| 00:02.03 | BSD_Tech | real-time is database drivenn |
| 00:03.14 | *** join/#asterisk vitaminmoo (n=vitaminm@70.58.177.109) |
| 00:06.15 | tuxd00d | Could one of you guys tell me which command I need to go through a list of files, and preform a action on each file..... so like a "for each" statement |
| 00:06.34 | tuxd00d | like foreach(*.call) {} |
| 00:06.49 | tuxd00d | I'm having a mental block |
| 00:10.02 | BSD_Tech | its sunday your not going to get alot of input |
| 00:10.21 | BSD_Tech | come back tomarrow morning |
| 00:10.23 | BSD_Tech | lol |
| 00:10.44 | *** join/#asterisk rmayorga_ (n=rmayorga@unaffiliated/rmayorga) |
| 00:11.13 | *** join/#asterisk zotz (n=zotz@24.244.163.157) |
| 00:11.33 | flenders | tuxd00d: for file in `ls` ; do command $file ; done |
| 00:13.02 | *** join/#asterisk webman (n=chatzill@124.246.8.196.static.nexnet.net.au) |
| 00:13.53 | tuxd00d | flenders: I need to mv a file, sleep 1 minute, and go to the next file, all with '.call' extentions. |
| 00:14.38 | flenders | tuxd00d: for file in `ls *.call` ; do mv $file /tmp/wherever_youwant ; sleep 1 ; done |
| 00:15.02 | tuxd00d | oh, groovy, thanks flenders |
| 00:15.57 | webman | anyone have experience with h323, and know which channel driver is the most reliable for h323 <-> h323 calls ? |
| 00:21.43 | *** join/#asterisk MrMister2 (n=mrmister@89-180-93-125.net.novis.pt) |
| 00:28.50 | *** join/#asterisk rmayorga (n=rmayorga@unaffiliated/rmayorga) |
| 00:31.49 | *** join/#asterisk tomcontr3 (n=tomcontr@200-77-246-201.adsl.terra.cl) |
| 00:32.03 | tomcontr3 | hi, Im having some problems with Voicemail Emails |
| 00:32.29 | tomcontr3 | every time I record a new voicemail, and asterisk sends the email, the email gets recognized as SPAM |
| 00:32.55 | JT | by what? |
| 00:33.00 | JT | and how's that an asterisk problem |
| 00:33.14 | webman | I can get IAX2 -> h323 working and h323 -> IAX2 working, but can't get h323 -> h323 working |
| 00:34.24 | webman | using the chan_h323 from asterisk |
| 00:35.14 | tomcontr3 | it isnt, but, If there is anyone that has this feature working and could give me a hand I would really appreciated |
| 00:35.23 | tomcontr3 | Im sending the emails to my gmail account |
| 00:35.37 | JT | then mark them as not spam |
| 00:35.39 | tomcontr3 | and all those emails get droped to the SPAM folfer |
| 00:35.46 | MrTelephone | do any of you get stale call-limits? |
| 00:36.00 | tomcontr3 | right, but I also tried other mail accoutns, and is the same storry |
| 00:38.40 | tomcontr3 | this is the headers http://pastebin.ca/671838 |
| 00:43.41 | tomcontr3 | ? |
| 00:47.30 | fujin | hey uh, should a queue go to the next priority of callers when the first 0 priority user doesn't answer? |
| 00:47.41 | fujin | members rather |
| 00:49.37 | *** join/#asterisk riddlebox (n=victoria@75-132-205-90.dhcp.stls.mo.charter.com) |
| 00:50.05 | riddlebox | is there a way to have the voicemail emails converted to mp3 instead of wav? |
| 00:52.14 | *** join/#asterisk redbaron1973 (n=redbaron@host55-226.rancor.birch.net) |
| 00:53.24 | redbaron1973 | I have a question regarding HDLC /c the TE420, anyone familiar with this? |
| 00:54.51 | flenders | riddlebox: out of the box, no |
| 00:55.25 | riddlebox | hrmm my moto q doesnt like the wav files that it sends in the emails |
| 00:56.27 | CCFL_Man2 | riddlebox: moto q ftl |
| 00:56.55 | riddlebox | CCFL_Man2, ftl? |
| 00:57.14 | *** join/#asterisk ptiggerdine (n=ptiggerd@123-243-144-208.tpgi.com.au) |
| 00:57.14 | flenders | not sure if changing the format from wav_49 to wav would help you |
| 00:57.31 | Sweeper | http://www.flickr.com/photos/mrneutron/sets/1568481/ <-- best. photoset. ever. |
| 00:57.34 | *** join/#asterisk kFuQ (n=somedude@c-67-185-112-20.hsd1.wa.comcast.net) |
| 00:59.40 | fujin | christ, still can't get this to work |
| 00:59.45 | fujin | can anyone confirm queue behaviour? |
| 01:00.03 | fujin | I'm trying to make it so that if someone sits in a queue, and the pentalty=0 person doesn't answer, it jumps to the penalty=1 people. |
| 01:00.09 | fujin | doesn't wanna work though |
| 01:00.14 | redbaron1973 | < in zaptel.conf, if I am wanting to bond 2 Data-t1's as an HDLC, would it be proper to set nethdlc=1-48 ? |
| 01:02.57 | CCFL_Man2 | Sweeper: i saw goatse for the 1st time two years ago but knew about it for years |
| 01:03.20 | CCFL_Man2 | and i must say i never want to see it again |
| 01:07.12 | *** join/#asterisk WilliamK (n=noc@static-71-170-144-3.dllstx.fios.verizon.net) |
| 01:13.22 | *** join/#asterisk rmayorga (n=rmayorga@unaffiliated/rmayorga) |
| 01:20.20 | *** join/#asterisk bminish (n=bminish@brenbox.westnet.ie) |
| 01:25.05 | bminish | stubit question |
| 01:25.16 | bminish | Just rebuilt asterisk wit the latest buld |
| 01:25.25 | bminish | now it has no awareness of my zaptel hardware |
| 01:25.46 | bminish | what stupid thing did I do? |
| 01:26.22 | bminish | no zaptel commands available in the asterisk console but modules loaded ok |
| 01:26.29 | bminish | ver 1.4.11 |
| 01:26.33 | webman | bminish: do a make menuconfig and ensure channels zap is incliuded?? and/or install that latest zaptel before you install asterisk |
| 01:26.36 | bminish | zaptel 1.4.0 |
| 01:27.06 | webman | possibly re-run ./configure in asterisk before you do the make menuconfig |
| 01:27.56 | bminish | hmm make meunconfig shows zap as unchoosable , i had forgotten about the menuconfig |
| 01:28.12 | bminish | doing a make clean ./configure |
| 01:28.21 | bminish | and see if ti picks it up this time |
| 01:28.41 | russellb | did you install the latest zaptel release as well? |
| 01:28.52 | bminish | latest one would not build for me udev related error |
| 01:29.06 | bminish | so rebuilt and reinstalled 1.4.0 |
| 01:29.25 | bminish | which is also linked in my src dir as zaptel |
| 01:30.08 | bminish | bugger chan_zap still not selectable |
| 01:30.40 | russellb | 1.4.0? |
| 01:30.53 | russellb | hm, try the one right before the latest one then |
| 01:30.59 | BSD_Tech | move to 1.4.10.1 and zaptel 1.4.5 |
| 01:31.01 | bminish | ok which was ? |
| 01:31.06 | russellb | or even better, try straight from svn |
| 01:31.15 | bminish | ok |
| 01:31.23 | russellb | svn co http://svn.digium.com/svn/zaptel/branches/1.4 zaptel-1.4 |
| 01:31.28 | *** join/#asterisk dw (n=dw@unaffiliated/dw) |
| 01:35.36 | bminish | my asterisk box is dog slow when it comes to compiling stuff |
| 01:40.45 | bminish | does asterisk ./config need to know where the current zaptel src is located? |
| 01:42.30 | *** join/#asterisk riddlebox (n=james@75-132-225-75.dhcp.stls.mo.charter.com) |
| 01:42.47 | bminish | bugger zaptel make install issue with svn version |
| 01:42.48 | bminish | build_tools/genudevrules: line 3: udevinfo: command not found |
| 01:42.48 | bminish | make: *** [devices] Error 1 |
| 01:43.21 | bminish | kernel 2.6.9-42 |
| 01:44.08 | bminish | a bit old I know but far from straight forward to move this box on to later kernel |
| 01:45.31 | russellb | bminish: hm, i guess file a bug on the bug tracker for your problems installing the latest code |
| 01:45.38 | russellb | bminish: inlude the distro information |
| 01:45.46 | *** join/#asterisk rmayorga (n=rmayorga@unaffiliated/rmayorga) |
| 01:45.54 | bminish | perhaps it's because i am missing udevinfo |
| 01:46.06 | SplasPood | Hrm... anyone happen to know if the Nokia N95 sip client supports video? (I know this is only tangentially related to asterisk) |
| 01:46.21 | bminish | e70 does not |
| 01:46.59 | SplasPood | Yea I don't think the N95 does either, which makes the video call support it *does* has totally worthless to me |
| 01:47.05 | SplasPood | s/has/have |
| 01:51.35 | hmmhesays | ~seen bkw_ |
| 01:51.43 | jbot | bkw_ <n=brian@pool-71-246-222-63.washdc.fios.verizon.net> was last seen on IRC in channel #asterisk, 8d 6h 47m ago, saying: 'Sweeper, just an FYI anything over say 4 v's is pointless'. |
| 01:52.21 | fujin | Hi look I'm trying to get a queue working with two levels of priorities, I'd like the first priority level (0) to ring for only 15 seconds before ringing the second priority level |
| 01:52.28 | fujin | can anyone tell me how to do this? |
| 01:52.43 | bminish | in the absence of udevinfo in the distro any suggestions as to how I can get this ti make install |
| 01:53.03 | bminish | I am abut messed up here, this needs to be up and running in 4 hours |
| 01:54.44 | bminish | OHH, dohh this box isn;t using udev |
| 01:55.00 | bminish | but how to get zaptel to install on dev |
| 02:01.11 | bminish | is it possible to build zaptel without udev |
| 02:04.10 | webman | bminish: I am pretty sure you can build zaptel without udev |
| 02:04.35 | bminish | delete ./build_tools/gendevrules |
| 02:05.23 | bminish | STILL no chan_zap available in asterisk make menuconfig |
| 02:05.46 | webman | bminish rerun ./configure |
| 02:06.33 | webman | bminish: did the zaptel make install complete successfully ? |
| 02:06.39 | *** join/#asterisk anthony[ (n=anthony@fl-71-49-118-147.dhcp.embarqhsd.net) |
| 02:06.44 | bminish | worst thing is that if I get this working the bloody phone will be ringing in a few hours, it's 3 am here |
| 02:06.57 | anthony[ | Hi, what do I have to do to record ALL PHONE calls? |
| 02:07.07 | bminish | yes installed with no errors it fell back to the dev install, guess it assumes that 2.6 kernel = udev |
| 02:07.35 | webman | bminish: what does ./configure say about zaptel (in the asterisk source dir) |
| 02:07.45 | jer | anthony[, consult a lawyer to make sure it's legal in your jurisdiction. for instance, here in Canada, it's legal so long as one party knows about it and that party is the one doing the recording (automated recording systems are illegal here)... this may or may not be similar in your jurisdiction |
| 02:08.01 | webman | anthony: show application mixmonitor and configure your diaplan appropriately |
| 02:08.56 | webman | jer: are you the author of chan_h323 or am I confusing you with someone else? |
| 02:09.05 | jer | webman, confusing me with someone else |
| 02:09.12 | bminish | lots of yesses and we are now good in make menuconfig |
| 02:09.15 | bminish | thanks |
| 02:09.32 | bminish | now to compile and see if we won ;-) |
| 02:09.36 | webman | bminish: good luck, don't forget to get some sleep before the phone rings :) |
| 02:10.04 | webman | jer: ok, sorry.... desperately trying to get h323 working, with rather limited success..... |
| 02:10.14 | jer | ah |
| 02:10.21 | jer | wish i could help, but i've never played with it |
| 02:15.21 | *** join/#asterisk Faithful (n=Faithful@ns.linuxterminal.com) |
| 02:17.05 | *** join/#asterisk tengulre (n=tengulre@222.90.66.10) |
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| 02:40.45 | kiscokid | My Polycom phone won't boot |
| 02:41.23 | mcab | kiscokid: can it contact its bootserver? |
| 02:41.58 | kiscokid | well, it seems to but it gets config file errors |
| 02:42.21 | kiscokid | 4020 or 10020 |
| 02:42.33 | kiscokid | can't find any doc for these |
| 02:43.40 | kiscokid | mcab: have you worked with Polycom phones? |
| 02:44.23 | mcab | kiscokid: yup, IIRC 4020 usually means it can't connect to/log into the boot server |
| 02:44.36 | mcab | 0x10020 I don't remember |
| 02:44.50 | kiscokid | where do you find these? |
| 02:45.10 | mcab | kiscokid: trial and error :-) |
| 02:45.51 | tengulre | how to config two asterisk box with HA software, I have two server box, no interface card, only provide sip services. |
| 02:46.27 | mcab | kiscokid: are you using FTP? |
| 02:46.35 | bminish | webman we are good good go, thanks for the help whilst I bumbled about |
| 02:47.18 | kiscokid | mcab: was using tftp, I was going to try ftp next |
| 02:48.06 | kiscokid | mcab: what command do you use in dhcpd.conf to tell it the address of the boot server? |
| 02:48.58 | kiscokid | only way I could get it to see the boot server was to use the phone interface |
| 02:49.16 | kiscokid | in the server menu |
| 02:50.07 | mcab | kiscokid: by default they use Option 66 with DHCP |
| 02:53.18 | mcab | if you use the phone gui to change the bootserver, you need to go into the DHCP menu and change the *mumble* option from 'Option 66' to 'static' |
| 02:53.37 | mcab | (the exact name eludes me right now) |
| 02:55.33 | kiscokid | is there anyway to get the phone to go back to factory defaults? All the methods say press something like "press 2 4 6 8 simultaneously" but they don't work |
| 02:55.54 | kiscokid | all I get now is the phone trys to boot over and over |
| 02:56.09 | kiscokid | and gets one of those errors |
| 02:56.40 | kiscokid | mcab: which phone model and firmware version are you using? |
| 02:57.28 | mcab | kiscokid: a variety :-) what model & firmware are you using? |
| 02:57.56 | kiscokid | IP 430 with 1.6.7.133 |
| 02:58.10 | kiscokid | that's what it came with |
| 02:58.27 | mcab | the reset to default is usually 4,6,8,*; but the 430 is different I think |
| 02:58.46 | mcab | 1,3,5,7 maybe? |
| 02:59.00 | mcab | what BootROM is it running? |
| 02:59.06 | kiscokid | at what point can you do that? |
| 02:59.18 | kiscokid | one sec, I gotta plug in the phone |
| 02:59.49 | mcab | kiscokid: pretty much at anypoint, I think - I usually do it at the count down screen just after the phone powers up |
| 02:59.59 | mcab | or when the phone is booted and idle |
| 03:00.29 | *** join/#asterisk santiago (n=santiago@debian/developer/santiago) |
| 03:01.20 | kiscokid | boot rom 3.2.2.0019 |
| 03:02.00 | mcab | ok, so the BootROM is fairly recent, the app you should upgrade (2.2 was just released, 2.1.2 was the previous version) |
| 03:02.45 | kiscokid | if I put the app and the default config files on the TFTP server should it boot? |
| 03:03.09 | mcab | kiscokid: yes, so long as it can contact the server :-) |
| 03:03.22 | mcab | kiscokid: unfortunately, dinner is ready so I have to take off |
| 03:03.23 | *** join/#asterisk luke802 (n=luke802@69.73.203.190) |
| 03:03.37 | kiscokid | right now I'd just like to get it to boot from the flash |
| 03:03.52 | kiscokid | mcab, ok thanks for the help |
| 03:03.52 | luke-jr_ | anyone know how to unlock a PAP2-NA? :/ |
| 03:04.09 | luke802 | Hi all |
| 03:04.23 | *** join/#asterisk stkn_ (n=stkn@gentoo/developer/pdpc.active.stkn) |
| 03:04.52 | luke802 | we currently support asterisk |
| 03:05.22 | luke802 | what is a reasonable cost for suport in the us? |
| 03:05.39 | luke802 | (to charge for support) |
| 03:07.01 | mmlj4 | luke802: depends. I charge about $100 per hour, regardless of what I'm working on, your market may vary |
| 03:08.08 | luke802 | mmlj4: thanks. We charge thereabouts currenlty in our region |
| 03:08.20 | mmlj4 | then I'd roll with that |
| 03:08.40 | russellb | i'll do it for $99 ! |
| 03:08.40 | *** join/#asterisk guillote_GNU (n=guillote@190.7.27.17) |
| 03:08.41 | luke802 | thank you. bbr. got a couple more qs |
| 03:08.50 | mmlj4 | but understand that people are used to being ripped off by traditional phone vendors |
| 03:09.09 | luke802 | yes i know. |
| 03:09.13 | mmlj4 | give them solid, sane service and don'and you'll make a name for yourself |
| 03:09.22 | mmlj4 | give them solid, sane service and don't try to gouge them, and you'll make a name for yourself |
| 03:09.24 | luke802 | we spent years selling nortel and toshiba |
| 03:09.27 | luke802 | we stopped doing that. |
| 03:09.32 | luke802 | we "consult" |
| 03:14.07 | *** join/#asterisk luke802 (n=luke802@69.73.203.190) |
| 03:14.31 | luke802 | sorry. got disconnected. |
| 03:15.20 | luke802 | we try not to posisiton ourselves as pbx vendors and work more on a consulting basis. With experience in several major pbx brands we sell all equipment at costs and work solely on service contract and consulting fees. |
| 03:16.23 | luke802 | now our prices are less than 50% of our competitors' proprietary solutions. so far, its been a good model for us. |
| 03:16.56 | *** join/#asterisk DarkRift (i=dark@bas10-montreal02-1177584025.dsl.bell.ca) |
| 03:18.13 | luke802 | another question I have though... does anyone have experience with Digium';s professional development services? |
| 03:18.44 | luke802 | i've read some mixed reviews on the overall service and eventual outcomes of projects. does anyone here have an opinion? |
| 03:19.58 | luke802 | why biased? |
| 03:20.03 | *** join/#asterisk CrazyTux[m] (n=CrazyTux@216-110-94-230.static.twtelecom.net) |
| 03:20.03 | luke802 | employee? |
| 03:22.57 | russellb | yes, i work there |
| 03:23.07 | russellb | not in that group, specifically |
| 03:24.13 | luke802 | ahh (i) |
| 03:24.36 | russellb | (i) ? |
| 03:27.13 | luke802 | nevermind.. using msn messenger icons ... |
| 03:27.18 | luke802 | ofcourse, they dont work on IRC |
| 03:27.19 | luke802 | lol |
| 03:27.49 | luke802 | so, biased russellb, is the digium boot camp all it's cracked up to be ? |
| 03:28.26 | luke802 | our company is scheduling training in october (hopefully). |
| 03:29.13 | russellb | yeah, i have heard a lot of positive reviews of it. |
| 03:29.38 | russellb | the guys that teach it are extremely knowledgable |
| 03:30.48 | *** join/#asterisk JerJer (n=PhatJ@pdpc/supporter/bronze/jerjer) |
| 03:31.11 | luke802 | well thats good to hear |
| 03:34.11 | *** join/#asterisk nclx (n=nightcal@carnivore.scrapshells.com) |
| 03:35.42 | nclx | well I guess my motherboard / HP-BIOS must really suck. Even if I disable HPET, set timer to 1000Hz in kernel, unload ztdummy and zaptel kernel mods, I am still getting about 30 syslogs per second saying rtc: missed some interrupts at 1024Hz. I give up, I'm trying asterisk on a different box. |
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| 03:48.37 | luke802 | Sorry if this is not the correct place to ask. Does anyone know of a retailer who provides rack mountable servers specifically built to run asterisk? |
| 03:49.24 | JerJer | Dell |
| 03:51.25 | nclx | look also a eracks.com (no I don't work for them) but they support Open Source OS's and it is branded all over there site |
| 03:52.18 | *** join/#asterisk heelios (n=heelios@onyx.6pixies.com) |
| 03:52.30 | heelios | can I use a nice, rotary phone with a PAP2? |
| 03:55.57 | nclx | My local cable company "Bright House Networks", is looking like a bunch of buffoons. Check out their new marketing site: http://www.asteriskhunters.com The idea is that they are eliminating the "Asterisk" IE all the fine print and fees associated with typical telecom accounts with their Digital Phone Service. However it is just funny the name they choose because obviously no body there had heard of Asterisk the uber popular Free PBX software |
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| 04:17.39 | itiliti | anyone feel like helping me out real quick> |
| 04:18.52 | *** join/#asterisk GiantPickle (n=GiantPic@S01060016b600537f.gv.shawcable.net) |
| 04:20.53 | *** join/#asterisk d4rkst4r75 (n=d4rkst4r@ip-172-69.sn3.eutelia.it) |
| 04:25.40 | *** join/#asterisk antlers (n=antlers@ip70-173-89-173.lv.lv.cox.net) |
| 04:30.48 | JerJer | itiliti: how about helping yourself by asking a specific question? |
| 04:31.13 | *** join/#asterisk mtaht4 (n=m@237-109-62-200.enitel.net.ni) |
| 04:31.26 | itiliti | True, true. |
| 04:31.44 | itiliti | Is there a way to write a group pickup to grab a rig group? |
| 04:31.47 | itiliti | ring* |
| 04:32.28 | itiliti | I have a ring group that rings on the main SIP trunk. I would like for the phones to be able to pick up the call if the person is not there.. |
| 04:32.33 | itiliti | from their phones.. |
| 04:32.54 | itiliti | it works great on DId tied to their direct extension, but when I am trying it with the 600 ring group.. |
| 04:32.55 | *** join/#asterisk onats (n=julian@122.53.135.14) |
| 04:33.31 | onats | hi, i installed asterisk on a debian machine using synaptics. do i have to explicitly enable traffic through port 5060? i'm really having a hard time registering my ip soft phones... |
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| 04:49.52 | tzafrir_laptop | onats, UDP port 5060 |
| 04:49.59 | tzafrir_laptop | netstat -lnup |
| 04:50.17 | *** part/#asterisk techie (n=techie@c-67-181-184-170.hsd1.ca.comcast.net) |
| 04:53.12 | onats | tzafrir, how do i enable it/open it? |
| 04:53.21 | onats | netstat -lnup shows 0.0.0.0:5060 |
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| 04:57.01 | variable_office | currently astersik dumps all the voicemail into spool with only root permissions, is there a way i can set it to do it with permissions of another user or give it 777 permissions/ |
| 04:57.25 | *** join/#asterisk L|NUX (n=linux@unaffiliated/lnux/x-10290) |
| 04:59.56 | *** join/#asterisk Mavvie (n=edwin@ppp59-167-20-105.lns2.syd7.internode.on.net) |
| 05:05.44 | Buhntz | dunno if asterisk supports it, but you could make a cronjob |
| 05:07.15 | boch | do you know why im getting cause 16 in hangup events trough AMI when the real cause is no answer ? |
| 05:07.39 | JT | boch: you really aren't giving us enough info to work on |
| 05:08.15 | *** join/#asterisk yotta- (i=omECd1UE@xen.yotta.org) |
| 05:08.51 | yotta- | hello |
| 05:09.41 | onats | tzafrir? |
| 05:10.41 | boch | JT: I have an script making calls trough AMI using originate command and the hangup events says the calls that arent answered are terminated ok (16- normal clearing) |
| 05:11.10 | JT | make calls to what? |
| 05:11.59 | boch | another peers |
| 05:12.25 | *** join/#asterisk robh71 (n=robh71@host-65-124-86-25.entouch.net) |
| 05:12.34 | onats | guys, does anyone know how to open up port 5060 udp? |
| 05:12.36 | boch | i let the phone ring and the call ends with 16 code, normal like answered |
| 05:22.31 | tzafrir_laptop | onats, asterisk should be listening on it |
| 05:22.36 | tzafrir_laptop | is asterisk running? |
| 05:22.44 | onats | tzafrir, yes.. i just run asterisk |
| 05:23.00 | tzafrir_laptop | can you connect to it with rasterisk ? |
| 05:24.20 | onats | yes i can... |
| 05:24.24 | onats | currently have console.. |
| 05:24.53 | onats | netstat -lnup says 0.0.0.0:5060.. and asterisk is listening.. |
| 05:25.03 | onats | shouldn't that have an ip address instead of all 0s? |
| 05:26.28 | yotta- | I'd like to set up an extension in asterisk that rings my cell phone, but if I don't answer, I'd like to handle voice mail on asterisk, rather then let it go to the voice mail at my cell provider. |
| 05:26.30 | onats | my other linux machine, when connecting, it gives an error that 5060 is being used... |
| 05:26.43 | yotta- | Is there a reasonable way to do that? |
| 05:28.02 | *** part/#asterisk sysreq (n=sysreq@modemcable064.127-81-70.mc.videotron.ca) |
| 05:30.08 | fujin | onats: 0.0.0.0 = all interfaces |
| 05:30.34 | onats | fujin, aw... thanks for the info.. |
| 05:30.43 | fujin | yotta-: sure, just check ${DIALSTATUS} after you dial. althouhg, I'm not sure you'll be able to stop your cellphone from picking up and going to answerphone |
| 05:30.51 | onats | i wonder why my phones can't register... |
| 05:31.05 | yotta- | fujin: well, that's what I'm trying to figure out... |
| 05:31.15 | yotta- | how to avoid it going to the answering service. |
| 05:31.20 | tzafrir_laptop | onats, 0.0.0.0 means: listening on all IP addresses |
| 05:31.27 | tzafrir_laptop | bound to all interfaces |
| 05:32.11 | heelios | yotta-: usually voicemail on cellphones is just a number it transfers to if you dont pickup. if you have an available number, you could just set it in your phone and you could pickup the call from there. dunno if im clear. <_< |
| 05:32.14 | onats | even a locally installed softphone on the server can't register... |
| 05:32.19 | tzafrir_laptop | onats, next thing: use 'sip debug' |
| 05:32.42 | tzafrir_laptop | do you see much spam from you phone when it tries to connect? |
| 05:32.51 | tzafrir_laptop | to disable: sip no debug |
| 05:32.56 | *** join/#asterisk sysreq (n=sysreq@modemcable064.127-81-70.mc.videotron.ca) |
| 05:32.56 | yotta- | (this is AT&T in the US if it matters) |
| 05:33.03 | onats | 1 sec... |
| 05:33.25 | yotta- | yeah, but then I have to set busy forward on my phone to somewhere else |
| 05:33.56 | heelios | yotta-: its the only way I can figure thatd work. |
| 05:34.05 | yotta- | hmm |
| 05:34.13 | fujin | yotta-: ask your cellphone provider if there is a way to disable voicemail |
| 05:34.17 | onats | tzafrir, wow it connected now.. i must've had missed something.. |
| 05:34.21 | onats | lol |
| 05:34.24 | heelios | yotta-: besides, a DID would be proably cheaper than your carrier's voicemail service anyhow. |
| 05:34.25 | JT | boch: "another peerS" - how informative |
| 05:34.28 | onats | tzafrir, thanks for the help! |
| 05:34.38 | onats | will be playing with it again now |
| 05:34.41 | fujin | yotta-: you could just dial for a set amount of seconds (i.e; 10) and make sure that your cellphone online picks up voicemail after longer than 10 seconds |
| 05:34.46 | yotta- | heelios: cell service is paid for by my emplayer |
| 05:34.51 | fujin | and otherwise drops back to voicemail on * |
| 05:35.02 | yotta- | fujin: I thought of that |
| 05:35.10 | yotta- | but if my phone is off it goes straight to VM. |
| 05:35.32 | yotta- | hmm |
| 05:35.44 | heelios | yotta-: if you have a really nice employer they might go along with your scheme and get you a DID instead of voicemail? :P |
| 05:35.45 | yotta- | would it be possible to require at least one ring and the time out after a while? |
| 05:36.30 | boch | JT, another SIP peers sorry, i also see the cancel request and the 487 answer from ata |
| 05:36.54 | yotta- | I _COULD_ do that for VM, but |
| 05:37.01 | fujin | run SIP on your phone |
| 05:37.04 | fujin | that's what I do :[ |
| 05:37.06 | JT | boch: a SIP peer, i see |
| 05:37.09 | yotta- | i'd like it to go to cell phone vm if my phone is called directly. |
| 05:37.27 | yotta- | fujin: no data plan |
| 05:37.30 | fujin | anyone know about the IN USE detection of Local channels? does it work, how does it work? |
| 05:37.32 | fujin | get a data plan -_- |
| 05:37.48 | yotta- | :/ |
| 05:38.03 | yotta- | I could just use an IPKall did |
| 05:38.09 | yotta- | but |
| 05:38.13 | yotta- | erg. |
| 05:38.32 | yotta- | I still want direct cell phone calls to go to AT&T vm |
| 05:38.53 | onats | one more question, only one ip phone can run on each machine right? |
| 05:39.22 | yotta- | I could patch asterisk to allow a DTMF to boot out of the Dial() app with a code that is special. |
| 05:39.30 | yotta- | but that would be a pain. |
| 05:39.47 | fujin | onats: no, incorrect |
| 05:40.07 | fujin | multiple IP phones can run on each machine, providing they use different logins and bind to differnt port ranges |
| 05:40.21 | fujin | yotta-: probably wouldn't require a patch at all |
| 05:40.23 | boch | JT, if the call is generated using Dial() without any timeout from the dialplan instead AMI originate, the call ends with cause 19- noanswer; But if some timeout is passed to Dial() the call ends with cause 0 when should be noanswer |
| 05:40.39 | onats | fujin, ah... there.. i think its both binding to 5060... |
| 05:40.41 | yotta- | fujin: how could i do it without a patch? |
| 05:41.13 | fujin | some dialplan logic should suffice |
| 05:41.15 | fujin | what are you trying to do? |
| 05:42.03 | fujin | I'm not going to spoon feed you; I'm most certain that what you're trying to do has been done. |
| 05:43.04 | yotta- | ok, well, I want some way to detect that the call has been answered by AT&T VM |
| 05:43.11 | yotta- | and instead transfer to VM on * |
| 05:43.30 | yotta- | when my cell is called directly, I'd like it to behave normaly. |
| 05:44.43 | *** join/#asterisk oej (n=olle@soll4-125.cust.blixtvik.net) |
| 05:45.53 | onats | is it ok for all phones to register to asterisk using 5060? |
| 05:46.00 | onats | i mean, they are using only one port? |
| 05:46.22 | heelios | onats: yes. why not? |
| 05:46.46 | onats | heelios, just asking... im trying to get another phone to connect |
| 05:47.03 | onats | the softphone is on the same machine as the asterisk server |
| 05:47.35 | heelios | onats: oh. i misread. it's not okay if they both try to listen on the same port. |
| 05:47.58 | onats | heelios, you mean the phone and the server right? |
| 05:48.03 | heelios | onats: yes. |
| 05:48.06 | onats | ok.. |
| 05:48.22 | heelios | onats: any two applications as a matter of fact. |
| 05:48.40 | mDuff | onats: it's like Apache; you have one apache daemon listening on port 80, and any number of www clients can connect to it. |
| 05:48.44 | onats | how do i test a callback on one phone? i mean, i just want it to ring.. dial its own extension #? |
| 05:48.54 | mDuff | onats: however, you can't have both Apache and another service listening on that port at the same time. |
| 05:48.55 | onats | mDuff, that's a good analogy.. understood |
| 05:49.08 | onats | thanks |
| 05:50.44 | onats | does anyone have a good tutorial/pdf? |
| 05:51.00 | onats | something i can start with... |
| 05:51.10 | heelios | onats: download asterisk - the future of telephony |
| 05:51.31 | onats | i have that.. is that a good guide? |
| 05:51.37 | heelios | onats: yeah. |
| 05:51.48 | onats | 503 Service Unavailable<--- what does this mean? |
| 05:53.48 | heelios | onats: exactly what it says. no one can debug this without some more info. and with that, im going to bed. |
| 05:55.38 | phix | Gey |
| 05:55.40 | phix | hey |
| 05:56.47 | phix | I have changed some hardware in my server and the g.729 codec licence is no longer valid. I would also like to purchase an additional licence. How should I go about this? order another licence first or revalidate my original one? |
| 05:57.03 | onats | hehehe |
| 05:57.05 | onats | sorry |
| 05:57.06 | onats | newbie |
| 05:57.21 | phix | ok |
| 05:57.55 | *** join/#asterisk robh71 (n=robh71@host-65-124-86-25.entouch.net) |
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| 06:23.06 | *** join/#asterisk AirCoder (n=Aircoder@adsl-69-237-145-213.dsl.irvnca.pacbell.net) |
| 06:23.36 | AirCoder | hello all. |
| 06:25.32 | *** join/#asterisk deegan (i=deegan@killer.coding.ninja.monkii.net) |
| 06:28.11 | AirCoder | any one have nvfax and rx tx fax working in 1.4.11? |
| 06:29.31 | yotta- | anyone know what needs to be done to get the privacy creener thing to work under debian? |
| 06:29.36 | yotta- | there's some permissions issue |
| 06:29.45 | *** join/#asterisk aces234 (n=aces234@ip70-173-52-152.lv.lv.cox.net) |
| 06:30.33 | aces234 | anyone here that runs an asterisk consulation business i would like to ask some advice from about starting my own asterisk consultation business. |
| 06:30.50 | many | haha |
| 06:31.12 | phix | so any ideas? |
| 06:32.41 | *** part/#asterisk dw (n=dw@unaffiliated/dw) |
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| 06:34.07 | AirCoder | so talkative! |
| 06:35.05 | mvanbaak | AirCoder: most people are sleeping now |
| 06:35.15 | AirCoder | people sleep? |
| 06:35.17 | AirCoder | no wy |
| 06:35.20 | AirCoder | err way |
| 06:35.33 | AirCoder | guess programmers are a rare breed. |
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| 06:46.14 | tzafrir | AirCoder, people sleep. That is not to say programmers sleep |
| 06:46.39 | pkunkra | well, considering tomorrow is the first day of work, programmers are most likely sleeping but also they're recovering from a weekend of drinking. |
| 06:46.52 | AirCoder | lol |
| 06:47.09 | AirCoder | im tanking up on coffie to recover. |
| 06:48.00 | AirCoder | first day of work? you mean there are people that done work on sunday? |
| 06:48.16 | AirCoder | wow im out of the loop |
| 06:51.47 | phix | hmmmmmm |
| 06:52.57 | *** join/#asterisk marexz (n=marexz@marexz.mil.lv) |
| 06:54.13 | AirCoder | any one have nvfax and rx tx fax working in 1.4.11? |
| 06:54.39 | AirCoder | I have it compiled but nvfax hangs on detection on 2 sip connections ive tested.. |
| 06:56.53 | matt_ | can i use a single sip entry for lots of SIP accounts ? |
| 06:57.10 | matt_ | like have a single sip entry and then have a list of number:passwd pairs |
| 06:57.13 | matt_ | maybe in a database |
| 06:57.17 | *** join/#asterisk Avalone (n=Avalone_@mail.kawkazrg.ru) |
| 06:57.43 | AirCoder | i beleive the registration is per trunk. |
| 06:58.11 | AirCoder | i know you can have mutiple proxys per trunk, but have never seen mutiple registrations. |
| 06:58.36 | *** join/#asterisk Aurs (n=Aurs@1ult2p8.ip.hipercom.no) |
| 06:58.37 | matt_ | per .. trunk ? |
| 06:58.52 | matt_ | this wouldn't involve trunks |
| 06:58.56 | AirCoder | each sip account that you register is techincaly called a trunk. |
| 06:59.01 | JT | no |
| 06:59.08 | JT | a sip account is not a trunk |
| 06:59.44 | matt_ | i was just thinking about large setups with 1000's of remote sip devices |
| 06:59.57 | AirCoder | maby im misunderstanding whats being asked. |
| 07:00.02 | matt_ | they would be one large sip file and there must be an easier way |
| 07:01.03 | AirCoder | ohhh sip devices. |
| 07:01.27 | AirCoder | ignore my previous then i misunderstood what you were asking. |
| 07:02.26 | matt_ | http://www.voip-info.org/wiki/index.php?page=Asterisk+sip+autocreatepeer |
| 07:02.31 | matt_ | like that |
| 07:02.48 | matt_ | but with authentication |
| 07:03.22 | matt_ | but it would be good if ou could have like groups |
| 07:03.35 | matt_ | so you can say [group1options] |
| 07:03.53 | matt_ | and then with a module you can have a database with user:passwd:groupname |
| 07:04.22 | matt_ | so you can have different devices but easily add devices without having large config files, this must be possiable |
| 07:06.25 | AirCoder | anything is posible. |
| 07:06.33 | matt_ | lol yea |
| 07:06.45 | matt_ | but i wont be able to program something like that |
| 07:07.06 | AirCoder | dont know of anything that exists but it could be. |
| 07:10.53 | matt_ | The Asterisk external configuration engine is the result of work by Anthony Minessale II, Mark Spencer, and Constantine Filin. It is designed to provide a flexible, seamless integration between Asterisk's internal configuration structure and external SQL databases (maybe even LDAP one day). |
| 07:10.55 | matt_ | :D |
| 07:12.49 | matt_ | AirCoder, i was thinking about that but it seems a little sloppy |
| 07:14.35 | matt_ | although using a database directly might cause more problems with database connections die'ing |
| 07:16.48 | *** join/#asterisk IgorG (n=FeedomPa@195.162.32.126) |
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| 07:30.55 | aces234 | matt you head of adhearsion? |
| 07:31.02 | aces234 | heard? |
| 07:31.26 | *** join/#asterisk qdk (n=qdk@213.150.62.32) |
| 07:32.20 | yotta- | is there any way to have DISA use a sound file instead of a dialtone? |
| 07:40.08 | tzafrir | Background? |
| 07:44.10 | AirCoder | any one get nvfaxdetect working on 1.4.11? I have it compiled but nvfax hangs on detection on 2 sip connections ive tested.. |
| 07:49.16 | *** join/#asterisk psk (n=psk@golia.caltanet.it) |
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| 07:52.42 | jhlee | considering asterisk server on collocation mainly as conferencing server. without zaptel hardware will MeetMe works good enough? |
| 07:54.02 | JT | probably not |
| 07:54.56 | jhlee | JT: any advise for conferencing? |
| 07:55.12 | JT | don't use MeetMe? |
| 07:55.39 | jhlee | i see. what's alternative? |
| 07:57.24 | JT | app_conference |
| 07:59.04 | *** join/#asterisk uwe (n=uwe@dogbert.palnet.com) |
| 08:00.15 | uwe | hello, im getting a lot of "acl.c: 255.255.255.0,0.0.0.0/0.0.0.0 is not a valid netmask" in the full log, but i have no idea what is generating it ? this comes with low voice quality , any idea what it could be ? |
| 08:00.47 | jhlee | JT:thx, there sourceforge page doesn't have anything in download section. do you have any idea to get the source or pkg? |
| 08:13.38 | jhlee | Jt: thank for you help. gotta leave now |
| 08:13.42 | *** part/#asterisk jhlee (n=jhlee111@ip68-2-164-102.ph.ph.cox.net) |
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| 08:23.56 | Renacor | anybody know if the Digium TE410P is 64 bit or 32bit pci? |
| 08:53.31 | *** join/#asterisk marexz (n=marexz@marexz.mil.lv) |
| 08:55.58 | JT | Renacor: 32bit would be my bet |
| 08:59.33 | Renacor | thanks |
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| 09:48.22 | Rsaman | hello all |
| 09:48.50 | Rsaman | how do i check if i have the correct module compiled ? Specifically cdr pgsql |
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| 10:06.35 | Rsaman | ? |
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| 10:16.44 | aikanaro79 | hi, if I put the following priorities in my dialplan will a call get to the conference app? 100,1,Answer() // 100,n,Dial(SIP/xpto) // 100,n,Conference(1234/MTV) |
| 10:19.23 | taupin974 | after the Dial(SIP/xpto)? |
| 10:21.04 | *** join/#asterisk Woifi1988 (n=anon@M1226P019.adsl.highway.telekom.at) |
| 10:21.06 | *** join/#asterisk saftsack (n=oliver@p54A7D419.dip.t-dialin.net) |
| 10:21.37 | Woifi1988 | hi! |
| 10:22.19 | Woifi1988 | i have a problem with compiling zaptel for asterisk! |
| 10:22.25 | Woifi1988 | can somebody help me? |
| 10:22.53 | *** part/#asterisk vlrk (n=vlrk@dns1.muppidis.com) |
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| 10:26.23 | *** join/#asterisk RsaMan2 (n=aa@196.210.154.3) |
| 10:26.25 | RsaMan2 | arrrg |
| 10:26.36 | RsaMan2 | My blind transfer is still messed up |
| 10:26.54 | RsaMan2 | tried to many combos |
| 10:27.00 | RsaMan2 | and no success |
| 10:27.18 | Woifi1988 | please |
| 10:27.26 | RsaMan2 | anyone familiar with transfering calls? |
| 10:27.45 | RsaMan2 | both parties get dropped, after transfer |
| 10:28.29 | RsaMan2 | does anyone know how to set a transfer rule in my dialplan, i cant rely on blind transfer because it is not working for me |
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| 10:29.00 | uwe | Woifi1988, state what the problem is , maybe this will remind someone with something |
| 10:29.22 | RsaMan2 | :( |
| 10:29.47 | RsaMan2 | i have had my problem for over a week, tried a million things, posted on the forum, but i still seem to be missing something |
| 10:30.20 | *** join/#asterisk guillote_GNU (n=guillote@host39.190-30-65.telecom.net.ar) |
| 10:31.47 | Woifi1988 | I try to do a make clean --> i get two errors but i don't mind. the ./configure worked, but with make i get the error, that i have no sources for linux-2.6.15-26-server installed, but i have installed these sources and i also created a softlink at /usr/src. I looked in some forums, but I didn't find any solution :( |
| 10:32.06 | *** join/#asterisk HaYZaM (n=helghara@62.117.45.169) |
| 10:32.44 | HaYZaM | can i use asterisk as a video conferencing tool if the cameras support SIP protocol ? |
| 10:33.56 | HaYZaM | heey you asterisk ppl |
| 10:36.14 | HaYZaM | gatko steen nela 3ala demaghko |
| 10:36.47 | RsaMan2 | ? |
| 10:38.08 | HaYZaM | can i use asterisk as a video conferencing tool if the cameras support SIP protocol ? |
| 10:38.12 | HaYZaM | can i use asterisk as a video conferencing tool if the cameras support SIP protocol ? |
| 10:38.13 | HaYZaM | can i use asterisk as a video conferencing tool if the cameras support SIP protocol ? |
| 10:38.19 | many | stop it. |
| 10:38.21 | RsaMan2 | piss off |
| 10:38.30 | HaYZaM | u r sleeping |
| 10:38.34 | RsaMan2 | the answer is : maybe |
| 10:38.40 | HaYZaM | ZzzZzzZZzz |
| 10:38.58 | HaYZaM | should i flood to answer |
| 10:39.07 | RsaMan2 | yes |
| 10:39.08 | HaYZaM | why dont you answer from the begining |
| 10:39.18 | RsaMan2 | any more questions? |
| 10:39.31 | HaYZaM | yes , ok , i am gonna use this way everytime i need something |
| 10:39.31 | *** join/#asterisk mbit (n=nothing9@218-214-57-65.people.net.au) |
| 10:39.49 | many | i have one question, why is hayzam such a d.ck? |
| 10:39.56 | tzafrir | HaYZaM, then you're going to be banned from here |
| 10:40.03 | HaYZaM | coz u r a p.ssy |
| 10:40.55 | RsaMan2 | shhh |
| 10:42.32 | Woifi1988 | I try to do a make clean --> i get two errors but i don't mind. the ./configure worked, but with make i get the error, that i have no sources for linux-2.6.15-26-server installed, but i have installed these sources and i also created a softlink at /usr/src. I looked in some forums, but I didn't find any solution :( |
| 10:42.42 | *** part/#asterisk mbit (n=nothing9@218-214-57-65.people.net.au) |
| 10:44.53 | *** join/#asterisk saftsack (n=oliver@p54A7CEBD.dip.t-dialin.net) |
| 10:48.05 | Woifi1988 | please |
| 10:48.08 | matt_ | HaYZaM, if you flood channels on purpose you will probuly end up getting klined |
| 10:48.20 | *** join/#asterisk michael-i (n=michael-@141.41.40.55) |
| 10:50.57 | RsaMan2 | is there anyone who knows how to setup a dialplan to transfer calls? |
| 10:51.16 | RsaMan2 | i am pretty lost, my # blind transfer drops both ends |
| 10:51.17 | RsaMan2 | :9 |
| 10:51.32 | HaYZaM | i have a useful PDF |
| 10:51.34 | *** join/#asterisk darkskiez (n=mbryars@195-11-205-216.suip.mezzonet.net) |
| 10:52.54 | RsaMan2 | called? |
| 10:53.27 | RsaMan2 | ~thebook |
| 10:53.27 | jbot | methinks thebook is a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 10:53.29 | RsaMan2 | ? |
| 10:54.42 | darkskiez | I'd really like a couple of patches that have been posted (bug 8824) but aren't available on the bug tracker to download, is there any way to contact the contributor off list, or bug them to do what they need to do to get it available ? |
| 11:02.42 | Woifi1988 | I try to do a make clean --> i get two errors but i don't mind. the ./configure worked, but with make i get the error, that i have no sources for linux-2.6.15-26-server installed, but i have installed these sources and i also created a softlink at /usr/src. I looked in some forums, but I didn't find any solution :( |
| 11:02.44 | Woifi1988 | please |
| 11:03.02 | lirakis | you need to symlink |
| 11:03.06 | lirakis | /usr/src/linux |
| 11:03.11 | lirakis | to the directory where your sources are |
| 11:03.34 | lirakis | ln -s /usr/src/linux-2.6.15-26-server /usr/src/linux |
| 11:03.52 | lirakis | make sure your link is correct |
| 11:04.01 | lirakis | ls -alh /usr/src |
| 11:04.07 | lirakis | will show you where your link is pointing |
| 11:06.59 | JT | HaYZaM: shutup moronic flooding idiot |
| 11:07.47 | Woifi1988 | lirakis the link should be named linux? |
| 11:07.52 | Woifi1988 | not linux-2.6 |
| 11:07.59 | matt_ | Woifi1988, yes |
| 11:08.04 | Woifi1988 | ohh |
| 11:08.06 | *** join/#asterisk nighty^ (n=nighty@sushi.rural-networks.com) |
| 11:08.09 | lirakis | Woifi1988: ;) |
| 11:08.20 | Woifi1988 | every tutorial say linux-2.6 |
| 11:08.28 | lirakis | Woifi1988: i doubt it |
| 11:08.30 | matt_ | lol k |
| 11:10.03 | tzafrir | no, a symlink should not be needed at all |
| 11:10.13 | Woifi1988 | but i get the same error :-( |
| 11:10.26 | tzafrir | Woifi1988, apt-get install linux-headers-`uname -r` |
| 11:10.36 | Woifi1988 | i have done this |
| 11:10.44 | Woifi1988 | i'll do it twice |
| 11:11.22 | tzafrir | What error do you get? |
| 11:11.39 | *** join/#asterisk DrAk0 (n=ljd@nelug/coreteam/luisjose) |
| 11:11.50 | Woifi1988 | You do not appear to have the sources for th 2.6.15-server installed |
| 11:12.00 | *** join/#asterisk zotz (n=zotz@24.244.163.157) |
| 11:12.24 | tzafrir | ls -l /lib/modules/`uname -r`/build/.config |
| 11:12.31 | michael-i | Quick question: I'm loading all of the zaptel modules so I can detect cards present in the system but this seems to be messing up my timing with ztdummy even if no cards were found. Are these two timing sources mutually exclusive or do they interfere even by being loaded? |
| 11:12.34 | tzafrir | what is the output / error? |
| 11:12.54 | tzafrir | michael-i, if you have a card you don't need ztdummy |
| 11:12.55 | lirakis | l8r all off to wrok |
| 11:13.03 | Woifi1988 | one moment it seems to work now |
| 11:13.04 | lirakis | s/wrok/work |
| 11:13.07 | Woifi1988 | it compiles |
| 11:13.23 | *** part/#asterisk lirakis (n=lirakis@cpe-68-175-38-65.nyc.res.rr.com) |
| 11:13.34 | Woifi1988 | maybe i have had the wrong headers! |
| 11:13.47 | michael-i | tzafrir, yes, of course but I'm saying that with no card present, simply loading the zaptel modules for detection purposes seems to interfere with ztdummy |
| 11:13.49 | Woifi1988 | because i didn't downloaded hem with ùname -r |
| 11:13.50 | Woifi1988 | ` |
| 11:14.04 | matt_ | Woifi1988, paste config.log |
| 11:14.08 | matt_ | not here |
| 11:14.10 | matt_ | to pastebin |
| 11:14.53 | Woifi1988 | the make did more but has also some errors |
| 11:15.00 | Woifi1988 | where is pastebin? |
| 11:15.02 | Woifi1988 | a channel? |
| 11:15.10 | matt_ | google |
| 11:15.38 | michael-i | http://pastebin.ca/ |
| 11:15.44 | Woifi1988 | ok ;-) one moment |
| 11:16.27 | tzafrir | michael-i, in what way does it interfere with ztdummy? |
| 11:18.41 | michael-i | tzafrir, people using AskoziaPBX have reported choppy, robot-like sound on channels needing transcoding / moh / conferencing after I added analog support to the latest version. Since that was one of the only major changes, I think it is my problem. |
| 11:18.43 | *** join/#asterisk blinky42 (n=me@67.106.175.130.ptr.us.xo.net) |
| 11:19.50 | Woifi1988 | i just need a moment |
| 11:19.55 | Woifi1988 | my virtual machine strikes |
| 11:21.21 | tzafrir | michael-i, ah, that's freebsd. |
| 11:22.04 | tzafrir | Sorry, but I don't know the low-level issues of zaptel there well |
| 11:22.29 | *** join/#asterisk clandmeter (n=Carlo@81.175.82.2) |
| 11:22.31 | michael-i | tzafrir, yes :) i'm also using a special ztdummy that was submitted to the asterisk-bsd list. In general, it's just not a good idea to have them both loaded, correct? |
| 11:22.56 | tzafrir | If you say so |
| 11:23.57 | michael-i | you were supposed to agree so I feel confident hacking in better detection! ;) I'll look into the code a bit more to see where / if any locking occurs between the timing sources |
| 11:24.40 | Woifi1988 | matt_ i can't get the config.h |
| 11:24.57 | Woifi1988 | i can only give you some lines |
| 11:25.18 | tzafrir | michael-i, I really don't know FreeBSD, but if a driver has any overhead without a hardware, it's a bug |
| 11:25.27 | tzafrir | (a driver for a card) |
| 11:29.06 | matt_ | config.log |
| 11:29.31 | matt_ | ./configure scripts create a config.log file that says what they attemped todo |
| 11:31.03 | RsaMan2 | how do i check if all is installed to connect cdr to a postgres database? |
| 11:31.43 | RsaMan2 | do i only the cdr pgsql module |
| 11:31.47 | RsaMan2 | or must i install odbc? |
| 11:33.07 | Woifi1988 | matt_ --> http://pastebin.ca/672207 |
| 11:33.36 | RsaMan2 | any ideas? |
| 11:34.04 | matt_ | # |
| 11:34.04 | matt_ | configure:4974: gcc -o conftest -g -O2 conftest.c -lusb >&5 |
| 11:34.04 | matt_ | # |
| 11:34.04 | matt_ | /usr/bin/ld: cannot find -lusb |
| 11:34.16 | matt_ | its looking for libusb |
| 11:34.38 | matt_ | Woifi1988, http://libusb.sourceforge.net/ |
| 11:35.17 | Woifi1988 | i thougt libusb is only needed for kernel version 2.4?! |
| 11:35.55 | matt_ | where did you read that? |
| 11:35.59 | *** join/#asterisk Dr-Linux (n=Nothing@202.125.139.198) |
| 11:36.08 | matt_ | 2.6 has usb support :) |
| 11:36.10 | Woifi1988 | in my asterisk book |
| 11:36.15 | *** join/#asterisk mitcheloc (n=mitchel@adsl-67-126-140-84.dsl.irvnca.pacbell.net) |
| 11:36.15 | Woifi1988 | okay |
| 11:36.19 | matt_ | anyway thats what that config script is telling me |
| 11:36.30 | Woifi1988 | i can't install this via apt-get |
| 11:36.32 | Woifi1988 | ? |
| 11:36.37 | matt_ | so i guess you can either install it or look at ./configure --help and see if you can take out usb support |
| 11:36.46 | matt_ | Woifi1988, yea sure |
| 11:37.15 | Woifi1988 | and where can i anable the ztdummy support? |
| 11:37.31 | matt_ | i have no idea |
| 11:37.51 | matt_ | i dont even know what ztdummy is lol |
| 11:37.51 | tzafrir | matt_, ./build_tools/install_prereq test |
| 11:38.13 | tzafrir | matt_, you just need ztdummy? |
| 11:38.24 | matt_ | no |
| 11:38.31 | matt_ | my askterisk setup works fine :) |
| 11:38.32 | Woifi1988 | ztdummy is a tool which provides clockrate for devices which have no hardware timer |
| 11:38.39 | matt_ | ahh ok |
| 11:38.46 | matt_ | ive never needed that |
| 11:39.02 | Dr-Linux | matt_: know C ? :) |
| 11:39.16 | tzafrir | libusb is needed for the firmware loader of a certain Zaptel device |
| 11:39.22 | matt_ | Dr-Linux, no |
| 11:39.24 | tzafrir | see the README |
| 11:39.35 | RsaMan2 | arg |
| 11:39.39 | Woifi1988 | what is the apt-get comman for libusb? |
| 11:39.44 | RsaMan2 | i am stuck with my same issues |
| 11:39.55 | RsaMan2 | cannot connect cdr to my postgres database |
| 11:39.56 | Dr-Linux | tzafrir: i wanna use argument /n here but where should i use in this below line: |
| 11:39.57 | Dr-Linux | <PROTECTED> |
| 11:40.21 | Woifi1988 | what is the apt-get commanD for libusb? |
| 11:40.45 | tzafrir | Woifi1988, apt-get install libusb-dev |
| 11:40.52 | Woifi1988 | thx |
| 11:40.57 | tzafrir | ./build_tools/install_prereq test |
| 11:41.03 | Woifi1988 | and now a make clean and configure again? |
| 11:41.11 | RsaMan2 | is anyone using a postgres database? |
| 11:41.14 | tzafrir | why make clean? |
| 11:41.26 | RsaMan2 | to store cdr data? |
| 11:41.26 | RsaMan2 | do i need to install the odbc module ? |
| 11:41.32 | Woifi1988 | i have to recompile the package, have i? |
| 11:41.47 | tzafrir | make |
| 11:41.54 | tzafrir | or: ./configure; make |
| 11:42.08 | Woifi1988 | whats up with the build_tools? |
| 11:42.21 | tzafrir | you asked how to install |
| 11:42.23 | tzafrir | try it |
| 11:42.28 | tzafrir | that was the answer |
| 11:42.53 | Woifi1988 | it showes test completed successfully |
| 11:42.59 | tzafrir | with "test" it prints the command, with "install", it runs the command |
| 11:43.23 | Woifi1988 | apt-get install -y |
| 11:43.26 | Woifi1988 | thats all |
| 11:44.01 | tzafrir | so everything is already installed |
| 11:44.25 | Woifi1988 | but i have an error again when i do make |
| 11:45.02 | Woifi1988 | compilation aborted at zt_registration on line 11 |
| 11:46.41 | tzafrir | what version of zaptel is that? |
| 11:46.54 | tzafrir | what distro? |
| 11:47.44 | *** join/#asterisk KDan (n=KDan@87-194-122-30.bethere.co.uk) |
| 11:47.48 | KDan | is there any way to use Asterisk to send SMS's for quasi-free? Or so you *have* to use an SMS gateway service that charges per SMS? |
| 11:48.08 | tzafrir | Woifi1988, what is the line right above the "compilation error"? missing module? |
| 11:48.51 | tzafrir | I suspect that this will be solved if you install "perl-modules". But this is slightly non-elegant |
| 11:50.59 | *** join/#asterisk DrAk0 (n=ljd@nelug/coreteam/luisjose) |
| 11:51.22 | tzanger | morning |
| 11:54.03 | Woifi1988 | aahm |
| 11:54.16 | Woifi1988 | zaptel 1.4.5.1 and ubuntu 6.06.1 server |
| 11:54.55 | tzafrir | KDan, european telcos? |
| 11:55.18 | Woifi1988 | tzanger no missing module |
| 11:55.25 | tzanger | good to hear |
| 11:55.30 | tzanger | but I think you mean tzafrir |
| 11:55.41 | Woifi1988 | make[2]: *** [pearlchek] Error 1 |
| 11:55.48 | Woifi1988 | yes sry ;-) |
| 11:56.08 | Woifi1988 | make[1]: *** [xpp-utils] Error 2 |
| 11:57.25 | *** join/#asterisk lirakis (n=etamme@65.200.191.253) |
| 11:57.42 | tzafrir | edit xpp/utils, look for all: , and remove "perlcheck" from that line |
| 11:57.54 | tzafrir | You could live without it |
| 11:58.08 | tzafrir | or install perl-modules |
| 11:58.20 | Woifi1988 | what does it do? |
| 11:58.54 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 11:59.36 | tzafrir | extra perl modules. The default Debian and Ubuntu come with just a package called "perl-base", which has a rather limited set of modules. |
| 11:59.45 | tzafrir | Good enough for your daily server needs |
| 12:00.07 | Woifi1988 | is it nessasary to make a compley dial-plan with api? |
| 12:00.11 | tzafrir | The "regular" perl package (perl + perl-modules) is larger |
| 12:00.16 | tzafrir | not at all |
| 12:00.19 | *** join/#asterisk steliosk (n=Stelios@ipa226.211.tellas.gr) |
| 12:00.19 | Woifi1988 | okay |
| 12:00.57 | *** join/#asterisk jarod14 (n=jarod14@mail.viatelecom.com) |
| 12:01.16 | *** join/#asterisk Vorondil (n=vorondil@unaffiliated/vorondil) |
| 12:01.17 | Woifi1988 | tzafrir i do a apt-get install pearl* |
| 12:01.30 | tzafrir | perl (no A) |
| 12:01.45 | tzafrir | apt-get install perl-modules |
| 12:02.15 | tzafrir | I think that a simple 'apt-get install perl' will do the same. |
| 12:04.36 | Woifi1988 | ok it compiles again |
| 12:04.42 | *** join/#asterisk caio1982 (i=caio1982@CAcert-br/caio1982) |
| 12:04.55 | *** part/#asterisk caio1982 (i=caio1982@CAcert-br/caio1982) |
| 12:05.14 | *** join/#asterisk CrazyTux[m] (n=CrazyTux@216-110-94-230.static.twtelecom.net) |
| 12:05.56 | Woifi1988 | tzafrir it's ready.. there is now error- but also no success message! |
| 12:06.06 | Woifi1988 | how can i improve that it works? |
| 12:06.12 | tzafrir | what errors? |
| 12:06.28 | *** join/#asterisk coppice (n=chatzill@54.197.17.210.dyn.pacific.net.hk) |
| 12:06.55 | Woifi1988 | now errors |
| 12:06.58 | Woifi1988 | no errors |
| 12:07.06 | Woifi1988 | i'm confused |
| 12:07.18 | Woifi1988 | there are no errors |
| 12:07.26 | Woifi1988 | but also no success messages |
| 12:09.30 | RsaMan2 | do i need cdr odbc for the cdr pgsql module |
| 12:10.42 | Woifi1988 | ok make install was success |
| 12:10.48 | RsaMan2 | ? |
| 12:10.58 | Woifi1988 | thx very much tzafrir |
| 12:11.08 | Woifi1988 | sorry Rs |
| 12:11.19 | Woifi1988 | sorry RsaMan2 i don't know |
| 12:11.49 | RsaMan2 | :( |
| 12:11.53 | RsaMan2 | i hate being stuck |
| 12:11.58 | RsaMan2 | soo stuck for over a week |
| 12:12.32 | Woifi1988 | yes mee too, but with zaptel |
| 12:12.39 | Woifi1988 | now it works :) |
| 12:14.01 | *** join/#asterisk shtoom (n=shtoom@123.252.144.92) |
| 12:14.20 | *** join/#asterisk [TK]D-Fender (n=joe@MTRLPQ02-1177745839.sdsl.bell.ca) |
| 12:16.22 | *** join/#asterisk tsurko (n=tsurko@78.90.100.214) |
| 12:17.54 | RsaMan2 | i have a bloody blind transer issue with no solution |
| 12:17.56 | RsaMan2 | :* |
| 12:18.24 | RsaMan2 | i am going to die soon |
| 12:18.24 | RsaMan2 | :) |
| 12:18.38 | coppice | we all are |
| 12:19.06 | RsaMan2 | i am gonna die sooner than most people |
| 12:19.10 | The_LightSide | RsaMan2 , i have seen something called __TRANSFER_CONTEXT not sure if it will help or not |
| 12:19.28 | The_LightSide | also having issues with blind transfers |
| 12:19.42 | coppice | RsaMan2: I like to see such gritty determination |
| 12:20.25 | *** join/#asterisk puzzled (n=patrick@puzzled.xs4all.nl) |
| 12:20.38 | puzzled | hi |
| 12:20.42 | hmmhesays | good lord itunes is a pain in the @$$ |
| 12:21.12 | DrAk0 | ~book |
| 12:21.12 | jbot | extra, extra, read all about it, book is a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 12:21.34 | [TK]D-Fender | "Life is a sexually transmitted disease, which is in all cases, fatal. If you're reading this now, you're already fucked." |
| 12:21.44 | hmmhesays | haha |
| 12:25.30 | *** join/#asterisk shay|work (n=shay@unaffiliated/shay) |
| 12:25.55 | tzafrir | RsaMan2, what exactly was your problem? wrong context or misdetected DTMF? |
| 12:29.00 | *** join/#asterisk DrAk0 (n=ljd@nelug/coreteam/luisjose) |
| 12:29.02 | RsaMan2 | erm |
| 12:29.03 | RsaMan2 | not sure |
| 12:29.11 | RsaMan2 | i compiled a new post for the forum |
| 12:29.18 | RsaMan2 | just need to double check my version |
| 12:29.18 | DrAk0 | is this book any good? http://www.asteriskguide.com/asterisk-configuration-guide-english1.html?gclid=CPH3mPzWlY4CFTyKOAodwlf8EA |
| 12:29.28 | RsaMan2 | how do i know what version of asterisk i am using |
| 12:29.29 | RsaMan2 | ? |
| 12:29.58 | DrAk0 | RsaMan2, asterisk -V |
| 12:30.14 | RsaMan2 | thanks |
| 12:30.14 | DrAk0 | RsaMan2, or `core show version` on the cli |
| 12:31.12 | RsaMan2 | thanks |
| 12:31.58 | RsaMan2 | i have tried to provide all the right info |
| 12:31.59 | RsaMan2 | http://forums.digium.com/viewtopic.php?p=56585#56585 |
| 12:32.07 | RsaMan2 | this is my problem with blind call transfer |
| 12:32.09 | RsaMan2 | in that post |
| 12:32.19 | Woifi1988 | i think a good book is www.das-asterisk-buch.de for people who speack german or asterisk - the future of telephony |
| 12:33.36 | Woifi1988 | When I installed zaptel - shouldn't it be listed in dpkg -l??? |
| 12:34.32 | tzafrir | Woifi1988, no. You have not installed it from a package |
| 12:34.55 | Woifi1988 | oh okay |
| 12:35.17 | DrAk0 | i need a book that covers 1.4 |
| 12:35.19 | Woifi1988 | is there any command to see all installed software and drivers? |
| 12:35.22 | Woifi1988 | Dr |
| 12:35.38 | Woifi1988 | DrAk0 Asterisk - The future of telephony |
| 12:35.44 | *** join/#asterisk guillote_GNU (n=bancaria@host185.190-136-203.telecom.net.ar) |
| 12:35.48 | RsaMan2 | gonna try upgrade my astersik |
| 12:36.01 | RsaMan2 | maybe that will resolve the issue |
| 12:36.10 | RsaMan2 | do i just download the lastest 1.4.* |
| 12:36.24 | RsaMan2 | then ./configure |
| 12:36.25 | RsaMan2 | make |
| 12:36.29 | RsaMan2 | make install |
| 12:36.32 | RsaMan2 | will this be safe ? |
| 12:37.59 | *** join/#asterisk pgarcia (n=pedgarci@c934ac83.virtua.com.br) |
| 12:38.56 | tzafrir | Woifi1988, package management commands work when you use packages |
| 12:39.03 | tzafrir | e.g: when you apt-get install zaptel |
| 12:39.08 | *** part/#asterisk pgarcia (n=pedgarci@c934ac83.virtua.com.br) |
| 12:39.12 | tzafrir | and asterisk |
| 12:39.52 | Woifi1988 | okay |
| 12:40.23 | Woifi1988 | but so i don't know what software and drivers are installed on the system |
| 12:42.19 | Woifi1988 | RsaMan2 try to do a make upgrade in the asterisk directory |
| 12:44.43 | *** join/#asterisk mitcheloc (n=mitchel@adsl-67-126-140-84.dsl.irvnca.pacbell.net) |
| 12:45.32 | [TK]D-Fender | RsaMan2: Do the entire normal install procedure EXCEPT for "make samples" and you'll be fine |
| 12:49.02 | *** join/#asterisk _ShrikE (n=ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
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| 12:51.15 | *** join/#asterisk SAL123 (i=ondrejj@work.salstar.sk) |
| 12:52.24 | SAL123 | Hello. Is there somebody who can help me with app_rxfax (fax receiving). I have configures asterisk with rx_fax, but there is only silence from my phone. |
| 12:59.54 | *** join/#asterisk Ryushin (i=proxy@windwalker.openinnovations.com) |
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| 13:05.41 | SAL123 | hmm, there is no live? :) |
| 13:06.52 | michael-i | ...probably just no one with app_rxfax experience and time at the moment ;) |
| 13:07.09 | tzafrir | SAL123, and your question isn't really clear |
| 13:07.35 | *** join/#asterisk robl^ (n=robl@pdpc/supporter/monthlybyte/robl) |
| 13:07.54 | tzafrir | rxfax expects faxes. It doesn't send them |
| 13:08.33 | *** join/#asterisk stoffell_w (n=stoffell@fw.catsanddogs.com) |
| 13:09.35 | The_LightSide | i have an issue where in the middle of a conversation, the call is dropped. from what i can see on the log files, it seems to be a normal release (sip to sip) any ideas of where i should start looking? * ver 1.2.19 |
| 13:09.38 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 13:09.49 | The_LightSide | sorry, 1.2.17 |
| 13:10.14 | SAL123 | I am calling to my fax number. I think I need to hear something similiar like from calling to modem. But there is no sound. |
| 13:10.33 | SAL123 | Call is holding with silence. |
| 13:12.43 | *** join/#asterisk florz (n=florz@2001:1a50:503c:0:0:0:0:1) |
| 13:14.25 | SAL123 | I see this in my log: |
| 13:14.26 | SAL123 | <PROTECTED> |
| 13:14.48 | SAL123 | but there is not /tmp/...tif file an no sounds from phone speaker. |
| 13:15.12 | RsaMan2 | arg |
| 13:15.15 | RsaMan2 | my asterisk wont update |
| 13:15.16 | RsaMan2 | chan_zap.c: In function ‘zt_call’: |
| 13:15.21 | RsaMan2 | chan_zap.c:2129: error: too few arguments to function ‘pri_sr_set_bearer’ |
| 13:15.21 | RsaMan2 | chan_zap.c: In function ‘zt_hangup’: |
| 13:15.21 | RsaMan2 | chan_zap.c:2596: error: too few arguments to function ‘pri_hangup’ |
| 13:15.21 | RsaMan2 | chan_zap.c:2616: error: too few arguments to function ‘pri_hangup’ |
| 13:15.21 | RsaMan2 | chan_zap.c: In function ‘zt_handle_event’: |
| 13:15.21 | RsaMan2 | chan_zap.c:3795: error: too few arguments to function ‘pri_hangup’ |
| 13:15.23 | RsaMan2 | chan_zap.c: In function ‘pri_dchannel’: |
| 13:15.25 | RsaMan2 | oops |
| 13:15.27 | RsaMan2 | sorry |
| 13:15.32 | RsaMan2 | that was meant for pastebin |
| 13:15.34 | RsaMan2 | my bad |
| 13:17.24 | file | upgrade libpri |
| 13:17.25 | *** join/#asterisk pgarcia (n=pedgarci@c934ac83.virtua.com.br) |
| 13:17.39 | RsaMan2 | is that my problem ? |
| 13:17.40 | [TK]D-Fender | SAL123: pastbin the FULL CLI output for your entire call. |
| 13:17.41 | [TK]D-Fender | ~pb |
| 13:17.42 | jbot | A Pastebin is a place to paste your stuff without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste |
| 13:17.44 | [TK]D-Fender | ^^^^^^^^^^^^^^^^^^^^^^ |
| 13:18.33 | RsaMan2 | think i have broken my asterisk ? |
| 13:18.39 | RsaMan2 | i can seem to update it now |
| 13:19.11 | SAL123 | [TK]D-Fender: http://pastebin.com/d40ac746 |
| 13:19.27 | *** join/#asterisk saftsack (n=oliver@p54A7E8A6.dip.t-dialin.net) |
| 13:19.29 | SAL123 | [TK]D-Fender: call is still in progress |
| 13:20.07 | [TK]D-Fender | Oh joy... Fax over VoIP...... |
| 13:20.52 | SAL123 | [TK]D-Fender: yes, it worked for me on asterisk-1.0 or 1.2, but after some upgrades I am unable to make ti work. :) |
| 13:21.25 | [TK]D-Fender | SAL123: rxfax/txfax are real problems on 1.4 |
| 13:21.40 | [TK]D-Fender | SAL123: Get Googling, this will not be easy I'm sure |
| 13:21.59 | SAL123 | [TK]D-Fender: Googling does not helped me. :( |
| 13:22.17 | SAL123 | [TK]D-Fender: Is there other solution to receive faxes over voip? |
| 13:22.30 | *** join/#asterisk waKKu (n=eurulo@unaffiliated/wakku) |
| 13:22.47 | x86 | SAL123: get a FoIP provider |
| 13:22.50 | tzafrir | RsaMan2, bristuffed vs. non-bristuffed? |
| 13:22.58 | *** join/#asterisk stkn|work (n=stkn|wor@gentoo/developer/pdpc.active.stkn) |
| 13:23.03 | x86 | tzafrir: bristuff always wins ;) |
| 13:23.05 | robl^ | SAL123: I just pay a fax service center $5 a month and have faxes delivered to email less hassle |
| 13:23.17 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 13:23.40 | x86 | robl^: do they support LNP? :) |
| 13:23.42 | tzafrir | bristuffed libpri and non-bristuffed asterisk? |
| 13:23.58 | robl^ | x86: LNP? |
| 13:24.33 | x86 | robl^: local number portability |
| 13:24.36 | lirakis | hmm... i have a call queue that seems to be working welll.. but what is weird is .. when the caller goes into the queue.. MOH starts playing fine.. then stops after like.. 3 or so seconds |
| 13:24.40 | lirakis | i dont know why |
| 13:24.47 | x86 | robl^: where i can use my existing DID with thier service |
| 13:24.59 | robl^ | x86: ahh! yeah. They do support it. |
| 13:25.06 | x86 | robl^: nice... |
| 13:25.19 | x86 | lirakis: do you have real timing? |
| 13:25.34 | lirakis | x86: .. not sure |
| 13:25.39 | lirakis | x86: how do i know? |
| 13:25.45 | x86 | do you have any zaptel hardware in the box? |
| 13:26.01 | robl^ | They are also a VoIP provider service. I have a block of 20 DIDs with them. One of that contiguous block is my fax |
| 13:26.15 | x86 | TDM2400P, TDM04B, etc |
| 13:26.23 | x86 | X100P ;) |
| 13:26.25 | lirakis | x86: i have a sangoma T1 card |
| 13:26.36 | x86 | lirakis: ok, then timing is not the issue... |
| 13:26.45 | x86 | lirakis: are you using MP3 or native files? |
| 13:27.07 | lirakis | x86: .. i beleive i set it to "files" .. but let me double check |
| 13:27.07 | robl^ | x100p?? I have 3 of those. They are usefull for fixing wobbley tables. shove one under a leg to keep it level. |
| 13:27.28 | x86 | lirakis: i've had the best luck using regular WAV files, or even converting them to ULAW |
| 13:27.28 | lirakis | x86: yes.. type=files |
| 13:27.38 | x86 | lirakis: type=native |
| 13:28.07 | lirakis | x86: hrmm.. yeah.. ive just been using the default sounds (mp3s) that came with * for the MOH |
| 13:28.07 | *** join/#asterisk bkw_ (n=brian@64.149.40.227) |
| 13:28.22 | RsaMan2 | tzafrir : i am not sure what u mean ? is the bristuff conflicting in some way? |
| 13:28.43 | x86 | lirakis: try type=native with the mp3's |
| 13:31.48 | lirakis | x86: ok ill give it a shot |
| 13:32.33 | [TK]D-Fender | x86: there is no "type=native" |
| 13:32.43 | [TK]D-Fender | x86: "type=files" = Native MoH |
| 13:33.06 | lirakis | x86: it does the same thing... plays about 2 seconds of music ... then cli says "STopped MOH" |
| 13:33.31 | lirakis | i think i may have done some thing stupid in my queues.conf .. :\ not sure |
| 13:34.05 | datachomper | What kinds of equipment do the ISTP's use to convert PSTN signals to sip traffic? |
| 13:35.03 | *** join/#asterisk Woifi1988 (n=anon@M1226P019.adsl.highway.telekom.at) |
| 13:35.21 | Woifi1988 | how can i start asterisk 1.4? |
| 13:35.26 | [TK]D-Fender | datachomper: AudioCodes Mediant, etc |
| 13:35.33 | Woifi1988 | there is no script in /etc/init.d |
| 13:35.39 | [TK]D-Fender | Woifi1988: "asterisk -gvvvvc |
| 13:35.40 | Woifi1988 | or something like this |
| 13:35.53 | [TK]D-Fender | Woifi1988: "make config" <- for scripts |
| 13:35.58 | Woifi1988 | gvvvvc? |
| 13:36.13 | Woifi1988 | what is the safe_asterisk option? |
| 13:36.37 | tzafrir | RsaMan2, I suspect that you have libpri from bristuff |
| 13:36.50 | datachomper | [TK]D-Fender, Thanks |
| 13:37.03 | tzafrir | try reinstalling original libpri. Or patch asterisk from bristuff |
| 13:37.05 | lirakis | sort of unrelated.... i set "announce = " files for annoucing to the agents the call type.. but those dont play.. and i get not cli output display saying it even tries to play any announcement |
| 13:37.19 | [TK]D-Fender | Woifi1988: Just to check if its FUNCTIONAL and aee the error if any. If it works, THEN its OK to stop & do "safe_asterisk &" |
| 13:38.37 | The_LightSide | i have noticed that safe_asterisk does not restart asterisk when it dies (1.4.4 svn branch) |
| 13:38.51 | Woifi1988 | okay! why there are so many v when you start asterisk? |
| 13:39.17 | tzafrir | use a proper init.d script, and not safe_asterisk :-( |
| 13:39.17 | *** join/#asterisk anthm (n=anthm@CPE-72-131-113-50.wi.res.rr.com) |
| 13:39.17 | *** mode/#asterisk [+o anthm] by ChanServ |
| 13:39.22 | tzafrir | bah |
| 13:39.44 | tzafrir | safe_asterisk is such a mess to work with |
| 13:40.01 | Woifi1988 | ok the best is /etc/init.d/asterisk start? |
| 13:40.18 | hmmhesays | using the start up script is the easiest |
| 13:40.23 | hmmhesays | most painless |
| 13:40.24 | tzafrir | (and disable safe_asterisk from there. At least IMHO) |
| 13:40.41 | Woifi1988 | and where do i get the output? |
| 13:40.48 | Woifi1988 | in the tty is started the script? |
| 13:40.55 | tzafrir | in /var/log/asterisk |
| 13:41.01 | The_LightSide | tzafrir, and when * dies? the init.d does not restart |
| 13:41.04 | Woifi1988 | okay thx! |
| 13:41.07 | hmmhesays | god I hate itunes m4p bullsh1at |
| 13:41.13 | tzafrir | The_LightSide, Why should asterisk die? |
| 13:41.24 | tzafrir | If it dies, chances are it will die again |
| 13:41.31 | The_LightSide | we have yet to find out, i beleive iax2 issues in 1.4.4 |
| 13:41.59 | The_LightSide | we have a queue server which randomly dies, restart and its fine for a few more days |
| 13:42.01 | tzafrir | And then the "restart" strategy will just hide your problem, and also make it impossible to work on the system while trying to fix things |
| 13:42.18 | tzafrir | The_LightSide, then fix the bug |
| 13:42.33 | tzafrir | Debug things |
| 13:42.38 | The_LightSide | tzafrir, if the system is down for more than a min, we got calls of y is it down |
| 13:42.38 | tzafrir | There's a problem hiding |
| 13:42.44 | *** join/#asterisk shinao1 (n=shinao1@217.20.242.50) |
| 13:42.56 | The_LightSide | we need to upgrade, theres been a lot of fixes since that release |
| 13:43.16 | The_LightSide | just wondering why a script that is supposed to restart * doesnt ;) |
| 13:43.49 | tzafrir | The_LightSide, maybe because there are two such scripts running in tandem |
| 13:44.00 | Woifi1988 | is it bad when i have a lot of warnigs in my /var/log/asterisk/messages that asterisk tries to include context that doesn't exist? |
| 13:44.22 | The_LightSide | nope, checked that, its alos visible on the cli if there are 2 running |
| 13:44.29 | *** join/#asterisk Dr-Linux (n=Nothing@202.125.139.198) |
| 13:44.37 | tzafrir | Woifi1988, it is not an error, but could indicate a typo in your dialplan |
| 13:44.52 | tzafrir | If you use freepbx - ognore those |
| 13:45.00 | Woifi1988 | i don't |
| 13:45.04 | Woifi1988 | i use ubuntu |
| 13:45.19 | Woifi1988 | maybe its because of the samples? |
| 13:45.38 | lirakis | It looks like when i call Queue(maint) .. the call flow does some wierd stuff.. it some how executes a macro function i wrote for dialing internal extensions... can some one take a look at this pastbin of cli |
| 13:45.39 | lirakis | http://pastebin.com/d36fd0585 |
| 13:46.12 | Woifi1988 | i've done a "make samples" |
| 13:46.13 | tzafrir | Woifi1988, well, pastebin the messages and your config if you really don't know what they are (config: extensions.conf in this case) |
| 13:46.37 | Woifi1988 | the extensions.conf is the sample file |
| 13:46.51 | Dr-Linux | tzafrir: where can i use /n option in local channel syntax? |
| 13:47.06 | tzafrir | Dr-Linux, why do you ask me? |
| 13:47.07 | lirakis | .. its like it is sending the call to the context of the extension.. then redialing it |
| 13:47.41 | Dr-Linux | tzafrir: bcoz i thought you maybe know that, since it's not comon thing |
| 13:47.55 | [TK]D-Fender | lirakis: BRILLIANT |
| 13:48.10 | [TK]D-Fender | lirakis: -- Executing Answer("Local/9003@internal-382f,2", "") in new stack <----- THIS is your problem |
| 13:48.43 | *** join/#asterisk RsaMan (n=aa@196.210.154.3) |
| 13:48.44 | lirakis | [TK]D-Fender: .. why is it doing that... i guess i dont understand how queues process calls |
| 13:48.47 | RsaMan | Hi |
| 13:48.49 | [TK]D-Fender | lirakis: NEVER use a Local channel in a queue that gets arbitrarily ANSWERED! No "Answer", no "Playback", no "Voicemail" <----------- |
| 13:48.51 | RsaMan | sorry was cut off |
| 13:49.01 | RsaMan | tzafrir : how would i reinstall the old libpri ? |
| 13:49.10 | tzafrir | make install |
| 13:49.15 | [TK]D-Fender | lirakis: the first thing your queue does is ANSWER the call and then it will NEVER cirulate to your other agents. |
| 13:49.49 | RsaMan | make install from the extracted asterisk source folder? |
| 13:49.50 | tzafrir | RsaMan, which version of asterisk do you run? (the one that complaied about chan_zap) |
| 13:49.55 | tzafrir | yes |
| 13:50.24 | RsaMan | Asterisk 1.4.10.1-BRIstuffed-0.4.0-test4 built by root @ localhost on a i686 running Linux on 2007-08-14 14:55:52 UTC |
| 13:50.32 | RsaMan | i get that error when i make install |
| 13:50.33 | RsaMan | .. |
| 13:50.57 | x86 | that's not an error |
| 13:51.05 | lirakis | [TK]D-Fender: .. the 9003 ext. that agent 100 is logged in on has a context of "internal" .. will you take a look at this pastebin again that includes that internal context from ext.conf. http://pastebin.com/d3ab93273 |
| 13:51.10 | RsaMan | noo |
| 13:51.28 | RsaMan | x86: lol i know , |
| 13:51.36 | RsaMan | x86: i posted the error further up |
| 13:52.07 | [TK]D-Fender | lirakis: No need. It is blatantly answering. Your queues is DOA |
| 13:52.30 | [TK]D-Fender | lirakis: get rid of the answer |
| 13:52.46 | lirakis | [TK]D-Fender: okay |
| 13:53.08 | [TK]D-Fender | lirakis: unfortunately your macro ALSO answers and I'll bet you may need that one... |
| 13:53.39 | [TK]D-Fender | lirakis: Noramlly you set up an entirely separate context for dialing your agents, because VM is a super no-no as well |
| 13:53.53 | lirakis | [TK]D-Fender: thats exactly what i was going to ask if i should do |
| 13:54.00 | lirakis | ok |
| 13:54.05 | lirakis | [TK]D-Fender: thanks a lot |
| 13:57.55 | *** join/#asterisk vgster (n=vgster@cpc2-ledn1-0-0-cust826.leed.cable.ntl.com) |
| 13:58.49 | lirakis | [TK]D-Fender:i just setup a dialing prefix.. so to actually dial using the macro..you have to dial 9 .. so now calls routed to _XXXX will just Dial(${EXTEN}) |
| 14:00.24 | [TK]D-Fender | lirakis: Using 1 dial to recurse into the dialplan through Local right after is just ridiculous. |
| 14:01.24 | lirakis | [TK]D-Fender: im confused... i dont know how to handle the call properly from the queue then... i though the queue would automagically dial extensions... |
| 14:01.28 | [TK]D-Fender | lirakis: exten => _XXXX,n,Macro(dial_ext) <- and this kind of completely exten driven macro with NO parameters assumes way too much. The point of macro's is PARAMETERS, and the ability to exit back for something useful. This doe NEITHER |
| 14:01.33 | lirakis | [TK]D-Fender: now i know it doesnt do that.. |
| 14:02.00 | [TK]D-Fender | lirakis: You are having it use a Local channel, and the n your "magic macro" does ANOTHER chan_local incursion. |
| 14:02.19 | lirakis | [TK]D-Fender: .. so how can i make the queue not use "local" |
| 14:02.26 | *** join/#asterisk svensk_neutrino (n=tze@static-213-115-44-90.sme.bredbandsbolaget.se) |
| 14:02.28 | lirakis | [TK]D-Fender: can i set it to "SIP" |
| 14:02.42 | lirakis | so it just dials the sip ext like i thought it would? |
| 14:02.47 | *** part/#asterisk The_LightSide (n=JBouncer@dsl-241-49-189.telkomadsl.co.za) |
| 14:02.52 | *** join/#asterisk bkruse_home (n=kruz@thuroros.wca-hsv.org) |
| 14:03.15 | [TK]D-Fender | lirakis: Yes, you can. That mean they don't log in or out any more, they are then STATIC |
| 14:03.16 | codec | hi all |
| 14:03.31 | codec | do I need to run my SIP stuff over STUN if I'm NAT'd? |
| 14:04.42 | lirakis | [TK]D-Fender: .. so if i have static members.. how do i make it so calls dont route to them when they go home for the night? |
| 14:04.59 | *** join/#asterisk shinao1 (n=shinao1@217.20.242.50) |
| 14:05.10 | [TK]D-Fender | lirakis: You'd use the "PauseQueueMember" app |
| 14:05.19 | *** join/#asterisk fakhir (n=Fakhir@unaffiliated/fakhir) |
| 14:05.23 | [TK]D-Fender | codec: No. Read this : |
| 14:05.25 | [TK]D-Fender | ~sipnat |
| 14:05.26 | jbot | [~sipnat] Quick guide on configuring * + SIP behind NAT : http://www.aocomputing.net/?p=3 otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
| 14:05.26 | RsaMan | how do i reinstall the original version of libpri ? |
| 14:05.26 | RsaMan | chan_zap.c: In function ‘zt_call’: |
| 14:05.26 | RsaMan | chan_zap.c:2129: error: too few arguments to function ‘pri_sr_set_bearer’ |
| 14:05.26 | RsaMan | chan_zap.c: In function ‘zt_hangup’: |
| 14:05.28 | [TK]D-Fender | ^^^^^^^^^^^^^^^^^^^^ |
| 14:05.37 | RsaMan | if i get those errors |
| 14:05.40 | codec | [TK]D-Fender: okay, thanks |
| 14:05.46 | [TK]D-Fender | RsaMan : the same way you would fron scratch |
| 14:05.49 | lirakis | [TK]D-Fender: .. okay .. i will do some reading |
| 14:08.41 | lirakis | [TK]D-Fender: .. quickly... so .. really my other alternative .. is to have a seperate context for agents? |
| 14:08.51 | [TK]D-Fender | lirakis: Correct |
| 14:09.20 | [TK]D-Fender | lirakis: Where it does not perform anything that would ANSWER the call except Dial being answered. |
| 14:09.24 | lirakis | [TK]D-Fender: but still then.. it would be using local channel .. and i would have to execute a Dial() command to ring the agents |
| 14:09.31 | *** join/#asterisk shinao1 (n=shinao1@217.20.242.50) |
| 14:09.52 | svensk_neutrino | is it possible to specify the ip,gw,dns-ip from the boot prompt for a linux machine? |
| 14:09.59 | [TK]D-Fender | lirakis: That Queue itself uses Local is fine. For you to RECURSE it yourself is NOT brilliant |
| 14:10.50 | [TK]D-Fender | svensk_neutrino: Depends on your distro. This isn't ##linux you know |
| 14:11.00 | lirakis | [TK]D-Fender: .. i just dont understand .. it seems like .. the queue.. sends the call to the context of the ext... so what else am i supposed to do with it besides send the call to the extension?? |
| 14:11.20 | [TK]D-Fender | lirakis: You need to pay attention to what your exten is DOING <- |
| 14:11.35 | *** join/#asterisk darkfires (n=lwhite@d38-37-41.commercial1.cgocable.net) |
| 14:11.49 | [TK]D-Fender | lirakis: Your's ANSWERS the call thereby denying its ability to cirulate to other agents. |
| 14:12.04 | lirakis | [TK]D-Fender: ... so if i have all agents members of the [agent] context .. and within there.. there is a exten s,1,Dial(${EXTEN}) .... |
| 14:12.27 | lirakis | [TK]D-Fender: right i understand .. and get that.. and have removed it |
| 14:12.40 | [TK]D-Fender | lirakis: pastebin the whole mess. |
| 14:13.24 | darkfires | does anyone know the best way to get rid of echo with digium cards? the hpec causes kernel lockups randomly... using athlon-xp on a athlon x2 |
| 14:13.52 | Qwell | darkfires: if HPEC is causing problems, you should contact support |
| 14:14.32 | *** join/#asterisk VoicePulse (n=contact@unaffiliated/voicepulse) |
| 14:16.28 | darkfires | heh |
| 14:16.37 | darkfires | so what you're saying is the only solution is hpec. |
| 14:16.41 | bkruse_home | darkfires: yes, they will jump on it and get it solved quick |
| 14:16.50 | bkruse_home | darkfires: digital card? |
| 14:16.53 | darkfires | tdm400p |
| 14:17.14 | darkfires | im even getting crossed lines |
| 14:17.23 | *** join/#asterisk MindTheGap (n=iote@c9505ffe.bhz.virtua.com.br) |
| 14:17.25 | *** join/#asterisk pa (n=Paolo@unaffiliated/pa) |
| 14:17.28 | bkruse_home | darkfires: do you have fxs ports? |
| 14:17.40 | darkfires | fxo |
| 14:17.56 | bkruse_home | darkfires: how far are the lines? |
| 14:18.02 | bkruse_home | use fxotune on the card |
| 14:18.03 | bkruse_home | done |
| 14:18.04 | [TK]D-Fender | "But Egon, I though you said crossing the streams was 'Baaaad'" |
| 14:18.13 | darkfires | how far are the lines ?? |
| 14:18.32 | Strom_M | well, technically, "how many thousand feet of copper are between your CPE and the telco's switch?" |
| 14:18.49 | darkfires | well with how the bell guy was telling me the phone lines were routed in this building |
| 14:18.49 | darkfires | lots |
| 14:19.11 | Strom_M | you should also make sure your receive gain is balanced against the attenuation on the loop |
| 14:19.21 | Strom_M | darkfires: "lots" isn't a number, sadly :) |
| 14:19.57 | codec | [TK]D-Fender: hmm, seems like i need a STUN server because my phone and my * is behind a NAT |
| 14:19.58 | darkfires | i know that but i dont have any equip that will tell me specifically ;) |
| 14:20.02 | bkruse_home | Strom_M: correct. |
| 14:20.13 | darkfires | Strom_M how do you balance the gain with the attenuation on the loop |
| 14:20.16 | bkruse_home | Strom_M: wouldnt you suggest fxotune? |
| 14:20.21 | darkfires | excuse my ignorance please. |
| 14:20.28 | bkruse_home | messing with fxotune and your rxgains/txgains could make a huge difference |
| 14:20.28 | darkfires | but yes im looking at fxotune right now heh |
| 14:20.42 | Strom_M | darkfires: call telco repair service and ask them to test the loop...but make sure you disconnect your CPE first |
| 14:21.00 | bkruse_home | darkfires: great, you can mess with that all day |
| 14:21.12 | bkruse_home | the gui will have fxotune options in there one day soon.... |
| 14:21.25 | darkfires | theres a gui ? |
| 14:21.35 | bkruse_home | darkfires: haha, yes. |
| 14:21.42 | bkruse_home | no one knows about it I suppose :/ |
| 14:22.05 | Strom_M | darkfires: you adjust rxgain by calling up the telco's milliwatt test number located in the same class 5 switch that serves your dial tone, then messing with rxgain until the level that shows up in ztmonitor is about 14,000 |
| 14:23.02 | darkfires | my phone company forwards all the sales and technical s upport to india |
| 14:23.10 | bkruse_home | Strom_M: correct. |
| 14:23.17 | *** join/#asterisk anonymouz666 (n=anonymou@189.25.172.179) |
| 14:23.28 | Strom_M | darkfires: your local exchange carrier |
| 14:23.31 | darkfires | Yes. |
| 14:23.39 | darkfires | Bell Canada |
| 14:23.39 | bkruse_home | Strom_M: 14,000 = magic number? |
| 14:23.52 | bkruse_home | fxotune WILL try to guess, but that would be the best way after you have those numbers set. |
| 14:24.11 | Strom_M | bkruse: yes; that's what happens when you call up the local milliwatt via a CAS T1 or ISDN PRI circuit |
| 14:24.44 | Strom_M | darkfires: not tech support. you want repair service |
| 14:24.45 | *** join/#asterisk ManxPower (n=manxpowe@032-393-107.area5.spcsdns.net) |
| 14:24.47 | bkruse_home | Strom_M: ahh, great |
| 14:25.03 | darkfires | ok |
| 14:25.15 | *** join/#asterisk rmayorga (n=rmayorga@unaffiliated/rmayorga) |
| 14:25.16 | Strom_M | usually it's on 611 |
| 14:25.22 | Strom_M | but i'd check your local directory |
| 14:25.27 | darkfires | so 611 will give me a miliwatt test number |
| 14:25.29 | Strom_M | no |
| 14:25.35 | Strom_M | 611 will give you repair service |
| 14:25.40 | darkfires | oh, right |
| 14:25.43 | Strom_M | assuming bell canada puts repair service on 611 |
| 14:25.49 | bkruse_home | 611 will give you a persn |
| 14:25.51 | bkruse_home | person* |
| 14:25.57 | lirakis | raar! |
| 14:26.00 | Strom_M | you ask them for the test number and hope and pray they give it to you :) |
| 14:26.24 | Strom_M | but they should tell you how many thousand feet long your local loop is |
| 14:26.59 | *** join/#asterisk Dan0maN_Work (n=dschuh@64.149.174.136) |
| 14:27.25 | *** join/#asterisk galeras (n=galeras@200.31.204.42) |
| 14:28.17 | *** join/#asterisk alexcf (i=alexcf@lexi.me.uk) |
| 14:28.25 | hmmhesays | wow that was a serious pain in the @$$ |
| 14:28.33 | alexcf | hi.. i've got a weird problem since i rebooted my asterisk box |
| 14:28.46 | alexcf | when a call comes into a queue, there's no MoH or announcements being sent |
| 14:28.55 | bkruse_home | hmmhesays: pretty much |
| 14:29.00 | bkruse_home | you can do fxotune and hope for thebest |
| 14:29.01 | alexcf | but, the call comes through, and when an agent answers it, it's fined |
| 14:29.04 | bkruse_home | tweak it yourself to find what sounds rihgt |
| 14:29.09 | bkruse_home | if not, then try it |
| 14:29.13 | alexcf | the user can then be placed on hold and MoH works fine |
| 14:29.17 | *** join/#asterisk zapp-branigan (n=zapp-bra@9.218.216.87.static.jazztel.es) |
| 14:29.37 | Strom_M | "tweak it yourself" almost never works |
| 14:29.50 | bkruse_home | Strom_M: worked for me |
| 14:29.53 | bkruse_home | and our beta partners |
| 14:30.11 | bkruse_home | Strom_M: but I most def will try your suggestions next time...sounds way more effecient |
| 14:31.55 | *** join/#asterisk zeppelin_ (n=zeppelin@201-35-136-14.paemt700.dsl.brasiltelecom.net.br) |
| 14:31.56 | ManxPower | The answers you seek are within yourself, grasshopper. |
| 14:32.02 | bkruse_home | ManxPower: nice. |
| 14:32.08 | alexcf | no one had that issue before? o0 |
| 14:32.24 | ManxPower | alexcf: no. |
| 14:32.29 | lirakis | [TK]D-Fender: .. it really wasnt so complex... i dont know if you meant for some thing else.. but this is working fine... http://pastebin.com/d20595a7b |
| 14:32.30 | alexcf | bummer :) |
| 14:33.22 | Strom_M | bkruse_home: well yeah, but you actually know what to listen for |
| 14:33.40 | Strom_M | most people just keep cranking that number up because they want MOAR LOUD |
| 14:34.01 | bkruse_home | Strom_M: ahh, this is true. Lol, nice with "MOAR LOUD" |
| 14:34.06 | darkfires | Strom_M what parameters do oyu use on ztmonitor to match up with the 14000 |
| 14:34.14 | darkfires | ztmonitor 1 -v ? |
| 14:35.32 | Strom_M | -vv IIRC |
| 14:35.55 | bkruse_home | Strom_M: Ima have to call you one of these days and get some more info on that |
| 14:36.03 | Strom_M | bkruse_home: alright |
| 14:36.05 | bkruse_home | When you coming back? |
| 14:36.09 | darkfires | Strom_M the TX? |
| 14:36.12 | bkruse_home | or in town for a little, your training now right? |
| 14:36.17 | Strom_M | september IIRC |
| 14:36.18 | alexcf | ah |
| 14:36.21 | alexcf | i was missing an "answer" |
| 14:36.23 | Strom_M | darkfires: no no, RX |
| 14:36.26 | bkruse_home | Strom_M: nice nice |
| 14:36.29 | bkruse_home | tell me when you do |
| 14:36.34 | bkruse_home | for the new building? |
| 14:36.35 | darkfires | RX is like 1-100 |
| 14:36.40 | darkfires | sometimes up to 300 |
| 14:36.47 | darkfires | oh went up to 1000 |
| 14:37.08 | Strom_M | darkfires: is the milliwatt number you're calling in the same class 5 switch as the other end of your copper? |
| 14:37.40 | Strom_M | bkruse_home: no, the september one will be at Atrium |
| 14:37.45 | bkruse_home | Strom_M: ahh :/ |
| 14:37.58 | Strom_M | yeah, the new building looks sweet |
| 14:38.26 | bkruse_home | Strom_M: have you seen the latest? its awesome. |
| 14:38.55 | Strom_M | i saw it in....july |
| 14:39.11 | bkruse_home | wow its come a long way even since then |
| 14:39.15 | bkruse_home | its excited |
| 14:39.17 | bkruse_home | exiting* |
| 14:39.22 | bkruse_home | s/excited/exciting/g |
| 14:39.28 | bkruse_home | :[ |
| 14:39.42 | Strom_M | catsex |
| 14:40.02 | bkruse_home | Strom_M: totally. |
| 14:40.04 | [TK]D-Fender | codec: No, you do NOT. Just read the guide. |
| 14:40.10 | [TK]D-Fender | codec: the FIRST one. |
| 14:40.25 | bkruse_home | Strom_M: remember that day we went to cheeburger and you ordered milk and the lady got confused? |
| 14:40.31 | *** join/#asterisk codazoda (n=Joel_Dar@mail.hurdmanivr.com) |
| 14:40.32 | *** join/#asterisk flujan (n=flujan@200-160-115-020.static.spo.ctbc.com.br) |
| 14:40.32 | bkruse_home | I am totally good friends with that chick now, lol |
| 14:40.53 | Strom_M | sweet |
| 14:41.19 | [TK]D-Fender | lirakis: No, I said make an entirely SEPARATE context, not just another exten in some "generic" one. |
| 14:41.38 | darkfires | Strom_M: thank you for all your help and putting up with my complete telephony ignorance. |
| 14:41.53 | bkruse_home | darkfires: youc all em yet? |
| 14:41.58 | darkfires | on hold |
| 14:42.07 | Strom_M | darkfires: no worries :) |
| 14:42.48 | codazoda | I'm installing Asterisk for the umptenth time. This install is on a slightly newer version of CentOS, with a slightly newer Asterisk, and a slightly newer PHP. When I run AGI scripts, relative include paths don't seem to work. For example, I'm doing "include './phpagi-2.14/phpagi.php'", and it fails to find the file. If I use the full path, it works fine. Does anyone know what dictates where PHP is being executed from? |
| 14:42.55 | lirakis | [TK]D-Fender: .. i know.. the thing is .. is that the only people on this pbx are call agents. I may totally remove the macro .. b/c as you seem so say so often.. its not needed (or rather improper) .. the only reason i have it is so i can easily do the whole vm thing with one call |
| 14:43.31 | [TK]D-Fender | lirakis: tahts FINE. leave that for NORMAL use and get your queues the HELL OUT OF THESE |
| 14:43.53 | [TK]D-Fender | lirakis: Make ANOTHER context and leave [internal] behind fo it own purpose! |
| 14:45.29 | lirakis | [TK]D-Fender: .. okay.. but if i make a "queueagent" context .. the queue agents still need to be able to dial ext->ext and to make outbound calls... so some how that functionality will have to get linked into thier "special" contexts. I guess i just dont understand how that is so different that this current setup.. but i do want to know |
| 14:45.49 | bkruse_home | codazoda: good point, I believe its run from that native directory, even if you do php -q /blah/phpscript.php, and include 'class.php' it should include it from /blah/include.php, not caring where it was executed from |
| 14:46.13 | bkruse_home | codazoda: try the obvious, remove the ./relative and just do include 'file.php' and put it in the same directory, and try that |
| 14:46.25 | bkruse_home | btw, the php_agi and php_manager libraries are incredible. |
| 14:46.36 | Dr-Linux | Qwell: any good news about 7935 with 1.4.x? :) |
| 14:46.45 | [TK]D-Fender | <PROTECTED> |
| 14:47.09 | [TK]D-Fender | lirakis: Your phones will still use [internal] or whatever.... |
| 14:48.03 | lirakis | [TK]D-Fender: i can only "force" the queue to use some thing other than local .. when i use static agents (which i dont want to do ) .. otherwise it uses local.. which will use whatever context the extension is under... correct? |
| 14:48.21 | Dr-Linux | file: :) |
| 14:48.42 | codazoda | It works if I move the phpagi.php to the same location as my calling php file. It also works if I use the full path. So, it seems like the working directory is just wrong when it's executed. |
| 14:49.32 | [TK]D-Fender | lirakis: No, it uses whatever context you TELL IT. Time to wake up and realize where you are SPECIFYING the context it uses.... |
| 14:49.45 | ManxPower | well make the php print out the current directory. that will tell you where the "current" directory |
| 14:50.20 | file | Dr-Linux: hrm? |
| 14:50.43 | *** join/#asterisk davixx (n=davixx@ASt-Lambert-151-1-9-194.w82-120.abo.wanadoo.fr) |
| 14:50.47 | *** join/#asterisk PepOSX (n=pepOSX@200.90.126.74) |
| 14:51.11 | lirakis | [TK]D-Fender: thats the part im missing.. how am i telling the queue what context to goto when it tries to reach a agent? |
| 14:51.16 | Dr-Linux | file: can i pastebin, that how i'm papulating varibales for type etc? |
| 14:51.18 | Dr-Linux | :) |
| 14:51.30 | [TK]D-Fender | lirakis: Go look where you log them IN <- |
| 14:51.39 | lirakis | [TK]D-Fender: okay |
| 14:51.44 | file | Dr-Linux: you could, but I have work to do ... and that doesn't include mucking with your code |
| 14:52.04 | darkfires | Strom_M the bell guy is here.... do i ask for a milliwatt test number |
| 14:52.08 | lirakis | [TK]D-Fender: ahh finally .. i see it... thank you.. |
| 14:52.43 | lirakis | [TK]D-Fender: i will remove the 9 prefix.. create a different context just for the agents .. and change the login context |
| 14:54.34 | Strom_M | darkfires: on your prem? |
| 14:54.34 | Strom_M | or on the phone |
| 14:54.34 | Strom_M | ask first for a loop length check (make sure you disconnect any and all CPE first) |
| 14:55.06 | ManxPower | "My loop length is 90,000 ft! Unplug your phone, idiot." |
| 14:55.29 | lirakis | [TK]D-Fender: ... so now.. another question.. should i do anything for the timout extension in the agents context? .. i dont want to disrupt the queue .. so i dont know if i should hangup() or anything |
| 14:55.34 | darkfires | cpe ?? |
| 14:55.54 | darkfires | copper pair |
| 14:56.13 | [TK]D-Fender | lirakis: No, just Dial. |
| 14:56.22 | [TK]D-Fender | lirakis: Or other "non-call" stuff |
| 14:56.22 | lirakis | [TK]D-Fender: okay |
| 14:56.36 | [TK]D-Fender | lirakis: Actaully "hangup probably shouldn't be bad. |
| 14:56.36 | lirakis | [TK]D-Fender: thanks very much for your patients. |
| 14:56.49 | lirakis | [TK]D-Fender: ill try it to see if it freaks out |
| 14:57.11 | Strom_M | darkfires: CPE is customer premise equipment |
| 14:57.19 | codazoda | Okay, I ran my PHP script and output the directory it's running FROM. I already knew it wasn't executing from the same directory as the PHP script. Now I know it's actually executing from /tmp. Can I change this behavior somehow? |
| 14:57.25 | Strom_M | phones, PBXes, answering machines, Call display adjunct units, etc etc etc etc |
| 14:57.31 | *** join/#asterisk saftsack (n=oliver@p54A7FFF2.dip.t-dialin.net) |
| 14:57.53 | ManxPower | codazoda: there is no default working directory as far as I know. |
| 14:58.07 | ManxPower | codazoda: How, exactly, are you starting Asterisk? |
| 14:58.21 | codazoda | ManxPower, I started it with safe_asterisk. |
| 14:58.41 | *** join/#asterisk Alric (n=nbowyer@masq.hyperusa.com) |
| 14:58.42 | *** join/#asterisk holiday_42 (n=no@spike.wcta.net) |
| 14:58.44 | ManxPower | I guess you know where to look, don't you. |
| 15:03.16 | *** join/#asterisk masus (n=tet@88.248.14.186) |
| 15:03.42 | darkfires | Strom_M he's telling me it wont affect the echo heh |
| 15:03.46 | darkfires | the repair guy is here |
| 15:03.57 | darkfires | Strom_M do i have asterisk running when they do the test |
| 15:04.18 | darkfires | er stupid q |
| 15:05.44 | *** join/#asterisk bbryant (n=brett@68.208.65.50) |
| 15:06.26 | bkruse_home | Strom_M: "he's telling me it wont affect the echo" |
| 15:06.38 | bkruse_home | typical sales/repair man |
| 15:06.41 | PepOSX | how i declare timemax for answer one call and go to the voicemail? |
| 15:06.50 | bkruse_home | did he ask you to upgrade from the 4 lines to a pri? :P |
| 15:07.28 | *** join/#asterisk blinky42 (n=me@67.106.175.130.ptr.us.xo.net) |
| 15:07.41 | *** join/#asterisk Juggie (n=Juggie@CPE00c049d9f271-CM000a73a18a20.cpe.net.cable.rogers.com) |
| 15:10.27 | Strom_M | darkfires: having your gains set correctly will help |
| 15:10.37 | *** join/#asterisk blinky42 (n=me@67.106.175.130.ptr.us.xo.net) |
| 15:11.41 | *** join/#asterisk Luch0 (n=luis@217.149.158.122) |
| 15:12.47 | darkfires | Strom_M they said theres no such thing as milliwatt test |
| 15:13.01 | masus | PepOSX: try this exten=1234,1,Dial,SIP/${EXTEN}|10 ; |
| 15:13.11 | ManxPower | darkfires: hole on |
| 15:13.20 | Strom_M | darkfires: yes there is |
| 15:13.37 | Strom_M | some telcos just don't like to give the number to customers |
| 15:13.39 | darkfires | i know |
| 15:13.43 | darkfires | what else is it called? |
| 15:13.46 | darkfires | terminating test number? |
| 15:13.48 | Strom_M | no |
| 15:13.55 | Strom_M | let me look up the specific type |
| 15:13.55 | Strom_M | hang on |
| 15:13.58 | darkfires | k |
| 15:14.38 | ManxPower | darkfires: it is also called a "type 102" test. |
| 15:14.51 | ManxPower | http://lists.digium.com/pipermail/asterisk-users/2004-September/056166.html |
| 15:15.05 | Strom_M | ManxPower: yeah, i was just grepping through "notes on the network" trying to determine which numbered test line type it is :) |
| 15:15.19 | Strom_M | yes, 102-type |
| 15:15.35 | Strom_M | notes on the networks page 593 :) |
| 15:15.39 | *** join/#asterisk riddlebox (n=JamesMid@75-128-170-26.static.stls.mo.charter.com) |
| 15:15.49 | darkfires | the bell guy here says hes never heard of it and hes calling another bell guy same thing |
| 15:15.50 | darkfires | fuckin cock |
| 15:15.52 | darkfires | i know they know |
| 15:15.55 | riddlebox | has anyone used any of the Quintum devices with asterisk? |
| 15:16.10 | Strom_M | darkfires: ask for the 102 type test line then :) |
| 15:16.19 | darkfires | he says its 1.5km |
| 15:16.20 | darkfires | distance |
| 15:16.24 | ManxPower | Also: http://www.cisco.com/en/US/products/hw/switches/ps1925/products_maintenance_guide_chapter09186a008008745c.html for cisco's take on it |
| 15:16.35 | Strom_M | darkfires: exactly? |
| 15:16.43 | *** join/#asterisk neverblue (n=neverblu@unaffiliated/neverblue) |
| 15:17.23 | *** join/#asterisk oej (n=olle@soll4-125.cust.blixtvik.net) |
| 15:17.31 | darkfires | yeah |
| 15:17.41 | Strom_M | ok, that's only about 5000 feet |
| 15:17.50 | Strom_M | so your loop shouldn't be attenuating too much |
| 15:19.07 | darkfires | hes going to call with that tes |
| 15:19.08 | darkfires | t |
| 15:19.18 | Strom_M | call you and give you the number? |
| 15:19.33 | Qwell[] | Strom_M: No, he's gonna call into the test line, and 3-way him in |
| 15:19.49 | Qwell[] | (I'm kidding, of course) |
| 15:19.50 | Strom_M | Qwell[]: well, i dont know; his statement was ambiguous at best |
| 15:20.09 | Qwell[] | Strom_M: sorry, that sounded sarcastic...it wasn't supposed to be, heh |
| 15:20.13 | Strom_M | :D |
| 15:20.37 | Qwell[] | just sounds like something a bell tech would try |
| 15:20.49 | *** join/#asterisk CrazyTux[m] (n=CrazyTux@216-110-94-230.static.twtelecom.net) |
| 15:21.07 | *** join/#asterisk jingles (n=foo@39.183.dowl.anc.borealisbroadband.net) |
| 15:22.08 | Strom_M | Qwell[]: I picked up an AT&T canvas tool bag the other week |
| 15:22.15 | Qwell[] | eh? |
| 15:22.24 | Qwell[] | the slick roll-up kind? |
| 15:22.39 | Qwell[] | not sure why I instantly pictured one of those |
| 15:23.17 | Strom_M | looking for a picture now; hold plzkthzomglolwtf |
| 15:25.11 | darkfires | they wont give me the number |
| 15:25.20 | darkfires | flat out refuse |
| 15:26.21 | zapp-branigan | hi who can i generate md5 asterisk passords ? |
| 15:26.29 | darkfires | 31051.745000] 130751745 Polarity reversed (0 -> 1) |
| 15:26.31 | darkfires | wtf |
| 15:26.34 | Strom_M | darkfires: oh well, at least knowing the length helps a bit |
| 15:26.38 | Strom_M | there are other ways around it |
| 15:26.56 | Strom_M | Qwell[]: one of these, except sans tools, and with the at&t logo on one side |
| 15:26.56 | Strom_M | http://www.gemline.com/gemline.web/shop-gemline/product_detail.aspx?productid=340 |
| 15:27.13 | Qwell[] | ahh, nice |
| 15:27.23 | Qwell[] | kinda like an old school doctors bag, except not leather :p |
| 15:27.34 | Strom_M | yeah |
| 15:27.37 | Strom_M | well it was $0 |
| 15:27.42 | Qwell[] | nice |
| 15:27.53 | *** join/#asterisk viKing78 (n=viking@cerberus.franklinamerican.com) |
| 15:27.59 | Strom_M | yeah, so i'm not complaining :D |
| 15:28.10 | Strom_M | i'm gonna keep all my phone tools in that |
| 15:28.24 | Sweeper | at&t logo is cool :3 |
| 15:28.45 | Sweeper | but that bag wouldn't hold all my tools~ |
| 15:29.50 | codazoda | safe_asterisk launches asterisk from /tmp (current and older versions). It seems that the active directory is now wherever you start asterisk from. This seems to be a change from previous versions, since my other systems also start in /tmp but that doesn't effect the working directory of AGI scripts. I also wonder if this might be a bug, since it seems more logical that AGI's should be executed with a working directory of wherever they are loc |
| 15:30.44 | darkfires | Strom_M should i be adjusting anything now that i know the length is 1.5km |
| 15:30.50 | Strom_M | Sweeper: well i've got the basics...buttset, tone/probe kit, d814 impact tool, pliers, screwdrivers, dykes, butt splices, screws, bridge clips, plastic T1 circuit protection caps, crimp tool, can wrench, etc etc |
| 15:31.09 | Qwell[] | circuit protection caps? |
| 15:31.37 | Strom_M | Qwell: yeah, you know those little red plastic caps they put on 66 blocks to prevent you accidentally taking down someone's T1? |
| 15:31.39 | coppice | condoms for circuits |
| 15:31.47 | Qwell[] | Strom_M: nop |
| 15:31.50 | Qwell[] | nope too |
| 15:31.56 | Strom_M | *blink* |
| 15:32.04 | Qwell[] | I'm a programmer, not a telcom guy :p |
| 15:32.15 | Strom_M | but but but you work on phone software :) |
| 15:32.32 | Qwell[] | I don't think I've ever needed to interface with a circuit protection cap :D |
| 15:32.34 | russellb | Qwell[]: me too :) |
| 15:32.35 | [TK]D-Fender | Strom_M: " dykes, butt splices, screws," : This is starting to sound a little too "racy" ;) |
| 15:32.49 | Qwell[] | [TK]D-Fender: telcom is full of innuendo |
| 15:33.01 | Strom_M | [TK]D-Fender: no, it gets racy when I start talking about what my boyfriend and I did last night |
| 15:33.04 | [TK]D-Fender | Qwell[]: In YOUR "Edno" ;) |
| 15:33.10 | [TK]D-Fender | Endo* |
| 15:33.41 | Strom_M | darkfires: lemme look and see what rxgain I'm using |
| 15:34.02 | codazoda | Anyone else running AGI under 2.4.11? |
| 15:34.16 | Qwell[] | kernel 2.4.11? |
| 15:34.22 | Strom_M | i have a setting of 5.3 on a loop of 7000 ft |
| 15:34.45 | Strom_M | so try setting it around 3.8-4.4 |
| 15:35.07 | codazoda | No, Asterisk 2.4.11. |
| 15:35.27 | Strom_M | asterisk 2.4.11?? |
| 15:35.29 | codazoda | Uhg. SOrry. 1.4.11 |
| 15:35.38 | robl^ | Strom_M: what about a punch down tool to go with the butt set? |
| 15:35.46 | Strom_M | i was about to go look at my calendar and make sure it wasn't 2011 |
| 15:35.52 | Strom_M | robl^: I said I have that |
| 15:35.59 | Strom_M | D814 impact tool :) |
| 15:36.26 | Strom_M | http://www.flukenetworks.com/fnet/en-us/products/D814+Series+Impact+Tools/?categorycode=PTB |
| 15:36.31 | darkfires | Strom_M are you using hpec or no |
| 15:36.37 | Strom_M | darkfires: no |
| 15:36.40 | *** join/#asterisk l2trace9999 (n=l2trace@fl-67-76-209-28.sta.embarqhsd.net) |
| 15:36.45 | Strom_M | though I should try it out |
| 15:37.53 | darkfires | i got the ANAC number |
| 15:37.57 | *** join/#asterisk outtolunc (n=me@adsl-66-218-53-172.dslextreme.com) |
| 15:38.02 | darkfires | do you think the other nubmer would be similiar |
| 15:38.06 | Strom_M | possibly |
| 15:38.18 | darkfires | 905 958 2622 is the anac |
| 15:38.36 | Strom_M | yeah, 958 and 959 are reserved prefixes for test numbers |
| 15:38.48 | darkfires | i tried 2623 |
| 15:38.53 | darkfires | some chick answered and said |
| 15:38.58 | darkfires | good morning, what number did you dial? |
| 15:39.04 | Strom_M | holy shit |
| 15:39.11 | Strom_M | you have a live intercept operator!?! |
| 15:39.18 | darkfires | is that what that is |
| 15:39.19 | Strom_M | that's AWESOME |
| 15:39.24 | darkfires | what can i do with that |
| 15:39.35 | *** join/#asterisk jmesquita (n=jmesquit@201.20.243.195.user.ajato.com.br) |
| 15:39.39 | Strom_M | you can tell her a number and she will tell you whether it's been changed or disconnected or whatnot |
| 15:39.39 | outtolunc | move to the big city <G> |
| 15:40.13 | Strom_M | but yeah, i didnt know bell canada still did live intercept |
| 15:40.18 | Strom_M | that's uber awesome |
| 15:41.07 | Qwell[] | What would she ask what number you dialed? You'd think she'd know that |
| 15:41.14 | darkfires | im not sure |
| 15:41.24 | Qwell[] | should've said "911" |
| 15:41.32 | darkfires | haha |
| 15:41.40 | *** join/#asterisk javb (n=javb@190.80.238.132) |
| 15:41.40 | Strom_M | Qwell[]: there's a good reason for that |
| 15:41.45 | *** join/#asterisk davevg-btwtech (n=davevg@nj-67-76-177-147.sta.embarqhsd.net) |
| 15:42.01 | javb | Any ideas of a board that supports a TE210P digium Dual t1 Card ? |
| 15:42.11 | javb | Must be PCI 3,3 Volts |
| 15:42.31 | Strom_M | if you route numbers like that to live intercept, then the operator can tell the customer whether they've misdialed or whether they just have the wrong number |
| 15:42.45 | Qwell[] | uh...huh |
| 15:42.49 | Strom_M | as a general rule, any time I get a wrong number call, I always ask what number they think they're dialing |
| 15:43.06 | *** join/#asterisk hfb (n=hfb@pool-72-87-254-188.lsanca.dsl-w.verizon.net) |
| 15:43.21 | Strom_M | that way I can tell them if they've misdialed, or I can tell then whether they have the wrong number in their directory |
| 15:44.19 | rudholm | I do it to reduce the chances that they're going to just call me again in 10 seconds. |
| 15:44.24 | *** join/#asterisk ctaloi (n=ctAloi@nat-66-218-1-47.usadatanet.com) |
| 15:44.48 | Strom_M | yes, same reason here |
| 15:44.56 | Qwell[] | meh, I tell people they mis-dialed, and they still call back like 3 times |
| 15:45.05 | *** join/#asterisk fatgoose (n=fg@206-248-135-39.dsl.teksavvy.com) |
| 15:45.12 | Qwell[] | "Nope, you're STILL dialing the wrong number..." |
| 15:45.26 | Strom_M | Qwell[]: do you ask them what number they think they're dialing? |
| 15:45.50 | riddlebox | should I put my co straight to my fax machine, or would it be ok to have the fax as an extension on an external fxs box? |
| 15:46.00 | Qwell[] | Strom_M: well, heh...yeah |
| 15:46.18 | Qwell[] | my phone number was once put into a Hispanic newspaper...as a mechanic |
| 15:46.24 | Qwell[] | ...with no area code |
| 15:46.30 | *** part/#asterisk codazoda (n=Joel_Dar@mail.hurdmanivr.com) |
| 15:46.35 | *** join/#asterisk duckz (n=duckz@81.180.83.75) |
| 15:46.38 | Qwell[] | well, s/once/repeatedly/ |
| 15:46.56 | lirakis | is there some thing like a "show agents" commant to see what agents are logged in? |
| 15:46.59 | Qwell[] | explaining it to them was a bit...uhh...difficult |
| 15:47.00 | darkfires | maybe ill setup asterisk to sit and call every # through 905-958 ... cause most of them are out of service |
| 15:47.01 | [TK]D-Fender | riddlebox: Straight if you know whats good for you... |
| 15:47.14 | lirakis | ah |
| 15:47.14 | *** part/#asterisk masus (n=tet@88.248.14.186) |
| 15:47.18 | Qwell[] | darkfires: gonna get busted for wardialing :p |
| 15:47.18 | lirakis | how about "show agents" |
| 15:47.18 | lirakis | duh |
| 15:47.19 | lirakis | lol |
| 15:47.28 | Qwell[] | wardialing test numbers seems like a bad idea |
| 15:47.37 | [TK]D-Fender | lirakis: "show queues" <- also very useful |
| 15:47.53 | darkfires | qwell seriously ? |
| 15:47.59 | riddlebox | [TK]D-Fender: thats kinda what I figured, I will buy a smart device to go in front of the line and then go into the fax or my asterisk box |
| 15:48.01 | Strom_M | ?Hola!?Bienvenidos a la casa de telefonos de Qwell! Si quieres habla con un mecanico, tienes un numero malo. |
| 15:48.02 | Qwell[] | dunno, but it sure seems silly |
| 15:48.19 | Qwell[] | Strom_M: the problem went away when I switched away from Sprint...heh |
| 15:48.24 | Strom_M | heh |
| 15:48.26 | Qwell[] | some other sucker has that number now |
| 15:48.33 | Qwell[] | probably, anyhow |
| 15:48.49 | `Sean | lol @ wardialing |
| 15:49.02 | Strom_M | 909-909-9009 |
| 15:49.09 | Qwell[] | 714 |
| 15:49.25 | Strom_M | damnit, Qwell, you're ruining my 909 joke |
| 15:49.34 | Qwell[] | 951 > 909 |
| 15:49.39 | Strom_M | yes I know |
| 15:49.48 | Strom_M | hence a 909 joke |
| 15:49.49 | Qwell[] | I never lived in the real 909 :p |
| 15:50.00 | Qwell[] | huh |
| 15:50.11 | Qwell[] | I don't think they ever released that number back into the pool |
| 15:50.27 | `Sean | lol i should war dial all of 710 :P |
| 15:50.28 | rudholm | Strom_M: you know how I have GLadstone5-XXXX, right? There's a restaurant in Westwood with HIghland5-XXXX and their customers often call me instead |
| 15:50.34 | `Sean | see wich numbers are active :D |
| 15:50.46 | Strom_M | Qwell[]: they tend to like to leave numbers unassigned for some time after disconnection |
| 15:50.55 | Strom_M | `Sean: there's only one active prefix in that area code |
| 15:51.03 | Qwell[] | Strom_M: it's going straight to vm... like it's still active |
| 15:51.18 | Strom_M | rudholm: yeah, I remember you've gotten calls for that restaurant while I was there |
| 15:51.20 | rudholm | Strom_M: so does that mean your phone number hasn't been reassigned yet? |
| 15:51.22 | Qwell[] | I'm gonna check my old password :p |
| 15:51.35 | Strom_M | rudholm: I think we were eating breakfast when it happened once |
| 15:51.43 | Strom_M | rudholm: no, I still have my number :) |
| 15:51.44 | `Sean | Strom_M someone has tried before? |
| 15:51.49 | Strom_M | i had my service reconnected |
| 15:52.08 | Strom_M | `Sean: uh, no, it's a matter of looking at prefix assignment records from NANPA and/or Telcordia |
| 15:52.09 | *** join/#asterisk aris_g (n=manager@200.71.48.212) |
| 15:52.14 | aris_g | Hello People.. |
| 15:52.52 | *** join/#asterisk Weezey (n=ohno@wan.iasloffice.iasl.com) |
| 15:53.17 | Qwell[] | eww |
| 15:53.20 | Qwell[] | Acer is buying Gateway |
| 15:53.44 | Strom_M | are they going to call the new company Aceway, or will it be Gacer? |
| 15:54.17 | *** join/#asterisk CunningPike (n=CunningP@dhcp-10-234.district.north-van.bc.ca) |
| 15:54.29 | aris_g | Friends ...i have a problem with Calls Hangup... My server is a gateway into 2 PRI... PSTN-----Asterisk----PBX.... and only see this .... |
| 15:54.32 | aris_g | http://pastebin.com/m5174492 |
| 15:56.32 | Qwell[] | Strom_M: If they're smart, they'll get rid of the gateway name |
| 15:57.25 | Strom_M | i like "Aceway" |
| 15:57.43 | *** join/#asterisk rmayorga_ (n=rmayorga@unaffiliated/rmayorga) |
| 15:57.58 | Strom_M | damnit, someone already owns aceway.com |
| 15:59.25 | hmmhesays | aceway.net? |
| 15:59.38 | Strom_M | taken |
| 16:00.32 | *** join/#asterisk aikanaro79 (n=noone@89-180-92-128.net.novis.pt) |
| 16:00.55 | Buhntz | .org .info .us is free |
| 16:01.03 | aikanaro79 | hi..how do I transfer a call? is it possible to transfer a call to other context just after someone has been dialed? |
| 16:01.07 | Weezey | I'm having a weird issue. Asterisk 1.4 shows the 1.2 box via IAX2 shows codec: 0xe004 (ulaw) and 1.2 shows the 1.4 box via IAX2 with codec: 0xf804 (ulaw). They seem to talk 1.2 -> 1.4 just fine, but when calls come in from 1.4 to 1.2 it complains that it doesn't understand the codecs. |
| 16:01.28 | *** join/#asterisk etix (n=etix@nala.l0cal.com) |
| 16:01.32 | aris_g | i see many time this in calls....is it normal?????? |
| 16:01.34 | aris_g | http://pastebin.com/m5719d488 |
| 16:01.59 | fatgoose | hi |
| 16:02.05 | fatgoose | anyone tried the Digium AA50 ? |
| 16:03.00 | darkfires | Strom_M when you dial a milliwatt test number is it just a loud long tone ? |
| 16:03.33 | Strom_M | darkfires: yes |
| 16:03.40 | Strom_M | it should be 1004hz |
| 16:04.07 | darkfires | i found some milliwatt test numbers that are long distance heh |
| 16:04.15 | Strom_M | that's not going to work |
| 16:04.23 | Strom_M | you need it to be in your class 5 office |
| 16:04.27 | ManxPower | The Asterisk milliwatt is 1000Hz |
| 16:04.28 | darkfires | i know |
| 16:04.43 | darkfires | Strom_M i just dialed it anyway to see what it sounded like so when i find the # for my own |
| 16:04.49 | Strom_M | :) |
| 16:05.02 | ManxPower | I feel that as long as the path to the test number is all digital it would be REASONABLY accurate. |
| 16:05.53 | Strom_M | ManxPower: yeah, that's not the case |
| 16:05.55 | aris_g | any Idea? |
| 16:05.58 | *** part/#asterisk shtoom (n=shtoom@123.252.144.92) |
| 16:06.09 | Strom_M | many IXC trunks will be calibrated to a -3dB attenuation |
| 16:07.07 | *** join/#asterisk Yourname` (n=chatzill@76-10-128-178.dsl.teksavvy.com) |
| 16:07.56 | Yourname` | Hi, queues are making me go wild. Especially when I type "show queues", it gives all these different values, like (dynamic) (Invalid) (Not in use) -> and I can't find any documentation telling me what these words mean on the CLI |
| 16:08.12 | Qwell[] | fatgoose: I have |
| 16:08.20 | *** join/#asterisk rtcg (n=rtcg@static-71-244-46-30.dllstx.fios.verizon.net) |
| 16:08.55 | fatgoose | qwell[]: comments? what you need to add if you want the fxo/fxs port? |
| 16:09.11 | Qwell[] | What do you mean, what do you need to add? |
| 16:09.25 | fatgoose | it comes without analog ports right? |
| 16:09.30 | Qwell[] | currently, yes |
| 16:09.57 | Qwell[] | If you want analog ports, wait for the one with the modules |
| 16:09.59 | fatgoose | ok, you can't expand it yet? |
| 16:10.02 | fatgoose | ok |
| 16:10.13 | [TK]D-Fender | fatgoose: Just build a real simple server and save yourself a lot of $ and greif |
| 16:10.15 | fatgoose | you known any similar product? |
| 16:10.18 | Qwell[] | should be "Real Soon Now (TM)" |
| 16:10.31 | [TK]D-Fender | Qwell[]: "Next Spring... Sharp!" |
| 16:10.35 | *** join/#asterisk bryanfe2 (n=chatzill@wsip-70-169-190-173.sb.sd.cox.net) |
| 16:10.51 | fatgoose | [tk]d-fender: don't have time to set it up, then maintain, just want something that work out of the box |
| 16:11.10 | Qwell[] | fatgoose: If I were you, I'd call sales and ask if they know when it will be shipping with analog modules |
| 16:11.39 | fatgoose | ok |
| 16:11.53 | [TK]D-Fender | fatgoose: You think that will configure ITSELF any faster? "Put down the crack-pipe" (c) JerJer |
| 16:12.10 | [TK]D-Fender | fatgoose: Go download Trixbox, and happy trails. |
| 16:12.35 | fatgoose | [tk]d-fender: at least I won't have to deal with hardware issue |
| 16:12.37 | bryanfe2 | hi folks... Can anyone tell me why Asterisk would send RTP audio data back to a sip client's internal IP address (behind a NAT), and not it's external? I have "nat=yes" correctly configured in sip.conf, but still, Asterisk is sending RTP data to the wrong IP. SIP data is going to the correct IP. I've tried this on the latest Asterisk 1.2 and 1.4. |
| 16:12.50 | [TK]D-Fender | fatgoose: Actually... you DO. |
| 16:13.17 | [TK]D-Fender | fatgoose: And the * GUI is not that comprehensive. as far as confg goes, nor as "mature" (*shudder*) |
| 16:13.38 | [TK]D-Fender | bryanfe2: Because there is a lto more you need to setup. Go read : |
| 16:13.41 | [TK]D-Fender | ~sipnat |
| 16:13.41 | jbot | [~sipnat] Quick guide on configuring * + SIP behind NAT : http://www.aocomputing.net/?p=3 otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
| 16:13.43 | [TK]D-Fender | ^^^^^^^^^^^^^^^^^^ |
| 16:13.50 | bryanfe2 | Fender I already went through all that |
| 16:13.54 | bryanfe2 | going crazy |
| 16:14.15 | [TK]D-Fender | bryanfe2: Then if you think that "nat=yes" = everything, that you clearly aren't paying attention. |
| 16:15.01 | bryanfe2 | Fender, what STUN server do people like to use with Asterisk? |
| 16:15.10 | [TK]D-Fender | bryanfe2: * does not SUPPORT STUN |
| 16:15.16 | [TK]D-Fender | bryanfe2: Nor do you need it. |
| 16:16.21 | bryanfe2 | we have this exact same configuration working elsewhere, the only difference I can see is that on the other setup we have a Cisco PIX firewall with a "sip rule" enabled, which seems to be working some magic. |
| 16:16.38 | JT | hahaha |
| 16:16.42 | JT | throw it out |
| 16:16.42 | aikanaro79 | if I dial a sip user can I, afterwards, transfer him to another context and/or extension? |
| 16:16.50 | JT | cisc PIX == absolute arse |
| 16:16.54 | JT | cisco |
| 16:16.59 | bryanfe2 | yeah i know |
| 16:17.04 | JT | cisc PIX with sip enabled mode == double absolute arse |
| 16:17.10 | JT | :) |
| 16:17.44 | bryanfe2 | fender the only real meat of that document is to set nat=yes, canreinvite=no |
| 16:18.02 | Yourname` | Can someone please tell me what does this mean? -> http://pastebin.ca/672468 |
| 16:18.11 | rtcg | What are the piece of the puzzle to get zaptel 1.4.5.1 wctdm and wcfxo drivers loaded in a 2.6 kernel? (I have a 1.4.5.1 created zaptel.rules file in /etc/udev/rules.d and my slackware box previously ran zaptel 1.2.x drivers(I removed the zaptel udev rules I had previously put in the main udev.rules file)) |
| 16:18.14 | JT | and to set cisco pic = removed |
| 16:18.23 | JT | pix |
| 16:18.35 | JT | they completely balls things up in sip aware mode |
| 16:18.44 | bryanfe2 | well it seems to be working better there for some reason |
| 16:19.00 | bryanfe2 | without the PIX (just a box straight on the 'net), asterisk is sending audio data to a sip client's internal IP addr |
| 16:19.02 | rtcg | I see the zaptel drivers on fedora core help page, but that doesn't help my slackware box very well. |
| 16:19.19 | bryanfe2 | even as it is sending SIP traffic to the correct address |
| 16:19.45 | [TK]D-Fender | bryanfe2: No, you clearly missed EXTERNIP <--------------- |
| 16:20.01 | bryanfe2 | i have that set too... |
| 16:20.05 | [TK]D-Fender | bryanfe2: And Cisco PIX NAT = flaming garbage and an extreme risk to functioning. |
| 16:20.31 | [TK]D-Fender | bryanfe2: Pastebin your configs. |
| 16:20.34 | bryanfe2 | i'm trying to get rid of the PIX believe me... but my box with the PIX is working (that is, it talks to SIP clients behind a NAT), and my box without the PIX, is not working.. |
| 16:21.14 | JT | what clients would these be? |
| 16:21.41 | *** join/#asterisk dg (i=dgl@otherwize.co.uk) [NETSPLIT VICTIM] |
| 16:21.53 | bryanfe2 | how do you pastebin? |
| 16:22.18 | JT | ~pb |
| 16:22.19 | jbot | A Pastebin is a place to paste your stuff without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste |
| 16:23.36 | bryanfe2 | fender: http://pastebin.com/d752e8cb0 |
| 16:25.14 | [TK]D-Fender | bryanfe2: You are clearly also missing the LOCALNET clause.... |
| 16:25.28 | [TK]D-Fender | bryanfe2: Which detemines WHEN to use the EXTERNIP |
| 16:25.41 | bryanfe2 | got it fender, let me try that... |
| 16:25.53 | JT | but he knows it all already ;) |
| 16:25.55 | bryanfe2 | however -- isn't that just for my own source addressing? Does it really interact with out the client is addressed? |
| 16:26.04 | JT | yes, it matters. |
| 16:26.05 | bryanfe2 | with how the client is addressed i mean? |
| 16:26.15 | JT | weird assumptions |
| 16:28.21 | bryanfe2 | ok, the correct EXTERNIP settting WORKED |
| 16:28.23 | bryanfe2 | I may be a putz |
| 16:28.50 | bryanfe2 | but I honestly thought that it had to do with how it puts the server's IP address in outbound packets, not how it addresses the clients' IP address. |
| 16:28.54 | [TK]D-Fender | bryanfe2: I hide things in the BIG PRINT :p |
| 16:29.23 | rtcg | hmm.. What would keep the zaptel 1.4.5.1 drivers from installing even after a 'make install' ?? |
| 16:29.43 | rtcg | there are no files in /lib/modules/$kernel/misc/* |
| 16:31.47 | pacneil | this brings up another question. If asterisk is NAT'ed in a DMZ will I still be able to connect from another private network or the DMZ network? |
| 16:31.55 | [TK]D-Fender | rtcg: Perhaps you should pastebin your complete attempt and show some backup in a pastebin... |
| 16:32.08 | [TK]D-Fender | pacneil: ----> |
| 16:32.11 | [TK]D-Fender | ~sipnat |
| 16:32.12 | jbot | [~sipnat] Quick guide on configuring * + SIP behind NAT : http://www.aocomputing.net/?p=3 otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
| 16:32.13 | [TK]D-Fender | ^^^^^^^^^^^^^^^^ |
| 16:33.27 | darkfires | i found a number 905-958-1114 (most of them say out of service) but this one says it wasn't correctly dialed but is a local call.... maybe need some prefix ? |
| 16:33.36 | Strom_M | ^^^^^^^^^^^^^^^^^^ <--------------- --------------------> vvvvvvvvvvvvvvvv |
| 16:33.37 | Yourname` | Hi, can I do AddQueueMember(testq|Agent/${CALLERID(num)}) -> Where the channel is actually Agent other than SIP? |
| 16:34.04 | Qwell[] | darkfires: congratulations - your area is getting a forced 10 digit dial |
| 16:34.14 | Qwell[] | erm, 11 |
| 16:34.24 | Qwell[] | you just found the 11 digit dial test number :P |
| 16:34.37 | pacneil | [TK]D-Fender: very nice, can you now answer the question about DMZ since it't not contained in either of those pages? |
| 16:34.55 | darkfires | qwell haha is that what it really is |
| 16:34.58 | Qwell[] | no |
| 16:35.01 | Qwell[] | but it sounded good, eh? |
| 16:35.22 | pacneil | I run m0n0wall |
| 16:35.25 | [TK]D-Fender | pacneil: yes, the answer IS contained in there. You clearly need a LOT more setting, and not jsut DMZ. It explicitly tells you waht ports are NEEDED, and if you want to throw MORE there, more power to you. |
| 16:35.59 | *** join/#asterisk riddlebox (n=JamesMid@75-128-170-26.static.stls.mo.charter.com) |
| 16:36.03 | riddlebox | hrmm I was given a Quintum ASM400 for free from my boss, it has 4 fxo and 4 fxs ports, but it programs weird, you setup a small huntgroup in the tenor, adding the extensions on the tenor, then you tell them what extensions they are in asterisk, why would you have two different extensions?? |
| 16:39.11 | pacneil | [TK]D-Fender: You tell me where in those two pages there is mention of DMZ solution. I just reread them both. |
| 16:39.23 | pacneil | or don't |
| 16:39.58 | [TK]D-Fender | pacneil: DMZ = all ports forwarded from a NAT-like routed solution. That guide says what you need for * to work behind NAT. Whats the issue? |
| 16:41.45 | Nugget | DMZ doesn't mean "behind nat" |
| 16:42.01 | [TK]D-Fender | Nugget: For all intents and purposes ...."WHATEVER" |
| 16:42.07 | Nugget | uh, not at all. |
| 16:42.31 | Nugget | I'd assume pacneil means he's trying to make asterisk work in a multi-homed environment, which would be the hallmark of a traditional DMZ. |
| 16:42.36 | pacneil | I have a m0n0wall router/firewall. I want to put * in the DMZ, some phones will be in the LAN, others will be behind other NAT routers. The only thing I don't read about is from LAN to DMZ connections |
| 16:43.00 | Nugget | A DMZ is pretty much the opposite of "behind nat" |
| 16:43.01 | pacneil | Nugget: exactly |
| 16:43.07 | pacneil | that too |
| 16:43.31 | rtcg | Would a blank "INSTALL_MOD_PATH= " cause the make script to not know where to install the files? ( results of a make install along with an empty search for the installed drivers - http://pastebin.ca/672499 ) |
| 16:43.53 | Nugget | I have had nothing but problems running asterisk in that manner, I'm sad to say. Not entirely due to asterisk deficiences, but that certainly contributed to my headaches. |
| 16:44.22 | pacneil | It's actually plain routing. Except the firewall intermediates and disallows connections from the DMZ to the LAN except in response to connections from LAN to DMZ |
| 16:44.29 | Nugget | I suggest against trying. You're certainly going a direction that the asterisk developers don't particularly care about if you do it that way |
| 16:45.00 | pacneil | I guess they aren't all that interested in real security then, are they? |
| 16:45.07 | Nugget | when I re-did our offices I switched to a more "normal" static nat mapping instead of a proper dmz and I get a lot less trouble from asterisk. |
| 16:45.16 | Nugget | (using pfsense, which is quite similar to your m0n0wall solution) |
| 16:45.29 | pacneil | I'm aware of pfsense |
| 16:46.00 | pacneil | so then the optimal solution is just to have asterisk in the LAN? |
| 16:46.05 | Nugget | asterisk seems to be fundamentally unaware of the fact that a machine can be on two networks at once, and seems to yield unpredictable results if you try. |
| 16:46.33 | Nugget | things that worked fine would spontaneously stop working when I'd update asterisk to a newer version because there's no testing at all being done for that sort of deployment. |
| 16:46.51 | pacneil | and hope the * developers did their work well in providing no exploits? |
| 16:47.01 | Nugget | to be fair, though, it was the more recent cisco builds that finally made me stop trying. |
| 16:47.07 | Nugget | and that's entirely a cisco issue |
| 16:47.21 | *** join/#asterisk rprince (n=robert@wsip-70-169-190-173.sb.sd.cox.net) |
| 16:47.34 | pacneil | security is just such an inconvenience for so many people |
| 16:47.40 | *** join/#asterisk thansen|laptop (n=thansen@19.243.sfcn.org) |
| 16:48.08 | pacneil | no wonder the internet is being taken over by scoundrels and thieves |
| 16:48.31 | [TK]D-Fender | pacneil: Well if connections go "one-way", you're pretty much DOA. |
| 16:48.47 | pacneil | [TK]D-Fender: configurable |
| 16:49.20 | [TK]D-Fender | pacneil: if it does either way then its not much of a DMZ is it? That'd sound more like the description of "just on another subnet" |
| 16:49.52 | [TK]D-Fender | pacneil: For which you'd only have to set a localnet clause and a gatway. |
| 16:49.54 | Nugget | a simplified example would be where the asterisk box has a public IP and also an IP on an rfc1918 network like 10.0.0.0/24 |
| 16:49.56 | pacneil | I'm still trying to get my head around * |
| 16:50.05 | Nugget | with phones talking to asterisk coming in from both places. |
| 16:50.17 | Nugget | (potentially behind nat, but not necessarily) |
| 16:50.23 | *** join/#asterisk wunderkin (n=wunderki@ip72-223-86-126.ph.ph.cox.net) |
| 16:50.31 | Nugget | asterisk totally fails to handle that situation with any sort of grace or predictability |
| 16:50.43 | *** join/#asterisk Mavvie (n=edwin@ppp121-44-109-49.lns10.syd6.internode.on.net) |
| 16:50.44 | pacneil | specifically Phones in 192.168.0 * in 17.16.1.0 and other interface is public IP providing NAT to DMZ |
| 16:50.59 | Nugget | for a long time it worked as long as you had two physical network interfaces for the differen segments, but didn't work at all if you were cheating by using eth aliases. |
| 16:51.00 | rprince | asteriks folks, I have a coding question (maybe - actually I hope it can be done currently with EAGI). I need to listen for DTMF signals with sub-100 ms timing. |
| 16:51.05 | Nugget | then it didn't work at all for a while |
| 16:51.13 | Nugget | now I think it's sort of working again, for some phones |
| 16:51.17 | rprince | Is this something I'd need to do in a separate app, passing the audio stream using EAGI? |
| 16:51.46 | Nugget | When I was running that way I faced an endless stream of weird problems where the internal IP would "bleed out" in SIP packets destined for the external interface (and vice versa) |
| 16:51.48 | pacneil | aliases are a bad implimentation. I've never done it that way. You can't trust the separation that aliases provide |
| 16:51.56 | [TK]D-Fender | Nugget: My work * box has 2 NIC's each with their own subnet, I have 2 other VPN's subnets they acecss through 1 of those 2, AND is behind NAT on the same tot he public internet. All works fine. |
| 16:51.58 | Nugget | agreed, I'm just providing context. |
| 16:53.13 | *** join/#asterisk qdk (n=qdk@0x50c627be.bynxx11.adsl-dhcp.tele.dk) |
| 16:53.28 | pacneil | VPNs should be not problem, they should be transparent if they're implimented right. |
| 16:53.43 | rprince | I've looked at the code in .../apps, and I don't see anything specific to what I'm trying to do. |
| 16:54.41 | pacneil | I presume that if that works, as long as my firewall rules are right then, DMZ/LAN should be essentially the same |
| 16:55.30 | pacneil | getting the rules right for DMZ -> LAN is going to be the trick, I guess. |
| 16:57.00 | Yourname` | Ok, weird. Using addqueuemember, I added members to the queue. But when I do agent show on the CLI, it says 0 online.. and makrs everyone as not logged in. |
| 16:57.07 | Yourname` | What's going bad again? |
| 16:57.31 | *** join/#asterisk h0 (n=Fakhir@unaffiliated/fakhir) |
| 16:57.42 | pacneil | Is there a debugger of some kind for * that allows you to check files for syntax errors and the like? |
| 16:57.55 | Nugget | [TK]D-Fender: curious, is that 1.2 or 1.4? |
| 16:58.12 | [TK]D-Fender | Nugget: 1.2 for this box |
| 16:58.15 | Nugget | *nod* |
| 16:58.17 | Nugget | interesting. |
| 16:58.28 | *** join/#asterisk chemikk (i=abap@real.wilbury.sk) |
| 16:58.35 | *** join/#asterisk scardinal (n=supreme@port816.ds1-suoe.adsl.cybercity.dk) |
| 16:58.40 | chemikk | hello |
| 16:59.50 | Yourname` | So no help on queues today at all eh.. :( |
| 17:00.20 | *** join/#asterisk gardo (n=gardo@121.97.214.226) |
| 17:00.55 | chemikk | anybody here speaking czech? |
| 17:01.09 | jingles | never while I'm sober. |
| 17:01.16 | [TK]D-Fender | Yourname`: pastebin = your friend. |
| 17:01.23 | Yourname` | I did, twice |
| 17:01.41 | Yourname` | First one was this.. where I didn't know how to interpret http://pastebin.ca/672468 |
| 17:01.47 | Yourname` | But let me do it again. |
| 17:01.51 | darkfires | OMG I FOUND IT |
| 17:01.55 | darkfires | Strom_M |
| 17:02.30 | ManxPower | and the problem was? |
| 17:03.08 | chemikk | a have problem with configuring voice menu, i meen problem with sending DTMF |
| 17:03.53 | [TK]D-Fender | Yourname`: and now "sip show peers" |
| 17:04.15 | Yourname` | [TK]D-Fender: Pastebinning everything for you.. one sec |
| 17:04.38 | chemikk | voice menu i functional localy but no when i calling from outside |
| 17:05.35 | *** join/#asterisk nclx (n=nightcal@carnivore.scrapshells.com) |
| 17:06.29 | Strom_M | darkfires: cool |
| 17:06.39 | Strom_M | are you sure it's in your switch? |
| 17:06.57 | darkfires | its in the same prefix as he used to check other shit on my line |
| 17:07.16 | Dr-Linux | remind me please, what's the cisco's 7960 new phone default password? |
| 17:07.21 | Yourname` | [TK]D-Fender: http://pastebin.ca/672521 -> queues.conf , http://pastebin.ca/672522 - agents.conf, http://pastebin.ca/672524 - extensions, http://pastebin.ca/672527 - cli commands |
| 17:07.25 | Dr-Linux | is it "*##" ? |
| 17:07.46 | Strom_M | darkfires: coolness |
| 17:08.22 | Qwell[] | Dr-Linux: it's different on sip and sccp - I think sccp is **# at the menu, and on sip, there is an unlock option in settings, and the password is "cisco" |
| 17:08.22 | Dr-Linux | Qwell[]: new often comes with SCCP |
| 17:08.27 | Dr-Linux | opss yeah gotcha |
| 17:08.39 | Dr-Linux | i was looking for **# |
| 17:08.42 | Dr-Linux | Thanks .. |
| 17:08.44 | Dr-Linux | hhm.. |
| 17:08.53 | Dr-Linux | Qwell: any good news about 7935 with 1.4.x? :) |
| 17:13.11 | *** join/#asterisk MdeP (n=mdep@200.124.36.28) |
| 17:13.29 | darkfires | is there anyway to turn off hpec without recompiling zaptel |
| 17:14.25 | *** join/#asterisk Op3r (n=Op3r@121.97.247.27) |
| 17:14.26 | tzanger | darkfires: echocancel=no? |
| 17:14.53 | Qwell[] | darkfires: no.. |
| 17:15.02 | Qwell[] | oh, there is that I guess |
| 17:15.11 | Qwell[] | I thought he meant switch to something else |
| 17:15.37 | darkfires | tzanger doesnt work |
| 17:15.48 | Qwell[] | darkfires: What did support say when you called? |
| 17:16.24 | darkfires | what do u mean |
| 17:16.32 | Qwell[] | You said it was crashing, right? |
| 17:17.03 | *** join/#asterisk _ShrikE (n=ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
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| 17:17.13 | darkfires | it was yeah but i havn't been able to reproduce it ... now i have a milliwatt test thing i wanna try tuning these gains so maybe i wont have to use hpec |
| 17:17.49 | darkfires | im gonna get fired if the phone system keeps locking up |
| 17:17.49 | darkfires | haha |
| 17:18.00 | *** join/#asterisk bryanfe2 (n=chatzill@wsip-70-169-190-173.sb.sd.cox.net) |
| 17:18.43 | bryanfe2 | does anyone know if there is an API call in Asterisk, for module developers, to wait for a specific tone? i.e. "wait for a 2000 hz tone lasting 1 second, then return"? |
| 17:20.59 | *** join/#asterisk hyphen (n=hyphen@dsl081-022-042.phl1.dsl.speakeasy.net) |
| 17:22.10 | [TK]D-Fender | *b00m* |
| 17:22.10 | outtolunc | weee |
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| 17:22.41 | *** mode/#asterisk [+o russellb] by irc.freenode.net |
| 17:24.05 | Dan0maN_Work | <PROTECTED> |
| 17:24.21 | Yourname` | [TK]D-Fender: Did you get my pastebins? |
| 17:24.41 | ManxPower | darkfires: many people just go back to 1.2. |
| 17:24.57 | ManxPower | A production enviroment is no place to test out software. |
| 17:25.01 | *** join/#asterisk umdstu_ (n=rfid@mobile-166-217-248-211.mycingular.net) |
| 17:25.56 | Qwell[] | Dan0maN_Work: eh, that was a tiny split |
| 17:26.12 | [TK]D-Fender | Yourname`: I don't see where the "SIP/" ones are coming from... |
| 17:26.24 | Qwell[] | real EFNet style splits unsync every server |
| 17:26.40 | [TK]D-Fender | Yourname`: Oh, jsut found them |
| 17:26.44 | Yourname` | k |
| 17:27.33 | [TK]D-Fender | Yourname`: I don't think app_queue likes the fact they are unmonitored... |
| 17:27.49 | chemikk | voice menu functional only localy not when i calling from outside, this si my extensions.conf: http://pastebin.ca/672552 |
| 17:27.57 | chemikk | please help |
| 17:28.05 | *** join/#asterisk techie (n=techie@c-67-181-184-170.hsd1.ca.comcast.net) |
| 17:28.06 | ManxPower | chemikk: how are the outside calls getting into Asterisk? |
| 17:28.21 | chemikk | ManxPower: from mobile phone |
| 17:28.37 | *** join/#asterisk fordfrog (n=fordfrog@gentoo/developer/fordfrog) |
| 17:28.39 | chemikk | ManxPower: or normal telephone, sorry my english very bad |
| 17:29.11 | Dr-Linux | Qwell[]: do you remember the to Disable default cisco call manager? |
| 17:29.14 | Dan0maN_Work | Qwell: i dont hang out much on irc. this is the largest room i've really hung around in |
| 17:29.16 | *** join/#asterisk jsmith (n=jsmith@000-143-916.area3.spcsdns.net) |
| 17:29.16 | *** mode/#asterisk [+o jsmith] by ChanServ |
| 17:29.21 | *** join/#asterisk brodiem (n=brodiem@67.110.68.66.ptr.us.xo.net) |
| 17:29.25 | Dr-Linux | actually it's try to find IP address from the Cisco CM |
| 17:29.44 | Dr-Linux | Qwell[]: i'm working on this phone remotely :) |
| 17:29.49 | Yourname` | [TK]D-Fender: lol wanna keep it that way |
| 17:30.14 | *** join/#asterisk guillote_GNU (n=guillote@190.7.27.17) |
| 17:32.04 | chemikk | which version asterisk you recommends?, i have 1.4.10.1 |
| 17:32.17 | Qwell[] | chemikk: 1.4.11 |
| 17:33.48 | *** join/#asterisk Aeudian (n=somewher@ip67-94-157-98.z157-94-67.customer.algx.net) |
| 17:34.19 | chemikk | and this version will fix my problem? :) |
| 17:34.30 | Qwell[] | I don't know what your problem is |
| 17:35.41 | chemikk | Qwell[]: my problem is voice menu, which is not functional when i calling from outside not accepted DTFM maybe, http://pastebin.ca/672552 |
| 17:37.33 | Dr-Linux | Qwell[]: the the 7960 phone is new and i want to disbale the Cisco callmanager, so it should not try to get IP from Cisco call manager |
| 17:39.19 | [TK]D-Fender | chemikk: "outside" doesn't say anything useful. We need to know what HARDWARE is involved, and to see the configuration of that channel. |
| 17:39.35 | [TK]D-Fender | chemikk: Pastbin your other configs and the CLI output of a failed attempt at verbose 10 |
| 17:39.37 | [TK]D-Fender | ~pb |
| 17:39.38 | jbot | A Pastebin is a place to paste your stuff without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste |
| 17:39.39 | [TK]D-Fender | ^^^^^^^^^^^^^^^^^^ |
| 17:42.53 | *** join/#asterisk vader-- (n=me@204.183.88.101) |
| 17:42.58 | vader-- | hello hello |
| 17:43.15 | vader-- | having some weird issues with a TDM2400 card that im using to power some analog phones |
| 17:43.54 | Dr-Linux | anybody know how can i disable Call manager on cisco 7960? |
| 17:44.02 | vader-- | the card seems to work fine but twice now i have had an issue where one channel on the card won't pick up a dialtone when you pick up the line |
| 17:44.13 | vader-- | if you call the channel the line is instantly picked up |
| 17:44.16 | vader-- | but you can't dial out |
| 17:44.21 | vader-- | there is no dial tone |
| 17:44.35 | vader-- | the only way i have seen to reset this problem is by powering off the server and back on |
| 17:44.48 | brodiem | To have MixMonitor store in g729 format, do I need to do anything more than specify the filename with a ".g729" ext? |
| 17:45.22 | [TK]D-Fender | brodiem: I would say a G.729 license to spare as well... |
| 17:45.39 | vader-- | tdk any ideas what i could try to reset this card without taking the whole telephone system down |
| 17:45.45 | brodiem | [TK]D-Fender ok yes besides that.. |
| 17:46.24 | [TK]D-Fender | vader--: Nope, I've had that happen before with FXO's on the same card.... inexplicable, and was part of what forced its replacement. |
| 17:46.35 | [TK]D-Fender | brodiem: Nothing I'm aware of. |
| 17:46.40 | *** part/#asterisk fatgoose (n=fg@206-248-135-39.dsl.teksavvy.com) |
| 17:46.58 | vader-- | ya the card was running for 230+ days when it flaked out like that |
| 17:47.18 | brodiem | [TK]D-Fender I'm trying to figure out why my g729 consumption is so high. If I make a call from one SIP ext to another, using g729 on both legs of the call and using MixMonitor w/ g729 format, it uses 1 enc and 2 dec |
| 17:47.22 | [TK]D-Fender | vader--: My clients we never so lucky :) |
| 17:47.57 | vader-- | ya now it's been up 62 days and this card decided to flake |
| 17:48.03 | [TK]D-Fender | brodiem: Because its decogin BOTH ends. |
| 17:48.11 | [TK]D-Fender | decoding* |
| 17:48.27 | vader-- | did you replace it with another card or the same card? |
| 17:48.48 | vader-- | i wish there was a way to reload the card |
| 17:48.51 | vader-- | or something |
| 17:48.55 | Qwell[] | brodiem: seems normal to me |
| 17:48.56 | [TK]D-Fender | vader--: You already know what happens in these cases :) |
| 17:48.59 | brodiem | [TK]D-Fender I thought if I'm going to save it _as_ g729 that I wouldn't need to use up decoder licenses since the stream is already 729? |
| 17:49.05 | vader-- | whats that |
| 17:49.07 | vader-- | ? |
| 17:49.12 | Qwell[] | brodiem: In order to mix, it has to go to signed linear first |
| 17:49.18 | brodiem | ahh |
| 17:49.19 | [TK]D-Fender | ^^^^^^^^^^^^^^^^^^^ |
| 17:49.42 | [TK]D-Fender | brodiem: Expect the same with MeetMe, etc... |
| 17:49.51 | *** join/#asterisk brodiem (n=brodiem@67.110.68.66.ptr.us.xo.net) |
| 17:50.03 | brodiem | damnit I hate when I hit ctrl-z in a BX session, kills it every time =/ |
| 17:50.18 | Qwell[] | brodiem: there's a simple fix for that... |
| 17:50.24 | Qwell[] | irssi :P |
| 17:50.51 | Qwell[] | I used to be a bx fanatic... then I used irssi once. |
| 17:51.04 | darkfires | BitchX-1.1-final+ by panasync - Linux 2.6.22.1 |
| 17:51.44 | brodiem | is it a gui or console? |
| 17:51.47 | Qwell[] | console |
| 17:51.57 | brodiem | cool |
| 17:52.04 | Qwell[] | it's pretty much the most awesome thing ever |
| 17:52.12 | brodiem | lol |
| 17:52.30 | brodiem | hmm |
| 17:53.04 | Qwell[] | You know how they say some things are the best thing since sliced bread? |
| 17:53.18 | Qwell[] | well, this is more awesome that sliced bread. That's why they say "since sliced bread" |
| 17:53.32 | Qwell[] | </exaggeration> |
| 17:53.40 | brodiem | haha |
| 17:53.42 | brodiem | I'll be sure to try it |
| 17:54.11 | brodiem | so I suppose there is no way to just nativevly store g729 without doing any transcoding? |
| 17:54.19 | Qwell[] | record |
| 17:54.30 | Qwell[] | mixmonitor actually mixes the audio, so has to go to slin first |
| 17:55.08 | Qwell[] | actually, maybe not record.. heh |
| 17:55.19 | Qwell[] | possibly monitor (without doing mixing) - I'm not sure quite how that works |
| 17:55.37 | *** join/#asterisk drwelby (n=mpfister@mail.enplan.com) |
| 17:56.22 | brodiem | won't hurt to try, doing it now |
| 17:56.26 | *** join/#asterisk tzafrir_laptop (n=tzafrir@bzq-79-180-14-100.red.bezeqint.net) |
| 17:56.42 | rtcg | under what circumstance would running zaptel's 'make install' cause the creation of new kernel module directory with '-ThisDoesNotExist' appended to it? (example: /lib/modules/2.6.17.13-ThisDoesNotExist ) |
| 17:57.21 | rtcg | is that a function of the make install or some other process ....like depmod ? |
| 17:57.29 | brodiem | aha! 0/0 used |
| 17:58.21 | brodiem | except it seems it can't mix them using 729 =/ |
| 17:59.03 | [TK]D-Fender | brodiem: To mix you have to decode/encode. End of story. |
| 17:59.52 | brodiem | i guess that makes sense, I was just thinking soxmix could do it so that it would be external but guess not |
| 18:02.05 | [TK]D-Fender | brodiem: Not sure what it takes for SOX to use G.729 natively. |
| 18:05.06 | *** join/#asterisk oej (n=olle@soll4-125.cust.blixtvik.net) |
| 18:05.23 | rtcg | How can I trace why zaptel's "make install" isn't installing the drivers in the /lib/modules/$KVERS directory? |
| 18:06.01 | [TK]D-Fender | rtcg: pastebin it and "uname -a" |
| 18:07.01 | variable_office | anyone using voicemail odbc storage with postgresql? |
| 18:08.26 | rtcg | [TK]D-Fender: drivers install into /lib/modules/2.6.17.13-ThisDoesNotExist http://pastebin.ca/672601 |
| 18:10.03 | *** join/#asterisk waKKu (n=eurulo@unaffiliated/wakku) |
| 18:10.08 | errr | Im having a problem with a number where when I dial it it goes out Zap/g0 then rings like 6 times, then I get a busy signal.. any idea what causes that? |
| 18:10.24 | lirakis | can anyone reccomend a good queue statistics package?.. preferably open source |
| 18:11.03 | [TK]D-Fender | rtcg: "uname -a" please... |
| 18:11.16 | [TK]D-Fender | errr: PASTEBIN <----- |
| 18:11.26 | lirakis | qmetrics .. is servlet based .. and i dont want to run tomcat or some junk like that... and *guru's queuestats is really poorly documented and runs on postgre... and id rather not run 2 databases for essentially no reason |
| 18:11.35 | errr | [TK]D-Fender: what do you want me to pastebin? |
| 18:11.54 | [TK]D-Fender | errr: the call at verbose 10 w/ channel debug and your channel configs clearly... |
| 18:12.15 | rtcg | [TK]D-Fender: Linux phone1 2.6.17.13 #1 Sat Sep 9 01:11:49 CDT 2006 i686 athlon-4 i386 GNU/Linux |
| 18:13.47 | errr | [TK]D-Fender: I have so many calls at once going on I cant get most of this. Is there anyway to not have to get this info from the cli? |
| 18:14.17 | errr | [TK]D-Fender: or to make all the other calls not show up in the cli and only this one |
| 18:14.30 | [TK]D-Fender | rtcg: did you provide any parms to "./configure" or "make"? |
| 18:14.44 | rtcg | [TK]D-Fender: no |
| 18:14.45 | [TK]D-Fender | errr: Let me sort that out |
| 18:15.11 | [TK]D-Fender | rtcg: I'd suggest redoing those steps and pastebin-ing the full process |
| 18:15.40 | rtcg | [TK]D-Fender: the first time around, I ran make menuconfig...but I've since wiped the directory and ran them 'plain-jane' |
| 18:15.53 | rtcg | them = ./configure && make && make install |
| 18:16.04 | [TK]D-Fender | rtcg: Trasht he folder and start from scratch |
| 18:16.09 | rtcg | doing it. |
| 18:16.16 | [TK]D-Fender | rtcg: do them seperately and pastebin them all |
| 18:17.59 | *** join/#asterisk Netgeeks (n=chris@pbx5.netgeeks.net) |
| 18:18.46 | AirCoder | any one get nvfaxdetect working on 1.4.11? I have it compiled but nvfax hangs on detection on 2 sip connections ive tested.. |
| 18:20.48 | variable_office | i wanted to go in and install odbc storage; i already have asterisk running and compiled and now i need to recompile it, what directory should i go into to do this? |
| 18:22.05 | jsmith | variable_office: Go into the Asterisk source directory, re-run "make menuselect", and choose the option there |
| 18:22.54 | variable_office | jsmith for 1.2 this work too? |
| 18:23.09 | jsmith | variable_office: No, ODBC voicemail storage only works on Asterisk 1.4 |
| 18:23.17 | rtcg | [TK]D-Fender: Do you want the config.status file in the pastebin? |
| 18:23.19 | jsmith | variable_office: At least, as far as I recall |
| 18:23.24 | variable_office | online it says it works in 1.2 as well |
| 18:23.28 | [TK]D-Fender | rtcg: More > less |
| 18:23.31 | variable_office | http://www.voip-info.org/wiki/view/Asterisk+Voicemail+ODBC+storage |
| 18:24.00 | jsmith | Oh, in that case, I think you have to change the source code in app_voicemail.c to enable it. |
| 18:25.35 | variable_office | jsmith it says i just have to edit the makefile in apps and rebuild; i found the lines in the makefule, but i dont know what directory i should be in to do that actual make? the same directory as the makefile? |
| 18:25.53 | jsmith | No, in the top Asterisk directory |
| 18:25.58 | jsmith | (not in the apps directory) |
| 18:26.43 | variable_office | so i edit the makefile in /usr/src/asterisk/apps but run make in /usr/src/asterisk/ ? |
| 18:31.06 | *** join/#asterisk darkfires (n=lwhite@d38-37-41.commercial1.cgocable.net) |
| 18:31.32 | darkfires | bah.... with hpec enabled, machine locked up again on an incoming call... just completely froze unresponsive |
| 18:33.01 | wunderkin | what version of hpec? |
| 18:33.42 | darkfires | hpec-9.00.003-athlon |
| 18:34.09 | wunderkin | i figured that still hasn't been fixed |
| 18:34.24 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 18:34.29 | darkfires | so you're saying theres nothing i can do about it |
| 18:34.37 | rtcg | [TK]D-Fender: http://pastebin.ca/672617 Here's the complete paste of the entire process. I've grepped for the phrase "ThisDoesNotExist" and have yet to find WHAT is supplying that phrase to whatever process is appending that when creating a new kernel version directory in /lib/modules |
| 18:34.44 | *** join/#asterisk jtknapp (n=skip@65-126-63-1.dia.static.qwest.net) |
| 18:35.13 | wunderkin | i don't know, good luck, i never get anywhere with... those people |
| 18:36.19 | darkfires | that sucks |
| 18:36.54 | darkfires | it locks up so hard software watchdog doesn't even reboot it |
| 18:38.01 | brodiem | [TK]D-Fender & Qwell[]: FYI I found a good solution for keeping the 729 licenses down with recording.. 1) Use Monitor() to record g729 in/out streams, then once the recording is complete 2) use CLI convert to convert them to Wav49 (takes only a few hundred ms), and 3) execute soxmix to combine into one stream |
| 18:38.13 | [TK]D-Fender | rtcg: DEPMOD 2.6.17.13-ThisDoesNotExist <? |
| 18:38.26 | rtcg | [TK]D-Fender: EXACTLY!!!! |
| 18:38.38 | [TK]D-Fender | rtcg: I might suggest moving files around manually |
| 18:39.00 | rtcg | HAHAHAH and I just did that.. and was able to get the modules installed (after a udev restart) |
| 18:39.14 | rtcg | ug! any CLUE as to WTF is going on?? |
| 18:39.50 | *** join/#asterisk bm-chs (n=bradley@64-151-58-154.static.everestkc.net) |
| 18:40.02 | *** join/#asterisk NightMonkey (n=NightMon@pdpc/supporter/sustaining/NightMonkey) |
| 18:40.12 | darkfires | wunderkin so you just dont have echo cancellation or something |
| 18:40.18 | [TK]D-Fender | rtcg: Try another version and see what happens :) |
| 18:41.01 | rtcg | will do... |
| 18:41.32 | wunderkin | darkfires, not atm, it is up to the client to get a card w/ ec now, should have started that way anyway but oh well, hpec worked well when it worked but we would have license key 'leaks' and kernel panics so f that |
| 18:41.58 | *** join/#asterisk mog (n=mog@c-71-207-200-130.hsd1.al.comcast.net) |
| 18:41.58 | *** mode/#asterisk [+o mog] by ChanServ |
| 18:42.35 | bm-chs | Has anyone ran into problems accessing AsteriskNow through a router? |
| 18:43.46 | darkfires | wunderkin what cards do you use that include ec |
| 18:44.12 | [TK]D-Fender | bm-chs: Yeah.. I have trouble fitting my arm through those little ventilation holes to get to the other side myself! |
| 18:44.31 | *** join/#asterisk johann8384 (n=johann83@gateway.myogre.com) |
| 18:45.38 | AirCoder | any one get nvfaxdetect working on 1.4.11? I have it compiled but nvfax hangs on detection on 2 sip connections ive tested..? |
| 18:45.43 | bm-chs | LOL. Uh, yea. For some reason I can't get it to respond to even the web interface through a Linksys router . . . thought maybe AT&T was blocking ports, but I can see in the router interface that requests are coming in, they just die soemwhere. |
| 18:46.24 | AirCoder | you try port forwarding? |
| 18:46.48 | holiday_42 | bm-chs, are you accessing from the outside or inside? |
| 18:46.50 | bm-chs | I can point port 80 to my Snom phone and see everything there, so that eliminates AT&T. |
| 18:46.54 | bm-chs | Outside |
| 18:47.29 | AirCoder | bm-chs can you DMS the ip address of the asterisk box? |
| 18:47.39 | holiday_42 | bm-chs, heh, does your asterisk box have correct gateway? |
| 18:47.39 | [TK]D-Fender | bm-chs: Have you checked the http server to see what ip/range it will ACCEPT connections from? |
| 18:47.43 | wunderkin | darkfires, i haven't ever used one with ec but it depends on how many ports you need |
| 18:47.43 | AirCoder | errr dmz |
| 18:48.33 | holiday_42 | bm-chs, check errant firewall rules on asterisk box too |
| 18:48.36 | bm-chs | Hmm - don't want to stick my * box in dmz. |
| 18:48.47 | *** join/#asterisk Lucky7 (n=Adam@207.200.28.175) |
| 18:48.54 | Lucky7 | http://rafb.net/p/DL3lTD90.html |
| 18:49.03 | AirCoder | just to see if you can hit it. |
| 18:49.10 | Lucky7 | when i call into the IVR, it plays the message and then immediately hangs up |
| 18:49.10 | AirCoder | i wouldnt leave dmz on perm. |
| 18:49.17 | Lucky7 | i'm going to assume its because of the auto fallthrough |
| 18:49.19 | bm-chs | I can't ssh to that either. |
| 18:49.26 | Lucky7 | any idea on why that would be happeneing |
| 18:49.58 | variable_office | i am trying to do odbc voicemail storage, but postgresql doesnt have type="blob" what can i make this instead? |
| 18:50.13 | [TK]D-Fender | Lucky7: == Auto fallthrough, channel 'SIP/140-b78500d8' status is 'UNKNOWN' <----------------- |
| 18:51.58 | [TK]D-Fender | Lucky7: you need to set "autofallthrough=no" in [general] |
| 18:52.12 | brodiem | or fix the dial plan so it doesn't run off the end |
| 18:54.41 | darkfires | wunderkin just 2... i have a tdm400p |
| 18:54.44 | rtcg | [TK]D-Fender: Well, zaptel versions 1.4.4 and 1.4.1 also do the weird "/lib/module/$KVERS-ThisDoesNotExist" path creation thing. Do you think this a bug in the zaptel install..or some other OS related bug? |
| 18:56.09 | brodiem | anyone know of a sip provider that will do orig/term t38? |
| 18:56.14 | [TK]D-Fender | rtcg: To be honest, I'm really not sure, and EVERYTHING is suspect..... I was hoping something might have stood out more, and that someone else may have noticed something more subtle that I'd ahev missed in asking you to provide all of that |
| 18:56.17 | brodiem | pay as you go |
| 18:56.48 | Lucky7 | [TK]D-Fender > in "general" which file is this context in? |
| 18:56.49 | rtcg | brodiem: premiervoice.net (t38). |
| 18:57.01 | rtcg | brodiem: ...I think.... |
| 18:57.05 | rtcg | check em out anyway. |
| 18:57.14 | brodiem | ty |
| 18:57.18 | [TK]D-Fender | Lucky7: extensions.conf |
| 18:57.20 | wunderkin | darkfires, i dunno i don't use the analog cards, i guess they don't have an echo can module for those and hpec is the 'solution' for that, they need to fix these problems somehow... |
| 18:57.27 | Lucky7 | ah, found it |
| 18:57.30 | Lucky7 | yea, thanks |
| 18:57.55 | Qwell[] | nothing we can do about it if nobody reports the problem to support |
| 18:58.09 | wunderkin | i have |
| 18:58.15 | AirCoder | any one get nvfaxdetect working on 1.4.11? I have it compiled but nvfax hangs on detection on 2 sip connections ive tested..? |
| 18:58.21 | wunderkin | you probably forgot about my ranting from months ago |
| 18:58.27 | darkfires | qwell what # should I call to report this |
| 18:58.34 | Qwell[] | darkfires: the normal support number |
| 18:58.42 | brodiem | AirCoder i have it running on 1.4.10.1 |
| 18:58.46 | wunderkin | darkfires, i only see an echo can module option for the 24 port analog cards, hpec uses too much cpu for > 8 calls |
| 18:59.46 | *** part/#asterisk brodiem (n=brodiem@67.110.68.66.ptr.us.xo.net) |
| 19:00.19 | *** join/#asterisk brodiem (n=brodiem@67.110.68.66.ptr.us.xo.net) |
| 19:00.34 | darkfires | has anyone had any luck with rhino fxo cards with ec built in |
| 19:01.03 | darkfires | ive had enough of digium cards |
| 19:01.04 | darkfires | heh |
| 19:01.19 | *** join/#asterisk bryanfe2 (n=chatzill@wsip-70-169-190-173.sb.sd.cox.net) |
| 19:02.00 | [TK]D-Fender | darkfires: I haven't heard any field reports on Rhino yet, but use Sangoma exclusively myself. |
| 19:02.02 | bryanfe2 | Guys, is there a module or other piece of software I could use (i.e. from within the Dialplan) to "wait for a specific tone" before continuing? i.e. "wait for 2000hz"? |
| 19:02.37 | *** join/#asterisk taupin974 (n=taupin97@89.237.79.244) |
| 19:03.19 | *** join/#asterisk masterisk (n=mascool@70.88.122.205) |
| 19:03.28 | *** join/#asterisk brodiem (n=brodiem@67.110.68.66.ptr.us.xo.net) |
| 19:04.35 | [TK]D-Fender | bryanfe2: I'm betting you'd have to mod up the dtmf detection code for that. |
| 19:04.52 | [TK]D-Fender | bryanfe2: Definately can't picture anything pre-existing. |
| 19:05.02 | *** join/#asterisk taupin974 (n=taupin97@89.237.79.244) |
| 19:05.27 | bryanfe2 | fender - wow I'm surprised after so much asterisk use, nobody else has needed a "wait for specific tone" module |
| 19:05.39 | wunderkin | darkfires, if you do call digium about the hpec kernel panics, please reference case 8664 so they know you aren't the only person |
| 19:05.51 | bryanfe2 | any developers for hire out there who could help me with this? |
| 19:06.18 | variable_office | jsmith i saw you posted in the bug report for postgresql not playing back voicemail, do you have this working |
| 19:07.39 | *** join/#asterisk Katty (n=Katty@hera.copi-rite.com) |
| 19:07.42 | Katty | allo. |
| 19:07.50 | *** join/#asterisk iBuMp (n=ibump@cpe-66-68-37-190.austin.res.rr.com) |
| 19:07.59 | [TK]D-Fender | bryanfe2: Whats so special about that specific tone? |
| 19:08.09 | *** join/#asterisk taupin974 (n=taupin97@89.237.79.244) |
| 19:08.10 | bryanfe2 | fender - it's special to my application ;) |
| 19:08.10 | [TK]D-Fender | Katty: Mew. |
| 19:08.11 | *** join/#asterisk ReDNeQ- (n=ibump@cpe-66-68-37-190.austin.res.rr.com) |
| 19:08.22 | bryanfe2 | we have to wait for it, then play an audio file. |
| 19:08.24 | wunderkin | the brown noise? |
| 19:08.37 | bryanfe2 | it's not a DTMF tone |
| 19:08.44 | [TK]D-Fender | bryanfe2: Well * was built with "real world" stuff in mind, and "special to my customer hardware/app/whatever" clearly didn't factor in :) |
| 19:08.57 | jsmith | variable_office: Yes, it's working great. |
| 19:08.58 | bryanfe2 | i know.. |
| 19:08.59 | Katty | [TK]D-Fender: herro (= |
| 19:09.00 | *** join/#asterisk taupin974 (n=taupin97@89.237.79.244) |
| 19:09.11 | jsmith | variable_office: Look at the instructions in the doc/ subdirectory of the Asterisk 1.4 source |
| 19:09.18 | bryanfe2 | was just wondering if there was a general purpose module, or API call within Asterisk itself, which could help with this. |
| 19:09.23 | jsmith | variable_office: I explain how to setup the tables, the triggers, etc. |
| 19:09.34 | variable_office | does it only work in 1.4 or should i still be fine with 1.2? |
| 19:09.46 | jsmith | It should work in 1.2 as well. |
| 19:09.56 | jsmith | (again, as long as you've recompiled Asterisk with ODBC voicemail support) |
| 19:10.13 | variable_office | yep, everything is working but it wont playback |
| 19:10.56 | jsmith | variable_office: Did you look at those instructions? |
| 19:11.08 | *** join/#asterisk ReDNeQ (n=ibump@cpe-66-68-37-190.austin.res.rr.com) |
| 19:11.14 | Katty | [TK]D-Fender: do you know how to make samba share out a directory to a samba user? |
| 19:11.28 | Katty | [TK]D-Fender: so my windows people can get to the server phone logs. |
| 19:11.34 | variable_office | jsmith not yet, going to now, i had been reading the old instructions in 1.2 |
| 19:11.36 | *** join/#asterisk iBuMp- (n=ibump@cpe-66-68-37-190.austin.res.rr.com) |
| 19:12.32 | [TK]D-Fender | Katty: Thats a wildly-Google-able topic with a million easy guides out there.. Go pick one ;) |
| 19:13.07 | *** join/#asterisk taupin974 (n=taupin97@89.237.79.244) |
| 19:13.12 | Katty | kay :) |
| 19:13.15 | Katty | i'm already reading one hehe |
| 19:14.19 | *** join/#asterisk taupin974 (n=taupin97@89.237.79.244) |
| 19:14.41 | bm-chs | Anyone familiar with security on AsteriskNow install? Does it do some weird security things? |
| 19:15.08 | bm-chs | I turned off iptables, thinking that might be it, still no love. |
| 19:16.52 | jsmith | bm-chs: What exactly are you trying to do? |
| 19:17.37 | bm-chs | I want to remotely administer, and at the very least register a phone to that box from the outside world. |
| 19:18.00 | jsmith | OK, that should work. |
| 19:18.29 | bm-chs | It's like the Asterisk box isn't responding to any requests from the router inward . . . . ??? |
| 19:18.43 | variable_office | jsmith so the fix is to insert those pgsql functions? is the problem fixed in newer versions of asterisk where you dont need those functions, or do you always need those functions? |
| 19:18.55 | jsmith | bm-chs: Did you set the default gateway in the IP configuration |
| 19:19.04 | jsmith | variable_office: You always need those functions |
| 19:19.15 | jsmith | variable_office: (because of the way PostgreSQL handles it's large objects) |
| 19:19.37 | bm-chs | Hmm -- I did . . . but come to think of it . . . I think I changed the IP of the router. |
| 19:19.50 | bm-chs | What file do I need to tweak to point it to the right place? |
| 19:20.20 | jsmith | Not sure... to be honest, I haven't done much with AsteriskNOW |
| 19:21.33 | variable_office | jsmith will that sql statements you gave work in postgres 7.4? |
| 19:21.46 | variable_office | i got error at char ';' @ 769 |
| 19:21.57 | bm-chs | resolv.conf file had 192.168.1.1 but my box is at .2 . . . |
| 19:21.59 | variable_office | or rather "ERROR: syntax error at or near ";" at character 769"; |
| 19:22.07 | jsmith | variable_office: Hmmmn... they should work. To be honest, I only tried them on 8.x, but someone else said they worked on 7.4 |
| 19:22.21 | jsmith | bm-chs: OK, that's only for DNS resolution |
| 19:22.43 | jsmith | bm-chs: I don't know how you set that up on Rpath linux... /etc/sysconfig/network maybe? |
| 19:22.44 | variable_office | jsmith whats the best way to figure out what char 769 is? |
| 19:23.02 | jsmith | variable_office: No, that's saying *line* 769, I think. |
| 19:23.20 | variable_office | oh, but it said character 769 |
| 19:23.43 | variable_office | theres not even 769 lines in that statement |
| 19:23.44 | bm-chs | ok . . .looking into that also. |
| 19:23.45 | [TK]D-Fender | bm-chs: read this : |
| 19:23.47 | [TK]D-Fender | ~sipnat |
| 19:23.48 | jbot | [~sipnat] Quick guide on configuring * + SIP behind NAT : http://www.aocomputing.net/?p=3 otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
| 19:23.49 | [TK]D-Fender | ^^^^^^^^^^^ |
| 19:24.15 | bm-chs | Many thanks!! |
| 19:24.30 | many | no prob |
| 19:25.10 | *** join/#asterisk bkruse (i=bkruse@nat/digium/x-a02a52c0611d0f88) |
| 19:26.18 | *** join/#asterisk supers (i=supers@Sia.AnimeNfo.com) |
| 19:26.36 | nclx | I setup asterisk 1.4-trunk on a new box and am trying to get the demo to pickup when dialing my broadvoice.com phone number. When I call I can see this on the console, then I get a message saying party is unavailable and asking to leave a message (I believe this is generated by broadvoice.) Thi i |
| 19:26.39 | *** join/#asterisk legis_ (n=legis@unaffiliated/legis) |
| 19:27.00 | supers | hey, i'm trying to use asterisk with a internet radio stream, but i'm having issues, anyone around to help? |
| 19:27.00 | nclx | this is all I see on the console when I dial in from PSTN via broadvoice: == Using TOS bits 0 |
| 19:27.00 | nclx | <PROTECTED> |
| 19:27.04 | legis_ | Hi, which linux client supports g729? |
| 19:27.56 | [TK]D-Fender | ncix : SIP debug info would help.... |
| 19:30.31 | *** join/#asterisk DrAk0 (n=ljd@nelug/coreteam/luisjose) |
| 19:32.08 | variable_office | jsmith it seems i get errors with "CREATE TRUSTED LANGUAGE plpgsql;" and the "$$" in the function vm_lo_cleanup() |
| 19:32.50 | jsmith | variable_office: Well, it obviously doesn't work in 7.4 then... there may be workarounds, but I really don't have time to work on it today |
| 19:33.52 | variable_office | not a problem, for now i just removed the function and will have to manually clean the table every now and again, maybe cron |
| 19:34.15 | jsmith | Yeah... |
| 19:36.22 | variable_office | i need to get around to upgrading to pg 8 one of these days, any idea on the difficulty, would i have to pg_dumpall and then restore them? |
| 19:37.16 | jsmith | Yeah, it's pretty easy... pg_dumpall and re-import your data |
| 19:38.39 | jsmith | variable_office: http://www.postgresql.org/docs/8.2/static/install-upgrading.html |
| 19:39.58 | hmmhesays | anyone running osx in here? |
| 19:40.08 | hmmhesays | I'm trying to find out where cisco vpn client stores its profile |
| 19:40.12 | *** join/#asterisk huey23 (n=huey23@64.192.209.132) |
| 19:40.17 | Qwell[] | hmmhesays: nope, but if you had sent me that macbook, I would be :P |
| 19:41.08 | huey23 | anyone have any insight to where i can find out how to do simple call forwarding in the dialplan? |
| 19:41.57 | Nugget | "the" dialplan? |
| 19:42.17 | file | Nugget: the one dialplan to rule them all... |
| 19:42.46 | [TK]D-Fender | MY PRECIOUS!!!!!!!!!!!!! |
| 19:43.00 | *** join/#asterisk bbryant (n=12243@c-68-59-20-153.hsd1.sc.comcast.net) |
| 19:43.16 | jsmith | file: But what if I have 15 different dialplans? |
| 19:43.39 | file | jsmith: then you lose |
| 19:44.06 | *** join/#asterisk Ryushin (i=proxy@windwalker.openinnovations.com) |
| 19:44.55 | variable_office | jsmith in the bug report it says the problem was fixed in revision 55158 ; any idea what version that equates to? |
| 19:45.07 | *** join/#asterisk darkfires (n=lwhite@d38-37-41.commercial1.cgocable.net) |
| 19:45.21 | jsmith | variable_office: It probably wasn't fixed in 1.2, if that's what you're asking... |
| 19:45.33 | jsmith | variable_office: since most people using ODBC VM storage are using 1.4 |
| 19:45.39 | darkfires | so the digium guy said to try the svn version of zaptel and asterisk ... hopefully it fixes it |
| 19:46.04 | variable_office | jsmith no it says "Fixed in 1.2 in revision 51158, merged to 1.4 in 51159, merged to trunk in 51160." |
| 19:46.15 | JunK-Y | that could be great on a release to announce, that new tarball is made with which specific svn version, that will avoid that kind of questions, aksed so much time, no? |
| 19:46.15 | jsmith | variable_office: Ah, how about that... that's good to know |
| 19:46.16 | wunderkin | darkfires, um if he is just guessing, no... |
| 19:46.35 | darkfires | i dont think it will fix it either |
| 19:46.37 | [TK]D-Fender | huey23: First you'll have to decide where you are going to stare the "memory" of who is forwarded where. Then you'll have to add the "check" functioning into whatever would NORMALLY dial your devices. Then you'll have to create some scripts to prompt the user as to where they want to be forwarded to (hopefully with some auth and q/c checks) |
| 19:46.54 | variable_office | so jsmith any idea how revision numbers equate to versions? |
| 19:47.08 | huey23 | i am looking to forward an exten to an outside number...can anyone point me in the right direction to find some info? |
| 19:47.10 | jsmith | variable_office: Yes... every change to any file gets a new revision number... |
| 19:47.30 | *** join/#asterisk zotz (n=zotz@24.244.163.157) |
| 19:47.33 | jsmith | variable_office: And they increase... at some point, a release is tagged with a version number |
| 19:48.27 | *** join/#asterisk fiber0pti (i=fiber0pt@216.31.101.41) |
| 19:48.57 | [TK]D-Fender | huey23: "show application read" , "show application gotoif", "show function DB" <- these are your core |
| 19:49.20 | variable_office | ya, looks like 1.2.15 or higher |
| 19:49.54 | *** join/#asterisk tsurko (n=tsurko@213.91.216.130) |
| 19:49.58 | huey23 | [TK]D-Fender: you have helped me before...but that might be over my head |
| 19:50.22 | Nugget | huey23: let me go out on a limb here |
| 19:50.28 | supers | i'm trying to use an external program for MOH, i've converted it to 8000hz w/ pcm codec but it seems "slow" |
| 19:50.32 | supers | anyone have any suggestions? |
| 19:50.38 | variable_office | jsmith the changelog just reads, added documentation on how to fix; so i doint know that anything was actually done |
| 19:50.54 | [TK]D-Fender | huey23: then I guess you're not actually looking for a hint on how to do it, but rather to have someone write the entire thing FOR you. |
| 19:50.56 | Nugget | you're using the term "forward" simply. what you want is just for users to be able to dial a particular extension and have it "go" to an external number and not an internal phone, right? |
| 19:51.01 | huey23 | [TK]D-Fender: i just want to add a line or 2 to the dialplan to automatically forward to an outside number when dialed |
| 19:51.23 | Nugget | Dial() is all you need to do that. |
| 19:51.39 | [TK]D-Fender | huey23: Ah, so not a "dynamic function" but for it to simply dial out on a fixed exten? |
| 19:51.45 | huey23 | [TK]D-Fender: i don't need it written for me...i just want a hint on where to put it |
| 19:51.55 | [TK]D-Fender | huey23: Yuo do it the same as any other dial you do for that same resource. |
| 19:52.08 | [TK]D-Fender | huey23: And "where is "extensions.conf" <- |
| 19:52.46 | huey23 | [TK]D-Fender: ok so instead of Dial(SIP/123,20,0) Dial(1234567890)? |
| 19:53.03 | [TK]D-Fender | huey23: What do you CURRENTLY do to dial "out"? |
| 19:53.15 | [TK]D-Fender | huey23: and the latter is CLEARLY wrong. |
| 19:53.19 | *** join/#asterisk zydrunas (n=Miranda@216.49.228-ip-184.ckt.net) |
| 19:53.31 | huey23 | [TK]D-Fender: that's y i come here :) |
| 19:53.34 | [TK]D-Fender | huey23: Former looks like you'd use for a SIP phone. |
| 19:53.39 | *** join/#asterisk NirS (n=NirS@87.68.144.5) |
| 19:53.41 | Nugget | the dial command will need to resemble your current dial entry which understands how to route out to the public phone network. |
| 19:53.44 | NirS | good evening everybody |
| 19:53.49 | [TK]D-Fender | huey23: Well? What do you do NOW to dial "out"? |
| 19:54.03 | huey23 | [TK]D-Fender: press 9 |
| 19:54.11 | [TK]D-Fender | huey23: CODE DMMIT. |
| 19:54.21 | huey23 | [TK]D-Fender: lol..just a second |
| 19:54.23 | [TK]D-Fender | huey23: What the hell does "9" say to us? |
| 19:54.29 | [TK]D-Fender | :p |
| 19:54.56 | huey23 | [TK]D-Fender: 9="Dial 'out'" :P |
| 19:55.24 | [TK]D-Fender | huey23: That means absolutely NOTHING. Show use the DIALPLAN code that allows you to "dial out" |
| 19:55.59 | darkfires | well svn of asterisk seems to have better sound quality |
| 19:57.17 | huey23 | [TK]D-Fender: ok...am i ok to paste 1 line here? |
| 19:57.27 | [TK]D-Fender | huey23: 1-3 |
| 19:57.52 | [TK]D-Fender | huey23: Anything more will leave you somewhat roasted. |
| 19:58.18 | huey23 | exten => _9NXXXXXX,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}}) |
| 19:58.36 | huey23 | [TK]D-Fender: i believe that's it |
| 19:59.19 | [TK]D-Fender | exten => [whateveryour'redoinghere],[somepriority],Dial(${TRUNK}/1234567890) |
| 19:59.21 | [TK]D-Fender | There |
| 19:59.45 | [TK]D-Fender | just replace the variable # with your FIXED number and put in place where appropriate |
| 20:01.01 | huey23 | ok |
| 20:01.39 | huey23 | [TK]D-Fender: i was willing to try and read up on it but thanks... |
| 20:02.44 | [TK]D-Fender | huey23: Well this is just a single dial. The only thing you needed to understand is to replace the ${EXTEN} part with the specific # you want it to go to. |
| 20:03.10 | [TK]D-Fender | exten => 4,1,Dial(${TRUNK}/1234567890) ; Yay, I dial 1234567890 when someone dials "4"! |
| 20:03.18 | *** join/#asterisk Ebola (n=Ebola@host86-138-6-48.range86-138.btcentralplus.com) |
| 20:03.26 | huey23 | <[TK]D-Fender: thanks again for the help...i'll stick around and let you know how it works out |
| 20:07.45 | nclx | I posted earlier, I have an inbound broadvoice number going to a 1.4-trunk asterisk box. I want for testing this call to just go to the default context and play the demo. But right now when I get a call on my broadvoice line, I can see something about TOS and CoS and then I hear a broadvoice generated the number is unavailable leave voicemail. |
| 20:07.52 | nclx | Someone suggested posting sip debug info |
| 20:08.07 | nclx | so I put it here if anyone can advise of what might be going on: http://rafb.net/p/m6fvsJ87.html |
| 20:08.29 | nclx | my broadvoice number has been sanitized to AAABBBCCCC, my public IP is AAA.BBB.CCC.DDD |
| 20:09.11 | *** part/#asterisk riddlebox (n=JamesMid@75-128-170-26.static.stls.mo.charter.com) |
| 20:09.44 | jsmith | nclx: OK, all that means is that you've registered with them... it doesn't mean that inbound calls are actually able to connect to your Asterisk box |
| 20:09.57 | huey23 | [TK]D-Fender: I was able to get it...thanks again |
| 20:10.01 | [TK]D-Fender | nclx: Indeed |
| 20:11.24 | bkruse | is bugs.digium.com down? or is firefox into trickery |
| 20:11.33 | Qwell[] | bkruse: it's all you |
| 20:11.45 | [TK]D-Fender | nclx: You need a sip peer/user/friend to receive calls against. If you see NOTHING on an incoming call at all I'd first guess your * server is behind NAT. |
| 20:11.53 | bkruse | thanks Qwell[] L[ |
| 20:11.54 | bkruse | ;] |
| 20:12.28 | [TK]D-Fender | bkruse: Its not Firefox... just YOU :) |
| 20:12.42 | Qwell[] | exactly |
| 20:14.03 | bkruse | this... |
| 20:14.06 | bkruse | could be true :/ lol |
| 20:14.29 | bkruse | Qwell[]: its ACTUALLY because im on the wireless, as I just found out. |
| 20:14.31 | nclx | it is behind NAT, I have ports 5060 and 10000-20000 forwarded to my asterisk server which is on 172.31.33.52 |
| 20:14.35 | bkruse | So im trying to go to the internal address :/ |
| 20:15.21 | nclx | I can tcpdump its eth0 interface on 5060 and watch when an incoming call attempts to see if it is getting packets, I'll let ya know in a few |
| 20:15.47 | bkruse | if your getting udp packets, is it still tcpdump? |
| 20:15.48 | [TK]D-Fender | nclx: ok, well you still need other entries in sip.conf for incoming calls to be processed. You saying you get NOT sip debug at all on incoming calls? |
| 20:15.59 | *** part/#asterisk huey23 (n=huey23@64.192.209.132) |
| 20:15.59 | jsmith | bkruse: Yeah, unfortunately... |
| 20:16.07 | bkruse | jsmith: hmm |
| 20:16.08 | [TK]D-Fender | nclx: no need for tcpdump, just use *'s "sip debug" |
| 20:16.14 | bkruse | [TK]D-Fender: correct |
| 20:18.16 | *** join/#asterisk bmg505 (n=leon@196.209.181.226) |
| 20:21.41 | nclx | So is this not an incoming call? (This is from sip debug right after I dialed my broadvoice number from PSTN) <--- SIP read from 147.135.4.128:5060 ---> |
| 20:21.41 | nclx | ACK sip:s@172.31.33.52:5060 SIP/2.0 |
| 20:21.42 | nclx | Call-ID: 5f00cf-5f@147.135.4.128 |
| 20:21.42 | nclx | CSeq: 1 ACK |
| 20:21.42 | nclx | From: "Mr. Caller"<sip:DDDEEEFFFF@147.135.4.128;user=phone>;tag=8bce |
| 20:21.43 | nclx | outtolunc: "Mr. Callee"<sip:s@172.31.33.52>;tag=as5151fc20 |
| 20:21.45 | nclx | Via: SIP/2.0/UDP 147.135.4.128:5060;received=AAA.BBB.CCC.DDD |
| 20:21.47 | nclx | Content-Length: 0 |
| 20:21.47 | bm-chs | On my asterisk box, I can't seem to ping anything outside my network -- no dns resolution? Any suggetsions? |
| 20:22.05 | nclx | sounds like no gateway |
| 20:22.36 | bm-chs | My gateway IP is 192.168.1.2 . . . I can ping that . . . I can ping phones . . . |
| 20:22.42 | bm-chs | Anything internally. |
| 20:23.44 | outtolunc | eh? what did i miss now |
| 20:24.26 | nclx | cat /etc/resolv.conf; is that correct for your nameserver? |
| 20:24.54 | *** join/#asterisk Shaun222 (n=shaun@ip68-4-127-67.oc.oc.cox.net) |
| 20:24.56 | jsmith | bm-chs: Type "route -n" at the Linux CLI |
| 20:25.03 | jsmith | bm-chs: Do you have a default route? |
| 20:25.08 | holiday_42 | bm-chs, when you ping does it resolve name to IP or no? |
| 20:25.18 | nclx | sudo route add default gw 192.168.1.2 |
| 20:25.46 | Shaun222 | is there really any good advantage to having a frac PRI T1 over say 8 POT lines? |
| 20:26.44 | bm-chs | Kernel IP routing table |
| 20:26.44 | bm-chs | Destination Gateway Genmask Flags Metric Ref Use Iface |
| 20:26.44 | bm-chs | 192.168.1.0 0.0.0.0 255.255.255.0 U 0 0 0 eth0 |
| 20:26.44 | bm-chs | 169.254.0.0 0.0.0.0 255.255.0.0 U 0 0 0 eth0 |
| 20:26.51 | Sweeper | Shaun222: yes |
| 20:27.01 | Sweeper | it probably costs less, and it's switched |
| 20:27.04 | holiday_42 | bm-chs, where's the default gw? |
| 20:27.12 | bm-chs | 192.168.1.2 |
| 20:27.19 | carrar | default gw's are over rated!! |
| 20:27.22 | holiday_42 | :) |
| 20:27.41 | holiday_42 | bm-chs, nclx nailed it. |
| 20:27.42 | Shaun222 | Sweeper: i can get pots cheaper... whats switched do for me? |
| 20:27.54 | holiday_42 | bm-chs, add yer default route |
| 20:28.19 | lirakis | later all |
| 20:28.19 | Sweeper | Shaun222: if you've got 8 lines, and someone calls your main number, nobody else will be able to call your main number |
| 20:28.40 | Sweeper | on a t1, it just comes in a different channel |
| 20:28.51 | Sweeper | well, on a pri |
| 20:28.55 | *** part/#asterisk lirakis (n=etamme@65.200.191.253) |
| 20:28.57 | Shaun222 | i was told by the phone company that they would just do a roll over dilio... |
| 20:29.19 | bm-chs | Ok -- that seems to help . . . how do I make that permanent? |
| 20:30.27 | Sweeper | Shaun222: ah. well, t1 hardware is a bit more reliable than FXO stuff, since it stays digital all the way. |
| 20:30.37 | *** join/#asterisk fakhir (n=Fakhir@unaffiliated/fakhir) |
| 20:30.44 | Sweeper | other than that, not a whole lot |
| 20:30.56 | Sweeper | a single t1 card is probably cheaper than an 8fxo |
| 20:34.35 | tzanger | more reliable too |
| 20:35.06 | Sweeper | yea. it's about $400 for a t1 card, $900 for an 8fx0 |
| 20:35.08 | variable_office | jsmith it looks like it wont be doable for me to run odbc storage with pg 7.4; any ideas on a next best thing with my goals being for a webserver to be able to access the voicemail stuff(voicemails are currently stored with root) |
| 20:35.33 | jsmith | variable_office: Upgrade to 1.4 and PostgreSQL 8.x :-) |
| 20:35.53 | jsmith | variable_office: Trust me... any other way becomes a logistical nightmare |
| 20:36.12 | variable_office | taking postgres down for that kindof timeframe would be terrible |
| 20:36.33 | variable_office | jsmith there is no way then to just make asterisk save the voicemails with looser permissions? |
| 20:36.35 | Sweeper | variable_office: there are web interfaces that work just fine with file-based voicemail storage |
| 20:36.45 | jsmith | variable_office: Not an easy way, no. |
| 20:36.57 | *** join/#asterisk kwame (n=kwame@209.213.194.7) |
| 20:37.02 | variable_office | Sweeper any idea on how they read the files, do they just run as root? |
| 20:37.07 | jsmith | Sweeper: And none of them that I've seen work that well... most have subtle race conditions that cause problems over time |
| 20:37.39 | Sweeper | variable_office: probably the latter. just use mysql :D |
| 20:37.43 | kwame | hi, in my /var/log/asterisk/messages I get this message [Aug 27 15:28:58] WARNING[4707] file.c: No such format 'h261' |
| 20:38.02 | kwame | when doing a call from one ekiga client to another, any idea what this error means? |
| 20:38.52 | variable_office | how hard is it to get odbc running on mysql? |
| 20:39.46 | jsmith | variable_office: Not that hard |
| 20:40.02 | *** join/#asterisk weasel00 (n=snowball@pencomsf.com) |
| 20:40.14 | weasel00 | where do i install phpagi? |
| 20:45.04 | KDan | anywhere you want, you don't need to put it in a special location |
| 20:45.10 | KDan | is there any way to use Asterisk to send SMS's for quasi-free? Or so you *have* to use an SMS gateway service that charges per SMS? |
| 20:47.45 | viKing78 | I'm having a problem with a PolyCom 330 in a reboot loop. It keeps giving me a "Duplicate IP" error. |
| 20:47.54 | viKing78 | I've checked and that IP is not in use |
| 20:48.40 | viKing78 | They both are showing a MAC of a windows server I have as the offending machine. |
| 20:49.16 | viKing78 | And that server does not reply to pings on the IP of the phone if I unplug it. |
| 20:49.37 | viKing78 | Any idea how the Polycom is checking for a duplicate? |
| 20:52.19 | variable_office | Sweeper all i need is for this to work for a day, what would i have to do to make the php script run as root when called from the webserver(i know it is not safe) |
| 20:52.29 | variable_office | its local area only though, so not too bad |
| 20:56.18 | *** join/#asterisk NinjaJon (n=jonathan@anya.northenden.ninja.org.uk) |
| 20:57.34 | NinjaJon | Hi! I'm having a problem with chan_zap - is this the best place to get some pointers as to where I can look to fix it? |
| 20:58.14 | JerJer | NinjaJon: if you actually ask a question, it might get answered |
| 20:58.51 | NinjaJon | Thanks JerJer - this is my first time on IRC, apologies. Question coming up.. |
| 20:59.00 | JerJer | no worries |
| 20:59.04 | NirS | Hey JerJer |
| 20:59.07 | NirS | wassup man ? |
| 20:59.09 | JerJer | meep meep |
| 20:59.11 | NirS | been a while |
| 20:59.15 | JerJer | lame ass carriers that suck |
| 20:59.24 | NirS | no shit |
| 20:59.33 | NirS | I've got a few lame ass people banging at the door |
| 20:59.54 | NinjaJon | I had a working Asterisk installation up until this morning. Asterisk 1.4.x, Zaptel (TDM400), all hunky dory. This morning, I edited features.conf and restarted Asterisk - all of a sudden I now get errors from chan_zap : |
| 20:59.54 | NinjaJon | ERROR[4086]: chan_zap.c:10472 build_channels: Unable to reconfigure channel '1' |
| 21:00.24 | NirS | hmmm... that is weird |
| 21:00.29 | NinjaJon | I've been tearing my hair out for a few hours now so figured it was time to ask for some pointers. I'm now running the latest zaptel drivers, the latest Asterisk (1.4.9) and the latest kernel |
| 21:00.31 | NirS | is channel 1 on your card an FXS port ? |
| 21:00.47 | NirS | Ninja, asterisk latest is 1.4.11 |
| 21:01.08 | NinjaJon | yup, channels 1-4 have phones plugged in to them; channel 5 is a x100p |
| 21:01.23 | JerJer | ahh the x100p |
| 21:01.31 | NinjaJon | ah, my mistake.. I downloaded asterisk this afternoon & it must be sorted numerically rather than alphabetically. |
| 21:01.34 | *** join/#asterisk denon (n=denon@tooth.decay.org) |
| 21:01.34 | *** mode/#asterisk [+o denon] by ChanServ |
| 21:01.44 | JerJer | i just found like 15-20 legit Digium sourced X100Ps in my other garage |
| 21:01.54 | hmmhesays | heh |
| 21:01.58 | *** part/#asterisk denon (n=denon@tooth.decay.org) |
| 21:02.14 | NinjaJon | But, my zapata.conf is now 3 lines long, for debugging this... it says '[channels]' 'context=sip' 'channel=>1' - which should work... |
| 21:02.23 | kwame | any ideas? |
| 21:02.29 | NirS | Ninja, can you please paste your features.conf file to pastebin.com ? |
| 21:02.41 | NinjaJon | I had an idea it might have been permissions on /dev/zap (don't ask me why) but chmod 666 hasn't done the trick |
| 21:02.43 | kwame | hi, in my /var/log/asterisk/messages I get this message [Aug 27 15:28:58] WARNING[4707] file.c: No such format 'h261' |
| 21:02.43 | JerJer | pastebin.ca is cooler |
| 21:03.09 | JerJer | kwame: do you intend on providing H.261 video? |
| 21:03.18 | NirS | kwame, h261 is a video codec if I'm not mistaken |
| 21:04.02 | NinjaJon | features.conf is now at http://pastebin.ca/672759, I have indicated the two lines I added this morning. (Backing the change out didn't help, sadly) |
| 21:04.53 | kwame | JerJer: mmhhhh, yes |
| 21:05.23 | NirS | Ninja, I don't believe that this has anything to do with the FXS ports |
| 21:05.40 | NirS | can you please paste your zapata.conf file? are you using TrixBox or something similar ? |
| 21:06.25 | NinjaJon | NirS - indeed, which is why I am so confused. I even tried power-cycling the box, as I couldn't figure out what has changed.. but no luck. I've now gone the other way and upgraded everything to current (or slightly older, as it happens), this hasn't helped.. zapata.conf coming right up |
| 21:07.07 | JerJer | NinjaJon: make sure to reload the linux kernel modules if you do any major upgrading or downgrading - especially with zaptel directly |
| 21:08.27 | NirS | I agree with JerJer, a modification of the zaptel kernel module always require a restart of zaptel |
| 21:08.27 | NinjaJon | ok, my (drastically reduced) zapata.conf is now at the end of that pastebin. JerJer, I did reload, thanks - although I have now spotted that my latest 'make install' of zaptel has wiped my /etc/zaptel.conf. I'm going to restore that and check again (I'll add it to pastebin as well) |
| 21:09.09 | JerJer | eh - it shouldn't have |
| 21:09.33 | NirS | URL please |
| 21:09.39 | NinjaJon | http://pastebin.ca/672765 |
| 21:09.56 | NinjaJon | ah, I see how pastebin works now! I thought the url would be the same, sorry |
| 21:10.31 | JerJer | pastebins are nice search engine whores |
| 21:10.40 | NinjaJon | JerJer - I agree.. but somehow it did. The timestamps on /etc/zaptel.conf and /lib/modules/2.6.22.2-42.fc6/misc/zaptel.ko are the same.. |
| 21:10.52 | JerJer | that's a bug then |
| 21:11.18 | *** join/#asterisk denon (n=denon@tooth.decay.org) |
| 21:11.18 | *** mode/#asterisk [+o denon] by ChanServ |
| 21:11.30 | Sweeper | variable_office: have you tried voicemail.cgi? |
| 21:11.32 | Sweeper | err |
| 21:11.35 | Sweeper | vmail.cgi |
| 21:11.43 | *** join/#asterisk Galeras (n=Galeras@201.245.18.122) |
| 21:12.49 | NirS | well, sounds like the make install portion of Makefile has a bug in it |
| 21:13.12 | NinjaJon | OK, /etc/zaptel.conf and some associated output is here: http://pastebin.ca/672769 |
| 21:13.19 | *** join/#asterisk drako (n=ljd@nelug/coreteam/luisjose) |
| 21:14.22 | NinjaJon | I mainly don't understand how/why it suddenly stopped working - that's what's bugging me. It has worked fine for about 2 years, until just now |
| 21:14.51 | *** part/#asterisk Sweeper (i=sweeper@softcheese.net) |
| 21:15.15 | Galeras | Dear Sirs, is gtalk/asterisk ready for production environments? |
| 21:15.28 | JerJer | define production envrionments |
| 21:15.46 | wunderkin | spip_ssip_2_2_0_release_sig.zip (40121056 bytes). BootRom_4_0_0_release_sig.zip (12734818 bytes). |
| 21:15.49 | wunderkin | holy sh1t |
| 21:16.33 | jsmith | New boot rom, eh? |
| 21:16.54 | wunderkin | yeah it isn't listed on the website... top secret i guess |
| 21:16.56 | bisybackson | does anyone know how to disable call waiting on snom 300's web interface? |
| 21:19.23 | Galeras | JerJer: from voip-info.org: "you should consider this feature to be in beta phase, there are still quite a number of problems and glitches around." |
| 21:19.47 | NirS | sorry, have to buzz |
| 21:19.49 | NirS | c'ya later |
| 21:19.57 | NinjaJon | thanks for your help NirS |
| 21:20.08 | NirS | you're welcome |
| 21:22.13 | *** join/#asterisk Super_Cat_Frog (n=bob@82-40-170-180.cable.ubr02.blac.blueyonder.co.uk) |
| 21:23.13 | Super_Cat_Frog | hi - im trying to create a new rule, where 6 digit numbers (local numbers) can be dialled without dialing 9 for an external line, and without the area code (01253). i dont think its matching, my phone is giving me error 484 - exten => _XXXXXX.,1,Dial(SIP/01253${EXTEN:1}@sipgate,60,tr) |
| 21:24.32 | NinjaJon | Super_Cat_Frog - have you tried it without the trailing '.' , and using SIP/01253${EXTEN} in your dial string? |
| 21:24.42 | Super_Cat_Frog | i'll try |
| 21:25.46 | Super_Cat_Frog | NinjaJon: didn't work - 484 again, To: <sip:313819@192.168.0.250>;tag=as6a8b7873 |
| 21:26.33 | NinjaJon | Super_Cat_Frog, I assume you're reloading the dialplan ("dialplan reload") after each change, and you are dialling from the correct context? Basic questions, I know.. |
| 21:27.18 | *** join/#asterisk QbY_ (n=Kelvin@66.236.241.67.ptr.us.xo.net) |
| 21:27.34 | QbY_ | is there a way to call an external script after a voicemail is left? |
| 21:27.41 | Super_Cat_Frog | NinjaJon: yes i'm reloading and using the correct context |
| 21:28.12 | NinjaJon | Super_Cat_Frog: Can you post your extensions.conf to pastebin.ca ? |
| 21:28.23 | Super_Cat_Frog | NinjaJon: shall do |
| 21:28.27 | *** join/#asterisk Mavvie (n=edwin@ppp121-44-38-133.lns3.syd7.internode.on.net) |
| 21:29.02 | NinjaJon | JerJer: If you're still happy to look at this weird chan_zap problem, I've set up a login on my box? |
| 21:29.34 | Super_Cat_Frog | NinjaJon: http://rafb.net/p/Tvq2G577.html |
| 21:30.41 | NinjaJon | Super_Cat_Frog - which context is your phone in? I assume dialling 9 + the number does work. |
| 21:31.13 | Super_Cat_Frog | NinjaJon: its in the default context. dialing 9....... works |
| 21:32.06 | variable_office | is there a way to change the permissions that asterisk makes the voicemails with so that other users can use them>? |
| 21:32.28 | Super_Cat_Frog | variable_office: you could umask the mount ? |
| 21:32.42 | *** join/#asterisk [hC] (n=hardcore@66.119.167.163) |
| 21:32.56 | NinjaJon | Super_Cat_Frog - I wonder if changing your _XXXXXX to _[0-8]XXXX would help.. i.e. tell it to match anything not starting with a 9 ? |
| 21:33.10 | NinjaJon | Super_Cat_Frog - everything else looks OK |
| 21:33.29 | variable_office | Super_Cat_Frog how would i do that? |
| 21:34.08 | *** join/#asterisk zotz (n=zotz@24.244.163.157) |
| 21:34.32 | *** join/#asterisk Trionnis (n=blah@65-117-172-195.dia.static.qwest.net) |
| 21:34.45 | Super_Cat_Frog | variable_office: google 'mount umask' - im not completely sure, just a suggestion that may possibly help |
| 21:34.50 | Super_Cat_Frog | NinjaJon: that didn't work either |
| 21:35.49 | Trionnis | Anyone have some info about the error "ACL error (permit/deny)" in a sip registration message? I'm not finding much on voip-info or google |
| 21:35.51 | _ShrikE | pdf |
| 21:37.22 | *** join/#asterisk NinjaJon (n=jonathan@anya.northenden.ninja.org.uk) |
| 21:37.55 | NinjaJon | sorry, client just crashed so may have missed some messages.. |
| 21:38.12 | Super_Cat_Frog | NinjaJon: that didn't work either |
| 21:39.17 | NinjaJon | Super_Cat_Frog - I have a very similar dialplan for my outgoing: http://rafb.net/p/UHtwpe81.html |
| 21:39.48 | *** part/#asterisk QbY_ (n=Kelvin@66.236.241.67.ptr.us.xo.net) |
| 21:40.24 | NinjaJon | Anybody here know about chan_zap internals? My Asterisk box decided to not reload chan_zap.so this morning; I haven't been able to figure out why.. |
| 21:41.06 | jsmith | NinjaJon: Did you update your kernel? Are the zaptel kernel modules loading? Did you change the signalling in zaptel.conf or zapata.conf? |
| 21:41.52 | NinjaJon | jsmith: the only change I am aware of making was an unrelated features.conf edit. When I reloaded Asterisk, chan_zap wouldn't load again. I backed out the features.conf change but no luck |
| 21:42.04 | Super_Cat_Frog | NinjaJon: its 503ing, which is probably a good thing - thanks |
| 21:42.06 | NinjaJon | jsmith: Since then I've upgraded Asterisk, upgraded kernel & zaptel etc.. but still the same |
| 21:42.22 | NinjaJon | Super_Cat_Frog - glad you're a bit further on, at least..! |
| 21:42.33 | jsmith | NinjaJon: What happens if you set core verbose and core debug to 9, then type "module unload chan_zap.so" then "module load chan_zap.so" |
| 21:42.45 | jsmith | NinjaJon: That should give you some useful information on why it's not loading |
| 21:44.17 | NinjaJon | jsmith, you are a genius.. I think that's given me the clue I needed. Will double-check now & then summarise for the irc archives.. |
| 21:44.31 | jsmith | NinjaJon: Cool... |
| 21:49.54 | *** join/#asterisk sergey (n=sergey@gw4-130.iks.ru) |
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| 21:51.46 | killfill | hi... |
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| 21:52.57 | killfill | ive setup in iax.conf my peer. its in a [xxx] section. |
| 21:53.29 | killfill | if i set "IAX2/xxx:pass@xxx/${EXTEN:1}" in my dialplan, it works ok. |
| 21:53.47 | killfill | any way i can only specify IAX2/xxx/${EXTEN:1} ?.. |
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| 21:57.05 | Trionnis | can anyone assist with a sip registration issue? you can see the debug here: http://rafb.net/p/fB2ZkW84.html Thanks |
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| 22:07.46 | NinjaJon | OK, a summary of my chan_zap problem and the solution. Hopefully this will end up archived & some other poor soul can search for it to easily fix their problem. |
| 22:07.54 | NinjaJon | Problem: When reloading chan_zap.so, I got ERROR[4404]: chan_zap.c:10472 build_channels: Unable to reconfigure channel |
| 22:08.00 | NinjaJon | Solution: I had removed the 'signalling' lines from zaptel.conf a few weeks ago, due to some misleading warning messages. Everything worked absolutely fine after this, as I didn't restart Asterisk (only reloaded chan_zap). When I restarted Asterisk today, the missing 'signalling' lines stopped things from working; this was confirmed by jsmith's suggestion of 'core set verbose 9, core set debug 9', 'module unload chan_zap.so', 'module loa |
| 22:08.20 | NinjaJon | Unloading and reloading is *DIFFERENT* to 'module reload chan_zap.so'. chan_zap.so processes the "signalling=" config statements from zaptel.conf ONCE only, on module load. It issues a warning on module reload, saying it is ignoring the "signalling=" statements - this is what made me think they were redundant and why I removed them completely... and this is what broke zaptel for me. |
| 22:08.20 | NinjaJon | Moral of the story - Just because chan_zap warns you on reload that it is ignoring a line in your configuration, doesn't mean that line is redundant and can be removed... |
| 22:08.59 | SplasPood | heh, I've always wondered about that ignoring msg |
| 22:09.11 | Qwell[] | it just means that it can't change the signalling once it's loaded |
| 22:09.26 | NinjaJon | I'm going to try and add a note to the voip-info.org wiki as well.. |
| 22:09.54 | NinjaJon | I'm not sure what else the message should say, though. Perhaps "ignoring parameter on module reload" would be clearer ? |
| 22:10.50 | sevard | Does anyone know of a sip client for windows that doesn't need installing? |
| 22:11.22 | *** join/#asterisk orvux (n=orvux@200.77.223.187.cable.dyn.cableonline.com.mx) |
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| 22:16.53 | orvux | hi everyboy, why asterisk has two branches??? 1.2* and 1.4*??? |
| 22:17.19 | JunK-Y | 1.2 is deprecated. |
| 22:17.32 | sevard | 1.2 isn't depreciated |
| 22:17.46 | Strom_M | sevard: deprecated, not depreciated |
| 22:17.50 | Strom_M | they're different words |
| 22:18.00 | sevard | you're deprecated. |
| 22:18.07 | Strom_M | "depreciated" is an accounting term, silly |
| 22:18.58 | file | play nice kids |
| 22:19.56 | coil | this laptop's screen is too small |
| 22:20.15 | Qwell[] | coil: how small? |
| 22:20.20 | coil | 13" |
| 22:20.30 | Qwell[] | what res? 13" should be good |
| 22:20.55 | coil | 1280x800 |
| 22:21.07 | Qwell[] | yeah, that's plenty for 13"... |
| 22:21.08 | mvanbaak | that's more then enough |
| 22:21.33 | mvanbaak | I have a 12' schmackbook |
| 22:21.40 | *** part/#asterisk NinjaJon (n=jonathan@anya.northenden.ninja.org.uk) |
| 22:21.40 | *** join/#asterisk Deeewayne (i=dwayne@nat/digium/x-a55a580cc53ef2ef) |
| 22:21.40 | *** mode/#asterisk [+o Deeewayne] by ChanServ |
| 22:21.51 | coil | well i have too many windows open :P |
| 22:22.01 | mvanbaak | it's nice, I can switch desktops by hitting my macbook on the side |
| 22:22.21 | mvanbaak | thanks to the 'motion sensor to protect HD' device in there |
| 22:22.26 | coil | lol |
| 22:22.36 | coil | what prog can you use to do that |
| 22:23.16 | mvanbaak | http://blog.medallia.com/2006/05/smacbook_pro.html |
| 22:23.23 | Trevor_b | hack into the accelerometer is a pretty cool thing, but not like it saves time when you remove you hand from keyboard to hit a monitor when you do the same with a mouse click ;) |
| 22:25.50 | mvanbaak | I have a thinkpad T61 now so I'm waiting for the linux module to support it so I can do it on both my macbook and thinkpad |
| 22:25.59 | mvanbaak | the T41 is already supported |
| 22:26.35 | orvux | well, then if i want to deploy an Asterisk PBX i should use the 1.4* version??? |
| 22:26.59 | mvanbaak | orvux: yeah, 1.2 is in 'security-fixes-only' mode |
| 22:27.01 | Nugget | I installed that on my macbook pro but I got rid of it because I found it got triggered too much through just normal jostling |
| 22:27.04 | mvanbaak | so 1.4 is your best bet |
| 22:27.07 | Nugget | maybe they've improved it |
| 22:27.15 | *** join/#asterisk thgood (n=thgood@web2b.profitboost.com) |
| 22:27.20 | mvanbaak | Nugget: it works pretty good here now |
| 22:27.27 | thgood | hi |
| 22:27.35 | orvux | ok, thanks.... |
| 22:27.37 | Nugget | cool. I haven't fooled with it since it was initially released |
| 22:28.03 | mvanbaak | you can tweak how sensitive it should be |
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| 22:35.03 | *** join/#asterisk Lucky7 (n=Adam@207.200.28.175) |
| 22:39.16 | Nugget | Asterisk needs more cowbell. |
| 22:39.39 | coil | moar cowbell!!! |
| 22:40.29 | killfill | hey guys... |
| 22:40.38 | killfill | why does this not work? Dial(IAX2/coneccion-pbxs/${EXTEN:1}) |
| 22:40.39 | Lucky7 | ugh |
| 22:40.42 | Lucky7 | i hate t1's |
| 22:40.43 | killfill | http://pastebin.ca/672829 |
| 22:41.06 | killfill | if i speficy the user:pass@.... then it works.. but i dont want to stick the pass in there... |
| 22:41.37 | killfill | <PROTECTED> |
| 22:41.39 | Nugget | what does it do while it's busy not working/ |
| 22:41.40 | Nugget | ? |
| 22:41.48 | Nugget | ah, I see. |
| 22:42.02 | killfill | and am i doing wrong?.. |
| 22:42.03 | Nugget | sounds like it's not correctly figuring out the peer to use from iax.conf |
| 22:42.46 | mvanbaak | looks like the username is not send to the peer |
| 22:42.59 | Nugget | or you don't have the username/secret set in iax.conf |
| 22:43.14 | mvanbaak | try: Dial(IAX2/username@peer_defined_in_iax.conf/number) |
| 22:44.04 | mvanbaak | to stick with your sample: Dial(IAX2/username@coneccion-pbxs/${EXTEN:1}) |
| 22:44.21 | mvanbaak | I had the exact same thing with Voop today ;) |
| 22:44.28 | mvanbaak | that's why I know the answer |
| 22:44.50 | killfill | actually.. in iax.conf of the remote pbx.. i have no username=.... |
| 22:45.01 | killfill | should i ?..:P |
| 22:45.09 | mvanbaak | look on the remote PBX what username it tries to use |
| 22:45.13 | Nugget | that would help. :) |
| 22:45.44 | mvanbaak | it was fun. we had this setup working for 4 months, and all of a sudden it stopped working |
| 22:45.54 | mvanbaak | Voop was telling us nothing changed on their side |
| 22:46.02 | mvanbaak | gheh |
| 22:46.07 | mvanbaak | try to debug that |
| 22:46.30 | mvanbaak | svn log did not show any changes on our side for weeks |
| 22:47.24 | boch | can i launch multiple calls using AMI originate ? |
| 22:47.58 | killfill | doesnt work.. :S |
| 22:48.22 | mvanbaak | boch: while true; do ./originate_call.sh; done |
| 22:48.43 | mvanbaak | you have to chain them |
| 22:49.08 | killfill | do i need trunk=yes?.. |
| 22:49.29 | mvanbaak | killfill: nope |
| 22:49.38 | mvanbaak | killfill: did you read: http://www.voip-info.org/wiki/view/Asterisk+Connect+2+servers |
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| 22:50.18 | mvanbaak | anywayz, I'm off to bed |
| 22:50.21 | mvanbaak | latero all |
| 22:50.31 | killfill | yup.. |
| 22:50.32 | killfill | later |
| 22:50.35 | killfill | damn.. have a meeting.. |
| 22:50.39 | killfill | later too. |
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| 22:53.59 | Lucky7 | does asterisk need call confirmation on a T1? |
| 22:54.11 | *** part/#asterisk kwame (n=kwame@209.213.194.7) |
| 22:54.13 | JT | say what? |
| 22:54.53 | Lucky7 | call confirmation, its a T1 switch echo back when a call is secessfully going out |
| 22:55.18 | JT | yeah you're still not making much sense |
| 22:59.08 | Yourname` | Hi, I'm currently using exten => 4190000000,1,Set(CALLERID(name="Inbound"). How can I set the incoming call's phone number as well when I rcv the call on 4190000000? |
| 23:00.31 | flenders | exten => s,5,SetCallerID("${CALLERID(name)}" <${CALLERID(num)}>) |
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| 23:00.44 | *** join/#asterisk CoolGuy21 (n=Tilt@cpe-76-175-234-137.socal.res.rr.com) |
| 23:00.45 | CoolGuy21 | hi |
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| 23:01.03 | JT | err |
| 23:01.08 | JT | that's deprecated |
| 23:01.09 | flenders | oops sorry |
| 23:01.26 | JT | you should use Set(CALLERID(name)=) or (num) etc |
| 23:01.36 | flenders | Set(CALLERID(num)=) |
| 23:01.40 | JT | have no idea if Set(CALLERID(name="Inbound") will work at all |
| 23:02.46 | CoolGuy21 | can someone help me with this http://pastebin.ca/672883 ? it shows registered and i can dial out. when i call from another phone to it i get a unreachable at this time and i pasted the sip debug. |
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| 23:13.29 | flenders | CoolGuy21: can you paste your register line with no passwords? and also your sip account from sip.conf, and also relevant parts of your extensions.conf |
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| 23:16.07 | CoolGuy21 | k fixed it |
| 23:18.02 | heelios | how would I go about amplifying a SIP call? |
| 23:18.28 | Yourname` | JT: Why not? |
| 23:18.40 | JT | Yourname`: why not what? |
| 23:18.55 | Yourname` | JT>have no idea if Set(CALLERID(name="Inbound") will work at all |
| 23:19.12 | JT | because as far as i know it is completely invalid use of the syntax |
| 23:19.15 | Yourname` | It's working. Basically, I want to set the calleridname of the inbound caller, but want the callerid number to be that of the inbound caller. |
| 23:19.47 | Yourname` | JT: Ohhh, I made a mistake on typing it out. |
| 23:19.53 | JT | Set(CALLERID(name)=Inbound) |
| 23:19.59 | JT | pleast don't type stuff out |
| 23:20.00 | Yourname` | Set(CALLERID(name)="Inbound" |
| 23:20.03 | JT | copy and paste |
| 23:20.05 | Yourname` | Yup, that's what it is.. |
| 23:20.08 | JT | quotes are unnecessary |
| 23:20.14 | Yourname` | It was just one line, so I did. Sorry. |
| 23:20.16 | Yourname` | Ok. |
| 23:20.23 | Yourname` | So what about the phone number thing then? |
| 23:20.40 | Yourname` | Because this lets me set the name as I want, but I want the phone number of the caller as is. |
| 23:21.17 | flenders | Set(CALLERID(num)=${CALLERID(num)}) |
| 23:21.48 | Yourname` | flenders: Gotcha, thanks. Will be trying it in a bit.. gotta run to pick up the wife. |
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| 23:35.34 | joe_bag_odonuts | Hi I just put up a box running Fedora Core 7. I compiled asterisk and installed it. Vers 1.4.11. I have another asterisk box running Asterisk@home (asterisk 1.2). I have an xlite device that registers just fine with the 1.2 box (the xlite device is being natted). However, I can't seem to get it to register with the 1.4 box. The sip.conf section for the device is identical on both boxes. I can register with my isp on th |
| 23:36.00 | *** join/#asterisk Galeras (n=Galeras@190.84.206.174) |
| 23:36.23 | joe_bag_odonuts | I don't even see the registration attempt when using the CLI. |
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| 23:37.57 | CoolGuy21 | how can i find out what did number it sees when i dial the did? i put the did number in the routing but it doesnt match so how do i see what it sees |
| 23:41.36 | flenders | CoolGuy21: put DID in the routing? |
| 23:42.04 | CoolGuy21 | yeah but it seems like its not matching |
| 23:42.15 | flenders | what is routing? |
| 23:43.46 | flenders | CoolGuy21: can you paste your register line with no passwords? and also your sip account from sip.conf, and also relevant parts of your extensions.conf |
| 23:44.45 | flenders | do you have calls coming in at all? |
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| 23:47.44 | flenders | jbot: he is great |
| 23:47.45 | jbot | ...but he is already something else... |
| 23:47.46 | flenders | :D |
| 23:48.04 | flenders | jbot: yeah, you should bome by and meet him someday |
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| 23:48.21 | JT | heh |
| 23:48.23 | flenders | hahaha |
| 23:48.44 | flenders | I wonder if people think jbot is real sometimes |
| 23:49.00 | JT | some people do |
| 23:49.10 | JT | must be some sort of mental illness |
| 23:49.14 | flenders | CoolGuy21: you back? |
| 23:50.07 | flenders | I've seen people thanking jbot when you do a ~book, for example |
| 23:50.41 | CoolGuy21 | yes |
| 23:50.52 | CoolGuy21 | g thnx flenders |
| 23:51.05 | CoolGuy21 | it was a small thing just wanted to see if anyone knew off the top of there head |
| 23:51.15 | flenders | is it working now? |
| 23:51.32 | CoolGuy21 | no |
| 23:51.44 | flenders | alright, wanna paste all that stuff so I can have a look? |
| 23:51.48 | CoolGuy21 | not sure what the tsp is sending as the did |
| 23:51.48 | flenders | pastebin I meant |
| 23:51.57 | weasel00 | i installed phpagi and now my server is dropping ssh and the gui when 2 voip clients connect to it |
| 23:51.58 | flenders | is it a SIP account? |
| 23:52.39 | flenders | CoolGuy21: ? |
| 23:53.06 | CoolGuy21 | yes it is |
| 23:54.17 | flenders | if you a '/1234567' at the end of your register line, and then a exten => 1234567,1,command() on the SAME context on extensions.conf, it should be fine |
| 23:54.26 | flenders | is that what your problem is? |
| 23:54.44 | flenders | just guessing, as you said you're trying to match DID |
| 23:55.35 | CoolGuy21 | no |
| 23:55.40 | CoolGuy21 | im trying to match the DID |
| 23:55.47 | CoolGuy21 | like i have 3 sip accounts |
| 23:56.03 | CoolGuy21 | and if 555 555 5555 dials i want to send it to exten 222 |
| 23:56.05 | CoolGuy21 | and so forth |
| 23:56.10 | CoolGuy21 | but its not matching up |
| 23:56.13 | flenders | well, that's not a DID then |
| 23:56.47 | flenders | Direct Inward Dialing |
| 23:56.53 | flenders | DID is your number |
| 23:57.15 | CoolGuy21 | yes |
| 23:57.18 | flenders | how are you trying to match the CID? |
| 23:57.27 | CoolGuy21 | if they are dialing through sip 1 sip 2 |
| 23:57.34 | CoolGuy21 | i have 3 different trunks |
| 23:57.45 | CoolGuy21 | and i want to filter from which trunk where the calls go |
| 23:58.10 | *** join/#asterisk riddlebox (n=james@75-132-225-75.dhcp.stls.mo.charter.com) |
| 23:58.12 | JT | sip is not a trunk |
| 23:58.16 | flenders | so, each SIP account with your ITSPs have a different DID, right? |
| 23:58.22 | CoolGuy21 | yes |
| 23:58.30 | flenders | and I assume you have 3 different register lines |
| 23:58.34 | CoolGuy21 | yes |
| 23:58.43 | flenders | so, as I said before: |
| 23:58.49 | flenders | if you a '/1234567' at the end of your register line, and then a exten => 1234567,1,command() on the SAME context on extensions.conf, it should be fine |
| 23:59.07 | CoolGuy21 | mine doesnt end with / |
| 23:59.16 | flenders | well, make it end then |
| 23:59.30 | CoolGuy21 | im using inbound routes |
| 23:59.43 | *** join/#asterisk DrAk0 (n=ljd@nelug/coreteam/luisjose) |
| 23:59.51 | flenders | register => asd:asd@host.domain.com/1 |
| 23:59.58 | flenders | then on extensions.conf you do: |
| 23:59.59 | JT | inbound routes, why do you make up strange terms for everything? |