| 00:00.08 | Mercestes | by likely, I mean definately. |
| 00:00.30 | Mercestes | definitely even |
| 00:02.30 | *** part/#asterisk TimGroe (n=LivedTyp@202.172.97.35) |
| 00:02.31 | duxy786 | thanks for the help Mercester...I'll get in touch with Crona on #SER |
| 00:02.47 | duxy786 | and I'll hopefully fill pi in here again later |
| 00:02.51 | duxy786 | c ya! |
| 00:03.08 | Mercestes | good luck. |
| 00:03.12 | *** join/#asterisk Yourname`` (n=Miranda@unaffiliated/yourname/x-837320) |
| 00:03.23 | killfill_ | how do i make agents join queues these days?.. |
| 00:03.27 | killfill_ | (1.4.11) |
| 00:05.00 | Mercestes | killfill_, I use tasers. |
| 00:05.31 | `Sauron | I like scopalamine (sp?) |
| 00:06.13 | killfill_ | tasers?.. |
| 00:06.20 | killfill_ | hm.. |
| 00:06.45 | Yourname`` | Someone somewhere told me that Asterisk cannot go over 300-400 channel, no matter what system config. An installation of asterisk on a quadcore cpu, 2 gig ram box was doing close to 800-1000 channels, and the provider said "yes, we're seeing about 600-800 channels that you guys are sending". Seemed about right. All of a sudden today, the performance isn't the best. AMD is slow, call connect time is slow even after a reboot. Any insigh |
| 00:08.24 | *** join/#asterisk khronos (n=khronos@c-66-229-159-201.hsd1.fl.comcast.net) |
| 00:09.27 | Mercestes | Yourname``, if you *were* running 800-1000 channels on a quad-core with 2 gigs of RAM, I'm going to assume that you were doing no transcoding. |
| 00:09.47 | Mercestes | Yourname``, it is possible that you are now doing transcoding because of something silly either you or your provider changed. |
| 00:10.15 | Mercestes | You also could've lost one of yoru drives in yoru array and now your running off of restore mode, one of your memory chips could have gone bad... |
| 00:10.22 | *** join/#asterisk lesouvage (n=lesouvag@cc341200-a.assen1.dr.home.nl) |
| 00:10.37 | Mercestes | Or, someone finally told your asterisk box that it could only do 300-400 channels and it's suffering from self-doubt. |
| 00:13.03 | Yourname`` | Mercestes: No transcoding, you're right. |
| 00:13.06 | tzanger | Mercestes: hahaha |
| 00:13.19 | Yourname`` | lol |
| 00:13.23 | *** join/#asterisk mindCrime (n=chatzill@cpe-065-190-188-124.nc.res.rr.com) |
| 00:13.28 | Tebi | install that viagra module ;) |
| 00:13.44 | Mercestes | but if it functions for more than 4 hours, consult your physician. |
| 00:14.55 | GreggB | Now I've really upset her... |
| 00:14.58 | Mercestes | So, Chuck Norris goes into the doctor's office. |
| 00:15.03 | Yourname`` | Ran IOstat on a few boxes, and "await" which is the avg wait time on this box was exceptionally high. |
| 00:15.41 | Mercestes | He says, "Doctor, I've been up for more than four hours..what do I do?" |
| 00:15.50 | Mercestes | The doctors says, "how long ago did you take the viagra?" |
| 00:15.57 | Mercestes | And Chuck Norris says, "Viagara? What's that?" |
| 00:17.14 | Mercestes | Yourname``, hrm. I would definately check for transcoding. |
| 00:17.19 | *** join/#asterisk kiscokid (n=ron@208.106.35.66) |
| 00:17.34 | Yourname`` | hmm |
| 00:18.18 | Yourname`` | All the files it's playing are in ulaw. |
| 00:18.22 | Yourname`` | Provider uses ulaw. |
| 00:18.24 | Yourname`` | Check |
| 00:18.49 | Mercestes | hrm, a mystery it is then. |
| 00:20.07 | Yourname`` | Yup. |
| 00:20.48 | *** join/#asterisk anthm (n=anthm@mbf0736d0.tmodns.net) |
| 00:20.48 | *** mode/#asterisk [+o anthm] by ChanServ |
| 00:21.36 | kiscokid | If I have a machine with a Sangoma A200 and replace it with a Sangoma A101 do I need to do anything besides running /usr/sbin/wancfg_zaptel? |
| 00:23.57 | *** join/#asterisk WindBack (n=Administ@host60.190-138-93.telecom.net.ar) |
| 00:24.05 | ManxPower | kiscokid: you should just be able to edit the configs by hand and remove the 2nd span. |
| 00:24.16 | ManxPower | or is the A200 analog? |
| 00:24.32 | kiscokid | the A200 is analog |
| 00:24.39 | WindBack | Are there a package with asterisk 1.4 for debain etch?? |
| 00:24.47 | ManxPower | ah. wancfg_zaptel is prolly where it's at. |
| 00:25.06 | kiscokid | ok, I'll give it a try |
| 00:25.11 | ManxPower | WindBack: I'm sure there is, but I would not recommend you use it. Asterisk changes too fast for packages to be up to date. |
| 00:25.55 | tzafrir_home | WindBack, at buildserver.net |
| 00:26.06 | tzafrir_home | or at updates.xorcom.com/rapid |
| 00:27.23 | ManxPower | as tzafrir_home actually has his own debian based distro, he would be the one to believe. |
| 00:29.23 | *** join/#asterisk WindBack (n=Administ@host60.190-138-93.telecom.net.ar) |
| 00:29.55 | WindBack | tzafrir_home, can You repeatme the page? |
| 00:30.35 | ManxPower | tzafrir_home: WindBack, at buildserver.net |
| 00:30.35 | ManxPower | tzafrir_home: or at updates.xorcom.com/rapid |
| 00:30.40 | ManxPower | ManxPower: as tzafrir_home actually has his own debian based distro, he would be the one to believe. |
| 00:30.51 | tzafrir_home | deb http://updates.xorcom.com/rapid etch main |
| 00:31.03 | tzafrir_home | or: |
| 00:31.25 | WindBack | ManxPower, are there a good guide who tellme how to install * 1.4 from the sources well |
| 00:31.27 | WindBack | ? |
| 00:31.29 | tzafrir_home | deb http://pkg-voip.buildserver.net/debian etch main |
| 00:31.42 | tzafrir_home | The former also has pre-compiled zaptel-modules... |
| 00:31.44 | WindBack | tzafrir_home, yes, thanks |
| 00:32.15 | WindBack | ManxPower, on debian etch |
| 00:32.28 | ManxPower | WindBack: seriously, if you don't know linux well enough to install asterisk from source, networking enough to set up your own router, telecom enough to know what "ESF/B8ZS" is, you might consider a turnkey PBX. |
| 00:33.31 | WindBack | ManxPower, I know linux |
| 00:33.49 | tzanger | every time I see "turnkey" I think "turkey" |
| 00:34.22 | WindBack | ManxPower, I want a good guide who explainme how to conver asterisk in a daemon runing as asterisk user |
| 00:34.31 | tzafrir_home | tzanger, well, for me it means Turkey (the country) |
| 00:34.34 | WindBack | convert |
| 00:34.47 | tzafrir_home | WindBack, install the debs :-) |
| 00:35.06 | *** join/#asterisk groogs[h] (n=gregm@cbl-66-102-80-229.wtccommunications.ca) |
| 00:35.24 | tzafrir_home | Or: tell asterisk to run as user, and look at the logs to see where it shouts for "permission denied"... |
| 00:36.15 | WindBack | tzafrir_home, yes, I did that |
| 00:36.35 | tzanger | I think my next home asterisk system is going to either be flash-based (USB) or running on a wrt. |
| 00:36.42 | tzafrir_home | WindBack, my form of documentation is the debs... |
| 00:37.16 | WindBack | debs.. the debian packages?? |
| 00:41.10 | tzanger | hmm |
| 00:41.54 | tzanger | what do y'all recommend for 2-4FXS, 1FXO T38 capable ATAs? |
| 00:42.51 | ManxPower | tzanger: you know my answer to that. |
| 00:42.57 | tzanger | I do? |
| 00:43.27 | ManxPower | An Adtran Total Access 750 from eBay. Maybe two of them, which would still be cheaper than a single new one. |
| 00:43.48 | tzanger | heh |
| 00:43.56 | tzanger | if I want to do channel bank I'll use the Adit600 I have here |
| 00:44.00 | tzanger | I'm looking to get rid of asterisk |
| 00:44.26 | ManxPower | Hell, if you can get cheap local loops to your customers, you could just backhaul all the sites to a couple of channel banks. |
| 00:44.36 | tzanger | no no this is for my home |
| 00:50.55 | killfill_ | hm.. my phones are not detecting the palm key. |
| 00:50.57 | killfill_ | (SIP) |
| 00:51.01 | Siya | Qwell: ? |
| 00:51.01 | killfill_ | what could it be?.. |
| 00:52.28 | ManxPower | "palm key"? |
| 00:58.10 | *** join/#asterisk tiav (n=tiav@ram94-3-82-225-11-215.fbx.proxad.net) |
| 01:00.01 | killfill_ | never mind.. :) |
| 01:00.11 | *** part/#asterisk rnovotny22 (n=root@h460dfd16.area2.spcsdns.net) |
| 01:02.10 | *** part/#asterisk atomicd (n=Atomicd@74-206-0-80.static-ip.m.telepacific.net) |
| 01:02.52 | *** join/#asterisk bryanfe2 (n=chatzill@wsip-72-215-181-171.sb.sd.cox.net) |
| 01:04.11 | *** join/#asterisk `Sean (i=Sean@CPE002211569301-CM0011e6be76d9.cpe.net.cable.rogers.com) |
| 01:04.42 | bryanfe2 | hey all... I need to Park a user, and tell another running app which extension they were parked into (i.e. save it to a DB). I noticed that ParkAndAnnounce.c will set the environ variable "PARKEDAT" to the extension they were parked into, but since the app blocks for the rest of the call, I don't see how I can possibly make use of the variable. (i.e. parkandannounce.c never exits). Am I... |
| 01:04.44 | bryanfe2 | ...missing something? |
| 01:05.24 | killfill_ | guys.. in zap incomming calls, im doing Playback(/myfile). |
| 01:05.28 | *** join/#asterisk gardo (n=gardo@121.97.193.78) |
| 01:05.51 | killfill_ | users need to let the sound finish to begin to dial the numbers (ie.. support, etc) |
| 01:06.02 | killfill_ | how do i change this, and let them dial the number right away? |
| 01:06.45 | killfill_ | im usign Playback() and then WaiExten(30) |
| 01:07.25 | *** join/#asterisk __freedom__lover (n=eduardo@201-42-51-148.dsl.telesp.net.br) |
| 01:09.16 | killfill_ | oh damn. there is background()... |
| 01:09.33 | bryanfe2 | from app_parkandannounce.c: The variable ${PARKEDAT} will contain the parking extension into which the call was placed. Use with the Local channel to allow the dialplan to make use of this information. |
| 01:09.50 | bryanfe2 | Can someone explain the 2nd sentence? I can't understand how to make use of the information, but I do need to. |
| 01:12.37 | *** join/#asterisk |dennis| (n=dennis@200.32.233.84) |
| 01:18.18 | Nivex | I'm confused... does the PAP2T-NA not come with a power adapter? |
| 01:20.58 | Strom_M | why would it not come with a power adapter? |
| 01:21.41 | Nivex | Strom_M: good question, but when I added it to my cart at voipsupply.com, they automatically added a power brick |
| 01:21.51 | Nivex | voxilla does not do this |
| 01:21.58 | Strom_M | perhaps because voipsupply is staffed by morons |
| 01:22.04 | Nivex | and neither site makes mention of needing an additional power source |
| 01:22.07 | Strom_M | just thinking out loud here |
| 01:22.08 | [TK]D-Fender | and their prices suck |
| 01:22.19 | Qwell | Strom_M: perhaps? |
| 01:22.29 | Strom_M | Qwell: yes |
| 01:22.30 | Strom_M | perhaps |
| 01:22.44 | Strom_M | i don't know if that's the causal relationship, hence perhaps |
| 01:23.35 | Nivex | "Connect the included power adapter" |
| 01:24.21 | Nivex | despite voxilla charging me more for shipping, I think I'll get it from them. I've bought from them before anyway |
| 01:24.38 | Strom_M | Nivex: try telephonydepot.com |
| 01:25.26 | Nivex | Strom_M: looking |
| 01:26.00 | Nivex | nice. $5 cheaper on product and the lower shipping rates |
| 01:28.30 | *** join/#asterisk r0d3nt (i=nobody@foster.stonedcoder.org) |
| 01:28.32 | Strom_M | i like telephonydepot :) |
| 01:29.26 | *** join/#asterisk techie (n=techie@adsl-76-240-176-149.dsl.lsan03.sbcglobal.net) |
| 01:29.44 | [TK]D-Fender | considerably better and many happy clients |
| 01:30.23 | Nivex | Thanks for the tip |
| 01:32.23 | *** join/#asterisk DaveCanoe (n=Dave@canoe404.dclg.ca) |
| 01:34.31 | Mercestes | So, anyone hiring Asterisk techs? |
| 01:35.00 | *** join/#asterisk [hC] (n=hardcore@mail.metrobridge.com) |
| 01:35.44 | *** join/#asterisk CrazyTux[m] (n=CrazyTux@adsl-68-90-41-25.dsl.hstntx.swbell.net) |
| 01:37.42 | Mercestes | guess not. |
| 01:47.14 | dmz | he |
| 01:47.15 | dmz | heh |
| 01:48.20 | Mercestes | Is it just me or does asterisk-jobs.com have no actual asterisk-jobs on their site? |
| 01:49.24 | __freedom__lover | \quir |
| 01:50.37 | Mercestes | I am not a quir! |
| 01:54.46 | Mw3 | <PROTECTED> |
| 01:56.21 | Mercestes | I put in a resume at Digium but....I don't think I should've admitted to who I was on IRC. |
| 01:56.34 | `Sean | lol |
| 01:57.25 | *** join/#asterisk [TK]D-Fender (n=Joe@64.235.216.2) |
| 01:58.21 | Mercestes | Welcome back, Joe. |
| 01:59.41 | *** part/#asterisk Alowishus (n=jpenix@fwsdo.projectdesign.com) |
| 02:00.19 | *** join/#asterisk kiscokid (n=ron@208.106.35.66) |
| 02:03.12 | kiscokid | I tried changing Sangoma cards and reconfiguring but now none of the zap commands are available |
| 02:08.41 | kiscokid | maybe I'll recompile everything |
| 02:10.33 | *** part/#asterisk kiscokid (n=ron@208.106.35.66) |
| 02:10.55 | *** join/#asterisk macros73_ (n=cs@c-24-3-246-27.hsd1.mn.comcast.net) |
| 02:11.11 | Mercestes | Obviously recompile didn't work either. |
| 02:17.53 | *** part/#asterisk techie (n=techie@adsl-76-240-176-149.dsl.lsan03.sbcglobal.net) |
| 02:19.59 | [TK]D-Fender | Mercestes, Anything but average ;) |
| 02:20.23 | Mercestes | [TK]D-Fender, ?? |
| 02:20.34 | [TK]D-Fender | Mercestes> Welcome back, Joe. |
| 02:20.55 | [TK]D-Fender | Mercestes, its Colloquial Friday! |
| 02:21.05 | Mercestes | Ah.... |
| 02:21.45 | JT | you guys are behind the times |
| 02:21.48 | JT | it's saturday! |
| 02:22.18 | Mercestes | JT: ....not in where I'm at. |
| 02:23.49 | Mercestes | Hey Fender, hook me up with a job. |
| 02:24.38 | [TK]D-Fender | Mercestes, Hooking.... ask your doctor for a D&D cert first :p |
| 02:24.49 | Mercestes | D&D cert? |
| 02:24.54 | coppice | for most people its saturday |
| 02:24.55 | [TK]D-Fender | Drug & Disease |
| 02:25.01 | Mercestes | Oh, I'm good there. |
| 02:25.17 | peanut- | anyone knowhow long a grandstream holds its setting for after having power removed? |
| 02:25.23 | peanut- | they have a battery? |
| 02:25.56 | coppice | settings held by battery is sooooo 1970s |
| 02:26.10 | [TK]D-Fender | peanut-, EEPROM FTW! |
| 02:26.45 | Mercestes | peanut-, Forever. For the settings that you put into your grandstream resulted in it sucking, and the grandstream will always suck. Therefore, the fruits of yoru labor are forever immortalized. |
| 02:26.59 | coppice | nah. you get 10 to 100 years, depending on the temperature |
| 02:27.38 | [TK]D-Fender | coppice, I'll take it blue & seared ;) |
| 02:30.43 | peanut- | neat |
| 02:31.02 | peanut- | just want to make sure it won't erase on it's way to krautland |
| 02:31.09 | coppice | Flash Gordon, saviour of the configuration data |
| 02:31.25 | peanut- | for something this cheap I found it possible that you might have to reset after a bit of no power |
| 02:31.52 | coppice | flash is cheaper than a battery |
| 02:35.34 | *** join/#asterisk linxroute (n=linx@203.190.164.152) |
| 02:35.59 | *** join/#asterisk [hC] (n=hardcore@mail.metrobridge.com) |
| 02:45.10 | *** join/#asterisk asdx (n=diego@adsl-148-181.click.com.py) |
| 02:45.37 | *** join/#asterisk pitbossy (n=frankjr@adsl-67-115-67-130.dsl.lsan03.pacbell.net) |
| 02:48.09 | *** join/#asterisk r0d3nt (i=nobody@foster.stonedcoder.org) |
| 02:49.36 | pitbossy | Hello! I'm brand new to Asterisk. My PRI was just turned on today. I have a couple of questions if someone has the time to try to help. Thanks in advance. #1) I have two test phones (Grandstream GXP-2000) running on a dedicated network (for the phones only). Whenever I connect through the zaptel channels (Sangoma cards), if I speak very loudly the sound overloads on the phone and squeaks. |
| 02:49.48 | pitbossy | almost like a microphone mixer that the levels are too high... |
| 02:50.38 | pitbossy | I thought maybe it was the phones so I got an astra 480ct. It seems to have the same problem. When I call SIP to SIP (no zaptel involved), I dont notice the same issue. |
| 02:53.14 | Mercestes | ~phones |
| 02:53.20 | jbot | phones is probably http://bani.anime.net/phones/. While personal preference will dictate which phone works best for you, general consensus on a rough order of quality and suggestibility is as follows: Polycom (any), Aastra 480i, Aastra 5i Series, Cisco 7940+, Linksys SPA-9XX, Snom, and finally everything else. Do not consider Grandstream phones. Ever. places ... |
| 02:53.34 | Mercestes | pitbossy, also see rxgain/txgain under zaptel.conf |
| 02:53.38 | *** join/#asterisk linagee (n=linagee@about/linux/staff/linagee) |
| 02:53.55 | pitbossy | I was looking today in zaptel.conf and the gains are both at 0.0 |
| 02:54.22 | pitbossy | As far as the phones go, I see a lot of people are puking on the grandstreams, and I am rapidly joining the consensus.. Thanks. |
| 02:56.41 | [TK]D-Fender | pitbossy, test with another phone if you can. GXP's are well known for flakey firmware and the revision you're running may be one of the particularly bad ones. |
| 02:57.31 | [TK]D-Fender | pitbossy, try to grab the latest, but do a test with a soft-phone & headset as a sanity chech that your PRI isn't "running hot". |
| 02:58.16 | pitbossy | The second issue I am having is with the console/dsp ....chan_alsa doesn't recognize the on board audio....chan_oss does....It says it is setting up the console....but I cannot get any sound from the sound card...I can dial the console from a sip phone, but the console is dead....I can play sound from linux through the sound card... |
| 02:58.21 | [TK]D-Fender | pitbossy, So you think its only your PRI that is "high"? |
| 02:59.12 | pitbossy | Well, I am so new at this, I really dont know. It only seems to be when a call goes from inside to the outside world (telco). Internal to internal calls doesn't seem to suffer |
| 03:00.29 | pitbossy | And being new to this, it doesn't help that my data T1 won't be on for another week...Tends to make it tough to search for resolutions because I cant be near the asterisk box when Im reading... |
| 03:01.58 | pitbossy | I also have FXO card (Sangoma)...i ran fxotune, and then check the line as a wiki indicated....I am getting .11 for echo...is that acceptable? |
| 03:02.39 | [TK]D-Fender | pitbossy, if the PRI is active, setup your dialplan to answer a DID and play one of the VM recording like 5 times or so in a loop to test and I'll tell you if it sounds high |
| 03:03.03 | [TK]D-Fender | pitbossy, I also only buy cards with HWEC. Wouldn't know on that point |
| 03:03.24 | pitbossy | that would be great except that I am at a different location right now. |
| 03:03.39 | pitbossy | there is a recording that will play if the system is called right now though.... |
| 03:03.51 | pitbossy | i cant loop it right now though... |
| 03:03.58 | pitbossy | sucks not being local to the machine... |
| 03:04.01 | [TK]D-Fender | pitbossy, no net access I hear you... |
| 03:04.18 | pitbossy | i did buy my pri card with HWEC. |
| 03:04.19 | [TK]D-Fender | pitbossy, if its a recording you did yourself I can't use it as a baseline. |
| 03:04.27 | `Sean | HWEC? |
| 03:04.30 | pitbossy | the FXO card is really just an emergency route. |
| 03:04.35 | `Sean | ~HWEC |
| 03:04.38 | pitbossy | Hardware Echo cancel |
| 03:04.40 | [TK]D-Fender | HardWare Echo Cancellation |
| 03:04.43 | `Sean | oh ok |
| 03:04.54 | pitbossy | I did do the recording my self |
| 03:05.12 | [TK]D-Fender | pitbossy, Ok, then that test won't do much. Lack of direct access blows. |
| 03:05.22 | pitbossy | absolutely! |
| 03:06.01 | pitbossy | Hopefully I am not a fool....It is a car dealership that will be running behind the box....and naturally I have a lot of nay sayers |
| 03:06.33 | *** join/#asterisk FireMac (n=firemac@CPE000d88ae88b9-CM0011ae8bb0ee.cpe.net.cable.rogers.com) |
| 03:06.41 | pitbossy | i got about 12 days to get the kinks out! |
| 03:06.45 | Mercestes | I've noticed that about car dealerships and their phones. |
| 03:07.03 | hesco | after a day of interruptions, I finally got around to installing my kernel headers and ztdummy. Its now loaded into the kernel, but I'm still seeing periodically: "NOTICE[16486]: res_musiconhold.c:515 monmp3thread: Request to schedule in the past?!?!" WHat have I missed in this? |
| 03:07.26 | pitbossy | Thanks all for the help! |
| 03:07.41 | Mercestes | pitbossy, Um, ....I hate to say it, but, have you considered a consultant? >.> |
| 03:07.52 | pitbossy | yes...late next week |
| 03:07.59 | Mercestes | ok. |
| 03:08.14 | pitbossy | being the geek, I cant help but try to get a jump though... |
| 03:08.16 | Mercestes | Just wanted to point out the obvious flaw there. 12 days + new at asterisk. |
| 03:08.51 | [TK]D-Fender | Mercestes, If you've heard what he's tested so far and his general awareness of * bits hes a damn fast learner. |
| 03:10.23 | pitbossy | well thank you...I have a little bit of a leg up though...I am a comp sci eng...and the web has a ton of info, if you just dig + all the nice folks such as yourself! |
| 03:10.38 | Qwell | pitbossy: Mercestes isn't nice. |
| 03:10.39 | Qwell | :p |
| 03:10.49 | Mercestes | I'm really not. |
| 03:10.53 | pitbossy | lol... |
| 03:10.57 | Mercestes | I'm a jerkface.. |
| 03:11.15 | Mercestes | Sometimes I slip up and accidentally help someone tho. |
| 03:11.53 | pitbossy | Well you guys all certainly are helping me! ty |
| 03:12.15 | Qwell | Hey, what do you guys think of this UMA/GAN stuff? |
| 03:12.22 | [TK]D-Fender | pitbossy, Ok, first thing to do is have access to a basic analog line (one of those backups) and pulg a normal phone to test from. Call some regular outside places for a test. Then dial into the PRI and answer looping a decent premade * recording or few a few times with a pause so you can get a sense of the overall gain. Start tweaking from there. |
| 03:12.51 | pitbossy | cool...Will try that tomorrow... |
| 03:12.56 | [TK]D-Fender | pitbossy, that is if the gain is constant to BOTH phone models you have already at around 1/2 |
| 03:13.46 | Mercestes | Qwell, well, my first thought was, "what's uma/gan" stuff. |
| 03:13.48 | [TK]D-Fender | pitbossy, and try to get your hands on the GXP firmwares (multiple revisions if you can) and prepare to field upgrade them |
| 03:14.05 | Mercestes | Qwell, hook me up with a job. =/ |
| 03:14.16 | Qwell | Mercestes: we have a job listing on digium.com |
| 03:14.20 | Qwell | or...we did |
| 03:14.27 | pitbossy | Ok. |
| 03:14.39 | Mercestes | Qwell, Yea, I put in my resume at digium.com but I made the mistake of admitting what my IRC nick is, so they deleted it. |
| 03:14.48 | Qwell | or, I thought we did, heh |
| 03:15.47 | pitbossy | I think Grandstream has released one newer version of firmware than I am running now...After reading all the nightmares about grandstream, I haven't upgraded to the newest one. I've read that many of the firmware upgrades fix 1 problem and screw up 3 more.. |
| 03:16.06 | Qwell | pitbossy: it's not really even the firmware that's the problem... |
| 03:16.19 | pitbossy | The mic gain on the GXP-2000 seems really, really high. |
| 03:16.41 | pitbossy | Qwell: is it the name Grandstream? That is what I seem to be reading! |
| 03:16.44 | Mercestes | It's kinda like, if Walmart made a SIp phone it'd be alot like hte grandstream. |
| 03:16.50 | pitbossy | lol |
| 03:16.50 | [TK]D-Fender | pitbossy, firmwares have evened that stuff out off & on. |
| 03:16.51 | Qwell | heh |
| 03:17.40 | hesco | I've installed kernel headers and ztdummy. lsmod says its there, but I'm still seeing periodically: "NOTICE[16486]: res_musiconhold.c:515 monmp3thread: Request to schedule in the past?!?!" What have I missed in this? |
| 03:18.01 | pitbossy | Well, I need to buy about 25 handsets. I got two GXP-2000 just to prove to the skeptics that * could work for us. I also got the aastra 480ct to try something different....It sounds like polycom is the way to go. |
| 03:19.27 | *** join/#asterisk heison (n=heison@CPE000a01d49e6f-CM00111a59bec2.cpe.net.cable.rogers.com) |
| 03:19.30 | Mercestes | hesco: DId your timestamps/date get messed up anywhere in the process? |
| 03:19.49 | coppice | all IP phones are overpriced crap, some just more than others. compare them to the price and quality of an entry level cellphone and they look pretty sick |
| 03:20.52 | pitbossy | Well I think cellphones don't exactly fit the term "emerging technology". The sell a few more of them than IP phones...Kind of unfair to compare their price points at this time... |
| 03:21.07 | hesco | Mercestes: how would I know? date seems to return something more or less correct, but three hours off. |
| 03:21.22 | Mercestes | hesco: Therein lies your problem. |
| 03:21.39 | Mercestes | pitbossy, coppice is the more optimistic of our group. |
| 03:21.48 | coppice | what is emerging about an IP phone? its all as mature as a cellphone, expect for a few minor bits of the software |
| 03:22.02 | pitbossy | lol |
| 03:22.48 | Mercestes | I've never had an IP phone randomly drop calls, the call waiting on my IP phone works, my IP phone doesn't hang out on a cal for an extra 10 minutes after I hang up randomly to bill me more minutes... |
| 03:23.05 | hesco | Mercestes: I just updated the system date with the date command. |
| 03:23.19 | hesco | I'll watch for the error again. |
| 03:23.21 | Mercestes | The only thing my cellphone has over my IP phone is my cell phone as a camera, a MP3 player, an email client, a keyboard, a swiss army knife, and it vibrates in my pocket. |
| 03:23.49 | [TK]D-Fender | Mercestes, yeah, like THAT'S where you keep it ;) |
| 03:23.54 | coppice | grandstream makes more phones than some of the small GSM players |
| 03:24.04 | Mercestes | [TK]D-Fender, >.> |
| 03:24.18 | [TK]D-Fender | Mercestes, <.< |
| 03:24.52 | pitbossy | As I said earlier, I'm in the auto industry...I was just looking up some stats from a recent class...In the US there are 219 million cellphones active...72% of the population has one...8.4% dont even have a home telephone.....102 Billion Text messages were sent in the first half of 2007... |
| 03:25.11 | Mercestes | that rumor isn't true...I just....like to talk on my phone in the bathroom, that's all. |
| 03:25.39 | Mercestes | pitbossy, don't listen to him. He doesn't even use asterisk. |
| 03:25.47 | pitbossy | lol...even better! |
| 03:26.08 | Mercestes | but, that being said, there are perils to using VoIP. |
| 03:26.11 | coppice | the small GSM players can be in the business because people like mediatek supply silicon packaged with a full complement of software. most IP phone people work the same way. grandstream is an exception. they develop everything from scratch |
| 03:26.11 | [TK]D-Fender | pitbossy, 43.7% of all statistics are made up on the spot..... |
| 03:26.14 | hesco | More with these errors: "Nov 9 22:24:28 NOTICE[16486]: res_musiconhold.c:515 monmp3thread: Request to schedule in the past?!?!" |
| 03:26.18 | Mercestes | But since I recall you saying "PRI" earlier, you are on pretty safe ground. |
| 03:27.07 | Mercestes | coppice, are you pro grandstream? |
| 03:27.16 | pitbossy | I agree 100% about stats...you can twist them however you want...The instructor may have made up the #s, but they were fed directly from Chrysler Corporation research....(Even more likely they are made up!) |
| 03:27.18 | Mercestes | that sounded almost pro-grandstream. |
| 03:27.19 | *** join/#asterisk uski (n=uski@wap.ST.HMC.Edu) |
| 03:27.48 | *** join/#asterisk kiscokid (n=Ron_Laut@adsl-216-101-109-187.dsl.snfc21.pacbell.net) |
| 03:27.49 | coppice | I think people give grandstream a hard time, like they are the runt in the pack, when the whole industry is hardly any better |
| 03:28.35 | Qwell | hey coppice, mind a PM? |
| 03:29.23 | coppice | I will say that when I debug a protocol problem with most VoIP stuff the other guy is at fault. when I debug with a grandstream it usually turns out to be me at fault |
| 03:29.33 | coppice | Qwell: go ahead |
| 03:29.52 | Mercestes | but...Grandstream really is the runt of the pack though.... |
| 03:30.39 | pitbossy | I would think at their cheap prices, some people (myself), would say...hell let me try....And then my Daddy reminds me....If it sounds to good to be true.....well you know the rest. |
| 03:31.05 | *** join/#asterisk kiscokid (n=Ron_Laut@adsl-216-101-109-187.dsl.snfc21.pacbell.net) |
| 03:31.50 | Qwell | ~cheap |
| 03:31.51 | jbot | well, cheap is a bad idea. If you're setting up an Asterisk system, one of the wisest pieces of advice you can take is this: DON'T BE A CHEAPSKATE. |
| 03:32.07 | Qwell | for like $30 more, you can get a polycom |
| 03:32.24 | pitbossy | Yeah...I'm figuring that out quickly... |
| 03:32.29 | hesco | that might could be generalized to address alll telephony applications. |
| 03:32.47 | hesco | I had a similar experinece with hylafax a couple of years ago. |
| 03:32.58 | kiscokid | How do you debug a situation where the zap commands don't appear in the asterisk console? |
| 03:33.24 | hesco | are they getting to the server? How? |
| 03:35.46 | dijungal | hi |
| 03:35.49 | Mercestes | hesco: Hylafax is pretty ok. Worked good for fax to email and email to fax applications. |
| 03:36.11 | dijungal | i am using mixmonitor to record a call and it's ending the monitoring before the call starts |
| 03:36.40 | dijungal | it's ending the recording before the call connects to the agent |
| 03:37.17 | dijungal | any idea why? |
| 03:38.13 | kiscokid | I think I've heard of this symptom before but I don't remember what the solution is |
| 03:38.32 | dijungal | when i use monitor it works |
| 03:39.12 | dijungal | i wonder if it has to do with the fact that i'm using group command to restrict one call at a time to the phone |
| 03:40.00 | coppice | don't mention fax to email. J2 might be listening :-) |
| 03:40.33 | kiscokid | what happens if J2 hears it? |
| 03:40.39 | Qwell | they sue you |
| 03:41.01 | pitbossy | Polycom IP 330 decent ?? |
| 03:41.03 | kiscokid | they? |
| 03:41.06 | coppice | They're gonna sue sue, |
| 03:41.08 | coppice | That's what they're gonna do |
| 03:41.16 | Qwell | J2 |
| 03:41.23 | Qwell | pitbossy: very |
| 03:41.32 | pitbossy | ty |
| 03:41.35 | Qwell | or 320 |
| 03:41.57 | kiscokid | does J2 have a company that has something to do with faxing? |
| 03:42.18 | coppice | they do |
| 03:42.37 | Qwell | efax |
| 03:42.42 | kiscokid | oh |
| 03:42.48 | coppice | e for evil |
| 03:42.59 | Qwell | Brand names marketed by j2 include eFax, jConnect, JFAX, eFax Corporate, UniFax, Onebox, Electric Mail, jBlast, eFax Broadcast, eVoice, PaperMaster, Consensus, M4 Internet, and Protofax. |
| 03:43.00 | kiscokid | they're too expensive |
| 03:43.02 | coppice | they are still patent trolls, though |
| 03:43.03 | Qwell | yeah, they do a little bit of fax |
| 03:43.42 | coppice | the acquired all the failed unified messaging companies, but I think they are still small |
| 03:44.05 | pitbossy | Here comes a subjective question.....What do you guys recommend for an operator SIP Phone about 30-40 extensions? |
| 03:44.21 | Qwell | pitbossy: polycom with the expansion modules |
| 03:44.34 | coppice | they have some really bogus "that's absolutely bloody obvious" patents which their team of lawyers do the Godfather act with |
| 03:44.55 | Qwell | coppice: not to mention the insane amount of prior art |
| 03:45.05 | pitbossy | lol...Qwell....Something earlier made me think you might work for Digium....But perhaps you are moonlighting for Polycom ?!?! |
| 03:45.09 | pitbossy | Sales? |
| 03:45.11 | pitbossy | lol |
| 03:45.21 | Qwell | pitbossy: no, they really are just that good |
| 03:45.36 | coppice | they have shaken down a few users of spandsp |
| 03:45.37 | Qwell | this is nothing though - you should see [TK]D-Fender evangelize. |
| 03:45.52 | pitbossy | I couldn't pass on the jab...sorry! |
| 03:46.00 | Qwell | pitbossy: I do work for Digium though :p |
| 03:46.30 | pitbossy | Well, my PRI went hot today, but the whole concept of * kicks ass. |
| 03:46.37 | [TK]D-Fender | And people swear I work for Polycom ;) |
| 03:46.44 | Qwell | [TK]D-Fender: no, but you should |
| 03:46.48 | [TK]D-Fender | Qwell, Indeed.... |
| 03:47.06 | coppice | polycom really sucks in asia |
| 03:47.07 | pitbossy | I love it when the big phone vendors call me and ask about coming in to pitch their nortels etc, and I mention *...They just shut up and get off the line as quick as they can |
| 03:47.28 | pitbossy | Their pricing models just dont compete... |
| 03:47.36 | pitbossy | Nor their features... |
| 03:47.42 | pitbossy | Nor their....well you get the idea! |
| 03:47.44 | [TK]D-Fender | pitbossy, Usually they attack the stability & support factors... |
| 03:48.13 | pitbossy | There are more people on IRC, blogs, wiki helping then they have in the support center.... |
| 03:48.15 | [TK]D-Fender | pitbossy, And for features, yeah they can compete, just not on the $. But managers tend to want accoutnability & stability and don't care so much about $ |
| 03:48.19 | kiscokid | how much does it cost to buy and install a 50 phone commercial pbx these days? |
| 03:48.25 | Qwell | kiscokid: far too much |
| 03:48.35 | Qwell | anywhere from $20k to $250k |
| 03:48.44 | pitbossy | hang on...PACBELL just quoted me on a NORTEL BCM...Lemme look it up. |
| 03:49.06 | Strom_M | "pac bell" hasn't existed for six years now |
| 03:49.19 | pitbossy | you get the idea...whoever they are today.... |
| 03:49.30 | pitbossy | SBC ...ATT...Cingular...Pacbell..MaBell |
| 03:49.33 | kiscokid | well pac bell sounds better than sbc or att |
| 03:50.03 | pitbossy | BCM 400...Bare Bones...40 Extensions with handsets...$27,873.15 |
| 03:50.18 | pitbossy | Thats installed. |
| 03:50.22 | kiscokid | voicemail included? |
| 03:50.29 | pitbossy | 5 seats |
| 03:50.34 | pitbossy | 5 seats of Voicemail |
| 03:51.20 | Qwell | and fifteen cents? |
| 03:51.28 | kiscokid | interesting |
| 03:51.31 | pitbossy | dont cut the 15 cents off...they wont make the deal |
| 03:51.59 | pitbossy | get this...the installation charge is $13,019.61 |
| 03:52.13 | *** join/#asterisk bmg505 (n=leon@196.209.183.44) |
| 03:52.14 | Qwell | so, $41k? |
| 03:52.21 | pitbossy | no...$27 installed. |
| 03:52.24 | Qwell | oh |
| 03:52.25 | kiscokid | wonder what the 61 cents is for |
| 03:52.31 | pitbossy | overtime |
| 03:52.38 | Qwell | I want to know what the $19 is for |
| 03:52.56 | *** join/#asterisk angom (n=angom@200.38.31.239.dsl.dyn.telnor.net) |
| 03:52.58 | kiscokid | the manual? |
| 03:53.24 | pitbossy | Well...The equipment came out to $24,293.80 but they were kind enough to offer up a $10,268.10 discount... for a net of 14,664.70 |
| 03:53.25 | Qwell | no, that's $1k |
| 03:53.30 | dmz | howdy, anyone know why function SHA1 wouldn't be in the debian 1.2 version? |
| 03:53.43 | coppice | kiscokid: basic statistics. the less certain you are, the more digits you use to try to hide that |
| 03:54.04 | pitbossy | SHA1 -- Perhaps export restrictions? |
| 03:54.13 | dmz | ahh license then |
| 03:54.24 | [TK]D-Fender | pitbossy, 39 x IP 330 =(39 x $85 = $3315), 1 x IP 650 & 3 Modules = ($276 + 187$x3 = $837). Sangoma A101d PRI = $900. Server = $2000 tops. 2 x 24 port PoE Switches (D-Link, Linksys, etc @400 ea = $800 Total $7854 |
| 03:54.26 | dmz | i wonder how i can get around having to recompile everything |
| 03:54.57 | pitbossy | Fender: I agree....Thats why I am taking the plunge...(Heaven save me!) |
| 03:55.05 | pitbossy | Not to mention the support costs.. |
| 03:55.16 | pitbossy | upgrades / maint etc.. |
| 03:55.26 | coppice | pitbossy: when people say the money is in service and not hardware, that $13,019.61 is the kind of thing they are talking about |
| 03:56.01 | pitbossy | i agree. That just starts the rock rolling down the hill...Everytime you need a change, no problem they will reconfig...for a charge... |
| 03:57.11 | pitbossy | was just looking at the quote...$222.64 per month maint. svcs after the 1st year |
| 03:57.25 | Qwell | per month? that's it? |
| 03:57.44 | pitbossy | But wait Qwell:...you also get this paring knife... |
| 03:58.05 | [TK]D-Fender | pitbossy, so they can bleed the last from you... |
| 03:58.15 | pitbossy | something like that... |
| 03:58.40 | coppice | they will *not* give you a paring knife, or any other tool you might use on them when they piss you off |
| 03:59.46 | pitbossy | They have done enough of that.....I ordered a PRI and a T1 data for our new location in August.....The PRI went hot today...the T1 data goes up next week....Unreal...I had to delay the opening of our business because of them...Remarkably they got the PRI in before the Data side.... |
| 03:59.58 | [TK]D-Fender | coppice, thats why I keep my old Bell Tactical Defense Rotary Phone handy during all "negociations" :D |
| 04:00.34 | kiscokid | pittbossy: who did you order that from? |
| 04:00.41 | pitbossy | ATT |
| 04:00.58 | pitbossy | They are being pretty fair to me I think as far as pricing goes... |
| 04:01.05 | nestAr | pitbossy: love telco's... it took mine over a month to reconfigure an existing circuit |
| 04:01.22 | pitbossy | PRI is $430 month. Full T1 is $343...Plus all the BS taxes of course |
| 04:01.33 | pitbossy | nine months!? Ouch... |
| 04:01.36 | nestAr | during this time, they implied that they were waiting for a new circuit to be installed by the ilec |
| 04:01.42 | kiscokid | They quoted me 28 days to get a T1 PRI |
| 04:02.01 | kiscokid | maybe it depends on location |
| 04:02.58 | pitbossy | Get this....When I went Nuclear about their install dates, they got a satellite Data provider to call me....They suggested that I install a satellite Data path in the interim so I could get opened...It would only be a measley $299 for the equip, $80 / month svc, 1 year commit.... |
| 04:03.22 | pitbossy | and I would have a backup route just in case the T went down. |
| 04:03.42 | Qwell | "At that price, why am I using you again?" |
| 04:03.59 | *** join/#asterisk stkn_ (n=stkn@gentoo/developer/pdpc.active.stkn) |
| 04:04.00 | kiscokid | is this way out in the woods? |
| 04:04.20 | hesco | Am I getting this right? A DID can somehow be redirected back to my IP address, permitting the PSTN to address the server? |
| 04:04.33 | hesco | Am I barking up the right tree? |
| 04:04.44 | pitbossy | well, the Satellite pulled off nearly 1.5 down, but the upstream rate was horrible, and dynamic IP's only... |
| 04:05.02 | *** join/#asterisk ekimus_ (n=mm@xover.htu.tuwien.ac.at) |
| 04:05.16 | Mercestes | hesco: only via a gateway, which ITSPs offer. So a DID can be redirected to an ITSP which can gateway from PSTN to IP for you. |
| 04:05.18 | Mercestes | ~itsp |
| 04:05.19 | jbot | [itsp] an Internet Telephony Service Provider, or a "VoIP Phone Company". |
| 04:05.23 | Mercestes | ~itsps |
| 04:05.38 | Mercestes | Aw...he used to list them all out and then make fun of them. |
| 04:05.50 | kiscokid | Voicepulse |
| 04:05.57 | kiscokid | Voipjet |
| 04:05.58 | Mercestes | Teliax. |
| 04:06.00 | kiscokid | etc |
| 04:06.34 | pitbossy | brb |
| 04:06.58 | hesco | Is diamnondcard a gateway? a ITSP? |
| 04:07.24 | dijungal | why would mixmonitor stop recording a call right before it bridges |
| 04:07.24 | *** join/#asterisk RypPn (i=TuMbL@rosscom.demon.co.uk) |
| 04:07.27 | *** join/#asterisk CunningPike (n=CunningP@S010600095b33697f.vc.shawcable.net) |
| 04:07.58 | dijungal | so i execute mixmonitor right before the dial command but it does not record the call |
| 04:09.39 | kiscokid | hesco: diamondcard looks like an itsp |
| 04:14.52 | [TK]D-Fender | dijungal, Never ask without already having your configs and CLI output in a pastebin waiting... |
| 04:15.36 | dijungal | awww man... hold |
| 04:15.54 | hesco | Will my server reach out identify itself to the ITSP? Or do I need somehow to configure things at the ITSP to find my server at my IP? |
| 04:16.22 | kiscokid | hesco: you register with the itsp |
| 04:16.40 | kiscokid | in your iax.conf or sip.conf |
| 04:17.06 | dijungal | http://pastebin.com/d5d2093bf |
| 04:17.14 | dijungal | there u go buddy.. |
| 04:19.54 | dijungal | as you can see the mixmonitor command is called before the the dial command but yet it ends the recordings before the call is bridged - http://pastebin.com/d5d2093bf |
| 04:20.10 | dijungal | any idea why? |
| 04:21.59 | [TK]D-Fender | dijungal, Sure looks like its recording it... |
| 04:22.54 | [TK]D-Fender | dijungal, and that pastebin shows it ends the recording when the call is ended. |
| 04:22.55 | dijungal | notice the "Â == End MixMonitor Recording Local/2025@agents-2f78,2" |
| 04:23.23 | dijungal | i ended the call when i noticed it was doing the same thing |
| 04:23.48 | dijungal | but notice it ends the recording, but it goes on to play queue-less-than and so forth |
| 04:23.53 | dijungal | telling the agent the hold time |
| 04:24.27 | dijungal | point is.. the call would continue had i not hungup, but the monitoring had ended long before that |
| 04:25.28 | dijungal | also notice... it ends the monitoring but the caller has not even reached the agent, "Stopped music on hold Zap/25-1" comes after the end recording |
| 04:26.07 | [TK]D-Fender | dijungal, No. Look : 26 Dial. 28 start recording. 24 ANSWERED. 39. Call terminated. 40. Recording terminated |
| 04:27.44 | [TK]D-Fender | dijungal, I think I see a flaw... |
| 04:27.47 | dijungal | ok then why are call my calls the same lenght and empty? |
| 04:28.12 | [TK]D-Fender | dijungal, |
| 04:28.12 | [TK]D-Fender | exten => _2xxx,n,Dial(SIP/${EXTEN},10)); <- you should not be putting a TIME LIMIT on your dial. Perhaps its being answered on the threshhold.... |
| 04:28.40 | dijungal | timeout??? |
| 04:28.44 | [TK]D-Fender | dijungal, Your queue agent dial timout should deal with that... |
| 04:28.49 | [TK]D-Fender | dijungal, the **10** |
| 04:28.57 | dijungal | hold let me take it out and try |
| 04:29.07 | *** join/#asterisk blq (n=Bl@dslb-088-064-131-012.pools.arcor-ip.net) |
| 04:29.07 | [TK]D-Fender | dijungal, You should not be doing a limit on your dial. The QUEUE will terminate the overall channel if needed. |
| 04:29.39 | dijungal | k |
| 04:31.14 | dijungal | queue timeout is set on the queue cmd? |
| 04:31.18 | *** join/#asterisk luke-jr (n=luke-jr@2002:440d:6de2:0:20e:a6ff:fec4:4e5d) |
| 04:31.28 | [TK]D-Fender | dijungal, no, in queues.conf. |
| 04:31.32 | luke-jr | I heard GrandCentral allows to call arbitrary SIP addresses… anyone know how? |
| 04:35.24 | dijungal | [TK]D-Fender: same problem |
| 04:35.38 | dijungal | even without the dial timeout |
| 04:37.10 | dijungal | oooh well i gotta head out.. i'll have to continue, testing this another time |
| 04:37.16 | dijungal | [TK]D-Fender: any more ideas |
| 04:37.20 | dijungal | ? |
| 04:37.20 | killfill_ | if in zapata.conf i have group=1, then calling throught g1 should be ok, no?.. |
| 04:37.40 | killfill_ | D[D[D[D[D[D(Zap/g1) |
| 04:38.18 | killfill_ | why would i get Channel 0/1, span 1 got hangup request, cause 1?.. |
| 04:38.29 | kiscokid | anyone know why the zap commands suddenly don't work in the CLI? |
| 04:39.03 | killfill_ | kiscokid: becouse you dont have zaptel drivers or card configured before runnign asterisk |
| 04:39.27 | [TK]D-Fender | dijungal, not ATM |
| 04:40.42 | [TK]D-Fender | kiscokid, because Zaptel isn't initialized maybe. or chan_zap just failed to load |
| 04:42.22 | kiscokid | how can I tell if zaptel isn't initialized? |
| 04:43.22 | *** join/#asterisk ghento (n=ghento@75.155.241.7) |
| 04:45.19 | [TK]D-Fender | kiscokid, try loading the module |
| 04:45.43 | killfill_ | ztcfg -v'it.. |
| 04:48.06 | kiscokid | as in modprobe zaptel? |
| 04:49.34 | kiscokid | ztcfg -vvvvv doesn't produce anything worrying |
| 04:51.01 | [TK]D-Fender | kiscokid, restart * then |
| 04:51.55 | *** join/#asterisk emist (n=emist@unaffiliated/emist) |
| 04:52.37 | kiscokid | still no joy |
| 04:54.04 | kiscokid | This happened after I replaced my Sangoma A200 (analog) card with a A101d (T1/E1) |
| 04:54.27 | kiscokid | did a wancfg_zaptel |
| 04:54.53 | killfill_ | oh.. im configured that exactly card too.. :P |
| 04:55.14 | kiscokid | A101d? |
| 04:55.19 | killfill_ | yup |
| 04:55.54 | emist | hey guys, are there any known issues with asterisk dying on the demo- playback? |
| 04:57.58 | [TK]D-Fender | kiscokid, Would be nice to see all the configs and "wanrouter status" , "ztcfg -vvvv" you know.... |
| 04:58.55 | kiscokid | ok |
| 04:59.33 | *** join/#asterisk stkn__ (n=stkn@gentoo/developer/pdpc.active.stkn) |
| 05:03.50 | pitbossy | hey guys.....I'm back, but got to go! Thanks for all the helpful info and comments! C ya soon! |
| 05:06.12 | kiscokid | http://pastebin.com/d7698d09a |
| 05:08.27 | [TK]D-Fender | wanpipe1 | AFT HDLC | N/A | Connecting | |
| 05:08.32 | [TK]D-Fender | not CONNECTED |
| 05:08.36 | [TK]D-Fender | not good. |
| 05:08.41 | killfill_ | whats a nice code for use for wifi phones?.. |
| 05:08.59 | kiscokid | yeah, its not plugged into the T1 yet |
| 05:09.24 | [TK]D-Fender | killfill_, code? |
| 05:09.37 | killfill_ | codec.. sorry.. :P |
| 05:09.49 | [TK]D-Fender | killfill_, G.711 |
| 05:09.59 | kiscokid | zaptel won't come up without plugging it into the T1? |
| 05:10.26 | [TK]D-Fender | kiscokid, What happened when you tried to reload zaptel at * CLI? |
| 05:11.05 | killfill_ | G.711 is one of the *-law? |
| 05:11.35 | kiscokid | Fender: can you tell me how to do that? |
| 05:12.22 | kiscokid | reload chan_zap.so ? |
| 05:16.13 | Maliuta | killfill_: both |
| 05:16.54 | kiscokid | <PROTECTED> |
| 05:16.54 | kiscokid | <PROTECTED> |
| 05:16.54 | kiscokid | [Nov 9 22:09:22] WARNING[5777]: chan_zap.c:11090 process_zap: Ignoring switchtype |
| 05:16.54 | kiscokid | [Nov 9 22:09:22] WARNING[5777]: chan_zap.c:11090 process_zap: Ignoring signalling |
| 05:16.54 | kiscokid | [Nov 9 22:09:22] ERROR[5777]: chan_zap.c:10442 build_channels: Unable to reconfigure channel '1-23' |
| 05:16.55 | kiscokid | [Nov 9 22:09:22] WARNING[5777]: chan_zap.c:11406 reload: Reload of chan_zap.so is unsuccessful! |
| 05:16.57 | killfill_ | with thouse, the sound is choppy.. we got some nokias n95/e65 |
| 05:16.59 | Maliuta | G711 is good, depend on how much bandwidth you have |
| 05:17.12 | Maliuta | and if the phones do something like G729 |
| 05:17.15 | killfill_ | maybe the bottle is on the router/net.. |
| 05:17.57 | Maliuta | if it's wifi then you could be getting radio interference from any number of other devices |
| 05:18.15 | hesco | Record() generates a .wav file, but Playback()'s folder seems filled with .gsm files. Is there a command line tool for converting one to another? |
| 05:18.41 | Maliuta | it's almost cheaper and better quality to get standard DECT sets in the 1.9Ghz or 5.8Ghz ranges and an ATA |
| 05:18.55 | kiscokid | playback will play a wav file |
| 05:18.55 | Maliuta | hesco: sox |
| 05:19.05 | hesco | Mailuta: thanks. |
| 05:19.40 | killfill_ | jitter could help? |
| 05:20.29 | Maliuta | jitter is caused when packets arrive out of order |
| 05:20.41 | Maliuta | or with latency delays |
| 05:20.43 | kiscokid | maybe I need a newer version of wanpipe |
| 05:20.53 | Maliuta | how could jitter actually help? |
| 05:22.25 | killfill_ | well.. 802.11 can alwais have some latency... |
| 05:22.26 | *** join/#asterisk [phl4k-x] (n=saludos@190.40.15.83) |
| 05:22.45 | hesco | Mailuta: wow! sox looks sweet, from a quick scan of the man page. Like a convert command for audio. Thanks again. |
| 05:23.10 | Maliuta | is the sound loss between phones that are on the local network? or only on things being routed out over the net? |
| 05:23.52 | Maliuta | even if you can only QoS one side of the link (i.e. packets leaving your network) it may still be worth doing |
| 05:24.17 | killfill_ | oh yah.. but the sound loss is on the internal net.. :) |
| 05:24.52 | killfill_ | some guys does some nice traffic frecuently.. |
| 05:25.07 | Maliuta | have you done a wireless survey to see who else may be interfering with your transmission? |
| 05:25.15 | killfill_ | maybe i should test it with an isolated wifi... |
| 05:25.18 | Maliuta | sounds like your WAP can't handle it |
| 05:25.20 | [phl4k-x] | hi |
| 05:25.20 | [phl4k-x] | <[phl4k-x]> holas |
| 05:25.20 | [phl4k-x] | <[phl4k-x]> hi for all |
| 05:25.20 | [phl4k-x] | <[phl4k-x]> I have a X100P with FXO connect to PSTN |
| 05:25.20 | [phl4k-x] | <[phl4k-x]> also I have another X100P with FXO conecc to to PBXAnalog Panasonic |
| 05:25.22 | [phl4k-x] | <[phl4k-x]> When I recived call from PSTN with 1 X100P I pass it to PBX Analogic panasonic extension |
| 05:25.24 | [phl4k-x] | <[phl4k-x]> but when both Hangup |
| 05:25.26 | [phl4k-x] | <[phl4k-x]> The 2 Channels FXO continues Busy |
| 05:25.28 | [phl4k-x] | <[phl4k-x]> How I can solutionated this problem? |
| 05:25.32 | Maliuta | [phl4k-x]: don't paste in here |
| 05:25.39 | [phl4k-x] | Maliuta sorry |
| 05:25.49 | [phl4k-x] | this is my problem |
| 05:25.53 | [phl4k-x] | Anyone can hel pe? |
| 05:25.58 | [phl4k-x] | help me? |
| 05:26.03 | Maliuta | yes pasting in here is your problem |
| 05:26.19 | [phl4k-x] | Maliuta I have problems with FXO interfaces |
| 05:26.40 | Maliuta | is "solutionated" even a real word? |
| 05:26.56 | [phl4k-x] | Maliuta I speak spanish |
| 05:27.04 | Maliuta | good for you |
| 05:27.04 | [phl4k-x] | my english is not very well |
| 05:27.08 | pepse | that depends on your definition of 'is' |
| 05:27.18 | [phl4k-x] | pepse of is? |
| 05:27.22 | [phl4k-x] | wher? |
| 05:27.25 | [phl4k-x] | where? |
| 05:27.27 | [phl4k-x] | in zapata.conf? |
| 05:28.31 | [phl4k-x] | pepse, in is??? |
| 05:28.32 | Maliuta | pepse: wow, I have never looked a dictionary definition of "is" before ... interesting |
| 05:28.52 | [phl4k-x] | Maliuta please, do you know something about it? |
| 05:29.19 | Maliuta | you don't have enough details to even start looking at it |
| 05:29.38 | [phl4k-x] | Maliuta |
| 05:29.43 | [phl4k-x] | what do you need? |
| 05:29.58 | [phl4k-x] | but |
| 05:30.23 | Maliuta | details on how the fxo lines are configured and how you are bridging them in the dialplan might be a good start |
| 05:30.33 | [phl4k-x] | is posible to configure a X100P to clean the channels when the Calls Hungup??? |
| 05:30.49 | Maliuta | how are you detecting the hangup? |
| 05:30.55 | [phl4k-x] | KS |
| 05:30.59 | [phl4k-x] | with KS |
| 05:31.02 | Maliuta | that will be part of how the lines are configured |
| 05:31.12 | killfill_ | [phl4k-x]: do you happend to found good spanish sounds?... |
| 05:31.19 | killfill_ | (for asterisk...) |
| 05:31.30 | [phl4k-x] | Maliuta |
| 05:32.05 | Maliuta | [phl4k-x]: so pastebin some of your config details if you want us to look at them |
| 05:32.38 | kiscokid | Fender, any ideas? |
| 05:32.46 | [phl4k-x] | When I transfer the call from PSTN Chanel 1 to a Analogic Extension in Channel 2 (PBX Panasonic), when both hungup the phones |
| 05:32.57 | [phl4k-x] | asterisk dont clean the channels |
| 05:33.05 | [phl4k-x] | and the channels continues busy |
| 05:33.27 | [phl4k-x] | I have to clean the channels with command: soft hanguo ZAP/1-1 |
| 05:33.46 | [phl4k-x] | Is posible to asterisk clean the channels automatic??? |
| 05:34.07 | [TK]D-Fender | [phl4k-x], Your PBX is not sending a disconnect indication to your card so it doesn't know the call has ended |
| 05:34.51 | [phl4k-x] | but not only my PBX, also mi PSTN Provider |
| 05:35.02 | [phl4k-x] | my PSTN Provider is Telefonica |
| 05:35.18 | [phl4k-x] | How I can resolution this problem? |
| 05:35.29 | killfill_ | telefonica.. chile?.. |
| 05:35.33 | [phl4k-x] | Peru |
| 05:35.38 | killfill_ | ah.. |
| 05:35.51 | Maliuta | are you sure using ks is the right signalling? |
| 05:35.54 | [phl4k-x] | Is any metod to resolve this problem? |
| 05:35.55 | *** part/#asterisk unixdog (n=unixdog@adsl-69-234-184-228.dsl.irvnca.pacbell.net) |
| 05:36.19 | [phl4k-x] | Maliuta KS, is the best detecting Disconnect indication |
| 05:36.33 | Maliuta | not on all PSTN systems |
| 05:37.16 | [phl4k-x] | Reconfigured channel 1, FXS Kewlstart signalling |
| 05:37.16 | [phl4k-x] | <PROTECTED> |
| 05:37.16 | [phl4k-x] | <PROTECTED> |
| 05:37.17 | [phl4k-x] | see |
| 05:37.31 | kiscokid | well, I'm going to make a frozen pizza and try a new version of the sangoma stuff |
| 05:37.33 | [phl4k-x] | that is when I put >> reload chan_zap.so |
| 05:38.07 | [phl4k-x] | Maliuta If I tried with LoopStar, this can resolve my problem? |
| 05:38.58 | *** part/#asterisk kiscokid (n=Ron_Laut@adsl-216-101-109-187.dsl.snfc21.pacbell.net) |
| 05:41.20 | [phl4k-x] | eyy |
| 05:41.34 | [phl4k-x] | But Its possible to asterisk clean the channels automatic??? |
| 05:42.00 | [phl4k-x] | only I have to tried different configurations???, Its posible??? |
| 05:42.13 | [TK]D-Fender | [phl4k-x], KS will only work if your TELCO used CDS |
| 05:43.10 | [phl4k-x] | [TK]D-Fender, lets, What I have to do? |
| 05:43.22 | [phl4k-x] | to asterisk clean the channels automatic? |
| 05:45.52 | [TK]D-Fender | [phl4k-x], I thought I was very clear. CDS is a service your TELCO has to ooffer you. |
| 05:47.38 | *** join/#asterisk Keltus (i=Keltus@about/cooking/nakedchef/beefstew/Keltus) |
| 05:47.54 | [phl4k-x] | CDS?? |
| 05:48.02 | [phl4k-x] | what is CDS?, give me a URL please |
| 05:48.24 | [phl4k-x] | But the call I recived from my PBX Panasonic |
| 05:48.26 | [phl4k-x] | lets |
| 05:48.47 | [phl4k-x] | the PBX Panasonic needs CDS yes? |
| 05:50.33 | Maliuta | if you have an old iron PBX why are you using asterisk to link to fxo channels? |
| 05:51.35 | [phl4k-x] | Maliuta for use and IVR |
| 05:51.41 | [phl4k-x] | do you know the solution? |
| 05:52.02 | *** join/#asterisk marl (n=marl@89.241.242.164) |
| 05:53.05 | marl | hi there, can anyone tell me if * can be compiled with its modules static, rather than having it opening up over 100 files for the modules when running? |
| 05:57.07 | killfill_ | hey guys. |
| 05:57.17 | killfill_ | is it me that i have CDR bad configured? |
| 05:57.39 | killfill_ | im saving cdr's in pgsql |
| 05:58.02 | killfill_ | and when i call in a queue |
| 05:58.04 | *** join/#asterisk metfan2007 (n=metfan20@189.180.217.155) |
| 05:58.22 | killfill_ | cdr data is only generating, when an agent hangsup or takes the phone. |
| 05:58.29 | metfan2007 | Hi all!!! How can I specify the load of modules in zaptel?? |
| 05:58.44 | killfill_ | i wish to get a cdr data when the agent's phone is ringing |
| 05:58.46 | killfill_ | is this possible? |
| 05:58.58 | killfill_ | or i should look outside cdr |
| 05:59.40 | metfan2007 | sorry for my question.... how can I specify the modules load order in zaptel... I need that my TE card loads before TDM card.. |
| 06:01.20 | killfill_ | Maliuta: any tips welcome.. :) |
| 06:02.05 | Maliuta | I think CDR is written out after a call terminates |
| 06:02.46 | Maliuta | It's kinda like apache logging, only happens after a request has finished being processed |
| 06:03.07 | killfill_ | yah.. thats my problem.. i got the support guys with a program that checks hes call, and get data out of our databases.. but they see the thing when the take the call.. :S |
| 06:04.06 | killfill_ | i guess ill need to execute a program on every call, that saves to another databse... |
| 06:04.45 | killfill_ | but not sure how to do it.. if i just execute a script.. it will open/close connection every time.. i bet performance wont be very good |
| 06:05.28 | *** join/#asterisk [TK]D-Fender (n=Joe@64.235.216.2) |
| 06:05.55 | killfill_ | too bad cdr cannot do it.. :S |
| 06:07.19 | *** join/#asterisk EnigmaCurry (n=user@67.166.72.245) |
| 06:12.44 | *** join/#asterisk PepOSX (n=pepOSX@190.72.153.45) |
| 06:39.01 | *** join/#asterisk implicit (n=implicit@ip68-105-92-210.sd.sd.cox.net) |
| 06:58.47 | *** join/#asterisk emist (n=emist@unaffiliated/emist) |
| 06:59.18 | emist | asterisk is remembering my old users/extensions even though i removed them, reloaded/restarted/restarted the whole box |
| 06:59.26 | emist | im stumped =[ |
| 06:59.44 | Mercestes | are you sure it's asterisk and not....trixbox or something retarded like that? |
| 06:59.59 | emist | yeah, its asterisk from source on a ubuntu box |
| 07:00.04 | Mercestes | I also got fooled once by copies of extensions.conf being in my home directory, and I was editting those insetad of hte ones in /etc/asterisk |
| 07:00.57 | emist | no, i reverted /etc/asterisk/sip.conf to the original sample |
| 07:01.01 | emist | still nothing =| |
| 07:01.56 | emist | hold up...i think i got it |
| 07:03.10 | emist | nah |
| 07:08.23 | emist | i even uninstalled-all it now and still nothing |
| 07:08.36 | *** join/#asterisk s0lid (n=_freq@7.246.50.60.brf03-home.tm.net.my) |
| 07:14.42 | *** join/#asterisk unixdog (n=unixdog@adsl-69-234-184-228.dsl.irvnca.pacbell.net) |
| 07:14.47 | unixdog | http://www.avalue.com.tw/Panel_PC/touch_panel_pc.cfm |
| 07:14.52 | unixdog | this is just sick |
| 07:19.02 | *** join/#asterisk ghento (n=ghento@75.155.241.7) |
| 07:19.30 | *** join/#asterisk emist (n=emist@unaffiliated/emist) |
| 07:19.51 | emist | ... |
| 07:21.52 | kaldemar | emist: check your astetcdir in asterisk.conf, then for #include's in your configuration files. asterisk is reading them from some file, it doesn't just remember them. check also users.conf, extensions.ael etc. |
| 07:22.20 | emist | will do kaldemar, thanks |
| 07:30.23 | emist | kaldemar, it seems asterisk is reading a completely different set of config files, i added a new extension to the extensions file and it doesn't seem to recognize it now |
| 07:30.28 | emist | any ideas as to how that could happen? |
| 07:31.42 | kaldemar | how are you starting asterisk? |
| 07:32.05 | emist | from the terminal, asterisk & |
| 07:33.42 | kaldemar | and your asterisk.conf has /etc/asterisk as astetcdir and you're editing files in that directory? |
| 07:34.20 | emist | yes, if i understand you correctly |
| 07:34.22 | emist | /etc/asterisk/asterisk.conf:astetcdir => /etc/asterisk |
| 07:35.19 | *** join/#asterisk techie (n=techie@adsl-76-214-29-227.dsl.lsan03.sbcglobal.net) |
| 07:38.58 | *** join/#asterisk scardinal (n=supreme@port816.ds1-suoe.adsl.cybercity.dk) |
| 07:40.41 | *** join/#asterisk h3x (i=Justino@64.192.116.17) |
| 07:41.04 | kaldemar | what if you start asterisk with asterisk -C /etc/asterisk/asterisk.conf ? |
| 07:41.37 | emist | same thing ='[ |
| 07:43.15 | kaldemar | would you pastebin your extensions.conf and sip.conf? |
| 07:43.28 | emist | sure, hold up a sec |
| 07:45.41 | emist | kaldemar, sip.conf |
| 07:45.42 | emist | http://pastebin.org/7639 |
| 07:45.45 | emist | and extensions.conf |
| 07:45.49 | emist | http://pastebin.org/7640 |
| 07:46.04 | emist | they're basically the sample ones |
| 07:47.41 | ai-a[afk] | emist: you should.. as a start NOT use the samplese. they are bad idea. instead use the asterisk book and start with an empty /etc/asterisk folder |
| 07:48.18 | emist | ai-a, will do ai-a, just kind of getting around to playing with asterisk for the first time |
| 07:48.19 | ai-a | your new ext is 1342 ? |
| 07:48.26 | ai-a | and context hello? |
| 07:48.30 | emist | yeah 1342 is the one i just added |
| 07:48.34 | emist | which can't be found |
| 07:48.43 | emist | yet 1235 is one i added before |
| 07:48.48 | emist | and that one is still functional |
| 07:48.50 | ai-a | what sip account name ? |
| 07:49.14 | emist | 104 is the account name, which is a working account somehow, even though it doesn't exist anymore |
| 07:49.25 | ai-a | sip.conf doesnt contain 104 |
| 07:49.28 | emist | somehow the previous extensions/sip accounts are still working on this system |
| 07:49.33 | emist | thats the problem ai-a |
| 07:49.39 | ai-a | emist: your must add sip. |
| 07:49.45 | ai-a | emist: your in a muddle. |
| 07:49.56 | ai-a | read the astrisk book, follow the examples in there.. |
| 07:49.59 | emist | maybe im explaining this wrong |
| 07:50.07 | emist | i have working extensions to accounts that don't exist |
| 07:50.12 | emist | and extensions that don't exist |
| 07:50.12 | ai-a | do this cd /etc/asterisk; mkdir samples; mv *.* samples |
| 07:50.25 | ai-a | your doing it wrong. |
| 07:50.35 | ai-a | use the sampleas as references to the conf files. |
| 07:50.41 | emist | ... |
| 07:50.51 | ai-a | use the book to make your sip / extentions / examples / context / zap.. system |
| 07:50.57 | emist | i set up some extensions and accounts that _worked_ fine, i removed them and they _still_ work fine |
| 07:51.00 | ai-a | you will learn 500 times faster, and do it right. |
| 07:51.00 | emist | that is my problem at the moment |
| 07:51.06 | ai-a | okay,, fine... bye |
| 07:51.17 | emist | alright, thanks for not even hearing what the problem is |
| 07:52.04 | ai-a[afk] | i will count the hours,, and i am betting within 3 hours, if people talk with you on here,, you will start with an empty folder. |
| 07:52.18 | ai-a[afk] | i dont have 3 hours to wait,, its saturday 8am here, im free. |
| 07:52.19 | emist | i did already...it doesn't change the configs |
| 07:52.26 | emist | everything still behaves the same |
| 07:52.33 | emist | i even uninstalled-all asterisk |
| 07:52.34 | emist | reinstalled |
| 07:52.36 | emist | etc |
| 07:52.47 | emist | stuff that was working before with a previously configured asterisk is still working |
| 07:52.51 | emist | and nothing new that i add changes anything |
| 07:53.24 | emist | thats the problem im having but you can count the hours im sure its going to be more than 3 |
| 07:55.29 | kaldemar | try removing all the files in /etc/asterisk then stop and start asterisk again, if it starts and works fine, you can be sure that it's not the place where it's reading the configs. |
| 07:57.41 | Mercestes | emist, what does it say in /etc/asterisk.conf? |
| 07:57.55 | Mercestes | before you umm...remove it. |
| 07:58.21 | emist | kaldemar, when i remove all the configs asterisk doesn't really respond after restart << which im guessing is whats supposed to happen |
| 07:58.31 | emist | Mercestes, http://pastebin.org/7639 |
| 07:58.48 | kaldemar | yes, it doesn't load any modules then. |
| 08:00.09 | Mercestes | 1: that doesn't look like asterisk.conf |
| 08:00.14 | *** join/#asterisk hijacked (i=rTZ2@66.255.220.17) |
| 08:00.18 | Mercestes | 2: ....you couldn't have grepped -v ; that first? |
| 08:00.38 | emist | ohh sorry, thats sip.conf |
| 08:00.45 | Mercestes | Your testing me, aren't you? |
| 08:00.53 | emist | asterisk.conf is just the default sample |
| 08:00.53 | Mercestes | You don't think I use asterisk! |
| 08:00.56 | emist | hehe |
| 08:01.00 | Mercestes | lol |
| 08:01.06 | emist | my shoulder is killing me =\ |
| 08:01.09 | Mercestes | Default asterisk.conf points your configdir=/etc/asterisk |
| 08:01.16 | emist | surgery =! good |
| 08:01.34 | kaldemar | Mercestes: you're testing us too, with your /etc/asterisk.conf ;) |
| 08:01.35 | Mercestes | ! surgery != good. |
| 08:01.47 | Mercestes | there's an /etc/asterisk.conf. |
| 08:02.03 | Mercestes | wait... |
| 08:02.05 | Mercestes | is it in extconfig.conf? |
| 08:02.14 | kaldemar | not in vanilla asterisk. |
| 08:02.46 | Mercestes | ? |
| 08:02.52 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 08:03.10 | kaldemar | there is no /etc/asterisk.conf in plain asterisk. |
| 08:03.16 | Mercestes | yea there is..... |
| 08:03.44 | emist | there is a /etc/asterisk/asterisk.conf |
| 08:03.49 | Mercestes | http://www.voip-info.org/wiki/index.php?page=Asterisk+config+asterisk.conf |
| 08:03.50 | emist | if thats what you mean |
| 08:04.04 | Mercestes | oh...ok, yea, /etc/asterisk/asterisk.conf .. |
| 08:04.11 | Mercestes | sorry >.> |
| 08:04.32 | Mercestes | Sorry, I'm working my way towards drunk tonight. |
| 08:04.43 | Mercestes | wife decided I was an evil bastard. |
| 08:04.47 | Mercestes | s/decided/discovered/ |
| 08:05.22 | kaldemar | i decided that my liver is an evil bastard and i have to punish him. but that was yesterday. |
| 08:05.27 | emist | Mercestes, this is all there is in my asterisk.conf |
| 08:05.28 | emist | <PROTECTED> |
| 08:05.28 | emist | <PROTECTED> |
| 08:05.28 | emist | <PROTECTED> |
| 08:05.28 | emist | <PROTECTED> |
| 08:05.29 | emist | <PROTECTED> |
| 08:05.30 | emist | <PROTECTED> |
| 08:05.32 | emist | <PROTECTED> |
| 08:05.34 | emist | <PROTECTED> |
| 08:05.36 | emist | <PROTECTED> |
| 08:05.38 | emist | <PROTECTED> |
| 08:05.40 | emist | <PROTECTED> |
| 08:05.40 | kaldemar | use pastebin. |
| 08:05.42 | emist | <PROTECTED> |
| 08:05.44 | emist | <PROTECTED> |
| 08:05.46 | emist | <PROTECTED> |
| 08:05.48 | emist | <PROTECTED> |
| 08:05.50 | emist | <PROTECTED> |
| 08:05.55 | emist | <PROTECTED> |
| 08:05.56 | emist | <PROTECTED> |
| 08:05.58 | emist | errr |
| 08:06.00 | emist | wrong paste |
| 08:06.02 | emist | http://pastebin.org/7641 |
| 08:06.04 | emist | yeah i had the wrong buffer |
| 08:06.06 | emist | sorry guys |
| 08:06.33 | Mercestes | Your stuff should be in /etc/astersk |
| 08:06.49 | emist | thats where i got them |
| 08:06.49 | Mercestes | instead of deleting it try chmod 000 * on /etc/asterisk and restart asterisk..assuming you have root access |
| 08:06.51 | Mercestes | otherwise, don't... |
| 08:06.55 | emist | alright |
| 08:07.10 | *** join/#asterisk khronos (n=khronos@c-66-229-159-201.hsd1.fl.comcast.net) |
| 08:07.15 | Mercestes | It should massively spasm and die |
| 08:07.39 | khronos | In 1.4.13 how do I run as user asterisk? |
| 08:07.55 | emist | not quite, it works |
| 08:08.05 | khronos | Is there an option in the Makefile that tells what user to run as? |
| 08:10.18 | Mercestes | khronos, You su asterisk |
| 08:10.27 | Mercestes | khronos, google run asterisk non-root |
| 08:12.16 | Mercestes | installing asterisk on sabayon is a PITA |
| 08:19.20 | Mercestes | whats a good win % in solitare? |
| 08:24.53 | Mercestes | anyone? |
| 08:35.32 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 08:46.24 | *** join/#asterisk BeeBuu (n=chatzill@125.95.250.63) |
| 08:46.30 | BeeBuu | hello,all |
| 08:47.20 | BeeBuu | can i use set(var=) to set the var to nothing? |
| 08:49.29 | ai-a[afk] | BeeBuu: can you determin if a variable is set or not set ? |
| 08:50.20 | *** join/#asterisk khronos (n=khronos@c-66-229-159-201.hsd1.fl.comcast.net) |
| 08:50.26 | BeeBuu | i just want to set one variable to nothing,how can i do that? |
| 08:51.00 | ai-a[afk] | why do that ? considering you cant do an if undefined. |
| 08:51.19 | ai-a[afk] | why not just consider "FooBarG" is undefined, and do set(var=FooBarG) |
| 08:51.37 | ai-a[afk] | then you can check if var is FooBarG and know its undefined. |
| 08:51.42 | ai-a[afk] | otherwise i dont see the point. |
| 08:52.01 | ai-a[afk] | on the other hand, a) try it, b) read the manual |
| 08:52.07 | BeeBuu | so how can i know a variable be seted? |
| 08:52.19 | ai-a[afk] | you CANT. |
| 08:52.22 | ai-a[afk] | thats the point im saying. |
| 08:52.23 | Mercestes | Most people do a If[${callerid(num)}"foo"="foo"]? to check for null. |
| 08:52.30 | ai-a[afk] | If trying to zero out the CALLERID(name) do not use empty quotes, use Set(CALLERID(name)=) |
| 08:52.47 | ai-a[afk] | the manual cearly states it quite clearly. |
| 08:52.59 | ai-a[afk] | s/cearly/clearly |
| 08:53.15 | BeeBuu | o,so i can do this set(myvar=) |
| 08:53.22 | ai-a[afk] | BeeBuu: RTFM |
| 08:53.25 | BeeBuu | thanks |
| 08:53.33 | Mercestes | s\/cearly\clearly/\/cearly\/clearly\// |
| 08:54.06 | BeeBuu | RTM |
| 08:54.12 | Mercestes | Yup. |
| 08:54.33 | ai-a[afk] | http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+Set |
| 08:59.14 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 09:10.08 | *** join/#asterisk duki (n=duki@host-85-27-58-159.brutele.be) |
| 09:37.35 | *** join/#asterisk ming_zym (n=ming_zym@124.254.56.192) |
| 09:51.21 | *** join/#asterisk dutchfish (n=wil@wc-35.r-195-35-167.atwork.nl) |
| 09:57.41 | *** join/#asterisk kv0s (n=kv0s@p4FD26598.dip.t-dialin.net) |
| 09:58.12 | kv0s | Hi! |
| 10:00.08 | kv0s | I've some clients with x-lite at my asterisk and plantronics headsets ... but all headsets are to "silent". it is possible to adjust the incoming loudness at asterisk? |
| 10:02.37 | dutchfish | kv0s, cant you just cranck up the volume on your client? |
| 10:03.31 | kv0s | dutchfish: Mhm. X-Lite - yes. But the maximum volume isn't really "loud" ... |
| 10:04.08 | dutchfish | kv0s, is it outgoing volume (microphone) or incoming in respect to client? |
| 10:04.20 | kv0s | The incoming ... |
| 10:05.38 | dutchfish | kv0s, so if speakers of clients headset at max i have no idee what can be done, i believe astrix, has something as autovolume as a setting, did you check that? |
| 10:06.42 | kv0s | That is what i mean ... but in which configfile and which parameters? |
| 10:07.22 | dutchfish | kv0s, i have no idee but i read it somewhere. I am newb in respect to astrix and wrestling to set it up on a debian system |
| 10:07.45 | kv0s | Okies. Thanks. |
| 10:07.56 | kv0s | I'll give google some tries... ;-) |
| 10:08.52 | dutchfish | kv0s, can yo help me out with 1 question (if you can)? |
| 10:08.59 | *** join/#asterisk duki (n=duki@host-85-27-58-159.brutele.be) |
| 10:10.04 | JT | dutchfish: please stop saying astrix |
| 10:10.15 | dutchfish | sorry, astriks |
| 10:10.27 | dutchfish | sorry again oops |
| 10:10.50 | kv0s | dutchfish: don't ask to ask questions ... ask! ;-) |
| 10:11.42 | dutchfish | ok, i having trouble with setting up a multiline sip trunk for astriks, are there any tutors? |
| 10:11.59 | JT | asterisk |
| 10:12.00 | JT | ffs |
| 10:12.49 | kv0s | ASTERISK!!!! |
| 10:12.51 | kv0s | :D |
| 10:13.07 | dutchfish | ok, i got it now, sorry for my dislexia? |
| 10:13.59 | dutchfish | I am having trouble with setting up a multiline sip trunk for astrisk, are there any tutors? |
| 10:15.21 | kv0s | Multiline? What do you mean? Many lines on only one trunk? |
| 10:15.32 | dutchfish | kv0s, yes |
| 10:15.54 | JT | no such thing as a sip trunk ;) |
| 10:17.04 | kv0s | U only need one sip trunk? |
| 10:17.11 | dutchfish | sorry being a total newby, but what about this http://www.siptrunk.org/ ? |
| 10:19.02 | kv0s | Mhm. |
| 10:19.27 | JT | dutchfish: a load of rubbish |
| 10:19.39 | JT | dutchfish: there is NO such thing as sip trunking |
| 10:19.48 | JT | regardless how much some try to insist there is |
| 10:19.51 | kv0s | If u have a sip-provider u need nothing else than your account information. set up your sip-trunk in asterisk and call your friends and customers ... ,-) |
| 10:19.53 | JT | mostly marketing types |
| 10:21.15 | dutchfish | ok, so how do i setup more then 1 simultanous line ( 50 par example) over just 1 IP adress on the other site? |
| 10:21.29 | JT | you use sip. |
| 10:21.40 | JT | the fact if it's one or 100 is irrelevant |
| 10:21.47 | JT | they all are seperate connections anyway |
| 10:21.57 | dutchfish | JT, yes |
| 10:22.03 | kv0s | dutchfish: one sip trunk for all connections. |
| 10:22.15 | JT | kv0s: what are you talking about? |
| 10:22.28 | kv0s | asterisk made for each call via sip a new connection with same sip-provider-account ... |
| 10:22.38 | JT | sure |
| 10:22.44 | JT | but they're seperate connections |
| 10:22.49 | JT | and they're not trunks |
| 10:22.58 | kv0s | Mhm. |
| 10:23.14 | kv0s | Jt: at asterisk it is named sip-trunk or not? |
| 10:23.17 | kv0s | Mhm. |
| 10:23.19 | JT | sip trunk is a misnomer from free-pbx |
| 10:23.20 | JT | MHM |
| 10:23.21 | JT | <HM |
| 10:23.23 | JT | NO |
| 10:23.24 | JT | it is not |
| 10:23.28 | JT | fucking freepbx does |
| 10:23.31 | JT | asterisk does not |
| 10:23.47 | kv0s | jt: oh. okay. i've set up freepbx ... *ieeehhh* |
| 10:23.56 | JT | FREEPBX IS NOT ASTERISK |
| 10:23.58 | JT | okay |
| 10:24.01 | JT | ~freepbx |
| 10:24.02 | jbot | [~freepbx] FreePBX is unable to be supported here. It is made up of complex dialplans and scripts which can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking there |
| 10:24.04 | dutchfish | ok, i grasp the fact that the established connection goes an exclsuive path, and that i only need 1 endpoint for negatiation, but test for example with voipbuster allow me only 1 connection at a time, is this astriks related or provider related? |
| 10:24.17 | JT | dutchfish: provider |
| 10:24.25 | JT | dutchfish: and it's still called "asterisk" |
| 10:25.04 | dutchfish | JT, ok how do i find a good asterisk supported provider that supports more then 1 connection? |
| 10:25.30 | JT | dutchfish: i imagine google to be useful |
| 10:25.56 | dutchfish | JT, i googled my buts off to find one in earope that is affordable |
| 10:26.43 | dutchfish | JT, most of them only want to sell solutions and equipment, wich i do need but still |
| 10:26.45 | JT | dutchfish: i can't advise there |
| 10:27.04 | dutchfish | JT, ok, fair enough |
| 10:27.24 | kv0s | JT: I really now what freepbx is. I thought it is named "sip trunk". Sorry for the wrong naming .. ;-) |
| 10:29.34 | dutchfish | JT, let me rephrase, does astrisk support this what i explained and is there some howto or tutor on this subject? |
| 10:31.34 | JT | dutchfish: multiple sip calls is completely standard and normal |
| 10:31.48 | JT | dutchfish: not allowing multiple is something to do with the provider |
| 10:33.47 | dutchfish | JT, ok, being newb, forgive me my stupid questions, but what about rtsp, does asterisk suport that? |
| 10:34.09 | JT | i don't think so |
| 10:34.31 | dutchfish | JT, i mean this http://en.wikipedia.org/wiki/Real_Time_Streaming_Protocol |
| 10:36.07 | dutchfish | JT, thank you for help so far |
| 10:37.04 | JT | no |
| 10:37.10 | JT | no rtsp |
| 10:37.16 | JT | i don't think much uses rtsp |
| 10:39.20 | dutchfish | JT, i will try anyway, debian is flexable enough to chain it up, so my rtsp services can be used for asterisk too, think about a voice library of translated voice messages etc etc |
| 10:40.46 | JT | i'm not sure what your distro being debian has to do with anything |
| 10:40.53 | JT | asterisk does NOT support rtsp |
| 10:40.59 | JT | i'm not sure what you're hoping for |
| 10:41.00 | dutchfish | JT, this will also pull off the load from the asterisk box |
| 10:41.13 | JT | i have no idea what you're trying to do |
| 10:41.26 | dutchfish | JT, thats why i ask |
| 10:41.43 | JT | what are you talking about with rtsp |
| 10:41.47 | JT | what are you doing? |
| 10:43.29 | dutchfish | JT, i want to use rtsp as a huge sound library and connect to asterisk, the library contains ~1.3 TB sound |
| 10:43.47 | JT | ~wglwat |
| 10:43.48 | jbot | methinks wglwat is well, good luck with all that |
| 10:44.14 | dutchfish | JT, the lib contains spoken instructions for disabled poeple |
| 10:44.20 | JT | uhuh |
| 10:44.42 | JT | what is so magical about this library? is it not just a collection of sound files? |
| 10:44.53 | dutchfish | JT, the lib is already in use for other purposes too |
| 10:45.33 | dutchfish | JT, the transport is done by rtsp |
| 10:45.51 | dutchfish | JT, i kow this is all offtopic |
| 10:45.53 | JT | AND ASTERISK DOES NOT SUPPORT IT |
| 10:45.57 | JT | i don't care about the lib |
| 10:46.05 | dutchfish | JT, ok sorry and thanks |
| 10:46.21 | JT | sip uses RTP |
| 10:46.23 | JT | not rtsp |
| 10:46.26 | JT | same for H.323 |
| 10:46.31 | JT | and IAX2... let's not go there |
| 10:47.09 | dutchfish | back then rtsp was choosen because its almost echo-less |
| 10:47.28 | JT | what the hell |
| 10:47.32 | JT | rtp does not have echo |
| 10:47.41 | dutchfish | ok |
| 10:47.41 | JT | echo is caused by analogue components |
| 10:47.47 | JT | not by the transport layer |
| 10:47.47 | dutchfish | yes |
| 10:47.59 | JT | in ip anyway |
| 10:48.07 | dutchfish | ok |
| 10:48.40 | JT | echo only happens at analogue sections, like analogue phone lines and handsets/speakers/mics etc |
| 11:17.23 | *** part/#asterisk dutchfish (n=wil@wc-35.r-195-35-167.atwork.nl) |
| 11:31.07 | *** join/#asterisk shinao1 (n=shinao1@217.20.242.50) |
| 11:34.01 | coppice | *unwanted* echo only happens at analogue sections. we can add much loved reverb, for dramatic effect, in the digital sections :-) |
| 11:34.18 | JT | yes indeed ;) |
| 11:34.59 | *** join/#asterisk [hC] (n=hardcore@S01060016b61c8983.vf.shawcable.net) |
| 11:39.16 | *** join/#asterisk dijungal (n=kdaniel@63.175.159.171) |
| 11:55.36 | *** join/#asterisk jm|laptop (n=jm|home@dilbert.jamiem.com) |
| 11:56.55 | *** join/#asterisk dijungal (n=kdaniel@63.175.159.171) |
| 11:58.18 | *** join/#asterisk fordfrog (n=fordfrog@gentoo/developer/fordfrog) |
| 11:58.58 | dijungal | back on the trail again... |
| 11:59.32 | dijungal | why would Mixmonitor stop recording before the call is bridged? http://pastebin.com/d5d2093bf |
| 12:12.29 | tzafrir_home | JT, one of those "analog components" is the handset of a VoIP phone |
| 12:13.10 | coppice | especially if someone turns up the handset volume |
| 12:23.01 | *** join/#asterisk duki (n=duki@host-85-27-58-159.brutele.be) |
| 12:26.19 | *** join/#asterisk shinao1 (n=shinao1@217.20.242.50) |
| 12:27.06 | *** join/#asterisk loompek (n=NoName@noname.rula.net) |
| 12:27.11 | loompek | morning 'yall |
| 12:29.36 | loompek | is it possible for asterisk to call back a user when he dials in? |
| 12:29.37 | loompek | http://rula.net/170 |
| 12:30.42 | loompek | i'd like asterisk to call me back and give me a 'free outgoing line' |
| 12:33.00 | *** join/#asterisk chrizz- (i=chrizz@82.196.215.94) |
| 12:37.12 | Mw3 | does anybody know a replacement for soxmix? its no longer in sox 14.0 |
| 12:37.55 | *** join/#asterisk xheliox (n=jeff@pdpc/supporter/active/xheliox) |
| 12:42.55 | dijungal | why would Mixmonitor stop recording before the call is bridged? http://pastebin.com/d5d2093bf |
| 12:59.34 | *** join/#asterisk agx (n=badpengu@81-174-45-156.dynamic.ngi.it) |
| 13:07.34 | *** join/#asterisk Mw3 (n=mw3@ip59934bd1.rubicom.hu) |
| 13:14.40 | *** join/#asterisk robl^ (n=robl@pdpc/supporter/monthlybyte/robl) |
| 13:26.23 | dijungal | does anyone know why would Mixmonitor stop recording before the call is bridged? http://pastebin.com/d5d2093bf |
| 13:30.29 | *** join/#asterisk curtn (n=curtis@cl-451.trn-01.it.sixxs.net) |
| 13:30.48 | moemoe | does anybody know why dijungal repeats himself every hour? |
| 13:31.14 | dijungal | moemoe: hoping that a smart person would join the room |
| 13:31.46 | *** join/#asterisk riddlebox (n=james@75-132-225-75.dhcp.stls.mo.charter.com) |
| 13:32.05 | curtn | does it exist a rule which says if we have to Answer() on an exten.. or not ? |
| 13:33.28 | curtn | for example : incoming calls need Answer() (but it seems to works without..) |
| 13:34.01 | curtn | before Dial.. we do need Answer() in general ? |
| 13:35.42 | curtn | this Answer () is a problem for me, if I have an analog phone in parallel with my FXO |
| 13:36.21 | *** join/#asterisk linxroute (n=linx@203.190.164.47) |
| 13:36.22 | curtn | (on the same pstn line) |
| 13:37.04 | agx | curtn, you only need Answer() before a Playback() or a Background() or wanna process a DTMF input |
| 13:38.27 | *** join/#asterisk rsd (n=chaos@200.181.133.130) |
| 13:38.47 | curtn | agx: does it mean that if I don't Answer, I can't use the timeout on Dial to "Playback(vm-nobodyavail)" ? |
| 13:39.24 | curtn | agx: some "Answer" seems to be automatic.. I'm a bit confusing.. |
| 13:39.26 | agx | you can Answer() after Dial() fails before the Playback() :-) |
| 13:40.12 | agx | if you have immediate=yes in your audio card config file the channell driver will answer for you before the call hit the dialplan |
| 13:40.16 | curtn | agx: for example... before Record, I don't need Answer ! (but, of course, Asterisk answer..) |
| 13:41.19 | agx | certain apps autoanswer the channel for you if i recall, make sure you have immediate=no in misdn.conf and zapata configs |
| 13:41.55 | *** part/#asterisk rsd (n=chaos@200.181.133.130) |
| 13:42.52 | curtn | agx: I use a SPA3102 |
| 13:44.41 | agx | curtn, uhm never used such fxo gateways |
| 13:46.02 | curtn | agx: maybe it would be easier to control via MGCP instead of SIP.. |
| 13:46.25 | curtn | agx: but SIP gives the gateway an "autonomy"... |
| 13:55.49 | curtn | does it exist a simple application to keep every ${CALLERID} in a file ? |
| 13:59.06 | curtn | (answer or not) |
| 14:00.52 | loompek | umm |
| 14:01.16 | robl^ | CDR does that. not an applicatio, but a feature. read up on cdr_custom. |
| 14:01.52 | loompek | yep... /var/log/asterisk/cdr-csv/Master.csv |
| 14:02.05 | loompek | it seems all the numbers are here |
| 14:02.49 | robl^ | you can also use cdr-custom to define exactly how the file is formatted |
| 14:03.49 | robl^ | check out the file /etc/asterisk/cdr_custom.conf |
| 14:05.57 | curtn | ok i see "${CDR(src)}", thanks |
| 14:06.38 | curtn | and why is ${CDR(clid)} empty ? |
| 14:42.11 | *** join/#asterisk tobias (n=tobias@cpe-069-134-226-227.nc.res.rr.com) |
| 14:44.58 | *** join/#asterisk bantu (n=Miranda@p54A30D8D.dip0.t-ipconnect.de) |
| 14:50.03 | *** join/#asterisk DarkRift (i=dark@bas10-montreal02-1177582580.dsl.bell.ca) |
| 15:00.43 | killfill_ | how do i see what codecs do i have enabled?.. |
| 15:01.35 | killfill_ | (not in the config file) |
| 15:02.02 | agx | killfill_, sip debug of an INVITE message |
| 15:02.33 | agx | uhm, or maybe "show translations" and watch at column with "-" instead of numbers |
| 15:03.20 | killfill_ | hm... |
| 15:03.27 | killfill_ | show translations? |
| 15:03.46 | agx | killfill_, show codec translation, cannot remember the exact syntax |
| 15:03.51 | *** part/#asterisk agx (n=badpengu@81-174-45-156.dynamic.ngi.it) |
| 15:04.25 | killfill_ | ah.. |
| 15:07.47 | killfill_ | what codec should i enable for granstream phones?.. i.e. gxp2000 |
| 15:08.30 | *** join/#asterisk linxroute (n=linx@203.190.164.47) |
| 15:08.54 | linxroute | . |
| 15:09.48 | codec | killfill_: http://www.asteriskguru.com/tutorials/gxp2000_grandstream_hardphone.html |
| 15:10.00 | codec | "Available codecs are: G.711u, G.711A, G.722, G.723, G.726, G.728, G.729 and iLBC" |
| 15:10.13 | killfill_ | wired.. when i call from these phones, out to zap (pstn). they cannot hear what people on the pstn talks.... |
| 15:10.28 | killfill_ | software sip works tho. (eyebeam) |
| 15:10.30 | killfill_ | and iax too |
| 15:12.23 | linxroute | . |
| 15:12.52 | killfill_ | what could it be?.. |
| 15:13.12 | *** join/#asterisk blq (i=Bl@dslb-088-067-043-101.pools.arcor-ip.net) |
| 15:18.25 | killfill_ | damed. |
| 15:24.15 | *** join/#asterisk icewaterman (n=immagine@i53874083.versanet.de) |
| 15:26.02 | icewaterman | hi, i want to use asterisk as a voip -> isdn gateway - meaning i want to create several sip accounts for asterisk (for me to connect to that asterisk) and then have asterisk use a normal isdn line from there on in both directions (so that i can use my voip phone for calling normal telephones and also getting called by normal telephones). |
| 15:26.45 | icewaterman | i have one hfcsusb device, will asterisk be able to do that? |
| 15:28.28 | killfill_ | what could be happening... |
| 15:29.06 | icewaterman | will i need the bristuff? |
| 15:29.11 | icewaterman | i guess so |
| 15:33.58 | killfill_ | http://pastebin.ca/768361 <-- is there something wired in there?.. |
| 15:34.39 | *** join/#asterisk omarc55 (n=omar@c-66-176-110-20.hsd1.fl.comcast.net) |
| 15:35.18 | icewaterman | killfill_: looks weired to me, but i'm new here :) |
| 15:35.28 | killfill_ | heh.. |
| 15:36.02 | killfill_ | why the heck does the sip phone not hear what ppl say in the pstn.. |
| 15:37.46 | *** join/#asterisk [TK]D-Fender (n=Joe@64.235.216.2) |
| 15:39.40 | killfill_ | why could this happend.. im not using nat.. they are on the same net.. |
| 15:39.42 | icewaterman | is there a howto for creating local sip accounts with asterisk? |
| 15:39.50 | *** join/#asterisk enalert (i=trelane@2001:4830:150c:0:20d:61ff:fe31:a58) |
| 15:39.59 | icewaterman | killfill_: there might be a packetfilter involved anyway |
| 15:40.12 | killfill_ | icewaterman: there are plenty... or you could try to use asterisk-gui.. |
| 15:40.17 | killfill_ | nope.. no firewall |
| 15:40.36 | icewaterman | killfill_: asterisk gui? is that a webinterface? |
| 15:41.05 | icewaterman | killfill_: my router does not have X11 installed |
| 15:41.16 | killfill_ | yah. not much people like to use guis in here.. but you could see how it writed configs and stuff |
| 15:41.46 | killfill_ | you dont need X |
| 15:41.51 | *** join/#asterisk dijungal (n=kdaniel@63.175.159.171) |
| 15:41.55 | icewaterman | ah ok |
| 15:43.20 | icewaterman | killfill_: X would also be a huge problem, as the box does not have screen/keyboard or mouse (only ssh and serial) |
| 15:43.53 | tzafrir_home | icewaterman, what's the status of the FreeBSD asterisk port? IIRC they had bristuff support. But I also recall it was aufully dated |
| 15:44.10 | icewaterman | tzafrir_home: how would i know? |
| 15:44.42 | tzafrir_home | you were using FreeBSD, right? Or do I confuse you with someone else? |
| 15:44.47 | killfill_ | i use * with a sangoma card on freebsd.... i have a problem tho. but i think its a configuration thingy |
| 15:44.52 | icewaterman | tzafrir_home: yes you do, i am using debian :) |
| 15:45.11 | tzafrir_home | well, the official debs are bristuffed as well |
| 15:45.21 | icewaterman | tzafrir_home: i always wanted to switch to freebsd but couldn't due to lack of hardware support for my isdn card |
| 15:45.38 | tzafrir_home | but I don't think anybody wrote zaptel drivers for hfcusb |
| 15:45.52 | tzafrir_home | would be interesting |
| 15:45.53 | icewaterman | tzafrir_home: would i need zaptel drivers? |
| 15:46.05 | tzafrir_home | either that or misdn |
| 15:46.15 | icewaterman | i have those misdn drivers working right now (not with asterisk though) |
| 15:46.28 | icewaterman | tzafrir_home: misdn drivers work |
| 15:46.28 | tzafrir_home | so you don't need bristuff |
| 15:46.55 | coppice | "misdn drivers work" is a very original comment :-) |
| 15:47.27 | tzafrir_home | yeah. They work much better htan the zaptel driver for this device |
| 15:47.44 | icewaterman | tzafrir_home: seems like debian packages will have bri-stuff enabled anyway, cant do much about it but it should not do much harm |
| 15:48.07 | tzafrir_home | It shouldn't do any harm, right |
| 15:48.18 | *** join/#asterisk |dennis| (n=dennis@200.32.236.10) |
| 15:49.06 | icewaterman | uhm btw. will asterisk also work with the isdn4linux driver? |
| 15:49.33 | tzafrir_home | there's some obsolete support. Not recommended |
| 15:49.48 | icewaterman | ok, then i have to stay with misdn. |
| 15:49.57 | tzafrir_home | (chan_modem_i4l) |
| 15:50.13 | icewaterman | apt-cache search chan_modem |
| 15:50.18 | icewaterman | ooops wrong window |
| 15:50.24 | tzafrir_home | or chan_capi . or chan_visdn ... |
| 15:50.32 | killfill_ | hey tzafrir_home would you know why would granstream phones not hear anything when zalling throught ZAP? over SIP it does work.. softSIP phones works and IAX ones too... |
| 15:50.47 | tzafrir_home | no, you really shouldn't waste time on chan_modem |
| 15:51.00 | icewaterman | tzafrir_home: chan_capi is there but i guess that one requires a capi driver like misdn |
| 15:51.00 | killfill_ | "over SIP it does work" i mean calling to another SIP phone... |
| 15:51.27 | tzafrir_home | IIRC you can use misdn as a capi driver, but I'm not sure |
| 15:51.29 | *** join/#asterisk riddlebox (n=james@75-132-225-75.dhcp.stls.mo.charter.com) |
| 15:52.11 | tzafrir_home | killfill_, for starters, check if you "see" audio in the relevant channel with ztmonitor NN -v |
| 15:52.47 | killfill_ | oh.. |
| 15:54.36 | icewaterman | tzafrir_home: the website for chan_capi says: This module (often declared as 'driver', but it isn't) provides the connection between the PBX software and ISDN Hardware which provides a CAPI 2.0 compatible interface. |
| 15:54.58 | icewaterman | looks to me as if it required capi support from the isdn-driver in the first place |
| 15:55.25 | *** join/#asterisk puga (n=puga@201-92-24-88.dsl.telesp.net.br) |
| 15:55.39 | tzafrir_home | icewaterman, again, I know less about misdn and capi, sorry |
| 15:56.01 | tzafrir_home | latest debs should support misdn, IIRC |
| 15:56.06 | killfill_ | tzafrir_home: i can see the the levels ok in both directions... |
| 15:56.06 | icewaterman | tzafrir_home: but you have a howto for creating sip accounts with asterisk? ;-) |
| 15:56.07 | puga | anyone here knows how to configure zaptel for MFC/R2 signalling?? |
| 15:56.11 | killfill_ | letms get the generated audio file |
| 15:56.34 | tzafrir_home | killfill_, use -v . you can easily see if there is or isn't audio |
| 15:56.36 | killfill_ | oh well.. its lenght 0.. but i saw the levelsok |
| 15:56.46 | *** join/#asterisk jameswf-home (n=that@ip72-204-228-104.ph.ph.cox.net) |
| 15:57.06 | killfill_ | yup. RX was up when peer at ZAP talked |
| 15:57.07 | tzafrir_home | "tx" is audio sent from asterisk, "rx" is audio recieved by Asterisk |
| 15:57.16 | killfill_ | yup. its recieving |
| 15:57.22 | killfill_ | so its asterisk/sip part |
| 15:58.34 | killfill_ | wired tho.. when i "-f output" output is 0 bytes. |
| 15:59.28 | omarc55 | Hi all, I am trying to get IAX transfer to work between 2 IAX servers using trunking but I see that the server stays in the media path. what can I check so I can get this to work? |
| 15:59.47 | *** join/#asterisk mattboll (n=mattboll@br137-1-82-228-156-113.fbx.proxad.net) |
| 15:59.52 | mattboll | hi |
| 15:59.53 | killfill_ | tzafrir_home: what else can i test?.. |
| 16:00.37 | tzafrir_home | killfill_, what exactly is the problem? Asterisk doesn't send audio? |
| 16:00.55 | tzafrir_home | puga, you need chan_unicall |
| 16:01.22 | killfill_ | when calling from hardphone SIP to an external ZAP phone, the sip phone cannot hear what the person on zap is. |
| 16:01.26 | killfill_ | but the zap does hear sip |
| 16:02.07 | *** join/#asterisk IPetrov2 (i=IPetrov2@ppp30-211.pppoe.mtu-net.ru) |
| 16:02.10 | killfill_ | with softphones it works tho. (and i just change the asterisk server.. no modification on the hard phones) |
| 16:03.13 | tzafrir_home | There used to be a page http://soft-switch.org/unicall/installing-mfcr2.html , but I have no idea where it went |
| 16:03.44 | BBHoss | does an echo test work? |
| 16:04.22 | killfill_ | BBHoss: between sip's it works okey (and IAX's too) |
| 16:04.59 | killfill_ | and the wired thing, is that when an incoming call comes from the zap, and get connected to the sip hardphone. everything works fine |
| 16:05.12 | mattboll | does anyone know if it is possible to talk to the person when we transfer a call ? |
| 16:05.29 | killfill_ | its just when calling to the pstn.... outgoing calls. |
| 16:05.41 | BBHoss | mattboll:which person, the person you're transferring to? |
| 16:05.56 | mattboll | "can I talk to X ? sure, wait... mister X someone for you... and then they talk together |
| 16:06.07 | BBHoss | you want attended xfer |
| 16:06.17 | BBHoss | you on zaptel or sip/iax? |
| 16:06.23 | mattboll | sip |
| 16:06.31 | BBHoss | what kind of phone |
| 16:07.23 | mattboll | softphone (ekiga) or some other but don't know which yet |
| 16:07.48 | BBHoss | hmm |
| 16:08.06 | BBHoss | im not sure how attended works or even IF it is supported by ekiga |
| 16:08.18 | BBHoss | you can use the built-in xfer though |
| 16:08.24 | BBHoss | as if you were on an analog phone |
| 16:09.00 | BBHoss | you just have to set up features.conf, then make sure your dial options allow for xfers |
| 16:09.16 | BBHoss | http://www.voip-info.org/wiki-Asterisk+config+features.conf |
| 16:10.05 | BBHoss | you need the t and T dial options |
| 16:10.31 | jameswf-home | ! google |
| 16:10.32 | mattboll | ok, thanks a lot, now that I know the name (attended xfer) it should be more easy ;) |
| 16:10.37 | mattboll | easier sorry |
| 16:10.55 | tzafrir_home | jameswf-home, you left out a space |
| 16:13.24 | *** join/#asterisk linxroute (n=linx@125.214.28.119) |
| 16:17.19 | *** join/#asterisk docelm0 (n=vircuser@c-76-99-153-242.hsd1.de.comcast.net) |
| 16:23.16 | *** join/#asterisk atomicd (n=atomicd@adsl-69-109-58-155.dsl.irvnca.pacbell.net) |
| 16:23.53 | *** join/#asterisk klauwhamer (n=felixdhc@ipd50af070.speed.planet.nl) |
| 16:24.41 | killfill_ | tzafrir_home: what else can i try?.. i can hear all okey with ztmonitor... |
| 16:24.48 | killfill_ | bu the damn phone doesn get any sound.. |
| 16:28.19 | atomicd | How can I pause the screen per page in the CLI to prevent the output of a command from scrolling off the screen? For commands at the system prompt, I can use the "more" command. Anything like that in Asterisk's CLI? |
| 16:31.03 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 16:31.23 | *** join/#asterisk lirakis (n=lirakis@cpe-68-175-38-65.nyc.res.rr.com) |
| 16:31.23 | killfill_ | the echo test works ok and all... |
| 16:31.25 | killfill_ | :S |
| 16:31.59 | *** join/#asterisk xheliox (n=jeff@pdpc/supporter/active/xheliox) |
| 16:36.47 | *** part/#asterisk lirakis (n=lirakis@cpe-68-175-38-65.nyc.res.rr.com) |
| 16:48.57 | *** join/#asterisk orkid (n=mike@unaffiliated/orkid) |
| 16:56.33 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 17:01.56 | *** join/#asterisk emist_ (n=emist@unaffiliated/emist) |
| 17:04.26 | *** join/#asterisk fordfrog (n=fordfrog@gentoo/developer/fordfrog) |
| 17:07.26 | *** join/#asterisk mordaunt (n=mordaunt@unaffiliated/mordaunt) |
| 17:10.22 | *** join/#asterisk CrazyTux[m] (n=CrazyTux@adsl-68-90-41-25.dsl.hstntx.swbell.net) |
| 17:20.25 | *** join/#asterisk nighty^ (n=nighty@sushi.rural-networks.com) |
| 17:22.27 | *** part/#asterisk ekimus_ (n=mm@xover.htu.tuwien.ac.at) |
| 17:35.33 | *** join/#asterisk ManxPower (n=manxpowe@218.sub-75-200-126.myvzw.com) |
| 17:40.11 | unixdog | ok your all fired |
| 17:44.37 | *** join/#asterisk CrazyTux[m] (n=CrazyTux@adsl-68-90-41-25.dsl.hstntx.swbell.net) |
| 17:46.18 | ManxPower | You can't fire us! We quit! |
| 17:46.40 | BBHoss | anyone here use Oreill'y Safari? |
| 17:46.46 | BBHoss | Oreilly's |
| 17:47.10 | *** join/#asterisk Hekt0r (n=HoraceX@ppp-70-246-228-73.dsl.ksc2mo.swbell.net) |
| 17:47.45 | ManxPower | I used to. |
| 17:50.03 | Hekt0r | Hi. I'm asstempting to upgrade to 1.4.13. After I disk make install I get warnings that a variety of modules were not installed by this version of asterisk and might be incompatible. They are app_addon_sql_mysql.so, app_cut.so, ... cdr_addon_mysql.so, ... pbx_functions.so, res_config_mysql.so. I reran menuselect and didn't see these as options. What am I doing wrong? |
| 17:50.46 | ManxPower | Hekt0r: nothing. those are modules YOU added after you installed 1.2.x, and they are not included in 1.4.x |
| 17:51.11 | ManxPower | i.e. 1.4.x does not include MySQL support, because of licensing issues, but those modules are available in asterisk-addons, if you need them. |
| 17:52.01 | ManxPower | rm those files in /usr/lib/asterisk/modules |
| 17:52.32 | BBHoss | !book |
| 17:52.33 | ManxPower | Some of those modules may have been in 1.2.x but are not in 1.4.x, for example app_cut.so was removed and replaced with a func_app_cut |
| 17:52.39 | BBHoss | ! book |
| 17:52.43 | ManxPower | ~book |
| 17:52.46 | 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 |
| 17:53.36 | Hekt0r | Are pbx_functions and app_math.so replaced? How do I know which ones are replaced and which ones I need find and compile? |
| 17:54.01 | ManxPower | Hekt0r: no real way if you don't remember which modules you added before. |
| 17:54.08 | Qwell | all of the ones that it complains about |
| 17:54.14 | Qwell | remove them all |
| 17:54.32 | ManxPower | from the list, it looks like the mysql things is the only thing you added that you MIGHT be needing now. Do you use the MySQL stuff in Asterisk? |
| 17:54.43 | Qwell | still need to rebuild it |
| 17:55.23 | Hekt0r | I'm not sure if mysql is used. I'd assume it would be fore some sort of reporting? |
| 17:55.40 | ManxPower | Hekt0r: You did not install this Asterisk did you? |
| 17:56.02 | Hekt0r | Nope. It was three guys ago. |
| 17:56.23 | ManxPower | Hekt0r: we really can't help you much if you don't know anything about the system. |
| 17:56.44 | Qwell | 3 people in less than 2 years? |
| 17:56.48 | Qwell | sounds like a crappy place to work |
| 17:56.51 | ManxPower | And really, if you know so little about the system, you're screwed anyway when you upgrade from 1.2.x to 1.4.x |
| 17:57.11 | Hekt0r | Why am I screwed? |
| 17:57.47 | ManxPower | Hekt0r: because there is a large chance the upgrade won't work out of the box and you don't know enough about the system to fix the problems. I assume you read UPGRADE.txt and the changelog for 1.4.x? |
| 17:58.24 | Hekt0r | Ok. thanks for the help. |
| 17:58.33 | ManxPower | Poor guy. |
| 17:58.43 | Qwell | I have no sympathy |
| 17:59.05 | Qwell | he's gonna get fired like the other 3 anyways :p |
| 17:59.11 | ManxPower | I wonder if the previous admins knew as much about Asterisk. |
| 17:59.17 | `Sean | lol |
| 17:59.45 | ManxPower | Qwell: People don't understand just how incredibly complex and technical a VoIP PBX is. |
| 18:00.01 | Qwell | you're preaching to the choir |
| 18:00.11 | ManxPower | I know. |
| 18:02.17 | ManxPower | I'll be back later, I need to VPN into a client and can't get to IRC from there. |
| 18:02.38 | Qwell | O.o |
| 18:03.11 | robl^ | But.. but.. PBX are simple! Just use a GUI |
| 18:07.41 | TJNII | Okay, this just confises me: |
| 18:07.42 | TJNII | Failed to execute '/var/lib/asterisk/agi-bin/wakeup.php': No such file or directory |
| 18:07.42 | TJNII | <PROTECTED> |
| 18:07.50 | TJNII | RanchNet:/etc/asterisk# ls -l /var/lib/asterisk/agi-bin/wakeup.php |
| 18:07.50 | TJNII | -rwxrwx--- 1 asterisk asterisk 20887 2007-09-25 20:39 /var/lib/asterisk/agi-bin/wakeup.php |
| 18:09.17 | Mw3 | head -1 /var/lib/asterisk/agi-bin/wakeup.php |
| 18:09.38 | Qwell | -n1 |
| 18:10.50 | TJNII | Aah. I know |
| 18:11.00 | TJNII | New server. PHP isn't installed yet |
| 18:11.35 | TJNII | Mw3: Thanks. If you hadn't told me to look at the shebang line I would have kept pounding at permissions. |
| 18:26.21 | BBHoss | anyone here ever use PIKA boards? |
| 18:28.13 | *** join/#asterisk duki (n=duki@host-85-27-58-159.brutele.be) |
| 18:28.36 | TJNII | Where is the default sounds directory specified? |
| 18:28.49 | unixdog | you where stepping on my paw |
| 18:29.01 | *** join/#asterisk stybba (n=stybba@201.204.40.106) |
| 18:30.02 | unixdog | anyone here ever use sipfoundry sipx in conjunction with asterisk |
| 18:33.16 | TJNII | The debian build has the sound directory pointed to /usr/share/asterisk and I want it pointed to /var/lib/asterisk/sounds.... |
| 18:33.24 | TJNII | Well, a symlink will work for now.... |
| 18:33.57 | unixdog | thats because you did not use the emerge to install |
| 18:34.05 | unixdog | and to update |
| 18:34.06 | TJNII | emerge is gentoo |
| 18:34.28 | unixdog | but deb does not use the standard layout |
| 18:34.52 | TJNII | As I've noticed. Is that compiled in? Where is that directory specified? |
| 18:35.07 | Strom_M | TJNII: asterisk.conf |
| 18:35.14 | unixdog | you have to use the deb pkg system to install |
| 18:35.15 | Strom_M | I believe it's /etc/asterisk.conf |
| 18:35.27 | unixdog | /etc/asterisk/asterisk.conf |
| 18:35.42 | Strom_M | disclaimer: i've just woken up |
| 18:35.48 | unixdog | but then again I work on freebsd and asterisk |
| 18:35.59 | unixdog | and i love it |
| 18:36.18 | TJNII | Hmmmm... I don't have an entry for it. Would it default to a specified directory? |
| 18:36.39 | unixdog | use locate |
| 18:36.42 | Strom_M | TJNII: run updatedb and then type "locate asterisk.conf" |
| 18:36.44 | unixdog | use find |
| 18:36.44 | TJNII | I copied all the confs off a system that works with /var/lib/asterisk, to this strikes me as a bit odd |
| 18:37.01 | unixdog | thats why |
| 18:37.58 | TJNII | Also a grep for /usr/share/asterisk in /etc yeilds nothing |
| 18:38.56 | TJNII | unixdog: what do you mean by "thats why" |
| 18:42.06 | *** join/#asterisk russellb (n=russell@asterisk/developer-and-stable-maintainer/drumkilla) |
| 18:42.06 | *** mode/#asterisk [+o russellb] by ChanServ |
| 18:44.17 | killfill_ | what could this mean? chan_sip.c:3625 sip_write: Asked to transmit frame type 4, while native formats is 0x1 (g723)(1) read/write = 0x8 (alaw)(8)/0x4 (ulaw)(4)? |
| 18:44.53 | *** join/#asterisk Chris-NB (n=chris@213162066163.public.t-mobile.at) |
| 18:46.05 | killfill_ | hm.. |
| 18:46.07 | killfill_ | channel.c:2991 set_format: Unable to find a codec translation path from g723 to alaw |
| 18:47.25 | killfill_ | how do i fix this?... |
| 18:48.25 | unixdog | do you have g723 installed |
| 18:49.18 | Strom_M | killfill_: for g723 you have to install the digium transcoder card |
| 18:49.29 | Strom_M | otherwise you can only do g723 in passthrough mode |
| 18:50.00 | killfill_ | hm.. in the practice what does this mean?.. |
| 18:50.22 | killfill_ | how do check if i have g723 working fine? |
| 18:50.33 | unixdog | g723 to g723 |
| 18:50.57 | unixdog | other then that no real way |
| 18:51.10 | killfill_ | http://pastebin.ca/768573 i see that |
| 18:51.12 | Strom_M | passthrough is exactly what you'd expect it to be |
| 18:51.22 | unixdog | unless you get the opensource g723 software |
| 18:51.38 | Strom_M | opensource my ass |
| 18:51.40 | Strom_M | it's patented |
| 18:51.55 | Strom_M | you don't go running around violating patents :) |
| 18:52.14 | unixdog | if its used for testing and or non profit its not |
| 18:52.17 | killfill_ | but wait. my problem begins when i take this granstream phone, and make a call. |
| 18:52.26 | Strom_M | killfill_: read your own messages |
| 18:52.27 | Strom_M | # |
| 18:52.28 | Strom_M | Disclaimer: this command is for informational purposes only. |
| 18:52.28 | Strom_M | # |
| 18:52.28 | Strom_M | <PROTECTED> |
| 18:52.32 | killfill_ | should i better make the phone choose a different codeco or something? |
| 18:52.39 | Strom_M | killfill_: yes |
| 18:52.52 | unixdog | g729/ulaw/alaw/gsm/ilbc |
| 18:54.01 | TJNII | I hate it when you're tinkering with something, and something else breaks in what you're tinkering with, but the two are unrelated. |
| 18:54.18 | TJNII | Then you're pounding the dirt going "What did I do!" |
| 18:54.30 | TJNII | Especially when the breakage isn't your fault |
| 18:55.22 | killfill_ | hm... |
| 18:55.36 | De_Mon | Strom_M maybe YOU dont... |
| 18:57.04 | killfill_ | yeah. the codec selection of the phone is 1: PCMU 2: PCMA 3: G724, 4: etc etc. |
| 18:57.06 | *** join/#asterisk ghento (n=ghento@75.155.241.7) |
| 18:57.19 | killfill_ | when i change G724 for GSM, then it selects GSM and it works.. |
| 18:57.25 | killfill_ | ^_^ |
| 18:57.35 | killfill_ | but.. gsm is like the worst quelity.. right?... |
| 18:57.54 | killfill_ | why doesnt pcmu/a get selected?.. thats u/a law right?.. |
| 18:58.59 | killfill_ | unixdog, oh.. the list of codecs you writed, are like orderen by quality of sound?.. |
| 19:08.10 | *** join/#asterisk Hadi- (n=Hadi@CPE001310492769-CM001225e00576.cpe.net.cable.rogers.com) |
| 19:08.14 | Hadi- | hi everyone |
| 19:08.37 | TJNII | hmmmm.... my iax port isn't open, even though it should be..... |
| 19:09.23 | Hadi- | just a quick question... we have a SIP PRI connected directly to our CISCO 2800 series router... we are sending some outgoing calls from asterisk to the Cisco 2800 series.. ans we are getting a lot of Got SIP response 486 "Busy here" |
| 19:11.45 | Hadi- | any ideas why |
| 19:14.37 | *** join/#asterisk techie (n=techie@adsl-76-214-29-227.dsl.lsan03.sbcglobal.net) |
| 19:19.18 | JT | killfill_: wow what a surprise, getting rid of G.724 works... |
| 19:20.00 | killfill_ | does people not like 724?... |
| 19:20.30 | JT | i have never heard of anyone using it, ever |
| 19:20.45 | JT | and most importantly... it is NOT supported by asterisk |
| 19:21.06 | bantu | a |
| 19:22.26 | Hadi- | do you guys recommand any good voip radios billing software? |
| 19:22.37 | Hadi- | that word with Cisco |
| 19:22.45 | Hadi- | supports.. |
| 19:23.02 | JT | radios billing software? |
| 19:23.07 | Strom_M | Cisco? |
| 19:23.44 | Hadi- | RADIUS |
| 19:23.45 | Hadi- | sorry |
| 19:23.46 | Hadi- | yes |
| 19:24.27 | Hadi- | calling card application as well as wholesale would be nice |
| 19:31.09 | *** join/#asterisk EclecticRob (n=Eclectic@adsl-6-81-50.msy.bellsouth.net) |
| 19:34.57 | EclecticRob | Hi all, I am experiencing a strange problem that I have not been able to solve. I have an asterisk server in a data-center and two computers at home behind a NAT DSL connection. One is an Ubuntu box and the other is a Mac. I can call out from both systems perfectly fine and everything works great. I can call the Mac box via a local extension from the Ubuntu box without any problems. When I call from the Mac to the Ubuntu box vi |
| 19:36.02 | JT | EclecticRob: try not to make so long questions |
| 19:36.46 | EclecticRob | Okay |
| 19:36.53 | TJNII | RTP traffic is all UDP, correct? |
| 19:37.05 | JT | the ircd cuts of lines after about a billion characters |
| 19:37.45 | EclecticRob | heh, sorry |
| 19:38.01 | Strom_M | TJNII: yes |
| 19:38.07 | *** join/#asterisk xpot (n=jim@c-71-195-241-115.hsd1.ma.comcast.net) |
| 19:38.45 | TJNII | Okay. That's probably why my SIP straffic wasn't working. I just discovered I set the RTP port forwarding to TCP instead of UDP |
| 19:38.59 | killfill_ | SIP/19-087b6000 recieved frame with invalid timing info: has_timing_info=0, len=0, ts=0, src=g729tolin_frameout |
| 19:39.05 | killfill_ | what means this? |
| 19:44.12 | *** join/#asterisk russellb (n=russell@asterisk/developer-and-stable-maintainer/drumkilla) [NETSPLIT VICTIM] |
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| 19:45.28 | endre | oh hi |
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| 19:48.11 | katsuodo | hello |
| 19:48.40 | katsuodo | receive error 3 no route destination for sip phone |
| 19:49.54 | TJNII | katsuodo: NAT? |
| 19:50.53 | katsuodo | linksys router |
| 19:51.05 | katsuodo | would this cause a problem |
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| 19:54.51 | TJNII | So phone -> (Nat) -> * server? |
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| 19:58.17 | katsuodo | TNJII yes the sip phone is plugged into linksys connected to the server |
| 20:00.30 | katsuodo | made change to sip.conf to allow nat=yes |
| 20:00.35 | katsuodo | one moment |
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| 20:06.03 | katsuodo | TJNII the message is as follows warning [4521]: app_dial.c:1106 dial_exec_full : Unable to create channel of type SIP (cause 3 no route to destination) |
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| 20:15.01 | [TK]D-Fender | katsuodo, Read this, now : |
| 20:15.03 | [TK]D-Fender | ~sipnat |
| 20:15.04 | jbot | [~sipnat] Quick guide on configuring * + SIP behind NAT : http://www.aocomputing.net/?p=3 otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
| 20:16.49 | katsuodo | okay |
| 20:16.57 | katsuodo | hey jbot |
| 20:17.04 | katsuodo | how you doing |
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| 20:17.47 | curtn | for g.729a... there is no free solution ? |
| 20:19.28 | *** join/#asterisk rsd (n=chaos@200.181.133.130) |
| 20:20.32 | curtn | mp3 is not free... but the licensing terms on G.729a seems to be a little bit "agressive" |
| 20:24.07 | hesco | I would have thought that ${CDR(dst)} would have given me the destination phone number. But instead I'm getting the number I put in the CID setting. How is it I log the numbers of the outgoing calls I'm making? |
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| 20:25.56 | [TK]D-Fender | katsuodo, don;'t talk to the bot... its too late for that much humour :) |
| 20:29.04 | katsuodo | understood |
| 20:30.06 | curtn | ~g729 |
| 20:30.07 | jbot | i heard g729 is an ITU-standard voice codec which operates at 8kbps and offers quality very similar to GSM. G.729 is patent-encumbered; those wishing to use it with Asterisk must buy a license from Digium. |
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| 20:59.00 | khronos | w |
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| 21:15.58 | MrTelephone | for you guys using openser don't you lose the * codes in chan_sip? |
| 21:16.38 | MrTelephone | like *69 |
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| 21:21.13 | Strom_M | chan_sip has vertical service codes? |
| 21:21.29 | becks` | hi, somebody ever used the protos sip test with an asterisk server? i wonder if asterisk would forward messages with corrupt headers to the device under test |
| 21:23.52 | MrTelephone | strom, is that what you call them? |
| 21:24.01 | Strom_M | ~vsc |
| 21:24.02 | jbot | extra, extra, read all about it, vsc is Vertical Service Codes such as *67, *69, *72, and *82. These codes are generally reserved for specific uses, and it's a bad idea to conflict with the official assignments. A list of assigned VSCs for North America is at http://nanpa.com/number_resource_info/vsc_assignments.html and ... |
| 21:25.50 | [TK]D-Fender | MrTelephone, No such thing unless you've coded your dialplan that way |
| 21:27.24 | MrTelephone | if you do the sdp properly and contact info properly for two nat clients.. do you necessarily have to proxy the rtp too? |
| 21:29.38 | MrTelephone | the more i read about it the more i don't think you need to proxy the rtp |
| 21:37.35 | Strom_M | who stood on the internet hose again |
| 21:37.35 | ricko73 | foot |
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| 21:47.49 | killfill_ | how do i check if a sip or iax user is connected?.. this way i could route: "IF the user is connected, ring him. If not, goto Recepcionist" |
| 21:49.34 | robl^ | chanisavail |
| 21:50.28 | robl^ | http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+ChanIsAvail |
| 21:57.41 | *** mode/#asterisk [-r] by Qwell |
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| 21:59.00 | killfill_ | hm.. |
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| 22:04.28 | Strom_M | you could also use qualify=yes |
| 22:05.30 | MrTelephone | i dunno i like asterisk better than openser |
| 22:07.29 | killfill_ | qualify? |
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| 23:11.53 | killfill_ | is there a way to define, that when a call get into a queue, and is there for 20 minutes, to refirect the call to another queue? |
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| 23:19.05 | Sunmoon__ | hello there |
| 23:19.10 | Sunmoon__ | anyone home |
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| 23:20.52 | [TK]D-Fender | killfill_, yes. "show application queue" |
| 23:21.54 | lvl- | killfill_, if a call isn't answered by the queue, the dialplan simply continues at the next priority |
| 23:22.44 | killfill_ | hm... you mean i could set a timeout there.. |
| 23:22.54 | killfill_ | (where i go to the queue) |
| 23:23.08 | lvl- | yep |
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| 23:39.35 | killfill_ | grate |
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| 23:43.55 | kopke | hi all, I'm new user of Asterisk. I've got a problem with VAD! I have a local asterisk, with 2 clients, a Cisco 7960 and a softphone. I set up MOH, I put enablevad=0 to my Cisco, and MOH works well between the two one. But when I use a VSP, to make call to french mobile operator, my MOH plays only when I speak or make noise, so I want to know if I can set up a global novad to asterisk? I tried silencesuppresion to yes or no, nothing changed, and trie |
| 23:43.55 | kopke | s some codecs but nothing better :( Anyone has an idea? |
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| 23:49.54 | [TK]D-Fender | kopke, * doesn |
| 23:50.08 | [TK]D-Fender | kopke, * doesn's SUPPORT VAD. You have to tell your PROVIDER to stop |
| 23:50.51 | kopke | OK, that what I was worry, so should'nt be possible! |
| 23:52.01 | kopke | I read that problem is that asterisk takes timer from the other side, to play the MOH, and read one time an option like internal_timer that should force asterisk to play it, but no precision? |
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| 23:52.52 | ymonsalvez | hi |
| 23:52.55 | ymonsalvez | i need help |
| 23:53.20 | ymonsalvez | sorry but english is bad |
| 23:53.54 | J4k3 | english is an awful language anyways |
| 23:54.00 | J4k3 | too much flow control, not enough content. |
| 23:55.11 | ymonsalvez | I have server asterisk with gateway gsm voiceblue 2n |
| 23:56.10 | ymonsalvez | for finished cell calls |
| 23:57.14 | ymonsalvez | how can hangup calls for a group of channels |
| 23:58.51 | ymonsalvez | with soft hangup can just hang a call from a specific channel |
| 23:59.30 | ymonsalvez | but for more than one channel there is something about that |
| 23:59.49 | ymonsalvez | thanks |