| 00:11.13 | *** part/#asterisk lowlevel (n=Stuart@lowlevel.ca) |
| 00:12.52 | *** join/#asterisk rcy (n=rcy@S010600207810532a.vc.shawcable.net) |
| 00:18.06 | *** join/#asterisk l3jj (n=chatzill@dsl093-001-038.det1.dsl.speakeasy.net) |
| 00:18.16 | *** part/#asterisk l3jj (n=chatzill@dsl093-001-038.det1.dsl.speakeasy.net) |
| 00:19.55 | *** join/#asterisk justdave (n=dave@unaffiliated/justdave) |
| 00:21.09 | *** part/#asterisk beek (n=klinebl@65.211.106.242) |
| 00:22.37 | *** part/#asterisk Simon-- (n=sim@staff-nat.netnation.com) |
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| 01:18.47 | esaym | man I am getting an error compiling zaptel 1.4.9.2: http://pastebin.com/m320bb91d |
| 01:18.50 | esaym | anyone have any idea |
| 01:19.21 | esaym | this is a ubuntu fiesty box with a real time 2.6.22 kernel |
| 01:20.12 | *** part/#asterisk tessier_ (n=treed@kernel-panic/sex-machines) |
| 01:20.17 | esaym | I got all the dependencies I think |
| 01:20.50 | JayTee52 | did you do a ./configure first? |
| 01:21.54 | rob0 | lacking kernel source? |
| 01:22.05 | rob0 | (or headers) |
| 01:22.13 | esaym | yea, headers are there |
| 01:22.26 | esaym | ./configure goes fine |
| 01:22.32 | *** join/#asterisk jmardonesk (n=jmardone@200-126-74-37.bk5-dsl.surnet.cl) |
| 01:22.32 | JayTee52 | what distro of linux? |
| 01:22.39 | rob0 | 01:19 < esaym> this is a ubuntu fiesty box with a real time 2.6.22 kernel |
| 01:22.40 | esaym | correction, this is a ubuntu gusty box |
| 01:23.11 | esaym | amd64 real time kernel |
| 01:24.51 | jmardonesk | hello to all, I have a very basic question about asterisk, is possible to make automated interactive surveys ? |
| 01:25.42 | jmardonesk | by example i make a call, a voice say a question, then the user press a number ... |
| 01:25.58 | rob0 | It is. Next question ... you, over there in the corner? |
| 01:27.44 | [TK]D-Fender | jmardonesk, Yes. |
| 01:28.08 | *** join/#asterisk nighty^ (n=nighty@v090008.dynamic.ppp.asahi-net.or.jp) |
| 01:30.06 | jmardonesk | [TK]D-Fender, but I can store the results?, this can be done using AGI? |
| 01:30.22 | [TK]D-Fender | jmardonesk, plenty of ways. |
| 01:31.00 | *** join/#asterisk BeeBuu (n=beebuu@219.132.191.147) |
| 01:31.40 | JayTee52 | esaym, enable multiverse and universe in your repos and search in Synaptic for Asterisk and Zaptel |
| 01:31.57 | esaym | JayTee52: they are old as heck |
| 01:32.45 | JayTee52 | then use this debian how-to as a checklist reference for any dependencies: http://www.asteriskguru.com/tutorials/asterisk_installation_compilation_debian.html |
| 01:33.30 | jmardonesk | [TK]D-Fender, can you tellme some ways, for search information. |
| 01:34.54 | [TK]D-Fender | jmardonesk, SQL in your dialplan, AstDB, Text output via "System", AGI, etc. |
| 01:37.22 | JayTee52 | esaym, here's a how-to that's more specific to Gutsy |
| 01:37.25 | JayTee52 | http://www.voip-info.org/wiki/view/How+to+install+Asterisk+1.4+and+FreePBX+2.3.1+in+Ubuntu+Linux&view_comment_id=15498 |
| 01:39.18 | *** join/#asterisk lowlevel (n=Stuart@lowlevel.ca) |
| 01:42.14 | esaym | JayTee52: I was using the dependency list from here: http://howtoforge.com/asterisk-zaptel-libpri-misdn-asterisk-addons-asterisk-gui-on-debian-etch-p2 |
| 01:42.30 | esaym | it works fine etch and ubuntu fiesty |
| 01:45.45 | JayTee52 | there are a few libs in the last link I posted that aren't in that link you just posted. |
| 01:47.02 | esaym | yea I see, I am going to try and install the full kernel source too |
| 01:47.40 | esaym | only libs I don't have from that list are : libncursesw5-dev ncurses-term |
| 01:47.50 | JayTee52 | openssl? |
| 01:48.04 | JayTee52 | and yeah, the full kernel sources are listed in the how-to |
| 01:48.44 | JayTee52 | sudo apt-get install linux-headers-2.6.22-14-server linux-image-2.6.22-14-server linux-source-2.6.22 |
| 01:49.00 | jmardonesk | [TK]D-Fender, thanks, i will read about that... |
| 01:49.48 | *** join/#asterisk bronson (n=bronson@adsl-68-122-117-135.dsl.pltn13.pacbell.net) |
| 01:53.59 | esaym | hmm freaking heck. I don't know. I found a list I made of the dependencies that I install on my other asterisk servers. This server has all of them install yet it is not compiling. I am wondering if this real time kernel is messing with it? |
| 01:54.22 | JayTee52 | install the generic kernel then |
| 02:07.01 | esaym | gerrr installing kernel source didn't work either |
| 02:07.19 | esaym | Im gunna dump this realtime kernel |
| 02:08.32 | adeel | esaym, can you pastebin the compile error |
| 02:08.45 | adeel | esaym, the kernel shouldn't have anything to do with asterisk not compiling |
| 02:09.02 | esaym | it is zaptel |
| 02:09.17 | adeel | ah, then, yeah it could be kernel related |
| 02:10.21 | esaym | One sec, will paste the whole log |
| 02:11.42 | esaym | http://pastebin.com/m5c3c5b8 full log |
| 02:11.52 | *** join/#asterisk profounded (n=Bryan_Ru@c-68-82-34-163.hsd1.nj.comcast.net) |
| 02:12.30 | esaym | Google shows me other apps that fail to compile and ending up with the "size of array âtype nameâ is negative" error but I haven't found a fix yet if that is the problem.. |
| 02:12.54 | adeel | that seems to be more of a bug in the zaptel ver |
| 02:13.06 | adeel | try using an older ver, or commenting out the wcte12xp stuff in the Makefile |
| 02:13.16 | adeel | unless you need the wcte12p driver |
| 02:14.34 | *** join/#asterisk fskrotzki_ (n=fskrotzk@cpe-74-74-245-250.rochester.res.rr.com) |
| 02:14.46 | esaym | oh it is a driver? I only need ztdummy |
| 02:15.07 | adeel | iirc, zaptel supports make menuconfig |
| 02:15.16 | adeel | so run that and disable drivers you don't need |
| 02:15.24 | adeel | then run make |
| 02:17.46 | esaym | I think you are right. I got to install ncurses though |
| 02:17.49 | *** join/#asterisk mikong (n=gones@203.193.37.251) |
| 02:17.59 | esaym | although I thought they were already installed |
| 02:18.14 | *** join/#asterisk fskrotzki_ (n=fskrotzk@cpe-74-74-245-250.rochester.res.rr.com) |
| 02:19.37 | adeel | rule 1...never assume anything |
| 02:20.29 | esaym | well libncurses5 is installed |
| 02:21.19 | adeel | doesn't mean ncurses itself is installed |
| 02:21.37 | adeel | the lib is installed |
| 02:21.49 | adeel | bbl |
| 02:21.55 | esaym | hmm |
| 02:26.46 | [TK]D-Fender | ~assume |
| 02:26.47 | jbot | About assumptions : http://www.moviewavs.com/php/sounds/?id=gog&media=WAVS&type=Movies&movie=Long_Kiss_Goodnight"e=assume.txt&file=assume.wav |
| 02:31.28 | JayTee52 | Assumption is the mother of all fuckups |
| 02:31.55 | esaym | adeel: "make menuselect" works |
| 02:32.26 | JayTee52 | yay! |
| 02:37.32 | vector | anyone usinng IAX with vitelity.net? |
| 02:37.49 | vector | even tho they apparently frown upon it |
| 02:38.19 | adeel | is there a way to determine the sip channel id of a call after it has completed? |
| 02:40.28 | rob0 | I was just looking at vitelity.net today, but I figured I'd use SIP if I sign up. |
| 02:41.16 | vector | they're not bad.. I've been with them for all of 3 days now... |
| 02:41.57 | vector | I get jittery oubound audio with sip... but when I set it up IAX the problem doesnt exist... but I am unable to make outbound calls via IAX |
| 02:42.04 | vector | thus far at least |
| 02:43.27 | rob0 | debuggery? |
| 02:43.48 | *** join/#asterisk test34 (n=test34@unaffiliated/test34) |
| 02:45.36 | vector | well so var I've just been looking at the console with verbosity at 3... I get a notice "Auto-congesting call due to slow response -- Hunup 'IAX2/vitel-outbound-2'" |
| 02:45.56 | rob0 | help iax2 set |
| 02:51.10 | adeel | vector, try raising the verbose level and maybe enable core/iax debug |
| 02:51.52 | vector | how does one enable that stuff? |
| 02:51.58 | vector | I'm relatively new with asterisk |
| 02:52.06 | vector | at least as far as solving issues like this LOL |
| 02:56.26 | *** join/#asterisk coppice (n=chatzill@99.166.17.210.dyn.pacific.net.hk) |
| 02:56.35 | esaym | adeel: yes disabling the extra drivers worked |
| 02:58.06 | rob0 | in the console type "help iax2 set" |
| 02:58.28 | rob0 | console means it's interactive |
| 03:00.09 | adeel | esaym, i'm glad |
| 03:01.00 | rob0 | afk |
| 03:01.10 | vector | ah. very good then... I'll give it a shot.. thx rob0 |
| 03:03.32 | esaym | anyway to disable dundi from starting? |
| 03:05.32 | JunK-Y | esaym: noload => pbx_dundi.so in ur modules.conf |
| 03:06.36 | adeel | or just remove the .so |
| 03:07.37 | esaym | that worked. Thanks! |
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| 03:17.03 | *** join/#asterisk PaRruDoO (n=cris@201.22.56.13.static.gvt.net.br) |
| 03:24.52 | adeel | how many digits are there in an international phone number? |
| 03:26.06 | *** join/#asterisk gitguy (n=diego@adsl-134-171.click.com.py) |
| 03:26.14 | *** part/#asterisk gitguy (n=diego@adsl-134-171.click.com.py) |
| 03:27.01 | [TK]D-Fender | adeel, as many as the can be angels dancing on the head of a pin. |
| 03:28.15 | adeel | fun...that only makes handling domestic and international calls that much easier |
| 03:29.41 | LARefugee | ~dundi |
| 03:29.42 | jbot | i heard dundi is at http://www.dundi.com. DUNDi, an optional Asterisk component, is a distributed, decentralized peer to peer network that provides routes to PSTNs between peers on the same DUNDi network. |
| 03:31.49 | *** join/#asterisk sysreq (n=sysreq@unaffiliated/sysreq) |
| 03:39.31 | *** join/#asterisk BeeBuu (n=beebuu@219.132.191.147) |
| 03:40.42 | BeeBuu | queue application with 't' or 'T' option,how to transfer the call? |
| 03:44.32 | BeeBuu | transfer to where? |
| 03:48.10 | *** part/#asterisk workaphobia (n=workapho@ool-457fa98d.dyn.optonline.net) |
| 03:50.48 | esaym | how you change the port for iax? |
| 03:51.13 | esaym | I can change the listen port but not the port the server uses to connect remotely |
| 03:52.53 | *** join/#asterisk erojasv (n=erojasv@190.40.56.67) |
| 03:56.32 | BeeBuu | any expert help,please |
| 03:57.23 | BeeBuu | application queue with 't' or 'T' option,how to transfer the call to a number? |
| 03:58.03 | plik | BeeBuu: it's set in features.conf (# by default I think) |
| 04:01.33 | BeeBuu | i got you,let me check. |
| 04:08.26 | *** part/#asterisk BeeBuu (n=beebuu@219.132.191.147) |
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| 04:15.11 | esaym | so know one knows about iax ports? |
| 04:16.19 | drmessano | Well |
| 04:16.27 | drmessano | IAX uses a port |
| 04:16.43 | drmessano | Why are you changing it? |
| 04:18.03 | esaym | I have 2 servers on one server |
| 04:19.13 | esaym | I see [general] takes the port= option. But what about the client/context? I can make it listen on a different port but I can't tell anything to use that port to connect to |
| 04:20.10 | esaym | been messing with it for like an hour >_< |
| 04:21.37 | drmessano | why do have two servers on one server? |
| 04:22.55 | djs | drmessano: It is an experiment to see how far off the timing can be thrown on multiple VMs... |
| 04:23.21 | esaym | we ran out of computers and we will be getting another one in a few months |
| 04:23.51 | plik | a few *MONTHS* !!?? |
| 04:24.04 | esaym | yea no rush |
| 04:24.08 | plik | introduces esaym to the idea of 'next day delivery' |
| 04:24.34 | drmessano | lol |
| 04:25.19 | esaym | beats head |
| 04:25.35 | drmessano | Umm |
| 04:25.49 | drmessano | You cant accomplish what youre doing with ONE asterisk instance? |
| 04:26.00 | esaym | it is a special instance |
| 04:26.34 | esaym | I don't know, screw it. I will just disable iax on the one that doesn't need it and stick to stock ports........... |
| 04:31.08 | esaym | see the user contexts in sip.conf can take a port= option. But that doens't work in iax.conf. I can set the bindport (listen port) but I can't figure out how to tell a remote asterisk box to connect to the new port that this box is now listening on. |
| 04:31.31 | esaym | make since? sinse? cense? |
| 04:32.18 | jbeez | sense |
| 04:34.56 | coppice | noncents |
| 04:35.28 | esaym | innocents |
| 04:35.48 | jackson__ | esaym, can't you bind multiple ips to the box and then have each instance of asterisk bind to the different ips? |
| 04:36.41 | esaym | yea I could. But since this is just temp I don't want to mess with it too much. I figured that changing the port would be very easy |
| 04:37.45 | esaym | blah, I am going to kill iax on the other instance. I don't need. the "other guy" is going to throw a fit for me messing with his config though |
| 04:38.00 | esaym | but I will throw hot coffee on him |
| 04:43.13 | drmessano | So let me get this right |
| 04:43.20 | drmessano | you wont change the bind address for the instance |
| 04:43.34 | drmessano | But youre willing to move the default IAX port? |
| 04:43.48 | drmessano | Because you dont want to mess with things too much |
| 04:45.36 | esaym | basically |
| 04:45.57 | esaym | I guess bind address woudn't be too hard either |
| 04:55.57 | *** join/#asterisk shital (n=shital@122.167.66.128) |
| 04:56.15 | drmessano | Your logic is extremely flawed |
| 04:59.11 | *** join/#asterisk Ritzerisk (n=Ritztech@24.120.190.66) |
| 05:24.01 | *** join/#asterisk kamanashisroy (n=kamanash@202.56.7.193) |
| 05:35.00 | Ritzerisk | sooo where could i get a auto dialer |
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| 07:04.14 | vector | anyone have outbound callerID working with IAX and Vitelity? |
| 07:18.55 | *** join/#asterisk steliosk (n=Stelios@85.75.201.253) |
| 07:19.41 | ectospasm | vector: Does Vitelity advertise that they respect whatever you put in CallerID? |
| 07:20.21 | ectospasm | I can't remember the VOIP provider I was using back in the day, but I'd be surprised if any of them allow arbitrary CallerID anymore, because of the potential for abuse. It still happens, no doubt |
| 07:21.13 | vector | I just solved my IAX problem (it was a typo on my part) |
| 07:21.47 | vector | and so far.. I've been able to send out whatever I want as CID and it reaches the other end :) |
| 07:22.09 | vector | but I dont think they advertise it pre se |
| 07:22.11 | vector | per* |
| 07:22.49 | jblack | Hmmm. What does "per se" mean? |
| 07:23.08 | vector | um... like "necessarily" |
| 07:23.11 | vector | sort of |
| 07:23.20 | jblack | No. It's latin for "by itself" |
| 07:24.02 | vector | in that sentance it would be replaceable with necessarily.. good enough for me |
| 07:24.08 | jblack | Just letting you know, so that you don't use misuse it in a job interview. :) |
| 07:24.29 | vector | "I dont think they advertise it by itself" .. still valid |
| 07:24.30 | vector | :) |
| 07:24.37 | vector | but good to know :D |
| 07:25.02 | jblack | Yeah. I went years misusing "for all intensive purposes" before someone took the time to tell me not to be a dope |
| 07:25.19 | vector | LOL |
| 07:25.53 | vector | you would have a hayday here.. I'm in south dakota... moved here 9 years ago.. everyone says "I seen ___ the other day" |
| 07:26.07 | vector | possibly the most irritating thing in the world |
| 07:27.43 | jblack | no. There's worse. |
| 07:28.13 | jblack | Here, they use "yous". LIke "How are yous". On rare occasion, you'll even hear "youses". As in, two or more groups of "yous" |
| 07:29.23 | *** join/#asterisk angeldavid (n=angeldav@nelug/coreteam/pepo) |
| 07:29.24 | vector | anyone ever axe you a question? |
| 07:29.49 | jblack | not often, but rarely. |
| 07:30.42 | jblack | I grounded my daughter for an entire day for overusing "like". =) |
| 07:30.59 | vector | LOL |
| 07:32.51 | vector | I'll have to remember that... mine's not old enough to hit that stage yet.. but when she gets there perhaps I shall do the same |
| 07:32.55 | jblack | Her latest is "oh my god". |
| 07:33.13 | jblack | I've been yelling at her for that one fror a couple months. Ironically, as I'm an athiest |
| 07:33.24 | vector | hehehe |
| 07:34.31 | vector | so do you antagonize her by exclaming "on your god!" when she's around? |
| 07:35.26 | jblack | Nah. |
| 07:37.39 | jblack | The misphrasing that ticks me off the most is pronouncing caramel as "karmel" |
| 07:38.47 | jblack | holy shit |
| 07:38.48 | vector | hrm... don't think I've heard that word in quite some time... but yeah.. that would get irritating |
| 07:38.53 | jblack | There's a wine 1.0 rc out. |
| 07:38.58 | vector | ?!?! |
| 07:39.03 | vector | realy? |
| 07:39.06 | jblack | Yeah. wine 1.0rc1 is out. |
| 07:39.07 | drmessano | Dont fool yourself |
| 07:39.13 | vector | fools self |
| 07:39.17 | drmessano | There will be 30 RCs |
| 07:39.22 | vector | doh... and right after you told me not to :( |
| 07:39.25 | drmessano | lol |
| 07:39.40 | jblack | Isn't that one of the signs of the apocolypse, wine 1.0 coming out? |
| 07:40.05 | drmessano | I thought the roadmap showing 1.1 was pretty funny |
| 07:40.06 | jblack | apocalypse. |
| 07:40.17 | drmessano | "Whoa.. dont get ahead of yourselves here" |
| 07:40.39 | jblack | looks for the 1.1 map |
| 07:41.08 | *** join/#asterisk phix (n=threat@123-243-44-131.tpgi.com.au) |
| 07:41.35 | jblack | Oh look... google, when asked for where the roadmap is, wants to give me driving directions to a website |
| 07:45.00 | jblack | I can't find the wine 1.1 plans. I bet they're good, though |
| 07:45.46 | drmessano | Seems it was part of the "release plan".. but I see just one vague mention of 1.1 in there |
| 07:45.57 | drmessano | Hmm. Forget now where I saw that |
| 07:46.03 | drmessano | One of their blogs maybe |
| 07:47.35 | *** join/#asterisk XnOSX (n=XnOSX@77.226.14.236) |
| 08:03.09 | jblack | what are they chasing next? directx? |
| 08:07.41 | jblack | Oh, and it looks like 9 superdelegates jumped on barak tonight. |
| 08:11.23 | *** join/#asterisk ^shark_ (n=^shark_@216.104.202.233) |
| 08:12.07 | ^shark_ | hi friends, i used sox to convert my mp3 to wav and now i am hearing echos in my moh files, any help in this regard |
| 08:12.31 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 08:13.27 | ^shark_ | i am running asterisk 1.4.19.1 |
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| 09:29.29 | jblack | Heh. Speaking of word abuse.... one of the mythtv developers is griping about a user that is asking about closed captioning for the "death and hard of hearing" |
| 09:30.02 | jblack | I called him a vitality bigot. |
| 09:32.04 | *** join/#asterisk gardo (n=gardo@210.4.12.160) |
| 09:32.27 | jblack | What do we want? Zombie Rights! When do we want them? "UUUrrrhhhhhhhhh!" |
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| 11:40.57 | jblack | christ. Know what microsoft uses for network passwords? md4. md-frigging-four. |
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| 13:36.48 | *** join/#asterisk cjk (n=cjk@d90-129-59-39.cust.tele2.lu) |
| 13:37.03 | cjk | hi, how do i distinguish between the channel structure on my analog xorcom device and the bri channels on my junghanns card |
| 13:38.00 | *** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
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| 14:13.45 | *** join/#asterisk joez212 (n=jhart@CPE001c101b40b5-CM0018c0d91624.cpe.net.cable.rogers.com) |
| 14:14.00 | joez212 | hello |
| 14:14.25 | joez212 | is it possible to use voxalot with an asterisk server? |
| 14:14.35 | joez212 | adding more features? |
| 14:14.48 | JayTee52 | what's voxalot? |
| 14:15.17 | joez212 | a VSP |
| 14:15.22 | joez212 | but with some unique features |
| 14:15.35 | joez212 | VoXaLot is a VSP hub that provides you with an easy and convenient way to manage all of your other VoIP providers from one central location. |
| 14:16.37 | joez212 | if i call my voxalot SIP, and I miss the call.. I get an email with a wave file of the voicemail! |
| 14:16.43 | joez212 | I have been told asterisk can do this |
| 14:16.45 | joez212 | but in mp3 format |
| 14:17.02 | joez212 | (assuming I leave voicemail) |
| 14:18.33 | rob0 | ~voipbuster |
| 14:18.50 | rob0 | ~botsnack |
| 14:18.50 | jbot | aw, gee, rob0 |
| 14:18.54 | joez212 | ? |
| 14:18.59 | [TK]D-Fender | joez212, Yes * can e-mail VM's. MP3 will take some extra special scripting as * can't natively encode in it. |
| 14:19.33 | JayTee52 | [TK]D-Fender, know of any good How-To's for that? |
| 14:19.54 | joez212 | thanks tk |
| 14:20.10 | joez212 | i guess the big thing right now is figuring out what * can do without any extra scripting/work |
| 14:20.20 | joez212 | i have an ata |
| 14:20.24 | [TK]D-Fender | JayTee52, not specificly where but I saw a guide about a month ago. GOOGLE <- |
| 14:20.32 | [TK]D-Fender | joez212, ... |
| 14:20.34 | [TK]D-Fender | ~book |
| 14:20.35 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com, or see ~buybook |
| 14:20.36 | [TK]D-Fender | ~wikis |
| 14:20.37 | jbot | [~wikis] VoIP Wiki covering Asterisk, FreeSWITCH, TrixBox, SER, OpenSER, sipX, CallWeaver, and YATE. http://www.voip-info.org (c) Arte Marketing Inc / CommPartners |
| 14:20.38 | [TK]D-Fender | ^^^^^^^^ |
| 14:22.10 | JayTee52 | [TK]D-Fender, thanks, I'll Google for it. Might be on Voip-info.org but I don't remember seeing it. |
| 14:26.38 | *** join/#asterisk stimpie (n=stimpie@84-104-6-10.cable.quicknet.nl) |
| 14:28.43 | *** join/#asterisk acxty (n=acxty@201.220.132.138) |
| 14:31.12 | acxty | I have a agi application that need to be run like 20 seconds. If the user hangup before that 20 seconds the agi script don't finish doing its work. So what I did was using a DeadAgi. I received a warning saying to don't use a DeadAgi on a live channel to use a Agi() that this may cause problem. Is it possible that a agi finish doing its job even if the user hangup? |
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| 14:46.20 | *** part/#asterisk jivco (n=jivco@85.187.217.6) |
| 14:47.28 | *** join/#asterisk russellb (n=russell@asterisk/developer-and-stable-maintainer/drumkilla) |
| 14:47.28 | *** mode/#asterisk [+o russellb] by ChanServ |
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| 15:27.45 | R-MAN | yo yo |
| 15:28.44 | RoyK | jau |
| 15:29.42 | RoyK | in sip realtime, the useragent is never stored in the db |
| 15:30.25 | RoyK | I wrote a patch to store it there as it's pretty nice for customer service to see what sort of equipment the customer is using... |
| 15:30.45 | russellb | yep, i have a patch pending for that |
| 15:30.51 | russellb | came from switchvox |
| 15:31.00 | russellb | haven't gotten around to merging it |
| 15:32.23 | *** join/#asterisk chronos_ (n=a23@adsl-99-141-170-109.dsl.emhril.sbcglobal.net) |
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| 15:37.39 | RoyK | russellb: should I send you mine? |
| 15:39.13 | RoyK | I merged it with callweaver a year or so back, but the progress in callweaver isn't really that good, so I had to use asterisk for this new setup - we're moving to sip termination now, so we won't need t38 gateway and then asterisk may be a better choice |
| 15:39.46 | russellb | thanks for the offer, but i've got it |
| 15:48.37 | Ritzerisk | i got a zap channell ringing and ringing sometimes and it finally answers and somettimes it just answers right away.... using the hylafax client... weird |
| 16:03.51 | errr | I am having a problem with the voicemail app. WARNING[18944] app_voicemail.c: Failed to open file: /var/spool/asterisk/voicemail/default/7005/tmp/3oLbRP.WAV: No such file or directory I get this every time I get a voicemail and then when the voicemail shows up in my inbox the attachment is 0 size |
| 16:04.32 | [TK]D-Fender | errr, goo look at the .txt file for it, and check your actual file. |
| 16:05.58 | errr | [TK]D-Fender: they are fine, the problem is in the make_email_file() function. When base_encode() is called thats where it exits with -1 |
| 16:06.32 | rob0 | permissions? |
| 16:06.37 | RoyK | errr: does the directory /var/spool/asterisk/voicemail/default/7005/tmp/ exist? |
| 16:06.46 | errr | yes |
| 16:06.49 | *** join/#asterisk af_ (n=getsmart@88-149-230-6.dynamic.ngi.it) |
| 16:06.51 | RoyK | [ -d /var/spool/asterisk/voicemail/default/7005/tmp/ ] && echo yes |
| 16:07.48 | RoyK | errr: is it writable by the asterisk user? |
| 16:08.08 | errr | http://fluxbox.pastebin.ca/1013809 |
| 16:08.09 | errr | yes |
| 16:08.54 | errr | the whole /var/spool/asterisk/ is asterisk:asterisk |
| 16:08.57 | RoyK | it's 775 - what's the group name? |
| 16:09.01 | RoyK | ok |
| 16:10.24 | errr | I also did verify it is running as "asterisk" |
| 16:10.28 | errr | root 18885 0.0 0.1 4448 532 pts/0 S 11:00 0:00 /bin/bash /usr/sbin/safe_asterisk -U asterisk -G asterisk |
| 16:10.31 | errr | asterisk 18896 0.6 1.8 25764 9532 pts/0 Sl 11:00 0:03 /usr/sbin/asterisk -f -U asterisk -G asterisk -vvvg -c |
| 16:11.08 | *** join/#asterisk anthm (n=anthm@adsl-76-229-232-205.dsl.milwwi.sbcglobal.net) |
| 16:15.43 | errr | its like the temp file it opens for writting is named one thing then when it trys to send the file to me it trys to open some other file name that was never written |
| 16:16.52 | errr | here is another example from a couple days ago: http://fluxbox.pastebin.ca/1012341 I consolidated the relevant lines I think |
| 16:17.43 | errr | Could I have some bad memory in my system that would cause this? |
| 16:19.18 | errr | its like the filename ref is getting messed up somehow.. |
| 16:21.09 | rob0 | Bad memory? Is the machine randomly crashing? |
| 16:21.48 | errr | no |
| 16:23.20 | errr | Im just grasping at straws though, I have the same setup exactly (same version of asterisk and same dial plan everything) on another system and this all works fine though |
| 16:29.43 | *** join/#asterisk shinao1 (n=shinao1@smtp.gtbank.com) |
| 16:35.00 | *** join/#asterisk adjohn (n=adjohn@i220-221-4-188.s05.a013.ap.plala.or.jp) |
| 16:37.20 | RoyK | errr: same version of asstrix? |
| 16:39.26 | errr | yep |
| 16:40.02 | rob0 | The voicemailbox for 7005 always fails like this? |
| 16:40.09 | errr | for any extension |
| 16:46.16 | errr | http://fluxbox.pastebin.ca/1013840 |
| 16:46.49 | *** part/#asterisk frocos11292 (n=ask@firewall.vipvoz.com) |
| 16:47.39 | *** join/#asterisk esaym (n=user@cpe-70-120-89-6.satx.res.rr.com) |
| 16:48.21 | errr | it opens the 3 temp files for writting just fine, its just when it finishes and then needs to open again to write the email its like it forgot the name of the file it opened first and so it makes up a new filename which isnt there |
| 16:54.44 | drmessano | Does it have read permissions as well as write for the temp folder? |
| 16:54.51 | errr | yes |
| 16:56.04 | errr | drmessano: but unless Im missing something the temp file it opened and wrote is named 1 thing, and the open it wants to open to create the email with is named another |
| 16:56.45 | errr | x=0, open writing: /var/spool/asterisk/voicemail/default/231/tmp/AfFoUu format: wav49, 0x932a458 |
| 16:56.46 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 16:56.48 | errr | app_voicemail.c: Failed to open file: /var/spool/asterisk/voicemail/default/231/tmp/atb5fz.WAV: No such file or directory |
| 16:56.52 | errr | see |
| 16:57.15 | errr | should it be trying to open AfFoUu.WAV |
| 16:57.51 | drmessano | Is it writing AfFoUu? |
| 16:59.53 | errr | *shrug* I guess I can run ll while I leave a message and see |
| 17:00.13 | drmessano | Lack of info = weak sauce |
| 17:02.26 | errr | yes it is |
| 17:02.38 | errr | heh, you stole my line ;) |
| 17:03.05 | drmessano | I know |
| 17:03.13 | drmessano | ..and then used it against you |
| 17:03.16 | drmessano | ;) |
| 17:04.05 | errr | http://fluxbox.pastebin.ca/1013855 |
| 17:05.00 | Madkiss | die lpi 101er-sachen sind ja wirklich basic ohne ende. |
| 17:05.45 | errr | I dont understand how the file name gets lost |
| 17:06.32 | drmessano | You got me there |
| 17:07.00 | Madkiss | oh. |
| 17:07.04 | Madkiss | this was the wrong channel. |
| 17:07.06 | Madkiss | sorry. |
| 17:09.36 | drmessano | Si Senor |
| 17:12.45 | Madkiss | it is so confusing to get along with 48 channel windows ;) |
| 17:14.29 | errr | not all on freenode too |
| 17:14.47 | errr | chumps will only let you connect to like 20 or so rooms max |
| 17:15.20 | errr | I get that stupid (You have joined to too many channels) message all the time |
| 17:16.01 | drmessano | I only get that when I join #asterisk-pr0n |
| 17:21.59 | [TK]D-Fender | drmessano, You've clrealy gone too far and must be stopped.... |
| 17:32.18 | drmessano | lol |
| 17:34.42 | *** join/#asterisk Greek-Boy (n=email@41.221.58.14) |
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| 18:06.56 | *** join/#asterisk teknoprep (n=chris@unaffiliated/teknoprep) |
| 18:07.02 | teknoprep | hey all |
| 18:07.04 | teknoprep | when setting up SIP NAT |
| 18:07.13 | teknoprep | localnet= i have more than one subnet to add |
| 18:07.27 | teknoprep | would it be.... localnet=192.168.0.0/16&10.0.0.0/8 |
| 18:07.33 | teknoprep | or would i have to lines of |
| 18:11.36 | drmessano | No |
| 18:11.38 | drmessano | Two lines |
| 18:13.16 | teknoprep | ahh |
| 18:13.22 | teknoprep | i think that would be the problem |
| 18:13.45 | teknoprep | thanx bro |
| 18:13.51 | teknoprep | i was getting random 1way audio |
| 18:21.19 | CCFL_Man2 | my candlestick phone sounds like crap with a sidetone subset |
| 18:43.19 | *** join/#asterisk djs (n=djs@unaffiliated/djs26) |
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| 19:01.02 | profounded | <PROTECTED> |
| 19:02.26 | rob0 | hax: no such user |
| 19:02.35 | profounded | thx lol |
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| 21:01.42 | *** join/#asterisk HighOctane (n=HighOcta@68-185-143-114.dhcp.jcsn.tn.charter.com) |
| 21:02.12 | HighOctane | hello, anyone available? |
| 21:02.27 | moy | HighOctane: available for? |
| 21:03.25 | HighOctane | Need a question answered: I am looking to possibly edit or add a new ring strategy in asterisk, was curious where to find the config? |
| 21:04.29 | moy | no clue |
| 21:05.32 | moy | I have disclaimed to Digium my code for chan_zap to have R2 support. The library required for this disclaimed code in chan_zap to work is LGPL and I wrote it too. If I want to add code from other LGPL lbrary to my LGPL library, there is no conflict, right? |
| 21:05.40 | HighOctane | has anyone here ever been able to transfer a call that has been picked up by their cell phone back "into the asterisk" pbx by initiating a transfer on the cell phone. |
| 21:06.12 | HighOctane | can you use a cell phone transparently as if it was part of the asterisk network? |
| 21:06.27 | moy | HighOctane: that sounds like a user question, try #asterisk |
| 21:06.34 | moy | oops |
| 21:06.38 | moy | wrong channel |
| 21:06.41 | moy | lol |
| 21:07.02 | HighOctane | no problem. I am in asterisk right? |
| 21:07.20 | moy | correct, I thought I was in asterisk-dev for a second |
| 21:07.24 | *** join/#asterisk Keltus (i=Keltus@about/cooking/nakedchef/beefstew/Keltus) |
| 21:10.37 | HighOctane | Still no takers on my question? Was my question clear enough? |
| 21:15.25 | *** join/#asterisk jeffspeff (n=jeff@c-68-53-81-14.hsd1.ky.comcast.net) |
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| 21:18.20 | aydiosmio | HighOctane: you question is indeed not clear |
| 21:18.33 | aydiosmio | do you mean use the phone as if it were an extension? |
| 21:21.41 | jeffspeff | how could i set it to where when somebody enters a conference bridge they can't speek, but they can only listen, but the marked caller (there will be only one) will be able to speek to all of them. |
| 21:21.58 | HighOctane | aydiosmio: yes, sir. I would like to use the phone as if it were an extension, being able to transfer the caller back to another extension on the PBX, rather than telling them to hang-up, redial our phone number, then enter the extension. I would like to initiate an attended and/or blind transfer from my cell phone. |
| 21:27.04 | jeffspeff | is that set using the peer, user and friend type options? |
| 21:32.32 | errr | What I think HighOctane wants is this: exten 100 rings, and has a follow me on it. His cell is in the list of follow me. He picks the call up from his cell. Now he wants to be able to transfer the call off to exten 101 because he is done helping them |
| 21:32.55 | errr | HighOctane: correct? |
| 21:37.05 | HighOctane | errr: That is correct. You got it. |
| 21:43.08 | *** join/#asterisk fx0ers (n=me@r74-195-218-125.msk1cmtc02.mskgok.ok.dh.suddenlink.net) |
| 21:44.38 | jeffspeff | errr, any ideas on my solution? |
| 21:47.49 | fx0ers | hi guys, I am having a problem getting zaptel to start after performing a yum update. |
| 21:48.06 | fx0ers | the error I am getting is: FATAL: Error inserting zaptel (/lib/modules/2.6.9-34.0.2.ELsmp/extra/zaptel.ko): Unknown symbol in module, or unknown parameter |
| 21:48.21 | *** join/#asterisk keulin (n=cray@247.126.64-86.rev.gaoland.net) |
| 21:48.56 | fx0ers | the machine is a phonebochs which is running CentOS 4.6 with kernel 2.6.9-34.0.2.ELsmp |
| 21:49.54 | MikeJ | you need to rebuild zaptel against the new kernel headers |
| 21:50.13 | fx0ers | MikeJ sorry for sounding dumb but how do I do that? |
| 21:50.27 | MikeJ | same way you did when you built zaptel the first time |
| 21:50.39 | fx0ers | MikeJ this was a machine pre-built from TrixBox |
| 21:50.48 | MikeJ | ahh.. |
| 21:50.48 | rob0 | Then you have a tricbox bug |
| 21:50.52 | fx0ers | so I did not initially build it |
| 21:51.03 | fx0ers | but everything worked great until after I yum updated |
| 21:51.09 | fx0ers | my guess is the kernel was updated |
| 21:51.11 | MikeJ | you can probably get away with downloading the src for the same zaptel version |
| 21:51.26 | MikeJ | make sure your kernel sources are there |
| 21:51.29 | MikeJ | headers |
| 21:51.30 | fx0ers | ok |
| 21:51.38 | MikeJ | and just configure and build the zaptel |
| 21:52.12 | fx0ers | hmm so I guess a yum remove then reinstall of zaptel won't work for me? It will have to be from source? |
| 21:54.46 | MikeJ | well. if the one in the repo was right.. it would have worked already :D |
| 21:55.23 | JayTee52 | and if the RPM for zaptel in the repo hasn't been updated since the kernel update was pushed it won't either |
| 21:56.29 | *** join/#asterisk comphappy_ (n=comphapp@c-71-227-212-6.hsd1.or.comcast.net) |
| 21:57.11 | comphappy_ | is there a way i can uses a serial modem to make a simple pre-recorded phone call |
| 21:58.29 | JayTee52 | simple answer: no |
| 21:58.33 | JayTee52 | not in asterisk |
| 21:58.59 | JayTee52 | you'd need an FXO capable card |
| 21:59.03 | *** join/#asterisk adr3nalin3 (n=afink@66.172.245.81) |
| 22:06.02 | *** join/#asterisk adelas (n=booger@rrcs-24-199-21-138.west.biz.rr.com) |
| 22:06.07 | adr3nalin3 | Hello everybody. I am having an issue with a Sangoma A101 T1/E1 card. I have installed the wanpipe drivers and everything but when I issue a wanrouter start I get this error message -> Configuring interfaces: w1g1 w1g1: unknown interface: No such device, any advice? Thanks. |
| 22:06.27 | adelas | Hi, anyone know about the polycom 320 feature of swaping out line 2 button for a message button? |
| 22:10.25 | adelas | in other words, how do i program button 2 to go to automatically dial a number? |
| 22:16.21 | Greek-Boy | has 12 cisco 7912's that are blown :( won't power up |
| 22:17.48 | JayTee52 | Power Over Ethernet from a POE equipped switch? |
| 22:23.13 | adr3nalin3 | Another issue is when I run ztcfg I get this error -> line 0: Unable to open master device '/dev/zap/ctl' ...Also when I do an ifconfig I don't see the Sangoma card. I should shouldn't I? |
| 22:25.16 | JayTee52 | no, it shouldn't show up in ifconfig |
| 22:25.16 | JayTee52 | have you done an lsmod to see if the sangoma module is loading? |
| 22:26.52 | adr3nalin3 | I see wanrouter is loaded. |
| 22:27.43 | adr3nalin3 | wanrouter 39592 5 wanec,wanpipe_lip,af_wanpipe,wanpipe,wanpipe_syncppp, sdladrv 60784 2 wanpipe,wanrouter |
| 22:28.40 | JayTee52 | adr3nalin3, check this and see if you missed something in the config. I use Digium's TE212P so I'm not familiar with setting up Sangoma PRI cards. http://www.voip-info.org/wiki/view/Sangoma+A200+Configuration |
| 22:32.34 | adr3nalin3 | I'm beginning to think maybe it is just giving me some misleading error messages. b/c wanrouter status shows the card active....I can't test b/c the ISP hasn't provisioned the PRI yet. |
| 22:34.10 | JayTee52 | is the A101 a single port card? |
| 22:39.53 | adr3nalin3 | yes it is |
| 22:41.22 | JayTee52 | ah, yes. pity that. I have a two span card from Digium so I was able to make a crossover cable, set one span to pri_net with clocking and loop the spans so I could send test calls out one span and direct the call incoming to a SIP phone for testing. |
| 22:42.26 | JayTee52 | that let me do alot of testing a deliberate breaking so I could see the various console errors etc so I had a better idea of what to expect when I connected up live to the telco's PRI's. |
| 22:47.42 | *** join/#asterisk YoungProGuru (n=root@cpe-24-198-43-238.buffalo.res.rr.com) |
| 23:09.04 | JT | JayTee52: an A200 is not a pri card |
| 23:18.49 | *** join/#asterisk mritchie (n=ritchiem@87-194-117-238.bethere.co.uk) |
| 23:23.55 | mritchie | Hello, I was just wondering if callerid on the X100P cards in the UK were supported now. I've seen a lot of talk about the 'history' patch but don't see it actually making it on to trunk. Is there a way to get callerid working on the X100P in the UK with the latest 1.6 beta? Thanks |
| 23:32.14 | *** join/#asterisk lmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 23:32.14 | *** mode/#asterisk [+o lmadsen] by ChanServ |
| 23:33.01 | mritchie | anyone here? |
| 23:35.14 | fluff | ye+ |
| 23:35.17 | fluff | ep |
| 23:35.18 | fluff | yep |
| 23:35.19 | fluff | goddamn |
| 23:35.22 | fluff | hard to type. |
| 23:37.54 | *** join/#asterisk Buzer (n=buzer@cs27038053.pp.htv.fi) |
| 23:40.27 | Buzer | Hello. I have setup pretty basic Asterisk setup on home-enviroment (only one outside SIP channel) with callback for my mobile (I call the system, it hangs up and calls back, asks passcode and then allows me to call outside). I'm using following exten for that: _8XXXXXXXXXX,1,Transfer(SIP/${EXTEN:1}@voiprovider) |
| 23:41.30 | Buzer | However, when Asterisk attempts to transfer the call, it sends a SIP request with two Max-Forwards: fields and other VOIP gateway naturally sends me SIP/2.0 400 Multiple Max-Forwards headers. Is this a bug on Asterisk or have I missed some config line? |
| 23:42.17 | Buzer | (I'm using 1.4.19.1) |
| 23:45.43 | *** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 23:49.03 | Buzer | Sample attempt: http://pastebin.ca/1014211 |
| 23:55.59 | *** join/#asterisk talntid (n=eric@c-67-185-237-158.hsd1.wa.comcast.net) |