| 00:14.41 | *** join/#asterisk The_LightSide (n=dialt@wbs-196-2-116-128.wbs.co.za) |
| 00:15.27 | justdave | is there a way to tell * to group three extensions together so that a call that was delivered to them via a ringall shows up as call in progress on the non-answering phones, and picking up those phones would join the call? |
| 00:15.43 | justdave | or do all three phones have to actually have the same extension number to do that? |
| 00:16.21 | *** join/#asterisk Davey (n=davey@unaffiliated/davey) |
| 00:16.59 | justdave | two of those phones can only handle one line presence, so I can't add a second line to them to handle the ringall stuff, but I want to preserve them being able to call each other |
| 00:21.08 | *** join/#asterisk The_LightSide (n=dialt@wbs-196-2-116-128.wbs.co.za) |
| 00:21.24 | *** join/#asterisk ivanfm (n=ivanfm@c93481ec.virtua.com.br) |
| 00:28.32 | *** join/#asterisk harryvv (n=none@S010600a0c93f6f7e.vs.shawcable.net) |
| 00:38.38 | *** join/#asterisk TripleFFFF (n=TripleFF@145-27.mc.cite.net) |
| 00:38.54 | TripleFFFF | anyonw nknow that compilation pof updates for windows ? |
| 00:38.57 | TripleFFFF | i cant remember name |
| 00:39.15 | TripleFFFF | n/m |
| 00:39.17 | TripleFFFF | autopatcher |
| 00:46.59 | *** join/#asterisk file2 (n=IrcNet@out.clearnet.com) |
| 00:47.00 | *** mode/#asterisk [+o file2] by ChanServ |
| 00:49.25 | file2 | very very quiet |
| 00:49.30 | Qwell | shh |
| 00:49.34 | Qwell | we're playing a game |
| 00:50.20 | file2 | i think i will use my phone to wave my friend to where i am... |
| 00:50.38 | file2 | yessssss |
| 00:51.02 | *** part/#asterisk TripleFFFF (n=TripleFF@145-27.mc.cite.net) |
| 00:51.04 | Strom_C | catsex |
| 00:51.21 | file2 | dogballs |
| 00:51.26 | harryvv | so file from my understanding you did a presentation at von? |
| 00:51.38 | file2 | i did a panel |
| 00:52.08 | file2 | with kevin delaney from skype and chris bond from counterpath |
| 00:52.23 | harryvv | good ;) I just got off vacation and to fly all the way to the east coast would be tiring. |
| 00:52.25 | file2 | and schuyler deerman from digium moderating it |
| 00:52.54 | harryvv | man skype is so fricken popular. When are thay going to start charging pc to pc calls? |
| 00:53.07 | file2 | who knows. |
| 00:54.05 | file2 | this airport sucks with radio signals... |
| 00:54.12 | harryvv | BTW Nortel is putting in some kind of adapter or card in there ancient pbxs and now thay are voip capable. |
| 00:54.14 | Nitrus^ | im using the t110p card with adtran TA750 12FXS and 12FXO and im using analog phones on the FXS and vonage on my FXO, when someone calls in and i answer they dont hear me say hello due to what seems like a bridging delay, what steps must i take to fix this? most people say hello once, then hang up on me since i dont hear them say it. |
| 00:54.27 | file2 | full signal outside barely any inside. |
| 00:54.28 | harryvv | file, go and buy a yagi antenna :) |
| 00:55.13 | file2 | yeah i will get right on that. uh huh |
| 00:55.18 | harryvv | :) |
| 00:55.48 | file2 | hrm that is not the suv... darn. |
| 00:56.35 | file2 | so did i miss anything? |
| 00:57.38 | *** join/#asterisk saftsack (n=saftsack@p54A7D717.dip.t-dialin.net) |
| 00:58.06 | benjk | Nitrus^, I'd say the first thing to do is try to narrow down where this delay occurs |
| 00:58.13 | file2 | guess not. |
| 00:58.16 | harryvv | problem with EHF frequencies is thay are very directional and suseptable to just about any kind of blockage. Now take my HF radio and I can hit japan from my car with no problem. BTW there is a very cool technoligy that been out for a few years in hamradio its called PSK31. Its db responce is much more sensitive and can cut though the noise. Sometimes I am on a freq and cannot hear anything yet it comes though. Its been tested on 144 mg |
| 00:58.45 | Nitrus^ | when i call out and someone picks up, i dont hear them say hello |
| 00:58.58 | harryvv | once i was typing to paris from seattle on psk31 |
| 00:59.01 | benjk | yes, I understand |
| 00:59.20 | benjk | what I am saying is you need to find out where the delay occurs |
| 00:59.35 | benjk | is it on the FXO-to-Vonage leg |
| 00:59.50 | benjk | or is it on the ADTRAN-to-* leg |
| 01:00.00 | benjk | that's what you should find out first |
| 01:00.38 | file2 | *"*"*0" *0" *0" *0" |
| 01:00.40 | file2 | <PROTECTED> |
| 01:00.55 | file2 | whoops |
| 01:01.13 | Nitrus^ | how would i go about doing that? |
| 01:02.05 | benjk | if you have a VoIP phone, you could configure that as a client on the * box then route incoming calls from the vonage to that phone and see if it is the same |
| 01:02.38 | Nitrus^ | ahh good idea |
| 01:02.39 | Nitrus^ | i can do that |
| 01:02.49 | Nitrus^ | ill give that a try |
| 01:02.51 | Nitrus^ | thanks |
| 01:03.25 | *** join/#asterisk ariel_ (n=Ariel@dsl-20-177.cofs.net) |
| 01:04.58 | *** join/#asterisk syzygyBSD (n=chatzill@66.226.228.204.cpe.speedyquick.net) |
| 01:06.14 | *** join/#asterisk justinu|laptop (n=Justin@dsl081-083-093.lax1.dsl.speakeasy.net) |
| 01:12.35 | *** join/#asterisk kusznir (n=kusznir@66-233-98-93.lew.clearwire-dns.net) |
| 01:20.39 | kusznir | Hi all: Can anyone tell me why CID from my sip device comes through as +1234567890 (instead of just 1234567890)? |
| 01:21.18 | *** join/#asterisk seebs (n=seebs@216-243-131-210.static.iphouse.net) |
| 01:21.40 | seebs | I have confusing output. |
| 01:21.42 | seebs | Module 0: Installed -- AUTO FXS |
| 01:21.42 | seebs | Module 1: Installed -- AUTO FXS |
| 01:21.42 | seebs | VoiceDAA did not bring up ISO link properly! |
| 01:21.42 | seebs | Module 2: Not installed |
| 01:21.42 | seebs | VoiceDAA did not bring up ISO link properly! |
| 01:21.43 | seebs | Module 3: Not installed |
| 01:22.08 | seebs | I do not know for sure what this means. I suspect that it means either that I have introduced a bug into the driver I am porting, or that FXO modules don't configure unless there's a line plugged into them. |
| 01:26.59 | *** join/#asterisk mtgh (n=chatzill@dsl093-001-038.det1.dsl.speakeasy.net) |
| 01:32.31 | *** join/#asterisk furby_singh (n=furby@n156.dkm.cz) |
| 01:32.35 | *** join/#asterisk Qwell_ (n=north@unaffiliated/qwell) |
| 01:32.35 | *** mode/#asterisk [+o Qwell_] by ChanServ |
| 01:32.51 | *** join/#asterisk nitrus^ (i=Nitrus_@ip70-187-148-2.oc.oc.cox.net) |
| 01:33.27 | furby_singh | Hi all :-) Can anyone force a clue on me about how to inject audio into an active call? it's driving me nuts :-( |
| 01:35.34 | *** join/#asterisk A-Tuin (n=a-tuin@steves.ip.v4.me.uk) |
| 01:35.47 | furby_singh | Is there anyone alive in here? |
| 01:35.59 | harryvv | yes |
| 01:36.20 | harryvv | what are your issues? |
| 01:36.39 | furby_singh | yay! Hello from Prague Harry :-) |
| 01:36.39 | nitrus^ | so im running a TA750 and im getting a bridging delay when i attempt to bridge FXO and FXS channels, when i bridge to SIP it's fine, anyone know what could be causing this? also when i dial out there is a large pause until i hear a ring when i call out on analog and sip lines |
| 01:36.47 | *** join/#asterisk brut- (n=brutwork@66.173.4.254) |
| 01:37.05 | furby_singh | Thanks for response, I need to play audio to a caller and callee on a bridged call, just a recorded message but damned if I can work out how... |
| 01:40.13 | furby_singh | <PROTECTED> |
| 01:40.22 | *** join/#asterisk bkw_ (n=bkw_@asterisk/friend-and-developer/bkw) |
| 01:40.30 | benjk | Nitrus, whats the exact call path |
| 01:40.40 | *** part/#asterisk bkw_ (n=bkw_@asterisk/friend-and-developer/bkw) |
| 01:40.43 | *** join/#asterisk r0d3nt (n=RatMan@foster.stonedcoder.org) |
| 01:40.43 | furby_singh | Alternativly does anyone know how to put both channels of a bridged call on hold through the manager interface? |
| 01:41.17 | benjk | like [vonage]---[fxo-adapter]---SIP----[asterisk]---[foo]---[bar]---[baz] |
| 01:42.39 | nitrus^ | k hold on |
| 01:42.43 | nitrus^ | ill type it up |
| 01:43.16 | *** join/#asterisk _Vile (i=_Vile@198.175.14.242) |
| 01:43.45 | nitrus^ | outgoing call (analog phone) -> TA750 (fxo channel) -> t1 -> asterisk -> t1 -> TA750(fxs channel) -> vonage fxo adapter -> voip |
| 01:43.59 | nitrus^ | they pick up and say hello but we dont here them until they say it a second time |
| 01:44.25 | benjk | is this on incoming and outgoing calls? |
| 01:44.29 | nitrus^ | outgoing |
| 01:44.39 | nitrus^ | i might have the fxo and fxs swapped for the TA750 |
| 01:44.42 | nitrus^ | theyre correct on the device |
| 01:45.00 | benjk | no, if you mismatched them, it wouldn't work at all |
| 01:45.08 | nitrus^ | right |
| 01:45.10 | nitrus^ | and they do |
| 01:45.21 | nitrus^ | im using kewlstart |
| 01:46.22 | benjk | this is difficult to debug |
| 01:46.41 | furby_singh | Anyone here used app_bridge ? |
| 01:46.48 | benjk | the trouble is that you have got multiple analog-digital-analog conversions in your path |
| 01:47.13 | benjk | everytime it adds some delay |
| 01:47.17 | nitrus^ | right |
| 01:47.38 | nitrus^ | but since sip doesnt seem to be affected i think it's only between the TA 750 channels |
| 01:48.14 | furby_singh | nitrus^: apologies for interrupting, just out of interest may I ask why you have this setup? I'm fascinated is all :-) |
| 01:48.28 | benjk | well, the SIP phone leg to asterisk is all IP so its very fast and any delay you are getting on the FXO is small enough not to be noticeable |
| 01:48.31 | nitrus^ | whch parts? |
| 01:48.34 | nitrus^ | it's for business phones |
| 01:49.12 | nitrus^ | i needed a channel bank and the TA was cheap |
| 01:49.21 | benjk | do you have a Sipura or some other FXO-SIP device there? |
| 01:49.35 | nitrus^ | nope |
| 01:49.39 | furby_singh | nitrus^: ok, I don't know much about vonage but I'm sure the reason is good. Many thanks. |
| 01:49.39 | benjk | just for testing another route |
| 01:49.53 | nitrus^ | it's cheap is all |
| 01:50.06 | benjk | if you can get hold of one, you could hook the vonage port to a SIP adapter and go from there into asterisk |
| 01:50.12 | benjk | then see if that makes a difference |
| 01:50.26 | benjk | like Sipura 3000 for example |
| 01:50.42 | *** join/#asterisk jontow (i=jontow@hijacked.us) |
| 01:51.02 | benjk | nitrus, how many channels with vonage do you have? |
| 01:51.06 | benjk | just one? |
| 01:51.08 | nitrus^ | 6 |
| 01:51.12 | benjk | ah |
| 01:51.34 | benjk | because Vonage do have a service package where you don't have to use their FXO adapter |
| 01:51.39 | benjk | you can go straight SIP |
| 01:51.44 | benjk | but it costs extra |
| 01:52.00 | benjk | 10 or 15 bucks or so per channel, IIRC |
| 01:52.10 | nitrus^ | hmmm |
| 01:52.11 | nitrus^ | i didnt know that |
| 01:52.15 | nitrus^ | i complained about that crap to them |
| 01:52.24 | benjk | it is intended to be an added service |
| 01:52.26 | nitrus^ | sip is sip i dont need their stupid hardware |
| 01:52.44 | benjk | for people who have the adapter at home and want to use their vonage account fro their laptop while on the road |
| 01:52.44 | nitrus^ | that would take all those stupid boxes out of my telco room |
| 01:53.13 | nitrus^ | then i can drop all my FXO adapters |
| 01:53.15 | nitrus^ | hmm |
| 01:53.22 | benjk | so the catch is that you have to pay for the standard service with the fxo adapter and then extra for the SIP add on |
| 01:53.43 | benjk | you'd then just use the sip one, and never pick up on the adapter |
| 01:53.50 | nitrus^ | right |
| 01:54.04 | nitrus^ | are you talking about their soft phone package? |
| 01:54.10 | benjk | yes |
| 01:54.20 | nitrus^ | i thought there were restrictions on minutes and stuff |
| 01:54.22 | benjk | you can configure asterisk to hook up to that |
| 01:54.30 | benjk | I don't know what the rules are |
| 01:54.33 | *** join/#asterisk lowlevel (n=Stuart@CPE0050ba71c82f-CM000f9f7d6742.cpe.net.cable.rogers.com) |
| 01:54.39 | benjk | I can only tell you about the technical side |
| 01:55.15 | nitrus^ | yeah you only get 500 local and long distance minutes |
| 01:55.31 | benjk | why did you chose Vonage anyway? |
| 01:55.49 | nitrus^ | it was cheap at the time |
| 01:55.54 | benjk | ok |
| 01:56.01 | nitrus^ | i would like a pure sip to analog provider if there is one |
| 01:56.07 | nitrus^ | i just dont know who is good |
| 01:56.18 | benjk | I generally tell my clients to shop for a provider who doesn't force them to use locked CPE gear |
| 01:56.43 | benjk | there are oodles of providers |
| 01:56.48 | benjk | both SIP and IAX |
| 01:57.07 | benjk | and most of them have pretty good coverage with DID numbers for the US |
| 01:57.08 | nitrus^ | can they roll my numbers over? |
| 01:57.26 | benjk | that I don't know, but I would think so |
| 01:57.34 | nitrus^ | yeah it's FCC rules i believe |
| 01:57.39 | *** join/#asterisk bjohnson (n=bjohnson@i216-58-42-41.cybersurf.com) |
| 01:57.41 | nitrus^ | hmm i should probably go with IAX |
| 01:57.45 | nitrus^ | can you recommend someone? |
| 01:57.46 | benjk | as far as I know its a legal requirement to allow numbers to be ported |
| 01:57.47 | nitrus^ | im in socal |
| 01:58.19 | benjk | you probably want to pick two or three providers |
| 01:58.33 | benjk | and use a package that doesn't charge you any fixed fee, only on usage |
| 01:58.40 | seebs | Hey, people! |
| 01:58.51 | benjk | or, pick one provider with a package deal, and another one with a usage-only deal |
| 01:58.51 | nitrus^ | yeah i should check my average minute usage per month |
| 01:59.02 | nitrus^ | yeah i could use some backup lines |
| 01:59.02 | seebs | I have successfully (I think!) gotten the Zaptel drivers ported to NetBSD again. |
| 01:59.04 | benjk | this way you can route calls to another provider if you cannot get though |
| 01:59.07 | jontow | anyone use libiax much? |
| 01:59.10 | nitrus^ | right |
| 01:59.12 | seebs | I now have stupid newbie questions about asterix configuration. |
| 01:59.12 | benjk | gives you higher availability |
| 01:59.22 | jontow | seebs; get wct1xxp done? ;) |
| 01:59.29 | nitrus^ | ill look some up then |
| 01:59.35 | nitrus^ | is pricing pretty good from what you've seen? |
| 01:59.37 | seebs | No, just wcfxs, ztdummy, zaptel. |
| 01:59.52 | jontow | good deal.. we were that far i think |
| 01:59.52 | benjk | the pricing is pretty much the same everywhere |
| 01:59.59 | jontow | but i suspect it got all sorts of stale |
| 02:00.04 | seebs | I redid the port, looking partially at riz's code. I improved on it, I think, in a couple of spots. |
| 02:00.06 | benjk | some providers go very low to fetch market share when they start |
| 02:00.08 | jontow | and he did the vast majority of the work |
| 02:00.16 | benjk | then everybody goes to sign up and you get lots of congestion |
| 02:00.19 | seebs | I had two major bugs, one of which was painfully stupid, the other of which was only very stupid. |
| 02:00.19 | jontow | i was QA, effectively |
| 02:00.19 | jontow | :P |
| 02:00.34 | benjk | so better use a provider with a little higher rate and get better service |
| 02:00.35 | nitrus^ | now imma have to find something to do with the TA750 |
| 02:00.37 | seebs | Anyway, I'm up to the point of needing to know something about asterisk hardware configuration and setup. |
| 02:00.40 | nitrus^ | we're switching to cisco phones soon |
| 02:00.47 | jontow | :) good deal |
| 02:00.47 | nitrus^ | and if i get an IAX or sip provider |
| 02:00.47 | jontow | whatcha need? |
| 02:00.49 | nitrus^ | that thing is a brick |
| 02:01.03 | seebs | VoiceDAA did not bring up ISO link properly! |
| 02:01.03 | seebs | Module 2: Not installed |
| 02:01.07 | benjk | nitrus, the TA750 might still be usefiul as an FXS gatreway |
| 02:01.17 | seebs | I get this for my two FXO modules. Should I expect it to go away once the machine's plugged into a phone line? |
| 02:01.23 | nitrus^ | i have intercom systems running on it |
| 02:01.26 | nitrus^ | so ill keep it in the loop |
| 02:01.33 | jontow | i think you do need 'em plugged in, yeah |
| 02:01.36 | jontow | but it should see them? |
| 02:02.00 | nitrus^ | but as far as my current issue is concerned are there settings to tweak to try and fix the delay issues |
| 02:02.09 | benjk | I used Voicepulse's service when they still accepted non-US credit cards |
| 02:02.10 | nitrus^ | from what i can tell it seems purely at the time i bridge the FXS and FXO lines together |
| 02:02.20 | benjk | since youre in the US you won't have a problem with that |
| 02:02.28 | benjk | they service is good, they actually have a phone hotline |
| 02:02.38 | benjk | many providers only do email |
| 02:02.47 | benjk | and they have redundant servers |
| 02:02.59 | benjk | little more expensive than some other providers |
| 02:03.02 | seebs | Well, it "sees" them enough to determine that it can't get the link up. |
| 02:03.03 | benjk | but they do SIP and IAX |
| 02:03.07 | seebs | I assume it's a sanity check. |
| 02:03.24 | seebs | Anyway, I am wondering if someone can help me with the absolutely trivial configuration I want. |
| 02:03.35 | benjk | and Voicepulse officially support Asterisk |
| 02:03.46 | seebs | I have two FXO and two FXS. I want anything coming in on the first FXO port to be routed directly to the first FXS port, preserving caller ID, and I want the same setup for the other pair. |
| 02:03.49 | nitrus^ | they dont seem to have rates on their site |
| 02:03.54 | benjk | so you wont get any shit from their support desk when you say you're using Asterisk |
| 02:04.06 | benjk | you need to use connect.voicepulse.com |
| 02:04.06 | furby_singh | anyone here know app_bridge? or how to add audio to a bridged call? (Sorry to pester) |
| 02:04.13 | benjk | that's for asterisk business users |
| 02:04.17 | benjk | the other is residential |
| 02:04.23 | seebs | I don't need any routing or anything; I just want a transparent gateway between lines and phones that can record calls and log caller ID info. |
| 02:05.01 | nitrus^ | ahh |
| 02:05.22 | nitrus^ | excellent i can call afghanistan for 48 cents a minute |
| 02:05.28 | benjk | heh |
| 02:05.53 | *** join/#asterisk coppice (n=chatzill@229.166.17.210.dyn.pacific.net.hk) |
| 02:06.01 | benjk | in any event, there are loads of providers |
| 02:06.19 | benjk | no need to get stuck with Vonage |
| 02:06.25 | nitrus^ | yeah i agree |
| 02:06.52 | nitrus^ | if i can port my numbers and so long as the transition is pretty much smooth ill probably do it |
| 02:06.53 | benjk | locking customers into a single type of device is just stupid |
| 02:06.58 | nitrus^ | i agree |
| 02:07.08 | nitrus^ | it's the best way to charge for more features though |
| 02:07.33 | benjk | but then they should at least offer something better than FXO for business customers |
| 02:07.34 | nitrus^ | though it seems like it costs them more in the end anyways |
| 02:07.50 | nitrus^ | yeah their FXO is pretty crappy IMO |
| 02:07.55 | benjk | like if you have 6 or 8 channels, they should provide you with a little box that gives you a virtual T1 with 8 channels |
| 02:08.01 | nitrus^ | i could never get fax to work through it |
| 02:08.04 | nitrus^ | i had to go to land line |
| 02:08.23 | benjk | fax over voip is still in its early beginnings |
| 02:08.44 | benjk | and asterisk isn't dping all too well with fax over voip either |
| 02:08.51 | *** join/#asterisk McDaver7 (n=tech_sys@CPE-70-94-239-225.mn.res.rr.com) |
| 02:09.38 | Lyfe | anyone know why i would get an error about configuring a zaptel clone card in the same machine as a sangoma a200? (the a200 is a 4-port unit, and occupies channels 1-4, the zap clone is on channel 5). |
| 02:09.49 | seebs | What error? |
| 02:09.57 | Lyfe | the drivers load fine, but ztcfg -vvv spits out "ZT_CHANCONFIG failed on channel 5: Invalid argument (22)" |
| 02:10.03 | seebs | Hmm. |
| 02:10.04 | nitrus^ | yeah they offer a fax line free |
| 02:10.05 | nitrus^ | no wonder |
| 02:10.07 | nitrus^ | it's crap. |
| 02:10.29 | Lyfe | it's trying to configure it with fxsks, so it's not that i chose the wrong signalling type. |
| 02:10.43 | seebs | Well, 22 is indeed EINVAL. Not sure where to go with that. Anything interesting in the driver load messages? |
| 02:11.19 | Lyfe | nope. it finds the wcfxo just fine. |
| 02:11.24 | seebs | Hmm. |
| 02:11.27 | *** join/#asterisk mick_linux (n=mick@adsl-153-159-186.mia.bellsouth.net) |
| 02:11.32 | seebs | Well, you have exceeded my very limited knowledge. |
| 02:11.56 | kusznir | are there any known issues with 1.2.7.1 that cause iax channel instability with some other remote ends? My connections to FWD (over iax) and voipjet are both very unstable in different ways. |
| 02:11.56 | Lyfe | what's bugging me is that i know the cards worked before i added the sangoma a200 in.. which is a shame, cause i like the sangoma hardware sofar. |
| 02:12.11 | justinu|laptop | iax unstable? impossible |
| 02:12.15 | kusznir | They worked find with 1.0.10 (I think...) |
| 02:12.15 | mick_linux | what is the best way to test asterisk? (with POTS, there is a test phone number that will tell you if the call was a success) |
| 02:13.00 | kusznir | FWD is returning busy on the lines whereas voipjet is taking a minute to complete the call. Every now and then, things will start to work for both providers, then stop again (without me changing or reloading anything) |
| 02:13.34 | seebs | Whee! ztcfg reboots the machine. :) |
| 02:13.42 | kusznir | The problems started when I upgraded asterisk. |
| 02:13.42 | seebs | Well, I have a lot of fun ahead of me. |
| 02:14.27 | seebs | And, just to be quite clear: If the first two channels on my card are probed as "auto FXS", that means I put "fxoks=1-2", right? |
| 02:15.16 | kusznir | benjk nitrus^ vonage does offer direct SIP termination to business-class customers. You have to call inside sales. When I looked into it, though it was about 10x what you would pay with a provider like voipjet, asterlink, or sixtel. |
| 02:15.51 | justinu|laptop | oh, vonage will deal if you decide you want to cancel |
| 02:15.54 | *** part/#asterisk McDaver7 (n=tech_sys@CPE-70-94-239-225.mn.res.rr.com) |
| 02:16.25 | *** join/#asterisk McDaver7 (n=tech_sys@CPE-70-94-239-225.mn.res.rr.com) |
| 02:16.43 | kusznir | Oh, yea..they also have a third plan that's less money (like $8/mo, IIRC) that they'll only offer you if you're on the $15 and you want to cancel :) |
| 02:17.00 | kusznir | (or at least, they did about 1.5 years ago) |
| 02:17.35 | justinu|laptop | i told them i would reconsider if they gave me the sip credentials |
| 02:17.35 | De_Mon | seebs youre so lucky! |
| 02:17.41 | *** join/#asterisk rene1 (n=rene1@gea-gye-internet.telconet.net) |
| 02:17.44 | rene1 | hello |
| 02:18.01 | McDaver7 | Hello, can somebody help me with a new Trixbox installation problem adding an FXO card? I'm receiving a ZT_chanconfig failed message. |
| 02:18.10 | seebs | I'm lucky? |
| 02:18.17 | rene1 | is there someone that is experienced on alcatel 4400 cdrs retrieving? |
| 02:18.53 | furby_singh | renel: an alcatel engineer? </sarcasm> |
| 02:18.58 | rene1 | i am looking at the so called tickets under /usr4/account |
| 02:19.18 | kusznir | Justinu|laptop: did they? |
| 02:19.22 | justinu|laptop | nope :) |
| 02:19.27 | kusznir | didn't think so :) |
| 02:19.44 | kusznir | Although I've seen some instructions for a few different ATAs on how to extract the credentials :) |
| 02:19.50 | justinu|laptop | heh, nice |
| 02:20.32 | kusznir | When I looked into it, though, it just wasn't worth it...there are companies out there that actually support asterisk and cost less (sometimes a lot less). So I went that route. |
| 02:20.43 | benjk | justinu, nice move |
| 02:20.50 | benjk | with vonage |
| 02:20.57 | benjk | threatening to cancel |
| 02:20.58 | benjk | heh |
| 02:21.41 | benjk | I have to reengineer ATA behaviour all the time here |
| 02:21.59 | benjk | cause in Japan this is the only way you get VoIP |
| 02:22.03 | justinu|laptop | i didn't threaten, i did |
| 02:22.08 | benjk | heh |
| 02:22.28 | benjk | in Japan everybody who has internet gets an analog jack in their cable or DSL modem |
| 02:22.42 | benjk | and the SIP adapter is built in to the modem |
| 02:22.52 | justinu|laptop | is it actually sip? |
| 02:22.57 | benjk | and your ISP is the only provider you can get service from |
| 02:23.12 | benjk | it is SIP in most cases, Yahoo use MGCP though |
| 02:23.16 | ceb` | good night ppl |
| 02:23.19 | benjk | Yahoo BB |
| 02:24.01 | benjk | it doesn't even travel through your modem |
| 02:24.16 | benjk | it has a separate IP address and is a separate PPPoE session |
| 02:24.36 | benjk | so basically you have internet on one session and VoIP on the other session |
| 02:24.53 | benjk | no way to even sniff the packets |
| 02:24.59 | justinu|laptop | do they garauntee qos? |
| 02:25.11 | benjk | not on the residential service |
| 02:25.22 | justinu|laptop | that sucks |
| 02:25.31 | benjk | but for all practical purposes as seen by the end user it is more a form of VoDSL |
| 02:25.37 | benjk | even though it uses IP |
| 02:26.14 | justinu|laptop | i thought DSL was ATM based... couldn't they set up a VC for the voice channel? |
| 02:26.21 | seebs | Well, cool! |
| 02:26.30 | benjk | I guess they could, but they dont |
| 02:26.33 | seebs | I ran ztcfg, got a warning about interfaces being configured with the wrong kind of signalling. |
| 02:26.44 | LoneShadow | justinu|laptop: why not sunrocket ? |
| 02:26.46 | seebs | So I reconfigured the other way, and got a kernel panic in zt_tone_free. WOOO! |
| 02:27.05 | justinu|laptop | i got something better |
| 02:27.29 | LoneShadow | how much better ? |
| 02:27.34 | benjk | Sunshine! |
| 02:27.54 | justinu|laptop | like the ppl who provide service to sunrocket and vonage ;) |
| 02:28.14 | *** join/#asterisk L|NUX (n=linux@unaffiliated/lnux/x-10290) |
| 02:28.34 | benjk | justinu is in California, he's got something better than sunrocket all year round: sunshine |
| 02:28.37 | LoneShadow | but what did they offer extra ? |
| 02:28.38 | *** part/#asterisk McDaver7 (n=tech_sys@CPE-70-94-239-225.mn.res.rr.com) |
| 02:28.42 | *** join/#asterisk techie (n=gus@voipops.net) |
| 02:28.55 | rene1 | many asterisk people used to work with propietary pbx systems in the past. so the posibility to bump into an expert in such devices in #asterisk is very real |
| 02:29.10 | rene1 | maybe at a different time |
| 02:29.14 | LoneShadow | <- from cali too :P |
| 02:29.15 | justinu|laptop | heh, too much sunshine... august was damn hot |
| 02:29.29 | benjk | same here |
| 02:29.45 | LoneShadow | from San Jose here |
| 02:29.56 | furby_singh | renel: you are right of course :-) I'm full of caffine and feeling mischevous |
| 02:30.08 | justinu|laptop | we saw 119F... new record for this area |
| 02:30.26 | seebs | Do we have anyone here who understands the zaptel driver? I am trying to figure out a crash involving free_tone_zone, and I don't understand the allocation/deallocation strategy for this driver. |
| 02:30.32 | LoneShadow | central or LA? |
| 02:30.33 | benjk | wouldnt that be about 25 EUR now? |
| 02:30.51 | rene1 | furby_singh: no problem |
| 02:30.57 | *** join/#asterisk Ryushin (i=proxy@windwalker.openinnovations.com) |
| 02:31.00 | justinu|laptop | LA burbs |
| 02:31.06 | benjk | seebs, zaptel is a total mystery |
| 02:31.10 | coppice | free_tone_zone with every 5 purchased :-) |
| 02:31.16 | seebs | I was afraid of that. |
| 02:31.25 | benjk | it is a wonder of magic |
| 02:31.28 | LoneShadow | used to be near Thousand oaks |
| 02:31.34 | justinu|laptop | yeah, i'm near TO |
| 02:31.34 | benjk | its amazing that it works at all |
| 02:31.40 | benjk | because it probably shouldnt |
| 02:32.00 | benjk | Linus Torvalds said he didn;t want "that crap in my kernel" |
| 02:32.02 | LoneShadow | WLV ? |
| 02:32.13 | Qwell | benjk: again - link? |
| 02:32.14 | seebs | I am not surprised, it looks like total crap. |
| 02:32.31 | *** join/#asterisk droops (n=root@adsl-147-224-61.jan.bellsouth.net) |
| 02:32.37 | Qwell | otherwise, you're trolling, and you will be removed |
| 02:32.43 | justinu|laptop | agoura |
| 02:33.01 | LoneShadow | I lived in Oakpark :P |
| 02:33.14 | justinu|laptop | yeah |
| 02:33.28 | LoneShadow | nice place |
| 02:33.31 | justinu|laptop | it's ok |
| 02:33.34 | justinu|laptop | kinda generic |
| 02:33.48 | coppice | Qwell: he did say that |
| 02:33.56 | Qwell | coppice: link? ;) |
| 02:33.57 | LoneShadow | like it better than San Jose, and not too far from sunset blvd :D |
| 02:34.20 | coppice | Qwell: somewhere in the history of the kernel mailing list |
| 02:34.20 | justinu|laptop | it's still a far drive from oakpark to agoura |
| 02:34.39 | Qwell | and how completely out of context was it? |
| 02:34.40 | justinu|laptop | if you get lucky with PCH you might make it in 30 minutes |
| 02:35.29 | *** join/#asterisk McDaver7 (n=tech_sys@CPE-70-94-239-225.mn.res.rr.com) |
| 02:36.00 | seebs | I dunno, but if I see "bcopy" in a driver, that always makes me nervous. |
| 02:36.50 | coppice | Qwell: not sure. he took a dislike to zapata from the original tormenta 1 card's need for 8000 interrupts per second. after that it might have been kneejerk reaction. he's not keen on anything with high unstoppable interrupt flows, though, and for good reason |
| 02:36.58 | LoneShadow | far drive from oakpark to aguoura ?? its right next to it, 10-15mins |
| 02:37.25 | justinu|laptop | oops, i meant oakpark to PCH |
| 02:37.28 | justinu|laptop | er sunset |
| 02:38.07 | LoneShadow | yea |
| 02:38.27 | coppice | Qwell: do job interviews at Digium test that you treat every comment at trolling? |
| 02:38.49 | Qwell | coppice: unsubstantiated comment, from somebody who trolls daily |
| 02:39.35 | *** part/#asterisk McDaver7 (n=tech_sys@CPE-70-94-239-225.mn.res.rr.com) |
| 02:39.35 | coppice | ah, so I guess they do check that |
| 02:39.56 | *** join/#asterisk wunderkin (n=kev@ip68-226-113-228.ph.ph.cox.net) |
| 02:40.32 | X-Rob | heya coppice |
| 02:40.55 | coppice | hiya. hhated any good colos lately? :-) |
| 02:41.05 | Qwell | coppice: Do you have a problem with the way I handle trolls? |
| 02:41.38 | X-Rob | coppice, no, it's been good. I hardly hate anyone at the moment! |
| 02:41.44 | coppice | Qwell: I have a problem with the work troll being brainlessly applied to any comment, even the simplest statement of fact |
| 02:41.53 | X-Rob | s/work/word/ |
| 02:41.59 | Qwell | coppice: an unsubstantiated comment is not fact |
| 02:42.18 | *** join/#asterisk McDaver7 (n=tech_sys@CPE-70-94-239-225.mn.res.rr.com) |
| 02:42.28 | coppice | Qwell: when you start differentiating I'll take you seriously |
| 02:43.00 | Qwell | coppice: I suggest you look through your logs, and find the last time I called somebody a troll (while actually being serious). :) |
| 02:43.29 | mog | you were asking for me Qwell |
| 02:43.33 | seebs | I can't say what Linus did or didn't say about the driver; I can say that I wouldn't include it in a kernel tree in its current state. |
| 02:43.40 | furby_singh | app_chan! app_chan! app_chan! To the theme if Batman, anyone here know anything about it? |
| 02:43.43 | Qwell | mog: yeah...like...earlier. It's all good |
| 02:43.43 | coppice | Kevin must really stick out at Digium. he's the only one who doesn't stoop to this kind of crap these days |
| 02:43.48 | furby_singh | s/if/of/ |
| 02:44.06 | Qwell | mog: ugh |
| 02:44.14 | mog | fevers rock |
| 02:45.18 | Qwell | coppice: okay, next time somebody says something that I feel is untrue, I'll just let it go, and let people believe that it is true. Happy? |
| 02:46.05 | seebs | I think there's a middle ground; you could correct them without threatening them with a kick. |
| 02:46.09 | seebs | That's what I always do. |
| 02:46.14 | Qwell | seebs: I did that the last few times |
| 02:46.32 | Qwell | seebs: with that exact same comment, from the exact same person |
| 02:46.37 | X-Rob | Qwell, how about saying 'I don't believe that's correct, can you offer any proof?' rather than calling troll? |
| 02:46.39 | furby_singh | does anyone know of a good asterisk nebies channel? |
| 02:46.47 | seebs | Well, I have no evidence either way on the topic, so I can't comment. |
| 02:46.50 | Qwell | X-Rob: logs say I have several times :) |
| 02:46.58 | X-Rob | Qwell, I don't know why I'm getting involved, actually |
| 02:47.02 | X-Rob | I should be coding. |
| 02:47.30 | seebs | So, anyway, can anyone confirm or deny my theory that "VoiceDAA did not bring up ISO link properly!" means "you cannot use this hardware without a phone line"? |
| 02:48.16 | furby_singh | how would I inject audio into an active call? I'll post anyone who helps a bottle of Czech beer! Becherovka if it works :-) |
| 02:48.40 | coppice | seebs: are you addressing actual porting issues, or is this stuff just as problematic on the original BSD? |
| 02:48.41 | seebs | Furby, can you clarify what you mean? Do you want to replace the active call, or add data to it? Does the data go to both parties? |
| 02:48.50 | seebs | coppice: I think I'm done with porting, I don't know. |
| 02:49.06 | Qwell | furby_singh: further, should it be triggered by one of the parties? |
| 02:49.10 | coppice | these don't seem like the fine differences between BSDs |
| 02:49.10 | seebs | At this point, I am getting no crashes, so I'm thinking it's nearly time to install the box. |
| 02:49.11 | furby_singh | thanks seebs :-) I need to play an audio file to a caller and callee on an active call, worst case even just the callee |
| 02:49.23 | Qwell | If so, you could use something in features.conf |
| 02:49.32 | furby_singh | Qwell: to be triggered ideally by caller or manager |
| 02:49.35 | seebs | Oh, yeah, I'm long past the porting differences, some of which were rather large. |
| 02:49.44 | Qwell | caller can do it with a feature code in features.conf |
| 02:49.59 | seebs | I'm down to trivia like "is it safe to call free on a potentially-null pointer", where I think NetBSD's kernel-malloc still has pre-ansi semantics. |
| 02:50.13 | furby_singh | Qwell: that's the angle I've looked at most but can't work out how to play to callee without dropping call after sample played (sample played through callplan) |
| 02:50.22 | seebs | But really, I'm to the point of "hey, should I have gotten this error message, or does it indicate a bug in my port". |
| 02:50.29 | Qwell | furby_singh: take a look at features.conf |
| 02:50.35 | Qwell | You could add a simple Playback() |
| 02:50.43 | furby_singh | Qwell: sorry, been looking at this too much, of course can play audio to callee with Playback() |
| 02:50.55 | furby_singh | Qwell: heh, beat me too it :-) |
| 02:51.01 | Qwell | in 1.4, you have the option of sending to both, I believe |
| 02:51.27 | Qwell | if not, there is a patch on the tracker that should still apply |
| 02:51.28 | furby_singh | Qwell: yeah, I hear that, what's your oppinion on using 1.4 for heavy use server? |
| 02:51.46 | furby_singh | Qwell: what is tha patch if you know? |
| 02:51.59 | Qwell | furby_singh: I don't recall if it was committed or not.. |
| 02:52.03 | *** join/#asterisk `Sauron (i=sauron@h-69-3-12-50.hstqtx02.covad.net) |
| 02:53.14 | furby_singh | Qwell, thanks for the input, I think I gotta wait for 1.4 it seems, do you know when it's due out? (I'm sure I should know this but don't) |
| 02:53.23 | seebs | So, anyone patient willing to walk me through a fairly trivial 2-line configuration? |
| 02:53.24 | coppice | seebs: do you end up with something that will build on any BSD, or do you have to fork? |
| 02:53.41 | seebs | I have #ifdef'd code for FreeBSD and NetBSD. |
| 02:54.06 | seebs | There's no way to eliminate ALL of the #ifdefs, but a lot of them can be reduced to single-line macros that can then be used in dozens of places without additional #ifdefs. |
| 02:57.53 | furby_singh | in features.conf is there anyway to simultainiously call Playback to caller AND callee? |
| 02:58.22 | furby_singh | or is there a way to prompt the feature to be called by the manager interface? |
| 03:00.51 | kusznir | Hi all: I've got a couple asterisk-specific questions (I know, a change in topic :) ) |
| 03:01.08 | seebs | It is, pretty much! |
| 03:01.32 | seebs | BTW, I am still looking for someone willing to pretty much handhold me through a simple configuration, since I don't know what's going on yet. |
| 03:02.30 | kusznir | I've finally isolated my troubles to CID values. As per the suggestion printed out when I upgraded asterisk, I changed from using SetCIDName and SetCIDNum to "CALLERID(number)=" and "CALLERID(name)=". However, when that gets set, two carriers won't place calls. |
| 03:03.17 | kusznir | I have one phone set up to bypass the CID setting commands, and it is able to complete calls without issue. (Although for some reason, my cell phone recieves the CID number with a "+" in front, breaking callback....haven't figured that out). |
| 03:03.17 | justinu|laptop | some carriers won't let you change the CID |
| 03:03.34 | justinu|laptop | + is E164 |
| 03:03.53 | kusznir | The carriers in question are FWD and voipjet. Previously, I was using the SetCIDName and SetCIDNumber without issue to set the values to the same thing I am now with the new methods. |
| 03:04.12 | kusznir | justinu|laptop: So I should be able to dial +<num> from my cell, and it should go through? |
| 03:04.22 | kusznir | Or only if I registered my number on E164? |
| 03:04.27 | justinu|laptop | yeah... + is the the international access code |
| 03:05.24 | kusznir | So is that happening because I don't have a 1 in my CID setting on the device? (eg, NXXNXXXXXX instead of 1NXXNXXXXXX)? |
| 03:05.36 | justinu|laptop | it's possible |
| 03:06.06 | kusznir | Any suggestions on the setting CID info with CALLERID(...) breaking some carriers? |
| 03:06.07 | justinu|laptop | i encourage you to experiment |
| 03:06.15 | justinu|laptop | no, that's not the problem |
| 03:06.35 | justinu|laptop | i also suggest turning on sip debug, and looking at the messages |
| 03:07.31 | kusznir | They're both going through IAX. Neither returns a useful message. voipjet completes the call, but after a 1+ minute delay; voipjet just sends a hangup code that translates to "normal hangup" or something like that (I think its IAX code 16) |
| 03:07.44 | justinu|laptop | ah, sorry, can't help you with IAX |
| 03:07.49 | kusznir | err..the second voipjet = fwd :) |
| 03:09.29 | kusznir | I just bypassed my CALLERID(...) lines in my extensions.conf, and calls now go through as they should. |
| 03:12.20 | kusznir | However, now the SetCIDNum and SetCIDName commands also are breaking the ability to call. |
| 03:12.45 | kusznir | (now = with asterisk 1.2.10; !now = asterisk 1.1.10) |
| 03:13.00 | kusznir | Is there a different way I should be setting my CID info? |
| 03:13.02 | justinu|laptop | that sucks |
| 03:18.26 | *** join/#asterisk twisted[asteria] (n=twisted@pdpc/supporter/active/twisted) |
| 03:18.26 | *** mode/#asterisk [+o twisted[asteria]] by ChanServ |
| 03:23.58 | harryvv | a fast busy at a iax provider typically means that all circuits are busy? I know that is the case in a telco. |
| 03:25.51 | *** part/#asterisk McDaver7 (n=tech_sys@CPE-70-94-239-225.mn.res.rr.com) |
| 03:28.01 | coppice | has anyone actually seen a Pika card working with Asterisk? |
| 03:29.30 | X-Rob | Pikachoo! |
| 03:29.46 | coppice | bless you |
| 03:30.17 | JunK-Y | coppice: ya, i saw it before. |
| 03:31.29 | De_Mon | kusznir use SET() |
| 03:31.29 | coppice | does it emulate zaptel, like the current sangoma, or does it use its own channel driver? |
| 03:31.49 | *** part/#asterisk mick_linux (n=mick@adsl-153-159-186.mia.bellsouth.net) |
| 03:31.56 | JunK-Y | ive no idea, it was like 1.5 years ago. |
| 03:32.26 | coppice | really? that long ago? they must have really been dragging this stuff out |
| 03:32.36 | De_Mon | kusznir http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+Set |
| 03:35.24 | furby_singh | can a feature be called by the manager interface? i.e. setup a feature in features.conf and call it without the punter having to hit buttons on the phone? |
| 03:35.38 | furby_singh | and if so, how? |
| 03:35.46 | furby_singh | and if how, thanks :-) |
| 03:36.39 | znoG | question.. if I have the following setup: [SIP provider]<-->[Asterisk server]<-->[SIP client] |
| 03:37.00 | znoG | If a call comes in from SIP provider to SIP client |
| 03:37.29 | znoG | should SIP provider talk directly to SIP client by-passing the Asterisk server? (reinvite??) |
| 03:37.42 | znoG | if so, should I configure the SIP account on Asterisk server for SIP provider AND SIP client so that they both do reinvite? (canreinvite=yes on both) |
| 03:50.58 | kusznir | znoG: Some things to consider re: your second question. If the call is bypassing your asterisk server, you do not have any means for logging CDR accurately, interrupting the call, transfering/redirecting/parking, etc. Also, for this to work, your sip device would have to be directly accessable by the sip provider (aka not behind a NAT/firewall, etc) |
| 03:51.09 | kusznir | (I don't think stun helps in this case, btw) |
| 03:51.32 | znoG | the Asterisk server is not behind NAT, the client is (SJPhone) |
| 03:52.01 | kusznir | So I don't think reinvite is doable then. From what I understand, few people actually use reinvite anyway. |
| 03:52.34 | znoG | the case here is that the SJPhone client is in a far away country from the Asterisk server, same as the SIP provider |
| 03:52.49 | znoG | so coming through here creates a long path which, if possible, should be avoided |
| 03:52.52 | *** join/#asterisk bmg505 (n=leon@c1-182-3.rndf.isadsl.co.za) |
| 03:53.00 | znoG | i think i'll probably setup a proxy server closer to where the SIP provider and the client are |
| 03:54.08 | znoG | s/proxy/asterisk |
| 03:59.42 | *** join/#asterisk Ryushin (i=proxy@windwalker.openinnovations.com) |
| 04:00.45 | *** join/#asterisk wasim (n=wasim@pdpc/supporter/active/wasim) |
| 04:07.14 | *** join/#asterisk jp_ast (n=jeep_jer@71-33-191-145.hlrn.qwest.net) |
| 04:08.16 | jp_ast | Is it possible for an agent to do an attended transfer on a SIP phone from a queue? |
| 04:08.53 | *** join/#asterisk DEader (n=ceo@ool-44c6ac15.dyn.optonline.net) |
| 04:09.20 | DEader | can someone tell me how to isntall the h264 video codec on asterisk |
| 04:10.07 | denon | DEader: http://www.google.com/search?sourceid=navclient&ie=UTF-8&rls=GGLF,GGLF:2005-15,GGLF:en&q=h%2e264+asterisk |
| 04:11.54 | DEader | denon |
| 04:11.59 | *** join/#asterisk Mike800 (n=mike800@ip68-231-212-10.oc.oc.cox.net) |
| 04:12.15 | DEader | what the diference between asterisk and the trunk version of asterisk |
| 04:12.24 | DEader | i can't seem to find where to download it |
| 04:12.28 | mog | trunk is the development branch |
| 04:12.39 | mog | svn co http://svn.digium.com/svn/asterisk/trunk asterisk-trunk |
| 04:12.57 | denon | mog: of he cant find the download link, you really expect svn commands to be helpful? |
| 04:13.12 | Mike800 | lol |
| 04:13.17 | mog | i just answered the question |
| 04:13.19 | DEader | that link don't work |
| 04:13.21 | denon | hehe |
| 04:13.26 | Mike800 | plenty of directions here: http://www.asterisk.org/download |
| 04:13.27 | denon | DEader: http://www.asterisk.org/download |
| 04:13.31 | Mike800 | :-D |
| 04:13.40 | mog | that was too funny |
| 04:13.54 | denon | mog: I have a 6th sense about these things |
| 04:13.59 | denon | 6th, sick, whatever |
| 04:14.02 | mog | oh i knew denon |
| 04:14.08 | mog | i just had to be truthful |
| 04:14.08 | DEader | doesn't say nothing about trunk |
| 04:14.16 | denon | DEader: forget about trunk |
| 04:14.20 | denon | just get the tarballs |
| 04:14.30 | denon | (but yes, it does talk about trunk, scroll down) |
| 04:14.58 | DEader | but does the tarball support h264 code |
| 04:14.59 | DEader | c |
| 04:16.10 | mog | no |
| 04:16.14 | mog | trunk does |
| 04:17.57 | DEader | how do i install trunk |
| 04:19.19 | denon | DEader: did you read that link? |
| 04:19.28 | denon | or would you like me to copy and paste the page into here? |
| 04:20.24 | DEader | copy |
| 04:20.41 | denon | whoo boy.. |
| 04:20.59 | mog | DEader: http://www.asterisk.org/download |
| 04:21.01 | mog | go there |
| 04:21.02 | mog | read |
| 04:21.07 | denon | rhetorical questions get you nowhere |
| 04:21.14 | mog | we are hear to serve, but not chew your food for ya |
| 04:21.16 | mog | ^_^ |
| 04:21.37 | denon | but for $300 and your ssh login, it'll be done in 30 minutes |
| 04:21.54 | denon | mog'll chew your food for a price |
| 04:21.56 | kusznir | DEader: trunk is just a name for the bleeding edge of asterisk. They're directing you to go get the source code, compile it, and install that version of asterisk. |
| 04:21.58 | mog | lol |
| 04:22.37 | kusznir | In some respects, trunk is so new, it doesn't even have a version number. The codec you're looking for isn't released yet in the normal builds... |
| 04:22.41 | Un1x | mog: when is TRUNk going to go Mainstream? |
| 04:23.23 | denon | isnt that sort of an oxymoron? |
| 04:23.33 | denon | the second trunk goes mainstream, ... trunk is newer again |
| 04:24.48 | Un1x | no denon sorry i ment when asterisk is going to be, 1.3.X |
| 04:25.05 | Qwell | Un1x: never |
| 04:25.09 | Un1x | err |
| 04:25.18 | Un1x | 4 too many istakes tonite |
| 04:25.20 | Un1x | *mistakes |
| 04:25.24 | Un1x | sorry sleepy :( |
| 04:25.55 | Corydon76-home | Odd minor versions are indications of development versions |
| 04:26.12 | DEader | kusznir do i have to rm the previous version of asterisk i have install before installing the trunk version |
| 04:26.42 | Un1x | i thought this wasn't tier 2 support :P |
| 04:27.00 | Corydon76-home | This isn't. Neither is -dev |
| 04:27.26 | *** join/#asterisk kusznir (n=kusznir@66-233-98-93.lew.clearwire-dns.net) |
| 04:27.30 | kusznir | Umm..its a good idea to rm it or move it to another location. Strictly speaking it shouldn't be required. |
| 04:31.08 | DEader | let me see if i understand this |
| 04:31.34 | DEader | all i have to do is download the svn trunk version |
| 04:31.42 | DEader | then install it has normal? |
| 04:32.00 | kusznir | download tarball, compile, and install. |
| 04:32.30 | DEader | tarball ? |
| 04:32.35 | kusznir | .tar.gz file |
| 04:32.40 | DEader | oh |
| 04:32.55 | *** join/#asterisk tessier (n=treed@adsl-75-5-99-178.dsl.sndg02.sbcglobal.net) |
| 04:33.05 | DEader | on the site i'm not seeing anywhere to download the truck tarball |
| 04:33.07 | kusznir | You should be able to find instructions for "installing asterisk from source" or "installing asterisk from svn". Those instructions are quite revelavent. |
| 04:33.45 | DEader | its telling me how to checked out the 1.2 branch |
| 04:34.00 | kusznir | Whats the svn command line? |
| 04:34.37 | DEader | svn checkout http://svn.digium.com/svn/asterisk/trunk asterisk |
| 04:34.59 | DEader | then it said |
| 04:34.59 | kusznir | Ok, that is what you want to do...that will get you trunk (note that trunk is at the end of the URL) |
| 04:35.16 | DEader | yeah |
| 04:35.22 | DEader | i run that command already |
| 04:35.39 | DEader | but i'm now confuse on what to do next |
| 04:35.43 | kusznir | Actually, the last part should be asterisk-trunk, not asterisk. |
| 04:36.04 | kusznir | Now you have the most up-to-date source for asterisk. You now need to compile and install it. |
| 04:36.32 | DEader | ok |
| 04:36.42 | DEader | did that command |
| 04:36.57 | DEader | download the trunk source |
| 04:38.14 | kusznir | Now you should be abe to cd into the directory, run ./configure, then when it finishes, run "make", and when it finishes, running "make install" will install it in your system (but it would be a good idea to remove your old asterisk before running make install, as it may install in a different place than your old one, and you don't know which version of asterisk your OS will find then) |
| 04:38.16 | *** join/#asterisk encode (n=encode@blah.i.hate.w1ndo.ws) |
| 04:38.29 | encode | hi people |
| 04:38.43 | encode | i'm trying to compile the zaptel driver for my digium tdm400p wildcard |
| 04:39.18 | DEader | ok |
| 04:39.20 | encode | ive got a 2.6.15 kernel that was compield with gcc-4.0, and after compiling the latest zaptel source I get the following error when trying to modprobe it: |
| 04:39.23 | encode | zaptel: disagrees about version of symbol copy_from_user |
| 04:39.26 | encode | zaptel: Unknown symbol copy_from_user |
| 04:39.40 | encode | latest source meaning zaptel-1.2.9.1 |
| 04:40.11 | encode | i also get some errors from wctdm |
| 04:40.22 | encode | any ideas as to what i should be looking at? |
| 04:40.27 | DEader | so just use the rm command to remove the folder oft he asterisk right |
| 04:40.35 | DEader | or there is another way? |
| 04:40.56 | encode | why would i want to remove the folder off the asterisk? |
| 04:41.00 | encode | i want to use the driver |
| 04:41.25 | benjk | LOL |
| 04:41.29 | kusznir | encode, DEader is not talking to you! |
| 04:41.30 | kusznir | :) |
| 04:41.45 | encode | oh, lol |
| 04:41.47 | encode | sorry |
| 04:41.59 | kusznir | DEader: depends on how your old asterisk got there...if it was a binary package (rpm, .deb, .ipkg, etc), then you want to use the package manager for your OS to remove it. |
| 04:42.10 | kusznir | If not, then life just got a lot harder :( |
| 04:42.25 | kusznir | (you have to go find all the places asterisk put stuff and remove them -- except for /etc/asterisk) |
| 04:47.05 | *** join/#asterisk murf (n=steve_mu@216.166.159.235) |
| 04:47.34 | benjk | you can use my uninstall script http://www.sunrise-tel.com/downloads/AsteriskShellScripts.zip |
| 04:47.41 | benjk | its bash |
| 04:48.11 | benjk | archives your configs and /var/lib/asterisk stuff too |
| 04:50.09 | *** join/#asterisk `Tingles` (n=tingles@S01060011d8ecb1d0.cg.shawcable.net) |
| 04:54.17 | *** join/#asterisk Ryushin (i=proxy@windwalker.openinnovations.com) |
| 04:55.47 | *** join/#asterisk Cyt (n=danielcy@athedsl-141532.otenet.gr) |
| 04:58.50 | justdave | I have three phones in a call group, so there's one number you can call that rings all three of them. Right now, when someone answers one of those, the call belongs to the phone that answers, and the other two get a dial tone when you pick them up. Is there a way to set up * so that if a call goes to that ring group, that all three phones can still join the call if they pick up? |
| 05:01.34 | denon | xfer the call to a meetme :) |
| 05:04.20 | *** join/#asterisk JunK-Y (n=junky@modemcable205.175-81-70.mc.videotron.ca) |
| 05:04.42 | jp_ast | is anyone able to do attended transfers for a call from a queue? i'm not finding jack on this topic |
| 05:10.26 | DEader | kusznir i finish install the trunk |
| 05:10.44 | DEader | how to i check the version thats running to make sure thats its the right one |
| 05:15.01 | denon | DEader: show version |
| 05:15.32 | denon | or asterisk -rvx "show version" from a command line |
| 05:16.24 | denon | er, -rx |
| 05:16.27 | denon | same diff |
| 05:16.34 | DEader | is Asterisk 1.2.11 the latest trunk version |
| 05:16.35 | DEader | ? |
| 05:16.48 | Corydon76-home | 1.2 is not trunk |
| 05:16.58 | Corydon76-home | 1.2 is the release branch |
| 05:17.08 | DEader | huh |
| 05:17.16 | Corydon76-home | And as the topic says, the latest release is 1.2.12.1 |
| 05:17.34 | denon | he actually wants trunk, for some video stuff |
| 05:18.04 | DEader | i thought i removed that version |
| 05:28.05 | *** join/#asterisk sai2 (n=sai@203-97-220-141.cable.telstraclear.net) |
| 05:29.30 | *** join/#asterisk LoneShadow (n=duh@59.92.222.184) |
| 05:29.48 | sai2 | I have a situation where asterisk works great for internal calls, and a call from somebody outside the firewall to somebody inside (or vice versa), but when two people outside the firewall try to talk to each other, they get no audio after the call connects. I have set canreinvite=no in sip.conf. Is there some other trick needed to make this work? |
| 05:32.15 | DEader | sai2 open port 10000 - 20000 on your router |
| 05:32.23 | DEader | for rtf |
| 05:32.30 | DEader | or something like that |
| 05:34.10 | sai2 | yes, already open |
| 05:41.16 | *** join/#asterisk x86_ (n=x86@p3m/member/x86) |
| 05:43.40 | *** join/#asterisk Rahail (i=rahail1@209.19.88.243) |
| 05:44.31 | *** join/#asterisk radovoip (n=radovoip@xd141.sstar.com) |
| 05:45.23 | *** join/#asterisk adelas (n=booger@rrcs-24-199-21-138.west.biz.rr.com) |
| 05:48.21 | *** join/#asterisk shtoom (n=godson@59.184.31.213) |
| 05:50.15 | *** join/#asterisk SwK (n=Silik0nJ@12-218-74-89.client.mchsi.com) |
| 05:55.28 | *** part/#asterisk justinu|laptop (n=Justin@dsl081-083-093.lax1.dsl.speakeasy.net) |
| 05:57.08 | *** join/#asterisk SwK_ (n=Silik0nJ@12-218-74-89.client.mchsi.com) |
| 06:06.53 | *** join/#asterisk Ryushin (i=proxy@windwalker.openinnovations.com) |
| 06:07.30 | DEader | ok i got this runnign SVN-trunk-r42982 |
| 06:07.37 | DEader | is that the latest trunk |
| 06:07.40 | DEader | ? |
| 06:18.26 | *** join/#asterisk jart (n=user@ool-4357221d.dyn.optonline.net) |
| 06:20.53 | tecnico | DEader: 43018 ? |
| 06:21.55 | radovoip | could use a little direction please |
| 06:22.37 | DEader | hrm |
| 06:22.50 | DEader | is the latest version 43018 |
| 06:22.54 | DEader | ? |
| 06:23.00 | drray | that's your PID? |
| 06:23.12 | radovoip | I'm running centOS 4.4 I normally like to dload my extera rpms from dag wieers but |
| 06:23.24 | radovoip | just ain't right |
| 06:23.26 | DEader | hrm |
| 06:23.29 | DEader | ok |
| 06:24.29 | drray | show version |
| 06:24.48 | drray | oops |
| 06:39.54 | X-Rob | DEader, if 'svn update' in /usr/src/asterisk says 'updated to ANUMBERHERE' then you're running the latest. |
| 06:53.32 | radovoip | X-Rob, are you familiar w/CentOS4.4 and asterisk? |
| 06:53.48 | X-Rob | radovoip, uh. a bit. |
| 06:54.00 | radovoip | lol |
| 06:54.03 | radovoip | thx |
| 06:54.24 | radovoip | I'm just trying to stark k |
| 06:55.00 | radovoip | I got my kernel right and I have the zaptel rpm k |
| 06:55.30 | radovoip | but...grrr I can't seem to find the right asterisk bin rpm |
| 06:55.38 | X-Rob | Don't bother with a RPM |
| 06:55.40 | X-Rob | compile from source. |
| 06:55.51 | X-Rob | http://www.aussievoip.com/wiki/freepbx-CentOS |
| 06:55.55 | radovoip | just go the srpms? |
| 06:55.57 | X-Rob | No |
| 06:56.02 | X-Rob | follow the instructions there |
| 06:56.02 | radovoip | oooooooo |
| 06:56.06 | justdave | last time I looked nobody's done any decent SRPMS for asterisk since really old versions |
| 06:56.17 | radovoip | oh man thx for this |
| 06:56.37 | justdave | Asterisk@Home (Preinstalled CentOS4.x + Asterisk) even ships source-compiled Asterisk |
| 06:56.38 | radovoip | I was goin nuts and I'm glad I just didn't jump into it |
| 06:56.43 | X-Rob | That also includes installing freepbx, which you may or may not want 8) |
| 06:56.55 | X-Rob | justdave, it's called trixbox now, and it's a pile of shit. |
| 06:57.01 | X-Rob | (the binary asterisk) |
| 06:57.04 | radovoip | I don't want @home |
| 06:57.45 | benjk | X-Rob, somebody in here was recently getting very angry with me insisting that "FreePBX" is a fork of Asterisk and should therefore be avoided |
| 06:58.03 | X-Rob | benjk, did they call you a troll? 8) |
| 06:58.03 | Juggie | openpbx is a fork |
| 06:58.04 | drray | there are any number of people that could be |
| 06:58.10 | justdave | heh |
| 06:58.12 | Juggie | not freepbx |
| 06:58.19 | benjk | I held that "whatever it is, whether we liked it or not, it was certainly not a fork" |
| 06:58.30 | jart | does anyone know how to code a c app to recover from SIGFPE? |
| 06:58.34 | benjk | X-Rob, they always call me a troll |
| 06:58.48 | X-Rob | May I must be good. Rewriting chan_sip in php. |
| 06:58.50 | X-Rob | I r0xx0r. |
| 06:59.03 | X-Rob | s/May/man/ |
| 06:59.07 | radovoip | X-Rob, appreciate ya man, thx |
| 06:59.09 | benjk | hehe |
| 06:59.14 | justdave | AAH is tribox now, or that's something else? |
| 06:59.21 | seebs | Eh, I am just discovering that I suck; I botched my attempt to figure out what a very elaborate bit of FreeBSD code was doing. |
| 06:59.23 | jart | i have this blob that is bringing my phone system down because of x/0 |
| 06:59.25 | benjk | I think it was Blitzrage but I may be wrong |
| 06:59.28 | X-Rob | radovoip, those prerequs are for freepbx. If you wanna use freepbx, feel free to hang out on #freepbx |
| 06:59.36 | X-Rob | (it's the webby config stuff for asterisk) |
| 07:00.12 | radovoip | X-Rob, look I want to go both ways w/this...will the freepbx do it? |
| 07:00.13 | benjk | No, its a fork |
| 07:00.30 | benjk | some resident Digium fanboy said so |
| 07:00.35 | radovoip | X-Rob, I want to be my own provider in other words |
| 07:00.44 | justdave | you can edit stuff manually to some degree with freepbx |
| 07:00.45 | X-Rob | radovoip, freepbx is just a front end for asterisk. All your asterisk stuff is there. It just helps you out by generating the config files |
| 07:01.04 | X-Rob | you can totally ignore it (eg, don't do the 'download and install freepbx' bit) |
| 07:01.15 | X-Rob | but if you 'wanna be your own provider' you've got a long way to go |
| 07:01.41 | radovoip | X-Rob, I know..it's a heavy challenge |
| 07:01.44 | benjk | Of course if Digium are launching their own web front end, then I can see why they would like to spread fud and say FreePBX is a fork |
| 07:01.52 | X-Rob | justdave, I'm slightly biased IRT freepbx, as I'm the main dev for it. You can edit whatever the hell you want with it, as long as you understand how asterisk works. |
| 07:02.10 | X-Rob | benjk, *shrug* doesn't bother me, honestly. |
| 07:02.14 | benjk | That makes me wonder ... can you fork something before the "original" has been released |
| 07:02.28 | drray | that's what a fork is |
| 07:02.42 | X-Rob | I'll just happily start porting all my stuff for openpbx and help you guys out more of the time 8) |
| 07:02.44 | radovoip | X-Rob, kk gonna go get after it now |
| 07:02.44 | benjk | BEFORE the original? |
| 07:02.55 | drray | it's from a previous version |
| 07:02.59 | drray | never to meet again |
| 07:03.06 | X-Rob | Anyway. I should get back to h4xx0r1ng. |
| 07:03.14 | benjk | er, there is no previous version of a Digium web front end |
| 07:03.19 | X-Rob | There's patches that have been sitting here for 2 damn years 8-( |
| 07:04.16 | justdave | We have an asterisk server at work, and I use one at home for my home phones as well. |
| 07:04.21 | X-Rob | benjk, oddly enough, about 6 months ago I had a big conf with all the guys at digium, with me going 'dude, why are your tech support guys so dead against freepbx? It's not hard, in fact it's easy.. here's how it all goes together' |
| 07:04.40 | X-Rob | 2 months later there's a httpd in asterisk and another 4 months they're announcing their own front end |
| 07:04.41 | Juggie | freepbx is a good idea |
| 07:04.45 | Juggie | implemented porely. |
| 07:04.48 | justdave | The one at work has some webmin thing on it for the config, that basically takes control of everything, and if you don't edit from webmin what you do gets overwritten next time someone does. |
| 07:04.53 | X-Rob | Juggie, please. Submit patches. |
| 07:05.01 | Juggie | X-Rob, i'd rather not. |
| 07:05.03 | justdave | My home system has freepbx on it |
| 07:05.04 | radovoip | X-Rob, oh boy...right off the bat they talking they wanted an auto partitioning but my drives are raid1...you forsee any probs? |
| 07:05.19 | drray | freepbx is a great idea, but it's like trying to fix a car with a swiss army knife, it's just not the right tool |
| 07:05.21 | X-Rob | radovoip, I have no idea what you're talking about. Trixbox possibly? |
| 07:05.34 | X-Rob | drray, why do you say that? |
| 07:05.44 | *** join/#asterisk HaMYaI (i=HaMYaI@61.47.126.29) |
| 07:05.48 | benjk | X-Rob, even without their own front end, they wouldn't want stuff to be easy because that is regarded as detrimental to selling support and training and all that |
| 07:05.49 | radovoip | no that installion walk thru you turned me on to |
| 07:05.53 | radovoip | freepbx |
| 07:05.56 | X-Rob | People keep putting shit on freepbx, and I say 'Why' and all they can do is make up similies. |
| 07:05.58 | Juggie | i like a web control pannel for things that exist in realtime, sip phones, iax trunks, etc... |
| 07:06.18 | Juggie | but for real settings... eg sip general, iax general, conf files is best. |
| 07:06.20 | HaMYaI | anyone usinf oh323? |
| 07:06.31 | X-Rob | radovoip, that's a walk through from starting from a bare system. please, read it all |
| 07:06.47 | radovoip | kk |
| 07:07.01 | justdave | Freepbx doesn't seem to have as many options as the webmin thing, but it's certainly easy to find things, and it usually doesn't screw with stuff you've edited manually (your manual edits show up in freepbx) |
| 07:07.03 | HaMYaI | it doesn't seem to accept calls in the dialplan |
| 07:07.11 | X-Rob | justdave, no they don't. |
| 07:07.21 | X-Rob | freepbx is write-only 8) |
| 07:07.42 | justdave | hmm, I've edited stuff and not had it break |
| 07:07.54 | X-Rob | justdave, hang on. Doesn't have as many options as webmin? |
| 07:07.55 | X-Rob | *BAHAHAHAHAHA* |
| 07:08.06 | justdave | I should finish that thought... |
| 07:08.13 | HaMYaI | my oh323 client keeps ringing even the SIP client has answered the call |
| 07:08.15 | justdave | download a few plugins and you get the stuff :) |
| 07:08.16 | Juggie | it still sucks. |
| 07:08.29 | X-Rob | Yeah I know. we've got about 30 plugins now for freepbx |
| 07:08.38 | X-Rob | Juggie, well, excuse me for not giving a flying fuck. |
| 07:08.55 | justdave | oh, it's the includes with the _additional stuff |
| 07:08.56 | X-Rob | The last 'real' suggestions I've implemented. |
| 07:09.06 | benjk | Juggie, that makes it a perfect fit for Asterisk then I suppose |
| 07:09.23 | justdave | each config is split, one chunk is managed by freepbx, the other chunk you can do what you want with |
| 07:09.24 | X-Rob | justdave, yeah. If you fuck with the _Additonal stuff it gets overwritten. The .conf and _custom.conf files are there to be messed with |
| 07:09.31 | Juggie | unless the web control pannel implements realtime i'm not intreasted in the least. |
| 07:09.50 | X-Rob | Juggie, astonishingly, I still don't give a fuck. |
| 07:10.07 | X-Rob | However, as I said earlier, if you were to start writing patches the I certainly would. |
| 07:10.09 | justdave | just click the little red bar a lot :) |
| 07:10.20 | justdave | if you want monitoring, there's the FOP or something |
| 07:10.29 | X-Rob | justdave, he's referring to asterisk's realtime implementation, it reads all the config from a database on the fly |
| 07:10.31 | HaMYaI | X-Rob, does the Freepbx include configs for h323 clients? |
| 07:10.34 | X-Rob | No |
| 07:10.37 | Juggie | X-Rob, i have more important things to address. |
| 07:10.42 | justdave | ah. |
| 07:11.03 | X-Rob | Juggie, so do I. So realtime support is _way_ down the list. |
| 07:11.08 | X-Rob | After, in fact, openpbx support. |
| 07:11.22 | Juggie | great, well why not run it with asterisk 1.0.10 for a laugh then too... |
| 07:11.37 | X-Rob | AMP runs with 1.0, freepbx with 1.2 |
| 07:11.55 | Juggie | what 1.2 features does it support |
| 07:11.56 | X-Rob | freepbx 3.0 will do all the 1.4 goodness, as well as openpbx support. |
| 07:12.41 | X-Rob | Juggie, you must have missed me saying I don't give a flying fuck about what you think. Go look at the source. www.freepbx.org/trac/browser |
| 07:13.20 | Juggie | your attitude is fantastic. |
| 07:13.51 | *** join/#asterisk websae (n=websae@adsl-64-149-206-121.dsl.milwwi.sbcglobal.net) |
| 07:13.56 | benjk | methinks his attitude changed this way after you called his work bad names |
| 07:14.04 | X-Rob | Heh |
| 07:14.23 | X-Rob | <X-Rob> Juggie, please. Submit patches. |
| 07:14.24 | X-Rob | <Juggie> X-Rob, i'd rather not. |
| 07:14.28 | X-Rob | Actually, that was it |
| 07:14.40 | Juggie | well, thats because, i dont use it. |
| 07:14.53 | Juggie | so, yah, i dont think i will be submitting patches for somerthing i dont use. |
| 07:15.00 | Juggie | *something |
| 07:15.15 | benjk | depends |
| 07:15.36 | benjk | I have been submitting contributions to opbx before using it |
| 07:16.17 | benjk | nice thing about some projects is that they actually take your contribution |
| 07:16.23 | benjk | imagine that |
| 07:16.50 | seebs | I submit tons of patches for things I won't use. |
| 07:17.10 | benjk | and though I haven't contributed to FreePBX, from my experience talking to X-Rob, I have the impression that his project is one of those projects which actually take contributions |
| 07:17.25 | seebs | *sigh* So, it turns out that the two different things in the FreeBSD zaptel port I didn't understand and thought were dumb were TOGETHER doing something. |
| 07:17.57 | *** join/#asterisk linagee (n=na@cpe-66-75-142-207.san.res.rr.com) |
| 07:18.27 | linagee | what is a good hardware phone that speaks SIP+IAX2, does POE, and has two ethernet ports, and is cheap? |
| 07:18.45 | Juggie | hah |
| 07:18.50 | Juggie | better write santa for that one. |
| 07:18.53 | *** part/#asterisk jart (n=user@ool-4357221d.dyn.optonline.net) |
| 07:18.54 | linagee | Juggie: lol |
| 07:19.13 | benjk | with asterisk its kinda odd, they email you for advice on Japanese language support only to give you a hard time when you actually do provide the advice and they end up ignoring ALL OF IT |
| 07:19.22 | linagee | Juggie: i am thinking maybe budget tone cheap ones if i keep them inside the network. (i'd really only need IAX if it's a remote phone for NAT traversal) |
| 07:19.37 | linagee | Juggie: hrm. "SIP works fine" :-D |
| 07:19.39 | Juggie | linagee, you dont need iax for nat transversal. |
| 07:19.57 | Juggie | and the best phones are definitally sip phones. |
| 07:19.57 | linagee | Juggie: you just need a sip proxy. which isn't something you'd find in any random person's cable modem router. :P |
| 07:20.04 | Juggie | linagee, also not true. |
| 07:20.12 | linagee | Juggie: what part |
| 07:20.18 | Juggie | that you need a sip proxy. |
| 07:20.30 | linagee | Juggie: how then |
| 07:20.43 | linagee | hm |
| 07:20.46 | Juggie | is your * server in a firewall or not? |
| 07:20.48 | seebs | Woot! Kernel panic. |
| 07:20.51 | Juggie | (eg dmz) |
| 07:20.52 | linagee | Juggie: sure |
| 07:21.00 | Juggie | does it have a public or private ip? |
| 07:21.08 | linagee | Juggie: i finally got * working. got a voip provider that speaks IAX. ;) |
| 07:21.22 | linagee | Juggie: private. i said firewall=yes. |
| 07:21.23 | benjk | At some point all I could do was to suggest to write to the Japanese Ministry of Cultural affairs to ask them if they would kindly change the Japanese language so it would fit the Asterisk project |
| 07:21.46 | linagee | Juggie: using pfsense. hehehe. more opensourceware. ;) |
| 07:22.06 | benjk | because "Asterisk doesn't work that way" is what they kept saying |
| 07:22.12 | linagee | Juggie: these, "pop in a CD dists" make it easy to recover. ;) |
| 07:22.27 | Juggie | linagee, ok, so, in sip.conf you do this, 1) set extenip= your external ip, set nat=yes for your clients, and make sure localnet and externnet are set, then, look @ rtp.conf take note of the port range. |
| 07:22.50 | benjk | seebs, another one :) |
| 07:22.52 | linagee | Juggie: set it to my external IP? it's dynamic though. :P |
| 07:23.02 | linagee | Juggie: dyndns. |
| 07:23.05 | Juggie | then set it to a externaldns |
| 07:23.16 | Juggie | its in sip.conf with a refresh time i might add. |
| 07:23.17 | linagee | hrm |
| 07:23.24 | Juggie | so you can force it to recheck the dns every so often |
| 07:23.32 | linagee | nice. i don't think that was in there last time i was handcrafting files |
| 07:23.45 | Juggie | make sure your firewall passes RTP on the same ports you opened in rtp.conf |
| 07:24.18 | seebs | Yeah, this one came somewhere in zt_timer_poll. |
| 07:24.25 | Juggie | so that is all, open firewall for sip & rtp, externip=whatever nat=yes and localnet/externnet set. |
| 07:24.34 | Juggie | so, what will happen is this. |
| 07:24.47 | justdave | oh, you can stick a domain name in there instead of an IP? |
| 07:24.54 | Juggie | theres another setting for a domain |
| 07:25.02 | linagee | justdave: yeah. first time i heard of it too. |
| 07:25.08 | linagee | justdave: dyndns rules. :-D |
| 07:25.25 | Juggie | oh if you look @ sip.conf.sample |
| 07:25.27 | Juggie | you will find it |
| 07:25.43 | linagee | justdave: i can pick up my servers and run anywhere and drop it into the network, and within minutes it's back there. :-D |
| 07:25.47 | justdave | heh. I went and wrote myself a cron job to check the IP and write it out to sip_nat_ip.conf periodically, and do a sip reload if it changed |
| 07:25.56 | linagee | justdave: ew |
| 07:25.59 | justdave | (and included that file from the main conf) |
| 07:26.05 | linagee | heh |
| 07:26.21 | Juggie | ;externhost=foo.dyndns.net |
| 07:26.29 | benjk | hope that sip reload won't crash your box |
| 07:26.33 | Juggie | ;externrefresh=10 ; How often to refresh externhost |
| 07:26.40 | *** join/#asterisk doolph (n=doolph@200.75.247.87) |
| 07:26.44 | doolph | hi there |
| 07:26.46 | linagee | benjk: crash the whole phone network. :-D |
| 07:26.51 | doolph | omg another realease? |
| 07:26.51 | justdave | heh, it hasn't yet |
| 07:26.53 | Juggie | anyways, linagee, thats all you need for sip behind nat... |
| 07:27.05 | linagee | Juggie: and then any remote users can connect? |
| 07:27.05 | benjk | it'll do a global lock on all data used by chan_sip |
| 07:27.06 | Juggie | no more, no less, no stun, no sip aware proxys, nothing. |
| 07:27.09 | Juggie | yes. |
| 07:27.16 | justdave | I have a dyndns domain already, so I can ssh in to get stuff |
| 07:27.18 | Juggie | they can connect due to the power of nat, yay. |
| 07:27.27 | benjk | asterisk does extremely "finegrained" locking |
| 07:27.38 | Juggie | when they register your port forward on your firewall will pass in the sip packet. |
| 07:27.39 | benjk | Rocky Mountain locking |
| 07:27.50 | Juggie | then when a phone call starts... this is what will happen |
| 07:27.50 | linagee | benjk: yeah. don't want to fsck up the users just because you want to add some silly extension |
| 07:27.59 | Juggie | your client will start sending audio and so will * |
| 07:28.06 | Juggie | a few packets from * wont get through |
| 07:28.17 | linagee | benjk: i wonder if freepbx takes advantage of fine grained locking, or just always does a reload... |
| 07:28.19 | Juggie | (not that its noticiable or matters) |
| 07:28.46 | Juggie | but the client will send packets to rtp port you are forwarding into your * server |
| 07:28.51 | benjk | linagee, how can it, if it is a fork, rewritten in PHP :P |
| 07:29.12 | Juggie | * will now respond back to the source port of these packets w/ another source port which is the symetric rtp port it wants the client to respond to. |
| 07:29.23 | Juggie | (nat stuff) |
| 07:29.32 | Juggie | some clients will change to that port, others will ignore it |
| 07:29.37 | Juggie | either way the call will function fine. |
| 07:30.09 | doolph | is the 1.2.12 and older causing my call transfer problems, it sometimes transfer a busy tone to the caller |
| 07:30.10 | linagee | i know you guys are going to hate me for mentioning trixbox, but it would be cool if it automatically controlled your pfsense firewall too. :-D |
| 07:30.28 | linagee | wizard it all into a single abstraction of clicking a button. :-D |
| 07:30.53 | benjk | single button operation |
| 07:30.59 | Juggie | pfsense? |
| 07:30.59 | benjk | [recover from crash] |
| 07:31.02 | linagee | benjk: for a home asterisk box, why not? |
| 07:31.06 | linagee | Juggie: m0n0wall fork |
| 07:31.21 | linagee | benjk: maybe it doesn't work when you have a bajillion users |
| 07:31.24 | benjk | yeah, the button would be called [recover from crash] |
| 07:31.48 | Juggie | ah, well none the less, no packet inspection is required to make sip w/ double nat work. |
| 07:32.02 | Juggie | all the fancy sip enabled firewalls dont accomplish anything that nat cant. |
| 07:32.07 | linagee | benjk: i just don't get why everyone is a trixbox hater. job security? |
| 07:32.25 | linagee | benjk: i mean even in a small business, say you have ten people, what's wrong with it? |
| 07:32.32 | benjk | especially since you probably load everyting into such an asterisk box, and the more modules, the more stuff it does the less stable it is, YMMV |
| 07:32.56 | linagee | benjk: that's why you hire an admin to fix it. see, that's where the job security comes in. ;-) |
| 07:33.09 | linagee | benjk: doing less work |
| 07:33.15 | benjk | I don;t know about trixbox, but I do know asterisk, and that isn't exactly the incarnation of stability |
| 07:33.21 | linagee | benjk: working smarter, not harder, all that |
| 07:33.34 | Juggie | benjk, then fix it or something, or move to openpbx |
| 07:33.49 | Juggie | stop bitching endlessly when no one is paying attention. |
| 07:33.52 | benjk | I am fixing it a lot |
| 07:34.03 | linagee | benjk: if you're directing 100+ calls like a voip provider might do, you probably want to have a stripped down asterisk on the fastest hardware you can budget. |
| 07:34.09 | linagee | benjk: stable versions too. |
| 07:34.15 | linagee | benjk: and constant monitoring jobs |
| 07:34.29 | Juggie | * isnt perfect, but not much software is. |
| 07:34.41 | benjk | linagee, like I said, in my experience, the more stuff you load into it the less stable it becomes |
| 07:34.44 | Juggie | i can crash a mitel voip switch as easily as i can crash *. |
| 07:34.47 | benjk | so I strip off whatever I can |
| 07:34.47 | doolph | asterisk is good |
| 07:34.50 | linagee | Juggie: that's easy. you just go up and kick it, it starts working again. :-D |
| 07:35.03 | linagee | benjk: i think that's true of just about any system. ;) |
| 07:35.36 | Juggie | as i said, i can crash a supposedly industry tested hardware machine as easy as i can break * |
| 07:35.37 | linagee | benjk: when i loaded trixbox, sugarcrm was fscked up. couldn't change the default password! lol. but i did an update and that little bug went away. |
| 07:35.38 | Juggie | so meh :) |
| 07:35.54 | linagee | benjk: (before it just presented a 404 when you clicked change password) |
| 07:36.05 | benjk | sure, but folks keep getting angry at me when I don't follow the general mass urge to upgrade every 5 mins because of some new geek feature |
| 07:36.11 | linagee | lol |
| 07:36.22 | linagee | benjk: see, that's why you have 10 people using it |
| 07:36.33 | linagee | benjk: and it's easy to upgrade and fix features if you know how to do it smart. |
| 07:36.58 | linagee | benjk: either have two boxes or run xen. have one a stripped down asterisk that just takes calls in and directs them all to one extension. |
| 07:37.20 | linagee | benjk: so you can flip between the two virtual machines when you're upgrading and not miss any calls. ;) |
| 07:37.27 | benjk | upgrading means you never get it replaced by something decent because you spend time fixing other people's stuff |
| 07:37.41 | linagee | benjk: a small amount of time |
| 07:37.45 | benjk | rewirting pieces properly is a better option |
| 07:37.49 | linagee | benjk: unless you want to pay tens of thousands. ;) |
| 07:38.30 | linagee | which might be better if the calls were more valuable, it just depends. heh |
| 07:38.31 | Juggie | upgrading depends on your scale |
| 07:38.47 | Juggie | i can turn off a machine and nothing is affected |
| 07:39.02 | linagee | Juggie: exactly. it all depends on scale. ;) |
| 07:39.13 | Juggie | multiple t1's into multiple * boxes |
| 07:39.19 | Juggie | did's all shared on a super group across those t1's |
| 07:39.50 | linagee | Juggie: maybe they should just put that on the screen when you install trixbox. then people wouldn't be freaking out about it. "Warning: This system is only designed for approximately ten users. Please use something more stable ($$$) if you plan on supporting more users." |
| 07:40.08 | linagee | Juggie: interesting... |
| 07:40.11 | Juggie | you can pull a box for maintenance and the phone company will automatically route calls to a t1 thats up. |
| 07:40.13 | benjk | trouble is I can't use a stock asterisk because the kind of support required for the geography I am in is categorically being refused by digium |
| 07:40.33 | linagee | Juggie: so in that case, it's really the phone company providing the fail over. heh |
| 07:40.36 | benjk | so if I upgrade I have to patch everything all over again and fix what might have broken in the process |
| 07:40.38 | Juggie | yep... |
| 07:40.59 | linagee | Juggie: i bet they charge up the ying yang for that. to spread a group of DIDs across multiple PRIs. |
| 07:41.03 | Juggie | linagee, its nice :) |
| 07:41.11 | Juggie | not really |
| 07:41.17 | Juggie | its just NFAS w/ more then one d channel |
| 07:41.39 | linagee | Juggie: how do the SIP devices handle that? do you just have DHCP tell devices what's going on, or ? |
| 07:41.41 | Juggie | i dont know the exact # but its nothing huge. |
| 07:41.42 | benjk | if you were in the same situation you would do the exact same thing I do |
| 07:42.12 | Juggie | linagee, i am lucky enough to not run sip phones :) so i dont have to worry about that. |
| 07:42.24 | Juggie | other then a small number i run for like myself at home etc. |
| 07:42.44 | linagee | Juggie: well that all depends. if it's small (residence), rebooting at night would be no biggy. if it's a small business, do the VM thing to fail over to one phone line. if it's large, yes, group DIDs across multiple PRIs makes sense. very near zero downtime. |
| 07:43.04 | linagee | Juggie: huh? LOL. PRI in, asterisk as the PBX, analog out? weird.... |
| 07:44.27 | *** join/#asterisk lowlevel (n=Stuart@CPE0050ba71c82f-CM000f9f7d6742.cpe.net.cable.rogers.com) |
| 07:44.30 | linagee | or are they digital phones like an old school PBX? |
| 07:44.44 | *** join/#asterisk Juggie (n=agony@CPE00c049d9f271-CM00137186c8d8.cpe.net.cable.rogers.com) |
| 07:44.54 | Juggie | awesome internet died for a sec |
| 07:44.58 | Juggie | yah, t1s in FastAGI out |
| 07:45.10 | linagee | not sure what FastAGI is... |
| 07:45.19 | Juggie | do you know what AGI is? |
| 07:45.28 | linagee | asterisk something interface? |
| 07:45.32 | Juggie | Gateway |
| 07:45.39 | linagee | yes |
| 07:45.41 | linagee | for programming |
| 07:45.46 | JT | Juggie: so it's all IVRs? |
| 07:46.05 | linagee | wait no. AGI isn't the programming part. that's another module, right? |
| 07:46.28 | JT | agi allows interfacing with scripting languages to control a call |
| 07:46.35 | linagee | aha. i was right. :-D |
| 07:46.36 | benjk | agi => external dialplan-script socket interface |
| 07:46.41 | JT | but is not an outside world telephony interface |
| 07:46.58 | linagee | JT: which scripting? python? |
| 07:47.11 | JT | linagee: ANYTHING, pretty much |
| 07:47.17 | linagee | JT: PHP? lol |
| 07:47.18 | JT | communicates via STDIN and STDOUT |
| 07:47.19 | benjk | any scripting language that can read from a socket |
| 07:47.20 | JT | yes |
| 07:47.24 | linagee | JT: hah |
| 07:47.56 | linagee | JT: so i can do database crap in PHP, output to AGI? if i was so crippled as to only know PHP... ;) |
| 07:48.11 | JT | i guess you could |
| 07:48.16 | linagee | lol |
| 07:48.40 | linagee | i had PHP parsing my GPS for a while there. lol. i was trying to record GPS stuff into a database. |
| 07:48.44 | benjk | fater than using the manager interface |
| 07:48.48 | benjk | faster |
| 07:49.01 | linagee | benjk: manager interface? i saw that, but ? |
| 07:49.19 | benjk | tcpip socket interface for remote control and monitoring |
| 07:49.19 | linagee | "FastAGI can be used to do the processing on a remote machine" |
| 07:49.31 | linagee | benjk: aha |
| 07:49.39 | *** join/#asterisk Juggie (n=agony@CPE00c049d9f271-CM00137186c8d8.cpe.net.cable.rogers.com) |
| 07:49.40 | benjk | but its a serious bottleneck |
| 07:49.48 | Juggie | back again |
| 07:49.53 | Juggie | isp lost a route there for a minute |
| 07:49.59 | linagee | wtf? deadagi? gives access to dead channels after hangup? why would you want that? |
| 07:50.01 | benjk | it goes all through a single thread |
| 07:50.11 | linagee | Juggie: is that the same ISP that's feeding your voip? lol |
| 07:50.21 | linagee | Juggie: oh wait, you said PRIs |
| 07:50.23 | benjk | in order to record information into a database |
| 07:50.28 | benjk | for example the hangup cause |
| 07:50.29 | linagee | benjk: hrm.. |
| 07:50.37 | Juggie | linagee, no. there is no voip. |
| 07:50.42 | benjk | or last called number and dureation |
| 07:50.47 | linagee | benjk: is there a hangup reason: user was pissed? lol |
| 07:50.48 | justdave | does freepbx 2.1.2 work with trunk asterisk right now? |
| 07:51.01 | doolph | justdave, wrong channel? |
| 07:51.03 | benjk | linagee there are about 100 hangup causes |
| 07:51.10 | linagee | yikes |
| 07:51.14 | Juggie | benjk, whats wrong w/ dead agi? |
| 07:51.14 | X-Rob | justdave, that's a #freepbx queston, and the answer is probably no. |
| 07:51.33 | benjk | http://trac.openpbx.org/cgi-bin/trac.cgi/browser/openpbx/trunk/include/openpbx/causes.h |
| 07:51.35 | doolph | hey rob lol |
| 07:51.51 | linagee | trunk asterisk? didn't freepbx work with trunks in 2.1.1 also? |
| 07:52.06 | benjk | its not that kinda trunk |
| 07:52.11 | benjk | its the trunk of |
| 07:52.14 | benjk | and elephant |
| 07:52.16 | justdave | ok, won't upgrade asterisk yet then :) |
| 07:52.20 | benjk | an elephant |
| 07:52.21 | linagee | benjk: wtf? lol |
| 07:52.26 | X-Rob | justdave, branches/1.2 is the best |
| 07:52.29 | benjk | white elephant |
| 07:52.39 | linagee | benjk: white elephant module? |
| 07:52.47 | justdave | oh, of asterisk? |
| 07:52.53 | benjk | yeah, the whole thing is a white elephant module |
| 07:53.01 | *** join/#asterisk nassy (n=nassy@207-38-197-201.c3-0.wsd-ubr1.qens-wsd.ny.cable.rcn.com) |
| 07:53.04 | benjk | especially sip stuff |
| 07:53.10 | justdave | my current install is a trunk checkout, but it's from a while ago. hopefully before they branched for 1.4 :) |
| 07:53.37 | justdave | identifies as SVN-trunk-r-1M |
| 07:53.45 | benjk | trunk checkout, an elephant in a supermarket checking out |
| 07:53.49 | linagee | benjk: make a child's game with AGI. :-D press a number, it will use a letter at random on that number. "you pressed, 6. M. The cow says Mooo!" |
| 07:54.03 | benjk | heh |
| 07:54.16 | benjk | you can do that without AGI though |
| 07:54.23 | linagee | benjk: i was just about to say that |
| 07:54.24 | benjk | within the dialplan |
| 07:54.30 | linagee | benjk: but then i thought about the random bit. heh |
| 07:54.45 | benjk | you can use the random function |
| 07:54.57 | linagee | benjk: ah. don't have that one in freepbx. ;) |
| 07:55.07 | benjk | or in earler versions you just strip off the seconds digit of TIMESTAMP |
| 07:55.11 | linagee | benjk: if i add things like that in extensions.conf, freepbx will clobber it, right? heh |
| 07:55.27 | benjk | not necessarily |
| 07:55.50 | benjk | X-Rob explained earlier that there was a part of the file you can touch and another one you dont |
| 07:55.56 | linagee | benjk: so the only reason to use AGI is for database stuff then? |
| 07:56.01 | benjk | or split into two files |
| 07:56.09 | benjk | not really |
| 07:56.12 | linagee | benjk: oic. so it's marked. |
| 07:56.23 | Juggie | there are lots of reasons to use agi |
| 07:56.27 | benjk | the dialplan language is very limited |
| 07:56.34 | benjk | you can use it for simple stuff |
| 07:56.39 | X-Rob | benjk/linagee - .conf files get overwritten with ./install_amp - _addtional.conf files get rewritten when you reconfigure freepbx. _custom.conf files never get overwritten |
| 07:56.45 | linagee | Juggie: what things can dial plans not handle?? |
| 07:56.53 | benjk | but when your tasks become a little more involved, then it shows its limitations |
| 07:57.02 | benjk | thats where AGI becomes handy |
| 07:57.05 | linagee | X-Rob: oic. thanks. :) |
| 07:57.07 | Juggie | hmmmmm.... accessing data on our mainframe for one. |
| 07:57.15 | Juggie | acessing microsoft sql |
| 07:57.22 | linagee | Juggie: didn't i say? pulling things out of a database? heh |
| 07:57.25 | *** join/#asterisk LoneShadow (n=duh@59.92.154.100) |
| 07:57.25 | justdave | sometimes it's easier to write a script in perl or python than it is to write a complex dialplan for something |
| 07:57.34 | justdave | grab a webpage and read the contents to you |
| 07:57.52 | Juggie | linagee, the bonus of writing FastAGI is that you can write an async server |
| 07:57.59 | linagee | justdave: you could just have a dialplan that says every single different word? LOL |
| 07:58.08 | justdave | I think it ships with a sample AGI script that grabs a weather report for New York :) |
| 07:58.11 | Juggie | so you can handle all your calls in one thread + worker threads |
| 07:58.22 | Juggie | rather then calls*threads |
| 07:58.26 | Juggie | as you would if you ran just local AGI |
| 07:58.37 | linagee | calls*threads? hrm |
| 07:58.47 | linagee | Juggie: so say you have 5 calls. 5 threads, right? |
| 07:58.56 | Juggie | er, thats wrong. what i just said |
| 07:59.04 | Juggie | but yes, if you had 5 calls running AGI locally |
| 07:59.09 | Juggie | you'd have 5 processes running |
| 07:59.15 | Juggie | whatever was runnning your agi script |
| 07:59.35 | linagee | Juggie: so again though, this doesn't matter on a 10 user box. ;) |
| 07:59.38 | benjk | though you get the overhead of fork() |
| 07:59.39 | Juggie | where as, via fastagi, all 5 of those calls are handled in one process. |
| 07:59.51 | linagee | Juggie: hrm. weird. |
| 08:00.00 | Juggie | well, because fastagi is a seperate server. |
| 08:00.13 | Juggie | its just like a web server, or a ssh server. |
| 08:00.26 | linagee | ah. so it might have children processes |
| 08:00.32 | Juggie | you could write it multithreaded, but our implementation is async. |
| 08:00.41 | Juggie | at least as far as handeling all the connections |
| 08:00.57 | benjk | it still doesn't allow you escape from * locking though |
| 08:01.07 | Juggie | explain. |
| 08:01.41 | benjk | well, if you have a sip call come in and you handle the other hand via FastAGI IVR, it still uses all the chan_sip stuff |
| 08:01.43 | linagee | benjk: locking up? :-D oh, wrong type. :-D |
| 08:02.03 | benjk | and its locks |
| 08:02.28 | Juggie | well of course its going to use whatever channel the call comes in on |
| 08:02.30 | benjk | and like I said earlier, the locks are often very coarse grain |
| 08:02.32 | Juggie | i'm not sure what you mean. |
| 08:03.05 | linagee | asterisk is going to sit there and wait until your process is done? :-D |
| 08:03.10 | benjk | the bottleneck is usually the locks when you get high loads |
| 08:03.37 | benjk | so moving some work off to another process doesn't help you much when you hit the ceiling |
| 08:03.49 | Juggie | linagee, theres no process for * to wait for w/ fastagi |
| 08:03.57 | Juggie | because its a seperate box. |
| 08:04.19 | linagee | Juggie: so instead you introduce network latency? ;) |
| 08:04.46 | benjk | on a local network that should be minimal, but the locks are what kill you |
| 08:05.14 | Juggie | linagee, theres no network latency |
| 08:05.15 | linagee | benjk: a lock might last longer than a network lag delay? |
| 08:05.22 | Juggie | nothing above 2ms anyways |
| 08:05.27 | linagee | Juggie: even if it's just a few milliseconds. ;) |
| 08:05.42 | benjk | a lock might last seconds |
| 08:05.51 | linagee | Juggie: 2ms for tiny little packets. what happens when you start pushing data through like mad? :-D |
| 08:05.54 | Juggie | linagee, only the control path of the call is transfered over agi, not the audio. |
| 08:05.56 | linagee | benjk: yikes. |
| 08:05.58 | benjk | especially when you do or watch stuff on the console |
| 08:06.06 | linagee | benjk: what does the caller hear over seconds? lol. *silence* |
| 08:06.11 | benjk | nothing |
| 08:06.19 | linagee | "Hello??" |
| 08:06.39 | benjk | everything stops while the console is priniting some stuff |
| 08:06.53 | benjk | when its done priniting stuff the calls will continue |
| 08:06.55 | Juggie | benjk, console output is another story. |
| 08:07.04 | Juggie | i agree w/ you there and i've heard this before. |
| 08:07.20 | Juggie | not sure if it was from you or not |
| 08:07.37 | benjk | cosole output is only showing the problem on a larger scale because it does the most coarse locking of all activities |
| 08:07.44 | linagee | Juggie: the best thing you can do for interactive latency is just to try to standardize it. so at least it's not a random wait. ;) |
| 08:08.09 | linagee | Juggie: better to make all users wait 2 seconds for a DB fetch than to have some 0.1 seconds, some 5 seconds. :-D |
| 08:08.16 | benjk | but the problem is there still without the console one noticeable scale once you have a certain call volume |
| 08:09.00 | seebs | Wow. |
| 08:09.02 | Juggie | benjk, theres no doubt that the output system should be changed, but i lack knowledge to an extent, but mostly time. |
| 08:09.04 | seebs | I almost have it working! Almost. |
| 08:09.09 | linagee | benjk: that's silly that everything would wait for console output. lol. |
| 08:09.25 | benjk | seebs, good work! |
| 08:09.26 | seebs | I can get a dialtone, but if I try to dial an actual number, I get a fast busy. |
| 08:09.37 | seebs | Does asterisk actually generate dialtones and busies, or is that really my outside line? |
| 08:09.41 | linagee | benjk: why not just have a thread with a buffer that handles system output. push stuff to it when you want output to the screen, then continue doing what you were doing. |
| 08:09.57 | benjk | its not just the console output, its the way how locks are done, they are really overdoing those |
| 08:10.18 | Juggie | benjk, is it just locks or is there blocking code as well |
| 08:10.27 | seebs | Hmm. |
| 08:10.31 | Juggie | or potentially blocking code. |
| 08:10.36 | linagee | seebs: it could be either. ;) |
| 08:10.46 | benjk | linagee the problem lies deeper, it starts with the way internal data is organised |
| 08:10.51 | seebs | Well, looking at cdr-csv, incoming calls are showing up correctly with caller-ID, but the in house phone doesn't ring. |
| 08:11.03 | benjk | everything is a linked list |
| 08:11.03 | linagee | seebs: if you were using freepbx and call features, i could recommend to try dialing 7777. ;) |
| 08:11.21 | seebs | Heh. |
| 08:11.31 | linagee | benjk: what's wrong with linked lists? |
| 08:11.32 | Juggie | benjk, well digium is still looking for developers :) |
| 08:11.36 | benjk | so in certain circumstances, in order to get to one element you have to traverse the list and they lock the whole list then |
| 08:12.07 | linagee | benjk: why not have a multidirectional linked list then? forward/backward at least |
| 08:12.07 | benjk | if other threads are trying to write to other elements in that list, they have to wait |
| 08:12.14 | seebs | [ Context 'telco2' created by 'pbx_config' ] |
| 08:12.14 | seebs | <PROTECTED> |
| 08:12.14 | seebs | <PROTECTED> |
| 08:12.14 | seebs | <PROTECTED> |
| 08:12.14 | seebs | <PROTECTED> |
| 08:12.33 | Juggie | benjk, theres a whole event system for * that never made it into 1.4 |
| 08:12.35 | benjk | linagee, the best way would be not to have a linked list at all, and use hashtables |
| 08:12.44 | benjk | which is what we are doing in openpbx |
| 08:12.48 | seebs | If I have understood this correctly, this should take incoming calls on the channel "telco2", record them to a file, and then connect them to Channel 2. |
| 08:12.53 | Juggie | i'm not sure how much that affects what you are talking about |
| 08:12.57 | linagee | benjk: hash tables are fine when you're dealing with a fsckload |
| 08:12.59 | seebs | Am I even close? :) |
| 08:13.10 | seebs | Hmm. |
| 08:13.14 | benjk | also, pthreads aren't exactly the best choice for asynchronous realtime servers such as telephny |
| 08:13.32 | seebs | What are we arguing about linked lists vs. hashtables FOR, anyway? Databases? Names? Birds? |
| 08:13.42 | linagee | benjk: why not just make a data structure that changes from linked lists to hash tables when the data structure gets too big? hehehe. hide it all from the application programmer. |
| 08:14.02 | benjk | linagee, that's how many hashtables work |
| 08:14.07 | benjk | the other way round though |
| 08:14.11 | linagee | hm |
| 08:14.19 | benjk | they are tables like arrays |
| 08:14.21 | seebs | If you need to know whether you're using lists or hashes, you've probably violated an abstraction layer. |
| 08:14.25 | linagee | benjk: n/m then. it appears it's already been done. ;) |
| 08:14.32 | linagee | benjk: an operator overloaded array. heh |
| 08:14.36 | benjk | and when they get full they link to a chain of smaller linked lists |
| 08:14.42 | benjk | its called external chaining |
| 08:15.17 | benjk | the hashtable library I wrote for openpbx is actually a two dimensional hashtable with a thrid dimension being linked lists |
| 08:15.24 | linagee | benjk: and then you have to know, "what's too big for this systemk |
| 08:15.44 | linagee | benjk: heh. it would have to time how long the linked list takes, or use a static table along with the cpu Mhz |
| 08:15.57 | benjk | the chance that you overflow both dimensions is very minimal, so there will not be many linked lists spawned and in cases where they are, they are very short, only a handful of elements |
| 08:16.32 | benjk | seebs, abstraction layer? in Asterisk? |
| 08:16.35 | benjk | ROTFL |
| 08:16.38 | seebs | Well, uhm. |
| 08:16.41 | seebs | I speak in the general case. |
| 08:16.48 | benjk | fair enough |
| 08:16.52 | seebs | So, uhm. Could I hit you up for more of your wonderful advice on debugging this? It's ALMOST working. |
| 08:17.00 | linagee | benjk: interesting.... so if you overflow the full 32-bits or whatever of the hash table address space, it uses linked lists off of those?? |
| 08:17.21 | benjk | linagee, in a nutshell yes |
| 08:17.40 | linagee | benjk: ew. wouldn't you constantly have to be sorting to know if it's full or something? |
| 08:17.44 | seebs | It's not uncommon to have a hash, even a smalish one, of linked lists; that way, you can compare only against 1/N of the items per trip. |
| 08:18.15 | seebs | In that case, you don't care whether it's REALLY full, you just assume that the hash will work well enough to reduce load in non-pathological cases. |
| 08:18.15 | linagee | benjk: and wouldn't sorts take forever if you have linked lists off of the hashes themselves? :( |
| 08:18.18 | benjk | lineagee, if you are interested how this works, you can look at this ... http://trac.openpbx.org/cgi-bin/trac.cgi/browser/openpbx/branches/benjk/opbx_dictionary.c |
| 08:18.35 | benjk | scroll down a bit, there are diagrams (ASCII art) of the datastructures |
| 08:18.52 | linagee | benjk: why even bother with hash tables and linked lists. why not just use a database and trust they already have the most efficient algorithms? :-D |
| 08:19.16 | linagee | benjk: are you talking about data structures like call logs? |
| 08:19.18 | benjk | because you probably get more overhead |
| 08:20.05 | benjk | no, applications (like app_dial, the Dial() command), variables, like ${CALLERID}, contexts, extensions, channels etc etc etc |
| 08:21.32 | linagee | is asterisk programmed in C? |
| 08:21.41 | benjk | yes |
| 08:21.48 | linagee | yikes |
| 08:21.50 | seebs | Very loosely speaking. :P |
| 08:21.56 | Juggie | i think i need to read up on hashtables & linked lists |
| 08:21.56 | benjk | well actually its more like Commodore VIC-20 BASIC |
| 08:22.02 | linagee | benjk: LOL |
| 08:22.03 | benjk | with C syntax |
| 08:22.07 | linagee | BASIC? |
| 08:22.17 | benjk | yes, lotsa loadsa gotos |
| 08:22.19 | linagee | 10 PRINT "HELLO ASTERISK" : 20 GOTO 10 |
| 08:22.23 | linagee | benjk: ha ha ha |
| 08:22.25 | benjk | 40 page functions |
| 08:22.32 | linagee | benjk: functions? lol |
| 08:22.42 | linagee | benjk: commodore BASIC didn't have functions. :P |
| 08:22.52 | benjk | *40 page* functions !!! |
| 08:23.04 | linagee | you just set a lot of variables and GOTO somewhere. :-D |
| 08:23.32 | Juggie | benjk, hashed tables & linked list is something c++ would hide from you right normally? eg, not necessairly required w/ c++ |
| 08:24.19 | seebs | Not normally, I wouldn't think. |
| 08:24.54 | *** join/#asterisk danalien (n=danalien@unaffiliated/danalien) |
| 08:25.16 | linagee | oh yeah. lol |
| 08:25.16 | linagee | when i was talking about operator[] overloading. heh. i guess that wouldn't apply to C |
| 08:25.16 | benjk | Juggie, I am not versed in C++, but I think there are hashtable libs in the standard C++ libraries |
| 08:25.17 | linagee | benjk: yes |
| 08:25.18 | seebs | Is there a good way to emit messages to console or /var/log/asterisk/messages from within a dialplan? |
| 08:25.26 | linagee | seebs: not normally what? |
| 08:25.29 | Juggie | a hashtable is just like a keyed array? |
| 08:25.29 | linagee | benjk: heh |
| 08:25.39 | Juggie | seebs, verbose or noop |
| 08:25.42 | seebs | Normally, C++ would not hide hashes or linked lists unless you told it to. |
| 08:25.50 | seebs | I can't find syntax for Verbose, how's it go? |
| 08:25.51 | benjk | Objective-C as in 90% Smalltalk 10% C |
| 08:26.03 | linagee | Juggie: see a basic CS class for info on data structures for hash table info. ;) |
| 08:26.16 | linagee | BASIC! :) |
| 08:26.25 | benjk | Juggie, read that link I posted |
| 08:26.28 | benjk | http://trac.openpbx.org/cgi-bin/trac.cgi/browser/openpbx/branches/benjk/opbx_dictionary.c |
| 08:26.30 | linagee | benjk: objective-c and smalltalk are ugly |
| 08:26.38 | benjk | unlike Asterisk code, its documented |
| 08:26.53 | benjk | you can actually figure out what it does by reading the text in there |
| 08:26.57 | linagee | benjk: C is self documenting! :-D |
| 08:27.04 | benjk | lol |
| 08:27.06 | seebs | Objective-C may be sorta ugly, but it's a very very elegant kind of ugly. |
| 08:27.08 | linagee | benjk: imagine if it was written in ASM. ;) |
| 08:27.19 | seebs | So, uhm. This Verbose() thing. It's not in the voip-info wiki, where should I find it? |
| 08:27.26 | benjk | linagee, wouldn't make much of a difference anymore |
| 08:27.33 | linagee | benjk: sure it would |
| 08:27.48 | linagee | benjk: it would make it a nightmare to edit. nobody would ever fix any bugs. lol |
| 08:27.53 | benjk | it would break more often, but it would be just as incomprehensible |
| 08:28.03 | linagee | benjk: that's a difference. :-D |
| 08:28.03 | benjk | ROTFL |
| 08:28.25 | linagee | benjk: backwards interpolate that to the land of C++ ;) |
| 08:28.44 | linagee | benjk: or refuse to like most others. :)O |
| 08:28.47 | benjk | like I said, C++ is not my terrain |
| 08:29.09 | linagee | benjk: C++ is awesome for self-documentability. |
| 08:29.29 | linagee | as long as you don't use names like thisOneFunctionIThoughtUp |
| 08:29.43 | benjk | I do however like the dynamic nature of ObjC and I know C++ is static, compile time bound |
| 08:30.35 | seebs | Hmmmm. |
| 08:30.49 | linagee | benjk: if you make a new instantion of something that's not static.. |
| 08:31.09 | linagee | something, that's |
| 08:31.21 | benjk | res = [origChannel brigdeTo: destChannel withTimeout: 60]; |
| 08:31.28 | benjk | that's what I like ;) |
| 08:31.32 | linagee | benjk: ew |
| 08:32.01 | Juggie | if wikipedia is telling me the truth, a hashtable is a way of doing a Associative Array/keyed array |
| 08:32.11 | Juggie | c++ does this w/ std::map |
| 08:32.19 | linagee | res = new Channel(destChannel, 60); // Channel(bridgeTo, withTimeout) |
| 08:32.21 | benjk | linagee, I am talking about dynamic late binding |
| 08:32.48 | benjk | like everything is a runtime object, nothing is a compiletime only object |
| 08:33.00 | linagee | benjk: sounds like java. heh |
| 08:33.03 | benjk | introspection |
| 08:33.20 | benjk | Java was modeled to some extend from ObjC |
| 08:33.22 | seebs | Well, caller ID may not be working right. |
| 08:33.28 | benjk | however, even Java isn't late bound |
| 08:33.37 | seebs | Apart from that, it looks as though I am successfully running my * system. |
| 08:33.54 | seebs | Next up, remove all the stupid debugging printfs from the driver. |
| 08:33.58 | benjk | seebs, do you get any output now from "zap show channel" |
| 08:34.04 | seebs | Yup. |
| 08:34.07 | seebs | And I can call in and out. |
| 08:34.10 | benjk | what does it say |
| 08:34.15 | seebs | Don't remember, but it's working. |
| 08:34.27 | linagee | seebs: you should have made a function like debug("Stuff"); instead, then you can turn it all on and off. ;) |
| 08:34.32 | benjk | heh |
| 08:34.40 | seebs | I don't really understand the language. |
| 08:34.43 | linagee | seebs: any programmer knows that. ;) |
| 08:35.00 | benjk | which language? |
| 08:35.06 | benjk | the one Asterisk is written in? |
| 08:35.14 | linagee | benjk: English? :) |
| 08:35.16 | benjk | Gibberish |
| 08:35.20 | linagee | lol |
| 08:35.24 | benjk | Cibberish |
| 08:35.28 | seebs | The language used for writing stuff in extensions. |
| 08:35.41 | linagee | "That one language"? |
| 08:35.41 | benjk | Dialplan Gibberish |
| 08:35.44 | seebs | Yeah. |
| 08:35.48 | benjk | heh |
| 08:36.01 | seebs | I have no clue how it works. I'm trying to set something so I get convenient logs. The Caller ID info isn't coming in sometimes. |
| 08:36.02 | linagee | benjk: is that an official term? add it to the wiki. :-D |
| 08:36.11 | benjk | so what does the console say, when you get an incoming call? |
| 08:36.19 | benjk | linagee, lol |
| 08:36.28 | seebs | I'm rebooting right now. |
| 08:36.32 | linagee | Dialplan Gibberish, see Dialplan Configuration |
| 08:36.58 | linagee | sometimes. hrm |
| 08:37.15 | benjk | linagee, btw, not all of asterisk is in English, there is some Spanish in there |
| 08:37.21 | linagee | benjk: wtf? |
| 08:37.27 | benjk | and the header file reads "Written by Mark Spencer" |
| 08:37.30 | benjk | go figure |
| 08:37.33 | seebs | Sep 16 03:37:04 DEBUG[1872]: chan_zap.c:6625 do_monitor: Monitor doohicky |
| 08:37.33 | seebs | got event Ring/Answered on channel 3 |
| 08:37.33 | seebs | Sep 16 03:37:04 DEBUG[1872]: app_queue.c:490 changethread: Device 'Zap/3' change |
| 08:37.33 | seebs | d to state '2' (In use) but we don't care because they're not a member of any qu |
| 08:37.34 | seebs | eue. |
| 08:37.34 | seebs | Sep 16 03:37:06 NOTICE[1872]: callerid.c:324 callerid_feed: Caller*ID failed checksum |
| 08:37.36 | seebs | Sep 16 03:37:10 NOTICE[1872]: chan_zap.c:6058 ss_thread: Got event 2 (Ring/Answered)... |
| 08:37.36 | linagee | who's that |
| 08:37.38 | seebs | Sep 16 03:37:10 DEBUG[1872]: pbx.c:1677 pbx_extension_helper: Launching 'NoOp' |
| 08:37.40 | seebs | Sep 16 03:37:10 DEBUG[1872]: pbx.c:1677 pbx_extension_helper: Launching 'SetVar' |
| 08:37.42 | seebs | Sep 16 03:37:10 DEBUG[1872]: pbx.c:1677 pbx_extension_helper: Launching 'MixMonitor' |
| 08:37.44 | seebs | Sep 16 03:37:10 DEBUG[1872]: channel.c:1014 ast_channel_spy_add: Spy MixMonitor added to channel Zap/3-1 |
| 08:37.46 | seebs | Sep 16 03:37:10 DEBUG[1872]: pbx.c:1677 pbx_extension_helper: Launching 'Dial' |
| 08:37.48 | seebs | Sep 16 03:37:10 DEBUG[1872]: chan_zap.c:7558 zt_request: Using channel 1 |
| 08:37.56 | benjk | people tell me the real author's name is Jesus Arias though, but of course I am a troll |
| 08:38.38 | linagee | <PROTECTED> |
| 08:38.50 | seebs | Well, anyway, it MOSTLY works. |
| 08:38.50 | benjk | seebs, I dont mind you flooding the channel, but jjust beware that the Digium Gestapo will take you away if you do that during your daytime ;) |
| 08:39.02 | seebs | It was a pretty small flood. |
| 08:39.06 | seebs | Okay, so. |
| 08:39.09 | benjk | yeah, no problem |
| 08:39.10 | linagee | benjk: everyone lives in PST? :-D |
| 08:39.18 | seebs | Next up, I want to figure out what's wrong with caller ID, and whether it's always wrong or just sometimes. |
| 08:39.39 | benjk | that's a tricky one to debug |
| 08:39.59 | linagee | hrm... on the east coast you can sleep in, because if you get up early, nobody on the west coast would even consider waking up. :-D |
| 08:40.02 | benjk | when did you buy that Digium card? |
| 08:40.31 | seebs | A couple of weeks back. |
| 08:40.43 | benjk | then you will be eligible to support |
| 08:40.45 | linagee | benjk: ah. is there a way to spy on people using asterisk? |
| 08:40.46 | seebs | Got the driver port mostly done, but it was crashing, so I sent out feelers to see if anyone wanted to debug it for me. |
| 08:40.56 | benjk | trouble is that you are running BSD and Digiuym won't lke htat |
| 08:41.04 | linagee | benjk: say you're the manager of a call center and you want to be able to listen in |
| 08:41.12 | seebs | Eh, I don't care. I got caller ID, so it may just be that the one time it failed was due to some other quirk. |
| 08:41.12 | benjk | so if you had another box lying around, set up Linux and Asterisk to test this |
| 08:41.25 | benjk | if you have the same issue then, you can get them to look at the caller ID stuff |
| 08:41.34 | linagee | yes. "well there's your problem!" hehehe. (BSD) |
| 08:41.46 | seebs | No, it's working now. |
| 08:41.52 | linagee | benjk: any way to spy in on call center workers? |
| 08:41.56 | seebs | Oh, interesting. |
| 08:42.20 | benjk | linagee yes you can, its called ChanSpy |
| 08:42.26 | linagee | hehehe |
| 08:42.27 | linagee | lol |
| 08:42.41 | benjk | seriously |
| 08:43.18 | linagee | "Usefull in a call center to monitor agents on the phone. " <--- or to listen in on your sister and run off giggling. :-D |
| 08:43.20 | benjk | seebs, it could be the card, or the module, or the driver, or even your POTS line |
| 08:43.30 | seebs | I get 0-byte files if I don't actually pick up an incoming call. |
| 08:43.34 | seebs | Which I think makes sense. |
| 08:44.02 | seebs | Anyway, I got them all but that one time, and frankly, the statement "Qwest cannot provide caller ID reliably even on non-blocked calls" is an article of faith to me. |
| 08:44.23 | benjk | yeah well, analog sucks anyway |
| 08:44.27 | seebs | If you told me that Qwest-provided copper was only conductive six days a week on the average, I would believe you. |
| 08:44.36 | benjk | hehe |
| 08:44.52 | seebs | We once had over ONE HUNDRED separate outages in a three month period, peaking at five lines out simultaneously. |
| 08:44.54 | Strom_C | seebs: that refers more to the fact that Qwest cannot guarantee that the originating end of the call will send a calling party number |
| 08:44.58 | benjk | coppice told me an interesting thing |
| 08:45.01 | seebs | No, no. |
| 08:45.11 | seebs | I'm using my cell phone, which sends caller ID very consistently. |
| 08:45.21 | seebs | If I call home on it ten times, I get caller ID about 8. |
| 08:45.29 | benjk | about dry lines being prone to erosion because they don't carry any voltage all the time |
| 08:45.39 | benjk | corosion |
| 08:45.42 | linagee | benjk: is that what those huge expansion module switch boards are for? so the operator can hit a button and spy on any extension? ChanSpy would be perfect! |
| 08:45.43 | Strom_C | where are you located, what kind of switch are you served out of, and how long is your loop? |
| 08:45.48 | linagee | Strom_C! |
| 08:45.48 | seebs | Strom, I don't know. |
| 08:45.55 | seebs | I mean, I know where I am. I'm in St. Paul. |
| 08:46.04 | seebs | Wherever I am, they can give me 7Mbps DSL that works pretty well. |
| 08:46.14 | seebs | But basic phone service? Not so much. |
| 08:46.16 | Strom_C | what kind of hardware are you using? |
| 08:46.19 | Strom_C | hey linagee |
| 08:46.21 | benjk | that sounds like you are close to the exchange |
| 08:46.26 | seebs | I've got a TDM400P. |
| 08:46.29 | Strom_C | ok |
| 08:46.38 | Strom_C | have you screwed with your rxgain settings at all? |
| 08:46.38 | seebs | Two channels in, two out, connected to a two-line cordless phone with multiple handsets. |
| 08:46.47 | seebs | Haven't touched anything. |
| 08:46.52 | benjk | Strom-C all default |
| 08:46.55 | Strom_C | hmmm |
| 08:47.01 | benjk | very minimalistic zaptel config |
| 08:47.15 | benjk | caller id checksum sometimes fails |
| 08:47.20 | linagee | seebs: you are using one of those filters for your DSL to not interfere, right...?? |
| 08:47.27 | Strom_C | does caller ID work reliably if you connect the pots line directly to a telephone set that has a caller id display? |
| 08:47.42 | seebs | No. |
| 08:47.48 | linagee | No?? |
| 08:47.54 | seebs | No to Strom. Yes to lin. |
| 08:47.59 | linagee | ah |
| 08:48.01 | seebs | I have a filter on the line that has the DSL. |
| 08:48.10 | seebs | And yeah, the caller ID has ALWAYS been dodgy here. |
| 08:48.15 | seebs | On both lines. |
| 08:48.17 | benjk | ok, seems your line doesn't deliver good caller id service then |
| 08:48.20 | seebs | Yup. |
| 08:48.28 | Strom_C | yeah, time to blame the line card at the telco |
| 08:48.37 | benjk | or the copper |
| 08:48.44 | Strom_C | if it were the copper, the DSL would be shit too |
| 08:48.50 | linagee | seebs: complain to your telco. if they don't fix it, then get a new telco (j/k. you're stuck. ;) ) |
| 08:48.53 | seebs | They replaced the local trunk lines recently. |
| 08:49.05 | seebs | I mean, a couple years back, now, but "recently" compared to the usual 25-year age. |
| 08:49.13 | benjk | depends, the copper may be ok on higher frequencies and bad on lower ones |
| 08:49.31 | Strom_C | and i somehow refuse to believe that 7 megabit DSL would work fine while 1200 baud FSK data wouldn't :) |
| 08:49.47 | linagee | Strom_C: if long distance providers like to sneak runs of voip to make their costs cheaper, what happens to overall call latency? hrm |
| 08:49.54 | benjk | depends on what kind of filters they have in between |
| 08:49.56 | Strom_C | linagee: it depends |
| 08:50.06 | linagee | Strom_C: like those 1010321 numbers |
| 08:50.21 | Strom_C | seebs: do you hear noise or attenuation on normal voice calls? |
| 08:50.22 | linagee | Strom_C: i'm sure they're using voip 99 times out of 100. ;) |
| 08:50.33 | Strom_C | linagee: well, feature group D technically has nothing to do with it |
| 08:50.45 | linagee | feature group D? |
| 08:50.49 | Strom_C | 101-XXXX codes |
| 08:50.50 | linagee | 1010 numbers |
| 08:51.15 | Strom_C | all that does is tell the switch to route your call through someone else's class four tandem |
| 08:51.23 | linagee | Strom_C: i don't mean the numbers themselves having anything to do with it. i mean, how else could they provide service at such low prices without doing voip or something |
| 08:51.32 | linagee | Strom_C: hrm |
| 08:52.03 | Strom_C | linagee: if they've already got excess capacity in their fiber, it doesn't cost them that much to provision more OC3s ;) |
| 08:52.04 | linagee | Strom_C: and i guess that routing table is shared among all the switches so you can be on any switch and have the 1010 numbers work. or else there's some sort of DNS-ish thing. |
| 08:52.25 | benjk | linagee, the average cost to complete a call for AT&T in 1997 was less than 1 cent |
| 08:52.26 | Strom_C | linagee: it's similar to DNS; there's an SS7 lookup that happens |
| 08:52.32 | linagee | hm |
| 08:52.35 | benjk | not per minute, per call |
| 08:52.48 | linagee | Strom_C: similar to DNS.. so there's a central database for that sort of thing. heh |
| 08:52.52 | benjk | including all long distance and any overseas calls |
| 08:53.04 | benjk | and that was in 1997 |
| 08:53.10 | benjk | almost 10 years ago |
| 08:53.20 | linagee | Strom_C: hrm. extra capacity... |
| 08:53.57 | Strom_C | dont forget that the major north american IXCs built out tons of capacity in the late 90s in anticipation that the dot-com era would keep roaring and there would be a gigantic spike in demand |
| 08:54.01 | benjk | before VoIP came along and prices began to move, telephone service was a license to print money |
| 08:54.14 | benjk | virtually no cost, and high revenue |
| 08:54.26 | Strom_C | "virtually no cost" my ass |
| 08:54.28 | linagee | Strom_C: so if i had an OC3, it's cheaper to have them run "PRIs" through that than to expand the bandwidth? hrm |
| 08:54.34 | Strom_C | do you have any idea how much a DMS-100 costs? :) |
| 08:54.42 | benjk | rolled out per total number of calls, virtually no cost |
| 08:54.43 | seebs | -rw-r--r-- 1 root wheel 72138 Sep 16 03:51 20060916-035043-IN-6608-"SEEBACH,PETER " <XXXYYYZZZZ>.gsm |
| 08:54.55 | Strom_C | linagee: I have no idea what you mean by that |
| 08:54.59 | seebs | Good Enough. |
| 08:55.01 | linagee | benjk: no cost to terminate, still costs to dial |
| 08:55.37 | linagee | Strom_C: DMS-100? oh, i have one of those in my garage. (j/k. hehehe) |
| 08:55.43 | seebs | Now, uhm. One last stupid question before I stop bugging you people: |
| 08:55.49 | seebs | What's a ".gsm" file and how do I read it? |
| 08:55.49 | linagee | Strom_C: uhm... $5 million. ;) |
| 08:56.03 | Strom_C | GSM-encoded audio |
| 08:56.17 | linagee | Strom_C: they make virtual PRIs over OC3? |
| 08:56.23 | seebs | So, what programs are good bets for interpreting GSM audio? |
| 08:56.28 | Strom_C | really, you should be recording your audio in wav or slin format unless you specifically need otherwise |
| 08:56.30 | Strom_C | seebs: asterisk |
| 08:56.49 | Strom_C | linagee: what do you mean "virtual"> |
| 08:56.50 | linagee | seebs: quicktime might work. ;) |
| 08:56.51 | benjk | seebs, sox will convert it to whatever yu want |
| 08:56.59 | linagee | Strom_C: well it's not copper coming out of the wall. ;) |
| 08:57.01 | seebs | Okay, I could do a different format. What bit-width and frequency is Asterisk going to record a WAV at? |
| 08:57.13 | Strom_C | linagee: a PRI does not necessarily have to be on T1 |
| 08:57.18 | linagee | hm |
| 08:57.27 | Strom_C | a PRI can be on DS3 and have maybe two or three D-channels |
| 08:57.29 | linagee | Strom_C: so it's not a "virtual PRI". it IS a PRI... |
| 08:57.42 | linagee | Strom_C: aha. what's his face was saying something about that then |
| 08:57.44 | linagee | i see... |
| 08:57.48 | Strom_C | also, PRI is typically only between CPE and the telco |
| 08:57.59 | Strom_C | the telco uses SS7 for interoffice signaling |
| 08:58.14 | linagee | SS7 is used for billing as well? |
| 08:58.19 | Strom_C | ISDN User Part is only a subset of SS7 |
| 08:58.21 | Strom_C | what? no |
| 08:58.27 | Strom_C | AMA records are used for billing |
| 08:58.31 | linagee | hm |
| 08:58.34 | Strom_C | SS7 is only for call setup and teardown |
| 08:58.38 | linagee | i see |
| 08:58.57 | linagee | Strom_C: so SS7 is what tells asterisk the DID for that channel... |
| 08:59.05 | Strom_C | what? nonono |
| 08:59.07 | *** join/#asterisk lukketto (n=lukketto@host78-154-dynamic.11-87-r.retail.telecomitalia.it) |
| 08:59.09 | linagee | hehehe |
| 08:59.12 | linagee | ;) |
| 08:59.13 | Strom_C | SS7 is an INTEROFFICE signaling system |
| 08:59.18 | Strom_C | used in the telco network |
| 08:59.21 | linagee | ah |
| 08:59.36 | Strom_C | ISDN PRI is what the telco will provision out to your prem |
| 08:59.51 | Strom_C | the DID information comes to your CPE in the ISDN SETUP message |
| 09:00.09 | Strom_C | on the telco side, it comes into the telco switch in the SS7 ISUP IAM |
| 09:00.15 | linagee | hm |
| 09:00.39 | linagee | i see... AMA and SS7 records correlate. |
| 09:00.45 | Strom_C | no no no |
| 09:00.50 | linagee | well by phone number. ;) |
| 09:00.52 | Strom_C | there are no such things as "ss7 records" |
| 09:01.01 | Strom_C | AMA records are generated as calls are set up and torn down |
| 09:01.06 | Strom_C | SS7 is used for signaling |
| 09:01.14 | linagee | don't kill the google messenger. :) |
| 09:01.18 | Strom_C | AMA records are /derived/ from SS7 information |
| 09:01.21 | benjk | I think there is a band called SS7 |
| 09:01.28 | benjk | they may well have made some records |
| 09:01.29 | Strom_C | linagee: you should read a document on my website |
| 09:01.30 | Strom_C | one moment... |
| 09:01.47 | Strom_C | http://www.stromcarlson.com/docs/basics/NTtelephony101.pdf |
| 09:01.53 | Strom_C | http://www.stromcarlson.com/docs/basics/pt_ss7_tutorial.pdf |
| 09:01.54 | *** join/#asterisk abatista (n=Ariel@dsl-20-177.cofs.net) |
| 09:01.55 | linagee | Strom_C: nice. this one looks good. http://www.ss7.com/down1.cfm#search=%22ss7%20ama%20how%20things%20work%22 |
| 09:02.00 | Strom_C | read those two and call me in the morning :) |
| 09:02.02 | linagee | Strom_C: SS7 pocket guide. :-D |
| 09:02.41 | linagee | "In most cases each |
| 09:02.41 | linagee | digit you dial places two tones on the line. |
| 09:02.41 | linagee | We call this “touch tone” dialing." |
| 09:02.47 | linagee | you don't say! (j/k) |
| 09:02.48 | linagee | lol |
| 09:02.52 | benjk | linage, you use WOOMERA to get your Asterisk box hooked up to SS7 |
| 09:03.01 | linagee | woomera? |
| 09:03.04 | benjk | that's all you need to know |
| 09:03.07 | benjk | yes, WOOMERA |
| 09:03.10 | linagee | woo? |
| 09:03.12 | seebs | I'm no finding much detail in the Asterisk docs on the sound formats available for recording. |
| 09:03.15 | benjk | WOOMERA |
| 09:03.29 | Strom_C | seebs: it's very simple |
| 09:03.29 | benjk | chan_woomera |
| 09:03.50 | Strom_C | seebs: telephone audio is 8khz 16-bit audio companded down to 8-bit samples |
| 09:03.57 | seebs | I sort of recall that. |
| 09:04.13 | seebs | So, if I have asterisk save WAV, I get what, 8KHz mono 16-bit data? |
| 09:04.17 | Strom_C | yes |
| 09:04.29 | linagee | unless you have two phones. then you can have stereo! |
| 09:04.36 | linagee | :-> |
| 09:05.40 | benjk | linagee, http://www.voxgratia.org/docs/woomera_desc.html |
| 09:05.42 | linagee | Strom_C: why does voicepulse send out a busy tone when your asterisk server is not logged in? |
| 09:05.52 | Strom_C | busy? or reorder? |
| 09:05.57 | linagee | Strom_C: is that the default phone switch thing to do, or is that something on their end? |
| 09:06.03 | linagee | Strom_C: can i tell by listening? :-D |
| 09:06.13 | Strom_C | reorder is twice as fast as a busy signal |
| 09:06.19 | linagee | Strom_C: cell phone says "Number is busy" |
| 09:06.26 | linagee | Strom_C: not fast busy |
| 09:06.32 | Strom_C | that's really not the thing they're supposed to be doing |
| 09:06.38 | linagee | hehehe |
| 09:06.51 | benjk | and also http://www.ss7box.com/asterisk.html |
| 09:06.53 | Strom_C | if the endpoint can't be reached, they're supposed to send a circuit-busy, congestion, or failure message |
| 09:07.00 | linagee | Strom_C: i guess it would make people all weireded out if they started throwing fast busy tones around |
| 09:07.09 | benjk | using WOOMERA to hook * to an SS7 SMG |
| 09:07.12 | Strom_C | why? that's what they're supposed to do |
| 09:07.17 | linagee | Strom_C: *shrug* |
| 09:07.22 | linagee | Strom_C: call them. ;-) |
| 09:07.44 | linagee | Strom_C: i'd like them to fail over to my cell instead of just going all busy if asterisk is not available. |
| 09:07.49 | linagee | oh well |
| 09:07.53 | Strom_C | or perhaps send inband progress audio: "Due to temporary equipment conditions, your call did not go through. Please try your call again later. This is a recording." |
| 09:08.07 | seebs | Well, if it's just 8KHz mono, I think I'll just use wav as my default format, and convert to something else Later. |
| 09:08.19 | benjk | Clearly a US message prompt |
| 09:08.28 | linagee | Strom_C: i'm allowed to use 4 channels. :-D |
| 09:08.31 | Strom_C | benjk: and clearly a message from thirty years ago :) |
| 09:08.33 | benjk | as evidenced by "This is a recording" |
| 09:08.54 | Strom_C | benjk: hey, I like "This is a recording." |
| 09:08.56 | linagee | Strom_C: how about "This is not a recording" :-D |
| 09:08.58 | linagee | lol |
| 09:09.15 | benjk | wonder why they don't have "Dont use your phone while taking a bath" |
| 09:09.16 | linagee | this recording will explode if not stored properly |
| 09:09.35 | linagee | benjk: using your phone is just fine as long as it doesn't ring. ;) |
| 09:10.01 | benjk | and don't try your wet cat in the microwave oven either |
| 09:10.04 | benjk | dry |
| 09:10.21 | Strom_C | benjk: well, thirty years ago, you might reach an intercept operator instead of an intercept recording |
| 09:10.26 | linagee | benjk: ew |
| 09:10.34 | linagee | Strom_C: wtf? |
| 09:10.35 | Strom_C | so it makes sense to clearly differentiate that what you are listening to is not a live human being |
| 09:10.37 | linagee | Strom_C: a person?? |
| 09:10.43 | benjk | linagee, there was a court case over that |
| 09:11.11 | linagee | Strom_C: i've heard that once you could call up numbers on the phone and get actual people. but they were all replaced with machines. |
| 09:11.13 | Strom_C | linagee: yes...they would typically ask what number you were trying to reach, and then let you know whether it was disconnected or changed or whether you just misdialed. |
| 09:11.18 | benjk | they fried they cat in the microwave oven after the cat had fallen into the bath tub and they though it might be a good idea to dry it in there |
| 09:11.28 | benjk | then they sued the manufacturer |
| 09:11.40 | linagee | benjk: that's nasty |
| 09:11.53 | linagee | benjk: do you realize how freaked out and how many screaming noises the cat would make? |
| 09:11.56 | benjk | and they won, because the users manual did not say that you cannot dry your wet cat in the microwave oven |
| 09:12.03 | Strom_C | linagee: listen to www.phonetrips.com if you don't believe me :) |
| 09:12.03 | *** join/#asterisk speedwagon (n=Ariel@dsl-20-177.cofs.net) |
| 09:12.09 | drray | not more than once |
| 09:12.19 | linagee | Strom_C: don't tell me there's a screaming cat in a microwave? :-/ |
| 09:12.24 | Strom_C | no no |
| 09:12.26 | linagee | good |
| 09:12.30 | *** join/#asterisk Gunnar (n=gunnar@101.Red-80-24-171.staticIP.rima-tde.net) |
| 09:12.36 | Strom_C | there are recordings of intercept operators :) |
| 09:12.39 | linagee | lol |
| 09:12.56 | linagee | Strom_C: were they actually paid money to do that? lol |
| 09:12.56 | linagee | Strom_C: just walk into work with a recording. lol |
| 09:13.01 | Strom_C | linagee: yes, they were AT&T employees |
| 09:13.05 | linagee | ah! |
| 09:13.06 | seebs | WOOT! My system works without manual intervention from a cold boot. |
| 09:13.09 | linagee | corporate. heh |
| 09:13.15 | linagee | i bet they even got full benefits. ;) |
| 09:13.18 | benjk | lineagee, if the microwave was on 2000W or so, it only takes a very short time for the cat's brain to fry and then the cat will scream no more |
| 09:13.39 | linagee | benjk: most microwaves don't do anything for like the first three seconds. |
| 09:13.44 | Strom_C | linagee: well, also, you have to realize that such things were very necessary with the switching technology of the time |
| 09:13.55 | linagee | benjk: the microwave emmitter is just starting up |
| 09:14.02 | benjk | yeah, maybe the cat felt comfortable during those first 3 secs |
| 09:14.06 | linagee | eww |
| 09:14.24 | benjk | like frogs if you boil em |
| 09:14.42 | linagee | benjk: you can test this by putting a cup of water and measuring the temp for the first three seconds. also, use a Kill-a-watt and you'll see this |
| 09:14.48 | benjk | if yo increase the temperature only very gradually, they will stay in the water and eventually get boiled |
| 09:15.10 | linagee | Strom_C: very necessary? |
| 09:15.14 | benjk | Kill-a-watt-cat |
| 09:15.15 | seebs | Apparently, that is an urban legend. |
| 09:15.22 | seebs | Frogs jump out when it gets hot. |
| 09:15.25 | seebs | Hmmm. |
| 09:15.27 | benjk | Watta-kill-cat |
| 09:15.34 | linagee | what happens if a frog is microwaved? does it jump? |
| 09:15.37 | seebs | I have possible flaw; when I do a MonitorMix on one of my calls, I record dialtone. |
| 09:15.38 | Strom_C | linagee: look, go listen to phonetrips.com for a while and you'll understand |
| 09:15.39 | benjk | is it? |
| 09:16.02 | benjk | Watta-kill-frog |
| 09:16.06 | seebs | Yeah. Tragic. |
| 09:16.08 | benjk | (tm) |
| 09:16.15 | seebs | Such a great analogy, but apparently only humans are that stupid. |
| 09:16.34 | benjk | Watta-kill-frog(tm) Microwave Oven |
| 09:16.42 | linagee | Strom_C: lol. the "They'll never catch you" poster |
| 09:16.58 | linagee | Strom_C: "using a blue box is bad, mmkay?" |
| 09:17.41 | benjk | linagee, ask, JDcrunchman about the blue box |
| 09:17.57 | linagee | Strom_C: you know, i think i'll start charging you for every letter you send me. if you refuse to pay, it's fraud!! up to $1000 chargable per offense! (seriously, what gives them the right? a pack of lawyers?) |
| 09:18.19 | Strom_C | benjk: that man is a total nutcase |
| 09:18.23 | benjk | heh |
| 09:18.26 | benjk | tell me about it |
| 09:18.31 | linagee | Strom_C: so they own the wires. if i own something and sue people, it doesn't mean a thing |
| 09:18.40 | Strom_C | benjk: evan doorbell is a much more reliable source :) |
| 09:18.46 | benjk | I went to his house in LA and installed Asterisk on his Mac and an IAXy |
| 09:18.47 | linagee | Strom_C: "you now have to pay $1000 if you step on my lawn. or even if your kids step on my lawn" |
| 09:18.53 | linagee | right. lol |
| 09:19.40 | linagee | what gives the phone company the right to prosecute over their property if i can't equally do the same? lol |
| 09:20.05 | Strom_C | linagee: because it's a federally regulated industry, and wire fraud is a federal crime. |
| 09:20.08 | benjk | but fun to listen to his stories anyway |
| 09:20.10 | Strom_C | that's why. |
| 09:20.13 | linagee | oh, and if i live on a state line, if you walk across my lawn, that's an *out of state crime* |
| 09:20.14 | benjk | like the Nixon prank |
| 09:20.39 | benjk | whenthey called some whitehouse crisis number |
| 09:20.40 | linagee | Strom_C: federally regulated industry. hrm. how do i get that? heh |
| 09:20.46 | linagee | Strom_C: $$$ |
| 09:20.48 | linagee | heh |
| 09:20.59 | Strom_C | benjk: I think 70% of his stories are only 10% true, and the other 30% are only 40% true |
| 09:21.09 | benjk | I don't really care |
| 09:21.16 | benjk | it was a hilarous story |
| 09:21.21 | linagee | Strom_C: wire fraud, what about wireless fraud? heh |
| 09:21.30 | Strom_C | linagee: you're getting really irritating. |
| 09:21.40 | linagee | :( |
| 09:22.33 | benjk | however, it can't be all made up because he went to prison for his bluebox shit, or did he not? |
| 09:23.26 | Strom_C | no, but I have several friends who knew him in those days, and their stories all corroborate with each other far more than they do with his |
| 09:23.45 | linagee | benjk: it would be nice to have some market around some service with such a stranglehold that you could use the law and huge penalties when people don't play by your rules |
| 09:23.45 | Strom_C | he's always had a tendency to massively embellish his stories, apparently |
| 09:23.49 | benjk | but he went to prison for real, right? |
| 09:24.00 | Strom_C | that part is true |
| 09:24.31 | seebs | Hmm. |
| 09:24.42 | Strom_C | linagee: you mean like...being a landlord? |
| 09:24.48 | seebs | So, I have an outgoing line, and I dial on it. If the other end hangs up, I get a busy signal, instead of being disconnected. |
| 09:24.49 | linagee | Strom_C: hrm! :-D |
| 09:24.49 | benjk | anyway, that Nixon prank story is hilarious, true or not, blown up or not |
| 09:25.20 | seebs | Is that normal, or did I misconfigure something? |
| 09:25.38 | benjk | seebs, thats how you get disconnect |
| 09:25.45 | Strom_C | seebs: is it a busy signal,or is it a reorder tone? |
| 09:25.47 | linagee | Strom_C: can a landlord say if you don't pay your rent that you have to pay out $1000 in fines or something? oh wait, another regulated industry. heh |
| 09:25.51 | seebs | I'm not sure. It could be a reorder tone. |
| 09:25.59 | Strom_C | seebs: a reorder tone is twice as fast as a busy signal |
| 09:25.59 | benjk | it play fast busy and only cuts off audio after a timeout |
| 09:26.10 | seebs | Hmm. |
| 09:26.16 | Strom_C | linagee: you're demonstrating a very poor understanding of how the telecommunications industry works. Please stop it. |
| 09:26.28 | seebs | If I wanted the behavior "hang up on the inside line when the outside line hangs up on Asterisk", how would I get that? |
| 09:26.51 | linagee | Strom_C: but it's all based on the cash, right? to even get a foot in the door of having a PRI requires huge cashola, right? |
| 09:26.53 | Strom_C | seebs: ask the telco to provision something called "far-end disconnect supervision" onto your line |
| 09:27.08 | Strom_C | linagee: what the hell are you talking about? |
| 09:27.10 | benjk | seebs, the fast busy comes when you do Hangup, that is what hangup does |
| 09:27.16 | linagee | Strom_C: nothing. sleeping. |
| 09:27.19 | seebs | Hmm. |
| 09:27.37 | seebs | Oh, so I assume that my regular phone, when it's connected to the line, was just magically knowing to hang up when that happened? |
| 09:27.59 | Strom_C | seebs: what do you mena |
| 09:28.03 | Strom_C | s/mena/mean/ |
| 09:28.16 | seebs | Okay. In the absence of *: |
| 09:28.16 | *** join/#asterisk S^P (n=Masood@203.148.73.236) |
| 09:28.30 | seebs | I call someone. We talk. He hangs up. Whether or not I hang up, my phone disconnects as though I had hung up. |
| 09:28.30 | benjk | no the central office will just withdraw the line so your phone may stay silent on that signal |
| 09:28.41 | seebs | Hmm. |
| 09:28.53 | Strom_C | seebs: did you provision your line with "ks" or "ls" signaling in asterisk? |
| 09:28.54 | benjk | but Asterisk gives you feedback for a short time |
| 09:28.56 | seebs | ks. |
| 09:29.12 | seebs | I don't know what the difference is, though; maybe I guessed wrong. |
| 09:29.29 | Strom_C | seebs: on a normal analog telephone, when the other party hangs up, does your talk battery drop for half a second? |
| 09:29.41 | seebs | Hmm. Maybe? |
| 09:29.45 | Strom_C | maybe. |
| 09:29.48 | seebs | I don't know what a "talk battery" is. |
| 09:29.51 | Strom_C | go plug an analog phone into the line |
| 09:29.57 | linagee | Strom_C: Kewl Signaling? :) |
| 09:30.00 | Strom_C | talk battery is the DC voltage on the line |
| 09:30.20 | seebs | Oh! I have no idea; I don't have anything that could measure this, but I bet it does, because my phones nearly always notice when they get hung up on. |
| 09:30.23 | seebs | ... |
| 09:30.29 | Strom_C | linagee: are you stoned or drunk or something? |
| 09:30.30 | seebs | Oh, wait. No they don't. I get that "if you'd like to make a call" thing. |
| 09:30.33 | seebs | I am just stupid tonight. |
| 09:30.49 | seebs | To put this in perspective, eighteen hours ago I said "I am going to debug this stupid driver today." |
| 09:31.00 | Strom_C | seebs: like I said: have the telco provision far-end disconnect supervision on your line |
| 09:31.06 | Strom_C | seebs: and take a break, man |
| 09:31.19 | linagee | Strom_C: naw. but brain is fried |
| 09:31.29 | seebs | Well, if I have the telco do that, will they charge me more? If anything, I'm hoping to REDUCE my phone bill now by dropping call waiting, which I really haven't used in years. |
| 09:31.31 | benjk | without a microwave oven? |
| 09:31.42 | linagee | benjk: cell phone |
| 09:31.43 | seebs | The only people left who call me know my other numbers. |
| 09:31.57 | Strom_C | seebs: there may be a one-time charge for provisioning, but i've never seen a recurring charge for that. |
| 09:32.14 | seebs | What does "far-end disconnect supervision" mean, and what do I do on my end to deal with it? |
| 09:32.19 | benjk | seebs, it will be affordable, only 100 USD per month |
| 09:32.38 | seebs | Does it mean that they signal in some way to my line that the other end hung up, so Asterisk can DTRT? |
| 09:32.46 | Strom_C | seebs: it means that when the far end disconnects, your talk battery will drop for half a second. |
| 09:32.56 | linagee | seebs: call waiting is a fear based service. "what if someone else is calling? i'll never known w/o this service!" as if any call was that important. hah. |
| 09:33.00 | seebs | Heh. |
| 09:33.05 | seebs | When I had only one line, it was worth it. |
| 09:33.10 | benjk | careful with that carpet when you drop the battery |
| 09:33.22 | seebs | Now, anyone who has any business trying to reach me in a hurry has my cell number, my email address, and my IM. |
| 09:33.41 | Strom_C | I despise my cellphone. |
| 09:33.53 | Strom_C | I never pay attention to it while i'm home :) |
| 09:33.53 | seebs | I love my current one. |
| 09:34.05 | benjk | you can always use the cellphone to drop the battery |
| 09:34.06 | seebs | It has two key features that make me like it. |
| 09:34.22 | seebs | The first is that the only cable it uses is a STANDARD CONNECTOR USB cable, for both power and connection to computer. |
| 09:34.24 | linagee | Strom_C: why do you despise it? |
| 09:34.37 | seebs | The second is that it can be switched between STANDARD USB mass storage device and STANDARD USB modem modes. |
| 09:34.37 | Strom_C | linagee: because I hate low-bandwidth audio codecs |
| 09:34.52 | seebs | No "sorry, we can only support Windows XP" crap. |
| 09:35.11 | linagee | Strom_C: oic. i think it would also introduce more latency intro your audio stream. heh. |
| 09:35.14 | seebs | It's a phone. It takes one cable to plug it in, I will never have to buy a special cable at extra cost for it, and it works as a modem under NetBSD |
| 09:35.16 | seebs | Whee! |
| 09:35.28 | Strom_C | seebs: if you capitalize "standard" one more time, I'm going to beat you to death with a copy of the los angeles yellow pages :) |
| 09:35.48 | benjk | seebs, mine works over Bluetoosh as a modem while I am talking on it |
| 09:35.55 | benjk | I was very surprised |
| 09:36.05 | seebs | Nice, ben! |
| 09:36.11 | benjk | I was connected through it and browsing and the phone rang |
| 09:36.18 | linagee | benjk: EVDV? |
| 09:36.19 | seebs | Strom: If you had dealt with the previous three or phones I had, you wouldn't feel that way. |
| 09:36.20 | Strom_C | right, I'm going to sleep. |
| 09:36.21 | benjk | I was really confused |
| 09:36.26 | linagee | Strom_C: gnite |
| 09:36.42 | benjk | I didn't expect it to ring |
| 09:37.01 | benjk | and I picked up the call and my connection was still going |
| 09:37.04 | linagee | benjk: were you currently transmitting data at that exact moment, or just connected? |
| 09:37.07 | seebs | G'night, Strom. |
| 09:37.21 | benjk | I was connected and browsing |
| 09:37.34 | linagee | benjk: it's normal to stay connected and be able to get an incoming call. but not to have data at the same moment |
| 09:37.42 | *** join/#asterisk adorah (n=admin@87.68.168.3.cable.012.net.il) |
| 09:37.50 | linagee | benjk: which phone? |
| 09:37.59 | benjk | yeah, but this one can do both voice and data at the same time |
| 09:38.06 | linagee | benjk: EVDV? |
| 09:38.07 | benjk | Nokia 6680 or so |
| 09:38.10 | benjk | UMTS |
| 09:38.13 | linagee | hm |
| 09:38.33 | linagee | maybe UMTS does that? never played with it |
| 09:38.42 | benjk | the Japanese phone companies always have to change all the model numbers and make up their own model numbers |
| 09:38.55 | benjk | so I can never remember what model I have |
| 09:39.02 | benjk | but I think its a 6680 |
| 09:39.11 | Strom_C | linagee: it's EV-DO, silly |
| 09:39.21 | linagee | Strom_C: EVDO != EVDV. ;) |
| 09:39.56 | benjk | yeah, I think it can only do that if its in a UMTS network |
| 09:40.14 | benjk | not sure if it can do the same trick on a GSM network |
| 09:40.58 | benjk | I actually tried it out when I was in the US and I can't remember now if it worked or not (both data and voice together) |
| 09:41.22 | benjk | that was only this spring, geez, I cant remember now |
| 09:42.55 | benjk | anyway, the bluetooth thing is great, not having to fiddle with cables, not even having to pull the phone out of the pocket to get online with the notebook |
| 09:43.07 | linagee | ah, here we go. http://www.cellphonestalk.com/glossary/1xev-dv.html |
| 09:43.56 | benjk | but thats based on CDMA2000 |
| 09:44.17 | benjk | mine is the GSM flavour of UMTS |
| 09:44.21 | benjk | W-CDMA |
| 09:44.35 | benjk | or 3GSM as they now call it |
| 09:44.57 | linagee | weird how they don't really show "data or voice" as so much of a feature anywhere... |
| 09:45.07 | linagee | you'd probably have to hunt in the user manual to find that sort of thing. |
| 09:45.24 | benjk | data or voice, take you pick, once picked you cannot revert :) |
| 09:45.30 | X-Rob | bluetooth chews batteries. It sucks severely. |
| 09:45.38 | benjk | :( |
| 09:45.46 | linagee | X-Rob: get a bigger battery. :-D |
| 09:46.01 | X-Rob | Mmm. Kinda defeats the whole no-wires thing |
| 09:46.05 | benjk | no, Strom said you need to drop the battery for 1/2 second |
| 09:46.20 | X-Rob | I'll stick with the USB adaptor that also charges the phone |
| 09:46.25 | linagee | benjk: the best way to do data + voice is to have two phones. oh joy. now you get to pay them twice. hah |
| 09:46.35 | benjk | yeah, charges the phone my ass |
| 09:46.48 | benjk | sucking my notebook battery dry in the process |
| 09:47.10 | benjk | that's a nogo |
| 09:47.36 | benjk | linagee I always have two phones when I travel |
| 09:47.43 | seebs | I really like the USB thing that charges the phone, because I virtually never run my laptop off battery. |
| 09:47.52 | linagee | benjk: is one a personal phone>? |
| 09:47.53 | seebs | I think my phone has an option to disable charging. |
| 09:47.54 | benjk | one for emergency roaming, the other for poor mans roaming |
| 09:48.12 | benjk | poor man's roaming => purchase a local SIM card |
| 09:48.20 | benjk | prepaid local SIM card |
| 09:49.39 | linagee | benjk: doesn't seem too hard. most charging circuits these days should do that already to "not overcharge a battery" |
| 09:49.40 | benjk | there were some phones in the mid 90s who had such a feature |
| 09:50.00 | benjk | you could disable the advice of charge signal on the phone |
| 09:50.19 | linagee | benjk: you could always take out the battery. ;) |
| 09:50.21 | benjk | and then talk without having to pay any more even if the balance went below 0 |
| 09:50.32 | linagee | ?? |
| 09:51.00 | benjk | I was working as a project manager on a prepaid mobile project when our customers discovered this |
| 09:51.08 | benjk | we lost about 10.000 USD per day |
| 09:51.17 | benjk | expensive overseas calls |
| 09:51.26 | linagee | that's silly. the timing of the prepaid mobile is based off the battery? lol |
| 09:51.34 | seebs | Ben, thanks again for all your help. |
| 09:51.40 | benjk | no its based of a signal called AoCC |
| 09:51.42 | seebs | I would NEVER have been able to do this without the input. |
| 09:51.51 | benjk | seebs, you're welcome |
| 09:52.24 | benjk | if you find the time, please consider contributing your NetBSD patches to the OpenPBX Zaptel branch |
| 09:52.29 | linagee | benjk: i'd think they would do all charge accounting on the cell provider side... |
| 09:52.42 | benjk | linagee, NOW they do :) |
| 09:52.48 | linagee | benjk: wtf? cool. lol |
| 09:53.01 | benjk | back then some didn't |
| 09:53.07 | benjk | hehe |
| 09:53.19 | linagee | benjk: let me guess. they stored the balance on the SIM card in cleartext. LOL |
| 09:53.23 | benjk | wouldn't work now because they all changed the prepaid platforms |
| 09:53.39 | benjk | they stored the balance on the SIM yes |
| 09:53.43 | linagee | wow |
| 09:53.59 | benjk | and the SIM was programmed to count down the balance when it received the AoC signal |
| 09:54.11 | linagee | AoC? |
| 09:54.16 | linagee | Advice of Charge? |
| 09:54.19 | benjk | but some smart engineers at Nokia put a back door in the phone firmware |
| 09:54.27 | benjk | Advice of Charge yes |
| 09:54.31 | linagee | lol |
| 09:54.55 | linagee | benjk: why would you need a back door? LOL. just take out the SIM, put it in a computer, screw with the values...? |
| 09:54.58 | benjk | so somehow it leaked out how you could disable the phone's listening to the AoC signal |
| 09:55.13 | benjk | no, that would have been much more difficult to do |
| 09:55.29 | *** join/#asterisk Ebola (n=Ebola@81-86-155-65.dsl.pipex.com) |
| 09:55.31 | benjk | all you had to do was punch in some 16 digit code at some prompt |
| 09:55.36 | linagee | hahaha |
| 09:55.37 | seebs | I might well have a look at it, anyway. |
| 09:55.39 | benjk | and the phone would no longer listen to the AoC |
| 09:55.40 | linagee | you're kidding. lol |
| 09:55.57 | benjk | it was meant for testing |
| 09:56.05 | benjk | not meant to be in the production firmware |
| 09:56.09 | benjk | and not meant to leak out |
| 09:56.25 | linagee | i wonder if they kept alive the accounts for those phones. lol |
| 09:56.29 | benjk | but it somehow leaked and some TV station in the Middle East broad cast it |
| 09:56.51 | benjk | and all of a sudden we had hourlong phone calls to countries in the Middle East |
| 09:57.00 | benjk | and the account balances went below 0 |
| 09:57.09 | benjk | it was "I told you so" |
| 09:57.30 | linagee | benjk: when i had my PPC6600, you could put it in a signal strength engineering mode. ##DEBUG. also, there's a mode where Sprint enters your provisioning info. heh. (CDMA doesn't have SIM cards, so yes they do this manually) |
| 09:57.35 | benjk | but then it was too late, we had purchase the prepaid platform and this was going on with a loss of about 10000 USD per day |
| 09:58.07 | benjk | I was lucky enough to hire a mathematics whiz |
| 09:58.13 | benjk | this was in Romania |
| 09:58.24 | linagee | what did the mathematics whiz do? |
| 09:58.31 | benjk | the salaries there were like 80 USD at university |
| 09:58.33 | linagee | compute losses? lol |
| 09:58.42 | benjk | and we paid them like 500-600 USD |
| 09:58.52 | benjk | so everybody wanted a job at the mobile phone company |
| 09:58.57 | linagee | dollars or thousand dollars? |
| 09:59.10 | seebs | What an awesome day. I am gonna go play video games to finish spinning down, then sleep. |
| 09:59.16 | benjk | we had a job fair to find call center staff and other general IT admin staff |
| 09:59.23 | linagee | benjk: $600k?? |
| 09:59.39 | benjk | so this mathematics PhD showed up and wanted a job in my support group |
| 09:59.46 | benjk | 600 USD per month |
| 09:59.56 | linagee | oh! |
| 10:00.03 | linagee | wow that's not much at all! |
| 10:00.21 | benjk | I said, I don;t have any job for you, but we just now have this problem with a massive fraud, can you solve it? |
| 10:00.35 | benjk | he was a PhD in statistical analysis |
| 10:00.45 | seebs | ... Ooooh, FUN. |
| 10:00.47 | linagee | benjk: did you have to collect all the old phones? |
| 10:00.48 | *** join/#asterisk DataCompBoy (n=datacomp@gw6.lvs.ru) |
| 10:00.53 | benjk | the software we bought for him was 4000 USD |
| 10:00.53 | DataCompBoy | Hello! |
| 10:00.57 | benjk | more than his salary |
| 10:01.25 | DataCompBoy | Can someone help me with smaaal but stupid problem? :) |
| 10:01.30 | benjk | within less than 2 weeks he had found almost all of the people who defrauded us |
| 10:01.47 | linagee | benjk: how? just by people charging over their average? |
| 10:01.55 | benjk | call analysis and movements around the city and stuff |
| 10:02.10 | linagee | benjk: what if someone was smart enough to only use their average minutes? heheh |
| 10:02.24 | benjk | those people all had post paid phones too |
| 10:02.31 | benjk | so we were able to correlate |
| 10:02.42 | benjk | by heavy statistical analysis |
| 10:02.44 | linagee | yikes. you mean they charged them? |
| 10:03.02 | benjk | no we were able to correlate to postpaid phones |
| 10:03.14 | DataCompBoy | I have dialplan: 111,1,Answer; 111,2,DateTime; 222,1,Answer; 222,2,Conference("222|Sd"). When I call to 111 from softphone (X-Line) I hear silence (while console write that it playing numbers). When I call to 222 I got correctly phrase that I'm alone in conference. What's wrong?! |
| 10:03.17 | benjk | like always in the same place always logging in and out |
| 10:03.27 | linagee | benjk: heavy statistical analysis doesn't show everything. if i only make a phone call a week and i make only a phone call a week after.. ? |
| 10:03.42 | benjk | you are constantly moving around though |
| 10:03.51 | benjk | and the cell towers triangulate all the time |
| 10:04.06 | benjk | the cell companies have logs of that |
| 10:04.17 | linagee | benjk: who has access to the giant map that shows where all the citizens are? LOL. |
| 10:04.27 | benjk | if you move around with two phones registered to the same company you can find out |
| 10:04.31 | DataCompBoy | Both times connected with G711u, and that the only allowed codec. |
| 10:04.37 | linagee | benjk: i bet the govt would drool at that if they don't have it already |
| 10:04.47 | benjk | the one phone is a postpaid phone for which we know who it is |
| 10:04.52 | E-bola | hi |
| 10:05.01 | DataCompBoy | E-bola: hi |
| 10:05.01 | benjk | the other phone is the prepaid one we want to find out who it belongs to |
| 10:05.35 | benjk | combine that with the numbers they call and the numbers they receive calls from, and you catch the fraudster |
| 10:05.54 | linagee | i see. and although you can't get millimeter accuracy, you know they are within the same cells for a period of time |
| 10:06.05 | benjk | yes |
| 10:06.13 | benjk | also they may make a call to 5551234 |
| 10:06.19 | benjk | a lot |
| 10:06.30 | linagee | 5551234? |
| 10:06.37 | *** join/#asterisk A-Tuin (n=a-tuin@steves.ip.v4.me.uk) |
| 10:06.39 | benjk | and the other phone that is always in the same spot, receives calls a lot from 5551234 |
| 10:07.11 | benjk | because they call out on the prepaid, and receive calls on the postpaid |
| 10:07.29 | linagee | benjk: did you have multiple provider's databases?? i mean, how many people carry around two cells from the same provider? |
| 10:07.46 | benjk | so over a period of about 10 days, you get a pretty accurate picture |
| 10:07.58 | benjk | there was only us and another provider |
| 10:08.14 | benjk | this was in Romania in 1997 |
| 10:08.34 | linagee | benjk: how long exactly did you keep records of where everyone was?? |
| 10:09.07 | linagee | or do they ever delete the records? LOL |
| 10:09.16 | benjk | those logs are not under the billing dept, they are kept by engineering folk and regularly rotated |
| 10:09.23 | linagee | can you trace down where everyone is for the past 30 years? hah |
| 10:09.32 | DataCompBoy | ugh:( |
| 10:09.33 | benjk | so I don't think we would have had those records for long |
| 10:09.40 | benjk | but this was fresh, fraud in progress |
| 10:10.23 | linagee | benjk: what if the govt really didn't care exactly where you are, but they want to make sure you follow for instance, residency laws. hah |
| 10:10.28 | benjk | but there was a shitload of data and without that mathematics whiz, we wouldn't have had a clue how to go about this |
| 10:10.47 | linagee | benjk: like what if they see your cell phone in another state for a year or something, and you haven't renewed your license plates in the correct state. hah |
| 10:10.54 | benjk | human resources complained that I paid him 50 USD over the usual |
| 10:11.08 | benjk | I was really pissed off abou tthat |
| 10:11.08 | linagee | benjk: 50 dollars?!?! |
| 10:11.15 | benjk | yeah 650 per month |
| 10:11.21 | linagee | benjk: 650 dollars? |
| 10:11.27 | linagee | benjk: where are you right now? |
| 10:11.29 | benjk | whereas the guidelines were 500-600 |
| 10:11.33 | linagee | what year was this? hah |
| 10:11.33 | benjk | in Japan |
| 10:11.38 | benjk | 1997 in Romania |
| 10:11.49 | benjk | Romania is rather poor you know |
| 10:11.55 | linagee | that seems like such a small amount of money! |
| 10:12.00 | benjk | extremely well educated people there |
| 10:12.05 | linagee | Romania, Japan, wow. |
| 10:12.06 | benjk | excellent universities |
| 10:12.27 | benjk | but the country was run down by a communist dictatorship |
| 10:12.27 | linagee | I wonder if they can track you going to an airport. hrm. |
| 10:12.44 | benjk | when we arrived there it was dark and greay |
| 10:12.47 | linagee | ok... citizen #95182312 is in the plane... lol |
| 10:13.30 | benjk | after a year, the competition and investment of the two mobuile phone companies (us, Vodafone and them, France Telecom) the place really lightened up, totally changed |
| 10:14.01 | adorah | ROmania is now moving fast to market economy..a lot of action regarding VOIP |
| 10:14.11 | linagee | adorah: why? |
| 10:14.58 | benjk | because we (the two mobile phone companies) pumped so much money in the local economy, it made a big difference |
| 10:14.58 | adorah | linagee: I know a lot of action goes there with communication..a lot of countrymen r staying there and making deals in telco |
| 10:15.07 | benjk | shows popped up all over the place |
| 10:15.19 | benjk | people got salaries and spend money in the shops |
| 10:15.26 | linagee | whoa |
| 10:15.29 | benjk | we hired local companies to do stuff for us |
| 10:15.29 | linagee | people got money>!?! |
| 10:15.35 | linagee | $600 USD?!?! |
| 10:15.37 | linagee | geez |
| 10:15.38 | benjk | those companies hired |
| 10:15.53 | benjk | their employees woudl buy mobile phones |
| 10:16.00 | linagee | i can't imagine $600 being a lot of money even back in '97 |
| 10:16.06 | benjk | so they money would circulate |
| 10:16.13 | benjk | not just the salaries |
| 10:16.20 | linagee | profit |
| 10:16.24 | adorah | In romania u could buy a nice house for 2000$ back than.. |
| 10:16.30 | benjk | we built a complete microwave transmission network spanning all of Romania |
| 10:16.33 | linagee | wtf? out of mud? |
| 10:16.42 | linagee | adorah: $2000??? |
| 10:16.45 | benjk | because we figured the government network wasn't good enough |
| 10:17.00 | linagee | adorah: i know people who pay that much A MONTH in rent! geez! |
| 10:17.05 | benjk | our CEO was elected Man of the year of Romania |
| 10:17.10 | adorah | yup..not inin some towns and villages not far from the capitol |
| 10:18.27 | benjk | we installed prepaid mobile public telephon boxes in villages which never had any telephone service before |
| 10:18.49 | linagee | mobile telephone box? |
| 10:19.09 | benjk | the local grocery store or post office would sell the prepaid SIM cards and people would go to the public telephone box (GSM) to make calls and listen to their voicemail |
| 10:19.16 | *** join/#asterisk daysmen3 (n=primus@host86-139-50-238.range86-139.btcentralplus.com) |
| 10:19.26 | adorah | that is a common parctice in many 3rd world countries..public mobule booths.. |
| 10:19.27 | benjk | and there was no phone lines into those villages |
| 10:19.35 | linagee | oic |
| 10:19.53 | linagee | so it's a telephone box/room that has a cellphone... |
| 10:19.53 | benjk | we could do it because we had built a microwave transmission network for the whole country |
| 10:20.05 | benjk | not a room, just a public phone that uses GSM |
| 10:20.16 | adorah | In Africa and countries like India and Indonesia large parts r covered only by cllular netorks |
| 10:20.38 | benjk | those people had to wait 20 years for a phone line |
| 10:20.49 | linagee | benjk: what's wrong with a public phone box that uses satellite? $10/minute. lol |
| 10:21.00 | benjk | when we rolled out GSM prepaid, they could just walk into a phone shop and buy a phone |
| 10:21.07 | benjk | and get connected right away |
| 10:21.20 | linagee | were you always expanding coverage or something? |
| 10:21.27 | linagee | benjk: were the phones recycled? |
| 10:21.36 | adorah | I worked in a company that develops fixed cellular gateways both for such places and VOIP providers.. |
| 10:21.37 | benjk | imagine if you want to open a business and you cannot get a phone line in 20 years |
| 10:21.40 | linagee | benjk: old used GSM phones? |
| 10:21.48 | benjk | you can't open a business then |
| 10:21.58 | benjk | no business, no job creation |
| 10:22.10 | linagee | benjk: you could say the same for internet. :-D |
| 10:22.18 | benjk | so when we started GSM service there, all those businesses popped up |
| 10:22.33 | benjk | little street shops, boutiques, cafes, everything |
| 10:22.39 | linagee | benjk: so the people went were the coverage was? lol. interesting. |
| 10:22.44 | benjk | the whole place changed totally over the course of a few months |
| 10:22.54 | benjk | we had good coverage |
| 10:23.21 | linagee | benjk: interesting too how it changed by using the media you installed. lol. (put in phone networks, people call their friends, tell them where more coverage can be found) |
| 10:23.23 | benjk | this was an incredible live lesson in market economics |
| 10:23.37 | benjk | you get told in school that this is what ought to happen |
| 10:23.41 | benjk | but its all academic |
| 10:23.52 | benjk | in Romania I have seen it happening before my own eyes |
| 10:24.08 | adorah | Well it didn't happen that way in Africa..despite GSM coverage.. |
| 10:24.10 | linagee | benjk: you mean huge demand (from word of mouth) and very little supply? heh |
| 10:24.21 | benjk | Africa has deeper problems |
| 10:24.50 | benjk | for example in Romania there were the Koreans who had bought the run down car manfucaturing company |
| 10:25.00 | benjk | they built a new car factory |
| 10:25.07 | benjk | and made cheap Korean cars |
| 10:25.11 | linagee | benjk: were the phones used or not? |
| 10:25.14 | benjk | again, lots of jobs |
| 10:25.28 | adorah | In Africa they use fixed cellular to contact their relatives in Europe vua VOIP to send them some more money..hehe |
| 10:25.35 | linagee | benjk: i'm just wondering if thats where all of our used phones are going. lol |
| 10:25.38 | benjk | so everthing was falling into place in ROmania in those years |
| 10:25.48 | *** join/#asterisk vgster (n=vgster@cpc2-ledn1-0-0-cust944.leed.cable.ntl.com) |
| 10:26.09 | benjk | and still Romania has a distance to go |
| 10:26.36 | benjk | you cant just make up for 50 years of communist mismanagement and corruption in a few years |
| 10:27.14 | linagee | benjk: so what happens when mesh really takes off and the whole world starts having free internet? :-D |
| 10:27.15 | adorah | Well at least thx to the late dictator they had a very low debt ratio.. |
| 10:27.46 | benjk | they have this building called the Palace of the people |
| 10:27.58 | benjk | its the largest building in the world in floorspace |
| 10:28.05 | benjk | bigger than the Pentagon |
| 10:28.09 | benjk | in floorspace |
| 10:28.17 | benjk | its huge |
| 10:28.25 | benjk | all in marble |
| 10:28.37 | benjk | but the folks didn't have enough to eat |
| 10:28.58 | benjk | they still don't know what to do with that buikding |
| 10:28.59 | linagee | so they went online and downloaded the schematics for more food? |
| 10:29.09 | adorah | bcoz they exported all the food to cut foreign debt..that was what brought him down.. |
| 10:29.11 | benjk | we hired a room in there for our Xmas party |
| 10:29.22 | benjk | one room for 1500 employees |
| 10:29.52 | benjk | beautiful architecture too |
| 10:31.22 | benjk | http://www.globalgayz.com/RomaniaGallerABucharest/IMG_0555.html |
| 10:31.49 | linagee | nice domain. heh |
| 10:32.22 | benjk | not my domain |
| 10:32.47 | benjk | I just googled for pictures of the House of People Bucharest |
| 10:33.53 | linagee | benjk: have you heard of olpc? |
| 10:34.01 | benjk | you are looking at one of the most fascinating buildings in the world and all you notice it a weird domain name? |
| 10:34.12 | benjk | no |
| 10:34.14 | linagee | :-D |
| 10:34.21 | linagee | benjk: http://laptop.org/ |
| 10:34.30 | *** join/#asterisk coppice (n=chatzill@229.166.17.210.dyn.pacific.net.hk) |
| 10:34.52 | linagee | benjk: an almost "crazy pipe dream" of putting a cheap laptop into the hands of every third world child in the world. |
| 10:35.00 | benjk | ah ok, yes read about it |
| 10:35.05 | linagee | benjk: they plan on making them all wireless mesh nodes. :-D |
| 10:35.49 | benjk | I like the idea |
| 10:36.34 | linagee | it's pretty neat |
| 10:36.46 | linagee | it reminds be of robocop though. lol. almost makes me think they got the idea from that. |
| 10:37.08 | linagee | the girl in the beginning of the movie had a "learning computer" (laptopish looking device) |
| 10:37.36 | linagee | it was supposed to portray a post-apocolyptic society. in that you don't even go outside your house for school |
| 10:38.22 | coppice | i thought not going to school was becoming the normal pattern in many places :-) |
| 10:38.50 | linagee | er, it's in Robocop 3 |
| 10:38.59 | linagee | (according to the wiki) |
| 10:40.52 | benjk | coppice, that's why you need extended regular expressions |
| 10:43.33 | linagee | nice. OLPC. http://wiki.laptop.org/go/Image:EToys_-_new_display.jpg |
| 10:45.38 | *** join/#asterisk saftsack (n=saftsack@p54A7C55A.dip.t-dialin.net) |
| 11:08.48 | *** join/#asterisk Geliman (n=scorpio@unaffiliated/drkshdw) |
| 11:08.50 | *** join/#asterisk ToTo (n=ToTo@host149-109-dynamic.58-82-r.retail.telecomitalia.it) |
| 11:11.46 | fish_ | re |
| 11:15.50 | *** join/#asterisk LoneShadow (n=duh@59.92.177.64) |
| 11:26.54 | *** join/#asterisk lukketto (n=lukketto@host84-83-dynamic.57-82-r.retail.telecomitalia.it) |
| 11:44.15 | *** join/#asterisk jmls (n=asterisk@host81-159-167-80.range81-159.btcentralplus.com) |
| 11:48.09 | jmls | Currently, I send a popup to an agent asking them to accept the call or not. However, they are complaining that they do not *see* the popup (it's onscreen for 7 seconds) and I was wanting (have been asked to provide) a facility where I can "ping" an agents phone before sending the popup. |
| 11:48.09 | jmls | This would make them aware that there is a call about to come in. The best I could come up with so far is a Dial(SIP/Exten,1,g). This seems to work, but I was wondering if there was another more elegant mechanism. We currently use Cisco 7960 phones. |
| 11:58.42 | E-bola | why do they need to know that a call is about to come in? |
| 11:58.51 | E-bola | doesnt that sort of defy the purpose of their phone ringing? |
| 12:00.21 | jmls | We allow our agents to decide if they want / can / need to take the call, so the popup allows them to reject or accept the call |
| 12:00.25 | *** join/#asterisk DanTMG (n=danielga@203-214-135-194.perm.iinet.net.au) |
| 12:01.22 | jmls | they are complaining that they don't see the poup, so we are getting a lot of "timeout" rejections. If it "pings" in their ear, they then know that a call is about to be offered to them |
| 12:01.42 | benjk | maybe they should see a doctor |
| 12:01.46 | jmls | ;) |
| 12:02.13 | jmls | I had thought of finding some way of building up an electrical charge in the earpiece |
| 12:02.15 | benjk | ringing ears can be a serious thing |
| 12:02.32 | jmls | and then "zapping" them. That would get their attention :> |
| 12:02.51 | benjk | molar cellphone implant |
| 12:03.08 | jhiver | mhhh, seems like a bit tricky to do |
| 12:03.16 | jmls | as long as I can cause them as much pain as they cause me :) |
| 12:03.34 | jhiver | ideally you'd want some kind of "visual" ring on the phone when a call is waiting on the queue, and a "normal" ring for other calls |
| 12:03.55 | benjk | its called a light bulb |
| 12:04.01 | jhiver | isn't it possible to use some kind of distinctive ring to to that? |
| 12:04.10 | jmls | yup, the cisco's have the big red flashing light |
| 12:04.14 | jhiver | well it would be better with a LED i would think ;) |
| 12:04.24 | jmls | I am using bellcore-dr3 for the 1s call. |
| 12:04.32 | benjk | Distinctive LED |
| 12:04.34 | jmls | it is slightly different, so it seems to be working. |
| 12:04.35 | benjk | DLED |
| 12:04.41 | jmls | Just wanted it more elegant |
| 12:05.30 | jhiver | yeah... i really have no idea on how to do that :) |
| 12:06.00 | jhiver | i know it's not very useful answer... |
| 12:06.17 | benjk | maybe not the right time of day now |
| 12:06.20 | jmls | no that's cool. It's not as if I have no options. |
| 12:06.39 | benjk | there are lots of distinctive ring specialists here during US daytime during the week |
| 12:07.55 | *** join/#asterisk zotz (n=zotz@24.244.163.225) |
| 12:08.30 | benjk | we have got half of chan_bluetooth working |
| 12:08.46 | benjk | you could always have your agents yell into a bluetooth headset |
| 12:08.55 | benjk | but they won't hear anything |
| 12:09.02 | *** join/#asterisk Winkie (n=urmom@cpc2-stre3-0-0-cust344.bagu.cable.ntl.com) |
| 12:09.02 | benjk | like I said, its half working |
| 12:09.06 | benjk | one way audio |
| 12:09.35 | jhiver | mind you... |
| 12:09.41 | jhiver | which way is it working? |
| 12:09.43 | benjk | yell once, take the call |
| 12:09.51 | benjk | yell twice, don't take the call |
| 12:09.55 | jhiver | bluetooth -> asterisk, or asterisk -> bluetooth? |
| 12:10.16 | jhiver | ok, bluetooth -> asterisk is ok then... |
| 12:10.17 | benjk | you can speak to the openpbx end from the bluetooth device |
| 12:10.25 | benjk | but you cant hear anything |
| 12:10.38 | jhiver | too bad... otherwise they would have been able to have a device that "rings in their ear" :) |
| 12:10.46 | HaMYaI | everytime I reboot my asterisk box the order in /proc/zaptel/* varies from time to time |
| 12:10.58 | HaMYaI | anyone experienced this before? |
| 12:11.09 | benjk | what do you mean by "order" |
| 12:11.16 | *** join/#asterisk Ebola (n=Ebola@user-54463be2.lns2-c12.dsl.pol.co.uk) |
| 12:11.27 | HaMYaI | benjk: the numbers "1" , "2" ... |
| 12:11.41 | *** join/#asterisk Winkie (n=urmom@cpc2-stre3-0-0-cust344.bagu.cable.ntl.com) |
| 12:12.05 | benjk | that order is solely determined by ls |
| 12:12.06 | HaMYaI | benjk: sometimes one gets loaded before another no matter which is loaded first in /etc/modprobe.d/zaptel |
| 12:13.06 | HaMYaI | benjk: ok, for example there are two cards in my box, TE110P and TDM400 with 4xFXS |
| 12:13.07 | benjk | depends on the device readiness |
| 12:13.48 | HaMYaI | benjk: sometimes "1" belongs to TE110P and sometimes it belongs to the TDM400 |
| 12:14.05 | HaMYaI | benjk: but I can't have this varies from time to time |
| 12:14.39 | HaMYaI | benjk: zaptel will not loaded correctly because span 1 in /etc/zaptel.conf is TE110P |
| 12:17.36 | HaMYaI | any knows if this can be fixed |
| 12:19.23 | benjk | can you not control this in zapata.conf |
| 12:22.11 | DanTMG | benjk: are you doing a project for this bluetooth thing? as we used to do installs for a company that have devices that allow for offices of just headsets and would be interested in assisting in any way with developing it especially since we have a decent amount of bluetooth networking hardware lying around here |
| 12:22.43 | benjk | CtRiX is working on chan_bluetooth now |
| 12:23.00 | *** join/#asterisk stkn_ (n=stkn@gentoo/developer/pdpc.active.stkn) |
| 12:23.18 | benjk | I am sure we would very much welcome some support for this |
| 12:25.37 | benjk | DanTMG? |
| 12:26.35 | DanTMG | yes |
| 12:28.35 | DanTMG | excuse my ignorance but is this working with jsr-82 for the software on the phone or is it all done through the headsets with voice rec on the Asterisk end? |
| 12:29.29 | X-Rob | chan_bluetooth gives me wood. |
| 12:29.39 | X-Rob | boiiing |
| 12:29.57 | benjk | DanTMG pm |
| 12:34.49 | *** join/#asterisk DasTech (n=DasTech@d47-69-168-46.col.wideopenwest.com) |
| 12:35.26 | DasTech | Kram said everyone was firedtill monday |
| 12:39.46 | *** join/#asterisk amitev (n=adrian@87.126.111.124) |
| 12:41.15 | *** part/#asterisk amitev (n=adrian@87.126.111.124) |
| 12:41.57 | HaMYaI | benjk: what do you mean by not to control it? can I make it dynamic? |
| 12:42.57 | benjk | you should make it static |
| 12:43.29 | benjk | assign channels 1,2,3,4 to one card and channels 5,6, 7... to the other |
| 12:45.53 | HaMYaI | benjk: yeah, they are like that now, 1-31 assigned to TE110P and 32-35 to TDM400 |
| 12:46.17 | benjk | and it doesn't follow that assignment? |
| 12:46.35 | *** join/#asterisk ppyy (n=lala@222.185.19.61) |
| 12:46.59 | HaMYaI | but it was a problem when the /proc/zaptel/1 changes to TDM400 |
| 12:47.29 | *** join/#asterisk stkn_ (n=stkn@gentoo/developer/pdpc.active.stkn) |
| 12:47.31 | benjk | it shouldn't be |
| 12:47.38 | HaMYaI | as i told you, /proc/zaptel/1 varied all the time |
| 12:48.22 | HaMYaI | yeah, it never happened before until I added a new card, TDM400 |
| 12:48.53 | benjk | well, with only one card it couldnt really happen, can it |
| 12:49.09 | De_Mon | jmls did you come up with any other solutions aside from Dial(SIP/Exten,1,g)? |
| 12:49.12 | *** join/#asterisk nassy (n=nassy@207-38-197-201.c3-0.wsd-ubr1.qens-wsd.ny.cable.rcn.com) |
| 12:49.50 | HaMYaI | benjk: I can't happen =) |
| 12:50.39 | HaMYaI | benjk: what about the order in /etc/modprobe.d/zaptel? do you think it has anything to do with this? |
| 12:51.16 | benjk | that order changed too? |
| 12:51.17 | *** join/#asterisk riddlebox (n=james@24-171-10-102.dhcp.stls.mo.charter.com) |
| 12:51.32 | *** join/#asterisk RoyK (n=roy@ti211310a080-3143.bb.online.no) |
| 12:53.38 | HaMYaI | benjk: nope, the wcte11xp came after that of wctdm |
| 12:53.53 | HaMYaI | I mean the order |
| 12:54.37 | benjk | so the order in /etc/modprobe is always the same, adn the order in /proc/zaptel changes from boot to boot |
| 12:54.45 | HaMYaI | I thought if wctdm gets loaded first, it might cause some problems |
| 12:55.03 | HaMYaI | benjk; yes, it is like that |
| 12:55.35 | HaMYaI | benjk: and the /etc/modprobe.d/zaptel gets overwritten everytime I compile zaptel |
| 12:55.40 | benjk | I dunno what to suggest, you could read the system log and see if there is anything that looks suspicious |
| 12:56.16 | HaMYaI | benjk: so, by the way, it's not convenient to change the order in /etc/modprobe.d/zaptel manually |
| 12:56.37 | benjk | I would study the system logs |
| 12:57.01 | HaMYaI | benjk; where's the log where these cards were loaded/ |
| 12:57.17 | benjk | the system log |
| 12:57.28 | HaMYaI | benjk: wondering how the pci cards were detected |
| 12:57.43 | benjk | all the logs are in /var/log |
| 12:58.04 | De_Mon | When someone is on a call, how can I "whisper" in one of the callers ears? ie tech is on the phone with customer and teir 2 is listening to the call and only talking to the teir3 support and not the customer? |
| 12:58.31 | De_Mon | replace teir3 with 'tech' |
| 12:59.34 | HaMYaI | benjk: changes were only made to wtmp,secure and lastlog. I don't seem to find what I look for |
| 13:00.03 | benjk | lastlog |
| 13:00.10 | benjk | also system.log |
| 13:00.48 | RoyK | HaMYaI: type 'dmesg' |
| 13:00.58 | RoyK | that shows the circular kernel log |
| 13:00.59 | jmls | de_mon: not yet, although it seems to work ok |
| 13:01.43 | RoyK | that log is also written to some log in /var/log - see /etc/syslog.conf for info about where |
| 13:03.09 | HaMYaI | RoyK: oh, Ok |
| 13:03.15 | De_Mon | jmls is that approach instead of using queues or, through the existing queue system? |
| 13:03.55 | RoyK | wtf? http://www.networkworld.com/news/2006/091206-von-sam-houston.html |
| 13:09.22 | E-bola | cool story royk |
| 13:12.26 | RoyK | "Forgive them for they know not what they do" |
| 13:18.22 | jmls | de_mon: with queues. We use jabber to send them an ack message with all the call details |
| 13:18.35 | Nivex | RoyK: I'm sure they'll be calling Digium for a nice fat support contract soon :) |
| 13:18.50 | Hymie | someone must untangle the secret mysteries of the uniden UIP-200, and make the lights come ALIVE! ALIVE I SAY! |
| 13:19.33 | RoyK | Nivex: poor souls |
| 13:22.30 | De_Mon | jmls that sounds really cool, can you set me up with the scripts / config snipits needed to do it? |
| 13:23.51 | *** join/#asterisk yogurt2ungue (n=yogurt2u@24-48-231-201.fibertel.com.ar) |
| 13:26.36 | *** join/#asterisk Borgon (n=l3orgon@host-64-234-31-131.nctv.com) |
| 13:26.58 | Borgon | hello am tryign to install asterisk through cvs, but cvs.digium.com is a unknown host. Where else can i get it? |
| 13:27.41 | De_Mon | Borgon http://www.asterisk.org/download |
| 13:27.43 | De_Mon | try harder! |
| 13:28.31 | fish_ | i have (still) a problem with asterisk. sometimes, well, most times my asterisk does not answer to isdn calls. the console shows -- Executing Wait("CAPI/contr1/3954675-1", "1") in new stack which is the first command of my voice menu, the second is answer but it never get answered. but sometimes it just works fine |
| 13:29.03 | *** join/#asterisk zeppelin_ (n=zeppelin@201.35.139.252) |
| 13:43.10 | *** join/#asterisk tzafrir_laptop (n=tzafrir@bzq-179-75-202.cust.bezeqint.net) |
| 13:45.36 | jhiver | Borgon, use svn, it's easier |
| 13:46.35 | jhiver | svn checkout http://svn.digium.com/svn/asterisk/branches/1.2 asterisk-1.2 |
| 13:46.37 | *** join/#asterisk `Sauron (i=sauron@h-69-3-12-50.hstqtx02.covad.net) |
| 13:47.22 | *** join/#asterisk daysmen3 (n=primus@host86-139-50-238.range86-139.btcentralplus.com) |
| 13:52.32 | *** join/#asterisk RoyK (n=roy@183.80-203-45.nextgentel.com) |
| 13:54.47 | *** part/#asterisk RoyK (n=roy@183.80-203-45.nextgentel.com) |
| 13:54.51 | *** join/#asterisk RoyK (n=roy@183.80-203-45.nextgentel.com) |
| 13:55.33 | *** join/#asterisk LoneShadow (n=duh@59.92.175.19) |
| 14:07.38 | Borgon | jhiver: thanks, i just built it from sources. |
| 14:08.29 | jhiver | np, that's usually how i do it to, svn is nice enough |
| 14:18.04 | *** part/#asterisk jmls (n=asterisk@host81-159-167-80.range81-159.btcentralplus.com) |
| 14:18.17 | *** join/#asterisk saftsack (n=saftsack@p54A7C55A.dip.t-dialin.net) |
| 14:19.23 | *** join/#asterisk wunderkin (n=kev@ip68-226-113-228.ph.ph.cox.net) |
| 14:22.26 | *** join/#asterisk andresmujica (n=andresmu@201.245.230.87) |
| 14:30.52 | RoyK | ~book |
| 14:30.57 | jbot | rumour has it, book is a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 14:31.17 | RoyK | ~docs |
| 14:31.18 | jbot | [docs] Documentation can be found at http://www.digium.com/index.php?menu=documentation or http://www.voip-info.org/wiki-Asterisk or http://www.asteriskdocs.org or http://www.oreilly.com/catalog/asterisk or http://www.asteriskguru.com, or http://www.astmasters.net/howtos.html |
| 14:31.20 | RoyK | ~rtfs |
| 14:31.21 | jbot | from memory, rtfs is probably read the f*cking source... |
| 14:31.23 | RoyK | :P |
| 14:31.45 | RoyK | andresmujica: take a look above |
| 14:33.03 | tzafrir_laptop | RoyK, though I usually let gcc read the source for me |
| 14:34.09 | RoyK | tzafrir_laptop: heh |
| 14:40.26 | *** join/#asterisk zotz (n=zotz@24.244.163.225) |
| 14:40.50 | *** join/#asterisk dalekurt (n=dalekurt@port0096-afo-adsl.cwjamaica.com) |
| 14:47.23 | *** join/#asterisk kannan (n=kanna@61.246.91.126) |
| 14:49.33 | *** join/#asterisk jeffik (n=Jeff@dialup-4.247.245.141.Dial1.Tampa1.Level3.net) |
| 14:55.05 | *** part/#asterisk jeffik (n=Jeff@dialup-4.247.245.141.Dial1.Tampa1.Level3.net) |
| 15:09.54 | *** join/#asterisk `Tingles` (n=tingles@S01060011d8ecb1d0.cg.shawcable.net) |
| 15:17.19 | *** join/#asterisk shap (n=shap@c-68-33-84-43.hsd1.md.comcast.net) |
| 15:18.57 | shap | Is there a way for a dial plan in extensions.conf to make reference to values that were set in the sip.conf config? In particular, I want to reference the value of 'vmexten' so that a user dialing the extension for VoiceMailMain automatically gets their own voicemail. |
| 15:22.29 | RoyK | shap: if it's in a variable somewhere, it is. if it isn't, it isn't unless you use AGI to parse sip.conf yourself |
| 15:22.45 | shap | So far, the only way I can see to do it is to set up a separate extensions context for each handset. |
| 15:22.57 | RoyK | which of course would be unbeleivably ugly :P |
| 15:23.23 | RoyK | wtf is vmexten anyway? |
| 15:23.37 | *** join/#asterisk zepmantra (i=whgat@AP-203.167.26.36.sysads.com) |
| 15:23.42 | RoyK | you could patch up chan_sip or whatever to have it set it as a channelvar |
| 15:24.05 | RoyK | shap: pastebin your config... there might be a workaround somewhere |
| 15:24.55 | shap | Royk: is there a list of existing channelvars someplace before I go digging into chan_sip? |
| 15:24.56 | *** part/#asterisk zepmantra (i=whgat@AP-203.167.26.36.sysads.com) |
| 15:32.12 | shap | Royk: Looks like I can get what I want by using "setvar" in the sip.conf entry. Thanks. That was a very useful pointer |
| 15:32.43 | *** join/#asterisk DarKnesS_WolF (n=wolf@196.218.72.38) |
| 15:33.36 | *** join/#asterisk ariel_ (n=Ariel@70-46-87-158.ftl.fdn.com) |
| 15:34.55 | De_Mon | shap you're working at it the wrong way |
| 15:35.35 | shap | De_Mon: I'm open to suggestions |
| 15:35.58 | De_Mon | sec |
| 15:38.03 | De_Mon | shap have you looked at http://www.voip-info.org/wiki/view/Asterisk+phone+snom |
| 15:38.10 | *** join/#asterisk Deeewayne (n=dwayne@ool-44c0d56e.dyn.optonline.net) |
| 15:38.32 | De_Mon | most notibly the line 'exten => asterisk,1,VoiceMailMain(${CALLERIDNUM}' |
| 15:38.41 | De_Mon | and 'vmexten = *97' |
| 15:39.28 | shap | Yes, but using CALLERIDNUM isn't correct. |
| 15:40.51 | De_Mon | its more correct than using vmexten... |
| 15:40.51 | shap | Yes, I agree. vmexten is not right either. |
| 15:40.51 | De_Mon | according to this vmexten tells the phone what extension to dial to reach the mailbox, not how to find which mailbox to reach ie *97 |
| 15:42.18 | shap | De_Mon: yes. I see that. One sec -- I'm testing something |
| 15:42.47 | shap | De_Mon: Yeah. CALLERID is buggered. There are actually two separate problems with it: |
| 15:43.17 | shap | de_mon: 1. There is no way to get the proper voicemail context for the voice mail box. ${CONTEXT} returns the *extension* context |
| 15:44.08 | shap | de_mon: 2. In many configurations, the proper setting for CALLERID isn't going to be your local extension number. It's going to be your external DID number (which doesn't help here at all). |
| 15:44.33 | *** join/#asterisk riddlebox (n=james@24-171-10-102.dhcp.stls.mo.charter.com) |
| 15:44.50 | De_Mon | really? i only set the external DID number when the call actually LEAVES the pbx not before |
| 15:44.51 | shap | de_mon: so the recommendation works for really simple examples, but not for richer examples. |
| 15:45.09 | shap | However, I just discovered something that does work more generally: |
| 15:45.27 | De_Mon | where are you setting callerid to the externaldid number such that it isn't the internal extension? |
| 15:46.28 | shap | de_mon: maybe I have callerid screwed up. Oh. Of course I do. I should be clobbering callerid to our main number only when I dial out onto PSTN. Thanks for the pointer. |
| 15:46.43 | De_Mon | ^_^ |
| 15:47.07 | shap | de_mon: however, this doesn't address the issue of not being able to get the voicemail context for the mailbox |
| 15:47.24 | De_Mon | go read about voicemailmain |
| 15:47.43 | shap | de_mon: I did. In many places. |
| 15:48.15 | De_Mon | how many voicemail contexts are you using? |
| 15:48.46 | shap | de_mon: today, one, and I know how to make that work. I'm trying to test out some stuff for a few weeks from now where I will need more than one. |
| 15:48.59 | De_Mon | and I assume the voicemail context doesn't match the callers context? |
| 15:49.03 | De_Mon | (which seems rather weird to begin with) |
| 15:49.42 | *** join/#asterisk docelmo (n=vircuser@55-65.126-70.tampabay.res.rr.com) |
| 15:49.47 | shap | When contexts start getting included around, it's not always obvious which context you are operating in. I want to name the voicemail contexts after the clients |
| 15:49.57 | De_Mon | you will probably need a different extension for each possible mailbox context if you're not keeping them in predictable locaations |
| 15:50.36 | shap | Yes, but the extension contexts are per-client as well. What's going on is that I may host some people on my PBX, and I want to keep them out of the voicemail for my company. |
| 15:51.04 | shap | Anyway, the following worked: |
| 15:51.30 | De_Mon | so if client a dials *97 asterisk sends them to voicemailmain($calidnum@companya) |
| 15:51.48 | shap | in the sip.conf entry: setvar=VM_MAILBOX=the-mailbox@the-vm-context |
| 15:52.04 | shap | in the extensions.conf: send them to voicemailman(${VM_MAILBOX} |
| 15:52.14 | De_Mon | yeah... but thats just more work |
| 15:52.21 | shap | But yes, using callidnum in that way would probably also work. |
| 15:52.29 | shap | Slightly more work, but *much* more explicit. |
| 15:53.09 | De_Mon | I'd be sending each customer into a separate context, and let asterisk figure out as much as it can |
| 15:53.15 | De_Mon | less things for *you* to break that way |
| 15:53.16 | *** join/#asterisk Ebola (n=Ebola@81-86-155-65.dsl.pipex.com) |
| 15:53.43 | shap | I mean, I'm looking at the sip config, and the extension is already appearing in multiple places. At a minimum, it appears in the mailbox= setting and callerid= setting. What's one more :-) |
| 15:54.16 | shap | But I like the customer context suggestion, and I will probably adopt it. |
| 15:54.18 | De_Mon | now you have to worry about setting this variable correctly (don't want to acidentally assign them to someone elses mailbox!) or, no mailbox at all... |
| 15:54.28 | shap | Oh. And thank you. :-) |
| 15:54.32 | De_Mon | or typos, I hate typos |
| 15:55.05 | De_Mon | best of luck to you, happy asterisking |
| 15:55.10 | shap | I hear you on that |
| 15:55.24 | shap | (that == typos) |
| 15:55.26 | shap | Thanks again |
| 15:57.09 | *** join/#asterisk shodan (n=shodan@ip121.96-113-216.pppoe1.joliette.intermonde.net) |
| 16:03.33 | shodan | if I want to dial *67 before calling , should I dial(zap/1/*67XXXXXXXXXX) ? |
| 16:04.19 | shodan | (*67 is to hide caller id) |
| 16:07.06 | shodan | my problem is that sometime this works , other time I get please press 1 before dialing this number is long distance (using the same local number) |
| 16:08.59 | *** join/#asterisk vgster (n=vgster@cpc2-ledn1-0-0-cust944.leed.cable.ntl.com) |
| 16:11.43 | *** join/#asterisk shap (n=shap@c-68-33-84-43.hsd1.md.comcast.net) |
| 16:12.03 | shap | Stupid question: what happened to the addmailbox script? |
| 16:12.52 | wunderkin | shap, not needed |
| 16:17.48 | *** join/#asterisk Druken (n=jdumais@CPE000854de4ec0-CM00137189cb0c.cpe.net.cable.rogers.com) |
| 16:18.03 | Druken | afternoon everyone..... |
| 16:18.51 | Druken | anyone feel like solving a stupid problem for me? trying to add in app dbodbc into my asterisk source, but the make is bitching about a missing seperator... |
| 16:21.19 | Druken | pfft, never mind... serves me right for cutting and pasting... |
| 16:21.40 | *** join/#asterisk aptura_ (n=none@S010600a0c93f6f7e.vs.shawcable.net) |
| 16:21.53 | shap | wunderkin: asterisk now builds the tree itself? |
| 16:22.11 | wunderkin | ya |
| 16:22.24 | *** join/#asterisk Davey (n=davey@unaffiliated/davey) |
| 16:22.36 | shap | wunderkin: interesting, because I just hit a glitch with that. Probably my own fault with the rpm spec. Updating now and I'll see. |
| 16:28.30 | *** join/#asterisk insomni (n=insomni@x1-6-00-e0-18-6f-34-ad.k455.webspeed.dk) |
| 16:34.01 | *** join/#asterisk [TK]D-Fender (n=joe@64.235.216.2) |
| 16:34.06 | *** join/#asterisk dahunter3 (n=dahunter@pool-71-110-86-116.lsanca.dsl-w.verizon.net) |
| 16:35.12 | shodan | hmm this is really weird , this problem rarely happens (only twice in 3 weeks) when I call an outgoing number this way analog phone -> fxs (linksys spa-2102) -> * -> fxo (x100p clone) -> pstn ,, the fxo dials and I can talk to the other person , but I can still hear ringing , and on the console I still get "Zap/1-1 is ringing" ... |
| 16:36.03 | *** join/#asterisk Tili (n=tili@202.133.67.109) |
| 16:36.09 | shodan | if I call sip phone (polycom 301) -> * -> fxo -> pstn , I don't hear ringing after the far side has answered and I get Zap/1-1 answered .... |
| 16:36.25 | shodan | if I reboot the fxs I'm pretty sure it'll stop doing that |
| 16:37.36 | shodan | how is that even possible ? that depending on the sip client the zap interface can or can't detect that the far side has answered ?! |
| 16:37.56 | *** join/#asterisk Mw3 (i=mw3@national.t-error.hu) |
| 16:39.27 | *** join/#asterisk riddlebox (n=james@24-171-10-102.dhcp.stls.mo.charter.com) |
| 16:39.32 | wunderkin | shodan, on your *67 question, you need to add a few "w" after the *67 |
| 16:39.58 | shodan | wundaboy, thanks , that solves that other problem :) |
| 16:46.58 | shodan | hmm I think I found the cause , sometime the * server doesn't detect ringing , I just made a call with the sip phone with the console opened and this time it kept ringing but the sip phone didn't make the ringing sound |
| 16:47.18 | [TK]D-Fender | shodan : The SPA-2102 is also a router and may be blocking off part of the response because of being used on the wrong port. (wan/lan) |
| 16:47.46 | [TK]D-Fender | shodan : Several of the router-based solutions only work on the WAN port for instance which is why I never recommend them. |
| 16:48.47 | Druken | [TK]D-Fender: the 2102 is good for customers sites usually... |
| 16:49.04 | shodan | I disabled all things rounter (set in bridge mode with the ethernet cable plugged in the wan port) |
| 16:49.52 | [TK]D-Fender | shodan Don't try to be "too smart" with dumb devices :/ |
| 16:50.26 | Druken | [TK]D-Fender: hehe my entire office is run off a wireless bridge :) |
| 16:50.31 | [TK]D-Fender | Druken : Yes, there is a very specific segment that that uniit applies to, but I always suggest a seperate ATA from router solution. |
| 16:54.12 | *** join/#asterisk rowter (n=Silver@201.135.9.97) |
| 16:58.40 | shodan | I would have gotten a 2100 but when I order voipdepot didn't stock them and I didn't have a gentek account yet |
| 16:59.13 | shodan | I think I found out why the fxs is ringing and not the sip phone , it looks like the fxs rings until it receive a answered sip message |
| 16:59.51 | shodan | not sure why but the sip phone even ignores sip ring message after the call is connected |
| 17:00.01 | shodan | even if it didn't receive a sip answered message |
| 17:01.03 | shodan | all I need is to find out why * isn't sending answered messages , maybe the gain is too low |
| 17:11.44 | Druken | shodan: that's kinda the deal with analog FXO.... doesn't do callpres very well... |
| 17:12.50 | shodan | callpres ? |
| 17:12.54 | shodan | ~callpres |
| 17:13.30 | shodan | things like ringing answering callerid etc? |
| 17:14.43 | Druken | my bad, call progress |
| 17:16.49 | shodan | when I dial something on the pstn , when the far side answers , is there a call progress tone for that ? how is * supposed to know that the far side has answered ? |
| 17:17.48 | Druken | i BELIVE it's known by the voltage over the line, but i don't know for sure.... |
| 17:18.30 | file | on what technology? |
| 17:18.34 | shodan | ah ok , so the x100p card has to tell * because * can't tell just by analyzing the audio ? |
| 17:18.41 | shodan | file, pstn |
| 17:18.49 | Druken | x100p |
| 17:19.04 | file | you don't know that it's answered, analog doesn't provide signalling like that |
| 17:20.32 | Druken | shodan: my reccomendation, if you need the analog, use them for incoming only, send out all your calls via voip |
| 17:21.12 | shodan | then , * should know it's answered because it's no longer ringing maybe ? (the problem I'm having is a call like this fxs(spa-2102) -> * -> fxo(x100p) -> pstn , when the far side has answered the fxs is still ringing) |
| 17:21.30 | file | it doesn't know it's ringing |
| 17:21.34 | file | UNLESS |
| 17:21.39 | file | you turn on the highly experimental callprogress option |
| 17:21.46 | file | that may cause weirdness, and calls to be hung up randomly |
| 17:22.07 | Druken | something tells me he has callprogress on.... |
| 17:22.18 | shodan | Druken, can't, I'm on a wisp , latency is too variable , and it would add at least 70ms |
| 17:22.25 | shodan | ... hmm I'll check |
| 17:22.55 | *** join/#asterisk Dibbler_ (n=Dibbler@dsl-217-155-254-174.zen.co.uk) |
| 17:22.58 | shodan | hmm callprogress is on |
| 17:23.05 | Druken | and the FXS on the 2102 is basically a voip phone... |
| 17:24.10 | Druken | if you remove the call progress, when you dial out the FXO, it'll just dial, without anything else, it'll assume it's been answered the second it's done dialing |
| 17:26.07 | shodan | yes , I turned callprogress off , now as soon I as dialout I get Zap/1-1 answered |
| 17:26.18 | shodan | but now my old echo problem is back :( |
| 17:27.13 | Druken | hehe, why not try using the 2102 as an fxo ? |
| 17:27.23 | Druken | if it's got one... |
| 17:28.25 | shodan | nah , it only has 2 fxs |
| 17:28.35 | *** join/#asterisk wunderkin (n=kev@ip68-226-113-228.ph.ph.cox.net) |
| 17:28.38 | shodan | I'm getting one spa-3102 soon |
| 17:29.13 | shodan | hmm if I turn callprogress back on , I get almost no echo , it's still there but it's muffled |
| 17:29.41 | [TK]D-Fender | shodan : the IP 301 is reacting properly. perhaps you should see about upgrading your SPA's firmware.... |
| 17:30.36 | *** join/#asterisk Ox0F0-0FF (n=pierre@200.216.238.226) |
| 17:30.48 | *** join/#asterisk f0urtyfive (i=f0urtyfi@c-67-165-5-232.hsd1.ct.comcast.net) |
| 17:31.48 | shodan | [TK]D-Fender, couldn't hurt I guess |
| 17:34.25 | *** join/#asterisk DrukenHME (n=jdumais@CPE000854de4ec0-CM00137189cb0c.cpe.net.cable.rogers.com) |
| 17:38.51 | De_Mon | handle_response_peerpoke: Peer 'sipura001' is now REACHABLE! |
| 17:38.56 | De_Mon | how do I turn these off? |
| 17:39.16 | De_Mon | debug is 0 verbose is 3 |
| 17:41.49 | DrukenHME | qualify=no ? |
| 17:42.15 | *** join/#asterisk LoneShadow (n=duh@59.92.177.216) |
| 17:43.25 | *** join/#asterisk profounded (n=pro@ool-44c4eae2.dyn.optonline.net) |
| 17:44.40 | *** join/#asterisk lowlevel (n=Stuart@CPE0050ba71c82f-CM000f9f7d6742.cpe.net.cable.rogers.com) |
| 17:45.21 | De_Mon | ah. is that what qualify did... |
| 17:45.48 | De_Mon | what does leaving it on do besides create those warnings? |
| 17:46.49 | *** join/#asterisk clyrrad (n=ddd@CPE001195f553c7-CM0011aea484a4.cpe.net.cable.rogers.com) |
| 17:46.59 | DrukenHME | if the phone isn't REACHABLE it goes directly to n+101, makes it much better for voicemail and such... |
| 17:47.53 | De_Mon | I've been using nat keepalives on the phone to keep the connection up, should I do both or disable nat keepalives maybe? |
| 17:48.32 | DrukenHME | no idea |
| 17:51.03 | De_Mon | cool, thanks. |
| 17:51.03 | *** join/#asterisk celophane (n=e@ip68-104-251-230.ph.ph.cox.net) |
| 17:51.06 | celophane | Hi...need HELP! - Trying to install Linux (tried poundkey, Fedora Core 4, Fedora Core 5, and rPath) and every time it hangs at first screenfull of information at "ACPI: Assume root bridge [\_SB_.PCI0] bus is 0". Installing on Intel board DG965WH. I need to use this board...if these distros can't/won't work...which one will? This machine will be used as a phone server and I've got to get it done today. ANY help would be MUCH appreciated |
| 17:51.18 | clyrrad | Does anyone here using 905 DID's? |
| 17:51.22 | celophane | I've got a SATA drive installed....but I disconnected it and attempted to use an IDE drive thinking maybe it couldn't find the SATA drive....but I got the same results. |
| 17:51.24 | clyrrad | either SIP or IAX? |
| 17:51.31 | celophane | I've basically turned everything off in the BIOS VT Technology (disabled), etc.... |
| 17:53.01 | De_Mon | celophane ask in your prefered distro's channel... |
| 17:53.32 | DrukenHME | clyrrad: tonnes of providers for 905... |
| 17:53.44 | celophane | I don't have a prefered distro. I'm setting up a Asterisk box (which is why I'm in this channel). |
| 17:53.58 | DrukenHME | 905 is a large areacode tho... any particular area? |
| 17:54.44 | celophane | The Asterisk distro of poundkey does the same thing all of the other distros I've tried using (mentioned above...using same kernel I think) do. |
| 17:55.13 | clyrrad | DrukenHME.... at this point I would look local to TO with 905 |
| 17:55.21 | [TK]D-Fender | celophane : I'd suggest CentOS. |
| 17:55.22 | clyrrad | I have not found a provider that offers them wholesale |
| 17:55.32 | clyrrad | they all want them on Plans |
| 17:55.47 | [TK]D-Fender | clyrrad : www.unlimitel.ca |
| 17:55.50 | clyrrad | I want some service like Unlimitel where you can buy the DID's at XXX per month then pay for whatever you use |
| 17:55.55 | celophane | [TK]: Why? |
| 17:55.56 | clyrrad | TKD LOL |
| 17:56.06 | clyrrad | I have lots of accounts with Unlimitel |
| 17:56.11 | clyrrad | but they dont offer 905 DID's |
| 17:56.27 | [TK]D-Fender | clyrrad : They have pay/volum and wholesale plans.. what is it exactly that they lack? |
| 17:56.36 | clyrrad | 905 DID's |
| 17:56.36 | [TK]D-Fender | clyrrad ;They don't? |
| 17:56.41 | clyrrad | NOPE |
| 17:56.49 | clyrrad | they dont offer 905 DID's |
| 17:57.03 | clyrrad | I want a provider that offers like Unlimitel - except one that can offer 905 DID's |
| 17:57.28 | [TK]D-Fender | 416 is better range anyways... whtas the difference? |
| 17:57.45 | clyrrad | the difference is lots of 905 customers will not change to VOIP or loose their 905 DID |
| 17:57.47 | clyrrad | need to keep them |
| 17:59.08 | clyrrad | I am happy with Unlimitel for everything else... just they cant get or port 905 DID's and I need that... |
| 18:00.39 | shodan | :\ it feels so wrong to use a 20$cad GE 900mhz cordless phone on a 90$ fxs ! yet most other phones are not as confortable/simple to use as this phone |
| 18:01.07 | clyrrad | TKD - do you know of a provider that can do this for 905's? |
| 18:01.17 | clyrrad | that will offer wholesale service like Unlimitel does? |
| 18:01.54 | *** join/#asterisk luchshiy (n=anonymou@d212-53-104-193.cust.tele2.ch) |
| 18:05.47 | *** join/#asterisk mohsenbasirat (n=mohsen@213.207.243.195) |
| 18:05.59 | mohsenbasirat | hi |
| 18:06.20 | mohsenbasirat | i use tiger type modem as fxo interface |
| 18:07.06 | mohsenbasirat | device working when i load driver but when running asterisk with proper configuration i get this error |
| 18:10.31 | *** part/#asterisk mohsenbasirat (n=mohsen@213.207.243.195) |
| 18:11.07 | profounded | i got a polycom smartphone 430 which is SIP enabled, is it possible to hook that up to a vonage softphone line without using asterisk? |
| 18:12.39 | [TK]D-Fender | profounded If you have the SIP connection detail, sure, just like any other. |
| 18:13.24 | clyrrad | TKD - did you have an anser about my question? |
| 18:14.55 | [TK]D-Fender | clyrrad : Not offhand... |
| 18:15.05 | clyrrad | ok - thanks bud |
| 18:19.27 | *** join/#asterisk P4C0 (n=ash@200.124.22.34) |
| 18:19.47 | P4C0 | hello, is there anyway to do a call from asterisk console? |
| 18:22.34 | mog | dial |
| 18:22.44 | *** join/#asterisk SomethingISODD (n=dan@h109.42.63.69.cable.ottr.cablerocket.net) |
| 18:22.47 | SomethingISODD | evening all |
| 18:23.12 | SomethingISODD | actually i guess afternoon :-) question can you run more then one asterisk on the same server with different ip addresses? |
| 18:23.53 | clyrrad | why would you want to do that? |
| 18:24.19 | SomethingISODD | something i am working on. can it be done? |
| 18:24.49 | LoneShadow | I wonder what would "asterisk -r" do if you were to run 2 copies on the same machine |
| 18:25.03 | clyrrad | exactly |
| 18:25.04 | SomethingISODD | i am building a Redundancy setup don`t have a second box to test with |
| 18:25.09 | clyrrad | how would it know what one to connect to |
| 18:25.29 | SomethingISODD | clyrrad i was thinking under different accounts |
| 18:25.39 | clyrrad | i dont see how it would work |
| 18:25.39 | SomethingISODD | i was reading Asterisk can run on non root accounts |
| 18:25.39 | LoneShadow | I think asterisk creates some running context under /var/run/asterisk or simillar folder |
| 18:25.52 | LoneShadow | running 2 copies at the same time may not work |
| 18:26.04 | SomethingISODD | LoneShadow good point. |
| 18:26.07 | clyrrad | it wont work as far as I can tell, bease also the configs are in /etc/asterisk |
| 18:26.22 | clyrrad | if you had 2 copieds of the same config files it would not work for obvious reasons |
| 18:26.25 | SomethingISODD | clyrrad oh so asterisk can`t run as a non root? |
| 18:26.31 | clyrrad | I dont think it can work like you are trying |
| 18:26.37 | clyrrad | unless you do it in a Virtual Machine |
| 18:26.44 | clyrrad | that would work |
| 18:26.58 | SomethingISODD | can i run a Virtual Machine on a remote server? |
| 18:27.05 | LoneShadow | hmm, actually I think you maybe able to run 2 copies, might need few modifications |
| 18:27.11 | clyrrad | I dont see why not |
| 18:27.20 | clyrrad | tunnel in with SSH |
| 18:27.25 | LoneShadow | you will need 2 different /etc/asterisk/asterisk.conf files |
| 18:27.38 | clyrrad | he will need 2 coipes of every config file |
| 18:27.54 | clyrrad | a virtual machine is best |
| 18:28.04 | SomethingISODD | clyrrad do u know where i can get info on tunneling, as i said this is just for testing not for production or anything |
| 18:28.04 | clyrrad | it will have its own IP address and seperate memory space |
| 18:28.13 | LoneShadow | how about chroot ? |
| 18:28.18 | clyrrad | Just use SSH |
| 18:28.35 | clyrrad | and for a Virtual Machine use Vmware |
| 18:28.48 | SomethingISODD | clyrrad ok thanks |
| 18:28.58 | SomethingISODD | clyrrad one more question |
| 18:29.10 | SomethingISODD | are you a php programmer by chance? |
| 18:29.17 | clyrrad | yes why? |
| 18:29.36 | SomethingISODD | i am having a small issue with a extensions script i was wondering if you could take a look and recommend a fix |
| 18:29.49 | LoneShadow | agi script ? |
| 18:29.54 | clyrrad | pastebin your script |
| 18:30.01 | clyrrad | and then describe your problem |
| 18:30.13 | LoneShadow | clyrrad: how difficult is to write a simple callback agi script ? |
| 18:30.26 | clyrrad | not hard at all |
| 18:30.35 | clyrrad | then again nothing is hard once you know how to do it :) |
| 18:30.59 | P4C0 | anyone in the mood to help me with nat issues? |
| 18:31.04 | LoneShadow | is it like couple of lines, to callback ? |
| 18:31.09 | SomethingISODD | clyrrad ok moment let me jump in to my server LoneShadow give me a half hour i have a script you are welcome to use. i just need to do something before i have a chance to find it |
| 18:31.50 | LoneShadow | I am trying to simulate a intercomm on my telephones attached to a ATA |
| 18:31.53 | clyrrad | LoneShadow what are you trying to do? |
| 18:32.13 | clyrrad | what do u need the call back to do? |
| 18:32.20 | clyrrad | have you ever written an AGI before? |
| 18:32.21 | LoneShadow | I have linksys/openwrt/asterisk <--> spa3k <--> 2 telephones |
| 18:32.49 | clyrrad | im guessing you want your AGI to interface with the Asterisk CML? |
| 18:32.55 | clyrrad | CLI* |
| 18:33.01 | LoneShadow | trying to dial a special # on the phone, the AGI script should hangup and callback |
| 18:33.05 | LoneShadow | yea thru cli |
| 18:33.08 | LoneShadow | never written one |
| 18:33.11 | clyrrad | in PHP? |
| 18:33.27 | LoneShadow | I am a c programmer, not that good with scripting |
| 18:33.46 | mog | why not just make an app then LoneShadow |
| 18:33.47 | clyrrad | well in c im not sure exactly I have done them in PHP |
| 18:33.55 | clyrrad | there are wrapper classes you can use with PHP |
| 18:33.56 | mog | and there is C agi bindings as well |
| 18:34.02 | mog | no need for php |
| 18:34.07 | LoneShadow | mog: Yea I noticed |
| 18:34.18 | clyrrad | yep if you do it in c you dont need php |
| 18:35.08 | clyrrad | but if you want a fast simple way in PHP you can use this: http://phpagi.sourceforge.net/ |
| 18:35.18 | LoneShadow | so the callback script would hangup, pause for few seconds, dial the sip/spa3k, and wait till the sip/spa3k hangs up ? |
| 18:35.24 | mog | what ar eyou trying to do LoneShadow |
| 18:35.58 | LoneShadow | thats pretty much what I am trying to do |
| 18:36.29 | LoneShadow | if I call from a sip client 200, the script should hangup, callback, and wait till the user picks up and hangs up |
| 18:37.17 | mog | thats easy use the h exten |
| 18:37.22 | mog | you can do this in dialplan |
| 18:37.28 | mog | no need to write anything |
| 18:37.31 | clyrrad | Does asterisk not already have an intercom feature? I am not sure have never looked into it... |
| 18:37.39 | mog | yes |
| 18:37.57 | LoneShadow | thats for specific phones with intercom feature |
| 18:38.20 | LoneShadow | I am trying to create a laymans intercomm |
| 18:38.22 | mog | sip phones support an intercom feature |
| 18:38.32 | mog | but you can also use chan_alsa for over head paging |
| 18:38.39 | LoneShadow | mine are the regular analog phones connected to a ATA |
| 18:39.28 | [TK]D-Fender | LoneShadow You will not be able to use ATA's to page through the phones. |
| 18:39.33 | mog | well you cant force an analog to answer |
| 18:39.42 | mog | but app_page will dial em |
| 18:39.48 | LoneShadow | ? |
| 18:39.48 | mog | and speak when they pick up |
| 18:39.53 | [TK]D-Fender | LoneShadow You CAN add a paging device onto an ATA that will use an overhead speaker however |
| 18:40.11 | [TK]D-Fender | mog : If you have to accept its not much of a "page" |
| 18:40.18 | *** join/#asterisk P4C0 (n=ash@200.124.22.34) |
| 18:40.46 | LoneShadow | we have one phone in the first floor and the other in the second |
| 18:40.47 | mog | well agreed |
| 18:40.54 | mog | but it allows you to push everyone into a meetme |
| 18:40.57 | clyrrad | so app_page is only working on SIP phones that support it? |
| 18:40.58 | mog | with you talking |
| 18:41.04 | LoneShadow | just trying to create a simple intercom system |
| 18:41.05 | mog | its as close to it as you can |
| 18:41.12 | LoneShadow | the agi script should be sufficient right ? |
| 18:41.28 | P4C0 | hello guys I have problem with outgoing calls, my asterisk server is inside a nat, the invite shows no private ip at all, but the reply from header of my provider has my private ip, I try to change that and it also changes so it's not hardcoded in the server side... what can I do? |
| 18:41.46 | LoneShadow | nat=yes ? |
| 18:41.56 | LoneShadow | Fender, will say I assume a lot :P |
| 18:42.18 | [TK]D-Fender | LoneShadow If they have to answer their phones then it isn't proper paging. Get a speaker system installed and use a pgaing device. |
| 18:42.38 | P4C0 | LoneShadow, server or client side? |
| 18:43.09 | LoneShadow | P4C0: arnt both sides behind a NAT ? |
| 18:43.39 | P4C0 | LoneShadow, yes... but I mean is my server provider the one that have to put nat for me? |
| 18:44.04 | LoneShadow | client has to |
| 18:44.21 | LoneShadow | probably on the server side too |
| 18:44.38 | LoneShadow | check voip-info.org |
| 18:46.48 | SomethingISODD | clyrrad this is the script i decribed the print out http://pastebin.ca/173242 |
| 18:47.16 | P4C0 | LoneShadow, http://rafb.net/paste/results/svWHuE44.html this is how it is |
| 18:48.22 | joe | anyone having issues building libsupertone on centos? |
| 18:49.08 | P4C0 | LoneShadow, as you can see, my invite have nothing about my private ip (192.168.6.10) even so my voip provider sent the reply with that ip... i can change the private ip to all and voip provider will keep sending it with the private ip, if I go outside the nat it works fine, the incoming calls works fine to.. so not sure how is it possible for my voip provider to get my private ip... |
| 18:49.18 | joe | or rather has anyone built libsupertone on centos? |
| 18:50.33 | SomethingISODD | clyrrad are you there? |
| 18:50.51 | DrukenHME | anyone got app_dbodbc working? |
| 18:52.42 | LoneShadow | P4C0: do you have externalip defined in sip.conf ? |
| 18:54.32 | SomethingISODD | any other php people in here that could help me resolve a issue with creating an extensions.conf flat file from a database? |
| 18:54.36 | clyrrad | yes im here |
| 18:54.40 | SomethingISODD | oh ok |
| 18:54.43 | clyrrad | what seems to be the problem? |
| 18:55.10 | SomethingISODD | clyrrad it will not print out the full lineup under each context before moving to the next context |
| 18:55.39 | clyrrad | what do you mean? |
| 18:55.53 | clyrrad | you mean $row['priority']."," prints nothing? |
| 18:55.57 | SomethingISODD | let me show you what the out put should have looked like |
| 18:55.58 | clyrrad | like its blank? |
| 18:56.21 | *** join/#asterisk anthm (n=anthmct@CPE-69-76-83-52.wi.res.rr.com) |
| 18:56.21 | *** mode/#asterisk [+o anthm] by ChanServ |
| 18:56.59 | SomethingISODD | clyrrad http://pastebin.ca/173255 |
| 18:57.35 | clyrrad | the top part of that is what you are trying to create obvously right? |
| 18:57.54 | P4C0 | LoneShadow, yes |
| 18:58.19 | SomethingISODD | clyrrad basically the issue is i can`t get it to finish off under the context before moving on, |
| 18:58.21 | P4C0 | LoneShadow, as you can see, the invite have no private ip... http://rafb.net/paste/results/svWHuE44.html |
| 18:58.27 | clyrrad | and your first pastebin is the results you are actually getting? |
| 18:58.32 | SomethingISODD | yes |
| 18:58.45 | clyrrad | echo this $row['priority'] |
| 18:58.49 | clyrrad | see if it has a value or not |
| 18:58.58 | SomethingISODD | it does already checked |
| 18:59.26 | LoneShadow | P4C0: that looks like the work from your router |
| 18:59.33 | clyrrad | what do you get if youe echo $ext? |
| 19:00.03 | SomethingISODD | everything comes up i believe its a issue with the way i am trying to collect the info with the loop in side of a loop |
| 19:00.21 | clyrrad | is your file being created? |
| 19:00.29 | SomethingISODD | ya |
| 19:00.34 | P4C0 | LoneShadow, humm, you mean that my router is modifing the sip data?? |
| 19:00.50 | clyrrad | also i see only one loop |
| 19:00.51 | SomethingISODD | and the first paste shows you what is actually being collected in the file |
| 19:01.09 | LoneShadow | not sip data |
| 19:01.10 | clyrrad | ah nm |
| 19:01.15 | SomethingISODD | :-) |
| 19:01.18 | clyrrad | i see the outer while |
| 19:01.26 | clyrrad | you need to indent your code better :p |
| 19:01.56 | SomethingISODD | ya i guess i do :-) |
| 19:02.01 | clyrrad | also take out your db connect stuff from the inner loop |
| 19:02.09 | clyrrad | that is terrible for performance |
| 19:02.14 | SomethingISODD | k |
| 19:02.20 | P4C0 | LoneShadow, what else then? the ip?? |
| 19:03.02 | clyrrad | also you need only one $connect - you dont need to do that twice |
| 19:03.26 | *** join/#asterisk RoyK (n=roy@ti211310a080-11263.bb.online.no) |
| 19:03.31 | P4C0 | LoneShadow, but whatisyourip.com works for me... |
| 19:03.31 | SomethingISODD | k |
| 19:03.59 | SomethingISODD | clyrrad do you think this would have any barring on why its not printing out alll data |
| 19:04.09 | clyrrad | yes its possible |
| 19:04.11 | *** join/#asterisk Ebola (n=Ebola@81-86-155-65.dsl.pipex.com) |
| 19:04.38 | clyrrad | re-work you code how i told you - then indent properly and pastebin again - this is aweful to look at how you have it now |
| 19:05.03 | clyrrad | gonna make a sandwich |
| 19:05.24 | *** join/#asterisk amdtech (n=amd011@ip70-179-175-131.dl.dl.cox.net) |
| 19:06.06 | SomethingISODD | k |
| 19:09.19 | P4C0 | LoneShadow, what do you exactly mean by the work of my router?? |
| 19:12.05 | SomethingISODD | clyrrad this is about as clean loop as i can do http://pastebin.ca/173269 |
| 19:13.21 | SomethingISODD | loop=look |
| 19:13.24 | *** join/#asterisk |Vulture| (n=_Vulture@101.222.121.70.cfl.res.rr.com) |
| 19:14.08 | |Vulture| | Anyone have a recommendation for VoIP LD service... needs to be reliable with a decent number of outbound channels... inbound not needed |
| 19:14.33 | *** join/#asterisk riddlebox (n=james@24-171-10-102.dhcp.stls.mo.charter.com) |
| 19:14.53 | aptura_ | something in my dialplan is not excepting my DID. Cli says -- Executing Answer("SIP/username-b1f9", "") in new stack |
| 19:14.53 | aptura_ | <PROTECTED> |
| 19:14.53 | aptura_ | <PROTECTED> |
| 19:15.24 | aptura_ | I substituted username and did number for security reasons. |
| 19:15.39 | |Vulture| | aptura_: pastebin the whole section of debug |
| 19:15.48 | aptura_ | btw |
| 19:15.54 | aptura_ | where is the debug log |
| 19:16.12 | P4C0 | brb |
| 19:16.12 | *** join/#asterisk Davey (n=davey@unaffiliated/davey) |
| 19:16.13 | clyrrad | SomethingISODD so the file is only writting one context right? |
| 19:16.14 | |Vulture| | just that output on your screen |
| 19:16.28 | aptura_ | it moves to quick but will try |
| 19:16.35 | SomethingISODD | clyrrad this should help better http://pastebin.ca/173271 |
| 19:16.41 | SomethingISODD | help explain i mean |
| 19:17.46 | clyrrad | echo $ext and paste me EXACTLY what output you see |
| 19:18.32 | SomethingISODD | second |
| 19:18.51 | aptura_ | is there a way to filter sip debug |
| 19:19.19 | De_Mon | filter? pipe it to a file and filter that |
| 19:19.26 | SomethingISODD | clyrrad want me to pastbin it |
| 19:20.07 | |Vulture| | dont use sip debug unless it is a really difficult issue |
| 19:20.17 | *** join/#asterisk profounded (n=pro@ool-44c4eae2.dyn.optonline.net) |
| 19:20.27 | clyrrad | yes |
| 19:20.32 | SomethingISODD | clyrrad http://pastebin.ca/173273 |
| 19:20.49 | aptura_ | |Vulture| well thats what you asked me to do :) |
| 19:21.44 | clyrrad | looks like its not gouping them properly |
| 19:21.51 | |Vulture| | aptura_: I just meant the standard console display |
| 19:22.00 | clyrrad | you probably need to add GROUP BY context |
| 19:22.01 | SomethingISODD | should i do group by context? |
| 19:22.03 | SomethingISODD | k |
| 19:22.08 | clyrrad | also |
| 19:22.17 | clyrrad | for a new link i think its \r\n not just \n |
| 19:22.28 | SomethingISODD | \n works perfect |
| 19:22.39 | clyrrad | not in ur pastebin it didint |
| 19:23.27 | aptura_ | |Vulture| thats what you saw threee lines |
| 19:23.55 | aptura_ | -- Executing Answer("SIP/username-b1f9", "") in new stack |
| 19:23.55 | aptura_ | <PROTECTED> |
| 19:23.55 | aptura_ | <PROTECTED> |
| 19:23.57 | SomethingISODD | clyrrad good point btw when i do a group by context, it removed the stuff under [default] |
| 19:24.31 | clyrrad | so add a WHERE clause to excluded when context = default |
| 19:24.45 | clyrrad | or |
| 19:24.50 | clyrrad | add an IF in your code |
| 19:25.18 | RoyK | :%s/asterisk/openpbx/g |
| 19:25.44 | aptura_ | perhaps I should change the verbosity |
| 19:25.47 | SomethingISODD | clyrrad already have a where clause select context,exten,priority,app,appdata from exten where context='$cons' group by context |
| 19:26.17 | *** join/#asterisk pdt (n=pdthome@c-68-53-40-50.hsd1.tn.comcast.net) |
| 19:26.54 | clyrrad | why not have your loop create an array |
| 19:27.04 | clyrrad | then you can format your array in the proper order / sequence |
| 19:27.09 | clyrrad | then dump the array to file |
| 19:27.11 | SomethingISODD | clyrrad i was thinking of that but i am not good at array`s |
| 19:27.26 | clyrrad | becase right now you are stuck with whatever order the DB reads and the current itteration of the loop |
| 19:27.29 | SomethingISODD | clyrrad could you show me an example on how to do it with arrays |
| 19:27.32 | clyrrad | arrays are simple |
| 19:27.58 | clyrrad | http://www.php.net/types.array |
| 19:28.42 | clyrrad | make an outer array called Contexts_Array, then nest inside that another array called Context_Data |
| 19:28.56 | clyrrad | that way each Context_Array element will have an array of Context_Data |
| 19:29.16 | clyrrad | you can then ittereate those arrays, dumping to file the [default] context first |
| 19:29.35 | clyrrad | then all the remaining contexts - this should preserve the order and intergrity of the file so asterisk can parse it and use it |
| 19:31.34 | SomethingISODD | ok i will play with this and see if i can figure out a way of doing this with the arrays thanks for the recommendations |
| 19:32.15 | clyrrad | welcome - and this problem you are having is an ideal candiate for arrays - if you use arrays you will solve this problem rater easily |
| 19:33.00 | SomethingISODD | clyrrad ya i was thinking that myself but i don`t have any clues about Arrays but i guess i am about to learn |
| 19:33.11 | clyrrad | they are not hard |
| 19:33.16 | clyrrad | that page i sent you shoudl get you started |
| 19:33.20 | clyrrad | it has some good examples |
| 19:33.36 | clyrrad | and php.net is a great site full of useful info and examples |
| 19:40.12 | Lyfe | anyon eknow why every now and then I see " -- B-channel 0/22 successfully restarted on span 1" (where it goes from 0/1-0/23) on the asterisk console for my PRI? |
| 19:41.09 | Juggie | for the very reason it says it |
| 19:41.15 | Juggie | just restarting the b-channel |
| 19:41.38 | *** join/#asterisk Ox0F0-0FF (n=pierre@200.216.238.226) |
| 19:41.44 | pdt | it keeps your pri so fresh and so clean clean ;) |
| 19:41.45 | Lyfe | well, i'm worried that it might reset a b-channel that's in use, really.. so i guess that's my real inquiry about it. |
| 19:41.56 | Juggie | no |
| 19:42.05 | Juggie | it doesnt restart b-channels that are in use |
| 19:42.17 | *** join/#asterisk ToTo (n=ToTo@host149-109-dynamic.58-82-r.retail.telecomitalia.it) |
| 19:42.19 | Lyfe | so, it's a normal thing then.. ok. |
| 19:42.25 | Juggie | yes |
| 19:42.28 | pdt | if you watch it for awhile you will see it skip the in use channels |
| 19:43.06 | Lyfe | gotcha. |
| 19:43.12 | *** join/#asterisk pifiu (n=someone@216.5.79.1) |
| 19:43.29 | Lyfe | it doesn't happen often enough, nor currently conflict with in-use calls enough that i could notice that easily. |
| 19:43.58 | *** join/#asterisk |Vulture| (n=_Vulture@101.222.121.70.cfl.res.rr.com) |
| 19:44.26 | pifiu | anyone know where i can find the changelog of polycom ip501 sip from 1.6.3.0067 to 1.6.6? |
| 19:46.47 | *** join/#asterisk Badenser (n=user@104-15-124-83.dsl.3u.net) |
| 19:47.10 | Lyfe | pifiu: first hit on a google search for "polycom changelog 1.6.6" got me a pdf with the info you might be after. |
| 19:47.55 | Badenser | Hi all, today I faced the following problem " Unable to find a codec translation path from ilbc to ulaw" ... never had this before ... was there a change to Asterisk 1.2.10? |
| 19:48.51 | Badenser | Sep 16 21:08:36 WARNING[16521]: channel.c:2706 ast_channel_make_compatible: No path to translate from IAX2/daniel-2(4) to SIP/daniel-gmx-081a00b0(1024) |
| 19:48.52 | Badenser | Sep 16 21:08:36 WARNING[16521]: app_dial.c:1595 dial_exec_full: Had to drop call because I couldn't make IAX2/daniel-2 compatible with SIP/daniel-gmx-081a00b0 |
| 19:48.56 | Badenser | :-( |
| 19:50.45 | Badenser | oh ... "show translation" prints only "-" at ilbc ... was there a major change at this codec??? |
| 19:51.05 | pifiu | lyfe damn you were feeling lucky! |
| 19:51.06 | pifiu | lol |
| 19:52.11 | Lyfe | hehe. |
| 19:52.24 | |Vulture| | Anyone have a VoIP that they have had good experiences with for outbound termination? |
| 19:54.10 | aptura_ | hi vulture yea..."Good experaince" |
| 19:54.12 | aptura_ | :) |
| 19:55.16 | |Vulture| | well VoicePulse is just sitting on my new account |
| 19:55.23 | *** part/#asterisk andresmujica (n=andresmu@201.245.230.87) |
| 19:55.36 | |Vulture| | cant figure out my PW to my year and a half old Asterlink account |
| 20:00.30 | *** part/#asterisk Davey (n=davey@unaffiliated/davey) |
| 20:01.53 | *** join/#asterisk Crashsys (i=Kumba@1-211.124-70.tampabay.res.rr.com) |
| 20:09.57 | Badenser | *arrggg* ok ... found it ... last update removed ilbc because its non-free |
| 20:10.41 | |Vulture| | Badenser: huh? |
| 20:11.00 | |Vulture| | ilibc was removed from *? |
| 20:11.00 | *** join/#asterisk Greek-Boy (n=Greek-Bo@196.46.109.224) |
| 20:11.13 | Badenser | it was removed from the Debian package |
| 20:11.33 | |Vulture| | ah strange |
| 20:11.58 | |Vulture| | I use ilibc between all of my locations... hope they dont plan on doing that to the source releases |
| 20:15.50 | *** join/#asterisk Ox0F0-0FF (n=pierre@200.216.238.226) |
| 20:17.10 | Strom_C | what, you're so bandwidth-stingy that you can't use somethng higher-quality like g726? |
| 20:17.35 | |Vulture| | of course |
| 20:17.46 | |Vulture| | plus we have had good experiences with ilibc |
| 20:18.07 | |Vulture| | we have 768k frac-t1 connections and I prefer to keep the badwidth down |
| 20:18.25 | Strom_C | how many concurrent calls do you handle at peak? |
| 20:18.50 | |Vulture| | voip? |
| 20:19.05 | Strom_C | over any given fractional t1 |
| 20:19.11 | |Vulture| | prolly 2-3 per location |
| 20:19.24 | Strom_C | oh come on man, you can totally afford g726 |
| 20:19.27 | |Vulture| | hahaha |
| 20:19.38 | Strom_C | your calls will sound WORLDS better |
| 20:19.52 | |Vulture| | they also have PRI for local calling thats why they are so low |
| 20:19.58 | |Vulture| | yea yea... Ill give it another shot |
| 20:20.05 | |Vulture| | its been almost 2 years since I played with codec |
| 20:20.06 | |Vulture| | s |
| 20:20.16 | SomethingISODD | clyrrad you still around? |
| 20:20.24 | Strom_C | yeah, the basic codec thing works like this: |
| 20:20.41 | Strom_C | g711 and g726 sound exactly like and almost exactly like real TDM telephone calls, respectively |
| 20:20.54 | Strom_C | everything else sounds like varying grades of abysmal |
| 20:21.00 | Strom_C | (except for slin, of course) |
| 20:22.17 | |Vulture| | now I am just trying to get a good VoIP provider for our LD |
| 20:22.51 | Strom_C | if you're doing interoffice trunking over known high-bandwidth pipes, you shouldn't be considering anything less than g726 unless you're being uber-stingy and somehow need to cram a hundred calls down a single T1 |
| 20:22.59 | *** join/#asterisk danalien (n=danalien@unaffiliated/danalien) |
| 20:23.52 | Strom_C | if you have, for example, traveling voip phones out in the field that will be connecting over mystery DSL, then low-bitrate is a good choice from a compatibility standpoint |
| 20:24.17 | |Vulture| | Strom_C: I have 729 for that |
| 20:24.28 | Strom_C | well, good |
| 20:24.40 | |Vulture| | yea works decent with the polycom phones |
| 20:24.50 | Strom_C | wait, polycom phones? are these traveling? |
| 20:25.01 | Strom_C | or are they sitting on known connections permanently |
| 20:25.06 | |Vulture| | known connections |
| 20:25.16 | Strom_C | and how fat are those pipes? |
| 20:25.17 | |Vulture| | we dont support traveling phones |
| 20:25.21 | |Vulture| | cable/DSL |
| 20:25.38 | Strom_C | but how fat are those pipes? |
| 20:25.51 | Strom_C | there's like sixty varieties of "cable/dsl" |
| 20:26.06 | |Vulture| | 40kbytes up and 800 down |
| 20:26.35 | Strom_C | what are they rated in kilobits, man...kilobytes mean NOTHING in the telecom world |
| 20:26.38 | SomethingISODD | does anyone here know php? |
| 20:27.03 | Strom_C | 40 kilobytes sounds vaguely like 512kbps |
| 20:27.04 | SomethingISODD | and would mind giving me a hand with a issue i can not figure out |
| 20:27.21 | |Vulture| | yea ~350kbps |
| 20:27.46 | Strom_C | ok, so you can totally afford to not have your remote employees on something that sounds like ass through a cheese grater. |
| 20:27.48 | |Vulture| | but being non-dedicated is a real strain |
| 20:27.59 | |Vulture| | :P |
| 20:29.21 | |Vulture| | I have no problem with the quality |
| 20:29.22 | Strom_C | because g729 and g711 sound worlds different to me |
| 20:29.30 | Strom_C | yeah, and thats your problem |
| 20:30.03 | Strom_C | low-bitrate codecs are there as a last resort for when regular-bandwidth calls just will not work under any circumstances whatsoever |
| 20:30.33 | |Vulture| | well Ill give 726 another shot |
| 20:31.06 | *** join/#asterisk radovoip (n=radovoip@xd141.sstar.com) |
| 20:31.37 | Strom_C | if you're doing intreroffice iax trunking, set trunk=yes in iax.conf and save yourself some IP overhead :) |
| 20:32.37 | clyrrad | anyone here know of a good provider for 905 IAX DID's? |
| 20:34.13 | radovoip | freepbx and asterisk for that matter...anyone do it in X? |
| 20:34.30 | *** join/#asterisk isaacs (n=isaac@201.103.46.67) |
| 20:34.32 | Greek-Boy | anyone know a of a good provider of IAX DID termination. I have my own 800 number that i want to use |
| 20:34.43 | *** join/#asterisk Jedirl (n=fdsa783f@154.Red-217-127-168.staticIP.rima-tde.net) |
| 20:36.12 | SomethingISODD | question if you have more then one context the same name it will not work in extensions conf will it |
| 20:36.41 | *** join/#asterisk ibob63 (n=hp@dsl-217-155-69-86.zen.co.uk) |
| 20:36.53 | Strom_C | SomethingISODD: why would you do that in the first place |
| 20:37.25 | isaacs | Is searching for asterisk experts to outsource a Call Center setup installation if someone is interested please contact me |
| 20:37.30 | SomethingISODD | Strom_C because i can`t figure out how to get everything under different contexts using pulling from mysql using php |
| 20:38.20 | Strom_C | isaacs: you'll have better luck posting a detailed spec requirement to the asterisk-biz list |
| 20:39.09 | SomethingISODD | Strom_C this is the way its currently coming out http://216.110.92.250/config.php?config=exten |
| 20:39.48 | Strom_C | SomethingISODD: yeah, that wont work |
| 20:40.03 | SomethingISODD | didn`t really think so |
| 20:40.04 | Greek-Boy | i need some cheap reliable wholesale IAX providers? any recommended? |
| 20:40.24 | SomethingISODD | Greek-Boy i heard class1 is |
| 20:40.35 | SomethingISODD | Strom_C you any good with php?? |
| 20:40.46 | Strom_C | never touched a line of php in my life |
| 20:40.58 | SomethingISODD | Strom_C ok thanks anyway |
| 20:41.06 | clyrrad | SomethingISODD - I already told you how to solve this problem |
| 20:41.37 | SomethingISODD | clyrrad yes but its not solving it |
| 20:41.50 | clyrrad | that means your not doing it properly then |
| 20:42.02 | clyrrad | you have not coded your arrays how I told you |
| 20:42.56 | SomethingISODD | clyrrad :-/ more then likely |
| 20:43.57 | clyrrad | paste your database table data in pastebin |
| 20:44.03 | clyrrad | I want to see what it looks like |
| 20:44.06 | *** join/#asterisk Adrian__ (n=foo@zux221-091-123.adsl.green.ch) |
| 20:44.18 | clyrrad | if its personal data PM me the link to the pastebin |
| 20:44.45 | SomethingISODD | clyrrad i don`t want to paste it lol it looks like crap and i would more then likly be told off for not doing it correctly |
| 20:44.59 | clyrrad | your database or your php? |
| 20:45.06 | clyrrad | or both LOL |
| 20:45.26 | SomethingISODD | na database is fine its this "stupid" array |
| 20:45.36 | clyrrad | I want to see the database |
| 20:45.41 | clyrrad | i want to see how its structured |
| 20:45.53 | aptura_ | When a incomming DID calls in on sip and you get a responce like this at cli - Spawn extension (inbound, 6044840607, 2) exited non-zero does that mean that the dial plan puked at the inbound context line 2? |
| 20:46.07 | SomethingISODD | oh moment |
| 20:47.38 | *** join/#asterisk Assid (i=assid@203.115.83.215) |
| 20:48.36 | SomethingISODD | clyrrad i removed some data but it gives the idea |
| 20:48.37 | SomethingISODD | http://pastebin.ca/173341 |
| 20:49.26 | clyrrad | yep so sould be pretty straigt forward then |
| 20:49.29 | clyrrad | 2 arrays |
| 20:49.36 | clyrrad | 1 of which is nested |
| 20:50.10 | SomethingISODD | clyrrad yup "should be" |
| 20:50.37 | clyrrad | are you looking for someone to code this for you? |
| 20:51.24 | isaacs | anyone here has experience with CRM integration for call centers? \ |
| 20:51.31 | SomethingISODD | what would you charge just for that part |
| 20:52.32 | ibob63 | I am having trouble passing call backways and forward between two servers. I can get things to go one way, but not the other. |
| 20:53.26 | ibob63 | Do I need both servers to register with each other? |
| 20:55.20 | ibob63 | I have put a shortened version of my extensions.conf and iaxs.conf here: http://www.pastebin.co.uk/1216 |
| 20:58.24 | *** join/#asterisk ibob63 (n=hp@dsl-217-155-69-86.zen.co.uk) |
| 20:58.41 | ibob63 | evening all |
| 20:59.18 | ibob63 | I think my computer crashed and none of the posted- I just put up appeared :( |
| 20:59.46 | aptura_ | found out what part of my problem is. commented out some parts of the dialplan for this did and changed the priorities. phone now rings just does not go to voicemail no biggy. |
| 21:03.12 | |Vulture| | Strom_C: you still around? |
| 21:03.16 | Strom_C | yes |
| 21:03.34 | |Vulture| | Im getting errors on g726... ever seen "WARNING[18063]: channel.c:2328 set_format: Unable to find a codec translation path from g726 to slin" |
| 21:03.54 | Strom_C | g726 on both ends? |
| 21:04.23 | |Vulture| | on the other end I get "WARNING[5620]: channel.c:2328 set_format: Unable to find a codec translation path from g726 to gsm" "WARNING[5620]: file.c:824 ast_streamfile: Unable to open transfer (format g726): No such file or directory" |
| 21:04.36 | |Vulture| | duno why it would bring it to gsm |
| 21:04.47 | file | <PROTECTED> |
| 21:04.59 | |Vulture| | its in the list of show codecs |
| 21:05.12 | file | that doesn't answer the question :) |
| 21:05.16 | |Vulture| | ... lol 1s |
| 21:05.18 | file | type load codec_g726.so |
| 21:05.50 | Strom_C | right, I'm off to the record store. |
| 21:06.07 | |Vulture| | that was it... |
| 21:06.15 | |Vulture| | file: Ill smack myself later |
| 21:06.39 | |Vulture| | yea it was a noload |
| 21:12.45 | *** join/#asterisk teknoprep (n=teknopre@unaffiliated/teknoprep) |
| 21:12.48 | *** part/#asterisk teknoprep (n=teknopre@unaffiliated/teknoprep) |
| 21:14.24 | *** join/#asterisk Gunnar (n=gunnar@101.Red-80-24-171.staticIP.rima-tde.net) |
| 21:14.24 | *** join/#asterisk teknoprep (n=teknopre@unaffiliated/teknoprep) |
| 21:14.35 | teknoprep | hmm.. i am having a problem with g729 |
| 21:14.46 | teknoprep | a remote client is nat'd through his nat.. connected to my asterisk box |
| 21:14.51 | *** join/#asterisk flashnet (i=flashnet@kbhn-vbrg-sr0-vl212-213-185-15-166.perspektivbredband.net) |
| 21:14.52 | teknoprep | my asterisk box is not nat'd |
| 21:15.08 | teknoprep | with G729 and only G729 he can have audio out.. but can hear no audio |
| 21:15.14 | teknoprep | with ulaw it works fine |
| 21:16.57 | teknoprep | on internal calls tho |
| 21:17.09 | teknoprep | say i use my extension i can call his extension directly |
| 21:17.17 | teknoprep | it works |
| 21:17.29 | teknoprep | on outside incomming calls it does not work |
| 21:18.31 | *** join/#asterisk dahunter3 (n=dahunter@16.sub-75-214-91.myvzw.com) |
| 21:19.28 | *** join/#asterisk bkw_ (n=bkw_@asterisk/friend-and-developer/bkw) |
| 21:19.30 | bkw_ | HOLY SHIT |
| 21:19.32 | bkw_ | lilo died |
| 21:20.17 | teknoprep | ? |
| 21:21.20 | bkw_ | [Global Notice] On the 12th September Rob Levin, known to many as Freenode's lilo, was hit by a car while riding his bike. He suffered head injuries and passed away in hospital on the 16th. For more information please visit #freenode-announce |
| 21:21.26 | *** part/#asterisk bkw_ (n=bkw_@asterisk/friend-and-developer/bkw) |
| 21:21.45 | Corydon76-home | lilo was Freenode's most visible and chief admistrator |
| 21:22.24 | *** join/#asterisk xnon (i=xnon@200.82.222.123) |
| 21:24.57 | Juggie | thats crazy stuff |
| 21:25.55 | tzafrir_laptop | don't bother joining that channel. THere's just junk traffic there now. |
| 21:26.01 | tzafrir_laptop | no useful information |
| 21:26.52 | *** join/#asterisk maciq (i=macros@2001:1528:122:0:213:d3ff:fee9:5fbc) |
| 21:28.16 | tzafrir_laptop | well, most previous global notices on this server were from lilo (his nick was unrelated to the LInux LOader, btw) |
| 21:29.39 | teknoprep | hmm |
| 21:30.27 | *** join/#asterisk PakiPenguin (n=uppal@linuxpakistan/admin/pakipenguin) |
| 21:30.30 | PakiPenguin | hello everyone |
| 21:30.34 | teknoprep | hi |
| 21:31.54 | tzafrir_laptop | http://en.wikipedia.org/wiki/Rob_Levin |
| 21:32.21 | macTijn | hmm |
| 21:32.31 | macTijn | that's the end of freenode then... |
| 21:33.41 | RaYmAn-Bx | why would it be the end of freenode? |
| 21:33.52 | ruskie | doubtfull |
| 21:33.57 | ruskie | he set the gears in motion |
| 21:34.03 | ruskie | and the staff is still around... |
| 21:42.28 | teknoprep | i don't know why everyone hates grandstream |
| 21:42.36 | teknoprep | i am using there ATA"s with 0 problem on ilbc codec |
| 21:42.42 | teknoprep | and there phones work great also |
| 21:42.50 | teknoprep | sipura ata's give me alot of problems |
| 21:46.12 | *** join/#asterisk Ebola (n=Ebola@81-86-155-65.dsl.pipex.com) |
| 21:46.20 | Ebola | . |
| 21:46.28 | Ebola | whoops |
| 21:46.33 | Crashsys | ... |
| 21:46.39 | Crashsys | Grandstream phones work great? |
| 21:46.55 | Crashsys | I had to use the Beta bios to eliminate echo... |
| 21:46.59 | Crashsys | that's not exactly great... |
| 21:47.07 | file | there is no echo... you merely think there is echo |
| 21:47.11 | Ebola | Yeah so xchat sucks for suddenly popping up 10 tabs when you're typing. |
| 21:49.25 | *** part/#asterisk Crashsys (i=Kumba@1-211.124-70.tampabay.res.rr.com) |
| 21:49.29 | *** join/#asterisk Crashsys (i=Kumba@1-211.124-70.tampabay.res.rr.com) |
| 21:59.33 | tzafrir_laptop | I see that this is a slow time here, so an OT Q: |
| 22:00.24 | tzafrir_laptop | who maintained the original rpmfind.net site (at the w3c server)? |
| 22:01.49 | *** join/#asterisk f0urtyfive (i=f0urtyfi@c-67-165-5-232.hsd1.ct.comcast.net) |
| 22:05.44 | *** join/#asterisk Ox0F0-0FF (n=pierre@200.216.238.226) |
| 22:09.06 | pifiu | this is so weird, my damn polycoms wont download their cfg files that are meant for them, they will only use the default phon1.cfg |
| 22:13.15 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 22:14.13 | *** join/#asterisk radovoip (n=radovoip@xd141.sstar.com) |
| 22:17.49 | radovoip | http://www.aussievoip.com/wiki/index.php?page=freePBX-Centos |
| 22:17.56 | *** join/#asterisk linlin (i=linlin@c-67-173-49-55.hsd1.il.comcast.net) |
| 22:17.58 | radovoip | I did a quick read of this doc |
| 22:18.22 | radovoip | and right on top he talks of disabling the iptables |
| 22:18.49 | radovoip | am I missing something or they actually run the system w/out a firewall? |
| 22:20.40 | radovoip | do you people run a firewall in front of your asterisk box? |
| 22:21.05 | radovoip | I am a gringo at this and trying to figure it out is all |
| 22:21.50 | *** join/#asterisk kusznir (n=kusznir@66-233-98-93.lew.clearwire-dns.net) |
| 22:22.56 | kusznir | Hi all: What could cause my iax interfaces to not see any iax hosts? (IE: I have some configured users and peers. When I try and place a call through a peer, it fails with "no route to channel"; iax2 show peers shows (Unspecified) for host when it normally shows an IP. |
| 22:28.46 | kusznir | The exact error is: dial_exec_full: Unable to create channel of type 'IAX2' (cause 3 - No route to destination) |
| 22:29.04 | *** join/#asterisk QMario_ (n=QMario@unaffiliated/QMario) |
| 22:31.33 | clyrrad | did you register => the DID in iax.conf? |
| 22:33.26 | *** part/#asterisk seebs (n=seebs@216-243-131-210.static.iphouse.net) |
| 22:33.35 | tzafrir_laptop | radovoip, "quick and dirty"? |
| 22:39.13 | *** join/#asterisk adorah (n=admin@87.68.168.3.cable.012.net.il) |
| 22:39.33 | pifiu | can anyone help me? my polycom config files work fine when i set the phone to boot from tftp, but when i switch it to http it doesnt work |
| 22:41.17 | *** join/#asterisk docelmo (n=vircuser@55-65.126-70.tampabay.res.rr.com) |
| 22:41.32 | *** join/#asterisk docelmo (n=vircuser@55-65.126-70.tampabay.res.rr.com) |
| 22:42.10 | *** join/#asterisk Defraz (n=t0tal@fw.centrisys.com) |
| 22:52.02 | *** join/#asterisk xAD (n=xAD@host144-199.pool8290.interbusiness.it) |
| 22:53.27 | radovoip | tzafrir_laptop, quick and dirty???? |
| 22:54.17 | tzafrir_laptop | This is the reason to disable firewall |
| 22:54.54 | r0d3nt|m | ... |
| 22:55.48 | radovoip | tzafrir_laptop, are you saying that it's considered standard practice to run asterisk w/out a firewall? |
| 22:56.18 | tzafrir_laptop | if the setup is quick&dirty: yes |
| 22:57.59 | radovoip | I am trying to find out some info please...I do not mind turning the iptables off for a very short time but no way could I consider running anything hot |
| 22:58.06 | radovoip | for very long time |
| 22:58.26 | radovoip | and I don't think I am out the norm w/this |
| 22:59.46 | radovoip | the author never comes back to say ok...let's set up the iptables or anything leading the reader to believe this is ok |
| 23:01.12 | radovoip | does anyone run asterisk inside X windows? |
| 23:01.35 | radovoip | I love to run X windows |
| 23:02.33 | radovoip | oh well thx anyway |
| 23:02.36 | maciq | erm |
| 23:03.04 | maciq | what do you mean by 'run astersik inside X windows' ? |
| 23:04.35 | *** join/#asterisk robin_sz (n=robin@adsl.redpoint.org.uk) |
| 23:05.00 | robin_sz | so ... what do people recommend for cordless voip phones? |
| 23:05.09 | *** join/#asterisk justinu|laptop (n=Justin@dsl081-083-093.lax1.dsl.speakeasy.net) |
| 23:05.12 | robin_sz | dect base plus an ata? |
| 23:05.22 | Crashsys | ata? |
| 23:06.19 | robin_sz | seems like a workable solution |
| 23:07.07 | Crashsys | Aastra is the only cordless sip phone i've seen |
| 23:07.49 | robin_sz | I have a zyxel prestige wireless sipp phone ... |
| 23:07.52 | robin_sz | but its shit. |
| 23:08.21 | robin_sz | is the aastra any good? |
| 23:08.30 | Crashsys | dunno, never coughed up $500 for it |
| 23:08.40 | robin_sz | coo. |
| 23:08.56 | robin_sz | dect base and an ata would be WAY cheaper :) |
| 23:09.27 | Crashsys | Yup... what i'm planning to do for a place that wants cordless... |
| 23:09.46 | Crashsys | just tie it to their base phone's extension context... |
| 23:13.00 | robin_sz | there is some Dutch company that does a DECT base with an ethernet port .. suports 6 handsets, each as a SIP or h323 channel |
| 23:13.35 | robin_sz | tried one once, but it was before they had SIP support, and I failed to understand h323 |
| 23:14.22 | *** join/#asterisk talljon84 (n=jonathan@pool-72-69-88-151.chi01.dsl-w.verizon.net) |
| 23:14.46 | talljon84 | what does the format_mp3.so module provide to linux? and more importantly, why would I be missing it from a standard asterisk build? |
| 23:16.42 | [TK]D-Fender | talljon84 : * doesn't come with MP3 decode support by default because of legal restrictions. |
| 23:17.05 | [TK]D-Fender | talljon84 : Its part of "asterisk-addons" which is linked right below anyways. |
| 23:18.13 | talljon84 | gotcha.. hmm, i apparently missed that. oops. lol |
| 23:18.19 | *** join/#asterisk justinu|laptop (n=Justin@dsl081-083-093.lax1.dsl.speakeasy.net) |
| 23:23.04 | *** join/#asterisk profounded (n=pro@ool-44c4eae2.dyn.optonline.net) |
| 23:33.16 | *** join/#asterisk wunderkin (n=kev@ip68-226-113-228.ph.ph.cox.net) |
| 23:37.33 | *** join/#asterisk zotz (n=zotz@24.244.163.225) |
| 23:37.35 | *** join/#asterisk profounded (n=pro@ool-44c4eae2.dyn.optonline.net) |
| 23:47.57 | *** join/#asterisk Monie (n=Monie@adsl-154-38-124.clt.bellsouth.net) |
| 23:49.00 | Monie | #NintendoXG |
| 23:52.48 | *** part/#asterisk Monie (n=Monie@adsl-154-38-124.clt.bellsouth.net) |
| 23:57.04 | wunderkin | kjakjdfksjdaflksdaflkasjfldsa linux intel blah |
| 23:58.36 | *** join/#asterisk Greek-Boy (n=Greek-Bo@196.46.109.237) |
