| 00:01.45 | flenders | knarfly: all tests I did with linksys phones were fine with speakerphones and all, but on production, we had lots of complaints, so changed the ones on boardrooms for polycoms IP430s and people are loving it |
| 00:03.41 | *** join/#asterisk darkskiez (n=mhb@cpc1-broo3-0-0-cust659.renf.cable.ntl.com) |
| 00:04.12 | nny | ERROR[4320] chan_zap.c: Unable to open channel 3: No such device or address |
| 00:04.12 | nny | here = 0, tmp->channel = 3, channel = 3 |
| 00:04.16 | nny | is a better error |
| 00:05.49 | knarfly | nny: that looks okay...and you should have to create anything...you're running Linux |
| 00:06.00 | nny | yeah i thought so |
| 00:06.22 | nny | knarfly: why the missing device errors in messages though?? |
| 00:06.23 | knarfly | and you do have two lines into this machine? |
| 00:06.28 | nny | yes |
| 00:07.07 | knarfly | I run FreeBSD...different kind of animal...but still you shouldn't have parts of /var disappearing...that's beyond me |
| 00:07.31 | knarfly | nny what happens when you run ztcfg? |
| 00:07.39 | knarfly | or zttool? |
| 00:07.50 | nny | says 2 channels configured |
| 00:08.08 | nny | 03 and 04 |
| 00:08.10 | knarfly | did you try pissing on a spark plug yet...! |
| 00:08.14 | nny | hehehe |
| 00:08.18 | nny | thats after i am done |
| 00:08.33 | knarfly | it should work...but I'm not really an expert on zaptel yet |
| 00:09.48 | mocker | sigh, my dialplan just got a ton more complicated because of damn softphones. |
| 00:10.27 | *** join/#asterisk kFuQ (n=somedude@c-67-185-112-20.hsd1.wa.comcast.net) |
| 00:10.47 | knarfly | mocker: well we all have our little problems....what softphone, X-Lite? |
| 00:11.26 | mocker | The software doesn't matter, the fact that people exit out of them matterse. |
| 00:11.29 | mocker | +spelling |
| 00:11.52 | mocker | When they exit, my dundi setup doesn't see that as a valid extension anymore kills the call. |
| 00:11.54 | knarfly | yes, but remember if it weren't for users...they wouldn't need techs |
| 00:12.02 | *** part/#asterisk atomicd (n=Atomicd@74-206-0-80.static-ip.m.telepacific.net) |
| 00:16.08 | *** join/#asterisk grimsy (n=chatzill@203-206-220-214.perm.iinet.net.au) |
| 00:21.16 | *** part/#asterisk nny (n=scott@64.203.239.83.static-pool-4.pool.hargray.net) |
| 00:30.09 | *** join/#asterisk nny (n=scott@64.203.239.83.static-pool-4.pool.hargray.net) |
| 00:30.39 | nny | can anyone tell me why /var/run/asterisk directory is not being created upon boot? (And why it is so &*@&#*( hard to secure asterisk) |
| 00:32.34 | nny | *in ubuntu at least* |
| 00:33.37 | nny | init script is apparenlty garbage |
| 00:35.48 | wiljacket | nyy: are you running feisty? |
| 00:36.13 | wiljacket | I remember having that problem with edgy |
| 00:39.07 | nny | nah 6.06 |
| 00:39.20 | nny | problem is the init.d scripts make config installs |
| 00:39.24 | nny | they suck big time |
| 00:39.34 | nny | using good ones from another server fixed a lot |
| 00:39.43 | nny | including using init.d to unload zaptel |
| 00:42.54 | drwelby | For a users.conf User, can you have hassip=yes and hasiax=yes and have both a sip phone and and iax phone use the same username? |
| 00:43.04 | drwelby | Or are the two mutually exclusive |
| 00:43.09 | wiljacket | hehe, trouble with 6.06 is what turned me over to just compiling * from cvs/source to get rid of all of the crap in the configs.. but if you need to work in ubuntu and apt, I hear good things about feisty and asterisk |
| 00:43.29 | wiljacket | that init script stuff is at least taken care of |
| 00:43.58 | *** join/#asterisk zeedo (n=zeedo@reboot-robot.net) |
| 00:44.43 | nny | i am using source on 6.06 |
| 00:44.46 | nny | it's a long story |
| 00:45.33 | nny | beer time.. short version.. init.d scripts made with make config don't create /var/run/asterisk dir for running as non-root AND even the init.d scripts for zaptel break.. used some scripts from another server i have, all is wonderful |
| 00:45.54 | nny | zaptel init.d script now even unloads modules |
| 00:45.58 | nny | tomorrow I post on wiki |
| 00:46.03 | nny | tonight I create hangover.so |
| 00:46.06 | _ShrikE | chan_lager.so |
| 00:46.09 | nny | lol |
| 00:46.14 | nny | later all |
| 00:46.20 | *** part/#asterisk nny (n=scott@64.203.239.83.static-pool-4.pool.hargray.net) |
| 00:47.15 | *** join/#asterisk famicon (i=scenesta@c51447ddc.cable.wanadoo.nl) |
| 00:53.20 | *** join/#asterisk brut- (n=brut@66.173.4.254) |
| 01:01.48 | *** join/#asterisk Raky-2 (n=John@220.157.75.246) |
| 01:03.13 | *** join/#asterisk asdx (n=diego@adsl-159-133.click.com.py) |
| 01:19.43 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 01:20.11 | mocker | Anyone know if there is a way to check if an extension exists? Sorta like ChanIsAvail but for an extension? |
| 01:23.06 | [TK]D-Fender | mocker, ChanIsAvail <- |
| 01:25.53 | mocker | [TK]D-Fender: But sometimes an extension is not a channel.. |
| 01:26.03 | [TK]D-Fender | mocker, And how is that? |
| 01:26.04 | mocker | if that makes sense.. |
| 01:27.00 | mocker | [TK]D-Fender: If I have an extension that just goes to say an IVR. |
| 01:27.16 | [TK]D-Fender | mocker, And yes I understood right from the start exactly what you're looking for and I'm hoping that your thinking about what KIND fo channel can represent that for the purpose of using that app :) |
| 01:27.41 | mocker | Hmm. |
| 01:27.51 | mocker | I tried Local/${EXTEN} |
| 01:27.55 | mocker | Thinking that might do it.. |
| 01:28.46 | [TK]D-Fender | ;) |
| 01:28.57 | [TK]D-Fender | getting warmer! |
| 01:29.08 | [TK]D-Fender | ALMOST there! |
| 01:30.24 | mocker | crap! |
| 01:30.38 | mocker | I'm hitting a wall I guess. |
| 01:32.25 | mocker | Well, it wasn't Zap... |
| 01:32.29 | mocker | But I didn't think it would be. |
| 01:32.35 | mocker | [TK]D-Fender: I'll get you for this. :) |
| 01:33.37 | [TK]D-Fender | mocker, here : ChanIsAvail(Local/12345@context/n) <---------- "/n" is literal. |
| 01:34.33 | mocker | Ohh, damn contexts! |
| 01:34.50 | Raky-2 | hey, that's a pretty cool feature. |
| 01:36.03 | mocker | [TK]D-Fender: Thanks, I'll try that. |
| 01:36.05 | *** join/#asterisk sysreq (n=sysreq@unaffiliated/sysreq) |
| 01:37.43 | *** join/#asterisk techie (n=techie@adsl-76-240-178-163.dsl.lsan03.sbcglobal.net) |
| 01:37.46 | *** join/#asterisk Dalbaech (n=Dalbaech@c-98-200-244-16.hsd1.tx.comcast.net) |
| 01:37.54 | Dalbaech | has anyone played with alarmreceiver much? |
| 01:38.12 | *** part/#asterisk |R (i=bob@modemcable241.28-203-24.mc.videotron.ca) |
| 01:38.15 | mocker | [TK]D-Fender: That worked. |
| 01:38.20 | mocker | [TK]D-Fender: You're the man. :) |
| 01:38.23 | [TK]D-Fender | mocker, You're welcome :) |
| 01:38.56 | mocker | [TK]D-Fender: Do you ever read/post on asterisk-useres? |
| 01:38.59 | mocker | +spelling |
| 01:39.06 | [TK]D-Fender | mocker, not usually |
| 01:39.50 | mocker | I talked to Jared Smith@digium about creating an asterisk-dcap list |
| 01:40.02 | mocker | Something with a little less volume. |
| 01:41.12 | [TK]D-Fender | mocker, I'm sure it'll find some kind of following.. |
| 01:41.29 | *** join/#asterisk droops (n=droops@adsl-074-245-001-031.sip.jan.bellsouth.net) |
| 01:41.44 | mocker | Never do major dialplan changes w/o regression testing. |
| 01:41.50 | mocker | saves my ass every time. |
| 01:43.10 | mocker | j/k |
| 01:43.34 | *** join/#asterisk rmayorga (n=rmayorga@unaffiliated/rmayorga) |
| 01:44.10 | Qwell | mocker: feel free to write test scripts and submit them |
| 01:44.12 | russellb | :-p |
| 01:44.19 | Qwell | russellb: see my email? |
| 01:44.23 | russellb | Qwell: not yet, no |
| 01:44.31 | mocker | Hah, I was at Astricon and it seemed to be a pretty hot topic. :-) |
| 01:44.38 | mocker | Hence the j/k |
| 01:44.41 | Qwell | just the mac address question earlier.. answer is yes, you do need it added |
| 01:44.58 | russellb | ah ha, thanks |
| 01:45.06 | Qwell | had to get mine added too, so did Kevin |
| 01:45.09 | russellb | mocker: hehe, yeah, it's all good |
| 01:45.26 | russellb | mocker: but seriously, i understand that it's something people want us to do |
| 01:45.27 | Qwell | russellb: I'd like to get a PoE switch for our private LAN, so we can do all of that stuff there |
| 01:45.29 | russellb | i have been thinking about it a good bit |
| 01:45.41 | *** join/#asterisk riddlebox (n=james@75-132-225-75.dhcp.stls.mo.charter.com) |
| 01:45.45 | russellb | Qwell: sounds like a good plan |
| 01:45.49 | russellb | Qwell: would save us a lot of pain |
| 01:45.52 | Qwell | indeed |
| 01:46.04 | Qwell | I mean, we don't "need" PoE, but...yeah :D |
| 01:46.39 | mocker | russellb: Has Jared's community project gotten much traction yet? |
| 01:46.45 | mocker | (I imagine he's been pretty busy) |
| 01:46.56 | russellb | mocker: not a lot yet, i think |
| 01:47.03 | russellb | i added a couple scripts to start it off :) |
| 01:47.10 | russellb | and made it generate a pdf, too |
| 01:47.10 | russellb | heh |
| 01:48.11 | mocker | [TK]D-Fender: I'm having to rewrite my dialplan because of softphones. :( |
| 01:48.29 | [TK]D-Fender | mocker, and why is that? |
| 01:49.08 | mocker | I had a great setup where sip registrations would be dynamically added to my dialplan by registering to a context. |
| 01:49.16 | mocker | Works great w/ hardphones. |
| 01:49.29 | [TK]D-Fender | mocker, regexten.... BLEH |
| 01:49.43 | mocker | Because the extension is always there.. Softphone laptop gets closed, and then that extension no longer exists! |
| 01:50.15 | mocker | It was great until they started giving me telecommuters! |
| 01:50.17 | mocker | :) |
| 01:50.26 | symlink | I hate you Rogers, I really really hate you |
| 01:50.40 | russellb | mocker: it's ok, i really like regexten :) |
| 01:51.15 | mocker | I like it too, I just wasn't planning on extensions disappearing. |
| 01:51.39 | mocker | Strangely people want their voicemail box to pick up even if their laptop is off. |
| 01:51.44 | mocker | :P |
| 01:51.47 | russellb | oh, heh, oops |
| 01:51.50 | russellb | get it fixed? |
| 01:52.05 | mocker | russellb: Yeah, but it's not as pretty as it used to be. |
| 01:52.07 | mocker | ;) |
| 01:52.49 | russellb | but the cool thing is, it was possible, right? ;) |
| 01:52.54 | Raky-2 | hey guys |
| 01:53.08 | russellb | asterisk dialplan is a weird world. |
| 01:53.12 | russellb | Raky-2: greetings |
| 01:53.18 | Raky-2 | any idea where the setting is, to always allow SIP traffic to go through the server. |
| 01:53.39 | mocker | russellb: Yeah, love that it's possible. :) |
| 01:54.15 | russellb | Raky-2: sounds like you're looking for the "allowguest=yes" option in the [general] section of sip.conf |
| 01:54.20 | mocker | russellb: Eventually I'll need to learn AEL so I can actually have indenting that makes sense! |
| 01:54.57 | Raky-2 | well basically, what i mean is |
| 01:55.15 | Raky-2 | currently when i make a direct call to someone, it will use the bandwidth at each user's connection |
| 01:55.20 | Raky-2 | however, when i connect to a conference |
| 01:55.28 | Raky-2 | it will use the server's bandwidth |
| 01:55.37 | Raky-2 | i was wondering if there's a way to always route SIP traffic to use the server's bandwidth |
| 01:55.40 | Raky-2 | if that makes any sense, haha. |
| 01:56.00 | mocker | Raky-2: canreinvite ? |
| 01:57.00 | Raky-2 | i currently have canreinvite set to no, let me read up on it. |
| 01:57.01 | Raky-2 | thanks heaps! |
| 01:57.31 | *** join/#asterisk Freman (n=freman@brdr-gw-01.benon.com) |
| 01:57.41 | Freman | heyas, got my hands on a couple of polycom phones |
| 01:57.49 | Freman | what firmware/bootrom should I be on? |
| 01:58.08 | Freman | oh... and how in blazes to I get the latest if needed? |
| 01:58.16 | Raky-2 | the problem which i'm having mocker |
| 01:58.27 | mocker | Freman: Polycom authorized reseller. :( |
| 01:58.30 | Raky-2 | is that i have two * machined trunked together |
| 01:58.35 | mocker | Freman: They don't just let you download the latest firmware. |
| 01:58.49 | Raky-2 | when i make local calls so let's say on Box.A to Box.B it works fine. |
| 01:58.58 | Raky-2 | when i make calls from Box.A to Box.B it's scrambled. |
| 01:59.10 | Raky-2 | however, if i connect from Box.A to Box.B-conference, the quality is fine? |
| 01:59.29 | mocker | Raky-2: Define scrambled? |
| 01:59.43 | Raky-2 | demonic. |
| 01:59.49 | Raky-2 | i often tell them they sound like satan. |
| 02:00.50 | mocker | Have you unloaded channel_satan.so? |
| 02:00.57 | Raky-2 | hahaha. |
| 02:01.32 | mocker | I guess what I would check is the codec being used across all the boxes to ensure you aren't transcoding 50 times. |
| 02:01.45 | mocker | disallow=all, allow=gsm |
| 02:01.49 | mocker | Or something like that. |
| 02:02.02 | mocker | Also look for any weird messages in your console. |
| 02:02.03 | Raky-2 | yeah, the machines are connected over gsm. |
| 02:02.15 | mocker | And the endpoints? |
| 02:02.22 | Raky-2 | Box.A uses alaw, Box.B uses ulaw |
| 02:02.45 | Raky-2 | but even if i use gsm across the board, it still has that sound. |
| 02:02.51 | mocker | Weird. |
| 02:03.29 | *** part/#asterisk kiwoneka (n=kiwoneka@KTNRON06-1168103823.sdsl.bell.ca) |
| 02:05.39 | mocker | Do you have a hardware clock on both boxes? |
| 02:05.44 | mocker | Or are you using ztdummy? |
| 02:05.48 | Raky-2 | ztdummy |
| 02:07.27 | *** join/#asterisk WilliamK (n=noc@static-71-170-144-3.dllstx.fios.verizon.net) |
| 02:07.42 | mocker | Raky-2: 2.6 kernel? |
| 02:08.09 | Raky-2 | yes |
| 02:08.36 | Raky-2 | one is 2.6.18, the other 2.6.21.5-smp |
| 02:09.09 | mocker | And nothing strange comes across the console? |
| 02:10.00 | *** join/#asterisk LakeSolon (n=blake@12-202-202-168.client.mchsi.com) |
| 02:10.24 | Raky-2 | nothing at al |
| 02:10.25 | Raky-2 | *all |
| 02:11.10 | mocker | Raky-2: Try running zttest.. |
| 02:11.29 | *** join/#asterisk Kumbang (n=kumbang@router-wmi.paume.itb.ac.id) |
| 02:17.49 | *** join/#asterisk bjohnson (n=bjohnson@67.212.10.134) |
| 02:23.40 | *** join/#asterisk blinky42 (n=me@c-71-230-47-244.hsd1.pa.comcast.net) |
| 02:24.41 | *** join/#asterisk outtolunc (n=me@c-67-174-216-60.hsd1.ca.comcast.net) |
| 02:39.03 | *** join/#asterisk famicon (i=redz@c51447ddc.cable.wanadoo.nl) |
| 02:48.05 | mocker | ~grandstreawm |
| 02:48.10 | mocker | ~grandstream |
| 02:48.11 | jbot | rumour has it, grandstream is the Yugo of VoIP hardware. Run. Run away now. |
| 02:51.21 | *** join/#asterisk sammy__98 (n=sacha@CPE004005521e21-CM00159a08ffe0.cpe.net.cable.rogers.com) |
| 02:51.28 | Dalbaech | hehe |
| 02:51.37 | Dalbaech | functional.... |
| 02:51.39 | Dalbaech | but ugly. |
| 02:52.12 | TrentCreek | fugly |
| 02:54.07 | [TK]D-Fender | Grandstream.... putting the "fun" back into Dysfunctional! |
| 02:54.19 | *** join/#asterisk mihinomenest (n=argh@66.255.220.22) |
| 02:54.36 | *** join/#asterisk waKKu (n=worth@unaffiliated/wakku) |
| 03:00.32 | J4k3 | ~gs |
| 03:00.32 | jbot | GrandSuck phones & gateways are cheap junk which should be avoided with extreme prejudice. |
| 03:00.40 | J4k3 | alas, my budgetone 101's work great |
| 03:00.45 | mocker | Anyone have a FWD account they can try to place a test call for me? |
| 03:00.49 | J4k3 | just remember they're half the price of a good phone |
| 03:00.52 | J4k3 | so expect half the performance |
| 03:00.58 | J4k3 | if you do that, you'll be very happy with the bt101 |
| 03:01.11 | J4k3 | as for the more expensive units, dunno... if I was spending real money, I'd buy a real phone. |
| 03:01.29 | riddlebox | J4k3, that is true, the GXP-2000 works great, until you use the speaker phone |
| 03:01.49 | J4k3 | you can get low end polycoms for the price of the 2000 (in the USA at least) |
| 03:04.03 | *** join/#asterisk CaRb0n^ (n=Omer@203.81.206.225) |
| 03:07.58 | *** join/#asterisk MdeP (n=mdep@200.125.91.223) |
| 03:09.26 | Dalbaech | yea mocker |
| 03:09.27 | Dalbaech | what's up? |
| 03:11.10 | mocker | Dalbaech: Just testing my setup. |
| 03:11.32 | mocker | Can you try to call 720298? |
| 03:13.04 | Dalbaech | <PROTECTED> |
| 03:13.13 | Dalbaech | <PROTECTED> |
| 03:13.39 | mocker | huh. |
| 03:13.45 | Dalbaech | that's the responses |
| 03:13.46 | Dalbaech | heh |
| 03:13.50 | Dalbaech | when calling |
| 03:13.52 | Dalbaech | try calling |
| 03:13.58 | Dalbaech | 760962 |
| 03:14.02 | mocker | I can't, I'm not at home where the box is.:) |
| 03:14.06 | Dalbaech | haha |
| 03:14.15 | Dalbaech | ok; I get a timeout when calling you |
| 03:14.18 | *** join/#asterisk PepOSX (n=pepOSX@190.72.149.12) |
| 03:14.44 | *** join/#asterisk sysreq (n=sysreq@unaffiliated/sysreq) |
| 03:15.20 | *** join/#asterisk saftsack (n=saftsack@pD9E0460B.dip.t-dialin.net) |
| 03:15.40 | *** join/#asterisk stkn__ (n=stkn@gentoo/developer/pdpc.active.stkn) |
| 03:17.25 | Dalbaech | ok; calling a FWD test # works |
| 03:17.28 | Dalbaech | so it's something with your setuip |
| 03:17.31 | Dalbaech | *setup |
| 03:17.56 | mocker | Dalbaech: Oh, I was pretty sure of that. :) |
| 03:18.04 | Dalbaech | well, I wasn't. |
| 03:18.04 | Dalbaech | hehe |
| 03:18.12 | Dalbaech | I hadn't been sending anything outbound to FWD in a LONG time |
| 03:18.12 | Dalbaech | hehe |
| 03:18.18 | Dalbaech | I had to add a new outgoing rule for it |
| 03:18.19 | Dalbaech | :P |
| 03:18.25 | mocker | heh, awesome. |
| 03:18.36 | Dalbaech | I use SIPbroker to do most of the dialing, except the PAP2's truncate *XXXXX to *XX |
| 03:18.36 | Dalbaech | hehe |
| 03:19.58 | [TK]D-Fender | Dalbaech, go fix its dialplan then. |
| 03:20.14 | Dalbaech | nah; I live without it for now. |
| 03:20.14 | Dalbaech | heh |
| 03:20.24 | Dalbaech | I mainly did it so I can use incoming to be sent (a sipbroker gateway) |
| 03:24.25 | mocker | Dalbaech: Can you try one more time? |
| 03:24.32 | Dalbaech | yea |
| 03:25.02 | Dalbaech | ummm |
| 03:25.03 | Dalbaech | busy? |
| 03:25.03 | Dalbaech | heh |
| 03:25.04 | Dalbaech | <PROTECTED> |
| 03:25.04 | Dalbaech | <PROTECTED> |
| 03:25.04 | Dalbaech | <PROTECTED> |
| 03:25.04 | Dalbaech | <PROTECTED> |
| 03:25.29 | mocker | heh. |
| 03:25.32 | mocker | great. |
| 03:26.30 | *** join/#asterisk mordaunt (n=mordaunt@unaffiliated/mordaunt) |
| 03:29.00 | *** join/#asterisk BBHoss (n=hoss@146.229.191.72) |
| 03:29.26 | Dalbaech | well, if you could decode what the "fast beep", etc meant |
| 03:29.39 | Dalbaech | you could jsut call it through one of my DIDs |
| 03:29.39 | Dalbaech | :P |
| 03:29.51 | Dalbaech | or a sipbroker # |
| 03:30.15 | [TK]D-Fender | perhaps some sip debug on the INBOUND attempt would say something........ |
| 03:30.24 | Dalbaech | true |
| 03:30.45 | mocker | [TK]D-Fender: That's the problem, I'm not seeing any attempts. |
| 03:30.57 | Dalbaech | silly question |
| 03:31.11 | Dalbaech | is your register line proper? |
| 03:31.11 | Dalbaech | heh |
| 03:31.20 | Dalbaech | and is it NAT'd? |
| 03:31.25 | mocker | fwd.pulver.com:5060 720298 105 Registered |
| 03:31.25 | mocker | <PROTECTED> |
| 03:31.41 | Dalbaech | set verbose 99 |
| 03:31.43 | [TK]D-Fender | mocker, nothing with SIP debug enabled? |
| 03:32.07 | [TK]D-Fender | mocker, Indeed... any NAT involved? |
| 03:32.16 | mocker | Yeah, NAT involved.. |
| 03:32.23 | mocker | I set the externip and localnet though. |
| 03:32.26 | [TK]D-Fender | mocker, pastebin your sip.conf masking only passwords |
| 03:32.35 | mocker | I just see register attempts.. |
| 03:32.56 | Dalbaech | X-Asterisk-HangupCause: Unallocated (unassigned) number |
| 03:33.28 | Dalbaech | SIP/2.0 404 Not Found |
| 03:35.17 | Dalbaech | I get not found when calling ya man |
| 03:35.27 | Dalbaech | something's up with the NAT or with the registration |
| 03:35.48 | mocker | http://pastebin.ca/731522 |
| 03:36.18 | [TK]D-Fender | BAD |
| 03:36.23 | [TK]D-Fender | mocker, ....tsk tsk |
| 03:37.25 | [TK]D-Fender | mocker, your extenip & localnet are being IGNORED because you put them AFTER the register statement. You have to do all your settings BEFORE your register statements. You also forgot "canreinvite=no", and "nat=yes". |
| 03:37.26 | mocker | crap, that means it's obvious. |
| 03:37.35 | [TK]D-Fender | Indeed it is. |
| 03:38.00 | Dalbaech | flying monkeys. |
| 03:38.21 | mocker | Monkeys? |
| 03:38.25 | Dalbaech | nevermind. |
| 03:38.42 | mocker | Heh, that's what you're supposed to hear when you call me. :() |
| 03:39.03 | mocker | [TK]D-Fender: What's funny is, the example sip.conf has register before the nat settings. |
| 03:39.48 | Dalbaech | **** happens. |
| 03:39.53 | Dalbaech | just try it |
| 03:39.54 | Dalbaech | :P |
| 03:39.55 | Dalbaech | hehe |
| 03:40.13 | [TK]D-Fender | ~sipnat |
| 03:40.13 | 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 |
| 03:40.14 | [TK]D-Fender | ^^^^^^^^^^^ |
| 03:40.49 | mocker | Dalbaech: Oh, I'm sure he's right. :) |
| 03:40.55 | mocker | He has an annoying habit of that. |
| 03:40.58 | mocker | :P |
| 03:41.53 | Dalbaech | hehe |
| 03:42.51 | mocker | Dalbaech: Can you try again? |
| 03:42.55 | mocker | Made those changes. |
| 03:43.11 | Dalbaech | no dice |
| 03:44.00 | Dalbaech | <-- SIP read from 69.90.155.70:5060: |
| 03:44.01 | Dalbaech | SIP/2.0 404 Not Found |
| 03:44.40 | [TK]D-Fender | mocker, you port forwarding? |
| 03:44.47 | Dalbaech | this is when calling 720298 |
| 03:44.57 | mocker | [TK]D-Fender: Yeah. |
| 03:45.05 | [TK]D-Fender | mocker, what exactly? |
| 03:45.24 | mocker | Lemme install a text mode browser real quick. :) |
| 03:45.27 | hmmhesays | transcoding a cell phone call into g729 sounds like chiat |
| 03:48.08 | *** join/#asterisk Strom_M (n=strom@pool-71-109-0-125.lsanca.dsl-w.verizon.net) |
| 03:48.29 | mocker | [TK]D-Fender: sorry, text mode browsers aren't going to cut it. |
| 03:48.41 | mocker | [TK]D-Fender: I'll revisit it when I'm home, thanks for the ideas though. |
| 03:48.52 | mocker | Sounds like NAT is my issue. |
| 03:49.09 | Dalbaech | indeed |
| 03:49.37 | [TK]D-Fender | mocker, well * should be set up right if you followed my corrections. the rest is up to your router |
| 03:49.54 | mocker | [TK]D-Fender: The weird thing is that I have another ITSP I use just fine w/ sip. |
| 03:50.04 | mocker | vitelity. |
| 03:50.20 | mocker | incoming call goes to tt-monkeys. |
| 03:50.31 | hmmhesays | i use vitelity |
| 03:50.45 | hmmhesays | god today was crazy, rtp streams getting mixed up going to the wrong places |
| 03:50.46 | hmmhesays | n shit |
| 03:50.59 | mocker | really? |
| 03:51.09 | mocker | But that's freaking awesome. |
| 03:51.23 | mocker | I read on the list about someone installing a SIP phone in a bank drive through window. |
| 03:51.45 | mocker | I cringed, hoping he knows that SIP isn't encrypted |
| 03:52.15 | hmmhesays | there are things you can do |
| 03:52.21 | Raky-2 | back |
| 03:52.23 | Raky-2 | sorry, what's zttest? |
| 03:52.27 | *** join/#asterisk bmg505 (n=leon@196.209.179.15) |
| 03:52.30 | mocker | hmmhesays: Yeah, you just have to know to do them. :) |
| 03:53.06 | mocker | wireshark, or, how to scare a new asterisk admin. |
| 03:53.34 | Raky-2 | i'm running it now, on both machines. |
| 03:54.55 | hmmhesays | and it depends if it is a sip phone running across the internet, lan or p2p |
| 03:57.50 | Raky-2 | mocker: what does zttest do? |
| 03:58.00 | Raky-2 | it's just shooting out percentages of accuracy? |
| 03:58.34 | mocker | Raky-2: I have to head home, but basically it checks for IRQ misses on your machine. |
| 03:59.05 | mocker | Is it running clean? |
| 03:59.19 | Raky-2 | yeah seems to be running clean |
| 03:59.19 | *** join/#asterisk AJaymn (n=mypocket@71-82-218-158.dhcp.mdsn.wi.charter.com) |
| 03:59.31 | Raky-2 | started tih 99.98% averaging around 99.95% |
| 03:59.32 | mocker | percentages? |
| 03:59.53 | Raky-2 | Best: 100.000000 -- Worst: 99.902344 -- Average: 99.953306 |
| 04:00.01 | Raky-2 | Best: 99.975586 -- Worst: 99.792480 -- Average: 99.957579 |
| 04:00.04 | Raky-2 | those are the two machines. |
| 04:00.37 | BBHoss | i hate freepbx |
| 04:00.39 | mocker | Those aren't great scores, but you might shoot your problem out again. |
| 04:00.52 | mocker | Raky-2: [TK]D-Fender is back. :) |
| 04:01.04 | Raky-2 | yooo BBHoss |
| 04:01.08 | mocker | G'night. |
| 04:01.08 | BBHoss | sup |
| 04:01.23 | *** join/#asterisk CunningPike (n=CunningP@S010600095b33697f.vc.shawcable.net) |
| 04:03.27 | *** join/#asterisk SexyKen (n=sexy@c-24-4-238-80.hsd1.ca.comcast.net) |
| 04:03.37 | SexyKen | Anyone know how to make the IP 600 connect to multiple SIP servers? |
| 04:03.44 | Dalbaech | anyone have a fax available? |
| 04:03.45 | Dalbaech | hehe |
| 04:04.17 | fujin | oh god |
| 04:04.23 | fujin | fax=sip=baadddddd |
| 04:04.25 | hmmhesays | yay faxes |
| 04:04.34 | Dalbaech | I really haven't had problems |
| 04:04.40 | hmmhesays | between like hardware running the same firmware releases its fine |
| 04:04.44 | hmmhesays | any other time, hells no |
| 04:04.44 | Dalbaech | it's a box with a REALLY good connection |
| 04:05.04 | hmmhesays | I fought for weeks with a shitty dsl connection |
| 04:05.11 | Dalbaech | hehe |
| 04:05.20 | Dalbaech | well, between the telco and my box.... there's.... |
| 04:05.32 | Dalbaech | 6ms |
| 04:05.58 | Dalbaech | I think i'm going to go do that sleep thing.... |
| 04:06.04 | Dalbaech | feel free to send me SPAM fax |
| 04:06.05 | Dalbaech | hehe |
| 04:06.20 | Dalbaech | 2025210428 |
| 04:06.21 | Dalbaech | :) |
| 04:09.07 | Dalbaech | yay... fax. |
| 04:09.07 | Dalbaech | hehe |
| 04:14.58 | *** join/#asterisk craigk (n=ckowald@58.174.122.198) |
| 04:16.36 | Dalbaech | good night all |
| 04:24.41 | *** join/#asterisk denon (n=denon@208.122.43.201) |
| 04:24.41 | *** mode/#asterisk [+o denon] by ChanServ |
| 04:30.21 | *** join/#asterisk techie (n=techie@adsl-76-240-178-163.dsl.lsan03.sbcglobal.net) |
| 04:35.28 | *** join/#asterisk |omni| (n=rob@c-67-185-56-106.hsd1.wa.comcast.net) |
| 04:35.34 | *** join/#asterisk PepOSX (n=pepOSX@190.72.149.12) |
| 04:36.51 | *** join/#asterisk DaveCanoe (n=Dave@canoe404.dclg.ca) |
| 04:40.48 | *** join/#asterisk lunaphyte__ (n=lunaphyt@0158ahost161.starwoodbroadband.com) |
| 04:53.12 | *** join/#asterisk psy65535 (n=psy65535@24-205-53-78.dhcp.gldl.ca.charter.com) |
| 04:53.45 | *** join/#asterisk TrentCreek (i=GeekBoy@cpe-70-117-207-168.rgv.res.rr.com) |
| 04:54.20 | *** join/#asterisk famicon (i=scenesta@c51447ddc.cable.wanadoo.nl) |
| 04:54.40 | *** join/#asterisk famicon (i=redz@c51447ddc.cable.wanadoo.nl) |
| 04:55.56 | *** join/#asterisk Tili (n=tili@cm48.gamma244.maxonline.com.sg) |
| 04:59.56 | *** join/#asterisk grimsy (n=chatzill@203-206-220-214.perm.iinet.net.au) |
| 05:01.42 | *** join/#asterisk Chris-NB (n=chris@ng1.kurtkrenn.com) |
| 05:02.34 | *** join/#asterisk [hC] (n=hardcore@S01060016b61c8983.vf.shawcable.net) |
| 05:03.30 | TrentCreek | anyone up for a question on extension.conf? |
| 05:04.07 | *** part/#asterisk techie (n=techie@adsl-76-240-178-163.dsl.lsan03.sbcglobal.net) |
| 05:04.10 | pepse | don't ask to ask |
| 05:04.14 | pepse | :) |
| 05:04.34 | TrentCreek | okay i set it up and it is functioning to make calls in the US |
| 05:04.41 | *** join/#asterisk twoshadetod (n=twoshade@c-66-177-74-0.hsd1.fl.comcast.net) |
| 05:04.50 | TrentCreek | but I cannot dial the extra numbers and call out of the country |
| 05:05.04 | TrentCreek | what is the format? |
| 05:05.24 | pepse | 011 + country code + city code + number |
| 05:07.02 | TrentCreek | curently it is setup like exten => _1NXXNXXXXXX,1,SetCallerID(1234567890) |
| 05:07.06 | twoshadetod | i read a very good looking guide http://chayden.net/Asterisk/SeUpAsteriskAtHome.htm that talks about using a cd to reformat the system and install it's own sort of os. But I also see asterisk in my synaptic list in ubuntu. I much rather have a system I'm used to , is the ubuntu way just as good? |
| 05:07.12 | TrentCreek | exten => _1NXXNXXXXXX,2,Dial,IAX2/ |
| 05:08.24 | pepse | you could do exten => _011.,2,Dialetc |
| 05:09.09 | pepse | or if you know the countries/cities you'll be dialing to you can get more specific |
| 05:09.20 | pepse | N is 2-9, X is anything, etc etc |
| 05:09.26 | pepse | at least i think N is 2-9. |
| 05:10.45 | TrentCreek | you mean add an additional exten => to that phone? |
| 05:11.43 | TrentCreek | or change what i got? |
| 05:14.31 | *** join/#asterisk karleeto (n=karl@209.194.99.178) |
| 05:21.54 | TrentCreek | i see a third one : exten => _011.,1,SetCallerID(1234567890) |
| 05:23.13 | TrentCreek | i think I got it |
| 05:25.55 | TrentCreek | Anyone, anyone? |
| 05:25.56 | TrentCreek | exten => _011.,1,SetCallerID(1234567890) |
| 05:25.57 | TrentCreek | exten => _011.,2,2,Dial,IAX2/ |
| 05:26.22 | TrentCreek | still can't call international |
| 05:29.34 | Strom_M | TrentCreek: which version of asterisk are you using? |
| 05:29.56 | TrentCreek | 1.14 |
| 05:30.13 | TrentCreek | was that extension done right? |
| 05:31.39 | *** join/#asterisk jarod14 (n=jarod14@mail.viatelecom.com) |
| 05:35.18 | Strom_M | there is no 1.14 |
| 05:35.35 | Strom_M | but if you're using 1.4, you shouldn't be reading examples that use 1.0 syntax |
| 05:36.42 | AJaymn | Anyone know of a good DID provider thats less then $4 per DID? |
| 05:36.52 | AJaymn | for unlimited invbound |
| 05:36.54 | TrentCreek | i di |
| 05:36.56 | TrentCreek | i do |
| 05:36.59 | jql | I presume you mean not in bulk |
| 05:37.06 | TrentCreek | bu tyou have to buy them in bulk |
| 05:37.14 | jql | heh, yeah. bulk is easy |
| 05:37.32 | AJaymn | well lets say 25 at a time :P |
| 05:37.32 | jql | you want 500 numbers? I got the hookup. heh |
| 05:37.54 | jql | (no, not really) |
| 05:43.38 | *** join/#asterisk DaveCanoe (n=Dave@canoe404.dclg.ca) |
| 05:47.31 | TrentCreek | i did get me international calling to function |
| 05:47.46 | TrentCreek | I had a typo in the exentions.conf file |
| 05:53.39 | TrentCreek | anyway to monitor the call quality? |
| 05:53.56 | jql | rtcp debug and stuff |
| 05:54.47 | TrentCreek | thanks |
| 05:55.18 | TrentCreek | RTCP (Real-time Control Protocol) |
| 05:55.19 | TrentCreek | <PROTECTED> |
| 05:55.19 | TrentCreek | <PROTECTED> |
| 05:55.19 | TrentCreek | <PROTECTED> |
| 05:55.19 | TrentCreek | <PROTECTED> |
| 05:55.19 | TrentCreek | <PROTECTED> |
| 05:55.21 | TrentCreek | <PROTECTED> |
| 05:55.23 | TrentCreek | <PROTECTED> |
| 05:55.25 | *** join/#asterisk gremzoid (n=gremzoid@d58-106-229-231.rdl5.qld.optusnet.com.au) |
| 05:58.08 | *** join/#asterisk Teln1100A (i=hello123@bas2-toronto12-1128738062.dsl.bell.ca) |
| 06:00.15 | *** join/#asterisk Somebee (n=sindre@80.232.5.97) |
| 06:01.07 | *** join/#asterisk karleeto (n=karl@207.191.91.242) |
| 06:02.18 | JT | TrentCreek: what was the point of that flood? |
| 06:05.53 | *** join/#asterisk chendy (n=chendy@121.76.132.123) |
| 06:07.00 | TrentCreek | oh..sorry |
| 06:07.54 | TrentCreek | Came from a great Speech tonight |
| 06:08.06 | TrentCreek | filmed the whole thing |
| 06:08.18 | TrentCreek | ol' Gorby |
| 06:09.06 | JT | i see |
| 06:09.53 | TrentCreek | nyet ;-) |
| 06:14.21 | Freman | whoot, found the firmware 2.2 for the polycom's |
| 06:14.27 | Freman | (raided the trixbox repo (c: ) |
| 06:15.38 | jql | I'm pretty happy with 2.2, so far |
| 06:15.48 | jql | I even dared to upgrade the bootrom |
| 06:15.50 | jql | scary stuff |
| 06:16.43 | awk | how do you lower music on hold vulume? |
| 06:16.45 | awk | volume |
| 06:17.13 | jql | sox? |
| 06:17.59 | awk | i'm using madplay |
| 06:18.04 | BBHoss | jql: anything different or any good features |
| 06:18.36 | jql | BBHoss: various bugfixes, and it was the first release I've used with nat ping |
| 06:18.44 | jql | which I eagerly turned on |
| 06:19.41 | jql | and it supports the 550, which is a kickass phone. damn |
| 06:20.30 | BBHoss | hmm |
| 06:20.34 | jql | also, it fixed the microbrowser to not look all bizzare, which tempts me to use it for stuff |
| 06:20.39 | BBHoss | basically a keepalive |
| 06:20.42 | jql | yes |
| 06:21.51 | *** join/#asterisk IgorG (n=FeedomPa@195.162.32.126) |
| 06:25.35 | *** join/#asterisk denon (n=denon@208.122.43.201) |
| 06:25.35 | *** mode/#asterisk [+o denon] by ChanServ |
| 06:25.47 | Freman | that's why I'm looking at it |
| 06:26.02 | Freman | 2.1 introduces microbrowser for the 430, 2.2 fixes it (c: |
| 06:26.45 | jql | lol, sounds right |
| 06:27.13 | Freman | holey hell.. the version I found of it is has a 13.8 meg sip.ld |
| 06:30.39 | Freman | now to work out that template stuff... |
| 06:31.23 | *** join/#asterisk Strom_M (n=strom@208.127.172.112) |
| 06:33.23 | *** join/#asterisk clandmeter (n=Carlo@81.175.82.2) |
| 06:36.17 | *** join/#asterisk kmarquez (n=kmarquez@62.141.48.57) |
| 06:36.19 | psy65535 | ok so I'm attached to a sip trunk, inbound calls work fine. testing my outbound dialplan I can get the record app to record voice but functions like WaitForSilence and AMD are reporting that they are not hearing anything. I've been up and down the firewall config, any ideas? |
| 06:38.31 | *** join/#asterisk denon (n=denon@208.122.43.201) |
| 06:38.31 | *** mode/#asterisk [+o denon] by ChanServ |
| 06:38.38 | *** join/#asterisk PepOSX (n=pepOSX@190.72.149.12) |
| 06:40.37 | TrentCreek | hmmm |
| 06:41.37 | TrentCreek | have you tried a WAOT before starting the call to allow the network to settle down first? |
| 06:41.42 | TrentCreek | oops.. WAIT |
| 06:42.02 | psy65535 | there really isn't a lot of traffic right now.. I'm doing testing with one channel |
| 06:42.12 | *** join/#asterisk ru_wild (n=xwild@81.26.90.134) |
| 06:42.58 | psy65535 | what is most strange is that I'm seeing all sorts of traffic but the udp stuff from the box I care about doesn't even show up in the firewall log |
| 06:43.20 | psy65535 | but I am registered and the phone does ring... |
| 06:43.35 | jql | asterisk sees the incoming rtp packets, though? |
| 06:43.38 | TrentCreek | yeah but |
| 06:43.53 | *** join/#asterisk syneus (n=syneus@syneus.aemcom.net) |
| 06:44.03 | psy65535 | hard to say. I do have that port range natted over to the asterisk host. |
| 06:44.05 | *** join/#asterisk Ethon (i=arne@Oldman.steinkamm.com) |
| 06:44.10 | TrentCreek | sometimes conections happen faster than what the rets of the system can repsond to |
| 06:44.16 | jql | rtp debug gives good info |
| 06:46.06 | psy65535 | indeed |
| 06:46.09 | psy65535 | nice flood |
| 06:46.24 | psy65535 | rtp definitely getting stuff |
| 06:46.38 | *** join/#asterisk [hC] (n=hardcore@S01060016b61c8983.vf.shawcable.net) |
| 06:46.54 | *** join/#asterisk kiscokid (n=Ron_Laut@adsl-216-101-109-187.dsl.snfc21.pacbell.net) |
| 06:47.00 | jql | well, that's good at least |
| 06:47.01 | psy65535 | can I paste a line from the output? |
| 06:47.07 | jql | a line is fine |
| 06:47.10 | psy65535 | app_waitforsilence.c:102 do_waiting: No audio available on SIP/ |
| 06:47.24 | jql | was that the entire line? |
| 06:47.44 | psy65535 | channel name and date time truncated |
| 06:47.51 | jql | because that line no channel |
| 06:48.04 | jql | okay |
| 06:48.08 | psy65535 | 1 more time. :) |
| 06:48.10 | psy65535 | Oct 9 23:45:11 WARNING[30170]: app_waitforsilence.c:102 do_waiting: No audio available on SIP/telasip-gw-0826b6b8?? |
| 06:49.17 | jql | well, that generally means 4 seconds have passed without audio |
| 06:49.27 | jql | which is weird, considering you're seeing packets fly in |
| 06:49.30 | jql | nat problem? |
| 06:51.07 | psy65535 | well there is definitely natting going on but I've got the entire rtp range forwarding |
| 06:51.37 | psy65535 | what else should I be looking for; or is the waitforsilence app using the channel differently? |
| 06:51.56 | jql | it seems like asterisk is not recognizing the incoming packets as belonging to *that* channel, which is weird if "rtp debug" displays the incoming packets |
| 06:52.06 | jql | that would tend to indicate that asterisk is listening on that port |
| 06:52.13 | jql | I dunno |
| 06:52.50 | psy65535 | I'm thinking about checking the code difference between channel use between the two applications (waitforsilence and record) |
| 06:53.57 | psy65535 | what is most curious is that AMD was working a couple weeks ago -- all of a sudden it stopped. Nothing changed in the fw config, in fact due to this I've started getting more involved with the firewall config and nat translation |
| 06:54.07 | psy65535 | I shrug too |
| 06:55.23 | jql | I had to do some funky nat stuff to get asterisk to talk on two networks at once. very annoying |
| 06:55.33 | jql | I really wish support was a bit better |
| 06:56.21 | jql | my iptables config is scary, even for me |
| 06:56.28 | psy65535 | kinda why we have each other |
| 06:56.33 | psy65535 | and open code |
| 06:56.50 | *** part/#asterisk kiscokid (n=Ron_Laut@adsl-216-101-109-187.dsl.snfc21.pacbell.net) |
| 06:58.40 | psy65535 | any suggestions working asterisk with ddd? |
| 06:58.53 | psy65535 | flags, options, caveats, tips? |
| 06:58.54 | *** part/#asterisk Raky-2 (n=John@220.157.75.246) |
| 06:58.59 | *** join/#asterisk munmun (n=mun_mun@203.80.176.168) |
| 06:59.02 | *** join/#asterisk KermitTheFragger (n=siepkes@118-197.bbned.dsl.internl.net) |
| 06:59.05 | *** join/#asterisk dominic1 (n=dob@213.221.82.242) |
| 06:59.07 | *** part/#asterisk dominic1 (n=dob@213.221.82.242) |
| 06:59.13 | TrentCreek | i got a quicky...Looking at the call progresses, I see no call total time..how can I see how much time per call? |
| 06:59.45 | psy65535 | I'm database driven but I believe that is in the CDR information |
| 06:59.54 | psy65535 | duration/billsec |
| 07:00.34 | *** part/#asterisk carrar (i=tim@osburn.com) |
| 07:02.59 | TrentCreek | thanks |
| 07:03.54 | *** join/#asterisk Kapsel (i=kapsel@62.242.240.33) |
| 07:04.03 | TrentCreek | i am too...but...SIP calls from the internal connection is not |
| 07:05.01 | psy65535 | they should all go into the cdr table then, internal or external |
| 07:06.06 | TrentCreek | okay..thanks for that info |
| 07:07.07 | TrentCreek | its only 1 or 2 people using inside...I am just curious |
| 07:07.30 | TrentCreek | can the aserisk box detect answer ? |
| 07:07.48 | jql | define detect and define answer |
| 07:07.55 | psy65535 | heh |
| 07:07.55 | jql | :) |
| 07:08.10 | psy65535 | careful you don't end up like me |
| 07:08.22 | psy65535 | reliant on AMD plugin and it no workie! |
| 07:08.23 | jql | asterisk knows when the call is answered. it's with a 200 OK message through SIP |
| 07:08.50 | TrentCreek | groovy///because I just noticed my privder is charging me even if I get a message saying "phone Not in Service" |
| 07:09.20 | jql | you shouldn't be. that's gimp |
| 07:09.36 | jql | messages like that should be part of the early audio |
| 07:09.41 | TrentCreek | no..i looked at the calls I made today |
| 07:10.11 | TrentCreek | only 1 cent, but when i get 100+ callers going, that will add up |
| 07:10.40 | TrentCreek | gimp? the Linux version of Photoshop? ;-) |
| 07:10.52 | TrentCreek | oh..thats THEgimp |
| 07:10.59 | awk | *Sigh* whenm will parking be fixed in asterisk |
| 07:10.59 | jql | no, more like the gimp from pulp fiction. :) |
| 07:11.02 | awk | [Oct 10 09:09:24] WARNING[28870]: chan_sip.c:12532 handle_response: Remote host can't match request BYE to call '08a33d2a6ac34fa03816aac86753e9e7@192.168.21.203'. Giving up. |
| 07:11.20 | *** join/#asterisk codec (n=codec@iglu.paranoid-penguin.de) |
| 07:11.23 | awk | ive wrote my own patches, ive tried it all, parking is just crap |
| 07:11.32 | jql | bleh. parking |
| 07:11.43 | TrentCreek | or Gunp |
| 07:11.47 | TrentCreek | Gump |
| 07:11.59 | jql | I'm at the point of just "escorting" people into a random conference via ami, at this point |
| 07:12.12 | jql | and that sucks |
| 07:12.32 | TrentCreek | why? |
| 07:19.33 | pif | hi, I'm trying to upgrade a thomson st2030's firmware but it keeps rebooting after loading the fw file through tftp, any idea? |
| 07:21.52 | karleeto | you guys think that a G3 iMac with Gentoo Linux that is built ONLY to need would serve well for my home system?? maybe just like 2 or 3 phones?? |
| 07:22.53 | karleeto | s/need/spec/ |
| 07:23.20 | karleeto | nice, i like jbot |
| 07:23.50 | TrentCreek | only 2-3 phones? You would be better just getting a 2 line SIP device |
| 07:24.44 | TrentCreek | and with the cost of energy going up..would save mor emoney |
| 07:29.21 | karleeto | TrentCreek: maybe, but ive had the imac in the garage for 2 years; i'm installing BASE gentoo linux on it now, and i was looking at it as a learning experience; i've always used trixbox beofre lately, so i'm actually tryinh yo learn asterisk ALONE and not trixbox to take over all my configs for me |
| 07:29.48 | J4k3 | karleeto: just make sure to power-save the CRT fully, and put a real fan in it. |
| 07:29.59 | J4k3 | imacs, especially 1st gens, like to overheat. |
| 07:30.09 | J4k3 | also make sure it doesn't have any blown capacitors |
| 07:31.11 | TrentCreek | I have a first gen with a 500Mhz Harmoni card in it and OX X 10.3.9 |
| 07:32.11 | TrentCreek | Flash video plays very bad on it, but strangly Final Cut Pro 4 runs finr |
| 07:32.16 | TrentCreek | *fine |
| 07:33.22 | TrentCreek | those PwerPC are a "double edged sword" |
| 07:34.00 | karleeto | it just killed my ssh that i was installing gentoo in cause it didnt have enough mem\ |
| 07:34.20 | *** join/#asterisk J4zen (n=Jvan4zen@a213-84-139-244.adsl.xs4all.nl) |
| 07:35.10 | karleeto | is there any way i can tell if that finished? it was # tar xvjf /mnt/gentoo/portage-latest.tar.bz2 -C /mnt/gentoo/usr |
| 07:35.17 | TrentCreek | they are RISC based, which means they can perform instructions faster, however if the instructions do not exist on the CPU, thye must be done in software which means slower operations |
| 07:35.18 | *** join/#asterisk zeeesh (i=zeeesh@202.166.161.45) |
| 07:35.42 | TrentCreek | yeah..when the USER PROMPT reapperars |
| 07:35.44 | jql | if the command itself finished? |
| 07:35.48 | karleeto | or should i just rm -rf usr/portage and rerun that extraction> |
| 07:35.51 | jql | yeah, what TrentCreek said |
| 07:36.11 | karleeto | jql: is was over a sshd, and it got killed |
| 07:36.37 | jql | then it probably got pipe stalled and SIGPIPEd |
| 07:36.49 | *** join/#asterisk _ShrikE (n=ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 07:36.50 | jql | which is usually fatal |
| 07:37.32 | jql | I run screen, because I ssh over wifi |
| 07:37.36 | jql | damn thing's always going down |
| 07:38.11 | TrentCreek | you could sell that iMac and buy a newer Intel Based system for the money |
| 07:38.32 | *** join/#asterisk bintut (n=bintut@203.125.63.150) |
| 07:39.05 | TrentCreek | Prob a PIII Coppermine running at 1Ghz |
| 07:39.33 | jql | damn, the dawn of the ghz race |
| 07:39.45 | TrentCreek | bno |
| 07:39.47 | TrentCreek | no |
| 07:40.09 | jql | and to think, we're still only at 2.x ghz all these years later |
| 07:40.13 | TrentCreek | the PII and original Athalon was at 1GHz |
| 07:40.16 | *** join/#asterisk duskot (n=dsk-o@194.209.212.4) |
| 07:40.51 | awk | I think i found the parking issue |
| 07:40.57 | TrentCreek | because speed is no longer an issue because of technology limitation, thus they are turning to other ways to get more speed |
| 07:40.57 | awk | it looks like a snom issue with park orbit |
| 07:41.01 | awk | anyone experienced it |
| 07:41.06 | TrentCreek | I mean performance |
| 07:41.21 | TrentCreek | no..never heard of it |
| 07:41.24 | jql | I don't support the snom, and I don't try very hard with parking |
| 07:41.31 | jql | sorry. :( |
| 07:41.47 | jql | although I do like my 360 |
| 07:42.03 | jql | any phone running linux is fine by me |
| 07:42.13 | karleeto | bbiaf, ivestigation the installation of linux on my NEW laptop! |
| 07:42.34 | awk | hrm |
| 07:43.58 | TrentCreek | Linux on an IBM 360? ;-) |
| 07:46.57 | BBHoss | ive been trying to get into the snom 320 |
| 07:47.09 | BBHoss | but the source on their site is outdated |
| 07:47.34 | BBHoss | im trying to write an LTP and IAX driver for the snom |
| 07:47.55 | BBHoss | they are just using an INCA-IP from infineon |
| 07:55.38 | J4zen | You still working on that BBHoss ? |
| 07:55.47 | BBHoss | yeah |
| 07:55.58 | BBHoss | next step is tearing into my snom 300 |
| 07:56.07 | J4zen | scary hehe |
| 07:56.08 | BBHoss | see if i can find the serial port |
| 07:56.13 | J4zen | yeah i wonder |
| 07:56.14 | *** join/#asterisk rati (n=rati@124.125.254.227) |
| 07:56.15 | BBHoss | it was free |
| 07:56.16 | J4zen | i have one on my desk right now |
| 07:56.26 | J4zen | please, do tell me if you find it |
| 07:56.26 | BBHoss | if you do decide to dive in |
| 07:56.40 | BBHoss | test the voltage on the serial port on the board |
| 07:56.51 | J4zen | for the rs232 ? |
| 07:56.51 | BBHoss | because its rumored to be 3.3v |
| 07:56.57 | BBHoss | and RS232 is 12v |
| 07:57.01 | J4zen | yeah i read that too |
| 07:57.07 | BBHoss | so we'd need a MAX3232 |
| 07:57.24 | J4zen | i think it stated that voltage in the manual even |
| 07:57.28 | J4zen | limited as it may be .. |
| 07:57.31 | BBHoss | where? |
| 07:57.41 | J4zen | one of the readme's in the dev pack |
| 07:57.43 | BBHoss | voltage for a serial port? |
| 07:57.45 | BBHoss | oh |
| 07:57.56 | J4zen | i might be mistaking tho, it was quite a maze |
| 07:58.02 | BBHoss | that dev pack is totally or mostly bogus i think |
| 07:58.06 | J4zen | did you ever find some proper documentation |
| 07:58.09 | BBHoss | nope |
| 07:58.47 | BBHoss | some info here |
| 07:58.47 | BBHoss | http://web.archive.org/web/*/www.openhardphone.org/ |
| 07:59.05 | BBHoss | and i found a tool that decompresses firmware, not sure if it recompresses |
| 07:59.12 | J4zen | good ol' archives |
| 07:59.45 | BBHoss | looks like we could just setup a toolchain for the r4k MIPS processor |
| 07:59.49 | BBHoss | and compile and go |
| 07:59.55 | BBHoss | for at least ssh and stuff |
| 08:00.13 | BBHoss | but from what ive seen |
| 08:00.28 | BBHoss | they package their whole device software into a binary file |
| 08:00.36 | BBHoss | so we probably will have to rip that out |
| 08:00.41 | BBHoss | and do a total rewrite |
| 08:01.07 | BBHoss | we would also have to learn how to address the hardware, since LCSserver usually did that |
| 08:01.42 | J4zen | wish i had time for it :( |
| 08:01.47 | BBHoss | yeah |
| 08:02.12 | BBHoss | looks like its going to be a huge task if snom doesnt want to help |
| 08:02.23 | BBHoss | maybe they'll throw us a bone |
| 08:02.35 | BBHoss | give us the source to their app |
| 08:02.43 | J4zen | doubt it, but worth a shot |
| 08:02.55 | J4zen | they're rather greedy, so i hear anyway |
| 08:02.58 | BBHoss | then all we would have to do is change out the sip stuff with iax stuff |
| 08:03.14 | BBHoss | i have some connections, at least with the american side of things |
| 08:03.29 | J4zen | yeah that'd be awesome |
| 08:03.40 | J4zen | connections? |
| 08:04.08 | J4zen | WARNING: Never connect the serial interface directly to a RS232 interface! The serial interface of the phone uses 3.3V and a RS232 uses 12V! |
| 08:05.13 | BBHoss | is that off of the archive |
| 08:05.24 | J4zen | Yes |
| 08:05.27 | J4zen | let me check that tar |
| 08:06.26 | J4zen | Was it around 300 megs? |
| 08:06.43 | J4zen | nm, got it |
| 08:06.48 | *** join/#asterisk duskot (n=dsk-o@194.209.212.4) |
| 08:11.36 | Guggemand | anyone know of any poblic available danish sounds for asterisk ? |
| 08:12.44 | J4zen | or now that we're on that topic, Dutch. Possibly paid even |
| 08:15.36 | *** join/#asterisk nfi|ermes (n=ErMeS@217.220.121.62) |
| 08:16.03 | *** part/#asterisk munmun (n=mun_mun@203.80.176.168) |
| 08:28.14 | *** join/#asterisk tengulre (n=tengulre@222.90.66.10) |
| 08:28.49 | *** join/#asterisk parag0n (n=parag0n@87-194-9-117.bethere.co.uk) |
| 08:28.57 | tengulre | hi,all |
| 08:29.13 | tengulre | anybody successful running SS7 with asterisk ? |
| 08:29.28 | *** join/#asterisk paljas (n=paljas@sarastro.cs.uu.nl) |
| 08:48.32 | *** join/#asterisk _pepo_ (n=Pepo@190.10.187.20) |
| 08:51.17 | *** join/#asterisk shinao1 (n=shinao1@41.222.65.161) |
| 08:52.51 | *** join/#asterisk tzafrir (n=tzafrir@62.90.10.53) |
| 08:58.50 | *** join/#asterisk Schumie (n=Steve@host90-152-2-2.ipv4.regusnet.com) |
| 09:00.25 | *** join/#asterisk Polis_ttt (n=Polis_tt@194-237-172-225-no48.business.telia.com) |
| 09:01.17 | _pepo_ | hi friends |
| 09:03.13 | Polis_ttt | i do got a litle problem. My asterisk-server, 1.2.17, do try to send some signals to offline phones, lite "keep alive" asterisk-cli doesn't print anything about it, but when i killed asterisk the signals stopped. What can i do to stop this? the phones are not registred, and show peers, tells me that they are unknown, just like i should do when they are offline |
| 09:04.21 | Polis_ttt | when i kill asterisk-server, the signals stops, so it is asterisk that send those signals, app. it uses 20kb/s bandwith/sip-account for those signals |
| 09:04.36 | Polis_ttt | and only udp traffic |
| 09:05.12 | BBHoss | youve got some problem there |
| 09:05.36 | BBHoss | i've never seen this with asterisk though |
| 09:05.51 | BBHoss | and asterisk dosent send anything out to extensions that aren't registered |
| 09:06.06 | BBHoss | and even if they are registered it rarely does |
| 09:06.19 | BBHoss | the phones initiate all conversations |
| 09:08.33 | Polis_ttt | i think it's very strange to, none of my other servers do send this signals, sometimes this server transmit around 300kb/s when theres no sip-account registred or unreachable on it, when stop asterisk, signals stop :( |
| 09:09.03 | BBHoss | where are you getting these kb/s values from? |
| 09:09.11 | Polis_ttt | it's also strange that it only sends udp-signals for this, no tcp signals on port 5060-5061 is even trying to connect |
| 09:09.19 | BBHoss | yeah |
| 09:09.23 | BBHoss | nothing goes over tcp |
| 09:09.30 | BBHoss | everything is udp |
| 09:09.40 | BBHoss | sip, iax, RTP, rtcp, etc |
| 09:10.54 | BBHoss | ive never seen this before |
| 09:11.29 | BBHoss | its most likely a bug with your exact setup, or a rootkit, or you misunderstanding a graph |
| 09:12.15 | Polis_ttt | BBHoss: using netstat or tshark, with grep on ip-number to coustumer, so only that traffic |
| 09:13.00 | BBHoss | what kind of packets is it sending |
| 09:13.06 | BBHoss | do you have a pcap file i can look at? |
| 09:13.32 | Polis_ttt | i don't missunderstand the traffic-log, iv'e used cacti log to, when server is totaly inactive, only those signals, server sends those signals 24/7 and my coustumers do only use server 9-21 |
| 09:13.47 | BBHoss | ok |
| 09:14.00 | BBHoss | are you using a voip provider? |
| 09:14.25 | BBHoss | or POTS/T1,E1,J1 etc |
| 09:15.00 | _pepo_ | hi friends |
| 09:15.51 | _pepo_ | I have problems, when I call to another extension the call is good but I cant hear the tones (DTMF) in any hear |
| 09:15.54 | Polis_ttt | BBHoss: there you got a log-file, from tshark, using command tshark -V dst host ***.***.***.*** |
| 09:16.01 | BBHoss | ok |
| 09:16.30 | _pepo_ | Do I have to use some especial codec? |
| 09:17.20 | Polis_ttt | _pepo_: strange, are you using sip och iax, and what voip-client do you use? |
| 09:18.04 | BBHoss | are you using inband DTMF signaling |
| 09:18.16 | _pepo_ | I am using SIP and it is the same with soft-phones and hard-phones |
| 09:18.48 | _pepo_ | yes I was using dtmf=inband |
| 09:18.49 | _pepo_ | dtmfmode=inband on [general] in my sip.conf |
| 09:19.07 | _pepo_ | do I have to use rfc2833 ? |
| 09:19.53 | Polis_ttt | _pepo_: try to ;-out that string, and se if it's any diffrence |
| 09:20.49 | BBHoss | if you're not using alaw/ulaw then inbadn wont work |
| 09:21.12 | BBHoss | rfc2833 is usually much better than inband |
| 09:22.51 | *** join/#asterisk munmun (n=mun_mun@203.80.176.168) |
| 09:28.39 | Fluor_ | is it? i have been told inband was the-standard-way :) |
| 09:30.15 | BBHoss | i hate inband |
| 09:30.25 | BBHoss | an you have to use a fullrate codec |
| 09:30.25 | J4k3 | death to inband |
| 09:30.29 | J4k3 | (doesn't inband eat cpu?) |
| 09:30.54 | BBHoss | dunno |
| 09:31.03 | BBHoss | im sure it uses more than rfc2833 |
| 09:31.06 | [hC] | rfc2833 is "the standard way" as far as asterisk goes. |
| 09:31.21 | BBHoss | 'rfc2833 is cool' |
| 09:31.22 | Chris-NB | hi |
| 09:31.23 | BBHoss | :) |
| 09:31.34 | Chris-NB | anyone using queues in an asterisk system? |
| 09:31.43 | BBHoss | lots of people do :) |
| 09:32.39 | Chris-NB | BBHoss, do you use them? |
| 09:32.52 | Chris-NB | my * crashed from time to time when using queues |
| 09:32.54 | Chris-NB | ... |
| 09:32.57 | BBHoss | i haven't used them on Asterisk |
| 09:33.06 | Chris-NB | and I've no clue why |
| 09:33.07 | BBHoss | how many channels are you running |
| 09:34.19 | *** join/#asterisk stkn|work (n=stkn|wor@gentoo/developer/pdpc.active.stkn) |
| 09:35.00 | JT | Fluor_: inband is not the standard way with VoIP |
| 09:35.16 | JT | does not work unless you use G.711 |
| 09:35.20 | JT | use RFC2833 |
| 09:35.29 | *** join/#asterisk fordfrog (n=fordfrog@gentoo/developer/fordfrog) |
| 09:36.19 | JT | Polis_ttt: what's weird about there being no TCP. asterisk does not do VoIP over TCP. |
| 09:37.34 | Chris-NB | BBHoss, between 10 and 30 |
| 09:37.40 | BBHoss | hmm |
| 09:37.44 | BBHoss | what kind of proc |
| 09:37.47 | *** join/#asterisk bird_of_Luck (n=melifaro@secured.by.ipfw.ru) |
| 09:37.52 | Chris-NB | BBHoss, overall, but only 1 -3 on queue |
| 09:38.06 | Chris-NB | BBHoss, a DL380-G5 (oversized) : D |
| 09:38.32 | JT | that's not the processor type |
| 09:38.38 | JT | but it would be some sort of xeon |
| 09:38.42 | BBHoss | yeah |
| 09:38.45 | BBHoss | should be enough |
| 09:38.50 | Chris-NB | should be ... |
| 09:38.51 | BBHoss | so it just crashes out of the blue |
| 09:38.55 | Chris-NB | jep |
| 09:39.09 | BBHoss | any certain load or anything, or is it totally random |
| 09:39.10 | J4k3 | xeon... with Pentium Pro 133/256's installed on slockets. |
| 09:39.21 | BBHoss | :) |
| 09:39.24 | J4k3 | hehe |
| 09:44.53 | parag0n | wait, dual pentium pros arent enought o run asterisk? |
| 09:45.19 | parag0n | why would someone sell em this server for ?10000 then! |
| 09:45.41 | J4k3 | pimpium |
| 09:47.20 | Chris-NB | BBHoss, totaly random. |
| 09:47.31 | Chris-NB | now it was 5 days up |
| 09:47.46 | Chris-NB | a week ago it happened 3 times during one day |
| 09:47.54 | Chris-NB | here is my config: http://pastebin.com/m3cfb9ac1 |
| 09:48.39 | BBHoss | what version are you running |
| 09:49.00 | Chris-NB | 1.2.24 |
| 09:49.23 | Chris-NB | but had the same behavior on 1.2.13 |
| 09:49.51 | BBHoss | hmm |
| 09:49.55 | BBHoss | have you tried 1.4 |
| 09:50.03 | BBHoss | many bugs are fixed (and created) |
| 09:50.29 | gremzoid | is there a reason for 1.2 stable and 1.4 stable? |
| 09:50.38 | BBHoss | heh |
| 09:50.44 | BBHoss | its totally fuxxed |
| 09:51.12 | gremzoid | ? |
| 09:51.19 | BBHoss | http://gremzoid.getfuxed.com/ |
| 09:51.35 | BBHoss | like that |
| 09:51.41 | BBHoss | :) |
| 09:51.51 | gremzoid | hey! how'd it'd know i look like that? |
| 09:52.10 | Chris-NB | BBHoss, everything else is working fine. it's a production box and highly customized so I don't want to 'try' 1.4 |
| 09:52.26 | BBHoss | LOL!! |
| 09:52.38 | Chris-NB | : / |
| 09:52.50 | BBHoss | hmm |
| 09:53.07 | BBHoss | its not trashbox or freepbx is it? |
| 09:53.18 | Chris-NB | no its a plain asterisk |
| 09:53.24 | Chris-NB | from src |
| 09:53.57 | BBHoss | hmm |
| 09:54.06 | BBHoss | are you running 64 bit? |
| 09:54.21 | Chris-NB | ähm ... have to look : ) |
| 09:54.49 | Chris-NB | no |
| 09:54.57 | BBHoss | thats wierd |
| 09:55.21 | Chris-NB | DualCore Xeon |
| 09:55.27 | Chris-NB | TOTALLY !! |
| 09:55.28 | Chris-NB | : / |
| 09:55.42 | BBHoss | could be a dual core problem, but i doubt it |
| 09:55.50 | Chris-NB | *hmmm |
| 09:56.08 | JT | that's a 64bit cpu btw |
| 09:56.23 | BBHoss | yeah but if hes not running x86_64 |
| 09:56.31 | BBHoss | then there shouldnt be a problem |
| 09:56.56 | BBHoss | ok, does asterisk dump core and all, or just stop responding when it 'crashes' |
| 09:56.58 | JT | shouldn't |
| 09:57.03 | JT | but then again, it is asterisk |
| 09:57.23 | Chris-NB | ok I'm running linux-2.6.18-5-686 |
| 09:57.32 | Chris-NB | no core dump |
| 09:57.38 | Chris-NB | just stops |
| 09:57.47 | BBHoss | it just basically stops responding to anything? |
| 09:58.03 | Chris-NB | no asterisk process any longer |
| 09:58.09 | Chris-NB | server is responding fine |
| 09:58.17 | BBHoss | hmm |
| 09:58.20 | Chris-NB | but nothing from asterisk |
| 09:58.24 | BBHoss | so it is a total crash then |
| 09:58.29 | Chris-NB | jep |
| 09:59.01 | BBHoss | im looking at bugs btw |
| 09:59.03 | Chris-NB | and the logs (full+sip+pri intense) print the last message and stops |
| 09:59.10 | Chris-NB | nothing more! |
| 09:59.16 | Chris-NB | no err, no oops .... |
| 09:59.31 | BBHoss | is the last message significant? |
| 09:59.47 | Chris-NB | nop |
| 09:59.49 | Chris-NB | nothing |
| 09:59.58 | Chris-NB | seems normal |
| 10:00.21 | Chris-NB | before i configured the queue the server was up for 1/2 a Year running just fine |
| 10:00.28 | BBHoss | hmm |
| 10:02.06 | BBHoss | maybe you ought to run it in gdb the do a bt full when it crashes |
| 10:02.35 | Chris-NB | but for that I've to rebuild it without optimization, right? |
| 10:02.46 | BBHoss | not sure |
| 10:02.56 | Chris-NB | read something about that |
| 10:03.01 | BBHoss | its my understanding you can run gdb on anything |
| 10:03.10 | BBHoss | but you might not get as much info |
| 10:03.15 | Chris-NB | mhm |
| 10:03.17 | parag0n | we are in the UK, and currently hae 6 BT lines into our building. this is costing us quite every month in line rental alone, so i'm looking at trying to replace it with a voip system |
| 10:03.18 | Chris-NB | how can I do that? |
| 10:03.22 | *** join/#asterisk ghenry (n=ghenry@suretecsystems.plus.com) |
| 10:03.50 | BBHoss | personally ive not used gdb that much |
| 10:04.17 | Chris-NB | personally ive never used gdb : ) |
| 10:04.30 | parag0n | can anyone tell me if we could replace all the current phone lines with a single line (ISDN?) that could plug into an asterisk box |
| 10:04.34 | BBHoss | im sure theres a tutorial |
| 10:04.47 | BBHoss | a BRI is 2 voice paths |
| 10:04.52 | Chris-NB | then I've to lookk at that : / |
| 10:04.53 | BBHoss | so for 6 POTS |
| 10:04.55 | JT | parag0n: no, an isdn bri line would provide only 2 channels |
| 10:05.01 | parag0n | ah |
| 10:05.02 | BBHoss | you need 3 BRI |
| 10:05.06 | parag0n | what other options are there? |
| 10:05.14 | BBHoss | not much with digital |
| 10:05.17 | BBHoss | you could get a pri |
| 10:05.21 | BBHoss | but thats overkill |
| 10:05.24 | Chris-NB | is there another 'alternative' for queues? |
| 10:05.37 | BBHoss | not really |
| 10:05.50 | BBHoss | i can only suggest trying a 1.4 version |
| 10:05.58 | BBHoss | they are mostly config compatible |
| 10:05.59 | Chris-NB | .... scared ... |
| 10:06.09 | BBHoss | back your shit up :) |
| 10:06.24 | Chris-NB | I've never tried 1.4 |
| 10:06.39 | BBHoss | not much different form 1.2 |
| 10:06.41 | Chris-NB | and I don't wanna try it in a production environment : / |
| 10:06.45 | BBHoss | couple new features |
| 10:06.53 | JT | parag0n: 3 BRIs would be 6 channels |
| 10:07.19 | parag0n | well, we'd need more channels, as we're expanding, probably 10-15 in the near term |
| 10:07.26 | hwt | hm, i am using SayNumber to read up phone numbers, but when it's supposed to read 06 (oh-six), it only reads up 6.. how to i fix that? |
| 10:07.27 | BBHoss | a PRI is 24 |
| 10:07.27 | JT | get a PRI then |
| 10:07.32 | hwt | i don't want to use SayDigits. |
| 10:07.32 | JT | you can start at 10 channels |
| 10:07.35 | BBHoss | or is it 23? |
| 10:07.36 | JT | BBHoss: nope, not in the UK |
| 10:07.39 | Chris-NB | BBHoss, pri is 30 in EU |
| 10:07.39 | parag0n | that sounds good |
| 10:07.40 | JT | and most place in the world |
| 10:07.42 | BBHoss | ok yeah30 |
| 10:07.49 | JT | it's 23B in the us |
| 10:08.06 | parag0n | so, we'd need a server, a PRI line, and an ISDN modem of some sort? |
| 10:08.14 | BBHoss | no |
| 10:08.18 | BBHoss | you need a server |
| 10:08.18 | JT | a pri card |
| 10:08.19 | BBHoss | a pri |
| 10:08.22 | BBHoss | and a pri card |
| 10:08.26 | JT | or a pri to sip gateway |
| 10:08.26 | parag0n | ok |
| 10:08.28 | BBHoss | AKA t1/e1/j1 card |
| 10:08.34 | BBHoss | i suggest a card |
| 10:08.44 | BBHoss | digium, sangoma, rhino |
| 10:08.47 | JT | gateways are more scalable |
| 10:08.52 | JT | lol @rhino |
| 10:09.12 | JT | gateways are overkill for most non-provider applications though |
| 10:09.18 | BBHoss | yeah |
| 10:09.26 | BBHoss | for a SMB you need a card |
| 10:09.34 | JT | need? |
| 10:09.40 | BBHoss | or want |
| 10:09.41 | BBHoss | i guess |
| 10:12.43 | parag0n | hmm, so probably about ?1000 for the card + server, plus line rental? |
| 10:13.15 | BBHoss | not quite $1000 |
| 10:13.36 | parag0n | the card is ?350 |
| 10:13.53 | parag0n | probably put it in a cheap dell rackmount |
| 10:14.11 | parag0n | so closer to ?700 or so for the server + card |
| 10:14.12 | BBHoss | i suggest the sangoma A101 |
| 10:14.12 | JT | get a card with hardware echo cancellation |
| 10:14.19 | JT | about USD$900 for a single port |
| 10:14.22 | BBHoss | A101D |
| 10:15.33 | BBHoss | they also have Pci-express models too |
| 10:16.17 | BBHoss | http://www.voipsupply.com/product_info.php?products_id=2945&gclid=CPm8yrCMhI8CFQ6CPAodeXKM2A |
| 10:17.03 | *** join/#asterisk gardo (n=gardo@124.107.37.42) |
| 10:17.14 | BBHoss | basically a FPGA with a PRI PHY |
| 10:17.58 | BBHoss | dunno why they want a g for it |
| 10:18.07 | BBHoss | i guess for thier fpga deign |
| 10:19.15 | BBHoss | http://www.voipsupply.com/product_info.php?products_id=2914 |
| 10:19.22 | BBHoss | for a digium model\ |
| 10:20.50 | BBHoss | i dont think digium has echo cancel for their one port cards |
| 10:20.55 | JT | www.telephonydepot.com |
| 10:20.57 | BBHoss | at least not hardware |
| 10:20.57 | JT | no they don't |
| 10:21.09 | BBHoss | they do offer the free octasic though |
| 10:21.30 | JT | not for 30 channels |
| 10:21.36 | BBHoss | yeah |
| 10:22.12 | *** join/#asterisk Ursa (n=Ursa@nic06-3-88-173-73-90.fbx.proxad.net) |
| 10:26.01 | Ursa | Can anyone please remind me of the extensions.conf variable to get the source IP address of an inbound SIP call? |
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| 10:29.23 | *** join/#asterisk BiG^DoG (n=BiG^DoG@c-71-204-211-58.hsd1.de.comcast.net) |
| 10:36.59 | *** part/#asterisk gremzoid (n=gremzoid@d58-106-229-231.rdl5.qld.optusnet.com.au) |
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| 10:37.30 | *** part/#asterisk dominic1 (n=dob@213.221.82.242) |
| 10:44.36 | Ursa | Can anyone tell me the difference between SIPCHANINFO(peerip) and SIPCHANINFO(recvip)? |
| 10:44.52 | Ursa | Is it due to possible NAT? |
| 10:49.34 | Ursa | Well, I've just done a test, and they seem to behave identically |
| 10:55.26 | *** join/#asterisk KermitTheFragger (n=siepkes@118-197.bbned.dsl.internl.net) |
| 11:03.19 | *** join/#asterisk yassaccan (n=yassacca@admin189.hgo.se) |
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| 11:17.01 | *** join/#asterisk DrAk0 (n=ljd@nelug/coreteam/luisjose) |
| 11:19.01 | Somebee | Hi guys. I have a problem with incoming calls. It does not manage to find the extension that matches the incoming number. I need to use a 's'-extension, and cannot find a way to differentiate between the 10 incoming lines |
| 11:19.05 | Somebee | what might be the problem? |
| 11:20.59 | *** join/#asterisk Kumbang (n=kumbang@router-wmi.paume.itb.ac.id) |
| 11:26.19 | *** join/#asterisk shinao1 (n=shinao1@196-220-27-23.netcomng.com) |
| 11:27.01 | JT | Somebee: ...what sort of lines... |
| 11:27.56 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 11:28.59 | rati | whats meaning of inbondcall and outbond call |
| 11:30.37 | Somebee | JT: SIP. Its a trunk-account with 10 siplines |
| 11:30.50 | *** join/#asterisk grEvenX (n=even@1ult2p8.ip.hipercom.no) |
| 11:32.00 | Somebee | JT: One of the numbers are lets say 21000000, and it does not find the extension: exten => _NXXXXXXX,1,Answer() |
| 11:32.28 | Somebee | JT: It's directed to the right context, and I have had it working with another provider |
| 11:34.03 | JT | so they probably transmit the number differently |
| 11:34.22 | Somebee | JT: Yep. Seems like it |
| 11:35.58 | rati | i am from india , where i ahve to get voip provider |
| 11:42.00 | *** join/#asterisk bantu (n=Miranda@p54A32B5A.dip0.t-ipconnect.de) |
| 11:49.42 | *** join/#asterisk blinky42 (n=me@67.106.175.130.ptr.us.xo.net) |
| 11:57.55 | *** join/#asterisk __freedom__lover (n=eduardo@clipper.provale.com.br) |
| 11:59.42 | *** join/#asterisk meppl (i=mephisto@meppl.net) |
| 12:00.11 | meppl | so, i downloaded the asterisk-source from from the asterisk-homepage |
| 12:00.21 | meppl | and compiled and installed it |
| 12:00.35 | *** join/#asterisk caio1982 (i=caio1982@CAcert-br/caio1982) |
| 12:00.39 | meppl | and after "make install" asterisk didnt create a init-script |
| 12:00.41 | meppl | start-script |
| 12:00.48 | meppl | is this the normal behaviour? |
| 12:01.07 | *** join/#asterisk harryr (n=harryr@77.240.56.17) |
| 12:01.21 | *** join/#asterisk reber (n=reber@LNeuilly-152-21-126-73.w193-253.abo.wanadoo.fr) |
| 12:02.42 | *** join/#asterisk eserra (i=nobody@89-96-52-24.ip10.fastwebnet.it) |
| 12:02.47 | eserra | hi all |
| 12:03.03 | *** join/#asterisk guillote_GNU (n=bancaria@host225.190-30-159.telecom.net.ar) |
| 12:03.12 | eserra | I'm having problem with ael syntax |
| 12:03.33 | eserra | has recently changed some syntax rule ? |
| 12:03.42 | *** join/#asterisk DarkFlib (n=mike@host90-152-23-30.ipv4.regusnet.com) |
| 12:03.54 | eserra | for example about double quotes usage |
| 12:04.08 | *** join/#asterisk zerohalo (n=zeroHalo@pool-72-70-79-233.bstnma.east.verizon.net) |
| 12:04.23 | *** join/#asterisk saftsack (n=saftsack@pD9E067C3.dip.t-dialin.net) |
| 12:04.31 | eserra | I'm testing a script done with ast 1.4.4 with current trunk and I get a lot of related errors |
| 12:04.33 | *** join/#asterisk zotz (n=zotz@24.244.163.157) |
| 12:04.54 | DarkFlib | hey, just a quickie... trying to match a * in an expression, but it always seems to read it as an operator that should be used not as a string to be compared as.... whats the correct way of escaping it? |
| 12:05.47 | *** join/#asterisk drako (n=ljd@nelug/coreteam/luisjose) |
| 12:07.03 | meppl | ah, in asterisk-src/contrib/init.d/ is no init-script for fedora |
| 12:07.17 | meppl | do you think i just just use the redhat-init-script? |
| 12:07.22 | harryr | meppl: yes |
| 12:07.27 | DarkFlib | meppl: yup |
| 12:07.44 | meppl | thank you |
| 12:08.02 | J4zen | Hm for some reason i am unable to simulate an incoming call using 7777 |
| 12:08.11 | J4zen | ( or any number for that matter ) |
| 12:08.41 | J4zen | This is where the problem lies , i think: |
| 12:08.42 | J4zen | Executing NoOp("SIP/100-09de26b8", "No DID or CID Match") in new stack |
| 12:09.31 | J4zen | As far as i know, it shouldn't require a DID or CID ? |
| 12:11.17 | Fluor_ | Look up what triggers the NoOp(), as the NoOp() itself does not do anything. |
| 12:11.56 | JT | J4zen: freepbx/trixbox? |
| 12:12.00 | *** join/#asterisk [TK]D-Fender (n=joe@MTRLPQ02-1177745839.sdsl.bell.ca) |
| 12:13.56 | *** join/#asterisk reber (n=reber@LNeuilly-152-21-126-73.w193-253.abo.wanadoo.fr) |
| 12:14.05 | *** join/#asterisk stkn|work (n=stkn|wor@gentoo/developer/pdpc.active.stkn) |
| 12:18.01 | J4zen | JT: Yes |
| 12:18.14 | J4zen | Freepbx in trixbox actually |
| 12:18.42 | J4zen | <PROTECTED> |
| 12:18.42 | J4zen | <PROTECTED> |
| 12:18.42 | J4zen | <PROTECTED> |
| 12:18.45 | J4zen | Thats it |
| 12:18.50 | J4zen | and it repeats that over and over again |
| 12:19.01 | [TK]D-Fender | ~freepbx |
| 12:19.02 | jbot | extra, extra, read all about it, freepbx is unable to be supported here. It's a complex piece of dialplan and GUI, and can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking there |
| 12:19.04 | [TK]D-Fender | ~trixbox |
| 12:19.05 | jbot | hmm... trixbox is a full linux distro that includes FreePBX, and other 3rd party add-ons. It is all this extra stuff which makes trixbox seriously painful to support and hence you will find little help here for it. Try asking in #trixbox or on their forums & WIKI at http://www.trixbox.org |
| 12:19.23 | [TK]D-Fender | J4zen: You are in the wrong place for that..... |
| 12:20.59 | J4zen | DID/CID is 100% Asterisk, im merely asking if i need to setup a DID/CID for the functionality to work |
| 12:21.10 | JT | we have no idea |
| 12:21.17 | JT | it's purely a freepbx support issue |
| 12:21.24 | *** join/#asterisk penguinFunk (n=penguin@unaffiliated/penguinfunk) |
| 12:21.31 | J4zen | Alright, i'll ask there |
| 12:21.32 | J4zen | Thanks |
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| 12:41.17 | *** join/#asterisk famicon (i=redz@c51447ddc.cable.wanadoo.nl) |
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| 12:55.43 | *** join/#asterisk waKKu (n=eurulo@unaffiliated/wakku) |
| 12:56.54 | _x86_ | morning |
| 12:58.23 | *** join/#asterisk hermuli (n=Eladamri@xdsl-83-145-207-63.nebulazone.fi) |
| 12:58.31 | *** join/#asterisk FlyboySR22 (n=rsears@hq.fw.americanis.net) |
| 12:59.33 | [TK]D-Fender | INDEED |
| 13:03.51 | *** join/#asterisk zerohalo (n=zeroHalo@pool-72-70-79-233.bstnma.east.verizon.net) |
| 13:05.06 | *** join/#asterisk felix_da_catz (n=felix@PixNationalGroup.phonoscope.com) |
| 13:09.17 | rbd | Hi guys. I'm having a problem with asterisk 1.4.10 (ubuntu 7.10 tribe5) segfaulting after a AMI connection with the starpy AMI interface library (for python). I can make a straight AMI telnet connection and it works fine. however, this same lib version worked fine with a previous version of asterisk. I will try hitting another asterisk box with it, but I wonder if anyone has seen this problem? |
| 13:09.18 | Nugget | telnet is eeeeeeevil! |
| 13:09.49 | *** join/#asterisk shido6 (n=shido6@204.126.120.132) |
| 13:10.36 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
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| 13:16.11 | *** join/#asterisk wubbla (n=wubbla@85-127-179-111.dynamic.adsl-line.inode.at) |
| 13:16.14 | wubbla | hi there! |
| 13:16.40 | *** join/#asterisk stkn|work (n=stkn|wor@gentoo/developer/pdpc.active.stkn) |
| 13:19.29 | __freedom__lover | \scan |
| 13:19.41 | *** join/#asterisk Carlos_Tico (n=grillo_v@c-98-195-31-182.hsd1.tx.comcast.net) |
| 13:19.48 | *** join/#asterisk lirakis (n=lirakis@65.200.191.253) |
| 13:19.50 | Carlos_Tico | i give up with these spa3102 |
| 13:20.11 | Carlos_Tico | is there any device better that can work to do pstn calls ? |
| 13:21.24 | *** join/#asterisk reber (n=reber@LNeuilly-152-21-126-73.w193-253.abo.wanadoo.fr) |
| 13:21.49 | *** join/#asterisk blitzrage (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 13:21.49 | *** mode/#asterisk [+o blitzrage] by ChanServ |
| 13:21.56 | *** join/#asterisk _ShrikE (n=ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 13:22.16 | _x86_ | hmm, zttool is reporting that both of my T1's going to my Sangoma A102D-x are internally clocked... |
| 13:22.58 | _x86_ | but in zaptel.conf, i set one up as 1,0,0,esf,b8zs and the other as 2,1,0,esf,b8zs |
| 13:23.19 | _x86_ | only span 2 should be internally clocked, as it goes to the PSTN |
| 13:23.27 | _x86_ | span 1 goes to a channel bank |
| 13:24.13 | _x86_ | is that normal? I don't have any IRQ misses or bipolar violations, yet seemingly randomly (and not terribly often), someone calling from the channel bank over to the PSTN gets a dropped call |
| 13:25.58 | *** join/#asterisk red9012 (n=marc3234@76-10-149-62.dsl.teksavvy.com) |
| 13:26.30 | *** join/#asterisk frigidzephyr (i=frigidze@nat/digium/x-37bef743cce6bff6) |
| 13:27.21 | [TK]D-Fender | Carlos_Tico: Digium TDM400P, Sangoma A200d |
| 13:28.46 | _x86_ | A20002D-x |
| 13:30.05 | [TK]D-Fender | _x86_: lol.... no thanks... PCI-X limits deployments and adds nothing of value. |
| 13:30.30 | [TK]D-Fender | _x86_: check your clocking in wanpipeX.conf <----- |
| 13:31.55 | _x86_ | [TK]D-Fender: wanpipe1 is set to MASTER (which is the T1 going to the channel bank, should be correct) |
| 13:32.03 | _x86_ | [TK]D-Fender: wanpipe2 is set to NORMAL |
| 13:32.09 | _x86_ | wanpipe2 goes to the PSTN |
| 13:35.32 | *** join/#asterisk putnopvut (i=putnopvu@nat/digium/x-443beb8869e20611) |
| 13:36.29 | *** join/#asterisk anonymouz666 (n=anonymou@201.19.168.48) |
| 13:36.37 | *** join/#asterisk mocker (n=user@198.247.173.227) |
| 13:37.46 | *** join/#asterisk L|NUX (n=linux@unaffiliated/lnux/x-10290) |
| 13:38.02 | [TK]D-Fender | :/ |
| 13:38.12 | *** join/#asterisk anthm (n=anthm@CPE-72-131-113-50.wi.res.rr.com) |
| 13:38.12 | *** mode/#asterisk [+o anthm] by ChanServ |
| 13:38.23 | _x86_ | [TK]D-Fender: any ideas? |
| 13:38.51 | [TK]D-Fender | _x86_: Only 1, and you know it already...... |
| 13:38.51 | _x86_ | [TK]D-Fender: also, wasn't talking about PCI-X at all... was talking about PCIe ;) |
| 13:39.06 | _x86_ | A20002D-x is PCIe, not PCI-X :P |
| 13:39.08 | [TK]D-Fender | also.... *whatever* :p |
| 13:39.12 | _x86_ | lolz |
| 13:39.42 | _x86_ | ok, now the guy is telling me it's not outbound calls dropping, it's inbound... on a A20002D-x |
| 13:39.57 | _x86_ | gah, it'd be insanely helpful if people knew how to report a problem properly |
| 13:39.57 | *** join/#asterisk Juggie (n=Juggie@99.224.171.141) |
| 13:40.07 | _x86_ | I CANT FIX SHIT IF YOU DONT TELL ME WHATS WRONG PEOPLE |
| 13:40.09 | _x86_ | heh |
| 13:40.18 | _x86_ | </rant> |
| 13:40.32 | jer | _x86_, how do you expect people to think of you as a miracle worker? |
| 13:40.43 | *** join/#asterisk duki (n=duki@host-85-27-58-159.brutele.be) |
| 13:41.27 | [TK]D-Fender | JerJer: Beam me up.... there's no intelligent life down here.... |
| 13:41.37 | [TK]D-Fender | jer : rather |
| 13:41.45 | jer | =] |
| 13:41.47 | JerJer | what who?! |
| 13:41.49 | JerJer | :D |
| 13:41.55 | jer | [TK]D-Fender, i was wondering how many people would get that reference |
| 13:43.36 | *** join/#asterisk akaast47 (i=0ca5bc82@gateway/web/cgi-irc/ircatwork.com/x-9b082d89df58dd91) |
| 13:45.24 | akaast47 | I try to set up a now box with asterisk 1.4 and I don't know which linux distribution to choose. I need ztdummy because I don't have hardware. |
| 13:45.56 | [TK]D-Fender | akaast47: Whichever you are most comfortable administering. |
| 13:47.25 | akaast47 | [TK]D-Fender: I am not a linux guru so I am open for any sugestion. |
| 13:48.27 | [TK]D-Fender | akaast47: You'll probably be well served with CentOS as its commonly used so lots of people can help you if there are issues |
| 13:49.25 | *** join/#asterisk bmg505 (n=leon@196.209.181.16) |
| 13:49.29 | akaast47 | [TK]D-Fender: ok. This would be my first choose too. Thanks. |
| 13:50.16 | akaast47 | [TK]D-Fender: Last time I mentioned that I have adio issues with meetme. I get a tdm01b card and looks like the issue is solved |
| 13:50.41 | [TK]D-Fender | akaast47: Glad to here.... this is ANOTHER server, right? |
| 13:50.45 | [TK]D-Fender | hear* |
| 13:50.59 | akaast47 | [TK]D-Fender: yes |
| 13:51.09 | *** join/#asterisk karlh626 (n=kharris@IP-66.249.227.5.indigital.net) |
| 13:51.35 | akaast47 | [TK]D-Fender: I have 44 calls limitation on the Asterisk BE what I don' |
| 13:51.44 | akaast47 | [TK]D-Fender: I have 44 calls limitation on the Asterisk BE what I don't like |
| 13:51.54 | rati | any one can help me |
| 13:51.57 | [TK]D-Fender | akaast47: I don't get it... how are you limited on the # of calls? |
| 13:52.00 | *** join/#asterisk jetlagmk2 (i=jetlag@pool-70-17-59-108.pskn.east.verizon.net) |
| 13:52.20 | akaast47 | [TK]D-Fender: because of the license |
| 13:52.26 | [TK]D-Fender | rati: No, not just anyone... I suggest a trained psychologist... |
| 13:52.27 | rati | i have setup the asterisk@home in VMware , |
| 13:52.36 | [TK]D-Fender | akaast47: WTF?!?! show me where it tells you that... |
| 13:52.43 | rati | i want test is it working or not |
| 13:52.46 | [TK]D-Fender | rati: Asterisk @ home is NOT supported here. |
| 13:52.48 | rati | how??? |
| 13:52.55 | JT | ~freepbx |
| 13:52.56 | jbot | well, freepbx is unable to be supported here. It's a complex piece of dialplan and GUI, and can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking there |
| 13:53.37 | akaast47 | [TK]D-Fender: We get 1 license for Asterisk BE 1.3 and says I have only max 44 calls |
| 13:54.07 | [TK]D-Fender | where does it say this? Thats crazy |
| 13:54.28 | *** join/#asterisk razu (n=razu@razu.data.ee) |
| 13:54.31 | akaast47 | <PROTECTED> |
| 13:54.51 | akaast47 | <PROTECTED> |
| 13:55.03 | JT | lol@abe |
| 13:55.19 | rati | [TK]D-Fender : then , where to i get help |
| 13:55.37 | JT | rati: i already said |
| 13:56.14 | [TK]D-Fender | rati: A@H is very old and outdated now. Trixbox replaced it a long time ago. Go take a look at www.voxilla.com ' s forums and see if they have a topic for that. |
| 13:56.21 | akaast47 | [TK]D-Fender:When we get the BE version I wasn't aware of this |
| 13:56.50 | [TK]D-Fender | akaast47: I've never heard of that till now.... that's Proprietary Crack Edition...... |
| 13:57.41 | akaast47 | [TK]D-Fender:It looks like which bothers me because we spend $1000 for this |
| 13:57.54 | [TK]D-Fender | akaast47: Poor you..... |
| 13:58.05 | [TK]D-Fender | akaast47: and I don't see this limit written anywhere on their site... |
| 13:58.18 | [TK]D-Fender | akaast47: I have to wonder if that has legal ramafications... |
| 13:59.10 | [TK]D-Fender | akaast47: Probably not, but they've buried it so deep that you'll probably sign before getting to it.. |
| 13:59.18 | [TK]D-Fender | akaast47: Nasty nonetheless |
| 13:59.49 | *** join/#asterisk theHub (n=theHub@69.177.93.21) |
| 14:03.40 | rati | <[TK]D-Fender> which one is stable version in trixbox , i open that sit |
| 14:04.02 | *** join/#asterisk Somebee (n=sindre@80.232.5.97) |
| 14:04.05 | [TK]D-Fender | rati: And TRIXBOX is not supported here either. |
| 14:04.07 | [TK]D-Fender | ~trixbox |
| 14:04.08 | jbot | from memory, trixbox is a full linux distro that includes FreePBX, and other 3rd party add-ons. It is all this extra stuff which makes trixbox seriously painful to support and hence you will find little help here for it. Try asking in #trixbox or on their forums & WIKI at http://www.trixbox.org |
| 14:04.50 | *** join/#asterisk ccesario (n=ccesario@189-19-9-100.dsl.telesp.net.br) |
| 14:10.42 | akaast47 | [TK]D-Fender:So now I try to build a new Asterisk 1.4 box |
| 14:11.49 | [TK]D-Fender | akaast47: Good idea.... |
| 14:12.25 | *** join/#asterisk anonymouz666 (n=anonymou@201.19.155.185) |
| 14:12.43 | *** join/#asterisk callguy (n=callguy@pool-72-70-79-233.bstnma.east.verizon.net) |
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| 14:24.47 | *** part/#asterisk jarod14 (n=jarod14@mail.viatelecom.com) |
| 14:27.18 | jgalvin1 | hello |
| 14:28.00 | jgalvin1 | I'm setting up an asterisk server on Debian Etch. It seems to have installed fine, and I'm dialling in using XLite, and I can leave voice mails and that works... problem is I get no sound in playback, no 'welcome' message, no echo |
| 14:28.04 | *** join/#asterisk saftsack (n=saftsack@p54A7417E.dip.t-dialin.net) |
| 14:28.13 | *** join/#asterisk jcmoore (n=jcmoore@unaffiliated/tgrman) |
| 14:29.21 | alrs | jgalvin1: Are you and the Debian machine behind the same NAT? Do you have a Zaptel card of some sort in the machine/ |
| 14:29.22 | alrs | ? |
| 14:29.38 | jgalvin1 | no NAT on the debian machine, yes on this client machine |
| 14:29.42 | jgalvin1 | no zaptel, i'm using FWD |
| 14:29.46 | jgalvin1 | just for testing |
| 14:30.38 | alrs | jgalvin1: I haven't used X-Lite, but I hear that it gets through firewalls pretty well. Still, it might help if you use a STUN server. |
| 14:31.28 | alrs | jgalvin1: Do you have a Linux desktop that you could use with Ekiga? Ekiga offers a public STUN server in the setup process. |
| 14:31.48 | jgalvin1 | don't have it set up at the moment, am on my Mac laptop |
| 14:32.22 | jgalvin1 | i never thought of the firewall here blocking the sounds |
| 14:32.27 | *** join/#asterisk rmayorga (n=rmayorga@unaffiliated/rmayorga) |
| 14:32.31 | alrs | Does the Debian machine have ZTDUMMY loaded/ |
| 14:32.32 | alrs | ? |
| 14:32.34 | jgalvin1 | that would be ok, as long as the server is working |
| 14:32.40 | jgalvin1 | ya i loaded ztdummy |
| 14:33.24 | [TK]D-Fender | jgalvin1: go read this, NOW : |
| 14:33.26 | [TK]D-Fender | ~sipnat |
| 14:33.27 | 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 |
| 14:33.27 | alrs | sometimes audio is lost because of transcoding problems, you might try forcing a different codec. |
| 14:33.28 | [TK]D-Fender | ^^^^^^^^^^^^^^^^^^^^^^ |
| 14:36.04 | jgalvin1 | great, that worked |
| 14:36.12 | jgalvin1 | thanks alrs and d-fender |
| 14:36.26 | jgalvin1 | now to change this annoying default voice... :) |
| 14:38.03 | [TK]D-Fender | jgalvin1: np |
| 14:43.46 | mocker | So what's everyone solution for telecommuters connecting to an office asterisk system? Standard VPN w/ softphone? |
| 14:44.08 | _x86_ | sRTP |
| 14:44.34 | _x86_ | if you can get everything involved to support it |
| 14:44.39 | mocker | _x86_: That to me? |
| 14:45.20 | [TK]D-Fender | _x86_: And some of us like not running trunk and the instability that that brings :) |
| 14:46.21 | mocker | Wow, yeah. svn checkout headaches |
| 14:46.25 | mocker | :) |
| 14:46.33 | _x86_ | [TK]D-Fender: ;) |
| 14:46.56 | _x86_ | [TK]D-Fender: hopefull by 1.10, asterisk natively supports sRTP ;) |
| 14:47.23 | _x86_ | natively and _stable-y_ |
| 14:47.24 | _x86_ | ;) |
| 14:47.33 | [TK]D-Fender | _x86_: Nah... chan_telepathy.so will have made everything else defunct ;) |
| 14:47.47 | *** part/#asterisk Joe_CoT (i=joe_cot@ubuntu/member/joeterranova) |
| 14:47.59 | _x86_ | chan_choakabitch.so |
| 14:49.09 | mocker | It'd be sweet if there was a hardphone that supported OpenVPN. |
| 14:49.20 | _x86_ | yuck |
| 14:49.35 | _x86_ | a phone should not have to ever worry about implementing VPN |
| 14:49.42 | *** join/#asterisk ApEtc (n=apetc@ip70-162-218-46.ph.ph.cox.net) |
| 14:51.05 | *** join/#asterisk xpot (n=jim@c-71-195-241-115.hsd1.ma.comcast.net) |
| 14:51.11 | Nugget | my iPhone does it, but that's not quite what you want, I'm sure. |
| 14:51.48 | *** join/#asterisk deeperror (n=deeperro@69-215-202-195.ded.ameritech.net) |
| 14:54.47 | Carlos_Tico | <[TK]D-Fender> Carlos_Tico: Digium TDM400P, Sangoma A200d --> i have a slimserver cannot connect any pci card i am looking something to put over the network |
| 14:54.49 | _x86_ | Nugget: you have an iPhone or an iBrick? |
| 14:54.57 | Nugget | an iPhone |
| 14:55.00 | _x86_ | nice |
| 14:55.17 | [TK]D-Fender | Carlos_Tico: Whats wrong with your SPA-3102? |
| 14:55.38 | Carlos_Tico | The Echo ...and the quality of PSTN calls Is horrible |
| 14:56.34 | _x86_ | Carlos_Tico: Astribank |
| 14:57.04 | _x86_ | Carlos_Tico: USB channel bank available in a number of different FXO/FXS configurations, up to 32 ports |
| 14:57.22 | _x86_ | Carlos_Tico: vanilla zaptel drivers support it |
| 14:58.13 | jameswf | good morning |
| 14:59.01 | _x86_ | http://www.xorcom.com/products/astribank/astribank_models |
| 14:59.19 | Carlos_Tico | dont have ... usb |
| 14:59.21 | Carlos_Tico | nothing |
| 14:59.33 | _x86_ | your server is crap |
| 14:59.39 | Carlos_Tico | yeah |
| 14:59.43 | Carlos_Tico | its a slim server |
| 14:59.48 | [TK]D-Fender | Carlos_Tico: then prepare to spend REAL money... |
| 14:59.49 | _x86_ | start over with a decent server |
| 15:00.47 | *** join/#asterisk mmlj4 (n=jkelly@ip70-171-92-106.no.no.cox.net) |
| 15:01.32 | *** join/#asterisk Kandinsky (n=cristi@perla2.tm.ew.ro) |
| 15:02.08 | *** join/#asterisk darkfires (n=lwhite@d38-37-97.commercial1.cgocable.net) |
| 15:02.15 | Kandinsky | hello. anyone using BRI ISDN with asterisk? |
| 15:02.15 | darkfires | asterisk[31221]: rc_avpair_new: unknown attribute 1490026597 |
| 15:02.21 | darkfires | getting those msgs like crazy...any ideas ? |
| 15:03.27 | [TK]D-Fender | darkfires: http://lists.digium.com/pipermail/asterisk-users/2007-September/196299.html |
| 15:03.47 | darkfires | i read that |
| 15:03.49 | *** join/#asterisk syzygyBSD (n=chatzill@66.226.228.204.cpe.speedyquick.net) |
| 15:04.01 | darkfires | it doesn't really provide anything useful. |
| 15:04.21 | [TK]D-Fender | darkfires: Trace the thread's responses |
| 15:04.51 | darkfires | I did, there is 1 response |
| 15:06.29 | darkfires | if that response was of any help i would not have asked in the first place |
| 15:06.31 | darkfires | :) |
| 15:06.39 | *** join/#asterisk joetester (n=joeteste@216.191.34.13) |
| 15:07.04 | [TK]D-Fender | darkfires: Well I tried..... |
| 15:07.09 | darkfires | thank you |
| 15:07.22 | darkfires | :) |
| 15:09.47 | darkfires | almost starting to wish i just got a nortel phone sys heh |
| 15:10.12 | *** join/#asterisk DeeJayTwo (n=deejay2@129-233.sh.cgocable.ca) |
| 15:10.35 | [TK]D-Fender | darkfires: What version? |
| 15:11.06 | darkfires | of asterisk? |
| 15:11.19 | [TK]D-Fender | yes |
| 15:11.29 | darkfires | Asterisk SVN-branch-1.4-r85242.... but it happened on r81323 or whatever too |
| 15:11.51 | [TK]D-Fender | darkfires: Try using a "full release" |
| 15:12.16 | _x86_ | darkfires: nortel sucks... at least get an inter-tel |
| 15:12.17 | darkfires | the reason i am using this is because the guy at digium couldn't figure out why zaptel was kernel panicking the system |
| 15:12.48 | *** join/#asterisk Neel007 (n=Nitesh@66.184.39.174) |
| 15:14.04 | _x86_ | darkfires: try a 1.2 release |
| 15:14.21 | _x86_ | darkfires: it's less bleeding edge, and more stable in my experiences |
| 15:14.53 | darkfires | i cant it kernel panics with hpec |
| 15:15.09 | darkfires | i don't even want to use hpec but |
| 15:15.23 | Neel007 | Hello All.... Can anyone please help me with RealTime SIP... I got everything setup for RealTime SIP and my SIP endpoints are also registering fine.... but after sometimes it will loose the registration. The endpoint will able to call out but wont able to receive the call until I have to restart the device. |
| 15:15.37 | darkfires | tdm400p doesn't have hardware echo cancellation |
| 15:16.23 | mocker | Neel007: Do you have qualify=yes for the devices? |
| 15:16.26 | Neel007 | on the *CLI when I do "sip show peers" it shows my endpoint is UNREACHABLE but after sometimes it will show REACHABLE |
| 15:16.47 | mocker | And any NAT keepalive turned on on the phones? |
| 15:17.00 | Neel007 | Yes.. but I changed it to NO... and still have the same problems |
| 15:17.00 | *** join/#asterisk hfb (n=hfb@pool-71-106-219-180.lsanca.dsl-w.verizon.net) |
| 15:18.14 | Neel007 | but why this problem when I only turn on RealTime SIP... using Flat-file config i dont have this problem |
| 15:21.54 | Neel007 | on the Db i have: NAT=yes, QUALIFY=no, CANREINVITE=yes |
| 15:22.08 | *** join/#asterisk nny (n=scott@64.203.239.83.static-pool-4.pool.hargray.net) |
| 15:22.54 | nny | so every time I boot I have to stop asterisk, and urn ztcfg -vvvv and start to get asterisk to recognize the card.. is this a software/asterisk issue or hardware/distro issue? |
| 15:23.00 | Neel007 | and Yes NAT Keep Alive is enabled on the SIP Device. |
| 15:23.01 | nny | well speculative issue |
| 15:24.32 | Carlos_Tico | any one using a spa3102 with asterisk here ? |
| 15:24.55 | [TK]D-Fender | Neel007: you need "qualify=yes", "canreinvite=no", and "nat=yes" |
| 15:25.15 | [TK]D-Fender | nny: Distro. |
| 15:25.24 | DeeJayTwo | Is there a problem with 3 hops reinvites on asterisk? |
| 15:25.29 | [TK]D-Fender | nny: make sure you have a zaptel startup script. |
| 15:25.37 | DeeJayTwo | I mean.. 3 asterisk server inviting each other in serial to a remote UA. |
| 15:25.40 | *** join/#asterisk Corydon76-home (i=five@pdpc/supporter/bronze/Corydon76-home) |
| 15:25.40 | *** mode/#asterisk [+o Corydon76-home] by ChanServ |
| 15:25.41 | DeeJayTwo | 3 middle server |
| 15:25.46 | DeeJayTwo | with SIP |
| 15:25.52 | DeeJayTwo | It's working with 1 server |
| 15:26.20 | DeeJayTwo | as soon as we dial thru 2 or 3 servers... the phone ring...but answering closes the communication in the consoles.. |
| 15:26.24 | *** join/#asterisk Corydon76-dig (i=grey@pdpc/supporter/bronze/Corydon76-home) |
| 15:26.25 | *** mode/#asterisk [+o Corydon76-dig] by ChanServ |
| 15:27.07 | *** join/#asterisk ajohnson (n=ajohnson@63.147.46.186) |
| 15:27.38 | Neel007 | Ok, did that and restarted the device and reloaded sip.. lets ee |
| 15:27.39 | nny | [TK]D-Fender: ok thanks |
| 15:28.06 | Carlos_Tico | any one using a spa3102 with asterisk here ? |
| 15:28.12 | [TK]D-Fender | Neel007: Should be restart SIP, THEn your device |
| 15:28.31 | *** join/#asterisk Dan0maN_Work (n=dschuh@64.149.174.136) |
| 15:28.49 | Kandinsky | anyone using BRI ISDN with asterisk? |
| 15:28.50 | nny | [TK]D-Fender: had a LOT of issues with scripts made by make config.. asterisk init.d scrtip wouldn't create the /var/run/asterisk directory and zaptel script wouldn't unload the modules (not that I need it to, but I have other boxes that do it).. I ended up swapping scripts from a newer distro |
| 15:29.09 | [TK]D-Fender | Carlos_Tico: http://voipedia.pl/index.php/SPA_3102_i_echo |
| 15:29.11 | Neel007 | OK now it says Peer '2488881234' is now REACHABLE! (61ms / 2000ms) |
| 15:29.24 | nny | [TK]D-Fender: biggest problem is we are using 6.06 server ubuntu right now.. trying to decide if it's worth sticking with or switch to Debian server or newer ubuntu |
| 15:29.32 | Neel007 | and now it says Peer '2488881234' is now UNREACHABLE! Last qualify: 61 |
| 15:30.06 | [TK]D-Fender | nny: Ubuntu doesn't use the standard boot process and its custom packages add some risk of their own... |
| 15:30.11 | Neel007 | On the SIP device I do see its registered and I can make calls out... but I can not receive calls in while its UNREACHABLE |
| 15:30.23 | nny | [TK]D-Fender: compiling from source |
| 15:30.28 | [TK]D-Fender | Neel007: is your * behind NAT as well? |
| 15:30.38 | Neel007 | NO its on Public IP |
| 15:30.50 | [TK]D-Fender | nny: Yeh, but * is expecting a SANE distro... and Ubuntu doesn't quite qualify |
| 15:30.52 | Neel007 | only the SIP device is behind NAT |
| 15:30.57 | Carlos_Tico | <[TK]D-Fender> Carlos_Tico: http://voipedia.pl/index.php/SPA_3102_i_echo --- >Thanks pal this echo is a nightmare |
| 15:30.58 | nny | [TK]D-Fender: lol agreed |
| 15:31.04 | [TK]D-Fender | Neel007: What router are they using? |
| 15:31.11 | nny | does anyone here use debian as a base o/s for *? |
| 15:31.14 | *** join/#asterisk Katty (n=Katty@hera.copi-rite.com) |
| 15:31.22 | Neel007 | LinksysWT54G |
| 15:31.24 | [TK]D-Fender | nny: plenty of people |
| 15:31.27 | mocker | nny: lots of people do |
| 15:31.34 | nny | better than ubuntu I imagine? |
| 15:31.35 | [TK]D-Fender | Neel007: Are you doing any port forwarding on it? |
| 15:31.38 | mocker | although I like CentOS for my * |
| 15:31.42 | [TK]D-Fender | nny: Naturally... |
| 15:31.51 | mocker | If TrixBox uses it, it can't be bad! |
| 15:31.52 | mocker | er.. |
| 15:31.54 | mocker | :) |
| 15:31.58 | nny | any particular version recommended for Asterisk? |
| 15:32.01 | Katty | hi. |
| 15:32.09 | Neel007 | No... I havent |
| 15:32.19 | Katty | i have question anyone know about asterisk??????????? |
| 15:32.51 | mocker | Katty: Hell no, we all just come here with questions! |
| 15:32.56 | Katty | *hee* |
| 15:33.08 | Katty | k, done being annoying. |
| 15:33.14 | mocker | blind leading the blind. |
| 15:33.15 | Neel007 | [TK]D-Fender: do I need to forward 5060 ports |
| 15:33.19 | Katty | ....for 5 minutes. |
| 15:33.35 | [TK]D-Fender | Neel007: Ok, what phone are you using? |
| 15:34.06 | Neel007 | [TK]D-Fender: Phone? |
| 15:34.24 | [TK]D-Fender | Neel007: I presume you have a remote phone behind NAT... what is it? |
| 15:35.10 | Neel007 | [TK]D-Fender: its a SPA-2102 connected with some analog telephone set Motorola |
| 15:35.23 | Neel007 | [TK]D-Fender: with US DID on it |
| 15:35.30 | Katty | twisted: ping? |
| 15:36.21 | Kandinsky | anyone using BRI ISDN with asterisk? |
| 15:37.03 | [TK]D-Fender | Neel007: Make sure to turn OFF any NAT settings on it. |
| 15:38.16 | Neel007 | [TK]D-Fender: there is non... just NAT keep alive enabled |
| 15:38.23 | [TK]D-Fender | Neel007: DISABLE that |
| 15:38.30 | Neel007 | ok |
| 15:38.36 | Katty | anonymouz666: :> |
| 15:38.40 | [TK]D-Fender | Neel007: don't tell the SPA anything about its being behind NAT |
| 15:38.45 | anonymouz666 | :> |
| 15:38.57 | Katty | anonymouz666: how's the NY thing coming along? progress? :> |
| 15:39.14 | Katty | [TK]D-Fender: about time! |
| 15:39.37 | anonymouz666 | Katty: yeeeessss! |
| 15:39.40 | _x86_ | hmm... i've got asterisk 1.4.12.1 installed now, but asterisk-addons-1.4 is not available on my distro |
| 15:39.46 | _x86_ | (1.4.12.1 actually was) |
| 15:39.59 | _x86_ | I need mysql support, do i still need asterisk-addons for that in 1.4? |
| 15:40.30 | Neel007 | [TK]D-Fender: Ok now its showing Peer '2488881234' is now REACHABLE! (73ms / 2000ms) |
| 15:40.35 | *** join/#asterisk ManxPower (n=manxpowe@120.sub-70-223-120.myvzw.com) |
| 15:40.52 | Neel007 | [TK]D-Fender: lets wait for couple mintues... and see |
| 15:40.58 | [TK]D-Fender | _x86_: ....DUH :p |
| 15:41.06 | Katty | anonymouz666: yay!! |
| 15:41.22 | Katty | _x86_: how's your mysterious dropped calls problem? :< |
| 15:41.30 | Neel007 | [TK]D-Fender: Peer '2488881234' is now UNREACHABLE! Last qualify: 55 |
| 15:41.39 | Neel007 | [TK]D-Fender: Didnt help.... |
| 15:41.51 | [TK]D-Fender | Neel007: Ok, unless some crazy firewalling is going on, I'm out of ideas |
| 15:42.11 | Neel007 | [TK]D-Fender: firewall on the Asterisk side? |
| 15:42.13 | _x86_ | [TK]D-Fender: didn't know if that got rolled into core or not... *shakes fist* |
| 15:42.31 | _x86_ | why cant they roll in mysql support like they do postgres? *shakes fist more* |
| 15:42.45 | _x86_ | Katty: still happening... mysteriously |
| 15:42.46 | _x86_ | ;) |
| 15:43.05 | [TK]D-Fender | Neel007: EITHER |
| 15:43.30 | Katty | how mysterious! :P |
| 15:44.30 | *** join/#asterisk seele_ (n=seele@1.101.60.190.host.ifxnetworks.com) |
| 15:46.52 | *** join/#asterisk flujan (n=flujan@200.160.115.20) |
| 15:47.28 | mocker | Guh, telecommuting user losing audio 2-3 minutes into a call. |
| 15:49.35 | seele_ | hello I'm trying to make a video call but the call hangs and the other terminal no rings ... my CLI show this with sip debug enabled http://www.pastebin.ca/731972, this is my sip.conf http://www.pastebin.ca/731985, I'm using two tornados M20 .... any suggest to make a success video call ?? |
| 15:49.46 | *** join/#asterisk dps (n=dps@133.64.30.213.rev.vodafone.pt) |
| 15:49.59 | dps | Hey dudes |
| 15:50.05 | Katty | ahem. |
| 15:50.19 | seele_ | the audio call works fine for me, the problem is the video call |
| 15:51.30 | dps | Anyone knows a softphone that can be used on a command line? |
| 15:52.24 | [TK]D-Fender | seele_: What codec did you set on your phones? |
| 15:52.46 | seele_ | H264 |
| 15:53.22 | *** join/#asterisk ted_brown (n=angel@212.145.176.154) |
| 15:53.52 | mocker | dps: linphonec I think? |
| 15:54.02 | dps | wuuuu |
| 15:54.05 | [TK]D-Fender | seele_: nos ure about this : maxcallbitrate=512 |
| 15:54.10 | dps | mocker: ty dude will chech |
| 15:54.13 | dps | check* |
| 15:55.47 | seele_ | [TK]D-Fender, I have tryed with and without this parameter and the result is the same |
| 15:55.47 | [TK]D-Fender | seele_: 403 = bad auth |
| 15:57.41 | *** join/#asterisk blackgecko (n=blackgec@200.36.96.215) |
| 15:57.51 | seele_ | [TK]D-Fender, I accept anonymous calls |
| 15:58.03 | *** join/#asterisk saftsack (n=saftsack@p54A7417E.dip.t-dialin.net) |
| 15:58.20 | [TK]D-Fender | seele_: yes, but it recoginizes the user so its not coming IN as un-auth'd |
| 15:59.11 | blackgecko | im having an issue with incoming calls been hanged up 10 minutes maximum, any idea where can i look ? |
| 16:01.02 | [TK]D-Fender | blackgecko: Yeah... pastebin the CLI output of the entire call with channel debug enabled for every channel type involved. |
| 16:01.05 | [TK]D-Fender | ~pb |
| 16:01.06 | jbot | A Pastebin is a place to paste your stuff 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 |
| 16:01.08 | [TK]D-Fender | ^^^^^^^^^^^^^^^^^^ |
| 16:01.12 | ManxPower | blackgecko: maximum or exactly? |
| 16:01.18 | blackgecko | exactly |
| 16:01.21 | *** join/#asterisk arekm (i=arekm@pld-linux/arekm) |
| 16:01.32 | arekm | hello, is QUOTE() safe for use for file-names quoting? |
| 16:01.34 | ManxPower | blackgecko: Are you setting any timeouts in the dialplan? |
| 16:01.40 | blackgecko | nop |
| 16:01.47 | arekm | uhm, rather no |
| 16:01.53 | ManxPower | blackgecko: then it must be a network or other issue like that. |
| 16:02.28 | blackgecko | mmm network is coneccted via optical fiber |
| 16:02.42 | Nugget | as opposed to what, acoustical fiber? :) |
| 16:02.58 | *** join/#asterisk fordfrog (n=fordfrog@gentoo/developer/fordfrog) |
| 16:03.15 | alrs | ours uses fiberglass |
| 16:03.17 | *** join/#asterisk NigelS (i=nigel@xdev.net) |
| 16:03.28 | alrs | it's from an old speedboat |
| 16:04.15 | [TK]D-Fender | Mine runs on Metamucil. |
| 16:04.30 | blackgecko | ill try to capture the CLI output but it is a time consumig task cause are over 30 simultaneous calls |
| 16:05.08 | *** join/#asterisk Dandre (n=daniel@was59-3-82-236-48-30.fbx.proxad.net) |
| 16:05.15 | Dandre | hello, |
| 16:06.20 | mocker | ~vpn |
| 16:08.02 | Dandre | I have a problem that I don't know how to solve: |
| 16:08.03 | Dandre | I have 2 softphones ekiga on one pc and sjphone on another. Both are registered and as context set to the same value. If I call sjphone from ekiga, the call is established but from sjphone to ekiga I get 404 error. I have no log and nothing in the asterisk console |
| 16:08.30 | [TK]D-Fender | Dandre: enable SIP DEBUG, and pastebin your failed call attempt |
| 16:09.21 | [TK]D-Fender | ~pb |
| 16:09.22 | jbot | A Pastebin is a place to paste your stuff 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 |
| 16:09.22 | [TK]D-Fender | A Pastebin is a place to paste your stuff 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 |
| 16:09.22 | [TK]D-Fender | ^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^ |
| 16:12.03 | arekm | it could be funny if setting callerid name to: ; rm -rf / ; would be possible ;/ |
| 16:12.44 | Dandre | http://pastebin.ca/732013 |
| 16:14.56 | *** join/#asterisk Agnt_0rnge (n=chatzill@mail.enplan.com) |
| 16:15.04 | Agnt_0rnge | <PROTECTED> |
| 16:15.10 | Agnt_0rnge | anyone know what this means? |
| 16:15.25 | *** join/#asterisk atomicd (n=Atomicd@74-206-0-80.static-ip.m.telepacific.net) |
| 16:15.36 | Kandinsky | anyone using BRI ISDN with asterisk? |
| 16:15.48 | Kandinsky | or know who to set up isdn dialplans |
| 16:17.55 | SexyKen | Hey ya'll |
| 16:18.00 | *** join/#asterisk TrentCreek (i=GeekBoy@cpe-70-117-207-168.rgv.res.rr.com) |
| 16:19.46 | SexyKen | Anyone know of a good NAS device for 40TB+? |
| 16:20.00 | harryr | 40+tb! |
| 16:20.30 | *** join/#asterisk atomicd (n=Atomicd@74-206-0-80.static-ip.m.telepacific.net) |
| 16:20.34 | harryr | I can only really suggest Sun StorageTek systems, can't recommend them though - never used one |
| 16:23.16 | harryr | although this one I'm looking at comes in at $130k for 44tb |
| 16:23.32 | Dandre | [TK]D-Fender: her is my pastebin : http://pastebin.ca/732013 |
| 16:24.41 | Agnt_0rnge | Anyone know why a system might constantly go on and off line? |
| 16:26.20 | Somebee | Hi. When I have a trunk-account with 10 sip-lines, and call one of them.. Should I not be able to direct them all to the same inbound-context (the context in [general]) and then go right to the first extension that matches the number that was called? |
| 16:26.24 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 16:27.00 | [TK]D-Fender | Dandre: looking for 6001 in kwdp-000000 (domain 192.168.0.40) <--------- SIP/2.0 404 Not Found |
| 16:27.07 | NigelS | hi guys; I have a volunteer organisation which I help out with - the committee are spread about geographically and face-to-face meetings are relatively rare or between just a few people at a time. Now, I've used asterisk a bit before and was wondering if ppl felt that it would be a good way of trying to get better co-operation/information flow going on. I was imagining giving them all their own acct on the server and holding conferences but also letting people |
| 16:28.11 | [TK]D-Fender | Agnt_0rnge: exactly what it says |
| 16:28.29 | Dandre | ah ok! |
| 16:28.34 | Dandre | thanks |
| 16:29.33 | ManxPower | 44tb? That could hold like all the porn in the world. |
| 16:29.36 | *** join/#asterisk Toerkeium (i=oo@201.216.206.221) |
| 16:29.50 | harryr | ManxPower: nowhere near that much, maybe all porn created in a week |
| 16:30.30 | [TK]D-Fender | ~cisco |
| 16:30.31 | jbot | cisco makes the routers that move the pr0n across the internet at a very high rate of exchange. a company full of patent protected punks! |
| 16:30.53 | ManxPower | [TK]D-Fender: you are such a jbot queen |
| 16:31.55 | [TK]D-Fender | ManxPower: you just love the sound of your own voice (or typeface). I'm EFFICIENT, and have other's do the menial shit for me :p |
| 16:32.04 | ManxPower | ~jbot |
| 16:32.04 | jbot | extra, extra, read all about it, jbot is a hack!, or known to have only said one useful thing. a tool, or dating the fembots, or [TK]D-Fender's b*tch |
| 16:32.25 | *** join/#asterisk grandpapa (n=null@adsl-070-148-053-190.sip.bhm.bellsouth.net) |
| 16:32.26 | *** part/#asterisk meppl (i=mephisto@meppl.net) |
| 16:32.29 | harryr | lol |
| 16:32.30 | grandpapa | Greets mighty baud warriors ... |
| 16:34.19 | *** join/#asterisk mkl1525 (n=qwertz@82.193.235.220) |
| 16:34.54 | dps | Any of you know any project regarding the log of failled calls? |
| 16:35.06 | *** join/#asterisk drwelby (n=mpfister@mail.enplan.com) |
| 16:35.18 | dps | Like... if a call failled, just write it to a file |
| 16:35.28 | lirakis | does any one know of good queue reporting software? ive been talking to the people at AsteriskGuru but they arent very responsive |
| 16:35.46 | [TK]D-Fender | dps: no such thing. |
| 16:35.57 | *** join/#asterisk wunderkin (n=wunderki@ip72-223-86-126.ph.ph.cox.net) |
| 16:36.07 | [TK]D-Fender | lirakis: There's a big list on the WIKI |
| 16:36.14 | lirakis | QueueMetrics.. is really quite expensive... € 1000.00 for 20 users |
| 16:37.03 | lirakis | [TK]D-Fender: ooo orderlystats looks new |
| 16:37.16 | [TK]D-Fender | lirakis: quite old actually... |
| 16:37.34 | mkl1525 | Hi, I'd like to install some (snom) phones on our asterisk, so that callers first enter their number + password and then * directs all calls to the phone - anyone know how to do this? |
| 16:38.55 | [TK]D-Fender | mkl1525: this is all dialplan..... how you do it is up to you. |
| 16:39.23 | *** join/#asterisk GoRK (n=gork@gw.amarillo.energynet.com) |
| 16:40.33 | *** join/#asterisk thieums (n=Mathieu@rny93-4-82-231-54-139.fbx.proxad.net) |
| 16:40.43 | *** join/#asterisk techie (n=techie@adsl-76-240-178-163.dsl.lsan03.sbcglobal.net) |
| 16:41.08 | lirakis | [TK]D-Fender: "big list" .. ? .. there are only like 3 packages that are for queue stats on there .. :\ |
| 16:41.09 | dps | Is it possible to start call from asterisk directly, without using an user agent? |
| 16:41.18 | lirakis | [TK]D-Fender: this is on http://www.voip-info.org/wiki-Asterisk+call+queues |
| 16:41.37 | [TK]D-Fender | dps: Yes. Lookup "call files" and "AMI Originate" on the WIKI |
| 16:42.03 | mkl1525 | [TK]D-Fender, thanks, any keyword to google for? atm the sip accounts are configured on the snoms any I don't know how to put this on the dialplan |
| 16:42.23 | [TK]D-Fender | lirakis: http://www.voip-info.org/wiki-Asterisk+GUI#CallCentreampContactCentreManagementSolu |
| 16:42.28 | GoRK | Hello; I'm having a problem where MixMonitor stops recording when an attended transfer is made. I have a test in my extension macro to start MixMonitor when certain extensions receive calls. So let's say extension 202 receives an incoming call and then begins an attended transfer to extension 201 which is an extension that triggers the MixMonitor recording .. the call announcement will be recorded but mixmonitor will stop when the original call is |
| 16:42.38 | nny | meh so far debian is about the same as ubuntu-server |
| 16:42.40 | thieums | anybody noticed a recent problem with fork application ? |
| 16:42.46 | [TK]D-Fender | mkl1525: Chicken & egg problem then.... this has nothign to do with what kind of PHONES you have. |
| 16:42.46 | ManxPower | GoRK: attended or blind transfer? |
| 16:42.47 | nny | shit i think they are practically the same software |
| 16:42.55 | nny | kernel is newer though |
| 16:42.57 | GoRK | manxpower: attended. Blind transfers record fine |
| 16:43.10 | [TK]D-Fender | mkl1525: You'll have to put some checks in all appropriate extens to see where they should be sending calls to. |
| 16:43.16 | ManxPower | GoRK: I would not expect attended transfers to be automagically recorded. |
| 16:43.39 | ManxPower | as an attended transfer is really a three-way call where 1 person drops out of the call. |
| 16:44.01 | lirakis | [TK]D-Fender: thanks |
| 16:44.08 | ManxPower | I can't think of an easy to do what you want to do, but I'm sure it is possible, just a matter of how much work is involved. |
| 16:44.28 | [TK]D-Fender | mkl1525: I would probably store that routing info using AstDB. So things to read : "show function DB", "show application gotoif", etc |
| 16:45.10 | deeperror | anyone from XPANCE.net in here? |
| 16:45.43 | *** join/#asterisk DrLeech (n=larae@201-212-163-95.net.prima.net.ar) |
| 16:46.10 | DrLeech | hi, anyone knows a good open source SIP software phone for Windows? |
| 16:46.29 | grandpapa | DrLeech: They all pretty much suck. I prefer the suckiness of EzTalk above others. |
| 16:46.40 | GoRK | manxpower: well I found http://bugs.digium.com/view.php?id=7717 which descirbes the problem but does not indicate that it was actually fixed or a workaround was discovered so I'm not sure what to do; the documentation on the Dial option 'n' is not helpful |
| 16:47.17 | [TK]D-Fender | DrLeech: Ekiga |
| 16:47.22 | GoRK | manxpower: in my instance I'm not even using a LOCAL channel |
| 16:47.44 | DrLeech | grandpapa: Yes, I know. But I need to create a custom for use internally in the company, between headquarters. And most of the users use Windows |
| 16:47.56 | DrLeech | [TK]D-Fender, ok, thanks I'll check this too |
| 16:50.06 | grandpapa | Someone needs to release an FTP config based SIP "Batphone", red, preferrably. |
| 16:50.35 | GoRK | manxpower: ah nevermind I think i figured it out; it will be tricky but it will work; the MixMonitor is getting put on the wrong channel.. it's put onto the originating channel instead of the receiving channel, so in reality i could use the local channel to force a mixmonitor to be attached to the channel which is receiving the call |
| 16:50.38 | anonymouz666 | FTP must die |
| 16:50.46 | ManxPower | GoRK: I think the issue is that nobody thinks it is a problem |
| 16:50.53 | grandpapa | Ok, http then, either one. |
| 16:51.24 | seele_ | ok some typo errors solved ... but noting happened this is my sip debug message in the CLI http://www.pastebin.ca/732052 and this is my sip.conf http://www.pastebin.ca/732055 any suggest to make a video call? |
| 16:51.31 | GoRK | manxpower: yeah i actually agree now. It's not intuitive but I believe the current behavior is correct |
| 16:52.36 | *** join/#asterisk saftsack (n=saftsack@pD9E067C3.dip.t-dialin.net) |
| 16:52.38 | [TK]D-Fender | seele_: you don't even have a call ATTEMPT in there.... |
| 16:53.02 | GoRK | manxpower: however I think that the documentation for MixMonitor should be explicit about which channel is going to be monitored and have an option to change which leg the monitor is attached to; I guess I'll put it on my todo list as it would probably be a very straightforward patch |
| 16:54.09 | nny | so debian users, is make config the ideal way to create the init.d scripts??? |
| 16:55.03 | [TK]D-Fender | nny: http://www.voip-info.org/wiki/index.php?page=Asterisk+Starting+and+Stopping |
| 16:55.35 | [TK]D-Fender | nny: "The solutions is to copy the init.d startup script from the contrib folder over to /etc/init.d. Then you’ll need to create the symbolic links by hand. I used the existing symbolic links for Apache2 as a template." |
| 16:55.48 | GoRK | nny: there is a debian init script in contrib/init.d/rc.debian.asterisk |
| 16:55.59 | [TK]D-Fender | nny: for the "make config" scripts that don't install for Debian |
| 16:56.11 | seele_ | [TK]D-Fender, but the audio call works fine .... then the problem are the phones ? |
| 16:56.38 | [TK]D-Fender | seele_: You showed NOTHING that we can comment on. |
| 16:56.40 | *** topic/#asterisk by russellb -> Asterisk: The Open Source Telephony Application Platform -=- Asterisk 1.4.13 (Oct. 10, 2007) -=- Join #asterisknow or #asterisk-gui for AsteriskNOW and Asterisk-GUI info -=- Join #asterisk-commits to monitor svn changes -=- Join #freepbx for freepbx/#trixbox for trixbox support. -=- http://www.digium.com/en/company/switchvox-acquisition-faq.php |
| 16:56.58 | alrs | nny: check out "update-rc.d". It's in /usr/sbin |
| 16:57.03 | alrs | su |
| 16:57.05 | nny | k |
| 16:57.59 | seele_ | someone used a tornado M20 phone ? |
| 16:58.03 | hmmhesays | bah I can't figure out how to filter a single call with rtp in wireshark |
| 16:58.59 | [TK]D-Fender | seele_: Nobody can help you until you provide a USEFUL pastebin. |
| 16:59.32 | GoRK | nny: copy that script to /etc/init.d/asterisk, edit it to suit your install, then install the symlinks with the command 'update-rc.d asterisk defaults' |
| 16:59.38 | seele_ | [TK]D-Fender, what do you need in pastebin ? |
| 17:00.32 | [TK]D-Fender | seele_: to see the entire CALL attempt from beginning to end with SIP debug and verbose 10. |
| 17:00.32 | [TK]D-Fender | seele_: And the configs for all related devices |
| 17:04.28 | seele_ | ok simple audio call |
| 17:04.29 | seele_ | http://www.pastebin.ca/732072 |
| 17:04.56 | nny | GoRK: thanks |
| 17:05.02 | nny | [TK]D-Fender: alrs thanks as well |
| 17:05.29 | ManxPower | seele_: We can't help you with TrixBox/AMP/FreePBX |
| 17:05.53 | ManxPower | and dialedparties.agi is like having a big sign on your back that says "Kick me, I use a GUI." |
| 17:06.03 | ManxPower | We point and laugh at people like that. |
| 17:06.07 | [TK]D-Fender | seele_: are you completely deaf? I said SIP DEBUG. Youa re showing LESS every time. |
| 17:07.19 | seele_ | [TK]D-Fender, yes sorry ... typo error |
| 17:07.31 | [TK]D-Fender | seele_: And no verbose either... |
| 17:09.40 | *** part/#asterisk Dandre (n=daniel@was59-3-82-236-48-30.fbx.proxad.net) |
| 17:09.50 | seele_ | [TK]D-Fender, http://www.pastebin.ca/732081 |
| 17:11.04 | *** join/#asterisk |omni| (n=rob@c-67-185-56-106.hsd1.wa.comcast.net) |
| 17:13.25 | [TK]D-Fender | seele_: and I said the ENTIRE call......... |
| 17:13.54 | nny | so i have asterisk running as root, but safe_asterisk runs as user asterisk, is this normal? |
| 17:15.43 | *** part/#asterisk DrLeech (n=larae@201-212-163-95.net.prima.net.ar) |
| 17:16.13 | [TK]D-Fender | nny: Go look at who has RIGHTS to asterisk, and what the script is doing... |
| 17:16.52 | seele_ | [TK]D-Fender, http://www.pastebin.ca/732090 ENTIRE call |
| 17:18.03 | [TK]D-Fender | seele_: And no verbose again.... I'm clearly wasting my time.... |
| 17:18.13 | *** join/#asterisk Kandinsky (n=cristi@perla2.tm.ew.ro) |
| 17:19.03 | seele_ | [TK]D-Fender, ok no verbose video call log |
| 17:19.07 | seele_ | [TK]D-Fender, http://www.pastebin.ca/732095 |
| 17:20.00 | [TK]D-Fender | seele_: Never mind, this process is just going in circles........ |
| 17:20.23 | [TK]D-Fender | seele_: Go ask on the forums at www.voxilla.com |
| 17:21.37 | seele_ | Entire voice call with no vebose .... http://www.pastebin.ca/732102 |
| 17:21.54 | darkfires | why the hell does NTP take hours and hours before it's ready to serve clients ? |
| 17:22.45 | darkfires | guess ill have to rip that out of the code heh |
| 17:22.53 | [TK]D-Fender | darkfires: ? |
| 17:23.02 | darkfires | my sip phones use ntp servers for time |
| 17:23.35 | darkfires | so if i have to reboot this machine, it takes 4-6 hours before ntp will give out time, otherwise says no servers suitable for synchronization |
| 17:24.03 | [TK]D-Fender | darkfires: NTP servers are supposed to REPSOND witht he time when asked, not jsut send it out blindly. |
| 17:24.22 | darkfires | yes i know that |
| 17:24.25 | darkfires | and responding it is not. |
| 17:24.37 | [TK]D-Fender | darkfires: So what exactly is polling what for NTP? |
| 17:25.21 | darkfires | My aastra 9133i phones poll ntp server XX.xx.xx.xx for the time on bootup, so instead because the box ntp is running on was rebooted it says Jan 1 1970 3:37am |
| 17:25.34 | darkfires | Oct 10 13:20:01 pbx ntpd[5724]: adjusting local clock by 2.242138s |
| 17:25.34 | darkfires | Oct 10 13:23:12 pbx ntpd[5724]: adjusting local clock by 2.178710s |
| 17:25.43 | darkfires | eventually it'll get down to 0.0000 and be ready |
| 17:25.45 | darkfires | and respond |
| 17:26.00 | [TK]D-Fender | DarkFWell if your NTP box can't enev keep its own clock I guess you'[re screwed :| |
| 17:26.11 | [TK]D-Fender | even* |
| 17:26.18 | darkfires | dood |
| 17:26.29 | seele_ | how can I see if my codec h264 is present or enabled? |
| 17:26.42 | darkfires | im going to shove a hayes 1200 up your ass |
| 17:27.23 | [TK]D-Fender | darkfires: I see your 1200 and raise you an accoustic-couple 75 baud + 1956 bellcore phone |
| 17:27.35 | darkfires | funny guy |
| 17:27.36 | darkfires | ;) |
| 17:27.45 | [TK]D-Fender | pwned |
| 17:28.00 | J4k3 | I'll raise you both with psk31 over a 80m shortwave rig. |
| 17:29.14 | *** join/#asterisk zeedo (n=zeedo@reboot-robot.net) |
| 17:29.57 | Kandinsky | anyone whoknows how to configure ISDN on asterisk? |
| 17:31.10 | *** join/#asterisk mitcheloc (n=mitchel@adsl-67-127-235-238.dsl.irvnca.pacbell.net) |
| 17:31.17 | nny | lol fuck me |
| 17:31.36 | nny | whole new system, whole new install, still need to do a ztcfg -vvvv before starting atserisk -_- |
| 17:32.03 | darkfires | um nny |
| 17:32.14 | darkfires | do you have zAPTEL init script ? |
| 17:32.19 | darkfires | sorry caps |
| 17:32.58 | darkfires | the zaptel init script does that for you. |
| 17:33.07 | nny | darkfires: yeah i have it |
| 17:33.15 | darkfires | well you should see why it's not running ztcfg |
| 17:33.18 | nny | used make config |
| 17:33.22 | darkfires | is ZTCFG= |
| 17:33.25 | darkfires | the right path ? |
| 17:33.37 | nny | no |
| 17:33.38 | darkfires | ZTCFG=/sbin/ztcfg |
| 17:33.38 | darkfires | ZTCFG_CMD="$ZTCFG" # e.g: for a custom zaptel.conf location |
| 17:33.42 | nny | thats what it wrong |
| 17:33.44 | nny | us of a! |
| 17:33.55 | darkfires | excuse me, bucko, canada just helped you |
| 17:33.59 | nny | lol |
| 17:34.00 | darkfires | us of A couldn't do jack shit for ya. |
| 17:34.04 | nny | hahaha |
| 17:34.06 | nny | thanks |
| 17:34.12 | darkfires | ;) |
| 17:35.16 | [TK]D-Fender | darkfires: actually US of A did EXACTLY jack-shit for him :p |
| 17:35.16 | nny | was being more facetious than anything else :) |
| 17:35.27 | darkfires | hahaha [TK]D-Fender |
| 17:35.33 | darkfires | touche |
| 17:35.39 | deeperror | is there a way to grep cli output? |
| 17:35.48 | darkfires | cmd | grep str |
| 17:35.49 | darkfires | ? |
| 17:36.06 | darkfires | oh asterisk cli |
| 17:36.13 | deeperror | like as it is in action |
| 17:36.25 | deeperror | like lines with only zap/5-1 in them |
| 17:36.33 | deeperror | or something like that |
| 17:36.41 | darkfires | setup console logging to a file |
| 17:36.43 | darkfires | and grep taht |
| 17:36.43 | nny | darkfires: actually yeah it is correct... woudn't auto complete the first time, but ztcfg is in /sbin/ |
| 17:36.59 | darkfires | nny what distro ? |
| 17:37.06 | darkfires | do you have mulitple ztcfg's ? |
| 17:37.09 | darkfires | multiple |
| 17:37.12 | nny | darkfires: debian |
| 17:37.23 | nny | darkfires: fresh install no multiples |
| 17:37.28 | darkfires | is your /etc/default/zaptel correct |
| 17:37.44 | nny | btw it says "Changing signalling on channel 4 from Unused to FXS Kewlstart" |
| 17:37.45 | darkfires | set DEBUG=yes in default/zaptel |
| 17:37.51 | *** join/#asterisk khronos (n=khronos@aquaman.perryinstitute.org) |
| 17:37.53 | nny | and likewise with channel 3 |
| 17:37.55 | darkfires | nny what hardware? |
| 17:37.56 | nny | when i run it |
| 17:37.59 | nny | tdm02b |
| 17:38.07 | darkfires | so tdm400p with 2 modules ? |
| 17:38.10 | nny | yes |
| 17:38.11 | nny | fxo |
| 17:38.15 | darkfires | the 2 modules are in slots 3 & 4 |
| 17:38.17 | darkfires | ? |
| 17:38.18 | nny | yes |
| 17:38.20 | nny | default |
| 17:38.22 | darkfires | you have to move them into slots 1 & 2 |
| 17:38.27 | nny | i do>? |
| 17:38.32 | darkfires | yep have the same card as you |
| 17:38.34 | darkfires | had so many issues |
| 17:38.38 | darkfires | until i moved them into 1 & 2 |
| 17:38.44 | nny | i have another card here where they are on 1 and 3 -_- |
| 17:38.52 | nny | thats kind of a bitch to predict |
| 17:39.07 | darkfires | it may not be the cure to your problem but it caused me problems |
| 17:39.42 | darkfires | that is my /etc/default/zaptel |
| 17:39.43 | nny | thanks i'll check it out |
| 17:40.02 | darkfires | my /etc/zaptel.conf |
| 17:40.07 | nny | mine has them all commented out except for the tdm driver |
| 17:40.15 | nny | wctdm |
| 17:40.25 | darkfires | yeah |
| 17:40.39 | darkfires | what ver of zaptel |
| 17:41.26 | *** join/#asterisk CunningPike (n=arodgers@204.239.12.183) |
| 17:42.34 | nny | 1.4.5 |
| 17:42.46 | darkfires | u know 1.4.5.1 is out right |
| 17:42.56 | *** join/#asterisk saftsack (n=saftsack@pD9E067C3.dip.t-dialin.net) |
| 17:42.59 | nny | yeah sry box is rebootingh |
| 17:43.01 | darkfires | k |
| 17:43.32 | nny | i have that i use wget to get the zaptel-1.4-current.tar.gz |
| 17:43.57 | darkfires | i have these in my /etc/modules too crc_ccitt and wctdm |
| 17:43.58 | nny | rebooted with zapata and zaptel.conf changed to 1&2, moved modules |
| 17:44.07 | darkfires | work?? |
| 17:44.11 | nny | whats crc_ccitt |
| 17:44.12 | nny | testing |
| 17:44.19 | darkfires | zaptel relies on crc_ccitt |
| 17:44.22 | darkfires | kernel module |
| 17:44.43 | *** part/#asterisk arekm (i=arekm@pld-linux/arekm) |
| 17:45.06 | nny | hmm don't have that first time I have seen it |
| 17:45.17 | darkfires | lsmod | grep crc |
| 17:45.33 | darkfires | zaptel 217328 10 zttranscode,wctdm |
| 17:45.33 | darkfires | crc_ccitt 2112 1 zaptel |
| 17:45.40 | nny | ahh yeah it's running |
| 17:45.45 | nny | crc_ccitt 2304 2 hisax,zaptel |
| 17:45.47 | darkfires | k |
| 17:46.01 | nny | testing now |
| 17:46.13 | nny | didn't have "channels=1-2" in zaptel.conf either |
| 17:46.19 | darkfires | ya thats why i sent u my configs |
| 17:46.23 | nny | only zone info and fxsks |
| 17:46.26 | nny | yeah reading them |
| 17:46.27 | darkfires | just use mine |
| 17:46.32 | darkfires | same config as you |
| 17:46.35 | darkfires | and mine works |
| 17:46.36 | darkfires | :) |
| 17:47.38 | nny | hrrm no love yet |
| 17:47.46 | nny | added channels to conf and rebooted |
| 17:49.16 | *** join/#asterisk VoipMasta (n=fabio@dial-148-240-53-213.zone-2.dial.net.mx) |
| 17:49.36 | VoipMasta | Hi |
| 17:49.51 | nny | darkfires: whelp..no luck there |
| 17:50.02 | VoipMasta | any main advantages/disadvantages of using qualify=no in my sip users definition? |
| 17:50.07 | darkfires | nny can u do md5sum /etc/init.d/zaptel |
| 17:50.25 | nny | 3494f88f8f6e148c995b4736e2312a2b |
| 17:50.40 | [TK]D-Fender | VoipMasta: if your remote client is behind NAT prepare for him to drop off the map.... |
| 17:51.08 | darkfires | also nny i have wctdm and crc_ccitt in /etc/modules, may help too (dont know i fi really need them there but its working) |
| 17:51.09 | VoipMasta | [TK]D-Fender: Most of my remote clients are behind NAT, so should I leave qualify=yes? |
| 17:51.12 | darkfires | that is my /etc/init.d/zaptel |
| 17:51.19 | darkfires | nny you can do sh -x /etc/init.d/zaptel start |
| 17:51.26 | darkfires | to debug it and find out whats going on |
| 17:51.29 | nny | darkfires: k |
| 17:53.47 | VoipMasta | [TK]D-Fender: If I should leave qualify=yes... is there a way to increase the default value? I'm asking because some of my customers have limited bandwidth and sometimes they get +1000ms lag |
| 17:54.37 | [TK]D-Fender | VoipMasta: "qualify=yes" = 2000 |
| 17:54.51 | VoipMasta | in RT? I thought it was 1000 |
| 17:58.15 | nny | god this sucks |
| 17:58.17 | nny | no luck |
| 17:58.23 | nny | need to sacrifice chicken |
| 17:58.57 | darkfires | nny |
| 17:59.03 | TrentCreek | Chicken-Man!!! He's everywher,eeverywhere! |
| 17:59.09 | nny | darkfires: whats the command to start zap in debug? |
| 17:59.18 | darkfires | sh -x /etc/init.d/zaptel start |
| 17:59.22 | darkfires | nny what is your default run level ? |
| 17:59.23 | nny | darkfires: positive note your init.d script actually unloads the drivers |
| 17:59.36 | darkfires | nny: your other one didnt ? |
| 17:59.51 | TrentCreek | SIP devices are so much cheaper and easier to setup |
| 17:59.54 | nny | no they never do, on multuple installs with ubuntu and debian |
| 17:59.58 | nny | multiple* |
| 18:00.03 | darkfires | nny maybe you need to try the svn of zaptel |
| 18:00.12 | nny | darkfires: possibly |
| 18:00.22 | nny | i just can't imagine why this works on every box but this one |
| 18:00.35 | darkfires | murphys law |
| 18:00.39 | nny | i have an install here, on ubuntu desktop 6.06 for god sake, which works fine |
| 18:00.44 | nny | how do I check run level |
| 18:00.47 | darkfires | type runlevel |
| 18:00.52 | nny | N 2 |
| 18:00.56 | darkfires | ls /etc/rc2.d |
| 18:00.59 | darkfires | and paste to me in pm |
| 18:01.04 | nny | -_- |
| 18:01.06 | nny | shit |
| 18:01.08 | nny | that did it |
| 18:01.16 | nny | i need to do an rc.update for zaptel |
| 18:01.17 | darkfires | i think asterisk might be starting before zaptel |
| 18:01.25 | nny | nothing in rc.2 for zaptel |
| 18:01.27 | darkfires | oh |
| 18:01.29 | GoRK | OK if anyone was curious about my mixmonitor/attended transfer problem, I solved the issue by attaching the MixMonitor to the *called* channel via a macro run using the dial option M(macro^args) .. convoluted but works. if anyone needs the specific dialplan lines let me know |
| 18:01.44 | darkfires | ln -s /etc/init.d/zaptel /etc/rc2.d/S15zaptel |
| 18:02.11 | *** join/#asterisk sexyman (n=davetroy@64.240.183.2) |
| 18:02.46 | nny | lol |
| 18:02.50 | nny | darkfires thanks |
| 18:02.56 | Katty | ha hhahahahaha |
| 18:03.13 | Katty | so, the office manager calls me up and says zomgangiei'mnotgettingemailsfromthephoneserveranymoreENDOFTHEWORLD |
| 18:03.28 | Katty | and so i switched it over to my email, tested okay. |
| 18:03.32 | nny | swear the sad thing is i am trying to write a debian/ubuntu 6.06 howto... the ones on the wiki are outdated,.... and I have mangled the shit out of it in the last two days |
| 18:03.34 | Katty | switched it back to hers and tested... |
| 18:03.38 | Katty | she marked the emails as junk mail. |
| 18:03.41 | nny | LOL |
| 18:03.46 | Katty | and then wonders why she's not getting her emails. |
| 18:03.56 | Katty | and top it all off, she tries to blame ME for setting it as voicemail |
| 18:04.00 | Katty | since, ya know, i setup the email server. heh |
| 18:04.10 | tzafrir | nny, what are the bugs you come accross that you have to document? |
| 18:04.12 | nny | heh wow I would have brought a shotgun to work by now |
| 18:05.03 | nny | tzafrir proper linking/ updating init.d scripts.. etc. I have a step by step howto for installing on a clean system.. me and my biz partner have been hammering it out for the last couple of days for our company |
| 18:05.04 | Katty | nny: oh, i have better stories. |
| 18:05.16 | Katty | nny: like the time our furniture sales rep accused me of screwing up his machine. |
| 18:05.25 | Katty | nny: and hacking inot his personal info.. |
| 18:05.29 | Katty | nny: by doing Defrag. |
| 18:05.44 | J4k3 | why why why why why |
| 18:05.47 | nny | lol |
| 18:06.01 | Katty | my personal favorite tho.. |
| 18:06.10 | Katty | the other IT guy that works here, who everything thinks is The Shit.. |
| 18:06.27 | J4k3 | Katty: let his followers find a pile of kiddie pr0n on his computer. |
| 18:06.29 | Katty | because he knows how to install a piece of software onto windows xp pro, put a card in a machine, and connect bnc cables to it... |
| 18:06.37 | Katty | but anytime a computer doesn't work, or a client calls in... |
| 18:06.42 | Katty | he's too scared to go out and fix it |
| 18:06.45 | nny | lol |
| 18:06.47 | Katty | much less reboot any of our servers in the back room |
| 18:06.55 | nny | there are people here who own companies like that |
| 18:07.02 | J4k3 | why should he stop doing what he's doing (wanking to kiddie pr0n) and go fix it? |
| 18:07.04 | nny | we are the only linux shop in the area |
| 18:07.07 | nny | with good reason |
| 18:07.09 | Katty | i was at the gas station getting a soda last week, and one of our servers needed rebooting... they asked him to do it and he said he didn't know how |
| 18:07.17 | Katty | HOW DO YOU NOT KNOW HOW TO REBOOT A WINDOWS SERVER?! |
| 18:07.24 | nny | LOL |
| 18:07.29 | Sci_05 | lmfao |
| 18:07.34 | J4k3 | is he a mac weenie? |
| 18:07.36 | J4k3 | haha |
| 18:07.37 | nny | well it is a very uncommonly used feature |
| 18:07.37 | Katty | no |
| 18:07.44 | Katty | he worked in radio |
| 18:07.50 | Katty | and did a few websites of a local company |
| 18:07.53 | J4k3 | I hope you recorded the conversation |
| 18:07.56 | J4k3 | :) |
| 18:07.57 | Katty | he needed a job, and he was a friend of mine, and i got him hired. |
| 18:08.08 | Katty | taught him how to do the video systems. |
| 18:08.09 | hmmhesays | doing? |
| 18:08.22 | Katty | and now he walks aorund like he's hot stuff. |
| 18:08.27 | hmmhesays | is he? |
| 18:08.28 | *** join/#asterisk kFuQ (n=somedude@c-67-185-112-20.hsd1.wa.comcast.net) |
| 18:08.35 | Katty | he's a fucking moron |
| 18:08.57 | Katty | he went to install a video system at a local daycare... |
| 18:09.12 | *** join/#asterisk joetester (n=joeteste@216.191.34.13) |
| 18:09.18 | nny | should show him zoneminder |
| 18:09.23 | nny | let him try and set that up |
| 18:09.25 | Katty | and the girl wanted to be able to access a workstation at a nother location |
| 18:09.29 | Katty | he sent it to me. |
| 18:09.39 | *** join/#asterisk Kandinsky (n=cristi@perla2.tm.ew.ro) |
| 18:09.47 | Katty | i guess he couldn't just get into the firewall, do a port forward, and turn on remote desktop |
| 18:09.54 | Katty | takes... 2minutes tops |
| 18:09.55 | *** join/#asterisk popvoxdave (n=popvoxda@64.240.183.2) |
| 18:10.02 | Katty | but no, "Oh, you'll have to call Angie for that" |
| 18:10.40 | Katty | he's terrified of my phone server. |
| 18:10.56 | Katty | i guess it has too much Importance for him to accidentally screw up ;) |
| 18:11.18 | nny | lol... i can see how you don't bring a shotgun, it must be highly amusing to watch them flounder |
| 18:11.24 | nny | the bofh would be proud |
| 18:11.26 | *** join/#asterisk apardo (n=apardo@248.64.220.87.dynamic.jazztel.es) |
| 18:11.39 | [TK]D-Fender | guns are too impersonal...... |
| 18:11.42 | *** join/#asterisk gardo (n=gardo@124.217.85.159) |
| 18:11.45 | Katty | i'm not sure what annoys me more. |
| 18:11.57 | Katty | him, or the fact the company just got him a company vehicle with a video surv. logo on the side. |
| 18:12.21 | VoipMasta | I have an issue here, I need to have a call transfered from a SIP phone to an external (PSTN) phone, which is pretty easy... however how can I transfer it back from the PSTN to another SIP extension? |
| 18:12.23 | Katty | he runs 1 xp pro machine as a server. i run 5 windows servers and a linux server. |
| 18:12.26 | Katty | >:( |
| 18:12.32 | Katty | not to mention all the clients stuff |
| 18:13.00 | [TK]D-Fender | VoipMasta: Dial that PSTN call with the "T" option. |
| 18:13.13 | *** join/#asterisk kkjoe (n=kvirc@p509893b3.dip0.t-ipconnect.de) |
| 18:13.23 | nny | darkfires: lol not it either |
| 18:13.35 | nny | damn it all to hell.. why must these machines mess with my emotions |
| 18:13.38 | nny | :) |
| 18:13.51 | VoipMasta | [TK]D-Fender: But can the transfer be initiated by the callee at the PSTN phone? |
| 18:14.05 | nny | Katty: sounds like he needs the blue screen wallpaper and hidden icons on his desktop |
| 18:14.15 | [TK]D-Fender | VoipMasta: When I just hand you the answer like that that its obviously a "YES" |
| 18:14.31 | kkjoe | is there an channel for libpri support ? |
| 18:14.46 | [TK]D-Fender | kkjoe: what are you doing with it? |
| 18:14.53 | VoipMasta | [TK]D-Fender: What happens is that I tried it with no success, I guess the callee phone needs a "flash" key, however mobile phones don't have one |
| 18:15.05 | [TK]D-Fender | VoipMasta: no, they DON'T. |
| 18:15.24 | [TK]D-Fender | VoipMasta: because the flash is only used between them and their TELCO. |
| 18:15.31 | VoipMasta | [TK]D-Fender: so how does the callee start the transfer? |
| 18:15.31 | [TK]D-Fender | VoipMasta: It'd never make it across. |
| 18:15.45 | [TK]D-Fender | VoipMasta: "show application dial" <----------- |
| 18:15.45 | Katty | nny: hahaha. i was thinking more like rename windows.com |
| 18:15.51 | [TK]D-Fender | VoipMasta: Go read |
| 18:15.54 | nny | so let me review.. init.d script doesn't run ztcfg for some reason, at least not properly.. stopping and starting it manually do the same thing, the only way to get it working is to stop asterisk, run ztcfg and restart asterisk.. all my conf files seem to be in order.. |
| 18:16.13 | nny | Katty: heh proxy the connection and flip all the images upside down |
| 18:16.18 | nny | saw that somewhere |
| 18:16.56 | nny | Katty: http://lifehacker.com/software/wifi/turn-your-wifi-piggybackers-internet-upside-down-190441.php |
| 18:16.59 | Katty | maybe i'll just get him fired. |
| 18:17.35 | nny | lol that's effective |
| 18:17.55 | nny | careful though he me get disgruntled and hack teh company internets |
| 18:17.59 | [TK]D-Fender | Katty: Would you like to know the 7 bytes of executable code it takes to wipe a HD clean in a few odd ms? ;) |
| 18:18.07 | *** join/#asterisk rpm (n=russell@75.155.167.90) |
| 18:18.13 | VoipMasta | [TK]D-Fender: Ok thanks, I got it... I wasn't incuding the features.conf file. BTW I see you're not in your most patient mood today :) |
| 18:18.25 | Katty | nny: he wouldn't know how. |
| 18:18.45 | nny | Katty: hehe my sarcasm thingy is switched off :) |
| 18:18.48 | Katty | nny: and i don't know who i'd replace him with.... |
| 18:18.55 | nny | yeah better to mess with him |
| 18:18.58 | [TK]D-Fender | VoipMasta: I'm plenty patient. I passed up all sorts of chances to simply say "RTFM" and hope that you'd even FIND the relevant documentation yourself :p |
| 18:19.04 | Katty | nny: maybe i'll just quit. |
| 18:19.06 | Katty | anyone hiring |
| 18:19.17 | Katty | yes, i do wear skirts. |
| 18:19.20 | nny | lol |
| 18:19.21 | Katty | no, i don't run cable in them. |
| 18:19.24 | nny | LOL |
| 18:19.48 | VoipMasta | Katty: It all depends on the length of those skirts ;) |
| 18:19.49 | Katty | moving to vegas could be fun! |
| 18:19.59 | J4k3 | heh |
| 18:20.03 | Katty | oook, not working for VoipMasta's company. |
| 18:20.05 | nny | lived there, suggest against it now a days |
| 18:20.13 | VoipMasta | lol |
| 18:20.13 | nny | they have crammed homes into every inch of the valley |
| 18:20.14 | J4k3 | I have ex-friends in vegas |
| 18:20.26 | J4k3 | vegas sucks worse than they do. |
| 18:20.29 | nny | lol |
| 18:20.29 | Katty | nny: maybe outside of vegas? |
| 18:20.34 | Katty | nny: but still within driving distance. |
| 18:20.42 | Katty | go hollow out a cactus maybe. |
| 18:20.45 | J4k3 | you realize you get far out of vegas |
| 18:20.48 | nny | Katty: now that'd be cool. other side of hoover dam would be nice |
| 18:20.48 | J4k3 | YOU LIVE IN HELL |
| 18:21.08 | Katty | well it's hell or misery |
| 18:21.11 | Katty | i mean missouri. |
| 18:21.12 | J4k3 | then again, vegas is the center of hell |
| 18:21.20 | J4k3 | personally you couldn't pay me enough to work in that shithole |
| 18:21.28 | J4k3 | I'd work in LA or NYC first |
| 18:21.34 | Katty | nyc is scary. |
| 18:21.40 | J4k3 | less crime, more money, more avoidable dumbshits |
| 18:21.42 | Katty | la is just... weird. |
| 18:21.44 | nny | i am fortunate enough to live on an island, and unfortunate enough to have it populated with complete idiots |
| 18:21.49 | Dan0maN_Work | try austin ;) |
| 18:21.49 | J4k3 | of course, less random tourists to hump |
| 18:21.58 | Katty | ugah. |
| 18:22.00 | Katty | i didn't want to read that. |
| 18:22.05 | J4k3 | Austin has absolutely no redeeming qualities. |
| 18:22.06 | nny | lol |
| 18:22.14 | Dan0maN_Work | lol |
| 18:22.14 | J4k3 | well, except emos |
| 18:22.20 | Katty | emos are everywhere now |
| 18:22.22 | J4k3 | no no |
| 18:22.27 | [TK]D-Fender | Katty: Salvation awaits you above the 49th parallel! |
| 18:22.30 | J4k3 | not Emos as in lamers |
| 18:22.35 | J4k3 | Emos as in emosaustin.com |
| 18:22.44 | Katty | i'll go to alaska! |
| 18:22.44 | Dan0maN_Work | 3 lakes, UT girls, over 350 bars in travis county alone. need i say more?!? |
| 18:23.06 | Katty | and hang out with the peinguins. |
| 18:23.08 | J4k3 | UT girls aren't girls at all |
| 18:23.10 | J4k3 | they just dress that way |
| 18:23.16 | Dan0maN_Work | heh |
| 18:23.17 | nny | is there a way to start the init.d zaptel script and watch what it does, like strace? |
| 18:23.37 | J4k3 | I dunno, maybe I'm defective |
| 18:23.44 | J4k3 | but I can have more fun in dinky little Lufkin, TX |
| 18:23.44 | Dan0maN_Work | not for everyone |
| 18:23.48 | J4k3 | than I ever have in Austin. |
| 18:23.48 | [TK]D-Fender | Dan0maN_Work: Rednecks, Televangelists, wannabe cowboys, I can think of a few on the OTHER side all right... |
| 18:24.25 | Dan0maN_Work | not many cowboys in austin. that's more dallas. |
| 18:24.28 | J4k3 | although I've driven faster in Austin than any other major city, of course you have to wait til 5am to do it because theres so goddamn much traffic. |
| 18:24.31 | Dan0maN_Work | but you're forgetting lesly |
| 18:24.39 | Dan0maN_Work | exactly |
| 18:24.41 | drwelby | Anybody run into the AA50 losing FXO channels, supposedly due to a memory leak in the 1.0.3.1 firmware? |
| 18:24.48 | VoipMasta | [TK]D-Fender: I've already googled and tried to find the info myself... but I couldn't find any, so here's the question: Is there a way to use RealTime with features.conf? I mean to have "dynamic" features? |
| 18:24.49 | *** join/#asterisk zapa (n=hzapa@189.129.201.34) |
| 18:24.56 | Dan0maN_Work | which is why i live right next door to work ;) |
| 18:24.57 | J4k3 | cowboys... sheeit, ever been to amarillo? :) |
| 18:25.03 | *** part/#asterisk zapa (n=hzapa@189.129.201.34) |
| 18:25.06 | J4k3 | the place even SMELLS like cowshit. |
| 18:25.19 | [TK]D-Fender | VoipMasta: This is a Dial PARAMETER. there is nothing beyond that! |
| 18:25.27 | *** join/#asterisk r0d3nt (i=nobody@foster.stonedcoder.org) |
| 18:25.30 | [TK]D-Fender | VoipMasta: Fix your DIAL line. |
| 18:25.36 | *** join/#asterisk exvito (n=exvito@195.245.132.93) |
| 18:25.41 | J4k3 | and dear lord don't try to use your cellphone there... its CellularOne on A and some ghettoest of ghetto CDMA providers on B. |
| 18:25.59 | J4k3 | the CDMA provider was so ghetto I couldn't even figure out who it was... kept eating 611 calls |
| 18:26.00 | Dan0maN_Work | lesly is the "homeless" guy that runs around downtown austin late at night in a thong that ran for mayor |
| 18:26.01 | VoipMasta | [TK]D-Fender: But if I want to change the DTMF sequences that trigger the call transfer (or other functions) dynamically? |
| 18:26.25 | J4k3 | Dan0maN_Work: see, my problem is... "Keep Austin Weird" turned into "Keep Austin Queer" which kinda sucks. |
| 18:26.35 | *** join/#asterisk puzzled (n=patrick@puzzled.xs4all.nl) |
| 18:26.49 | [TK]D-Fender | VoipMasta: AFAIK transfer is a fixed option. ATTENDED transfer is something else. |
| 18:26.51 | Dan0maN_Work | i never was much for the wierd |
| 18:27.09 | J4k3 | and there are just as many posers in Austin now as any other major city. |
| 18:27.13 | [TK]D-Fender | J4k3: What was that comment about "qweers & steers"? ;) |
| 18:27.25 | Dan0maN_Work | ~s/ie/ei/ |
| 18:27.41 | Dan0maN_Work | (saw that work once. dunno how to use it) |
| 18:27.50 | [TK]D-Fender | Dan0maN_Work: no "~" in front |
| 18:27.58 | [TK]D-Fender | Dan0maN_Work: Too late now :) |
| 18:28.05 | Dan0maN_Work | gotcha |
| 18:28.06 | nny | ok so anyone got any bright ideas on how to figure out this zaptel issue |
| 18:28.38 | J4k3 | [TK]D-Fender: not really... basic raw facts of life. |
| 18:28.40 | exvito | hi, does anyone have experience/feedback on Patton's SmartNode 4960 VoIP gateway + SmartLink M-ATAs for doing T.38 FAX over IP ? |
| 18:32.51 | puzzled | exvito: I know http://www.asterisk.pl/ sells them. perhaps call or email them |
| 18:33.25 | exvito | ...I'll have a look puzzled, thanks. |
| 18:34.10 | _ShrikE | exvito: I have used several other patton products, and cant say I have been very pleased with their support. |
| 18:35.49 | exvito | _ShrikE: ...so you've had a not so good support experience; what about the products themselves, would you classify them as below or above average ? (whatever that means) |
| 18:37.07 | _ShrikE | exvito: Given our experiences, we try to stay away from Patton when possible. |
| 18:37.41 | exvito | _ShrikE: ok, thanks for the feedback. :) |
| 18:38.11 | _ShrikE | exvito: We ended up using Audiocodes mediant and MP gateways. They have their own issues, but better than Patton. |
| 18:38.38 | _ShrikE | and are pretty friendly with other devices as far at T.38 goes. |
| 18:38.50 | _ShrikE | as* |
| 18:38.59 | *** join/#asterisk zotz (n=zotz@24.244.163.157) |
| 18:43.18 | nny | meh |
| 18:43.20 | nny | i give up |
| 18:43.25 | nny | this is a bitch |
| 18:43.45 | nny | i fail to grasp what it preventing the init.d script to simply run ztcfg |
| 18:44.12 | VoipMasta | Does anyone know any company offering worldwide t.38 termination? |
| 18:46.55 | nny | heh at least by reading the init script I see that zaptel's script starts hpec -_- |
| 18:47.57 | *** join/#asterisk nestAr (i=nester@makes.all.the.girlies.go.wewt.wewt.net) |
| 18:49.13 | nestAr | Opinion Time.. Dual-Core Opteron 1.8ghz, more than enough for 20 phones, 8 lines (PRI) basic menus and voicemail? |
| 18:49.35 | Strom_M | nestAr: way more than enough |
| 18:49.43 | nestAr | i figured as much |
| 18:50.07 | nny | why god! |
| 18:50.20 | nny | ok bout to put a bounty on this issue |
| 18:51.50 | *** join/#asterisk duki (n=duki@host-85-27-58-159.brutele.be) |
| 18:52.14 | nny | if ANYONE has an idea why i have to manually run ztcfg before starting asterisk, please let me know... i am tired of fucking with this one issue |
| 18:54.38 | [TK]D-Fender | nny: You know you could simply CHEAP and be done with it,,, |
| 18:54.42 | [TK]D-Fender | CHEAT* |
| 18:55.30 | nny | [TK]D-Fender: lol oh i have considered it :) |
| 18:55.56 | [TK]D-Fender | nny: Good... for everything else there's #drphil . get packing! :p |
| 18:56.05 | nny | [TK]D-Fender: but this is going to be our method for many installs, and having this hack in it seems.. wrong i guess |
| 18:56.06 | nny | lol |
| 18:56.09 | nny | is that a real channel |
| 18:56.41 | [TK]D-Fender | Sure it is! |
| 18:57.13 | nny | http://s221.photobucket.com/albums/dd312/clownvan2/?action=view¤t=horridmonkey.jpg |
| 18:57.42 | nny | niice |
| 18:57.46 | [TK]D-Fender | nny: uNF! |
| 18:57.50 | Sci_05 | nny what os are you running? |
| 18:58.11 | nny | Sci_05: debian |
| 18:59.36 | Sci_05 | all you should have to do is toss a simple bash script into /etc/rc3.d/ Name it something like S99ztcfg and do a chmod u+x to that file and it should run when it boots up |
| 18:59.52 | nny | Sci_05: I think it goes to run level 2 |
| 18:59.57 | *** join/#asterisk CVirus (n=GoD@196.205.193.193) |
| 19:00.17 | nny | Sci_05: yeah that was teh hack i was thinking of.. fuck it if it works, so be it |
| 19:00.29 | nny | I can get back to drinking MD20/20 and smoking crack |
| 19:00.39 | Sci_05 | I got mine in 3, After I run the ztcgf stuff I just asterisk -p next so it loads the zt stuff first then * |
| 19:00.53 | Kandinsky | anyone who knows how to configure bri isdn on asterisk? |
| 19:01.32 | *** join/#asterisk outtolunc (n=me@adsl-66-218-53-172.dslextreme.com) |
| 19:01.45 | *** join/#asterisk kFuQ (n=somedude@c-67-185-112-20.hsd1.wa.comcast.net) |
| 19:03.07 | nny | Sci_05: appears there are symlinks to asterisk and zaptel in rc0 through 5, is this normal? |
| 19:03.36 | Sci_05 | nny did you install from debian sources or build from source? |
| 19:03.44 | nny | Sci_05: build from source |
| 19:03.53 | nny | used update-rc to add asterisk and zaptel |
| 19:04.51 | nny | nm I see how rc directories work |
| 19:04.54 | nny | reading READMES |
| 19:04.58 | tzafrir | Sci_05, "zt stuff first"? What exactly is zt stuff by your definition? |
| 19:05.02 | nny | 0 is shutdown |
| 19:05.06 | nny | 1 is single |
| 19:05.10 | nny | 2 is run level on this box |
| 19:05.17 | nny | K# is disabled, S# is start |
| 19:05.21 | tzafrir | nny, I prefer the init.d script in the debs (at least in the recent versions) |
| 19:05.50 | nny | tzafrir yeah considered downloading the package and picking through it |
| 19:06.10 | nny | going to to just have ztcfg run before asterisk and after zaptel in init for now |
| 19:06.29 | *** join/#asterisk ectospasm (n=ectospas@c-68-62-214-33.hsd1.al.comcast.net) |
| 19:06.37 | tzafrir | The zaptel init.d script should simply be run before the asterisk one |
| 19:06.43 | Sci_05 | nny give me a sec and I will post mine for you |
| 19:07.04 | nny | hmm they both are numbered 20 |
| 19:07.06 | Sci_05 | nny your right rc2.d |
| 19:07.22 | nny | but even if i manually invoke both of them in order, it needs to have ztcfg run seperately |
| 19:07.24 | tzafrir | Recent versions of ubuntu should support dependencies between init.d scripts (as in SuSE for very long) |
| 19:07.38 | nny | well this is debian now |
| 19:07.51 | tzafrir | nny, do you have any zaptel hardware of just ztdummy? |
| 19:07.56 | nny | ubuntu got thrown out this morning after trying to fix this error.. |
| 19:08.01 | nny | yes tdm400 with 2 modfules |
| 19:08.06 | nny | modules* |
| 19:08.38 | nny | shit even asterisk starts, but unless I run ztcfg, it never sees the call and logs say "ERROR[26006] chan_zap.c: Unable to open channel 4: No such device or $ |
| 19:08.38 | nny | here = 0, tmp->channel = 4, channel = 4 |
| 19:08.55 | nny | but if i run ztcfg and restart asterisk, all is lovely and wonderful |
| 19:09.04 | Sci_05 | nny: http://pastebin.ca/732199 |
| 19:09.40 | tzafrir | nny, is the zaptel script being run before the asterisk script? |
| 19:10.13 | tzafrir | ls /etc/rc2.d/*asterisk /etc/rc2.d/*zaptel |
| 19:10.24 | nny | tzafrir they both have s20 as their start.. but even if i manually start one than the other it fails |
| 19:10.51 | tzafrir | They both have 20 . And "a" comes before "z" , so Asterisk is run first |
| 19:10.56 | nny | so if i do an /etc/init.d/zaptel start and /etc/init.d/asterisk start i still have to run ztcfg in between there |
| 19:11.11 | tzafrir | So either make Asterisk run later, or make Zaptel run earleier |
| 19:11.12 | nny | i can symlink zaptel to start firts |
| 19:11.15 | nny | first* |
| 19:11.25 | nny | but wouldn't manually running them fix the issue if this was the case??? |
| 19:11.49 | tzafrir | your issue is at boot, right? |
| 19:11.54 | nny | issue is regardless |
| 19:12.30 | nny | in other words i can stop both after boot, and start zaptel, than asterisk, and still have no communication and that error |
| 19:12.43 | nny | but if i run zaptel, do a ztcfg -vvv and then run asterisk, it works |
| 19:13.04 | nny | this is on two different installs, one ubuntu 6.06 and one debian |
| 19:13.15 | nny | i have even checked for IRQ conflicts and whatnot with lspci |
| 19:13.33 | *** join/#asterisk drako (n=ljd@nelug/coreteam/luisjose) |
| 19:13.35 | *** join/#asterisk DrAk0 (n=ljd@nelug/coreteam/luisjose) |
| 19:13.47 | *** part/#asterisk exvito (n=exvito@195.245.132.93) |
| 19:13.54 | nny | I have a feeling Sci_05's suggestion of running it manually between in the init sequence would fix it, and at this point i may just accept it |
| 19:13.56 | TrentCreek | Using my termiantion service, internation extension rings are not occuring. Can asterisk do this? |
| 19:13.57 | Sci_05 | nny look at my pastebin, that should take care of what your looking to do, just uncomment out what you want and it should be all good |
| 19:14.30 | nny | but I am writing a howto for our company and to link to the wiki, and hate to spread the hack.. so i may just word it proper on the wiki and see if others respond |
| 19:14.36 | nny | Sci_05: yeah working on that now |
| 19:14.48 | tzafrir | nny, Just run the zaptel one first. Or The Asterisk one later. What's so complicated abotu this? |
| 19:15.04 | tzafrir | Why try complicated things? |
| 19:15.04 | nny | tzafrir that won't fix it though |
| 19:15.10 | tzafrir | Why? |
| 19:15.12 | nny | tzafrir ok i will try now |
| 19:15.30 | nny | tzafri becuase manually replicating that exact scenario still breaks things |
| 19:16.41 | Norm | has anyone used a D-Link DIV-140 to bridge with PSTN? |
| 19:17.09 | nny | tzafrir rebooting with asterisk set as S30 |
| 19:17.27 | nny | and then i will inject the ztcfg script at 25 (after S20zaptel) is it doesn't work |
| 19:17.35 | nny | which i have a feeling it won't |
| 19:17.44 | nny | I am wondering if maybe HPEC is causing any shit |
| 19:19.30 | nny | tzafrir no worky |
| 19:19.45 | nny | time for Sci_05's suggestion, slightly modified |
| 19:20.18 | *** join/#asterisk saftsack (n=saftsack@pD9E067C3.dip.t-dialin.net) |
| 19:21.21 | *** join/#asterisk keulin (n=cray@ifth-pdcr2.infotheme.fr) |
| 19:21.35 | *** join/#asterisk davevg-btwtech (n=davevg-b@nj-67-76-177-147.sta.embarqhsd.net) |
| 19:23.19 | tzafrir | nny, ztcfg fails? |
| 19:23.22 | Somebee | If I have a "trunk-account" with 10 siplines from my provider, should I need to register more than once (register => ...) to get them all working with inbound calls? |
| 19:24.06 | Somebee | only one of the numbers gets routed to the server, the other 9 does not send one single package to server, even though asterisk says the account is registered ok |
| 19:24.59 | nny | fail |
| 19:25.03 | nny | now i have no love at all |
| 19:25.17 | nny | too much fuzting with shit has only broadened the issue |
| 19:25.21 | nny | let me check my confs |
| 19:26.10 | tzafrir | What error does ztcfg fail with? |
| 19:26.19 | nny | no it works |
| 19:26.36 | nny | asterisk never sees the call coming in |
| 19:27.04 | tzafrir | nny, if it works then you have no problem. But you said earlier that it fails |
| 19:27.26 | nny | no it loads, but if I call the line, astrisk console shows nothing |
| 19:27.40 | nny | before, if i stopped asterisk, ran ztcfg -vvvv and restarted, it would work fine |
| 19:27.53 | *** join/#asterisk tripps (n=ss@66.60.235.100) |
| 19:28.14 | nny | but i have been changing things trying to get to the issue, i just restored my config tarball and ran my handmade permissions script.. trying to get back tothat point |
| 19:29.26 | nny | ok still nothing now |
| 19:29.53 | nny | sonofa |
| 19:29.54 | tzafrir | zap show channels |
| 19:29.56 | tzafrir | anything? |
| 19:29.58 | nny | no udev |
| 19:30.03 | nny | permissions wrong |
| 19:30.04 | nny | -_- |
| 19:30.58 | nny | gah udev rules are correct though -_- |
| 19:31.15 | *** join/#asterisk [hC] (n=hardcore@S01060016b61c8983.vf.shawcable.net) |
| 19:31.16 | nny | asterisk is running as non root btw |
| 19:31.47 | tzafrir | nny, what is the output of: zap show channels |
| 19:31.51 | tzafrir | in the asterisk CLI |
| 19:32.24 | nny | No such command 'zap show' (type 'help' for help) |
| 19:32.50 | nny | asterisk messages says permission denied for opening channels, and the perms in /dev/zap are root |
| 19:33.09 | nny | it appears that the install process doesn't drop the udev rules proper in for debian |
| 19:34.13 | __freedom__lover | hey, i have an intel celeron 2.6 with 256 of mem running freebsd 6.1. i want to know how many calls i can manage simultanely? |
| 19:35.26 | RypPn | I'd bet two before you start dropping handsets |
| 19:35.56 | *** join/#asterisk r0d3nt (n=astrutt@foster.stonedcoder.org) |
| 19:37.16 | [TK]D-Fender | load res_octopoid.so! |
| 19:37.16 | tzafrir | nny, Debian actually has proper zaptel udev rules |
| 19:37.33 | tzafrir | Just add asterisk to the group dialout |
| 19:37.56 | [TK]D-Fender | nny: I am thoroghly impressed. You have turned this little init script issue into a fullday conniption! |
| 19:38.08 | nny | [TK]D-Fender: oh not half as impressed as I am |
| 19:38.18 | nny | it would be fucking nice if I didn't have to chase down these issues |
| 19:39.04 | nny | tzafrir i see that dialout has group perms in that dev node.. just wondering why the zaptel.rules file in udev isn't getting read.. it seems it's name isn't what debian expects it to be |
| 19:39.16 | nny | which is a god dammed bug as far as I am concerned |
| 19:39.31 | Strom_C | nny: odd; ive been running asterisk on debian for three years without a lick of trouble |
| 19:39.57 | nny | Strom_C: so why do I have to fuck with udev rules from a stock install |
| 19:40.01 | tzafrir | nny, you just don't need it. It's unnecessary. I'm also not sure if it has any effect |
| 19:40.09 | Strom_C | nny: I have no idea. |
| 19:40.16 | nny | Strom_C I can send you our howto which line for line dictates the install process |
| 19:40.17 | Strom_C | is this your first time installing asterisk? |
| 19:40.18 | CCFL_Man2 | Strom_C: i figured how to put back together my 5H dial |
| 19:40.24 | nny | Strom_C no, and what version are you running |
| 19:40.28 | Strom_C | nny: sure |
| 19:40.39 | Strom_C | nny: asterisk 1.4 svn branch on debian 4.0 stable |
| 19:41.02 | CCFL_Man2 | Strom_C: now it returns faster than it did since i oiled it, is that a normal thing? |
| 19:41.11 | Strom_C | CCFL_Man2: I have no idea. |
| 19:41.39 | nny | Strom_C well than you can look at our howto and point out the glaringly obvious mistake i am making that you in your year sof apparent glorious use have not run into |
| 19:42.05 | Strom_C | nny: with an attitude like that, I'm apt to tell you to go fuck yourself rather than offer help |
| 19:42.07 | *** part/#asterisk Fluor_ (i=ssmeenk@dot.freshdot.net) |
| 19:42.17 | Strom_C | nny: so chill the hell out |
| 19:42.48 | nny | tzafrir fwiw all /dev/zap/* files have group dialout EXCEPT for transcode.. which has the proper asterisk:asterisk |
| 19:43.09 | nny | but i am adding asterisk to group dialout and adding it to our howto |
| 19:43.42 | Strom_C | nny: please link me to your howto |
| 19:44.19 | tzafrir | nny, not that you need /dev/zap/transcode for anything... |
| 19:45.39 | __freedom__lover | \q |
| 19:50.15 | *** join/#asterisk duki (n=duki@host-85-27-58-159.brutele.be) |
| 19:51.27 | *** join/#asterisk blitzrage (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 19:51.27 | *** mode/#asterisk [+o blitzrage] by ChanServ |
| 19:51.44 | *** join/#asterisk zeedo (n=zeedo@reboot-robot.net) |
| 19:52.34 | *** part/#asterisk zerohalo (n=zeroHalo@pool-72-70-79-233.bstnma.east.verizon.net) |
| 19:53.07 | nny | http://pastebin.com/f168dad95 |
| 19:53.33 | nny | it may be missing some things I have been working on today, been too caught up in these last issues to update it today |
| 19:54.11 | Strom_C | ok |
| 19:54.13 | Strom_C | this is not debian |
| 19:54.15 | Strom_C | this is ubuntu |
| 19:54.18 | nny | actually |
| 19:54.20 | nny | it is debian |
| 19:54.24 | nny | that was for ubuntu |
| 19:54.27 | nny | originally |
| 19:54.54 | nny | but we switched, because ubuntu server wasn't up to par, at least that was suggested |
| 19:55.45 | nny | and* the same exact issue was present in ubuntu fwiw |
| 19:56.07 | Strom_C | ok |
| 19:56.15 | nny | (needing to run ztcfg after zaptel but before asterisk, otherise asterisk would say the channels didn't exist) |
| 19:56.22 | nny | granted now i have permissions issues with udev |
| 19:56.33 | Strom_C | let me wipe this box and try modifying your instructions |
| 19:56.46 | nny | which has been corrected by adding asterisk to group dialout |
| 19:56.59 | nny | (i should* add that on there at least) |
| 19:58.21 | [hC] | nny: how was ubuntu server not up to par? Ive been thinking of moving from debian to it |
| 19:58.28 | [hC] | since debian is getting so stale |
| 19:58.32 | Strom_C | stale |
| 19:58.39 | nny | [hC]: dunno, i have two systems running it in production right now |
| 19:58.44 | Strom_C | what the hell do you need bleeding-edge for on your production servers? |
| 19:58.57 | nny | our office phones have had ubuntu 6.06 desktop (desktop for cryingout loud) running non stop since feburary |
| 19:58.58 | [TK]D-Fender | [hC]: GETTING stale? It is by definition "stale" |
| 19:58.59 | Strom_C | for production servers, i value stability over zomgnew |
| 19:59.15 | [hC] | Strom_C: I dont, but when they include packages that have security holes in them and boxes get rooted, i kinda want out. |
| 19:59.45 | Strom_C | *shrug* i've never had that problem |
| 19:59.54 | [hC] | i hadnt either.. |
| 19:59.58 | [hC] | :) |
| 20:00.02 | nny | [hC]: at this point, I haven't made a concrete decision on which to use, right now, I just am trying to formailize the install process so other monkeys can do what i have done |
| 20:00.18 | Strom_C | and the stable release seems to be up on backporting the security fixes to the existing packages |
| 20:00.26 | [hC] | I mean dont get me wrong i love debian... im not about to try to throw some shiny zomgnew into the mix, but if they do a better job.. |
| 20:01.04 | blitzrage | booooo debian |
| 20:01.11 | [hC] | haha |
| 20:01.18 | [hC] | wait wait.. let me guess.. |
| 20:01.20 | blitzrage | ya that's right -- I said it! |
| 20:01.22 | [hC] | Solaris, javaman? solaris? |
| 20:01.33 | nny | gentoo eh? |
| 20:01.33 | blitzrage | OSX! :D |
| 20:01.37 | nny | windows!!!1 |
| 20:01.40 | blitzrage | gentoo is worse than debian |
| 20:01.44 | nny | lol |
| 20:01.48 | [hC] | blitzrage: yeah! *hugs his osx boxes* |
| 20:02.12 | [hC] | i actually run 1.4 on my laptop in OSX... it works :) |
| 20:02.35 | tzafrir | nny, apt-get install asterisk zaptel-source |
| 20:02.40 | nny | hmm added asterisk to group dialout |
| 20:02.45 | tzafrir | nny, apt-get install asterisk zaptel-source build-essential |
| 20:02.47 | nny | yeah not using 1.2 |
| 20:02.50 | tzafrir | m-a a-i zaptel |
| 20:02.55 | blitzrage | I like Fedora for a desktop, and CentOS for a server... I mainly don't like debian because I don't understand it as well as RH based stuff |
| 20:02.56 | *** join/#asterisk JerJer (n=PhatJ@pdpc/supporter/bronze/jerjer) |
| 20:03.06 | [hC] | blitzrage: get this... chris and I freaked out the airline people, i fired up asteirsk 1.4 and was playing around with asterisk-gui on the plane... I associated a wifi sip phone to my laptop, and chris connected to me with a softphone/headset, and we were talking to each other -- so of course people dont 'get it' and think we're on cell phones/being terrorists |
| 20:03.18 | tzafrir | nny, this would be the same, but with different repos... |
| 20:03.25 | blitzrage | [hC]: lol... hawt :) |
| 20:03.28 | [hC] | on the way back from astricon of course. |
| 20:03.37 | tzafrir | Maintain your own repo |
| 20:03.45 | [hC] | i figured i should put it away when i had about 5 eyes STARING at me |
| 20:04.03 | Strom_C | [hC]: you know, they do tell you to turn off wifi and bluetooth on your laptops |
| 20:04.34 | nny | brb |
| 20:05.29 | *** join/#asterisk dimmik (n=dimmik@static062038217245.dsl.hol.gr) |
| 20:05.31 | [hC] | Strom_C: oh.. i know that... but, who listens? |
| 20:06.06 | Strom_C | you, if you think it's important to comply with federal law |
| 20:06.11 | *** join/#asterisk flewid (n=flewid@mail.flewid.ca) |
| 20:06.14 | flewid | gooday |
| 20:06.31 | [hC] | Well I guess its a good thing im not that concerned about federal law! |
| 20:06.37 | flewid | quick question, is there a simple way to setup an email alert or similar event, when a trunk registration fails? |
| 20:06.46 | [hC] | if i thought it was actually going to hurt anything... |
| 20:06.53 | flewid | (we have multiple providers, and we'd like an email to be sent when one of them is un-registered or can't connect) |
| 20:06.58 | [TK]D-Fender | flewid: No/ |
| 20:07.12 | flewid | yeah i figured as much :/ |
| 20:07.24 | Strom_C | [TK]D-Fender: those tags are legal for you, as the consumer, to remove :) |
| 20:07.28 | flewid | without doing some crazy middleman thing that'd give a status for the registration out to an email |
| 20:07.29 | karleeto | i have 10 polycom 501's, and i need to map my Directory key to a speeddial or something.. i've already mapped the company's 3rd line key to *51 (thier overhead paging key), but now i need to map the directory key to transfer->70 to make it easier for them to park calls. their freakin 80s pbx could park with 1 button push, and i dont want their new expensive system to be harder for them to use |
| 20:08.01 | [TK]D-Fender | karleeto: Not happening. |
| 20:08.13 | karleeto | [TK]D-Fender: why? |
| 20:08.25 | [TK]D-Fender | karleeto: because you can't map multiple actions like that. |
| 20:08.32 | Katty | [TK]D-Fender: linkedin? |
| 20:08.42 | [hC] | Strom_C: nah, I just dont see why anyone should get so worked up.. Its been proven time and time again that those devices dont interfere with anything, its just their way of blanketing the entier group of people and covering their own asses from a perception standpoint... how many people do you think leave that stuff on without even realizing it? or cell phones? I would be 'good about it' and turn it off, like i said, if i thought i |
| 20:08.42 | [hC] | t even mattered. |
| 20:08.53 | Strom_C | karleeto: why are you using *51 for overhead paging? that code is reserved for another use |
| 20:08.57 | karleeto | i couldnt make a function to transfer to 70, and make it *700, then map the key to speeddial *700?? |
| 20:09.05 | Strom_C | [hC]: blah blah blah blah blah |
| 20:09.08 | [hC] | Strom_C: to me, that kind of rule is like being told to drive with your hands at 10 and 2. is it really that necessary? no. |
| 20:09.12 | [TK]D-Fender | karleeto: You'd have to make another linke-key for 700 and then do [transfer] [speed dial to 700] [transfer] |
| 20:09.27 | [TK]D-Fender | Katty: huh? |
| 20:09.34 | dimmik | hey everyone. I am trying to figure this out. When a sip phone is redirecting calls via 302 moved is there any way to restrict it to a specific content. I tried with __TRANSFER_CONTEXT with no luck. |
| 20:09.45 | Katty | [TK]D-Fender: linkedin.com? |
| 20:09.48 | [TK]D-Fender | Strom_C: Don't go all CLASS on his ass..... |
| 20:09.48 | Katty | [TK]D-Fender: are you on there? |
| 20:10.07 | [TK]D-Fender | Katty: nope. |
| 20:10.08 | twisted | ARRRRRGH |
| 20:10.10 | twisted | i hate liknedin |
| 20:10.19 | twisted | it's like the social networking site that never could. |
| 20:10.56 | *** join/#asterisk Chuji (n=brian@mail.point3media.com) |
| 20:11.04 | flewid | [tk]: looks like we were wrong, i just found someone else doing the same thing via a cron script |
| 20:11.30 | Chuji | Say I wanted to busy out a zap channel for a bit. What's the easiest way to get it to join an empty meetme |
| 20:11.40 | [TK]D-Fender | flewid: You haven't validated the EASY part :) |
| 20:11.58 | tripps | hello all . . . i have a mediant 1000 sip gateway behind a FW on the same network as the * box and all the cisco 79xx sip endpoints. is there a reason i shouldn't or can't enable reinvite on all devices? |
| 20:12.04 | *** join/#asterisk stimpie (n=michiel@84-104-5-115.cable.quicknet.nl) |
| 20:12.35 | karleeto | [TK]D-Fender: i could make an extension say *420, then make a little app in extensions.conf for *420 to transfer->70, then map directory key to speeddial *420, right? |
| 20:12.37 | flewid | tk: haha |
| 20:12.49 | flewid | [tk]: copying some script is easy imho :p |
| 20:12.51 | flewid | as long as she works |
| 20:13.01 | [TK]D-Fender | karleeto: No. |
| 20:13.18 | Strom_C | karleeto: read this please |
| 20:13.19 | Strom_C | http://nanpa.com/number_resource_info/vsc_assignments.html |
| 20:13.20 | [TK]D-Fender | flewid: So more like "Was hard for THEM, but my stealing it is EASY!" |
| 20:13.21 | Strom_C | ~vsc |
| 20:13.22 | jbot | [vsc] Vertical Service Codes such as *67, *69, *72, and *82. These codes are generally reserved for specific uses, and it's a bad idea to conflict with the official assignments. A list of assigned VSCs for North America is at http://nanpa.com/number_resource_info/vsc_assignments.html and http://www.nanpa.com/number_resource_info/vsc_definitions.html |
| 20:13.40 | flewid | [tk]: it's not stealing if it's posted on voip-info to use :p |
| 20:13.42 | flewid | but essentially, yes. |
| 20:13.57 | [TK]D-Fender | flewid: Well more power to you then... |
| 20:14.31 | nestAr | lol |
| 20:14.56 | karleeto | [TK]D-Fender: OK, then a 420 app in extentions.conf to transfer->70,wait 7 seconds so they get the parking spot, then transfer again.. why wouldnt that work?? |
| 20:15.01 | [TK]D-Fender | karleeto: Time to tell them TFB <----- |
| 20:15.09 | Katty | twisted: oh :< |
| 20:15.14 | *** join/#asterisk jm|laptop (n=jm|home@dilbert.jamiem.com) |
| 20:15.15 | Katty | twisted: but, i just added you on linkedin :< |
| 20:15.25 | [TK]D-Fender | karleeto: NOTHING is going to make your PHONE tranfer that other call in progress. Its a dead end. Forget about it, |
| 20:15.56 | tripps | [hC]: heh funny about the phones . . . i'm a pilot and you're right about them not interfering. of course i hope they don't change the laws since i would hate to be on a plane and have everyone talking on their cell phones . . . |
| 20:16.28 | karleeto | hmmmm. ok |
| 20:17.32 | *** join/#asterisk ToTo (n=ToTo@62.123.184.142) |
| 20:17.51 | *** join/#asterisk ToTo (n=ToTo@62.123.184.142) |
| 20:21.08 | *** join/#asterisk Assid (n=assid@unaffiliated/assid) |
| 20:21.10 | Assid | hey |
| 20:21.30 | Assid | is there a way to hear gsm codec files on a symbian phone ? if anyoine has tried it |
| 20:21.44 | karleeto | [TK]D-Fender: http://rafb.net/p/tPMX6U63.html |
| 20:21.48 | [hC] | tripps: i agree |
| 20:21.55 | karleeto | WTF is that about then? |
| 20:22.35 | [TK]D-Fender | karleeto: Since when does that let you INTEGRATE the transfer along WITH the number? |
| 20:22.53 | [TK]D-Fender | karleeto: You only get HALF the job done there.... |
| 20:23.12 | [TK]D-Fender | karleeto: You keep looking for "one-touch" and are really jsut not getting it.... |
| 20:23.16 | *** join/#asterisk CrazyTux[m] (n=CrazyTux@adsl-68-92-214-219.dsl.hstntx.swbell.net) |
| 20:24.19 | [TK]D-Fender | karleeto: it would take a minimum of 3 presses to park a call. |
| 20:24.41 | [TK]D-Fender | karleeto: Doesn't matter which way you want to remap them, its still THREE. |
| 20:25.09 | nestAr | but 3 buttons, man, that's a lot! |
| 20:25.10 | nestAr | ;) |
| 20:25.22 | karleeto | [TK]D-Fender: because i could make a macro where you dial *1000 in extensions.conf to transfer->70,wait few seconds,transfer,hangup; then make a speeddial to *1000 |
| 20:25.34 | karleeto | i dont see why that wouldnt work?!!?! |
| 20:25.42 | [TK]D-Fender | nestAr: Like I always say... "cry me a river..... so I can hold your HEAD UNDER" |
| 20:25.48 | nestAr | lol |
| 20:26.14 | [TK]D-Fender | karleeto: No, that doesn't work, because if its a speedi-dial, it will open a NEW channel! that does not affect you CURRENT CALL. |
| 20:26.35 | Chuji | Can anyone think of a way to busy out a zap channel with making it dial an outside number? |
| 20:27.06 | [TK]D-Fender | Chuji: You'd have to invent a way modding chan_zap.so |
| 20:27.21 | Chuji | [TK]D-Fender : OK, that's not going to happen :) |
| 20:27.42 | GoRK | the speed dial could call an AGI that parks the other call on that endpoint via the AMI; but it would be tricky |
| 20:27.45 | [TK]D-Fender | Chuji: actually.... I think you COULD do "Dial(ZAP/1) raw like that... not sure if your telco would cause a disconnect on that though... |
| 20:28.00 | [TK]D-Fender | Chuji: but that might very well do it.. |
| 20:28.15 | Chuji | [TK]D-Fender : With = without |
| 20:28.20 | Chuji | I don't want it to dial out |
| 20:28.31 | Chuji | like have it go offhook directly to a meet me or something |
| 20:28.36 | *** join/#asterisk anonymouz666 (n=anonymou@201.19.160.16) |
| 20:28.50 | [TK]D-Fender | GoRK: Yeah, you COULD try to "guesstimate" the channel and do a hostile transfer on it.... but that is MORE than fugly (risky!) and then again... you don't get the parking lot! |
| 20:28.54 | [TK]D-Fender | (number) |
| 20:29.01 | *** join/#asterisk HCEvan (i=edude@pool-72-77-226-91.tampfl.fios.verizon.net) |
| 20:29.24 | [TK]D-Fender | Chuji: well that IS taking it off-hook, and NOT dialing a number... seems to do what you want... |
| 20:29.45 | GoRK | [TK]D-Fender: unless you have it call the phone back with a ring-answer or auto answer or something then read the number |
| 20:29.47 | GoRK | :) |
| 20:29.54 | *** join/#asterisk zotz (n=zotz@24.244.163.157) |
| 20:30.06 | GoRK | it would carry a challenge though, yes |
| 20:30.19 | karleeto | [karl@asterisk1 ~]$ screen -r |
| 20:30.19 | karleeto | <PROTECTED> |
| 20:30.21 | [TK]D-Fender | GoRK: "terminally impractical" pretty much sums that up :) |
| 20:30.23 | karleeto | 15:21 < karleeto> WTF is that about then? |
| 20:30.25 | karleeto | 15:22 -!- anonymouz666 [n=anonymou@201.19.155.185] has quit [Connection timed out] |
| 20:30.28 | karleeto | 15:22 < [TK]D-Fender> karleeto: Since when does that let you INTEGRATE the transfer along WITH the number? |
| 20:30.31 | karleeto | 15:22 < [TK]D-Fender> karleeto: You only get HALF the job done there.... |
| 20:30.33 | karleeto | 15:23 < [TK]D-Fender> karleeto: You keep looking for "one-touch" and are really jsut not getting it.... |
| 20:30.40 | Assid | hey tkd! how goes it |
| 20:30.40 | [TK]D-Fender | ..... |
| 20:30.43 | Assid | ltns |
| 20:31.28 | [TK]D-Fender | Poor deluded schmuck :p |
| 20:31.41 | [TK]D-Fender | Assid: (not you) Getting by.... |
| 20:32.23 | [TK]D-Fender | Failing to take my advise is typically a very bad thing... people should learn this! |
| 20:32.31 | Assid | hehe.. yep |
| 20:32.46 | Assid | when taking advice.. generally its best to play "leave your brains at home today" |
| 20:33.46 | [TK]D-Fender | ok, heading home.. BBIAB |
| 20:34.45 | *** part/#asterisk lirakis (n=lirakis@65.200.191.253) |
| 20:34.48 | _ShrikE | katty: tomato and green onion reduction over bow tie pasta... mmmmm.. |
| 20:35.27 | Assid | Katty: spinach |
| 20:35.34 | Assid | with some salads |
| 20:35.49 | Assid | or just have a BBQ |
| 20:35.57 | Katty | _ShrikE: "reduction"? |
| 20:36.36 | _ShrikE | dice the tomato and cook it and the diced onions down in their own juices.. plus just a bit of olive oil |
| 20:36.48 | Katty | oh. |
| 20:37.11 | _ShrikE | its light but flavorful |
| 20:38.55 | *** join/#asterisk lesouvage (n=lesouvag@cc341200-a.assen1.dr.home.nl) |
| 20:40.41 | *** join/#asterisk codec (n=codec@iglu.paranoid-penguin.de) |
| 20:43.03 | *** part/#asterisk techie (n=techie@adsl-76-240-178-163.dsl.lsan03.sbcglobal.net) |
| 20:44.14 | Assid | _ShrikE: you a chef? |
| 20:44.44 | nny_away | back |
| 20:44.52 | _ShrikE | not officially but I love cooking. |
| 20:45.10 | Strom_C | nny: I'm building everything and documenting it carefully |
| 20:45.14 | nny | k |
| 20:45.48 | nny | Strom_C still getting permission denied when asterisk is trying to access /dev/zap even though asterisk user is in dialout group and group perms are rw |
| 20:46.07 | Strom_C | nny: yeah, just sit tight |
| 20:46.17 | Strom_C | let me concentrate on this :) |
| 20:46.28 | nny | k |
| 20:47.06 | PSU_Boss | ok, so i have asterisk set up, and i'm on a different network right now using a utstarcom f1000g connected to the asterisk box. if i call other extensions, i hear it ring, and then when i talk, or voicemail answers.. i can't hear it. it's using the u-law codec.. |
| 20:47.44 | PSU_Boss | anyone have an idea as to why it's doing that? also, the softphones i have connected to it work fine.. |
| 20:48.10 | Strom_C | PSU_Boss: ringing is not rtp :) |
| 20:48.18 | Strom_C | PSU_Boss: i'm guessing a sip / nat issue |
| 20:49.43 | hmmhesays | anyone know where I can get a ulaw *.pcap file for SIPP? |
| 20:49.48 | *** join/#asterisk mitcheloc (n=mitchel@adsl-67-127-235-238.dsl.irvnca.pacbell.net) |
| 20:52.39 | lesouvage | Does ${CALERIDNUM} contains the number of the one that starts the phone call or the one that is called? |
| 20:52.59 | lesouvage | ${CALLERIDNUM} |
| 20:53.15 | nestAr | for a call originating in going out? |
| 20:53.22 | nestAr | or for a inbound call |
| 20:53.33 | lesouvage | outbound |
| 20:53.40 | nestAr | the one that starts the call |
| 20:53.52 | nestAr | ${EXTEN} would be the number they are dialing |
| 20:54.28 | nestAr | ex: Dial(IAX2/user@peer/${EXTEN}) |
| 20:54.38 | lesouvage | OK, thanks. I have to change a setting without the change of testing it right now, but this should be the one to choose. |
| 20:55.25 | Katty | i need a nap :< |
| 20:57.44 | Strom_C | lesouvage: ${CALLERID(num)} |
| 20:59.49 | PSU_Boss | <Strom_C> PSU_Boss: i'm guessing a sip / nat issue <-- how can it be a nat issue if it connects to the asterisk server. and i can call the utstarcom phone from one of my other phones and the utstarcom can answer it, but can't talk. |
| 21:00.10 | Strom_C | PSU_Boss: you said you were on a different network |
| 21:00.16 | Strom_C | therefore...it might be NAT |
| 21:00.20 | nestAr | yeah, i'm still using old asterisk, so i'm behind on the variables.. :) |
| 21:02.08 | PSU_Boss | Strom_C, any idea on how i can go about fixing that? |
| 21:02.22 | PSU_Boss | i forwarded the port 5060 on my router to my asterisk box |
| 21:02.42 | Strom_C | PSU_Boss: you also need to forward ports 10000-20000 for the RTP media |
| 21:03.33 | Strom_C | 5060 is signaling only |
| 21:05.20 | PSU_Boss | is that udp or tcp? |
| 21:05.25 | Strom_C | udp |
| 21:06.07 | *** part/#asterisk karlh626 (n=kharris@IP-66.249.227.5.indigital.net) |
| 21:06.39 | *** part/#asterisk deeperror (n=deeperro@69-215-202-195.ded.ameritech.net) |
| 21:07.27 | nny | well got the system stable at least |
| 21:08.19 | nny | (by adding chown -R asterisk:asterisk /dev/zap* to my init scripts -_-) |
| 21:08.29 | Strom_C | nny: I am about two steps away from having a workable solution for you |
| 21:08.52 | nny | Strom_C: cool for what it's worth I am enjoying the education |
| 21:10.45 | nny | so is real time for asterisk ideal or is there certain situations where it works better? |
| 21:11.31 | nny | most of the systems we deploy are under 20 phones, using newer hardware (dual core) gig + of ram etc |
| 21:13.14 | *** join/#asterisk [TK]D-Fender (n=Joe@64.235.216.2) |
| 21:14.50 | hmmhesays | overkill for 20 phones |
| 21:17.17 | blitzrage | nny: generally... the main purpose of realtime is to be able to update configurations from a web gui (in my experience) |
| 21:17.27 | blitzrage | or some other external application that configures the system |
| 21:17.49 | nny | blitzrage: gotcha |
| 21:17.55 | nny | will keep that in mind |
| 21:18.06 | nny | i was just looking at the 550 |
| 21:18.16 | nny | typically sell 501s, but the 550 is pretty damn sharp |
| 21:18.18 | blitzrage | for systems like yours, you probably keep everything in a flat file, and for 20 sets, realtime would just add complexity without any advantages |
| 21:19.03 | nny | hmmhesays: yeah overkill, but the pricing on those systems is 500-600 bucks... considering what people pay for mitel pbx or even old school systems |
| 21:19.33 | *** join/#asterisk duckz (n=duckz@81-31-157-49.vm.dnshosting.it) |
| 21:19.39 | flewid | exit |
| 21:20.03 | [TK]D-Fender | nny, IP 550 is virtually un-suggestable. |
| 21:20.09 | nny | working on a quote right now for a client.. 8 501 phones, optional T1 card or 8 FXO (Leaving that up to them) |
| 21:20.11 | nny | awwww |
| 21:20.15 | nny | but it's so shinny |
| 21:20.19 | nny | guess it's a bitch eh? |
| 21:20.31 | nny | really the only thing I liked most was the backlit lcd |
| 21:20.40 | blitzrage | it's HD voice I think |
| 21:20.41 | blitzrage | 16 bit |
| 21:20.43 | nny | er liked was |
| 21:20.48 | blitzrage | thus... $$$$ |
| 21:20.50 | nny | jesus brain scrambled |
| 21:20.51 | [TK]D-Fender | nny, IP 501 is only suggestable for non PoE environments and those with more than basic needs that the IP 330 can't handle |
| 21:20.59 | nny | hmm |
| 21:21.15 | nny | yeah definitely non poe |
| 21:21.19 | blitzrage | ya... 501 isn't really the best choice for a set anymore... 330 does everything for cheaper |
| 21:21.23 | alrs | [TK]D-Fender: does the 330 have the same echo can stuff that the 501 has? |
| 21:21.49 | nestAr | what's wrong with the 550's? |
| 21:22.02 | jcanfield | would be nice to see g.722 on the 330. |
| 21:22.10 | [TK]D-Fender | nestAr, simply not worth the money compared to otehr models. |
| 21:22.26 | nestAr | still in the box at this point... |
| 21:22.31 | blitzrage | G.722 doesn't make a lot of sense unless someone else has it :) |
| 21:22.32 | [TK]D-Fender | G.722 = only in-office, and we're talking about a bloody PHONE here... |
| 21:22.52 | jcanfield | yup...but makes for nice in-office calls. |
| 21:22.53 | PSU_Boss | Strom_C, i forwarded port 10010 to the asterisk box, which is the port that my softphone and the f1000g is using. but when i make a call from the softphone, that port in iptables doesn't show any activity. |
| 21:23.09 | GoRK | g.722 goes on ISDN also |
| 21:23.12 | [TK]D-Fender | blitzrage, I advocate the 320 over the 330. Invest the difference in WIRING. |
| 21:23.17 | jcanfield | 90% is calls here are inter-office. |
| 21:23.24 | nny | so 330 is poe only? |
| 21:23.30 | PSU_Boss | i also forwarded the entire port range, and that shows a little bit of activity, but not at the moments when i make the calls |
| 21:23.43 | blitzrage | [TK]D-Fender: I think I meant the 320... not the 330 :) |
| 21:23.45 | JT | GoRK: since when? |
| 21:23.46 | Dan0maN_Work | nny: you can buy an adapter for it |
| 21:23.48 | [TK]D-Fender | nny, no, it does PoE natively, and you can get a brick cheap for it |
| 21:23.50 | jcanfield | nny: it has wall wart. |
| 21:24.51 | nny | anyone get a chance to setup the Kirks yet? |
| 21:24.55 | mcab | 301/320/330/430/501/550/601/650/4000 |
| 21:24.58 | nny | have a doctors office who wants em |
| 21:25.02 | JT | GoRK: ISDN is G.711 |
| 21:25.02 | *** join/#asterisk Corydon76-vcch (n=tilghman@pdpc/supporter/bronze/Corydon76-home) |
| 21:25.03 | *** mode/#asterisk [+o Corydon76-vcch] by ChanServ |
| 21:25.33 | [TK]D-Fender | IP 301 = no point at all any more |
| 21:26.16 | Strom_C | nny: http://pastebin.com/d6c00b208 |
| 21:26.33 | Strom_C | i have it working just fine on the box to my left |
| 21:26.41 | nny | Strom_C i'll try it out, thanks, and I appreciate the help |
| 21:26.42 | _charly_ | hi, is there a 64bit version of idefisk or zoiper somewhere? i tried the 32bit version, but after starting it i only get a "Floating point exception" |
| 21:26.59 | Strom_C | nny: please let me know if it works |
| 21:27.00 | _charly_ | a 64bit version for linux |
| 21:27.07 | nny | Strom_C: will do |
| 21:29.01 | GoRK | JT: More importantly ISDN is 64kb. You can run other codecs over it if you want. I did some radio remote stuff 9 years ago and we could use L2 MPEG or G.722 with the unit we had |
| 21:29.44 | GoRK | JT: "normal" voice calls made over ISDN use uLaw though.. but a call between ISDN endpoints can do all kinds of stuff |
| 21:30.23 | Strom_C | nny: also, "debian sarge" should be "debian 4" |
| 21:30.33 | Strom_C | brain was running on autopilot for that one :) |
| 21:30.50 | Sci_05 | !book |
| 21:30.53 | Sci_05 | ~book |
| 21:30.54 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf |
| 21:33.44 | nny | Strom_C: installing right now using debian-40r1 iso |
| 21:33.51 | Strom_C | nny: great |
| 21:33.53 | nny | Strom_C: we are also installing hpec |
| 21:34.05 | Strom_C | ok |
| 21:34.11 | nny | should be able to tell you how it went in about 20 minutes |
| 21:34.18 | Strom_C | i dont have hpec here, so you'll have to modify the instructions to work with that |
| 21:34.28 | Strom_C | (perhaps) |
| 21:34.49 | nny | Strom_C: no problem,.. if all works well I can put it on the wiki somewhere as well, i'll be sure to credit you |
| 21:35.03 | BadHorsie | so should i just go straight for that book? or would it be good to read first TFOT 1st edition? |
| 21:35.12 | Qwell | 2nd edition |
| 21:35.32 | BadHorsie | nice |
| 21:35.51 | Strom_C | nny: no, i'll take care of doing that |
| 21:35.59 | Strom_C | but thanks for the offer |
| 21:36.55 | *** join/#asterisk |R (i=bob@modemcable241.28-203-24.mc.videotron.ca) |
| 21:37.41 | *** join/#asterisk zeedo (n=zeedo@reboot-robot.net) |
| 21:40.18 | Agnt_0rnge | is there a command to ping everyones phone so they will register? |
| 21:41.24 | wishes | why would their client not register? |
| 21:41.33 | wishes | or you mean re-register ? |
| 21:41.48 | Agnt_0rnge | ya |
| 21:42.16 | Agnt_0rnge | so when the phones get disconnected you dont have to wait and or have them unplug and replug in the phone. |
| 21:42.34 | wishes | most phones you can set to re-register at x amount seconds |
| 21:42.48 | wishes | or hangup/pickup a couple times makes it happen |
| 21:42.59 | wishes | or was it calling somebody? something like that |
| 21:46.03 | Agnt_0rnge | It seems when changes are made in the system and you save them, some people loose their connection. |
| 21:46.26 | *** join/#asterisk lunaphyte__ (n=lunaphyt@0158ahost161.starwoodbroadband.com) |
| 21:46.57 | Agnt_0rnge | so instead of having them power cycle can I ping them and force the phone to re-register....not sure if register is the right term |
| 21:50.00 | nny | Strom_C: wondering if using SVN will maigcally fix some of the issues i was having |
| 21:50.42 | Strom_C | nny: i dont think it was asterisk's fault ;) |
| 21:50.53 | Strom_C | nny: these directions should work fine with tarballs too |
| 21:51.24 | nny | Strom_C: hmm I will use svn, although other than the part we had for editing the asterisk makefile to use /var/run/asterisk, the rest seems very similar |
| 21:51.38 | *** join/#asterisk Tommy3 (n=Tommy2@66.0.46.210) |
| 21:52.15 | PSU_Boss | Strom_C, i forwarded the ports required and it still doesn't work. |
| 21:53.13 | PSU_Boss | the thing i don't understand is how a softphone connected to the same network that the utstarcom f1000g is, works when the f1000g doesn't |
| 21:58.29 | *** join/#asterisk Flauto (n=zhao@71.194.141.225) |
| 21:59.10 | nny | Strom_C: on make menuselect for zaptel.. I assume apart from the wctdm i need little else |
| 21:59.15 | Flauto | hi all |
| 21:59.16 | nny | or nothing else for that matter |
| 21:59.24 | Strom_C | nny: that depends on what you have installed |
| 21:59.33 | nny | tdm02b (tdm400 with 2 fxo) |
| 21:59.35 | Strom_C | you probably want local channel support |
| 21:59.52 | nny | whats pciradio? |
| 21:59.56 | Strom_C | unnecessary |
| 22:00.24 | nny | whats local channel support module? |
| 22:00.31 | *** join/#asterisk kgx (n=kgx@60.234.20.178) |
| 22:00.32 | Strom_C | for using the LOCAL channel type |
| 22:00.37 | Strom_C | i'd recommend you compile that |
| 22:00.38 | nny | no i mean which is it |
| 22:00.44 | Strom_C | it's at the bottom |
| 22:00.49 | Strom_C | it'll say "local" |
| 22:00.57 | nny | ztd-loc ? |
| 22:01.03 | Strom_C | yeah, i think so |
| 22:01.04 | nny | nothing specifically says local |
| 22:01.14 | Strom_C | the description will say "local channel support" |
| 22:01.15 | nny | what about ztd-eth ? |
| 22:01.35 | Strom_C | ......is there not a description field at the bottom of the screen as you move the cursor to each of these? |
| 22:01.38 | nny | in zaptel modules? |
| 22:01.44 | nny | yes but nothing says local channel support |
| 22:02.08 | Strom_C | ztd-loc |
| 22:02.09 | Strom_C | says |
| 22:02.13 | Strom_C | Local Virtual Span |
| 22:02.19 | Strom_C | i'd say that's pretty damned clear |
| 22:02.25 | nny | heh Local Virtual Span ! local channel support |
| 22:02.44 | nny | i mean i am sure they mean the same, i was taking you literally |
| 22:02.57 | Strom_C | sigh |
| 22:03.02 | *** part/#asterisk Tommy3 (n=Tommy2@66.0.46.210) |
| 22:03.09 | Flauto | does 1.4 come with some kind of call recording function? |
| 22:03.17 | Strom_C | no, i do not have the zaptel menuselect screen memorized |
| 22:03.19 | nny | what about zttranscode |
| 22:03.19 | Strom_C | Flauto: yes |
| 22:03.29 | Strom_C | nny: you dont need it unless you have a transcoder card |
| 22:03.29 | Flauto | strom, anything i can read on? |
| 22:03.35 | Strom_C | Flauto: MixMonitor() |
| 22:03.45 | nny | i didn't think so, i was being cautious as to follow your instructions as precise as possible |
| 22:04.39 | Strom_C | then really all you need is wctdm, ztd-loc, and ztdynamic |
| 22:04.44 | kgx | hey. i need to call an agi script when someone picks up a phone from a que. also need to pass the sip_id of the person who pick up the call. anyone knows how i can do this? this doesnt work: Queue(support-queue|trn|||15|AGI(script.php|sip_id=${CHANNEL})) |
| 22:04.50 | nny | k |
| 22:04.51 | Flauto | thanks, i am searching for mixmonitor now. strom_c |
| 22:05.17 | Strom_C | kgx: because ${CHANNEL} is the channel of the calling party, not the called party |
| 22:05.49 | nny | Strom_C: processor is am2, which hpec module (nothing quite says K7) is best, they have i686, 586, 386 athlon, athlonxp |
| 22:06.01 | nny | caution again to avoid any further issues |
| 22:06.09 | Strom_C | nny: I have no idea; i've never used hpec |
| 22:06.20 | kgx | Strom_C: hmm, it did seem to work for my macros though. but thanks for letting me know. so how do i pass the sip id? |
| 22:06.44 | Strom_C | kgx: beats me |
| 22:07.13 | anonymouz666 | Oct 10 19:05:26 DEBUG[25388]: chan_sip.c:11539 sipsock_read: SIP message could not be handled |
| 22:07.15 | anonymouz666 | nice |
| 22:07.32 | anonymouz666 | asterisk can't just handle REFER SIP Messages |
| 22:10.27 | *** join/#asterisk adorah (n=Michael@87.69.57.246.cable.012.net.il) |
| 22:17.05 | BadHorsie | i wonder if this channel have some cheat sheet for asterisk, say, for the sake of space and the lack of paper for printing |
| 22:17.19 | BadHorsie | some sort of objective view of asterisk i meant. |
| 22:17.30 | fujin | ~thebook |
| 22:17.30 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf |
| 22:17.44 | fujin | cheat sheet = your brain |
| 22:18.58 | jameswf | google = my cheatsheet |
| 22:20.08 | Flauto | is there a way that i can turn on/off mixmonitor in the middle of a conversation? |
| 22:20.49 | nny | Strom_C: rebooting |
| 22:21.11 | *** join/#asterisk zeedo (n=zeedo@reboot-robot.net) |
| 22:21.13 | nny | Strom_C: followed it pretty much to a t |
| 22:21.19 | nny | didn't even use asterisk addons |
| 22:21.25 | nny | (no need really) |
| 22:21.30 | *** join/#asterisk flujan (n=flujan@200-160-115-020.static.spo.ctbc.com.br) |
| 22:21.41 | flujan | hi guys... anyone seeing this error: |
| 22:21.42 | flujan | http://forums.digium.com/viewtopic.php?t=18413&highlight=&sid=7971631e60b4729c53c23d9efb717222 |
| 22:21.50 | flujan | I am having the same problem here!!! |
| 22:22.32 | Strom_C | flujan: its telling you that it doesn't need to write an unnecessary CDR entry |
| 22:22.48 | flujan | hum... so it now problem using it? |
| 22:23.01 | Strom_C | huh? |
| 22:23.06 | flujan | I also noticed that my connection with the pgsql is consuming 100% of the CPU. |
| 22:23.07 | *** join/#asterisk metfan2007 (n=metfan20@189.135.175.112) |
| 22:23.26 | flujan | 5480 postgres 25 0 77944 68m 67m R 100 1.8 7:06.49 postgres |
| 22:23.33 | nny | Strom_C: perm errors in asterisk messages for /dev/zap |
| 22:23.42 | metfan2007 | hey, there's a bug in the last zaptel 1.4.5 version... do you know it?? |
| 22:23.45 | nny | Strom_C: same problem i was experiencing before :) |
| 22:23.53 | metfan2007 | it does not create the /etc/zaptel.conf file!! |
| 22:23.56 | flujan | postgres 5480 99.5 1.7 77944 70476 ? Rs 19:16 7:19 \_ postgres: totalip totalipdb 192.168.1.8(34078) INSERT |
| 22:24.03 | metfan2007 | I mean, zaptel 1.4.5.1 |
| 22:24.08 | Strom_C | nny: odd; what did you do differently than my instructions? |
| 22:24.13 | nny | no!!! |
| 22:24.14 | flujan | Strom_C: this behavior appear after 1.4.12 upgrade |
| 22:24.15 | flujan | :( |
| 22:24.37 | nny | and i checked, asterisk is member of dialout, and dialout has group perms on /dev/zap |
| 22:26.23 | Strom_C | nny: I sent you a PM |
| 22:26.43 | nny | Strom_C: got it.. answered, pidign sucks |
| 22:27.28 | nny | Strom_C: Stand by, I loaded my stock premade configs, but haven't changed zapata.conf yet, someone suggested i had to have modules in 1 and 2 in order to work, old premades still have 3,4 |
| 22:28.08 | nny | so not same issue |
| 22:28.17 | Strom_C | nny: well yeah, the instructions do kind of assume you have zapata and zaptel configured correctly :) |
| 22:28.23 | nny | hopefully all is well now, and your howto works.. personally I wanna move forward myself |
| 22:28.23 | [TK]D-Fender | metfan2007, And why would it be creating /etc/zaptel.conf? |
| 22:28.42 | nny | Strom_C: heh they were before reinstall, but i have a tarball here with dialplans, etc already setup |
| 22:29.04 | nny | Strom_C: and by default, the tdm02b comes on channel 3,4 from digium |
| 22:29.24 | nny | but someone earlier stated that wouldn't work (i tried, in spit of my instinct and the fact i have other boxes on 3,4) |
| 22:29.43 | Strom_C | nny: whoever said it wouldn't work is a moron |
| 22:30.01 | anonymouz666 | ok, I was wrong. |
| 22:30.02 | anonymouz666 | * can handle incoming refer requests |
| 22:30.06 | nny | lol i think they were just trying to help, but yeah our office * box has 1 and 3 (don't ask why) |
| 22:30.06 | anonymouz666 | not sure about outgoing. |
| 22:30.26 | nny | Strom_C: ok still permission denied |
| 22:30.46 | nny | <PROTECTED> |
| 22:30.46 | nny | here = 0, tmp->channel = 1, channel = 1 |
| 22:30.57 | Strom_C | nny: ok, give me ssh access |
| 22:31.12 | nny | crw-rw---- 1 root dialout 196, 1 2007-10-10 18:28 1 |
| 22:31.25 | nny | stand by, ddwrt firewall was giving me shit the other day |
| 22:31.46 | nny | asterisk : asterisk dialout |
| 22:32.08 | nny | for some reason I can't PM in pidign.. let me install xchat |
| 22:33.29 | *** join/#asterisk Kunnis (n=someone@cpe-70-112-252-73.austin.res.rr.com) |
| 22:34.36 | Kunnis | Hey, I'm just wondering rough numbers here, but how do most telco's bill for 1-800 numbers? |
| 22:34.59 | Kunnis | And waht's the monthly fees like |
| 22:35.33 | Kunnis | Or know any good sites that would explain it |
| 22:35.36 | nny | brb switching to xchat |
| 22:36.35 | Sci_05 | Kunnis: usually 800 numbers get billed to the Long Distance account |
| 22:36.47 | TrentCreek | it varies |
| 22:37.17 | GoRK | Kunnis: 800 numbers are a little bit complicated but normally they just have a small monthly fee to add the number to your regular LD account and then a per minute rate.. there are special circumstances where you would have PRI's or something for only the 800 number but if this is what you need you probably wouldnt be asking us :) |
| 22:37.23 | TrentCreek | Some do not and just charge unlimted fee, unless call originated from payphone |
| 22:37.34 | *** join/#asterisk nny (n=scott@64.203.239.83.static-pool-4.pool.hargray.net) |
| 22:37.45 | *** part/#asterisk popvoxdave (n=popvoxda@64.240.183.2) |
| 22:37.54 | GoRK | Kunnis: you will have different rates for different origination such as pay phones or canada.. or you can disallow origination from these or other areas also |
| 22:38.37 | *** join/#asterisk blq (n=Bl@dslb-088-064-146-061.pools.arcor-ip.net) |
| 22:39.28 | Guggemand | anyone having trouble using "sip notify snom-check-cfg" in 1.4.11? |
| 22:39.45 | wishes | ok i have a queues/AGI query, I have a queue(.....AGI script) the script should trigger on answer (according to the docs), what im trying to pass to the AGI is the extention or username of the person who answers it - any ideas? |
| 22:39.47 | Guggemand | after issuing the command once nothing else works until i reconnect to the console and run a reload |
| 22:42.21 | hmmhesays | hey guys have any of you used one of these d945GCNL boards with zaptel hardware? |
| 22:44.57 | *** join/#asterisk pots_line (n=bryan@66-43-34-50.misn.com) |
| 22:46.02 | hmmhesays | I'm also curious if it matters what voltage pci slot you plug an a200 card into |
| 22:46.08 | [TK]D-Fender | wishes, it doesn't take parameters. All it can do is access channel variables. |
| 22:46.17 | [TK]D-Fender | hmmhesays, 3.3v / 5v |
| 22:46.35 | *** join/#asterisk jetlagmk2 (i=jetlag@pool-70-17-59-108.pskn.east.verizon.net) |
| 22:46.38 | [TK]D-Fender | hmmhesays, All Sangoma PCI cards are 3.3/5 copatible |
| 22:46.47 | Kunnis | GoRK That's bascailly what I was wondering. I worked on asterisk a long time ago, and figured this would be a good place to ask. |
| 22:46.56 | *** join/#asterisk nny (n=scott@64.203.239.83.static-pool-4.pool.hargray.net) |
| 22:47.03 | Kunnis | I actually got sphix+asterisk == voice recogition working. |
| 22:47.28 | wishes | [TK]D-Fender: yeah you can call it like .. |
| 22:47.29 | wishes | exten => 2,n,Queue(support-queue|trn|||15|AGI(/var/lib/asterisk/agi-bin/csr_call_logger.php|sip_id=${EXTEN}|callerid=${CALLERIDNUM})) |
| 22:47.51 | wishes | it does work, i just cant find the correct sip_id var to give it the user extention that picked up |
| 22:48.05 | wishes | Kunnis: nice |
| 22:48.09 | wishes | was it hard? |
| 22:48.14 | Kunnis | But I hit a wall on getting recogition decent. |
| 22:48.28 | wishes | but it does basic 'yes' and 'no' etc ? |
| 22:48.35 | Kunnis | I spent 2 weeks paid work on it. look up the jasterisk project on sourceforge.net |
| 22:48.42 | Kunnis | so with my code it's easy... but it's a fork of asterisk |
| 22:48.55 | hmmhesays | i'm wondering if I can run a 4 fxo port card on a mini pci board |
| 22:49.14 | Kunnis | Someone started Jasterisk, and I just cleaned it up. |
| 22:49.34 | Kunnis | I'd rate it as hard because of the java<->C bridge |
| 22:49.44 | JT | GoRK: right, normal voice calls over ISDN. |
| 22:49.52 | JT | what else would people be talking about? |
| 22:49.57 | JT | calls to the telco |
| 22:50.10 | JT | GoRK: it's G.711, not necessarily Mu-Law |
| 22:50.36 | *** part/#asterisk pots_line (n=bryan@66-43-34-50.misn.com) |
| 22:50.44 | [TK]D-Fender | wishes, you should really try reading up a bit more... "New in Asterisk 1.4: The MEMBERINTERFACE channel variable holds information about which queue member received the call. " |
| 22:51.36 | Kunnis | But bascially it's now just download and build jasterisk, and download and build sphinx, and it magically works. |
| 22:51.49 | *** join/#asterisk nny (n=scott@64.203.239.83.static-pool-4.pool.hargray.net) |
| 22:51.58 | wishes | [TK]D-Fender: using 1.2 still |
| 22:52.13 | Kunnis | I started pulling jasterisk back into a module, but I got pulled off the project, so I didn't do any more development. |
| 22:52.15 | wishes | upgrading would requrie working for a weekend whilst nobody is using it |
| 22:52.46 | wishes | though ive serious been tempted - but atm im moving house because i burnt the last one half down so im kinda busy on the weekend cleaning up the old place :) |
| 22:53.12 | NovceGuru | Weird, absolutely no config changes and I'm getting WARNING[47194]: app_dial.c:1106 dial_exec_full: Unable to create channel of type 'SIP' (cause 3 - No route to destination) |
| 22:53.30 | NovceGuru | out of the blue |
| 22:53.49 | [TK]D-Fender | wishes, Queue(queuename[|options[|URL][|announceoverride][|timeout]]): |
| 22:53.49 | [TK]D-Fender | <PROTECTED> |
| 22:53.55 | JT | check that the network connection is working properly, NovceGuru |
| 22:54.09 | wishes | [TK]D-Fender: nasty nasty nasty hacked/patched 1.2 |
| 22:54.16 | wishes | 1.2.18 |
| 22:54.44 | [TK]D-Fender | wishes, Sorry.... your warranty is completely DEAD with us now :) |
| 22:54.45 | wishes | hence why i want to upgrade it fully and get rid of all the crap out of here :/ |
| 22:54.49 | wishes | hehe |
| 22:55.16 | wishes | im dealing with the previous legacy network managers - one who had a clue but left a couple eyars ago, the other who just like to fuck shit up and left me to deal with it |
| 22:55.16 | [TK]D-Fender | wishes, Hey, go hack it in yourself then while you're at it.... |
| 22:55.19 | NovceGuru | JT: seems fine, *checks again* |
| 22:55.27 | hmmhesays | [TK]D-Fender: I ordered the h261 with m22 |
| 22:55.28 | NovceGuru | try hacking 127.0.0.1 |
| 22:55.32 | NovceGuru | =) |
| 22:55.36 | PSU_Boss | do you have to actually forward the ports 10000-20000 on the router to the box that has asterisk on it? or just open the ports on the firewall on the asterisk box? |
| 22:55.56 | wishes | naw, i want to reinstall from scratch etc - but i know that the configs will change a fair bit :D |
| 22:55.59 | [TK]D-Fender | hmmhesays, They'll be happy with it. Binaural really helps you concentrate on your caller. |
| 22:56.05 | hmmhesays | good |
| 22:56.10 | PSU_Boss | because i did a tcpdump on the asterisk box and port 10010 is being accessed, and there are no hits to the port forward on the router |
| 22:56.20 | wishes | i never knew i looked so angry all the time until i got a webcamera up |
| 22:56.23 | *** join/#asterisk jm|laptop (n=jamie@dilbert.jamiem.com) |
| 22:56.30 | hmmhesays | i'm hoping that I can use a sangoma card with this mini atx board |
| 22:56.40 | jm|laptop | anyone got a Cisco 7905/7912 ? |
| 22:57.18 | jm|laptop | with SIP firmware |
| 22:59.20 | hmmhesays | oops its a micro atx |
| 22:59.39 | hmmhesays | anyone used any sangoma hardware with a micro atx board? |
| 22:59.52 | [TK]D-Fender | hmmhesays, My server does |
| 22:59.58 | [TK]D-Fender | hmmhesays, works fine |
| 23:00.03 | hmmhesays | [TK]D-Fender: what board? |
| 23:00.29 | [TK]D-Fender | hmmhesays, A7V8X-MX |
| 23:01.27 | hmmhesays | I have an intel d945gcnl |
| 23:01.28 | [TK]D-Fender | jm|laptop, What about them? |
| 23:01.36 | jm|laptop | meh |
| 23:01.53 | jm|laptop | I was wondering if new firmware allows dial alphabetical sip uri stuff |
| 23:01.57 | jm|laptop | I no longer have CCO |
| 23:02.30 | [TK]D-Fender | jm|laptop, no clue. Cisco.... bleh |
| 23:02.35 | jm|laptop | :/ |
| 23:02.47 | jm|laptop | Cisco schmisco |
| 23:02.50 | jm|laptop | Cisco are good :/ |
| 23:03.06 | [TK]D-Fender | more like "could do worse" |
| 23:03.14 | jm|laptop | (: |
| 23:03.23 | *** join/#asterisk anonymouz666 (n=anonymou@201.19.159.138) |
| 23:05.23 | *** join/#asterisk ectospasm (n=ectospas@c-68-62-214-33.hsd1.al.comcast.net) |
| 23:06.31 | hmmhesays | if I only have 2 ip 501's I'd rather just get an ac adapter for them |
| 23:06.35 | hmmhesays | but I can't seem to find any |
| 23:06.48 | JT | cisco are shite |
| 23:06.55 | jm|laptop | thanks JT |
| 23:07.55 | *** join/#asterisk |dennis| (n=dennis@200.32.236.18) |
| 23:09.21 | [TK]D-Fender | hmmhesays, they came with the PoE-only cable? |
| 23:09.56 | hmmhesays | [TK]D-Fender: i'm trying to figure out what the 501's come with |
| 23:09.59 | hmmhesays | for power |
| 23:10.06 | hmmhesays | I have to order some tomorrow |
| 23:10.29 | jm|laptop | my 7912 came with PoE only |
| 23:10.29 | [TK]D-Fender | hmmhesays, I'm not talking about "the" IP 501's, I'm talking about "YOUR" IP 501's. |
| 23:10.42 | hmmhesays | I don't have any |
| 23:10.49 | *** join/#asterisk zeedo (n=zeedo@reboot-robot.net) |
| 23:11.14 | hmmhesays | why do you think i'm always asking questions about them |
| 23:11.25 | hmmhesays | I finally get to order some tomorrow |
| 23:11.57 | [TK]D-Fender | hmmhesays, They come with either a ) a cable with a PoE "nugget" in-line circuit , or b ) a special cable that you plug the wall wart (which comes with the cable) into the MIDDLE of. |
| 23:12.22 | hmmhesays | it doesn't say on voipsupply |
| 23:12.35 | [TK]D-Fender | hmmhesays, odds are if it doesn't say "PoE cable bundle" then you are going to get "B" |
| 23:12.52 | [TK]D-Fender | hmmhesays, link it. Oh, and Voipsupply's prices suck on Polycom. |
| 23:12.52 | *** join/#asterisk riddlebox (n=james@75-132-225-75.dhcp.stls.mo.charter.com) |
| 23:13.24 | hmmhesays | [TK]D-Fender where do you recommend? |
| 23:13.33 | [TK]D-Fender | hmmhesays, www.telephonydepot.com |
| 23:13.43 | hmmhesays | they US? |
| 23:14.49 | [TK]D-Fender | hmmhesays, yup |
| 23:15.34 | *** join/#asterisk DaveCanoe (n=Dave@canoe404.dclg.ca) |
| 23:15.50 | [TK]D-Fender | hmmhesays, fast shipping to Canada and they were nice to deal with on the phone (I don't order direct from sites if at all possible) |
| 23:16.29 | blitzrage | I hate ordering from the US... custom duties always make it cheaper to order from Canada |
| 23:16.31 | hmmhesays | gotcha, so what are my power possibilities on here, either power bundle, is there a regular ac adapter? |
| 23:16.46 | blitzrage | and with the Canadian dollar stronger than the USD, it makes sense even less so :) |
| 23:17.16 | rpm | go canada! |
| 23:17.18 | [TK]D-Fender | hmmhesays, "B" 's cable comes with a brick you plug INTO the cable mid-way through |
| 23:17.29 | [TK]D-Fender | hmmhesays, kind of like an IV drip at the hospital |
| 23:17.40 | jm|laptop | A vodka one? |
| 23:18.05 | hmmhesays | it doesn't say on telephony depot |
| 23:18.29 | [TK]D-Fender | blitzrage, tip : Canadian prices aren't synching to match the USD exchange rate. Therefor US goods are becoming CHEAPER and MORE worthwhile to import. Check your math :p |
| 23:18.40 | _x86_ | i never thought i would see the day when a republican ran the US dollar into the dirt so bad |
| 23:19.03 | jm|laptop | "I love you long time for top Euro!" |
| 23:19.09 | BBHoss | lol |
| 23:19.15 | hmmhesays | just not as funny |
| 23:19.18 | [TK]D-Fender | hmmhesays, look on page 2 of "polycom phones". You'll see 2 x IP 501's 1 with PoE bundle, 1 without |
| 23:19.21 | jm|laptop | "just one Eulo!" |
| 23:19.45 | BBHoss | yeah i love 'conservatism' :) more like 'spendatism' |
| 23:19.53 | [TK]D-Fender | _x86_, He isn't a republican, he's a FASCIST <------ |
| 23:19.56 | hmmhesays | so the one with the poe bundle has everything I need to power it |
| 23:20.55 | [TK]D-Fender | hmmhesays, No. it doesn't come with the "Power+ethernet" cable, it comes with the "cable with PoE support nugget" ONLY |
| 23:21.08 | JT | the australian dollar is expected to equal the US dollar some time next year :D |
| 23:21.13 | [TK]D-Fender | hmmhesays, it SUBSTITUTES the cable, not ADDING. |
| 23:21.26 | NovceGuru | JT: that extension can dial me, but I can't dial them, he calls me, can't hear either way, but we both can call a conference |
| 23:22.02 | [TK]D-Fender | JT : From what little news of AU passes my eyes its looking like your gov't is becoming much more "1984"-like all the time. Is that a fair assessment? |
| 23:22.15 | *** join/#asterisk Corydon76-vcch (n=tilghman@pdpc/supporter/bronze/Corydon76-home) |
| 23:22.15 | *** mode/#asterisk [+o Corydon76-vcch] by ChanServ |
| 23:22.21 | NovceGuru | also, that extension doesn't show up in sip show peers, |
| 23:22.28 | [TK]D-Fender | NovceGuru, "canreinvite=no" <-------------- |
| 23:22.28 | NovceGuru | erm, its ping doesn't show up |
| 23:22.31 | blitzrage | [TK]D-Fender: actually... ya... I guess that's true since that's the reason I bought my MacBook in the US while I was there |
| 23:22.35 | [TK]D-Fender | NovceGuru, classic NAT issue |
| 23:22.49 | NovceGuru | fender, they are both set to no :| |
| 23:22.52 | JT | [TK]D-Fender: not really, nothing like the US |
| 23:22.58 | [TK]D-Fender | NovceGuru, here : |
| 23:23.00 | [TK]D-Fender | ~sipnat |
| 23:23.00 | 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 |
| 23:23.07 | hmmhesays | [TK]D-Fender: so I have to get a poe hub or injector to power it right? |
| 23:23.18 | [TK]D-Fender | JT : Never said it had to be as bad, just that its a continuing trend. |
| 23:23.37 | [TK]D-Fender | hmmhesays, I've tried to be painfully clear on this... you jsut don't seem to be getting it :/ |
| 23:23.52 | *** join/#asterisk Teln1100A (i=hello123@bas2-toronto12-1167860795.dsl.bell.ca) |
| 23:23.56 | *** join/#asterisk luke-jr (n=luke-jr@2002:440d:6de2:0:20e:a6ff:fec4:4e5d) |
| 23:24.08 | luke-jr | Any ideas on why faxing is suddenly not working? :/ |
| 23:24.18 | luke-jr | does RTP packet size and such matter? |
| 23:24.36 | NovceGuru | [TK]D-Fender: both behind a basic nat and the server isn't natted |
| 23:24.37 | twisted | hmmhesays: quick tip: the polycom 501's w/PoE already have the poe injecters |
| 23:24.44 | luke-jr | fax machine -> PAP2 -> LAN -> Asterisk |
| 23:24.45 | twisted | hmmhesays the ones that don't say w/PoE do NOT |
| 23:25.21 | *** join/#asterisk Strom_M (n=strom@208.127.172.112) |
| 23:25.44 | JT | [TK]D-Fender: nothing really that worrying yet |
| 23:25.48 | twisted | also, you could look at pricepoint. generally, unless there's a special, the ones that cost more have everything you need |
| 23:25.52 | *** join/#asterisk anonymouz666 (n=anonymou@201.19.148.36) |
| 23:26.10 | rpm | can asterisk 1.4 do t.38 passthrough? |
| 23:28.36 | BBHoss | yes i think so |
| 23:29.40 | BBHoss | http://www.voip-info.org/wiki/view/Asterisk+T.38 |
| 23:31.17 | blitzrage | rpm: yes |
| 23:31.31 | BBHoss | its not perfect though |
| 23:33.46 | NovceGuru | if I sip client can connect, and register, then it should show a ping with qualify=yes? |
| 23:34.02 | *** join/#asterisk saftsack (n=saftsack@pD9E067C3.dip.t-dialin.net) |
| 23:34.05 | BBHoss | not always |
| 23:34.37 | blitzrage | twisted: omg you're alive |
| 23:34.42 | riddlebox | wohoo charter is going to replace my modem tomorow, because it isnt doing disconnect supervision, now I wont get text messages all the time when there is a hangup |
| 23:35.12 | *** join/#asterisk Qapf (n=Qapf@stevenson-17-105.resnet.ucsc.edu) |
| 23:36.48 | twisted | blitzrage, yeah, i live. |
| 23:37.12 | Qapf | Hey, I have a branch server connected to my main one, and on my main server i have an ivr where people can direct dial an extension number and get there. extensions on the branch office are not on the extensions list of the main server and as a result the ivr says invalid extension, is there any way to make the ivr aware of the branch office so those extensions can be dialed directly? |
| 23:38.02 | *** join/#asterisk ectospasm (n=ectospas@c-68-62-214-33.hsd1.al.comcast.net) |
| 23:38.28 | BBHoss | hang on qapf ill get you an example |
| 23:39.25 | BBHoss | its as simple as adding a line to extensions.conf |
| 23:39.25 | *** join/#asterisk drwelby (n=mpfister@mail.enplan.com) |
| 23:40.56 | riddlebox | what is a different name for disconnect supervision? |
| 23:41.17 | *** join/#asterisk Itiliti (n=Itiliti@c-76-29-86-174.hsd1.il.comcast.net) |
| 23:41.52 | Qapf | BBHoss, thanks |
| 23:42.03 | Itiliti | How can I compile res_bonjour into asterisk? |
| 23:43.27 | BBHoss | exten => _5XXX.,1,Dial(IAX2/yourpeer1/${EXTEN}) |
| 23:43.57 | BBHoss | replace 5XXX with whatever pattern you use to call down there |
| 23:44.11 | BBHoss | just put that line in the context that handles normal calls |
| 23:48.38 | NovceGuru | so with ekiga it's working better, the peer shows a ping, and I can call and we can hear each other (stupid echo or loop inside the soundcard) but its tellin him invalid pin |
| 23:48.41 | NovceGuru | in a conference |
| 23:49.14 | [TK]D-Fender | NovceGuru, pastebin it. |
| 23:49.38 | NovceGuru | let me get a fresh attempt |
| 23:50.10 | Qapf | and i assume replace yourpeer1 with the real name of the iax2 trunk |
| 23:50.15 | BBHoss | yes |
| 23:50.27 | Qapf | ok, ill give it a go |
| 23:50.36 | Qapf | thanks |
| 23:50.39 | BBHoss | anything that has a 5 and 3 following 0-9 numbers will go through that trunk |
| 23:50.42 | BBHoss | sure |
| 23:51.54 | NovceGuru | fender, asterisk -vvvvvc should be enough output? |
| 23:52.32 | [TK]D-Fender | NovceGuru, to start |
| 23:55.27 | [TK]D-Fender | BBHoss, ... ALMOST ;) |
| 23:55.42 | NovceGuru | blah, if I login with his account it works fine, must be something in ekiga |
| 23:55.51 | BBHoss | ? |
| 23:56.07 | [TK]D-Fender | BBHoss> anything that has a 5 and 3 following 0-9 numbers will go through that trunk <--- ALMOST right :) |
| 23:56.25 | BBHoss | what did i miss |
| 23:56.50 | [TK]D-Fender | BBHoss> exten => _5XXX.,1,Dial(IAX2/yourpeer1/${EXTEN}) <----- read this carefully, and you tell ME what you missed :) |
| 23:57.06 | BBHoss | shit |
| 23:57.12 | BBHoss | the dot on the end |
| 23:57.18 | fujin | ;p |
| 23:57.22 | [TK]D-Fender | :p |
| 23:57.29 | fujin | DOT ON THE END INDEED |
| 23:57.36 | BBHoss | that will fuck things up :) |
| 23:57.43 | [TK]D-Fender | only a littl! |
| 23:57.45 | [TK]D-Fender | little* |
| 23:58.12 | rpm | which option do i need to turn on in zapata.conf so asterisk will accept digits to be dialed? i can call my analog phone connected to my wctdm fxs module, but can't dial i immediately get a congestion tone. |
| 23:59.00 | VoipMasta | rpm: set a context that holds a valid extension for the digits you're trying to dial |
| 23:59.01 | [TK]D-Fender | rpm, usually this is doing something silly like pointing it to a context that doesn't exist or have anything that's sane to match against <-- |
| 23:59.41 | rpm | it doesn't wait for a timeout period before matching against a number/extension? it starts matching immediately? |
