| 00:00.26 | Dovid | l-fy: and are there any other settings that i have to set ? |
| 00:00.29 | Dovid | for qos ? |
| 00:00.59 | l-fy | not that i know |
| 00:01.05 | l-fy | but i never use it |
| 00:01.46 | Dovid | what number do i have to change 160 to ? |
| 00:01.54 | l-fy | try 10 |
| 00:02.27 | anthm | avoided deadlock is a while loop around a mutex trylock to prevent permenantly blocking in some situation where the code is not comfortable trusting the mutex to event release |
| 00:02.36 | Dovid | what does the number signafy ? |
| 00:03.08 | Dovid | brb. |
| 00:07.31 | *** join/#asterisk oadaeh (n=jason@las-static-208.57.199.83.mpowercom.net) |
| 00:08.48 | Dovid | back |
| 00:08.55 | Dovid | what does the 10 signafy ?> |
| 00:09.04 | rowter | anthm, ooh |
| 00:09.15 | rowter | anthm, but it was blocked at last. |
| 00:09.36 | *** join/#asterisk brockj49464_home (n=chatzill@63.87.56.153) |
| 00:09.40 | anthm | 10 was how many times in the loop |
| 00:09.57 | Dovid | loop meaning ? |
| 00:10.05 | anthm | for(x = 0; x < 10; x++) if (still_wont_lock) punt(); |
| 00:10.22 | Dovid | r we talkin about the snom anthm ? |
| 00:10.55 | anthm | nah i was talking about the "what does avoiding deadlock mean" |
| 00:11.22 | Dovid | ah ok |
| 00:11.30 | Dovid | anyone know how to do QOS on the SNOM's ? |
| 00:11.38 | l-fy | anthm > there are better ways to avoid deadlocks |
| 00:11.56 | anthm | that's for sure ;) |
| 00:12.06 | l-fy | like |
| 00:12.07 | *** part/#asterisk brockj49464_home (n=chatzill@63.87.56.153) |
| 00:12.10 | l-fy | good design |
| 00:12.31 | rowter | l-fy, a good design of what? |
| 00:12.46 | l-fy | of the software |
| 00:13.18 | l-fy | deadlock is a design bug always |
| 00:13.23 | l-fy | so it has to be fixed there |
| 00:13.27 | rowter | anthm, how could I avoid this deadlocks in this case? |
| 00:13.47 | rowter | l-fy, I never had a deadlock on 1.0.x |
| 00:14.16 | l-fy | rowter > i don't use asterisk so don't ask me :) |
| 00:14.25 | anthm | http://www.cluecon.com/dev.txt |
| 00:14.51 | dlynes_laptop | l-fy, that kinda begs the question well why are you in this channel, then? :) |
| 00:15.00 | l-fy | dlynes_laptop > to troll of course :) |
| 00:15.04 | *** join/#asterisk oadaeh (n=jason@las-static-208.57.199.83.mpowercom.net) |
| 00:15.21 | dlynes_laptop | l-fy, ah...thought maybe it was to recruit for yate :) |
| 00:15.34 | rob0 | What is asterisk? |
| 00:15.44 | dlynes_laptop | ~asterisk |
| 00:15.46 | jbot | asterisk is the best free PBX in the world. |
| 00:16.03 | Qwell[] | and jbot is never wrong |
| 00:16.08 | l-fy | that should be updated to the most used |
| 00:16.12 | rowter | anthm, am on a dead end if I have deadlocks hehe then? |
| 00:16.23 | rob0 | Sometimes slow though! Did someone reboot him? |
| 00:16.32 | Qwell[] | rob0: must've |
| 00:18.05 | dlynes_laptop | rob0, must have...he wasn't even logged on, earlier |
| 00:19.59 | anthm | you are avoiding them that is what the message said but i can't promise you how good of a result you will get =D |
| 00:20.05 | *** join/#asterisk oadaeh (n=jason@las-static-208.57.199.83.mpowercom.net) |
| 00:21.05 | rowter | anthm, the result was a hung of asterisk.. hehe |
| 00:21.17 | *** join/#asterisk oadaeh (n=jason@las-static-208.57.199.83.mpowercom.net) |
| 00:21.21 | *** join/#asterisk Katty (n=Administ@dialup-4.244.180.237.Dial1.StLouis1.Level3.net) |
| 00:21.23 | *** join/#asterisk ariel_ (n=Ariel@dsl-20-177.cofs.net) |
| 00:21.26 | l-fy | rowter > what are the threads that get locked? |
| 00:21.59 | anthm | think of it like: I let my gf walk home in the dark, i will call her 10 times to make sure she got home safe, if she doesnt answer after 10 I will assume she is in bed sleeping ;) |
| 00:22.12 | Katty | hi lads. |
| 00:22.12 | Qwell[] | anthm: That is a great analogy :p |
| 00:23.05 | anthm | it's channel.c so i assume it's the channel struct linked list destroy fiasco |
| 00:23.57 | *** join/#asterisk trbldwine (i=troubled@71.194.161.170) |
| 00:24.05 | *** join/#asterisk jeebusmobile (n=jeebusmo@29palms-cuda1-68-170-42-234.losaca.adelphia.net) |
| 00:24.15 | anthm | when a channel hangs up, it locks it, unlocks it then destroys the lock, if something else is happening at the same time that locks it it prints that |
| 00:24.33 | anthm | cos the hangup expects it should not stay locked |
| 00:24.43 | rowter | anthm, ooh, am using rhinos there are lots of zap's |
| 00:24.44 | l-fy | anthm > but this is wrong |
| 00:24.51 | anthm | so after 10 it just goes ahead and nukes it |
| 00:25.01 | *** join/#asterisk rene- (n=rene-@gea-gye-internet.telconet.net) |
| 00:25.04 | l-fy | it's normal to have hangup locked for a while to have time to clean other objects related to hangup of the channel |
| 00:25.27 | l-fy | i think the real problem is not the deadlock |
| 00:25.28 | rowter | anthm, ahh maybe where am calling each zap I should add a congestion or hangup.. maybe there is the prob.. let me add it. |
| 00:25.45 | l-fy | i think the real question is why dosen't the channel destroy until 10? |
| 00:25.45 | rene- | hi, how do i set a variable based on the output of an external system command? |
| 00:26.08 | *** join/#asterisk oadaeh (n=jason@las-static-208.57.199.83.mpowercom.net) |
| 00:26.37 | Qwell[] | rene-: can't currently, without like an agi |
| 00:26.55 | anthm | get app_backticks http://from pbxfreeware.org |
| 00:27.20 | rene- | thks |
| 00:27.37 | rene- | k pex rowter |
| 00:27.40 | Qwell[] | actually, you know what |
| 00:27.59 | anthm | oh it's only in the channel_find_locked now |
| 00:28.05 | rowter | rene-, a problem with deadlocks hehe on asterisk 1.2.10 |
| 00:28.19 | rene- | damn |
| 00:28.34 | rene- | Qwell[]: what |
| 00:28.42 | Qwell[] | rene-: nothing, just thinking out loud |
| 00:28.46 | anthm | all the channels are in a big linked list and everything locks them constantly |
| 00:28.49 | Qwell[] | I thought somebody did a SYSTEM function |
| 00:28.54 | anthm | like show channels command etc |
| 00:28.58 | Qwell[] | but I'm not seeing it in trunk |
| 00:29.05 | rene- | ahh |
| 00:29.06 | anthm | it's more or less russian roluette |
| 00:29.26 | l-fy | of course not anthm |
| 00:29.31 | anthm | there is a system func too but that wont put the result in a var |
| 00:29.32 | *** join/#asterisk oadaeh (n=jason@las-static-208.57.199.83.mpowercom.net) |
| 00:29.37 | l-fy | it should try until it gets the lock |
| 00:29.41 | Qwell[] | anthm: no, function, not app :) |
| 00:29.48 | rowter | anthm, yeah something got wrong on that list, cause I killed and tried to start it but it told me that channells where busy, and had to restart.. |
| 00:30.08 | l-fy | if it can't get the lock of the channel list something else is fucked up much more and it dosen't matter if one channel is freazing |
| 00:30.34 | anthm | hense like yoda says: try not, do or do not, there is no try |
| 00:30.38 | anthm | =D |
| 00:31.03 | rene- | haha |
| 00:31.25 | rowter | ooh god.. well I-ll add this congestion after each call to zap's and unicall.. hope it fix something out.. |
| 00:31.31 | l-fy | anthm > just that people don't like "try to make the pbx to work, do or do not, have or not have money" |
| 00:31.44 | l-fy | rowter > don't |
| 00:31.51 | l-fy | is not gonna fix the problem |
| 00:31.54 | l-fy | it will just cover it |
| 00:32.03 | anthm | be careful w congestion it's eternal |
| 00:32.21 | rowter | then hangup |
| 00:32.31 | anthm | instead of sending the specific signal to the channel it emulated a busy sig forever and never hangs up |
| 00:32.40 | rowter | ooh |
| 00:32.54 | anthm | it's a black hole you cant exit it unless the far end hangs up |
| 00:33.00 | anthm | and it can get stuck |
| 00:33.08 | l-fy | o well, this is how asterisk gets a little bit worse |
| 00:34.05 | rene- | does app_backticks supports passing parameters to the file it executes? |
| 00:34.12 | rowter | but a hangup after each call to zap and unicall should help no? |
| 00:34.16 | rene- | installing backticks is easy http://henryjunior.com/blog/?p=18 |
| 00:34.39 | anthm | wow power docs |
| 00:34.55 | anthm | they killed astxs now i think |
| 00:35.04 | rene- | :( |
| 00:35.10 | Qwell[] | anthm: yeah, it needs to be fixed up for trunk/1.4.. |
| 00:36.25 | l-fy | ogurki |
| 00:38.27 | anthm | hangup should do no good cos if the call completes it will never execute hangup anyway |
| 00:38.44 | *** join/#asterisk dprevite (n=dprevite@cpe-66-61-129-18.insight.res.rr.com) |
| 00:38.53 | anthm | unless you have the g flag and the far end initiated the terminate |
| 00:40.08 | anthm | are you doing anything else fancy like that chan_loco it's pretty notorious for causing commotion |
| 00:42.15 | rene- | chan_loco |
| 00:42.20 | rene- | =) |
| 00:43.04 | rowter | anthm, no antm, anything fancy.. am changing the music and hold to g711 also.. |
| 00:43.14 | rene- | later |
| 00:43.20 | rowter | see ya |
| 00:44.21 | rowter | anthm, far end always initate the termination in this case, cause they are using modems to answer the zap as extensions |
| 00:46.02 | dlynes_laptop | bbiab |
| 00:46.03 | *** join/#asterisk batphone (n=bugz@cpe-70-123-122-41.houston.res.rr.com) |
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| 00:47.16 | anthm | yah so you will never get past the dial line |
| 00:47.38 | anthm | next time it happens instead of killing it you can connect to gdb and get a trace |
| 00:47.56 | anthm | or use that thing that sucks the core out of a running process |
| 00:48.18 | anthm | and make a full backtrace on every thread and report it to the bug tracker where someone can eval it |
| 00:48.19 | rowter | ooh let me take a look on voipinfo.. |
| 00:48.27 | batphone | http://pastebin.ca/100328 |
| 00:49.17 | anthm | make sure it's 100% pure fresh checkout unaltered cos if not you may be sent back to repeat |
| 00:50.06 | *** join/#asterisk oadaeh (n=jason@las-static-208.57.199.83.mpowercom.net) |
| 00:50.33 | rowter | anthm, am looking at the steps to make a backtrace and gdb, seems complicatd hehe, there is no core in /tmp right now.. let me see. |
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| 01:14.38 | Zodiacal | anyone know if theres a way to name linekeys in my polycom 601 differently? |
| 01:14.43 | Zodiacal | i.e. line1, line2, etc.. |
| 01:14.51 | Zodiacal | and still have them ring down during call waiting |
| 01:14.57 | Zodiacal | does that make any sence? |
| 01:15.04 | Zodiacal | sense |
| 01:17.01 | *** part/#asterisk Entegrity (n=Entegrit@c-65-96-116-56.hsd1.ma.comcast.net) |
| 01:38.12 | *** join/#asterisk LakeSolon (n=blake@12-227-169-99.client.mchsi.com) |
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| 01:42.14 | *** part/#asterisk CANO-1982 (n=alejandr@190.48.72.107) |
| 01:43.29 | *** join/#asterisk shmaltz (n=mybox@mail.dmaven.com) |
| 01:43.45 | shmaltz | hi everyone |
| 01:43.50 | shmaltz | jbot you awake? |
| 01:44.02 | shmaltz | ~ time |
| 01:44.03 | jbot | it has been said that time is 1 dimensional, or everlasting, an illusion, or 2006.07.26 1:44:03 GMT |
| 01:44.13 | shmaltz | ~ date |
| 01:44.16 | jbot | It is now 2006.07.26 1:44:16 GMT |
| 01:44.19 | shmaltz | ~ seen tzafrir |
| 01:44.37 | jbot | tzafrir <n=tzafrir@local.xorcom.com> was last seen on IRC in channel #asterisk, 9d 22h 5m 20s ago, saying: 'http://the-edge.blogspot.com/2005/10/worldss-smallest-ip-pbx-at-astricon.html'. |
| 01:44.37 | shmaltz | ~ seen tzafrir_laptop |
| 01:44.42 | jbot | tzafrir_laptop <n=tzafrir@bzq-88-152-23-125.red.bezeqint.net> was last seen on IRC in channel #asterisk, 2d 19h 59m 27s ago, saying: 'that said, #freepbx may be a better place'. |
| 01:47.04 | *** join/#asterisk Johnnie (n=john@pdpc/supporter/active/Johnnie) |
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| 01:53.18 | *** join/#asterisk dlynes_laptop (n=dlynes@216.251.149.66) |
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| 02:17.11 | *** join/#asterisk NoRemorse (n=bah@203-214-13-168.dyn.iinet.net.au) |
| 02:17.13 | NoRemorse | hi all |
| 02:17.44 | NoRemorse | I am tryiung to provision an iaxy, but one leg of the ip network is wireless and I am getting no response from the unit, any ideas please? |
| 02:18.03 | NoRemorse | oh, do i have to unplug the phone> |
| 02:25.34 | reticent | hmm, has anyone ever encountered a situation where the Asterisk comedian voicemail module would put new voice messages in the Old folder instead of the INBOX folder (this is new voicemail)? |
| 02:26.21 | reticent | i have two systems running with the same NFS backend and the same configuration (even the same binaries) however one puts any new messages left for a user into the old folder and the other into the INBOX folder |
| 02:27.20 | reticent | this is with Asterisk 1.2.10 |
| 02:27.47 | dlynes_laptop | reticent, yes, yes i have |
| 02:27.52 | dlynes_laptop | reticent, and as it turned out |
| 02:28.03 | dlynes_laptop | reticent, it was my stupidity :) |
| 02:28.16 | dlynes_laptop | reticent, check to make sure you haven't reassigned their mailbox |
| 02:30.39 | reticent | nope, same database even (mysql replicated cluster) |
| 02:31.34 | *** join/#asterisk JunK-Y (n=junky@modemcable205.175-81-70.mc.videotron.ca) |
| 02:31.42 | reticent | probably stupidity though, uck.. so hard to discover ones own stupidity somtimes |
| 02:32.01 | dlynes_laptop | reticent, i would check your logs |
| 02:32.17 | dlynes_laptop | reticent, and try logging in to that person's mailbox, yourself |
| 02:32.25 | dlynes_laptop | reticent, watch the log while you're doigng it |
| 02:32.37 | dlynes_laptop | reticent, make sure it's logging into the correct mailbox |
| 02:32.51 | dlynes_laptop | reticent, if it is, get the end user to enter the mailbox |
| 02:32.56 | dlynes_laptop | reticent, check hte logs |
| 02:33.06 | dlynes_laptop | reticent, and make sure they're also going into the correct mailbox |
| 02:34.45 | reticent | everything looks good, just a sporadic "Jul 26 02:32:44 NOTICE[20758] sched.c: Scheduled event in 0 ms?" |
| 02:34.54 | reticent | i'm going to strace the event |
| 02:35.00 | reticent | lucky this is a dev system |
| 02:35.12 | reticent | *cringes at the memory of stracing active systems* |
| 02:35.32 | dlynes_laptop | reticent, that looks normal |
| 02:36.38 | *** join/#asterisk ManxPower (i=ewieling@97.sub-70-210-153.myvzw.com) |
| 02:38.40 | batphone | rp -an |grep -v `for i in \`ls 000*.cfg|grep -v phone|tr [a-z] [A-Z]|cut -f 1 -d .|sed -e "s/../&:/g" -e "s/.$//"\`; do echo "$i\|";done|tr -d '\012'|sed -e s/..$// -e s/^/\'/ -e s/$/\'/` |
| 02:38.42 | Qwell | file: there, there |
| 02:38.46 | batphone | arp |
| 02:38.47 | batphone | arp |
| 02:38.53 | batphone | ;) |
| 02:39.01 | batphone | arp dammit! |
| 02:39.05 | batphone | stupid phone.. |
| 02:39.45 | dlynes_laptop | wtf is metaboo? |
| 02:39.49 | dlynes_laptop | ~metaboo |
| 02:40.00 | dlynes_laptop | ~wiki metaboo |
| 02:41.01 | Qwell | ha |
| 02:42.36 | shmaltz | stupid: http://www.stupid.com/index.html |
| 02:43.44 | shmaltz | I like this headset: |
| 02:43.46 | shmaltz | http://www.stupid.com/stat/PLEZ.html |
| 02:50.14 | *** join/#asterisk Nugget (i=nugget@dazed.slacker.com) |
| 02:50.31 | file | Nugget! |
| 02:50.35 | Nugget | moo |
| 02:50.52 | file | what's the sauce today? |
| 02:50.56 | *** join/#asterisk tengulre11 (n=tengulre@61.185.224.66) |
| 02:50.57 | Nugget | honey |
| 02:53.30 | file | only honey? |
| 02:55.47 | mrdigital | anyone else wanna take a stab |
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| 03:00.32 | *** mode/#asterisk [+o mog_home] by ChanServ |
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| 03:06.26 | clyrrad | how can I completely get rid of mpg123? Anyone have a good link? Or know how to completely get it out of Asterisk? |
| 03:06.50 | *** join/#asterisk EyeCue (n=eyecue@unaffiliated/eyecue) |
| 03:09.59 | Nugget | it's not "in" asterisk in the sense that I think you mean. |
| 03:10.13 | Nugget | rm `whcih mpg123` ought to be all you need to do. |
| 03:10.22 | Nugget | er, rm `which mpg123` |
| 03:10.43 | Nugget | but you can just pretend it's not there to the same effect |
| 03:11.06 | clyrrad | I have tried that |
| 03:11.10 | clyrrad | changed mode=files |
| 03:11.17 | clyrrad | but on the CLI i still get mpg123 errors |
| 03:11.27 | clyrrad | I have also rm -rf'ed the directory from /sbin |
| 03:11.40 | clyrrad | still asterisk see's it and tries to use it and i get issues |
| 03:13.26 | fgwaller | native music on hold : http://www.voip-info.org/wiki-Asterisk+config+musiconhold.conf |
| 03:13.56 | fgwaller | running without mpg123 in asterisk, just use it for converting to native slin or ulaw |
| 03:14.10 | fgwaller | night |
| 03:15.06 | *** join/#asterisk Kumba_ (n=Kumba@241-208.124-70.tampabay.res.rr.com) |
| 03:16.20 | Kumba_ | Will daisy-chaining phones together cause any problems? (Polycom's) |
| 03:16.59 | clyrrad | fgwaller - yes i have done all that |
| 03:17.11 | clyrrad | my configs are as shown in your link |
| 03:17.19 | clyrrad | i beleive its the encoding of the mp3s that the issue |
| 03:18.03 | clyrrad | I would need to conver them - I am just now sure how to do that - do you have any more links? |
| 03:18.33 | *** join/#asterisk [TK]D-Fender (n=joe@66.11.164.239) |
| 03:18.52 | clyrrad | yea it must be the encoding of the MP3's lots of errors on the CML |
| 03:19.00 | clyrrad | not ERRORS - but WARNINGS |
| 03:19.14 | clyrrad | is it possible to convert them and leave them in mp3 format? |
| 03:22.14 | *** join/#asterisk blebleble (i=godie@caesar.godie.net) |
| 03:22.28 | blebleble | anyone offer consulting services for implementing a fax solution to asterisk? |
| 03:22.47 | clyrrad | does it have to be in Asterisk? |
| 03:22.50 | Kumba_ | Whenever I try to go to voicemail to check my inbox, I get an error that says it cant create the lockfile... is that because the Mailbox is new and no one has ever left a message before? |
| 03:23.03 | blebleble | clyrrad: well my thoughts are hylafax + asterisk |
| 03:23.11 | clyrrad | pure VOIP? |
| 03:23.20 | Kumba_ | I cant find anything regarding initializing the VM Boxes |
| 03:23.35 | clyrrad | Kumba_ just leave a voicemail and let asterisk do the work |
| 03:23.43 | Kumba_ | Ahhh... K |
| 03:26.59 | *** join/#asterisk mhnoyes (n=mhnoyes@dialup-4.246.233.169.Dial1.SanJose1.Level3.net) |
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| 03:32.02 | type0 | who do you suggest for a provider |
| 03:32.06 | type0 | for a DID |
| 03:32.08 | type0 | with good rates |
| 03:32.26 | type0 | with 2 calls at once |
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| 03:42.13 | b00mer | I am having some trouble with local digitmaps on Polycoms... anyone familar? |
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| 04:10.20 | file | quiet night |
| 04:10.40 | *** join/#asterisk CrashHD (i=CrashHD@c-67-182-167-222.hsd1.ca.comcast.net) |
| 04:10.59 | CrashHD | anyone have problems with their asterisk boxes freezing? |
| 04:11.07 | CrashHD | unresponsive |
| 04:11.12 | CrashHD | but the pri is still working on the machine |
| 04:11.20 | Kumba_ | I had horrific problems caused by a $2 home depot RJ45 coupler :) |
| 04:11.49 | CrashHD | .... |
| 04:13.39 | *** join/#asterisk brc_ (n=brc_@pdpc/supporter/basic/brc) |
| 04:14.35 | viler | Anyone knows if is possible to execute an agi while a call enters into queue ? |
| 04:22.42 | dlynes_laptop | CrashHD, define "but the pri is still working" |
| 04:23.21 | CrashHD | calls pass through the system just fine pri port 1 through to a legacy pbx on pri port 2 |
| 04:23.26 | CrashHD | I have 2 machines doing this |
| 04:23.45 | dlynes_laptop | CrashHD, but you can't ssh into it or something? |
| 04:23.52 | CrashHD | correct |
| 04:23.56 | CrashHD | even the console is dead |
| 04:24.13 | dlynes_laptop | how much time have you given it to respond? |
| 04:24.19 | CrashHD | hour |
| 04:24.23 | CrashHD | I left for dinner |
| 04:24.23 | dlynes_laptop | oh wow |
| 04:24.29 | CrashHD | deffinitely dead |
| 04:24.40 | CrashHD | I run asterisk processes in realtime |
| 04:24.43 | dlynes_laptop | but calls are still happening |
| 04:24.46 | dlynes_laptop | ewww |
| 04:24.51 | CrashHD | could they be spinning out of control? |
| 04:24.57 | dlynes_laptop | why is it that people run it in real time? |
| 04:25.02 | dlynes_laptop | I don't see the point to that, personally |
| 04:25.08 | CrashHD | the sip phones on one system will not connect |
| 04:25.13 | CrashHD | but pri calls on the other system are fine |
| 04:25.27 | CrashHD | so ip = dead |
| 04:25.30 | CrashHD | tdm = ok |
| 04:25.37 | CrashHD | I'm going to be pulling out of realtime |
| 04:25.54 | dlynes_laptop | are all the tdm calls on one card? |
| 04:26.00 | CrashHD | yes |
| 04:26.01 | dlynes_laptop | or are any of them going between cards? |
| 04:26.07 | CunningPike | Anyone else's Shaw connection in the crapper? |
| 04:26.09 | *** join/#asterisk brc__ (n=brc_@pdpc/supporter/basic/brc) |
| 04:26.18 | dlynes_laptop | CunningPike, Beauty! |
| 04:26.19 | CrashHD | one a104d card |
| 04:26.27 | CrashHD | with newest driver |
| 04:26.40 | dlynes_laptop | CrashHD, ah...new driver's been out for about 2 weeks now |
| 04:26.48 | dlynes_laptop | CrashHD, i haven't had the courage to install it yet :) |
| 04:26.54 | *** join/#asterisk scardinal (n=supreme@port816.ds1-suoe.adsl.cybercity.dk) |
| 04:27.02 | CrashHD | they have a ways to go |
| 04:27.06 | dlynes_laptop | i'm starting to think laziness is a virtue now :) |
| 04:27.08 | CrashHD | I'm still getting some pops and clicks |
| 04:27.16 | CrashHD | which driver are you running? |
| 04:27.21 | dlynes_laptop | CrashHD, really? I have no problems with it |
| 04:27.38 | CrashHD | what are you getting on your zttest scores? |
| 04:27.54 | dlynes_laptop | -rw-r--r-- 1 root root 3362237 2006-06-20 17:19 hwec-utils-beta4-2.3.4.tgz |
| 04:27.54 | dlynes_laptop | -rw-r--r-- 1 root root 6634563 2006-06-20 17:19 wanpipe-beta4-2.3.4.tgz |
| 04:28.11 | CrashHD | they had me upgrade to beta5 then beta6 for troubleshooting |
| 04:28.17 | CrashHD | but haven't gotten back to me yet |
| 04:28.19 | dlynes_laptop | i can't even recall what zttest is offhand, now |
| 04:28.55 | dlynes_laptop | CrashHD, what kernel are you using? |
| 04:29.16 | CrashHD | 2.6.9-34.0.2.ELsmp |
| 04:29.21 | dlynes_laptop | ah |
| 04:29.21 | CrashHD | on both machines |
| 04:29.22 | dlynes_laptop | yeah |
| 04:29.28 | CunningPike | dlynes_laptop: Your Shaw connection is OK? |
| 04:29.31 | dlynes_laptop | I'm running 2.6.15.5 on all my sangoma machines |
| 04:29.37 | dlynes_laptop | CunningPike, wouldn't know |
| 04:29.47 | CrashHD | dlynes_laptop: I'm lazy...up2date is my friend |
| 04:29.47 | CunningPike | dlynes_laptop: Telus> |
| 04:29.53 | dlynes_laptop | CunningPike, our sprinkler system popped a sprinkler thursday or friday night |
| 04:30.01 | dlynes_laptop | CunningPike, i've been without internet at home since |
| 04:30.08 | CunningPike | dlynes_laptop: Bummer |
| 04:30.13 | dlynes_laptop | CunningPike, yeah, no doubt |
| 04:30.22 | dlynes_laptop | CunningPike, so i'm sleeping out in the backyard in a tent right now :) |
| 04:30.29 | CunningPike | lol |
| 04:30.32 | CunningPike | Really? |
| 04:30.33 | CrashHD | lol |
| 04:30.35 | dlynes_laptop | yeah |
| 04:30.50 | dlynes_laptop | why pay for a hotel room, if you don't have to? |
| 04:30.52 | CunningPike | dlynes_laptop: Good job the weather's nice :D |
| 04:31.08 | dlynes_laptop | I've got a four man tent...may as well make use of it |
| 04:31.20 | dlynes_laptop | big enough for my blankets, pillows, and a mini office |
| 04:31.28 | CrashHD | dlynes_laptop: any specific reason I should upgrade my kernel, think it would help? |
| 04:31.29 | *** join/#asterisk jcims (n=jcims@cpe-24-210-60-100.columbus.res.rr.com) |
| 04:31.38 | dlynes_laptop | got a couple crates for a desk, and a chair out there |
| 04:31.43 | CrashHD | this freezing is driving me nuts |
| 04:31.43 | reticent | dlynes_laptop: turns out it was my own fault |
| 04:31.44 | dlynes_laptop | just throw my laptop on top of the crates |
| 04:31.46 | CunningPike | dlynes_laptop: Aha - Shaw seems fixed again - I was seeing 50% packet loss and latency of 5000ms |
| 04:31.46 | dlynes_laptop | and away i go |
| 04:32.04 | dlynes_laptop | reticent, see? I told ya :) |
| 04:32.12 | reticent | although it wasn't so obvious |
| 04:32.17 | dlynes_laptop | CrashHD, well, just curious |
| 04:32.18 | CrashHD | CunningPike: they were just giving you a chance to catch up with yourself...that's all. |
| 04:32.25 | dlynes_laptop | CrashHD, i'm running 2.6.15.5 without problems on sangoma |
| 04:32.33 | CunningPike | CrashHD: Well, I am fast |
| 04:32.38 | dlynes_laptop | CrashHD, you're running a patched 2.6.9 and you're having problems |
| 04:32.42 | CunningPike | CrashHD: And cunning |
| 04:32.51 | CrashHD | nods |
| 04:32.57 | dlynes_laptop | reticent, so what was the problem, then? |
| 04:32.58 | reticent | i made some code modifications so that voicemail files and directories has specific permissions, when i upgraded to 1.2.10 my patch applied one of the changes to the wrong line and that was causing the files to be written to Old instead of new |
| 04:33.00 | CrashHD | and different asterisk/zaptel version on the machines |
| 04:33.10 | CrashHD | hmmmm |
| 04:33.11 | dlynes_laptop | reticent, cool beans |
| 04:33.19 | CrashHD | dlynes_laptop: you mind checking your zttest scores? |
| 04:33.26 | CrashHD | I'm wondering how far I am off |
| 04:33.30 | CrashHD | never ran sangoma before |
| 04:33.35 | CrashHD | this is my first experience |
| 04:33.45 | dlynes_laptop | CrashHD, which version of zaptel are you using? |
| 04:33.51 | dlynes_laptop | CrashHD, i'm using 1.2.6 |
| 04:34.03 | CrashHD | 1.2.7 on machine with 1.2.10 |
| 04:34.04 | dlynes_laptop | CrashHD, and libpri 1.2.3 |
| 04:34.10 | CrashHD | ya |
| 04:34.11 | dlynes_laptop | CrashHD, and asterisk 1.2.9.1 |
| 04:34.27 | dlynes_laptop | i was thinking about upgrading to 1.2.7 and 1.2.10 tonight |
| 04:34.40 | dlynes_laptop | i can't remember |
| 04:34.45 | dlynes_laptop | can you run zttest while the pbx is up? |
| 04:34.52 | CrashHD | I believe so |
| 04:34.55 | CrashHD | I usually do |
| 04:35.03 | CrashHD | scores don't seem affected |
| 04:35.15 | CrashHD | don't quote me though |
| 04:36.06 | dlynes_laptop | --- Results after 24 passes --- |
| 04:36.07 | dlynes_laptop | Best: 100.000000 -- Worst: 99.987793 -- Average: 99.988302 |
| 04:36.23 | dlynes_laptop | and 1) i'm not running realtime |
| 04:36.28 | dlynes_laptop | and 2) I'm using an a101 |
| 04:36.45 | CrashHD | I'm getting 99.987 |
| 04:36.52 | CrashHD | I'm getting 99.987793 |
| 04:36.54 | dlynes_laptop | so what's wrong with that? |
| 04:36.58 | CrashHD | *shrugs* |
| 04:37.01 | dlynes_laptop | I only got 100 once |
| 04:37.08 | dlynes_laptop | every other time it was 99.987793 |
| 04:37.15 | CrashHD | thought I read something about a certain score you needed |
| 04:37.21 | CrashHD | maybe just for digium |
| 04:37.22 | dlynes_laptop | 99.7 or higher |
| 04:37.28 | CrashHD | oh |
| 04:37.30 | CrashHD | ok |
| 04:37.33 | CrashHD | I feel better about that then |
| 04:37.47 | dlynes_laptop | i often got 88's with digium hardware |
| 04:37.51 | dlynes_laptop | it was horrible |
| 04:37.56 | CrashHD | eek |
| 04:38.14 | dlynes_laptop | my range was generally 88-92 |
| 04:38.22 | dlynes_laptop | and the odd time I'd get 96 point something |
| 04:38.43 | dlynes_laptop | i got tired of playing russian roulette with our business |
| 04:38.49 | CrashHD | if I could just figure out this lockup issue I'd be sold |
| 04:38.50 | CrashHD | heh |
| 04:38.55 | CrashHD | that is pretty much what I'm playing now |
| 04:39.07 | CrashHD | making me go mad |
| 04:39.09 | dlynes_laptop | CrashHD, i would suggest running it normally |
| 04:39.13 | dlynes_laptop | CrashHD, forget about realtime for now |
| 04:39.17 | CrashHD | just made the switch |
| 04:39.20 | CrashHD | while we were chatting |
| 04:39.43 | dlynes_laptop | CrashHD, also edit your makefile for asterisk so that it uses astmm |
| 04:39.54 | dlynes_laptop | CrashHD, then you can watch for memory leaks, too |
| 04:40.22 | CrashHD | astmm? |
| 04:40.38 | CrashHD | file's always laughing at me |
| 04:40.38 | file | dlynes_laptop: now that I told you about that, are you all happy? :) |
| 04:40.57 | dlynes_laptop | heh |
| 04:41.01 | CrashHD | :) |
| 04:41.07 | dlynes_laptop | well, i'll be happy if it points out where the problem is :) |
| 04:41.22 | dlynes_laptop | cause then i can fix it, and submit a patch |
| 04:42.16 | CrashHD | what command would I run once astmm has been compiled? |
| 04:42.18 | CrashHD | to see |
| 04:42.40 | CrashHD | show mem |
| 04:42.41 | CrashHD | ahh |
| 04:45.49 | CrashHD | recompiling |
| 04:47.38 | *** join/#asterisk postel (n=jp@unaffiliated/postel) |
| 04:51.09 | CrashHD | asterisk is so much fun |
| 04:51.12 | CrashHD | lol |
| 04:51.17 | docelmo | No its not.. ACK! |
| 04:51.18 | CrashHD | it's like a kid |
| 04:51.27 | CrashHD | so much joy....yet so much anguish |
| 04:55.34 | CrashHD | file: any reason why dlynes_laptop kernel would be working better than mine? |
| 04:56.01 | dlynes_laptop | CrashHD, because mine doesn't have redschmat patches applied to it? |
| 04:56.15 | CrashHD | those bad ? lol |
| 04:56.40 | dlynes_laptop | CrashHD, i run a plain vanilla kernel..no patches applied |
| 04:57.14 | CrashHD | how much bearing does that have on asterisk I suppose was my question |
| 04:57.26 | CrashHD | like could that be causing my problems |
| 04:57.34 | CrashHD | running the redwh0re patched kernel |
| 04:57.36 | file | I think it's karma... |
| 04:57.54 | dlynes_laptop | CrashHD, it's compiled for smp, low latency optimizations, wan router, ... |
| 04:58.01 | nailbags|work | CrashHD: it has a fairly big bearing on things that use kernel modules, such as zaptel |
| 04:58.23 | dlynes_laptop | CrashHD, and i'm also running on a Dual PIII |
| 04:58.30 | dlynes_laptop | CrashHD, what are you running on? |
| 04:58.47 | CrashHD | it's a dell 1850, single cpu (ht enabled) |
| 04:58.52 | dlynes_laptop | aha! |
| 04:58.59 | dlynes_laptop | there's the culprit right there, I would imagine |
| 04:59.09 | dlynes_laptop | zaptel/asterisk do not play nice with HT |
| 04:59.15 | CrashHD | oh |
| 04:59.20 | CrashHD | *slap in the face* |
| 04:59.26 | dlynes_laptop | HT is fake smp |
| 04:59.36 | dlynes_laptop | it's not real smp |
| 04:59.38 | nailbags|work | dlynes_laptop: ah, i didn't know that. but all my askerisk hardware is too old for HT |
| 05:00.04 | dlynes_laptop | nailbags, yeah..everyone i see on here having severe hardware issues is usually running HT |
| 05:00.32 | dlynes_laptop | besides |
| 05:00.40 | dlynes_laptop | I would never ever ever run a server on a PIV |
| 05:00.43 | *** join/#asterisk ronaldl79 (n=chatzill@d198-53-139-22.abhsia.telus.net) |
| 05:00.53 | nailbags|work | dlynes_laptop: do you know why it doesn't play nice? is it the driver? or is it asterisk not being threaded in such a way to use HT well? |
| 05:00.53 | CrashHD | ? |
| 05:00.55 | ronaldl79 | Howdy |
| 05:01.02 | dlynes_laptop | It's just asking for trouble running a PIV on a telecom server |
| 05:01.10 | CrashHD | dlynes_laptop: you'll have to sustantiate that comment now |
| 05:01.14 | CrashHD | lol |
| 05:01.17 | CrashHD | I'm sooo curious |
| 05:01.19 | CrashHD | as to why |
| 05:01.21 | dlynes_laptop | CrashHD, the HT, or the PIV? |
| 05:01.24 | Qwell | Because it's Intel :P |
| 05:01.26 | CrashHD | P4 |
| 05:01.37 | CrashHD | and do you feel the same for xeon? |
| 05:01.41 | Qwell | CrashHD: P4 is for desktops, not telecom servers |
| 05:01.42 | dlynes_laptop | P4's when they get overloaded kick out every clock cycle to cool down |
| 05:01.53 | dlynes_laptop | erm every other clock cycle i mean |
| 05:01.53 | Qwell | dlynes_laptop: nice |
| 05:02.10 | dlynes_laptop | CrashHD, no, the problem doesn't exist for Xeons...only the regular PIV's |
| 05:02.14 | [TK]D-Fender | Yeah, everyone knows C3 is where it's at, right dlynes_laptop? ;) |
| 05:02.22 | dlynes_laptop | [TK]D-Fender, lol |
| 05:02.22 | CrashHD | via c3? |
| 05:02.30 | dlynes_laptop | CrashHD, major POS's |
| 05:02.33 | CrashHD | lol |
| 05:02.37 | file | StrongARM... that's where it's @ |
| 05:02.38 | nailbags|work | is asterisk better with preempt on or off? |
| 05:02.43 | nailbags|work | off right? |
| 05:02.58 | dlynes_laptop | nailbags, ummmm...you can't even run asterisk on a real mode OS |
| 05:03.13 | nailbags|work | file: going to try running asterisk on an IXP420 soon |
| 05:03.35 | nailbags|work | dlynes_laptop: huh? |
| 05:03.53 | dlynes_laptop | preempt on/off == preemptive multitasking right? |
| 05:04.18 | nailbags|work | dlynes_laptop: nah i meant pre-emtable linux kernel |
| 05:04.42 | dlynes_laptop | aren't all linux kernels preemptable? |
| 05:05.07 | nailbags|work | dlynes_laptop: no you can choose whether the kernel can be preempted or not (not talking about userspace) |
| 05:05.16 | dlynes_laptop | ah |
| 05:06.00 | CrashHD | dlynes_laptop: wanpipe beta6 seems to be ok |
| 05:06.04 | CrashHD | today was the first day running it |
| 05:06.08 | CrashHD | running some fax tests now |
| 05:06.22 | dlynes_laptop | CrashHD, thought you said you were having issues with it not allowing you to ssh in? |
| 05:06.26 | CrashHD | the only thing I notice is there is a slight hiccup when I make an outbound call to our carrier |
| 05:06.37 | CrashHD | well that could be the driver |
| 05:06.46 | ronaldl79 | Any chatter about the progress of 1.4 beta? |
| 05:06.49 | CrashHD | I was speaking more to the timing issues pops/clicks we were seeing |
| 05:07.01 | dlynes_laptop | ah |
| 05:07.06 | CrashHD | they found that when clocking = MASTER for any card |
| 05:07.09 | CrashHD | the 1.4 would add bits |
| 05:07.09 | *** join/#asterisk forrestc{hm} (n=forrestc@206.127.117.45) |
| 05:07.13 | *** join/#asterisk brc_ (n=brc_@pdpc/supporter/basic/brc) |
| 05:07.14 | CrashHD | that would cause bad timing |
| 05:07.16 | CrashHD | and frame slips |
| 05:07.39 | CrashHD | you are probably just a slave to your telco? |
| 05:07.43 | CrashHD | so it wouldn't have affected you |
| 05:08.16 | dlynes_laptop | yeah |
| 05:08.28 | dlynes_laptop | It's hooked into a pri at the telco colo |
| 05:08.33 | *** join/#asterisk _santiago_ (i=santiago@debian/developer/santiago) |
| 05:08.57 | forrestc{hm} | so what we trying to troubleshoot tonight? |
| 05:09.14 | *** join/#asterisk Un1x (i=Sean@72.61.82.242) |
| 05:09.16 | CrashHD | what is our purpose in life |
| 05:09.29 | dlynes_laptop | forrestc{hm}, you can troubleshoot my memory leaks, my SIP 489's and SIP 500's, BLF, and the world of flakiness in general |
| 05:09.37 | CrashHD | lol |
| 05:09.46 | CrashHD | inconsistant bugs |
| 05:09.48 | CrashHD | them bastages |
| 05:09.58 | CrashHD | also |
| 05:10.00 | CrashHD | another question |
| 05:10.02 | CrashHD | just popped in my head |
| 05:10.06 | forrestc{hm} | Nah... I have enough problem with my Linksys ATA's. |
| 05:10.18 | forrestc{hm} | And echo. |
| 05:10.19 | dlynes_laptop | forrestc{hm}, get a sipura 2002 instead |
| 05:10.19 | CrashHD | these astra phones |
| 05:10.26 | CrashHD | waht dtmf mode should I be running? |
| 05:10.31 | dlynes_laptop | forrestc{hm}, you'll see it's much easier to work with than the PAP2-NA's |
| 05:10.35 | CrashHD | for all my asterisk servers in general |
| 05:10.44 | dlynes_laptop | CrashHD, rfc2833 |
| 05:10.47 | forrestc{hm} | and in what way would a 2002 be easier? |
| 05:11.16 | dlynes_laptop | forrestc{hm}, the software works better, and the dtmf (****) menu actually works |
| 05:11.33 | CrashHD | fun |
| 05:11.36 | CrashHD | I'm running that |
| 05:11.41 | CrashHD | dtmf doesn't work very well lol |
| 05:11.42 | dlynes_laptop | forrestc{hm}, i was never able to get **** to work on the PAP2-NA's, personally |
| 05:11.49 | *** join/#asterisk linlin (n=linlin@c-67-184-229-183.hsd1.il.comcast.net) |
| 05:12.03 | dlynes_laptop | forrestc{hm}, has the code changed for those devices, or do you know? |
| 05:12.20 | forrestc{hm} | I've got about 60 -NA's (PAP2, RT31P2, WRT54GP2 mix), and they all seem to work well, excep the @(#*$ echo issue. |
| 05:12.38 | forrestc{hm} | I'm running something quite new . |
| 05:12.43 | forrestc{hm} | Several weeks old. |
| 05:12.44 | dlynes_laptop | yeah...I never had an issue with echo on the 2000's, 2002's, or PAP2-NA's |
| 05:12.48 | CrashHD | you know what I need |
| 05:12.50 | CrashHD | for asterisk |
| 05:12.56 | dlynes_laptop | CrashHD, a nice buxom blonde? |
| 05:13.13 | CrashHD | that big machine that answers questions from the movie "hitch hikers guide to the galaxy" |
| 05:13.18 | CrashHD | "computer, answer me this" |
| 05:13.24 | CrashHD | although |
| 05:13.26 | CrashHD | knowing asterisk |
| 05:13.27 | forrestc{hm} | The problem is echo cancellation in the local hybrid. AKA echo is toward the PSTN |
| 05:13.41 | CrashHD | it would spit out "please come back in 100,000 million years for the correct answer" |
| 05:13.51 | dlynes_laptop | forrestc{hm}, you're running 3.1.7? |
| 05:13.53 | forrestc{hm} | ATA user doesn't hear echo PSTN user does. |
| 05:14.04 | *** join/#asterisk Clausian (i=reginald@203-206-55-63.dyn.iinet.net.au) |
| 05:14.30 | dlynes_laptop | how is pstn connected? Are you using an asterisk box? |
| 05:15.05 | forrestc{hm} | 3.1.10 right now |
| 05:15.11 | CrashHD | and there is not dtmf mode setting for iax correct? |
| 05:15.18 | dlynes_laptop | CrashHD, correct |
| 05:16.27 | CrashHD | what are the hidden key combo's that asterisk uses? |
| 05:16.29 | CrashHD | like ##? |
| 05:16.37 | dlynes_laptop | umm |
| 05:16.39 | forrestc{hm} | I know that 3.1.12 is out, haven't had a chance to get it on there. |
| 05:16.40 | dlynes_laptop | it doesn't have any? |
| 05:16.49 | CrashHD | hmm |
| 05:16.58 | CrashHD | something is making dtmf not be sent |
| 05:17.01 | dlynes_laptop | CrashHD, i'm guessing you mean easter eggs? |
| 05:17.01 | forrestc{hm} | 3.1.7 had stability issues which 3.1.10 seems to be better with . |
| 05:17.03 | CrashHD | thought it could be that |
| 05:17.06 | CrashHD | lol |
| 05:17.28 | dlynes_laptop | forrestc{hm}, ah...yeah...i'm happily plugging along on 3.1.7 without any issues |
| 05:18.00 | dlynes_laptop | forrestc{hm}, what do you have to terminate onto pstn? |
| 05:18.29 | *** join/#asterisk Chris-NB (n=chris@ng1.kurtkrenn.com) |
| 05:18.30 | forrestc{hm} | PRI's |
| 05:18.42 | dlynes_laptop | forrestc{hm}, and do you have echo cancellation on the pri's? |
| 05:18.46 | forrestc{hm} | yes |
| 05:19.04 | dlynes_laptop | forrestc{hm}, have you tried changing the ec algorithm? Or are you using hwec? |
| 05:19.07 | forrestc{hm} | And the echo is *for sure* coming from the PAP'2's since I can hear it on other SIP/IAX phones. |
| 05:19.16 | forrestc{hm} | And on local calls. |
| 05:19.16 | dlynes_laptop | forrestc{hm}, ah |
| 05:19.22 | dlynes_laptop | forrestc{hm}, what I would check then |
| 05:19.33 | dlynes_laptop | forrestc{hm}, is the handset on the phone that's connected to the sipura unit |
| 05:19.39 | reticent | you don't happen to have an impedence mismatch between the handsets and ata? |
| 05:20.04 | dlynes_laptop | forrestc{hm}, try replacing the handset, and see if that fixes the problem |
| 05:20.21 | forrestc{hm} | I've played with the impedance settings. We've swapped PAP'2s We've had 6-7 phones. It happens on *all* of the ATA's. |
| 05:20.35 | forrestc{hm} | Not every call. |
| 05:20.50 | forrestc{hm} | It's like the PAP2 EC firmware just plain sucks. |
| 05:21.15 | dlynes_laptop | forrestc{hm}, how long is the line cord between the pap2's and the phones? |
| 05:21.16 | forrestc{hm} | Ican make 3 calls from the same SIP phone to the PAP2 with no other changes... 2 of the calls will echo, then one won't |
| 05:21.26 | forrestc{hm} | 3' |
| 05:21.28 | forrestc{hm} | 2' |
| 05:21.31 | dlynes_laptop | ah |
| 05:21.34 | dlynes_laptop | so nothing major then |
| 05:21.44 | dlynes_laptop | send the units back |
| 05:21.48 | dlynes_laptop | ask for replacements |
| 05:21.49 | forrestc{hm} | Like I said, this is around 60 different PAP units. |
| 05:22.02 | forrestc{hm} | from 3 different batches. |
| 05:22.02 | dlynes_laptop | ok, or ask for a huge refund |
| 05:22.19 | dlynes_laptop | do you have syslogging turned on? |
| 05:22.33 | forrestc{hm} | I think so. |
| 05:22.41 | dlynes_laptop | have you checked your syslog? |
| 05:23.01 | forrestc{hm} | yep.. We're actually running 56K modem calls through the same asterisk box. |
| 05:23.04 | dlynes_laptop | well, i guess the answer to the second question is no |
| 05:23.12 | dlynes_laptop | because you didn't know if you had it turned on, or not :) |
| 05:23.29 | forrestc{hm} | Yes, I'm syslogging.. |
| 05:23.38 | forrestc{hm} | AM I syslogging the right stufff.. Well, maybe. |
| 05:23.56 | dlynes_laptop | and no abnormalities showing up in the logs? |
| 05:23.58 | forrestc{hm} | no problems.. No dropped modem calls. |
| 05:24.01 | forrestc{hm} | that is correct. |
| 05:24.15 | dlynes_laptop | have you tried fiddling with the ec settings on the ata? |
| 05:24.22 | nailbags|work | forrestc{hm}: i have 11 pap2s from 3 different batches and have never had echo |
| 05:24.25 | forrestc{hm} | Yep, every knowb. |
| 05:24.48 | dlynes_laptop | weird |
| 05:24.53 | forrestc{hm} | have *you* never heard echo (as the PAP2 user) or has the people calling you never heard eacho. |
| 05:25.00 | forrestc{hm} | er echo |
| 05:25.20 | forrestc{hm} | BTW silence supression is evil on a PAP2. |
| 05:25.22 | dlynes_laptop | forrestc{hm}, the only time anyone on my system ever had echo from a pap2 or sipura 2000 or sipura 2002 was when I had an issue with EC on the pri |
| 05:25.30 | nailbags|work | forrestc{hm}: i've tested them thoroughly. i call that office every days and talk to someone who is on a pap2 |
| 05:25.35 | dlynes_laptop | forrestc{hm}, maybe because asterisk doesn't support silence suppression? |
| 05:25.45 | forrestc{hm} | what I mean was "echo supression" |
| 05:25.59 | forrestc{hm} | 2 different beasts |
| 05:26.20 | forrestc{hm} | The lastest .10 firmware does seem to have made it better. |
| 05:26.29 | dlynes_laptop | try .12 |
| 05:26.33 | dlynes_laptop | see if it makes it worse |
| 05:26.38 | forrestc{hm} | That's where I'm going next. |
| 05:27.56 | forrestc{hm} | The other thing I'm going to do is to capture the rtp stream with tethereal and listen to it directly to 100% verify the echo is where I think it is. |
| 05:28.02 | CrashHD | hmmmmmmmm |
| 05:28.03 | CrashHD | odd |
| 05:28.17 | CrashHD | there is no rfc2833 option in the astra web interface (config) |
| 05:28.22 | CrashHD | only rtp and info |
| 05:28.37 | CrashHD | there is a check box for rfc2833 though |
| 05:28.54 | CrashHD | but the dtmf method selection doesn't include it |
| 05:29.00 | dlynes_laptop | CrashHD, rtp == rfc2833 |
| 05:29.14 | CrashHD | ahh |
| 05:29.18 | CrashHD | thought rtp was in band |
| 05:29.28 | dlynes_laptop | CrashHD, rtp == out of band |
| 05:29.29 | CrashHD | isn't sip info out of band? |
| 05:29.38 | dlynes_laptop | CrashHD, no...sip info is sip info |
| 05:29.50 | dlynes_laptop | CrashHD, rfc2833 is also known as AVT and out of band |
| 05:30.08 | CrashHD | hmm |
| 05:30.10 | CrashHD | all confused now |
| 05:30.13 | CrashHD | so what is inband |
| 05:30.14 | CrashHD | lol |
| 05:30.21 | dlynes_laptop | inband is within the audio |
| 05:30.24 | CrashHD | ya |
| 05:30.26 | CrashHD | but how would I set it |
| 05:30.29 | dlynes_laptop | it's only possible with ulaw and alaw |
| 05:30.33 | CrashHD | to do that |
| 05:30.41 | CrashHD | cause I have three options |
| 05:30.45 | dlynes_laptop | one second |
| 05:30.49 | *** join/#asterisk benjk (n=benjamin@f8a01-0357.din.or.jp) |
| 05:30.59 | CrashHD | well 1 check box and a drop down with sip info, rtp and auto |
| 05:31.12 | dlynes_laptop | CrashHD, which firmware version are you using? |
| 05:31.13 | CrashHD | so you have eliminated any of those options being inband |
| 05:31.16 | CrashHD | 1.4 |
| 05:31.27 | forrestc{hm} | sip info = rfc2833 |
| 05:31.31 | forrestc{hm} | I believe |
| 05:31.34 | dlynes_laptop | forrestc{hm}, no |
| 05:31.37 | CrashHD | heh |
| 05:31.45 | dlynes_laptop | forrestc{hm}, sip info and rfc2833 are two completely different beasts |
| 05:31.56 | dlynes_laptop | forrestc{hm}, they're both sent out of band |
| 05:32.07 | forrestc{hm} | ok. |
| 05:32.14 | forrestc{hm} | I thought I was wrong as soon as I had said it. |
| 05:32.19 | dlynes_laptop | lol |
| 05:32.35 | dlynes_laptop | anyways...inband is dtmf tones as you hear it in the phone |
| 05:32.50 | dlynes_laptop | sip info and rfc2833 are both voip methods of relaying dtmf |
| 05:33.13 | dlynes_laptop | rfc2833 relays it inside the rtp I think |
| 05:33.20 | CrashHD | so how would I set the phone to use inband |
| 05:33.23 | dlynes_laptop | and sip info relays it as xml messages within the sip protocol |
| 05:33.26 | CrashHD | they don't make that an option is seems |
| 05:33.31 | dlynes_laptop | CrashHD, correct |
| 05:33.32 | forrestc{hm} | dlynes: I pulled rfc2833 and that's right. |
| 05:33.52 | dlynes_laptop | forrestc{hm}, i don't understand what you said |
| 05:33.55 | dlynes_laptop | forrestc{hm}, pulled? |
| 05:34.07 | forrestc{hm} | rfc2833 = dtmf in RTP headers or "RTP Payload for DTMF Digits, Telephony Tones and Telephony Signals" |
| 05:34.15 | forrestc{hm} | pulled as in looked at |
| 05:34.19 | dlynes_laptop | ah |
| 05:34.50 | CrashHD | so rtp |
| 05:34.51 | CrashHD | is udp |
| 05:34.51 | dlynes_laptop | yeah...like i said rfc2833 == out of band/rtp dtmf/avt |
| 05:35.02 | CrashHD | aren't the sip headers tcp? |
| 05:35.02 | dlynes_laptop | CrashHD, not necessarily |
| 05:35.08 | CrashHD | well rtp with asterisk is udp |
| 05:35.11 | dlynes_laptop | CrashHD, you can send rtp over tcp, too |
| 05:35.19 | dlynes_laptop | CrashHD, asterisk just doesn't do tcp |
| 05:35.23 | CrashHD | *nods* |
| 05:35.32 | CrashHD | aren't the sip messages sent over tcp? |
| 05:35.38 | dlynes_laptop | CrashHD, no |
| 05:35.51 | CrashHD | ahh no |
| 05:35.55 | CrashHD | 5060 udp |
| 05:36.03 | CrashHD | took the wind right out of them sails |
| 05:36.26 | CrashHD | and rtp is better than info because....? |
| 05:36.35 | forrestc{hm} | it works? |
| 05:36.41 | dlynes_laptop | because 99.9% of your upstreams use it |
| 05:36.48 | dlynes_laptop | almost nobody uses sip info |
| 05:37.01 | dlynes_laptop | why convert, if you don't need to? |
| 05:37.06 | *** part/#asterisk suma (n=kans@61.14.86.23) |
| 05:37.15 | CrashHD | *shrugs* I'm always the guy that asked why |
| 05:37.25 | dlynes_laptop | also, sip info doesn't say how long the tones are |
| 05:37.25 | CrashHD | telcommone actually uses sip info |
| 05:37.32 | dlynes_laptop | it just says play this tone |
| 05:37.54 | dlynes_laptop | with rfc2833, when you create the dtmf message, you can state how long it is |
| 05:38.15 | CrashHD | what would the "both" setting do.... |
| 05:38.16 | CrashHD | ? |
| 05:38.28 | dlynes_laptop | it means autodetect what is being sent |
| 05:38.41 | dlynes_laptop | and what the receiving host supports |
| 05:38.51 | dlynes_laptop | I always force everything to rtp, personally |
| 05:39.03 | dlynes_laptop | I was having issues with info working properly on Aastras |
| 05:39.41 | CrashHD | well your experience is good enough for me |
| 05:40.21 | dlynes_laptop | well, those problems were with 1.3.1 firmware |
| 05:40.26 | dlynes_laptop | I haven't bothered to try it on 1.4.0 |
| 05:40.36 | dlynes_laptop | i figure if it works on 1.3.1 i'm gonna use it on 1.4.0 too |
| 05:40.42 | CrashHD | :) |
| 05:40.42 | dlynes_laptop | I don't feel like gambling |
| 05:40.46 | CrashHD | 1.4 is nice |
| 05:40.50 | CrashHD | so far anyway |
| 05:40.58 | dlynes_laptop | well |
| 05:41.10 | dlynes_laptop | blf is still a super buggy piece of fscking sh*t |
| 05:41.22 | CrashHD | working great for my call parking |
| 05:41.22 | dlynes_laptop | i'd stay well away from it for now |
| 05:41.31 | CrashHD | not monitoring extensions though |
| 05:41.39 | dlynes_laptop | yeah...the aastras in general are working fine for me, too |
| 05:41.50 | CrashHD | blf for my call parking is working great that is |
| 05:41.58 | dlynes_laptop | i've got 22 aastra 9133i's deployed |
| 05:42.08 | dlynes_laptop | erm 23 i mean |
| 05:42.16 | CrashHD | we had 10 we were testing |
| 05:42.20 | CrashHD | just sent those to a client |
| 05:42.22 | CrashHD | and using 480's now |
| 05:42.24 | dlynes_laptop | 6 at one customer's, 6 at another's, and 11 at another's |
| 05:42.26 | CrashHD | WARNING: Freeing unused memory at (nil), in ast_yyfree of ast_expr2f.c, line 2882 |
| 05:42.30 | CrashHD | keeps coming up |
| 05:42.38 | dlynes_laptop | COOL BEANS!!!! |
| 05:43.01 | CrashHD | freeing memory is a good thing I'm guessing |
| 05:43.05 | CrashHD | warning? |
| 05:43.06 | dlynes_laptop | yeah |
| 05:43.09 | *** join/#asterisk Assid (i=assid@203.115.83.214) |
| 05:43.09 | CrashHD | how about notice |
| 05:43.16 | dlynes_laptop | but freeing memory from an unknown location? |
| 05:43.33 | dlynes_laptop | i woudl tend to think that's not a good thing |
| 05:43.40 | *** join/#asterisk pdtmobile (n=ptinsley@c-68-53-40-50.hsd1.tn.comcast.net) |
| 05:43.45 | CrashHD | hence the "warning" I suppose? |
| 05:43.45 | dlynes_laptop | I would take a look at that line of code |
| 05:44.14 | CrashHD | it would look like shangralian to me |
| 05:44.19 | CrashHD | lol |
| 05:44.39 | dlynes_laptop | ah |
| 05:44.46 | *** join/#asterisk MikeJ (n=vircuser@d14-69-8-30.try.wideopenwest.com) |
| 05:44.49 | dlynes_laptop | you don't know how to program at all? |
| 05:45.10 | CrashHD | a bit of java |
| 05:45.12 | CrashHD | mostly php though |
| 05:45.19 | CrashHD | I can make my way around C |
| 05:45.20 | dlynes_laptop | so just take a look at the code then |
| 05:45.20 | CrashHD | to read through it |
| 05:45.24 | dlynes_laptop | just read it |
| 05:45.26 | CrashHD | brb I will |
| 05:45.27 | dlynes_laptop | not asking you to change it |
| 05:46.15 | CrashHD | free( (char *) ptr ); /* see ast_yyrealloc() for (char *) cast */ |
| 05:46.22 | CrashHD | checking function now |
| 05:47.27 | CrashHD | return (void *) realloc( (char *) ptr, size ); |
| 05:47.55 | CrashHD | looks like it is adjusting memory? |
| 05:48.06 | dlynes_laptop | file, you still there? |
| 05:48.17 | CrashHD | yyalloc malloc's( size) |
| 05:48.49 | dlynes_laptop | not sure...haven't looked at it, myself |
| 05:49.09 | CrashHD | then realloc runs to adjust the mem space |
| 05:49.12 | *** join/#asterisk Stephnie (i=Stephnie@u15157627.onlinehome-server.com) |
| 05:49.15 | CrashHD | then yyfree free's up the old mem space |
| 05:49.17 | CrashHD | just a guess though |
| 05:49.20 | Stephnie | hi |
| 05:49.28 | dlynes_laptop | Good morning, Steph |
| 05:49.36 | Stephnie | gud morning |
| 05:49.43 | *** join/#asterisk Un1x (i=Sean@72.61.82.242) |
| 05:50.24 | Stephnie | SIP/outgoing-08899ec0 is ringing ......but I dont get rings in my X-lite |
| 05:50.29 | *** join/#asterisk colarte (n=colarte@200.124.175.245) |
| 05:51.03 | dlynes_laptop | Stephnie, perhaps your sip/outgoing trunk is not passing along call progression to you |
| 05:51.53 | CrashHD | ahh crap |
| 05:51.56 | CrashHD | that's the other thing I forgot to do |
| 05:52.06 | CrashHD | need to handle busy's being sent back to the user |
| 05:52.12 | Stephnie | dlynes_laptop : any solution ? |
| 05:52.55 | dlynes_laptop | Stephnie, well, if it's an issue of your sip provider not passing call progression back to you, i'd bitch and complain at them |
| 05:53.16 | Stephnie | :-/ |
| 05:53.20 | colarte | hello, can it be possible to connect all existing normal telephone extensions (working on a normal pbx) to an asterisk based pbx server?? |
| 05:53.23 | dlynes_laptop | Stephnie, so as i understand it, you're not getting ringing at all? |
| 05:53.45 | dlynes_laptop | Stephnie, or you're just not getting ringing for ten or fifteen seconds or so? |
| 05:53.45 | *** part/#asterisk forrestc{hm} (n=forrestc@206.127.117.45) |
| 05:53.59 | Stephnie | not getting ringing at all |
| 05:53.59 | dlynes_laptop | colarte, digital or analog extensions? |
| 05:54.15 | colarte | analog .. |
| 05:54.18 | *** join/#asterisk redondos_ (n=redondos@190.48.37.12) |
| 05:54.19 | dlynes_laptop | Stephnie, do you get the other end asnwering at all? |
| 05:54.28 | dlynes_laptop | colarte, yeah...it's straight forward, then |
| 05:54.36 | Stephnie | yes the otherend is answering |
| 05:54.49 | dlynes_laptop | Stephnie, so you can hear them pick up then? |
| 05:54.56 | dlynes_laptop | Stephnie, and you can talk to them? |
| 05:55.02 | Stephnie | hmmmm let me be sure :) |
| 05:55.20 | dlynes_laptop | Stephnie, call your cell phone then |
| 05:55.30 | dlynes_laptop | Stephnie, that way you know the exact result |
| 05:55.47 | dlynes_laptop | Stephnie, or your home phone if you have an analog phone, is even better |
| 05:56.44 | dlynes_laptop | colarte, as i understand it though, it's better to use channel banks and a pri card, as the fxo->fxs bridging is apparently not very good |
| 05:58.12 | Stephnie | dlynes_laptop : yep.....problem ... destination cant hear .... |
| 05:58.13 | CrashHD | brb |
| 05:58.34 | dlynes_laptop | Stephnie, can the originator hear? |
| 05:58.45 | Stephnie | dlynes_laptop : yes |
| 05:58.59 | dlynes_laptop | Stephnie, you've got at least one end behind a router, right? |
| 05:59.50 | Stephnie | well...I remember when I change the TRUNK ...it works...but let me change carrier and check.. |
| 06:00.01 | dlynes_laptop | trunk? |
| 06:00.36 | dlynes_laptop | what are you referring to as your trunk? |
| 06:00.44 | Stephnie | I mean carrier :) |
| 06:00.54 | *** join/#asterisk brc__ (n=brc_@pdpc/supporter/basic/brc) |
| 06:01.59 | dlynes_laptop | holy freaking heat wave |
| 06:04.00 | Stephnie | yep problem is only with the rings....I have set NAT=YES so that both can hear |
| 06:04.40 | Stephnie | now the problem is only with the ring....I have two different carrier ...one provides ringing but other doesnt.. |
| 06:10.36 | *** join/#asterisk wunderkin (n=kev@ip68-226-113-228.ph.ph.cox.net) |
| 06:10.39 | dlynes_laptop | Stephnie, are you using port mapping? |
| 06:10.48 | dlynes_laptop | Stephnie, and are you using canreinvite=no? |
| 06:11.22 | *** join/#asterisk colarte (n=colarte@200.124.175.245) |
| 06:11.24 | Stephnie | no |
| 06:11.31 | dlynes_laptop | no to both? |
| 06:11.38 | Stephnie | yes no to both |
| 06:11.54 | dlynes_laptop | set revinite=no |
| 06:12.00 | Assid | k back |
| 06:12.04 | Assid | router issue dlynes_laptop? |
| 06:12.08 | dlynes_laptop | Assid, is that a threat? |
| 06:12.19 | Stephnie | ok |
| 06:12.22 | Assid | yep.. im here to fart till i drop.. |
| 06:12.22 | dlynes_laptop | Assid, not for me...for steph |
| 06:12.41 | Assid | canreinvite=no |
| 06:12.49 | Assid | and also nat=yes or nat=route |
| 06:12.51 | Stephnie | reinvite= no in SIP/outgoing? |
| 06:13.07 | dlynes_laptop | Stephnie, no....canreinvite=no, not reinvite=no |
| 06:13.24 | Assid | dont forget to sip reload |
| 06:13.31 | Assid | and then remake the call |
| 06:15.24 | Stephnie | okey |
| 06:19.03 | CrashHD | hmmm |
| 06:19.13 | CrashHD | the # sound doesn't seem to be sent very well |
| 06:19.21 | CrashHD | but everythign else is fine |
| 06:20.26 | *** join/#asterisk daysmen3 (n=primus@host86-138-235-88.range86-138.btcentralplus.com) |
| 06:21.31 | Clausian | how can i check the registration status of asterisk? |
| 06:23.07 | *** join/#asterisk daysmen3 (n=primus@host86-138-235-88.range86-138.btcentralplus.com) |
| 06:23.40 | CrashHD | how do I debug rtp dtmf? |
| 06:24.33 | *** join/#asterisk Kumbang (n=kumbang@167.205.24.67) |
| 06:24.43 | Kumbang | hi |
| 06:25.04 | dlynes_laptop | CrashHD, rtp debug on |
| 06:25.18 | dlynes_laptop | Clausian, sip show registry |
| 06:25.44 | CrashHD | rtp debug ip |
| 06:25.48 | CrashHD | how do I debug all? |
| 06:25.56 | dlynes_laptop | eh? |
| 06:26.05 | CrashHD | rtp doesn't have a debug on? |
| 06:26.15 | dlynes_laptop | CrashHD, just "rtp debug" |
| 06:26.20 | dlynes_laptop | CrashHD, forget the ip |
| 06:26.31 | Kumbang | guys, how can i register g729 codec in astlinux |
| 06:26.59 | CrashHD | the damn dtmf just isn't reliable |
| 06:27.01 | *** join/#asterisk drega (n=dforeman@mail.amcat.co.uk) |
| 06:27.05 | CrashHD | sometimes it picks up fine |
| 06:27.08 | CrashHD | other times it doesn't |
| 06:27.10 | CrashHD | any suggestions |
| 06:27.18 | *** join/#asterisk kmilitzer (n=km@office-gw.westend.com) |
| 06:27.29 | dlynes_laptop | CrashHD, i seem to recall some other peeps having issues with that, too |
| 06:27.31 | dlynes_laptop | CrashHD, one sec |
| 06:29.50 | *** join/#asterisk topping (n=topping@207.47.6.136.static.nextweb.net) |
| 06:32.04 | *** join/#asterisk Un1x (i=Sean@72.61.82.242) |
| 06:35.17 | dlynes_laptop | CrashHD, try this: http://bugs.digium.com/view.php?id=5970 |
| 06:35.22 | dlynes_laptop | anwyays....gotta run |
| 06:35.24 | dlynes_laptop | need coffee |
| 06:35.27 | CrashHD | thanks bud |
| 06:35.30 | CrashHD | will take a look |
| 06:35.31 | dlynes_laptop | need CAAAAFFFEIIINEE... |
| 06:35.37 | CrashHD | you and I both lol |
| 06:36.08 | *** join/#asterisk drega (n=dforeman@mail.amcat.co.uk) |
| 06:36.49 | benjk | dlynes, do you have any idea how to use pbx_builtin_setvar_helper() ? |
| 06:38.18 | dlynes_laptop | SetVar(benjk=null) |
| 06:38.24 | *** join/#asterisk RestLessGemini (n=rLg@202.141.247.150) |
| 06:38.34 | benjk | no that won't work |
| 06:38.55 | benjk | I added some code to libpri to fetch some data from the the PRI messages the telco sends |
| 06:39.21 | dlynes_laptop | sorry |
| 06:39.31 | benjk | and now I need to stuff it programmatically (in chan_zap.c) into a new channel variable so it becomes accessible in the dialplan |
| 06:39.32 | dlynes_laptop | Set(benjk=null) |
| 06:39.41 | dlynes_laptop | SetVar's deprecated |
| 06:39.56 | benjk | I am not talking about setting it inside the dialplan |
| 06:40.09 | *** join/#asterisk carrar (i=tim@osburn.com) |
| 06:40.10 | carrar | w00t |
| 06:40.24 | benjk | I am talking about creating a new dialplan variable that is set automatically by chan_zap |
| 06:40.25 | dlynes_laptop | benjk, oh |
| 06:40.38 | carrar | any have BLF on a Grandstream to monitor a ZAP channel? |
| 06:40.41 | dlynes_laptop | benjk, i would ask in #asterisk-dev, then |
| 06:40.43 | carrar | err anyone |
| 06:40.58 | *** join/#asterisk [Airwolf] (n=airwolf@dsl51B79ED6.pool.t-online.hu) |
| 06:41.01 | carrar | not to sure that even works |
| 06:41.04 | benjk | yeah I did already |
| 06:41.07 | dlynes_laptop | benjk, nvm...i see you've already asked there |
| 06:41.20 | benjk | I am so stuck with this |
| 06:41.29 | dlynes_laptop | benjk, well, if i wasn't heading out right away |
| 06:41.38 | dlynes_laptop | benjk, i'd probably be able to help |
| 06:41.58 | benjk | fair enough, thanks for trying anyway |
| 06:42.09 | dlynes_laptop | benjk, well, how long are you going to be here for? |
| 06:42.35 | benjk | I have it 3:40 pm so it'll be another 10 hours at least ;) |
| 06:44.20 | dlynes_laptop | ah, ok |
| 06:44.28 | dlynes_laptop | well i'll be back in about 2 hours or so, probably |
| 06:44.39 | dlynes_laptop | or sooner if i can get internet in the coffee shop |
| 06:44.45 | CrashHD | crap! |
| 06:44.45 | benjk | ok, then ttyl |
| 06:44.50 | dlynes_laptop | but if there's internet there, i probably won't have the time to delve into the code |
| 06:44.53 | CrashHD | ok so |
| 06:45.06 | CrashHD | so in the box that is doing the iax2 to sip conversion |
| 06:45.09 | CrashHD | to a sip provider |
| 06:45.21 | CrashHD | the dtmf is in stream I think |
| 06:45.36 | CrashHD | even though rfc2833 |
| 06:45.47 | CrashHD | is set |
| 06:45.49 | CrashHD | !!!!!!!!!!!!1 |
| 06:45.51 | CrashHD | wtf |
| 07:03.21 | *** join/#asterisk parag_ast (n=root@dxb-b16506.alshamil.net.ae) |
| 07:03.35 | parag_ast | anybody using polycom ip500 |
| 07:06.50 | E-bola | morning |
| 07:07.34 | *** join/#asterisk DarKnesS_WolF (n=wolf@212.103.170.135) |
| 07:10.00 | *** join/#asterisk vgster (n=vgster@cpc2-ledn1-0-0-cust944.leed.cable.ntl.com) |
| 07:15.44 | *** join/#asterisk UlbabraB (n=UlbabraB@host241-43-static.72-81-b.business.telecomitalia.it) |
| 07:16.08 | *** join/#asterisk kristalino (n=kristali@84-50-84-146-dsl.trt.estpak.ee) |
| 07:18.16 | *** join/#asterisk angom_h (n=angom@red-corp-200.38.30.201.telnor.net) |
| 07:18.26 | Assid | parag_ast: ask your qustion .. |
| 07:18.55 | Assid | sup dlynes |
| 07:25.34 | carrar | w00t |
| 07:26.04 | *** join/#asterisk qdk (n=qdk@213.237.44.34) |
| 07:26.32 | benjk | dlynes left |
| 07:27.26 | *** join/#asterisk L|NUX (n=linux@202.5.145.56) |
| 07:29.04 | *** join/#asterisk [pyro] (i=pyro@tor/regular/bracketed-pyro) |
| 07:29.38 | *** join/#asterisk ditesh|cassini (n=ditesh@60.51.217.61) |
| 07:32.34 | Clausian | Jul 26 17:17:10 WARNING[4422] loader.c: Loading module format_au.so failed! |
| 07:33.03 | Clausian | according to http://lists.digium.com/pipermail/asterisk-commits/2006-February/001447.html that was merged into another |
| 07:33.12 | Clausian | why is it trying to load it? |
| 07:33.47 | *** join/#asterisk X-Gen (n=X-Gen@dsl-145-255-112.telkomadsl.co.za) |
| 07:37.07 | russellb | Clausian: because you either have it listed in modules.conf, or you have not cleaned out your /usr/lib/asterisk/modules directory. |
| 07:38.17 | *** join/#asterisk SimoAmi (n=SimoAmi@207-38-248-221.c3-0.wsd-ubr1.qens-wsd.ny.cable.rcn.com) |
| 07:38.31 | SimoAmi | hi there |
| 07:38.46 | SimoAmi | quick question |
| 07:39.25 | SimoAmi | for the tdm2400 cards do I need a male or female amphenol cable? |
| 07:39.36 | Clausian | female |
| 07:39.54 | SimoAmi | so the card has a male? |
| 07:39.57 | Clausian | yes |
| 07:40.56 | SimoAmi | so to use with a 24 Port Telephone Patch Panel (2 conductor), a need a female to female cable ? |
| 07:41.07 | Clausian | yes |
| 07:41.18 | SimoAmi | nice, thanks |
| 07:41.28 | Clausian | np |
| 07:41.41 | SimoAmi | was going to make a mistake and buy a mle to female ;) |
| 07:41.47 | Clausian | :o |
| 07:43.16 | type0 | i hate amp cables |
| 07:43.19 | type0 | i have a bunch of them |
| 07:43.41 | type0 | and some dialogic analog hardware too |
| 07:51.30 | *** join/#asterisk moon06 (n=michael@cim06-1-82-228-240-97.fbx.proxad.net) |
| 07:53.00 | hads|home | > -------------------------- |
| 07:53.36 | waba | I'm a bit puzzled, sip.conf.sample has type=peer for proxy both in/out categories. Shouldn't it be user for [sip_proxy] and peer for [sip_proxy-out] ? |
| 07:53.53 | waba | (1.2 sample, haven't checked in trunk) |
| 07:54.48 | hads|home | Man I'm glad irssi has paste control :) |
| 07:56.05 | linlin | is it possible to play a shoutcast stream to a caller? |
| 07:56.27 | type0 | sure |
| 07:56.34 | linlin | cause when i'm at work and i'm at break i'd like to be able to tune into my favorite talk radio stream :) |
| 07:56.42 | type0 | yeah its easy |
| 07:57.02 | type0 | instead of setting mpg123 to playing an mp3 |
| 07:57.07 | type0 | have it play the pls of the station |
| 07:57.27 | linlin | know of any guides anywhere? |
| 07:57.38 | type0 | you shouldnt need a guide |
| 07:57.43 | type0 | do you know the pls file |
| 07:57.45 | type0 | for the radio station? |
| 07:57.47 | linlin | yeah |
| 07:57.52 | type0 | ok |
| 07:58.06 | type0 | are you talking about straight calling in |
| 07:58.10 | type0 | and getting a shoutcast stream? |
| 07:58.23 | *** join/#asterisk ToTo (n=ToTo@host212-207.pool872.interbusiness.it) |
| 07:58.35 | linlin | well right now i have an ivr, i'd like to add it to an option on the iv |
| 07:58.41 | parag_ast | How can i assign the Sip username and Password in Polycom IP 500 |
| 07:58.48 | linlin | but i know how to do that part |
| 07:59.05 | type0 | just add the pls to the mpg123 dir then |
| 07:59.47 | parag_ast | How can i assign the Sip username and Password in Polycom IP 500 ??? |
| 08:01.29 | linlin | ive never had to write anything to use mpg123 |
| 08:02.07 | waba | hm, that's type=peer too in the trunk's sip.conf.sample. That's wrong, isn't it ? Or ? |
| 08:03.18 | Assid | not sure of ip500 .. but should be similar to 501 |
| 08:03.31 | Assid | are you provisioning ??? or are you inputing the information straight tot he phone |
| 08:04.14 | parag_ast | there is no option in the phone to put |
| 08:04.34 | Assid | should be in administration |
| 08:06.06 | *** join/#asterisk Assid (i=assid@203.115.83.214) |
| 08:07.53 | parag_ast | where in administration |
| 08:07.55 | parag_ast | there is nothing |
| 08:08.27 | Assid | go to the menu, settings, advanced |
| 08:10.02 | parag_ast | no its not there |
| 08:10.06 | parag_ast | settings is not there |
| 08:10.12 | parag_ast | on menu> setup |
| 08:19.47 | *** join/#asterisk topping (n=topping@207.47.6.136.static.nextweb.net) |
| 08:30.13 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 08:33.07 | *** join/#asterisk Modcuts (n=bob@lan.proporta.com) |
| 08:41.46 | *** join/#asterisk creativx (n=creadure@196.82-134-19.bkkb.no) |
| 08:53.03 | *** join/#asterisk LapTop006 (n=laptop00@sparc006.chriskaine.com.au) |
| 08:53.33 | *** join/#asterisk Gunnar (n=gunnar@62.97.242.6) |
| 09:00.03 | docelmo | Does anyone know where I can get the swift code for asterisk? |
| 09:01.00 | *** join/#asterisk Azrael (n=Azrael@orion.negativeblue.com) |
| 09:02.56 | *** join/#asterisk abatista (n=Ariel@dsl-20-177.cofs.net) |
| 09:04.20 | *** join/#asterisk r_marvin (n=marvin@83.166.220.142) |
| 09:05.29 | r_marvin | i'd need some tips on enabling g723 on debian sarge, has anyone here used that? |
| 09:07.20 | *** join/#asterisk kumamoto (n=kumamoto@24-178-3-69.dhcp.ftwo.tx.charter.com) |
| 09:11.54 | *** join/#asterisk RoyK (n=roy@80.239.107.70) |
| 09:12.40 | *** join/#asterisk speedwagon (n=Ariel@dsl-20-177.cofs.net) |
| 09:20.34 | x86 | r_marvin: you need to purchase licenses from digium |
| 09:21.03 | r_marvin | yes, i just found out |
| 09:22.00 | r_marvin | are there evaluation licenses? i'll need to make some tests before buying |
| 09:22.12 | r_marvin | g729 too, preferably |
| 09:22.29 | *** join/#asterisk dlynes_laptop (n=dlynes@zz212094.cipherkey.net) |
| 09:22.53 | dlynes_laptop | good morning, peeps |
| 09:27.20 | hads|home | mornin dlynes_laptop |
| 09:27.40 | hads|home | Early morning for you no? |
| 09:27.51 | dlynes_laptop | yep |
| 09:27.59 | dlynes_laptop | almost 2:30am |
| 09:28.09 | hads|home | How'd the house/apartment end up after the water? |
| 09:28.13 | dlynes_laptop | planning to stay up most of the night |
| 09:28.37 | hads|home | Getting too much sleep lately or something? :) |
| 09:28.51 | dlynes_laptop | nah...not enough |
| 09:28.56 | dlynes_laptop | too much work and not enough time |
| 09:29.06 | hads|home | Hate that |
| 09:29.10 | dlynes_laptop | and the stupid sprinkler that went off in our house the other day didn't help matters |
| 09:29.29 | hads|home | 21:28:05 < hads|home> How'd the house/apartment end up after the water? |
| 09:29.34 | kumamoto | anyone taken that dCAP certification thingie |
| 09:29.57 | docelmo | yes why? |
| 09:30.30 | dlynes_laptop | hads, about $25K worth of damage |
| 09:30.38 | hads|home | Ouch. |
| 09:31.02 | dlynes_laptop | hads, well, that's almost purely damage to the landlord's property and the house |
| 09:31.16 | dlynes_laptop | almost zero problems with my stuff |
| 09:31.23 | dlynes_laptop | luckily it was in one of the front rooms |
| 09:31.33 | dlynes_laptop | if it was in the back rooms, my computers would have been toast |
| 09:31.40 | hads|home | Lucky |
| 09:31.43 | kumamoto | can pass it using experience or you have to go to one of those bootcamp or buy the oreilly asterisk book> |
| 09:31.45 | dlynes_laptop | yep |
| 09:31.46 | benjk | toast? |
| 09:31.49 | benjk | from the water? |
| 09:31.50 | docelmo | ahh well back to dev.. |
| 09:32.04 | benjk | toast from the heat, I'd understand |
| 09:32.04 | dlynes_laptop | benjk, well, last i checked, computers didn't like water too much |
| 09:32.19 | benjk | yeah, but toast? |
| 09:32.21 | docelmo | kumamoto I had 2 years solid experience with asterisk before taking the test. I dont know if the bootcamp or book will help you |
| 09:32.38 | benjk | if its hot water, then boiled might be the right word |
| 09:32.46 | docelmo | my experience is some dev, admin, and configuration |
| 09:34.48 | kumamoto | docelmo: was it daunting when you started using asterisk? |
| 09:36.48 | *** join/#asterisk ghenry (n=ghenry@suretecsystems.plus.com) |
| 09:37.04 | *** join/#asterisk MikeJ (n=vircuser@d14-69-8-30.try.wideopenwest.com) |
| 09:37.32 | Assid | docelmo: you using realtime? |
| 09:37.58 | *** join/#asterisk viler (i=1000@200.114.70.228) |
| 09:38.07 | docelmo | yep |
| 09:38.13 | docelmo | a highly modded version but yes |
| 09:38.22 | Assid | modded version? |
| 09:38.34 | docelmo | My version isnt part of the source tree |
| 09:39.10 | Assid | i c |
| 09:39.19 | benjk | eggshell-aluminum casing with speclial glitter spray colour |
| 09:39.28 | creativx | <3 glitter |
| 09:39.41 | Assid | docelmo: pgsql database im guessing? |
| 09:40.23 | kumamoto | I was wondering if anyone has experience with polycom ip phones |
| 09:40.59 | Assid | kumamoto: just ask your question regarding it.. if anyone knows.. hey might help you |
| 09:40.59 | Assid | brb.. |
| 09:42.37 | RoyK | http://karlsbakk.net/fun/Spyware_infestation.png |
| 09:42.46 | kumamoto | my phone an ip301 has 2 lines for the love of all things digital can seem to 1 extension to each line |
| 09:43.05 | kumamoto | each line with it's own extension |
| 09:45.49 | dlynes_laptop | docelmo, thought you were going back to work? :) |
| 09:46.47 | Assid | kumamoto: are you provisioning it? |
| 09:47.15 | docelmo | I am.. Im coding and I am not liking it. And I need a question answered bout ast_verbose() |
| 09:48.02 | Assid | RoyK: believe or not.. ihad a friend who used to have a desktop like that |
| 09:48.08 | Assid | with bonzibuddy and everything |
| 09:48.44 | Assid | he had this funny thing.. if he finds something interactive or remotely interesting.. he will install it |
| 09:49.45 | RoyK | hehe. http://www.ronpatrickstuff.com/ |
| 09:49.50 | Assid | there was this other software too he had.. with the animated girls stripping.. something similar to bonzibuddy |
| 09:52.12 | *** join/#asterisk Sonderblade (n=mah@static-213.131.147.169.addr.tdcsong.se) |
| 09:54.38 | *** join/#asterisk cluetrain (n=root@212.33.151.25) |
| 09:55.53 | Assid | damn.. i want that car |
| 09:57.01 | *** join/#asterisk Chris-NB (n=chris@ng1.kurtkrenn.com) |
| 09:58.05 | dlynes_laptop | docelmo, how are you trying to use it? |
| 09:58.38 | docelmo | I have ast_verbose set to ast_verbose_3 btu when I set verbose to 0 in the cli shit still shows.. Thats my problem |
| 09:58.50 | RoyK | Assid: check the scooter at the bottom of the page :) |
| 09:59.18 | Assid | yeah i did |
| 09:59.28 | Assid | look mean!.. i wonder if i can add those to my bike |
| 10:00.26 | Assid | http://www.mypulsar.com/ <-- my bike |
| 10:00.43 | Assid | puny underpowered crap |
| 10:00.56 | CrashHD | operator=yes in my voicemail.conf does not seem to be working...any ideas? |
| 10:01.16 | Assid | man. i forgot what operator=yes does |
| 10:01.17 | CrashHD | when leaving a voicemail 0 will not do anything |
| 10:01.42 | CrashHD | it allows someone leaving a voicemail the option of 0 to reach an operator |
| 10:02.09 | Assid | have you programmed 0 in your dialplan before spawning off to voicemail? |
| 10:02.41 | CrashHD | if I spawn to voicemail from a macro will 0 need to be available in the macro? |
| 10:03.12 | Assid | dont think so.. i think its gotta be the dialplan before the macro is invoked.. but then i could be wrong here |
| 10:04.48 | CrashHD | I'll test it out |
| 10:05.42 | CrashHD | works if I have o,1, in the macro |
| 10:05.59 | CrashHD | or if o,1, is in the source context that launches the macro |
| 10:07.36 | Assid | yep |
| 10:07.42 | Assid | thats what u need |
| 10:08.04 | Assid | put it in the source context where the macro is invoked |
| 10:08.14 | CrashHD | either or will work |
| 10:08.17 | CrashHD | is what I was saying |
| 10:11.38 | *** join/#asterisk xxoxx (n=xxoxx@tor/regular/xxoxx) |
| 10:13.01 | *** join/#asterisk daysmen3 (n=primus@host86-138-235-88.range86-138.btcentralplus.com) |
| 10:16.02 | *** join/#asterisk _4d4m_ (n=adam@62.69.102.99) |
| 10:18.39 | cluetrain | Hello; I have a context that recieves an incoming Zap call (FXS port is on a legacy pbx), and then Dial()s a SIP softphone. Is there any way to create an extension for buttons pressed on the _softphone_ during this conversation? |
| 10:18.56 | *** part/#asterisk nettie (i=[U2FsdGV@85-18-54-38.ip.fastwebnet.it) |
| 10:19.19 | *** join/#asterisk nettie (i=[U2FsdGV@85-18-54-38.ip.fastwebnet.it) |
| 10:19.29 | nettie | hey guys anyone using trunk with JB enabled on chan_sip please? |
| 10:19.30 | Assid | cluetrain: i guess you could use Read() buti think you would need an extension to answer the call.. not a sip softphone |
| 10:21.52 | cluetrain | Assid: Hmm didn't quite understand that, - you're saying I shouldn't Dial(), but rather jump to a different context? I can't see how I'd read() something from the softphone without it answering a call first? |
| 10:27.22 | Assid | no |
| 10:27.29 | Assid | i said softphone wotn accept read |
| 10:27.40 | Assid | you need an extension which does.. and that means a virtual extension |
| 10:31.42 | cluetrain | Hm, ok I'm not too savvy with this yet, I'll read some docs and give it a go, thanks |
| 10:32.01 | parag_ast | anybody experienced with Quintum + asterisk |
| 10:51.45 | *** join/#asterisk razu_ (n=razu@tln-kontor.norby.ee) |
| 10:57.42 | *** join/#asterisk pa (n=Paolo@unaffiliated/pa) |
| 10:59.24 | *** join/#asterisk ghinckley6 (n=Spam@209.26.206.110) |
| 10:59.52 | ghinckley6 | need help compiling the zaptel drivers under mandriva 2006 |
| 11:02.56 | CrashHD | I need to be able to set a goto to a non-existent extension and have the dialplan continue if it does |
| 11:03.40 | CrashHD | or beable to do an if based on extension exists |
| 11:04.31 | CrashHD | it's going to go to i righ? |
| 11:05.09 | CrashHD | hmm it doesn't |
| 11:06.59 | ghinckley6 | is one here |
| 11:07.47 | ghinckley6 | any one know how to get mandriva to compile the zaptel1.2.7 |
| 11:08.12 | *** part/#asterisk champster (n=asterisk@AH.tescogroup.com) |
| 11:09.53 | *** join/#asterisk Whisk (n=a@195.60.27.193) |
| 11:12.51 | *** join/#asterisk razu_ (n=razu@tln-kontor.norby.ee) |
| 11:18.24 | *** join/#asterisk skeffling (n=andrew@andrew.1ec.aaisp.net.uk) |
| 11:19.31 | *** join/#asterisk ma_dzen (n=ma_dzen@217.66.17.141) |
| 11:19.41 | ma_dzen | Âñåì ñàëÿì |
| 11:19.52 | skeffling | a dum question, but how can I find the actual version number of zaptel that I'm running? |
| 11:20.37 | *** join/#asterisk lilwookie (n=jw@modemcable219.9-70-69.static.videotron.ca) |
| 11:23.18 | ma_dzen | how about supports fax (T.38) and CNG ? |
| 11:44.51 | *** join/#asterisk coppice (n=chatzill@127.166.17.210.dyn.pacific.net.hk) |
| 11:47.29 | ericsmythe | <PROTECTED> |
| 11:49.29 | *** join/#asterisk sndguru (n=irc@dsl-202-72-164-252.wa.westnet.com.au) |
| 11:49.48 | cluetrain | skeffling: dmesg|grep -i zap |
| 11:51.17 | sndguru | question for any pros out there - I'm interested in setting up an asterisk server |
| 11:51.28 | sndguru | I was hoping if if can handle |
| 11:51.36 | sndguru | dial-in server for internet connection |
| 11:51.37 | *** join/#asterisk [TK]D-Fender (n=joe@toronto-HSE-ppp4122655.sympatico.ca) |
| 11:51.55 | sndguru | instead of me paying a isp for a dial-up connection for my laptop |
| 11:52.06 | sndguru | being that I have a adsl connection at home |
| 11:52.10 | sndguru | any ideas |
| 11:52.11 | sndguru | ?? |
| 11:52.31 | E-bola | in denmark dialup connections are free |
| 11:52.34 | *** join/#asterisk ma_dzen (n=ma_dzen@217.66.17.141) |
| 11:52.38 | E-bola | i think u can get free subscriptions in most countries? |
| 11:52.46 | E-bola | u just pay for the minutes |
| 11:52.49 | sndguru | :P i'll just phone then from australia |
| 11:52.50 | sndguru | :P |
| 11:52.54 | rob0 | So there you have it, move to Denmark. |
| 11:53.03 | sndguru | :P |
| 11:53.13 | sndguru | but that can explain if it doesn't do it |
| 11:53.22 | E-bola | :) |
| 11:53.34 | E-bola | I just dont get what advantage ur own dialin server woudl give u? |
| 11:53.41 | rob0 | You want to plug a modem in a zaptel FXS port? |
| 11:53.58 | sndguru | it's it's not really required as the rest of the world is so ahead of my home country |
| 11:53.59 | sndguru | :P |
| 11:54.06 | sndguru | no surprise thou |
| 11:55.05 | *** join/#asterisk ariel_ (n=Ariel@70-46-87-158.ftl.fdn.com) |
| 11:55.26 | rob0 | sndguru: try searching the wiki. |
| 11:55.34 | rob0 | ~wikis |
| 11:55.39 | jbot | i guess wikis is http://www.voip-info.org |
| 11:56.32 | sndguru | there's a wiki |
| 11:56.34 | sndguru | where |
| 11:56.54 | sndguru | there's not much on asterisk.org |
| 11:57.14 | E-bola | what does asterisk have to do with a dialin server? |
| 11:57.29 | E-bola | just put a cheap pci modem in it and run pppd |
| 11:57.47 | sndguru | rob0: sorry missed your post - yes I'd like to dial up my HOME copper line as a dialup connection from my laptop |
| 11:58.03 | rob0 | Nothing directly. I guess he wants to use FXO ports for the incoming line[s], and plug a modem into one of those. |
| 11:58.39 | sndguru | :P linux and the asterisk server will answer my only copper line |
| 11:59.24 | sndguru | nice - never heard of a pppd sounds cool |
| 11:59.24 | *** join/#asterisk NoRemorse (n=bah@203-214-13-168.dyn.iinet.net.au) |
| 11:59.26 | NoRemorse | hi all |
| 11:59.30 | sndguru | might look into that |
| 11:59.48 | rob0 | If I was going to have a dedicated modem line, I'd bypass the FXO and just use the modem. |
| 11:59.50 | sndguru | yes please |
| 11:59.53 | NoRemorse | Is there anything I have to do to enable call xfer? I press # and the call doesnt park, just keeps going... |
| 12:00.06 | ma_dzen | In new version of ast CNG & VAD is supported? |
| 12:00.28 | sndguru | rob0: cool so you don't think the FXO would do it!? |
| 12:00.40 | sndguru | that give me an idea |
| 12:00.58 | E-bola | its pretty easy to setup a dialin server with a normal modem |
| 12:01.00 | rob0 | I didn't say that. I just said I would bypass it if possible, if a dedicated line. |
| 12:01.02 | sndguru | oh where is the wiki |
| 12:01.20 | nailbags | sndguru: i don't see why you'd want asterisk and an FXO to do that |
| 12:01.36 | NoRemorse | Is there anything I have to do to enable call xfer? I press # and the call doesnt park, just keeps going... |
| 12:01.42 | sndguru | oh!!!!!! the air is free for your help rob0 |
| 12:01.45 | jalsot | hi |
| 12:02.21 | jalsot | does anybody know any better [mean: faster] PCM wav->GSM6.10 wav conversion tool than sox? |
| 12:02.25 | *** join/#asterisk buzzdeee (n=buzzdee@ogo.rapideye.de) |
| 12:02.38 | sndguru | the reason I like the asterisk bit is because I have ONE line |
| 12:03.09 | sndguru | and i don't want every caller get a pci modem answering |
| 12:03.09 | nailbags | sndguru: are you saying you want to use it for incoming/outgoing zap calls as well? |
| 12:03.29 | sndguru | zap??? sorry what |
| 12:03.39 | nailbags | sndguru: sorry, i mean fxo |
| 12:04.03 | sndguru | :P I've like to plug my copper fxo into a sipura 3000 |
| 12:04.05 | *** join/#asterisk Khaloudiii (n=k@212.98.141.200) |
| 12:04.10 | sndguru | then asterisk it up |
| 12:04.14 | Khaloudiii | hi |
| 12:04.27 | sndguru | and plug my other phones into the fx? |
| 12:04.30 | sndguru | what is it |
| 12:04.33 | sndguru | ummm |
| 12:04.36 | sndguru | fxs |
| 12:04.41 | sndguru | anyway |
| 12:04.50 | Khaloudiii | any one know how to let the cdr at asterisk at home save the ip address of the user |
| 12:04.54 | nailbags | sndguru: i think the way to do it would be to plug the line into a pci modem as well, then use asterisk to run an external command to tell the other modem when to pick up |
| 12:05.15 | sndguru | and have dial-in internet, voice, and answer machine from it |
| 12:05.33 | sndguru | nice that can work |
| 12:05.55 | sndguru | umm, i wonder if the laptop modem would hold off for that long |
| 12:06.11 | sndguru | i guess if wouldn't be that long really |
| 12:06.15 | Khaloudiii | ? |
| 12:06.17 | nailbags | sndguru: huh? |
| 12:06.37 | sndguru | the handshake of the two modems |
| 12:06.51 | ManxPower | Khaloudiii, "show applications" in the Asterisk CLI. Pay attention to the CDR stuff. Also see /path/to/src/asterisk/docs/README.variables |
| 12:06.56 | nailbags | sndguru: ah, yes ... didn't think about the client modem ... |
| 12:07.24 | nailbags | sndguru: there might be some way to do it with a special modem string on the client |
| 12:07.25 | Khaloudiii | ManxPoweri read it even i modified cdr.c |
| 12:07.28 | NoRemorse | Is there anything I have to do to enable call xfer? I press # and the call doesnt park, just keeps going... |
| 12:07.32 | Khaloudiii | but there is trick |
| 12:07.36 | sndguru | i mean I'd only be saving like $5-10 a month in ISP costs but it would help |
| 12:07.40 | ManxPower | Khaloudiii, Why did you modify cdr.c?????? |
| 12:07.47 | *** join/#asterisk Chris-NB (n=chris@ng1.kurtkrenn.com) |
| 12:07.50 | ManxPower | There is an app for adding custom CDR fields. |
| 12:08.05 | nailbags | sndguru: oh and i feel your pain about tel$tra |
| 12:08.20 | sndguru | :P hay a fellow australian ?? |
| 12:08.37 | ManxPower | NoRemorse, You are trying to use "# transfers", not the transfer feature built into the device. |
| 12:08.52 | nailbags | yeah i'm in sydney. have an asterisk pabx set up for an office |
| 12:09.00 | sndguru | nice |
| 12:09.11 | sndguru | i watch systm and got hooked |
| 12:09.26 | sndguru | just trying to find a place to buy a sipura 30000 |
| 12:09.34 | nailbags | sndguru: just doing it at home? u in perth? |
| 12:09.48 | sndguru | and or unlock my engin linksys wireless router |
| 12:09.59 | sndguru | but that hasn't got fxo on it |
| 12:10.06 | sndguru | yep perth |
| 12:10.26 | nailbags | nice place, want to get back there soon |
| 12:10.34 | nailbags | i got a digium card from techtopia |
| 12:10.38 | sndguru | :P i always hear people say that |
| 12:10.53 | sndguru | tis why i'm not interested in leaving :P |
| 12:11.05 | sndguru | pci card |
| 12:11.20 | nailbags | yeah |
| 12:11.29 | nailbags | they have a lot of voip gear iirc |
| 12:11.38 | sndguru | i'm running out of pci slot in my server and the cat5 cable in my roof is right next to the phone lines |
| 12:12.02 | NoRemorse | Is there anything I have to do to enable call xfer? I press # and the call doesnt park, just keeps going... |
| 12:12.14 | sndguru | :P i've already got a rj12 on the master line in |
| 12:12.33 | sndguru | it's will be click click when i get the gear |
| 12:12.51 | ManxPower | NoRemorse, did you put the correct options on the dial lone? |
| 12:12.55 | ManxPower | ..er.. line? |
| 12:13.20 | sndguru | nailbags: techtopia is that australia based |
| 12:13.36 | nailbags | sndguru: i just did dial-in prompts, internal directory and voicemail on the weekend for this company. i love asterisk even more. |
| 12:13.45 | nailbags | nailbags: yeah |
| 12:13.50 | nailbags | oops |
| 12:13.52 | nailbags | lol |
| 12:13.53 | sndguru | :P |
| 12:14.07 | nailbags | yeah techtopia is aussie |
| 12:14.18 | sndguru | systm made it look so easy |
| 12:14.19 | ManxPower | Ah, I see NoRemorse doesn't actually want help, he just wants to annoy us. |
| 12:14.22 | nailbags | i think they were the only ones that sold digium at the time |
| 12:14.36 | NoRemorse | lol sorry, missed that |
| 12:14.37 | sndguru | coolies techtopia.com.au ????? |
| 12:14.42 | nailbags | yeah |
| 12:14.54 | NoRemorse | now tell me what a dial lone is! |
| 12:15.17 | sndguru | oh and the wiki for asterisk what's that |
| 12:15.21 | *** join/#asterisk asteriskmonkey (n=phil@h216-235-8-130.host.egate.net) |
| 12:15.31 | ManxPower | NoRemorse, Dial LINE. |
| 12:16.01 | NoRemorse | ah ok sorry, will check options now cos answer is no I'd say... |
| 12:16.02 | ManxPower | i.e. in extensions.conf: Dial(Zap/1/${EXTEN},25,abcd) |
| 12:16.19 | ManxPower | NoRemorse, what documentation have you read? |
| 12:16.37 | NoRemorse | Dial cmd, wasnt thinking straight tho forgot about option. |
| 12:16.39 | ManxPower | try "show application dial" in the Asterisk Command Line Interface Terminal. |
| 12:17.05 | sndguru | nailbags: $139 for the siprua 3000 that's not bad |
| 12:17.06 | NoRemorse | ohh look at that, t missing, thanks manx |
| 12:17.16 | *** join/#asterisk bjohnson (n=bjohnson@jecinc.tor.istop.com) |
| 12:18.25 | asteriskmonkey | typos are the #1 learning curve killer :D |
| 12:21.12 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 12:23.42 | skeffling | thanks cluetrain (sorry for delay in replying!) |
| 12:23.58 | *** join/#asterisk kinker12 (n=kinker12@www.yellox.de) |
| 12:24.04 | kinker12 | hello |
| 12:26.07 | kinker12 | 2 questions i got: 1. smsq - how can i use it with 1.1.10 ? |
| 12:26.18 | *** join/#asterisk beyond (n=beyond@200.192.160.100) |
| 12:26.24 | kinker12 | 2. eventhandler for SIP-registration |
| 12:30.45 | *** join/#asterisk s0lid (n=jlq@124.107.26.37) |
| 12:37.39 | *** join/#asterisk Assid (i=assid@203.115.83.215) |
| 12:38.12 | Assid | anyone know how people make a rate deck ? |
| 12:38.28 | *** part/#asterisk sndguru (n=irc@dsl-202-72-164-252.wa.westnet.com.au) |
| 12:43.22 | *** join/#asterisk puzzled (n=patrick@puzzled.xs4all.nl) |
| 12:43.28 | Assid | btw: anyone know a decent UK provider? |
| 12:45.22 | *** join/#asterisk gandhijee (n=gandhije@mail.win-ent.com) |
| 12:48.00 | *** join/#asterisk Vorondil (n=jkiffmey@miro.voltaiccommerce.com) |
| 12:58.44 | ManxPower | ~ecfo |
| 12:58.51 | jbot | Echo Canceler Freak Out, this happens when the rxgain is too high and the echo canceler freaks out. Some users describe it as "screeching", "feedback", "static", or other useless terms. If users report "static" on a system where there cannot be static (all digital, PRI, SIP, etc), you might be experiencing ECFO. |
| 12:59.35 | coppice | definitely not G.168 compliant :-) |
| 12:59.44 | *** join/#asterisk Vorondil (n=jkiffmey@miro.voltaiccommerce.com) |
| 13:00.45 | *** join/#asterisk my007ms (n=noor@217.139.224.195) |
| 13:01.09 | coppice | some people really love setting rx gain very high, until everything overloads badly |
| 13:01.21 | RoyK | rotfl. telenor, the (former?) telemonopolist in .no have some 70k customers on VoIP. their VoIP rig failed around midnight during an upgrade and it's still down :) |
| 13:01.34 | [TK]D-Fender | Setting gains for EC... how "quaint" ;) |
| 13:01.35 | *** join/#asterisk anthm (n=anthmct@CPE-69-76-83-52.wi.res.rr.com) |
| 13:01.35 | *** mode/#asterisk [+o anthm] by ChanServ |
| 13:02.27 | *** join/#asterisk benjk (n=benjamin@f8a01-0357.din.or.jp) |
| 13:03.23 | ManxPower | [TK]D-Fender, Yeah. I prefer leaving the gains alone and installing an external Tellabs echo canceler |
| 13:05.38 | *** join/#asterisk tamp4x (n=tampon@63.138.1.178) |
| 13:05.52 | tamp4x | anyone know how to force snoms to release their dhcp? |
| 13:05.59 | *** join/#asterisk MacWeenie (n=chatzill@82-35-73-28.cable.ubr02.dals.blueyonder.co.uk) |
| 13:06.17 | nailbags | ManxPower: can't u just redirect port 80 to the proxy? |
| 13:06.52 | ManxPower | nailbags, that breaks proxy authentication |
| 13:07.10 | ManxPower | well, breaks web site authentication when proxy authentication is also required. |
| 13:07.14 | *** join/#asterisk gade_pt (n=IceChat7@195-23-46-90.net.novis.pt) |
| 13:07.31 | nailbags | ManxPower: i see |
| 13:08.24 | gade_pt | Hi everyone |
| 13:11.19 | gade_pt | i'am trying to configure a Dialogic D4PCIUF on a TrixBox (formely Asterisk@home). Anyone had the same kind of experience? I had a lot of problems on how to understand the way i can connect Asterisk with the Intel/Dialogic hardware. |
| 13:11.56 | gade_pt | if someone could help ... It will be really nice |
| 13:13.39 | benjk | you need commercial software from Digium for Dialogic support |
| 13:14.13 | gade_pt | damn... My boss will not like this answer :| |
| 13:14.28 | gade_pt | so well... |
| 13:14.32 | gade_pt | if i had to.... |
| 13:14.41 | benjk | you can always use Bayonne |
| 13:14.56 | gade_pt | i've read about bayonne |
| 13:15.12 | benjk | let me just ask David Sugar about that card ... |
| 13:15.30 | gade_pt | but bayonne is not like Asterisk isn't it ? |
| 13:16.21 | *** join/#asterisk ghenry (n=ghenry@suretecsystems.plus.com) |
| 13:17.01 | benjk | it is like Asterisk in a some ways, it is unlike Asterisk in other ways |
| 13:18.21 | gade_pt | As I understand, Bayonne is for making a telephony center for Linux, Asterisk is more to dedicated to VoIP |
| 13:18.52 | benjk | You can do pretty much all the things you can do with Asterisk |
| 13:19.02 | anthm | come to cluecon next wednesday and see a presentation on it by the author himself =D |
| 13:19.03 | benjk | the concepts are different |
| 13:19.18 | coppice | asterisk isn't really a VoIP platform at all |
| 13:19.24 | gade_pt | At this moment i've already VoIP working perfectly with softphones. Now, the objective is to make a phone call from a softphone to the portuguese network trought the Dialogic interface |
| 13:19.37 | *** join/#asterisk nortex (n=nortex@64.136.65.144) |
| 13:19.39 | benjk | David Sugar says "I am not immediately familiar with that specific card model, but if it is supported by the runtime and has voice resources, it should work" |
| 13:20.02 | gade_pt | Thanks Benjk :) |
| 13:20.31 | gade_pt | By the way, what he means with "runtime" ? |
| 13:20.39 | gade_pt | is the Linux OS? |
| 13:20.48 | benjk | probably Dialogic's framework |
| 13:21.36 | gade_pt | it's a really good news :) |
| 13:21.42 | anthm | btw you have it reversesd, bayonne is multi-platform, multi-arch Asterisk is way more Linux-centric than bayonne by a longshot |
| 13:21.44 | coppice | what is a D4PCIUF? A D4PCI should be supported by the old Dialogic drivers, which were free. If its a newer card it will require paid for drivers |
| 13:22.09 | *** join/#asterisk |dennis| (i=dennis@200.32.215.84) |
| 13:22.25 | *** join/#asterisk nortex (n=nortex@64.136.65.144) |
| 13:22.56 | gade_pt | coppice: i use the Evaluation Driver for the D4PCIUF (only works for 10 hours. after that I have to restart the service) |
| 13:22.58 | benjk | David is just heading out for lunch but will be back later, I gave him your IRC nick, so if you want to hang out for another hour or so, you can talk to him directly |
| 13:23.14 | gade_pt | Thank benjk. |
| 13:23.21 | gade_pt | I'm also going to launch |
| 13:23.23 | benjk | he'll contact you by PM |
| 13:23.39 | coppice | last time i looked the paid for drivers were damned expensive |
| 13:23.40 | gade_pt | Ok |
| 13:24.03 | gade_pt | I know coppice... Is for that reason i've installed the evalutation drivers |
| 13:24.22 | gade_pt | well.... I have to launch. It's 14:24 here... |
| 13:24.23 | coppice | so you plan on restarting every 10 hours? :-) |
| 13:24.24 | gade_pt | :P |
| 13:24.37 | gade_pt | Do you have another solution ? |
| 13:24.44 | gade_pt | there is another kind of drivers ? |
| 13:25.03 | benjk | talk to David Sugar after lunch |
| 13:25.17 | gade_pt | As far I read, the only the Dialogic 6.1 drivers for Linux |
| 13:25.18 | RoyK | charging money for drivers is sick |
| 13:25.32 | gade_pt | benjk: what is the David sugar Nickname ? |
| 13:25.33 | coppice | they seem to be open sourcing something, but its not the whole driver package. I dunno what you really have to do now. charging for the drivers really shot Intel in the foot |
| 13:25.35 | *** join/#asterisk mhnoyes (n=mhnoyes@dialup-4.246.6.57.Dial1.SanJose1.Level3.net) |
| 13:25.55 | coppice | the 5.1 drivers were free |
| 13:25.59 | benjk | dyfet |
| 13:26.55 | gade_pt | All right benjk. After the launch i'll talk to him. If he will be available to talk with me :P |
| 13:27.18 | coppice | dialogic is for masochistic |
| 13:27.28 | gade_pt | lol |
| 13:27.41 | benjk | :D |
| 13:27.47 | gade_pt | coppice, as i can see... ! It's really hard to work with :| |
| 13:28.18 | gade_pt | But, i haven't the choice |
| 13:28.26 | coppice | 10 years ago dialogic were pretty reasonable. after that they went berserk pissing off their users |
| 13:28.57 | kinker12 | how can i use smsq with 1.1.10 ? |
| 13:29.58 | gade_pt | well... i have to go. See you later guys. Thank you for the help :) |
| 13:30.03 | gade_pt | .: gone |
| 13:30.21 | *** join/#asterisk Dr-Linux|work (n=Linux@202.125.139.198) |
| 13:36.15 | waba | anyone here configured succesfully a annatel.net account on * ? |
| 13:36.26 | *** join/#asterisk QbY (n=Kelvin@cm-64-221-170-205.dhcp.southerncoastalcable.net) |
| 13:41.18 | *** join/#asterisk niteowldave (n=dave@CPE-203-51-61-57.nsw.bigpond.net.au) |
| 13:43.08 | *** join/#asterisk daysmen3 (n=primus@host86-138-235-88.range86-138.btcentralplus.com) |
| 13:45.13 | Assid | im just curious.. how do people do call routing.. like if i say 18004444444 -- how does it know its a US based call and not a country with 18 as the country code? |
| 13:45.35 | creativx | dial prefixing |
| 13:46.02 | creativx | eg dial 00 intl, 0 local dial |
| 13:46.15 | *** join/#asterisk Dovid (n=dovi5988@pool-71-250-66-243.nwrknj.east.verizon.net) |
| 13:46.23 | Assid | no no.. okay.. suppose i say .. something like australia |
| 13:46.50 | Assid | we call 01161222222222 how does it know its australia and not a country with 6 or 612 |
| 13:47.15 | creativx | you could force your users to dial an extension to get a local line, and another for an international line |
| 13:47.18 | Dovid | anyone know where i can a simple agi script that connects to PHP ? i need to have the user call an exten and set 1 and place a value in the db or press 2 and put a value in the db ??? |
| 13:47.36 | Assid | err.. no dude.. thats not my question |
| 13:48.14 | Assid | how do telecom companies know this is australia based call and not say the likes of newzealand or whoever else has a country code of 6 |
| 13:48.43 | creativx | ah |
| 13:49.23 | Dovid | if the 6 is for multiple countries then it must be the next digit |
| 13:49.29 | creativx | how do you dial +47 22 22 30 30 ? |
| 13:49.42 | Dovid | from the US ? |
| 13:49.49 | creativx | yeah or au |
| 13:50.02 | ManxPower | Dovid, 011-47-22-22-30-30 |
| 13:50.04 | Dovid | from the US u do 011 47 22 22 30 30 |
| 13:50.15 | Dovid | i know some else asked, i was responding |
| 13:50.24 | creativx | yeah and from australia Assid? |
| 13:50.31 | ManxPower | Assid, each country has it's own country code. |
| 13:50.35 | creativx | in .no i would do 00 <countrycode> <number> |
| 13:50.36 | Assid | right for UK for example.. they have 44 as 1 rate. but 447 or something as another rate |
| 13:50.55 | Dovid | ManxPower: do u know anything about snom QoS ? |
| 13:50.56 | creativx | you would never be able to mix local numbers and international country codes |
| 13:51.31 | ManxPower | Dovid, No. I have come to believe that my body somehow generates a field that breaks any QoS I try to put into place on any network I manage. |
| 13:51.46 | Nugget | heh |
| 13:51.49 | Dovid | hrehe |
| 13:52.15 | Dovid | ManxPower: I have a client with snom phones that as soon as he starts a big download his phone call becomes unclear |
| 13:52.26 | *** join/#asterisk Ebola (n=Ebola@81-86-155-65.dsl.pipex.com) |
| 13:52.47 | ManxPower | Dovid, sounds like standard issue with lack of QoS on consumer DSL/cablemodem service. |
| 13:52.54 | *** join/#asterisk tsurk0 (n=tsurko@85.187.160.157) |
| 13:53.00 | Dovid | meaning ? |
| 13:53.15 | ManxPower | Dovid, meaning he is screwed. |
| 13:53.30 | Assid | creativx: then how do they route the calls.. thats the question.. how do they know the call is bound to UK or wherever and not anywhere else with a similar country code |
| 13:53.37 | ManxPower | What internet service does he have, what IP phone does he have, what router does he have, what "modem" does he have? |
| 13:53.44 | Dovid | the phone plugs in to the router (only device plugged into the router) and then the computer runs thru the phone. is there anything tha I can set on the phone that gives Prioroty to the phone ? |
| 13:54.04 | ManxPower | Assid, each carrier has a list of all that stuff. |
| 13:54.25 | ManxPower | Dovid, I don't know anything about SNOM. |
| 13:54.31 | Dovid | okies. thanks |
| 13:54.53 | ManxPower | Dovid, chances are he will need to do the QoS on the router |
| 13:54.54 | Dovid | is there a router that I can get that will give prioroty to aspecific port on the router ? |
| 13:55.03 | Dovid | he has a POS router |
| 13:55.12 | ManxPower | Dovid, yes. some Linksys routers can do that. |
| 13:55.57 | Dovid | does the VOIP provider have to haev QOS as well or the router is smart enough ? |
| 13:57.33 | Dovid | brb. Need a smoke |
| 13:57.52 | ManxPower | If you want real QoS you need every single router between the phone and the VoIP server to have QoS. |
| 13:57.59 | *** join/#asterisk trbldwine (i=troubled@adam.ur.northwestern.edu) |
| 13:58.15 | ManxPower | Most people see significant improvement by doing QoS on their home/office router. |
| 13:58.24 | *** join/#asterisk parag_ast (n=root@dxb-b16506.alshamil.net.ae) |
| 13:59.09 | *** join/#asterisk nortex (n=breeves@snapper.titanspecialties.com) |
| 13:59.17 | parag_ast | hi...i m getting this error with polycom IP501 "not enough room for applications" while downloading the tftp sip.ld image.... |
| 13:59.56 | Assid | ManxPower: what if you have the gateway/dsl line dedicated only for asterisk? |
| 14:00.08 | mutilator | i need more cowbell! |
| 14:00.58 | *** join/#asterisk MatsK (n=mats@141.221.181.62.in-addr.dgcsystems.net) |
| 14:00.59 | *** join/#asterisk pif (n=ldm@zenon.apartia.fr) |
| 14:01.06 | *** part/#asterisk kmilitzer (n=km@office-gw.westend.com) |
| 14:02.05 | Dovid | ManxPower: I was using a vonage box that QoS in use and that did nothing (it had no vonage service on it) |
| 14:03.04 | *** join/#asterisk asteriskmonkey (n=phil@h216-235-8-130.host.egate.net) |
| 14:05.34 | *** join/#asterisk aldente (n=ed@pool-72-67-217-107.lsanca.fios.verizon.net) |
| 14:05.52 | nortex | Is there an online site that makes it easy to get a list of area codes local to a phone number? I'm trying to build a list of local exchanges for a Dundi routing setup. |
| 14:06.21 | buzzdeee | hi, I have a problem with my PRI interface and DID here in germany, overlapdial=yes and immediate=no in zapata.conf, but asterisk sometimes doesnt wait for the last one or two digits, any idea? |
| 14:10.05 | *** part/#asterisk wwalker (n=wwalker@pdpc/supporter/sustaining/wwalker) |
| 14:11.54 | *** join/#asterisk eKo1 (n=eKo1@190.4.7.90) |
| 14:15.35 | ManxPower | buzzdeee, you didn't get any responses when you asked on the mailing lists? |
| 14:16.24 | *** join/#asterisk Syrus_ (n=pascal@tahiti.mpl.rullier.net) |
| 14:16.39 | trelane_ | cdr_odbc still loses CDR's when the database connection goes down doesn't it? |
| 14:18.14 | eKo1 | trelane_: of course, but that is with all database driven cdr_* modules. |
| 14:18.41 | clyrrad | Does anyone have a good link that shows how to convert/prepare mp3s to be played over asterisk with out all the CLI warnings of 'Junk at the begining of frame xxxxxx'? |
| 14:18.50 | Assid | okay what would eat up more resources? a CURL branch off.. or an AGI branch off |
| 14:19.02 | MatsK | I thought I saw a notice that cdr_postgre was caching CDR's when the connection failed |
| 14:19.04 | tamp4x | does auto answer have to be turned on to using paging? |
| 14:21.52 | rpm | i think im on my way to crazy or i may just be there. |
| 14:22.10 | *** join/#asterisk rogier (n=rogier@83-67-dsl.ipact.nl) |
| 14:22.49 | tamp4x | i upgraded asterisk, now i get this....2006-07-26 10:21:00 WARNING[17043]: codec_g729.c:249 lintog729_framein: Out of G.729 Encoder Licenses! |
| 14:23.09 | *** part/#asterisk file (n=file@216.237.114.82) |
| 14:23.12 | *** join/#asterisk file (n=file@216.237.114.82) |
| 14:23.12 | *** mode/#asterisk [+o file] by ChanServ |
| 14:23.14 | tamp4x | anyreason why? |
| 14:23.34 | rpm | tamp4x: tamp4x, because you have no licenses.. you need a license per channel. |
| 14:24.05 | clyrrad | does anyone know how to convert mp3 for playback over asterisk? |
| 14:24.05 | tamp4x | i did buy the lcienses |
| 14:24.10 | tamp4x | licenses |
| 14:24.13 | tamp4x | ahh shit |
| 14:24.21 | tamp4x | i think when i use paging i run out of lcienses |
| 14:24.57 | pif | anyone has recent cisco 79xx firmware? |
| 14:25.02 | rpm | i used to use g729 but found it sucked and would rather use the bit of extra bandwidth for gsm. |
| 14:25.13 | *** join/#asterisk malverian (n=malveria@gentoo/developer/malverian) |
| 14:25.28 | *** join/#asterisk BryanFE2 (n=chatzill@ip72-194-209-224.sb.sd.cox.net) |
| 14:27.21 | *** join/#asterisk wunderkin (n=kev@ip68-226-113-228.ph.ph.cox.net) |
| 14:27.34 | malverian | Corydon-w, Did you ever mention anything with regards to the res_agi change that I spoke of a while back? |
| 14:28.08 | buzzdeee | Hi ManxPower, yes, I got answers, but they did not helped |
| 14:28.09 | malverian | The one that pertained to record_file returning -1 on hangup and killing even DeadAGI scripts. |
| 14:28.19 | trelane_ | eKo1, why hasn't that been fixed? |
| 14:28.22 | rpm | php+agi is going to kill me.. using the get_variable function is so flakey. |
| 14:28.48 | buzzdeee | they said, play with pridialplan and switchtype |
| 14:29.24 | sevard | Is there anyway to setup what default shell asterisk will run when you just hit ! at the CLI? |
| 14:29.45 | rpm | buzzdeee: i didn't see in my scrollback what your issue was. whats wrong with the pri circuit? |
| 14:29.56 | buzzdeee | hi, I have a problem with my PRI interface and DID here in germany, overlapdial=yes and immediate=no in zapata.conf, but asterisk sometimes doesnt wait for the last one or two digits, any idea? |
| 14:30.13 | nortex | buzzdeee, Does it allways drop the same number of digits? |
| 14:30.25 | buzzdeee | no, sometimes one, sometimes two |
| 14:31.02 | rpm | buzzdeee: how many dtmf digits does the dce side look for? |
| 14:31.22 | benjk | you can change the delay in libpri, I think it's set to 1000 ms |
| 14:31.41 | buzzdeee | three, the first is always coming |
| 14:31.52 | benjk | I ran across the code while looking for something else during the last couple of days |
| 14:32.17 | buzzdeee | oh, is there some #define statement somewhere? |
| 14:32.23 | buzzdeee | that sounds interesting |
| 14:32.26 | eKo1 | trelane_: because it can't? |
| 14:32.38 | trelane_ | eKo1, I don't buy that |
| 14:32.39 | benjk | q931.c in the libpri directory, then search for overlapdial |
| 14:32.48 | eKo1 | trelane_: well then you fix it. |
| 14:33.04 | BryanFE2 | question: Is there an app which will wait for a fixed frequency hz tone (specified as a parameter), then exit? |
| 14:33.05 | benjk | might even be able to influence this within chan_zap.c |
| 14:34.17 | *** join/#asterisk pdtmobile (n=ptinsley@c-68-53-40-50.hsd1.tn.comcast.net) |
| 14:36.18 | buzzdeee | benjk, there is the FUNC_SEND, but I do not see any timeout to set |
| 14:36.27 | coppice | BryanFE2: that usually turns out to be a more complex problem than it sounds |
| 14:37.29 | BryanFE2 | coppice: that's a bummer. Any suggestions? I need to make a dialplan which waits for a tone, then plays an audio file. It's important to wait for the tone, I can't just use a delay. |
| 14:38.54 | coppice | simply looking for a tone is easy. distinguishing that it is a pure tone within a set of tolerances makes it a more customised problem |
| 14:39.21 | BryanFE2 | is there an app which does the simple version? |
| 14:39.40 | *** join/#asterisk acrg (n=aragon@decoder.geek.sh) |
| 14:39.40 | acrg | hi |
| 14:39.50 | acrg | is anyone using an AVM C2 or C4 with asterisk ? |
| 14:39.51 | benjk | buzzdee, I can't tell you where it is, it's just that I have been going over libpri and chan_zap a lot during the last two days to do something unrelated to overlapdial, but I saw the overlapdial code there pop up again and again and there was some delay with a comment or something like that |
| 14:40.18 | benjk | arcg, is it HFC based? |
| 14:40.47 | acrg | benjk not afaik - it's an active capi card |
| 14:40.52 | coppice | BryanFE2: i don't think so, but you'll find it pretty useless anyway |
| 14:40.58 | benjk | that means nothing |
| 14:41.25 | benjk | capi is a software API, has nothing to do with the chipset on the card |
| 14:41.33 | acrg | I know |
| 14:41.35 | *** join/#asterisk jbalcomb (n=jbalcomb@216.28.180.158) |
| 14:41.42 | *** join/#asterisk smackus (n=ckwall@63.149.122.93) |
| 14:41.49 | acrg | but HFC based cards are all passive ? |
| 14:41.55 | buzzdeee | benjk, thanks, then I need to dig a bit deeper |
| 14:41.59 | Dovid | what does this mean ? |
| 14:42.00 | benjk | no |
| 14:42.03 | *** join/#asterisk cytrak (n=kvirc@adelphi.geofocus.com) |
| 14:42.08 | Dovid | Incoming call: Got SIP response 500 "Internal Server Error" back from 10.10.10.101 |
| 14:42.12 | benjk | there are both active and passive cards with HFC chips |
| 14:42.26 | buzzdeee | benjk, at least on a first glance it doesn't seem to be in the file you mentioned |
| 14:42.46 | benjk | check out chan_zap.c in asterisk/channels |
| 14:42.57 | buzzdeee | yeah, will do that |
| 14:42.57 | acrg | benjk http://www.avm.de/en/index.php3?Produkte/Server-Produkte/C4/index.js.html |
| 14:43.09 | benjk | both q931.c in libpri and chan_zap.c in channels, I don't know which one I saw the stuff in |
| 14:44.30 | benjk | that chip doesn't look like an HFC chip, doesn't seem to have the Cologne logo on it |
| 14:44.43 | benjk | anyway, you should be able to use the card with chan_capi |
| 14:44.48 | acrg | yea |
| 14:44.57 | benjk | BRIstuff with ZapHFC would have been easier though |
| 14:45.11 | acrg | ok kewl |
| 14:47.10 | macTijn | AVM, isn't that the diva-series ? |
| 14:47.11 | Dovid | is digium still using SVN ? |
| 14:47.15 | Assid | hey anyone know if a rate card shows 30/6 0.012 does that mean its .012 for 30 seconds or per minute? |
| 14:48.09 | file | Dovid: yes... |
| 14:48.37 | Dovid | file: is svn the newer system or the older ? i know they switched from one system to a diff one. |
| 14:48.42 | E-bola | Anybody now any cool tools for asterisk on win32? |
| 14:48.48 | E-bola | snap crashes cuz im international |
| 14:48.48 | file | Dovid: newer |
| 14:48.54 | E-bola | and calling circles seems really buggy |
| 14:48.56 | Dovid | thanks |
| 14:49.01 | Dovid | what was the old one ? |
| 14:49.04 | *** join/#asterisk FaithX (n=FaithX@ns.linuxterminal.com) |
| 14:49.04 | E-bola | i tried asterisk desktop manager that also only work 50% or so |
| 14:49.09 | Dovid | oh |
| 14:49.10 | Dovid | nnm |
| 14:49.12 | Dovid | cvs |
| 14:49.48 | smackus | is anyone here using astguiclient |
| 14:49.59 | Assid | anyone know any decent providers for the likes of UK ? |
| 14:50.05 | Assid | and australia |
| 14:53.21 | buzzdeee | benjk, I think I shall add some time to the matchdigittimeout and gendigittimeout defines in chan_zap.c |
| 14:53.23 | eKo1 | E-bola: what do you expect? It's Winblows |
| 14:54.06 | Ebola | heh |
| 14:54.08 | Ebola | oh right |
| 14:54.36 | benjk | matchdigittimeout alone would seem to be sufficient |
| 14:55.28 | *** join/#asterisk hmmhesays (n=ohyeah@66.173.103.110) |
| 14:55.29 | buzzdeee | benjk, thanks for the hint, I'll let you know whether it worked, or not |
| 14:55.31 | hmmhesays | wee haw |
| 14:55.46 | hmmhesays | well the switch to a linux desktop is going smoothly |
| 14:55.52 | tzanger | bah humbug |
| 14:55.55 | hmmhesays | except I can't figure out how to edit my applications menu in gnome |
| 14:56.14 | benjk | good luck |
| 14:56.17 | buzzdeee | but generally, is there something I can define in one of the config files to raise the timeout for overlap dials? |
| 14:56.29 | benjk | maybe not |
| 14:56.36 | benjk | you could add that ;) |
| 14:56.44 | eKo1 | hmmhesays: i just leave it alone |
| 14:57.01 | hmmhesays | i don't want to leave it alone, I want xmms in my sounds menu |
| 14:57.20 | nortex | Ebola, What version of snap did you try? |
| 14:57.42 | tzanger | trying to get rogers cable for a residential address but I want business service (no cap) |
| 14:57.44 | benjk | I think astguiclient is abandonware |
| 14:57.45 | buzzdeee | ah, well, I can read c source, and change some values, but for me to add sth. like this, in a way that it will get into the source tree, i do not think so (: |
| 14:58.21 | benjk | why would it need to get into the source tree? |
| 14:58.45 | eKo1 | hmmhesays: what distro? |
| 14:58.50 | benjk | if you feel like it, you can make a patch and put it up for download |
| 14:58.50 | tzanger | adsl can't get me better than 2m/512k where I am :-( |
| 14:59.10 | buzzdeee | just for downloading the tree, and not need to patch it every time when I upgrade |
| 14:59.33 | eKo1 | tzanger: why the sad face? |
| 14:59.41 | tzanger | eKo1: because I prefer DSL to cable |
| 14:59.46 | *** join/#asterisk Sonderblade (n=mah@static-213.131.147.169.addr.tdcsong.se) |
| 14:59.52 | benjk | if you are running a production box you shouldn't upgrade that often |
| 14:59.54 | eKo1 | so, get dsl |
| 14:59.55 | *** join/#asterisk xheliox (n=jeff@pdpc/supporter/active/xheliox) |
| 14:59.58 | Ebola | nortex: Think you mean E-bola |
| 15:00.07 | tzanger | eKo1: as I said, I cannot get better than 2m/512k where I am |
| 15:00.12 | benjk | if you are running a development, testing box, it shouldn't matter |
| 15:00.25 | nortex | Ebola, sorry |
| 15:00.27 | eKo1 | what more do you want? |
| 15:00.30 | Ebola | No problem |
| 15:00.32 | tzanger | eKo1: 5m/800k |
| 15:00.34 | [TK]D-Fender | tzanger : Thats livable by my standards... I'd rather have slower and unburdened (cable overload + contracts), and be able to use my S518 and get my connection with a ton of cheap frills... |
| 15:00.35 | nortex | E-bola, What version of snap did you try? |
| 15:00.54 | Ebola | Haven't been in a channel with someone using such a similar nick before :p |
| 15:01.02 | tzanger | [TK]D-Fender: yeah I want the S518 as well... |
| 15:01.10 | tzanger | [TK]D-Fender: I <3 my 518 |
| 15:01.26 | [TK]D-Fender | tzanger : 5000/800 is what I'm getting :) Gravy... don't need it all... I'd be just fine with yours or less. Technically I'd prefer T1 grade speeds to yours. |
| 15:01.48 | buzzdeee | benjk, I know, but just in case, I do not like to add my own patches, and maybe others will need that as well, they too will not like to patch, I assume |
| 15:01.49 | tzanger | yeah I'd rather have 1.5 in both directions |
| 15:01.54 | [TK]D-Fender | Cable co's are SO full of shit its not funny. |
| 15:02.09 | tzanger | yeah I signed up if I can't get business service I'll stick with the 2meg |
| 15:02.20 | eKo1 | is that dedicated? |
| 15:02.20 | Ebola | hmmhesays: http://www.ubuntuforums.org/showthread.php?t=82537 Might be useful |
| 15:02.24 | benjk | well, good luck getting past the bugtracker gestapo then |
| 15:02.38 | benjk | I think its a waste of time |
| 15:02.51 | E-bola | nortex: newest binary on their page |
| 15:02.58 | E-bola | it crashes before it ever gets going |
| 15:03.00 | benjk | but that's just me |
| 15:03.53 | buzzdeee | benjk, at first I'll raise the values in the define and recompile, if it helps, then I can think again wheter I will add a config option or not (: |
| 15:04.12 | E-bola | is sourceforge down? |
| 15:04.13 | benjk | you can always submit your patches to Junghanns |
| 15:07.14 | jbalcomb | [TK]D-Fender: Search for Parannoid Penguin on Linux Journal. They're articles from a guy named Mick Bauer. I think you would appreciate them. |
| 15:08.24 | E-bola | lol |
| 15:08.28 | E-bola | snap seems to crash ona font issue |
| 15:09.59 | Ebola | hmmhesays: on the other hand, I have a menu editor under system tools but that may be because I did a complete install of dropline this morning |
| 15:11.39 | *** join/#asterisk Mw3 (i=mw3@national.t-error.hu) |
| 15:12.29 | *** join/#asterisk mutilator (n=animenod@65.111.222.120) |
| 15:16.12 | *** join/#asterisk vgster (n=vgster@cpc2-ledn1-0-0-cust944.leed.cable.ntl.com) |
| 15:16.38 | *** join/#asterisk bjohnson_ (n=bjohnson@jecinc.tor.istop.com) |
| 15:17.51 | *** join/#asterisk ToTo (n=ToTo@host212-207.pool872.interbusiness.it) |
| 15:18.49 | *** join/#asterisk stkn (n=stkn@gentoo/developer/pdpc.active.stkn) |
| 15:19.41 | hmmhesays | bah how to I check what is causing one of my mount points to be busy when I try to umount |
| 15:20.27 | Qwell | hmmhesays: bash |
| 15:20.47 | hmmhesays | um... |
| 15:20.51 | Qwell | lsof can show you |
| 15:20.59 | Qwell | but, the likely culprit is a shell.. |
| 15:21.06 | Qwell | cd / && umount /mnt/blah |
| 15:21.41 | *** part/#asterisk parag_ast (n=root@dxb-b16506.alshamil.net.ae) |
| 15:22.14 | hmmhesays | bah fc kind of sucks some days |
| 15:22.31 | sevard | this guy called me complaining that he can't send out email, swearing "i've been using email for 35 years, i can't believe i can't use it from your connection" |
| 15:22.45 | hmmhesays | 35 years? |
| 15:22.52 | hmmhesays | cool |
| 15:23.04 | hmmhesays | must have been a Uni prof |
| 15:23.10 | sevard | after several traceroutes i find it's a problem with his ISP, a router is busted. |
| 15:23.51 | sevard | i have to hand hold him through reconfiguring his eudora client his words "everyone working on the internet standard uses eudora" and he comes across this and says |
| 15:23.58 | sevard | "now, what's this setting, what's this, pop3, what's that" |
| 15:24.00 | sevard | .................. |
| 15:26.30 | hmmhesays | bah, my PATH is all farked up |
| 15:27.02 | hmmhesays | sevard: where did you find this guy? |
| 15:27.13 | sevard | he called in and i'm the only one working |
| 15:27.16 | hmmhesays | "everyone working on the internet standard?" |
| 15:27.36 | sevard | "i've been around the world all over and never had troubles ever emailing ever." |
| 15:27.39 | *** join/#asterisk Loceur (n=noneya@vsas.veedix.com) |
| 15:28.29 | Loceur | hey guys, I need to troubleshoot a T1 PRI callerid issue. What tools are out there to troubleshoot your callerid on the T1? (It's never worked, we're setting it up) |
| 15:29.12 | *** join/#asterisk IvyUK (n=mark@194.201.148.200) |
| 15:29.46 | hmmhesays | pretty awesome dude there |
| 15:30.22 | *** join/#asterisk salviadud (n=ralfalfa@201.123.130.161) |
| 15:30.38 | *** join/#asterisk klasstek (n=nunyobiz@ip67-154-143-190.z143-154-67.customer.algx.net) |
| 15:31.06 | *** join/#asterisk kfudge (n=kzymxa@24-217-137-21.dhcp.stls.mo.charter.com) |
| 15:31.25 | kfudge | I have a question about an IAX2 trunk |
| 15:31.35 | salviadud | go ahead and ask |
| 15:32.13 | kfudge | It is configured between two servers that are on a vpn. each box can ping the other but in the cli the are unreachable |
| 15:33.00 | salviadud | what kind of vpn are we talking about here? |
| 15:33.05 | salviadud | stunnel perhaps? |
| 15:33.14 | hmmhesays | leeerooooy jenkins |
| 15:33.22 | sevard | No! |
| 15:33.23 | hmmhesays | so I got asterisk running on my wrt |
| 15:33.27 | hmmhesays | its a pretty sweet little setup |
| 15:33.38 | sevard | wow neat i did that like 2 million years ago |
| 15:33.49 | hmmhesays | I am going to kick you in the head |
| 15:33.51 | hmmhesays | hard |
| 15:33.53 | sevard | try. |
| 15:33.58 | hmmhesays | not the one on your shoulders either |
| 15:34.00 | kfudge | it is one maintained on a fortinet device through my isp. the issue is that it worked yesterday |
| 15:34.09 | sevard | it's a big target |
| 15:34.43 | hmmhesays | lol, if happen to be carrying a pair of binoculars |
| 15:34.44 | salviadud | kfudge, do you think your isp might be up to something? |
| 15:34.53 | sevard | hmmhesays: you can't even find me to buy me a beer |
| 15:35.01 | hmmhesays | where are you at? |
| 15:35.06 | hmmhesays | and you're a bit underage |
| 15:35.06 | sevard | pisshaw. |
| 15:35.15 | sevard | your mom doesn't seem to mind |
| 15:35.25 | kfudge | not really we have five facilites all using it, configured as a mesh |
| 15:35.43 | kfudge | and only these two facilities can seem to connect to each other |
| 15:35.58 | hmmhesays | yeah she talks about you all the time |
| 15:36.00 | salviadud | pastebin iax.conf for both sides |
| 15:36.02 | sevard | i kind of made her world |
| 15:36.09 | salviadud | without passwords, of course |
| 15:36.28 | sevard | include passwords, plz |
| 15:36.48 | *** join/#asterisk fgwaller (n=frank@65.105.5.40.ptr.us.xo.net) |
| 15:37.43 | kfudge | nevermind, i changed the config for the external ips and the trunk works |
| 15:37.51 | kfudge | just not through the vpn |
| 15:38.18 | kfudge | but thanks |
| 15:38.42 | kfudge | I was hoping to not have to do it this way but it is showing that there is an issue with the vpn |
| 15:38.54 | *** join/#asterisk X-Rob (n=rob-x@dsl-202-173-151-24.qld.westnet.com.au) |
| 15:39.15 | sevard | hmmhesays: is NetRange: 31.0.0.0 - 31.255.255.255 reserved? |
| 15:41.26 | *** join/#asterisk gandhijee (n=gandhije@mail.win-ent.com) |
| 15:45.51 | hmmhesays | anyone ever run into this zaptel.c:188: warning: ‘fcstab’ defined but not used ? |
| 15:46.21 | *** join/#asterisk marv0997 (n=marv@190.4.2.82) |
| 15:47.34 | [TK]D-Fender | hmmhesays : All the time on FC5. |
| 15:47.48 | [TK]D-Fender | hmmhesays : I have concluded that FC sucks. COMPLETELY |
| 15:48.09 | [TK]D-Fender | hmmhesays : I since replaces with CentOS to much happiness |
| 15:48.15 | coppice | FC gets worse with each release. i used to like RH |
| 15:48.20 | sevard | hmmhesays: do you kno? |
| 15:48.24 | salviadud | rpms maaaaan |
| 15:48.35 | hmmhesays | [TK]D-Fender: you ever fix it? |
| 15:48.49 | [TK]D-Fender | hmmhesays : Yes, inserting a few CentOS disks solved EVERYTHING :) |
| 15:48.52 | *** join/#asterisk topping (n=topping@207.47.6.136.static.nextweb.net) |
| 15:49.00 | hmmhesays | [TK]D-Fender: haha, besides that |
| 15:49.01 | *** join/#asterisk SplasPood (n=jwb@gate.lga2.us.voxel.net) |
| 15:49.16 | [TK]D-Fender | hmmhesays : nope... I gave up when I can't do zaptel without bloodshed |
| 15:50.08 | sevard | [TK]D-Fender: do you know if 31.0.0.0/8 is a special reserved range? |
| 15:50.55 | [TK]D-Fender | sevard : not AFAIK |
| 15:51.00 | key2 | 69 |
| 15:51.24 | sevard | i think it's either special use or private use |
| 15:51.30 | salviadud | i had problems with zaptel the other day, took me a couple of kernel recompiles, but no blood was spilled |
| 15:51.34 | sevard | and i think somebody from it is trying to udp flood me |
| 15:52.15 | sevard | but rfc 3330 and rfc 1918 don't mention it |
| 15:53.13 | salviadud | sevard, could it be some law enforcement agency? |
| 15:53.29 | sevard | NetType: IANA Reserved |
| 15:53.40 | sevard | grr, reserved for what ;/ |
| 15:56.01 | *** join/#asterisk [Airwolf] (n=airwolf@dsl51B67C38.pool.t-online.hu) |
| 15:56.01 | *** join/#asterisk Avalone (n=Avalone_@dial-140.vl-cen-as3.avtlg.ru) |
| 15:56.40 | *** join/#asterisk Toerkeium (i=oo@201.216.206.221) |
| 15:56.57 | sevard | 031/8 Apr 99 IANA - Reserved |
| 15:57.08 | sevard | http://www.iana.org/assignments/ipv4-address-space |
| 15:57.14 | sevard | straight from the horse's mouth |
| 15:57.34 | coppice | reserved for UDP flooders :-) |
| 15:57.37 | hmmhesays | problem found and fixed |
| 15:57.41 | sevard | heh |
| 15:57.41 | watchy | anyone here that can help fix echo issues on a TDM400p? |
| 15:57.52 | sevard | my router ought to not be letting that through |
| 15:58.13 | hmmhesays | zaptel now compiles |
| 15:58.15 | hmmhesays | woohOO |
| 15:58.41 | watchy | whoever helps me fix my tdm400p echo issues gets a hug |
| 15:58.48 | Qwell | file: ^^ |
| 15:58.54 | *** join/#asterisk DarKnesS_WolF (n=wolf@82.201.233.171) |
| 15:58.55 | hmmhesays | a hug is not going to pay for my hooker tonight |
| 15:59.06 | Loceur | depends on where you go |
| 15:59.06 | file | hmmhesays: you have to pay for those? but you're a nice lookin' guy! 'tsk 'tsk |
| 15:59.12 | watchy | how about $25 then :( |
| 15:59.16 | watchy | and a hug |
| 15:59.20 | Qwell | "Guys, seriously...the hooker gave the money back" |
| 15:59.33 | hmmhesays | file: it was a joke |
| 15:59.38 | file | hmmhesays: :D |
| 15:59.49 | watchy | this echo issue is driving me freaking insane |
| 15:59.55 | salviadud | insane? |
| 16:00.02 | *** join/#asterisk Ebola (n=Ebola@81-86-155-65.dsl.pipex.com) |
| 16:00.02 | salviadud | i would like to see that... |
| 16:00.03 | coppice | watchy: don't they just? |
| 16:00.25 | watchy | im starting to think digiums cards suck or something |
| 16:00.37 | Qwell | watchy: no, analog telephony sucks |
| 16:00.44 | coppice | well, the tdm400p has issues |
| 16:00.45 | watchy | qwell: i agree |
| 16:00.48 | salviadud | that's right |
| 16:00.53 | salviadud | analog is old |
| 16:00.57 | Qwell | besides, echo comes from the far end |
| 16:01.05 | coppice | analog is proven |
| 16:01.12 | Qwell | watchy: just mess with your rxgain settings |
| 16:01.16 | Qwell | bbl, work |
| 16:01.35 | watchy | i get echo when people call out but only for a few seconds |
| 16:01.39 | watchy | it goes away after that |
| 16:02.11 | *** join/#asterisk MstlyHrmls (n=mh@66.195.193.151) |
| 16:02.42 | tzanger | watchy: that's the echo canceller training |
| 16:03.12 | coppice | its training to be a PITA :-) |
| 16:03.14 | watchy | whats the best way of testing each line for echo and changing settings |
| 16:03.18 | tzanger | you can try to pretrain it with echotraining=800 but personally i don't like that. you can also try fxotune to try and get the simple filter in the FXO module better matched too |
| 16:03.24 | wangster | watchy, look on voip-info under "causes of echo". |
| 16:03.53 | watchy | so what needs to be done first? fxotune? |
| 16:04.29 | sevard | pppfffft |
| 16:04.37 | watchy | DGM-TE110P |
| 16:04.43 | watchy | should i just get a t1 card? |
| 16:04.58 | *** join/#asterisk _alex_mx_ (n=_alex_mx@200.78.229.18) |
| 16:05.19 | sevard | send me the money for your t1 card |
| 16:05.49 | watchy | and youll fix my echo? |
| 16:07.46 | *** join/#asterisk future (i=future@future.unknown.sk.cx) |
| 16:08.39 | *** join/#asterisk javar (n=javar@Dynamic-IP-cr20011876227.cable.net.co) |
| 16:08.43 | *** part/#asterisk future (i=future@future.unknown.sk.cx) |
| 16:09.02 | hmmhesays | the zaptel problem is a kernel issue |
| 16:09.06 | hmmhesays | *was a |
| 16:10.05 | sevard | so, who access to subscriber information off of revesed telephone numbers |
| 16:10.32 | *** join/#asterisk CrashHD (i=CrashHD@c-67-182-167-222.hsd1.ca.comcast.net) |
| 16:11.02 | *** part/#asterisk smackus (n=ckwall@63.149.122.93) |
| 16:12.27 | hmmhesays | she's a mine oh my, my minds made up yeah i'm so sure, cause theres none so fine |
| 16:12.58 | Loceur | anyone have any good tutorials on callerid setup? |
| 16:13.06 | Loceur | links, that is |
| 16:13.09 | *** join/#asterisk mog (i=ejabberd@68.62.237.103) |
| 16:13.09 | *** mode/#asterisk [+o mog] by ChanServ |
| 16:13.11 | hmmhesays | what do you need to setup? |
| 16:13.19 | Loceur | callerid on a PRI |
| 16:13.25 | Loceur | hey guys, I need to troubleshoot a T1 PRI callerid issue. What tools are out there to troubleshoot your callerid on the T1? (It's never worked, we're setting it up) |
| 16:13.33 | Loceur | ^^ original question |
| 16:13.47 | sevard | she's gone she's gone she's gone oh man let it go |
| 16:14.01 | Loceur | it may be a config issue too, I just need a direction on the callerid thing, and voip-info.org is ellusive at best |
| 16:14.14 | hmmhesays | i'm going to go to the pawn shop and buy all the chumbawumba cd's |
| 16:14.17 | sevard | XFCE is the best window manager. I'm going to have to use it until E17 lands home. |
| 16:14.22 | sevard | hmmhesays: you're gay like that. |
| 16:14.49 | hmmhesays | stuff it biatch |
| 16:14.49 | Nugget | OS X > * |
| 16:14.58 | sevard | gimmedrunkz |
| 16:15.03 | hmmhesays | i would have to say mac has the best windows manager |
| 16:15.12 | sevard | Nugget: only if it didn't crash as much as my 98 box |
| 16:15.20 | hmmhesays | it would be kickass if someone would port it |
| 16:15.38 | sevard | it'sn it aqua proprietary? |
| 16:15.42 | sevard | isn't* |
| 16:15.44 | hmmhesays | yeah |
| 16:15.56 | sevard | port it then. |
| 16:16.03 | Nugget | Linux lacks the susbsystems you'd need to do aqua anyway. |
| 16:16.13 | sevard | jebus |
| 16:16.15 | sevard | 017/8 Jul 92 Apple Computer Inc. |
| 16:16.37 | hmmhesays | ok what is the difference between su and su - |
| 16:16.51 | Nugget | "su -" replicates a login, with the target user's full environment |
| 16:16.57 | Nugget | su without the - just sets euid. |
| 16:17.05 | sevard | su - replaces the enviroment |
| 16:17.06 | [TK]D-Fender | hmmhesays : The "-" sets env vars (like path), etc |
| 16:17.09 | sevard | i'm lagged. |
| 16:17.23 | hmmhesays | I am in shock and awe right now |
| 16:17.27 | hmmhesays | I expected to be berated |
| 16:17.31 | [TK]D-Fender | hmmhesays : as part of what everyone else said :) |
| 16:17.33 | sevard | because everyone knew that and you're and itiot? |
| 16:17.35 | sevard | idiot***** |
| 16:17.37 | *** join/#asterisk henk (n=marius@s5593c2e9.adsl.wanadoo.nl) |
| 16:17.42 | hmmhesays | now I feel at home |
| 16:17.47 | sevard | you SUCK! |
| 16:17.52 | MikeJ | hmmhesays!! |
| 16:17.58 | hmmhesays | hola MikeJ |
| 16:18.09 | hmmhesays | [TK]D-Fender: I did fix my zaptel problem in FC5 |
| 16:18.14 | [TK]D-Fender | I SUCK at linux and the only reason I knew this is one of my cusomers set up an account for me for maintenance and I had to "sudo su -" to get root |
| 16:18.25 | [TK]D-Fender | hmmhesays : Congrats, what did it take? |
| 16:18.53 | sevard | 'sudo su' is gay, just edit /etc/suauth |
| 16:19.35 | hmmhesays | http://lists.digium.com/pipermail/asterisk-dev/2006-February/018702.html |
| 16:19.50 | [TK]D-Fender | sevard : I barely know enough to EXECUTE it... go back to your Jurassic little world you 2,000,000 year old admin! ;) |
| 16:20.00 | sevard | :( |
| 16:20.26 | sevard | you're the freaking one who has to type in CAPS all day |
| 16:20.31 | henk | I have asterisk and php-agi set up to work together, very cool stuf!. If I call in to my server a php script answers the phone does text2speech all interactive very nice. However I'm trying to get it to call out. I'm dropping a call file into the spool and it call my cellphone but as soon as the call is placed the agi php scripts starts running. It does not wait for me to pick up the phone. How can I detect is the line is open on both ends ? |
| 16:21.08 | sevard | henk: have your sister admin your box |
| 16:21.43 | henk | what kind of answer is that !? |
| 16:21.51 | eKo1 | lol |
| 16:22.31 | nettie | Hi guys I'm using trunk with jitterbuffer on chan_sip but I'm still hanving huuuuuge voice quality issues. Anyone know what could be wrong please? |
| 16:22.41 | anthm | try reversing the call file so it makes the call first then connects the script the other way is broken |
| 16:22.44 | sevard | nettie: codec/available bandwidth |
| 16:22.56 | nettie | 256kb/sec |
| 16:23.00 | nettie | 711u |
| 16:23.12 | nettie | I'm having issues of course when the lin kis lagged |
| 16:23.15 | henk | anthm, how do i do that ? |
| 16:23.17 | sevard | is that 256k/b if upstream? |
| 16:23.25 | sevard | s/if/of |
| 16:23.25 | nettie | but jitterbuffer is supposed to handle this |
| 16:23.27 | nettie | yes |
| 16:23.30 | nettie | sevard |
| 16:23.32 | nettie | yer |
| 16:23.35 | nettie | 1280 in |
| 16:23.38 | nettie | 256 out |
| 16:23.45 | sevard | that's a lot of bandwidth. |
| 16:23.47 | nettie | DSLAN is synchronized at 320 |
| 16:23.53 | nettie | exaclty |
| 16:24.03 | nettie | the sip server is at the colo |
| 16:24.03 | anthm | is the channel in the call file what's calling the phone line and the app is your script? |
| 16:24.07 | sevard | are you translating codecs at all? |
| 16:24.24 | *** join/#asterisk anonymouz666 (n=anonymou@20151125001.user.veloxzone.com.br) |
| 16:24.37 | sevard | does your colo run QoS on their gateway? |
| 16:24.41 | nettie | no, not at all, just to keep it as simple as possible I created an extensin which plays an mp3 file |
| 16:24.51 | henk | anthm, this is my call file : http://pastebin.ca/100974 |
| 16:24.59 | nettie | sevard I'm not sure but I have 5Mbit there |
| 16:25.07 | nettie | and we're talking about a single cal |
| 16:25.08 | sevard | wow |
| 16:25.11 | sevard | yeah. |
| 16:25.16 | sevard | i'm talking about your CPU |
| 16:25.19 | nettie | at home I have 10mbit fiber |
| 16:25.24 | nettie | and voice is crystal clear |
| 16:25.28 | nettie | it's the DSL |
| 16:25.36 | nettie | if the link is not lagged voice is great |
| 16:25.50 | eKo1 | yummy fiber |
| 16:25.54 | sevard | so the same server worked great at your house |
| 16:25.55 | anthm | does that exten have an answer in it? |
| 16:26.00 | nettie | CPU? the box has a P4 2.8GHz |
| 16:26.00 | sevard | but it doesn't work so great at your colo |
| 16:26.05 | nettie | sure it does |
| 16:26.06 | nettie | no |
| 16:26.06 | sevard | did i get that right |
| 16:26.06 | nettie | no |
| 16:26.14 | henk | anthm, yes the php agi does an $agi->answer |
| 16:26.15 | nettie | serever is ALWAYS at the colo |
| 16:26.21 | nettie | phones might be behind the DSL |
| 16:26.22 | nettie | or |
| 16:26.32 | nettie | behind fiber |
| 16:26.41 | anthm | try skipping that |
| 16:26.45 | nettie | I'm sure it's not an issued of bw at the colo |
| 16:26.48 | henk | anthm, just a sec |
| 16:27.03 | nettie | the issue is the dsl |
| 16:27.33 | nettie | when it's busy |
| 16:27.36 | nettie | voice gets choppy |
| 16:27.40 | anthm | does the sip in turn call out somewhere ? |
| 16:27.40 | nettie | JB should preevent this |
| 16:28.15 | anthm | if the other side of that sip call answers then calls somewhere else it will trick the call file into completing the call too soon |
| 16:28.16 | eKo1 | nettie: you need to stop downloading pr0n when you're on the phone |
| 16:28.22 | *** join/#asterisk Mercestes (n=Merceste@216.54.143.2) |
| 16:28.42 | anthm | so nm about the answer line thing |
| 16:28.54 | anthm | it's probably on the other end of your sip call that is making it happen |
| 16:29.19 | Mercestes | anyone seen this before? WARNING[14950]: ast_expr2.fl:183 ast_yyerror: ast_yyerror(): syntax error: syntax error, unexpected $end, expecting TOK_MINUS or TOK_COMPL or TOK_LP or TOKEN; I'm using Asterisk 1.2.9. |
| 16:29.39 | henk | anthm, hmm that had no effect the script starts running anyway |
| 16:29.53 | anthm | yah read above |
| 16:30.35 | nettie | eKo1 ehehe thanx, this will definitely help |
| 16:30.36 | nettie | :) |
| 16:30.38 | anthm | that spooler code in the call file thing should wait for the channel to be in answer state before starting the script so whatever you are calling on sip is doing answer when it shouldnt probably |
| 16:31.45 | anthm | the answer signalling should originate from the far end and translate back to the call file and trigger it to move on and if anything in between sends an answer signal then it will jump the gun |
| 16:31.57 | henk | anthm, the call files sip line is a call to my cellphone though a sip provider here in holland. I dont think anyone in between is answering |
| 16:31.57 | rpm | Mercestes: looks like you've got a syntax error in your dialplan, or its not filling a variable. |
| 16:32.10 | *** join/#asterisk Qwell[] (i=north@unaffiliated/qwell) |
| 16:32.10 | *** mode/#asterisk [+o Qwell[]] by ChanServ |
| 16:32.25 | Mercestes | rpm: Oh....so it's dialplan specific? yay! I thought my source code was fargered. |
| 16:32.48 | anthm | trace the sip and see if you get a 200 from the sip provider before your phone is offhook |
| 16:33.00 | henk | anthm, ok hang on |
| 16:33.16 | anthm | the outgoing_exten func in the code should be waiting for an answer which in sip is a 200 |
| 16:34.25 | *** part/#asterisk acrg (n=aragon@decoder.geek.sh) |
| 16:36.52 | jbalcomb | I'm happy that people knew the different between `su` and `su -` but sad that one wouldn't just `man su` =) |
| 16:37.27 | *** join/#asterisk laichzeit (n=immortal@dsl-145-177-185.telkomadsl.co.za) |
| 16:37.59 | *** join/#asterisk mtaht4 (n=m@c-71-198-23-124.hsd1.ca.comcast.net) |
| 16:38.02 | laichzeit | Hi everyone, I'm getting a "call forward uncondtional" message from the asterisk-lady everytime I try and dial an extension, anyone know what setting I'm missing? |
| 16:38.20 | jbalcomb | [TK]D-Fender: Do you need anything more from me to begin our work? If not, when can we get the ball rolling? |
| 16:38.26 | *** join/#asterisk RoyK (n=roy@183.80-203-45.nextgentel.com) |
| 16:38.37 | [TK]D-Fender | jbalcomb : PM |
| 16:38.51 | hmmhesays | wiicky wicky wild wild |
| 16:38.52 | sevard | jbalcomb: but '`man su`' is massivly incorrect |
| 16:39.32 | jbalcomb | sevard: `man su` "OPTION - make this a login shell" |
| 16:39.47 | sevard | jbalcomb: but '`man su`' is massivly incorrect |
| 16:39.51 | jbalcomb | sevard: though I suppose that is a terrible explanation. ;) |
| 16:39.59 | sevard | jbalcomb: but '`man su`' is massivly incorrect |
| 16:40.05 | henk | anthm, pff that is a lot of output and I am seeing some 200 messages but i'm not quite sure where they are comming from |
| 16:40.23 | anthm | you have tethereal |
| 16:40.51 | anthm | you can do tethereal -i eth0 udp and port 5060 |
| 16:41.07 | *** join/#asterisk wasim (n=wasim@pdpc/supporter/active/wasim) |
| 16:41.51 | sevard | `man su` |
| 16:41.51 | sevard | bash: SU(1): command not found |
| 16:41.56 | sevard | you fail at bash, jbalcomb |
| 16:42.05 | *** part/#asterisk mhnoyes (n=mhnoyes@dialup-4.246.6.57.Dial1.SanJose1.Level3.net) |
| 16:42.23 | anthm | man man lol |
| 16:42.44 | sevard | there are lots of man options. you might need to man man sometime. |
| 16:42.59 | *** join/#asterisk Bullseye_Network (n=info@216.143.192.69) |
| 16:43.18 | henk | anthm, as far as I can tell the 200 comes from my sip provider as soon as my phone starts ringing.. Does that mean they are badly configured ? |
| 16:43.21 | anthm | or man bash to learn how to formulate a command line to man man |
| 16:43.48 | anthm | yes that is what it means |
| 16:44.27 | anthm | they answer the call as soon as they gateway to the pstn rather than when the pstn answers it |
| 16:45.34 | sevard | the tab key taught me linux. |
| 16:45.51 | henk | anthm, can you recomend me a better sip provider ? |
| 16:46.02 | anthm | not in holland |
| 16:46.23 | henk | anthm, I dont care, Most of these calls are going to be international anyway |
| 16:46.34 | E-bola | Do anybody understand the ACK stuff asterisk seems to be doing regularly? |
| 16:46.40 | E-bola | its a bit spamming in my logs |
| 16:46.55 | E-bola | i asume its something similar to tcp/ip ack packets |
| 16:47.20 | anthm | well we have asterlink in VA usa but if you start at holland to come to us then back intl then that's an issue if you are terminating in us pstn give us a call |
| 16:48.39 | *** part/#asterisk nettie (i=[U2FsdGV@85-18-54-38.ip.fastwebnet.it) |
| 16:49.16 | henk | anthm, anyway thank you for your help. I thought I was doing it all wrong, turns out i'm not that dumb ;). I'm going to get some food now. bye |
| 16:50.33 | Loceur | anyone have any good tutorials/help on callerid and PRIs? |
| 16:50.45 | Qwell[] | Loceur: in what regard? |
| 16:51.36 | Loceur | we have a PRI but don't seem to be getting incoming callerids... |
| 16:51.40 | Loceur | can't find out why |
| 16:52.07 | Loceur | -- Accepting call from '' to '2344202' on channel 0/1, span 1 |
| 16:52.09 | Qwell[] | Loceur: 1) Do you get CID from your telco? |
| 16:52.19 | Qwell[] | 2) Do you have usecallerid=yes? |
| 16:52.32 | Loceur | Qwell: 1)we SHOULD be, 2) yes |
| 16:52.41 | Loceur | I'd like to verify 1 though |
| 16:52.42 | Qwell[] | Loceur: verify |
| 16:53.02 | Loceur | any way to from our end? |
| 16:53.15 | Qwell[] | besides picking up a phone, and calling them? :) |
| 16:53.29 | Loceur | yes... that's much more trouble than you know |
| 16:53.34 | Loceur | but I'm doing that too |
| 16:53.52 | Loceur | our "Global Accounts Rep" is less than usful |
| 16:53.54 | Loceur | useful |
| 16:54.19 | Loceur | you gotta love a monopoly... |
| 16:56.06 | Loceur | Local Customer Care: 1800... busy |
| 16:56.33 | E-bola | chan_sip.c:1401 __sip_ack: Stopping retransmission on '0fafef114a7aa0111dbb4fc76b99c518@213.150.48.236' of Request 102: Match Found |
| 16:56.39 | E-bola | can any1 explain to me what that means? |
| 16:57.04 | Loceur | ebola... lol nice nick |
| 16:57.31 | _alex_mx_ | Loceur, is it isdn pri? |
| 16:57.48 | file | E-bola: it means chan_sip sent out a packet, and needed to know that it was sent... by receipt of an ACK or whatever, so it received the ACK and won't resend the original packet |
| 16:58.11 | E-bola | file: so its a way for asterisk to "keep track" of the sip device or what? |
| 16:58.17 | E-bola | i mean to verify they are still there or? |
| 16:58.50 | file | no... it's to make sure that the device received the SIP packet |
| 16:58.55 | file | what the packet is about I don't know |
| 16:59.20 | E-bola | How could i find out? |
| 16:59.42 | file | sip debug |
| 16:59.47 | E-bola | i got that turned on |
| 17:00.10 | E-bola | but there dont seam to be any text about the 102 extension |
| 17:00.15 | file | so watch for the packets... |
| 17:00.20 | E-bola | which i asume the "Request 102: Match Found" is about |
| 17:00.23 | file | no |
| 17:00.30 | E-bola | SIP/2.0 200 OK |
| 17:00.33 | E-bola | ohh thats the 200? |
| 17:00.39 | E-bola | some sort of sip protocol code? |
| 17:00.47 | file | yes... |
| 17:01.07 | E-bola | knwo a place i can see a list of the codes meanings? |
| 17:01.17 | file | Google. |
| 17:01.30 | E-bola | well so far i've only gotten that it means ok |
| 17:01.36 | E-bola | no idea whats ok or why it keeps sending it |
| 17:01.44 | Loceur | _alex_mx_: yes PRI |
| 17:01.55 | file | E-bola: why are you so interested? |
| 17:01.56 | Loceur | 23 + 1 D |
| 17:02.13 | E-bola | file: because i dont like seeing it in my console |
| 17:02.22 | file | so turn off debug, and turn off sip debug |
| 17:02.23 | E-bola | + i'd like to know whats going on in general |
| 17:02.33 | E-bola | i normally got sip debug off |
| 17:02.46 | E-bola | i connect to asterisk like: asterisk -rvvv |
| 17:02.56 | E-bola | turning off debug means using less v's ? |
| 17:03.14 | file | it means go into logger.conf and make sure debug isn't being printed to console |
| 17:03.41 | file | if you really want to know what's going on, then you'll have to learn about SIP |
| 17:03.44 | E-bola | Well i could do that, but i'd still like to know why its sending those packages :) |
| 17:03.59 | file | are you using the qualify option? |
| 17:04.07 | E-bola | yes |
| 17:04.12 | file | that's probably why |
| 17:04.14 | _alex_mx_ | Loceur, at the CLI...pri set debug file /tmp/pri_debug, then pri intense debug span 1, place call, stop debugging, unset debug file, look at file and look in the SETUP section of the call for CID |
| 17:04.19 | E-bola | sounds logical ya |
| 17:04.35 | file | chan_sip is sending out an OPTIONS packet to make sure the device is still there and to gauge it's response time, the device is sending back a 200 OK on acknowledgment that it received the OPTIONS |
| 17:04.46 | E-bola | Yep thats prolly it |
| 17:04.48 | *** join/#asterisk _4d4m_ (n=adam@62.69.102.99) |
| 17:04.48 | file | chan_sip then stops the retransmit timer on the OPTIONS packet since it got the 200 OK |
| 17:04.56 | E-bola | Are there any reasons to use qualify on a small lan? |
| 17:04.57 | file | dialog is finished and done |
| 17:05.16 | file | not really, but I'm not you |
| 17:05.41 | *** join/#asterisk |dennis| (n=dennis@200.32.215.82) |
| 17:05.52 | E-bola | guess its not completely useless since my devices are behind a nat |
| 17:05.54 | hmmhesays | time to see if I can tank this wrt |
| 17:06.00 | E-bola | it makes sure the firewall doesnt kill the sessions |
| 17:06.08 | Corydon-w | qualify is nice in that it tells you immediately if somebody's phone just rebooted |
| 17:06.44 | E-bola | I guess i can leave it on now that i know what the messages mean |
| 17:07.01 | E-bola | DEBUG[16003]: chan_sip.c:1323 __sip_autodestruct: Auto destroying call '15a97e6368cb365652c66acc13259b7d@127.0.1.1' |
| 17:07.01 | E-bola | Destroying call '15a97e6368cb365652c66acc13259b7d@127.0.1.1' |
| 17:07.09 | E-bola | i wonder if thats comming from the process |
| 17:07.49 | file | yes, yes it is and I'm not even going to ask about the IP... |
| 17:08.04 | Qwell[] | 127.0.1.1 is valid :D |
| 17:08.18 | Qwell[] | 127.0.0.0/8 should be valid everywhere |
| 17:08.20 | E-bola | why is it calling it self? |
| 17:08.56 | E-bola | i dont got any device registering on the asterisk server it self |
| 17:09.02 | file | E-bola: it's not, you just don't know what all this means |
| 17:09.05 | *** join/#asterisk Kast (n=Kast@CPE001195d72b23-CM0011e6bddd29.cpe.net.cable.rogers.com) |
| 17:09.11 | file | or how SIP works |
| 17:09.15 | E-bola | file: Im trying to learn :) |
| 17:09.23 | Kast | anyone had this problem before --> Incoming call: Got SIP response 500 "Internal Server Error" back from 192.168.1.101 |
| 17:09.39 | E-bola | kast: sounds like a http server hehe |
| 17:09.49 | Kast | ya, that's what i though |
| 17:10.00 | file | E-bola: there's tons of docs out there on SIP, tons of RFCs, tons of pages on how it works in the basic sense... and you have the code for chan_sip |
| 17:10.22 | *** part/#asterisk websae (n=websae@adsl-64-149-206-121.dsl.milwwi.sbcglobal.net) |
| 17:10.35 | file | you should take a gander and Google if you really want to learn aboot it |
| 17:10.36 | E-bola | file: if i read some sip rfc's it would tell me my asterisk is telling me its auto destroying a call every few secs? |
| 17:11.13 | file | you would know that it's destroying a dialog created by the OPTIONS since it received a response |
| 17:11.29 | file | you would also know what a dialog is and what it consists of |
| 17:11.32 | E-bola | so asterisk considers any sip dialog a call? |
| 17:12.02 | file | that's just what the message says, it could easily be changed to say dialog |
| 17:12.14 | file | just nobody but developers usually see it, and we don't complain about the word "call" instead of "dialog" |
| 17:12.26 | E-bola | ok |
| 17:12.40 | E-bola | I guess i should lower my verbosity for now |
| 17:13.01 | file | it's amusing when people turn up verbosity and turn on all sorts of debug |
| 17:13.07 | file | they start freaking out because they see strange things |
| 17:13.10 | E-bola | I find that asterisk is pretty reasonably docuemented for an OS program, its just not very well organised documentation |
| 17:13.21 | laichzeit | anyone know how to turn off call forwarding for an extension? I can't dial between extensions asterisk keeps ending up doing a call-fwd-unconditional |
| 17:13.27 | E-bola | file: well thats how it is when u dont know what it means |
| 17:13.35 | E-bola | my first thought was somebody was trying to hack the server :P |
| 17:13.40 | file | E-bola: yes, but under normal circumstances someone shouldn't touch those :) |
| 17:13.48 | file | laichzeit: is it forwarded on the phone? |
| 17:13.55 | laichzeit | file, nope |
| 17:13.55 | *** join/#asterisk trimor (n=kvirc@202.125.143.66) |
| 17:14.10 | E-bola | file: well thanks for explaining the meanings to me now |
| 17:14.11 | *** join/#asterisk viler (i=1000@200.114.70.228) |
| 17:14.12 | trimor | hi ! can ne body tell from where i can get the addmail utility |
| 17:14.15 | file | laichzeit: are you using AMP? Asterisk@Home? Trixbox? FreePBX? |
| 17:14.15 | trimor | ?? |
| 17:14.16 | E-bola | u learn something new everyday :) |
| 17:14.23 | laichzeit | file, Trixbox |
| 17:14.46 | file | laichzeit: they implement their own dialplan logic and stuff for doing call forwarding in Asterisk, you'd best join their channel to see where it is stored and if you can change it |
| 17:14.53 | *** part/#asterisk javar (n=javar@Dynamic-IP-cr20011876227.cable.net.co) |
| 17:14.54 | trimor | i want to setup voicemail |
| 17:14.58 | file | laichzeit: it's not part of a regular Asterisk install |
| 17:15.05 | trimor | i cant find addmailbox |
| 17:15.07 | laichzeit | ok, thanks |
| 17:15.17 | trimor | ?? |
| 17:15.23 | file | trimor: that's been deprecated for some time, the first time someone leaves a message the folders and stuff are automatically created |
| 17:15.36 | E-bola | I gave up on pbx and trixbox cuz it wasnt at all the same way u do things as with pure asterisk |
| 17:15.38 | trimor | thanx |
| 17:15.45 | E-bola | and my bet was there was more docs on a normal asterisk |
| 17:15.53 | *** join/#asterisk generalhan (i=general_@ip67-90-64-2.z64-90-67.customer.algx.net) |
| 17:15.55 | *** join/#asterisk WeirdM (n=weirdm@udp079073uds.hawaiiantel.net) |
| 17:16.29 | file | E-bola: configuring a normal Asterisk install forces you to think... good exercise |
| 17:16.32 | trimor | well i get login incorrect message when i try to connect to the voicemail box |
| 17:16.35 | laichzeit | E-bola, TrixBox "does the right thing" for most situations. |
| 17:16.46 | file | trimor: is it setup in voicemail.conf? |
| 17:17.12 | E-bola | file: thats why i chose it |
| 17:17.21 | trimor | yes |
| 17:17.23 | E-bola | to better understand the concepts involved |
| 17:17.31 | laichzeit | but I need the AMP webgui thingie so clients can add their own extensions without bothering me ;) |
| 17:17.35 | *** join/#asterisk dandan (i=dandan@pacanka.com) |
| 17:17.37 | file | trimor: pastebin your voicemail.conf and the dialplan portion that is going to voicemail |
| 17:17.40 | dandan | :) hello |
| 17:17.43 | file | trimor: http://www.pastebin.ca/ |
| 17:17.47 | file | dandan: greetings |
| 17:17.58 | file | laichzeit: hehe just when they want more then it offers... then freak out! |
| 17:18.04 | trimor | ok |
| 17:18.11 | dandan | i have a question regarding pressing 0 while in vm... it doesn't work properly... it doesn't work at all... |
| 17:18.21 | dandan | anyone having a working example? |
| 17:18.42 | file | dandan: voicemail or voicemailmain? |
| 17:18.56 | dandan | hm, operator=yes |
| 17:19.04 | file | I meant where are you hitting 0 from |
| 17:19.09 | dandan | voicemail |
| 17:19.23 | dandan | while i am listening to a greeting/while leaving a msg... |
| 17:19.27 | dandan | afterwards... |
| 17:19.40 | dandan | it says to put a o extension somewhere... |
| 17:19.47 | dandan | and wiki is vague... :) |
| 17:19.49 | *** join/#asterisk DarKnesS_WolF (n=wolf@196.218.72.125) |
| 17:20.23 | file | dandan: labbing it up right now |
| 17:20.29 | dandan | ty |
| 17:20.48 | wasim | anybody seen citats lately? |
| 17:21.06 | dandan | ~citats |
| 17:21.19 | dandan | ~seen citats |
| 17:21.35 | jbot | citats <n=james@bgp925576bgs.brghtn01.mi.comcast.net> was last seen on IRC in channel #asterisk, 283d 11h 35m 42s ago, saying: 'ciscos data stuff is solid. voice gear is a different software, its loaded with bugs'. |
| 17:21.36 | tzanger | haven't seen him in ages |
| 17:22.01 | wasim | yeah, i miss him |
| 17:22.48 | tzanger | I knew him years and years before asterisk.. him and his brother both |
| 17:23.51 | *** join/#asterisk pdtmobile (n=ptinsley@c-68-53-40-50.hsd1.tn.comcast.net) |
| 17:24.26 | trimor | @file pastebin is producing errors and not working |
| 17:24.36 | trimor | so please check the private message |
| 17:26.17 | file | dandan: looks like per context isn't working, setting it in general does |
| 17:27.10 | file | unless... |
| 17:28.05 | file | oh, duh |
| 17:28.43 | file | it does work |
| 17:29.14 | *** join/#asterisk juanjoc (n=juanjoc@200.73.189.82) |
| 17:34.03 | file | dandan: where do you have operator=yes exactly? |
| 17:34.35 | *** join/#asterisk |dennis| (i=dennis@200.32.215.84) |
| 17:35.38 | *** join/#asterisk kFuQ (n=somedude@c-67-185-114-199.hsd1.wa.comcast.net) |
| 17:37.54 | *** join/#asterisk asterisk-newbie1 (n=asteri@59.93.79.150) |
| 17:40.34 | hmmhesays | not to try and get a usb soundcard to work on this wrt |
| 17:40.35 | hmmhesays | fun |
| 17:40.46 | *** join/#asterisk _problem_ (n=lokesh_k@estrela.nortenet.pt) |
| 17:41.02 | WeirdM | Is there a good web page or document on configuring asterisk somewhere? |
| 17:41.30 | dandan | ~docs |
| 17:41.32 | jbot | i heard docs is probably Documentation can be found at http://www.digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or http://www.voip-info.org/wiki-Asterisk or http://www.asteriskdocs.org or http://www.oreilly.com/catalog/asterisk or http://www.asteriskguru.com, or http://www.astmasters.net/howtos.html |
| 17:41.34 | *** join/#asterisk trimor (n=kvirc@202.125.143.69) |
| 17:41.38 | *** join/#asterisk inv_arp[work] (i=junya@c-67-191-62-53.hsd1.fl.comcast.net) |
| 17:41.41 | dandan | ~books |
| 17:41.52 | dandan | ~book |
| 17:41.53 | jbot | hmm... book is a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 17:41.58 | MikeJ | ~thebook |
| 17:41.59 | jbot | extra, extra, read all about it, thebook is a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 17:42.38 | hmmhesays | It is always kind of comical when people ask if there is a good configuration guide |
| 17:42.38 | WeirdM | Thank You! |
| 17:43.01 | asterisk-newbie1 | hi qwell or some expert on asterisk: A quick question for you. I have MANY numbers which change dynamically and all these are listed as contexts in real time extensions in the database. Since there are lot of numbers and they change dynamically, I cannot have them listed in extensions.conf as it is not a viable option. An incoming number comes with context and extensions as DID. I have set a rule in default [_xxxxxxxx] -> switch to realtime/${CON |
| 17:43.34 | file | you can't pattern match a context |
| 17:43.50 | asterisk-newbie1 | However, since extensions.conf is loaded during start, it does not handle real time context for real time database. Hardcoding to number did in place works |
| 17:44.08 | *** join/#asterisk tsurk0 (n=tsurko@85.187.160.157) |
| 17:44.24 | asterisk-newbie1 | file: Any suggestions how I can handle that |
| 17:44.38 | file | out of curiosity, why are you getting a context with your number? |
| 17:45.15 | Qwell[] | pattern matched contexts...heh, that's funny |
| 17:45.20 | asterisk-newbie1 | thats how the sip provider does. We have lot of numbers from them and an incoming call comes to that extension (DID) and context (DID) |
| 17:45.35 | hmmhesays | you could pattern match a context with an agi if you really wanted |
| 17:45.35 | asterisk-newbie1 | qwell: Good day today! |
| 17:45.36 | file | but you can't send contexts over SIP |
| 17:46.05 | hmmhesays | you could if you had something like sip:pattern#context@host |
| 17:46.29 | file | hmmhesays: that's cheating, still sends it as the extension then he would have to dissect it on his side and do a Goto or something |
| 17:46.38 | asterisk-newbie1 | hmmhesays: I have investigated the option of using AGI. However, I cannot switch to realtime from agi. Isn;t it so? |
| 17:46.38 | file | or you could send it as a custom header, but anyway |
| 17:46.41 | hmmhesays | file: cheating maybe, but possible, yes |
| 17:46.59 | hmmhesays | pattern/context is arbitrary to hard you parse it |
| 17:47.06 | hmmhesays | *how you parse it |
| 17:47.11 | carrar | Life is all about cheating death! |
| 17:47.51 | asterisk-newbie1 | I let everything go to default context, I match it with [_XXXX...] -> switch to realtime(). I am not sure how sip:pattern#context@host would work? |
| 17:48.07 | generalhan | hey guys ... im running asterisk-1.2.9.1 and i need to get the addon pkg can i use the 1.2.3 addons version with this version of asterisk ? |
| 17:48.24 | Qwell[] | generalhan: use whatever the latest version of -addons is |
| 17:48.48 | generalhan | Qwell[]: thanks i didnt know if i had to use the version that came out with this ver. of * |
| 17:48.56 | asterisk-newbie1 | hmmhesays: How can I use agi for this purpose? |
| 17:48.58 | file | asterisk-newbie1: sounds like you're overcomplicating things... if all calls are going to the same incoming context, why won't doing a realtime switch there work? |
| 17:49.39 | asterisk-newbie1 | file: I have a seperate context for each did. I need to send the call to the appropriate context |
| 17:50.04 | asterisk-newbie1 | <PROTECTED> |
| 17:50.19 | tamp4x | chan_zap.c:6131 ss_thread: CallerID returned with error on channel 'Zap/2-1' |
| 17:50.28 | file | okay, but a context has to exist in extensions.conf for the realtime switch to work, and you can't do a pattern match on a context |
| 17:50.29 | tamp4x | anyone know the reason for that? |
| 17:51.05 | linlin | anyone know of a softphone that would be suitable for a radio station type setup? Like have a bunch of people on hold and jsut hit hte button and htey are live |
| 17:51.15 | dandan | anyone doing different ringtones with gxp-2000? |
| 17:51.39 | asterisk-newbie1 | file: yep file. Thats the problem here. I tried adding another extension. Went to realtime databse and even goto there is looking for that context in realtime. |
| 17:52.31 | file | asterisk-newbie1: and there's no answer to it, the context has to exist in extensions.conf - now there is a patch on the bug tracker somewhere to override that and make realtime the predominant thing for the PBX core... but it's completely unsupported at this time |
| 17:52.34 | file | so use at your own risk |
| 17:52.43 | asterisk-newbie1 | file: As I am not experienced, what would be the best way to handle lot of toll free numbers. Do we need to specify the number in the extensions.conf or is there a better way of doing it (using ael etc.) |
| 17:53.23 | CunningPike | Kast: Polycom phones? |
| 17:53.34 | asterisk-newbie1 | hmm |
| 17:54.26 | syzygyBSD | if I dont' have a zaptel card, but want /dev/zap/pseudo, how do I have to configure /etc/zaptel.conf? |
| 17:54.38 | asterisk-newbie1 | file: agi probably? |
| 17:54.41 | file | Qwell[]: pattern matching context names would hurt my head slightly |
| 17:54.42 | dandan | anyone doing different ringtones with gxp-2000? |
| 17:55.00 | file | asterisk-newbie1: maybe, give it a try... |
| 17:55.16 | Qwell[] | file: indeed.. |
| 17:55.18 | *** join/#asterisk BZBW (i=BZBW@ip67-153-142-109.z142-153-67.customer.algx.net) |
| 17:55.40 | asterisk-newbie1 | BTW, is there a realtime music on hold? :) |
| 17:55.56 | tamp4x | would anyone know what would cause callerid error? im using fxs signalling |
| 17:56.06 | BZBW | anyone knows if I can do "exten => _62XX,hint,SIP/${EXTEN}", I have about 120 extensions, and hate to put all the hints in extensions.conf:) |
| 17:56.12 | tamp4x | or do i have stop asterisk to see the logger debug messages? |
| 17:56.31 | CunningPike | BZBW: You have no idea how much we'd love to do the same thing. I use an included file instad |
| 17:56.34 | file | asterisk-newbie1: what you see is what there is, if it's not there - it probably doesn't exist for you to get |
| 17:57.17 | CunningPike | s/instad/instead/ |
| 17:57.33 | BZBW | CunningPike: It's just odd that we can not do this, and instead, you have to use a include file? |
| 17:57.56 | asterisk-newbie1 | file: ok file, thanks. We will live with hardcoding the contexts for the time being. I know its pain. |
| 17:58.00 | file | asterisk-newbie1: you could also write an external script to make the contexts and switches... and use #exec to do it... but it wouldn't be truly realtime then |
| 17:58.08 | CunningPike | BZBW: Yes - it would make sense to do it that way |
| 17:59.17 | asterisk-newbie1 | file: are globals $exten and $context valid in musiconhold.conf? I was curious? |
| 17:59.37 | file | asterisk-newbie1: wha? no |
| 17:59.52 | *** part/#asterisk Loceur (n=noneya@vsas.veedix.com) |
| 18:00.26 | BZBW | CunningPike: Emm..., this puzzles me, as I can do most other stuff similar way, so why not make this works as well:( |
| 18:00.36 | asterisk-newbie1 | file: is there a way that I can send variables to musiconhold. 'application' in musiconhold needs that |
| 18:00.38 | CunningPike | BZBW: You can! Submit a patch :D |
| 18:00.53 | *** join/#asterisk [hC] (n=hardcore@S01060004e21ea953.vc.shawcable.net) |
| 18:01.18 | file | asterisk-newbie1: send variables? what are you talking about? |
| 18:02.23 | BZBW | CunningPike: Ha, this is a bug(I think). Anyway, can you show me how you do with include file? |
| 18:02.39 | *** join/#asterisk tempest1 (n=asf@adsl-144-58-113.chs.bellsouth.net) |
| 18:02.52 | file | my screen is all red from people saying "file" |
| 18:03.31 | [TK]D-Fender | file FILE FILE file FILE FILE file FILE FILE file FILE FILE file FILE FILE file FILE FILE file FILE FILE file FILE FILE file FILE FILE |
| 18:03.37 | CunningPike | BZBW: I simply have a file that contains exten => 1234,hint,SIP/1234 (one line for each extension) - the only reason for the include file is to keep the clutter out of my extensions.conf |
| 18:03.51 | file | yay red |
| 18:04.00 | asterisk-newbie1 | I am trying to invoke application (bin) in musiconhold instead of giving it as a file. Depending on the context and extension, I would want the bin file to change the file type depending on params to it. I would love to give context and extension as param to this bin file. Hence this question |
| 18:04.19 | tamp4x | anyone know how to reload changed in logger.conf with out shutting downa sterisk? |
| 18:04.29 | file | tamp4x: logger reload |
| 18:04.52 | tamp4x | thanks |
| 18:04.59 | file | asterisk-newbie1: ummm res_musiconhold just loads musiconhold.conf when it starts or when it's told to and parses out the given data... you don't give it anything |
| 18:05.12 | BZBW | CunningPike: thx. BTW, what's command in extensions.conf for include file? |
| 18:06.39 | tamp4x | file, i allowed debug settigngs for the console but i do not see anything of why i get this error: |
| 18:06.39 | tamp4x | <PROTECTED> |
| 18:06.44 | asterisk-newbie1 | file: will have to change the source code then. I looked at the source, this was a small change (not adding globals support) but to return a new class (with new filepath) when class does not exist |
| 18:06.48 | tamp4x | do you ahve any idea where i should look |
| 18:06.51 | tamp4x | have |
| 18:06.55 | file | tamp4x: that's not an error, that's dialplan logic |
| 18:06.57 | Bullseye_Network | #include "included.file" |
| 18:07.17 | *** join/#asterisk Seravat (i=Luft@bl5-45-94.dsl.telepac.pt) |
| 18:07.23 | Seravat | hello Lokesh!! |
| 18:07.28 | _problem_ | hello |
| 18:07.31 | CunningPike | BZBW: #include <relative path to file> |
| 18:07.39 | tamp4x | oops, sorry file, wrong line copied |
| 18:07.43 | file | must be a reason.... to hold on to what we've got! |
| 18:07.46 | BZBW | CunningPike: Thx! |
| 18:07.54 | CunningPike | BZBW: np |
| 18:08.17 | Seravat | good to be back in mIRC |
| 18:08.26 | tamp4x | file: ss_thread: CallerID returned with error on channel 'Zap/2-1 |
| 18:08.27 | _problem_ | Seravat:ya |
| 18:08.36 | WeirdM | Would anyone mind taking a look at the errors I am getting running make on zaptel? http://pastebin.ca/101081 |
| 18:08.44 | file | tamp4x: I would think callerid returned with error, but then again I'm not a zaptel person - sorry! |
| 18:09.03 | tamp4x | o i should just call digium |
| 18:09.16 | syzygyBSD | WeirdM: what distro/kernel you using? |
| 18:09.26 | syzygyBSD | Fedora core 5 and 2.6.15? |
| 18:09.31 | file | might not have the right method set for callerid, can depend on country |
| 18:09.49 | WeirdM | 5 and 2.6.15-1 |
| 18:09.50 | tamp4x | its an intergrated t1 |
| 18:09.58 | WeirdM | 2.6.15-1.2054_FC5 |
| 18:10.06 | file | an "integrated t1"? that's a new one |
| 18:10.12 | tamp4x | upgrade the kernel |
| 18:10.21 | tamp4x | file: voice/data t1 |
| 18:10.22 | salviadud | how do you upgrade the kernel in fedora core |
| 18:10.24 | WeirdM | k |
| 18:10.24 | salviadud | ? |
| 18:10.27 | WeirdM | I'll give that a try |
| 18:10.39 | salviadud | i know how to do it in slackware though... |
| 18:10.42 | tamp4x | yum install kernel |
| 18:10.52 | salviadud | that's not even fun |
| 18:11.04 | salviadud | i wouldn't use FC... |
| 18:11.17 | WeirdM | I installed the kernel dev package earlier cause it was complaining that the source wasn't installed |
| 18:11.24 | tamp4x | yea fc5 is not very fin |
| 18:11.28 | tamp4x | fun |
| 18:11.40 | salviadud | down here in mexico |
| 18:11.42 | salviadud | we call it |
| 18:11.45 | WeirdM | I figured it would install the latest stuff but I just ran yum update and it looks like it is updating the dev package |
| 18:11.49 | salviadud | Pedorra Core |
| 18:12.15 | salviadud | and it translates to Farting Core |
| 18:12.16 | syzygyBSD | WeirdM: they custom compiled that kernel with "fixes" for usb that wern't released until .16 |
| 18:12.27 | syzygyBSD | upgrade your kernel at least one version and it will work |
| 18:12.56 | syzygyBSD | or you can do what I did and grep the code for 2.6.16 and replace with 2.6.15 |
| 18:13.01 | syzygyBSD | then recompile |
| 18:13.56 | WeirdM | ok, i'll upgrade the kernel |
| 18:14.04 | syzygyBSD | I was about to submit my patch but checked the kernel patch noted on kernel.org to make sure when that fix was patched into the kernel tree, it was .16 |
| 18:14.20 | WeirdM | cool, thanks for the info |
| 18:16.23 | jbalcomb | The volume when I dial an extension directly is perfect but when I call its DID number the volume is about half what it should be. What gives? |
| 18:16.30 | syzygyBSD | I should really post that to the list so someone can search for it.... |
| 18:16.50 | *** join/#asterisk [Airwolf] (n=airwolf@dsl5402DE1C.pool.t-online.hu) |
| 18:17.00 | syzygyBSD | I have just never gotten around to subscribing to the list |
| 18:18.07 | jbalcomb | Is anyone else here running QueueMetrics? If so, what distro are you running it on and do you have notes on installing it properly? |
| 18:18.42 | file | jbalcomb: this DID number is delivered over what? |
| 18:19.23 | jbalcomb | I'm thinking about setting up a XenU running Gentoo or Ubuntu because they seem to have package support for Tomcat and Sun Java. |
| 18:19.30 | jbalcomb | file: PRI |
| 18:20.00 | file | exciting |
| 18:20.20 | jbalcomb | file: as I understand it the call would be going out my Asterisk server to the telco, the telco would send it back, and my Asterisk server would route it to the queue. |
| 18:20.26 | *** part/#asterisk Seravat (i=Luft@bl5-45-94.dsl.telepac.pt) |
| 18:21.21 | jbalcomb | file: I don't know if that would be attenuation issues or even Asterisk or the telco adjusting the gain. :/ |
| 18:21.34 | file | Asterisk won't touch something like that |
| 18:21.44 | jbalcomb | file: what about the txgain and rxgain settings? |
| 18:21.48 | file | unless it's told to |
| 18:22.34 | jbalcomb | file: I don't think the telco would do anything either and attentuation seems pretty weak given that call originating outside work fine. |
| 18:23.12 | jbalcomb | file: I supposed the PRI card might be involved but that segment of operation is a knowledge void for me |
| 18:23.25 | asterisk-newbie1 | qwell: Yesterday, you have suggested to use a Macro to achieve the call connection only after callee has entered #. I looked at the macro but unfortunately, the macro cannot take any input from the callee, it can take only from the caller |
| 18:24.24 | jbalcomb | asterisk-newbie1: there are several pages that cover having a callee press # to reject or accept calls. |
| 18:24.51 | jbalcomb | asterisk-newbie1: The coverage often seems related to screening telemarketers. |
| 18:25.14 | asterisk-newbie1 | jbalcomb: I would be glad if you can provide a link or tell me where I can find that |
| 18:25.58 | asterisk-newbie1 | I am trying to dial a cell phone for one extension and user often complains that it is taking the callers to his carrier vm. So, we thought of coming up with an idea to connect the call only if you dial any digit |
| 18:28.38 | jbalcomb | asterisk-newbie1: google.com or voip-wiki.com would be as specific as I could direct you. |
| 18:29.41 | Mercestes | asterisk-newbie1: The problem your going to run into is the cell phone carrier is going to immediately provide ring back whether the cellphone is ringing or not. |
| 18:29.58 | hmmhesays | yeaahhaw |
| 18:30.05 | Mercestes | asterisk-newbie1: So the caller could here from 4-8 rings regardless of whether the cell phone picks up on four rings or not. |
| 18:30.06 | DaveHope | Anyone ever come across "SIP/2.0 484 Address Incomplete " with outgoing SIP calls ? |
| 18:30.37 | Mercestes | asterisk-newbie1: so what you have is an unknown amount of time before it goes to the cell phone voicemail. |
| 18:30.44 | asterisk-newbie1 | jbalcomb: I might have gone through almost everything on this. There is an option like c/... but it works only with zaptel |
| 18:30.56 | Mercestes | asterisk-newbie1: Also if hte cellphone shows offline or cannot be found at all, the cell phone will go straight to voicemail which Asterisk will see as an answer. |
| 18:31.09 | asterisk-newbie1 | Mercestes: correct. If it is roaming and people are using like callshield on their cells, it goes to vm immediately |
| 18:31.12 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 18:31.50 | file | asterisk-newbie1: you want to have the person have to hit a key in order to accept the call? |
| 18:31.58 | asterisk-newbie1 | file: yes |
| 18:34.06 | Mercestes | asterisk-newbie1: Well, your going to have to accept the incomming call and place it on hold...then ring the device, prompt for a key, and upon recieving that key, bridge the incoming call to the outgoing number.. |
| 18:34.49 | Mercestes | asterisk-newbie1: with a time out of course because if it goes to Vm..you won't recieve that key. not even sure what that dialplan will look like. |
| 18:35.14 | hmmhesays | man I am just having no luck with spandsp at all |
| 18:35.37 | hmmhesays | and rxfax and txfax |
| 18:35.51 | asterisk-newbie1 | Mercestes, how do I bridge the incoming call to outgoing? Should I use some meetme? |
| 18:36.15 | viperdude | hi guys |
| 18:37.05 | Mercestes | asterisk-newbie1: possible. I don't know if there is a transfer executable for extensions.conf. |
| 18:37.09 | viperdude | has anyone every managed to connect a Sony Ericsson F250m to asterisk? |
| 18:37.53 | *** join/#asterisk Ebola (n=Ebola@81-86-155-65.dsl.pipex.com) |
| 18:37.54 | Mercestes | asterisk-newbie1: Could use a call park as well and call a cell phone with a call park extension. might be a quick and dirty way. Kicking them back out is the issue there. You'd have to use a timeout extension that says "sorry, not available" *click* |
| 18:38.26 | Mercestes | asterisk-newbie1: If your coding savvy at all, operator flash panel has code that makes one phone call another phone and you can use similar code to do what yoru trying to do. |
| 18:39.15 | Mercestes | asterisk-newbie1: Or you can write an AGi that you can call into that stores the caller ID and hangs up on the callee..takes the DNID and calls the cell phone, prompts for a # and upon recieving that #, calls the caller back. |
| 18:39.19 | asterisk-newbie1 | I am *very* code savvy :) but totally new to asterisk. I was thinking about call aprt |
| 18:39.40 | asterisk-newbie1 | Mercestes: thats not going to work - calling back |
| 18:39.54 | asterisk-newbie1 | Mercestes: though the easiest way |
| 18:39.56 | Mercestes | asterisk-newbie1: Just brainstorming..:) |
| 18:40.06 | asterisk-newbie1 | :) |
| 18:40.28 | asterisk-newbie1 | Mercestes: call a cell phone with a call park extension? |
| 18:40.32 | Mercestes | asterisk-newbie1: A meet me might work. Just a 2 person meetme. |
| 18:40.54 | asterisk-newbie1 | Mercestes: Whats up with this call park extension? |
| 18:41.04 | Mercestes | asterisk-newbie1: Yea, use hte parkandannounce code on the wiki and call a cellphone with it. Problem is, you'll get voicemails that say "701" all the time. |
| 18:41.36 | Mercestes | asterisk-newbie1: call parking basically parks a call, and delivers an extension, usually 701 for the first caller..then someone can pick that call up with 701. |
| 18:42.00 | Mercestes | asterisk-newbie1: you *may* once you establish a voice path, grab that 701 from a cell phoen but I'm not certain...I've never tried it. |
| 18:42.30 | Mercestes | asterisk-newbie1: once asterisk hears "701" or whatever the parked extension is...it should deliver the call. |
| 18:42.36 | *** join/#asterisk Assid (i=assid@203.115.83.215) |
| 18:43.02 | *** join/#asterisk AuPix (n=AuPix@adsl-04-85.abel.net.uk) |
| 18:43.05 | Assid | heya |
| 18:43.30 | Assid | if if i wanna use realtime.. but i want it to use postgresql.. do i have to use the likes of ODBC ? or can i use a pgsql driver? |
| 18:43.35 | asterisk-newbie1 | Mercestes: Let me check parkandannounce. This is interesting |
| 18:45.01 | tamp4x | when i dial a parked call, i get address incomplete... but yet i have everythign definte in features.conf....any reason why? |
| 18:45.44 | tamp4x | defined |
| 18:45.53 | *** join/#asterisk Chotaire (i=chotaire@chotaire.net) |
| 18:46.03 | asterisk-newbie1 | Mercestes: Checked parkandannounce but it is limited to one per box except we go for 1.3 which will not fully tested and may not be stable (I believe) |
| 18:46.47 | Mercestes | asterisk-newbie1: You should be able to handle around 99 parked calls...that's why it starts at 701 and goes up. |
| 18:47.57 | *** join/#asterisk smackus (n=ckwall@63.149.122.93) |
| 18:48.10 | *** join/#asterisk Dovid (n=dovi5988@pool-71-250-66-243.nwrknj.east.verizon.net) |
| 18:48.32 | smackus | question... is it possible to change the ring tone of a polycom soundpoint ip phone? if so, does anyone know where to do that? |
| 18:48.34 | asterisk-newbie1 | Mercestes: "In the current Asterisk version, there is one parking lot per PBX. This leads to problems when you have multiple user groups in your PBX. With this patch, you can enable multiple parking lots, in order to" |
| 18:49.02 | Mercestes | asterisk-newbie1: Oh, that...yea, that's true.... |
| 18:49.12 | Dovid | I have a problem with NAT |
| 18:49.17 | Mercestes | asterisk-newbie1: Don't tell me your using one monster PBX instead of multiple on site PBX's... |
| 18:49.27 | Dovid | i have nat=yes and for some reason I dont hear the voice at all |
| 18:49.32 | Mercestes | Dovid: Nat is the devil..what's your problem? |
| 18:49.33 | Dovid | the call comes in but nothing happens |
| 18:49.42 | Mercestes | Dovid: Firewall/port issue. |
| 18:49.49 | *** join/#asterisk sky1234 (n=sky1234@12.44.122.130) |
| 18:49.49 | Dovid | I have 5000-6000 and 10000-20000 forwarding |
| 18:49.59 | Dovid | for both TCP and UDP |
| 18:50.01 | CunningPike | smackus: You can do it right on the phone |
| 18:50.04 | Mercestes | Dovid: I think the sip ports go on through 30k |
| 18:50.13 | CunningPike | smackus: You can also do it via the config files |
| 18:50.24 | CunningPike | Is there a feature list for 1.4? |
| 18:50.27 | Dovid | nah |
| 18:50.30 | Dovid | 10-20 |
| 18:50.30 | Mercestes | asterisk-newbie1: Asterisk is a PBX, not a phone switch. That's why it's called the open source PBX..:P |
| 18:50.41 | smackus | so could It be done for just one particular sip.cfg file then? |
| 18:50.42 | Dovid | but can be used as a switch |
| 18:50.50 | Mercestes | Dovid: So can hubs..:P |
| 18:51.00 | smackus | so i did not have to do it per phone, but so i did not affect all phones? |
| 18:51.08 | *** join/#asterisk paolob (n=donpaolo@pri-214-b7.codetel.net.do) |
| 18:51.11 | asterisk-newbie1 | Mercestes: It is not a monster PBX :) but we have around 100 clients per box |
| 18:51.13 | *** part/#asterisk paolob (n=donpaolo@pri-214-b7.codetel.net.do) |
| 18:51.17 | Dovid | haha |
| 18:51.19 | sky1234 | hello. I am acutally working on some automation related items.. using kickstart. anyone aware that after the rw_lock change, why the modules for zaptel wouldnt get loaded or built...it seems to do the rest though...ie...properly moves stuff into init.d, and does the chkconfig...but no good on the modules.. any ideas?? |
| 18:51.26 | [TK]D-Fender | Anyone here know the default root password for CentOS 4.3? |
| 18:51.31 | Dovid | all ports (5060 as well as RTP) are UDP only ? |
| 18:51.41 | CunningPike | smackus: Mutually exclusive - you either set it for all phones in sip.cfg, or per phone in <macaddress.-phone.cfg |
| 18:51.52 | Dovid | TK: default root for a@H or just plain simple root ? |
| 18:52.20 | Dovid | tk: when ever i install centos it forces me to put in a pass for the root |
| 18:53.27 | CunningPike | [TK]D-Fender: You would have supplied one at install time |
| 18:53.35 | [TK]D-Fender | Dovid : CentOS general, not A@H. Plain root |
| 18:53.51 | [TK]D-Fender | CunningPike : Actually... I don't recall it asking... jsut for a USER. |
| 18:54.23 | CunningPike | [TK]D-Fender: Weird - it always asks me for a root password....... |
| 18:54.36 | [TK]D-Fender | NVM... brain like a siv today.... Just remembered |
| 18:55.28 | CunningPike | Anyway, is there a feature list for 1.4? |
| 18:56.13 | [TK]D-Fender | CunningPike : Stunning idea... |
| 18:56.25 | CunningPike | [TK]D-Fender: rofl |
| 18:56.45 | Dovid | what does this mean ? |
| 18:56.49 | Dovid | Jul 26 14:57:06 NOTICE[9908]: chan_sip.c:11244 handle_request: Unknown SIP command 'PUBLISH' from '68.192.81.173' |
| 18:57.16 | eKo1 | It means what it means. |
| 18:57.19 | file | chan_sip doesn't support the SIP PUBLISH method |
| 18:57.29 | file | so it's all like, "eep I don't understand you!" |
| 18:57.35 | Dovid | that i got |
| 18:57.36 | *** join/#asterisk Maroder (n=drago@fanatici.net) |
| 18:57.46 | Dovid | but what is the phone tryin to send that it dosent understand |
| 18:57.52 | Dovid | and will this affect my phone usage ? |
| 18:57.56 | Maroder | hello |
| 18:58.21 | Maroder | can anybody help me with asterisk and gnugk |
| 18:59.18 | asterisk-newbie1 | Merc: parkandannounce looks interesting. Should I park the present caller to some extension (can I park to some random extension or should I park the call and it gives me some random extension for me to connect later)? |
| 18:59.49 | *** join/#asterisk Vulture- (n=Vulture@64.238.204.68.cfl.res.rr.com) |
| 18:59.59 | *** join/#asterisk Bobcat_1966 (n=chatzill@cpe-069-132-139-254.carolina.res.rr.com) |
| 19:00.03 | Vulture- | Anyone here using Asterlink or VoicePulse Connect? |
| 19:00.19 | *** join/#asterisk wunderkin (n=wunderki@216-19-202-13.getnet.net) |
| 19:01.25 | *** join/#asterisk javar (n=javar@Dynamic-IP-cr20011876227.cable.net.co) |
| 19:01.36 | dandan | anyone doing different ringtones with gxp-2000? |
| 19:04.55 | Assid | <PROTECTED> |
| 19:07.06 | [TK]D-Fender | exit |
| 19:07.31 | Assid | err.. which package does it come with |
| 19:07.42 | Mercestes | Assid: Asterisk-addons |
| 19:07.51 | Assid | hrmm.. thats what i installled |
| 19:08.16 | Mercestes | Assid: I think, if you used the cvs version, that the makefile checks for DB socks like mysql.sock...if it doesn't see it then it doesn't install the stuff for it. |
| 19:08.31 | Mercestes | Assid: Might wanna check the permissions on it too in case it does exist and is owned by root or something stupid. |
| 19:09.05 | *** part/#asterisk _alex_mx_ (n=_alex_mx@200.78.229.18) |
| 19:09.32 | *** part/#asterisk asteriskmonkey (n=phil@h216-235-8-130.host.egate.net) |
| 19:09.34 | Assid | err. trying to use odbc |
| 19:10.44 | Assid | Vulture-: yes me |
| 19:12.58 | *** join/#asterisk Ebola (n=Ebola@81-86-155-65.dsl.pipex.com) |
| 19:14.57 | Maroder | eoo |
| 19:15.00 | *** part/#asterisk javar (n=javar@Dynamic-IP-cr20011876227.cable.net.co) |
| 19:15.11 | Maroder | i want to link gnugk with asterisk |
| 19:15.23 | *** join/#asterisk Meaty (n=meaty3@office.abi.ca) |
| 19:15.38 | Maroder | where i can read about asteris+gnugk |
| 19:17.12 | *** join/#asterisk clive- (n=pirch@dsl-145-52-134.telkomadsl.co.za) |
| 19:24.55 | clive- | has anyone had any wierd (ie.very slow) zaptel timming hapenning on a dual xeon? |
| 19:26.20 | *** join/#asterisk MikeJ (n=vircuser@adsl-75-9-243-230.dsl.sfldmi.sbcglobal.net) |
| 19:28.16 | *** join/#asterisk Ebola (n=Ebola@81-86-155-65.dsl.pipex.com) |
| 19:28.20 | Assid | have you loaded the right zaptel device? |
| 19:28.59 | clive- | Assid yup, its says loaded up fine |
| 19:29.09 | mrdigital | can anyone help me connect my pstn line to asterisk via a ata? |
| 19:29.37 | Assid | lsmod |grep zaptel |
| 19:30.28 | Assid | mrdigital: have you setup the sip account correctly? |
| 19:30.49 | Zodiacal | anyone know how i can get my cisco 7960's dialplan.xml to dial users mailboxes immediatly? i.e. *97 dials the mailbox. but theres a timeout that takes a while before it gets sent unless they press #.. |
| 19:30.54 | clive- | zaptel 194180 1 wcfxo |
| 19:30.55 | clive- | crc_ccitt 6081 1 zaptel |
| 19:31.08 | Assid | you have a fxo device? |
| 19:31.12 | brad_mssw | mrdigital: uhh, most likely that won't work, unless you have an ata which supports FXO ... most ATAs are FXS-only (meaning they can only connect to a telephone, not a pstn line from your telco) |
| 19:31.23 | clive- | assid its a x100p that i want to use for timming purposes |
| 19:32.03 | Assid | okay.. if you are planning to use it only for timing.. you could stick to ztdummy instead |
| 19:32.05 | brad_mssw | mrdigital: notably the sipura spa-3000 has both an FXO and FXS port, but not many others do |
| 19:32.13 | Dovid | hi all |
| 19:32.31 | hmmhesays | anyone ever assign two ip addresses to one network interface in linux |
| 19:32.36 | hmmhesays | like you can do with windoze |
| 19:32.37 | Dovid | yes |
| 19:32.40 | clive- | assid, I have had bad expereinces with ztdummy, it loses accuarcy with small load increases |
| 19:32.41 | Assid | oh wait.. i thought he meant normal telephone with ata |
| 19:32.59 | Dovid | hmmhesays: yes u can |
| 19:33.00 | Assid | clive-: does the CLI give you any errors? |
| 19:33.20 | Assid | hmmhesays: virtual eth interface.. ala eth0:0 eth0:1 |
| 19:33.21 | clive- | the CLI is fine, but the sound is garbelled...my zttest is like 34% |
| 19:33.21 | hmmhesays | having a hard time with google figuring out how though |
| 19:33.31 | hmmhesays | ahh |
| 19:33.49 | Dovid | you have to edit a certain file |
| 19:33.51 | Dovid | one sec. |
| 19:34.19 | Assid | hmmhesays: debian? |
| 19:34.39 | Assid | clive-: you tried using wideband codec like ulaw? |
| 19:34.50 | clive- | hmmmhesays go with assid, the eth0:0 is what I use also |
| 19:35.12 | hmmhesays | yeah if I can find the freaking config file in fc5 |
| 19:35.28 | *** join/#asterisk klasstek (n=nunyobiz@ip67-154-143-190.z143-154-67.customer.algx.net) |
| 19:35.38 | Dovid | hmmhesays: i forgot the name of the file that u have to edit |
| 19:35.38 | clive- | assid, the calls sound fine when bridged, but the IVR sounds terrible, which is what zaptel timming seems to do |
| 19:36.29 | hmmhesays | in every other linux /etc/network/interfaces |
| 19:36.51 | clive- | my plan of action is to swap the x100p for another one, and see if that works, or maybe swap pci slots....just so strange |
| 19:37.04 | Assid | clive-: right.. too slow ? |
| 19:37.07 | eKo1 | hmmhesays: fc? |
| 19:37.35 | clive- | assid, way too slow, it shojuld be about 99.9% , so 34% is not doing it |
| 19:37.55 | Assid | you tried working with zapata / zaptel |
| 19:38.11 | hmmhesays | fc5 eKo1 |
| 19:38.31 | clive- | assid, what do you suggest one could do in zapata to make things better ? |
| 19:38.43 | eKo1 | hmmhesays: cd /etc/sysconfig/network-scripts |
| 19:38.43 | Dovid | hmmhesays: /etc/sysconfig/network-scripts/ifcfg-eth0 |
| 19:39.04 | eKo1 | cp ifcfg-eth0 ifcfg-eth0:0 |
| 19:39.12 | eKo1 | and edit appropriately |
| 19:40.12 | Assid | clive-: frankly i dont know.. it was just a suggestion |
| 19:41.27 | clive- | assid, thanks for your help, I am going to try swapping cards/slots tomorrow and see if that helps...its a wacky one |
| 19:43.07 | Assid | yeah |
| 19:43.20 | Assid | actualy.. i think there is something with the kernel to |
| 19:43.23 | Assid | pre-empt |
| 19:43.28 | *** join/#asterisk Ebola (n=Ebola@81-86-155-65.dsl.pipex.com) |
| 19:43.32 | Assid | you could try somehing with that |
| 19:44.04 | clive- | assid now you have gone past my knowledge...:), what is pre-emt ? |
| 19:46.35 | *** join/#asterisk Scrye (n=ryan@2001:470:1f00:2514:280:c8ff:fec9:96d8) |
| 19:47.27 | Assid | its part of the kernel |
| 19:47.39 | Assid | did you install your kernel from source? |
| 19:48.07 | clive- | nope, its a centos 4.3 kernel ...2.6.9.ELsomthingsmp |
| 19:49.22 | Assid | okay then i dunno |
| 19:49.23 | Assid | try installing the kernel from suorce. and check the pre-empt |
| 19:49.32 | Assid | there is somethign that mentiones the Hz.. set it to 1000 |
| 19:50.32 | clive- | that may be an option....a whole new kernel ove version 2.6.14 I belive does much better timming on ztdummy |
| 19:50.52 | Assid | well try that |
| 19:50.55 | clive- | just nervous to install a new kernel from scratch |
| 19:52.54 | Assid | damn.. anyone got the the dump for sip.conf config for postgresql realtime? |
| 19:53.06 | Corydon-w | As long as you don't install over the old kernel, you should be fine |
| 19:53.25 | Assid | yeah. just name it as something else.. and grub your way in |
| 19:53.34 | clive- | corydon Thanks, do you know af a good "howto" somewhere |
| 19:54.22 | *** join/#asterisk TripleFFFF (n=TripleFF@147-102.mc.cite.net) |
| 19:54.25 | TripleFFFF | ~centos_bug |
| 19:54.30 | TripleFFFF | ~centos |
| 19:54.32 | jbot | rumour has it, centos is better than Fedora Core except for that silly bug, see ~centosbug for details |
| 19:54.36 | Corydon-w | I generally just install kernels that are packaged with my distro. Then I know that the configuration was previously tested and I have to spend literally no time at all messing with it |
| 19:54.51 | TripleFFFF | ~centosbug |
| 19:54.52 | jbot | methinks centosbug is a problem with the latest Centos kernels (4.2 and 4.3). To fix it, paste everything inside the quotes into a root shell: "sed -i s/rw_lock/rwlock/ /usr/src/kernels/`uname -r`-`uname -m`/include/linux/spinlock.h" |
| 19:56.38 | clive- | corydon yup, thats why I havent attempted to install a new kernel, I belive the stock centos kernel should be more stable |
| 19:57.38 | TripleFFFF | hmmmhmm patch does work cool finaly somthing that works |
| 19:57.42 | *** join/#asterisk rogier (n=rogier@83-67-dsl.ipact.nl) |
| 19:57.43 | TripleFFFF | any vicidual users out her e? |
| 20:05.39 | Mw3 | do you know about some kind of push to talk over voip service? |
| 20:07.06 | watchy | i use gentoo for asterisk boxes, is it a good choice? |
| 20:07.38 | rob0 | Gentoo fans will say definitely so. :) |
| 20:07.50 | Dovid | what does the canreinvite function do ? |
| 20:08.07 | Dovid | for some reason when its set to yes I cant hear the other party |
| 20:08.58 | eKo1 | watchy: any distro will do |
| 20:09.02 | TripleFFFF | invite to a part |
| 20:09.04 | TripleFFFF | y |
| 20:09.15 | clive- | dovid, it tries to move asterisk out the media path....doesnt work so well, esp with NAT |
| 20:09.36 | *** join/#asterisk SimoAmi (n=SimoAmi@cpe-24-29-135-83.nyc.res.rr.com) |
| 20:10.21 | Dovid | ah ok |
| 20:10.26 | Dovid | cause it worked fine locally |
| 20:10.42 | watchy | im having issues with echo, could it be because the machine is a fucking emachine? |
| 20:10.50 | Dovid | but when i called in from a voip line it didnt work |
| 20:13.43 | rob0 | The E in Emachine stands for echo. |
| 20:14.34 | *** join/#asterisk tsurk0 (n=tsurko@85.187.160.157) |
| 20:16.18 | *** join/#asterisk dasenjo_ (n=dasenjo@208.195.215.158) |
| 20:17.55 | Dovid | what does thie mean ? |
| 20:17.58 | Dovid | Incoming call: Got SIP response 500 "Internal Server Error" back from 10.10.10.107 |
| 20:18.14 | *** part/#asterisk tamp4x (n=tampon@63.138.1.178) |
| 20:18.32 | [TK]D-Fender | Dovid : Just a small SIP comm error with Polycom's... nothing to get worked up over |
| 20:18.43 | Dovid | thnks |
| 20:19.04 | Dovid | thought it was nothing but the person i am doing it for is anal and he saw it and he wanted to know what it is |
| 20:19.46 | *** join/#asterisk florz (i=nobody@2001:1a50:503c:0:0:0:0:1) |
| 20:26.28 | *** join/#asterisk mitcheloc (n=mitchelo@020.131-62-69.ftth.swbr.surewest.net) |
| 20:28.36 | *** join/#asterisk ast_freak (n=ast_frea@68-112-130-237.dhcp.stcd.mn.charter.com) |
| 20:30.38 | pdt | Dovid: I see that sometimes when the polycoms get screwed up on the buddy list stuff |
| 20:30.44 | *** join/#asterisk santiago (i=santiago@debian/developer/santiago) |
| 20:30.57 | *** join/#asterisk flujan (n=flujan@internet.nube.com.br) |
| 20:31.20 | *** join/#asterisk WeirdM (n=weirdm@udp079073uds.hawaiiantel.net) |
| 20:32.22 | flujan | guys, i am having problem in compiling mpg123 to work with asterisk and slackware-current |
| 20:32.43 | *** join/#asterisk tempest1 (n=asf@adsl-0-232-223.chs.bellsouth.net) |
| 20:34.08 | flujan | someone have mpg123 working with slackware? |
| 20:35.46 | *** join/#asterisk likwid-- (i=likwid@c-24-99-14-123.hsd1.ga.comcast.net) |
| 20:36.08 | *** part/#asterisk likwid-- (i=likwid@c-24-99-14-123.hsd1.ga.comcast.net) |
| 20:36.24 | Dovid | pdt: thanks |
| 20:36.30 | file | flujan: what problem shows up? |
| 20:36.40 | Dovid | pdt: the buddies list acts up after too many asterisk reloads |
| 20:36.51 | Dovid | file: the 500 error from the polycoms |
| 20:37.00 | file | Dovid: wasn't talking to you ;) |
| 20:37.10 | *** part/#asterisk TripleFFFF (n=TripleFF@147-102.mc.cite.net) |
| 20:37.43 | *** join/#asterisk russellb_ (n=russell@asterisk/developer-and-stable-maintainer/drumkilla) |
| 20:37.43 | *** mode/#asterisk [+o russellb_] by ChanServ |
| 20:38.04 | flujan | file, http://pastebin.ca/101366 |
| 20:38.08 | file | russellb_: back for good? |
| 20:38.17 | russellb_ | unknown |
| 20:38.20 | russellb_ | serious storm action going on |
| 20:38.44 | jbalcomb | Is anyone else here running QueueMetrics? If so, what distro are you running it on and do you have notes on installing it properly? |
| 20:38.45 | file | flujan: interesting... |
| 20:38.47 | *** join/#asterisk champster (n=asterisk@AH.tescogroup.com) |
| 20:38.48 | russellb_ | power flickering ... house shaking ... yay summer |
| 20:38.48 | mitcheloc | we could ask god over in #religion to let off the storm for a while ;) |
| 20:39.03 | flujan | file, its anoying... :( |
| 20:39.08 | flujan | file, :) |
| 20:39.53 | file | flujan: what about doing it manually? |
| 20:40.41 | flujan | hum... how can i do it? |
| 20:40.43 | file | that's odd |
| 20:40.44 | flujan | :) |
| 20:40.57 | file | russellb_: still there? |
| 20:41.34 | flujan | file, how can i do it manually? |
| 20:42.38 | file | russellb_ lost power to his house |
| 20:43.41 | jbalcomb | flujan: Have you seen all the mentions of mpg123 not working with ALSA anymore? |
| 20:44.24 | flujan | jbalcomb, no. :( So, how can I make it work with asterisk? |
| 20:45.26 | jbalcomb | flujan: As I understand it you need to use OSS instead of ALSA or use ALSA 0.9 or earlier |
| 20:46.12 | file | sorry, was talking to russellb on the phone! |
| 20:47.37 | flujan | jbalcomb, it is sad that a cool app like asterisk depends of a deprecated program like mpg123 :( |
| 20:47.50 | file | it doesn't anymore |
| 20:48.06 | jbalcomb | flujan: it doesn't. |
| 20:48.26 | nortex | Go native man :) |
| 20:48.34 | flujan | jbalcomb, ??? how can i have music on hold working without it? |
| 20:48.36 | jbalcomb | flujan: can't you just remove mpg123 as a dependency and use madplay instead? |
| 20:49.10 | flujan | i dunno how to use madplay instead... i just need to add a application line to the musiconhold.conf? |
| 20:49.24 | jbalcomb | flujan: default => custom:/asterisk/moh/classical,/usr/bin/madplay -A -6 --downsample --mono -R 8000 --output=raw:- |
| 20:50.17 | jbalcomb | FYI: putting the asterisk directory in / pisses me off |
| 20:50.25 | file | lol |
| 20:50.42 | jbalcomb | its like some dumbass win311 app |
| 20:52.07 | jbalcomb | then again, having moh OR ringing as the options going into the queue, it probably is a win311 app |
| 20:52.09 | flujan | jbalcomb, thanks... i will tri it... :) |
| 20:52.18 | jbalcomb | flujan: np. works for us. |
| 20:52.41 | flujan | jbalcomb, it is working... thanks guys... :) |
| 20:52.48 | jbalcomb | =) |
| 20:53.00 | jbalcomb | i like helping people |
| 20:53.22 | hmmhesays | so I think i bricked this wrt |
| 20:54.00 | jbalcomb | hmmhesays: there is a guy on ebay's want to buy section that'll give you $20 for it |
| 20:56.22 | *** join/#asterisk JaysonS (n=jayson@200.207.205.131) |
| 20:57.15 | [TK]D-Fender | BBIAB |
| 20:58.02 | JaysonS | Hi People, sorry for this question, so can I mount a free gateway for connection with Asterisk - Conventional Telefophony ? |
| 20:58.28 | Assid | if i wanna set something which should be used in CDR within the sip.conf per user/peer/friend conext.. how do i do this? |
| 20:59.45 | smackus | I am trying to set up a custom ring tone on my polycom phones. I have put the wav file in the directory where my sip.cfg file is and have edited the line in sip.cfg that reads: <sampled_audio saf.1="SoundPointIPWelcome.wav" saf.2="" to include: saf.2="CTU.wav" |
| 21:00.00 | smackus | i expected to find CTU.wav in the ring type menu after rebooting, but do not. |
| 21:00.10 | smackus | has anyone done this successfully that can help me out? |
| 21:00.53 | anthm | you can use the cdr_custom thing I made last year with the CDR() varialble |
| 21:02.12 | jbalcomb | bah |
| 21:03.08 | Assid | ? |
| 21:03.21 | WeirdM | Can someone take a look at the errors I am getting doing "make install" for Zaptel? http://pastebin.ca/101384 |
| 21:04.12 | Assid | smackus: <sampled_audio saf.1="SoundPointIPWelcome.wav" saf.2="ctu.wav" saf.3="" saf.4="" saf.5="" saf.6="" saf.7="" saf.8="" saf.9="" saf.10="" saf.11="" saf.12="" saf.13="" saf.14="" saf.15="" saf.16="" saf.17="" saf.18="" saf.19="" saf.20="" saf.21="" saf.22="" saf.23="" saf.24=""/> |
| 21:04.46 | *** join/#asterisk knight_ (n=knight@209.85.11.98) |
| 21:04.56 | knight_ | anyone know where i can pick up a couple Cisco 7960's for cheap? |
| 21:04.59 | knight_ | used is fine |
| 21:05.00 | Assid | and also.. make sure your sox'd ctu.wav for the right format |
| 21:05.07 | Assid | anthm ? |
| 21:05.15 | anthm | yes?/ |
| 21:05.46 | Assid | what did you make last year? |
| 21:06.08 | WeirdM | What did I make? |
| 21:06.19 | Assid | are you anthm? |
| 21:06.22 | WeirdM | No |
| 21:06.36 | anthm | cdr_custom |
| 21:06.43 | anthm | it's still in there isnt it |
| 21:06.50 | Assid | think so |
| 21:07.06 | WeirdM | Anyone have any ideas why Zaptel isn't installing for me? |
| 21:07.13 | WeirdM | http://pastebin.ca/101384 |
| 21:07.22 | Assid | umm.. what do i do if im using realtime? |
| 21:07.37 | Un1x | i wish zaptel dev's would stop being lazy and try and fix the drivers for FreeBSD :) |
| 21:07.40 | Assid | with cdr_odbc |
| 21:08.13 | anthm | the cdr has it's own variables which is probably not exposed in that mod |
| 21:08.35 | anthm | but in the code before you post the cdr you cal get vars from it |
| 21:08.40 | anthm | that you set from the dialplan |
| 21:09.09 | Assid | yeah trying to see if i can set(cdr(accountcode) = ${ACCOUNTCODE}) |
| 21:09.11 | Assid | to do the trick |
| 21:09.31 | Assid | nah |
| 21:09.55 | Assid | didnt work |
| 21:10.03 | anthm | you should be able to set(CDR(anything)=anything) but getting to it is the trick |
| 21:10.28 | anthm | cos none of the cdr mods besides cdr_custom and the one in res_perl can get to them |
| 21:11.34 | anthm | we have a cdr_dumper out of tree mod that writes 1 file to the disk per record in a perl Data::Dumper hash format |
| 21:12.09 | anthm | that one puts the cdr record as a hash with another hash inside of that with all the cdr vars |
| 21:12.10 | Assid | what if you gotta use the likes of cdr_odbc? |
| 21:12.27 | anthm | gotta ? |
| 21:12.49 | Assid | shouldnt it simply match the variable with the column name.. and just update/insert that as needed? |
| 21:13.04 | Assid | cdr_dumper i guess is file based? |
| 21:13.13 | anthm | its 1 file per record |
| 21:13.57 | Assid | oh |
| 21:14.12 | Assid | thats not good |
| 21:15.09 | *** join/#asterisk linlin (n=linlin@c-67-184-229-183.hsd1.il.comcast.net) |
| 21:15.17 | anthm | http://www.cluecon.com/asterisk_code/cdr_sample.txt |
| 21:15.56 | smackus | Assid: yep, thats how I have it. |
| 21:16.08 | smackus | aside from the case of CTU.wav |
| 21:16.17 | smackus | instead of ctu.wav |
| 21:16.39 | vader-- | have any of you guys used the directory function in asterisk? |
| 21:16.53 | vader-- | it can't seem to find anything in my directory |
| 21:16.54 | watchy | i just put it in today |
| 21:17.02 | vader-- | watchy did you have to do anything special? |
| 21:17.33 | watchy | put directory(voicemailcontext|extcontext) |
| 21:17.49 | Assid | hehe.. too much info |
| 21:17.55 | anthm | the point is you can aggregate a shared dir from a cluster of boxes into a transaction block of sql stmts and you have way more columns than that hardcoded stuff |
| 21:18.44 | Assid | hrmm true |
| 21:18.47 | anthm | otherwise you are going to have to look at cdr_custom and whatever one you use |
| 21:18.53 | anthm | db one |
| 21:18.58 | anthm | and make a hybrid |
| 21:18.58 | watchy | work for ya vader? |
| 21:19.14 | anthm | where you can specify a sql template that would be expanded with the var names |
| 21:19.27 | anthm | so you could outline the cols |
| 21:19.41 | *** join/#asterisk [TK]D-Fender (n=joe@66.11.164.239) |
| 21:19.50 | Assid | well. i could use what this has.. and make it into an array column |
| 21:19.52 | watchy | i love you fender |
| 21:19.58 | *** join/#asterisk generalhan (i=general_@ip67-90-64-2.z64-90-67.customer.algx.net) |
| 21:20.02 | generalhan | hey guys ... |
| 21:20.33 | generalhan | can some one explain to me the best way to import my .csv CDR into the mysql DB i created ? |
| 21:20.33 | Assid | but would it catch the accountcode ? |
| 21:20.50 | watchy | be my internet wife |
| 21:21.04 | Assid | hhahaha.. [TK]D-Fender a wife! |
| 21:21.11 | Assid | rofl |
| 21:21.23 | anthm | accountcode is already a cdr col isnt it ? |
| 21:21.33 | [TK]D-Fender | watchy : Where are you located? |
| 21:21.40 | watchy | should 6ms of latency be bothersum for asterisk? |
| 21:21.41 | anthm | it's not really a variable it's a member in the cdr struct |
| 21:21.46 | Assid | yes. but it doesnt grab anything i send .. even if i use cdr() |
| 21:21.47 | watchy | arkansas |
| 21:21.59 | Assid | always shows as empty |
| 21:22.03 | anthm | there is some explicit set_account or some crap isnt there |
| 21:22.06 | anthm | it's a 1 way var |
| 21:22.32 | [TK]D-Fender | watchy : Ah yes.. the state famous for the question "if a man & a woman get divorced ..... ar they still brother & sister?" :O |
| 21:22.44 | watchy | haha :( |
| 21:23.04 | [TK]D-Fender | watchy : And with Canada being on top, that'd make YOU my bitch! :D |
| 21:23.13 | Assid | anthm: actually i thinn k its my bad |
| 21:23.15 | generalhan | HAHAHAHAHAHAHAHAHA |
| 21:23.34 | [TK]D-Fender | <- teh winnar |
| 21:24.50 | [TK]D-Fender | anthm : What the approximate "ready for public consumtion" date for FreeSWITCH looking like lately? |
| 21:25.03 | watchy | im moving to canada so we can be together |
| 21:25.08 | anthm | from the looks of things Set(CDR(accountcode)=blah) would work, or at least it did when i wrote it |
| 21:25.34 | generalhan | [TK]D-Fender: i just started up the mysql for the cdr, but now i have a huge .csv file that has all my data in it .. whats the easiest way for me to get that data to the new DB ? |
| 21:25.42 | anthm | *shrug* september with any luck |
| 21:25.46 | [TK]D-Fender | watchy : Seen "Total Recall"? |
| 21:25.50 | watchy | yes |
| 21:26.09 | generalhan | best movie ever ! lol |
| 21:26.14 | [TK]D-Fender | watchy : "Condsider this a divorce *BANG*" |
| 21:26.18 | generalhan | lol |
| 21:26.43 | [TK]D-Fender | Ah-nuld! Ah-nuld! Ah-nuld! |
| 21:27.02 | generalhan | [TK]D-Fender: WHAT are you on today ? |
| 21:27.08 | [TK]D-Fender | Mein Gropen-furher! |
| 21:27.09 | generalhan | you are in a strange mood ! |
| 21:27.17 | file | [TK]D-Fender: o m g... you rock |
| 21:27.22 | Assid | anthm: yeah.. apparently i set the accountcode of the sip account to 1.. which never got registered |
| 21:27.35 | *** join/#asterisk AsterLost (n=Owner@69-12-144-146.dsl.static.sonic.net) |
| 21:27.38 | Assid | therefore the accountcode always showed as empty |
| 21:27.56 | [TK]D-Fender | generalhan : I'm in a "give even better than I take, easily due to UNDERQUALIFACTION" mood :) its a good day indeed. |
| 21:28.18 | generalhan | lol |
| 21:28.22 | AsterLost | hello |
| 21:28.50 | generalhan | i cant remember the last "good day" ive had ... i need a new job ! lol |
| 21:29.29 | AsterLost | i'm having 1-way audio issues when connecting a SIP phone from outside firewall |
| 21:29.42 | watchy | tk: i'm gonna email polycom to see if there is a way to make a 2nd incoming call ring instead of call waiting beep, i know your god of polys because youve help me alot but i gotta find this out :( |
| 21:29.57 | Assid | damn: ERROR: column "accountcode" cannot be cast to type "pg_catalog.int8" |
| 21:30.24 | AsterLost | i have forwarded tons of ports to the * server... still 1-way audio... anyone have a suggestion? thx |
| 21:30.34 | anthm | isnt accountcode a string ? |
| 21:30.51 | Assid | it is.. i want to make it an int |
| 21:33.11 | *** join/#asterisk CANO-1982 (i=alejandr@190.48.72.107) |
| 21:33.14 | [TK]D-Fender | watchy : Well I told you you can make it ring on the current audio device, but not forcibly the speaker AFAIK |
| 21:33.50 | watchy | tk: yea the ring would on the current audio would just be a RING sound sent to the callwaiting function instead of the beep correct? |
| 21:33.54 | [TK]D-Fender | watchy : And by that you can also up the volume. Your problem CAN also be solved by sending that deaf bitch to a ENT M.D. |
| 21:34.08 | watchy | tk: hahaha |
| 21:34.16 | [TK]D-Fender | watchy : Correct |
| 21:34.31 | Assid | okay im outta here.. need sleep |
| 21:34.48 | watchy | 6ms latency shouldn't be a big issue for * should it tk? |
| 21:34.52 | Assid | gnight guys |
| 21:35.14 | watchy | I wonder if its causing my echo problems im having |
| 21:35.15 | [TK]D-Fender | watchy : For RAM... HIDEOUS! |
| 21:35.42 | watchy | ram? no no, I run a wireless ISP and their phone system is at their maybe building |
| 21:36.00 | [TK]D-Fender | watchy : Audio engineering says it takes 10ms to be audible. |
| 21:36.06 | watchy | I have connected them together using 5.8ghz 3mbit radios and the latency is 6ms |
| 21:36.29 | MikeJ | watchy, what;s the cost on those.. and range? |
| 21:36.40 | watchy | range like 20 miles LOS |
| 21:37.03 | watchy | I think its about $1k for a 54meg unit were using 3megs for this co and 54megs for some other |
| 21:37.36 | watchy | were using Alvarion equipment. There backhaul shit is great but i don't like there fucking CPE units for consumers |
| 21:37.48 | watchy | i really hate them. they run on 900mhz and are pieces of shit |
| 21:37.54 | watchy | were having so many problems with them |
| 21:38.11 | eKo1 | I use motorola canopy. |
| 21:38.48 | eKo1 | Expensive, but it works pretty good as a backhaul. |
| 21:38.58 | watchy | well our backhaul is great |
| 21:39.03 | watchy | what sucks is our client stuff |
| 21:39.14 | watchy | you checked out this www.nextnet.com stuff? it looks great |
| 21:39.38 | watchy | www.nextnetwireless.com i mean |
| 21:40.20 | watchy | it runs on licensed freqs though but I think i want to look into it |
| 21:40.21 | watchy | i;m tired of our 900mhz equiment sucking |
| 21:40.32 | MikeJ | thanks for the info... |
| 21:40.34 | jbalcomb | I was just asked to make Randy Estelle our hold music. WTF! It's christian contemporary/gospel. |
| 21:40.38 | AsterLost | can anyone help me with a 1-way audio problem? |
| 21:40.43 | MikeJ | is that 1k for each side.. or for a pair? |
| 21:41.02 | watchy | mikej: 1 side without aentenna as far as i know at 54mbit |
| 21:41.11 | [TK]D-Fender | watchy : What is the full link chain of an echo-y call? |
| 21:41.40 | watchy | full link chain? |
| 21:41.47 | watchy | im retarded. english please |
| 21:42.05 | [TK]D-Fender | watchy : Every piece or equipement and reencoding from 1 end to the other. |
| 21:42.16 | [TK]D-Fender | watchy : phones, links and all |
| 21:43.06 | AsterLost | please? |
| 21:43.09 | watchy | I got a amd athlon 1900 asterisk box connected by gbit to a dell gbit switch to a linksys rv016 router connected to the 3mbit link |
| 21:43.24 | eKo1 | watchy: ever tried terabeam? |
| 21:43.39 | watchy | the other end is a 3mbit link to a linksys rv082 vpn router to a switch to a polycom 501/601 |
| 21:43.42 | *** join/#asterisk Vorondil (n=jkiffmey@miro.voltaiccommerce.com) |
| 21:43.56 | watchy | the lines are connected from AT&T 8 analogs to 2 TDM400ps |
| 21:44.04 | watchy | eko1: no that for backhaul or what? |
| 21:44.32 | jbalcomb | watchy: Is that a Dell PowerConnect? What model number? |
| 21:44.35 | eKo1 | watchy: yeah |
| 21:44.58 | *** join/#asterisk arkonadev (n=chatzill@65.203.186.131) |
| 21:45.03 | watchy | jbal: no idea but yea its a powerconnect |
| 21:45.12 | watchy | dunno the model |
| 21:45.36 | watchy | 24port gbit from what i remember |
| 21:46.05 | arkonadev | is anyone here familiar with fonality? |
| 21:46.07 | [TK]D-Fender | watchy : So LAN + wireless to a remote Polycom? |
| 21:46.13 | eKo1 | I use teletronic ezbridges for cpe's |
| 21:46.23 | [TK]D-Fender | watchy : TDM400P sounds to be the problem..... |
| 21:46.28 | jbalcomb | watchy 3324 or 5324 prolly. You ever update the firmware on that thing? |
| 21:46.41 | [TK]D-Fender | watchy : Notorious |
| 21:47.18 | nestar | anyone know how the Rhino analog cards do? |
| 21:47.19 | arkonadev | does anyone know why the callerid would not be showing up from an inbound call? |
| 21:47.23 | watchy | tk: yea i think your probably right about the TDMs. gotta be some way to fix the damn echo though |
| 21:47.24 | nestar | better worse the same? |
| 21:47.34 | *** join/#asterisk Alric (n=nbowyer@ppp-db.1stel.com) |
| 21:47.49 | jbalcomb | nestar: haven't heard anything |
| 21:47.59 | *** join/#asterisk SwK (n=Silik0nJ@64.89.118.139) |
| 21:48.23 | watchy | jbal: should i upgrade the firmware on em? |
| 21:48.25 | nestar | i want to use asterisk at my new job, but we can't justify a t1, but i'm concerned with the analog cards |
| 21:48.49 | nestar | not enough bandwidth to do voip over the dsl |
| 21:48.52 | nestar | so dunno |
| 21:48.59 | watchy | nestar: i aint had nothing but issues with my tdm400ps, you can get cards with hardware echo cancelation though |
| 21:49.03 | *** join/#asterisk nettie (i=[U2FsdGV@85-18-54-38.ip.fastwebnet.it) |
| 21:49.04 | watchy | i think thats the way to go |
| 21:49.06 | nettie | Hi guys, does anyone know if trunk supports VAD/CNG pls? |
| 21:49.32 | nestar | seems like a channel bank would be the way to fo, but they are spendy. $$$$ |
| 21:49.57 | nestar | ILEC here doesn't offer fractional PRI's |
| 21:50.12 | watchy | id pay someone cash to fix my fucking echo issues |
| 21:51.21 | viler | Hello there, does anyone know if Is possible to execute an agi when a call arrives at the member queue? |
| 21:52.04 | *** join/#asterisk santiago (i=santiago@debian/developer/santiago) |
| 21:52.38 | [TK]D-Fender | watchy : A200 |
| 21:53.25 | jbalcomb | viler: you want to so some agi after you use the queue function? any reason you can't do it right before that? |
| 21:53.32 | *** join/#asterisk ArkonaDev-Mike (n=somewher@65.203.186.131) |
| 21:54.20 | jbalcomb | viler: have you tried putting it in the context specified by the queues.conf? |
| 21:54.24 | watchy | I would think about it if digium would give me a fucking refund |
| 21:55.21 | watchy | but i already wasted $500 a piece for the 4 port cards that i cant get to stop echoing |
| 21:55.24 | *** join/#asterisk MikeJ (n=vircuser@d14-69-8-30.try.wideopenwest.com) |
| 21:56.51 | watchy | god i want one of these sangomas now |
| 21:57.08 | viler | jbalcom: I need to know who is the person (member queue) that answer the call, and execute an agi to show data agent browser |
| 21:57.39 | viler | to show data in agent browser |
| 21:58.01 | *** join/#asterisk tempest1 (n=asf@adsl-12-51-176.chs.bellsouth.net) |
| 21:59.01 | jbalcomb | What gives? "queue_set_param: Unknown keyword in queue 'tesuto-queue': eventmemberstatusoff" |
| 21:59.19 | jbalcomb | eventmemberstatusoff is listed in the queues.conf.sample |
| 21:59.25 | watchy | i kinda wish we woulda gone with a T1 card tk |
| 22:00.11 | eKo1 | watchy: You can still have echo problems with a T1 card. |
| 22:00.31 | jbalcomb | nestar: VoIP takes up less than 8K of bandwidth... |
| 22:00.43 | watchy | yea but i woulda got hw echo cancelation |
| 22:01.05 | jbalcomb | watchy: you still should and sell the TDM400s on ebay. |
| 22:01.09 | watchy | instead of this faggot ass digium no hw echo cancell and have the worst echo ever in a fucking production enviroment |
| 22:01.21 | mrdigital | on a analog phone how do i transfer a call |
| 22:01.24 | eKo1 | no no, keep the tdm400 and use it for testing |
| 22:01.37 | jbalcomb | eKo1: testing what? |
| 22:01.45 | jbalcomb | "Yep, still echoes" |
| 22:01.57 | eKo1 | like testing your pir |
| 22:01.59 | eKo1 | err, pri |
| 22:02.24 | jbalcomb | you cant test your PRI on a TE411P? |
| 22:02.30 | nestar | jbalcomb: kilobytes or bits... in any event without some sort of QoS.. surfing will kill said voip. |
| 22:02.49 | eKo1 | jbalcomb: i'm talking about testing each zap channel |
| 22:03.00 | eKo1 | on the pri of course |
| 22:03.11 | jbalcomb | nestar: k/Kb is bits, K/KB is bytes |
| 22:03.34 | jbalcomb | nestar: many people do it regardless and why can't you set up QoS? |
| 22:04.27 | Cresl1n | watchy did you run fxotune? |
| 22:04.31 | jbalcomb | eKo1: i guess i'm in the dark cause i don't see how that's working out better for you |
| 22:04.44 | *** join/#asterisk russellb (n=russell@asterisk/developer-and-stable-maintainer/drumkilla) |
| 22:04.44 | *** mode/#asterisk [+o russellb] by ChanServ |
| 22:05.18 | jbalcomb | russellb: any trees fall in your living room |
| 22:05.38 | eKo1 | jbalcomb: it is understandable, I have my pri's sitting behind an ss7 gateway |
| 22:07.27 | CrashHD | I have an issue where rtp debug is not showing any rfc2833 packets being sent to to another asterisk even though both servers are clearly set to rfc2833, anyone have any thoughts on how to troubleshoot this? |
| 22:07.45 | CrashHD | I do however see them recieved from the phone |
| 22:08.39 | *** join/#asterisk tempest1 (n=asf@adsl-12-51-176.chs.bellsouth.net) |
| 22:12.38 | Toerkeium | guys, is there anyway or anyone knows a product to connecy a cell phone to a landline ? something like a GSM/GPRS access point ? |
| 22:13.04 | *** join/#asterisk sb_mx (n=sb_mx@200.78.229.18) |
| 22:14.33 | *** join/#asterisk insomni (n=insomni@x1-6-00-e0-18-6f-34-ad.k455.webspeed.dk) |
| 22:14.36 | Sedorox | Toerkeium: there are plenty of them if you look on ebay |
| 22:14.45 | Bullseye_Network | Toerkeium: http://www.voip-info.org/wiki/view/Asterisk+Connecting+to+the+Cellular+Network |
| 22:15.04 | Bullseye_Network | this will connect asterisk to the cell network |
| 22:15.29 | Bullseye_Network | or landline |
| 22:15.58 | Toerkeium | great.. |
| 22:16.05 | Toerkeium | have you working with that ever? |
| 22:16.16 | *** join/#asterisk |dennis| (i=dennis@200.32.215.84) |
| 22:16.50 | *** join/#asterisk tempest1 (n=asf@adsl-12-51-176.chs.bellsouth.net) |
| 22:17.42 | *** join/#asterisk Tall-guy (i=tall-guy@207-195-103-110.regn.static.sasknet.sk.ca) |
| 22:23.01 | Tall-guy | ok, quick one kids....Sipura 942, or Polycom 501......which should I buy ? |
| 22:23.11 | Toerkeium | I want to make empoys of a business call with throug * with their cell phones using a landline or a GPRS modem connected to asterisk.. anyone know if this is possible? I see only asterisk > GPRS/GSM, not GSM device > asterisk.. any idea? |
| 22:25.51 | *** part/#asterisk CANO-1982 (i=alejandr@190.48.72.107) |
| 22:26.25 | [TK]D-Fender | Tall-guy : IP 501 hands down. |
| 22:27.13 | Tall-guy | reasoning? |
| 22:28.16 | *** join/#asterisk CANO-1982 (n=alejandr@190.48.72.107) |
| 22:32.59 | Tall-guy | fender? |
| 22:34.59 | insomni | does anyone know if it is possible to connect a cell phone to asterisk via a bluetooth interface on the server to place cell calls and sms? |
| 22:35.27 | *** join/#asterisk shmaltz (n=mybox@mail.dmaven.com) |
| 22:36.00 | shmaltz | ~seen tzafrir |
| 22:36.15 | jbot | tzafrir <n=tzafrir@local.xorcom.com> was last seen on IRC in channel #asterisk, 10d 18h 56m 58s ago, saying: 'http://the-edge.blogspot.com/2005/10/worldss-smallest-ip-pbx-at-astricon.html'. |
| 22:36.15 | shmaltz | ~seen tzafrir_laptop |
| 22:36.17 | jbot | tzafrir_laptop <n=tzafrir@bzq-88-152-23-125.red.bezeqint.net> was last seen on IRC in channel #asterisk, 3d 16h 51m 2s ago, saying: 'that said, #freepbx may be a better place'. |
| 22:36.46 | *** part/#asterisk Tall-guy (i=tall-guy@207-195-103-110.regn.static.sasknet.sk.ca) |
| 22:39.16 | *** part/#asterisk CANO-1982 (n=alejandr@190.48.72.107) |
| 22:40.11 | insomni | about my bluetooth question above, all i found on the wiki was this: http://lists.digium.com/pipermail/asterisk-dev/2004-June/thread.html#4703 not very up to date and not that informative |
| 22:40.39 | *** join/#asterisk RoyK (n=roy@122.80-203-78.nextgentel.com) |
| 22:41.49 | *** join/#asterisk dprevite (n=dprevite@cpe-66-61-129-18.insight.res.rr.com) |
| 22:42.26 | *** join/#asterisk CANO-1982 (n=alejandr@190.48.72.107) |
| 22:45.29 | WeirdM | Is there an easy way to tell whether Asterisk is recognizing my X100P? |
| 22:46.08 | RoyK | thunder! |
| 22:46.17 | RoyK | WeirdM: zap show channels |
| 22:46.22 | RoyK | or something |
| 22:46.36 | WeirdM | In the Asterisk CLI? |
| 22:48.52 | sb_mx | WeirdM, yes |
| 22:52.16 | *** join/#asterisk ariel_ (n=ariel_@74.8.35.2) |
| 22:52.20 | ariel_ | hello everyone |
| 22:53.29 | WeirdM | I don't even have a zap command in my CLI. Do I have something configured wrong? |
| 23:02.41 | *** join/#asterisk Arno[Slack] (n=hellSOUN@master.infinityperl.org) |
| 23:04.43 | *** part/#asterisk ariel_ (n=ariel_@74.8.35.2) |
| 23:07.04 | *** join/#asterisk stkn_ (i=nobody@gentoo/developer/pdpc.active.stkn) |
| 23:17.29 | *** join/#asterisk acctor (n=heh@user-2525.l4.c5.dsl.pol.co.uk) |
| 23:17.39 | Zodiacal | anyone know what the polycom softkey "split" is for? |
| 23:17.45 | Zodiacal | it shows up when i am trying to transfer a call |
| 23:18.01 | *** join/#asterisk smash- (i=smash@remainsilent.org) |
| 23:18.06 | smash- | Hello |
| 23:18.19 | acctor | hello - I have .sb2 and .loads files for the firmware for my cisco 7960 but the phone is looking for .sbn. Can I get it to look for the loads/sb2 pair instead |
| 23:22.12 | shmaltz | acctor, what does your OS97xx.txt look like? |
| 23:23.07 | shmaltz | anybody here dealing with voipsupply? |
| 23:23.35 | *** join/#asterisk javar (n=javar@Dynamic-IP-cr20011876227.cable.net.co) |
| 23:23.42 | javar | hi |
| 23:24.28 | javar | how change: register => 1111:11xx@xx.xx.xx.xx/1111 to field on extensions_conf using ARA ? |
| 23:25.22 | javar | somebody help me, pls? |
| 23:25.44 | *** join/#asterisk santiago (i=santiago@debian/developer/santiago) |
| 23:26.12 | acctor | shmaltz: nm I've got it |
| 23:26.14 | acctor | thanks anyway |
| 23:26.52 | reticent | wow. |
| 23:26.52 | javar | hi |
| 23:27.02 | reticent | mysql federated tables are quite terrible |
| 23:27.19 | *** join/#asterisk santiago (i=santiago@debian/developer/santiago) |
| 23:28.40 | reticent | for any query you execute it transfers the ENTIRE TABLE from the source database into memory on the database the federated table exists, returns your results (after sorting, limiting, joining, etc) and then dumps all the data. Pulling all the data anew for the next query |
| 23:28.55 | smash- | hey does anyone wanna buy a sangoma card =P or trade it for a red-fone |
| 23:29.30 | *** part/#asterisk javar (n=javar@Dynamic-IP-cr20011876227.cable.net.co) |
| 23:29.42 | smash- | Haas anyone used a Red-fone tdm device? |
| 23:29.56 | smash- | <PROTECTED> |
| 23:33.19 | *** join/#asterisk N3GLV (n=n3glv@monrovll-cuda1-24-53-251-235.pittpa.adelphia.net) |
| 23:33.23 | N3GLV | CRAP |
| 23:33.34 | N3GLV | dropped kbd, it closed window |
| 23:33.38 | N3GLV | ooops |
| 23:33.40 | N3GLV | sorry |
| 23:33.42 | *** part/#asterisk N3GLV (n=n3glv@monrovll-cuda1-24-53-251-235.pittpa.adelphia.net) |
| 23:40.53 | *** join/#asterisk jbot_ (i=ibot@pdpc/supporter/active/TimRiker/bot/apt) |
| 23:40.53 | *** topic/#asterisk is Asterisk: The Open Source PBX -=- http://www.asterisk.org -=- Asterisk 1.2.10 and Zaptel 1.2.7 released! (July 14, 2006) -=- FreePBX/AMP/Asterisk@Home/Trixbox support in #freepbx |
| 23:42.09 | *** join/#asterisk hads|home (n=hads@mail.nice.net.nz) |
| 23:45.40 | Dovid | can I use the t extension in a macro ? |
| 23:47.05 | *** join/#asterisk Bobcat_1966 (n=chatzill@cpe-069-132-139-254.carolina.res.rr.com) |
| 23:51.47 | Toerkeium | does anyone know USB to RJ11 adapters? I have a GSM modem which I would like to connect to a pap, so I can receive calls from the GSM network to *.. anyone knos? |
| 23:56.57 | *** join/#asterisk ph|ber (n=phiber@slackwaresupport.com) |
| 23:57.26 | ph|ber | anyone have a prob with polycom 601 not pulling the -directory.xml ?? |
| 23:57.53 | ph|ber | is John in here.. from * ? |
| 23:58.43 | carrar | I'm your huckleberry! |
