| 00:01.21 | *** join/#asterisk keulin (n=cray@bne75-5-82-231-224-34.fbx.proxad.net) |
| 00:02.20 | tmiw_ | hi. so i installed spandsp and tried to add app_fax to the asterisk showconfig |
| 00:02.26 | tmiw_ | and I get this: |
| 00:02.27 | tmiw_ | app_fax.c:312: error: storage size of 'fax' isn't known |
| 00:02.35 | tmiw_ | is this a known issue? |
| 00:02.53 | drmessano | Which version of spandsp? |
| 00:03.06 | tmiw_ | 0.0.6pre2 |
| 00:03.12 | drmessano | You need 0.0.5 |
| 00:03.18 | drmessano | the last 0.0.5 |
| 00:03.28 | tmiw_ | ah, i'll try that |
| 00:03.29 | tmiw_ | one sec |
| 00:06.45 | tmiw_ | drmessano: cool, compiling is continuing now. thanks! |
| 00:09.24 | *** join/#asterisk RypPn (i=TuMbL@rosscom.demon.co.uk) |
| 00:20.09 | *** join/#asterisk jebmpls (n=user@75.146.149.21) |
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| 00:25.00 | *** part/#asterisk jebmpls (n=user@75.146.149.21) |
| 00:31.49 | protocols | [TK]D-Fender, but it does not recognize my changes |
| 00:32.12 | protocols | open("/etc/asterisk/asterisk.conf", O_RDONLY|O_LARGEFILE) = 4 |
| 00:32.18 | protocols | it reads my config |
| 00:33.43 | [TK]D-Fender | protocols: You aren't showing anything complete and useful |
| 00:34.06 | protocols | what would be useful? |
| 00:34.19 | [TK]D-Fender | protocols: I don't see your complete configs. Nor any folder dumpts. Nor CLI show evidence of not following your configs. |
| 00:34.23 | protocols | I showed, that via strace asterisk does not have any problems with opening the asterisk.conf |
| 00:34.46 | protocols | Unable to open pid file '/var/run/asterisk.pid': Permission denied |
| 00:35.06 | protocols | but: astrundir => /var/run/asterisk |
| 00:35.32 | protocols | ah ok my mistake |
| 00:35.48 | protocols | [directories] ; remove the (!) to enable this |
| 00:36.04 | protocols | damn those boogey traps |
| 00:36.27 | [TK]D-Fender | protocols: In my world we call that "the big print" |
| 00:36.58 | joat | vogon construction notice (you gotta look) |
| 00:36.58 | protocols | :D |
| 00:37.00 | Carlos_PHX | Holy shit, this could get ugly: http://www.earthtimes.org/articles/show/sprint-nextel-severs-its-internet-connection-to-cogent-communications,603138.shtml |
| 00:38.52 | *** join/#asterisk jksM (i=jks@193.189.93.254) |
| 00:41.37 | drmessano | God I hated sprint |
| 00:41.41 | *** join/#asterisk edoceo (n=edoceo@c-71-197-244-147.hsd1.or.comcast.net) |
| 00:42.21 | Carlos_PHX | Hated? |
| 00:42.26 | Carlos_PHX | You like them now? |
| 00:43.21 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 00:43.44 | *** join/#asterisk jer (n=jer@unaffiliated/jer) |
| 00:45.09 | *** join/#asterisk andrewy (i=andrewy@209.126.180.153) |
| 00:51.22 | drmessano | No, I dont have to deal with them anymore |
| 00:51.32 | Carlos_PHX | Ah, lucky you. |
| 00:51.48 | Carlos_PHX | I considered them, but they took three months to follow up with a quote. |
| 00:51.52 | Carlos_PHX | I think I got lucky. |
| 00:52.27 | drmessano | We used to have Sprint for our corporate MPLS |
| 00:52.35 | drmessano | Back at my old job |
| 01:12.37 | *** join/#asterisk Coolthreads (n=Coolthre@203-97-238-71.cable.telstraclear.net) |
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| 01:34.22 | *** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 01:43.26 | *** join/#asterisk jjg_ (n=jjg@76.21.4.40) |
| 01:43.42 | jjg_ | hi, anyone know of some simple command line based rtp tools? |
| 01:44.19 | drmessano | As in? |
| 02:17.44 | *** join/#asterisk heedly (n=heedly@purplehaze.lamedomain.net) |
| 02:18.07 | heedly | hi, anyone know where to get a VPS with access to a PRI |
| 02:25.52 | jjg | anyone used sipp to generate rtp traffic? |
| 02:29.20 | *** join/#asterisk reno139 (n=reno@68.51.17.119) |
| 02:31.04 | reno139 | hey all, i am having a problem getting g729 codec to work. removed the default g729 .so and replaced it with another, restarted, and it's showing - where it should show 6 or 8 i believe when i do a 'core show translation recalc 10', any suggestions? |
| 02:39.51 | *** join/#asterisk [gnubie] (n=bintut@cm61.omega118.maxonline.com.sg) |
| 02:42.47 | drmessano | Which one did you replace it with? |
| 02:43.16 | reno139 | codec_g729-ast14-gcc4-glibc-athlon-sse.so |
| 02:44.17 | drmessano | You realize that codec isn't legal, right? |
| 02:44.28 | drmessano | Unsupported to say the least |
| 02:44.31 | reno139 | i thought that was only is you used it commercially |
| 02:44.40 | drmessano | No |
| 02:44.51 | reno139 | well then, another option will be considered |
| 02:45.05 | drmessano | Testing only.. Non-production us |
| 02:45.06 | drmessano | Testing only.. Non-production use |
| 02:45.19 | drmessano | Get the licensed one from digium, its worth it |
| 02:45.40 | reno139 | you happen to know offhand how much? |
| 02:45.56 | drmessano | $10 per channel |
| 02:46.02 | reno139 | ah |
| 02:46.09 | reno139 | not so bad |
| 02:47.29 | *** join/#asterisk jicksta (n=jicksta@c-24-6-87-187.hsd1.ca.comcast.net) |
| 02:47.29 | reno139 | do you have a suggestion for a comparable coedc that is oss? this is just a home box that i'm on a rather strict budget, but may be making more that 4 calls at a time, and i'm looking at $80 at that point aren't i? |
| 02:47.58 | *** join/#asterisk tgrman (n=jcmoore@unaffiliated/tgrman) |
| 02:48.34 | drmessano | Depends |
| 02:48.40 | drmessano | If the endpoints are using G729, no |
| 02:48.44 | drmessano | Its only for transcoding |
| 02:49.02 | reno139 | gxp2000 and x-lite |
| 02:49.08 | drmessano | ok |
| 02:49.35 | *** part/#asterisk tgrman (n=jcmoore@unaffiliated/tgrman) |
| 02:50.04 | drmessano | If your ITSP is using G729 and you're using G729 on the devices you wont be transcoding |
| 02:50.17 | drmessano | If any leg of the call needs to be transcoded, thats 1 license |
| 02:50.30 | drmessano | So g729 <> ITSP w/g711 would be 1 |
| 02:50.32 | reno139 | they are |
| 02:51.06 | reno139 | ok, so if i'm end to end g729, no license required? |
| 02:51.23 | drmessano | Theres cases like voicemail where you would be transcoding |
| 02:51.32 | reno139 | i see |
| 02:51.46 | drmessano | So you may be able to get by with 2 licenses, or 3 |
| 02:52.09 | reno139 | yeah, because i seriously doubt that i would be getting more than one VM at a time |
| 02:52.13 | drmessano | Maybe less.. If you're always end to end G729, you may need that 1 for the occasional voicemail or inbound SIP call thats non-g729 |
| 02:52.15 | reno139 | or, more than 2 or 3 a day |
| 02:52.46 | reno139 | but, if my trunk is g729, then they would be handing that, correct? |
| 02:53.13 | drmessano | If your ITSP is using G729, and you are using G729, there is no transcoding involved |
| 02:53.38 | reno139 | and thanks, this is the most informative bit of info i've received so far! |
| 02:53.49 | drmessano | If it were me |
| 02:53.57 | drmessano | Having the volume of calls you expressed |
| 02:54.33 | drmessano | I would get 2.. It's a 20 spot.. |
| 02:54.57 | reno139 | gotcha |
| 02:54.57 | drmessano | If you're aggressive and max volume is 1, then you have 1 "just in case" |
| 02:56.20 | *** join/#asterisk tobias (n=tobias@user-0ce2hvs.cable.mindspring.com) |
| 02:56.43 | reno139 | thanks for the info! off to do some more reading. |
| 02:56.50 | drmessano | No probs |
| 02:59.39 | *** join/#asterisk jeev (n=email@unaffiliated/jeev) |
| 02:59.55 | jeev | shit |
| 03:10.27 | *** join/#asterisk shmaltz (n=chatzill@mail.dmaven.com) |
| 03:27.16 | *** join/#asterisk rcy (n=rcy@S01060002553240a8.vc.shawcable.net) |
| 03:30.13 | tmiw_ | neat, I got T.38 working :D |
| 03:39.18 | *** join/#asterisk [TK]D-Fender (n=chatzill@64.235.218.194) |
| 03:51.26 | jeev | pump it up |
| 03:51.29 | jeev | i saw W. HILARIOUS |
| 03:51.34 | jeev | i saw W., HILARIOUS |
| 03:51.35 | jeev | ! |
| 03:58.05 | *** join/#asterisk osiris (n=osiris@c-71-205-27-88.hsd1.mi.comcast.net) |
| 03:59.51 | drmessano | Affleck doing Olbermann = VERY WIN |
| 04:00.00 | drmessano | Funniest SNL skit I have seen in a LOOONG time |
| 04:00.00 | *** join/#asterisk freakazoid0223 (n=mattc@pool-68-238-187-201.phil.east.verizon.net) |
| 04:01.11 | *** join/#asterisk cesar_CR (n=cesar@celord.ice.co.cr) |
| 04:15.45 | AndyML | I'm having trouble with Dahdi - anyone figured it out that wants to help? |
| 04:16.53 | AndyML | I have a quad span digium card and I get this when I run dahdi_cfg -vvv |
| 04:16.54 | AndyML | line 0: Unable to open master device '/dev/dahdi/ctl' |
| 04:18.59 | *** part/#asterisk heedly (n=heedly@purplehaze.lamedomain.net) |
| 04:21.04 | *** join/#asterisk CunningPike (n=arodgers@S010600095b33697f.vc.shawcable.net) |
| 04:24.11 | *** join/#asterisk beek (n=klinebl@static-71-240-222-58.alt.east.verizon.net) |
| 04:26.47 | beek | Hello guys. I have a Sangoma A200d card -- when dialing out I frequently get issues with the DTMF being recognized by Verizon... so 50% of the time the call does not get placed. I'm using the latest dahdi/asterisk combo on Linux. I have not found anything to delay dialing once the line is connected. Other than changing parameters in kernel.h in dahdi-linux, do you have any suggestions? |
| 04:29.39 | [TK]D-Fender | beek: You mean DTMF before the call is fully started, or after its really in progress? |
| 04:30.27 | beek | I'm doing a Dial(DAHDI/G0/xxxx). Half the time I get a completed call, the other times I get |
| 04:30.45 | beek | the three-tones, then "you're call cannot be completed as dialed." |
| 04:31.15 | beek | There are four FXOs on this card and only one of the three lines is consistently good. |
| 04:31.22 | [TK]D-Fender | beek: Dial(DAHDI/G0/wwxxxx) |
| 04:31.38 | [TK]D-Fender | beek: "w"'s in front of the number add 1/2s ea |
| 04:31.47 | [TK]D-Fender | beek: that should do it |
| 04:31.51 | beek | That was what I was looking for! Thanks. |
| 04:32.51 | beek | [TK]D-Fender: Got the A104d installed... really easy to get going. Thanks again for that, as well. |
| 04:33.06 | [TK]D-Fender | bekkQuite welcome. |
| 04:33.11 | [TK]D-Fender | beek* |
| 04:35.30 | x86 | [TK]D-Fender: I thought the w's added 1/4 second each |
| 04:35.42 | beek | [TK]D-Fender: Whlie I have you... I also have a problem with callerid on these lines. One line is a private line and CallerID comes in every time. The other three are in a hunt group and verizon delays the FSK for CallerID and it frequently gets ignored. Anything I can do to delay the actual anwer? I have a wait(8) before the actual answer(); |
| 04:35.57 | *** join/#asterisk mindCrime (n=chatzill@cpe-075-177-141-190.nc.res.rr.com) |
| 04:36.06 | beek | s/anwer/answer/ |
| 04:36.52 | [TK]D-Fender | beek: You shouldn't have to wait at all... just asking * for CID should do it... and if they are fubar'd, then bitch to the telco. |
| 04:37.18 | beek | [TK]D-Fender: Will do. Thanks very much for all of your help -- have a good night. |
| 04:37.25 | *** join/#asterisk voxter (n=voxter@S01060016b6b53c0c.vc.shawcable.net) |
| 04:48.18 | *** join/#asterisk terracon (n=garry@CPE001cf097a750-CM0012254076d6.cpe.net.cable.rogers.com) |
| 04:50.24 | *** join/#asterisk jer (n=jer@unaffiliated/jer) |
| 04:53.52 | AndyML | [TK]D-Fender: wanna help me with a T1 config in dahdi? something just isn't right but I think i'm close... |
| 05:01.12 | angler | beek, Yea, Asterisk will have already looked for the CID before it even executes your dial plan so using a Wait won't have any affect |
| 05:01.28 | jeev | damn |
| 05:01.34 | jeev | angler, you new or am i not paying attention to my op list |
| 05:01.56 | beek | angler: Thanks... This SNAFU is a PITA. |
| 05:01.59 | angler | you must not have been paying attention to the op list for years :) |
| 05:02.32 | jeev | ahh ok cool |
| 05:02.41 | jeev | i was going to introduce myself as the guy who beats up russellb |
| 05:02.44 | jeev | but now you wont beliee me |
| 05:02.45 | jeev | believe |
| 05:03.12 | angler | -been around for 5.5 years :) |
| 05:03.20 | jeev | damn |
| 05:03.21 | angler | jeev, lol |
| 05:04.00 | angler | I don't get much time on IRC anymore |
| 05:04.21 | jeev | ahh |
| 05:04.26 | angler | workin late tonight :) |
| 05:04.38 | jeev | ah |
| 05:04.40 | jeev | is russell there? |
| 05:04.44 | angler | nope |
| 05:04.50 | angler | i think im the only one here |
| 05:04.55 | jeev | go leave him a gift on his desk |
| 05:04.58 | jeev | #2 |
| 05:05.02 | angler | lmfao! |
| 05:05.05 | jeev | if it's diarrhea |
| 05:05.07 | jeev | even better |
| 05:05.41 | angler | lol |
| 05:05.44 | jeev | lo |
| 05:05.44 | jeev | kl |
| 05:05.51 | angler | heck I didn't know he had a flag in his office |
| 05:06.29 | angler | wonders if file is still awake |
| 05:06.39 | file | O.O |
| 05:06.47 | angler | yay! |
| 05:06.55 | angler | file, what time is it there? |
| 05:07.09 | file | it went from 2AM to 1AM 7 minute sago |
| 05:07.38 | angler | ah. thats right the time moves back an hour tonight |
| 05:07.44 | file | indeed |
| 05:07.55 | angler | ill prolly forget and end up working even more tonight |
| 05:08.02 | file | but now I go to sleep |
| 05:08.11 | angler | :( |
| 05:08.17 | angler | tty monday then |
| 05:08.31 | *** part/#asterisk hadronzoo (n=user@ppp-70-247-170-199.dsl.rcsntx.swbell.net) |
| 05:08.44 | angler | can blast music since no one else is here. |
| 05:09.59 | jeev | or porn |
| 05:10.52 | jeev | bbiab |
| 05:10.54 | angler | hahaha |
| 05:11.09 | jeev | and dont blast john denver for christ sake |
| 05:11.26 | *** join/#asterisk russellb (n=russell@asterisk/digium-open-source-team-lead/russellb) |
| 05:11.26 | *** mode/#asterisk [+o russellb] by ChanServ |
| 05:17.21 | Carlos_PHX | Time change, what a weird concept. |
| 05:20.04 | angler | pokes russellb |
| 05:21.53 | russellb | falls over |
| 05:23.00 | *** join/#asterisk jicksta (n=jicksta@c-24-6-87-187.hsd1.ca.comcast.net) |
| 05:23.07 | angler | sweet |
| 05:23.35 | angler | russellb, you should join me at work |
| 05:24.59 | *** join/#asterisk jjshoe (i=jjshoe@cpe-76-175-157-237.socal.res.rr.com) |
| 05:28.26 | russellb | angler: heh, um ... i think i'll stay home |
| 05:28.32 | russellb | do you need something? or are you just bored :) |
| 05:28.42 | angler | bored :) |
| 05:28.59 | angler | well not really, working on some perl scripts and dial plan |
| 05:29.19 | angler | russellb, is marko in town? |
| 05:30.09 | russellb | cool ... i don't know. he was last night, at least |
| 05:30.46 | angler | you go to his place last night? I wanted to but ended up being really tired. |
| 05:31.03 | russellb | yeah, i did, it was fun |
| 05:31.13 | angler | lot of people? |
| 05:31.48 | russellb | hmm ... maybe 30 |
| 05:32.26 | angler | cool |
| 05:56.43 | beek | angler: I'm really beating my head against the wall over this. This is the error on the asterisk console: |
| 05:56.47 | beek | [Nov 2 01:55:45] ERROR[7030]: callerid.c:564 callerid_feed: No start bit found in fsk data. |
| 05:58.03 | angler | beek, I bet the telco uses a distinctive ring to differentiate the line it came in on since they are in a hunt group |
| 05:58.55 | beek | Yes, they do. |
| 05:59.14 | beek | So, is there a workaround for this? |
| 06:00.00 | russellb | not really, there is no way for asterisk to predict which ring type they are going to use |
| 06:00.12 | russellb | asterisk is just going to try to look for callerid after the first ring ... |
| 06:00.49 | *** join/#asterisk hadronzoo (n=user@ppp-70-247-170-199.dsl.rcsntx.swbell.net) |
| 06:01.17 | hadronzoo | Hello, is it possible to use jabber.conf to connect to multiple gtalk accounts concurrently? |
| 06:02.15 | russellb | yes |
| 06:02.31 | hadronzoo | russellb: how do I separate the accounts? Do I use separate [sections]? |
| 06:02.35 | beek | russellb: Thanks... |
| 06:02.39 | jeev | wow |
| 06:02.42 | jeev | i got my ass torn in red alert 3 |
| 06:03.01 | russellb | hadronzoo: correct |
| 06:03.10 | jeev | russellfer. |
| 06:03.13 | Carlos_PHX | I thought that game was for little girls? |
| 06:03.37 | jeev | no way |
| 06:03.53 | hadronzoo | russellb: each with its own context? |
| 06:04.22 | russellb | i think so |
| 06:04.31 | jeev | russelldorf |
| 06:04.46 | hadronzoo | russellb: thanks for your help |
| 06:05.10 | russellb | np |
| 06:05.56 | jeev | knocks on russellb's head |
| 06:05.58 | jeev | hello????? |
| 06:07.50 | *** join/#asterisk kamanashisroy (n=kamanash@119.30.34.8) |
| 06:10.37 | russellb | continues to ignore jeev |
| 06:11.00 | jeev | to TRY |
| 06:11.01 | jeev | to ignore |
| 06:11.54 | jeev | come on russellb. |
| 06:12.40 | *** join/#asterisk luckyaba (n=lucky@ip70-177-7-204.sb.sd.cox.net) |
| 06:16.27 | jeev | russeldorf |
| 06:16.28 | jeev | i'm hungry |
| 06:16.49 | *** join/#asterisk implicit (n=bayan@unaffiliated/implicit) |
| 06:24.43 | [gnubie] | is anyone on this channel tried this => http://www.voip-info.org/wiki/view/Asterisk+cmd+DTMFToText on Asterisk-1.4.x ? |
| 06:27.12 | jeev | russellb, feed me |
| 06:28.19 | drmessano | [gnubie]: Were you not here earlier when the AUTHOR of that app told you it was an ugly hack? |
| 06:28.52 | [gnubie] | drmessano: nope. sorry. |
| 06:29.19 | drmessano | [gnubie]: Actually, you were... and ignored it.. like the rest of the waste of time I spent typing earlier lol |
| 06:29.42 | [gnubie] | drmessano: i don't know. may i know his nick on this channel? |
| 06:29.50 | drmessano | Anyway.. Even coppice said it was thrown together and didn't have much faith in it |
| 06:29.55 | drmessano | and he WROTE IT |
| 06:30.08 | drmessano | Not sure if that tells you something |
| 06:30.22 | hadronzoo | does anyone know how to associate a particular jabber connection with a context? |
| 06:30.32 | [gnubie] | drmessano: i don't know. |
| 06:30.54 | drmessano | What dont you know? |
| 06:31.24 | [gnubie] | drmessano: i just found out that the callweaver project is using it and it is updated since last month |
| 06:32.34 | drmessano | ok |
| 06:32.38 | drmessano | Well |
| 06:32.54 | [gnubie] | drmessano: that the author wrote somewhere that he didn't have faith in what he wrote. i actually was trying to find the reason why it was gone |
| 06:33.14 | drmessano | You should have asked him earlier |
| 06:33.42 | [gnubie] | drmessano: honestly, i didn't noticed him and i don't know his nick |
| 06:33.46 | drmessano | "It works, but it's a poor first resort" were almost his exact words |
| 06:35.20 | drmessano | ok, [gnubie].. Word of advice.. if you're gonna ask for input or help with something you want to actually READ what is said BACK to you.. Not only did you waste my time, but you wasted the time of the guy who wrote it who was talking to you about it.. or so he thought |
| 06:36.51 | [gnubie] | drmessano: i am.. but earlier when i was online, i was researching about that app_dtmftotext and why it was gone.. maybe i didn't noticed it when i was online.. |
| 06:37.34 | [gnubie] | drmessano: and i was also looking for alternative way of doing similar solution to my need |
| 06:37.35 | drmessano | [gnubie]: You were researching it and the guy who wrote it was was trying to talk to you about it.. Does this make any sense to you? |
| 06:38.00 | [gnubie] | drmessano: as i've said earlier, i don't know his nick and sorry about that |
| 06:40.40 | [gnubie] | drmessano: ok, i just checked my history.. it was you and him whom i chatted earlier.. sorry about that.. |
| 06:45.15 | hadronzoo | do jabber.conf and gtalk.conf follow a standard format? If so, could someone point me in the right direction? |
| 06:46.11 | *** join/#asterisk ltd (n=z@pat.transact.net.au) |
| 06:48.57 | drmessano | hadronzoo: Try the samples in the tarball |
| 06:50.23 | hadronzoo | drmessano: I have it working for one account, and I can have two accounts logged in and working. My question is how do I assign a context to a jabber connection. Or, how to jabber labels map to the gtalk.conf file? |
| 06:51.12 | hadronzoo | drmessano: s/to/do |
| 06:51.50 | drmessano | hadronzoo: The setup in gtalk.conf is associated with a connection in jabber.conf |
| 06:51.57 | drmessano | Its all well laid out in the samples |
| 06:53.03 | drmessano | gtalk.conf defines the contexts, allow lines, etc |
| 06:53.09 | hadronzoo | drmessano: can you point me to the samples you are looking at (I've had to install this manually)? |
| 06:53.28 | drmessano | O.o |
| 06:53.51 | drmessano | They're in the tarball |
| 06:54.23 | hadronzoo | drmessano: the iksemel tarball? |
| 06:54.41 | drmessano | no |
| 06:54.48 | drmessano | These are ASTERISK config files |
| 06:54.55 | drmessano | They are in the ASTERISK tarball |
| 06:55.48 | *** join/#asterisk pikachu2000 (n=pikachu2@196-209-198-105-rrdg-esr-2.dynamic.isadsl.co.za) |
| 06:55.53 | hadronzoo | ok, I'll take a look. Thanks. |
| 07:00.29 | hadronzoo | drmessano: OK, I'm looking at these samples. How is gtalk.conf tied to jabber.conf for an incoming call? |
| 07:01.15 | *** join/#asterisk Ericounet (n=Ericoune@ACaen-151-1-78-94.w86-215.abo.wanadoo.fr) |
| 07:01.30 | hadronzoo | drmessano: In other words, how can I assign seperate contexts for the different jabber connections? |
| 07:01.32 | Ericounet | hello |
| 07:02.05 | drmessano | Its all defined in gtalk.conf |
| 07:02.32 | drmessano | the associated connection statement == connection in jabber.conf |
| 07:03.35 | hadronzoo | drmessano: ok, in this case under [ogorman] label |
| 07:04.32 | hadronzoo | drmessano: so for each jabber connection, I just create a new label, where each label ties to a jabber connection and defines a new context |
| 07:04.54 | *** mode/#asterisk [+b %jeev!*@*] by russellb |
| 07:05.03 | russellb | jeev: congrats on earning another ban |
| 07:06.33 | drmessano | hadronzoo: IIRC the intention was that jabber.conf is your Jabber/XMPP connection and gtalk was the jingle bit of Gtalk |
| 07:06.54 | drmessano | I know the application names have been clarified in 1.6 now |
| 07:09.14 | hadronzoo | drmessano: thanks for clarifying this for me. I think I understand how they tie together now. |
| 07:09.56 | drmessano | hadronzoo: Thats saying a lot.. I had to stare at them for a bit and hold my mouth just right to get it when I took it on some time back |
| 07:11.19 | hadronzoo | drmessano: well, than perhaps "understand" is overstating things :) |
| 07:11.30 | *** join/#asterisk af_ (n=getsmart@88-149-241-170.dynamic.ngi.it) |
| 07:11.43 | drmessano | "Ah, I see what you did there" <-- is close enough |
| 07:12.02 | hadronzoo | right |
| 07:12.20 | [gnubie] | gtg for now.. |
| 07:12.22 | drmessano | Once you get it working its actually all slick how it works together |
| 07:12.26 | [gnubie] | thanks drmessano.. ;) |
| 07:12.40 | *** join/#asterisk feeds (n=feeds@85-135-236-242.adsl.slovanet.sk) |
| 07:12.46 | drmessano | Especially when your asterisk bot starts spamming your IM client with messages from the dialplan |
| 07:13.22 | hadronzoo | ha. I'm already impressed, and I'm just beginning. Making calls from GTalk is really cool |
| 07:14.41 | hadronzoo | (and having those calls terminate to my cell phone) |
| 07:15.44 | drmessano | As soon as AOL gets off their ass and finishes the SIP features they have been working on, now you've got massive IM integration with Asterisk |
| 07:18.41 | hadronzoo | drmessano: when do you think that will occur? |
| 07:19.45 | drmessano | Right now you can register a SIP client to AIM, make SIP calls out with it.. No calling in, no calling other users.. Im really not sure at this point.. They're pussyfooting around for some reason |
| 07:20.55 | hadronzoo | drmessano: what about the other popular services? I know skype is proprietary. What about Yahoo? |
| 07:22.17 | drmessano | Yahoo uses SIP, but it's not accessible |
| 07:22.38 | angler | thinks about sleeping under his desk... |
| 07:22.43 | hadronzoo | drmessano: that's unfortunate |
| 07:22.48 | drmessano | Really, they use SIP because it's there.. no plans to make it open or anything |
| 07:23.28 | drmessano | IIRC MSN used sip at one point |
| 07:23.37 | drmessano | Not sure about now |
| 07:23.46 | *** part/#asterisk feeds (n=feeds@85-135-236-242.adsl.slovanet.sk) |
| 07:25.45 | hadronzoo | drmessano: I know that the point of Asterisk is voice and not messaging, but has someone attempted to write a messaging client for Asterisk that connects to these proprietary protocols? |
| 07:26.28 | jjshoe | msn also yanked sip out |
| 07:26.43 | drmessano | MSN was Net2Phone.. I was just reading it |
| 07:26.52 | drmessano | They used Net2Phone in 2.0 |
| 07:27.53 | *** join/#asterisk tzafrir_laptop (n=tzafrir@local.xorcom.com) |
| 07:28.13 | drmessano | hadronzoo: Effort vs ROI.. Gtalk/Jabber was open, they partnered with Skype, and if AOL comes to the table, it will be due to their implmentation of SIP.. beyond that, it doesnt make a lot of sense |
| 07:28.48 | drmessano | Yahoo and MSN are not widely know for their phone features.. AIM barely is, but they've also made it an issue, not just an accessory |
| 07:29.26 | hadronzoo | drmessano: Right, but the messaging side is understood, right (i.e. Pidgin)? |
| 07:29.57 | drmessano | At some level.. it's all done with reverse engineering |
| 07:30.18 | jjshoe | http://news.aol.com/health/article/bowler-dies-after-rolling-perfect-game/234243 |
| 07:30.20 | jjshoe | poor donnie |
| 07:30.21 | jjshoe | lol |
| 07:30.32 | drmessano | jjshoe: I thought that myself lol |
| 07:30.48 | drmessano | Nihlists got him |
| 07:30.50 | drmessano | lol |
| 07:32.05 | jjshoe | this isn't nam smokie, we have rules here |
| 07:32.20 | drmessano | HE WAS OVER THE LINE |
| 07:32.37 | drmessano | Who the hell brings a gun to a bowling alley |
| 07:33.20 | drmessano | Walter Sobchak, that's who |
| 07:36.15 | drmessano | Digg has a story about a moving skyscraper for NYC |
| 07:36.48 | drmessano | First thought: Oh crap, here they come again.. wait for it... wait for it.... MOVE IT LEFT, NOW |
| 07:37.09 | *** part/#asterisk sivadnz (n=sivad@202-78-149-14.cable.telstraclear.net) |
| 07:37.57 | drmessano | Im still waiting for the Ben Affleck skit from SNL to get posted |
| 07:39.44 | *** join/#asterisk xacatecas (n=jkroon@dsl-240-163-76.telkomadsl.co.za) |
| 07:47.12 | xacatecas | dahdi seems to have both hw and sw gain, why would I need both? |
| 08:00.54 | tzafrir_laptop | xacatecas, at the moment only FXO modules on wctdm/wctdm24xxp seem to support hwgain |
| 08:01.25 | tzafrir_laptop | (yeah, yeah, we have it on our TODO list) |
| 08:01.38 | xacatecas | tzafrir_laptop, not complaining. i only use FXO modules anyway. |
| 08:01.42 | xacatecas | just wondering. |
| 08:02.02 | xacatecas | i personally think * is a really cool product. |
| 08:02.06 | tzafrir_laptop | theoretically hwgain should be the one to use if it available |
| 08:02.22 | xacatecas | makes sense. |
| 08:03.05 | tzafrir_laptop | but test it . we tested it here (by direct registers manipulation, that is) and it had bad effects |
| 08:03.27 | xacatecas | i wouldn't even know how to do direct register manipulation. |
| 08:03.50 | xacatecas | i just set rx/txgain in chan_dahdi.conf and get good results. i leave it alone. |
| 08:04.08 | tzafrir_laptop | it is basically manipulation on the analog waveform, and hence should not be as lossy as the digital software gain |
| 08:05.34 | xacatecas | on a totally different angle, Monitor vs MixMonitor, I like that Monitor allows me to record each channel separately (g729) and then later convert and mix it back, but I take it there are risks to that? |
| 08:08.14 | xacatecas | on chan_dahdi.conf, I'm trying to get the section based configuration working, however, it doesn't seem to be working, how can I go about trouble-shooting this? i've got verbosity at 10 but no luck so far. |
| 08:08.50 | *** join/#asterisk radgrills (n=acoetzee@dsl-145-66-181.telkomadsl.co.za) |
| 08:10.57 | tzafrir_laptop | xacatecas, what version of asterisk do you have? sections-based config works as of 1.6.1 |
| 08:11.21 | radgrills | hi there, i'm a complete n00b to asterisk and I just checked out the latest svn trunk and compiled it on my gentoo ~x86 system. |
| 08:11.23 | tzafrir_laptop | the sample chan_dahdi.conf in 1.6.0 accidentally included them |
| 08:11.35 | xacatecas | 1.6.0 with dahdi 2.0.0 |
| 08:11.36 | tzafrir_laptop | (and the sample file was fixed in 1.6.0.1) |
| 08:12.22 | radgrills | i need some help with a floating point error when I try to dial from a sip phone |
| 08:12.30 | tzafrir_laptop | xacatecas, in fact, the change of that sample file was the sole change from 1.6.0 to 1.6.0.1 |
| 08:12.43 | xacatecas | tzafrir_laptop, ok well, then I'm just going to use the non-section stuff for now, was just thinking it's a cleaner mechanism. |
| 08:12.53 | xacatecas | too lazy to go and compile asterisk-1.6.1 right now. |
| 08:13.39 | tzafrir_laptop | xacatecas, alternatively, if you don't use users.conf for anything, use it as your chan_dahdi.conf |
| 08:14.52 | xacatecas | i don't. how would I go about that? I've only got like three lines per section and I've already written it without the section stuff, but it most definitely would be "safer" to use the sectionized stuff. |
| 08:15.02 | xacatecas | reads users.conf |
| 08:15.21 | tzafrir_laptop | just put the sections you wanted to put in chan_dahdi.conf, in users.conf instead |
| 08:15.41 | xacatecas | this will work for fxo channels too? |
| 08:15.56 | tzafrir_laptop | don't try to have that generate dialpan, sip, or whatever |
| 08:16.09 | tzafrir_laptop | yes |
| 08:16.16 | xacatecas | tries |
| 08:16.26 | tzafrir_laptop | chan_dahdi.so reads users.conf and reads each section separately |
| 08:16.47 | tzafrir_laptop | radgrills, floating point error? where? |
| 08:16.53 | xacatecas | ok, so I should also set hassip=no and hasiax=no in [general] in users.conf? |
| 08:17.15 | radgrills | tzafrir, as soon as i try to dial anything |
| 08:17.18 | tzafrir_laptop | xacatecas, I think it's the default. but maybe it would be more clear that way |
| 08:17.32 | radgrills | my dialplan works fine with version 1.4 |
| 08:17.41 | tzafrir_laptop | floating point error? do you have a backtrace? |
| 08:17.51 | radgrills | i ran an strace, but i don't really understand the output |
| 08:18.29 | tzafrir_laptop | radgrills, I would put that in the, pardon the expression, "strange gentoo errors dept" |
| 08:18.30 | radgrills | may i post the last few lines of the strace output? |
| 08:18.40 | radgrills | lol |
| 08:19.03 | radgrills | my svn version is Asterisk SVN-trunk-r153577 |
| 08:19.17 | tzafrir_laptop | You gentoos experiment with all sorts of combinations of build options, and hence expose all that should not be done :-) |
| 08:19.26 | xacatecas | strace gives a list of system calls, probably not too useful for floating point errors, gdb would be more useful (if you've compiled with debugging info). |
| 08:19.29 | tzafrir_laptop | radgrills, does it crash asterisk? |
| 08:19.38 | radgrills | yes it crashes |
| 08:19.40 | xacatecas | tzafrir_laptop, i use gentoo too, have very, very few "strange gentoo errors" |
| 08:19.43 | tzafrir_laptop | if so, next thing to do would be a backtrace |
| 08:19.59 | radgrills | i don't mind recompiling |
| 08:20.39 | radgrills | i'm really keen to try out chan_mobile |
| 08:20.52 | radgrills | that's why i wanted the SVN trunk |
| 08:21.38 | xacatecas | radgrills, i discovered that thing too on friday - haven't yet had time to play, but it looks really useful. |
| 08:21.43 | radgrills | the CLI works fine, I can execute various commands, the crash only happens when I try to dial from a sip phone |
| 08:21.54 | tzafrir_laptop | xacatecas, here's a similar error that is, in fact, not on a gentoo system: http://lists.digium.com/pipermail/asterisk-users/2008-October/221093.html |
| 08:21.56 | xacatecas | to chan_mobile? |
| 08:22.12 | radgrills | no, to any number (valid or invalid) |
| 08:22.28 | radgrills | sorry, extenstion instead of "number" |
| 08:22.37 | xacatecas | compile with debugging info, run asterisk inside gdb and get a backtrace on the crash. |
| 08:22.44 | tzafrir_laptop | (someone played with a custom kernel, and actually on a Debian system, and ran into an issue with the scheduler) |
| 08:22.47 | radgrills | ok will try |
| 08:22.52 | xacatecas | tzafrir_laptop, users.conf works! |
| 08:23.13 | tzafrir_laptop | xacatecas, also: check that it does not create any dialplan |
| 08:23.40 | radgrills | i did do my own kernel, so it could well be something there |
| 08:24.02 | tzafrir_laptop | radgrills, it would look at userspace first |
| 08:24.17 | tzafrir_laptop | kernel normally does not touch floating point |
| 08:24.17 | xacatecas | tzafrir_laptop, i scanned dialplan show and didn't see anything, is there a more reliable way to confirm this? |
| 08:24.17 | radgrills | the other thing is, i don't have any zaptel/dahdi components, and i'm not using ztdummy |
| 08:24.31 | radgrills | tazfir, ok thanks for that tip |
| 08:24.52 | radgrills | sorry for the finger trouble |
| 08:25.29 | radgrills | fwiw, my kernel is 2.6.26-gentoo-r1 |
| 08:27.00 | radgrills | ok, i just need to emerge gdb, and check the debug compile option |
| 08:27.25 | tzafrir_laptop | the standard build is with debug information |
| 08:28.00 | tzafrir_laptop | try: file /usr/sbin/asterisk (or whereever it is). If it is "not stripped", it is with debug info |
| 08:28.56 | radgrills | tzafrir, how do i check that it is "not stripped"? |
| 08:29.33 | radgrills | i didn't change any selections (standard .configure, make, make install) |
| 08:30.03 | radgrills | so i guess it has the debug symbols in it then |
| 08:31.25 | radgrills | i've never used gdb before, so I might need some hand-holding |
| 08:33.19 | tzafrir_laptop | gdb -c path/to/core.file /usr/sbin/asterisk |
| 08:33.20 | radgrills | oh yeah, i should mention, my hardware is a little low on spec (PIII 700 MHz, with 256Mb RAM) |
| 08:33.25 | tzafrir_laptop | (gdb) bt |
| 08:33.32 | tzafrir_laptop | (gdb) bt full |
| 08:33.48 | radgrills | thanks |
| 08:34.09 | tzafrir_laptop | radgrills, if it was good enough to build asterisk, it is sure way good enough to run gdb |
| 08:34.20 | radgrills | lol ... i bet |
| 08:34.40 | radgrills | no, i meant that might bear some relation to my FPE |
| 08:35.13 | tzafrir_laptop | I want to see what libraries were involved |
| 08:35.17 | xacatecas | radgrills, 700MHz? 256MB RAM? not too long ago that was a "high spec" machine. |
| 08:35.31 | radgrills | you're right |
| 08:35.32 | tzafrir_laptop | in fact, the bt would be more interesting for starters |
| 08:35.43 | radgrills | ok |
| 08:35.54 | xacatecas | remembers in 2005 still working with a dual MMX200 with 256MB RAM. With a memory hole at 16MB causing me severe strangeness. |
| 08:36.01 | radgrills | still busy compiling gdb ... |
| 08:36.17 | tzafrir_laptop | xacatecas, err... "not long ago" as in 7 years ago? |
| 08:36.37 | xacatecas | 2005 == 3 yeas ago. |
| 08:36.40 | tzafrir_laptop | is used to simply apt-get install ... |
| 08:36.53 | xacatecas | but yes, the machine was already like 5 years old at the time. |
| 08:37.11 | xacatecas | likes debian too. but not as much as his gentoo. |
| 08:38.05 | radgrills | i "discovered" linux through Knoppix (Debian) ... but I really like gentoo now |
| 08:38.44 | xacatecas | what would this imply: [Nov 2 10:38:26] WARNING[21144]: chan_dahdi.c:4290 handle_alarms: Detected alarm on channel 2: Red Alarm |
| 08:39.05 | xacatecas | it has it's pros and cons. long compile times is a con. |
| 08:40.42 | radgrills | i agree, but at least it minimizes precompiled shared lib issues, and dependency issues |
| 08:40.54 | radgrills | i just wish there was an ebuild for asterisk-svn |
| 08:42.59 | xacatecas | create one. |
| 08:43.13 | xacatecas | i submitted a bug report regarding chan_mobile on friday, it's fixed. |
| 08:43.25 | radgrills | oh great |
| 08:43.36 | xacatecas | no wait, that was the pkgconfig bug ... |
| 08:43.44 | xacatecas | had a LONG day yesterday. |
| 08:44.10 | xacatecas | compile of asterisk-addons currently fails if the mobile useflag is not set (missing deps) |
| 08:44.43 | xacatecas | and i really don't want all the bluetooth jaz on my server where I don't have bluetooth to begin with. |
| 08:45.08 | radgrills | oh, i built it without the using ebuild |
| 08:45.20 | radgrills | i just have a bluetooth dongle |
| 08:45.55 | radgrills | but i'm hoping i will be able to route cellular calls through there |
| 08:46.26 | xacatecas | hehe, i'm hoping to do exactly the opposite, if/when i walk into the office, have asterisk pick up my calls coming in off my cell phone. |
| 08:46.43 | xacatecas | but i'll use my laptop as a "jump" point. |
| 08:47.07 | radgrills | ok |
| 08:47.28 | radgrills | i have another question ... |
| 08:47.32 | xacatecas | or fake a bluetooth device on the server that actually connects to the real bluetooth device on my laptop via tcp or something (local lan, shouldn't be a problem) |
| 08:47.55 | radgrills | yesterday, i build a little IVR plan ... |
| 08:48.22 | radgrills | Using the Background() application |
| 08:48.54 | xacatecas | why not Read()? |
| 08:48.59 | radgrills | I would like it to automatically detect if the incoming call from the PSTN is a fax, and go ahead and receive the fax |
| 08:49.07 | drmessano | I got tired of messing with 1.4 SVN and Chan_mobile |
| 08:49.13 | drmessano | 1.6 makes it much nicer |
| 08:49.19 | *** join/#asterisk MrNaz (n=mrnaz@ppp118-208-228-96.lns10.mel6.internode.on.net) |
| 08:49.34 | radgrills | Is trunk = 1.6 ? |
| 08:49.45 | xacatecas | NVFaxDetect iirc, then create an extension called fax in the same context. if the call jumps to that extension then you've got a fax. |
| 08:49.48 | radgrills | I don't see 1.6 in the version |
| 08:50.03 | radgrills | yes, I do have a fax extension |
| 08:50.18 | xacatecas | i believe NVFaxDetect is part of asterisk-addons. |
| 08:50.23 | radgrills | Is NVFaxDetect an application? |
| 08:50.27 | radgrills | Or a module? |
| 08:50.40 | xacatecas | it's an application that gets loaded as part of some module. |
| 08:50.41 | drmessano | 1.6 is current |
| 08:51.06 | radgrills | so perhaps I should try that instead of trunk |
| 08:51.28 | radgrills | Do I need to call NVFaxDetect in my dialplan? |
| 08:51.47 | xacatecas | yes, just before/after Answer() I believe. |
| 08:52.04 | radgrills | xacatecas, ok i'll add that, thanks |
| 08:52.22 | drmessano | Trunk is a little too cutting edge for me |
| 08:52.28 | xacatecas | I usually have dedicated fax lines/extensions so I haven't used it before as such, but I have seen that that is how things like freepbx does it. (except it seems to crash more often than not on trixbox) |
| 08:52.44 | drmessano | I prefer the stable + SVN |
| 08:53.22 | xacatecas | drmessano, agreed. but I like to just stick with the tagged versions. |
| 08:53.33 | xacatecas | i'll custom patch here + there for specific needs. |
| 08:53.37 | radgrills | drmessano, I'm beginning to understand why you do ... |
| 08:54.05 | drmessano | Well, having commits before tagged versions is much better IMO |
| 08:55.23 | drmessano | and tagged versions are too binding.. Isnt a version number kinda subjective? What makes 1.4.23 better than 1.4.22? Few updates.. and someone saying "Ok, lets release here".. What about the next commit.. does it make it unstable? No |
| 08:55.43 | drmessano | Trunk is a bit different |
| 08:57.07 | xacatecas | drmessano, following svn on a number of different servers becomes a pain in the backside. |
| 08:57.30 | xacatecas | so whilst perfectly do-able for in-house it's a serious pain if you need to do this for 20+ clients. |
| 08:57.33 | drmessano | lol |
| 08:57.56 | drmessano | Who says anything about "keeping up?" |
| 08:58.06 | drmessano | and again |
| 08:58.21 | xacatecas | ah, you're just installing and leaving, and updating if people complain about particular problems :p. smart man. |
| 08:58.30 | drmessano | Uh no |
| 08:58.49 | xacatecas | ok ? |
| 08:59.38 | drmessano | If I decide to upgrade 20+ servers to 1.4.18, and dont upgrade those 20 again until 1.4.22 what makes 1.4.22 any better than when I ran svn update, make and make install 2 weeks after after 1.4.22 and had the latest 1.4.x commits? |
| 09:00.15 | xacatecas | no difference really. |
| 09:00.37 | xacatecas | i think i'm beginning to see your side. it makes sense. |
| 09:02.02 | drmessano | None at all.. and theres no "keeping up", since you're not gonna tell me you've "kept up" anyway.. you decided now was a good time to make it all current, who cares if it's 1.4.22.0.0.0.0 or post 1.4.22.. If you decided to sleep in this week, it would have been 1.4.22.1 |
| 09:04.02 | drmessano | With 1.6 it becomes even more interesting because of the new release strategy.. I think specific version numbers are even less important now |
| 09:05.30 | radgrills | ok, gdb is compiled |
| 09:06.28 | radgrills | tzafrir, any idea where I might look for the core dump? |
| 09:09.24 | tzafrir_laptop | radgrills, post (pastebin?) it |
| 09:10.01 | radgrills | i don't know where to find it? |
| 09:10.15 | radgrills | can I run asterisk from within gdb? |
| 09:10.18 | tzafrir_laptop | hmm... you need to run asterisk with -g |
| 09:10.24 | radgrills | oh ok |
| 09:11.29 | radgrills | i did that, and crashed it by dialing 1 digit on my sip phone. |
| 09:12.42 | radgrills | will it dump core to the current folder, or to /var/spool/asterisk ? |
| 09:12.58 | drmessano | xacatecas: Its funny you mentioned trixbox earlier.. and talk about "keeping up".. Fonality creates a pseudo-uncertainly layer with their yum updates for trixbox almost on an hourly basis.. Do you know if you have 1.4.22.1-1,1.4.22.1-2, or 1.4.22.1-3? Probably not, but they're updating that RPM under nose fixing crap here and there and also increasing the chances they'll bork one. You can already be on 1.4.22.1 and see 3 RPM updates t |
| 09:13.16 | drmessano | Id much rather deal with SVN |
| 09:14.33 | xacatecas | drmessano, agreed, which is exactly why I'm moving away from that crap as quickly as i possibly can. |
| 09:14.57 | xacatecas | my quality is better without their BS. i don't know, I don't have half the functionality yet, but what I have is "just working" now. |
| 09:15.24 | xacatecas | radgrills, current folder. |
| 09:15.48 | radgrills | ok, well then there's no dump |
| 09:16.01 | radgrills | according to strace it exits with a SIGFPE |
| 09:16.09 | drmessano | Indeed.. all that fluff n Trixbox is valued added shit anyway.. Only thing really that FreePBX doesnt provide is provisioning. |
| 09:16.13 | xacatecas | yes, that's most likely a division by zero. |
| 09:16.25 | drmessano | Everything else is just green wrapper for FreePBX and assloads of broken RPMs |
| 09:16.57 | radgrills | well, I just checked out version 1.6.1 and compiled that, so if that works, I can try chan_mobile with that |
| 09:17.03 | xacatecas | drmessano, provisioning is one thing I suspect I'm after. This having to flippen configure every button on every phone by hand is driving my technicians up the wall. |
| 09:18.20 | drmessano | xacatecas: It's not that much work to edit some XMLs.. |
| 09:18.20 | xacatecas | anyway, enough slandering, let's rather focus on a great product, called * and see how we can make it even greater still. |
| 09:19.02 | radgrills | yeah, great product for sure, amazing |
| 09:19.08 | tzafrir_laptop | radgrills, what codec was used there? |
| 09:19.10 | xacatecas | drmessano, well, i first need to get some basic things going, but xml is not difficult to do, just need something to work from. |
| 09:19.11 | drmessano | Slandering would imply things that arent true.. Nothing I said about is not based on 100% fact |
| 09:19.17 | radgrills | g711u |
| 09:19.25 | xacatecas | drmessano, :) |
| 09:21.54 | xacatecas | ok, with the cascading sections I can now declare a section like [line](!) and then [company](line), is it possible to "sub" the [company] one again? in other words, [line1](company) ? |
| 09:24.42 | xacatecas | I've got say 8 incoming lines that needs to be split between about 5 companies that share the pbx, and there are obvious stuff that holds for all lines (like context) and then per company (group, pickupgroup, callgroup) and then per-line (DID and gains) stuff that needs to be set. |
| 09:25.46 | xacatecas | context is shared and goes to a context that only contains "exten => s,1,Goto(incoming,${DAHDI_DID},1)" where DAHDI_DID is set within the line-specific section. |
| 09:29.43 | radgrills | tzafrir, this make install is taking forever ... i can't wait to see if 1.6.1 will be the answer to my problems |
| 09:31.04 | tzafrir_laptop | xacatecas, you can use setenv in some channel drivers . I think that in chan_dahdi 1.6.1 as well |
| 09:31.14 | xacatecas | in 1.6.0 already. |
| 09:31.31 | tzafrir_laptop | even better |
| 09:32.00 | xacatecas | i make use of those, it's just that a bunch of things are set at each of those three levels and I'm one of those fanatics that hate duplicating even the smallest of bits and pieces of configuration. |
| 09:32.42 | drmessano | ouch |
| 09:32.51 | xacatecas | anyway, so I want to create templates from templates. |
| 09:32.52 | drmessano | I wouldnt be using 1.6.1 yet.. |
| 09:32.56 | tzafrir_laptop | Cascading Configuration Sections . Almost the same as CSS. Any better TLA? |
| 09:32.57 | drmessano | I couldnt get addons to compile with it |
| 09:33.27 | tzafrir_laptop | drmessano, there was a recent addons release, IIRC |
| 09:33.40 | tzafrir_laptop | anyway, could you point me to an error message? |
| 09:33.44 | xacatecas | tzafrir_laptop, argh! no, i can't think of any. |
| 09:34.13 | drmessano | I tried addons SVN last weekend and it didnt work.. unless they've changed something |
| 09:34.44 | tzafrir_laptop | xacatecas, that's as opposed to Configuration All in one Section (CAS) |
| 09:35.07 | drmessano | Nope.. no updates |
| 09:35.24 | xacatecas | this cascading thing is actually more configuration but it reads a heck of a lot simpler. |
| 09:37.59 | radgrills | brb |
| 09:40.48 | rednul | I have a situation where I have multiple accounts with the same provider. I setup a sip trunk (using FreePBX) for each account. Incoming calls seem to work fine. However, outgoing calls fail on any trunk but the first (seems to have an issue authenticating properly). It seems that it due to multiple sip connections going to the same server ip & port. Any advice getting asterisk to work with these multiple accounts to t |
| 09:42.45 | xacatecas | rednul, one of my clients had exactly the opposite problem. please pastebin your configuration. |
| 09:43.24 | xacatecas | drew, tzafrir_laptop - what is the purpose of having multiple "contexts" for voicemail? |
| 09:43.45 | xacatecas | each user is only supposed to have access to his/her own voicemail anyway? |
| 09:43.48 | tzafrir_laptop | multiple namespaces |
| 09:44.03 | tzafrir_laptop | e.g.: for hosting multiple companies |
| 09:44.17 | xacatecas | so there are no security implications or something similar that i should be aware of? |
| 09:44.31 | xacatecas | so it's purely a namespace thing? |
| 09:44.34 | tzafrir_laptop | yes |
| 09:45.06 | tzafrir_laptop | As for security: if you can change the dialplan, (or even originate calls) you have just about full control over asterisk |
| 09:45.12 | xacatecas | so voice mailboxes need only be unique within a specific namespace? eg, i can have a voice mailbox 123 in both context "a" and "b"? |
| 09:45.40 | tzafrir_laptop | yes |
| 09:45.58 | xacatecas | tzafrir_laptop, i know that, which is why I'm the only person with access to the dialplan :). |
| 09:46.03 | tzafrir_laptop | err.. you can have two different mailboxes with the name 123 in contexts a and b |
| 09:46.19 | xacatecas | yes, that's what i understood. |
| 09:46.30 | xacatecas | my question was unclear. |
| 09:47.12 | xacatecas | ok, so essentially if I then have a number to dial to get to voicemail in theory I can tell it that the caller only have access to voice mail boxes in a specific context? |
| 09:48.03 | xacatecas | hmm, looks like it based on the description of VoiceMailMain |
| 09:48.45 | rednul | xacatecas: http://pastebin.com/m3908c011 |
| 09:54.41 | radgrills | tzafrir, great news .... 1.6.1 doesn't crash |
| 09:55.22 | radgrills | tzafrir, so for now I'll forget about trunk and stick with 1.6.1 |
| 09:55.37 | radgrills | thanks for all the help |
| 09:57.45 | *** join/#asterisk Bladerunner05 (n=feelme@81-174-56-54.static.ngi.it) |
| 09:59.10 | drmessano | Which addons did you use? |
| 10:01.17 | Bladerunner05 | i'm looking for web-interface to see asterisk log order by callerid or extension time etcetc |
| 10:02.09 | radgrills | Bladerunner05, I was looking at asterisk-stat yesterday, looks quite interesting |
| 10:03.16 | radgrills | http://www.areski.net/asterisk-stat-v2/about.php |
| 10:05.55 | xacatecas | rednul, looks good as far as I can tell. you say line-1 is working but the rest not? |
| 10:06.32 | xacatecas | authname <-- is this different to authuser? |
| 10:06.58 | rednul | xacatecas: correct... if i disable line 1.. then line 2 works but not 3... and so on.... my provider has a secondary IP address I can connect to.. if i use that for line 2, it works, but then line 3 doesn't.. on either IP address.... |
| 10:07.21 | rednul | xacatecas: no....it was just something I added from reading one of the wikis.. hoping to help :) |
| 10:07.27 | xacatecas | incoming calls always works ... |
| 10:07.36 | rednul | xacatecas: yep |
| 10:07.51 | rednul | xacatecas: as far as i can tell... |
| 10:07.56 | xacatecas | on the topic, what exactly does the /foo after a register line do? |
| 10:08.12 | xacatecas | rednul, what does sip set debug on reveal? |
| 10:08.17 | radgrills | aaaargh, now I have a new problem with 1.6.1 |
| 10:09.02 | xacatecas | lol @ radgrills - you're just a problematic person today. |
| 10:09.10 | radgrills | My Ekiga softphone fails to authenticate, hardware sip phones are working fine |
| 10:09.10 | xacatecas | anyway, I've got to run, i'm late for an appointment. |
| 10:09.17 | radgrills | lol |
| 10:10.01 | radgrills | i can dial my Ekiga phone from a hardware phone ... clearly something wrong in my Ekiga configuration |
| 10:11.06 | radgrills | ok cheers xacateca, thanks for the support |
| 10:12.03 | radgrills | This was working perfectly with 1.4 |
| 10:14.23 | radgrills | <PROTECTED> |
| 10:16.01 | Bladerunner05 | thanks radgrills |
| 10:16.45 | radgrills | Bladerunner05, it needs mySQL or Postgres |
| 10:17.07 | radgrills | but I like the screenshots |
| 10:17.21 | Bladerunner05 | I see |
| 10:18.00 | drmessano | radgrills: No problems here with 1.6.0 |
| 10:18.16 | *** join/#asterisk tmiw (i=mooneer@voldemort.lifeafterking.org) |
| 10:19.12 | radgrills | drmessano, yes, I don't think its an asterisk problem this time - because my other hardware phones are working fine |
| 10:19.56 | radgrills | i'm using a minimal sip.conf, so I'm probably missing something in my config |
| 10:22.56 | radgrills | sip show peers sees my ekiga phone as "online" |
| 10:23.16 | radgrills | moment i dial from ekiga, i get "security check error" on the ekiga status bar |
| 10:23.50 | radgrills | and that "handle_request_invite" error in the CLI |
| 10:33.55 | radgrills | oh well, insecuer=invite,port fixes that problem :) |
| 10:34.04 | radgrills | insecure even |
| 10:35.47 | radgrills | right, now to tackle chan_mobile |
| 10:37.20 | radgrills | can i use the trunk version of addons with the 1.6.1 branch? |
| 10:41.00 | *** join/#asterisk ltd (n=z@pat.transact.net.au) |
| 10:43.55 | radgrills | thinks i'll find out soon enough |
| 11:12.21 | radgrills | ok, looks like that works :-) |
| 11:36.05 | *** join/#asterisk Daejeo (n=chatzill@118.221.248.5) |
| 11:38.16 | Daejeo | hello guys, i made a phone verification system for eliminating anonymous web registration/feedback abuse . |
| 11:38.48 | *** join/#asterisk rnode (n=rdn@dehghany.demon.co.uk) |
| 11:38.51 | rnode | Hi guys |
| 11:39.00 | rnode | can you integrate asterisk with cisco call manager? |
| 11:39.07 | Daejeo | i want to evaluate the performance |
| 11:39.45 | Daejeo | i would appreciate if few people can test it |
| 11:40.11 | Daejeo | i can provide the URL by PM |
| 11:42.24 | Daejeo | mode? |
| 11:42.51 | Daejeo | ah |
| 11:42.52 | Daejeo | rnode |
| 11:42.58 | Daejeo | :) |
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| 11:56.04 | donnib | hi |
| 11:56.05 | donnib | i have exten => _X.,1,Set(CALLERID(num)=0049${CALLERID(num)}) but the num is having a 0 in front. how can i remove that ? |
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| 11:58.37 | donnib | anyone ? |
| 12:01.23 | tzafrir_laptop | {CALLERID(num):1} ? |
| 12:01.31 | tzafrir_laptop | I'm not sure this syntax works |
| 12:04.24 | donnib | it does thank you :) |
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| 13:03.23 | LND | exit |
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| 13:31.22 | hi365 | anyone got the new 'rining' icon to work with polycom blf? (the icon that shows you that a remote extension is ringing) |
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| 14:45.29 | Dovid | hi guys. |
| 14:46.19 | Dovid | working on creating a system where if some one calls an extension and they do not pick up say after 30 seconds then i wan to be able to see if they were on the phone and that is why they did not pick up or if the person was not at their desk and they did not pick up |
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| 14:57.29 | seanbright | Dovid: 'core show application Dial' at the asterisk CLI |
| 14:57.40 | seanbright | look for the timeout option and the DIALSTATUS variable |
| 14:58.07 | *** join/#asterisk ngvoice (n=linxrout@58.186.250.2) |
| 14:58.23 | ngvoice | ; |
| 14:58.44 | ngvoice | hi there |
| 14:59.14 | ngvoice | no one |
| 15:00.55 | ngvoice | so silence |
| 15:01.00 | tzafrir_laptop | -ENONE |
| 15:02.14 | Dovid | seanbright: I tried it. got 0 back every time. |
| 15:02.22 | Dovid | do i need hinting working for it ? |
| 15:02.37 | seanbright | no |
| 15:02.49 | seanbright | pastebin your extensions.conf |
| 15:02.55 | seanbright | ~pb |
| 15:02.56 | 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 |
| 15:04.07 | *** join/#asterisk [TK]D-Fender (n=chatzill@64.235.218.194) |
| 15:04.56 | ngvoice | hi |
| 15:05.25 | ngvoice | have anyone able to show the callee name on display |
| 15:05.54 | ngvoice | like when i press 6000 to call Mr A, his name show up on the phone |
| 15:06.04 | ngvoice | just like cisco call manager |
| 15:06.46 | [TK]D-Fender | ngvoice: there is a path being worked on, but nothing final. |
| 15:06.51 | [TK]D-Fender | patch* |
| 15:06.56 | [TK]D-Fender | ~cpid |
| 15:06.56 | jbot | [~cpid] Called-Party ID is possible with * using patches on Mantis. See : http://bugs.digium.com/view.php?id=8824 |
| 15:07.11 | Dovid | sean: http://pastebin.ca/1243230 |
| 15:07.51 | ngvoice | thanks TK |
| 15:09.03 | seanbright | Dovid: 'core set verbose 10' |
| 15:09.06 | seanbright | Dovid: do a test call |
| 15:09.11 | seanbright | Dovid: pastebin the CLI output |
| 15:09.50 | Dovid | sean:I always get noanswer |
| 15:09.58 | Dovid | even when that extension is on the phone |
| 15:10.00 | seanbright | ohhh |
| 15:10.09 | Dovid | i am trying to see if that user was on the phone or not already when they get this call |
| 15:10.31 | seanbright | is "that user" a local user? |
| 15:11.38 | seanbright | unless you use either call-limit along with hints or the GROUP()/GROUP_COUNT() functions, it will always be NOANSWER |
| 15:12.05 | Dovid | seanbright: I have hints set up |
| 15:12.39 | seanbright | what about call-limit on your SIP friends? |
| 15:12.50 | Dovid | seanbright: u see here the hints updating: http://pastebin.ca/1243233 |
| 15:12.59 | Dovid | call-limit = 100 |
| 15:13.02 | Dovid | it is set |
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| 15:13.21 | seanbright | right, at 100 |
| 15:13.34 | Dovid | http://pastebin.ca/1243235 |
| 15:13.40 | seanbright | so asterisk will deliver 100 calls before it returns a BUSY |
| 15:13.51 | Dovid | ok. if i set it to 1 |
| 15:14.11 | Dovid | then it would let me know that they were busy however then i wouldnt be able to call their extension so its of no use |
| 15:14.18 | [TK]D-Fender | Dovid: what are you expecting to happen here? |
| 15:15.20 | Dovid | TK: Call extension 100, if user was not on the phone at the time say "user was away....... 1 for foo 2 for bar" if user was on the phone when they got this second call say "user was on the phone and busy...... 1 for x 2 for y" |
| 15:15.40 | seanbright | right... so set call-limit to 1 and you're done |
| 15:15.42 | [TK]D-Fender | Dovid: Well you didn't do any of that. |
| 15:15.50 | seanbright | or use GROUP/GROUP_COUNT |
| 15:16.05 | [TK]D-Fender | Dovid: You aren't checking the status before you dail and you aren't calling ChanIsAvail properly in the first place. |
| 15:16.36 | Dovid | TK: I did actually Noop it out b4 but it didnt work so i tried it after. |
| 15:17.10 | [TK]D-Fender | Dovid: Did I say anything about the variable? No. You are calling the APP wrong. |
| 15:17.37 | Dovid | TK: then what am i doing wrong ? |
| 15:17.54 | [TK]D-Fender | Dovid: You are calling the app wrong <- |
| 15:20.18 | Dovid | TK: That may be but I dont see how. from looking at the wiki example "exten => s,1,ChanIsAvail(Zap/2&Zap/1) " |
| 15:21.34 | [TK]D-Fender | Dovid: Those aren't its instruction. How many more times do we have to say that the wiki is outdated crap 99% of the time? |
| 15:21.45 | seanbright | core show application ChanIsAvail |
| 15:21.50 | Dovid | TK: I get the same form core show application chanisavail |
| 15:21.53 | Dovid | "core show application chanisavail" |
| 15:22.08 | seanbright | look at the options |
| 15:22.18 | [TK]D-Fender | face-palm |
| 15:23.03 | Dovid | seanbright: I can use s but then i wont know if the extension was actually busy OR if they were on the phone. |
| 15:23.05 | Dovid | but OK |
| 15:23.13 | seanbright | what is the difference? |
| 15:23.19 | [TK]D-Fender | Dovid: O RLY? |
| 15:23.35 | Dovid | TK: so i have seen but let me test again |
| 15:23.40 | [TK]D-Fender | Dovid: What is a "real" busy" mean? |
| 15:23.57 | seanbright | i am confused by the difference between "actually busy" and "they were on the phone" |
| 15:24.03 | [TK]D-Fender | Dovid: You just don't seem to get it. |
| 15:24.21 | [TK]D-Fender | Dovid: "busy" is when the PHONE tells you to "fuck off". |
| 15:24.41 | Carlos_PHX | Add this to notes for future use. |
| 15:24.45 | Carlos_PHX | ROFL |
| 15:24.48 | [TK]D-Fender | Dovid: the phone WANTS to accept another call even if its on a call already because it CAN <- |
| 15:24.51 | seanbright | Carlos_PHX: it's already part of the SIP standard ;) |
| 15:25.18 | seanbright | RFC 12312: Fuck-Off Support in SIP Devices |
| 15:25.24 | Dovid | ;) |
| 15:25.45 | [TK]D-Fender | Dovid: "busy" isn't "I'm on a call right now", its "I couldn't take you if I wanted to" |
| 15:26.05 | Carlos_PHX | I should set that as one of the DND text options. |
| 15:27.36 | Dovid | TK: let me explain again. I want to let the caller know if the person they called was away from their desk and they are just lazy or if they were actually on a call, their phone rang but they didnt pick upbecause they were on another call |
| 15:28.06 | [TK]D-Fender | Dovid: Yes, you answered that earlier |
| 15:28.09 | Dovid | TK: I guess i was looking for something that would tell me (user is on the phone) |
| 15:28.17 | [TK]D-Fender | Dovid: It does! |
| 15:30.45 | Dovid | TK: ok then if I have http://pastebin.ca/1243250 (whcih I hope I using it correctly) the variable should change based on the state. it is always 0 |
| 15:31.36 | [TK]D-Fender | Dovid: Whart ver of *? |
| 15:31.45 | seanbright | 0.9 |
| 15:32.43 | Dovid | hahah |
| 15:32.43 | Dovid | 1.4 |
| 15:33.25 | Dovid | 1.4.20.1 to be exact |
| 15:33.36 | [TK]D-Fender | Dovid: first change your "|" to a ",".... that's dumped in 1.6, then pastebin a call including a channel dump PRIOR as well as a hint dump. |
| 15:34.03 | [TK]D-Fender | Dovid: And you should just test "availchan" as being blank or not |
| 15:34.20 | Dovid | TK: I have "," and not "|" |
| 15:34.36 | Dovid | sorry for asking wut do u menay by hunt dump ? |
| 15:34.42 | Dovid | and channel dump |
| 15:35.01 | Dovid | nm. i see by the |S |
| 15:35.34 | [TK]D-Fender | Dovid: And stop with the extra white-space in your exten lines. |
| 15:35.50 | [TK]D-Fender | Dovid: "core show channels" <- |
| 15:36.24 | Dovid | hint dump = ? |
| 15:36.56 | [TK]D-Fender | Dovid: core show hint". FFS |
| 15:36.59 | Dovid | white spaces make issues ? |
| 15:37.13 | [TK]D-Fender | Dovid: Whitespace fucks up all sort of things. |
| 15:37.26 | Dovid | good to know |
| 15:37.30 | Dovid | let me try with out |
| 15:37.32 | [TK]D-Fender | Dovid: You don't read instructions and take liberties all over the place. |
| 15:37.54 | Dovid | TK: I actually do it based on whom i learnt from ;) i guess i went wrong there |
| 15:38.28 | Carlos_PHX | I'd recommend reading the sample files, you learn a lot, and follow their style. |
| 15:39.02 | Carlos_PHX | Never heard: "Asterisk is so forgiving in how I write my dialplan." |
| 15:39.48 | [TK]D-Fender | * variable usage is the dumbest thing I've ever seen and I've written considerably better 15 years ago. |
| 15:40.36 | [TK]D-Fender | Whitespace will screw GotoIF's, et and all sorts of fun things. |
| 15:41.23 | Dovid | TK: http://pastebin.ca/1243253 |
| 15:41.32 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 15:42.39 | Dovid | wut was wierd now is with using the US when the second call comes in it goes no where. i should still ring. U see there that i have to soft hang it up |
| 15:43.26 | [TK]D-Fender | Dovid: No, I don't see that. What I see is 1 call to chanisavail and no checking of the result. |
| 15:43.48 | [TK]D-Fender | Dovid: I see my time is completely wasted. |
| 15:43.54 | Dovid | TK: I used a noop to display the status. i start slowly |
| 15:44.03 | Dovid | i guess i just need to start fresh tomorrow |
| 15:44.58 | [TK]D-Fender | Dovid: your test is bad |
| 15:45.12 | [TK]D-Fender | You are not showing me the status BEFORE chanisavail is called |
| 15:45.22 | Dovid | Throws hands in the air and gives up for the day. (or atleast wants to) |
| 15:45.29 | [TK]D-Fender | Dovid: And you are not checking the variable I told you to. |
| 15:46.15 | Dovid | TK: Ok. you want {AVAILSTATUS} before i call chanisavail and what other variable where ? |
| 15:46.21 | Carlos_PHX | Dovid: Suggestion...follow [TK]D-Fender's help closely, he knows what he's saying and you're getting valuable help... |
| 15:46.33 | Dovid | oh i see now: availchan" |
| 15:46.39 | [TK]D-Fender | Dovid: You cehked if they were on the phone. They WEREN'T. You then went forward to call them. they ANSWERD. THEN you deicide its time to do a channel dump? |
| 15:46.48 | [TK]D-Fender | Dovid: after which we see ONLY that call in progress! |
| 15:47.00 | [TK]D-Fender | Dovid: which means they WEREN'T on the phone BEFORE and are NOW. |
| 15:47.18 | hi365 | ive configure a mysql cdr. how can i test if the connect is succesfull? is there a command to see the status? |
| 15:48.42 | Dovid | TK: baby steps for me. i should call in and ignore what it tells me then call in again (second time) and then run the dumps ? |
| 15:48.58 | *** join/#asterisk maxxim (n=maxxxim@93-96-96-106.zone4.bethere.co.uk) |
| 15:49.19 | [TK]D-Fender | Dovid: Why are you testing chaanisavail when they are on the &^$#$ phone? |
| 15:49.37 | [TK]D-Fender | aren't* |
| 15:49.42 | maxxim | hi, how to configure asterisk, if the line is congested, the clirnt to be put on musoconhold, and automaticaly to transfer to the line, if it became free. |
| 15:50.05 | [TK]D-Fender | maxxim: this is up to you to code in the dialplan. |
| 15:50.10 | Dovid | TK: I want to save it for later to see where to send the user if the called person did not pick up |
| 15:50.32 | hi365 | hmm, cdr mysql status shows Not currently connected to a MySQL server. does asterisk only connect when it has something to deposit in the db? |
| 15:50.41 | [TK]D-Fender | maxxim: You have to Dial, check the result, play MoH for X time. try again. Or use queues |
| 15:51.07 | Carlos_PHX | hi365: That's set in your DB config. Read the sample file. |
| 15:51.11 | [TK]D-Fender | Dovid: Check before you dial. You have that var through the call. |
| 15:51.48 | Carlos_PHX | hi365: pre-connect => yes |
| 15:52.04 | Dovid | Tk: check ${AVAILCHAN} |
| 15:52.07 | maxxim | [TK]D-Fender> where can i read more about Queues? |
| 15:52.25 | Carlos_PHX | On voip-info.org or in the queues.conf sample |
| 15:52.30 | *** join/#asterisk jer (n=jer@unaffiliated/jer) |
| 15:52.35 | maxxim | thanks |
| 15:52.45 | [TK]D-Fender | maxxim: The sample configs, the book, then the WIKI |
| 15:53.24 | Carlos_PHX | Queues are powerful but ... What is the word... Particular to how they are conigured. |
| 15:53.57 | Dovid | configs are only good as the one creating them, in my case i am up shits creek |
| 15:54.22 | maxxim | thanks to all, i'll try to study it :) |
| 15:54.34 | jaytee | maxxim, the book is your paddle |
| 15:54.39 | jaytee | ~boo |
| 15:54.39 | jbot | for heaven's sake, jaytee, don't do that! |
| 15:54.39 | jaytee | ~book |
| 15:54.40 | jbot | from memory, book is Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com or see ~buybook |
| 15:54.46 | jaytee | hehehe |
| 15:55.05 | maxxim | thanks :) |
| 15:55.38 | *** join/#asterisk Zeeek (n=Zeeek@pdpc/supporter/active/Zeeek) |
| 15:56.17 | hi365 | Carlos_PHX: what file do you see pre-connect => yes in? |
| 15:56.28 | hi365 | it sure aint here: http://svn.digium.com/view/asterisk-addons/branches/1.4/configs/cdr_mysql.conf.sample?view=markup |
| 15:57.59 | Carlos_PHX | res_odbc.conf |
| 15:58.15 | Carlos_PHX | I'm looking at a 1.6 server though. |
| 15:58.43 | hi365 | ah |
| 15:58.56 | Carlos_PHX | Fairly certain in 1.4 it's the same file same thing |
| 15:59.13 | hi365 | ill just assume that the thing will connect when it has what to send (and im not using odbc, im using mysql) |
| 16:04.09 | *** join/#asterisk neurosys (n=neurosys@adsl-225-163-83.mia.bellsouth.net) |
| 16:17.00 | Dovid | TK: I applied the devstate and its working for when that user is on a call :) |
| 16:17.08 | Dovid | the devstate patch* |
| 16:17.30 | Dovid | but when they are on they make a call it does not show they are on the phone. will work on it ;) |
| 16:17.33 | Dovid | thanks for all the help |
| 16:18.10 | [TK]D-Fender | ? |
| 16:18.34 | [TK]D-Fender | Dovid: Devstate is NOT phone check if a device is busy. Its for creating a VIRTUAL device so you can mess with presence indicators |
| 16:18.47 | [TK]D-Fender | Dovid: they are the opposite of what you are looking for. |
| 16:19.27 | Dovid | ok. |
| 16:19.59 | Dovid | either way its cool if i wana play around |
| 16:20.47 | [TK]D-Fender | Dovid: Yes, it can be a very useful thing for reporting dailplan flag status for how it will handle calls, etc. Night-mode indicators, and other unusual things. |
| 16:21.04 | maxxim | hi, i want to use L parameter in Dial, in order to play a sound every X seconds till the end of the call. unfortunatley it doens't works propoely for me. the call is aborted after the specified time, but the worning sound is not played during the call. please look here http://rafb.net/p/JBIGbt24.html |
| 16:25.07 | Dovid | TK: If i did: http://pastebin.ca/1243287 |
| 16:25.38 | Dovid | then I can subscribe to Custom:lamp1 ? |
| 16:28.20 | Dovid | TK: nm. i need to then subscribe to 1234. works ;) |
| 16:38.34 | *** join/#asterisk xnixan (n=xnixan@unaffiliated/xnixan) |
| 16:39.27 | write__erase | hi365, Are there SQL create table scripts somewhere in SVN ? for CDR ? |
| 16:39.48 | hi365 | duno |
| 16:41.01 | write__erase | ok thx |
| 16:41.34 | carrar | asterisk/contrib/scripts/postgres_cdr.sql |
| 16:41.56 | write__erase | great |
| 16:43.19 | maxxim | i;ve found the problem by looking into sourse code of *, it is a new syntacs deliter, L(20000:10000:3000) |
| 16:46.02 | *** join/#asterisk grEvenX (n=even@cB78D5AC1.dhcp.bluecom.no) |
| 16:47.09 | *** join/#asterisk jaytee (n=jforde05@unaffiliated/jaytee) |
| 16:50.14 | drmessano | jaytee: Have you seen affleck doing Olbermann on SNL? |
| 16:50.59 | *** join/#asterisk russellb (n=russell@asterisk/digium-open-source-team-lead/russellb) |
| 16:50.59 | *** mode/#asterisk [+o russellb] by ChanServ |
| 16:51.18 | jaytee | drmessano, nope. I missed it. |
| 16:51.38 | jaytee | hi russell |
| 16:55.19 | drmessano | http://www.nbc.com/Saturday_Night_Live/video/clips/countdown-with-keith-olbermann/805561/ |
| 16:55.29 | drmessano | Quite honestly.. funniest SNL skit in years |
| 16:55.36 | drmessano | This week.. |
| 16:56.37 | Carlos_PHX | Has anyone used the Noojee Firefox plugin? (Click to dial from Firefox) |
| 16:58.35 | drmessano | circles the room three times around Carlos_PHX, gaining speed and momentum, flies over, grabs the edge of his underoo's and applies a MASSIVELY PAINFUL NOOOOOJEEEEEEEE!!!!!1111!!!!ONES!!!!!11!!!! |
| 16:59.08 | drmessano | Ahem |
| 16:59.14 | drmessano | No, I have not |
| 17:01.13 | Carlos_PHX | Wonders why he assumes I'm wearing underwear. Or anything at all really. |
| 17:06.34 | jjshoe | I can't recall the last time i wore underwear |
| 17:06.37 | *** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 17:07.28 | drmessano | I had a really good line about how disturbing that was, followed by an attempt to one-up it, however, there's still time in life for me to show I am more than just another Jeev with a Cobalt Praetorian American Express card |
| 17:07.37 | drmessano | "Don't leave Rome without it" |
| 17:08.16 | jaytee | hahaha, Affleck nailed it!!!! |
| 17:09.01 | drmessano | HOW DARE YOU SIR, HOW DARE YOU |
| 17:09.04 | Carlos_PHX | WTF. Sometimes SIP sucks. |
| 17:10.22 | jaytee | drmessano, I have a Pink Diamond Praetorian Card, the preferred card for gay rollerbladers everywhere, "Don't skate to the gloryholes without it!" |
| 17:10.56 | drmessano | lol |
| 17:12.25 | jaytee | drmessano, what's really sad is that sometime the real Olbermann is dead on with his facts and analysis but unfortunately his delivery makes him look as big an assclown from the left as Bill O'Reilly is from the right. Just another bloviated bag of bullshit. |
| 17:13.26 | drmessano | That was really the best part.. Using the same inflection bashing Bush as he did the president of the coop board |
| 17:13.33 | drmessano | Calling for his immediate resignation |
| 17:13.35 | drmessano | toooo funny |
| 17:13.54 | drmessano | "Their silence is deafening" |
| 17:14.50 | [TK]D-Fender | drmessano: Yes, good piece. I like what Olberman has been able to do mind you, and the facts really just say it. Mind you his inflection is harsh.... kinda like what a huge amount of the population feels but couldn't say themselves. |
| 17:15.30 | [TK]D-Fender | Rachel Maddow is the calmer quirkier side of it esp as Olberman is moving out from the Countdown |
| 17:15.41 | jjshoe | I can't wait until we elect the next idiot so people stop talking about politics for a while :) |
| 17:16.07 | drmessano | Obama endorsed Asterisk |
| 17:16.34 | drmessano | Sure Ron Paul runs his pacemaker on it |
| 17:16.38 | drmessano | But still, cool |
| 17:16.44 | jjshoe | he's also half hasidic jew / half catholic |
| 17:17.50 | [TK]D-Fender | jjshoe: thats a lot of guilt in one man.... |
| 17:18.24 | jaytee | one side does guilt the other does shame. |
| 17:19.29 | *** join/#asterisk ManxPower (n=manxpowe@37.sub-75-202-117.myvzw.com) |
| 17:20.36 | drmessano | They pick a day of the week |
| 17:20.58 | drmessano | Shame on Wednesday, guilt Thursday |
| 17:21.07 | drmessano | God, sounds like some S&M club |
| 17:21.26 | Carlos_PHX | Fine line between that and a nun paddling you. |
| 17:21.43 | drmessano | Thats Tuesdays |
| 17:22.08 | Carlos_PHX | I got thrown out of Catholic school when I was 13 for punching a nun. That sounds bad but I swear she had it coming. |
| 17:22.12 | Carlos_PHX | She started it. |
| 17:23.22 | stintel | :D |
| 17:24.23 | jaytee | I was forced by family pressure to convert to Catholicism at age 12. The nuns in my CCD classes hated me because I asked too many intelligent questions. I left the church after my 16th birthday. |
| 17:28.05 | drmessano | I was forced to join a catholic church on Easter 3 or 4 times as a child. |
| 17:28.09 | Carlos_PHX | The questions are what started the problems for me. |
| 17:30.40 | *** join/#asterisk R-Guy (n=ron-mirc@exmail.mcleodnet.com) |
| 17:31.41 | R-Guy | Anyone know where I can get a DID for Mumbai? |
| 17:32.10 | jaytee | for me it was having a deep love of science and I'd just finished reading a biography of Galileo Galilei and his persecution, being forced to recant his heliocentric theory of the solar system and his eventual house arrest for the remainder of his life was the final straw for me. |
| 17:35.32 | *** join/#asterisk Carlos_PHX (n=Carlos@ip68-3-162-244.ph.ph.cox.net) |
| 17:36.30 | jaytee | Carlos_PHX, did you miss my last statement in response to your line about questions? |
| 17:36.35 | ManxPower | "The Catholic Church does not make mistakes! " --The movie Dogma |
| 17:36.51 | Carlos_PHX | About Galileo, no, saw that. It's but one example. |
| 17:36.54 | jaytee | ManxPower, I have a Buddy Christ dashboard statue |
| 17:37.07 | jaytee | and you can get a bobblehead version now |
| 17:37.08 | Carlos_PHX | "But sister, why did the church kill all these people?" |
| 17:40.00 | *** part/#asterisk Zeeek (n=Zeeek@pdpc/supporter/active/Zeeek) |
| 17:42.07 | jaytee | Carlos_PHX, the honest answer to that question would have been something like, "They weren't team players and we cannot allow that." but the more likely answer would have been some juicy rationalization based no dogma and some obscure scriptural passages. |
| 17:42.24 | jaytee | s/no/on/ |
| 17:42.26 | Carlos_PHX | "They were heathens" |
| 17:42.53 | jaytee | hmm, it didn't correct it. |
| 17:45.38 | *** join/#asterisk amiads (n=ami@bzq-84-108-224-206.cablep.bezeqint.net) |
| 17:45.56 | amiads | hello |
| 17:47.06 | amiads | I'm a new user and have a question about Asterisk |
| 17:47.25 | jaytee | so ask |
| 17:47.27 | amiads | to see if it's the platform that fit my needs |
| 17:47.48 | amiads | I want to have a system that calls all of my contacts |
| 17:48.03 | amiads | and plays them a recorded message |
| 17:48.12 | *** join/#asterisk petchaw (n=petchaw@c-66-229-58-42.hsd1.fl.comcast.net) |
| 17:48.13 | amiads | in the time i choose |
| 17:48.34 | *** join/#asterisk telecos (n=sergio@84.166.219.87.dynamic.jazztel.es) |
| 17:48.47 | amiads | ( a bunch in the same time is prefarrable) |
| 17:48.56 | amiads | can I do it with asterisk? |
| 17:49.02 | jaytee | amiads, yes it's doable |
| 17:49.24 | amiads | what is the hardware needs if any? |
| 17:49.27 | rob0 | hopes he's not on amiads' contact list |
| 17:49.42 | amiads | can i do it over ip? |
| 17:49.48 | jaytee | you could use either a sql database on the local host or use SugarCRM which alot of people use with Asterisk. |
| 17:50.18 | jaytee | amiads, if you have an ITSP and enough accounts to handle that many concurrent calls, then yes. |
| 17:50.21 | amiads | sql is good for me... |
| 17:51.07 | ManxPower | *sniff* *sniff* I smell a telemarketer. |
| 17:51.20 | rob0 | rob0calling |
| 17:51.25 | amiads | actually i got that as a freelance job |
| 17:51.39 | jaytee | is that what that foul stench is? damn! I'm aiding and abbetting the enemy! |
| 17:51.43 | amiads | never been in telecom business before |
| 17:52.02 | ManxPower | amiads: Asterisk is not really a PBX. Asterisk is a TOOLKIT that lets you build a PBX. |
| 17:52.17 | jaytee | Telecom Legos |
| 17:52.30 | amiads | ok... |
| 17:52.38 | amiads | so what does it mean? |
| 17:52.43 | ManxPower | amiads: You should generally expect to spend about a month learning enough about Asterisk to so what you are wanting to do. |
| 17:52.47 | jaytee | amiads, for more info read this: |
| 17:52.49 | amiads | what do i need in order to build that pbx |
| 17:52.50 | jaytee | ~book |
| 17:52.51 | jbot | [book] Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com or see ~buybook |
| 17:52.51 | amiads | ? |
| 17:53.14 | amiads | yes i've started to read it |
| 17:53.35 | jaytee | another resource is the WIKI at voip-info.org |
| 17:53.37 | amiads | just asking here for reference in hours and possibilities |
| 17:53.58 | [TK]D-Fender | amiads: hours depends on you. Possibilities as well |
| 17:54.17 | [TK]D-Fender | amiads: My * makes me COFFEE. What you do with yours is up to you. |
| 17:54.28 | amiads | but you say that i can build such system (as explained above) in asterisk without needed hardware? |
| 17:54.51 | [TK]D-Fender | amiads: You need hardware if you want * to talk to the PST using a physical line you have. |
| 17:54.53 | jaytee | amiads, of course you need hardware. a server |
| 17:54.54 | Carlos_PHX | Yes |
| 17:54.56 | [TK]D-Fender | PSTN* |
| 17:55.26 | amiads | i don't mean a server - this I have |
| 17:55.29 | [TK]D-Fender | amiads: To get to the PSTN you either need local hardware or use something like a VoIP termination provier or ITSP |
| 17:55.31 | [TK]D-Fender | ~itsp |
| 17:55.31 | 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 |
| 17:55.35 | amiads | I mean a special telecom card |
| 17:55.48 | [TK]D-Fender | amiads: only if you want to use a physical LINE you have on-site |
| 17:56.25 | amiads | i don't |
| 17:56.31 | amiads | I prefer voip |
| 17:56.33 | petchaw | ~itsplist-us |
| 17:56.34 | jbot | [~itsplist-us] Here are some popular ITSPs (USA) starting with the more respected ones : http://www.teliax.com , http://connect.voicepulse.com , http://www.nufone.net , http://www.broadvoice.com, http://www.jnctn.com , http://www.bandwidth.com , http://vitelity.net |
| 17:56.47 | ManxPower | amiads: Dude, you don't even know enough to know what you prefer. |
| 17:56.49 | *** join/#asterisk hi365_m (n=hi365@213.151.36.235) |
| 17:56.49 | amiads | what are the servers in israel? |
| 17:57.25 | ManxPower | I don't know anything about wine but I prefer Chateau Noir 1952. |
| 17:57.29 | amiads | ~itsplist-il |
| 17:57.43 | ManxPower | amiads: those are the only 2 lists there are. |
| 17:57.51 | petchaw | ~itsplist-ca |
| 17:57.52 | jbot | [~itsplist-ca] Here are some popular ITSPs (Canada) starting with the more respected ones : http://www.unlimitel.com , http://www.les.net , http://www.babytel.ca |
| 17:58.00 | jaytee | and Canada |
| 17:58.07 | jaytee | but that's it I think |
| 17:58.11 | amiads | ok |
| 17:58.12 | [TK]D-Fender | jaytee: Off to a camera show.. back later :) |
| 17:58.20 | jaytee | [TK]D-Fender, have fun! |
| 17:58.26 | Carlos_PHX | Spend money! |
| 17:58.27 | amiads | i'm going to say the client i need about a month to build such system |
| 17:58.34 | [TK]D-Fender | Carlos_PHX: trying not to! |
| 17:58.44 | jaytee | he'll come back with a new toy to make me more jealous, I just know it :-) |
| 17:59.01 | jaytee | needs to find a better paying job :-( |
| 17:59.14 | ManxPower | amiads: It could be less, we really don't know how smart you are. But the fact that you accepted a job without even knowing how you will do the job does not bode well. |
| 17:59.29 | amiads | i havent |
| 17:59.39 | amiads | i just said i looked into it |
| 17:59.51 | amiads | i'm a fast learner |
| 18:00.09 | amiads | the 600 pages book will take me a week |
| 18:00.11 | jaytee | "Heck, yeah! I can build you a nuclear reactor! No problem!" "Pssst! Hey, what's this fission stuff about?" |
| 18:00.19 | amiads | cuz english is not my native language |
| 18:01.19 | ManxPower | amiads: most people here will be more generous if your native language is not english. That's not a problem. Stupidity knows no language. 8-) |
| 18:02.12 | ManxPower | amiads: the book is only the beginning. Don't be distracted by things like Asterisk Realtime or users.conf or anything that is not really required for the job. |
| 18:02.12 | amiads | My question is: After finishing the book is the knowledge in it enough to build such system? |
| 18:03.02 | amiads | ManxPower: so what in it I need? |
| 18:04.10 | carrar | should be |
| 18:04.10 | carrar | Your requirements were simple |
| 18:04.21 | carrar | <amiads> I want to have a system that calls all of my contacts |
| 18:04.21 | carrar | <amiads> and plays them a recorded message |
| 18:04.38 | amiads | yes |
| 18:04.41 | carrar | thats very easy |
| 18:05.07 | amiads | I'm a web programmer so if I could integrate it in a web enviorment will be gooddd |
| 18:05.48 | amiads | is this possible |
| 18:05.55 | amiads | ? |
| 18:06.10 | petchaw | i built a system for that already |
| 18:06.10 | ManxPower | amiads: anything is possible with Asterisk. |
| 18:06.24 | carrar | very easy |
| 18:06.26 | petchaw | i have a web interface that connects to asterisk and does that |
| 18:06.35 | ManxPower | petchaw: how long did it take you from know nothing about asterisk to building a web thingy? |
| 18:06.36 | carrar | especially if the web server is on the same box |
| 18:06.54 | amiads | it is |
| 18:07.09 | amiads | petchaw: can you show it to me? |
| 18:07.26 | amiads | <ManxPower> petchaw: how long did it take you from know nothing about asterisk to building a web thingy? |
| 18:07.35 | petchaw | took me about a month |
| 18:07.54 | petchaw | i started using asterisk in may |
| 18:07.58 | amiads | and your prior knowledge was nothing? |
| 18:08.08 | petchaw | and in july I did it |
| 18:08.18 | petchaw | it is pretty easy |
| 18:08.22 | petchaw | yep |
| 18:08.26 | petchaw | it was nothing |
| 18:08.41 | petchaw | first thing i did with asterisk was ivr applications |
| 18:08.52 | amiads | can you show me your system? |
| 18:08.55 | amiads | what is ivr? |
| 18:09.07 | amiads | I 'm really noob in the world of telcom |
| 18:09.11 | petchaw | interactive voice response |
| 18:09.19 | carrar | amiads: http://www.voip-info.org/wiki-Asterisk+auto-dial+out |
| 18:09.46 | amiads | i think i'll go ahead and read the book |
| 18:09.49 | petchaw | like you call your bank, an automated attendant answer, ask you to input your act number and pin code then read back your account balance |
| 18:09.52 | petchaw | stuff like that |
| 18:09.59 | carrar | pop a file in there when you want it to dialout and then have it play your recording |
| 18:10.19 | carrar | doesn't get any easier |
| 18:10.26 | *** join/#asterisk tkbeat (n=tk@p54B965CB.dip.t-dialin.net) |
| 18:10.30 | petchaw | i didnt use auto dial out function |
| 18:10.44 | petchaw | i didnt even know there was one |
| 18:10.48 | amiads | yeap but that's bring me back to my question |
| 18:10.49 | carrar | thats what he wants to do per his request |
| 18:10.49 | *** join/#asterisk shmaltz (n=chatzill@mail.dmaven.com) |
| 18:10.58 | amiads | is the book enough to build such systems? |
| 18:10.59 | shmaltz | wow, this is really good |
| 18:11.18 | carrar | <amiads> I want to have a system that calls all of my contacts |
| 18:11.18 | carrar | <amiads> and plays them a recorded message |
| 18:11.18 | carrar | <amiads> in the time i choose |
| 18:11.18 | carrar | <amiads> ( a bunch in the same time is prefarrable) |
| 18:11.18 | carrar | <amiads> can I do it with asterisk? |
| 18:11.19 | petchaw | as long as you know how call files work, your system just has to be able to create the file and make the calls |
| 18:11.26 | shmaltz | using a AMD K62 400MHZ with 128MB RAM to pump out 8000 calls, works like a charm |
| 18:11.45 | shmaltz | pechaw, who needs help with call files? |
| 18:11.51 | shmaltz | thats exaclty what I'm doing now |
| 18:11.56 | carrar | but read the Orielly Asterisk book |
| 18:12.01 | carrar | it's just good info to know |
| 18:12.07 | carrar | it will help you |
| 18:12.09 | amiads | i will |
| 18:12.13 | petchaw | you can have a database with all your contacts, the your web application would grab the numbers one by one and create the call file |
| 18:12.38 | shmaltz | petchaw, who you talking to? who needs help with call files? Thats exactly what I'm doingnow |
| 18:12.51 | amiads | so the way calls are maid here is by a file that is created and then executed in asterisk? |
| 18:13.16 | shmaltz | I have a VBScript that pools numbers from a text files, creates the call files then moves them over to asteirsk spool directory using pscp over ssh |
| 18:13.19 | petchaw | amiads wanted a system that that calls all his contacts and play a recorded message |
| 18:13.23 | carrar | amiads, yes |
| 18:13.31 | petchaw | so i told him he can do it with call files |
| 18:13.32 | shmaltz | amaids, you have Windows? |
| 18:13.45 | shmaltz | amaids, you have a working asterisk system? |
| 18:14.05 | amiads | nope |
| 18:14.09 | shmaltz | I can give you my scirpts |
| 18:14.13 | shmaltz | amaids, no for what? |
| 18:14.19 | amiads | i'm so new to the the term Asterisk |
| 18:14.24 | *** join/#asterisk tmiw_ (i=thinknot@cpe-66-75-245-86.san.res.rr.com) |
| 18:14.24 | amiads | just heard of today |
| 18:14.36 | carrar | It's awesome powerfull stuff |
| 18:14.40 | shmaltz | amaids, welcome, it's nothing really, it's just a symbol on your keyboard :) |
| 18:14.45 | amiads | i don't have a working station |
| 18:15.01 | shmaltz | amiads, ok where are your contacts? |
| 18:15.03 | amiads | yes glad i have you here |
| 18:15.10 | amiads | i was a web programmer |
| 18:15.14 | amiads | nothing to do with telephone |
| 18:15.23 | shmaltz | ok, I'll be posting the scripts on the wiki |
| 18:15.23 | amiads | SQL |
| 18:15.26 | amiads | MYSQL |
| 18:15.31 | carrar | perl/php? |
| 18:15.35 | amiads | php |
| 18:15.39 | shmaltz | amiads, you using windows? |
| 18:15.42 | carrar | You can write call plans in php |
| 18:15.46 | carrar | through AGI |
| 18:15.53 | shmaltz | but I don't know php/perl |
| 18:16.01 | amiads | unfortunately yes |
| 18:16.02 | carrar | I write mine in perl |
| 18:16.40 | amiads | is color allowed in this channel? |
| 18:17.06 | carrar | it's frowned |
| 18:17.21 | shmaltz | look at this baby: |
| 18:17.22 | shmaltz | http://pastebin.ca/1243344 |
| 18:17.24 | shmaltz | all with an AMD K62400Mhz and 128 MB machine |
| 18:17.33 | amiads | how come - it's easier reading that way |
| 18:17.53 | carrar | Using the wrong client if you are having a hard time reading |
| 18:18.08 | amiads | just mirc |
| 18:18.17 | amiads | what client do you suggest? |
| 18:18.27 | carrar | I use ScrollZ on my unix box |
| 18:20.07 | amiads | ok so to get things straight - I need to read the book and than i'll have enough knowledge in order to write the code for that system? |
| 18:20.44 | amiads | or the book is just general stuff? |
| 18:21.25 | carrar | I never read the book and I did it |
| 18:22.15 | Corydon76-dig | The book focuses on everything from the general concepts needed to understand the system all the way to the specifics of how to do things |
| 18:22.20 | carrar | If you are smart you can figure it out |
| 18:22.27 | Daejeo | hello guys, i made a phone verification system for eliminating anonymous web registration/feedback abuse, want to evaluate the performance> i would appreciate if few people can test it. |
| 18:22.43 | Daejeo | http://w2.pcu.ac.kr/~singh/papers/abstract.pdf |
| 18:22.47 | amiads | ok |
| 18:22.53 | amiads | thanks a lot you guys |
| 18:22.58 | amiads | helped me big time |
| 18:23.07 | amiads | ==]] |
| 18:23.19 | amiads | Kudos 2 u |
| 18:23.34 | Corydon76-dig | What's that supposed to be, ASCII art of a penis? |
| 18:24.17 | jaytee | and a square headed penis no less! |
| 18:24.29 | Corydon76-dig | No, wait, it's a rocket ship! |
| 18:26.56 | amiads | <PROTECTED> |
| 18:27.29 | Corydon76-dig | That's a rather elongated head |
| 18:29.19 | jaytee | long headed guy with double chin? |
| 18:29.46 | Corydon76-dig | and four eyes? |
| 18:30.09 | jaytee | possibly |
| 18:30.28 | Corydon76-dig | Purple People Eater? |
| 18:30.34 | *** join/#asterisk talntid (n=eric@c-67-185-239-175.hsd1.wa.comcast.net) |
| 18:30.45 | jaytee | that only had one eye |
| 18:31.01 | Corydon76-dig | 4 Purple People Eaters! |
| 18:31.16 | jaytee | with double chins, lol |
| 18:32.20 | jaytee | just one more week and I will be making my Hajj to the Mecca of OSS VOIP, Huntsville, AL. woo-hoo!!! |
| 18:33.03 | Corydon76-dig | bootcamp? |
| 18:33.15 | jaytee | Advanced Asterisk |
| 18:33.29 | Corydon76-dig | Ah, I should take that class sometime |
| 18:33.58 | Corydon76-dig | Make myself certifiable |
| 18:34.02 | petchaw | will you take the dCAP exam? |
| 18:34.15 | jaytee | figured since I've been using * for almost 2 years now and already have a server in production with 34 clients I'd just skip the Bootcamp class. |
| 18:34.51 | jaytee | petchaw, I'm still deciding on that. I talked to Jan at Digium and she said I could wait until the week of the class to decide. |
| 18:35.02 | petchaw | yea true |
| 18:35.23 | ManxPower | jaytee: As I understand it, Bootcamp is no longer done. The course work was split into 2 classes, Fast Start and Advanced. |
| 18:35.24 | petchaw | they dont offer the bootcamp anymore though |
| 18:35.25 | jaytee | and it's not exactly as in demand as an MCSE or other certifications. |
| 18:35.31 | petchaw | it is only asterisk advance now |
| 18:35.37 | jaytee | ManxPower, that makes sense |
| 18:37.04 | petchaw | if you gonna be doing work on asterisk, or work for an itsp, the dCAP is a big plus |
| 18:37.20 | petchaw | do you know who will be teaching the class? |
| 18:37.32 | jaytee | petchaw, Jared I believe |
| 18:37.40 | petchaw | oh ok |
| 18:37.51 | petchaw | you gonna be in good hands then |
| 18:37.52 | jaytee | and hopefully ManxPower will be assisting |
| 18:38.27 | ManxPower | jaytee: I need to e-mail them tomorrow to confirm |
| 18:39.12 | jaytee | ManxPower, I hope you end up assisting. That'll make for a killer class. |
| 18:40.14 | jaytee | plus it will give me an opportunity that's long overdue. I figure I owe you several beers at least for all the help you've been over the last couple years. |
| 18:44.26 | petchaw | ~did |
| 18:44.26 | jbot | extra, extra, read all about it, did is Direct Inward Dialing, or just a phone number |
| 18:45.03 | petchaw | ~lumenvox |
| 18:45.09 | petchaw | ~asterisk |
| 18:45.10 | jbot | i heard asterisk is a free PBX, or #asterisk on irc.freenode.net, or http://www.asterisk.org, or just like a mini-mall |
| 18:45.22 | petchaw | lol |
| 18:45.49 | *** join/#asterisk steliosk-laptop (n=stelios@ipa107.2.tellas.gr) |
| 18:46.07 | jaytee | petchaw, you use LumenVox? |
| 18:50.59 | petchaw | i am trying it |
| 18:51.37 | petchaw | I have set it up and everything |
| 18:51.53 | *** join/#asterisk kamanashisroy (n=kamanash@202.56.7.183) |
| 18:51.58 | petchaw | but i have another problem with my box that i am trying to fix |
| 18:52.45 | jaytee | petchaw, we have an speech recognition IVR from Liaison (now owned by Nuance) for our Nortel PBX. I'm building a replacement using Lumenvox on Asterisk. So far so good. |
| 18:54.56 | petchaw | cool |
| 18:58.39 | file | jaytee: are you loading grammars in the dialplan? |
| 18:59.05 | jaytee | file yes I am |
| 18:59.27 | file | wonders why people do that |
| 18:59.58 | jaytee | file, do you recommend loading all grammars prior and then just activating and deactivating as needed? |
| 19:00.03 | file | yes |
| 19:00.38 | file | because if you don't unload then you will potentially leak memory on every call, and it also takes a bit of time to compile the grammar internally and set it up versus only activating |
| 19:00.42 | *** join/#asterisk tkbeat (n=tk@p54B965CB.dip.t-dialin.net) |
| 19:01.05 | file | and on a loaded system this can add up and cause delays |
| 19:02.39 | petchaw | be right back |
| 19:02.40 | rob0 | And with a loaded system operator, you get silly mistakes. |
| 19:06.35 | jaytee | file, could you refresh my tired and aging old brain and tell me where I preload the grammars? is it in lumvenvox.conf? |
| 19:06.43 | file | jaytee: yes. |
| 19:07.44 | *** join/#asterisk protocols (n=protocol@ip-88-153-205-180.unitymediagroup.de) |
| 19:08.08 | jaytee | ah, I'm going to try that then. it makes sense. I've got all my grammars activating in each section of the IVR as it's called and deactivating after a speech result but eliminating delays is definitely a good thing :-) |
| 19:09.39 | drmessano | file: Has any of the G726 behavior changed in 1.6 in regard to support for aal2 and the config options? |
| 19:10.04 | file | drmessano: nope? |
| 19:10.21 | drmessano | Just an open question... not a leading one |
| 19:10.53 | drmessano | I was getting back to "I want to play around with G726" the other day, and had documented a convo we had |
| 19:11.06 | drmessano | had to relearn the workarounds lol |
| 19:14.30 | *** join/#asterisk TOrrIeri (n=torrieri@nelug/crew/torrieri) |
| 19:16.13 | *** part/#asterisk LemensTS (i=LemensTS@adsl-70-238-149-98.dsl.stlsmo.sbcglobal.net) |
| 19:22.19 | *** join/#asterisk tkbeat (n=tk@p54B9BF3B.dip0.t-ipconnect.de) |
| 19:22.47 | *** join/#asterisk Defraz (n=T0tal@24-117-156-21.cpe.cableone.net) |
| 19:43.54 | *** join/#asterisk tkbeat (n=tk@p54B9BF3B.dip0.t-ipconnect.de) |
| 19:47.07 | *** join/#asterisk pikachu2000 (n=pikachu2@196-209-199-191-rrdg-esr-2.dynamic.isadsl.co.za) |
| 19:55.08 | *** join/#asterisk hunmonk_ (n=hunmonk@drupal.org/user/22079/view) |
| 19:55.58 | hunmonk_ | can anybody tell me exactly what the 'Refresh' column in the output of "sip show registry" means? |
| 19:57.50 | *** join/#asterisk elvisthedj (n=kris@67.61.125.156) |
| 19:58.22 | [netman] | how often checks whether the peer is on-line |
| 19:58.33 | mchou | in seconds |
| 19:58.37 | mchou | :) |
| 19:58.41 | elvisthedj | Can someone tell me why these statements are evaluating as false? http://pastebin.com/m10fb3145 |
| 19:59.28 | elvisthedj | i've tried with quotes, without, with quotes only on the string (not around the variable).. always false |
| 20:00.02 | hunmonk_ | [netman]: is that a setting somewhere? b/c my output changed recently when i made changes to my registration strings. Refresh used to be 105, and now it's 3585 -- i think that's like 5 days in seconds... :| |
| 20:03.03 | mchou | hunmonk_: you need to do some math. 1hr. has 3600 seconds |
| 20:03.15 | mchou | hunmonk_: that's hardly 5 days |
| 20:03.42 | [netman] | hunmonk_: you can change the refresh time in sip.conf |
| 20:04.37 | mchou | hunmonk_: and 1 hr. is fine as long as your firewall doesnt time out on whatever sip port you're using |
| 20:04.47 | elvisthedj | 3600 secs feels like 5 days when you're hoping someone will look at your pastebin :) |
| 20:05.07 | hunmonk_ | mchou: you're right on the math. |
| 20:05.21 | hunmonk_ | dunno what i missed originally there |
| 20:05.37 | mchou | lol |
| 20:05.42 | hunmonk_ | elvisthedj: i checked it, i'm not seeing anything out of the ordinary |
| 20:06.28 | elvisthedj | hunmonk_: Well, as you can see the caller id name and num were both anonymous... one of the two statements should have caught it... but they evaluate as false |
| 20:06.31 | hunmonk_ | elvisthedj: have you tried NoOp(${CALLERID(name)}), etc to make sure there's something in those vars? |
| 20:06.53 | elvisthedj | hunmonk_: look at the first statement :) |
| 20:07.13 | elvisthedj | exten => s,n,NoOp(Caller ID Name was ${CALLERID(name)} and number was ${CALLERID(num)}) |
| 20:07.13 | hunmonk_ | elvisthedj: ah, right |
| 20:07.26 | elvisthedj | <PROTECTED> |
| 20:07.35 | elvisthedj | :( |
| 20:08.26 | hunmonk_ | elvisthedj: i dunno. i use AEL, and never have issues w/ evaluating expressions |
| 20:08.50 | hunmonk_ | elvisthedj: i use the surrounding quotes in AEL |
| 20:09.17 | hunmonk_ | elvisthedj: like if ( "${foo}" = "bar" ) { |
| 20:09.31 | hunmonk_ | elvisthedj: so what you're doing looks right to me |
| 20:09.46 | elvisthedj | yeah.. me too.. but it's not working |
| 20:09.54 | elvisthedj | somethin stupid i'm doing no doubt |
| 20:10.29 | hunmonk_ | elvisthedj: try removing the spaces around the = |
| 20:10.36 | elvisthedj | i know it's been discussed in the bugs, but i think privacymanager should have the option to ignore Unavailable, Private and Anonynmous |
| 20:10.43 | elvisthedj | those could never be useful... |
| 20:10.51 | Katty | mmm, ice cream. |
| 20:11.00 | *** join/#asterisk zchaos (n=none@CPE0080c828f609-CM0019474d28a8.cpe.net.cable.rogers.com) |
| 20:11.19 | elvisthedj | hunmonk_: I'm pretty sure i've tried w/o spaces, but i'll try it again :) |
| 20:11.34 | zchaos | my voip provider (acana) sent me my voip details but i'm not sure how to plug them into the asterisk box to start accepting calls... can anyone tell me how? |
| 20:11.49 | hunmonk_ | elvisthedj: i know at times the parser can be finnicky, that's all i really can see to try after looking at your code |
| 20:12.25 | elvisthedj | hunmonk_: thanks :) i'll just play around i guess til i find something it likes |
| 20:12.49 | elvisthedj | zchaos: if your provider is sip, search the wiki (voip-info.org) for sip.conf |
| 20:13.21 | hunmonk_ | elvisthedj: fwiw, i can't imagine doing a regular dialplan after having worked w/ AEL now |
| 20:14.15 | elvisthedj | hunmonk_: My dial plan is 1053 lines.. i just can't imagine taking the time to switch to ael |
| 20:14.41 | elvisthedj | hunmonk_: i know that's not a lot of lines for a big company.. but for me.. it's a lot |
| 20:15.02 | zchaos | elvisthedj: i think its sip, how do i confirm? |
| 20:15.15 | hunmonk_ | [netman]: defaultexpirey <-- is that the sip.conf setting you're referring to set the refresh time? |
| 20:15.17 | elvisthedj | zchaos: it's most likely sip |
| 20:16.02 | zchaos | elvisthedj: sip oppose to what |
| 20:16.26 | jaytee | elvisthedj, are your incoming calls from the PSTN or SIP from an ITSP? |
| 20:16.33 | elvisthedj | zchaos: as opposed to IAX .. among others |
| 20:16.37 | elvisthedj | jaytee: sip |
| 20:16.43 | jaytee | ah, ok. |
| 20:16.56 | elvisthedj | jaytee: y?? something to do with my pastebin? |
| 20:17.32 | jaytee | no, just curious because with my system we have PRI and to extract CID I have to use callerid(ani) |
| 20:18.36 | [netman] | hunmonk_: yes It is |
| 20:18.38 | jaytee | although it looks like you're referencing a labeled priority that isn't in that block of code anywhere. whoru. is that supposed to jump to priority 8? |
| 20:18.47 | zchaos | elvisthedj: so how do i link my voip provider (sip) with asterisk.... |
| 20:19.45 | jaytee | zchaos, pages 97 to 104 of the book |
| 20:20.19 | jaytee | zchaos, and if you're behind a NAT'd firewall you'll want to read this too. |
| 20:20.22 | elvisthedj | jaytee: whoru is a different context.. i didn't think i needed the exten/priority if it's s,1 .. but still, the expressions are coming up false |
| 20:20.22 | jaytee | ~sipnat |
| 20:20.23 | 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 |
| 20:20.54 | zchaos | jaytee thanks i'll check it out |
| 20:20.59 | zchaos | i'm only using a router |
| 20:21.01 | zchaos | no nat |
| 20:21.03 | jaytee | elvisthedj, and NoOp is returning a null? |
| 20:21.36 | elvisthedj | jaytee: Nope.. the variables are set. it's returning Anonymous for the name and anonymous for the number |
| 20:21.56 | elvisthedj | exten => s,n,NoOp(Caller ID Name was ${CALLERID(name)} and number was ${CALLERID(num)}) |
| 20:22.02 | *** join/#asterisk tndev (n=tndev@196.203.35.178) |
| 20:22.09 | tndev | Hi |
| 20:22.10 | elvisthedj | returns "Caller ID Name was Anonymous and number was anonymous") |
| 20:22.53 | tndev | i have gnudialer and asterisk 1.4.18, i had applied the patchs for channel.c and manager.c but i have always the problem of transfer of calls |
| 20:22.54 | elvisthedj | so, GotoIf($["${CALLERID(name)}" = "Anonymous"]? should return true |
| 20:23.00 | tndev | agents dont get the calls |
| 20:23.52 | tndev | the log of gnudialer says "Tranfering - to agent 300" ==> the call is not asign to a channel |
| 20:24.05 | jaytee | elvisthedj, try it without the quotes around ${callerid(name)} |
| 20:24.06 | tndev | plz if any one know this bug, can you help ME |
| 20:24.16 | *** join/#asterisk bijit (n=benji@200.122.188.156) |
| 20:24.23 | elvisthedj | jaytee: ok, here goes :) |
| 20:24.36 | jaytee | elvisthedj, and the number as well |
| 20:25.44 | elvisthedj | jaytee: just the varialble, right.. leave the quotes around "Anonymous" |
| 20:25.55 | tndev | PLZ ANY ONE KNOW GNUDIALER ?? |
| 20:26.36 | jaytee | elvisthedj, yes, just the variable but wait.....first try moving the first " in the caller id variable BEFORE the $[ and try that first. |
| 20:26.37 | elvisthedj | tndev: I don't, but it looks like they have a channel |
| 20:26.41 | drmessano | ZOMG ALL CAPS MAKES ME KNOW IT NOW |
| 20:27.13 | jaytee | tndev, I'VE NEVER USED GNUDIALER, SORRY! BEST OF LUCK!!!! |
| 20:27.22 | tndev | thnks jaytee |
| 20:27.33 | jaytee | yw, tndev |
| 20:27.45 | tndev | jaytee just for the answer :) |
| 20:28.12 | jaytee | tndev, might be something on the WIKI at voip-info.org. fwiw alot of the stuff there is out of date though. |
| 20:28.17 | tndev | at least u answred me, even its a negative answer but always an answer better then nothind |
| 20:28.19 | tndev | nothing |
| 20:28.30 | drmessano | http://tinyurl.com/gnudialer |
| 20:28.33 | elvisthedj | Join us in #gnudialer on freenode.net. |
| 20:28.49 | tndev | in #gnudialer theres no one connected |
| 20:30.12 | jaytee | tndev, if no one answered you immediately it means they probably have no experience with gnudialer and typing in all caps is considered shouting and rude manners. especially if you only wait 3 minutes for an answer. |
| 20:30.39 | drmessano | I show 2 mins here |
| 20:30.45 | tndev | just a question, i wana know at least what kind of modules can be the cause of this probleme, i mean for example is zaptel can be one of the cause |
| 20:30.54 | drmessano | WE DONT KNOW |
| 20:31.53 | *** part/#asterisk Carlos_PHX (n=Carlos@ip68-3-162-244.ph.ph.cox.net) |
| 20:32.10 | drmessano | Gnudialer sounds like a metering device used to determine the shears needed to castrate a bull... Beyond that, i'm bunnypancake on the subject |
| 20:32.17 | *** join/#asterisk Carlos_PHX (n=Carlos@ip68-3-162-244.ph.ph.cox.net) |
| 20:32.31 | jaytee | hehehehe, bunnypancake |
| 20:33.08 | hunmonk_ | [netman]: lemme see if i have this straight. if Refresh is 1 hour, and the server i'm registered to tanks, it's possible the output of "sip show registry" might falsely show me as registered for up to an hour? |
| 20:40.25 | edoceo | tndev: See #gnudialer |
| 20:42.51 | elvisthedj | I hate when i get stuck with something so dumb.. I guess I will use an agi to evaluate this since i can't figure out gotoif |
| 20:43.22 | elvisthedj | here's an update of all i've tried if anyone has a thought http://pastebin.com/m3e05e23 |
| 20:43.27 | jaytee | elvisthedj, trying both of those failed? |
| 20:44.04 | elvisthedj | jaytee: yep.. i have other gotoif's in my dialplan that work fine.. so i don't know what's up here |
| 20:44.24 | drmessano | Are you Elvis Duran from Z100? |
| 20:45.00 | elvisthedj | drmessano: No, I'm way better than that guy :p |
| 20:45.16 | drmessano | Do you know who he is? |
| 20:45.23 | elvisthedj | drmessano: Yep |
| 20:45.42 | drmessano | Do you have a morning zoo? |
| 20:46.13 | jaytee | elvisthedj, your callerid(num) is returning anonymous in lower case but your test condition is looking for uppercase. Might make a diff, but the name one returns a cap so it should at least match on that as true first. |
| 20:46.18 | elvisthedj | jaytee: I've got an idea.. since i seem to be able to successfully evaluate expressions when i'm comparing to varialbes, I'll just set up a global with the value of Anonymous and compare to that |
| 20:46.39 | elvisthedj | jaytee: I've tried both cases.. the name is "A", the num is "a" |
| 20:47.02 | elvisthedj | drmessano: I did.. but I set them free |
| 20:47.41 | jaytee | elvisthedj, but your GotoIf is still looking for Anonymous for the callerid(num) |
| 20:47.47 | [netman] | hunmonk_: you are right |
| 20:48.43 | *** join/#asterisk lesouvage (n=lesouvag@cc341200-a.assen1.dr.home.nl) |
| 20:48.45 | [TK]D-Fender | jaytee: Got a 62mm Circ Polariser @ $50, and a body cap & extra lens cap for $3 |
| 20:48.48 | [TK]D-Fender | jaytee: Not bad... |
| 20:49.24 | jaytee | see! see! I told you all he'd come back with a new toy just to make me jealous. :-) |
| 20:50.26 | [TK]D-Fender | elvisthedj: The priorities you show us in your dialplan do not match the CLI execution. Frankly I'd distruct that entire pastebin. Go show us something consistent and sane. In that PB you also has a "n" as the first one listed followed by a #2 *** WTF** |
| 20:50.51 | jaytee | elvisthedj, in Revision 1 you put the leading quote in between the $ and the [ bracket, not in front of the $[ |
| 20:51.30 | lesouvage | If a system is running well on 1.4.18.1 is there an urgent reason to move to 1.4.22. I tried to find some readable change logs but it seems that I don't know where to look. |
| 20:52.07 | [TK]D-Fender | lesouvage: Blatantly obvious on the HTTP file server... |
| 20:52.19 | jaytee | lesouvage, if it ain't broke don't fix it and I don't think there's anything compelling between the two minor versions. |
| 20:52.42 | lesouvage | jaytee: thanks |
| 20:53.32 | elvisthedj | [TK]D-Fender: That's because I just stuck the NoOp in between 1 and 2 to test the variables that i am evaluating.. Isn't that the point of "n" priority?? so you don't have to renumber everything.. |
| 20:53.45 | jaytee | to [TK]D-Fender it's blatantly obvious. To people like me it's more like "Where the hell are my car keys???" looks down at what he's holding in his hand, "Oh! duh!" |
| 20:54.08 | [TK]D-Fender | elvisthedj: You can't follow "n" with HARD priorities like that |
| 20:54.09 | jaytee | oh, jesus I missed that. dammit |
| 20:54.16 | [TK]D-Fender | elvisthedj: You are running over yourself |
| 20:54.19 | jaytee | sorry elvisthedj |
| 20:54.32 | [TK]D-Fender | elvisthedj: You probably killed the only one of those that would WORK |
| 20:55.12 | elvisthedj | [TK]D-Fender: Ok.. I'll fix the priorities.. which expression should i use? |
| 20:55.21 | [TK]D-Fender | jaytee: Captain Obvious strikes again! |
| 20:55.46 | jaytee | [TK]D-Fender, is it the priority 2 that has the "'s in the right place? |
| 20:55.59 | lesouvage | [TK]D-Fender: could you and are you willing to be a little bit more specific? Is there just the technical oriented changlog or is there somewhere on this particular HTTP fiel server (I assume you ment www.asterisk.org) a document with change info for non developers? |
| 20:56.02 | [TK]D-Fender | elvisthedj: Tell you what... go clean up your mess NOW and go see. |
| 20:56.16 | elvisthedj | [TK]D-Fender: What doesn't make sense to me is that the gotoif's are running, but returning false |
| 20:56.36 | elvisthedj | [TK]D-Fender: it's not skipping over.. but i'll go renumber everything and put up a new paste .. just for you |
| 20:56.51 | [TK]D-Fender | lesouvage: http://downloads.digium.com/pub/asterisk/ |
| 20:57.00 | [TK]D-Fender | lesouvage: read the changelog file(s) |
| 20:57.20 | [TK]D-Fender | lesouvage: Some detail is technical, though most of even those is clear as to it impact. |
| 20:59.56 | lesouvage | [TK]D-Fender: thanks, it is kind of hard to find out if it is just an adjustment/improvement or a kind of critical security/stablility fix. I will read the info. |
| 21:00.51 | [TK]D-Fender | lesouvage: thats all it is usualy. New features make it to the new version unless they sneak in with another fix. |
| 21:02.12 | *** part/#asterisk hunmonk_ (n=hunmonk@drupal.org/user/22079/view) |
| 21:12.35 | protocols | I get: pp_dial.c:1242 dial_exec_full: Unable to create channel of type 'Zap' (cause 0 - Unknown on 1.4.22 is this some zaptel/dahdi issue? |
| 21:12.41 | protocols | did not have this with 1.4.21 |
| 21:12.46 | *** join/#asterisk Steve_J-obs (n=Chris123@ip70-173-64-212.lv.lv.cox.net) |
| 21:12.58 | jaytee | off to have some gumbo and watch some football, be back later |
| 21:14.36 | zchaos | can anyone here help me with a small asterisk project/setup? I already have asterisk installed and up and running |
| 21:15.10 | *** join/#asterisk rcy`` (n=rcy@S01060002553240a8.vc.shawcable.net) |
| 21:16.40 | zchaos | let me know what you want in return... |
| 21:16.43 | [TK]D-Fender | protocols: Which are you using now? |
| 21:16.53 | [TK]D-Fender | zchaos: Details would help. |
| 21:16.58 | protocols | 1.4.22 |
| 21:17.06 | zchaos | fender... all i did waas install asterisknow |
| 21:17.06 | protocols | with latest zaptel |
| 21:17.18 | protocols | similar to this: http://www.asteriskguru.com/board/channel-not-implemented-vt3654.html |
| 21:17.26 | protocols | or is the asterisk-gui broken for 1.4.22? |
| 21:18.10 | hi365_m | anyone running polycoms with 3.1? |
| 21:19.54 | [TK]D-Fender | protocols: would help if you showed the configs, actual CLI output, proof that your device's driver is loaded, that ztcfg checks out ok, etc |
| 21:20.01 | *** join/#asterisk xchaos (n=none@CPE0080c828f609-CM0019474d28a8.cpe.net.cable.rogers.com) |
| 21:20.04 | Katty | blehs. |
| 21:20.08 | xchaos | sorry i disconnected fender... |
| 21:20.08 | xchaos | fender... all i did waas install asterisknow, i need to some how setup my VOIP provider on asterisknow, and setup a dialplan |
| 21:20.22 | [TK]D-Fender | xchaos: GUI's not supported here. |
| 21:20.33 | Katty | [TK]D-Fender: fun times, eh? |
| 21:20.42 | [TK]D-Fender | Katty: Keeps getting better... |
| 21:20.45 | xchaos | well maybe someone else can help me |
| 21:21.08 | protocols | zttool says "ok", module is loaded - it is actually pretty the same setup as 1.4.21 (which worked) |
| 21:21.27 | [TK]D-Fender | protocols: Less talk, more show... |
| 21:21.33 | protocols | :D |
| 21:22.37 | protocols | ztcfg: http://pastebin.ca/1243480 |
| 21:22.42 | *** join/#asterisk cmslaght (n=cmslaght@mail.issohio.com) |
| 21:22.50 | *** join/#asterisk tkbeat (n=tk@p54B965CB.dip.t-dialin.net) |
| 21:23.24 | Steve_J-obs | Hello Everybody! |
| 21:23.36 | Katty | hi |
| 21:23.49 | Steve_J-obs | hi Kathy |
| 21:24.07 | drmessano | lol |
| 21:24.08 | lesouvage | xchaos: have you tried #asterisknow for help? |
| 21:24.28 | protocols | http://pastebin.ca/1243481 <- the error log when dialing from sip to pstn |
| 21:24.54 | *** join/#asterisk voxter (n=voxter@S01060016b6b53c0c.vc.shawcable.net) |
| 21:25.41 | protocols | lsmod: http://pastebin.ca/1243482 |
| 21:27.53 | protocols | hm ok |
| 21:28.02 | protocols | [Nov 2 22:02:11] ERROR[2041] chan_dahdi.c: Unable to load zapata.conf |
| 21:28.02 | protocols | [Nov 2 22:02:11] ERROR[2041] codec_dahdi.c: Failed to open /dev/zap/transcode: No such file or directory |
| 21:28.10 | protocols | that does not look right.. |
| 21:28.25 | *** part/#asterisk drew (i=drew@whitehat.org) |
| 21:29.07 | hardwire | protocols: you, sir, need to load some modules |
| 21:29.16 | hardwire | and make some files |
| 21:29.27 | protocols | hm what modules are missing? |
| 21:29.30 | hardwire | and read the INSTALL |
| 21:29.53 | [TK]D-Fender | protocols: you are MIXING zaptel & DAHDI.. You should not be doing that |
| 21:30.24 | protocols | I thought 1.4.22 was compatible with zaptel modules + naming? |
| 21:31.12 | [TK]D-Fender | protocols: you run one or the other in their entirely, not both |
| 21:31.27 | protocols | hmm where do you see I am running both? |
| 21:32.48 | [TK]D-Fender | protocols: # ztcfg -vv <- this sure as shit ain't DAHDI, nor is 1] Dial("SIP/6000-08cb8788", "Zap/g1/0285515746") in new stack |
| 21:33.15 | protocols | yes thats true |
| 21:33.31 | [TK]D-Fender | protocols: protocols read the docs, use the new configs. |
| 21:33.44 | protocols | but I read in the "Dahdi-zaptel.txt" that I should be able to use old zaptel with 1.4.22 without any probs |
| 21:33.52 | [TK]D-Fender | Some people wouldn't get consistency if it ran up and bit them in the face. |
| 21:34.26 | protocols | yes but what you quoted up is consistent usage of zap, or not? |
| 21:34.32 | hardwire | [TK]D-Fender: most snakes don't run. |
| 21:34.34 | protocols | thats what I am actually trying.. |
| 21:34.36 | [TK]D-Fender | protocols: One REPLACES the other, hybridize them and you're bound to mess stuff up. |
| 21:34.47 | protocols | yep I know.. I want to stay with zap ;) |
| 21:35.03 | protocols | and I thought I did everything.. there is nothing dahdi in my config |
| 21:35.12 | [TK]D-Fender | protocols: Zaptel is gone. Get over it. Read the docs and ditch the old zaptel stuff. |
| 21:35.28 | [TK]D-Fender | protocols: You need chan_fluxcapacitor.so then |
| 21:35.36 | protocols | doh.. but dahdi does not work with asterisk-gui |
| 21:40.12 | *** join/#asterisk DagMoller (n=aguirre@unaffiliated/dagmoller) |
| 21:40.15 | *** join/#asterisk ManxPower (n=manxpowe@15.sub-70-223-43.myvzw.com) |
| 21:40.33 | jjshoe | re |
| 21:40.34 | *** part/#asterisk DagMoller (n=aguirre@unaffiliated/dagmoller) |
| 21:45.24 | *** join/#asterisk esaym (n=user@cpe-70-120-89-6.satx.res.rr.com) |
| 21:54.17 | *** join/#asterisk Zhad (n=tom@cpc4-sout6-0-0-cust165.sotn.cable.ntl.com) |
| 21:56.14 | Zhad | I have 2 lines going into a TDM400P, one BT, one Bulldog. |
| 21:56.24 | Zhad | The BT one detects hangups and the bulldog one doesn't |
| 21:56.36 | Zhad | they probably use different signalling methods. |
| 21:56.55 | Zhad | (With zaptel) Can the signalling method beconfiguredd as different for each card? |
| 21:57.56 | xchaos | anyone here interested in helping with a small asterisknow project? I have the system all setup and ready to go... i'll provide details if interested.... msg me if your interested and what it will cost |
| 22:02.22 | drmessano | Anyone have provisioning info on Linksys RTP300 boxes? |
| 22:03.13 | write__erase | Hi, Is there a way to execute commands (inserts in database) when DHCPd offers a lease ? I'd like to have MAC:IP mapping in a database so I can reboot the phones if required . thx |
| 22:09.42 | *** join/#asterisk freakazoid0223 (n=mattc@pool-68-238-187-201.phil.east.verizon.net) |
| 22:15.46 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 22:17.53 | *** join/#asterisk Dovid (n=annon@tony09-118-62.inter.net.il) |
| 22:20.00 | *** join/#asterisk joobie (n=joobie@201.023.dsl.mel.iprimus.net.au) |
| 22:27.17 | [TK]D-Fender | write__erase: Wrong channel... try in ##linux or something... |
| 22:27.45 | write__erase | thx |
| 22:29.05 | Dovid | a bit OT but does anyone know where I can start if I want to write a VB app to register as a sip user and subscribe an extension status so i can see who is ringing, on a call etc. |
| 22:29.06 | Dovid | ? |
| 22:30.50 | [TK]D-Fender | Dovid: You don't... |
| 22:30.55 | Dovid | ;) |
| 22:31.00 | [TK]D-Fender | Dovid: Wrong tool for the job. |
| 22:31.03 | Dovid | i know after today..... |
| 22:31.16 | Dovid | TK: What do you suggest ? |
| 22:31.32 | Dovid | I want to have something that users can install on their desktops |
| 22:31.34 | [TK]D-Fender | Dovid: AMI is what you'd use to get system onfo out of *. |
| 22:31.42 | [TK]D-Fender | Dovid: Phones know nothing by comparison |
| 22:31.53 | [TK]D-Fender | Dovid: They're jsut phones after all |
| 22:32.18 | Dovid | TK: I need to work off subscribe since its working off OpenSER with a custom "BLF" script so I cant use the AMI |
| 22:32.31 | write__erase | record some sip communications with wireshark or something like that, then write your client with by hand (open TCP/UDP connexion , read, write ...) boring job though... There might be some SIP Client libs, but none for V.B. ! |
| 22:32.51 | Dovid | weite_rease: was my last resort but I may do that |
| 22:32.55 | [TK]D-Fender | Dovid: Oh well... Best of luck with that :) I'm sure you can google up some VB SIP code if you try hard enough |
| 22:33.06 | joobie | guys is there a table or something that gives rough guides to which processors and which codecs can handle X amount of calls at a given time? |
| 22:33.18 | [TK]D-Fender | joobie: No. |
| 22:33.55 | Dovid | TK: been googlin for a bit. |
| 22:34.04 | Dovid | will see what happens. looks like its wireshark time |
| 22:34.04 | [TK]D-Fender | dovthats what "more" means. |
| 22:34.09 | joobie | Fender, my dillema is i have a single processor.. can upgrade it to dual processor but not sure if it's necessary to handle the calls iw ant |
| 22:34.12 | Dovid | lol |
| 22:34.15 | write__erase | Perl : http://search.cpan.org/~sullr/Net-SIP-0.50/lib/Net/SIP/Simple.pod |
| 22:34.19 | joobie | or not sure how much improvement in terms of the number of calls it will give me |
| 22:34.25 | joobie | any idea how to go about tackling this? |
| 22:34.29 | [TK]D-Fender | joobie: And you are capable of everything but actually describing your NEEDS it seems |
| 22:34.49 | [TK]D-Fender | write__erase: If only Perl were VB... |
| 22:35.40 | lesouvage | joobie: just discribe how many calls, what codecs, recording or not and in what format, conferences etc. etc. |
| 22:35.44 | write__erase | Writing a simple SIP client in VB should not be that long in fact ... open socket, send invite packet ... |
| 22:36.10 | Dovid | write_erase: thanks. i am going to stat off with that and see where it goes |
| 22:36.21 | joobie | Fender, im putting together a voip server to handle multiple sip calls.. not sure how many calls it will support at the moment, but i have an option to upgrade the cpu to dual processor.. just wondering if i do, how many more calls it will handle.. codecs i want to use are either g729 or ulaw.. no call recording for the moment.. conference call support |
| 22:36.29 | lesouvage | people here do have some professional guessing skills and they might want to hare there thoughts with you. It is inpossible to make hard calcualations |
| 22:37.01 | *** join/#asterisk Nasra (n=maxshipp@CPE001217b1920e-CM00159a010eda.cpe.net.cable.rogers.com) |
| 22:37.36 | Dovid | write_erase: may i PM ? |
| 22:37.39 | joobie | not after hard figures though les.. just trying to get a ball park |
| 22:37.53 | joobie | like a single processor with g729 can handle 30-50 calls.. dual processor 80-90 .. |
| 22:37.54 | lesouvage | joobie: there is a big difference in cpu load between g729 (compressed heavely) and ulaw (uncompressed) |
| 22:37.55 | [TK]D-Fender | joobie: Wow.. everything but a relevent number. No further ahead... |
| 22:38.07 | joobie | or is it more like single process handling 10-15 calls.. dual with 30-40.. |
| 22:38.31 | joobie | Fender, i dont know what info you need to determine this man.. |
| 22:38.44 | joobie | if you can ask me what you need for this it would be better.. have no idea |
| 22:38.52 | joobie | i'm no xpert |
| 22:39.21 | jksM | joobie, if you want an answer, reverse the question... ask how many concurrent calls other people have been able to sustain on their single cpu servers |
| 22:39.22 | lesouvage | joobie: it is always a trade off between bandwidth needed and number of calls the server can handle. |
| 22:39.38 | jksM | joobie, then afterwards, try to guess if they have a 350 Mhz Celeron Mobile or a 3,6 Ghz Xeon Quad-Core |
| 22:40.01 | joobie | ahh my bad |
| 22:40.05 | joobie | sec ill get the processor |
| 22:40.24 | jksM | (I actually meant it... it would be fun to know) |
| 22:41.14 | lesouvage | Joobie: try the question. I have a servr with ... gb memory and 2 core .... mhz processor. I will use a sip trunk and the provider uses codec ....... . I will record al/none/... % of all calls in ..... format. |
| 22:41.57 | joobie | The processor is a QuadCore Intel XEON E5405 (2GHz) 2 x 6MB.. |
| 22:42.00 | *** join/#asterisk voxter (n=voxter@S01060016b6b53c0c.vc.shawcable.net) |
| 22:42.16 | lesouvage | Nobody can calculate when the server colapse/brake but a professional gues will give you some direction. |
| 22:42.18 | joobie | 4GB memory |
| 22:42.44 | Zhad | when the CPU cores start boiling? |
| 22:44.07 | lesouvage | and what about codecs, switching between codecs, recording, expected number of calls, conferences and the number of participants etc. |
| 22:44.15 | joobie | I have a Intel QuadCore (E5405 @ 2GHZ) with 4GB of memory.. Just trying to figured out roughly how many g729 SIP calls it can handle concurrently before CPU would start to be an issue.. no call recording.. conference call support (not heavily used but say applied to 25% of the calls).. uplink will be various sip providers |
| 22:44.28 | Zhad | lots and lots |
| 22:44.38 | Katty | consumes pizza. |
| 22:45.11 | joobie | les, it wil be SIP to the proivder and SIP to the phone.. expected number of calls is unknown.. it will start with 5 phones.. but i want it to be able to scale out to 100's if possible... |
| 22:45.14 | Zhad | G.729 isn't really *that* computationally intensive, it only looks like that when you compare it to ulaw/alaw . |
| 22:47.06 | lesouvage | joobie: see http://www.voip-info.org/wiki/view/Asterisk+dimensioning |
| 22:48.00 | jksM | joobie, let's just say, that I'm doing 100 concurrent calls on g.729 with much less hardware than that |
| 22:48.36 | joobie | thanks jksM |
| 22:49.14 | *** join/#asterisk jov4n (n=jovan@87.19.108.219) |
| 22:49.23 | jov4n | Hi |
| 22:49.35 | jksM | joobie, I'm not saying that you'll be able to do the same, however |
| 22:53.37 | *** join/#asterisk StephenF[W] (n=none@198.144.201.106) |
| 22:54.15 | *** join/#asterisk kisu (n=kisu@ip70-179-88-179.dc.dc.cox.net) |
| 22:59.21 | *** join/#asterisk Arsenick- (n=ar@modemcable026.33-70-69.static.videotron.ca) |
| 23:01.39 | Arsenick- | someone had worked with dahdi ? |
| 23:01.48 | xchaos | what would you guys recommend.... asterisk-gui or freepbx? Because i installed the asterisknow flash which came with freepbx and it seems like all the support is for the asterisk-gui |
| 23:01.53 | xchaos | am i screwed? do i have to reinstall? |
| 23:02.14 | Dovid | if ur gona use a gui and u wana learn go with asterisknow |
| 23:02.22 | Dovid | if u want a quick fix then freepbx |
| 23:02.34 | Arsenick- | yeah but don't update to dahdi :p |
| 23:03.06 | ManxPower | ~zeeek |
| 23:03.07 | jbot | zeeek is, like, someone who once said "learning asterisk using a GUI is like learning sex through masturbation. You'll never get to the good stuff." |
| 23:03.25 | xchaos | dovid... i installed the asterisknow FLASH and it installed freepbx |
| 23:03.31 | jov4n | ahahhahha |
| 23:03.33 | xchaos | so i'm not sure what you are recommending... |
| 23:04.22 | xchaos | dovid i installed - AsteriskNOW 1.5.0 Beta1 (32-bit) |
| 23:04.30 | xchaos | which installed freepbx |
| 23:04.34 | xchaos | i dont see the asterisk-gui |
| 23:04.44 | Arsenick- | rofl |
| 23:05.43 | ManxPower | AsteriskNOW should have installed AsteriskGUI, IIRC. |
| 23:06.41 | xchaos | how do i cueck... when i go to the lan IP addres i get taken to the freepbx admin panel |
| 23:06.47 | Arsenick- | ManxPower, did you ever use asterisk-gui with another distrib ? I mean not asterisk-now just the gui ? |
| 23:06.50 | Maliuta | wait for it ... |
| 23:07.04 | Maliuta | AsteriskNOW and GUI aren't supported here |
| 23:07.20 | ManxPower | Arsenick-: I have never and will never use a GUI for Asterisk. |
| 23:07.42 | Arsenick- | yeah.. sound slike a good idea.. |
| 23:07.45 | ManxPower | Most people here have never touched an Asterisk GUI |
| 23:07.48 | Maliuta | ManxPower: don't you use ssh and vim for your gui? I know I do |
| 23:08.12 | ManxPower | Maliuta: yes. |
| 23:08.12 | Dovid | ;) |
| 23:08.19 | ManxPower | well joe, nit vim |
| 23:08.25 | Arsenick- | I've decided to install the gui just for other admin who don't know asterisk can do basic task, like adding new sip user etc.. |
| 23:08.59 | ManxPower | Arsenick-: You will regret that decision as soon as you try to do any customization. |
| 23:09.17 | Arsenick- | ahhh... it's already too late |
| 23:09.18 | Arsenick- | lol |
| 23:09.48 | Arsenick- | Asterisk-gui is too dumb to get my anallog card information using dahdi... |
| 23:11.11 | ManxPower | why are you even trying to use dahdi? |
| 23:11.44 | Arsenick- | I had problem today with my iax provider, and I updated asterisk |
| 23:12.05 | Arsenick- | from asterisk 1.4.17 to 1.4.22 .. |
| 23:12.27 | Arsenick- | and then I had problem with my analog card.. |
| 23:12.44 | ManxPower | just upgrade your zaptel |
| 23:12.53 | Arsenick- | I've read a little bit a see that asterisk 1.4.22 i using dahdi |
| 23:14.00 | ManxPower | No. Asterisk 1.4 supports both Zaptel and DAHDI. Asterisk 1.6 only supports DAHDI |
| 23:14.35 | Arsenick- | yeah I know but I though this would fix my problem... lol |
| 23:17.33 | *** join/#asterisk jer (n=jer@unaffiliated/jer) |
| 23:17.39 | *** join/#asterisk protocols (n=protocol@ip-88-153-205-180.unitymediagroup.de) |
| 23:18.27 | protocols | strange.. when my server has a virtual-interface configured, I am not able to sip-register from remote |
| 23:18.40 | protocols | if I remove that virtual-interface everything works.. |
| 23:18.55 | xchaos | which asterisknow flash version has the asterisk-gui....? **** I installed AsteriskNOW 1.5.0 Beta1 (32-bit) and that flash did not install asterisk-gui it installed FreePBX... can anyone help? tell me whcih asterisknow has the asterisk-gui? |
| 23:20.06 | protocols | 1.0x |
| 23:20.21 | xchaos | why the hell did 1.5 go to freepbx? |
| 23:20.28 | protocols | but I thought both were shipped now |
| 23:20.30 | xchaos | is free pbx suppose to be ebtter than asterisk-gui? |
| 23:20.30 | protocols | dunno |
| 23:20.40 | protocols | apparently.. |
| 23:21.09 | protocols | personally freepbx just looks like a webversion of vim/nano |
| 23:21.18 | protocols | except for that flash... |
| 23:21.23 | ManxPower | xchaos: You misunderstand. We don't use GUIs here. |
| 23:21.24 | drmessano | web version of nano? |
| 23:21.30 | ManxPower | ~freepbx |
| 23:21.30 | jbot | [~freepbx] FreePBX is unable to be supported here. It is made up of complex dialplans and scripts which can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking there |
| 23:21.32 | ManxPower | ~trixbox |
| 23:21.33 | jbot | somebody said trixbox was a full linux distro that includes FreePBX and other 3rd party add-ons. It is all this extra stuff which makes trixbox VERY difficult to support, and is not supported in #asterisk. Try asking in #trixbox or on their forums & wiki at http://www.trixbox.org/. We do not recommend using it. |
| 23:21.35 | ManxPower | ~asteriskgui |
| 23:21.41 | protocols | ha! |
| 23:21.42 | protocols | :D |
| 23:21.48 | *** join/#asterisk af_ (n=getsmart@88-149-230-65.dynamic.ngi.it) |
| 23:21.52 | protocols | ~asterisk-gui |
| 23:21.53 | jbot | [~asterisk-gui] The Asterisk-GUI is an open source project lead by Digium. It's client-driven, and its only dependency is Asterisk itself. Check out the new version from svn here: $ svn co http://svn.digium.com/svn/asterisk-gui/branches/2.0 asterisk-gui-2.0. For support go to #asterisk-gui |
| 23:21.54 | drmessano | Um yeah |
| 23:22.23 | drmessano | How is FreePBX the web version of NANO? |
| 23:22.56 | protocols | because its just shows the content of the config files |
| 23:23.12 | protocols | no improved usability |
| 23:23.18 | drmessano | Uh no |
| 23:23.28 | drmessano | Your info is wrong |
| 23:23.30 | protocols | at least compared to asterisk-gui |
| 23:23.37 | *** join/#asterisk wonderworld (n=ww@ip-62-143-16-59.unitymediagroup.de) |
| 23:23.40 | stintel | ~pbxinaflash |
| 23:23.41 | jbot | somebody said pbxinaflash was Ward Mundy's toy assembled by joe roper visit pbxinaflash.org or #pbxinaflash |
| 23:23.56 | ManxPower | drmessano: dude, they are GUI people. Thos sorts are never logical |
| 23:23.57 | drmessano | Its a full config application |
| 23:24.24 | drmessano | protocols: You've obviously no idea what you're talking about |
| 23:24.31 | protocols | yupp |
| 23:25.11 | wonderworld | hi, I try to hang up a sip channel with the ASterisk Manager Api. My problem is, that i don't know the full channel name, just th extension. do i need to parse the output of SIPpeers or is there a smarter way to hangup a channel that is used by a known extension (like 101) |
| 23:25.36 | [TK]D-Fender | wonderworld: "core show channels consice" |
| 23:26.06 | [TK]D-Fender | wonderworld: "core show channels concise" |
| 23:26.11 | wonderworld | can cli commands sent thru AMI too? |
| 23:26.36 | [TK]D-Fender | wonderworld: If you read the command list you'd have already known the answer was "yes" |
| 23:26.45 | wonderworld | tnx a lot |
| 23:28.44 | wonderworld | wow, the first hit on google doesn't look promising: http://bugs.digium.com/view.php?id=11181 |
| 23:30.13 | [TK]D-Fender | wonderworld: Applies to OLD versions, and not necessarily the way I might do this. |
| 23:30.46 | [TK]D-Fender | wonderworld: Go read the API list again |
| 23:30.59 | *** join/#asterisk protocols (n=protocol@ip-88-153-205-180.unitymediagroup.de) |
| 23:31.03 | ManxPower | There ARE AMI docs included in the Asterisk source dir. |
| 23:31.29 | wonderworld | ok i will. sorry for asking dumb questions. need to get something done quickly and used the api never berfore. don't want to get on your nerves. will go reading now. |
| 23:31.57 | ManxPower | wonderworld: then you will fail. Telecom is not something you can rush. |
| 23:32.43 | *** join/#asterisk RobertLaptop (n=rmiddle@pool-72-81-212-249.bltmmd.fios.verizon.net) |
| 23:33.22 | wonderworld | i am not new to asterisk. i wrote several agi-apps. but i never used the management interface before |
| 23:42.34 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 23:48.10 | *** join/#asterisk Koshatul (n=evangeli@ppp188-33.static.internode.on.net) |
| 23:50.44 | *** join/#asterisk tobias (n=tobias@user-0ce2hvs.cable.mindspring.com) |
| 23:51.06 | *** join/#asterisk protocols (n=protocol@ip-88-153-205-180.unitymediagroup.de) |
| 23:53.22 | *** join/#asterisk jer (n=jer@unaffiliated/jer) |
| 23:55.34 | Carlos_PHX | I swear if I see one more of these I'm going to burn the building down. 0004F201148F |
| 23:55.46 | Carlos_PHX | - Incoming call: Got SIP response 500 "Internal Server Error" back from 192.168.100.82 |
| 23:55.58 | Carlos_PHX | Anyone know why Polycoms do this randomly? |