| 00:00.16 | Qwell | BBHoss: what, like AEL or Lua? |
| 00:00.29 | *** join/#asterisk root52 (n=root52@ip70-191-120-39.cl.ri.cox.net) |
| 00:00.40 | BBHoss | Qwell: i was thinking more on the lines of ruby or python |
| 00:00.45 | [TK]D-Fender | jaytee: Or just looked at the grouping on asterisk.org where clearly there is no zaptel in that pile. |
| 00:00.50 | Qwell | That's why we have AGI. |
| 00:00.56 | Qwell | Feel free to use whatever suits you best. |
| 00:01.10 | [TK]D-Fender | jaytee: or any of the large announcements, airplane banners or 30-minute infomercials |
| 00:01.28 | *** join/#asterisk CrazyTux (n=brandon@user-vcaur3m.dsl.mindspring.com) |
| 00:01.47 | BBHoss | Qwell: I do, I just think it would be valuable to have a built in ruby or python parser, so we don't have to drop into agi every time we want to do something semi-complex |
| 00:01.48 | jaytee | [TK]D-Fender, it's like blaming the French for WWII in Europe and letting the Germans off the hook |
| 00:02.03 | harry_v | TK, again as I have stated. All configs were correct. Just the module chan_zap.so was missing in the lib directly. |
| 00:02.16 | kerx | how do you set CallerID w/ Dial() ? |
| 00:02.26 | Qwell | BBHoss: so, what, you want it inline with extensions.conf? |
| 00:02.36 | [TK]D-Fender | harry_v: well 1.6 doesn't use chan_zap |
| 00:02.51 | harry_v | I did state it was 1.6 |
| 00:02.52 | harry_v | :) |
| 00:03.03 | Qwell | BBHoss: You're going to have to "drop into" something. |
| 00:03.08 | BBHoss | Qwell: not inline with it, like a whole different language, so where i could build classes and all sorts of cool stuff |
| 00:03.14 | Qwell | like AGI. |
| 00:03.16 | [TK]D-Fender | harry_v: harry_v And you installed it without reading any of the big print & import docs. Kudos! |
| 00:03.38 | BBHoss | Qwell: sure, but something just feels dirty about agi |
| 00:03.46 | jaytee | kerx, you'd use Set(CALLERID(num)=$"somevariable") in the priority right before the one that uses the Dial app |
| 00:03.52 | Qwell | How else would you plan on doing it? |
| 00:03.57 | kerx | why somevariable? |
| 00:04.02 | [TK]D-Fender | jaytee: yeah... uh huh.. that'd work ;) |
| 00:04.05 | kerx | jaytee, I want to force the callerid to work |
| 00:04.17 | kerx | [TK]D-Fender, you are master boss big-dick |
| 00:04.25 | kerx | how are you today [TK]D-Fender ? |
| 00:04.28 | harry_v | for the most part i do read the README |
| 00:04.28 | kerx | Big-Dick Asterisk Master? |
| 00:04.38 | [TK]D-Fender | BDAM! |
| 00:04.51 | kerx | hehe |
| 00:04.57 | kerx | DAMBINO! |
| 00:05.01 | kerx | joo know |
| 00:05.03 | kerx | joo know |
| 00:05.37 | ratmandu | god, thats a horrible ISP |
| 00:05.46 | jaytee | kerx, somevariable is just my text as a placeholder for the variable name you'd use and assign a callerid number that you want to use. but forcing it? forcing it where? |
| 00:06.11 | kerx | Set(CALLERID(num)="18185551212") |
| 00:06.18 | kerx | i'll give that a shot |
| 00:06.21 | [TK]D-Fender | kerno quotes <- |
| 00:06.27 | [TK]D-Fender | kerx: no quotes <- |
| 00:06.35 | kerx | oops :P |
| 00:06.36 | *** join/#asterisk andresmujica (n=andresmu@190.25.104.186) |
| 00:06.38 | jaytee | kerx, yeah, no quotes |
| 00:06.52 | kerx | btw, the realtime dialplan really affects asterisk's speed |
| 00:06.56 | jaytee | or you could use a variable you've set elsewhere if you wanted |
| 00:07.00 | [TK]D-Fender | jaytee: guess you miseed why I wrote that before... silly broken variable reference! |
| 00:07.03 | kerx | i've had to convert from realtime extensions over to plain old textfiles again |
| 00:07.07 | [TK]D-Fender | jaytee: take another mark off your score! |
| 00:07.21 | jaytee | [TK]D-Fender, I'm slow tonight, it was a long day |
| 00:07.30 | BBHoss | Qwell: well instead of having contexts, i would have a class for users in sip.conf for example, and write some sort of case statement or something for the calls |
| 00:07.36 | jaytee | arthritic fingers don't lend themselves to fast typing |
| 00:08.39 | Qwell | BBHoss: Patches welcome. |
| 00:08.52 | BBHoss | Qwell: yeah it would be interesting wouldn't it |
| 00:08.55 | [TK]D-Fender | jaytee: I burned an extra 2.5 work-days on that ^#%$ing budge this weekend... I should be friend myself, but I'm reenergized by having completed it and starting to clear off my plate and making inroads at migrating from NetWare5 to Samba instead of Win2K3 sometime mid next year |
| 00:09.00 | jaytee | I ran my extensions.conf file through astograph.py and the png file it generated was really interesting. most all of my contexts all point to the context named [DEAD_END] and that points to [RETARD!] |
| 00:09.00 | Qwell | I can't see a use for it, no. |
| 00:09.19 | BBHoss | also maybe do something in erlang to help distribute load |
| 00:09.19 | [TK]D-Fender | s/friend/fried/ |
| 00:09.25 | Qwell | BBHoss: AGI |
| 00:09.56 | jaytee | so you're gonna use Samba on linux instead of Win2K3? Awesome!!!!! |
| 00:09.59 | drmessano^ | Erlang? |
| 00:10.09 | jaytee | gesundheit |
| 00:10.16 | [TK]D-Fender | jaytee: I'm pushing for it... |
| 00:10.24 | drmessano^ | jaytee: I am using NFS on Win2k3 R2 instead of Linux! |
| 00:10.45 | BBHoss | drmessano^: yeah like put the dialplans and configs in mnesia and have the asterisk nodes communicate about load and all of the calls going on |
| 00:10.52 | [TK]D-Fender | jaytee: All out marketing stuff is on it for 3 years now and they run 100% and account for 2/3 of our data storage needs as it is... |
| 00:10.53 | jaytee | drmessano^, don't make me come back there!!! |
| 00:11.15 | Qwell | BBHoss: #exec, dundi, AGI. |
| 00:11.16 | drmessano^ | Well, mentioning erlang in public is like doing a George Michael |
| 00:11.53 | jaytee | seriously, when did the uber-lords of Redmond add NFS support to Win2K3? in R2? I don't remember seeing that. |
| 00:12.04 | BBHoss | dundi in my experience is fragile |
| 00:12.10 | drmessano^ | But in any case, you'll have to wait in line.. Qwell promised me he would commit my COBOL parser to 1.6 |
| 00:12.16 | drmessano^ | jaytee: R2 |
| 00:12.38 | jaytee | it's so hard to tell whether he's being serious or snarky. I usually just go with the latter |
| 00:12.49 | drmessano^ | lol |
| 00:12.59 | drmessano^ | R2 did add NFS |
| 00:12.59 | BBHoss | i remember i upgraded to one of the 1.6 betas and it just flat out didnt work |
| 00:13.05 | drmessano^ | I was a bit shocked too |
| 00:13.15 | drmessano^ | 1.6 seems to work for most of us |
| 00:13.34 | drmessano^ | did you uncomment broken=very_yes? |
| 00:13.54 | drmessano^ | Asterisk likes to feel dirty |
| 00:14.20 | jaytee | drmessano, wow! you are serious! I just found it. I hadn't heard that and didn't see it in the general What's New shit for R2. |
| 00:14.26 | [TK]D-Fender | I wish I could commit my parser to replace the dialplan app parser.... |
| 00:14.50 | drmessano^ | jaytee: They pretty much merged the services for Unix in, and then some |
| 00:15.01 | drmessano^ | jaytee: Its like a Linux box, but you get laid too |
| 00:15.08 | jaytee | [TK]D-Fender, but no one really wants to code in Pascal :-) |
| 00:15.26 | [TK]D-Fender | jaytee: just the stuff INSIDE the () silly! |
| 00:15.41 | jaytee | ah! ok, I down wid dat! |
| 00:15.48 | drmessano^ | I think it would be cool to write an erlang parser in AGI that takes AGI commands as arguments |
| 00:15.52 | [TK]D-Fender | jaytee: that allows for typed variables, mixed data types, clear text separation, etc |
| 00:16.21 | jaytee | [TK]D-Fender, instead of just dumb ascii strings |
| 00:16.31 | [TK]D-Fender | jaytee: yup |
| 00:17.53 | *** part/#asterisk Deeewayne (n=Deeewayn@nat/digium/x-5d265ee7041eddd5) |
| 00:18.02 | jaytee | [TK]D-Fender, it would add just a little extra vertical to the learning curve but it would be worth it. My gut tells me that 1) 50% or a little more would love it and the rest would hate it and 2) I shouldn't have had the ravioli for dinner. |
| 00:18.22 | [TK]D-Fender | jaytee: Mine interprets strings, hec, dec, float, binary, etc. |
| 00:18.43 | jaytee | hec=hex |
| 00:18.43 | [TK]D-Fender | jaytee: I'll go with #2 |
| 00:18.51 | [TK]D-Fender | yes, hex. |
| 00:19.03 | [TK]D-Fender | jaytee: OH, and asci chars. |
| 00:19.06 | drmessano^ | XML is the new ASCII |
| 00:19.18 | drmessano^ | Not sure what the hell that means |
| 00:19.23 | drmessano^ | But YESH! |
| 00:19.28 | *** join/#asterisk [netman] (n=netman@238.Red-88-23-119.staticIP.rima-tde.net) |
| 00:19.51 | jaytee | drmessano^, why stop there? why not go with EBDCDIC? |
| 00:19.52 | drmessano^ | Asterisk should be able to parse XML from myspace if it plans to compete |
| 00:20.08 | jaytee | ooops, meant EBCDIC |
| 00:20.23 | drmessano^ | ICUPAEIOU to you too |
| 00:20.37 | jaytee | damn, I can't type for shit tonight |
| 00:20.58 | [TK]D-Fender | ... |
| 00:21.05 | jaytee | one more time. EBCIDIC, IBM cruft |
| 00:22.24 | echelon | anyone know an ATA that allows you to change its user-agent string or headers? |
| 00:29.05 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 00:31.40 | [TK]D-Fender | zchaos: *ping |
| 00:36.13 | *** join/#asterisk rcy (n=rcy@S01060002553240a8.vc.shawcable.net) |
| 00:42.50 | *** join/#asterisk jeffspeff (i=jeffspef@c-98-240-113-191.hsd1.ky.comcast.net) |
| 00:43.54 | jeffspeff | anybody here familiar with astbill or a similar solution? |
| 00:49.40 | *** join/#asterisk UnixDawg (n=UnixDawg@cpe-98-28-149-231.cinci.res.rr.com) |
| 00:53.12 | *** join/#asterisk anthm (n=anthm@freeswitch/developer/anthm) |
| 00:54.20 | *** join/#asterisk ming_zym (n=ming_zym@nat/yahoo/x-af1b82e077989a16) |
| 01:00.13 | *** join/#asterisk Bilano (n=no@66.54.249.50) |
| 01:04.11 | *** join/#asterisk TrentCreek (n=kvirc@ppp-70-244-17-140.dsl.hrlntx.swbell.net) |
| 01:04.33 | *** join/#asterisk brut- (n=brut@h66-173-4-254.mntimn.dedicated.static.tds.net) |
| 01:11.54 | neurosys | [TK]D-Fender: which distro you use? |
| 01:12.39 | [TK]D-Fender | neurosys: Primarily CentOS, Still have an old love for Slackware though. I use Ubuntu as my home desktop (when I don't boot to WinXP like 99.999% of the time) |
| 01:13.06 | neurosys | [TK]D-Fender: Slack run asterisk nicely? |
| 01:13.22 | [TK]D-Fender | neurosys: 100% Every time |
| 01:13.48 | neurosys | [TK]D-Fender: Thx :) |
| 01:13.50 | [TK]D-Fender | neurosys: Mind you so has every time with CentOS... I just came to like certin bits of it better than slack. |
| 01:14.18 | [TK]D-Fender | neurosys: SysV inits, startup, etc. |
| 01:14.31 | neurosys | [TK]D-Fender: Oh More like debian? |
| 01:15.19 | neurosys | [TK]D-Fender: Will CentOS install without the GUI> |
| 01:15.21 | [TK]D-Fender | neurosys: Debian I'm sure I could be just as happy with, but their approach for stable vs what we consider "normal" is annoying. |
| 01:15.27 | [TK]D-Fender | neurosys: Certainly |
| 01:15.38 | [TK]D-Fender | neurosys: CentOS = RHEL you know... |
| 01:16.09 | neurosys | [TK]D-Fender: Yeah, But the one thing i recall loving slack for was it minimal install :) |
| 01:17.01 | [TK]D-Fender | neurosys: Yes, I miss where 9.0 fit all on 1 CD. |
| 01:17.12 | [TK]D-Fender | neurosys: and that was INCLUDING X |
| 01:18.06 | neurosys | [TK]D-Fender: heh. well... starting my CentOS torrent. |
| 01:18.19 | neurosys | [TK]D-Fender: Gotta try new things once in a while ;) |
| 01:18.55 | [TK]D-Fender | neurosys: What are you coming from? |
| 01:19.13 | *** join/#asterisk echelon (i=Unknown@gateway/tor/x-1d748cfc6eebed6b) |
| 01:19.20 | neurosys | [TK]D-Fender: FBSD |
| 01:19.24 | echelon | anyone here use an ATA device? |
| 01:19.38 | *** join/#asterisk RB2 (n=RB2@pool-71-127-212-121.nwrknj.east.verizon.net) |
| 01:20.02 | [TK]D-Fender | echoI hope this isn't leading to the question you asked 1 hr ago.... |
| 01:20.28 | [TK]D-Fender | echelon: I hope this isn't leading to the question you asked 1 hr ago.... |
| 01:21.07 | echelon | [TK]D-Fender: why? i never got an answer to it |
| 01:22.21 | [TK]D-Fender | echoBecause this new "Anyone use an aATA' as a lead in for that is like "Hey anyone tried ice-cream before? Yes? Good! Now what I really want to know is about advanced particle physics... you know ice-cream is made of particles so you must all be able to answer my questions!" |
| 01:23.15 | [TK]D-Fender | echelon: that is a really bad bat&switch lead in question. |
| 01:23.23 | [TK]D-Fender | echelon: Over 99% of ATA users would never think of changing the user-agent |
| 01:23.28 | [TK]D-Fender | bait* |
| 01:23.45 | echelon | [TK]D-Fender: i just want to ask if they've seen this setting on their UI |
| 01:24.19 | [TK]D-Fender | echelon: Check the manuals for models you think might be good |
| 01:25.19 | tzanger | any tdm experts here? |
| 01:25.20 | *** join/#asterisk jer (n=jer@unaffiliated/jer) |
| 01:25.33 | tzanger | not zaptel specific, just tdm in general |
| 01:25.49 | echelon | thanks for the suggestion :) |
| 01:27.47 | *** join/#asterisk Kumbang (n=unknown@125.163.83.153) |
| 01:28.54 | echelon | umm.. anyone know a good voip store site? most of the ones i've come across have ridiculously priced shipping rates |
| 01:29.19 | echelon | they're meant for companies that buy them in builk |
| 01:29.22 | echelon | *bulk |
| 01:31.51 | jeffspeff | anybody familiar with Freeside billing system? http://www.freeside.biz/freeside/index.html Hoping to get some recommendations on what to use or what to stay away from. |
| 01:31.57 | [TK]D-Fender | echelon: Depends where yo are, now doesn't it? |
| 01:33.06 | echelon | in the east coast |
| 01:33.39 | [TK]D-Fender | echelon: Lots of places have an east coast.... |
| 01:33.56 | jaytee | except for Kansas |
| 01:33.58 | echelon | usa |
| 01:34.04 | [TK]D-Fender | echelon: www.telephonydepot.com <- good for most of North America |
| 01:34.07 | jeffspeff | echelon: you are IN the east coast? thank god for those mobile wifi cards! lol |
| 01:34.31 | [TK]D-Fender | holds his head under until the squirming and bubbles stop |
| 01:34.40 | jaytee | decent rates unless you're a cheap prick that want's to make a couple of my buddies lose their jobs at UPS due to cutbacks |
| 01:36.08 | echelon | [TK]D-Fender: yeah, that's one of those sites.. there isn't a shipping rate below 10 bucks |
| 01:36.41 | [TK]D-Fender | echelon: And what kind of store anywhere ships stuff under 10$? |
| 01:37.06 | echelon | umm.. newegg.. overstock.. chiefvalue |
| 01:37.13 | [TK]D-Fender | echelon: How little are you looking to buy to penny-pinch so much? |
| 01:37.18 | echelon | but their supply of voip products is limited |
| 01:37.33 | echelon | just one ATA -_- |
| 01:38.34 | [TK]D-Fender | echelon: I just entered my head-office in CT for shipping... came up free |
| 01:38.55 | echelon | they should include the weight of the products before they caclulate shipping |
| 01:39.08 | [TK]D-Fender | echelon: the margin is so low on 1 stupid little box where do you think they can afford to make shipping cheap for you from? |
| 01:39.12 | echelon | from telephony depot? ^_- |
| 01:39.17 | [TK]D-Fender | echelon: yes |
| 01:39.26 | echelon | i'm in ny |
| 01:39.42 | [TK]D-Fender | echelon: I added the Sangoma A101d to the cart, entered their state & zip, and came up 0$ |
| 01:40.06 | [TK]D-Fender | echelon: WOW.. first we start with east cost, then get a COUNTRY, and only now get a state. |
| 01:40.13 | [TK]D-Fender | echelon: Paradoid much? |
| 01:40.24 | [TK]D-Fender | echelon: Not that your use of TOR isn't a complete give-away |
| 01:40.29 | [TK]D-Fender | paranoid* |
| 01:40.34 | echelon | that's why i'm on tor :D |
| 01:40.44 | echelon | lol |
| 01:40.52 | [TK]D-Fender | echelon: Great, so paranoid AND cheap. Good basis. |
| 01:41.23 | *** join/#asterisk shinao1 (n=shinao1@62.173.48.42) |
| 01:41.48 | echelon | yeah.. same rate |
| 01:42.02 | echelon | nothing below $10 |
| 01:42.17 | [TK]D-Fender | echelon: I just changed the order to 1 ATA and now its $7.22 |
| 01:42.32 | [TK]D-Fender | echelon: Which ATA did you pick? |
| 01:42.54 | echelon | grandstream ht 286 |
| 01:42.59 | [TK]D-Fender | echelon: GREAT! |
| 01:43.01 | [TK]D-Fender | ~gs |
| 01:43.02 | jbot | GrandSuck phones & gateways are cheap junk which should be avoided with extreme prejudice. |
| 01:43.17 | [TK]D-Fender | echelon: You also had to pick the worst POS product they had |
| 01:43.45 | echelon | lol, what's wrong with it? |
| 01:43.51 | jaytee | ROFL, serves him right!!!! |
| 01:44.06 | jaytee | enjoy the pain! |
| 01:44.16 | [TK]D-Fender | jaytee: Yup, I'm just backing off of this case... total losing scenario. |
| 01:44.17 | echelon | what's wrong with it?? |
| 01:44.25 | echelon | everyone keeps telling me they suck |
| 01:44.31 | jaytee | ~gs |
| 01:44.32 | jbot | GrandSuck phones & gateways are cheap junk which should be avoided with extreme prejudice. |
| 01:44.33 | [TK]D-Fender | echelon: and you're not listening? |
| 01:44.41 | jaytee | ~grandstream |
| 01:44.41 | jbot | well, grandstream is the Yugo of VoIP hardware. Run. Run away now. |
| 01:45.10 | [TK]D-Fender | goes off to do something productive... |
| 01:45.37 | echelon | linksys are overpriced |
| 01:45.38 | jaytee | echelon, I highly recommend Grandstream to anyone that likes lots of jitter and silence suppression that you can't turn off without a firmware update. |
| 01:45.59 | jaytee | Linksys are not overpriced. |
| 01:46.16 | [TK]D-Fender | echelon: Says who? |
| 01:46.37 | jaytee | but go ahead and order a bunch of Grandstream equipment, just don't come in here whining about it afterwards |
| 01:46.47 | [TK]D-Fender | echelon: You mean more expensive than CRAP? I hope so... means they spend mony hopefully making it NOT crap. |
| 01:47.00 | echelon | the cheapest linksys product i could find is $12.. and it's a power adapter |
| 01:47.22 | [TK]D-Fender | ~cheap |
| 01:47.23 | jbot | rumour has it, cheap is a bad idea. If you're setting up an Asterisk system, one of the wisest pieces of advice you can take is this: DON'T BE A CHEAPSKATE. |
| 01:47.26 | [TK]D-Fender | ^^^^^^^^^^^^^^^^^^ |
| 01:47.32 | jaytee | I have two tin cans with string that have better audio quality than 2 GSX2000's on a 100MB lan with no other traffic. |
| 01:48.19 | *** join/#asterisk UnluckyAlf (i=me@5.dollar.sucky.sucky.at.shellium.org) |
| 01:48.34 | UnluckyAlf | Hi I was wondering if someone could help with a dialplan please |
| 01:48.37 | UnluckyAlf | What is the asterisk dial plan for calling Gizmo's conference rooms? Such as 1222 and 12221234567 (I know how to add them individually, but is there a dialplan that will take them all no matter how many digits there are after the 1222? |
| 01:48.39 | jaytee | echelon, the PAP2 is 44.95 or so at telephonydepot. that's maybe 10 to 14 bucks more than a HT286 and gives you 2 FXS ports instead of 1 |
| 01:48.47 | [TK]D-Fender | UnluckyAlf: Ask a specific question, get a specific answer... |
| 01:48.51 | *** join/#asterisk interfaithquest (n=chatzill@ip67-88-184-130.z184-88-67.customer.algx.net) |
| 01:49.11 | [TK]D-Fender | UnluckyAlf: "_1222!" |
| 01:49.19 | UnluckyAlf | Thank you [TK]D-Fender |
| 01:49.35 | [TK]D-Fender | UnluckyAlf: 1222 + none or any number of extra chars |
| 01:49.47 | interfaithquest | sip.conf needs in global nat=yes, for sip devices on the local lan , strange ? |
| 01:49.54 | UnluckyAlf | and the same will work for 1747 I presume? |
| 01:50.00 | kerx | WTF is OutgoingSpoolFailed? |
| 01:50.00 | echelon | jaytee: does it let you change the user-agent string? |
| 01:50.03 | echelon | or header? |
| 01:50.07 | [TK]D-Fender | interfaithquest: No, it doesn't |
| 01:50.24 | [TK]D-Fender | echelon: Go read its manual |
| 01:51.00 | jaytee | echelon, don't know about that. On Asteriskt the user agent string is a general setting, not a channel device setting |
| 01:51.01 | echelon | UnluckyAlf: http://support.gizmoproject.com/index.php?_m=knowledgebase&_a=viewarticle&kbarticleid=190 |
| 01:51.02 | interfaithquest | well it is bizarre, as when nat=no, then audio is just half duplex for some sip phones, odd but evident |
| 01:52.13 | [TK]D-Fender | interfaithquest: same subnet? |
| 01:52.22 | echelon | i don't think i'll be able to run asterisk.. i only have 24mb ram.. 152mhz cpu |
| 01:52.37 | jaytee | wtf? |
| 01:52.49 | [TK]D-Fender | SMRT |
| 01:52.57 | TrentCreek | sure you can..maybe only one user |
| 01:53.13 | echelon | yeah, it'll be for home use |
| 01:53.32 | [TK]D-Fender | echelon: and what do you need to set the UA for? |
| 01:53.35 | jaytee | you can find faster equipment with more RAM at a yard sale in most places on the east coast for probably the price of a HandyTone 286 |
| 01:53.48 | UnluckyAlf | I'm doing it through the GUI echelon, I don't understand all the editing of files yet, my aim is to get it set up and then look at the conf files |
| 01:54.41 | echelon | [TK]D-Fender: umm.. to break TOS? |
| 01:54.58 | [TK]D-Fender | echelon: With whom? |
| 01:55.05 | echelon | UnluckyAlf: oops.. http://www.voip-info.org/tiki-index.php?page=Asterisk+settings+Gizmo |
| 01:55.10 | echelon | [TK]D-Fender: MagicJack |
| 01:56.04 | [TK]D-Fender | Yup... a COMPLETE WRITEOFF |
| 01:56.08 | [TK]D-Fender | echelon: Well... |
| 01:56.10 | [TK]D-Fender | ~wglwat |
| 01:56.11 | jbot | extra, extra, read all about it, wglwat is well, good luck with all that |
| 01:56.23 | interfaithquest | are some providers using asterisk to handle 1000's of subscribers ? via one machine ? |
| 01:56.23 | jaytee | what a friggen joke! |
| 01:56.45 | [TK]D-Fender | echelon: A cheap-skate to the end. You've branded yourself in the worst possible way with this one... |
| 01:56.48 | echelon | interfaithquest: no, they're using freeswitch |
| 01:56.51 | interfaithquest | do they replace some .conf files with a database |
| 01:56.53 | interfaithquest | oh |
| 01:57.02 | interfaithquest | freeswitch can do iax ? |
| 01:57.45 | echelon | [TK]D-Fender: i was able to retrieve my SIP password |
| 01:57.49 | jaytee | [TK]D-Fender, do you know of any well known businesses running Asterisk on a large scale? say over 500 users on a network? not necessarily on the same server. |
| 01:58.03 | [TK]D-Fender | interfaithquest: the vast majority of the VoIP world does not give a rat's ass about IAX. You are hooked on dreamland protocols..... Faith-based VoIP = FAILING VOIP |
| 01:58.15 | *** join/#asterisk Kumbang (n=unknown@125.163.83.153) |
| 01:58.20 | interfaithquest | junction networks ? |
| 01:58.29 | echelon | UnluckyAlf: you've heard of gizmo's backdoor dialing feature? |
| 01:58.52 | interfaithquest | so the big boys like xo , broadvox.. are only doing sip /mgcp ? |
| 01:58.53 | UnluckyAlf | yeah, I have the rules set up to take advantage of that with the 0101 :) |
| 01:59.29 | interfaithquest | then asterisk is good for proxy to sip upstream |
| 01:59.48 | echelon | lol, i'm setting up the dialplan right now with my softphone :D |
| 02:00.02 | UnluckyAlf | :) |
| 02:00.25 | UnluckyAlf | I have been able to set up outgoing calls via Gizmo, now I just need to get the incoming ones to Gizmo and my asterisk extension working |
| 02:00.34 | interfaithquest | is there a website for voip carriers to deal cards ? |
| 02:01.15 | UnluckyAlf | Can you guys tell me, when I make a call from my extension to say 17471234567 does the call go through the asterisk server (using the servers bandwidth), or does it simply hand off directly? |
| 02:01.41 | UnluckyAlf | It's a hosted asterisk server, not a local one |
| 02:02.06 | echelon | ok |
| 02:02.19 | [TK]D-Fender | UnluckyAlf: Only if both ends can survive a reinvite which is almost never the case. |
| 02:02.44 | UnluckyAlf | So it's the servers bandwidth then right? |
| 02:03.01 | [TK]D-Fender | UnluckyAlf: both ways |
| 02:03.02 | interfaithquest | asterisk can handle 100's of calls at one time ->http://blogs.zdnet.com/ip-telephony/?p=1229 |
| 02:03.15 | interfaithquest | asterisk can serve as a softswitch |
| 02:03.18 | UnluckyAlf | I'll keep an eye on that, I've only got 200GBS/mo to play with, lol |
| 02:03.32 | interfaithquest | the mini carrier is on the verge of undoing the big boys |
| 02:04.39 | [TK]D-Fender | interfaithquest: Grey on the terminology, but much is possible |
| 02:04.46 | echelon | interfaithquest: it's propaganda |
| 02:05.13 | interfaithquest | so freeswitch is the best open source road to a softswitch ? |
| 02:05.30 | echelon | i would say so |
| 02:05.31 | interfaithquest | and freeswitch is a derivative of asterisk ? |
| 02:05.49 | interfaithquest | just a better scaleable architecture ? |
| 02:05.49 | echelon | no, it's based on its own code |
| 02:06.02 | echelon | yeah, cleaner |
| 02:06.04 | interfaithquest | that's quite a trick |
| 02:06.23 | interfaithquest | and 100's of providers are using it now ? |
| 02:06.29 | interfaithquest | nice |
| 02:06.34 | *** join/#asterisk RobertLaptop (n=rmiddle@96.244.48.200) |
| 02:06.43 | echelon | they should |
| 02:07.22 | Micc | Is there anything that asterisk does better than freeswitch? |
| 02:07.23 | interfaithquest | as i see it iax is the easy way at this time to get out from the nat at home.. then proxy to sip to the big provider like broadvox |
| 02:07.24 | [TK]D-Fender | interfaithquest: I'd go qualify that if I were you. |
| 02:07.40 | [TK]D-Fender | interfaithquest: Most don't use * as a front-end anyways, more as a back-end app server, etc/. |
| 02:07.54 | [TK]D-Fender | interfaithquest: Almost any serious setup has SER or the like in front. |
| 02:08.31 | interfaithquest | yes i have gone with SER.. before.. a sip call setup proxy |
| 02:08.37 | [TK]D-Fender | Micc: If you want to drink the cool-aid, go try it yourself, and you tell US. |
| 02:09.07 | RB2 | [TK]D-Fender, fyi, this is the attendant feature I was going on about yesterday: http://bugs.digium.com/view.php?id=10354 |
| 02:09.14 | Micc | TKD-Fender, I'm not sure I have the time to learn another system. |
| 02:09.39 | jaytee | they had orange KoolAid at Digium. It was yummy. I had seconds :-) |
| 02:11.13 | interfaithquest | so is there a chat group for freeswitch ? |
| 02:11.48 | jaytee | try /join #freeswitch |
| 02:11.57 | interfaithquest | nice |
| 02:12.10 | [TK]D-Fender | RB2: in terms of seeing RING, etc? because the individual subscribe basis works... |
| 02:12.34 | [TK]D-Fender | RB2: This just looks like an alternative. |
| 02:13.23 | [TK]D-Fender | RB2: an ok here it is im trying to get the loose ends iorned out please note this will require TCP support (0004903) so 1.6 only, and the kind of patch that'll only make it to 1.6.1 or higher |
| 02:13.54 | [TK]D-Fender | RB2: I'm sure its quite doable though... but certainly more hallse |
| 02:13.58 | [TK]D-Fender | hassle* |
| 02:14.56 | RB2 | [TK]D-Fender, I saw that. So, I'm not pursuing it any farther. I just can't get the poly w/ the bw on it to give me any other indication but in-use or not-in-use |
| 02:15.29 | UnluckyAlf | When calling my SiPPhone number from SiPBroker.com it rings and hangs up immediately, where should I be looking for a solution to this? |
| 02:15.36 | [TK]D-Fender | RB2: Yes, that as I've said is all its ever reported... now others like Aastra have full info available and indicate ringing, etc |
| 02:15.57 | [TK]D-Fender | UnluckyAlf: SIP DEBUG at * CLI |
| 02:16.34 | [TK]D-Fender | UnluckyAlf: and if you are behind NAT theres a lot more to do... |
| 02:17.01 | UnluckyAlf | Ah yes I am behind a NAT |
| 02:17.28 | UnluckyAlf | I could negate that with port forwarding right? |
| 02:17.51 | [TK]D-Fender | ~sipnat |
| 02:17.51 | jbot | [~sipnat] Quick guide on configuring * + SIP behind NAT : http://www.aocomputing.net/?p=3 , otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
| 02:17.53 | [TK]D-Fender | ^^^ |
| 02:17.58 | [TK]D-Fender | UnluckyAlf: MANY more settings to do |
| 02:18.33 | RB2 | [TK]D-Fender, using the RFC4662, the poly supports full information. The Poly 650 will actually do it via UDP, but it's the only model. |
| 02:18.34 | UnluckyAlf | Ouch! lol |
| 02:18.52 | [TK]D-Fender | RB2: retarded... |
| 02:21.40 | UnluckyAlf | Ah, it's just the client that's behind a NAT, as far as I know, the server isn't |
| 02:21.40 | *** join/#asterisk dynaguy (n=gao@d154-20-51-140.bchsia.telus.net) |
| 02:22.18 | *** join/#asterisk Defraz (n=T0tal@24-117-156-21.cpe.cableone.net) |
| 02:23.07 | *** join/#asterisk jtodd (n=jtodd@blob.fox-den.com) |
| 02:23.26 | *** join/#asterisk mindCrime (n=chatzill@cpe-075-177-141-190.nc.res.rr.com) |
| 02:24.38 | [TK]D-Fender | UnluckyAlf: then you still have other settings to do... |
| 02:24.46 | [TK]D-Fender | UnluckyAlf: also in that doc. |
| 02:25.07 | [TK]D-Fender | UnluckyAlf: And pretty much forget abour reinvites |
| 02:25.49 | UnluckyAlf | Yeah, I just tried allowing all traffic from the server to the Grandstream and it wouldn't have it |
| 02:25.57 | UnluckyAlf | It won't even accept incoming now |
| 02:27.29 | UnluckyAlf | Right where do I begin because that page just lists the different types of nat, I have tried a few of the links but all but one are dead now |
| 02:28.28 | [TK]D-Fender | UnluckyAlf: the FIRST LINK |
| 02:29.19 | UnluckyAlf | SipExpress router? |
| 02:30.05 | [TK]D-Fender | ~sipnat |
| 02:30.06 | jbot | [~sipnat] Quick guide on configuring * + SIP behind NAT : http://www.aocomputing.net/?p=3 , otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
| 02:30.39 | UnluckyAlf | Oooooooh, that first link, I thought you meant the first link on the wiki, my apologies |
| 02:30.55 | [TK]D-Fender | I'm going to go have than aneurysm now. I've earned it... |
| 02:32.15 | UnluckyAlf | lol |
| 02:32.17 | UnluckyAlf | Sorry mate |
| 02:33.04 | trnzmeta | someone's been watching australia too much |
| 02:35.55 | *** join/#asterisk JonBach (n=chatzill@70.102.12.123) |
| 02:36.54 | [TK]D-Fender | convulses |
| 02:37.18 | jaytee | lotta helmet-headed mouth breathers around tonight |
| 02:37.42 | JonBach | So I'm in a bind -- our IT guy who manages voip is out of town, and it seems email voicemail is not delivering. Someone up for a quick consulting gig? |
| 02:39.43 | *** join/#asterisk andresmujica (n=andresmu@190.25.104.186) |
| 02:42.27 | *** join/#asterisk jameswf (n=james@ip68-3-60-66.ph.ph.cox.net) |
| 02:43.14 | *** join/#asterisk Titanous (n=titanous@unaffiliated/titanous) |
| 02:43.30 | jameswf | I just saw a promo that said "brought to you my windows, life without walls" If you have no walls wtf do you need windows for |
| 02:44.09 | RB2 | lol |
| 02:47.26 | Titanous | I've got an Aastra phone on a remote IP, when I call it from my grandstream on the local network (where the asterisk box is) I get full two way audio, when the aastra phone tries to call out, I see it working in the console, but no audion in either direction |
| 02:47.31 | Titanous | suggestions? |
| 02:48.21 | *** join/#asterisk ghento (n=ghento@141.117.160.235) |
| 02:49.57 | UnluckyAlf | Oh it receives calls, I have call screening switched on somewhere which is blocking calls from sipbroker |
| 02:52.31 | JonBach | No takers on a paid gig? I can also just pose my question: email delivery of voicemails was working fine until Friday. I need to know how to check the outgoing queue, to verify that they're sitting there, just not being delivered. |
| 02:52.34 | jameswf | disgusting http://www.microsoft-watch.com/walls2.jpg |
| 02:52.38 | UnluckyAlf | Okay so I can receive calls from trunks :) Can anybody point me in the direction of receiving calls to my SIP URI please? |
| 02:52.51 | JonBach | Which I know is a noob Linux questions...but I can take the mockery :) |
| 02:52.57 | UnluckyAlf | I have set up a sip alias |
| 02:53.06 | UnluckyAlf | user@mydomain.tld |
| 02:53.25 | UnluckyAlf | I have set that directly on the trunk, do I also set it as a DID? |
| 02:57.44 | *** join/#asterisk axisys (n=axisys@ip68-98-177-71.dc.dc.cox.net) |
| 02:58.47 | JonBach | hm |
| 03:00.38 | UnluckyAlf | SRV Records? |
| 03:00.56 | jaytee | JonBach, perhaps you're getting no takers because you've left things pretty vague. You haven't even mentioned whether you're running Asterisk or something else. |
| 03:01.38 | JonBach | Ah, sorry. I'm running Asterisk-now. I'm unsure of what details are useful, I'm afraid. |
| 03:02.09 | jaytee | this channel isn't for AsteriskNOW. it has it's own support channel |
| 03:02.14 | *** join/#asterisk vicom (n=Sam@ool-44c76f10.dyn.optonline.net) |
| 03:02.17 | JonBach | Hah. Thanks. |
| 03:02.55 | AwayML | http://downloads.digium.com/pub/asterisk/ChangeLog-1.4.23-rc1 - on 2008-10-28 at 22:32 +0000, Tilghman Lesher patched apps/app_dial.c. Is there anyway to patch 1.4.22.3 with that fix do you think? |
| 03:03.06 | jaytee | JonBach, do you know anything about linux? |
| 03:03.21 | JonBach | Just enough to get around. |
| 03:03.36 | JonBach | I assume asterisknow uses sendmail, at least it seems so |
| 03:04.41 | jaytee | check the logs in /var/log/asterisk and the mail logs in /var/log/ to see if there are any error indications. it could be a disk space problem. you could also just do a restart of AsteriskNOW |
| 03:04.50 | jaytee | that might fix it |
| 03:04.52 | *** join/#asterisk Kumbang (n=unknown@125.163.83.153) |
| 03:05.25 | AndyML | http://downloads.digium.com/pub/asterisk/ChangeLog-1.4.23-rc1 - on 2008-10-28 at 22:32 +0000, Tilghman Lesher patched apps/app_dial.c. Is there anyway to patch 1.4.22.3 with that fix do you think? |
| 03:08.29 | [TK]D-Fender | AndyML: unload chan_echo.so |
| 03:08.51 | JonBach | Wow, it might just be a DNS issue |
| 03:09.11 | AndyML | [TK]D-Fender: what do you think chan_echo.so is doing? |
| 03:09.39 | jaytee | [TK]D-Fender: what do you think chan_echo.so is doing? |
| 03:10.17 | jaytee | :-) |
| 03:13.57 | *** join/#asterisk moy (n=moy@189.169.60.228) |
| 03:14.14 | AndyML | chan_echo.so isn't loaded... |
| 03:15.07 | [TK]D-Fender | ... |
| 03:15.40 | AndyML | i'm getting weird blip-like calls to my queue memebers. a caller goes to the queue. agent01 gets a ring for 15 seconds, then the CLI shows 'nobody answers in 15000ms' then agent02 gets a blip for a split second and then the CLI shows that nobody answers in 15000ms - all within 1 second. |
| 03:16.20 | [TK]D-Fender | AndyML: Show us the call with complete debug. |
| 03:16.52 | AndyML | you're nothing if not consistant. I appreciate that. http://pastebin.com/m6829b5d6 |
| 03:18.03 | AndyML | line 1145 shows the end of the first call |
| 03:18.11 | AndyML | at 21:20:04 |
| 03:18.28 | AndyML | then 1277 shows the end of the second call at 21:20:04 |
| 03:18.54 | AndyML | then again 1586 again at 21:20:04 |
| 03:19.21 | [TK]D-Fender | AndyML: try again, no CORE debug, just SIP /IAX, etc where applicable. |
| 03:19.33 | AndyML | ok. will do |
| 03:22.08 | AndyML | http://pastebin.com/m78b30a59 |
| 03:23.31 | AndyML | so, line 414, 455, and 549 |
| 03:24.27 | AndyML | only two agents logged in - 691 and 997. |
| 03:26.08 | [TK]D-Fender | AndyML: First, I don't see any sip debug for -- Executing [s@macro-dial:8] Dial("Local/691@from-internal-00b2,2", "SIP/691||trM(auto-blkvm)") in new stack |
| 03:26.15 | [TK]D-Fender | SIP/691 |
| 03:26.21 | [TK]D-Fender | anyWhy not? |
| 03:26.26 | [TK]D-Fender | AndyML: Why not? |
| 03:26.45 | [TK]D-Fender | AndyML: And what about a peer dump? |
| 03:26.51 | [TK]D-Fender | AndyML: You don't seem to looking too hard |
| 03:27.02 | AndyML | I ran 'sip set debug peer 691' then 'sip set debug 997' |
| 03:27.35 | AndyML | http://pastebin.com/m4b215358 |
| 03:27.53 | AndyML | can you enable sip debugging for more than one peer? |
| 03:28.02 | AndyML | http://pastebin.com/m4b215358 - sip show peers |
| 03:29.51 | [TK]D-Fender | AndyML: 691/691 172.18.0.66 D N 64403 OK (167 ms) <-only oddball port of the bunch and the worst ping. now why would THAT be? |
| 03:30.17 | [TK]D-Fender | AndyML: What is this device, and where is it? |
| 03:30.46 | AndyML | 997 and 691 are both through a VPN. 691 is a softphone. this strange timeout behavior happens with hard-devices on the localnetwork though - we just set this up for afterhours testing without going into the office. |
| 03:31.14 | [TK]D-Fender | AndyML: debug your DEFECTIVE phone. |
| 03:32.07 | AndyML | [TK]D-Fender: it happens when these phones are not logged in. JUST like I'm showing you. if you are content to write the problem off as a figment of my imagination simply because I don't have two hard-phone to test it on, then I'll hook up another hard-phone for you. |
| 03:33.17 | [TK]D-Fender | AndyML: You aren't showing me debug for the phone you say should be ringing but isn't. Do you want help or not? |
| 03:34.11 | AndyML | they're both supposed to ring, and they do - just after a blip-ring (<.2 seconds). I do want help of course. |
| 03:34.21 | echelon | i'm working on getting magicjack to work on linux :) |
| 03:34.53 | AndyML | i'll rinse and repeat the test scenerio until I get the right debug information for you... would you prefer it from /var/log/asterisk/full with time stamps? or is the CLI sufficient? |
| 03:35.29 | [TK]D-Fender | AndyML: CLI verbose 10+ SIP only |
| 03:37.11 | AndyML | [TK]D-Fender: to be clear for my benefit. You want "core set verbore 10"+, and "sip set debug peer 997" (assuming 997 is the one that blips. |
| 03:37.22 | [TK]D-Fender | AndyML: yes |
| 03:37.24 | AndyML | thanks |
| 03:37.25 | AndyML | brb |
| 03:40.45 | *** part/#asterisk echelon (i=Unknown@gateway/tor/x-1d748cfc6eebed6b) |
| 03:43.07 | *** part/#asterisk vicom (n=Sam@ool-44c76f10.dyn.optonline.net) |
| 03:43.25 | AndyML | ok - there are several itterations. both 996 and 997 are logged into the queue. in this example, both 996 and 997 are supposed to ring at one point or another and blip before the queue rings the other (showing that the one supposed to ring has already rung for 15000ms.) http://pastebin.com/m3ff6d9f4 |
| 03:44.09 | AndyML | here is an update 'sip show peers' |
| 03:44.10 | AndyML | http://pastebin.com/m14ac9e43 |
| 03:44.13 | AndyML | *updated |
| 03:45.11 | *** join/#asterisk synchris (n=synchris@athedsl-154703.home.otenet.gr) |
| 03:45.11 | *** join/#asterisk n3glv (n=n3glv@c-98-219-138-80.hsd1.pa.comcast.net) |
| 03:47.40 | *** join/#asterisk riddlebox (n=james@75-132-225-75.dhcp.stls.mo.charter.com) |
| 03:48.36 | *** join/#asterisk luke-jr (n=luke-jr@2002:48c4:141a:0:20e:a6ff:fec4:4e5d) |
| 03:48.43 | luke-jr | Anyone know about this "Door King" thing? |
| 03:48.56 | luke-jr | and if it will mess with a regular ATA? |
| 03:54.20 | [TK]D-Fender | AndyML: that entire subnet is looking NAT'd |
| 03:54.37 | AndyML | no - extended subnet. |
| 03:54.43 | AndyML | it isn't /24 |
| 03:55.13 | [TK]D-Fender | AndyML: why am I seeing 996 & 997 traffic on the same IP then? |
| 03:55.35 | AndyML | two sip peers on one device so you can rule out the latency |
| 03:55.43 | [TK]D-Fender | AndyML: 1010 1029 |
| 03:56.02 | [TK]D-Fender | AndyML: and the same PORT? Not sane |
| 03:56.41 | [TK]D-Fender | AndyML: Um... wait.. ,multiple regs on a single phone? |
| 03:56.44 | AndyML | the symptoms are the same whether we're on the defective softphone, the insane multiline Aastra, or the production environment. |
| 03:56.53 | [TK]D-Fender | AndyML: You're trying to emulate mutliple devices with a test phone? |
| 03:57.55 | AndyML | not emulate so much as implement... but if you're not buying it, i can register some agent on the production environment, rinse and repeat... |
| 03:58.07 | [TK]D-Fender | AndyML: No... it is adding up now at least |
| 03:58.14 | [TK]D-Fender | AndyML: Looking a little more |
| 03:58.23 | *** join/#asterisk kisu (n=kisu@daniel1117.broker.freenet6.net) |
| 04:00.01 | *** join/#asterisk esaym (n=user@cpe-70-120-89-6.satx.res.rr.com) |
| 04:00.36 | [TK]D-Fender | AndyML: Whatever it is... I'm just not seeing it at this point... |
| 04:01.02 | AndyML | it looks like a bug in app_dial where the timeout isn't resetting between the calls. |
| 04:03.38 | [TK]D-Fender | AndyML: Highly doubt that... |
| 04:03.46 | [TK]D-Fender | AndyML: Everybody would get hit with that |
| 04:04.21 | AndyML | i'm not convinced that they're not - there are notes here in the 1.4.23-rc1 changelogs that allude to a patch that could solve this - theoretically of course. |
| 04:04.36 | [TK]D-Fender | AndyML: this is calls froma queue isn't it? |
| 04:04.41 | AndyML | yeah |
| 04:04.59 | [TK]D-Fender | AndyML: the timeout isn't from app_dial FWICT |
| 04:05.11 | [TK]D-Fender | AndyML: the channel timeout should be from app_queue |
| 04:05.17 | AndyML | well app_queue then |
| 04:07.28 | *** part/#asterisk philippel (n=p_lindhe@pool-98-111-70-106.sttlwa.fios.verizon.net) |
| 04:08.46 | AndyML | [TK]D-Fender: thanks for looking at it. I'm pretty boggled by it. |
| 04:09.24 | *** part/#asterisk root52 (n=root52@ip70-191-120-39.cl.ri.cox.net) |
| 04:19.30 | *** join/#asterisk Coolthreads (n=Coolthre@203-97-238-71.cable.telstraclear.net) |
| 04:21.29 | *** join/#asterisk kisu (n=kisu@daniel1117.broker.freenet6.net) |
| 04:24.54 | xieliwei | Agk... Need help with chan_mobile |
| 04:24.59 | xieliwei | it does not seem to work with any of my mobiles, even those listed as compatible |
| 04:25.02 | xieliwei | it just lists them as headsets and says they're not usable |
| 04:25.08 | xieliwei | have tried a motorola V3, nokia e51 and htc p3600 running windows mobile 6.1 |
| 04:25.12 | xieliwei | tried it on different systems with both asterisk 1.4 and 1.6 trunk running opensuse 10.2 and 10.3 |
| 04:25.19 | xieliwei | the only thing i can't change is the dongle, but its csr based and I've got many of them used for other linux bluetooth apps on other systems |
| 04:25.25 | xieliwei | i did try using a different dongle (but same model), no difference |
| 04:27.40 | Maliuta | anyone know if a PAP2-NA can be made to do 1FXO+1FXS? |
| 04:28.07 | [TK]D-Fender | Maliuta: lol |
| 04:28.18 | [TK]D-Fender | Maliuta: Doesn't work that way. |
| 04:28.41 | [TK]D-Fender | You need FXO, get a proper device. SPA-3102 would do it |
| 04:29.17 | drmessano^ | xieliwei: Is the dongle being detected? |
| 04:29.55 | Maliuta | [TK]D-Fender: I use a TDM400 here ... just after something to put into my parents place. It will need to deal with an FXO |
| 04:30.15 | [TK]D-Fender | Maliuta: then the SPA-3102 is perfect |
| 04:30.25 | *** join/#asterisk styelz (n=yoohoo@m0o0.mooo.com) |
| 04:30.31 | *** join/#asterisk tpak (n=tpak@c-75-70-54-170.hsd1.co.comcast.net) |
| 04:38.46 | xieliwei | drmessano^ yea, dongle is detected |
| 04:38.57 | xieliwei | hcitools sees my phones too |
| 04:39.07 | xieliwei | and my phones bind to the server okay as well |
| 04:39.26 | drmessano^ | Ok, so whats the problem? |
| 04:39.36 | xieliwei | the phones can't be used |
| 04:39.42 | drmessano^ | How so> |
| 04:39.47 | drmessano^ | Have you set them up? |
| 04:39.50 | drmessano^ | What happens? |
| 04:39.57 | xieliwei | supposedly the voip-info page says the wm6 and e51 phones are supported as handsets |
| 04:39.59 | drmessano^ | Do they show up in Asterisk? |
| 04:40.05 | xieliwei | i do mobile search |
| 04:40.08 | xieliwei | they show up |
| 04:40.11 | drmessano^ | Forget the wiki, its crap |
| 04:40.13 | xieliwei | but they are reported as handsets |
| 04:40.23 | drmessano^ | Then you have them set up that way |
| 04:40.28 | xieliwei | and the "usable" column are all 0 |
| 04:40.30 | xieliwei | *no |
| 04:40.34 | drmessano^ | Dont set them up as handsets |
| 04:40.48 | xieliwei | nope, that was before i set them up |
| 04:40.58 | *** join/#asterisk riddlebox (n=james@75-132-225-75.dhcp.stls.mo.charter.com) |
| 04:41.12 | xieliwei | i was supposed to do mobile search, find their bt address and port and then set them up in mobile.conf |
| 04:41.26 | drmessano^ | ok |
| 04:41.28 | xieliwei | but all three phones I've tried so far says they cannot be used |
| 04:42.00 | jameswf | can anyone confirm http://bugs.digium.com/view.php?id=13912 |
| 04:42.01 | drmessano^ | Then you have some really obscure phones.. I have a Samsung, a Motor Razr v1 and Blackberry that all work |
| 04:42.14 | xieliwei | i have the motor razr v1 |
| 04:42.25 | xieliwei | its detected as handset and not usable |
| 04:42.27 | xieliwei | *headset |
| 04:42.47 | xieliwei | so i'm not sure what's wrong |
| 04:43.35 | drmessano^ | I dunno what to tell you.. maybe its the dongle |
| 04:43.57 | xieliwei | hmm, that means i should get another one |
| 04:43.58 | drmessano^ | Seanbright FTW |
| 04:44.03 | drmessano^ | "This is by design." |
| 04:44.04 | xieliwei | but could it be any configuration? |
| 04:44.12 | drmessano^ | xieliwei: Sure it could |
| 04:44.15 | drmessano^ | I told you that |
| 04:44.24 | xieliwei | hmm |
| 04:44.32 | xieliwei | i mean bluetooth config? |
| 04:44.44 | *** join/#asterisk rfernandez (n=rfernand@189.136.65.226) |
| 04:44.47 | drmessano^ | Possibly.. |
| 04:45.06 | rfernandez | hi!! for a hundred sip extensions running via lan in g711u which switch do you prefer? |
| 04:45.10 | xieliwei | would you be able to get me your bluetooth configuration files? |
| 04:45.19 | xieliwei | minus all the private stuff of course |
| 04:46.00 | xieliwei | by the way, i followed this guide: http://www.voipphreak.ca/2008/10/30/installing-and-configuring-chan_mobile-for-bluetooth-presence-support-in-asterisk-16/ |
| 04:46.05 | drmessano^ | Mine look just like the default config.. I have the mac of the phone, port=4, a context, an adapter name |
| 04:46.25 | xieliwei | what about your hci.conf? |
| 04:46.33 | xieliwei | *hcid.conf |
| 04:47.50 | drmessano^ | OS default |
| 04:47.51 | drmessano^ | No changes |
| 04:48.01 | xieliwei | okay... |
| 04:48.05 | drmessano^ | I've done very little to make this work.. it just 'worked' |
| 04:48.29 | xieliwei | dang, i need more of those apple "It just works" mojo |
| 04:48.43 | trnzmeta | if only everything was like that |
| 04:48.47 | xieliwei | asterisk has always been a pain for me |
| 04:49.21 | xieliwei | oh lookie lookie, i found another different brand dongle... lemmi plug it in and try |
| 04:50.05 | *** join/#asterisk freakazoid0223 (n=mattc@pool-71-242-219-34.phlapa.east.verizon.net) |
| 04:51.33 | xieliwei | ah shucks, the case broke into pieces... but its in nevertheless |
| 04:52.33 | xieliwei | (lesson learnt: don't plug in unknown dongles without checking) |
| 04:52.40 | xieliwei | apparently the server stopped responding |
| 04:53.23 | drmessano^ | ok |
| 04:54.39 | xieliwei | okay, a kernel panic occurred... i'll just get it rebooted and try it with this dongle and see what happens |
| 04:54.43 | xieliwei | thanks drmessano& |
| 04:54.52 | xieliwei | * drmessano^ |
| 04:55.07 | drmessano | bah |
| 04:55.10 | drmessano | I hate carrots |
| 04:55.32 | xieliwei | haha |
| 04:58.38 | xieliwei | hmm, while waiting for the server to come back up, I wonder has anyone tried doing something similar to chan_mobile using the serial+audio cable method? |
| 04:59.05 | xieliwei | i'd look into that if its more stable (most likely since it is wired) |
| 05:00.04 | xieliwei | i see celliax, but it seems kind of messy now |
| 05:00.29 | xieliwei | and i'm not sure what handsets will work, so i'll need to find out |
| 05:06.02 | drmessano | chan_mobile uses a common platform to support all devices via bluetooth.. When it comes to hardwired setup, things are all over the place |
| 05:06.14 | drmessano | I would not want an audio coupled cell phone |
| 05:06.19 | drmessano | Too my RFI |
| 05:07.57 | riddlebox | does anyone use a sip phone on a nokia n810? |
| 05:07.59 | xieliwei | i think so too... but if chan_mobile won't work, i'll have no choice |
| 05:11.08 | xieliwei | weird, its not finding my phone now |
| 05:11.17 | drmessano | Get a decent dongle and make it work |
| 05:11.23 | drmessano | $5 on ebay |
| 05:11.40 | xieliwei | supposedly the first dongle i used is decent |
| 05:12.12 | drmessano | Well, here are the options |
| 05:12.12 | xieliwei | but i'll go get one tomorrow and see if things get better |
| 05:12.32 | drmessano | The dongles you have done work, your config is wrong, or your asterisk install is hosed |
| 05:12.44 | drmessano | dont* |
| 05:13.05 | xieliwei | okay, so dongles is still a possibility, config too |
| 05:13.11 | xieliwei | asterisk install can be eliminated |
| 05:13.13 | *** join/#asterisk aliraja (n=aliraja@202.125.156.122) |
| 05:13.35 | xieliwei | i installed asterisk on another machine and tried too |
| 05:13.43 | xieliwei | fresh trunk |
| 05:14.07 | *** join/#asterisk rrrobert (n=rrrobert@202.125.156.122) |
| 05:14.24 | xieliwei | but you said the razr v1 should work right? that should confirm something is wrong on my side |
| 05:15.16 | drmessano | indeed |
| 05:15.47 | aliraja | hi all ,can any one explain me difference b/w call tranfer feature using Dial command and using transfer application... |
| 05:16.06 | xieliwei | hmm, just one more point to eliminate... how do i check my usb bus speed? |
| 05:16.38 | xieliwei | i fear my server my be running on 1.1, its kinda old |
| 05:22.45 | [TK]D-Fender | aliraja: Dial is NOT a "transfer" |
| 05:23.43 | [TK]D-Fender | aliraja: "Transfer" is for telling * to take the call and THROW it off the server. If the originating call is a SIP call and the target channel is a SIP call then it will tell the originating channel to "leave" |
| 05:24.48 | *** join/#asterisk workdraft (n=acxide@203.215.94.239) |
| 05:26.26 | aliraja | [TK]D-Fender, I means to say passing t argument to dial application and using transfer application .,, |
| 05:29.12 | [TK]D-Fender | aliraja: same ting |
| 05:29.46 | [TK]D-Fender | aliraja: "t" allowsyou to transfer the call to another extension INSIDE of asterisk. "Transfer" is to completely EJECT the call from your server |
| 05:29.53 | rrrobert | hello aliraja |
| 05:30.10 | rrrobert | how r u bro? |
| 05:30.21 | aliraja | [TK]D-Fender, Thanks alot |
| 05:30.33 | aliraja | rrrobert, yes |
| 05:31.12 | rrrobert | aliraja, how is it going with yr yada, |
| 05:32.41 | rfernandez | hi!! for a hundred sip extensions running via lan in g711u which switch do you prefer? |
| 05:32.58 | aliraja | rrrobert, its almost finished |
| 05:33.53 | rrrobert | great job man, now finally you are able to handle cdr with oracle, it would be nice if you write a small howto page on yada :-) |
| 05:34.12 | rfernandez | hi!! just a little question, if i need to deploy 100 sip extensions running ulaw which switch do i need to get best performance (cause are all softphones) |
| 05:35.12 | aliraja | rrrobert, sure |
| 05:36.02 | rrrobert | so sweet.. |
| 05:47.22 | *** join/#asterisk ghento (n=ghento@CPE0014bfaf7d7a-CM00159a03312c.cpe.net.cable.rogers.com) |
| 05:47.29 | [TK]D-Fender | rfernandez: its a switch. Ordds are 100mbit minimum. ULAW = 85kBIT. that means you don't have to care. at all. |
| 05:48.06 | rfernandez | [TK]D-Fender, for 100 sip extensions runnig ulaw? |
| 05:48.31 | [TK]D-Fender | rfernandez: Yes |
| 05:48.43 | rfernandez | okies =D a gigabit should be fine then? |
| 05:48.54 | [TK]D-Fender | rfernandez: and thats even assuming that all 100 are talking at once, AND going through * for RTP |
| 05:49.10 | rfernandez | oh ok! |
| 05:49.19 | rfernandez | so the switch should be no problem? |
| 05:49.33 | rfernandez | if the customer wants to use laptops (no desktops) in wifi mode? shouldnt be a trouble right? |
| 05:49.51 | rfernandez | cause in the shop recommend to use a layer 3 switch but i feel its too big =S |
| 05:54.11 | [TK]D-Fender | rfernandez: WIFI = bleh.... not true full duplux... |
| 05:54.18 | [TK]D-Fender | rfernandez: and softphones SUCK |
| 05:54.25 | [TK]D-Fender | rfernandez: I pity your users |
| 05:54.30 | rfernandez | [TK]D-Fender, lol |
| 05:54.49 | *** join/#asterisk chendy (n=chatzill@121.34.152.233) |
| 05:54.57 | rfernandez | [TK]D-Fender, me too i thought about cat 6 wire and wire the access points... |
| 05:55.05 | rfernandez | all in cat 6 for best efford |
| 05:55.38 | [TK]D-Fender | the AP is your weak link no, the wire you run from it |
| 05:56.06 | [TK]D-Fender | Wireless G at best is 54mbip, which you never really get... the rest is split across a LOT of connections.. |
| 05:56.13 | rfernandez | [TK]D-Fender, i was thinking a 108 mbps... |
| 05:56.16 | [TK]D-Fender | I really don't even want to to think about such a deployment. |
| 05:56.33 | [TK]D-Fender | rfernandez: And your enpoints all support N? |
| 05:56.48 | rfernandez | [TK]D-Fender, i prefer to wire all the office but he wants to use the laptop even in the bathroom (sarcasm) cause they are mobile agents... |
| 05:56.49 | [TK]D-Fender | rfernandez: This topology is really not advisable |
| 05:57.11 | rfernandez | [TK]D-Fender, sure the topology its a lot of insecure.... |
| 05:57.18 | rfernandez | im afraiding of the ap |
| 05:58.02 | drmessano | Well |
| 05:58.09 | drmessano | What you need to do |
| 05:58.32 | drmessano | Is put a biscuit jack on top of every laptop users desk that they can plug a cat5 cable into |
| 05:58.34 | drmessano | and can the wireless |
| 05:58.39 | drmessano | Except for execs |
| 05:58.43 | rfernandez | the customer have 100 employees al with laptops, he wants to use softphones... |
| 05:58.52 | rfernandez | yep |
| 05:59.08 | rfernandez | im thinking about wire every station and the wireless using as a "floating" point |
| 05:59.16 | drmessano | That whole "Its a laptop, so I need wireless" line is laziness on their end.. they want to be "cute" |
| 05:59.23 | drmessano | yes |
| 05:59.36 | [TK]D-Fender | drmessano: or cheap on wiring |
| 05:59.42 | [TK]D-Fender | drmessano: then again... SOFTPHONES. |
| 05:59.44 | [TK]D-Fender | BLEH |
| 05:59.45 | rfernandez | ironi of life: the 100 ppl are security informatic experts, lol! |
| 05:59.55 | rfernandez | xD |
| 06:00.32 | rfernandez | conclusion: a gigabit switch (whatever i want) will work fine sure? |
| 06:00.41 | rfernandez | assumming 100 ppl talking simmultanoeusly? |
| 06:01.16 | [TK]D-Fender | rfernandez: Again, who cares about the wired part when the WIRELESS is where you are at risk |
| 06:01.21 | rfernandez | cause i thinked a bout if its 80 kbps the ulaw codec... 100 ppl are 8 gigabits... |
| 06:01.42 | rfernandez | [TK]D-Fender, no problem ill convince the boss to wire everything |
| 06:01.51 | rfernandez | and dont use the wifi |
| 06:02.19 | [TK]D-Fender | And still leave your poor chumps on soft-phones |
| 06:02.45 | rfernandez | well im guessing he wants "the "cheaper" option (not the true option |
| 06:02.57 | rfernandez | a true option includes aastra hardphones =D |
| 06:04.31 | workdraft | is there a recommended sound card for Sip clients? |
| 06:04.32 | rfernandez | id the implementation goes well ill write a white paper for the eternity lol! |
| 06:04.39 | rfernandez | *if |
| 06:05.42 | rfernandez | well good ppl ill go to the kitchen im starving |
| 06:05.44 | rfernandez | see ya later |
| 06:05.48 | rfernandez | and thanks! =D |
| 06:06.17 | [TK]D-Fender | workdraft: huh? |
| 06:06.32 | rfernandez | [TK]D-Fender, drmessano thank you! =D |
| 06:06.33 | [TK]D-Fender | workdraft: SIP is a network protocol. What does this have to do with a sound card? |
| 06:07.03 | workdraft | sip clients or sip phones in desktop |
| 06:07.23 | workdraft | i must have said it wrong |
| 06:08.20 | [TK]D-Fender | workdraft: You mean a soft-phone? |
| 06:08.35 | workdraft | what sound card would you recommend installed in a desktop computer that is using a soft phone, specifically a sip client soft phone. |
| 06:09.14 | [TK]D-Fender | workdraft: Ah well... something with a better SNR ratio (Creative Labls, ASUS, etc), but seriouly... if you;re ven thinking about the sound card you should jsut get thema hard phone. |
| 06:09.20 | [TK]D-Fender | Not that you shouldn't do that ANYWAYS |
| 06:09.23 | [TK]D-Fender | softphones SUCK. |
| 06:10.59 | workdraft | then ill look for hard phones that is a headset pluggable. |
| 06:11.15 | workdraft | or hardphones with headset. |
| 06:12.01 | workdraft | any specific hard phone to recommend? |
| 06:15.34 | [TK]D-Fender | workdraft: Polycom IP 32.330 for your typical user |
| 06:15.39 | [TK]D-Fender | 320/330 |
| 06:15.46 | workdraft | thnx. |
| 06:16.01 | [TK]D-Fender | workdraft: Outside of Nort America, Linksys is an econimcal option |
| 06:16.15 | [TK]D-Fender | ok, checkout time... later all |
| 06:16.31 | workdraft | k |
| 06:26.50 | *** join/#asterisk zeeesh (n=zeeesh@203.215.179.43) |
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| 06:47.59 | *** mode/#asterisk [+o denon] by ChanServ |
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| 06:48.37 | *** mode/#asterisk [+o denon] by ChanServ |
| 06:56.34 | *** join/#asterisk sosperec (n=david@office.axpnet.com) |
| 06:56.39 | sosperec | hello |
| 07:04.24 | *** part/#asterisk sosperec (n=david@office.axpnet.com) |
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| 08:21.52 | JonCup | Any body around? |
| 08:23.40 | *** join/#asterisk fcois93 (n=Administ@bagnolet.acropolistelecom.net) |
| 08:23.45 | fcois93 | hello all! |
| 08:24.10 | fcois93 | is it possible to do 'sip show peers' in the dialplan to know if a peer is online ? |
| 08:25.08 | *** join/#asterisk oej (n=olle@ns.webway.se) |
| 08:25.29 | kaldemar | fcois93: core show function SIPPEER |
| 08:29.02 | JonCup | Hey guys, any one know anything about hardware virtualization. I wanna build a server running an asterisk server, a apache server, and windows server 2008, all sharing the same hardware, but I want to make sure the asterisk server has full priority over the hardware, can any one give me any insight or ideas in this |
| 08:29.08 | JonCup | the hardware is quad core zeon 2.66 ghz, 8 GB ram, 4x500 GB disks ( preconfigured in a 1.5 TB raid 5 array |
| 08:29.11 | fcois93 | kaldemar: ok, I will have a look,,thank you |
| 08:30.18 | fcois93 | kaldemar: it is exactualy what I needed :-) |
| 08:30.24 | JonCup | Is this going to work? The asterisk config is simple, 5 sip phones (and maybe a few softphones) and 9 SIP lines from bandwith.com |
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| 08:44.59 | Karlitoo | hey guys I'm trying to authenticate on a local asterisk server but all of the sip softphones that are available send the auth request as sip:pass@domain instead of user:pass@domain |
| 08:45.11 | Karlitoo | can any 1 give me a hint on what to do |
| 08:46.18 | kaldemar | change your configuration. |
| 08:47.06 | kaldemar | on the phones, that is. |
| 08:47.16 | *** join/#asterisk angryuser (n=gdobrovo@LPuteaux-151-42-35-99.w193-251.abo.wanadoo.fr) |
| 08:48.05 | Karlitoo | don't know exactly where I'm using twinkle |
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| 09:09.03 | ecm | Hello. If, for a sip device, I've configured calllimit=3, is there any way to see if any of those 3 calls are currently being used |
| 09:09.29 | ecm | ? because DIALSTATUS works only with calllimit=1, as far as I can see |
| 09:12.28 | *** join/#asterisk lupino3 (n=andrea@static-217-133-45-108.clienti.tiscali.it) |
| 09:14.48 | angryuser | ecm hm, core show function SIPPEER |
| 09:15.50 | angryuser | ecm: check if there any way to check the number of concurrent connections, if no you can use group() and groupcount() |
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| 09:32.52 | ecm | angryuser, thank you very much. I'll try your suggestions. |
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| 10:11.12 | hi365 | what argumanets does genzaptelconf need to generate zaptel and zapata? |
| 10:20.23 | ecm | SIPPEER with curcalls works great, thanks again angryuser |
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| 10:25.17 | ertyui | hi 2 all |
| 10:25.37 | ertyui | my asterisk system work |
| 10:25.45 | ertyui | well |
| 10:26.09 | ertyui | i just got a little question |
| 10:27.04 | ertyui | is there any phone where we enter the calling destination fees |
| 10:27.06 | ertyui | ? |
| 10:27.28 | ertyui | first is there anyone ihere ? |
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| 10:35.15 | Daejeo | In which directory I can find extensions messages such as name, busy message and temp greetings |
| 10:35.24 | Daejeo | ? |
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| 10:47.10 | ghostknife | anyone know what function I can use to create an extension that simply echos back what I say a second or so after I stop talking? |
| 10:48.21 | ghostknife | It's pissing people off to constantly phone them and ask them to repeat stupid things like 1-2-3, testing, hallo and i r baboon |
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| 10:50.17 | DarKnesS_WolF | tzafrir_laptop: there? |
| 10:50.56 | SwK | ghento, the echo app |
| 10:53.17 | DarKnesS_WolF | tzafrir_laptop: having a very strange problem with Ericsson G36 GSM and the astribank... when i call from * to any other mobile via teh GSM gateway the call get dics. after 3 secounds but this disconnection only from mobile but the channel between the asterisk and teh GSM gateway not off and i get this messages |
| 10:53.38 | DarKnesS_WolF | [Nov 17 11:46:36] DEBUG[19698] chan_zap.c: Ignoring Polarity switch to IDLE on channel 6, state 6 |
| 10:53.41 | DarKnesS_WolF | [Nov 17 11:46:36] DEBUG[19698] chan_zap.c: Polarity Reversal event occured - DEBUG 2: channel 6, state 6, pol= 0, aonp= 0, honp= 0, pdelay= 600, tv= -1445449907 |
| 10:53.45 | DarKnesS_WolF | [Nov 17 11:46:39] DEBUG[19698] chan_zap.c: Ignore switch to REVERSED Polarity on channel 6, state 6 |
| 10:53.55 | ghostknife | wow! I found the echo back dialplan application!! it's called... "Echo" :> |
| 10:54.40 | ghostknife | when I edited extensions.conf, what "asterisk -r" command can reload it? |
| 10:59.12 | tzafrir_laptop | DarKnesS_WolF, here |
| 10:59.24 | DarKnesS_WolF | tzafrir_laptop: i find it is a bug ? but that back in 2005 |
| 10:59.29 | DarKnesS_WolF | http://bugs.digium.com/file_download.php?file_id=4354&type=bug |
| 10:59.40 | DarKnesS_WolF | +;answeronpolarityswitch=yes or so :-s but not sure what do u think ? |
| 11:00.49 | tzafrir_laptop | http://bugs.digium.com/view.php?id=13917 |
| 11:02.01 | tzafrir_laptop | Maybe play with polarityonanswerdelay ? |
| 11:02.43 | *** join/#asterisk shine (n=stroll@lamantin.achamo.net) |
| 11:05.19 | DarKnesS_WolF | tzafrir_laptop: yes maybe any options i can find :)? |
| 11:06.03 | DarKnesS_WolF | tzafrir_laptop: i'll need to patch / compile ? |
| 11:06.24 | tzafrir_laptop | for that option? no |
| 11:07.05 | DarKnesS_WolF | so just answeronpolarityswitch=yes |
| 11:07.05 | DarKnesS_WolF | hanguponpolarityswitch=yes |
| 11:07.07 | DarKnesS_WolF | <PROTECTED> |
| 11:07.11 | DarKnesS_WolF | ok will try |
| 11:09.53 | tzafrir_laptop | DarKnesS_WolF, if you don't have answeronpolarityswitch, it shouldn't matter |
| 11:13.33 | *** join/#asterisk caio1982 (i=caio1982@CAcert-br/caio1982) |
| 11:19.16 | *** join/#asterisk kamanashisroy (n=kamanash@119.30.34.7) |
| 11:19.25 | Daejeo | tzafrir_laptop: In which directory I can find extensions messages such as name, busy message and temp greetings |
| 11:19.44 | *** part/#asterisk orly_owl (n=DavoDink@c122-108-147-164.sunsh1.vic.optusnet.com.au) |
| 11:19.51 | tzafrir_laptop | under voicemail? |
| 11:20.40 | Daejeo | :) |
| 11:23.23 | tzafrir_laptop | Daejeo, in the sounds directory |
| 11:23.45 | tzafrir_laptop | though I'm not sure which ones you refer to |
| 11:24.04 | Daejeo | extension: 4545 |
| 11:24.13 | Daejeo | *98 |
| 11:24.16 | DarKnesS_WolF | tzafrir_laptop: sorry didnt get it it is an GSM gateway may be they are using it i don't know i'll just enable this options and test. |
| 11:24.52 | *** join/#asterisk LND (n=lee@92-233-208-244.cable.ubr08.gate.blueyonder.co.uk) |
| 11:24.53 | dominic1 | why isn't asterisk running by default with safe_asterisk. In earlier versions I had in the initscript of debian the cli command to run wirh safe_asterisk. Now I saw that it changed to /usr/sbin/asterisk |
| 11:25.02 | dominic1 | I have currently the problem that asterisk is not writing coredumps on crashes |
| 11:25.35 | dominic1 | I am currentyl using the new asterisk initscript. With the older one it was no problem |
| 11:25.38 | *** join/#asterisk DJ_HaMsTa (i=DJ_HaMsT@c-69-136-240-75.hsd1.nj.comcast.net) |
| 11:25.44 | DJ_HaMsTa | does avaya colect royalties to use their system ? |
| 11:26.34 | Daejeo | "/var/spool/asterisk/voicemail/default" |
| 11:30.48 | *** join/#asterisk mbranca (n=matteo@93-62-230-243.ip24.fastwebnet.it) |
| 11:42.50 | *** join/#asterisk seanbright (n=sean@asterisk/contributor-and-bug-marshal/seanbright) |
| 11:43.34 | dominic1 | ? |
| 11:59.35 | Daejeo | Is it possible to have both a busy and an away message when the call |
| 11:59.37 | Daejeo | waiting feature is enabled? |
| 12:07.38 | *** join/#asterisk coppice (n=chatzill@33.155.17.210.dyn.pacific.net.hk) |
| 12:12.30 | *** join/#asterisk bartpbx (n=bartpbx@217.24.210.202) |
| 12:16.14 | *** join/#asterisk farkus (n=chatzill@cpe-72-225-212-219.nyc.res.rr.com) |
| 12:19.47 | *** join/#asterisk gaetronik (n=gaetan@200.111.138.170) |
| 12:20.12 | dominic1 | how can I activate core dumps on crash? I am using the debian initscript from 1.4.21.2 |
| 12:20.31 | dominic1 | if I adjust in /etc/asterisk/asterisk.conf nothing happens |
| 12:20.36 | dominic1 | I did not get any dumps |
| 12:20.48 | dominic1 | with my older asteriskversion and safe_asterisk that worked.... |
| 12:26.08 | kaldemar | is asterisk using /etc/asterisk/asterisk.conf? there's an option -C for for the asterisk binary to use some other file. |
| 12:29.54 | *** join/#asterisk jer (n=jer@unaffiliated/jer) |
| 12:31.59 | *** join/#asterisk kisu (n=kisu@daniel1117.broker.freenet6.net) |
| 12:32.36 | *** join/#asterisk raasdnil (n=mikel@60-241-138-146.static.tpgi.com.au) |
| 12:32.39 | raasdnil | evening all |
| 12:40.38 | dominic1 | I didn't have a look I thought the initscript defaults to asterisk.conf |
| 12:40.50 | dominic1 | I read something about ulimit for dumps.... |
| 12:41.04 | dominic1 | this is only used in the safe asterisk script |
| 12:41.22 | dominic1 | in the newer version of the asterisk init safe_asterisk isn't used anymore |
| 12:42.39 | *** join/#asterisk sosperec (n=david@office.axpnet.com) |
| 12:42.44 | sosperec | Hello! |
| 12:44.41 | sosperec | What should I do to convince asterisk not to authenticate always? I have a central asterisk and a remote one. The remote handles the extensions. When I try to call out, I get this on the central: NOTICE[3061025680]: chan_sip.c:13409 handle_request_invite: Failed to authenticate user "Tanacs David" <sip:52@remote.ip.address>;tag=as755b8893 |
| 12:46.29 | kaldemar | sosperec: take a look at parameter insecure in sip.conf. |
| 12:46.49 | raasdnil | heya anyone... had any problems getting a fax machine to send dtmf through asterisk? Seems like it is autodialing too fast and asterisk is not getting all the dtmf tones properly |
| 12:48.21 | sosperec | kaldemar: insecure=very in both sip.conf files |
| 12:49.59 | sosperec | kaldemar: http://pastebin.com/d50c83df3 |
| 12:50.03 | sosperec | this is the central |
| 12:53.14 | *** part/#asterisk LND (n=lee@92-233-208-244.cable.ubr08.gate.blueyonder.co.uk) |
| 12:53.19 | sosperec | kaldemar: this is the remote station: http://pastebin.com/d6299c963 |
| 12:54.16 | Titanous | I've got an Aastra phone on a remote IP, when I call it from my grandstream on the local network (where the asterisk box is) I get full two way audio, when the aastra phone tries to call out, The call is processed by asterisk, but no audio in either direction |
| 12:54.20 | Titanous | suggestions? |
| 12:54.40 | kaldemar | sosperec: what version of asterisk are you using? |
| 13:00.25 | *** join/#asterisk mRCUTEO (n=info@64.235.47.76) |
| 13:01.12 | kaldemar | sosperec: the insecure=very might have been removed at some point. you need the insecure in the peer's context in the central asterisk. insecure=invite,port. |
| 13:01.14 | mRCUTEO | hi what is asteriksnow? and how to make a simple server client voip communication using asterisk? can anyone help? |
| 13:03.35 | sosperec | kaldemar: 1.2 |
| 13:04.21 | sosperec | kaldemar: very changed to invite,port , but still not working. I got a message, that the remote peer is now reachable |
| 13:04.38 | sosperec | kaldemar: at a call, still the same, failed to auth |
| 13:21.47 | x86 | is there a way I can hook up an external loud ringer to a Polycom IP330 phone? |
| 13:22.03 | x86 | all I can seem to find is external loud ringers for analog phones :( |
| 13:22.18 | x86 | I've got a phone in a warehouse that no one can hear when it rings |
| 13:24.41 | DarKnesS_WolF | x86: huh :P what is loud ringers :P |
| 13:25.01 | *** join/#asterisk Dr-Linux|home (n=Nothing@119.63.130.34) |
| 13:25.23 | Dr-Linux|home | anybody is using SRV record with Asterisk? |
| 13:26.23 | dominic1 | where will I have to adjust the startparameters of asterisk to set the var "$ASTARGS in /etc/init.d/asterisk under debian |
| 13:26.46 | dominic1 | is it better to use the initscript from debian than from asterisk, in debian I can adjust the parameters in /etc/default/asterisk |
| 13:26.58 | dominic1 | http://bugs.digium.com/view.php?id=9843 |
| 13:28.56 | *** join/#asterisk jer (n=jer@unaffiliated/jer) |
| 13:33.03 | x86 | DarKnesS_WolF: an external speaker that can be mounted high on a wall in a factory, so people can hear the phone ring ;) |
| 13:33.56 | x86 | looks like they only make them analog, so I'll have to get an ATA, an analog external loud ringer, and set the ATA to be in the same ring group as the Polycom phone |
| 13:39.17 | tzafrir_laptop | dominic1, I personally prefer the packaged one... |
| 13:40.52 | n3glv | x86: softphones ring out the sound card (or can) |
| 13:41.24 | *** part/#asterisk n3glv (n=n3glv@c-98-219-138-80.hsd1.pa.comcast.net) |
| 13:49.26 | dominic1 | ehm, the debian packaged? |
| 13:51.06 | *** join/#asterisk [TK]D-Fender (n=joe@216.191.106.163) |
| 13:52.20 | DarKnesS_WolF | x86: mmmmm nice idea get a dect phone :P |
| 13:53.53 | *** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 13:59.43 | *** join/#asterisk Mark17 (n=mark@vnc.tt.streamservice.nl) |
| 13:59.49 | Mark17 | hello |
| 14:00.05 | *** join/#asterisk lmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 14:00.05 | *** mode/#asterisk [+o lmadsen] by ChanServ |
| 14:01.24 | *** join/#asterisk n3hxs (n=HAMming@static-151-196-93-200.balt.east.verizon.net) |
| 14:01.36 | Mark17 | how should it look in my log files if asterisk is correctly connected to a mobile phone (using bluetooth with an usb dongle)? |
| 14:05.46 | dominic1 | tzafrir_laptop, do you mean the debian or the asterisk package? I had the problem with the asterisk initscript, that I didn't found an option to activate coredumps except hardcoding the -g parameter.... |
| 14:06.17 | tzafrir_laptop | dominic1, at least in the latest version it is possible |
| 14:07.10 | dominic1 | I tried with 1.4.21.2 and 1.4.22. Are there any newer scripts? |
| 14:07.25 | tzafrir_laptop | I mean: of the Debian package |
| 14:07.27 | Katty | yawn. |
| 14:08.25 | dominic1 | ah okay, thank you, now I am using http://svn.debian.org/wsvn/pkg-voip/asterisk/trunk/debian/asterisk.init?op=file&rev=0&sc=0 |
| 14:08.39 | Mark17 | did someone manage to get a bluetooth connection working on asterisk? |
| 14:08.41 | dominic1 | I found them in the bugreport posted earlier |
| 14:08.45 | Mark17 | with a connection to a mobile phone |
| 14:08.56 | Katty | Qwell: tuesdays :< |
| 14:10.42 | jsmith | ~tuesday |
| 14:10.42 | jbot | Tuesday sucks, because it follows Monday (see monday). |
| 14:11.31 | Katty | we need to put something down about server maintenance on tuesdays. |
| 14:12.14 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 14:12.39 | dominic1 | ~monday |
| 14:12.40 | jbot | methinks monday is when everything breaks for no apparent reason, creating so many problems that it takes you until friday to get back to normal. at which point one more thing breaks that takes you the whole weekend to fix |
| 14:13.51 | Katty | pouts |
| 14:14.00 | tzanger | heh |
| 14:14.15 | *** join/#asterisk synchris (n=synchris@athedsl-154703.home.otenet.gr) |
| 14:14.25 | Mark17 | ~monday |
| 14:14.26 | jbot | monday is, like, when everything breaks for no apparent reason, creating so many problems that it takes you until friday to get back to normal. at which point one more thing breaks that takes you the whole weekend to fix |
| 14:14.33 | *** part/#asterisk bartpbx (n=bartpbx@217.24.210.202) |
| 14:14.51 | Katty | file: oh happy day! |
| 14:15.09 | file | Katty: yes! |
| 14:16.39 | Daejeo | Meow, Meow :) |
| 14:16.47 | Katty | mrrrrow |
| 14:17.55 | *** join/#asterisk theHub (n=theHub@69.177.93.21) |
| 14:19.15 | Mark17 | did someone manage to get a bluetooth connection working on asterisk? so it connects to a mobile phone? |
| 14:19.31 | Katty | baroo? |
| 14:19.36 | Katty | why would you do that? |
| 14:19.43 | Katty | bluetooth has limited range. |
| 14:20.05 | yang | Mark17: bluetooth is being managed by bluetooth daemon, I don't see any relevance to asterisk |
| 14:20.11 | Mark17 | the phone is within 2M of the bluetooth dongle |
| 14:20.21 | Katty | oh |
| 14:20.26 | Katty | yes, we were able to get that to work. |
| 14:20.30 | Katty | with polycom 501s |
| 14:20.33 | Katty | and voyager headsets |
| 14:20.39 | Katty | and cellphones |
| 14:20.47 | Mark17 | it is for making cheaper calls (read: free) to certain numbers |
| 14:20.58 | Katty | ^_- |
| 14:21.04 | Mark17 | how did you do that Katty? |
| 14:21.05 | Katty | k, maybe i don't get it. that does not parse either. |
| 14:21.27 | DJ_HaMsTa | Katty cook ? |
| 14:21.32 | Katty | DJ_HaMsTa: yes i do. |
| 14:21.40 | DJ_HaMsTa | er nvm |
| 14:21.44 | Katty | k |
| 14:21.51 | DJ_HaMsTa | shes our manager for the voip team :P |
| 14:21.56 | file | tickles Katty |
| 14:21.59 | Mark17 | with the mobile phone it is free of charge to call to certain phonenumber and with the other connection (sip trunk) it is free to make calls to other numbers |
| 14:22.03 | Katty | hai file! |
| 14:22.16 | Katty | DJ_HaMsTa: while i do manage all things voip here, for the most part, my last name is not cook :P |
| 14:22.18 | DJ_HaMsTa | file is a shark! |
| 14:22.27 | Katty | file is mah brother |
| 14:22.30 | Katty | from another mutter |
| 14:22.42 | DJ_HaMsTa | hai = shark in german |
| 14:22.43 | Katty | or possibly my twin. |
| 14:22.47 | Katty | orly? |
| 14:22.50 | Katty | files away in memory |
| 14:23.06 | file | Katty: I may not be your twin but I am lmadsen |
| 14:23.12 | Mark17 | and it is haai in dutch ;) |
| 14:23.12 | Titanous | I've got an Aastra phone on a remote IP, when I call it from my grandstream on the local network (where the asterisk box is) I get full two way audio, when the aastra phone tries to call out, The call is processed by asterisk, but no audio in either direction |
| 14:23.16 | Titanous | suggestions? |
| 14:23.40 | Katty | Titanous: check your rtp ports. |
| 14:23.44 | Daejeo | grandstream ? |
| 14:23.50 | Mark17 | Katty: how did you manage it to work? |
| 14:23.54 | Daejeo | ? grandstream |
| 14:23.56 | *** join/#asterisk jeffik (n=jeffik@69-196-165-181.dsl.teksavvy.com) |
| 14:23.56 | Katty | Titanous: rtp.conf must match with the ports you have opened/forwarded in your firewall |
| 14:24.07 | Katty | Mark17: i don't know what you're doing. |
| 14:24.12 | Katty | Mark17: your description does not parse. |
| 14:24.20 | Daejeo | Mark17 : Mark Spencer ? |
| 14:24.29 | Katty | Daejeo: no, that's not mark spencer |
| 14:24.46 | lmadsen | mark spencer == kram |
| 14:24.55 | lmadsen | who doesn't really go on IRC anymore |
| 14:25.00 | lmadsen | haven't seen him here for a couple years now |
| 14:25.02 | Katty | indeed |
| 14:25.06 | Katty | but i want his shirt. |
| 14:25.09 | *** join/#asterisk ddunavant (n=David@75.145.240.14) |
| 14:25.13 | Katty | his I'm A Super Duper Programmer shirt |
| 14:25.32 | Katty | mark's kinda weird. |
| 14:25.35 | *** join/#asterisk Dovid (n=annon@tony09-121-90.inter.net.il) |
| 14:25.39 | Katty | we had drinks once tho (= |
| 14:25.44 | Dovid | how many peole here are using 1.6.X ? |
| 14:25.45 | Titanous | Katty: all packets from the asterisk IP are forwarded directly to the phone (no matter what port), and the ports in rtp.conf/sip/iax2 etc are all forwarded to asterisk |
| 14:25.59 | Mark17 | Katty: i am working on the following setup: 1 asterisk server with debain and a bluetooth dongle, 1 mobile phone (with bluetooth) |
| 14:26.02 | Katty | Dovid: i started using it, but discovered the ami changed so much that isymphony wouldn't work |
| 14:26.04 | *** join/#asterisk telnettech (n=bsimpson@d192-24-95-65.col.wideopenwest.com) |
| 14:26.05 | Titanous | I have full two way audio when I initiate the call |
| 14:26.08 | Katty | Dovid: therefor, we went back to 1.4 |
| 14:26.16 | Titanous | no audio when the remote phone initiates the call |
| 14:26.30 | Daejeo | Katty: are you dating M? |
| 14:26.31 | kerx | any suggested asterisk server optimization guides? Can't find much w/ Google searches on those keywords. Any suggestions or references would be highly appreciated. |
| 14:26.50 | Mark17 | if sip calls come in on the sip trunk it should connect to the mobile phone and send the number to the mobile phone to call |
| 14:26.56 | DJ_HaMsTa | lets say i have asterisk set up on my server fully functional, where can i get a number so people can call me? |
| 14:27.04 | Mark17 | if calls come in on the mobile phone it should connect them to the sip trunk |
| 14:27.41 | Mark17 | DJ_HaMsTa: for example at voipbuster, but there are many more options |
| 14:28.19 | Titanous | Daejeo: Grandstream GXP-2000 |
| 14:28.28 | Titanous | crappy phone, nice backlight |
| 14:28.47 | DJ_HaMsTa | Mark17 voipbuster, how much would it cost a month for the service ? |
| 14:29.22 | Dovid | Katty: were there any other issues ? I need it just to do SIP -> H.323 |
| 14:29.26 | DJ_HaMsTa | ah its free |
| 14:29.33 | Mark17 | DJ_HaMsTa: i dont know, see voipbuster.com |
| 14:29.37 | Dovid | on 1.4 it crashed way to often. seems to be many fixes to it for 1.6.X |
| 14:29.39 | DJ_HaMsTa | even better then what im paying now, ($5 a month) |
| 14:29.49 | kerx | Anyone seen any improvement performances using multiple context's in the dialplan that are duplicated? |
| 14:30.03 | *** part/#asterisk fcois93 (n=Administ@bagnolet.acropolistelecom.net) |
| 14:30.05 | kerx | Anyone know if it's possible to load 2 different Asterisk instances on one high end piece of hardware for high performance gain? |
| 14:30.26 | kerx | Any suggestions or guidance to Asterisk Optimization would be appreciated, thanks in advance. |
| 14:30.40 | jsmith | kerx: It's certainly possible, but you may see better results from loading 2 different Asterisk instances on two different pieces of hardware |
| 14:31.17 | kerx | jsmith, Thanks, I've taken this into consideration, at this time we are not in capacity with our current datacenter to be able to grow out our current Rack |
| 14:31.30 | kerx | I am trying to squeeze as much performance possible with 1 Full rack and high-end equipment |
| 14:31.59 | kerx | I've noticed that Asterisk doesn't seem to be squeezing the juice out of a single server, similar to let's say a bogged down Apache web server |
| 14:32.01 | Katty | Dovid: never got that far. |
| 14:32.07 | Katty | Dovid: so not sure. |
| 14:32.09 | *** join/#asterisk jer (n=jer@unaffiliated/jer) |
| 14:32.19 | Katty | Daejeo: also, no. |
| 14:32.29 | kerx | Also, has anyone seen 1.4 > 1.6 for really basic dialplan functionality? I've heard this rumoring a bit... |
| 14:32.35 | jsmith | kerx: In a nutshell, you can start asterisk with the -C option to point it at a different set of config files. The second instance would obviously need to listen on different network ports, etc. |
| 14:32.43 | jsmith | kerx: No, I haven't. |
| 14:33.00 | kerx | I see, I think this method would not be a bad idea... |
| 14:33.13 | kerx | What do you think about the diff. context's (duplicated) |
| 14:33.16 | jsmith | kerx: If that is indeed the case (and you can reproduce it), then it's worth filing a bug |
| 14:33.18 | *** join/#asterisk telnettech (n=bsimpson@d192-24-95-65.col.wideopenwest.com) |
| 14:33.20 | kerx | I've heard some stuff regarding linked lists, etc.. |
| 14:33.34 | jsmith | kerx: I didn't understand the question regarding duplicated contexts |
| 14:34.09 | kerx | Well, I'm not too familiar with the internals of Asterisk, I wasn't really able to find a laymens guide to it without going into the C Code (which I can't do unfortunately) |
| 14:34.22 | Daejeo | Is it possible to have both a busy and an away message when the call waiting feature is enabled? |
| 14:34.46 | kerx | I did lots of archived mailing list searches and noticed that it may be possible to use separate context's to avoid deadlocks, or locking problems that Asterisk is known for |
| 14:34.56 | kerx | I just don't know if this is fact, or false statement |
| 14:35.30 | kerx | I also checked out this brief explanation: http://www.voip-info.org/wiki/index.php?page=Asterisk+ss7+internals |
| 14:35.53 | kerx | I noticed each dialplan has it's own thread |
| 14:36.24 | Katty | Daejeo: whatcha mean by call waiting? |
| 14:36.31 | jsmith | kerx: I don't see how having separate contexts would avoid deadlocking. |
| 14:36.35 | *** join/#asterisk fskrotzki (n=fskrotzk@host198.textwise.com) |
| 14:36.39 | Katty | hugs fskrotzki |
| 14:37.01 | kerx | jsmith, Ok, I will do more research, are you certain about this? |
| 14:37.07 | jsmith | kerx: That being said, every reproducible deadlock has been fixed (as far as I know), and the code for dealing with deadlock avoidance has been greatly improved |
| 14:37.22 | kerx | What about this Linked List issue I've heard about before? |
| 14:37.34 | kerx | respect to jsmith |
| 14:37.37 | jsmith | kerx: Well, who's to say I know anything more than the next guy... I've just been doing Asterisk stuff for 6 years |
| 14:37.49 | kerx | Oh, just for 6 years |
| 14:38.06 | kerx | I see... I guess I'll ask the other thousands of people who have done it for hrmm.. let's say 20 years or so :P |
| 14:38.20 | kerx | ;) |
| 14:38.26 | kerx | Thanks jsmith |
| 14:38.28 | jsmith | kerx: Many of the structures in the core of Asterisk were linked lists in Asterisk 1.2 and earlier. In Asterisk 1.6, many of the structures were converted to hash tables, etc. |
| 14:38.49 | kerx | Oh, so it must be performance improvements from 1.2 , 1.4 to 1.6? |
| 14:39.04 | jsmith | kerx: Many of the changes between 1.4 and 1.6 were simply making the internal plumbing work better at higher loads, etc. |
| 14:39.06 | kerx | I have used 1.4 in production environment and with 300 channels I received high load and lots of call failures |
| 14:39.28 | jsmith | kerx: Absolutely... in some cases, we saw more than a 10x improvement in certain areas |
| 14:39.34 | Dr-Linux|home | anybody is using SRV record with Asterisk? |
| 14:39.35 | kerx | Many outgoingspoolfailed |
| 14:39.46 | Katty | jsmith: ya got two years on me ;) |
| 14:39.48 | kerx | I researched outgoingspoolfailed it seems like this can be performance related |
| 14:39.50 | *** join/#asterisk codefreeze-lap (n=murf@72.21.67.40) |
| 14:39.55 | Katty | codefreeze-lap: ohai |
| 14:39.58 | kerx | Katty, you have 3years and 1 month on me ;) |
| 14:40.00 | Katty | codefreeze-lap: apparently, hai means shark. |
| 14:40.07 | kerx | damn |
| 14:40.08 | kerx | sorry |
| 14:40.14 | kerx | I meant 3years and 11 month's |
| 14:40.17 | kerx | heh |
| 14:40.23 | Katty | hehe |
| 14:40.24 | tuxx- | Katty: idd, in dutch haai = shark ;-) |
| 14:40.25 | kerx | awake 2 long |
| 14:40.48 | kerx | jsmith, Appreciate it... I will migrate over to 1.6 |
| 14:40.53 | jsmith | kerx: For example, see http://www.asterisk.org/node/112 |
| 14:40.54 | kerx | BTW, big note to anyone |
| 14:41.04 | kerx | if you use Asterisk Realtime MySQL extensions |
| 14:41.09 | kerx | you will have lots of performance degradation |
| 14:41.17 | jsmith | coughs.... *ODBC* |
| 14:41.20 | kerx | hehe |
| 14:41.24 | kerx | slaps himself |
| 14:41.38 | jsmith | prefers ODBC and/or PostgreSQL |
| 14:41.49 | Daejeo | Katty: *70 Activate Call Waiting (deactivated by default) |
| 14:41.50 | kerx | jsmith, what type of performance improvements have you seen from 1.2/1.4 to 1.6 as far as channel capacity ? |
| 14:42.23 | jsmith | kerx: It completely depends on the situation. VoIP channels? TDM channels? Analog channels? Doing any transcoding? Doing call recording? Doing audio conferencing? |
| 14:42.52 | *** join/#asterisk mbranca (n=matteo@93-62-230-243.ip24.fastwebnet.it) |
| 14:43.06 | jsmith | kerx: There are too many variables to give you a simple answer, but in general, my experience is that 1.6 seems to be able to handle about 20% higher number of calls in general than say 1.2 |
| 14:43.07 | kerx | jsmith, sending SIP, basic dialplan for audio, AMD and call transfering (typical call-center operation) |
| 14:43.41 | kerx | i wish there was a way for me strip out ton's of the un-necessary functionality that this specific environment doesn't need out of Asterisk |
| 14:43.43 | jsmith | Ah... call centers... how I love (to hate!) thee... |
| 14:43.54 | kerx | i know.. |
| 14:43.58 | Katty | Daejeo: i don't think i've ever used that. |
| 14:44.02 | jsmith | kerx: Well, one easy way is to not use "autoload" in modules.conf |
| 14:44.06 | kerx | i am learning quite a bit at the expense of people who buy from them though :) |
| 14:44.15 | jsmith | kerx: and only load the modules you need |
| 14:44.20 | kerx | this is what is keeping me from avoiding the truth :) |
| 14:44.34 | kerx | knowledge is fun |
| 14:44.38 | kerx | voip is really fun |
| 14:44.39 | Katty | vodka is good for that too |
| 14:44.40 | lmadsen | don't use the mysql-addons stuff for anything -- I've already had at least 2 clients who have had constant crashing issues at high load, who saw them miraculously disappear when switching to ODBC and func_odbc (if using MYSQL() application) |
| 14:44.46 | jsmith | kerx: Also, depending on your architecture, try removing CDRs (or at least batching them, turning off the generation of CSV CDRs, etc.) |
| 14:44.47 | kerx | katty, heh |
| 14:44.51 | lmadsen | Katty: I have some of that in the freezer! :) |
| 14:44.57 | Katty | lmadsen: i'll be right over. |
| 14:45.04 | kerx | jsmith, good idea |
| 14:45.04 | lmadsen | Katty: I'll be here! |
| 14:45.13 | *** join/#asterisk subdolus (n=subdolus@subby.afraid.org) |
| 14:45.25 | *** mode/#asterisk [+o awk_r] by lmadsen |
| 14:45.25 | kerx | i will strip autoload and more from modules.conf and i can even disable cdr because another device will do CDR |
| 14:45.33 | kerx | do you know if System() is expensive ? |
| 14:45.36 | jsmith | kerx: If you're playing a lot of sound files, make sure they're in the proper formats, to avoid transcoding |
| 14:45.51 | kerx | I converted the audio into 8000 using sox |
| 14:45.51 | jsmith | kerx: It can be, as it fires off a shell each time it's run |
| 14:45.53 | neurosys | is there a real improvement in 64bit over 32bit within asterisk? |
| 14:46.08 | jsmith | neurosys: Not a huge difference |
| 14:46.09 | Katty | wonddddddderrrrr boyyyyyy, what is the secret of your power... |
| 14:46.10 | kerx | Yes, I have lots of System() to create my own real-time reporting |
| 14:46.30 | lmadsen | kerx: huh? why not write to a DB using func_odbc? |
| 14:46.32 | Katty | does vodka get songs out of your head? |
| 14:46.36 | lmadsen | that operation has to be much cheaper |
| 14:46.36 | neurosys | jsmith: ok. thx :) |
| 14:46.37 | jsmith | kerx: What are you running from System()? Can you use something like FastAGI or func_odbc instead? |
| 14:46.46 | lmadsen | jsmith: great minds think a like! |
| 14:46.56 | lmadsen | jsmith: I'm just sorry to say it's my mind you're thinking like |
| 14:47.00 | jsmith | lmadsen: Yeah.... we should write a book or something |
| 14:47.03 | telnettech | whois jsmith |
| 14:47.03 | Katty | great minds are in a collective? |
| 14:47.04 | lmadsen | jsmith: heck ya |
| 14:47.05 | kerx | it was easiest for me to use like System(/path/to/perl_script.pl actionMade phoneNumber timeStamp) |
| 14:47.15 | jsmith | telnettech: I'm just a clone of lmadsen |
| 14:47.17 | lmadsen | kerx: that should be a FastAGI |
| 14:47.21 | jsmith | ~jsmith |
| 14:47.21 | jbot | i heard jsmith is perpetually hungry, or the co-author of Asterisk: The Future of Telephony |
| 14:47.27 | Titanous | I have a remote Aastra phone connecting over the internet to Asterisk. When I call the remote phone from the internal Asterisk network, I get full two-way audio. When the Aastra phone initiates the call, there is no audio in either direction. |
| 14:47.27 | lmadsen | ~blitzrage |
| 14:47.28 | jbot | rumour has it, blitzrage is a super cool fellow |
| 14:47.31 | Titanous | All rtp.conf ports are forwarded to Asterisk, and all packets from Asterisk are forwarded directly to the phone by (NAT) firewall |
| 14:47.32 | lmadsen | ~lmadsen |
| 14:47.33 | jbot | hmm... lmadsen is dreamy http://blogs.dfw.com/startle_grams/images/leif75_4.jpg |
| 14:47.36 | Titanous | Suggestions? |
| 14:47.40 | lmadsen | oh ya, I always forget about that, lol |
| 14:47.49 | jsmith | Titanous: A firewall in between the devices isn't passing your RTP audio along |
| 14:48.14 | Titanous | jsmith: why can I call it with full audio but not vice versa? |
| 14:48.39 | lmadsen | because of the FW on the other end |
| 14:48.42 | jsmith | Titanous: Because firewalls are typically more generous is passing outbound RTP than they are in receiving RTP audio |
| 14:48.52 | jsmith | Titanous: Here, let me try to explain |
| 14:48.57 | lmadsen | the FW will like it better when you initiate the call |
| 14:49.12 | jsmith | Titanous: Let's say you have an Asterisk box, which I'll call Alice, out on the public internet (not behind a firewall) |
| 14:49.28 | jsmith | Titanous: And then let's say you have a phone, called Bob, sitting behind a firewall. |
| 14:49.31 | telnettech | sorry im new to this IRC chat and am playing with the options |
| 14:49.44 | jsmith | Titanous: And let's say you're using the SIP protocol. With me so far? |
| 14:49.57 | kerx | neurosys, I see two articles regarding your question: |
| 14:49.57 | kerx | http://www.trixbox.org/forums/trixbox-forums/open-discussion/64-bit-asterisk-trixbox |
| 14:49.58 | Titanous | jsmith: yeah |
| 14:50.01 | kerx | http://www.mail-archive.com/asterisk-users@lists.digium.com/msg132567.html |
| 14:50.09 | Titanous | jsmith: I think it must be the firwall on the asterisk end |
| 14:50.18 | lmadsen | trusts nothing on the trixbox forums |
| 14:50.28 | jsmith | Titanous: Stay with me, and then I'll explain. OK, Bob goes to make a call to Alice. |
| 14:50.35 | Katty | http://howto.wired.com/wiki/Make_Cake_in_a_Mug <- Pure. Joy. |
| 14:50.36 | lmadsen | remember that RTP is in a port range outside of the SIP signaling port |
| 14:50.36 | neurosys | kerx: Thanks! |
| 14:50.51 | jsmith | Titanous: Bob's phone sends Alice a message on UDP port 5060 and says "Hey, let's talk!" |
| 14:50.52 | Daejeo | if i deactivate Call Waiting then caller can hear the busy message |
| 14:50.57 | Titanous | lmadsen: yeah I've got like 10000-20000 forwarded to asterisk |
| 14:51.00 | kerx | neurosys, It seems like it will, especially for transcoding and audio formats |
| 14:51.11 | kerx | neurosys, the others here can more confidently answer this :) |
| 14:51.29 | Katty | jsmith: uh oh. you're doing Kat-like descriptions now. |
| 14:51.31 | jsmith | Titanous: But in that message, Bob also says "Hey, I support the ulaw codec, and I'll be listening for your audio on port 12345", where 12345 is a random high-numbered port |
| 14:51.37 | jsmith | Katty: Scary, isn't it |
| 14:51.42 | Katty | jsmith: i knowes. |
| 14:52.13 | jsmith | Titanous: Alice says "Ok, that's great. I'll listen for your audio on port 24242" (again, where 24242 could be any available high-numbered port) |
| 14:52.41 | Katty | giggles. |
| 14:52.45 | jsmith | Titanous: Bob sends his audio to Alice's port 24242, and since Bob's firewall lets outbound UDP out on high-numbered ports, it gets to Asterisk just fine |
| 14:53.19 | jsmith | Titanous: Alice starts sending her audio to Bob, but Bob's firewall blocks the audio, as it doesn't realize that it needs to allow that audio in and forward it back to Bob's phone. |
| 14:54.22 | jsmith | Titanous: Hence, Alice can hear Bob, but Bob cannot hear Alice. |
| 14:54.50 | jsmith | Titanous: Stick a firewall in front of Alice, and you have double the problems, as now Alice's firewall is blocking Bob's audio, and Bob's firewall is blocking Alice's audio. |
| 14:55.06 | lmadsen | ugh... the light off the lake is blinding me |
| 14:55.42 | Katty | lmadsen: post gifs. |
| 14:55.46 | jsmith | Titanous: And sure... in a perfect world you could just go open the entire RTP range on both firewalls, but that's just a security nightmare waiting to happen. You might as well leave your door unlocked at night as well, and leave money strewn around your front lawn |
| 14:55.48 | Katty | lmadsen: optionally, png. |
| 14:56.10 | jsmith | lmadsen: Well, that's what you get for being so bright... close the blinds so you don't blind the neighbors |
| 14:56.33 | Titanous | jsmith: What's weird here is that I'm using a variety of providers (Gizmo, etc) with incoming calls/audio working fine, but there is no audio coming through on either end if the Aastra initiates the call (SIP is coming through fine). Calls from Asterisk to the Aastra work great |
| 14:56.43 | lmadsen | jsmith: I have no curtains in my living room :) |
| 14:56.53 | lmadsen | is waiting for christmas when his mom makes him some |
| 14:57.01 | jsmith | lmadsen: I wonder what your neighbors think.... |
| 14:57.05 | *** join/#asterisk wwalker (n=wwalker@mailbox.eclipsing.com) |
| 14:57.13 | jsmith | lmadsen: On second thought, I don't wanna be grossed out, so I'm not going there |
| 14:57.16 | *** join/#asterisk IsUp (n=nocturne@unaffiliated/isup) |
| 14:57.25 | lmadsen | jsmith: luckily most of them are in offices, so they are not home at night :) |
| 14:57.37 | Titanous | jsmith: is it possible that the Aastra remote phone is using a RTP port to send audio to that isn't in rtp.conf? |
| 14:57.39 | lmadsen | or rather... most of them are offices, so no one is there at night |
| 14:57.56 | jsmith | Titanous: Absolutely! The phone can't go out and read rtp.conf on the Asterisk box |
| 14:58.09 | jsmith | Titanous: rtp.conf *only* controls what ports Asterisk uses... |
| 14:58.18 | Titanous | jsmith: to send? |
| 14:59.29 | jsmith | Titanous: Or to receive... |
| 14:59.48 | IsUp | any ideas about 3G video calls over SS7? |
| 14:59.55 | Katty | lmadsen: your mom makes you curtains? |
| 15:00.03 | Titanous | jsmith: so why are all the other providers working, but not the aastra phone? Do they negotiate what ports to use? |
| 15:00.31 | lmadsen | Katty: my mom has made me quilts before |
| 15:00.40 | wwalker | Hi. My inbound and outbound calls work fine, but when someone in the office sets their Polycom to forward, and it is called, it sends a 302 to asterisk and asterisk makes an outbound call that has no audio either direction. I've tried to compare a tcpdump of the referred call to a working outbound but am not seeing the problem |
| 15:00.41 | Katty | lmadsen: aww (= |
| 15:00.46 | lmadsen | Katty: sorry... I have no idea where my mini-usb cable is... so can't get these pics off my cam |
| 15:00.52 | Katty | kk |
| 15:00.57 | lmadsen | Katty: she's very good at interior decorating and making things |
| 15:01.00 | wwalker | Anyone know what usually causes this rather than me starting at a sip trace all day? |
| 15:01.02 | jsmith | Titanous: Again, I think it's a problem with the firewall between your Asterisk box and the Aastra |
| 15:01.14 | Katty | lmadsen: oh. so is she the one that decorated your place? |
| 15:01.21 | lmadsen | Katty: I once went away for 3 days to a friends house, and came back to a new deck on the front of our house (she's also very good with power tools) |
| 15:01.32 | lmadsen | Katty: no, I decorated it |
| 15:01.35 | Katty | oh. |
| 15:01.36 | Katty | k'then |
| 15:01.43 | lmadsen | I get it from my mom I guess :) |
| 15:01.53 | Katty | mayhaps! |
| 15:02.00 | lmadsen | she helped pick out the paint colours though... actually went with a theme idea from my dad |
| 15:02.01 | Katty | you can come redecorate my house next. |
| 15:02.05 | lmadsen | heh |
| 15:02.05 | file | all I got was cleaning... |
| 15:02.17 | lmadsen | Katty: it'll cost ya :) |
| 15:02.19 | Katty | file: http://howto.wired.com/wiki/Make_Cake_in_a_Mug |
| 15:02.24 | Katty | lmadsen: i'll make you spaghetti. |
| 15:02.31 | file | Katty: eh! |
| 15:02.44 | Katty | file: optionally, muffins. |
| 15:03.43 | Katty | lmadsen: right now i just want some white sheer snowflake curtains for the upstairs den |
| 15:03.51 | Katty | lmadsen: something sparkly |
| 15:04.58 | *** join/#asterisk nicox (n=nicox@dsl-82-106.utaonline.at) |
| 15:06.18 | neurosys | We used to make cakes in mugs in prison |
| 15:07.05 | nicox | i'm searching for a solution of a problem, is anyone open to help out a little bit? |
| 15:07.19 | lmadsen | ~ask |
| 15:07.19 | jbot | i heard ask is Questions in the channel should be specific, informative, complete, concise, and on-topic. Don't ask if you can ask a question first. Don't ask if a person is there; just ask what you intended to ask them. Better questions more frequently yield better answers. We are all here voluntarily or against our will. |
| 15:07.55 | neurosys | lol |
| 15:08.14 | nicox | oh yeah, and in 99% noone answers |
| 15:08.24 | lmadsen | maybe no one knows the answer |
| 15:08.43 | [TK]D-Fender | nicox: And asking to ask is SO much better. |
| 15:08.52 | lmadsen | #asterisk-dev is not tier 2 or 3.14159 support |
| 15:08.54 | [TK]D-Fender | nicox: so just spit it out already :p |
| 15:09.00 | [TK]D-Fender | mmmmmmmm PIE! |
| 15:09.05 | neurosys | I love PI |
| 15:09.09 | lmadsen | poon-tang flavoured? |
| 15:09.17 | lmadsen | has gone too far... again |
| 15:09.48 | *** join/#asterisk anthm (n=anthm@freeswitch/developer/anthm) |
| 15:09.48 | nicox | i have some asterisk-machines which speak IAX between them, and soem of the calls between the boxes are rejected |
| 15:10.01 | nicox | and i have no idea why |
| 15:10.07 | *** join/#asterisk ocnarf (n=chatzill@125.252.90.5) |
| 15:10.09 | Katty | hugs anthm |
| 15:10.10 | nicox | so, where can i start to debug? |
| 15:10.28 | lmadsen | iax debug ? |
| 15:10.59 | lmadsen | review your configuration files and make sure they are calling with the right authentication? |
| 15:11.05 | ocnarf | hi everyone, may i ask if anyone had experience "UNREACHABLE" on sip peers? |
| 15:11.24 | lmadsen | ocnarf: means the peer isn't responding to our NOTIFY |
| 15:11.28 | ocnarf | this happens to me like every 5mins or less |
| 15:11.33 | lmadsen | (which might mean the peer isn't seeing it) |
| 15:11.47 | ocnarf | what do you think maybe the culprit? |
| 15:11.55 | lmadsen | firewall/nat? |
| 15:11.59 | lmadsen | almost always |
| 15:12.10 | ocnarf | hmmm.. |
| 15:12.19 | nicox | there are thousand working calls |
| 15:12.26 | ocnarf | first it lagged then UNREACHABLE then goes back to REACHABLE again |
| 15:12.27 | nicox | and then there is a call which is rejected |
| 15:12.40 | nicox | so the configuration i think is not the problem |
| 15:12.52 | lmadsen | nicox: I got that the other day -- usually means the other end didn't respond to a critical packet |
| 15:12.52 | WimpMan | bandwidth? |
| 15:13.12 | lmadsen | nicox: look at the iax debug trace between the two calls and see what is different |
| 15:13.35 | lmadsen | I bet you don't get a response to a NEW when you should |
| 15:14.09 | nicox | lmadsen iax debug will be complicated because of 80+ concurrent calls |
| 15:14.16 | lmadsen | nicox: I'm answering here... respond here pls |
| 15:14.21 | ocnarf | lmadsen: so this could be all on the network? not on asterisk? |
| 15:14.29 | lmadsen | nicox: use wireshark so you can filter |
| 15:14.40 | lmadsen | ocnarf: most likely it is the network and not asterisk |
| 15:14.57 | ocnarf | lmadsen: thanks for that info |
| 15:15.00 | nicox | its a gigabit-switched local network... |
| 15:15.16 | nicox | but, a good thing to start |
| 15:15.34 | lmadsen | of course |
| 15:15.52 | *** join/#asterisk SibRphrek (i=SibrPhre@cpe-67-243-43-136.nyc.res.rr.com) |
| 15:16.15 | nicox | or the destination server is overloaded, is that also possible? |
| 15:16.39 | lmadsen | anything is possible |
| 15:16.46 | lmadsen | except for the Leafs winning the Stanley Cup |
| 15:16.48 | Katty | wow. |
| 15:17.03 | Katty | a girl here at work, just called me up to ask how to put pictures on myspace. |
| 15:17.15 | Katty | this is abuse of my position |
| 15:17.48 | lmadsen | indeed |
| 15:18.33 | *** join/#asterisk SibRphrek (i=SibrPhre@cpe-67-243-43-136.nyc.res.rr.com) |
| 15:19.00 | *** join/#asterisk gr0mit (n=tim@dhcp3.zuk40.mot-tools.co.uk) |
| 15:19.22 | nicox | thanks a lot for the idea! |
| 15:19.38 | *** join/#asterisk SibRphrek (i=SibrPhre@cpe-67-243-43-136.nyc.res.rr.com) |
| 15:21.15 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 15:21.55 | nicox | lmadsen: oh wait, also the destination asterisk says in the log ost 10.x.y.z failed to authenticate as xxx |
| 15:23.18 | *** join/#asterisk jameswf (n=james@dsl093-157-131.phx1.dsl.speakeasy.net) |
| 15:23.25 | *** join/#asterisk mindCrime (n=chatzill@216.27.62.2) |
| 15:26.39 | *** join/#asterisk Karlitoo (n=proscom@213.137.110.67) |
| 15:27.00 | nicox | any idea on this? |
| 15:27.30 | *** join/#asterisk esaym (n=user@cpe-70-120-89-6.satx.res.rr.com) |
| 15:27.49 | lmadsen | no idea |
| 15:28.21 | *** join/#asterisk SuPrSluG (n=SuPrSluG@72.237.213.162) |
| 15:28.33 | SuPrSluG | hello |
| 15:28.56 | Katty | ohai |
| 15:28.59 | *** join/#asterisk ddunavant (n=David@75.145.240.14) |
| 15:29.07 | Katty | sometimes, i feel like the door greeter at walmart. |
| 15:29.09 | Karlitoo | I have a little problem with channals any ideas http://pastebin.com/d2700ed7b |
| 15:29.47 | nicox | H323 is not registered in asterisk |
| 15:29.54 | nicox | check if the module is loaded |
| 15:30.03 | lmadsen | nicox: you keep bouncing between channels... pls just keep the discussion here |
| 15:30.17 | nicox | or which module you are using for H323 |
| 15:32.01 | nicox | lmadsen: yeah, different problems, different channels of course |
| 15:32.16 | lmadsen | none of your problems are #asterisk-dev related |
| 15:33.13 | nicox | hm, the bug-report is not asterisk-dev related? |
| 15:33.42 | lmadsen | #asterisk-bugs |
| 15:33.58 | Karlitoo | you were right nicox I don't have that module loaded |
| 15:33.59 | nicox | oh, great, so i will spam there :-) |
| 15:33.59 | Karlitoo | :) |
| 15:34.01 | *** join/#asterisk UnixDawg (n=UnixDawg@cpe-98-28-149-231.cinci.res.rr.com) |
| 15:34.04 | SuPrSluG | i can't make calls between extensions |
| 15:34.07 | Karlitoo | sry I'm a asterisk newb |
| 15:34.11 | UnixDawg | ok having a issue with asterisk |
| 15:34.22 | nicox | no prob., your welcome |
| 15:34.29 | UnixDawg | it seems 1.4.22 is not phrasing correctly |
| 15:34.37 | *** join/#asterisk korihor (n=korihor@201.210.239.172) |
| 15:34.41 | *** join/#asterisk agx (n=AGX@host63-216-static.34-88-b.business.telecomitalia.it) |
| 15:34.57 | UnixDawg | I have outbound matching to 1NXXNXXXXXX 1+NXXNXXXXXX |
| 15:34.57 | agx | hi, the TE120P does support only 75 ohm or has a jumper for serting 120 ohm? |
| 15:35.23 | UnixDawg | and if you dial a 10 digit number you get all curcuts are busy |
| 15:35.24 | SuPrSluG | i am on our private net and have the * box on a public. it sees all extension coming from the same address. http://pastebin.ca/1260447 any ideas? |
| 15:36.20 | coppice | agx: why would the card do 75 ohms? they only do that when they have a coax termination. twisted pair is normally 110 to 120 ohms |
| 15:36.40 | agx | coppice, the default looks 75 ohm |
| 15:36.54 | coppice | measured how? |
| 15:37.23 | agx | coppice, connecting to another pbx with 75 ohm work, with 120 noes |
| 15:37.47 | coppice | that's not a measure of impedance :-) |
| 15:38.39 | Karlitoo | for implementing asterisk with avaya over h323 should I use the oh323, ooh323c or woomera |
| 15:38.44 | agx | coppice, so you mean the card's own default is 120 ohm not 75? |
| 15:39.58 | *** join/#asterisk mog (n=mog@nat/digium/x-71d11feb5c78c174) |
| 15:39.58 | *** mode/#asterisk [+o mog] by ChanServ |
| 15:39.59 | coppice | cards with RJ48C connectors are normally 110-120, and may have a hi-impedance option, for tapping purposes. 75 ohms is used for coax, which you won't find used for T1, but often find used for E1 |
| 15:40.33 | WimpMan | E1 in coax??? |
| 15:41.03 | coppice | at one time all E1 was coax |
| 15:41.43 | WimpMan | Ok, I'm only in that business for nine years. |
| 15:42.14 | coppice | well, 9 years ago coax was still very common for E1 |
| 15:42.44 | Karlitoo | which h323 channel driver for asterisk is compatible with 1.6 |
| 15:42.50 | WimpMan | Never seen such a thing. Usually TP and occasionally glas. |
| 15:42.54 | agx | coppice, its E1 and i usually connect it to a cisco using a normal ethernet cablwe |
| 15:43.32 | coppice | WimpMan: I guess that's the kind of sheltered life wimps lead |
| 15:43.50 | WimpMan | Then I must be lucky :-) |
| 15:44.22 | WimpMan | I have seen Coax interfaces on NASes but never a fitting line. |
| 15:44.35 | agx | coppice, is a 5 euro tester enough to test the impedance? :) |
| 15:45.20 | coppice | agx: its usually a pure resistance, so any old meter can check it |
| 15:45.59 | SuPrSluG | when i dial from the console all calls go to the same extension. bizarre |
| 15:46.27 | SuPrSluG | 3 extens regged and all go to the same phonel |
| 15:46.32 | SuPrSluG | er phone |
| 15:47.26 | agx | coppice, thanks, lets go to the supermarket :) |
| 15:50.22 | *** join/#asterisk murdock_ut (n=chatzill@mail.kimballequipment.com) |
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| 15:51.48 | *** mode/#asterisk [+o Deeewayne] by ChanServ |
| 15:58.12 | *** join/#asterisk rwaite (n=richard@rrcs-74-218-125-86.central.biz.rr.com) |
| 15:58.30 | Katty | [TK]D-Fender: http://www.petsmart.com/product/index.jsp?productId=3135694 |
| 15:59.18 | *** join/#asterisk rwaite (n=richard@rrcs-74-218-125-86.central.biz.rr.com) |
| 16:06.05 | *** join/#asterisk osiris (n=osiris@c-71-205-27-88.hsd1.mi.comcast.net) |
| 16:09.31 | *** join/#asterisk Ritzerisk (n=Ritztech@nv-65-40-156-46.sta.embarqhsd.net) |
| 16:10.05 | [TK]D-Fender | Katty: I see your imploding economy has curbed your consumerism :) |
| 16:10.25 | Katty | [TK]D-Fender: you have no idea. did you know they have a police dog outfit? |
| 16:10.36 | Katty | plots evil things. |
| 16:10.41 | [TK]D-Fender | Katty: I wouldn't doubt it |
| 16:10.57 | Katty | [TK]D-Fender: i need an authentic seeing eye dog in training jacket. |
| 16:11.08 | Katty | [TK]D-Fender: that way puppeh can come with me to regular stores ;P |
| 16:11.50 | *** join/#asterisk ChkDigit (n=mike@static24-72-71-175.regina.accesscomm.ca) |
| 16:12.06 | Qwell | Katty: mmhmm |
| 16:12.20 | Katty | Qwell: still down :< /tear |
| 16:13.03 | Qwell | supposed to stay up until 3am on Monday nights! |
| 16:13.36 | Katty | :< |
| 16:18.09 | *** join/#asterisk anonymouz666 (n=anonymou@201.19.199.65) |
| 16:19.21 | Katty | hugs anonymouz666 |
| 16:19.27 | *** join/#asterisk putnopvut (n=putnopvu@nat/digium/x-d91de4d1a36c624c) |
| 16:19.27 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 16:20.57 | [TK]D-Fender | Katty: And get you busted, fined or worse |
| 16:21.12 | Katty | [TK]D-Fender: probably worse ;) |
| 16:21.40 | anonymouz666 | Katty! |
| 16:22.53 | *** join/#asterisk c4t3l (n=c4t3l@c-98-200-2-241.hsd1.tx.comcast.net) |
| 16:23.05 | c4t3l | hello world |
| 16:25.02 | wwalker | Hi. My inbound and outbound calls work fine, but when someone in the office sets their Polycom to forward, and it is called, it sends a 302 to asterisk and asterisk makes an outbound call that has no audio either direction. I've tried to compare a tcpdump of the referred call to a working outbound but am not seeing the problem |
| 16:25.03 | neurosys | [TK]D-Fender: Hehe just found out there is no seperate SMP kernel for centos anymore. Its all included in kernel-devel. ugh :) |
| 16:25.03 | Katty | ello. |
| 16:25.37 | c4t3l | howdy! |
| 16:29.00 | *** join/#asterisk c4t3l (n=c4t3l@c-98-200-2-241.hsd1.tx.comcast.net) |
| 16:29.07 | c4t3l | wow that was fun! |
| 16:29.34 | Katty | rehi. |
| 16:29.55 | Ritzerisk | anyone know how to unlock the web admin access for a linksys 2102 sipura |
| 16:30.39 | c4t3l | hmmm... nope. |
| 16:31.02 | *** join/#asterisk colulu (n=jg@58.251.78.185) |
| 16:31.15 | neurosys | Probably a good ole fashion reset will work ;-D |
| 16:31.25 | c4t3l | you can try to dial in thru the IVR and activate it. If i remember its 80# or somesuch |
| 16:31.44 | Katty | wow, obama rhinestone shirts for dogs. |
| 16:32.07 | neurosys | nice! My pit has his xmas gift |
| 16:32.29 | Katty | what'd you get him for christmas? |
| 16:32.38 | *** join/#asterisk SuPrSluG (n=SuPrSluG@72.237.213.162) |
| 16:33.15 | neurosys | I meant I'll be getting him the obama gear ;) |
| 16:33.20 | Katty | oh ; |
| 16:33.22 | Katty | ;) |
| 16:33.34 | wwalker | he'd prefer a leg of mailman |
| 16:33.56 | Ritzerisk | haha its a China with italkbb but i need to edit some settings and use it to work with my system via Fxs port |
| 16:33.57 | neurosys | Funny part is.. on South Beach, he may prefer the shirt |
| 16:34.06 | Katty | i just want a plain red shirt for riddick. |
| 16:34.11 | *** join/#asterisk Segnale007 (n=Pietro@host21-255-dynamic.7-87-r.retail.telecomitalia.it) |
| 16:34.13 | Katty | can't find one. |
| 16:34.28 | Katty | found a blue one. but not red )= |
| 16:36.49 | Katty | ooh! i found one! |
| 16:38.38 | *** join/#asterisk nighty^ (n=nighty@x122091.ppp.asahi-net.or.jp) |
| 16:39.29 | *** join/#asterisk fcois93 (n=Administ@bagnolet.acropolistelecom.net) |
| 16:39.46 | *** join/#asterisk demonist (n=hi@jstickland.ca) |
| 16:40.04 | fcois93 | I need help with gotoif. with goto I can do goto(incoming,s,1) can I do the same with gotoif ? |
| 16:40.06 | colulu | does anyone use any good hardware gateway? |
| 16:40.38 | demonist | Hello, i dont know of any other telephony related channels, so i will ask here: does anyone know the correct term for a loop connecting two sites point to point, using the telcos wires. |
| 16:41.13 | demonist | like, not going onto the pstn |
| 16:41.16 | [TK]D-Fender | "trunk" sometimes "LAN extension" |
| 16:41.17 | Katty | fcois93: exten => s,1,GotoIfTime(8:00-17:00|mon-fri|*|*?weeeeeeeeee,s,1) |
| 16:41.31 | demonist | but just a circuit connecting one NID to another NID |
| 16:41.32 | Carlos_PHX | demonist: Point to point T1? |
| 16:41.36 | demonist | nope, not T1 |
| 16:41.41 | demonist | sort of on an analog line still |
| 16:41.46 | demonist | no pstn on it |
| 16:41.57 | demonist | just a pair running from location a to b, using the telcos wires |
| 16:42.18 | demonist | so i can use my own dslam and dsl modem on that loop |
| 16:42.24 | Katty | fcois93: exten => h,1,GotoIf($["${Variable}" != ""]?yessssssssss,s,1:nooooooooooooo,s,1) |
| 16:42.30 | Carlos_PHX | Can't do that. |
| 16:42.30 | Carlos_PHX | Won't work. |
| 16:42.35 | Katty | fcois93: etc. |
| 16:42.48 | Carlos_PHX | The amps and intermediary devices will make DSL not work. |
| 16:42.48 | demonist | Carlos, are you refering to what im talking about? |
| 16:43.11 | Carlos_PHX | I mean, I suppose a really short haul with nothing in between might, but probably not. |
| 16:43.14 | Carlos_PHX | demonist: Yes |
| 16:43.34 | demonist | so theres no such thing as connecting two houses together using telco pairs |
| 16:43.41 | demonist | for a private circuit |
| 16:43.54 | demonist | with a dslam on one end, and a modem on the other end. |
| 16:44.02 | Carlos_PHX | Yes there is in some places, but putting DSL on it is another thing. |
| 16:44.11 | demonist | strange....i read an article that this is okay |
| 16:44.14 | Carlos_PHX | I don't think it's even generally available to get a dry loop. |
| 16:44.24 | demonist | and the manufacturer of the dslam said its okay too |
| 16:44.27 | demonist | is on a dry loop |
| 16:44.46 | Carlos_PHX | I'd love to read the article, because knowing about how difficult it can be to make DSL work on engineered circuits, I can't picture it working ad-hoc. |
| 16:44.57 | jsmith | demonist: It *can* work, but won't *necessarily* work |
| 16:44.59 | demonist | well |
| 16:45.08 | demonist | how about asking the phone company |
| 16:45.10 | jsmith | has seen it work before |
| 16:45.16 | Carlos_PHX | Right, it can, but I believe is unlikely. |
| 16:45.18 | jsmith | has seen it fail before too |
| 16:45.19 | demonist | "can you connect two loops together at the frame |
| 16:45.23 | demonist | no pstn, no dslam" |
| 16:45.32 | demonist | just NID to NID |
| 16:45.38 | [TK]D-Fender | demonist: Ah : Leased Line |
| 16:45.48 | Carlos_PHX | Sounds like a fun project. |
| 16:46.00 | demonist | then, at one NID, have a DSLAM |
| 16:46.04 | demonist | a mini IP dslam |
| 16:46.10 | demonist | at the other nid, dsl modem |
| 16:46.48 | demonist | thing im wondering though...where do we get the power for the pair...would my dslam provide that... |
| 16:46.51 | [TK]D-Fender | demonist: Don't really need a DSLAM |
| 16:46.58 | SuPrSluG | i have a phone going out from our private net to our * server with a public. I have 3 extensions created. 1 line per phone. sip show peers shows all 3 registered although only 1 phone is powered up. all from the same address ( the firewall). why won't each phone register by itself? |
| 16:47.13 | demonist | no, not for a point to point...but if i got more and more circuits a mini dslam might make more sense |
| 16:47.22 | demonist | economic sense |
| 16:47.23 | [TK]D-Fender | demonist: If you effectively ahve a straight par you can just get 2 Sangoma S519 's or something... |
| 16:47.35 | demonist | ok |
| 16:47.35 | [TK]D-Fender | demonist: Yeah, depending how you scale. |
| 16:47.58 | *** join/#asterisk Tuxguy (n=jimi@rrcs-70-63-90-226.midsouth.biz.rr.com) |
| 16:48.03 | demonist | tk, so i would ask for a leased line? |
| 16:48.20 | Tuxguy | When I start asterisk and try to connect a softphone locally, I do not get a connection, and I do not see the connection attempting in CLI w/ sip set debug on |
| 16:48.52 | Katty | goes to lunch |
| 16:49.07 | demonist | feels like banging on the central office door and asking for their input |
| 16:49.18 | demonist | it would probably be quicker to get information from them than talking to sales |
| 16:49.28 | demonist | sales would probably want to say "okay, heres your t1" |
| 16:49.30 | [TK]D-Fender | tuxeither a networking, firewall or configuration issue |
| 16:50.00 | Tuxguy | The asterisk box is on the same machine as the softphone. The bind address = 0.0.0.0 |
| 16:50.01 | [TK]D-Fender | demonist: Most telcos offer LAN Extension options through their network. |
| 16:50.12 | *** join/#asterisk subdolus (n=subdolus@subby.afraid.org) |
| 16:50.41 | [TK]D-Fender | Tuxguy: Then THAT is likely the prblem. Your softphone needs to be told to use 5061, not 5060 for tis port, and you need to set your peer entry to reflect that |
| 16:52.20 | neurosys | [TK]D-Fender: Hmm. Just finished getting asterisk on centos. Isnt the sip modules already on by default? |
| 16:52.40 | [TK]D-Fender | neurosys: HUH? |
| 16:53.36 | Tuxguy | Xlite doesnt give me an option to set the port. |
| 16:53.44 | neurosys | [TK]D-Fender: Well. copied over my basic sip.conf and extensions.conf files.. but its not working for some reason. |
| 16:53.52 | [TK]D-Fender | Tuxguy: then pick one that does |
| 16:54.11 | [TK]D-Fender | neurosys: Guess you'd better find something of substance to show us. |
| 16:54.35 | [TK]D-Fender | neurosys: "its not working" = doesn't help us help you. |
| 16:54.43 | neurosys | [TK]D-Fender: yeah. Pretty vauge huh. Im digging.. |
| 16:55.09 | [TK]D-Fender | neurosys: Go verify that chan_sip is loaded. check your peers, check your netowrking, check your FIREWALL. And of course show us your SIP debug of comm attempts |
| 16:55.38 | [TK]D-Fender | neurosys: Describe in detail what yuo are TRYING that is failing. What is calling what? |
| 16:57.21 | neurosys | [TK]D-Fender: SIP is loaded. sip.conf has a peer configured. Same one that worked previously. called the ITSP #, but it simply rings, then the ITSP times it out.. The extensions.conf file is also the old one that worked previosly. |
| 16:57.48 | neurosys | [TK]D-Fender: There are no warnings or errors in the SIP debug |
| 16:58.17 | [TK]D-Fender | neurosys: Gee and no more infor if NAT is involved, SIP debug of your register attempts, info about your router if any, etc. |
| 16:58.38 | [TK]D-Fender | neurosys: Please try to be COMPLETE in your breakdon of how you are set up.... |
| 16:58.44 | [TK]D-Fender | PASTEBIN IS YOUR FRIEND |
| 16:58.46 | [TK]D-Fender | ~pb |
| 16:58.47 | 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, or , http://bin.cakephp.org/ |
| 16:58.48 | [TK]D-Fender | ^^^^^^^^^^^ |
| 16:59.08 | neurosys | [TK]D-Fender: The box is in a DMZ. One the same exact machine it worked on before i installed CentOS. |
| 16:59.29 | [TK]D-Fender | neurosys: MORE meaningless info |
| 16:59.43 | [TK]D-Fender | neoShow configs, SYSTEM SETTING, firewall dumps, etc |
| 16:59.45 | *** join/#asterisk CunningPike (n=arodgers@204.239.8.157) |
| 17:00.51 | *** part/#asterisk nicox (n=nicox@dsl-82-106.utaonline.at) |
| 17:05.54 | neurosys | [TK]D-Fender: gee. CentOS comes with a preconfigured firewall. |
| 17:09.20 | *** join/#asterisk hi365_m (n=hi365@213.151.38.153) |
| 17:12.49 | *** join/#asterisk jaytee (n=jforde@unaffiliated/jaytee) |
| 17:14.23 | *** join/#asterisk mgdm (n=michael@river.mgdm.net) |
| 17:14.32 | *** part/#asterisk mgdm (n=michael@river.mgdm.net) |
| 17:14.39 | *** join/#asterisk dandre (n=daniel@was59-3-82-236-48-30.fbx.proxad.net) |
| 17:16.02 | *** join/#asterisk outtolunc (n=me@adsl-66-218-53-172.dslextreme.com) |
| 17:20.14 | *** join/#asterisk esaym (n=user@cpe-70-120-89-6.satx.res.rr.com) |
| 17:22.35 | *** join/#asterisk Spirits-Sight (n=christop@c-24-218-128-159.hsd1.ma.comcast.net) |
| 17:23.25 | *** join/#asterisk root52 (n=f745082a@ip70-191-120-39.cl.ri.cox.net) |
| 17:24.37 | Spirits-Sight | [TK]D-Fender: I have been reading the book, and wow, some things seem to be easy to understand and other thing not, I am trying to setup a very very basic dail plain for voice pulse and not sure what to do, I have downloaded the once they give but it full of stuff I don't know about and have not learned yet, can you please help me so I can make out going calls on one ext so I can get rid of the one I have right now |
| 17:25.02 | [TK]D-Fender | ~jerjerguide |
| 17:25.03 | jbot | [~jerjerguide ] Jeremy McNamara's quick sample guide to setting up * : http://www.jeremy-mcnamara.com/2007/02/26/how-to-configure-asterisk-your-first-installation/ |
| 17:25.14 | root52 | Quick ? I am sure I have seen this before but just can not remeber. If I want to wrap on line in * config file down to the next line. Is there a special char. Or will * just keep proccessing? |
| 17:25.28 | [TK]D-Fender | Spirits-Sight: There is a good minimal system for a different provider. The basics should be incredibly quick to adapt |
| 17:25.42 | *** join/#asterisk joako (n=joako@adsl-074-170-252-213.sip.gnv.bellsouth.net) |
| 17:25.44 | [TK]D-Fender | root52: no such thing as wrapping |
| 17:25.55 | root52 | ok that answers that Thanks!! |
| 17:25.59 | [TK]D-Fender | root52: this isn't bash, perl, etc |
| 17:26.08 | Spirits-Sight | I am going confured with the outbound stuff, thanks for the information |
| 17:26.18 | joako | I keep on seeing Remote UNIX connection / Remote UNIX connection disconnected on one server... what can cause this? |
| 17:26.21 | [TK]D-Fender | root52: *'s parsers are the dumbest we could find/hack/make/steal |
| 17:26.32 | [TK]D-Fender | joaAMI or CLI process connecting. |
| 17:26.39 | root52 | :-) |
| 17:26.57 | [TK]D-Fender | joako: You should already know what processes related to * you have installed on your system. |
| 17:27.17 | joako | Yes... but there is no reason for it to be doing that... |
| 17:27.30 | *** join/#asterisk marc7 (n=marc@S0106001c1024382d.gv.shawcable.net) |
| 17:27.51 | *** join/#asterisk ElSonico (n=tav@dondo.ampiainen.net) |
| 17:28.29 | [TK]D-Fender | joako: Yes there is... because of software YOU installed and enabled |
| 17:28.58 | jaytee | or he's been p0wned |
| 17:29.25 | joako | So when the other machines connect via IAX that message is normal? |
| 17:29.50 | [TK]D-Fender | joako: no |
| 17:31.57 | ricko73 | I believe I've found a bug in parkandannounce |
| 17:32.39 | ricko73 | if a call is parked, the external party is able to initate a transfer by pressing the transfer feature key |
| 17:32.53 | Spirits-Sight | [TK]D-Fender: when I have a choice between IAX and SIP which one is the better one to use and why? |
| 17:33.25 | [TK]D-Fender | Spirits-Sight: IAX has been known to have issues with audio quality, trunked calls when the trunk fails, you lose all calls. |
| 17:33.30 | *** join/#asterisk erth64net (n=erth64ne@mail.opensourcery.com) |
| 17:33.32 | [TK]D-Fender | Spirits-Sight: Basically use IAX if you HAVE TO. |
| 17:33.54 | Spirits-Sight | [TK]D-Fender: you got it, thanks very much |
| 17:34.03 | [TK]D-Fender | ricko73: Which feature key? |
| 17:34.25 | ricko73 | the client has * configured in features.conf |
| 17:34.30 | [TK]D-Fender | ricko73: please share a pastebin of a complete sample call from CLI |
| 17:35.22 | ricko73 | I'll have to redo the call with the client in 20-30 minutes. The receptionist has to get the owner packed up and out the door |
| 17:36.56 | jaytee | [TK]D-Fender must have modeled his business after Milo Minderbinder's M&M Enterprises in Catch-22 because he spends all his time in here helping people for free yet still manages to somehow make a living. :-) |
| 17:37.25 | neurosys | Love that book. |
| 17:38.34 | jaytee | the part where Milo makes a deal where they and the Germans each bomb their own runways and sell the aviation fuel they would have used on the black market always cracked me up. |
| 17:38.46 | jaytee | Global capitalism at it's finest. |
| 17:39.30 | *** join/#asterisk ManxPower (n=manxpowe@78.sub-70-220-220.myvzw.com) |
| 17:44.19 | *** part/#asterisk dandre (n=daniel@was59-3-82-236-48-30.fbx.proxad.net) |
| 17:45.36 | *** join/#asterisk Tuxguy (n=root@rrcs-70-63-90-226.midsouth.biz.rr.com) |
| 17:46.28 | Tuxguy | I just did a netstat -ua and netstat -ta and did not see anything binding to 5060, although, asterisk is running and accesible through the CLI |
| 17:47.26 | giovani | Tuxguy: netstat will say "sip" rather than 5060 |
| 17:47.32 | giovani | you sure you didn't just skip overi t? |
| 17:47.44 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 17:47.48 | ManxPower | Try this: netstat -an | grep 5060 | grep udp |
| 17:48.00 | giovani | you can do a netstat -unap | grep 5060 |
| 17:48.01 | giovani | heh |
| 17:48.08 | Katty | so, i was driving back to work from lunch, and this SUV alaskan plates on it was in front of me. I thought about running it off the road on the off-chance that it was the governor (or family) and i could get some free clothes out of the deal. |
| 17:48.10 | giovani | ManxPower: no reason to grep on udp -- that's what -u is for |
| 17:48.18 | Tuxguy | udp 109308 0 0.0.0.0:5060 0.0.0.0:* 30800/xtensoftphone |
| 17:48.20 | ManxPower | giovani: Ah |
| 17:48.26 | ManxPower | Tuxthere you go |
| 17:48.27 | Tuxguy | Looks like my softphone has it, but is over powering it... even though its not running |
| 17:48.31 | Tuxguy | Maybe it didnt close properly |
| 17:48.45 | giovani | it's clearly running |
| 17:49.42 | Tuxguy | damn |
| 17:49.55 | giovani | just issue kill -9 30800 |
| 17:50.01 | giovani | it's probably a hung process |
| 17:50.04 | Tuxguy | thanks, i did |
| 17:50.09 | Tuxguy | That made asterisk work when i reloaded it |
| 17:50.14 | giovani | :) |
| 17:50.36 | Katty | file: beans and bytes > * |
| 17:51.37 | Tuxguy | So I need to use 5061 for my softphone then, and just set that up in the sip.conf for that usrename |
| 17:51.40 | Tuxguy | username |
| 17:52.58 | Katty | [TK]D-Fender: ping? |
| 17:53.34 | [TK]D-Fender | Katty: pong? |
| 17:53.38 | giovani | Tuxguy: you shouldn't need to reconfigure asterisk to do that |
| 17:53.44 | Katty | [TK]D-Fender: ohai. you didn't nack |
| 17:54.05 | Katty | [TK]D-Fender: tango, much? |
| 17:54.07 | [TK]D-Fender | Katty: no, this was an ACK. resource found! |
| 17:54.15 | [TK]D-Fender | Katty: Horizontal mambo ;) |
| 17:54.20 | Katty | haha |
| 17:54.28 | ManxPower | Tuxguy: you only need to set the SOURCE port on the phone, not the DEST port |
| 17:55.09 | jaytee | an ACK is better than a NACK but still not as good as a SNACK |
| 17:55.11 | Tuxguy | I dont see that option in XLITE, do you know of a way to do it, or can recommend another softphone for linux? |
| 17:55.44 | [TK]D-Fender | Tuxguy: Ekiga |
| 17:56.01 | giovani | Tuxguy: twinkle? |
| 17:56.06 | Tuxguy | ty |
| 17:56.19 | jaytee | Ekiga? isn't that the Finnish word for vomit? |
| 17:56.32 | giovani | http://www.xs4all.nl/~mfnboer/twinkle/index.html |
| 17:56.50 | giovani | soft phones generally suck |
| 17:56.54 | giovani | I've yet to see a really nice one |
| 17:57.10 | Katty | maybe i should've named Riddick Ekiga |
| 17:57.20 | jaytee | Office Communicator......? (ducks) |
| 17:57.34 | jaytee | does the puppy puke up alot? |
| 17:57.41 | Katty | only in the car :/ |
| 17:58.00 | Tuxguy | ekiga doesnt allow to change the source port |
| 17:59.51 | jaytee | some people only take their dog in the car when they take them to a vet so the dog gets stressed and carsick. if you take the dog someplace fun like a park to play and then back home several times then they don't automatically associate being in the car with a trip to the vet. |
| 18:00.12 | giovani | running a softphone on the same system as asterisk is definitely not a common setup |
| 18:00.55 | Tuxguy | Well, for testing I dont really have another option. |
| 18:01.10 | giovani | twinkle does allow you to set the SIP port |
| 18:01.24 | Tuxguy | I can't install twinkle yet, stupid dependency issues. |
| 18:01.27 | jaytee | if you run X-lite with defaults you just have to ensure that you start X-lite after Asterisk has started so * will grab 5060 and xlite will grab 5061 |
| 18:01.32 | Tuxguy | Mixing RPMs + source :D |
| 18:02.04 | kaldemar | jaytee: finnish word for vomit? where did you come up with that one? :D |
| 18:02.52 | jaytee | kaldemar, I dunno. just kinda sounded finnish and Ekiga IS vomit as far as softphones go so...... there ya have it in a nutshell |
| 18:03.17 | [TK]D-Fender | Tuxguy: Ekiga The listen ports : The main port listening for incoming connections in Ekiga for SIP is port 5060 (UDP), while 1720 (TCP) is used by H.323. To change those ports you need to load "gconf-editor". Open gconf-editor, select apps from the left hand side menu and then select Ekiga. Then select "sip" or "h323", it should give you a list in the corresponding window to your right.... |
| 18:03.19 | [TK]D-Fender | ...Select listen_port and change it to your desired value. You can also change the UDP/RTP port ranges. |
| 18:03.42 | [TK]D-Fender | Tuxguy: Next time actually use the HELP menu to look for settings your want to change when you can't see where |
| 18:03.56 | *** join/#asterisk nosbig (n=nosbig@cpe-75-185-59-24.columbus.res.rr.com) |
| 18:03.56 | kaldemar | sorry if your world falls to pieces, but ekiga isn't either vomit in or anything else in finnish. :) |
| 18:04.01 | Tuxguy | Thanks |
| 18:04.11 | *** join/#asterisk km2 (n=x@mobile-166-217-013-009.mycingular.net) |
| 18:04.12 | Tuxguy | It should be an option under network settings or something else though. |
| 18:04.42 | Tuxguy | bbiab, testing |
| 18:04.45 | [TK]D-Fender | Tuxguy: Superfluous. Next time spend more than 5 seconds looking. |
| 18:05.06 | *** join/#asterisk Ericounet (n=Ericoune@ACaen-151-1-37-217.w86-215.abo.wanadoo.fr) |
| 18:05.37 | [TK]D-Fender | Amazing how fast I found it hardly ever having used it and only jsut configured it on my work server jsut now and using Putty + xming on my desktop. |
| 18:05.48 | [TK]D-Fender | Katty: You run Xming? |
| 18:06.19 | Katty | [TK]D-Fender: no |
| 18:06.34 | [TK]D-Fender | Katty: you should, its the shiznit y0! |
| 18:06.42 | Katty | [TK]D-Fender: wai |
| 18:06.45 | *** join/#asterisk fskrotzki_ (n=fskrotzk@cpe-74-74-245-250.rochester.res.rr.com) |
| 18:07.17 | [TK]D-Fender | Katty: run X-apps on your PC via PuTTY |
| 18:07.19 | [TK]D-Fender | Katty: http://sourceforge.net/projects/xming |
| 18:07.30 | Katty | ooo |
| 18:07.32 | [TK]D-Fender | Katty: Great for "direct on server" wireshark, etc |
| 18:07.42 | *** join/#asterisk shazaum (n=shazaum@unaffiliated/shazaum) |
| 18:07.46 | [TK]D-Fender | Katty: just enable SSH forwarding in your PuTTY entry and voila |
| 18:07.59 | Katty | sounds super duper |
| 18:08.02 | [TK]D-Fender | loves SSH tunneling too |
| 18:08.15 | ricko73 | [TK]D-Fender: I just verified the bug on my local system (the ability to initiate a transfer externally) |
| 18:08.34 | [TK]D-Fender | ricko73: And after all this time you still have nothing to show us for it? |
| 18:08.47 | ricko73 | I'm putting the pastebin together |
| 18:08.50 | ricko73 | keep your pants on |
| 18:09.51 | *** join/#asterisk StephenF (n=none@198.144.201.106) |
| 18:10.25 | *** join/#asterisk etfonhomey (n=chatzill@www2.askpri.org) |
| 18:10.43 | Katty | fender? pants on? |
| 18:10.52 | Katty | i don't think i'm even going to make a comment. |
| 18:10.57 | *** join/#asterisk protocols (n=protocol@p5791FD5D.dip.t-dialin.net) |
| 18:11.03 | protocols | hi all |
| 18:11.38 | protocols | is there a difference in receiving faxes from an isdn fax machine and an analog fax machine? |
| 18:11.57 | ricko73 | http://pastebin.com/m6fe749e4 |
| 18:12.03 | *** join/#asterisk Assimilate (n=Assimila@72.22.242.66) |
| 18:13.13 | [TK]D-Fender | mambo #5! |
| 18:13.31 | *** join/#asterisk sack (n=sack@62.Red-81-33-114.dynamicIP.rima-tde.net) |
| 18:13.33 | Katty | one two three four five |
| 18:13.37 | Katty | everybody in the car |
| 18:13.38 | Katty | oh |
| 18:13.42 | Katty | i'm going to stop now |
| 18:14.52 | lou_gr | does anyone know openvox products? are they compatible with asterisk? |
| 18:14.54 | [TK]D-Fender | ricko73: interesting. |
| 18:15.01 | Qwell | lou_gr: define "compatible" |
| 18:15.05 | ricko73 | yeah, that's not what the client thinks |
| 18:15.21 | ricko73 | [TK]D-Fender: I believe their exact words were 'toyish' |
| 18:15.32 | ricko73 | or amatuer |
| 18:15.36 | ricko73 | I forget ;) |
| 18:15.41 | [TK]D-Fender | lou_gr: Yes, but they are knockoffs of old Digium designs and the warranty is spotty. They may work, or they may suck , and worse still if they suck and you have problems getting them replaced, repaired, or returned |
| 18:15.46 | lou_gr | Qwell, work properly |
| 18:15.49 | Qwell | then no |
| 18:16.38 | *** join/#asterisk lanning (n=lanning@66.151.128.195) |
| 18:17.05 | lou_gr | can you suggest me please similar pci cards? |
| 18:17.35 | protocols | or are there any suggestions why I can't receive faxes from some sources |
| 18:18.05 | ricko73 | lou_gr: if you are looking at Openvox because of price, then you get what you pay for |
| 18:18.47 | ricko73 | in my opinion, you're better off spending the money for either Digium or Rhino cards and be able to talk with good tech support people on the phone |
| 18:18.49 | lou_gr | am not looking cheap products. But I 'm intresting in the pci interface |
| 18:18.50 | Qwell | lou_gr: Digium. |
| 18:18.59 | Qwell | There are other vendors as well, but I'm slightly biased. |
| 18:19.13 | ricko73 | Qwell: really? who would have guessed ;) |
| 18:19.17 | Qwell | lou_gr: There are 2 (or 3, if jameswf is around) "major" vendors. |
| 18:19.28 | ricko73 | (but who can blame you--they do sign your pay check) |
| 18:19.28 | Qwell | well, 3 that do pci. |
| 18:19.32 | Qwell | ricko73: indeed |
| 18:19.44 | lmadsen | s/check/cheque/g |
| 18:19.51 | Qwell | lmadsen: Get out. |
| 18:19.56 | lmadsen | nevah! |
| 18:19.58 | ricko73 | plus the newer Digium cards do not have the same issues that the original TDM400 cards did |
| 18:20.08 | lmadsen | new chipset w00t |
| 18:20.38 | ricko73 | so what's the best way to report this fancy new bug I found? |
| 18:20.42 | Qwell | bugs.digium.com |
| 18:20.46 | ricko73 | Mailing list or is there a bug tracker... |
| 18:20.46 | lmadsen | bugs.digium.com? :) |
| 18:20.47 | ricko73 | ko |
| 18:20.55 | lou_gr | thank you. |
| 18:21.01 | lmadsen | however I'm closing all new bugs because asterisk has none |
| 18:21.19 | jtodd | closes lmadsen |
| 18:21.25 | lmadsen | shuts his mouth |
| 18:21.35 | colulu | from a resource utalization perspective, is it a big difference between PCI card and external gateway? |
| 18:21.37 | lmadsen | goes back to testing bugs! |
| 18:21.52 | colulu | I am looking for ways to bridge PSTN |
| 18:21.54 | lmadsen | I'm just waiting for asterisk to compile on the VM |
| 18:21.55 | Qwell | colulu: what do you mean by external? |
| 18:22.09 | colulu | like a physical hardware such as audio codes |
| 18:22.18 | Qwell | like a SIP gateway? |
| 18:22.21 | colulu | yes |
| 18:22.43 | [TK]D-Fender | colulu: External gateway is virtually no load unless * has to transcode the call |
| 18:22.45 | Qwell | sure, but it's not cheap for a quad PRI box |
| 18:23.02 | colulu | if I put it inside the same box using PCI card, then it reduce the resource for the socket connection. Does that make much a difference for resource utilization and voice quality? |
| 18:23.04 | [TK]D-Fender | colulu: Its also a great way to deal with redundency, etc |
| 18:23.18 | [TK]D-Fender | colulu: Quality should not be an issue |
| 18:23.43 | [TK]D-Fender | colulu: it all gets turned to VoIP at some point, shouldn't really matter whose side |
| 18:23.58 | [TK]D-Fender | colulu: Which is the better choice depends on your specific needs, budget, etc |
| 18:24.06 | colulu | TK: so the two options do not make any difference in terms of voice quality and # of concurrent channels right? |
| 18:24.10 | *** join/#asterisk axisys (n=axisys@155.70.141.45) |
| 18:24.15 | [TK]D-Fender | colulu: What kind & size of install are you looking at? |
| 18:24.30 | colulu | TK: i am looking for a 8 E1 |
| 18:24.45 | colulu | but I can also have 2 PCI cards or one audio codes |
| 18:24.48 | Qwell | colulu: that is going to be a fair bit more expensive :) |
| 18:25.09 | [TK]D-Fender | colulu: PCI solution often off-load EC and other things to the host CPU therefore external gateways naturally would allow you more calls ont he same base CPU with transcoding not being a factor, etc |
| 18:25.16 | Qwell | here's the way I see it (and I'm sure [TK]D-Fender will disagree): |
| 18:25.21 | [TK]D-Fender | colulu: Or 1 PCI card. |
| 18:25.24 | *** join/#asterisk apeiron (n=apeiron@c-76-124-253-149.hsd1.pa.comcast.net) |
| 18:25.34 | Qwell | If you're only using a few channels, (say 4-12), the load is very low |
| 18:25.53 | Qwell | if you're using a lot of channels (say 12-24 or more), the external box will get very expensive |
| 18:25.54 | apeiron | Hm, is this a help channel or should I go elsewhere? |
| 18:25.56 | [TK]D-Fender | Qwell: He's lookint at 8 E1 <- :) |
| 18:25.59 | Qwell | ~ask |
| 18:25.59 | jbot | well, ask is Questions in the channel should be specific, informative, complete, concise, and on-topic. Don't ask if you can ask a question first. Don't ask if a person is there; just ask what you intended to ask them. Better questions more frequently yield better answers. We are all here voluntarily or against our will. |
| 18:26.05 | Qwell | [TK]D-Fender: that's gonna be one expensive gateway... |
| 18:26.09 | apeiron | Qwell, Cool, thanks. |
| 18:26.09 | [TK]D-Fender | apeiron: Depends what you want help with :) |
| 18:26.18 | [TK]D-Fender | Qwell: Oh hell yeah :) |
| 18:26.26 | Qwell | SO, yeah... PCI :p |
| 18:26.28 | jameswf | only 2 u.s. companies |
| 18:26.32 | [TK]D-Fender | Qwell: and I don't disagree withyou :) |
| 18:26.40 | Qwell | [TK]D-Fender: for 8 channels, you'd go PCI? |
| 18:26.52 | [TK]D-Fender | colulu: What codec are your calls going to be in? |
| 18:27.01 | jameswf | is working on getting in to the automotive supply chain so i can grab some bail out bucks |
| 18:27.07 | apeiron | I've got videoconferencing set up in my Ubuntu 1.4.21 setup, as well as voicemail. I own the machine and the account where the voicemail is stored. Can I view the video part somehow? |
| 18:27.11 | colulu | 711 most probably |
| 18:27.48 | [TK]D-Fender | colulu: Ok, well PCI could do it without too much of an issue... do make sure its a HWEC card though |
| 18:28.20 | apeiron | What I'm trying to do is use something like totem or mplayer to view the voicemail without having to go through Asterisk to view it. |
| 18:28.33 | ricko73 | ran into a nightmare last night at a new client |
| 18:28.41 | Qwell | apeiron: video conferencing? O.o |
| 18:28.45 | apeiron | Qwell, Yes. |
| 18:28.47 | [TK]D-Fender | apeiron: This is #asterisk. We support ASTERISK here. |
| 18:28.52 | ricko73 | they have a 'phone system' that runs on top of Windows XP! |
| 18:28.55 | apeiron | okay, well |
| 18:29.03 | [TK]D-Fender | apeiron: We do not know about your software OR the service you are connecting to. |
| 18:29.06 | ricko73 | can't wait for it to totally crash and die so I can replace it |
| 18:29.06 | Qwell | apeiron: it's just a file on the fs. look in /var/spool/asterisk/voicemail/ |
| 18:29.14 | apeiron | [TK]D-Fender, Asterisk / QuteCom. |
| 18:29.18 | [TK]D-Fender | Qwell: Apparently its not on * |
| 18:29.23 | Qwell | O.o |
| 18:29.25 | apeiron | Qwell, I tried playing it, file format isn't recognized. |
| 18:29.28 | *** join/#asterisk neurosys (n=neurosys@c-66-229-81-161.hsd1.fl.comcast.net) |
| 18:29.32 | colulu | TK: from Asterisk's perspective, does it process faster and use less resource if the data is obtained directly from PCI card vs an external server via SIP ( for a local lan case)? |
| 18:29.33 | [TK]D-Fender | Qwell: At least, thats what it reads as |
| 18:30.00 | ratmandu | <PROTECTED> |
| 18:30.16 | [TK]D-Fender | colulu: for stright ULAW routing (* will BE a gateway), you could be OK, but I'd recommend a beefy CPU and fill up the ram to 4g |
| 18:30.18 | *** join/#asterisk donnib (n=donnib@0x555281d0.adsl.cybercity.dk) |
| 18:30.18 | apeiron | [TK]D-Fender, I enabled videoconferencing in my sip.conf. I enabled voicemail in voicemail.conf. I set up a dialplan in extensions.conf. I called myself. Left a voicemail with video. I want to view it. |
| 18:30.23 | jameswf | mplayer plays asterisk VM |
| 18:30.24 | apeiron | [TK]D-Fender, How is this not an Asterisk question? |
| 18:30.26 | donnib | hi |
| 18:30.29 | [TK]D-Fender | apeiron: Better |
| 18:30.30 | apeiron | jameswf, Ah, cool. |
| 18:30.49 | apeiron | hm. totem just must not have the codec for it. |
| 18:30.50 | [TK]D-Fender | apeiron: Your wording left it in question. Good to clarify. |
| 18:31.08 | apeiron | [TK]D-Fender, Sorry for the ambiguity, then. |
| 18:31.10 | [TK]D-Fender | apeiron: You want to download the VM as a video file? |
| 18:31.25 | donnib | i have a freepbx setup where all clients are on the same network. some clients are in india and some are in europe where the server is. i have problems with the clients in india. i get UNREACHABLE. i have a 280 ms latency. |
| 18:31.28 | apeiron | [TK]D-Fender, It's a very small installation that's private between me and a friend. So I'm running a desktop on it too. |
| 18:31.30 | [TK]D-Fender | apeiron: that'd take a mixing app to take the split audio & video codec recording and transcode them |
| 18:31.37 | lmadsen | don't you just login to the voicemail to playback the voicemail w/ video? |
| 18:31.39 | donnib | tried to set QUALIFY to 10000 but didn't help |
| 18:31.41 | [TK]D-Fender | apeiron: * cannot do this, it'd take a bunch of external work to do it. |
| 18:31.42 | apeiron | [TK]D-Fender, Right, yes. |
| 18:31.53 | donnib | how can i find out what the problem is ? |
| 18:31.55 | apeiron | [TK]D-Fender, I understand that * is not a multimedia platform. :) |
| 18:32.00 | [TK]D-Fender | apeiron: but * is effectively out of the picture.. |
| 18:32.02 | lmadsen | donnib: qualify=no |
| 18:32.13 | lmadsen | donnib: it's not an asterisk issue -- its a network issue |
| 18:32.17 | *** join/#asterisk voxter (n=voxter@S01060016b6b53c0c.vc.shawcable.net) |
| 18:32.18 | apeiron | [TK]D-Fender, Okie. Was just wondering if people here had any experience with opening those files directly. Thanks. :) |
| 18:32.19 | donnib | lmadsen: what does Qualify actually mean ? |
| 18:32.25 | [TK]D-Fender | apeiron: Sure its a multimedia platform... so long as you're only working with ne kind of media at a time ;) |
| 18:32.35 | lmadsen | donnib: essentially "ping" |
| 18:32.40 | lmadsen | donnib: it checks the latency to the phone |
| 18:32.42 | donnib | i was actually thinking about the network |
| 18:32.43 | apeiron | [TK]D-Fender, Well I mean it's not like iMovie or something. :) |
| 18:32.46 | lmadsen | to determine if it is available |
| 18:32.55 | donnib | but if if does not get back then it won't call ? |
| 18:32.56 | lmadsen | donnib: it absolutely is the network |
| 18:32.59 | [TK]D-Fender | apeiron: Ok, well It hink we're all clear on the answer here then.. |
| 18:33.00 | donnib | because i can't reach them |
| 18:33.00 | lmadsen | donnib: correct |
| 18:33.04 | donnib | ah ok |
| 18:33.04 | apeiron | [TK]D-Fender, Yes. |
| 18:33.08 | donnib | will try now |
| 18:33.13 | lmadsen | donnib: if UNREACHABLE then asterisk will not attempt to call |
| 18:33.18 | donnib | ok |
| 18:33.20 | lmadsen | if Unmonitored it'll try regardless |
| 18:33.27 | donnib | so is this good practice ? |
| 18:33.29 | [TK]D-Fender | apeiron: You CAN have voicemail.conf trigger a "cleanup" script that will call whatever tool you have configured to do your dirty-work, but its really out of *'s hands |
| 18:33.29 | colulu | TK: sorry, i was a bit lost. so for Asterisk, is it faster to process calls directly via PCI card than via an external sip gateway? |
| 18:33.35 | lmadsen | donnib: it's a practice |
| 18:33.40 | apeiron | [TK]D-Fender, Yeah, I saw that. |
| 18:34.08 | [TK]D-Fender | colulu: I'd probably guess that gatway would be a little faster since pure SIP setup is jsut networking and there is no BUS issue, etc |
| 18:34.21 | lmadsen | donnib: if they are behind a NAT, then it is possible the NAT connections will close on you and you still won't be able to reach them. You'll need the client to register fairly often |
| 18:34.42 | donnib | i am not behind any firewall so that should not be the problem |
| 18:34.51 | lmadsen | you're not.. but the phones might be |
| 18:34.51 | donnib | i am on the same network but different subnets |
| 18:34.59 | donnib | neither are the phones |
| 18:35.08 | lmadsen | ok, then just turn off qualify |
| 18:35.24 | ricko73 | http://bugs.digium.com/view.php?id=13923 |
| 18:35.34 | donnib | ok will do |
| 18:35.38 | donnib | :) |
| 18:36.02 | colulu | TK: with PCI card, Asterisk won't need to utilize resource for socket connection. Do you think that makes any different difference at all? |
| 18:36.08 | [TK]D-Fender | donnib: pastebin your sip.conf masking only passwords. |
| 18:36.09 | [TK]D-Fender | ~pb |
| 18:36.10 | 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, or , http://bin.cakephp.org/ |
| 18:36.12 | [TK]D-Fender | ^^^^ |
| 18:36.40 | colulu | TK: i know I may be paranoid |
| 18:36.48 | [TK]D-Fender | colulu: Trust me, a PCI resource to interface witht he card would be worse than an IP socket. Linux kernel was kinda designed with IP in mind you know ;) |
| 18:37.01 | [TK]D-Fender | colulu: Oh no... we ARE out to get you! ;) |
| 18:38.03 | *** part/#asterisk lmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 18:38.24 | donnib | [TK]D - Fender: but i solved the problem, well at least i think |
| 18:39.13 | donnib | if a client is not behind NAT but in the config of the asterisk it has a filed NAT=yes then will it still work |
| 18:39.14 | donnib | ? |
| 18:39.26 | bmoraca | it depends on the phone |
| 18:39.28 | ricko73 | woah, this bug is even nastier than I thought. I removed all instances of 't or T' in my dial statements and the caller is still able to tranfer a call after it's been parked |
| 18:39.30 | donnib | xlite |
| 18:39.34 | bmoraca | not sure |
| 18:39.37 | bmoraca | couldn't hurt to try |
| 18:39.39 | colulu | TK: thanks alot |
| 18:39.43 | donnib | ok thx |
| 18:39.46 | bmoraca | i know that polycoms work OK in that config |
| 18:39.52 | bmoraca | but i've had problems with Cisco phones |
| 18:39.54 | colulu | TK: you input is great. I know what i need to do now |
| 18:40.04 | donnib | well i will try to see |
| 18:40.11 | *** join/#asterisk lmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 18:40.11 | *** mode/#asterisk [+o lmadsen] by ChanServ |
| 18:40.16 | *** part/#asterisk raasdnil (n=mikel@60-241-138-146.static.tpgi.com.au) |
| 18:40.37 | [TK]D-Fender | donnib: GIANT "depends" on that |
| 18:40.45 | jameswf | cant run asterisk as he has Ubuntu 8.10 |
| 18:41.00 | lmadsen | there used to be an extconfig.txt file in the doc/ directory of the asterisk source in 1.4 that has the table layouts for realtime... but I don't seem to see that file anymore in trunk. Doesn't appear to be in doc/tex/ either... any idea where it went or was renamed to? |
| 18:41.11 | donnib | huh ? |
| 18:41.40 | [TK]D-Fender | donnib: PB your config and I'll tell you what'll happen |
| 18:42.03 | kaldemar | lmadsen: configs/extconfig.conf.sample |
| 18:42.28 | kaldemar | that it? |
| 18:42.40 | lmadsen | kaldemar: no, that doesn't define the table structure |
| 18:42.48 | kaldemar | oh, silly me. |
| 18:43.14 | [TK]D-Fender | jameswf: and why is that? |
| 18:43.25 | donnib | don't have a sip.conf since i am running freepbx where data is in mysql |
| 18:43.40 | lmadsen | uh oh... you said the deadly word |
| 18:43.52 | donnib | i know |
| 18:43.54 | [TK]D-Fender | donnib: FreePBX GENERATES sip.conf |
| 18:43.59 | donnib | i am sorry |
| 18:44.12 | [TK]D-Fender | WTF. |
| 18:44.17 | [TK]D-Fender | No seriously, WTF.... |
| 18:44.22 | [TK]D-Fender | sighs |
| 18:44.28 | donnib | but my question did not have anything to do with freepbx therefore did not mention it |
| 18:44.46 | donnib | ok where is that file ? some temporary place then ? |
| 18:45.05 | donnib | it's gotta be generated from the mysql thingy |
| 18:45.09 | [TK]D-Fender | donnib: FreePBX builds you configs. I don't care if your configs are deposited directly from the CONFIGURATION FAERIE, I just asked to see them. |
| 18:45.11 | jsmith | donnib: If you're using FreePBX, the files it generates are in /etc/asterisk/sip*.conf |
| 18:45.40 | [TK]D-Fender | jsmith: these days you can't even GIVE help away! |
| 18:45.42 | bmoraca | donnib, use winscp to log in to your server and navigate to /etc/asterisk and locate the sip_additional.conf file. that's where your extensions are |
| 18:46.10 | jameswf | [TK]D-Fender: http://bugs.digium.com/view.php?id=13912 |
| 18:46.25 | bmoraca | i'd take all the help I can get resolving my nvfaxdetect issue...though I think my setup fundamentally precludes me from using that application anyway |
| 18:46.46 | jsmith | [TK]D-Fender: No kidding... why do I even try? |
| 18:46.51 | donnib | hey easy now boys |
| 18:46.52 | [TK]D-Fender | ~cluebat jameswf |
| 18:46.52 | jbot | ACTION pulls out a ClueBat (tm) and thwaps jameswf. |
| 18:46.56 | jsmith | wishes he had more time to give away help |
| 18:47.09 | ricko73 | jameswf: duct tape might help |
| 18:47.18 | ricko73 | will contain the explosion |
| 18:47.29 | donnib | i know about sip.conf...i know how to gain access to it but it's not that easy because freepbx includes other files so actually you will need like 10 files before you make sip.conf |
| 18:47.56 | bmoraca | freepbx stores its configuration in mysql, but it doesn't use asterisk realtime |
| 18:48.11 | bmoraca | when you click the "Apply Settings" bar, it writes normal asterisk config files |
| 18:48.29 | *** join/#asterisk Segnale007 (n=Pietro@host218-252-dynamic.18-79-r.retail.telecomitalia.it) |
| 18:48.56 | bmoraca | where it puts its custom SIP extensions is sip_additional.conf |
| 18:49.18 | bmoraca | sip.conf includes sip_*.conf |
| 18:49.33 | donnib | i know |
| 18:49.41 | donnib | and that was what i was trying to say |
| 18:49.47 | bmoraca | right. so the only file you need to show us is sip_additional.conf |
| 18:49.54 | *** join/#asterisk rgrrbbt (n=roger2k1@catv-86-101-104-2.catv.broadband.hu) |
| 18:50.00 | rgrrbbt | hi |
| 18:50.15 | *** join/#asterisk pecanha (n=e@189.106.180.89) |
| 18:50.46 | rgrrbbt | could please anybody help me with asterisk.ctl? |
| 18:50.49 | *** join/#asterisk saftsack (n=oliver@g228010009.adsl.alicedsl.de) |
| 18:51.32 | [TK]D-Fender | bmoraca: Umm... NO |
| 18:51.38 | pecanha | hey guys, I'm using trixbox, my SIP is registered, but when someone calls me it doesn't work. How can I debug and what I need to look for? |
| 18:52.08 | [TK]D-Fender | pecanha: Go to the trixbox or FreePBX site and follow the GUI for getting it to work from behind NAT |
| 18:52.49 | [TK]D-Fender | guide* |
| 18:52.53 | pecanha | [TK]D-Fender: ok, I'll look for it |
| 18:53.19 | bmoraca | eh? |
| 18:53.23 | pecanha | [TK]D-Fender: do you know where this NAT options is located? |
| 18:53.50 | [TK]D-Fender | sip_custom.conf / sip_nat.conf |
| 18:54.00 | [TK]D-Fender | pecanha: Something like that |
| 18:54.15 | [TK]D-Fender | pecanha: Its all in the docs. Go visit their sites. |
| 18:54.28 | [TK]D-Fender | pecanha: I know FreePBX has a decent guide for it, Trixbox should as well. |
| 18:54.41 | [TK]D-Fender | pecanha: Esp as they add layers of BS on top |
| 18:55.14 | *** join/#asterisk Ritzerisk (n=Ritztech@nv-65-40-156-46.sta.embarqhsd.net) |
| 18:55.41 | rgrrbbt | nobody can help with that? |
| 18:55.46 | Ritzerisk | anyone know how to dial out a regular local number locally in hong kong haha weird question though |
| 18:55.54 | donnib | ok here it is http://pastebin.com/d571d17e7 |
| 18:57.13 | bmoraca | that's set with nat=no. if you're connecting locally, you shouldn't have a problem. |
| 18:58.09 | donnib | i connect local on the same network |
| 19:03.32 | [TK]D-Fender | and the peer alone doesn't hold the answer |
| 19:03.49 | [TK]D-Fender | rgrrbbt: ... |
| 19:03.51 | [TK]D-Fender | ~ask |
| 19:03.52 | jbot | rumour has it, ask is Questions in the channel should be specific, informative, complete, concise, and on-topic. Don't ask if you can ask a question first. Don't ask if a person is there; just ask what you intended to ask them. Better questions more frequently yield better answers. We are all here voluntarily or against our will. |
| 19:06.30 | Spirits-Sight | pbx_ael.c:4157 pbx_load_module: AEL load process: verified config file name '/etc/asterisk/extensions.ael'. |
| 19:06.30 | Spirits-Sight | whats this mean? and why? |
| 19:07.17 | codefreeze-lap | Spirits-Sight: means that the AEL module loaded, and most likely read in the default example config file you installed... |
| 19:07.18 | [TK]D-Fender | Spirits-Sight: AEL is an alternative to extensions.conf |
| 19:07.43 | [TK]D-Fender | Spirits-Sight: empty out the contents of extensions.ael if you have no intention of using it |
| 19:08.12 | Spirits-Sight | ok, but why would this load when I have the extensions.conf file |
| 19:08.57 | [TK]D-Fender | Spirits-Sight: and add "noload pbx_ael.so" to modules.conf |
| 19:09.07 | [TK]D-Fender | Spirits-Sight: because you can use BOTH at the same time |
| 19:09.50 | Spirits-Sight | oo so get rid of it or do the noload pbx_ael.so in the modules.conf file with stop it from using that file |
| 19:11.10 | [TK]D-Fender | Spirits-Sight: the noload alone is enough |
| 19:11.32 | Daejeo | ~ask |
| 19:11.33 | jbot | extra, extra, read all about it, ask is Questions in the channel should be specific, informative, complete, concise, and on-topic. Don't ask if you can ask a question first. Don't ask if a person is there; just ask what you intended to ask them. Better questions more frequently yield better answers. We are all here voluntarily or against our will. |
| 19:11.37 | Spirits-Sight | ok thats what I throught just making sure |
| 19:12.01 | Daejeo | ~grandstream |
| 19:12.02 | jbot | well, grandstream is the Yugo of VoIP hardware. Run. Run away now. |
| 19:12.17 | rgrrbbt | if i run asterisk with -c or -f, eveything is fine, but without any options the asterisk.ctl doesn't get created. has anybody an idea why this happens? |
| 19:12.24 | Spirits-Sight | how do I reload now after doing that change |
| 19:12.34 | [TK]D-Fender | Spirits-Sight: completely restart * |
| 19:12.50 | Spirits-Sight | how do I do this |
| 19:13.09 | *** join/#asterisk artemmakhutov (n=artemmak@e181198008.adsl.alicedsl.de) |
| 19:13.20 | [TK]D-Fender | rgrrbbt: You should be using safe_asterisk to start it as a daemon, or via an init script it installed |
| 19:13.29 | [TK]D-Fender | Spirits-Sight: restart now |
| 19:14.14 | Spirits-Sight | cool thanks |
| 19:14.16 | rgrrbbt | same problem with the init script. but asterisk.pid is there in both cases |
| 19:14.34 | [TK]D-Fender | rgrrbbt: who are you starting * as? |
| 19:14.41 | rgrrbbt | root |
| 19:14.50 | rgrrbbt | it's an openwrt router |
| 19:14.56 | [TK]D-Fender | rgrrbbt: still go check your scrip to see which user its calling as, and verify your perms |
| 19:15.27 | artemmakhutov | Hello, is someone familiar with asterisk and TLS? |
| 19:16.21 | rgrrbbt | isn't that the same user who creates .pid and .ctl? |
| 19:16.35 | jsmith | artemmakhutov: Just ask your question, and if somebody knows the answer, they'll shout out an answer |
| 19:16.38 | artemmakhutov | I am trying to setup TLS, but the clients are always getting a 403 - Forbidden error |
| 19:16.40 | *** join/#asterisk oej (n=olle@ns.webway.se) |
| 19:17.05 | *** join/#asterisk Deeewayne (n=Deeewayn@nat/digium/x-547c915ba480e7ce) |
| 19:17.05 | *** mode/#asterisk [+o Deeewayne] by ChanServ |
| 19:17.08 | artemmakhutov | without TLS everything works fine. The certificate seems also to work |
| 19:17.54 | ricko73 | [TK]D-Fender: looks like that bug I just reported has been fixed already (in 1.4.23-rc) |
| 19:17.56 | *** join/#asterisk murdock_ut (n=chatzill@mail.kimballequipment.com) |
| 19:18.15 | [TK]D-Fender | ricko73: Don't worry, your wheel will be rounder ;) |
| 19:18.36 | ricko73 | wheel will be rounder? |
| 19:18.49 | ricko73 | never heard that saying before |
| 19:19.18 | jsmith | artemmakhutov: Does anything show up on the Asterisk CLI? |
| 19:19.44 | jsmith | artemmakhutov: What if you turn on debug messages in logger.conf and then do a "logger reload" and then "core set debug 4" and then try again? |
| 19:19.56 | Spirits-Sight | I can make a call outgoing but I can not call 866 numbers here is my very very simple dail plain (less then two lines) |
| 19:20.00 | [TK]D-Fender | ricko73: An implication of "reinventing the wheel".... |
| 19:20.01 | artemmakhutov | no, nothing, only when I enable the sip debug I can see the communication |
| 19:20.24 | Spirits-Sight | exten => _1NXXNXXXXXX,1,Set(CALLERID(num)=xxxxxxxxxx) |
| 19:20.44 | jsmith | artemmakhutov: If you've turned on debug messages in logger.conf, then you *should* see all kinds of things |
| 19:20.55 | [TK]D-Fender | Spirits-Sight: PASTEBIN is your friend. Use it. |
| 19:20.57 | [TK]D-Fender | ~pb |
| 19:20.57 | 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, or , http://bin.cakephp.org/ |
| 19:21.02 | [TK]D-Fender | Spirits-Sight: show us the failure |
| 19:22.01 | Spirits-Sight | http://pastebin.com/m6d9fd397 |
| 19:22.16 | rgrrbbt | i set /var/run to be writable for everyone but it still doesn't work |
| 19:22.49 | artemmakhutov | I will try, thx |
| 19:24.07 | rgrrbbt | the init script is at http://pastebin.com/d88b03a8 |
| 19:25.43 | Spirits-Sight | [TK]D-Fender: here is the failure http://pastebin.com/m6d9fd397 |
| 19:25.43 | Katty | runs around. |
| 19:25.44 | [TK]D-Fender | Spirits-Sight: thats EVERYTHING? |
| 19:25.49 | Katty | shreds curtains. |
| 19:26.20 | c4t3l | shreds on his guitar |
| 19:26.23 | Spirits-Sight | thats all it says, it says the same thing three times which is how many times I tryed it |
| 19:26.26 | Katty | shredder. |
| 19:26.41 | [TK]D-Fender | Spirits-Sight: then your dialplan is wrong. |
| 19:26.55 | c4t3l | splinter? |
| 19:27.04 | Katty | splinter of turkey. |
| 19:27.11 | Katty | that's what i wanted, when i was 4, around thanksgiving. |
| 19:27.13 | ghento | Hi all. Is there a way to get more information about the execution of an agi script? the one i'm working with doesn't seem to be functioning, and all i see is "AGI Script foo.agi completed, returning 0" |
| 19:27.19 | c4t3l | nice! |
| 19:27.28 | *** join/#asterisk Tuxguy (n=jimi@rrcs-70-63-90-226.midsouth.biz.rr.com) |
| 19:27.29 | Spirits-Sight | I am going to have to finish this issue later I am now leaving |
| 19:28.15 | Tuxguy | Can someone point me at documentation for setting up MGCP SIP devices? Like what d001 is, etc |
| 19:28.30 | rgrrbbt | i cannot understand what is the difference between running with and without -f (besides not forking). with -f, the .ctl file gets created in /var/run, wihtout it, the file is missing |
| 19:28.47 | Katty | c4t3l: http://flickr.com/photos/izaah/3023011327/in/set-72157608822215475/ <- 4ish. |
| 19:29.08 | Katty | c4t3l: don't eve ASK about that dress. |
| 19:29.33 | *** part/#asterisk mog (n=mog@nat/digium/x-71d11feb5c78c174) |
| 19:29.37 | *** join/#asterisk mog (n=mog@nat/digium/x-71d11feb5c78c174) |
| 19:29.37 | *** mode/#asterisk [+o mog] by ChanServ |
| 19:30.22 | c4t3l | lol. every kid has the same kind of pic. neutral background. folded hands and such. |
| 19:30.27 | *** join/#asterisk jameswf (n=james@dsl093-157-131.phx1.dsl.speakeasy.net) |
| 19:30.51 | artemmakhutov | I have found the TLS problem: I had also to set "transport=tls" in the device configuration. Just enabeling it globally is not enough. |
| 19:31.06 | Katty | c4t3l: and an insane dress with BANGS?! |
| 19:31.11 | artemmakhutov | Thx again! |
| 19:31.16 | c4t3l | hehe |
| 19:31.25 | *** join/#asterisk esaym (n=user@cpe-70-120-89-6.satx.res.rr.com) |
| 19:31.49 | esaym | hello,k is there anyway that I can start recording a channel from the asterisk -r command line? |
| 19:31.56 | jameswf | aparently "GNU audio editor" is a command alias for "Crash X" |
| 19:32.24 | *** join/#asterisk ElSonico (n=tav@dondo.ampiainen.net) |
| 19:32.35 | Katty | c4t3l: oh well, happy memories. |
| 19:32.46 | Katty | c4t3l: swinging! chasing ducks around! |
| 19:32.54 | Katty | c4t3l: sesame street! |
| 19:33.27 | c4t3l | god! when i was a kid there was this duck that lived at a pond near my house that would freakin terrorize me |
| 19:34.09 | *** join/#asterisk ManxPower (n=manxpowe@112.sub-75-202-76.myvzw.com) |
| 19:34.11 | Katty | :< |
| 19:34.27 | Katty | gives c4t3l an anti-duck necklace charm |
| 19:34.32 | Katty | :> |
| 19:36.23 | [TK]D-Fender | esaym: No, but you can do it from AMI |
| 19:36.48 | esaym | what is AMI? |
| 19:36.49 | *** join/#asterisk fukz (n=fukz@p5B063D38.dip.t-dialin.net) |
| 19:36.50 | esaym | :-/ |
| 19:36.59 | esaym | oh the interfaced |
| 19:37.01 | esaym | interface |
| 19:37.11 | esaym | hmm, not sure if I have it enabled |
| 19:37.50 | [TK]D-Fender | esaym: It is unless you specifically disabled it |
| 19:38.40 | jameswf | wow oprah is a nut job.. |
| 19:40.15 | *** join/#asterisk ziram19 (n=chatzill@41.226.252.220) |
| 19:41.07 | ziram19 | since one month i have a pb to configure my thomson st2030 |
| 19:41.23 | ziram19 | i need to have a payant consultant |
| 19:42.07 | [TK]D-Fender | ziram19: Pardon? |
| 19:42.35 | ziram19 | u speek frensh TK? |
| 19:43.04 | [TK]D-Fender | ziram19: Yes, but the word is the same in english as well |
| 19:44.29 | StephenF | is there a way to configure the dial by name directory to match first AND last names? |
| 19:44.37 | StephenF | or is it only one or the other |
| 19:44.57 | [TK]D-Fender | StephenF: either/or |
| 19:44.57 | StephenF | nvm, I just found the answer |
| 19:45.03 | StephenF | :) thx |
| 19:45.03 | [TK]D-Fender | (last I read) |
| 19:45.35 | StephenF | oh I just read in the wiki, there is an option to allow both |
| 19:45.43 | StephenF | http://www.voip-info.org/wiki/view/Asterisk+cmd+Directory |
| 19:46.06 | esaym | ok ty |
| 19:46.50 | [TK]D-Fender | StephenF: I wouldn't trust the wiki jsut yet. check the instruction at CLI |
| 19:47.00 | StephenF | alright, will do |
| 19:47.10 | *** join/#asterisk sah-work (n=Bawbatos@140.221.249.201) |
| 19:47.51 | *** join/#asterisk ManxPower (n=manxpowe@124.sub-75-203-172.myvzw.com) |
| 19:48.11 | ManxPower | Message I sent to them: "I can no longer access the developer lounge. No matter what I do, no matter how many times I change my password it won't let me in. Error messages on the page include: "Notice: Trying to get property of non-object in /var/www/html/plugins/user/joomla.php on line 109" and "E_NOLOGIN_ACCESS" First you forgot to include the sample source code with the Linux SDK, now your developers lounge is inaccessable. This |
| 19:48.11 | *** join/#asterisk lesouvage (n=lesouvag@cc341200-a.assen1.dr.home.nl) |
| 19:48.20 | StephenF | yup no 'b' option mentioned in the CLI |
| 19:48.21 | ManxPower | Well, THAT was sent to the wrong channel. |
| 19:48.45 | StephenF | its either/or |
| 19:50.00 | Tuxguy | Can someone point me at documentation for setting up MGCP SIP devices? Like what d001 is, etc |
| 19:50.43 | ManxPower | Tuxguy: mgcp.conf.sample is not useful? |
| 19:52.18 | StephenF | ahh the 'b' option may be in TRUNK, looks like there is a patch to allow it to work |
| 19:52.34 | ManxPower | which "b" option are you referring to? |
| 19:53.00 | StephenF | ManxPower dial by name directory, 'b' option allows to match first AND last name |
| 19:56.25 | ManxPower | That is a 1.6 option I believe. I'm checking right now |
| 19:56.57 | StephenF | probably is, because it looks like it was accepted into 1.4 trunk. I would assume it would find its way to 1.6 then as well |
| 19:58.17 | ManxPower | well there's no reference to it in the changelog, I bet the reference is to the bug id in the changelog |
| 19:58.50 | jsmith | No, I doubt it was added to 1.4 |
| 19:58.54 | ghento | Hi all. Has anyone successflly got a python script to work with agi? I'm trying, and the agi file doesn't even seem to be executing. in the console it says it's returning 0 |
| 19:59.12 | jsmith | StephenF: In fact, it's not even in 1.6.0 |
| 19:59.18 | StephenF | hmm |
| 19:59.25 | ManxPower | *nod* I just discovered that. |
| 19:59.45 | StephenF | heres the bug: http://bugs.digium.com/view.php?id=7151 |
| 19:59.56 | ManxPower | jsmith: I sometimes what people THINK "1.4 gets no new features" actually means. |
| 20:00.13 | ManxPower | They seem to think "1.4 gets new features" |
| 20:00.44 | [TK]D-Fender | ManxPower: It dioes... just never included in digium releases :) |
| 20:01.06 | StephenF | i know its not gonna be in official 1.4 release |
| 20:01.13 | ManxPower | Looks like it may have gone into 1.6.0-beta4 |
| 20:01.31 | jsmith | StephenF: It was only merged into tr1unk... meaning it's not in 1.6.0 either, but probably made it in time for 1.6. |
| 20:01.40 | jsmith | 1.6.1, that is |
| 20:01.50 | ManxPower | 1,6 does get pretty much any new feature anyone seems to want to throw into it. |
| 20:02.00 | StephenF | cool, well i should be able to apply the patch to my 1.4 install |
| 20:02.17 | ManxPower | I'm sure that will improve stability. But I'm not bitter about it. |
| 20:02.54 | *** join/#asterisk highzeth (n=highzeth@hoiseth.no) |
| 20:03.07 | jsmith | ManxPower: Not quite true... 1.6.<next version> will get many new features |
| 20:03.37 | ManxPower | are they going to backport fixes into the previous release? |
| 20:04.15 | [TK]D-Fender | its not in 1.6.0.1 |
| 20:04.21 | jsmith | ManxPower: No. |
| 20:04.31 | jsmith | ManxPower: So, let's talk about right now, for example. |
| 20:04.44 | jsmith | ManxPower: If I added a new feature, it would get added to TRUNK first |
| 20:04.50 | ManxPower | so if I want a bug fix I have to upgrade to a version with a bunch of new features? |
| 20:04.58 | jsmith | ManxPower: It wouldn't go into 1.6.0, as that's already feature frozen |
| 20:04.59 | ManxPower | jsmith: nobody downloads TRUNK 8-| |
| 20:05.11 | jsmith | ManxPower: It also wouldn't go into 1.6.1, as it's already been feature frozen as well |
| 20:05.14 | bearded_blitz | just did |
| 20:05.28 | jsmith | ManxPower: It *would* go into 1.6.2, as that hasn't been branched from TRUNK yet |
| 20:05.28 | luke-jr | ManxPower: that's how Linux works now too â¹ |
| 20:06.02 | ManxPower | luke-jr: That's hardly a shining example of a history of stable releases. |
| 20:06.04 | bearded_blitz | ManxPower: bug fixes go into 1.6.x.y where 'x' is your current version, and 'y' is the version with new bug fixes |
| 20:06.13 | ManxPower | and look at how many zillion patches distros backport. |
| 20:06.35 | ManxPower | bearded_blitz: that directly contradicts jsmith |
| 20:07.00 | jsmith | ManxPower: No, I was talking about *features*. He was talking about *bug fixes* |
| 20:07.00 | bearded_blitz | oh wait... bug fixes != regressions |
| 20:07.05 | ManxPower | He just said fixes won't be backported. |
| 20:07.06 | bearded_blitz | and that |
| 20:07.07 | jsmith | ManxPower: See the difference? |
| 20:07.16 | bearded_blitz | fixes != features |
| 20:07.24 | ManxPower | jsmith: and I was talking about bug fixes. |
| 20:07.35 | jsmith | ManxPower: Bug fixes will only be backported if they're regressions |
| 20:07.43 | bearded_blitz | bug fix regressions are supposed to go back into the previous 3 released versions (as it sits now -- that policy may change) |
| 20:07.44 | jsmith | (which is really hard to have in a .0 release) |
| 20:08.08 | ManxPower | I run production Asterisk servers. I would prefer not to spend a month testing every new release because it's full of new bugs. I would like to be able to download just a "bug fix" update. |
| 20:08.37 | ManxPower | jsmith: So no new bug fixes, just fixes to things that used to work and were broken? |
| 20:09.19 | Tuxguy | ManxPower: I dont know what d001 is though, or some of the other examples |
| 20:09.23 | jsmith | ManxPower: Say something was working in 1.6.0 but broken in 1.6.1 and then fixed in 1.6.2. Since there was a regression, it would get backported to the 1.6.1 branch |
| 20:09.32 | ManxPower | bearded_blitz: that policy has already changed. 1.2 has only gotten SECURITY fixes for quite a while now. |
| 20:09.41 | bearded_blitz | ok... |
| 20:09.43 | StephenF | wher would the app_directory.c file be located? I want to apply that patch to it... |
| 20:09.45 | jsmith | ManxPower: But if the fix only fixes a bug that wasn't a regression, it won't be backported |
| 20:09.54 | Tuxguy | ManxPower: Also there arent any .sample files in my /etc/asterisk/ |
| 20:09.55 | bearded_blitz | StephenF: in the apps/ subdir of your asterisk source |
| 20:09.58 | jsmith | StephenF: In the apps/ subdirectory of the source code |
| 20:10.05 | StephenF | so i have to reinstall? |
| 20:10.08 | ManxPower | jsmith: but if something was broken since 1.4.6 then the bug will not be fixed? |
| 20:10.09 | jsmith | bearded_blitz: You copying me!?! |
| 20:10.15 | bearded_blitz | StephenF: you have to make the change and then run 'make install' |
| 20:10.20 | StephenF | gotcha |
| 20:10.24 | jsmith | ManxPower: It will be fixed in the 1.4 branch, but not in 1.6.0 |
| 20:10.34 | ManxPower | Tuxguy: try /path/to/src/asterisk/configs |
| 20:10.35 | bearded_blitz | StephenF: it'll only recompile the stuff that changed if you had previously compiled everything in that dir |
| 20:10.37 | jsmith | ManxPower: (I don't necessarily agree with that part, but that's the way it is) |
| 20:10.44 | bearded_blitz | jsmith: of course!@ |
| 20:10.46 | StephenF | perfect, thanks |
| 20:11.39 | ManxPower | jsmith: so if in like 6 months something broke in 1.6.21 and remained broken thru the current release we'll randomly call it 1.6.45 then the breakage won't be fixed? |
| 20:12.13 | bearded_blitz | sure it would... that would be a regression |
| 20:12.36 | ManxPower | Tuxguy: I know it's not "cool" these days to look int eh tarball for the docs, but that is where the best, most current, and most accurate Asterisk docs are |
| 20:12.52 | bearded_blitz | but only 1.6.45 - 3 point releases (as it stands now) |
| 20:13.06 | Tuxguy | oh |
| 20:13.14 | Tuxguy | I installed via rpm, ill try to find it |
| 20:13.18 | *** join/#asterisk guilherme-jorge (n=guilherm@mail.danresa.com.br) |
| 20:13.28 | ManxPower | Tuxguy: and that is why we don't like packages here. |
| 20:13.45 | ManxPower | go find out where the packager put the docs and hope they put all the docs there. |
| 20:14.20 | Katty | bearded_blitz: bearded?! |
| 20:14.21 | jsmith | ManxPower: If it's a reversion, it gets backported. If it's not a reversion, it doesn't. I could waste all day making theoretical knots and logic pretzels, but that's why we have a development team. They're the ones that make a judgment call (and yes, it is a judgment call) on what it qualifies as |
| 20:14.35 | bearded_blitz | Katty: I have not shaved in over a week! |
| 20:14.39 | Katty | bearded_blitz: oh. |
| 20:14.41 | jsmith | ManxPower: a reversion and what doesn't. |
| 20:14.43 | Katty | bearded_blitz: :> |
| 20:14.53 | jsmith | bearded_blitz: And I'll bet I still have a better beard than you! |
| 20:15.04 | bearded_blitz | jsmith: most likely! damn these blonde hairs :) |
| 20:15.17 | bearded_blitz | oh well... lucky for me the g/f likes facial hair, so I don't have to shave every day :) |
| 20:15.58 | jsmith | bearded_blitz: Oh, she's now an official g/f, is she?\ |
| 20:16.06 | jsmith | bearded_blitz: You sly devil, you... |
| 20:16.11 | bearded_blitz | jsmith: I figured after 3 months I can give her that title :) |
| 20:16.15 | jsmith | watches another heart get broken |
| 20:17.05 | bearded_blitz | jsmith: I just haven't called anyone my g/f in like... a few years... so it's just a little weird for me to say it :) |
| 20:18.36 | *** join/#asterisk littlepinkdot (n=thedot@69.7.43.20) |
| 20:20.22 | *** join/#asterisk acxty (n=glax@201.220.136.117) |
| 20:20.26 | acxty | Hi guys, |
| 20:20.45 | acxty | A good store where I can get telephones, cards, etc.. compatible with asterisk |
| 20:20.46 | *** join/#asterisk X-Rob (n=Rob@ppp214-210.static.internode.on.net) |
| 20:21.04 | X-Rob | Well. It's morning. |
| 20:21.37 | bearded_blitz | acxty: www.digium.com? :) |
| 20:21.39 | X-Rob | would someone (Qwell, mog?) like to make a call that http://bugs.digium.com/view.php?id=13786 should be split into two bugs? |
| 20:21.51 | X-Rob | I'm quite happy to create a new bug |
| 20:21.53 | bearded_blitz | acxty: I think there are links to distributors too |
| 20:22.16 | bearded_blitz | X-Rob: what are the two issues? |
| 20:22.33 | X-Rob | bearded_blitz, I'll let you read it |
| 20:22.54 | bearded_blitz | well I read it, but I don't understand the two issues |
| 20:23.04 | X-Rob | exactly my point. _IS_ it two issues? |
| 20:23.15 | X-Rob | tzafir's not around to poke him about it |
| 20:23.30 | bearded_blitz | ya he is in israel I think |
| 20:24.12 | Tuxguy | ManxPower: Still searching, but what does aaln mean? |
| 20:24.18 | Ritzerisk | is it possilbe to connect a FXS linksys sipura 2102 to a LS trunk on a mitel ?? |
| 20:25.19 | [TK]D-Fender | Ritzerisk: Should be fine |
| 20:27.45 | *** join/#asterisk mindCrime (n=chatzill@216.27.62.2) |
| 20:27.45 | ghento | Hi all. Has anyone successflly got a python script to work with agi? I'm trying, and the agi file doesn't even seem to be executing. in the console it says it's returning 0 |
| 20:29.14 | [TK]D-Fender | ghento: pastebin is your friend. and AGI doesn't care what language you use for it as long is it inputs & outputs to the right devices |
| 20:29.22 | *** join/#asterisk JonCup (n=JonCup@cpe-67-49-245-26.dc.res.rr.com) |
| 20:32.07 | *** join/#asterisk lesouvage (n=lesouvag@cc341200-a.assen1.dr.home.nl) |
| 20:35.30 | *** join/#asterisk Arsenick- (n=rpurcell@modemcable026.33-70-69.static.videotron.ca) |
| 20:43.21 | *** join/#asterisk kisu (n=kisu@daniel1117.broker.freenet6.net) |
| 20:46.23 | *** join/#asterisk rwaite (n=richard@rrcs-74-218-125-86.central.biz.rr.com) |
| 20:47.29 | *** join/#asterisk rwaite (n=richard@rrcs-74-218-125-86.central.biz.rr.com) |
| 20:47.36 | *** join/#asterisk florz (i=nobody@2001:1a50:503c:0:0:0:0:1) |
| 20:51.13 | *** join/#asterisk kisu (n=kisu@daniel1117.broker.freenet6.net) |
| 20:53.47 | sprbck | hello. I have a Digium TDM800P card, where all FXS ports (4) just froze. All 4 analog extensions gave no dial tone. After a reboot all was fine. Restarting asterisk and reloading modules did not help. Any hint about this? |
| 20:54.31 | *** join/#asterisk tzafrir_laptop (n=tzafrir@bzq-179-75-202.static.bezeqint.net) |
| 20:55.38 | *** join/#asterisk erth64net (n=erth64ne@mail.opensourcery.com) |
| 20:56.13 | *** join/#asterisk mnicholson (n=mnichols@nat/digium/x-be808b8e97f46420) |
| 20:57.10 | jaytee | Katty, how come your website's down? |
| 20:59.20 | *** join/#asterisk asteriskmonkey (n=philip@69.77.169.14) |
| 20:59.23 | *** part/#asterisk kisu (n=kisu@daniel1117.broker.freenet6.net) |
| 20:59.25 | donnib | how do i force use of a specific codec ? disallow: all allow: gsm is that correct ? |
| 20:59.50 | asteriskmonkey | with sip "friends" since out-going limit was taken away in 1.4, how do you limit outgoing calls? |
| 20:59.53 | [TK]D-Fender | donnib: "=", not ":" |
| 21:00.10 | donnib | yes sorry |
| 21:00.15 | donnib | that's what i meant |
| 21:01.55 | donnib | does this mean it's running GSM codec ? http://pastebin.com/d21d4036a |
| 21:02.35 | asteriskmonkey | <PROTECTED> |
| 21:02.41 | [TK]D-Fender | donnib: Sure looks like. |
| 21:03.00 | donnib | and another question. which codec uses less BW ? |
| 21:03.03 | asteriskmonkey | donnib : add allow=ulaw before the gsm line it will give ulaw priority |
| 21:03.49 | asteriskmonkey | [TK]D-Fender: with sip "friends" since out-going limit was taken away in 1.4, how do you limit outgoing calls? do i have to arse about abit with dial plan |
| 21:04.22 | [TK]D-Fender | asteriskmonkey: Go read the sample config. its in there. |
| 21:04.29 | [TK]D-Fender | asteriskmonkey: and in the upgrade.txt, etc |
| 21:04.44 | asteriskmonkey | ah if is see a group thing ill be upset |
| 21:06.06 | *** join/#asterisk pecanha (n=e@189.106.46.162) |
| 21:06.08 | donnib | if i can choose between Codecs: G.711a/u-law and GSM which one uses less BW ? |
| 21:06.56 | jsmith | donnib: GSM uses less bandwidth |
| 21:07.02 | donnib | ok |
| 21:07.20 | jsmith | donnib: It's approximately 13kbps (plus IP overhead) as opposed to G.711's 64kbps (plus overhead) |
| 21:08.25 | asteriskmonkey | doe missing my limitonpeers, that is so misleading :/ |
| 21:13.16 | Tuxguy | Anyone know an MGCP softphone? |
| 21:14.53 | lesouvage | If I receive this message in the cli "Got SIP response 500 "Internal Server Error" back from 82.146.xxx.xx" does this indicate that the provider has a problem? The IP is the IP of the SIP provider. |
| 21:16.04 | lesouvage | If I do sip show registry the sip account seem to be registered |
| 21:17.33 | [TK]D-Fender | lesouvage: in response to WHAT? |
| 21:17.48 | [TK]D-Fender | Tuxguy: Now why would you want to go and do something silly like that? |
| 21:18.07 | *** join/#asterisk erth64net (n=erth64ne@mail.opensourcery.com) |
| 21:18.36 | Tuxguy | [TK]D-Fender: Trying to trouble shoot a hardphone connecting to asterisk. Its a Nortel 6812 MGCP device. |
| 21:18.49 | lesouvage | [TK]D-Fender: a Dial() statement that used to work until 30 minutes ago. |
| 21:19.27 | [TK]D-Fender | Tuxguy: I've never seen one personally. Why do you feel you'd need an MGCP softphone to debug a hardphone's issue? |
| 21:20.15 | *** join/#asterisk esdrasbeleza (n=esdras@sisyphus.dreamhost.com) |
| 21:20.22 | Tuxguy | Because if I can connect w/ the softphone, and see some of the messages etc, then i can compare the two's settings and see the difference etc |
| 21:21.57 | [TK]D-Fender | Tuxguy: Do you see anything coming in from the phone? |
| 21:22.39 | Tuxguy | Nothing. I enabled debug on mgcp, and rebooted the phone. Still havent seen anything coming through. |
| 21:23.05 | esdrasbeleza | hi, I have a question about voicemail. If I delete an single message from /var/spool/asterisk/voicemail/LOCAL/user/INBOX, do I need to rename the other files to keep them in sequence? |
| 21:25.50 | Tuxguy | [TK]D-Fender: I have the call server as 192.168.1.109 and port 2427 , which is what mgcp.conf is using |
| 21:26.36 | [TK]D-Fender | ok, checkout time, heading home. |
| 21:26.39 | [TK]D-Fender | BBIAB |
| 21:27.00 | justdave | I have a switch statement doing a dundi lookup in an IVR menu, and we discovered this morning that only some of the numbers that are available via dundi are working. They're all three digit numbers, and with some specific patterns, it cuts the user off after the 2nd digit and tells them it's invalid. (it's using the Background() application for the menu) |
| 21:27.14 | justdave | is that expected, and is there some way to work around that? |
| 21:27.21 | jaytee | Tuxguy, if you do a netstat -ua is Asterisk listening on that port? |
| 21:27.37 | *** join/#asterisk voxter (n=voxter@76.77.95.2) |
| 21:27.43 | Tuxguy | Yes |
| 21:27.52 | Tuxguy | udp 0 0 0.0.0.0:2427 0.0.0.0:* 17853/asterisk |
| 21:28.37 | *** part/#asterisk root52 (n=f745082a@ip70-191-120-39.cl.ri.cox.net) |
| 21:29.25 | Tuxguy | However, I do not see anything in CLI when the phone connects. |
| 21:29.33 | jaytee | Tuxguy, what if you do a mgcp show endpoints ? |
| 21:29.51 | Tuxguy | Gateway '192.168.1.84' at 192.168.1.84 (Static) -- 'd00/1@192.168.1.84 in 'default' is idle |
| 21:30.40 | *** join/#asterisk sah-work (n=Bawbatos@commodity-noc-201.sc08.org) |
| 21:31.00 | jaytee | Tuxguy, is your subnet mask 255.255.255.0 ? |
| 21:31.09 | Tuxguy | yes |
| 21:31.26 | Tuxguy | inet addr:192.168.1.109 Bcast:192.168.1.255 Mask:255.255.255.0 |
| 21:32.19 | jaytee | Tuxguy, did it work before or is this a new setup? |
| 21:32.26 | Tuxguy | New setup |
| 21:32.40 | Tuxguy | The phones work on another MGCP network though through bandwidth.com |
| 21:32.47 | Tuxguy | But, I am trying to do an in-house testing |
| 21:33.58 | jaytee | what's the verbose level of the CLI? |
| 21:34.17 | Tuxguy | 10 |
| 21:34.31 | Tuxguy | er |
| 21:34.31 | Tuxguy | Verbosity is at least 29 |
| 21:34.33 | *** join/#asterisk CrazyTux (n=brandon@rrcs-67-52-124-226.west.biz.rr.com) |
| 21:34.57 | jaytee | even at 4 it should show a call attempt. it sounds like it's never reaching asterisk |
| 21:35.20 | jaytee | is there a firewall between the two? |
| 21:37.16 | jaytee | Tuxguy, I'd recheck all the settings on the phone itself. make sure it's not trying to use some other address as a proxy, etc. |
| 21:37.16 | Tuxguy | no |
| 21:37.36 | Tuxguy | I can run a sip connection from anywhere on the network to this device, and it works fine. |
| 21:38.02 | Tuxguy | I checked, it has 192.168.1.109 as the call server, and port 2427 as the mgcp port, not sure how else to diagnose it |
| 21:38.09 | Tuxguy | would that connection attempt show up in netstat? |
| 21:39.02 | jaytee | Tuxguy, no |
| 21:39.56 | Tuxguy | Is there any way to test the issue? Thats why I was looking for a MGCP softphone. |
| 21:39.58 | jaytee | you'd have to use wireshark or something similar to see actual traffic |
| 21:40.33 | jaytee | does your server have a gui? probably not |
| 21:40.50 | jaytee | I mean like Gnome or Kde, not an asterisk gui |
| 21:42.57 | Tuxguy | Yes |
| 21:43.00 | Tuxguy | Gnome |
| 21:45.13 | jaytee | download wireshark and set it to capture any packets to and from the phone's address, then you can analyze them. if you can do a capture on your other network on a similar phone to a call through bandwidth.com you'd have something you could compare. |
| 21:45.37 | Tuxguy | Installing now |
| 21:45.54 | Tuxguy | Do I need a promiscuous router? or, switch? |
| 21:46.22 | *** join/#asterisk anthm (n=anthm@freeswitch/developer/anthm) |
| 21:47.29 | jaytee | if you're running wireshark on your asterisk box then all you need is to make sure your NIC supports running in promiscuous mode which most do |
| 21:48.00 | Tuxguy | ok |
| 21:48.32 | Tuxguy | Installed wireshark.. whatis the command? wireshark isn't the name of the binary file |
| 21:49.32 | jaytee | Tuxguy, based on the IP address of the phone and the server they're both on the same subnet. If you're trying to capture traffic from device A to device B on device B no problem. If you need to capture traffic from a third node you need to have both it and one of the endpoints on a hub cascaded usually. there are other tricks though. |
| 21:50.09 | jaytee | it should install a link in your Applications/System Tools menu |
| 21:50.16 | Tuxguy | ah i got it |
| 21:50.16 | Tuxguy | <PROTECTED> |
| 21:50.20 | Tuxguy | <PROTECTED> |
| 21:51.39 | jaytee | Tuxguy, what linux distro? |
| 21:51.49 | Tuxguy | Centos |
| 21:52.05 | Tuxguy | Weird that asterisk isnt showing that attempt |
| 21:52.24 | jaytee | Tuxguy, http://wiki.wireshark.org/VoIP_calls |
| 21:52.58 | Tuxguy | hmm, my wireshark isnt a GUI like that |
| 21:53.02 | jaytee | there's a section in there about MGCP. It's quittin time for me so I've gotta leave in a minute. |
| 21:53.18 | Tuxguy | ok |
| 21:53.24 | Tuxguy | Ill wait here until another day :P j/p |
| 21:53.58 | jaytee | Tuxguy, it used to be called Ethereal and it comes in both Windows and Linux flavors. Each one looks just a teensy bit different |
| 21:54.52 | Tuxguy | ah found wireshark-gnome |
| 21:55.35 | Tuxguy | Only getting the same errors |
| 21:56.20 | jaytee | try typing service iptables stop to temporarily kill your firewall and retest a call. |
| 21:56.34 | Tuxguy | its not running |
| 21:56.50 | jaytee | well, good luck with that. I've gotta head out. |
| 21:57.24 | Tuxguy | ty |
| 22:00.00 | Carlos_PHX | Trying to shop SIP carriers is worse than having to vote for either Obama or McCain. |
| 22:00.28 | Carlos_PHX | You know no matter what you are screwed, but you are currently being screwed and hope for some lube in your future screwing. |
| 22:00.39 | *** join/#asterisk [TK]D-Fender (n=chatzill@64.235.218.194) |
| 22:00.44 | *** join/#asterisk kornelak (n=karl@199.33.79.4) |
| 22:01.21 | Carlos_PHX | It's the bane of my existence these days. |
| 22:02.31 | Tuxguy | I do not have a firewall running, but I am getting an error over wireshark that my MGCP phone is unreachable. |
| 22:03.44 | Tuxguy | "692","448.496570","192.168.1.109","192.168.1.84","ICMP","Destination unreachable (Port unreachable)" |
| 22:04.30 | *** join/#asterisk johann8384 (n=johann83@intra.netlogic.net) |
| 22:05.28 | [TK]D-Fender | Tuxguy: Do you see * listening on MGCP? |
| 22:05.42 | Tuxguy | Yes |
| 22:06.27 | [TK]D-Fender | Tuxguy: pastebin your firewall dump, your mgcp.conf, CLI output at verbose 10, mgcp debug enabled, etc |
| 22:06.45 | *** join/#asterisk shinao1 (n=shinao1@62.173.48.42) |
| 22:07.26 | Tuxguy | I do not have a firewall, but I can do the rest. Also, mgcp debug doesn't ever say anything |
| 22:09.42 | Tuxguy | [TK]D-Fender: http://pastebin.ca/1260808 this is mostly just mgcp.conf , because there isnt anything in CLI except the command prompts |
| 22:11.02 | [TK]D-Fender | Tuxguy: include an iptables dump, a netstat -an dump, an "mgcp show users", etc |
| 22:13.59 | Tuxguy | No such command mgcp show users |
| 22:14.14 | Tuxguy | show endpoints? |
| 22:15.09 | [TK]D-Fender | Tuxguy: "help mgcp" |
| 22:16.00 | Tuxguy | http://pastebin.com/m1b668d78 |
| 22:17.14 | Tuxguy | Is there anything else needed with that apte , [TK]D-Fender ? |
| 22:19.30 | [TK]D-Fender | Tuxguy: [192.168.1.84] <- wonder if this should be some kind of username |
| 22:19.34 | [TK]D-Fender | MAC, etc |
| 22:19.43 | Tuxguy | IP address |
| 22:19.59 | Tuxguy | hmm |
| 22:20.14 | Tuxguy | I can change it to the MAC address of the phone and retry. |
| 22:20.26 | [TK]D-Fender | Tuxguy: try a few different things |
| 22:21.14 | Tuxguy | Ok, i have changed it to use the MAC address. What else do you suggest i try? |
| 22:22.38 | Tuxguy | [00405A141243] |
| 22:23.25 | [TK]D-Fender | Tuxguy: not sure |
| 22:23.40 | Tuxguy | Running wireshark again on the IP address of the phone. |
| 22:23.51 | Tuxguy | This is why I was hoping that I could find an MGCP softphone. |
| 22:24.17 | [TK]D-Fender | tux you see no incoming traffic on starting up the phone? |
| 22:24.23 | [TK]D-Fender | Tuxguy: or dialing? |
| 22:24.32 | Tuxguy | Not in the CLI with Verbosity set to 10, and MGCP set debug on |
| 22:25.02 | Tuxguy | In wireshark, I see a bunch of ICMP Destination unreachable when connecting from Asterisk->the phone |
| 22:25.14 | Tuxguy | Also a lot of Phone -> Asterisk RSIP messages |
| 22:26.14 | [TK]D-Fender | Tuxguy: nmap the phone... see anything unusual? |
| 22:27.15 | [TK]D-Fender | Tuxcomplete port scan, TCP & UDP |
| 22:29.42 | *** join/#asterisk etfonhomey (n=chatzill@74-143-196-254.static.insightbb.com) |
| 22:32.11 | Tuxguy | The only ports that are open are 6000 and 8000 |
| 22:32.12 | Tuxguy | weird |
| 22:32.35 | [TK]D-Fender | Tuxguy: I'm trusting it less and less |
| 22:33.00 | *** join/#asterisk cvnet (n=dahitler@74.210.103.241) |
| 22:33.04 | cvnet | hello hello |
| 22:33.38 | cvnet | is there a way to find out the pattern of payphones? |
| 22:33.50 | Tuxguy | [TK]D-Fender: These phones work with bandwidth.com though, wich is weird. |
| 22:40.56 | Tuxguy | This makes absolutely no sense. Shouldnt that port be open so that asterisk can talk to it? |
| 22:41.13 | tompaw | hi there |
| 22:41.18 | tompaw | which phones are you discussing? |
| 22:42.04 | *** join/#asterisk ManxPower (n=manxpowe@116.sub-70-223-239.myvzw.com) |
| 22:42.24 | tompaw | has anyone used asterisk + openser as a load balancer? |
| 22:42.50 | Tuxguy | tompaw: Nortel 6812 |
| 22:42.53 | Tuxguy | MGCP |
| 22:43.05 | tompaw | ouch :) |
| 22:43.24 | Tuxguy | Familiar with this phone? |
| 22:43.56 | tompaw | nope, but I just noticed you guys were discussing mgcp |
| 22:45.23 | Tuxguy | This phone doesnt have an incoming port, lol |
| 22:48.06 | tompaw | does it say on the package that it supports answering calls, too? |
| 22:48.14 | tompaw | maybe is has an asterisk somewhere ;-) |
| 22:48.25 | tompaw | * - this model is only designed for making calls. |
| 22:49.34 | ManxPower | It's going to be a cold night here in northern AL |
| 22:49.59 | tompaw | what's AL? |
| 22:50.03 | tompaw | it is a cold night here as well |
| 22:50.16 | tompaw | and they say that true winter starts this weekend |
| 22:50.28 | tompaw | with snow, santa and stuff... |
| 22:50.31 | *** join/#asterisk shinao1 (n=shinao1@62.173.48.42) |
| 22:51.11 | tompaw | luckily, my Nokians WR G2 + quattro drive are giving me a piece of mind ;) |
| 22:53.10 | *** join/#asterisk shinao1 (n=shinao1@62.173.48.42) |
| 22:54.51 | *** join/#asterisk Carlos_PHX (n=Carlos@68.3.162.244) |
| 22:58.19 | tompaw | ManxPower: what's AL? |
| 22:58.45 | ManxPower | Alabama |
| 23:02.52 | cvnet | questions: if a calls comes in from one of DID in asterisk box, and there is no match in sip.conf where does it go on extensions.conf ? |
| 23:09.10 | murdock_ut | cvnet: Depends on your dialplan. |
| 23:09.29 | *** join/#asterisk C4away (n=DJpyro@66.185.107.193) |
| 23:09.41 | C4away | is there a way to send to two email addresses from voicemail.conf? |
| 23:10.22 | C4away | 1234 => 1234,Joe User,joe@1234.com&joe@5678.com |
| 23:11.07 | C4away | or maybe with a ; |
| 23:11.14 | highzeth | or make a mail alias |
| 23:11.20 | C4away | comma and pipe are obviously out |
| 23:11.24 | C4away | yea, I can do that |
| 23:11.29 | C4away | if I have to |
| 23:12.01 | C4away | hell, I'm too lazy to google to see if anyone has done this, you think I'm going to go make a mail alias? |
| 23:13.06 | murdock_ut | C4away: That is what I do. |
| 23:15.21 | C4away | in 2006 there was talk on asterisk-dev mailing list to do a space or : seperated list of email addresses |
| 23:16.51 | *** join/#asterisk coppice (n=chatzill@33.155.17.210.dyn.pacific.net.hk) |
| 23:18.53 | C4away | hah |
| 23:18.55 | C4away | funny me |
| 23:19.19 | C4away | I"ll just modify the voicemail thing to point to Voicemail(123&456) |
| 23:19.29 | C4away | why didn't I think of that earlier? |
| 23:20.39 | *** join/#asterisk gpowers (n=chatzill@99.152.249.221) |
| 23:21.27 | *** part/#asterisk Deeewayne (n=Deeewayn@nat/digium/x-547c915ba480e7ce) |
| 23:25.53 | *** join/#asterisk synchris (n=synchris@athedsl-154703.home.otenet.gr) |
| 23:25.58 | unpaidbill | hrm, my chan_dahdi module loads, i see it reads the config fine 'Registered channel 1, FXO Kewlstart signalling' etc.. but i have no dahdi commands in the console |
| 23:26.14 | unpaidbill | i cant figure out what is broken :/ |
| 23:27.20 | *** join/#asterisk jaytee (n=jforde05@unaffiliated/jaytee) |
| 23:27.26 | unpaidbill | it appears to not be even trying to load chan_dahdi... |
| 23:28.01 | ManxPower | cvnet: it will normally be sent to the contest listed in sip.conf [general] |
| 23:36.22 | unpaidbill | hmm ok so it is loading chan_dahdi.so apparently, i can do a module load chan_dahdi and it does the whole registered channel 1.... and so on, but i dont get any dahdi * commands on the console and I cant dial any of the channels on the tdm |
| 23:36.30 | unpaidbill | how nice |
| 23:38.16 | *** join/#asterisk root52 (n=root52@ip70-191-120-39.cl.ri.cox.net) |
| 23:39.30 | *** join/#asterisk [netman] (n=netman@238.Red-88-23-119.staticIP.rima-tde.net) |
| 23:43.22 | *** join/#asterisk DarkRift (n=dark@65.92.251.135) |
| 23:44.44 | jameswf | unpaidbill: RTFL |
| 23:52.50 | *** join/#asterisk MrNaz (n=mrnaz@ppp118-208-142-142.lns10.mel4.internode.on.net) |
