:08.10 | brettnem | link? :) |
| 18:08.14 | harryvv | What are the typical line of site distances for voip phones? |
| 18:08.25 | brettnem | harryvv: how far can you throw it? |
| 18:08.35 | Ahrimanes | xp_prg: hm i have some idea.. |
| 18:08.57 | ManxPower | harryvv, your question makes no sense. "How far can I throw air?" |
| 18:08.58 | harryvv | brettnem with a wifi antenna that is perhaps 30 feet in the air. |
| 18:09.12 | justinu|laptop | [TK]D-Fender: wake up, polycom support needed =) |
| 18:09.23 | harryvv | Manx makes complete sence. Ever |
| 18:09.31 | r_evolution | what the hell? |
| 18:09.35 | brettnem | fight! |
| 18:09.36 | r_evolution | sorry I just came back on this one |
| 18:09.43 | feld | crap i cant find any proper documentation. can anyone point me to some info that explains how to make 2 asterisk servers talk to each other / route calls? |
| 18:09.45 | r_evolution | typical LOS distances for voip phones? |
| 18:09.50 | Ahrimanes | my stomach is throwing air a lot.. |
| 18:09.52 | r_evolution | typically voip phones have these things... |
| 18:09.54 | r_evolution | called cords. |
| 18:10.05 | r_evolution | i don't think LOS is an issue :) |
| 18:10.11 | harryvv | Well obviosly thay have a market |
| 18:10.16 | r_evolution | now if you're talking about WiFi VoIP phones |
| 18:10.37 | r_evolution | no... voip phones pretty much always have a cord... it's one of those little 'important' things... right up there with power |
| 18:10.48 | r_evolution | WiFi VoIP phones = different subject matter :) |
| 18:10.52 | harryvv | I was refering to wifi voip phones. |
| 18:11.01 | ManxPower | r_evolution, PoEoWiFi |
| 18:11.03 | ManxPower | !!! |
| 18:11.08 | r_evolution | hahah SCORE FOR TECHNOLOGY! |
| 18:11.12 | harryvv | Somone in here does not have the logic to fill in the blanks :) |
| 18:11.18 | r_evolution | then how do you have an antenna that is 30 feet in the air |
| 18:11.24 | r_evolution | someone does not have the logic to ask a properly formed question |
| 18:11.28 | *** part/#asterisk a1fa (n=a1fa@207.210.210.202) |
| 18:11.32 | ManxPower | harryvv, no, someone here is not thinking thru the question |
| 18:11.35 | r_evolution | if you're asking the range on the typical wifi phone |
| 18:11.36 | harryvv | r_evolution ummm, a repeater tower? |
| 18:11.45 | r_evolution | then I think you need to look at the range of the WiFi hotspot |
| 18:11.48 | ManxPower | It's WiFi, it doesn't matter what you run over it. |
| 18:12.04 | r_evolution | exactly... that would be called a hotspot harry |
| 18:12.08 | r_evolution | quick lesson on wireless |
| 18:12.14 | r_evolution | you have different kinds |
| 18:12.20 | brettnem | "hotspot" sound like a marketing term |
| 18:12.26 | r_evolution | what you're referring to with the wifi phone |
| 18:12.33 | r_evolution | that wireless is typically called a 'hotspot' |
| 18:12.41 | harryvv | In a direct line of site examples say in a farm it should have some range and distance if the wifi repeater antenna is on a tower say 30 feet up or greater. |
| 18:12.43 | ManxPower | r_evolution, The only lesson he needs is to learn that WiFi sucks. It's unreliable, problem prone, and tends to have MASSIVE jitter. |
| 18:12.43 | r_evolution | and is usually broadcast around 2.4GHz (unlicensed spectrum) |
| 18:12.44 | nettie | wifi phones :) cool topic eheh |
| 18:12.52 | r_evolution | shhhh and listen |
| 18:13.05 | r_evolution | now... where you're talking about LOS... is more of the Point to Point |
| 18:13.16 | justinu|laptop | Ahrimanes: what model? |
| 18:13.20 | websae | I have a wifi phone ;) |
| 18:13.26 | r_evolution | Alvarion... SkyPilot... Infinet... AirSpan... etc. |
| 18:13.30 | Ahrimanes | justinu|laptop: qtek 8310/imate sp5 |
| 18:13.35 | r_evolution | in which case -- your wifi phone will NOT work with them |
| 18:13.43 | r_evolution | hotspot... if you can pick up a signal... you're good |
| 18:13.45 | justinu|laptop | i'm waiting for the Nokia E61 to be available here |
| 18:13.49 | harryvv | ManxPower mmmm mabey from your examples but in one case, there has been one example where three oil recovery ships where tied with wifi voip and asterisk. |
| 18:13.53 | r_evolution | point to point -- if you have the equipment you're good |
| 18:13.55 | nettie | and you need an hi performances, well surveyed and supercovered wifi network to deploy it |
| 18:14.00 | nettie | hi=high |
| 18:14.02 | Ahrimanes | justinu|laptop: yeah we're waiting for that as well |
| 18:14.18 | r_evolution | so think of it like this harry... |
| 18:14.19 | iCEBrkr | Anyone in need of a Sangoma EC quad-span TDM card? |
| 18:14.29 | iCEBrkr | Ooo, I said the dirty 'Sangoma' word :D |
| 18:14.33 | r_evolution | if you're getting a good/excellent signal with COMPUTER WiFi cards... |
| 18:14.34 | Ahrimanes | justinu|laptop: but man it's a struggle getting a sip client for windows mobile 5 smartphone.. you need to be an operator and pay big $$ |
| 18:14.35 | justinu|laptop | send it over :) |
| 18:14.41 | [TK]D-Fender | iCEBrkr : "d" or regular? |
| 18:14.45 | r_evolution | you'll probably get a good/excellent signal from your VoWiFi phone |
| 18:14.51 | r_evolution | but ya know... those things kinda suck anyway ;x |
| 18:14.52 | harryvv | nettie I am only talking about a localized example where the phones are contained within a given range and never leave that area say a farm or mine. |
| 18:14.57 | nettie | I suggest ou a good book, 802.11 Wireless Network site Survey and installation (ciscopress) |
| 18:14.57 | justinu|laptop | Ahrimanes: E61 is symbian... no idea what the SIP client landscape is there |
| 18:15.02 | r_evolution | yessssss |
| 18:15.08 | r_evolution | that's a GREAT book nettie. |
| 18:15.15 | websae | *websae sells WI-FI phones :) |
| 18:15.20 | Ahrimanes | justinu|laptop: kind of the same afaik.. but i hear rumors that E60/E61 migth ship with one |
| 18:15.21 | r_evolution | hey harry did you read what i wrote? |
| 18:15.22 | nettie | r_evolution yeah incredible well done |
| 18:15.27 | *** part/#asterisk gursikh (n=guriskh1@dsl254-123-245.nyc1.dsl.speakeasy.net) |
| 18:15.33 | justinu|laptop | Ahrimanes: interesting |
| 18:15.43 | r_evolution | or typed as it were |
| 18:15.50 | nettie | harryvv you want a suggestion? |
| 18:15.50 | iCEBrkr | [TK]D-Fender: grr, lemme check.. |
| 18:16.04 | harryvv | nettie, comming from a rf background in repair shure. |
| 18:16.22 | harryvv | sure go a head |
| 18:16.29 | nettie | buy a Dect :) |
| 18:16.30 | Ahrimanes | justinu|laptop: one producer of a voip client for the qtek quoted us 50.000 euro's for the first licensed client |
| 18:16.39 | justinu|laptop | heh |
| 18:16.47 | justinu|laptop | chump change :) |
| 18:17.00 | iCEBrkr | [TK]D-Fender: All it has marked on the card is A104 |
| 18:17.01 | nettie | harryvv how many users are you planning having? |
| 18:17.05 | Ahrimanes | justinu|laptop: after the first one, it gets slightly more reasonable (50 euros afaik per client) but damn.. |
| 18:17.09 | justinu|laptop | heh |
| 18:17.16 | iCEBrkr | [TK]D-Fender: But I kow it has echo-cancelling on it |
| 18:17.19 | [TK]D-Fender | iCEBrkr : Old one or the EC one I suggested way back? |
| 18:17.30 | harryvv | nettie possibly up to 5 mabey 10. no more then what the bandwith of 802.3 can handle. |
| 18:17.31 | [TK]D-Fender | iCEBrkr : Ok, A104d. Why's it up for sale? |
| 18:17.39 | iCEBrkr | [TK]D-Fender: Got 1 too many |
| 18:17.54 | [TK]D-Fender | iCEBrkr : Apparently! Just a "safety spare"? |
| 18:17.58 | nettie | harryvv anyway if you read that book you'll definitely get an idea of how hard, complex and problemprone voice over wifi is gonna be |
| 18:18.03 | _Sam-- | i am trying to figure out what *50 does....when i dial that from any of my grandstream phones through *, it is some kind of code...but the phone features have nothing noted for *50, and neither does *...any ideas? |
| 18:18.14 | iCEBrkr | [TK]D-Fender: Well, our quota manager throttles are calls now. So we're not dialing for dollars |
| 18:18.26 | iCEBrkr | [TK]D-Fender: We expected we needed more T1s.. In reality we don't |
| 18:18.44 | r_evolution | no kidding nettie... ;x |
| 18:18.45 | harryvv | nettie in your examples yes. |
| 18:19.03 | nettie | harryvv nevertheless if I had no experience I would only implement it with very specified gears |
| 18:19.24 | [TK]D-Fender | iCEBrkr : Ah. Thought that place canned you, or is this a new project? |
| 18:19.25 | nettie | which have: support, knowledgebase, case history. |
| 18:19.27 | harryvv | There are full wifi voip examples used in a contained enviroment where path production is allways assured. |
| 18:19.49 | iCEBrkr | [TK]D-Fender: New project |
| 18:20.07 | iCEBrkr | [TK]D-Fender: I've been doing some contract work to stay afloat |
| 18:20.09 | nettie | if you already decided to go for it I strongly suggest you to read that book |
| 18:20.26 | *** join/#asterisk DOffset (n=none@201.226.130.55) |
| 18:20.35 | nettie | ah, you definitely need an additional infrastructure for data traffic |
| 18:20.42 | sevard | VoIP over wifi? no. VoIP over wimax? yeah. |
| 18:21.05 | [TK]D-Fender | iCEBrkr : Have you been happy with those cards since you started working with them? |
| 18:21.13 | Katty | mister fender! |
| 18:21.14 | iCEBrkr | Yeah |
| 18:21.17 | r_evolution | i still keep waiting for WiMAX to be more than talk :-\ |
| 18:21.20 | *** join/#asterisk MacDome (n=eseidel@A17-205-15-32.apple.com) |
| 18:21.23 | DOffset | Hello... i have a question regarding Dialogic cards to work with Asterisk@home |
| 18:21.27 | iCEBrkr | [TK]D-Fender: Works flawlessly |
| 18:21.46 | Katty | twisted[asteria]: is still in your background? |
| 18:22.10 | nettie | harryvv you must do the survey with the phones you're going to use to be sure to accomplish positive results, buy phones which support survey |
| 18:23.01 | nettie | r_evolution yeah it's years ppl talk about wimax and in the meantime there're new emerging technologies which are very interesting.. |
| 18:23.07 | [TK]D-Fender | Katty : Passed you a hug way back there :) |
| 18:23.43 | [TK]D-Fender | Katty : I'm optics free now.... eyes aren't doing so hot today though, a regression. told its normal through the fisrt few weeks |
| 18:23.50 | [TK]D-Fender | iCEBrkr : Glad to hear. |
| 18:24.20 | *** join/#asterisk kbitter (i=_kbitter@201.64.192.132) |
| 18:24.29 | *** join/#asterisk gmaruzz (n=gmaruzz@217-133-80-112.b2b.tiscali.it) |
| 18:24.31 | Katty | [TK]D-Fender: did the surgery go ok? |
| 18:24.34 | r_evolution | kinda how i'm feeling netie... i keep hearing 'yeah we're about to have a WiMAX product' |
| 18:24.41 | r_evolution | but it's the same old talk... promises with no delivary |
| 18:24.52 | [TK]D-Fender | Katty : thats what I'm referring to. |
| 18:25.53 | iCEBrkr | [TK]D-Fender: The only problems I had was Asterisk would segfault when I'd drop the callfiles in the directory. I think the create/modify date was screwing it up. So I put a .5 sec sleep() inbetween the creation of the call files. |
| 18:25.53 | *** join/#asterisk viLeR (i=1000@200.114.70.228) |
| 18:26.09 | justinu|laptop | lol |
| 18:26.11 | justinu|laptop | ouch |
| 18:26.22 | kbitter | hi all, im knew to asterisk.. i want to enable the g729 codec on my calls, but * does not come with it, how can i proceed? |
| 18:26.33 | *** join/#asterisk mtaht3 (n=m@reserve-64-79-114-30.wiline.com) |
| 18:26.51 | *** part/#asterisk mtaht3 (n=m@reserve-64-79-114-30.wiline.com) |
| 18:26.53 | r_evolution | jeez TK... feeling a little violent? |
| 18:27.06 | [TK]D-Fender | iCEBrkr : Well every source I've read about .call files says that its tricky placing them in there and to use a rename to move them into the folder as it only changes the inode so you don't get a partial file. nothing to do with hardware... |
| 18:27.22 | brettnem | so anyone got that latest polycom firmware and feeling in a sharing mood? :D |
| 18:27.23 | [TK]D-Fender | r_evolution : You taunt, I retaliate ;) |
| 18:27.37 | [TK]D-Fender | r_evolution : We should all be thankful I can't wield a sword in here ;) |
| 18:27.46 | iCEBrkr | [TK]D-Fender: Yea. I dump the call files in a queue directory and then move them.. Yea.. |
| 18:27.47 | brettnem | [TK]D-Fender I've eard the same |
| 18:27.50 | Katty | [TK]D-Fender: but no complications or anything? |
| 18:28.05 | iCEBrkr | [TK]D-Fender: But I still think the issue was the file date. Cuz when I put the delay in there, my problems went away |
| 18:28.39 | [TK]D-Fender | Katty : Only complication is today I have strain and things are blurry. Should clear up though I'm told and checkup is in 2 days |
| 18:29.03 | [TK]D-Fender | Katty : trouble focusing. |
| 18:29.20 | justinu|laptop | staring at your monitor is probably doing wonders :) |
| 18:29.24 | justinu|laptop | lol |
| 18:29.27 | Katty | [TK]D-Fender: close your eyes and try to relax every 30 minutes |
| 18:29.28 | salviadud | it's all emotional |
| 18:29.36 | Katty | [TK]D-Fender: don't strain it too much...your eyes are /trying/ to heal. |
| 18:29.39 | salviadud | i'd recommend fender to get a massage |
| 18:29.42 | Katty | [TK]D-Fender: 30 minute breaks will make recovery go quicker. |
| 18:29.56 | Katty | [TK]D-Fender: yes, and a massage will do wonders. |
| 18:30.00 | Katty | [TK]D-Fender: also, ice cream! |
| 18:30.04 | r_evolution | and sex! |
| 18:30.13 | r_evolution | :) |
| 18:30.14 | Katty | [TK]D-Fender: drive out here, and i'll see to it you get both (= |
| 18:30.15 | *** join/#asterisk epablo (n=epablo@200.84.4.217) |
| 18:30.30 | epablo | Hi people.. how is going |
| 18:30.31 | r_evolution | sex and ice cream? |
| 18:30.33 | r_evolution | kinky. |
| 18:30.37 | Katty | r_evolution: no |
| 18:30.43 | r_evolution | no fun. |
| 18:30.46 | Katty | r_evolution: you're clearly not a trained massage therapist. |
| 18:30.51 | brettnem | haha.. is this really happening? |
| 18:30.55 | r_evolution | clearly not... |
| 18:31.02 | [TK]D-Fender | Katty : Thats a nasty overhead for my food & entertainment budget... 500mi drive :) |
| 18:31.03 | Katty | r_evolution: go take a class, it's great info |
| 18:31.08 | r_evolution | i am however a reformed rave kid |
| 18:31.14 | *** join/#asterisk stack_ (n=stack@63.239.190.202) |
| 18:31.15 | r_evolution | so that means i did a lot of E once upon a time ;x |
| 18:31.27 | stack_ | Afternoon everyone... |
| 18:31.28 | Katty | r_evolution: and the deep muscle tissue massage is /perfect/ for guys, if you're gay |
| 18:31.35 | r_evolution | only if it comes with the happy ending Katty |
| 18:31.37 | epablo | I got a problem with an IAX setup. A nasty No authorization found Code 50. |
| 18:31.39 | r_evolution | in regards to the class |
| 18:31.50 | Katty | r_evolution: you get a certificate thing |
| 18:32.00 | *** join/#asterisk A-Tuin (n=a-tuin@steves.ip.v4.me.uk) |
| 18:32.00 | r_evolution | i dont want a certificate... i want a happy ending! |
| 18:32.11 | brettnem | if you're gay? what? |
| 18:32.13 | r_evolution | but not from some girl who has more body hair than me |
| 18:32.18 | r_evolution | that's scary. |
| 18:32.21 | Katty | brettnem: males respond well to deep tissue massage. |
| 18:32.30 | Katty | brettnem: specially the guys who have all the muscle in their neck and upper back. |
| 18:32.35 | epablo | Can anyone swing me, what that error refers to. other than a problem in my setup? |
| 18:32.40 | brettnem | Katty: what does that have to do with being gay? |
| 18:32.42 | stack_ | We are having problems with people dialing out through our PRI. Every once in a while, I get "Channel 0/22, span 2 got hangup request" when making an outbound call. If they try again, it works fine... any ideas? |
| 18:32.45 | justinu|laptop | i had an ex girlfriend who was a message therapist |
| 18:32.46 | Katty | brettnem: so, if r_evolution is gay, then he should take special interest in deep tissue massage. |
| 18:32.53 | brettnem | I'm not gay, but I do like a massage? Is that possible? |
| 18:33.00 | Katty | brettnem: because girls don't respond well to deep tissue massage |
| 18:33.01 | epablo | Yes... |
| 18:33.02 | papo | If I want to dial ${EXTEN}@ethz.ch using the peer I defined in [ekiga.net], how should my Dial() application look like? |
| 18:33.06 | Katty | brettnem: it's usually too rough. |
| 18:33.07 | r_evolution | on that note... i dont think ill take a special interest in deep tissue massage |
| 18:33.15 | brettnem | Katty: oh I think I get it. |
| 18:33.19 | Katty | brettnem: good for you! |
| 18:33.19 | jpabuyer | this is #asterisk not #assterisk |
| 18:33.22 | r_evolution | Katty... introduce E into the scenario... and nothing is too rough |
| 18:33.24 | brettnem | katty: ;) |
| 18:33.45 | Katty | why does everyone presume sex when someone mentions massage? |
| 18:33.46 | Katty | meesha. |
| 18:33.54 | brettnem | hell I didn't. :P |
| 18:33.56 | r_evolution | I'm telling you Katty... it's the happy ending. |
| 18:34.00 | brettnem | haha |
| 18:34.01 | epablo | Can anyone swing me, what code 50 error is on IAX refers to? other than a problem in my setup? |
| 18:34.04 | iDunno | did someone say sex? |
| 18:34.05 | iDunno | :) |
| 18:34.11 | Katty | iDunno: no, but i do want a hug. |
| 18:34.19 | r_evolution | katty was talking about sex and ice cream earlier... |
| 18:34.21 | Katty | iDunno: then again, when do ever not want a hug. |
| 18:34.29 | Katty | r_evolution: /you/ were talking about sex and ice cream. |
| 18:34.43 | r_evolution | you participated |
| 18:35.03 | *** join/#asterisk Dark_kbitter (i=_kbitter@201.64.192.132) |
| 18:35.23 | Katty | iDunno: are you all bouncy today? |
| 18:35.50 | iDunno | not really, tired and hungry and feeling a bit ill :/ |
| 18:36.12 | *** join/#asterisk rabelais (n=blank@santamonica-cuda4-24-55-43-249.vnnyca.adelphia.net) |
| 18:36.13 | Katty | :< |
| 18:36.28 | iDunno | s'all right though - pizza has been ordered to fix this :) |
| 18:36.48 | r_evolution | may I advise some choc milk with that pizza iDunno? |
| 18:36.50 | *** join/#asterisk oej (n=oej@apollo.webway.se) |
| 18:37.05 | justinu|laptop | or just straight glucose |
| 18:37.05 | iDunno | banana would be better ;) |
| 18:37.20 | Katty | potassium! |
| 18:37.53 | *** join/#asterisk MacDome (n=eseidel@A17-205-15-32.apple.com) |
| 18:39.48 | Dark_kbitter | hi all, does any one know how to add the g729 call support to asterisk? |
| 18:39.58 | file | Dark_kbitter: huh? |
| 18:40.14 | justinu|laptop | buy the licenses from digium |
| 18:40.26 | r_evolution | buy licenses... follow read-me |
| 18:40.26 | iDunno | and yeah - what justinu|laptop said :) |
| 18:40.28 | r_evolution | simple. :) |
| 18:40.40 | brettnem | yeah, what iDunno said |
| 18:40.48 | justinu|laptop | yeah, what brettnem said |
| 18:41.03 | r_evolution | oh no. not the eternal loop. |
| 18:41.04 | iDunno | yay! cyclic :) |
| 18:41.10 | Dark_kbitter | file: justinu answered my question.. :-D thanks |
| 18:41.18 | r_evolution | Do While :( |
| 18:41.23 | Katty | do while /not/ |
| 18:41.42 | r_evolution | O_o |
| 18:41.55 | Katty | brettnem: kinky. |
| 18:41.56 | r_evolution | disagreeing with me !=allowed |
| 18:42.03 | *** join/#asterisk jtodd (n=jtodd@68-185-24-2.static.mdfd.or.charter.com) |
| 18:42.05 | *** join/#asterisk harryvv (n=none@S010600a0c93f6f7e.vs.shawcable.net) |
| 18:42.35 | harryvv | work station crashed leaving my old..this one here. What was that 802.3 wifi book somone mentioned in here? |
| 18:43.10 | justinu|laptop | r_evolution: i'd have to disagree with that statement |
| 18:43.27 | r_evolution | <PROTECTED> |
| 18:43.33 | r_evolution | harry : 802.11 Wireless Network site Survey and installation |
| 18:43.41 | r_evolution | you shall die justin. :-\ |
| 18:43.44 | *** part/#asterisk sandos (n=sandos@me-0-50-da-e0-bb-36.2.cust.bredband2.com) |
| 18:43.58 | harryvv | All right thanks r_evolution |
| 18:44.11 | justinu|laptop | i die a little every day |
| 18:44.14 | justinu|laptop | :P |
| 18:44.31 | r_evolution | feeling a little existential today? |
| 18:44.44 | *** join/#asterisk jeffgus (n=jeffgus@alpha.zimage.com) |
| 18:45.13 | justinu|laptop | always |
| 18:45.30 | [TK]D-Fender | justinu : Well hurry up slacker! ;) |
| 18:46.04 | justinu|laptop | this seems like the stupidest idea i've heard in a long while |
| 18:46.04 | justinu|laptop | http://www.newscientisttech.com/article/dn9176-flybywireless-plane-takes-to-the-air.html |
| 18:46.24 | r_evolution | i take it you werent alive for the 'pet rock' craze then justin |
| 18:46.36 | justinu|laptop | i think i was in diapers |
| 18:47.00 | Ahrimanes | yesterday? |
| 18:47.11 | Qwell[] | sorry...you'll have to excuse my ignorance here |
| 18:47.21 | Qwell[] | don't ALL planes fly without wires? |
| 18:47.32 | stack_ | We are having problems with people dialing out through our PRI. Every once in a while, I get "Channel 0/22, span 2 got hangup request" when making an outbound call. If they try again, it works fine... any ideas? |
| 18:47.35 | Qwell[] | haven't ALL planes flown without wires, since the first Wright Brothers plane? |
| 18:47.51 | justinu|laptop | they're talking about using wireless networks to link the controls with the control surface actuators |
| 18:48.04 | Qwell[] | and what, praytell, were they using before? |
| 18:48.18 | justinu|laptop | cables, hydraulics, etc. |
| 18:48.27 | Qwell[] | so the wifi is inside the plane? |
| 18:48.31 | justinu|laptop | sounds like it |
| 18:48.34 | Qwell[] | what' |
| 18:48.35 | harryvv | a small model plane |
| 18:48.37 | Qwell[] | s the point? |
| 18:48.40 | justinu|laptop | no idea |
| 18:48.49 | harryvv | easy configuration and saving weight |
| 18:48.54 | BadPacket | does anyone here use www.iterus.com? |
| 18:49.00 | brettnem | hey, anyone know why audio would drop out in the middle of a call... assuming no network problems? |
| 18:49.13 | brettnem | oops.. sorry, I mean one way audio |
| 18:49.37 | harryvv | When I was a heavy lift helicopter crew chief our wire bundles would exceed 4 inches in diameter. A rats nest if it was ever taken apart. |
| 18:49.40 | BadPacket | i guess that's a no |
| 18:50.20 | justinu|laptop | harryvv: would you fly on a helicopter who's control system used RF? |
| 18:50.25 | justinu|laptop | what if someone jammed it? |
| 18:50.35 | justinu|laptop | what if solar flares or something freaked it out |
| 18:50.53 | [TK]D-Fender | BadPacket : I know a guy who works there. |
| 18:51.09 | *** join/#asterisk mind__ (n=mind@host236-233.pool878.interbusiness.it) |
| 18:51.11 | harryvv | justinu no :) but in a drone it would work. Say the rf was contained within the fuselodge of the aircraft then it would be shielded from outside interferance. |
| 18:51.14 | mind__ | hi |
| 18:51.37 | mind__ | What is the module chan_h323.os? I can't find on google |
| 18:51.37 | brettnem | crazy |
| 18:51.42 | *** join/#asterisk [hC] (n=hardcore@mail.metrobridge.com) |
| 18:51.58 | harryvv | A aircrafts cabin is one big fariday cage so it would be imune to interferance. |
| 18:52.09 | BadPacket | [TK]D-Fender: so do I - websae, who keeps bothering me with PM's |
| 18:52.37 | BadPacket | just trying to figure out if they're for real... so far I don't think so |
| 18:52.49 | brettnem | totally immune |
| 18:52.50 | [TK]D-Fender | BadPacket : I know another guy there. What do you mean "for real"? |
| 18:52.53 | justinu|laptop | harryvv: the germans lost two Tornado fighters because they flew too close to an FM radio transmitter |
| 18:53.02 | BadPacket | real != nufone |
| 18:53.15 | BadPacket | if they're actually more than one guy |
| 18:53.43 | brettnem | better make that two |
| 18:53.55 | harryvv | justinu yea even the h-60s is not totally imune from fr interferance. |
| 18:54.00 | lunk | electronic brakes? |
| 18:54.05 | lunk | helll no |
| 18:54.08 | brettnem | I'd qualify it with "do they use asterisk or not" |
| 18:54.09 | lunk | next. |
| 18:54.27 | *** join/#asterisk Cybertoy (n=maxim@dsl254-123-241.nyc1.dsl.speakeasy.net) |
| 18:54.30 | sremington | Anybody seen a "FXO phone off hook in weird state 3??" error before? |
| 18:54.59 | Cybertoy | hi .. quick question... I use enum lookup and it seems that tf.voipmich.com routes many toll-free numbers that I lookup on e164.org... |
| 18:55.13 | Cybertoy | however since about a week I get a "Forbidden - wrong password..." error ... |
| 18:55.23 | harryvv | my asterisk is now acting a little strange. it does not respond when dialing some numbers..only 10 seconds is there cli activity does it show up. |
| 18:55.30 | sremington | channel.h defines 3 as AST_STATE_DIALING /*! Digits (or equivalent) have been dialed */ |
| 18:55.31 | Cybertoy | I don't expect that I need to authenticate if I use enum entries... or am I missing something? |
| 18:55.41 | brettnem | asterisk always acts strange |
| 18:55.56 | *** join/#asterisk marl (n=matt@albacom.plus.com) |
| 18:56.02 | sremington | but zap channel is not dialing any digits when this happens |
| 18:56.07 | brettnem | Sometimes I get calls that attach to the wrong SDP stream.. asterisk is just flaky like that |
| 18:56.40 | stack_ | Our PRI is just hanging up on people... would this be an asterisk thing or a PRI thing. It happens randomly to different people, but it's been terrible today. |
| 18:56.42 | marl | can anyone point me to a good 'press # to accept this call' script? please? |
| 18:57.02 | harryvv | im outa here |
| 18:57.25 | marl | stack_, have you changed anything in the pbx over the last few days? |
| 18:57.40 | brettnem | ugh. now I'm getting customers complaining that transfered calls sometimes don't establish 2 way talk paths.. grr asterisk.. |
| 18:57.53 | stack_ | marl, just the echo cancel algorithm, but I was getting the hang ups before that as well |
| 18:58.12 | brettnem | sometimes the two way talk path goes one way in the middle of the call. |
| 18:58.22 | stack_ | marl, and it is on inbound and outbound calls |
| 18:58.22 | justinu|laptop | weird |
| 18:58.27 | stoffell | stack_, what verison you're using? |
| 18:58.31 | *** join/#asterisk clive- (n=pirch@dsl-145-61-54.telkomadsl.co.za) |
| 18:58.36 | stack_ | stoffell, 1.2.7.1 |
| 18:58.37 | justinu|laptop | berttnem: SIP? |
| 18:58.37 | marl | did it become worse after you change the echo settings? |
| 18:58.46 | brettnem | justinu: yes, SIP->SIP |
| 18:58.52 | justinu|laptop | reinvite? |
| 18:58.53 | brettnem | asteirsk 1.2.0 |
| 18:58.56 | stack_ | marl, about the same, we have a higher call volume today, so it's hard to tell |
| 18:58.58 | brettnem | justinu: no |
| 18:59.02 | stoffell | stack_, are the calls just hungup? or does asterisk/pri hang completely? |
| 18:59.03 | justinu|laptop | so asterisk is proxying the rtp? |
| 18:59.07 | brettnem | justinu: yes |
| 18:59.28 | justinu|laptop | brettnem: any chance of getting a pcap of an affected call? |
| 18:59.31 | xp_prg | ok here is my confusion, the management api can make calls and play a pre-recorded message, yet what is this dialplan thing? Do I need it? Do I connect and write an extentions.conf then disconnect, help me understand this management api, all I want to do is dial a phone # and play a pre-recorded message |
| 18:59.36 | brettnem | justinu: I have a couple of customers on different * boxes complaining of this. |
| 18:59.44 | justinu|laptop | all 1.2.0? |
| 18:59.45 | marl | i have come accross this in the past with standard PBX's, it can be down to noise/interferance on the phone line (althow it could be * as well!!!) |
| 18:59.51 | stack_ | stoffell, I either get a 'Hungup 'Zap/XX-1' or something stating the other end has hung up and then the hangup notice |
| 19:00.00 | brettnem | justinu: intermitent.. no some are 1.2.6 |
| 19:00.12 | justinu|laptop | how much traffic you running? |
| 19:00.13 | stack_ | we've been getting crackling on the line sometimes too |
| 19:00.20 | brettnem | not much.. couple hundred calls a day |
| 19:00.27 | brettnem | maybe a couple thousand.. |
| 19:00.30 | stoffell | stack_, hm, make sure you set verbosity of logging high enough, this might even be related to the telco.. |
| 19:00.32 | brettnem | but it's low |
| 19:00.36 | justinu|laptop | brettnem: i don't have that issue... do sip->sip exclusively |
| 19:00.46 | *** join/#asterisk DrPete (n=Pete@host-84-9-255-194.bulldogdsl.com) |
| 19:00.53 | brettnem | justinu: I'm only SIP->SIP 100% |
| 19:01.03 | stack_ | stoffell, that's what we are thinking, but I don't want this to because of my ignorance some how ;p |
| 19:01.14 | brettnem | justinu: just lost audio on a call.. then it came back.. it's still up... |
| 19:01.34 | marl | stack_, sounds more like a line fault than anything in *, althow i stand corrected if anyone has come accross this before, im only new to * :) have worked with prop fone systems on both ananlog and isdn/pri/bri |
| 19:02.10 | marl | stack_, what telco u with? |
| 19:02.12 | justinu|laptop | brettnem: nothing like VAD happening? |
| 19:02.17 | stack_ | marl, airespring |
| 19:02.25 | stoffell | stack_, it wouldn't be bad to ask your telco to do a line test. you could set logging high enough, maybe that gives an idea... |
| 19:02.32 | brettnem | justinu: polycom->asterisk->ser->asterisk->polycom |
| 19:02.48 | justinu|laptop | ser is on the signalling plane only tho, right? |
| 19:02.50 | *** join/#asterisk zparta_ (i=zparta@146.net90.skekraft.net) |
| 19:02.56 | stack_ | stoffell, I've set the console to 10 right now... |
| 19:03.06 | *** part/#asterisk epablo (n=epablo@200.84.4.217) |
| 19:03.06 | marl | id ask fo rthe line test as well, telecos dont normally charge for that kinda thing, so its a good thing to check |
| 19:03.19 | stoffell | stack_, yep, verbosity 10 and debug maybe 5 or so |
| 19:03.22 | *** part/#asterisk ghento (n=ghento@CPE0014bfaf7d7a-CM00111ae4684c.cpe.net.cable.rogers.com) |
| 19:03.36 | brettnem | justinu: correct |
| 19:03.41 | brettnem | justinu: no rtp proxy |
| 19:04.06 | brettnem | justinu: it's up now, can you think of anything for me to poke at? |
| 19:04.07 | stack_ | I've got it cranked up now... we'll see... |
| 19:04.08 | clive- | justinu I hear rumours of VAD, maybe in about 6 months something will surface |
| 19:04.14 | marl | BUT, if they turn around and say theres nothing wrong with the line, DONT just asume there rite and * is playing up, telcos DONT like admitting there are faults sometimes, it depends on who u get ont he end of the phone and way kinda day they have had :( |
| 19:04.35 | justinu|laptop | brettnem: i usually rely on network traffic caps, and ethreal to diagnose crap like that. |
| 19:05.06 | brettnem | justinu: rigt, just trying to figure out what to do with an existing call |
| 19:05.08 | marl | so can anyone recomend a good 'press # to accept this call' script? |
| 19:05.46 | justinu|laptop | brettnem: still happening on your call? |
| 19:05.54 | adept_ | marl: sure send me your server login info and $500 |
| 19:06.13 | adept_ | and I will put a single fucking line in your extensions.conf |
| 19:06.18 | brettnem | justinu: it's OK now.. but it is the same call with the problem |
| 19:06.35 | r_evolution | i think adept needs a hug ;x |
| 19:06.38 | justinu|laptop | well, you're not gonna be able to figure out what happened, i don't think |
| 19:06.43 | marl | adept_, why u swairing at me? i only asked a question |
| 19:06.54 | *** join/#asterisk CanyonMan (i=cepasp@grace.isc.rit.edu) |
| 19:06.59 | adept_ | :O the blogosphere isn't going to like this... |
| 19:07.01 | r_evolution | adept = ball of joy and ray of sunshine |
| 19:07.18 | CanyonMan | Do any of you have an SPA-3102 ? |
| 19:07.40 | brettnem | justinu: blah.. well thanks.. let me know if you come up with any ideas.. |
| 19:07.54 | kbitter | justino: i dont have any pci card in *. I just need sip-to-sip calls. do i still have to buy the g729 license from digium to use this codec during the calls? |
| 19:07.55 | CanyonMan | I have everything working EXCEPT inbound CID. |
| 19:08.11 | justinu|laptop | the only thing I can think is that for whatever reason, one phone's TX RTP stops getting thru |
| 19:08.19 | justinu|laptop | with stock asterisk, thats enough to break the entire RTP chain |
| 19:08.36 | brettnem | justinu: is there a way to prevent that?? |
| 19:08.37 | jpabuyer | if going to g729 <-> g729, no... only when transcoding |
| 19:08.53 | justinu|laptop | brettnem: there's some patches in the bug tracker that allow async rtp |
| 19:08.56 | justinu|laptop | i use them |
| 19:09.07 | *** join/#asterisk xai (n=pasta@about/networking/0.0.0.0/xai) |
| 19:09.23 | clive- | how would you guys begin to trace a fault of random call hangups |
| 19:09.28 | xai | ManxPower: you workin? |
| 19:09.29 | justinu|laptop | but I would have to figure out why RTP is getting interrupted |
| 19:09.44 | DOffset | Im trying to setup asterisk as my pbx provider and vonage as my phone provider... |
| 19:09.58 | brettnem | justinu: do you get much of a performance improvement with that?? |
| 19:10.08 | brettnem | justinu: can't jut internet latency cause that kind of stuff? |
| 19:10.09 | DOffset | and i have a Dialogic D120JTC-LS card i need some help on this please |
| 19:10.19 | justinu|laptop | brettnem: no, i use it for other reasons (early media) |
| 19:10.31 | justinu|laptop | internet latency, or other connectivity issues... sure |
| 19:10.42 | justinu|laptop | in that case, there's basically nothing you can do |
| 19:10.47 | clive- | doffset afaik it needs some special drivers to work with asterisk |
| 19:10.49 | brettnem | justinu: 5162? |
| 19:10.52 | xai | ManxPower: where do you go when you need a sizing design for a call center? |
| 19:10.58 | brettnem | justinu: but do you think that patch would help out? |
| 19:11.40 | *** join/#asterisk Cybertoy (n=maxim@dsl254-123-241.nyc1.dsl.speakeasy.net) |
| 19:11.55 | DOffset | *clive-* do you know where to get it?... i read something about a license |
| 19:12.36 | justinu|laptop | brettnem: no, that's not it... and i'm not finding it at the moment |
| 19:12.51 | kbitter | jpabuyer: ok, but how do i enable the g729 codec support on * just to g729 <-> g729, without transcoding. do i have to compile something? |
| 19:12.51 | clive- | doffset i'd start by calling digium, otherwise buy a digium or sangoma card |
| 19:13.49 | DOffset | *clive-* they said that as far they are concerned.. dialogic doesn't work with asterisk... :( |
| 19:14.30 | clive- | doffset I hear about some drivers, but only for certain dialogic cards, not sure which....check the wiki |
| 19:14.37 | kbitter | jpabuyer: everytime i try to use the g729 on my sip-to-sip calls, it does not work! |
| 19:16.14 | *** join/#asterisk djMax (n=chatzill@artsalliancelabs.com) |
| 19:16.18 | justinu|laptop | brettnem: if you can get a good capture during a bad call, we might be able to get a difinitive answer as to what's happening |
| 19:16.23 | justinu|laptop | otherwise, tough to say |
| 19:16.24 | brettnem | justinu: I'm being told that on one of the asterisk boxes a station to station call did this |
| 19:16.48 | jpabuyer | kbitter, I don't remember that.. look in voip-info.org |
| 19:17.04 | DOffset | *clive-* other question... can i use Softphones with my sip provider(voipjet) as extensions to transfer call to them? |
| 19:17.13 | *** join/#asterisk RonaldGibson (n=ronaldgi@adsl-66-218-62-5.dslextreme.com) |
| 19:17.16 | jpabuyer | kbitter, maybe you do have to buy a $10 license :) |
| 19:18.09 | Supaplex | the "don't work" excuse is an easy crutch. =) |
| 19:19.44 | clive- | doffset your softphones must be registered as clients of your asterisk box |
| 19:21.56 | DOffset | *clive-* the thing is i have people working for me outside of the country and i want to transfer calls to them... do i need an static ip address on my asterisk server for them to use the softphone? |
| 19:22.45 | clive- | doffset there are workarounds, but if you want it to work reliably, you need a static ip |
| 19:23.58 | DOffset | *clive-* wich softphone do you recomend? one that is easy to configure and integrate with asterisk |
| 19:25.53 | *** join/#asterisk djacob-wcgs (n=djac@65.105.215.226.ptr.us.xo.net) |
| 19:26.17 | *** join/#asterisk mtaht3 (n=m@reserve-64-79-114-30.wiline.com) |
| 19:26.30 | *** join/#asterisk LL0rd (n=mb@devserver.info) |
| 19:26.34 | LL0rd | hi |
| 19:26.39 | clive- | I am not so clued up about soft phones.... |
| 19:26.57 | clive- | x-lite is popular with sip |
| 19:27.11 | djacob-wcgs | i am using the x-ten phone and it works good |
| 19:27.21 | djacob-wcgs | i am a noob though :) |
| 19:27.59 | brettnem | anyone know what the status on the jitterbuffer is? |
| 19:28.49 | DOffset | well that's what i got :D |
| 19:28.54 | djacob-wcgs | be back in a few |
| 19:29.05 | r_evolution | i need a nap :( |
| 19:29.13 | *** join/#asterisk gmaruz1 (n=gmaruzz@217-133-80-112.b2b.tiscali.it) |
| 19:30.32 | brettnem | justinu: same call just lost audio again.. for 25 seconds |
| 19:30.50 | *** join/#asterisk tanvalmg (n=mguevarr@67-134-234-194.dia.static.qwest.net) |
| 19:31.09 | justinu|laptop | i'd have to suspect some kind of network anomaly |
| 19:31.44 | tanvalmg | Question on Directory() Application...Is this application valid for Asterisk v.1.0.6 ? |
| 19:31.57 | Zodiacal | anyone know if theres a way to get the cisco 7960 phones to display info using only one line of text insted of two lines? it takes up too much of the screen real estate |
| 19:33.11 | DOffset | i know is a dumb question but... how do i download a file in linux?...specially CenteOS for the asterisk |
| 19:33.52 | iulius | wget |
| 19:34.08 | syle | wget http://blah.blah.blah/file.txt |
| 19:34.34 | DOffset | and then how do i open a directory |
| 19:34.49 | DOffset | i know is like cd something on MS-DOS |
| 19:34.55 | Supaplex | same |
| 19:34.55 | brettnem | man this is so weird |
| 19:35.04 | Supaplex | man bash :) |
| 19:35.07 | LL0rd | i need help! after Playback() I can't hear caller and the caller can't hear me |
| 19:35.12 | DOffset | and to open up a file? to execute it |
| 19:35.18 | syle | google unix commands |
| 19:35.29 | robl^ | "man man" |
| 19:35.46 | syle | try #linux |
| 19:36.07 | Zodiacal | or #centos |
| 19:38.49 | *** join/#asterisk Greek-Boy (n=Greek-Bo@193.220.93.162) |
| 19:39.38 | DOffset | thanks |
| 19:39.50 | znoG | anyone use a SPA-841? |
| 19:40.37 | znoG | i have this problem that it works fine, mostly, then it remains registered to Asterisk but, when you call the phone, it doesn't detect an incoming call yet the caller hears the ringing .. |
| 19:40.45 | znoG | when looking at the Sipura info page, it says the device is idle |
| 19:41.09 | _Sam-- | i am trying to figure out what *50 does....when i dial that from any of my grandstream gxp2000 phones through *, it is some kind of code ...but the phone features have nothing noted for *50, and neither does *...does anyone know what *50 does? |
| 19:41.24 | websae | I have many SPA-841 |
| 19:41.32 | websae | the first addition was terrible |
| 19:41.37 | *** join/#asterisk anderiv (n=anderiv@207.67.87.34) |
| 19:41.44 | _Sam-- | edition |
| 19:42.02 | websae | never placed in the spelling bee |
| 19:42.18 | _Sam-- | apparently only the math bee for addition |
| 19:42.34 | websae | clearly yes |
| 19:44.34 | ManxPower | xai, not a call center |
| 19:44.49 | ManxPower | xai, Service location: 1125 Loop Road, Steele, AL 35987 256-538-8517 |
| 19:45.14 | *** part/#asterisk BadPacket (n=root@unaffiliated/badpacket) |
| 19:45.24 | ManxPower | Standard internet access. SLA would be nice. Would run OVERFLOW calls over it. It's stupid to run all your voice over VoIPoInternet |
| 19:47.18 | *** join/#asterisk |dennis| (i=dennis@200.32.215.84) |
| 19:47.21 | ManxPower | BTW, any ISPs or CLECs here that offer CHEAP T-1 internet service? /msg me. Service location: 1125 Loop Road, Steele, AL 35987 256-538-8517 |
| 19:47.25 | xai | ManxPower: is that your new office? |
| 19:47.36 | DrPete | whats the best way to get asterisk? I remeber I used cvs when i used it months ago, I see I can just download it |
| 19:47.37 | ManxPower | xai, something like that |
| 19:47.53 | ManxPower | DrPete, production or developement? |
| 19:48.06 | Qwell[] | ManxPower: having trouble getting a loop on loop road? |
| 19:48.12 | Greek-Boy | I ordered cisco phones without lience, I hope i'm not going to have problems |
| 19:48.21 | ManxPower | Qwell, Trouble getting a cheap loop on loop road |
| 19:48.26 | DrPete | ManxPower: Well its just for the house at the moment, but I want it stable |
| 19:48.40 | ManxPower | DrPete, then go with 1.2.x |
| 19:48.42 | xai | ManxPower: ok.. we're trying to get into callcenter sizing/design work. asterisk or otherwise. |
| 19:49.19 | SplasPood | Heh, I'm trying to design a system to generate asterisk config from a db... Now I've moved onto extensions.conf and I'm banging my head against a wall trying to figure out how to automate the generation of that, yet still allow for a reasonable level of flexibility |
| 19:49.39 | ManxPower | Will be selling Internet service and telephone service and cable service to the people that live at the campground |
| 19:49.43 | xai | websae: no.. |
| 19:50.28 | DrPete | ManxPower: Thanks |
| 19:50.50 | brettnem | justinu: you there.. new information.. this is interesting.. every 30 MINUTES of call duration on the mark.. the audio goes one way for 15-30 seconds |
| 19:51.21 | brettnem | [TK]D-Fender: what do you think of that? |
| 19:51.52 | xai | ManxPower: how many are living there now? |
| 19:51.54 | [TK]D-Fender | brettnem : Freaky, and nothing I can explain. |
| 19:51.59 | brettnem | weird |
| 19:52.09 | [TK]D-Fender | brettnem : But far to consistent to blame on * ;) |
| 19:52.19 | brettnem | HAHA |
| 19:52.23 | brettnem | good point |
| 19:52.26 | ManxPower | xai, 3 "households" |
| 19:52.30 | [TK]D-Fender | I am SO bad :D |
| 19:52.57 | *** join/#asterisk OloBola (n=not@netblock-68-183-67-158.dslextreme.com) |
| 19:53.39 | [TK]D-Fender | SplasPood : Take note of how the varying GUI's out there break up the process... the logic is quite easy. |
| 19:53.56 | SplasPood | [TK]D-Fender: Everything seems so restrictive |
| 19:54.26 | *** part/#asterisk fjean (n=fjean@201.29.140.206) |
| 19:55.20 | justinu | who says you can't LNP numbers away from vonage? |
| 19:55.31 | brettnem | SplasPood: Go check out phonecall http://www.vecsector.com/phonecall |
| 19:55.52 | justinu | brettnem: so start your capture, lets see what's happening |
| 19:56.04 | websae | you can LNP numbers from vonage |
| 19:56.08 | websae | I do it all the time |
| 19:56.24 | justinu | yeah, i just did too... i heard they wouldn't do it |
| 19:56.27 | [TK]D-Fender | SplasPood : Thats the point of a GUI! |
| 19:56.57 | *** join/#asterisk ToyMan (n=stuq@user-12lcqia.cable.mindspring.com) |
| 19:57.40 | [TK]D-Fender | SplasPood : Press a digit, go to a specific phone/queue/menu/vmbox, etc. Wash/rinse/repeat. |
| 19:57.49 | brettnem | justinu: I'm restablishing the call and doing an ngrep on 5060 |
| 19:57.53 | *** join/#asterisk crich1999 (n=crich@port-212-202-0-42.dynamic.qsc.de) |
| 19:59.29 | justinu | 5060 is just signalling, so we're not gonna see any issues there, i doubt. |
| 20:01.14 | brettnem | justinu: suggestion? |
| 20:01.49 | justinu | tcpdump -s0 -w full.cap |
| 20:02.01 | *** join/#asterisk MacDome (n=eseidel@A17-205-15-32.apple.com) |
| 20:02.05 | justinu | then we sort it out in ethereal |
| 20:02.35 | ManxPower | xai, if I can get it for under $250/month I can afford the line, and just offset some of the cost by selling internet access to the people here. |
| 20:02.57 | _Sam-- | justinu: i know this is a stupid question...but you unloaded the GXP2000s from ages ago? |
| 20:02.59 | ManxPower | and there are some CLECs with VERY good pricing |
| 20:03.12 | justinu | i still have 3 sitting here, sam |
| 20:03.34 | brettnem | justinu: at this rate, it'll be several gigs big |
| 20:03.41 | justinu | you take that much traffic? |
| 20:03.45 | justinu | you said it was light |
| 20:03.56 | brettnem | well there is liek maybe 5 calls up |
| 20:04.02 | *** join/#asterisk japerry (n=japerry@216.231.51.208) |
| 20:04.45 | brettnem | it's at 10M already.. been up for a minute or so.. |
| 20:04.58 | brettnem | needs to be up for 30.. ok.. so that's not a gig.. haha |
| 20:05.32 | justinu | if you can figure out what src/dst udp ports that call is on, you could just capture the rtp for that call |
| 20:05.39 | justinu | not sure if sip show channel will tell you that |
| 20:06.52 | djMax | anybody know if it's possible to SMS a number served by a PRI? |
| 20:06.57 | *** join/#asterisk Smi|k (i=Smilk@cpe-72-130-156-132.san.res.rr.com) |
| 20:07.03 | djMax | (with * on it) |
| 20:07.45 | brettnem | actually I'm at 30M now |
| 20:08.19 | brettnem | justinu: So after it does this, how do I filter this data to look for problems? |
| 20:08.46 | justinu | got ethereal installed on a workstation? |
| 20:09.59 | *** join/#asterisk bzbw (i=bwz@ip67-153-142-109.z142-153-67.customer.algx.net) |
| 20:10.37 | *** join/#asterisk pa (n=Paolo@unaffiliated/pa) |
| 20:10.43 | brettnem | justinu: I don't do X.. hmm. I can get it on my PC I guess. blah |
| 20:11.00 | brettnem | do these pcap files compress? |
| 20:11.07 | justinu | they should, a bit |
| 20:13.55 | justinu | if you can get it into ethereal, load it up and pick Analyze/RTP/Show all RTP streams |
| 20:14.45 | brettnem | justinu: ok we lost audio.. |
| 20:14.59 | [TK]D-Fender | brettnem : tethereal for CLI use |
| 20:15.23 | justinu | yeah, but tethreal doesn't have all that 1337 analyzer stuff |
| 20:16.04 | brettnem | yeah I don't think it can do that..but tell me if it can.. |
| 20:16.06 | brettnem | ugh 70 M! |
| 20:16.31 | justinu | what are you on dialup or something? |
| 20:16.34 | brettnem | no |
| 20:16.37 | brettnem | heh |
| 20:17.36 | paolob | guys, I can't get musiconhold on my asterisk... where can I enable it? |
| 20:18.43 | *** join/#asterisk CrummyGummy (n=wayne@dsl-145-117-03.telkomadsl.co.za) |
| 20:19.07 | *** join/#asterisk Dr-Linux (n=Linux@202.59.73.131) |
| 20:25.32 | *** join/#asterisk _Paulo_ (n=Paulo@c906d4b0.virtua.com.br) |
| 20:26.53 | *** join/#asterisk pjz (n=pj@zachs.place.org) |
| 20:27.19 | pjz | anyone know why I get like a second of silence when people first pick up when I call them from my asterisk system? |
| 20:27.35 | blitzrage | pjz: because it takes some time to answer the line |
| 20:28.12 | pjz | blitzrage: hrm, I didn't describe it correctly. I call someone, they pick up and say 'hello?' and I don't hear that. |
| 20:28.24 | pjz | blitzrage: but then like 1s later I can hear them fine. |
| 20:28.38 | paolob | Guys, what's this error: "res_musiconhold.c:421 spawn_mp3: Found no files in '/usr/share/asterisk/mohmp3' |
| 20:28.38 | paolob | May 16 16:26:16 WARNING[23382]: res_musiconhold.c:488 monmp3thread: Unable to spawn mp3player" ? thank you! |
| 20:28.40 | blitzrage | pjz: calling through an ITSP? probably a lag in the call setup time and the RTP getting through |
| 20:28.57 | blitzrage | paolob: it means res_musiconhold.so didn't find any files |
| 20:29.01 | pjz | blitzrage: no, not calling thru an ITSP |
| 20:29.11 | pjz | blitzrage: I've got a PRI into my asterisk box |
| 20:29.22 | paolob | blitzrage, I installed the asterisk-mp3 package, do I lack some more? |
| 20:29.24 | blitzrage | pjz: other end is another SIP phone? |
| 20:29.27 | pjz | blitzrage: and then gigE from there to the polycom 500 packages |
| 20:29.32 | pjz | blitzrage: nope, other end can be anything |
| 20:29.34 | blitzrage | paolob: check the dir it just told you where it can't find the files |
| 20:29.44 | paolob | blitzrage, it's void |
| 20:29.51 | blitzrage | pjz: ok... anything is pretty vague -- what are you connecting to? |
| 20:30.45 | obiwanmikenolte | echotraining? |
| 20:30.46 | pjz | blitzrage: I pick up my polycom 501 which is connected via gigE to my asterisk server that has a PRI card in it and dial any phone number. When the remote end picks up across the PSTN, I get a moment where I can't hear them. |
| 20:31.05 | pjz | obiwanmikenolte: ooooo, could be. Is there a way to adjust that? |
| 20:31.12 | blitzrage | obiwanmikenolte: good call |
| 20:31.16 | obiwanmikenolte | echotraining=n |
| 20:31.21 | obiwanmikenolte | http://kb.digium.com/entry/1/1/ |
| 20:31.24 | ManxPower | I wish I could find DECT phones in the USA |
| 20:32.10 | websae | blitzrage: need that phone? |
| 20:32.26 | blitzrage | websae: probably going to try and find it in Canada so I don't get screwed on duty and such |
| 20:32.35 | LL0rd | comon guys, can nobody help me? after Playback() I can't hear caller and the caller can't hear me |
| 20:32.36 | *** part/#asterisk sremington (n=sremingt@rrcs-24-123-247-29.central.biz.rr.com) |
| 20:33.00 | blitzrage | LL0rd: bitching like that doesn't make people want to help you |
| 20:33.35 | *** join/#asterisk mhnoyes (n=mhnoyes@dialup-4.246.21.50.Dial1.SanJose1.Level3.net) |
| 20:33.55 | obiwanmikenolte | LL0rd: I don't see where you described your problem |
| 20:34.11 | obiwanmikenolte | Who are you calling? |
| 20:34.12 | *** join/#asterisk MacDome (n=eseidel@A17-205-15-32.apple.com) |
| 20:34.46 | LL0rd | a call is comes in... |
| 20:34.55 | pjz | hrm, echotraining was set to 800 |
| 20:35.00 | pjz | what's a reasonable setting for it? |
| 20:35.09 | pjz | b/c I suspect that was ms, which would explain the silence |
| 20:35.14 | LL0rd | exten => 100,3,Dial(SIP/3000,40,trm) |
| 20:35.43 | brettnem | justinu: loading trace into ethereal now |
| 20:35.59 | LL0rd | doesn't work, if exten => 100,2,Playback(port29_weiter) is in the extensions.conf |
| 20:36.08 | obiwanmikenolte | That article says to start with 100 and increment by 100 but not to exceed 1200 |
| 20:36.45 | obiwanmikenolte | LL0rd: a call comes in from where? ARe you calling from outside of your box? |
| 20:36.53 | LL0rd | yes |
| 20:37.08 | *** part/#asterisk mhnoyes (n=mhnoyes@dialup-4.246.21.50.Dial1.SanJose1.Level3.net) |
| 20:37.09 | obiwanmikenolte | LL0rd: what kind of connection do you have to the outside? |
| 20:37.25 | brettnem | justinu: how do I do the RTP trace.. I don't see it in analyze where I used to see it. |
| 20:37.37 | ManxPower | LL0rd, you realize that using the "r" option to dial is like having a sign on your back that says "moron", right? |
| 20:37.43 | paolob | blitzrage, what should I have in /usr/share/asterisk/mohmp3 ? |
| 20:37.45 | r_evolution | shhhh Manx |
| 20:37.45 | justinu|laptop | analyze, RTP, show all streams |
| 20:37.51 | stack_ | I've been debugging my PRI and this message popped up: "NEW_HANGUP DEBUG: Calling q931_hangup, ourstate Disconnect Indication, peerstate Disconnect Request"... can anyone decypher this for me? |
| 20:37.56 | brettnem | justinu: it's not there anymore. |
| 20:38.03 | justinu|laptop | RTP option not there? |
| 20:38.05 | LL0rd | a fritz box is taking the calls and pass it via SIP to asterisk |
| 20:38.22 | brettnem | justinu: its in statistics now |
| 20:38.40 | blitzrage | paolob: ummmm... mp3s? |
| 20:38.43 | brettnem | justinu: oh crap |
| 20:38.44 | justinu|laptop | my bad, it's under statistics |
| 20:38.55 | brettnem | justinu: didn't get the SDP! son of a ... |
| 20:38.59 | justinu|laptop | that's ok |
| 20:39.00 | obiwanmikenolte | LL0rd: so you don't have any zaptel hardware? Try "modprobe ztdummy" from a shell |
| 20:39.01 | LL0rd | ManxPower i also tried it without any options, same |
| 20:39.27 | paolob | blitzrage, I think so, and I have the debian package asterisk-mp3 installed, I'd think it should install those mp3, but it doesn't... |
| 20:39.28 | justinu|laptop | well, it kinds sucks, because now we don't know what calls are associated with what invites |
| 20:39.44 | brettnem | right |
| 20:39.50 | brettnem | now it's happening on another server. |
| 20:39.59 | blitzrage | paolob: I don't use debian -- good luck! |
| 20:40.05 | LL0rd | obiwanmikenolte: no i have no zaptel hardware... |
| 20:40.12 | brettnem | it's not decoding these packets at all |
| 20:40.20 | r_evolution | booo @ mp3s ;x |
| 20:40.25 | r_evolution | raw! |
| 20:40.28 | obiwanmikenolte | LL0rd: You'll have to load ztdummy. Then try the call. |
| 20:40.48 | justinu|laptop | brettnem, you can force ethereal to decode the UDP packets as RTP |
| 20:41.01 | justinu|laptop | brettnem: is there any common network gear for the affected servers? |
| 20:41.09 | r_evolution | dammit justin |
| 20:41.09 | LL0rd | obiwanmikenolte: ok, mompl |
| 20:41.17 | r_evolution | you switched names again |
| 20:41.24 | *** join/#asterisk doughecka_ (n=Miranda@unaffiliated/doughecka) |
| 20:41.59 | *** join/#asterisk MGSsancho (n=user@adsl-67-126-128-145.dsl.irvnca.pacbell.net) |
| 20:43.24 | brettnem | justinu: sure there is. it's a big honkin router.. but there doesn't appear to be any problems with it |
| 20:49.10 | Hmmhesays | or you can use ethereal properly |
| 20:49.26 | Hmmhesays | and have it decode rtp packets as rtp |
| 20:50.56 | justinu|laptop | yes, you can force it to decode those packets as RTP |
| 20:52.30 | *** join/#asterisk [Airwolf] (n=airwolf@83.98.235.219) |
| 20:52.45 | *** join/#asterisk PoWeRKiLL (n=PoWeRKiL@84.205.154.179) |
| 20:56.14 | *** join/#asterisk swytch (n=ezcall@d213-103-231-75.cust.tele2.fr) |
| 20:56.16 | *** join/#asterisk [TK]D-Fender (n=joe@66.11.164.239) |
| 20:56.49 | *** join/#asterisk scubes13 (n=klanders@cpe-071-068-198-068.sc.res.rr.com) |
| 20:56.55 | *** part/#asterisk scubes13 (n=klanders@cpe-071-068-198-068.sc.res.rr.com) |
| 20:57.25 | paolob | Guys, what's this error: "res_musiconhold.c:507 monmp3thread: Request to schedule in the past?!?!" ? |
| 20:57.52 | *** join/#asterisk vechers (n=103326C9@64.61.117.138) |
| 20:58.23 | r_evolution|afk | it means God hates you paolob :) |
| 20:58.43 | paolob | r_evolution|afk, :-( but what is due to? |
| 20:59.08 | r_evolution|afk | i suppose an attempt to open up a moh channel that's already open? |
| 20:59.11 | obiwanmikenolte | Have you tried Google? |
| 20:59.16 | obiwanmikenolte | http://www.voip-info.org/wiki-Asterisk+Request+to+schedule+in+the+past |
| 21:00.22 | justinu|laptop | r_evolution|afk: go to bed!!! |
| 21:01.03 | r_evolution|afk | no kiddddiiiiinngg |
| 21:01.05 | r_evolution|afk | im out. p.z |
| 21:03.48 | *** join/#asterisk xcoyote (n=farfan@201.154.238.108) |
| 21:04.48 | *** part/#asterisk stack_ (n=stack@63.239.190.202) |
| 21:04.49 | Hmmhesays | shake that @$$ for me shake that @$$ for me |
| 21:05.08 | xcoyote | question: i have built my own program (app_myapp.c) to integrate to asterisk (Makefile) but i do not know where to include the file name in order to compile it when executing make command |
| 21:05.11 | xcoyote | any idea? |
| 21:05.48 | xp_prg | what is the asterisk manager api call to say dial a # then play a pre-recorded message? |
| 21:06.04 | Hmmhesays | originate |
| 21:06.06 | Qwell[] | xcoyote: don't need to, unless you need special compile options. Just drop it in the apps/ dir, and it'll compile |
| 21:06.47 | xcoyote | then i do not need to modify any file? |
| 21:06.52 | Qwell[] | shouldn't |
| 21:07.12 | xcoyote | because i can't see any line about compiling my app |
| 21:07.20 | xcoyote | when executing make |
| 21:07.20 | swytch | im a newbee to asterisk and will soon dive into the doc i can find. but does anyone have a hint for me, at this stage, of how best to manage a variable set of sip-callers. potencially for blocking their accounts (part of prepayment system). is it stable to periodically re-generate some file included from the sip-config ? |
| 21:07.42 | Qwell[] | xcoyote: 1.2? |
| 21:08.39 | swytch | or is there another better way to accomplish this. what should i be alert at in my future browsing of available docs ? |
| 21:09.33 | xcoyote | it is |
| 21:10.06 | Hmmhesays | there is no one answer to that question |
| 21:10.22 | Hmmhesays | theres about eleventy billion ways you can do that |
| 21:10.34 | X-Rob | swytch, dialplans AGI and realtime are the three asterisk concepts you should be reading up on |
| 21:10.51 | robl^ | and database ;-) |
| 21:11.58 | swytch | X-Rob: "dialplans AGI" ... ok. "realtime" is that specific enough? |
| 21:12.03 | *** part/#asterisk xcoyote (n=farfan@201.154.238.108) |
| 21:12.07 | X-Rob | swytch, three concepts. |
| 21:12.13 | X-Rob | not two |
| 21:12.22 | Smi|k | anyone know of any irc channels good for business owners who enjoy being on the edge of various technologies |
| 21:12.33 | xp_prg | ok I am looking at the originate asterisk manager api, I see how to dial a phone # but how do you then tell it to play a pre-recorded message? |
| 21:12.34 | *** join/#asterisk Gunnar (n=gunnar@34.80-203-209.nextgentel.com) |
| 21:12.34 | brettnem | justinu: I see 3 sequence errors.. that wouldn't cause 15 seconds of silence would it? |
| 21:12.38 | X-Rob | Smi|k, #imacrappybusinessowner |
| 21:12.44 | Smi|k | X-Rob, lol |
| 21:12.49 | justinu | no, definitely not |
| 21:12.55 | swytch | ok, so "dialplans" (i suppose thats the .conf files), AGI, /and/ that "realtime". ok. |
| 21:12.57 | Smi|k | not crappy, just *clueless* |
| 21:13.00 | X-Rob | perhaps #techsneeded |
| 21:13.05 | Smi|k | but at least I know it unlike 99.99% of others... |
| 21:13.06 | X-Rob | #positionsvacant |
| 21:13.13 | justinu | well... how many lost packets? |
| 21:13.42 | brettnem | 3 |
| 21:13.53 | xp_prg | Smi|k probably #linpeople but I don't know |
| 21:16.35 | swytch | X-Rob: thanx. that is very helpfull. tomorrow morning i know what to read. (-8 |
| 21:17.06 | X-Rob | swytch, read the asterisk book, and spend a week or so at voip-info.org |
| 21:17.31 | swytch | X-Rob: i dont have a week. i have until friday to put together the stuff. |
| 21:17.46 | X-Rob | swytch, you better be hiring a consultant quick smart then.l |
| 21:18.31 | X-Rob | because, really, you don't have anywhere NEAR enough time to learn how to do what you want to do. |
| 21:18.41 | *** part/#asterisk trbldwine (n=trbldwin@adam.ur.northwestern.edu) |
| 21:18.51 | swytch | X-Rob: and i read slowly. i will make it. the config generating idea is probably quicker that attaching to AGI or realtime for now. i will synch the config with my own db. |
| 21:19.27 | swytch | X-Rob: so i guess my dialplan hack is the best opt |
| 21:19.53 | X-Rob | swytch, yeah, except for the whole prepayment part. |
| 21:20.29 | swytch | X-Rob: i have a system that does all the logic and serves the clients app. i now has it hooked onto a voicemaster. its about to crash, so... |
| 21:20.58 | X-Rob | swytch, just because asterisk is libre, doens't mean it's easy. |
| 21:21.04 | X-Rob | in fact, libre usually means difficult. |
| 21:21.25 | Smi|k | I havent checked in awhile, has anyone linked asterisk to skype out yet? |
| 21:21.33 | Smi|k | ie. free calling |
| 21:21.40 | swytch | X-Rob: i know. and you now say what i was horribly afraid of was the truth )-8 is it really so? |
| 21:22.20 | X-Rob | swytch, as I said. spend a couple of weeks with some books and a couple of test asterisk servers, and a couple of soft phones and you'll pick up on how stuff works. If you don't have the time, hire a consultant. |
| 21:23.02 | syle | a couple weeks |
| 21:23.09 | syle | thats generous |
| 21:23.18 | X-Rob | doing exclusively nothing else |
| 21:23.45 | Nivex | It's taken me YEARS to become the mediocre Asterisk hacker I am today! :) |
| 21:23.49 | *** join/#asterisk nagl (n=nagl@86.59.54.237) |
| 21:24.05 | syle | might get an idea of how configs work and calls functions by that point, i doubt anything about making stuff dynamic or learning all the builtin apps as well |
| 21:24.24 | *** part/#asterisk gmaruz1 (n=gmaruzz@217-133-80-112.b2b.tiscali.it) |
| 21:24.33 | syle | or the real fun part, agi's :) |
| 21:24.41 | tainted_ | i love it when people boast how quickly they are going to learn asterisk |
| 21:25.01 | [TK]D-Fender | tainted_ : PM |
| 21:25.07 | adept_ | tainted in 15 minutes I am gonna know way more than you so keep chuckling! |
| 21:25.11 | brettnem | ~gwypf |
| 21:25.15 | jbot | [gwypf] Get What You Pay For - this channel is full of volunteers who are here to help you. However, we can't hold your hand. If you need a specific problem solved immediately, there is a list of for-hire consultants located at: http://www.voip-info.org/tiki-index.php?page=Asterisk+Consultants |
| 21:26.04 | adept_ | agi's... because cgi sucked too |
| 21:26.42 | paolob | Guys, experimenting with musiconhold(): asterisk executes it, but after saying " Started music on hold, class 'default', on channel 'SIP/oficina-7929'", it inmediately says "Stopped music on hold on SIP/oficina-7929". I have a mp3 in the mohmp3 dir, what's wrong? |
| 21:27.03 | paolob | Obviously I can't hear the mp3 music |
| 21:28.18 | *** join/#asterisk GerbilWrk (i=GerbilNu@65.88.144.41) |
| 21:28.20 | X-Rob | paolob, 1: _stop and restart_ asterisk. 2: check to make sure it's readable (check /var/log/asterisk/full) |
| 21:28.37 | *** join/#asterisk LL0rd (n=mb@devserver.info) |
| 21:28.48 | LL0rd | obiwanmikenolte: re |
| 21:29.10 | obiwanmikenolte | LL0rd: ? |
| 21:29.11 | Assid | argh |
| 21:29.13 | LL0rd | i load the module, but it doesn't help |
| 21:29.14 | Assid | stupid voicepulse |
| 21:29.27 | Assid | not letting me set the caller id on this particular account |
| 21:30.17 | obiwanmikenolte | LL0rd: You may need to restart ASterisk, but I can't remember. Guys? When you load/unload modules, do you need to restart Asterisk? |
| 21:30.47 | GerbilWrk | someone mind explaining what this would do? exten => 2600,5,System(/root/supportstatus techsupport specialrecording_on) |
| 21:31.19 | LL0rd | i had to rebuild my kernel, because of rtc part... then i load the module and started asterisk.. |
| 21:31.34 | xp_prg | what port does the manager api listen on? |
| 21:32.31 | *** join/#asterisk mjmac (n=mjmac@pdpc/supporter/active/mjmac) |
| 21:32.52 | paolob | X-Rob, now it works, thank you! |
| 21:34.15 | Dr-Linux | check someone >> http://pastebin.ca/56361 |
| 21:34.24 | Dr-Linux | why three timing ringing ringing? |
| 21:34.25 | obiwanmikenolte | GerbilWrk: System executes a shell command. What's at /root/support/status? |
| 21:35.44 | obiwanmikenolte | LL0rd: If you take the other phone out of the equation and just try to dial something that calls Playback(), does it work? Like, exten => 666,1,Playback(you-sound-cute) |
| 21:36.16 | LL0rd | yes |
| 21:36.27 | LL0rd | mom |
| 21:36.36 | obiwanmikenolte | LL0rd: Guess it's a codec problem? |
| 21:36.56 | LL0rd | i think it works..... |
| 21:37.05 | LL0rd | i have no two SIP phones.... |
| 21:37.22 | obiwanmikenolte | I'm not sure what the means |
| 21:37.22 | LL0rd | but i can heat the hold music |
| 21:37.29 | LL0rd | hear |
| 21:38.14 | obiwanmikenolte | I'm not sure what you're saying. When do you hear the hold music? |
| 21:38.32 | obiwanmikenolte | Does "no two sip phones" mean that there's only 1 sip phone? Is this a riddle? |
| 21:39.07 | Dr-Linux | anybody looked into my pastebin? |
| 21:39.41 | tamp4x | `/j #mysql |
| 21:39.45 | obiwanmikenolte | Doc: Yeah, you're going to have to tell us what the macro does |
| 21:39.46 | *** join/#asterisk MacDome (n=eseidel@A17-205-15-32.apple.com) |
| 21:39.51 | LL0rd | i have a AVM VoIP adapter called fritz box, one SIP phone is connected to it... |
| 21:40.45 | LL0rd | the box dials 100 if someone calls on ISDN |
| 21:40.47 | Dr-Linux | obiwanmikenolte: you are talking to me? : |
| 21:40.47 | obiwanmikenolte | LL0rd: Where are you from? |
| 21:40.57 | LL0rd | germany |
| 21:40.59 | obiwanmikenolte | Yeah, can I call you Doc? |
| 21:41.00 | obiwanmikenolte | Heh |
| 21:41.09 | Dr-Linux | obiwanmikenolte: sure |
| 21:41.28 | obiwanmikenolte | I'm not sure what AVM is |
| 21:41.43 | *** join/#asterisk lkimes (n=lkimes@rh-la-32-99.rhythm.com) |
| 21:41.57 | obiwanmikenolte | I guess no one else feels like responding, so I'm just going to lead as many of you astray as I can |
| 21:42.07 | Dr-Linux | obiwanmikenolte: macro has just 3 lines, first calls the particular extensions for 20 sec, then voicemail, then hangup |
| 21:42.19 | LL0rd | obiwanmikenolte: its a vendor |
| 21:42.22 | syle | anyone know where i can find source code to background application in tree ? |
| 21:42.43 | Dr-Linux | obiwanmikenolte: there is nothing wrong with macro, this server is running since 5 months and fine |
| 21:43.43 | obiwanmikenolte | LL0rd: Oh. I'm definitely thinking it's a codec issue, but I'd try adding that line to your dialplan and try dialing it from your phone |
| 21:44.19 | obiwanmikenolte | LL0rd: I don't really understand why you'd have a device between your SIP phone and your ASterisk server that isn't a switch/hub |
| 21:45.51 | LL0rd | obiwanmikenolte: ISDN -> VoIP Box -> Switch -> asterisk -> switch -> VoIP Box -> analog phone |
| 21:46.02 | xp_prg | ok I think I might be onto something, if I do Action: Originate, once it connects can I then send a PlayBack action to get it to send the pre-recorded message? |
| 21:46.30 | justinu|laptop | yes |
| 21:46.49 | justinu|laptop | but you're doing it the wrong way, just send it to an extension in your dialplan that plays the file |
| 21:47.16 | GerbilWrk | obiwanmikenolte, no clue, was from a sample extensions we saw for setting up the phone system to be called, a recording made, and then adding to the database so a check is made against it to either play the recording, or play the normal recording |
| 21:47.43 | LL0rd | obiwanmikenolte: if it is a codec problem, why should asterisk change the codec during a dialplan? |
| 21:47.46 | xp_prg | justinu|laptop I am still new to asterisk can you help me to understand what your saying more, I am not new to cti, but not on asterisk |
| 21:48.14 | *** join/#asterisk sevard (i=sev@merrill-49-29.resnet.ucsc.edu) |
| 21:48.56 | LL0rd | obiwanmikenolte: and the same works, if i use IAX instead of SIP |
| 21:49.03 | justinu|laptop | one of the parameters to the Originate action is the "extension" |
| 21:49.35 | xp_prg | so once it connects it immediately sends them to that extension? |
| 21:49.43 | justinu|laptop | yes |
| 21:50.02 | xp_prg | and then I define the extention in my extentions.conf? |
| 21:50.18 | *** part/#asterisk swytch (n=ezcall@d213-103-231-75.cust.tele2.fr) |
| 21:51.30 | *** join/#asterisk watchy (n=watchy@70.238.57.237) |
| 21:51.33 | obiwanmikenolte | GerbilWrk: So it probably just sets a value to on, then |
| 21:51.54 | watchy | anyone got the polycom sip 1.6.6 firmware? ive contacted voipsuppy but havent gotten a reply yet |
| 21:52.06 | obiwanmikenolte | Dr-Linux: I have no idea why it would say that it's ringing 3 times |
| 21:52.28 | Dr-Linux | obiwanmikenolte: i knew :) |
| 21:58.27 | *** join/#asterisk miguel3239 (n=chatzill@ns1.nashuacs.com) |
| 22:03.29 | *** join/#asterisk Brumle (n=brumle@brumle.com) |
| 22:03.45 | ManxPower | watchy, did you buy the polycoms from Avenue Computer Supply? |
| 22:04.18 | Assid | atacomm? |
| 22:04.30 | Assid | whats that site address again? |
| 22:04.41 | ManxPower | www.acs-comp.com |
| 22:04.46 | Assid | no no |
| 22:04.48 | Assid | the other one |
| 22:05.17 | ManxPower | I dunno, I wasn't talking about them |
| 22:05.52 | Assid | click on products |
| 22:05.54 | Assid | it dies out |
| 22:06.16 | ManxPower | you really have to call them |
| 22:06.33 | ManxPower | since Katrina they have pretty much stopped selling anything via their web site. |
| 22:06.48 | ManxPower | But if you get Polycom phones from them, they can provide the firmware |
| 22:06.59 | ManxPower | Well, they would call me and tell me to give you the firmware at least 8-) |
| 22:07.19 | Assid | how come |
| 22:07.33 | marl | can anyone tell me if there is a way to dump all curently set vars when in the * CLI? |
| 22:07.46 | ManxPower | marl, not that I know of |
| 22:07.57 | marl | :( ok thanks |
| 22:08.18 | Assid | http://voipstore.atacomm.com/Shops/Browse.aspx/27934028032-36931336704.htm <-- here these guys.. |
| 22:08.36 | marl | ok one other question :) is there a time varaible? that can be used to format a timestamnp for recording? |
| 22:08.59 | ManxPower | marl, see README.variables |
| 22:09.10 | Assid | but will keep acs in mind |
| 22:09.38 | Assid | err.. if you use a PoE cable.. do you just need a cable? or anything else? |
| 22:10.03 | ManxPower | just the cable and a PoE injector or PoE switch |
| 22:10.09 | marl | thanks ManxPower :) was sure id seen it before, but couldnt find it |
| 22:11.04 | *** join/#asterisk Cybertoy (n=maxim@dsl254-123-241.nyc1.dsl.speakeasy.net) |
| 22:11.32 | *** join/#asterisk robby (n=robby@adsl-ull-31-247.46-151.net24.it) |
| 22:11.49 | robby | hi |
| 22:11.52 | Assid | err.. i ave this one button.. on my poly301.. i want to assign it my contact list.. is it possible? |
| 22:12.36 | brockj49464_home | Is there any way to get the sip register= to have the LOCAL port number in it? |
| 22:15.41 | *** join/#asterisk Tigger-- (n=tigger@msabir.gotadsl.co.uk) |
| 22:17.15 | *** join/#asterisk MoRpHeUz (n=artur@tux14.ltc.ic.unicamp.br) |
| 22:17.34 | ManxPower | brockj49464_home, Why would you need that? |
| 22:17.50 | ManxPower | you should not need to specify a local port for a registration. |
| 22:18.16 | brockj49464_home | I have three accounts with the same provider and I need to seperate them out and not by DID |
| 22:18.16 | ManxPower | since ALL registration does it inform the far end what the IP address associated with that user/dsecret is |
| 22:18.19 | ManxPower | secret too |
| 22:18.31 | Malthus | hey |
| 22:18.33 | ManxPower | brockj49464_home, It sucks to be you. |
| 22:19.12 | Malthus | the voip-info wiki is a bit skippy on zaptel data support with kernel 2.6 |
| 22:19.23 | Malthus | anybody know if it works? |
| 22:19.51 | ManxPower | Malthus, data as in DACs or data as in PPP/HDLC? |
| 22:19.52 | brockj49464_home | Any other ideas? SPA's use different local ports 5060 and 5061 for example so how do I do that with *? |
| 22:20.30 | Malthus | HDLC I suppose |
| 22:20.35 | ManxPower | brockj49464_home, try putting it in the he [happysipentry] section |
| 22:20.43 | MoRpHeUz | anybody here knows any softphone that has conference support and is under GPL ? |
| 22:20.44 | ManxPower | Malthus, did you check the ZAPTEL readme? |
| 22:20.49 | Malthus | have a T1 with 14 voice chans and 10 64k data chans |
| 22:21.00 | Malthus | :) |
| 22:21.02 | Malthus | checking now |
| 22:21.48 | Malthus | actually the zaptel REAMDE doesn't mention anything |
| 22:21.49 | *** join/#asterisk Harlyman^ (n=skrot@torino.crystalnet.no) |
| 22:21.53 | ManxPower | Malthus, Ah. Can't help you with that. For ALL Voice/Data T-1s we have, we split them out using an Adtran |
| 22:22.08 | ManxPower | send the data into a router |
| 22:22.16 | MoRpHeUz | nobody ? =( |
| 22:22.21 | terrapen | `/p |
| 22:22.23 | *** part/#asterisk terrapen (n=cjs@166.70.183.109) |
| 22:22.24 | Malthus | and then use another channel bank to connect the analog chans to a router? |
| 22:22.37 | ManxPower | Malthus, Hmm? |
| 22:23.05 | ManxPower | T-1 -> Adtran, which spits out a V.35 for Data into the Cisco and DXS-1 to connect to Asterisk |
| 22:23.05 | Malthus | you split them to T1s or analog with the adtran? |
| 22:23.12 | Malthus | oh ok |
| 22:23.17 | ManxPower | if we need analog we put an Adtran Channel ban/ |
| 22:23.24 | ManxPower | the first Adtran is one of their TSUs |
| 22:23.40 | ManxPower | It's more expensive, but we like it |
| 22:23.49 | *** join/#asterisk driz (i=driz@spack.eblish.org) |
| 22:23.50 | Malthus | ever tried to do it with zaptel/asterisk? |
| 22:24.06 | ManxPower | Malthus, I don't like pain nearly that much |
| 22:24.22 | Malthus | I hear you |
| 22:24.29 | Malthus | well I'm gonna try |
| 22:24.44 | Malthus | hopefully I won't die in the process |
| 22:25.47 | ghenry | THat is SO cool! |
| 22:25.48 | *** join/#asterisk MoutaPT (n=MoutaPT@85.139.196.147) |
| 22:26.10 | *** join/#asterisk redondos_ (n=redondos@190.48.52.62) |
| 22:26.21 | ghenry | Just had our first sip call, routed to our other office via IAX2. No echo, clear as crystal! * is such a good product |
| 22:26.38 | Malthus | :) |
| 22:26.40 | Malthus | great |
| 22:26.46 | driz | good deal |
| 22:26.49 | ManxPower | ghenry, just wait until you have to interface to the PSTN |
| 22:26.54 | driz | i am reading AsteriskTFOT.pdf front to back |
| 22:26.54 | Malthus | lol |
| 22:27.02 | driz | i know absolutely nothing :) |
| 22:27.04 | Malthus | PSTN sucks |
| 22:27.13 | Malthus | source of ALL my problems |
| 22:27.16 | brockj49464_home | ManxPower as bindport? |
| 22:27.26 | ghenry | ManxPower: Yeah. Set up the conferencing and queues. All this for free as in speech! |
| 22:28.49 | MoutaPT | any channel for experience with OpenSER connected to Asterisk? |
| 22:30.23 | ManxPower | brockj49464_home, no, as in port= |
| 22:30.55 | *** join/#asterisk ad3 (n=adenardo@70.57.36.217) |
| 22:30.58 | ghenry | Is it true that IAX2 uses quite a bit more bw then SIP calls? I swa botu 15k/s for our call just now? |
| 22:31.04 | ghenry | saw about |
| 22:31.40 | ManxPower | no. |
| 22:31.44 | Malthus | it should be about the same bandwidth SIP or IAX2 |
| 22:31.59 | Assid | hrmm.i dont mind a non free DID.. say $2-$3/mo ? with unlimited incoming ? anyoneknow? |
| 22:32.07 | Assid | free DID's dont really last in quality |
| 22:32.21 | Malthus | unless you have several channels and use IAX2 trunking, in which case IAX2 will use less bandwidth |
| 22:32.45 | Malthus | there is a page with experimental data on the wiki at http:voip-info.org |
| 22:33.49 | ghenry | ah, ok. Well, it's like Phone<->SIP<->*<->IAX2<->*<->SIP<->Phone<-> |
| 22:33.56 | ghenry | minus the last <-> |
| 22:34.42 | brockj49464_home | manxpower: Neither worked... |
| 22:34.55 | *** join/#asterisk MacDome (n=eseidel@A17-255-96-84.apple.com) |
| 22:35.02 | ManxPower | brockj49464_home, I doubt it can be done |
| 22:35.18 | ManxPower | brockj49464_home, if you have NAT it's not going to do much good anyway |
| 22:36.27 | keyhack | Maybe not the best question to ask here, but, does anyone know the performance of Asterisk? I'm basically going to use it for people to call in and listen to recordings, I know the SIP overhead itself is minimal, but does anyone know how many simultaenous Playbacks it can do on an average CPU? |
| 22:36.39 | keyhack | could always measure it myself (which I will do) ;-) |
| 22:40.42 | brockj49464_home | no my box is on a direct static ip |
| 22:42.10 | MoutaPT | OpenSer and Asterisk in the same box, any one here with experience? |
| 22:42.33 | websae | anyone in here going to Cluecon in august? |
| 22:48.05 | ad3 | I just got asterisk compiled on my debian box. I have an FXO card and a sipura device. before starting asterisk I did a modprobe zaptel (successfully)..there is one more module I'm supposed to load, but can't seem to find it in my notes. any help? |
| 22:48.22 | obiwanmikenolte | wctdm |
| 22:48.28 | ad3 | obanta, thx |
| 22:49.04 | ad3 | obiwanmikenolte, thx |
| 22:51.16 | *** join/#asterisk Freman (n=twitsrus@jaguar.wbs.net.au) |
| 22:51.25 | DrPete | Is it possible to use asterisk revieve and make calls to skype users? |
| 22:51.43 | Freman | hey... extensions.ael (AEL in general) looks Damn sexy... how do I go about making asterisk use the ael instead of .conf? |
| 22:52.43 | obiwanmikenolte | Does anyone know whether I should set echocancelwhenbridged=yes on a PRI? It's my understanding that PRI should already have echo cancellation, so setting it to yes could actually make it worse. Yet, people swear to me that calls echo when I turn it off. Any opinions? |
| 22:54.45 | *** join/#asterisk pattieja (n=pcxuser@c-67-163-29-136.hsd1.il.comcast.net) |
| 22:55.47 | *** join/#asterisk Ariel_ (n=Ariel@dsl-20-177.cofs.net) |
| 22:59.19 | pattieja | hello |
| 22:59.19 | pattieja | anybody know what happened to GoIAX? |
| 22:59.19 | pattieja | website doesn't seem to be displaying anything |
| 22:59.19 | *** join/#asterisk |dennis| (i=dennis@200.32.215.84) |
| 23:01.34 | robl^ | GoneIAX? |
| 23:01.35 | pattieja | robl^: heh |
| 23:01.35 | *** join/#asterisk Winkie (n=urmom@cpc3-stre1-0-0-cust656.bagu.cable.ntl.com) |
| 23:02.54 | delmar | Hi everyone. I have a problem I am trying to track down which people are blaming on ztdummy. Is there anyone here that has good knowledge of the ztdummy function and can comment? The problem is that the main IVR is intermittantly "choppy" which most people would attribute to a lack of timing, but I am being told that ztdummy is the cause. So for the past weeks we have been running without it and for the most part this hasn't bee |
| 23:02.55 | delmar | n too bad but the chop does still come and go. Anyone have any comments about this? |
| 23:03.13 | *** join/#asterisk mhnoyes (n=mhnoyes@dialup-4.246.236.159.Dial1.SanJose1.Level3.net) |
| 23:08.49 | *** join/#asterisk malverian (n=malveria@gentoo/developer/malverian) |
| 23:12.44 | *** join/#asterisk malverian (n=malveria@gentoo/developer/malverian) |
| 23:13.43 | pjchilds | delmar: what type of channels are you using, what is the number of calls concurrently, what type of hardware spec, and version of linux / asterisk? |
| 23:14.01 | *** join/#asterisk malverian (n=malveria@gentoo/developer/malverian) |
| 23:15.01 | *** join/#asterisk dwclarkNU (n=dwclark@c-24-60-209-35.hsd1.ma.comcast.net) |
| 23:15.22 | dwclarkNU | is it possible to have two devices register on teh same extesion? so they would both ring when that extension is dialed? |
| 23:15.51 | dlynes_office | dwclarkNU: when one registers, it knocks the other off |
| 23:16.04 | dwclarkNU | dlynes_office: is it possible to disable this? |
| 23:16.18 | delmar | pjchilds, tried SIP/ulaw and SIP/g729 ... usually only 1-2 calls at a time. the box is a 2gig celeron, so plenty cpu.. its a redhat distro, asterisk 1.2.x zaptel 1.2.5 i think... need to grab my login info and login so i can confirm more details if u need. |
| 23:16.19 | dlynes_office | dwclarkNU: not afaik, but i'm far from being the sip expert, too |
| 23:16.53 | delmar | pjchilds, i guess what im after here is some comments regarding the fact that I'm being told.. this problem is being caused by ztdummy. |
| 23:17.45 | delmar | i can find nothing anywhere to backup this statement... everything i read says that ztdummy is required for timing... and under the 2.6.x kernel which this box does have... it will be using the high res timer provided by the rtc etc. |
| 23:18.53 | *** join/#asterisk malverian (n=malveria@gentoo/developer/malverian) |
| 23:19.14 | *** join/#asterisk generalhan (i=general_@ip67-90-64-2.z64-90-67.customer.algx.net) |
| 23:19.22 | dlynes_office | delmar: If there's a timing source available, asterisk will use it |
| 23:19.30 | dlynes_office | delmar: have you tried running without ztdummy? |
| 23:19.57 | generalhan | whats up all ! |
| 23:20.00 | dlynes_office | delmar: i know for a fact that the timing source does affect the ivr, even though it's not documented as such |
| 23:20.02 | generalhan | anyone working with .call files in here ? |
| 23:20.07 | delmar | dlynes_home, yes. chop issues still happen. No Asterisk wont use a timing source if its there unless u load the zaptel modules... if there is no zaptel hardware... you aparantly need ztdummy to provide asterisk with timing |
| 23:20.21 | *** join/#asterisk Telamon (i=telamon@blk-222-22-126.eastlink.ca) |
| 23:20.28 | *** join/#asterisk malverian (n=malveria@gentoo/developer/malverian) |
| 23:20.29 | delmar | dlynes_home, so.. no timing source at all.. and its crap yes? |
| 23:20.33 | dlynes_office | delmar: When my ztdummy driver was loaded and ztcfg failed on it, I couldn't even hear my ivr, much less have it be choppy |
| 23:20.50 | delmar | there is NO zaptel hardware in this box |
| 23:21.01 | delmar | dlynes_home, so u had issues with ztdummy did u log a bug? |
| 23:21.05 | dlynes_office | delmar: I had assumed that |
| 23:21.25 | dlynes_office | delmar: no. No point in logging a bug...I found out what the reason was for why ztdummy wasn't loading |
| 23:21.38 | dlynes_office | delmar: well...it was loadin g....ztcfg was just failing against it |
| 23:21.49 | delmar | ok |
| 23:21.55 | dlynes_office | delmar: it was a kernel/driver mismatch |
| 23:22.02 | delmar | dlynes_home, do u use ztdummy? do u have any zaptel hardware ? |
| 23:22.13 | dlynes_office | delmar: i use ztdummy on a number of systems |
| 23:22.21 | dlynes_office | delmar: i also run zaptel hardware on a number of systems |
| 23:22.27 | delmar | yep |
| 23:22.39 | dlynes_office | take your pick |
| 23:23.17 | *** part/#asterisk Telamon (i=telamon@blk-222-22-126.eastlink.ca) |
| 23:23.23 | dwclarkNU | is there a way to connect to voicemailmain and when it prompts you for the mailbox number, be able to hit # to accept your extension? similar to meridian? |
| 23:23.24 | dlynes_office | ztdummy on a 2.6 kernel is probably a more reliable timing source than an x100p card, I would imagine, though |
| 23:23.37 | *** join/#asterisk [Airwolf] (n=airwolf@83.98.235.219) |
| 23:23.57 | dlynes_office | dwclarkNU: on meridian, or norstar? |
| 23:24.45 | dlynes_office | dwclarkNU: well..not that it matters |
| 23:24.45 | dlynes_office | dwclarkNU: but why would you want to have asterisk act like a nortel system? |
| 23:24.48 | dwclarkNU | dlynes_office: what do you mean? i've used a meridian traditional pbx and thats how it works, but i was wondering if it was possible for asterisk to do that |
| 23:24.50 | *** part/#asterisk mhnoyes (n=mhnoyes@dialup-4.246.236.159.Dial1.SanJose1.Level3.net) |
| 23:25.02 | dlynes_office | dwclarkNU: anything's possible, if you modify the C code |
| 23:25.27 | dwclarkNU | dlynes_office: true, i was just wondering if it was a feature that could be turned on |
| 23:25.36 | dlynes_office | dwclarkNU: not that i'm aware of, no |
| 23:25.54 | dlynes_office | dwclarkNU: you can get it to act like a cics or a mics, though |
| 23:25.55 | dwclarkNU | dlynes_office: i want to be able to check other voicemails, but dont want to ahve ot type if my extension every time i check mine |
| 23:26.14 | dlynes_office | dwclarkNU: i.e. startalk flash/norstar voicemail |
| 23:26.33 | dwclarkNU | dlynes_office: yes |
| 23:26.43 | dlynes_office | dwclarkNU: well, actually |
| 23:26.48 | dlynes_office | dwclarkNU: there is one way |
| 23:27.31 | dlynes_office | Background(please-press-pound-to-enter-your-mailbox-or-mailbox-number-to-enter-someone-elses) |
| 23:27.40 | robl^ | ok.. what did I miss? I used to manage a fairly large Norstar switch and startalk system. what is it you want asterisk to do like it? |
| 23:28.04 | dlynes_office | #,1,VoicemailMain(uMyMailBoxNumber) |
| 23:28.08 | dlynes_office | #,2,Hangup |
| 23:28.40 | dwclarkNU | robl^: i want to be prompted for my mailbox number and be able ot hit pound to enter mine |
| 23:28.40 | dlynes_office | _X.,1,VoicemailMain(${EXTEN}) |
| 23:28.43 | dlynes_office | _X.,2,Hangup |
| 23:28.48 | pjchilds | delmar: I doubt its a ztdummy issue... sounds like a load'ish issue... check DMA on your harddrives, then check sar, load, etc... |
| 23:29.21 | dwclarkNU | dlynes_office: nice work around |
| 23:29.32 | robl^ | dwclarkNU: that can be done easily in extensions.conf just use about 5 lines of a dialplan |
| 23:29.42 | delmar | pjchilds, sar? |
| 23:30.07 | pjchilds | delmar, man 'sar' .. its part of the sysstat package and can track performance over time... |
| 23:30.32 | delmar | pjchilds, the box hardly does anything. passes a couple of calls at most. |
| 23:31.32 | Freman | I need to confirm that the phone number caller*id has presented is the number that the caller wants a callback on, and ask them for a new number if it's not... is there any built in function to do this? (before I go re-inventing the wheel again (c: ) |
| 23:31.35 | robl^ | sounds like my cowarking.. he doesn't do much.. just sits there idle.. and passes gas |
| 23:32.06 | pjchilds | delmar, then I would check DMA on your harddrives, and perhaps ensure your recordings you use are transcoded already into the appropriate format (so you don't need to do it on the fly...) |
| 23:36.41 | delmar | pjchilds, ok. gonna login and take a look at a couple of things now then. |
| 23:40.55 | *** join/#asterisk MacDome (n=eseidel@A17-255-96-84.apple.com) |
| 23:42.33 | delmar | pjchilds, using_dma = 1 (on) |
| 23:42.55 | *** join/#asterisk Lino` (n=Lino@i577BD937.versanet.de) |
| 23:43.12 | websae | anyone going to Cluecon at all??? anyone? |
| 23:43.23 | websae | I mean does anyone have any idea about freeswitch? |
| 23:43.28 | delmar | pjchilds, /dev/hda: |
| 23:43.29 | delmar | <PROTECTED> |
| 23:43.29 | delmar | <PROTECTED> |
| 23:43.38 | delmar | drive looks about as good as any |
| 23:43.52 | websae | Question of the night --- Are you going to Cluecon!? |
| 23:44.09 | Lino` | answer of the night: no |
| 23:44.16 | file | if I'm told to go, yes |
| 23:44.18 | websae | Lino: why? |
| 23:44.18 | file | if I'm not, no |
| 23:44.20 | *** join/#asterisk malverian (n=malveria@gentoo/developer/malverian) |
| 23:44.28 | websae | file: hehehe, I am sure you're gonna go |
| 23:44.36 | file | don't bet on it |
| 23:44.39 | Lino` | there were times when i flew 10000 miles for a lan party |
| 23:44.42 | file | :D |
| 23:44.45 | websae | file: I don't gamble |
| 23:44.47 | Lino` | but i'm not going that way for a freakin con |
| 23:44.47 | Lino` | :D |
| 23:45.15 | Lino` | although the flight would be free |
| 23:45.17 | dlynes_office | I'm not |
| 23:45.20 | websae | Lino: it's revolutionizing the way VoIP will be handled, free switch that is |
| 23:45.33 | dlynes_office | but, isn't cluecon about voip in general, not freeswitch, specificallY? |
| 23:45.43 | Lino` | :-P |
| 23:45.49 | Lino` | besides, i dont have the time |
| 23:45.53 | Lino` | :D |
| 23:45.56 | websae | Freeswitch will be an active part of it |
| 23:46.03 | dlynes_office | yeah, i realize that |
| 23:46.04 | Lino` | i'm a busy man ;) |
| 23:46.11 | websae | but yep...the big picture = VoIP |
| 23:46.11 | dlynes_office | but freeswitch won't be the only exhibitor |
| 23:46.25 | websae | Lino: you can have my free plane ticket |
| 23:46.26 | dlynes_office | digium will also be there propping up asterisk afaik too, right? |
| 23:46.50 | Lino` | (although that is not really credible, because whoever does IRC must be unemployed, stupid or have some strange psychic disease) |
| 23:46.52 | dlynes_office | and sangoma will probably be there propping up both |
| 23:46.53 | Lino` | :D |
| 23:47.01 | Lino` | just dont mention sangoma |
| 23:47.06 | Lino` | and if theres a guy from poland |
| 23:47.09 | Lino` | greet him |
| 23:47.09 | Lino` | :D |
| 23:47.11 | Lino` | he hates me |
| 23:47.37 | delmar | pjchilds, checked IVR sounds and they are all .gsm |
| 23:47.55 | dlynes_office | man |
| 23:47.58 | dlynes_office | this heat is killing me |
| 23:48.07 | dlynes_office | it must be about 85 or 90 outside today |
| 23:48.21 | *** part/#asterisk dwclarkNU (n=dwclark@c-24-60-209-35.hsd1.ma.comcast.net) |
| 23:48.25 | websae | haha |
| 23:49.03 | pjchilds | delmar: what does hdparm -i |
| 23:49.03 | pjchilds | /dev/IDE Device say... |
| 23:49.08 | *** join/#asterisk |dennis| (i=dennis@200.32.215.84) |
| 23:49.22 | *** join/#asterisk mog_dc (n=achika54@64.241.203.209) |
| 23:50.36 | delmar | pjchilds, little bit more than i would want to paste on here .. anything specific im lookin for in that lot? |
| 23:50.47 | *** join/#asterisk malverian (n=malveria@gentoo/developer/malverian) |
| 23:50.57 | websae | ~pastebin |
| 23:50.58 | jbot | i guess pastebin is a place to paste your stuff without flooding the channel - try http://pastebin.com/, or http://pastebin.ca, or http://channels.debian.net/paste, or for #oe use http://oe.pastebin.com |
| 23:51.47 | delmar | pjchilds, http://pastebin.ca/56433 |
| 23:53.22 | pjchilds | delmar: hmm... well udma5 is set... |
| 23:53.42 | delmar | yep |
| 23:53.49 | pjchilds | delmar: so I guess if you have choppyness perhaps it is not the PC :) |
| 23:54.00 | delmar | im thinking less likely |
| 23:54.25 | delmar | pjchilds, i have some other things to try so I will go do that and see what happens over the next few days. |
| 23:54.36 | pjchilds | delmar: good luck ! |
| 23:54.48 | delmar | I honestly thing its a chop issue caused upstream.. by the provider or beyond. |
| 23:55.02 | *** join/#asterisk MacDome (n=eseidel@A17-255-96-84.apple.com) |
| 23:55.59 | pjchilds | delmar: well if you have a large amount of money then you could put hammer VOIP analayser over the link... |
| 23:56.19 | generalhan | anyone working with .call files in here ? |
| 23:56.20 | pjchilds | delmar: or the latest version of ethereal have some interesting VOIP stuff (looks a jitter etc...) |
| 23:56.31 | pjchilds | generalhan: yup... |
| 23:57.04 | generalhan | pjchilds: im having serious issues with mine ... it wont let me "move" the file into the dir. it says permission denied in my httpd error logs |
| 23:57.39 | generalhan | pjchilds: but i did a chown apache for the /var/spool/asterisk/ dir and the /var/www/html/ dir (where the script IS) and still permissions denied ? |
| 23:57.50 | generalhan | did you have anything like that happen to you ? |
| 23:58.43 | delmar | pjchilds, i think we can get to a point where we have covered all bases, and can blame the upstream provider. |
| 23:59.46 | pjchilds | generalhan, nope... more of a apache type problem ? |
