| 00:00.39 | drmessano | We're talking about the same price and feature space |
| 00:00.46 | awannabe | ohh ok |
| 00:00.48 | drmessano | OpenWRT blows away a PIX box |
| 00:00.58 | awannabe | was gonna say, umm a 6509 is not like a linksys guys :) |
| 00:01.01 | drmessano | and its $400 cheaper |
| 00:01.18 | drmessano | Ummm.. I dont think anyone meant THE WHOLE PRODUCT LINE |
| 00:01.27 | awannabe | haha, just checking |
| 00:01.30 | drmessano | I dont think Linksys makes carrier grade equipment |
| 00:01.36 | drmessano | yeah, we are all dumbasses |
| 00:01.42 | awannabe | wasnt saying that |
| 00:01.55 | drmessano | just checking :) |
| 00:02.05 | jameswf | ~nowwhat |
| 00:02.06 | jbot | So you just installed asterisk and arent sure what to do now? visit http://www.a1b2c3.com/suilodge/metfun1.htm |
| 00:02.35 | drmessano | We're in the process of replacing Cisco VPN with Juniper stuff |
| 00:02.40 | jameswf | ahh 5pm run away |
| 00:02.42 | drmessano | ZOMFG it's NIGHT AND DAY |
| 00:02.51 | *** part/#asterisk jameswf (n=james@dsl093-157-131.phx1.dsl.speakeasy.net) |
| 00:02.55 | *** join/#asterisk coppice (n=chatzill@175.203.17.210.dyn.pacific.net.hk) |
| 00:03.01 | Qwell | Juniper++ |
| 00:03.26 | lmadsen | heck ya... but holy crap do you need to know a lot to get juniper certs |
| 00:03.32 | lmadsen | you actually need to know what you're doing :) |
| 00:03.34 | awannabe | lol |
| 00:03.37 | JT | are they really that hard? |
| 00:03.39 | drmessano | Every Cisco windows VPN client I have uninstalled has either screwed the IP stack to where I had to run a Winsock reset on it, or BSOD'ed the machine |
| 00:03.44 | *** join/#asterisk nighty^ (n=nighty@210.188.173.245) |
| 00:03.59 | drmessano | or both |
| 00:04.29 | drmessano | Oh, and don't touch a PIX box between October and March |
| 00:04.41 | drmessano | Static from 18 miles away will fry one |
| 00:04.45 | JT | they have a period then? |
| 00:04.58 | drmessano | No, it's dry :) |
| 00:05.06 | drmessano | Static season, as I call it |
| 00:05.27 | *** join/#asterisk riddlebox (n=james@75-132-225-75.dhcp.stls.mo.charter.com) |
| 00:07.10 | JT | is that the driest time of the year in the us? |
| 00:07.21 | drmessano | yes |
| 00:07.56 | drmessano | I usually spray all the carpets down with my antistatic mix |
| 00:08.38 | drmessano | One capfull of fabric softener to 1 gallon of water, throw it in a garden sprayer, and hose down everything |
| 00:08.56 | drmessano | usually keeps the static down |
| 00:09.20 | Qwell | drmessano: why do I not believe you? |
| 00:09.34 | drmessano | Im dead serious |
| 00:09.47 | drmessano | Use bounce or something like that, in the bottle |
| 00:09.59 | drmessano | Just a capful to a gallon |
| 00:10.16 | drmessano | Trick I learned many years ago.. |
| 00:11.33 | drmessano | Im guessing the ultra would need less.. half-capful |
| 00:11.34 | drmessano | That concentrate stuff |
| 00:12.07 | drmessano | You can get the store brand that comes in a cardboard "milk carton" type container as a "refill" and its dirt cheap |
| 00:14.53 | *** join/#asterisk techie (n=techie@adsl-76-214-27-74.dsl.lsan03.sbcglobal.net) |
| 00:14.55 | drmessano | They say you can tuck a dryer sheet in the waistband of your pants and keep the static down too.. But then you're walking around all day with a "Mountain Fresh" dryer sheet tucked in your pants... not cool |
| 00:16.25 | drmessano | To eliminate static shock when you walk across your carpet, spray the carpet with a fabric softener solution. Dilute 1 cup softener with 2 1/2 quarts (2.5 liters) water; fill a spray bottle and lightly spritz the carpet. Take care not to saturate it and damage the carpet backing. Spray in the evening and let the carpet dry overnight before walking on it. The effect should last for several weeks. |
| 00:16.32 | drmessano | Ive used much less |
| 00:17.14 | drmessano | half cup to a gallon, I guess |
| 00:17.16 | coppice | just move to a jungle |
| 00:17.24 | *** join/#asterisk b1ch0 (i=b1ch0@static-200-105-150-60.acelerate.net) |
| 00:18.10 | coppice | you'll be fine until MacDonalds clears it to make cheap burgers |
| 00:18.36 | drmessano | If you use a bug sprayer, like you get at Wal Mart, you can make the mist fine enough to not worry about it soaking the carpet |
| 00:18.41 | drmessano | ha |
| 00:18.47 | drmessano | Good ole McDonalds |
| 00:19.38 | b1ch0 | hi, just updated to 1.4.17 .... but now *8 function core to hunt calls is not working anymore |
| 00:20.09 | b1ch0 | any idea? ** plus extension works fine |
| 00:20.54 | drmessano | Is this a Trixbox? |
| 00:21.34 | b1ch0 | hi dr, yes still with this F#$% box |
| 00:21.50 | drmessano | You're asking in the wrong place :) |
| 00:22.36 | b1ch0 | yes i know, i know .... but still thinking that best people is here and not inthe others channels |
| 00:22.41 | JT | wow i've never taken any static countermeasures like that |
| 00:22.57 | drmessano | JT: We have bad problems here with it |
| 00:23.11 | draygon2 | anyone here use route5060.com? |
| 00:23.22 | JT | get an antistatic wrist strap and touch the connector for that to doorhandles first |
| 00:23.25 | JT | ;) |
| 00:23.29 | coppice | it depends where you live. we have 90%+ humidity for 9 months a year, so static is not something we see too much |
| 00:23.30 | drmessano | lol |
| 00:24.46 | drmessano | My friends townhome has static problems so bad that we grounded his equipment shelves with 4 inch strap, sprayed the carpets, and all sort of crap. |
| 00:24.53 | *** join/#asterisk Jaxxan (n=Jaxxan@202.70.125.111) |
| 00:24.55 | drmessano | He was losing switches left and right |
| 00:25.15 | drmessano | Even now you have to ground yourself to the shelf before you pick anything off it |
| 00:25.15 | JT | go fibre |
| 00:25.15 | Jaxxan | hey guys |
| 00:25.21 | drmessano | heh |
| 00:25.24 | *** join/#asterisk joat (n=joat@ip70-160-147-169.hr.hr.cox.net) |
| 00:25.25 | drmessano | Thats practical |
| 00:25.33 | JT | sure why not :) |
| 00:25.44 | drmessano | I'm gonna need some VC |
| 00:25.55 | BBHoss | drmessano, thats why you should rip the carpet up and get conductive tiling |
| 00:26.16 | drmessano | My house isn't going to have carpet anyway.. heh |
| 00:26.24 | coppice | fibre is glass. it can hold a lot of static :-) |
| 00:26.29 | denon | electrify the flooring of the whole house, 12VDC |
| 00:26.36 | BBHoss | i hate carpet, its always a liability |
| 00:27.30 | drmessano | They traded out some carpet at work a few years back.. and I am pretty sure this stuff is made of recycled balloons |
| 00:27.41 | denon | hehe |
| 00:27.48 | denon | esd-wise? |
| 00:27.56 | drmessano | yeah |
| 00:28.20 | drmessano | If I lick a 9v battery, I can lay flat on it and static can lift me 6 inches high |
| 00:28.21 | denon | I can't remember the details, but there are lots of carpets out there specifically designed to actually reduce esd as you walk on em |
| 00:28.23 | drmessano | no, but close |
| 00:28.46 | denon | we've got it in the offices and stuff |
| 00:29.24 | drmessano | The only thing the carpet we got was resistant to is.. being good carpet |
| 00:29.34 | *** join/#asterisk Igbothom_III (n=Hilton@office.quarkgroup.com.au) |
| 00:29.34 | denon | hah |
| 00:29.50 | drmessano | It looks like church carpet |
| 00:29.55 | drmessano | I swear to god.. same color |
| 00:29.59 | drmessano | Burgundy |
| 00:30.14 | *** join/#asterisk Igbothom_III (n=Hilton@office.quarkgroup.com.au) |
| 00:30.41 | *** join/#asterisk styelz (n=yoohoo@m0o0.mooo.com) |
| 00:30.48 | denon | I like my new carpet at home .. |
| 00:30.59 | denon | dark tan with brown threads |
| 00:31.03 | denon | kind of a mixture of stuff in between |
| 00:31.05 | drmessano | Cool |
| 00:31.15 | denon | looks really nice, but more importantly, you could lose like a whole cat in it |
| 00:31.20 | denon | and it'd not make a stain |
| 00:31.27 | drmessano | heh |
| 00:31.27 | denon | stuff always looks clean |
| 00:31.34 | drmessano | Stain resistant FTFW |
| 00:31.40 | drmessano | Well |
| 00:31.41 | denon | well, kinda by design |
| 00:31.44 | drmessano | Stain camo |
| 00:31.46 | drmessano | yeah |
| 00:31.48 | denon | but I also put that rubberized stuff underneath |
| 00:31.53 | denon | so that liquid wouldnt soak into the padding |
| 00:31.57 | Jaxxan | so i'm currently running 1.2.16 on my PBX. I'm building a sip gateway now, so I expect I should start it with 1.4.18. there's probably significant changes between these two versions that I'm not aware of. My main question is: Is there a 32 or 64-bit version? or does it matter? |
| 00:32.19 | denon | and I put a layer of anti-bacteria stuff just under it .. which is cool, active bacteria that eat odors and stuff |
| 00:32.24 | drmessano | no, yes, yes, no, no |
| 00:32.27 | denon | er anti-odor/stain/whatever |
| 00:32.50 | drmessano | Wait |
| 00:32.57 | *** join/#asterisk b1shop (n=b1shop@c-71-194-197-216.hsd1.il.comcast.net) |
| 00:33.07 | drmessano | Yes, yes, yes, no, no |
| 00:33.11 | denon | didn't you give him more answers than questions? |
| 00:33.12 | denon | hehe |
| 00:33.21 | drmessano | yes |
| 00:33.29 | drmessano | or no |
| 00:33.36 | Jaxxan | were those answers for me ? |
| 00:33.39 | drmessano | I love those questions.. |
| 00:33.49 | denon | yes |
| 00:33.53 | drmessano | Denon, do you want the steak, or do you want the chicken? |
| 00:33.54 | drmessano | 'yes" |
| 00:33.57 | denon | Jaxxan: yes you should use 1.4 |
| 00:34.05 | denon | Jaxxan: yes there are major differences |
| 00:34.13 | Jaxxan | 1.4 used the IEL right ? |
| 00:34.15 | denon | Jaxxan: no, you don't need a platform-specific build |
| 00:34.32 | denon | and yes there are probably differences you aren't aware of |
| 00:34.34 | Jaxxan | sorry, ael |
| 00:34.36 | denon | if that was a question |
| 00:34.44 | denon | Jaxxan: why do you want ael? |
| 00:34.56 | Jaxxan | i dont, i'm perfectly content with the context model |
| 00:35.07 | denon | you have some sort of wish to inflict pain on yourself? |
| 00:35.12 | denon | ah ic |
| 00:35.12 | denon | good man |
| 00:35.16 | drmessano | Im skipping 64 bit, waiting for 128 bit |
| 00:35.32 | denon | yeah, 'cause we all have the need to access like 500TB of ram |
| 00:35.51 | denon | though, it would be nice .. |
| 00:35.52 | drmessano | 618TB actually |
| 00:35.57 | denon | close 'nuff |
| 00:36.01 | drmessano | Wait |
| 00:36.08 | drmessano | We're up to 619TB now |
| 00:36.08 | husimon | hey drmessano sup |
| 00:36.17 | *** join/#asterisk ming_zym (n=ming_zym@nat/yahoo/x-c9c4acd858969897) |
| 00:36.17 | drmessano | sup husimon |
| 00:36.21 | denon | if you can find me a chipset that'll do it .. |
| 00:36.28 | drmessano | lol |
| 00:36.29 | husimon | wtf do you have that uses 500tb of ram? |
| 00:36.42 | drmessano | Firefox for Windows |
| 00:36.48 | husimon | LOL |
| 00:37.12 | drmessano | Can I get a "hell yes" please? |
| 00:37.18 | husimon | i've run xmms for 3 months and it had a memory leak |
| 00:37.28 | husimon | was using 1.5gb of ram |
| 00:37.34 | drmessano | hah |
| 00:37.44 | Jaxxan | ok, so i'm building a sip server with 8gb ram and a 2.3ghz quadcore. that should be enough to process about 3000 calls an hour via sip to a Cisco AS5350 yeah ? |
| 00:37.49 | b1ch0 | guys, still bothering all of you .... anyone got *8 call pickup problem afe=ter upgrading to 1.4.17 ? |
| 00:37.51 | husimon | My boxen was all slow and i was like wtf, xmms is using 80% cpu and 1.5gb of ram |
| 00:38.02 | husimon | Jaxxan, you only have one ? |
| 00:38.17 | Jaxxan | one what ? |
| 00:38.19 | drmessano | I had Firefox up to a GB once |
| 00:38.20 | husimon | server? |
| 00:38.20 | denon | Jaxxan: well, you plan on g729'ing all the calls? |
| 00:38.23 | BBHoss | Jaxxan, how many concurrent calls |
| 00:38.38 | Robba | hi guys |
| 00:38.40 | drmessano | I have 4GB RAM, and for some reason Firefox using 1GB made the system unusable |
| 00:38.41 | husimon | I mean you do want a backup.... |
| 00:39.09 | Robba | in the queues.conf file, what specifies it to play MOH after calling in? |
| 00:39.40 | Jaxxan | I have another server that i'll build as a backup. i expect like. maybe 200-400 concurrent calls. |
| 00:39.49 | Jaxxan | think i need more ram ? |
| 00:40.04 | BBHoss | anything over 200 is iffy, i would cluster some servers together |
| 00:40.05 | Jaxxan | i haven't decided on a codec yet. |
| 00:40.28 | denon | ulaw |
| 00:40.33 | denon | codec of champions |
| 00:40.37 | drmessano | alaw |
| 00:40.39 | drmessano | heh |
| 00:40.44 | denon | darn foreigners |
| 00:40.48 | Jaxxan | well, i'm gonna single server it for now, my call volume may not be that high initially. if it becomes a problem then i'll consider clustering |
| 00:40.54 | BBHoss | Jaxxan, http://www.voip-info.org/wiki-Asterisk+dimensioning |
| 00:40.59 | BBHoss | that will help you out |
| 00:41.02 | drmessano | I would love to be a douche and insist on alaw |
| 00:41.03 | Jaxxan | sweet |
| 00:41.05 | chavigny | whats the best asterisk graphing application?, perhaps like to show demographics |
| 00:41.08 | drmessano | "But this is the US?" |
| 00:41.10 | drmessano | "So" |
| 00:41.19 | denon | drmessano: but ulaw rox your sox and you know it |
| 00:41.30 | drmessano | ulaw is how we roll |
| 00:41.58 | drmessano | "I found ulaw and, ulaw won.." |
| 00:42.18 | drmessano | I finally got speex going |
| 00:42.20 | drmessano | Its pretty decent |
| 00:42.25 | denon | yeah, its ok |
| 00:42.29 | denon | it's kinda nice for flaky stuff |
| 00:42.33 | drmessano | yeah |
| 00:42.39 | drmessano | Like X-Lite |
| 00:42.40 | Robba | lol |
| 00:42.44 | denon | where you'd use 729 due to bandwidth constraints, but ulaw due to poor routing .. |
| 00:42.47 | denon | speex is a good mix |
| 00:43.08 | denon | it adapts pretty well |
| 00:43.14 | drmessano | My PAP2 don't speak Speex :( |
| 00:43.21 | denon | nothin hardware does |
| 00:43.30 | denon | idefisk/zoiper does though |
| 00:43.42 | drmessano | X-Lite seems to work well with Speex |
| 00:43.50 | denon | I havent used x-line in a few years |
| 00:43.56 | denon | last time I did it was needlessly complex to configure |
| 00:44.01 | drmessano | They took GSM out apparently, whch was good |
| 00:44.02 | denon | idefisk is like 3 clicks |
| 00:44.04 | denon | and iax works |
| 00:44.07 | drmessano | X-lite is better than it was |
| 00:44.07 | denon | iax2 |
| 00:44.09 | drmessano | 2.0 sucked |
| 00:44.13 | drmessano | 3.0 is way better |
| 00:44.21 | Robba | my pap2 loves the ulaw... |
| 00:44.23 | SwK | G723.1 > * |
| 00:44.28 | drmessano | It will do ilbc and speex |
| 00:44.39 | denon | drmessano: played much with zoiper? |
| 00:44.44 | denon | its just so clean and simple |
| 00:44.59 | drmessano | I head "Zoiper" sucks.. and to stick to old Idefisks |
| 00:45.11 | drmessano | I dunno.. Ive used idefisk |
| 00:45.15 | drmessano | Its decent |
| 00:45.25 | drmessano | heard* |
| 00:45.26 | denon | well, zoiper has some stuff disabled, for business edition only |
| 00:45.33 | drmessano | ah |
| 00:45.39 | denon | but .. it also has echo can |
| 00:45.43 | denon | and better jitter stuff |
| 00:45.53 | denon | and as I understand it, does more with the fun new stuff in iax2 for 1.4 |
| 00:46.11 | denon | Ive not spent enough time with zoiper yet, but Ive just set a few tests people up on it, that were on idefisk |
| 00:46.20 | denon | people in weird parts of the world, as far as Internet goes |
| 00:46.34 | drmessano | yeah |
| 00:46.46 | JT | SwK: why G.723.1? |
| 00:47.09 | Qwell | JT: because it's like .5kb/hour |
| 00:47.11 | denon | drmessano: business edition also has an option for 729 |
| 00:47.17 | drmessano | I found X-Lite 3.x to be VERY easy to config.. so its been my choice and what ive gotten others to use on my systems |
| 00:47.22 | denon | but it does speex, ilibc, gsm, ulaw, alaw, etc free |
| 00:47.34 | denon | drmessano: but no iax2 support? |
| 00:47.47 | drmessano | IAX on a softphone doesnt impress me |
| 00:47.48 | denon | zoiper is literally server IP, user, pass |
| 00:47.58 | denon | caller ID name and number optional |
| 00:48.23 | denon | drmessano: well, if you have users on laptops, going to odd networks that may bork your sip stuff .. |
| 00:48.26 | denon | you might like iax a bit more |
| 00:48.45 | denon | that and I'd rather not open up any more sip stuff than I have to |
| 00:48.47 | drmessano | I really haven't had any of those problems.. So SIP has been fine |
| 00:48.51 | *** join/#asterisk PepOSX (n=angeldav@190.72.132.44) |
| 00:48.52 | denon | iax2 has a bit lower exposure as far as exploits |
| 00:49.09 | denon | and less chance that nosy admins are listening in, if they do it outside the tunnels |
| 00:49.40 | denon | shrugs, guess I should spend a little time with xlite again |
| 00:50.01 | drmessano | I dont doubt Zoiper is good |
| 00:50.09 | Jaxxan | i'll bump it up to 16gb of ram and drop in another quad core processor if it starts to buckle |
| 00:50.14 | drmessano | and I dont doubt the older Xlite sucked |
| 00:50.15 | drmessano | it did |
| 00:50.29 | drmessano | I just have been happy enough with the newer xlite that ive not had a reason to switch |
| 00:50.56 | denon | but no speex? |
| 00:51.04 | denon | oh, no gsm you said |
| 00:51.39 | drmessano | They took speex out |
| 00:51.42 | drmessano | ERRR |
| 00:51.42 | drmessano | No |
| 00:51.47 | drmessano | They took GSM out |
| 00:51.55 | drmessano | Added Speex, wideband Speex, Speex FEC |
| 00:52.00 | denon | I guess I dont care much about gsm |
| 00:52.12 | denon | I either use ulaw, speex or g729 in those situations |
| 00:52.14 | drmessano | Id rather have ilbc and speex as options than GSM |
| 00:52.30 | denon | 729's really not bad |
| 00:52.51 | denon | just that I find overseas the latency fluctuates too much, and you lose packets .. then 729 gets ugly fast |
| 00:52.54 | denon | and ulaw shines |
| 00:53.30 | drmessano | Yeah |
| 00:54.08 | drmessano | Hmmm |
| 00:54.23 | drmessano | I dont see anything "greyed out" in Zoiper, so thats good |
| 00:55.11 | drmessano | NM |
| 00:55.18 | drmessano | Call recording |
| 00:55.47 | denon | attended xfer is disabled as well |
| 00:56.05 | denon | and you can only do a couple lines, can't do local bridging, stuff like that |
| 00:56.13 | drmessano | Yeah, I see that now |
| 00:56.30 | drmessano | Id rather them just leave the buttons off |
| 00:56.31 | drmessano | lol |
| 00:57.04 | *** join/#asterisk angryuser (i=nononon@df01t2-212-194-207-119.d4.club-internet.fr) |
| 00:57.16 | denon | well, of course ya would |
| 00:57.22 | denon | but then they wouldn't sell anything :) |
| 00:57.27 | drmessano | yeah lol |
| 00:57.41 | denon | (they did leave the buttons off on idefisk, which is probably why all your free-as-in-beer friends say to avoid zoiper) |
| 00:58.08 | drmessano | May be.. not sure where I heard it, but that probably why |
| 00:59.45 | Robba | does anyone know what would cause Unknown RTP codec 100 received from 'IP ADDRESS'? |
| 01:00.21 | *** join/#asterisk b11d (n=no@234-200-29-134.hcc.mnscu.edu) |
| 01:00.29 | b11d | hello chaps |
| 01:00.54 | drmessano | ZOMG |
| 01:00.55 | b11d | welp FAXing is broken for me again.. I think im going to have to use rxfax and txfax.. sigh.. |
| 01:01.00 | drmessano | ZOIPER SUPPORTS T38 |
| 01:01.06 | Qwell | drmessano: really? |
| 01:01.10 | drmessano | Hah yes |
| 01:01.24 | Qwell | neat |
| 01:01.48 | drmessano | Thats a knucklebuster of a little feature |
| 01:02.07 | denon | yeah, thats a neat little deal |
| 01:02.36 | drmessano | Oh ncie |
| 01:02.38 | drmessano | nice |
| 01:02.44 | drmessano | Outlook integration on windows |
| 01:02.47 | *** part/#asterisk joat (n=joat@ip70-160-147-169.hr.hr.cox.net) |
| 01:02.52 | drmessano | Thunderbird integration on linux |
| 01:02.59 | drmessano | but NOT Thunderbird integration on Windows |
| 01:04.34 | drmessano | Whats the exchange rate of dollars to euros now? |
| 01:04.53 | drmessano | 1800 USD to 1 EURO |
| 01:05.45 | drmessano | $52 USD for Zoiper Business Edition |
| 01:05.51 | drmessano | They're proud of that, aren't they |
| 01:06.08 | *** join/#asterisk jetlagmk2 (n=jetlag@pool-70-17-48-245.pskn.east.verizon.net) |
| 01:06.19 | denon | 52? I didnt think it was nearly that high |
| 01:06.54 | denon | oh, for qty 1 |
| 01:07.18 | drmessano | I'll go ahead and order 100 for my callcenter |
| 01:07.47 | drmessano | Hmm.. make that 102.. might need spares |
| 01:07.51 | denon | actually, I'd like to get the OEM one |
| 01:08.28 | drmessano | Screw it, i'll stick to the softphone that comes with Vista |
| 01:08.37 | drmessano | Windows Messenger FTFW |
| 01:08.39 | cmantito | vista comes with a softphone? |
| 01:08.51 | drmessano | Windows Messenger |
| 01:08.56 | cmantito | what does it supporT? |
| 01:09.00 | cmantito | *support? |
| 01:09.12 | drmessano | SIP.. not sure of the codec.. I guess ulaw |
| 01:09.27 | cmantito | I never woulda seen that coming from Mickeysoft |
| 01:09.41 | drmessano | As if it was a pioneering innovation |
| 01:09.45 | cmantito | haha |
| 01:09.47 | WilliamK | it's part of their office portion I believe... |
| 01:09.53 | denon | you kidding? MSN was the first soft client we recommended in here |
| 01:09.53 | WilliamK | I saw it first in MS Communicator |
| 01:09.54 | drmessano | No |
| 01:09.56 | denon | years and years ago |
| 01:09.58 | drmessano | Its in Windows Messenger |
| 01:10.13 | cmantito | heh |
| 01:10.14 | Robba | indeed |
| 01:10.26 | WilliamK | Windows Messenger probably only did h323 though |
| 01:10.29 | WilliamK | prior |
| 01:10.31 | Robba | but now they are moving to the unified communications platform |
| 01:10.43 | denon | WilliamK: no, it was one of the only sip clients |
| 01:10.46 | denon | hence the reason we recommended it |
| 01:10.47 | drmessano | Windows Messenger always did SIP |
| 01:11.00 | denon | it wasn't Windows Messenger, it was MSN at the time |
| 01:11.00 | drmessano | They did it as a companion to LCS |
| 01:11.01 | denon | but yeah |
| 01:11.18 | plik | Netmeeting wasn't it? |
| 01:11.20 | drmessano | No |
| 01:11.21 | denon | netmeeting was h323 back then |
| 01:11.23 | Robba | no windows messenger 4.? did sip |
| 01:11.24 | denon | but msn was sip |
| 01:11.40 | cmantito | I remember netmeeting being h323, I had fun with that :P |
| 01:11.57 | denon | bash msft all ya want, but they've been in the voip game an awful long time |
| 01:12.03 | drmessano | 4.0 was the split off, I beleiver |
| 01:12.05 | plik | yeah - and the shared whiteboard ..oO |
| 01:12.06 | drmessano | 4.0 was the split off, I believe |
| 01:12.12 | plik | memories |
| 01:12.31 | drmessano | Messenger kept SIP and the Exchange Messaging |
| 01:12.32 | JT | what's wrong with h.323? :) |
| 01:12.45 | denon | JT: that phrase should be an auto-kick |
| 01:12.51 | JT | why? |
| 01:12.57 | JT | because asterisk is no good at it? |
| 01:13.06 | denon | because h323 is no good at it :) |
| 01:13.17 | JT | it's a far superior protocol to sip |
| 01:13.25 | JT | esepcially at the carrier level |
| 01:13.44 | drmessano | I remember playing with LCS 2003 and Messenger.. LCS sucked.. but I guess a SIP client is a SIP client |
| 01:13.59 | JT | it uses Q.931 cause codes |
| 01:13.59 | denon | to each their own |
| 01:14.04 | JT | has efficient binary signalling |
| 01:14.15 | JT | instead of ambiguous wastefull ascii signalling |
| 01:14.30 | denon | you forgot "easy to troubleshoot" |
| 01:15.05 | JT | that's bogus, packet sniffers can easily troubleshoot either |
| 01:15.18 | JT | and h.323 is really well defined |
| 01:15.37 | JT | sip is a matter of which rfc? which supplement? |
| 01:15.41 | JT | whose implementation |
| 01:16.09 | denon | as I said, to each their own |
| 01:16.18 | JT | well these are real reasons |
| 01:16.18 | drmessano | Thats because h.323 was defined back when only 3 computers existed on the planet |
| 01:16.35 | JT | h.323 is an itu standard, that's why it's well defined |
| 01:16.45 | JT | it's a proper telecommunications standard |
| 01:17.01 | drmessano | SO why are you using Asterisk again? |
| 01:17.03 | drmessano | :) |
| 01:17.09 | JT | why not? |
| 01:17.25 | b11d | why cant FAXing in Asterisk be easy? |
| 01:17.43 | drmessano | Because FAX sucks |
| 01:17.45 | b11d | PRI -> Asterisk -> T1 -> Channel Bank -> FAX |
| 01:17.47 | b11d | why!@(!U@ |
| 01:17.50 | drmessano | Its an OU812 standard |
| 01:17.50 | b11d | it should be simple :) |
| 01:17.59 | *** join/#asterisk [hC] (n=hardcore@mail.metrobridge.com) |
| 01:18.16 | *** join/#asterisk jm|home (n=jm|home@zen.jamiem.com) |
| 01:18.24 | drmessano | However, Fax over h.323 is very FTW |
| 01:18.32 | b11d | i keep getting "poor line condition" errors :( |
| 01:18.56 | b11d | internal works.. so it must be to do with the PRI. |
| 01:19.32 | denon | faxing in asterisk isn't bad .. it's when people try to shove a protocol designed for low-loss conditions over IP |
| 01:19.37 | denon | and then over the intarweb |
| 01:19.46 | denon | and then run their torrent client |
| 01:19.57 | b11d | hah.. |
| 01:20.06 | b11d | my FAXes arent even touching IP and they dont work :( |
| 01:20.10 | drmessano | If you print an ASCII picture on a dot matrix and fax it over h.323, Family Ties will come back on the air on NBC next season |
| 01:20.11 | denon | faxing over TDM on asterisk isn't bad |
| 01:20.13 | Nivex | fax is so last century anyway |
| 01:20.30 | denon | b11d: fax attached to TDM, and PRIs goin out? |
| 01:20.37 | b11d | correct |
| 01:20.43 | denon | what kinda problems? |
| 01:20.55 | *** join/#asterisk rbd (n=rbd@adsl-074-229-183-112.sip.rmo.bellsouth.net) |
| 01:21.01 | denon | (don't echo can when bridged) |
| 01:21.03 | b11d | poor line condition.. faxes never complete transmission |
| 01:21.08 | JT | fax to tdm with asterisk usually works |
| 01:21.11 | b11d | single page works.. multiple pages dont |
| 01:21.16 | JT | analogue cards seem to have less reliability |
| 01:21.22 | b11d | echocancelwhenbridged=no is set on the fax channels |
| 01:21.35 | denon | b11d: poor line conditions, but you have a real PRI goin out? |
| 01:21.35 | b11d | echocancelwhenbridged IS yes onthe PRI channels though.. is this a mistake? |
| 01:21.44 | b11d | denon.. right. |
| 01:21.46 | denon | I mean, real PRI as in, not some pri to voip thing from your cable company |
| 01:21.54 | b11d | no its PRI to my telco |
| 01:22.04 | JT | b11d: when a fax is detected, the echo cancellation *should* be disabled |
| 01:22.05 | denon | b11d: it's probably disabling when you try to fax, but yeah, you dont want echo can on a fax |
| 01:22.13 | denon | but force it off, just to test |
| 01:22.17 | rbd | hey guys. ubuntu 7.10, x64, asterisk 1.4.10. I've been trying to compile app_swift for awhile and have been getting some WEIRD errors: http://pastebin.com/d21b4d39f ...can anyone offer any ideas ...looks to me like something screwy with the kernel headers (or which kernel headers app_swift is referencing) |
| 01:22.24 | b11d | so I should have "echocancelwhenbridged=no" on my PRI? |
| 01:22.31 | b11d | it IS set to yes right now. |
| 01:22.49 | denon | well, just glance at the channel info during a faxc |
| 01:22.50 | JT | b11d: read again |
| 01:22.50 | denon | -c |
| 01:23.04 | denon | it'll tell you if it's disabling automagically or not |
| 01:23.11 | drmessano | YAY Vulnerability in SNOM web interface --> http://blogs.zdnet.com/ip-telephony/?p=3218 |
| 01:23.22 | denon | oh joy |
| 01:23.33 | denon | glad we don't have much snom out there :) |
| 01:23.55 | bkw_ | frame slips kill fax like mad |
| 01:23.59 | b11d | i dont know what im looking for here really.. one sec |
| 01:24.17 | denon | b11d: show channel zap/50 or whatever |
| 01:25.02 | b11d | yeah "echo cancellation: 0 taps unless TDM bridged" |
| 01:25.12 | denon | no, do it during a fax |
| 01:25.18 | denon | and it'll tell you it's current status |
| 01:25.25 | b11d | one sec.. |
| 01:25.35 | denon | Ive gotta bail |
| 01:25.45 | denon | everyone and their dog knows what I'm gettin at in here though |
| 01:25.50 | *** join/#asterisk Kumbang (n=dsp@rusnas.paume.ITB.ac.id) |
| 01:25.51 | denon | ttyl |
| 01:25.55 | drmessano | heh |
| 01:26.06 | b11d | thanks denon |
| 01:26.09 | denon | no sweat |
| 01:26.15 | drmessano | l8r |
| 01:27.14 | bkw_ | drmessano: I suspect that will be fixed soon.. I think the 6 or so bugs I opened with Snom are fixed in 7.1.33 |
| 01:27.30 | bkw_ | all the bugs I found in the snom were related to TLS and SRTP interop issues |
| 01:27.33 | bkw_ | and they were real bugs |
| 01:27.53 | drmessano | Thats cool |
| 01:28.06 | Robba | does anyone know the cause of 'ast_rtp_read: Unknown RTP codec 100 received from' in the asterisk cli |
| 01:28.15 | bkw_ | Robba: talking to a cisco box? |
| 01:28.19 | b11d | hmm.. yeah.. it shows EC as OFF on both channels |
| 01:28.24 | bkw_ | that is sending dtmf on 100 instead of 101 |
| 01:28.42 | Robba | linksys = cisco same diff different car |
| 01:28.49 | bkw_ | asterisk still has that bug? it doesn't adjust the based on the rtp map in the dynamic range |
| 01:29.10 | bkw_ | if I recall asterisk compares the payload number instead of the name in the sdp compare unless thats changed recently |
| 01:29.40 | bkw_ | If I invite and say telephony-event is on 98 asterisk should honor that.. but it doesn't |
| 01:30.43 | Robba | will it cause problems? |
| 01:30.53 | bkw_ | well dtmf prob. won't work |
| 01:31.41 | Robba | i have probably done the worst thing here cause hooked up to that PAP2 box is a Fax lol |
| 01:32.06 | Robba | and the fax seems to work fine |
| 01:32.10 | bkw_ | hehe |
| 01:32.23 | Robba | just every now and then that line appears in the CLI |
| 01:32.51 | Robba | is there a fix? |
| 01:33.29 | bkw_ | in the pap2 you should be able to tell it 101 for dtmf |
| 01:34.37 | bkw_ | Ok I give up in chan_sip.c |
| 01:34.46 | Robba | NSE Dynamic Payload is the only thinkg that references 100 |
| 01:35.12 | bkw_ | NSE asterisk doesn't support that |
| 01:35.14 | bkw_ | DTMF AVT? |
| 01:35.17 | bkw_ | do you see that as an option? |
| 01:36.56 | Robba | is AVT Dynamic Payload? |
| 01:37.03 | Robba | thats currently set to 101 |
| 01:37.23 | bkw_ | ok then you shoudl be ok.. asterisk will just complain about 100 |
| 01:37.27 | bkw_ | which you can ignore since it doesn't support it |
| 01:38.10 | husimon | on the linksys pap2t what the hell is the username once you set a password? |
| 01:38.24 | drmessano | user |
| 01:38.25 | drmessano | or |
| 01:38.26 | drmessano | admin |
| 01:38.30 | husimon | ok |
| 01:38.47 | plik | husimon: howdy :) |
| 01:39.03 | husimon | hi |
| 01:39.22 | husimon | drmessano, i find it funny that there is no-where to set the admin password |
| 01:39.30 | drmessano | Sure there is |
| 01:39.33 | drmessano | Click Advanced |
| 01:39.52 | plik | new toys huh? |
| 01:39.57 | bkw_ | the madness |
| 01:40.12 | bkw_ | duplication is the theme |
| 01:40.39 | drmessano | lol |
| 01:40.45 | drmessano | Great.. brb, reboot |
| 01:52.42 | *** join/#asterisk edisonxl (n=root@58.37.227.24) |
| 01:57.26 | *** join/#asterisk obnauticus (n=obnautic@c-67-160-183-109.hsd1.or.comcast.net) |
| 01:57.59 | *** join/#asterisk outtolunc (n=me@c-67-174-216-60.hsd1.ca.comcast.net) |
| 01:59.55 | dmz | howdy y'all, i have been tracing down issues w/my queues and think i've found the problem. I have a queue that's members are set to be direct dial ; ex Member => SIP/<phone>@<voip>/ext,1 |
| 02:00.00 | dmz | this "use to" work |
| 02:00.25 | dmz | but now when a call comes into the queue and it calls out to the members it never transfers the call and it ends up timing out in the queue |
| 02:01.12 | dmz | any suggestions? (other than using regular agents, I liked this as it allowed for me to get lots of incoming calls on with each being able to be answered directly as an extension ff my phone |
| 02:02.17 | dmz | Members can be direct channels, i.e. phones connected to Asterisk. You can also define members as individuals that login from any connection to receive calls. |
| 02:02.23 | *** join/#asterisk Faithful (n=Faithful@ns.linuxterminal.com) |
| 02:02.28 | dmz | that's what i'm trying to do, but it just isn't working anymore |
| 02:04.25 | *** join/#asterisk luckyone (n=hidden@CPE-65-28-6-188.kc.res.rr.com) |
| 02:05.31 | luckyone | hello all - why does /var/run/asterisk get removed when I restart my system? and when that directory isn't there, why isn't it created when Asterisk is started |
| 02:06.23 | luckyone | it causes problems because /var/run/asterisk/asterisk.ctl isn't there |
| 02:14.59 | Jaxxan | dmz: which version of asterisk are you using ? |
| 02:16.17 | Jaxxan | i use: member=>SIP/<number>@<voip> in my queues.conf for mobile phones to accept calls in a queue. |
| 02:16.35 | Jaxxan | and it works for me |
| 02:16.52 | Jaxxan | i'm running 1.2.16 |
| 02:20.08 | *** join/#asterisk drmessano (n=nonya@c-76-125-26-150.hsd1.ga.comcast.net) |
| 02:20.15 | Jaxxan | in fact, i setup a operator queue for a non-profit telethon a few months ago. gave them 12 mobile phones, setup a queue for them in asterisk and voila, mobile telethon. |
| 02:20.51 | *** join/#asterisk b1shop (n=b1shop@c-71-194-197-216.hsd1.il.comcast.net) |
| 02:20.57 | cmantito | ooh, I *like* that idea |
| 02:21.24 | cmantito | that's good :) |
| 02:21.38 | drmessano | Windows. pffft |
| 02:23.20 | *** join/#asterisk outtolunc (n=me@c-67-174-216-60.hsd1.ca.comcast.net) |
| 02:23.34 | *** join/#asterisk [gquit]bombadil (n=dana@CPE-70-94-44-157.wi.res.rr.com) |
| 02:24.16 | *** join/#asterisk qthrul (n=qthrul@adsl-074-184-195-117.sip.asm.bellsouth.net) |
| 02:25.36 | *** join/#asterisk FL1SK (n=FL1SK@72.24.30.153) |
| 02:25.53 | FL1SK | Hi |
| 02:26.21 | FL1SK | Need to know what the best VOIP router to use is |
| 02:26.32 | drmessano | Yeah, thats not subjective |
| 02:26.40 | cmantito | hehe |
| 02:26.45 | FL1SK | I have this damn Vonage RT31P2 Router |
| 02:26.49 | FL1SK | damn this is locked |
| 02:26.55 | FL1SK | ahhhh |
| 02:27.14 | drmessano | Are you looking for something to use with Asterisk? |
| 02:27.16 | FL1SK | sorry guys kind of broad |
| 02:27.25 | FL1SK | Yes i want to use with Asterisk |
| 02:27.54 | luckyone | all I had to do was create a script to make that directory |
| 02:27.57 | outtolunc | the best *one* is an unlocked *one* <G> |
| 02:27.59 | drmessano | Then get a real ATA or a real PHONE |
| 02:28.14 | FL1SK | real ATA |
| 02:28.22 | FL1SK | unlocked one drmessano |
| 02:28.26 | drmessano | VoOP routers are the sort of things you put at yours parents houses |
| 02:28.30 | drmessano | No, A locked one |
| 02:28.31 | drmessano | Duh |
| 02:28.46 | cmantito | I reccomend an unlocked PAP2 if you can get hold of it |
| 02:28.47 | drmessano | s/VoOP/VoIP/ |
| 02:28.48 | FL1SK | :) |
| 02:28.54 | drmessano | Or BUY an ATA |
| 02:28.55 | cmantito | or rther |
| 02:29.00 | drmessano | Like a PAP2T |
| 02:29.03 | cmantito | or rather a never-was-locked PAP2 |
| 02:29.14 | FL1SK | where can i get them |
| 02:29.16 | luckyone | where do you put the wav you want to play back for a voicemail greeting? |
| 02:29.17 | FL1SK | give me brand |
| 02:29.19 | cmantito | since unlocked PAP2s that were once locked are never actually "unlocked" |
| 02:29.23 | cmantito | Linksys PAP2T |
| 02:29.25 | drmessano | Linksys PAP2T Google |
| 02:29.29 | luckyone | FL1SK: Linksys PAP2 |
| 02:29.44 | FL1SK | ok |
| 02:29.45 | cmantito | http://www.google.com/products?q=PAP2T&btnG=Search+Products |
| 02:29.55 | FL1SK | are those newer than my RT31P2 |
| 02:30.14 | luckyone | can someone help a newbie get his voicemail setup with a wav he just recorded? |
| 02:30.28 | FL1SK | damn those are 57 dollars |
| 02:30.29 | FL1SK | ahhhh |
| 02:30.36 | drmessano | Welcome to VoIP |
| 02:30.37 | *** join/#asterisk St1ckm4n (i=St1ckm4n@75.145.72.133) |
| 02:30.40 | cmantito | luckyone: what's up? |
| 02:30.53 | FL1SK | wish i could unlock this damn RT31P2 |
| 02:31.05 | luckyone | cmantito: where do I put my wav file? in what file do I need to modify to tell it to use this file? |
| 02:31.11 | drmessano | If you unlock it, you'll still have a RT31P2 |
| 02:31.14 | cmantito | FL1SK: you don't want to, even after it's unlocked, it still won't work right |
| 02:31.18 | drmessano | HA |
| 02:31.20 | drmessano | jinx |
| 02:31.23 | cmantito | cmantito: to use as a voicemail greeting |
| 02:31.24 | cmantito | hehe |
| 02:31.26 | cmantito | sljfdalsdj |
| 02:31.30 | luckyone | cmantito: yes |
| 02:31.31 | FL1SK | cmantito: understood |
| 02:31.32 | cmantito | sorry, head's in 32 directions |
| 02:31.52 | FL1SK | ok guys i guess i will look for a cheap one somewhere's |
| 02:32.06 | FL1SK | but that PAP2t is the best then eh? |
| 02:32.07 | drmessano | Yep, cheap is always the way to go with VoIP |
| 02:32.17 | drmessano | Buy something chinese |
| 02:32.22 | FL1SK | lol |
| 02:32.30 | drmessano | A FungXu 8100L is good |
| 02:32.35 | cmantito | luckyone: /var/spool/asterisk/voicemail/<voicemail context>/<voicemail user>/unavail.wav for an unavailable message |
| 02:32.41 | cmantito | or busy.wav for an "is on the phone" type message |
| 02:32.42 | FL1SK | i need to have a PAP2T with the -NA right |
| 02:32.47 | drmessano | No |
| 02:32.48 | cmantito | usually that's where it goes anyway |
| 02:32.51 | drmessano | PAP2T |
| 02:32.55 | drmessano | there is no PAP2T-NA |
| 02:33.02 | FL1SK | hmmm |
| 02:33.05 | FL1SK | i see a bunch |
| 02:33.15 | drmessano | How many? |
| 02:33.17 | cmantito | it's just a north american designated pap2t iirc |
| 02:33.32 | FL1SK | oh i gotcha |
| 02:33.33 | cmantito | as in, pap2t and pap2t-na mean the same thing except one was "packaged" for north america |
| 02:33.40 | drmessano | How many do you need, FL1SK? |
| 02:33.50 | FL1SK | oh just 1 |
| 02:33.51 | cmantito | just like if you look around, you'll occasionally see pap2t-eu's |
| 02:33.55 | St1ckm4n | I appologize if this is a stupid question but I can't figure out how to do a switch/case statement in my dialplan without using AEL |
| 02:34.11 | FL1SK | linksys site only shows a PAP2 no PAP2T |
| 02:34.11 | JT | GoToIf |
| 02:34.15 | FL1SK | whats the diff |
| 02:34.21 | luckyone | cmantito: so, per mailbox, I create a busy.wav and and a unavil.wav in the mailbox dir |
| 02:34.29 | drmessano | One is in English, one is in Euros |
| 02:34.37 | St1ckm4n | that's what I've been doing is nesting a bunch of gotoif's but thought there must be another way |
| 02:34.40 | FL1SK | seriously |
| 02:34.42 | FL1SK | heh heh |
| 02:34.59 | luckyone | cmantito: and if there isn't a file there, then it uses the ones from default? |
| 02:35.04 | cmantito | yes |
| 02:35.06 | cmantito | also |
| 02:35.19 | cmantito | you can have it make the appropriate files from within the voicemail system |
| 02:35.23 | drmessano | FL1SK, you can always use a softphone to get your feet wet |
| 02:35.29 | drmessano | Like X-Lite or Zoiper |
| 02:35.32 | cmantito | mailbox options, greetings, then choose the relevant greeting and it'll let you record it right over the phone |
| 02:36.05 | FL1SK | nah |
| 02:36.13 | FL1SK | im goin all out dood |
| 02:36.22 | FL1SK | i have vonage now |
| 02:36.25 | FL1SK | im getting rid of it |
| 02:36.28 | drmessano | Oh hell yes |
| 02:36.48 | cmantito | luckyone: basically, if you set up a voicemail extension that calls VoiceMailMain() for the ext it'll not only let you check your messages but set mailbox options including greetings |
| 02:36.48 | FL1SK | :) |
| 02:36.54 | drmessano | Then get a PAP2T and a 5 Jigawatt Asterisk box |
| 02:37.04 | FL1SK | for sure |
| 02:37.04 | drmessano | Free calls forever |
| 02:37.07 | FL1SK | thats what im doin |
| 02:37.14 | drmessano | ZAWESOME |
| 02:37.19 | FL1SK | :) |
| 02:37.37 | drmessano | I dont know much about Asterisk |
| 02:37.48 | drmessano | I just hang out here when my Vontage is broken |
| 02:38.08 | outtolunc | anyone got any 5's... go fish |
| 02:38.14 | FL1SK | :) |
| 02:39.00 | luckyone | cmantito: hmmm, I moved the files there, chowned them to asterisk:asterisk and called my extension, it played back vm-intro |
| 02:39.15 | cmantito | the filename is unavail.wav? |
| 02:39.23 | cmantito | and what did you use to refer it to voicemail in the dialplan? |
| 02:39.33 | luckyone | cmantito: one second |
| 02:39.44 | luckyone | cmantito: and thanks for the help/patience! |
| 02:39.48 | cmantito | np |
| 02:40.11 | drmessano | http://www.telephonydepot.com/product_p/105-054-pap.htm <-- $52 PAP2 |
| 02:40.30 | cmantito | just outta curiosity, is anyone here dCAP certified? |
| 02:40.38 | luckyone | cmantito: VoiceMail(9999@get-open) |
| 02:40.57 | luckyone | cmantito: this is for inbound callers |
| 02:41.16 | cmantito | luckyone: ok, try Voicemail(9999@get-open|u) |
| 02:41.28 | cmantito | the u option specifies you want to play the unavailable greeting if there is one |
| 02:41.37 | luckyone | cmantito: I am running 1.4.17 |
| 02:42.44 | *** join/#asterisk jmesquita (n=jmesquit@201.20.243.195.user.ajato.com.br) |
| 02:42.55 | luckyone | cmantito: now it says - the person at extenstion 9 9 9 9 is unavailable... |
| 02:43.10 | cmantito | and you named it unavail.wav? |
| 02:43.49 | luckyone | yeah, I think I didn't put it in the unavail directory, just at the root of 9999 |
| 02:43.55 | luckyone | will move it now |
| 02:43.56 | cmantito | right |
| 02:44.00 | cmantito | no, that's the right place |
| 02:44.25 | cmantito | but try calling it while you're connected to the asterisk console and see if there's any messages printed out that look relevant |
| 02:44.27 | cmantito | (asterisk -r) |
| 02:44.41 | cmantito | (even better is asterisk -rvvvvv beacuse of the extra detail) |
| 02:44.52 | luckyone | cmantito: yeah, I see lots of relevant info - connected that way to cli |
| 02:45.07 | cmantito | anything specifically relevant to the voicemail greeting? |
| 02:45.11 | cmantito | IE, unable to play, unable to open, etc |
| 02:46.34 | luckyone | cmantito: cool, now I see it is the wrong freq |
| 02:46.42 | luckyone | what freq does it need to be? |
| 02:46.45 | cmantito | it may work if you try changing to .WAV |
| 02:46.57 | cmantito | surprisingly, .wav and .WAV are different formats to asterisk |
| 02:47.06 | cmantito | and what one PC calls a "wave file" may actually be a .WAV |
| 02:47.07 | luckyone | <PROTECTED> |
| 02:47.36 | cmantito | try changing the file ext first |
| 02:48.30 | luckyone | cmantito: yeah, now it is complaining about the header size |
| 02:48.41 | cmantito | what did you use to record this? lol |
| 02:49.17 | luckyone | cmantito: sound recorder then touched it up in audacity |
| 02:50.01 | cmantito | rename unavail.wav to unavail-2.wav and then do "sox unavail-2.wav unavail.wav" -- sometimes if there's bad information sox can correct it by just running it through it |
| 02:50.43 | cmantito | if you're lucky, that is ;) |
| 02:51.03 | luckyone | cmantito: that's my name ;) |
| 02:51.08 | cmantito | haha |
| 02:52.37 | luckyone | cmantito: no dice... |
| 02:53.17 | cmantito | damn |
| 02:53.26 | cmantito | audacity can do some weird stuff |
| 02:54.11 | alrs | http://cgi.ebay.com/Chinese-outstanding-teak-carving-dragon-telephone_W0QQitemZ150214837270QQihZ005QQcategoryZ985QQssPageNameZWDVWQQrdZ1QQcmdZViewItem |
| 02:54.32 | cmantito | umm |
| 02:54.50 | cmantito | I'm not sure offhand what wav file format asterisk expects, someone else around here would probably know that. |
| 02:55.35 | luckyone | I probably just need to setup a VoiceMailMain() extension anyway... |
| 02:55.53 | cmantito | that's what I reccomend |
| 02:55.58 | FL1SK | so what if the Notorious DogFace05 unlocks my RT31P2 |
| 02:56.02 | cmantito | I find I get better quality using the phones to record it anyway |
| 02:56.09 | cmantito | FL1SK: it'll still be crap |
| 02:56.15 | FL1SK | ok |
| 02:56.20 | FL1SK | thanks cmantito |
| 02:56.26 | drmessano | You'll still have a fucking RT31P2 |
| 02:56.32 | FL1SK | heh heh |
| 02:56.39 | drmessano | Im not joking |
| 02:56.42 | drmessano | They're CRAP |
| 02:56.54 | drmessano | Send it to umm |
| 02:56.57 | drmessano | a1fa |
| 02:57.03 | drmessano | He needs an ATA |
| 02:57.18 | cmantito | haha |
| 02:57.35 | cmantito | seriously, an unlocked rt31p2 is a useless rt31p2 |
| 02:57.39 | cmantito | either way |
| 02:57.42 | cmantito | it's still a doorstop |
| 02:59.15 | *** join/#asterisk andresmujica (n=andresmu@190.25.100.212) |
| 02:59.55 | cmantito | luckyone: good luck. just ask if you need anything |
| 03:00.17 | andresmujica | hi!, anyone has ever tried (or hear about) safari c3?? i'm trying to connect an asterisk box to that softswitch. It's using xten with a SIP account with digest auth. But it's not working with my asterisk box... |
| 03:01.05 | andresmujica | i mean, i ve got a sip line with xten pro using that softswitch and whant to put it on my asterisk |
| 03:01.23 | obnauticus | Which twisted pair in cat5e is not being used by default? |
| 03:01.41 | obnauticus | because i want to route regular phone through it. |
| 03:01.56 | andresmujica | if you're using 100M you can take the blue pair |
| 03:02.05 | obnauticus | k |
| 03:02.15 | obnauticus | I might as well just use voip then L\ |
| 03:02.17 | andresmujica | if you're with 1000 there's no room available |
| 03:02.22 | obnauticus | ya |
| 03:02.34 | obnauticus | and i can use the blue pair for POE too right? |
| 03:02.50 | andresmujica | hmm.. don't know much about that yet. but probably yes. |
| 03:04.23 | cmantito | yes |
| 03:04.24 | drmessano | Brown is PoE |
| 03:04.32 | cmantito | but standard is brown |
| 03:04.33 | cmantito | 48v |
| 03:04.36 | obnauticus | k |
| 03:05.29 | luckyone | cmantito: what context should my internal voice mailbox management extensions be defined under? |
| 03:05.43 | andresmujica | ohh. now we know. |
| 03:05.53 | andresmujica | the brown also is used for autonegotiation... |
| 03:06.05 | obnauticus | oioo |
| 03:06.21 | cmantito | luckyone: whatever you'd like, mine's simply 'voicemail' |
| 03:06.46 | luckyone | cmantito: do I have to define anything special for it in sip.conf ? |
| 03:07.02 | luckyone | cmantito: if I was to create a new context for instance? |
| 03:07.09 | cmantito | nope |
| 03:07.13 | cmantito | just set the mailbox option for each client |
| 03:08.54 | luckyone | cmantito: cool - can I make them enter a mailbox number? |
| 03:09.24 | luckyone | so I just have one extension for managing mailboxes, not multiple extensions for each mailbox? |
| 03:09.40 | cmantito | you mean for checking the messages? |
| 03:10.37 | *** join/#asterisk adjohn (n=adjohn@p6081-ipad53marunouchi.tokyo.ocn.ne.jp) |
| 03:10.39 | luckyone | cmantito: for managing them |
| 03:10.46 | luckyone | cmantito: the mailboxes |
| 03:10.55 | luckyone | also - what is the best sip client for gnome? |
| 03:12.28 | cmantito | then you can just point an extension at VoiceMailMain() |
| 03:12.28 | andresmujica | ekiga is the default (not necessarily the best but is teh default one) |
| 03:12.59 | *** join/#asterisk hmmhesays (n=hmmhesay@24-119-176-74.cpe.cableone.net) |
| 03:13.02 | cmantito | exten => 604,1,VoiceMailMain(@voicemail) |
| 03:13.08 | cmantito | for example, where @voicemail is the context |
| 03:13.22 | cmantito | and it'll prompt the user to enter ext & pin |
| 03:14.24 | *** join/#asterisk L|NUX (n=linux@unaffiliated/lnux/x-10290) |
| 03:21.49 | luckyone | cmantito: hmmm, it says my extension is not defined... |
| 03:22.16 | cmantito | when you enter the extension to VoiceMailMain? |
| 03:22.44 | luckyone | no, I am trying to get to voicemail main... |
| 03:22.57 | luckyone | exten => 9000,1,VoiceMailMain(9000@get-open) |
| 03:23.24 | outtolunc | hehe |
| 03:23.30 | luckyone | I have a voicemailbox defined for 9000 (can I not use the same number?) |
| 03:23.39 | cmantito | no, you don't want to use the same number |
| 03:23.53 | luckyone | how confusing I must be for my poor system... |
| 03:24.01 | luckyone | hehe |
| 03:24.44 | cmantito | haha |
| 03:25.05 | cmantito | I reccomend doing something like 600,1,VoiceMailMain(@get-open) |
| 03:25.13 | cmantito | and letting it figure out the login :P |
| 03:27.07 | outtolunc | i prefer something like exten => _85XX,1,VoiceMailMain(${EXTEN:1}@context) and the user boxes are 500 series, first box 500 being a common |
| 03:27.11 | luckyone | hmmm, created that in extenstions.conf saved it, then did a reload in cli |
| 03:27.17 | luckyone | still no dice |
| 03:27.52 | cmantito | what does the console say about it? |
| 03:28.15 | *** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 03:28.18 | luckyone | <PROTECTED> |
| 03:28.28 | luckyone | I defined mine on 500 |
| 03:28.37 | luckyone | from extensions.conf... |
| 03:28.56 | luckyone | exten => 500,1,VoiceMailMain(@get-open) |
| 03:29.00 | outtolunc | did you create it in voicemail.conf |
| 03:29.18 | luckyone | let me see (and let's hope so) |
| 03:29.21 | cmantito | and that's in the [get-open] context |
| 03:29.22 | cmantito | ? |
| 03:30.40 | luckyone | do I need a mailbox defined at 500? |
| 03:30.49 | luckyone | I have 4 mailboxes defined |
| 03:30.57 | luckyone | 9999, 9000, 9001, 9002 |
| 03:31.15 | cmantito | no you don't |
| 03:31.32 | cmantito | pastebin your extensions.conf and voicemail.conf? |
| 03:33.04 | riddlebox | cool, I am getting closer to having asterisk be a voicemail solution for the Partner systems |
| 03:34.02 | *** join/#asterisk rajiv (n=rajiv@gentoo/developer/rajiv) |
| 03:34.55 | luckyone | cmantito: http://pastebin.org/19352 |
| 03:35.55 | Jaxxan | i replaced a glenayre voicemail box with asterisk. it was pimp. |
| 03:35.56 | cmantito | k, hang on |
| 03:36.07 | *** join/#asterisk scottsmith77 (n=ssmith@doc-24-32-205-137.ellsworth.ks.cebridge.net) |
| 03:36.15 | Jaxxan | i said goodbye to the 20k annual support for glenayre |
| 03:36.17 | cmantito | right, 500 needs to be defined in the extensiosn.conf context that the phones are in |
| 03:36.53 | luckyone | cmantito: like the inbound context? |
| 03:37.03 | cmantito | well, where are your SIP client's context? |
| 03:37.36 | flush | grssh mayday |
| 03:37.47 | flush | im looking to buy a TDM400P card |
| 03:38.18 | flush | can i get it with 4 FXS modules on it, and order a FXO module apart so i can remove a FXS module and replace it by the FXO module if needed |
| 03:38.36 | luckyone | cmantito: I pretty much just have did's that do different things |
| 03:38.49 | cmantito | ahh |
| 03:38.53 | cmantito | well.....hm |
| 03:39.00 | cmantito | then yes |
| 03:39.04 | cmantito | it needs to be in your incoming context |
| 03:39.38 | luckyone | cmantito: ok, could I define something new in sip.conf or something to make like an internal context? |
| 03:39.53 | cmantito | you could, but you'd need a SIP phone |
| 03:39.56 | cmantito | or something |
| 03:40.56 | luckyone | cmantito: that was dumb of me, that just defines my clients/connections over sip.. |
| 03:41.34 | cmantito | hehe |
| 03:42.17 | luckyone | so, yeah, I don't get why I can't just create an extension in extensions.conf under this new voicemail context and be able to dial it |
| 03:42.47 | cmantito | because you have to have a DID that directs to the voicemail context if you have no clients |
| 03:43.11 | luckyone | I have clients defined too - like my ekiga softphone |
| 03:43.59 | Jaxxan | luckyone: would you like to see my configs for the voicemail box i created for mobile phone users ? |
| 03:44.05 | cmantito | what context is that connected to? |
| 03:44.29 | luckyone | Jaxxan: sure, do you mind posting them (hopefully it won't blow my mind) |
| 03:44.46 | luckyone | cmantito: I have two, homephone and get-open |
| 03:44.59 | cmantito | ok, then put the 500 in the get-open context and not the voicemail context |
| 03:45.33 | luckyone | cool, will do |
| 03:46.40 | watchy | are hints in asterisk 1.4 different? |
| 03:47.12 | watchy | i upgraded a client to 1.4 and they say there poly 601s arent showing busy |
| 03:48.23 | Jaxxan | http://www.pastebin.ca/902088 |
| 03:49.36 | Jaxxan | i authenticate via callerid |
| 03:49.45 | Jaxxan | IE: if you're not on my network you can't check voicemail. |
| 03:50.05 | Nugget | IE: if you can set your caller id, you can check anyone's voicemail. :) |
| 03:50.22 | Jaxxan | well, that's what passwords are for |
| 03:51.02 | Nugget | so you just disallow your users from checking their voicemail when they're away, calling from their mobiles, etc? |
| 03:51.14 | *** join/#asterisk jameswf-home (n=james@ip72-204-221-181.ph.ph.cox.net) |
| 03:51.27 | Jaxxan | i said that wrong, they have to enter in their full phone number with the voicemailmain app |
| 03:51.39 | Jaxxan | but if they're calling from their mobile phones it just prompts them for their password |
| 03:52.46 | Jaxxan | so for people leaving a voicemail number, the voicemail box is determined by the RDNIS |
| 03:53.15 | Jaxxan | if you're calling the voicemail number your callerid determines the voicemail box you go to |
| 03:53.18 | *** join/#asterisk _mwoodj_ (n=MWoodJ@pdpc/sponsor/digium/hyper-eye) |
| 03:53.19 | Jaxxan | then you enter your password |
| 03:53.39 | Jaxxan | if your callerid is not a recognized voicemailbox number, then you're prompted to enter your phone number and password |
| 03:55.07 | luckyone | does reload on the cli not reload my info? |
| 03:55.22 | Jaxxan | which info ? |
| 03:55.30 | cmantito | it should |
| 03:55.37 | luckyone | extensions |
| 03:55.41 | Jaxxan | it does |
| 03:55.43 | cmantito | definitely should |
| 03:55.50 | luckyone | I don't see how I don't have an extension defined for 500 |
| 03:56.05 | Jaxxan | you probably have a typo somewhere |
| 03:56.18 | Jaxxan | did you save your extensions.conf before the reload ? |
| 03:56.19 | cmantito | core show dialplan |
| 03:57.35 | luckyone | dialplan show shows a lot of stuff... |
| 03:58.14 | luckyone | luckyseven*CLI> dialplan show get-open |
| 03:58.14 | luckyone | [ Context 'get-open' created by 'pbx_config' ] '500' => 1. VoiceMailMain(9999@get-open) [pbx_config] '9100' => 1. VoiceMailMain(9000@get-open) [pbx_config] |
| 03:58.21 | obnauticus | pastebin ftw. |
| 03:58.28 | luckyone | sorry |
| 03:58.32 | obnauticus | <3 |
| 03:58.32 | cmantito | should work then |
| 03:58.55 | luckyone | I don't get it for the life of me... |
| 03:59.29 | *** join/#asterisk ShadowHntr (n=sentinel@wikipedia/Shadowhntr) |
| 04:00.02 | cmantito | and hitting 500 on a phone who's sip.conf context=get-open doesn't connect you? |
| 04:00.59 | *** join/#asterisk pkunkra (n=chris@cpe-74-73-10-32.nyc.res.rr.com) |
| 04:01.12 | luckyone | yeah, I just disconnected my homephone account in ekiga to be sure |
| 04:01.14 | *** join/#asterisk ahbritto (n=guest@adsl-69-104-3-183.dsl.pltn13.pacbell.net) |
| 04:01.18 | riddlebox | if someone has this in their dialplan, exten => _#11###,2,Set(DB(IVR/NIGHT)=1) does the DB stand for database? |
| 04:01.28 | Jaxxan | yes |
| 04:03.50 | riddlebox | Jaxxan, my guess is that they use that play the night greeting, would that be safe to assume? |
| 04:03.56 | cmantito | if I have 2 SIP channels that are bridged in a call, can I disconnect one of those channels from the CLI? |
| 04:04.06 | cmantito | say an inbound call was transferred to another SIP address |
| 04:04.36 | outtolunc | soft hangup tech/dev-occurance |
| 04:04.49 | cmantito | where do you get the dev-occurance from? |
| 04:04.50 | Jaxxan | riddlebox: looks like they dial an extension to set the database IVR to the night greeting. |
| 04:05.07 | Jaxxan | or something |
| 04:05.10 | outtolunc | [core] show channels |
| 04:05.25 | cmantito | so, for example, SIP/inerail-trunk-00something? |
| 04:05.25 | outtolunc | the whole channel name = tech/dev-occurance |
| 04:05.32 | riddlebox | Jaxxan, I am getting this from, http://www.voip-info.org/wiki/index.php?page=Asterisk-Partner+ACS+for+Voicemail |
| 04:05.39 | cmantito | damn, it's truncated, haha |
| 04:06.06 | outtolunc | hit tab |
| 04:06.06 | Jaxxan | riddlebox: looks like you're right on the money |
| 04:06.16 | cmantito | ahh tahnks :P |
| 04:06.41 | riddlebox | Jaxxan, the thing I dont understand is, in that link, there is no IVR/Night context? |
| 04:07.50 | [TK]D-Fender | riddlebox, that is NOT a context |
| 04:08.01 | [TK]D-Fender | riddlebox, that is a DB2 family/key pair |
| 04:10.18 | jameswf-home | dah dah dah |
| 04:11.12 | riddlebox | [TK]D-Fender, how you would get the night greeting to play using that family/key pair? |
| 04:12.27 | Robba | in the queues.conf file, what specifies the queue to play MOH after calling in? |
| 04:13.10 | Jaxxan | music=default |
| 04:13.11 | [TK]D-Fender | riddlebox, you do a Gotoif if checking its value |
| 04:13.18 | Jaxxan | music=newMOHdirectory |
| 04:13.49 | Jaxxan | you specify the MOH classes in your musiconhold.conf |
| 04:13.49 | Robba | hmmm |
| 04:13.57 | riddlebox | [TK]D-Fender, thats kind of what I thought, but the existing link doesnt have that, I guess its time to play around with it then |
| 04:14.06 | Robba | music=default plays silence |
| 04:14.08 | [TK]D-Fender | riddlebox, what "link"? |
| 04:14.20 | riddlebox | [TK]D-Fender, http://www.voip-info.org/wiki/index.php?page=Asterisk-Partner+ACS+for+Voicemail |
| 04:14.21 | Jaxxan | Robba: if it plays silence, then you dont have any music in your moh directory |
| 04:14.35 | Robba | if i park a call they get music |
| 04:14.45 | Robba | or put a call on hold |
| 04:15.02 | Jaxxan | the call coming through zap or sip ? |
| 04:15.09 | Robba | either |
| 04:15.20 | Jaxxan | in your zapata.conf you have to specify moh as well |
| 04:15.26 | Jaxxan | for incoming zap calls |
| 04:15.31 | Jaxxan | same for sip i belive |
| 04:15.37 | [TK]D-Fender | riddlebox, yes they show you code for enabling/disabling night moode as they define by that key-pair. They simply don't show themselves doing anything with it. |
| 04:16.05 | Jaxxan | well, maybe not for sip.conf |
| 04:16.33 | Jaxxan | wait, yeah there's a musicclass define in sip.conf |
| 04:17.01 | riddlebox | [TK]D-Fender, I am reading about the DB stuff, I will play with it and get it going, the only other thing I have to figure out is a way to have the customer record their day/night greetings |
| 04:17.46 | Robba | Jaxxan: what has to be set in zapata? |
| 04:17.47 | [TK]D-Fender | riddlebox, .... |
| 04:17.57 | Jaxxan | musiconhold = default |
| 04:18.04 | Robba | thanks |
| 04:19.04 | riddlebox | [TK]D-Fender, I am just trying to get an easy way for them to record the prompts |
| 04:19.24 | Jaxxan | riddlebox: you're trying to emulate an avaya vm box ? |
| 04:19.28 | Robba | hmmmm |
| 04:19.36 | [TK]D-Fender | riddlebox, You seriously have no slue do you? Have you even looked at the list of * applications? |
| 04:19.37 | Robba | still doesn't seem to work |
| 04:19.39 | [TK]D-Fender | clue* |
| 04:19.40 | Jaxxan | robba, you'll prolly have to restart asterisk |
| 04:19.49 | Robba | not just reload? |
| 04:19.49 | Jaxxan | a simple reload wont do it i dont think |
| 04:19.55 | Robba | ahhh |
| 04:19.57 | riddlebox | [TK]D-Fender, I know |
| 04:20.02 | Robba | point taken |
| 04:20.43 | Jaxxan | robba: whenever you're dealing with any of the zaptel and zapata configs, you usually need a restart |
| 04:20.54 | Jaxxan | at least that's my experience |
| 04:21.09 | drmessano | I just sent a Fax over H.323 and "Silver Spoons" got renewed on NBC |
| 04:21.22 | riddlebox | Jaxxan, just trying to provide a voicemail solution for the avaya partner |
| 04:22.10 | Robba | restarted and still nothing |
| 04:22.23 | Robba | want me to pb my queues.conf file? |
| 04:22.28 | Jaxxan | BKW_ still around ? |
| 04:22.39 | bkw_ | yes |
| 04:22.43 | Jaxxan | hey man (= |
| 04:22.49 | Jaxxan | just saying hi, it's been awhile |
| 04:23.04 | Jaxxan | you helped me get my PRI up with a DMS100 5 years ago |
| 04:23.12 | bkw_ | oh yes I don't think your nick was this one was it? |
| 04:23.18 | Jaxxan | yeah it was |
| 04:23.27 | bkw_ | my memory is a bit fragmented |
| 04:23.29 | bkw_ | :P |
| 04:23.33 | Jaxxan | i can imagine |
| 04:23.43 | bkw_ | you're the guy that didn't have the right line card on the DMS100? |
| 04:23.50 | bkw_ | and you work at a CLEC? |
| 04:24.18 | Jaxxan | no, my configs were just wrong, you logged into my box and fixed them. i work for a telco |
| 04:24.24 | *** join/#asterisk egecko (n=sam@cpe-76-176-206-245.san.res.rr.com) |
| 04:24.26 | bkw_ | oh yes |
| 04:24.32 | bkw_ | close :) |
| 04:24.35 | Jaxxan | (= |
| 04:24.37 | jameswf-home | Jaxxan: not bell canada is it |
| 04:24.48 | Jaxxan | no, i work for blue sky communications in american samoa |
| 04:25.02 | *** join/#asterisk Frogzoo (n=Frogzoo@144.137.116.10) |
| 04:25.21 | bkw_ | Jaxxan: well I don't work much on Asterisk anymore.. |
| 04:25.38 | Jaxxan | whatcha into these days ? |
| 04:25.45 | bkw_ | www.freeswitch.org |
| 04:25.52 | Jaxxan | i like the sound of that |
| 04:26.02 | riddlebox | [TK]D-Fender, its not that I dont know how or what to use, I am just trying to figure out a way in this dialplan to do it |
| 04:26.19 | bkw_ | Jaxxan: well join #freeswitch .. I don't like to have the FS conversations spill into this channel... people might get mad at me |
| 04:26.27 | Jaxxan | kk |
| 04:26.57 | Jaxxan | well i'm off. ttyl |
| 04:27.21 | bkw_ | hehe |
| 04:28.23 | Robba | [TK]: do you have any clues as to my question before? about the MOH in queues? |
| 04:31.23 | *** join/#asterisk [hC] (n=hardcore@S01060016b6b53c0c.vc.shawcable.net) |
| 04:36.32 | *** join/#asterisk emist (n=emist@unaffiliated/emist) |
| 04:37.17 | *** join/#asterisk CunningPike (n=CunningP@S010600095b33697f.vc.shawcable.net) |
| 04:41.44 | *** join/#asterisk ph0ne (n=ph0ne@dsl-207-112-19-129.tor.primus.ca) |
| 04:48.23 | *** join/#asterisk frogzoo (n=Frogzoo@CPE-144-137-116-10.nsw.bigpond.net.au) |
| 04:50.15 | St1ckm4n | do you guys have any opinion of the dialparties.agi script should I use it to determine a phones status before sending it calls? |
| 04:51.15 | St1ckm4n | our *@Home version uses it but has deadlock issues so I'm trying to make a clean dialplan and would prefer not to run such a script, but didn't know if there was a better way to see a phones status |
| 04:51.48 | JT | i don't see why dialparties.agi is at all necessary |
| 04:51.56 | JT | just seems like a weird freepbx quirk |
| 04:52.37 | St1ckm4n | JT:is there anyway to check a phones status in my extensions.conf before I send a call to it? |
| 04:53.47 | St1ckm4n | I thought about creating a status variable that would get set based on any action a phone did e.g)dial out local, long distance, intl ... but couldn't think of an easy way to reset that variable immediately after they hungup |
| 04:54.26 | JT | there is no need usually |
| 04:54.29 | JT | if the dial fails |
| 04:54.43 | JT | handle the failure to do something else |
| 04:55.48 | St1ckm4n | I'm trying to handle the call before dial hits the timeout, right now if one phone is busy the call still rings for x seconds before hitting voicemail, ideally I'ld want asterisk to recognize that phone is on another call and send directly to voicemail |
| 04:55.53 | *** join/#asterisk [gquit]bombadil (n=dana@CPE-70-94-44-157.wi.res.rr.com) |
| 04:56.34 | St1ckm4n | I tried disabling call waiting but that didn't seem to work |
| 04:56.49 | *** join/#asterisk azidenth (n=aby_azid@185.219.50.60.cbj04-home.tm.net.my) |
| 04:57.22 | azidenth | hello, |
| 04:58.52 | azidenth | im having problem send SIP call to another asterisk server |
| 04:59.52 | azidenth | im getting error chan_sip.c:8373 check_auth: username mismatch, have <100778208>, digest has <kl_asterisk> |
| 05:00.24 | azidenth | how do i bypass this? |
| 05:01.23 | riddlebox | [TK]D-Fender, last night when I was having those weird problems with the zaptel drivers, I reloaded ubuntu tonight and installed zaptel first, configured it, then installed asterisk and now its working perfectly |
| 05:03.00 | azidenth | really appreciate if someone can help me.. |
| 05:06.36 | cmantito | can you post your sip.confs? |
| 05:06.56 | cmantito | as well as the dialplan command you're using to send the call? |
| 05:07.21 | cmantito | use pastebin |
| 05:07.31 | ShadowHntr | azidenth: i don't know enough about * to be sure, but either make sure that the user you specify in your channels is correct on their end or read further into the documentation. |
| 05:07.49 | ShadowHntr | if i were on broadband right now i'd dive into the documentation for you right now |
| 05:08.24 | azidenth | hi..the thing is I have to asterisk box connected together |
| 05:08.37 | azidenth | i mean i have 2 asterisk box connected together |
| 05:08.48 | cmantito | can you post the sip.conf from each and the dialplan command you're using to send the call to pastebin? |
| 05:09.58 | azidenth | i successfully connected both server via register |
| 05:10.14 | cmantito | azidenth: my last msg please. |
| 05:10.57 | azidenth | how do i send it to pastebin cmantito? |
| 05:11.11 | azidenth | never use it |
| 05:11.13 | cmantito | paste it in pastebin.org |
| 05:11.17 | cmantito | and the it'll give you al ink |
| 05:11.21 | cmantito | *link |
| 05:11.23 | cmantito | paste the link here |
| 05:11.49 | azidenth | ok |
| 05:21.33 | azidenth | cmantito here the link: http://pastebin.org/19356 |
| 05:22.40 | *** join/#asterisk pkunkra (n=chris@cpe-74-73-10-32.nyc.res.rr.com) |
| 05:22.58 | cmantito | and what command are you using to send the call to the other server? |
| 05:23.39 | azidenth | dial command |
| 05:23.57 | azidenth | im using Realtime SIP |
| 05:24.01 | cmantito | the whole thing |
| 05:24.04 | cmantito | what is the whole dial command |
| 05:24.53 | azidenth | if im from kl and dial to HK the command is exten => _100.,1,Dial(SIP/hk_asterisk/${EXTEN}) |
| 05:25.32 | cmantito | try SIP/${EXTEN}@hk_asterisk |
| 05:25.32 | azidenth | cmantito this happend when I login as Realtime SIP user |
| 05:26.05 | azidenth | if i login from sip user from flatfile i dont get this error |
| 05:26.40 | cmantito | hmm, I'm afraid I'm not sure what's going on, sorry |
| 05:27.02 | azidenth | no worries..thanks anyway |
| 05:28.45 | azidenth | are they any differences sip user from flat file and Realtime SIP users? |
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| 05:36.37 | *** part/#asterisk ^shark_ (n=^shark_@217.194.147.193) |
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| 06:31.32 | Corydon76-dig | ~thebook |
| 06:31.32 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com |
| 06:31.51 | ShadowHntr | Corydon76-dig: not sure if it covers what i want to know, but i'll check ita gain... |
| 06:32.09 | sweeper | woo second edition |
| 06:33.12 | Corydon76-dig | Buy a copy, feed the authors |
| 06:33.43 | Corydon76-dig | They occasionally want to eat something other than hog slop |
| 06:34.11 | ShadowHntr | hmm... |
| 06:34.24 | azidenth | already bought it |
| 06:34.27 | ShadowHntr | Corydon76-dig: the book mentions what i want to do, but just says that it's been done before. |
| 06:34.39 | ShadowHntr | i'd like to talk to some people who have used the UNISTIM modules for Asterisk |
| 06:34.42 | *** join/#asterisk adorah (n=Michael@87.69.130.248) |
| 06:35.04 | Corydon76-dig | Very few people have |
| 06:35.10 | ShadowHntr | cause i've seen Nortel IP phones for rather cheap, and they're (build quality) about as good as Polycom. don't know about features, but would like to look into it further. |
| 06:36.54 | ShadowHntr | actually |
| 06:36.57 | ShadowHntr | given my environment |
| 06:37.10 | ShadowHntr | i wouldn't mind having one Nortel device and perhaps another IP phone from another vendor |
| 06:37.13 | ShadowHntr | to experiment |
| 06:37.18 | ShadowHntr | this is for a home environment |
| 06:45.28 | azidenth | how do we bypass authentication on incoming INVITE? |
| 06:50.17 | *** join/#asterisk Jaxxan (n=Jaxxan@leone-canopy05.bluelink.as) |
| 06:50.22 | Jaxxan | hey guys |
| 06:53.00 | *** join/#asterisk glen2 (n=glen@87-194-2-134.bethere.co.uk) |
| 06:59.09 | azidenth | help i'm stuck |
| 07:00.35 | kaldemar | azidenth: http://www.voip-info.org/wiki-Asterisk+sip+insecure |
| 07:01.48 | *** join/#asterisk swankier (i=swank@S01060011d81a9ada.vf.shawcable.net) |
| 07:02.46 | azidenth | thanks |
| 07:04.00 | *** join/#asterisk erojasv (n=erojasv@201.240.80.53) |
| 07:14.57 | awk | hmm, anyone havea feature list of 1.6 |
| 07:16.39 | patrick-- | can anyone tell me anything on this error: mISDN_close: fid(18) isize(131072) inbuf(0xb717f008) irp(0xb717f008) iend(0xb717f008) |
| 07:19.49 | the_5th_wheel | exten => s, 100, Queue(support||||16) <-- is there any reason why when i use this bit to send someone to the que,it dumps the in the front of the que? |
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| 07:24.59 | *** join/#asterisk juliusspencer (n=julius@60-234-130-224.bitstream.orcon.net.nz) |
| 07:25.47 | juliusspencer | howdy, I have a simple question. I have a Digium TDM400P and am wondering if it's possible to stop the card from answering calls. |
| 07:26.39 | kaldemar | pull the plug |
| 07:27.04 | juliusspencer | heh... sorry yeah didn't specify that I want to be able to continue to make outbound calls |
| 07:27.30 | kaldemar | it's not the card that answers calls, but the software behind it. configure your dialplan not to answer any incoming calls. |
| 07:27.41 | juliusspencer | ah dialplan |
| 07:27.54 | Corydon76-dig | juliusspencer: set the context in zapata.conf to a context that does not exist (and ensure there is no "default" context |
| 07:27.57 | juliusspencer | ok so that would be in extensions.conf? |
| 07:29.00 | *** join/#asterisk worgil (n=worgil@78.166.127.48) |
| 07:29.02 | juliusspencer | ok, so if I set it to a context that doesn't exist do you reckon it will still be possible to make outgoing calls? |
| 07:29.13 | Corydon76-dig | Yes, certainly |
| 07:29.21 | Corydon76-dig | context is only needed for incoming calls |
| 07:29.27 | juliusspencer | oh sweet |
| 07:29.57 | Corydon76-dig | just remember that it falls back to "default" if it can't find the context to which you set it |
| 07:30.07 | juliusspencer | thanks heaps, I'm quite the n00b at it. I managed to get my brother SIP in to make an outgoing call, it just started answering calls :) |
| 07:30.35 | juliusspencer | ok so I need to get that default part out of the situation |
| 07:31.05 | Corydon76-dig | Of course, if the aim is to prevent the lines from being tied up, you might want to use a context that answers the call and immediately hangs up |
| 07:31.28 | juliusspencer | ok I have from-internal (which is cool) and I have from-pstn (which is the one I should rename) |
| 07:32.28 | Corydon76-dig | I like "does-not-exist" as a context name |
| 07:32.29 | juliusspencer | oh no no, aim is to allow incoming calls to be answered by phones manually, but allow authenticated SIP users outbound access |
| 07:32.43 | juliusspencer | :) nice nice |
| 07:32.56 | Corydon76-dig | It ensures that nobody comes in and defines that context later |
| 07:33.53 | Corydon76-dig | btw, if you aren't already, best practices are to keep your config files in a source control repository, such as Subversion |
| 07:34.02 | juliusspencer | that's great, I'll give it a go, once the rpms (for kmdl etc) catch up with the latest version of kernel for the distribution |
| 07:34.18 | Corydon76-dig | It ensures that if somebody screws around with your configs, you can easily revert to a known working config |
| 07:34.39 | juliusspencer | oh right, it's just one machine which I'm looking after. I just use svn for documentation and java. |
| 07:34.47 | juliusspencer | it is handy as a backup :) |
| 07:35.05 | juliusspencer | something to keep in mind |
| 07:35.12 | juliusspencer | I tend to keep configs in a wiki... |
| 07:35.16 | juliusspencer | which I backup |
| 07:35.38 | juliusspencer | thanks once again :) |
| 07:40.40 | *** join/#asterisk Chris-NB (n=chris@ng1.kurtkrenn.com) |
| 07:40.44 | Chris-NB | hi |
| 07:40.55 | *** join/#asterisk Kapsel (i=kapsel@62.242.240.33) |
| 07:41.07 | Chris-NB | someone usint a Sangoma A10X and syncing clock to a A500 (or A400)? |
| 07:41.27 | Chris-NB | I've a few questions concerning the clock syncing |
| 07:41.37 | Chris-NB | anyone have done this? |
| 07:41.50 | JT | done what specifically? |
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| 08:03.20 | RedStalker_Mike | hi all! |
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| 08:13.13 | alrs | shtoom: is it me, or is this list totally dead? http://news.gmane.org/gmane.comp.voip.shtoom |
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| 09:11.51 | jhiver | hi |
| 09:12.47 | jhiver | is there a way to distiguish a 'busy' as in 'the guy on the phone is chatting' from a 'fast busy' as in '503 no circuit' or '603 declined'? |
| 09:13.01 | jhiver | because it seems asterisk always returns 'busy' |
| 09:13.06 | jhiver | which is kind of silly |
| 09:13.19 | *** join/#asterisk [hC] (n=hardcore@S01060016b6b53c0c.vc.shawcable.net) |
| 09:14.43 | adeel | anyone know where i can find a 2u redundant power supply? |
| 09:14.52 | adeel | i seem to be having some trouble locating one |
| 09:15.01 | adeel | that isn't like 350 bucks or more |
| 09:17.10 | JT | it's usually easier to get a brand name server equipped with a redundant power supply than to buy an individual psu |
| 09:17.46 | *** join/#asterisk dominic1 (n=dob@213.221.82.242) |
| 09:17.48 | *** part/#asterisk dominic1 (n=dob@213.221.82.242) |
| 09:17.55 | adeel | yeah, i've tried that route...the server they spec out for me runs over 5,000, when i can build it myself for less than 2,300 |
| 09:18.29 | JT | with not exactly the same warranty |
| 09:18.38 | JT | at least build it with a supermicro case or similar |
| 09:19.08 | adeel | yeah i'm looking for something like that...i found a supermicro barebones that has the chasis/ps i want...but the barebones is like 1800 bucks |
| 09:19.25 | JT | does it have the motherboard? |
| 09:19.32 | adeel | not the board i want, but it has a board |
| 09:19.47 | *** join/#asterisk glen2 (n=glen@212.54.184.253) |
| 09:19.55 | JT | surely they have a combo that comes close to your requirements |
| 09:19.58 | JT | they have so many |
| 09:20.46 | adeel | i'll keep looking, if you know of any place that specializes in supermicro, it'd be appreciated....i've checked newegg, ewiz, pricewatch, no real luck |
| 09:21.08 | JT | i have no idea what country you're in |
| 09:21.45 | adeel | USA |
| 09:21.47 | adeel | california |
| 09:21.58 | adeel | actually, i think i may have found something on their site |
| 09:22.11 | JT | i'd say find a couple of candidates o their site, then go asking some distributors |
| 09:22.50 | *** join/#asterisk Iamnacho (i=Iamnacho@ip68-103-153-140.ks.ok.cox.net) |
| 09:23.06 | adeel | thanks, will do that |
| 09:23.10 | cmantito | adeel: rackmountpro.com |
| 09:23.41 | adeel | cmantito, thanks, will check |
| 09:23.43 | *** join/#asterisk sergey (n=sergey@91.189.233.66) |
| 09:23.45 | cmantito | np |
| 09:23.56 | cmantito | just shouted over to our hw guy, he reccomended there :P |
| 09:25.09 | adeel | haha nice |
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| 09:29.50 | defswork | is there anything obvious that would explain oneway voice only between to sip clients on the same server ? |
| 09:30.10 | defswork | only from one handset and only internally |
| 09:30.24 | cmantito | is NAT involved? |
| 09:30.31 | defswork | no |
| 09:30.35 | defswork | flat lan |
| 09:30.39 | cmantito | hmo |
| 09:30.41 | cmantito | *hm |
| 09:30.45 | cmantito | bad handset? :p |
| 09:30.49 | *** join/#asterisk KermitTheFragger (n=siepkes@118-197.bbned.dsl.internl.net) |
| 09:31.29 | defswork | it's only started happening today |
| 09:31.34 | cmantito | that's quite strange |
| 09:31.37 | cmantito | what changed? |
| 09:31.51 | defswork | thought yesterday the receptionist reckon betweeen 1 and 2 people sounded like they were underwater |
| 09:31.59 | defswork | nothing has changed |
| 09:32.18 | cmantito | bad handset/swtich/switchport/routing rule/STP rules? |
| 09:33.01 | defswork | no problem on external calls though |
| 09:33.05 | cmantito | hrmmm |
| 09:33.13 | cmantito | unusual surge in internal network traffic of late? |
| 09:33.42 | cmantito | with internal calls it's twice the network traffic of an outbound call (generically speaking) |
| 09:33.51 | *** join/#asterisk vale-ICS (n=vale@icsnet.demon.co.uk) |
| 09:34.13 | defswork | theres a pc daisy chained off the handset |
| 09:34.30 | cmantito | check if it's doing anything unusual, virii, spyware, or even games ;) |
| 09:34.32 | defswork | but all that could be doing is maybe a virus checker update |
| 09:34.34 | cmantito | streaming media, etc. |
| 09:35.01 | defswork | even that would be limited to their broadband speed though which is onyl 2 meg |
| 09:35.40 | cmantito | is there any other possible internal network traffic that's ... more than usual? |
| 09:36.18 | defswork | theres shouldn't be |
| 09:36.23 | defswork | I'll have to go there |
| 09:36.32 | cmantito | wireshark =p |
| 09:36.37 | cmantito | over an ssh tunnel lol |
| 09:38.20 | sweeper | cmantito: uh, maybe a reinvite is happening, and the phones are using a different codec? |
| 09:38.40 | sweeper | oh, oneway voice... |
| 09:38.41 | cmantito | yeah but all of a sudden? that's why I asked if anything changed ;) |
| 09:39.18 | sweeper | yea, try a different handset |
| 09:39.58 | cmantito | here's a question..how do you all organize complex dialplans? split it into multiple files? pull from MySQL? |
| 09:40.19 | cmantito | I have a dialplan with about 8 contexts and it's growing, rapidly, and I dunno how to keep it organized and readable |
| 09:40.45 | cmantito | right now it's split into included files -- <context_name>.extensions.conf // <context_name>.voicemail.conf // <context_name>.sip.conf, etc. |
| 09:40.53 | cmantito | but even that feel messy lol |
| 09:41.47 | defswork | this install has 6 analog lines |
| 09:42.02 | defswork | and a sangoma a200 |
| 09:42.18 | defswork | it's not a problem that it is simply plugged into a master socket is it ? |
| 09:42.33 | cmantito | I'll brb |
| 09:42.50 | defswork | (as opposed to being kroned in direct) |
| 09:43.50 | *** part/#asterisk mattzerah (n=matt@121.50.220.20) |
| 09:45.52 | cmantito | back |
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| 09:47.59 | cmantito | I'm not certain :S if it happened randomly then it's probably nothing software related |
| 09:48.18 | defswork | that's my feeling too |
| 09:48.33 | defswork | they've got one duff line by the sounds of it |
| 09:49.12 | defswork | and I've got a problem with hangup detection |
| 09:49.21 | cmantito | oh? |
| 09:49.39 | defswork | works on some lines - not on others |
| 09:49.44 | cmantito | *shrug* |
| 09:49.49 | cmantito | I'm trying to organise my dialplan :P |
| 09:49.50 | defswork | I've got the telco to check that clear disconnect is on all lines |
| 10:15.09 | *** join/#asterisk craigk (n=craigk@58.174.150.119) |
| 10:23.37 | *** join/#asterisk borgie (n=borgie@217.150.124.10) |
| 10:23.43 | borgie | Hello people |
| 10:23.57 | borgie | i have a problem which is a bit out of my depth |
| 10:24.07 | *** join/#asterisk keulin (n=cray@ifth-pdcr2.infotheme.fr) |
| 10:24.41 | borgie | One of our machines suffered a power failure, and now i can't load zaptel. I just get "/dev/zap" not found when running the init script |
| 10:25.22 | badcfe | when im rotating the asterisk logs, asterisk keep wrinting to the old files even if i do logger reload. how do i make asterisk re-open the logfiles ? |
| 10:26.44 | defswork | badcfe: modules aren't loaded |
| 10:26.54 | defswork | oops - borgie even |
| 10:27.29 | defswork | borgie: could be a fried card if they used to autoload |
| 10:27.39 | borgie | defswork: damn, what's what i was scared of |
| 10:27.43 | defswork | otherwise try loading them manually |
| 10:27.50 | defswork | and see if you get errors |
| 10:27.50 | borgie | defswork: how would i load htem manually |
| 10:27.56 | defswork | borgie: what vard ? |
| 10:27.58 | defswork | card* |
| 10:27.58 | borgie | modprobe zaptel etc? |
| 10:28.04 | borgie | defsword: TE220P |
| 10:28.36 | defswork | not sure what modules te220p uses - I've not used one yet |
| 10:28.46 | defswork | do you see it in lspci ? |
| 10:28.52 | borgie | defswork: yes i do |
| 10:29.01 | defswork | not fried then :) |
| 10:29.08 | borgie | okay, thats reassuring |
| 10:29.19 | borgie | i've been working with asterisk and zaptel for over a year now but this has me stumped |
| 10:29.41 | borgie | i'll try and load the modules manually and let you know |
| 10:29.51 | defswork | nothing interesting in /var/log/message from startup ? |
| 10:29.56 | *** join/#asterisk sergee (n=serg@voip1.west-call.com) |
| 10:30.10 | borgie | ill take a look, just rebooted the machine |
| 10:30.10 | borgie | <PROTECTED> |
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| 10:31.45 | badcfe | defswork: ? what should i do |
| 10:32.07 | defswork | badcfe: does asterisk use syslog ? |
| 10:32.22 | badcfe | defswork: no, is that the only way to accomplish this? |
| 10:32.27 | defswork | no |
| 10:32.33 | *** join/#asterisk sasch (n=bo@host117-234-static.4-79-b.business.telecomitalia.it) |
| 10:32.36 | sasch | hi all |
| 10:32.44 | sasch | anyone use java-asterisk ?? |
| 10:32.45 | defswork | my logs rotate ok |
| 10:32.48 | badcfe | the anser is its not doing it via syslog, no |
| 10:33.03 | badcfe | defswork: u use syslog to write em? |
| 10:33.36 | defswork | no |
| 10:33.44 | borgie | defswork: okay, so it inserts the zaptel module |
| 10:33.53 | borgie | but the other modules wont insert becuase of missing /dev/zap |
| 10:33.57 | defswork | hang on - just looking at the script that manages it on a trixbox |
| 10:33.58 | badcfe | defswork: using logrotate? (i can see logrotate has moved the file and created a new one, but asterisks still pointing to the rotated one) |
| 10:34.14 | defswork | borgie: isn't /dev/zap provided by the card's modules ? |
| 10:34.30 | borgie | defswork: Ill show you the output |
| 10:35.23 | borgie | [root@punepbx01 ~]# modprobe wct4xxp t1e1override=0xFF |
| 10:35.24 | borgie | Notice: Configuration file is /etc/zaptel.conf |
| 10:35.24 | borgie | line 0: Unable to open master device '/dev/zap/ctl' |
| 10:36.02 | badcfe | defswork: heres my logrotate conf maybe im doing some config unfitted for asterisk? http://pastebin.ca/902291 |
| 10:36.50 | badcfe | borgie: do you have the zaptel source tree then you could do make install once again .. |
| 10:37.03 | borgie | ive already done that |
| 10:37.06 | badcfe | borgie: you tried doing modprobe your_thing manually? |
| 10:37.18 | badcfe | borgie: and "make config" |
| 10:37.25 | borgie | yea |
| 10:37.28 | borgie | but i will try again |
| 10:37.49 | borgie | after a complete clean insteall |
| 10:37.52 | borgie | it still does it |
| 10:37.53 | badcfe | http://pastebin.ca/902291 -- is this logrotate config fit for asterisk? |
| 10:38.48 | borgie | defswork: i have an output for you |
| 10:38.54 | borgie | of var/log/messages |
| 10:38.58 | borgie | and all seems normal |
| 10:49.22 | phix | G'day |
| 10:49.37 | phix | I finally have the TDM400p card so I can do some more testing |
| 10:49.54 | phix | someone said try compiling zaptel moduoles from source, what else should I try? asterisk from source too? |
| 10:51.32 | *** join/#asterisk Muazzam (n=ahgi@static-host210-2-171-178.link.net.pk) |
| 10:53.35 | borgie | [root@punepbx01 zaptel-1.4.8]# lsmod |grep zap |
| 10:53.35 | borgie | zaptel 189316 11 xpp,wcusb,pciradio,wcfxo,wctdm,wctdm24xxp,tor2,wct1xxp,wcte11xp,wcte12xp,wct4xxp |
| 10:53.35 | borgie | crc_ccitt 6337 1 zaptel |
| 10:53.41 | borgie | all the modules seem to be loaded. |
| 10:53.44 | phix | ice |
| 10:53.44 | borgie | but no /dev/zap?! |
| 10:59.21 | *** join/#asterisk craigk (n=craigk@58.174.150.119) |
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| 11:02.51 | *** join/#asterisk Azam (n=azamzia@58-65-160-140.nayatel.pk) |
| 11:03.15 | Azam | Hi can anyone please help me, i want to configure my asterisk behind NAT |
| 11:03.55 | Azam | I want my asterisk to send calls to another asterisk. My asterisk is behind NAT the other one is global |
| 11:04.51 | tzafrir | bogar, what kernel version? |
| 11:05.47 | tzafrir | uname -r |
| 11:06.01 | borgie | [root@punepbx01 log]# uname -r |
| 11:06.01 | borgie | 2.6.21-1.3194.fc7 |
| 11:06.20 | borgie | tzafrir: 2.6.21-1.3194.fc7 |
| 11:06.52 | tzafrir | borgie, do you see /sys/class/zaptel/zapctl ? |
| 11:07.44 | *** join/#asterisk lftsy (n=lftsy@anj75-1-81-57-0-22.fbx.proxad.net) |
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| 11:14.40 | borgie | tsafrir: hold on a moment, the box just went down |
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| 11:26.51 | borgie | Hey thanks for all of your help, figured the problem |
| 11:27.06 | borgie | the filesystem was corrupted, but very minorly, so forced an fsck and now is fine |
| 11:27.53 | b1ch0 | hi, after upgrading to 1.4.17 i dont have core call pickup (*8 ) enabled ... anyone knows how to re enable it or is it a version issue ? |
| 11:36.38 | sty|work | b1cho: what does it say in the asterisk console. when you type show features |
| 11:37.09 | sty|work | under Current |
| 11:37.33 | sty|work | b1ch0: . |
| 11:37.40 | *** join/#asterisk Sinar (n=bwoSinar@81.144.146.38) |
| 11:39.53 | Sinar | Hi guys. I'm using FastAGI to process my calls. I'm wondering if its possible to play out a voice menu whilst still allowing people to press the digits before its finished? At the moment I'm using "EXEC PLAYBACK localfile" followed by a "WAIT FOR DIGITS 5000". Background() looks like it passes the DTMF processing back to the Dialplan which seems to defeat the advantage of using AGI to create an interactive dialplan. |
| 11:39.57 | *** join/#asterisk GBR_ (n=gbr@200.103.96.98) |
| 11:40.27 | Sinar | Any ideas what I can use instead of Playback to allow me to break out of the message before its finished and process the DTMF digit pressed? |
| 11:44.21 | *** join/#asterisk adjohn (n=adjohn@38.99.101.133) |
| 11:47.51 | tzafrir | Sinar, Background ? |
| 11:48.22 | tzafrir | WaitExten? |
| 11:48.46 | Sinar | What are you trying to ask, tzafrir? |
| 11:49.10 | tzafrir | things to use instead of PlayBack |
| 11:49.31 | Sinar | I've not tried using Background yet, but I've read that it'll process the DTMF tones as an extension and then switch to that in the extensions list? |
| 11:49.43 | Sinar | which would redirect things away from the current AGI request |
| 11:50.04 | *** join/#asterisk anonymouz666 (n=anonymou@201.19.127.250) |
| 11:50.53 | Sinar | the example they give is : exten => s,1, Answer exten => s,2,Background(thanks) exten => 1,1, Goto(submenu,s,1) exten => 2, 1, Hangup |
| 11:51.06 | Sinar | which implies that Background will accept the Digit and then redirect to the rule in the dialplan accordingly |
| 11:52.16 | phix | k weird |
| 11:52.18 | Sinar | if I've just got a single dialplan rule which pushes out to my FastAGI application, then executing this command might force Asterisk to look for the number in the dialplan which would take control away from my AGI script... Maybe I just need to try it |
| 11:52.55 | phix | Under debian, I can only get 2 out of 3 FXS modules working on my TDM400p. Under Ubuntu all of them work. |
| 11:53.24 | phix | what would cause zaptel / asterisk not to pick up my 3rd module under a different distro? |
| 11:53.57 | Sinar | What I was hoping for was something like SAY NUMBER where you can specify escape digits to stop it saying them... so that my prerendered voice menu could be interrupted by the user pressing the right key |
| 11:54.09 | phix | The only difference is the asterisk version, in debian it is 1.2.x under ubuntu it is 1.4.x |
| 11:55.53 | *** join/#asterisk windback (n=windback@proxy.unc.edu.ar) |
| 11:56.11 | phix | is there much difference between asterisk 1.2.x and 1.4.x? |
| 11:56.18 | *** join/#asterisk Silent-X (n=silentx@unaffiliated/silent-x) |
| 11:56.43 | Sinar | So both Background and WaitExten appear to throw the control back to the dialplan rather than AGI |
| 11:58.22 | phix | hi |
| 11:59.17 | windback | I'm triyng to enable the sqlite module from the make menuselect of asterisk, It apears as XXX, the help, tellme that it depends on sqlite. I have installed sqlite package, but it continues as XXX. Can somebody helpme please?? I have also installed the libsqlite-dev and always I ran the ./configure script before run make menuselect |
| 12:01.02 | tzanger | morning |
| 12:01.14 | tzanger | windback: did you install the sqlite-devel package? |
| 12:01.25 | tzanger | generally speaking if you want to build something that *uses* something else, you need the -devel package too |
| 12:06.52 | tzafrir | (and don't forget to run ./configure after that) |
| 12:07.34 | windback | tzanger, thank you, it found !! |
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| 12:14.59 | windback | tzafrir, If i have an asterisk installed, and i want to install a new version, Do I have to do make unistall in the sources of the old version?? |
| 12:15.34 | windback | tzafrir, or just install over the old version with make install |
| 12:15.47 | windback | in the new sources? |
| 12:15.59 | tzanger | windback: I generally clean out sources |
| 12:16.08 | tzanger | but the installed binaries and sounds I leave in place and let them be overwritten |
| 12:16.23 | tzanger | the install part of the makefile is smart-ish and will warn you about extra binaries that were from the old system |
| 12:17.37 | *** part/#asterisk simbol76 (n=simbol@ip-212-18.sn1.eutelia.it) |
| 12:18.34 | windback | tzanger, what you mean with clean out sources?? |
| 12:18.50 | windback | tzanger, make uninstall?? |
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| 12:24.43 | tzanger | windback: no, I blow away the source tree |
| 12:24.47 | tzanger | (or rather build in another) |
| 12:25.03 | penguinFunk | is there a better application for call pickup? |
| 12:25.20 | penguinFunk | pickup_exten is not so good as it requires you to know the extensions that is ringing before you can pick it up |
| 12:25.47 | penguinFunk | is there an easy way to pickup any phone that is ringing? |
| 12:25.53 | penguinFunk | without knowing the extension? |
| 12:27.07 | penguinFunk | currently using: exten => _8.,1,Pickup(${EXTEN:1}) |
| 12:27.11 | penguinFunk | in extensions.conf |
| 12:28.02 | tzanger | penguinFunk: I just use *8 in features.conf and I can pick up any phone I'm in the ringgroup with |
| 12:28.03 | penguinFunk | maybe there is a wildcard for the Pickup application |
| 12:28.21 | penguinFunk | tzanger: and you do not have an entry in extensions.conf? |
| 12:28.26 | tzanger | nope |
| 12:28.32 | penguinFunk | hmm, that didn't work for me |
| 12:28.40 | penguinFunk | unkown extensions |
| 12:29.22 | sty|work | remove the _8 |
| 12:30.26 | *** join/#asterisk coppice (n=chatzill@175.203.17.210.dyn.pacific.net.hk) |
| 12:30.37 | penguinFunk | k thanx |
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| 12:32.33 | *** join/#asterisk Wayhigh (i=noid@www.kevinlynn.com) |
| 12:32.39 | Wayhigh | good morning everyone |
| 12:32.50 | *** join/#asterisk puzzled (n=patrick@puzzled.xs4all.nl) |
| 12:33.05 | penguinFunk | i got *8 in features.conf, but when i try and pickup a call using *8 i get 'unavailable' |
| 12:33.18 | penguinFunk | is this phone specific or something? |
| 12:33.41 | sty|work | what phone? |
| 12:33.56 | penguinFunk | ive also set pickupgroup=1 in sip.conf |
| 12:34.06 | penguinFunk | we're using snom300's, this should work with them right? |
| 12:35.11 | *** join/#asterisk cpm (n=Chip@pdpc/supporter/active/cpm) |
| 12:37.28 | sty|work | not sure |
| 12:40.30 | *** part/#asterisk PepOSX (n=angeldav@190.72.132.44) |
| 12:41.11 | *** part/#asterisk shtoom (n=godson@59.93.124.40) |
| 12:41.16 | sty|work | check the phones dialplan i guess |
| 12:41.28 | sty|work | make sure it has *. or *x in it |
| 12:41.33 | sty|work | at least |
| 12:41.45 | penguinFunk | if you use *8 then you do not need an entry in extensions.conf ? |
| 12:42.18 | penguinFunk | or if you dont use *8 (in features.conf) then you can do what i did in extensions.conf (exten => _8.,1,Pickup(${EXTEN:1})) |
| 12:46.40 | *** join/#asterisk DrAk0 (n=thinkpad@nelug/coreteam/luisjose) |
| 12:48.43 | plik | ~book |
| 12:48.43 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com, or see ~buybook |
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| 12:58.42 | *** mode/#asterisk [+o russellb] by ChanServ |
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| 13:39.16 | eric2 | if I have a string like: SIP/mgr-2005&SIP/4165551234@vpri is there a way to split it into an array inside a dial plan? |
| 13:39.42 | eric2 | need to run this through a macro |
| 13:39.55 | *** join/#asterisk [TK]D-Fender (n=joe@MTRLPQ02-1177745839.sdsl.bell.ca) |
| 13:41.12 | eric2 | if I have a string like: SIP/mgr-2005&SIP/4165551234@vpri is there a way to split it into an array inside a dial plan? |
| 13:42.03 | [TK]D-Fender | eric2: "core show function CUT" |
| 13:42.11 | eric2 | k, tx |
| 13:42.42 | [TK]D-Fender | eric2: Now's a good time for you to do "core show functions" and "core sho applications", because if you did you'd have found your answer all by yourself. |
| 13:43.06 | eric2 | no way!? |
| 13:43.08 | dmz | hey i'm having problems with my call queues |
| 13:43.18 | eric2 | lots of goodies in there :) |
| 13:43.21 | dmz | i upgraded from 1.2 -> 1.4 (and yes I changed all the areas I believe were effected by the upgrade) |
| 13:43.43 | dmz | but now when calls come in they don't "connect" with the agent or member (channel) they are trying to connect to |
| 13:44.00 | dmz | just get "dead air" and it never announces queue or connects/bridges in the caller |
| 13:44.17 | dmz | i'm about to go back to 1.2 :( |
| 13:44.53 | eric2 | I've used 1.2 but don't know enough about the differences between 1.2 & 1.4 |
| 13:45.04 | eric2 | keep moving forward... don't go back |
| 13:45.08 | eric2 | if you can help it.. |
| 13:45.18 | dmz | i'm losing about 20 calls / day :( |
| 13:45.20 | dmz | not good for business |
| 13:45.47 | eric2 | that is a problem |
| 13:45.57 | eric2 | don't you have a development area/machine? |
| 13:46.17 | dmz | it works sometimes but most of the time it just doesn't connect the caller; the queue shows the agent as busy and the caller still in the queue; so it sees the agent pick up just doesn't bridge the 2 together |
| 13:46.24 | *** join/#asterisk jm|home (n=jm|home@zen.jamiem.com) |
| 13:46.27 | dmz | i should :) but don't :( |
| 13:46.53 | dmz | everything is working great in 1.4 except for my call queues :( |
| 13:46.54 | russellb | what version? |
| 13:46.57 | eric2 | get one setup would be my advice, don't want to loose business |
| 13:47.13 | dmz | debian: Asterisk 1.4.17~dfsg-2+b1 built by buildd @ ninsei on a i686 running Linux on 2008-01-26 04:30:00 UTC |
| 13:47.16 | *** join/#asterisk inadaptado (n=matias@32.59.64.129) |
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| 13:47.41 | russellb | dmz: upgrade to 1.4.18 |
| 13:47.43 | dmz | i always see who is calling in & I can call back; so not losing the "call" just a pain cause i can't answer it when it comes in |
| 13:47.53 | Elfe | hi, I run an asterisk server behind a nat device (all ports forwarded) the server is doing a qualify to the external phone but the problem is that the server is sending an incorrect port 1024 in the Via and Contact header, the phone tries to answer to port 1024 which results in icmp unreachable, so is there a way to get asterisk to send the correct port in the header lines? (1.2.17) |
| 13:47.57 | dmz | russellb is this an issue w/1.4.17? |
| 13:48.19 | dmz | i saw that 1.2 had some calling issues w/a january date problem |
| 13:48.24 | russellb | plenty of issues :) |
| 13:48.43 | dmz | i'll harass the developer to update to 18 |
| 13:48.45 | russellb | there have been 175 fixes to asterisk 1.4 sine 1.4.17 |
| 13:48.49 | dmz | ah |
| 13:49.00 | russellb | asterisk moves _very_ quickly ... |
| 13:49.03 | dmz | ok let me go harass the developer; trying to avoid source compiles :) |
| 13:49.04 | dmz | yeah i know |
| 13:49.26 | dmz | but does this problem sound liek it's a real problem, or a config issue on my side? |
| 13:49.32 | *** join/#asterisk newbie1 (n=rose@202.161.189.44) |
| 13:49.48 | russellb | compilingdon't know ... but going to the latest version is always step 1 :) |
| 13:49.49 | [TK]D-Fender | dmz: True to form you have not shown us anything |
| 13:49.58 | dmz | heh thanks |
| 13:49.59 | russellb | compiling from source with asterisk really isn't a big deal |
| 13:50.16 | *** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 13:50.18 | dmz | i know but when you choose to use a distro it's nice to help it keep it's distro methods |
| 13:50.33 | dmz | D-Fender how do I show something when it is a connection problem in the pbx, i've described what the problem was |
| 13:51.11 | dmz | how can i describe it better to get away from my form of not providing enough info? |
| 13:51.15 | dmz | :-D |
| 13:52.25 | [TK]D-Fender | dmz: pastebin your configs, ueue status, member status, etc. Then do another with CLI output of a failed call, etc. Inlude debug info for your agents and veify their account setups as well. |
| 13:52.46 | *** join/#asterisk af_ (n=getsmart@88-149-240-211.dynamic.ngi.it) |
| 13:53.05 | dmz | what is pastebin url? |
| 13:53.33 | Qwell | ~pb |
| 13:53.34 | jbot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste |
| 13:53.36 | dmz | thanks |
| 13:56.57 | dmz | should I paste just my queue config, or what would be helpful? I have a somewhat large extensions.conf, i could paste portions of it but don't want to leave anything out and don't want to paste hundreds of lines of stuff that may/may not be helpful |
| 13:58.10 | Qwell | too much is better than not enough |
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| 14:02.04 | sabbir | hi, im having problem on asterisk realtime configuration |
| 14:02.28 | sabbir | can anyone help me ? |
| 14:03.46 | sabbir | i saw that the asterisk ARA was configured properly but when im going to register by iax client then getting this message " NOTICE[31747] chan_iax2.c: Restricting registration for peer 'user1' to 60 seconds (requested 300)" |
| 14:04.33 | *** join/#asterisk simbol76 (n=simbol@ip-212-18.sn1.eutelia.it) |
| 14:06.21 | Elfe | http://www.pastebin.ca/902425 would be an example for my port 1024 problem |
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| 14:13.52 | dmz | how do i kill a call sitting dead in a queue? |
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| 14:15.16 | sabbir | will anyone plx respond me :-X |
| 14:16.00 | sabbir | :'( |
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| 14:16.22 | *** mode/#asterisk [+o russellb] by ChanServ |
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| 14:18.28 | frogzoo | anyone managed to make a .deb from 1.6? |
| 14:18.51 | dmz | wierd not getting any debug info in one box :( |
| 14:19.51 | dmz | http://pastebin.com/d1c797d4a |
| 14:19.52 | Elfe | sabbir: I guess your client is trying to register for 5 minutes while the server only allows 60 seconds (or check maxregexpire in the iax config part) |
| 14:20.01 | dmz | ok that's the configs & details for my queue problems |
| 14:20.31 | sabbir | Elfe:: let me check |
| 14:20.49 | sabbir | can u refer any doc for that purpose ? |
| 14:21.04 | mvanbaak | dmz: if you dont get debug output check /etc/asterisk/logger.conf |
| 14:21.09 | defswork | channel.c: Didn't get a frame from channel: SIP/203-b7909200 < any ideas what that might be ? |
| 14:21.12 | dmz | k |
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| 14:22.40 | sabbir | Elfe: but i did not get this message when the configuration was not as ARA, static configuration did no show this message |
| 14:22.48 | sabbir | Elfe: what do u think about that |
| 14:22.57 | dmz | ah yeah that was it, just need to wait for call to timeout now an di can get more debug info |
| 14:24.48 | dmz | should i get more info? lots more available :) any thoughts would be helpful |
| 14:24.55 | dmz | i'm 90% sure it's something i did :) |
| 14:24.56 | Elfe | sabbir: can't help with that :( |
| 14:25.05 | sabbir | :) |
| 14:25.13 | sabbir | let me check |
| 14:25.30 | dmz | wierd i can add/remove members from queue in run-time just can't remove a caller |
| 14:25.43 | sabbir | can u give ur email so that i can send email if u can then reply me , nothing more |
| 14:25.57 | *** join/#asterisk adjohn (n=adjohn@p6081-ipad53marunouchi.tokyo.ocn.ne.jp) |
| 14:26.39 | sabbir | Elfe:: i have changed the maxregex , did not get message now |
| 14:27.06 | sabbir | Elfe: is there any command to get the current loged in users ? |
| 14:27.09 | *** join/#asterisk drako (n=thinkpad@nelug/coreteam/luisjose) |
| 14:27.42 | Elfe | sip show peers works, so maybe iax show peers as well |
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| 14:29.49 | sabbir | Elfe:: will u see the extension and the iax user tables , i think i have sometime wrong in the table data ??? can i ? |
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| 14:34.57 | Elfe | sry don't know anything about asterisk and db usage |
| 14:35.37 | *** join/#asterisk mmlj4 (n=jkelly@ip70-171-94-246.no.no.cox.net) |
| 14:35.41 | sabbir | great thanks for the help Elfe :-D |
| 14:35.58 | sabbir | Elfe:: let me check if anyone can help |
| 14:37.30 | sabbir | Can anyone help on ::[Feb 13 15:19:36] NOTICE[2932]: chan_iax2.c:7785 socket_process: Rejected connect attempt from 192.168.1.20, request '2222@xxxx' does not exist |
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| 14:38.37 | sabbir | Can anyone help on ::[Feb 13 15:19:36] NOTICE[2932]: chan_iax2.c:7785 socket_process: Rejected connect attempt from 192.168.1.20, request '2222@xxxx' does not exist .its shown when i was calling another user with extension 2222 for xxxx peers and context is mycontext |
| 14:40.53 | *** join/#asterisk putnopvut (n=putnopvu@216.207.245.1) |
| 14:40.59 | sabbir | may be i have configured wrong . so i have got this error |
| 14:40.59 | sabbir | [Feb 13 15:19:36] NOTICE[2932]: chan_iax2.c:7785 socket_process: Rejected connect attempt from 192.168.1.20, request '2222@xxxx' does not exist |
| 14:40.59 | sabbir | Tx-Frame Retry[000] -- OSeqno: 001 ISeqno: 002 Type: IAX Subclass: REJECT |
| 14:40.59 | sabbir | <PROTECTED> |
| 14:40.59 | sabbir | <PROTECTED> |
| 14:41.01 | sabbir | <PROTECTED> |
| 14:41.03 | sabbir | Rx-Frame Retry[ No] -- OSeqno: 002 ISeqno: 002 Type: IAX Subclass: ACK |
| 14:41.05 | sabbir | <PROTECTED> |
| 14:42.25 | dmz | more debug info: http://pastebin.com/d6189c428 |
| 14:42.53 | dmz | so if anyone has any suggestions i'd be eternally grateful :) |
| 14:44.19 | *** join/#asterisk mchou (n=mchou@c-71-198-127-234.hsd1.ca.comcast.net) |
| 14:44.53 | dmz | could this be it?? : # |
| 14:44.53 | dmz | Feb 13 14:37:44] DEBUG[10594] chan_iax2.c: Ooh, voice format changed to 4 |
| 14:44.53 | dmz | # |
| 14:44.53 | dmz | [Feb 13 14:37:44] DEBUG[11645] channel.c: Internal timing is disabled (option_internal_timing=0 chan->timingfd=-1) |
| 14:44.53 | dmz | # |
| 14:44.55 | dmz | [Feb 13 14:37:44] DEBUG[11645] app_queue.c: Dunno what to do with control type -1 |
| 14:45.11 | dmz | meeting time, guess i'll ask again in an hour |
| 14:45.19 | defswork | I'm getting a load of weird problems at a new install |
| 14:45.30 | *** join/#asterisk tnt_ (n=tnt_@8.253-244-81.adsl-static.isp.belgacom.be) |
| 14:46.07 | tnt_ | Hi. What is the best option to handle roaming extensions with asterisk ? (people 'logging' in and out. Possibly several user on one phone ...) |
| 14:46.34 | iCEBrkr | roaming extensions? |
| 14:46.43 | mchou | Recently got a pap2 and looking for voip provider after trying out ekiga softphone. anyone have any experience/feedback, good or bad, regarding diamondcard.us? |
| 14:47.44 | tnt_ | iCEBrkr: I'm not sure of the 'exact' name. For example, I go at work, then sit up at any workbench, then compose a special number on the phone and 'login' my extension on that phone so that all call for me are routed to that phone. |
| 14:49.14 | iCEBrkr | So you want people to be able to 'login' at different phones without moving the phone. |
| 14:49.35 | iCEBrkr | So if Joe sits at Desk A, he can login and get calls.. But tomorrow, Joe sits at Desk B.. |
| 14:49.41 | *** join/#asterisk Faithful (n=Faithful@ns.linuxterminal.com) |
| 14:49.50 | tnt_ | iCEBrkr: Yes, precisely. |
| 14:49.59 | iCEBrkr | tnt_: Sounds like an 'Agent' thing. |
| 14:50.29 | tnt_ | iCEBrkr: Ok, thanks for th keyword, I'll search using that. |
| 14:50.37 | iCEBrkr | tnt_: I'm not really familiar with how the AgentLogin thing works. But you could www.voip-info.org and check it out |
| 14:51.24 | Sinar | http://www.voip-info.org/wiki/index.php?page=Asterisk+agents |
| 14:51.40 | iCEBrkr | I've been out of the loop for a good 7-8mo. Working for a crappy M$ shop has consumed a lot of my time and created a lot of headache so I don't geek when I get home. I escape realtiy. |
| 14:51.56 | iCEBrkr | But that'll all change come Monday. |
| 14:52.03 | Sinar | what's happening Monday? |
| 14:52.07 | iCEBrkr | New jobby job |
| 14:52.11 | iCEBrkr | Working with Asterisk. |
| 14:52.15 | Sinar | great! |
| 14:52.18 | iCEBrkr | I think so |
| 14:52.21 | russellb | iCEBrkr: cool ... who are you working for? |
| 14:52.47 | iCEBrkr | russellb: I'll be working with Kristian and AstLinux. |
| 14:52.51 | Sinar | My job's recently moved to needing to use Asterisk. Able to keep my Linux side happy as well as doing C# stuff with the other hand |
| 14:52.55 | russellb | iCEBrkr: cool :) |
| 14:53.23 | iCEBrkr | russellb: I think it's a double-win. So not only am I back in my element (linux/opensource), I get to work with some cool people. |
| 14:53.40 | Sinar | can tell from two sentences that its a welcome relief to you |
| 14:53.42 | russellb | yeah, sounds like it |
| 14:54.04 | iCEBrkr | I only took this job because of a friend. It's crappy classic ASP and they're moving to .NET |
| 14:54.11 | iCEBrkr | THe more I work with .NET the more I think it sucks. |
| 14:54.16 | iCEBrkr | I can't deal with working in the box. |
| 14:54.28 | iCEBrkr | When there are debugging issues and problems, there's no strace |
| 14:54.34 | iCEBrkr | There's no logging. |
| 14:54.35 | Sinar | like many big libraries it brings a whole heap of unknowns |
| 14:54.37 | iCEBrkr | It just sucks. |
| 14:54.48 | sabbir | Can anyone give me any reference for realtime configuration. i have configured that and mysql engin is load successfully but i think i have configured somethik wrong. so im getting message >>[Feb 13 15:19:36] NOTICE[2932]: chan_iax2.c:7785 socket_process: Rejected connect attempt from 192.168.1.20, request '2222@xxxx' does not exist |
| 14:54.49 | sabbir | Tx-Frame Retry[000] -- OSeqno: 001 ISeqno: 002 Type: IAX Subclass: REJECT |
| 14:54.49 | sabbir | Timestamp: 00011ms SCall: 00003 DCall: 15808 [192.168.1.20:4569] |
| 14:54.49 | sabbir | CAUSE : No such context/extension |
| 14:54.49 | sabbir | CAUSE CODE : 3 |
| 14:54.58 | iCEBrkr | I just know when there's an issue with a linux box, diagnosing the problem is a lot easier. |
| 14:55.04 | tnt_ | Yes, that agent stuff is definitly what I was looking for. I just needed the right 'keyword' :) Thanks. |
| 14:55.09 | Sinar | iCEBrkr: yes |
| 14:55.12 | iCEBrkr | tnt_: :) np |
| 14:55.13 | russellb | sabbir: stop pasting all of that stuff in here ... |
| 14:55.21 | russellb | sabbir: but it's just saying that the extension doesn't exist |
| 14:55.21 | sabbir | ok |
| 14:55.22 | iCEBrkr | sabbir: You're fired. |
| 14:55.32 | russellb | sabbir: realtime iax or extensions? |
| 14:55.37 | *** join/#asterisk FlyboySR22 (n=rsears@hq.fw.americanis.net) |
| 14:55.40 | sabbir | iax |
| 14:55.47 | russellb | ok, well the problem is your dialplan |
| 14:56.01 | *** join/#asterisk jm|home (n=jm|home@zen.jamiem.com) |
| 14:56.06 | sabbir | <PROTECTED> |
| 14:56.17 | russellb | um ... you need to make 2222@xxxx exist. |
| 14:56.23 | iCEBrkr | The other problem with working at this current place is that people dont' even understand how the internet works (dns/email/http). We sell a web-based CMS/information portal!!!! |
| 14:56.31 | iCEBrkr | It makes me itch |
| 14:56.33 | russellb | [xxxx] exten => 2222,1,OMGHI2U |
| 14:57.12 | sabbir | its in db like >>1 tb_iax_users 1111 1 Dial IAX2/user1@xxxx/${EXTEN} |
| 14:57.37 | *** join/#asterisk lmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 14:57.37 | *** mode/#asterisk [+o lmadsen] by ChanServ |
| 14:57.40 | Sinar | iCEBrkr: Sometimes makes me wonder how people get jobs. Must be lying on their resumes |
| 14:58.34 | sabbir | this is the reference what i have followed http://www.voip-info.org/wiki/view/Asterisk+RealTime+IAX |
| 14:59.41 | defswork | anyone here in the uk used sangoma a200 ? |
| 15:00.00 | iCEBrkr | haha |
| 15:00.20 | *** join/#asterisk hijacked (n=argh@66.255.220.17) |
| 15:00.25 | *** join/#asterisk bkw_ (n=brian@adsl-70-142-54-162.dsl.tul2ok.sbcglobal.net) |
| 15:00.45 | iCEBrkr | bkw??? Welp, there goes the neighborhood. |
| 15:01.00 | drmessano | I had a dream someone was talking about me |
| 15:01.02 | drmessano | WHO WAS IT? |
| 15:01.42 | sabbir | russellb:: the configuration for iax.conf file in the http://www.voip-info.org/wiki/view/Asterisk+RealTime+IAX is OK?? |
| 15:03.15 | sabbir | russellb: is the xxxx part is ok as peer ? |
| 15:03.47 | russellb | i ... don't know |
| 15:03.59 | russellb | i can't look at it any longer, sorry |
| 15:05.02 | drmessano | If russellb looked at one of my config files, I wouldnt wash my monitor for a month |
| 15:05.05 | sabbir | russellb:: thanks |
| 15:05.06 | drmessano | You should feel honored |
| 15:05.15 | mvanbaak | lol drmessano |
| 15:05.50 | drmessano | ~russellb |
| 15:05.51 | jbot | russellb is, like, Russell Bryant <russell@digium.com>, or not a fan of jbot |
| 15:06.10 | *** join/#asterisk fskrotzki (n=fskrotzk@host198.textwise.com) |
| 15:06.22 | *** join/#asterisk fskrotzki_ (n=fskrotzk@cpe-74-74-245-250.rochester.res.rr.com) |
| 15:06.44 | *** join/#asterisk jameswf-home (n=james@dsl093-157-131.phx1.dsl.speakeasy.net) |
| 15:06.51 | russellb | drmessano: <3 |
| 15:07.05 | timeshell | How much longer will 1.6 be in beta? |
| 15:07.19 | russellb | somewhat undefined ... but the results we have seen have been good |
| 15:07.45 | jameswf | Most products remain in beta until they are no longer in beta |
| 15:08.03 | russellb | jameswf: good answer |
| 15:08.15 | drmessano | It's always funny |
| 15:08.18 | russellb | it will be in beta until the beta period is deemed complete |
| 15:08.44 | jameswf | at which point it will enter RC status which also is indeffinate |
| 15:08.52 | drmessano | "Hurry up, faster" Then when it comes out of beta and there's the obvious .0 bugs that go with any project "You guys suck, should have kept it in beta longer" |
| 15:08.57 | drmessano | You can't win.. ever |
| 15:09.13 | russellb | it's especially arbitrary for an open source project, where there is no product release schedule to meet |
| 15:09.25 | russellb | no market requirements pushing things for a date ... |
| 15:09.29 | jameswf | there are no bugs, our software is perfect, it must be you, whatever you are doing STOP IT |
| 15:09.33 | russellb | it's done when the technology is there |
| 15:09.34 | drmessano | lol |
| 15:09.45 | drmessano | jameswf: pastebin your config files please |
| 15:09.53 | drmessano | jameswf: what version are you on |
| 15:09.59 | drmessano | jameswf: have you rebooted? |
| 15:10.18 | jameswf | I am running web 4.0 alpha is asterisk compatible |
| 15:10.28 | drmessano | ha |
| 15:10.37 | drmessano | ~jameswf |
| 15:10.37 | jbot | somebody said jameswf was he has way to much time on his hands, or a GOD |
| 15:10.44 | russellb | the shinanigans level in this channel is quite high :) |
| 15:10.47 | drmessano | I need to change that |
| 15:10.56 | drmessano | jbot: forget jameswf |
| 15:10.56 | jbot | i forgot jameswf, drmessano |
| 15:11.09 | drmessano | jbot: jameswf loves unsolicited technical support |
| 15:11.28 | drmessano | jbot: jameswf loves unsolicited technical support |
| 15:11.32 | drmessano | ~jameswf |
| 15:11.33 | Qwell | fail |
| 15:11.38 | drmessano | fail? |
| 15:11.40 | drmessano | :( |
| 15:11.45 | jameswf | jbot responds to is |
| 15:11.56 | tzafrir | jbot, no, jameswf is <reply> jameswf loves unsolicited technical support |
| 15:11.57 | jbot | tzafrir: okay |
| 15:11.57 | russellb | jbot: jameswf is <reply> jameswf loves unsolicited technical support |
| 15:11.58 | jbot | i already had it that way, russellb |
| 15:11.58 | drmessano | oh |
| 15:12.03 | russellb | darn, i lose |
| 15:12.06 | drmessano | duh, ok |
| 15:12.11 | drmessano | ~jameswf |
| 15:12.11 | jbot | jameswf loves unsolicited technical support |
| 15:12.11 | jameswf | wow that was cool |
| 15:12.16 | drmessano | got it |
| 15:12.54 | russellb | jbot: drmessano is also a jbot junky in training |
| 15:12.55 | jbot | russellb: okay |
| 15:13.08 | drmessano | ~drmessano |
| 15:13.09 | jbot | methinks drmessano is the leading cause of censorship in #asterisk, or a jbot junky in training |
| 15:13.09 | russellb | hrm, junky is also the nick of someone ... that doesn't go well |
| 15:13.26 | jameswf | I think the hologen fluid in my flux cap is low would that cause asterisk to dial 900 numbers at 3am |
| 15:14.29 | drmessano | Blackberry's suck |
| 15:14.33 | drmessano | OT, but FWIW |
| 15:15.01 | jameswf | someone in the lists said his box calls the equiv of 911 I wonder if that is his box crying for help |
| 15:15.15 | drmessano | "Oops sorry we took down your business while upgrading from 2.7.21 to 2.7.22 Mr HIGHPOWERBUSINESSMANPAYINGCUSTOMERGUY" |
| 15:15.52 | mchou | haha, has RIM provided an explanation yet? |
| 15:16.02 | drmessano | Botched upgrade |
| 15:16.02 | jameswf | My blackberry has memory leaks, when a blackberry runs out of memory does it close apps? NO it deletes text messages and call logs, good job rim |
| 15:16.15 | drmessano | A f***** "Oops, sorry" |
| 15:16.31 | drmessano | Like the guy pulling the plug out of the runway lights in the movie "Airplane" |
| 15:16.51 | drmessano | Actually, that was a "Just kidding" |
| 15:17.08 | jameswf | fortunately we are not on a BIS server cause we dont do windoze here |
| 15:18.07 | *** join/#asterisk javar (n=javar@69.79.134.24) |
| 15:18.30 | drmessano | Mobile VoIP back to the Asterisk box at "home" along with ActiveSync back to a LINUX box at "home" would kill |
| 15:19.02 | drmessano | All these windows based phones with activesync and no serious ActiveSync killer for Linux? |
| 15:19.13 | jameswf | when hired at research in motion is that considdered getting a rim job |
| 15:19.21 | drmessano | lol |
| 15:19.22 | Qwell | ... |
| 15:19.25 | mchou | haha |
| 15:19.31 | drmessano | damn right |
| 15:19.34 | Qwell | jameswf: welcome to bash.org |
| 15:19.46 | mchou | a canadian rim job no less :) |
| 15:19.52 | russellb | O.O |
| 15:20.02 | mchou | that's gotta be exciting :) |
| 15:20.06 | jameswf | is that different than... oh never mind |
| 15:20.11 | *** join/#asterisk anotherkarvan (n=kvirc@213.170.201.2) |
| 15:20.23 | Qwell | jameswf: do you want me to change your nick when I submit this? :p |
| 15:20.29 | Qwell | to, err, protect the guilty |
| 15:20.36 | jameswf | if you go up north to the mugeon rim there is a store called rim liquer |
| 15:20.45 | *** join/#asterisk saftsack (n=oliver@p54A710DE.dip0.t-ipconnect.de) |
| 15:20.48 | anotherkarvan | Can anyone tell me if Asterisk 1.4.x supports VAD and CNG? And to what degree? |
| 15:20.56 | drmessano | change the nick to "russellb" |
| 15:21.23 | jameswf | my name is ron paul and i approved this message |
| 15:21.27 | drmessano | lol |
| 15:21.45 | *** join/#asterisk tobias (n=tobias@cpe-066-026-085-055.nc.res.rr.com) |
| 15:21.46 | [TK]D-Fender | anotherkarvan: "no" , "zero degrees" |
| 15:22.14 | drmessano | Alright. time to head to the office.. if that drmessano-LT guy comes in here, ban him.. he's a loser |
| 15:22.19 | Qwell | http://qdb.us/141703 |
| 15:22.38 | [TK]D-Fender | mchou: Yes RIM makes the blackberry phones and is a Canadian company. You are quite astute. |
| 15:23.19 | mintee | anyone ever setup the wanpipe modules on debian? |
| 15:27.20 | anotherkarvan | [TK]D-Fender: Thanks for the reply. |
| 15:30.30 | *** part/#asterisk anotherkarvan (n=kvirc@213.170.201.2) |
| 15:41.50 | *** join/#asterisk Winkie (n=urmom@ur.fa.gs) |
| 15:42.16 | *** join/#asterisk fskrotzki_ (n=fskrotzk@cpe-74-74-245-250.rochester.res.rr.com) |
| 15:43.01 | *** join/#asterisk angler (n=angler@pdpc/sponsor/digium/angler) |
| 15:43.01 | *** mode/#asterisk [+o angler] by ChanServ |
| 15:43.02 | *** join/#asterisk fskrotzki (n=fskrotzk@host.textwise.com) |
| 15:44.43 | b11d | if I run 'ztcfg' and then 'module reload chan_zap.so' -- Asterisk should get any changes i've made to zapata.conf right? |
| 15:44.45 | *** join/#asterisk flujan (n=flujan@200-160-115-020.static.spo.ctbc.com.br) |
| 15:44.58 | b11d | I dont want to cold restart asterisk, but I do need to make zapata.conf changes |
| 15:45.38 | *** join/#asterisk b1ch0 (i=b1ch0@static-200-105-150-37.acelerate.net) |
| 15:46.42 | jameswf | b11d: sometimes not if you dont see changes restart |
| 15:46.59 | b1ch0 | after upgrading to 1.4.17 the call pickup core *8 is not present anymore |
| 15:47.03 | b1ch0 | any idea ? |
| 15:47.20 | jameswf | upgrade to 1.4.18 |
| 15:47.23 | *** join/#asterisk adjohn (n=adjohn@p6081-ipad53marunouchi.tokyo.ocn.ne.jp) |
| 15:47.35 | b11d | james.. i want to modify the tx/rx gain on a zap channel.. i dont see that in 'zap show channel' so im not certain I can tell if the change was picked up or not. |
| 15:47.38 | jameswf | check features.conf |
| 15:47.42 | b1ch0 | is it a bug ? |
| 15:47.46 | *** join/#asterisk ajohnson (n=ajohnson@63.147.46.186) |
| 15:50.25 | jameswf | could be a poorly done upgrade, could be allot of things... |
| 15:51.19 | b11d | i'd kill for a way to easily filter asterisk CLI output :( |
| 15:52.04 | b11d | or.. maybe instead of killing my way to a solution, i could work on that myself.. |
| 15:52.29 | tzafrir | ztcfg is not related to zapata.conf |
| 15:52.32 | *** join/#asterisk michael-i (n=michael-@141.41.40.225) |
| 15:52.46 | b11d | i thought i had to run ztcfg when adding or removing channels in zapata.conf |
| 15:53.31 | b11d | ztcfg just looks at zaptel.conf then? |
| 15:53.47 | jameswf | DANGER a haxor can steal root access to your box if they have logged in as another user and run a small program..... If someone is in your pc dont you have bigger concerns |
| 15:54.07 | tzafrir | b11d, right |
| 15:54.28 | b11d | ok.. still.. any way to tell if asterisk modified the tx/rx gain on a zap channel when issuing a reload on chan_zap? |
| 15:55.50 | [TK]D-Fender | b11d: if its modified from the last that * used, no. You can merely see what it says right then and there. |
| 15:56.39 | jameswf | wow I have kernel updates availible I wonder if it is for that... |
| 15:57.18 | b11d | i see.. |
| 15:57.28 | b11d | thanks. |
| 15:57.34 | b11d | Damn FAX problems to hell :) |
| 15:57.39 | jameswf | OHZ NOZ tzafrir is going to steal my pron |
| 15:57.54 | b11d | PRI -> Asterisk -> T1 -> Rhino CB-24 -> FAX |
| 15:58.06 | b11d | i constantly get "poor line quality" fax errors.. |
| 15:58.15 | b11d | its not being converted to SIP or anything.. |
| 15:58.18 | b11d | no EC anywhere |
| 15:58.36 | jameswf | b11d: are you raising or dropping your gains |
| 15:58.51 | b11d | i had them set to zero for both, just modified them now to test.. |
| 15:58.59 | *** join/#asterisk klauwhamer (n=felixdhc@ipd50af070.speed.planet.nl) |
| 15:59.29 | jameswf | ~faxing |
| 15:59.30 | jbot | methinks faxing is 8% knowledge, 5% skill, 11% luck, and 76% voodoo |
| 15:59.30 | b11d | sangoma claims on their asterisk fax page that I should have txgain = 8 and rxgain = 1 -- i set 7 and 1 to test. Although I doubt it will help any. |
| 15:59.36 | b11d | well im working voodoo right now.. |
| 15:59.49 | anonymouz666 | hahahaha |
| 15:59.57 | anonymouz666 | nice one, jbot :D |
| 15:59.57 | jameswf | I would think jumping gains would cause clipping |
| 16:00.21 | coppice | b11d: was that sangoma recommendation written on 1st April? |
| 16:00.35 | b11d | its here.. http://wiki.sangoma.com/wanpipe-linux-asterisk-faxing |
| 16:01.01 | coppice | yeah, I've seen it. its stupid |
| 16:02.07 | b11d | yeah.. I need to do their clock recommendation though.. that actually makes sense. |
| 16:02.15 | jameswf | this is why you dont let windows people near linux bah |
| 16:02.53 | *** join/#asterisk glen2 (n=glen@212.54.184.253) |
| 16:02.54 | coppice | b11d: are both T1s on the same card? |
| 16:02.56 | b11d | yes |
| 16:03.19 | b11d | its a Sangoma A104d.. PRI on port1 (wag1) and T1 to CB24 on port2 (wbg1) |
| 16:03.19 | coppice | then sync to the PSTN, and it should work. |
| 16:03.37 | b11d | well I thought it was.. but looking at my wanpipe1.conf, it isnt.. |
| 16:03.42 | b11d | so.. i'll be modifying those. |
| 16:03.55 | b11d | but.. problem is I cant drop my phone system during the day while people are here.. :( |
| 16:04.04 | b11d | losers.. why cant everyone just go home when Im working on a problem? |
| 16:04.04 | b11d | :) |
| 16:04.22 | jameswf | b11d: just kill it tell em it was terrorist but they are gone now |
| 16:05.00 | b11d | haha.. I'll just say "remember 9/11" and it'll all go away |
| 16:05.10 | b11d | but i'll follow it up with a "ron paul 2008" and they'll come back.. |
| 16:05.42 | jameswf | tell them had they voted for ron paul there would be no issues |
| 16:06.23 | b11d | haha agreed |
| 16:06.23 | jameswf | ron paul can to t.3a translation in his head and perfectly put it on paper with a crayon |
| 16:06.27 | b11d | Ron Paul for FAXing 2008 |
| 16:06.32 | jameswf | *t.38 |
| 16:06.42 | *** join/#asterisk seanbright (n=elixer@65.207.74.18) |
| 16:07.44 | b11d | haha |
| 16:08.02 | jameswf-home | hmm |
| 16:08.03 | b11d | im going to schedule a restart of my PBX at 10:30 to fuck with my timing sources.. |
| 16:09.39 | [TK]D-Fender | b11d: No, just tell them it is to accomodate another unwarranted wire-tapping order that will soon to be granted retroactive immunity from prosecution. That and if they show the slightest sign of concern they will be helping the "terrists" and the MIBs will get them in their sleep. |
| 16:09.55 | jameswf | updating my kernel.... you only live once |
| 16:10.18 | [TK]D-Fender | jameswf: But you can die a little every day for a long time. |
| 16:10.44 | [TK]D-Fender | jameswf: maybe even in bigger steps. That line can be stretched so very far.... |
| 16:10.51 | jameswf | every time Bush says Nuclear god kills a bunny |
| 16:11.40 | jameswf | the trixbox bunny is over 28 and therefore cant be drafted |
| 16:11.55 | b11d | TK.. what an excellent excuse. I'll be sure to include it in my email :) |
| 16:12.02 | mintee | b11d, i see your using the wanpipe mods... have you gotten around the kernel panic upon unloading the modules? |
| 16:12.15 | b11d | i dont get a kernel panic.. |
| 16:12.21 | b11d | I did back in 2.x but not in 3.x anyway |
| 16:12.22 | mintee | 2.6 kernel? |
| 16:12.28 | b11d | i run FreeBSD, not Linux. |
| 16:12.35 | Nugget | Yay FreeBSD. |
| 16:12.38 | mintee | ah |
| 16:13.13 | mintee | that's an idea. fbsd ports up to date rather well with asterisk, etc? |
| 16:13.19 | mintee | or do you manually build? |
| 16:13.23 | b11d | i dont use the ports.. im a source man. |
| 16:13.27 | mintee | ; |
| 16:13.29 | mintee | gotcha |
| 16:13.41 | b11d | im really not a fan of the 'ports' system personally.. |
| 16:13.51 | b11d | usually out of date.. or adds out of date dependancies, which anger me. |
| 16:14.01 | mintee | i like ports for certain things, but i don't like pkg's |
| 16:14.02 | b11d | i do like the ports simply for finding software though. |
| 16:14.07 | *** join/#asterisk AndyGraybeal (n=andy@node78.38.251.72.1dial.com) |
| 16:14.16 | mintee | <3 make search whatever |
| 16:14.22 | *** join/#asterisk drmessano-LT (n=nonya@207.230.140.240) |
| 16:14.22 | b11d | make search key="string" rocks |
| 16:18.35 | mintee | ah yeah, that's it... my qmail server is fbsd.... haven't really had to touch it recently |
| 16:19.05 | b11d | qmail is cool.. I wish there'd be some development with it though... seems kinda dead lately doesnt it |
| 16:19.08 | plik | b11d: so did Asterisk et al compile nicely from source without using ports at all? |
| 16:19.12 | mintee | ok, well it seems if i unload wanpipe modules before i unload the zaptel modules, then I get a clean reboot... |
| 16:19.14 | b11d | i like the djbdns package he does |
| 16:19.24 | plik | on FreeBSD of course |
| 16:19.24 | mintee | qmail hasn't been touched in about 6 years iirc |
| 16:19.27 | b11d | plik.. asterisk, libpri, zaptel-bsd, etc.. all works tits. |
| 16:19.32 | *** join/#asterisk worgil (n=worgil@88.230.178.73) |
| 16:19.33 | russellb | we have asterisk on our automated build cluster building on freebsd every hour |
| 16:19.45 | russellb | with no modifications |
| 16:20.08 | plik | cool... I built from ports at home but would like to try from source, so I'll give it a go sometime |
| 16:20.14 | b11d | its super easy.. |
| 16:20.20 | b11d | i recommend it |
| 16:20.28 | mintee | russellb? what? why are you building asterisk every hour? |
| 16:20.28 | b11d | just be sure to install 'gmake' first |
| 16:20.44 | *** join/#asterisk rlx (n=edward@c-24-22-183-194.hsd1.mn.comcast.net) |
| 16:20.47 | russellb | mintee: so it yells at us when we break it for some platform |
| 16:20.59 | plik | just now I'm making amends to an office system that started as AsteriskNOW - so does anyone have any pointers for Docs on how to migrate away from useres.conf please? |
| 16:20.59 | russellb | us (digium developers) |
| 16:21.35 | mintee | russellb, ah, ok that makes sense then.... |
| 16:22.21 | jameswf | Where can i download zaptel.exe for windows |
| 16:22.50 | Nugget | there's a zaptel.exe for windows? wow. |
| 16:22.54 | russellb | jameswf: just run windows update |
| 16:23.19 | jameswf | i cant im on a pirated versio of vista sp6 |
| 16:23.29 | drmessano-LT | :( |
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| 16:23.54 | jameswf | I wonder is i can make winepass WGA |
| 16:23.56 | jameswf | :) |
| 16:24.21 | Nugget | heh |
| 16:24.29 | jameswf | s/is/if/ s/winepass/wine pass/ |
| 16:24.31 | jameswf | dang |
| 16:25.10 | jameswf | congrats your copy of wine is genuine |
| 16:28.06 | drmessano-LT | HAHAH |
| 16:28.17 | drmessano-LT | Now THAT is BASHworthy |
| 16:28.49 | *** join/#asterisk fnordus (n=dnall@24.84.160.227) |
| 16:28.55 | drmessano-LT | Even if Wine does contain a bunch of Windows 95 source code :/ |
| 16:30.14 | clyrrad | Has anyone setup FollowMe in 1.4 using Realtime Pgsql? |
| 16:30.55 | *** join/#asterisk SteveTotaro (n=root@209.213.170.178) |
| 16:31.36 | clyrrad | looking for a way to set the follow me numbers in a Pgsql database instead of the ASTDB |
| 16:31.48 | *** join/#asterisk esaym (n=user@72.183.198.134) |
| 16:31.52 | drmessano-LT | I wonder how getting Comcast VoIP will affect my other VoIP use at home |
| 16:32.45 | drmessano-LT | I wonder if they'll apply some QoS that makes all my VoIP better |
| 16:32.47 | drmessano-LT | Hmm |
| 16:33.27 | drmessano-LT | I would imagine they do it at the node and per node port, so anything on my connection would be better |
| 16:33.37 | *** join/#asterisk CunningPike (n=arodgers@204.239.12.183) |
| 16:33.48 | drmessano-LT | If they do it at all |
| 16:34.56 | kyron | Anyone has favorable/unfavorable comments about the `D-LINK DVG-1120`...I need an FXO for home use (so Digium cards, although probably really good, are out of the question) |
| 16:36.00 | jameswf | i told wga to run on windows 3.1 it didnt know what to do |
| 16:37.51 | tzafrir | kyron, x100p? |
| 16:39.00 | *** part/#asterisk FL1SK (n=FL1SK@72.24.30.153) |
| 16:39.33 | kyron | tzafrir, x100P seem like more of a headache than I feel like dealing with given both seem to be about the same price |
| 16:40.24 | kyron | tzafrir, I can get a DVG-1120 for about 34$ before shipping...and an x100p is about 30$ before shipping.. |
| 16:41.26 | kyron | with the DVG, I know it "works" and I even get 2 FXS ports with it (not that I really need em given I have a Mediatrix 1104...) |
| 16:41.56 | kyron | tzafrir, but this becomes completely mute if DVG-1120 are known to be a pain in the @@ with *... |
| 16:42.39 | drmessano-LT | Get an SPA-3102 |
| 16:43.29 | kyron | that's close/over 100$ :/ |
| 16:44.07 | drmessano-LT | WHAT? |
| 16:44.10 | *** join/#asterisk |omni| (n=rob@70.89.211.34) |
| 16:44.16 | drmessano-LT | Where is it $100? |
| 16:45.24 | kyron | well..off e-bay.. and $67.99 off google shopping.. |
| 16:45.31 | kyron | before shipping... |
| 16:45.32 | drmessano-LT | ok |
| 16:45.34 | drmessano-LT | $70 |
| 16:45.42 | kyron | so unless you have one for sale.. |
| 16:45.58 | drmessano-LT | For a Unix admin, you're a cheap bastard, kyron :) |
| 16:46.07 | kyron | yeah well, the lower priced items tend to have outrageous shipping and "handling" costs.. |
| 16:46.36 | kyron | drmessano-LT, for a student with a newborn and the mother also being a student...maybe not so much ;) |
| 16:46.36 | drmessano-LT | You can get a grandstream HT488 |
| 16:46.50 | kyron | drmessano-LT, now that's just plain mean |
| 16:47.29 | drmessano-LT | Maybe FungXu has some nice FXO devices |
| 16:47.35 | kyron | makes me wonder what UnixDog got out of the meeting with them ;) |
| 16:47.43 | drmessano-LT | Check eBay for "LQQK FXO LOW PRICE" |
| 16:47.52 | kyron | FungXu? |
| 16:48.03 | kyron | lqqk O_o |
| 16:48.17 | drmessano-LT | Yeah |
| 16:49.06 | kyron | drmessano-LT, your sarcasm is hard to gage through IRC ;) |
| 16:49.21 | drmessano-LT | I wasn't joking about the LQQK |
| 16:49.25 | drmessano-LT | :) |
| 16:50.09 | drmessano-LT | That's eBay speak for "ZOMG THIS ISNT FAKE AT ALL LINKSIS PAP2 VERY YES |
| 16:50.35 | kyron | LOL "Very YES!" (origin: Hong Kong) |
| 16:50.36 | *** join/#asterisk docelmo (n=vircuser@h74.196.180.12.cable.rstb.cablerocket.net) |
| 16:50.43 | drmessano-LT | I found some FXS, FXO, Router box from China on Bay for $19 |
| 16:50.49 | drmessano-LT | Brand new |
| 16:50.51 | kyron | so I'd get my FXO in 3months...heheh |
| 16:51.00 | drmessano-LT | Actually that shit comes fast |
| 16:51.02 | drmessano-LT | 10 days maybe |
| 16:51.36 | drmessano-LT | I bought a few chinese knock off things on eBay and never had them take more than a week |
| 16:51.41 | kyron | drmessano-LT, Might as well get my FXO through: www.nxtvox.com <--still, would be 100$ |
| 16:52.21 | *** join/#asterisk outtolunc (n=me@adsl-66-218-53-172.dslextreme.com) |
| 16:52.23 | drmessano-LT | Im tempted to get a LQQKCom LINKSIS WTF31P4 |
| 16:52.40 | drmessano-LT | For $20.. if it sucks, I can let my cat play with it |
| 16:53.32 | drmessano-LT | Im waiting for a chinese knock off trimline VoIP phone |
| 16:53.41 | drmessano-LT | I know Linksys makes a trimline |
| 16:53.55 | drmessano-LT | Id like to see a $15 trimline VoIP phone from LQQKCom |
| 16:54.17 | kyron | drmessano-LT, hmm...didn't get any hit on ebay for lqqk |
| 16:54.58 | drmessano-LT | I got 5626... but none for VoIP boxes :( |
| 16:55.49 | drmessano-LT | Im trying to remember how I found that $20 box |
| 16:58.17 | drmessano-LT | It was a silver case.. looked VERY cheap |
| 16:59.05 | b1ch0 | hi messano, i connected a panasonic fax machine to my fxs port, i can send faxes but when i try to receive one, i can hear bip from fax .... putting it in phone mode i can make and receive calls |
| 16:59.26 | b1ch0 | ... and i know that it is not the right channel |
| 16:59.28 | *** join/#asterisk davevg-btwtech (n=davevg-b@nj-67-76-177-147.sta.embarqhsd.net) |
| 17:00.36 | drmessano-LT | Bah |
| 17:00.54 | drmessano-LT | FAX over ATA = Suxors |
| 17:01.03 | drmessano-LT | Be ashamed you even asked |
| 17:01.09 | drmessano-LT | :) |
| 17:01.53 | *** join/#asterisk CunningPike (n=arodgers@204.239.12.183) |
| 17:02.01 | *** join/#asterisk esaym (n=user@72.183.198.134) |
| 17:02.33 | hmodes | try setting it to 9600bps, but yeah, fax over voip is crap |
| 17:03.03 | b1ch0 | it is not an external ata, i am using fxs port of internal tdm card .... Hylafax works great, but customer still want receive faxes at theyir old machine |
| 17:03.43 | coppice | its a myh that 9600bps helps. |
| 17:03.59 | b1ch0 | fax machine doesnt send "bip" to start |
| 17:03.59 | coppice | b1ch0: TDM400P card? |
| 17:04.16 | b1ch0 | no, zapmicro, but with FXS digium module |
| 17:04.40 | drmessano-LT | oh |
| 17:04.50 | drmessano-LT | You weren't using a PAP2? Sorry |
| 17:05.15 | coppice | b1ch0: same thing, really. they give endless trouble with FAX. |
| 17:05.36 | *** join/#asterisk CoffeeIV_ (n=CoffeeIV@adsl-99-162-117-1.dsl.austtx.sbcglobal.net) |
| 17:07.46 | *** join/#asterisk kristijan (n=kristija@dslb-084-059-129-096.pools.arcor-ip.net) |
| 17:08.41 | CoffeeIV_ | I am trying to do fax detection on a zap channel coming in on a T1/PRI with a digium card. I have "faxdetect=incoming" in the zapata.conf, but it is not working. Is this the standard way to do fax detect, or should I set up that NVFaxDetect() stuff ? If I do NVFaxDetect() will it also work on my IAX2 calls ? |
| 17:08.42 | *** join/#asterisk beek (n=klinebl@65.211.106.243) |
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| 17:12.25 | tzafrir | CoffeeIV_, it is |
| 17:12.34 | tzafrir | How do you know it doesn't work? |
| 17:13.14 | tzafrir | what happens when a fax comes in? a trace, please |
| 17:13.30 | *** join/#asterisk ctp (n=ctp@brsg-4d07bc6a.pool.mediaWays.net) |
| 17:15.49 | ctp | hi folks. i have a strange trouble with my * box. i've defined 3 phones in my sip.conf. calling hardphone from softphone works fine, the opposite way i get a "Call from '10' to extension '12' rejected because extension not found.". strange. this is my sip.conf snippet for calling within lan: |
| 17:15.54 | ctp | exten => _1X,1,NoCDR() |
| 17:15.54 | ctp | exten => _1X,n,Dial,SIP/${EXTEN}|55|Ttr |
| 17:16.09 | ctp | anyone here has a hint whats going wrong? |
| 17:16.11 | *** part/#asterisk kristijan (n=kristija@dslb-084-059-129-096.pools.arcor-ip.net) |
| 17:16.26 | *** join/#asterisk |omni| (n=rob@70.89.211.34) |
| 17:17.10 | *** join/#asterisk apocn (n=apo@unaffiliated/apocn) |
| 17:17.23 | apocn | Is anyone here experienced with QueueMetrics? |
| 17:19.23 | apocn | anyway Im using MixMonitor to record the conversations of the agents logged on the Queue. |
| 17:19.38 | apocn | now I want to listen to the agents conversation in real time, should I use ChanSpy? |
| 17:20.00 | CoffeeIV_ | tzafrir: I know the fax doesn't work because when I send one it, it never goes to the fax extension. the *CLI> and full log show the same thing as a normal call. These .conf's and dialplan and this card did fax detection at some point in the past, but since then asterisk has been upgraded, and the server was moved to a different T1 line, so it makes sense that something changed and broke it |
| 17:20.08 | apocn | or ExtenSpy? |
| 17:20.58 | *** join/#asterisk vale-ICS (n=vale@icsnet.demon.co.uk) |
| 17:21.47 | *** part/#asterisk rlx (n=edward@c-24-22-183-194.hsd1.mn.comcast.net) |
| 17:22.06 | tzafrir | CoffeeIV_, in the output of: zap show channel NNN, do you see 'Fax Handled: incoming'? |
| 17:23.49 | CoffeeIV_ | tzafrir: I see "Fax Handled: no" which is probably my problem |
| 17:25.04 | [TK]D-Fender | ctp: that is not stuff for sip.conf, taht is EXTENSIONS.CONF |
| 17:25.41 | [TK]D-Fender | ctp: and we'd have to see your full CLI output in a PASTEBIN along with your complete dialplan context and peer config |
| 17:27.41 | ctp | [TK]D-Fender: was a typo. the two lines are part of my extensions.conf |
| 17:27.58 | [TK]D-Fender | ctp: Please provide everything I have just requested. |
| 17:28.47 | CoffeeIV_ | tzafrir: compairing to another asterisk I administer, I see that faxdetect=incoming is above the group= line in the zapata.conf. I left myself a comment saying that was necessary there but I have no memory of it . . . should faxdetect= line be above the group= line ? |
| 17:29.38 | tzafrir | CoffeeIV_, doesn't matter |
| 17:29.53 | tzafrir | As I mentioned, you can see its actual value |
| 17:30.06 | tzafrir | At runtime |
| 17:30.14 | CoffeeIV_ | yes |
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| 17:37.09 | dijungal | app_queue sets MEMBERINTERFACEjust after it would be useful to use in MixMonitor file name variable substitution <- this was patched by qwell.... but i'm using Asterisk 1.4.15 and it has that issue still |
| 17:37.17 | dijungal | which version of asterisk was this fixed on? |
| 17:38.14 | *** join/#asterisk lunaphyte_ (n=lunaphyt@70.90.148.3) |
| 17:38.15 | BCS-Satori | Our system has been running for 77days without any issues on 1.4.11 and in the past few days we have been experiencing phones continuing ringing even when the phone call is answered by another phone. We have rebooted the system, switches, & routers and it typically goes away for several hours and then returns. Any ideas why they continue to ring even when the call is answered? each user has to walk around to all phones and hang them up |
| 17:38.21 | CoffeeIV_ | tzafrir: I moved the faxdetect= line higher than the group= and channel= lines in zapata.conf, and now a fax call produces "chan_zap.c: Fax detected, but no fax extension" in the log. So I think that order did make a difference. Of course now I have to figure out where my fax extension went. |
| 17:38.25 | ctp | [TK]D-Fender: ok, here is my config: http://rafb.net/p/Jwmtcy82.html |
| 17:39.10 | tzafrir | CoffeeIV_, it *must* be above the 'channel' line. No relation to the group line |
| 17:39.44 | tzafrir | The context to look at: the one set by context= |
| 17:41.45 | CoffeeIV_ | tzafrir: thanks, that makes sense. I appreciate your help a lot |
| 17:42.14 | ctp | [TK]D-Fender: argh, i see one mistake now. context=default in [12] instead of context=sipphones. changed that i get get new error now: |
| 17:42.17 | ctp | [Feb 13 18:40:23] WARNING[2118] chan_sip.c: Remote host can't match request NOTIFY to call '0b8f005058c53cb8227b3bca25cf7d5b@192.168.5.2'. Giving up. |
| 17:42.17 | ctp | [Feb 13 18:40:26] WARNING[2153] app_dial.c: Unable to create channel of type 'SIP' (cause 3 - No route to destination) |
| 17:42.39 | *** join/#asterisk emist_ (n=emist@unaffiliated/emist) |
| 17:44.55 | *** join/#asterisk patrick-- (n=patrick@devnull.biz) |
| 17:48.50 | *** join/#asterisk admin0 (n=admin@116.90.228.34) |
| 17:50.18 | x86 | would impedance cause zaptel not to hear digits / not break dialtone when a channel starts dialing? |
| 17:54.19 | jm|laptop | hello :) |
| 17:54.37 | jm|laptop | so today I got my bluetooth headset working with BlueZ |
| 17:55.06 | jm|laptop | does anyone know a soft sip phone that supports bluetooth headsets :/ |
| 17:55.06 | patrick-- | Hey, i have 2 ISDN Phones on a BN8S0. When i pickup one phone, i get a dialtone.. when i puickup the other i dont. what could cause this? |
| 17:55.18 | jm|laptop | I tried linphone and ekiga and they just can see the default ALSA device |
| 17:55.32 | [TK]D-Fender | ctp: please note the rather blatant error. In [12] you have "context = sipphones". This is NOT the context you have in mind. Pay attention to where you point things to. |
| 17:55.36 | *** join/#asterisk [hC] (n=hardcore@S01060016b6b53c0c.vc.shawcable.net) |
| 17:55.57 | patrick-- | btw [TK]D-Fender thanks for your support :) i got a lot further with just figuring things out myself :) |
| 17:55.59 | x86 | i'm in the US, and the impedance on this Adit is set to 900 ohms |
| 17:56.26 | [TK]D-Fender | x86: Gain & line quality can disrupt DTMF. What ver of * are you on? |
| 17:56.44 | [TK]D-Fender | x86: isn't 600 the standard for North America? |
| 17:57.57 | x86 | that's why I'm asking -- someone told me to use 600 ohms |
| 17:58.31 | [TK]D-Fender | x86: taht could be it... |
| 17:58.32 | x86 | but the Adit CLI says 900 ohms + 2.16uF is US standard |
| 17:58.56 | [TK]D-Fender | patrick--: You had mISDN issue before, right? |
| 17:59.29 | x86 | [TK]D-Fender: do i want 600 ohms straight up or 600 ohms + 2.16uF? |
| 18:00.37 | [TK]D-Fender | x86: Well I'm speaking from general hearsay no, excessive personal technical teste knowledge. Its the number I remember seeing in some ATA's, and thrown around. I will reserve any judgement based on that. Just contributing my input as a question, not an answer |
| 18:01.18 | *** part/#asterisk sabbir (n=SABBIR@210.4.73.156) |
| 18:01.27 | patrick-- | [TK]D-Fender: thats right. |
| 18:02.13 | x86 | hmm |
| 18:02.28 | x86 | anyone ever changed impedance on an Adit 600 channel bank? |
| 18:02.40 | tzanger | x86: I believe i Have |
| 18:02.48 | *** join/#asterisk lunaphyte_ (n=lunaphyt@70.90.148.3) |
| 18:03.15 | x86 | tzanger: you know how to do it? |
| 18:03.43 | tzanger | it was just a command I believe |
| 18:03.45 | tzanger | I might be wrong though |
| 18:03.59 | x86 | right, do you remember the command? :) |
| 18:04.36 | [TK]D-Fender | patrick--: Yeah, sorry, wish I could have offered some more advice, but I have never worked with it before. What issues do you have left? |
| 18:05.06 | patrick-- | [TK]D-Fender: im a bit confused with my HFC card's ports. there are 2 S0 ports on each RJ45 connector. i connected 2 isdn phones, one gets a dialtone straight on pickp, the other doesnt... |
| 18:05.38 | tzanger | x86: all the commands are there with help I think |
| 18:06.09 | x86 | tzanger: thats.... not helpful ;P |
| 18:06.20 | drmessano-LT | 900 ohms + 2.16uF is US Standard? |
| 18:06.28 | drmessano-LT | ..since when? |
| 18:06.57 | jameswf | Im not that smart but i will be staying at a holliday in |
| 18:07.43 | dijungal | hi guys.... i wanna install 1.4.18 but i have 1.4.11 now installed.... how do i go proceed? |
| 18:07.58 | dijungal | do i just go ahead and install over the current version? |
| 18:08.54 | x86 | drmessano-LT: since CAC said so? :P |
| 18:09.04 | patrick-- | [TK]D-Fender: any ideas? :D |
| 18:09.07 | x86 | drmessano-LT: do you know how I would go about changing impedance on an Adit 600? |
| 18:09.23 | drmessano-LT | US Standard phone impedence is 600 ohms, purely resistive |
| 18:09.40 | x86 | the only "pure 600 ohms" option the Adit 600 has is A-law |
| 18:10.00 | x86 | the only thing that comes close is 600 ohms + 2.16uF which can be done u-Law |
| 18:10.02 | Wayhigh | do people here primarily config * using only the config files or do more people use something like freepbx? and why? (trying to decide which method to use) |
| 18:10.03 | *** join/#asterisk ManxPower (n=manxpowe@15.sub-75-202-227.myvzw.com) |
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| 18:11.21 | drmessano-LT | Well, 2.16uF shouldnt make too much of a difference.. its not likely the line is exactly 600 anyway.. A little reactance is ok |
| 18:11.50 | x86 | ok, so how do i change it? :P |
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| 18:12.07 | drmessano-LT | You're asking the wrong person.. I dont even know what the fuck that thing is |
| 18:12.24 | drmessano-LT | But you asked about standards, and I am Mr KnowItAll, so there you go |
| 18:12.38 | ManxPower | any change to the impeedence is dont by the device with the analog ports on it. |
| 18:12.43 | drmessano-LT | Anything else will run you $122.50 rounded up to the first our |
| 18:12.45 | ManxPower | dont == done |
| 18:12.51 | drmessano-LT | hour |
| 18:13.32 | drmessano-LT | ManxPower: Thats Obvious here |
| 18:13.46 | drmessano-LT | x86 is trying to figure out how to do it on a specific device |
| 18:15.35 | drmessano-LT | I couldn't find it with a quick google search, x86 |
| 18:16.43 | *** join/#asterisk vrwttnmtu (n=eryktykt@82-69-161-137.dsl.in-addr.zen.co.uk) |
| 18:16.45 | _ShrikE | For FXS card in slot 1 it would be... |
| 18:16.59 | _ShrikE | set 1:1-8 impedance {the number you want} |
| 18:17.14 | _ShrikE | the number found by doing show impedance |
| 18:17.20 | tzafrir | Wayhigh, I think more people *here* edit config files |
| 18:18.14 | *** part/#asterisk simbol76 (n=simbol@ip-212-18.sn1.eutelia.it) |
| 18:18.29 | sponger | Is there anyone in here that could give me a pointer on SLA and polycoms. I have my SLA all setup and it rings on incoming calls and it can be answered. I am certain that the SLA is all properly configured. I am having issues with the configuration files of the polycoms and having them subscribe to the hints on the server. sip show subscriptions is blank |
| 18:18.55 | vrwttnmtu | I've just found something very strange. 2 Asterisk servers, domain1, and domain2, all working and configured fine. user@domain1 can SIP call user2@domain2, but can't call a user with the same username without getting a 407 Proxy Auth error |
| 18:19.13 | *** join/#asterisk really_phukt (n=chatzill@static-72-77-217-74.tampfl.dsl-w.verizon.net) |
| 18:19.47 | mvanbaak | hhmm |
| 18:20.04 | mvanbaak | I see the default config files changed from Zap/g1 to Zap/G2 |
| 18:20.07 | mvanbaak | why is that ? |
| 18:21.11 | *** join/#asterisk ph0ne (n=ph0ne@dsl-207-112-19-129.tor.primus.ca) |
| 18:22.38 | Wayhigh | tzafrir: if I was asking freepbx config questions I'd certainly ask there.. |
| 18:23.26 | Wayhigh | I'm just wondering if there's really a good reason to not use something like freepbx.. |
| 18:23.45 | tzafrir | patrick--, what card is it? |
| 18:24.11 | ManxPower | ~freepbx |
| 18:24.12 | jbot | [~freepbx] FreePBX is unable to be supported here. It is made up of complex dialplans and scripts which can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking there |
| 18:24.18 | patrick-- | tzafrir: BN8S0 |
| 18:24.20 | *** join/#asterisk ArM-eye (i=elf@12-205-155-236.client.mchsi.com) |
| 18:24.23 | ManxPower | there is your reason to not use FreePBX. |
| 18:25.27 | patrick-- | tzafrir: when i put another phone onto the same port it wont even power up |
| 18:25.33 | really_phukt | Wayhigh, I like to have control over my dialplan, so I had scratch the idea of freepbx when I saw how many config files the damn thing makes... |
| 18:25.44 | AndyGraybeal | Wayhigh... from #santacruz on efnet? |
| 18:26.11 | patrick-- | tzafrir: it seems as if the card doesnt actually "recognize" the phone, cause the phone that works with sip and such has a different writing on the LCD |
| 18:27.58 | tzafrir | patrick--, so it seems that the phone is not powered? |
| 18:27.59 | x86 | any ideas as to why i'm not able to break dialtone on a zap channel unless I dial a 6 or above first? |
| 18:28.09 | *** join/#asterisk GBR_ (n=gbr@200.103.96.98) |
| 18:28.22 | ManxPower | x86: the dialplan, of course. |
| 18:28.57 | x86 | dialplan has something to do with that? |
| 18:29.33 | x86 | for example, I have one channel in it's own context, and the only thing in the context is exten => _X.,NoOp(${EXTEN}) |
| 18:30.00 | x86 | the only time i can get asterisk to even stop giving me dial tone is if i dial 6, 7, 8, 9, * or # |
| 18:30.10 | x86 | 1, 2, 3, 4, 5 do nothing |
| 18:30.54 | ManxPower | x86: Well, without a priority it won't do anything |
| 18:31.12 | really_phukt | ignorepat? |
| 18:31.16 | x86 | ManxPower: bah, it had a priority ;) |
| 18:31.21 | x86 | ManxPower: it's not the dialplan |
| 18:31.30 | x86 | ManxPower: it's never actually hitting asterisk |
| 18:31.54 | x86 | ManxPower: if i debug DTMF in logger.conf, i never see any digits passed in unless i start dialing with a 6, 7, 8, 9, *, or # |
| 18:32.00 | dijungal | ! |
| 18:32.27 | [TK]D-Fender | x86: sounds like something is distorting across the top & bottoms DTMF rows. thats a duo-tone mismatch. |
| 18:32.44 | flush | ahoy |
| 18:32.56 | *** join/#asterisk tobias (n=tobias@cpe-066-026-085-055.nc.res.rr.com) |
| 18:32.56 | flush | i just ordered a TDM400P with 3x FXS and one FXO module.. |
| 18:33.08 | [TK]D-Fender | x86: Again, what * are you on? |
| 18:33.08 | flush | is it a good move to set up my first asterix/pbx box |
| 18:33.25 | [TK]D-Fender | flush: It is if thats what you want. |
| 18:33.38 | [TK]D-Fender | flush: And the FXS is a waste |
| 18:33.47 | flush | ? |
| 18:33.56 | [TK]D-Fender | flush: I would have advised a different hardware scenario |
| 18:34.01 | dijungal | what s this error abou, hot do i fix it? Internal RTCP NTP clock skew detected |
| 18:34.04 | dijungal | "Internal RTCP NTP clock skew detected" |
| 18:34.08 | flush | [TK]D-Fender please tell me |
| 18:34.08 | *** join/#asterisk angom (n=angom@200.56.104.87.dsl.dyn.telnor.net) |
| 18:34.12 | flush | i can still change the setup of the card |
| 18:34.18 | dijungal | "Internal RTCP NTP clock skew detected: lsr=3059514815, now=3059607361, dlsr=131000 (1:998ms), diff=38454" |
| 18:34.21 | flush | just ordered on ebay like 2 hours ago |
| 18:34.25 | flush | not shipped yet for sure |
| 18:34.25 | *** join/#asterisk putnopvut (n=putnopvu@216.207.245.1) |
| 18:34.30 | [TK]D-Fender | flush: Zaptel FXS is only a source of unnecessary hardware, configuration and economic issues |
| 18:34.47 | flush | i thought FXS module was to plug a normal phone in it |
| 18:34.56 | flush | and that FXO was used to connect to the phone line in the wall |
| 18:35.00 | drmessano-LT | Its a waste using up a space on a card for FXS |
| 18:35.01 | [TK]D-Fender | flush: What are you buying the card for use in? home or business? What kind of expansion is planned? |
| 18:35.05 | flush | home busines |
| 18:35.10 | flush | brb somoene knocking |
| 18:35.17 | [TK]D-Fender | flush: Planning on more than 1 line? |
| 18:35.32 | x86 | [TK]D-Fender: 1.4.12 |
| 18:35.51 | x86 | [TK]D-Fender: so what do i do about the duo-tone mismatch? |
| 18:35.54 | dijungal | how do I fix this: Internal RTCP NTP clock skew detected: lsr=3059514815, now=3059607361, dlsr=131000 (1:998ms), diff=38454 |
| 18:36.08 | *** join/#asterisk guillote_GNU (n=guillote@host157.201-253-55.telecom.net.ar) |
| 18:36.34 | flush | [TK]D-Fender no only one line |
| 18:36.43 | flush | but maybe plug 2 or 3 phones to the box.. |
| 18:36.47 | [TK]D-Fender | flush: What kind of call volume? |
| 18:36.49 | flush | will it work ? |
| 18:36.54 | flush | not much at all |
| 18:37.02 | flush | im doing it just for fun to be true.. |
| 18:37.12 | x86 | [TK]D-Fender: i mean it's not a huge deal, i just tell everyone to start the dial with a 7 "to get an outside line" haha |
| 18:37.20 | [TK]D-Fender | flush: What I'd advise would probably be from Linksys : 1 x SPA-3102, and 1 x SPA-2102 |
| 18:37.30 | x86 | [TK]D-Fender: which works most of the time, but sometimes it misses digits in the middle of the number |
| 18:37.47 | [TK]D-Fender | x86: Except that its not just the first digit, its the inbetweens tuff that'll kill you. |
| 18:38.10 | x86 | right |
| 18:38.12 | jameswf | T1 troubleshooting Step 1 is it plugged in |
| 18:38.14 | [TK]D-Fender | x86: Is this across multiple phones as well? (make, model, and port) |
| 18:38.22 | x86 | [TK]D-Fender: yes |
| 18:38.33 | flush | [TK]D-Fender okay.. but for what im planning to do, have 3 phones plugged in the box, with one line, is it okay to have 3x FXS and one FXO ? |
| 18:38.38 | [TK]D-Fender | x86: Does swapping out the CB solve it? |
| 18:38.51 | x86 | i've tried playing with zaptel's rxgain, as well as rxgain on the channel bank (although not both at the same time) |
| 18:38.54 | x86 | [TK]D-Fender: nope |
| 18:39.12 | flush | damnit, brb food is calling |
| 18:39.15 | x86 | [TK]D-Fender: happens with both a Rhino as well as an Adit |
| 18:39.55 | ManxPower | x86: put your zaptel.conf and zapata.conf on pastebin.ca |
| 18:41.49 | ManxPower | x86: So you have Analog phone -> channel bank -> Asterisk right? |
| 18:42.18 | x86 | http://pastebin.ca/902717 |
| 18:42.25 | x86 | ManxPower: yessir |
| 18:42.46 | x86 | ManxPower: i've got two channel banks hooked up to this, one is a rhino and one is an adit |
| 18:42.54 | x86 | ManxPower: problem persists across both |
| 18:43.07 | ManxPower | what spans are they on? |
| 18:43.49 | x86 | rhino = span2 |
| 18:43.55 | x86 | adit = span3 |
| 18:43.58 | x86 | telco = span1 |
| 18:44.20 | x86 | they are all labeled ;) |
| 18:44.33 | *** join/#asterisk keulin (n=cray@ifth-pdcr2.infotheme.fr) |
| 18:45.04 | *** join/#asterisk XnOSX (i=d491ac7f@gateway/web/ajax/mibbit.com/x-b81f1fff52585f78) |
| 18:45.31 | ManxPower | you need a 1 in the 2nd field of span 1 |
| 18:45.38 | x86 | no, i dont |
| 18:45.41 | x86 | not according to sangoma |
| 18:45.56 | x86 | they told me that's handled in the wanpipe1 configuration |
| 18:46.05 | x86 | which has an option for clock |
| 18:46.26 | x86 | i'll try it, but i'm not having any problems at all with span 1 |
| 18:46.31 | ManxPower | then it can't hurt, can it. |
| 18:46.40 | x86 | sure, but it can't help either ;) |
| 18:46.47 | ManxPower | I see nothing wrong with your setup. Exactly how are you testing this? |
| 18:47.20 | mvanbaak | I never see anything wrong in my setup. but still it almost breaks at first try |
| 18:47.23 | mvanbaak | ;) |
| 18:47.41 | x86 | ok, i've got a patch panel with (24) RJ11 ports on the front, and (1) AMP25 on the back |
| 18:47.46 | patrick-- | tei_mux: wrong mt 2 <-- what does this mean? |
| 18:47.55 | x86 | the AMP25 connects the patch panel to the channel bank |
| 18:48.12 | cmantito | in ast 1.4.5, what's the correct way to set the accounting code for CDRs? is it Set(CDR(accountcode)) or SetAccount()? |
| 18:48.18 | x86 | so i plug a standard analog phone (doesn't matter the make or model) into any of the ports, and try dialing |
| 18:48.19 | ManxPower | That was not my question |
| 18:48.26 | ManxPower | ah, that does. |
| 18:48.30 | x86 | ManxPower: i wasn't done ;) |
| 18:48.37 | mvanbaak | cmantito: CDR(accountcode) |
| 18:48.50 | ManxPower | So you pick up the phone and dial what? |
| 18:49.15 | cmantito | and what module would that belong to? |
| 18:49.17 | Nivex | No you dial who!</costello> |
| 18:49.23 | x86 | i try dialing anything... first i try a single "1", still have dial tone, then just "2", still dial tone... all the way until I get to "7", which breaks the dial tone |
| 18:49.27 | *** join/#asterisk Docfxit (n=none@ip-64-32-143-214.lax.megapath.net) |
| 18:49.43 | ManxPower | x86: then I guess we also need your extensions.conf |
| 18:49.54 | mvanbaak | cmantito: it's a standard dialplan function |
| 18:49.57 | x86 | ManxPower: how is dialtone and extensions.conf related? |
| 18:50.13 | ManxPower | x86: they are completly related on FXS ports. |
| 18:50.25 | x86 | hmm |
| 18:50.27 | ManxPower | in fact it deternins all dialing strings, breaking dialtone, etc. |
| 18:50.31 | *** join/#asterisk mmmToop (n=michaelt@dsl-243-248-143.telkomadsl.co.za) |
| 18:50.31 | cmantito | mvanbaak: thanks |
| 18:50.50 | patrick-- | does anyone know what this means? tei_mux: wrong mt |
| 18:50.59 | ManxPower | this is your first experience with analog and fxs isn't it? |
| 18:51.06 | x86 | ManxPower: well the same thing happens when (like i was saying before), i single out a random channel and put it in it's own context, then do exten => _X.,1,NoOP(${EXTEN}) |
| 18:51.09 | ManxPower | patrick--: it's BRI error, not many people run BRI here. |
| 18:51.14 | Wayhigh | thank you |
| 18:51.16 | x86 | ManxPower: nope |
| 18:51.27 | patrick-- | ManxPower: so what does it mean? |
| 18:51.28 | ManxPower | x86: I cannot help you if you don't show us extensions.conf |
| 18:51.32 | x86 | ManxPower: first time I've had this issue though |
| 18:51.36 | x86 | ok, hold on |
| 18:52.02 | really_phukt | ignorepat? |
| 18:52.04 | Wayhigh | btw.. anyone else found vonage selling their email addresses? I'm getting spam to a one-time email used w/ vonage.. bastads |
| 18:52.30 | ManxPower | really_phukt: We will find out when we see his zapata.conf |
| 18:52.37 | patrick-- | really_phukt: pardon me? |
| 18:52.43 | x86 | ManxPower: http://pastebin.ca/902733 |
| 18:52.58 | x86 | ManxPower: I put channel 61 in the test context in zapata.conf |
| 18:53.15 | x86 | ManxPower: then i used just this extensions.conf with no other |
| 18:53.29 | x86 | ManxPower: only a 7, 8, 9, *, or # will break the dial tone |
| 18:53.56 | ManxPower | x86: you're an asshole. There is no way the extensions.conf you gave me will work with the zapata.conf and zaptel.conf you gave me |
| 18:53.58 | *** join/#asterisk oej (n=olle@soll4-125.cust.blixtvik.net) |
| 18:53.59 | x86 | ManxPower: clearly, i'm not trying to exclude or include any one particular digit(s) |
| 18:54.16 | x86 | ManxPower: i just told you that i put channel 61 in that context! |
| 18:54.17 | ManxPower | in fact, I doubt the extensions.conf you gave us will work with ANYTHNIG. |
| 18:54.27 | x86 | sure it will |
| 18:54.29 | x86 | and it does |
| 18:54.31 | really_phukt | patrick--, in extensions.conf "ignorepat" can be used to keep the dial tone going. this is in regard to x86's problem |
| 18:54.45 | ManxPower | nope. There is no [general] there is no [global] section either |
| 18:55.04 | ManxPower | Now either give us the ACTUAL files or don't expect help. |
| 18:55.08 | x86 | ManxPower: those are both optional |
| 18:55.15 | ManxPower | x86: not in my experience. |
| 18:55.35 | x86 | really_phukt: no ignorepat in any of my confs |
| 18:55.36 | ManxPower | x86: Your problem is sometging subtle. If we can't find where that subtle problem is we can'thelp. |
| 18:55.44 | patrick-- | really_phukt: but why is only one of my phones working? |
| 18:55.46 | ManxPower | and we can't do that unless we see current config files. |
| 18:55.51 | x86 | ManxPower: i gave you my conf... not sure what else you want |
| 18:56.00 | ManxPower | not some config files, then you make a change and not show us anything more. |
| 18:56.11 | x86 | ok, hold on |
| 18:56.15 | ManxPower | x86: well giving us updated zapata.conf and zaptel.conf would be astart |
| 18:56.18 | really_phukt | patrick--, because the other one is phukt... ;) :D |
| 18:57.31 | b1ch0 | hi everybody, here again with my incoming fax problem over fxs module (internal tdm card) |
| 18:58.16 | b1ch0 | FXS port, answer, but cant hear fax tone from Fax Machine |
| 18:58.25 | b1ch0 | any idea ? |
| 18:58.32 | ManxPower | b1ch0: "can't hear"????? |
| 18:59.01 | b1ch0 | if i replace fax machine with a normal phone i can make and receive calls |
| 18:59.34 | b1ch0 | yes, exactly |
| 18:59.36 | ManxPower | b11d: and you have faxdetect turned off, or not enabled in the zap config, right? |
| 18:59.59 | b1ch0 | no it is ok, i got: |
| 19:00.04 | ManxPower | b11d: and if you send a fax to that number, you can hear the fax tones when you pick up the phone right? |
| 19:00.12 | ManxPower | I assume this is a dedicated number? |
| 19:00.19 | patrick-- | really_phukt: cant be :D |
| 19:00.22 | b1ch0 | faxdetect=both |
| 19:00.22 | b1ch0 | busydetect=yes |
| 19:00.22 | b1ch0 | busycount=5 |
| 19:00.25 | *** join/#asterisk Overshard (n=Overshar@nc-205-240-45-138.sta.embarqhsd.net) |
| 19:00.32 | ManxPower | b11d: turn all of those off |
| 19:00.58 | ManxPower | you only want fax detection if you want a combined voice/fax number. |
| 19:01.02 | b1ch0 | even faxdetect ?? |
| 19:01.15 | *** join/#asterisk angryuser[A] (i=nononon@df01t2-213-44-151-248.d4.club-internet.fr) |
| 19:01.35 | ManxPower | b11d: do you want Asterisk to magically route your call based on detected fax tone, or do you want the call to go to where you want it? |
| 19:01.40 | b1ch0 | on that port, just fax |
| 19:01.56 | ManxPower | b11d: then stop arguing and remove all three options |
| 19:02.00 | x86 | ManxPower: http://pastebin.ca/902741 |
| 19:02.04 | x86 | ManxPower: that's my zapata.conf |
| 19:02.07 | b1ch0 | but these are overal zapata.conf |
| 19:02.12 | x86 | ManxPower: zaptel.conf was not modified |
| 19:03.00 | x86 | ManxPower: http://pastebin.ca/902743 |
| 19:03.01 | ManxPower | x86: now pastebin the output of a failed call. |
| 19:03.11 | x86 | ManxPower: that's my complete extensions.conf, merged with the test channels |
| 19:03.14 | x86 | channel* |
| 19:03.28 | x86 | ManxPower: err, there is nothing to fail? asterisk doesn't see the digits |
| 19:03.44 | x86 | ManxPower: i could pastebin a blank page and call it the dtmf debug output, if you like... |
| 19:03.58 | flush | yo |
| 19:04.17 | b1ch0 | cant remove last 2 options |
| 19:04.20 | ManxPower | x86: Does Asterisk say "starying simple switch on zap/68"? |
| 19:04.22 | flush | whats the difference between FXS and FXO, one is for plugging normal phones to it and the other is to plug in the wall mount ? |
| 19:04.32 | x86 | ManxPower: yessir, and that's all |
| 19:04.39 | ManxPower | ~fxofxs |
| 19:04.39 | jbot | extra, extra, read all about it, fxofxs is 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. |
| 19:04.53 | b1ch0 | i was having channel problem before, card was not able to detect properly hangup event from pstn |
| 19:04.55 | x86 | ManxPower: are you insane? |
| 19:04.59 | x86 | ManxPower: you saw my signalling! |
| 19:05.14 | ManxPower | x86: I have only one suggestion left -- try a different phone |
| 19:05.18 | x86 | ManxPower: i'm doing FXO signalling on my FXS ports |
| 19:05.22 | x86 | ManxPower: tried 5 different phones |
| 19:05.32 | x86 | all different brands |
| 19:05.36 | ManxPower | x86: then I guess I have no more suggestion. |
| 19:05.42 | x86 | hmm ok |
| 19:05.43 | patrick-- | I have 2 beroNet cards. if my phones are connected to the one card, and my NTBA is connected to the other, will i be able to make outgoing calls without a bridging cable? |
| 19:05.45 | x86 | thanks anyway |
| 19:06.03 | ManxPower | too bad I didn't get to see your CLI output, I might have seen something. |
| 19:06.09 | ManxPower | oh well. |
| 19:06.12 | *** part/#asterisk lmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 19:06.21 | *** join/#asterisk metfan2007 (n=metfan20@201.103.115.64) |
| 19:06.43 | *** join/#asterisk simbol76 (n=simbol@host198-234-dynamic.33-79-r.retail.telecomitalia.it) |
| 19:07.00 | ManxPower | b1ch0: Removing faxdetect fixed the problem, I assume. |
| 19:07.20 | ManxPower | b1ch0: Do you understand what faxdetect does? |
| 19:07.36 | metfan2007 | hi all!!!, anyone knows if is possible to configure Asterisk to listen on multiple ports for SIP? |
| 19:08.05 | bkw_ | metfan2007: is that possible? |
| 19:08.08 | *** join/#asterisk emist (n=emist@unaffiliated/emist) |
| 19:08.11 | ManxPower | metfan2007: I think only in 1.6 |
| 19:08.14 | x86 | ManxPower: if you can interpret something else out of "Starting simple switch on Zap/68", other than the phone went off hook, you must have superpowers |
| 19:08.46 | x86 | metfan2007: you can also redirect ports with iptables |
| 19:08.58 | bkw_ | x86: that doens't work well with sip |
| 19:09.07 | tzafrir | x86, what exactly is your problem? |
| 19:09.13 | x86 | bkw_: sure it does |
| 19:09.21 | tzafrir | I'm trying to separate higer-level Asterisk from lower-level stuff |
| 19:09.25 | x86 | bkw_: doesn't work well with RTP though ;) |
| 19:09.34 | bkw_ | x86: not when the sip messages have port numbers in them |
| 19:09.55 | x86 | bkw_: ah... right |
| 19:09.57 | bkw_ | x86: that really breaks things |
| 19:09.58 | x86 | bkw_: good point :) |
| 19:10.03 | *** join/#asterisk SteveTotaro (n=root@209.213.170.178) |
| 19:10.05 | x86 | bkw_: just dont use rport |
| 19:10.09 | x86 | tzafrir: /query ? |
| 19:10.13 | metfan2007 | x86, mmmm, maybe... the problem here is that the other endpoint (Avaya) says that can only handle 30 calls per port, so they want to use 5 different ports for send calls to Asterisk |
| 19:10.27 | bkw_ | x86: in FreeSWITCH you can launch multiple profiles on as many ports and ip's as you want |
| 19:10.40 | x86 | oh that's kinda cool actually |
| 19:10.48 | tzafrir | even if you forward the port on the kernel level, you might send out a wrong port number |
| 19:10.57 | bkw_ | that whole INADDR_ANY thing is just not a good idea for SIP |
| 19:11.04 | SteveTotaro | anyone have any luck/experience setting up SIP with a Teles switch, the provider is telling me that it will not accept SIP Invites |
| 19:11.23 | SteveTotaro | and they are just timing out after 6 retransmissions |
| 19:12.02 | SteveTotaro | Destroying call '7d22f1a302476bb01e221701154c30fa@88.198.10.70' |
| 19:12.02 | SteveTotaro | Retransmitting #6 (no NAT) to xx.xx.xx.xx:5060: |
| 19:12.02 | SteveTotaro | INVITE sip:49672387116@xx.xx.xx.xx SIP/2.0 |
| 19:12.13 | *** join/#asterisk codefreeze (n=steve_mu@216.166.159.235) |
| 19:12.14 | *** mode/#asterisk [+o codefreeze] by ChanServ |
| 19:12.15 | mintee | what's peoples opinions on aserisk now? |
| 19:12.21 | ManxPower | SteveTotaro: it looks like ASTERISK is timing out. |
| 19:12.35 | ManxPower | mintee: We don't have an opinion, as we don't support it here. |
| 19:12.40 | mintee | *ahem* asteriskNow |
| 19:12.42 | *** join/#asterisk JenniferAkemi (n=akemi@76-10-147-54.dsl.teksavvy.com) |
| 19:12.52 | ManxPower | *ahem* Not supported here. |
| 19:12.52 | bkw_ | ManxPower: looks like the other side doesn't respond thus asterisk goes into retransmitting |
| 19:13.03 | SteveTotaro | yes but I am told by the provider that they do not take SIP invites on their teles switch, not sure how to work around this |
| 19:13.19 | ManxPower | SteveTotaro: you can't make a SIP call without an invite |
| 19:13.29 | SteveTotaro | that is what I thought |
| 19:13.36 | metfan2007 | x86, jejeje, sorry, it is H323, no SIP xD, do you think redirecting port will help? |
| 19:13.39 | ManxPower | perhaps they don't support REinvites. |
| 19:13.53 | SteveTotaro | i have canreinvite set to no |
| 19:14.05 | SteveTotaro | but they are not answering my invites as you can see |
| 19:14.32 | ManxPower | xx.xx.xx.xx looks like a NAT network address to me, but I can't tell for sure. I think my monitor is dirty. |
| 19:14.51 | SteveTotaro | it is all routable IPs |
| 19:15.09 | ManxPower | perhaps you have a packet filtering issue. |
| 19:15.17 | SteveTotaro | no NAT or port forwarding, no firewalls |
| 19:15.35 | ManxPower | if NAT is running on the same box a asterisk, that could be an issue, even if Asterisk is using the public IP. |
| 19:15.47 | ManxPower | SteveTotaro: get it working with a different carrier |
| 19:15.53 | SteveTotaro | it is |
| 19:16.04 | SteveTotaro | BT works fine |
| 19:16.17 | ManxPower | British Telecom has SIP service? |
| 19:16.28 | *** part/#asterisk really_phukt (n=chatzill@static-72-77-217-74.tampfl.dsl-w.verizon.net) |
| 19:16.30 | SteveTotaro | yeah |
| 19:16.37 | SteveTotaro | and it works fine with a few others |
| 19:16.43 | *** join/#asterisk skipper2 (n=DK@chello084113018116.7.12.vie.surfer.at) |
| 19:17.47 | b1ch0 | ManxPower: faxdetect=both means that zap channel is able to detect |
| 19:17.58 | b1ch0 | inconing and outgoing |
| 19:18.02 | b1ch0 | call |
| 19:18.14 | ManxPower | b1ch0: Not just DETECT, DETECT and REROUTE |
| 19:18.46 | ManxPower | It is only useful if you have a DID you want to use with both voice and fax. In you have a dedicated fax number, then faxdetect will just screw things up. |
| 19:19.19 | ManxPower | You don't WANT Asterisk to detect the fax, you want asterisk to just route the call. Now have you removed faxtetect from your config? |
| 19:23.00 | docelmo | Where is the max calls for asterisk located? What config file? |
| 19:23.28 | ManxPower | docelmo: there is none |
| 19:23.37 | skipper2 | hi..one of my sip phones is sending INFO methods (non dtmf) inside a dialog. asterisk seems to not forward them, even though its sending a 200 Ok back to the phone. is there a way to make this work? |
| 19:24.02 | ManxPower | skipper2: tell the phone to use RFC2833 or tell asterisk to use INFO |
| 19:24.44 | skipper2 | manxpower: the user is configured to use info, but it seems anything but dtmf-relay is not forwarded |
| 19:24.52 | *** join/#asterisk [hC] (n=hardcore@vpn.voxter.com) |
| 19:25.14 | ManxPower | skipper2: then configure asterisk for info DTMF in sip.conf for that device. |
| 19:26.10 | *** join/#asterisk zobia (n=laurashr@222.212.77.227) |
| 19:26.13 | skipper2 | manxpower: it is...but again, it only forwards dtmf-relay contenttype, if I send plain/text it will not forward it |
| 19:26.17 | zobia | hello everyone |
| 19:26.43 | ManxPower | skipper2: and again, set Asterisk to dtmf mode INFO. Asterisk won't pass INFO packets if it's not configured for INFO |
| 19:26.44 | zobia | anyone have experience with config 7960 with sip in asterisk ? |
| 19:26.45 | *** join/#asterisk emist (n=emist@unaffiliated/emist) |
| 19:27.20 | *** join/#asterisk IPNorte (n=ircap8@pc-4-150-45-190.cm.vtr.net) |
| 19:27.34 | ManxPower | skipper2: dtmf-relay is rfc2833. You must have Asterisk configured for RFC2833 for that device and since the device is sending INFO packets, asterisk will ignore them. |
| 19:27.39 | IPNorte | Hello |
| 19:29.11 | IPNorte | I've buyed a TDM400P but it's only offhook the phone, no dial, anyone can help me please? |
| 19:30.06 | ArM-eye | amon-ra everyone |
| 19:32.00 | dijungal | where can i see the new asterisk 1.6 features? |
| 19:32.29 | skipper2 | manxpower: rfc 2833 is rtp dtmf payloading...i'm talking about the sip header ContentType: application/dtmf-relay |
| 19:33.39 | skipper2 | manxpower: and that if the ContentType is anything but application/dtmf-relay it will not forward that request |
| 19:33.51 | dijungal | does asterisk support SRTP? |
| 19:34.28 | *** join/#asterisk ManxPower (n=manxpowe@15.sub-75-202-227.myvzw.com) |
| 19:35.29 | dijungal | does asterisk support SRTP? |
| 19:35.48 | x86 | beta |
| 19:37.34 | *** join/#asterisk Overshard (n=Overshar@nc-205-240-45-138.sta.embarqhsd.net) |
| 19:45.32 | mintee | shouldn't anything coming in [from-pstn] match exten => s,1,ringing |
| 19:46.22 | [TK]D-Fender | mintee: In from where? |
| 19:46.42 | mintee | if a write exten => 88865484131,1,ringing (the number i dial, it rings |
| 19:46.47 | *** join/#asterisk saftsack (n=oliver@p54A710DE.dip0.t-ipconnect.de) |
| 19:46.55 | ManxPower | mintee: that depends on many things, but yes, if it's configured right and you are using analog FXO. |
| 19:47.01 | mintee | but if i just use 's' for a type of "catch-all" it doesn't |
| 19:47.14 | mintee | ManxPower PRI |
| 19:47.14 | [TK]D-Fender | mintee: "s" is NOT a cath-all |
| 19:47.35 | [TK]D-Fender | mintee: PRI's target DID's and you must have an exten taht will match. This is all in chapter 5... |
| 19:47.42 | ManxPower | mintee: With a PRI you know what number is dialed. "s" means "no number dialed" |
| 19:48.34 | mintee | oh... strange, I had it working like that with trixbox... but now i'm going back to the basics and just writing them myself. |
| 19:48.53 | ManxPower | mintee: A call to a PRI will never match "s". |
| 19:48.58 | mintee | [TK]D-Fender, lol, chapter 5 of what. |
| 19:49.05 | [TK]D-Fender | ~book |
| 19:49.06 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com, or see ~buybook |
| 19:49.09 | [TK]D-Fender | ^^^^^^^^^^^ |
| 19:49.14 | ManxPower | I don't care if it's trixbox Asterisk or Psychic Friends Asterisk |
| 19:51.31 | mintee | ManxPower, i'm sure there was some other extensions that piped it down to the rule that matched the s. That's why i'm trying from scratch... trixbox was full of crap and confusing |
| 19:51.57 | ManxPower | mintee: start by reading the book, then come back |
| 19:52.29 | mintee | i started reading the book a few weeks back... |
| 19:52.39 | ManxPower | then finish it |
| 19:52.50 | mintee | but i'd rather cut my eyes out with a 2x4 than read an o'reily book |
| 19:52.57 | mintee | imo |
| 19:53.04 | JenniferAkemi | i don't think that's really the way to get help here mintee :P |
| 19:53.46 | mintee | i've recieved the help that I need thus far JenniferAkemi.... Just meerly making a joke. |
| 19:54.06 | ManxPower | Thou Shalt Not Joke About The Good Asterisk Book. |
| 19:54.13 | *** join/#asterisk timeshell (n=Khoja@gw.lusi.on.ca) |
| 19:54.21 | JenniferAkemi | heh |
| 19:54.25 | timeshell | Greetings |
| 19:55.44 | patrick-- | Is there anyone that has experience with HFC cards and mISDN ? |
| 19:59.26 | timeshell | I need a little assistance with understand how devices connect at the IP level |
| 20:00.04 | [TK]D-Fender | timeshell: think a little higher and preface with some actual details of a precice scenario. |
| 20:00.18 | timeshell | I've been trying to get a Polycom to register on port 5061. I've done this before with a PAP2. I just ran Ethereal and watched the REGISTER attempts from the polycom and it is indeed attempting to login on port 5061 |
| 20:00.29 | timeshell | TKD: Patience...I'm getting there |
| 20:00.55 | timeshell | I'm trying to explain this in the most logical way |
| 20:00.57 | ManxPower | timeshell: you realize that what you are trying to do is not normally needed, right? |
| 20:01.14 | timeshell | Manx: It is if you are logging in twice from the same device |
| 20:01.28 | ManxPower | timeshell: no it isn't. |
| 20:01.53 | timeshell | Manx: Then explain to me how to get rid of the digest errors I get when I log in |
| 20:02.10 | patrick-- | Is there anyone that has experience with ISDN / BRI / mISDN? |
| 20:02.13 | timeshell | Using the pap2 before, the only way was to set one line to 5060 and the second to 5061 |
| 20:03.03 | timeshell | I've already tried on the Polycom. They both logon using 5060, but I get digest errors when using the second line. |
| 20:03.33 | ManxPower | using different userids |
| 20:03.37 | timeshell | yes |
| 20:04.32 | timeshell | Line 1 is 5221, line 2 is 5121. Get error something like "Auth is <5221>, digest has <5121>" |
| 20:04.56 | timeshell | I got around this before on the pap2 just by having the second line use port 5061 |
| 20:05.03 | *** join/#asterisk smackd00d (n=smc@CPE00500417f78f-CM00137186e4ae.cpe.net.cable.rogers.com) |
| 20:05.13 | *** join/#asterisk lunaphyte__ (n=lunaphyt@207.106.12.118) |
| 20:07.16 | timeshell | So, as I was saying, I see the REGISTER request going out from the phone on to myip:5061, but I see nothing coming back...not registering on the asterisk server. |
| 20:07.45 | *** join/#asterisk droops (n=droops@74.193.237.138) |
| 20:08.06 | timeshell | I'm at a loss as to why since the same server accepts port 5061 from my pap2 |
| 20:09.03 | ManxPower | we use polycoms all the time, one registration per line, no issues. |
| 20:09.31 | timeshell | Well, I'm open to suggestions. |
| 20:09.44 | timeshell | I've tried numerous variations of the config. |
| 20:10.17 | timeshell | I'm using Polycom 301 SIP2.2.2 |
| 20:11.11 | timeshell | You registering 2 lines on one polycom with the same asterisk server, each line with it's own userid? |
| 20:11.55 | timeshell | And on the same outgoing port? |
| 20:12.20 | ManxPower | yes. we do not change the port number. |
| 20:12.43 | timeshell | You make outbound calls from both lines? |
| 20:12.43 | AmR-eye | amon-ra everyone |
| 20:12.52 | AmR-eye | Would you like to read the reveal? |
| 20:12.58 | AmR-eye | It involves biblical figures |
| 20:13.16 | AmR-eye | yes or no? |
| 20:13.24 | ManxPower | On some phones we have all six lines registered to the same server different userids |
| 20:13.30 | AmR-eye | it must not be given against will |
| 20:13.46 | *** join/#asterisk anthm (n=anthm@CPE-72-131-113-50.wi.res.rr.com) |
| 20:13.46 | *** mode/#asterisk [+o anthm] by ChanServ |
| 20:13.51 | timeshell | And you can make outbound calls on any one of those lines? |
| 20:14.00 | AmR-eye | BEGINNITIO REVELATIONEM |
| 20:14.01 | AmR-eye | DECODE BIBLIA |
| 20:14.01 | AmR-eye | Revel - Abraham is associated with the Egyptian pharaoh Amenemhat I (translates: amen is the head) who worshiped the god Amun (Amen). Abraham god then be associated with in the Abrahamic religions god as amun, amon, omon, amen and the deity aamon. Abraham/Amenemhet I |
| 20:14.06 | AmR-eye | Revel - Jacob = King Yakubher |
| 20:14.08 | AmR-eye | Revel - Moses = Thutmose III |
| 20:14.11 | AmR-eye | Revel - David = Psusennes I |
| 20:14.13 | AmR-eye | Revel - Solomon = Siamun (translates: son of amun) |
| 20:14.16 | AmR-eye | Revel - James = Ptolemy Philadelphus |
| 20:14.18 | AmR-eye | Revel - Thomas Judas Didymus = Alexander Helios |
| 20:14.21 | AmR-eye | Revel - Mary Magdalene = Cleopatra Selene II |
| 20:14.23 | AmR-eye | FINALIZE REVELATIONEM |
| 20:14.26 | AmR-eye | BIBLIA CHARACTER 1 |
| 20:14.28 | *** mode/#asterisk [+b *!*i=elf@*.client.mchsi.com] by Qwell |
| 20:14.28 | *** kick/#asterisk [AmR-eye!i=north@pdpc/sponsor/digium/Qwell] by Qwell (Qwell) |
| 20:15.02 | [hC] | Aw damnit, i was learning about pharaohs! |
| 20:15.20 | Qwell | RTFB |
| 20:15.21 | Qwell | :D |
| 20:16.27 | [hC] | hahah |
| 20:16.30 | timeshell | AmR-eye: What do you base your "Revel"s on? |
| 20:19.05 | b11d | I finally got my FAXing to work perfectly.. |
| 20:19.09 | b11d | I was so stupid :) |
| 20:19.27 | mvanbaak | b11d: exten => fax,1,Hangup() ? |
| 20:19.28 | ManxPower | b11d: what was it? |
| 20:20.01 | b11d | I had two T1 ports on a single four port T1 card.. Port1 was PRI to telco, synched the clock from the telco.. the port2 was connected to a Rhino CB24 channel bank.. the Rhino was acting as a MASTER clock for the 2nd t1 port.. I disabled that.. and made port2 slave to port1s clock.. works tits now. |
| 20:20.06 | b11d | mvanbaak.. :) hahaha |
| 20:20.07 | timeshell | So then Manx, why do I get digest errors when I attempt to do the same thing that works for you? |
| 20:20.31 | ManxPower | timeshell: you are doiing something wrong 8-) |
| 20:20.43 | mvanbaak | hahahahahaha |
| 20:20.51 | timeshell | Manx: Evidentally...guidance on what that may be? |
| 20:21.08 | jameswf | http://video.stumbleupon.com/#p=9yrn62ou2o gah |
| 20:21.15 | jameswf | wait no cancel that |
| 20:21.36 | ManxPower | timeshell: I don't know enough about your config. Diagram it for us. |
| 20:21.37 | b11d | strike that, reverse it. |
| 20:22.19 | *** join/#asterisk tparcina (n=tparcina@78-3-87-77.adsl.net.t-com.hr) |
| 20:22.26 | *** part/#asterisk tparcina (n=tparcina@78-3-87-77.adsl.net.t-com.hr) |
| 20:24.37 | *** join/#asterisk rafiks (n=rafiks@c-68-56-23-83.hsd1.fl.comcast.net) |
| 20:24.54 | ManxPower | timeshell: so you have Polycom -> Local LAN -> Asterisk |
| 20:25.26 | timeshell | Yes |
| 20:25.38 | mocker | Guh, anyone had problems w/ Polycom IP330s freezing up? |
| 20:25.46 | mocker | rebooting randomly? |
| 20:25.54 | b11d | what version of SIP & bootrom mocker? |
| 20:26.23 | timeshell | I'm reconfiguring to give the exact digest error |
| 20:26.25 | mocker | Varies across pones.. |
| 20:26.29 | mocker | Some at 3.2.3.0021 |
| 20:26.34 | b11d | and they ALL reboot randomly? |
| 20:26.36 | mocker | Some at 2.2 |
| 20:26.44 | mocker | Actually it's at the same time. |
| 20:26.47 | rafiks | hey! |
| 20:26.53 | mocker | So I'll have like 30 phones freeze |
| 20:26.59 | b11d | all connected to the same PoE switch by chance? |
| 20:27.07 | timeshell | heh |
| 20:27.07 | mocker | Two different PoE switches. |
| 20:27.11 | mocker | But it's *just* the IP330s. |
| 20:27.12 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 20:27.17 | b11d | thats bizarre.. |
| 20:27.18 | mocker | IP550s, etc.. are fine. |
| 20:27.25 | mocker | b11d: I know. :( |
| 20:27.29 | b11d | upgrade to the latest bootom & SIP i guess.. |
| 20:27.29 | cmantito | bad grounding possible |
| 20:27.33 | cmantito | possibly?* |
| 20:27.36 | mocker | What's the latest firmware on the IP330? |
| 20:27.36 | rafiks | whats the most common cause of voip calls being dropped |
| 20:27.45 | droops | hangups |
| 20:27.45 | b11d | isnt 3.0.0 available for the 330? |
| 20:27.51 | Qwell | droops: you beat me to it :) |
| 20:28.02 | mocker | I'm on hold w/ my vendor trying to get the latest release. |
| 20:28.04 | b11d | rafiks.. admins stopping and restarting asterisk? :P |
| 20:28.07 | droops | i have been waiting for like 2 years on that one |
| 20:28.20 | [hC] | rafiks: packet loss, in my experience |
| 20:28.33 | [hC] | rafiks: or link outages |
| 20:28.50 | rafiks | i have an asterisk box here and I've talked to my provider and he tells me to try calling directly to see if tis my box .. |
| 20:28.54 | [hC] | rafiks: or actually, cpu load on the box can do it too. |
| 20:28.56 | *** join/#asterisk DarWin_vcch (n=daryl@205.241.238.3) |
| 20:30.23 | cmantito | ok, any thoughts on this? |
| 20:30.23 | cmantito | tom*CLI> core show function cdr |
| 20:30.23 | cmantito | No function by that name registered. |
| 20:30.28 | *** join/#asterisk ACiDV (n=joel@122-205-229.dr.cgocable.ca) |
| 20:31.00 | rafiks | [hC] : i have a p4 1.8 ghz box..how much more do i need? |
| 20:31.05 | cmantito | case sensitive, sorry XD |
| 20:31.13 | [hC] | rafiks: its not about how much you need, mpg123 can peg your cpu on anything. |
| 20:31.31 | timeshell | <PROTECTED> |
| 20:31.31 | timeshell | , have <5121>, digest has <5221> |
| 20:31.31 | timeshell | <PROTECTED> |
| 20:31.31 | timeshell | <PROTECTED> |
| 20:31.46 | apocn | Hello, Im using MixMonitor to record the conversations of the agents logged on the Queue. Now I want to listen to them live, should I use ExtenSpy? |
| 20:32.07 | [TK]D-Fender | apocn: probably chanspy |
| 20:32.35 | timeshell | That is what I get when they both logon using the same port. |
| 20:32.48 | timeshell | I used to get that on the PAP2 until I changed line2 port to 5061 |
| 20:33.00 | JenniferAkemi | what are the thoughts on the gui in general |
| 20:33.05 | JenniferAkemi | is it useful? |
| 20:33.18 | ManxPower | timeshell: are the two lines configured for different users? |
| 20:33.21 | rafiks | [hC] : its not like mpg123 is always running on the backgrounf ..how can this be an issue.. this machine is basically a dedicated box |
| 20:33.25 | timeshell | Manx: Yes |
| 20:33.38 | timeshell | Manx: Line1 is 5221, Line2 is 5121 |
| 20:33.48 | [hC] | rafiks: you arent hearing me. |
| 20:33.50 | timeshell | Those are the actual user ID's |
| 20:33.51 | ManxPower | put your MAC-phone.cfg and your sip.conf on pastebin.ca |
| 20:34.01 | [hC] | rafiks: when i said pegging your cpu, you want to look for stuff like runaway processes |
| 20:34.15 | [hC] | rafiks: im not saying it would always be pegged, but its possible that it could happen by accident without you realizing it |
| 20:34.26 | [hC] | rafiks: by means of a runaway process, not by actual USAGE |
| 20:34.54 | rafiks | [hC] ok gotcha.. |
| 20:35.10 | b11d | whats wrong mocker? |
| 20:35.13 | mocker | All I want is the latest firmware! :) |
| 20:35.24 | mocker | 15 minutes on hold in the support queue. |
| 20:35.26 | ManxPower | timeshell: see also http://www.fnords.org/~eric/polycom-config-examples |
| 20:35.29 | timeshell | http://www.pastebin.ca/902871 |
| 20:36.47 | *** join/#asterisk ZaVoid (n=zavoid@66-95-182-90.client.dsl.net) |
| 20:36.50 | ZaVoid | hey guys |
| 20:36.55 | ManxPower | timeshell: sip.conf not sip.cfg |
| 20:37.02 | timeshell | oops |
| 20:37.02 | timeshell | sorry |
| 20:37.06 | ZaVoid | what would caouse the g729 register tool to say the g729 license is registered but show g729 doesn't work |
| 20:37.11 | timeshell | My sip.conf is empty...I use users.conf |
| 20:37.14 | ManxPower | timeshell: look at the example url I sent you |
| 20:37.18 | timeshell | I am |
| 20:37.27 | ManxPower | timeshell: I cannot help you then. |
| 20:37.39 | ManxPower | sip.conf is what we use around here. |
| 20:37.54 | ManxPower | So I cannot help you. |
| 20:38.01 | timeshell | Can you send sample of your sip.conf? |
| 20:38.48 | patrick-- | I have a weird Problem: one of my ISDN Phones works on a HFC NT Port on my asterisk, but another wont... could that be cause of the phones software version? |
| 20:39.11 | jameswf | I miss the days when people would patch their own kernels.... |
| 20:39.27 | b11d | i still patch mine manually |
| 20:39.49 | ManxPower | timeshell: hold on |
| 20:40.01 | *** join/#asterisk c4t3l (n=c4t3l@74.95.210.124) |
| 20:42.17 | ManxPower | timeshell: I put part of a production sip.conf on that polycon examples url |
| 20:43.57 | timeshell | Manx: This still has me baffled. My original pap2 config used sip.conf and all in all, was virtually configured that same way. |
| 20:44.05 | timeshell | But I still had the problem on my pap2 as well back then |
| 20:44.38 | timeshell | anyway...afk for 10 mins... |
| 20:44.39 | ManxPower | timeshell: I have over 80 Polycom phones registering to the same server, each with at least 2 lines, some with 6 lines, never had to change any port numbers |
| 20:44.41 | timeshell | coffee time |
| 20:45.03 | timeshell | Manx: Not doubting you...looking for an answer as to why mine doesn't work. |
| 20:45.05 | timeshell | brb |
| 20:48.08 | b11d | ive got 250 polycoms registering to the same server.. no port changes. |
| 20:49.04 | c4t3l | hello all. was there a major change in the way polycom displays caller info from sip v 1.0 to sip v 2.1? |
| 20:49.22 | b11d | read the changelog i guess |
| 20:49.37 | b11d | sip 1.. what year was that released? 3 just came out. |
| 20:49.59 | b11d | and you went from sip 1 to 2.1 without reading the changelog first? wow.. ballsy :) |
| 20:50.08 | c4t3l | sorry, sorry, sip 2.0 to sip 2.1 |
| 20:50.12 | b11d | oh.. ok :) |
| 20:50.30 | b11d | hmm... i dont think it changed at all then. |
| 20:50.38 | ManxPower | still should have read the changelog, |
| 20:50.44 | b11d | aye |
| 20:50.46 | c4t3l | i have read the changelog |
| 20:51.25 | c4t3l | but the problem is display. I'm working with a company that wont give me direct access to the phone.cfg or sip.cfg files |
| 20:51.44 | b11d | whats the issue exactly? |
| 20:52.04 | c4t3l | customers are complaining that after the upgrade from 2.0 to 2.1 that muliple line caller ID has disappeared |
| 20:52.55 | *** join/#asterisk jackm1944 (n=IceChat7@CPE000f664f0f37-CM0014045a95ea.cpe.net.cable.rogers.com) |
| 20:53.02 | c4t3l | i cant get a straight answer from the supplier. so I'm kinda in a spot. I really just want to know for my own benifit |
| 20:53.23 | b11d | hmm.. |
| 20:53.29 | ManxPower | c4t3l: you might want to come back when you can troubleshoot the problem. |
| 20:53.36 | *** join/#asterisk Maxous (n=Maxous@74.7.13.242) |
| 20:53.46 | b11d | i'll say that I didnt notice any change in callerID display.. |
| 20:53.53 | b11d | but cant confirm it.. and sounds like neither can you |
| 20:55.26 | *** join/#asterisk hi365_m (n=hi365@213.151.62.64) |
| 20:55.56 | tzafrir | partial subject on a mailing list message: "Asterisk Manager and Vi" |
| 20:56.14 | tzafrir | Looked promising |
| 20:57.26 | *** join/#asterisk deeperror (n=deeperro@69-215-202-195.ded.ameritech.net) |
| 20:58.49 | c4t3l | ManxPower: sorry, I'm just a bit frustrated cuz I cant see the configs. |
| 20:59.02 | deeperror | Would switching from channel banks connected to t1 cards use more or less CPU than running the same amount of calls with sip clients? |
| 20:59.24 | b11d | SIP calls use more CPU.. |
| 20:59.39 | deeperror | thats what i thought due to software ec? |
| 20:59.44 | b11d | T1 cards to their own processing.. SIP is done in CPU. |
| 20:59.53 | b11d | to = do |
| 21:00.01 | ManxPower | Zaptel cards do their processing on the CPU |
| 21:00.31 | ManxPower | Switching from channel bank to SIP would cause a little bit more CPU usage, but if everything else stays the same, it should not be much more CPU |
| 21:00.32 | deeperror | currently we run about 50 channels on zap and i have a load average around 1-2 that would probably go way up then |
| 21:00.47 | b11d | load averages are not based on CPU usage.. |
| 21:00.49 | b11d | read up |
| 21:00.57 | ManxPower | deeperror: you must have a VERY slow system |
| 21:00.57 | deeperror | cpu load |
| 21:01.38 | deeperror | it is agree...this is due to a limitation in the t1 cards we are using |
| 21:01.51 | ManxPower | deeperror: which T-1 cards are you using? |
| 21:01.58 | deeperror | started with rhino r4t1 |
| 21:02.07 | deeperror | now using 3x r1t1 |
| 21:02.26 | deeperror | pushed 157,000 minutes last month on it |
| 21:02.39 | ManxPower | I don't know enough about the Rhino cards to comment. Are they based on the original Zapata reference design? |
| 21:02.46 | deeperror | yea they are |
| 21:02.52 | deeperror | it was something to do with their on board ecm |
| 21:02.58 | ManxPower | Then they are the worst possible choice. |
| 21:03.13 | zobia | hello every one |
| 21:03.17 | ManxPower | Sangoma and modern Digium cards are much better. |
| 21:03.27 | ManxPower | Also, if you have three cards in the system, you will have 3x the overhead |
| 21:03.31 | deeperror | well were making the move to 100% sip solution in 3-4 months |
| 21:03.32 | b11d | I heart my Sangoma A104d's |
| 21:03.38 | zobia | any one knows what code asterisk send to turn the message light on? |
| 21:03.38 | deeperror | so they will be getting phased out soon |
| 21:03.50 | b11d | watch your modems and FAXes deeperror.. if any. |
| 21:04.15 | zobia | any one knows what code asterisk send to turn the message light on? |
| 21:04.23 | b11d | they dont "do so well" in an all SIP environment |
| 21:05.17 | deeperror | yea all agents in a callcenter |
| 21:05.22 | b11d | ok |
| 21:06.04 | zobia | any one knows what code asterisk send to turn the message light on? i want to manually set one phone's light on |
| 21:06.33 | b11d | "set polycom EXTENSION msg ind light enable=1" at the CLI |
| 21:06.46 | b11d | ok.. that was a lie :) |
| 21:07.08 | ManxPower | zobia: you cannot manually set an MWI light on |
| 21:07.16 | ManxPower | MAYBE in 1.6, but not in 1.4 or earlier |
| 21:07.27 | JerJer | sure you can - just not with asterisk :) |
| 21:07.30 | b11d | whats coming down the line in 1.6 anyways? |
| 21:07.35 | JerJer | sipsak works great |
| 21:07.44 | ManxPower | b11d: pretty much everything listed in the changelog |
| 21:07.51 | b11d | oh.. neat.. is that what that is? |
| 21:07.52 | b11d | :) |
| 21:08.02 | zobia | ManxPower: in 1.6 how to manually set the MWI light on? |
| 21:08.04 | b11d | a log of the changes? |
| 21:08.08 | b11d | :) |
| 21:08.18 | ManxPower | zobia: I have no idea. 1.6 has not been released. Its still in beta |
| 21:08.33 | *** join/#asterisk glen2 (n=glen@87-194-2-134.bethere.co.uk) |
| 21:08.34 | ManxPower | and I said MIGHT |
| 21:08.41 | *** join/#asterisk generalhan (n=asd@ip67-90-64-2.z64-90-67.customer.algx.net) |
| 21:09.06 | zobia | ManxPower:thank you. if without asterisk hwo to set the MWI on? |
| 21:09.18 | ManxPower | zobia: Ask JerJer |
| 21:09.34 | deeperror | i think we will get a quad core server when making the switch to 100% sip that should handle the load. Were now just using junk to interface with our analog junk. |
| 21:09.48 | b11d | hahaha |
| 21:09.51 | b11d | thats SO overkill |
| 21:10.07 | zobia | JerJer: how to set MWI on without asterisk? |
| 21:10.16 | deeperror | haha |
| 21:10.17 | deeperror | dual core? |
| 21:10.22 | generalhan | hey all, im currently running a 1.2 and am thinking about upgrading to 1.4.18 ... will the change log for the 1.4.18 tell me of the changes from 1.2 to 1.4 ? or will i have to get my hands on a 1.4.0 changelog/upgrade ? |
| 21:10.28 | fujin | hiyas |
| 21:10.31 | b11d | dual is smart. |
| 21:10.36 | fujin | I currently have this: http://rafb.net/p/l83uZo46.html |
| 21:10.46 | ManxPower | generalhan: you should look at the upgrade.txt in BOTH 1.2 and 1.4 |
| 21:10.49 | fujin | I want to make it so that if an agent is available (they take DND off), they're delivered the call in less than 5 seconds |
| 21:10.51 | b11d | generalhan.. i just did the same thing.. didnt run into much.. I have a pretty simple dialplan thouhg. |
| 21:10.52 | fujin | what setting sdo I need?? |
| 21:10.59 | *** join/#asterisk ShakaGoldSaint (n=eleazar@190.38.75.102) |
| 21:11.00 | zobia | JerJer: Are you there? |
| 21:11.11 | JerJer | i am just a fig newton of your imagination |
| 21:11.29 | generalhan | ManxPower: the upgrade for 1.2.23 and 1.4.18 will give me all the info i need ? or do i need to get the 1.4.0 upgrade ? |
| 21:11.40 | b11d | i went from 1.2.12 to 1.4.17 just fine.. |
| 21:11.43 | ManxPower | generalhan: you do not need anything from 1.4.0 |
| 21:11.48 | b11d | but i tested my config on another machine first.. |
| 21:11.53 | generalhan | b11d: my dialplan is pretty involved right now and its production so i need to be sure there are no mistakes |
| 21:12.02 | generalhan | ManxPower: thank you ! |
| 21:12.07 | b11d | then you better setup a test box and learn the issues ahead of time |
| 21:12.40 | zobia | JerJer: hope it can be possible in 1.6 |
| 21:12.41 | b11d | i had to ditch digitTimeout and that was about it.. |
| 21:12.56 | b11d | zobia.. i have to know.. why do you want to do that anyway? |
| 21:14.50 | mintee | ok, so i've read the chapter on dialplans... |
| 21:14.59 | mintee | and it specifically said to use s |
| 21:15.23 | mintee | but asterisk is still rejecting the call |
| 21:15.25 | ManxPower | mintee: what page? |
| 21:15.28 | ManxPower | ~book |
| 21:15.29 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com, or see ~buybook |
| 21:15.35 | mintee | 127 |
| 21:15.45 | ManxPower | downloading it now. |
| 21:16.05 | mintee | i have 23 openchannels with the context of [from-pstn] |
| 21:16.38 | mintee | and the dial plan set just like on page 127 |
| 21:16.49 | mintee | "If it doesn’t work, check the Asterisk console for error messages, |
| 21:16.49 | mintee | and make sure your channels are assigned to the [incoming] context." |
| 21:17.06 | mintee | I'm not sure what it means by "channels are assigned to the [incoming] context." |
| 21:17.10 | zobia | b11d: i want to do it because we receive voicemail on asterisk. but the phone is not registered on asterisk. it 's registered on callmanager |
| 21:17.18 | ManxPower | perhaps if you stopped talking and waited for me to find that spot in the book, evernyone will be happy. |
| 21:18.31 | b11d | ahhh.. |
| 21:18.33 | b11d | tough one :) |
| 21:18.44 | zobia | b11d: yes. to tough |
| 21:18.49 | zobia | too tough |
| 21:19.41 | ManxPower | mintee: that example is not for PRI |
| 21:20.18 | ManxPower | You can argue, scream, throw a temper tantrum, but that will not change the fact that you should not use exten "s" on a PRI. |
| 21:20.33 | mintee | O_o |
| 21:20.53 | mintee | your the one that's throwing a tantrum... I'm just trying to figure this out. |
| 21:21.02 | mintee | you tell me to read the book, so I do |
| 21:21.07 | ManxPower | If you look at the top of page 125 you will see the example is for an FXO port. |
| 21:21.23 | b11d | i rock a PRI and use the s extension in some limited sense.. but certainly not in the default context. |
| 21:21.28 | b11d | if thats what is meant by that |
| 21:21.46 | ManxPower | mintee: you didn't read the book, you found a part of the book, then tried to use that information out of context. |
| 21:22.19 | ManxPower | b11d: No, what is meant is that an incoming call on a PRI won't match the empty extension (extension "s" is the "empty extension") |
| 21:23.01 | mintee | that doesn't say the example is for an FXO... reread that sentance |
| 21:23.23 | mintee | regardless, it's not helping... |
| 21:23.40 | b11d | ahh |
| 21:23.45 | mintee | i know if i specifically put exten => s,8872721464,dosomething() |
| 21:23.48 | mintee | err |
| 21:23.51 | b11d | lol |
| 21:23.59 | mintee | i know if i specifically put exten => 8872721464,1,dosomething() |
| 21:24.15 | mintee | it will dosomething() when I dial 8872721464 |
| 21:24.39 | ManxPower | That is correct. |
| 21:24.42 | mintee | what I'm looking to do is dosomething() based on the number that is being dialed. |
| 21:24.44 | ManxPower | That is the way to set it up. |
| 21:24.50 | mintee | i have 4000 numbers |
| 21:25.02 | ManxPower | then I guess a pattern match is in your future. |
| 21:25.06 | mintee | i'm not writing 4000 extensions |
| 21:25.31 | ManxPower | if you want each of the 4000 extensions to do something different you will have to write 4000 extensions, if you don't then a wildcard is just fine. |
| 21:26.16 | mintee | well, once i capture the number i'll call it with a variable to route... |
| 21:26.45 | mintee | ${DNID}: |
| 21:26.50 | ManxPower | Page 137-140 talk about pattern matching |
| 21:26.56 | mintee | k |
| 21:27.13 | mintee | damnit... i closed the book on accident. |
| 21:27.14 | ManxPower | Now, read the rest of the book |
| 21:28.22 | mintee | all i know, is somehow, trixbox setup a catchall, and i foolishly didn't save the extension.conf file |
| 21:28.46 | ManxPower | there is a catch all, it's not "s" |
| 21:29.09 | ManxPower | The fact that Trixbox does something should indicate yo you what NOT to do. |
| 21:29.20 | mintee | lol, true |
| 21:30.28 | b11d | exten => _.,BLAH |
| 21:30.30 | b11d | catch all :P |
| 21:30.37 | jackm1944 | i am using asterisk 1.2, not Trixbox, does anyone know how to do ondemand recording? |
| 21:30.48 | b11d | record() |
| 21:31.01 | ManxPower | b11d: You will, of course, help him when that pattern screws up his dialplan, right? |
| 21:31.02 | jackm1944 | while I am on the phone? |
| 21:31.07 | b11d | no :) |
| 21:31.18 | ManxPower | jackm1944: look up one touch recording for asteris, |
| 21:31.32 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 21:31.38 | jackm1944 | one touch recording??? |
| 21:32.29 | ManxPower | Results 1 - 10 of about 72 from lists.digium.com for "one touch recording". AND "too lazy to use google" (0.32 seconds)Â |
| 21:32.29 | jackm1944 | http://archives.free.net.ph/message/20060117.055348.0ac66584.en.html |
| 21:33.06 | b11d | Manx.. you rule |
| 21:33.40 | ManxPower | jackm1944: that message does not contain all the information you need. You also have to setup features.conf |
| 21:35.28 | jackm1944 | thanks but if I have two asterisk machines, suppose calls from like this Call -> Asterisk 1 -> Asterisk 2, do I need to put the option w on both dial command in both Asterisk or just in only one that is originating the call? |
| 21:35.53 | *** part/#asterisk deeperror (n=deeperro@69-215-202-195.ded.ameritech.net) |
| 21:36.08 | ManxPower | put it on the system where you want the recording to happen |
| 21:36.27 | jackm1944 | ok, thank you |
| 21:37.33 | *** join/#asterisk riddlebox (n=james@75-132-225-75.dhcp.stls.mo.charter.com) |
| 21:40.16 | *** join/#asterisk sarthor (n=sarthor@77.64.65.102) |
| 21:41.34 | mintee | :D |
| 21:41.35 | mintee | exten => _X.,1,Goto(incoming|${DNID}|1,) |
| 21:42.20 | ManxPower | mintee: That should crash your PBX if that exten line is in the [incoming] context. |
| 21:42.37 | ManxPower | because DNID will match _X. |
| 21:43.29 | mintee | icey. currently I don't have any specifics in the [incoming] just more _X.'s |
| 21:43.49 | mintee | thanks for your stubborn help thus far ;) |
| 21:46.15 | mintee | nah, it didn't crash |
| 21:46.28 | *** join/#asterisk anthm (n=anthm@CPE-72-131-113-50.wi.res.rr.com) |
| 21:46.28 | *** mode/#asterisk [+o anthm] by ChanServ |
| 21:46.37 | mintee | i just set a specific [incoming] extension for the number i dial |
| 21:46.42 | mintee | and it went thru fine |
| 21:47.53 | generalhan | hmm... seems like a lot of of things are going to need to be changed in my dialplan :( |
| 21:48.31 | JenniferAkemi | i thoguth adpcm WAS g726 |
| 21:48.35 | JenniferAkemi | is that wrong? |
| 21:48.42 | generalhan | i need to figure out a way to rig a test box ... i cant unplug the PRI line to my production machine, so maybe ill have to wait until the weekned to test this new box |
| 21:49.03 | *** join/#asterisk tecnico (n=tecnico@user-24-214-56-217.knology.net) |
| 21:49.31 | ManxPower | no, adpcm is adpcm |
| 21:49.42 | *** join/#asterisk gvasterisk (n=gvasteri@200.69.249.33) |
| 21:49.57 | *** part/#asterisk gvasterisk (n=gvasteri@200.69.249.33) |
| 21:50.04 | JenniferAkemi | how come the book says It is also known as Adaptive Differential |
| 21:50.04 | JenniferAkemi | Pulse-Code Modulation (ADPCM), and it can run at several bitrates. |
| 21:50.23 | JenniferAkemi | on page 195 |
| 21:50.32 | ManxPower | oh, it might use adpcm as part of the codec. |
| 21:50.46 | ManxPower | part |
| 21:51.04 | JenniferAkemi | strange. |
| 21:51.10 | JenniferAkemi | i wonder what the difference is |
| 21:51.56 | *** part/#asterisk dijungal (n=kdaniel@63.175.159.171) |
| 21:52.14 | ManxPower | several codecs use adpcm |
| 21:53.34 | *** join/#asterisk teknoprep (n=Chris@unaffiliated/teknoprep) |
| 21:53.34 | *** join/#asterisk grandpapadot (n=no@adsl-074-185-089-046.sip.bhm.bellsouth.net) |
| 21:53.35 | teknoprep | hey all |
| 21:53.41 | teknoprep | i just bought an IP650 from polycom |
| 21:53.53 | teknoprep | and it just keeps rebooting on its own while on a call... every call makes it reboot |
| 21:54.09 | teknoprep | i check the app.log in the tftp server and its just a bunch of numbres |
| 21:54.18 | Qwell | teknoprep: call polycom? |
| 21:54.29 | *** part/#asterisk Maxous (n=Maxous@74.7.13.242) |
| 21:54.38 | teknoprep | Qwell, i have to call my supplier as polycom will not help end-users |
| 21:54.45 | teknoprep | Qwell, or so there technical support line says |
| 21:55.49 | Qwell | call your supplier then |
| 21:56.03 | teknoprep | yeah already did.. not very smart ppl |
| 21:56.16 | ManxPower | JenniferAkemi: http://forskningsnett.uninett.no/voip/codec.html |
| 21:57.46 | ManxPower | teknoprep: what version of firmware? |
| 21:57.51 | teknoprep | 3.2.3 |
| 21:57.54 | ManxPower | and what version of config file? |
| 21:57.57 | teknoprep | then i upfraded to 4.0 |
| 21:58.01 | teknoprep | same problem |
| 21:58.07 | teknoprep | version of config file ? |
| 21:58.20 | ManxPower | and downgrading to a known good release (2.1 or 2.2) does not fix it either? |
| 21:58.20 | sarthor | Hi, i am using linux "ubuntu Gutsy", trying to use Ekiga for lowaratevoip.com, to call. Working fine, but there is not balance now in my account, the pakage have free minuts, i can use that free minuts from xp on the lowratevoip dialer , but not on Ekiga?? help Using this toturail " http://didier.misson.net/didier/index.php?2007/09/17/138-sip-ekiga-avec-low-rate-voip " |
| 21:58.26 | teknoprep | i have the newest sip firmware also installed |
| 21:58.42 | *** join/#asterisk d00gster (n=doughant@bas1-toronto12-1088937109.dsl.bell.ca) |
| 21:58.43 | teknoprep | let me try 2.1 then |
| 21:58.46 | teknoprep | or 2.2 |
| 21:58.52 | ManxPower | and are your sip.cfg and phone1.cfg from that version of the firmware? |
| 21:59.12 | ManxPower | you don't want to run a sip.cfg from 2.1 on a 3.x phone, etc |
| 21:59.30 | teknoprep | let me check |
| 21:59.54 | ManxPower | sarthor: This is an Asterisk support channel. Not a voip service support channel |
| 22:00.33 | sarthor | ManxPower, some one in linuxhelp chan told me to ask here, So my friend where to ask on irc? please guide me |
| 22:01.14 | ManxPower | sarthor: we don't know anything about anything you are using. |
| 22:01.36 | ManxPower | sarthor: your asking here makes as much sense as asking for Microsoft Windows help at a political rally. |
| 22:03.45 | sarthor | ManxPower, Strange. I already Got. What do you want to prove? if there is any satisfaction for you in this.. So Speak, i will listen, but if you want to tell me that i am on wrong place. So i am not asking again and again. |
| 22:05.02 | *** join/#asterisk hesco (n=hesco@c-76-17-99-50.hsd1.ga.comcast.net) |
| 22:05.21 | hesco | Using the dial command, how is it I instruct the server to dial an extension, after a remote phone system has answered? |
| 22:06.12 | ManxPower | Dial(Zap/g1/5551212,,D(1234)) IIRC |
| 22:06.27 | hesco | thanks ManxPower! |
| 22:06.44 | ManxPower | hesco: you need to look at "core show application dial" |
| 22:09.42 | JenniferAkemi | when i type in "sip show peer 6004" is it getting the information from the phone config or from the config in users.conf and sip.conf? in particular, the codec part. |
| 22:10.15 | ManxPower | JenniferAkemi: Asterisk knows nothing about the phone config. |
| 22:10.49 | JenniferAkemi | ok |
| 22:10.52 | hesco | how would you do that from inside a call file? Any ideas? |
| 22:11.44 | JenniferAkemi | thanks ManxPower |
| 22:13.03 | *** join/#asterisk b1shop (n=b1shop@dsl081-149-253.chi1.dsl.speakeasy.net) |
| 22:15.37 | *** join/#asterisk Legend (n=Legend@65.75.85.82) |
| 22:15.42 | *** join/#asterisk darkskiez (i=mhb@cpc1-broo3-0-0-cust659.renf.cable.ntl.com) |
| 22:16.44 | *** join/#asterisk jm|home (n=jamie@zen.jamiem.com) |
| 22:17.22 | jm|home | I'm nearly there with chan_mobile but I can't get my V3r to see my bluetooth/asterisk server |
| 22:18.12 | *** join/#asterisk [TK]D-Fender (n=Joe@64.235.216.2) |
| 22:18.23 | jm|home | hello [TK]D-Fender |
| 22:18.42 | jm|home | [TK]D-Fender, can you help me with chan_mobile? |
| 22:19.39 | jm|home | hm |
| 22:20.23 | jm|home | it's doing something now |
| 22:20.35 | mintee | doesn't the following supposed to spit out the DIALEDTIME onto the CLI ??? exten => _X.,1,NoOp(${DIALEDTIME}) |
| 22:21.49 | mintee | because all i'm getting is -- Executing [888xxxxxxx@from-pstn:1] NoOp("Zap/1-1", "") in new stack |
| 22:24.06 | Legend | is 8.2 the latest firmware for the cisco 7900s? |
| 22:24.09 | *** join/#asterisk |dennis| (n=dennis@200.32.233.84) |
| 22:25.40 | grandpapadot | 8.2 is the latest *working* software for the 7940/7960 if you're using NAT and Asterisk. |
| 22:25.58 | *** join/#asterisk sergey (n=sergey@sergey.iks.ru) |
| 22:26.19 | Legend | no nat, the phone is on the same subnet as the asterisk |
| 22:26.20 | *** join/#asterisk rhombus (n=sfbosch@dsl-vlan435-66-18-218-36.nucleus.com) |
| 22:26.26 | grandpapadot | There are some minor bugs, for example, you can't initiate a g729a conference call with 8.2 but you can put the first caller on hold and initiate a second g729a call. This bug doesn't exist when using ulaw. |
| 22:26.36 | rhombus | what's the purpose of format_mp3.so? |
| 22:26.40 | Legend | these phones have been sitting for a while, and are at sip 7.1, just wondering if i am missing anything |
| 22:26.41 | grandpapadot | Legend: 8.6 or 8.8 is your best bet, then. |
| 22:27.01 | [TK]D-Fender | jm|home, nope. |
| 22:27.25 | jm|home | thanks dude :) |
| 22:27.32 | Legend | grandpapadot: mind sharing the magic search string to find those releases on the site? i do have a valid CCO login |
| 22:28.07 | grandpapadot | I run 8.2, doesn't require CCO. It's under the Unified Phone something-or-another, though |
| 22:28.28 | Legend | ok, ill keep digging |
| 22:28.40 | Legend | when i had these phones deployed tow years ago, you needed a CCO login for firmware |
| 22:28.42 | seanbright | rhombus: to save VMs/recordings in MP3 format? |
| 22:28.48 | [TK]D-Fender | mintee, that gets set after you call Dial. I can't see it being any use as priority 1 like that |
| 22:28.51 | grandpapadot | You do for everything but 8.2 |
| 22:28.58 | Legend | grandpapadot: ah |
| 22:29.19 | rhombus | seanbright: well, it's part of addons. i'm asking because I've got an asterisk installation that seems to die the moment it tries to load the module |
| 22:29.43 | rhombus | but I've also got mp3 MOH, so just disabling it is not an option if it means that MOH won't work |
| 22:29.46 | seanbright | rhombus: using compatible versions of asterisk and asterisk-addons |
| 22:29.48 | seanbright | ? |
| 22:30.10 | rhombus | seanbright: well, they're both 1.2.x |
| 22:30.21 | *** join/#asterisk tristanbob (n=tristanr@oalug/member/tristanbob) |
| 22:30.26 | rhombus | Asterisk is 1.2.22 and addons is...well, I don't actually know |
| 22:30.33 | [TK]D-Fender | rhombus, the point of format_mp3.so is to allow * to playback MP3's |
| 22:30.38 | grandpapadot | The same g729a conference call bug exists in 8.8 I just found out from one of the other engineers here. |
| 22:31.08 | Legend | grandpapadot: ok, well im all ulaw on a lan |
| 22:31.15 | rhombus | [TK]D-Fender: if it's absent, Asterisk won't play mp3s, then? |
| 22:31.17 | grandpapadot | Yep, no worries for U! |
| 22:31.33 | [TK]D-Fender | rhombus, obvious reversal of the definition I just gave you, yes.... |
| 22:32.10 | rhombus | Well, not really obvious -- there are different ways of playing mp3s in Asterisk. |
| 22:32.12 | seanbright | rhombus: how are you playing MOH with MP3s when the format_mp3.so module is killing asterisk? |
| 22:32.38 | [TK]D-Fender | rhombus, when I say it allws you play MP3, not having it would clearly NOT let you do that. |
| 22:33.07 | [TK]D-Fender | rhombus, Because the "other ways" are not having * play them back, but rather some other external process. |
| 22:34.05 | rhombus | seanbright: I have a live vanilla asterisk system and I'm trying to install FreePBX in place |
| 22:34.15 | rhombus | seanbright: with the vanilla configs, it works fine |
| 22:34.23 | seanbright | rhombus: using mpg123? |
| 22:35.06 | rhombus | seanbright: that I don't know -- what i know is that it's the one that doesn't restart the MOH file every time someone is put on hold |
| 22:35.19 | seanbright | rhombus: right. you don't need format_mp3.so |
| 22:35.25 | seanbright | as [TK]D-Fender said. |
| 22:35.29 | grandpapadot | sox somefile.mp3 somefile.wav, done |
| 22:35.41 | Legend | grandpapadot: upgraded, thanks - still no damn softkeys though :-\ |
| 22:35.49 | *** part/#asterisk beek (n=klinebl@65.211.106.243) |
| 22:36.09 | grandpapadot | Eh? Mine has soft-keys. Do you have a mal-formed config? |
| 22:36.19 | rhombus | seanbright: so removing it from modules.conf is not going to impact MOH in my case, then, correct? |
| 22:36.34 | Legend | grandpapadot: i meant user assignable, like speed dials and features and stuff, not the canned "redial, newcall, and call forward" |
| 22:36.38 | seanbright | rhombus: correct. |
| 22:36.47 | grandpapadot | Ah.. Never tried to add any myself.. |
| 22:37.14 | rhombus | seanbright: what's puzzling is that the modules.conf for both the vanilla asterisk and the FreePBX asterisk call format_mp3.so |
| 22:38.55 | seanbright | rhombus: by "vanilla asterisk" you mean an asterisk release from asterisk.org? |
| 22:39.20 | rhombus | seanbright: yeah, without the funky FreePBX configuration files :) |
| 22:39.26 | seanbright | rhombus: because i'm looking at the sample config for 1.2 and don't see format_mp3.so mentioned |
| 22:39.35 | *** join/#asterisk juanant (n=chatzill@190.156.245.114) |
| 22:39.59 | rhombus | seanbright: it's part of addons... this is why I was uncertain about whether it was even necessary |
| 22:40.05 | juanant | hi all i am from colombia |
| 22:40.07 | seanbright | rhombus: long story short: its not. |
| 22:40.23 | juanant | i need some help |
| 22:40.30 | *** part/#asterisk SteveTotaro (n=root@209.213.170.178) |
| 22:40.38 | juanant | can anybody help my? |
| 22:40.44 | seanbright | juanant: just ask your question |
| 22:40.44 | rhombus | seanbright: okay -- I'm getting it now -- if I've got mpg123 in the process list, I'm not using format_mp3.so. |
| 22:40.52 | seanbright | rhombus: correct |
| 22:41.02 | rhombus | seanbright: thank you for your help |
| 22:41.04 | seanbright | rhombus: mpg123 is doing the heavy lifting, asterisk just pipes the audio |
| 22:41.07 | seanbright | rhombus: np |
| 22:41.26 | juanant | sean |
| 22:41.32 | juanant | seanbright |
| 22:41.38 | seanbright | juan |
| 22:41.40 | seanbright | juanant |
| 22:41.44 | juanant | Tanks |
| 22:42.08 | juanant | look i installed fedora and updated asterisk to version 1.4.17-1 |
| 22:42.42 | juanant | i download sounds and place it on /var/lib/asterisk |
| 22:43.00 | juanant | as root change the owner and group to asterisk |
| 22:43.34 | juanant | but in the log file it still saying: coulnd find file |
| 22:43.49 | juanant | is this a bug??? |
| 22:44.13 | seanbright | juanant: pastebin the part of your extensions.conf file where you are trying to play the sound, and the CLI output |
| 22:44.15 | seanbright | ~pb |
| 22:44.15 | jbot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste |
| 22:44.48 | juanant | i can playback sounds with the full path but not with the name |
| 22:45.02 | seanbright | juanant: pastebin the part of your extensions.conf file where you are trying to play the sound, and the CLI output |
| 22:45.08 | defsdoor | can I create a test call file that calls me at home and play moh or something until I hang up ? trying to test something on the line |
| 22:46.01 | Legend | is there a preferred iax did provider? it used to be nufone and voicepulse, but i guess the market has grown? |
| 22:46.20 | juanant | many thanks |
| 22:46.28 | seanbright | juanant: good talk. |
| 22:46.38 | juanant | jajajajaja |
| 22:47.00 | seanbright | defsdoor: yes, there is a file called callfiles.txt in the doc/ directory of the asterisk tarball |
| 22:47.57 | seanbright | juanant: did you pastebin your conf file and CLI output as i asked? |
| 22:48.06 | juanant | i am in that.... |
| 22:48.14 | juanant | it is in the server.... |
| 22:48.57 | seanbright | what is in the server? |
| 22:48.57 | defsdoor | seanbright: I've got the gist of the call file syntax etc.. - just stuck on getting it to talk to me till I hang up :) |
| 22:48.57 | seanbright | the conf file? the CLI? yes, they are in the server. |
| 22:48.57 | mintee | splitsy |
| 22:50.20 | seanbright | defsdoor: ah, well you just create an extension that does a WaitExten(100000) or something and then have the call file reference that |
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| 22:53.32 | juanant | HI sean i pasted it |
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| 22:55.35 | jm|laptop | ohai |
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| 22:55.40 | outtolunc | titanic! |
| 22:56.24 | juanant | sean are you there???? |
| 22:56.24 | juanant | seanbright |
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| 22:56.26 | juanant | :) |
| 22:57.39 | jameswf | boing |
| 22:57.49 | juanant | ping seanbright |
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| 22:58.17 | juanant | there is another person who can helpmy????? |
| 22:59.02 | juanant | http://pastebin.com/m33d75f58 |
| 23:00.06 | juanant | i cant talk very well but a good undertander few words need |
| 23:00.11 | mintee | File demo-congrats does not exist |
| 23:00.33 | jameswf | use tt-monkeys |
| 23:00.44 | mintee | or all-you-base |
| 23:00.51 | mintee | all-your-base |
| 23:01.04 | jameswf | all your base loses something when she says it |
| 23:01.23 | mintee | not if you wait(2) |
| 23:02.07 | outtolunc | you also need to reload your extensions |
| 23:02.07 | mintee | i need to be able to dial * when I'm in the greeting of VoiceMail() so that it sends me to VoiceMailMain |
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| 23:02.25 | mintee | i've just looked everywhere and I can't find a way to do it |
| 23:02.33 | outtolunc | as you have demo-congrats and menu commented out in dialplan, but it is calling them in the cli output |
| 23:03.24 | juanant | i will come tomoroow |
| 23:03.31 | juanant | tanks a lot bye |
| 23:03.36 | mintee | lol |
| 23:03.59 | outtolunc | no tanks needed <G> |
| 23:04.07 | mintee | anyway... yeah, dial * to reach voicemailmain() anyone? |
| 23:04.47 | outtolunc | features? |
| 23:05.13 | outtolunc | its either set it there, or disable it so you can do it in dialplan |
| 23:05.48 | mintee | hum... it's not in my sample... |
| 23:05.59 | draygon | I have this little IAX device that converts my analog to VOIP |
| 23:06.01 | ManxPower | mintee: Yes, it's trivial. |
| 23:06.07 | draygon | Anyone know what software I would use to configure it? |
| 23:06.10 | ManxPower | "show application voicemail" |
| 23:06.36 | ManxPower | draygon: Do you mean an IAXy? |
| 23:08.18 | draygon | hm |
| 23:08.24 | draygon | Not sure whats its called |
| 23:08.33 | draygon | Its a small device, I plug in my cat 5 |
| 23:08.37 | draygon | and plug in my normal phone line |
| 23:09.14 | ManxPower | I will assume it's an IAXy. There should be config info on Digium's web site or voip-info.org |
| 23:09.24 | draygon | Hang on |
| 23:09.26 | draygon | I will tell you |
| 23:09.28 | mintee | ah, nice... i get it now. It was trivial. |
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| 23:14.00 | draygon | Ah ManxPower |
| 23:14.02 | draygon | You might be right |
| 23:14.08 | draygon | All it says is digium on it |
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| 23:15.05 | draygon | How do I configure it? |
| 23:15.11 | draygon | I can't find any info anywhere |
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| 23:19.38 | [TK]D-Fender | draygon, the IAXY is documented in the book, and on the WIKI |
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| 23:23.31 | draygon | Do you have a link? |
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| 23:30.43 | [hC] | so whats everyones big beef with snom phones? I just got my hands on a 320, but it seems to be designed fairly nicely |
| 23:30.54 | [hC] | better imho than the aastra 5xi series! (physically, i mean) |
| 23:31.15 | [hC] | draygon: www.voip-info.org will get you a search box to find out the info on the iaxy |
| 23:33.08 | draygon | OK I dont understand this |
| 23:33.18 | draygon | Do I configure the device |
| 23:33.21 | draygon | or the iax.conf? |
| 23:33.29 | *** part/#asterisk lirakis (i=lirakis@66.252.24.133) |
| 23:33.37 | [hC] | er.. probably both. ive never used one, your guess is as good as mine. |
| 23:36.00 | draygon | yeah i think the device may need to be configured |
| 23:37.28 | andresmujica | draygon: you can use a windows tool for configuring it it's on voip-info |
| 23:38.02 | draygon | hrm, I can't seem to find it anywhere |
| 23:38.08 | andresmujica | or from linux you can use the iaxyprov |
| 23:38.36 | andresmujica | let me check where is it... |
| 23:38.53 | draygon | windows would be better for me |
| 23:38.58 | andresmujica | http://asterisk.gnuinter.net/files/digium/iaxyprov/ |
| 23:38.59 | draygon | you rock andresmujica ;) |
| 23:39.03 | draygon | thanks allot. |
| 23:39.18 | draygon | welp, i need the one for windows |
| 23:39.35 | andresmujica | gimme a sec.. |
| 23:41.12 | andresmujica | http://dacosta.dynip.com/asterisk/ |
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| 23:41.50 | draygon | okay |
| 23:42.07 | draygon | now do you know if i need to configure the device or iax.conf? |
| 23:42.26 | andresmujica | the device. you need an iax extension thou |
| 23:42.52 | draygon | okay cool |
| 23:42.58 | draygon | thanks a bunch |
| 23:43.01 | draygon | I have it configured |
| 23:43.06 | draygon | I just hope i dont mess anything up |
| 23:43.47 | andresmujica | probably not |
| 23:44.17 | draygon | i just pm'd you real quick |
| 23:44.22 | draygon | quick question |
| 23:45.28 | [TK]D-Fender | ~book |
| 23:45.29 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com, or see ~buybook |
| 23:45.30 | [TK]D-Fender | ~wikis |
| 23:45.31 | jbot | [~wikis] VoIP Wiki covering Asterisk, FreeSWITCH, TrixBox, SER, OpenSER, sipX, CallWeaver, and YATE. http://www.voip-info.org (c) Arte Marketing Inc / CommPartners |
| 23:45.36 | [TK]D-Fender | draygon, ^^^^^^^^ |
| 23:46.16 | *** join/#asterisk AndyGraybeal_ (n=andy@node138.35.251.72.1dial.com) |
| 23:50.22 | *** part/#asterisk kraptv (n=ryan@magic.skylab.org) |
| 23:51.06 | andresmujica | anyone here has ever configured an asterisk box with a SAFARI C3 CEDAR SIP extension ??? |
| 23:58.08 | *** join/#asterisk BBHoss (n=hoss@c-71-207-173-38.hsd1.al.comcast.net) |