| 00:00.07 | *** join/#asterisk CrazyTux[m] (n=CrazyTux@cpe-76-175-64-30.socal.res.rr.com) |
| 00:09.32 | *** join/#asterisk mightnare (n=mike@s230165.ppp.asahi-net.or.jp) |
| 00:11.45 | mightnare | i'm getting "NOTICE-xpp: XBUS-02: A PCM packet within a Non-PCM xframe" with zaptel 1.4.4 |
| 00:18.24 | *** join/#asterisk nuonguy (n=john@c-24-6-175-26.hsd1.ca.comcast.net) |
| 00:25.54 | *** join/#asterisk nuonguy (n=john@c-24-6-175-26.hsd1.ca.comcast.net) |
| 00:31.08 | *** join/#asterisk nuonguy (n=john@c-24-6-175-26.hsd1.ca.comcast.net) |
| 00:39.12 | *** join/#asterisk nuonguy (n=john@c-24-6-175-26.hsd1.ca.comcast.net) |
| 00:45.20 | *** join/#asterisk nuonguy (n=john@c-24-6-175-26.hsd1.ca.comcast.net) |
| 00:51.08 | *** join/#asterisk Huwawa (n=Huwawa@adsl-69-104-2-215.dsl.pltn13.pacbell.net) |
| 01:01.19 | *** join/#asterisk pkempgen (n=pkempgen@ACB053E8.ipt.aol.com) |
| 01:01.33 | *** join/#asterisk bkruse_home (n=kruz@69.73.127.92) |
| 01:02.22 | *** join/#asterisk CVirus (n=GoD@82.201.174.99) |
| 01:02.32 | CVirus | [Jul 21 04:00:01] NOTICE[12792]: chan_iax2.c:5636 update_registry: Restricting registration for peer 'test1' to 60 seconds (requested 300) |
| 01:02.45 | CVirus | why is asterisk restricting the registration to 60 seconds ? |
| 01:02.49 | *** part/#asterisk pkempgen (n=pkempgen@ACB053E8.ipt.aol.com) |
| 01:05.39 | carrar | what did you set it too? |
| 01:06.48 | carrar | CV |
| 01:06.53 | carrar | let me ask this |
| 01:06.58 | carrar | did you google this? |
| 01:07.00 | carrar | heh |
| 01:07.40 | carrar | or have you read the ORIGINAL sip.conf |
| 01:07.53 | carrar | or ix.conf |
| 01:07.55 | carrar | iax |
| 01:08.18 | Innatech | <PROTECTED> |
| 01:08.27 | CVirus | carrar: sorry |
| 01:08.36 | carrar | Sometimes I find reading the config fire 1st gives me the answer |
| 01:08.39 | carrar | file |
| 01:08.51 | CVirus | carrar: axregexpire |
| 01:08.59 | CVirus | maxregexpire* |
| 01:09.01 | carrar | also try www.voip-info.org |
| 01:09.12 | carrar | You will find answers there to a lot of your questions |
| 01:10.29 | carrar | Innatech, run ntpd on your asterisk box and point all your phones to that |
| 01:10.57 | *** join/#asterisk mitcheloc (n=mitchelo@adsl-68-120-226-30.dsl.irvnca.pacbell.net) |
| 01:11.36 | Innatech | Yes, that's the plan eventually. However, I'd feel much better if I could figure out why one of each of the models can't get out to pool.ntp.org..... |
| 01:11.52 | carrar | quite possible they are restricting you |
| 01:12.03 | carrar | so as to avoide excessive abuse like that |
| 01:12.26 | *** join/#asterisk [hC] (n=hardcore@S01060016b61c8983.vf.shawcable.net) |
| 01:13.11 | Innatech | oooh. Hadn't thought of that. Is there a published policy on acceptable connection frequency? |
| 01:13.20 | carrar | once you rule out the part you have no control of you can narrow things down |
| 01:14.03 | *** join/#asterisk [hC] (n=hardcore@S01060016b61c8983.vf.shawcable.net) |
| 01:14.16 | *** join/#asterisk sudoer (n=jtoy@c-24-60-25-28.hsd1.ma.comcast.net) |
| 01:14.18 | carrar | can't say for sure as pool is arbitary group of addresses |
| 01:14.38 | Innatech | yeah, I need to get it running on the * box. It just seems odd---I have four Polys. I power up all of em at the same time--one of each gets NTP, the other two doesnt. Strange. I'd like to chase down whatever gremlin is responsible if only for my own peace of mind. |
| 01:15.01 | carrar | chase it down with known values |
| 01:15.16 | [hC] | all identical config and all of them work? |
| 01:15.22 | [hC] | just 3 dont get ntp? |
| 01:15.32 | Innatech | all 4 work, 2 don't get ntp. |
| 01:15.41 | Innatech | one of the 501s and one of the 430s. |
| 01:15.56 | [hC] | tried a different ntp server just to see if that works? |
| 01:16.09 | Innatech | I'm using pool.ntp.org....round robin DNS, I think. |
| 01:16.25 | [hC] | ah yes. maybe try forcing to one, incase their round robin is hosed? |
| 01:16.28 | [hC] | tick.mit.edu |
| 01:16.29 | [TK]D-Fender | Innatech, verify your time zone settings and local overrides on each. |
| 01:16.48 | [hC] | that too. |
| 01:16.51 | Innatech | will do. |
| 01:16.56 | Innatech | afk for a minute. |
| 01:18.05 | [TK]D-Fender | Innatech, MAIN>STATUS>NETWORK>TCP/IP |
| 01:20.55 | Mad|Cow | Does anyone know how to make a Cisco 7960 phone register itself on multiple proxy's? |
| 01:22.51 | Nugget | I haven't run that way in a while, but I do recall one bug where the cisco won't figure out what you want if you're trying to use the same username on the different proxies. |
| 01:23.13 | Nugget | and it absolutely won't work if one proxy is local and the other is on the far side of nat. |
| 01:23.22 | Nugget | since the nat yes/no setting is global for the phone |
| 01:23.38 | Mad|Cow | Nugget: I actually can dial out... but when I try and call the extension, it only works with the extension of the last registered proxy |
| 01:23.47 | Nugget | weird |
| 01:23.53 | Mad|Cow | tell me about it |
| 01:23.55 | Mad|Cow | ;-) |
| 01:26.26 | *** join/#asterisk bkruse_home (n=kruz@69.73.127.92) |
| 01:28.17 | *** join/#asterisk bkruse_home (n=kruz@69.73.127.92) |
| 01:33.12 | Hmmhesays | so [TK]D-Fender: the custom greeting thing was super simple in the dialplan |
| 01:33.26 | [TK]D-Fender | Hmmhesays, naturally |
| 01:33.44 | Hmmhesays | the only thing I wish voicemailmain did i'm going to have to patch it |
| 01:33.56 | [TK]D-Fender | Hmmhesays, namely? |
| 01:34.10 | Hmmhesays | is read of the name of the person calling before playing the voicemail back if a file exists with their name |
| 01:34.11 | [TK]D-Fender | Hmmhesays, THIS doesn't qualify as "worth it" |
| 01:34.47 | [TK]D-Fender | Hmmhesays, it deserves to be delegated to the dialplan. How many people really care? |
| 01:34.50 | mightnare | just upgraded to zaptel 1.4.4 and I'm getting a lot of these "NOTICE-xpp: XBUS-02: A PCM packet within a Non-PCM xframe"... anyone? |
| 01:34.59 | Hmmhesays | [TK]D-Fender: I do and that matters the most |
| 01:35.03 | Hmmhesays | ;) |
| 01:36.05 | *** join/#asterisk phessler_ (n=phessler@gir.theapt.org) |
| 01:36.57 | Hmmhesays | it should be easy enough to change that |
| 01:38.49 | *** join/#asterisk AtomicDawg (n=atomicda@74-206-0-81.static-ip.m.telepacific.net) |
| 01:43.06 | *** join/#asterisk fakhir (n=Fakhir@unaffiliated/fakhir) |
| 01:46.01 | *** join/#asterisk zydrunas_ (n=zydrunas@24-119-29-130.cpe.cableone.net) |
| 01:46.09 | *** join/#asterisk [TK]D-Fender (n=Joe@64.235.216.2) |
| 01:51.20 | Innatech | [TK]D-Fender , [hC] , carrar: no luck. Offset verified on each, SNTP setting verified on each. Tried using tick.mit.edu insteal of pool.ntp.org as well as using the numerical IP for tick.mit.edu . Same behavior: phone starts up and works normally, but sets itself to 4:00 Dec 31. |
| 01:52.00 | *** join/#asterisk tako-san (n=Tako-san@24.68.129.29) |
| 01:52.18 | el_critter | are digium cards pci-express compatible? |
| 01:52.20 | Innatech | [TK]D-Fender , [hC] , carrar: behavior is the same regardless of whether SNTP settings are specified in the provisioning files or entered via the web interface. |
| 01:52.39 | carrar | -carrar |
| 01:53.15 | phessler_ | Innatech: can the phone connect to pool.ntp.org, or is it not leaving your network? |
| 01:53.27 | carrar | And you read the ntp section of the polycom admin manual? |
| 01:53.54 | Innatech | carrar: yep. Although I should probably go read it again since I'm losing my mind. |
| 01:54.09 | carrar | Are you giving out ntp-server ip's in dhcp? |
| 01:54.11 | Innatech | phessler_: I have two Poly 501s and two 430s. One of each is connecting. |
| 01:54.25 | Innatech | carrar: no, that phones have static IPs. |
| 01:54.40 | Innatech | *the |
| 01:55.07 | *** join/#asterisk `Sean (i=Un1x@CPE000c256d416d-CM0012c9213a06.cpe.net.cable.rogers.com) |
| 01:55.10 | phessler_ | the phones that work, static ip or dhcp? ditto with phones that don't work. |
| 01:55.11 | [TK]D-Fender | verify your GMT offset in each |
| 01:55.20 | Innatech | all are static, all are at -8 GMT. |
| 01:57.06 | carrar | GONNA make me look at my configs are ya! |
| 01:57.09 | [TK]D-Fender | Innatech, Same firmware versions? |
| 01:57.09 | tako-san | aaagh. i have just added another fxo card to my channel bank made the necessary changes to zaptel and zapata and now i keep getting the message "all circuits are busy now". i can see the lights on the fxo card and asterisk is not even getting to the point were it seizes the line. suggestions? |
| 01:57.42 | [TK]D-Fender | tako-san, SHPASTEBIN is your friend. |
| 01:57.43 | [TK]D-Fender | ~pb |
| 01:57.44 | jbot | A Pastebin is a place to paste your stuff without flooding the channel. Here are links to a few : http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org |
| 01:57.45 | [TK]D-Fender | ^^^^^^^^^^^^^^^^^^^^^^^^ |
| 01:57.54 | carrar | tcpIpApp.sntp.gmtOffset? |
| 01:58.09 | tako-san | [TK]D-Fender: I know what it is. What do you want to see? Zaptel and zapata? anything else? |
| 01:58.18 | CVirus | My iax call hangs up after a couple of seconds of initialization ... Here's is the output http://rafb.net/p/xXTMlT14.html |
| 01:58.44 | [TK]D-Fender | tako-san, both + CLI output at verbose 10 of your failed call and a channel dump |
| 01:58.53 | Innatech | carrar: yep, tcpIpApp.sntp.gmtOffset is set to -8 in sip.conf |
| 01:58.54 | tako-san | okey dokey |
| 01:58.56 | tako-san | brb |
| 01:59.00 | carrar | -8 seconds? |
| 01:59.21 | tako-san | [TK]D-Fender: can i include it all in one pastebin or do you prefer seperate ones for each section? |
| 01:59.26 | Innatech | [TK]D-Fender yep, same firmware. I've only ever downloaded it once. These phones are a couple weeks old. |
| 01:59.29 | carrar | in sipMAC.cfg |
| 01:59.35 | [TK]D-Fender | tako-san, All together witha bit of space |
| 01:59.49 | tako-san | [TK]D-Fender: Roger |
| 01:59.57 | Innatech | MAC.cfg files include a reference to sip.cfg |
| 01:59.59 | [TK]D-Fender | Innatech, indeed, get your UOM correct |
| 02:00.20 | [TK]D-Fender | Innatech, check your phone-local as well. |
| 02:00.25 | Innatech | UOM? |
| 02:01.12 | Innatech | [TK]D-Fender by phone-local do you mean the web interface? I'm a little slow on the polycom jargon. |
| 02:01.13 | CVirus | Anybody ? |
| 02:01.43 | [TK]D-Fender | Innatech, web / PHYSICAL on the phone + provisioning. |
| 02:02.10 | [TK]D-Fender | CVirus, Looks like some sort of transfer was started and you didn't provide an IAX2 debug info. |
| 02:02.13 | carrar | CV, you allowing GSM? |
| 02:02.27 | CVirus | carrar: yes |
| 02:02.43 | carrar | allow ulaw? |
| 02:02.46 | CVirus | carrar: allow=all |
| 02:02.47 | Innatech | [TK]D-Fender: OK. Should they be redundant or left blank? |
| 02:03.12 | carrar | maybe the path |
| 02:03.14 | *** join/#asterisk Avero (n=louisr@24.96.142.67) |
| 02:04.06 | [TK]D-Fender | Innatech, frankly you should never EVERY configure anything but your provisioning server on the phone itself, and NEVER use the web interface EVER. |
| 02:04.17 | [TK]D-Fender | Innatech, redo your configs and FLUCH the phones. |
| 02:04.45 | Innatech | Very well. |
| 02:05.03 | CVirus | carrar: so ? |
| 02:05.04 | [TK]D-Fender | FLUSH* |
| 02:05.17 | carrar | I like FLUCH |
| 02:05.48 | carrar | CV, [TK]D-Fender> CVirus, Looks like some sort of transfer was started and you didn't provide an IAX2 debug info. |
| 02:05.49 | *** join/#asterisk rmayorga (n=rmayorga@unaffiliated/rmayorga) |
| 02:06.10 | CVirus | carrar: how can I provide IAX2 debug info ? |
| 02:06.30 | [TK]D-Fender | *CLI> iax2 debug on |
| 02:06.52 | tako-san | [TK]D-Fender: http://pastebin.com/d25c592bd |
| 02:07.09 | CVirus | No such command 'iax2 debug on' (type 'help' for help) |
| 02:07.24 | CVirus | iax2 set debug |
| 02:07.26 | carrar | did you type help? |
| 02:07.34 | CVirus | done |
| 02:07.45 | carrar | (we don't need play by play) |
| 02:07.47 | carrar | heh |
| 02:09.44 | [TK]D-Fender | tako-san, You did not provide a channel dump at the time of the call "show channels concise", "zap show channels", and your channel specs look WHACKED. you have *9* ports at the start of your CB (2-10), what happened to 1? Also 11 ports at the end? |
| 02:10.15 | tako-san | [TK]D-Fender: I have purposefully removed channel 1 as it seemed to be causing some strange issues. |
| 02:10.17 | [TK]D-Fender | tako-san, And you are not setting a timing source on your channel bank's span. |
| 02:10.20 | *** join/#asterisk JunK-Y (n=junky@modemcable094.9-83-70.mc.videotron.ca) |
| 02:10.35 | [TK]D-Fender | tako-san, I am losing faith in your hardware second by second... |
| 02:10.36 | tako-san | I have 2 8 port FXO cards but we only have 9 inbound FXO lines |
| 02:11.11 | [TK]D-Fender | bad cac on the 11 at the end, indeed 8. |
| 02:11.15 | [TK]D-Fender | calc* |
| 02:11.36 | [TK]D-Fender | tako-san, swap your modules and reduce your channels. |
| 02:12.05 | tako-san | [TK]D-Fender: you mean for testing purposes? reduce to 1 channel per card or? |
| 02:12.24 | CVirus | [TK]D-Fender: carrar: http://rafb.net/p/NzemM279.html |
| 02:12.27 | [TK]D-Fender | tako-san, yes for testing. |
| 02:12.31 | tako-san | [TK]D-Fender: ok |
| 02:14.00 | [TK]D-Fender | CVirus, doesn't tell me anything, sorry. |
| 02:14.33 | CVirus | [TK]D-Fender: it seems that I lost a portion ... let me paste it all |
| 02:14.48 | CVirus | [TK]D-Fender: no .. it's there ... thanks anyways |
| 02:14.51 | *** join/#asterisk MrMister2 (n=mrmister@89-180-44-18.net.novis.pt) |
| 02:15.56 | CVirus | carrar: any idea ? |
| 02:16.11 | carrar | read more |
| 02:16.31 | carrar | google more |
| 02:16.39 | CVirus | will do |
| 02:16.47 | carrar | take more breaks |
| 02:17.01 | carrar | (it's Friday) |
| 02:17.04 | carrar | have a beer |
| 02:17.07 | carrar | that sometimes helps |
| 02:17.34 | carrar | maybe start with something more simple |
| 02:19.01 | CVirus | what's more simple than using iax to make 2 softphones communicate !! |
| 02:19.21 | carrar | using 1 phone |
| 02:19.34 | carrar | or not using iax |
| 02:20.45 | CVirus | If no sound is transferred ... does asterisk hang up the call automatically ? |
| 02:22.26 | carrar | research that, let me know what you find out |
| 02:22.31 | CVirus | sure |
| 02:23.53 | *** part/#asterisk phessler_ (n=phessler@gir.theapt.org) |
| 02:23.54 | Innatech | [TK]D-Fender , [hC] , carrar: Did a local settings reset and then a full device clear. No change, except that the physical config menu no longer shows ANY sntp settings at all. And the 430 reset to 12:00 Jan 1 rather than 4:00 Dec 31. |
| 02:24.30 | tako-san | [TK]D-Fender: I edited zaptel and zapata so that only channels 2 & 9 are used and also changed zaptel so that the Sangoma card is now acting as the timing device (doh). Same problem. |
| 02:24.33 | [TK]D-Fender | Innatech, you need to double then triple check your configs. Something sis screwed three |
| 02:24.44 | Innatech | yeah, seems that way to me too. |
| 02:25.24 | carrar | just the 1st time |
| 02:25.42 | tako-san | [TK]D-Fender: I cannot give you the dump information that you want because the commands as given do not seem to be available. I am using Asterisk 1.4.4 if that makes a difference. |
| 02:25.44 | carrar | or untill you write a script to fill in data |
| 02:26.17 | Innatech | Well, the first two went swimmingly. I copied the config files, renamed and altered them where appropriate and for some reason the SNTP went to hell. Everything else is fine. |
| 02:26.21 | [TK]D-Fender | tako-san, "core show channels concise" the other should be fine |
| 02:26.33 | tako-san | ok |
| 02:27.04 | Innatech | I probably need to knock off and pick it up tomorrow. At this point I'm likely not seeing the forest for the trees. Or not seeing the trees for the forest. Or something. |
| 02:27.11 | carrar | Innatech, maybe they have the same IP? |
| 02:27.28 | carrar | or did you say they make calls fine? |
| 02:27.31 | Innatech | carrar: naw, I checked for all that kind of stuff. Good thought, though. |
| 02:27.35 | Innatech | Yeah, calling is fine. |
| 02:28.10 | Innatech | I probably need to slap tethereal (or whatever its called now) on the router and watch the traffic. |
| 02:28.17 | tako-san | [TK]D-Fender: asterisk*CLI> zap show channels |
| 02:28.17 | tako-san | No such command 'zap show' (type 'help' for help) |
| 02:28.48 | tako-san | [TK]D-Fender: asterisk*CLI> core show channels concise |
| 02:28.48 | tako-san | SIP/242-09ab22c0!macro-outisbusy!s!1!Ring!Playback!all-circuits-busy-now|noanswer!242!!3!0!(None) |
| 02:28.55 | [TK]D-Fender | tako-san, I'm wondering if zap is even loaded.... |
| 02:29.11 | [TK]D-Fender | tako-san, "module load chan_zap.so" |
| 02:29.35 | tako-san | asterisk*CLI> module load chan_zap.so |
| 02:29.35 | tako-san | <PROTECTED> |
| 02:29.35 | tako-san | <PROTECTED> |
| 02:29.35 | tako-san | <PROTECTED> |
| 02:29.36 | tako-san | ?! |
| 02:29.52 | [TK]D-Fender | tako-san, looks like it wasn't... |
| 02:29.56 | [TK]D-Fender | tako-san, try a call again |
| 02:29.58 | tako-san | now why would it not be loading... it was loading fine before I made the one change to zaptel and zapata |
| 02:31.01 | tako-san | [TK]D-Fender: Was working just fine before I decided to not use channel 1 and use 9 & 10 instead |
| 02:32.41 | [TK]D-Fender | Why only 2 channels? |
| 02:33.27 | [TK]D-Fender | tako-san, redo 1 whole module alone, all ports. Then stop *, restart zaptel, then restart * |
| 02:33.34 | tako-san | [TK]D-Fender: why only 2 on the second card? or why only 2 right now during testing? <- i thought you wanted me to do that for testing |
| 02:35.06 | *** join/#asterisk MrBIOS (n=aperez@adsl-76-227-18-165.dsl.pltn13.sbcglobal.net) |
| 02:35.47 | tako-san | [TK]D-Fender: fyi, when i put it back to FXO card 1 with channels 1-8 only it will be back to its orignal state |
| 02:37.16 | [TK]D-Fender | tako-san, Ok, well hopefull this gives you some insight.... |
| 02:37.29 | tako-san | [TK]D-Fender: Don't you FXO card 2? ;-) |
| 02:37.49 | *** join/#asterisk coppice (n=chatzill@93.195.17.210.dyn.pacific.net.hk) |
| 02:38.42 | tako-san | [TK]D-Fender: *But* if I swap the 2 cards I can use the other FXO card without a problem. Perhaps there is an issue with the card slot in the channel bank. I have an identical back up channel bank that i will swap as the next phase of testing. |
| 02:40.18 | Andy_G | good evening |
| 02:41.49 | [TK]D-Fender | tako-san, go clean the connections, reseat things, retest that first port, etc. |
| 02:42.03 | [TK]D-Fender | tako-san, sometimes equiment just feels NEEDY. |
| 02:42.25 | tako-san | [TK]D-Fender: Yep, I am just doing that now. I am going to pull the 2nd FXO card as well just to make sure there is no monkey business. |
| 02:44.02 | *** join/#asterisk JackEStorm (n=no@ip68-225-77-136.no.no.cox.net) |
| 02:50.53 | AtomicDawg | How come the state of a phone never changes? I type "core show hints" and my phones always show idle?!? |
| 02:51.56 | tako-san | [TK]D-Fender: i dont understand it. still having the same issue... |
| 02:52.48 | [TK]D-Fender | AtomicDawg, you need to use "type=peer", and set "call-limit=100" or some other helthy number foe each of your phones w/ * 1.4 |
| 02:54.24 | AtomicDawg | K...let me try it... |
| 02:59.09 | *** part/#asterisk CoolGuy21 (n=Tilt@cpe-76-81-1-73.socal.res.rr.com) |
| 03:00.39 | AtomicDawg | [TK]D-Fender: You rule... BLF is now working on my SNOM 320. (On the sidecar no less!) Thanx a billion! |
| 03:01.30 | tako-san | [TK]D-Fender: Ok, I found a mistake in zaptel.conf that i correct and it is now working again. |
| 03:01.49 | tako-san | [TK]D-Fender: I was referring the to FXS channels as FXO... |
| 03:01.58 | [TK]D-Fender | ... |
| 03:02.23 | tako-san | [TK]D-Fender: But that does not solve the issue when I was using both FXO cards |
| 03:02.33 | tako-san | [TK]D-Fender: the error happened after I removed the 2nd card |
| 03:03.31 | [TK]D-Fender | tako-san, take a giant step back and break down your tests into the smallest and most unique steps |
| 03:03.44 | [TK]D-Fender | tako-san, each individually |
| 03:03.54 | tako-san | [TK]D-Fender: Good thinking. I think I need to take a short 5 minutes break too to clean out the cobwebs. |
| 03:04.24 | *** join/#asterisk stkn__ (n=stkn@gentoo/developer/pdpc.active.stkn) |
| 03:07.19 | *** join/#asterisk thesndguru (n=Nathan@202.61.168.243) |
| 03:07.50 | *** join/#asterisk jm|laptop (n=jm|home@dilbert.jamiem.com) |
| 03:09.26 | thesndguru | exten => s,2,GotoIf($[DB(home/dnd)]=off)?50:2) |
| 03:09.35 | thesndguru | can anyone help what I'm doing wrong |
| 03:10.38 | thesndguru | i'm missing a bracket |
| 03:10.40 | thesndguru | hrmm |
| 03:11.27 | [TK]D-Fender | thesndguru, and that endless loop possibility is a little ugley too ;) |
| 03:11.54 | thesndguru | oh where |
| 03:12.07 | *** join/#asterisk lisandropm (n=lisandro@cpe-22-76.bvconline.com.ar) |
| 03:12.16 | thesndguru | oh two |
| 03:12.17 | thesndguru | dar |
| 03:12.26 | thesndguru | thankz i'll try that too |
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| 03:16.51 | [TK]D-Fender | thesndguru, exten => s,2,GotoIf($["${DB(home/dnd)}"="off")?50) |
| 03:17.23 | thesndguru | oh ok |
| 03:18.14 | thesndguru | oh the quotes are need |
| 03:18.17 | thesndguru | ed |
| 03:18.28 | thesndguru | werid I didn't see that in the book i read |
| 03:18.48 | thesndguru | and two $ |
| 03:19.12 | thesndguru | ?? where's the closed } |
| 03:19.14 | thesndguru | sorry |
| 03:19.15 | thesndguru | ] |
| 03:21.23 | [TK]D-Fender | thesndguru, exten => s,2,GotoIf($["${DB(home/dnd)}"="off"]?50) |
| 03:21.41 | thesndguru | yep ok |
| 03:22.09 | [TK]D-Fender | thesndguru, $[] = evaluate expression, ${} = evaulate variable or function |
| 03:22.11 | thesndguru | therefore my next line is |
| 03:22.13 | thesndguru | exten => s,3,GotoIf($["${DB(home/dnd)}"="study"]?55:51) |
| 03:22.15 | tako-san | [TK]D-Fender: Ok, so far I have been able to get Asterisk to recognize and dial out with ports 2-8 on FXO card 1 and with port 1 on FXO card 2. Going to tweak things a little more. Looks like things may be stabilizing a little now. Thanks for your help. |
| 03:22.17 | *** join/#asterisk |dennis| (n=dennis@200.32.236.20) |
| 03:22.32 | [TK]D-Fender | tako-san, np |
| 03:22.39 | thesndguru | cool that's what I was missing |
| 03:22.46 | thesndguru | thankz [TK]D-Fender |
| 03:25.05 | tako-san | [TK]D-Fender: One more thing if I may. Do you know of a way to simulate a bunch of incoming calls to test load performance? |
| 03:25.28 | [TK]D-Fender | tako-san, ask someone with a PRI :) |
| 03:25.32 | Innatech | [TK]D-Fender , [hC] , carrar: Thanks for all your help. Knocking off for tonight.... |
| 03:25.37 | Innatech | cyas all tomorrow. |
| 03:25.47 | tako-san | [TK]D-Fender: LOL |
| 03:25.53 | tako-san | [TK]D-Fender: Of course ;) |
| 03:26.12 | thesndguru | [TK]D-Fender, thankyou so much, tis working now, so I've just got to learn these {'s & ['s |
| 03:26.22 | JackEStorm | hey was happened to set verbose <0-255> and set debug? |
| 03:26.46 | [TK]D-Fender | thesndguru, yup |
| 03:27.26 | thesndguru | it's been playing with all these bracket for like a week |
| 03:27.34 | thesndguru | but asterisk is like wicked |
| 03:27.46 | thesndguru | hopefully bye bye telemarkets |
| 03:27.48 | thesndguru | hehehe |
| 03:28.00 | *** join/#asterisk MdeP (n=MdeP@181-94-22-190.adsl.tie.cl) |
| 03:28.20 | tako-san | [TK]D-Fender: Aargh, one more perhaps? |
| 03:28.28 | [TK]D-Fender | tako-san, ? |
| 03:28.56 | [TK]D-Fender | tako-san, www.goats........ nevermind ;) |
| 03:29.08 | tako-san | Doh |
| 03:29.23 | [TK]D-Fender | tako-san, http://bestpicever.com/pic-129-My-first-Goatse |
| 03:29.30 | [TK]D-Fender | (SFW!) |
| 03:29.46 | tako-san | [TK]D-Fender: I am having a random recurring issue where at times when I dial out a number the call somehow terminates and I get a dialtone instead...?? |
| 03:30.39 | [TK]D-Fender | tako-san, Add some delay at the start "ww8005551212" |
| 03:30.40 | tako-san | [TK]D-Fender: I thought it was specific to line 1 of the 1st FXO card (thus I was not going to use it) but that does not seem to be the case after all. |
| 03:31.11 | tako-san | [TK]D-Fender: Ok, i will give that a try. |
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| 03:43.00 | coppice | [TK]D-Fender: they could well be labelled "My First Code Review" |
| 03:43.27 | [TK]D-Fender | coppice, ? |
| 03:44.02 | coppice | the picture |
| 03:44.17 | [TK]D-Fender | coppice, which? |
| 03:44.57 | coppice | http://bestpicever.com/pic-129-My-first-Goatse "You never forget your first peek at the code for <insert something nasty>" |
| 03:47.33 | [TK]D-Fender | coppice, Ah. |
| 03:49.01 | coppice | [lossy compresses the waves, and responds goodnight in only 1/4 of the bits] |
| 03:49.17 | [TK]D-Fender | lol |
| 03:49.40 | [TK]D-Fender | New Polycom pricing is SICK.... |
| 03:50.00 | *** join/#asterisk fx0 (n=fx0@cypher.punk.net) |
| 03:50.47 | fx0 | Hello, does anyone remember the name of that company in the UK that has a webphone that integrates with asterisk via iax ? |
| 03:51.41 | coppice | [TK]D-Fender: what is sick about their pricing? |
| 03:52.08 | [TK]D-Fender | coppice, IP 320 @ 87.50$USD |
| 03:52.28 | coppice | and what is sick about that? its still pretty high |
| 03:52.46 | [TK]D-Fender | coppice, Wihle yes I understand that the physical materials cost isn't technically high, market value is REALLY good, esp for their quality |
| 03:53.00 | [TK]D-Fender | coppice, as a matter of perspective. |
| 03:53.06 | *** join/#asterisk bmg505 (n=leon@196.209.183.182) |
| 03:53.15 | coppice | they'll never be mass market while they are above $50 |
| 03:53.27 | coppice | you can buy cellphones for less than that |
| 03:53.31 | [TK]D-Fender | coppice, And VoIP isn't mass-market. |
| 03:53.40 | coppice | exactly |
| 03:53.58 | [TK]D-Fender | coppice, thus is life.... but it IS improving rather well now... |
| 03:54.13 | *** join/#asterisk NirS_ (n=Nir@87.68.26.6.cable.012.net.il) |
| 03:54.19 | [TK]D-Fender | coppice, I'm waiting for OpenMoko Phase 2's release |
| 03:59.28 | AtomicDawg | I like the Polycom 320/330s... Speaker phone is awesome. Simple for people to use. Small deskspace. |
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| 04:01.54 | coppice | never seen one. polycom seemed to disappear from the market here when IP arrived. we used to see their stuff quite a bit before that |
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| 04:15.39 | thesndguru | anyone recommend a ip wireless phone and where to buy |
| 04:15.42 | [TK]D-Fender | coppice, What brands prevail in your area? |
| 04:15.48 | [TK]D-Fender | thesndguru, ... |
| 04:15.50 | [TK]D-Fender | ~wifisip |
| 04:15.51 | jbot | Wi-Fi SIP phones suck. All of them. HARD. Some only slightly less than others... |
| 04:16.06 | thesndguru | oh bummer |
| 04:16.15 | thesndguru | what about the sip apps on mobiles |
| 04:16.34 | coppice | if IP wireless phone == wifi, they suck |
| 04:16.35 | coppice | if IP wireless phone == DECT with an IP base station they are fine |
| 04:16.56 | thesndguru | i was thinking wifi no base required then |
| 04:17.10 | thesndguru | but if they still shooking then i'll wait |
| 04:17.12 | coppice | then you want something that sucks. |
| 04:17.31 | coppice | they won't stop sucking. wifi makes them suck |
| 04:17.36 | thesndguru | anyone use there mobiles to connect to asterisk |
| 04:17.55 | thesndguru | hanging out for a new mobile to do that with |
| 04:19.12 | thesndguru | but what about just in home, fair enough out and about |
| 04:19.30 | thesndguru | i'm in wa and wifi spots are hard enough to find |
| 04:26.26 | *** join/#asterisk m4sk (n=gabriel@201008158136.user.veloxzone.com.br) |
| 04:28.44 | [TK]D-Fender | thesndguru, for around home, just get an ATA + regular cordless phone. |
| 04:29.13 | *** join/#asterisk drako (n=ljd@nelug/coreteam/luisjose) |
| 04:29.16 | thesndguru | got three, just figure wants next |
| 04:30.10 | thesndguru | is that all you guys use |
| 04:33.56 | coppice | a DECT IP phone works well |
| 04:34.58 | tzanger | coppice: yeah I saw the SIP DECT stuff form aastra yesterday |
| 04:35.25 | *** join/#asterisk CoolGuy21 (n=Tilt@cpe-76-81-1-73.socal.res.rr.com) |
| 04:35.26 | CoolGuy21 | hi |
| 04:35.38 | CoolGuy21 | how do i configure my Rhino 4 port fxo on asterisk please help? |
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| 05:03.54 | bkruse_home | hey |
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| 05:22.12 | dlynes_laptop | CoolGuy21: same way as a tdm400p |
| 05:22.45 | dlynes_laptop | CoolGuy21: If you want, try talking to Dr-Linux...I know he was using a Rhino 4 port before |
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| 06:03.25 | DragoraN | iam using cisco ip communicator with asterisk and i cannot place a call but i can dial numbers |
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| 06:17.57 | blshkv | hi people, I have several SIP accounts and want to use them all in the same time by using asterisk and my linksys IP phone at home. |
| 06:18.00 | blshkv | So far the only problem is that I have half way sound (SIP->PSTN only) with pfingoTalk. The rest of SIP accounts and asterisk's demo echo test (extention 600) is just working fine. |
| 06:18.59 | blshkv | i'm stuck right now and have no idea what to try. Any hints? |
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| 06:32.33 | DarKnesS_WolF | anyone have problem with zaptel 1.2.19 ? |
| 06:35.40 | DarKnesS_WolF | ZT_CHANCONFIG failed on channel 4: No such device or address (6) |
| 06:35.45 | DarKnesS_WolF | even everything was working :-s |
| 06:46.58 | *** join/#asterisk fordfrog (n=fordfrog@gentoo/developer/fordfrog) |
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| 06:51.27 | kikoy | guys among the voicemail formats: which utilizes lower cpu? wav49,gsm,wav? |
| 06:51.54 | DragoraN | kikoy: more correct question would be "on which side" (server,client) |
| 06:52.06 | kikoy | DragoraN: server side |
| 06:52.24 | DarKnesS_WolF | wav as i think |
| 06:52.28 | DragoraN | kikoy: lower cpu - wav, but high bandwidth |
| 06:52.47 | DragoraN | wav is near to raw (uncompressed) format |
| 06:52.51 | kikoy | high bandwidth could also mean high IO... |
| 06:52.53 | kikoy | right? |
| 06:53.01 | DarKnesS_WolF | IO ? |
| 06:53.05 | DragoraN | input/output |
| 06:53.07 | DarKnesS_WolF | kikoy: no he means connetion bandwidth |
| 06:53.22 | DarKnesS_WolF | ah i thought IRQ or so ;-) |
| 06:53.29 | DragoraN | kikoy: yes.. but once it is used asterisk holds it in RAM i think, so no problem |
| 06:54.04 | DragoraN | i prefer gsm |
| 06:54.05 | kikoy | yeah i understand its about connection bandwidth... but with wav, files tends to be bigger. so recording those files eats your HD. |
| 06:54.10 | kikoy | ^^ |
| 06:54.27 | DarKnesS_WolF | kikoy: use GSM |
| 06:54.27 | DragoraN | of mp3 ;) |
| 06:54.33 | kikoy | but I guess with a striped HD, this can be overcome. |
| 06:54.34 | DarKnesS_WolF | it's in between ;-) |
| 06:54.50 | kikoy | GSM uses more cpu power. |
| 06:54.52 | DarKnesS_WolF | DragoraN: i cen use format_mp3 to record in mp3 ? the voicemail ? |
| 06:54.53 | *** join/#asterisk |dennis| (n=dennis@200.32.236.20) |
| 06:54.56 | kikoy | or resources |
| 06:55.04 | DragoraN | DarKnesS_WolF: i think you can |
| 06:55.08 | DarKnesS_WolF | kikoy: yes but low bandwidth |
| 06:55.09 | DarKnesS_WolF | coool |
| 06:55.21 | kikoy | thanks... i understand now. |
| 06:55.23 | DragoraN | kikoy: for example some users cannot allow email more than 3MB into the inbox.. |
| 06:56.00 | kikoy | DragoraN: my users will be accessing their voicemail locally so bandwidth is not an issue. |
| 06:56.21 | *** join/#asterisk implicit (n=bayan@210.16.55.38) |
| 06:56.23 | DragoraN | kikoy: ok, but you will need big voicemail storage |
| 06:56.52 | kikoy | i just want my VoIP server to be able to handle larger number of concurrent calls. |
| 06:57.03 | kikoy | AFAIK asterisk has a limit of about 200. |
| 06:57.47 | DragoraN | kikoy: if your hardware fits it, no problem to get more than 200 |
| 06:57.49 | DragoraN | i think.. |
| 06:58.19 | DragoraN | kikoy: look.. asterisk it just for signaling until you have access to PSTN via asterisk |
| 06:58.28 | kikoy | oh really? cool. |
| 06:58.30 | DragoraN | calls travel through LAN |
| 06:58.40 | kikoy | DragoraN: like peer to peer? |
| 06:58.44 | DragoraN | kikoy: yes |
| 06:58.50 | kikoy | oh i see. |
| 06:59.00 | DragoraN | wasnt that ironic? :) |
| 06:59.38 | kikoy | hmmm... but how about if i'm doing outbound calls via zaptel device? |
| 07:00.06 | DragoraN | this is done via asterisk.. |
| 07:00.07 | DragoraN | but |
| 07:00.16 | DragoraN | zaptel, what is zaptel |
| 07:00.48 | DragoraN | it is BRI/PRI ? |
| 07:02.05 | Nugget | a zaptel device is any of the digium cards. POTS or PRI |
| 07:02.12 | DragoraN | cool.. |
| 07:02.39 | DragoraN | BRI has 2 (B) bearer channels and 1 (D) signaling channel |
| 07:02.49 | DragoraN | so with BRI you have 2 concurrent calls allowed |
| 07:03.02 | DragoraN | but with PRI, you have 22B+1D (in USA) |
| 07:03.14 | DragoraN | 30B in EU |
| 07:03.22 | DragoraN | so, 22 concurrent calls |
| 07:03.43 | DragoraN | do you have a card(s) which will make more than 200 concurrent calls |
| 07:06.02 | Nugget | the most dense card I know of is a 4xPRI which would be 4x23 (in the states) totalling 92. |
| 07:06.16 | DragoraN | ... :) |
| 07:06.27 | DragoraN | kikoy: so you wont make 200 concurrent calls ever :) |
| 07:06.34 | Nugget | not sure how much luck you'd have running more then one in the same machine. |
| 07:06.43 | DragoraN | :) |
| 07:06.48 | Nugget | the old zaptel cards were really flaky for that, but the newer stuff seems a lot less awfu |
| 07:06.51 | Nugget | er, awful |
| 07:07.05 | DragoraN | kikoy: dont worry about the asterisk server |
| 07:07.17 | DragoraN | nice soloution is cisco + asterisk |
| 07:07.24 | Nugget | for purely VoIP there's no such limit. |
| 07:07.28 | DragoraN | cisco for access to PSTN ak gatekeeper |
| 07:07.41 | Nugget | but if you're needing 200 concurrent calls you're probably not going to be happy with VoIP over Internet. |
| 07:07.47 | DragoraN | power of VoIP is p2p |
| 07:08.21 | *** join/#asterisk scardinal (n=supreme@port816.ds1-suoe.adsl.cybercity.dk) |
| 07:08.51 | kikoy | Nugget: there's a Sangoma Card - the A108DX which can handle up to 8 E1 ports. so that's 8x30. |
| 07:08.56 | Nugget | spiffy |
| 07:08.57 | DragoraN | i work as technician in Zurich Insurance call centre (Zurich Connect) and there are 300 agents.. :) |
| 07:09.02 | kikoy | making it 240 |
| 07:09.24 | kikoy | DragoraN: tell us your experience |
| 07:09.31 | DragoraN | kikoy: but the coding/decoding is done on there cards, not on the asterisk machine CPU |
| 07:10.02 | DragoraN | kikoy: :) now are just working 30agents.. but there are a few PRI cards in Avaya |
| 07:10.10 | kikoy | DragoraN: a few guys told me that if i'm gonna do transcoding... my concurrent calls will be limited. |
| 07:10.18 | DragoraN | kikoy: we use Avaya, no Asterisk, no Cisco, but no difference |
| 07:10.29 | kikoy | hmmm.. |
| 07:10.45 | DragoraN | kikoy: if transcoding is done on server's CPU, its an bottleneck |
| 07:11.05 | DragoraN | for example SIP for 300 users :) thats _bad_ |
| 07:11.06 | kikoy | Well, i won't know all these stuff unless I try. I'm also working on a Call Center. |
| 07:11.17 | kikoy | right now it's a 50-seater call center. |
| 07:11.31 | DragoraN | kikoy: asterisk? |
| 07:11.37 | kikoy | CW |
| 07:11.40 | DragoraN | CW? |
| 07:11.44 | kikoy | yeah |
| 07:11.47 | kikoy | CallWeaver |
| 07:12.04 | DragoraN | i dont know this :) |
| 07:12.12 | DragoraN | openpbx.. |
| 07:12.17 | *** join/#asterisk Vorondil (n=vorondil@unaffiliated/vorondil) |
| 07:12.18 | kikoy | i guess this is a moderated channel... so its not a good idea to talk about it |
| 07:12.26 | DragoraN | :)) |
| 07:12.28 | kikoy | yeah... formerly OpenPBX |
| 07:12.33 | kikoy | ^^ |
| 07:12.49 | DragoraN | there is no matter if you use OpenPBX, Cisco, Avaya, Asterisk... its all the same, just management is different |
| 07:12.53 | kikoy | but i guess they're the same... since CW is a fork of asterisk. |
| 07:13.04 | kikoy | yeah. |
| 07:13.33 | kikoy | but i'll request for another server if i hit the *200 limit. |
| 07:13.44 | DragoraN | for example with avaya, you can replace interfaces on fly :) |
| 07:13.57 | DragoraN | kikoy: it runs on x86 PC? |
| 07:14.15 | kikoy | yeah... but convenience comes with a cost. |
| 07:14.22 | kikoy | ^^ |
| 07:14.33 | DragoraN | :) |
| 07:14.33 | kikoy | Avaya equipments are more pricey. |
| 07:14.34 | DragoraN | yes |
| 07:14.45 | DragoraN | i hate avaya :) |
| 07:14.51 | kikoy | :D |
| 07:14.52 | kikoy | lol |
| 07:15.16 | DragoraN | iam just 1st line support.. |
| 07:15.25 | DragoraN | Telefonica build the infrastructure |
| 07:15.28 | kikoy | DragoraN: have u used a predictive dialer during your stay on the call center? |
| 07:15.34 | kikoy | oh i see... |
| 07:15.41 | DragoraN | kikoy: oh, i dont know what is it |
| 07:16.05 | kikoy | it's a cool equipment |
| 07:16.13 | DragoraN | formely iam not user of the phone, iam admin.. i place calls only when i need to create a ticket with Spanish helpdesk |
| 07:16.22 | kikoy | what it does is it simply calls a lead a head of the agent. |
| 07:17.12 | kikoy | so agents won't be dialing the tel number of leads... the predictive dialer does. |
| 07:17.23 | DragoraN | kikoy: oh, predictive dialers... so call distribution for agents |
| 07:17.25 | DragoraN | yes... |
| 07:17.31 | kikoy | it gives the agent more talk time. |
| 07:17.47 | kikoy | Call Distribution is inbound. Predictive dialer is outbound. |
| 07:17.49 | kikoy | ^^ |
| 07:17.56 | DragoraN | oh |
| 07:18.03 | DragoraN | no outbound calls from agents are made |
| 07:18.07 | DragoraN | only inbound |
| 07:18.34 | kikoy | http://www.tutorial-reports.com/internet/telephony/predictive-dialer/working.php?PHPSESSID=a47d1a30e165612599f1a7dbcbf8721d |
| 07:18.38 | kikoy | that may be helpful |
| 07:19.45 | kikoy | our call center haven't started operating yet. telephone lines are still being installed. probably next week, we'll be operating full blast. |
| 07:19.47 | kikoy | ^^ |
| 07:20.11 | DragoraN | oh i unserstand |
| 07:20.15 | DragoraN | understand.. |
| 07:20.29 | DragoraN | kikoy: where are you located? |
| 07:20.35 | kikoy | Philippines here. |
| 07:20.49 | DragoraN | :) Slovakia, Europe |
| 07:21.02 | kikoy | oh... you're on the other end. |
| 07:21.07 | kikoy | Hello Europe! |
| 07:21.12 | DragoraN | :) |
| 07:22.06 | kikoy | but the funny thing is... on our initial operation we won't be using PBX stuff. the agents will be using an ordinary handset. |
| 07:22.35 | kikoy | until we get our Sangoma T1/E1 card(s) and a couple of channel banks. |
| 07:22.44 | DragoraN | :) |
| 07:23.06 | DragoraN | for example we have only headsets |
| 07:23.07 | kikoy | so i'm sort of experimenting at the moment. |
| 07:23.18 | kikoy | how about in your Call Center? |
| 07:23.20 | *** join/#asterisk newsmafia (n=newsmafi@wsip-68-224-174-204.sd.sd.cox.net) |
| 07:23.30 | DragoraN | and it is quite annoying when someon calls and you need to install the headset on your head quickly |
| 07:23.42 | kikoy | lol. |
| 07:24.00 | DragoraN | headphone with avaya proprietary audio jacks :D |
| 07:24.09 | kikoy | oh i see. cool! |
| 07:24.50 | DragoraN | http://www.headsetexperts.com/photos/49368-01-2.jpg |
| 07:24.55 | DragoraN | the thing in the middle |
| 07:25.06 | kikoy | the problem i see with ordinary handsets is that at the end of the agent's duty. they're gonna have stiff neck. |
| 07:25.51 | kikoy | so how do u dial with that device? |
| 07:26.12 | DragoraN | http://www.net54.hu/img/4622.jpg |
| 07:26.34 | kikoy | cool! |
| 07:26.43 | kikoy | that must be expensive. |
| 07:26.52 | DragoraN | it is :) |
| 07:27.13 | DragoraN | cisco's are better |
| 07:27.33 | DragoraN | avaya is like stoneage |
| 07:27.39 | DragoraN | or iceage :)] |
| 07:28.05 | kikoy | ^^ |
| 07:28.07 | kikoy | lol |
| 07:28.12 | DragoraN | when PoE switch comes down, 300 phone die |
| 07:28.17 | DragoraN | phone=phones |
| 07:28.24 | DragoraN | when it is back up |
| 07:28.40 | DragoraN | you need to go personally to each phone and login to an particular extension |
| 07:28.54 | DragoraN | using security code |
| 07:29.06 | DragoraN | (talking about avaya) |
| 07:29.12 | DragoraN | cisco - no problem |
| 07:29.21 | kikoy | oh... that must be very tedious thing to do. dont you allow your agents to do those stuff? |
| 07:30.06 | DragoraN | this is not a problem |
| 07:30.23 | DragoraN | its just stoneage |
| 07:30.28 | kikoy | lol |
| 07:30.32 | DragoraN | phone it ment to be thin client |
| 07:30.52 | DragoraN | specially in call centre with 300 agents |
| 07:31.11 | kikoy | DragoraN: thanks for the chat... i'll be right back. the telco guys have arrived. |
| 07:31.18 | DragoraN | ;) |
| 07:31.25 | DragoraN | give them my regards |
| 07:34.30 | *** join/#asterisk steliosk (n=Stelios@ipa226.211.tellas.gr) |
| 07:35.01 | DragoraN | ok kikoy, i am on duty also today, have a nice day! |
| 07:38.20 | *** join/#asterisk af_ (n=getsmart@81-174-8-254.dynamic.ngi.it) |
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| 08:18.24 | *** join/#asterisk RSAMan (n=a@dsl-244-242-13.telkomadsl.co.za) |
| 08:18.28 | RSAMan | hello guys |
| 08:18.35 | RSAMan | quick question : |
| 08:19.18 | RSAMan | i am noob.. i want to download asterisk and am presented with 2 versions # Asterisk 1.2.22 |
| 08:19.19 | RSAMan | # Asterisk 1.4.8 |
| 08:19.29 | RSAMan | is the earlier one the latest stable version ? |
| 08:32.17 | r0d3nt | yes |
| 08:33.30 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 08:35.59 | RSAMan | are u from Sa ? |
| 08:36.06 | RSAMan | r0d3nt: |
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| 09:43.03 | *** join/#asterisk delmar (n=delmar@ip-58-28-158-154.ubs-dsl.xnet.co.nz) |
| 09:44.22 | delmar | hey everyone. are there any known callerID issues with Asterisk1.4.x or Zaptel 1.4.x with tdm400 FXO trunk? |
| 09:45.11 | delmar | I ask because I finally got some time to look into why my callerID has been absent. rolled back from Asterisk 1.4.8 with Zaptel 1.4.4 to the latest 1.2.x versions, and my callerID has returned. |
| 09:45.38 | delmar | perhaps there is a config change I need to apply.. |
| 09:49.45 | delmar | maybe one of the variable names has changed and my config needs a tweak... |
| 09:51.11 | *** join/#asterisk af_ (n=getsmart@81-174-8-254.dynamic.ngi.it) |
| 10:36.17 | delmar | yep. thats all it was... a few small changes needed to comply with 1.4.x etc. |
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| 11:18.45 | Jiboneus | hello,... |
| 11:22.09 | Jiboneus | any body know of any call dispatcher based on asterisk? |
| 11:34.05 | mvanbaak | autodialer you mean ? |
| 11:34.35 | Jiboneus | an operator console of some sort |
| 11:35.25 | mvanbaak | try fop |
| 11:35.29 | mvanbaak | flash operator panel |
| 11:37.26 | Jiboneus | ok thanks,.. looking at it right now |
| 11:45.02 | *** join/#asterisk dikdust (n=dikdust@2001:1418:1c7:0:211:11ff:fed3:ecc7) |
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| 12:18.20 | mvanbaak | n/13 |
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| 12:40.27 | af_ | there is speech recognition sw that works with *? |
| 12:43.48 | *** join/#asterisk sysreq (n=sysreq@modemcable064.127-81-70.mc.videotron.ca) |
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| 13:10.43 | *** join/#asterisk [TK]D-Fender (n=Joe@64.235.216.2) |
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| 13:27.03 | *** join/#asterisk Daejeo1 (n=chatzill@124.62.146.40) |
| 13:27.40 | Daejeo1 | can anyone recommend me a hard video phone for asterisk? |
| 13:28.30 | shido6 | GXV-3000 |
| 13:28.35 | shido6 | $234.95 |
| 13:28.51 | Daejeo1 | is it really good? |
| 13:29.40 | Daejeo1 | I did not heard much good thing sabout grandstream |
| 13:29.46 | *** join/#asterisk friedrich| (n=friedric@e177245074.adsl.alicedsl.de) |
| 13:30.08 | [TK]D-Fender | Daejeo1, They all suck. |
| 13:30.11 | Daejeo1 | any other brand? |
| 13:30.54 | Corydon76-home | Any work with video phones is pretty much going to be cutting edge, as most people don't use them |
| 13:31.11 | Daejeo1 | [TK]D-Fender: right |
| 13:31.34 | Corydon76-home | Cisco apparently makes one, if their commercials are to be believed |
| 13:31.38 | coppice | the bottom line is video phones are pretty useless |
| 13:31.50 | Daejeo1 | someone sold a new GXV-3000 phone on ebay for 99$ |
| 13:32.01 | [TK]D-Fender | Daejeo1, means their worth 12$ ;) |
| 13:32.27 | shido6 | you could always go with Teleprescence |
| 13:32.48 | shido6 | w/ 3 65" flat panels |
| 13:33.19 | shido6 | its only about $300k |
| 13:35.46 | shido6 | or polycoms eye to eye |
| 13:35.56 | shido6 | eyeconnect |
| 13:36.07 | Daejeo1 | how much? |
| 13:36.11 | shido6 | muahahahah |
| 13:36.20 | shido6 | got your lottery ticket? |
| 13:36.24 | Daejeo1 | no |
| 13:36.35 | Daejeo1 | well I have to see on ebay |
| 13:36.41 | shido6 | um... |
| 13:36.43 | shido6 | yeah... |
| 13:36.50 | [TK]D-Fender | X-Lite / eyeBeam seem to be fine for me.... |
| 13:36.55 | Daejeo1 | maybe someone will sell " stolen one" |
| 13:37.03 | Daejeo1 | haha |
| 13:37.08 | shido6 | you know what... |
| 13:37.12 | shido6 | you may have something there.. |
| 13:37.54 | Daejeo1 | recently i got a new cisco 7060g for 67$ |
| 13:42.31 | shido6 | well |
| 13:42.42 | shido6 | there's the Polycom RPX HD system |
| 13:43.01 | shido6 | which gives you that Imax look and feel |
| 13:44.31 | shido6 | then there's the "24" version by CIT called teh Personal Video Interactive Companion |
| 13:44.51 | shido6 | for those incognito conferences in the field |
| 13:45.28 | shido6 | but you will need to break out the corporate card or some serious plastic for these |
| 13:49.42 | *** join/#asterisk keulin (n=cray@AMontpellier-152-1-64-87.w83-197.abo.wanadoo.fr) |
| 13:49.46 | tzanger | heh |
| 13:51.50 | *** join/#asterisk Sci_05 (i=Sci_05@ts.bestserversllc.net) |
| 13:51.55 | Sci_05 | morning all |
| 13:52.11 | Sci_05 | anyone know if the linksys ata's are any better then the grandstream ones? |
| 13:55.38 | [TK]D-Fender | ~gs |
| 13:55.39 | jbot | GrandSuck phones & gateways are cheap junk which should be avoided with extreme prejudice. |
| 13:55.40 | [TK]D-Fender | ^^^^^^^^^^^^^^^^^^^^^^ |
| 13:56.58 | Sci_05 | [TK]D-Fender: yes I have knowed that for some time now, i was woundering if the linksys are any better as an alternitive |
| 13:57.13 | [TK]D-Fender | Sci_05, definately. |
| 13:57.31 | Sci_05 | ok |
| 13:58.27 | Daejeo1 | [TK]D-Fender: what hard IP phones do you have? |
| 13:58.44 | [TK]D-Fender | Daejeo1, Polycom almost exclusively. |
| 13:59.02 | [TK]D-Fender | Daejeo1, personally IP501 & IP301 |
| 13:59.23 | Sci_05 | [TK]D-Fender: ever use an ip 4000? |
| 13:59.34 | [TK]D-Fender | Daejeo1, at work, IP 301,430,600,601, some Uniden UIP-200's, and an Aastra 57i CT |
| 13:59.44 | [TK]D-Fender | Sci_05, Nope, but I've configured them for a client. |
| 13:59.59 | Sci_05 | i used them a few times, they are damn nice |
| 14:00.04 | [TK]D-Fender | Sci_05, I use a SoundStation2W w/ SPA-2002 personally. |
| 14:00.21 | Sci_05 | i will check thoes out |
| 14:00.24 | [TK]D-Fender | Sci_05, wireless non-IP restricted fun. |
| 14:00.41 | Daejeo1 | [TK]D-Fender: I need one for my home, which one do you recommend ? |
| 14:00.55 | Daejeo1 | I just need a basic one |
| 14:01.13 | Sci_05 | here is one for you, with the polycom provisioning scrips, is there a way to set ringtones in it? |
| 14:01.17 | [TK]D-Fender | Daejeo1, what, jsut a normal phone? IP320 then : http://www.telephonydepot.com/product_p/105-058-320.htm |
| 14:01.39 | Daejeo1 | I mean |
| 14:01.43 | Daejeo1 | for home use |
| 14:01.47 | [TK]D-Fender | Sci_05, Yes, you can convert whatever you want to ulaw and use it as a ring tone. I have a large collection. |
| 14:02.01 | [TK]D-Fender | Daejeo1, See above. SICK price.... |
| 14:02.11 | Sci_05 | what section of the script is that at? |
| 14:02.37 | Daejeo1 | got it |
| 14:03.13 | [TK]D-Fender | Sci_05, under "sounds" and "sampled". Go read the admin guide |
| 14:03.16 | Daejeo1 | [TK]D-Fender: did you ever played with cisco phones on asterisk? |
| 14:03.27 | Daejeo1 | play* |
| 14:03.50 | [TK]D-Fender | Daejeo1, Nope. Overpriced, poorer call handling, PoE non-standard on most, paid firmware? ICK. |
| 14:03.54 | Corydon76-home | Cisco phones tend to be more trouble than they're worth |
| 14:04.03 | Hymie | hey, has anyone got the distinctive ring working with the polycoms? I've tried about 4 billion variations on alert-info, no joy for me... |
| 14:04.15 | Daejeo1 | firmware you can get for free |
| 14:04.17 | [TK]D-Fender | Hymie, yup |
| 14:04.22 | Daejeo1 | I got 8.6 |
| 14:04.24 | Corydon76-home | Hymie: yes, but you have to define the classes in the config file first |
| 14:04.31 | Daejeo1 | it is working very nicely |
| 14:04.35 | Hymie | hmmmm |
| 14:04.38 | [TK]D-Fender | ok, I'm off for a while. |
| 14:04.39 | [TK]D-Fender | BBL |
| 14:04.44 | Corydon76-home | Hymie: in the Polycom XML config file |
| 14:05.09 | Hymie | Corydon76-home: sec, I'll show you what I added to me config, if you have a moment |
| 14:05.20 | Corydon76-home | ~pb |
| 14:05.20 | jbot | A Pastebin is a place to paste your stuff without flooding the channel. Here are links to a few : http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org |
| 14:07.20 | Hymie | hmm, nuts |
| 14:07.48 | Hymie | http://pastebin.ca/628647 |
| 14:08.02 | Hymie | I had two lines *like* this in my global.cfg file |
| 14:08.10 | Hymie | those are for paging though, I know.... |
| 14:08.37 | Hymie | but the ones I had provided no joy, I deleted them in a fit of rage at their imputent bitliness |
| 14:10.00 | Corydon76-home | Hymie: the ringer number has to match a defined class in ringType |
| 14:11.17 | Corydon76-home | Hymie: I've never seen anyone use spaces in the alertInfo value, though |
| 14:11.32 | Hymie | ok, checking these things (thanks) |
| 14:11.56 | Hymie | so |
| 14:12.05 | Hymie | I could match to CUSTOM_1 I guess? |
| 14:12.14 | Corydon76-home | Any of them is fine |
| 14:13.06 | Corydon76-home | Each of those is different, though if you actually want ring-answer, you'll need to either customize one or use class 4, which is already ring-answer |
| 14:13.31 | Hymie | ok, I think I get it.... so, I really shouldn't have to define anything but the alertInfo line in my own config, and just reference a ringer that exists in sip.cfg |
| 14:13.42 | Corydon76-home | Hymie: are you doing a Set(__ALERT_INFO=Ring Answer) ? |
| 14:13.57 | Hymie | I don't want to do ring answer, I was just testing (that worked, btw) for something else |
| 14:14.13 | Hymie | now, aside from ring answer, I'm just looking for distinctive ring |
| 14:14.34 | Corydon76-home | Each of those provides a different ring cadence |
| 14:15.10 | Hymie | so, I can just put "<alertInfo voIpProt.SIP.alertInfo.1.value="Booger" voIpProt.SIP.alertInfo.1.class="8"/>"" in my config? |
| 14:15.19 | Hymie | and use booger for the alert info? |
| 14:15.31 | Corydon76-home | case-sensitive, but yes |
| 14:15.48 | Hymie | then, <CUSTOM_1 se.rt.8.name= should ring (the 8 matching the class)? |
| 14:15.55 | Corydon76-home | Correct |
| 14:15.58 | Hymie | hmm |
| 14:16.06 | ManxPower | I learned everything I know about Polycom ringtone setting from the Wiki |
| 14:16.08 | Hymie | why be evil and call it a 'class' polycom :P |
| 14:16.19 | Hymie | ManxPower: I found zero info there on these specifics |
| 14:16.31 | Hymie | ManxPower: the alert_info part, sure.. but this part? |
| 14:16.35 | Corydon76-home | Polycom actually offers a 90-minute training video via the web |
| 14:16.42 | Hymie | Corydon76-home: really? |
| 14:17.13 | Hymie | Corydon76-home: I imagine you must pay 4 billion dollars and a carrabibean island too? |
| 14:17.15 | *** join/#asterisk riddlebox (n=james@75-132-225-75.dhcp.stls.mo.charter.com) |
| 14:17.16 | Corydon76-home | Yeah, it's how you become a certified reseller |
| 14:17.18 | *** join/#asterisk Infested (n=infested@24.148.112.10) |
| 14:17.32 | Hymie | doh |
| 14:17.54 | ManxPower | I think it is free |
| 14:18.11 | Corydon76-home | which makes sense, actually. You should know how to configure their phones if you're going to resell them |
| 14:18.58 | Hymie | Corydon76-home: yes, but it's silly for it not to be free, so fewer people are mucking about, and more people want to install them .. but ManxPower may have brought a ray of sun to this conversation |
| 14:19.02 | ManxPower | All I did was take a web based test or two to become polycom certified |
| 14:19.08 | Hymie | no money? |
| 14:19.13 | Hymie | no boatloads of money |
| 14:19.27 | ManxPower | But everything was setup beforehand by the vendor for which I was becoming the certified tech for. |
| 14:19.28 | Corydon76-home | Hymie: I was saying yes to your 'really' not yes to your 4 billion comment |
| 14:19.32 | Hymie | carried to them via a gaggle of midget women? |
| 14:19.36 | ManxPower | So I have NO idea if there is any money involved. |
| 14:20.03 | Corydon76-home | One of our guys took the training yesterday while I was in the office |
| 14:20.11 | Hymie | ok guys, I'll have to look into this.. |
| 14:20.20 | Hymie | Corydon76-home: do yo uknow if there are no costs, some costs? |
| 14:20.27 | Corydon76-home | I don't know, no |
| 14:22.03 | Hymie | ok, well.. thanks guys.. now to test my new aural delights ;) |
| 14:26.20 | *** join/#asterisk uwe (n=uwe@dogbert.palnet.com) |
| 14:28.39 | Sci_05 | anyone ever use the Patton SmartLink M-ATA before? anygood? |
| 14:29.08 | *** join/#asterisk CrazyTux[m] (n=CrazyTux@cpe-76-175-64-30.socal.res.rr.com) |
| 14:36.17 | uwe | hello, ive been suffering from echo/delay and bad sound quality with asterisk on a dell poweredge with SAS harddisks, ive tried to do hdparam -t /dev/sda5 (its configured as raid5) and zttest reported as low as 93.652344 with avarage 99.864159 ... i dont know what to say, i disabled whatever hardware i could, i redistributed the inturrupts but i still have problems, not to mention that asterisk sometimes simply stops sending the calls to the queue and i g |
| 14:36.18 | uwe | et an evergrowing queue ... no, i cant get an expert to do the work, and yes, im willing to do what ever reading/debugging/research that can get things to work, and yes, i currently have only remote access to the machine, and yes its a production machine ... if someone knows of what odd things might possibly cause this to happen, please tell me where to go and read more, and appologies for continuesly searching for help here ... |
| 14:37.56 | Sci_05 | uwe: what phones are you using, using a provider for termination or a card (pri,pots,etc)? |
| 14:38.50 | uwe | Sci_05, its TE410P card |
| 14:39.37 | Sci_05 | phones? |
| 14:39.40 | ManxPower | uwe: what verision of Asterisk? |
| 14:40.21 | uwe | its now 1.2.16 |
| 14:40.51 | ManxPower | uwe: are you using onboard ethernet |
| 14:40.57 | uwe | ive been trying since 1.2.14 or 15, i tried up to 1.2.17 |
| 14:41.11 | uwe | yes, there were 2 and i disabled one of them |
| 14:41.17 | Sci_05 | do you have the echocancelation turned on in your zapata.conf |
| 14:42.05 | uwe | echocancel=yes |
| 14:42.11 | ManxPower | disable them both and put in a PCI ethernet card. |
| 14:42.25 | ManxPower | onboard ethernet (expecially GigE) frequently cause issues. |
| 14:43.17 | Sci_05 | echotraining, what is that set to? |
| 14:43.30 | uwe | i have this card : 07:08.0 Ethernet controller: Intel Corporation 82541GI/PI Gigabit Ethernet Controller (rev 05) , what card do you suggest i should use ? |
| 14:44.02 | uwe | it was set to 800, and to 128, and now its completely removed |
| 14:44.35 | ManxPower | uwe: We use ALL 3-Com cards, but most any PCI card should work |
| 14:44.49 | Hymie | uwe: by the way, is it just that the echo cancellation is the problem, or that you get other sound problems? ... I run into this sort of problem with 1.2.x all the time, if I have a lot of voicemails being recorded, or if I'm recording all the conversations.. because the disk can't keep up when playing prompts (reading from the disk) and writng at the same time |
| 14:45.25 | uwe | ok, great, ill get a 3-Com as soon as possible and try it out, is there any symptoms that can confirm this suspision ? |
| 14:45.28 | Hymie | it also stutters the audio if you are talking, and the recording to disk of that conversation gets lagged by slow drive access |
| 14:46.49 | uwe | Hymie, its mainly echo and delays, stutters happen too, but around twice a week and when that happens i usually restart asterisk |
| 14:47.19 | Hymie | uwe: I will say this.. while 1.4.x is annoyingly betaish, it does seem to handle echo cancellation 1000 times better than 1.2.x |
| 14:47.36 | Hymie | uwe: there were places I could not get rid of echo, that now is gone with 1.4.x |
| 14:47.42 | ManxPower | Hymie: Most anything handles EC better than Asterisk 1.2 |
| 14:47.59 | ManxPower | uwe: you need to fix your audio quality issues before you work on the echo cancel problem. |
| 14:48.16 | ManxPower | you will NEVER get good EC until your other audio issues are resolved |
| 14:48.34 | uwe | ok, so my agenda should include a) trying 3-com nic B) switching to asterisk 1.4 ... |
| 14:48.54 | Hymie | uwe: again, are you writing a lot to the disk? recording all calls or some such? |
| 14:49.00 | ManxPower | uwe: if you don't have a 3-COM NIC handy, try any other decent nic |
| 14:49.06 | Hymie | uwe: that's a major problem for asterisk, or was with 1.2 .. not sure about 1.4 |
| 14:49.07 | uwe | i have all recording disabled |
| 14:49.11 | Hymie | ok |
| 14:49.24 | Hymie | uwe: is your box swapping a lot? how much ram to do you have |
| 14:49.33 | Hymie | uwe: you want to keep swapping to an utter minimum |
| 14:50.05 | uwe | 2 GB total, using 1687924k |
| 14:50.11 | *** join/#asterisk Ebola (n=Ebola@host86-138-208-67.range86-138.btcentralplus.com) |
| 14:50.22 | uwe | no swap is used (48k only) |
| 14:50.52 | uwe | i never noticed it used swap |
| 14:52.07 | Hymie | ok, that's good at least |
| 14:52.32 | Hymie | biab |
| 14:53.11 | uwe | ok |
| 14:56.55 | *** join/#asterisk zeedo (n=zeedo@reboot-robot.net) |
| 15:08.41 | uwe | Thank you all, ill leave to get a 100 Mb 3-com NIC, a pci-x i suppose is what is should get, or would a pci suffice? |
| 15:10.51 | *** join/#asterisk minkus (n=minkus@pool-71-182-32-236.clrkwv.east.verizon.net) |
| 15:12.54 | ManxPower | I would suggest PCI |
| 15:14.06 | Hymie | this may not be what you want to hear, either.. but dell doesn't necessarily mean quality. I"ve always.. always had problems with crappy chipsets, does that motherboard have a lame, evil, crappy via chipset on it? or perhaps a less lame, but sometimes crappy sis chipset on it? |
| 15:15.03 | Hymie | in multiple installations, I've seen customers with heavy interrupt (like asterisk) or DMA applications have nothing but pain with a via chipset, I'd never touch one with a 10000 foot pole |
| 15:16.30 | uwe | i have to check that Hymie , i'm not sure today, ill try after hours |
| 15:17.09 | Hymie | uwe: type 'lspci' and it all you see is via this and via that .. you'll know |
| 15:19.24 | uwe | lspci | grep -i "\(sis\|via\)" yields nothing, so i think things are good |
| 15:19.49 | Hymie | good |
| 15:19.53 | Hymie | freaking via bastards |
| 15:19.55 | coppice | Intel have produced more crappy chipsets than VIA |
| 15:20.42 | Hymie | only because intel has been around longer, and produces more chipsets |
| 15:21.07 | Hymie | and even then, you're full of it ;) |
| 15:21.07 | Hymie | yes, intel has made some mistakes |
| 15:21.09 | Hymie | but via is a steaming pile of crap |
| 15:21.14 | uwe | ManxPower, is their a particular reason why pci would fit better than pci-x ? |
| 15:23.50 | *** join/#asterisk r0d3nt (i=nobody@foster.stonedcoder.org) |
| 15:24.28 | uwe | anyway, thank you Hymie , ManxPower , Sci_05 and everyone, i hope ill report back with good news :) |
| 15:25.17 | *** join/#asterisk danalien (n=danalien@unaffiliated/danalien) |
| 15:29.23 | *** join/#asterisk karleeto (i=karl@gentoo.karlhaines.com) |
| 15:32.17 | *** join/#asterisk zeedo (n=zeedo@reboot-robot.net) |
| 15:49.20 | *** join/#asterisk fakhir (n=Fakhir@unaffiliated/fakhir) |
| 15:56.17 | *** join/#asterisk santiago (i=santiago@debian/developer/santiago) |
| 15:58.39 | *** join/#asterisk luke-jr_work (n=luke-jr@ip72-206-114-236.om.om.cox.net) |
| 15:58.48 | luke-jr_work | Anyone seen a PAP2-NA with blinking blue power LED? |
| 16:01.01 | *** join/#asterisk cpm (n=Chip@pdpc/supporter/active/cpm) |
| 16:05.19 | *** join/#asterisk booray (n=ray@adsl-75-7-150-84.dsl.irvnca.sbcglobal.net) |
| 16:07.14 | *** join/#asterisk CrazyTux[m] (n=CrazyTux@cpe-76-175-64-30.socal.res.rr.com) |
| 16:07.15 | booray | Random voip provider question; with a single DID routed to an asterisk setup, will it ever ring busy or just ring through? i.e. I want to know if I only *need* one number do I need multiple DIDs to orig/term multiple calls? |
| 16:08.03 | booray | that is, if anyone is awake |
| 16:08.11 | booray | probably too hungover... |
| 16:09.25 | luke-jr_work | booray: you can't route a DID |
| 16:09.41 | luke-jr_work | a DID is just the # |
| 16:09.45 | luke-jr_work | the calls come in on channels |
| 16:09.54 | luke-jr_work | a single DID can use multiple channels or not, depends on your ITSP |
| 16:09.59 | booray | gotcha |
| 16:10.20 | luke-jr_work | generally, if you have unlimited minutes, they will only do 1 channel |
| 16:10.26 | luke-jr_work | and if you pay per minute, unlimited |
| 16:10.34 | booray | makes sense |
| 16:11.01 | booray | thanks for the clarification luke |
| 16:13.41 | booray | now if I could just choose between this massive list of providers |
| 16:18.29 | *** join/#asterisk cpm (n=Chip@pdpc/supporter/active/cpm) |
| 16:21.09 | *** join/#asterisk arcanine (n=saxon_m2@210.4.36.175) |
| 16:21.34 | luke-jr_work | booray: hard choice |
| 16:21.42 | luke-jr_work | they all suck in at least one way, IME |
| 16:21.58 | luke-jr_work | well, except Voipjet-- never had problem with them |
| 16:22.04 | luke-jr_work | but they're termination only |
| 16:22.34 | booray | yeah, need at least one did.. :-/ |
| 16:23.00 | luke-jr_work | $1/mo from SellVoIP, if you don't mind a complete lack of tech support and months of downtime ;p |
| 16:23.06 | booray | haha |
| 16:23.28 | booray | I'm looking at les.net... haven't seen anything negative about them and Ive been googling for a bit |
| 16:23.28 | jbroome | what a bargin! |
| 16:23.34 | luke-jr_work | $6/mo from MyPhoneCompany with 1 channel/unlimited incoming, if you don't mind Flash website |
| 16:23.40 | booray | cheap enough to try for a little |
| 16:23.57 | luke-jr_work | never heard of les |
| 16:24.22 | booray | as long as I can use lynx, right? |
| 16:24.54 | luke-jr_work | Flash isn't lynx-compatible :) |
| 16:24.55 | cpm | I can't get voipjet to reset my lost password, and I can't just open another account, as the info is already there. I like voipjet, just wish they'd respond to email |
| 16:25.24 | luke-jr_work | voipjet is for businesses |
| 16:25.28 | luke-jr_work | lost password is suspicious |
| 16:25.31 | booray | 530 mailbox full |
| 16:25.37 | luke-jr_work | wtf? |
| 16:25.43 | booray | just guessting |
| 16:26.57 | luke-jr_work | dunno... Voipjet is very good about responding when I email, and even sending out notification of stuff before it happens |
| 16:27.41 | cpm | luke-jr_work, they are excellent at keeping folks up to date on what is going on, agreed. But none of us are supposed to admit we are using them :) |
| 16:28.18 | luke-jr_work | wow, les.net looks better than sellvoip based on prices... |
| 16:28.28 | luke-jr_work | ignoring the crap about 4 channel limit |
| 16:28.41 | luke-jr_work | cpm: why not? |
| 16:28.56 | cpm | didn't you read your user agreement? |
| 16:29.02 | luke-jr_work | probably |
| 16:29.04 | luke-jr_work | why? |
| 16:29.17 | cpm | It states clearly right there, not to tell folks |
| 16:29.22 | luke-jr_work | O.o |
| 16:29.28 | cpm | :) |
| 16:29.37 | luke-jr_work | maybe you signed up after me or smth |
| 16:29.39 | luke-jr_work | :p |
| 16:30.19 | cpm | I signed up, , err, back in, , , wow, has it been that long? apr '05 |
| 16:30.41 | luke-jr_work | les.net looks like the breakpoint between per-minute and unlimited is 273 minutes or more |
| 16:31.04 | cpm | Ahh, now they have password recovery! |
| 16:31.13 | *** join/#asterisk rmayorga (n=rmayorga@unaffiliated/rmayorga) |
| 16:31.16 | cpm | luke-jr_work, thanks for bringing this up/ |
| 16:32.01 | cpm | Doh! |
| 16:32.06 | luke-jr_work | :p |
| 16:32.13 | cpm | it wasn't my password that I forgot!, (sheesh, what a moron) |
| 16:33.32 | luke-jr_work | any idea what exchanges les.net has DIDs for? |
| 16:39.16 | cpm | hrmm, looks like they want you to sign up before they will tell you |
| 16:43.52 | *** join/#asterisk ToyMan (n=Stuart@user-12lcquh.cable.mindspring.com) |
| 16:45.39 | karleeto | i'm about to be doing a voip job thats going to span accross 3 cities using iax trunks (if thats the best way to go), all calls will need to get routed to the main location, then will be distributed to the proper place by the secretary at that main location |
| 16:46.07 | karleeto | anyone have any reccomendations on some reading i could do? or maybe what approach i should take? |
| 16:48.57 | *** join/#asterisk BudgetDedicated (n=BudgetDe@s5593c2e9.adsl.wanadoo.nl) |
| 16:49.04 | luke-jr_work | cpm: I'll call Les-- want me to ask about any specific one for you? |
| 16:49.06 | luke-jr_work | or booray |
| 16:49.20 | BudgetDedicated | once I have xtdummy loaded do I need to do anything to have astersik make good use of it? |
| 16:49.42 | karleeto | les.net is ABSOLUTELY HORRIBLE |
| 16:49.47 | luke-jr_work | karleeto: ?!?! |
| 16:49.47 | karleeto | i know from experience |
| 16:50.23 | karleeto | man, i paid them up 60 dollars and after two months, they still hadn't ported my DIDs |
| 16:50.52 | karleeto | i run a business here, i can't wait for two months, my phone numbers ARE my business |
| 16:51.40 | luke-jr_work | I'm not porting :p |
| 16:51.51 | karleeto | and the tech support is absolutly horrible, "Jason" i beleive it was, was just uncaring and really just not very nice |
| 16:52.06 | karleeto | luke-jr_work: well, good luck getting support when it doesnt work |
| 16:52.22 | luke-jr_work | sounds like sellvoip :x |
| 16:52.38 | luke-jr_work | how about VoiceStick |
| 16:52.41 | karleeto | luke-jr_work: and you'll remember me saying this once you're unsatisfied like i am |
| 16:53.15 | karleeto | luke-jr_work: i haven't tried anyone else yet, but the people in my LUG reccomended someone, want to me go check my inbox and tell you what they reccomended? |
| 16:54.33 | *** join/#asterisk dmaxwell (n=david@h216-235-8-214.host.egate.net) |
| 16:55.12 | dmaxwell | Any members of the Asterisk project around? |
| 16:55.23 | luke-jr_work | karleeto: sure |
| 16:55.30 | karleeto | luke-jr_work: k, gimmie one sec |
| 16:56.07 | luke-jr_work | VoiceStick? |
| 16:56.47 | karleeto | luke-jr_work: VOIPStreet, http://www.voipstreet.com/ |
| 16:57.14 | karleeto | luke-jr_work: also, Junction Networks, http://www.junctionnetworks.com/ |
| 16:57.34 | luke-jr_work | Junction lacks DIDs in my area |
| 16:57.39 | karleeto | luke-jr_work: my friends all use VOIPStreet and Junktion Networks for a backup trunk |
| 16:57.42 | luke-jr_work | VoipStreet's website is b0rked :p |
| 16:58.00 | karleeto | luke-jr_work: maybe, but they have a great staff apparently |
| 16:58.00 | luke-jr_work | wtf, VoipStreet has Japan DIDs ;p |
| 16:58.38 | *** join/#asterisk Rachad (n=a@41.250.48.135) |
| 16:58.55 | luke-jr_work | VoipStreet is on the expensive end |
| 16:59.18 | *** join/#asterisk rbd (n=rbd@adsl-074-229-183-112.sip.rmo.bellsouth.net) |
| 16:59.38 | *** join/#asterisk newsmafia (n=newsmafi@wsip-68-224-174-204.sd.sd.cox.net) |
| 17:00.42 | *** join/#asterisk aptura (n=jondoe@S010600a0c93f6f7e.vs.shawcable.net) |
| 17:00.48 | rbd | the asterisk manager 'Redirect' call can be used to transfer calls off to a different server (SIP for instance), right? I'm asking because all of the examples I've seen do a local transfer where the call stays on the same asterisk server |
| 17:03.55 | *** join/#asterisk wunderkin- (n=wunderki@ip68-2-62-143.ph.ph.cox.net) |
| 17:26.55 | *** join/#asterisk JT_ (n=jon@unaffiliated/jt) |
| 17:32.19 | *** part/#asterisk zotz (n=zotz@24.244.163.157) |
| 17:36.02 | *** join/#asterisk saftsack (n=oliver@p54A7F5CA.dip.t-dialin.net) |
| 17:51.05 | *** join/#asterisk zeedo (n=zeedo@reboot-robot.net) |
| 17:59.16 | *** join/#asterisk jj56 (n=jflo@a80-127-56-82.adsl.xs4all.nl) |
| 18:01.11 | jj56 | Hi All! |
| 18:01.28 | jj56 | I'm having some trouble getting the callerid to display on my budgettone 100.. |
| 18:01.38 | jj56 | I just use Set(CALLERID(all)=Joe Smith <2095551212>) |
| 18:01.48 | jj56 | then Dial(Sip/141|30|tmo) |
| 18:01.55 | jj56 | That should be enough, right? |
| 18:03.27 | *** join/#asterisk kirberich (n=robert@i53871CB6.versanet.de) |
| 18:03.31 | kirberich | good evening |
| 18:04.51 | kirberich | i have a problem with a zaphfc pci card. I'm trying to run it in nt-mode, but the interaction between card and telephone doesn't seem to work |
| 18:05.20 | kirberich | softwarewise everything seems fine, dmesg says the card runs in nt-mode, zaptel doesn't report any errors, ztcfg shows the card just fine |
| 18:05.55 | jj56 | but asterisk doesnt pick up incoming calls? |
| 18:06.16 | kirberich | there are two jumpers on the card (a billion bipac), but they don't seem to be documented anywhere |
| 18:06.20 | kirberich | right jj56 |
| 18:06.40 | kirberich | ztmonitor also doesn't recieve anything when i pick up the phone or try to dial something |
| 18:06.43 | jj56 | I had that too the other day, |
| 18:06.59 | jj56 | make sure you have an extension that matches everything to see if it reacts: |
| 18:07.07 | jj56 | exten => _.,1,DoYourStuff |
| 18:07.51 | kirberich | i started asterisk with a lot of v's, it should tell me if it doesn't find an extension |
| 18:08.26 | jj56 | i assumed that too, had the same prob, solved it for me.. |
| 18:08.34 | kirberich | hm ok, let me try |
| 18:10.08 | Aces1Up | if i wanted to know when someone joined, or left a conference via an event, what would i use to notify me? ami? |
| 18:14.10 | kirberich | jj56, i got something interesting |
| 18:14.12 | kirberich | in dmesg: |
| 18:14.13 | kirberich | zaphfc: empty HDLC frame or bad CRC received (framelen = 171, stat = 0xff, card = 0). |
| 18:14.14 | kirberich | zaphfc: empty HDLC frame or bad CRC received (framelen = 2, stat = 0xff, card = 0). |
| 18:14.14 | kirberich | zaphfc: empty HDLC frame or bad CRC received (framelen = 85, stat = 0xff, card = 0) |
| 18:14.17 | kirberich | and a lot more of those |
| 18:15.08 | jj56 | hmmm, i'm sorry, i use a fritz/fcpci/chan_capi, don't know much about zaphfc |
| 18:17.57 | [TK]D-Fender | Aces1Up, One way. |
| 18:18.28 | Aces1Up | tkd what way? |
| 18:18.58 | Aces1Up | i'm looking at the events at voip-info looks like i want to capture event meetme join, and meetme leave. |
| 18:19.11 | Aces1Up | are these events always sent to any ami listening? |
| 18:19.18 | Aces1Up | or do i have to request them? |
| 18:20.15 | [TK]D-Fender | Aces1Up, To any with listen priviledges |
| 18:20.30 | Aces1Up | cool, so they just flow on in? right on.. |
| 18:20.39 | Aces1Up | just gotta catch em. |
| 18:20.52 | kirberich | brb |
| 18:21.13 | [TK]D-Fender | Aces1Up, You could do a DIFF on the list on a given frequency. I have a web-monitory for MeetMe so i can see who's in/out. |
| 18:21.31 | [TK]D-Fender | Aces1Up, I'm preparing a semi-private #asterisk MeetMe conf on my server |
| 18:22.41 | Aces1Up | tkd cool, when will that be ready? |
| 18:23.38 | [TK]D-Fender | Aces1Up, Well its functional NOW for basic viewing, but I want to add alick-to-kick, etc to it. It currently jsut lists who's on, how long, and who's talking (colour coded) |
| 18:25.07 | *** join/#asterisk kirberich (n=robert@i53871CB6.versanet.de) |
| 18:25.19 | kirberich | jj56, you don't use your card in nt-mode, right? |
| 18:28.17 | jj56 | kirberich: i hooked it up to my phoneline, not to a phone |
| 18:29.31 | *** join/#asterisk Cyon (n=cyon@216.179.31.170) |
| 18:30.30 | *** join/#asterisk phace (n=phace@SE400.PPPoE-2250.sa.bih.net.ba) |
| 18:30.51 | phace | hi all... anyone managed to run asterisk with eicon diva ? is there an manual for it ? |
| 18:31.55 | jj56 | Is there anything else I need to do besides calling 'Set(CALLERID(all)=Joe Smith <2095551212>)' to get the digits on the display of a SIP budgettone ? I'm using Asterisk 1.4.8 |
| 18:32.36 | jj56 | After a lot of googling that's all I can find about it |
| 18:33.13 | jj56 | When I Dial the bugettone, the display shows the callerid of the phone itself, not from the caller, or the one I specified with Set |
| 18:33.19 | jj56 | That's weird, right? |
| 18:34.40 | *** join/#asterisk sysreq (n=sysreq@modemcable064.127-81-70.mc.videotron.ca) |
| 18:34.47 | jj56 | i'm using a zap device to make the call btw, but that should matter afaik |
| 18:45.01 | mvanbaak | jj56: it's because of the budgettone |
| 18:45.06 | mvanbaak | budgetcrap |
| 18:45.17 | mvanbaak | that phone is so darn bad |
| 18:45.38 | jj56 | heh, i know it sucks, voicequality is crap too, but I'm using it for testing.. |
| 18:45.48 | mvanbaak | install x-lite |
| 18:45.50 | jj56 | I'm doing the right thing tho? |
| 18:45.53 | mvanbaak | way better for testing |
| 18:46.02 | mvanbaak | that Set line looks ok indeed |
| 18:46.13 | mvanbaak | I would put it in 2 lines though |
| 18:46.21 | mvanbaak | one for CALLERID(name) |
| 18:46.27 | mvanbaak | and one for CALLERID(num) |
| 18:46.27 | jj56 | ok, thanks, that saves me some headaches ;) |
| 18:46.40 | jj56 | Tried that, no luck |
| 18:46.44 | rbd | can the asterisk manager 'redirect' command be used to transfer calls to an external source (e.g. from one asterisk server to another via SIP) (the presence of a 'context' parameter in it makes me wonder...) |
| 18:46.49 | jj56 | ohwell, lets blame it on the phone.. |
| 18:47.21 | mvanbaak | lol |
| 18:47.37 | mvanbaak | sure rbd |
| 18:47.50 | rbd | ok |
| 18:47.58 | mvanbaak | redirect it to <some_nr> in the context used for outbound calling |
| 18:48.26 | mvanbaak | bbl, food |
| 18:49.02 | jj56 | eet smakelijk |
| 18:49.22 | jj56 | works in x-lite indeed, thanks |
| 18:53.09 | *** join/#asterisk |dennis| (n=dennis@200.32.236.20) |
| 18:57.13 | rbd | I need to transfer a call out to another SIP system via the manager API's redirect. So it looks like how I would do that would be to call redirect with the destination extn/context being an outbound dial rule. Unfortunately I don't know exactly which SIP system I need to transfer to until I call the redirect command, so I can't put the sip host/port in my dial plan hardcoded. Any way to forward this sip server information along in re |
| 19:00.28 | rbd | I guess when the call is transferred to the outbound dial rule, it could run an agi script to dynamically find out which sip server to use.... |
| 19:13.45 | *** join/#asterisk MoutaPT (n=MoutaPT@a212-113-161-241.netcabo.pt) |
| 19:15.53 | MoutaPT | Hi guys, there's almost 6 months since my asterisk deploy, I'm installing a new one and using TE2XXP card, currently my Meetme rooms are ok, but some times the sound quality is not that good, on show channels during Meetme call I see ZAP/pseudo, is this expected? My timer source should be TE2XXP so no Zap/pseudo?? am I wrong? |
| 19:17.37 | MoutaPT | any one here??? |
| 19:19.14 | *** join/#asterisk zeedo (n=zeedo@reboot-robot.net) |
| 19:19.34 | BudgetDedicated | How do i make a call. wait for the line to answer, call a second number and connect the two lines for the duration of their call? |
| 19:20.09 | MoutaPT | u can use dialout call files or asterisk manager interface |
| 19:22.11 | *** join/#asterisk Op3r (n=Op3r@121.97.193.145) |
| 19:24.25 | BudgetDedicated | MoutaPT: i know about dialing out but what is the command for connecting them |
| 19:24.29 | BudgetDedicated | i use call files |
| 19:24.58 | MoutaPT | the idea of dialout call files is that |
| 19:25.03 | MoutaPT | make two call legs |
| 19:25.07 | MoutaPT | and bridge them |
| 19:25.38 | MoutaPT | if you want both calls to be outbound pstn call you may need to use local channel |
| 19:25.45 | MoutaPT | as work around for it |
| 19:26.04 | BudgetDedicated | yes both are outbound |
| 19:26.13 | MoutaPT | so make one like |
| 19:26.17 | MoutaPT | ZAP/1/XXXXXXXXX |
| 19:26.22 | MoutaPT | and other |
| 19:26.29 | MoutaPT | local/XXXXX@yourcontext |
| 19:26.43 | MoutaPT | and in your context make the Dial(ZAP/XXXXXX) |
| 19:26.46 | MoutaPT | got it? |
| 19:27.06 | BudgetDedicated | ok i'll try to get that thank you |
| 19:27.15 | MoutaPT | ;) |
| 19:27.19 | MoutaPT | good luck |
| 19:27.44 | MoutaPT | http://www.voip-info.org/wiki/index.php?page=Asterisk+Local+channels |
| 19:28.33 | *** join/#asterisk |dennis| (n=dennis@200.32.236.20) |
| 19:28.46 | kirberich | if there is anyone here who successfully uses a hfc pci card in nt-mode, i really need some help with this. I can't get the telephone to do anything and I don't see an error anywhere |
| 19:29.40 | *** join/#asterisk Putzz (n=me@CPE001a707d4d4e-CM00111ae07ac2.cpe.net.cable.rogers.com) |
| 19:35.12 | *** join/#asterisk UQlev (n=yuriy@ykulyev.logos.cy.net) |
| 19:36.55 | *** join/#asterisk Koshatul (n=evangeli@ppp188-33.static.internode.on.net) |
| 19:37.27 | MoutaPT | can any one tell me if ZAP/Pseudo is related with ztdummy and not when using Digium cards??? |
| 19:37.32 | Hymie | hmm, has anyone noticed that the ring volume on the polycom 330s sucks donkey balls? |
| 20:00.10 | *** join/#asterisk tako-san (n=Tako-san@154.5.212.245) |
| 20:20.25 | *** join/#asterisk bkruse_home (n=kruz@69.73.127.92) |
| 20:22.31 | *** part/#asterisk bkruse_home (n=kruz@69.73.127.92) |
| 20:29.04 | *** join/#asterisk bkruse_home (n=kruz@69.73.127.92) |
| 20:30.12 | luke-jr_work | Is there any open source firmware IP phone? |
| 20:30.24 | *** part/#asterisk bkruse_home (n=kruz@69.73.127.92) |
| 20:32.04 | killfill | hey |
| 20:32.23 | killfill | <PROTECTED> |
| 20:33.13 | killfill | <PROTECTED> |
| 20:33.36 | *** part/#asterisk MoutaPT (n=MoutaPT@a212-113-161-241.netcabo.pt) |
| 20:33.47 | *** join/#asterisk CunningPike (n=CunningP@S010600095b33697f.vc.shawcable.net) |
| 20:38.14 | Qwell | luke-jr_work: openmoko |
| 20:39.42 | *** part/#asterisk UQlev (n=yuriy@ykulyev.logos.cy.net) |
| 20:41.13 | killfill | ah.. it was the nat.. :P |
| 20:43.39 | *** join/#asterisk fakhir (n=Fakhir@unaffiliated/fakhir) |
| 20:43.44 | *** join/#asterisk Givemelove (n=bozoo@216.70.173.176) |
| 20:43.50 | Givemelove | hello there |
| 20:44.03 | Givemelove | anybody available to help me on a 401 issue? |
| 20:44.28 | Givemelove | I'm having some issues authenticating on Asterisk with a Cisco 7941 phone |
| 20:45.29 | killfill | guys.. i have a bnch of grandstream GXP2000, and cannot find where to change its language. anyone knows where is this option?.. |
| 20:45.36 | *** join/#asterisk bbryant_ (n=Brett@user-24-214-124-177.knology.net) |
| 20:45.49 | Givemelove | This is the logs I have regarding that 401 : http://pastebin.com/m207ceba |
| 20:47.10 | bbryant_ | Givemelove, that's how the sip handshake works, first it sends a register without the auth info, then the unauthorized packet tells it how to send the password |
| 20:47.24 | Givemelove | ? |
| 20:47.24 | bbryant_ | that part always happens |
| 20:47.32 | Givemelove | yeah, but there's nothing more |
| 20:47.36 | *** join/#asterisk rmayorga_ (n=rmayorga@unaffiliated/rmayorga) |
| 20:47.42 | Givemelove | the phone does not send another request |
| 20:47.51 | Givemelove | it just keeps trying with the same packets |
| 20:48.09 | bbryant_ | ah, did you configure it with a sip account? |
| 20:48.17 | Givemelove | yeah |
| 20:48.20 | Givemelove | user 102 |
| 20:48.53 | Givemelove | this issue is related to asterisk AND a cisco 7941 |
| 20:52.03 | jj56 | Does AEL(2) have anything to seperate contexts into namespaces? Or do all contexts always end up in the same 'global' context? |
| 20:52.50 | jj56 | The idea is to have multiple users define their own dialplans, without the risk of having them define and call eachothers contexts.. |
| 20:53.13 | jj56 | this way you can use a single asterisk installation to host multiple dialplans.. |
| 20:59.44 | luke-jr_work | Qwell: Openmoko supports some IP phone? |
| 21:01.39 | Qwell | it's an open source firmware for phones, that has an IP stack |
| 21:01.42 | Qwell | so, technically..yes |
| 21:03.26 | killfill | hey.. this phone support "Busy Line Field". but its not working. where should be the configuration in asterisk? |
| 21:08.39 | *** join/#asterisk drako (n=ljd@nelug/coreteam/luisjose) |
| 21:09.49 | *** join/#asterisk af_ (n=getsmart@81-174-8-254.dynamic.ngi.it) |
| 21:11.29 | luke-jr_work | Qwell: technically, IP phones must use the IP stack for the telephony |
| 21:11.48 | Qwell | nothing stopping you from putting a SIP client in openmoko |
| 21:11.55 | luke-jr_work | AFAIK, no phone supported by OpenMoko has the bandwidth for it |
| 21:12.50 | Qwell | eh? |
| 21:13.54 | luke-jr_work | they all only support the very-low-bandwidth IP connection |
| 21:14.04 | luke-jr_work | like 56kbit speeds |
| 21:22.43 | *** join/#asterisk riddlebox (n=james@75-132-225-75.dhcp.stls.mo.charter.com) |
| 21:26.29 | *** join/#asterisk andresmujica (n=andresmu@201.245.236.215) |
| 21:37.21 | *** join/#asterisk _DAW (n=_DAW@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 21:39.18 | *** part/#asterisk andresmujica (n=andresmu@201.245.236.215) |
| 21:48.42 | CoolGuy21 | ok |
| 21:59.42 | riddlebox | are the artdio phones any good? |
| 22:03.56 | JackEStorm | if I have a TDM400P and a TE12xP in my system, what should be the loading order for layer 1? |
| 22:05.11 | *** join/#asterisk rmayorga (n=rmayorga@unaffiliated/rmayorga) |
| 22:07.21 | *** join/#asterisk m4sk (n=gabriel@189.13.97.138) |
| 22:13.25 | *** join/#asterisk michaelo (n=michaelo@adsl-068-159-111-129.sip.gsp.bellsouth.net) |
| 22:13.49 | *** join/#asterisk x86 (n=x86@p3m/member/x86) |
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| 22:50.59 | *** join/#asterisk s1icksho3s (n=slick_sh@c-71-227-143-180.hsd1.wa.comcast.net) |
| 22:51.03 | s1icksho3s | Hello. |
| 22:53.09 | s1icksho3s | I'm running asterisk 1.4.8 on Debian 4.0. No configure/make issues, and everything seems to work. However, asterisk crashes when an IAX call is answered, and module res_features.so is not loaded. If I load res_features.so, the call goes through. |
| 22:54.16 | s1icksho3s | I believe res_features.so is used for call parking, correct? I'm not using call parking, so I'm not sure why res_features.so needs to be loaded. Does anybody have any ideas? |
| 22:56.59 | shido6 | wow |
| 22:57.09 | shido6 | wtf man |
| 22:57.13 | shido6 | just laod it and leave it |
| 22:57.15 | shido6 | whats the prob? |
| 22:57.16 | shido6 | :) |
| 22:57.36 | s1icksho3s | Ha. I suppose. But I'd like to know what's going on. |
| 22:57.43 | shido6 | set verbose 50 |
| 22:57.50 | shido6 | go into logger.conf turn debug on |
| 22:57.54 | shido6 | make a call |
| 22:58.25 | s1icksho3s | Let me give that a shot. |
| 22:58.30 | *** join/#asterisk Cybertoy (n=cybertoy@dsl254-123-112.nyc1.dsl.speakeasy.net) |
| 23:06.05 | s1icksho3s | Ah. When running asterisk with -c (as opposed to -r) I got an error telling me ast_bridge_call was undefined. And sure enought, ast_bridge_call is defined in res_features.so. |
| 23:06.22 | s1icksho3s | s/enought/enough |
| 23:09.13 | tzafrir_home | s1icksho3s, so another module needs it |
| 23:09.43 | tzafrir_home | res_* modules are modules that may export functions for other modules |
| 23:11.15 | s1icksho3s | I see, thanks. |
| 23:12.36 | *** join/#asterisk |dennis| (n=dennis@200.32.236.20) |
| 23:14.10 | *** join/#asterisk aptura (n=jondoe@S010600a0c93f6f7e.vs.shawcable.net) |
| 23:15.15 | CoolGuy21 | how |
| 23:15.22 | russellb | how! |
| 23:15.51 | JackEStorm | ugh this PRI is killing me |
| 23:17.25 | *** join/#asterisk santiago (i=santiago@debian/developer/santiago) |
| 23:18.06 | shido6 | y? |
| 23:18.31 | JackEStorm | its sitting in yellow all the time |
| 23:19.11 | *** part/#asterisk Cybertoy (n=cybertoy@dsl254-123-112.nyc1.dsl.speakeasy.net) |
| 23:19.20 | shido6 | dont be afraid to get some help from the teclo |
| 23:19.26 | shido6 | its your money |
| 23:19.40 | shido6 | and digium. |
| 23:19.40 | JackEStorm | been trying to get it |
| 23:21.01 | saftsack | hi, does wctdm allocate much memory while loading it? |
| 23:21.08 | saftsack | i want to load it on a machine with 32mb of ram |
| 23:28.51 | JackEStorm | shido6: I really think it's something at the CO or at the smartjack...but waiting on ATT is like watching the grassgrow. |
| 23:29.03 | aptura | Jack whats the issue? |
| 23:29.15 | CoolGuy21 | shido6 are u here? |
| 23:29.19 | JackEStorm | and on top of that, I'm trying to look up all the opcodes and return codes for my card so I can figure out what it really going on. |
| 23:31.23 | JackEStorm | aptura: trying to figure out why my TE120P is sitting in yellow (per the driver) but red per zttool, and then after that why I can't configure with my TDM400P in the system |
| 23:31.47 | JackEStorm | s/configure with/configure it with/ |
| 23:33.22 | *** join/#asterisk NightMonkey (n=NightMon@pdpc/supporter/sustaining/NightMonkey) |
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| 23:43.20 | *** join/#asterisk Berny_ (n=Berny_@ip-89-168-5-83.cust.homechoice.net) |
| 23:43.25 | Berny_ | hi all |
| 23:43.32 | s1icksho3s | Hello, Berny_ |
| 23:43.32 | saftsack | hi |
| 23:43.33 | Berny_ | I have an implementation question... |
| 23:44.05 | Berny_ | I was thinking of using asterisk as a VoIP gateway with a Sipura unit |
| 23:44.54 | Berny_ | so the sipura unit registers and works with the asterisk server, and the asterisk server would relay the call out to different providers depending on the country I was calling |
| 23:45.09 | Berny_ | possible? Or I need to look at this a little more? |
| 23:50.19 | x86 | definitely possible |
| 23:50.39 | x86 | asterisk can do Least Cost Routing (LCR) with little problems |
| 23:51.11 | x86 | you can also daisy-chain dial statements or even use a loop to iterate over a list of providers to try, for failover and redundancy |
| 23:51.53 | x86 | although when you start trying to combine LCR with redundant dialplan setups, it tends to get kinda complex ;-) |
| 23:51.56 | *** join/#asterisk MoutaPT (n=Blink@a213-22-40-195.cpe.netcabo.pt) |
| 23:53.07 | MoutaPT | hi Guys, when I'm joining a Meetme room on my * I notice ZAP/Pseudo has been created, I'm using TE2XXP, so this shouldn't appear as the timer source is Digium card and not ztdummy, am I wrong? |
| 23:54.37 | x86 | MoutaPT: if you're using a zaptel card, you shouldn't be using ztdummy at all anyway |
| 23:54.48 | Berny_ | x86, thanks for the info |
| 23:54.57 | x86 | Berny_: no prob :) |
| 23:55.05 | Berny_ | just wanted to see how it was going to go, presumably it would work as well on the inbound? |
| 23:55.24 | x86 | no |
| 23:55.29 | Berny_ | so calls from VoIP providers, I could pass back to the sipura box? |
| 23:55.35 | x86 | ah, yes |
| 23:55.57 | x86 | but you don't have control over the redundancy at that point (assuming VoIP inbound) |
| 23:56.10 | Berny_ | ok, when I move to Ireland next month, looks like I am installing asterisk on my ubuntu box (Currently sitting in storage)! |
| 23:56.24 | x86 | sounds fun :) |
| 23:56.28 | x86 | where are you moving from? |
| 23:56.31 | Berny_ | london |
| 23:56.38 | x86 | ah |
| 23:56.53 | Berny_ | so I am going to get a VoIP provider here, and I have one in Aus (That's where I am from) |
| 23:57.02 | x86 | moving up to better beer and less facism ;) |
| 23:57.15 | Berny_ | the Sipura box I have doesn't deal with multiple providers very well |
| 23:57.26 | x86 | no, generally not |
| 23:57.33 | x86 | sipura is crap in my experience |
| 23:57.42 | Berny_ | why is that? |
| 23:57.51 | x86 | you should really get an IP phone and have it talk natively to asterisk |
| 23:58.25 | x86 | i've got three polycom 601's sitting here at my house :) |
| 23:58.44 | Berny_ | the only problem I have had with the sipura box is that every now and again it used to not route calls properly |
| 23:59.02 | Berny_ | so I put a cron job on my server to reboot it at 3am every morning, no more problems |
| 23:59.11 | x86 | yeah that's a hack ;) |
| 23:59.13 | Berny_ | yes |
| 23:59.39 | x86 | i've had to do similar things in the past to get it to do DTMF handling properly, etc |
| 23:59.50 | Berny_ | odd |
| 23:59.59 | Berny_ | it was a pain in the ass to get it configured right too |
