| 00:02.38 | nnY_2 | crap crap |
| 00:02.52 | nnY_2 | wtf is wrong with this phone. Not registering, but has the appropriate info |
| 00:03.14 | nnY_2 | i would just chalk it up to being broken, but that's strange state to be broken |
| 00:03.23 | *** part/#asterisk astassistant (n=skip@65-126-63-1.dia.static.qwest.net) |
| 00:09.56 | *** join/#asterisk stencil (n=stencil@d193-237-37.home3.cgocable.net) |
| 00:12.43 | nnY_2 | hmm sip info has a bunch of messages with Unknown Caller.. over and over.. ok maybe thats part of it |
| 00:15.04 | *** join/#asterisk ming_zym (n=ming_zym@nat/yahoo/x-6da63567fb90e7fb) |
| 00:17.42 | nnY_2 | er nm thats just asterisk querying OPTIONS |
| 00:18.58 | drmessano | grandstream? |
| 00:21.29 | *** join/#asterisk msaa (n=dlink514@89-212-1-27.dynamic.dsl.t-2.net) |
| 00:21.41 | *** join/#asterisk Gershwin (n=fake@63.250.233.162) |
| 00:22.24 | *** join/#asterisk bkw_ (n=bkw_@freeswitch/developer/bkw) |
| 00:22.27 | *** join/#asterisk CtRiX (n=CtRiX@aretha.navynet.it) |
| 00:22.33 | *** join/#asterisk [intra]lanman (n=lanman@freeswitch/developer/intralanman) |
| 00:22.49 | bkw_ | ~centos52bug |
| 00:22.50 | jbot | [~centos52bug] There is a bug compiling Zaptel up to 1.4.11/1.2.26 on CentOS/RHEL 5.2: error in xpp/xdefs.h. Workaround: uncheck "xpp" (Astribank) driver in menuselect. Fixed in SVN. Patches: http://bugs.digium.com/12889 |
| 00:26.29 | nnY_2 | drmessano no aastra |
| 00:26.40 | nnY_2 | drmessano I honestly think the phone is just broken some how |
| 00:27.05 | C4away | what's wrong with the aastra? |
| 00:27.12 | C4away | aastras are the best, they never break |
| 00:27.16 | C4away | it must be your configuration |
| 00:27.20 | nnY_2 | it worked up until today for months, and now it won't register. I have restored the factory defaults, reprovisioned it, sacrificed many many animals |
| 00:27.21 | nnY_2 | nope |
| 00:27.24 | drmessano | No polycoms are |
| 00:27.28 | nnY_2 | for one, it just stopped working |
| 00:27.31 | C4away | oh yea |
| 00:27.32 | drmessano | ~polycommunist |
| 00:27.32 | jbot | A polycommunist is someone who believes Polycom phones can do no wrong.. that Polycom's are so over and above anything else, that what you are using is surely crap, and the mere fact you mention another brand name is of great insult to the channel, the community, and the world. They may also be getting a 10% kickback. |
| 00:27.35 | nnY_2 | ha |
| 00:27.37 | C4away | sorry, it's probably the phone |
| 00:27.41 | nnY_2 | heh |
| 00:27.46 | C4away | i've nearly RMAd a few aaastrtas |
| 00:27.47 | nnY_2 | i have a new one en route for tomorrow am |
| 00:27.49 | bkw_ | C4away: Aastra has bugs. |
| 00:27.49 | C4away | wow |
| 00:27.50 | Alton2 | hah |
| 00:27.52 | C4away | let me try that again |
| 00:27.54 | C4away | aastras |
| 00:27.54 | nnY_2 | yeah i stopped using them |
| 00:28.06 | nnY_2 | using snom m3 for cordless for now |
| 00:28.09 | bkw_ | I found two bugs with Aastra phones with in 4 min of taking it out of the box |
| 00:28.20 | nnY_2 | yeah the old firmware use to crash when you pressed the left arrow >< |
| 00:28.22 | C4away | bkw_: I know, I was turning the poly retorhic against them |
| 00:28.46 | nnY_2 | yeah i won't buy them anymore, that's for certain |
| 00:28.55 | nnY_2 | i had to rma the handsets already for stupid reasons too |
| 00:28.58 | *** join/#asterisk axisys (n=axisys@ip68-98-177-71.dc.dc.cox.net) |
| 00:28.59 | C4away | hahaha, I have discovered sarcasm and satire ... I call it stcarcasm |
| 00:29.08 | C4away | satirecasm |
| 00:29.14 | C4away | sartire |
| 00:29.16 | C4away | there we go |
| 00:29.21 | nnY_2 | i would have said sartire :) |
| 00:29.27 | C4away | lol |
| 00:29.31 | C4away | took me a second, but I got it |
| 00:30.09 | nnY_2 | at this point i just want to know what is actually not working |
| 00:30.20 | nnY_2 | i have the phone syslogging to the server, but the output is useless |
| 00:30.25 | nnY_2 | i get this from sip debugging |
| 00:30.26 | nnY_2 | http://pastebin.com/m13709c53 |
| 00:30.49 | C4away | btw bkw_ wtf you doin here? got your fishing nets out or something? |
| 00:31.50 | bkw_ | C4away: haha |
| 00:32.18 | bkw_ | C4away: what poly retorhic? |
| 00:32.32 | C4away | polycoms are the best, never blame the phone, nothing is superior |
| 00:32.32 | C4away | etc |
| 00:32.51 | bkw_ | no my list goes Snom, Polycom everything else is pure crap |
| 00:32.59 | C4away | heh |
| 00:33.04 | bkw_ | snom is high on my list because they support ipv6 |
| 00:33.18 | *** join/#asterisk DarkRift (n=dark@bas10-montreal02-1177582806.dsl.bell.ca) |
| 00:33.25 | C4away | my list starts with "crap" and goes to "absolutely abysmal shit" |
| 00:33.39 | C4away | the phones are pretty evenly spaced out in there |
| 00:34.07 | C4away | at the present moment I have rated the grandstream gxp2000 highest on the list |
| 00:34.11 | *** join/#asterisk Kumbang (n=kumbang@router-wmi.paume.itb.ac.id) |
| 00:34.23 | C4away | every once in a while I throw the names back in the hat and mix them up a bit |
| 00:34.28 | *** join/#asterisk Dovid (n=Dovid@tony09-121-90.inter.net.il) |
| 00:34.39 | bkw_ | C4away: the shit list? |
| 00:34.47 | C4away | "oh wow, this week pingtel phones are the highest rated? interesting" |
| 00:34.55 | bkw_ | because every single grandstream product is pure shit |
| 00:35.03 | bkw_ | they work mostly |
| 00:35.07 | bkw_ | but they dont' live up to my stnadards |
| 00:35.15 | C4away | ah, poly is on the shit list because of their crap user interface |
| 00:35.28 | bkw_ | you don't config poly from the user interface for starters |
| 00:35.34 | C4away | grandstream because I have found more bugs than features ... and that's bad because they have a list of features the size of my head |
| 00:35.42 | bkw_ | I found 39 bugs in one day |
| 00:35.47 | bkw_ | I just got pissed and set it to the side |
| 00:35.48 | C4away | I mean DTMF keys, message button, programmable buttons ... etc |
| 00:35.58 | C4away | USER interface, not web interface |
| 00:36.07 | C4away | and yes, our polys are always configured by FTP |
| 00:36.09 | *** join/#asterisk s0lid (n=s0lid@60.52.253.84) |
| 00:36.31 | nnY_2 | linksys is what i use now |
| 00:36.50 | C4away | ciscos because they sound like crap unless you use sccp/skinny/mgcp/etc |
| 00:36.58 | C4away | intentionally |
| 00:37.11 | C4away | yea, we support sip, but what you really need is our call manager |
| 00:37.19 | bkw_ | linkshit is what I call them |
| 00:37.22 | nnY_2 | yeah i have a cisco at the office along with grandstream, poly, snom and a linksys 962 and it sucks the most |
| 00:37.29 | bkw_ | or stinksys |
| 00:37.40 | nnY_2 | hmm i don't have any problems with them |
| 00:37.42 | C4away | linksys because they are crappy ciscos or souped-up SPAs |
| 00:37.50 | C4away | SPA because they are crap |
| 00:38.02 | C4away | and see linksys |
| 00:38.22 | Alton2 | Ciscos and some others are a pain when you lose the settings. At least you can reset the cheaper phones and get them going. |
| 00:38.27 | Alton2 | What use is a super phone set to the side? |
| 00:38.31 | C4away | aastra because they are buggy and have features in the admin interface that are there, but just don't work |
| 00:39.12 | bkw_ | I love the aastra dtmf bug |
| 00:39.18 | C4away | which is this? |
| 00:40.00 | bkw_ | it fucks up and sends a 4 min duration for an rfc2833 button press |
| 00:40.00 | bkw_ | nice bug |
| 00:40.00 | C4away | wow |
| 00:40.00 | C4away | I found one |
| 00:40.02 | *** join/#asterisk hmmhesays (n=hmmhesay@70-57-193-123.farg.qwest.net) |
| 00:40.29 | C4away | different sip accounts per line indicator, you make/receive a call on let's say L4, then a call comes in on L1, you bridge them together with the "conf" button, the green light follows from L4 to L1, guess what, you have to reset to get L1 not to be L4 anymore |
| 00:40.42 | C4away | L1 inherrits the settings for the L4 button if you bridge them together |
| 00:41.18 | bkw_ | how does that shit make it past QA? |
| 00:41.19 | C4away | but only if the green light hits that button and they are bridged there ... if you put L4 on hold, then go to L1 and then bridge L1 to L4, instead of the other way around, you don't have the problem because they are bridged on L4 and the green light stays there |
| 00:41.22 | bkw_ | I guess i'm just too damn picky |
| 00:41.33 | C4away | I am too, those things piss me off |
| 00:41.40 | bkw_ | brb |
| 00:41.42 | C4away | which is why I say every manufacturer is on the shit list |
| 00:41.46 | nnY_2 | which phone is that? |
| 00:41.52 | C4away | 480i |
| 00:41.55 | nnY_2 | ha nice |
| 00:41.58 | C4away | I will be testing on the 9133i shortly |
| 00:42.02 | nnY_2 | yeah i no longer use aastra |
| 00:42.09 | C4away | I"m moving to only aastra |
| 00:42.30 | C4away | they are all shit, might as well pick one and dive in |
| 00:42.33 | nnY_2 | hahaha |
| 00:42.37 | nnY_2 | that is the absolute truth |
| 00:42.43 | C4away | at least then I know one set of bugs to deal with |
| 00:42.48 | nnY_2 | indeed |
| 00:44.46 | C4away | well I do need to go, like 40 minutes ago |
| 00:44.57 | C4away | I shall be back later, I assure you all of this fact |
| 00:47.52 | Gershwin | prediction |
| 00:56.34 | *** join/#asterisk ManxPower (n=manxpowe@adsl-222-29-13.msy.bellsouth.net) |
| 00:57.43 | *** join/#asterisk lowtek (n=no@adsl-074-185-089-046.sip.bhm.bellsouth.net) |
| 00:59.33 | *** join/#asterisk iewebguy (n=mark@65.19.81.253) |
| 01:03.21 | Dovid | ManxPower: When updating the kernel can I rebuild zaptel b4 the reboot or after ? |
| 01:04.54 | Dovid | rather can I rebuild it b4 rebooting the box ? |
| 01:05.37 | [TK]D-Fender | Dovid: after |
| 01:05.50 | Alton2 | what's the reason for that? |
| 01:07.10 | iewebguy | Hello, I have this same problem (again) with a new asterisk install .... http://lists.digium.com/pipermail/asterisk-users/2007-February/181272.html Can't make "local" calls on a brand new PRI |
| 01:07.29 | Alton2 | how many digits are you dialing? |
| 01:09.30 | Alton2 | Happy to help! :-) |
| 01:09.55 | iewebguy | 7 |
| 01:10.18 | Alton2 | sometimes that's an option that you need to have the carrier offer, that's the case with our carrier anyway |
| 01:10.28 | Alton2 | just dial 10 digits, costs the same. |
| 01:12.55 | iewebguy | it rejects 10 digits with a recording. (you do not need to dial the AC or 1 when calling this number) |
| 01:13.10 | Alton2 | golly |
| 01:14.09 | iewebguy | my frustration is that I had this problem last year and I can't recall the solution |
| 01:14.40 | *** join/#asterisk moy (n=moy@189.169.208.120) |
| 01:14.43 | Alton2 | trying to think, although we haven't had this problem |
| 01:15.33 | Alton2 | I wonder if it's something odd, like you're not providing proper caller id or the dtmf isn't quite right. |
| 01:16.25 | iewebguy | I have the old asterisk 1.2 system working and copied all the setup over |
| 01:16.47 | Alton2 | check everything related, mostly codec used and dtmf type |
| 01:16.54 | *** join/#asterisk jaytee (n=jforde05@unaffiliated/jaytee) |
| 01:19.07 | [TK]D-Fender | iewebguy: "prilocaldialplan=national" + "pridialplan=national" -> zapata.conf |
| 01:19.24 | iewebguy | tried that. |
| 01:19.31 | iewebguy | tried all the choices |
| 01:19.47 | Alton2 | isn't switch type listed somewhere? N1-A or something, can't remember |
| 01:20.09 | Alton2 | looking here |
| 01:20.49 | Alton2 | in zapata.conf, switchtype = national |
| 01:22.19 | iewebguy | I have it set to national |
| 01:22.25 | iewebguy | same as old system. |
| 01:23.35 | Alton2 | signalling= setting? |
| 01:23.37 | jaytee | national or NI1 should work. I've also seen NI2 but I'm not sure if * will recognize that |
| 01:24.10 | jaytee | pri_cpe on the * side usually |
| 01:24.13 | Alton2 | Is this the same hardware you used before, when it worked? Just curious. |
| 01:24.17 | iewebguy | signalling=pri_cpe |
| 01:24.53 | iewebguy | same card from same company. but new firmware sangoma 101D I think it is |
| 01:25.28 | Alton2 | I wouldn't know if that might be relevant, but someone else might. |
| 01:25.38 | JT | national in zapata == NI2 |
| 01:26.25 | jaytee | that's what I thought too |
| 01:27.56 | ManxPower | So it also rejects it for 7-digit calls? |
| 01:28.27 | ManxPower | [TK]D-Fender: No! pridialplan=unknown |
| 01:29.03 | iewebguy | 7 digit calls yup |
| 01:29.11 | ManxPower | or leave it out. That's sort of why in zapata.conf.sample the commends for pridialplan= and prilocaldialplan= says "you almost never need to set this" |
| 01:29.32 | iewebguy | I left them out. - no work |
| 01:29.44 | iewebguy | I tried unknown |
| 01:29.48 | iewebguy | no work |
| 01:29.52 | ManxPower | iewebguy: and you either stopped and started asterisk (not reload) or unload/load chan_zap.so ? |
| 01:30.17 | iewebguy | ya I stop asterisk |
| 01:30.30 | ManxPower | iewebguy: call your carrier 8-) |
| 01:30.32 | iewebguy | how do I unload chan_zap? |
| 01:30.38 | ManxPower | unload chan_zap.so |
| 01:31.54 | drmessano | IE Web Guy? |
| 01:32.23 | iewebguy | We have "joe's" telecom here :( |
| 01:32.28 | iewebguy | 802 496 |
| 01:32.55 | ManxPower | iewebguy: Asterisk can't change the settings on the telco side of the PRI so maybe you should call them? |
| 01:33.36 | iewebguy | my OLD asterisk install 1.2 works like a charm (except for 2% dropped calls) |
| 01:34.07 | ManxPower | what version of 1.4 are you using? |
| 01:35.00 | drmessano | Where the hell is RC1? |
| 01:35.12 | Qwell | of? |
| 01:35.23 | drmessano | 1.6 |
| 01:35.33 | drmessano | I was promised a RC1 last week by Russell |
| 01:35.42 | drmessano | He said he demanded an RC1 |
| 01:35.48 | drmessano | That it WOULD be done |
| 01:35.58 | drmessano | Did you guys fall asleep on the job? |
| 01:36.02 | Qwell | It was done. |
| 01:36.06 | Qwell | you missed it |
| 01:36.42 | drmessano | Using Digium numbering again? |
| 01:36.44 | drmessano | Errr |
| 01:36.46 | drmessano | HA |
| 01:36.49 | drmessano | Slip of the tongue |
| 01:36.50 | Qwell | rc3 |
| 01:36.55 | *** join/#asterisk riddlebox (n=james@75-132-225-75.dhcp.stls.mo.charter.com) |
| 01:36.55 | drmessano | Fonality numbering |
| 01:36.59 | drmessano | Beta9 = RC1 |
| 01:37.05 | drmessano | Beta16 = RC2 |
| 01:37.07 | drmessano | ? |
| 01:37.09 | Qwell | rc3 |
| 01:37.13 | iewebguy | the latest 1.4 downloaded about 4 days back |
| 01:37.35 | drmessano | um, 1.6 dood |
| 01:37.47 | Qwell | yeah? |
| 01:37.54 | drmessano | Russell told me RC1 would be out |
| 01:38.03 | ManxPower | iewebguy: Some questions have an option of a vague answer. However, "what version you are using" is *not* one of them. |
| 01:38.05 | Qwell | it won't be |
| 01:38.06 | drmessano | he said it was a drop dead daete |
| 01:38.13 | drmessano | date* |
| 01:38.16 | drmessano | It wont be? |
| 01:38.18 | drmessano | Why is that? |
| 01:38.21 | Qwell | rc3 |
| 01:38.33 | drmessano | So where is RC3? |
| 01:38.38 | Qwell | svn |
| 01:38.50 | ManxPower | Just release it so people can start finding bugs. 8-| |
| 01:39.18 | drmessano | RC3 is in SVN? |
| 01:39.18 | Qwell | ManxPower: it's ready to go.. nobody has hit the big red button though |
| 01:39.29 | ManxPower | Qwell: Ah. Pussies. |
| 01:39.45 | drmessano | I'm calling shenanigans |
| 01:40.42 | drmessano | Until RC1 is out, I am going to swear up and down it hasn't been released due to the last minute inclusion of someone module used to do nasty things to users |
| 01:40.50 | drmessano | Errr |
| 01:40.53 | drmessano | some |
| 01:41.02 | drmessano | Yes, indeed |
| 01:41.12 | ManxPower | Qwell: But I understand what you mean. In a meeting today the new (like his first week) IT manager at my largest customer. He asked what needs to be done to upgrade the OS on their web server. I said I would not touch it and the other two people in the room also flat out said "I'm not touching that". |
| 01:41.30 | iewebguy | Ok it looks like the final answer is: pridialplan=local prilocaldialplan=local |
| 01:41.43 | iewebguy | thanks for all of your ideas! |
| 01:41.47 | ManxPower | iewebguy: and you can dial toll calls? |
| 01:41.50 | drmessano | RC1 is going to contain the codename garrykerrison module that makes your PBX post lies on forums |
| 01:41.58 | ManxPower | because usually with a local dialplan you can't dial toll calls |
| 01:42.07 | iewebguy | yes, LD worked since the start |
| 01:42.21 | iewebguy | I can still make LD calls |
| 01:42.40 | iewebguy | my telco is in the dark ages. |
| 01:47.03 | iewebguy | I think they got the switch they use on ebay ;) |
| 01:47.51 | *** join/#asterisk M-I-A (n=chacha@bas21-toronto12-1242562547.dsl.bell.ca) |
| 01:48.37 | *** join/#asterisk irieKen (n=chatzill@rrcs-74-87-28-55.west.biz.rr.com) |
| 01:50.27 | jaytee | so when I move from 1.4 to 1.6 I'll have to DAHDI all my ZAP stuff and probably DIDDLE a bunch of other shit too I imagine. |
| 01:50.33 | *** join/#asterisk outtolunc (n=me@c-67-170-211-86.hsd1.ca.comcast.net) |
| 01:51.09 | jaytee | PowerNaps are good :-) |
| 01:51.22 | drmessano | jaytee: 1.6 is months away |
| 01:51.26 | drmessano | jaytee: Maybe years |
| 01:51.51 | jaytee | drmessano, so is a stable LTS version of Ubuntu :-) |
| 01:52.58 | jaytee | I was at the book store tonight and saw a current issue of Linux + magazine that had an Ubuntu 7.10 CD in it. I went WTF? |
| 01:54.09 | drmessano | I woke up this morning, no 1.6 RC1 again |
| 01:54.12 | drmessano | I was all like |
| 01:54.18 | drmessano | *cough*trixbox*cough* |
| 01:54.25 | jaytee | lol |
| 01:54.26 | irieKen | does anyone know how to make Asterisk save settings to flash (commit current config files that are in the ramdisk to flash)? |
| 01:54.34 | irieKen | Hehe:S |
| 01:54.36 | irieKen | *:D |
| 01:54.41 | lowtek | cp? |
| 01:54.53 | drmessano | Save settings? |
| 01:55.00 | drmessano | They're config files on the file system |
| 01:55.01 | lowtek | cp /dev/ramfs /dev/flash |
| 01:55.04 | lowtek | or whatever |
| 01:55.05 | drmessano | yeah lol |
| 01:55.15 | irieKen | drmessano: I have an AA50, so it's all in ramfs. |
| 01:55.22 | irieKen | lowtek: let me take a look:D |
| 01:55.59 | jaytee | I much prefer the AA70 because it has the built in expresso maker and microwave |
| 01:56.16 | irieKen | lowtek: Hmm, no /dev/flash. |
| 01:56.42 | jaytee | can you SCP them to another box? |
| 01:56.49 | lowtek | Or from wherever the files are ... |
| 01:57.12 | irieKen | jaytee: No SCP on the AA50, as far as I know, but I can FTP (it crashes the AA50 half the time though). |
| 01:57.32 | irieKen | lowtek: I can't seem to figure out where the flash mountpoint would be. |
| 01:57.35 | jaytee | can you ssh into it? if so, try using FileZilla from another system |
| 01:57.55 | jaytee | irieKen, what about /media? |
| 01:58.20 | lowtek | Or just use rsync |
| 01:58.31 | irieKen | jaytee: nope, no /media |
| 01:59.01 | jaytee | sounds like a totally worthless crappy appliance then |
| 01:59.26 | irieKen | jayteee:S |
| 01:59.56 | jaytee | I looked at it on Digium's site and thought, hmmmm, wonder if Toys 'R Us carries those? |
| 02:00.13 | irieKen | Honestly, all I want to do is find a way to set the gain levels, and not have the device wipe them out when it gets rebooted. |
| 02:00.30 | *** join/#asterisk philippel (n=p_lindhe@pool-98-111-70-106.sttlwa.fios.verizon.net) |
| 02:00.44 | irieKen | I set rxgain in fxotune.conf, and it seems to have held... but it isn't applying them on reboot. |
| 02:01.04 | jaytee | is fxotune running at startup? |
| 02:01.37 | irieKen | shouldn't it be? |
| 02:02.22 | philippel | question, I call background() in a macro, but explicitly set the macro's context. If I only have single digit extensions as options for the background command, it works. If I have multiple digit options, it fails and dies. I do NOT have the 'm' option set (which seems to not make a difference in this scenario. Is this known or expected behavior or a potential bug? |
| 02:03.04 | irieKen | Where is gain traditionally set? |
| 02:03.13 | lowtek | philippel: error 99 |
| 02:03.21 | jaytee | what distro does the AA50 run? Rpath? |
| 02:03.26 | lowtek | pastbin yer code dude |
| 02:04.06 | philippel | lowtek I think I got rid of it already (and did it a different way) let me check |
| 02:04.48 | irieKen | jaytee: No idea. What's the command to find out?:D |
| 02:04.59 | *** join/#asterisk Sargun (n=xbmodder@atarack/staff/sargun) |
| 02:05.22 | lowtek | Right now FireFox 3 is using 30% of my CPU with no plugins sitting on google.com |
| 02:05.24 | jaytee | uname -r |
| 02:05.42 | irieKen | lowtek: Are you running a pentium II 200? |
| 02:06.05 | lowtek | irieKen: No, Quad Core Extreme Edition oc'd to 4GHz |
| 02:06.14 | lowtek | FF is bugged |
| 02:06.33 | lowtek | QX6850 |
| 02:06.33 | philippel | lowlevel I did get rid of it - but back to the question (before I report a bug) shoudl the background application work with multi-digit options in a macro, when the macro-context is set and the 'm' option is NOT present, or not? Then I can create an example failing applicaion to put in a bug if it shoudl work |
| 02:06.49 | irieKen | jaytee: It doesn't give a distro name... Just says linux:S |
| 02:06.50 | philippel | it's just not clear - given the know problems with background in macros to begin with |
| 02:07.36 | lowtek | I have no idea since I haven't seen your code and no clue what version of asterisk your running but I don't know of such issues ... |
| 02:07.44 | lowtek | code=dialplan |
| 02:08.12 | philippel | lowtek if you don't know of the issues with background and macro, then don't worry - they are well known (the issues) |
| 02:08.54 | lowtek | What version of asterisk? |
| 02:09.26 | *** join/#asterisk s0lid (n=s0lid@60.52.253.84) |
| 02:09.29 | philippel | 1.4.20.1 - but the issue is present in 1.2 I believe (the background + macro if you don't specify a context) |
| 02:10.44 | irieKen | Anyone know how I can change what the current gain level is set to on a channel in Asterisk? |
| 02:10.52 | [TK]D-Fender | do NOT try to make IVR's in macros |
| 02:10.59 | irieKen | *how I can tell what the current gain level is |
| 02:11.31 | [TK]D-Fender | irieKen: There is no "gain" except on a zaptel channel, and thats fixed |
| 02:11.31 | iewebguy | rxgain=0.0 |
| 02:11.31 | iewebguy | txgain=0.0 |
| 02:11.31 | iewebguy | in zapata.conf |
| 02:12.06 | iewebguy | what do you mean fixed? |
| 02:12.10 | philippel | [TK]D-Fender I am aware, none the less I'm trying to determine if the issue is a bug |
| 02:12.20 | jaytee | irieKen, sorry, to find out the distro it's running type cat /etc/*-release |
| 02:12.21 | [TK]D-Fender | philippel: Which issue? |
| 02:12.34 | irieKen | [TK]D-Fender: I can change the hwgain via "zap set hwgain rx ..." . I just want to know if there is a way to check what it is currently set to:) |
| 02:12.51 | philippel | I put background() in a macro, explicititly set the context to that macro, and explicitly did NOT have 'm' option set |
| 02:12.56 | *** part/#asterisk bkw_ (n=bkw_@freeswitch/developer/bkw) |
| 02:12.57 | *** join/#asterisk vonkleist (n=gcontrer@201.116.65.115) |
| 02:13.01 | philippel | however, only single digit options worked |
| 02:13.14 | vonkleist | hi |
| 02:13.15 | [TK]D-Fender | philippel: pastebin it |
| 02:13.18 | philippel | if I tried to do more than one digit options, ithe channel just died |
| 02:13.19 | irieKen | jaytee: /etc/*-release doesn't exist:S |
| 02:13.21 | vonkleist | could somebody helpme with this? |
| 02:13.24 | vonkleist | zap show statusDescription Alarms IRQ bpviol CRC4 Fra Codi Options LBOWildcard TE121 Card 0 RED/NOP 1 0 0 CAS HDB3 CRC4/YEL 0 db (CSU)/0-133 feet (DSX-1)Wildcard TDM400P REV I Board 1 OK 0 0 0 CAS Unk YEL 0 db (CSU)/0-133 feet (DSX-1)Wildcard TDM400P REV I Board 2 OK 0 0 0 CAS Unk YEL 0 db (CSU)/0-133 feet (DSX |
| 02:13.38 | philippel | [TK]D-Fender I don't have it any longer, so the question is this: |
| 02:13.41 | vonkleist | what does that alarms mean on the TE121 Card? |
| 02:13.56 | jaytee | irieKen, that's odd. |
| 02:14.08 | philippel | if it is suppose to work, I can make a dialplan that reproduces it so I can fiel a bug, but if it is not suppose to work, then no bother other than the help menu should be clarified |
| 02:14.24 | jaytee | every REAL linux distro supports the cat /etc/*-release to show distro info |
| 02:14.48 | jaytee | but this is an appliance so Digium probably stripped everything that wasn't crucial out of it. |
| 02:15.12 | *** join/#asterisk kamanashisroy (n=kamanash@202.56.7.193) |
| 02:15.30 | jaytee | vonkleist, pastebin your stuff, don't paste it in here |
| 02:15.39 | vonkleist | sorry, jaytee |
| 02:15.53 | irieKen | jaytee: Digium has done some odd things with it, and didn't bother to provide documentation:S |
| 02:15.55 | jaytee | it isn't really readable in here and you flood the channel |
| 02:16.28 | jaytee | irieKen, there is a user guide for the AA50 but it isn't very useful |
| 02:16.29 | vonkleist | here, http://gera.pastebin.ca/1181251 |
| 02:16.34 | vonkleist | can somebody help me with it? |
| 02:17.35 | jaytee | vonkleist, is this a new setup or was it working before? |
| 02:17.45 | vonkleist | new setup |
| 02:18.54 | vonkleist | we've a MFC/R2 on the other side |
| 02:19.54 | jaytee | the output indicates the TE121 card is having problems with interrupts. I'd call Digium's tech support. |
| 02:20.09 | vonkleist | :O |
| 02:21.10 | vonkleist | is it because of the red alarm? |
| 02:21.20 | jaytee | vonkleist, one thing to try is to check what interrupt it's set to and make sure it's not sharing it. |
| 02:21.31 | vonkleist | ok |
| 02:21.35 | vonkleist | let me check |
| 02:22.45 | vonkleist | http://gera.pastebin.ca/1181251 |
| 02:23.23 | vonkleist | it seem's it's shared with uhci_hcd and wctdm |
| 02:24.22 | jaytee | vonkleist, check to see if you can change the assignment in your system's BIOS setup |
| 02:24.44 | vonkleist | ouuuu |
| 02:24.47 | vonkleist | remote machine... |
| 02:24.50 | vonkleist | that's baaad |
| 02:24.54 | jaytee | yep |
| 02:25.05 | jaytee | how remote? |
| 02:25.15 | vonkleist | so remote |
| 02:25.52 | jaytee | was it tested before it was deployed or did someone set it up there and install the card? |
| 02:26.04 | vonkleist | No, it wasn't tested before |
| 02:26.22 | jaytee | vonkleist, what can I say? sucks to be you |
| 02:27.35 | jaytee | I'd still give Digium a call if I were in your shoes. They can verify your configuration and might have a workaround |
| 02:28.28 | vonkleist | ok |
| 02:28.30 | vonkleist | will try that |
| 02:28.33 | vonkleist | thank you |
| 02:31.26 | jaytee | damn!! I got chips and salsa take-out from Chili's and forgot to pick up some Tums earlier. bbiab |
| 02:31.58 | jblack | Milk can make for a standby. |
| 02:32.04 | jeev | barfs |
| 02:32.32 | jaytee | milk? what's that? |
| 02:32.36 | *** join/#asterisk fordfrog (n=fordfrog@gentoo/developer/fordfrog) |
| 02:33.07 | jaytee | oh, wait! that's that stuff that the Half and Half I put in my coffee is partly made of :-) |
| 02:33.43 | jaytee | this old carcass needs the extra strength shit to put the flames out. milk wouldn't even dent my heartburn |
| 02:33.53 | jaytee | but thanks for the suggestion |
| 02:34.00 | jeev | heart burn is for losers |
| 02:37.17 | *** join/#asterisk axisys (n=axisys@ip68-98-177-71.dc.dc.cox.net) |
| 02:45.44 | *** join/#asterisk xenonex (n=xenonex@89.218.236.51) |
| 02:48.59 | *** join/#asterisk xenonex (n=xenonex@89.218.236.51) |
| 02:49.03 | drmessano | jeev=trollbair |
| 02:49.05 | drmessano | jeev=trollbait |
| 02:52.25 | vonkleist | hey, jaytee |
| 02:52.34 | jaytee | vonkleist, yeah? |
| 02:52.36 | vonkleist | I unloaded the conflicting modules and see: http://gera.pastebin.ca/1181264 |
| 02:52.51 | *** join/#asterisk xenonex (n=xenonex@89.218.236.51) |
| 02:55.51 | vonkleist | does it look the same? |
| 02:55.56 | jaytee | vonkleist, yes |
| 02:56.24 | drmessano | 1.6 is such a ripoff |
| 02:56.25 | jaytee | vonkleist, did you restart zaptel ? |
| 02:56.30 | drmessano | Same friggin GUI |
| 02:56.35 | vonkleist | no |
| 02:56.42 | jaytee | vonkleist, do it |
| 02:56.56 | drmessano | 1.0 --> 1.2 --> 1.4 --> 1.6 <--- SAME GUI |
| 02:57.08 | jaytee | and the GUI sucks |
| 02:57.26 | vonkleist | ok, let me try |
| 02:57.31 | drmessano | Apparently someone at Digium loves MS-DOS |
| 02:57.34 | drmessano | Same samn UI |
| 02:57.36 | jaytee | I want a gui that will let me add sip accounts and extensions and leave the damn extensions.conf alone |
| 02:58.19 | jaytee | but everything I've seen overwrites extensions.conf, sip.conf etc. after each reboot |
| 02:58.45 | Alton2 | drink a little baking soda in water to kill the acid |
| 02:58.49 | Alton2 | works for me |
| 02:59.00 | jaytee | I bought Tums already so lets move on |
| 02:59.12 | Alton2 | suit yourself |
| 02:59.20 | *** join/#asterisk oilinki (n=oil@ppp-124-120-1-131.revip2.asianet.co.th) |
| 02:59.33 | jaytee | but thanks for the idea. if I'm ever out again I'll try it |
| 02:59.42 | Alton2 | thaniks |
| 02:59.43 | Alton2 | thanks |
| 03:01.31 | vonkleist | jaytee, I unloaded zaptel |
| 03:01.33 | vonkleist | now what? |
| 03:02.19 | jaytee | reload it then reload asterisk and do zap show status again |
| 03:02.30 | jaytee | and look for errors when reload zaptel |
| 03:02.54 | vonkleist | zttool still shows a red alarm on the te121 |
| 03:03.19 | jaytee | something is amiss in your config then |
| 03:04.23 | jaytee | and since it appears you're using E1 BRI I'm not gonna be any help in that area. |
| 03:04.51 | jaytee | since it's a remote site you can't even verify cabling issues. |
| 03:05.13 | JT | E1 BRI? |
| 03:05.13 | vonkleist | yeah, that's right... |
| 03:06.11 | vonkleist | Tomorroy I'll make somebody go there and check if there's still any cable connected to the asterisk box |
| 03:06.36 | JT | jaytee: no such thing as an E1 BRI |
| 03:06.56 | jaytee | JT, ok, just E1 then |
| 03:07.43 | jaytee | I don't mess with E1's since I'm in the US |
| 03:08.07 | JT | he mentioned something about MFC/R2, so not sure if it's E1 PRI or E1 MFC/R2 |
| 03:08.16 | JT | sucks to be stuck in a country using E1 MFC/R2 |
| 03:08.32 | vonkleist | JT, it's MFC/R2, as long as I know |
| 03:09.40 | JT | vonkleist: are you using chan_unicall? |
| 03:10.10 | vonkleist | that was the first thing we tried |
| 03:10.20 | vonkleist | We found a page about that and tried it |
| 03:10.39 | vonkleist | later we saw that it (the page) was writen 2 years ago |
| 03:11.06 | vonkleist | isn't there any universal meaning for a red alarm? |
| 03:11.58 | JT | yes |
| 03:11.59 | jaytee | red alarm could indicate many critical condition. lost of clock timing, frame slips exceeding threshold setting etc. |
| 03:12.00 | JT | no signal |
| 03:12.13 | JT | make sure your cable is correct |
| 03:12.24 | jaytee | JT, it's a remote site |
| 03:12.40 | JT | well it's a red alarm, someone will need to go plug it in |
| 03:12.43 | JT | or check it |
| 03:13.27 | vonkleist | ok |
| 03:13.37 | *** join/#asterisk xenonex (n=xenonex@89.218.236.51) |
| 03:14.04 | vonkleist | the telco provider put the cable there, but well... I wouldn't swear that cable is still connected... |
| 03:14.42 | jaytee | if the cable is unplugged or the wrong type of cable it would red alarm |
| 03:15.17 | vonkleist | could we say that it's the most common cause of red alarms? |
| 03:15.47 | jaytee | usually yeah, or something screwed up in the provisioning at the telco's end that they'll never admit to |
| 03:16.03 | vonkleist | damn... |
| 03:16.15 | vonkleist | ok... there's so many work to do there, then... |
| 03:17.04 | jaytee | that's why I love having a dual port T1/E1 card. I can set the signalling on one span to be PRI_NET and use a crossover cable to test the card vs. the telco's circuit. |
| 03:21.14 | vonkleist | well guys... you've been a GREAT HELP for me... thank you very much... |
| 03:21.41 | jaytee | don't see how but you're welcome anyways and good luck with it |
| 03:22.01 | jaytee | vonkleist, where you at if you don't mind me asking? |
| 03:22.17 | vonkleist | mexico... :S |
| 03:22.32 | jaytee | ah, cool |
| 03:22.34 | vonkleist | with a big and monopolic telco... |
| 03:23.06 | vonkleist | using mfc r2 |
| 03:23.08 | vonkleist | :P |
| 03:23.57 | jaytee | I live in Indianapolis and my neighborhood is predominately hispanic |
| 03:24.24 | vonkleist | indianapolis?? really? |
| 03:25.06 | jaytee | lots of good taquerias around and I can get real Coca-Cola with cane sugar at the supermercado instead of the regular US Coca-Cola with high fructose corn syrup. |
| 03:25.25 | jaytee | family across the hall from me is from mexico |
| 03:25.28 | vonkleist | hahahaha |
| 03:25.38 | vonkleist | really??? |
| 03:25.54 | jaytee | lotta illegals here but most of us don't care. "illegal" is a stupid concept anyways. |
| 03:26.22 | vonkleist | mainly if they apply it to human beings, don't you think? |
| 03:26.43 | moy | vonkleist: if you have red alarm or yellow alarm it has nothing to do with R2 .... is this Telmex? |
| 03:26.56 | vonkleist | moy, you're right... |
| 03:27.11 | vonkleist | are u the same moy that wrote *that* article?? |
| 03:27.22 | *** join/#asterisk obnauticus (n=obnautic@about/windows/regular/obnauticus) |
| 03:27.40 | moy | hum, don't know what article you mean? I wrote some stuff about R2, but not sure if it's the same you are referring to |
| 03:27.49 | vonkleist | I think so |
| 03:27.51 | jaytee | http://www.voip-info.org/wiki/view/Asterisk+MFC+R2 |
| 03:27.57 | vonkleist | I think I saw it on zarzamora |
| 03:28.20 | jaytee | dunno if you have looked at that or if it would help you any |
| 03:29.36 | *** join/#asterisk xenonex (n=xenonex@89.218.236.51) |
| 03:29.40 | moy | vonkleist: nope, the zarzamora article was written by other guy ... I wrote an R2 troubleshooting PDF, nothing on installation properly |
| 03:29.46 | vonkleist | jaytee, thanks... that's one of the articles we folllowed, with no luck |
| 03:29.51 | vonkleist | ah, got it |
| 03:30.05 | vonkleist | do you think that the red alarm could something more physical? |
| 03:30.43 | moy | definitely, like having the connections wrong, some ppl put the balun connections backwards :P |
| 03:30.44 | vonkleist | moy, this is telmex, indeed |
| 03:31.10 | vonkleist | didn't got it... |
| 03:31.11 | *** join/#asterisk dmz (n=dmz@12.149.3.162) |
| 03:31.14 | vonkleist | balun connections? |
| 03:31.44 | moy | how are you connecting the E1? you know what a balun is? |
| 03:32.02 | vonkleist | :S |
| 03:32.16 | vonkleist | I declare myself ignorant on this issue |
| 03:32.38 | *** join/#asterisk jameswf-home (n=james@ip72-223-0-183.ph.ph.cox.net) |
| 03:32.46 | vonkleist | it seems telmex left an alcatel demux with a 20ohms output |
| 03:32.50 | vonkleist | RJ45 |
| 03:32.51 | jaytee | vonkleist, it's basically a passive transformer for impedance matching between balanced and unbalanced circuits |
| 03:33.07 | vonkleist | but... |
| 03:33.08 | vonkleist | ummm |
| 03:33.30 | moy | as jaytee said, telmex usually leaves you a pair of coaxial connectors |
| 03:33.32 | vonkleist | I mean... got it, but don't know how this rf45 connector can be put backwards |
| 03:33.36 | moy | which you connect to the balun |
| 03:33.46 | moy | and then from the balun to the E1 card via Rj45 |
| 03:34.10 | vonkleist | ok, let me check with the guy that was there when they connected it |
| 03:34.29 | moy | do you have access right now to the E1? |
| 03:34.50 | jaytee | moy, his system is at a remote location |
| 03:34.53 | vonkleist | moy, no it's far way from here |
| 03:35.04 | vonkleist | far away, I mean |
| 03:35.32 | vonkleist | But we only told telmex where to connect the rf45 connector |
| 03:36.20 | moy | I see, well, I believe they connected the wrong cable, all Telmex installations (or Axtel, Maxcomm whatever), have 2 coaxial connectors (rx and tx) where you should plug the balun, on the other side of the balun there is a rj45 connection that should go to the E1 |
| 03:36.38 | vonkleist | yes |
| 03:37.37 | moy | so, that's most likely your issue, wrong connections ... if you ever one to try MFC/R2 in chan_zap btw, you can go here http://www.libopenr2.org/ |
| 03:37.46 | moy | s/one/want |
| 03:38.40 | vonkleist | moy, shouldn't we HAVE TO go with mfc/r2? |
| 03:39.09 | moy | hu? |
| 03:39.49 | vonkleist | I mean... we have a MFC R2, so I think that's anyway the way to go... isn't it? |
| 03:40.19 | moy | there is 2 ways of using MFC/R2 in Asterisk (that I know of), it's using Unicall or OpenR2 |
| 03:40.30 | vonkleist | is unicall still alive? |
| 03:40.43 | moy | with Unicall you install chan_unicall, with OpenR2 you use chan_zap |
| 03:40.50 | *** part/#asterisk lowlevel (n=Stuart@lowlevel.ca) |
| 03:40.53 | moy | yes, unicall as a framework is alive |
| 03:41.05 | vonkleist | I think we already compiled with libopenr2 |
| 03:41.36 | moy | and I give occasional maintenance to chan_unicall driver, but I spend most of my time now developing the libopenr2 implementation |
| 03:42.23 | vonkleist | confirmed... we compiled using libopenr2 |
| 03:42.38 | jblack | Anybody know of any public conferences? |
| 03:42.47 | jblack | chat lines, so to speak? |
| 03:42.59 | moy | ah ok, nice :) ... lemme know if you have any problems with it, I want to squeeze out all bugs ASAP |
| 03:43.47 | vonkleist | well, yes... zap channel status shows a horrible red alarm and... |
| 03:43.49 | vonkleist | hehehehe |
| 03:44.11 | drmessano | jblack: Doesnt sipphone/gizmo5 have a few? |
| 03:44.38 | moy | oh well, but that is not a bug :) ... but user error, or sounds like it anyway |
| 03:44.55 | jblack | It's been broken for years, and unused. They play loud jungle sounds on them. |
| 03:46.14 | jblack | I'm always mildly surprised we don't hae one for this channel. |
| 03:47.00 | *** join/#asterisk [T]ank (n=chwall@71-219-148-249.slkc.qwest.net) |
| 03:47.49 | [T]ank | i am used to dialing using IAX2 and I think the syntax should be different for sip. here is what I am trying to do: |
| 03:48.16 | [T]ank | exten => _NXXXXXX,n,Dial(SIP/XXXXXXXXXX:passwod@sip.binfone.com/1801${EXTEN}||) |
| 03:48.22 | [T]ank | sip/username |
| 03:48.56 | drmessano | jblack: Not possible.. everyone in here has a broken system |
| 03:49.05 | jblack | Not the regulars. |
| 03:50.08 | *** join/#asterisk blepsoaf (n=pbaker@ool-43562836.dyn.optonline.net) |
| 03:50.14 | drmessano | Joke? |
| 03:50.35 | jblack | Your system works. Mine works. TK's works. I'm sure most do. |
| 03:50.46 | jblack | Your system does work, no? |
| 03:50.47 | jaytee | my system works |
| 03:51.27 | jaytee | I could VPN into work and pull CDR data to tell you how many calls it handled today |
| 03:51.32 | jaytee | but I'm lazy |
| 03:51.37 | jaytee | and it's bedtime |
| 03:51.39 | Maliuta | my system works |
| 03:51.53 | jblack | [T]ank: Asterisk doesn't define the format of the sip url. |
| 03:52.09 | jaytee | if I want a broken system I'll just add asterisk-gui |
| 03:52.21 | jaytee | or if I want to totally hose it up, freepbx |
| 03:53.00 | jblack | I haven't heard a human voice over the ate of 14 in 3 days. Unless you count the ex, who Im' not sure is human. |
| 03:53.25 | jaytee | I detect bitterness |
| 03:53.29 | mchou | haha |
| 03:53.40 | jblack | No. I have nothing but respect for that slut. |
| 03:53.55 | jaytee | it's time to forgive and move on, just forget the bitch |
| 03:53.59 | mchou | jblack: http://www.neospeech.com/ |
| 03:54.11 | [T]ank | jblack: trying to find the right syntax however... |
| 03:54.16 | mchou | jblack: give 'kate' a try |
| 03:54.31 | jblack | jaytee: Not a choice. My daughter is a blood relation to the demon |
| 03:54.39 | jblack | mchou: You showed me that last night. |
| 03:54.41 | jaytee | I've managed to get my emotional baggage down to the size of a small carryon but when we first split I was hauling the equivalent of a full set of Tourister luggage. |
| 03:54.53 | mchou | jblack: haha, ok |
| 03:55.06 | jblack | It's cool. |
| 03:55.12 | *** join/#asterisk SanityIO (n=SanityIO@77.242.105.93) |
| 03:55.13 | *** join/#asterisk SanityIO_ (n=SanityIO@77.242.105.93) |
| 03:55.24 | *** join/#asterisk vonkleist (n=gcontrer@201.116.65.115) |
| 03:55.26 | jblack | wonders how many concurrent calls fits into 300 gigabytes. |
| 03:56.55 | vonkleist | moy, uh oh... |
| 03:56.55 | jblack | redoes the math |
| 03:57.04 | vonkleist | It seems somebody did something nasty |
| 03:57.16 | vonkleist | take a look to what we found when we got there |
| 03:57.17 | vonkleist | http://gera.com.mx/IMG_0004.jpg |
| 03:57.37 | jaytee | WTF? a coaxial loopback? |
| 03:57.52 | moy | haha, that's funny |
| 03:57.54 | vonkleist | that's the way the telco left this |
| 03:58.04 | vonkleist | worse: |
| 03:58.29 | vonkleist | the guy that went to work with the * box got out the loop cable and left the coaxial cables alone |
| 03:58.33 | vonkleist | and... |
| 03:59.05 | vonkleist | changed the rj45 cable to the 120ohms port |
| 03:59.26 | vonkleist | so... |
| 03:59.28 | jblack | That's more like it. 82KB a sec... figure 128kbit for ulaw (64 each dir), that works about to 5 concurrent calls. |
| 04:00.03 | moy | that could make more sense, if you see, there is a G703 line covering both the coaxials and the rj45 with 120ohm port |
| 04:00.27 | vonkleist | ok, so connecting to that rj45 port is good? |
| 04:00.31 | moy | I don't know what the other port is for, but eth sounds like screaming, ethernet here, not E1 |
| 04:00.38 | vonkleist | ok |
| 04:00.43 | vonkleist | but still... |
| 04:00.48 | vonkleist | what about the coaxial cables??? |
| 04:01.01 | *** join/#asterisk [T]an2 (n=chwall@71-219-131-152.slkc.qwest.net) |
| 04:01.03 | vonkleist | right now they are unterminated... |
| 04:01.16 | vonkleist | the connectors are connected to nothing |
| 04:01.30 | moy | regarding the 120 ohm rj35 port, I don't know ... I'd connect the coaxials to the balun (that you need to buy) and from the balun to the card connect the rj45 cable |
| 04:01.38 | *** join/#asterisk [intra]lanman (n=lanman@freeswitch/developer/intralanman) |
| 04:01.56 | vonkleist | oh |
| 04:02.00 | moy | those 2 coaxials need to go to the balun (I believe is 120 to 75 ohms) |
| 04:02.20 | moy | and in the other side of the balun there's a rj45 port, connect the cable there to the card |
| 04:02.44 | vonkleist | so, what should we do with the 120ohm port? |
| 04:02.52 | moy | http://www.data-connect.com/Patton_460.htm |
| 04:02.59 | JT | why would you need to get a balun? |
| 04:03.01 | moy | I believe that's the one |
| 04:03.04 | JT | simply unplug the coax |
| 04:03.20 | JT | plug a patch cable unto the 120Ohm socket, and connect the other end to your digium card |
| 04:03.38 | JT | if that doesn't work, try an E1 crossover cable instead of normal patch cable |
| 04:03.46 | JT | if neither works, get a balun |
| 04:03.56 | *** part/#asterisk [T]an2 (n=chwall@71-219-131-152.slkc.qwest.net) |
| 04:03.59 | moy | JT, I never connected directly, so I did not want to advice on something I had never used :P |
| 04:04.16 | *** join/#asterisk [T]an2 (n=chwall@71-219-131-152.slkc.qwest.net) |
| 04:04.16 | jaytee | are the pinouts for an E1 RJ45 the same as a T1? |
| 04:04.18 | vonkleist | JT, right now, the coax connectors are connected to nothing |
| 04:04.24 | JT | moy: it's cheaper to test my theory first |
| 04:04.29 | moy | sure |
| 04:04.42 | JT | vonkleist: if you don't have the modem hooked to anything, leave the coax plugged in |
| 04:04.44 | vonkleist | and the cat5 cable is connected from the 120ohm port to the digium card |
| 04:04.50 | JT | otherwise they telco may turn down the link |
| 04:04.55 | JT | due to alarms on their NMS |
| 04:04.59 | vonkleist | oh! |
| 04:05.03 | JT | oh |
| 04:05.08 | JT | you have cat5 hooked up? |
| 04:05.10 | JT | that's ok |
| 04:05.13 | JT | still red alarm? |
| 04:05.36 | vonkleist | yes, I have it hooked up |
| 04:05.38 | vonkleist | still red alarm |
| 04:05.48 | JT | go back to loopback while you get/make a E1 crossover cable |
| 04:06.28 | *** join/#asterisk aliver (n=aliver@ip-216-17-149-97.rev.frii.com) |
| 04:06.47 | jaytee | if the pinouts for E1 are the same as T1 it's pin 1 to pin 4 and pin 2 to pin 5 |
| 04:06.54 | JT | it is the same |
| 04:07.31 | aliver | I want to make several phones ring when one extension is called. Can I do that with extensions.conf ? Also, once it's picked up the other two people should ring if the extension is called again. Will that work? |
| 04:07.37 | [T]an2 | ok, so if i have a sip friend from a sip provider. I am registered to that provider and can receive calls. How do i set up the outbound dial. I can do the iax just fine, just not sip. I have never dialed out using sip before. I am doing "exten => XXXXXXX,1,Dial(SIP/user:pass@host/${EXTEN}) |
| 04:07.56 | vonkleist | aliver, you could use a queue to do that |
| 04:08.08 | jblack | aliver: exten => exten,n,Dial(first1&second2&thirdone2,60) |
| 04:08.13 | aliver | That looks complicated. I'll do it, but is there an easier way? |
| 04:08.25 | aliver | jblack now that looks easy! |
| 04:09.11 | jblack | I suggest you consider trying just one line for 5 seconds first. |
| 04:09.21 | jblack | It can get annoying to have 10 phones go off all at the same time. |
| 04:09.26 | mchou | hey, asterisk macros can be called with a variable number of arguments, right? I mean if you define a specific macro that takes 3 arguments, is it ok to call Macro(mymacro(1))? (as long as arg2 and arg3 are verified to be empty within the defined macro) |
| 04:09.45 | jblack | Yes, macros can call macros. |
| 04:10.05 | mchou | jblack: umm, that's not what I asked |
| 04:10.10 | jblack | i know it works when you pass through ${ARG1} ${ARG2} etc etc. |
| 04:10.47 | mchou | I meant Macro(macro,1)) |
| 04:10.51 | jblack | I don't know that you can just transparently forward all macros. |
| 04:10.57 | jblack | pardon, all arguments. |
| 04:12.34 | mchou | jblack: are you and I talking about the same thing? |
| 04:12.50 | jblack | possibly not. |
| 04:13.07 | jaytee | it may sound transparent but it sure looks opaque :-) |
| 04:13.24 | vonkleist | Well... we got out the loop last friday, so it may be the telco just disconnected us because of the alarms? |
| 04:13.35 | jaytee | vonkleist, could be |
| 04:13.58 | mchou | I want to know if * macro can be kind of like like a c function that takes a variable list of arguments |
| 04:14.30 | jblack | that's where we're not on the same page. Last I checked, you didn't explicitely declare macro arguments at all. |
| 04:14.34 | mchou | jblack: kind of like how printf might work in c, if you get my droft |
| 04:14.36 | [TK]D-Fender | mchou: a macro is not a "function" and does not have a return value. |
| 04:15.08 | [TK]D-Fender | mchou: Dialplan is not a higher level language. *I'VE* written better parsers. |
| 04:15.09 | mchou | [TK]D-Fender: well, that why I said "kind of" |
| 04:15.14 | jblack | I would think that if you pass MacroA 5 arguments, you'd have to call MacroB with 5 arguments as well. |
| 04:15.39 | jaytee | nite all, don't stay up too late |
| 04:15.42 | vonkleist | Ok, so... having this physical setup: http://gera.com.mx/IMG_0004.jpg (without the coax loop and having the cat5 cable hooked to the 120ohm connector) , what do you recommend me to do next? |
| 04:16.01 | jblack | It's actually C like. If a variadic function calls a variadic function, you still have to pass the arguments you want to go through. |
| 04:16.32 | jblack | In this case, s/function/macro |
| 04:17.19 | [TK]D-Fender | And when you think about nesting macros just remember that there is no "stack" for the ARGS. They will mash the shit out of each other. |
| 04:17.47 | jblack | To try to be more clear, you want MacroA(One,Two,Three), and MacroA calls MacroB, you want it to get One, Two, Three with a MacroB, and not a MacroB(${ARG1},${ARG2},${ARG3}), correct? |
| 04:17.52 | *** join/#asterisk tzafrir_laptop (n=tzafrir@bzq-179-75-202.static.bezeqint.net) |
| 04:18.05 | mchou | nono |
| 04:18.15 | mchou | there is only MacroA |
| 04:19.18 | mchou | is MacroA variadic in the sense exten=>12345,Macro(1,2,3) |
| 04:19.23 | jblack | Are you looking for an iterator for macro arguments ? |
| 04:19.58 | mchou | and exten=>12345,Macro(1) legit? |
| 04:20.17 | jblack | You can call macro with any number of arguments you want. |
| 04:20.21 | mchou | sorry, meant to say Macro(MacroA(...)) |
| 04:20.45 | [TK]D-Fender | mchou: Your second set of () is not valid |
| 04:20.55 | mchou | jblack: I mean here a SPECIFIC macro |
| 04:21.10 | [TK]D-Fender | mchou: You are not being clear at all. |
| 04:21.28 | jblack | Yes. You can right now, call Macro(MacroA), then 5 extensions later, call Macro(MacroA,1) and yet again, call Macro(MacroA,1,2,3) |
| 04:21.47 | mchou | what's not clear about variadic function? |
| 04:21.50 | jblack | It's up to your argument to make sure to not use arguments it's not given, of course. |
| 04:22.10 | jblack | pardon, it's up to your macro to make sure you don't use arguments that don't exist. |
| 04:22.22 | mchou | jblack: bingo! |
| 04:22.33 | jblack | Ok. Then you weren't clear in the least. |
| 04:22.41 | jblack | just feedback |
| 04:23.03 | [TK]D-Fender | jblack: and watcht he fun when you pass 3 args on the first call, and only 2 on the next only to miss that ARG3 is still SET from the first. |
| 04:23.10 | mchou | dude, that why I gave the two exten examples |
| 04:23.38 | jblack | don't get snitty with someone trying to help. :P Besides, I answered that question at least once, and I think twice. :) |
| 04:24.00 | jblack | 00:14 < jblack> that's where we're not on the same page. Last I checked, you didn't explicitely declare macro arguments at all. |
| 04:24.05 | [TK]D-Fender | mchou: And you are still not clear. Your wording goes in circles. |
| 04:24.10 | jblack | That naturally follows from macros not having prototypes |
| 04:24.14 | mchou | jblack: I didn't get snitty. you answered the wrong question |
| 04:24.31 | jblack | I answered many wrong questions. |
| 04:24.38 | mchou | jblack: you though I meant nested macros |
| 04:24.42 | [TK]D-Fender | And there is nothing "explicit" about macros period. All of * dialplan is the most adhoc scope-less mess ever |
| 04:24.54 | mchou | question have zip to do with nesting |
| 04:25.26 | jblack | Ok. Well, I'm sorry that I wasted your time. I'll be sure to remain hushed the next time you're looking for help. |
| 04:26.17 | jblack | [TK]D-Fender: If I put up a permament conference for the channel, do you think it would get use? |
| 04:26.44 | mchou | jblack: lol. read how I phrased my question originally. There is no reference to nesting, only variable # of arguments |
| 04:26.49 | *** join/#asterisk kamanashisroy (n=kamanash@202.56.7.131) |
| 04:27.01 | jblack | mchou: Fine. You were right, I was wrong. Can you leave me alone now? |
| 04:27.03 | [TK]D-Fender | mchou: And I do now get your question. Yes it is safe to call a macro with any number of arguments. |
| 04:27.24 | [TK]D-Fender | mchou: Where safe = dangerous. |
| 04:27.26 | [TK]D-Fender | ^^^ |
| 04:28.05 | *** join/#asterisk Bonix (n=Bonix@196-lo1.rt2.isimples.com.br) |
| 04:28.13 | [TK]D-Fender | mchou: Lets say you call 1 macro and set 3 ARGS. then you call ANOTHER macro (no, not nested or anything), and pass it 2 args. ARG3 is STILL FILLED from the first time you called a macro with a 3rd arg |
| 04:28.48 | jblack | Huh. I didn't know that. |
| 04:29.04 | mchou | that makes no sense whatsoever |
| 04:29.08 | [TK]D-Fender | mchou: So you better know exactly what SNAFU's you might run into when you start trying to use this stuff. |
| 04:29.40 | mchou | where do the arguments of a macro "live?" |
| 04:29.48 | [TK]D-Fender | mchou: Exten => 123,1,Macro(macroone,a,b,c) |
| 04:30.01 | [TK]D-Fender | mchou: Exten => 123,2,Macro(macrotwo,a,b) |
| 04:30.14 | mchou | what macrotwo |
| 04:30.22 | mchou | it's still macroone |
| 04:30.24 | [TK]D-Fender | mchou: Exten => 123,3,NoOp($(ARG3)) <--- will give you "c" |
| 04:30.33 | mchou | on the second line |
| 04:30.35 | [TK]D-Fender | mchou: DIFFERNT LESSON |
| 04:30.42 | jblack | chuckles |
| 04:30.47 | *** join/#asterisk bbryant (n=brett@c-68-59-20-153.hsd1.sc.comcast.net) |
| 04:31.07 | [TK]D-Fender | mchou: I announced this twice. Pay attention. This is highlighting the fact that calling 2 macros can FUBAR you due to "leftovers" |
| 04:31.31 | mchou | I am paying attention. [21:29] <mchou> where do the arguments of a macro "live?" |
| 04:31.43 | jblack | [TK]D-Fender: Gee, it's as if you're saying arguments aren't reset between macro calls |
| 04:32.00 | [TK]D-Fender | mchou: and the term "arguments" is not valid. They are assigned to completely dumb, and scope-less CHANNEL VARIABLES. |
| 04:32.15 | mchou | what?? |
| 04:32.16 | mchou | lol |
| 04:32.23 | [TK]D-Fender | jblack: OMG, almost like I used LANGUAGE to convey coherent theories! Oh noes!!! |
| 04:32.31 | jblack | ORLY? |
| 04:32.34 | mchou | consider * documentation calles them arguments |
| 04:32.36 | [TK]D-Fender | RLY! |
| 04:32.44 | jblack | OWOW KTHX BAI! |
| 04:32.55 | [TK]D-Fender | mchou: A junk term. Stop thinking so much. Dialplan is DUMB SHIT |
| 04:33.19 | [TK]D-Fender | mchou: Think linear or it will bite you in the ass. |
| 04:33.31 | jblack | This is why I think the channel needs a conference, so that we can lart people |
| 04:33.42 | mchou | lart? |
| 04:33.48 | mchou | what's that? |
| 04:33.50 | [TK]D-Fender | jblack: I swear sometimes I just want to kill people. |
| 04:33.51 | aliver | It seems like only patterns at the top of my extension.conf contexts are matching. |
| 04:33.56 | [TK]D-Fender | ~lart mchou |
| 04:33.56 | jbot | shoots mchou in his sleep |
| 04:34.04 | aliver | Is there a likely reason for that? |
| 04:34.16 | [TK]D-Fender | aliver: load res_psychic.so |
| 04:34.20 | jblack | aliver: do you know what contexts are yet? |
| 04:34.31 | aliver | yes, I think so. |
| 04:34.45 | aliver | "[iamAcontext]" |
| 04:34.59 | aliver | things pointed to from sip.conf context= |
| 04:35.04 | jblack | You won't go between contexts (normally a dialplay will have anywhre from half a dozen to a couple dozen) without a Goto to jump between them. |
| 04:35.04 | aliver | and stuff like Goto() |
| 04:35.10 | jblack | Ok. Good. |
| 04:35.28 | jblack | It sounds to me like you aren't doing gotos, but you know what goto is, so please pastebin your dialplan |
| 04:35.36 | JT | vonkleist: it's not still plugged into ETH is it? |
| 04:35.49 | aliver | okay just a sec, and thanks. |
| 04:35.57 | jblack | aliver: You can also run asterisk -r, set debug 10, set verbose 10, and watch calls happen |
| 04:36.06 | mchou | [TK]D-Fender: so this is where I'm messed up. What's the mechanism in a dialplan to pass variable arguments into a macrolike routine where some inner function/application might require vaiable arguments? |
| 04:36.23 | mchou | variable* |
| 04:37.36 | [TK]D-Fender | mchou: You can pass as many args as you want to a macro (within some sort of general limit). its that they are not PROTECTED from carrying into another call. |
| 04:37.38 | aliver | http://pastebin.com/d4cf3dff1 |
| 04:37.57 | aliver | jblack the sip debug stuff just DoS's me I can't tell what the hell is happening. |
| 04:38.08 | mchou | [TK]D-Fender: like say for example I call Dial() in some macro-like routine |
| 04:38.15 | jblack | Uh, where did the ascii art come from? |
| 04:38.27 | aliver | http://pastebin.com/d4cf3dff1 <-- [internalphones] doesn't get past the first few patterns |
| 04:38.34 | aliver | figlet made the ascii titles |
| 04:38.42 | [TK]D-Fender | aliver: HUH?!?! |
| 04:38.52 | jblack | Ok, so we don't have some funky front end generating this, right? |
| 04:40.00 | mchou | [TK]D-Fender: Dial takes variable arguments. In the same macro I want to call Dial different ways...... |
| 04:40.00 | [TK]D-Fender | mchou: please explain. That last one did not encmpass a complete idea. |
| 04:40.00 | *** join/#asterisk steliosk (n=Stelios@athedsl-123365.home.otenet.gr) |
| 04:40.00 | [TK]D-Fender | mchou: Macros have NO fixed number of "arguments". |
| 04:40.00 | aliver | jblack no, I pasted that stuff in. |
| 04:40.07 | aliver | figlet is just a CLI tool. |
| 04:40.10 | jblack | aliver: Ok, what context is jims in? |
| 04:40.36 | [TK]D-Fender | mchou: You could call Macro(mymacro,123456) in one place and Macro(mymacro,SIP/100,2,abc,wtf) somewhere else. |
| 04:40.49 | aliver | jblack he's in the default context called internalphones |
| 04:41.05 | jblack | Ok. And jims gets stuck in internalphones, 'a few lines in', right? |
| 04:41.36 | [TK]D-Fender | mchou: It doesn't matter if you pass anything along with a macro or not. If you DO, then those channel vars get set/updated. |
| 04:42.09 | aliver | jblack well for example, I can call in from outside and ring that line. However, if I try to dial out, it says 404 not found even though I know that the pattern exists in internalphones. |
| 04:42.22 | mchou | [TK]D-Fender: so where is the scope issue? |
| 04:42.59 | jblack | Yeah. Well, did you notice yet that you only have 1 context? |
| 04:43.17 | [TK]D-Fender | aliver: ok, I'm going to try not to hold back too much..... SHOW US THE FUCKING FAILURE. No, we DON't trust your description, or your pastebin, or the "content" of what you claim is coming in. SHOW US. |
| 04:43.17 | jblack | as far as I can see, you only have [internalphones] |
| 04:43.29 | aliver | jblack sure, and I could change that, but shouldn't I be able to have more than one pattern per context? |
| 04:43.49 | jblack | This is a mess. |
| 04:44.03 | aliver | no problem I can paste the failure. |
| 04:44.04 | jblack | You can, but it gets confusing. |
| 04:44.16 | jblack | aliver: Here's what I think you should have, which I learned from tk a long while ago. |
| 04:44.23 | jblack | Have a [phones] context, that just list phones. |
| 04:44.34 | [TK]D-Fender | aliver: you can have a MILLION patterns. Hopefully you are dumb enough to make them overlap and fight with each other. |
| 04:44.40 | jblack | With simple dials |
| 04:44.53 | jblack | THen, have an [internal] context, and include phones. |
| 04:45.09 | jblack | then, have a [public] context, which can have your ivr, and include [phones] |
| 04:45.10 | [TK]D-Fender | mchou: Calll ANY macro passing 3 args. then call another macro (not nested even) and pass it only 2. ARG3 is STILL POPULATED |
| 04:45.23 | jblack | Then, have an [outgoing] context, that sets up dialing out. |
| 04:46.31 | aliver | http://pastebin.com/d15fc820a <-- okay there is the actual failure |
| 04:46.42 | aliver | what idiocy have I engaged in to make this happen? |
| 04:47.03 | mchou | [TK]D-Fender: so what if on 2nd call of macro I just do this? Macro(Macroone,1,2,) |
| 04:47.06 | [TK]D-Fender | aliver: Looking for 6908268 in internalphones (domain sip.coloradovnet.com) |
| 04:47.18 | mchou | [TK]D-Fender: Is that legit? |
| 04:47.28 | [TK]D-Fender | aliver: what line in this pastebin of yours do you think it should match? http://pastebin.com/d4cf3dff1 |
| 04:47.38 | aliver | jblack are you saying that I should have another context for dialing outbound, what difference will that make since the patterns will match either way, right? |
| 04:47.53 | [TK]D-Fender | mchou: yes legit, and sets ARG3 to "blank" |
| 04:48.04 | aliver | [TK]D-Fender lml |
| 04:48.11 | [TK]D-Fender | aliver: lml? |
| 04:48.15 | [TK]D-Fender | nvm |
| 04:48.36 | aliver | 112 |
| 04:48.40 | jameswf-home | i like my dialplans to be modular |
| 04:48.45 | [TK]D-Fender | aliver: You shouldn't have to look. You seem to think it should work so you should already have the answer on what you thought it should have matched |
| 04:49.02 | aliver | Just trying to answer your Q |
| 04:49.16 | aliver | I think it's all screwed up, too. |
| 04:49.24 | jblack | aliver: http://pastebin.com/mcf4a728 |
| 04:49.26 | [TK]D-Fender | aliver: You think "exten => _NXXXXXX,1,Dial(SIP/+1970${EXTEN}@bandwidth.com_outbound,60)" should match "Looking for 6908268 in internalphones" ??? |
| 04:49.28 | aliver | I think line 112 would match. Why do you believe it wont? |
| 04:49.32 | jblack | here's a quick, not proof-read, example |
| 04:49.46 | aliver | [TK]D-Fender Yes, I do. |
| 04:50.14 | aliver | [TK]D-Fender it's seven digits and the pattern looks right to me. |
| 04:50.26 | aliver | [TK]D-Fender there are no other patterns in my extensions.conf |
| 04:50.52 | jblack | aliver: did you look at my example yet? |
| 04:51.11 | aliver | jblack yes, I'm reading it now |
| 04:51.24 | jameswf-home | ~ch5 |
| 04:51.25 | jbot | Read about extensions DialPlans etc.. in Chapter 5 of Asterisk: The Future of Telephony 2nd Edition http://www.oreilly.com/catalog/9780596510480/ |
| 04:51.27 | [TK]D-Fender | aliver: now PB "dialplan show internalphones" |
| 04:51.33 | jameswf-home | ~ch6 |
| 04:51.33 | jbot | Read about Advanced extensions DialPlans etc.. in Chapter 6 of Asterisk: The Future of Telephony 2nd Edition http://www.oreilly.com/catalog/9780596510480/ |
| 04:52.31 | aliver | jbot okay |
| 04:52.32 | jbot | okay is probably fine |
| 04:53.07 | jameswf-home | jbot: whos your daddy |
| 04:53.07 | jbot | ACTION who's your daddy now? huh? huh? |
| 04:53.09 | [TK]D-Fender | jbot fine |
| 04:53.09 | jbot | Alrighty then =) |
| 04:53.18 | jblack | jbot: sleep |
| 04:53.18 | jbot | sleep is, like, overrated, and a poor substitute for caffeine. |
| 04:53.42 | jameswf-home | jbot dropdatabase; |
| 04:53.43 | jbot | So you think you could kill me, did you. You thought you could come in here, in my own home, and take me down without a fight. I would have given you a chance, really. But now, you have unleashed hell. May god have mercy on your soul. |
| 04:53.48 | jblack | A needed command for jbot is the ability to tell it to go silent for 5 minutes. |
| 04:54.22 | mchou | ~stfu |
| 04:54.22 | jbot | :( |
| 04:54.35 | mchou | lol |
| 04:54.55 | aliver | okay, I'm reading the book now |
| 04:55.31 | [TK]D-Fender | aliver: Wheres the new PB?> |
| 04:55.52 | jblack | I need to cut down my dialplan. |
| 04:56.12 | jblack | 223 lines for a building with 2 people is silly. |
| 04:56.29 | aliver | [TK]D-Fender sorry, I didn't realize that you meant to paste the output of that command one sec. I'll do it. |
| 04:56.55 | aliver | No such command 'dialplan' (type 'help' for help) |
| 04:57.03 | aliver | this is ast 1.2 |
| 04:57.12 | jblack | sigh |
| 04:57.14 | mchou | "show dialplan" :) |
| 04:57.20 | aliver | whoops |
| 04:57.34 | jblack | Is there any particular reason that yo're running a really old asterisk? |
| 04:57.50 | [TK]D-Fender | And tonight he's going to party like its 1699... |
| 04:58.05 | mchou | jblack: maybe he's running embedded :) |
| 04:58.23 | jblack | Don't joke with me. I'm still mad at you. |
| 04:58.28 | aliver | [TK]D-Fender http://pastebin.com/m5412c291 |
| 04:58.52 | aliver | jblack 1.4 won't compile on my platform. |
| 04:59.00 | aliver | it blows up in a million ways. |
| 04:59.13 | [TK]D-Fender | aliver: Well now... no lines there.... do a RELOAD and try again. |
| 04:59.56 | aliver | [TK]D-Fender the output is the same after I issue the reload command. |
| 05:00.08 | aliver | or do you mean the pastebin was empty? |
| 05:00.14 | [TK]D-Fender | aliver: dump your entire dialplan that way |
| 05:00.29 | [TK]D-Fender | aliver: You're missing all kinds of stuff. |
| 05:00.31 | mchou | aliver: what platform is this? |
| 05:01.11 | aliver | I typed "show dialplan" |
| 05:01.16 | aliver | http://pastebin.com/d4140594b <-- here is the output |
| 05:01.22 | aliver | this is BSD |
| 05:02.21 | [TK]D-Fender | aliver: backupt your dialplan and wipe that ascii art crap out |
| 05:02.53 | aliver | [TK]D-Fender okay |
| 05:04.04 | aliver | I'll recreate it real quick using that output |
| 05:04.53 | mchou | grep -v works :) |
| 05:05.09 | aliver | I'm just wondering why it's ignoring half my dialplan |
| 05:05.24 | aliver | what 'show dialplan' shows is NOT what's in extensions.conf |
| 05:05.27 | aliver | errors? |
| 05:05.40 | [TK]D-Fender | aliver: you should see them on "reload" |
| 05:08.04 | jameswf-home | ~bsd |
| 05:08.04 | jbot | BSD is a UNIX operating system. An asterisk port is currently availible if you feel you must, or a way to set your pc back 30 years, progress is overrated |
| 05:08.05 | kamanashisroy | hmmmmmm .. dead friends .. :) |
| 05:08.49 | kamanashisroy | LOLz |
| 05:11.03 | jblack | I have decided the best part of star wars is when the yeti knocks out luke |
| 05:11.40 | [TK]D-Fender | jblack: Unfortunately he DOESN'T get eaten |
| 05:11.52 | jblack | That's where things take a turn for the worse. |
| 05:12.08 | jblack | Moot point. I changed my mind, and decided to watch natalie portman instead |
| 05:13.53 | [TK]D-Fender | jblack: http://i36.photobucket.com/albums/e23/The_Shwa/demotivational/denial.jpg |
| 05:15.53 | *** join/#asterisk Koshatul (n=evangeli@ppp188-33.static.internode.on.net) |
| 05:17.25 | jblack | heh |
| 05:17.44 | *** join/#asterisk rcy (n=rcy@S01060002553240a8.vc.shawcable.net) |
| 05:18.17 | *** join/#asterisk rcy (n=rcy@S01060002553240a8.vc.shawcable.net) |
| 05:21.50 | [TK]D-Fender | checkout time, later all. |
| 05:24.52 | *** join/#asterisk LND (n=Lee@89.192.211.83) |
| 05:36.34 | *** join/#asterisk Maxous (n=Maxous@74.7.13.242) |
| 05:37.13 | *** join/#asterisk totalimpact8 (n=Miranda@72.169.145.198) |
| 05:38.44 | *** join/#asterisk steliosk (n=Stelios@athedsl-123365.home.otenet.gr) |
| 05:38.46 | totalimpact8 | anyone know how to read DIALSTATUS info on a zap channel without using a DIAL cmd? |
| 05:38.46 | Maxous | Hello |
| 05:39.05 | totalimpact8 | such as for a centrex transfer |
| 05:39.24 | Maxous | who knows what to set the jitters SIP settings to overcome scratchy voice quality on received side only? |
| 05:40.46 | totalimpact8 | or how to perform a DIAL cmd on a currently open zap channel? |
| 05:40.48 | Maliuta | Maxous: are you sure it's jitter? |
| 05:40.52 | jblack | there are jitter settings for sip? |
| 05:41.13 | Maliuta | Maxous: the jitterbuffer settings are the same as the iax ones IIRC |
| 05:41.18 | Maliuta | jblack: since 1.4 |
| 05:41.36 | Maxous | Maliuta: Fairly Sure. |
| 05:42.11 | Maxous | Maliuta: I am asking this on behalf of another, so I am not sure of the full depth of the problem. I do apologize. |
| 05:44.05 | Maxous | Maxous: Do you have an idea for what a good start would be to change it to? |
| 05:44.21 | Maxous | Maliuta:Do you have an idea for what a good start would be to change it to? |
| 05:44.54 | Maliuta | Maxous: look on the wikki, the settings I use are based on the IAX ones |
| 05:45.16 | Maxous | Maliuta: Will do, TY. |
| 05:47.01 | *** join/#asterisk cvox (n=chatzill@c-71-199-54-34.hsd1.co.comcast.net) |
| 05:47.51 | cvox | I have an interesting situation.. placing a call from asterisk out a Verizon e&m wink circuit some calls go out but not all... telco says they see the connection but no data (on the numbers I cannot reach). |
| 05:47.52 | *** join/#asterisk miloux (n=miloux@static-213.88.173.45.addr.tdcsong.se) |
| 05:47.57 | cvox | inbound dialing works flawless. |
| 05:48.49 | *** join/#asterisk _gm (n=gmustafa@202.133.78.60) |
| 05:50.35 | *** join/#asterisk chendy (n=chatzill@58.60.220.146) |
| 06:02.36 | *** join/#asterisk djin (n=djin@84-104-110-116.cable.quicknet.nl) |
| 06:06.59 | *** part/#asterisk Maxous (n=Maxous@74.7.13.242) |
| 06:14.52 | *** part/#asterisk cvox (n=chatzill@c-71-199-54-34.hsd1.co.comcast.net) |
| 06:22.18 | *** join/#asterisk [T]ank (n=chwall@c-71-199-25-239.hsd1.co.comcast.net) |
| 06:22.35 | [T]ank | i am trying to dial out using a new sip account and I am not making it work correctly |
| 06:22.41 | [T]ank | http://pastebin.ca/1181374 are the details |
| 06:22.56 | [T]ank | can anyone assist me in the correct syntax to dial out with this account? |
| 06:23.02 | [T]ank | i can dial into this account no problem. |
| 06:23.16 | jblack | Looks like failed authentication on line 59 |
| 06:23.17 | [T]ank | but i am not successful in calling outbound |
| 06:23.30 | [T]ank | yeah... i am not getting the syntax |
| 06:23.31 | jblack | You can try "sip debug" to get more info |
| 06:23.37 | [T]ank | i have tried a hunred different things. |
| 06:24.11 | jblack | Dial(extension@PROVIDER,60) |
| 06:24.18 | jblack | pardon. |
| 06:24.22 | jblack | Dial(SIP/extension@PROVIDER,60) |
| 06:24.28 | [T]ank | i dont need to use the user name and pass? |
| 06:24.40 | jblack | That's defined in the sip.conf |
| 06:24.48 | *** join/#asterisk nighty^ (n=nighty@host217-37-109-17.in-addr.btopenworld.com) |
| 06:24.55 | jblack | PROVIDER will be the sip context that holds the authentication info |
| 06:25.01 | nnY_2 | you'll have 3 entries.. |
| 06:25.16 | nnY_2 | register, and an inbound and outbound |
| 06:26.59 | [T]ank | so i have changed it to be exten => _1XXXXXXXXXX,n,Dial(SIP/${EXTEN}@8018204181) |
| 06:27.06 | [T]ank | i have not changed the sip.conf |
| 06:27.24 | jblack | Is registration working? |
| 06:27.37 | [T]ank | they are registered, yes |
| 06:27.51 | jblack | look at lines 17-19. You disabled authentication there. |
| 06:27.57 | [T]ank | in sip? |
| 06:28.12 | jblack | Yes. I'm giving the line numbers to your own pastebin |
| 06:28.32 | jblack | How they map to your sip.conf, I don't know. |
| 06:28.41 | [T]ank | yeah... hang on... lemme try |
| 06:29.04 | jblack | dont' forget to reload your sip.conf and extensions.conf after changing them. |
| 06:29.33 | kaldemar | [T]ank: i sincerely hope those are not your real secrets... |
| 06:29.39 | [T]ank | nice... its always something REALLY stupid |
| 06:30.35 | jblack | kaldemar: Shhhh. I need free phone calls |
| 06:31.04 | jblack | If he weren't giving us his authentication, I wouldn't be able to call the vatican and ask them what the pope wears under his robes |
| 06:31.07 | jeev | he |
| 06:31.07 | jeev | hey |
| 06:32.54 | *** join/#asterisk simoncpu (n=soulfury@58.71.34.137) |
| 06:33.05 | simoncpu | uh.... |
| 06:33.21 | jblack | here we go. |
| 06:33.26 | simoncpu | can you guys recommend a good web ui for call detail records? |
| 06:33.29 | simoncpu | uhm... |
| 06:33.36 | Maliuta | xterm+vim |
| 06:34.02 | Maliuta | or just point apache at the CDR dir and use moddir and a browser |
| 06:34.10 | jblack | Heh. Answers.com can't answer "why does answers.com suck" |
| 06:34.34 | simoncpu | lol |
| 06:34.41 | simoncpu | apache can't do analysis and stuff |
| 06:34.55 | jblack | I usually put something together for clients that match their needs. |
| 06:35.01 | simoncpu | i'm using pbxnsip, btw, not asterisk |
| 06:35.09 | jblack | Then you're in the wrong channel. |
| 06:35.10 | simoncpu | but there seems to be no channel for pbxnsip |
| 06:35.53 | jblack | You don't take a toyota to a ford dealership just because there aren't any Toyota dealers about. |
| 06:36.15 | sn9 | unless you plan to trade it for a ford |
| 06:36.36 | simoncpu | nope, but the guys at ford may be able to help me on general car problems |
| 06:36.36 | simoncpu | hehehe |
| 06:36.38 | jblack | even then. It's better to sell it in the paper. |
| 06:37.07 | jblack | besides, I maintain that if you're trading a toyota for a ford, you've got a mental handicap |
| 06:37.36 | jblack | simoncpu: Sure, if you pay them twice as much. |
| 06:37.50 | simoncpu | hahaha |
| 06:37.51 | *** join/#asterisk jer (n=jtregunn@unaffiliated/jer) |
| 06:38.12 | jblack | simoncpu: So, following in that meme, if you paypal me a $500 nonrefundable deposit, I'll take the time to care about pbxnsip. |
| 06:38.17 | *** part/#asterisk [T]ank (n=chwall@c-71-199-25-239.hsd1.co.comcast.net) |
| 06:38.51 | jblack | Otherwise, you're just a good way to stave off boredom by being a target for "he got foreign" jokes. |
| 06:38.56 | simoncpu | but i'm here in soviet russia; |
| 06:39.02 | simoncpu | you paypal me, not me to you |
| 06:39.03 | simoncpu | hehehe |
| 06:39.32 | jblack | Shouldn't you be off in georgia, knocking over police cars with tanks? |
| 06:40.16 | simoncpu | i went to atlanta yesterday but i didn't see any tanks |
| 06:40.44 | jblack | You were in atlanta yesterday, and in russia today? |
| 06:40.52 | simoncpu | yeah |
| 06:40.55 | simoncpu | :) |
| 06:41.13 | jblack | wrong georgia. |
| 06:41.54 | jblack | .ph .. .ph... |
| 06:41.55 | sn9 | the state department is an asshat for not renaming the republic of georgia |
| 06:42.21 | simoncpu | i'm just using a shell |
| 06:42.35 | simoncpu | ei, thanks for the time guys |
| 06:42.41 | simoncpu | i guess i'll just google for answers or something |
| 06:42.46 | jblack | That's probably best. |
| 06:43.01 | sn9 | or, you could try asterisk, too |
| 06:43.12 | jblack | then we'd have to help him |
| 06:43.17 | simoncpu | hihiih |
| 06:43.31 | simoncpu | i'll definitely try asterisk someday |
| 06:43.48 | simoncpu | (next month, probably, as soon as i get a spare box to play with) |
| 06:44.01 | simoncpu | is looking for used soekris boxes on ebay |
| 06:44.20 | jblack | That was almost fun for a bit. |
| 06:44.32 | *** join/#asterisk s0lid (n=s0lid@60.52.204.197) |
| 06:45.13 | sn9 | you could also use something cheaper than a soekris |
| 06:46.39 | simoncpu | uhm... what's cheaper than soekris? |
| 06:46.41 | *** join/#asterisk shinao1 (n=shinao1@62.173.48.176) |
| 06:46.45 | sn9 | an old pc |
| 06:46.50 | simoncpu | ahhhh |
| 06:47.49 | jblack | Some day, I'm going to get an mini-itx w/ wireless, hook up a polycom to it, and make my own "car phone" |
| 06:48.48 | sn9 | "forget buses and trains -- i'll take the phone" |
| 06:49.04 | jblack | I mean a phone in my car. |
| 06:49.09 | simoncpu | i want to hook up mini-itx wireless with a bomb to make my own "wi-fi activated bomb" |
| 06:49.11 | jblack | such as used to be common in the 90s. |
| 06:49.12 | simoncpu | very cooool |
| 06:50.00 | jblack | bah. Anyone can do a remote detonator with a cell phone and 2 bucks in parts. |
| 06:50.30 | sn9 | ah, but the wi-fi would make it "cool" ... |
| 06:50.50 | jblack | only if you want to be close enough to get caught. |
| 06:50.58 | simoncpu | and the victim can surf the net while he waits for certain death... |
| 06:51.19 | sn9 | hooray for useless features |
| 06:52.42 | jblack | I think tomorrow, I'm going to walk up to a meter maid and try to use jedi mind tricks on them. |
| 06:52.48 | switchcat | You guys know this channel is logged, right? :P |
| 06:52.50 | jblack | Just to see their response. |
| 06:53.03 | jblack | switchcat: In more places than one. |
| 06:53.25 | jblack | speaking of which.. Dear NSA. I _still_ want my helicopter-black coffee cup. |
| 06:53.52 | switchcat | Is it a silent coffee cup? |
| 06:53.56 | drmessano | jblack |
| 06:54.03 | drmessano | They sent it to you.. you just cant see it |
| 06:54.05 | jblack | switchcat: You never see it comin' |
| 06:54.30 | sn9 | hmm, stealth coffee... |
| 06:54.32 | switchcat | What amazes me is how visible the NSA is |
| 06:54.34 | drmessano | I have like two dozen of them.. Scattered all over the house.. not that i've ever seen one or tripped over one |
| 06:54.41 | simoncpu | iraq nsa bush bomb nuclear assasinate war terrorism bomb |
| 06:54.44 | switchcat | I mean they have a big honking sign on the beltway that says "this way to the NSA" |
| 06:55.04 | drmessano | simoncpu: You left off 'troll' and 'wtf' |
| 06:55.11 | sn9 | you said bomb twice |
| 06:55.15 | jblack | simoncpu: If you actually do do international trouble, I'd knock it off. |
| 06:55.33 | jblack | that's coming from experience. |
| 06:56.18 | sn9 | more likely to trouble the international do do |
| 06:57.32 | jblack | I was doing a lot of international travel when i worked for the guys that make ubuntu. I started making too many coffee cup requests, and then the next 2-3 times I few, I was detained every single time for a "random search" |
| 06:58.00 | jblack | s/few/flew |
| 06:58.38 | *** join/#asterisk stimpie (n=stimpie@84-104-5-14.cable.quicknet.nl) |
| 06:59.18 | drmessano | Word of advice: Dont order Anthrax CDs from Amazon |
| 06:59.19 | jblack | which just pisses me off all the more. All I want is my coffee cup! How long would it take some spook to paint a $.99 coffee cup in stealth-bomber paint, and ship it usps ? |
| 06:59.25 | simoncpu | yikes... i hope that troll won't get me in trouble at the airport in the next few weeks |
| 06:59.28 | simoncpu | :( |
| 06:59.44 | jblack | You'll know if your tickets start getting SSSS on them a lot. |
| 07:00.01 | drmessano | We already have your IP address simon |
| 07:00.08 | drmessano | carry on |
| 07:00.13 | simoncpu | hehehe |
| 07:00.40 | jblack | I'm still bored. Can we let louisiana flood again? |
| 07:01.48 | *** join/#asterisk Gary (i=gary@freenode/staff/colchester-lug.gary) |
| 07:10.27 | switchcat | jeev - quit thinking about it, and just DO it. |
| 07:10.45 | switchcat | jblack - coffee cup requests? |
| 07:10.55 | jeev | what |
| 07:11.04 | switchcat | jeev - just install a distro |
| 07:11.05 | simoncpu | switchcat: black coffee |
| 07:11.14 | switchcat | you're thinking about it too much |
| 07:11.25 | jeev | wow |
| 07:11.28 | jeev | you scroll back too much! |
| 07:11.31 | jeev | i just did 4.6 32bit. |
| 07:11.32 | jeev | :) |
| 07:11.39 | drmessano | ROFL |
| 07:11.44 | drmessano | Centos 4.6? |
| 07:11.58 | jeev | i need to stress test it more |
| 07:12.01 | jeev | bonnie++ has been running for hours |
| 07:12.07 | drmessano | Good god |
| 07:12.12 | drmessano | 5.2 is 10x better |
| 07:12.15 | switchcat | jeev - oh, heh, I had did a /lastlog on a word, and your line came up.. I thought you were repeating yourself from earlier. :P |
| 07:13.50 | jeev | lol |
| 07:14.04 | switchcat | jblack - I don't get it, is this supposed to be a joke, or is some part of the government actually giving out promotional coffee cups? |
| 07:16.32 | jeev | boooo |
| 07:17.09 | jblack | switchcat: Oh, it's completely serious. |
| 07:17.52 | jblack | next time you're going through the line at the airport, demand the TSA cup that's due you. |
| 07:18.00 | jblack | It's part of the budget. You have to be firm in the request, though |
| 07:18.54 | jblack | They started offering them to soften their image, but realized it cost too much. |
| 07:20.12 | *** join/#asterisk blogbasti (n=blogbast@calypso.planet-ic.de) |
| 07:22.19 | *** join/#asterisk ice_croft (n=nolan@213.132.86.246) |
| 07:23.21 | jblack | switchcat: Want to see a copy of my mug? |
| 07:26.19 | *** join/#asterisk ToTo (n=ToTo@207.176.6.37) |
| 07:29.00 | *** join/#asterisk zydoon (n=zydoon@41.225.157.25) |
| 07:30.11 | *** part/#asterisk zydoon (n=zydoon@41.225.157.25) |
| 07:32.45 | switchcat | uhm sure |
| 07:32.51 | jblack | k. I'll upload it |
| 07:36.00 | *** join/#asterisk zydoon (n=zydoon@41.225.157.25) |
| 07:36.13 | *** part/#asterisk zydoon (n=zydoon@41.225.157.25) |
| 07:37.00 | switchcat | doing a websearch for TSA cup and TSA mug turns up next to nothing apart from some stories about some sippy cup incident |
| 07:38.04 | jblack | hold on. I'm looking for my camera |
| 07:43.13 | *** join/#asterisk steliosk (n=Stelios@emile.ath.forthnet.gr) |
| 07:44.46 | *** join/#asterisk mintos (n=mvaliyav@nat/redhat-in/x-07f371d70485fcfa) |
| 07:44.57 | jblack | I found it. my kid took it |
| 07:45.02 | jblack | now I just need to find the batteries |
| 07:45.19 | C4away | I can't even find anything for "TSA Mug" or "TSA Cup" on ebay |
| 07:46.06 | *** join/#asterisk linuxstb (n=linuxstb@rockbox/developer/linuxstb) |
| 07:46.24 | simoncpu | http://www.nles.com/store/customer/product.php?productid=950 |
| 07:47.49 | jblack | ok. |
| 07:48.13 | jblack | http://linuxguru.net/~jblack/mytsamug.jpg |
| 07:49.25 | simoncpu | lol |
| 07:49.30 | simoncpu | photoshop? |
| 07:49.31 | simoncpu | hehehe |
| 07:49.37 | jblack | I don't have photoshop |
| 07:49.51 | jblack | I don't even have windows. |
| 07:49.53 | simoncpu | paintbrush? |
| 07:49.59 | simoncpu | gimp |
| 07:50.07 | jblack | well, I do have gimp. |
| 07:51.59 | C4away | heh, that does look photoshopped |
| 07:52.27 | C4away | however, the ports on the laptop look photoshopped to, so I'll assume it is a "color correction" built into the camera |
| 07:53.30 | switchcat | dude that crashed my browser |
| 07:53.43 | jblack | I don't see how. It's just a jpeg. |
| 07:54.07 | jblack | Sorry it's not a great camera. I don't exactly buy my kid high end equipment. |
| 07:54.30 | simoncpu | switchcat: the jpeg contains a browser exploit |
| 07:54.48 | jblack | Look. YOu can tell it's not photoshop, because you can see the reflection in the desk on the left. |
| 07:55.27 | C4away | yea |
| 07:55.42 | C4away | you had over 3 minutes between the time you said "hold on let me get my camera" and when you uploaded the pic |
| 07:55.55 | C4away | that is plenty of time to add reflections, shading, etc |
| 07:55.57 | jblack | I was trying to not wake my kid up. |
| 07:56.01 | sn9 | "Bad Photoshop/No Reflection" |
| 07:56.27 | C4away | I saw that pic already on digg anyway |
| 07:56.38 | switchcat | yeah really bad fake :P |
| 07:56.38 | jblack | of my mug? |
| 07:56.47 | jblack | http://gallery.linuxguru.net/cool-stuff/img_0633 |
| 07:57.12 | jblack | I got the TSA logo from their site, and the coffee cup off of cafepress. |
| 07:57.21 | C4away | haha nice |
| 07:57.40 | C4away | thought the bottom edge was a bit jagged |
| 07:57.50 | jblack | Yeah. THat's what gives it away. |
| 07:57.58 | jblack | if I had had time, I wouldh ave added a couple coffee stains to it. |
| 07:58.10 | C4away | nice |
| 07:58.12 | switchcat | the bottom edge, the top edge, and the handle edge are all jagged |
| 07:58.19 | jblack | I was in a rush. |
| 07:58.25 | C4away | the pen was too big anyway |
| 07:58.29 | C4away | the scale was off |
| 07:58.43 | C4away | It's just fun saying "photoshopped" and "seen it" though |
| 07:58.52 | switchcat | next time you wanna fake a mug, put a real mug on your desk, take the photo, and then just put in the logo, it will look more realistic |
| 07:59.04 | jblack | in 3 minutes? CMON |
| 07:59.06 | C4away | and blur it a bit more |
| 07:59.12 | C4away | just a bit to take the edge off |
| 07:59.24 | jblack | besides, my camera really is in my kids room. |
| 07:59.25 | C4away | actually the text on that one looks pretty good |
| 08:00.23 | switchcat | well at least thats better than sending me to goatse.cx |
| 08:00.45 | *** join/#asterisk darkskiez (n=mhb@cpc1-broo3-0-0-cust659.renf.cable.ntl.com) |
| 08:01.25 | jblack | That's not even my current laptop. |
| 08:02.38 | jblack | now I'm bored again |
| 08:03.35 | switchcat | uhm, go hack the zipit z2 messenger to work with VoIP apps. |
| 08:03.59 | jblack | nobody wants to call me, much less message me. |
| 08:04.46 | switchcat | dude.. it's a $36 linux pda essentially |
| 08:04.52 | switchcat | besides your kid will love it |
| 08:04.54 | jblack | what is? |
| 08:04.59 | jblack | a zipit? |
| 08:05.08 | C4away | I had a zip 250 once |
| 08:05.12 | C4away | it was usb and pcmcia |
| 08:05.26 | C4away | but the power socket was loose |
| 08:05.31 | C4away | would kick on and off |
| 08:05.33 | jblack | does it do ssh? |
| 08:05.41 | C4away | a zip 250? no |
| 08:05.58 | jblack | Give me a cheap device that does ssh, and I'd be interested. |
| 08:06.10 | jblack | but I still couldn't afford it. |
| 08:06.58 | C4away | my desktop computer does ssh |
| 08:07.03 | C4away | so does my palm treo |
| 08:08.12 | switchcat | c4away - the zipit messenger is a device aimed at instant messaging, its run on an PXA270 ARM processor with a 340x240 (?) standard pda type screen, and keyboard. it's $50 retail or like $36 on amazon.com |
| 08:08.57 | switchcat | the z2 model has a miniSD card slot so you can do all sorts of things with it, and the company that put it out supports and encourages hacking it |
| 08:11.07 | switchcat | the z1 is hackable too (available for $10-20 on ebay), but doesnt have the SD card slot and is b&w as opposed to the z2's color backlit screen. |
| 08:11.18 | C4away | wifi? bluetooth? evdo? |
| 08:11.21 | switchcat | wifi |
| 08:11.32 | C4away | only wifi? |
| 08:11.33 | switchcat | and it's got a custom port also |
| 08:11.40 | switchcat | and a headset jack |
| 08:11.51 | C4away | well what good is that? |
| 08:12.12 | C4away | if i have to be home to use it then why don't I just use my pc |
| 08:12.19 | C4away | or the media center computer in the living room |
| 08:12.19 | switchcat | it's only 36 bucks, please find me a similar linux based pda device for $36 |
| 08:12.35 | C4away | how about $0 than I'm already paying for my desktop |
| 08:12.36 | switchcat | you dont have to be home to use it |
| 08:13.06 | C4away | yea, heh, I can see getting arrested standing in the bushes outside someone's house trying to connect to their Linksys Channel 6 |
| 08:13.19 | C4away | "I was trying to chat with my friends" |
| 08:13.37 | switchcat | I dunno about you but I happen to live in a city with free municipal wifi. :P |
| 08:13.44 | C4away | would be nice |
| 08:13.50 | C4away | lucky bastard |
| 08:13.54 | switchcat | besides, lots of people have open access systems for just that purpose |
| 08:14.08 | switchcat | and you can also get on at numerous commercial wifi points |
| 08:14.16 | C4away | the only free wifi here is the aforementioned Linksys Chan 6 |
| 08:14.21 | simoncpu | our free wifi here requires me to buy a coffee |
| 08:14.43 | simoncpu | there used to be a lot of places that offer free wifi, |
| 08:14.48 | simoncpu | but they suddenly stopped using wep |
| 08:14.49 | simoncpu | :( |
| 08:14.54 | C4away | haha |
| 08:15.02 | C4away | yea |
| 08:15.05 | C4away | I know what you mean |
| 08:15.11 | C4away | ... I mean ... uh ... I've heard about that |
| 08:15.11 | simoncpu | hehehe |
| 08:15.20 | *** join/#asterisk KermitTheFragger (n=KermitTh@118-197.bbned.dsl.internl.net) |
| 08:15.28 | C4away | my friend told me about that ¬¬ |
| 08:20.47 | *** join/#asterisk sdam (n=sdam@80.72.151.229) |
| 08:29.58 | *** join/#asterisk ToTo (n=ToTo@207.176.6.37) |
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| 09:38.14 | jblack | does meetme still require ztdummy? |
| 09:38.32 | kaldemar | yes |
| 09:38.35 | *** join/#asterisk Shotygun (n=thorn@84-16-228-45.internetserviceteam.com) |
| 09:38.42 | kaldemar | or digium hardware, of course. |
| 09:38.55 | DarKnesS_WolF | jblack: still needs timing source |
| 09:39.02 | jblack | there goes that on that machine then. |
| 09:39.10 | *** join/#asterisk laarka (n=lak@194.192.93.101) |
| 09:40.38 | kaldemar | jblack: try app_conference |
| 09:40.39 | laarka | hi everyone. i could really really need a hand with a nat setup on my trixbox. i've got trunks working, internal phones, and a SIP client on the outside can call in and get two way audio. a call from the inside to outside gets no audio |
| 09:40.52 | jblack | yeah. I've used that before. I'm looking for it now. |
| 09:40.59 | jblack | looks like it got pulled out of ubuntu |
| 09:41.32 | DarKnesS_WolF | jblack: mmm u don't compile asterisk ;-)? |
| 09:41.34 | kaldemar | http://sourceforge.net/projects/appconference/ |
| 09:41.44 | jblack | Not on production systems. |
| 09:41.59 | DarKnesS_WolF | kbwhy ? i do compile on pridcutions |
| 09:42.19 | DarKnesS_WolF | laarka: the trixbox behind NAT ? |
| 09:42.38 | kaldemar | laarka: you'll be more likely to get help in #trixbox |
| 09:42.45 | jblack | because reproduction after failure becomes difficult. |
| 09:43.10 | laarka | thanks, ill try to bother them in #trixbox first then come back if that fails |
| 09:43.24 | jblack | No, don't bother coming back. Trixbox isn't supported here at all. |
| 09:43.45 | DarKnesS_WolF | jblack: who said so ? |
| 09:43.53 | kaldemar | everyone says so. |
| 09:44.19 | DarKnesS_WolF | laarka: most lickly the externalip in ur sip.conf not correct don't know how trixbox handels that but i think this might be the problem |
| 09:44.22 | DarKnesS_WolF | kaldemar: ah ic :P |
| 09:44.40 | jblack | You won't hurt my feelings if you don't believe me, but all you'll get is picked on. |
| 09:45.07 | kaldemar | DarKnesS_WolF: you just said it yourself, no one here knows how trixbox handles stuff. |
| 09:45.26 | DarKnesS_WolF | kaldemar: i don't but i do know how elastix handels it ;-) |
| 09:45.45 | laarka | hmm, isnt trixbox just a webinterface put on top of asterisk? if i edit config files directly whats the difference to you? |
| 09:46.18 | laarka | (bring on the hail-storm) |
| 09:46.20 | kaldemar | trixbox messes up the configuration file structure and makes configurations complex. |
| 09:46.31 | DarKnesS_WolF | laarka: nop trixbox is a full distro the webinterface is just FreePBX |
| 09:46.42 | DarKnesS_WolF | kaldemar: that is also true ;-) |
| 09:48.21 | *** join/#asterisk prologin (n=prologin@ero34-1-88-161-76-133.fbx.proxad.net) |
| 09:51.41 | laarka | now that you got that off your chest, and the trixbox ppl are sleeping in their channel, could we possibly view this as an asterisk problem? :-) |
| 09:53.10 | DarKnesS_WolF | laarka: in that case ur sip.conf extranip should point to ur router IP |
| 09:53.19 | DarKnesS_WolF | or i think there is another option like extreanhost |
| 09:53.21 | laarka | i've forwarded UDP 4569,5004:5037,5039:5082,1027:1028,10000:20000 to my asterisk |
| 09:53.36 | DarKnesS_WolF | may be alos ur router /firewall not opining UPNP ports ? |
| 09:53.52 | DarKnesS_WolF | try to fwd 10000-20000 to ur asterisk prts / UDP for sure |
| 09:54.01 | DarKnesS_WolF | laarka: also u might wanna try IAX ;) |
| 09:54.10 | laarka | i have this in sip_nap: |
| 09:54.36 | DarKnesS_WolF | then it may be the other end firewall not doing upnp ? |
| 09:54.44 | DarKnesS_WolF | laarka: try from diffrent locations |
| 09:55.18 | laarka | the firewall does not do upnp, do i really need that when ive forwarded a massive load of udp manually? |
| 09:55.49 | DarKnesS_WolF | laarka: no i mean the other router the other end |
| 09:55.53 | DarKnesS_WolF | wherre the sip phone is there |
| 09:56.16 | laarka | neither firewall does upnp |
| 09:56.25 | kaldemar | forget upnp. |
| 09:57.10 | laarka | upnp defies the purpose of a firewall. if someone sneaks malware inside the network, its wide open, as it can just ask the firewall to open ports |
| 09:57.56 | DarKnesS_WolF | huh ? |
| 09:58.09 | laarka | if asterisk calls the client, does the client connect to asterisk to get audio or is it always the initiator that establishes audio first? |
| 09:58.30 | kaldemar | depends on your sip configuration. |
| 09:59.10 | laarka | for the client or general asterisk? i'd like the client to always initiate audio |
| 10:00.15 | kaldemar | if you put allowtransfer=no in your sip.conf, asterisk stays on the media path. |
| 10:00.24 | laarka | i have in sip_nat.conf |
| 10:00.24 | laarka | externhost = voip.mydomain.com |
| 10:00.24 | laarka | externip=87.55.xxx.xxx |
| 10:00.24 | laarka | localnet=10.199.0.0/255.255.255.0 |
| 10:00.24 | laarka | nat=yes |
| 10:00.25 | laarka | port = 5060 |
| 10:00.34 | kaldemar | ~pb |
| 10:00.35 | jbot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste |
| 10:00.41 | laarka | sorry |
| 10:00.51 | laarka | irc n00b obviously |
| 10:03.10 | kaldemar | oh crap, forget about allowtransfer. reinvite parameters are the ones. |
| 10:04.13 | kaldemar | canreinvite=no that is. |
| 10:06.50 | laarka | sorry already have canreinvite = no |
| 10:10.35 | *** join/#asterisk ToTo (n=ToTo@207.176.6.138) |
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| 10:12.14 | *** part/#asterisk zydoon (n=zydoon@41.225.157.25) |
| 10:12.16 | kaldemar | well, the asterisk way to proceed would be "sip set debug" and make a call attempt. then go though the trace and see what happens. |
| 10:16.14 | mchou | laarka: does your firewall have a shell interface? |
| 10:16.32 | laarka | mchou: sure, nothing gets logged as blocked |
| 10:17.15 | mchou | laarka: you say inbot to outbound doesnt get inbound audio? |
| 10:17.23 | mchou | inbound* |
| 10:17.37 | mchou | umm, internal phone* |
| 10:17.41 | laarka | from asterisk to client gets no audio at all |
| 10:18.17 | mchou | asterisk is public pi pr behind nat? |
| 10:18.24 | mchou | ip or* |
| 10:18.34 | laarka | (ie a call from a real phone number via sip trunk to client sip phone |
| 10:18.36 | kaldemar | is the audio going from asterisk to the client, or is the outside client trying to send directly to the client? |
| 10:19.22 | laarka | thats a good question. being naive i thought the audio always went through asterisk |
| 10:19.23 | mchou | laarka: whats rtp denug look like on * box? |
| 10:19.26 | kaldemar | find that out by looking at the trace and dumping the asterisk box's interface. |
| 10:19.39 | mchou | debug* |
| 10:23.12 | laarka | rtp debug: http://pastebin.com/d67b9bde7 |
| 10:24.24 | laarka | i called from a regular phone to 8749xxxx (my phone number at my sip provider), this goes to a ring group and called 110 which is my mobile behind nat at an outside network. i answered the phone, no audio in either direction, and hung up |
| 10:24.48 | mchou | thats not rtp |
| 10:25.14 | mchou | there is no rtp info in the PB |
| 10:27.54 | laarka | hm - this is sip debug (i hope): http://pastebin.com/d5b4e4308 |
| 10:28.28 | mchou | where is rtp debug? |
| 10:28.54 | laarka | i did "rtp debug" and the first pastebin is what i got |
| 10:28.55 | *** join/#asterisk ToTo (n=ToTo@209.8.41.146) |
| 10:30.24 | laarka | aaaah |
| 10:30.40 | *** join/#asterisk s0lid (n=s0lid@60.52.253.84) |
| 10:30.56 | laarka | perhaps my providers sip trunk tries to connect directly to the sip client phone?? |
| 10:33.02 | laarka | calling from the client to echotest gives rtp debug output, calling from a real phone to my nat client does not |
| 10:33.04 | mchou | What's your canreinvite setting? |
| 10:33.25 | laarka | for the sip phone its no |
| 10:33.36 | laarka | im not sure about the trunk |
| 10:34.55 | mchou | I dunno. You're just going to have to ask someone else |
| 10:36.00 | mchou | ppl would be in a better position to help you if you posted your sip.conf (user/password blanked) |
| 10:36.24 | mchou | I'm inclined to think its some trixbox bullshit |
| 10:41.59 | *** join/#asterisk mort_gib (n=mjensen@16.Red-83-36-63.staticIP.rima-tde.net) |
| 10:53.09 | *** join/#asterisk Delvar (n=Delvar@87-194-74-202.bethere.co.uk) |
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| 11:02.21 | *** join/#asterisk microw (n=mike@219.133.105.235) |
| 11:04.09 | microw | hi, which is the best sip client softphone for an embedded linux(arm chip)? |
| 11:04.30 | simoncpu | iphone!!! |
| 11:04.40 | sn9 | yuck |
| 11:04.44 | simoncpu | hahahahaha |
| 11:05.03 | microw | simoncpu: iphone?? |
| 11:05.21 | sn9 | apple's iphone has an arm chip |
| 11:05.23 | simoncpu | (i don't even know if it works as a sip client) |
| 11:06.17 | simoncpu | i guess you can install one, in theory |
| 11:07.02 | microw | i seeked in google, and found the SipXtapi, it is a sip client sdk, is it good to use it to write my own softphone for embedded linux?? |
| 11:07.29 | *** join/#asterisk gabunga (n=chatzill@p54A7A938.dip0.t-ipconnect.de) |
| 11:07.40 | gabunga | hi all |
| 11:08.06 | gabunga | i am trying to create a voip account on our voip server |
| 11:08.14 | gabunga | i am completly newbee |
| 11:08.31 | gabunga | any idea how i can pass through quickly? |
| 11:08.35 | microw | i also found an "linphone", it gives some lib details about what libraries to use when developing your own softphone |
| 11:09.32 | microw | anyone has any suggestions about developing a softphone on an embedded linux (i prefer to use qt as gui)? |
| 11:11.01 | microw | since SipXtapi is from SipX, i don't know if it could interoperate with asterisk |
| 11:12.00 | microw | hi all, any idea about developing an softphone for embedded linux? |
| 11:13.01 | microw | or is there any popular and good open source references? |
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| 11:48.23 | *** join/#asterisk dlewis (n=dlewis@about/security/staff/dlewis) |
| 11:49.15 | dlewis | hey, has anyone hooked up an asterisk box with a Panasonic KX-TD1232-7 and a KX-TVS120? |
| 11:49.31 | *** join/#asterisk flush (n=SYN_SENT@ip216-239-79-48.vif.net) |
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| 11:54.01 | *** join/#asterisk mike8861 (i=chatzill@221.124.81.21) |
| 11:54.11 | mike8861 | hello all, |
| 11:54.29 | mike8861 | does asterisk 1. |
| 11:54.44 | mike8861 | does asterisk 1.6 beta 9 support SIMPLE or IM ? |
| 11:57.06 | *** join/#asterisk scampbell (n=scampbel@199.105.195.156) |
| 12:00.46 | *** join/#asterisk gilli (n=alec@dslnet.212-29-38.ip15.dokom.de) |
| 12:00.58 | gilli | hello |
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| 12:03.14 | *** join/#asterisk gabunga (n=chatzill@p54A7A938.dip0.t-ipconnect.de) |
| 12:03.46 | gabunga | hi all can someone tell me how i can create a account for a voip on a asterisk server |
| 12:04.00 | gabunga | to get password and user name |
| 12:07.39 | DarKnesS_WolF | gabunga: ?? what ? |
| 12:09.31 | *** join/#asterisk koenvi (i=d988fccb@gateway/web/ajax/mibbit.com/x-3aef402c42da9a02) |
| 12:10.34 | gabunga | ahh ok |
| 12:10.47 | gabunga | i have hier a wrking asterisk server |
| 12:10.48 | koenvi | hi all |
| 12:10.55 | gabunga | everything is working fine |
| 12:11.03 | gabunga | not done by me |
| 12:11.21 | gabunga | now i just would like to add another phone |
| 12:11.36 | koenvi | I just tried asterisk 1.6 rc on an Ubuntu hardy ... got choppy sound when doing first demo test through IAX2 |
| 12:11.49 | gabunga | i am trying to create a new account |
| 12:11.54 | *** join/#asterisk russellb (n=russellb@asterisk/developer-and-stable-maintainer/drumkilla) |
| 12:11.54 | *** mode/#asterisk [+o russellb] by ChanServ |
| 12:11.59 | koenvi | anybody else has this running successfuly? |
| 12:20.32 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 12:30.34 | DarKnesS_WolF | russellb: mmm u get up so early!!! |
| 12:30.39 | DarKnesS_WolF | ah it is not weekend overthere :P |
| 12:30.44 | russellb | heh |
| 12:30.51 | russellb | but I did get to the office at 7 AM |
| 12:30.59 | DarKnesS_WolF | russellb: even in weekend i get up early |
| 12:31.04 | russellb | been up since 4 AM today |
| 12:31.04 | DarKnesS_WolF | russellb: u working hours starting from 7 !? |
| 12:31.12 | russellb | I don't _have_ to be in that early |
| 12:31.16 | russellb | I choose to |
| 12:31.22 | DarKnesS_WolF | damn it u did drink lots of redbull !? |
| 12:31.32 | russellb | heh ... coffee in the morning |
| 12:31.36 | DarKnesS_WolF | i can tell ;-) |
| 12:31.48 | DarKnesS_WolF | if i have a car i don't think i'll ever leave the office ;-) |
| 12:31.58 | DarKnesS_WolF | i love where i work :-) and i enjoy it |
| 12:33.13 | russellb | that's a good thing |
| 12:33.27 | russellb | I can get a lot done early in the morning when nobody is here yet ... |
| 12:35.25 | *** join/#asterisk XnOSX (n=XnOSX@212.145.173.80) |
| 12:36.06 | *** part/#asterisk roxlu_ (n=Roxlu@90-145-42-196.wxdsl.nl) |
| 12:37.09 | dlewis | hey russellb, do you have any experience with hooking up a Panasonic KX-TD1232-7 and a KX-TVS120 with asterisk? |
| 12:37.23 | russellb | Nope |
| 12:37.31 | russellb | I don't actually use this stuff :-p |
| 12:37.32 | dlewis | ok |
| 12:37.35 | dlewis | lol |
| 12:37.35 | dlewis | ok |
| 12:38.03 | russellb | I just help make it. I rely on talking to you guys to figure out what needs to be added, fixed, or improved. :) |
| 12:39.55 | *** join/#asterisk rwaite (n=richard@rrcs-74-218-125-86.central.biz.rr.com) |
| 12:40.41 | DarKnesS_WolF | russellb: lots actually :-) |
| 12:40.41 | DarKnesS_WolF | russellb: there is this killing bug where if u have asterisk and sip phones in the same network and then =----> internet =----- >sip provider |
| 12:40.49 | DarKnesS_WolF | if the internet goese down the sip phones lose the functions of registering " actually it do register " but it can't use zap / other provider or anything |
| 12:40.52 | russellb | the internet is overrated |
| 12:41.06 | DarKnesS_WolF | due to the internet is down and i have to create ad ummy sip register on teh same box |
| 12:41.14 | russellb | have you enabled the DNS manager? |
| 12:41.19 | DarKnesS_WolF | even the phones can't talk to eachothers |
| 12:41.20 | dlewis | ok, cool |
| 12:41.21 | russellb | sounds like a dns related issue ... |
| 12:41.24 | DarKnesS_WolF | nop didn't |
| 12:42.03 | DarKnesS_WolF | what i did read that asterisk tryes to check teh SIP providers 1st before even making internal calls :-s so what i have to do is to crete ad ummy sip proivder on teh same box 127.0.0.1 and everythign works if the internet is down |
| 12:42.09 | DarKnesS_WolF | russellb: looks / sounds strange |
| 12:42.17 | DarKnesS_WolF | but happen with me in 2 installations so far 1.2 and 1.4 |
| 12:42.45 | *** join/#asterisk lmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 12:42.45 | *** mode/#asterisk [+o lmadsen] by ChanServ |
| 12:43.11 | DarKnesS_WolF | russellb: heard of that before ? |
| 12:43.44 | russellb | you're working around the DNS issue when you do that |
| 12:43.58 | russellb | what you said is not quite how it works, though |
| 12:44.06 | DarKnesS_WolF | mmm that waht i did read online :-D |
| 12:44.15 | DarKnesS_WolF | so it is only when the DNS manager is enabled ? |
| 12:44.16 | *** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 12:44.19 | russellb | not everything you read online is correct |
| 12:44.29 | DarKnesS_WolF | if not local SIP phones should works fine ? |
| 12:44.33 | russellb | no, I"m saying that the reason you are having trouble is when Asterisk hangs on DNS lookups |
| 12:44.43 | russellb | enabling the dns manager will help some of that, but doesn't entirely fix it |
| 12:44.48 | russellb | but it should help |
| 12:44.51 | DarKnesS_WolF | russellb: yes i thought so also and what really drive me crazy if there is a cretical bug like that since 1.2 why it is not fixed i can't believe it |
| 12:45.10 | russellb | the bug is your broken network |
| 12:45.11 | russellb | :-p |
| 12:45.16 | DarKnesS_WolF | russellb: m ok i c |
| 12:45.19 | DarKnesS_WolF | russellb: yes :P i am bad :P |
| 12:45.20 | russellb | but i understand, and that situation has been vastly improved since 1.2 |
| 12:45.27 | russellb | keep in mind, 1.2 is ancient in our eyes at this point ... |
| 12:45.31 | russellb | our, being the developers |
| 12:45.35 | DarKnesS_WolF | sure |
| 12:45.51 | DarKnesS_WolF | russellb: so any final fix yet? |
| 12:45.53 | *** join/#asterisk [TK]D-Fender (n=joe@216.191.106.163) |
| 12:45.57 | russellb | I checked yesterday, we make more than twice as many changes to Asterisk 1.6 than 1.4 these days :) |
| 12:46.13 | russellb | I think some changes in 1.6 will fix what you're running in to |
| 12:46.46 | DarKnesS_WolF | perfect |
| 12:46.57 | DarKnesS_WolF | russellb: so what exactly u code in asterisk ? |
| 12:47.04 | russellb | everything |
| 12:47.19 | DarKnesS_WolF | russellb: :-D perfect !! so any ideas about new fetuers? |
| 12:47.22 | russellb | yes. |
| 12:47.23 | russellb | :-p |
| 12:47.28 | DarKnesS_WolF | ok shoot :D |
| 12:47.57 | russellb | Here: http://svn.digium.com/view/asterisk/trunk/CHANGES?view=markup |
| 12:47.58 | russellb | heh |
| 12:48.01 | DarKnesS_WolF | russellb: new about chan_bluetooth ? used it long ago with 1.2 but never worked :-) |
| 12:48.02 | russellb | that's what is already done, anyway |
| 12:48.15 | russellb | Asterisk 1.6 (-addons) has chan_mobile |
| 12:48.17 | DarKnesS_WolF | ok will read :-) |
| 12:48.22 | DarKnesS_WolF | stable enough now :D? |
| 12:48.32 | russellb | never used it ... |
| 12:49.12 | DarKnesS_WolF | Directory now permits both first and last names to be matched at the same 66 time |
| 12:49.24 | DarKnesS_WolF | hehe already did that :P just add a 2nd line with same voicemail box and switch the name :P |
| 12:49.39 | DarKnesS_WolF | last name then 1st name in one and then 1st name and last name on the other :D |
| 12:50.38 | *** join/#asterisk nn (n=nn@unaffiliated/nn) |
| 12:52.13 | *** join/#asterisk caio1982 (i=caio1982@CAcert-br/caio1982) |
| 12:52.40 | *** join/#asterisk Entr4nced (n=IMG001@67-129-213-39.dia.static.qwest.net) |
| 12:52.50 | DarKnesS_WolF | russellb: mmm nice , also any new news about SRTP ? |
| 12:53.19 | russellb | We are working very hard to do what we have to do to get it finished up |
| 12:53.23 | *** join/#asterisk ar3dam (n=ar3dam@189.156.217.142) |
| 12:53.32 | DarKnesS_WolF | russellb: u make me happy :D |
| 12:53.36 | russellb | I can't give you an estimate on a timeframe right now |
| 12:53.57 | DarKnesS_WolF | russellb: don't worry i have been waiting for some things sinec asterisk 1.0 and even cvs before that :P |
| 12:54.05 | russellb | heh |
| 12:54.16 | russellb | 1.0 <3 |
| 12:54.50 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 12:54.58 | DarKnesS_WolF | russellb: one of the things but i don't think it is related to u is no one is porting zaptel to openbsd anymore |
| 12:55.13 | *** part/#asterisk sn9 (n=danielg4@gimpelevich.san-francisco.ca.us) |
| 12:55.14 | DarKnesS_WolF | i am sure u can get teh freebsd one running on openbsd but after like i don't know how much effort |
| 12:55.22 | ar3dam | hi there ... i have a question, the time of linux is the same time of asterisk? because when i see the Master.csv not match the time... |
| 12:55.52 | ar3dam | how can set same time on asterisk with time of linux? |
| 12:56.06 | russellb | by default, it uses GMT |
| 12:56.10 | russellb | to use localtime, edit cdr.conf |
| 12:56.15 | russellb | there is an option for that |
| 12:56.31 | DarKnesS_WolF | russellb: gtg eat lanch will be back later ;-) |
| 12:56.37 | russellb | enjoy |
| 12:57.25 | ar3dam | ok, @russellb thanks for tips |
| 12:57.36 | russellb | np |
| 12:58.43 | ManxPowerNap | russellb: I though 1.6 would ship when Mark got tired of everyone whining about it? |
| 12:58.55 | russellb | not quite |
| 12:59.36 | dlewis | how does asterisks compare to trixbox in your opinion russellb? |
| 12:59.49 | russellb | trixbox _includes_ asterisk |
| 12:59.51 | russellb | ~trixbox |
| 12:59.52 | jbot | [trixbox] a full linux distro that includes FreePBX and other 3rd party add-ons. It is all this extra stuff which makes trixbox VERY difficult to support, and is not supported in #asterisk. Try asking in #trixbox or on their forums & wiki at http://www.trixbox.org/. We do not recommend using it. |
| 12:59.55 | [TK]D-Fender | dlewis: How do tires compare to CARS? |
| 13:00.02 | [TK]D-Fender | lol |
| 13:00.19 | russellb | trixbox is crap, though, btw. |
| 13:00.34 | russellb | asterisk is awesome, and I even like FreePBX |
| 13:00.39 | russellb | but trixbox is teh evil |
| 13:00.52 | dlewis | why? |
| 13:01.08 | dlewis | [TK]D-Fender: how does not funny compare to corny? |
| 13:01.18 | mvanbaak | DarKnesS_WolF: porting freebsd zaptel to OpenBSD is not as easy as it looks like |
| 13:01.28 | ar3dam | excuse me, about my experiencie with trixbox, i compare in 2 events, and i prefer use asterisk. |
| 13:01.35 | russellb | the management of that project has been very poor. I would recommend using one of the other options for a distro that includes asterisk and FreePBX |
| 13:02.00 | dlewis | ok |
| 13:02.14 | russellb | (or just using asterisk by itself) |
| 13:02.16 | [TK]D-Fender | I would recommend running the distro of YOUR choice and installing * on that. |
| 13:02.18 | russellb | it depends on what you're looking for |
| 13:02.35 | ar3dam | trixbox maybe u can use to learn some about extension or voip |
| 13:02.51 | russellb | be careful, it eats babies |
| 13:03.01 | russellb | and kills kittens |
| 13:03.15 | mvanbaak | ok, dont use it |
| 13:03.20 | russellb | lol |
| 13:03.21 | [TK]D-Fender | ar3dam>trixbox maybe u can use to learn some about extension or voip <-- huh?! |
| 13:03.25 | mvanbaak | with the first thing I can live, but killing kittens .... |
| 13:03.30 | ar3dam | yes, dont use it |
| 13:04.12 | ar3dam | yes if is beginner, maybe can learn little |
| 13:04.29 | brodiem | ... |
| 13:04.38 | russellb | maybe ... but in that case, use one of the other FreePBX options, or Switchvox Free ... |
| 13:04.46 | mvanbaak | the only thing trixbox can learn you is how it shouldn't be done |
| 13:04.54 | russellb | or take the steep curve (but higher peak) learning option and use asterisk by itself |
| 13:05.22 | [TK]D-Fender | ar3dam: Quick answer usually because that trixbox and * GUI's in general teach you virtually NOTHING and fill your head with improper terms that will only screw you over more when you come here. |
| 13:05.51 | lmadsen | or use Asterisk with the Asterisk-GUI which lets you actually create dialplan |
| 13:06.10 | russellb | ah yes, that GUI is a good learning tool, as well |
| 13:06.14 | russellb | ~asterisk-gui |
| 13:06.17 | russellb | hrm... |
| 13:06.50 | ManxPowerNap | ~zeeek |
| 13:06.51 | jbot | extra, extra, read all about it, zeeek is someone who once said "learning asterisk using a GUI is like learning sex through masturbation. You'll never get to the good stuff." |
| 13:07.04 | ManxPowerNap | You cannot learn Asterisk by using a GUI. |
| 13:07.10 | russellb | jbot: asterisk-gui is The Asterisk-GUI is an open source project lead by Digium. It's client-driven, and its only dependency is Asterisk itself. Check out the new version from svn here: $ svn co http://svn.digium.com/svn/asterisk-gui/branches/2.0 asterisk-gui-2.0 |
| 13:07.10 | jbot | russellb: okay |
| 13:07.18 | *** join/#asterisk [intra]lanman (n=lanman@freeswitch/developer/intralanman) |
| 13:07.52 | *** join/#asterisk Nate187 (n=Nate187@gw.bigrivertel.net) |
| 13:11.08 | [TK]D-Fender | ~asterisk-gui |
| 13:11.08 | jbot | [~asterisk-gui] The Asterisk-GUI is an open source project lead by Digium. It's client-driven, and its only dependency is Asterisk itself. Check out the new version from svn here: $ svn co http://svn.digium.com/svn/asterisk-gui/branches/2.0 asterisk-gui-2.0 |
| 13:11.52 | *** join/#asterisk mort_gib (n=mjensen@dsl-p4-177.gibconnect.com) |
| 13:13.14 | mort_gib | HI, I'm trying to install * on a fresh CentOS 5.2 |
| 13:13.25 | mort_gib | When I try make menuselect |
| 13:14.03 | mort_gib | I get : *** Install ncurses to use the menu interface! *** |
| 13:14.26 | [TK]D-Fender | mort_gib: "yum install ncurses*" |
| 13:14.44 | mort_gib | rpm -q ncurses gives: ncurses-5.5-24.20060715 |
| 13:14.47 | ManxPowerNap | mort_gib: Install ncurses-dev or ncurses-devel or whatever your distro uses for the ncurses development header files. |
| 13:14.56 | mort_gib | So ncurses IS installed |
| 13:15.05 | ManxPowerNap | mort_gib: but ncurses-devel is not |
| 13:15.26 | mort_gib | No, doing that now |
| 13:16.09 | mort_gib | Thanks, worked |
| 13:16.42 | [TK]D-Fender | mort_gib: http://www.asterisk.org/support/install |
| 13:16.48 | ManxPowerNap | Personally, I think the message should be changed to be more clear. |
| 13:17.01 | [TK]D-Fender | mort_gib: Old, but that tells you stuff we've needed for a long time... |
| 13:17.14 | ManxPowerNap | *** Install ncurses library and header files to use the menu interface! *** |
| 13:17.39 | mort_gib | I started there :-) must have overlooked that one |
| 13:17.49 | ManxPowerNap | It is just as confusing to newbies, but much clearer for non-newbies |
| 13:17.52 | [TK]D-Fender | mort_gib: AMAZING.... |
| 13:17.57 | mort_gib | Thanks anyway |
| 13:19.53 | ManxPowerNap | mort_gib: missing a single option in the asterisk config could allow someone to use your system to make calls billed to you. You should read more carefully. |
| 13:20.40 | mort_gib | I know, I know |
| 13:21.57 | ManxPowerNap | FEMA recently had their new PBX voicemail system hacked and something like a quarter of a million dollars of calls were run thru the system before someone discovered the issue. |
| 13:22.41 | Nivex | yeah, but that's FEMA. We've come to expect that level of incompetence and apathy from them. |
| 13:22.47 | mort_gib | How do you check for that?? |
| 13:23.05 | mort_gib | Is there a Nessus for * :-) |
| 13:23.11 | ManxPowerNap | Sorry, it was only $12,000 |
| 13:23.14 | ManxPowerNap | http://www.msnbc.msn.com/id/26319201/ |
| 13:23.20 | *** join/#asterisk candyban_ (n=candyban@5.192-136-217.adsl-dyn.isp.belgacom.be) |
| 13:23.29 | candyban_ | Hi all |
| 13:23.30 | ManxPowerNap | mort_gib: you design your dialplan carefully |
| 13:24.03 | *** part/#asterisk switchcat (n=center@pool-72-94-244-136.phlapa.east.verizon.net) |
| 13:24.52 | candyban_ | Can anyone point me to some good docs with regards to using cisco 7940s with asterisk? The phone constantly reboots ... and I have no idea why |
| 13:25.05 | mort_gib | Yeah, I actually did play around with context a while... I'm sure I'm no expert but it DID dawn on me what might happen if your IVR ends up in the wrong place |
| 13:25.33 | *** join/#asterisk bradleyprice86 (n=bradleyp@fw.datafax.net) |
| 13:25.54 | bradleyprice86 | Can't create numeric folder in linux. Any suggestions? |
| 13:26.00 | candyban_ | this is a new setup ( no phones yet ) and I only installed (a working) asterisk like a year ago (so I'm very new at asterisk) |
| 13:26.39 | [TK]D-Fender | candyban_: Go review the WIKI guides on setting them up. |
| 13:26.58 | [TK]D-Fender | bradleyprice86>Can't create numeric folder in linux. Any suggestions? <-- huh? |
| 13:27.09 | rwaite | hah. 'mkdir 7' |
| 13:27.23 | *** join/#asterisk moy (n=moy@189.169.67.95) |
| 13:27.25 | [TK]D-Fender | Sure worked for me... |
| 13:27.36 | bradleyprice86 | It says mkdir: cannot create directory `7': No such file or directory |
| 13:27.52 | candyban_ | [TK]D-Fender, I did (it loads the SIP image properly ... but it constantly reboots after that) |
| 13:28.03 | *** join/#asterisk n3hxs (n=HAMming@151.196.87.132) |
| 13:28.16 | bradleyprice86 | Everytime I create extensions it will not create the voicemail folder to go along with it. |
| 13:28.23 | [TK]D-Fender | bradleyprice86: Please note your QUOTES don't match |
| 13:28.45 | [TK]D-Fender | bradleyprice86: And thats a RIGHTS problem, |
| 13:28.56 | [TK]D-Fender | bradleyprice86: More FreePBX BS |
| 13:29.02 | candyban_ | [TK]D-Fender, is there a way to get more information (anywhere) about a reason for the reboot? |
| 13:29.16 | [TK]D-Fender | candyban_: www.google.com |
| 13:29.20 | bradleyprice86 | [TK]D-Fender: Ye |
| 13:29.24 | [TK]D-Fender | candyban_: this isn't #cisco you know. |
| 13:30.22 | *** join/#asterisk MrNaz (n=naz@124-168-102-159.dyn.iinet.net.au) |
| 13:30.23 | candyban_ | [TK]D-Fender, I know ... but you are familiar with all kinds of phones ... perhaps it was a common/known issue? |
| 13:30.44 | bradleyprice86 | [TK]D-Fender: root should be able to create it. |
| 13:31.04 | [TK]D-Fender | candyban_: Constant reboots would usually indicate a serious config file problem, unstable firmware or dead phone. |
| 13:31.12 | [TK]D-Fender | candyban_: And you have no useful details. |
| 13:32.06 | [TK]D-Fender | bradleyprice86: either * isn't running as root, or you are perhaps using a read-only location to start from... like that LINKED folder I recall mention of earlier |
| 13:32.24 | candyban_ | [TK]D-Fender, I tried with 3 or 4 different firmwares ... so it's not the firmare I suppose ... |
| 13:33.17 | *** join/#asterisk Lordrack (n=lordrack@200150013197.static.corp.wayinternet.com.br) |
| 13:33.43 | candyban_ | [TK]D-Fender, I currently use a config which I downloaded (and customized) ... so likely the problem is there ... I'll try with a bare minimum config |
| 13:34.09 | *** join/#asterisk theHub (n=theHub@69.177.93.21) |
| 13:35.47 | candyban_ | [TK]D-Fender, ok ... found the problem ... SIPDefault.cnf had tftp_cfg_dir: /cisco/ ... while it should have been tftp_cfg_dir: cisco/ |
| 13:35.48 | bradleyprice86 | [TK]D-Fender: I can't create a numeric folder anywhere on this linux box. It doesn't matter which directory I am in. |
| 13:36.37 | [TK]D-Fender | bradleyprice86: pastebin the attempts and "whoami" and ls -la" and "mount" |
| 13:36.41 | [TK]D-Fender | ~pb |
| 13:36.43 | jbot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste |
| 13:36.47 | candyban_ | [TK]D-Fender, thanks for your help :o) |
| 13:37.19 | ar3dam | hi there, how i can test from cli if echo cancellation is on? |
| 13:38.01 | candyban_ | [TK]D-Fender, I was constantly looking at the wrong file : SIP<mac> ... when using a minimum config didn't change anything I had to look for other config files |
| 13:38.22 | [TK]D-Fender | ar3dam: On for what? |
| 13:38.46 | bradleyprice86 | [TK]D-Fender: http://pastebin.com/m20fd6667 |
| 13:39.02 | ManxPowerNap | ar3dam: "zap show channel X" where X is the channel you want to look at. |
| 13:39.21 | ar3dam | ManxPowerNap, thks.. |
| 13:39.40 | Lordrack | ManxPowerNap, how can i turn it on? |
| 13:40.05 | ar3dam | echo cancellation is currently off... |
| 13:40.06 | ManxPowerNap | Lordrack: in /etc/asterisk/zapata.conf then restart Asterisk |
| 13:40.14 | ManxPowerNap | ar3dam: is that channel in use? |
| 13:40.29 | Lordrack | <PROTECTED> |
| 13:40.31 | *** join/#asterisk tkbeat (n=tk@p578b2de9.dip0.t-ipconnect.de) |
| 13:40.46 | ManxPowerNap | You will NOT have echo canceling enabled on channels that are not in use. |
| 13:41.02 | *** join/#asterisk zeeqy (n=zeeqy@196-209-147-186-tbnb-esr-3.dynamic.isadsl.co.za) |
| 13:41.07 | ManxPowerNap | Do you really want Asterisk to be doing the very CPU intensive stuff for EC on a channel that is not being used? |
| 13:41.08 | [TK]D-Fender | bradAnd if you create ABC where you are? |
| 13:41.39 | zeeqy | Some help regarding port forwarding for remote extention...??? |
| 13:41.44 | ar3dam | mmmm so what i can do? |
| 13:42.09 | bradleyprice86 | [TK]D-Fender: http://pastebin.com/m1e78587 -- worked fine |
| 13:43.21 | ManxPowerNap | ar3dam: You do not have a problem |
| 13:43.57 | zeeqy | anyone ever had luck with remote extension on E61 with Asterisk??? |
| 13:44.07 | [TK]D-Fender | bradleyprice86: mkdir: cannot create directory `4240': No such file or directory <- the quotes around 4240 don't match. Something screwy is happening here. |
| 13:44.41 | [TK]D-Fender | ~sipnat |
| 13:44.42 | jbot | [~sipnat] Quick guide on configuring * + SIP behind NAT : http://www.aocomputing.net/?p=3 , otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
| 13:44.44 | [TK]D-Fender | zeeqy: ^^^ |
| 13:45.24 | ar3dam | ManxPowerNap, ok, so i try to enabled |
| 13:45.26 | [TK]D-Fender | bradleyprice86: try : mkdir "123" |
| 13:45.42 | gilli | hi everyone. i'm sorry if I'm asking this on the wrong channel, so just let me know. is there anywhere a documentation of the zaptel drivers, so I can use them as a framework to build an own application based on that? |
| 13:45.46 | zeeqy | thanks Fender!!! |
| 13:46.13 | [TK]D-Fender | gilli: The source is pretty much all you've got. |
| 13:46.29 | bradleyprice86 | [TK]D-Fender: http://pastebin.com/m7d2c06a --- didn't work either |
| 13:46.52 | [TK]D-Fender | bradleyprice86: Go ask in ##linux |
| 13:47.28 | bradleyprice86 | [TK]D-Fender: Ok. |
| 13:49.48 | brodiem | bradleyprice86: only caught the last piece of what you guys are discussing, but usually that mkdir error gets spit out when your current working directory no longer exists. |
| 13:50.10 | *** part/#asterisk zeeqy (n=zeeqy@196-209-147-186-tbnb-esr-3.dynamic.isadsl.co.za) |
| 13:51.08 | [TK]D-Fender | that is a thoght. |
| 13:52.11 | jameswf-home | why are you quoting? |
| 13:52.19 | *** join/#asterisk LeddyHM (i=leddy@da-club.with.my.beerandcondoms.com) |
| 13:52.30 | [TK]D-Fender | jameswf-home: Sanity check from previous failures |
| 13:52.32 | gilli | [TK]D-Fender: thanks for the answer. Do you know of any simple applications that use some basic zaptel interfaces just for studying? |
| 13:52.54 | [TK]D-Fender | gilli: Asterisk :) |
| 13:53.01 | jameswf-home | i guess dont matter works both ways here |
| 13:53.07 | *** join/#asterisk putnopvut (n=putnopvu@nat/digium/x-5f9afe07c0719e29) |
| 13:53.07 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 13:53.07 | bradleyprice86 | brodiem: I know the directory is there. It is the asterisk voicemail folder I am trying to create a folder for extension 4240, because it strangely dissappeard |
| 13:53.43 | gilli | [TK]D-Fender: any...simpler? :) |
| 13:54.07 | [TK]D-Fender | gilli: Doubt it. Who else cares about Zaptel? |
| 13:54.10 | a1fa | oh wow |
| 13:54.13 | a1fa | [TK]D-Fender never sleeps |
| 13:54.16 | jameswf-home | gilli: zaptel has alot of apps, zttool ztcfg ztmonitor |
| 13:54.17 | [TK]D-Fender | gilli: Perhaps FreeSWITCH |
| 13:54.46 | a1fa | damn, if I had a cent for everytime [TK]D-Fender answered a question... I'd be brazillionare :p |
| 13:55.05 | [TK]D-Fender | a1fa: Damn South-Americans. |
| 13:55.10 | a1fa | ;) |
| 13:55.11 | a1fa | not me |
| 13:55.15 | jameswf-home | ah but if everytime he answered 1 right :) |
| 13:55.15 | a1fa | <- Arkansas |
| 13:55.17 | gilli | [TK]D-Fender: ah, that might be a lot of help already..thanks a bunch! |
| 13:55.28 | brodiem | bradleyprice86: Pastebin: pwd; cd `pwd`; mkdir test; whoami; mount; df -m; uname -a |
| 13:55.48 | a1fa | hehe.. you know this joke.. Bush was in the office, and Secretary of State comes to him and tells him Two brazillian people died infront of the WH. |
| 13:55.56 | a1fa | Bush asks how many is two brazillian? |
| 13:55.59 | a1fa | haha |
| 13:56.18 | *** join/#asterisk tobias (n=tobias@user-0c2hj2e.cable.mindspring.com) |
| 13:57.28 | a1fa | not funny :( |
| 13:59.10 | Lordrack | a1fa, not funny. I am brazillian. |
| 13:59.41 | DarKnesS_WolF | lol |
| 14:00.58 | *** join/#asterisk mog (n=mog@nat/digium/x-09ef9d9ef8156eab) |
| 14:00.58 | *** mode/#asterisk [+o mog] by ChanServ |
| 14:02.10 | a1fa | haha |
| 14:02.23 | a1fa | Lordrack : hehe |
| 14:02.28 | Lordrack | :D |
| 14:02.45 | tobias | all calls and music playback sound fine, but when i enter a conference, it's /really/ choppy. any ideas? |
| 14:03.02 | tobias | even with just 2 people |
| 14:03.10 | a1fa | timing |
| 14:03.12 | [TK]D-Fender | tobias: Zaptel timer isn't stable |
| 14:04.47 | *** join/#asterisk codefreeze-lap (n=murf@72.21.67.40) |
| 14:05.12 | *** join/#asterisk jaytee (n=jforde@unaffiliated/jaytee) |
| 14:06.18 | tobias | [TK]D-Fender: i'm running it in a VM whose kernel has CONFIG_HZ=1000, and i built zaptel with RTC disabled |
| 14:06.27 | Lordrack | My /var/log/asterisk/message's getting too many warnings like "WARNING[3469] chan_sip.c: sip_xmit of 0xb656aa4c (len 568) to xxx.xxx.xxx.xxx(unknow public ip):32946 returned -2: Network is unreachable" any ideas? |
| 14:06.32 | tobias | it worked fine for asterisk 1.2 - could that have changed for 1.4? |
| 14:06.45 | [TK]D-Fender | tobias: in a VM? Good luck with all that... |
| 14:07.20 | tobias | [TK]D-Fender: it worked flawlessly before i upgraded to 1.4 |
| 14:07.29 | tobias | + freepbx |
| 14:07.36 | jaytee | "I built a nuclear weapon in Second Life and I just don't understand why it doesn't destroy anything real when I detonate it." |
| 14:08.35 | jaytee | asterisk and a girlfriend, two things better left real versus virtual |
| 14:09.42 | *** join/#asterisk anthm (n=anthm@freeswitch/developer/anthm) |
| 14:10.00 | *** join/#asterisk shazaum (n=shazaum@200.175.61.250.static.gvt.net.br) |
| 14:11.30 | nnY_2 | ok now i am confused |
| 14:11.49 | nnY_2 | so i had a phone that would not register yesterday, after having worked for 4 months |
| 14:12.04 | nnY_2 | thought that the phone itself had shat the bed, swapped it out today, and still no luck |
| 14:12.19 | *** join/#asterisk anonymouz666 (n=anonymou@201.19.95.130) |
| 14:12.44 | *** join/#asterisk codefreeze-lap (n=murf@72.21.67.40) |
| 14:13.25 | *** join/#asterisk axisys (n=axisys@155.70.141.45) |
| 14:15.21 | nnY_2 | 100/100 192.168.100.105 D 5060 UNREACHABLE |
| 14:15.44 | nnY_2 | i could not qualify it but it still doesn't show up in the registry |
| 14:16.12 | *** join/#asterisk esaym (n=user@cpe-70-120-89-6.satx.res.rr.com) |
| 14:16.27 | brodiem | nnY_2: you don't have nat enabled; do you not need nat support? |
| 14:17.11 | tobias | jaytee: asterisk does not use near the resources my girlfriend does :) |
| 14:17.25 | nnY_2 | brodiem they are both on the same local subnet, no iptables or anything on the box |
| 14:17.52 | [TK]D-Fender | nnY_2: Can you ping it? Also try calling it directly. |
| 14:18.19 | nnY_2 | pings fine |
| 14:18.24 | nnY_2 | one sec on calling it |
| 14:18.25 | brodiem | nnY_2: dump your sip traffic to see if anything is being passed, and if so, the reason for it not authenticating |
| 14:18.39 | nnY_2 | will do one sec |
| 14:19.39 | jaytee | tobias, it's a sad but true fact that the HM models of girlfriends are much more resource hungry than the LM models but the LM models are usually uglier :-) |
| 14:21.22 | nnY_2 | http://pastebin.com/m7cf734ad |
| 14:21.30 | nnY_2 | the phone can call out too |
| 14:23.16 | [TK]D-Fender | nnY_2: SIP/2.0 401 Unauthorized |
| 14:23.31 | [TK]D-Fender | nnY_2: register auth = bad so no qualify. CALL auth good = can CALL. |
| 14:24.03 | *** join/#asterisk Mawkee (n=Mawkee@200.152.178.136) |
| 14:24.25 | *** part/#asterisk Mawkee (n=Mawkee@200.152.178.136) |
| 14:24.37 | nnY_2 | hmm wth it just stopped working... |
| 14:24.39 | nnY_2 | http://pastebin.com/m7f9f713e |
| 14:24.42 | nnY_2 | the configs match |
| 14:26.24 | [TK]D-Fender | nnY_2: When have you rebooted the phone last? |
| 14:27.17 | brodiem | seems a bit odd that the aastra stops responding |
| 14:27.34 | nnY_2 | it was just put in today. We had another phone there that stopped working. Assuming it was broken, as all the configs matched and it worked, we had a new one sent over. It booted up, provisioned, got new firmware etc and the same problem popped up. So now I am gonna have to dig around and see why it is not authing |
| 14:27.40 | [TK]D-Fender | perhaps the phone is locked. |
| 14:28.51 | brodiem | although I find aastra very picky about the content of the SIP messages.. i.e. I cannot get mine to re-register any less than than the default of 1hr because it doesn't like how the registration response from Ast indicates the expire time and won't follow it |
| 14:29.47 | brodiem | although according to aastra was due to Ast doing its NAT rewriting |
| 14:30.40 | iCEBrkr | [TK]D-Fender: Know any Windows people I could use for stability testing for my call manager project? |
| 14:31.04 | [TK]D-Fender | iceSorry, I don't know any stable Windows people ;) |
| 14:31.09 | iCEBrkr | :( |
| 14:31.10 | iCEBrkr | lol |
| 14:31.37 | DarKnesS_WolF | windows Awwww :-s |
| 14:31.44 | *** join/#asterisk stimpie (n=stimpie@84-104-5-14.cable.quicknet.nl) |
| 14:32.16 | iCEBrkr | Yeah well.. |
| 14:33.20 | iCEBrkr | I should try and compile this thing with Mono :) |
| 14:33.49 | nnY_2 | not reallt sure where to go from here |
| 14:33.54 | nnY_2 | really |
| 14:34.03 | nnY_2 | gonna get aastra support on the phone |
| 14:34.39 | brodiem | nnY_2: for the hell of it turn on nat in your sip friend definition, even though it shouldn't do anything differently |
| 14:40.15 | *** join/#asterisk nny_1 (n=nny_1@64.203.237.47) |
| 14:40.21 | nny_1 | mIRC blows goats |
| 14:40.46 | nny_1 | YES! Please aastra, spam me with adverts while on hold for your crappy product! |
| 14:42.39 | *** join/#asterisk linuxstb (n=linuxstb@rockbox/developer/linuxstb) |
| 14:44.23 | nny_1 | omg |
| 14:44.24 | nny_1 | this sucks |
| 14:44.30 | nny_1 | their phone system sucks ass |
| 14:44.32 | iCEBrkr | nny_1: Punch it! |
| 14:44.34 | nny_1 | hehe |
| 14:45.54 | *** join/#asterisk wonderworld (n=ww@ip-62-143-216-14.hsi.ish.de) |
| 14:45.55 | *** join/#asterisk zydoon (n=zydoon@41.225.157.25) |
| 14:46.09 | *** part/#asterisk zydoon (n=zydoon@41.225.157.25) |
| 14:48.18 | *** join/#asterisk AdamWest (n=Leif@CPE001d7e2f9574-CM0012c9db3d2e.cpe.net.cable.rogers.com) |
| 14:48.35 | *** mode/#asterisk [+o lmadsen] by ChanServ |
| 14:49.58 | [TK]D-Fender | lmadsen: na na na na na! |
| 14:50.27 | lmadsen | I don't want to meet your mom! I just want... |
| 14:51.59 | [TK]D-Fender | lmadsen: ! ! ! |
| 14:52.09 | creativx | god damn |
| 14:52.14 | *** join/#asterisk afink (n=afink@65-120-137-2.dia.static.qwest.net) |
| 14:53.16 | nnY_2 | brodiem i'll try that |
| 14:53.25 | nnY_2 | brodiem i think i tried that last night |
| 14:54.13 | nnY_2 | yeah still UNREACHABLE |
| 14:54.19 | nnY_2 | and sip show registry shows nada |
| 15:00.10 | brodiem | nnY_2: are you providing your auth credentials both in the 'global sip' or whatever it was called as well as the 'line' section? |
| 15:00.26 | [TK]D-Fender | nnY_2: iT WON'T, BECAUSE THAT COMMAND SHOWS THINGS asterisk IS REGISTERED TO |
| 15:00.34 | [TK]D-Fender | darn caps |
| 15:00.49 | brodiem | [TK]D-Fender: heh you must not look at your monitor while you type :) |
| 15:01.30 | *** join/#asterisk korihor (n=korihor@190.78.32.60) |
| 15:01.49 | Maliuta | brodiem: you don't touch type while looking at another of your 3 monitors? |
| 15:04.30 | *** join/#asterisk spokra (n=spokra@host093-179-187.sea0.speakeasy.net) |
| 15:04.44 | brodiem | Maliuta: no not really, not when typing sentances.. |
| 15:04.52 | [TK]D-Fender | brodiem: Nope. |
| 15:04.57 | Maliuta | brodiem: how unco of you |
| 15:06.32 | brodiem | Maliuta: I do ~100WPM, so it's quite alright with me how I type |
| 15:07.52 | brodiem | and I have typeracer.com screen shots to prove it, lol |
| 15:08.09 | brodiem | when I did 140 heh |
| 15:12.28 | *** join/#asterisk Zeeek (n=Zeeek@pdpc/supporter/active/Zeeek) |
| 15:12.49 | Zeeek | i love asterix |
| 15:13.10 | Zeeek | but it is less configurable than asterisk |
| 15:13.39 | [TK]D-Fender | Zeeek: Obelix > you |
| 15:15.38 | *** join/#asterisk DerbyD (n=cueto@159.90.26.202) |
| 15:15.43 | Zeeek | Only one left. One week of vacation. Just one. |
| 15:16.11 | [TK]D-Fender | Zeeek: I haven't touched any of my 4 weeks this year yet... |
| 15:16.23 | Zeeek | Well, I just used up 5 of mine |
| 15:16.51 | Zeeek | diskutil resizeVacation +100 days |
| 15:17.22 | creativx | speaking of vacation |
| 15:17.24 | creativx | its weeeeeeeeeekend |
| 15:17.45 | [TK]D-Fender | Everybody's working for the weekend! |
| 15:17.48 | [TK]D-Fender | rocks out |
| 15:17.57 | nnY_2 | heh |
| 15:18.04 | Zeeek | yeeessssss. Friday! And Fridays at 12 Noon EDT, 9AM PDT there's the big conference call http://voipusersconference.org |
| 15:18.06 | creativx | im actually leaving for home right now. hehe |
| 15:18.07 | creativx | bbl |
| 15:18.11 | nnY_2 | so what's the best way to check if a sip device is registered? |
| 15:18.11 | *** join/#asterisk jpeeler (n=jpeeler@asterisk/digium-software-dev/jpeeler) |
| 15:18.12 | Maliuta | [TK]D-Fender: I'm afraid the boys are back in town |
| 15:18.17 | DerbyD | hi |
| 15:18.24 | nnY_2 | i know of a couple, wondering which is the prefered |
| 15:18.24 | Zeeek | sip show reg<tab> |
| 15:18.52 | nnY_2 | empty |
| 15:18.53 | Zeeek | no, sip show peers |
| 15:18.56 | [TK]D-Fender | nnY_2: "SIP SHOW PEER [PEER]" |
| 15:18.57 | *** join/#asterisk n3hxs (n=HAMming@151.196.87.132) |
| 15:18.58 | DerbyD | Hi there, IHi |
| 15:19.02 | nnY_2 | [TK]D-Fender thanks |
| 15:19.10 | *** join/#asterisk mort_gib (n=mjensen@16.Red-83-36-63.staticIP.rima-tde.net) |
| 15:19.13 | DerbyD | I wanted to get some help of you guys |
| 15:19.35 | Zeeek | the next available agent is [TK]D-Fender |
| 15:20.17 | nnY_2 | http://pastebin.com/m31b21211 |
| 15:20.18 | DerbyD | I am doing some testing in my job with a AX100P ATCOM card |
| 15:20.27 | nnY_2 | that pastebin is the info for the aastra' |
| 15:20.28 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 15:20.34 | nnY_2 | looks.. abnormal |
| 15:20.56 | DerbyD | I need to know if this card is able to work with digital telephone lines |
| 15:21.14 | *** join/#asterisk Sargun (n=xbmodder@atarack/staff/sargun) |
| 15:21.36 | [TK]D-Fender | nnY_2: Addr->IP : (Unspecified) Port 0 <--- says it all |
| 15:21.41 | DerbyD | I already configured and got IVR working ok, but in analogic line, and need to get caller ID |
| 15:21.49 | [TK]D-Fender | DerbyD: No. |
| 15:22.54 | DerbyD | thanks for answer [TK]D-Fender, I switched line to digital yesterday and Asterisk stop working |
| 15:23.13 | [TK]D-Fender | DerbyD: And how do you switch a line to "digital"? |
| 15:23.26 | [TK]D-Fender | DerbyD: that can mean a lot of things.. |
| 15:24.07 | DerbyD | I am in a company, Central guys said that they can switch my line to digital in the "central Hardware" i guess |
| 15:25.13 | [TK]D-Fender | DerbyD: It does not become "digital" on the same line just like that from a switch at central. Go find out what you actually HAVE. |
| 15:25.25 | nnY_2 | [TK]D-Fender is this the phone goofing up? |
| 15:25.34 | DerbyD | they said that central always transmits the callerid, but with analogic line, i only got "SAP/1-1" |
| 15:26.03 | Qwell | [TK]D-Fender: they're transmitting the callerid over digital copper. |
| 15:26.21 | *** join/#asterisk stimpie (n=stimpie@84-104-5-14.cable.quicknet.nl) |
| 15:27.26 | DerbyD | I really don't understand, I assume that they switched the line in hardware |
| 15:28.07 | *** join/#asterisk CunningPike (n=arodgers@204.239.8.157) |
| 15:28.21 | DerbyD | back to the beginning, AX100P is not able to work in digital lines, that's why asterisk stop ansering the calls |
| 15:28.29 | [TK]D-Fender | DerbyD: "digital" isn't a term you can just throw around. Go find out the EXACT signalling the line is using. |
| 15:28.30 | DerbyD | *answering |
| 15:28.53 | Qwell | [TK]D-Fender: fxs_dgtl |
| 15:29.09 | [TK]D-Fender | Qwell: ..... |
| 15:29.22 | [TK]D-Fender | Qwell: Snarky today aren't we? ;P |
| 15:29.25 | Qwell | a bit |
| 15:30.10 | DerbyD | AX100P specification said that the card is able to "interpret" caller ID, I am really confuse in this subject, because it's a FXO used to plug in Analogic lines |
| 15:30.24 | Qwell | Who is Analogic? |
| 15:30.50 | *** join/#asterisk micheluntu (n=michele@dhcp197-193.cu.mi.it) |
| 15:32.14 | DerbyD | Sorry Qwell, I am sure a mess with the terms |
| 15:32.40 | Qwell | Don't use terms you don't understand. |
| 15:32.53 | [TK]D-Fender | DerbyD: First you mention CallerID problems, then you're takling about changing the entire line TECHNOLOGY. Make up your damn mind. |
| 15:33.13 | [TK]D-Fender | DerbyD: Go call your telco and have them explain everything to you |
| 15:33.16 | DerbyD | Qwell, do you think that if I change zaptel.conf to use fxs_dgtl, card will be able to work? |
| 15:33.28 | [TK]D-Fender | Qwell: Oh now you've done it... |
| 15:34.33 | DerbyD | I already called my telco, but it's a company, guys don't know about asterisk or maybe they don't wanted to help |
| 15:34.50 | [TK]D-Fender | DerbyD: That is not a valid answer. |
| 15:35.18 | [TK]D-Fender | DerbyD: They don't HAVE to know Asterisk. They should know what SERVICE they are providing you and be able to give you the precise terms for everything |
| 15:36.06 | [TK]D-Fender | DerbyD: I don't need to know the recipes a restaurant is going to prepare in order to sell them SALT. |
| 15:36.07 | *** join/#asterisk keith4 (n=keith@lust.CC.Lehigh.EDU) |
| 15:36.12 | *** join/#asterisk rwaite (n=richard@rrcs-74-218-125-86.central.biz.rr.com) |
| 15:36.32 | Zeeek | OMG I missed a session of semantics from [TK]D-Fender ? Damn |
| 15:39.27 | [TK]D-Fender | ~cluebat Zeeek |
| 15:39.28 | jbot | ACTION pulls out a ClueBat (tm) and thwaps Zeeek. |
| 15:39.37 | [TK]D-Fender | ClueBat NEVER misses! |
| 15:40.15 | kaldemar | some legendary stuff here. :) |
| 15:41.29 | Zeeek | Ahhhhhhhhhhhhhhhh! |
| 15:41.39 | Zeeek | starts his own channel |
| 15:42.01 | Zeeek | join us now for the VoIP Users Conference: http://x2z.eu |
| 15:42.10 | DerbyD | I got the point |
| 15:42.13 | Zeeek | IRC #voip-users-conference |
| 15:42.28 | DerbyD | I just wanted some help, thanks |
| 15:42.28 | Zeeek | voip-users is NOT a registered trademark while asterisk is |
| 15:43.13 | [TK]D-Fender | DerbyD: Want a clue before wanting help :) |
| 15:43.52 | nnY_2 | ~cluebat Aastra Support |
| 15:43.53 | jbot | ACTION pulls out a ClueBat (tm) and thwaps Aastra Support. |
| 15:43.56 | nnY_2 | ty |
| 15:44.04 | nnY_2 | they have no idea why either heh |
| 15:44.22 | nnY_2 | i think [TK]D-Fender knows more about this than they do |
| 15:44.41 | nnY_2 | then* |
| 15:44.47 | nnY_2 | need coffee |
| 15:46.07 | Zeeek | want clues |
| 15:46.11 | Zeeek | want money |
| 15:46.22 | Zeeek | want wine, women, song |
| 15:46.52 | *** join/#asterisk stimpie (n=stimpie@84-104-5-14.cable.quicknet.nl) |
| 15:48.18 | DarKnesS_WolF | nnY_2: what is the problem again ?may be i can help |
| 15:50.44 | *** join/#asterisk microw (i=microw@58.60.35.66) |
| 15:51.05 | microw | hi all, |
| 15:51.17 | microw | what sip client softphone are you using? |
| 15:52.00 | [TK]D-Fender | "target not found" |
| 15:54.03 | *** part/#asterisk micheluntu (n=michele@dhcp197-193.cu.mi.it) |
| 15:55.27 | *** join/#asterisk moy (n=moy@189.169.67.95) |
| 16:01.04 | *** join/#asterisk Gershwin (n=fake@63.250.233.162) |
| 16:01.39 | DarKnesS_WolF | microw: xlite / zoiper |
| 16:01.48 | DarKnesS_WolF | sometimes egika |
| 16:02.30 | *** join/#asterisk lzhang (n=lzhang@rrcs-67-78-33-170.sw.biz.rr.com) |
| 16:03.51 | lzhang | hello, I'm getting DTMF issues with double keypresses and false positives... |
| 16:04.16 | lzhang | relax dtmf is off; are there any other options I could try to fix it? |
| 16:04.33 | lzhang | this is over a T1 channel |
| 16:05.19 | microw | DarKnesS_Wolf: i am developing an softphone on a embedded linux, is there any good open source reference ?? |
| 16:08.40 | *** join/#asterisk |dennis| (n=Dennis@200.32.231.2) |
| 16:08.59 | DarKnesS_WolF | microw: i think iaxclient should have some good refrnces |
| 16:09.04 | DarKnesS_WolF | microw: or ur doing SIP onl ? |
| 16:09.16 | DarKnesS_WolF | microw: u'll do it java or wat ? |
| 16:09.18 | DarKnesS_WolF | what * |
| 16:11.02 | microw | DarKness_wolf: i am doing qt c++ on an arm chip with embedded linux |
| 16:11.17 | microw | iaxclient? where can i find it? |
| 16:12.46 | errr | nnY_2: who at aastra did you talk to? |
| 16:15.09 | microw | i am developing an voip softphone on an embedded linux on an arm chip, any good reference?? |
| 16:15.37 | *** join/#asterisk totalimpact8 (n=Miranda@72.169.145.198) |
| 16:19.02 | jeev | pump, pump the jam, pump it up |
| 16:20.22 | *** join/#asterisk [netman] (n=netman@107.Red-83-63-171.staticIP.rima-tde.net) |
| 16:20.31 | *** join/#asterisk sah-work (n=Bawbatos@65-119-47-100.dia.static.qwest.net) |
| 16:22.39 | DarKnesS_WolF | microw: just google iaxcleint |
| 16:22.43 | DarKnesS_WolF | iaxclient * |
| 16:22.54 | DarKnesS_WolF | microw: i am sure there is lots of refrefaces |
| 16:23.14 | DarKnesS_WolF | microw: and what i understand that all u need is teh compile the softphone with arm-gcc or arm-g+++ whatever the name is |
| 16:23.39 | DarKnesS_WolF | microw: i helped in some arm development not coding but compiling stuff for arm handheld devices |
| 16:24.47 | DarKnesS_WolF | microw: and also there are some arm images for qemu so u can test on the emulator ur code |
| 16:24.53 | *** join/#asterisk nicox (n=nicox@vie-nas-ge-0-2.onenet.at) |
| 16:27.37 | microw | so far my platform is qt on linux on a samsung2440 |
| 16:27.41 | *** join/#asterisk lowtek (n=no@adsl-074-185-089-046.sip.bhm.bellsouth.net) |
| 16:28.22 | JerJer | zoiper |
| 16:28.23 | microw | the boad is not handheld but wallheld, |
| 16:28.31 | microw | board * |
| 16:28.45 | JerJer | borat ? |
| 16:29.08 | microw | with a touch screen, calling audio and video |
| 16:30.14 | microw | zoiper? |
| 16:30.23 | DarKnesS_WolF | microw: get any opesource softphone try to compile it aginest arm platfrom using arm-gcc armg++ |
| 16:31.44 | microw | yuck |
| 16:32.42 | *** join/#asterisk gaetronik (n=gaetan@eficaz2.manquehue.net) |
| 16:32.49 | lowtek | thinks all softphones suck ... |
| 16:33.15 | gaetronik | hi there |
| 16:33.36 | gaetronik | i've an issue qith monitor |
| 16:34.16 | gaetronik | when i read the sound file i've one way sound for a bunch of files but not all |
| 16:34.16 | DarKnesS_WolF | gaetronik: show dialplan and error messges / output of the dialplan in pastebin.ca |
| 16:34.56 | gaetronik | it's a prod server it's a funcking mess the logs |
| 16:35.09 | gaetronik | but i wille send the dialplan in pastebin |
| 16:35.12 | gaetronik | ~pb |
| 16:35.13 | jbot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste |
| 16:35.21 | Anonissimus | anyone in here that designs solutions in asterisk? |
| 16:35.36 | tzanger | Anonissimus: what kind of solutions? |
| 16:35.50 | Anonissimus | dialing system for small company |
| 16:36.09 | [TK]D-Fender | Anonissimus: I found a solution to the "what am i going to sue for a PBX?" in Asterisk... |
| 16:36.22 | [TK]D-Fender | use* |
| 16:36.36 | gaetronik | http://rafb.net/p/m44TbT43.html |
| 16:36.36 | microw | hi all, is sipx better than asterisk on performance?? |
| 16:37.22 | gaetronik | DarKnesS_WolF, |
| 16:37.25 | Anonissimus | D-Fender: ?? |
| 16:37.47 | nny_1 | aastra is perplexed |
| 16:37.56 | nny_1 | this issue is apparently very common |
| 16:37.58 | Anonissimus | I'm looking for someone who wants to discuss an autodialing system |
| 16:38.02 | lowtek | nny_1: What's the issue? |
| 16:38.03 | nny_1 | or rather this issue IS the norm |
| 16:38.37 | *** part/#asterisk lzhang (n=lzhang@rrcs-67-78-33-170.sw.biz.rr.com) |
| 16:39.04 | nny_1 | lowtek one sec now it is working |
| 16:42.08 | microw | Anonissimus: what kind of autodialing, what's your scenerio? |
| 16:42.33 | Anonissimus | one that dials numbers from a database |
| 16:42.52 | microw | what's the problem? |
| 16:42.55 | [TK]D-Fender | Anonissimus: Easy enough so far |
| 16:42.56 | gaetronik | Anonissimus, script+auto dialout |
| 16:43.20 | gaetronik | Anonissimus, http://www.voip-info.org/wiki/view/Asterisk+auto-dial+out+deliver+message |
| 16:43.27 | Anonissimus | as in: now we have people calling potential customers |
| 16:43.38 | nny_1 | lowtek er nm. Phone is not registering properly (480i ct) |
| 16:43.38 | Anonissimus | but they are nto working really hard without supervision |
| 16:44.05 | Anonissimus | so now I'm trying to find some kind of system that dials number after number from the directory |
| 16:44.24 | nny_1 | lowtek have aastra ont he phone, sending them syslog dumps from the phone but so far they have no info |
| 16:44.37 | nny_1 | lowtek if i call it asterisk says ringing 100 but the phone doesn't actually ring |
| 16:44.40 | Anonissimus | so that callgirls only have to take up the phone do the talking and hang up |
| 16:44.44 | microw | Anonissimus: junk call :-) |
| 16:44.51 | DarKnesS_WolF | gaetronik: why u don't try mixmonitor ? |
| 16:44.58 | DarKnesS_WolF | gaetronik: http://www.voip-info.org/wiki/view/MixMonitor |
| 16:45.44 | Anonissimus | noooo :) marketing |
| 16:46.30 | gaetronik | DarKnesS_WolF, because i used to use Monitor |
| 16:48.00 | gaetronik | but in what format are stored |
| 16:48.05 | gaetronik | with Mixmonitor |
| 16:49.39 | gaetronik | DarKnesS_WolF, ? |
| 16:49.46 | *** join/#asterisk hfb (n=hfb@pool-96-247-108-198.lsanca.dsl-w.verizon.net) |
| 16:51.48 | *** part/#asterisk korihor (n=korihor@190.78.32.60) |
| 16:52.48 | DarKnesS_WolF | gaetronik: wav |
| 16:53.00 | DarKnesS_WolF | core show application mixmonitor\ |
| 16:53.22 | *** join/#asterisk adrianXXX (i=xpl@84.126.207.62.dyn.user.ono.com) |
| 16:53.28 | gaetronik | DarKnesS_WolF, ok thanks |
| 16:54.34 | gaetronik | there is no way to write in WAV |
| 16:54.54 | gaetronik | to avoid useless disk use |
| 16:55.47 | *** join/#asterisk paige (n=Paige@208.89.241.31) |
| 16:56.19 | DarKnesS_WolF | gaetronik: ? |
| 16:56.27 | *** join/#asterisk mort_gib (n=mjensen@16.Red-83-36-63.staticIP.rima-tde.net) |
| 16:56.33 | DarKnesS_WolF | gaetronik: u can convert to mp3 with lame if u like |
| 16:56.51 | gaetronik | DarKnesS_WolF, in monitor the codec depend on the extension |
| 16:56.56 | gaetronik | is this the case in mixmonitor |
| 16:57.05 | DarKnesS_WolF | mmm ic |
| 16:57.10 | paige | do moh files have to be mp3 or can they be wav? |
| 16:57.32 | DarKnesS_WolF | not sure i did the recording once using mixmonitor and the client is happy and everything working perfect |
| 16:57.37 | DarKnesS_WolF | paige: wav / mp3 works |
| 16:58.12 | paige | mp3 adds an overhead on the switch from what i have read |
| 16:58.25 | gaetronik | DarKnesS_WolF, i want to avoid 1Go daily |
| 16:58.25 | DarKnesS_WolF | paige: true |
| 16:58.33 | [TK]D-Fender | paige: Any format * can read. |
| 16:58.34 | DarKnesS_WolF | gaetronik: convert to mp3 :-s |
| 16:58.44 | [TK]D-Fender | paige: Prefferably the format of the channel listening to it. |
| 16:58.46 | DarKnesS_WolF | [TK]D-Fender: mixmonitor can't write mp3 directly right ? |
| 16:59.05 | [TK]D-Fender | DarKnesS_WolF: No. |
| 16:59.52 | paige | ok, so I can plunk a ton of wavs into /usr/local/asterisk/moh and they will work then? |
| 17:00.51 | DarKnesS_WolF | i used to have STAIND as mp3 MOH :P |
| 17:00.58 | DarKnesS_WolF | it was like press 1 for coldplay press 2 for staind :-D |
| 17:01.09 | DarKnesS_WolF | ahhh damn old playing days :'( |
| 17:02.56 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 17:03.12 | *** join/#asterisk qdk (n=qdk@0x573fe57e.bynqu2.static.dsl.tele.dk) |
| 17:03.25 | jeev | is 7 digit dialing not recommended? |
| 17:03.38 | paige | is there a prefered bitrate for them? ie, 8k, 16k, 32k? |
| 17:04.24 | microw | <PROTECTED> |
| 17:04.41 | gaetronik | paige, 8k |
| 17:05.03 | paige | kthnx |
| 17:06.41 | scooby2 | Anyone know how you would block caller id when calling 800 numbers? We keep getting harassing calls from someone to our 800#'s and it is showing up with no callerid. |
| 17:07.19 | paige | k thank worked thanks |
| 17:10.48 | *** join/#asterisk juanjoc (n=juanjoc@host84.190-225-170.telecom.net.ar) |
| 17:11.12 | *** join/#asterisk DrAk0 (n=ljd@nelug/coreteam/luisjose) |
| 17:13.30 | nny_1 | wow wtf |
| 17:13.39 | nny_1 | so we put a new phone, and the new phone doesn't ring |
| 17:13.58 | nny_1 | apply changes to old phone and it does, so diagnostic with the new phone fails cause apparently it is broken |
| 17:14.02 | nny_1 | aastra :( |
| 17:14.04 | nny_1 | no mas |
| 17:14.15 | nny_1 | ~aastra |
| 17:14.18 | errr | nny_1: which aastra tech are you talking to? |
| 17:14.19 | Dudds | Don't you hate hardware failures? |
| 17:14.38 | nny_1 | errr jberendt |
| 17:15.00 | [TK]D-Fender | jeev: All of my systems are 7-10-11 digit transparent to the user |
| 17:15.14 | errr | nny_1: hmm, never worked with that one, I normally get Layne Monson, he is really really good |
| 17:15.23 | gaetronik | jeev, [TK]D-Fender what does 7 digit dialing mean |
| 17:15.33 | nny_1 | yesterday the old phone wouldn't register. Still don't understand why. I cleaned up the config files, removed the default XML entires that were bad (yay default config fail!) and removed the comments |
| 17:15.41 | [TK]D-Fender | gaetronik: Means dialing a 7 digit number |
| 17:15.45 | jeev | thanks fenderino |
| 17:15.58 | jeev | i guess i'll set up a dialmap on the phone. |
| 17:16.02 | gaetronik | for a non us citizen what mean? |
| 17:16.18 | jeev | dood, in the US, we call xxx-xxxx for a local number |
| 17:16.22 | jeev | as in, someone in my same area code |
| 17:16.22 | nny_1 | added a specific entry for line 1 registrar and proxy, which by default only exists globally |
| 17:16.26 | gaetronik | not using prefix so |
| 17:16.44 | errr | jeev: depends where you live, some of us have to do 10 digit dilaing for local numbers |
| 17:16.49 | jeev | oh, ok |
| 17:16.59 | gaetronik | like here |
| 17:17.05 | jeev | i guess that sucks ofor you |
| 17:17.06 | jeev | for you |
| 17:17.24 | *** part/#asterisk Zeeek (n=Zeeek@pdpc/supporter/active/Zeeek) |
| 17:17.25 | errr | jeev: it will be that way for everyone soon I bet |
| 17:17.25 | gaetronik | for that i miss frnace where all num have the same size |
| 17:30.33 | nny_1 | ok wth |
| 17:31.14 | *** join/#asterisk gr0mit (n=tim@81.187.32.146) |
| 17:31.32 | nny_1 | so any calls coming in on zap says Ringing 100, but 100 never rings, and goes to voicemail |
| 17:32.04 | nny_1 | if I call in on a SIP channel, it rings the phone and it works |
| 17:32.17 | nny_1 | i.e. If i call in from vitelity |
| 17:33.18 | nny_1 | both land in the same context (default) |
| 17:33.36 | nny_1 | so I ask you, good people.. wtf? |
| 17:34.03 | *** join/#asterisk rafael-ec (n=rafael@190.9.164.86) |
| 17:34.30 | *** join/#asterisk oilinki3 (n=oil@ppp-124-120-10-143.revip2.asianet.co.th) |
| 17:34.34 | nny_1 | and if i call into zap and forward out to a cell phone on SIP it works |
| 17:34.47 | jblack | look at your the macro you made to send people to voicemail after calling. You probably aren't actually calling. |
| 17:34.52 | jblack | You can debug with asterisk -r |
| 17:35.21 | nny_1 | so the only time the phone won't ring is when i call into ZAP and asterisk connects to SIP/100. If I connect a 2nd client via Xlite it works |
| 17:35.30 | nny_1 | jblack oh yeah i am a lot deeper than that |
| 17:35.39 | nny_1 | jblack voicemail is showing up in console |
| 17:35.41 | *** join/#asterisk tobias (n=tobias@user-0c2hj2e.cable.mindspring.com) |
| 17:35.54 | jblack | verbose 9, debug 9, and walk the dialplan. |
| 17:35.58 | jblack | Make sure you're in the context you think, etc. |
| 17:36.04 | nny_1 | i am |
| 17:36.20 | jblack | then the error is staring you in the face. You just need to look back |
| 17:39.37 | nny_1 | jblack maybe i am overlooking it http://pastebin.com/m7a885c |
| 17:40.15 | jblack | no doubt someone will look that over |
| 17:40.18 | *** join/#asterisk keith4 (n=keith@lust.CC.Lehigh.EDU) |
| 17:40.44 | jeev | i can't believe this company i've waited 4 days for.. they told me they dont do vlan's for colo. |
| 17:40.49 | jeev | just set up on netmask .128 and that's it. |
| 17:40.50 | jeev | wtf? |
| 17:41.30 | microw | <PROTECTED> |
| 17:42.24 | nny_1 | can someone look this over and tell me why SIP/100 doesn't ring and goes to VM if the call comes in on ZAP? |
| 17:42.25 | nny_1 | http://pastebin.com/m7a885c |
| 17:42.57 | jblack | did you do a seperate sip debug yet? |
| 17:44.23 | jblack | I want different debug. |
| 17:44.31 | jblack | I want you to run asterisk -r. First, turn debug to 0, and verbose to 9, and try a call. |
| 17:44.50 | jblack | then, I want you to turn debug to 9, and verbose to 0, and try a call. Then I want you to turn both of those off, and turn on sip debug. |
| 17:45.20 | jblack | I want what's logged to the console, not to a file. |
| 17:45.57 | jblack | but offhand, it looks to me like zap is broken. |
| 17:46.07 | nny_1 | jblack http://pastebin.com/m688557f9 |
| 17:47.05 | jblack | Looks like sip is working, though you're not using auth? |
| 17:47.21 | jblack | # |
| 17:47.22 | jblack | [Aug 22 13:45:31] WARNING[12352]: chan_sip.c:1950 retrans_pkt: Maximum retries exceeded on transmission 27fb65393358f8244e89eb7049284349@192.168.100.5 for seqno 102 (Non-critical Request) |
| 17:47.43 | nny_1 | i am using auth |
| 17:47.43 | *** join/#asterisk |||Mad||| (n=mad@mail.rubbusa.com) |
| 17:47.51 | [TK]D-Fender | nnY_2: PB "route -an" |
| 17:48.17 | [TK]D-Fender | nnY_2: and "iptables --list" |
| 17:48.18 | nny_1 | route: invalid option -- a |
| 17:48.37 | |||Mad||| | Hello, all! Quick question: is it easy to get the Asterisk voicemail system to say the name rather than spell it? |
| 17:48.43 | jblack | is there nat anywhere involved in this? Anywhere within 100 miles? |
| 17:48.53 | nny_1 | no |
| 17:49.03 | *** join/#asterisk dr_gogeta86 (n=gogeta@ppp-151-251.32-151.iol.it) |
| 17:49.15 | jblack | |||Mad|||: Nobody has written support for that. |
| 17:49.34 | *** join/#asterisk berspolis (n=berspoli@190.25.228.235) |
| 17:49.43 | berspolis | hi |
| 17:49.45 | jblack | what is the ip of the phone, and the ip of the asterisk server? |
| 17:49.48 | nny_1 | wait |
| 17:49.49 | |||Mad||| | OK, thanks jblack... |
| 17:50.03 | nny_1 | there is nat etc involved with vitelity SIP, but thats not broken here |
| 17:50.05 | nny_1 | http://pastebin.com/m282479ee |
| 17:50.16 | nny_1 | well thats not the current issue per se |
| 17:50.34 | nny_1 | ZAP --> SIP/100 is |
| 17:50.44 | *** join/#asterisk nicox (n=nicox@213-33-7-92.adsl.highway.telekom.at) |
| 17:50.53 | jblack | Ok. are the phone and * both in 192.168.100.0/24 ? |
| 17:50.56 | nny_1 | yes |
| 17:51.01 | nny_1 | on a simple switch |
| 17:51.04 | jblack | k |
| 17:51.05 | nny_1 | no layer 3 etc |
| 17:51.42 | nny_1 | jblack correct that there is no auth for vitelity inbound or outbound SIP |
| 17:51.49 | berspolis | anyone can tell me why i can make calls from an iax server (let´s call him 1) to another but fails in the other way? |
| 17:51.49 | jblack | when you call in zap, you get the ivr, etc? It just breaks when trying to reach sip/100 ? |
| 17:51.54 | nny_1 | but there is auth for phones locally |
| 17:51.58 | nny_1 | jblack yes |
| 17:52.09 | [TK]D-Fender | |||Mad|||: Yes, * can say the name |
| 17:52.19 | nny_1 | jblack and i can (and am) forwarding calls this way to 8435551212@vitel-outbound |
| 17:52.21 | jblack | [TK]D-Fender: I think he wants the _caller_ to say the name. |
| 17:52.31 | jblack | Like "I want to leave tk a voicemail", but just the tk part |
| 17:52.32 | nny_1 | basically ZAP --> VITELITY SIP |
| 17:52.42 | [TK]D-Fender | jblack: Not from what he wrote. |
| 17:52.49 | nny_1 | works fine |
| 17:53.03 | nny_1 | and VITELITY SIP --> SIP/100 works |
| 17:53.07 | jblack | berspolis: Can you restate waht you're asking for? |
| 17:53.22 | jblack | nny_1: You're confusing me. |
| 17:53.38 | nny_1 | jblack the client has 1 zap channel and vitelity itsp service |
| 17:53.40 | jblack | You've got a provider coming in via zap, i.e. a PRI or somesuch. |
| 17:53.53 | jblack | You have an ip phone that uses sip. |
| 17:53.57 | nny_1 | yes |
| 17:53.59 | jblack | Are both of these true? |
| 17:54.10 | nny_1 | correct |
| 17:54.12 | jblack | then why are you talking about "vitelity sip" ? |
| 17:54.15 | nny_1 | and they also have an ITSP |
| 17:54.17 | nny_1 | sorry |
| 17:54.43 | nny_1 | they have a 1800# that comes in (non authed) over SIP from outside the network |
| 17:54.52 | nny_1 | the company name is Vitelity |
| 17:54.57 | jblack | Gotcha. |
| 17:55.06 | nny_1 | and if I call that number, the phone rings >< |
| 17:55.15 | jblack | (P , P ) -> * -> (IP Phone) ? |
| 17:55.21 | berspolis | ok, i have two asterisk servers that are connected by internet, i have configured the iax.conf as friends and i have configured the dial plan perfectly, the calls from asterisk1 to asterisk2 work fine, but fail in the other way |
| 17:55.26 | nny_1 | jblack yes |
| 17:55.26 | berspolis | this is the error |
| 17:55.29 | berspolis | Unable to create channel of type 'IAX2' (cause 3 - No route to destination) |
| 17:56.10 | nny_1 | P , P = (ZAP/2 (local pots) , (SIP (Vitelilty) -> * -> (IPPhone) |
| 17:56.20 | jblack | Ok, so we define PZ as the zap provider, and PS as the sip provider, and the IP phone as local. |
| 17:56.28 | nny_1 | ok |
| 17:56.53 | nny_1 | so (PZ , PS) -> * -> (IPPHONE) |
| 17:56.54 | jblack | Calls from PS->everything are fine. PZ->IVR are fine. PZ->local are broken. Correct? |
| 17:57.07 | nny_1 | jblack correct |
| 17:57.35 | jblack | can you reverify PS->IPPHONE works? |
| 17:57.38 | lowtek | Hey guys what's the difference between duration and billsec in the cdr? Does billsec start when the call is ANSWERED()? |
| 17:57.41 | nny_1 | jblack yes |
| 17:57.51 | jblack | thanks. I'll go smoke while you do that |
| 17:57.58 | nny_1 | jblack heh ok |
| 17:58.21 | |||Mad||| | [TK]D-Fender: yes, I would like to have the directory say the person's name rather than spell it out. It can either use the recorded name from voicemail or say it phonetically |
| 17:58.41 | [TK]D-Fender | |||Mad|||: It will say the name. |
| 17:59.15 | |||Mad||| | Oh! Can you tell me how to make it say it instead or spell it? Or where I can read up on how to do it? |
| 17:59.17 | nny_1 | jblack when you get back, FYI, i have tested both FXO ports on the card in * |
| 17:59.22 | nny_1 | tried/tested |
| 17:59.30 | |||Mad||| | I didn't see it in the books |
| 17:59.50 | [TK]D-Fender | |||Mad|||: Go record it. |
| 18:00.05 | [TK]D-Fender | |||Mad|||: it isn't in the book because it isn't an OPTION, its a FACT |
| 18:00.09 | *** part/#asterisk rafael-ec (n=rafael@190.9.164.86) |
| 18:00.56 | |||Mad||| | Ahh, so it only spells it if the name hasn't been recorded? Funny, I thought I had recorded the name... |
| 18:01.26 | jblack | nny_1: That doesn't make any sense to me. |
| 18:01.45 | [TK]D-Fender | |||Mad|||: Less thought, more look. |
| 18:02.13 | jblack | can you paste your pertinant config files, with passwords changed to XXX? I'll want at least extensions.conf, zapata.conf, zaptel.conf, sip.conf |
| 18:02.15 | nny_1 | jblack I have 2 FXo ports on the TDM card. I moved the phone line over to the 2nd one, and edited zapata.conf to channel 2, to test to see if the module wasn't an issue |
| 18:02.21 | nny_1 | jblack yes |
| 18:02.48 | jblack | make sure the config files you paste are the ones you're testing? (i.e. verify you did a reload before you observed the problem) |
| 18:04.11 | nny_1 | jblack comedy: If i set up xlite as the SIP client 100, it rings :X |
| 18:04.21 | nny_1 | i tested this last night and can test again |
| 18:04.43 | nny_1 | this is through NAT, from my office to the site FWIW |
| 18:05.07 | jblack | we're getting close to "your problem is bad karma" |
| 18:05.26 | nny_1 | jblack ha i suspect ghosts |
| 18:05.56 | |||Mad||| | [TK]D-Fender: I just re-recorded my name in voicemail and the directory still spells it. I must be missing something |
| 18:07.22 | daniev | excuse my ignorance, but asterisk works behind NAT? i ask this question, because once a dCap told me that is not possible get work asterisk behind NAT |
| 18:07.46 | jblack | daniev: It's a pain in the ass. YOu have to forward the ports that are used through nat. |
| 18:07.53 | seanbright | ~sipnat |
| 18:07.54 | jbot | [~sipnat] Quick guide on configuring * + SIP behind NAT : http://www.aocomputing.net/?p=3 , otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
| 18:08.06 | daniev | mmm i see jblack |
| 18:08.11 | daniev | too many ports ??? |
| 18:08.23 | nny_1 | jblack http://pastebin.com/m273461af |
| 18:08.26 | berspolis | exit |
| 18:08.34 | jblack | daniev: iax is simpler to punch through (it's a single tcp pport), but sip is doable too (you're looking at punching through a single tcp port, and anywhere from 50 - 1000 udp ports) |
| 18:08.59 | *** join/#asterisk Katty (n=asterisk@hera.copi-rite.com) |
| 18:09.02 | Katty | hai! |
| 18:09.06 | daniev | jblack: thank you |
| 18:09.10 | [TK]D-Fender | |||Mad|||: Yes, you're missing a pastebin. |
| 18:09.10 | jblack | nny_1: Thanks. Next time, put "=======================" between conf files, to make them obvious? |
| 18:09.17 | [TK]D-Fender | Katty: Mew. |
| 18:09.20 | nny_1 | sorry I can do that now |
| 18:09.25 | jblack | if you don't mind |
| 18:09.25 | florz | jblack: TCP? is that IAX3? |
| 18:09.28 | Katty | [TK]D-Fender: mew. |
| 18:09.44 | seanbright | florz: you are kidding i hope |
| 18:09.49 | Katty | [TK]D-Fender: does anything seem unusual about this line: exten => _XXXXXXXXXXX,1,Set(CALLERID(num)=5733344439) |
| 18:09.50 | jblack | florz: It's the TCP in "TCP/IP" |
| 18:10.05 | Katty | [TK]D-Fender: anything about the syntax that doesn't settle well |
| 18:10.18 | florz | seanbright: erm, what exactly? |
| 18:10.24 | nny_1 | jblack http://pastebin.com/m18d0cc75 |
| 18:10.27 | |||Mad||| | [TK]D-Fender: pastebin of what? |
| 18:10.28 | jblack | thanks |
| 18:10.30 | florz | jblack: well, but what IAX are you talking about? |
| 18:10.37 | seanbright | florz: <florz> jblack: TCP? is that IAX3? |
| 18:10.38 | Kobaz | select count(*) from dnc.masterlist; 169048892 |
| 18:10.41 | Kobaz | that's alot of numbers |
| 18:10.49 | seanbright | Katty: no |
| 18:10.51 | [TK]D-Fender | Katty: That you have an 11-digit match with no 1 in front, or N's where they are normal for North America, yes... I'd say that's "irregular" |
| 18:11.11 | jblack | florz: IAX is the "Inter Asterisk eXchange protocol". It's an optional protocol for asterisk servers to route calls between one another. Think of it as a competitor to sip. |
| 18:11.31 | jblack | florz: Hold the rest of the questions for me for later please. I'm doing something difficult |
| 18:11.32 | nny_1 | jblack the extra config file (ext_mainmenu.conf) can be overwritten by a IVR style one with a button press and a script locally. It is not used right now |
| 18:11.32 | Katty | [TK]D-Fender: so what would you suggest it needs to look like? |
| 18:11.34 | daniev | jblack: for using IAX i need IAX supported phones i guess |
| 18:11.36 | [TK]D-Fender | |||Mad|||: Folder dumps, call attempt, voicemail config |
| 18:11.43 | Katty | [TK]D-Fender: exten => _NXXXXXXXXXX? |
| 18:11.45 | florz | jblack: well, yeah - and now, where does that TCP thingy come in? |
| 18:12.01 | jblack | nny_1: I can assume you're not hiding a trixbox or something else from me, right? |
| 18:12.08 | Katty | [TK]D-Fender: or perhaps exten => _1XXXXXXXXXX |
| 18:12.09 | [TK]D-Fender | Katty: for 11-digit standard : exten => _1NXXNXXXXXX,1,Set(CALLERID(num)=5733344439) |
| 18:12.16 | nny_1 | jblack hell no :) |
| 18:12.25 | |||Mad||| | OK, lemmee get what I think is relevant... |
| 18:12.27 | Katty | hmm. okay |
| 18:12.31 | jblack | EDESYnC |
| 18:12.31 | Kobaz | nny_1: you gotta say it right... it's "hellz no foo" |
| 18:12.36 | nny_1 | haha |
| 18:12.44 | [TK]D-Fender | |||Mad|||: Unless its MORE than I asked, don't |
| 18:12.47 | nny_1 | hellls no foo! I pitty the foo who uses trixbox |
| 18:12.58 | jblack | Are you hiding trixbox, callweaver, anything that's more/less/different from straight asterisk? |
| 18:12.59 | [TK]D-Fender | I have NO pity for trixbox users. |
| 18:13.09 | nny_1 | jblack none |
| 18:13.20 | jblack | Ok. So nothing that could muck up my basic assumptions? |
| 18:13.22 | Katty | [TK]D-Fender: and what would a standard 7 digit number look like |
| 18:13.29 | nny_1 | jblack asterisk 1.4.21.2, zaptel 1.4.11 asterisk addons on a CENTOS based box |
| 18:13.30 | Katty | [TK]D-Fender: exten => _Nxxxxxx |
| 18:13.36 | jblack | k |
| 18:13.37 | florz | seanbright: well, not really ... I'm just wondering where jblack has seen some IAX that's working over TCP, must be a new version I assume ... |
| 18:13.38 | nny_1 | jblack none |
| 18:13.41 | [TK]D-Fender | Katty: Yes |
| 18:13.43 | Katty | [TK]D-Fender: kk |
| 18:13.48 | nny_1 | jblack no SEL etc either |
| 18:13.57 | seanbright | florz: ohhhh, so you're just being a know-it-all. gotcha. |
| 18:14.05 | [TK]D-Fender | nnY_2: y-a-tu du poivre? |
| 18:14.06 | seanbright | florz: could have saved us all the trouble of thinking you an idiot. |
| 18:14.25 | jblack | Line 40 is a problem, but probably not your problem. |
| 18:14.50 | jblack | It's redundant. 39 will cover it |
| 18:14.55 | nny_1 | jblack ok I see what you mean |
| 18:14.58 | *** join/#asterisk AdamWest (n=Leif@CPE001d7e2f9574-CM0012c9db3d2e.cpe.net.cable.rogers.com) |
| 18:15.11 | nny_1 | jblack it means _(one digit) which is pointless |
| 18:15.19 | jblack | ok, so zap drops into default.... |
| 18:15.35 | jblack | to mainmenu,s,1 |
| 18:15.40 | jblack | which includes ext_main... |
| 18:15.44 | jblack | Hold up. call |
| 18:16.00 | nny_1 | jblack np |
| 18:16.00 | florz | seanbright: could even have done that yourself, I assume - given the fact that basic understanding of VoIP suggests that VoIP over TCP makes very little sense, indeed, it would be pretty obvious that I am referring to the pretty small probability of there being some "IAX over TCP" ... |
| 18:16.29 | seanbright | florz: huh? |
| 18:17.10 | Kobaz | seanbright: florz is correct. tcp does not make sense for voip |
| 18:17.21 | seanbright | right? and? |
| 18:17.28 | seanbright | not sure what any of this has to do with me? |
| 18:17.32 | Kobaz | dunno |
| 18:18.18 | *** join/#asterisk stencil (n=stencil@d193-237-37.home3.cgocable.net) |
| 18:19.32 | seanbright | florz: the statement in question: "<florz> jblack: TCP? is that IAX3?" |
| 18:19.46 | seanbright | in my mind, that read as "jblack... is TCP the same thing as IAX3?" |
| 18:20.05 | *** join/#asterisk uski (n=uski@bre01-1-88-162-0-210.fbx.proxad.net) |
| 18:20.17 | florz | seanbright: well, given the context that jblack claimed that you needed to port-forward a tcp-port in order for iax to work through NAT ... |
| 18:20.21 | [TK]D-Fender | daniev>excuse my ignorance, but asterisk works behind NAT? i ask this question, because once a dCap told me that is not possible get work asterisk behind NAT <- BS. Rarely any real trouble. |
| 18:21.00 | seanbright | jblack: florz corrected you |
| 18:21.01 | Kobaz | ipv6 will solve everything :P |
| 18:21.07 | seanbright | florz: you corrected jblack |
| 18:21.09 | seanbright | there |
| 18:21.13 | seanbright | we're all caught up |
| 18:21.45 | [TK]D-Fender | Kobaz: I'll believe that when I get my robot maid and flying car I was promised decades ago... |
| 18:21.55 | Kobaz | you didn't get that yet? |
| 18:22.00 | Kobaz | damn... |
| 18:22.03 | |||Mad||| | [TK]D-Fender: You want more than you asked for?!? Right now I have voicemail.conf, extensions.conf & a sample call into the directory |
| 18:22.19 | [TK]D-Fender | |||Mad|||: Don't even think of giving me LESS than I asked for. |
| 18:23.16 | |||Mad||| | OK, looks like the only thing I'm missing is a folder dump - which one? The mailbox greetings? |
| 18:23.48 | *** part/#asterisk Nivex (n=kjotte@user-0c8hvoj.cable.mindspring.com) |
| 18:24.41 | [TK]D-Fender | |||Mad|||: box & everything under it |
| 18:24.52 | |||Mad||| | OK |
| 18:25.30 | Katty | [TK]D-Fender: is the callerid supposed to be 10 digits or 11 digits? |
| 18:25.38 | Katty | [TK]D-Fender: i presume no 1? |
| 18:25.43 | [TK]D-Fender | Katty: 10 normally. |
| 18:25.46 | Katty | [TK]D-Fender: kk |
| 18:26.21 | |||Mad||| | Odd, all my recorded greetings are gone, and greet.gsm is in the mailbox root, not the greet folder |
| 18:26.41 | florz | Katty: without any further restrictions, the callerid is supposed to be of variable length |
| 18:28.07 | [TK]D-Fender | |||Mad|||: and after 15 minutes I still don't have a pastebin. |
| 18:28.59 | |||Mad||| | [TK]D-Fender: I'm a-working on it :) Wanna make sure I have no lass than what you asked for as to not annoy you further |
| 18:29.12 | |||Mad||| | http://pastebin.com/d357b5800 |
| 18:29.24 | |||Mad||| | Hopefully that is adequate |
| 18:31.36 | [TK]D-Fender | |||Mad|||: -- Executing [5@voicemenu-custom-1:1] Directory("Zap/3-1", "employees") in new stack || admin@rubb4 ~]$ cd /var/spool/asterisk/voicemail/default/103 |
| 18:31.46 | [TK]D-Fender | |||Mad|||: uh huh... go fix it... |
| 18:32.16 | [TK]D-Fender | |||Mad|||: "employees" != "default" |
| 18:32.25 | |||Mad||| | Ahhh |
| 18:32.26 | codefreeze-lap | lowtek: did anyone answer that CDR question? CDR's record 3 times: start, answer, and end. When end time is set, the billsec is also set to end-answer; and duration is set to end-start |
| 18:32.51 | |||Mad||| | [TK]D-Fender: Stupid mistake |
| 18:33.05 | [TK]D-Fender | NEXT!!!@!@ONEATATONEATONEONE!!! |
| 18:33.27 | nnY_2 | heh |
| 18:33.35 | [TK]D-Fender | (c) BKW |
| 18:33.53 | nnY_2 | I seriously think my issue lies in the phone still |
| 18:34.05 | nnY_2 | i mean shit, x lite works if i swap it out for the phone |
| 18:37.07 | *** join/#asterisk mvanbaak_ (n=michiel@asterisk/contributor-and-bug-marshal/mvanbaak) |
| 18:37.53 | jblack | nny_1: Ok, I'm back. |
| 18:38.01 | jblack | nny_1: but I'm going to smoke first. |
| 18:38.19 | seanbright | jblack: you smoke a lot |
| 18:39.34 | *** mode/#asterisk [+o lmadsen] by ChanServ |
| 18:39.50 | seanbright | i always think of family guy when i see AdamWest |
| 18:40.32 | |||Mad||| | [TK]D-Fender: Thank you very much, that seems to have fixed things up nicely!! I should have realized each context had a separate folder. |
| 18:40.39 | brodiem | oh my god, I'm a tomato! |
| 18:40.43 | gaetronik | nnY_2, you mean that if you configure a x lite like the phone and pull the phone out it works? |
| 18:40.57 | gaetronik | and nnY_2 are you nny_1 |
| 18:40.58 | gaetronik | ? |
| 18:41.05 | [TK]D-Fender | |||Mad|||: You're welcome. |
| 18:42.34 | nnY_2 | gaetronik yes |
| 18:42.48 | nnY_2 | gaetronik must have left self logged in at work |
| 18:43.03 | gaetronik | do you have a auto transfer on the phone |
| 18:43.10 | gaetronik | last time i had an issue like yours |
| 18:43.21 | nnY_2 | gaetronik no auto transfer |
| 18:43.25 | gaetronik | sure? |
| 18:43.40 | nnY_2 | gaetronik yeah yeah, set to factory defaults and loaded config |
| 18:43.42 | gaetronik | not a fucking thing with a *73776765 |
| 18:43.46 | jblack | nny_1: You're still screwed, right? |
| 18:43.47 | nnY_2 | gaetronik also have tested two phones |
| 18:44.00 | nnY_2 | jblack yup like a one legged man in an ass kicking contest |
| 18:44.15 | jblack | Ok. When you're done with gaetronik, I can go back to working with you on it exlusively |
| 18:44.27 | gaetronik | jblack, i ve a lot of work |
| 18:44.31 | gaetronik | so help him |
| 18:44.39 | jblack | it'll take a while. |
| 18:44.56 | nnY_2 | jblack get this, if I swap either Aastra for an xlite client here it works from Zap |
| 18:45.06 | gaetronik | the bad kama option is the most probable |
| 18:45.09 | nnY_2 | jblack so (PZ) -> * -> Xlite works |
| 18:45.17 | nnY_2 | meh aastra is teh devil |
| 18:45.26 | jblack | nny2: Oh, line 45. |
| 18:45.29 | nnY_2 | i also have syslog info from aastra phone |
| 18:45.43 | jblack | Is that a #include, instead of an include => ? |
| 18:45.47 | nnY_2 | yes |
| 18:46.03 | jblack | That looks like a C include, not a extensions include |
| 18:46.15 | nnY_2 | it's including an outside file, not a context locally |
| 18:46.49 | Qwell | #include <stdio.conf> |
| 18:47.05 | gaetronik | while(1){fork()} |
| 18:47.20 | Qwell | :(){ :|:& };: |
| 18:47.22 | Qwell | less chars |
| 18:47.28 | gaetronik | yes |
| 18:47.31 | nnY_2 | if you scroll down there is a System(autoattendant-enable) etc call that swaps ext_mainmenu.conf with a more complex one |
| 18:47.40 | jblack | I'd replace all of transfer with a exten => _1XX,1,Macro .... |
| 18:47.49 | jblack | And then add a Noop in there to make sure you're getting there. |
| 18:48.16 | nnY_2 | jblack so you think the DP could have an issue even if Xlite works but Aastra doesn't? |
| 18:48.18 | *** join/#asterisk vonkleist (n=gcontrer@201.116.65.115) |
| 18:48.58 | brodiem | I thought your problem was your aastra not registering in the first place |
| 18:49.02 | jblack | When things are broken, but everything is "right" the proper path is to call facts into question. |
| 18:49.21 | nnY_2 | brodiem itis registering now |
| 18:49.32 | nnY_2 | jblack understandable |
| 18:49.34 | jblack | Zap isn't broken, the DP isn't broken, the phone isn't broken, but it doesn't work. |
| 18:49.36 | brodiem | ah ok |
| 18:49.52 | nnY_2 | jblack i still suspect the phone a bit, but yeah |
| 18:49.53 | jblack | something is lying. We need to know what. |
| 18:49.56 | nnY_2 | jblack roger |
| 18:50.01 | nnY_2 | jblack ok changing conext |
| 18:50.14 | nnY_2 | transfer to _1XX,1,Macro and Noop |
| 18:50.28 | jblack | I'd do ...,1,NoOp, and ...,2,Macro |
| 18:50.40 | jblack | or n, or whatnot |
| 18:51.05 | nnY_2 | jblack honestly this is the only phone on that network btw |
| 18:51.19 | nnY_2 | jblack the other entries are part of our standard deployed configs |
| 18:51.33 | jblack | Ok. |
| 18:52.14 | nnY_2 | jblack so exten+ _1XX,1,Macro(stdexten,${EXTEN},SIP/${EXTEN}) ? |
| 18:52.21 | nnY_2 | er exten => |
| 18:52.31 | jblack | perfect |
| 18:53.40 | jblack | pulls out his pipe and magnifying glass |
| 18:53.56 | *** join/#asterisk aliver (n=aliver@ip-216-17-149-97.rev.frii.com) |
| 18:54.42 | jblack | waits patiently as nny checks to verify the noop is running |
| 18:55.27 | aliver | When I try to transfer a call to voice mail to an extension which doesn't have it's SIP phone online I get this error: |
| 18:55.29 | aliver | Aug 22 12:53:17 NOTICE[700]: app_dial.c:1076 dial_exec_full: Unable to create channel of type 'SIP' (cause 3 - No route to destination) |
| 18:55.36 | aliver | Is there a way to avoid that? |
| 18:56.04 | jblack | aliver: other than keeping the phone on, no. |
| 18:56.16 | [TK]D-Fender | aliver: yes. Stop looking at CLI |
| 18:56.16 | jblack | But you're welcome to try other extensions after the first one fails, or send to voice mail, etc etc |
| 18:56.41 | aliver | [TK]D-Fender stop looking at the CLI? |
| 18:56.43 | aliver | why? |
| 18:56.56 | jblack | aliver: The 'error' is appropriate. |
| 18:56.57 | [TK]D-Fender | aliver: So you won't have to see that harmlesss warning |
| 18:57.08 | [TK]D-Fender | hands aliver a spoon |
| 18:57.10 | aliver | jblack I gotcha. |
| 18:57.29 | seanbright | it's not an error... it's a "NOTICE" |
| 18:57.38 | nnY_2 | jblack whats the best way to incorporate a noop in this again? |
| 18:57.39 | seanbright | were it an error it would say "ERROR" |
| 18:57.49 | aliver | So, I have to have the dialplan be smart enough to avoid them just getting dumped. |
| 18:57.58 | jblack | nnY_2: just move the macro from 1 to 2, and add a noop in 1. |
| 18:57.58 | aliver | heh, tough crowd. |
| 18:58.13 | seanbright | aliver: they shouldn't be getting dumped, they should be going to the user's voicemail |
| 18:58.22 | seanbright | aliver: assuming you have your dialplan set up correctly |
| 18:58.31 | [TK]D-Fender | aliver: What do you mean "dumped"? Just because a dial fails doesn't mean you can't continue to do other things. |
| 18:58.37 | jblack | exten => s,1,NoOp(I am in the macro: ${ARG1} ${ARG2}) |
| 18:58.39 | twisted | hmmm |
| 18:59.02 | nnY_2 | jblack oh yeah replace app with noop, thanks :) |
| 18:59.04 | twisted | i wonder if there's a better way to figure out what extension a channel's in other than *chan->exten or whatever it is |
| 18:59.14 | twisted | oops, i need to be in -dev |
| 18:59.30 | jblack | nah. You can be here, twisted. It's ok. |
| 18:59.32 | aliver | seanbright from what jblack was saying, it sounds like I have a dialplan problem. |
| 18:59.40 | aliver | http://pastebin.com/d1e78b347 <-- my dialplan |
| 19:00.08 | jblack | No, You misunderstood me. From what you've described, you have no problem at all. |
| 19:00.39 | seanbright | aliver: when you transfer the call do they get to the user's voicemail? |
| 19:00.52 | aliver | Hmm, okay. Well, what happens is that if a SIP phone with an extension is turned off and someone dials that extension then they get a fastbusy rather than the voicemail. |
| 19:01.06 | [TK]D-Fender | aliver: show us the CALL! |
| 19:01.15 | jblack | aliver: Ok. paste here, the _one_ line that tries to dial the phone. |
| 19:01.16 | aliver | [TK]D-Fender I'm gathering the info now. |
| 19:02.05 | jblack | [TK]D-Fender: buddy, think we're gonna need all that info? |
| 19:02.42 | jblack | sounds to me he just needs an exten..n..voicemail() line |
| 19:03.01 | nnY_2 | jblack http://pastebin.com/ma28bc27 |
| 19:03.02 | [TK]D-Fender | ~cluebat jblack |
| 19:03.03 | jbot | ACTION pulls out a ClueBat (tm) and thwaps jblack. |
| 19:03.04 | [TK]D-Fender | <emeril> BAM!!!!!!! |
| 19:03.06 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 19:03.08 | seanbright | aliver: there is nothing in your dialplan that actually calls VoiceMail() other than the _+197020337XX extension |
| 19:03.12 | jblack | Ouch! |
| 19:03.13 | *** join/#asterisk lesouvage (n=lesouvag@cc341200-a.assen1.dr.home.nl) |
| 19:03.16 | aliver | I have other issues going on, my SIP provider is limiting my number of trunks. I'll come back when I have my shit together. |
| 19:03.29 | seanbright | aliver: sounds great. |
| 19:03.29 | aliver | seanbright ah, well that would be a problem, too. |
| 19:03.38 | seanbright | aliver: ya think? ;) |
| 19:04.21 | jblack | nnY_2: I can't seem to find the noop running. |
| 19:04.36 | jblack | Oh, there it is. It's executing, but your noops aren't printing. |
| 19:04.52 | jblack | <PROTECTED> |
| 19:04.53 | jblack | <PROTECTED> |
| 19:05.02 | nnY_2 | jblack yeah but the phone never rings |
| 19:05.15 | *** join/#asterisk iotashan (n=shan@adsl-71-150-254-145.dsl.mdsnwi.sbcglobal.net) |
| 19:05.30 | lesouvage | I have autoprovisoning setup and it seems to work (I work remotely) but in the log it looks like the provisioned phone (Polycom IP330) keeps downloading the firmware and other files. See http://www.pastebin.be/13323 . They show up in sip show peers as registered. |
| 19:05.32 | jblack | nnY_2: I don't get this. I thought you were trying to dial a SIP phone, no? |
| 19:05.33 | nnY_2 | jblack I can pastebin sip show peer 100, but if it wasn't registered, asterisk would just bitch and go to vm |
| 19:06.13 | nnY_2 | jblack yes |
| 19:06.16 | nnY_2 | jblack a sip phone |
| 19:06.19 | jblack | Then they is it dialing zap ? |
| 19:06.23 | jblack | s/they/why |
| 19:06.36 | lesouvage | What does the log indicate? I assume this is not normal but am I right? |
| 19:06.42 | nnY_2 | jblack good question let me see what you see |
| 19:07.07 | aliver | ugh. I make the most idiotic mistakes when 5 people are breathing down my throat asking "When will we be able to do XXX with the phones" every 5 minutes. |
| 19:07.12 | jblack | I'm looking at line 12 |
| 19:07.21 | aliver | and I can make some doozies without the pressure. |
| 19:08.07 | nnY_2 | jblack i have no clue the macro states exten => s,1,Dial(${ARG2},20) |
| 19:08.27 | nnY_2 | jblack and ARG2 is SIP/100 (or SIP/${EXTEN}) |
| 19:08.40 | nnY_2 | there is no Dial (Zap/etc) |
| 19:09.16 | jblack | I can't help but wonder if that file include is screwing up arguments. |
| 19:09.38 | nnY_2 | jblack I can try ot move it, but why would it work with one phone consistently but not the other |
| 19:09.46 | nnY_2 | jblack that c include is common in 90% of our systems |
| 19:09.46 | jblack | btw, why are your noops not printing? |
| 19:10.04 | nnY_2 | jblack hmm exten => _1XX,1,Noop(stdexten,${EXTEN},SIP/${EXTEN}) is what i put |
| 19:10.22 | nnY_2 | jblack is it case sensitive? |
| 19:10.24 | jblack | Oh, I was expecting something in english. |
| 19:10.30 | nnY_2 | heh |
| 19:10.39 | jblack | Something like "Dialing ${EXTEN}" |
| 19:11.11 | jblack | # |
| 19:11.12 | jblack | <PROTECTED> |
| 19:11.23 | nnY_2 | yeah i see that |
| 19:11.32 | lesouvage | nnY_2: if you do show applications on the cli is noop in the output? |
| 19:11.37 | jblack | Yeah. I put it here for my benefit. |
| 19:11.57 | nnY_2 | <PROTECTED> |
| 19:12.20 | jblack | noop is supposed to log its output to the cli. It's missing here. |
| 19:13.38 | jblack | Let's try something brute force. |
| 19:13.46 | jblack | in trasfer,1, replace it with a Dial(SIP/100) |
| 19:13.50 | jblack | see if you get a ring |
| 19:13.51 | brodiem | ... according to what jblack and nny just pasted, they are both priority 1.. |
| 19:13.58 | *** join/#asterisk Ryushin (i=proxy@windwalker.openinnovations.com) |
| 19:14.12 | jblack | brodiem: Hmm? The noop is 1, and the macro is 2 |
| 19:14.24 | brodiem | ok |
| 19:14.32 | nnY_2 | <PROTECTED> |
| 19:14.48 | nnY_2 | verbsoity is 10 |
| 19:14.52 | nnY_2 | verbosity too |
| 19:15.01 | jblack | nnY_2: You changed it, but forgot to reload your extensions.conf ? |
| 19:15.12 | nnY_2 | no I reloaded |
| 19:15.23 | lesouvage | nnY_2: I know, but it is jst an application like all the other applications. Checking if it is actually part of the system seems to be a logical step. You can do this by entering "show applications" on the cli. |
| 19:15.34 | jblack | You're telling me transfer,1 is still a noop. |
| 19:15.35 | aliver | What * cmd do I need to use to create a pause before the voicemail message. The folks can't hear the "leave a message" message. |
| 19:15.37 | nnY_2 | lesouvage sorry i confirmed it up above |
| 19:15.40 | jblack | I want it to be a straight Dial(SIP/100) |
| 19:15.52 | nnY_2 | jblack oh ok missed that sorry |
| 19:15.53 | aliver | I can read the docs, but just need to know the name of the cmd. |
| 19:15.54 | jblack | exten => transfer,1,Dial(SIP/100) |
| 19:16.00 | *** join/#asterisk lanning (n=lanning@66.151.128.195) |
| 19:16.05 | jblack | pardon, I hope yo uknow waht I meant. |
| 19:16.12 | vonkleist | Hi guys |
| 19:16.32 | jblack | At this point, it's probably _1XX,1,Dial(SIP/100) |
| 19:16.35 | vonkleist | What does "BLU" and "REC" means as alerts? (on a E1 card) |
| 19:16.44 | nnY_2 | jblack correct |
| 19:17.29 | jblack | You get a ring with that? |
| 19:18.00 | *** join/#asterisk shinao1 (n=shinao1@smtp.gtbank.com) |
| 19:19.02 | jblack | Are you really irc'ing from your mailserver? |
| 19:19.14 | jblack | No wonder gtbank has a 'SCAM E-MAILS ALERT' on the front page. |
| 19:19.49 | jblack | nnY_2: The anticipation is killing me |
| 19:20.00 | nnY_2 | nny_1 heh indeed one sec |
| 19:20.20 | nnY_2 | er um |
| 19:22.32 | nnY_2 | jblack nope doesnt ring |
| 19:22.50 | *** join/#asterisk korihor (n=korihor@201.211.168.130) |
| 19:23.32 | nnY_2 | wtf |
| 19:23.38 | nnY_2 | missed one call |
| 19:23.40 | nnY_2 | at 3:19 |
| 19:23.44 | nnY_2 | pastebinning 3:;19 |
| 19:23.56 | nnY_2 | fuck! |
| 19:24.02 | nnY_2 | stupid fucking scroll limit |
| 19:24.14 | seanbright | settle down |
| 19:24.30 | nnY_2 | it was a call using Dial(SIP/100) |
| 19:24.41 | nnY_2 | jblack and it worked once out of 6 or 7 attempts |
| 19:25.03 | nnY_2 | lights aastra phone on fire, pisses on remains, buries face down in desert |
| 19:26.57 | jblack | If you're reliably dialing SIP/100, and the phone answers 1 out of 7 times, the answer is "the phone is broken" |
| 19:27.15 | jblack | or the cable to it. |
| 19:28.23 | nnY_2 | jblack testing other phone on premise with different cable, same method |
| 19:29.17 | nnY_2 | jblack going to ring it a bunch |
| 19:29.35 | jblack | k |
| 19:30.20 | nnY_2 | jblack but only when PZ is used? |
| 19:30.27 | nnY_2 | none of this makes any sense to me |
| 19:31.01 | nnY_2 | i guess if PS is used than it hands PS --> (IPPHONE) |
| 19:31.22 | nnY_2 | which means it only happens when it's * --> (IPPHONE) |
| 19:31.26 | jblack | It may be that there's a redirect happening with PS. |
| 19:31.50 | nnY_2 | i think most sip calls like that just connect the two sip clients together |
| 19:31.53 | nnY_2 | i could be wrong |
| 19:31.56 | jblack | Built into SIP is the option to 'you guys want to talk, cut me out of the loop' |
| 19:32.38 | *** join/#asterisk zoid_99 (n=chris@router.asteriasgi.com) |
| 19:32.51 | jblack | but other than hardcoding dials in various places, I'm out of ideas. |
| 19:32.57 | zoid_99 | any tips on a ringall followme? |
| 19:33.17 | zoid_99 | concurrent followme that is instead of sequential |
| 19:33.31 | jblack | zoid_99: You can just dial all the phones at once, no problem. |
| 19:33.45 | jblack | Dial(SIP/1&SIP/2&SIP/3,20) |
| 19:33.46 | zoid_99 | jblack: yes |
| 19:34.08 | zoid_99 | jblack: but if someone presses 2 to reject it hangs up all the calls |
| 19:34.17 | jblack | that shouldn't be. |
| 19:34.27 | zoid_99 | jblack: we are using the followme app |
| 19:34.38 | jblack | I'm suggesting not using followme at all. |
| 19:35.06 | zoid_99 | we need to use followme as we need confirmation that a human answered via the press 1 |
| 19:35.08 | jblack | or breaking the process up in 2. |
| 19:35.14 | nnY_2 | jblack ok two different phones, different cables, different port on the switch same issue |
| 19:35.29 | jblack | say, ..,1,Dial(1&2&3&4) ...,2,Followme... |
| 19:35.37 | jblack | nnY_2: I give up |
| 19:35.45 | nnY_2 | jblack k thanks |
| 19:35.53 | jblack | nnY_2: I'd just try putting the dial earlier and earlier. |
| 19:35.57 | zoid_99 | that breaks when voicemail answers |
| 19:36.02 | jblack | Basicly cut that extension logic out of the equation. |
| 19:36.02 | nnY_2 | jblack have it after answer |
| 19:37.00 | jblack | zoid; Dial() doesn't call voicemail, and if the phone is going to voicemail on it's own, either don't group dial it, or set the dial timeout to before voicemail kicks in. |
| 19:37.10 | nnY_2 | jblack putting it in default |
| 19:37.53 | nnY_2 | jblack still doesn't work. Meh I hate aastra |
| 19:38.54 | jblack | we dont know that we can blame the aastra. |
| 19:39.19 | jblack | zoid_99: In simpler terms, if verizon wireless sends a call to voice mail, it counts as "answered". You need to beat it to the punch |
| 19:42.00 | nnY_2 | jblack um so where else would i look? |
| 19:42.08 | nnY_2 | jblack i have a dial in my default context |
| 19:42.16 | nnY_2 | unless asterisk is not compatible with aastra |
| 19:42.20 | nnY_2 | which they state they are |
| 19:43.58 | jblack | If you're alreayd at the top, and dial fails... |
| 19:44.16 | jblack | Look at the authentication. |
| 19:44.26 | jblack | debug the sip. see if you're getting error codes, espeically authentication |
| 19:44.31 | *** join/#asterisk Nugget (i=nugget@carrera.macnugget.org) |
| 19:45.01 | jblack | at least now you know you can discount the dialplan |
| 19:45.50 | nnY_2 | jblack ok |
| 19:45.55 | *** join/#asterisk ibercom (i=581c60f2@gateway/web/ajax/mibbit.com/x-7997cead7f2c8e56) |
| 19:46.12 | *** join/#asterisk ManxPower (n=manxpowe@209.16.72.135) |
| 19:46.38 | Kobaz | [Aug 22 15:44:58] WARNING[27534]: chan_sip.c:12621 handle_response: Remote host can't match request NOTIFY to call '2a96309e652694ab13dcd667026d5854@192.168.24.12'. |
| 19:46.47 | zoid_99 | jblack: that's why we use followme so that a human has to press 1 to confirm that they will take the call |
| 19:46.48 | Kobaz | every so often I get one of those from an audiocodes gateway |
| 19:46.49 | ManxPower | Does anyone know if Asterisk's cdr_sqlite works with sqlite v3? |
| 19:47.22 | zoid_99 | jblack: the reject button causes all of the calls to be rejected... |
| 19:47.36 | jblack | zoid_99: It seems to me that concurrent dialing and follow me are slightly incompatible with each other. |
| 19:47.38 | zoid_99 | found the solution: disable the reject key |
| 19:47.50 | nnY_2 | jblack http://pastebin.com/m521abcee |
| 19:47.52 | jblack | Ok. I'll keep that in mind the next time someone asks. |
| 19:48.12 | zoid_99 | jblack: it seems to work fine if the person receiving the call just hangs up |
| 19:48.31 | ManxPower | This is going to be one of these days where the fact that I spend hundreds of hours helping people here and nobody can help with my question |
| 19:48.53 | zoid_99 | ManxPower: what was the question? |
| 19:49.13 | jblack | zoid_99: if manx says no one can answer it, it's usually true. |
| 19:49.27 | jblack | nnY_2: I gave up. sorry. |
| 19:49.30 | nnY_2 | jblack roger |
| 19:49.32 | zoid_99 | Manx: I see it.. sorry, we use postgres for cdrs |
| 19:49.33 | *** join/#asterisk Assimilate (n=Assimila@216.83.78.108) |
| 19:49.42 | nnY_2 | fuck! |
| 19:49.45 | nnY_2 | i figured it out |
| 19:49.47 | nnY_2 | shoot me |
| 19:49.48 | jblack | ? |
| 19:49.50 | nnY_2 | Uknown Caller |
| 19:49.55 | nnY_2 | the space breaks Aastra |
| 19:50.04 | jblack | LOL |
| 19:50.13 | jblack | 6 out of 7 times? |
| 19:50.19 | zoid_99 | jblack: I'm aware of ManxPower's knowledge of asterisk |
| 19:50.23 | nnY_2 | CID is broken on telco end |
| 19:50.27 | jblack | zoid_99: Ok. sorry. |
| 19:50.52 | jblack | nnY_2: Hmm. you might have just helped me with a problem. |
| 19:51.12 | jblack | It's seemed to me that CID was really broken for a long time. I never considered that it may be spaces in the name for cid. |
| 19:51.16 | *** join/#asterisk bbryant (n=brett@c-68-59-20-153.hsd1.sc.comcast.net) |
| 19:51.42 | [TK]D-Fender | nnY_2: \o/ |
| 19:51.49 | nnY_2 | lol |
| 19:51.51 | nnY_2 | testing to confirm |
| 19:51.56 | *** part/#asterisk |||Mad||| (n=mad@mail.rubbusa.com) |
| 19:52.14 | jblack | looks for a way to take some credit. |
| 19:52.21 | nnY_2 | jblack you had it right |
| 19:52.26 | nnY_2 | it was staring me right in the face |
| 19:52.28 | nnY_2 | in sip.conf |
| 19:52.44 | nnY_2 | i did that months ago cause people what say "Well who is asterisk?" |
| 19:52.49 | nnY_2 | and i would tell them what |
| 19:52.55 | nnY_2 | and they would still look confused |
| 19:53.20 | nnY_2 | I need to put this on the wiki |
| 19:53.41 | nnY_2 | sand that explains why PS worked |
| 19:53.45 | nnY_2 | cause CID works on PS |
| 19:54.58 | nnY_2 | and why xlite worked |
| 19:55.08 | jblack | life makes sense again. |
| 19:55.08 | nnY_2 | so to recap Aastra cannot have spaces in CID |
| 19:55.12 | nnY_2 | that's super gay |
| 19:55.23 | ManxPower | zoid_99: (2:46:49 PM) ManxPower: Does anyone know if Asterisk's cdr_sqlite works with sqlite v3? |
| 19:55.29 | nnY_2 | (no offense to any gay people) I am not calling you aastra |
| 19:55.40 | jblack | wait. Aastra can't, or your pri provider chokes? |
| 19:55.48 | nnY_2 | ManxPower I saw support in 1.6 for lite |
| 19:55.54 | ManxPower | nnY_2: At least %30 of the channel is gay |
| 19:55.54 | nnY_2 | ManxPower in the notes |
| 19:56.06 | nnY_2 | cool |
| 19:56.14 | ManxPower | nnY_2: I'll assume since it's noteworthy, it's not supported in 1.2 |
| 19:56.48 | nnY_2 | i'll find a new derogatory term to associate aastra with. They're crackertastic! |
| 19:57.13 | jblack | ManxPower: Surely it must be through odbc |
| 19:57.59 | nnY_2 | ManxPower * Added a new CDR module, cdr_sqlite3_custom. |
| 19:58.06 | nnY_2 | ManxPower from http://svn.digium.com/view/asterisk/tags/1.6.0-beta1/CHANGES?view=co |
| 19:58.15 | nnY_2 | ManxPower trying hard to help for once :) |
| 19:58.18 | [TK]D-Fender | nnY_2: "Shitacular" |
| 19:58.26 | nnY_2 | Craptatsic! |
| 19:58.34 | nnY_2 | Craptastic too! |
| 19:59.20 | jblack | Comcastic. :) |
| 20:00.30 | nnY_2 | time to call brenedt at aastra back\ |
| 20:00.54 | jblack | tell him he owes me one for testing purposes. |
| 20:01.01 | nnY_2 | haha |
| 20:01.03 | nnY_2 | i owe ya one shit |
| 20:01.07 | nnY_2 | whats your email addy? |
| 20:01.23 | jblack | jblack@linuxguru.net |
| 20:01.28 | nnY_2 | you have paypal? |
| 20:02.01 | jblack | Actually, I do, but it's a different address |
| 20:02.12 | jblack | which I'm looking up |
| 20:02.35 | jblack | uh oh. paypal is crashing mozilla |
| 20:02.45 | ManxPower | nnY_2: We use 1.2 |
| 20:02.56 | ManxPower | But thanks for checking. |
| 20:02.58 | jblack | wtf |
| 20:03.02 | *** part/#asterisk ManxPower (n=manxpowe@209.16.72.135) |
| 20:03.21 | *** join/#asterisk rivalmel (n=zdraper@64-142-43-180.dsl.static.sonic.net) |
| 20:03.31 | jblack | Segmentation fault (core dumped) |
| 20:03.38 | nnY_2 | woot |
| 20:03.41 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 20:04.52 | jblack | hopes there's a new mozilla available |
| 20:05.17 | *** join/#asterisk propellerhead (n=yogurt2u@host170.190-30-192.telecom.net.ar) |
| 20:05.46 | *** join/#asterisk CanWood (n=chatzill@24.108.64.80) |
| 20:05.56 | propellerhead | ~centos52bug |
| 20:05.57 | jbot | [~centos52bug] There is a bug compiling Zaptel up to 1.4.11/1.2.26 on CentOS/RHEL 5.2: error in xpp/xdefs.h. Workaround: uncheck "xpp" (Astribank) driver in menuselect. Fixed in SVN. Patches: http://bugs.digium.com/12889 |
| 20:06.33 | jblack | Here we go. bug #252229 |
| 20:08.49 | jblack | nnY_2: It's jblack@merconline.com |
| 20:09.54 | *** join/#asterisk luxxx77 (n=luxxx77@g226195083.adsl.alicedsl.de) |
| 20:10.16 | nnY_2 | jblack k |
| 20:10.57 | nnY_2 | jblack btw it seems it only happens when there is no numerical CID info available, and if someone replaces the default "asterisk" with something with a space in it |
| 20:11.03 | nnY_2 | jblack so far only aastra is affected |
| 20:11.16 | nnY_2 | i have a grandstream, polycom, linksys 962 at the office i can test with as well |
| 20:11.35 | nnY_2 | brb smoke |
| 20:11.38 | jblack | It's a problem I'm having with a provider or two. |
| 20:11.40 | jblack | yeah. smoke for me too |
| 20:12.03 | *** join/#asterisk drako (n=ljd@nelug/coreteam/luisjose) |
| 20:13.05 | vonkleist | moy, do u know the correct parameters for telmex on zaptel.conf ? |
| 20:13.24 | luxxx77 | in the old configuration style i needed to write "register => USER:PASS@sip.carpo.de/carpo". What do i have to use now? The SPECIAL thing is the last part, which uses not the USERNAME again, but instead the word "capo"... |
| 20:16.23 | luxxx77 | so which variable do i have to set, so the SIP registration will be done with "...@sip.carpo.de/carpo"? |
| 20:18.59 | *** join/#asterisk CanWood (n=chatzill@24.108.64.80) |
| 20:19.36 | nnY_2 | jblack would this be a SIP issue? |
| 20:19.42 | moy | vonkleist: the same as for any other carrier cas=1,1,0,cas,hdb3 |
| 20:19.43 | nnY_2 | jblack i am editing the voip-info entry |
| 20:19.49 | moy | vonkleist: read http://www.voip-info.org/wiki/view/Asterisk+MFC+R2 |
| 20:19.56 | moy | zaptel.conf config is like that |
| 20:20.00 | jblack | nnY_2: It's just the aastra, right? |
| 20:20.29 | jblack | I'd put a note on the callerid page and the astra page that some aastras break in difficult to debug ways if there is a space in the name portion of callerid |
| 20:25.03 | jblack | nnY_2: Thanks for the tip! |
| 20:26.55 | [TK]D-Fender | ok, checkout time. BBL |
| 20:27.08 | *** join/#asterisk ghenry (n=ghenry@ghenry.plus.com) |
| 20:27.11 | nnY_2 | jblack pp'd ya thanks for the help |
| 20:27.24 | jblack | yeah. it already went through |
| 20:27.29 | ghenry | Can any recommended a technique for calling to/from office in UK/US/Asia? |
| 20:27.37 | nnY_2 | jblack yeah i put it under SIP issues |
| 20:27.44 | ghenry | Some hops inbetween? |
| 20:27.47 | nnY_2 | i'll add it to the callerid page too |
| 20:28.23 | jblack | ghenry: Can you rephrase that? It's too ambigious |
| 20:28.34 | ghenry | hi jblack |
| 20:28.45 | ghenry | company has HQ office in Glasgow |
| 20:28.47 | jblack | Hi. |
| 20:29.07 | jblack | Keep going... |
| 20:29.11 | ghenry | they have other offices in Seol, Tokoyo, San Fran |
| 20:29.14 | ghenry | Bejing |
| 20:29.25 | ghenry | Best way to make VoIP bteween them |
| 20:29.29 | nnY_2 | should i put it under the general callerid info page or the func callerid page |
| 20:29.37 | jblack | And you want to know if you can have asterisk phone systems in each office, and do calls between them for free. |
| 20:29.43 | jblack | ghenry: Definitely, yes. |
| 20:29.57 | jblack | nnY_2: Hmmmm. |
| 20:30.12 | ghenry | jblack: Yes, I know VoIP is the best way |
| 20:30.13 | ghenry | ;-) |
| 20:30.18 | ghenry | But latency probs |
| 20:30.18 | jblack | I'd put it on the func page. That's the last chance someone has to find out before needing aspirin. |
| 20:30.23 | ghenry | some offices are > 300ms |
| 20:30.32 | jblack | you're screwed on those offices. |
| 20:30.59 | jblack | You can try using IAX and setting a good jitter buffer, and it'll "kinda work", but you'll get plenty of complaints about people talking over each other. |
| 20:31.00 | ghenry | so can't do hopes in between to bring it down? |
| 20:31.25 | jblack | Unfortunately, there's no "MakeInternetGoFasterOption=yes" option. :9 |
| 20:31.27 | jblack | :( |
| 20:31.34 | ghenry | :-( |
| 20:31.47 | ghenry | but intermediate way points that forward? |
| 20:31.59 | jblack | You can try switching isp's at one or more offices, and hope for better peering backbones. |
| 20:32.12 | ghenry | right |
| 20:32.47 | jblack | Being europe, perhaps put everyone on orange, and have them treat you like one big company, instead of lots of small companies. |
| 20:33.35 | ghenry | hmmm. |
| 20:33.45 | ghenry | orange broadband? |
| 20:34.11 | jblack | sure. Just a random company name drop. I'm sure there's other companies that might work for you. |
| 20:34.28 | jblack | I'm in the United States, so I'm not deeply familiar with the providers available to you. |
| 20:34.46 | ghenry | I'll ping the * biz list |
| 20:34.49 | ghenry | thanks jblack |
| 20:35.01 | jblack | but if you can get everyone on the same provider, you might be able to get high volume discounts, and you should stay on the same network, cutting latency down significantly |
| 20:35.03 | *** join/#asterisk ibercom (i=581c6376@gateway/web/ajax/mibbit.com/x-49c79f44b7a66a45) |
| 20:35.38 | jblack | but double check that to make sure they're a large network, and not just a conglomerate of networks |
| 20:35.55 | jblack | There's another option. |
| 20:36.16 | jblack | Buy 15,000 miles of fiber optic cable, and drape it over everyone's house between your offices. |
| 20:37.18 | nnY_2 | ghenry may wanna see if they offer any kind of "metro ethernet" |
| 20:37.30 | nnY_2 | ghenry basically point to point connection that acts like a layer 2 switch |
| 20:37.36 | *** join/#asterisk Shotygun (n=thorn@82.166.241.50) |
| 20:37.55 | ghenry | yeah, thanks nnY_2 |
| 20:38.01 | jblack | I'd jus try and get him on the same network, and have him build the vpn himself. |
| 20:38.14 | jblack | Then he has control over traffic shaping, routing, etc etc. |
| 20:38.15 | ghenry | I've pinged the biz list to see what comanies chirp up |
| 20:38.30 | nnY_2 | yeah although here our latency inter network is still 91ms |
| 20:38.44 | ghenry | what locations nny_2? |
| 20:38.55 | jblack | Which for him would be a savings of about 70% |
| 20:39.02 | nnY_2 | hilton head island, South carolina, USA |
| 20:39.17 | nnY_2 | yeah the 91 ms is how i have my house phone connected to the office and it works great |
| 20:39.33 | nnY_2 | i agree highly, the metro E stuff is about 300 per side here |
| 20:39.51 | jblack | I vpn to one of my clients from pennsylvania to washington state. I typically see 140ms, and calls still work fine. |
| 20:40.11 | ghenry | yeah, about same here nny_2 |
| 20:40.37 | ghenry | yeah. less than 150ms or 200ms recommended? I always say 150 |
| 20:41.20 | nnY_2 | yeah the only latency issues i have had is when it's a large amount of hops. I have a client who uses Internetcalls.com with asterisk, server here, connecting to some freaking place in europe, and you get the lag between |
| 20:41.20 | nnY_2 | like hi!... pause.. Hello! |
| 20:41.20 | ghenry | nnY_2: that's oru prob |
| 20:41.20 | kfife | Simple Question: I'm calling a known IVR. When it anserws, I want to drill down with some DTMF tones. I'm trying to use the SendDTMF() application, with the G flag in the Dial() appliciton to direct the call after the called party answers. I'm a little confused as to why the G flag sends the called and calling parties to TWO different locations: Prioirity and Priority+1. I can understand the need to do different things with each call leg, bu |
| 20:41.20 | kfife | o send them to n and n+1?? Am I doing this wrong? What am I missing here? Sorry for the wordy question! Thanks! |
| 20:41.20 | jblack | ghenry: I try to shoot under 150ms for voip, and start solving problems by 200. |
| 20:41.29 | ghenry | Edinburgh, Kuala Lumpur, Singapore, Tokyo, Seoul, Beijing, San Francisco |
| 20:41.36 | ghenry | that's or offices |
| 20:41.38 | nnY_2 | ghenry ooh |
| 20:41.51 | jblack | ghenry: btw.... |
| 20:41.59 | nnY_2 | heh time to start running fiber over houses |
| 20:42.02 | jblack | ghenry: I'd check the ping times between all of the offices to all of the other offices. |
| 20:42.18 | jblack | It's possible that one office could have excellent timings to the rest of them. You could possibly route all calls through a central point. |
| 20:42.46 | jblack | I'd check San Fran and Tokyo first. |
| 20:42.55 | nnY_2 | heh i got bored one day and set my computer up to use my cell phone as a modem. (CDMA), and then connected to our box |
| 20:42.59 | nnY_2 | it was hilarious |
| 20:43.02 | nnY_2 | and useless |
| 20:43.21 | jblack | they intentionally push latency up on CDMA to prevent voip. |
| 20:43.33 | ghenry | jblack, good idea |
| 20:43.34 | nnY_2 | bless their black hearts |
| 20:44.06 | jblack | aye. |
| 20:44.15 | nnY_2 | i have a sjphone setup at the office using the local wireless and it sucks too. wonder if there is some trickery involved there |
| 20:44.15 | Shotygun | What's the latency you guys get over cdma? |
| 20:44.22 | nnY_2 | somethingl ike 600ms |
| 20:44.40 | Shotygun | jes.. here I get about 200ms average over the cdma network. |
| 20:44.50 | jblack | Depending upon providers, I've seen ranges of 550 to 1200 ms. |
| 20:44.55 | Shotygun | 100 to 300, depends on the area. |
| 20:45.41 | jblack | they can't charge you $1.50 a minute for international calls if you use sip on their network. |
| 20:45.53 | nnY_2 | i have yet to see a softphone on a pda or cell phone that works |
| 20:46.03 | nnY_2 | haven't tried iphone, probably never will |
| 20:46.09 | jblack | I've made it work. It wasn't pretty. |
| 20:46.18 | Shotygun | What about the nokia sip support? |
| 20:46.26 | nnY_2 | i have gotten it working, but it sounded awful |
| 20:46.27 | Shotygun | (Never tried it, didn't get the chance to play with one yet) |
| 20:46.32 | nnY_2 | thats what i was wanting to try |
| 20:46.33 | jblack | I don't know. Things are starting to change, though. |
| 20:46.35 | nnY_2 | the n80 or something |
| 20:46.51 | *** join/#asterisk hoegaatit (n=laa@c-24-5-27-120.hsd1.ca.comcast.net) |
| 20:47.02 | *** join/#asterisk jnfuller (n=jnfuller@S0106001217db850e.vs.shawcable.net) |
| 20:47.11 | nnY_2 | someone has to say hey! Everyone else is scared of this, I can make a business model out of it and get all their customers! |
| 20:47.17 | jnfuller | hi all |
| 20:47.27 | jblack | sounds like what nokia's is trying, nnY_2 |
| 20:47.28 | kfife | howdy |
| 20:47.34 | nnY_2 | jblack yay |
| 20:48.19 | jblack | google will kick ass, no doubt |
| 20:48.23 | hoegaatit | I would like to add lines like "exten => 100,7,Dial(IAX2/foo)" dynamically to my dialplan. Is AGI the way to go? |
| 20:48.27 | nnY_2 | yah thats what i am hoping for |
| 20:48.38 | jblack | hoegaatit: "dynamically" ? Look at AGI. |
| 20:48.47 | nnY_2 | aren't they pushing for the space in between the old VHF spectrum or something |
| 20:48.55 | hoegaatit | jblack: yeah from a script |
| 20:49.03 | jblack | hoegaatit: Yeah, go with agi |
| 20:49.12 | hoegaatit | when a new agent registers I'd like to add their exten to my dialplan |
| 20:49.13 | hoegaatit | k |
| 20:49.14 | Shotygun | He can also use real-time db |
| 20:49.15 | *** join/#asterisk [TK]D-Fender (n=chatzill@64.235.218.194) |
| 20:49.18 | jblack | nnY_2: Yeah. them, m$, intel, a couple others. |
| 20:49.38 | jnfuller | why not flexible patttern matching? |
| 20:50.06 | hoegaatit | how do you mean? |
| 20:50.17 | nnY_2 | fuck if m$ gets it we'll have clippy the phone assistant "Hey i see you are dialing drunk, want me to screw up and dial the number to your parent's instead?" |
| 20:50.18 | jblack | Yesterday, the issue was the wireless mic companies that make mics for churches and stadiums. They were bitching their use of the airwaves were gonna get screwed. |
| 20:50.32 | nnY_2 | yeah saw that |
| 20:50.42 | jblack | google came back to the FCC with "this is licensed spectrum, and they don't have a license". And now FCC is coming down on them like a ton of bricks |
| 20:50.50 | jnfuller | if agents have similar extensions the matching can be loose instead of per extension |
| 20:51.23 | jblack | ars has the latest |
| 20:51.36 | jblack | http://arstechnica.com/news.ars/post/20080822-fcc-wants-wireless-mic-ban-at-700mhz-to-boost-broadband.html |
| 20:52.29 | jblack | pardon, PISC pushed the FCC, not google. |
| 20:52.52 | adrianXXX | who can i uninstall asterisk from Ubuntu 7.04... |
| 20:53.02 | jblack | adrianXXX: You |
| 20:53.03 | russellb | did you ask that question yesterday? |
| 20:53.11 | adrianXXX | jblack : yes |
| 20:53.14 | russellb | and got a very detailed answer ... |
| 20:53.26 | jblack | adrianXXX: Thank you Come Again! |
| 20:53.29 | adrianXXX | who me ? |
| 20:53.32 | russellb | yes |
| 20:53.42 | adrianXXX | no |
| 20:53.48 | adrianXXX | from ubuntu its diferent |
| 20:53.49 | Qwell | yes |
| 20:53.58 | russellb | >_< |
| 20:54.01 | Qwell | it's not different |
| 20:54.08 | adrianXXX | bicause i installed with the apt-get.. |
| 20:54.11 | jblack | perhaps there was some different, completely unrelated adrianWWW that visited? |
| 20:54.16 | Qwell | so...remove it with apt-get |
| 20:54.22 | adrianXXX | who |
| 20:54.23 | adrianXXX | ? |
| 20:54.26 | jblack | adrianxxx: They can and will help you in #ubuntu |
| 20:54.30 | Qwell | THE GUY THAT JUST ASKED THE QUESTION |
| 20:54.45 | gaetronik | may be s/who/how |
| 20:55.01 | Qwell | gaetronik: unlikely |
| 20:55.17 | gaetronik | aptitude purge asterisk might be the answer |
| 20:55.20 | kfife | Question: Trying to send DTMF tones AFTER a called party picks up to drill down into an IVR. Should I use the D flag in Dial() or SendDTMF()? |
| 20:55.22 | rivalmel | sudo apt-get remove asterisk |
| 20:55.37 | adrianXXX | thanks rivalmel |
| 20:55.44 | rivalmel | np |
| 20:55.44 | jnfuller | http://www.google.com/search?hl=en&safe=off&client=safari&rls=en-us&q=uninstall+asterisk+from+ubuntu+7.04&btnG=Search |
| 20:55.52 | mchou | kfife: that's supposedly what it's for |
| 20:55.56 | [TK]D-Fender | kfife: D() |
| 20:56.47 | *** join/#asterisk edwin_quijada (n=macaruch@25.116.88.200.m.sta.codetel.net.do) |
| 20:57.22 | edwin_quijada | I run safe_asterisk and I get error as Asterisk died with code 1 |
| 20:57.41 | edwin_quijada | mpg123: no process killed |
| 20:57.47 | edwin_quijada | what does mean? |
| 20:57.50 | jblack | red herring. |
| 20:57.58 | jblack | It's probably file permissions or something. |
| 20:58.08 | jnfuller | asterisk -vvvvvvvvgc |
| 20:58.13 | jblack | check the logs in /var/log too |
| 20:58.16 | jnfuller | see where it really dies |
| 20:58.21 | [TK]D-Fender | edwin_quijada: Means "I should know better and run * MANUALLY to see whats actually happening" |
| 20:58.26 | kfife | Doc says: D() sends the DTMF digits BEFORE the call gets bridged. To clarify, that means the calling party WON'T HEAR them? It doesn't mean for example they're sent out-of-band before the audio channel is established, is that correct? Thanks! |
| 20:58.41 | [TK]D-Fender | kfife: Correct |
| 20:58.46 | kfife | Thank you! |
| 20:58.51 | jblack | kfife: Correct. as far as I know, there's no way to do dtmf after the call is established. |
| 20:58.55 | edwin_quijada | [TK]D-Fender: what commando to use to run manually |
| 20:58.57 | jblack | Which sucks, but is what it is. |
| 20:59.07 | [TK]D-Fender | jblack: There is. |
| 20:59.30 | kfife | You've really helped me out. Thanks |
| 20:59.32 | [TK]D-Fender | jblack: Can't see much of a need to however |
| 20:59.58 | jblack | I've had a need several times. |
| 20:59.59 | edwin_quijada | it says zaptel configuration but i dont have zaptel cards |
| 21:00.35 | [TK]D-Fender | edwin_quijada: Go fix it, whatever it is. Probably ztdummy... |
| 21:02.45 | kfife | Can anyone confirm that the 500ms wait 'w' from SendDTMF() is supported in the Dial() D() flag? |
| 21:03.12 | [TK]D-Fender | kfife: AFAIK, it isn't supported in either |
| 21:03.21 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 21:03.28 | kfife | Doc shows it in SendDTMF() |
| 21:03.38 | [TK]D-Fender | kfife: ok. |
| 21:05.23 | edwin_quijada | what can i use to simulate zaptel services ? ztdummy |
| 21:05.24 | edwin_quijada | ? |
| 21:05.54 | kfife | Thanks! Everyone can have a cold beer on me to show you my gratitude for the pontier! http://dooleyman.com/images/cold_beer.jpg |
| 21:06.16 | [TK]D-Fender | edwin_quijada: what "services"? |
| 21:07.34 | [TK]D-Fender | kfife: ick... you know what American beer and sex in a canoe have in common? |
| 21:09.02 | [TK]D-Fender | kfife: They're both fucking close to water! |
| 21:09.24 | rivalmel | lol |
| 21:11.21 | kfife | [TK]D-Fender: I won't show you what I gave the guys in the other forum. |
| 21:11.27 | kfife | :-) |
| 21:15.27 | *** join/#asterisk kensuke_ (i=c829e4f4@gateway/web/ajax/mibbit.com/x-449d2bc78cf44611) |
| 21:15.41 | nnY_2 | hmm |
| 21:16.28 | nnY_2 | talking with Mercestes about my issue today, and he noticed setting callerid=Unknown Caller (now just Unknown) edits the user field in the SIP dialog |
| 21:16.29 | nnY_2 | From: "Unknown Caller" <sip:Unknown Caller@192.168.100.5>;tag=as57f4f2ce |
| 21:17.11 | jblack | what did he have to say? |
| 21:18.08 | nnY_2 | Mercestes: SHoudl be "Unknown Caller" <sip:User@IP Address>;tag=something |
| 21:18.39 | jblack | so we're back to bad authentication |
| 21:19.02 | nnY_2 | i need to find out how I should be changing the unknown incoming CID appropiately, I don't remember how that method came about |
| 21:19.18 | nnY_2 | but if i remove the space it works :\ |
| 21:19.19 | edwin_quijada | how can i unload modules zaptel |
| 21:19.25 | jblack | try setting it in default in sip.conf |
| 21:19.32 | nnY_2 | haha that's where it is |
| 21:19.50 | [TK]D-Fender | edwin_quijada: "noload => chan_zap.so" <- in modules.conf |
| 21:19.57 | nnY_2 | so maybe an asterisk bug? Or that is not the intended use for it |
| 21:20.20 | nnY_2 | right now with no space it works fine, maybe i should file a bug report and see what comes of it |
| 21:20.31 | jblack | It's worth a try. |
| 21:20.48 | jblack | Like I mentioned, callerid has been fubar for me since 1.4.9 or so |
| 21:21.23 | jblack | It's different behaviours for me, though. |
| 21:21.45 | jblack | In my case, setting callerid(name) causes callerid to not be set at all. |
| 21:23.24 | jblack | I haven't filed a bug about it yet, because I wasn't able to pin it down. A half dozen providers give a half dozen different types of broken. |
| 21:24.29 | edwin_quijada | [TK]D-Fender: I need to unload everything about zaptel |
| 21:24.36 | edwin_quijada | how do you do |
| 21:24.40 | nnY_2 | would this be considered an issue with chan_sip? |
| 21:24.46 | [TK]D-Fender | edwin_quijada: the command I gave you will stop it from loading. |
| 21:24.55 | edwin_quijada | I unload in modules.conf chan_so |
| 21:25.02 | [TK]D-Fender | nnY_2: No, definitle an Aastra issue |
| 21:25.19 | edwin_quijada | i did it but when I run asterisk -vvvgc i get the same error about zaptel |
| 21:25.24 | nnY_2 | [TK]D-Fender hmm someone noticed that asterisk was mangling the user field in the CID |
| 21:25.28 | nnY_2 | mercestes actually |
| 21:25.31 | [TK]D-Fender | edwin_quijada: pastebin your modules.conf |
| 21:25.39 | edwin_quijada | ok |
| 21:25.44 | nnY_2 | [TK]D-Fender From: "Unknown Caller" <sip:Unknown Caller@192.168.100.5>;tag=as57f4f2ce |
| 21:26.12 | nnY_2 | which i think yeah. the fact a space make aastra pissy is their issue |
| 21:26.16 | [TK]D-Fender | nnY_2: pastebin your call w/ sip debug. |
| 21:26.31 | [TK]D-Fender | nnY_2: And your peer configs |
| 21:26.53 | jblack | watches the last 2 hours turn full circle |
| 21:27.09 | nnY_2 | lol |
| 21:27.24 | nnY_2 | at this point the issue is gone, but for the sake of scientific discovery I can |
| 21:27.42 | nnY_2 | really thinking I should at least make it known, and make sure I know what the issue is/was |
| 21:28.30 | edwin_quijada | [TK]D-Fender: http://pastebin.com/m243eda8c this is my modules.conf |
| 21:29.05 | *** join/#asterisk LND (n=Lee@89.192.135.129) |
| 21:29.33 | [TK]D-Fender | edwin_quijada: And now "ztcfg -vvvv" and "asterisk -gvvvvvvvvc" |
| 21:29.53 | *** join/#asterisk drwelby (n=mpfister@mail.enplan.com) |
| 21:30.14 | *** join/#asterisk Paige_ (n=Paige@2001:470:b:aa:21f:c6ff:fe48:8ce3) |
| 21:31.18 | drwelby | Anyone ever played with one of these: http://www.888voipstore.com/fuzion-ip-pbx-1u-rack-server-pr-19372.html ? I'm curious if the GUI is as idiot-proof as it looks. |
| 21:33.35 | *** join/#asterisk chadcrew (n=chadcrew@c-76-19-8-236.hsd1.ma.comcast.net) |
| 21:33.41 | nnY_2 | [TK]D-Fender if you really want to here's my conf files http://pastebin.com/m18d0cc75 and here is my debug http://pastebin.com/m7a885c and here is the sip dialog http://pastebin.com/m688557f9 |
| 21:33.59 | nnY_2 | [TK]D-Fender please note it works now, the solution was to remove the space between Unknown and Caller in sip.conf |
| 21:34.09 | nnY_2 | [TK]D-Fender and that it only breaks Aastra phones afaik |
| 21:34.29 | nnY_2 | however the comment was made that if asterisk is mangling the user field to Unknown Caller than that it bad too |
| 21:35.39 | nnY_2 | is* |
| 21:35.53 | nnY_2 | i put notes about it in voip-info's entires for CID func and the 480i |
| 21:36.28 | nnY_2 | just trying to put as much effort into realying the issue as people did helping me try to fix it ^^ |
| 21:36.33 | edwin_quijada | http://pastebin.com/m3980a13 |
| 21:36.33 | nnY_2 | relaying* |
| 21:36.43 | edwin_quijada | [TK]D-Fender: http://pastebin.com/m3980a13 |
| 21:37.24 | nnY_2 | jblack ha btw the problem arose only when CID stopped working yesterday, and i have a line ticket in for the client |
| 21:37.57 | [TK]D-Fender | nny_call from the BEGINNING please. |
| 21:38.06 | jblack | Seems like an odd thing to work on weds, and break on thurs. |
| 21:38.48 | anonymouz666 | [TK]D-Fender: http://pastebin.com/m6b225b0f --> care to comment on this? |
| 21:39.36 | [TK]D-Fender | anonymouz666: nope. |
| 21:39.55 | jblack | anonymouz666: Your /etc/zaptel.conf is wrong. |
| 21:40.19 | jblack | probably the bchan and dchan lines. Ask your provider where your dchan is, and adjust accordingly. |
| 21:40.25 | *** join/#asterisk Arck-FR (n=Arck-FR@cvl92-2-82-228-145-232.fbx.proxad.net) |
| 21:40.43 | nnY_2 | [TK]D-Fender crap must of gotten cut off by the scroll limit. I am unable to reproduce it now |
| 21:41.06 | kfife | Syntax correct? Dial(SIP/${out},99,D(7123#)) - Called party gets only one DTMF tone |
| 21:41.18 | nnY_2 | jblack yeah not sure how CID could just stop working either |
| 21:41.28 | kfife | Dial(SIP/${out},99,D(7123343434)) - Also gets just one little tone! |
| 21:41.36 | kfife | Why is that?? Any ideas? |
| 21:41.52 | jblack | kfife: Nope. I never could get D() to work |
| 21:42.31 | kfife | jblack: But you had the same problem? |
| 21:42.59 | jblack | similiar, yes. |
| 21:43.13 | jblack | it's been about six months since I tried, though |
| 21:43.50 | kfife | Also odd: Dial(SIP/${out},99,D(7123#:7123#)) the called party gets one tone, the callING party gets nothing. |
| 21:44.42 | kfife | Same for D(:7123#) even D(:71) -- callING party gets nothing |
| 21:44.45 | nnY_2 | jblack ha http://pastebin.com/m2975b99b |
| 21:45.01 | nnY_2 | [TK]D-Fender http://pastebin.com/m2975b99b was a test call letting asterisk use asterisk for unknown CID |
| 21:45.13 | nnY_2 | it also replaces the user field with asterisk |
| 21:46.28 | jblack | That looks like a working, but rejected, call to me |
| 21:46.49 | nnY_2 | yeah i hung up |
| 21:46.54 | nnY_2 | noone there to answer |
| 21:58.14 | adrianXXX | someone can tell me who can y enabled the colors on asterisk console ? |
| 21:59.22 | jblack | you can. |
| 22:03.06 | aliver | how do I turn off the output from "sip debug" ? |
| 22:03.13 | nnY_2 | core set debug 0 |
| 22:03.20 | nnY_2 | er |
| 22:03.20 | aliver | thanks |
| 22:03.21 | nnY_2 | no |
| 22:03.23 | nnY_2 | sorry |
| 22:03.26 | aliver | k |
| 22:03.30 | nnY_2 | sip set debug peer (name) |
| 22:03.34 | nnY_2 | toggles it on and off |
| 22:04.01 | nnY_2 | aliver man don;t listen to me |
| 22:04.03 | nnY_2 | i spread lies |
| 22:04.10 | aliver | I have to turn it off per-peer after turning it on with "sip debug"? |
| 22:04.14 | nnY_2 | sip set debug off |
| 22:04.20 | nnY_2 | nah |
| 22:04.20 | aliver | thanks |
| 22:04.24 | aliver | ???? |
| 22:04.34 | nnY_2 | that turns it off lol |
| 22:04.42 | nnY_2 | I was wrong the first two times.. been along day |
| 22:04.53 | nnY_2 | a long* |
| 22:04.54 | kfife | jblack: D() appears to be a DTMF issue. If I change my dtmfmode paramter to inband or info I get multiple tones. Bugs not yet worked out, but FYI |
| 22:05.30 | nnY_2 | goes to look up what D() does |
| 22:05.47 | jblack | supposed to dial digits after a call is established. |
| 22:05.49 | kfife | it's a flag for the Dial() applicatoin |
| 22:05.55 | nnY_2 | ooh that's useful |
| 22:06.09 | jblack | be more useful if it worked with rfc2833 |
| 22:06.12 | nnY_2 | i can call someone and send funky town to them |
| 22:06.27 | nnY_2 | ooh joy dtmf issues.. yeah i have plenty of those |
| 22:06.39 | nnY_2 | mostly with our itsp, but it's getting better |
| 22:06.50 | kfife | jblack: indeed, why doesn't it? |
| 22:06.56 | kfife | why dozen tit? |
| 22:07.06 | jblack | because kittens are cute. |
| 22:07.13 | kfife | I see |
| 22:07.16 | nnY_2 | how do we test this theory? |
| 22:07.21 | *** join/#asterisk wizard171 (n=wizard17@dialup-4.226.111.208.Dial1.Dallas1.Level3.net) |
| 22:07.35 | kfife | because 6 pair is better than 1? |
| 22:07.39 | jblack | by drowning every cute kitten in the world, and trying D() with rfc2833. |
| 22:08.05 | nnY_2 | is there an RFC for the standard of "cute" |
| 22:08.22 | nnY_2 | i mean, we just can't kill them all, and hope we got the cute ones |
| 22:08.25 | jblack | no. it's like pornography. It can't be defined, but you know it when you see it. |
| 22:08.28 | kfife | is that an asterisk limitation or do I need to complain to my ITSP? |
| 22:08.30 | jblack | sure we can. |
| 22:08.30 | nnY_2 | hah |
| 22:08.48 | nnY_2 | can you do it manually? |
| 22:08.56 | jblack | Take out all the chihuahuas while you're at it. |
| 22:09.23 | jblack | The purpose is to remove all the kittens for proper 2833 testing. The method to remove them is immaterial. |
| 22:09.28 | nnY_2 | maybe i can combine the efforts, load the chihuahuas with explosives and have them seek the kittens |
| 22:09.28 | MooingLemur | yokie arrow taco bell |
| 22:09.59 | jblack | Better yet, load explosives into the chihuahas, then load the chihuahas into the kittens. |
| 22:10.14 | nnY_2 | like a kitten chalupa! |
| 22:10.15 | jblack | you'll save money on det caps that way |
| 22:10.28 | jblack | lol |
| 22:11.28 | nix8n82 | no.no. just pass a law giving tax rebate if the chinese use kitty and mexican a rebate for chihuahuas, and it all works out in the end either you eat a little pussy or choke on a weiner |
| 22:11.55 | *** join/#asterisk ibercom (i=581c6376@gateway/web/ajax/mibbit.com/x-9b4d70826e028d84) |
| 22:11.58 | jblack | Offer a free iphone for each kitty corpse. That'll do it in short order. |
| 22:12.44 | nix8n82 | yeah that might be quicker |
| 22:12.55 | nnY_2 | my irc fu is weak, any way to log out my other account ? |
| 22:13.11 | jblack | identify to it with nickserv, and ghost it. |
| 22:14.38 | nnY_2 | well ok |
| 22:16.14 | *** join/#asterisk nnY_2 (n=nny_1@64.203.237.47) |
| 22:17.01 | nnY_2 | ~centos52bug |
| 22:17.02 | jbot | [~centos52bug] There is a bug compiling Zaptel up to 1.4.11/1.2.26 on CentOS/RHEL 5.2: error in xpp/xdefs.h. Workaround: uncheck "xpp" (Astribank) driver in menuselect. Fixed in SVN. Patches: http://bugs.digium.com/12889, or trouble compiling x86_64 packages? Report a bug to CentOS! ".i386 packages should not satisfy dependencies for .x86_64 packages." |
| 22:17.13 | nnY_2 | interesting |
| 22:19.59 | *** topic/#asterisk by Qwell -> Asterisk: The Open Source PBX and Telephony Platform (asterisk.org) -=- Zaptel or 64-bit compile troubles:type ~centos52bug -=- Asterisk 1.6.0-beta9 (2008/05/14), 1.4.21.2 (2008/07/22), *-Addons 1.6.0-beta4 (2008/06/04), 1.4.7 (2008/06/04), Zaptel 1.4.11 (2008/05/28), Libpri 1.4.7 (2008/08/05) -=- Related channels: #asterisknow #asterisk-gui #switchvox #freepbx #asterisk-commits #asterisk-bugs #asterisk-dev |
| 22:20.27 | jblack | what just bumped? |
| 22:20.35 | *** join/#asterisk lesouvage (n=lesouvag@cc341200-a.assen1.dr.home.nl) |
| 22:20.44 | *** topic/#asterisk by Qwell -> Asterisk: The Open Source PBX and Telephony Platform (asterisk.org) -=- CentOS 5.2 Zaptel or x86_64 compile troubles:type ~centos52bug -=- Asterisk 1.6.0-beta9 (2008/05/14), 1.4.21.2 (2008/07/22), *-Addons 1.6.0-beta4 (2008/06/04), 1.4.7 (2008/06/04), Zaptel 1.4.11 (2008/05/28), Libpri 1.4.7 (2008/08/05) -=- Related channels: #asterisknow #asterisk-gui #switchvox #freepbx #asterisk-commits #asterisk-bugs #asterisk-dev |
| 22:20.55 | Qwell | adding another reason to type centos52bug |
| 22:21.44 | jblack | the best fix for that centos bug is to replace centos |
| 22:22.00 | Qwell | or get them to finally acknowledge that it's a bug, and do something about it |
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| 22:30.42 | lesouvage | I just read www.argreenhouse.com/papers/stanm/sip-iptel2001.pdf a document about network enabled devices others then phones. Like a sip enabled microwave, frig, doorlocks, lightswitch, garage door, alarm to water a bananatree etc. Are there any reallife examples of sip enabled devices other then phones that can be used in combination with Asterisk? |
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| 22:55.48 | aliver | Is there a global variable in extensions.conf for the caller id of the person initiating the call? |
| 22:56.11 | Qwell | it would be pretty useless if it was global |
| 22:56.36 | aliver | Funny, 'cause I'd sure like to use it. |
| 22:57.00 | Qwell | it would change out from under you all the time, and wouldn't relate to the call at all |
| 22:57.03 | aliver | or maybe you are taking issue with my misuse of global? |
| 22:57.22 | Qwell | ${CALLERID(name)} or num |
| 22:58.23 | aliver | Okay, pedantic-pants, let me rephrase. Is there a variable of any kind, sort, or incarnation that represents the caller id of the caller or anything that might be construed as such by a person not as wise, worldly and supremely intelligent as yourself? |
| 22:58.36 | Qwell | see above |
| 22:58.45 | aliver | Thanks! |
| 22:58.59 | Qwell | with technology, you have to be pedantic. |
| 22:59.21 | Qwell | If you actually wanted a global callerid variable, the answer would be very different |
| 22:59.56 | Qwell | (it would be useless, but possible nonetheless) |
| 23:00.21 | aliver | I understand I'm just being pissy, never mind my PMS. |
| 23:00.26 | aliver | and thanks. |
| 23:01.20 | lesouvage | Qwell: You could use it as a kind of mechanism for a lottery to pick the winner. |
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| 23:13.20 | irieKen | Anyone know why Asterisk, on its own, would append new entries to the zapata.conf file? |
| 23:15.13 | jaytee | Asterisk? no. Asterisk with Asterisk-GUI? yeah. because it's retarded |
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| 23:28.40 | irieKen_ | Jaytee: sorry, got disconnected. |
| 23:29.07 | jaytee | no need to apologize, you're not on my payroll :-) |
| 23:30.01 | Gershwin | what skills would one need to be put on your payroll? |
| 23:30.29 | jaytee | the ability to work miracles for free mostly since I couldn't afford to pay anyone :-) |
| 23:30.41 | Gershwin | points to his shiny shoes, his black paste and well-worn shoe brush. |
| 23:31.13 | Gershwin | so much for your work force :/ |
| 23:31.41 | jaytee | Gershwin, perhaps with that particular skill set and accoutrements you should seek employement as a salesman at Foot Locker or Payless? |
| 23:33.37 | jaytee | I'm the only one on my payroll and I'm less than satisfied with my performance. I'm afraid I'm going to have to give myself a less than satisfactory rating when I write my annual review. |
| 23:35.15 | Gershwin | Let's hope that this doesn't warrant a demotion or a drop in your pay scale |
| 23:35.16 | jaytee | excellent, only 43 minutes remaining on my 79GB image backup of my dual boot CentOS server / Xubuntu web kiosk system. |
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| 23:46.09 | dlynes | Does anyone know what OpenSER is called now? Is it Kamailio, or OpenSIPS? |
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| 23:50.29 | jaytee | dlynes, I believe it is now called Kamailio but I'm also confused about that because I've seen the OpenSIPS site come up in searches too |
| 23:51.25 | _ShrikE | jaytee: openser was forked into both |
| 23:51.25 | jaytee | why both? |
| 23:51.37 | _ShrikE | politics I guess.. so they went two directions |
| 23:52.39 | jaytee | jeez, hellofa netsplit goin on |
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| 23:52.43 | _ShrikE | aye |
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| 23:54.51 | ectospasm | grrr... I hate Trixbox |
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| 23:56.07 | metfan2007 | hi all |
| 23:56.21 | jaytee | ectospasm, well you'll probably find little sympathy here for your angst |
| 23:56.41 | jaytee | venting in #trixbox might be the better venue |
| 23:56.48 | dlynes | So basically half the amount of effort in each |
| 23:56.58 | metfan2007 | I get a new TE121 card, and I'm trying to make it works in a new server, but every time I try to start zaptel I get an "ZT_SPANCONFIG failed on span 1: Invalid argument (22)" |
| 23:57.03 | dlynes | I wonder which one to follow, then |
| 23:57.05 | metfan2007 | any idea? pls |
| 23:58.46 | ectospasm | jaytee: it's not me that's using trixbox, it's a customer of mine! |
| 23:59.01 | jaytee | ectospasm, more's the pity then |
| 23:59.03 | ectospasm | and I'm just venting, nothing more |
| 23:59.23 | ectospasm | long day... still got an hour left of work. |
| 23:59.59 | dlynes | metfan2007: you don't have the device driver loaded, more than likely |
