| 00:04.26 | *** join/#asterisk dwery (n=dwery@nslu2-linux/dwery) |
| 00:09.30 | *** join/#asterisk axisys (n=axisys@ip68-98-177-71.dc.dc.cox.net) |
| 00:11.34 | SkykingOH | Anybody care to give me a hand, if I evaluate to a numeric it works, I can't seem to get the gotoif to evaluate an alpha to true |
| 00:13.03 | jaytee | SkykingOH, maybe your neice Penny could help :-) |
| 00:13.20 | jaytee | SkykingOH, just kidding. |
| 00:13.43 | SkykingOH | That's where the NIC came from |
| 00:13.57 | SkykingOH | I got it to work with 10 gotoif's the code is just ugly |
| 00:15.39 | jaytee | SkykingOH, if your evaluating alpha as two strings then you need to enclose each string in quotes as in GotoIf($["${STRINGVAR1}" = "${STRINGVAR2}]?somelabeledpriority) |
| 00:16.05 | jaytee | oops, left off the closing " between the } and the ] |
| 00:16.55 | SkykingOH | I have two quotes |
| 00:17.00 | SkykingOH | GotoIf($["${CALLERID(num):0:2}" = ","]?digitstrip:nodigitstrip) |
| 00:17.12 | SkykingOH | I am evaluating CALLERID if it begins with the comma |
| 00:18.44 | SkykingOH | Even though the CID beigns with , (I checked with a NoOP) it is still going to nodigitstrip |
| 00:19.28 | [TK]D-Fender | SkykingOH: pastebin your complete dilapln exten (all prios) and the CLI output of your failed attempt |
| 00:19.30 | [TK]D-Fender | ~pb |
| 00:19.31 | jbot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste, or , http://bin.cakephp.org/ |
| 00:20.30 | [TK]D-Fender | SkyActually I see it in #freepbx |
| 00:20.50 | [TK]D-Fender | SkykingOH: your problem... you are take the first TWO charaters off the front, not ONE |
| 00:21.04 | [TK]D-Fender | SkykingOH: "${CALLERID(num):0:2} <--- *2* |
| 00:21.14 | SkykingOH | I am trying to free myself of the bondage of FreePBX <<g>> |
| 00:21.16 | [TK]D-Fender | SkykingOH: Should only be 1 |
| 00:21.57 | SkykingOH | I see that, that was leftover from when I was trying to evaluate ,1 ... Let me reload and test |
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| 00:25.42 | jaytee | brought to you by Nabisco |
| 00:26.03 | docid | so, i got an 'interesting' issue, ive got a context set up to pass all calls from 1 t1 port out the second t1 port preserving caller id etc....now that that works just fine i need it to catch certain numbers coming from t1 #1 and catch other numbers from T1 #2 olny when they are incoming from port 2, |
| 00:26.47 | SkykingOH | No difference |
| 00:26.51 | SkykingOH | http://www.pastebin.ca/1353088 |
| 00:27.06 | docid | i need those numbers caught to go to seperate voicemail bboxwa |
| 00:27.09 | docid | boxes |
| 00:27.50 | *** join/#asterisk nullable_type (n=kumana@hq.verbx.net) |
| 00:28.02 | SkykingOH | docid - You need to put the channels from the first PRI in one context and the second in another so you can treat the traffic the way you want |
| 00:28.02 | docid | ugg, im not sure how to state that more clearly |
| 00:28.16 | nullable_type | Hey guys, While in CLI, how do i test a variable like echo ${DATETIME} |
| 00:28.21 | nullable_type | what command |
| 00:28.28 | docid | yes, they are seperate context, but somewhere along the line they get shifted to from-pstn and im haveing trouble finding where |
| 00:30.12 | [TK]D-Fender | SkykingOH: pastebin the new CLI output please |
| 00:30.35 | [TK]D-Fender | nullable_type: You can't. That is a channel variable. CLI isn't a channel |
| 00:30.46 | nullable_type | ic, thanks |
| 00:31.01 | docid | no pri's here, were on old school mf signaling |
| 00:31.03 | dwery | hardwire: I crashed :) |
| 00:31.36 | dwery | hardwire: I'll have to contact sangoma's tech support.... |
| 00:31.42 | *** join/#asterisk coppice (n=chatzill@46.166.17.210.dyn.pacific.net.hk) |
| 00:31.46 | SkykingOH | D-Fender - This system is very busy - Will the output from full | grep TEST do??? That is what I put the NoOP's in for |
| 00:31.57 | SkykingOH | I can certainly sanitize the log |
| 00:32.09 | [TK]D-Fender | docid: Take your incoming call. Look at the CID. If it matches something you care about Goto an exten to do VM. Otherwise dial out your other T1 port |
| 00:32.50 | [TK]D-Fender | SkykingOH: I want to see the failure. also put "" around your CID in the noop |
| 00:33.01 | *** join/#asterisk bmoraca (n=chatzill@209.60.253.58) |
| 00:33.05 | bmoraca | arg |
| 00:33.10 | bmoraca | pacwest sucks |
| 00:33.17 | SkykingOH | Ok will do and I will clean it up |
| 00:34.01 | eric2 | is there something that will recognize the caller saying 'yes' or 'no' as opposed to pressing '1' or '2' ? |
| 00:34.26 | [TK]D-Fender | eric2: CMU Sphinx, or other 3rd party VR packages |
| 00:34.35 | eric2 | aye |
| 00:34.36 | eric2 | tx |
| 00:38.32 | docid | thanks....i keep tryin this |
| 00:41.15 | SkykingOH | D-Fender |
| 00:41.18 | SkykingOH | Updated post |
| 00:41.22 | SkykingOH | http://www.pastebin.ca/1353100 |
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| 00:47.36 | *** join/#asterisk prpa1982 (n=bla@93-136-30-93.adsl.net.t-com.hr) |
| 00:47.47 | prpa1982 | hi all |
| 00:48.11 | prpa1982 | quick question....i am using LinuxMCE which uses Asterisk for its telephony section |
| 00:48.58 | prpa1982 | since i live in Croatia, and here is no VOIP provider, just POTS |
| 00:49.07 | prpa1982 | i was wonderinf if You could help me a little |
| 00:49.26 | prpa1982 | wondering |
| 00:50.49 | [TK]D-Fender | SkykingOH: Only idea coming to mind is that perhaps the "," is being parsed funny and needs to be escaped. |
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| 00:51.20 | SkykingOH | You are suggesting I supply the ASCII code for a comma |
| 00:51.37 | jaytee | # |
| 00:51.37 | jaytee | [Mar 4 19:35:12] DEBUG[6583] pbx.c: Function result is ',12164082009' |
| 00:51.37 | jaytee | # |
| 00:51.37 | jaytee | [Mar 4 19:35:12] DEBUG[6583] pbx.c: Expression result is '0' |
| 00:51.55 | SkykingOH | Yes, that is why I was so confused that it was evaluating wrong |
| 00:51.57 | jaytee | look at the function result, it's like it's not stripping at all |
| 00:52.14 | NovceGuru | patton make decent ATAs? |
| 00:52.15 | SkykingOH | Do I just supply the ascii # in decimal without the quotes |
| 00:53.12 | SkykingOH | jaytee- the fuction result in the Noop, I know I am printing 2 characters of the string |
| 00:53.28 | jaytee | yeah, i saw that |
| 00:53.49 | [TK]D-Fender | SkykingOH: do a "dialplan show from-trunk-avaya" |
| 00:53.52 | jaytee | but the function result in the GotoIf is returning the whole number |
| 00:55.21 | SkykingOH | Let me do the dialplan show |
| 00:55.40 | dwery | given that I have no hope to get the usbfxo working soon, anyone can suggest a SIP ATA with 2x FXO ? |
| 00:56.15 | SkykingOH | If you look the output of the function proceeds the execution of the gotoif |
| 00:56.27 | SkykingOH | I thought that was due to the NoOP |
| 00:56.58 | SkykingOH | While I am making the change to get you the dial plan output what was the syntax to evaluate agains an ascii |
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| 01:00.23 | edoceo | do you all use sox or ffmpeg to resample the a mp3 or wav to gsm? |
| 01:02.07 | SkykingOH | Ok I reposted http://www.pastebin.ca/1353100 |
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| 01:05.00 | orkid_ | i'm wondering what the problem can be. i have: sipura3102->asterisk_server->firewall->asterisk_server_on_internet->betamax |
| 01:05.13 | orkid_ | if i register with asterisk_server_on_internet and call through betamax i can do DTMF |
| 01:05.23 | orkid_ | if i reg with asterisk_server behind firewall i can't |
| 01:05.42 | orkid_ | although I can call just fine, and during key presses asterisk_server reports packet2packet iirc |
| 01:06.06 | orkid_ | but no dmtf. all dtmf stuff is set to rfc2833 in the servers, and nat is on i think in those accounts too |
| 01:06.24 | orkid_ | (although not needed if i go sipura->asterisk_server) |
| 01:06.54 | orkid_ | any ideas on what I could try? Could it be an issue that i'm doing sipura->1.4->1.6->betamax (the version mismatch) |
| 01:06.56 | orkid_ | ? |
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| 01:12.24 | SkykingOH | jaytee and defender....Thanks for your help. Unless you have additional input I am going to try this on a clean install. I have it working with 10 Gotoif statements to test the numerics 0-9 then fall through to the digit strip routing. That accomplishes my goal it just disrupts my entire sense of order in the universe to have so much code for something so simple. |
| 01:13.44 | nullable_type | exten => 1000,n,GotoIf($["${DIALSTATUS}" = "ANSWER"] ? CallAnswered) |
| 01:13.44 | nullable_type | exten => 1000,n(CallAnswered),DoSomething() |
| 01:13.44 | nullable_type | For some reason when i answer the call the above lines doesn't get into stack. Anyone know why? |
| 01:14.33 | *** join/#asterisk areay (n=areay@93-97-161-123.zone5.bethere.co.uk) |
| 01:14.52 | SkykingOH | Laughing, expressions are the problem of the night |
| 01:14.57 | SkykingOH | Thanks again.. |
| 01:15.33 | *** join/#asterisk rhombus (n=rhombus@dsl-vlan435-66-18-218-36.nucleus.com) |
| 01:15.56 | jaytee | SkykingOH, sorry I couldn't be more help. I tested a similar expression but without the comma, just to see if the first 6 digits matched my area code and nxx of my cell and called my * box. it evaluated correctly. I just have no way to send it a comma from my cell. |
| 01:16.15 | rhombus | Has anybody ever encountered a situation where a SIP peer appears in the 'sip show peers' list with absolutely the wrong IP address? |
| 01:16.58 | rhombus | I've got one Aastra set that shows up as having the default gateway address, when it clearly doesn't. |
| 01:19.48 | areay | i downloaded the asteriskNOW vmware image to test the capabilities of asterisk... i've managed to get it working making internal calls between extensions (using ekiga softphone), and i've managed to get ekiga softfone talking to my voip provider... but i can't get asterisk to work with my voip provider... i've set up all the calling rules but ekiga won't connect to an outside line thru asterisk. |
| 01:19.54 | jaytee | [TK]D-Fender, I think you were on to something, it had to be the way the retarded parser was reading the "," because a pure numeric value in the second string worked when I tested it. Can't send myself a comma though to actually prove it. |
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| 01:22.37 | *** mode/#asterisk [+o jtodd] by ChanServ |
| 01:24.30 | areay | i setup the service providers section of the asterisk web gui properly (the same as i have with my softphone, which works)... but it still wont work |
| 01:25.58 | SkykingOH | It's this stupid Avaya sending a comma. I am doing a phased migration of 6 buildings so I had to put this hack on the inter machine trunks. |
| 01:26.03 | SkykingOH | I really appreciate the effort |
| 01:26.05 | SkykingOH | Have a good night |
| 01:30.04 | *** join/#asterisk andresmujica (n=andresmu@ubuntu/member/andresmujica) |
| 01:36.25 | *** join/#asterisk BeeBuu (n=beebuu@218.13.87.175) |
| 01:36.54 | *** join/#asterisk shmaltz (n=chatzill@mail.dmaven.com) |
| 01:37.11 | shmaltz | jbot helo |
| 01:37.12 | jbot | it has been said that helo is the first command issued during smtp |
| 01:37.26 | shmaltz | ~hi |
| 01:37.27 | jbot | hello, shmaltz |
| 01:38.13 | *** join/#asterisk davevg (n=davevg__@nj-67-76-177-147.sta.embarqhsd.net) |
| 01:38.35 | shmaltz | ~everyone |
| 01:38.36 | jbot | i guess everyone is Disconnect, or to suspicious of bobtheavenger, or a communiss |
| 01:39.30 | shmaltz | ~sex |
| 01:39.57 | shmaltz | ~pregnant |
| 01:39.57 | jbot | Yes, shmaltz, and it's your child. |
| 01:39.59 | *** join/#asterisk Deeewayne (n=dwayne@c-76-29-245-9.hsd1.al.comcast.net) |
| 01:39.59 | *** mode/#asterisk [+o Deeewayne] by ChanServ |
| 01:40.15 | shmaltz | ~fun |
| 01:40.16 | jbot | methinks fun is what I'm not |
| 01:40.25 | shmaltz | ~busy |
| 01:41.21 | *** part/#asterisk jbjuly (n=joelbrya@203.177.143.137) |
| 01:42.45 | Corydon76-dig | ~congestion |
| 01:43.44 | shmaltz | looks like jbot is sleepy tonight |
| 01:43.58 | shmaltz | he used to know exactly how to have geeky sex |
| 01:44.01 | shmaltz | he forgot |
| 01:44.47 | *** join/#asterisk LiNeTuX (n=LiNeTuX@64.132.248.206) |
| 01:48.29 | Corydon76-dig | I'm sure Tim Riker would prefer if you didn't teach it back again |
| 01:50.12 | thehar | haha |
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| 02:02.29 | jaytee | ~botsnack |
| 02:02.29 | jbot | jaytee: thanks |
| 02:05.11 | BeeBuu | ~botwork |
| 02:05.58 | *** join/#asterisk andresmujica (n=andresmu@ubuntu/member/andresmujica) |
| 02:08.22 | shmaltz | ah here it is: |
| 02:08.23 | shmaltz | ~sex is alias sex "updatedb; locate; talk; date; cd; strip; look; touch; finger; unzip |
| 02:08.24 | jbot | okay, shmaltz |
| 02:08.29 | shmaltz | ~sex |
| 02:08.30 | jbot | methinks sex is alias sex "updatedb; locate; talk; date; cd; strip; look; touch; finger; unzip |
| 02:09.04 | andresmujica | that's nice!!! |
| 02:09.04 | FunkyGMT | lolll |
| 02:09.30 | *** join/#asterisk ming_zym (n=ming_zym@nat/yahoo/x-407968df81f4836d) |
| 02:09.46 | shmaltz | anyone here using backup exec? |
| 02:10.06 | drmessano | Yeah why |
| 02:10.44 | shmaltz | drmessano, for how long have you been using it? |
| 02:10.58 | shmaltz | I have been using it since version 7 I beleive around 10 years ago |
| 02:11.14 | drmessano | Version 7 was the first I used |
| 02:11.15 | lanning | shmaltz, you forgot "yes" |
| 02:11.21 | shmaltz | since Symantec took it over I have had way more trouble with it |
| 02:11.26 | shmaltz | lanning you are right |
| 02:11.44 | shmaltz | drmessano, what version you using now? |
| 02:11.54 | drmessano | Varies.. 10d mostly |
| 02:12.12 | shmaltz | just upgraded from 10.0 to 10d, what a pain |
| 02:12.17 | shmaltz | running my test backup now |
| 02:12.47 | drmessano | I've done a few dozen of those upgrades..Some easy, some near impossible |
| 02:13.24 | shmaltz | the only reason I ran this upgrade is because we upgraded from SQL 2000 to 2005, since then backup failed. |
| 02:13.44 | drmessano | Make sure you're on the very latest patchlevel too |
| 02:13.47 | shmaltz | but Symantec decided to require you to upgrade instead of having a patch that works to a know problem |
| 02:14.02 | shmaltz | why? some know issues? |
| 02:14.08 | drmessano | Yes, tons |
| 02:14.27 | shmaltz | running them now, thanks |
| 02:14.56 | *** join/#asterisk plasmid2 (n=noway@c-68-80-84-98.hsd1.pa.comcast.net) |
| 02:15.34 | drmessano | BE isnt hard to admin.. Just make sure you dont pussyfoot when fixing a problem.. Resetting credentials, restarting all services, etc.. Do them ALL when you have a problem.. not just one and hope for the best |
| 02:16.45 | drmessano | Fixing BE is like herding cats.. you get 2 in the box, and then go for 1 more, and those 2 jump out.. Best to get a pitchfork, stab 4 or 5 of the cats at once, and throw them all in the box |
| 02:18.20 | shmaltz | ROFL |
| 02:18.36 | shmaltz | anyhow, I have to wait till the backup is done first shoudln't take much longer |
| 02:19.15 | shmaltz | if/when it runs successfuly its a beauty |
| 02:19.20 | *** join/#asterisk ReDNeQ (n=ReDNeQ@70.114.226.70) |
| 02:19.29 | shmaltz | however if you see an error on the logs then you know you are in for a long one |
| 02:19.36 | jaytee | best description of BE I've ever heard ROFL |
| 02:19.41 | shmaltz | it takes hours to troubleshoot |
| 02:20.19 | shmaltz | since a backup takes quite long to run |
| 02:20.22 | shmaltz | then rerun it |
| 02:20.32 | shmaltz | this is my 3rd test run in the last 3 hours |
| 02:20.32 | drmessano | "Hey, I backed up 3 out of 5 days successfully, whats the differential if I dont do all 5?" |
| 02:21.32 | shmaltz | well, one doesn't really need a complete successfull backup EVERY night, BUT if that night that it failed is the night before your server room gets flooded then look for a new job |
| 02:21.45 | jaytee | I remember using Palindrome to backup Netware. It used a Towers of Hanoi algorithm in it's rotation scheme. Grandfather, father, son. You go to restore and it's Grandma, second cousin twice removed and "who the fuck are you?" |
| 02:22.17 | shmaltz | I dont' even do that if I have the choice, I just run a full backup everynight |
| 02:22.24 | shmaltz | the only rotation I do is the media |
| 02:22.35 | shmaltz | I have around 1 month on tape |
| 02:22.40 | drmessano | Backup and Restore kicks ass on BE |
| 02:22.45 | shmaltz | I'm trying to upgrade here to iSCSI |
| 02:22.45 | drmessano | Better than Acronis, IMO |
| 02:22.50 | jaytee | we backup to disk now instead of tape |
| 02:23.06 | jaytee | I still like Acronis for images |
| 02:23.11 | shmaltz | jaytee, it has it's advantages and disadvantages |
| 02:23.15 | drmessano | Acronis is great for images |
| 02:23.19 | drmessano | and bare metal restore |
| 02:23.23 | drmessano | But its wonky |
| 02:23.36 | shmaltz | jaytee, disk lacks good media management, it's expensive |
| 02:23.48 | *** join/#asterisk moy (n=moy@CPE001cdfec4cee-CM00080dab8485.cpe.net.cable.rogers.com) |
| 02:24.04 | jaytee | but we use BE with removable iSATA drives and we used to have so many problems with DLT with Dell Powervaults. |
| 02:24.34 | shmaltz | jaytee, now tell me, which physical drive can you find the backup that was done on Feb 17th? |
| 02:24.46 | drmessano | HAD |
| 02:24.49 | drmessano | -D |
| 02:24.56 | drmessano | Dell robotic drives |
| 02:25.02 | drmessano | ^^^ SUXOR |
| 02:25.03 | shmaltz | whats HAD? |
| 02:25.07 | drmessano | -D |
| 02:25.09 | drmessano | MINUS D |
| 02:25.11 | drmessano | MARK IT ZERO |
| 02:25.45 | shmaltz | I never knew that machine had so much data on it |
| 02:25.54 | shmaltz | it's been running now for 25 minutes over 30GB |
| 02:26.03 | jaytee | shmaltz, have you ever used disk with BE? it's not any more complicated than using DLT tape cartridges and you can do this really cool thing called LABELING. |
| 02:26.14 | drmessano | WUTS A LABEL? |
| 02:26.17 | shmaltz | jaytee, really? |
| 02:26.21 | shmaltz | on disks? |
| 02:26.29 | drmessano | We use disks with BE.. works great |
| 02:26.30 | shmaltz | using partition lables? |
| 02:26.35 | shmaltz | or media lables? |
| 02:26.36 | *** join/#asterisk killown (n=ukendt@unaffiliated/killown) |
| 02:26.43 | jaytee | so, yeah. If you really wanted me to go and find the backup from Feb 17th I could but it would cost you! |
| 02:26.54 | shmaltz | jaytee, seriously how? |
| 02:26.59 | shmaltz | I've tried it and gave up |
| 02:27.12 | shmaltz | jaytee, you using b2d? |
| 02:27.25 | drmessano | We use 2g1c |
| 02:27.40 | shmaltz | what the hell is 2g1c? |
| 02:27.50 | shmaltz | or is that for logic? |
| 02:28.04 | jaytee | 2 girls 1 cup |
| 02:28.22 | shmaltz | :P |
| 02:28.36 | drmessano | http://www.urbandictionary.com/define.php?term=2g1c |
| 02:28.39 | haps | Hello hello. |
| 02:28.40 | shmaltz | jaytee, you use b2d? |
| 02:28.49 | haps | the newb still kicking this dead horse |
| 02:28.59 | haps | [Mar 4 21:26:54] NOTICE[954]: chan_sip.c:14397 handle_request_invite: Call from 'gs102' to extension '5142439603' rejected because extension not found. |
| 02:29.20 | haps | I'm guessing that something is wrong with my extensions.conf |
| 02:29.21 | haps | ? |
| 02:29.23 | jaytee | shmaltz, look I'm not gonna get in a geek arguement here. just use your damn tapes and feel all smug and superior. see if I give a rat's ass! |
| 02:29.29 | haps | I have my external provider and my phone registered |
| 02:29.35 | haps | but i get 404 when i dial anything |
| 02:29.44 | haps | the provider is unlimitel |
| 02:30.00 | haps | and i'm using the extensions.conf they build for you |
| 02:30.14 | haps | I'm sure it's something obvious, I just don't have a clue. |
| 02:30.18 | shmaltz | jaytee, on the contrary |
| 02:30.25 | shmaltz | I want to learn how I went wrong |
| 02:31.01 | jaytee | Colonel Mustard, in the atrium, with a crescent wrench |
| 02:31.08 | shmaltz | my biggest problem was that I tried using b2d, and when swapping drives I had to restart BE services for it to work properly, then when trying to restore I had no clue which drive had it |
| 02:31.25 | k-man | it would be good if there was a directory (in aus.) of 13xxx numbers and their normal number counterpart |
| 02:31.34 | k-man | as 13xxx numbers cost more to call than a normal landline |
| 02:31.57 | shmaltz | k-man, and how will those guys make money? |
| 02:32.36 | shmaltz | in any event I would assume it's like an 800 number in the states wheere you don't really know where it rings to, although it's possible to look it up using SS7 |
| 02:33.39 | k-man | shmaltz: its the phone companies who make money from 13 numbers in Aus. not the receiving company |
| 02:34.34 | k-man | shmaltz: similar to a 1800, its charged at a flat rate to the caller, any aditional cost is passed onto the receiver, however, the flat rate is about $0.30 which is considerably more than a landline call these days |
| 02:35.04 | shmaltz | it aint similar to an 800 number at all |
| 02:35.11 | shmaltz | caller pays nothing for 800 |
| 02:35.38 | k-man | err... well, its kind of similar |
| 02:35.40 | k-man | imho |
| 02:35.46 | k-man | just the caller pays a flat fee also |
| 02:35.53 | k-man | we have 1800 numbers here too which are free for the caller |
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| 02:37.58 | *** join/#asterisk ber___ (i=brad@neu.cow.org) |
| 02:39.03 | ber___ | I would like to create an outgoing call and once it is answered, play an IVR and bridge it to an incoming call |
| 02:39.10 | ber___ | what is the best way to accomplish this? |
| 02:39.26 | [TK]D-Fender | ber___: "call file", or AMI Originate |
| 02:40.32 | ber___ | does that allow me to bridge in an incoming call? |
| 02:40.41 | ber___ | i have used that before to originate a call |
| 02:42.22 | [TK]D-Fender | ber___: How are you getting from "have * place an outgoing call" to having anything to do with an INCOMING call? |
| 02:42.43 | ber___ | i would like to bridge them together |
| 02:42.48 | ber___ | but the only way i can think of doing it |
| 02:42.51 | *** join/#asterisk Defraz (n=T0tal@24-117-236-174.cpe.cableone.net) |
| 02:42.55 | ber___ | is creating a conference for the outgoing call |
| 02:42.56 | [TK]D-Fender | ber___: Your direction makes no sense |
| 02:43.05 | ber___ | then dumping the incoming call into it |
| 02:43.25 | [TK]D-Fender | ber___: STILL makes no sense |
| 02:43.26 | ber___ | ok another example is this |
| 02:43.32 | ber___ | incoming call comes in |
| 02:43.36 | ber___ | i bridge it to its destination |
| 02:44.01 | ber___ | i would like the destination party to have to go thru an IVR before we hangup |
| 02:44.03 | *** join/#asterisk FunkyGMT (n=Hayes@modemcable107.105-58-74.mc.videotron.ca) |
| 02:44.13 | ber___ | but this can be after the incoming party hangs up |
| 02:44.28 | ber___ | essentially the IVR woudl be post-call |
| 02:44.29 | [TK]D-Fender | W-T-F |
| 02:45.12 | shmaltz | ber___, use the h extension |
| 02:45.14 | ber___ | the only way i can figure it out is to dump everyone into a conference |
| 02:46.10 | ber___ | shmaltz it says that is only usable after a call has been terminated i.e. i have hung up on the destination leg |
| 02:46.30 | ber___ | i want to keep the desination on the line once the call is effectively completed (incoming leg hangs up) |
| 02:46.33 | ber___ | and run them thru an IVR |
| 02:46.49 | ber___ | the only way i can think of is to dump everyone into a conference and have IVR be a participant int he conference |
| 02:46.53 | ber___ | so essentially a 3 leg call |
| 02:46.56 | shmaltz | ber___, does that destination person know about it? |
| 02:47.00 | ber___ | yes they do |
| 02:47.13 | shmaltz | ber____, then use the h extension |
| 02:47.35 | shmaltz | it will make sure that when the other party hangs up the remaining one endsup over there |
| 02:47.55 | shmaltz | well maybe I'm wrong and you have to use the g option from app_dial |
| 02:48.07 | shmaltz | in any event one of those should work |
| 02:48.11 | ber___ | ok but there is some mechanism to bridge the still active leg |
| 02:48.15 | ber___ | into some other processing |
| 02:48.46 | shmaltz | ber___, english please? |
| 02:49.03 | shmaltz | for your scenario above I answered you, what else you want? |
| 02:49.17 | ber___ | it was a statement not a question |
| 02:49.20 | ber___ | hence the lack of ? |
| 02:49.45 | ber___ | dial with 'g' option talks about the 'called' party hanging up |
| 02:50.05 | shmaltz | just try it |
| 02:50.09 | shmaltz | one of them will work |
| 02:50.10 | ber___ | i will |
| 02:50.12 | ber___ | thank you |
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| 02:53.31 | docelmo | Has anyone ever experienced with the sip qualify option = yes where it does nothing but scroll on the screen that the peer is reachable? |
| 02:53.40 | docelmo | then not reachable.. it never stops.. |
| 02:54.09 | *** join/#asterisk RichardLynch (n=RichardL@c-98-193-37-55.hsd1.il.comcast.net) |
| 02:54.36 | RichardLynch | Does anybody have a patch to allow Festival() to specify which voice to use? I can't be the only person needing this... |
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| 02:58.11 | docelmo | Why do you want to use Festival when you can get quality voices for almost nothing |
| 02:59.31 | docelmo | check out www.cepstral.com |
| 02:59.43 | russellb | docelmo: well, i guess cepstral _is_ infinitely more expensive than Asterisk :-) |
| 02:59.59 | russellb | (and festival) |
| 03:00.13 | russellb | but yeah, <3 cepstral |
| 03:00.42 | RichardLynch | Boss told me to use festival, I used festival... I'm happy to check out cepstral. Does the asterisk cepstral app allow voice picking from the dial plan? |
| 03:01.40 | RichardLynch | Or am I hand-rolling an AGI to get cepstral? |
| 03:01.49 | [TK]D-Fender | RichardLynch: no, that'd be voice recognition, not TTS |
| 03:05.08 | shmaltz | anyone here ever tried screwing with a known telemarketer that violates do not call lists? |
| 03:06.19 | RichardLynch | shmaltz: Small claims court with a log of what you did and when they called will net you the maximum fine small claims will let you do, probably maybe. |
| 03:06.54 | shmaltz | RichardLynch, that will never happen, first I don't have time for litigation |
| 03:07.28 | shmaltz | I have done other things |
| 03:07.30 | jaytee | do they block their callerid? |
| 03:07.33 | shmaltz | worked quite well so far |
| 03:07.39 | shmaltz | jaytee, some times |
| 03:07.48 | shmaltz | however they are advertising something |
| 03:07.57 | shmaltz | so I get that number of what they are advertising |
| 03:08.24 | shmaltz | I then call that company asking them who they used |
| 03:08.35 | jaytee | if not I'd filter on their number and send their call to a recording of a high pitched squeal with the volume amped and some faint talking in the background ("Don't call here again, asshole!" |
| 03:09.26 | shmaltz | if they refuse to tell me, I have created a little script that changes to random caller id and calls as many simultaneous calls to that company, playing a recording that they should press 1 when they are ready to give me some info |
| 03:09.32 | shmaltz | which in turn call me |
| 03:09.55 | shmaltz | when they call me I aks them again who they used |
| 03:10.01 | shmaltz | if they answer I stop the script |
| 03:10.08 | shmaltz | if they don't I keep running it |
| 03:10.16 | shmaltz | I never had to run it more than 3 minutes |
| 03:10.32 | shmaltz | they realized quite quickly that they have no phones till they answer me |
| 03:10.39 | shmaltz | which has created another thing |
| 03:10.55 | shmaltz | those companies will never ever use that telemarketing company again |
| 03:11.22 | shmaltz | I refuse to tell them what my number is so that *they can remove it* I just tell them scrub it agains the do not call from the gov |
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| 03:16.03 | RichardLynch | Okay, at $30 a pop per voice, cepstral may not be an option for what we are doing... |
| 03:16.19 | shmaltz | RichardLynch, what are you doing? |
| 03:16.42 | RichardLynch | So is there a patch to get Festival() to use a different voice from the dialplan? |
| 03:17.03 | k-man | can one "dial" another SIP device directly on the the internet from asterisk? |
| 03:17.04 | shmaltz | RichardLynch, is there a way to do that from the shell? |
| 03:17.20 | shmaltz | k-man, why not? |
| 03:17.26 | shmaltz | app_dial should work |
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| 03:18.03 | *** mode/#asterisk [+o denon] by ChanServ |
| 03:18.05 | RichardLynch | shmaltz: I might have to write a Lisp wrapper around it, but, yeah, I can do that from the shell. |
| 03:18.12 | k-man | shmaltz: so you'd need to know their ip address to do it right? |
| 03:18.28 | shmaltz | RichardLynch, then just run System from dialplan that changes it |
| 03:18.42 | shmaltz | k-man or hostname |
| 03:18.57 | RichardLynch | A bit of a race condition there, for simultaneous calls, no? |
| 03:19.02 | k-man | is there a way to map real phone numbers to hostnames/ip addresses? |
| 03:19.10 | shmaltz | RichardLynch, I guess it is |
| 03:19.24 | shmaltz | k-man enum |
| 03:19.26 | shmaltz | dundi |
| 03:19.31 | shmaltz | ~enum |
| 03:19.32 | jbot | methinks enum is http://www.voip-info.org/wiki-Enum |
| 03:19.33 | shmaltz | ~dundi |
| 03:19.34 | jbot | somebody said dundi was at http://www.dundi.com. DUNDi, an optional Asterisk component, is a distributed, decentralized peer to peer network that provides routes to PSTNs between peers on the same DUNDi network. |
| 03:20.34 | RichardLynch | So down in the guts of asterisk, Festival() in a dialplan calls some C code or what? |
| 03:21.07 | *** join/#asterisk ghento (n=ghento@d75-157-192-235.bchsia.telus.net) |
| 03:21.07 | russellb | Festival is an "Asterisk application", implemented in C, which uses the festival library. |
| 03:21.14 | RichardLynch | I'm downloading source to try to find out now, but digging through source can take awhile... |
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| 03:21.36 | *** join/#asterisk luke-jr (n=luke-jr@2002:46bb:1a76:0:20e:a6ff:fec4:4e5d) |
| 03:21.55 | luke-jr | Am I the only one getting silent calls from random numbers to "0010963112326111#" ? |
| 03:22.20 | russellb | RichardLynch: it doesn't actually use a lib, but anyway .... apps/app_festival.c |
| 03:22.42 | RichardLynch | russelb: Thanks. So I'd have to add another parameter in the C code to pass it through to festival library, which passes it on out to the Lisp function tts_asterisktext or whatever it is I copied into festival.scm |
| 03:22.45 | RichardLynch | Thanks! |
| 03:23.37 | russellb | shrugs |
| 03:23.44 | russellb | i haven't looked at that code in forever (if ever) |
| 03:23.59 | russellb | but i know where it is in the source tree at least, heh |
| 03:24.02 | [TK]D-Fender | RichardLynch: Its bad karma to mention "lisp" when talking about TTS :p |
| 03:25.03 | k-man | thanks shmaltz |
| 03:25.04 | RichardLynch | I'll call it scheme, then, since I think it's actually in scheme. Though it looks pretty much the same to me. |
| 03:25.45 | RichardLynch | Would a patch to app_festival to allow "voice" as a 3rd parameter be likely to be accepted, or would we forever be out of sync? |
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| 03:28.50 | luke-jr | RichardLynch: if it's accepted, it'd be in 1.8 at the soonest |
| 03:29.45 | *** join/#asterisk vncsnvs (n=vncsnvs@189.6.67.94) |
| 03:29.57 | vncsnvs | in wich language asterisk is writen? |
| 03:30.02 | russellb | mostly C |
| 03:30.13 | russellb | luke-jr: that's not true |
| 03:30.25 | luke-jr | should be* |
| 03:30.26 | russellb | we're adding features into each new 1.6.X release |
| 03:30.27 | vncsnvs | russellb, mostly C and what more? |
| 03:30.45 | russellb | vncsnvs: a tiny bit of C++ ... might as well call it all C |
| 03:31.05 | vncsnvs | russellb: thx, gn |
| 03:32.46 | drmessano | http://www.youtube.com/watch?v=Nej4xJe4Tdg <-- Off topic |
| 03:33.03 | vncsnvs | asterisk is very powerful! |
| 03:33.15 | vncsnvs | i'm proud to have opportunity to work with it |
| 03:33.22 | vncsnvs | everyday i learn something new |
| 03:33.45 | drmessano | I bake brownies with it |
| 03:33.52 | drmessano | Nothing it cant do |
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| 03:34.56 | russellb | vncsnvs: i'm glad you're enjoying it :-) |
| 03:35.01 | drmessano | One thing I will say about Asterisk.. There may be things that are lacking, or do not YET work the way they should, but generally regression is limited, and stability is there, constantly |
| 03:35.19 | *** join/#asterisk hsv-al (n=pluhngz@75.76.166.94) |
| 03:35.20 | hsv-al | . |
| 03:35.32 | russellb | drmessano: i'll take that :-) |
| 03:35.58 | russellb | and if you want to rant about things that it doesn't have or things that don't work the way they should, feel free to email me anytime |
| 03:36.12 | russellb | i may not have a chance to respond, but I can promise to read it and seriously consider all points made |
| 03:36.21 | russellb | (that goes for anyone else, too) |
| 03:36.48 | RichardLynch | Okay, so I'm reading app_festival.c, and I'm thinking festival_exec() is what does the work, but I can't see how the second args (intkeys) is tied in... |
| 03:37.32 | drmessano | Theres alternatives, some that do some things better, some just different, but none of them have a solid track record.. Generally I may bitch about things that dont work as they should, or things that are incomplete, but I seldom worry about random crashes, major regressions, and other nightmares with something that is supposed to be a SOLID part of your enterprise. |
| 03:37.38 | k-man | russellb: i really had trouble convincing Asterisk to make my coffee properly ;) do you think you could do something about that? |
| 03:37.59 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 03:38.06 | drmessano | I dont need asterisk to support every scripting language that rocks this week 4eva very yes, I need it to work |
| 03:38.12 | russellb | k-man: exten => foo,1,System(make-coffee) |
| 03:38.16 | k-man | russellb: ah! ill give that a go |
| 03:38.17 | RichardLynch | Ah. I *can* find it, and think there's a bit of a bug there... If you call Festival('','any') the test in line 322 to insure non-empty text to send will "pass" because it's done before text|intkeys is parsed and broken up. |
| 03:38.20 | russellb | the implementation of "make-coffee" is left as an exercise to the dialplan programmer. |
| 03:38.54 | k-man | russellb: i'm sure K&R will have a section on that |
| 03:38.58 | russellb | RichardLynch: sounds like an idea for your first patch submission to the project :-) |
| 03:39.07 | drmessano | app_wife is kinda buggy, but likely will take less of your money when it crashes.. I urge major development of this application, and quickly |
| 03:39.23 | RichardLynch | I'm gonna try, but I am rusty as all get out in C. Don't blame me if it looks more like PHP before I'm done :-) |
| 03:39.50 | drmessano | RichardLynch: The beauty part of coding.. if it looks like shit, its probably valid perl |
| 03:39.54 | drmessano | So go for it |
| 03:40.14 | russellb | RichardLynch: no worries, patches go to bugs.digium.com. If you have questions about the code, feel free to hit up #asterisk-dev or #asterisk-bugs |
| 03:41.21 | FunkyGMT | lol drmessano |
| 03:42.25 | hsv-al | drmessano, I thought you got casted into the pigs, and the pigs rolled down the hill into the river |
| 03:42.29 | hsv-al | :) |
| 03:46.35 | drmessano | hsv-al: Why would you say that? |
| 03:47.08 | drmessano | hsv-al: Arent you Mr "Hi everyone who has no life, unlike me".. Maybe I got busy? |
| 03:47.35 | drmessano | wraps hsv-al in Windows 95 CDs and rolls him into a con |
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| 03:54.52 | ashic | hi |
| 03:55.09 | iamy_china | hi? |
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| 04:03.59 | ashic | exit |
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| 04:06.20 | blackest_mamba | in sip.conf, the line permit=192.168.1.0/255.255.255.0 will only allow phones with an in 192.168.1.0/24 to regsiter. Is that correct? |
| 04:07.36 | *** join/#asterisk andresmujica (n=andresmu@ubuntu/member/andresmujica) |
| 04:11.48 | hsv-al | pokes drmessano |
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| 04:15.36 | retentiveboy | Finding confusing postings on Polycom SLA with *. Can someone tell me, before I spent too much time on it, if this is do'able yet? |
| 04:16.42 | k-man | anyone know a good per timed call voip provider in aus.? |
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| 04:31.51 | Micc | anyone know of a good iptables web admin? |
| 04:38.42 | *** join/#asterisk Sargun (n=Sargun@atarack/Staff/Sargun) |
| 04:39.29 | k-man | this may sound like a stupid question |
| 04:39.45 | k-man | but if i have an DID, can i receive more than one call simultaneously from that DID? |
| 04:39.49 | k-man | over sip |
| 04:40.05 | *** join/#asterisk killown (n=ukendt@unaffiliated/killown) |
| 04:44.00 | lanning | yes, but the number of channels is limited to what your ITSP and your bandwidth allow. |
| 04:44.13 | RichardLynch | anybody have any experiences good or bad with trying Sphinx, simulscribe, or other speech to text technologies? |
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| 04:47.13 | mazpe | how reliable is t.38 for faxing from fax-machine to fax-machine via asterisk/voip |
| 04:47.15 | mazpe | ? |
| 04:47.48 | RichardLynch | luke-jr: Yes, I am getting random calls to extension 0010963112326111# and similar-looking extensions. Not sure where/how. I'd like to lock down the asterisk server to only accept input from one other server. Is that strictly iptables, or is there a setting in /etc/asterisk that will also limit input? |
| 04:48.12 | *** join/#asterisk L|NUX (n=linux@unaffiliated/lnux/x-10290) |
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| 04:52.33 | toodlelou | anyone know a PBX solution to extend Magic Jack? |
| 04:52.41 | *** join/#asterisk RobertLaptop (n=rmiddle@174-153-128-77.pools.spcsdns.net) |
| 04:52.51 | stabler | extend? |
| 04:53.11 | toodlelou | yeah Magic Jack doesn't come with it |
| 04:53.26 | stabler | i have implemented MJ on asterisk |
| 04:53.34 | stabler | as a sip trunk |
| 04:54.00 | toodlelou | mind if I msg you? |
| 04:54.19 | stabler | sure |
| 04:55.20 | Micc | stabler, did you get your fist call problem worked out? |
| 04:55.43 | stabler | no :( |
| 04:56.11 | stabler | i have found other people running * with the MJ as a sip trunk with no issues |
| 04:56.22 | stabler | so it has to be something with my *box |
| 04:57.16 | k-man | vlc does not recognise wav49 files |
| 04:57.39 | RichardLynch | Is there a way to ask asterisk what settings were passed to ./configure, if you installed from packages? (PHP phpinfo() function does this for php installs) |
| 04:58.06 | pdmmm | what are you trying to determine? |
| 04:58.13 | k-man | lanning: what is ITSP? |
| 04:58.38 | L|NUX | can some one take a look into this http://www.pastie.org/private/ncbhivp7s8iz04j4cww when i hangup call asterisk will not hangup call can any one give some suggestions ? |
| 04:59.04 | RichardLynch | I'm going to compile from source this time, and would like the same settings to start with as I had from the packaged version. |
| 04:59.59 | pdmmm | RichardLynch: its been my experience that the packages are pretty vanilla - unless you're using IMAP voicemail storage or something like that |
| 05:00.25 | stabler | L|NUX: what kind of phone are you using? |
| 05:00.58 | k-man | anyone know how to get vlc to play a wav49 file? |
| 05:01.19 | stabler | k-man: install the proper codecs |
| 05:01.35 | k-man | stabler: which codec for vlc? thats what I'm having trouble determining |
| 05:01.50 | stabler | k-man: on a linux distro of windows/ |
| 05:01.56 | stabler | *or |
| 05:01.56 | k-man | windows |
| 05:01.57 | RichardLynch | No, nothing funky. Well I was using ODBC to tie into PostgreSQL because I was too stoopid to catch the "Variables" arg in Originate, but I need to get rid of that anyway. |
| 05:02.11 | pdmmm | RichardLynch: ya, then u should be fine :) |
| 05:02.24 | *** join/#asterisk ingenius (n=alektro@host62.190-224-108.telecom.net.ar) |
| 05:02.36 | stabler | k-man: try this http://www.cccp-project.net/ |
| 05:03.25 | RichardLynch | I would be fine if apt-get install libtermcap didn't say it couldn't be found... apt-get install termcap ditto. Which package actually provides termcap for asterisk? |
| 05:03.46 | stabler | RichardLynch: what distro/ |
| 05:03.56 | RichardLynch | Ubuntu. |
| 05:04.08 | stabler | hmm weird |
| 05:04.32 | RichardLynch | Found a 2003 email about this issue... http://lists.digium.com/pipermail/asterisk-users/2003-May/003970.html |
| 05:04.43 | k-man | stabler: what is it? |
| 05:04.51 | *** join/#asterisk Xen^ (n=linux@unaffiliated/lnux/x-10290) |
| 05:04.58 | RichardLynch | Perhaps termcap got rolled into some bigger meta-package... |
| 05:05.15 | k-man | a codec pack? |
| 05:05.25 | pdmmm | i dont know ubuntu :( |
| 05:05.27 | stabler | k-man: yes.. youll be very satisfied with it |
| 05:05.37 | lanning | try looking for ncurses |
| 05:05.55 | stabler | RichardLynch: it should be part of the termcap-compat package |
| 05:06.41 | RichardLynch | I Just tried that. apt-get seems to think it exists, but doesn't know where to get it. |
| 05:07.29 | k-man | stabler: didn't work |
| 05:07.45 | Xen^ | stabler: analog phone |
| 05:07.46 | stabler | k-man: still fail? |
| 05:08.03 | Xen^ | <stabler>: i got dc, this is L|NUX |
| 05:08.15 | k-man | stabler: yeah - vlc says The format of 'C:\DOCUME~1\jason\LOCALS~1\Temp\msg0005-2.WAV' cannot be detected. |
| 05:08.40 | stabler | k-man: try another player |
| 05:08.48 | k-man | stabler: yeah ok |
| 05:09.12 | Xen^ | stabler: Telekom Malaysia |
| 05:09.36 | stabler | Xen^: hmm not sure |
| 05:10.01 | Xen^ | stabler: humm |
| 05:10.07 | Xen^ | any one else ? |
| 05:10.11 | Xen^ | can some one take a look into this http://www.pastie.org/private/ncbhivp7s8iz04j4cww when i hangup call asterisk will not hangup call can any one give some suggestions ? |
| 05:10.48 | stabler | Xen^: you may want to add what kind of phone and fxs card/interface youre using |
| 05:11.20 | Xen^ | stabler: it is Sangoma A400 |
| 05:13.19 | k-man | god damn wmp is annoying |
| 05:14.52 | RichardLynch | Hmmf. I installed a lot of other stuff and somehow libtermcap got in there I suppose, since it works fine now. Go figure. |
| 05:15.38 | k-man | cool, i just received my first voicemail emailed to me from asterisk |
| 05:15.45 | k-man | like a real one, not a test one |
| 05:16.38 | stabler | k-man: woo |
| 05:16.46 | stabler | what mail service are you using |
| 05:17.01 | k-man | stabler: err.... not sure i understand your question |
| 05:17.12 | stabler | do you use sendmail? |
| 05:17.20 | k-man | stabler: postfix |
| 05:17.22 | stabler | oh |
| 05:17.27 | k-man | why do you ask? |
| 05:17.27 | stabler | how easy was the config for it |
| 05:17.36 | stabler | im getting ready to set mine up |
| 05:17.37 | lanning | RichardLynch: did you install ncurses? |
| 05:17.43 | k-man | postfix? pretty easy - a heck of a lot easier that sendmail imho |
| 05:17.53 | stabler | and getting sendmail to use gmail servers looks gnarly |
| 05:18.10 | haps | if I'm calling a voip number that I'm setting up (unlimitel) from a rogers cell phone and it tells me 'not allowed' is that indicative of a screwup in my .conf files (outgoing works) or is it rogers being dickheads? |
| 05:18.19 | k-man | stabler: i use my ISPs mailserver as the smarthost |
| 05:18.49 | k-man | stabler: getting your outgoing email address correct was the tricky part for me as the machine does not have an MX record |
| 05:18.57 | k-man | and cannot receive email from the internet |
| 05:18.59 | RichardLynch | Yes, I installed ncurses and all kinds of stuff. http://www.asteriskguru.com/tutorials/asterisk_installation_compilation_debian.html Everything there, except the stuff apt-get blew up on. Compiling now, with my patch, and blowing up in my face. (as expected, as I forgot to declare my vars after a decade of PHP work) |
| 05:19.13 | k-man | stabler: also, there is quite an active #postfix channel too |
| 05:19.35 | stabler | k-man: good to know |
| 05:19.36 | rob0 | But nobody there knows anything. |
| 05:19.51 | k-man | rob0: that might be true actually |
| 05:20.02 | lanning | heh. I believe termcap has been rolled into the ncurses project. |
| 05:20.19 | k-man | apparantly there is a bug in VLC about it being unable to play wavs encoded with GSM 1.6 - but i can't check track as its down |
| 05:20.42 | k-man | http://forum.videolan.org/search.php?st=0&sk=t&sd=d&keywords=gsm+6+1 |
| 05:20.52 | RichardLynch | Sounds quite reasonable. It got rolled into something in that list. |
| 05:29.19 | *** part/#asterisk brunner (n=chris@68-119-87-106.dhcp.mtgm.al.charter.com) |
| 05:29.51 | RichardLynch | I would like to run a second asterisk binary on a different port than usual, but using the same /etc/asterisk/*.conf files otherwise. What is the easiest way to achieve that? |
| 05:31.21 | *** join/#asterisk IcyB0x (n=frozen@d141-111-209.home.cgocable.net) |
| 05:31.38 | rob0 | uh, port? For SIP? |
| 05:31.44 | k-man | stabler: which distro do you run? |
| 05:33.33 | stabler | k-man: on what box :) |
| 05:33.35 | rob0 | I guess you'd need a different config directory and settings in asterisk.conf, plus different bindport settings in the appropriate network protocol configs. |
| 05:33.50 | *** join/#asterisk Gopaul (n=chatzill@61.95.156.130) |
| 05:34.21 | stabler | k-man: i have 1 ubuntu server, 2 unbuntu desktops, 1 ubuntu lappy, and a gentoo server |
| 05:34.40 | RichardLynch | Yeah, I just realized just how many ports were getting used and decided that killing the original and running my hacked one was a better idea. The client will just have live with it not being something he can play with for awhile. |
| 05:34.44 | stabler | k-man: i run * on my ubuntu server |
| 05:35.09 | RichardLynch | Whoo hoo! I wrote code that managed to segfault! (no surprise) |
| 05:35.45 | k-man | stabler: your an ubuntu kinda person then? |
| 05:36.11 | stabler | k-man: yes.. for the most part.. ive recently been sticking my hand in the gentoo pie |
| 05:38.03 | k-man | ive not really tried gentoo |
| 05:38.14 | k-man | i'm a debian kind of guy mostly |
| 05:38.34 | stabler | i had a debian server at one point |
| 05:38.46 | RichardLynch | Can anybody translate this into English: [Mar 4 23:31:25] NOTICE[12535] channel.c: Dropping incompatible voice frame on SIP/voicepulse-primary-0a180af0 of format ulaw since our native format has changed to 0x0 (nothing) |
| 05:38.54 | stabler | of course ubuntu is built on debian sooo pretty much the same |
| 05:39.22 | stabler | RichardLynch: is your primary codec ulaw? |
| 05:39.44 | RichardLynch | stable: I dunno. How would I find out? |
| 05:40.30 | stabler | RichardLynch: i seriously doubt you use ulaw as your primary codex |
| 05:40.33 | stabler | *codex |
| 05:40.36 | stabler | *codec |
| 05:40.36 | RichardLynch | Or how would I re-compiled it so I could say "yes" |
| 05:40.44 | RichardLynch | s/yes/no/ |
| 05:40.56 | stabler | which probably means youre getting a ulaw voice frame and the server does know what to do with it |
| 05:41.02 | stabler | but i could be wrong |
| 05:41.24 | RichardLynch | Festival is supposed to be outputting audio, which might maybe supposed to be in ulaw format... |
| 05:41.52 | RichardLynch | Of course, I just tried to hack Festival, which is likely the ultimate source of the problem... |
| 05:42.00 | stabler | lol |
| 05:42.03 | stabler | *doesnt |
| 05:45.47 | RichardLynch | I have log entries about "could not find preferred codec...going for the best...0x4 (ulaw)...preferred incoming 0x40 (slin) ... Joint capabilities 0x0 (nothing)" |
| 05:46.13 | RichardLynch | I'm guessing I am missing some kinda libs/-devel packages for audio stuff to compile in with asterisk? |
| 05:48.14 | *** join/#asterisk The_LightSide (n=lightie@dsl-241-44-37.telkomadsl.co.za) |
| 05:52.16 | RichardLynch | What packages should I install to maximize asterisk audio capabilities? mpg123??? arts??? alsa??? |
| 05:52.43 | *** join/#asterisk mikkel (n=mikkel@84-238-113-66.u.parknet.dk) |
| 05:53.32 | stabler | im not positive |
| 05:53.48 | stabler | but i think there are codec core packages you need to compile |
| 05:54.13 | stabler | i bet some quick googleing would reveal some answers |
| 05:56.50 | stabler | if you do a make menuselect when comiling * you have the option to add more codec cores |
| 05:58.49 | *** join/#asterisk _gm (n=gmustafa@115.186.106.37) |
| 05:59.14 | *** join/#asterisk IcyB0x (n=frozen@d141-111-209.home.cgocable.net) |
| 05:59.50 | RichardLynch | Sweet! |
| 06:00.00 | RichardLynch | Compiling a whole bunch of audio stuff now! :-) |
| 06:00.59 | *** join/#asterisk Sargun (n=Sargun@atarack/Staff/Sargun) |
| 06:01.59 | *** join/#asterisk llragsll (i=3b636995@gateway/web/ajax/mibbit.com/x-8184d8bf9a74f77f) |
| 06:02.35 | k-man | so like, how much bandwidth do you need per call? |
| 06:02.55 | llragsll | Hello, I need to change the On hold music by uploadin an mp3 but it throws up an error |
| 06:03.03 | llragsll | mpg123 not found... |
| 06:03.22 | k-man | install mg123 |
| 06:03.50 | llragsll | well I tried tht..got the package but having problems there too... |
| 06:04.16 | llragsll | the make command throws up a bunch of errors.. |
| 06:04.17 | k-man | which distro? |
| 06:04.32 | mahiti-irc | HI Anyone have GUI setup guide for asterisknow ? |
| 06:04.40 | llragsll | asterix-NOW(cent os) |
| 06:04.51 | k-man | llragsll: try mpg321, its a freeer version of mpg123 afaik |
| 06:05.01 | *** join/#asterisk [intra]lanman (n=intralan@freeswitch/developer/intralanman) |
| 06:05.05 | k-man | i don't know anything about cent-os so i can't help you there, sorry |
| 06:05.23 | lanning | yum install mpg321 |
| 06:05.51 | llragsll | is mpg321 same as 123??:/ |
| 06:06.01 | Corydon76-dig | NO! |
| 06:06.09 | k-man | llragsll: no, but close enough |
| 06:06.20 | Corydon76-dig | It is NOT close enough |
| 06:06.34 | Corydon76-dig | mpg321 is missing a critical feature |
| 06:06.38 | k-man | is it? |
| 06:06.40 | k-man | oh |
| 06:06.43 | lanning | you mean they haven't fixed it yet? |
| 06:06.49 | k-man | llragsll: please ignore me - i have no idea |
| 06:06.52 | stabler | llragsll: why are you using mp3s? |
| 06:07.10 | Corydon76-dig | mpg321 is out of active development |
| 06:07.11 | llragsll | any other format?? |
| 06:07.16 | llragsll | I can conver it too/ |
| 06:07.25 | llragsll | mp3 to ? |
| 06:07.32 | stabler | llragsll: it is not recommended to use mp3s |
| 06:07.36 | stabler | wav |
| 06:07.43 | Corydon76-dig | 8000Hz mono wav |
| 06:07.45 | stabler | mono wav |
| 06:08.02 | llragsll | so if I upload the wav file it will work right? |
| 06:08.03 | stabler | mp3's are complex sound files |
| 06:08.03 | Corydon76-dig | It HAS to be 8000Hz |
| 06:08.11 | stabler | and will create excessive load on your server |
| 06:08.23 | llragsll | wht is the native format? |
| 06:08.33 | stabler | mono wav |
| 06:08.37 | llragsll | wht does asterisk use internally? |
| 06:08.39 | llragsll | ok |
| 06:08.55 | Corydon76-dig | WAV encompasses multiple bitrates. The format isn't enough. |
| 06:08.59 | stabler | mono wav will wokr out of the box |
| 06:09.15 | Corydon76-dig | stabler: but only at 8000Hz |
| 06:09.26 | stabler | Corydon76-dig: yes... as hes saying it much be 8000Hz |
| 06:09.32 | stabler | 8must |
| 06:09.48 | llragsll | hmm..ok..I'll convert the mp3 to mono wave...maybe SOX can do it? |
| 06:10.04 | Corydon76-dig | Assuming sox has mp3 support on your box, yes |
| 06:10.20 | Corydon76-dig | sox foo.mpg -r 8000 -c 1 foo.wav |
| 06:11.56 | llragsll | sox: do not understand format : mp3 :( |
| 06:12.38 | k-man | can lame do it? |
| 06:12.47 | k-man | or does lame only go wav->mp3? |
| 06:13.47 | Corydon76-dig | lame --decode |
| 06:15.05 | k-man | llragsll: try lame maybe? |
| 06:15.45 | llragsll | yes...trying to get it now..wht is the package name? |
| 06:15.57 | RichardLynch | If you can go back to your original source before it got turned into an MP3, you'll have better quality audio, almost for sure... |
| 06:16.34 | llragsll | no...I just have the mp3 file... |
| 06:16.35 | RichardLynch | lame is the package name for lame... |
| 06:17.21 | RichardLynch | Hmmm I'm still having the same error about codec 0x0 (nothing) |
| 06:19.10 | *** join/#asterisk GameGamer43 (n=GameGame@69-2-83-45.static.networktel.net) |
| 06:19.47 | k-man | you might have to install lame from source - i think the distros are funny about distributing binaries of it for some reason |
| 06:20.21 | k-man | llragsll: but your distro might have an automated way of installing it from source. perhaps ask on #centos? |
| 06:21.12 | llragsll | I tried yum..giving an error |
| 06:21.19 | llragsll | anyway i got the rpm |
| 06:22.25 | RichardLynch | I went back to the original source code, and am still getting errors about codec 0x0 (nothing) and then a segfault... |
| 06:22.32 | stabler | sigh... centos... |
| 06:23.06 | luke-jr | Corydon76-dig: WAV encompasses multiple codecs, too. it's possible to have MP3 data inside a WAV file |
| 06:24.37 | RichardLynch | The WAV format pre-dates MP3 by quite some time... They retro-fitted mp3 to fit into it or what?... |
| 06:25.26 | Corydon76-dig | The wav format is expansive... you can come up with more formats for it than filetypes for MSDOS |
| 06:26.17 | stabler | such as GSM is a type of wav file |
| 06:26.21 | RichardLynch | It's been a long time since I've dug around inside of a .wav file I guess. |
| 06:26.36 | llragsll | y did they use cent-os I wonder...wasn't it rpath before? |
| 06:26.55 | stabler | cent-os=fail |
| 06:27.04 | stabler | in my opinion |
| 06:27.34 | llragsll | ah..installed lame |
| 06:27.37 | [TK]D-Fender | stabler: they're like assholes.... everybody's got one... |
| 06:28.06 | stabler | [TK]D-Fender: :) |
| 06:28.14 | stabler | [TK]D-Fender: im just letting everyone see mine |
| 06:28.34 | [TK]D-Fender | stabler: Dangerous words, especially in these parts |
| 06:28.59 | stabler | [TK]D-Fender: i aint skurd |
| 06:29.50 | k-man | is the linksys spa3102 any good? |
| 06:31.06 | RichardLynch | WARNING: /modules contains modules not installed by this version...insure compatibilty... How would I do that? The package manager put them there. They are v 1.24.21, I think, and I just installed 1.24.23.1 from source. |
| 06:31.39 | Corydon76-dig | Neither of those version numbers correspond to Asterisk packages |
| 06:31.48 | stabler | lol |
| 06:31.58 | Corydon76-dig | There's 1.2.21 and there's 1.4.21 |
| 06:32.39 | stabler | RichardLynch: soundsl like youre working up a tornado of version on one machine |
| 06:32.48 | stabler | =/ |
| 06:32.57 | stabler | *versions |
| 06:33.25 | Corydon76-dig | stabler: everytime I see your nick, I think of the fictional character Eliot Stabler on Law & Order |
| 06:33.35 | RichardLynch | Gah. 1.4.21 and 1.4.23.1 Sorry! |
| 06:33.56 | stabler | Corydon76-dig: i get that all the time |
| 06:34.00 | stabler | Corydon76-dig: :D |
| 06:34.15 | llragsll | Just for info : installed lame, lame --decode foo.mp3 bar.wav, used sox foo.wav -r 8000 -c bar.wav |
| 06:34.27 | stabler | llragsll: still fail? |
| 06:34.28 | Corydon76-dig | stabler: Nice that you have a sense of humor about it |
| 06:34.33 | *** join/#asterisk lmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 06:34.33 | *** mode/#asterisk [+o lmadsen] by ChanServ |
| 06:34.52 | llragsll | no..WIN..:D |
| 06:34.52 | stabler | woo |
| 06:35.08 | stabler | Corydon76-dig: its actually my last name |
| 06:35.48 | Corydon76-dig | stabler: I figured |
| 06:36.09 | stabler | there is also a famous football player with the last name stabler |
| 06:36.20 | stabler | but i dont really know who he is or who he played for |
| 06:36.25 | drmessano | Corydon76-dig: We call him UN-stabler |
| 06:36.26 | stabler | because im not a sports fan |
| 06:36.40 | *** join/#asterisk denon (i=denon@synapse.subneural.net) |
| 06:36.40 | *** mode/#asterisk [+o denon] by ChanServ |
| 06:36.46 | stabler | heh.. drmessano and i were just talking about this the other day |
| 06:36.48 | rob0 | Oh my I remember Kenny Sabler |
| 06:36.54 | drmessano | "Rut ro, Eliot's interrogating someone... here comes the face punch" |
| 06:36.55 | rob0 | *Stabler |
| 06:36.58 | stabler | rob0: yes! |
| 06:37.24 | lmadsen | Corydon76-dig: it's late |
| 06:37.25 | rob0 | Oakland Raiders (long before the LA move) |
| 06:37.29 | stabler | rob0: i am supposably related |
| 06:37.42 | stabler | dont know how closely |
| 06:37.44 | Corydon76-dig | lmadsen: indeed |
| 06:37.53 | drmessano | Im related to lmadsen |
| 06:38.01 | rob0 | hasn't followed football ... since Kenny Stabler's days |
| 06:38.03 | lmadsen | lies! |
| 06:38.11 | drmessano | Im 1/37th viking, bro |
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| 06:38.16 | *** mode/#asterisk [+o denon] by ChanServ |
| 06:38.37 | Corydon76-dig | was hoping to be related to Leif... but he evaded my advances... |
| 06:38.46 | drmessano | LOL |
| 06:40.44 | *** join/#asterisk brunner (n=chris@68-119-87-106.dhcp.mtgm.al.charter.com) |
| 06:42.12 | RichardLynch | Whooo Hoooo! I have my hack "in" and it didn't segfault! Course, I didn't try to actually pass in that third arg or anything, but still... Progress |
| 06:42.34 | brunner | I'm sure this is a really stupid question. I'm trying to set up IAXmodem on localhost so I can route calls to it when I detect fax tones. How do I remedy this error? [Mar 5 00:34:01] NOTICE[32070]: chan_iax2.c:5499 register_verify: No registration for peer 'faxer' (from 127.0.0.1) |
| 06:42.38 | brunner | iax.conf: http://pastebin.com/m2e90588d and iaxmodem-cfg.ttyIAX: http://pastebin.com/m140b5ad1 |
| 06:44.17 | brunner | the message only appears when I start IAXmodem |
| 06:44.44 | brunner | What's wrong in my config that's preventing IAXmodem from being able to register with asterisk? |
| 06:45.03 | *** join/#asterisk fiddur (i=fiddur@c042.rit.se) |
| 06:48.48 | RichardLynch | Anybody know of an IRC channel for festival?... |
| 06:49.31 | brunner | RichardLynch: if it doesn't show up here, it probably doesn't exist: http://tinyurl.com/avjj5e |
| 06:51.04 | RichardLynch | ah well I'll join the mailing list and go pester them for awhile :-) |
| 06:51.41 | stabler | lol |
| 06:53.00 | *** join/#asterisk scientes (n=scientes@75-165-95-28.tukw.qwest.net) |
| 06:57.21 | *** join/#asterisk IcyB0x (n=frozen@d141-111-209.home.cgocable.net) |
| 06:59.57 | brunner | so I'm sure my issue with IAXmodem registering is pretty elementary. Could someone take a look and see what I'm doing wrong? |
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| 07:04.11 | itakinet | * 1.4.23.1, cdr_addon_mysql inserts the CDR record about half a second before my AGI script completes resulting in unknown variables, PHP AGI Code works great in * 1.4.21.2.... any advice ? |
| 07:05.24 | *** join/#asterisk tzafrir_laptop (n=tzafrir@local.xorcom.com) |
| 07:17.36 | lmadsen | itakinet: ya, there are some inconsistencies between 1.4.20-ish to 1.4.22-ish+. You can check the open issues on the bug tracker for CDR's and see if any of them match your issue |
| 07:18.02 | lmadsen | if not, you could try opening a new issue, and it will get looked at as soon as someone is free |
| 07:18.33 | lmadsen | heads of to bed |
| 07:19.48 | itakinet | Yeah All I found was change log stuff from Steve, trying to just hack it now so the CDR write executes after the AGI finishes the dump |
| 07:20.02 | itakinet | goodnight |
| 07:21.32 | brunner | are there any premium-rate area codes in the Caribbean? |
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| 08:12.21 | j_kroon | hi all, i've got an asterisk 1.2.7.1 install in front of me (BRIstuffed-0.3.0-PRE-1p). Having trouble with a SNOM360 that's not subscribing properly to extensions. |
| 08:12.48 | *** join/#asterisk atomi (n=bradb@pool-98-119-254-164.lsanca.dsl-w.verizon.net) |
| 08:12.50 | j_kroon | IE: The blinking lights stopped working and I can't get them working again (rebooting the phone didn't solve it). |
| 08:13.00 | *** join/#asterisk fcois93 (n=Administ@bagnolet.acropolistelecom.net) |
| 08:13.05 | j_kroon | any ideas what I can look at? I suspect somebody broke it and the config only just reloaded now ... |
| 08:13.16 | j_kroon | sip show subscriptions doesn't list anything. |
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| 10:13.06 | dumezil | im having some problems with a TDM400P card. dahdi_hardware shows the card and dahdi_scan shows the card and dahdi_cfg configures the card, but asterisk doesn't see it. |
| 10:13.40 | dumezil | if i do a chan_dahdi is loaded, but its not seeing the card |
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| 10:22.11 | tamiel | dumezil: /etc/asterisk/chan_dahdi.conf is configured ? |
| 10:22.52 | tamiel | dumezil: Try to start asterisk in debug mode : asterisk -dddd -vvvvv |
| 10:34.13 | dumezil | now its very odd... i've restarted asterisk and when i do a module unload chan_dahdi.so, i get "/usr/sbin/safe_asterisk: line 125: 5360 Segmentation fault (core dumped) nice -n $PRIORITY ${ASTSBINDIR}/asterisk -f ${CLIARGS} ${ASTARGS} >&/dev/${TTY} </dev/${TTY}" |
| 10:34.23 | dumezil | and "Asterisk ended with exit status 139" |
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| 11:02.23 | stmaher | Hi guys.. how do i enable colours in the CLI? |
| 11:06.11 | fcois93 | stmaher: I can do like that asterisk -rcddddddddddddd |
| 11:06.28 | fcois93 | stmaher: try others, but I thin that it is the 'c' |
| 11:06.52 | fcois93 | stmaher: 'c' like 'color' ^^ |
| 11:07.18 | stmaher | herm.. doesnt work? |
| 11:08.20 | jaytee | c is not for color |
| 11:08.42 | jaytee | c is only if you are connecting to the console when * is not already running as a service |
| 11:09.53 | stmaher | asterisk is running |
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| 11:11.30 | stmaher | jaytee ok.. asterisk run with -c not from service works.. |
| 11:11.36 | stmaher | how do i get an asterisk -r working? |
| 11:12.53 | jaytee | asterisk -r is only used to connect to an already running instance of asterisk which starts as a service using an init script |
| 11:14.41 | jaytee | I only intended to point out that -c does not provide color. color is set by default by whatever terminal method you're using. I use Putty for ssh into my asterisk server. on a standard tty session on the server I get no color. if i run * with Gnome loading in runlevel 5 and I start asterisk from an xterm session I get color. |
| 11:17.27 | tzafrir_laptop | 'c' is for 'console' |
| 11:18.29 | stmaher | so if you start asterisk in an xterm-color anything after that in any other terminal with xterm-color will work? |
| 11:18.33 | jaytee | if you're not getting color at all look in your /etc/init.d/asterisk file and make sure you have the line that says DAEMON=$AST_SBIN/asterisk TERM=linux asterisk |
| 11:18.39 | stmaher | if not started in an xterm and daemonised.. thats the problem |
| 11:18.45 | dandre | hello |
| 11:19.16 | dandre | is there anyone who can help me configuring sangoma drivers for asterisk? |
| 11:19.28 | jaytee | Sangoma might |
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| 11:46.42 | Daejeo | 1200 calls, cps-140- asterisk @freebsd-system- 2.4 ghz ram512 |
| 11:47.35 | Daejeo | art- less 10ms |
| 11:47.55 | tzafrir_laptop | 1200 calls, 120cps, so the system was up for 10s? |
| 11:48.24 | tzafrir_laptop | err... 140 cps: even worse ;-) |
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| 11:49.52 | Daejeo | do you know what is art? |
| 11:50.17 | Daejeo | have you done testing? |
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| 12:03.13 | k-man | what are the options available in voicemail.conf for the format? |
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| 12:03.29 | k-man | is there something that the iphone could play natively? it doesn't like wav49 |
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| 12:05.20 | jaytee | wav49, gsm and wav |
| 12:05.28 | jaytee | at least with 1.6 |
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| 12:10.37 | k-man | oh |
| 12:10.40 | k-man | im on 1.4 |
| 12:11.11 | k-man | i saw somewhere you can use ogg also - but i tried that and it would send 0 byte .ogg files |
| 12:11.34 | k-man | i might have to use .wav files - |
| 12:13.46 | *** join/#asterisk Dovid (n=annon@212.199.133.250.static.012.net.il) |
| 12:14.16 | Dovid | hi. if in dahdi_tool it keeps flipping between red and yello alarm it means the telco has an issue ? |
| 12:14.49 | Dovid | it flips between red/rec and yel/red/rec |
| 12:15.44 | jaytee | not necessarily, it may mean the card has an issue, or a problem with the software configuration |
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| 12:16.26 | Dovid | jaytee: That would be in system.conf or chan_dahdi.conf ? |
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| 12:17.11 | jaytee | both, but what type of card and circuit? |
| 12:17.54 | Dovid | E1 |
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| 12:18.03 | double_cheesburg | Anyone have thoughts on the differences between asterisk 1.4 and 1.6 ?? |
| 12:18.15 | double_cheesburg | performance or otherwise ?? |
| 12:18.29 | mosty | double_cheesburg, have you read the changes document? |
| 12:18.30 | russellb | it should perform better, and it has more features. :-) |
| 12:18.33 | Dovid | Te100P I think |
| 12:18.42 | raasdnil | Hi all, I am doing a playtone while in the call setup phase. Right now playtones stops when Dial app connects. But how can I get it to stop on the "making progress" response from my upstream? |
| 12:18.49 | awk | hi, hmm, maybe one of you can advise... for a 'security' point of view and recordings with asterisk. 1 PBX company says this, We BLA BLA BLA, herbeby confirm that our Communications Recording System ..... And.... are using an ASC-proprietary format for achiving of recorded calls. The algorithm of that format has been developed by ... and is securely stored at .... R&D deparment. This algorithm is not avaiable for any third party... |
| 12:18.53 | double_cheesburg | mosty: No. |
| 12:19.02 | double_cheesburg | mosty: Anything stand out to you, though? |
| 12:19.09 | awk | now what about in a court of law using .wav they could say we could 'manipulate' it |
| 12:19.21 | russellb | ~asterisk16 |
| 12:19.22 | jbot | new features in Asterisk 1.6 are listed at http://svn.digium.com/view/asterisk/trunk/CHANGES?view=markup |
| 12:19.56 | double_cheesburg | jbot: thx |
| 12:19.57 | jbot | np, or eeeerrruuusssshhhwwwaaaaaAAAAAAAAA!!!!11 |
| 12:20.06 | jaytee | Dovid, has the system been in service for awhile and just started experiencing problems or is this your very first E1 circuit and Asterisk install |
| 12:20.37 | Dovid | jaytee: out of the blue it acted up |
| 12:21.23 | jaytee | the best new feature in 1.6 is the russellb() app. just add it as the first priority in any section of the dialplan and it will fix any errors that occur automagically :-) |
| 12:21.48 | Daejeo | tzafrir_laptop: for you http://netmedia.paichai.ac.kr/test3.html |
| 12:21.58 | double_cheesburg | Wow. You can use cURL with asterisk now? Does that pretty much make the AGI obsolete? |
| 12:22.17 | Daejeo | see asterisk 1.4 @freebsd 7.1 |
| 12:22.20 | double_cheesburg | I mean, I guess it should make PHP AGI obsolete |
| 12:22.37 | russellb | double_cheesburg: maybe, it depends on what you use the AGI for |
| 12:22.39 | jaytee | Dovid, so that answer was kind ummm.....vague? I'm going out on a limb here and assuming you meant it's been in operation for "awhile" |
| 12:23.13 | double_cheesburg | russellb: wow. I had no idea. Is cURL support new in 1.6 ?? |
| 12:23.48 | russellb | Some of it is, there may have been some support for it in 1.4 as well, I don't remember off of the top of my head ... |
| 12:23.52 | Dovid | a while = 6 months or more |
| 12:23.58 | russellb | using curl as a realtime backend is definitely new in 1.6 |
| 12:24.05 | russellb | using curl from the dialplan may also be new |
| 12:24.11 | double_cheesburg | That is so sweet |
| 12:24.14 | double_cheesburg | I had no idea |
| 12:24.25 | double_cheesburg | I was looking over the AGI documentation on VOIP-info |
| 12:24.31 | russellb | dialplan support for curl is in 1.4, as well |
| 12:24.32 | double_cheesburg | I'm so glad I found this |
| 12:24.41 | jaytee | Dovid, if it's been running ok for six months or more then it's either an issue with the hardware (card or server it's in) or the telco. |
| 12:24.53 | russellb | it's more extensive in 1.6, though |
| 12:25.09 | double_cheesburg | nice. I am stoked. |
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| 12:25.17 | Dovid | jaytee: I guess the best thing would be is to go there and create my own loop and see what happens |
| 12:25.46 | jaytee | Dovid, yes logically the first step in troubleshooting the circuit is put it in loopback and see if the alarms clear |
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| 12:32.42 | poller | Hello, I have som trouble with monitoring calls and soxmix for merging the two sound files. |
| 12:33.04 | poller | soxmix doesn't "overlay" the sound, i puts one track efter the other. |
| 12:33.34 | poller | So first i hear my voice, and then the other persons voice. Not at the same time as it should be |
| 12:34.55 | *** part/#asterisk raasdnil (n=mikel@60-240-58-103.tpgi.com.au) |
| 12:35.59 | mosty | poller, that's strange- what version of asterisk and what version of sox? |
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| 12:54.57 | poller | mosty: Nevermind, MixMonitor instead of Monitor did the trick :) |
| 12:55.01 | poller | Thanks :) |
| 12:56.16 | mosty | i have had problems with both Monitor and MixMonitor in the past, so far the safest solution is to use Monitor and my own script to manually mix the channels |
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| 13:16.25 | michael-i | I'm having trouble compiling asterisk, ld cannot find stdc++. I was wondering which optional pieces of asterisk require this library, so I can disable them, as I don't wish to build it for my embedded target. |
| 13:16.39 | *** join/#asterisk jaytee (n=jforde@unaffiliated/jaytee) |
| 13:23.37 | tzafrir_laptop | michael-i, you built yourself? binary distribution? |
| 13:23.58 | tzafrir_laptop | C++ is only needed for a few channel drivers, IIRC |
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| 13:24.05 | tzafrir_laptop | chan_h323 and vhan_vpb |
| 13:24.28 | tzafrir_laptop | ./configure --without-pwlib --without-vpb |
| 13:24.38 | michael-i | tzafrir_laptop: I'm building under t2 (t2-project.org) |
| 13:25.11 | tzafrir_laptop | What exactly is being linked? What target? |
| 13:26.54 | michael-i | I have already disabled those channels in my menuselect.makeopts but perhaps the additional configure options are also required |
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| 13:29.43 | tzafrir_laptop | michael-i, again, what's the exact error? |
| 13:29.48 | tzafrir_laptop | what target? |
| 13:30.30 | juan | what do they call an answering service between regular voice mail and a PBX? It's really just recorded messages on key-entry. No forwarding or out calling. |
| 13:30.39 | juan | Is there a name for this? |
| 13:31.17 | juan | Press 1 to hear this option, Press 2 to hear that option. <plays pre-recorded audio file> that's all |
| 13:31.20 | michael-i | tzafrir_laptop: http://pastebin.ca/1353516 (sorry for the t2 extras in there) |
| 13:32.45 | mosty | juan, "IVR" refers to those "press 1 to do <blah>" type menus |
| 13:33.47 | juan | Im trying to get one of those to work with the magic jack |
| 13:34.29 | juan | they have hard IVRs right? |
| 13:34.32 | juan | I can buy used |
| 13:34.47 | michael-i | tzafrir_laptop: ...and my makeopts (http://pastebin.ca/1353517) |
| 13:34.50 | tzafrir_laptop | michael-i, hmm... you can make Asterisk provide the complete build environment using: make NOISY_BUILD=yes |
| 13:35.36 | michael-i | tzafrir_laptop: I'll try...I'm still learning the t2 environment and how to append/modify options. Give me a sec. |
| 13:35.40 | *** join/#asterisk HermesNeto (i=HermesNe@200.249.176.44) |
| 13:36.21 | tzafrir_laptop | I recall Asterisk passes an escessive list of libraries in the link command. But IIRC it was just for the asterisk binary itself |
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| 13:38.45 | mosty | juan, what is "the magic jack"? |
| 13:39.36 | michael-i | tzafrir_laptop: a bit more verbosity : http://pastebin.ca/1353519 |
| 13:40.48 | juan | its a little USB box that installs a soft phones and connects to RJ22 |
| 13:40.51 | juan | RJ23 |
| 13:40.57 | juan | plug and talk |
| 13:41.04 | juan | they give you your own # and voice mail system too |
| 13:41.21 | juan | $29 a year unlimited calls in the US out and in |
| 13:42.32 | *** join/#asterisk [TK]D-Fender (n=joe@216.191.106.163) |
| 13:44.18 | tzafrir_laptop | michael-i, well, it is obvious, now |
| 13:44.24 | *** join/#asterisk SwK (n=SwK@freeswitch/developer/swk) |
| 13:44.38 | tzafrir_laptop | just look at the end of Makefile.rules :-( |
| 13:45.01 | tzafrir_laptop | %: %.o is built with $(CXX) |
| 13:45.23 | michael-i | so 1.4 links with g++? |
| 13:45.28 | jermey_g | my e1 port is giving LOS alarm. |
| 13:45.46 | *** join/#asterisk RypPn (i=TuMbL@rosscom.demon.co.uk) |
| 13:45.51 | tzafrir_laptop | michael-i, change that $(CXX) to $(CC) and see if things work |
| 13:45.59 | juan | I need a chepa hard-answering machine that does the IVR |
| 13:46.02 | juan | cheap |
| 13:46.06 | jermey_g | are there any good wifi IP phones out |
| 13:46.23 | *** join/#asterisk qdk (n=qdk@81.7.168.130) |
| 13:47.09 | jblack | jermey_g: Combining a "wireless game adaptor" with an IP phone can achieve that result. |
| 13:48.08 | [TK]D-Fender | jermey_g: What is the functional requirement? |
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| 13:58.22 | juan | looks like there is no IVR Hardware |
| 13:58.27 | littlerock | I connect asterisk to avaya SES using sip trunk. Avaya also has a PSTN trunk, however, it promt the call is restricted when I call PSTN number from asterisk sip phone. any idea? |
| 13:58.28 | juan | under like $1000 |
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| 14:01.11 | [TK]D-Fender | juan : WTH is "IVR hardware"? |
| 14:01.30 | *** join/#asterisk juanjoc (n=juanjoc@200.69.219.113) |
| 14:01.30 | juan | why cant I just have an answering machine with IVR on it? |
| 14:01.38 | [TK]D-Fender | littlerock: what is "it"? |
| 14:01.46 | mosty | juan, why not use asterisk? |
| 14:02.13 | juan | well at this point my business setup is not ready for a dedicated box |
| 14:02.25 | juan | I just need a temp solution for right now as I research the issue |
| 14:02.37 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 14:02.47 | mosty | juan, look for a hosted pbx provider |
| 14:02.57 | juan | press 1 to hear my speech on this, press 2 to hear my speech on that, press 3 for this. |
| 14:03.16 | *** join/#asterisk jasonwoot (n=some@bookit-dev.com) |
| 14:03.21 | [TK]D-Fender | juan : * is more than fine on this... |
| 14:03.28 | juan | you said it perfect |
| 14:03.35 | juan | its MORE THAN fine :) |
| 14:03.47 | juan | a little too much for 3 pre-recorded messages |
| 14:03.58 | juan | (i think) |
| 14:04.15 | littlerock | [TK]D-Fender: avaya's attendant |
| 14:04.43 | [TK]D-Fender | littlerock: Feel free to complain that * is too good for you |
| 14:04.50 | [TK]D-Fender | littlerock: We can use the reinforcement. |
| 14:05.10 | russellb | lol |
| 14:05.21 | *** join/#asterisk GameGamer43 (n=GameGame@nat/digium/x-559bfdd17226fc7c) |
| 14:05.24 | mort_gib | I haven an issue seeing incoming calls using a TDM400P with Dahdi |
| 14:06.21 | *** join/#asterisk SirThomas (n=tomc@mail.kendeco.com) |
| 14:06.27 | mort_gib | I can see the hardware using dahdi_hardware and lsmod shows dahdi_echocan_mg2,wctdm |
| 14:08.42 | littlerock | which codec should I choose for WAN |
| 14:08.51 | tzafrir_laptop | mort_gib, does dahdi_hardware show the device as having the module loaded (+) or unloaded (-) ? |
| 14:09.04 | tzafrir_laptop | michael-i, did this work? |
| 14:09.19 | littlerock | G.729 doesnt work, right? |
| 14:09.20 | *** join/#asterisk bbryant (n=brett@68.208.65.50) |
| 14:09.38 | tzafrir_laptop | Why? Is G.729 limited to 500 miles? |
| 14:09.41 | mort_gib | pci:0000:00:0e.0 wctdm+ e159:0001 Wildcard TDM400P REV I |
| 14:09.59 | tzafrir_laptop | mort_gib, so you should see a span for it |
| 14:10.05 | mort_gib | ?? |
| 14:10.18 | *** join/#asterisk RobertLaptop (n=rmiddle@pool-72-72-237-218.altnpa.east.verizon.net) |
| 14:10.23 | mort_gib | tzafrir_laptop: That's all I get |
| 14:10.24 | tzafrir_laptop | (lsdahdi / cat /proc/dahdi/* ) |
| 14:10.43 | mort_gib | and dahdi show channels in the clie shows no channels |
| 14:10.49 | littlerock | does asterisk support G.729 |
| 14:11.06 | *** join/#asterisk coppice (n=chatzill@46.166.17.210.dyn.pacific.net.hk) |
| 14:11.23 | *** join/#asterisk SwK (n=SwK@freeswitch/developer/swk) [NETSPLIT VICTIM] |
| 14:11.26 | mort_gib | tzafrir_laptop: That gives me the two modules I have installed |
| 14:11.36 | mort_gib | 1 FXO and 4 FXS |
| 14:12.08 | *** join/#asterisk intralanman (n=intralan@va-67-76-163-209.sta.embarqhsd.net) |
| 14:12.09 | *** join/#asterisk caio1982 (i=caio1982@CAcert-br/caio1982) |
| 14:12.52 | michael-i | tzafrir_laptop: this seems to have worked but now asterisk is complaining about missing zaptel deps. My problem, not yours. I'll let you know once I've worked around those deps to see if the CXX -> CC fix produces a usable binary |
| 14:13.07 | michael-i | tzafrir_laptop: many thanks |
| 14:14.09 | tzafrir_laptop | michael-i, zaptel (build time) deps are rather simple: the zaptel.h include file, and libtonezone |
| 14:14.42 | tzafrir_laptop | Or rather, libtonezone is only required for chan_zap/chan_dahdi and not for using zap as a timing source |
| 14:15.04 | michael-i | tzafrir_laptop: the problem is getting those to build in my sandbox in t2...i have an outdated patch to deal with and some other updates to make before zaptel builds |
| 14:21.24 | michael-i | tzafrir_laptop: got through the build...still need to test but this saved me including stdc++. thanks again! (time for a coffee) |
| 14:21.27 | *** join/#asterisk verywiseman (n=verywise@unaffiliated/verywiseman) |
| 14:21.48 | *** join/#asterisk sysreq (n=sysreq@unaffiliated/sysreq) |
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| 14:26.22 | [TK]D-Fender | littlerock: Yes, as a PAID & LICEnseD codec. |
| 14:26.28 | [TK]D-Fender | littlerock: www.digium.com |
| 14:29.13 | LuisTorres | Hi |
| 14:29.58 | LuisTorres | just upgraded my box from 1.4.21.2 to 1.4.23.1 and now everytime I pickup a call with *8 it cuts off the caller..., any ideas? |
| 14:30.31 | *** join/#asterisk anonymouz666 (n=anonymou@189.24.33.119) |
| 14:37.16 | phix | [TK]D-Fender: :D |
| 14:39.02 | jasonwoot | anyone using agentCallBackLogin? I need a way to route incoming calls to agentIDs rather than extensions |
| 14:39.08 | *** join/#asterisk magenbrot (n=magenbro@home.magenbrot.net) |
| 14:43.21 | *** join/#asterisk E-bola (i=psybnc@ip181.rev112.brygge.net) |
| 14:43.25 | E-bola | Hi all |
| 14:47.56 | jaytee | jasonwoot, it works in 1.4 but it's deprecated in 1.6 in favor of using an ugly AEL hack |
| 14:48.19 | *** join/#asterisk magenbrot (n=magenbro@home.magenbrot.net) |
| 14:50.18 | *** join/#asterisk axisys (n=axisys@155.70.141.45) |
| 14:51.36 | *** join/#asterisk KavanS (i=proxy@static-71-117-242-28.ptldor.dsl-w.verizon.net) |
| 14:53.47 | E-bola | I am looking for a gsm gateway product to use with our asterisk |
| 14:53.57 | E-bola | Do anybody know any products thats recomendable? |
| 14:54.33 | *** join/#asterisk stevetotaro (n=Steve@c-71-206-29-240.hsd1.md.comcast.net) |
| 14:57.23 | *** join/#asterisk kannan (n=kannan@121.246.242.95) |
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| 15:02.51 | LuisTorres | hey.., sorry for the stupid question, but how apply a xxx.patch on channel.c ? |
| 15:07.37 | *** join/#asterisk anthm (n=anthm@freeswitch/developer/anthm) |
| 15:09.33 | *** join/#asterisk nima0102 (n=nima@91.98.208.7) |
| 15:10.11 | *** join/#asterisk doolph (n=doolph@201.218.108.74) |
| 15:10.14 | *** join/#asterisk Defraz (n=T0tal@fw-poky.fuzecore.com) |
| 15:10.59 | doolph | hello, there's anyway to change the SIP provider error code?, I got a very stupid provider giving me busy code when it is not busy, it's only unvailable or no channels availables by them. |
| 15:12.31 | *** join/#asterisk codefreeze-lap (n=murf@72.21.67.40) |
| 15:12.40 | *** part/#asterisk nima0102 (n=nima@91.98.208.7) |
| 15:13.13 | *** join/#asterisk Jeff_phillips (n=ceramics@66-112-49-13.stat.centurytel.net) |
| 15:13.14 | Jeff_phillips | hi |
| 15:14.38 | *** join/#asterisk Mog (n=mog@c-68-62-218-186.hsd1.al.comcast.net) |
| 15:14.38 | *** mode/#asterisk [+o Mog] by ChanServ |
| 15:14.40 | *** join/#asterisk jpcansa (n=jpbenavi@201.198.231.210) |
| 15:16.22 | *** join/#asterisk gulden (n=gulden@av2-84-90-24-170.netvisao.pt) |
| 15:16.34 | doolph | sup |
| 15:20.05 | *** join/#asterisk davevg (n=davevg__@74.94.3.214) |
| 15:20.50 | *** join/#asterisk lowtek (n=no@adsl-074-185-089-046.sip.bhm.bellsouth.net) |
| 15:21.15 | lowtek | Hi all. Is there a list somewhere that explains the param1,param2,param3 parameters for each of the events in the queues log file? |
| 15:24.00 | *** join/#asterisk Chesther (n=cam2@cam2-mac.cit.cornell.edu) |
| 15:24.19 | *** join/#asterisk ghento (n=ghento@d75-157-192-235.bchsia.telus.net) |
| 15:26.44 | E-bola | dead room today |
| 15:26.52 | [TK]D-Fender | lowYes, in the DOC folder in your source tarball |
| 15:27.28 | *** join/#asterisk killown (n=ukendt@unaffiliated/killown) |
| 15:28.02 | lowtek | [TK]D-Fender: Thanks, will rtfm, lol ;p |
| 15:28.40 | *** join/#asterisk hfb (n=hfb@pool-96-247-109-136.lsanca.dsl-w.verizon.net) |
| 15:29.36 | *** join/#asterisk mcargile (n=mikec@rrcs-24-173-156-170.se.biz.rr.com) |
| 15:30.41 | Chesther | Is anyone on who has experience working with SIP over TLS on * 1.6? |
| 15:30.51 | mcargile | Aside from setting a timeout in the dial command, what can send a SIP call to the timeout extension? |
| 15:31.17 | mcargile | I have a feeling my providers equipment is screwing up but I have very little evidence to back that up. |
| 15:31.42 | *** join/#asterisk n3hxs (n=HAMming@adsl-70-128-62-214.dsl.ltrkar.swbell.net) |
| 15:32.39 | *** join/#asterisk esaym (n=user@cpe-70-120-89-6.satx.res.rr.com) |
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| 15:34.17 | *** mode/#asterisk [+o Deeewayne] by ChanServ |
| 15:36.51 | *** join/#asterisk mgdm (n=michael@river.mgdm.net) |
| 15:40.27 | *** join/#asterisk FireSlash (n=dsfargeg@rrcs-96-11-129-63.central.biz.rr.com) |
| 15:41.10 | FireSlash | I'm trying to send a text only message to other phones via the Manager interface. I've seen some macros to do this, but I've not seen any documentation on how to do it from the manager (at least not a complete working example) |
| 15:41.42 | *** join/#asterisk moy (n=chatzill@74.12.124.89) |
| 15:42.21 | [TK]D-Fender | FireSlash: AFAIK only Cisco's support pushed messages |
| 15:42.50 | FireSlash | [TK]D-Fender, I'm pretty sure our snom 390s do as well, but I could be wrong. I remember reading that SendMessage worked with them |
| 15:43.05 | FireSlash | 360* |
| 15:44.18 | lowtek | Woot! The new sip software for the nortel ip phones works great with asterisk and NAT. |
| 15:44.44 | FireSlash | http://asterisk.snom.com/index.php/Asterisk_1.4/Text_Messages <- There's an example of doing it through the dianplan. However, I'm trying to do it via a manager connection through Python :/ |
| 15:46.20 | *** join/#asterisk harry_v (n=pcsuppor@S0106001d7e52cc78.vs.shawcable.net) |
| 15:48.22 | *** join/#asterisk mazpe (n=mazpe@c-76-109-233-122.hsd1.fl.comcast.net) |
| 15:48.28 | RichardLynch | I've installed asterisk from source, instead of from apt-get. My init.d script has /var/run/asterisk/asterisk.pid (etc) in it. asterisk binary, however, seems to be trying to use /var/run/asterisk.pid, which is root-only. I looked at ./configure --help for a "rundir" or somesuch to no avail. Oooh. Maybe service asterisk start isn't actually using init.d, but some *other* start script??? |
| 15:50.16 | RichardLynch | mcargile: I think there is a DEFAULT timeout for Dial() that will send you there. |
| 15:50.34 | BlargMaN00 | [TH]D-Fender: did you ever get a chance to look at that pastebin that I did?? |
| 15:50.41 | BlargMaN00 | er [TK] |
| 15:52.29 | *** join/#asterisk KU0N (n=kuon@alragore.goyman.com) |
| 15:54.25 | *** join/#asterisk path_ (n=path_@190.21.122.17) |
| 15:54.50 | E-bola | Nobody uses a gsm gateway with asterisk? |
| 15:55.04 | *** join/#asterisk nima0102 (n=nima@91.98.208.7) |
| 15:57.25 | BlargMaN00 | RichardLynch: check out this article... it will tell you everything you need... http://www.voip-info.org/wiki/index.php?page=Asterisk+non-root |
| 15:58.18 | *** join/#asterisk mosty (n=mosty@213-66-224-163-no22.tbcn.telia.com) |
| 15:59.48 | *** join/#asterisk putnopvut (n=putnopvu@asterisk/master-of-queues/mmichelson) |
| 15:59.48 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 16:00.13 | RichardLynch | Yes, but my astrundir is *already* /var/run/asterisk I believe all that stuff has already been done by the original package (Ubuntu) install... |
| 16:00.44 | lowtek | The debian/ubuntu packages will run asterisk as a user not root. |
| 16:00.46 | RichardLynch | Why is asterisk still trying to use /var/run instead? |
| 16:01.02 | lowtek | RichardLynch: Did you install the package and then after the fact decide to download and compile the sources? |
| 16:02.52 | RichardLynch | lowtek: Yes, in fact, I did. |
| 16:03.08 | lowtek | RichardLynch: Did you run it for the first time as root without the -U and -G options? |
| 16:03.20 | RichardLynch | Yes. |
| 16:03.38 | lowtek | RichardLynch: You need to fix all the permissions before doing anything else. |
| 16:04.35 | lowtek | RichardLynch: oh, and remove the package and any of it's reminance, dpkg --purge |
| 16:04.40 | RichardLynch | It seemed to work fine last night. Oh well. Fix them which way at this point? I have a slight preference for keeping it as user asterisk. |
| 16:04.52 | RichardLynch | I do not want to lose the .conf files... |
| 16:05.04 | lowtek | RichardLynch: Back them up? |
| 16:05.12 | eppigy | hello |
| 16:05.14 | eppigy | i am dave |
| 16:05.47 | brunner | Is there a way to pass a caller ID privacy flag with asterisk, but still send the real number? |
| 16:05.50 | lowtek | eppigy: hi dave, I am lowtek, how may we serve you today? |
| 16:08.14 | *** join/#asterisk L|NUX (n=linux@unaffiliated/lnux/x-10290) |
| 16:08.24 | *** join/#asterisk bmoraca (n=chatzill@209.60.253.58) |
| 16:10.29 | *** join/#asterisk flujan (n=flujan@201-68-191-1.dsl.telesp.net.br) |
| 16:13.07 | jpcansa | my asterisk is crashing for some reason, how can i set it up to restart if the asterisk service stops?? |
| 16:13.24 | lowtek | jpcansa: probably best to figure out why it's crashing and fix it. |
| 16:13.36 | jpcansa | thats what i´m doing now |
| 16:13.44 | mort_gib | jpcansa: I use monit |
| 16:13.54 | jpcansa | but i need it to set that up as a temp workaround |
| 16:14.12 | mosty | jpcansa, use safe_asterisk (a shell script wrapper for asterisk, that does what you want) |
| 16:14.27 | E-bola | mort_gib: ever used a gsm gateway? |
| 16:14.38 | mazpe | how can i force my ivr a timeout or something that will allow me to dial extension 101, instead of having it to go option 1 in the ivr |
| 16:14.39 | mazpe | http://rafb.net/p/phuX7E45.html |
| 16:14.44 | jpcansa | thank mort_gib and mosty |
| 16:14.49 | jpcansa | i´ll take a look |
| 16:15.04 | lowtek | mazpe: t,1,goto(101) |
| 16:15.21 | flujan | hello guys |
| 16:15.32 | flujan | i am trying to trunk two asterisk boxes using sip. |
| 16:15.44 | mazpe | lowtek: i have to do that for all extensions? |
| 16:15.45 | flujan | machine one has the digital E1 boards |
| 16:15.48 | seanbright | RichardLynch: PM me your root password and public IP. |
| 16:15.49 | seanbright | :P |
| 16:15.55 | lowtek | flujan: sip doesn't do trunks, only channels |
| 16:16.07 | flujan | and machine two has the pabx with mixmonitor and the stuff. |
| 16:16.08 | mosty | RichardLynch, so you installed the asterisk .deb files and then compiled asterisk manually also? |
| 16:16.09 | mazpe | lowtek: like 102, 103, etc |
| 16:16.13 | RichardLynch | seanbright: It's not my box... |
| 16:16.20 | seanbright | (i was kidding anyway) |
| 16:16.30 | flujan | so the problem is the following: i am receiving calls on machine one and it is passing to machine 2. |
| 16:16.30 | seanbright | i think mosty is heading the right way, however. |
| 16:16.35 | lowtek | mazpe: Maybe you should give more detail for what you're looking to do. |
| 16:16.36 | RichardLynch | Yes. I'm actually needing to hack * source to do something with Festival (voice choice in dial plan) |
| 16:16.36 | flujan | today it stop working. |
| 16:16.48 | flujan | machine 2 keep saying that there is no user. |
| 16:17.00 | lowtek | flujan: what's changed since it was working? |
| 16:17.06 | mort_gib | E-bola: Sure |
| 16:17.24 | mazpe | lowtek: When someone calls the ivr, it should allow you to dial the extension 101, 102, etc or just 1 for a submenu of the ivr |
| 16:17.38 | *** join/#asterisk joesuffceren (n=chatzill@ip70-190-146-85.ph.ph.cox.net) |
| 16:18.02 | mosty | RichardLynch, try /usr/sbin/asterisk -r |
| 16:18.08 | mazpe | lowtek: at the moment when i dial 101, goes to option 1 of the ivr and then it dials 0 |
| 16:18.27 | E-bola | mort_gib: anything you can recomend? |
| 16:18.34 | flujan | lowtek: nothing that is the point |
| 16:18.41 | E-bola | Im looking for a gsm gateway to use with asterisk, and there is like a million different products |
| 16:18.42 | flujan | i am pasting the configuration files . |
| 16:18.44 | lowtek | That's going to be confusing as you're waiting for 1XX. You need to have your extensions seperate from your IVR options such as 8XX so there's no odd timeouts when dialing just a "1" or do something like "press 4 to dail an extension" and then go into another context where the 1XX extensions are. |
| 16:19.00 | *** join/#asterisk gulden (n=gulden@av2-84-90-24-170.netvisao.pt) |
| 16:19.10 | mort_gib | E-Bola: Depends, do you want a SIP/GMS or FXO/GSM gateway?? |
| 16:19.11 | joesuffceren | zapata.conf / PRI question: I need to change the number of channels on my PRI from 23 to 21 (I've had the telco provider pull two channels off for use as 'analog' lines). I've made thenecessary change in zapata.conf, but I would like to effect the change without restarting asterisk or terminating existing zap calls |
| 16:19.17 | RichardLynch | usr/sbin/asterisk does the same thing -- I'm reasonably certain there is only the one binary asterisk on the box to run at all. (Not counting usr/src/asterisk*/main/asterisk |
| 16:19.18 | *** join/#asterisk pwell (n=domin8@pool-72-80-203-194.nycmny.east.verizon.net) |
| 16:19.20 | E-bola | mort_gib: SIP/GSM |
| 16:19.26 | *** join/#asterisk orkid_ (n=orkid@unaffiliated/orkid) |
| 16:19.31 | flujan | lowtek: can u check it? http://pastie.org/408405 |
| 16:19.32 | E-bola | to be able to call out over GSM to mobile numbers, to save money |
| 16:19.33 | joesuffceren | I think doing a module reload on chan_zap will kill all existing calls |
| 16:19.38 | flujan | the main problem is |
| 16:19.40 | joesuffceren | any other way to reload that ingo? |
| 16:19.42 | joesuffceren | info* |
| 16:19.46 | E-bola | But im worried about the quality of the various products |
| 16:19.50 | flujan | when i have the parameter fromuser on the gateway machine it works |
| 16:19.56 | RichardLynch | I gotta go -- Now that I know what it is, I can probably track it down and much with chown/chmod enough to make it go. |
| 16:19.57 | flujan | when i don't have it |
| 16:20.00 | flujan | it does not work |
| 16:20.06 | RichardLynch | Thanks all! |
| 16:20.08 | mort_gib | E-bola: I always use FXo/GSM I know it ends up more expensive, but... |
| 16:20.12 | mazpe | lowtek: following your logic is a bad idea then to have extension 101 and 1010 |
| 16:20.29 | E-bola | mort_gib: sip/gsm should also be better quality since u onyl convert the signal once or something... |
| 16:21.16 | mort_gib | E-bola: True I suppose.... I have used Portech's gateways, they seems to be decent enough |
| 16:21.34 | coppice | most of the SIP GSM gadgets still go through an analogue step, as most GSM modems have no digital input for voice |
| 16:22.00 | E-bola | Portech seems very cheap? Very taiwan "like" |
| 16:22.06 | E-bola | they may work fine though |
| 16:22.15 | E-bola | Thats my problem, i got no idea which products are best |
| 16:22.38 | E-bola | we got a distributor who has some called teles |
| 16:22.44 | mort_gib | E-bola: I prefer TelecomFM's products though |
| 16:22.48 | E-bola | they seem pretty good, but again i cant find much info on it |
| 16:22.49 | mort_gib | -Looking for a model |
| 16:22.58 | flujan | i am getting this message: chan_sip.c:12483 handle_response_invite: Received response: "Forbidden" from '"1199514177" <sip:1199514177@172.16.0.24>;tag=as368667ef' |
| 16:23.07 | flujan | any tips? |
| 16:23.30 | mort_gib | E-bola: http://www.voipon.co.uk/telecomfm-onestream-g-p-1143.html |
| 16:23.55 | *** part/#asterisk gulden (n=gulden@av2-84-90-24-170.netvisao.pt) |
| 16:23.58 | ber___ | does anyone have a suggestion if i wanted to hang up one leg of a call then send the remaining leg to an IVR? |
| 16:24.25 | E-bola | mort_gib>: I need atleast 4 channels, maybe more |
| 16:24.33 | mort_gib | E-bola: The Portech's are a bit "plastic", I mean they work but.... |
| 16:24.40 | E-bola | problem is unless i can find some1 who tried a givwen product i got no idea which works best |
| 16:24.58 | E-bola | also i wonder how fast they work |
| 16:25.02 | mort_gib | Okay, I can RECOMMEND the TelecomFM's |
| 16:25.17 | E-bola | like i'd have planned to simply have a dialplan on asterisk that directed calls to callphone numbers through the gateway |
| 16:25.27 | E-bola | but i wonder how long a delay it will cause before the user hears ringing |
| 16:26.01 | mort_gib | E-bola: They work fairly fast, like not much of a delay |
| 16:27.08 | E-bola | how many do you use in paralel? Or have u just used it for ur own setup? |
| 16:27.32 | mort_gib | E-bola: biggest install was 2 X GSM |
| 16:27.57 | *** join/#asterisk mosty (n=mosty@213-66-224-163-no22.tbcn.telia.com) |
| 16:28.05 | mort_gib | -So far |
| 16:28.05 | E-bola | So 2 hardware boxes each with 1 sim card? |
| 16:28.35 | mort_gib | E-bola: I wish :-) Each (GSM) line needs one SIM card with the right contract etc |
| 16:28.53 | pwell | how does one get a local telephone number, not line? |
| 16:28.56 | Gido-E | current down: 6660 :-) |
| 16:29.05 | E-bola | mort_gib: sure thats what i meant |
| 16:29.08 | E-bola | but How do you get asterisk to "spread out" the calls between them then? |
| 16:29.12 | E-bola | thats also a concern for me |
| 16:29.25 | jpcansa | any one familiar with opensuse 10.2 ?? i cannot make dahdi to load on boot |
| 16:30.18 | lowtek | E-bola: dial(sip/ext1&sip/ext2) |
| 16:30.20 | mort_gib | E-Bola: Easy with the FXO/GMS's |
| 16:30.37 | E-bola | mm so if one of the channels is busy it just uses the other one? |
| 16:31.01 | *** part/#asterisk retentiveboy (n=pdugas@74.95.28.37) |
| 16:31.09 | mort_gib | E-bola: Sure, don't know if you have two devices, but it should be straight forward |
| 16:31.10 | E-bola | I wonder how the devices with 2,4 or 8 sim cards handle it |
| 16:32.01 | mort_gib | E-bola: They handle it internally, or you can handle it.... They come with quite advanced routing in the Gateways |
| 16:32.04 | E-bola | wish he had some devices to test |
| 16:32.17 | mort_gib | E-bola: Where are you?? |
| 16:32.22 | E-bola | Denmark |
| 16:32.28 | mort_gib | ah... |
| 16:33.03 | E-bola | our distributor offered to borrow me a 4 channel device |
| 16:33.10 | E-bola | but he couldnt get it for me before week 13 |
| 16:33.13 | E-bola | sux to wait :) |
| 16:33.19 | mort_gib | E-bola: Gotta love DK! |
| 16:33.35 | E-bola | Cant imagine Gibraltar being better hehe |
| 16:34.06 | RichardLynch | I'm back. Okay guys, 'splain this: root@Astro:/etc/asterisk# /etc/init.d/asterisk start |
| 16:34.06 | RichardLynch | PIDFILE /var/run/asterisk/asterisk.pid |
| 16:34.06 | RichardLynch | Starting Asterisk PBX: Unable to open pid file '/var/run/asterisk.pid': Permission denied |
| 16:34.22 | lowtek | chown -R asterisk:asterisk /var/run/asterisk |
| 16:34.28 | mort_gib | E-bola: Some things are better, some far worse |
| 16:34.37 | RichardLynch | Why/how is it getting /var/run/asterisk.pid instead of /var/run/asterisk/asterisk.pid |
| 16:34.38 | mort_gib | E-bola: Spain is REALLY hostile |
| 16:34.47 | E-bola | I wonder if i shoudl just try to call our via a bluetooth mobile |
| 16:34.52 | lowtek | Check /etc/asterisk/asterisk.conf |
| 16:34.56 | E-bola | mort_gib: ive had soem crappy experiences in spain |
| 16:35.03 | E-bola | like their border police are complete idiots |
| 16:35.11 | RichardLynch | /etc/asterisk/asterisk.conf: http://pastebin.com/d43107828 |
| 16:35.34 | RichardLynch | root@Astro:/etc/asterisk# ls -als /var/run | grep aster |
| 16:35.34 | RichardLynch | 0 drwxr-xr-x 2 asterisk asterisk 40 2009-03-04 23:30 asterisk |
| 16:35.39 | lowtek | RichardLynch: Are you starting via an init script? |
| 16:36.07 | mort_gib | E-bola: Me too, and I live here! |
| 16:36.17 | RichardLynch | Yes /etc/init.d/asterisk, which has always had -U asterisk and I just added -G asterisk in the faint hope it would help. |
| 16:36.21 | *** join/#asterisk outtolunc (n=me@adsl-66-218-53-172.dslextreme.com) |
| 16:36.25 | *** join/#asterisk glak (n=glak@bergen.nextgentel.net) |
| 16:36.35 | lowtek | RichardLynch: See if it's the culprit, pastebin it. |
| 16:36.55 | lowtek | brb |
| 16:37.21 | mort_gib | E-bola: Policia Local are really scary... Torpe con armas |
| 16:37.34 | glak | Hi! I want to setup an Asterisk, but I dont know how. I have a dedicated computer with Ubuntu 8.10. What else do I need, exept the software ofcourse? |
| 16:37.35 | E-bola | hehe |
| 16:37.37 | E-bola | im off later |
| 16:37.40 | E-bola | thanks for the info mort |
| 16:37.56 | *** join/#asterisk t_corr (n=richard@rrcs-74-218-125-86.central.biz.rr.com) |
| 16:38.28 | t_corr | Does anyone know if there's a built in way to say a length of time? (example: 1 hour, 32 minutes, 17 seconds) like a person would? |
| 16:39.29 | RichardLynch | etc/init.d/asterisk: http://pastebin.com/d747d8d21 |
| 16:39.39 | RichardLynch | glak: Time and patience. :-) |
| 16:40.01 | glak | hehe |
| 16:40.04 | glak | dont have :p |
| 16:40.25 | glak | but seriously, do I need anything else? |
| 16:40.29 | *** join/#asterisk NewCastleScott (n=Frank_Ri@76-235-245-36.lightspeed.cicril.sbcglobal.net) |
| 16:40.32 | RichardLynch | glak: Setting up asterisk to run is easy. Getting a dialplan that does exactly what you want... Depends on the dialplan. |
| 16:40.41 | lowtek | Got me RichardLynch ... |
| 16:40.59 | glak | Im going to use this only to do calls out, private usage |
| 16:41.21 | NewCastleScott | hello all, speaking of setting up * can someone give me a link to simplify the config? |
| 16:42.28 | RichardLynch | Could I have messed up compiling in some what that it would ignore /etc/asterisk/asterisk.conf ??? |
| 16:42.42 | RichardLynch | What is the default dir for config files from source compile? |
| 16:43.22 | RichardLynch | Is it like /usr/local/lib/asterisk or something? |
| 16:43.36 | eppigy | /etc/asterisk/ |
| 16:44.23 | RichardLynch | Afraid not, according to ./configure --help. It's PREFIX/asterisk, and PREFIX defaults to /usr/local, damn me. |
| 16:45.23 | *** join/#asterisk fcois93 (n=Administ@bagnolet.acropolistelecom.net) |
| 16:45.47 | *** join/#asterisk CunningPike (n=arodgers@204.239.10.119) |
| 16:47.18 | *** join/#asterisk docelmo (n=vircuser@pool-141-152-222-192.lyn.east.verizon.net) |
| 16:49.38 | *** join/#asterisk freakazoid0223 (n=matt@pool-71-246-17-74.phlapa.fios.verizon.net) |
| 16:51.25 | fcois93 | I need to do a server to do a lots of conferences. I hear that without digium card, it is hard to have good performance. is it right ? |
| 16:51.29 | glak | Is there a simplifyed "how-to" regarding setup of the Asterisk? |
| 16:54.05 | carrar | glak, like a book? |
| 16:54.34 | glak | carrar: Well, yes :p |
| 16:54.39 | carrar | ~book |
| 16:54.40 | jbot | book is, like, probably Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://tfot.leifmadsen.com or see ~buybook |
| 16:54.52 | glak | is that the one with 600 pages? |
| 16:55.16 | carrar | no, it's about 576 pages |
| 16:55.21 | glak | ack |
| 16:55.23 | glak | too much |
| 16:55.29 | glak | i need like one page :p |
| 16:55.50 | carrar | so you don't really want to learn how to set it up is what you're saying |
| 16:56.10 | glak | hehe, I do, but Im lacking patience |
| 16:56.37 | carrar | Asterisk is going to drive you nuts then |
| 16:56.43 | carrar | if you don't have patience |
| 16:56.57 | glak | ack |
| 16:57.35 | *** join/#asterisk jameswf (n=james@unaffiliated/jameswf-home) |
| 16:57.45 | *** join/#asterisk kannan (n=kannan@121.246.242.95) |
| 16:58.03 | *** join/#asterisk Assimilate (n=Assimila@72.22.242.66) |
| 16:58.44 | RichardLynch | Do I want ./configure --sysconfdir=/etc or ./configure --sysconfdir=/etc/asterisk ??? |
| 16:58.45 | carrar | Just google |
| 16:58.59 | carrar | you'll find lots of asterisk howto for the short of patience glak |
| 16:59.19 | carrar | lowtek? |
| 16:59.24 | fcois93 | I need to do a server to do a lots of conferences. I hear that without digium card, it is hard to have good performance. is it right ? |
| 16:59.26 | carrar | *lowtek* Yea, do what we did, use IronVoice (http://ironvoice.com), it's a hosted solution, freakin great! |
| 16:59.26 | carrar | <PROTECTED> |
| 16:59.34 | fcois93 | what is the limit without the card? |
| 16:59.38 | carrar | whats with the spamming your commercial product in here? |
| 17:00.07 | mosty | RichardLynch, why don't you just use the ubuntu packages? |
| 17:00.11 | lowtek | carrar: lol, no spam here, just a great service, you should check them out. |
| 17:00.21 | RichardLynch | glak: Installing it and making it "go" is easy. Getting it to do what you WANT is the 600 pages. |
| 17:00.21 | carrar | I didn't ask |
| 17:00.35 | lowtek | carrar: well, there's nothing preventing me from suggestion, cheers |
| 17:00.57 | *** join/#asterisk keulin (n=cray@bne75-5-82-231-224-34.fbx.proxad.net) |
| 17:01.01 | RichardLynch | mosty: I'm altering /usr/src/asterisk/app/app_festival.c to allow me to choose a voice from the dial plan... I don't think Ubuntu has a package with source code I wrote last night :-) |
| 17:01.05 | carrar | Marketing tactics like that reflect poorly on whatever you are selling |
| 17:01.16 | *** join/#asterisk jtodd (n=jtodd@41.sub-75-208-159.myvzw.com) |
| 17:01.16 | *** mode/#asterisk [+o jtodd] by ChanServ |
| 17:01.23 | lowtek | carrar: I'm not selling anything jackass, I was just making a referral. |
| 17:01.25 | glak | RichardLynch: yea, the only thing I need, is to make calls out. Dont need my own number ect |
| 17:01.31 | *** join/#asterisk stevetotaro (n=Steve@c-71-206-29-240.hsd1.md.comcast.net) |
| 17:01.43 | carrar | looks like marketing to me |
| 17:01.45 | mosty | RichardLynch, ok, so in that case i recommend that you remove the ubuntu asterisk pacakges if you're compiling manually |
| 17:01.47 | carrar | advertising |
| 17:02.29 | mosty | so there's no confusion as to which version you're starting, and which version you're trying to connect with |
| 17:02.32 | RichardLynch | glak: Make your own personal calls out, or for a large office complex or what? Just making your own calls out... Get skype :-) But it should not be TOO hard to setup just for that. You can skip about 500 pages. |
| 17:02.41 | *** join/#asterisk nicoAMG (i=asgalt@201.203.96.42) |
| 17:02.51 | carrar | fcois93, you can use a cheap card like x100p or something |
| 17:04.10 | glak | RichardLynch: yea, private usage only. Friends and family :) |
| 17:04.10 | fcois93 | carrar: yes, I know, but if I dont use card, is there a harder limit than if I had a card ? |
| 17:04.10 | glak | Skype is not free tho.. :/ |
| 17:05.12 | RichardLynch | glak: How much is your time worth? Do you flip burgers for a living? Up to you. The hardware and the hours will add up, and you have to maintain it... |
| 17:06.17 | doolph | No such command 'sip show peers' (type 'help sip show' for other possible commands) |
| 17:07.50 | doolph | nvm I had to reboot |
| 17:08.04 | *** join/#asterisk pdmmm (i=pdm@pdm.sh) |
| 17:09.46 | glak | RichardLynch: well, doing my bachelors degree, working 60%, have a baby and a girfriend.. Can work late sundays :p |
| 17:10.19 | glak | I do have some time, but is this going to take 1 or 10 hours? |
| 17:10.48 | glak | dont want to start a project, finding out that I fail big time and throw my computer out the window in flaming anger |
| 17:11.39 | glak | but I do have a dedicated computer for this project only, with linux on |
| 17:13.05 | pdmmm | project? |
| 17:13.15 | [TK]D-Fender | glak: * has a learning curve to it and MANY conf files and things to learn. |
| 17:13.24 | [TK]D-Fender | glak: This will take days most likely |
| 17:13.33 | glak | woot, seriously? |
| 17:14.03 | *** join/#asterisk af_ (n=getsmart@88-149-230-241.dynamic.ngi.it) |
| 17:14.18 | glak | I do have some skills reading and understandig conf files |
| 17:14.32 | glak | but is there alot for confing to do? |
| 17:14.36 | glak | of* |
| 17:15.07 | mort_gib | glak: Tk is right, but you will have fun from the beginning, you only start having a not so nice time much later when you dive in deeper :-) |
| 17:15.18 | glak | hehehe |
| 17:15.57 | glak | but only for simple usage, calling from one phone out, will this take som much time? |
| 17:16.22 | *** join/#asterisk outtolunc (n=me@adsl-66-218-53-172.dslextreme.com) |
| 17:17.04 | mort_gib | glak: Well, depends... I had my first call working in about 4 hours (not including OS) |
| 17:17.16 | glak | hm |
| 17:17.19 | glak | not too bad |
| 17:18.00 | mort_gib | glak: Yeah, well you later find out how much you need to find out! |
| 17:18.28 | mort_gib | glak: All good fun, only when you have to deal with Telcos the pain starts... |
| 17:18.29 | glak | but after 4 hours, the phone worked to call out, thats it? |
| 17:18.38 | glak | telcos? |
| 17:18.56 | *** join/#asterisk outtolunc (n=me@adsl-66-218-53-172.dslextreme.com) |
| 17:19.51 | mort_gib | glak: In my case Gibtelecom, also known as Gibdisconnect... |
| 17:20.37 | glak | ah, but they cant do anything. Besides, telecompanyes in Norway (Telenor) is not too smart anyway |
| 17:21.00 | mort_gib | glak: That is my very problem with Gibdisconnect |
| 17:21.17 | glak | hm |
| 17:21.21 | mort_gib | -Not too damn smart, naa saa du er en Fjaelabe :-) |
| 17:21.42 | glak | hahaha |
| 17:22.18 | glak | fjellape, hysterisk :D |
| 17:22.33 | mort_gib | I though so :-) |
| 17:22.37 | glak | hehe |
| 17:22.50 | glak | but how can they interfeer? |
| 17:23.06 | glak | urr, not spelled right, you know what I mean :p |
| 17:23.49 | mort_gib | as long as stick to one analogue line, you don't need CID you'll be fine |
| 17:24.17 | tzafrir_laptop | hmm... how *does* caller ID work in Norway? |
| 17:25.01 | tzafrir_laptop | mort_gib, but sticking with analouge means you miss all the real fun |
| 17:25.01 | glak | tzafrir_laptop: good question :9 |
| 17:25.46 | mort_gib | tzafrir_laptop: Yes :-) Although there is a certain, childish pleasure in following any call in the CLI :-) |
| 17:26.18 | glak | mort_gib: Off topic; do danish people call us norwegians for mountian monkeys? (really funny tho) :) |
| 17:26.19 | mort_gib | I connected my "office" setup to Voipstunt a few minutes back... |
| 17:26.40 | mort_gib | glak: Well, only when you are not around to bicth about it :-) |
| 17:26.46 | glak | haha |
| 17:26.49 | mort_gib | :-) |
| 17:27.28 | tzafrir_laptop | glak, I just call all .no folks "trolls" :-) |
| 17:27.34 | glak | hehehe |
| 17:27.35 | mort_gib | If it's any consolation, we like Norwegians MUCH better than those other guys next to you... |
| 17:27.56 | glak | well, the sweds of finns or russia? |
| 17:28.07 | mort_gib | Swedish |
| 17:28.35 | mort_gib | Yeah, so most of the girls are blond, but do you HAVE to wear so LOUD shirts! |
| 17:28.40 | glak | yea, damn hard to understand in phone (I work at NextGenTel, Norways next biggest broadband supplier) |
| 17:28.51 | glak | mort_gib: I know |
| 17:29.18 | glak | but when danish people call, the most used sentence is: can you repeat that? :p |
| 17:29.20 | mort_gib | Well, Skaane is like another language all together, Stockholm i understand better.. Go figure |
| 17:29.28 | tzafrir_laptop | wonders if mort_gib actually wants his SIP connection to work well |
| 17:29.43 | glak | Skaane is insane :p |
| 17:29.55 | mort_gib | well, I always understood .no better |
| 17:30.01 | mort_gib | tzafrir_laptop: -Sorry?? |
| 17:30.59 | glak | *afk* |
| 17:34.34 | glak | back to topic: I did manage to get halfway through BASIC and AutoCAD for dummies... Chance I'll get asterisk working for simple use? |
| 17:35.41 | Jeff_phillips | anyone ever use chan_mobile here? |
| 17:36.11 | lowtek | glak: you can try trixbox, it may be better suited to your need for quick setup, join #trixbox or #freepbx |
| 17:36.49 | glak | trixbox hvaing the same function as asterisk? |
| 17:37.09 | lowtek | It's based on asterisk. |
| 17:37.27 | mort_gib | glak: Stay with asterisk vanilla |
| 17:37.27 | hardwire | sigh |
| 17:37.37 | glak | asterisk vanilla? |
| 17:37.59 | tzafrir_laptop | glak, actually, FreePBX is quite confusing if you want to learn Asterisk |
| 17:38.09 | Corydon76-dig | prefers Asterisk chocolate |
| 17:38.57 | glak | well, I just want to use the free phone idea :) |
| 17:39.14 | hardwire | I'd love to get my hands on an open source fax machine |
| 17:39.20 | hardwire | add hylafax support |
| 17:39.26 | hardwire | put playdo in the pstn ports |
| 17:40.12 | tzafrir_laptop | hardwire, iaxmodem? |
| 17:40.30 | mort_gib | glak: Get a Soekris 5501, Debin, asterisk 1.6, Dahdi and some really strong Coffee!! |
| 17:40.31 | Chesther | Just find an old printer and hack in an embedded linux box somewhere inside the case. |
| 17:40.35 | hardwire | tzafrir_laptop: I just want to address the issue of people needing a fax machine |
| 17:40.55 | hardwire | like.. if hp open sourced just one firmware.. that model would be incredibly popular |
| 17:41.01 | tzafrir_laptop | actually for embedded, Alix looks quite nice |
| 17:41.10 | hardwire | is ordering some alix boards tomorrow |
| 17:41.11 | hardwire | woot |
| 17:41.53 | glak | hm, took a look at trixbox. seems like a easy install |
| 17:45.19 | coppice | hardwire: what would you change in a FAX machine if you had the source code? |
| 17:45.28 | mort_gib | glak: Please don't |
| 17:46.27 | glak | hehe |
| 17:47.07 | glak | I should rather fiddle with asterisk? |
| 17:47.19 | hardwire | coppice: add ip fax support |
| 17:47.36 | hardwire | make an e164 like directory for number->email |
| 17:47.40 | coppice | why not just use the IP FAX support hey have? |
| 17:48.01 | mort_gib | glak: How about AsteriskNow?? I haven't tried it, but.... |
| 17:48.07 | hardwire | afaik most of the fax machines I've worked with only work with pstn |
| 17:48.22 | glak | mort_gib: I'll google it |
| 17:48.24 | hardwire | coppice: can you shoot me some info on fax machines that support say.. hylafax or similar? |
| 17:48.29 | hardwire | I've googled.. and failed |
| 17:48.36 | tzafrir_laptop | glak, do you have asterisk installed? |
| 17:48.36 | coppice | many of the cheap multi-function machines with ethernet support can do IP FAX |
| 17:48.45 | hardwire | coppice: what kind of IP FAX? |
| 17:48.46 | tzafrir_laptop | What Linux distribution do you have right now? |
| 17:48.57 | coppice | T.37 usually |
| 17:49.00 | hardwire | err |
| 17:49.12 | hardwire | so it still uses a POTS connection |
| 17:49.12 | glak | tzafrir_laptop: no, I have a Dell OptiPlex with Ubuntu 8.10 |
| 17:49.29 | coppice | why would it? |
| 17:49.47 | hardwire | coppice: so how does it negotiate a call with a server? |
| 17:49.49 | hardwire | SIP? |
| 17:49.56 | tzafrir_laptop | The Ubuntu 8.10 Asterisk package is a relatively good start |
| 17:50.09 | coppice | T.37 is a standard for embedding FAXes in e-mails |
| 17:50.25 | hardwire | coppice: ah |
| 17:50.26 | *** join/#asterisk n3hxs (n=HAMming@adsl-70-128-62-214.dsl.ltrkar.swbell.net) |
| 17:50.31 | hardwire | coppice: so scan to email? |
| 17:50.44 | coppice | or print from email |
| 17:51.01 | hardwire | but it never goes to a destination POTS fax machine |
| 17:51.08 | hardwire | which is what I'd be hoping to get working |
| 17:51.17 | glak | tzafrir_laptop: I'll look in to it |
| 17:51.23 | hardwire | by using say a hylafax server. |
| 17:51.33 | coppice | so you want to do T.38 from the FAX machine? |
| 17:52.01 | hardwire | coppice: not really.. I would want to make a fax machine into a hylafax client. |
| 17:52.07 | hardwire | which isn't t.38 |
| 17:53.16 | hardwire | and potentially ammend the setup by adding support for fax->email based on destination fax number and a global directory. |
| 17:54.12 | coppice | you can configure a multi-function machine to behave in that way with hylafax. scan to hylafax and print from hylafax |
| 17:54.28 | mort_gib | hardwire: Why do you need a Fax machine for that?? |
| 17:54.57 | hardwire | this has nothing to do with me and everything to do with crazy old people. |
| 17:54.57 | Jeff_phillips | I need to get something together for sending faxes as well |
| 17:55.37 | glak | tzafrir_laptop: Where can I find more info regarding Ubuntu 8.10 and Asterisk? |
| 17:56.19 | tzafrir_laptop | glak, start with /usr/share/doc/asterisk , I guess |
| 17:56.34 | hardwire | mort_gib: wouldn't it be nice to offer a hard fax solution that acts as a hylafax client? |
| 17:56.37 | hardwire | no pots |
| 17:56.46 | tzafrir_laptop | glak, see README.Debian |
| 17:56.48 | mort_gib | ?? No |
| 17:56.55 | hardwire | why not? |
| 17:57.04 | hardwire | I don't quite understand why this wouldn't be more popular. |
| 17:57.24 | mort_gib | Well WHY would you need a "hard fax solution" ?? |
| 17:57.26 | mvanbaak | because fax machines are annoying |
| 17:57.43 | mort_gib | Yeah, so get rid of them already! |
| 17:57.49 | mvanbaak | coppiers, printers, faxmachines |
| 17:58.01 | mvanbaak | they all give me the same feeling as the dude in Office Space |
| 17:58.05 | hardwire | mort_gib: an idealist eh? |
| 17:58.10 | hardwire | no room for you in common b2b! |
| 17:58.13 | hardwire | back to your cave! |
| 17:58.21 | hardwire | or cloud!.. whatever. |
| 17:58.23 | mort_gib | I have done a lot of Hylafax work and I can't see any justification! |
| 17:58.30 | mort_gib | Yeah, that's me |
| 17:58.36 | mort_gib | Cage geek! |
| 17:58.40 | hardwire | heh |
| 17:58.43 | mort_gib | eh, cave geek |
| 17:58.48 | mort_gib | :-) |
| 17:58.54 | glak | tzafrir_laptop: you are referring to the installfile now right? |
| 17:58.55 | hardwire | mort_gib: we have like 4 fax machines here |
| 17:59.08 | hardwire | we get orders in, fax stuff to insurance companies and the like |
| 17:59.17 | hardwire | if we were cool we'd never need them.. but we're not. |
| 17:59.29 | hardwire | I'd just set up a comm station with a phone, scanner, printer, and that's that. |
| 17:59.38 | hardwire | but it would be nice to wrap that all into existing packaging |
| 17:59.41 | hardwire | vs tossing fax machines |
| 17:59.45 | tzafrir_laptop | glak, start with: aptitude install asterisk |
| 17:59.56 | tzafrir_laptop | (that is: install the package asterisk) |
| 17:59.59 | mort_gib | configured an old Optiplex as a hylafax server for a company, got hold of an 8 port modemcard, 6 months later they were receiving some 2.000 Faxe's a month |
| 18:00.11 | hardwire | mort_gib: yup |
| 18:00.15 | glak | tzafrir_laptop: well, Im not at home now, cant do much :p |
| 18:00.16 | pwell | any of you heard of magic jack? If so.. how are they able to give you a local number? |
| 18:00.21 | hardwire | we have hylafax hooked up to a t1 through asterisk and 8 iaxmodems |
| 18:00.25 | hardwire | but it's not on site. |
| 18:00.31 | loather | pwell:avoid magic jack like the plague |
| 18:00.33 | hardwire | hylafax clients are tcp/lossless |
| 18:00.44 | coppice | 2000 faxes a month is chicken feed :-) |
| 18:00.55 | hardwire | using a directory that correlates a number to a preferred email destination is more progressive in my opinion than just telling the world to toss their fax machines. |
| 18:00.56 | pwell | 3I just dont understand how to create my own cheap solution. |
| 18:01.05 | mort_gib | They would use Adobe standard to "digitally sign" -sic- the Faxes and forward them, to the fax server in the other office that would send the emails to the same server |
| 18:01.08 | hardwire | like.. a "next step" scenario |
| 18:01.37 | hardwire | mort_gib: I'd love to set things up that way.. but people here just wanna bang out a few numbers and hit the go button |
| 18:01.43 | hardwire | that's the ease of fax. |
| 18:01.55 | mvanbaak | till it eats your paper |
| 18:02.11 | hardwire | mvanbaak: I agree with both sides here :P |
| 18:02.15 | hardwire | I'm trying to find a compromise. |
| 18:02.18 | mort_gib | well, I configured an Oracle server with all the contacts, used a Windows hylafax client that plugged into that server |
| 18:02.48 | mort_gib | So a 10 min PHP interface managed their "speeddials" |
| 18:03.23 | mort_gib | I later dumped a "customer database into that Oracle server every night... |
| 18:04.21 | mort_gib | so everybody has all contacts when they used their Hylafax client whfc or something |
| 18:04.27 | hardwire | yeh |
| 18:04.30 | hardwire | ideally that would be nice |
| 18:04.34 | hardwire | we use avantfax here |
| 18:04.41 | hardwire | I'm replacing it with my own software soon |
| 18:04.47 | mort_gib | That was really easy... |
| 18:04.53 | hardwire | but it would still be gravy if a MFC were a hylafax client |
| 18:04.56 | Jeff_phillips | i think we need to create a custom context to route the fax machine directly to the dumpster |
| 18:05.05 | pwell | this sucks, I just want an IVR solution with MagicJack |
| 18:05.05 | hardwire | like.. make customer integration a snap |
| 18:05.19 | hardwire | Jeff_phillips: store and forward |
| 18:05.31 | hardwire | Jeff_phillips: accept the call.. store.. then dial them back and poop it back at them |
| 18:05.55 | mort_gib | hardwire: Use the HP 7680 |
| 18:06.05 | hardwire | mort_gib: rawr? |
| 18:06.07 | hardwire | howcome |
| 18:06.13 | coppice | I seem to remember one of hylafax clients can make an MFC behave as a FAX machine. dunno which one though |
| 18:06.28 | hardwire | coppice: that would be clever |
| 18:06.32 | mort_gib | That can Scan a PDF to a (Linux) SAMBA folder, pick it up from Hylafax |
| 18:06.43 | hardwire | I can pimp out an MF2727 to do things similar.. but not wholy like that |
| 18:06.47 | mort_gib | Major setup pain tho |
| 18:06.56 | hardwire | mort_gib: how is the destination addressing done? |
| 18:07.05 | hardwire | btw we have like 4 7680s |
| 18:07.16 | hardwire | they can't scan to folder when they are out of ink |
| 18:07.18 | hardwire | amazing.. right? |
| 18:07.20 | mort_gib | by selecting the destination folder |
| 18:07.30 | glak | hm, after checking out the wiki-part of asterisk, I think there is a flaw in my great plan... What kind of hardware do I need to do this? |
| 18:07.39 | hardwire | mort_gib: so the destination folder = arbitrary phone number? |
| 18:07.49 | mort_gib | That would be one idea.... |
| 18:07.49 | hardwire | does it auto-create destination directories? |
| 18:08.21 | mort_gib | Dunno.... |
| 18:09.01 | mort_gib | Ideally you would have "speeddials" |
| 18:09.09 | mort_gib | and that wouldn't work |
| 18:10.53 | hardwire | mort_gib: so yeh.. having an open source to one or even several popular MFC firmwares |
| 18:10.56 | hardwire | that would be pimp |
| 18:11.12 | hardwire | just flash and do a little dance. |
| 18:11.26 | hardwire | they would never release the 7680 firmware.. that's their bread/butter |
| 18:11.27 | mort_gib | Another idea is scanning to an email address on the Hylafax server like 00123456789@hylafax and then do some scripting in Hylafax |
| 18:11.38 | hardwire | "out of ink - cannot make scans - go buy more HP ink" |
| 18:11.47 | hardwire | mort_gib: I'd thought of that. |
| 18:11.51 | mort_gib | That would be nice, but the MFC's are ont longlived, I doubt it would be worth while |
| 18:12.09 | hardwire | mort_gib: but it could at least start there |
| 18:12.31 | hardwire | what bugs me is that it's just not an option |
| 18:12.42 | mort_gib | Yeah, but you must be majorly bored at work! |
| 18:12.44 | hardwire | people buy these mfc's all the time and use them for fax.. pay the telco for an extra pots line, etc.. |
| 18:12.44 | mort_gib | :-) |
| 18:12.54 | hardwire | mort_gib: naw.. I'm an idealist too.. just different. |
| 18:13.08 | mort_gib | THAT is true, I see that ALL the time |
| 18:13.21 | mort_gib | Yeah, you like pain more than I do :-) |
| 18:13.24 | hardwire | yes |
| 18:13.33 | hardwire | my therapist says I need to fight. |
| 18:13.33 | hardwire | heh |
| 18:14.05 | hardwire | like.. as a diagnosis.. not as a solution. |
| 18:14.11 | mort_gib | Well... Trying to get a new OS written for an £350 printer... |
| 18:14.20 | mort_gib | Uhm, passa tiempo |
| 18:14.21 | hardwire | mort_gib: your right.. insane |
| 18:14.32 | hardwire | new os's for say.. pdas.. that are sub $200.. |
| 18:14.34 | hardwire | insanity |
| 18:14.35 | hardwire | heh |
| 18:14.39 | hardwire | point to openembedded project |
| 18:14.40 | mort_gib | You said it! |
| 18:15.35 | mort_gib | Easier to stick a "gateway" in between the Fax and the POTS |
| 18:17.11 | *** join/#asterisk Micho123 (n=mcho123@94.187.70.48) |
| 18:17.49 | glak | If I run X-Lite on my computer with Asterisk, do I need anything else og hardware and/or software? |
| 18:17.51 | Micho123 | Hi all,When trying to send fax to an extension registered on asterisk I got the following error..handle_response_invite: Got error on T.38 re-invite. Bad configuration. Peer needs to have T.38 disabled....How can I fix this?any Suggestion will be apprecuated |
| 18:18.10 | jaytee | I expect the economy to totally tank now that Michael Jackson had announced he's making a comeback. |
| 18:19.24 | coppice | rather than come back maybe he should try a new career. entertainer, maybe |
| 18:20.47 | t_corr | Does * not have a singular 'hour', 'minute', or 'second' sound file? |
| 18:20.55 | t_corr | I can only find the plural |
| 18:24.49 | *** join/#asterisk orkid_ (n=orkid@unaffiliated/orkid) |
| 18:29.24 | t_corr | Nobody? |
| 18:29.31 | t_corr | must be lunch time |
| 18:29.47 | *** join/#asterisk riddlebox (n=james@75-132-225-75.dhcp.stls.mo.charter.com) |
| 18:35.08 | *** join/#asterisk vncsnvs (n=vncsnvs_@189.27.17.115.dynamic.adsl.gvt.net.br) |
| 18:43.06 | *** join/#asterisk denon (i=denon@synapse.subneural.net) |
| 18:43.06 | *** mode/#asterisk [+o denon] by ChanServ |
| 18:47.15 | glak | Can I use a softphone with Asterisk on a Linux machine? |
| 18:47.38 | guax | yes |
| 18:47.41 | guax | try twinkle |
| 18:47.57 | [TK]D-Fender | glak: What did you want to use * for again? |
| 18:48.29 | glak | private use at home |
| 18:48.36 | *** part/#asterisk vncsnvs (n=vncsnvs_@189.27.17.115.dynamic.adsl.gvt.net.br) |
| 18:48.59 | [TK]D-Fender | glak: Not quite what I'm looking for. What FUNCTIONALITY? |
| 18:49.20 | *** join/#asterisk killown (n=ukendt@unaffiliated/killown) |
| 18:50.49 | glak | err, I want to call out, thats it |
| 18:51.32 | *** join/#asterisk vncsnvs (n=vncsnvs_@189.27.17.115.dynamic.adsl.gvt.net.br) |
| 18:51.47 | [TK]D-Fender | glak: Via what? |
| 18:52.03 | glak | my Linux computer? |
| 18:52.15 | glak | or, Dell w/ Linux :p |
| 18:52.18 | [TK]D-Fender | glak: call WHERE? Via what technology or service? |
| 18:52.34 | glak | ah, cellphones and land lines in norway |
| 18:52.46 | [TK]D-Fender | glak: and the 2nd question? |
| 18:52.50 | glak | via my broadband connection |
| 18:52.56 | [TK]D-Fender | glak: and ITSP then? |
| 18:53.06 | glak | ITSP? |
| 18:53.07 | [TK]D-Fender | ~itsp |
| 18:53.08 | jbot | [~itsp] An ITSP is an Internet telephony Service Provider or "VoIP Telephone company". The allow you to either PLACE calls to the PSTN (called Termination), RECEIVE calls from the PSTN (called Origination), or both. Some offer fixed rates, others $/min. Enter ~itsplist-us (USA) or ~itsplist-ca (Canada) for a listing of popular ITSPs |
| 18:53.34 | glak | aha, well, I dont have one, do I need it? |
| 18:53.43 | glak | (can get) |
| 18:53.53 | [TK]D-Fender | glak: How do YOU think calls magically go from your computer to somebody's cellphone? |
| 18:54.38 | Qwell | [TK]D-Fender: intertubes |
| 18:55.26 | glak | magic I guess :p |
| 18:55.41 | [TK]D-Fender | glak: Time to find a clue then. |
| 18:55.42 | *** part/#asterisk vncsnvs (n=vncsnvs_@189.27.17.115.dynamic.adsl.gvt.net.br) |
| 18:56.16 | glak | just heard from my professor that he installed this on his linux, no ITSP, just via computer, he could ring everybody |
| 18:56.17 | [TK]D-Fender | glak: So far I see no need for * at all. You might jsut as well install a softphone and connect it directly to an ITSP account |
| 18:56.46 | [TK]D-Fender | glak: understand that "linux" does not magically allow you to call somebody's PHONE |
| 18:56.54 | glak | ofcourse not |
| 18:57.07 | glak | this is where asterisk come in? |
| 18:57.19 | [TK]D-Fender | glak: You are either using a hardware interface to use a telco line, or accessing a service that does this for you, typically charged |
| 18:58.06 | [TK]D-Fender | glak: Does owning an answering machine let you place calls? No, you need the LINE IT IS GOING TO USE <- |
| 18:58.16 | glak | I do have a telco line, but with no number. If I order a number, will it be magic then? |
| 18:58.34 | [TK]D-Fender | NFC.... |
| 18:59.23 | [TK]D-Fender | glak: If you can't even work your way around these simple questions don't bother with *. |
| 18:59.39 | *** join/#asterisk stevetotaro (n=Steve@c-71-206-29-240.hsd1.md.comcast.net) |
| 18:59.45 | [TK]D-Fender | glak: * sits in between the things you choose to interface with it. |
| 18:59.50 | *** join/#asterisk Dovid (n=annon@tony09-121-90.inter.net.il) |
| 19:01.04 | glak | well, not too easy to awnser questions when english is not my primary language |
| 19:01.45 | [TK]D-Fender | glak: Well the easy question remains. What service are you looking to use to call the PSTN? |
| 19:02.02 | glak | PSTN beeing? |
| 19:02.06 | [TK]D-Fender | ~pstn |
| 19:02.07 | jbot | methinks pstn is Public Switched Telephone Network, or "please stop the nonsense" |
| 19:02.36 | Dovid | afternoon TK |
| 19:02.47 | [TK]D-Fender | Dovid: Yes... it is afternoon here.... :p |
| 19:03.25 | Dovid | which is y i said afternoon and not evening ;) |
| 19:03.25 | glak | still dont know what that means in norwegian :S |
| 19:03.54 | [TK]D-Fender | glak: PSTN = NORMAL ^%$#ING PHONE NETWORK <- |
| 19:06.15 | glak | hm... I'll just call and bother my professor :p He made it sound so easy |
| 19:06.32 | FireSlash | Where would I go to enable the SIP command MESSAGE? |
| 19:06.49 | [TK]D-Fender | glak: I'm talking about how you're going to call a simple TELEPHONE. |
| 19:07.41 | glak | cant I install asterisk on my Linux to make it act like a server and call via that to a regular telephone via ex Telenor? |
| 19:07.54 | glak | via the regular land lines? |
| 19:08.14 | glak | and the regular land lines are where i get my broadband |
| 19:08.26 | Dovid | glak: yes. but how do you want to call ? VOIP, PRI, POTS ? |
| 19:08.30 | glak | oh |
| 19:08.31 | [TK]D-Fender | glak: glglaI asked you 5 times what service you wanted to use to access the PSTN.... |
| 19:08.42 | glak | why dont you say it like Dovid ? |
| 19:08.49 | [TK]D-Fender | glak: TELENOR IS THAT $#&!ING SERIVCE! |
| 19:08.59 | [TK]D-Fender | glak: You do not need Asterisk at all. |
| 19:09.18 | glak | The clue of this is that I want to call for free, right? |
| 19:09.21 | [TK]D-Fender | glak: Install a softphone. Connect it to Telenor's service. THE END |
| 19:09.30 | jaytee | [TK]D-Fender, step away from the keyboard and count to 20, life is just too short :-) |
| 19:09.56 | [TK]D-Fender | glak: You want free? go find a sevice that is free. Asterisk does not PROVIDE you this service, it jsut lets you USE it once you have it |
| 19:10.20 | [TK]D-Fender | jaytee: Other people's could be immensely shorter.... |
| 19:10.22 | [TK]D-Fender | goes to grab Mr. Pointy |
| 19:10.28 | *** join/#asterisk joshaidan (n=joshaida@S01060090f8009fa6.tb.shawcable.net) |
| 19:10.28 | jaytee | [TK]D-Fender, hehehe |
| 19:10.42 | jaytee | run glak, run! |
| 19:11.26 | Dovid | TK: I told some one may he live past 120 today and his response ? why are you cursing me .... thats way too long |
| 19:11.56 | glak | [TK]D-Fender: not that I like to argue with a angry person (I work at IT support), but your questions are on "expert" level. Dovid made it on the first try. You cant expect everyone to know what a PSTN or whatever is. English is NOT my mother tounge. |
| 19:11.59 | Micho123 | I'm getting the following error when receieving a FAX on Asterisk...ERROR[11320]: chan_sip.c:12441 handle_response_invite: Got error on T.38 re-invite. Bad configuration. Peer needs to have T.38 disabled. |
| 19:12.06 | Micho123 | Anyone can help here please? |
| 19:12.08 | [TK]D-Fender | Dovid: I know... thats why i don't usually offer death threats. I offer LIFE threats. |
| 19:12.15 | Dovid | haha |
| 19:12.41 | loather | [TK]D-Fender: haha, here, have a baby! |
| 19:12.46 | coppice | Dovid: it is a curse. they are making bodies survive to 120, but the brains are still failing at the same age |
| 19:12.58 | [TK]D-Fender | glak: And jbot gave you the answer to that and I explained several times myself . the REGULAR TELEPHONE NETWORK. |
| 19:13.24 | glak | <PROTECTED> |
| 19:13.34 | Dovid | coppice: lets hop science fixes that real soon |
| 19:13.44 | Dovid | glak: I did ;) |
| 19:13.50 | Dovid | it's called doing ur home work |
| 19:14.09 | glak | Dovid: English beeing your mother tounge? |
| 19:14.17 | coppice | glak: yeah, everyone understands that, so all the ITU specs call it the GSTN to reintroduce confusion :-) |
| 19:14.19 | Dovid | ask TK: I am still clueless some times. but i keep learning |
| 19:14.29 | *** part/#asterisk Mog (n=mog@c-68-62-218-186.hsd1.al.comcast.net) |
| 19:14.42 | *** join/#asterisk muiro (n=muiro@cpe-173-89-177-15.neo.res.rr.com) |
| 19:14.49 | [TK]D-Fender | [14:03]<[TK]D-Fender>glak: PSTN = NORMAL ^%$#ING PHONE NETWORK <- |
| 19:14.49 | glak | Dovid: seems like its illegal to learn here. Either you know, or get the hell out of here |
| 19:14.50 | [TK]D-Fender | [14:06]<glak>hm... I'll just call and bother my professor :p He made it sound so easy |
| 19:14.51 | Dovid | glak: English is my main language, how ever PSTN is not something that most people know. its something that you learn like the rest of us did |
| 19:14.54 | jaytee | everyone has a mother tongue but no one seems to have a father tongue |
| 19:15.12 | [TK]D-Fender | glak: What is hard about RegULAR PHONE NETWORK? Normal phones <- like you're neighbour. |
| 19:15.15 | Dovid | glak: Not at all. I have gotten my fair share of beatings here but you have to ask smartly and listen |
| 19:15.31 | coppice | jaytee: we do in our house. mother tongue is cantonese. father tongue is english |
| 19:15.33 | [TK]D-Fender | glak: But again, you have no need of Asterisk that I can see. |
| 19:15.33 | jaytee | my mother tongue is English, my father tongue is *$%&*$ $#*(#@ Damn #*(#~!~~~ |
| 19:15.59 | Dovid | glak: TK explained it to you several times. don't worry hang out here long enough and you will learn |
| 19:15.59 | glak | [TK]D-Fender: yes, but a "normal" phone network can mean supplier og the fysical copper lying around |
| 19:16.00 | jaytee | I'd say asterisk is a bit over his level, doncha think? |
| 19:16.20 | glak | og = or |
| 19:16.30 | muiro | is there a channel variable that holds the context name from sip.conf/iax.conf that originated the channel? |
| 19:16.30 | jaytee | fysical=physical |
| 19:16.35 | Dovid | jaytee: 6 years ago I was clueless. I am still finding my way.... |
| 19:16.39 | [TK]D-Fender | glak: normal phone network menas the ability to call people who use it. like BILLIONS DO RIGHT NOW |
| 19:16.58 | [TK]D-Fender | jaytee: I'm thinking rotay-dial phones are too far out right now... |
| 19:17.00 | jaytee | Dovid, Colonel Mustard, drawing room, candlestick! Get on it! |
| 19:17.03 | mosty | muiro, i don't think so, but you could set a channel variable in the incoming context yourself |
| 19:17.07 | Dovid | muiro: I don't think there is such a variable. you may want to set it in the begining |
| 19:17.19 | Dovid | lol |
| 19:17.19 | muiro | mosty, Dovid: thanks |
| 19:18.02 | glak | jaytee: I can read, my initial question entering this channel, was: Do I need to read ALL the 600 pages, or can I shorten it a bit? |
| 19:18.06 | Dovid | muiro: when I need to do that i do something like start_context=${CONTEXT} and use it all over |
| 19:18.40 | Dovid | glak: of the book ? you can do what ever you want. if you want to know asterisk well read the entire book and them some |
| 19:18.42 | [TK]D-Fender | glak: You do not need asterisk. At all. Period. |
| 19:19.06 | jaytee | glak, I never said you couldn't read. Show me where I said that! I never did. I just implied subtly that you couldn't SPELL |
| 19:19.09 | [TK]D-Fender | glak: you want to place calls tot he PSTN via an ITSP. You can install a softphone and read NOTHING and get it working. |
| 19:19.27 | muiro | Dovid: I want to use it to pass along to VoiceMailMain() to shoot the user directly into their own vm box. easy. |
| 19:19.53 | Dovid | muro: I personally use agi's with mysql. |
| 19:20.07 | Dovid | so i can have one context and look up by CID, or code etc. |
| 19:20.19 | Dovid | but yes you can set it in the incoming context and then pass it along |
| 19:20.21 | Dovid | e.g. |
| 19:20.32 | glak | jaytee: I didt say you said I couldt read, my "I can read" was towards the "I can read the manual" |
| 19:20.32 | jaytee | say what you will about proprietary, closed source software but at least the price tag keeps the room temperature I.Q wannabe techs out of the mix. Open source being free just seems to let in all the Forrest Gump Goes Becomes A VOIP Engineer wannabes. |
| 19:20.45 | Dovid | Exten => s,1,Set(start_context=${CONTEXT}) |
| 19:20.50 | [TK]D-Fender | muiro>is there a channel variable that holds the context name from sip.conf/iax.conf that originated the channel? <- "SetVar=ThePeer=thenameofthispeerentry" |
| 19:20.59 | [TK]D-Fender | Dovid: NO. |
| 19:21.02 | muiro | [TK]D-Fender: thanks, that's what I'm doing |
| 19:21.07 | Dovid | TK: why not ? |
| 19:21.15 | glak | [TK]D-Fender: I know that, but the ITSP is still going to charge me big time for calling (not so cheap here) |
| 19:21.23 | jaytee | glak, then go read the manual and have fun! Chapter 5 is a must! Chapters 6 through 8 are full of juicy goodness. |
| 19:21.26 | [TK]D-Fender | muiro: Keep doing it. Just about the only other option is parsing the channel name, and thats ugly and simply not worth it |
| 19:21.27 | Dovid | glak: Nothing in this world is free ;) |
| 19:21.36 | [TK]D-Fender | glak: Asterisk does not make it FREE! |
| 19:21.48 | [TK]D-Fender | glak: glak You have no need of * in this |
| 19:22.04 | Dovid | TK: What do you like about what I posted ? |
| 19:22.18 | glak | still I cant figure out what * is... Ive tried alot tho |
| 19:22.38 | muiro | [TK]D-Fender: that was the other option I was going to look at. I could use ${EXTEN:X:X} for it because I know where and how long the channel names will be, but this is much easier |
| 19:22.38 | Dovid | glak: it's a PBX that you can do many things with |
| 19:22.46 | Dovid | glak: What is your mother tounge ? |
| 19:22.57 | glak | Dovid: Norwegian |
| 19:23.19 | [TK]D-Fender | glax : Asterisk can be compared to a telephone PBX like Seimens, Nortel Norstar, etc. |
| 19:23.33 | [TK]D-Fender | glak: It is not a "service" |
| 19:23.34 | *** join/#asterisk _pepo_ (n=_pepo_@200.55.224.148) |
| 19:23.40 | _pepo_ | hi friends |
| 19:23.47 | [TK]D-Fender | glak: Picture it like a stupid answering machine. |
| 19:24.01 | glak | I understood it as a phone central (direct translation from norwegian) |
| 19:24.03 | [TK]D-Fender | glak: It doesn't give you free telephone calls. That is something else you have to find. |
| 19:24.21 | muiro | good, that worked perfectly. Thanks Dovid, mosty, [TK]D-Fender |
| 19:24.36 | [TK]D-Fender | glak: You do not need anything else between your phone and your ability to place calls. |
| 19:25.47 | glak | [TK]D-Fender: allright |
| 19:26.12 | Dovid | glak: http://en.wikipedia.org/wiki/Asterisk_PBX |
| 19:26.18 | Dovid | description on asterisk |
| 19:26.23 | *** join/#asterisk lanning (n=lanning@173.8.187.197) |
| 19:26.32 | [TK]D-Fender | glak: You talk into the softphone (via your computer), it uses an ITSP to reach the PSTN. the end. |
| 19:26.38 | Dovid | not sure if they have Norwegian on there |
| 19:26.49 | [TK]D-Fender | glak: you plug 3-4 values into the softphone to configure it. No books. |
| 19:27.13 | glak | [TK]D-Fender: I know, read about 5 different solutions |
| 19:27.23 | [TK]D-Fender | glak: Solutions for what? |
| 19:27.29 | glak | softphones |
| 19:27.40 | glak | err, 5 different softphones then |
| 19:27.47 | [TK]D-Fender | glak: Take your pick, to you they'll all be almost the same thing. |
| 19:28.11 | [TK]D-Fender | glak: Or you can get a dedicated VoIP-capable phone or converter box |
| 19:28.23 | glak | ATA-box? |
| 19:28.45 | Dovid | ~ata |
| 19:28.45 | jbot | well, ata is Analogue Terminal Adapter which provides an FXS and/or FXO and ethernet, see http://www.voip-info.org/wiki/view/ATA |
| 19:29.12 | [TK]D-Fender | glak: Yes |
| 19:29.33 | bmoraca | what would be the total power draw of a Sangoma A200 w/ 4 FXS ports? |
| 19:29.34 | glak | kk, can get one with a number for 0,- a month, but cost for usage |
| 19:29.54 | bmoraca | nm, found it |
| 19:30.47 | Dovid | glak: depends on the ITSP that you use.... |
| 19:32.17 | glak | I'll try a softphone at first, and take it from there |
| 19:32.39 | *** join/#asterisk killown (n=ukendt@unaffiliated/killown) |
| 19:33.33 | *** join/#asterisk n3hxs (n=HAMming@adsl-70-128-62-214.dsl.ltrkar.swbell.net) |
| 19:34.47 | *** join/#asterisk tamiel (n=tamiel@ip-176.net-89-2-117.rev.numericable.fr) |
| 19:35.06 | *** join/#asterisk etfonhomey (n=chatzill@74-143-192-75.static.insightbb.com) |
| 19:35.22 | *** join/#asterisk stevetotaro (n=Steve@c-71-206-29-240.hsd1.md.comcast.net) |
| 19:36.45 | etfonhomey | [TK]D-Fender, when provisioning Polycom's, I've only ever used tftp. So, I've always set DHCP option 66 to the IP address of my tftp server as just the IP (NOT a URL). What's the correct way to provision via FTP via DHCP Option 66? |
| 19:37.36 | etfonhomey | [TK]D-Fender, can I set DHCP option 66 to ftp://username@password:myftpserverip ? |
| 19:39.23 | *** join/#asterisk iEatChildren (n=WaffleMu@asa.redglaze.com) |
| 19:40.13 | iEatChildren | this might be a dumb question but here it goes. im looking in the cdr-csv log and i noticed it has entries such as "2009-03-05 19:10:59". .... why is the time so far off in the log? |
| 19:40.19 | iEatChildren | or am i just reading something wrong here? |
| 19:40.40 | iEatChildren | its about 7.5 hours off |
| 19:40.53 | doolph | any good/reliable/cheap sip provider? |
| 19:41.54 | Dovid | iEatChildren: check to make sure that the time on the server is correct |
| 19:42.01 | iEatChildren | it is correct |
| 19:42.04 | Dovid | doolph: Depends to where |
| 19:42.12 | iEatChildren | we have the gui as well, and it shows the correct time when i log in |
| 19:42.24 | Dovid | iEatChildren: did u change it after you started asterisk ? |
| 19:42.33 | iEatChildren | nope |
| 19:42.59 | Dovid | iEatChildren: try restarting asterisk and see what happens |
| 19:43.35 | iEatChildren | cant restart right now, ill try it tonight and check back |
| 19:44.29 | thehar | oh i LOVE children |
| 19:44.32 | thehar | nom nom |
| 19:47.01 | eppigy | hi |
| 19:47.06 | bmoraca | iEatChildren: is your server set to use UTC time, perhaps? |
| 19:48.53 | *** join/#asterisk j_kroon (n=jkroon@dsl-240-151-156.telkomadsl.co.za) |
| 19:50.29 | j_kroon | hi all, when using time specs, is it possible to do things like mon|wed|fri ? |
| 19:50.38 | *** join/#asterisk stevetotaro (n=Steve@c-71-206-29-240.hsd1.md.comcast.net) |
| 19:52.20 | Shaun222 | iEatChildren: if your gui shows the right time and the logs dont, my guess is that the logs are doing UTC and your log viewer is taking your time zone into account and converting. |
| 19:52.47 | *** join/#asterisk datafirm (n=dp@74.94.79.145) |
| 19:52.49 | datafirm | hello |
| 19:53.33 | datafirm | Im trying to get my vm to get emailed to us, but I think the mailboxcmd is not correct. Its set to sendmail local path with a " -t" but there are not errors and no deliveries. |
| 19:53.39 | datafirm | any more debugging I can try? |
| 19:54.41 | tzafrir_laptop | datafirm, normally you should not change mailboxcmd |
| 19:54.56 | tzafrir_laptop | Unless you have a very non-standard substitute of sendmail |
| 19:55.09 | tzafrir_laptop | What sendmail program do you use? |
| 19:55.10 | datafirm | tzafrir_laptop: mailcmd i meant |
| 19:55.14 | datafirm | postfix |
| 19:55.32 | tzafrir_laptop | Should just work. Can you send a mail from the command line? |
| 19:55.39 | datafirm | yes. |
| 19:55.47 | datafirm | but if I dont set it, i get errors. |
| 19:55.52 | datafirm | its trying an incorrect path |
| 19:55.57 | *** join/#asterisk orkid_ (n=orkid@unaffiliated/orkid) |
| 19:56.17 | *** part/#asterisk muiro (n=muiro@cpe-173-89-177-15.neo.res.rr.com) |
| 19:56.34 | datafirm | I have it set like so: mailcmd=/opt/local/bin/sendmail -t |
| 19:57.41 | *** join/#asterisk macros73 (n=cs@dsl093-063-232.pit1.dsl.speakeasy.net) |
| 20:00.51 | datafirm | fixed it.. my new path was incorrect. :-/ |
| 20:01.14 | *** part/#asterisk mcargile (n=mikec@rrcs-24-173-156-170.se.biz.rr.com) |
| 20:03.35 | *** join/#asterisk killown (n=ukendt@unaffiliated/killown) |
| 20:04.51 | path_ | branch 1.6.1 is considered stable ? |
| 20:07.36 | *** join/#asterisk eppigy (n=Dave@plasticlobster.com) |
| 20:07.42 | eppigy | THROW YOUR HOOD UP |
| 20:08.37 | *** join/#asterisk nima0102 (n=nima@91.98.208.7) |
| 20:09.23 | haps | path_: dude, 1.6.0.6 is stable, check out the website |
| 20:09.56 | path_ | haps, I'm looking at the svn |
| 20:10.32 | *** join/#asterisk stevetotaro (n=Steve@c-71-206-29-240.hsd1.md.comcast.net) |
| 20:10.59 | path_ | never mind I got the right url |
| 20:11.05 | haps | path_: why would svn be stable? latest release is 1.6.1-rc1 |
| 20:11.26 | path_ | thats what Im asking |
| 20:11.38 | path_ | if 1.6.1 is considered stable or not |
| 20:12.17 | *** join/#asterisk killown (n=ukendt@unaffiliated/killown) |
| 20:12.40 | haps | is stability important? |
| 20:14.05 | haps | because if it is you should be running a stable release, ie 1.6.0.6 |
| 20:14.15 | eppigy | some women seem to think so |
| 20:14.38 | haps | i think what you wanted to ask was 'is 1.6.1 svn likely to crash' and the answer is no, but there's no guarantee |
| 20:14.51 | haps | if you grab it after a bad patch is submitted it could go haywire |
| 20:15.06 | haps | the answer is, svn head is not stable. |
| 20:15.10 | haps | by definition |
| 20:15.23 | haps | stability implies well tested which it can't be |
| 20:15.45 | haps | rc1 should be pretty stable, since it's called 'rc1' so they're pretty confident it's got what it take |
| 20:15.48 | haps | s |
| 20:15.54 | haps | but obviously they are still working shitz out |
| 20:16.01 | haps | otherwise it would be a release, not an rc |
| 20:16.06 | *** join/#asterisk muiro (n=muiro@cpe-173-89-177-15.neo.res.rr.com) |
| 20:16.32 | muiro | do sip and iax send registration messages clear text only or can I make it a little bit more secure? |
| 20:17.38 | jplank | the register is clear text, I think the PW is hashed though |
| 20:17.49 | muiro | if the PW is hashed that's good enough |
| 20:19.38 | jplank | I'm pretty sure its hashed, I don't have a sip capture to back that up though. we usually authenticate by IP |
| 20:20.08 | haps | you can store md5 of the secret too |
| 20:20.13 | haps | http://www.voip-info.org/wiki/view/Asterisk+sip+md5secret |
| 20:20.32 | jaytee | haps> because if it is you should be running a stable release, ie 1.6.0.6 wow, I didn't even know Microsoft had a 1.6.0.6 release of Internet Explorer |
| 20:20.50 | jplank | haps: thanks for pointing that out, I've never seen that option |
| 20:21.18 | haps | jaytee: what? |
| 20:21.36 | jaytee | haps, just being snarky and cute! |
| 20:21.54 | jplank | muiro: it seems you have to specify you are using a hash with IAX |
| 20:21.57 | haps | dude, everyone knows that stable MSIE release is 1.666 |
| 20:22.05 | jplank | auth=md5 |
| 20:22.11 | muiro | jplank: okay, thanks |
| 20:22.27 | haps | jplank: isn't there a way to use AES as well? |
| 20:22.29 | jplank | then you have to put the md5 hashed secret instead of plain text |
| 20:22.35 | *** join/#asterisk theHub (n=theHub@69.177.93.21) |
| 20:22.49 | *** join/#asterisk DrAk0 (n=luisjose@nelug/coreteam/luisjose) |
| 20:23.25 | jplank | a quick look at http://www.voip-info.org/wiki/view/Asterisk+IAX+authentication shows only RSA and MD5, I don't know about AES |
| 20:23.58 | haps | hmm, just thought i'd seeen it somewhere |
| 20:24.07 | haps | very new to this |
| 20:24.37 | jplank | haps then you'll be sad to find out SIP by default is a very insecure protocol |
| 20:24.54 | jplank | its based off of HTTP which is completely plaintext |
| 20:25.26 | muiro | <3 iax |
| 20:25.52 | jplank | oh wait, forget what I said haps, muiro was asking about it :P |
| 20:26.17 | jplank | is there TCP support for SIP in 1.6? |
| 20:26.26 | jplank | maybe TCP w/ SSL support? |
| 20:26.48 | muiro | I read on teh wiki that no, not yet |
| 20:27.06 | jplank | i just did too, but that seemed to be from 2006 |
| 20:27.11 | jplank | unless your reading something else |
| 20:27.22 | muiro | oh, yeah, I didn't check the date |
| 20:27.32 | muiro | why did I associated it with 1.6... |
| 20:27.36 | muiro | who knows |
| 20:28.21 | KavanS | can someone point me to a good "multi-ring" macro in which a caller dials in....and after a certain time it goes to voicemail |
| 20:28.25 | KavanS | or is it called followme? |
| 20:29.37 | jplank | exten => s,1,Dial(sip/user1&sip/user2) |
| 20:29.55 | jplank | exten => s,2,Voicemail(sip/user1) |
| 20:30.01 | KavanS | ahh damn, that easy? shit |
| 20:30.11 | KavanS | thanks jplank |
| 20:30.15 | KavanS | I'll try 'er out :) |
| 20:30.25 | jplank | you can't copy and past that, but thats pretty much the just of it |
| 20:31.02 | jplank | you have to remember to set the timeout in dial (if the top of my head I think its Dial(sip/user1&sip/user2,,120) for 120 seconds |
| 20:31.22 | KavanS | yeah I'm tired of being out of hte office and missing calls |
| 20:31.38 | KavanS | so I figure it's time to get some multi-ring going |
| 20:31.43 | jplank | yes, but you'll get real tired of being on the phone, and your cellphone ringing |
| 20:31.46 | KavanS | is that the term in asterisk? |
| 20:31.53 | KavanS | haha yeah....well I'll just turn cell phone off |
| 20:32.10 | jplank | then all your calls will goto your cellphones VM |
| 20:32.28 | jplank | :) |
| 20:32.34 | KavanS | yeah I tried this "follow me" feature awhile back with asterisk-gui |
| 20:32.37 | *** join/#asterisk killown (n=ukendt@unaffiliated/killown) |
| 20:32.40 | KavanS | that you had to press 1 to accept call |
| 20:32.43 | jplank | follow me should work |
| 20:32.56 | jplank | yea, I always do a press 1 to accept thing |
| 20:33.16 | KavanS | yeah I'd prefer to utilize something like that, so I can easily forward to voicemail in case I'm just "laying low" for the da |
| 20:33.18 | KavanS | *day |
| 20:33.33 | muiro | in 1.6 the noanswer timeout is the first option after the tech/resource |
| 20:33.42 | jplank | is it? |
| 20:34.04 | jplank | oh |
| 20:34.05 | jplank | heh |
| 20:34.07 | jplank | 1.4 too |
| 20:34.27 | j_kroon | ok, can somebody explain this one (asterisk 1.6.0.3): 0 active channels, 1 active call. |
| 20:34.37 | jplank | like I said, don't copy and past what I put in |
| 20:36.41 | muiro | j_kroon: core show channels verbose |
| 20:37.13 | j_kroon | muiro, nothing listed. |
| 20:37.36 | j_kroon | there is just more column headings, but still no listed channels :) |
| 20:38.18 | j_kroon | raced counter perhaps? |
| 20:38.23 | muiro | maybe |
| 20:38.33 | j_kroon | not going to break his head. |
| 20:38.39 | scooby2 | weird i can seem to find a ftp module for everything except for Net::FTP |
| 20:38.40 | muiro | it should clear up on it's own |
| 20:38.48 | j_kroon | so asterisk has a bug ... whoopdi doo. |
| 20:39.05 | muiro | go check in the bug repo for it |
| 20:39.56 | muiro | would anyone here like me to write up a doc on how I got multiple asterisk instances running on one server/OS? |
| 20:40.03 | scooby2 | ack |
| 20:41.08 | jplank | why would you do that? |
| 20:41.31 | Chesther | I could think of a handful of uses, though none of them production. |
| 20:41.36 | muiro | multiple clients |
| 20:41.46 | Chesther | I'm sandboxing enough it would be handy for me to run 1.4 and 1.6 in parallel. |
| 20:41.48 | jplank | you could do multiple clients with one asterisk install |
| 20:41.56 | j_kroon | muiro, yes please. |
| 20:42.04 | jplank | if its not production, then why not use VM? |
| 20:42.07 | Chesther | Not to mention the level of isolation needed for multiple instances would be a good thing in general. |
| 20:42.20 | muiro | vm has a higher footprint than running multiple instances of asterisk |
| 20:42.29 | j_kroon | jpeeler, many reasons. The most important one to me separation of concerns for multiple clients. |
| 20:42.31 | Chesther | why should I manage multiple OS instances just to be able to run two daemons? |
| 20:42.53 | j_kroon | *jplank sorry |
| 20:43.17 | muiro | and running one instance per box is a drastic misuse of resources |
| 20:43.32 | jplank | explain that? |
| 20:43.35 | muiro | especially when you're talking about clients that might have 10-15 concurrent calls |
| 20:43.46 | j_kroon | jplank, let's say i'm hosting pbx setups for multiple clients. |
| 20:44.02 | jplank | ok |
| 20:44.10 | j_kroon | one client messing around and "breaking" their server shouldn't affect other clients. |
| 20:44.23 | muiro | also, we build our asterisk setup into our webapps. users and queues set up within our app |
| 20:44.38 | jplank | clients shouldn't have access to the PBX if they are hosted, thats the whole point of hosted |
| 20:44.40 | j_kroon | and they absolutely MUST NOT be able to use each other's contexts ... |
| 20:44.57 | muiro | to have all of our clients pile all of their setup into one databes for realtime config, making us need to frame out of our app to a shared database... messy |
| 20:45.06 | j_kroon | jplank, fair enough, but shit does go wrong, and you want to contain the damage when it does. |
| 20:45.12 | jplank | i get that muiro |
| 20:45.34 | jplank | I dont know, we've been running asterisk as a hosted platform for about 2 years now |
| 20:45.36 | j_kroon | muiro, i'd be very interested in that. |
| 20:45.52 | *** join/#asterisk harry_v (n=pcsuppor@S0106001d7e52cc78.vs.shawcable.net) |
| 20:45.54 | jplank | never had the need to run multiple instances of asterisk |
| 20:45.55 | muiro | j_kroon: stay tuned. I'll have it up in a day or two. work permitting |
| 20:45.59 | muiro | it's easier tahn you think |
| 20:45.59 | j_kroon | it would enable me to re-use a lot of work that i've already done. |
| 20:46.19 | j_kroon | muiro, i don't get in this # too often ... do you maybe have a url I could monitor? |
| 20:46.29 | muiro | ah sure, hold on |
| 20:46.32 | jplank | what I'd like to see is two separate asterisk installs with a heartbeat between them |
| 20:46.50 | muiro | jplank: that's up for R&D early next week, heh |
| 20:46.53 | j_kroon | jplank, that one is easier than you think :) |
| 20:47.00 | j_kroon | except for astdb ... |
| 20:47.01 | jplank | j_kroon: so I've been told |
| 20:47.10 | jplank | but no one wants to let the cat out of the bag |
| 20:47.25 | muiro | j_kroon: I'll put it on http://www.muiro.org |
| 20:47.26 | harry_v | Anyone care to explain why at times something like dial tone or vmmain would delay in responding when given a extention or number to dial? |
| 20:47.27 | jplank | I don't just mean two servers just replicating conf files |
| 20:47.44 | [TK]D-Fender | etfonhomey: I think I saw something like that one.... |
| 20:47.46 | [TK]D-Fender | once* |
| 20:47.50 | muiro | jplank: when I figure it out, I'll tell you how |
| 20:47.53 | j_kroon | jplank, what does your config look like? how much of it is really dependent on "dynamic" stuff? |
| 20:47.55 | jplank | I'm talking a real hearbeat between the two |
| 20:48.06 | j_kroon | so heartbeat detects one server going down, it brings the other up. |
| 20:48.07 | harry_v | muiro, i have seen it hooked up like that. |
| 20:48.12 | jplank | yea |
| 20:48.16 | jplank | same IP's |
| 20:48.19 | jplank | same everything |
| 20:48.21 | j_kroon | and you're realtime data you can replicate using normal replication mechanism. |
| 20:48.34 | jplank | just replicating conf and vm and whatever is the easy part |
| 20:48.44 | j_kroon | heartbeat has a "unique" IP for each server, and then a shared IP. |
| 20:48.56 | jplank | j_kroon: exactly |
| 20:49.07 | j_kroon | so what more info do you need? |
| 20:49.28 | j_kroon | you know how to replicate the configs, heartbeat detects failure ... what more is there? |
| 20:49.38 | j_kroon | anyway, i'm off to bed. |
| 20:49.45 | jplank | heartbeat part |
| 20:49.45 | j_kroon | before 23:00 for the first time in weeks. |
| 20:49.53 | j_kroon | read the heartbeat howtos. |
| 20:49.55 | muiro | j_kroon: watch muiro.org for the doc |
| 20:50.13 | jplank | heh, I was using heartbeat as a term, not an application |
| 20:50.17 | jplank | but now I will |
| 20:50.25 | tAnkOSX | :) |
| 20:51.13 | harry_v | how about issues with slow dtmf responce as in my case? |
| 20:51.32 | *** join/#asterisk keulin (n=cray@bne75-5-82-231-224-34.fbx.proxad.net) |
| 20:51.32 | harry_v | its been a anoying issue for a long time. Comes and goes. |
| 20:51.55 | harry_v | Cannot use hearbeat in a case such as this. |
| 20:52.42 | jplank | harry thats a very general question |
| 20:52.57 | harry_v | yes it is. |
| 20:53.03 | jplank | are you saying you dial a call, and sometimes it takes longer then other for the call to complete? |
| 20:53.21 | harry_v | yes, some times 10 seconds before I hear voicemail main |
| 20:53.23 | jplank | are you saying while you are on a call, and you dial DTMF it sometimes takes a while to push |
| 20:53.33 | jplank | what does asterisk see? |
| 20:53.51 | muiro | harry_v: are you using a speech recognition engine? |
| 20:53.55 | harry_v | no |
| 20:54.00 | harry_v | standard install. |
| 20:54.35 | *** join/#asterisk medjr (n=medjr@41.224.150.128) |
| 20:54.56 | harry_v | But Honestly, it may be a hardware issue because this issue has not been resolved from one ver to another. |
| 20:56.36 | *** join/#asterisk killown (n=ukendt@unaffiliated/killown) |
| 20:57.20 | pwell | anything that does IVR for free with Asterisk? |
| 20:57.31 | muiro | pwell: extensions.conf |
| 20:57.40 | pwell | whats with will's Manager Light, etc |
| 20:58.35 | doolph | anyone using AsteriskNow with asterisk 1.6? |
| 20:59.08 | doolph | the freepbx report is not working |
| 21:04.26 | *** join/#asterisk mazpe (n=mazpe@c-71-196-32-43.hsd1.fl.comcast.net) |
| 21:04.26 | *** join/#asterisk CrazyTux (n=brandon@c-98-196-6-54.hsd1.tx.comcast.net) |
| 21:07.19 | stabler | doolph: asteriskwin32=fail |
| 21:10.05 | jaytee | asteriskwin64 rocks! |
| 21:10.08 | *** join/#asterisk killown (n=ukendt@unaffiliated/killown) |
| 21:10.28 | *** join/#asterisk bsdfreak (n=alex@ninja.paranode.net) |
| 21:10.31 | stabler | jaytee: sure.... |
| 21:11.49 | jaytee | stabler, runs awesome as long as you don't try doing 2 concurrent calls while running anything else on the server and unfortunately it does crash whenever Windows Update runs. |
| 21:12.06 | stabler | lol |
| 21:12.08 | muiro | lol |
| 21:12.19 | stabler | sounds like any other windows app |
| 21:12.36 | stabler | smothered in fail sauce |
| 21:12.38 | *** join/#asterisk Mog (n=mog@c-68-62-218-186.hsd1.al.comcast.net) |
| 21:12.38 | *** mode/#asterisk [+o Mog] by ChanServ |
| 21:12.52 | jaytee | I'd rather run Asterisk on Ubuntu on an old Mac SE with the little 8" mono screen |
| 21:13.51 | [TK]D-Fender | stabler: Why did you bring up AsteriskWin32? |
| 21:14.04 | [TK]D-Fender | stabler: No-one else did... |
| 21:20.39 | *** join/#asterisk murdock_ut (n=chatzill@mail.kimballequipment.com) |
| 21:21.13 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 21:21.24 | *** join/#asterisk stevetotaro (n=Steve@c-71-206-29-240.hsd1.md.comcast.net) |
| 21:23.38 | *** join/#asterisk [T]ank (n=ckwall@206.71.78.158) |
| 21:24.08 | mazpe | how can i make an extension available to all context? for example... 9999 to access voicemail? |
| 21:25.05 | muiro | mazpe: write a context containing that extension, then you can include that context in each other context by using include |
| 21:25.54 | muiro | include => voicemail |
| 21:26.32 | muiro | [voicemail] being the name of our voicemail context |
| 21:26.48 | mazpe | thats what i'm doing now.. was wondering if there was any other hidden trick :) |
| 21:27.03 | muiro | not as far as I know |
| 21:27.06 | [TK]D-Fender | mazpe: No, contexts is what the dialplan is about. Everything is in ther |
| 21:28.05 | [TK]D-Fender | Checkout time, heading home, BBIAB |
| 21:28.12 | [T]ank | I am trying to set up a "tie line" between two systems. But I am not sure of the specific settings I should use. If I am receiving the T1 as esf b8zs and NI2, What would my other side have to send it as? Would it be esf b8zs using NI2? |
| 21:30.59 | *** join/#asterisk killown (n=ukendt@unaffiliated/killown) |
| 21:32.11 | *** join/#asterisk Mw3 (n=mw3@ip59934bd1.rubicom.hu) |
| 21:35.52 | jaytee | [T]ank, one side has to match the other. If it's PRI though, the signalling has to be pri_net on one side and pri_cpe on the other end. |
| 21:36.11 | jaytee | and timing normally 99% of the time has to come from the NET side |
| 21:37.28 | [T]ank | jaytee: ok, thank you |
| 21:37.49 | [T]ank | pri_net being the sending side and pri_cpe on the receiving side, right? |
| 21:39.34 | jaytee | [T]ank, nope, it's bi-directional actually. The telco is usually the net side. I have two spans for my PRI and one is normally used for outbound traffic while the other is used for inbound but I can make calls out of either span if I choose to. |
| 21:39.52 | *** join/#asterisk takashi_85 (n=ahmed@41.196.80.104) |
| 21:40.29 | jaytee | [T]ank, and when I say the telco is usually the NET side that's 99.99999% of the time. |
| 21:44.09 | *** join/#asterisk takashi_85 (n=taka@41.196.80.104) |
| 21:44.13 | *** part/#asterisk takashi_85 (n=taka@41.196.80.104) |
| 21:44.43 | harry_v | hiya jaytee |
| 21:45.11 | jaytee | hiya harry_v |
| 21:45.34 | harry_v | seeing whats up ..or in this day and age whats down like the dow :) |
| 21:45.59 | jaytee | "Obama will save us!!!!"(tm) |
| 21:46.44 | harry_v | well, this is not his problem. This is a loong isssue with debt that has caught up with the country. |
| 21:46.56 | harry_v | And china buying our debt which has recently stopped. |
| 21:47.04 | jaytee | I'm thinking of just stopping my 403b contributions altogether. What's the point? everything I've put in over the last 5 years is gone. |
| 21:47.12 | *** part/#asterisk _pepo_ (n=_pepo_@200.55.224.148) |
| 21:47.40 | *** join/#asterisk docid (n=eris@whthyt253-26.northwestel.net) |
| 21:48.35 | docid | hrmm, cant seem to find the command to show what type of signaling my spans are currently running (i know what i set them to just trying to find a way to display if they are in fact reflecting that kind of signaling) |
| 21:48.46 | harry_v | bummer |
| 21:49.16 | jaytee | pri show span "span number" |
| 21:49.38 | docid | even if they arnt pri spans? (no pri available in the artic) |
| 21:49.57 | jaytee | that will show you the signalling, it should be pri_cpe unless you're a telco or connecting that end to another pbx in your facility. |
| 21:50.13 | jaytee | docid, ah not pri? but running Asterisk? |
| 21:50.13 | docid | umm, we cant get pri from the telco here |
| 21:50.34 | docid | ive had a few different nonb-pri lines hooked up here |
| 21:50.50 | docid | MF signaling, gotta use some odd settings.. |
| 21:51.17 | jaytee | look in either /etc/asterisk/zapata.conf or /etc/asterisk/chan_dahdi.conf if you've got it already setup, that's where your signalling would be set. |
| 21:51.29 | docid | damn...the mr clean is too intense in here..cleaning day in the office, its horrid |
| 21:51.55 | jaytee | I prefer being overwhelmed by Pine-Sol myself. |
| 21:51.56 | docid | yes, i know its set there, i want to make sure it actually took, so i can get past the signaliung option being the problem or what |
| 21:52.02 | docid | i HATE cleanersw |
| 21:52.10 | docid | im gunna go hide in the shop for a bit |
| 21:52.28 | jaytee | quittin time for me. be back later |
| 21:53.11 | *** join/#asterisk [TK]D-Fender (n=joe@64.235.218.194) |
| 21:53.20 | *** join/#asterisk takashi_85 (n=takashi_@41.196.80.104) |
| 21:57.42 | *** join/#asterisk BadHAL (n=nn@cpe-72-179-194-139.stx.res.rr.com) |
| 21:58.20 | *** part/#asterisk [T]ank (n=ckwall@206.71.78.158) |
| 21:59.50 | kb3ien | i shortened my 'secret' for a sip peer, but the the old longer string still validates. is this a know bug? (yes i did a restart). |
| 21:59.54 | bsdfreak | anyone having problems with asterlink.com in the last two days? |
| 22:00.18 | kb3ien | i wanted to stop the registration without touching the client. |
| 22:04.01 | *** join/#asterisk defsdoor (n=andy@defsdoor.gotadsl.co.uk) |
| 22:05.01 | *** join/#asterisk puzzled (n=patrick@puzzled.xs4all.nl) |
| 22:06.23 | *** join/#asterisk path_ (n=path_@pc-15-190-86-200.cm.vtr.net) |
| 22:07.59 | *** join/#asterisk killown (n=ukendt@unaffiliated/killown) |
| 22:08.22 | *** join/#asterisk djMax (n=chatzill@66.92.91.133) |
| 22:08.37 | djMax | is there a current "best" sugarCRM/asterisk integration layer? |
| 22:08.53 | *** join/#asterisk takashi_85 (n=takashi_@41.196.80.104) |
| 22:11.10 | t_corr | Hmm, I have this Perl script that implements a mutex lock for a shared voicemail box and I never thought of it, but the built in AstDB would work better. |
| 22:11.41 | *** join/#asterisk bisonit (n=randall@59.21.124.24.cm.sunflower.com) |
| 22:11.48 | bisonit | hello |
| 22:12.12 | mvanbaak | djMax: talk-to.outdare.com I think. Might have the url wrong |
| 22:12.41 | *** join/#asterisk sjobeck (n=sjobeck@64.122.41.37) |
| 22:13.17 | djMax | yeah, that's the non-free one, but was afraid that was the answer. :) thanks |
| 22:15.10 | bisonit | where would be a good place to ask configuration questions about the extensions.conf file |
| 22:15.15 | *** join/#asterisk double_cheesburg (n=chatzill@ip68-98-36-177.ph.ph.cox.net) |
| 22:15.53 | [TK]D-Fender | bisonit: Here |
| 22:16.42 | bisonit | sweet. I am a totoal neweb. I have a extension that I want to twist it up a little. If it is called I want it to play a recorded message if not answered and then forward the caller back to the main IVR |
| 22:17.05 | bisonit | I have looked at the examples.. but mine is full of variables and stuff. I am a little confused |
| 22:17.37 | [TK]D-Fender | bisonit: ... |
| 22:17.39 | [TK]D-Fender | ~pb |
| 22:17.40 | jbot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste, or , http://bin.cakephp.org/ |
| 22:17.41 | [TK]D-Fender | ^^^^ |
| 22:17.55 | [TK]D-Fender | bisnShow us what you've got and we'll see what needs adjusting. |
| 22:18.04 | bisonit | ok.. |
| 22:18.33 | bisonit | Iam going to install IRC on the server that has network access to the asterisk server so I can cut and paste from edit pad.. stby |
| 22:19.14 | [TK]D-Fender | bisonit: SSH works jsut fine for this |
| 22:24.35 | *** join/#asterisk stevetotaro (n=Steve@c-71-206-29-240.hsd1.md.comcast.net) |
| 22:25.25 | double_cheesburg | I just rebuilt asterisk and I get the error:: Unable to connect to remote asterisk (does /var/run/asterisk.ctl exist?) |
| 22:25.27 | double_cheesburg | No. It doesn't exist. Is rebuilding asterisk by using { [1] ./configure [2] make [3] make install } supposed to result in /var/run/asterisk.ctl existing ??? I just attempted to upgrade from asterisk 1.42 to 1.6. |
| 22:26.16 | double_cheesburg | Asterisk was originally installed using asteriskNOW |
| 22:26.35 | *** join/#asterisk bisonit (n=randall@59.21.124.24.cm.sunflower.com) |
| 22:26.42 | bisonit | ok.. I am back |
| 22:26.57 | *** join/#asterisk aksyn (n=aksyn@gw.na.nu) |
| 22:27.37 | bisonit | what was the link to post my extensions.conf file |
| 22:27.45 | [TK]D-Fender | double_cheesburg: No, that only installs *, it doesn't START IT |
| 22:28.18 | double_cheesburg | [TK] D-Fender :: So how do you install the asterisk.ctl file ?? |
| 22:28.29 | [TK]D-Fender | double_cheesburg: Just because you bought a new car and at was delivered off the back of a flat-bed truck doesn't mean the egine is running <- |
| 22:28.48 | [TK]D-Fender | double_cheesburg: It gets created when you START ASTERISK |
| 22:28.57 | double_cheesburg | Ok. Well I haven't seen this instruction in any of the tutorials I've read... |
| 22:29.07 | [TK]D-Fender | double_cheesburg: "asterisk -gvvvvvc" |
| 22:29.38 | double_cheesburg | Cool. I just ran that. What did it do? |
| 22:29.48 | [TK]D-Fender | double_cheesburg: Or to start it as a daemon (after confirming that a manual start succeeds) do "safe_asterisk &" |
| 22:29.48 | double_cheesburg | I'm at the CLI now |
| 22:30.16 | *** join/#asterisk voxter (n=voxter@76.77.95.2) |
| 22:30.31 | [TK]D-Fender | double_cheesburg: You have just started * manually and are stuck there for the moment. If everything is fine you can "stop now" and start it as a daemon after which you can use "-r" to connect to is afterwards |
| 22:30.38 | [TK]D-Fender | ~pb |
| 22:30.39 | jbot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste, or , http://bin.cakephp.org/ |
| 22:30.42 | [TK]D-Fender | bisonit: ^^^ |
| 22:32.40 | *** join/#asterisk takashi_85 (n=ahmed@41.196.80.104) |
| 22:32.57 | bisonit | ok.. I just posted it |
| 22:33.03 | [TK]D-Fender | LINK PLEASE |
| 22:33.18 | bisonit | It is still uploading |
| 22:33.26 | bisonit | ......... sigh |
| 22:33.30 | *** part/#asterisk mohsen-ece (n=ahmed@41.196.80.104) |
| 22:35.09 | bisonit | John bigalo at Digium did most of the hacking on this thing. |
| 22:35.22 | kb3ien | i have a phone that wants to send its RTP packets not to the astbox but to the carrier, what option prevents that? |
| 22:36.09 | [TK]D-Fender | kb3ien: "canreinvite=no" |
| 22:36.50 | bisonit | http://pastebin.com/d7f05c0f7 |
| 22:37.09 | bisonit | The extension is 599 and it is at a remote office across a VPN |
| 22:38.54 | *** join/#asterisk stevetotaro (n=Steve@c-71-206-29-240.hsd1.md.comcast.net) |
| 22:39.00 | [TK]D-Fender | bisonit: We can see that 599 just calls that fixed-function macro. All it is desinged to do is the stuff you don't want. So instead of calling that macro, go change the exte to dial the device, then play the message, then Goto wherever you want. |
| 22:39.03 | bisonit | I have recorded a wav file for it |
| 22:39.39 | [TK]D-Fender | bisonit: Star making your new exten from #159 |
| 22:39.42 | [TK]D-Fender | start* |
| 22:40.14 | bisonit | man please bare with me.. I am going to need some extra hand holding here. I started figuring this thing out 3 hours ago. |
| 22:40.29 | bisonit | read the hand book v2 |
| 22:40.35 | *** join/#asterisk tobias (n=tobias@user-0ce2hu8.cable.mindspring.com) |
| 22:41.03 | [TK]D-Fender | bisonit: Where "just started" is 100% literal and you have zero concept of * and this is somehow dropped in your lap? |
| 22:41.28 | stabler | ~sipnat |
| 22:41.29 | jbot | [~sipnat] Quick guide on configuring * + SIP behind NAT : http://www.aocomputing.net/?p=3 , otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
| 22:42.12 | bisonit | pretty much. I worked with a guy that took care of things and he is sort of out of the picture. |
| 22:42.21 | bisonit | baptism by fire. |
| 22:42.34 | *** join/#asterisk killown (n=ukendt@unaffiliated/killown) |
| 22:42.47 | voxter | any of you guys know much about t1 pri interface signals? I'm looking for acceptable dB signal strengths for longer cable runs... |
| 22:42.49 | bisonit | think of Taladaga nights.. "I am on fire" "I am on fire" |
| 22:43.18 | [TK]D-Fender | bisonit: This passes "hand holding" and enters the "do it for me realm". |
| 22:43.42 | bisonit | ok.. I understand |
| 22:44.20 | *** join/#asterisk nny_1 (n=Scott@64.203.237.47) |
| 22:44.21 | [TK]D-Fender | bisonit: And this dialplan is somewhat broken... |
| 22:45.00 | bisonit | why are the extensions listed 3 times? |
| 22:45.04 | nny_1 | quick question I am seeing strange Caller ID strings on calls hitting SIP/MYIP (for example: 90056009712130799780) |
| 22:45.05 | bisonit | part of the flow? |
| 22:45.29 | *** join/#asterisk docelmo (n=vircuser@pool-151-199-186-145.lyn.east.verizon.net) |
| 22:45.38 | nny_1 | i assume this is shennaigans (hey whats that place you like to eat?) but not sure how to check to make sure nothing bad (tm) has happened |
| 22:46.15 | bisonit | Do you guys have any good dialplans I can look at to learn from? |
| 22:46.23 | *** join/#asterisk Torrieri (n=Torrieri@nelug/crew/torrieri) |
| 22:46.23 | [TK]D-Fender | bisonit: http://pastebin.com/m570afc42 |
| 22:46.34 | double_cheesburg | Alright. I {[1] asterisk -gvvvvvc [2] stop now [3] asterisk -r} and still get the same error message. |
| 22:46.35 | double_cheesburg | When you say "start asterisk as a daemon you mean {[1] asterisk -r} correct ? |
| 22:46.37 | double_cheesburg | Also, how does one set asterisk to automatically run on startup? |
| 22:46.39 | harry_v | nny_1 ive seen it from time to time. like 00000000 on my cid. Somone who does not know how to settup a asterisk box or wants to be anonymous |
| 22:46.46 | [TK]D-Fender | bisonit: 3 silly little lines. Substiture your macro call with that, and rename for the appropriate message file |
| 22:46.57 | [TK]D-Fender | bisonit: I hope you picked an * compatible format. |
| 22:47.13 | [TK]D-Fender | double_cheesburg: NO |
| 22:47.32 | [TK]D-Fender | [17:29]<[TK]D-Fender>double_cheesburg: Or to start it as a daemon (after confirming that a manual start succeeds) do "safe_asterisk &" |
| 22:47.41 | double_cheesburg | Error => Unable to connect to remote asterisk (does /var/run/asterisk.ctl exist?) |
| 22:47.43 | [TK]D-Fender | double_cheesburg: Can I be any clearer? |
| 22:47.58 | nny_1 | harry_v yeah except this is coming in as a SIP call to asterisk, no auth |
| 22:48.00 | double_cheesburg | Ok |
| 22:48.35 | bisonit | why are the extensions listed 3 times in the extensions.conf file? |
| 22:49.02 | nny_1 | here is the asterisk log entry http://pastebin.com/ |
| 22:49.24 | double_cheesburg | That's really interesting. Yes. It works. But what does this do, essentially? |
| 22:50.04 | [TK]D-Fender | double_cheesburg: ... starts * as a daemon. A background process..... |
| 22:50.18 | [TK]D-Fender | bisonit: because this exten does 3 STEPS |
| 22:50.35 | [TK]D-Fender | bisonit: You lack the most important basics. Go read chapter 5 of THE BOOK |
| 22:50.36 | [TK]D-Fender | ~book |
| 22:50.37 | jbot | it has been said that book is probably Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://tfot.leifmadsen.com or see ~buybook |
| 22:51.12 | bisonit | thanks .. I will! I feel so helpless. Driving me nutz |
| 22:51.39 | *** join/#asterisk stevetotaro (n=Steve@c-71-206-29-240.hsd1.md.comcast.net) |
| 22:52.58 | double_cheesburg | [TK]D-Fender :: What steps do I need to take to set Asterisk for automatic startup ?? |
| 22:57.27 | double_cheesburg | Alright, alright I'm looking |
| 22:58.23 | *** join/#asterisk mortum (n=fjbf@TELEYECLA-89-29-236-3.teleyecla.net) |
| 22:59.28 | path_ | bisonit, first you need to get some basic concepts and then it will become easier |
| 22:59.52 | path_ | avoid any gui configurator thats my advice :-) |
| 23:02.06 | [TK]D-Fender | Time for martial arts, BBIAB |
| 23:04.19 | SwK | ok heres a stupid question... |
| 23:04.39 | SwK | I need to set a channel var in a macro thats available in the calling context |
| 23:05.03 | SwK | but I can see it set in the macro, and i can noop in the macro and see the value, but immediate after I exit the macro the value is gone... any hints? |
| 23:05.20 | *** join/#asterisk docid (n=eris@69.196.68.142) |
| 23:05.29 | Qwell | SwK: prefix setting it with a _? |
| 23:05.40 | SwK | you know I didnt try that |
| 23:05.53 | Qwell | also try __ for fun if it doesn't work with just _ |
| 23:05.57 | Qwell | (2 vs 1) |
| 23:05.59 | SwK | yeah |
| 23:06.08 | SwK | i dont have to screw with macros too much any more hah |
| 23:06.20 | SwK | i just write the stuff in AGI and no dialplan for me |
| 23:07.44 | *** join/#asterisk Sazenchan (n=sazencha@85-18-219-234.ip.fastwebnet.it) |
| 23:08.05 | *** join/#asterisk Bonix (n=Bonix@200-195-41-212.isimples.com.br) |
| 23:08.11 | SwK | Qwell, nope and nope |
| 23:08.23 | SwK | if it matters if a macro call from dial |
| 23:08.39 | SwK | ie: dial(sip/foo||M(somemacro)) |
| 23:11.21 | mazpe | what am i doing wrong here? http://rafb.net/p/rsV2JC16.html |
| 23:11.34 | mazpe | after the reponse time out it goes to status is unkonwn |
| 23:11.43 | mazpe | unknown on the CLI |
| 23:11.50 | *** join/#asterisk orkid_ (n=orkid@unaffiliated/orkid) |
| 23:11.55 | mazpe | -- Auto fallthrough, channel 'SIP/gbpbx_102-08319b38' status is 'UNKNOWN' |
| 23:12.25 | mazpe | it should be expecting 1 or 2 |
| 23:18.40 | *** join/#asterisk mohsen-ece (n=ahmed@41.196.80.104) |
| 23:18.54 | *** part/#asterisk mohsen-ece (n=ahmed@41.196.80.104) |
| 23:22.27 | bisonit | This stuff is wicked cool. Thanks [TK]D-Fender |
| 23:22.48 | harry_v | mazpe, disable autofall though. Google it and then disable it. |
| 23:27.28 | bisonit | . |
| 23:28.01 | *** join/#asterisk bob_slacker (n=bob_slac@189.6.67.94) |
| 23:35.58 | *** join/#asterisk MaliutaLap (n=biteme@203.171.192.125) |
| 23:40.12 | voxter | any of you with polycom ip601's ever experience an issue where when attempting to transfer a call, you get "SIP" displayed on the screen and are unable to dial/transfer? |
| 23:45.52 | *** join/#asterisk Maliuta (n=scooby@kiev.lusan.id.au) |
| 23:50.30 | double_cheesburg | asterisk.ctl does not build on system startup, but only when asterisk resarted. Anyone know why? |
| 23:56.55 | *** join/#asterisk norgs (n=ken@WoWUIDev/Norganna/Administrator/Norgs) |
| 23:57.02 | norgs | Greetings |
| 23:57.53 | norgs | I have a question regarding app-pickup which doesn't seem to be working for my custom ivr |
| 23:58.18 | norgs | Here are the relevant portions of my dialplan: http://norgs.pastey.net/109556 |
| 23:59.25 | bisonit | Well that GUI that comes with the Aserisk Digium business is crap |
| 23:59.43 | bisonit | you can't configure half the crap you need to |