| 00:00.58 | [av]bani | Qwell: 7985 street price is only $2480 !! |
| 00:01.08 | Qwell | not too shabby |
| 00:01.34 | Qwell | [av]bani: When should I expect mine in the mail? |
| 00:01.47 | [av]bani | after you knock over a couple convenience stores |
| 00:02.19 | Qwell | want a video of it, or? |
| 00:02.24 | *** join/#asterisk cpm (n=Chip@pdpc/supporter/sustaining/cpm) |
| 00:10.21 | *** part/#asterisk SwK (n=Silik0nJ@12-219-147-107.client.mchsi.com) |
| 00:17.40 | *** join/#asterisk MikeJ[Laptop] (n=vircuser@d14-69-8-30.try.wideopenwest.com) |
| 00:18.15 | [av]bani | ~internet |
| 00:18.17 | jbot | it has been said that internet is for porn |
| 00:18.25 | Qwell | what's porn? |
| 00:18.33 | [TK]D-Fender | ~porn |
| 00:18.34 | jbot | Porn remains one of the largest problems with Open Source Software. Often causing development delays, flooded links and, in extreme cases, disabling programmers ability to type. |
| 00:18.42 | Qwell | haha |
| 00:18.56 | Qwell | That's awesome |
| 00:18.59 | [TK]D-Fender | jbot knows all, sees all, tells little ;) |
| 00:21.14 | bronze | what else does jbot do? |
| 00:21.29 | Qwell | ~lart bronze |
| 00:21.36 | bronze | ~jbot |
| 00:21.39 | jbot | hmm... jbot is only marginally useful at best, or a silly little bugger |
| 00:21.42 | SarahEmm | ~sarahemm |
| 00:21.43 | jbot | you are probably kitrichernesses. |
| 00:22.06 | bronze | JBot is another Java-based IRC bot but with a twist. JBot is architected as a small kernel augmented by dynamically loadable and updatable plugins |
| 00:22.42 | *** join/#asterisk _m_ (n=m@fbta199.fbta.uni-karlsruhe.de) |
| 00:26.57 | bronze | Current plugins include: Anagram Bullshit Command Confession Ebonify Factoid Google Idle Insult Javadoc Karma Nickometer Patent Regex Seen Slang Statistics Time TinyURL Wikipedia |
| 00:31.39 | digime | does anyone have a standard telco contract for phone service |
| 00:31.42 | Qwell | uh oh |
| 00:31.49 | Qwell | bkw_: You're almost out of DIDs :p |
| 00:32.02 | bkw_ | Qwell, I just commited some build system nitpicks for freeswitch on the Mac |
| 00:32.10 | bkw_ | .dylib on the mac not .so mmmkay |
| 00:32.12 | Qwell | fun |
| 00:33.58 | asterboy | Removed some FXO cards from * and updated zaptel.conf & zapata.conf to comment out the channels, now I'm getting: "Failed to register zone" "Inappropriate ioctl for device" |
| 00:34.09 | SarahEmm | y |
| 00:34.30 | asterboy | Any suggestions aprecieated |
| 00:35.03 | *** join/#asterisk littleball (n=littleba@26.203-123-30.leased.qala.com.sg) |
| 00:35.19 | asterboy | For now, I'm putting back what I took out. |
| 00:43.24 | *** join/#asterisk dgorski (n=dgorski@c-69-245-111-167.hsd1.mi.comcast.net) |
| 00:44.54 | *** join/#asterisk litage (n=nick@203.220.55.70) |
| 00:47.41 | asterboy | ~g729 |
| 00:47.42 | jbot | [g729] It was in November 1995 that the G.729 standard, also referred to as CS-ACELP was adopted by the ITU, a United Nations organization. Similar, quality-wise, to 32 kbps ADPCM, G.729 offers toll quality speech. Furthermore, being only an 8 kbps codec, G.729 offers opportunities for significant increases in bandwidth utilization to existing telephony and ... |
| 00:48.04 | asterboy | and ... ? |
| 00:48.36 | asterboy | Anyway, the fail to register zone is fixed once the FXO cards are reinstalled. |
| 00:48.57 | asterboy | Why the heck should they have to be installed to load a zone file? |
| 00:49.14 | *** join/#asterisk SwK (n=Silik0nJ@12-219-147-107.client.mchsi.com) |
| 00:55.48 | *** join/#asterisk froguz (n=froguz@252-130-222-201.adsl.terra.cl) |
| 00:56.13 | asterboy | Getting [modprobe: FATAL: Error running install command for wcfxo and wctdm] |
| 00:56.18 | asterboy | On system boot |
| 00:56.26 | Qwell | asterboy: take them out of your init script |
| 00:56.29 | asterboy | Doesn't seem to stop the cards from working. |
| 00:59.25 | asterboy | well modules is the only thing in init |
| 00:59.37 | Qwell | What distro? |
| 00:59.37 | asterboy | so take them out of /etc/modules.conf then? |
| 01:00.05 | asterboy | err...modprobe.conf |
| 01:02.00 | asterboy | LFS, similar to debian |
| 01:03.07 | zigman | hey nice to know i'm not the only one using lfs ;) |
| 01:06.38 | dgorski | it seems like besides redhat and suse, everything else is similar to or based in part on Debian |
| 01:06.59 | asterboy | closet anyways |
| 01:07.54 | dgorski | gentoo an ubnto both openly admit it |
| 01:08.02 | dgorski | I'm a gentoo fan personally |
| 01:08.28 | Nugget | Linux is poo/ |
| 01:08.49 | _Sam-- | [av]bani : how is your teliax working right now? |
| 01:09.01 | dgorski | nugget, poo, I get it |
| 01:09.03 | *** join/#asterisk pengyong (n=lala@210.21.32.85) |
| 01:13.34 | Qwell | dgorski: sorry, did you just say Gentoo admitted they're like Debian? |
| 01:17.01 | litage | if you get an ISDN line into your equipment in a data centre, can you dial into that ISDN line from a POTS line at home? |
| 01:18.38 | [TK]D-Fender | And what about Slackware, huh? |
| 01:21.32 | file[laptop] | eeeeep |
| 01:21.52 | file[laptop] | Nugget: I'm not made of chicken :( |
| 01:22.27 | Qwell | file[McNugget] |
| 01:22.29 | dgorski | Qwell: sorry: I did not mean to say that. I was implying it's roots were in debian, as in "I hate debian, I'm gonna build gentoo" |
| 01:22.46 | dgorski | I used to love slackware |
| 01:22.59 | dgorski | it was my first distro... before the kernel was 2.x |
| 01:23.21 | Nugget | I still use slackware because it stays out of my way, unlike all the other distros I've tried. |
| 01:23.31 | dgorski | try gentpp |
| 01:23.33 | dgorski | gentoo |
| 01:23.36 | Nugget | I have, I hate it. |
| 01:23.39 | dgorski | wow |
| 01:23.49 | Nugget | it's like bsd, but not as nice. |
| 01:23.52 | dgorski | that was it's purpose, stay out of your way |
| 01:23.53 | file[laptop] | Nugget: I'm going to sacrifice a chicken nugget at VON, just for you |
| 01:23.56 | dgorski | wow |
| 01:24.16 | dgorski | it's funny to hear someone say something I so totally disagree with |
| 01:24.58 | Nugget | I'm down to just two linux boxes now, though, so it's not too bad. zaptel and my gridmp box, both require linux. |
| 01:25.38 | dgorski | so you prefer... ? |
| 01:25.48 | *** join/#asterisk robin_sz (n=nospam@adsl.redpoint.org.uk) |
| 01:25.57 | Nugget | depends on what I'm doing. Anything you can do with linux you can do better with something else. |
| 01:26.13 | dgorski | that's the most untrue statement I've ever heard |
| 01:26.14 | Nugget | a mac's a better desktop, freebsd's a better server, and openbsd's a better firewall. |
| 01:26.24 | Nugget | and windows is better for gaming |
| 01:26.39 | *** join/#asterisk RoyK (n=roy@10.80-203-106.nextgentel.com) |
| 01:26.44 | file[laptop] | everything has it's place |
| 01:26.53 | Nugget | linux is best for zaptel drivers. |
| 01:26.57 | dgorski | I can't disagree with gaming and desktop, but I prefer linux for a server, but more than anything as a toolbox platform |
| 01:26.58 | [TK]D-Fender | Except WinME..... unless you are thinking of the Dumpster ;) |
| 01:27.19 | file[laptop] | [TK]D-Fender: no, that has it's place... in hell |
| 01:27.34 | [TK]D-Fender | :D |
| 01:28.16 | file[laptop] | hell consists of Windows ME running on P166 machines with 16MB of RAM... and a single 9600bps dialup connection for all inhabitants |
| 01:28.20 | RoyK | hi |
| 01:28.46 | RoyK | anyone here doing massive sip works? |
| 01:29.36 | RoyK | there seems to be a few leaks in chan_sip |
| 01:30.23 | dgorski | leaks? |
| 01:30.45 | RoyK | memory leaks |
| 01:31.19 | dgorski | I don't have that issue with chan_sip |
| 01:31.21 | RoyK | i have a rig with 1.2.5 and after a few thousand calls, it eats 300 megs |
| 01:31.45 | RoyK | dgorski: what volume? |
| 01:32.09 | dgorski | 70 stations off a small ast box with 1 T1 to in/out |
| 01:32.16 | dgorski | runs forever |
| 01:32.21 | RoyK | ok |
| 01:32.37 | *** join/#asterisk tengulre (n=tengulre@61.185.224.66) |
| 01:32.42 | *** join/#asterisk MooingLemur (n=troy@shells200.pinchaser.com) |
| 01:32.55 | RoyK | this is, say, 10k calls after a few hours, and it leaks, beleive me |
| 01:33.31 | *** join/#asterisk irieKEN (i=irieKEN@ppp-71-128-136-86.dsl.sndg02.pacbell.net) |
| 01:34.00 | dgorski | and you're sure it's SIP and not another part of your rig? |
| 01:34.05 | robin_sz | milky milky .... |
| 01:34.41 | file[laptop] | so what were you running before that didn't leak? |
| 01:35.05 | asterboy | figured it out. |
| 01:35.19 | asterboy | the zaptel.conf file sure does not like ";" for comments |
| 01:35.31 | RoyK | dgorski: i don't know, but the rig is only doing sip |
| 01:35.32 | asterboy | whereas the zapata.conf does not mind at all |
| 01:35.33 | *** join/#asterisk veto_laptop (i=mdkuser@cpe-66-69-38-192.satx.res.rr.com) |
| 01:35.49 | RoyK | file[laptop]: i ran 1.0 |
| 01:36.18 | asterboy | otherwise you get the symptoms of the modprobe fatals on boot and the failure to load region on phone pickup. |
| 01:37.09 | dgorski | RoyK: so you did not specifically trace it to the code in chan_sip? |
| 01:37.39 | *** join/#asterisk octothorpe_ (n=octothor@c-67-186-207-234.hsd1.ut.comcast.net) |
| 01:38.07 | RoyK | dgorski: i've tried, but i can't be sure. it's the closest i can get |
| 01:38.35 | RoyK | http://bugs.digium.com/view.php?id=6652 |
| 01:38.42 | asterboy | ok , so how do I get the dialtone to come on again after I hit "9". |
| 01:38.44 | asterboy | ??? |
| 01:39.21 | asterboy | works to dial out, but it would be nice to have some feedback after pressing 9 |
| 01:40.05 | dgorski | asterboy: isn't that what ignorepat= is for? |
| 01:40.11 | [TK]D-Fender | asterboy : IGNOREPAT => 9 |
| 01:40.35 | dgorski | RoyK: so you have channels piling up while your 'leak' is occurring? |
| 01:41.12 | RoyK | yes and no |
| 01:41.54 | asterboy | sweet, thanks Fender. |
| 01:42.00 | RoyK | the leak is constant, and the channel piling is occuring, but with a lower rate |
| 01:42.52 | asterboy | Also, after and incoming call, the voicemail from the VOIP provider kicks in when I hang up. |
| 01:43.01 | asterboy | How do I make the call just hangup? |
| 01:43.01 | dgorski | I would try to figure out why they are piling up. That really sounds like your problem. I don't have SIP channels pile up - I've never seen that. |
| 01:43.40 | file | they should disappear... |
| 01:44.29 | RoyK | dgorski: some channels are piling up, but i don't think that'ss the main problem |
| 01:45.15 | RoyK | dgorski: that is a problem indeed, but it's not the one leaking all the memory |
| 01:47.20 | dgorski | hung channels tie up both memory, and also threads |
| 01:49.01 | asterboy | What is the difference between "_9." and just "9"? |
| 01:51.04 | zigman | nothing |
| 01:55.43 | wunderkin | yes there is, notice the ., it will do a pattern match for anything starting with 9 |
| 01:56.07 | dgorski | the _ should be clue #1 |
| 01:56.23 | dgorski | that their not the same |
| 02:06.28 | *** join/#asterisk RV-Dioxide (i=appleboy@ip68-231-211-153.oc.oc.cox.net) |
| 02:16.08 | *** join/#asterisk asteriskmonkey (n=admin@69.158.149.183) |
| 02:16.34 | asteriskmonkey | is it possible to set two different echo settings on a single pri? |
| 02:16.49 | asteriskmonkey | ie have channels 1-10 with these settings and 11-23 some other settings? |
| 02:18.04 | *** join/#asterisk DeeJay[2] (n=bleh@37-179.sh.cgocable.ca) |
| 02:19.00 | DeeJay[2] | If I'm using multiple redundant servers for SIP clients, is there a pratical way for calls coming from a PRI to find the right server where the SIP client is connected so that I can forward the call? |
| 02:19.28 | I-MOD | asteriskmonkey: it should work just fine |
| 02:19.48 | DeeJay[2] | Let's say server A has a PRI ..and server B and C are hosting SIP registrations. |
| 02:20.39 | DeeJay[2] | A call comes from the PRI to server A but how server A can figure out which of server B or C has the user registered on? |
| 02:20.44 | I-MOD | you could setup dundi between them |
| 02:20.56 | I-MOD | s/between/among |
| 02:21.03 | DeeJay[2] | ok, dundi can achieve it seemlessly? ;) |
| 02:21.19 | asteriskmonkey | yes |
| 02:21.21 | I-MOD | define seamlessly |
| 02:21.27 | I-MOD | :) |
| 02:21.39 | DeeJay[2] | The best already implemented way to achieve it ;) |
| 02:21.40 | asteriskmonkey | set you expires to 0 |
| 02:22.02 | I-MOD | yeah, dundi should take care of your setup just fine |
| 02:22.19 | DeeJay[2] | ok... do I need some special provisioning? |
| 02:22.51 | asteriskmonkey | anyonw know if i can set 2 different sets of channels on a single pri .. ie i want to split it into two in my zapata so i can play with a serparate set of echo cancellors for a batch of channels |
| 02:22.52 | *** join/#asterisk jhnjwng (n=wj1918@pool-70-21-166-133.nwrk.east.verizon.net) |
| 02:23.14 | I-MOD | for the phone itself on iax.conf or sip.conf? no, but you should add a user specifically for dundi on all the servers |
| 02:23.26 | I-MOD | all the other setup is done in dundi.conf |
| 02:24.04 | dgorski | asteriskmonkey: did you try it? |
| 02:24.11 | I-MOD | asteriskmonkey: yes |
| 02:24.13 | DeeJay[2] | my users are in a database (thru realtime) |
| 02:24.21 | asteriskmonkey | try what? |
| 02:24.23 | DeeJay[2] | but both server B and C are using the same database. |
| 02:24.33 | dgorski | "anyonw know if i can set 2 different sets of channels on a" |
| 02:24.35 | dgorski | etc. |
| 02:24.38 | DeeJay[2] | with NFS for voicemail (MWI will work that way) |
| 02:24.40 | asteriskmonkey | no i want to |
| 02:24.43 | dgorski | what you are asking about |
| 02:24.44 | asteriskmonkey | im looking for info how to |
| 02:25.01 | asteriskmonkey | have you done it? |
| 02:25.24 | I-MOD | just specify two different channel => whatever and use different settings for each |
| 02:26.07 | asteriskmonkey | im not sure if i follow i want to set different gains and echo cancel options for a small group of channels.. |
| 02:26.09 | asteriskmonkey | is that possible? |
| 02:26.32 | dgorski | look at docs/README in the section "ABOUT CONFIGURATION FILES" |
| 02:26.37 | asteriskmonkey | :P |
| 02:26.51 | *** join/#asterisk Husk (n=tcg@202.55.153.169) |
| 02:26.58 | I-MOD | s/ look at docs/README in the section "ABOUT CONFIGURATION FILES" |
| 02:27.03 | I-MOD | crud..... |
| 02:27.10 | I-MOD | <PROTECTED> |
| 02:27.50 | Husk | Hey, when putting auth=md5 in my sip.conf file i get add_realm_authentication: Format for authentication entry is user[:secret]@realm |
| 02:28.00 | asteriskmonkey | damn i cant spell |
| 02:28.05 | Husk | the documentation im following says i need auth=md5 |
| 02:28.08 | dgorski | relax |
| 02:28.09 | Qwell | throat? |
| 02:28.12 | I-MOD | asteriskmonkey: lol |
| 02:28.16 | asteriskmonkey | yes |
| 02:28.17 | asteriskmonkey | lol |
| 02:28.29 | I-MOD | dont ya just love words? |
| 02:28.40 | dgorski | it will explain the less than obvious config format... asterisk configs are weird when compared to other config files (IMHO) |
| 02:29.19 | dgorski | in short... |
| 02:29.37 | dgorski | configure, specify channels, change config, specify more channels |
| 02:29.52 | dgorski | channel specs inherit config previously specified |
| 02:30.32 | I-MOD | basically, once you set something, it stays set for the rest of the file unless you set it to something else |
| 02:30.38 | dgorski | from README: |
| 02:30.39 | dgorski | <PROTECTED> |
| 02:30.39 | dgorski | configuration file read: |
| 02:30.39 | dgorski | <PROTECTED> |
| 02:30.39 | dgorski | <PROTECTED> |
| 02:30.39 | dgorski | <PROTECTED> |
| 02:30.41 | dgorski | <PROTECTED> |
| 02:30.43 | dgorski | <PROTECTED> |
| 02:30.45 | dgorski | Then, the "national" switchtype would be applied to channels one through |
| 02:30.47 | dgorski | four and channels 10 through 12, whereas the "dms100" switchtype would |
| 02:30.49 | dgorski | apply to channels 25 through 47. |
| 02:31.12 | asteriskmonkey | here ill show you my config you can seewhy i am confused |
| 02:31.15 | asteriskmonkey | http://pastebin.ca/44611 |
| 02:32.06 | asteriskmonkey | see i have channels define the whole batch |
| 02:32.27 | dgorski | ok, so break out the ones you want to be different |
| 02:32.28 | asteriskmonkey | im trying to split it in 2 so 2 different contect 2 different echo can properties and tx rx gain |
| 02:32.44 | I-MOD | not hard |
| 02:32.49 | asteriskmonkey | do they still stay under the channels context? |
| 02:33.03 | I-MOD | yes, if you don't change it |
| 02:33.03 | dgorski | if you want |
| 02:33.22 | dgorski | http://pastebin.ca/44612 |
| 02:34.39 | asteriskmonkey | so as soon as you say channel => whatever anything after that is considered config specific to those channels? |
| 02:35.00 | dgorski | before |
| 02:35.04 | I-MOD | man...... |
| 02:35.19 | dgorski | anything BEFORE the channel=> applies to the channel=> statement |
| 02:35.38 | asteriskmonkey | oh ok |
| 02:35.54 | asteriskmonkey | so i have to sepcify all the previous settings including signalling over again |
| 02:36.04 | I-MOD | no......no |
| 02:36.21 | *** join/#asterisk iq (n=iq@71-38-79-126.omah.qwest.net) |
| 02:36.30 | I-MOD | <insert all your current settings here> |
| 02:36.37 | I-MOD | <first channel assignment> |
| 02:36.43 | I-MOD | <more settings> |
| 02:36.52 | I-MOD | <second channel assignment> |
| 02:37.19 | I-MOD | the first group of settings will apply to both channel assignments |
| 02:37.26 | dgorski | override them |
| 02:37.37 | asteriskmonkey | ah ok |
| 02:37.38 | dgorski | anything you specified is good until you respecify it differently |
| 02:37.50 | I-MOD | the second group of settings will apply only to the second and any following channel assignments |
| 02:38.17 | dgorski | or, settings apply only to channel statements FOLLOWING them |
| 02:38.58 | dgorski | ;) |
| 02:39.10 | asteriskmonkey | ok that makes more sense now :) |
| 02:39.11 | asteriskmonkey | thanks |
| 02:39.26 | dgorski | sure. Now, go read docs/README |
| 02:39.29 | dgorski | promise |
| 02:39.30 | dgorski | ? |
| 02:39.30 | asteriskmonkey | confusing have just on context [channels] then splitting it in same context |
| 02:39.32 | asteriskmonkey | :D |
| 02:39.37 | *** join/#asterisk Skumling (n=skumling@fw.sg12.dk) |
| 02:39.43 | asteriskmonkey | sure |
| 02:39.57 | dgorski | thank you! ;) |
| 02:40.05 | asteriskmonkey | thanks |
| 02:40.27 | dgorski | it's not really a "context" in the case of zapata.conf |
| 02:41.44 | asteriskmonkey | thats what was throwing me off |
| 02:42.27 | asteriskmonkey | whole reason i wanted to split the groups is that im having horrid echo issues with some calls |
| 02:42.40 | russellb | someone remind me, do we have any other files that follow the general syntax used in zapata.conf? |
| 02:42.41 | asteriskmonkey | and im using a te411 top of the line digium card |
| 02:42.44 | russellb | i think it's the only one ... |
| 02:42.54 | I-MOD | i think so to |
| 02:42.56 | I-MOD | too |
| 02:43.11 | dgorski | echo... |
| 02:43.12 | russellb | ok, cool. I'm writing docs :-p |
| 02:43.16 | *** join/#asterisk pauldy (n=pauldy@24-155-86-154.ip.grandenetworks.net) |
| 02:43.39 | file | russellb: back to work! |
| 02:43.47 | russellb | file: YOU |
| 02:43.53 | file | russellb: ME |
| 02:44.21 | I-MOD | you two: what's this work you speak of? |
| 02:44.21 | russellb | just so everyone knows, file is super cool |
| 02:44.27 | asteriskmonkey | dont suppose there is a way of making all did;s go to one channel group and a spefic did to a test channel group |
| 02:44.28 | file | awww you're sweet |
| 02:44.40 | russellb | I-MOD: secret projects alpha and Z81..3b |
| 02:44.40 | file | russellb: you're super cool too |
| 02:45.08 | I-MOD | you two: those again? pfffft, thats not work, its FUN!! |
| 02:45.18 | asteriskmonkey | anyone going to von |
| 02:45.19 | file | it is fun |
| 02:45.21 | russellb | yup |
| 02:45.25 | file | asteriskmonkey: Yes |
| 02:45.35 | asteriskmonkey | krams going ? |
| 02:45.37 | russellb | file is trying to get me to go |
| 02:45.39 | file | naturally |
| 02:45.48 | file | don't forget to drop by the Digium/Asterisk pavilion! |
| 02:45.49 | Snake-Eyes | is there a way to have asterisk have array or read from text file? |
| 02:46.06 | file | pick up swag |
| 02:46.07 | russellb | Snake-Eyes: there is a ReadFile application, i think |
| 02:46.14 | russellb | Snake-Eyes: and there is an ARRAY dialplan function |
| 02:46.21 | asteriskmonkey | what gooes will digium have this year? |
| 02:46.30 | file | asteriskmonkey: wait and see! |
| 02:46.38 | asteriskmonkey | ill be there in a booth :P |
| 02:46.39 | I-MOD | mmmmm....gooes |
| 02:46.40 | russellb | pfft, don't go for the schwag, go to find out what cool new stuff is going on :) |
| 02:46.49 | russellb | asteriskmonkey: which one |
| 02:46.53 | asteriskmonkey | williams |
| 02:47.01 | Qwell | what do they do? |
| 02:47.11 | file | I'll swing by and make fun of you :P |
| 02:47.15 | Snake-Eyes | russellb, thanks, i look those two commands up |
| 02:47.21 | asteriskmonkey | lol really you dont know? there the only canadian digium distributor |
| 02:47.23 | Qwell | s#they#you/they# |
| 02:47.46 | Qwell | sounds like an open market :p |
| 02:47.55 | asteriskmonkey | ? |
| 02:48.00 | Qwell | nothing |
| 02:48.01 | russellb | file: you need to put me on speakerphone at the booth |
| 02:48.07 | file | russellb: ok! |
| 02:48.08 | russellb | so I can talk to people :) |
| 02:48.14 | russellb | eyebeam! |
| 02:48.17 | zigman | cebit ? |
| 02:48.20 | Qwell | russellb: There you go |
| 02:48.25 | file | "Talk to Russell Bryant, dial 5!" |
| 02:48.33 | Qwell | leave it running all day |
| 02:48.34 | russellb | lol, that's cool with me :) |
| 02:48.38 | I-MOD | and insert a coin! |
| 02:48.44 | file | my Powerbook will be streaming video live |
| 02:48.51 | russellb | coolio |
| 02:48.58 | Qwell | ooo...instant finding file |
| 02:49.16 | Qwell | http://whereisfile.com/stream/ |
| 02:49.21 | file | :D |
| 02:50.22 | asteriskmonkey | have a big server to stram that too ? :D |
| 02:50.31 | file | two servers |
| 02:50.44 | file | one primary, second relay :) |
| 02:50.51 | asteriskmonkey | sweet? where 151 front? |
| 02:51.00 | wunderkin | 5! 5! |
| 02:51.03 | file | in... San Jose! |
| 02:51.08 | Qwell | file: ack! |
| 02:51.21 | file | at the MAE-West building :D |
| 02:51.27 | asteriskmonkey | thats a fair hop |
| 02:51.32 | Qwell | What does SBC stand for? |
| 02:51.33 | file | so I should have good connectivity |
| 02:51.38 | asteriskmonkey | how can i become a digium mirror? |
| 02:51.43 | Qwell | I kinda assumed Southern Bell Corp, but...I'm obviously wrong, heh |
| 02:51.45 | file | sacrifice a goat! |
| 02:52.14 | asteriskmonkey | russelb: you work @ digium? |
| 02:53.22 | asteriskmonkey | file what streaming soft you use on the servers |
| 02:53.34 | file | Darwin Streaming Server |
| 02:53.58 | asteriskmonkey | what you use to ship it off ? ffmpeg? |
| 02:54.09 | file | QuickTime Broadcaster |
| 02:54.13 | russellb | asteriskmonkey: yes, i do |
| 02:54.18 | russellb | part time, anyway |
| 02:54.22 | russellb | I'm still a full-time student |
| 02:54.35 | asteriskmonkey | ah i want to work for digium :) |
| 02:54.39 | file | and full-time good guy |
| 02:54.52 | Qwell | asteriskmonkey: they hire occasionally... |
| 02:54.54 | russellb | asteriskmonkey: there is an employment page linked off of the digium main page ;) |
| 02:55.02 | asteriskmonkey | ask mark if he wants a toronto svn mirror |
| 02:55.11 | russellb | mark doesn't handle that stuff, heh |
| 02:55.13 | file | http://www.digium.com/index.php?menu=employment meep |
| 02:55.25 | file | lots of positions... okay well not lots but meh |
| 02:55.29 | russellb | asteriskmonkey: you talking about mirroring the ftp or what? |
| 02:55.50 | russellb | even if something there isn't exactly what you want, you can still send in a resume saying what you are interested in |
| 02:56.04 | russellb | it's a growing company that's always expanding ... |
| 02:56.09 | file | "Interested in conquering the world." |
| 02:56.12 | russellb | exactly |
| 02:56.13 | file | "YOU'RE HIRED!" |
| 02:56.17 | russellb | yup. |
| 02:56.27 | asteriskmonkey | yes |
| 02:56.40 | russellb | file: can I send you my resume? |
| 02:56.49 | file | russellb: yes, doesn't mean I'll look at it |
| 02:56.54 | russellb | darn |
| 02:57.01 | russellb | I'll code asterisk for foooood |
| 02:57.05 | asteriskmonkey | lol |
| 02:57.09 | file | I'd only accept it if you gave it to me at VON :P |
| 02:57.11 | Qwell | russellb: send it this way - I'll get you a job :p |
| 02:57.17 | russellb | hehe :) |
| 02:57.21 | asteriskmonkey | ill burn sangoma cards to kep warm |
| 02:57.26 | russellb | lol |
| 02:57.44 | russellb | now you're just sucking up :-p |
| 02:58.52 | asteriskmonkey | gah i wish asterisk had a sliding taps feature |
| 02:59.02 | Qwell | write one |
| 02:59.10 | SarahEmm | sliding taps? |
| 02:59.20 | asteriskmonkey | taps |
| 02:59.21 | Qwell | file: Just wait. I'm gonna get you. ;) |
| 02:59.22 | asteriskmonkey | echo taps |
| 02:59.27 | SarahEmm | ahhh |
| 02:59.35 | file | Qwell: I'll stuff you in a garbage can :P |
| 02:59.41 | asteriskmonkey | 32 64 128 256 |
| 02:59.44 | Qwell | I so won't fit |
| 02:59.55 | asteriskmonkey | long distance calls get mad echo if not set properly |
| 03:00.03 | file | Qwell: we can try! |
| 03:00.10 | Qwell | I would so fit :p |
| 03:00.19 | file | I'll do it!!! |
| 03:00.22 | dgorski | it's the far end... check your gains |
| 03:00.31 | dgorski | god that was cryptic |
| 03:00.36 | asteriskmonkey | gosh darn it my gains are like 100% perfect |
| 03:00.47 | asteriskmonkey | even attenuated them to the test line number from telco |
| 03:00.52 | asteriskmonkey | and i stil get echo |
| 03:00.53 | asteriskmonkey | grrr |
| 03:00.58 | Qwell | speaking of VON...anybody seen tuxinator_linux? |
| 03:01.03 | dgorski | i feel your pain |
| 03:01.21 | file | Qwell: not lately |
| 03:01.42 | Qwell | need to work out ride details with him sometime soon, heh |
| 03:01.42 | asteriskmonkey | went from a te110p to a 411 and no difference |
| 03:01.47 | asteriskmonkey | grr |
| 03:02.12 | *** join/#asterisk Mw3_ (n=mw3@ip5993414e.bp-1031.rubicom.hu) |
| 03:02.35 | asteriskmonkey | gots to be a dirty pri |
| 03:03.11 | I-MOD | how much have you tweaked the echocan stuff? |
| 03:03.26 | dgorski | I don't think there is such a thing as a dirty pri - it's digital |
| 03:03.41 | dgorski | sorry... voice channels are digital |
| 03:03.43 | asteriskmonkey | nor do i |
| 03:03.46 | asteriskmonkey | ok then |
| 03:03.49 | asteriskmonkey | here is the kicker |
| 03:04.05 | asteriskmonkey | my rx and tx are at -8 |
| 03:04.10 | I-MOD | woah |
| 03:04.18 | I-MOD | and you can still hear people? |
| 03:04.21 | dgorski | woa |
| 03:04.21 | asteriskmonkey | yes |
| 03:04.27 | asteriskmonkey | its at a colo |
| 03:04.31 | I-MOD | how is that possible? |
| 03:04.37 | dgorski | does DTMF work? |
| 03:04.39 | asteriskmonkey | yes |
| 03:04.43 | dgorski | wow |
| 03:04.58 | asteriskmonkey | yes |
| 03:05.01 | dgorski | what does your line build out look like? |
| 03:05.02 | *** join/#asterisk veto_laptop (i=mdkuser@cpe-66-69-38-192.satx.res.rr.com) |
| 03:05.04 | I-MOD | something has got to be bunk on that system |
| 03:05.10 | asteriskmonkey | less than 100ft |
| 03:05.16 | dgorski | <PROTECTED> |
| 03:05.53 | asteriskmonkey | span=1,1,0,esf,b8zs |
| 03:06.42 | asteriskmonkey | thats my line build out |
| 03:06.52 | asteriskmonkey | ? why speechless? |
| 03:06.59 | I-MOD | not cause of the lbo |
| 03:07.05 | dgorski | how about the provider's equipment... are they assuming you're miles and miles away? |
| 03:07.13 | dgorski | -8 is crazy |
| 03:07.24 | asteriskmonkey | no i called in a test |
| 03:07.34 | asteriskmonkey | there only 1 floor down |
| 03:07.39 | asteriskmonkey | they said they get no echo |
| 03:07.55 | dgorski | also, I'm surprised they are both the same (rx/tx at -8) |
| 03:07.55 | asteriskmonkey | im getting mad echo at anything higher in the gains |
| 03:08.05 | asteriskmonkey | shit ive had them as low as -16 |
| 03:08.23 | I-MOD | it only changes down to -9 i think |
| 03:08.42 | asteriskmonkey | no ive had -40 when testing |
| 03:09.39 | asteriskmonkey | any ideas or tests i could do to possible help get this tuned better |
| 03:09.48 | asteriskmonkey | ive used the ztmonitor to the max at this point |
| 03:09.56 | Qwell | how many multiples are you going at once? |
| 03:10.04 | asteriskmonkey | 1's |
| 03:10.05 | Qwell | 1 is a lot |
| 03:10.08 | Qwell | try like .5.. |
| 03:10.26 | asteriskmonkey | its tuned using ztmonitor -vv |
| 03:10.57 | asteriskmonkey | ah wait will zt monitor show anything over the max 14555 |
| 03:15.57 | dgorski | <PROTECTED> |
| 03:15.57 | dgorski | <PROTECTED> |
| 03:16.04 | dgorski | from chan_zap.c |
| 03:16.22 | dgorski | k for gain |
| 03:16.25 | asteriskmonkey | ok thats screwy check this out |
| 03:16.54 | asteriskmonkey | my tx gets some activity on it when the rx (the tone comming in from test number) is pulsed |
| 03:17.00 | dgorski | so -40 is reasonable (I guess) seems a little out there though - I would say telco issue |
| 03:17.06 | dgorski | because that's easier... ;) |
| 03:17.34 | dgorski | but it's gain, so bigger numbers make little sense... I don't know seems fishy |
| 03:18.43 | asteriskmonkey | i set my gain in zapatel.conf should i bee setting it in chan_zap.c? |
| 03:18.52 | asteriskmonkey | here is my results when i dial the test tone line Rx: 5817 ( 5837) Tx: 4137 ( 5604) |
| 03:22.06 | russellb | i didn't know there was a zapatel.conf! |
| 03:22.59 | Qwell | russellb: it's to config zapateller |
| 03:23.05 | Qwell | duh, sheesh. :) |
| 03:23.09 | russellb | d'oh |
| 03:23.57 | asteriskmonkey | ok so why is my rx spilling into the tx |
| 03:24.53 | *** join/#asterisk Djeli (n=djeli@ppp157-243.static.internode.on.net) |
| 03:26.30 | octothorpe_ | there's weasels in your phone system, crossing the signals |
| 03:28.59 | asteriskmonkey | so 14500 is supposed to be the sweet spot right? |
| 03:29.53 | dgorski | ha ha |
| 03:30.11 | dgorski | gains are set in zapata.conf, not zaptel.conf... |
| 03:30.44 | octothorpe_ | that's how it is in my system |
| 03:30.53 | asteriskmonkey | yes soryr i misspelled |
| 03:31.11 | dgorski | zapata.conf configures chan_zap.c |
| 03:31.17 | asteriskmonkey | why is my tx getting audio |
| 03:31.23 | asteriskmonkey | im not saying anything |
| 03:31.31 | asteriskmonkey | i even uplug the handset |
| 03:32.02 | dgorski | what kind of handset, analog, SIP ?? |
| 03:32.06 | asteriskmonkey | yes sup |
| 03:32.10 | asteriskmonkey | sip |
| 03:33.03 | dgorski | what kind of phone, grandstream? |
| 03:33.13 | dgorski | I had all kinds of problems with those |
| 03:33.35 | asteriskmonkey | no an atcom |
| 03:33.42 | asteriskmonkey | ive got an iaxy around here |
| 03:33.49 | dgorski | not familiar |
| 03:33.52 | asteriskmonkey | its ok |
| 03:33.57 | asteriskmonkey | its not a problem with the phone anyhow |
| 03:34.12 | asteriskmonkey | any ideas? |
| 03:35.26 | dgorski | nope |
| 03:35.30 | asteriskmonkey | like here check this out Rx: 11589 (11607) Tx: 10495 (10495) |
| 03:35.57 | asteriskmonkey | thats with the handset unplugged |
| 03:36.02 | asteriskmonkey | funny thing is |
| 03:36.05 | asteriskmonkey | its starts ok |
| 03:36.20 | asteriskmonkey | like tx 0 then bumps around a 1000 then starts to spill like mad |
| 03:36.43 | asteriskmonkey | russelb: you have any idea? |
| 03:36.56 | dgorski | are you sure it's the right channel? |
| 03:37.02 | asteriskmonkey | yes |
| 03:37.08 | asteriskmonkey | i do a show channels first |
| 03:38.19 | asteriskmonkey | its running on an amd 2800 sempron with 512 megs or ram and a te411 |
| 03:39.47 | dgorski | so, you did a show channel to ensure that your SIP/xxxx is bridged to ZAP/x and then you ztmonitor x |
| 03:39.49 | dgorski | ? |
| 03:40.03 | asteriskmonkey | yes |
| 03:40.46 | dgorski | an you see tx traffic? that's crazy... |
| 03:40.59 | asteriskmonkey | yes |
| 03:41.26 | asteriskmonkey | so could it be the provider |
| 03:41.33 | asteriskmonkey | like i thought pri's where flawless |
| 03:41.42 | dgorski | are you using a sheilded cable? sounds like maybe you are getting crosstalk |
| 03:41.58 | dgorski | that makes no sense, nevermind |
| 03:44.02 | asteriskmonkey | iv been banging my head on this for 8months now and i even spurged on the uber digium card |
| 03:44.05 | asteriskmonkey | so.. |
| 03:44.13 | asteriskmonkey | anything else i could test. |
| 03:44.31 | asteriskmonkey | (the only thing i havnt tried is changing line build out .. but why i thinks).. |
| 03:45.06 | dgorski | you could have your provide put a t1 line monitor on the line - that may help see what's happeneing. They have devices that will allow you to listen to the channels... |
| 03:45.42 | asteriskmonkey | they said they tested my pri and got no echo |
| 03:46.01 | dgorski | whatever that means |
| 03:46.48 | asteriskmonkey | i yelled at mci told em to fix my damn pri and they said it was all working no echo when they tested there end |
| 03:46.52 | *** join/#asterisk Vyeperman (n=Vye@ip68-8-174-154.sd.sd.cox.net) |
| 03:46.55 | asteriskmonkey | i know condfusing |
| 03:47.08 | *** join/#asterisk mtgh (n=chatzill@dsl093-001-038.det1.dsl.speakeasy.net) |
| 03:49.17 | *** join/#asterisk joelsolanki (i=joelsola@202.160.161.93) |
| 03:49.49 | asteriskmonkey | brb |
| 03:51.50 | *** join/#asterisk bmg505 (n=leon@dsl-165-158-31.telkomadsl.co.za) |
| 03:53.10 | Husk | asterisk is using 99.9% of the cpu yet it doesnt appear to be doing anything, any way of checking whats actually going on? |
| 03:53.41 | JunK-Y | gdb |
| 03:56.12 | tengulre | anybody have english resume template, I need. |
| 03:57.36 | *** join/#asterisk dseeb_ (n=dcb@CPE-144-131-35-116.vic.bigpond.net.au) |
| 03:59.22 | *** join/#asterisk asteriskmonkey (n=phil@69.158.149.183) |
| 03:59.23 | SarahEmm | templated resumes look like... templated resumes.... |
| 03:59.27 | asteriskmonkey | ok back |
| 03:59.50 | MikeJ[Laptop] | ok front |
| 03:59.57 | Qwell | ok left side |
| 04:00.05 | MikeJ[Laptop] | FLIP! |
| 04:00.13 | MikeJ[Laptop] | JunK-Y!!! |
| 04:00.17 | I-MOD | flip side? |
| 04:00.37 | MikeJ[Laptop] | yes |
| 04:01.21 | I-MOD | asteriskmonkey: any way you could borrow a channel bank from someone and test the lines out? |
| 04:01.35 | asteriskmonkey | I-MOD: i wish |
| 04:01.43 | I-MOD | have you tried other spans on the card? |
| 04:01.54 | asteriskmonkey | ive tried 2 different cards |
| 04:02.17 | I-MOD | but have you tried multiple spans on the te411p? |
| 04:02.28 | asteriskmonkey | no |
| 04:02.35 | asteriskmonkey | ive tried a single span te110p tough |
| 04:02.54 | I-MOD | you have a really odd problem there |
| 04:02.59 | asteriskmonkey | yes i know |
| 04:03.06 | asteriskmonkey | her is the funny thing |
| 04:03.21 | asteriskmonkey | echo can happen on either side of the call |
| 04:03.39 | asteriskmonkey | so if someone calls me i echo but if i call them they echo |
| 04:03.42 | asteriskmonkey | wierd eh/ |
| 04:03.58 | I-MOD | get the telco to come back out and make them let you listen |
| 04:04.15 | I-MOD | yeah, that's about the weirdest i've ever seen |
| 04:04.34 | asteriskmonkey | yes just worried if i get them to the cage they will be like its cause your using pc stuff not a telco gear |
| 04:05.14 | I-MOD | hide the pc under a sign that says "move along, nothing to see here" |
| 04:05.16 | I-MOD | :) |
| 04:07.01 | asteriskmonkey | lol |
| 04:09.57 | JunK-Y | mikeJ |
| 04:10.04 | JunK-Y | im with my new bomb now :) |
| 04:10.26 | JunK-Y | whats up? |
| 04:16.09 | *** join/#asterisk veto_laptop (i=mdkuser@cpe-66-69-38-192.satx.res.rr.com) |
| 04:16.53 | Pegger | has anyone set up billing system with asterisk? |
| 04:17.00 | [av]bani | http://www.nerdnetworks.org/~bistromath/photos/misc/DeLorean2.JPG |
| 04:18.24 | *** join/#asterisk mog_home (n=mogorman@68.62.237.103) |
| 04:20.49 | asteriskmonkey | i wrote my own |
| 04:21.00 | *** join/#asterisk Nukemizer (n=Nuke@160.7.249.15) |
| 04:24.01 | *** join/#asterisk droops (n=droops@adsl-065-005-212-128.sip.jan.bellsouth.net) |
| 04:34.40 | *** join/#asterisk Rocket- (n=Rocket@c-67-165-17-97.hsd1.ct.comcast.net) |
| 04:35.08 | Pegger | asteriskmonkey, can i ask you some questions about it |
| 04:38.46 | *** join/#asterisk nayyares (n=nayyar@58.65.151.218) |
| 04:39.09 | nayyares | hi guys! |
| 04:39.17 | Qwell | [av]bani: useless information - that picture was taken on the 405 fwy, where it meets the 55 fwy |
| 04:40.53 | kuku5 | Qwell: How come my sccp fw 7940 doesnt want to connect to asterisk |
| 04:41.16 | Pegger | nayyares, yo, I am trying to figure out the best way to set up a billing system |
| 04:41.18 | Qwell | kuku5: because you're doing something wrong. Until you give more information, that's about all I can say.. |
| 04:41.26 | russellb | nayyares: WELCOEM!!!!!!! |
| 04:41.30 | russellb | WELCOME, even |
| 04:41.31 | russellb | :( |
| 04:41.35 | Qwell | weclome |
| 04:41.38 | kuku5 | It asks for the tones and font file, then it grabs the cnf.xml file and over again |
| 04:41.51 | Qwell | kuku5: yeah, it'll skip those two. That's normaly |
| 04:41.52 | Qwell | -y |
| 04:41.53 | kuku5 | Sorry, after tones it asks for the tlv |
| 04:42.06 | Qwell | touch SEPblah.tlv |
| 04:42.07 | Qwell | reboot |
| 04:42.09 | Qwell | rm SEPblah.tlv |
| 04:42.21 | kuku5 | no, CTLSEPxxx.tlv |
| 04:42.24 | Qwell | whatever |
| 04:42.42 | Qwell | create it, let it read it, then delete it |
| 04:42.58 | nayyares | russellb, Qwell thanks buddy....! |
| 04:44.46 | Qwell | russellb: mind a quick msg? |
| 04:44.58 | russellb | Qwell: sure |
| 04:45.18 | *** join/#asterisk Ad-Hoc (n=Nimbus@ppp77-adsl-111.ath.forthnet.gr) |
| 04:45.32 | nayyares | is there any reference manual to Asterisk modules, that can be used to hack its code? |
| 04:45.42 | SarahEmm | *giggles* |
| 04:45.43 | SarahEmm | manual? ;) |
| 04:45.46 | kuku5 | Qwell: now its going in circles again, still asking for the tlv fonts, tones |
| 04:46.08 | kuku5 | IT briefly shows "opening 192.168.2.90" ( ip of the pbx ) |
| 04:46.21 | kuku5 | Qwell: sccp debug doesnt show anythign |
| 04:46.22 | Qwell | oh, it gets past the tlv part? |
| 04:46.49 | kuku5 | "past" ? I dont think so, it asks for the SEPxxx.cnf.xml file again |
| 04:47.16 | nayyares | SarahEmm, yeah, any document can help me to understand Asterisk low level issues? |
| 04:47.19 | Pegger | kuku5, you trying to set up a cisco phone? |
| 04:47.47 | russellb | nayyares: http://www.asterisk.org/doxygen/ |
| 04:47.49 | kuku5 | Qwell: look at the log http://pastebin.ca/44621 |
| 04:47.56 | SarahEmm | nayyares: the voip-info.org wiki is about the best i knew of last time i did devel stuff.. |
| 04:47.57 | kuku5 | Pegger: yes - sccp image |
| 04:47.59 | Qwell | kuku5: can't right now... |
| 04:48.04 | Qwell | kuku5: latest sccp firmware? |
| 04:48.05 | SarahEmm | ooh, that's new. nifty |
| 04:48.07 | kuku5 | yes |
| 04:48.12 | russellb | nayyares: and http://www.asterisk.org/developers/ |
| 04:48.19 | Pegger | kuku5, so are cisco phones that hard to set up just to use |
| 04:48.26 | kuku5 | Qwell: 0307020400 |
| 04:48.42 | kuku5 | Pegger ? |
| 04:48.45 | Qwell | close enough |
| 04:48.53 | Qwell | Pegger: They're usually easy |
| 04:49.14 | kuku5 | Pegger: you have to learn how to dance with them - after that is a piece of cake |
| 04:49.59 | nayyares | russellb, SarahEmm thanks, let me give it a try...:) |
| 04:57.29 | JunK-Y | Day changed to 06 Mar 2006 <-- just a way to say: hey goto bed, u work TODAY! |
| 04:57.38 | kuku5 | Qwell: any pointers? |
| 04:57.59 | Qwell | kuku5: just a pointer to a null array |
| 04:58.52 | kuku5 | yeh |
| 04:59.06 | kuku5 | Office -- SEP0016C77CD45C None |
| 04:59.15 | kuku5 | This is what i have for sccp show devices |
| 04:59.35 | kuku5 | Weird thing is if I take out theSEP , it doenst show up |
| 05:00.05 | Qwell | yeah, the SEP is needed |
| 05:00.15 | Qwell | the phone isn't hitting it. You need to figure out why. |
| 05:02.36 | kuku5 | Qwell: The phone is fetching SEP0016C77CD45C.cnf.xmlk |
| 05:02.46 | Qwell | yeah, that's right |
| 05:03.04 | kuku5 | <PROTECTED> |
| 05:03.26 | kuku5 | it should touch the server |
| 05:03.41 | Qwell | no nat/firewall issues in the way? |
| 05:03.44 | kuku5 | nope |
| 05:03.54 | kuku5 | same network |
| 05:04.15 | kuku5 | different switch... HM |
| 05:05.10 | kuku5 | The phone shows going to .90 for like half a second, then goes back to 192.168.2.1 ( which is the router ) |
| 05:05.12 | Qwell | is the asterisk box also the tftpd? |
| 05:05.27 | kuku5 | asterisk box is not tftpd 192.168.2.2 is |
| 05:05.36 | kuku5 | I dont know why the phone tries to go to the router |
| 05:05.41 | Qwell | ah hah, so nat/firewall issue is possible |
| 05:05.49 | Qwell | because it goes to the dhcp server |
| 05:05.54 | Qwell | which, in your case, is also the router |
| 05:06.02 | kuku5 | except the dhcp server is 192.168.2.2 |
| 05:06.22 | Qwell | oh, maybe it always does the .1 |
| 05:06.26 | kuku5 | router isnt running dhccp |
| 05:06.45 | Qwell | see if your * box can ping the phone? |
| 05:06.54 | kuku5 | But dhcpd log shows the ip being taken |
| 05:06.56 | Qwell | and check that the right port (2000, tcp) is open |
| 05:07.01 | Qwell | on the * box |
| 05:07.34 | kuku5 | it pings |
| 05:07.48 | kuku5 | 2000 isnt open |
| 05:07.52 | kuku5 | duh |
| 05:08.09 | kuku5 | Where do I set it to open the port - I thought the module did that ... |
| 05:08.15 | Qwell | your firewall |
| 05:08.17 | kuku5 | ( no fw ) |
| 05:08.25 | Qwell | where are you seeing 2000 isn't open? |
| 05:08.28 | Qwell | Can you nmap it? |
| 05:08.29 | kuku5 | netstat -a |
| 05:08.39 | Qwell | oh, look for sccp or something instead |
| 05:08.43 | Qwell | maybe cisco |
| 05:08.49 | Qwell | grep 2000 /etc/services |
| 05:08.54 | Qwell | whatever that says, it'll say in netstat |
| 05:09.18 | Qwell | You can disable naming ports in netstat with some switch too...I forget which one though |
| 05:09.23 | rabelais | -n |
| 05:09.25 | kuku5 | 2000 didnt find anything |
| 05:09.30 | Qwell | -n |
| 05:09.32 | Qwell | yeah |
| 05:09.38 | Qwell | netstat -n | grep 2000 |
| 05:09.44 | kuku5 | 2000 isnt open |
| 05:09.57 | dgorski | netstat -an | grep 2000 |
| 05:10.03 | rabelais | but you gotta use -a |
| 05:10.08 | Qwell | -a would help, yeah :p |
| 05:10.11 | rabelais | otherwise it won't list listening ports |
| 05:10.23 | Qwell | yeah..I had an established connection, so it showed up. my bad |
| 05:10.54 | kuku5 | Which services runs the 2000 |
| 05:10.55 | rabelais | to be super anal: netstat -an |grep LISTEN | grep 2000 ; P |
| 05:11.08 | asteriskmonkey | whats it called when the rx leaks in to the tx and vise versa |
| 05:11.10 | dgorski | grep -i listen |
| 05:11.10 | Qwell | kuku5: sccp |
| 05:11.16 | dgorski | to be super anal |
| 05:11.41 | dgorski | you may wish to verify you are loading the module in asterisk |
| 05:11.44 | dgorski | show modules |
| 05:12.02 | Qwell | it's loaded |
| 05:12.28 | kuku5 | chan_sccp.so Skinny Client Control Protocol (SCCP). R 0 |
| 05:12.43 | Qwell | does netstat -an show it? |
| 05:12.46 | kuku5 | nope |
| 05:12.55 | Qwell | if the module is loaded, there is no reason for it not to have |
| 05:13.15 | kuku5 | wait... |
| 05:13.16 | kuku5 | 1 sec |
| 05:13.26 | kuku5 | im a dumb fuck! |
| 05:13.36 | Qwell | agreed |
| 05:14.04 | kuku5 | Now I know why it was going to .1 |
| 05:14.07 | Qwell | erm, wrong channel |
| 05:14.16 | kuku5 | it wanted to get routed to 1.90 |
| 05:14.21 | kuku5 | but the ip of the pbx is 2.90 |
| 05:14.29 | kuku5 | it booted right off the bat |
| 05:15.17 | kuku5 | Wow - sccp is nice. |
| 05:15.20 | Qwell | indeed |
| 05:15.32 | kuku5 | No call forward button |
| 05:15.45 | Qwell | kuku5: pick up the handset |
| 05:15.48 | Qwell | then do it |
| 05:16.22 | kuku5 | ? - but there is no call forward - so the phone rings somewhere else |
| 05:16.38 | Qwell | yeah, cfwd appears when you pick up |
| 05:16.52 | Qwell | assuming it's enabled in the config.. |
| 05:16.54 | kuku5 | it doenst on mine |
| 05:16.58 | kuku5 | ah |
| 05:17.10 | Qwell | cfwdall=yes, or something |
| 05:17.17 | Qwell | cfwdall and cfwdbusy are valid, iirc |
| 05:17.20 | kuku5 | i jsut fell in love |
| 05:17.51 | Qwell | since jbot sucks... see below |
| 05:17.52 | iq | anyone doing CTI with asterisk? |
| 05:17.54 | Qwell | <Qwell> !wtf thanks? |
| 05:17.54 | Qwell | <LinuxHOW2> Just send money...... |
| 05:17.58 | Qwell | :D |
| 05:18.16 | SarahEmm | iq: uhhh... that's what asterisk is... |
| 05:18.23 | SarahEmm | it integrates computers with telephony :) |
| 05:18.26 | kuku5 | Thanks qwell :) |
| 05:18.37 | iq | SarahEmm, sure? |
| 05:18.41 | SarahEmm | err |
| 05:18.43 | Qwell | kuku5: np |
| 05:18.47 | SarahEmm | what's your definition of cti? |
| 05:18.50 | Qwell | I like sccp, so I try to spread it :p |
| 05:18.58 | trixter | spread it like VD |
| 05:19.07 | SarahEmm | well, one polycom |
| 05:19.08 | dgorski | iq: see AGI |
| 05:19.11 | kuku5 | Qwell: I dunno wtf I had to spend 2 days to learn to convert the phone to sip 6 months ago |
| 05:19.29 | kuku5 | And now learn this dance - with the phone. |
| 05:19.35 | iq | SarahEmm, I meant TSAPI, JTAPI, CSTA, etc... implementation |
| 05:19.43 | iq | dgorski, I'll look at AGI |
| 05:19.48 | dgorski | iq: I started a JTAPI... |
| 05:19.51 | iq | dgorski, its a scripting language, right? |
| 05:19.54 | dgorski | call model is different |
| 05:19.54 | kuku5 | Qwell: any infor on the config setup of these phone s? |
| 05:19.56 | SarahEmm | look at AGI, yeah... |
| 05:20.06 | Qwell | kuku5: it's all pretty self explanitory, really |
| 05:20.08 | SarahEmm | what's so good about sccp Qwell? |
| 05:20.10 | dgorski | AGI, no it's like CGI - any language you choose |
| 05:20.20 | Qwell | SarahEmm: it's one of those things, where you'd have to try it |
| 05:20.28 | kuku5 | Qwell: but like better phone directory or somethign |
| 05:20.50 | iq | dgorski, u writing ur own implementation for JTAPI? |
| 05:21.13 | Qwell | SarahEmm: once you see the code, you'd probably get it. TONS of potential for doing fun stuff |
| 05:21.14 | SarahEmm | okay qwell. |
| 05:21.18 | SarahEmm | i can't, so i'll stick with sip :) |
| 05:21.21 | SarahEmm | i don't have cisco gear |
| 05:21.43 | Qwell | with SIP, you're basically stuck to what the device can do |
| 05:21.47 | docelmo | Qwell, you said it dude |
| 05:22.07 | dgorski | I was, it stalled, as I said, call model is different and it's hard to match bridged calls in time |
| 05:22.07 | Qwell | with sccp, even though it IS a proprietary protocol...it's quite open for hacking on |
| 05:22.18 | JunK-Y | dont trust qwell, hes a bitch. |
| 05:22.22 | dgorski | I stopped active delevelopment a while ago |
| 05:22.23 | Qwell | well, I never! |
| 05:22.24 | kuku5 | :) |
| 05:22.28 | docelmo | SarahEmm, stay away from H323 & asterisk as its a bit unstable w/ a load |
| 05:22.34 | SarahEmm | cool Qwell :) |
| 05:22.37 | SarahEmm | docelmo: i use SIP. |
| 05:22.42 | dgorski | it's probably better now, have been thinking about picking it up |
| 05:22.54 | SarahEmm | docelmo: you can only get the polycoms in h323 from specific vendors as part of a system, evidently. otherwise just sip or mgcp |
| 05:22.54 | Qwell | but...everybody who ever uses sccp... |
| 05:22.56 | SarahEmm | mine's SIP |
| 05:23.04 | Qwell | at first ([av]bani), they knock it...a lot |
| 05:23.12 | Qwell | then, when they finally use it...it's like..."oh, wow!" |
| 05:23.13 | dgorski | iq: look at the manager interface. You can get telephony events - just not as formal as something like JTAPI |
| 05:23.16 | kuku5 | Qwell: I have 1 thing to say - since sccp is the main firwmare, I bet I can do more with cisco phones and sccp than sip |
| 05:23.25 | Qwell | kuku5: for sure |
| 05:23.27 | SarahEmm | i bet you can kuku5 |
| 05:23.37 | kuku5 | Where do I start reading? |
| 05:23.41 | SarahEmm | they're a lot more $$ than polycoms tho heh |
| 05:23.43 | dgorski | iq: coming from an Avaya world? |
| 05:23.50 | iq | dgorski, yup |
| 05:24.15 | Qwell | not by too much |
| 05:24.18 | docelmo | Hay Qwell what do you think about a chat @ astericon on Asterisk in the Carrier envorionment? |
| 05:24.19 | Qwell | what's a 601, like $220? |
| 05:24.21 | dgorski | tsapi/csta - I wrote some neat stuff foor the CT stuff... |
| 05:24.29 | dgorski | in jtapi of course |
| 05:24.32 | iq | dgorski, manager interface is a gui, right? |
| 05:24.37 | dgorski | no |
| 05:24.37 | Qwell | docelmo: I doubt I'll be going to astricon anymore |
| 05:24.38 | iq | dgorski, currently I'm using TSAPI |
| 05:24.44 | kuku5 | Qwell: How amny amps is your power supply - the 7940/60 g need .2, voipsupply sent me 50 power supplies with .38A |
| 05:24.47 | docelmo | Qwell why? |
| 05:24.52 | Qwell | kuku5: dunno |
| 05:24.55 | Qwell | docelmo: reasons... |
| 05:25.03 | docelmo | Qwell horrible.. |
| 05:25.05 | SarahEmm | Qwell: more than that i think. |
| 05:25.12 | asteriskmonkey | does anyone else ztmonitor spill into tx or rx when running milliwatt / co test number |
| 05:25.13 | docelmo | Qwell now I know who you are well shit this sucks |
| 05:25.13 | iq | is this chat logged ;) ??? |
| 05:25.20 | docelmo | iq yes |
| 05:25.21 | Qwell | SarahEmm: well, a 7940 can be had for about $250, iirc |
| 05:25.27 | SarahEmm | SoundPoint IP 601 (NA PSU and PoE)|2200-11631-001|$399 |
| 05:25.32 | Qwell | what?! |
| 05:25.32 | iq | oh... so will have to be careful :) |
| 05:25.36 | kuku5 | Qwell: 240 |
| 05:25.40 | SarahEmm | that's according to voip-info.org tho |
| 05:25.40 | Qwell | That must be CAD |
| 05:25.42 | docelmo | iq about that? |
| 05:25.44 | dgorski | iq: http://www.voip-info.org/wiki-Asterisk+manager+experience |
| 05:25.44 | SarahEmm | i have no idea |
| 05:25.46 | asteriskmonkey | dude Aastra 480i for the win |
| 05:25.49 | Qwell | weird... |
| 05:25.49 | kuku5 | Qwell: 240$ - out of warranty |
| 05:25.59 | Qwell | That's WAY expensive. You can get a 7970 for that price, just about |
| 05:26.00 | dgorski | there's more out there... |
| 05:26.09 | Qwell | kuku5: sounds about right |
| 05:26.10 | SarahEmm | Qwell: i dunno. i got my 501 for.. $250ish CAD i think? |
| 05:26.13 | dgorski | you really just have to play with it a bit to get the feel |
| 05:26.20 | Qwell | SarahEmm: sounds about right. That's like $200USD? |
| 05:26.21 | kuku5 | Qwell: where do you buy your 7970 ? |
| 05:26.26 | Qwell | kuku5: I don't. My boss does |
| 05:26.28 | SarahEmm | i got it over the cisco mainly because it was easy to get, i couldn't find a local dealer of ciscos |
| 05:26.32 | SarahEmm | Qwell: yeah, about that |
| 05:26.38 | kuku5 | Qwell: can you let me know ? |
| 05:26.44 | Qwell | kuku5: ebay |
| 05:26.47 | iq | dgorski, thats good... thanks :) |
| 05:26.54 | Qwell | You can get one on ebay for $450, easily |
| 05:27.06 | SarahEmm | Qwell: i might pick up a cisco at some point, but it's not a priority as the phones are basically just to tinker with and develop with |
| 05:27.21 | Qwell | SarahEmm: you'd have fun with xml service, I bet |
| 05:27.28 | Qwell | imagine...tty on your cisco :) |
| 05:27.31 | SarahEmm | hehe |
| 05:27.34 | SarahEmm | yeah ,but no keyboard... |
| 05:27.40 | SarahEmm | but yes, i'm sure i'd have fun |
| 05:27.40 | Qwell | well, no, but.. |
| 05:27.48 | SarahEmm | being able to get network status of various things and such right from the phone would rock |
| 05:27.52 | SarahEmm | the 601 has that too, but not the 501 |
| 05:27.52 | Qwell | yep |
| 05:27.54 | *** join/#asterisk scardinal (n=supreme@port816.ds1-suoe.adsl.cybercity.dk) |
| 05:27.57 | Qwell | it does rock |
| 05:28.06 | SarahEmm | it has a button for it, but nothing behind it. button just doesn't do anything. (??? seems odd..) |
| 05:28.15 | Qwell | heh |
| 05:28.25 | Qwell | isn't the 501 supposedly gonna get the minibrowser in a new firmware? |
| 05:28.27 | Qwell | That was the rumor |
| 05:28.29 | SarahEmm | *shrug* i dunno |
| 05:28.32 | SarahEmm | it'd rock if it did |
| 05:28.44 | Qwell | after this long, and now that the 601 is out...doubtful |
| 05:28.49 | SarahEmm | yeah... |
| 05:28.53 | Qwell | unfortunate, but...that's business |
| 05:28.57 | SarahEmm | the 601 is cool, but was too expensive for me at the time |
| 05:29.02 | SarahEmm | i find it hard to justify spending a lot on this stuff heh |
| 05:29.21 | Qwell | It's hard not to ;) |
| 05:29.40 | SarahEmm | hehee |
| 05:29.41 | SarahEmm | *nods* |
| 05:29.49 | iq | dgorski, would you recommend any sip gateway that can do cpa without use of tdm? |
| 05:30.12 | docelmo | cpa? |
| 05:30.22 | dgorski | cpa? |
| 05:30.26 | docelmo | hehe |
| 05:30.31 | iq | call progress analysis |
| 05:30.43 | iq | busy, answering machine, ring no answer, fax, etc... |
| 05:30.47 | trixter | SarahEmm: I think cisco tty stuff translates it to a byte data stream and the other end point will convert it back to tones |
| 05:30.54 | dgorski | ahh |
| 05:31.12 | trixter | which alievates jitter and other stuff... what I heard is that cisco on some models do support tty apps and in a sense better than doing it inband |
| 05:31.14 | dgorski | no I can't say that I would be able to answer that |
| 05:31.15 | docelmo | iq just use asterisk.. doesnt do answeing machine but works |
| 05:31.26 | docelmo | you could code it in |
| 05:31.30 | trixter | asterisk does answering machines |
| 05:31.30 | Qwell | trixter: nah, do it via xml services |
| 05:31.30 | dgorski | he meanst to detect at the far end... right? |
| 05:31.33 | trixter | if you want it to anyway |
| 05:31.37 | iq | docelmo, but it require tdm to do all that, right? |
| 05:31.41 | docelmo | nope |
| 05:31.44 | iq | dgorski, yes |
| 05:31.49 | iq | for out bound calls |
| 05:31.53 | docelmo | asterisk gives back like 6 dispositions |
| 05:31.59 | trixter | yeah asterisk detects answering machines that way |
| 05:32.05 | dgorski | asterisk can detect fax, and you can tell when you don't get answered... |
| 05:32.06 | dgorski | don |
| 05:32.08 | trixter | if you want it to anyway |
| 05:32.09 | dgorski | don |
| 05:32.12 | dgorski | crap |
| 05:32.25 | dgorski | don't know about answering machine detection... |
| 05:32.35 | SarahEmm | trixter: that's good.. but i can't talk from a phone, as i have no keyboard :) |
| 05:32.35 | iq | dgorski, using tdm with asterisk? |
| 05:32.37 | docelmo | doesnt do answering machine.. |
| 05:32.52 | dgorski | and busy |
| 05:32.58 | trixter | Qwell: well the thing with the cisco tty stuff was so that a pstn call (which cant easily do xml services) can be answered, then transmitted voip to maybe another pstn gateway where it must again be tones |
| 05:33.09 | docelmo | iq Im probably running the largest asterisk installation in this room.. it works just fine for dispositions |
| 05:33.10 | dgorski | yes, you can, but sip et. al all have call progress |
| 05:33.28 | dgorski | so you'll get busy indications, ringno, etc... and can detect fax as well |
| 05:33.32 | trixter | and it makes sense to do it that way, net traversal as data instead of audio encoded into data |
| 05:33.41 | trixter | SarahEmm: well you should get that fixed! :P |
| 05:33.49 | iq | i c |
| 05:33.52 | *** join/#asterisk Pegger (n=peg@pool-68-163-144-11.bos.east.verizon.net) |
| 05:33.56 | SarahEmm | trixter: yep it does |
| 05:33.59 | SarahEmm | trixter: get what fixed? |
| 05:34.04 | trixter | SarahEmm: write abunch of code for sphynx so it can detect what people are saying and have a tts engine in there so you can reply |
| 05:34.10 | SarahEmm | uhh |
| 05:34.11 | trixter | get the keyboard stuff fixed |
| 05:34.15 | SarahEmm | that wouldn't help me |
| 05:34.26 | SarahEmm | (voice rec on my end) |
| 05:34.36 | trixter | if a remote person was speech2text via sphins or something and you replied via tts? |
| 05:34.40 | SarahEmm | trixter: err |
| 05:34.47 | trixter | er sphinx |
| 05:34.48 | SarahEmm | trixter: umm, that wouldn't help me so much :) |
| 05:35.01 | Qwell | trixter: congratulations, you've just reinvented relay services :P |
| 05:35.02 | trixter | then get a keyboard for your phone :P |
| 05:35.05 | Qwell | which are free... |
| 05:35.12 | trixter | you have dtmf, tap out hex bytes and let them encode them properly! |
| 05:35.12 | SarahEmm | Qwell: right, with eliminating the relay operator. |
| 05:35.15 | SarahEmm | lol trixter |
| 05:35.16 | Qwell | well, sure |
| 05:35.21 | Qwell | but relay ops rock |
| 05:35.26 | SarahEmm | trixter: i spend most of my life not speaking. TTS doesn't help me much ;) |
| 05:35.32 | Qwell | the last one I talked to, totally got mad at me |
| 05:35.38 | SarahEmm | Qwell: err, why mad? |
| 05:35.38 | trixter | it is for your reply! |
| 05:35.40 | SarahEmm | and yes, they rock lots |
| 05:35.41 | Qwell | umm |
| 05:35.44 | SarahEmm | trixter: what? *confused* |
| 05:35.47 | SarahEmm | trixter: how? |
| 05:35.52 | Qwell | 'Sir, anything you say, I'm obligated to repeat." |
| 05:35.55 | Qwell | :( |
| 05:35.58 | SarahEmm | umm, yes |
| 05:35.59 | dgorski | ;) |
| 05:36.03 | SarahEmm | ... what were you saying, Qwell? |
| 05:36.04 | trixter | ok, someone calls you voice, they go through a speech recognition program (which needs a lot of work) your replies to them are via TTS |
| 05:36.09 | Qwell | umm |
| 05:36.13 | SarahEmm | trixter: err... |
| 05:36.16 | Qwell | I was trying to be nice... |
| 05:36.23 | SarahEmm | trixter: how would the TTS work? |
| 05:36.25 | Qwell | polite, rather |
| 05:36.29 | trixter | you type it speaks |
| 05:36.29 | dgorski | sure |
| 05:36.30 | SarahEmm | trixter: it would require speech on my end |
| 05:36.32 | SarahEmm | ooh |
| 05:36.33 | SarahEmm | tts. |
| 05:36.35 | SarahEmm | not stt. |
| 05:36.38 | SarahEmm | duh. |
| 05:36.41 | trixter | that is for their end |
| 05:36.42 | Qwell | I asked her if she could like...have the person on the other end type instead of talk |
| 05:36.44 | SarahEmm | trixter: umm, but i don't have a keyboard ;) |
| 05:36.51 | trixter | but that code doesnt work well with dictionaries larger than about 10 words |
| 05:36.53 | trixter | :/ |
| 05:36.56 | Qwell | yeah, it was probably rude, but...it would have gotten things done quicker |
| 05:36.57 | SarahEmm | Qwell: ask them that directly, don't ask the relay op :) |
| 05:37.01 | trixter | well you need to get that fixed then! |
| 05:37.05 | Qwell | I was trying to be polite :p |
| 05:37.14 | SarahEmm | trixter: hehe :) |
| 05:37.14 | dgorski | I can appreciate that... ;) |
| 05:37.14 | SarahEmm | Qwell: well, that's not polite.... |
| 05:37.19 | SarahEmm | polite would be asking the other person directly ;) |
| 05:37.21 | Qwell | That's why I asked the relay op! :( |
| 05:37.25 | Qwell | meh :P |
| 05:37.29 | SarahEmm | heh |
| 05:37.37 | Qwell | That whole call was just...wow |
| 05:37.45 | Qwell | first (and only) relay experience |
| 05:37.51 | SarahEmm | ahhh okay Qwell |
| 05:38.01 | trixter | I knew a bunch of people that would call relay ops into conferences and be really anoying with it |
| 05:38.09 | trixter | just to get the relay op to say something as if its a challenge or something |
| 05:38.17 | Qwell | first thing I say "Yes Ma'am, we have a TTD number that you can call." |
| 05:38.25 | kuku5 | Qwell: When I try dialing, it only grabs the first character - and says unknown number |
| 05:38.34 | SarahEmm | Qwell: hopefully you say TDD or TTY and not TTD? ;) |
| 05:38.44 | Qwell | I read whatever the script said :P |
| 05:38.49 | SarahEmm | hehe okay :) |
| 05:39.05 | Qwell | I am so NEVER doing that job again though, I tell you |
| 05:39.22 | Qwell | voter registrar...2000 presidential election |
| 05:39.25 | Qwell | NOT fun! |
| 05:39.45 | Qwell | You wanna know the absolute most common question I got was? |
| 05:39.51 | SarahEmm | what? |
| 05:39.56 | Qwell | "What party am I?" |
| 05:40.00 | SarahEmm | lol |
| 05:40.17 | docelmo | Im independent.. I say the elephant and donkey can have theirs |
| 05:40.31 | Qwell | "Sir, it's on your voter registration card. We are not allowed to give out that information." |
| 05:40.50 | Qwell | "Well, I need to know." "Sir, it's on the voter registration card you received when you registered to vote." |
| 05:40.53 | Qwell | yeah |
| 05:40.54 | trixter | I am more affiliated with the bull |
| 05:41.04 | Qwell | I don't recommend that job to any of you, heh |
| 05:41.24 | Qwell | this was for the largest county of CA too...UGH! |
| 05:41.26 | trixter | capitalism is my religion! |
| 05:41.37 | Qwell | non-fucking-stop calls, for like...2 weeks. 8 hours a day |
| 05:41.46 | Qwell | then like 12+ hours a day on the last few days |
| 05:42.02 | *** part/#asterisk dseeb_ (n=dcb@CPE-144-131-35-116.vic.bigpond.net.au) |
| 05:42.49 | kuku5 | ok... sip blows |
| 05:42.54 | kuku5 | sccp rules |
| 05:43.30 | SarahEmm | gah |
| 05:43.38 | SarahEmm | stop saying that, it's going to end up being expensive for me to hear that ;0 |
| 05:43.41 | SarahEmm | ;) even |
| 05:44.03 | kuku5 | cisco makes good things - I dont know why i was forced to use this sip crap |
| 05:44.19 | kuku5 | It actually works better than sip |
| 05:44.29 | kuku5 | the UI - I havent touched the rest yet. |
| 05:57.21 | asteriskmonkey | night |
| 05:57.47 | Snake-Eyes | whats the best way to store numbers in extension.conf, i want to be able to have it where callerid(number) can equal a different number using a file, array or hash. eg person phones out and has ext 101 and then the callerid get converted to 80800101 so the outside person can redial back (missed calls) |
| 05:58.25 | Snake-Eyes | would love to have hash(101) = 80800100 :) |
| 05:58.53 | Pegger | anyone here in the voip business? |
| 05:59.23 | xevo | I doubt it |
| 05:59.30 | xevo | we're all in the porn business here |
| 05:59.43 | xevo | I'm a registered fluffer |
| 06:00.36 | russellb | what's voip? |
| 06:01.05 | russellb | file[laptop]: do you know? |
| 06:01.39 | xevo | kinky |
| 06:01.57 | Pegger | xevo, thanks for being a smart ass |
| 06:02.22 | xevo | Pegger: if it wasn't me... it was going to be somebody else |
| 06:02.42 | SarahEmm | Pegger: i'm not. i'm in the network operations business |
| 06:02.58 | SarahEmm | honestly i'd like to be in the voip business, but i'm not sure there are a lot of people that'd hire me for that. |
| 06:03.14 | Pegger | xevo, but I am sure some people know more about voip then others |
| 06:03.34 | Pegger | SarahEmm, and i am in the systems administartion business |
| 06:03.43 | SarahEmm | ahh |
| 06:03.59 | xevo | what do you want to know about the voip business? |
| 06:04.03 | SarahEmm | i run the networks that voip stuff runs on, but i don't work directly in it |
| 06:04.06 | Pegger | but I am looking fo radvse on what I need to do to be profitable in the voip business |
| 06:04.35 | SarahEmm | lol |
| 06:04.36 | SarahEmm | ummm |
| 06:04.37 | xevo | voip is a pretty broad term |
| 06:04.38 | Pegger | SarahEmm, can you offer any advise on buying termination? |
| 06:04.40 | SarahEmm | what do you do in the business? |
| 06:04.51 | *** join/#asterisk tengulre (n=tengulre@61.185.224.66) |
| 06:04.51 | SarahEmm | or are you just thinking 'i want to make money with voip'? |
| 06:04.59 | SarahEmm | i have no idea pegger |
| 06:05.00 | Pegger | SarahEmm, i am a systems admin, but I am looking to start a voip business on the side |
| 06:05.07 | SarahEmm | i do some devel on * stuff, but i'm not the person to ask about actually using it |
| 06:05.20 | Pegger | SarahEmm, why not i got the expertiese |
| 06:06.01 | SarahEmm | why not what? |
| 06:06.44 | Pegger | why not get into the business |
| 06:06.50 | SarahEmm | err |
| 06:06.52 | SarahEmm | you, or me? |
| 06:06.57 | Pegger | meeee |
| 06:06.58 | xevo | lol |
| 06:07.05 | SarahEmm | err |
| 06:07.11 | SarahEmm | i never said anything about you not getting into it |
| 06:07.15 | SarahEmm | you just .. don't exactly seem to have a plan |
| 06:07.18 | SarahEmm | what about the voip business? |
| 06:07.20 | SarahEmm | it's a big market |
| 06:08.05 | xevo | Pegger: I think you should sell IAX phones that are comparable to Cisco 7960s but for $30/each |
| 06:08.06 | Pegger | SarahEmm, whell I want to sell termination |
| 06:08.14 | *** join/#asterisk sl16 (n=blah@tv.neterra.net) |
| 06:08.19 | kuku5 | Which is better for dtmf iax or sip ? |
| 06:08.31 | Pegger | xevo, what I would love is a decent wireless iax phone |
| 06:08.40 | Pegger | kuku5, iax is better |
| 06:08.50 | xevo | Pegger: What are you going to offer that's better than the 30 other people selling termination? |
| 06:09.16 | kuku5 | Pegger: if I loose quick dtmf tones what does that mean ? |
| 06:10.36 | glm2k | Pegger: nothing negative intended but there's a world of difference between an expert sysadmin and an expert telco engineer. |
| 06:11.09 | Pegger | glm2k, well I a working on educating my self one day at a time |
| 06:11.33 | Pegger | and looking to hear what other peoples experinces has been |
| 06:11.48 | Pegger | so maby I can learn from their exerience |
| 06:12.01 | xevo | haha |
| 06:12.14 | glm2k | Pegger: that's all well and good. the mailing lists, specially the biz list has a lot of information. |
| 06:15.23 | Pegger | like one of my big questions is, I know that there is a small profit margin, what needs to be done it actully be able to make a profit |
| 06:16.33 | SarahEmm | give something nobody else does |
| 06:16.42 | SarahEmm | and charge enough you don't lose money |
| 06:16.48 | SarahEmm | voip isn't some magic market. it's just like everything else. |
| 06:17.09 | SarahEmm | if anyone could give you a magic formula to make VoIPMonEY, they'd be doing it themself :) |
| 06:17.44 | xevo | I'll sell you a cheap DS3 loop to get you started |
| 06:18.35 | xevo | and I'll throw in a free Adtran M13 multiplexor for a three year contract |
| 06:22.41 | *** join/#asterisk Haris (n=Haris@fiber10-228.brain.net.pk) |
| 06:22.48 | Haris | lol |
| 06:22.53 | Haris | already 1.2.5 released. |
| 06:23.02 | Haris | I just downloaded 1.2.4 befor the weekend |
| 06:23.18 | *** join/#asterisk clive- (n=pirch@dsl-145-3-60.telkomadsl.co.za) |
| 06:23.42 | *** join/#asterisk LapTop006 (n=laptop00@sparc006.chriskaine.com.au) |
| 06:24.00 | Haris | I'm getting compile errors for 1.2.4 on Fedora Core 3. Its a PIII 666 MHz box with 256 MB memory on it and ample space. |
| 06:24.50 | SarahEmm | what errors? |
| 06:25.23 | Haris | I'm downloading 1.2.5 release and compiling it. |
| 06:25.45 | Haris | => /usr/bin/ld: cannot find -lidn |
| 06:25.50 | Haris | collect2: ld returned 1 exit status |
| 06:26.48 | Haris | => http://pastebin.com/586473 |
| 06:27.10 | Haris | I think its complaining some library is missing |
| 06:27.25 | SarahEmm | libidn is missing..... |
| 06:27.37 | Haris | libidn-devel? |
| 06:28.08 | *** join/#asterisk Qwell (n=north@unaffiliated/qwell) |
| 06:28.12 | Haris | the devel package is not installed on my box. |
| 06:29.28 | Juggie | thats the problem |
| 06:29.31 | Juggie | install libidn-devel |
| 06:30.17 | Juggie | apt-get install libidn-devel or yum install libidn-devel |
| 06:30.24 | Juggie | depending on your distro! :) |
| 06:30.27 | *** join/#asterisk puzzled (n=patrick@puzzled.xs4all.nl) |
| 06:31.24 | puzzled | morning |
| 06:31.33 | Haris | done. |
| 06:32.19 | SarahEmm | morning puzzled ;) |
| 06:33.39 | clive- | What can one do to prevent tons of debug messages appearing, even though debug is off |
| 06:38.47 | *** join/#asterisk mcnobody (n=laaksola@laaksola.net) |
| 06:41.35 | sarah_away | bbiaf.. |
| 06:44.01 | Haris | Does asterisk require, that I have a mail server? |
| 06:44.18 | diLLec | if you're not using voicemail - no |
| 06:44.18 | sarah_away | err, no |
| 06:45.08 | Qwell | even if you are using voicemail...no |
| 06:46.23 | *** join/#asterisk liew123 (n=goh_mail@60.49.6.190) |
| 06:47.22 | liew123 | hello, how can I load modprobe zaptel and wctdm when linux is boot? |
| 06:52.29 | Haris | Cool => +--------- Asterisk Build Complete ---------+ |
| 06:56.11 | Haris | install complete, make progdocs complete. |
| 06:56.46 | Haris | Ok, /etc/asterisk does not contain any files at all.. |
| 06:56.57 | Abydos313 | make samples |
| 06:56.59 | Haris | Should I do a make samples for it to generate default config files? |
| 06:57.08 | Abydos313 | liew123 check pm |
| 06:57.14 | Haris | wow! |
| 06:57.28 | Haris | 160 config files |
| 06:57.29 | Haris | lol |
| 06:58.06 | Haris | correction: 45 |
| 06:58.28 | Haris | Does asterisk install a service start/stop script in /etc/rc.d on redhat systems? |
| 06:58.53 | Abydos313 | Also, in the Asterisk sources, you'll find a |
| 06:58.53 | Abydos313 | contrib/init.d/rc.redhat.asterisk file that can be used to start |
| 06:58.53 | Abydos313 | Asterisk up during boot. |
| 06:58.53 | Abydos313 | cp rc.redhat.asterisk /etc/rc.d/init.d/asterisk |
| 06:58.53 | Abydos313 | chmod +x /etc/rc.d/init.d/asterisk |
| 06:58.54 | Abydos313 | chkconfig --add asterisk |
| 06:58.56 | Abydos313 | chkconfig asterisk on |
| 06:59.16 | SarahEmm | rehi |
| 07:00.45 | *** join/#asterisk littlejohn (n=little@host23-67.pool8255.interbusiness.it) |
| 07:00.58 | liew123 | Abydos313:thanks |
| 07:01.04 | Abydos313 | np |
| 07:01.23 | Abydos313 | pm and email addy and i'll send the doc that has it for zaptel and astersik |
| 07:02.00 | Haris | that last statement, makes asterisk be turned on automatically once the system boots? |
| 07:02.29 | Abydos313 | i think so, i just follow the directions and it all works like any other service |
| 07:02.57 | *** join/#asterisk chapeaurouge (n=chap@vilhost1.vision.lu) |
| 07:05.25 | *** join/#asterisk Koshatul (n=evangeli@ip157-65-132.cust.bit.net.au) |
| 07:06.47 | glm2k | www.carballo.biz |
| 07:06.50 | *** join/#asterisk dmaust (n=dmaust@adsl-67-120-175-165.dsl.lsan03.pacbell.net) |
| 07:08.08 | Haris | I'v installed asterisk successfully... |
| 07:11.10 | *** join/#asterisk sener (n=sener@p85.212.17.152.tisdip.tiscali.de) |
| 07:11.28 | Abydos313 | congrats |
| 07:13.55 | *** join/#asterisk oej (n=oej@apollo.webway.se) |
| 07:17.44 | *** join/#asterisk ozant (n=zet@mail.grid.com.tr) |
| 07:22.07 | litage | without having a FWD account, can Asterisk redirect calls to sip:FWD_NUMBER@fwd.pulver.com:5060 ? |
| 07:22.19 | Haris | Does the asterisk docs/wiki cover or have a walk-through for first time starters.. |
| 07:22.31 | litage | or would the redirected call not be allowed since the Asterisk box isn't registered with FWD? |
| 07:22.48 | Haris | For e.g. 1) install asterisk 2) create user accounts 3) register accounts in asterisk 4) specify call paths |
| 07:22.55 | litage | Haris: yes |
| 07:22.57 | X-Rob_ | Can anyone recommend something like a SPA-3000 that isn't just 'passthrough'? |
| 07:23.06 | litage | Haris: search the voip-info.org wiki |
| 07:24.12 | *** join/#asterisk Qwell (n=north@unaffiliated/qwell) |
| 07:27.39 | Haris | point 2, 3 is the focus of my Q. |
| 07:29.56 | SarahEmm | what do you mean by 'user accounts'? |
| 07:30.21 | litage | SarahEmm: i'm guessing he means sip/iax/etc users and peers |
| 07:31.51 | Haris | I'v installed it the first time. I want to test it in levels. Level 1) Software based PC <-> PC (over LAN) , Level 2) PC-To-Phone |
| 07:32.07 | Haris | I'm told, I have to add user accounts and register them under asterisk. |
| 07:32.15 | Haris | I'm new! to asterisk |
| 07:32.28 | Haris | levels = phase |
| 07:32.44 | Juggie | www.voip-info.org |
| 07:32.47 | Juggie | read. |
| 07:32.47 | litage | Haris: that's correct. a "user account" doesn't really exist though. what you do is create users/peers/friends for a given protocol (IE: sip, iax, h323) |
| 07:33.13 | Haris | litage: I see! |
| 07:33.48 | litage | Haris: if you create a SIP friend called "bob", then you can plug bob's username and password, as well as Asterisk's ip address and sip port, into a softphone |
| 07:35.24 | litage | Haris: the best place for information is www.voip-info.org |
| 07:36.18 | *** join/#asterisk sulex (n=sulex@pdpc/supporter/active/sulex) |
| 07:36.31 | litage | Haris: use voip-info.org's search facility to find what you're looking for. Eg: http://www.voip-info.org/wiki-Asterisk+config+sip.conf |
| 07:36.33 | Haris | Reading! ... |
| 07:37.19 | litage | great |
| 07:37.39 | *** join/#asterisk TonyM (n=TonyM@adsl-solo-80-168-225-93.claranet.co.uk) |
| 07:38.31 | YaP | i'm using asterisk with eicon 4 bri but during a call there is a lot of noise, what could be the problem? |
| 07:45.01 | puzzled | YaP: which version of chan_capi-cm do you use? |
| 07:47.35 | *** join/#asterisk astar` (n=astar@ANantes-154-1-29-146.w81-53.abo.wanadoo.fr) |
| 07:50.31 | *** join/#asterisk Qwell (n=north@unaffiliated/qwell) |
| 07:53.56 | Haris | Are mailboxes needed for using x-lite? (sip.conf, default section [xlite1] from make samples) |
| 07:55.25 | Qwell | no |
| 07:56.02 | Qwell | AT&T+SBC buying Bellsouth |
| 07:56.18 | puzzled | yeah just read about it |
| 07:56.19 | Qwell | "This action by AT&T will consolidate more than half of the original Bell System into a single entity, leaving only Verizon and Qwest as remaining Bell family competitors." |
| 07:56.31 | puzzled | so long for competition |
| 07:56.39 | Qwell | I give it 6-12 months before Verizon buys Bell |
| 07:56.43 | Qwell | "Bell" |
| 07:57.00 | Qwell | Then within 18-24, it'll be Sprint/Nextel |
| 07:57.03 | Qwell | mark my words |
| 07:57.16 | Qwell | we'll be in a WORSE situation than we were 30 years ago |
| 07:57.21 | puzzled | is the FCC sleeping or what? |
| 07:57.24 | Qwell | (as if we aren't already) |
| 07:57.40 | newl | puzzled: They're waiting for AT&T to get fatter first. :) |
| 07:57.50 | Haris | ... |
| 07:58.05 | newl | The process will then have come full circle and have to be repeated again with the original BFH. :) |
| 07:58.13 | puzzled | hehe |
| 08:02.30 | *** join/#asterisk oej (n=oej@apollo.webway.se) |
| 08:04.11 | *** join/#asterisk vgster (n=vg@host217-45-221-53.in-addr.btopenworld.com) |
| 08:10.05 | nayyares | which one is the best billing for Asterisk? especially PHP/mySQL ? |
| 08:10.53 | *** join/#asterisk Defraz (n=t0tal@fw.centrisys.com) |
| 08:11.01 | *** join/#asterisk BhaalWK (i=bhaal@freenode/staff/bhaal) |
| 08:20.41 | *** join/#asterisk pengyong (n=lala@222.188.138.165) |
| 08:36.47 | *** join/#asterisk X-Rob_ (n=Rob@dsl-220-235-83-83.vic.westnet.com.au) |
| 08:37.07 | *** join/#asterisk jradford (n=jradford@hoss.npl.com) |
| 08:43.34 | *** join/#asterisk P0L0 (n=tekn0@62-43-65-175.user.ono.com) |
| 08:47.10 | *** join/#asterisk Kernel_Core (i=Kernel_C@217.218.80.248) |
| 08:59.45 | *** join/#asterisk cypromis (n=michael@asterisk.pl) |
| 09:01.38 | *** join/#asterisk astar` (n=astar@ANantes-154-1-3-68.w81-53.abo.wanadoo.fr) |
| 09:02.07 | *** join/#asterisk viperdude (n=jon@borat.enta.net) |
| 09:02.11 | *** join/#asterisk jorgito (n=jorge@snat2.arachne.czfree.net) |
| 09:02.12 | jorgito | hi |
| 09:02.37 | jorgito | what sip termination for international calls do you recommend me ? i am in europe, |
| 09:04.15 | puzzled | jorgito: there is a list on www.voip-info.org |
| 09:06.28 | littlejohn | jorgito, skypho.net |
| 09:10.02 | *** join/#asterisk nagl (n=nagl@rih.zid-nw.wu-wien.ac.at) |
| 09:10.59 | *** join/#asterisk RoyK (n=roy@80.239.107.70) |
| 09:11.12 | puzzled | morning RoyK |
| 09:11.56 | RoyK | morning |
| 09:13.28 | RoyK | fuck jerjer |
| 09:14.04 | puzzled | ? |
| 09:14.05 | SarahEmm | RoyK: ? |
| 09:16.45 | SarahEmm | meep! |
| 09:16.48 | SarahEmm | *tips over* |
| 09:16.54 | SarahEmm | *fires some baudot back* |
| 09:17.36 | trixter | heh |
| 09:18.21 | RoyK | well. he's just banned me from #asterisk-dev for answering him |
| 09:18.21 | puzzled | battle of the bits |
| 09:18.59 | puzzled | don't see a ban msg but maybe I joined later |
| 09:19.15 | trixter | what was the question? |
| 09:20.07 | puzzled | how do I use realtime with radius :) |
| 09:20.13 | trixter | ahh |
| 09:20.23 | puzzled | nah, just messing around |
| 09:20.26 | trixter | be careful the mailbomb doesnt start |
| 09:20.33 | puzzled | hehe |
| 09:20.46 | trixter | and the call to your isp becuase you are accepting all these emails from him so your access should be turned off |
| 09:20.52 | trixter | I laughed when my isp called me over that one |
| 09:21.04 | RoyK | puzzled: there is one |
| 09:21.11 | RoyK | anyone with +o on -dev around? |
| 09:21.26 | puzzled | don't have o |
| 09:21.47 | jorgito | odes anybody know some hylafax channel in irc / |
| 09:21.48 | trixter | its a username ban change ident :P |
| 09:22.17 | puzzled | change it to "I love realtime and radius" |
| 09:22.44 | RoyK | puzzled: cc: jerjer |
| 09:22.44 | trixter | mine handled it fine, it was 6k/hr but the claim was 60k/hr |
| 09:23.29 | puzzled | and how long did it last? |
| 09:23.40 | trixter | few hours |
| 09:24.08 | trixter | I had all the logs though so my isp didnt care and thought it was stupid that he called them to get my service shut off |
| 09:24.33 | puzzled | hehe |
| 09:25.32 | RoyK | anyone who remembers the remove ban syntax? oej needs some help...... |
| 09:26.24 | trixter | <PROTECTED> |
| 09:26.37 | trixter | <PROTECTED> |
| 09:26.44 | trixter | there were enough of em there for ya |
| 09:26.45 | trixter | :/ |
| 09:26.58 | trixter | the last 2 are prolly it, but meh |
| 09:27.03 | *** join/#asterisk A-Tuin|work (n=A-Tuin@212.41.185.81) |
| 09:27.26 | trixter | ahh last 3 at least |
| 09:27.37 | RoyK | jerjer...... |
| 09:27.46 | trixter | all of those were by jer |
| 09:27.52 | RoyK | yes |
| 09:28.22 | trixter | 3 were all placed within minutes of each other the other two later on |
| 09:29.49 | *** join/#asterisk welles (n=welles@61.150.43.114) |
| 09:35.56 | *** join/#asterisk wellng (n=welles@61.150.43.114) |
| 09:36.23 | *** part/#asterisk wellng (n=welles@61.150.43.114) |
| 09:36.30 | *** part/#asterisk welles (n=welles@61.150.43.114) |
| 09:37.12 | *** join/#asterisk welles (n=welles@61.150.43.114) |
| 09:40.57 | *** join/#asterisk welles (n=welles@61.150.43.114) |
| 09:43.55 | *** join/#asterisk zeedo (n=zeedo@reboot-robot.net) |
| 09:44.12 | jorgito | somebady using hylafax with diva server e1 card ? |
| 09:45.46 | liew123 | I got a question, I dunno why my IAX phone can't hear what another analog phone form digium wctdm card said when I it is only dial from anolog phone to IAX but not other way round. |
| 09:48.26 | liew123 | Please advice |
| 09:51.40 | *** join/#asterisk pif (n=ldm@zenon.apartia.fr) |
| 09:53.59 | Haris | http://sial.org/pbot/16175 |
| 09:54.46 | *** join/#asterisk simulated (n=joker@71.196.10.2) |
| 09:54.58 | simulated | hi ho |
| 09:55.26 | *** join/#asterisk blkremedy (n=ur3rdeye@142M28.oasis.mediatti.net) |
| 09:55.34 | blkremedy | hello |
| 09:55.35 | simulated | anyone have any luck compiling g726 into a ast_h323 driver ? |
| 09:56.16 | blkremedy | anyone know how to get tdm400p to hangup properly on fxo module? |
| 09:56.47 | *** join/#asterisk tengulre (n=tengulre@61.185.224.66) |
| 09:58.07 | *** join/#asterisk Enth (i=wc@kahlan.skyforge.net) |
| 10:03.12 | *** join/#asterisk gwik (n=gwik@ATuileries-151-1-13-24.w82-123.abo.wanadoo.fr) |
| 10:03.22 | gwik | Hi |
| 10:04.41 | gwik | I'd like to know what are the restristion (if there are) bout the ip phone ?? |
| 10:04.45 | tzafrir | blkremedy, do you use loopstart or kewlstart? |
| 10:05.12 | *** join/#asterisk areski (n=areski@polar.es6.egwn.net) |
| 10:05.14 | blkremedy | I use kewlstart |
| 10:05.58 | simulated | gwik: idefisk = best iax softphone |
| 10:06.59 | simulated | iaxComm is heavily outdated |
| 10:07.04 | simulated | unfortunatly |
| 10:07.29 | gwik | simulated: I meant real Ip phones not soft phones |
| 10:07.53 | simulated | like the cisco 7940? |
| 10:10.25 | gwik | simulated: yep |
| 10:10.53 | simulated | good phone |
| 10:13.00 | gwik | simulated: quite expensive, in fact i'd like to know if features like annuaries depends on the phone |
| 10:13.09 | simulated | annuaries? |
| 10:14.04 | gwik | sorry, I meant directories |
| 10:15.08 | *** join/#asterisk Money5ack (i=moneysac@wer.will.spontanficken.de) |
| 10:15.27 | simulated | services? |
| 10:15.36 | simulated | i have a farm of 7940's and 7960's |
| 10:15.38 | simulated | i love them |
| 10:15.42 | simulated | in SCCP mode and SIP mode |
| 10:16.08 | gwik | ldap-like service |
| 10:17.34 | simulated | what are you looking to do? |
| 10:17.53 | simulated | the 79xx's are maintained with tftp conf files |
| 10:18.03 | *** join/#asterisk littlejohn (n=little@host23-67.pool8255.interbusiness.it) |
| 10:18.34 | gwik | I want some phones that are able to connect to a ldap server for example and dial from it |
| 10:18.56 | welles | hi trixter ,there? |
| 10:19.26 | gwik | ldap or any database |
| 10:20.29 | simulated | gwik, there is definetly a pathway to do what you want to do, it is a long one tho |
| 10:20.59 | trixter | yes |
| 10:21.09 | gwik | simulated: ok i see |
| 10:21.51 | *** join/#asterisk UlbabraB (n=salama@host241-43.pool8172.interbusiness.it) |
| 10:22.03 | gwik | simulated: Can you dial from your Computer |
| 10:22.07 | gwik | simulated: ?? |
| 10:22.44 | welles | trixter, when asterisk try to connect another asterisk and a iaxclient which behind a nat. it always failed. |
| 10:22.51 | gwik | simulated: and do the call with your phone |
| 10:23.17 | welles | trixter, have any ideas? |
| 10:23.53 | trixter | not really |
| 10:24.01 | trixter | unless your nat box is not letting the packets through |
| 10:24.05 | trixter | how exactly does it fail? |
| 10:24.23 | simulated | gwik: i can do anything |
| 10:25.11 | gwik | simulated: Is it cisco specific or is it something that depends on the protocol ?? |
| 10:25.43 | simulated | gwik there's a variety of softphones and ATAs |
| 10:26.29 | welles | iaxclient will use another port to send the packet to the another asterisk B,but the asterisk B will send the packet to iaxclient to port which he use to registration on asterisk A |
| 10:27.12 | welles | trixter, so they always fail |
| 10:27.18 | *** join/#asterisk fenlander (n=fenlande@82.152.81.57) |
| 10:27.24 | gwik | simulated: Thank you for your help I'll try to figure out what I need more precisly before asking again |
| 10:30.47 | *** join/#asterisk X-Rob_ (n=Rob@dsl-220-235-83-83.vic.westnet.com.au) |
| 10:31.03 | welles | anyone can help me about my issue? |
| 10:32.53 | *** join/#asterisk Dico_ (n=niko@60.51.217.58) |
| 10:33.02 | Dico_ | hello :) |
| 10:33.32 | trixter | welles: I dunno ... |
| 10:35.05 | *** part/#asterisk Dico_ (n=niko@60.51.217.58) |
| 10:37.14 | *** join/#asterisk svenna_ (n=svenna@p548D42EA.dip0.t-ipconnect.de) |
| 10:49.00 | jorgito | hi |
| 10:51.30 | *** join/#asterisk mbranca (n=matteo@host-210-mi.linuxserver.it) |
| 10:53.23 | jorgito | can i do music on hold while dialing ? i mean in the same moment that if i call that extension i will hear moh and simultanously it will dial number , when reach number moh stops... any ideas ? |
| 10:53.47 | Delvar | show application dial |
| 10:53.56 | Delvar | 'm' i think is the option |
| 10:54.23 | jorgito | Delvar, i have the book i will take look thanks |
| 11:00.59 | jorgito | Delvar, Dial(SIP/728043760@sip001,m[default]) or Dial(SIP/728043760@sip001,,m[default]) |
| 11:04.37 | Delvar | the second one |
| 11:06.03 | jorgito | Delvar, working thanks |
| 11:07.34 | Delvar | np |
| 11:22.47 | RoyK | ~seen zoa |
| 11:22.49 | jbot | zoa <n=kkk@pirus.securax.be> was last seen on IRC in channel #asterisk, 13d 17h 15m 24s ago, saying: '1000 interrupts per second are ok'. |
| 11:24.00 | *** join/#asterisk arcraig (n=arcraig@c-67-187-191-82.hsd1.ca.comcast.net) |
| 11:24.36 | *** join/#asterisk backblue (n=igor@82.102.1.42) |
| 11:25.06 | backblue | hi, does anyone knows any software (simple) to see pgsql or mysql cdr from asterisk? i only need cdr. |
| 11:25.25 | backblue | (not areski please) |
| 11:28.17 | areski | backblue, :) |
| 11:28.35 | backblue | :) |
| 11:28.41 | X-Rob_ | ~seen my car keys? |
| 11:28.43 | jbot | i haven't seen 'my car keys', X-Rob_ |
| 11:28.49 | X-Rob_ | Bugger. |
| 11:28.51 | backblue | X-Rob_: hehe |
| 11:28.52 | areski | backblue, u dont like it ? |
| 11:29.08 | RoyK | ~lart X-Rob_ |
| 11:29.15 | X-Rob_ | 8) |
| 11:29.23 | backblue | areski: to many information, i need something more simple, and dont have language suporte. |
| 11:29.46 | backblue | and i dont need graphs :D |
| 11:30.13 | *** join/#asterisk Dibbler (n=Dibbler@zidane.pi-net.net) |
| 11:30.26 | areski | backblue, you can extract part u need really quickly |
| 11:30.31 | *** join/#asterisk fulgas (n=fulgas@209.8.233.248) |
| 11:31.14 | backblue | areski: yes i know, but before that, i'm searching for another one more simple! :D rather spend 10min searching then 1h changing! :) |
| 11:32.15 | areski | right |
| 11:32.18 | areski | :) |
| 11:36.58 | backblue | areski: are you the author? |
| 11:37.29 | x86 | hmm |
| 11:37.33 | RoyK | hmmmmmmmmm |
| 11:37.41 | x86 | how do i play hold music for the duration of a System() call? |
| 11:37.54 | x86 | but make it stop when the System() is finished |
| 11:37.56 | areski | backblue, yes :-) |
| 11:38.41 | x86 | is it possible? |
| 11:38.46 | backblue | areski: good work. |
| 11:39.14 | backblue | x86: do you really need system() ? : |
| 11:39.15 | backblue | :o |
| 11:39.32 | x86 | backblue: yeah, why? |
| 11:41.10 | areski | backblue, thx u :D |
| 11:41.13 | backblue | x86: i know playing the music will you run system, but i dont know if it stops, or when it stops, but check background() documentation. |
| 11:41.15 | x86 | this is what i want: user calls in, dials an extension, user is put on hold, the extension is setup to execute a system command to fetch the latest weather and then pipe it to text2wave to conver it to wave, then it takes user off hold and playsback the weather, then deletes the wave |
| 11:41.29 | backblue | i use it for ivr proposes. |
| 11:41.45 | *** part/#asterisk svenna_ (n=svenna@p548D42EA.dip0.t-ipconnect.de) |
| 11:41.46 | x86 | ah yes... but can i Background my MOH? |
| 11:41.56 | x86 | i would think i would have to choose just one file :( |
| 11:42.06 | x86 | and even then |
| 11:42.12 | x86 | it would not stop when System was done ;) |
| 11:42.18 | backblue | you can use some bash script to do the same job, as moh |
| 11:42.24 | x86 | it would only stop when the file was done playing |
| 11:42.39 | x86 | hmm |
| 11:42.52 | x86 | i could also make a 20 second segment of one of my MOH files and be done ;) |
| 11:44.12 | *** join/#asterisk stoffell (n=stoffell@fw.catsanddogs.com) |
| 11:45.55 | welles | anyone can help me? |
| 11:46.12 | backblue | welles: dont ask to ask. |
| 11:46.40 | welles | when asterisk try to connect another asterisk and a iaxclient which behind a nat. it always failed. |
| 11:48.01 | welles | iaxclient will use another port to send the packet to the another asterisk B,but the asterisk B will send the packet to iaxclient to port which he use to registration on asterisk A |
| 11:48.36 | welles | backblue, have any suggestion？ |
| 11:48.56 | *** join/#asterisk umay (n=chris@70-101-61-50.dsl2-plymouth.roc.ny.frontiernet.net) |
| 11:49.59 | backblue | welles: iax works better behind firewall than sip |
| 11:50.21 | backblue | because it works everything inside the same package |
| 11:50.39 | backblue | i have no problems iax behind nat |
| 11:54.34 | welles | i set notransfer = yes in iax.conf for a peer .but asterisk still send TXREQ,why ?by the way i use realtime configure for iax in extconf.conf |
| 11:54.43 | x86 | backblue: even if i use Background, it waits to do the System command until after background is done... |
| 11:54.51 | RoyK | ~seen zoa |
| 11:54.53 | jbot | zoa <n=kkk@pirus.securax.be> was last seen on IRC in channel #asterisk, 13d 17h 47m 28s ago, saying: '1000 interrupts per second are ok'. |
| 11:55.02 | x86 | backblue: i think i need to park the call, do the system, then retrieve the call.... is this possible? |
| 11:55.25 | *** join/#asterisk _deg_ (n=deg@200.139.117.63.adsl.gvt.net.br) |
| 11:57.58 | welles | backblue, i use dial app's L parameter. and call another asterisk B''s extension ,it still try to connect my iaxclient and asterisk B directly, it is right? |
| 12:00.26 | *** join/#asterisk Sternn (n=sternn@user-0c938ku.cable.mindspring.com) |
| 12:03.04 | *** join/#asterisk welles (n=welles@61.150.43.114) |
| 12:03.17 | *** join/#asterisk cpm (n=Chip@pdpc/supporter/sustaining/cpm) |
| 12:11.27 | *** join/#asterisk UdontKnow (i=udontkno@freenode/staff/udontknow) |
| 12:12.12 | jorgito | Agent logs on, hears music on hold, waits for call. agent must wait for call listening moh ? cannt be done that the call will ring agenst phone ? |
| 12:12.44 | *** join/#asterisk Kernel_Core (i=Kernel_C@217.218.80.157) |
| 12:12.47 | Kernel_Core | hi all |
| 12:12.52 | Kernel_Core | anybody familiar with H.323 ?! |
| 12:13.33 | Kernel_Core | connection establishes, but I hear nothing .... |
| 12:15.59 | *** join/#asterisk stock (n=stock@stokkie.demon.nl) |
| 12:16.04 | stock | hello |
| 12:16.17 | UdontKnow | 'lo |
| 12:16.46 | UdontKnow | hmmm, so I was seeing that e164.org is starting to have translations |
| 12:16.57 | UdontKnow | but many translations are missing yet :/ |
| 12:16.59 | stock | i'm looking for a sip phone to work with asterisk, not to install a call manager server .. what sip phones are recommended? |
| 12:17.37 | umay | Kernel_Core: could be codec mismatch |
| 12:17.50 | UdontKnow | I guess they miss some voluntaries |
| 12:19.17 | *** join/#asterisk Dantix (n=Dantix@200.68.70.100) |
| 12:19.37 | Dantix | help |
| 12:19.44 | Dantix | sorry! |
| 12:19.47 | *** part/#asterisk Dantix (n=Dantix@200.68.70.100) |
| 12:20.21 | Kernel_Core | umay: how do I set my codecs ? |
| 12:20.33 | Kernel_Core | umay: allow=all |
| 12:20.44 | *** join/#asterisk anandbabu (i=ab@69.12.132.138) |
| 12:20.50 | stock | so which sip phone should one buy? |
| 12:21.30 | *** part/#asterisk octothorpe_ (n=octothor@c-67-186-207-234.hsd1.ut.comcast.net) |
| 12:21.41 | umay | stock: not grandstream |
| 12:21.41 | *** join/#asterisk Shoragan (n=shoragan@d072.apm.etc.tu-bs.de) |
| 12:21.54 | stock | oh ok.. why's that? |
| 12:21.55 | umay | Kernel_Core: what H.323 implementation ? are both sides Asterisk ? |
| 12:22.35 | stock | nowadays just mentioning 'should work with asterisk' should be enough |
| 12:22.55 | stock | so, any list of sip phones somewhere? |
| 12:23.01 | Kernel_Core | umay: I am running 1.2.4 and I compiled channels/h323/ |
| 12:23.07 | Kernel_Core | umay: I get this error : -- H323/amir-1 is making progress passing it to SIP/25-875c |
| 12:23.08 | Kernel_Core | Mar 6 06:16:55 WARNING[24454]: channel.c:787 channel_find_locked: Avoided initial deadlock for '0x81a1e40', 10 retries! |
| 12:23.34 | Kernel_Core | umay: connectoin establishes but I get this warning .... |
| 12:24.21 | umay | try the ooh323 in asterisk-addons |
| 12:24.30 | Kernel_Core | umay: == Registered channel type 'H323' (The NuFone Network's Open H.323 Channel Driver) |
| 12:24.42 | umay | sip phones are probably listed on the wiki |
| 12:24.47 | Kernel_Core | umay: I heard h323 is the most stable channel ... |
| 12:24.58 | Kernel_Core | umay: I am useing Eyebeam SIP Phone |
| 12:25.26 | jorgito | somebody here built call center? |
| 12:26.03 | umay | if nufone's was the most stable, then digium wouldn't have paid for objective systems to write an interface to ooh323 stack |
| 12:26.33 | trixter | ha |
| 12:26.37 | anandbabu | other than NAT issues, how is SIP compared to IAX2 when it comes to quality? I started with IAXy for my voip network. Not sure if SIP is any better |
| 12:26.44 | Kernel_Core | UdontKnow: I don't know ... |
| 12:26.52 | jorgito | somebody here built call center? |
| 12:26.53 | stock | umay ah ok... well lets make a example, for instance, TLS, http://www.thelinuxshow.com/ is makeing their reappearance using asterisk based sip phone radio forum.... so _they_ provide the asterisk 'hookup'. i as a hummble listener only need a sip phone... whats a good and affordable sip phone for such things? |
| 12:27.10 | umay | jorgito: are you building one ? |
| 12:27.21 | trixter | stock: www.sjlabs.com - free |
| 12:27.39 | jorgito | umay, yes |
| 12:28.05 | stock | hmm |
| 12:28.21 | umay | jorgito: how big ? |
| 12:29.28 | umay | anandbabu: SIP is supported by the telecom industry, IAX is supported by Asterisk |
| 12:30.48 | jorgito | umay, for about 100 people... , well |
| 12:30.53 | anandbabu | umay, So do you recommend IAX2 if I am building a VoIP network within my own company? |
| 12:31.11 | *** join/#asterisk octothorpe_ (n=octothor@c-67-186-207-234.hsd1.ut.comcast.net) |
| 12:31.29 | octothorpe_ | ~confessiom |
| 12:31.32 | octothorpe_ | ~confession |
| 12:31.38 | anandbabu | anandbabu, umay I use Asterisk on Debian |
| 12:32.40 | *** join/#asterisk Tili (n=Tili@82-217-236-131.cable.quicknet.nl) |
| 12:32.46 | Tili | hi ScaredyCat |
| 12:32.58 | tzafrir | anandbabu, sure. Quite recommended |
| 12:33.50 | umay | if theres nothing but asterisk boxes yeah why not |
| 12:34.28 | Tili | ~seen ScaredyCat |
| 12:34.52 | jbot | scaredycat <n=ScaredyC@net-pbx.demon.nl> was last seen on IRC in channel #asterisk, 5d 21h 21m 6s ago, saying: 'weak...'. |
| 12:34.52 | UdontKnow | umay: well... |
| 12:34.54 | UdontKnow | one can IAX between all the asterisks and SIP to the other hardware |
| 12:35.02 | anandbabu | thanks for the info. It is pure asterisk network and I will stick to IAX2. |
| 12:35.07 | UdontKnow | but when its asterisk-asterisk, IAX is recommended |
| 12:35.11 | UdontKnow | IAX2 even |
| 12:36.03 | anandbabu | other than IAXy, any other IAX ATAs or IP phones recommended? |
| 12:37.29 | *** join/#asterisk _Paulo_ (n=Paulo@200-168-112-132.dsl.telesp.net.br) |
| 12:37.48 | _Paulo_ | Morning |
| 12:38.31 | x86 | anyway to make an extension park a call, do something, and when finished doing something retrieve the parked call? |
| 12:40.58 | _Paulo_ | x86, why are you trying to do that? |
| 12:41.24 | backblue | x86: dont know. |
| 12:41.29 | backblue | x86: never done that. |
| 12:41.59 | backblue | why do you need so much kung fu? what are you trying to do with your system? |
| 12:42.25 | _Paulo_ | x86, hint us... :-) |
| 12:45.08 | *** join/#asterisk linville (n=linville@azure.tuxdriver.com) |
| 12:45.24 | *** join/#asterisk Eitch (n=hugo@unaffiliated/eitch) |
| 12:46.24 | Eitch | hau |
| 12:46.29 | *** join/#asterisk fourcheeze (n=rich@82.153.215.21) |
| 12:46.41 | *** join/#asterisk _deg_ (n=deg@200.250.222.8) |
| 12:47.50 | Kernel_Core | umay: with h323 module , I could make connection but there was no sound , with ooh323 I can't even make connection ( I get busy ...) |
| 12:49.27 | RoyK | try yate :) |
| 12:49.38 | RoyK | iirc yate works stably with h323 |
| 12:49.42 | RoyK | * does not |
| 12:50.07 | umay | yate is a lot less stable |
| 12:50.21 | Kernel_Core | :| |
| 12:50.39 | *** join/#asterisk m_a_g_o (i=maxgluck@201.243.97.246) |
| 12:50.45 | clive- | kernel core how about woomera ? |
| 12:50.49 | Kernel_Core | RoyK: what is yate ?! |
| 12:51.01 | RoyK | Kernel_Core: use google :) |
| 12:51.02 | RoyK | ~yate |
| 12:51.10 | RoyK | ~wiki yate |
| 12:51.14 | RoyK | ~google yate |
| 12:51.26 | umay | it doesnt exist :P |
| 12:51.36 | Kernel_Core | RoyK: is it kind of GK ? |
| 12:51.47 | *** join/#asterisk antiloopj (n=jip@i18218.upc-i.chello.nl) |
| 12:52.02 | m_a_g_o | anyone selling IP phones capable of displaying realtime (may be XML) content on the LCD display? |
| 12:52.05 | antiloopj | anyone working here with hfc cards and a PBX behind asterisk, got some issues that asterisk does not pass 'hangup' to my pbx? |
| 12:52.12 | RoyK | Kernel_Core: rtfw |
| 12:52.32 | Kernel_Core | :) |
| 12:53.27 | Kernel_Core | RoyK: is it modified asterisk ? |
| 12:53.31 | umay | m_a_g_o: i am looking for that too if you find one |
| 12:53.39 | backblue | i say, forget h323 :o |
| 12:54.01 | backblue | antiloopj: yes, what do you need. |
| 12:54.30 | Kernel_Core | backblue: why ? |
| 12:54.34 | backblue | m_a_g_o: cisco 794x does that? |
| 12:54.54 | *** join/#asterisk PakiPenguin (n=uppal@linuxpakistan/admin/pakipenguin) |
| 12:55.03 | backblue | Kernel_Core: because, i think its dead, i mean, it's the past and not the future! |
| 12:55.13 | RoyK | Kernel_Core: rtfw |
| 12:55.26 | RoyK | Kernel_Core: it's not. it's a whole new idea |
| 12:55.27 | fourcheeze | Is there a speaking clock for asterisk? I don't just mean DateTime() but something that says "On the third beep the time will be 12:55 and 30 seconds... beep beep beep" |
| 12:55.31 | backblue | unless you really really really need it |
| 12:55.31 | RoyK | Kernel_Core: written in c++ |
| 12:55.34 | RoyK | very modular |
| 12:55.34 | RoyK | etc |
| 12:56.04 | RoyK | h.323 is very, very annoying |
| 12:56.04 | *** join/#asterisk anandbabu (i=ab@69-12-132-138.dsl.static.sonic.net) |
| 12:56.07 | umay | cisco 794x takes arbitrary text and displays it immediately on the LCD ? |
| 12:56.49 | backblue | umay: what voip phones do you know, that do that in realtime? that would be a waste of resources :P |
| 12:56.52 | Kernel_Core | RoyK: is it stable ? on 2.8GHZ Xeon Dual , 2GB Ram , without transcodeing how many h323 channels may I have ? |
| 12:57.05 | RoyK | Kernel_Core: nfi |
| 12:57.08 | RoyK | Kernel_Core: try /j #yate |
| 12:57.36 | RoyK | it would surprise me if it scales any worse than asterisk |
| 12:57.52 | umay | backblue: it would be a great way to present IVRs |
| 12:58.21 | *** join/#asterisk YaP (n=YaP@host-84-223-138-54.cust-adsl.tiscali.it) |
| 12:58.23 | YaP | re |
| 12:58.26 | backblue | umay: yes. |
| 12:59.24 | m_a_g_o | backblue: with *? I thougth it only worked with the CM |
| 12:59.59 | backblue | m_a_g_o: what? |
| 13:00.14 | antiloopj | backblue when the line is 'hangup' asterisk says '-- Channel 0/1, span 1 got hangup ACK' fine.. but it does not pass this message to the PBX connected to the asterisk |
| 13:01.21 | m_a_g_o | I thought the 79xxs only displayed content if configured through Cisco Call Manager, not with * |
| 13:01.28 | backblue | antiloopj: are you trying to trunk to another pbx with hfc cards? |
| 13:01.35 | antiloopj | backblue fyi, i have 2 HFC cards connected to the phone company (TE Mode), and 2 HFC cards connected to the PBX (NT mode) |
| 13:01.44 | *** join/#asterisk doughecka_ (n=Miranda@unaffiliated/doughecka) |
| 13:01.46 | backblue | m_a_g_o: i was asking. :) |
| 13:02.27 | m_a_g_o | backblue: I was hoping they worked : ( |
| 13:02.31 | backblue | antiloopj: i have a lot of the same. and works great, does you have a hangup()? |
| 13:02.44 | *** join/#asterisk pawal (n=pawal@212.247.14.36) |
| 13:03.29 | *** join/#asterisk oej (n=oej@apollo.webway.se) |
| 13:03.45 | simulated | h323 rox |
| 13:04.01 | simulated | only problem is that in *, it's lacking codecs |
| 13:04.01 | *** join/#asterisk simes (n=simes@80-235-142-253.cable.ubr14.nail.blueyonder.co.uk) |
| 13:04.04 | backblue | m_a_g_o: realtime information, its only possible you have some kind of comunication protocol, you cannot have a cisco phone getting xml from a server in 1sec to another. to many resources, i think CM and cisco voip phones, uses some protocol. |
| 13:04.16 | simulated | SCCP |
| 13:04.24 | backblue | h323 has no future |
| 13:04.49 | cpm | and has a dodgy past. |
| 13:04.52 | simulated | h323 and related familes h245 q931 h450 h225 can do just about eerything |
| 13:05.16 | backblue | something that sip dont? |
| 13:05.16 | *** join/#asterisk coppice (n=chatzill@210.22.134.146) |
| 13:05.17 | UdontKnow | simulated: including give you a wrong bill |
| 13:05.27 | backblue | i dont know h323, sorry about my questions |
| 13:05.27 | *** part/#asterisk kimc (n=freenode@c-68-43-224-10.hsd1.mi.comcast.net) |
| 13:05.36 | simulated | backblue h323 is the original voip signaling protocol spec |
| 13:05.46 | simulated | they are different |
| 13:05.49 | simulated | sip/h323 |
| 13:05.54 | UdontKnow | simulated: the biggest trouble my friends had with h323 was it left the "line open |
| 13:06.04 | UdontKnow | " and kept billing after both parties hung up |
| 13:06.06 | simulated | but h323 is quite a powerful signaling protocol, and not to be just thrown away |
| 13:06.09 | cpm | right, sip is broadly supported, h323 has been idling along for years. |
| 13:06.27 | antiloopj | backblue i have the macro hangupcall (i'm using freepbx/amp stuff to configure) |
| 13:06.29 | simulated | i dont know if you've ever used a cisco voip gateway |
| 13:06.32 | cpm | it's nice stuff though, if you have bucketloads of bandwidth |
| 13:06.56 | simulated | i started voip in late 98 with cisco as5300's |
| 13:06.57 | backblue | antiloopj: hoo, sorry i dont know amp or whatever, i dot it by hand. it works everything great! |
| 13:07.03 | simulated | h323 was the thing back then |
| 13:07.06 | antiloopj | backblue pbx <> sip client stuff hangsup like it should |
| 13:07.38 | YaP | i'm using asterisk with eicon 4 bri but during a call there is a lot of noise, what could be the problem? |
| 13:07.58 | m_a_g_o | http://www.voip-info.org/tiki-index.php?page=Asterisk+Cisco+79xx+XML+Services |
| 13:08.01 | YaP | i'm using chan-capi-cm 0.6.4 |
| 13:08.38 | simulated | echo? |
| 13:09.27 | simes | Hello. Could someone please give me some advice? I'm using asterisk 1.2.4, and in my sip.conf, I register with the sipgate.co.uk provider. I'm passing all incoming calls direct to a voicemail box. The box records for 50 seconds, but then I get a message (CSeq: 103 Bye) from sipgate, and the call cuts off. Even if I set the extension to just wait for 2 minutes, the call cuts off with the same message. I wonder if sipga |
| 13:09.27 | simes | ut then with another provider (voipfone), I get exactly the same thing. Has anyone else had this problem? |
| 13:11.28 | jorgito | how can i make the agent to be logget all the time ? |
| 13:12.48 | *** join/#asterisk scanna (n=scannach@81-174-16-211.f5.ngi.it) |
| 13:12.51 | *** join/#asterisk tdonahue (n=tdonahue@208.51.101.201) |
| 13:12.57 | *** join/#asterisk [TK]D-Fender (n=joe@toronto-HSE-ppp4122655.sympatico.ca) |
| 13:13.25 | stock | hmm speakin of cisco... i have a couple of 'old' cisco IP Phones 12 SP+ .. can they made be of good use? |
| 13:13.35 | simulated | stock yes |
| 13:13.44 | simulated | with some patience and hax0ring |
| 13:13.44 | stock | oh great! |
| 13:13.48 | simulated | you can get them to work with * |
| 13:13.59 | stock | right |
| 13:14.10 | simulated | youll have to hex edit config files ;) |
| 13:14.16 | cpm | As an aside, just in case anyone cares, I am not pleased with the S100-FX v2, would not recommend it. |
| 13:14.44 | stock | well i asked about this before... and they said you would need to reflash them |
| 13:17.03 | RoyK | anyone here ever setup gnugk? |
| 13:17.47 | simulated | ive set it up... getting it to work proper no |
| 13:18.00 | [TK]D-Fender | cpm : Thats the "new" IAXY? |
| 13:18.50 | cpm | That's the box that is what we all wished the iaxy was. |
| 13:18.59 | jontow | reflash? jeez.. not quite.. they do load their configs via a TFTP server, though.. |
| 13:19.01 | cpm | but the sound quality is awful. |
| 13:19.22 | jontow | they're a bear the first time you get 'em working, and some strange things happen that you'll want to be able to track down -- but they work quite well, overall |
| 13:19.23 | cpm | calls take a long time to set up (vs the iaxy) and sound like crap (vs the iaxy) |
| 13:19.40 | stock | simulated any good doc on that ? |
| 13:19.48 | cpm | http://www.x100p.com/products_2.htm |
| 13:19.49 | jontow | i would have to say: try chan_sccp vs. the in-tree chan_skinny, though. |
| 13:20.04 | jontow | stock; i had a writeup once.. let me see if i can get you a quick run-down |
| 13:20.22 | jontow | rather, i did the write-up :) |
| 13:20.40 | *** join/#asterisk trelane` (n=trelane@mail.allthingsit.com) |
| 13:21.04 | stock | wow , well sounds like it was quite a hack then ;) |
| 13:21.07 | *** join/#asterisk jsharp (n=jsharp@65.88.255.245) |
| 13:21.43 | stock | a friend, CCIE, told me it wasn't possible |
| 13:22.31 | jontow | he's fulla shit |
| 13:22.38 | jontow | don't believe every certification you find |
| 13:23.20 | jontow | http://secure-bsd.be/~jontow/12sp.html |
| 13:23.28 | jontow | theres the rundown, values you need to edit, and means to calculate them in bold. |
| 13:23.41 | *** join/#asterisk bweschke (n=bweschke@37.sub-70-192-21.myvzw.com) |
| 13:23.49 | jontow | and, fwiw, i can throw my SEPDefault.cnf in the same spot |
| 13:24.54 | stock | oh ic, you need to hex insert your ip number |
| 13:25.04 | jontow | exactly |
| 13:26.08 | jontow | in that example, iirc, its 192.2.2.168, or somesuch nonsense |
| 13:26.31 | stock | ok! |
| 13:26.41 | UdontKnow | jontow: 192.2? wrong example, unless you own an 192.2 network |
| 13:26.53 | jontow | oh, now.. 192.168.2.1 -- the example shows wrong |
| 13:26.56 | jontow | let me get the correct one up |
| 13:27.02 | stock | normally the cisco ccm hands these out |
| 13:27.11 | jontow | yes it does, it build the config file |
| 13:27.13 | jontow | asterisk doesn't |
| 13:27.47 | *** join/#asterisk tzanger (n=tzanger@mixdown.ca) |
| 13:28.02 | jontow | there, reload the doc |
| 13:28.03 | jontow | :) |
| 13:28.04 | jontow | my bad |
| 13:28.19 | jontow | hexdump didn't behave like i thought it would |
| 13:28.23 | jontow | but 'hexedit' works well |
| 13:29.22 | jontow | took quite a while to figure that out; nobody had any info at the time i was doing it :) |
| 13:29.23 | stock | sure.. well , if this works i will laugh my balls off ;)) "hey dude, mr CCIE, do i have news for you!" :) |
| 13:29.36 | jontow | i have one on my desk at home.. 30 VIP / 12SP+ |
| 13:29.38 | jontow | and one here at work |
| 13:29.44 | jontow | and i had it working with 1.0.1 |
| 13:29.44 | jontow | hehe |
| 13:31.10 | *** join/#asterisk Bambr (n=Bambr@213-35-233-174-dsl.end.estpak.ee) |
| 13:33.11 | jontow | honestly, i just created a bunch of SEPDefault.cnf's .. one for each asterisk server i'd like to register to, put them in /tftpboot like this: SEPDefault.cnf.192.168.2.1, etc.. then symlink the one you need at the moment to SEPDefault.cnf |
| 13:33.34 | stock | oh here's the wiki page also http://www.voip-info.org/wiki/view/Configuring+Cisco+12SP+phones+with+Asterisk |
| 13:33.51 | jontow | you also need to create a SEPDefault.cnf that is instead named: SEPmacaddr.cnf |
| 13:33.59 | jontow | once again, i just symlinked it to SEPDefault.cnf |
| 13:34.10 | stock | ah ok |
| 13:34.13 | jontow | lrwxrwxrwx 1 root root 14 Oct 18 11:36 SEP00D0BA03C001.cnf -> SEPDefault.cnf |
| 13:34.22 | jontow | to configure the phone, i believe you hit ** |
| 13:34.27 | jontow | or similar, thats documented on the wiki.. |
| 13:37.00 | simes | Hello again. Does anyone have any idea about this voicemail cut off? I've just created an account with "voipuser" and mapped a PSTN number direct to my asterisk box. Exactly the same problem. 50 second recording, then cut off. |
| 13:39.02 | stock | hmm jontow i see that the latest asterisk versions use SEPDefault.xml... so these versions cannot work with the 12 SP+ ? |
| 13:40.51 | antiloopj | simes can you receive at all phonecalls with a duration longer than 50 seconds? |
| 13:42.47 | jontow | stock; i've never had .xml working -- .cnf is what i've used successfully |
| 13:42.59 | x86 | anyway to make an extension park a call, do something, and when finished doing something retrieve the parked call? |
| 13:43.08 | jontow | http://secure-bsd.be/~jontow/SEPDefault.cnf |
| 13:43.08 | simes | yes, it seems that if there is speech in both directions, the connection stays alive, but if asterisk is not outputting any sound the remote side drops. I've tried three providers now, and all do the same :-( However, if I open a softphone on my workstation and connect the call to that, I can talk for 30 minutes, no problems. |
| 13:43.55 | stock | ok, but thats tied to a a specific asterisk version? like asterisk 1.2.x and newer only work with SEPDefault.xml ? |
| 13:44.02 | jontow | no.. |
| 13:44.08 | jontow | SEPDefault.* is never read by asterisk |
| 13:44.14 | jontow | only by the phone, which grabs it via TFTP |
| 13:44.18 | stock | ic |
| 13:44.36 | stock | its a feature of the newer cisco phones? |
| 13:44.52 | antiloopj | simes behind a NAT? |
| 13:45.10 | jontow | its a feature of almost all cisco phones.. and the 12SP+ is an oldass phone |
| 13:46.13 | simes | tried both - on my firewall itself and on a box behind it. Both do the same. |
| 13:46.40 | *** join/#asterisk Druken (n=druken@CPE00121716da99-CM00159a090acc.cpe.net.cable.rogers.com) |
| 13:46.52 | Druken | morning peoples |
| 13:48.17 | stock | moin |
| 13:49.16 | Druken | who wants to build me a new website for free?? :) i'll give asterisk support.. heheh |
| 13:50.15 | antiloopj | simes who hangs up your asterisk of the voipprovider? |
| 13:50.58 | simes | the "103 Bye" message appears to come from all of the providers. |
| 13:52.56 | simes | ...and I get a standard message in the console saying "user hung up", even though I didn't! |
| 13:57.36 | *** join/#asterisk fugitivo (n=ajf@201.255.179.22) |
| 13:57.53 | fugitivo | hello |
| 13:58.32 | *** join/#asterisk jaiger (n=jaiger@c-67-165-4-34.hsd1.ct.comcast.net) |
| 14:00.07 | *** join/#asterisk jimmy_deanPB (n=jhodapp@indianalifesciences.com) |
| 14:00.42 | *** join/#asterisk Druken (n=druken@CPE00121716da99-CM00159a090acc.cpe.net.cable.rogers.com) |
| 14:02.52 | *** join/#asterisk Dr-Linux (n=Nothing@202.125.141.6) |
| 14:03.09 | Dr-Linux | i'm facing a strange problem with a queue |
| 14:03.17 | Dr-Linux | can anybody discuss with me? |
| 14:03.41 | fugitivo | what problem? |
| 14:04.01 | *** join/#asterisk pv2b (n=pvz@c80-216-45-134.cm-upc.chello.se) |
| 14:04.34 | Dr-Linux | fugitivo: i have 2 queues fc-sales and tc-sales , both are same in queueus.conf |
| 14:04.42 | Dr-Linux | and same in extensions.conf |
| 14:04.54 | Dr-Linux | but both behavior is kinda different :S |
| 14:04.57 | *** join/#asterisk jeffik (n=Jeff@CPE0050babf4cd6-CM014350000760.cpe.net.cable.rogers.com) |
| 14:05.06 | fugitivo | for example? |
| 14:05.43 | x86 | anyway to make an extension park a call, do something, and when finished doing something retrieve the parked call? |
| 14:06.01 | Dr-Linux | fugitivo: if the caller goes to tc-sales queue, he/she listen rings, while a phone is ringing.. and after 20 seconds he gets voicemail as defined |
| 14:06.10 | *** join/#asterisk Mike9 (n=sturdee@ireland.pathwaynet.com) |
| 14:06.48 | Dr-Linux | but in fc-sales, caller listen music, and only music and i see on CLI, the particular phone is rining ... |
| 14:06.52 | Dr-Linux | and never time out |
| 14:07.08 | Dr-Linux | even both queue have same members (2 extensions) |
| 14:07.46 | fourcheeze | is there a rapid double beep anywhere in asterisk sounds - like a "BIBIP" |
| 14:08.04 | Dr-Linux | fugitivo: can you understand something :S what could be happend :S |
| 14:08.20 | Dr-Linux | even both members have same phone :S |
| 14:08.33 | Dr-Linux | everything is same .. but behave different :S |
| 14:09.15 | Dr-Linux | fugitivo: would you like to see my queues.conf or anything? :S |
| 14:09.16 | *** join/#asterisk octothorpe_ (n=octothor@198.60.73.230) |
| 14:09.32 | *** join/#asterisk tamp4x (n=Lab@64.201.13.170) |
| 14:09.43 | *** part/#asterisk octothorpe_ (n=octothor@198.60.73.230) |
| 14:10.35 | *** join/#asterisk f^bar (n=octothor@198.60.73.230) |
| 14:11.23 | f^bar | ~commands |
| 14:11.34 | Druken | uhmm.... |
| 14:13.16 | Enth | heh |
| 14:14.06 | x86 | w00t |
| 14:14.30 | Enth | Anyone know how to add a personalised greeting to Voicemail? |
| 14:14.36 | Enth | James! |
| 14:15.02 | Enth | sup man |
| 14:15.08 | x86 | i got my asterisk to make a phantom extension that runs an application, downloads the latest weather forcast from NOAA, trims out all the extra crap, passes it into Festival which speaks it out, then plays it back to the user |
| 14:15.34 | Druken | x86: care to share? :) |
| 14:16.15 | Enth | sweet. |
| 14:17.12 | jsharp | I had something similar. It generated an outbound call when/if the system got a severe weather watch/warning off my NOAAPORT feed. |
| 14:17.31 | Enth | fn~Druken: how to add a personalised greeting to Voicemail? Option 3 for Advanced Settings doesnt have the option to record a new greeting for each user. |
| 14:18.06 | Druken | Enth: each user has to record their own greeting |
| 14:18.24 | Enth | yes but how? |
| 14:18.58 | Druken | option 0 |
| 14:19.10 | [TK]D-Fender | Enth : Try option "0" (mailbox options) ..... its kinda obvious... |
| 14:19.10 | Enth | oh |
| 14:19.26 | Enth | I really need to wake up. |
| 14:19.27 | *** join/#asterisk Fedoracore6 (n=deddd@60.50.141.168) |
| 14:19.53 | Druken | how old are ya now? 13? :( |
| 14:19.56 | Druken | :) |
| 14:19.57 | Enth | lol |
| 14:19.58 | Enth | 27 |
| 14:20.05 | Fedoracore6 | hai all |
| 14:20.31 | Enth | sup Fedora |
| 14:20.32 | jsharp | Nah. If he was 13, he'd be hanging out in #gentoo |
| 14:20.38 | Enth | lol |
| 14:21.04 | *** join/#asterisk Rhizome (n=rhizome@149.9.0.21) |
| 14:22.02 | f^bar | ~command |
| 14:22.06 | Druken | slack :) |
| 14:22.27 | Enth | I do have a slack running somewhere. |
| 14:22.42 | *** part/#asterisk f^bar (n=octothor@198.60.73.230) |
| 14:22.42 | Enth | and CentOS. |
| 14:23.20 | Enth | but i hardly get time to mess with systems as I'm busy with Cisco/Nortel tools |
| 14:23.25 | *** join/#asterisk octothorpe_ (n=octothor@198.60.73.230) |
| 14:23.40 | Enth | and the fact that I manage a team doesnt give me time. |
| 14:23.51 | *** join/#asterisk mjmac (n=mjmac@pdpc/supporter/active/mjmac) |
| 14:24.04 | Enth | why would I need to brag? |
| 14:24.04 | Enth | heh |
| 14:24.07 | Enth | christ. |
| 14:24.12 | Druken | :) |
| 14:24.31 | Enth | anyway, forget the above. |
| 14:24.46 | Enth | :) |
| 14:25.01 | Enth | soooooooooo, what's new |
| 14:25.47 | *** join/#asterisk Samoied (n=Samoied@popeye.opens.com.br) |
| 14:25.59 | Druken | nuttin |
| 14:26.07 | Druken | still bored... at home |
| 14:26.08 | Enth | lol |
| 14:26.13 | Enth | no work today? |
| 14:26.23 | Druken | work everyday... just office at home |
| 14:26.27 | Enth | I took a week off. really tired. |
| 14:26.58 | Druken | hehe need a vacation from the vacation eh? |
| 14:27.01 | fugitivo | anyone using the Z-PLEX-10-24S? |
| 14:27.48 | Enth | Drunken: yeah. It's been a while. Might consider going out of town but not sure where. |
| 14:28.02 | *** join/#asterisk PakiPenguin (n=uppal@linuxpakistan/admin/pakipenguin) |
| 14:28.54 | *** join/#asterisk lalune (n=lalune@pc-205-21.scpe.quickclic.net) |
| 14:29.50 | *** join/#asterisk Hmmhesays (n=Neg@72.24.227.83) |
| 14:30.06 | Hmmhesays | morning |
| 14:31.05 | *** join/#asterisk bigbootay (n=bigboota@209.250.156.214) |
| 14:32.06 | Druken | Enth: does it matter where?? :) ROAD TRIP!!!! |
| 14:32.26 | *** join/#asterisk fu3 (n=kaa@234-200-29-134.hcc.mnscu.edu) |
| 14:32.33 | Hmmhesays | i love road trips |
| 14:32.37 | Hmmhesays | specially back country style |
| 14:32.49 | Hmmhesays | loud tunes a case of bud |
| 14:33.25 | fu3 | fuck ya |
| 14:33.30 | fu3 | we called those "road cappers" back in the dya |
| 14:33.34 | fu3 | back in my day anyway |
| 14:33.51 | *** join/#asterisk anthm (n=anthmct@CPE-69-76-83-52.wi.res.rr.com) |
| 14:33.51 | *** mode/#asterisk [+o anthm] by ChanServ |
| 14:33.52 | fu3 | "woop! Another cherry Boys!" |
| 14:33.57 | fu3 | and then crack that beer |
| 14:34.04 | [TK]D-Fender | Yeah but its so much safer now without the swooping pteradactyls! ;) |
| 14:34.07 | bigbootay | Hmmhesays, you forgot the 13 year old girls. |
| 14:34.29 | bigbootay | at least in saskatchewan :) |
| 14:34.40 | fu3 | god I miss Canada :) |
| 14:34.46 | vgster | me too |
| 14:35.12 | bigbootay | you bet. in Sask, the 13 yos _bring_ the beer. |
| 14:35.23 | fu3 | thats the best thing ive heard all week |
| 14:35.28 | *** join/#asterisk docelm0 (n=docelmo@66.239.192.34.ptr.us.xo.net) |
| 14:35.45 | fu3 | I' |
| 14:35.58 | fu3 | i'll be heading up to Regina this summer |
| 14:36.13 | fu3 | we're gonna make it look like saskatoon! |
| 14:36.20 | Hmmhesays | wow the new guns n roses songs suck |
| 14:37.02 | Hmmhesays | gnr without slash is no gnr at all |
| 14:37.24 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 14:38.19 | RoyK | ~seen zoa |
| 14:38.27 | jbot | zoa <n=kkk@pirus.securax.be> was last seen on IRC in channel #asterisk, 13d 20h 31m 2s ago, saying: '1000 interrupts per second are ok'. |
| 14:38.45 | RoyK | ~lart himself |
| 14:38.51 | fugitivo | anyone using the Z-PLEX-10-24S channel bank? |
| 14:39.11 | jsharp | mmmm krispy kreme. |
| 14:39.12 | fu3 | thats great |
| 14:39.15 | *** join/#asterisk __chris (n=chris@unaffiliated/redlined) |
| 14:39.16 | Enth | Drunken: yeah road trip. Hopefully I'll bump into some hot women. |
| 14:39.16 | fu3 | zoa's last words were said to me |
| 14:39.24 | Enth | then again my gf wouldnt like that :/ |
| 14:39.26 | *** join/#asterisk htims (n=pd@Vf44a.v.pppool.de) |
| 14:39.52 | *** join/#asterisk anderiv (n=anderiv@207-67-87-34.gen.twtelecom.net) |
| 14:40.02 | jsharp | threesomes. |
| 14:40.03 | jsharp | er |
| 14:40.04 | Druken | Enth: i love downtown toronto for that :) |
| 14:40.09 | Enth | hh |
| 14:40.12 | Enth | hehe |
| 14:40.14 | Druken | even in the winter the bitches dress like hoes |
| 14:40.18 | Enth | lol |
| 14:40.51 | Enth | hehe |
| 14:41.15 | fu3 | I remember being snowed in for an entire week in TO. |
| 14:41.19 | Druken | hehe a MOtel |
| 14:41.48 | fu3 | I stayed at some hotel on young street.. Delta Chelsea if I remember correctly. |
| 14:41.58 | fu3 | man there were a lot of prostitutes |
| 14:42.26 | fu3 | I remember cracking up at all the ad's.. it was like being in vegas |
| 14:42.31 | Enth | hah |
| 14:43.06 | Enth | "Foor a good time, call 555-555-555" and in small font "Voice Powered by Asterisk" |
| 14:43.12 | Enth | for/foor |
| 14:43.28 | Curus | Now there's an application for festival... |
| 14:43.30 | x86 | hahaha |
| 14:43.30 | jsharp | App_PhoneSex |
| 14:43.35 | x86 | rofl |
| 14:43.46 | Enth | hehe |
| 14:43.48 | fourcheeze | chan_penis.so |
| 14:43.57 | Enth | callgirl.so |
| 14:44.12 | Druken | hehe |
| 14:44.15 | x86 | i thought of a cool name for a softphone... GhettoSip v2.0 -- Now includes brown bag! |
| 14:44.20 | tzanger | hahaha |
| 14:44.21 | [TK]D-Fender | Enth : Thats like my "Jenny Bangs" <514> 867-5309 script ;) |
| 14:44.31 | Enth | Actually, imagine if someone targetted sex lines and gave them a cheaper voice solutionm powered by Asterisk. |
| 14:44.32 | tzanger | GhettoSIP -- twine and cans not included |
| 14:44.34 | Druken | i wanted todo a santa clause voice activated system.... |
| 14:44.42 | Enth | he/she could make a few $$$ |
| 14:44.48 | Druken | i thought it'd be cool |
| 14:44.50 | jsharp | Enth: I considered it, but my wife said no. |
| 14:45.02 | *** join/#asterisk devel (n=devel@wiggum.digitalcoven.com) |
| 14:45.07 | Enth | well |
| 14:45.11 | x86 | hmm |
| 14:45.13 | Druken | [TK]D-Fender: jenny bangs? |
| 14:45.15 | Enth | ask her if she wants some of the profits |
| 14:45.21 | Enth | she wont say no to a handbag. |
| 14:45.26 | Enth | shoes etc |
| 14:45.36 | x86 | i always like spoofing my CID to Johnny Blaze ;) |
| 14:45.39 | Curus | I wonder if there are any providers which specialise in 90x-numbers |
| 14:45.51 | Enth | Curus: I'm sure there are. |
| 14:45.58 | jsharp | Enth: Nah. I'd have to convince her I'd buy her a new Mac with the profits. |
| 14:46.06 | Enth | Ebay it |
| 14:46.12 | Enth | she wont know the difference ;) |
| 14:46.38 | *** part/#asterisk lalune (n=lalune@pc-205-21.scpe.quickclic.net) |
| 14:46.44 | Enth | hrmmm speaking of Asterisk... |
| 14:46.46 | [TK]D-Fender | Druken : Well.. she never had a last name, just thing about the phone # ;) |
| 14:46.50 | [TK]D-Fender | think* |
| 14:47.12 | Enth | anyone used that before? |
| 14:47.20 | octothorpe_ | yes |
| 14:47.22 | *** join/#asterisk bartpbx (n=bartpbx@p54B00658.dip0.t-ipconnect.de) |
| 14:47.24 | Enth | ideal for hot desking |
| 14:47.28 | Druken | [TK]D-Fender: i must be slow today... i dun get it... :) |
| 14:47.32 | bartpbx | hello. |
| 14:47.36 | octothorpe_ | hot desking? |
| 14:47.47 | Enth | yeah |
| 14:48.11 | octothorpe_ | what is "hot desking" |
| 14:48.30 | Enth | working between different sites/locations of the same office or another but assigned to one desk. |
| 14:48.34 | [TK]D-Fender | Druken : Google it.... |
| 14:48.44 | Druken | google the phone number? |
| 14:48.44 | Enth | thus, yuor voice port is not static but your extension number is. |
| 14:48.44 | bartpbx | how can i get the codec used for a call? I have a deadAgi Applikation and would like to know which codec was uses |
| 14:49.00 | tzanger | octothorpe_: you come in and grab any available desk. you have no desk of your own, just one of a common "pool" |
| 14:49.07 | Enth | yeah |
| 14:49.11 | octothorpe_ | enth: yes, that is what I use it for |
| 14:49.36 | Enth | You can even work at home and still be connected to the corporate telephony network and receive calls as you would at work. |
| 14:49.48 | [TK]D-Fender | Druken : YES |
| 14:49.52 | Enth | hence "hot desking" or in some cases "flexible working" |
| 14:50.20 | tzanger | hot desking is just another way to make you feel like a cog in a big machine. you are replacable |
| 14:50.25 | Enth | However, does anyone have any idea of the limitations of Asterisk's Find me Follow me service? |
| 14:50.40 | Enth | tzanger: lol yeah |
| 14:50.47 | Dr-Linux | Druken: did you findout something? |
| 14:50.48 | [TK]D-Fender | Druken : http://www.letssingit.com/?/tommy-tutone-8675309-jenny-vcs7dgp.html |
| 14:50.56 | Druken | [TK]D-Fender: tommy tutone is a bit before my time... |
| 14:51.04 | x86 | w00t! |
| 14:51.05 | Hmmhesays | Enth: what type of limitations, you can make your dialplan do just about anything |
| 14:51.14 | x86 | my SIP hard phone will be here today! |
| 14:51.20 | Dr-Linux | i just want the queue member should get right, and caller should also listen the being ring |
| 14:51.39 | *** join/#asterisk mocker (n=ks@mocker.org) |
| 14:51.53 | [TK]D-Fender | x86 : We're all entitled to our first mistakes... the rest you need to apply for credit ;) |
| 14:52.02 | mocker | Can a user in sip.conf have multiple contexts? |
| 14:52.12 | x86 | [TK]D-Fender: heh |
| 14:52.13 | Hmmhesays | mocker: yesh |
| 14:52.26 | x86 | [TK]D-Fender: i think for as little as i'll use it, it'll be OK :P |
| 14:52.27 | mocker | Hmmhesays: Just multiple lines, or comman delimeted? |
| 14:52.31 | Enth | Hmmhesays: Well, like can one say incorporate the ability for the user to "register" his current location and "pull" calls through? |
| 14:52.32 | mocker | +spelling |
| 14:52.51 | Hmmhesays | Enth define "pull calls throuh" |
| 14:52.59 | Hmmhesays | mocker comma delimited if I remember correctly |
| 14:53.09 | mocker | Hmmhesays: Thanks. |
| 14:53.34 | Druken | [TK]D-Fender: http://packetstorm.linuxsecurity.com/wardialers/Jenny.htm |
| 14:53.57 | Druken | someone called every areacode... hehehe people with too much time |
| 14:54.20 | Enth | Well, let's say at 9 am your at work, your calls come through to your umm Cisco 7960, then an hour later, you have an urgent meeting and are mobile, and forgot to divert calls to mobile, so can one actually "dial in" his/her asterisk box and define where he/she wants the calls routed to? |
| 14:54.41 | jsharp | remote call forwarding |
| 14:54.56 | Enth | how does it work? |
| 14:55.02 | jsharp | Wouldn't be that hard to implement. |
| 14:55.11 | Enth | true. |
| 14:55.12 | Hmmhesays | so we had a serious rock out session on saturday, damn that was fun |
| 14:55.42 | tzanger | Hmmhesays: oh? |
| 14:55.54 | Enth | Anyone like Green Tea? :P |
| 14:56.04 | jsharp | I'd do it through some DBPut/DBGet calls. |
| 14:56.16 | Hmmhesays | tzanger yeah, band practice is usually a good time |
| 14:56.17 | Druken | i have remote call forwarding |
| 14:56.22 | Hmmhesays | especially when I odn't have a hangover |
| 14:56.22 | Druken | it's just done with the database |
| 14:56.23 | tzanger | Hmmhesays: always |
| 14:56.29 | Enth | ahh |
| 14:56.45 | Enth | yes but that would mean pre-set numbers |
| 14:56.53 | jsharp | Not at all. |
| 14:57.02 | Enth | what if you say suddenly happen to be at a client's office? |
| 14:57.03 | Druken | nah... you can use a read |
| 14:57.10 | Druken | or even use the CID of the incoming call |
| 14:57.57 | jsharp | read numbers, stuff into database with DBGet. |
| 14:58.01 | jsharp | Er, DBPut. |
| 14:58.09 | *** join/#asterisk kpettit (n=keith@69.15.174.113) |
| 14:58.19 | Druken | or odbcput/get :) |
| 14:58.51 | Enth | so you're saying that it is possible to even get calls through even if the number is not in the database? |
| 14:59.10 | Enth | Like even to a payphone? |
| 15:00.21 | Hmmhesays | the db stuff doesn't work so hot if you're using multiple asterisk boxes in a cluster style formation |
| 15:00.47 | Enth | hrmmm |
| 15:02.34 | *** join/#asterisk mhnoyes (n=mhnoyes@user-2ivfila.dialup.mindspring.com) |
| 15:02.36 | Hmmhesays | ah yes i'm all lit up again, on the couch, in my bed |
| 15:02.40 | Druken | Hmmhesays: why not? if your using an external database.... |
| 15:03.04 | Hmmhesays | and how would you go about doing that? |
| 15:03.40 | jsharp | Enth: With the appropriate dialplan magic, you can remote call forward your extension to any number your * box can dial. |
| 15:04.06 | Hmmhesays | yes you can even do cascading dial, I just set that up for a client |
| 15:04.26 | Hmmhesays | it dials all the cheapest endpoints first, then on to the pstn while the original calls are ringing |
| 15:04.28 | Enth | neat. |
| 15:05.56 | jsharp | I have my extension here at the office to ring on 2 7940s at the office, my ATA at the house, and my cell phone...all simultaneously. |
| 15:06.11 | jsharp | So whichever one I answer gets the call. |
| 15:06.17 | Enth | ye sbut that's pre-set |
| 15:06.31 | Enth | what if you suddenly want it to ring umm say your gf's mobile? |
| 15:06.40 | *** join/#asterisk zeedo (n=zeedo@reboot-robot.net) |
| 15:06.49 | jsharp | That would take some dialplan voodoo. |
| 15:06.51 | Hmmhesays | then you set your dialplan to do so |
| 15:06.54 | Enth | ahhh |
| 15:07.00 | __chris | any idea how I can sort my cisco 7940 out - people are complaining I'm too quiet on the phone, shouting and they can't hear very well, transferred to another internal extension (same phone model) and it seems ok - have tried increasing rx/txgain to no avail |
| 15:07.02 | Enth | exactly |
| 15:07.32 | jsharp | But I could also set it with the dialplan so that if there's a remote call forwarding set, then ring to that number...otherwise, ring to all the other numbers. |
| 15:07.36 | Enth | __chris: what firmware? |
| 15:07.48 | Enth | ah. |
| 15:08.01 | __chris | sip 7.5 |
| 15:08.56 | Hmmhesays | heh, trying to undercut us huh? |
| 15:08.59 | Enth | __chris: hrmmm, that's the current version too. Can you unplug the phone and plug it elsewhere? |
| 15:09.04 | Hmmhesays | the wiki is just fine |
| 15:09.12 | *** join/#asterisk sysdebug (n=sysdebug@200.250.222.8) |
| 15:09.14 | Enth | lol |
| 15:09.16 | Enth | well |
| 15:09.23 | Enth | I'll be happy to put them anywhere. |
| 15:09.29 | Hmmhesays | you spoon feed people dialplans and they'll never earn how they actually work, then we'll get more asinine questions in here |
| 15:09.36 | Hmmhesays | *never learn |
| 15:09.36 | Enth | good point. |
| 15:09.42 | simes | does anyone else use a SIP provider to deliver a PSTN number to their asterisk box? |
| 15:10.00 | Hmmhesays | yes |
| 15:10.10 | xtrvd | Hmmhesays: I went from not knowning anything to learning off of other people's dialplans. |
| 15:10.12 | __chris | plug in phone elsewhere, yep can do - to narrow out what though, cables? |
| 15:10.27 | Hmmhesays | xtrvd: probably partial dialpans though |
| 15:10.33 | xtrvd | Of course. |
| 15:10.35 | Fedoracore6 | yes... |
| 15:10.37 | Enth | __chris: yes and the port itself. |
| 15:10.40 | *** part/#asterisk mhnoyes (n=mhnoyes@user-2ivfila.dialup.mindspring.com) |
| 15:10.42 | Hmmhesays | putting full examples on the wiki is useless |
| 15:10.58 | Fedoracore6 | i .... get connect to data base |
| 15:10.59 | xtrvd | I'll agree there, =) |
| 15:11.00 | Fedoracore6 | CLI> realtime mysql status |
| 15:11.00 | Fedoracore6 | Connected to ivr@localhost, port 3306 with username edifier for 33 seconds. |
| 15:11.07 | Fedoracore6 | yes.... i happay |
| 15:11.10 | Fedoracore6 | wakakka 5 day |
| 15:11.12 | jorgito | what is the meaning of local extension? |
| 15:11.45 | Hmmhesays | thats a wiki question if i've ever heard one ;) |
| 15:11.57 | simes | for those that do use a SIP provider to deliver a PSTN number to their *, if the line diverts to your * voicemail, do you get the 50 second cut off "bug" that I'm getting? |
| 15:12.00 | __chris | ok will do - is there anyway I can alter the gain per phone via telnet or the conf file per mac address? |
| 15:12.36 | Enth | __chris: One the phone itself or on the switch or Asterisk? |
| 15:12.43 | jaiger | __chris, my polycom phones allow me to change their gains |
| 15:12.59 | *** join/#asterisk fulgas (n=fulgas@209.8.233.208) |
| 15:13.26 | __chris | ive changed gain in * but that doesn't seem to have had much affect, hence id like to change on the 7940 itself |
| 15:13.49 | Enth | Well, I'm sure you're aware of how Cisco operate. :) |
| 15:14.10 | Enth | the answer is yes, you can change the gain on the 7940 but it's a slight pain. |
| 15:14.32 | *** join/#asterisk Darwin35 (n=Darwin@sta-208-139-193-162.rockynet.com) |
| 15:14.44 | Darwin35 | Morning |
| 15:14.46 | Druken | Enth: is it a pain in the rectum? |
| 15:15.08 | Enth | __chris: I'd start by first plugging this 7940 on another port, preferably on the same port where the other 7940 works well. |
| 15:15.21 | Enth | Drunken: hah yeah. |
| 15:15.35 | simes | can someone volunteer to ring me and try to leave a message of over 50 seconds? |
| 15:16.49 | *** join/#asterisk marv[work] (n=timr@64.89.118.139) |
| 15:17.08 | Druken | Enth: i had them call and offer me a volume discount on it.. :) |
| 15:17.13 | Enth | apart from X-lite, any really good sofphones for Linux out there? |
| 15:17.30 | Enth | Drunken: how much? |
| 15:17.50 | Enth | but Eyebeam doesnt work on Linux. |
| 15:18.02 | Druken | i think it was down to like 30 a license |
| 15:18.07 | Enth | damn. |
| 15:18.11 | Enth | hrmmm |
| 15:18.22 | Druken | but they wanted 10,000 licences.. hehehe |
| 15:18.34 | simes | anyone... call me IP to IP? I need to establish if this problem only occurs through SIP providers? |
| 15:18.36 | Enth | if I could code well, I'd have built a really cool softphone for Linux |
| 15:18.46 | Enth | lol 10k licenses. |
| 15:19.40 | *** join/#asterisk mover (n=dlu@gw-dus-net.dus.de.ncore.net) |
| 15:19.41 | Enth | Cisco should at least consider making their old softphone code opensource. |
| 15:19.49 | *** join/#asterisk MikeJ[Laptop] (n=vircuser@d14-69-8-30.try.wideopenwest.com) |
| 15:20.02 | Enth | and I bet one of you code gurus would come up with a good one :) |
| 15:20.22 | mover | hi |
| 15:20.43 | Enth | simes: I would but my * is down for maintenance. |
| 15:20.45 | mover | it is possible to limit simultaneously used channels via iax2? |
| 15:21.10 | Druken | i belive so |
| 15:21.34 | Druken | same as in sip... incominglimit= and outgoinglimit= |
| 15:21.47 | simes | i only need one person out of the 259 on the channel... anyone? |
| 15:22.07 | [TK]D-Fender | simes : Pastebin your sip.conf |
| 15:22.18 | [TK]D-Fender | simes : I'll see if anything in there looks out of place. |
| 15:22.34 | Druken | simes, what do you need? |
| 15:22.46 | *** join/#asterisk epablo (n=epablo@WLL-24-pppoe203.t-net.net.ve) |
| 15:22.54 | *** join/#asterisk Katty (n=angela@64.82.232.54) |
| 15:22.57 | Katty | allo. |
| 15:23.02 | epablo | Hi people.. how's it going? |
| 15:23.18 | simes | Druken: I just want someone to call me (which will divert to voicemail) and try to leave a message over 50 seconds long. |
| 15:23.23 | fu3 | ola |
| 15:23.33 | Enth | epablo: Si. |
| 15:23.44 | Hmmhesays | simes why don't you do it? |
| 15:24.38 | simes | because I can't ring myself over IP. I can only ring into my box via a SIP provider (using PSTN) and that's when the problem occurs. |
| 15:24.51 | Hmmhesays | why can't you ring yourself over ip? |
| 15:25.02 | Enth | I bet it's a scam - premium rate over IP! |
| 15:25.04 | epablo | I'm having some problems ending a call in a T1. It comes from h323 in g729. I have the licenses but I get an error: Unable to find a codec translation path from g729 to unknown |
| 15:25.04 | Enth | hehe |
| 15:25.22 | mover | Druken: is sip its now call-limit |
| 15:25.41 | Darwin35 | asterisk+freebsd+te410 = rocking server |
| 15:25.49 | Enth | heh |
| 15:25.54 | Enth | fbsd 6? |
| 15:25.59 | Darwin35 | yep |
| 15:26.01 | Druken | mover: good to know... can ya tell i haven't upgraded in a while? |
| 15:26.03 | Enth | cool |
| 15:26.28 | Darwin35 | and we use realtime |
| 15:26.28 | [TK]D-Fender | epablo : G729 is a licensed codec that you have to pay for in *. |
| 15:26.40 | [TK]D-Fender | epablo : Which by default you have not. |
| 15:26.47 | epablo | [TK]D-Fender: Did so.. and registered the codec.. |
| 15:26.55 | epablo | 0/0 encoders/decoders of 24 licensed channels are currently in use |
| 15:27.39 | [TK]D-Fender | I'm guessing the 'UNKOWN" part is whats killing the call.... |
| 15:27.40 | *** join/#asterisk viLeR (i=1000@66.128.47.232) |
| 15:28.03 | Druken | probably |
| 15:28.10 | epablo | How do I force it to use g711 via zaptel.. |
| 15:28.12 | epablo | ? |
| 15:28.30 | [TK]D-Fender | epablo : "Unable to find a codec translation path from g729 to unknown" the other side isn't offering a codec list or something... |
| 15:28.53 | RoyK | kill -9 -1 |
| 15:28.54 | [TK]D-Fender | epablo : depends on how you set up both ends. What are on both ends of this failed call? |
| 15:29.12 | epablo | [TK]D-Fender: The other side is a T1 on a te4100 |
| 15:29.13 | *** join/#asterisk klasstek (n=nunyobiz@sta-206-168-96-89.rockynet.com) |
| 15:29.59 | Enth | Guys, isnt IAX2 used to connect to other * boxes? |
| 15:30.08 | Druken | generally |
| 15:30.18 | epablo | Enth: it's a good way to do it |
| 15:30.18 | UdontKnow | or to IAXy |
| 15:30.20 | Druken | there are a few devices that can speak it... |
| 15:30.42 | Enth | X-lite supports IAX right? |
| 15:30.49 | epablo | Nop |
| 15:30.54 | Enth | shit. |
| 15:30.56 | epablo | use g711 or ilbc |
| 15:31.11 | Enth | x-lite supports them? |
| 15:31.29 | epablo | the later goes easy on your bandwidth.. Yes x-lite supports them both |
| 15:31.40 | Enth | sweet. |
| 15:32.16 | Enth | so ideally ilbc is better for communicating with another * and plus using x-lite? |
| 15:32.36 | [TK]D-Fender | ILBC = ick.... |
| 15:32.51 | Enth | heh |
| 15:32.55 | epablo | Enth: I normally use g729 |
| 15:32.59 | *** join/#asterisk mtgh (n=chatzill@dsl093-001-038.det1.dsl.speakeasy.net) |
| 15:33.05 | epablo | but you have to pay the prices |
| 15:33.08 | Enth | yeah |
| 15:33.27 | Enth | so I guess g711, ilbc and iax2 are the options |
| 15:33.36 | Enth | *whistle* |
| 15:33.36 | [TK]D-Fender | IAX@ is not a codec... |
| 15:33.47 | *** join/#asterisk mikefoo (n=mikefoo@64.124.169.2) |
| 15:33.58 | Druken | ~voip-info |
| 15:34.00 | jbot | hmm... voip-info is the Voice Over IP wiki. It is a community resource which will answer all of your questions, from Asterisk to ZTDummy. You can find it over at http://www.voip-info.org - well worth bookmarking |
| 15:34.02 | Druken | bone up |
| 15:34.03 | Druken | :) |
| 15:34.19 | Enth | yes yes. |
| 15:34.22 | Enth | :P |
| 15:34.26 | *** join/#asterisk bweschke (n=bweschke@c-68-36-51-213.hsd1.nj.comcast.net) |
| 15:34.37 | Enth | I'll send dialplans there :P |
| 15:34.54 | Enth | mvuahaha! *thunder in background* |
| 15:34.58 | [TK]D-Fender | Enth : On X-Lite you have 2 real choices, G711 and GSM. if you aren't too worried about bandwidth G711, everything else = GSM. |
| 15:35.05 | Druken | i'm afraid to upgrade.. hehehe i'd have to redo all my god damn dialplans |
| 15:35.14 | *** join/#asterisk fu3 (n=kaa@234-200-29-134.hcc.mnscu.edu) |
| 15:35.26 | fu3 | anyone hear of a "ciruit pack" ? |
| 15:35.36 | fu3 | The telco says they ordered two to fix voltage issues here |
| 15:35.49 | Hmmhesays | hello to you too fu3 |
| 15:35.57 | fu3 | hi :) |
| 15:36.11 | fu3 | I was in here just like 5 mins ago.. I need to say hello every time? :) |
| 15:36.30 | Hmmhesays | fu3: simmer down now |
| 15:36.30 | Enth | fsck you. I was gonna say katana :( |
| 15:36.31 | *** join/#asterisk msw (n=msw@rdu-nat.rpath.com) |
| 15:36.39 | fu3 | im simmah'd |
| 15:36.45 | Hmmhesays | fu3: lol |
| 15:36.49 | fu3 | :) |
| 15:37.01 | GerbilWrk | has anyone ever seen this happen with a pri? http://pastebin.com/587029 |
| 15:37.04 | Hmmhesays | why do I have the sudden urge to play mortal kombat |
| 15:37.25 | Enth | hah |
| 15:37.37 | x86 | anyway to make an extension park a call, do something, and when finished doing something retrieve the parked call? |
| 15:37.43 | file | GerbilWrk: that's normal |
| 15:37.45 | jsharp | GerbilWrk: Yeah. Some times telco providers will restart idle spans. |
| 15:37.45 | [TK]D-Fender | GerbilWrk : Perfectly normal when the link comes up. |
| 15:38.00 | jsharp | Just to make sure everything is synced up and happy. |
| 15:38.07 | file | chan_zap restarts them periodically to make sure everything is happy |
| 15:38.15 | file | jsharp: bah |
| 15:38.27 | Katty | goodness. |
| 15:39.04 | GerbilWrk | even if it happens when calls are in progress? |
| 15:39.16 | *** join/#asterisk znoG (n=gs@109-130-89-200.fibertel.com.ar) |
| 15:39.50 | file | it won't reset the B channel of an active call |
| 15:40.18 | GerbilWrk | it has before |
| 15:40.56 | GerbilWrk | since i swapped the PCI slots and updated the Zaptel drivers, i haven't witnessed it hang up on a call |
| 15:41.02 | GerbilWrk | but i did see that just now so figured i'd check |
| 15:41.41 | *** join/#asterisk azzie (n=az@azzie.net) |
| 15:42.13 | *** join/#asterisk coolhp (n=crap@modemcable240.139-203-24.mc.videotron.ca) |
| 15:42.28 | fu3 | What do I know about Diamonds? |
| 15:42.38 | Enth | well |
| 15:42.41 | Enth | they are a girl's best friend |
| 15:42.58 | Enth | and a man's worst nightmare. |
| 15:43.01 | fu3 | haha |
| 15:43.07 | Hmmhesays | and all this time I thought it was peanut butter |
| 15:43.10 | Enth | specially if yuor gf happens to be hinting to get married for a long time. |
| 15:43.11 | fu3 | Is that the 2007 deBeers marketing lineup? |
| 15:43.19 | file | Hmmhesays: !!! |
| 15:43.21 | Enth | hah |
| 15:43.23 | Hmmhesays | file |
| 15:43.29 | file | hi. |
| 15:43.33 | Hmmhesays | hola |
| 15:43.38 | Enth | God I love being a man. |
| 15:43.42 | dgorski | GerbilWrk: resetinterval=never |
| 15:44.01 | dgorski | will stop the zap channel from restarting the channels |
| 15:44.03 | GerbilWrk | dgorski, in zaptel or zapata? |
| 15:44.05 | coolhp | Good day all. |
| 15:44.07 | dgorski | zapata |
| 15:44.10 | Fedoracore6 | hola |
| 15:44.22 | dgorski | no isdn settings in zaptel.conf |
| 15:45.07 | coolhp | Is anyone experiencing problems with Cisco 79XX phones and SIP ? I've got about 40 phones here and they seem to be loosing they registration on a regular basis. |
| 15:45.19 | *** join/#asterisk kamuix (n=kamuix@195.78.4.174) |
| 15:45.47 | dgorski | GerbilWrk: http://www.voip-info.org/wiki/index.php?page=Asterisk+config+zapata.conf |
| 15:46.01 | *** join/#asterisk htims (n=pd@Vf44a.v.pppool.de) |
| 15:46.05 | dgorski | GerbilWrk: look just under "ISDN PRI Switch Configuration" |
| 15:46.30 | dgorski | just above the isreal comments |
| 15:46.47 | *** join/#asterisk corruptor (n=andrew55@www.tae.ru) |
| 15:47.38 | epablo | coolhp: Are they behind NAT? |
| 15:47.57 | epablo | coolhp: set reginterval to about 120sec |
| 15:49.00 | coolhp | Hi epablo. |
| 15:49.08 | coolhp | No, I'm not behind a NAT. |
| 15:49.17 | coolhp | Reginterval ? |
| 15:49.25 | coolhp | Is that on the cisco side or asterisk side ? |
| 15:49.29 | epablo | cisco |
| 15:49.40 | *** join/#asterisk jbalcomb (n=jbalcomb@gateway.imtco.com) |
| 15:50.08 | epablo | It's tells the phone how often to register on to asterisk. |
| 15:50.09 | *** join/#asterisk ozant (n=zet@mail.grid.com.tr) |
| 15:50.24 | jbalcomb | anyone here anything about SNOMs being so mismanufactured right now that vendors aren't selling them? |
| 15:50.27 | coolhp | Is that timer_register_expires ? |
| 15:50.32 | epablo | yeap |
| 15:50.46 | Hmmhesays | ugh I can't find a good tutorial for bind9 |
| 15:50.47 | *** join/#asterisk Tall-guy (i=tall-guy@207-195-103-110.regn.hssx.sasknet.sk.ca) |
| 15:50.53 | coolhp | That is so weird... I've had people tell me the exact oposite ... LOL |
| 15:51.00 | coolhp | My original config was 3600. |
| 15:51.03 | zamba | Hmmhesays: what's your problem? |
| 15:51.20 | coolhp | Someone told me to set it to 30000... that didnt help |
| 15:51.22 | jbalcomb | Hmmhesays there are no good tutorials for BIND anything |
| 15:51.37 | coolhp | Do you think 120 will help ? |
| 15:51.41 | Hmmhesays | zamba: on problems, just never used bind before in my life, and I have found myself in need of a dns server |
| 15:51.54 | zamba | Hmmhesays: apt-get install bind9 :) |
| 15:52.09 | jbalcomb | Hmmhesays check out MyDNS or perhaps Windows Server |
| 15:52.29 | epablo | coolhp: Not sure, but I have it set that way.. I had to do it to fix NAT issues |
| 15:52.34 | fourcheeze | jbalcomb: !!! |
| 15:52.39 | fourcheeze | there's nothing at all wrong with bind |
| 15:52.53 | jbalcomb | haha.. only that its less friendly than Sendmail |
| 15:53.03 | fourcheeze | do you use it? |
| 15:53.09 | jbalcomb | yes |
| 15:53.16 | fourcheeze | I don't understand how you can think that |
| 15:53.21 | fourcheeze | there's only 2 things to edit |
| 15:53.29 | [TK]D-Fender | jbalcomb : There are a number of "easy" lightweight Linux DNS servers out there... |
| 15:53.31 | fourcheeze | and if you're really lazy you can get webmin to do it for you |
| 15:53.53 | epablo | I like postfix.. |
| 15:53.58 | jbalcomb | fourcheeze i only think it based on what I experienced initially and what I've seen other people go through |
| 15:54.03 | fourcheeze | Hmmhesays: what is your criteria for choosing a dns server? |
| 15:54.06 | Tall-guy | "....a BIND is a terrible thing to waste...." |
| 15:54.17 | _Sam-- | bind isnt bad once you get your zone files set, and there are enough examples. |
| 15:54.19 | fourcheeze | epablo: pity postfix isn't a DNS server |
| 15:54.21 | Hmmhesays | not much, just have to support srv records |
| 15:54.26 | jbalcomb | fourcheeze I do think it's the best though which is why I still use it |
| 15:54.36 | jbalcomb | fourcheeze epablo haha |
| 15:54.44 | epablo | <PROTECTED> |
| 15:54.48 | pif | any happy destar user around? |
| 15:55.45 | fourcheeze | Hmmhesays: http://www.tldp.org/LDP/lame/LAME/linux-admin-made-easy/domain-name-server.html should tell you what you need to know |
| 15:56.06 | fu3 | djbdns rules |
| 15:56.14 | fu3 | http://cr.yp.to |
| 15:56.17 | [TK]D-Fender | fu3 : Yeah, thats the one I was thinking about... |
| 15:56.17 | fourcheeze | Hmmhesays: are you conversant with the concept of what dns does? |
| 15:56.22 | fu3 | I love it |
| 15:56.25 | fu3 | ive been using it for YEARS. |
| 15:56.27 | fourcheeze | nah, crappy license |
| 15:56.32 | jsharp | :ughs at anything written by DJB. |
| 15:56.37 | fu3 | why? |
| 15:56.57 | fourcheeze | I don't think it's as good as bind - certainly not as quick |
| 15:57.16 | jsharp | Just makes me want to stab myself with a rusty trout. Gotta install a bunch of other crap, like daemontools. |
| 15:57.18 | *** join/#asterisk yjoe (n=yjoe@65.222.176.9) |
| 15:57.32 | x86 | anyway to make an extension park a call, do something, and when finished doing something retrieve the parked call? |
| 15:57.33 | x86 | heh |
| 15:57.38 | fourcheeze | jsharp: yeah and all that srv stuff |
| 15:57.43 | fourcheeze | service |
| 15:57.44 | fourcheeze | whatever |
| 15:57.47 | _Paulo_ | pdns is good |
| 15:58.00 | htims | hello i've got a problem when i try call someone via sip. the called phone just rings about 25 secs. and then i get at the calling side a busy signal. when the called person answers the call before i get the busy signal, everything is ok. does someone has a hint for me? |
| 15:58.20 | Hmmhesays | fourcheeze: haha who me? |
| 15:58.26 | Hmmhesays | yeah I'd say so |
| 15:58.29 | fourcheeze | ok |
| 15:58.34 | fourcheeze | so you know what an A record is? |
| 15:58.39 | Hmmhesays | whats that? |
| 15:59.00 | Hmmhesays | j/k, yes i'm familiar with dns and its inner workings |
| 15:59.02 | *** part/#asterisk Rhizome (n=rhizome@tor/session/x-5c51173685caa0e4) |
| 15:59.03 | fourcheeze | :-) |
| 15:59.07 | fourcheeze | ok, so just go for bind |
| 15:59.08 | *** join/#asterisk FlatFoot (n=simon@80.88.192.113) |
| 15:59.17 | fourcheeze | basically all you do is edit named.conf |
| 15:59.20 | fourcheeze | add your zone to that |
| 15:59.26 | fourcheeze | then create a file for the zone |
| 15:59.28 | FlatFoot | good afternoon all |
| 15:59.39 | fourcheeze | plenty of examples to be going on wiht |
| 15:59.50 | *** part/#asterisk Tall-guy (i=tall-guy@207-195-103-110.regn.hssx.sasknet.sk.ca) |
| 16:00.25 | FlatFoot | does anyone know where i might find , or have any sensible ringtones to use with the snom . just looking for a normal ring |
| 16:00.38 | epablo | Well gotta go.. c'ya |
| 16:00.40 | *** join/#asterisk dingbat- (n=dingbat@cD9089849.inet.catch.no) |
| 16:00.40 | *** part/#asterisk epablo (n=epablo@WLL-24-pppoe203.t-net.net.ve) |
| 16:00.50 | fourcheeze | FlatFoot: there's a cool CTU one out there |
| 16:00.59 | FlatFoot | CTU ?? |
| 16:01.16 | fourcheeze | but then you have to say things like "CTU, Almeda" when you pick it up |
| 16:01.22 | dingbat- | Does anyone have a general idea of cpu and ram usage of a asterisk server with 100users, probably 10concurrent calls? |
| 16:01.57 | FlatFoot | need a boring one it's for a solicitors office |
| 16:02.20 | [TK]D-Fender | dingbat- : Just a normal decent P4 type system with a decent MB. |
| 16:02.22 | jsharp | dingbat-: You could run that off any decent P4. |
| 16:02.34 | dingbat- | cool, sure hope so :) |
| 16:02.41 | dingbat- | thanks |
| 16:02.44 | [TK]D-Fender | dingbat- : Nothing terribly special, just give it like 2 gig ram and I guess it'll be happy. |
| 16:02.45 | Hmmhesays | Hey Katty |
| 16:02.51 | dingbat- | hehe |
| 16:02.53 | [TK]D-Fender | Katty: mew. |
| 16:03.15 | Katty | [TK]D-Fender: mew |
| 16:03.53 | Druken | hmm..... is $92 good for a pIII 1.0 ? |
| 16:04.00 | Druken | CDN |
| 16:04.16 | jsharp | For a whole machine? |
| 16:04.23 | Druken | yeah |
| 16:04.27 | backblue | Druken: yes. |
| 16:04.28 | Druken | 128m 20gig |
| 16:04.39 | jsharp | Yeah, that's decent. |
| 16:04.41 | backblue | 128 its not much, increase it |
| 16:04.46 | [TK]D-Fender | Druken : not really... |
| 16:04.58 | FlatFoot | anyone want any athlon boards ( XP ) got a few spare |
| 16:05.01 | Druken | [TK]D-Fender: you got a better supplier? |
| 16:05.07 | backblue | FlatFoot: for free? |
| 16:05.16 | FlatFoot | i wish |
| 16:05.30 | backblue | FlatFoot: ok, say your price. |
| 16:05.41 | FlatFoot | only got about 300 left |
| 16:05.47 | backblue | 300? |
| 16:05.48 | x86 | FlatFoot: with or without CPU? |
| 16:05.51 | [TK]D-Fender | Druken : I get wholesale and I see great Sempron deals. |
| 16:05.53 | backblue | omg |
| 16:05.54 | FlatFoot | without cpu |
| 16:06.03 | x86 | weak ;) |
| 16:06.10 | backblue | FlatFoot: 300 motherboards? hehe where the hell do you work? |
| 16:06.12 | x86 | how much are you wanting? |
| 16:06.17 | *** join/#asterisk SplasPood (n=jwb@206.252.198.100) |
| 16:06.25 | Druken | [TK]D-Fender: so do i... but i don't want anything special... it's just for my 4 yr old son |
| 16:06.41 | FlatFoot | just come across a good deal nto sure really what they are worth , and indeed if they all work |
| 16:06.55 | FlatFoot | to a random selection over the weekend and they work |
| 16:07.05 | *** join/#asterisk dgorski (n=dgorski@c-69-245-111-167.hsd1.mi.comcast.net) |
| 16:07.17 | backblue | FlatFoot: dont have MP processors? |
| 16:07.21 | backblue | i need 2 MP |
| 16:07.36 | backblue | i have a tyan tiger mp in "stock" because i dont have proc |
| 16:07.58 | FlatFoot | might also have a couple of hundred xp procs |
| 16:08.15 | [TK]D-Fender | Druken : http://www.tigerdirect.ca/applications/SearchTools/item-details.asp?EdpNo=1819500&Sku=MBM-P4V8X-MX-340 |
| 16:08.16 | FlatFoot | not tested them yet |
| 16:08.34 | FlatFoot | want more info then email simon@ses-networks.co.uk |
| 16:09.02 | backblue | FlatFoot: and what should i say? i speak with FlatFoot in the irC? :o |
| 16:09.35 | FlatFoot | i got a list on the home pc more info there |
| 16:09.42 | [TK]D-Fender | Druken : Oh, for a full machine? Hell yeah, do it... |
| 16:10.54 | FlatFoot | anyway back to why i started this irc up today , ringtones any sensible ones out there |
| 16:11.19 | Druken | [TK]D-Fender: yeah a fll machine |
| 16:11.25 | Druken | not just the processor |
| 16:12.03 | [TK]D-Fender | Druken : Do it... it'll be fine. |
| 16:12.11 | [TK]D-Fender | Assuing its got enough ram, etc. |
| 16:12.33 | *** join/#asterisk puzzled (n=yeahrigh@puzzled.xs4all.nl) |
| 16:12.52 | [TK]D-Fender | FlatFoot : You've heard the answer already... the Cisco "24" warble is rather unobtrusive. |
| 16:13.19 | FlatFoot | aha thanks v much i shall wander off and find that |
| 16:13.22 | Druken | [TK]D-Fender: 128m 20 gig |
| 16:13.49 | *** join/#asterisk iDunno (i=brettp@miranda.sommitrealweird.co.uk) |
| 16:13.58 | [TK]D-Fender | Druken : Thats tight... up to 256 and it'll be fine |
| 16:14.44 | Druken | [TK]D-Fender: i get shit like this all the time... |
| 16:15.01 | Druken | 2 weeks ago i had a couple 2 gigs for like 400 |
| 16:15.03 | *** join/#asterisk wunderkin (i=kev@ip68-226-113-228.ph.ph.cox.net) |
| 16:16.11 | *** join/#asterisk PakiPenguin (n=uppal@linuxpakistan/admin/pakipenguin) |
| 16:21.34 | trixter | why is 'gay' and 'iax' the same on the dtmf keypad? |
| 16:22.00 | bkw_ | because its the "gay" protocol |
| 16:23.10 | *** join/#asterisk SwK[Work] (n=SwK@64.89.118.139) |
| 16:23.58 | jsharp | AST_FRAME_TYPE_GAY? |
| 16:24.05 | trixter | haha |
| 16:24.12 | trixter | ghey |
| 16:24.20 | trixter | I am sure that all the gay men out there dont want to be associated with iax |
| 16:24.37 | *** join/#asterisk Defraz (n=t0tal@tim.mychoice.cc) |
| 16:24.51 | stock | geez, don't dirty up sauce code! |
| 16:24.53 | stock | later |
| 16:26.07 | x86 | anyway to make an extension park a call, do something, and when finished doing something retrieve the parked call? |
| 16:26.31 | trixter | isnt that the core functionality of parking? |
| 16:26.47 | Enth | Guys, is it possible to get a sip client to connect to an asterisk server behind NAT ? |
| 16:26.55 | [TK]D-Fender | Enth : Certainly. |
| 16:27.19 | *** join/#asterisk madounet (n=mad_net@juv34-2-82-226-155-19.fbx.proxad.net) |
| 16:27.31 | Hmmhesays | well that was easy |
| 16:27.31 | Hmmhesays | LOL |
| 16:28.09 | jsharp | Set nat=yes in sip.conf |
| 16:28.13 | jsharp | And pray. |
| 16:28.16 | Enth | fn~[TK]D-Fender: Ok, here's the scene: Person A is running X-lite at his place. Person B has broadband, an asterisk server behind NAT. Now can person A call Person B? (voip ob course) |
| 16:28.27 | jsharp | Ohwait. |
| 16:28.30 | jsharp | Server behind NAT. |
| 16:28.38 | jsharp | Yes, but takes a little more configuration voodoo. |
| 16:28.42 | trixter | yeah just set the nat box properly and forget it |
| 16:28.47 | [TK]D-Fender | Enth : Everyone and anyone can be behind NAT and still function. |
| 16:28.58 | trixter | you should do externip=xxx and localnet=xxx |
| 16:29.01 | trixter | both |
| 16:29.11 | Enth | ok so what IP should person A put on his sip client? |
| 16:29.25 | jsharp | The external address. |
| 16:29.43 | jsharp | Whatever globally routable IP the NAT server is given by the ISP. |
| 16:29.44 | Enth | and if its dynamic ip like most broadband are |
| 16:29.49 | Enth | hehe |
| 16:29.55 | jsharp | Then you'll be playing "Chase the IP" |
| 16:30.00 | Hmmhesays | anyone have an example srv record for bind? |
| 16:30.04 | Enth | gathered as much. |
| 16:30.09 | Enth | *whistle* |
| 16:30.13 | [TK]D-Fender | Enth : If * is behind nat you need "nat=yes", either "externip" or "externhost" + "externrefresh", "canreinvite=no" and "localnet" in sip.conf [general], and "nat=yes" on the peer entries. |
| 16:30.24 | x86 | trixter: well |
| 16:30.54 | [TK]D-Fender | BRB... system unstable... rebooting... |
| 16:31.00 | Enth | brb |
| 16:32.25 | x86 | trixter: say i have an IVR, user dials an option, i want to park the call (so that MOH is played), do some processing in the background (get text and convert it to sound), retreive the call, playback the temp file |
| 16:32.31 | *** join/#asterisk steve___ (n=steve@store-fw.porchlight.ca) |
| 16:32.55 | trixter | ahh do it from the ivr? |
| 16:33.04 | x86 | right |
| 16:33.10 | x86 | no human intervention |
| 16:33.24 | trixter | havent looked at how to od that iirc you can park a call from the ivr, I dont know how you would retreive that specific call |
| 16:33.31 | backblue | x86: hehehe text to speech? i will do that, probably in the next weeks |
| 16:33.49 | trixter | however I know the ghetto way would be to do it in an agi so you can background the TTS parts while doing a 'musiconhold' so they hear music |
| 16:33.51 | backblue | but i will do it in realtime,. |
| 16:33.53 | trixter | and you never lose the call |
| 16:34.05 | trixter | but that has a higher cost on the system than doing everything internally |
| 16:34.08 | Druken | can't you run an agi, with moh in the background? |
| 16:34.21 | jsharp | AGI can use "set music on", right? |
| 16:34.25 | x86 | i want to park the call, because if i try to do MusicOnHold, it never executes the System... If I do Background(some-small-music-file), it plays the sound, and then executes the System |
| 16:34.32 | *** join/#asterisk GyrosGeier (n=richter@p54994CEE.dip.t-dialin.net) |
| 16:34.34 | x86 | same with playback |
| 16:34.35 | GyrosGeier | hi |
| 16:34.54 | trixter | you arent understanding |
| 16:34.54 | GyrosGeier | did Asterisk's behaviour with regard to missing translated voice prompts change? |
| 16:35.00 | trixter | an agi can fork/spawn a new thread |
| 16:35.06 | x86 | oooo nice |
| 16:35.13 | trixter | so if it tells asterisk to do musiconhold for that call it can still go off and do whatever it needs |
| 16:35.26 | x86 | trixter: how would i go about doing that? |
| 16:35.37 | x86 | that's exactly what i need ;) |
| 16:35.39 | GyrosGeier | I just got a bug report that due to a missing 1f.gsm, voicemail calls were immediately terminated when a single message was waiting |
| 16:36.07 | x86 | GyrosGeier: hahahaha |
| 16:36.11 | *** join/#asterisk [TK]D-Fender (n=joe@toronto-HSE-ppp4122655.sympatico.ca) |
| 16:36.34 | *** join/#asterisk PakiPenguin_ (n=uppal@linuxpakistan/admin/pakipenguin) |
| 16:36.38 | Enth | wonder if using dyndns.org could help with the issue of * behind NAT but on a connection on a dynamic IP externally |
| 16:36.56 | GyrosGeier | x86, well, it sucks if you have to sort out the mess |
| 16:37.01 | [TK]D-Fender | Enth : yes it should... |
| 16:37.35 | [TK]D-Fender | Enth : then use "externhost" + "externrefresh".... |
| 16:37.55 | Enth | where? |
| 16:37.56 | *** join/#asterisk klasstek (n=nunyobiz@sta-206-168-96-89.rockynet.com) |
| 16:37.57 | x86 | trixter: do you have an example of doing such a thing? |
| 16:37.58 | Enth | sip.conf? |
| 16:38.04 | [TK]D-Fender | Enth : yup |
| 16:38.08 | Enth | cool |
| 16:38.09 | Enth | brb |
| 16:39.26 | *** join/#asterisk mog_work (n=mogorman@gateway.digium.com) |
| 16:40.48 | *** join/#asterisk cthompson (n=ct@66.98.136.16) |
| 16:42.57 | *** join/#asterisk stkn (i=nobody@gentoo/developer/pdpc.active.stkn) |
| 16:43.34 | Hmmhesays | heh, well that was painless |
| 16:43.36 | Hmmhesays | I'm suprised |
| 16:44.57 | Enth | which where? |
| 16:45.43 | Hmmhesays | setting up bind |
| 16:45.58 | *** join/#asterisk Bambr (n=Bambr@213-35-233-174-dsl.end.estpak.ee) |
| 16:46.06 | Enth | ah |
| 16:46.18 | *** join/#asterisk denon (i=denon@synapse.subneural.net) |
| 16:46.18 | *** mode/#asterisk [+o denon] by ChanServ |
| 16:46.22 | Enth | GUI setup ? :P |
| 16:46.33 | Hmmhesays | no, named.conf and then the associated zone file |
| 16:47.38 | fourcheeze | yep - way too easy |
| 16:47.49 | Enth | heh |
| 16:47.55 | Hmmhesays | granted the setup I require is super simple, it seems to be working right |
| 16:48.04 | *** join/#asterisk octothorpe_ (n=octothor@198.60.73.230) |
| 16:48.13 | fourcheeze | you can now have fun benchmarking it ;-) |
| 16:48.48 | Hmmhesays | i don't think I need to benchmark it, I don't need to handle more than 50 or so queries per second |
| 16:49.11 | Hmmhesays | you should be able to do that on a p1 |
| 16:49.32 | [TK]D-Fender | Enth : You can keep it reasonable with externhost & externrefresh, but yeah, it is easier with a fixed IP somewhere in the path |
| 16:49.54 | Enth | true |
| 16:50.56 | Enth | but then, there will be issues with having a static IP. |
| 16:51.10 | jsharp | Issues? |
| 16:51.10 | [TK]D-Fender | Enth : its worth it... and few issues I've encountered... |
| 16:51.21 | fourcheeze | what's the problem with a static IP |
| 16:51.28 | fourcheeze | it's just like a dynamic one, except it doesn't change |
| 16:51.28 | [TK]D-Fender | FUD about attacks :) |
| 16:51.46 | [TK]D-Fender | Just don't be stupid about whats visible on it... |
| 16:52.01 | Enth | Well, irc can be a pain with static IP |
| 16:52.13 | [TK]D-Fender | I don't run things on normal ports and don't run unnecessary services |
| 16:52.18 | Enth | not that I use my real ip here anwyay |
| 16:52.52 | fourcheeze | there's no difference between random attacks whether your IP is dynamic or static |
| 16:53.05 | Enth | I agree but then.... |
| 16:53.12 | [TK]D-Fender | Enth : I never had a problem with a fixed IP and IRC before... and that accounts for Windows + SonicWALL at work, and Windows/Linux + Linux NAT (and direct on gateway) at home |
| 16:53.12 | fourcheeze | people who do this just scan a large range and look for open ports |
| 16:53.36 | Enth | yeah |
| 16:54.12 | fourcheeze | if you leave a hole they can get in whether you're static or not |
| 16:54.18 | fourcheeze | just don't leave any holes |
| 16:54.58 | Enth | 99% of the time I'm on the bnc. |
| 16:55.22 | Enth | and once again, the box is not located here. it's elsewhere. |
| 16:55.25 | Enth | so hrmmm |
| 16:55.36 | Enth | damn it, I'll go for a static ip then |
| 16:55.44 | Enth | brb |
| 16:56.19 | *** join/#asterisk dpolitech (n=Owner@207.224.48.130) |
| 17:01.40 | *** join/#asterisk spunz_ (n=spunz@h081217096096.dyn.cm.kabsi.at) |
| 17:09.05 | *** join/#asterisk salviadud (n=ralfalfa@dsl-201-133-198-176.prod-infinitum.com.mx) |
| 17:09.29 | *** join/#asterisk PakiPenguin_ (n=uppal@linuxpakistan/admin/pakipenguin) |
| 17:15.26 | *** join/#asterisk Lino` (n=Lino@i577BDDA8.versanet.de) |
| 17:15.44 | *** join/#asterisk brockj49464 (n=brockj49@63.87.56.235) |
| 17:19.21 | Hmmhesays | does chan_sip not handle redirects very well or what? |
| 17:19.27 | *** join/#asterisk riksta (n=rick@213.121.151.210) |
| 17:19.53 | salviadud | i hear them crickets today |
| 17:20.03 | riksta | Hi there, can someone confirm that on an x86_64 box running fedora 4, I should have no problems using the zaptel dummy kernel module (and compiling it ok) ? Thanks a lot |
| 17:20.08 | salviadud | does anyone want to have an interesting conversation via FWD? |
| 17:21.44 | Hmmhesays | i send back a redirect and everything goes all crazy |
| 17:21.56 | salviadud | crazy? |
| 17:22.09 | *** join/#asterisk hypa7ia (i=hypatia@wsip-24-234-241-145.lv.lv.cox.net) |
| 17:22.12 | salviadud | what, asterisk trying to call al-qaeda via iax2? |
| 17:22.12 | file | Hmmhesays: depends |
| 17:22.38 | salviadud | that's not crazy, that's dangerous man. |
| 17:22.49 | Hmmhesays | when I send my redirect back from my peer asterisk forwards the call to local/{$EXTEN}@<peer context> |
| 17:23.00 | file | yup |
| 17:23.04 | salviadud | .... |
| 17:23.07 | Hmmhesays | thats no good |
| 17:23.16 | salviadud | have you tried switch => ??? |
| 17:23.17 | file | there's an option to make it use the SIP URI given back |
| 17:23.30 | Hmmhesays | where is this? |
| 17:23.36 | *** join/#asterisk pryk (n=tmalkut@fw.orasoft.net.pl) |
| 17:23.41 | file | promiscredir |
| 17:23.44 | file | I think |
| 17:23.44 | *** join/#asterisk pixolex (n=chatzill@87-196-215-18.net.novis.pt) |
| 17:24.00 | salviadud | what's the deal with asterisk 1.2.5 |
| 17:24.26 | salviadud | i have 1.2.4... and the changelog is not that convincing, bunch of bug fixes for slowlaris |
| 17:24.36 | file | then don't upgrade... |
| 17:24.52 | salviadud | i won't! no way maaaaaaaan |
| 17:24.53 | file | the 1.2 branch is just for bug fixes really... :) |
| 17:25.03 | salviadud | i love this chan |
| 17:25.15 | FuriousGeorge | anyone messing with that valet parking hack? |
| 17:25.16 | salviadud | makes me feel like part of some wild crazy experiment |
| 17:25.21 | Hmmhesays | that was it file |
| 17:25.22 | Hmmhesays | thanks |
| 17:25.27 | file | yw |
| 17:25.33 | FuriousGeorge | anyone know if it got ported to 1.2 |
| 17:25.52 | Hmmhesays | this asterisk ser coupling is going pretty well so far |
| 17:26.20 | file | Hmmhesays: the only thing with promiscredir is that you can't forward internally in the Asterisk, only remote destinations |
| 17:26.50 | Hmmhesays | file: thats ok with me because all of my redirects will be to users registered to SER |
| 17:26.55 | file | k |
| 17:27.03 | file | then that's fine |
| 17:27.38 | *** join/#asterisk shido6 (n=shido6@d221-68-216.commercial.cgocable.net) |
| 17:28.52 | *** join/#asterisk vmedrano (n=vmedrano@196.32.128.206) |
| 17:31.24 | jbalcomb | I can not appreciate that the install guides for Asterisk all seem to suggest 'automatically partition' for the Linux install |
| 17:32.35 | salviadud | asterisk is not intended to be used by linux newbs... sorry mang |
| 17:32.44 | jbalcomb | It seems that /var ought to be significantly larger than the default 512mb |
| 17:33.28 | jbalcomb | salviadud wouldn't that seem to be exactly why the doc should suggest automatic? |
| 17:33.28 | salviadud | hmmmmm, are you running solaris? |
| 17:33.48 | jbalcomb | Debain right now but I'm install RH ES 4 |
| 17:33.48 | salviadud | my /var does not have a space limit |
| 17:33.52 | fu3 | hahahaha |
| 17:33.59 | fu3 | prepare for ownership transition |
| 17:34.09 | salviadud | i'd use slackware |
| 17:34.17 | fu3 | there is a REASON for breaking up your mount points |
| 17:34.28 | jbalcomb | i'd use whatever digium suggests i should use so my support contract is valid |
| 17:34.59 | salviadud | well, fu3, what's your distro of choice? |
| 17:35.00 | *** join/#asterisk ManxPower (n=ewieling@64-247-239-154.wan.networktel.net) |
| 17:35.05 | fu3 | FreeBSD |
| 17:35.12 | [TK]D-Fender | jbalcomb : Got chumped into paying for Linux... sheesh.... |
| 17:35.13 | salviadud | see! |
| 17:35.20 | fu3 | see what? |
| 17:35.21 | salviadud | the more unix. the better |
| 17:35.23 | jbalcomb | the auto partition wants a 25GB usr, 17GB home, and 512MB for var |
| 17:35.25 | [TK]D-Fender | jbalcomb : Should have just gone with CentOS. |
| 17:35.39 | [TK]D-Fender | jbalcomb : On a server for * you should HAVE users! |
| 17:35.44 | [TK]D-Fender | shouldn't* |
| 17:35.50 | jbalcomb | haha.. i forgot to put on my pads before i joined the channel.. hold on, let me suite up |
| 17:35.57 | fu3 | yeah im only using FreeBSD until I finally come to a problem that I cannot overcome. |
| 17:36.03 | fu3 | then i'll switch to Linux. Grudgingly. |
| 17:36.14 | jbalcomb | s |
| 17:36.15 | salviadud | freebsd ain't that good |
| 17:36.17 | fu3 | but whatever works best.. i'll do it. |
| 17:36.22 | fu3 | FreeBSD is great.. sorry. |
| 17:36.37 | salviadud | recompiling a freebsd kernel is a whole other story |
| 17:36.42 | fu3 | it takes three commands |
| 17:36.42 | fu3 | woo |
| 17:36.54 | salviadud | and besides. the options are not that great. |
| 17:36.55 | jbalcomb | if they lied to me about Asterisk working best with RH ES so be it. I'll learn my lesson. |
| 17:37.07 | fu3 | whats wrong with them? |
| 17:37.11 | fu3 | what options arent there? |
| 17:37.17 | fu3 | what cant be tuned via sysctl? |
| 17:37.22 | salviadud | i wante to try infrared |
| 17:37.28 | salviadud | on freebsd |
| 17:37.43 | salviadud | then, a friend of mine wanted to get a wireless card to work... such pain and agony |
| 17:37.52 | salviadud | we rather use ndiswrapper |
| 17:37.59 | fu3 | well.. that sucks. I can understand the personal experiences thing. |
| 17:38.07 | fu3 | But, for me, it's always been nearly perfect |
| 17:38.12 | fu3 | but thats how it goes |
| 17:38.31 | salviadud | i think it's a great OS for an apache web server and a good firewall. |
| 17:38.32 | [TK]D-Fender | jbalcomb : * works fine on ANY stable linux platform. It runs mighty fine on Slackware for me. So frankly you're paying for something that has little to do with anything. |
| 17:38.39 | jbalcomb | why do you guys spend any amount of time arguing about what *nix to run? |
| 17:38.40 | salviadud | never goes down... |
| 17:38.47 | salviadud | unless of course, power failure |
| 17:38.56 | fu3 | yeah.. that'll happen. |
| 17:39.14 | fu3 | hell.. my gear is protected by a big natural gas generator. |
| 17:39.16 | ManxPower | jbalcomb, It will keep happening until everyone realizes that Mandrake is the best. |
| 17:39.27 | fu3 | it didnt start up during the last power outage :/ |
| 17:39.29 | fu3 | because one $2.50 sensor went bad. |
| 17:39.31 | jbalcomb | [TK]D-Fender i get it. i don't care. i've got RH ES so let's just take it from there eh? |
| 17:39.39 | salviadud | that's unlucky |
| 17:39.45 | fu3 | yeah.. tell me about it! |
| 17:39.51 | *** join/#asterisk vr_mex (n=vr_mex@dsl-201-144-0-18.prod-infinitum.com.mx) |
| 17:39.55 | salviadud | what are you running anyway? |
| 17:40.08 | fu3 | ? |
| 17:40.09 | salviadud | is it a company server? |
| 17:40.14 | fu3 | what server? |
| 17:40.33 | salviadud | are you talking about your "home" gear? |
| 17:40.35 | fu3 | no. |
| 17:40.38 | fu3 | this is a college. |
| 17:40.49 | salviadud | you a teacher? |
| 17:40.51 | fu3 | fuck no. |
| 17:40.58 | salviadud | haha |
| 17:41.04 | salviadud | student then? |
| 17:41.04 | fu3 | I would *never* teach here. |
| 17:41.09 | fu3 | no. i'm not a student either. |
| 17:41.11 | fu3 | just an employee. |
| 17:41.19 | salviadud | what college is it? |
| 17:41.23 | *** join/#asterisk asterboy (n=Snake@S01060204ee2b6007.ed.shawcable.net) |
| 17:41.31 | fu3 | does it matter? |
| 17:41.41 | salviadud | no, yet i am curious |
| 17:41.43 | fu3 | haha |
| 17:41.58 | asterboy | Anyone, have the skinny on upgrading from 1.2.0 to 1.2.5? Is it a simple download, compile, install? |
| 17:42.37 | jbalcomb | ok, /boot=128MB, /home=10GB, /usr=10GB, /var=10GB, /tmp=1024MB, SWAP=2048. Is that good? Am I missing anything? |
| 17:42.44 | fu3 | tmp should match swap. |
| 17:42.52 | vr_mex | how do i do the cabling to connect 16 analog phones to a tdm400p digium card? |
| 17:42.52 | fu3 | depending on things.. |
| 17:42.56 | fu3 | i'll keep my yapper shut |
| 17:42.59 | *** join/#asterisk cuban (n=djimenez@johnny5.cuban.cc) |
| 17:43.04 | jbalcomb | fu3 ok, tmp=2048MB |
| 17:43.18 | cuban | Hi all, I'm having trouble understanding the syntax of the IF statement. The documentation on the wiki is sorta confusing. |
| 17:43.27 | salviadud | cuban |
| 17:43.33 | salviadud | www.asteriskguru.com |
| 17:43.37 | jbalcomb | fu3 well, i've got 67GB RAID5 so no worries on an extra GB |
| 17:43.40 | fu3 | :) |
| 17:43.49 | *** join/#asterisk oej (n=oej@apollo.webway.se) |
| 17:43.55 | fu3 | looks good. but are you going to be doing all of your logging to /var? |
| 17:44.03 | fu3 | if so, how often are you going to rotate the logs? |
| 17:44.07 | fu3 | because /var will fill up fast. |
| 17:44.21 | fu3 | again, depending on things. |
| 17:44.25 | jbalcomb | fu3 yes, logging as well as VM, IVR prompt, reports, etc |
| 17:44.26 | cuban | salviadud I checked that out and didn't see IF under functions |
| 17:44.42 | salviadud | ? |
| 17:44.48 | jbalcomb | fu3 ive got 120 users now for 4 months and its at 2.1 GB used |
| 17:44.58 | fu3 | oh thats not bad at all |
| 17:45.15 | cuban | salviadud It's not there. I'm looking to write an if statement to perform a different action based on the callerid of the incoming caller |
| 17:45.16 | salviadud | yes, you are right indeed |
| 17:45.17 | fu3 | just think about where you and this system will be in another 24 months. |
| 17:45.21 | jbalcomb | fu3 I assume if I dont assign the other 40GB of the RAID I can grow a partition later? |
| 17:45.22 | [TK]D-Fender | cuban : Describe the kind of statement you have in mind. |
| 17:45.23 | salviadud | i thought it was a dialplan app |
| 17:45.24 | vr_mex | how do i do the cabling to connect 16 analog phones to a tdm400p digium card; that is how do I connect physically the 16 analog phones to the fxs port? |
| 17:45.38 | fu3 | jbalcomb.. but why worry about growing it later? why not just allocate the space now? |
| 17:45.41 | [TK]D-Fender | vr_mex : You wouldn't use TDM400' |
| 17:45.43 | [TK]D-Fender | ss for that |
| 17:45.58 | cuban | [TK]D-Fender Basically if I call from X number, go to 2,1 if I call from Y number go to 3,1. I realize it's not 100% secure, but it's more of speed/convience thing then anything else |
| 17:46.00 | jbalcomb | fu3 flexibility, easier to do than undo |
| 17:46.01 | vr_mex | how should i do it with what card and hos? |
| 17:46.05 | fu3 | you can at most, have four analog phones on a tdm400p. |
| 17:46.14 | fu3 | cool.. well.. you know your system :) |
| 17:46.17 | vr_mex | Oh |
| 17:46.37 | *** join/#asterisk spatulamaan (n=ggilmore@ip66-107-33-196.z33-107-66.customer.algx.net) |
| 17:46.50 | jbalcomb | vr_mex you might get single PRI card and have an external adtran (or other device) bring the lines together |
| 17:47.37 | *** join/#asterisk darby_t (n=tom@abbh191.neoplus.adsl.tpnet.pl) |
| 17:47.51 | vr_mex | jbalcomb: thanks, what is a adtran? and what is a PRI card? |
| 17:47.53 | jbalcomb | fu3 i hope so cause if this doesn't go well i'm going to end up eating mostly ramen for a while |
| 17:48.07 | fu3 | what are you concerned about? |
| 17:48.10 | jbalcomb | vr_mex www.google.com is smarter than me |
| 17:48.12 | fu3 | the system going down in a few months or something? |
| 17:48.24 | vr_mex | jbalcomb: thanks a lot |
| 17:48.59 | jbalcomb | fu3 i'm taking over an install and i have made purchases and recommendation for resolving the host of issues that were experienced. my system has to work better. |
| 17:49.06 | fu3 | ahhh :) |
| 17:49.11 | fu3 | yeah im in that boat.. kinda. |
| 17:49.34 | fu3 | you seem to know your shit so i'll wager it'll be better :) |
| 17:49.47 | jbalcomb | fu3 i haven't been a linux admin in 4 years and started using asterisk 3 months ago.. so FUD is well in play here. |
| 17:49.58 | fu3 | hahahah |
| 17:50.06 | fu3 | it'll be fine |
| 17:50.20 | fu3 | one lesson my dad taught me: When you dont know what you're doing.. just get started. It'll work out. |
| 17:50.24 | jbalcomb | fu3 that's my expectation. i'm mad for getting things done and done right. |
| 17:50.30 | fu3 | always has :) |
| 17:50.40 | jbalcomb | always will. =) |
| 17:51.02 | fu3 | the biggest thing i've found is just keep listening.. |
| 17:53.21 | fu3 | ugh.. my sangoma a104d card should be here tomorrow. |
| 17:53.27 | fu3 | so nervous. |
| 17:54.07 | fu3 | Anyonek now what a "circuit pack" is? |
| 17:54.09 | fu3 | doh |
| 17:54.12 | fu3 | Anyone Know |
| 17:55.58 | *** join/#asterisk backblue (n=igor@82.102.1.42) |
| 17:56.20 | backblue | ppl, anyone running CLI from remote asterisk pbx? |
| 18:00.16 | ManxPower | fu3, That's the way it usually is. i.e. whatever backup facilities you have, they won't work when you need them. |
| 18:00.28 | *** join/#asterisk saftsack (n=oliver@p54A7FD17.dip.t-dialin.net) |
| 18:04.37 | cuban | All I'm looking to setup an "IF" statement. For example, if I call from X number, go to 2,1 if I call from Y number go to 3,1. I realize it's not 100% secure, but it's more of speed/convience thing then anything else |
| 18:08.05 | file | exten => 500/145,1,Dial(SIP/dingo) |
| 18:08.11 | file | exten => 500/146,1,Dial(SIP/kaboom) |
| 18:08.25 | file | dial 500 with a callerid of 145 it does the first thing, call it with a callerid of 146 and it does the second |
| 18:09.22 | *** join/#asterisk clive- (n=pirch@dsl-145-37-44.telkomadsl.co.za) |
| 18:09.39 | *** join/#asterisk mroth_imm (n=chatzill@63.65.26.220) |
| 18:09.56 | mroth_imm | anyone know a command to look at which process is holding open file handles on a box? |
| 18:10.13 | *** join/#asterisk Pablin (i=pablito@200.40.228.68) |
| 18:10.15 | Pablin | hi all |
| 18:10.42 | sulex | mroth_imm: lsof? |
| 18:10.44 | Pablin | exist any command to give asterisk to hangup when the line is in silence for example 30 sec ? |
| 18:10.47 | clive- | hi, anyone using Sirrix cards here ? |
| 18:10.54 | mroth_imm | gotit...thanks sulex...brain fart : ) |
| 18:10.59 | *** join/#asterisk MikeJ__ (n=vircuser@d14-69-8-30.try.wideopenwest.com) |
| 18:12.26 | *** join/#asterisk Dr-Linux (n=Nothing@202.125.141.6) |
| 18:12.57 | Pablin | ? |
| 18:13.37 | dgorski | you might look at the voicemail code to see how it's done there... I don't know of a generic way in the dialplan to do that... |
| 18:13.48 | dgorski | Pablin: ^^^^ |
| 18:13.49 | *** part/#asterisk bartpbx (n=bartpbx@p54B00658.dip0.t-ipconnect.de) |
| 18:14.03 | Hmmhesays | ahh nothing like the -F flag in patch |
| 18:14.06 | *** join/#asterisk A-Tuin (n=a-tuin@steves.ip.v4.me.uk) |
| 18:14.30 | Hmmhesays | time to see if my chan_sip asplodes |
| 18:15.52 | clive- | bartpbx hi do you use sirrix ? |
| 18:16.32 | Pablin | my problem is simple... people from outside call me, but asterisk never hangup when the people hang. :( |
| 18:17.24 | dgorski | that's a different issue than detecting silence |
| 18:17.35 | *** join/#asterisk terrapen (n=cjs@166.70.183.108) |
| 18:17.40 | Pablin | mmm i think silence are my solution .... |
| 18:17.41 | dgorski | how do they call you? ZAP / SIP / IAX / ??? |
| 18:18.03 | *** join/#asterisk zapa (n=zant@200.66.16.96) |
| 18:18.05 | dgorski | Pablin: how do they call you? ZAP / SIP / IAX / ??? |
| 18:18.06 | Pablin | PBX.... to sip phones or meetme |
| 18:18.18 | dgorski | how does the call get to your PBX? |
| 18:18.26 | *** join/#asterisk stoffell (n=stoffell@d5153FED3.access.telenet.be) |
| 18:18.33 | Pablin | throw a te110p |
| 18:18.38 | dgorski | ZAP |
| 18:18.40 | Pablin | yep |
| 18:18.40 | terrapen | who is the least sucky mobile phone provider in the US? |
| 18:18.42 | dgorski | PRI ? |
| 18:18.47 | Pablin | yep |
| 18:18.51 | *** join/#asterisk yo_tyler (n=tyrivard@d207-216-24-81.bchsia.telus.net) |
| 18:18.56 | yo_tyler | hello |
| 18:19.04 | terrapen | i gotta get me a new mobile phone |
| 18:19.35 | dgorski | Pablin: have you tried 'pri debug span X' and look at the D channel messaging ? Are you getting the release messages from the telco? |
| 18:19.43 | *** join/#asterisk Zodiacal (i=hehehe@bdsl.66.14.242.199.gte.net) |
| 18:19.59 | yo_tyler | what is the cheepest card that i can use to connect asterisk to POTS? |
| 18:20.05 | Pablin | dgorski: i will try now... |
| 18:20.21 | Zodiacal | anyone know why my voicemail greetings are quiet for incoming pstn callers? but they are fine for internal sip users... |
| 18:20.29 | _Sam-- | [av]bani : you around? |
| 18:20.35 | dgorski | terrapen: I don't like my nextel and am happy with my cingular - my friends also tell me they like T-mobile |
| 18:20.46 | terrapen | yeah, i hate my Sprint |
| 18:20.55 | terrapen | not because the service is bad (it's alright, actually) |
| 18:21.03 | dgorski | yes, the consensus of my frinds with sprint is that it sucks |
| 18:21.07 | terrapen | but because they screw me with $100 of overages every month |
| 18:21.13 | dgorski | my wife has no problem with her sprint phone though... |
| 18:21.14 | terrapen | and i can't change my plan without re-upping 2 years |
| 18:21.17 | dgorski | which is weird |
| 18:21.31 | terrapen | so i'm looking at Cingular...but they really aren't any better! |
| 18:21.38 | dgorski | tarrapen: they all do that |
| 18:21.56 | dgorski | terrapen: (force you to renew when changin your plan) |
| 18:22.00 | terrapen | do you use a blackberry? |
| 18:22.10 | dgorski | yes, and it's the nexten one - and it sucks |
| 18:22.20 | terrapen | hmmm |
| 18:22.21 | dgorski | I can't say for SURE that it's Nextel or BB, but it sucks |
| 18:22.34 | terrapen | the blackberry data plans with cingular are a fucking rip-off |
| 18:22.40 | Pablin | dgorski... with the debug i dont have a hangup by the PBX thats is my problem no? |
| 18:23.24 | dgorski | is the remote called hangs up, the TelCo should send a disconnect on the D channel for that call, signifying to asterisk that the caller is gone and the channel is closing |
| 18:23.27 | *** join/#asterisk justinu (n=Justin@dsl081-083-093.lax1.dsl.speakeasy.net) |
| 18:23.31 | yo_tyler | does anybody know of a softphone for windows moble 5? |
| 18:23.35 | dgorski | otherwise there's no way for asterisk to know that it hung up |
| 18:23.59 | *** join/#asterisk Ad-Hoc (n=Nimbus@ppp40-adsl-80.ath.forthnet.gr) |
| 18:24.02 | dgorski | it looks like this: |
| 18:25.38 | dgorski | > Protocol Discriminator: Q.931 (8) len=9 |
| 18:25.38 | dgorski | > Call Ref: len= 2 (reference 3/0x3) (Terminator) |
| 18:25.38 | dgorski | > Message type: RELEASE (77) |
| 18:25.38 | dgorski | > [08 02 81 90] |
| 18:25.39 | dgorski | > Cause (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard (0) 0: 0 Location: Private network serving the local user (1) |
| 18:25.40 | dgorski | > Ext: 1 Cause: Unknown (16), class = Normal Event (1) ] |
| 18:25.42 | dgorski | <PROTECTED> |
| 18:26.04 | Pablin | dgorski: mmmmmm i understand... but if i insert a parameter in asterisk with silence > 30sec hangup... i solve this problem. have any parameter in asterisk to do that? |
| 18:26.13 | dgorski | no |
| 18:26.32 | Pablin | ok |
| 18:29.00 | *** part/#asterisk bigbootay (n=bigboota@209.250.156.214) |
| 18:29.09 | dgorski | Pablin: I think you should ask your provider why you don't get disconnect messages |
| 18:29.17 | *** join/#asterisk ^Accie^ (n=chrake@212.73.28.130) |
| 18:30.53 | dgorski | Pablin: I think that silence hack would really cause you lots of problems... what if someone on the meetme is on a rant, everyone else would get disconnected because they didn't speak for 30 seconds... that sounds like a bad idea |
| 18:31.16 | Pablin | yep but the proider didnt know any about asterisk and signaling..... |
| 18:31.19 | Pablin | :( |
| 18:31.29 | dgorski | it's not asterisk signalling, it's ISDN signalling |
| 18:31.45 | Pablin | yeeeee. but i can increase the time.... |
| 18:31.48 | *** join/#asterisk blitzrage (n=blitzrag@asterisk/documenteur-extraordinaire/blitzrage) |
| 18:32.04 | jsharp | If you're not getting a disconnect from your ISDN provider, your provider is broken. |
| 18:32.09 | dgorski | make sure your switchtype= matches them, "national" seems to work for many, but "4ESS or 5ESS" are more apprirpaite for ATT/Lucent type ISDN providers |
| 18:32.14 | dgorski | appropriate |
| 18:32.17 | Pablin | dgorsky.. worse.... im living in uruguay with a Siemens PBX and the tecnichians are really newbies in PBXs.... |
| 18:32.31 | dgorski | EuroISDN ? |
| 18:32.40 | Pablin | EUROISDN are my config |
| 18:32.42 | dgorski | what are you using for 'switchtype=" |
| 18:32.48 | Pablin | euroisdn |
| 18:33.05 | dgorski | huh. That's what I use in my lab for my te1xxx cards (set at E1) |
| 18:33.17 | dgorski | the example I pasted was euroisdn |
| 18:34.07 | dgorski | what does your ' |
| 18:34.17 | Pablin | i think i have many problems with siemens..... :( and is impossible to solve in latin america |
| 18:34.28 | dgorski | what do your 'channel =>' statements from zapata.conf look like |
| 18:34.41 | *** join/#asterisk chapeaurouge (n=chap@user-85-201-82-146.tvcablenet.be) |
| 18:35.50 | Hmmhesays | ok how trivial would it be to replace a the username with a mailbox number in the notify messages chan_sip sends out for mwi? |
| 18:36.19 | Pablin | dgorski: http://pastebin.com/587374 |
| 18:37.12 | dgorski | Pablin: shouldn't you be signalling=pri_cpe |
| 18:37.29 | saftsack | a few days left until cebit :) |
| 18:37.35 | saftsack | will digium show something brand new? |
| 18:37.43 | Pablin | if i change to pri_cpe asterisk give me a error |
| 18:38.28 | jsharp | His PBX is the CPE side, so asterisk has to pretend to be the NET side. |
| 18:38.29 | dgorski | because your /etc/zaptel.conf says 1,0,0 instead of 1,1,0 for span= |
| 18:38.31 | Pablin | the PBX is the cpe |
| 18:38.45 | dgorski | net = local timing, cpe = remote timing |
| 18:38.46 | Katty | to burn calories! |
| 18:39.13 | dgorski | span=1,0,0 is local timing, span=1,1,0 is remote timing |
| 18:39.19 | iDunno | of course :) |
| 18:39.20 | *** join/#asterisk kippi (n=chris@untrust-gct.equinoxit.net) |
| 18:39.22 | kippi | hey |
| 18:39.23 | trixter | use a lighter |
| 18:39.26 | trixter | burns calories just fine! |
| 18:39.29 | Pablin | dgorsky ok. i cahnge that |
| 18:39.30 | *** join/#asterisk dpolitech (n=Owner@207.224.48.130) |
| 18:39.32 | Pablin | 110 |
| 18:39.37 | *** join/#asterisk Telamon (i=telamon@blk-222-22-126.eastlink.ca) |
| 18:39.45 | kippi | is there away to use * commands to transfer a call? |
| 18:39.46 | trixter | you can literally watch the fat melt right off |
| 18:40.04 | dgorski | kippi: you can via manager interface |
| 18:40.33 | Pablin | in zaptel.conf i have span=1,1,0 |
| 18:41.18 | kippi | dgorski: manager interface? |
| 18:41.42 | Telamon | I'm having problem with the IAX protocol: phones that use IAX are having voice transmit problems (basically, they break up and get dead air, although they don't disconnect.) SIP phones all work fine. Any ideas what can be causing it? The problem has existed through different versions, from 1.2.0 to 1.2.4, and seems to go away for a week or two then come back. |
| 18:41.52 | Pablin | dgorski: in zaptel.conf i have span=1,1,0 |
| 18:45.05 | fu3 | Yeah Tommy. Before ze germans get here. |
| 18:45.28 | riksta | snatch++ |
| 18:45.33 | fu3 | you know it |
| 18:45.41 | fu3 | such a great movie |
| 18:46.31 | riksta | my fav |
| 18:46.32 | justinu | "got anything to declare?" |
| 18:46.37 | fu3 | yeah |
| 18:46.38 | justinu | "yeah.... don't go to england!" |
| 18:46.40 | fu3 | hahahaha |
| 18:46.47 | riksta | haha oi, im from england |
| 18:46.56 | justinu | "london?" |
| 18:47.01 | riksta | manchester |
| 18:47.01 | fu3 | say.. do Pikeys really exist? |
| 18:47.07 | riksta | yeah man they do |
| 18:47.11 | fu3 | I fuckin hate pikeys |
| 18:47.12 | justinu | "yeah, london... you know, bad food, worse weather... marry fuckin' poppins... london!" |
| 18:47.48 | cypromis | lol |
| 18:47.54 | riksta | "i dont like leaving my country doug, especially unless its for warm sandy beaches....and cocktails with little straw hats" - "we got sandy beaches" - "so who the fuck wants to see em" |
| 18:48.02 | fu3 | hahahaha |
| 18:48.04 | fu3 | classic! |
| 18:48.11 | fu3 | the dialog is so well written |
| 18:48.20 | kippi | how would I get asterisk to pickup a * command |
| 18:48.26 | riksta | you seen revolver? its another guy ritche, its good too |
| 18:48.32 | fu3 | no I havent |
| 18:48.36 | Hmmhesays | ~seen oej |
| 18:48.41 | jbot | oej is currently on #asterisk (1h 4m 52s), last said: 'Current asterisk is Asterisk version 1.2'. |
| 18:48.41 | fu3 | I saw Lock Stock and Two Smoking Barrels |
| 18:48.41 | fu3 | that was great |
| 18:49.05 | oej | I am here |
| 18:49.09 | riksta | 1.2.5 :) |
| 18:49.36 | Hmmhesays | oej, is it you thats done a lot of work on chan_sip? or am I thinking of someone else |
| 18:49.51 | fu3 | Oh nothing tommy. It's tip-top. I'm just not sure about the colour. |
| 18:50.34 | fu3 | "Why do they call him the bullet dodger?" - "Because he dodges bullets". |
| 18:50.37 | fu3 | ok i'm done now :) |
| 18:50.39 | *** join/#asterisk octothorpe_ (n=octothor@198.60.73.230) |
| 18:51.01 | *** join/#asterisk mphill (n=mphill@CPE-70-92-245-5.wi.res.rr.com) |
| 18:51.03 | justinu | "what are you doing tony?" |
| 18:51.12 | justinu | "driving with your head stuck in the my window... what do you think i'm doing?" |
| 18:51.21 | mphill | I am getting some echo when talking on my local PBX, any ideas or pointers? |
| 18:51.34 | justinu | ~gain |
| 18:51.37 | jbot | gaim? |
| 18:51.38 | justinu | ~gains |
| 18:51.43 | justinu | ~rxgain |
| 18:51.46 | justinu | ~txgain |
| 18:51.49 | justinu | ~echo |
| 18:51.50 | jbot | [echo] Displays the given arguments on the screen. Syntax: echo (arg1) (arg2) ..(argN). Where arg1 through argN are the arguments to echo. Example: echo "Hello World" displays the string "Hello World". "Why echo occurs" at http://lists.digium.com/pipermail/asterisk-users/2005-February/088794.html |
| 18:51.53 | justinu | lame |
| 18:52.13 | fu3 | i get echo on my POTS side of the PBX. |
| 18:52.23 | fu3 | voip to pots, echo. Pots to Pots. Echo. Voip to Voip. Perfect every time. |
| 18:52.33 | fu3 | but im workin on it |
| 18:52.39 | Hmmhesays | i'm trying to modify the chan_sip sends out a the notify message |
| 18:52.41 | mphill | fu3: i read on the net to uncomment agressive_suppression |
| 18:52.46 | fu3 | doen and done |
| 18:52.48 | fu3 | didnt work for me |
| 18:53.24 | fu3 | but im doing things outside of the recommendations, so I cannot complain |
| 18:53.34 | riksta | you just did ;) |
| 18:53.43 | mphill | fu3: do you have the loop back audio on or off? |
| 18:53.51 | fu3 | I made a statement that didnt include emotion or anything. |
| 18:54.01 | fu3 | loopback audio? I can shut that off? |
| 18:54.06 | fu3 | wheres that at? |
| 18:54.25 | mphill | on the phone |
| 18:54.28 | mphill | its might help |
| 18:54.35 | mphill | with unbalanced POTS |
| 18:54.40 | fu3 | oh |
| 18:54.41 | mphill | what sort phone are you using? |
| 18:54.43 | fu3 | yeah thats all taken care of |
| 18:54.50 | fu3 | i've tried three different analog phones. |
| 18:54.54 | mphill | humm... echo is gay |
| 18:55.06 | fu3 | at&t, astra, and a weird generic from walmar. |
| 18:55.07 | fu3 | t |
| 18:55.07 | *** join/#asterisk Defraz (n=t0tal@tim.mychoice.cc) |
| 18:55.25 | fu3 | just because those fixes didnt work for me, doesnt mean they wont work for you. |
| 18:57.31 | fu3 | man.. it's really easy to exploit vendors into sending you free shit |
| 18:57.42 | dgorski | Hmmhesays: can't you just use the username as the mailbox? <username> => 1234,Foo User |
| 18:57.57 | PakiPenguin_ | fu3, teach me :p |
| 18:57.57 | Hmmhesays | dgorski, negative |
| 18:58.20 | dgorski | VoiceMail(foo) instead of VoiceMail |
| 18:58.23 | dgorski | (1234) |
| 18:58.26 | dgorski | ? no? |
| 18:58.36 | dgorski | that seems silly |
| 18:58.51 | fu3 | haha |
| 18:58.52 | fu3 | just call them up |
| 18:59.00 | PakiPenguin_ | :) |
| 18:59.04 | fu3 | say you're interested in a medium voip deployment |
| 18:59.11 | fu3 | tell them you need desk phones |
| 18:59.13 | fu3 | and they send you them |
| 18:59.28 | *** join/#asterisk jcollie[home] (n=jeff@2001:470:1f01:252:0:0:0:2) |
| 18:59.33 | fu3 | because they are under the illusion that you are serious about doing business with them |
| 18:59.34 | fu3 | but you're not. |
| 18:59.46 | Hmmhesays | dgorski, no |
| 18:59.52 | *** part/#asterisk Pablin (i=pablito@200.40.228.68) |
| 19:00.01 | fu3 | dont get me wrong, it's actually not cool to push them too far. |
| 19:00.04 | fu3 | but still |
| 19:00.12 | riksta | fu3: you wouldn't be happy if someone did that to your business though |
| 19:00.15 | riksta | its a bit underhanded |
| 19:00.16 | *** join/#asterisk mrac (n=test@p54B6D1BA.dip.t-dialin.net) |
| 19:00.34 | fu3 | I know |
| 19:00.38 | fu3 | thats why i wrote: |
| 19:00.40 | fu3 | dont get me wrong, it's actually not cool to push them too far. |
| 19:00.50 | riksta | i was writing at the time of that :) |
| 19:00.53 | fu3 | ahh |
| 19:01.02 | octothorpe_ | can you spell unethical |
| 19:01.07 | fu3 | yeah believe me.. I talk it up, but I really wont screw anyone over. |
| 19:01.11 | Hmmhesays | in sip_send_mwi_to_peer adding ast_copy_string(p->username, peer->mailbox); should do the trick |
| 19:01.11 | fu3 | plus.. I am serious about buying them |
| 19:01.27 | fu3 | I will always let vendors compete for my business |
| 19:01.35 | fu3 | and I *will* play them against one another. |
| 19:01.45 | fu3 | I dont see anything wrong with that. |
| 19:01.49 | fu3 | Business IS Business |
| 19:02.01 | riksta | thats ok imo |
| 19:02.09 | *** part/#asterisk jcollie[home] (n=jeff@2001:470:1f01:252:0:0:0:2) |
| 19:02.58 | fu3 | I see the State of MN is ripping off Asterisk |
| 19:02.58 | [av]bani | fu3: i like when you get samples from a company, then 3mo later they go bankrupt. |
| 19:03.10 | fu3 | [av]bani.. haha.. why do you have to send them back or something? |
| 19:03.18 | [av]bani | no, they never called me back |
| 19:03.20 | fu3 | :) |
| 19:03.26 | [av]bani | that was 6 years ago |
| 19:03.31 | [av]bani | oh well |
| 19:03.32 | [av]bani | :) |
| 19:03.37 | fu3 | so far i've got some cisco and polycomm phones enroute. |
| 19:03.48 | *** join/#asterisk jpablo (n=jpablo@200.94.130.194) |
| 19:04.17 | fu3 | i'll wager i'll end up going with Polycomm. Cisco's cost so god damn much (or is that a lie?) |
| 19:04.41 | [av]bani | depends |
| 19:04.43 | fu3 | say.. do Cisco VOIP phones get covered with SMARTnet? |
| 19:04.45 | [av]bani | some do, some dont |
| 19:04.57 | [av]bani | for instance |
| 19:05.00 | [av]bani | http://www.cisco.com/en/US/products/hw/phones/ps379/products_data_sheet09186a00800c835a.html |
| 19:05.04 | [av]bani | this cisco is only $115 |
| 19:05.10 | fu3 | really? |
| 19:05.14 | fu3 | that seems really reasonable |
| 19:05.19 | [av]bani | yep.. go look at the url |
| 19:05.26 | fu3 | i am there now |
| 19:05.35 | mroth_imm | anyone know how many concurrent calls the default rtp port range 10000-20000 allows? |
| 19:05.48 | justinu | i just tried to order 20 phones from atacomm, and they want 300 bucks shipping... what a joke |
| 19:05.51 | fu3 | whats the EOL on this particular model? |
| 19:05.55 | [av]bani | 1 line, no blf, 10mbit, no poe, monitor-only speakerphone |
| 19:06.00 | [av]bani | :) |
| 19:06.05 | fu3 | ah.. I do need PoE. |
| 19:06.08 | fu3 | well actually |
| 19:06.11 | fu3 | let me explain |
| 19:06.32 | jaiger | [av]bani, sounds like the polycom 301 |
| 19:06.32 | [av]bani | http://www.cisco.com/en/US/products/hw/phones/ps379/products_data_sheet09186a0080159de2.html |
| 19:06.33 | fu3 | we do not want to sacrifice the availability of our phone system. As it stands, we can use it when the power is out. |
| 19:06.37 | [av]bani | this one i expect is even cheaper |
| 19:06.40 | fu3 | not necessarily so with VoIP. |
| 19:06.42 | [av]bani | it doesnt even have a display :)) |
| 19:07.20 | [av]bani | jaiger: the 301 has dual 10/100 and 2 blf |
| 19:07.23 | fu3 | How well do those PoE phones work? |
| 19:07.30 | fu3 | having a resiliant system is important to us |
| 19:07.49 | [av]bani | depends on the phone and the poe source |
| 19:07.58 | [av]bani | but if you want resilient, poe isnt it in general |
| 19:08.08 | [av]bani | poe = convenience poe != resilience |
| 19:09.34 | octothorpe_ | anybody familiar with a decent iax2 hardphone that is wall mountable and supports poe? |
| 19:10.11 | Zodiacal | anyone know whats wrong with this syntax? exten => *66,2,GotoIf([$mytrunk = "yes"]?3:5) |
| 19:10.18 | Zodiacal | the conditial isn't working |
| 19:10.42 | Telamon | octothorpe_: I think the Polycom 501 is. I've got one on order, so I haven't tried it yet. |
| 19:10.56 | Zodiacal | prevous line: exten => *66,1,SetVar(mytrunk="no") |
| 19:11.08 | Zodiacal | just testing stuff.. |
| 19:12.15 | Telamon | Zodiacal: Try *66,2,GotoIf($[${mytrunk} == "yes"]?3:5) |
| 19:12.43 | octothorpe_ | Telemon: I will check it out, thanks. I'll ask you about it in a week or so and see if you like it |
| 19:13.02 | Telamon | Zodiacal: Or *66,2,GotoIf($[ "${mytrunk}" == "yes" ]?3:5) If the first doesn't work. |
| 19:15.04 | oej | Hmmhesays: that's me |
| 19:15.19 | *** join/#asterisk Qwell[] (i=north@unaffiliated/qwell) |
| 19:16.34 | *** join/#asterisk apardo (n=apardo@87.218.44.29) |
| 19:16.58 | *** join/#asterisk MikeJ[Laptop] (n=vircuser@d14-69-8-30.try.wideopenwest.com) |
| 19:17.13 | Qwell[] | eww, it's MikeJ |
| 19:17.46 | Zodiacal | telamon still not working for some reason :/ do you know of a way to print a variable in the cli verbose mode? |
| 19:17.53 | Zodiacal | so that i can watch what my variable is? |
| 19:17.54 | *** join/#asterisk visba (n=dca[lapt@sta-208-139-193-162.rockynet.com) |
| 19:18.10 | MikeJ[Laptop] | Qwell.. see wahat happens :P |
| 19:18.18 | Zodiacal | is it NoOp()? |
| 19:19.34 | *** join/#asterisk ^Accie^ (n=chrake@212.73.28.130) |
| 19:20.25 | oej | Zodiacal: Yes, use Noop |
| 19:20.39 | Zodiacal | hrmm its still printing "" like my variables not getting set |
| 19:20.56 | Zodiacal | is this the right way to set them? exten => *66,1,SetVar(mytrunk="no") |
| 19:21.43 | fugitivo | anyone using channel banks with asterisk? |
| 19:21.44 | Zodiacal | i think i found the prob |
| 19:23.43 | *** join/#asterisk shido6 (n=shido6@d221-68-216.commercial.cgocable.net) |
| 19:23.43 | salviadud | are those channel banks PRI? |
| 19:24.24 | *** join/#asterisk ke4qqq (n=chatzill@68-115-212-155.static.spbg.sc.charter.com) |
| 19:24.52 | *** join/#asterisk RoyK (n=roy@10.80-203-106.nextgentel.com) |
| 19:24.59 | fugitivo | what channel banks? |
| 19:25.02 | ^Accie^ | Hi all. got an asterisk installation here, an AVM C2 card and p2p BRI setup. in need of someone who could give me some pointers on getting it to work, managed to dial in to the demo voice from the outside but somehow I messed up the configs. willing to pay for some consulting. anyone? |
| 19:25.13 | salviadud | the one's you are using? |
| 19:25.21 | dgorski | fugitivo: rhino works good for us |
| 19:25.22 | fugitivo | i'm not using |
| 19:25.26 | fugitivo | just asking |
| 19:25.38 | fugitivo | i want to know if adtran total access 750 is a good option |
| 19:25.41 | fugitivo | rhino is too expensive |
| 19:25.50 | Telamon | Zodiacal: Sorry, was on the phone. Yep, your setvar syntax is correct. |
| 19:25.59 | *** join/#asterisk pawal (n=pawal@c-65fee253.203-1-64736c11.cust.bredbandsbolaget.se) |
| 19:26.22 | dgorski | fugitivo |
| 19:26.32 | fugitivo | ? |
| 19:26.35 | dgorski | fugitivo: probably all work fine, haven't really heard of any that didn't |
| 19:26.44 | Zodiacal | telamon okie thank you! i still have some tweaking to do, but ill play with it... just learning.. |
| 19:26.46 | fugitivo | http://www.voip-info.org/tiki-index.php?page=Asterisk+Channel+Bank |
| 19:26.47 | jaiger | fugitivo, I use a TA750 |
| 19:26.49 | dgorski | fugitivo: I liked the rhino so I got me one |
| 19:26.55 | fugitivo | jaiger: any problem with it? |
| 19:27.10 | jaiger | no, I kindof like it but I have nothing to compare it to |
| 19:27.34 | jaiger | only pseudo-issue is echo but I don't think that's the fault of the CB |
| 19:27.43 | dpolitech | are the 750s that much cheaper then the Rhinos? |
| 19:28.00 | fugitivo | jaiger: any feature is missing, echo, or anything else i should take in consideration? |
| 19:28.03 | dpolitech | I'm shopping around right now as well |
| 19:29.05 | jaiger | fugitivo, you'll need/want hardware ecan |
| 19:29.16 | fugitivo | jaiger: hmmm |
| 19:29.24 | fugitivo | jaiger: how much is that? |
| 19:29.49 | jaiger | I got mine on ebay. If I were to buy new for a customer I'd probably go rhino or something else from voipsupply |
| 19:30.01 | cpm | rhino |
| 19:30.09 | *** join/#asterisk vgster (n=vg@spc1-ledn1-3-0-cust136.seac.broadband.ntl.com) |
| 19:30.21 | fugitivo | rhino is at same price as an audiocodes gateway here |
| 19:30.33 | jaiger | fugitivo, you can get it on the T1 cards from digium or a few hundred $US on ebay for Tellabs equipment |
| 19:30.39 | ^Accie^ | anyone with experience in BRI / AVM C2 here? |
| 19:30.46 | cpm | the rhino comes with a T1 card |
| 19:31.26 | jaiger | cpm, what do you mean it comes with a T1 card? |
| 19:31.47 | *** join/#asterisk bprice20 (n=brandon@Dynamic-216.120.224.156.hrnoc.net) |
| 19:32.13 | fugitivo | isn't an audiocodes gateway at the same price as a rhino channel bank? |
| 19:32.15 | *** join/#asterisk htims (n=pd@p5486F1ED.dip.t-dialin.net) |
| 19:32.36 | bprice20 | what is the variable to set the max length of a call? |
| 19:33.01 | justinu | voipsupply.com rocks! |
| 19:33.19 | [av]bani | audiocodes are way spendy |
| 19:33.21 | jaiger | I've probably spent ~US$400 on tellabs hardware ecan and another US$500 on TA750 w/12xFXS+4xFXO |
| 19:33.38 | jaiger | all on ebay |
| 19:33.43 | [av]bani | yay? |
| 19:34.13 | *** part/#asterisk krisguy (n=krisguy@h216-170-039-057.adsl.navix.net) |
| 19:34.38 | Zodiacal | telamon the goto if isn't working.. |
| 19:35.03 | Zodiacal | telamon the first one you gave me allways returns true, the second one you gave me allways returns false |
| 19:35.06 | Zodiacal | it seems |
| 19:38.04 | bprice20 | I got it its absolute timeout |
| 19:38.04 | mroth_imm | anyone know what type of channel this message is describing: Avoided deadlock for '0x94d100', 10 retries! |
| 19:39.26 | dgorski | quit |
| 19:39.54 | *** join/#asterisk [Outcast] (n=outcast@222-152-110-218.jetstream.xtra.co.nz) |
| 19:40.14 | salviadud | who has fwd? |
| 19:40.16 | salviadud | i wanna talk |
| 19:40.18 | salviadud | im so boooooored |
| 19:42.06 | dgorski | Hmmhesays: I just set up a voicemail box for USER instead of 1234 and was able to dial in and leave voicemail. So what part are you saying doesn't work? |
| 19:42.48 | salviadud | how come the ladies don't use asterisk? |
| 19:42.50 | salviadud | hugh! |
| 19:43.00 | *** join/#asterisk justnulling2 (i=justnull@ool-18bab443.dyn.optonline.net) |
| 19:43.43 | justnulling2 | i am getting protocol application invalid on 7960 and it is not pulling any files from tftp, how can i fix it? |
| 19:44.03 | clive- | is anyone using Sirrix cards here ? |
| 19:44.10 | zigman | yes |
| 19:44.12 | zigman | clive- |
| 19:44.13 | ^Accie^ | Hi all. got an asterisk installation here, an AVM C2 card and p2p BRI setup. in need of someone who could give me some pointers on getting it to work, managed to dial in to the demo voice from the outside but somehow I messed up the configs. willing to pay for some consulting. anyone? |
| 19:44.49 | clive- | zigman, hi, I am looking for the new drivers, and the URL is very well hidden, do you know where I can find them ? |
| 19:45.00 | Zodiacal | anyone know whats wrong with my gotoif syntax? heres the simple test code: http://pastebin.com/587490 |
| 19:45.51 | zigman | clive- try to send them an email |
| 19:46.16 | clive- | zigman, I did, I was hoping to work on it tonight, oh well...:( |
| 19:47.00 | *** join/#asterisk sulex (n=sulex@pdpc/supporter/active/sulex) |
| 19:47.55 | dgorski | costrand: docs/README |
| 19:48.04 | zigman | clive- hold on a sec |
| 19:48.06 | dgorski | costrand: I know seems obvious... |
| 19:48.24 | dgorski | constrand: see costrand... ;) |
| 19:49.26 | zigman | nope sorry |
| 19:49.37 | zigman | i though i might have a new version |
| 19:49.40 | zigman | but i don't |
| 19:49.42 | zigman | sorry |
| 19:50.26 | clive- | zigman thanks, I wish these sirrix people would put a link to the driver downlaod page on their website |
| 19:50.40 | zigman | get a quadbri instead ;) |
| 19:52.04 | stoffell | hey ^Accie^, where are you from.. ? |
| 19:52.24 | clive- | I prefer eicon's wiith chan_capi truthfully, but I already havethese sirrix cards |
| 19:54.19 | *** join/#asterisk frenzy (n=frenzy@196.45.144.40) |
| 19:54.24 | frenzy | hello all |
| 19:54.32 | clive- | howzit frenzy |
| 19:54.37 | frenzy | whats the correct path for svn release * 1.2.5? |
| 19:54.44 | frenzy | svn checkout http://svn.digium.com/svn/asterisk/branches/1.2 asterisk-1.2.5 |
| 19:54.46 | frenzy | ? |
| 19:54.53 | clive- | looks right |
| 19:54.58 | *** join/#asterisk backblue (n=moo@87-196-12-102.net.novis.pt) |
| 19:55.01 | frenzy | clive-: good you? |
| 19:55.10 | Abydos313 | svn 1.2.0 works for me..not 1.2.5 |
| 19:55.24 | frenzy | howcome? |
| 19:55.25 | clive- | fine thanks:), just anoyed with myself that I cant find sirrix download page |
| 19:55.47 | clive- | abydos that sounds strange |
| 19:56.10 | Abydos313 | says file not found i think |
| 19:56.35 | dgorski | clive-: when you find it you should update the wiki then you'll know where to look ;) |
| 19:56.41 | Abydos313 | yet 1.2.0 still compiles 1.24 |
| 19:57.24 | Abydos313 | i went back to cvs :)) |
| 19:57.28 | dgorski | what about 'wget ftp://ftp.digium.com/pub/telephony/asterisk/asterisk-1.2.5.tar.gz' ;) |
| 19:57.30 | frenzy | lol |
| 19:58.14 | dgorski | constrand: did you look in the source under docs/README ? |
| 19:58.37 | dgorski | constrand: there is a whole section on config format |
| 19:58.42 | clive- | dgorski , yup:) |
| 19:58.51 | *** join/#asterisk Gunnar (n=gunnar@34.80-203-209.nextgentel.com) |
| 19:59.15 | *** join/#asterisk ac3c (n=chatzill@dhcp2-211.si.umich.edu) |
| 19:59.28 | *** join/#asterisk jcims (n=jcims@rrcs-24-172-217-2.central.biz.rr.com) |
| 20:00.24 | *** join/#asterisk chr|s_ (n=chris@217.171.52.61) |
| 20:00.54 | *** join/#asterisk [Airwolf] (n=airwolf@attilla.nl) |
| 20:03.04 | frenzy | Asterisk SVN-branch-1.2-r12072, Copyright (C) 1999 - 2006 Digium, Inc. and others. |
| 20:03.10 | frenzy | is that 1.2.5 ? |
| 20:04.46 | *** join/#asterisk constrand (n=constran@rrcs-67-79-88-194.sw.biz.rr.com) |
| 20:04.51 | chr|s_ | frenzy, cat .version |
| 20:05.05 | *** join/#asterisk Qwell[] (i=north@unaffiliated/qwell) |
| 20:05.12 | Juggie | frenzy, no thats not 1.2.5 |
| 20:05.24 | *** join/#asterisk UdontKnow (i=udontkno@freenode/staff/udontknow) |
| 20:05.42 | *** join/#asterisk Deep6 (n=DEEP6@208.38.35.162) |
| 20:08.35 | Zodiacal | anyone knwo whats wrong with this gotoif statment? exten => *66,2,GotoIf($["${mytrunk}" = "yes"]?3:6) |
| 20:08.47 | dpryo | Is it possible to make calls with asterisk+soundcard? (mic+speakers).. Or do one need a, say sip-client? |
| 20:08.48 | frenzy | chr|s_: |
| 20:08.56 | frenzy | chr|s_: cat .version in cli? |
| 20:08.59 | chr|s_ | hey frenzy |
| 20:09.00 | chr|s_ | yes |
| 20:09.08 | frenzy | no such command |
| 20:09.10 | chr|s_ | I am just looking, is 1.2.5 has a .version |
| 20:09.16 | Katty | hi lads. |
| 20:09.26 | Beirdo | Heya, Katty |
| 20:09.29 | *** join/#asterisk Rhizome (n=rhizome@imta1.csuohio.edu) |
| 20:09.34 | chr|s_ | frenzy, you have cat |
| 20:09.35 | Katty | Beirdo: (= |
| 20:09.47 | frenzy | chr|s_: what do you mean |
| 20:09.52 | constrand | never seen GotoIf |
| 20:10.07 | chr|s_ | cat just shows a file contents, but it seems 1.2.5 doesn't have a .version |
| 20:10.15 | chr|s_ | or is that made on first build? I dunno |
| 20:10.38 | frenzy | I grabbed 1.2.5 using svn checkout http://svn.digium.com/svn/asterisk/branches/1.2 asterisk-1.2.5 |
| 20:10.58 | frenzy | shoudnt it be /branches/1.2.5 ? |
| 20:11.02 | chr|s_ | aaah |
| 20:11.06 | chr|s_ | no branch is always 1.2 |
| 20:11.07 | octothorpe_ | katty: what kind of hardphones do you use? |
| 20:11.10 | chr|s_ | all 1.2.* is 1.2 |
| 20:11.12 | chr|s_ | branch |
| 20:11.17 | frenzy | aha |
| 20:11.19 | chr|s_ | that is how versions work (commonly) |
| 20:11.28 | kuku5 | Qwell[]: My * is hanging with this sccp |
| 20:11.36 | frenzy | so i should be running the latest version |
| 20:12.30 | dgorski | *CLI> show version |
| 20:13.02 | Zodiacal | any ideas? |
| 20:13.14 | frenzy | Asterisk SVN-branch-1.2-r12072 built by root |
| 20:13.23 | *** join/#asterisk bcnl (n=mike@S01060002e30bf999.vs.shawcable.net) |
| 20:13.24 | dgorski | What is *66 ? |
| 20:13.27 | constrand | anyone know of a realy good how to that explanes the .conf in detail |
| 20:13.28 | dgorski | that's not valid |
| 20:13.33 | dgorski | sorry, nevermind |
| 20:14.20 | Zodiacal | dgorksi just a test |
| 20:14.36 | Zodiacal | heres my simple 5 line test of gotoif http://pastebin.com/587490 |
| 20:14.41 | Zodiacal | know whats wrong off hand? |
| 20:14.57 | Zodiacal | simpe 9 line test :P |
| 20:14.58 | Zodiacal | whatever |
| 20:15.02 | Zodiacal | simpe = simple |
| 20:15.33 | Zodiacal | it seems that no mater how i write that gotoif statment its either allways true or allways false |
| 20:15.38 | Zodiacal | its so frustrating :) |
| 20:15.40 | Katty | octothorpe_: polycom 500 |
| 20:16.14 | *** join/#asterisk robin_z (n=robin@adsl.redpoint.org.uk) |
| 20:16.20 | *** join/#asterisk usn (n=usn@p54B5B1E9.dip0.t-ipconnect.de) |
| 20:16.21 | robin_z | greeting |
| 20:16.26 | usn | hi there |
| 20:16.28 | octothorpe_ | katty: how do you like it? I umderstand that it supports POE, is this correct? |
| 20:16.56 | Katty | octothorpe_: i suggest googling. |
| 20:16.58 | [TK]D-Fender | Zodiacal : Do a NoOp of that var before your goto to prove whats in it |
| 20:16.59 | Katty | octothorpe_: i like it just fine. |
| 20:17.03 | usn | just try to connect first time to my brand-new-built 1.2.5. Never seen asterisk before, pray for me :) |
| 20:17.19 | [TK]D-Fender | octothorpe_ : The 50x needs a special cable to support PoE |
| 20:17.23 | robin_z | so I have * running nice for internal phones .. connecting external phones throught the firewall, i see packets coming into * towars port 5060 and .. * ignores them. what setting am I missing? |
| 20:17.30 | Zodiacal | tkd-fender i do, well, after a goto i read it.. it shows yes |
| 20:17.49 | Zodiacal | tkd-fender Executing NoOp("SIP/100-94da", ""yes"") in new stack |
| 20:17.57 | Zodiacal | hrm "" is odd |
| 20:18.07 | [TK]D-Fender | Zodiacal : thats your problem :) |
| 20:18.08 | *** join/#asterisk dja (n=alden@zaphod.math.ohio-state.edu) |
| 20:18.15 | [TK]D-Fender | Zodiacal : Show me where you set that value |
| 20:18.19 | *** join/#asterisk SwK[Work] (n=SwK@64.89.118.139) |
| 20:18.26 | Zodiacal | <PROTECTED> |
| 20:18.32 | Zodiacal | well mytrunk |
| 20:18.37 | Zodiacal | i was changing it around |
| 20:18.39 | robin_z | it seems to only respond to sip packets from the local network, packets from outside, even though they make it across the firewall and to the * box are ignored |
| 20:18.47 | [TK]D-Fender | Zodiacal : Not "show me in CLI", show me in extensions.conf |
| 20:18.53 | *** join/#asterisk prongils (n=el_terri@67.58.10.46) |
| 20:19.03 | Zodiacal | tk-d-fender http://pastebin.com/587490 |
| 20:19.10 | octothorpe_ | Katty: I tried googling, but I like real-world, end-user stuff better, some say they support POE, but they really only support "fake POE" (not standard 48V) |
| 20:19.27 | fu3 | what do you guys think about using saccharin in coffee? |
| 20:19.37 | RoyK | ~seen zoa |
| 20:19.42 | jbot | zoa <n=kkk@pirus.securax.be> was last seen on IRC in channel #asterisk, 14d 2h 12m 17s ago, saying: '1000 interrupts per second are ok'. |
| 20:19.42 | fu3 | i find it delicious |
| 20:19.47 | constrand | saccharin? |
| 20:19.50 | fu3 | yeah |
| 20:19.52 | Zodiacal | fu3 it used to have warning lables that said it caused cancer to labritory animals |
| 20:19.54 | fu3 | sweet-n-low |
| 20:19.58 | fu3 | has saccharin. |
| 20:20.22 | fu3 | yeah but that was because they gave 50,000 times the regular dosage to a rat, and then found out that rats have a protein in their bladders that react to it and cause the cancer. |
| 20:20.23 | Zodiacal | fu3 http://www.cspinet.org/new/saccharn.htm |
| 20:20.28 | fu3 | thats why the label is gone |
| 20:20.47 | fu3 | god dammit |
| 20:20.52 | fu3 | i'm going to have cancer :( |
| 20:20.56 | Zodiacal | :) |
| 20:21.09 | fu3 | oh well.. either this or another thing. |
| 20:21.16 | *** join/#asterisk |omni| (i=rob@c-67-185-96-86.hsd1.wa.comcast.net) |
| 20:21.28 | fu3 | see.. I wish I could go without any kind of sweetner |
| 20:21.32 | fu3 | but I think the coffee here sucks so much |
| 20:21.37 | *** join/#asterisk multispirit (n=tankist@82.209.226.193) |
| 20:21.37 | fu3 | that I cant drink it really without. |
| 20:21.38 | *** join/#asterisk [TK]D-Fender (n=joe@toronto-HSE-ppp4122655.sympatico.ca) |
| 20:21.41 | [TK]D-Fender | eek |
| 20:21.43 | |omni| | anyone have problems getting 79xx to register from behind NAT |
| 20:21.48 | fu3 | every now and again, someone brings in delicious coffee and it rules. |
| 20:21.48 | *** part/#asterisk jcims (n=jcims@rrcs-24-172-217-2.central.biz.rr.com) |
| 20:21.49 | robin_z | I used to have 2 sugars in my tea |
| 20:21.50 | fu3 | you can drink that with no sugar. |
| 20:22.01 | robin_z | indeed |
| 20:22.07 | fu3 | tea rules. |
| 20:22.09 | fu3 | no question about it. |
| 20:22.19 | robin_z | now I do, and I would never manage to drink it with sugar again .. |
| 20:22.22 | robin_z | its awful |
| 20:22.26 | fu3 | perhaps that is the solution.. switch to tea. |
| 20:22.39 | RoyK | fu3: i buy beans and grind it myself and create the coffee with a coffe press mug. that is good |
| 20:22.44 | constrand | not enof caffene |
| 20:22.46 | RoyK | no need for sugar in that |
| 20:22.55 | octothorpe_ | |omni| is that with the SIP or SCCP image? |
| 20:22.55 | fu3 | yeah.. i'd like to do that RoyK.. but I dont know how really. or even where to start. |
| 20:23.00 | robin_z | anyway ... |
| 20:23.00 | Zodiacal | tkd-fender okie, when i set the var it has to be no "" and when i read it, it should have "" :) works great now thanks! |
| 20:23.03 | |omni| | SIP image |
| 20:23.07 | robin_z | back to my SIP problem ... |
| 20:23.09 | robin_z | :) |
| 20:23.09 | fu3 | haha |
| 20:23.09 | constrand | get a french press |
| 20:23.09 | *** join/#asterisk Jestre (n=ack@dargo.trilug.org) |
| 20:23.11 | fu3 | sorry for the distration |
| 20:23.17 | fu3 | distraction :) |
| 20:23.24 | |omni| | 7.5 (which apparently has a NAT related bug) but I can't even get it to reg once |
| 20:23.32 | |omni| | Sipura works fine |
| 20:23.33 | RoyK | fu3: as constrand says. it doesn't cost much |
| 20:23.35 | |omni| | on the same LAN |
| 20:23.36 | robin_z | why does * ignore external sip packets, but respond perfectly to local network ones? |
| 20:23.38 | RoyK | fu3: and get good coffee |
| 20:23.47 | *** join/#asterisk Possible (n=Babbel@babbel.xs4all.nl) |
| 20:23.55 | fu3 | yeah.. I dont know where to get good coffee then. |
| 20:24.01 | fu3 | I doubt the 'beans' at walmart are good |
| 20:24.02 | prongils | Hello, anyone know of a way on the CISCO 79XX to move the softkeys around, for example to move the Transfer softkey to the 1st page where you dont have to push "More..." to get to it |
| 20:24.06 | fu3 | they're the same ones since 2000 :) |
| 20:24.10 | Possible | Lino`: you around ? |
| 20:24.29 | constrand | $15 bag at starbucks is always good |
| 20:24.30 | octothorpe_ | |omni| : Haven't had issues registering, but have had RTP problems with poor NAT devices |
| 20:24.43 | fu3 | no starbucks here |
| 20:24.46 | fu3 | nor is there one within 100 miles |
| 20:24.51 | Jestre | When checking voicemail, doesn't asterisk use the context from the sip definition? |
| 20:24.54 | robin_z | I see this coming in: |
| 20:25.01 | RoyK | fu3: mail order! |
| 20:25.03 | robin_z | 21:21:02.179819 IP adsl.redpoint.org.uk.15708 > localnet.sip: UDP, length: 698 |
| 20:25.06 | fu3 | from where? |
| 20:25.10 | fu3 | what kind of coffee is good? |
| 20:25.14 | robin_z | but noi response |
| 20:25.21 | |omni| | okay...something is goofy..I better check my TFTP.. the settings are loading..sort of..but it still pointing to the wrong CM...that will do it :P |
| 20:25.22 | robin_z | my tcpdump filter is: |
| 20:25.35 | robin_z | tcpdump host 212.18.243.31 and port 5060 |
| 20:25.46 | Katty | octothorpe_: k |
| 20:26.01 | octothorpe_ | yes Katty |
| 20:26.01 | RoyK | fu3: blue java, monsoon malabar, burbon jungle..... |
| 20:26.02 | [av]bani | \o> |
| 20:26.03 | RoyK | lots |
| 20:26.11 | fu3 | cool.. |
| 20:26.18 | robin_z | I dont see any "bad registration from blah" in the console either |
| 20:26.19 | RoyK | blue java is safe |
| 20:26.20 | fu3 | i just want good coffee with minimal hassle |
| 20:26.21 | Katty | octothorpe_: ? |
| 20:26.27 | *** join/#asterisk darkskiez (n=darkskie@bb-87-81-62-203.ukonline.co.uk) |
| 20:26.43 | RoyK | malabar is strong and dark, full of flavour |
| 20:27.03 | RoyK | it's really a lot of difference |
| 20:27.22 | octothorpe_ | Katty, I am here |
| 20:27.32 | Katty | octothorpe_: i can see that. |
| 20:28.23 | octothorpe_ | I saw [tk]D-Fender's message, that helps, that is what I call "fake" voip |
| 20:28.48 | octothorpe_ | excuse me, "fake POE" |
| 20:28.57 | *** join/#asterisk psymin (n=psymin@psymin.mso.montana.com) |
| 20:29.33 | octothorpe_ | katty and fender: thanks |
| 20:30.21 | constrand | is it neccicary for a voip servec plane |
| 20:30.37 | Qwell[] | prongils: sccp? You'll have to change the code |
| 20:30.42 | octothorpe_ | constrand: ? |
| 20:30.57 | constrand | sorry |
| 20:30.58 | Qwell[] | It *IS* possible to make it changable via a setting...but, you'll have to write some code |
| 20:31.19 | robin_z | ahh pooo |
| 20:31.30 | constrand | do i have to have a voip call plan |
| 20:31.54 | prongils | Qwell[] damn, its using SIP to an asterisk box |
| 20:32.02 | Qwell[] | then you're screwed. ;) |
| 20:32.04 | octothorpe_ | constrand: to make outbound calls, yes |
| 20:32.10 | prongils | damn u cisco :( |
| 20:32.13 | constrand | with my nbx it take care of everything |
| 20:32.21 | robin_z | OK, my sip phone on my netwrok at home is on 192.168.1.160 ... it accesses the remote site through some NAT, and at the far end the * server tries to respond not to me, remoteley, but to the local 192.168.1.160 ... |
| 20:32.47 | robin_z | is this where STUN comes into it? |
| 20:32.49 | jsharp | Do you have nat=yes in your sip.conf for the phone? |
| 20:32.54 | constrand | how much is it for unlimited long distance usaly |
| 20:32.55 | robin_z | hmmm ... |
| 20:33.06 | robin_z | constrand: unlimited amounts |
| 20:33.07 | *** part/#asterisk Samoied (n=Samoied@popeye.opens.com.br) |
| 20:33.17 | Qwell[] | ~unlimited |
| 20:33.19 | jbot | methinks unlimited is <Nugget> unlimited voip == punch the monkey to win a free ipod |
| 20:33.50 | octothorpe_ | ~google unlimited voip |
| 20:33.52 | constrand | ok any sugestions on a voip company |
| 20:34.27 | octothorpe_ | I like iax.cc / sixtel for US calls, not to sure about international |
| 20:34.30 | [TK]D-Fender | octothorpe_ : It's not quite "fake", but not "out of the box" either. |
| 20:34.40 | [TK]D-Fender | octothorpe_ : What are you lookingfor in a phone? |
| 20:34.45 | Qwell[] | [av]bani: die |
| 20:34.52 | file | [TK]D-Fender: !!!!!!! |
| 20:34.54 | file | Qwell[]: !!!!!! |
| 20:35.01 | Qwell[] | file: !!!! |
| 20:35.02 | _Sam-- | constrand : its hard for people to tell you which voip companies are good, since alot of it will be detemermined by your routes the provider(s). |
| 20:35.05 | [av]bani | Qwell[]: do you have smartnet? |
| 20:35.06 | Qwell[] | one ! per letter |
| 20:35.10 | constrand | mostly cheep long distance |
| 20:35.14 | Qwell[] | [av]bani: umm...let's pretend I do |
| 20:35.25 | [av]bani | Qwell[]: rumor has it a sip image for 7970 was released today |
| 20:35.34 | Qwell[] | Mr File...can you do me a favor? |
| 20:35.37 | Qwell[] | [av]bani: oh? |
| 20:35.45 | psymin | If anyone has advice with Local Number Portability that they'd like to share, please message me =] .. having difficulty with a local provider |
| 20:35.46 | file | like what? |
| 20:35.48 | Qwell[] | I can access the site - I just can't download |
| 20:35.52 | [av]bani | Qwell[]: yep, if you have smartnet you could go check if its deception&lies or not |
| 20:36.03 | Qwell[] | file: Pass a message along to tuxinator_linux, if you see him? |
| 20:36.07 | Qwell[] | [av]bani: Lemme take a look |
| 20:36.10 | file | Qwell[]: sure |
| 20:36.12 | *** join/#asterisk guilherme-jorge (n=guilherm@200.155.185.1) |
| 20:36.13 | [TK]D-Fender | file: !!! |
| 20:36.14 | octothorpe_ | [TK]D-Fender: Alot, I woud like it to have out-of-the-box POE (I have a midspan poe that is unused), and it needs to be wall-mountable, I prefer IAX2, but understand I probably have to accept SIP |
| 20:36.29 | file | I haven't seen his wife either :\ |
| 20:36.33 | [av]bani | Qwell[]: supposedly cisco has finally unified their sip builds across all phones with the latest 7.5 update |
| 20:36.33 | Qwell[] | his wife? |
| 20:36.42 | file | yes, he's married :P |
| 20:36.44 | [av]bani | Qwell[]: and hopefulyl fixed the serious 7.5 bugs :) |
| 20:36.50 | Qwell[] | yes, I know that. But, his wife on IRC? |
| 20:36.57 | file | on IRC? hehe no |
| 20:37.00 | Qwell[] | oh :p |
| 20:37.00 | guilherme-jorge | hello all, I would like to discover the codec used in a call,trough |
| 20:37.03 | file | Google Talk/AIM |
| 20:37.06 | Qwell[] | ahh |
| 20:37.19 | Qwell[] | I guess they're moving this weekend or something, or maybe this past weekend |
| 20:37.28 | [TK]D-Fender | octothorpe_ : Wall mountable sounds like just a basic phone then (emergency / courteousy). The IP 301 is wall mountable and with a PoE cable would be manageable. |
| 20:37.31 | Qwell[] | recall him saying "the weekend before VON" |
| 20:37.38 | guilherme-jorge | hello all, I'ld like to discover the codec used in a call, through AGI script. Is it possible? |
| 20:38.27 | *** join/#asterisk zotz (n=zotz@24.231.32.85) |
| 20:39.16 | [TK]D-Fender | guilherme-jorge : sure. Might take a few steps though. First you'd probably pull off the channel name and then inquire on it and parse out the codec. |
| 20:40.00 | octothorpe_ | [TK]D-Fender: Yes, but this is actually for a customers home phone, they don't want "extra" cables, and I would like normal POE, I know, I'm picky, although a basic phone is all that is really needed. |
| 20:40.51 | octothorpe_ | [TK]D-Fender: I will show them the IP 301 and see what they think, thanks |
| 20:41.05 | guilherme-jorge | <[TK]D-Fender>: how do I do this? |
| 20:41.39 | [TK]D-Fender | octothorpe_ : Ther are other models, but they either cost more than I suspect you'd like to pay, or too seriously comprimise on quality. |
| 20:41.47 | constrand | vonage has a really nice small buisness plan 49.99 |
| 20:42.12 | [TK]D-Fender | guilherme-jorge : You'll have to look into this. AGI passes the channel name, then if its SIP then parse out a "sip show channels" to find the call. |
| 20:42.35 | octothorpe_ | [TK]D-Fender: Thanks |
| 20:43.07 | robin_z | OK :) |
| 20:43.19 | robin_z | so .. now I have a connection, but |
| 20:43.23 | robin_z | no audio :( |
| 20:43.37 | Qwell[] | [av]bani: Well, I did find sccp 8.0 |
| 20:43.38 | robin_z | I get a ringing .. but thats a SIP thing right? |
| 20:43.46 | Qwell[] | 8.0.2, for almost all phones |
| 20:43.57 | Qwell[] | http://www.cisco.com/cgi-bin/tablebuild.pl/ip-7900ser You can see them with just a cco login (free) |
| 20:44.47 | [av]bani | Qwell[]: sccp 8.0? o_o |
| 20:44.59 | Qwell[] | yeah |
| 20:45.07 | [av]bani | is that a beta or somethign? |
| 20:45.18 | Qwell[] | nope.. |
| 20:45.28 | Qwell[] | released Mar 6th |
| 20:45.30 | [av]bani | i had someone check sccp for 7970 last night and it was 7.0(3) |
| 20:45.32 | [av]bani | hah |
| 20:45.42 | Qwell[] | They're newbs |
| 20:45.50 | [av]bani | hmmm i guess i shall have to poke this person about the sip image |
| 20:46.00 | Qwell[] | <PROTECTED> |
| 20:46.01 | Qwell[] | Firmware files only for 7970/71 IP Phone SCCP - for CCM releases through v5.0 |
| 20:46.12 | [av]bani | ahaha.. youhave to have a developer support contract |
| 20:46.18 | Qwell[] | can't find the SIP page, except for 7940/7960 |
| 20:46.19 | [av]bani | to get access to sip |
| 20:46.24 | Qwell[] | no, I have a regular cco login |
| 20:46.58 | Qwell[] | got the link to it? |
| 20:47.15 | [av]bani | i mean, you have to have a developer contract to get access to sip 7970 |
| 20:47.20 | [av]bani | a regular smartnet wont do it |
| 20:47.22 | Qwell[] | well, show me the link, and we'll see :p |
| 20:47.57 | guilherme-jorge | <[TK]D-Fender>: My problem is related to codec g729, I need a route to specific way when the calling send the call using g729, so I need discover the codec in the moment that SIP terminate the codec negociation... Any idea? |
| 20:48.28 | robin_z | jsharp: ok setting nat=yes partially did it ... |
| 20:48.29 | [av]bani | Qwell[]: says something about .cop files |
| 20:48.48 | robin_z | jsharp: I can dial ... it rings .. but then no audio on the call |
| 20:48.51 | [TK]D-Fender | guilherme-jorge : I gave you a hint on how, so start exploring it. |
| 20:49.01 | *** join/#asterisk Los415 (n=los@ssf-office.corp.race.com) |
| 20:49.07 | Qwell[] | oh shit, wait |
| 20:49.17 | Qwell[] | <PROTECTED> |
| 20:49.17 | Qwell[] | 7970/71 IP Phone SIP phone load - for CCM v5.0 |
| 20:49.23 | [av]bani | Qwell[]: :)) |
| 20:49.33 | Qwell[] | I knew I saw that somewhere :p |
| 20:49.47 | Qwell[] | 8.x probably started supporting it |
| 20:50.01 | *** join/#asterisk marv[work] (n=timr@64.89.118.139) |
| 20:50.10 | [av]bani | wonder if it works at all |
| 20:50.19 | robin_z | sigh .... |
| 20:50.20 | [av]bani | i guess they got sip/sccp unified on the xml front also |
| 20:50.29 | Qwell[] | good |
| 20:50.33 | [av]bani | eg, same code now runs the minibrowser so theres no difference in functionality |
| 20:50.37 | robin_z | connecting stuff should not be this hard :( |
| 20:50.38 | Qwell[] | did 7911 support sip before? |
| 20:51.02 | [av]bani | dont think so |
| 20:51.08 | [av]bani | does it now? |
| 20:51.08 | Qwell[] | 7911, 7941/7961, 7971 are all sip-ified |
| 20:51.12 | [av]bani | oooo |
| 20:51.12 | robin_z | sip // nat .. :( |
| 20:51.16 | [av]bani | nifty |
| 20:51.23 | Qwell[] | get a cco, and check out that link I posted |
| 20:51.32 | [av]bani | what link? the filename? |
| 20:51.32 | robin_z | this is not an office, its a hell with flourescent lights |
| 20:51.35 | *** join/#asterisk _Vile (n=vile@90.b160.bendtel.net) |
| 20:51.36 | Qwell[] | http://www.cisco.com/cgi-bin/tablebuild.pl/ip-7900ser |
| 20:51.52 | Qwell[] | can't download, but you can see what's available |
| 20:52.58 | [av]bani | <PROTECTED> |
| 20:52.58 | [av]bani | There are currently no files for this type. |
| 20:53.00 | UdontKnow | robin_z: hell should use incandescent |
| 20:53.04 | Qwell[] | eh? |
| 20:53.10 | [av]bani | thats what i get on that page |
| 20:53.12 | Qwell[] | nice |
| 20:53.28 | [av]bani | cisco's pages have a LOT of issues |
| 20:53.30 | Qwell[] | it looks like the .cop stuff is for ccm5 only |
| 20:53.31 | [av]bani | a lot of pages link to 404 |
| 20:53.41 | [av]bani | .cop are just renamed tar.gz |
| 20:53.49 | Qwell[] | nice |
| 20:53.54 | [av]bani | like .jar |
| 20:53.58 | [av]bani | (go cisco! yay!) |
| 20:54.07 | GerbilWrk | has anyone used voipreach.net? |
| 20:54.17 | Qwell[] | I wonder if these 7941/7961 images will work on 7940/7960 |
| 20:54.23 | UdontKnow | just remember suffix isnt extension |
| 20:54.26 | [av]bani | not likely, they are differnt phones |
| 20:54.39 | UdontKnow | suffix is merely informative |
| 20:55.10 | [av]bani | Qwell[]: if you feel like bricking your 7940/7960s, go ahead and try :)) |
| 20:55.57 | Qwell[] | heh |
| 20:56.03 | robin_z | OK, so ... |
| 20:56.08 | Qwell[] | I might try when I get home |
| 20:56.18 | Qwell[] | brb |
| 20:56.27 | robin_z | sip wise, it seems to be sending the port 5060 stuff bacwards and forwarrds OK, |
| 20:56.41 | robin_z | but the audio data its still sending towards local net 192.168.1.160 |
| 20:56.50 | RoyK | nat?yes |
| 20:56.50 | robin_z | instead of towars me, remotely |
| 20:56.54 | robin_z | yeah |
| 20:57.02 | robin_z | nat on both ends |
| 20:57.51 | robin_z | dialling works |
| 20:57.56 | robin_z | but audio doesnt |
| 20:58.06 | robin_z | I can see the packets coming out of * |
| 20:58.17 | RoyK | client (nat) (internet) (nat) asterisk? |
| 20:58.26 | robin_z | yezs |
| 20:58.42 | [TK]D-Fender | robin_sz : Its a workable situation... just need to get a few settings right |
| 20:58.47 | robin_z | on the * I see the audio data going towars 192.168.1.160 |
| 20:58.56 | robin_z | which is the local ip of my client |
| 20:59.14 | robin_z | on the * netwoirk its probalky the ip of the level detector in the girls toilets |
| 20:59.16 | constrand | is there any free GUI's i know of a couple pay ones |
| 21:00.13 | robin_z | [TK]D-Fender: well, I have the 'sontrol' part working OK, I can dial ...and the remote phone in the office rings |
| 21:00.13 | constrand | my boss is a moron and if theres any command line work he freaks out and callsme every five mins |
| 21:00.32 | [TK]D-Fender | robin_sz : pastebin your sip.conf |
| 21:00.36 | [TK]D-Fender | ~pb |
| 21:00.40 | jbot | pb is, like, a place to paste your stuff without flooding the channel - try http://pastebin.com/, or http://pastebin.ca |
| 21:00.50 | robin_z | [TK]D-Fender: all of it? |
| 21:01.02 | robin_z | or just that phone? |
| 21:01.06 | [TK]D-Fender | robin_sz : General + phone defs |
| 21:01.16 | [TK]D-Fender | both ends of the failed conversation. |
| 21:01.29 | robin_z | ok ... some passwrod mangling required ;) |
| 21:01.56 | Hmmhesays | oh chan_sip is giving me hell today |
| 21:02.27 | [Outcast] | me too! |
| 21:02.37 | Hmmhesays | oh? |
| 21:03.30 | [Outcast] | i upgrade to 1.2.5 now i am having some issues with devices behind nat. |
| 21:03.58 | *** part/#asterisk Jestre (n=ack@dargo.trilug.org) |
| 21:04.55 | constrand | sorry about asking newbie questions |
| 21:05.04 | justinu | anyone here know definity stuff? |
| 21:05.32 | [Outcast] | you talking about the citel gateway |
| 21:06.16 | Qwell[] | [av]bani: http://www.cisco.com/univercd/cc/td/doc/product/voice/c_ipphon/english/ipp7961/relnotes/index.htm |
| 21:07.24 | Qwell[] | http://www.cisco.com/univercd/cc/td/doc/product/voice/c_ipphon/english/ipp7960/relnotes/80_100rn.htm |
| 21:07.48 | robin_z | [TK]D-Fender: http://pastebin.ca/44688 |
| 21:08.23 | *** join/#asterisk Tamarisk (n=adrian@user-8054.l6.c4.dsl.pol.co.uk) |
| 21:08.23 | Qwell[] | http://www.cisco.com/en/US/products/hw/phones/ps379/prod_release_notes_list.html |
| 21:08.25 | Qwell[] | wee! |
| 21:08.36 | robin_z | wee? |
| 21:08.48 | Qwell[] | indeed! |
| 21:09.24 | GerbilWrk | i need a reliable provider with great quality and uptime. Teliax isn't cutting the mustard anymore. Any recommendations? |
| 21:09.38 | robin_z | switzernet! |
| 21:10.22 | GerbilWrk | what is this switzernet you speak of? |
| 21:10.30 | jsharp | sneakernet |
| 21:10.47 | *** join/#asterisk nDuff (n=ccd@64.128.31.220) |
| 21:10.49 | robin_z | its a reliable provider with great quality and uptime. |
| 21:11.03 | [av]bani | Qwell[]: hmm, prod_release_notes_list doesnt list sip for 7970 |
| 21:11.03 | robin_z | with good rates in and around Switzerland |
| 21:11.10 | [TK]D-Fender | robin_sz : You are missing your localnet declaration in [general]. Its essential for it to work. |
| 21:11.15 | GerbilWrk | ok, need good rates in and in the US |
| 21:11.29 | robin_z | [TK]D-Fender: ok .. wait .. Might just not hav pastedy |
| 21:11.31 | [TK]D-Fender | robin_sz : And you should have nat=yes in [general] as well |
| 21:11.42 | robin_z | in general too? |
| 21:11.46 | [av]bani | er yes it does |
| 21:11.48 | nDuff | Users at my site using SIP phones are complaining that using *67 to disable Caller ID has no effect (while it works correctly for folks using chan_zap). Any hints as to how I can make this work for the SIP phones? |
| 21:11.50 | [av]bani | wtf, feb 9? |
| 21:12.06 | robin_z | [TK]D-Fender: 99% of my phones are non-nat though |
| 21:12.09 | [TK]D-Fender | robin_sz : yes, as well as the peer |
| 21:12.20 | robin_z | oh, ok |
| 21:12.21 | Qwell[] | it does too |
| 21:12.22 | robin_z | :) |
| 21:12.22 | [TK]D-Fender | robin_sz : Localnet will tell * whats what. |
| 21:12.28 | robin_z | righty |
| 21:12.36 | Qwell[] | Cisco Unified IP Phone 7970G/7971G-GE Release Notes for Firmware Release 8.0(2) for Cisco Unified CallManager 5.0, 4.2, 4.1, 4.0, 3.3 (SIP) New! |
| 21:12.40 | Qwell[] | http://www.cisco.com/en/US/products/hw/phones/ps379/prod_release_note09186a00806208e3.html |
| 21:14.31 | *** join/#asterisk Possible (n=Babbel@babbel.xs4all.nl) |
| 21:15.01 | [av]bani | wonder if the new sccp fixes the placed calls list |
| 21:16.11 | robin_z | [TK]D-Fender: ok, those changes made, the problem is unchanged |
| 21:16.19 | robin_z | [TK]D-Fender: yes I did restart it |
| 21:16.48 | [TK]D-Fender | robin_sz : pastebin a better picture for me if you can then... |
| 21:17.02 | robin_z | sureley, because the remote and local networks are both 192.168.1.* it wont have any luck deciding based on that rule? |
| 21:17.34 | [TK]D-Fender | robin_sz : won't matter. What is on the receiving end? |
| 21:17.41 | robin_z | if I do tcpdump host 192.168.1.160 and udp |
| 21:17.52 | robin_z | when the voicmail answers, I see lots of packets |
| 21:18.09 | robin_z | .160 is the ip of my client |
| 21:18.44 | *** part/#asterisk mroth_imm (n=chatzill@63.65.26.220) |
| 21:18.45 | *** join/#asterisk ikey (n=kiran@213.210.245.87) |
| 21:18.47 | _Sam-- | [av]bani : i am making outgoing calls....SIP-->Softswitch-->PSTN-->cell phone the cell phone rings...but when i answer the cell phone, the remote asterisk side doesnt see the call answered any ideas? |
| 21:18.52 | robin_z | they *shold* be heading towards 212.foo |
| 21:19.35 | [TK]D-Fender | robin_sz : I mean what kind of phone, etc? |
| 21:19.43 | chr|s_ | just a query, i am looking fora definitive installation guide for asterisk, anyone read any good ones? |
| 21:19.49 | [TK]D-Fender | robin_sz : It'll have to know for itself that its behind NAT as well. |
| 21:19.50 | robin_z | oh .. GXP2000 :( |
| 21:20.08 | robin_z | with new sefl-destrcut firmware |
| 21:20.16 | chr|s_ | I have checked asterisk.org, but the install guide isn't even what you get in the book, I want to make sure I can reproduce a 'clean' install each time without irregularities |
| 21:20.17 | [TK]D-Fender | chr|s_ : Installing * is a piece of cake... CONFIGURING it is a slightly different matter |
| 21:20.23 | [TK]D-Fender | chr|s_ : Try these |
| 21:20.25 | [TK]D-Fender | ~thebook |
| 21:20.26 | jbot | i guess thebook is Asterisk: The Future of Telephony, released under a Creative Commons license and available at http://www.asteriskdocs.org << Read the book online! |
| 21:20.28 | [TK]D-Fender | This FIRST. |
| 21:20.31 | [TK]D-Fender | then... |
| 21:20.32 | [TK]D-Fender | ~docs |
| 21:20.33 | jbot | extra, extra, read all about it, docs is probably Documentation can be found at http://www.digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or http://www.voip-info.org/wiki-Asterisk or http://www.asteriskdocs.org or http://www.oreilly.com/catalog/asterisk or http://www.asteriskguru.com, or http://www.astmasters.net/howtos.html |
| 21:20.57 | chr|s_ | [TK]D-Fender, I have installed it once before, but it was so much fudging with stuff that wasn't mentioned, you never know if something you do will cause problems later on |
| 21:21.07 | [TK]D-Fender | I follow the FTP method for grabbing stable releases personally and it works without a hitch for me every time. |
| 21:21.21 | russellb | [TK]D-Fender: awesome :) |
| 21:21.39 | chr|s_ | same here - just in the build process, sometimes it says first modprobe wctdm before you build pibpri |
| 21:21.45 | [TK]D-Fender | russellb : A glowing review to be sure! |
| 21:22.00 | *** join/#asterisk darby_t (n=tom@abbh191.neoplus.adsl.tpnet.pl) |
| 21:22.05 | chr|s_ | but I found I also had to modprobe zaptel,which weasnt mentioned, this stuff really annoys me when you feel you had to hack a build, rather than follow simple stesp |
| 21:22.08 | [TK]D-Fender | chr|s_ : I always compile everything first, THEN modprobe things. |
| 21:22.16 | russellb | that's awesome to hear. :) There was no "releases" really when I got into this :-p |
| 21:22.23 | chr|s_ | aaah, I wonder why the documentation DOES show it like that |
| 21:22.31 | chr|s_ | which is why i wondered if someone had written a simple sane version |
| 21:22.37 | chr|s_ | thanks |
| 21:22.45 | [TK]D-Fender | russellb : Or whatever the term one would choose for whats linked right off asterisk.org :) |
| 21:23.08 | [TK]D-Fender | chr|s_ : For an extra step that falls out of order it shouldn't be a big deal... |
| 21:23.10 | russellb | well, yeah, heh. the ftp releases are tags on the release branch when we decide it's ready :) |
| 21:23.21 | russellb | you're not supposed to have to worry about how they get there, sorry :) |
| 21:24.00 | Qwell[] | [av]bani: It might... |
| 21:24.02 | [TK]D-Fender | russellb : Rest assured that I live with it... saves me about haveing to think about having SVN installed and I can DL them on windows PC's if need be as well. |
| 21:24.18 | robin_z | [TK]D-Fender: OK, I set "use NAT ip" on the grandstream .. I now see packets head out towards my ip ... they dont get here ,but hey .. they head off int he right direction, so its a start |
| 21:24.46 | robin_z | [TK]D-Fender: obviously, that has stopped the phoen working with the local * box, but .. its a start :) |
| 21:25.07 | [TK]D-Fender | robin_sz : Good, keep working at it. you should be forwarding 5060, 10000-20000 UDP to *, and nothing on the client side. |
| 21:25.43 | robin_z | thats probably right so far ... |
| 21:26.07 | robin_z | its a Sonicwall on the * side ... so yu just tick "sip" in the voicemail conf and pray |
| 21:27.10 | *** join/#asterisk Defraz (i=t0tal@tim.mychoice.cc) |
| 21:27.29 | [TK]D-Fender | robin_sz : you might want to try settinga different SIP port on it as well.... |
| 21:29.30 | Zodiacal | anyone know why my FOP trunks vanished? |
| 21:29.48 | Zodiacal | nm i think i know why |
| 21:31.01 | Zodiacal | acctualy i don't know |
| 21:31.01 | Zodiacal | :P |
| 21:31.22 | jsharp | Someone slathered them with vanishing creme? |
| 21:31.27 | robin_z | [TK]D-Fender: I think the packets leave the romote net and traverse the internet cloud .. but the client side nat doesnt know where to send them when they arrive ... i dopnt see them on the network after the router with tcpdump |
| 21:32.28 | [TK]D-Fender | robin_sz : just noticed, you missed the extremely important "qualify=yes" in the phone's sip.conf entry :) |
| 21:32.39 | robin_z | oh, whats that do? |
| 21:32.49 | [TK]D-Fender | That keeps the UDP like "alive" |
| 21:32.53 | nDuff | Can I use the SIP headers to determine the caller ID to be used for a call initiated from a SIP phone? |
| 21:32.59 | [TK]D-Fender | otherwise it'll go "unreachable" |
| 21:33.00 | robin_z | uh huh ... |
| 21:33.06 | [TK]D-Fender | yeah.. ESSENTIAL ;) |
| 21:34.49 | robin_z | mmm ... ok |
| 21:34.52 | robin_z | but no change |
| 21:35.23 | robin_z | I use this phoen with other networks, like, voipgate etc, no problemo |
| 21:38.13 | [TK]D-Fender | hmmmm, ok, well that all I'll be able to do for now, BBL |
| 21:38.28 | nDuff | When dialing with *67, my (Sipura) phone puts "Anonymous" in the SIP headers (From, Contact, etc) -- but because the name is specified in sip.conf on the Asterisk box, the directive is ignored. |
| 21:39.14 | RoyK | nDuff: does the sipura set callingpres? |
| 21:40.47 | nDuff | RoyK: is that a separate header? If so, no. |
| 21:41.37 | RoyK | dunno enought sip |
| 21:41.38 | RoyK | sorry |
| 21:41.56 | UdontKnow | hmmm |
| 21:42.09 | UdontKnow | I need a simple config for an asterisk |
| 21:42.11 | nDuff | RoyK: hrm. If I had a mechanism to look at the SIP headers and invoke SetCallingPres in my dialplan, that would work -- but the headers would need to be exposed. |
| 21:42.19 | UdontKnow | gonna have 2 softphones and 1 ATA together |
| 21:42.22 | UdontKnow | only that |
| 21:42.24 | UdontKnow | nothing else |
| 21:42.36 | UdontKnow | no dialing out anywhere |
| 21:42.51 | UdontKnow | anyone know where I can get something like that? |
| 21:43.23 | UdontKnow | or willing to show me how to do it? I dont remember anything from when I configured asterisk the first times... |
| 21:43.58 | RoyK | nDuff: try to 'sip show channel xxx' to see if it's set on incoming. i think it should say |
| 21:46.15 | justnulling2 | anyone knows cisco 7960? |
| 21:47.01 | Katty | Hmmhesays: mew? |
| 21:47.03 | guilherme-jorge | Hello all, are there a way to execute any command (for example, "sip show channel") in a AGI script? |
| 21:47.40 | RoyK | guilherme-jorge: lookup the wiki agi page. it tells you what you need :) |
| 21:47.59 | guilherme-jorge | RoyK: mmm... do u have a link? |
| 21:48.00 | dgorski | nDuff: show application sipgetheader |
| 21:48.12 | x86 | hmm |
| 21:48.22 | nDuff | dgorski: I don't have it installed. |
| 21:48.22 | x86 | can you have aliases for inbound SIP extensions? |
| 21:48.33 | RoyK | ~wiki agi |
| 21:48.57 | *** part/#asterisk Tamarisk (n=adrian@user-8054.l6.c4.dsl.pol.co.uk) |
| 21:49.10 | x86 | like have the main extension be the person's actual numerical extension, and have an alias of the person's name which points to the numerical extension? |
| 21:49.25 | dgorski | Goto |
| 21:49.30 | nDuff | dgorski: ahh, that's an Asterisk 1.1 addition -- I'm still on 1.0.7 |
| 21:49.43 | x86 | so instead of 105652356990137@some.sip.server.co.uk, you can have mynameisbob@some.sip.server.co.uk |
| 21:49.50 | dgorski | nDuff: understand, I was looking at a 1.2.4 box |
| 21:50.03 | nDuff | (management won't allocate me time to upgrade... grr) |
| 21:50.34 | dgorski | nDuff: it's never good to rush into an upgrade anyway... |
| 21:50.53 | Qwell[] | "rush"...heh |
| 21:50.54 | dgorski | 1.0.7 is good - I've got 1.0.8 on most stable boxes |
| 21:50.57 | Qwell[] | 1.2.5... |
| 21:51.06 | Qwell[] | You're way beyond "rush"ing |
| 21:51.13 | dgorski | lab |
| 21:51.17 | *** part/#asterisk dpolitech (n=Owner@207.224.48.130) |
| 21:51.28 | dgorski | I don't use 1.2 anywhere I care about just yet |
| 21:52.03 | *** join/#asterisk |omni| (i=cathode@c-67-185-96-86.hsd1.wa.comcast.net) |
| 21:52.22 | Hmmhesays | lad dee da dee da |
| 21:53.15 | dgorski | x86: exten => mynameisbob,1,Goto(105652356990137,1) |
| 21:55.15 | Katty | Hmmhesays: eeeeelp. |
| 21:55.17 | Katty | Hmmhesays: sort of. |
| 21:55.43 | Katty | Hmmhesays: asterisk is trying to tell me something and it does not parse. |
| 21:56.40 | Katty | i think a module has gone out :< |
| 21:59.09 | DaPrivateer | anyone have an example DP to manage IVR greetings on *? |
| 21:59.26 | Hmmhesays | whats your error message? |
| 22:02.00 | x86 | dgorski: ah... cool :) |
| 22:02.43 | x86 | dgorski: can i put the aliases in their own context? if so, how do i call the other context from Goto? |
| 22:05.24 | x86 | dgorski: also, do i have to define mynameisbob somewhere in sip.conf? |
| 22:05.54 | x86 | what i'm trying to do is make it so someone can use sip:mynameisbob@some.server.co.uk as a SIP URI instead of having to remember the number |
| 22:06.17 | x86 | ~wiki goto |
| 22:06.45 | [av]bani | so helpful, that jbot. |
| 22:06.49 | x86 | ~docs goto |
| 22:07.08 | x86 | ~goto |
| 22:07.10 | jbot | well, goto is considered harmful |
| 22:07.40 | jsharp | Boot to the head. |
| 22:07.46 | X-Rob_ | ~lart jbot |
| 22:07.51 | X-Rob_ | d'oh. |
| 22:07.58 | mikefoo | Anyone know off hand of a rtp capturing tool, to convert into wav files, therefore having recorded calls. |
| 22:10.14 | GerbilWrk | has anyone used Terravon.com for DID? |
| 22:12.02 | UdontKnow | dpryo: I once set up asterisk for a friend of mine who is blind... she used the asterisk console to place the calls |
| 22:12.03 | *** part/#asterisk psymin (n=psymin@psymin.mso.montana.com) |
| 22:12.11 | UdontKnow | dpryo: using alsa/oss you can, sure |
| 22:12.53 | brad_mssw | GerbilWrk: nope, looks interesting though, don't happen to have an ip to traceroute, do you? |
| 22:12.56 | *** part/#asterisk frenzy (n=frenzy@196.45.144.40) |
| 22:13.19 | GerbilWrk | not yet |
| 22:13.50 | *** join/#asterisk masonf (n=masonf@dungle.vineyard.net) |
| 22:14.56 | brad_mssw | GerbilWrk: used iax.cc/sixtel, teliax, junctionnetworks, sellvoip.net though |
| 22:15.26 | masonf | is there any way I can make my music on hold music louder? |
| 22:15.32 | *** join/#asterisk HamYaI (i=HamYai@125.24.9.224) |
| 22:15.35 | masonf | its barly audible. |
| 22:15.43 | UdontKnow | masonf: normalize your audio files? |
| 22:16.08 | masonf | explain? or should I google? |
| 22:16.10 | fu3 | masonf eh |
| 22:16.12 | fu3 | im an fmason |
| 22:16.29 | *** join/#asterisk shido6 (n=shido6@d221-68-216.commercial.cgocable.net) |
| 22:16.37 | masonf | fu3: ? |
| 22:16.41 | masonf | fu3: free mason? |
| 22:16.42 | fu3 | yes |
| 22:16.46 | _Sam-- | the mode in musiconhold.conf can change the volume of music on hold mp3 |
| 22:17.05 | UdontKnow | masonf: well, there are many audio processing programs that can normalize your audio files |
| 22:17.22 | UdontKnow | masonf: I dont remember one on the tip of the head, but you can find them easily |
| 22:17.38 | _Sam-- | mode=mp3 = loud |
| 22:17.44 | _Sam-- | mode=queitmp3 = soft |
| 22:17.51 | _Sam-- | mode=queitmp3 = soft = default |
| 22:17.57 | GerbilWrk | brad_mssw, which did you find to be the most reliable with the best call quality? |
| 22:18.23 | x86 | can anyone try to hit me with sip at sip:bryce@shellshark.net please? |
| 22:18.27 | brad_mssw | GerbilWrk: well, teliax truthfully ... but latency and packetloss can be issues at times ... still waiting on an east-coast server |
| 22:18.37 | GerbilWrk | :( |
| 22:18.42 | _Sam-- | brad_mssw : i switched to SIP at teliax, and my troubles are long gone |
| 22:18.45 | GerbilWrk | was afraid you were going to say Teliax |
| 22:18.54 | _Sam-- | in fact, all my providers who were using IAX sounded twice as nice using SIP |
| 22:18.57 | brad_mssw | GerbilWrk: just in the beginning stages with sellvoip, they look promising, lowest latency, just trying to sort out IAX DTMF issues |
| 22:19.08 | x86 | my real server is not "shellshark.net" but i added an SRV record to my DNS this morning and it should direct |
| 22:19.25 | brad_mssw | GerbilWrk: junction is ok, but latency is all over the board, and I wouldn't trust sixtel at all for origination |
| 22:19.41 | x86 | ooo |
| 22:19.45 | _Sam-- | brad_mssw : do you use iax or sip at teliax? |
| 22:19.48 | x86 | use telasip for origination :) |
| 22:19.57 | x86 | _Sam--: i use IAX with teliax |
| 22:20.08 | _Sam-- | try sip, i think you will be pleasantly surprised. |
| 22:20.12 | brad_mssw | _Sam--: yeah, using iax still |
| 22:20.18 | _Sam-- | my calls sound like POTs lines now |
| 22:20.19 | GerbilWrk | I'm using SIP with teliax and i'm still experiencing problems |
| 22:20.23 | x86 | my IAX setup works fine with Teliax |
| 22:20.26 | x86 | no problems at all |
| 22:20.41 | brad_mssw | _Sam--: I'm pretty much in a holding pattern waiting to see what teliax comes up with for an eastcoast server |
| 22:20.41 | x86 | GerbilWrk: what kind of upstream bandwidth do you have? |
| 22:20.56 | _Sam-- | brad_mssw : im on the east, and connect to colorado fine now |
| 22:20.57 | GerbilWrk | DS3 |
| 22:21.09 | x86 | GerbilWrk: how saturated is it? |
| 22:21.11 | _Sam-- | im telling you, for as long as takes to try SIP, you are missing out |
| 22:21.24 | _Sam-- | just switch an outgoing call to SIP and see for yourself |
| 22:21.31 | x86 | _Sam--: the first time i have a lost packet i might try it ;) |
| 22:21.43 | brad_mssw | _Sam--: well, it's possible with SIP and re-invites, that the RTP audio path would be much shorter, since it wouldn't have to relay through teliax |
| 22:21.43 | *** join/#asterisk insomnia4 (n=dasver96@c-68-46-200-53.hsd1.ks.comcast.net) |
| 22:21.44 | _Sam-- | x86 : you are 1 out 10 with no problems IAX to teliax. |
| 22:21.46 | x86 | i've not had a single problem with IAX and Teliax |
| 22:21.52 | brad_mssw | _Sam--: so I'd need to enable reinvites I suppose |
| 22:21.52 | _Sam-- | everyone else has problems |
| 22:22.02 | [av]bani | i have problems with teliax |
| 22:22.03 | x86 | i'm using u-law |
| 22:22.24 | [av]bani | i think teliax just has crappy jitter buffers or something |
| 22:22.24 | GerbilWrk | x86, most of the time the calls are going through we are only at about 50-70% usage on the ds3 |
| 22:22.25 | _Sam-- | anywho, im winding down my teliax days, i can connect 2ms away to pSTN now |
| 22:22.31 | brad_mssw | _Sam--: it's probably worth a shot ... |
| 22:22.35 | x86 | can someone please try a test SIP call to sip:bryce@shellshark.net please? |
| 22:22.40 | [av]bani | _Sam--: got your voip all working sip'd up then? |
| 22:22.46 | brad_mssw | _Sam--: to who ? |
| 22:22.48 | _Sam-- | [av]bani : yes i did |
| 22:22.52 | _Sam-- | brad_mssw : my ISP is also a CLEC |
| 22:22.57 | [av]bani | and they finally figured out how to remove their head from their ass? |
| 22:23.10 | _Sam-- | [av]bani : i had to pull it out slowly without causing too much pain. |
| 22:23.14 | [av]bani | they finally realized they didnt need your mac address? |
| 22:23.18 | _Sam-- | but they figured it out, and the first call i made, worked. |
| 22:23.25 | _Sam-- | ya, its all IP based auth |
| 22:23.49 | [av]bani | you tried reinvite with latest fw yet? |
| 22:23.53 | _Sam-- | next i have to port my numbers off teliax |
| 22:24.04 | _Sam-- | figures, just when it starts working perfectly for me |
| 22:24.15 | *** join/#asterisk kpettit (n=keith@69.15.174.113) |
| 22:24.25 | _Sam-- | im all canreinvite = no |
| 22:24.26 | [av]bani | so how is your clec's EC? |
| 22:24.59 | _Sam-- | ive only made about 5 calls so far, but no echo at all yet. |
| 22:25.08 | _Sam-- | right now they have me locked down to local calls for testing |
| 22:25.14 | [av]bani | :) |
| 22:25.37 | _Sam-- | but by showing them how to do this, i just showed them how they can do this in the future without needing those 'iad' devices |
| 22:25.45 | _Sam-- | they are about as excited as i am really |
| 22:26.22 | _Sam-- | the only thing so far i havent been able to do so far is set my caller id for outgoing calls |
| 22:26.34 | _Sam-- | it seems like they've got that locked down somehow |
| 22:26.37 | [av]bani | :) |
| 22:26.48 | *** join/#asterisk gerbrandoo (n=gerbrand@godzilla.xs4all.nl) |
| 22:27.02 | [av]bani | i bet they just wanted to sell you a pbx and phones |
| 22:27.07 | [av]bani | hence they were pimping the iad |
| 22:27.23 | _Sam-- | nah, they knew i wasnt going to buy that crap...and they had no idea how else to do it without an iad. |
| 22:27.41 | _Sam-- | when i was describing how we would do it, they were just blank looks all around |
| 22:27.52 | [av]bani | yep sounds like a clec |
| 22:27.54 | justinu | so aren't you back to where you started? no outbound clid? |
| 22:27.55 | _Sam-- | like you said, that tekelec softswitch is nothing more than a crappy asterisk box |
| 22:28.00 | [av]bani | sip? der? whats a packet? |
| 22:28.13 | [av]bani | oh, its running * ? |
| 22:28.37 | _Sam-- | justinu : yes. except instead of a PRI and a crappy clec who doesnt know anything....i am back with a plain t1 instead of PRI, with no caller ID, and a clec that doesnt know anything :) |
| 22:29.02 | _Sam-- | so yes...i am back to square 1, except i have 24 vs 23 channels, and can use g729 etc to send more calls over the same pipe |
| 22:29.48 | _Sam-- | [av]bani : no its not an asterisk box that i connect to...but it could be. instead of using that tekelec thing, asterisk would do the same functions. |
| 22:29.56 | [av]bani | and you can use the pipe for intarweb |
| 22:29.59 | justinu | tekelec is pretty pimp |
| 22:30.05 | *** join/#asterisk StevePL (n=stevel@216.62.85.65) |
| 22:30.09 | justinu | i'd like one |
| 22:30.23 | gerbrandoo | anyone here that knows a bit about the time-out function of the Dial command? when i use Dial on a Zap interface, the timeout never happens. Asterisk bridges the calls when the phone is still ringing. I wish to set a timeout so i never get a voicemail system, instead it should try a different number after around 15 seconds. how could i do that? |
| 22:30.30 | _Sam-- | the one im connecting to, in my opinion at least, really does nothing at all that we couldnt do with an asterisk machine. |
| 22:30.35 | _Sam-- | again, that is my opinion, not fact. |
| 22:30.49 | justinu | probably not, but it can do a lot more volume, and be more reliable |
| 22:31.01 | [av]bani | so you need to call tekelec and have them instruct your clec how to passthrough remote cid |
| 22:31.55 | x86 | can someone please try a test SIP call to sip:bryce@shellshark.net? |
| 22:32.17 | terrapen | does anybody know a good wiki page that can assist me in building me a distributed Asterisk architecture? |
| 22:32.30 | [av]bani | x86: whats that, hot ph0n3 s3xx0r? |
| 22:32.39 | terrapen | I want to separate my voicemail server from the main server |
| 22:32.47 | x86 | [av]bani: you know ;) |
| 22:33.01 | x86 | [av]bani: it's only set verbose 99999999999999999999 per minute ;) |
| 22:33.06 | terrapen | I'm not sure how you can transfer calls off to another Asterisk server |
| 22:33.21 | x86 | terrapen: using SIP REINVITE? :) |
| 22:33.23 | terrapen | I'm thinking that I want separate VM servers at each of our location |
| 22:34.41 | justinu | you guys ever work with the guys at voipconnection? |
| 22:34.44 | x86 | usually a good idea ;) |
| 22:34.51 | terrapen | i had hoped that the ORA Asterisk box would tell more about it |
| 22:34.54 | justinu | they're pretty nice guys to work with |
| 22:34.56 | terrapen | after all, it gave me the idea |
| 22:35.18 | justinu | cut me a nice discount on ip301s + poe cables, and they don't try and charge 300 bucks shipping like atacomm does |
| 22:35.40 | terrapen | justin: website? |
| 22:35.46 | justinu | voipsupply.com |
| 22:35.55 | *** part/#asterisk HamYaI (i=HamYai@125.24.9.224) |
| 22:35.56 | x86 | atacomm are bitches! |
| 22:36.08 | x86 | i bought a phone from them last monday and i'm still waiting on it |
| 22:36.39 | x86 | friday i call them up, they had arranged for fedex pickup of it to be shipped out, gotten a tracking number, and FORGOT TO PUT THE DAMN THING ON THE TRUCK |
| 22:36.45 | justinu | i haven't had problems with them shipping, but they charge a lot |
| 22:36.47 | justinu | for shipping |
| 22:36.58 | [av]bani | justinu: $40 to ship an spa3000. hah. |
| 22:37.00 | x86 | justinu: cheapest i could find for a BT101 ;) |
| 22:37.06 | justinu | they wanted 300 bucks for my 19 phone order. |
| 22:37.12 | [av]bani | yeah, insane |
| 22:37.20 | justinu | voipsupply , 30 bucks |
| 22:37.25 | [av]bani | they must not have wanted your 19 phone order very much :) |
| 22:37.25 | *** join/#asterisk |omni| (i=rob@c-67-185-96-86.hsd1.wa.comcast.net) |
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| 22:37.38 | justinu | nope, and I was happy to do business with voipsupply, after meeting them at astricon |
| 22:37.53 | [av]bani | well, voipsupply completely borked my order |
| 22:37.56 | x86 | i would never recommend atacomm to anyone... |
| 22:37.56 | [av]bani | but they made up for it |
| 22:38.05 | justinu | christ |
| 22:38.08 | justinu | der |
| 22:38.13 | justinu | i mean voipconnection.com |
| 22:38.15 | justinu | not voipsupply |
| 22:38.23 | x86 | then, after they screwed up my shipping, they wouldnt pay to have a phone expedited to me! |
| 22:38.24 | [av]bani | haha |
| 22:38.26 | justinu | pimping the wrong folks |
| 22:38.40 | *** part/#asterisk UlbabraB (n=salama@host241-43.pool8172.interbusiness.it) |
| 22:38.54 | *** part/#asterisk darby_t (n=tom@abbh191.neoplus.adsl.tpnet.pl) |
| 22:38.58 | gerbrandoo | hmm. no solutions for my Dial timeout problem then? |
| 22:39.36 | justinu | so anyways, call up voipconnection if you need help |
| 22:39.38 | [av]bani | polycom should remove(headfromass) and put minibrowser on teh 501 |
| 22:39.39 | Katty | Hmmhesays: heh, someone unplugged line 2. |
| 22:39.47 | justinu | also, i had an emergency situation where I needed an IP301, next day |
| 22:39.57 | justinu | I called voipconnection at 6pm EST, and they still got me my phone next day |
| 22:40.12 | justinu | and the shipping wasn't 100 bucks, either |
| 22:40.22 | [av]bani | sounds like how newegg used to be, before they switched shippers :( |
| 22:40.34 | Hmmhesays | Katty: lovely |
| 22:40.40 | Hmmhesays | i'm still fighting with chan_sip |
| 22:41.28 | kippi | from a handset that is using IAX2 is there away to transfer calls on it |
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| 22:48.51 | gerbrandoo | I will describe my challenge again. I wish to Dial a certain number, if it isn't answered within 15 seconds, call number 2. When I use Dial(Zap/4/0301234567,5), Asterisk says 'Called 4/0301234567' and also immediately 'Zap4-1 answered Zap/1-1', 'Attempting native bridge of Zap/1-1 and Zap/4-1'. But all of this happens when the phone is still ringing, and no one (or voicemail) has actually answered. Where do I start looking dor a solution for thi |
| 22:49.27 | gerbrandoo | So although I set a timeout, it will never ever reach the timeout. |
| 22:49.55 | gerbrandoo | (I use a PSTN line in the netherlands) |
| 22:50.09 | gerbrandoo | and i am quite stuck |
| 22:51.13 | |omni| | so your internal handsets are on zap extensions too? |
| 22:51.23 | *** join/#asterisk squinky86 (n=ASGjon@unaffiliated/squinky86) |
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| 22:51.36 | gerbrandoo | yes |
| 22:51.39 | jenwa | heya |
| 22:52.24 | gerbrandoo | i need a kind of 'hotline' fuctionality. pick up the handset, lookup number in database, dial the number. no quick answer, then lookup next number, dial it, etc. |
| 22:52.50 | backblue | gerbrandoo: hotline its done with the s extension. |
| 22:52.54 | |omni| | so...outbound line is zap/2, handset is zap/1 and when zap/2 calls in to your asterisk box you Dial(Zap/1,5) and it immediately answers? |
| 22:53.05 | *** join/#asterisk justnulling2 (i=justnull@ool-18bab443.dyn.optonline.net) |
| 22:53.22 | jenwa | is asterisk a very viable option for a production environment? |
| 22:53.29 | Hmmhesays | if you know what you're doing |
| 22:53.29 | backblue | jenwa: y |
| 22:53.31 | |omni| | jenwa: , yes |
| 22:53.45 | backblue | jenwa: dumb question, google-> asterisk! |
| 22:54.03 | gerbrandoo | it's the other way around. handset is internal, on zap/1. zap/4 is outbound line. hotline calls are from inside to the world out there |
| 22:54.29 | backblue | how does a software not ready for production have so much community, and so much partners and so much everything...? |
| 22:54.47 | backblue | gerbrandoo: yes, put s extension. |
| 22:55.01 | jenwa | doesnt mean anything |
| 22:55.02 | robin_z | ummm ... if your 12 port 100baseT switch in your spare bedroom is full, does that mean your house os uber geeked? |
| 22:55.04 | jenwa | :-) |
| 22:55.07 | backblue | exten _X.,s,Dial(SIP/blah,5) |
| 22:55.12 | backblue | jenwa: yes it does. |
| 22:55.34 | gerbrandoo | ah ok. great thanks, will try that. |
| 22:55.39 | jenwa | cisco has voip but their system is shit imo |
| 22:56.02 | Hmmhesays | robin_z that could mean many things.... |
| 22:56.09 | Hmmhesays | amongst having to find a girlfriend |
| 22:56.10 | backblue | gerbrandoo: when you pickup your phone, it goes to s extension, in s you do whatever you need. that's hotline feature. |
| 22:56.59 | backblue | jenwa: yes offcourse, cisco its not "very viable option for a production environment" has you sayed. Cisco like to spend money. :P |
| 22:57.19 | gerbrandoo | backblue. yes i have that working: the phone is picked up, and the other number is dialed, and the calls are bridged. the problem is the timeout of the Dial command |
| 22:57.31 | jenwa | well i suppose i will fsck around with it and find out |
| 22:57.33 | kippi | where is a good place to get cheap cisco handsets in the UK? |
| 22:57.48 | jenwa | is there a calculator to determine how much speed you need? |
| 22:57.52 | backblue | gerbrandoo: you have ael and a timeout in dial(). |
| 22:58.11 | backblue | jenwa: speed? which one? |
| 22:58.13 | [av]bani | robin_z: no.. it would have to be 1000bt to be uber |
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| 22:58.30 | Qwell[] | [av]bani: got the firmware yet? :p |
| 22:58.42 | [av]bani | Qwell[]: i was waiting for you to get it for me! |
| 22:58.49 | Qwell[] | not gonna happen :p |
| 22:59.04 | [av]bani | ha |
| 22:59.08 | jenwa | internet speed for one |
| 22:59.23 | backblue | [av]bani: your wrong, myrinet to be ultra geek. |
| 22:59.30 | gerbrandoo | backblue: ael? i use "exten => s,3,Dial(Zap/4/0301234567,15); should I do anything else to really make asterisk enforce the 15 second timeout? |
| 23:00.29 | backblue | gerbrandoo: i dont know when you want to do that, you can code a timeout with ael, or just simply use the timeout feature like you are saying, try both, and use the one you need. |
| 23:01.12 | gerbrandoo | backblue: i am not sure what you mean with 'ael' |
| 23:02.03 | gerbrandoo | backblue: found it |
| 23:02.36 | robin_z | [av]bani: 1000baseT? over copper? euch, how terribly 2005. fibre dude, its the only way ;0 |
| 23:02.51 | robin_z | sigh .... |
| 23:02.58 | UdontKnow | robin_z: 1000 works on copper |
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| 23:03.10 | UdontKnow | robin_z: cat5e and cat6 cables were created for that |
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| 23:03.34 | grendal_prime | hey can you config asterisk to act as a dataprovider for a web enabled phone? |
| 23:03.40 | robin_z | UdontKnow: yeah, ive doen firewire on copper .. |
| 23:03.50 | robin_z | in Avid systems |
| 23:05.29 | robin_z | dear grandstream, any chance of some firmware that can keep the display alive for more than 3 minutes. please? pretty please? yours etc, robin. |
| 23:05.33 | robin_z | oops worng window |
| 23:07.02 | [av]bani | sure :) |
| 23:07.07 | [av]bani | my grandstreams are all fine \o/ |
| 23:07.25 | robin_z | heh, wish mine were |
| 23:07.50 | [av]bani | bah. unless you've discovered at least 20 you arent leet |
| 23:08.06 | *** join/#asterisk sevard (i=pegasus@24-179-181-160.dhcp.dlth.mn.charter.com) |
| 23:08.11 | guilherme-jorge | hello all, I get the channel used in a call, through AGI script, but I'ld like to know which codec it's using. How to do this? I get the channel using $AGI->get_variable("CHANNEL") |
| 23:08.45 | robin_z | when use nat ip is set, it sends that ip in the sip message, even when sending to addresses that fall inside the ip/netmask for the local (non-nat) network |
| 23:09.44 | sevard | Question: I have a MSQL problem. We have the exact same information in the mysql database as in the sip.conf but it seems to break when we use the database. |
| 23:10.18 | [av]bani | robin_z: well you _did_ say "use nat ip" :) |
| 23:10.55 | robin_z | [av]bani: sure .. but make s it tricky when one account is on in internal box, the other account is on an external beyond-the-nat box |
| 23:14.07 | kippi | can anyone tell me what this error is? |
| 23:14.07 | kippi | Mar 6 23:12:56 WARNING[3065]: pbx.c:4664 ast_add_extension2: Unable to register extension '687142', priority 1 in 'incoming', already in use |
| 23:14.53 | gerbrandoo | i found a reference to my problem, and a possible solution: http://lists.digium.com/pipermail/asterisk-users/2004-September/061352.html |
| 23:16.32 | kippi | also does anyone know what this means? |
| 23:16.32 | kippi | Mar 6 23:13:23 DEBUG[3065]: iax2-provision.c:235 iax_provision_version: Unable to create provisioning packet for 'si-00036400160b' |
| 23:16.52 | [av]bani | robin_z: so basically you want a per-account nat ip setting |
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| 23:36.48 | jenwa | ok i think i am missing something here |
| 23:37.11 | jenwa | asterisk is just a package correct? |
| 23:37.35 | jenwa | what os is recommended? |
| 23:37.41 | msw | jenwa: pound key! |
| 23:37.48 | msw | (sorry, shameless plug) |
| 23:37.53 | jenwa | haha |
| 23:38.08 | jenwa | kudos |
| 23:38.51 | Zodiacal | anyone know where recordings are saved when i press *77? |
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| 23:38.56 | jenwa | obviously it can be compiled but i was wondering if one worked better than another |
| 23:39.03 | Zodiacal | i can hear them when i press *99. so they do get saved some where i think |
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| 23:40.00 | msw | jenwa: there are distro-related complications from time to time -- best to start with one that you're familiar with (if asterisk is already available as a precompiled package) or go with an asterisk-specific OS |
| 23:40.28 | msw | jenwa: (like pound key -- which we need to update at some point to get the latest shiny asterisk) |
| 23:40.59 | asterisk99 | Any hints on missing termcap support? I'm using ubuntu and the only termcap package is termcap-compat ... My asterisk compile fails with msg 'termcap support not found' |
| 23:41.08 | *** join/#asterisk lo2 (n=lo2@ti112210a080-1106.bb.online.no) |
| 23:43.23 | riksta | asterisk99: install termcap-devel or libtermcap-devel |
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| 23:44.10 | Pegger | has anyone set up real time billing? |
| 23:46.40 | asterisk99 | riska: ubuntu apt-get cannot find termcap-devel nor libtermcap-devel |
| 23:46.53 | asterisk99 | riksta: ubuntu apt-get cannot find termcap-devel nor libtermcap-devel |
| 23:48.16 | riksta | i know nothing about ubuntu |
| 23:48.43 | twisla | asterisk99: libncurses-dev |
| 23:50.01 | twisla | asterisk99: to be precise: libncurses5-dev - Developer's libraries and docs for ncurses |
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| 23:52.27 | riddlebox | does anyone do any agi programming in python? |
| 23:52.30 | FLeiXiuS | How would I go about getting DTMF being recognized any time during the IVR? |
| 23:53.27 | jenwa | anyone used asterisk in a call center environment |
| 23:54.05 | jenwa | or have any stories/websites about it |
| 23:54.46 | asterisk99 | twisla: yuppers. that worked... I got further... nor it cannot find -lssl |
| 23:54.57 | twisla | asterisk99: libssl-dev |
| 23:55.00 | asterisk99 | twisla: yuppers. that worked... I got further... noW it cannot find -lssl |
