| 00:00.02 | SeRi | p3nguin: ping |
| 00:01.12 | *** join/#asterisk felimwhiteley (~quassel@46.7.101.58) |
| 00:02.05 | p3nguin | http://www.speedtest.net/result/1624447643.png |
| 00:03.03 | SeRi | fucking sick! |
| 00:03.07 | SeRi | I was there once :( |
| 00:04.03 | p3nguin | I'm tellin' ya, I just need to reboot the router and restart networking a few more times and I'll be over 100. :) |
| 00:06.39 | SeRi | how is that making a difference? |
| 00:06.41 | SeRi | lol |
| 00:07.46 | p3nguin | It isn't. It's a joke. |
| 00:08.13 | SeRi | You are convincing! |
| 00:08.13 | SeRi | lol |
| 00:08.13 | _N1X_ | http://www.speedtest.net/result/1624447643.png < ? > |
| 00:08.18 | _N1X_ | 85.75 M |
| 00:08.18 | SeRi | well shit let me try and reboot my router |
| 00:09.38 | SeRi | http://www.speedtest.net/result/1624455796.png :( |
| 00:10.11 | p3nguin | At least you're faster than 77% of the US. |
| 00:10.26 | p3nguin | It's not as fast as my 93%, but still not bad. |
| 00:11.20 | SeRi | rofl |
| 00:11.26 | p3nguin | ;) |
| 00:11.37 | SeRi | Yea now that I sont have extream ya want to show off! |
| 00:11.44 | SeRi | s/sont/dont/ |
| 00:11.51 | p3nguin | don't |
| 00:12.11 | p3nguin | When did you downgrade? |
| 00:12.27 | SeRi | leate last month. |
| 00:12.34 | SeRi | It is worthless in comcast |
| 00:12.54 | SeRi | our line have caps so why the fuck do I need to pay over 150 dollars for bandwith I cant use |
| 00:13.00 | mirco | Hey guy's I need a hand with asterisk-gui: "The GUI does not have necessary privileges. Please check the manager permissions for the user !" But manager and http.conf seem to be fine.. |
| 00:13.26 | p3nguin | We don't support the Asterisk GUI here. |
| 00:13.36 | p3nguin | Try #asterisk-gui |
| 00:13.42 | WIMPy | mirco: #asterisk-gui |
| 00:13.42 | mirco | p3nguin: thx |
| 00:13.56 | WIMPy | I told you before. |
| 00:13.58 | mirco | WIMPy: thx to you too |
| 00:14.29 | mirco | as I said I didn't expect it to be something external… :-( |
| 00:16.44 | SeRi | p3nguin: * is now all ok? |
| 00:16.54 | p3nguin | How would I know? |
| 00:17.02 | *** join/#asterisk _r33dtard (~r33dtard@gateway/tor-sasl/r33dtard) |
| 00:17.08 | SeRi | no more drops.... :/ |
| 00:17.53 | p3nguin | Not since the last time you asked. |
| 00:18.08 | SeRi | p3nguin: Thats what I meant.... If your issues of calls been drop has not presented it self... |
| 00:18.17 | SeRi | I havent. |
| 00:18.23 | p3nguin | Haven't had any calls since the last time you asked. |
| 00:18.55 | p3nguin | Ask again in two days for an accurate answer. |
| 00:19.19 | SeRi | :/ ok :/ lol |
| 00:19.25 | *** join/#asterisk coppice (~chatzilla@host86-136-94-225.range86-136.btcentralplus.com) |
| 00:20.11 | SeRi | p3nguin: so you all ways had those speeds and you didnt know about it? |
| 00:20.31 | p3nguin | The last time I checked my speed, it was around 60 Mbit. |
| 00:20.47 | p3nguin | For several weeks (at least), my shit has been feeling very slow. |
| 00:21.20 | *** join/#asterisk mnicholson (~mnicholso@nat/digium/x-aewxovuztlpdmeeu) |
| 00:21.20 | *** mode/#asterisk [+o mnicholson] by ChanServ |
| 00:21.34 | p3nguin | It was never enough of a problem until today to try to fix it. |
| 00:22.06 | SeRi | o wow.... nice speeds man even at 60Mbps |
| 00:22.16 | SeRi | business line? |
| 00:22.20 | p3nguin | I didn't know we could get over 60. |
| 00:22.33 | p3nguin | No, business class is too expensive for crap service. |
| 00:22.58 | SeRi | indeed |
| 00:23.04 | SeRi | same here in comcastic land |
| 00:23.28 | p3nguin | The only advantage is you get priority in the event of an outage. |
| 00:23.44 | p3nguin | But if there's an outage that affects me, they are going to work on it quickly anyway. |
| 00:23.58 | SeRi | I am in no need of that service... |
| 00:24.01 | SeRi | rofl I bet |
| 00:24.16 | p3nguin | They don't like to have outages. |
| 00:24.30 | SeRi | comcas loves them |
| 00:25.28 | p3nguin | While I wait on the results of my network changes and dropping calls, I'd like to make sure I have a good shaper policy. Do you think 24% to IAX2, 24% to SIP, 24% to RTP, and 24% to default (everything else) is sensible? |
| 00:26.02 | p3nguin | I'm allowing 2Mbit out of my actual 3Mbit upload. |
| 00:26.15 | SeRi | 24% is more than enough. Make sure your prioritys are set correctly and you should be golden |
| 00:26.38 | SeRi | I have mine set a 15% and 20% max |
| 00:26.39 | p3nguin | I could probably go up to 2.50Mbit or even more without a problem, but I figured 2.00 was good enough. |
| 00:26.49 | SeRi | indeed is. |
| 00:27.01 | p3nguin | I don't see any way to set priorities. |
| 00:27.38 | SeRi | ooo ok. |
| 00:27.50 | p3nguin | It's just a shaper, not QoS. |
| 00:28.04 | SeRi | yes for got about that. |
| 00:28.18 | SeRi | s/for got/forgot/ |
| 00:29.31 | p3nguin | When I was researching how to configure it, I kept running across things for vyatta referring to qos policy or something, but my version does not have that, it only has traffic-policy. I don't know if that's something the subscription version has or if it was something in a previous version. |
| 00:29.31 | SeRi | Can't connect to IMAP4 server: imap.mail.r***** |
| 00:30.06 | SeRi | I am sure it must be for the paid version.... |
| 00:30.25 | SeRi | p3nguin: one sec. I brb. |
| 00:31.46 | p3nguin | http://pastebin.com/bGBUBpWR |
| 00:33.03 | *** join/#asterisk tompaw (~tompaw@slave30.tesserakt.eu) |
| 00:33.56 | p3nguin | I think this shaper policy should be reasonable, but I just don't know. |
| 00:34.11 | wcselby | wow, forgot this was open |
| 00:34.14 | wcselby | o/ later all |
| 00:34.31 | *** join/#asterisk radic (~radic@dslb-094-216-250-020.pools.arcor-ip.net) |
| 00:34.56 | *** join/#asterisk mirco (~mirco@p5B282D9C.dip.t-dialin.net) |
| 00:35.09 | *** join/#asterisk master_of_master (~master_of@p57B544AF.dip.t-dialin.net) |
| 00:43.06 | dijib | p3nguin: did you eva figure out that mixmon audio sync. not sure if still having it, but my shaping izza shaping |
| 00:43.46 | SeRi | ok back |
| 00:43.51 | SeRi | waz up dijib |
| 00:45.12 | p3nguin | The last time I checked, it was still out of sync. |
| 00:45.17 | SeRi | p3nguin: Class Based Queueing is a classful qdisc that implements a rich linksharing hierarchy of classes. It contains shaping elements as well as prioritizing capabilities |
| 00:45.35 | SeRi | so you can use priority's |
| 00:45.42 | p3nguin | priorities |
| 00:45.47 | SeRi | *** |
| 00:45.51 | SeRi | Yes Sr! |
| 00:46.00 | dijib | hey all |
| 00:46.12 | p3nguin | We don't use an apostrophe to make words plural. |
| 00:46.57 | SeRi | Yes Sr. |
| 00:47.01 | SeRi | waz up dijib |
| 00:47.02 | p3nguin | For priority settings, I will have to look at what traffic-policy offers. |
| 00:47.09 | dijib | whats up Fro |
| 00:47.12 | dijib | :D |
| 00:47.18 | SeRi | p3nguin: I see. |
| 00:47.36 | dijib | im using prioritiez |
| 00:47.44 | dijib | lol |
| 00:47.47 | dijib | yes i know |
| 00:48.02 | dijib | my head hurts. |
| 00:48.14 | dijib | why must the legal system continually be a joke |
| 00:48.25 | dijib | use computers nimrods |
| 00:49.25 | p3nguin | Oh, it does have a priority setting. |
| 00:49.40 | SeRi | p3nguin: Is it "class"? |
| 00:49.54 | p3nguin | set traffic-policy shaper VoIP-out class 20 priority |
| 00:49.55 | p3nguin | Possible completions: <0-7> Priority order for bandwidth pool |
| 00:50.04 | SeRi | ah. nice |
| 00:50.29 | p3nguin | Now I need to think about setting some priority. |
| 00:50.33 | SeRi | The higher the better. I have mine set at 7 |
| 00:50.37 | dijib | what are you using? your appliance |
| 00:50.49 | p3nguin | vyatta |
| 00:50.54 | dijib | mines the other way. lower, is higher priority |
| 00:50.59 | dijib | pretty |
| 00:51.00 | dijib | ;) |
| 00:51.11 | dijib | vyatta |
| 00:51.15 | WIMPy | is with dijib |
| 00:51.23 | dijib | lol |
| 00:51.44 | p3nguin | Should IAX2, SIP, and RTP all have the same priority? |
| 00:51.58 | SeRi | Yes. That should be ok |
| 00:52.05 | dijib | no clue. |
| 00:52.12 | dijib | rtp should be more? |
| 00:52.16 | SeRi | everything else in the queue's could have lower priority |
| 00:52.22 | SeRi | stuff you dont care for... |
| 00:52.37 | SeRi | dijib: 7 is the highest |
| 00:52.49 | dijib | 0 is highest |
| 00:53.00 | dijib | for my non vyetta, lincoln |
| 00:53.28 | dijib | traffic control |
| 00:53.29 | SeRi | p3nguin: maybe yours is set the same way since it uses tc... not sure. I know on bsd 7 is highest |
| 00:53.40 | p3nguin | For now, everything is fair-queue. Since it is dedicated to my asterisk system, I think it is okay to not use priority. |
| 00:53.42 | dijib | and come to think of it, i should be producing a backup box. |
| 00:53.51 | dijib | i mean another backup box. |
| 00:54.01 | SeRi | p3nguin: +1 I agree. if its the only device in that segment |
| 00:54.19 | p3nguin | It is the only system using that router as a gateway. |
| 00:54.28 | dijib | my script still spits out warnings everywhere but its working well from what i can tell. |
| 00:54.32 | SeRi | You should be ok. |
| 00:54.35 | WIMPy | There are implied priorities. |
| 00:54.42 | SeRi | dijib: LOL |
| 00:55.03 | dijib | srsly dude |
| 00:55.08 | dijib | shit works, |
| 00:55.14 | p3nguin | But eventually, this topology will change and everything will use the vyatta as a gateway. At that point, I may have to look at priorities a little closer. |
| 00:55.15 | SeRi | I dont doubt you. |
| 00:55.18 | dijib | or poop was it |
| 00:55.35 | SeRi | p3nguin: I see. |
| 00:55.40 | p3nguin | poop shaper |
| 00:55.44 | dijib | lol |
| 00:55.48 | dijib | yeh that was it |
| 00:55.50 | SeRi | shit shaper |
| 00:56.00 | dijib | sfinkter |
| 00:56.10 | dijib | shaper |
| 00:56.22 | p3nguin | I will eliminate a router which is sucking up power. |
| 00:56.28 | p3nguin | over 110 Watts. |
| 00:56.32 | WIMPy | Just imagine putting nozzels up your ass. |
| 00:56.38 | p3nguin | The Vyatta uses 22 Watts. |
| 00:56.45 | SeRi | p3nguin: nice |
| 00:56.52 | dijib | you actually have the appliance? |
| 00:56.57 | SeRi | My asterisk uses 12v :P |
| 00:57.01 | dijib | i figured you were just running their soft. |
| 00:57.03 | p3nguin | Mine, too. |
| 00:57.10 | SeRi | p3nguin: Nice! |
| 00:57.12 | dijib | yeah but thats not consumption seri |
| 00:57.20 | WIMPy | Yes. Network equipment is evil. That's why I replaced the switch with a quad HME. |
| 00:57.28 | SeRi | d00d trust me it does not go over 12v |
| 00:57.42 | p3nguin | Both the asterisk and vyatta boxes are 12V. |
| 00:57.49 | dijib | thats just voltage of v x amps = watts was that the math? |
| 00:58.00 | p3nguin | I have not measured the usage of the asterisk box, but I did on the other. |
| 00:58.02 | SeRi | its a 12v embedded system |
| 00:58.09 | SeRi | max 1.2amps |
| 00:58.14 | SeRi | if I am not mistaken |
| 00:58.26 | SeRi | dijib: I know |
| 00:58.34 | SeRi | what I mean is that the system it self is 12v |
| 00:58.47 | dijib | 12.4 watts? |
| 00:58.58 | p3nguin | Could be! |
| 00:59.01 | SeRi | I know that it runs the watts over comsumption time |
| 00:59.19 | SeRi | messuring devices like that is stupid ebcause everything comsumes electricity |
| 00:59.25 | dijib | Watts is a unit of power having the dimensions (energy per unit time) |
| 00:59.30 | WIMPy | Do they share a PSU? |
| 00:59.30 | dijib | M L2 / T2 divided by T = M L 2 / T 3 |
| 01:00.15 | p3nguin | I have two separate systems, so no they don't share a PSU. |
| 01:00.41 | SeRi | p3nguin: you built them correct? |
| 01:00.47 | p3nguin | no |
| 01:00.48 | p3nguin | HP did |
| 01:00.53 | SeRi | ah there hp. |
| 01:00.54 | WIMPy | One large PSU uses less power than several small ones. |
| 01:01.19 | dijib | WIMPy: i hear that |
| 01:01.31 | SeRi | <PROTECTED> |
| 01:01.37 | dijib | espacially if you have battery banks and generation solution |
| 01:01.38 | dijib | s |
| 01:02.20 | WIMPy | I saved some power just bu connecting the wifi AP to the PC instead of using the dedicated PSU. |
| 01:02.33 | dijib | and nuclear silo's, they help too |
| 01:02.37 | WIMPy | And my LED lighting is also connected to the PC. |
| 01:02.38 | p3nguin | silos |
| 01:02.45 | p3nguin | not "silo is" |
| 01:04.02 | dijib | i would think the plural is a thing, has a plural using 's |
| 01:04.18 | dijib | i know that made no sense |
| 01:04.24 | dijib | ok like silo is a thing |
| 01:04.27 | p3nguin | You don't us apostrophe for making something plural. |
| 01:04.29 | dijib | 2 silo's |
| 01:04.31 | dijib | is thus |
| 01:04.32 | p3nguin | no |
| 01:04.36 | p3nguin | two silos |
| 01:04.52 | dijib | not thinking so |
| 01:04.53 | p3nguin | Apostrophes are for possession and contractions. |
| 01:04.54 | dijib | lol |
| 01:05.18 | dijib | you a prof or something, how can you pull that out like that? |
| 01:05.20 | p3nguin | the silo's walls |
| 01:05.23 | dijib | <PROTECTED> |
| 01:05.46 | p3nguin | The silos are for shaping shit. |
| 01:05.54 | SeRi | WIMPy: did you see my question? |
| 01:06.02 | dijib | dude my shit shaper is shapin shit just fine thank you |
| 01:06.03 | WIMPy | Plural is not a contraction, either. |
| 01:06.18 | WIMPy | But it can be a condition if it's about your personality. |
| 01:06.33 | p3nguin | one silo's thing |
| 01:06.38 | p3nguin | two silos' things |
| 01:06.46 | dijib | too greek for me dude |
| 01:06.52 | p3nguin | It's English. |
| 01:06.57 | coppice | WIMPy: many of the current generation of small PSUs are extremely efficient, although older ones can be pretty poor |
| 01:07.03 | WIMPy | SeRi: Yes. That's why I connect everything I can to the PC. |
| 01:07.22 | SeRi | Ok. Than I am good. |
| 01:07.42 | WIMPy | coppice: Yes, but the statement remains true none the less. |
| 01:08.30 | coppice | no it doesn't. few PC PSUs are much above 80% efficient. 90+ is common for small wall warts now |
| 01:08.37 | WIMPy | And non-switching PSUs should definitely be avoided. |
| 01:08.56 | p3nguin | My 12V bricks are very similar and the CPUs are similar, so I'd estimate the consumption of those two systems to be near the same if not equal. |
| 01:09.15 | WIMPy | I didn't say PC PSU are the best. Just that a big one is better than several small ones. |
| 01:09.41 | p3nguin | Actually, I think the CPUs are exactly the same. |
| 01:09.47 | WIMPy | PC stuff generelly tends to be cheap and inefficient. |
| 01:10.57 | dijib | why are they bricks?!?! jtag |
| 01:11.06 | SeRi | rofl!!!!!!!!!!! |
| 01:11.08 | SeRi | hahahahaha |
| 01:11.12 | dijib | oh power bricks |
| 01:11.15 | dijib | not following |
| 01:11.16 | SeRi | brick = psu |
| 01:11.22 | SeRi | hahaha |
| 01:11.25 | p3nguin | Both are VIA Nehemiah CentaurHauls 800 MHz. |
| 01:11.42 | dijib | 256? |
| 01:12.01 | p3nguin | Just two. |
| 01:12.03 | WIMPy | You can get the PC inside the PSU, like the SheevaPlugs, etc.. |
| 01:12.25 | dijib | dude, rasberrypi |
| 01:12.28 | dijib | seen it? |
| 01:12.34 | WIMPy | yes |
| 01:12.39 | dijib | everywhere. |
| 01:13.01 | WIMPy | Lots of nice hardware comming up. |
| 01:13.08 | dijib | could handle small asterisk deployments |
| 01:13.16 | WIMPy | Beagleboard, etc. |
| 01:13.47 | WIMPy | They're just missing interfaces. |
| 01:13.52 | WIMPy | usually. |
| 01:19.03 | dijib | anybody else cold? |
| 01:20.24 | dijib | -1c? |
| 01:20.42 | SeRi | synology nas are cool. |
| 01:21.07 | dijib | ok so i use for text2speech but the text2wave is brutal |
| 01:21.19 | dijib | so how do i go about getting what was it swift? |
| 01:22.34 | p3nguin | 4 C over here. |
| 01:23.03 | p3nguin | It will be 0 before the night is over. |
| 01:23.12 | dijib | snow? |
| 01:23.20 | p3nguin | Not yet. |
| 01:23.32 | dijib | we got about 8 inches the other night |
| 01:23.44 | dijib | still sticking around. |
| 01:23.50 | dijib | not happy about it |
| 01:24.22 | WIMPy | likes snow, but we are at a warm 6.6°C here. |
| 01:24.31 | dijib | so when installing cepstral voices what version do i download? |
| 01:24.40 | dijib | for the asterisk box? |
| 01:24.49 | dijib | http://downloads.cepstral.com/cepstral/i386-linux/Cepstral_Allison_i386-linux_5.1.0.tar.gz |
| 01:24.54 | dijib | ??? |
| 01:25.23 | p3nguin | Use. Your. Package. Manager. |
| 01:28.46 | dijib | your mean. |
| 01:28.57 | dijib | you know the packet manager doesnt have cepstral |
| 01:29.01 | dijib | does it? |
| 01:29.05 | dijib | i doubt it |
| 01:29.18 | dijib | im downloading and building, but do i have to compile allison. |
| 01:29.18 | WIMPy | Be your own package manager. |
| 01:29.38 | dijib | libraries man libraries |
| 01:29.45 | dijib | 23% |
| 01:29.50 | p3nguin | If they don't offer an RPM, build your own. |
| 01:30.03 | dijib | ive attempted to build * b4 and had issue being my own manager |
| 01:30.18 | dijib | if they dont. what about just skip to build your own? |
| 01:31.08 | dijib | wait you can have dog voices? |
| 01:31.11 | dijib | <PROTECTED> |
| 01:31.26 | p3nguin | In the install instructions, just replace "make install" with "checkinstall" |
| 01:31.44 | dijib | y? |
| 01:32.13 | p3nguin | That's the n00b's way to build a package. |
| 01:32.27 | dijib | whats it dO? |
| 01:32.30 | p3nguin | Then you can manage it correctly with your package manager. |
| 01:33.01 | dijib | whats the non noob way to install? |
| 01:33.12 | p3nguin | checkinstall |
| 01:33.45 | p3nguin | Non-noob way? Write your own srpm. |
| 01:34.04 | p3nguin | It's not that hard, but you'll never be able to do it. |
| 01:34.16 | dijib | i think your right |
| 01:34.21 | p3nguin | not my left? |
| 01:34.35 | dijib | i think your tuesday actually p3nguin |
| 01:34.52 | p3nguin | Today is my Thursday, though. |
| 01:35.06 | dijib | i think this thursdays mine |
| 01:35.12 | dijib | but apparently not, its yours |
| 01:35.13 | p3nguin | I'll get the next one |
| 01:35.18 | dijib | im actually really mad at this day |
| 01:35.25 | dijib | this fucking cunttree is a joke |
| 01:35.26 | p3nguin | Slap it around a bit. |
| 01:35.32 | dijib | the wife? |
| 01:35.35 | p3nguin | Sure. |
| 01:35.39 | dijib | heh |
| 01:35.46 | dijib | fucking day |
| 01:35.55 | dijib | 1st of fuck you december '11 |
| 01:35.57 | dijib | pricks |
| 01:35.59 | dijib | lol |
| 01:36.20 | dijib | not a good day |
| 01:36.25 | dijib | 73% |
| 01:36.36 | dijib | come to my rescue alison |
| 01:36.48 | dijib | bring something good to this day |
| 01:37.01 | SeRi | dijib: conf? |
| 01:37.12 | dijib | meh. |
| 01:37.14 | dijib | i guess |
| 01:37.21 | dijib | i should move to the back office |
| 01:37.30 | dijib | and i should also call that the back oriface |
| 01:37.35 | dijib | because technically |
| 01:37.37 | dijib | its one |
| 01:37.42 | dijib | it's |
| 01:37.44 | dijib | pricks |
| 01:38.11 | dijib | crap i didnt screen my terminal |
| 01:38.15 | SeRi | no stress if you are bussy |
| 01:38.19 | dijib | im finding screen pretty invaluble |
| 01:38.31 | dijib | im not but im going to move the workstation |
| 01:38.34 | dijib | " " |
| 01:38.40 | dijib | in 93s |
| 01:38.45 | dijib | waiting for this download to finish |
| 01:38.51 | dijib | 80s |
| 01:39.03 | dijib | i see you |
| 01:39.06 | p3nguin | Going backward? |
| 01:39.10 | WIMPy | It will abort at 99% anyway. |
| 01:39.12 | SeRi | lmao |
| 01:39.16 | SeRi | hahaha |
| 01:39.22 | dijib | .54 SeRi |
| 01:39.30 | SeRi | yes |
| 01:39.39 | dijib | 169.54 |
| 01:39.54 | dijib | how did you stop MOH? |
| 01:39.55 | p3nguin | I'd rather use dtach for a lot of things many people use screen for. |
| 01:40.06 | dijib | dtach ive never heard of |
| 01:40.17 | p3nguin | yum -y install dtach |
| 01:40.22 | p3nguin | man dtach |
| 01:41.18 | SeRi | o wow they where here ^^ |
| 01:41.24 | *** join/#asterisk woleium (~woleium@S0106002369a9537f.ok.shawcable.net) |
| 01:41.25 | p3nguin | haha |
| 01:41.34 | SeRi | lol |
| 01:41.37 | p3nguin | Someone called TheCops. |
| 01:41.43 | SeRi | hahaha |
| 01:41.52 | SeRi | dijib: did you call them? |
| 01:41.57 | p3nguin | Fortunately for dijib, TheCops left. |
| 01:41.59 | dijib | nope |
| 01:42.01 | SeRi | hahahaha |
| 01:42.06 | dijib | k going to the back, brb |
| 01:44.01 | SeRi | p3nguin: you busy? |
| 01:46.06 | dijib | back |
| 01:47.36 | p3nguin | only a little. |
| 01:47.37 | dijib | seri did you leave? |
| 01:47.49 | SeRi | dijib: yes one sec |
| 01:47.53 | dijib | i might have called the cops |
| 01:47.55 | SeRi | ill be there in 2min |
| 01:47.58 | SeRi | lol |
| 01:48.12 | dijib | it better fucking warm up in here.. i think i need a baseboard heater |
| 01:48.42 | WIMPy | That's what happens if you reduce the electrical heating. |
| 01:49.40 | dijib | its go a gas outlet but i would have to keep this door open for the system to know where |
| 01:53.24 | *** join/#asterisk TheCops (~mdb@72.55.132.180) |
| 01:55.03 | SeRi | TheCops: are back! |
| 01:55.39 | *** join/#asterisk fisted (~fisted@unaffiliated/fisted) |
| 01:56.46 | WIMPy | As soo as TheCops are back, you get fisted. |
| 01:57.02 | SeRi | rofl!!!!!!!!!!!!!!!!!!!!!!!! |
| 01:57.04 | SeRi | hahaha |
| 01:58.56 | _N1X_ | please i need to setup dial plan for trunk001 in context001 with prefix 99|. |
| 01:59.02 | _N1X_ | how to do it ! |
| 02:00.31 | p3nguin | Good luck dialing a pipeline/vertical bar from your keypad. |
| 02:03.32 | [TK]D-Fender | _N1X_: #freepbx <- |
| 02:03.51 | [TK]D-Fender | _N1X_: And stop hand editing the config files. |
| 02:10.16 | *** join/#asterisk rpluto (~rpluto@a95-92-60-159.cpe.netcabo.pt) |
| 02:12.36 | *** join/#asterisk mintos (~mvaliyav@117.206.21.68) |
| 02:18.37 | p3nguin | Hmm, that's a first. I just saw an iPod Touch commercial on TV. |
| 02:19.35 | rpluto | i never see in my country apple ads |
| 02:19.41 | rpluto | on tv |
| 02:19.52 | SeRi | seriously? I see them all the time.... well they stop for a while... |
| 02:20.04 | p3nguin | I see iPhone all the time, but never an iPod Touch one. |
| 02:20.06 | [TK]D-Fender | I've been broadcast TV free for over 6 years now.... The few times I see commercials now I straigt-up cringe... |
| 02:20.58 | rpluto | but truth to be told, i dont i see lot of tv, only series and movies |
| 02:21.14 | jaytee | cringing can cause arterial sclerosis |
| 02:21.22 | *** join/#asterisk cbwest (~cbwest@nat/cisco/x-alaqywgzsetvozjq) |
| 02:21.23 | p3nguin | Hey, look! It's that Cisco guy, cbwest, again. |
| 02:22.13 | SeRi | lol |
| 02:22.13 | *** join/#asterisk master_of_master (~master_of@p57B543EC.dip.t-dialin.net) |
| 02:22.54 | s[X] | ahoi SeRi and [TK]D-Fender |
| 02:23.06 | SeRi | s[X]: hola! |
| 02:23.40 | rpluto | i need some alcatel guy to make me clear how i can integrate my asterisk sip with the old pbx |
| 02:24.06 | p3nguin | Does the legacy PBX support SIP? |
| 02:24.25 | rpluto | with cisco its everything easy |
| 02:24.39 | rpluto | because u have products for everything |
| 02:25.21 | rpluto | p3nguin: right now i have one voip card for 8 channels |
| 02:25.27 | rpluto | maby is on sip |
| 02:25.48 | rpluto | but i can find configurations for |
| 02:26.38 | s[X] | rpluto: how old is it ? |
| 02:26.56 | rpluto | the pbx, is recent |
| 02:27.03 | rpluto | 3 for years |
| 02:27.16 | rpluto | 3 / 4 years |
| 02:27.40 | p3nguin | 3/4? Is that .75 year? |
| 02:28.22 | s[X] | 3 or 4 years i think he means |
| 02:31.30 | rpluto | yes 3 or 4 |
| 02:32.05 | rpluto | is a Alcatel OmniPCX with 7.xxx firmware i think |
| 02:35.01 | rpluto | i want virtualized asterix, because i dont need fxo or fxs i need direct connection to the pbx and make it works, its a interesting project |
| 02:36.04 | rpluto | and after that have the possibilitie to have sip:email to the ext of the coloborator |
| 02:36.11 | rpluto | with that email |
| 02:36.32 | rpluto | office extension or mobile extension |
| 02:38.09 | *** join/#asterisk fisted (~fisted@unaffiliated/fisted) |
| 02:40.08 | SeRi | dijib: you cut off? |
| 02:48.02 | SeRi | p3nguin:!!!!!!!!!!!!!!!!!! |
| 02:48.20 | p3nguin | YAY!! |
| 02:48.29 | SeRi | hahahahaha! |
| 02:48.40 | Sedorox | OK... so... I've used asterisk for probably 6-7 years now... and I love it.. it's awesome.. and I finally got the chance to do a VoIP Deployment for a company, and I picked Switchvox... so far it's Freaking Awesome! |
| 02:49.40 | p3nguin | Is that the Digium appliance? |
| 02:50.51 | Sedorox | yup |
| 02:51.00 | rpluto | Sedorox: ist possible to integrade with old things |
| 02:51.04 | dijib | p3nguin: how can i load app_swift |
| 02:51.14 | dijib | its not in module show |
| 02:51.17 | *** join/#asterisk bmg505 (~leon@196-209-123-3.dynamic.isadsl.co.za) |
| 02:51.20 | Sedorox | rpluto: eh? |
| 02:51.39 | p3nguin | If you loaded it, it would be in module show. |
| 02:51.58 | rpluto | old pbx Sedorox |
| 02:52.06 | dijib | i did module reload, it: install -m 755 app_swift.so /usr/lib/asterisk/modules |
| 02:52.31 | Sedorox | rpluto: you mean to integrate with the old PBX system? so your running both? |
| 02:52.51 | p3nguin | Why would you run "module reload"? |
| 02:53.06 | rpluto | yes. to use old phone and other things |
| 02:53.13 | p3nguin | Install the module. Load the module. |
| 02:53.41 | rpluto | but give the possibitie to the new ipbx features |
| 02:54.28 | Sedorox | I know you can, and I looked into it.. but in this instance.. they want to get rid of the old PBX (Nortel NorStar system that is dying slowly), so we're going to do a full swapout over a weekend |
| 02:54.50 | Sedorox | that and to do the integration, woudl have ment buying other cards for the nortel system, and not worth it when they are switching out |
| 02:55.32 | rpluto | ya that is a good thing but i cant do that no money for that |
| 02:56.14 | Sedorox | hehe yea, I was lucky.. 15 hardwire phones, 25 extensions total.. so in the scheme of things it wasn't too bad |
| 02:56.14 | rpluto | i need to study the best solution for that with the minimum investment |
| 02:56.19 | Sedorox | and they had the money too :p |
| 02:56.47 | rpluto | ehehe i am talking about 100 phones |
| 02:57.01 | Sedorox | rpluto: oh, did you ask if it was possible to integrate? I thought you said "it is possible", not in the question form |
| 02:57.08 | rpluto | 2 gsm gateways |
| 02:57.27 | Sedorox | nice |
| 02:58.00 | Sedorox | the best way to integrate would be a T1 trunk |
| 02:58.01 | rpluto | yes is possible, but i need to do for my self, not money to use |
| 02:58.05 | Sedorox | I would think |
| 02:58.11 | Sedorox | ah |
| 02:59.07 | rpluto | and some configurations on old pbx i ask for support and only for that is lot of money to spend |
| 02:59.31 | Sedorox | oh? |
| 02:59.52 | rpluto | t1 trunk u talk about bandwith? |
| 03:00.30 | Sedorox | well physical T1.. would give you 24 channels between the systems.. most will allow you to extend extensions over a T1 for system links |
| 03:01.46 | rpluto | dont know the bandwith of a T1 |
| 03:02.06 | Sedorox | 1.544Mbit |
| 03:02.36 | rpluto | but onsite we have 1Gbit and between sites we have 4Mbit |
| 03:03.18 | rpluto | and all the site have 30/3 dwl upl internet |
| 03:03.48 | rpluto | 30Mbit download 3Mbit upload |
| 03:04.11 | rpluto | but for evrything not only for voip |
| 03:05.04 | Sedorox | honestly it really depends on what your old system is, what you pick for the new one, and what your options are for interconnecting the two |
| 03:05.42 | rpluto | we will contact some asterisk or digium resaller or partner to help me with |
| 03:06.30 | rpluto | sure, the old pbx is a alcatel OmniPCX |
| 03:06.43 | rpluto | in booth sites |
| 03:06.59 | rpluto | we have 2 of them |
| 03:07.30 | Sedorox | ah, not familar with it so I can't really help |
| 03:08.11 | Sedorox | all I know is I'm impressed with what Digium has put together with the GUI and such... I know the configs fairly well, so it took me a bit to get familar with how they are doing things GUI wise.. but it's sweet |
| 03:08.15 | Sedorox | I'm impressed |
| 03:08.38 | rpluto | no problem i am here to talk about this and get ideas or help someone in things i can help |
| 03:09.30 | [TK]D-Fender | Sedorox: Which GUi are you referring to? |
| 03:09.31 | rpluto | Sedorox: can u say the price for it |
| 03:10.11 | rpluto | [TK]D-Fender: possible Digium own GUI!!! |
| 03:10.28 | rpluto | to config the apliance |
| 03:10.36 | [TK]D-Fender | rpluto: I'm asking for him to be specific in terms of which one he is referring to. |
| 03:10.46 | [TK]D-Fender | Also depends on "which appliance" |
| 03:11.08 | [TK]D-Fender | The old AA50 ran AsteriskGUI. That one is just short of dead developmentally. |
| 03:11.25 | [TK]D-Fender | Their paid commerical product, Switchvox, is another matter |
| 03:11.50 | rpluto | is that one |
| 03:12.00 | rpluto | Switchvox |
| 03:12.19 | rpluto | 02:49 Sedorox OK... so... I've used asterisk for probably 6-7 years now... and I love it.. it's awesome.. and I finally got the chance to do a VoIP Deployment for a company, and I picked Switchvox... so far it's Freaking Awesome! |
| 03:13.35 | [TK]D-Fender | Ok, that used to be from a separate company then Digium bought them out. It's closed and commercial... not sure what you can do with it compared to other solutions |
| 03:13.51 | rpluto | someone knows one good companie for this jobs in north of portugal(porto) |
| 03:15.47 | rpluto | me neither |
| 03:16.22 | puzzled | rpluto: http://www.asterisk.pt/ http://www.voip-info.org/wiki/view/Asterisk+Consultants+Portugal |
| 03:17.47 | rpluto | puzzled: thx |
| 03:19.59 | Sedorox | [TK]D-Fender: It's the Switchvox GUI.. as was pointed out, it's the commerical side of Digium.. and I know there's other solutions, but since this is consulting, and I'm not there full time, I wanted something with full support |
| 03:20.20 | Sedorox | This is a SMB65 unit they got |
| 03:21.06 | Sedorox | rpluto: I got the SMB65, 25 extensions, 15 phone packs, TDM800 (for 8 incoming analog lines), and 4 years extended support... was around $8.5k |
| 03:21.10 | Sedorox | priced through digium's website |
| 03:21.17 | [TK]D-Fender | Sedorox: I just missed the line where you mentioned it. I jsut saw "GUI" all over the conversation with a name. Now it's clear |
| 03:21.29 | rpluto | Sedorox: thx |
| 03:22.05 | Sedorox | ah |
| 03:22.05 | Sedorox | :) |
| 03:22.52 | rpluto | someone have try or see lync festures? |
| 03:22.54 | Sedorox | yea, I'm comfortable with just throwing plain Asterisk on a Linux box, but I wanted something that if I were to leave the area for a different job, or even now, they can click a few things and add an extension, or change settings, etc |
| 03:23.03 | rpluto | *features |
| 03:24.13 | Sedorox | my normal job is starting to deploy lync, but it doesn't have more then a handful of people (and isn't complete yet) |
| 03:31.34 | dijib | CLI> module show like app_swift |
| 03:31.34 | dijib | Module Description Use Count |
| 03:31.37 | dijib | 0 modules loaded |
| 03:32.00 | p3nguin | Did you ever bother to load it? |
| 03:32.11 | dijib | but its in /usr/lib/asterisk/modules |
| 03:32.17 | dijib | and yes ive bothered |
| 03:32.21 | p3nguin | Yeah? So? |
| 03:32.31 | p3nguin | Having the module does not make it a loaded module. |
| 03:32.34 | dijib | unable to load. |
| 03:32.42 | dijib | i realise |
| 03:32.43 | p3nguin | Then why would you expect module show to show it? |
| 03:32.47 | dijib | ive tried to load.. its failing |
| 03:32.51 | p3nguin | Then why would you expect module show to show it? |
| 03:33.01 | dijib | becuase i core restart now |
| 03:33.04 | dijib | 'ed |
| 03:33.21 | dijib | hit it with a hammer |
| 03:33.33 | dijib | i had already tried to module load app_swift.so |
| 03:33.38 | dijib | fails |
| 03:33.53 | p3nguin | And yet you still expect module show to show it. |
| 03:34.07 | dijib | :@ no i dont ok |
| 03:34.16 | dijib | permissions are 755 on the file |
| 03:34.26 | dijib | do i need to config swift.conf? |
| 03:35.56 | *** join/#asterisk g_r_eek (~g_r_eek@173-9-142-122-Miami.FL.hfc.comcastbusiness.net) |
| 03:36.52 | *** join/#asterisk g_r_eek (~g_r_eek@173-9-142-122-Miami.FL.hfc.comcastbusiness.net) |
| 03:45.29 | SeRi | wants a Raspberry Pi |
| 03:46.06 | *** join/#asterisk mindCrime (~chatzilla@cpe-076-182-089-009.nc.res.rr.com) |
| 03:54.16 | SeRi | p3nguin: did you hit 100Mbps yet? |
| 03:55.22 | p3nguin | I think I still new a couple more reboots. |
| 03:56.02 | SeRi | lol! |
| 03:57.32 | SeRi | I am still working at it.... I am not having the same results though :/ |
| 03:58.48 | SeRi | p3nguin: any low budget speaker phone that you recomend for the kids? |
| 03:58.56 | SeRi | room* |
| 03:59.57 | SeRi | I had a grandstream in there room and it died.... |
| 04:00.05 | SeRi | s/there/their/ |
| 04:00.14 | p3nguin | ~grandstream |
| 04:00.14 | infobot | [grandstream] the Yugo of VoIP hardware. Run... Run away now. Though, therealcircut says that they're not that bad. |
| 04:00.51 | SeRi | LOL I know but hey it was free and you cant say no to free :P |
| 04:01.29 | SeRi | I need it to be a sip phone so i can activate the speaker |
| 04:01.35 | SeRi | with auto answer |
| 04:01.52 | p3nguin | Or an SCCP phone. |
| 04:02.02 | SeRi | ? |
| 04:02.08 | p3nguin | ~sccp |
| 04:02.08 | infobot | [sccp] Proprietary protocol used between Cisco Call Manager and Cisco VOIP phones. Also supported by some other vendors. Also Signaling Connection Control Part (SCCP), a routing protocol in SS7 protocol suite in layer 4, provides end-to-end routing for TCAP messages to their proper database. |
| 04:02.19 | *** join/#asterisk StaRetji (~BigEight@80.93.240.171) |
| 04:02.50 | SeRi | ah I see. sounds expensive :/ |
| 04:08.37 | *** join/#asterisk AgroTemp (~Agro@108-79-20-223.lightspeed.hstntx.sbcglobal.net) |
| 04:09.05 | AgroTemp | I was writing a function in C for wait_for_digit, but when I try to return an array of char*, it won't work... |
| 04:09.28 | AgroTemp | When I don't return anything, it works fine... but I need that array. |
| 04:10.14 | p3nguin | Do you want a phone with speakerphone, or do you want a conference phone? |
| 04:10.28 | AgroTemp | I mean, returning the char* array works for stream_file, say_number, but it doesn't work when I try wait_for_digit |
| 04:13.05 | AgroTemp | Does anyone know why this is happening? The WAIT FOR DIGIT command doesn't even make it to Asterisk's buffer, but it hasn't even returned anything since then... |
| 04:13.08 | SeRi | p3nguin: a phone with speaker |
| 04:13.23 | p3nguin | How much do you want to spend? |
| 04:14.50 | SeRi | p3nguin: well no more than 50 dollars.... I guess.... All I need is for the phone to auto answer so I can call them down stairs instead of having to shout |
| 04:17.36 | p3nguin | You can get a Cisco 7940 or one of several Polycom devices for that price. |
| 04:18.35 | SeRi | Really? Where? The cheapest polycom I found was the 321 for 80.00 |
| 04:19.40 | *** join/#asterisk kikohnl (~kotis@udp019436uds.hawaiiantel.net) |
| 04:21.03 | p3nguin | Ebay... 330, 430, 321, 301, 501, and several soundstation conf phones... $50 or less. |
| 04:22.54 | p3nguin | Too bad you don't want a conference phone. http://www.ebay.com/itm/ws/eBayISAPI.dll?ViewItem&item=330647987873 |
| 04:24.03 | SeRi | p3nguin: Is it worth it? would it work for what I need? |
| 04:24.07 | SeRi | Ill but it now |
| 04:24.11 | SeRi | Thats cheap as hell! |
| 04:24.18 | SeRi | I though it was more epensive! |
| 04:25.00 | p3nguin | It's just a table-top conference phone. It works like a regular phone, but it is a 360 degree speakerphone only. |
| 04:25.24 | p3nguin | You call it and make calls from it just like a handset phone. |
| 04:25.36 | SeRi | well sold. |
| 04:25.44 | p3nguin | You may even be able to wall mount it. |
| 04:26.27 | p3nguin | Send it an auto-answer, and it'll be like a whole room intercom. |
| 04:26.43 | dijib | http://www.ebay.com/itm/ws/eBayISAPI.dll?ViewItem&item=330647987873 |
| 04:27.25 | *** join/#asterisk irroot (~gregory@197.170.73.99) |
| 04:28.08 | SeRi | I can send an auto answer to a phone via asterisk? |
| 04:28.20 | p3nguin | Well sure. |
| 04:28.24 | p3nguin | How did you do it before? |
| 04:28.33 | irroot | SeRi ?? like itercom ? |
| 04:28.41 | SeRi | the phone had an option to just auto answer everything |
| 04:28.46 | p3nguin | Oh. |
| 04:29.08 | p3nguin | You can send a SIP header that tells the phone to answer. |
| 04:29.16 | SeRi | but if I can just do it vi asterisk even better! |
| 04:29.33 | SeRi | irroot: Its for the kids room |
| 04:29.36 | p3nguin | For intercom, I just prefix the extension number with *00. |
| 04:29.36 | SeRi | irroot: Yes |
| 04:29.53 | p3nguin | So if the extension for the phone is 123, I'd dial *00123 to make it an intercom. |
| 04:29.58 | SeRi | p3nguin: so this polycom will work with asterisk? |
| 04:30.02 | p3nguin | Sure. |
| 04:30.03 | dijib | you could remotely yell at the kids |
| 04:30.16 | SeRi | p3nguin: ok I jjust bought it |
| 04:31.01 | irroot | SeRi i did some ju-ju with the snom worked quite well was while back |
| 04:31.19 | irroot | aint done so on polycom |
| 04:31.31 | p3nguin | It's just a SIP header to make it answer. |
| 04:32.21 | p3nguin | same => n(autoanswer),Set(_ALERT_INFO="RA"); This is for the Polycoms |
| 04:32.26 | p3nguin | same => n,Dial(${DEVICE}); |
| 04:32.50 | p3nguin | or |
| 04:32.52 | p3nguin | same => n,SIPAddHeader(Call-Info: Answer-After=0); This is for the Grandstream, Snoms, and Others |
| 04:33.11 | dijib | why would app_module.so not load? |
| 04:33.20 | p3nguin | RA = Ring Answer |
| 04:33.21 | dijib | also res_fax_digium.so doesnt load |
| 04:34.55 | irroot | p3nguin i know on the snom you need to enable it in the phone explicitly or its ignored SeRi heads up may need to check the .cfg |
| 04:35.07 | irroot | dijib is app_fax loaded ?? |
| 04:35.19 | irroot | what error it give |
| 04:36.29 | SeRi | p3nguin: p3nguin msg me please |
| 04:36.46 | p3nguin | seri: http://pastebin.com/PyYdDA2S |
| 04:36.47 | dijib | no its not |
| 04:37.09 | dijib | asterisk.conf? |
| 04:38.14 | irroot | dijib modules.conf |
| 04:38.54 | SeRi | I just bought it |
| 04:39.05 | SeRi | if you go back to the seller is no longer avail :) |
| 04:39.43 | SeRi | I hope I can get the configs for it.... |
| 04:43.01 | p3nguin | You didn't check that before buying? |
| 04:43.15 | SeRi | I just tought about that right now :( |
| 04:43.32 | p3nguin | If it doesn't work out, put it back on ebay. |
| 04:44.10 | *** join/#asterisk sawgood (~sawgood@173-13-158-27-sfba.hfc.comcastbusiness.net) |
| 04:44.23 | p3nguin | Or talk the seller into cancelation. |
| 04:45.05 | SeRi | well fuck looks like this are not sip :( |
| 04:45.25 | p3nguin | SoundStation isn't SIP? Since when? |
| 04:46.09 | SeRi | I am reading something about DCP |
| 04:46.31 | SeRi | Wall module is DCP Module (Digital Communications Protocol) |
| 04:46.46 | p3nguin | what the ... |
| 04:47.51 | SeRi | I have no clue :/ |
| 04:48.30 | p3nguin | Maybe you should ask for cancelation. |
| 04:49.27 | p3nguin | "Oh nose! I bought the bad phone! Please undo!" |
| 04:51.02 | dijib | ok why dont my .so's load? |
| 04:51.26 | p3nguin | It's December 1, 2011. |
| 04:51.39 | p3nguin | Try again tomorrow. |
| 04:51.48 | dijib | huh? |
| 04:52.15 | p3nguin | I thought you told me earlier that today was a bad day and nothing was working out for you. |
| 04:52.23 | dijib | oh it was |
| 04:55.09 | sawgood | When needing to send information from Asterisk to a SMS gateway (would using a TCP base d API be the better choice vs a http syntax method)? |
| 04:55.53 | SeRi | p3nguin: so I can use the auto answer and the phone wont ring it would just activate? |
| 04:56.26 | p3nguin | I don't know if there's any ring at all, but when you send the correct SIP header, it auto answers. |
| 04:56.47 | p3nguin | On my Ciscos, the ring before answer is configurable. |
| 04:57.14 | irroot | in the asterisk cli with debug / verbose |
| 04:57.23 | irroot | module load XXXX.so |
| 04:57.29 | irroot | what error it show |
| 04:57.46 | SeRi | I contacted the seller I hope for two things... That he comes back and say. It does work with asterisk! or Sure no harm no foul here is your money back :) |
| 04:57.54 | p3nguin | Did you request an undo? |
| 04:58.06 | SeRi | how? |
| 04:58.16 | p3nguin | Beg, I guess. |
| 04:58.26 | SeRi | O yes I did |
| 04:58.28 | SeRi | lol |
| 05:01.56 | SeRi | p3nguin: http://pastebin.com/nA4jPkTG what you think? |
| 05:02.16 | SeRi | p3nguin: I also looked at the sellers policie and he accepts returns within 7 days |
| 05:02.26 | p3nguin | line 33 = not right |
| 05:02.29 | SeRi | so maybe he would cancell... |
| 05:02.40 | p3nguin | You're at 0 days, so I'd say so. |
| 05:02.56 | SeRi | lol |
| 05:04.17 | p3nguin | http://pastebin.com/BsZqRisb |
| 05:04.27 | dijib | load app_swift.so |
| 05:06.26 | SeRi | Thanks p3nguin! |
| 05:07.25 | p3nguin | Try it against your Polycom. |
| 05:09.59 | SeRi | p3nguin: I will in a min |
| 05:10.05 | SeRi | have the brother on the phone now |
| 05:10.49 | dijib | could anybody point me in the right direction for app_swift |
| 05:11.08 | dijib | ive compiled, checked permissions, rechecked permissions, still wont load |
| 05:11.49 | SeRi | [TK]D-Fender: you around? |
| 05:12.26 | SeRi | [TK]D-Fender: You know anything about this conf phone? http://www.ebay.com/itm/330647987873?ssPageName=STRK:MEWNX:IT&_trksid=p3984.m1497.l2649#ht_500wt_1413 |
| 05:14.51 | *** join/#asterisk _N1X_ (~z03r0c00l@41.34.221.51) |
| 05:15.15 | _N1X_ | hello all |
| 05:15.52 | vader-- | what is the average timeframe to port a number? I am using flowroute and attempting to port two numbers from verizon |
| 05:16.12 | _N1X_ | how can i set the username:pass@provider in call file (auto dialing) |
| 05:17.48 | *** join/#asterisk timahvo1 (~rogue@197.176.207.234) |
| 05:19.38 | p3nguin | vader--: up to 3 weeks |
| 05:19.50 | p3nguin | Usually more like a week or less. |
| 05:20.06 | p3nguin | Probably 2-3 days. |
| 05:20.15 | dijib | hey p3nguin i just checked that sync issue i was having with the mixmon and its corrected itself |
| 05:20.18 | dijib | i dont know |
| 05:20.39 | vader-- | will they tell you when it will cut over? |
| 05:20.48 | vader-- | just trying to figure out for loss of service issues or what not |
| 05:23.15 | p3nguin | Preconfigure your system. It'll just start working, and then you'll get an email saying it has completed. |
| 05:27.04 | vader-- | true |
| 05:28.50 | *** join/#asterisk WiretapSeven (~Wiretap@unaffiliated/wiretap) |
| 05:31.00 | vader-- | is it me or do the polycom phone's interfaces seem weird |
| 05:31.16 | vader-- | i just installed a few SoundPoint 335's and their interfaces were tricky |
| 05:31.23 | vader-- | not intuitive |
| 05:36.25 | SeRi | nut can be a pita! |
| 05:36.31 | SeRi | well all working again |
| 05:37.32 | SeRi | p3nguin: !!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!1 |
| 05:37.36 | SeRi | Its cancelled! |
| 05:37.38 | SeRi | :D |
| 05:37.54 | SeRi | he said that I was right... no workie with sip :( |
| 05:37.54 | p3nguin | Nice. |
| 05:38.22 | p3nguin | Maybe you should try for one of the 330s or something. |
| 05:38.46 | p3nguin | Or a SoundStation that has IP in the model number? |
| 05:38.58 | SeRi | yea he said he will send me a list of what supports sip that he sells |
| 05:42.16 | p3nguin | If you search for polycom and set a price limit of $50, you'll see several phones. |
| 05:44.09 | SeRi | Yea I just saw a 501 .... the only problem is 501 can not be wall mounted |
| 05:44.27 | p3nguin | :/ |
| 05:45.44 | SeRi | http://www.ebay.com/itm/Polycom-SoundPoint-IP-321-SIP-Phone-/250939878312?pt=LH_DefaultDomain_0&hash=item3a6d2eaba8#ht_500wt_1180 |
| 05:45.56 | SeRi | the 321 comes witha wall mount option |
| 05:47.34 | SeRi | http://www.ebay.com/itm/Polycom-SoundPoint-IP-335-Telephone-2-Line-New-Box-/180764742973?pt=LH_DefaultDomain_0&hash=item2a166b153d#ht_500wt_1413 <---- new in box? |
| 05:50.03 | p3nguin | That's what it says. |
| 05:50.10 | p3nguin | You're not going to get it for $50, though. |
| 05:50.23 | p3nguin | It's already at 46 and has over three days left. |
| 05:50.33 | p3nguin | plus shipping. |
| 05:51.08 | SeRi | true. |
| 05:53.46 | p3nguin | "If you took this drug and suffered blood clots and died, call us right now." |
| 05:54.37 | vader-- | i know you guys don't like freepbx questions, but with freepbx installed do you happen to know where the template for the voicemail email is? |
| 05:55.05 | p3nguin | Sure. #freepbx |
| 05:55.17 | vader-- | hehe ya i know, i asked in there but no one seems to be around |
| 05:55.26 | p3nguin | Since, you know, we don't support FreePBX here, and all. |
| 05:55.33 | vader-- | i know :-) |
| 05:55.38 | vader-- | was worth a shot |
| 05:56.07 | p3nguin | When someone shows up there, they will answer you. |
| 05:56.26 | p3nguin | Actually, you did get an answer. |
| 05:56.34 | SeRi | p3nguin: I dial *001003 and the call failed with no such number in context phones |
| 05:56.40 | vader-- | ya it's not in the gui |
| 05:56.47 | *** join/#asterisk gajini (~root@61.12.17.170) |
| 05:57.10 | p3nguin | seri: You have internal included in phones? |
| 05:57.14 | SeRi | p3nguin: to extension '*001003' rejected because extension not found in context 'phones'. |
| 05:57.17 | SeRi | Yes I do |
| 05:57.20 | p3nguin | And you did dialplan reload? |
| 05:57.24 | SeRi | yes |
| 05:57.46 | p3nguin | What if you try *00 only? |
| 05:58.06 | SeRi | one sec |
| 05:58.58 | s[X] | and boom goes the dynamite |
| 05:59.00 | SeRi | p3nguin: that worked. I got a beep but no dial or auto answer |
| 05:59.16 | p3nguin | If you enter 1003 after the beep, what happened? |
| 05:59.47 | SeRi | right when I dail 1 it hangs up |
| 05:59.53 | SeRi | it does not let me finish |
| 05:59.57 | p3nguin | hmm |
| 06:00.10 | SeRi | Invalid extension '1', but no rule 'i' or 'e' in context 'intercom' |
| 06:00.59 | p3nguin | Very weird. |
| 06:02.05 | gajini | Hi, I have configured PRI card with asterisk, Hw can i make dialplan to get dialtone if i press any prefix number? |
| 06:03.57 | p3nguin | What does "dialplan show *001003@internal" show you? |
| 06:04.41 | SeRi | failed. |
| 06:06.33 | p3nguin | I guess you have an error in your dial plan. |
| 06:07.37 | SeRi | on the context. everything else works :) |
| 06:07.41 | p3nguin | Here's mine: http://pastebin.com/wCuSSesb |
| 06:09.56 | p3nguin | edited, reload. |
| 06:10.20 | SeRi | I did the XXXX which is the only difference |
| 06:10.55 | p3nguin | And it doesn't work? |
| 06:11.33 | SeRi | exten => _*00XXXX,1,Goto(intercom,${EXTEN:3},1); |
| 06:11.55 | SeRi | fails hard... lol well one sec |
| 06:12.02 | p3nguin | Okay, so if you entered *001003, it would go to 1003 in intercom. |
| 06:14.00 | SeRi | p3nguin: ok typo |
| 06:14.05 | SeRi | got it now |
| 06:14.10 | p3nguin | Yeah? What did you typo? |
| 06:14.19 | SeRi | missed one X on intercom |
| 06:14.38 | p3nguin | You see... I pasted EXACTLY what you needed. All you have to do is COPY AND PASTE. |
| 06:14.44 | p3nguin | There's no typo in a copy and paste. |
| 06:14.57 | SeRi | sorry but it was yours and didnt work p3nguin |
| 06:15.14 | SeRi | I had to change it from _*00NXX |
| 06:15.20 | SeRi | to _*00XXXX |
| 06:15.27 | p3nguin | I didn't give you NXX. |
| 06:15.34 | SeRi | yes you did. |
| 06:15.40 | p3nguin | No, I didn't. |
| 06:15.52 | p3nguin | Did you look at http://pastebin.com/BsZqRisb when I gave it to you? |
| 06:16.57 | SeRi | yes |
| 06:17.15 | SeRi | I copy pasta that |
| 06:17.28 | p3nguin | I don't think you did. |
| 06:17.37 | p3nguin | If you did, you would not have had NXX in it. |
| 06:18.03 | SeRi | well fuck.... lol maybe I confused it with another window... I did had all of them open |
| 06:18.11 | p3nguin | http://pastebin.com/skBcpUD9 |
| 06:18.17 | p3nguin | That's what I gave you. |
| 06:18.54 | *** join/#asterisk gurra (~gurra__@unaffiliated/gurra) |
| 06:18.56 | SeRi | wow I didnt see that p3nguin |
| 06:19.01 | p3nguin | Apparently. |
| 06:19.06 | SeRi | I guess you are right I didnt see that pb |
| 06:19.28 | p3nguin | (2302.26) <p3nguin> line 33 = not right |
| 06:19.29 | p3nguin | (2304.17) <p3nguin> http://pastebin.com/BsZqRisb |
| 06:19.31 | p3nguin | (2307.25) <p3nguin> Try it against your Polycom. |
| 06:21.12 | SeRi | ahhh lol |
| 06:21.14 | SeRi | ok |
| 06:21.16 | SeRi | sorry |
| 06:21.32 | SeRi | it works but it wont auto answer |
| 06:21.38 | p3nguin | No auto answer? |
| 06:21.53 | p3nguin | You may have to enable it in the phone, I'm not sure. |
| 06:22.11 | SeRi | ok one sec |
| 06:22.24 | p3nguin | There's also an older method: |
| 06:22.34 | p3nguin | SIPAddHeader(Alert-Info: Ring Answer) |
| 06:23.19 | p3nguin | It might be a good idea to consult voip-info on the matter. |
| 06:23.30 | SeRi | I found the option :) |
| 06:23.43 | SeRi | http://www.freepbx.org/support/documentation/module-documentation/paging-and-intercom |
| 06:25.01 | SeRi | now this onlu auto answers if asterisk tells it to right? |
| 06:25.56 | p3nguin | As far as I know, if you don't send the alert info, it won't auto answer. |
| 06:26.55 | p3nguin | Since I don't have any Polycom phones, I can't say with 100% certainty what does what and how it works. |
| 06:27.35 | p3nguin | Ask me about Cisco 7940/7960 with SCCP, and you'll get much more definitive answers. |
| 06:27.41 | SeRi | got me a new mouse today.... Logitech Performance MX |
| 06:28.09 | *** join/#asterisk sawgood (~sawgood@173-13-158-27-sfba.hfc.comcastbusiness.net) |
| 06:28.19 | SeRi | replacing my old ass MX Revolution 1000 |
| 06:28.25 | p3nguin | Dial(${DEVICE}/aa=2wc) <-- auto answer, 2-way, reject with congestion. |
| 06:28.56 | p3nguin | I don't even know what model my mouse is. |
| 06:30.16 | p3nguin | That freepbx doc you referred to says no intercom... |
| 06:30.33 | p3nguin | Why would paging work but intercom not work? |
| 06:32.02 | SeRi | not sure I sort of ignored it becuase I dont see why it wouldnt work |
| 06:32.24 | p3nguin | For my phones, I can use 1w for paging and 2w for intercom. |
| 06:32.31 | p3nguin | 1-way or 2-way |
| 06:32.36 | [TK]D-Fender | SeRi: that conf phone is a Lucent system Polycom. Which should mean that it uses Lucent (Avaya) standard digital signalling and is unusable for most commodity gear |
| 06:33.20 | SeRi | [TK]D-Fender: Thanks for the info. I cancel the order and we are in route now to a new sip phone |
| 06:34.06 | p3nguin | Are you waiting for the phone to reboot with the new setting? |
| 06:34.24 | SeRi | p3nguin: It did and did not worked |
| 06:34.47 | p3nguin | :S |
| 06:35.08 | p3nguin | Callcentric is happy to announce that we have released an Android app. |
| 06:35.18 | p3nguin | Screw you, callcentric. Screw you. |
| 06:35.57 | SeRi | rofl! |
| 06:36.03 | SeRi | I didnt even bother |
| 06:37.22 | SeRi | callcentric = next vonage |
| 06:37.45 | p3nguin | I'm starting to hate this D-Link phone. |
| 06:38.03 | SeRi | oh oh..... |
| 06:38.10 | p3nguin | It's offline more than it is online. And when it's online, the battery goes dead all the damn time. |
| 06:38.10 | *** join/#asterisk gerhard7 (~gerhard7@82-171-103-215.ip.telfort.nl) |
| 06:38.30 | SeRi | :( |
| 06:38.59 | SeRi | ok so it does not auto answer |
| 06:39.16 | p3nguin | And the guy basically lied on the listing. Said brand new, never used, but when I got it, it had a password on the network profile and it had IP addresses in the SIP account. |
| 06:39.34 | SeRi | bastard! |
| 06:39.37 | SeRi | I hate that shit |
| 06:40.01 | s[X] | wants a nice SIP phone |
| 06:44.23 | p3nguin | I think a lot of people have that same wish list. |
| 06:44.34 | [TK]D-Fender | The d-link and linksys Wifi phones. Suck. Hard. ALL OF THEM |
| 06:44.42 | s[X] | yeah i wouldnt touch either of them |
| 06:44.54 | s[X] | I wanted to get a Cisco until someone shot them down |
| 06:44.54 | p3nguin | It was cheap and I wanted to give it a try. |
| 06:45.02 | [TK]D-Fender | Do you feel dirty? |
| 06:45.10 | p3nguin | Nah. |
| 06:45.17 | [TK]D-Fender | It'll come :) |
| 06:45.18 | p3nguin | Just kind of ripped off a little. |
| 06:45.33 | p3nguin | I was looking for a 7925G, but I didn't want to spend that much on a phone I don't really need. |
| 06:46.00 | [TK]D-Fender | Might be better to get an Aastra w/ DECT |
| 06:46.14 | *** join/#asterisk oej (~olle@87.96.134.129) |
| 06:46.21 | p3nguin | I had to pass on a 7920G becaues they are only wirless B and don't do WPA2. |
| 06:46.22 | [TK]D-Fender | I use a top end model for my warehouse shipping manager |
| 06:46.37 | [TK]D-Fender | Yeah, that blows |
| 06:46.59 | p3nguin | I'm not going to reduce my speeds and security just for a silly wifi phone. |
| 06:47.03 | [TK]D-Fender | And on that note... beed time... |
| 06:47.08 | [TK]D-Fender | bed * |
| 06:47.18 | [TK]D-Fender | 'Nite all |
| 06:47.27 | s[X] | [TK]D-Fender, Own company ? |
| 06:47.28 | SeRi | g/n |
| 06:47.38 | [TK]D-Fender | s[X]: ? |
| 06:47.38 | s[X] | Night [TK]D-Fender |
| 06:47.41 | SeRi | p3nguin: I think I found the issue |
| 06:47.49 | p3nguin | Monkeys? |
| 06:47.54 | s[X] | When you said warehouse shipping manager, i was just curious |
| 06:48.07 | [TK]D-Fender | s[X]: Oh, no, not my own company, just where I'm employed |
| 06:48.18 | s[X] | We are in e-commerce |
| 06:48.20 | s[X] | yourself ? |
| 06:48.27 | [TK]D-Fender | the range on the DECT handset is just kinda sick |
| 06:48.48 | p3nguin | If I could find one for under $50, I'd consider it. |
| 06:48.57 | [TK]D-Fender | at least a 60,000 sqft warehouse full of steel and concrete |
| 06:49.03 | [TK]D-Fender | p3nguin: Yeah, tall order |
| 06:49.09 | [TK]D-Fender | ok, definitely out now... |
| 06:49.24 | p3nguin | The ones I looked at started around $150-ish. |
| 06:52.50 | kikohnl | It's nice to work for a Cisco partner, we just got some 7945G's < $150 each Not for Retail, internal use only |
| 06:53.25 | kikohnl | work much better than the Polycom 560's |
| 06:55.28 | SeRi | p3nguin: I fixed the auto answer |
| 06:55.37 | p3nguin | And the problem was... |
| 06:55.56 | SeRi | same => n(autoanswer),SIPAddHeader(Alert-Info: Ring Answer); This is for the Polycoms |
| 06:56.10 | p3nguin | So you needed the old method. |
| 06:56.15 | SeRi | Yes Sr |
| 06:56.31 | p3nguin | Makes sense, considering it's an older phone. |
| 06:56.35 | SeRi | I did some research and appertnly the new method is not supported in old polycoms |
| 06:56.46 | SeRi | p3nguin: indeed |
| 06:56.56 | p3nguin | That's why I also told you about the old method. |
| 06:57.16 | SeRi | yes and thats why I went and consulted Dr.voip-infp |
| 06:57.30 | SeRi | s/voip-infp/voip-info/ |
| 06:58.06 | SeRi | as you adviced |
| 06:59.34 | p3nguin | Does it have 2-way audio when you intercom it? |
| 07:00.28 | SeRi | yes |
| 07:00.31 | p3nguin | Nice. |
| 07:00.37 | SeRi | indeed very nice |
| 07:01.18 | SeRi | I could buy another 501 and tape the hand set and just use the speaker and buy the wall mount option at voiplink.com |
| 07:01.41 | SeRi | lol getthoriged |
| 07:01.52 | *** join/#asterisk xpot (~xpot@166-70-100-198.ip.xmission.com) |
| 07:01.52 | p3nguin | You can't find an SoundStation IP something? |
| 07:02.30 | SeRi | let me try |
| 07:02.50 | p3nguin | Even an old Cisco conf phone, which is made by Polycom, would be okay. |
| 07:02.59 | p3nguin | Like a 7935 or something. |
| 07:03.29 | p3nguin | Cisco IP Conference Station 7935 |
| 07:04.11 | p3nguin | WHAT?! 1 new from $1,499.00 (amazon.com) |
| 07:04.39 | p3nguin | Good lord that's a lot of money. |
| 07:04.41 | SeRi | lol |
| 07:04.48 | SeRi | http://www.ebay.com/itm/Cisco-Polycom-IP-CP-7935-Conference-Station-Telephone-Microphone-/360412970361?pt=LH_DefaultDomain_0&hash=item53ea497d79#ht_1682wt_1165 <---- thoughts? |
| 07:06.04 | p3nguin | http://www.ebay.com/itm/Cisco-Polycom-IP-Conference-Station-7935-CP-7935-Conference-VOIP-IP-Phone-/120817056807?pt=LH_DefaultDomain_0&hash=item1c2141fc27 |
| 07:06.23 | p3nguin | Yours says for parts not working. |
| 07:07.49 | *** join/#asterisk mnicholson (~mnicholso@nat/digium/x-tlawacfyvfoiyzha) |
| 07:07.49 | *** mode/#asterisk [+o mnicholson] by ChanServ |
| 07:07.50 | SeRi | ah true |
| 07:08.17 | p3nguin | I'm also not sure if there is a SIP firmware for it. It may only do SCCP. |
| 07:08.22 | SeRi | mhhh for that much I can buy a new polycom 321.. |
| 07:08.26 | p3nguin | Yeah. |
| 07:08.43 | SeRi | ah fuck it.... Ill just buy the damn 321 |
| 07:09.59 | SeRi | https://www.voiplink.com/ProductDetails.asp?ProductCode=POLYCOM-321&CartID=1 |
| 07:14.32 | p3nguin | I'd buy one on ebay and save my money. |
| 07:14.49 | p3nguin | not made of money |
| 07:15.48 | SeRi | me and you both.... |
| 07:16.10 | p3nguin | http://cgi.ebay.com/ws/eBayISAPI.dll?ViewItem&item=250939878312 |
| 07:16.58 | p3nguin | http://cgi.ebay.com/ws/eBayISAPI.dll?ViewItem&item=140649279488 |
| 07:17.22 | *** join/#asterisk Ad-Hoc (~nimbus@194.30.241.105) |
| 07:19.03 | SeRi | for the first one shipping is 20.00 |
| 07:19.12 | SeRi | I am going to place a bid right at 30.00 and see |
| 07:19.55 | SeRi | I find the shipping outrageous.... |
| 07:21.19 | p3nguin | Here's your 501 wall mount bracket: http://cgi.ebay.com/ws/eBayISAPI.dll?ViewItem&item=260877061788 |
| 07:22.20 | p3nguin | And here's one cheaper: http://cgi.ebay.com/ws/eBayISAPI.dll?ViewItem&item=250928867712 |
| 07:22.34 | SeRi | yeap I seen it. the problem is that the handset does not hold so if you wall mount it it will fall.... |
| 07:23.02 | *** join/#asterisk ollii (~risker@2001:470:1f15:1384:62eb:69ff:fe31:b4) |
| 07:23.15 | SeRi | the 501 handset just sort of lay on top of the system there is nothing holding the phone besides a small gimp ass grove |
| 07:23.41 | p3nguin | There's a hook for the handset. |
| 07:24.49 | SeRi | on the base? |
| 07:25.22 | ollii | g'mornin |
| 07:26.38 | p3nguin | Someone was just talking about that a few days ago. |
| 07:27.00 | p3nguin | Said you have to poke something through somewhere to make it come up. |
| 07:27.17 | p3nguin | still not a Polycom user |
| 07:27.59 | *** join/#asterisk c0rnoTa (~c0rnoTa@78.24.154.190) |
| 07:28.06 | SeRi | got it |
| 07:28.52 | p3nguin | Check out this: http://www.ebay.com/itm/Polycom-2201-01900-001-IP-Soundstation-Premier-/330637077755?pt=LH_DefaultDomain_0&hash=item4cfb816cfb |
| 07:29.34 | *** join/#asterisk schmidts (~schmidts@Lo20-rt01.lm33.sil.at) |
| 07:29.36 | schmidts | good morning |
| 07:29.47 | p3nguin | It says IP, so I don't know if that means SIP or not. |
| 07:31.03 | *** part/#asterisk c0rnoTa (~c0rnoTa@78.24.154.190) |
| 07:32.38 | SeRi | It should |
| 07:33.27 | p3nguin | They don't know, so you'd have to look it up. |
| 07:34.34 | kikohnl | I dont think they sip unless they are SoundPoint IP |
| 07:37.02 | p3nguin | It's a conference phone, so it's going to be a SoundStation. |
| 07:37.22 | p3nguin | And it says IP. |
| 07:37.23 | kikohnl | sorry that's what I meant |
| 07:38.05 | p3nguin | But it doesn't say IP6000 or IP4000 or whatever normal SIP conference phone models they have. |
| 07:38.07 | *** join/#asterisk god-x6 (~poorelanc@funtoo/user/godmachine-x6) |
| 07:38.24 | p3nguin | Google might know, but I don't. |
| 07:39.51 | SeRi | I am looking at it..... I dont like the "Sold as is" |
| 07:40.13 | SeRi | I place a bid on the last two I am going to wait and see. |
| 07:40.30 | SeRi | p3nguin: Thanks for the help.... it's been exausting |
| 07:40.37 | SeRi | did I use it's right? |
| 07:40.42 | p3nguin | YES! |
| 07:40.45 | p3nguin | :) |
| 07:40.48 | SeRi | HA! |
| 07:40.51 | SeRi | :D |
| 07:41.19 | p3nguin | it's = it is, it has |
| 07:41.40 | SeRi | Yes Sr. :) |
| 07:41.44 | *** join/#asterisk mirco (~mirco@p5B283710.dip.t-dialin.net) |
| 07:43.48 | SeRi | brb going for something to eat... hungry |
| 07:44.00 | SeRi | s/to/to get/ |
| 07:44.10 | SeRi | rofl |
| 07:44.19 | SeRi | That was all sorts of fucked up |
| 07:44.53 | *** join/#asterisk qakhan (~qakhan@182.185.131.154) |
| 07:45.22 | qakhan | hi all |
| 07:46.29 | qakhan | i want to setup timing in dialplan, 8AM to 8PM call can be go to a queue, 8pm to 8am calls go to voice mail |
| 07:47.40 | ChannelZ | use GotoIfTime |
| 07:48.35 | qakhan | plz send me some example |
| 07:49.44 | dym | Does anyone know why i have the functions SendFAX and RecieveFAX in 1.8 available, even though i compiled with spandsp and its support? I'm missing rxfax and txfax |
| 07:50.13 | p3nguin | Those aren't functions, they're applications. |
| 07:50.48 | p3nguin | Did you turn off res_fax and turn on app_fax? |
| 07:50.54 | ChannelZ | GotoIfTime(08:00-20:00,*,*,*?gotoqueue:gotovoicemail) where gotoqueue and gotovoicemail are priorities in the same exten which can do whatever (see Goto for other ways to jump to other contexts/extens) |
| 07:51.40 | *** join/#asterisk bjornts (~BTS@it010226.klientdrift.uib.no) |
| 07:52.43 | qakhan | ok thank you very much Channelz |
| 07:52.56 | qakhan | you always help :) |
| 07:52.56 | ChannelZ | sure |
| 07:54.38 | *** join/#asterisk gavimobile (~user@bzq-84-108-104-165.cablep.bezeqint.net) |
| 07:54.42 | SeRi | I cant drink my night medicine and it sucks not been able today :( |
| 07:54.42 | gavimobile | I have an sbs server running behind a router. the sbs server gets mail and receives mail *@mysampledomain.com. I would like to a SIP (asterisks) server behind the router as well. if I use the name sip.mysampledomain.com for my hostname, will I have networking problems with my network or servers? |
| 07:55.13 | dym | gavimobile: I dont see why. |
| 07:55.17 | dym | So: No. |
| 07:55.34 | dym | You will have to utilize NAT though |
| 07:55.50 | dym | Is the router your gateway to the internet? |
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| 08:04.19 | gavimobile | dym: my router is the gateway for wireless devices and servers only |
| 08:04.39 | gavimobile | but the telephones and the workstations all get their ip from the sbs server |
| 08:05.14 | gavimobile | when you say "utilize nat" you are refering to the asterisks conf settings (I think its sip_nat.conf) |
| 08:12.24 | *** join/#asterisk binbash_ (~peter@server.digitog.nl) |
| 08:12.25 | ChannelZ | the question is does your Asterisk server have a real IP or is the router doing NAT/masquerading |
| 08:17.12 | ollii | hey |
| 08:17.20 | ollii | i have a question about queues and holdtime |
| 08:18.01 | ollii | what is excatly described by "holdtime" ? is that an average count? |
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| 08:28.53 | ChannelZ | As a status, yes. |
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| 08:45.31 | arekm | hello, what does this mean? -- Channel 0/30, span 4 received AOC-E charging 30 units |
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| 08:52.47 | bulkorok | arkem: https://wiki.asterisk.org/wiki/display/AST/Advice+of+Charge |
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| 08:57.18 | bulkorok | oh... sorry arekm |
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| 08:59.14 | arekm | bulkorok: thanks. I wonder what "unit" it is here then |
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| 09:30.15 | dym | How can i view the options a Dialplan Application uses? |
| 09:30.48 | singler | core show application <application name> |
| 09:30.58 | singler | same valid to functions too |
| 09:31.38 | dym | negatory |
| 09:31.45 | dym | core show application Recievefax fails |
| 09:34.32 | kaldemar | "Recieve" vs. "Receive" |
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| 09:35.04 | ChannelZ | "I before E except before Fax" |
| 09:35.11 | kaldemar | if it is spelled correctly and still there is no output, the providing module is not loaded. |
| 09:35.45 | singler | also you can TAB-complete names, if no beginning is given then all apps/functions will be showed |
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| 09:52.00 | *** join/#asterisk elliot98 (~elliot@unaffiliated/elliot98) |
| 09:52.22 | elliot98 | on DTMFs, it states "duration 0 ms" and sends out an emulated DTMF |
| 09:52.37 | elliot98 | is this correct? Isn't there some sort of minimum duration set up? |
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| 10:44.01 | apten | hi |
| 10:44.36 | apten | i encouter issues with res_odbc and an external mysql server |
| 10:45.23 | apten | if the server is available at start-up and then stops beeing available just during run-time of the asterisk there doesn't seem to be a timeout writing cdr |
| 10:45.52 | apten | the dialplan seems to work fine but actually nothing happes (e.g. connecting to a queue) |
| 10:46.24 | apten | it seems that asterisk is waiting for the odbc connection to become available again but blocks any other operation |
| 10:46.35 | apten | any ideas? |
| 10:47.09 | apten | asterisk is 1.8.7 |
| 10:53.10 | schmidts | apten if asterisk couldnt reach your database server the call is blocked until it can reach it |
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| 11:12.02 | apten | @schmidts yes |
| 11:12.34 | *** join/#asterisk gavimobile (~user@bzq-84-108-104-165.cablep.bezeqint.net) |
| 11:12.55 | apten | but there is a connection timeout configured - so that shouldn't happen |
| 11:13.08 | apten | at least from my point of view |
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| 11:18.22 | schmidts | apten i dont use odbc but i had this problem with normal mysql too |
| 11:18.46 | schmidts | does the call hang at the connection or when doing a query? |
| 11:20.50 | BlackBishop | anyone here using a gs ht503 ? :) I'm trying it to at least make it get into an extension upon an incoming call so I can do stuff but .. nothing .. |
| 11:22.06 | jacc0 | it would not be so smart to allaw calls when database connection is failing; you would miss CDR and can't bill the user for the call |
| 11:22.33 | apten | looking at the cli & full log i can't actually see any details because it's about writing cdr. so by looking at CLI it just the call which hangs - but probably in the background the process writing log/cdr hangs |
| 11:23.40 | apten | but anyway i don't see any reason why writing of cdr should block a call |
| 11:30.13 | apten | @jacc0: but it can't be an good idea either to block incoming calls just because it's not possible to write cdr right now |
| 11:33.27 | dym | why doesnt _12345XX match 1234566 ? |
| 11:34.42 | WIMPy | Because you wrote it in the wrong context? |
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| 11:34.56 | dym | why would you hurt me? :( |
| 11:34.59 | dym | i surely did not |
| 11:35.32 | WIMPy | Show us |
| 11:37.24 | puzzled | hi WIMPy |
| 11:38.11 | dym | http://pastebin.com/EZJPX3ne |
| 11:38.12 | dym | there! |
| 11:41.12 | singler | wrong syntax |
| 11:41.30 | WIMPy | No syntax? |
| 11:42.29 | jacc0 | exten => 2._12345XX,1,NoOp(*** Hi! ***) |
| 11:42.34 | jacc0 | sorry |
| 11:42.40 | jacc0 | exten => _12345XX,1,NoOp(*** Hi! ***) |
| 11:42.54 | jacc0 | you should put exten => in front of it |
| 11:43.01 | dym | well |
| 11:43.08 | dym | rofl |
| 11:43.13 | dym | head - desk |
| 11:43.14 | dym | yeah |
| 11:43.18 | dym | mr obvious strikes again |
| 11:43.20 | dym | thanks |
| 11:43.25 | jacc0 | WIMPy can you hurt him one more time ? :P |
| 11:43.27 | dym | thank GOD its friday |
| 11:43.31 | dym | NOOOO :( |
| 11:43.34 | jacc0 | :P |
| 11:43.42 | jacc0 | jk |
| 11:43.56 | WIMPy | Let me choose the right LART... |
| 11:45.02 | dym | actually |
| 11:45.05 | dym | still same error message... |
| 11:45.08 | dym | oddmuch |
| 11:45.48 | apten | @dym: you did reload the dialplan? ;) |
| 11:46.06 | WIMPy | It will work when my parcel has arrived. |
| 11:46.49 | *** join/#asterisk gavimobile (~user@bzq-84-108-104-165.cablep.bezeqint.net) |
| 11:46.59 | WIMPy | And did you save it before reloading it? |
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| 11:51.40 | singler | And did you put it in right context this time? :) |
| 11:51.59 | WIMPy | And are you logged in to the right box? |
| 11:52.16 | singler | :) |
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| 12:11.24 | *** join/#asterisk joobie (~joobie@unaffiliated/moo0o0ooo00o0o0o) |
| 12:11.39 | joobie | hey guys, is there a shortcut key to rbeoot the linksys spa942? |
| 12:11.49 | joobie | like polycom 321 has volumedown, volumeup, speaker, hold |
| 12:12.08 | WIMPy | Like pulling the plug? |
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| 12:17.29 | jacc0 | http://forum.voxilla.com/cisco-linksys-sipura-support-forum/spa2102-unlock-admin-login-hard-reset-23017.html |
| 12:17.36 | jacc0 | you could try that |
| 12:17.50 | BlackBishop | can I make a call from the cli ? |
| 12:19.47 | WIMPy | BlackBishop: 'channel originate ...' |
| 12:20.17 | BlackBishop | mhm |
| 12:21.56 | plundra | joobie: Yes. |
| 12:22.49 | ollii | console dial xy@context is also usable...but with sangoma there was once a bug where the whole system was freezed after a console dial |
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| 12:25.05 | plundra | joobie: Ok that was shitty help, can't find it :-) But there is a combo, I'm sure of it. |
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| 12:27.58 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
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| 12:37.22 | jacc0 | @ollii: Sangoma no longer supports asterisk 1.8.x at all : they tell you to use SMGv3 |
| 12:38.04 | WIMPy | Huh? I thought that was the old way of doing it? |
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| 12:41.34 | elliot98 | hello |
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| 12:45.33 | elliot98 | how does one check if a Digium card is using DTMF detection? |
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| 13:05.23 | orn | elliot98: The DTMF problems you're having is with a Digium card? |
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| 13:07.58 | [koss] | are there any asterisk distros with auto provisining out of the box like Druid had? |
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| 13:19.14 | BlackBishop | I'll be damned if I understand how to set up this ht503 thingy :) |
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| 13:21.17 | ijpalmer | hello all, I'm on *1.8.5 and using sip realtime, the problem is that when I restart Asterisk I have to reboot all the phones to get them to register again, is there a way around this. Thanks |
| 13:21.22 | voipeng | is there a command to see how many g729 concurent calls your license provides? It says i have 20 channels but im not sure how many concurrent calls are available |
| 13:21.28 | [TK]D-Fender | [koss], The FreePBX ISO comes with EPM |
| 13:22.12 | leifmadsen | ijpalmer: uhhhhh.... set the registration timeout to be shorter |
| 13:22.22 | leifmadsen | it's probably at 3600 seconds (1 hour) by default) |
| 13:22.40 | BlackBishop | anyone any idea on how can I make the HT503 forward all the incoming calls to incoming_number@mysip ? :/ as in .. If it gets a call from xxxxxxx ( where each x is a number from 0 to 9 :)) ) to xxxxxxx@mysipserver .. |
| 13:22.41 | leifmadsen | at least that's what polycoms are set at by default -- I usually change it to 120 seconds |
| 13:24.12 | ijpalmer | leifmadsen: thanks, you're rigt it is set to 3600, I'll change it as you suggest |
| 13:24.26 | BlackBishop | or .. how could I make a call through it ? :/ I got the cable that comes from my telco in the "Line" ( FXS !? ) port |
| 13:25.13 | leifmadsen | ijpalmer: rebooting the phones was unnecessary -- you just weren't patient enough :) |
| 13:26.19 | ijpalmer | leifmadsen: you're absolutely right, why does it all seem so obvious once you've spoken with someone else |
| 13:26.57 | *** join/#asterisk cVsup (~cVsup@187.127.102.184) |
| 13:26.59 | cVsup | <PROTECTED> |
| 13:28.44 | [TK]D-Fender | BlackBishop, http://www.fonality.com/trixbox/forums/trixbox-forums/trunks/howto-set-pstn-trunk-grandstream-ht-503-tb-ce-280 |
| 13:30.15 | [TK]D-Fender | cVsup, Please rephrase your question and provide more detail |
| 13:31.19 | BlackBishop | doesn't look like extensions.conf and other file setup to me :/ |
| 13:31.58 | *** join/#asterisk PoWeRKiLL (~powerkill@77.125.87.36) |
| 13:32.09 | ollii | jacc0: really?! they told us to use wanpipe 3.5.2x with dahdi support and smg v3 is no longer supported |
| 13:32.33 | ollii | [koss]: try gemeinschaft |
| 13:32.50 | ollii | im not quite sure, but they have auto provisioning |
| 13:33.19 | cVsup | BlackBishop: i have wctdm interface with 3 fxo and one fxs |
| 13:33.42 | [TK]D-Fender | BlackBishop, Look closer |
| 13:33.56 | cVsup | i need set fxs as extension |
| 13:35.22 | [TK]D-Fender | cVsup, exten => 100,1,Dial(DAHDI/4,30) |
| 13:35.29 | fprior | Hi all: can you help me to simplify this dialplan part ? pastebin.com/SrcQyDtK |
| 13:35.48 | [TK]D-Fender | cVsup, There... you now have an extension to dial the FXS port (in this case assuming it's on port 4) |
| 13:36.35 | BlackBishop | [TK]D-Fender: trying, trying, don't get how .. |
| 13:36.52 | [TK]D-Fender | BlackBishop, Their trunk info = sip.conf |
| 13:37.13 | [TK]D-Fender | fprior, "core show application macro" <- |
| 13:37.20 | ollii | exten => _XXX.,1,Dial(SIP/spa400b/L3${EXTEN},300,t) |
| 13:37.23 | *** join/#asterisk mjordan (~mjordan@nat/digium/x-bohxxizvymjlpomd) |
| 13:37.24 | ollii | fprior: |
| 13:37.29 | ollii | "." after _XXX |
| 13:37.38 | [TK]D-Fender | ollii, No. |
| 13:38.40 | *** part/#asterisk ijpalmer (~IceChat7@host81-137-172-233.in-addr.btopenworld.com) |
| 13:38.40 | [TK]D-Fender | removes access to only half of what he wants, and allows a nearly infinite amount of what he has not stated as wanting. |
| 13:39.16 | cVsup | [TK]D-Fender: http://pastebin.com/h52uYAQN |
| 13:39.21 | cVsup | my dahdi sets |
| 13:40.54 | [TK]D-Fender | cVsup, That defines the ports, but doesn't configure * to use them. |
| 13:41.24 | [TK]D-Fender | cVsup, And you're running Elastix. This is a ZAP/DAHDI Extension in the GUI |
| 13:41.39 | [TK]D-Fender | cVsup, and as I mentioned, not the kind of thing supported here. |
| 13:41.59 | cVsup | [TK]D-Fender: i need help with dahdi sets |
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| 13:42.17 | [TK]D-Fender | cVsup, that file looks fine.. its the others you need to look at |
| 13:42.36 | [TK]D-Fender | cVsup, chan_dahdi.conf is what defines what and how * will use your card |
| 13:46.45 | cVsup | [TK]D-Fender: for fxs i need create extension ZAP or Dahdi? |
| 13:46.57 | elliot98 | orn: well, does digium have onboard DTMF detection? |
| 13:51.29 | [TK]D-Fender | cVsup, Yes |
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| 13:54.34 | nfi|ermes | on Centos 5.7 with kernel 2.6.18-274.el5 i installed asterisk 1.8.7.1 and dahdi-linux-complete 2.5.0. But i receive error FATAL: Module wctdm not found. on /etc/init.d/dahdi start . Is there a compatibility issue between dahdi 2.5 and kernel 2.6.18-274.el5 ? |
| 13:54.54 | BlackBishop | [TK]D-Fender: neah, I still don't get it .. :| |
| 13:55.05 | [TK]D-Fender | BlackBishop, What don't you get? |
| 13:55.20 | fprior | [TK]D-Fender: now, with macro, result: http://pastebin.com/40V2q7rp (is possible reduce the code ?) |
| 13:56.10 | [TK]D-Fender | fprior, 1 macro.. 8 lines. You did this backwards |
| 13:56.52 | *** part/#asterisk gajini (~root@61.12.17.170) |
| 13:56.53 | BlackBishop | so, step 1 is to create a sip trunk, so I created "[314110724] context=from-trunk host=dynamic type=friend port=5062" |
| 13:57.39 | BlackBishop | then under the fxo port section of ht503's web interface I made those settings... |
| 13:59.07 | BlackBishop | now, I don't get about that inbound route ( creating and assigning ) |
| 13:59.47 | [TK]D-Fender | that is your dialplan |
| 13:59.59 | BlackBishop | in .. extensions.ael ? |
| 14:00.11 | [TK]D-Fender | make an extension to match the [pstn muber] they refer to |
| 14:00.17 | [TK]D-Fender | AEL is best forgotten |
| 14:00.39 | *** join/#asterisk gavimobile (~user@bzq-218-196-30.red.bezeqint.net) |
| 14:00.53 | orn | I often wonder whether anyone is using it |
| 14:00.54 | BlackBishop | I like it ! :/ it's a hell of alot easier for me to write it there than the default way of doing stuff .. |
| 14:01.08 | BlackBishop | because "line numbering" remembers me of z80 days |
| 14:01.34 | *** join/#asterisk FiReSTaRT (~dlyh@unaffiliated/firestart) |
| 14:01.34 | orn | i agree, it seems like a better way to go, but once you've gotten used to the other method it's just hard to get yourself to switch |
| 14:02.04 | BlackBishop | I switched in like 20 minutes to ael everything that took days to write in the default way |
| 14:02.16 | BlackBishop | so .. it's ok :) |
| 14:02.16 | [TK]D-Fender | AEL parses back to standard logic which is hard to compare when debugging, has limitations as to what you can do because of its nature. |
| 14:02.27 | BlackBishop | debuging it isn't a problem... |
| 14:02.41 | singler | I also use AEL, it is easier to use more advanced stuff (if, loops, etc), also syntax checked weeds out some stupid mistakes |
| 14:02.44 | BlackBishop | so far didn't get into the limitation of what I can do because I'm not doing that complex stuff |
| 14:03.08 | singler | also no need to repeat extension on each line |
| 14:03.09 | BlackBishop | yeah, the loops and ifs are awesome |
| 14:03.15 | BlackBishop | that too. |
| 14:03.52 | [TK]D-Fender | I ahven't seen a dialplan that really calls for anything like that yet |
| 14:03.53 | *** join/#asterisk gerhard7 (~gerhard7@82-171-103-215.ip.telfort.nl) |
| 14:03.58 | [TK]D-Fender | but "whatever" |
| 14:04.26 | *** join/#asterisk malcolmd (~malcolmd@pdpc/sponsor/digium/malcolmd) |
| 14:04.26 | *** mode/#asterisk [+o malcolmd] by ChanServ |
| 14:04.35 | orn | Yeah, I haven't run into anything that I need to do that I can do with AEL and can't with the standard one |
| 14:04.57 | [TK]D-Fender | orn, You never will. As I said, it can only do less. |
| 14:05.09 | BlackBishop | well, to each itsown .. ael seems alot easier to me ! :) |
| 14:05.13 | *** join/#asterisk serafie (~erin@nat/digium/x-mwqfrvxlyxzseqoe) |
| 14:05.27 | BlackBishop | having more code done in php .. it looks like it and I like it :) |
| 14:05.36 | orn | If I'm doing really complex stuff anyway, I usually resort to AGIs |
| 14:07.41 | singler | AEL is useful not only for complex stuff, for example if's. in standart way you need to use GotoIf, labels, maybe additional goto (if you need else part), adding another if case to same exten may be more difficult |
| 14:08.03 | singler | because you need to account for all jumps in exten |
| 14:08.12 | nfi|ermes | on Centos 5.7 with kernel 2.6.18-274.el5 i installed asterisk 1.8.7.1 and dahdi-linux-complete 2.5.0. But i receive error FATAL: Module wctdm not found. on /etc/init.d/dahdi start . Is there a compatibility issue between dahdi 2.5 and kernel 2.6.18-274.el5 ? |
| 14:08.23 | fprior | [TK]D-Fender: yeah, now http://pastebin.com/xJr39Q2b |
| 14:08.24 | singler | it's like ASM ;) |
| 14:08.49 | *** join/#asterisk kamikazemicrowav (~kamikazem@seraph.contegix.com) |
| 14:09.14 | [TK]D-Fender | singler, I wrote a language once... looked like a cross of Pascal & ASM :) |
| 14:09.28 | *** join/#asterisk knorkeknie (~hans@p5496C94C.dip.t-dialin.net) |
| 14:09.39 | knorkeknie | hi there |
| 14:10.03 | WIMPy | nfi|ermes: There is no compatibility with kernel modules. They need to exactely match your kernel. |
| 14:10.24 | [TK]D-Fender | fprior, Looks like you're catching on... |
| 14:10.47 | BlackBishop | [TK]D-Fender: when I call .. I see it as Call from '314110724' (86.121.76.232:5062) to extension '314110724 |
| 14:10.59 | nfi|ermes | so, shoud i install dahdi-linux-complete 2.6 ? |
| 14:11.01 | *** join/#asterisk filo1234 (~filo@unaffiliated/filo1234) |
| 14:11.02 | BlackBishop | I want to see the incoming number too :/ |
| 14:11.11 | BlackBishop | as in, who is actually calling ! |
| 14:11.24 | [TK]D-Fender | BlackBishop, You probably have it |
| 14:11.43 | filo1234 | hi |
| 14:12.21 | BlackBishop | in what ?! |
| 14:12.32 | WIMPy | nfi|ermes: You need the dahdi modules for EXACTELY your kernel. The dahdi version doesn't really matter. |
| 14:12.48 | [TK]D-Fender | fprior, Working nicely now? |
| 14:12.58 | [TK]D-Fender | fprior, Sure looks a lot cleaner, doesn't it? |
| 14:13.14 | [TK]D-Fender | BlackBishop, ...... CALLER ID <- |
| 14:13.45 | nfi|ermes | WIMPy, where can i find the dahdi module for 2.6.18-274.el5 ? |
| 14:14.21 | BlackBishop | I don't understand .. :| |
| 14:14.32 | WIMPy | nfi|ermes: You need to ask p3nguin, he likes package manager. I have no theory as to how that could work without building it yourself. |
| 14:14.34 | fprior | [TK]D-Fender: yes, thanks |
| 14:14.36 | [TK]D-Fender | nfi|ermes, Did you modprobe it? Have you potentially upgraded kernels since you isntalled DAHDI? |
| 14:15.13 | BlackBishop | 314110724 is the type=friend I created, I don't see the number I'm calling from ( my cell ) anywhere in the log .. |
| 14:15.36 | [TK]D-Fender | BlackBishop, What don't you understand? When the devices sends the call to * it targets a number and the caller ID = the CALLER ID |
| 14:16.05 | [TK]D-Fender | BlackBishop, When I yell "Hey John!" from across the room, I am calling John.. I am not saying that *I* am John. |
| 14:16.28 | [TK]D-Fender | Other sides name = CALLERID, not the extension they dial. |
| 14:16.55 | BlackBishop | well, it's not sending the call to * .. it's sending it to 314110724@mysip ! |
| 14:16.56 | BlackBishop | :| |
| 14:17.11 | [TK]D-Fender | to extension '314110724 <------- where you told it to |
| 14:17.25 | nfi|ermes | [TK]D-Fender, i have not upgraded, it's a frssh installation and i builded dahdi from source; no error building |
| 14:17.45 | BlackBishop | so there's no way get it to extension 'incoming_number' :| |
| 14:17.55 | [TK]D-Fender | Call from '314110724' <--- this might mean you didn't set up the HT to grab callerid. Hard to say. did you tell it to wait enough rings to get it? |
| 14:17.57 | BlackBishop | if I'm calling from 1298731927319 I want it to call 1298731927319 |
| 14:18.12 | [TK]D-Fender | <BlackBishop> if I'm calling from 1298731927319 I want it to call 1298731927319 <- never happening |
| 14:18.23 | BlackBishop | why not ? :/ |
| 14:18.34 | [TK]D-Fender | BlackBishop, The call is to a fixed target. the target is never the caller's number. that is the CALLERID, not the EXTENSION |
| 14:18.36 | BlackBishop | to call 1298731927319@mysip ! |
| 14:18.42 | [TK]D-Fender | ..... |
| 14:18.45 | singler | BlackBishop: did you check CALLERID(num) variable? |
| 14:18.47 | WIMPy | nfi|ermes: Why didn;t you tell us, you did them yourself? What happens if you modprobe them? |
| 14:18.50 | [TK]D-Fender | that is the ctarget, not the CALLERID |
| 14:20.01 | nfi|ermes | [root@centralino dahdi-linux-complete-2.4.1.2+2.4.1]# modprobe wctdm |
| 14:20.01 | nfi|ermes | FATAL: Module wctdm not found. |
| 14:20.15 | BlackBishop | so I can't make it send the call from 1298731927319 to 1298731927319@mysip :( |
| 14:20.17 | nfi|ermes | [root@centralino dahdi-linux-complete-2.4.1.2+2.4.1]# modprobe dahdi |
| 14:20.17 | nfi|ermes | FATAL: Module dahdi not found. |
| 14:20.25 | *** join/#asterisk elred_ (~elred_@unaffiliated/elred-/x-5010831) |
| 14:20.44 | nfi|ermes | insmod /lib/modules/2.6.18-274.12.1.el5/dahdi/wctdm.ko |
| 14:20.44 | nfi|ermes | insmod: error inserting '/lib/modules/2.6.18-274.12.1.el5/dahdi/wctdm.ko': -1 Unknown symbol in module |
| 14:20.46 | WIMPy | nfi|ermes: Did you install them? |
| 14:20.48 | singler | nfi|ermes: did you made make install for dahdi? |
| 14:20.59 | nfi|ermes | of course |
| 14:21.32 | WIMPy | nfi|ermes: Looks like the kernel source you used is not that of the kernel you're running. |
| 14:21.54 | nfi|ermes | uname -r |
| 14:21.55 | nfi|ermes | 2.6.18-274.el5 |
| 14:22.28 | [TK]D-Fender | BlackBishop, the inbound extension is never ther caller's phone number. When you dial from a SIP phone you set up as [100] do they only dial 100? |
| 14:22.50 | [TK]D-Fender | BlackBishop, You don't seem to be comprehending the difference between who is calling and who they are calling. |
| 14:23.10 | WIMPy | nfi|ermes: Where is the "12.1"? |
| 14:23.22 | *** join/#asterisk felimwhiteley (~quassel@46.7.101.58) |
| 14:23.26 | nfi|ermes | :| |
| 14:24.06 | [TK]D-Fender | BlackBishop, If you have 10 phone numbers and one of those if for "customer service". You'll want to know that the caller is dialing that customer service phone # so you can process them properly. Th call comes in targeting YOUR customer service number. not the caller's phone number. Othewise yuo have to knwo everybody phone number... Like the entire planet |
| 14:24.59 | [TK]D-Fender | BlackBishop, the call is to the phone number attached to the line you plugged into the HT503. THAT is what was called. the CALLER's name an number are int he CALLER ID of the call. |
| 14:25.32 | *** join/#asterisk asteriskATmarmuD (~Mundt@193.158.65.23) |
| 14:26.06 | asteriskATmarmuD | hi guys. I am looking for a solution for fax detection. came across nvfaxdetect. what is best for outgoing call fax detection |
| 14:26.07 | BlackBishop | or, you don't understand what I'm trying to do .. |
| 14:26.11 | asteriskATmarmuD | thanks in advance |
| 14:26.34 | BlackBishop | when a call gets into the HT503, I want it to forward that call to extension with the same number in my asterisk |
| 14:26.43 | [TK]D-Fender | BlackBishop, it sends to one fixed number there is no "option" |
| 14:26.53 | BlackBishop | :/ |
| 14:26.55 | BlackBishop | mhm.. |
| 14:27.02 | [TK]D-Fender | BlackBishop, No device has that option |
| 14:27.19 | [TK]D-Fender | BlackBishop, You have the caller id jump BASED on that. |
| 14:27.38 | [TK]D-Fender | BlackBishop, Take it and do what you want with it, but the initial target is a fixed number |
| 14:27.51 | BlackBishop | ahuh .. |
| 14:29.11 | *** join/#asterisk nighty- (~nighty@static-68-179-124-161.ptr.terago.net) |
| 14:29.11 | [TK]D-Fender | BlackBishop, Also.. that line of CLI doesn't actually prove what the callerID was on that call. |
| 14:29.14 | SeRi | good morning |
| 14:29.20 | [TK]D-Fender | BlackBishop, It could have actually been right |
| 14:29.34 | [TK]D-Fender | BlackBishop, But we haven't gotten to look at a complete call |
| 14:32.51 | *** join/#asterisk devmikey (~irc@unaffiliated/devmikey) |
| 14:35.40 | *** join/#asterisk irroot (~gregory@196-210-140-50.dynamic.isadsl.co.za) |
| 14:37.58 | BlackBishop | [2011-12-02 16:36:57] NOTICE[28063]: chan_sip.c:22866 handle_request_invite: Call from '314110724' (86.121.76.232:5062) to extension '314110724' rejected because extension not found in context 'from-trunk'. |
| 14:38.21 | BlackBishop | context from-trunk { 314110724 => { Hangup(); }; }; |
| 14:38.29 | [TK]D-Fender | Extension does not exist as it says. |
| 14:38.39 | *** join/#asterisk gavimobile (~user@bzq-218-196-30.red.bezeqint.net) |
| 14:39.41 | [TK]D-Fender | Something is wrong with your AEL. Here comes that "debugging" part I was talking about... |
| 14:40.10 | BlackBishop | it's a damn simple context .. shouldn't need debugging ! :) |
| 14:40.27 | [TK]D-Fender | BlackBishop, Perhaps you should pastebin the whole mess and show me it being loaded |
| 14:41.29 | *** join/#asterisk gavimobile (~user@bzq-218-196-30.red.bezeqint.net) |
| 14:41.31 | BlackBishop | I just showed it to ya', for this test, I deleted everything in the extensions.ael and placed just this thing .. |
| 14:42.07 | BlackBishop | Added a Verbose(Incoming CALL from ${CALLERID(num)}); now in the 314110724 => { ... }; |
| 14:42.47 | nfi|ermes | thx so much WIMPy and [TK]D-Fender !!! |
| 14:42.54 | nfi|ermes | now it's working |
| 14:43.20 | *** part/#asterisk wesphillips (~wphill04@adsl-75-53-136-233.dsl.hstntx.sbcglobal.net) |
| 14:43.49 | BlackBishop | http://pastebin.com/bt95yjCs |
| 14:44.39 | BlackBishop | http://pastebin.com/hH9E3cmN ( including the loading part ) |
| 14:45.54 | TheCops | 314110724 is 8 caracther and _XXXXXXXXX => { is 9 ? |
| 14:46.18 | [TK]D-Fender | BlackBishop, Doesn't look like the "not found" you showed earlier |
| 14:46.19 | TheCops | oh no |
| 14:46.23 | TheCops | sorry hehe |
| 14:46.31 | TheCops | forgot the 3 :) |
| 14:46.43 | BlackBishop | restarted asterisk |
| 14:46.50 | [TK]D-Fender | BlackBishop, Sure loks like the call is making it in. |
| 14:46.50 | BlackBishop | I only did the "dialplan reload" 'till now :/ |
| 14:47.00 | TheCops | lol |
| 14:47.03 | BlackBishop | but should it show the incomming call at each ring !? |
| 14:47.07 | BlackBishop | :| I only called once ! |
| 14:47.34 | [TK]D-Fender | ~gs |
| 14:47.34 | infobot | GrandSuck phones & gateways are cheap junk which should be avoided with extreme prejudice. |
| 14:47.54 | *** join/#asterisk Juggie (~Juggie@unaffiliated/juggie) |
| 14:47.56 | TheCops | lol |
| 14:48.33 | BlackBishop | will do next time. |
| 14:48.51 | BlackBishop | a linksys spa_something will be commin' next month |
| 14:48.57 | mandla | irroot: its a friday! |
| 14:49.55 | mandla | Its a friday Asterisk GURU's! All of you can come to my house for some wine/brandy/whiskey |
| 14:51.43 | BlackBishop | besides the fact that I get that verbose like 5 times ( untill I hang up ) .. it doesn't show up the caller id .. which the grandstream should relay ! |
| 14:52.05 | BlackBishop | ~linksys |
| 14:52.05 | infobot | from memory, linksys is a tool of satan |
| 14:52.19 | BlackBishop | what is recommended then !? :/ |
| 14:55.13 | [TK]D-Fender | BlackBishop, Answer before you hangup. |
| 14:55.31 | [TK]D-Fender | BlackBishop, change the number of rings before forwarding, etc |
| 14:56.24 | *** join/#asterisk sysreq (~sysreq@unaffiliated/sysreq) |
| 14:57.05 | BlackBishop | I want them forwarded immediately .. why should I wait before forwarding ? |
| 14:57.20 | BlackBishop | and for this test .. to work, I decided to autodeny the call, why should I answer ? :| |
| 14:57.59 | *** join/#asterisk gavimobile (~user@bzq-218-196-30.red.bezeqint.net) |
| 14:58.05 | [TK]D-Fender | to get rid of them. |
| 14:58.14 | [TK]D-Fender | So just ignore it till it goes away |
| 14:58.21 | [TK]D-Fender | Since you're effetively doing that anyway |
| 14:58.40 | BlackBishop | No, I'm not ignoring it .. I want to deny it ! |
| 14:58.43 | BlackBishop | ow, fsck ! |
| 14:58.47 | BlackBishop | I remember ! |
| 14:58.55 | BlackBishop | you can't deny it like you do on a normal phone |
| 14:58.56 | BlackBishop | :| |
| 14:59.01 | BlackBishop | I mean, cell phone |
| 14:59.02 | BlackBishop | :| |
| 14:59.21 | BlackBishop | so true ! |
| 14:59.51 | [TK]D-Fender | Helps when you remember what you're dealing with. Analog has no "reject" button. |
| 15:00.00 | BlackBishop | yeah |
| 15:00.14 | BlackBishop | now to figure out the caller id thing ! :/ |
| 15:00.24 | BlackBishop | that verbose I put there should show me the number, right ? |
| 15:02.28 | [TK]D-Fender | If the CID is provided and the HT was configured and able to pick it up |
| 15:04.03 | BlackBishop | "Caller ID Scheme:" .. what should I select for Romania !? :| |
| 15:05.13 | [TK]D-Fender | Not being from anywhere near there... who knows |
| 15:05.49 | *** join/#asterisk gavimobile (~user@bzq-218-196-30.red.bezeqint.net) |
| 15:05.53 | [TK]D-Fender | As a Romanian :) |
| 15:05.59 | [TK]D-Fender | Ask* |
| 15:06.24 | SeRi | God give me strength to continue dealing with comcast. Amen. |
| 15:06.25 | *** join/#asterisk fisted_ (~fisted@unaffiliated/fisted) |
| 15:07.12 | SeRi | ^^ Thats about right with comcast |
| 15:07.40 | MrTelephone | I just won the most worst programmer of the year award |
| 15:07.59 | MrTelephone | is now known as mr spaghetti |
| 15:08.23 | [TK]D-Fender | MrTelephone, Pastafarianism may be right for you... |
| 15:08.55 | MrTelephone | maybe it is because I don't plan. too much cut and pasting and less thinking about consolidating functions and subs |
| 15:09.28 | MrTelephone | is there a good perl editor for linux? |
| 15:10.01 | MrTelephone | it would be nice if there was something to show you when your missing brackets and whatever instead of executing to see the errors |
| 15:10.22 | *** join/#asterisk joshaidan (~brianj@24.109.210.41) |
| 15:13.59 | SeRi | MrTelephone: There is some perl validators out there |
| 15:16.38 | SeRi | well looks like I wont the auction. |
| 15:16.54 | *** join/#asterisk gavimobile (~user@bzq-218-196-30.red.bezeqint.net) |
| 15:16.59 | WIMPy | What did you get? |
| 15:17.00 | SeRi | Polycom SP IP 321 |
| 15:17.06 | SeRi | 24 dollars |
| 15:17.22 | SeRi | perefect phone for the kids room |
| 15:18.25 | BlackBishop | [TK]D-Fender: the thing is that .. it should at least show unknown .. not the name of my own extension ! :| |
| 15:18.31 | SeRi | it's wall mountable and has auto answer and speaker phone. Is all I need :) |
| 15:19.22 | BlackBishop | I'm calling from 07something .. not 314110724 .. |
| 15:19.53 | BlackBishop | 314110724 is the number of the line plugged in the HT .. and the name of the extension I assigned to it to forward calls to my asterisk |
| 15:20.24 | *** join/#asterisk gavimobile (~user@bzq-218-196-30.red.bezeqint.net) |
| 15:20.47 | [TK]D-Fender | BlackBishop, tht doesn't address any of my previous points... |
| 15:21.12 | [TK]D-Fender | BlackBishop, And you still don't seem to understand that no devices dials the CALLER'S phone number into * as the extnsion |
| 15:21.31 | BlackBishop | Ok, I got that part ! |
| 15:21.37 | [TK]D-Fender | BlackBishop, When I dial from my SIP phone with a callerID of 100, do I use exten => 100,1, for calls from that phone? No |
| 15:21.44 | [TK]D-Fender | BlackBishop, You don't seem to have. |
| 15:21.45 | BlackBishop | but I want to see in asterisk the caller id the handytone forwards the call from ! |
| 15:22.15 | [TK]D-Fender | BlackBishop, Maybe it isn't picking it up... because of ... I dunno.. having no idea how to interpret the signalling on a Romanian phone line <- |
| 15:22.29 | BlackBishop | but if it isn't picking it up .. shouldn't it show it as unknown !? |
| 15:22.32 | BlackBishop | or '' ! |
| 15:22.49 | [TK]D-Fender | BlackBishop, Maybethat's just the way this thing works. |
| 15:23.01 | [TK]D-Fender | BlackBishop, Maybe you configured something wrong on it... hard to say. |
| 15:23.20 | WIMPy | BlackBishop: What's your setup like and what is it you don't like? |
| 15:23.51 | [TK]D-Fender | WIMPy, HT503 for PSTN access, Romanian phone line, no callerID. Ask at your own peril :p |
| 15:24.27 | WIMPy | None? Oh, I just read that as the wrong wone. |
| 15:24.33 | BlackBishop | WIMPy: http://pastebin.com/hH9E3cmN , trying to make asterisk see the caller id of the incomming call in the HT503 |
| 15:25.01 | *** join/#asterisk zerohalo (~zerohalo@74-94-141-204-ma.hfc.comcastbusiness.net) |
| 15:25.03 | BlackBishop | "Incoming CALL from 314110724" isn't the right one ofcourse ! :| |
| 15:25.28 | BlackBishop | the ht503 has the 314110724 sip user id registered on the fxo port |
| 15:25.44 | [TK]D-Fender | BlackBishop, You've described it as looping through every ring. In north America CID arrives between the 1st and second ring which means you need to wait at least 2 full ringins before answering to get it |
| 15:25.56 | WIMPy | Ok, so it's about the devices configuration. |
| 15:26.04 | [TK]D-Fender | BlackBishop, and look at the SIP DEBUG for this call to see if it is in there anywhere |
| 15:26.10 | BlackBishop | and under the basic settings page Unconditional Call Forward to VOIP: 314110724@my_asteriskbox |
| 15:26.17 | MrTelephone | You guys have polycoms for your house? that's crazy |
| 15:26.18 | MrTelephone | lol |
| 15:26.49 | WIMPy | Yes, sip debug is a good idea. Maybe it's using PAI. |
| 15:26.55 | [TK]D-Fender | No... using Polycom is an investment. Grandstream is crazy :p |
| 15:27.05 | BlackBishop | sip set debug on |
| 15:27.06 | BlackBishop | lets see |
| 15:27.25 | *** join/#asterisk puzzled (~patrick@2001:980:5e31:1:6ef0:49ff:fe50:659d) |
| 15:27.58 | *** join/#asterisk KavanS (~KavanS@LINBIT/KavanS) |
| 15:28.10 | *** join/#asterisk jollie (~james@174-22-74-215.sxfl.qwest.net) |
| 15:28.55 | TheCops | MrTelephone, 335 is only 130$ |
| 15:29.11 | BlackBishop | http://pastebin.com/ZShvqduT |
| 15:29.27 | [TK]D-Fender | <SeRi> Polycom SP IP 321 - 24 dollars - perefect phone for the kids room <---- |
| 15:29.33 | TheCops | ya |
| 15:29.38 | Kobaz | i have some polycoms at my house |
| 15:29.47 | TheCops | hi koba |
| 15:29.48 | TheCops | z |
| 15:29.49 | TheCops | :) |
| 15:29.51 | Kobaz | like 6 |
| 15:29.56 | Kobaz | yello |
| 15:29.57 | SeRi | :) |
| 15:30.09 | TheCops | hehe i have a 670 at my desk with console to monitor some call center agent :p |
| 15:30.21 | TheCops | polycom phone rocks |
| 15:31.06 | [TK]D-Fender | BlackBishop, Nope, no callerID in there. Make sure it is set to wait 3 rings |
| 15:31.21 | *** join/#asterisk aidinb (~aidin@unaffiliated/aidinb) |
| 15:31.42 | WIMPy | Or maybe better get something decent. |
| 15:31.48 | BlackBishop | like what ? |
| 15:32.02 | *** join/#asterisk The_Boy_Wonder (~vossel@asterisk/batman-developer/dvossel) |
| 15:32.30 | WIMPy | Soemthing digital. |
| 15:33.26 | BlackBishop | well, I need something with the same features as this GS .. one FXS .. one FXO .. and the ability to use my asterisk for incomming calls in the small box .. and to call through it |
| 15:33.52 | [TK]D-Fender | BlackBishop, Go ask Grandstream support what scheme to use and if Romania's standards are even supported. |
| 15:34.54 | WIMPy | Get ISDN or VOIP. That will make your life a lot easier. |
| 15:35.30 | BlackBishop | I don't understand ... I'm talking about a hardware box, changing the provider line isn't an option ! |
| 15:35.50 | [TK]D-Fender | BlackBishop, Go ask Grandstream support what scheme to use and if Romania's standards are even supported. |
| 15:35.59 | BlackBishop | it's one of the 3 national providers here doubt there's a problem from them since an actual 10$ phone with caller id support shows the caller id ! |
| 15:36.10 | *** join/#asterisk putnopvut (~putnopvut@asterisk/master-of-queues/mmichelson) |
| 15:36.10 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 15:37.27 | BlackBishop | Ok, posted on the forums. |
| 15:37.57 | BlackBishop | google says romania uses etsi-fsk |
| 15:37.59 | [TK]D-Fender | http://www.fonality.com/trixbox/forums/vendor-forums-non-certified/grandstream/ht-503-fxo-port-trunk-asterisk-using-freepbx-front-en |
| 15:38.20 | [TK]D-Fender | http://www.google.ca/#sclient=psy-ab&hl=en&source=hp&q=HT503+callerid&pbx=1&oq=HT503+callerid&aq=f&aqi=g-v1&aql=&gs_sm=e&gs_upl=3685l3685l1l4818l1l1l0l0l0l0l338l338l3-1l1l0&bav=on.2,or.r_gc.r_pw.r_cp.,cf.osb&fp=e9f5a5308858f637&biw=1600&bih=927 |
| 15:38.21 | *** join/#asterisk LiuYan (~liu.yan@211.154.128.171) |
| 15:38.24 | [TK]D-Fender | Happy hunting... |
| 15:40.21 | *** join/#asterisk Naikrovek (~mbluth@unaffiliated/naikrovek) |
| 15:40.40 | BlackBishop | I don't think the problem is there though, I think I'm configuring it wrong |
| 15:40.59 | BlackBishop | asterisk sees as incomming the same sip user id the HT loggins with ! |
| 15:41.11 | [TK]D-Fender | Plenty of stated issues with the devie along with suggestions for tweaking. ICD is a PITA in many places |
| 15:41.14 | *** join/#asterisk gavimobile (~user@bzq-218-196-30.red.bezeqint.net) |
| 15:41.37 | [TK]D-Fender | BlackBishop, Yes, other more general setting may be wrong. Go read the manual check thier support chans, etc |
| 15:41.56 | [TK]D-Fender | BlackBishop, Nobody I know of who ever had one of these stuck with it |
| 15:42.10 | BlackBishop | what do you recommend gettig then ? |
| 15:42.37 | [TK]D-Fender | Linksys SPA-3102 is the next ste up |
| 15:43.37 | akrohn | has a spa3201 on my desk. they are nifty |
| 15:43.42 | *** join/#asterisk jkroon (~jkroon@dsl-241-252-251.telkomadsl.co.za) |
| 15:44.00 | akrohn | 3102* |
| 15:45.02 | *** join/#asterisk chazzam (~chazz@50-81-150-34.client.mchsi.com) |
| 15:45.13 | *** join/#asterisk BuenGenio (~Gene@cm61-15-218-59.hkcable.com.hk) |
| 15:45.48 | BlackBishop | [TK]D-Fender: you think that would make what I want easier ? :) |
| 15:45.53 | BlackBishop | and would probably work better ? |
| 15:46.12 | BlackBishop | or at least .. would it do what I want ? :)) |
| 15:47.12 | [TK]D-Fender | Better odds |
| 15:47.20 | [TK]D-Fender | I would always checkw ith the vendor first |
| 15:47.36 | *** join/#asterisk oej (~olle@87.96.134.129) |
| 15:52.02 | asteriskATmarmuD | how do you detect fax machines on outgoing calls? |
| 15:52.23 | BlackBishop | I don't :)) |
| 15:55.11 | leifmadsen | you could probably use M() or U() and execute some dialplan that did a Wait(3) or something with a fax extension, and handle the call there if a fax detection was found in the intial call setup, otherwise just return from the subroutine and handle the call normally |
| 15:55.56 | SeRi | leifmadsen: Did you get 2.3.x loaded on your phone? |
| 15:56.12 | tuxxie | I am looking for a guide to using the AMI. I have been unable to find what needs to be passed to commands. How can I find a list of commands with a detailed discriptions and a list of arguments that can be passed? |
| 15:56.36 | tuxx- | tuxxie: asterisk -rx "manager show commands" |
| 15:57.13 | tuxx- | for detailed information do asterisk -rx "manager show command <command>" |
| 15:58.26 | leifmadsen | SeRi: I have 2.3.4 installed, but no cyanogenmod because I have the SGH-T959P |
| 15:58.32 | leifmadsen | no firmware available for it |
| 15:58.45 | leifmadsen | lucky I didn't brick it while trying |
| 15:58.50 | leifmadsen | although it is now rooted |
| 15:59.01 | leifmadsen | and has the recovery software |
| 15:59.07 | tuxxie | gotcha thanks |
| 15:59.39 | SeRi | wow I see. cool! now you have unlocked another domension to your phone. Your phone has now become useful! :) |
| 15:59.53 | leifmadsen | SeRi: well, it being rooted does nothing for me :) |
| 16:00.26 | SeRi | You should be able to use application locked by your carrier from the market |
| 16:00.36 | SeRi | well only if your carrier had it that way |
| 16:00.38 | leifmadsen | I hadn't run into anything blocked |
| 16:00.48 | SeRi | I know AT&T does that |
| 16:00.55 | SeRi | ah Isee. |
| 16:00.58 | leifmadsen | ya, maybe Telus does, but I haven't found any software that was blocked |
| 16:01.06 | leifmadsen | maybe I am missing out :) |
| 16:01.19 | leifmadsen | now just need to figure out how to move applications to the SD card |
| 16:01.57 | SeRi | cool. settints ---- manage applications ----- lick on application and on the options move to sd card |
| 16:02.09 | SeRi | s/lick/click/ |
| 16:02.25 | *** part/#asterisk apten (~apten@carbon.gonicus.de) |
| 16:02.55 | leifmadsen | SeRi: ya just found it -- looks like most apps when I reinstalled put themselves on the SD anyways |
| 16:02.58 | leifmadsen | so yay for that |
| 16:03.09 | SeRi | lol nice |
| 16:05.36 | *** part/#asterisk asteriskATmarmuD (~Mundt@193.158.65.23) |
| 16:10.19 | *** join/#asterisk xSmurf (~MrSmurf@unaffiliated/mrsmurf) |
| 16:13.17 | *** part/#asterisk jollie (~james@174-22-74-215.sxfl.qwest.net) |
| 16:13.44 | *** join/#asterisk gavimobile (~user@bzq-218-196-30.red.bezeqint.net) |
| 16:15.20 | MrTelephone | it's -4 F outside |
| 16:15.25 | MrTelephone | I'm freezing my balls off |
| 16:16.34 | *** join/#asterisk gavimobile (~user@bzq-218-196-30.red.bezeqint.net) |
| 16:17.32 | *** join/#asterisk _Corey_ (~chatzilla@64.121.4.75) |
| 16:22.02 | SeRi | MrTelephone: where you at? |
| 16:22.42 | TheCops | very cold over here also |
| 16:22.51 | dijib | shit its the cops |
| 16:23.03 | TheCops | old joke |
| 16:23.15 | TheCops | :p |
| 16:23.26 | dijib | k shower ytime |
| 16:24.04 | *** join/#asterisk gavimobile (~user@bzq-218-196-30.red.bezeqint.net) |
| 16:25.46 | *** join/#asterisk tzafrir_laptop (~tzafrir@212.179.75.202) |
| 16:28.51 | tuxxie | I am using the AMI Action: CoreShowChannels, is there a way for me to limit the responces to only "Application: Queue" |
| 16:30.59 | *** join/#asterisk gavimobile (~user@bzq-218-196-30.red.bezeqint.net) |
| 16:41.21 | *** join/#asterisk gavimobile (~user@bzq-218-196-30.red.bezeqint.net) |
| 16:44.07 | *** join/#asterisk anthm (~anthm@freeswitch/developer/anthm) |
| 16:45.51 | *** join/#asterisk gavimobile (~user@bzq-218-196-30.red.bezeqint.net) |
| 16:49.06 | *** join/#asterisk jrose_atDigium (~jon@nat/digium/x-slyekteowtngarya) |
| 16:51.36 | *** join/#asterisk gavimobile (~user@bzq-218-196-30.red.bezeqint.net) |
| 16:52.33 | *** join/#asterisk krotos (~androirc@83.224.73.33) |
| 16:53.35 | krotos | Hi |
| 16:54.23 | krotos | I need |
| 16:54.36 | p3nguin | Type more words before pressing Enter. |
| 16:54.46 | [TK]D-Fender | </kirk> |
| 16:54.55 | *** join/#asterisk brdude (~brdude@12.155.183.30) |
| 16:55.11 | BlackBishop | [TK]D-Fender: P-Asserted-Identity: <sip:314110724@sip.sms1.ro> |
| 16:55.22 | BlackBishop | it seems that that's what it sends as an Id |
| 16:55.38 | BlackBishop | I selected "Caller ID Transport Type:" Relay via P-Asserted-Identity |
| 16:55.55 | p3nguin | Did I assert that in a Shatner voice? :) |
| 16:57.05 | [TK]D-Fender | p3nguin, I do a very good impression myself... |
| 16:57.11 | WIMPy | BlackBishop: What other options do you have? |
| 16:57.26 | SeRi | lol |
| 16:58.58 | BlackBishop | Relay via SIP From |
| 16:59.02 | BlackBishop | Send Anonymous |
| 16:59.05 | BlackBishop | Disable |
| 16:59.44 | WIMPy | From sounds like a good option. |
| 16:59.47 | tzanger | asterisk needs a SetIf() application. Most of my dialplan is GotoIf(${test}) stuff for setting |
| 17:00.09 | p3nguin | Just use ExecIf(?Set()) |
| 17:00.34 | [TK]D-Fender | tzanger, ExecIf |
| 17:00.47 | tzanger | Execif? that seems extreme |
| 17:01.02 | BlackBishop | WIMPy: yeah, it seems like the HT isn't sending the caller id right ! :| |
| 17:01.03 | p3nguin | If something, execute the set. How is that extreme? |
| 17:01.06 | krotos | I need to extract multiple remote party id headers from a call, but if i use sip_header(remote-party-id) I get only the first |
| 17:01.09 | [TK]D-Fender | tzanger, It's what we've got. |
| 17:01.15 | p3nguin | It's EXAXCTLY what you want to do. |
| 17:01.36 | [TK]D-Fender | p3nguin, No, he'd like to knock it back 1 level more... |
| 17:01.43 | [TK]D-Fender | p3nguin, But lets not get greedy :) |
| 17:01.44 | tzanger | [TK]D-Fender: no I hear you, it just seems... excessive to be able to encode dialplan applications "laterally" |
| 17:01.49 | tzanger | but yes that'd certainly work |
| 17:02.13 | p3nguin | I wish I would have thought of it. |
| 17:02.20 | WIMPy | krotos: You get the same header with different content in one message? |
| 17:02.26 | p3nguin | Oh, wait, I did. |
| 17:02.48 | [TK]D-Fender | tzanger, Exec is a parallel to If / Then / Else. SetIf would be a concept I've never seen in any language |
| 17:03.36 | tzanger | [TK]D-Fender: you've never used ternary operators? bar = (foo == 1) ? baz : quux |
| 17:03.44 | krotos | Is there any way to get the next rpid headers? Or the enteire sip header? |
| 17:03.52 | WIMPy | The dialplan is a language? |
| 17:03.52 | [TK]D-Fender | tzanger, Oh God.... |
| 17:04.13 | p3nguin | The dial plan is a language. |
| 17:04.19 | [TK]D-Fender | WIMPy, Technically, yes... |
| 17:04.38 | tzanger | [TK]D-Fender: how is that worse than bar = (foo == 1) ? doit(baz) : doit(quux) which is what execif is |
| 17:05.01 | [TK]D-Fender | Sure I've done better as a teen just entering college almost 20 years ago.. but that's besides the point :) |
| 17:05.33 | BlackBishop | I think I got it !!!!!! |
| 17:05.35 | BlackBishop | WHOOOHOOOO |
| 17:05.37 | *** join/#asterisk TimeRider (~steve@92.40.253.200.threembb.co.uk) |
| 17:06.36 | krotos | Wimpy, same header but different data ... |
| 17:06.54 | WIMPy | krotos: Sounds evil |
| 17:07.40 | p3nguin | ExecIf(?Set()) is your non-existent SetIf(), so use it or continue jumping around in dial plan unnecessarily. |
| 17:09.46 | krotos | The call come comes from a nortel ..and in the successive rpid headers contain the hop of a call that was redirected |
| 17:11.08 | *** join/#asterisk ferdna (~ferdna@cpe-67-10-220-35.elp.res.rr.com) |
| 17:14.10 | [TK]D-Fender | krotos, I don't believe there is a stock means of doing this. I'd take a look inside func_sip_header and see if the data store it hits has those available for parsing, and submit a patch to make pulling multiple records possible. |
| 17:16.13 | krotos | I m asking this Because during a sip dump i 've noticed this : 1 |
| 17:16.57 | *** join/#asterisk gavimobile (~user@bzq-218-196-30.red.bezeqint.net) |
| 17:18.44 | *** join/#asterisk dtcrshr (~datacrush@unaffiliated/datacrusher) |
| 17:19.03 | BlackBishop | and .. broke it again :| |
| 17:19.12 | BlackBishop | dunno what it was .. but for a couple of rings, I saw my number ! |
| 17:19.26 | *** join/#asterisk gavimobile (~user@bzq-218-196-30.red.bezeqint.net) |
| 17:21.17 | krotos | <PROTECTED> |
| 17:22.07 | *** join/#asterisk Cesar_B (~chatzilla@201.200.175.218) |
| 17:22.27 | BlackBishop | krotos: awesome ! |
| 17:22.52 | Cesar_B | hello to all, can anyone help a litlle bit with a ss7 problem? i dont understand what my telco its trying to do |
| 17:24.33 | Cesar_B | http://pastebin.com/FgzLyn4T |
| 17:25.50 | krotos | Ahshshs ..so strange!! If is not possible to extract multiples rpid headers., there is some way to get entire sip packet? I can invoke a script with this as param |
| 17:27.24 | [TK]D-Fender | krotos, as I said, look at the function's source |
| 17:28.01 | *** join/#asterisk wesphillips (~wphill04@137-237-233-124.harris.com) |
| 17:28.35 | krotos | Ok :) now i 'm from phone ..sorry if i write so slow |
| 17:30.23 | Cesar_B | people anyone from the presents know something about ss7 to help me with this issue? |
| 17:33.41 | BlackBishop | [TK]D-Fender, WIMPy : http://pastebin.com/wSEbLTVC |
| 17:33.56 | *** join/#asterisk gavimobile (~user@bzq-218-196-30.red.bezeqint.net) |
| 17:33.58 | BlackBishop | the From: is set right .. the first ring and a half, then the Hangup() comes in .. |
| 17:34.10 | BlackBishop | then it gets set wrong ( that is just one call there .. no answering .. no nothing ) |
| 17:34.54 | BlackBishop | so there is a bug in the firmware too .. I think .. |
| 17:35.25 | [TK]D-Fender | [2011-12-02 19:32:24] VERBOSE[30870] chan_sip.c: No matching peer for '0760905294' from '86.121.76.232:5062' |
| 17:35.35 | [TK]D-Fender | [2011-12-02 19:32:24] VERBOSE[30870] chan_sip.c: Looking for 314110724 in default (domain sip.sms1.ro:5060) |
| 17:35.40 | [TK]D-Fender | SIP/2.0 404 Not Found |
| 17:35.56 | BlackBishop | that's why it sets another from !? |
| 17:36.04 | [TK]D-Fender | Not matching your peer, therefor hitting the [general] context, not your peer, and landing in a place that has no target |
| 17:36.33 | [TK]D-Fender | That isn't "why" it sets it. |
| 17:36.47 | [TK]D-Fender | I do not yet see a reason to associate these 2 facts |
| 17:37.57 | BlackBishop | well, the lil' error about not finding the peer and stuff .. isn't a problem yet :) |
| 17:38.14 | BlackBishop | Just wondering why it fscks it up on the next rings after the hangup() |
| 17:38.41 | p3nguin | ring... after hangup? |
| 17:39.43 | [TK]D-Fender | Thre is no Hangup() |
| 17:39.47 | [TK]D-Fender | your dialplan isn't gettinghit |
| 17:40.03 | BlackBishop | ahm :| |
| 17:41.05 | *** join/#asterisk gavimobile (~user@bzq-218-196-30.red.bezeqint.net) |
| 17:41.07 | p3nguin | dijib: Are you on the conf today? |
| 17:41.26 | p3nguin | seri: You too? |
| 17:41.41 | SeRi | p3nguin: I I can jump in. |
| 17:42.32 | SeRi | p3nguin: wich one? |
| 17:42.55 | p3nguin | Every week, dijib says he didn't know about the VUC. But it's at the exact same time every single Friday. Today, I'm reminding him (an hour late). |
| 17:43.06 | p3nguin | The VUC, of course. |
| 17:43.36 | SeRi | LOL |
| 17:44.00 | p3nguin | 200901@login.zipdx.com |
| 17:44.00 | SeRi | VUC? |
| 17:44.06 | SeRi | ah! |
| 17:44.11 | SeRi | one sec |
| 17:44.20 | SeRi | are you in? |
| 17:44.37 | p3nguin | ~vuc |
| 17:44.37 | infobot | VUC is the VoIP Users Conference |
| 17:45.02 | p3nguin | I'm there, but I usually only listen. |
| 17:45.26 | p3nguin | Today, they are talking about PBX hacking. |
| 17:45.58 | p3nguin | The topic was AT&T's losses through hacked PBX services. |
| 17:46.00 | anonymouz666 | I'd talk if my english was good |
| 17:46.24 | SeRi | p3nguin: Nice! |
| 17:46.26 | SeRi | I am in. |
| 17:47.11 | gordonjcp | anonymouz666: I'm willing to bet that your English is better than most other people here's Portuguese |
| 17:47.41 | SeRi | p3nguin: what time does this starts? |
| 17:49.25 | *** part/#asterisk LiuYan1 (~LiuYan@222.125.130.16) |
| 17:49.34 | p3nguin | 11 AM EST every Friday. You can dial in up to 15 minutes early; I dial in at 10:50 automatically. |
| 17:50.10 | p3nguin | Wait. |
| 17:50.18 | p3nguin | 11 AM CST, sorry. |
| 17:50.37 | p3nguin | Forgot what TZ I am in for a minute. |
| 17:50.45 | SeRi | You auto dial? |
| 17:50.48 | SeRi | :) |
| 17:50.48 | p3nguin | Yes. |
| 17:51.21 | SeRi | will like to doa that. I will look in to it. |
| 17:51.40 | gordonjcp | p3nguin: sweet, so I can register a call to that and dial up from home? |
| 17:51.42 | p3nguin | It starts at 11 AM CST, noon EST, every Friday. |
| 17:51.55 | gordonjcp | what's CST? |
| 17:51.58 | BlackBishop | [TK]D-Fender: http://pastebin.com/iNuWRPMg .. notice the app_verbose.c |
| 17:52.04 | p3nguin | I don't know what "register a call" means. |
| 17:52.06 | gordonjcp | -5? |
| 17:52.13 | SeRi | -6 |
| 17:52.14 | p3nguin | GMT -6 |
| 17:52.19 | SeRi | ^^ |
| 17:52.23 | gordonjcp | ah, so starting in ten minutes |
| 17:52.30 | p3nguin | Started an hour ago. |
| 17:52.54 | SeRi | p3nguin: how can I talk? |
| 17:52.57 | gordonjcp | oh, 11am -6 hours, I see |
| 17:53.01 | p3nguin | Unmute. Talk. |
| 17:53.10 | SeRi | ok. |
| 17:53.53 | p3nguin | Topic: AT&T Fraud and Terrorism |
| 17:54.43 | [TK]D-Fender | BlackBishop, What about it? |
| 17:55.22 | BlackBishop | it gets in one context at first, then in another :| |
| 17:55.27 | p3nguin | seri: Are you dialed in from a 223 phone number? |
| 17:55.28 | BlackBishop | without me answering or doing anything at all |
| 17:56.28 | p3nguin | It's either that or a 1978 number. |
| 17:57.06 | [TK]D-Fender | BlackBishop, And you can see that it is coming in unauthed one time, and not on the next |
| 17:57.20 | [TK]D-Fender | BlackBishop, Along the way you are continuing to break your auth setup for your device |
| 17:58.16 | BlackBishop | by hanging up ? |
| 18:00.31 | BlackBishop | modified it to print the asserted identity too :| |
| 18:00.35 | BlackBishop | Incoming CALL from 314110724 - <sip:0760905294@sip.sms1.ro> |
| 18:00.39 | BlackBishop | Incoming CALL from 314110724 - <sip:314110724@sip.sms1.ro> |
| 18:00.41 | BlackBishop | :| |
| 18:04.17 | *** join/#asterisk gavimobile (~user@bzq-218-196-30.red.bezeqint.net) |
| 18:04.29 | Cesar_B | any ss7 expert here? |
| 18:05.39 | p3nguin | seri: Hey! Did you join MY conf rather than connecting to zipdx yourself? |
| 18:06.02 | WIMPy | Cesar_B: I'm not that observant here atm, but did you already ask a question? |
| 18:06.14 | p3nguin | Why'd you do that?! Weirdo! |
| 18:06.17 | Cesar_B | yes |
| 18:06.48 | Cesar_B | i dont know what its wrong in this: http://pastebin.com/FgzLyn4T |
| 18:06.56 | BlackBishop | same thing happens without the hangup(); too ! :)) |
| 18:07.37 | *** join/#asterisk gavimobile (~user@bzq-218-196-30.red.bezeqint.net) |
| 18:07.44 | BlackBishop | should it send that message each time it rings ? ( even without a hangup(); ? ) |
| 18:08.21 | SeRi | lol p3nguin |
| 18:08.25 | WIMPy | Cesar_B: It says "no route to specified transit network". Is that what you wanted to know? |
| 18:08.29 | p3nguin | Why would you do that? |
| 18:08.40 | p3nguin | (1143.59) <p3nguin> 200901@login.zipdx.com |
| 18:08.47 | Cesar_B | yes |
| 18:08.56 | Cesar_B | the telco its doing a test, that he says |
| 18:09.16 | Cesar_B | and when the telco do that, that its the output |
| 18:09.32 | Cesar_B | and they say "you have something wrong" |
| 18:10.18 | WIMPy | The routing or point code as it seems. |
| 18:10.19 | SeRi | p3nguin: I won the phone |
| 18:10.35 | Cesar_B | i m looking in google, and i see that the are doing its , LPA and CRC, Loop Back Acknowledgement, Continuity Check Request. |
| 18:10.48 | p3nguin | I redirected you to their conf. |
| 18:10.51 | Cesar_B | what WIMPy ? |
| 18:10.59 | p3nguin | Now I see you on the dashboard by name. |
| 18:11.09 | SeRi | p3nguin: Me? |
| 18:11.27 | p3nguin | yes |
| 18:11.33 | WIMPy | Cesar_B: Someone tries to talk to an unreachabel network. |
| 18:11.33 | SeRi | how did you do that? |
| 18:11.38 | p3nguin | magic |
| 18:11.43 | p3nguin | channel redirect ... |
| 18:11.49 | SeRi | you have access? |
| 18:12.04 | p3nguin | Of course. |
| 18:12.14 | SeRi | o shit. nice |
| 18:12.32 | SeRi | I notice I was moved. lol |
| 18:12.36 | SeRi | I was like wtf |
| 18:12.36 | Cesar_B | but they are doing a test, i supposed i need to answer that doing something like "test ok, dont do that again moron" jeje |
| 18:12.49 | p3nguin | What do yo mean you noticed? |
| 18:12.49 | Cesar_B | that its not a call, its a test over a voice channel |
| 18:13.10 | SeRi | I heard my self dialing in again and got the annoucement |
| 18:13.19 | p3nguin | hmm |
| 18:13.29 | p3nguin | I don't really understand. |
| 18:14.16 | SeRi | well when you did that I got some what disconnected and than the phone was ringing for a sec and than heard the systems options |
| 18:14.26 | SeRi | and I was in |
| 18:14.42 | p3nguin | I suggested that you called the vuc. Instead, you called me, so you did not appear in the dashboard. I didn't want you to appear as me, so I redirected you to zipdx directly. |
| 18:14.52 | SeRi | ah! |
| 18:14.55 | SeRi | wtf. |
| 18:14.57 | SeRi | really |
| 18:14.59 | SeRi | I called you? |
| 18:15.03 | p3nguin | Yes. |
| 18:15.09 | SeRi | well fuck.... Sorry! |
| 18:15.14 | SeRi | :/ dumb ass me |
| 18:15.16 | WIMPy | Cesar_B: I'm not into SS7, but I'd guess you don't have your point code correct. |
| 18:15.29 | SeRi | sorry man.... meds here |
| 18:15.42 | p3nguin | After I redirected you, then I see you in the dashboard as Your Name (1003). |
| 18:15.52 | *** join/#asterisk _Corey_ (~chatzilla@64.121.4.75) |
| 18:16.03 | SeRi | where is the dashboard? |
| 18:16.07 | Cesar_B | i confirmed with the carrier and the point code its correct , and the calls are good, inbound and outbound, WIMPy |
| 18:16.27 | Cesar_B | the only thing that are wrong its when the telco do that fucking test |
| 18:16.27 | p3nguin | zipdx.com |
| 18:16.46 | p3nguin | If you want to talk, you'll have to unmute yourself. |
| 18:16.49 | Cesar_B | and if i cannot solve this, they are shuting down the e1 ss7 lines |
| 18:16.54 | WIMPy | Cesar_B: Ah |
| 18:17.38 | *** join/#asterisk singler (~singler@84.15.129.49) |
| 18:17.45 | SeRi | p3nguin: got it. and sorry about that earlier |
| 18:17.57 | WIMPy | Sorry, but I definitely have no clue about what features you get. |
| 18:18.01 | p3nguin | Does your phone support wideband? |
| 18:18.15 | Cesar_B | not problem WIMPy thx anyway |
| 18:18.23 | SeRi | not sure.... |
| 18:18.23 | Cesar_B | you know who can hell me? |
| 18:18.27 | p3nguin | I'm connected to zipdx using g722, but you're connected through me using ulaw. |
| 18:18.30 | SeRi | satan? |
| 18:18.32 | *** join/#asterisk nix8n82 (~hmg@71-32-140-75.chyn.qwest.net) |
| 18:18.41 | p3nguin | heh |
| 18:19.10 | *** join/#asterisk leifmadsen (~Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 18:19.10 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 18:19.31 | SeRi | p3nguin: how can I change the codec just for them? |
| 18:19.42 | Cesar_B | help me, sorry |
| 18:19.48 | p3nguin | Why worry about that? Just use g722 all the time. |
| 18:20.08 | SeRi | p3nguin: Mhhhhh ok |
| 18:20.21 | SeRi | disallow all allow g722? |
| 18:20.27 | p3nguin | Surely your network where your phone and asterisk live isn't so saturated that g722 from one phone would bring it down. |
| 18:20.30 | p3nguin | yes |
| 18:20.52 | SeRi | does voip.ms supports it? |
| 18:20.57 | p3nguin | no |
| 18:21.08 | p3nguin | It wouldn't do any good for them to support it. |
| 18:21.16 | WIMPy | Cesar_B: I don't know who's in to that. I only know Schmidts is using it. |
| 18:21.27 | p3nguin | The PSTN is ulaw at best, so it would be a waste. |
| 18:21.34 | SeRi | so how do I use both ulaw and g722? |
| 18:21.42 | p3nguin | Why worry about it? |
| 18:22.00 | *** join/#asterisk mindCrime (~chatzilla@24.106.207.82) |
| 18:22.37 | SeRi | p3nguin: I am confused I can used g722 all the time and if a network does not support it will decode down to ulaw? |
| 18:22.50 | p3nguin | Asterisk will transcode. |
| 18:23.07 | SeRi | I see |
| 18:23.38 | SeRi | so i enable it on the peer for my phone right? |
| 18:23.44 | p3nguin | Correct. |
| 18:24.35 | SeRi | ok so is now is enabled on peer for polycom |
| 18:25.05 | p3nguin | Are you also allowing g722 in general? |
| 18:25.07 | SeRi | how can I see is using it? |
| 18:25.12 | SeRi | p3nguin: now |
| 18:25.15 | SeRi | now |
| 18:25.18 | SeRi | no* |
| 18:25.20 | SeRi | damn |
| 18:25.28 | p3nguin | sip show channels |
| 18:26.02 | p3nguin | In your sip.conf general, be sure you allow g722. |
| 18:26.08 | *** join/#asterisk Cesar_B (~chatzilla@201.200.175.218) |
| 18:26.14 | Cesar_B | -.- |
| 18:27.31 | SeRi | p3nguin: http://pastebin.com/rw7VzUUe |
| 18:28.18 | *** join/#asterisk gavimobile (~user@bzq-218-196-30.red.bezeqint.net) |
| 18:29.54 | p3nguin | seri: http://pastebin.com/r8MxsQKt |
| 18:30.29 | p3nguin | Also, why are you still qualifying your phone that is on your LAN? |
| 18:32.23 | SeRi | p3nguin: ooo just testing I have been lazy and not taken it out. ill do that now |
| 18:32.24 | *** join/#asterisk jaminja (~jaminja@unaffiliated/jaminja) |
| 18:32.30 | *** join/#asterisk TimeRider (~steve@92.40.253.200.threembb.co.uk) |
| 18:33.11 | p3nguin | If you take an hour to make this simple codec change, the conf is going to be closed before you have a chance to test it. |
| 18:33.27 | p3nguin | Over 50% of the people have already left. |
| 18:33.39 | p3nguin | Was 46, now 18. |
| 18:33.52 | SeRi | ok I am in |
| 18:34.06 | p3nguin | sip show channels |
| 18:34.10 | SeRi | nice g722! |
| 18:34.17 | p3nguin | You should see your codecs on both legs. |
| 18:34.27 | p3nguin | Do they both show g722? |
| 18:34.29 | SeRi | :( |
| 18:34.33 | *** join/#asterisk mateu (~mateu@missoula.org) |
| 18:34.34 | p3nguin | One is ulaw? |
| 18:34.40 | SeRi | Looks like I have to enable it on the phone |
| 18:34.58 | SeRi | I have to see how on my phone |
| 18:35.01 | p3nguin | I don't know if the phone supports it or not. That's why I asked if it does. |
| 18:36.01 | [TK]D-Fender | SeRi, what model? |
| 18:36.13 | p3nguin | I think it's a 501. |
| 18:36.13 | SeRi | [TK]D-Fender: 501 |
| 18:36.14 | [TK]D-Fender | SeRi, No G.722 on that thing.... |
| 18:36.16 | [TK]D-Fender | ^^^ |
| 18:36.20 | SeRi | ah ok |
| 18:36.22 | SeRi | lol |
| 18:36.25 | [TK]D-Fender | Now stop wasting your time. |
| 18:36.26 | SeRi | to old I guess |
| 18:36.27 | p3nguin | haha |
| 18:36.31 | [TK]D-Fender | WAY too old |
| 18:36.32 | SeRi | I wonder if the pap2 would |
| 18:36.35 | [TK]D-Fender | NO |
| 18:36.37 | p3nguin | no |
| 18:36.39 | SeRi | lol |
| 18:36.40 | p3nguin | too cheap |
| 18:37.50 | SeRi | :( |
| 18:37.56 | singler | :) |
| 18:38.12 | SeRi | damn cheapness!!!! |
| 18:38.46 | SeRi | *6 unmute? |
| 18:40.42 | *** join/#asterisk luckman212 (~irc@pool-108-41-8-176.nycmny.fios.verizon.net) |
| 18:41.44 | *** join/#asterisk gavimobile (~user@bzq-218-196-30.red.bezeqint.net) |
| 18:41.58 | p3nguin | yes, *6 to unmute. |
| 18:43.02 | luckman212 | Ast 1.8.8-rc4, Polycom IP331/550/650 running UC firmware 3.3.3 - whenever I initiate a reboot of a phone via Menu->3-1-6, I get this line in my *CLI> [2011-12-02 13:24:11] WARNING[9453]: chan_sip.c:20532 handle_response: Forbidden - maybe wrong password on authentication for NOTIFY |
| 18:43.26 | luckman212 | tried turning on SIP debug for one of those peers but it doesn't reveal anything immediately obvious to me |
| 18:43.26 | *** part/#asterisk gavimobile (~user@bzq-218-196-30.red.bezeqint.net) |
| 18:43.47 | luckman212 | no big deal just seems odd, wondering if anyone knows. Google turned up nil. |
| 18:44.40 | *** join/#asterisk leifmadsen (~Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 18:44.40 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 18:47.50 | BlackBishop | anyhow, how, I wonder how could I make a call through it ! :) |
| 18:48.21 | *** join/#asterisk FiReSTaRT (~dlyh@unaffiliated/firestart) |
| 18:49.14 | [TK]D-Fender | BlackBishop, Go make a call through it |
| 18:49.30 | BlackBishop | Dial(what ); |
| 18:49.31 | BlackBishop | ? |
| 18:50.23 | *** join/#asterisk joshaidan (~brianj@S0106000c6e79821d.tb.shawcable.net) |
| 18:50.52 | SeRi | facepalms |
| 18:50.58 | BlackBishop | headbangs ! |
| 18:53.02 | [TK]D-Fender | BlackBishop, SIP is SIP. Dial it like any other provider |
| 18:54.17 | BlackBishop | what I don't understand is .. how do I make my username ( blackbishop logged in through sip ) make the HT a call to where I want :| |
| 18:56.50 | [TK]D-Fender | Just like any other provider <- |
| 18:59.15 | BlackBishop | destination => Dial(SIP/314110724); ?! |
| 19:00.09 | [TK]D-Fender | BlackBishop, Have you ever set up * to dial out of anything else? |
| 19:00.22 | BlackBishop | Yeah, through a datacard... |
| 19:00.44 | [TK]D-Fender | Well I'm sure you'll recognize that you have to tell Dial() the number you want to dial out of it. |
| 19:00.44 | BlackBishop | Dial(Dongle/number); |
| 19:01.18 | [TK]D-Fender | Dial([tech]/[peer or channel]/[number to dial]) |
| 19:01.30 | BlackBishop | Mmm !!!!! |
| 19:01.34 | BlackBishop | tries it out |
| 19:03.07 | *** join/#asterisk irroot (~gregory@197.172.84.224) |
| 19:03.31 | n3hxs | we have a very busy system and when we want to get some info out of the CLI, it goes past so fast you can't see it. Is there a way to capture the stream from CLI/ |
| 19:03.33 | n3hxs | ? |
| 19:03.46 | BlackBishop | logger.conf ? |
| 19:04.43 | JunK-Y | n3hxs: yes, its called full in /var/log/asterisk/ (be sure to enable it in logger.conf, before) |
| 19:05.26 | *** join/#asterisk krotos (~androirc@83.224.73.33) |
| 19:05.59 | singler | and then use "logger reload" comman to apply config |
| 19:06.04 | singler | *command |
| 19:06.15 | n3hxs | Thanks. |
| 19:06.52 | *** join/#asterisk ruied (~ruied@po-217-129-155-146.netvisao.pt) |
| 19:08.13 | *** join/#asterisk FiReSTaRT (~dlyh@unaffiliated/firestart) |
| 19:09.06 | *** join/#asterisk beta2k (~Beta2K@d24-36-128-84.home1.cgocable.net) |
| 19:09.09 | beta2k | Hello all |
| 19:09.17 | beta2k | Anyone around running chan_sccp_b? |
| 19:09.32 | beta2k | I can't seem to get the realtime db to work with it |
| 19:09.58 | beta2k | Unfortunately their website doesn't explain why you're doing what it tells you too well :) |
| 19:10.08 | p3nguin | I use chan-sccp-b, but not with realtime. |
| 19:10.28 | beta2k | Or I'm missing something... |
| 19:10.54 | beta2k | I'm thinking about saying the heck with it and just doing it in the config :) |
| 19:11.05 | BlackBishop | [TK]D-Fender: sounds good, but .. trying to call my cell says "welcome to [my pstn telco], the number you have dialed is not in service" |
| 19:11.08 | BlackBishop | :/ |
| 19:11.11 | BlackBishop | or any other number infact.. |
| 19:12.22 | *** join/#asterisk neurosys (~neurosys@50.20.70.17) |
| 19:12.54 | [TK]D-Fender | BlackBishop, If you say so. |
| 19:13.14 | BlackBishop | :/ wonder what it tries to dial :| |
| 19:13.21 | [TK]D-Fender | "wonder"? |
| 19:13.30 | [TK]D-Fender | You are dialing it. there shouldn't be any "wonder" |
| 19:13.36 | p3nguin | ~faith |
| 19:13.36 | infobot | Telephony is not faith-based. Look. Always look, and then show. |
| 19:13.36 | BlackBishop | wwell, I'm dialing the right number. |
| 19:13.52 | [TK]D-Fender | And we don't see what yuor dialplan is doing. |
| 19:13.59 | BlackBishop | Dial(SIP/314110724/mycell); |
| 19:14.08 | *** join/#asterisk sezuan (bouncer@irc.scheff32.de) |
| 19:14.18 | [TK]D-Fender | BlackBishop, Then they don't like the number |
| 19:14.28 | *** join/#asterisk kikohnl (~kotis@ext-dip-166.hnl.cdsinc.com) |
| 19:14.43 | [TK]D-Fender | Your formatting is probably wrong for how you would need to dial it. |
| 19:15.02 | BlackBishop | ahm... |
| 19:15.10 | p3nguin | 314110724 is the peer name as defined in sip.conf? |
| 19:15.11 | [TK]D-Fender | And that isn't CLI output from an actual attempt, nor dialplan code |
| 19:15.14 | p3nguin | [314110724] |
| 19:15.16 | BlackBishop | yup |
| 19:15.40 | BlackBishop | http://pastebin.com/dGcdqCpP |
| 19:16.47 | beta2k | .v..'vv''vf'v;ptrying to call my cell says "welcome to [my pstn telco], the number you have dialed is not in service" |
| 19:16.50 | beta2k | trying to call my cell says "welcome to [my pstn telco], the number you have dialed is not in service" |
| 19:16.53 | beta2k | trying to call my cell says "welcome to [my pstn telco], the number you have dialed is not in service" |
| 19:16.59 | beta2k | trying to call my cell says "welcome to [my pstn telco], the number you have dialed is not in service" |
| 19:17.02 | beta2k | trying to call my cell says "welcome to [my pstn telco], the number you have dialed is not in service" |
| 19:17.05 | beta2k | sorry.... |
| 19:17.07 | beta2k | baby got the netbook :) |
| 19:17.09 | SeRi | ~pb |
| 19:17.09 | infobot | [~pb] A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few: http://www.pastebin.com , http://pastebin.us , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://bin.cakephp.org/ , http://asterisk.pastey.net/ ; or install pastebinit with yum or aptitude. |
| 19:17.20 | SeRi | :) |
| 19:17.27 | kikohnl | Good Morning! |
| 19:17.44 | [TK]D-Fender | BlackBishop, Well the telco doesn't like that number. If you're sure the format is right for them to dial exactly like that.... then call the telco. |
| 19:17.45 | beta2k | He sure loves keyboards lol |
| 19:18.10 | beta2k | Atleast he's not like his older brother, at this age he was removing the keys from every laptop keyboard he could get... |
| 19:19.44 | p3nguin | I guess the kid likes the combination of up arrow, then enter, repeat. |
| 19:19.50 | BlackBishop | it's the same format I'd call if I'd have the cable plugged in an actual phone ! |
| 19:20.11 | SeRi | lol |
| 19:23.47 | SeRi | p3nguin: You aint missing out much... |
| 19:23.54 | p3nguin | oh |
| 19:24.08 | SeRi | I feal like puking. |
| 19:24.15 | p3nguin | Just lame chatter? |
| 19:24.22 | SeRi | the damn screen keeps flipin... |
| 19:24.26 | SeRi | p3nguin: Yes. |
| 19:25.02 | SeRi | so when audio activates the screen flips to that persons video feed |
| 19:25.12 | p3nguin | ick |
| 19:25.13 | SeRi | imagine what happens when they all start talking |
| 19:25.19 | SeRi | barfs |
| 19:25.33 | p3nguin | lunch time |
| 19:25.39 | SeRi | I am about to have a seizure |
| 19:25.53 | Cesar_B | sorry for re questioning this, but i need help to solve one problem using libss7 with "Continuity Check Procedure", anyone can help me with this? |
| 19:31.16 | Cesar_B | no one? |
| 19:33.37 | fprior | Hi all; my scenario is Zoiper --> Asterisk --> LinkSys Spa400 --> POTS; during an outside call, when clients pickup, I receive very bad and strong noise. Check this: http://soundcloud.com/a13051922 ; what cause this ? |
| 19:39.25 | SeRi | p3nguin: I left that place... maybe next friday it will be interesting... the conf for sure was |
| 19:40.22 | p3nguin | Call it at 10:45-ish to get there when it starts. |
| 19:40.36 | p3nguin | They have a main topic of discussion with a guest speaker most of the time. |
| 19:40.56 | p3nguin | But that only lasts for the first hour (approximately). |
| 19:41.29 | beta2k | fprior, are your analog lines balanced? |
| 19:42.02 | beta2k | We get nasty hum on one of our ATA's due to really long loop length and it's imballanced |
| 19:42.06 | p3nguin | And I would like to say that my $2 Subway meatball sub was yummy. |
| 19:42.16 | Qwell | $2? WTF sorcery is this? |
| 19:42.33 | *** join/#asterisk jetlag (jetlag@pool-71-168-248-89.cmdnnj.east.verizon.net) |
| 19:42.59 | p3nguin | December is customer appreciation month at Subway. Meatball and cold cut combo subs are just $2 for 6-inch. |
| 19:43.13 | beta2k | us only I assume? |
| 19:43.15 | jrose_atDigium | So I heard on the radio this morning. |
| 19:43.33 | JunK-Y | p3nguin: only in US, cause i ate the same and wasnt at 2$ :( |
| 19:43.35 | p3nguin | I couldn't say if they appreciate their customers in other countries or not. |
| 19:44.28 | p3nguin | I guess they don't. |
| 19:44.50 | fprior | beta2k: how can I check if line is balanced ? |
| 19:44.53 | beta2k | Lets veto Subway Junk-Y for not appreciating us :) |
| 19:45.41 | beta2k | he telco |
| 19:45.45 | beta2k | You'd need a timset (sp?), check it with a POTS phone and if it's still there complain at the telco |
| 19:48.39 | *** join/#asterisk fisted_ (~fisted@unaffiliated/fisted) |
| 19:49.59 | *** join/#asterisk irroot (~gregory@197.169.71.128) |
| 19:50.03 | SeRi | p3nguin: got it. |
| 19:50.07 | SeRi | out to lunch |
| 19:50.47 | *** join/#asterisk neurosys_ (~neurosys@adsl-98-77-82-79.mia.bellsouth.net) |
| 19:51.24 | fprior | beta2k: sorry, I don't understand, timset ? |
| 19:51.52 | beta2k | It's a test set for ballanced lines |
| 19:52.09 | beta2k | Just check it with a regular phone and if it's there let the telco deal with it |
| 19:52.45 | p3nguin | seri: $2 Subway sub? |
| 19:52.53 | p3nguin | Get two and have a $4 footlong! |
| 19:53.14 | SeRi | p3nguin: lol nope. |
| 19:53.23 | SeRi | Tacos. |
| 19:53.27 | SeRi | Fajita :) |
| 19:53.43 | SeRi | or Fajaits |
| 19:57.28 | *** join/#asterisk malcolmd (~malcolmd@pdpc/sponsor/digium/malcolmd) |
| 19:57.29 | *** mode/#asterisk [+o malcolmd] by ChanServ |
| 19:58.00 | *** join/#asterisk kerx (~kerx@76-240-161-60.lightspeed.irvnca.sbcglobal.net) |
| 19:58.50 | SeRi | well shit... no more fajitas :( |
| 19:59.23 | fprior | beta2k, I've checket with analog phone, directly connected with a POTS. There is no bad sound but in place of it exists imperceptible noise. this mean line is unbalanced ? |
| 20:02.31 | *** join/#asterisk emedia (~chatzilla@201.200.175.218) |
| 20:02.51 | *** join/#asterisk malcolmd (~malcolmd@pdpc/sponsor/digium/malcolmd) |
| 20:02.51 | *** mode/#asterisk [+o malcolmd] by ChanServ |
| 20:03.47 | *** join/#asterisk joshaidan (~brianj@S0106000c6e79821d.tb.shawcable.net) |
| 20:04.22 | *** join/#asterisk hajekd (~hajekd@82.208.11.91) |
| 20:16.28 | [TK]D-Fender | fprior, Card issue |
| 20:17.00 | [TK]D-Fender | This would be the SPA400 apparently |
| 20:22.19 | *** join/#asterisk vpopov (~happylife@46.251.80.89) |
| 20:23.40 | fprior | [TK]D-Fender, beta2k: I've found problem. is line. I've 4 line from same telco and only 1 has noise. If I change spa400 port, noise keep on same line, thanks |
| 20:24.03 | [TK]D-Fender | ok |
| 20:25.09 | *** join/#asterisk malcolmd (~malcolmd@pdpc/sponsor/digium/malcolmd) |
| 20:25.09 | *** mode/#asterisk [+o malcolmd] by ChanServ |
| 20:26.48 | *** join/#asterisk jimi_ (~jimi@unaffiliated/tuxguy) |
| 20:28.10 | jimi_ | How can I tell if a queue call has ended? I want to call an external application once it's ended. |
| 20:33.45 | BlackBishop | nope .. it's not the telco, plugged in the cable into a phone and tried to dial the exact number, works .. |
| 20:34.02 | BlackBishop | so .. I'm using Dial(); wrong .. or the HT does something to it .. |
| 20:35.56 | [TK]D-Fender | look at the number. if you're passing it right, then then HT is doing something wrong |
| 20:36.57 | BlackBishop | definetly HT is doing something wrong .. I know my own number.. |
| 20:38.26 | BlackBishop | but .. since it doesn't have a DEBUG windows :| I can't see what it does .. |
| 20:39.21 | p3nguin | There's a debug feature in asterisk. That would be helpful to see what a device is sending to asterisk. |
| 20:40.24 | BlackBishop | mhm, but wouldn't help to see what the device is sending to the telco |
| 20:40.47 | BlackBishop | since it's making a connection to the telco, and I hear the telco robot "welcome to [telco], the number you have dialed is not in service" |
| 20:40.50 | BlackBishop | :| |
| 20:43.48 | [TK]D-Fender | BlackBishop, Pick up an analog phone in parallele and listen to hear if you get all the digits. Also it may be dialing too fast |
| 20:44.45 | *** join/#asterisk brdude (~brdude@12.155.183.30) |
| 20:52.38 | *** join/#asterisk voxter (~voxter@macpro.daytonhome.voxter.net) |
| 20:56.07 | *** join/#asterisk irroot (~gregory@197.174.44.193) |
| 20:56.28 | *** part/#asterisk StaRetji (~BigEight@80.93.240.171) |
| 20:58.34 | *** join/#asterisk TimeRider (~steve@92.40.253.200.threembb.co.uk) |
| 21:09.59 | *** join/#asterisk jrose_atDigium (~jon@nat/digium/x-harrvcndzctkhfcn) |
| 21:20.59 | *** join/#asterisk Ionic (ionic@ionic.de) |
| 21:24.31 | *** join/#asterisk Bidik (~bidik@74.117.156.225) |
| 21:25.50 | *** join/#asterisk joshaidan (~brianj@S010600095bfe15df.tb.shawcable.net) |
| 21:26.25 | [TK]D-Fender | Checkout time, BBIAB |
| 21:29.31 | *** join/#asterisk nighty- (~nighty@static-68-179-124-161.ptr.terago.net) |
| 21:36.02 | *** join/#asterisk Bidik (~bidik@74.117.156.225) |
| 21:38.29 | *** join/#asterisk IsUp (IsUp@unaffiliated/isup) |
| 21:38.42 | IsUp | hello |
| 21:40.11 | *** join/#asterisk CVirus (~GoD@41.130.183.209) |
| 21:41.23 | CVirus | Asterisk is segfaulting when I try to connect to it http://pastie.org/2957006 |
| 21:42.37 | jrose_atDigium | CVirus: You'll probably want to generate a backtrace. |
| 21:42.51 | IsUp | ChanServ: anything in /var/log/messages ? |
| 21:43.07 | jrose_atDigium | CVirus: https://wiki.asterisk.org/wiki/display/AST/Getting+a+Backtrace |
| 21:46.01 | CVirus | IsUp: http://pastie.org/2957030 |
| 21:47.32 | CVirus | let me get the backtrace |
| 21:48.35 | IsUp | CVirus: its Ubuntu, right? |
| 21:48.41 | CVirus | by the way asterisk is running normally .. the segfault only happens when I connect to it |
| 21:48.45 | *** join/#asterisk PoWeRKiLL (~powerkill@77.125.87.36) |
| 21:48.48 | CVirus | IsUp: no this is another machine running Debian |
| 21:49.02 | CVirus | even after the segfault .. asterisk is still running |
| 21:49.30 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.218.194) |
| 21:51.17 | IsUp | hello [TK]D-Fender |
| 21:53.26 | CVirus | gdb -se "asterisk" -ex "bt full" -ex "thread apply all bt" --batch -c core > /tmp/backtrace.txt |
| 21:53.32 | CVirus | /root/core: No such file or directory. |
| 21:53.33 | CVirus | No stack |
| 21:54.06 | CVirus | err |
| 21:54.35 | WIMPy | I'm not sure you can make the remote console do a core dump. You may have to run it with gdb. |
| 21:55.19 | CVirus | I misunderstood that part ... well asterisk doesn't actually crash in my case |
| 21:55.26 | CVirus | I only fail to connect to it |
| 21:55.35 | CVirus | using asterisk -rvvvvvvv |
| 21:58.21 | CVirus | http://us.generation-nt.com/answer/bug-649431-asterisk-segmentation-fault-asterisk-help-205498151.html |
| 21:58.23 | CVirus | :-) |
| 21:59.52 | Qwell | tl;dr: You should have upgraded before asking. |
| 22:03.55 | CVirus | Qwell: I'm already up-to-date on my branch |
| 22:05.11 | p3nguin | 1.8.7.1? |
| 22:06.01 | CVirus | p3nguin: Asterisk 1.8.7.1~dfsg-1 |
| 22:07.36 | *** join/#asterisk TJNII (~TJNII@tjnii.com) |
| 22:07.59 | *** join/#asterisk evil_gordita (robert@ip70-188-50-186.rn.hr.cox.net) |
| 22:09.30 | *** join/#asterisk parasitodelsur (~parasitod@c-76-31-169-54.hsd1.tx.comcast.net) |
| 22:13.15 | *** join/#asterisk irroot (~gregory@197.108.118.96) |
| 22:17.39 | *** part/#asterisk wesphillips (~wphill04@137-237-233-124.harris.com) |
| 22:22.13 | *** join/#asterisk Naikrovek (~mbluth@unaffiliated/naikrovek) |
| 22:23.23 | *** join/#asterisk bmg505 (~leon@196-209-123-3.dynamic.isadsl.co.za) |
| 22:27.45 | BlackBishop | what's the default insecure value *10 ? :/ |
| 22:32.18 | *** join/#asterisk joshaidan (~brianj@24.109.210.41) |
| 22:34.30 | p3nguin | Rephrase. |
| 22:43.07 | BlackBishop | the default value for insecure= for a peer if I don't specify any |
| 22:46.03 | p3nguin | If none is specified, there is no value. |
| 22:46.24 | p3nguin | I guess that would be insecure=no |
| 22:59.48 | p3nguin | This doesn't make any sense at all. I applied the following patch, and core show application ConfBridge still shows * rather than # for the menu. http://svnview.digium.com/svn/asterisk/branches/1.8/apps/app_confbridge.c?view=patch&r1=345545&r2=345544&pathrev=345545 |
| 23:00.10 | p3nguin | app_confbridge.c is patched. |
| 23:00.31 | p3nguin | A new app_confbridge.o and app_confbridge.so are built. |
| 23:01.04 | p3nguin | Install app_confbridge.so, restart (or reload app_confbridge.so), and it still shows the wrong character. |
| 23:01.51 | p3nguin | grepping the entire source tree only finds the line in app_confbridge.c and doc/core-en_US.xml, which are both now reflecting # correctly. |
| 23:01.57 | p3nguin | How it that possible? |
| 23:06.58 | dijib | i dont know, same shite with mine and app_SWIFT |
| 23:17.43 | *** join/#asterisk s[X] (~mark@ppp118-208-82-70.lns20.bne4.internode.on.net) |
| 23:31.54 | *** part/#asterisk mjordan (~mjordan@nat/digium/x-bohxxizvymjlpomd) |
| 23:35.45 | *** join/#asterisk bmg505 (~leon@196-209-123-3.dynamic.isadsl.co.za) |
| 23:39.55 | p3nguin | Did swift compile successfully? |
| 23:42.39 | *** join/#asterisk s[X] (~mark@ppp118-208-82-70.lns20.bne4.internode.on.net) |
| 23:44.02 | dijib | yes it did p3nguin |
| 23:44.13 | p3nguin | No errors? |
| 23:44.34 | p3nguin | Any warnings? |
| 23:46.50 | *** join/#asterisk s[X] (~mark@ppp118-208-82-70.lns20.bne4.internode.on.net) |
| 23:50.03 | *** join/#asterisk FiReSTaRT (~dlyh@unaffiliated/firestart) |
| 23:55.38 | *** join/#asterisk golikwid|mac (~chrislees@64.45.192.200) |
