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| 00:29.07 | *** mode/#asterisk [+o russellb] by ChanServ |
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| 01:02.29 | cmw_ | is there any known bugs with iax2 that prevent the bind & source port being the same? |
| 01:08.57 | [TK]D-Fender | jaytee: Nope. |
| 01:10.24 | BrianR___ | [TK]D-Fender: I noticed a difference between 1.0.9 and 1.4 - where in an IVR dialplan there seems to be a bit of a delay after entering the target extension |
| 01:10.36 | BrianR___ | and if I enter a valid extension followed by more digits, it's considered invalid... |
| 01:10.56 | BrianR___ | Where as in 1.0.x it completed as soon as enough digits were entered to match an extension. |
| 01:11.02 | [TK]D-Fender | BrianR___: pastebin |
| 01:11.21 | [TK]D-Fender | BrianR___: And no, the way digits are accepted is not different. |
| 01:11.40 | [TK]D-Fender | BrianR___: So its just a question of what the "oops" was that you've done that created this outcome. |
| 01:11.41 | BrianR___ | [TK]D-Fender: Well.. WHat about auto fall through or whatever it's called? |
| 01:12.05 | [TK]D-Fender | BrianR___: "autofallthrough=no" = old style "s" IVR's |
| 01:12.11 | *** part/#asterisk cmw_ (n=ice@167.206.176.233) |
| 01:12.26 | BrianR___ | In 1.0.x, the default at the end of running through all of the priorities for a given extension was to listen for another extension, right? |
| 01:12.37 | [TK]D-Fender | BrianR___: And in considerations, all extens should have a "hangup" at the end |
| 01:12.59 | BrianR___ | Hmm.. I wonder if I have something stupid like a _. or something. |
| 01:13.05 | [TK]D-Fender | BrianR___: Nope. In 1.0 at the end of an extension OTHER than "s" it would auto-disconnect |
| 01:13.15 | [TK]D-Fender | BrianR___: "s" = IVR so it'd wait |
| 01:13.25 | BrianR___ | [TK]D-Fender: Aah. |
| 01:13.27 | [TK]D-Fender | BrianR___: go look closely :) |
| 01:14.34 | BrianR___ | So what's the behavior for a "s" extension in 1.4.x? I've gone and added a WaitExten() as the last priority... |
| 01:16.26 | *** join/#asterisk vonkleist (n=vonkleis@189.155.176.253) |
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| 01:19.21 | *** part/#asterisk LemensTS (i=LemensTS@adsl-70-238-160-24.dsl.stlsmo.sbcglobal.net) |
| 01:21.23 | BrianR___ | [TK]D-Fender: found it. Fat fingers with an unlimited length extension somewhere. |
| 01:21.52 | [TK]D-Fender | BrianR___: You're welcome. |
| 01:22.04 | *** part/#asterisk `paul (n=paul@125.252.70.126) |
| 01:22.05 | BrianR___ | [TK]D-Fender: Thanks... |
| 01:22.14 | BrianR___ | [TK]D-Fender: I really should have kept on top of this versio thign better. :( |
| 01:22.29 | [TK]D-Fender | BrianR___: au disable auto-fallthrough and still run IVR's off "s" the old-fashioned way myself. WaitExten is a largely worthless app.... |
| 01:24.24 | *** join/#asterisk vipcarrier (n=vipcarri@ool-44c65232.dyn.optonline.net) |
| 01:26.43 | vipcarrier | any openser gury's? |
| 01:33.17 | rpm | vipcarrier, openser gurus or sip guru's? |
| 01:36.51 | *** join/#asterisk deltaray2 (n=deltaray@1.79.244.66.sdsl.sta.smithvilledsl.net) |
| 01:37.30 | deltaray2 | I have an asterisk server that is working great, it is hooked up to a four port FXO card with two pots lines hooked up to it. |
| 01:38.09 | deltaray2 | But once in a while, we receive calls here the call goes through the calling plan and by the time we get it, it is the recording saying "If you'd like to make a call, please hang up and try again....." |
| 01:38.37 | justdave | sounds like your FXO card isn't recognizing that the calling party hung up |
| 01:38.39 | deltaray2 | Upon listening to a recording of the whole call, it seems that immediately when the call starts, a dialtone is sounding from the other side. |
| 01:39.07 | deltaray2 | Well, I was wondering if this happens when the calling part hangs up in between the time that Asterisk detects a call and it picks up the line. |
| 01:39.22 | justdave | or someone calls, it rings, they hang up before it answers, and the card answers after they hung up |
| 01:39.28 | deltaray2 | Other than that, FXO card works fine (Its a Rhino) |
| 01:39.44 | deltaray2 | Is there any way for Asterisk to detect this? |
| 01:40.10 | justdave | in theory it ought to be able to tell it's getting a dialtone when it picks up |
| 01:40.17 | *** join/#asterisk tobias (n=tobias@user-0ce2hvs.cable.mindspring.com) |
| 01:40.20 | justdave | but I haven't had to mess with that much so I'm not sure what the related settings are |
| 01:40.57 | justdave | not familiar with rhino, either, is that zaptel-compatible or does it have another driver layer between it and the zap drivers? |
| 01:41.05 | deltaray2 | I tried searching for this problem, but too many other results come up and its hard to narrow down. |
| 01:42.23 | deltaray2 | I think its zaptel compatible. |
| 01:42.23 | deltaray2 | Anyone else know? |
| 01:43.38 | justdave | if it's zaptel compatible, the settings in question would probably be in /etc/asterisk/zapata.conf or /etc/zaptel.conf |
| 01:43.44 | tzafrir_laptop | deltaray2, from which country are you? |
| 01:47.00 | *** join/#asterisk drdrain (n=kimmyd@cpe-066-057-105-080.nc.res.rr.com) |
| 01:48.07 | drdrain | Just out of curiosity, how does a ITSP aquire DIDs in a particular market? Do they lease them from the FCC or something? |
| 01:49.17 | drdrain | Is there a central administation somewhere that assigns number somehow? |
| 01:51.21 | *** join/#asterisk Chicago (n=Chicago@c-98-223-62-171.hsd1.in.comcast.net) |
| 01:51.30 | Chicago | Is there a recommend Windows Mobile pocket pc SIP client? |
| 01:51.37 | *** join/#asterisk CrazyTux[m] (n=CrazyTux@65-60-108-170.static-ip.telepacific.net) |
| 01:51.37 | Chicago | Or an asterisk port for WM6? |
| 01:53.09 | [TK]D-Fender | lol |
| 01:53.44 | Chicago | yeah, I am laughing on the inside. |
| 01:53.58 | Chicago | Too bad there isn't a reliable toolchain for Windows crap. |
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| 01:56.23 | *** join/#asterisk korihor (n=korihor@190.78.32.60) |
| 01:56.29 | [TK]D-Fender | Chicago: Not a lot of WM users to get recommendations from. Maybe tour download.com or something based on ratings. |
| 01:57.12 | Chicago | Looking for the FSF to provide something :) |
| 01:58.20 | tzafrir_laptop | Chicago, sure: an image of GNU/Linux to run on that device :-) |
| 01:58.51 | *** part/#asterisk Deeewayne (n=Deeewayn@nat/digium/x-540cf4e7d365abcc) |
| 01:59.56 | deltaray2 | tzafrir_laptop, United States |
| 02:00.31 | tzafrir_laptop | make sure you use fxoks . |
| 02:00.34 | *** join/#asterisk Kumbang (n=kumbang@router-wmi.paume.itb.ac.id) |
| 02:00.47 | tzafrir_laptop | If that doesn't help, you'll have to resort to busydetect |
| 02:00.52 | nosbig | Chicago, I am looking at a project called Fring... Seems to be an integrated contact manager for local phone contact, AIM, ICQ, Skype, MSN, SIP, and a few other technologies. |
| 02:00.58 | Chicago | tzafrir_laptop: Do you know of 2.6 kernel ROMS for my device; an XV6800? |
| 02:01.30 | Chicago | You left out jingle... is Fring doing things jingle isn't capable? |
| 02:01.30 | Nugget | what the fark is a "kernel ROM"? |
| 02:01.54 | deltaray2 | tzafrir_laptop, I think I am, I mean I have fxsks=1 and fxsks in my /etc/zaptel.conf |
| 02:01.55 | Chicago | Nugget: kernel ROM... probably is best translated as a stage 2 loader with a bzImage. |
| 02:02.28 | deltaray2 | So at least I'm using the koolstart protocol, right? |
| 02:03.34 | nosbig | Chicago, it appears to allow a direct SIP connection, which should allow me to, in theory, make free phone calls using my own * server and SIP trunks... |
| 02:03.38 | jaytee | fxs modules use fxo signalling, fxo modules use fxs signalling |
| 02:04.06 | Chicago | nosbig: ding ding ding |
| 02:04.06 | tzafrir_laptop | deltaray2, right, fxsks |
| 02:04.10 | deltaray2 | right |
| 02:04.37 | nosbig | jaytee, I understand what you are talking about; I've set up a few analog cards before... but that has to be the most convoluted and confusing terminaology for people to handle... |
| 02:05.04 | deltaray2 | Is the problem that I have a common one? When I've "used" other phone systems I've encounted that. Like you have a voicemail with just a fast busy signal or something. |
| 02:05.20 | Chicago | What is the license for Fring? |
| 02:05.25 | CrazyTux[m] | Hey guys anyone have experience with Snom360? I want the following call flow, A calls -> B, B calls -> C, B conferences (A,B,C), -> B drops itself out of dialog, and A -> C are in call |
| 02:05.27 | jaytee | it seems like it should be the reverse logically but it's kind of a throwback to the way telecoms setup their networks. |
| 02:05.27 | Nugget | it's not as convolutes as "terminaology" :) |
| 02:05.37 | Chicago | http://www.fring.com/license.htm |
| 02:05.58 | nosbig | Chicago, I have an unlimited data plan but very few airtime minutes... I just wonder if the EVDO in my area will have high enough bandwidth and low enough latency to handle the calle... |
| 02:06.32 | Chicago | nosbig: Please keep me posted if you can monitor your latency. I am also on a CDMA network and just bought a new handset to test with. |
| 02:07.15 | Chicago | nosbig: The bandwidth should be there, but I doubt if it will push g.711 ulaw |
| 02:08.14 | nosbig | Yeah, I hope to be able to use gsm... |
| 02:08.46 | Chicago | Is EVDO just a CDMA concept? |
| 02:09.50 | nosbig | EVDO is available on CDMA networks, as long as the equipment is capable of supporting it... I, unfortunately, know too little of the layering of cellular protocols and standards to know exactly how it fits in... |
| 02:10.55 | Chicago | nosbig: About the Fring software, I just joined #fsf to ask about FOSS to act as a SIP RFC 3261 client. |
| 02:11.05 | Chicago | I prefer something I can compile myself. |
| 02:11.36 | nosbig | Chicago, I am updating my * box right now, but I will add a SIP peer as soon as that is done... |
| 02:11.59 | nosbig | Yeah, I prefer FLOSS... I am really looking for a proof of concept at this point. |
| 02:14.45 | *** join/#asterisk Nasra (n=maxshipp@CPE001217b1920e-CM00159a010eda.cpe.net.cable.rogers.com) |
| 02:15.49 | nosbig | I believe I have the peer set up... Waiting on Fring to log in... |
| 02:15.57 | Chicago | Why isn't there a visual studio project for asterisk? |
| 02:16.01 | Chicago | Asterisk! * |
| 02:17.53 | [TK]D-Fender | Chicago: I nominate you to go make one... |
| 02:18.08 | Chicago | [TK]D-Fender: Thank you for your nomination. |
| 02:18.18 | Chicago | Shall I use a git repository or SVN? |
| 02:18.45 | CrazyTux[m] | [TK]D-Fender, any experience with SNOM360? I'm trying to do A calls -> B, B calls -> C, B connects, A B C together, and then B drops itself out and A -> C remain |
| 02:19.12 | nosbig | I have a 92ms ping to fring... |
| 02:19.23 | [TK]D-Fender | CrazyTux[m]: Maybe Snom simply doesn't support it. |
| 02:19.23 | Chicago | There are only 253 hits on google for "RFC 3261 WM6" |
| 02:19.34 | CrazyTux[m] | [TK]D-Fender, really? |
| 02:22.00 | [TK]D-Fender | CrazyTux[m]: Amazing isn't it? Some products support more features than others and are of varying quality as well thus inducing us to make an evaluative judgment! ! |
| 02:23.29 | *** join/#asterisk jameswf-home (n=james@ip68-2-99-240.ph.ph.cox.net) |
| 02:27.00 | *** join/#asterisk esaym (n=user@cpe-70-120-89-6.satx.res.rr.com) |
| 02:27.31 | Chicago | What what? Looks like Linux on an XV6800 -> o,o http://forum.xda-developers.com/showthread.php?t=341903&page=2 |
| 02:28.51 | *** join/#asterisk jmacz (n=jmacz@201.244.195.13) |
| 02:29.04 | Chicago | Even better!!! -> http://forum.ppcgeeks.com/showthread.php?t=26378 |
| 02:30.19 | nosbig | Well, fring sounded pretty good on the audio quality... The delay is pretty bad, though... It's about 1 second over cellular... |
| 02:30.34 | nosbig | And my WM5 device has no wi-fi. |
| 02:31.02 | drdrain | Hey James |
| 02:32.47 | [TK]D-Fender | hugs his WM6.1 upgraded HTC Touch |
| 02:33.13 | Chicago | [TK]D-Fender: This is an HTC Titan. I am looking for a procedure for the unlocking and bootloading. |
| 02:33.16 | nosbig | Well, I will personally avoid any other WM5 devices in the future... |
| 02:34.03 | [TK]D-Fender | nosbig: Of course that only means you won't be visiting the SMITHSONIAN any more ;) |
| 02:35.21 | nosbig | I know... But that is what the moto q has installed... |
| 02:35.39 | nosbig | The next phone I have, I want to have Linux embedded in it... |
| 02:36.12 | [TK]D-Fender | "Q" .... Shudder |
| 02:36.35 | [TK]D-Fender | a "smart phone".... with no touch-screen running off a jog-wheel |
| 02:38.07 | nosbig | Yeah.... It's better than the LG VX-6100 which was my previous phone... At least I have real mobile Internet... Verizon just has had nothing else which has excited me lately... |
| 02:38.09 | Nugget | shudders at the though of a linux phone |
| 02:38.19 | Nugget | "thought" even. |
| 02:42.11 | *** join/#asterisk xpl (i=xpl@84.126.212.17.dyn.user.ono.com) |
| 02:42.35 | nosbig | Well, with Android and the OpenMoko project, it is looking more and more like that is going to happen... And it can't happen soon enough. This Q like to reboot randomly, the mobile IE is total crap, and the battery life is terrible... |
| 02:44.29 | nosbig | like --> likes |
| 02:44.59 | *** part/#asterisk drdrain (n=kimmyd@cpe-066-057-105-080.nc.res.rr.com) |
| 02:47.03 | jameswf-home | bkruse: uses openmoko |
| 02:47.36 | rpm | anyone have tech specs for a motorola sbv5220 ? |
| 02:47.49 | jameswf-home | rpm: google |
| 02:48.31 | nosbig | Actually, a buddy and I were talking, and we thought it would be cool to have * on your cell phone to help make decisions as to whether or not to use a Wi-Fi or Cellular option... Or to automatically hairpin calls through a traditional * install for privacyt. |
| 02:48.45 | nosbig | Turn the cell phone into a glorified SIP client and use the data plan... ;-) |
| 02:48.54 | *** join/#asterisk drmessano (n=nonya@c-76-125-29-30.hsd1.ga.comcast.net) |
| 02:49.49 | frogonwheels | nosbig: I have * running on a router - but how much mem does a phone have... Would certainly be an interesting idea |
| 02:50.18 | frogonwheels | nosbig: I certainly wouldn't want to be doing transcoding (avoid it on my router install) |
| 02:53.10 | nosbig | I used to run * on a Linksys NSLU2 before my current Mini-ITX solution... It has 32MB RAM... |
| 02:54.00 | nosbig | It looks as it I have 64MB in my Q... |
| 02:54.12 | frogonwheels | yeah - mine has 32mb |
| 02:55.17 | frogonwheels | I'm starting to get more familiar with asterisk, but still have a few points I am just missing on.. |
| 02:55.54 | frogonwheels | For a home system - I would like to be able to dial in and pick up voicemail. I know how I could differentiate based on callerid.. |
| 02:55.58 | nosbig | On my current * installation (minimalist Debian), including a TFTP server and HTTP server, I am using 55MB of RAM on that system... When I have three g.711 calls in a conference room... |
| 02:56.07 | nosbig | frogonwheels, I do exactly that.... |
| 02:56.38 | frogonwheels | but what asterisk app do you use so that I can press a key to go to a different menu |
| 02:57.02 | frogonwheels | Ie it still rings locally, but if I press a * or something it will drop to a system menu. |
| 02:57.11 | *** join/#asterisk [TK]D-Fender (n=chatzill@64.235.218.194) |
| 02:57.37 | nosbig | Use the following application to throw you to a custom menu based upon callerID: GotoIf($[${CALLERID(num)}=2025551212]?menu|s|1:internal|1000|1) |
| 02:58.09 | frogonwheels | nosbig: easier in 1.4 .. just do 1,_X./2025552122, |
| 02:59.03 | frogonwheels | but what about pressing a key while it rings locally to switch to another menu? |
| 02:59.04 | nosbig | That says that if the CallerID number matches 202-555-1212, then send me to my personal menu, otherwise, call my internal extension, based upon my dialplan. |
| 02:59.07 | frogonwheels | or context |
| 02:59.37 | nosbig | I think the syntax you posted will match based upon the DID which was called, not the inbound CallerID... |
| 02:59.45 | [TK]D-Fender | frogonwheels: CID matching on exten predates 1.0.... |
| 02:59.48 | frogonwheels | nosbig: no - the / does inbound |
| 03:00.05 | [TK]D-Fender | and frogonwheels sample acts on CID |
| 03:00.05 | frogonwheels | I've tested it, it works :) |
| 03:00.17 | nosbig | There is an extension that you enable, as far as I recall... And yes, I am obviously more familiar with 1.2.... ;-) |
| 03:02.42 | frogonwheels | [TK]D-Fender: so can I have an incoming ring locally (and connect if picked up) and at the same time wait for a DTMF key to drop to a different context |
| 03:02.55 | nosbig | Since I have the menu, it is rare for me to need that button to break out of the ringing and into voicemail. |
| 03:02.59 | frogonwheels | .. and if that all times out go on to voicemail? |
| 03:04.03 | *** join/#asterisk xloafx (n=xxloaf@pool-71-186-196-154.bflony.east.verizon.net) |
| 03:04.48 | [TK]D-Fender | frogonwheels: its your dialplan, you can do anything you want |
| 03:05.35 | frogonwheels | [TK]D-Fender: ok.. so my sticking point is I need some pointers on how to wait for both a DTMF key _and_ still be ringing.. |
| 03:06.36 | [TK]D-Fender | frogonwheels: "core show application dial" <- go read its instructions |
| 03:07.18 | frogonwheels | [TK]D-Fender: I've been banging my head against it - and as soon as I said that last sentence - I worked out that Dial was probably the key and was just about to do that. |
| 03:07.29 | frogonwheels | so thanks - and I'll go read now. |
| 03:08.20 | frogonwheels | yeah. obvious now. |
| 03:09.04 | [TK]D-Fender | frogonwheels: Loke so much else we hid it in the "big print" :) |
| 03:09.07 | [TK]D-Fender | like* |
| 03:09.46 | frogonwheels | [TK]D-Fender: I got hung up on finding an application to do it - I forgot that I was calling Dial() and as soon as I made that connection .. |
| 03:11.02 | frogonwheels | [TK]D-Fender: that's gonna require me to ANswer straight away I assume |
| 03:11.18 | [TK]D-Fender | frogonwheels: Clearly |
| 03:11.52 | *** part/#asterisk Damin (n=damin@nucleus.nacs.net) |
| 03:12.12 | *** join/#asterisk Damin (n=damin@nucleus.nacs.net) |
| 03:12.15 | Damin | Morning.. |
| 03:13.21 | frogonwheels | [TK]D-Fender: would it be sane to do Dial(SIP/ ,5) Answer Dial(SIP/ ,10,dr) |
| 03:13.51 | [TK]D-Fender | frogonwheels: you tell me. |
| 03:14.28 | frogonwheels | [TK]D-Fender: I guess I'll suck it and see . thanks |
| 03:16.34 | *** join/#asterisk Thorn (n=thorn@unaffiliated/thorn) |
| 03:18.04 | jameswf-home | discovered bug 1123 |
| 03:18.14 | jameswf-home | argh |
| 03:20.38 | [TK]D-Fender | Nugget: you about? |
| 03:22.55 | xloafx | how would I go about playing back the music on hold when I actually put a call on hold? at the moment I useing a SIP softphone and it just goes the silence |
| 03:23.32 | frogonwheels | xloafx: did you test your moh was working? |
| 03:24.12 | xloafx | yeah if I setup a Dial string and give it the "m" attribute it plays it back |
| 03:25.19 | xloafx | oh wait maybe it is working |
| 03:25.42 | xloafx | Ill have to do a little more testing here |
| 03:25.42 | *** join/#asterisk SanityIO (n=SanityIO@77.242.105.93) |
| 03:25.43 | *** join/#asterisk SanityIO_ (n=SanityIO@77.242.105.93) |
| 03:28.40 | bkw_ | anyone know who runs voipmich.com and sip-happens.com? |
| 03:30.15 | *** join/#asterisk unpaidbill (i=bill@420nugs.info) |
| 03:31.33 | jameswf-home | voipmich is probably michael White @ e4 |
| 03:33.04 | *** join/#asterisk growltiger (n=growltig@ip70-179-54-235.sd.sd.cox.net) |
| 03:33.29 | jameswf-home | Most things voip + michigan have his hands on em |
| 03:42.47 | *** join/#asterisk nixfreak (n=nixfreak@mn-10k-dhcp1-5913.dsl.hickorytech.net) |
| 03:53.25 | *** join/#asterisk russellb (n=russell@asterisk/digium-open-source-team-lead/russellb) |
| 03:53.25 | *** mode/#asterisk [+o russellb] by ChanServ |
| 03:58.33 | *** join/#asterisk BWS (n=wang@CPE001ee57a9471-CM001a6680be10.cpe.net.cable.rogers.com) |
| 03:58.35 | BWS | hello |
| 03:58.47 | BWS | when I try to compile zaptel, I get make: *** No rule to make target `menuselect/menuselect.c', needed by `menuselect/menuselect'. Stop. can anyone help |
| 03:59.53 | drmessano | Did you run a configure? |
| 04:00.37 | BWS | yes, I did |
| 04:01.02 | BWS | I did a ./configure and then make |
| 04:03.45 | BWS | ne? |
| 04:03.50 | BWS | this is on ubuntu |
| 04:05.39 | yang | BWS: why didn't you get it via apt-get ? |
| 04:06.41 | bkw_ | wonders |
| 04:06.42 | frogonwheels | I BWS use module-assistant |
| 04:09.50 | *** join/#asterisk Nasra (n=maxshipp@CPE001217b1920e-CM00159a010eda.cpe.net.cable.rogers.com) |
| 04:10.28 | jameswf-home | <PROTECTED> |
| 04:12.22 | jameswf-home | take that |
| 04:18.01 | *** join/#asterisk anthonyuniversal (n=anthonyu@user-0cej2sa.cable.mindspring.com) |
| 04:18.05 | anthonyuniversal | Hola! :] |
| 04:18.41 | *** join/#asterisk Winkie (n=urmom@ur.fa.gs) |
| 04:19.12 | anthonyuniversal | Hey so does asterisk have the capability of voice response menues? |
| 04:19.48 | bkw_ | not for free |
| 04:21.15 | *** join/#asterisk Defraz (n=T0tal@24-117-156-21.cpe.cableone.net) |
| 04:21.23 | *** part/#asterisk anthonyuniversal (n=anthonyu@user-0cej2sa.cable.mindspring.com) |
| 04:21.39 | [TK]D-Fender | Actually, it was "yes" |
| 04:22.17 | jaytee | for free? |
| 04:23.23 | drmessano | Asterisk doesn't have the capability... third party apps + asterisk do.. |
| 04:23.41 | jaytee | I use Lumenvox, it ain't free but it works |
| 04:26.19 | [TK]D-Fender | drmessano: Yes, that is technically more accurate. |
| 04:26.37 | [TK]D-Fender | Sphinx.... how to get it to work... is a MYSTERY! |
| 04:31.16 | jameswf-home | ok got something for telemarketers and cnam anything else on my list |
| 04:31.57 | bkw_ | uses pocketsphinx |
| 04:31.58 | bkw_ | works great |
| 04:41.58 | *** join/#asterisk ball (n=ball@216.176.80.183) |
| 04:46.11 | ball | Daviey! Stop following me ;-) |
| 04:59.01 | *** join/#asterisk russellb (n=russell@asterisk/digium-open-source-team-lead/russellb) |
| 04:59.01 | *** mode/#asterisk [+o russellb] by ChanServ |
| 05:10.48 | jameswf-home | drmessano: you use gizmo |
| 05:12.17 | phix | hai |
| 05:12.24 | russellb | ohi2u2 |
| 05:12.30 | phix | lolololol |
| 05:12.43 | phix | <3 |
| 05:12.46 | russellb | i can haz voip/ |
| 05:12.55 | phix | heh |
| 05:13.09 | drmessano | I do |
| 05:13.13 | phix | urinmaloop |
| 05:13.19 | phix | lolcats programming language <3 |
| 05:13.49 | *** join/#asterisk imcdona (i=imcdona@imcdona.broker.freenet6.net) |
| 05:17.10 | Nasra | just out for the rest of the night....what you think? |
| 05:17.34 | drmessano | what you say?? |
| 05:17.39 | drmessano | main screen turn on?? |
| 05:17.46 | *** join/#asterisk ftp4 (n=ftp3@68-26-134-148.area1.spcsdns.net) |
| 05:18.00 | Nasra | hi drmessano |
| 05:18.09 | drmessano | We get signal? |
| 05:18.37 | drmessano | How are you gentlemen? |
| 05:18.44 | drmessano | All your base are belong to us? |
| 05:19.02 | ftp4 | any of you guys use a2billing? |
| 05:19.21 | drmessano | No, we have no chance to survive. Make your time. |
| 05:20.35 | russellb | quack |
| 05:21.00 | ftp4 | in case someone has a quick answer for me :-) i installed a2billing, but it seems that "call-labs.com" is the default domain in there for lots of stuff.. and I am having a heck of a time finding the file(s) it is in.. anyone have a clue for me? |
| 05:21.28 | russellb | grep? |
| 05:21.29 | russellb | shrugs |
| 05:22.06 | drmessano | Only guessing, probably not a "file" |
| 05:22.11 | drmessano | Likely something in SQL |
| 05:22.31 | drmessano | Which would correspond to a checkbox |
| 05:22.35 | drmessano | or the like |
| 05:23.14 | drmessano | Unless there's an /etc/a2billing/zomgwtf.conf |
| 05:23.45 | drmessano | or it's something you have lacked defining in their config file, so it's using a hard coded default |
| 05:24.10 | *** join/#asterisk devhen (n=devhen@66.236.68.230.ptr.us.xo.net) |
| 05:25.25 | *** part/#asterisk ball (n=ball@216.176.80.183) |
| 05:26.59 | *** join/#asterisk chaozer (n=snurresp@c83-254-163-148.bredband.comhem.se) |
| 05:28.54 | ftp4 | found it. Thanks ;-) |
| 05:29.06 | *** join/#asterisk TrentCreek (n=kvirc@ppp-70-249-225-174.dsl.hrlntx.swbell.net) |
| 05:30.02 | TrentCreek | "is anyone out there?...." |
| 05:30.50 | TrentCreek | wake up |
| 05:31.36 | chaozer | i have a question ..... sometimes when someone calls into asterisk i sometimes forward the calls to an external number .... .. on the external phone however I ofcourse see the number of the outgoing trunk instead of the number that called into asterisk..... is there a way around this ? |
| 05:32.22 | chaozer | TrentCreek, good morning to you aswell ;) |
| 05:32.33 | frogonwheels | chaozer: I kinda hope not |
| 05:32.34 | TrentCreek | That is because it is sending your CID programmed into * on that truck |
| 05:32.36 | chaozer | just woke up :p |
| 05:32.41 | TrentCreek | chaozer: grassyass |
| 05:33.16 | TrentCreek | chaozer: You bette get Pepsi Max |
| 05:33.25 | chaozer | TrentCreek, yeah i got that.... is there... a way to somehow spoof it ;) ? via SIP ? |
| 05:34.13 | TrentCreek | I am sure there is a way to program it do it, but I am no expert on * scripting or programming |
| 05:35.11 | *** join/#asterisk ManxPower (n=manxpowe@118.sub-75-201-88.myvzw.com) |
| 05:35.28 | TrentCreek | I am sure ManxPower would know |
| 05:37.22 | TrentCreek | I guess you would have to take the value of the incoming CID and assign it to the varible of it going out..but I do not how it handles that. |
| 05:38.04 | chaozer | yeah .. the question is how ;) |
| 05:38.09 | chaozer | TrentCreek, thanks though :) |
| 05:38.57 | chaozer | seems CALLERID(num) can be set.... |
| 05:39.00 | TrentCreek | Maybe the answer is in "The Book" |
| 05:39.26 | TrentCreek | yes, but you would need to know it handles incoming CID. |
| 05:39.52 | TrentCreek | I am sure it is easy |
| 05:39.55 | *** join/#asterisk oilinki (n=oil@ppp-124-121-246-144.revip2.asianet.co.th) |
| 05:40.04 | chaozer | probably :) |
| 05:40.24 | frogonwheels | TrentCreek: somehow, it sounds dangerous - you could then spoof anybody's phone number |
| 05:41.02 | TrentCreek | You can..you can put anyone's number you want in...I have done it before to people who did not want to take my call.. |
| 05:41.03 | chaozer | afaik its possible spoof phone numbers ? :) |
| 05:41.30 | TrentCreek | I put in a number of one of their family members, and BINGO! |
| 05:41.42 | chaozer | yeah me too ... i think i even had some service where they had a #<command> for it ;) |
| 05:42.08 | TrentCreek | However, some companies have some sort of database that checks names and numbers |
| 05:42.51 | TrentCreek | I used to have that AOL Phone line and I tried it..I used the cell number of my ol lady, but my nam ein it,, the AOL Service override my name and put hers in it |
| 05:43.44 | TrentCreek | I am sure CID how CID is handeled is in the book |
| 05:44.12 | TrentCreek | Since you can have a record of call activity |
| 05:44.13 | chaozer | i'll check it out :) |
| 05:44.58 | TrentCreek | I am just too busy right now studying C++, Visual Basic and other internet tecnologies to read about * |
| 05:45.58 | chaozer | neat... shout if you have a c++ question .. :) |
| 05:46.18 | TrentCreek | while on that subject.who knows who to turn on CDR ? |
| 05:46.57 | chaozer | it hought it always was on ? :) |
| 05:47.02 | chaozer | I |
| 05:47.03 | TrentCreek | chaozer: sure..I have to have a small OS written by the morning with some sections written in assembly. |
| 05:47.22 | TrentCreek | No..CDR has to be setup..it goes in MySQL database |
| 05:47.54 | TrentCreek | i know I set it up when I did everything |
| 05:48.00 | TrentCreek | or thought I did |
| 05:49.07 | chaozer | oh youre talking about the logging of CDR ? |
| 05:49.17 | *** join/#asterisk Rico29 (n=Rico@static-120-146.blueline.mg) |
| 05:49.56 | TrentCreek | yes |
| 05:50.19 | chaozer | mine is just logged into some file somewhere |
| 05:50.32 | chaozer | didnt bother with sql just yet |
| 05:50.55 | *** join/#asterisk UD (n=stevo@unaffiliated/underdawg) |
| 05:51.07 | TrentCreek | oh.. |
| 05:51.45 | TrentCreek | I almost have my * server up to where I want it. |
| 05:51.53 | chaozer | but it should be easy |
| 05:52.20 | chaozer | jsut enable in cdr.conf I guess .. and then setup everything in cdr_<whateveryouareusing>.conf. shouldnt that be everything ? :) |
| 05:53.36 | chaozer | iirc i just use the csv stuff anyway right now ;) |
| 05:54.34 | TrentCreek | text files are too slow |
| 05:54.42 | chaozer | I agree ;) |
| 05:54.53 | chaozer | i'll do something about it eventually. |
| 05:56.20 | chaozer | btw.. is anyone using non-t.38 FOIP in here ? :) |
| 05:57.59 | TrentCreek | FIOP? Files over IP? ;-) |
| 05:59.07 | drmessano | There's no such thing |
| 05:59.30 | TrentCreek | sure there is ..I send files over IP everyday :-p |
| 05:59.32 | drmessano | T.38 makes Faxing over VoIP almost somewhat probable |
| 05:59.41 | drmessano | Wasnt talking to you |
| 06:00.01 | TrentCreek | I have sent faxes on VOIP |
| 06:00.13 | TrentCreek | using my * bpx |
| 06:00.15 | drmessano | non-t.38 faxing over VoIP is a unicorn |
| 06:00.18 | TrentCreek | *box |
| 06:00.24 | chaozer | well |
| 06:00.51 | chaozer | i got it working ... 100% on incoming faxes .. pretty much never fails.... atleast not more than on an analogue line |
| 06:01.00 | TrentCreek | just plugged my fax into my ATA and "BrrrrrAhhhhhZZaaaa." |
| 06:01.11 | chaozer | but outgoing faxes .. not a chance ;) pretty much every transmission fails. |
| 06:01.24 | TrentCreek | Every fax I send goes out fine |
| 06:01.28 | chaozer | i guess im wondering why :) |
| 06:01.49 | chaozer | im doing it via asterisk and hylafax + iaxmodem. |
| 06:02.17 | TrentCreek | I'm doing it HP OfficeJet, Linksys ATA, & * |
| 06:02.26 | drmessano | Which ATA? |
| 06:02.32 | TrentCreek | PAP2T |
| 06:02.36 | drmessano | heh |
| 06:02.43 | drmessano | And it's not using T.38? |
| 06:02.56 | TrentCreek | I dont know....I guess I can log in and look |
| 06:03.25 | chaozer | what I dont understand is why it works so well one way .. but fails completely the other way around ;) |
| 06:04.27 | TrentCreek | I don;t see anythign about T.38 |
| 06:04.55 | chaozer | my provider doesnt support t38 :( |
| 06:05.09 | chaozer | they use asterisk actually ;) |
| 06:05.23 | drmessano | Ah, they suck |
| 06:05.37 | *** join/#asterisk L|NUX (n=linux@unaffiliated/lnux/x-10290) |
| 06:05.59 | TrentCreek | The entire range of Linksys adaptors support fax pass-through using the g.711a codec. It is necessary to |
| 06:06.00 | TrentCreek | alter a few settings in the line tab of the web interface of the Linksys adaptor. You only need to alter the line |
| 06:06.00 | TrentCreek | the fax will be plugged into. " |
| 06:06.34 | TrentCreek | I have it set to auto select codec.. |
| 06:07.11 | chaozer | g.711a ... thats alaw ... right ? |
| 06:07.23 | TrentCreek | Network jitter buffer: very high |
| 06:07.23 | TrentCreek | • Jitter buffer adjustment: disable |
| 06:07.23 | TrentCreek | • Call Waiting: no |
| 06:07.23 | TrentCreek | • 3 Way Calling: no |
| 06:07.23 | TrentCreek | • Echo Canceller: no |
| 06:07.24 | TrentCreek | • Silence suppression: no |
| 06:07.26 | TrentCreek | • Preferred Codec: G711a |
| 06:07.28 | TrentCreek | • Use pref. codec only: yes |
| 06:07.34 | drmessano | yay paste spam |
| 06:07.37 | chaozer | TrentCreek, well.. the thing is that its not supposed to work very well ;) |
| 06:07.49 | TrentCreek | yum..spam |
| 06:08.01 | chaozer | any theories on why it works on incoming but not on outgoing for me ? ;) |
| 06:08.11 | TrentCreek | Well it worked for me..I sent many faxes to Mexico where they have crappy lines |
| 06:08.19 | TrentCreek | and have received |
| 06:08.37 | drmessano | Who is your provider? |
| 06:08.51 | chaozer | mine is RixTelecom. |
| 06:08.57 | TrentCreek | Rapid Vox |
| 06:09.29 | TrentCreek | I have no complaints about their service to Land Lines. |
| 06:11.58 | chaozer | maybe iaxmodem is the problem for me .... i honestly didnt try with a real fax machine for outgoing faxes yet ;) |
| 06:13.05 | *** join/#asterisk SuPrSluG (n=SuPrSluG@24.75.47.130) |
| 06:13.46 | *** join/#asterisk centrex (n=centrex@nat/digium/x-e97097b2b17cc829) |
| 06:13.50 | chaozer | i guess maybe its better to just invest in a good faxmodem and an analogue line... |
| 06:13.52 | TrentCreek | I found using REAL fax machines is the only way to go, but who really uses facsimilies anymore? |
| 06:14.20 | chaozer | TrentCreek, this installation is on a law firm ... and well.. they use it alot ;) many faxes daily. |
| 06:14.47 | TrentCreek | chaozer: old timers then,,courts have gone to "eFile" |
| 06:15.01 | chaozer | TrentCreek, not swedish courts :) |
| 06:15.46 | TrentCreek | Oh..and they say the US is a "third world with technology" |
| 06:16.22 | chaozer | TrentCreek, they main reason they dont send it with email f.ex is because they use the confirmation receipt from the fax as "proof" for sendouts. |
| 06:17.03 | TrentCreek | It is not done by email here either...courts's web site |
| 06:17.09 | chaozer | ah :) |
| 06:17.19 | TrentCreek | Login...Click Upload...done |
| 06:17.26 | chaozer | yeah that would probably be better ;) |
| 06:17.39 | chaozer | TrentCreek, the courts here in sweden actually have such a system called VERA... |
| 06:17.56 | TrentCreek | Verboten.. es ... |
| 06:17.57 | chaozer | TrentCreek, but for some reason its just used between courts.. |
| 06:18.09 | TrentCreek | secret stuff |
| 06:19.22 | chaozer | imo it would be nice if facsimilies just went away tomorrow ;) |
| 06:19.35 | chaozer | it refuses to die |
| 06:20.41 | TrentCreek | yeah..I think Teletypes finally died in the 90s in Europe |
| 06:21.55 | *** join/#asterisk imcdona (i=imcdona@imcdona.broker.freenet6.net) |
| 06:22.10 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 06:25.32 | TrentCreek | Oh, or you did you call them "Teleprinters?" |
| 06:26.47 | *** join/#asterisk mrva (n=marv@gw129.klimnet.org) |
| 06:30.09 | jameswf-home | anyone have a non freepbx agi for gizmo? |
| 06:31.09 | *** join/#asterisk trnzmeta (n=bleh@secure27.lnk.telstra.net) |
| 06:36.38 | TrentCreek | nope |
| 06:38.52 | *** join/#asterisk mort_gib (n=mjensen@16.Red-83-36-63.staticIP.rima-tde.net) |
| 06:56.47 | jameswf-home | yay sip:16416418004664411@sip.tollfreegateway.com <free goog411 via sip |
| 06:58.54 | TrentCreek | i will cal it and run al your credit out |
| 07:03.30 | jameswf-home | i has no credit |
| 07:04.41 | TrentCreek | darn |
| 07:11.31 | *** join/#asterisk smooth_penguin (n=smooth_p@59.95.55.226) |
| 07:14.55 | *** join/#asterisk disco_y2k (n=ddd@rrcs-97-77-178-239.sw.biz.rr.com) |
| 07:21.07 | mchou | jameswf-home: there is no need for all that junk. You know sipbroker has free gw to all toll free numbers? |
| 07:22.45 | mchou | jameswf-home: and you can include it in you dial plan so all you have to do is dial 1-8XX-NXX-XXXX and be connected appropriately |
| 07:24.23 | TrentCreek | mchou: yeah, but you have to dial access codes |
| 07:24.43 | mchou | TrentCreek: what access codes??? |
| 07:24.56 | mchou | TrentCreek: I dont dial any access codes |
| 07:25.05 | TrentCreek | " You know sipbroker has free gw to all toll free numbers?" |
| 07:25.32 | mchou | TrentCreek: right. there is no acsess code |
| 07:25.36 | mchou | access* |
| 07:25.51 | mchou | *1800-NXX-NXXXX |
| 07:25.55 | mchou | that's it |
| 07:26.28 | TrentCreek | oh? |
| 07:26.37 | mchou | embed a pattern in you dialplan and just dial 1800-NXX-NXXX. That's it |
| 07:26.52 | mchou | it aint rocket science |
| 07:26.59 | *** join/#asterisk Adrellias (n=random@196.211.85.165) |
| 07:27.22 | Adrellias | hola guys :D sorry to bother got a weird problem on my asterix box |
| 07:27.24 | Adrellias | for some reason |
| 07:27.42 | Adrellias | we keep on getting a unknown number phoning in thats just a loopback test |
| 07:27.47 | TrentCreek | Also, if you don't have access to a VoIP phone, you can dial any one of our PSTN access numbers available in over 20 countries around the world. At the voice prompt, just dial the VoIP number as described above. |
| 07:27.47 | mchou | TrentCreek: you have asterisk and you worried about acess codes? lol |
| 07:28.35 | mchou | access* |
| 07:28.35 | mchou | TrentCreek: I've never used that. |
| 07:28.36 | mchou | TrentCreek: I have no doubt it will be handy for some folks |
| 07:28.49 | TrentCreek | i see no other way on sipbroker on how to use their system besides that |
| 07:28.58 | mchou | TrentCreek: what??? |
| 07:29.00 | mchou | lol |
| 07:29.11 | *** join/#asterisk growltiger_ (n=growltig@ip70-179-54-235.sd.sd.cox.net) |
| 07:29.14 | jblack | One thing that drives me crazy is that cdr logs don't always match monitor file names. |
| 07:29.30 | jblack | date wise |
| 07:29.52 | TrentCreek | SIP Broker is a free service that provides unlimited VoIP calls to millions of people around the world. No signup is necessary. |
| 07:29.56 | mchou | TrentCreek: http://faq.sipbroker.com/tiki-index.php?page=Asterisk+Configuration |
| 07:30.29 | mchou | TrentCreek: stop parroting sipbroker's web page. most people here can read |
| 07:30.37 | Adrellias | So anybody have a idea ? |
| 07:30.42 | Adrellias | about my ghost calls |
| 07:30.51 | drmessano | Void where prohibited by law. |
| 07:30.53 | TrentCreek | Well them tell us where the "free" TOLL FREE numbers are |
| 07:30.57 | jblack | TrentCreek: That free stuff is sipbroker<->sipbroker calls, mostly. |
| 07:30.57 | drmessano | Some restrictions may apply. |
| 07:31.17 | jblack | They're not the only one the does it, they're just the only one that thinks it has market value. :) |
| 07:31.34 | drmessano | I love my free tollfree calls |
| 07:31.37 | drmessano | Very FTW |
| 07:31.48 | mchou | TrentCreek: wtf. http://www.sipbroker.com/sipbroker/action/providerWhitePages;jsessionid=966FE20CD91AE87831F02EFAC9AEC802 |
| 07:31.51 | drmessano | Although I don't use Sipbroker for that |
| 07:32.10 | mchou | search for "800" |
| 07:32.47 | mchou | wondering what people use asterisk for??? |
| 07:32.58 | awk | anyone using shared wrauptime in asterisk 1.4.21? |
| 07:32.58 | mchou | wtf |
| 07:33.03 | drmessano | I use it for Skype |
| 07:33.19 | TrentCreek | list of gateways |
| 07:33.31 | mchou | oh lord |
| 07:33.39 | awk | are you patching your asterisk? |
| 07:33.39 | awk | or it just works? |
| 07:33.55 | mchou | TrentCreek: get on the frigging clue train |
| 07:33.57 | awk | drmessano ? |
| 07:34.11 | drmessano | $6 wal mart phone --> Linksys ATA ---> Asterisk --> SPA3102 ---> Skype Adapter --> Windows Vista Ultimate |
| 07:34.16 | drmessano | Works like a champ |
| 07:34.22 | mchou | drmessano: haha |
| 07:35.01 | mchou | drmessano: $6 Walmart phone doesnt have caller ID :) |
| 07:35.11 | mchou | drmessano: get w/it :) |
| 07:35.17 | TrentCreek | the 8.95 one does |
| 07:35.25 | drmessano | yeah, the $8 GE does |
| 07:35.46 | drmessano | The 6 Emerson has a flashing Neon light on the back :) |
| 07:35.49 | drmessano | $6 |
| 07:36.30 | jblack | flashy lights are the the killer stylistic touch. |
| 07:36.43 | mchou | more like be killed |
| 07:36.59 | jblack | "reach out.. reach out and blind someone!" |
| 07:37.32 | *** join/#asterisk horsesgofaster (n=dcantera@pool-72-82-199-151.cmdnnj.east.verizon.net) |
| 07:37.37 | TrentCreek | i got the 8.95 one two days ago |
| 07:37.56 | *** join/#asterisk suvir (n=suvir@ppp-124-120-246-111.revip2.asianet.co.th) |
| 07:37.58 | drmessano | ~cheap |
| 07:38.01 | jbot | somebody said cheap was 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. |
| 07:38.02 | mchou | TrentCreek: linky pls |
| 07:38.16 | jblack | I have more phones than weekly calls. |
| 07:38.34 | drmessano | I have a sinking feeling about TrentCreek |
| 07:38.43 | TrentCreek | mchou: Wally World site is down |
| 07:38.43 | mchou | drmessano: walmart has pap2s on clearance for $15 |
| 07:38.56 | mort_gib | How do I get caller id on incoming call on a Zap channal ?? |
| 07:39.03 | TrentCreek | mchou: I already inquired about those...Linksys and not easy to crack |
| 07:39.16 | TrentCreek | for Vonage |
| 07:39.23 | mchou | TrentCreek: easy to crack |
| 07:39.37 | TrentCreek | Someone told me the new ones are hard |
| 07:39.45 | mchou | key is not to be hooked to inet b4 cracking |
| 07:40.01 | mchou | then you'd be fscked |
| 07:40.30 | TrentCreek | that's a lot of work just to save $20 for a NEW & Unlocked one |
| 07:40.44 | mchou | TrentCreek: we know it's hard for you :) |
| 07:41.18 | mchou | TrentCreek: don't forget to add shipping & handling |
| 07:41.29 | TrentCreek | did not say hard...just a LOT of work |
| 07:41.30 | drmessano | Version 1 or 2s? |
| 07:41.48 | mchou | drmessano: dunno |
| 07:41.52 | TrentCreek | Version 2....they are the dark plastic bands |
| 07:42.00 | drmessano | There's an important difference |
| 07:42.05 | drmessano | the Version 2s blow |
| 07:42.19 | TrentCreek | And that is what is on clearnance at Wally World |
| 07:42.19 | mchou | drmessano: yeah, but you cant find version 1 new |
| 07:42.30 | drmessano | I know |
| 07:42.38 | mchou | drmessano: only fleabay and the like |
| 07:42.50 | drmessano | Version 2s aren't worth it for me |
| 07:43.16 | drmessano | I got one, hated it |
| 07:43.41 | drmessano | If you could completely pwn it, that would be one thing |
| 07:43.57 | drmessano | But you're dealing with it being "mostly unlocked" |
| 07:44.03 | drmessano | That annoys me |
| 07:44.28 | TrentCreek | Here is the Waly World phone http://www.home-electronics.net/ge/pc/viewPrd.asp?idcategory=32&idproduct=33 |
| 07:44.36 | mchou | bah. wrtp54g's can be completely unlocked |
| 07:44.58 | TrentCreek | @ 8.95 |
| 07:45.11 | mchou | but they take up too much space |
| 07:45.31 | drmessano | I got enough PAP2 v1s to last me a while |
| 07:45.32 | mchou | damn, that phone is fugly |
| 07:46.15 | mchou | I got uniden dect 3 handset for $15 |
| 07:46.31 | TrentCreek | mchou: the color is a bit diferent now. |
| 07:46.36 | mchou | much prefer that over crappy GE |
| 07:46.51 | mchou | TrentCreek: lol. That's hardly the issue |
| 07:46.55 | TrentCreek | the buttons are black as well as the area aroudn the scren |
| 07:46.59 | drmessano | I'm not a uniden fan |
| 07:47.10 | TrentCreek | and it looks better than that $6 Wally World phone |
| 07:47.20 | drmessano | GE is hit and miss.. I've had some really good ones, and some really bad ones |
| 07:47.33 | mchou | drmessano: why is that? uniden SIP suck, but their DECTS are good |
| 07:47.41 | chaozer | btw.. im experiencing some very weird behaviour with queues here when ringinuse=no ... sometimes when the user finishes a call the next call in the queue never gets through..... in the CLI however it says "ringing" on the specific phone... even though it isnt ringing.... |
| 07:47.51 | mchou | of course I'd prefer panasonics |
| 07:48.15 | chaozer | i managed to "fix" it by setting ringinuse=yes and raise all the delays, wrapuptime etc... |
| 07:48.19 | mchou | panasonic and philips use standard AAA battries now |
| 07:48.32 | chaozer | anyone else experienced something similar ? ;) |
| 07:48.43 | mchou | to me that makes real good sense |
| 07:49.14 | drmessano | I have never been a fan of Uniden phones.. Just thought they did a lousy job with them.. Scanners, yes.. phones, NAY |
| 07:51.51 | jblack | I second that. |
| 07:52.02 | jblack | Great CB radios. |
| 07:52.18 | TrentCreek | CB simulator |
| 07:54.00 | *** join/#asterisk _gm (n=gmustafa@202.133.78.60) |
| 07:58.20 | TrentCreek | JUst order the MagicJack and we wont need Asterisk! |
| 07:58.54 | jblack | They play that annoying, 2 minute, commercial on bloomberg about 12 times a day. |
| 07:59.22 | TrentCreek | lol |
| 07:59.50 | jblack | winces through the pedipaws commercial. |
| 07:59.51 | TrentCreek | like it's Home Shopping Network |
| 08:00.08 | jblack | Yeah, that's the one. WIth that silly timer, and "units sold" |
| 08:00.49 | TrentCreek | http://www.youtube.com/watch?v=bsR3b2F9sUA |
| 08:01.08 | jblack | I was thinking about writing magicjack "Hey! I love your home shopping show, but y u keep showing the same magicjack? 50 times, and you still ain't sold 10,000 of them" |
| 08:01.14 | *** join/#asterisk gr0mit (n=tim@dhcp3.zuk40.mot-tools.co.uk) |
| 08:01.51 | mchou | what I dont understand is how majic jack has money for advertising on TV |
| 08:02.01 | mchou | magic* |
| 08:02.24 | mchou | what do they charge? $10? |
| 08:02.39 | TrentCreek | There was an expose on those TV ads last year |
| 08:03.00 | mchou | TrentCreek: what did they say? |
| 08:03.03 | TrentCreek | The iventors make almost NOTHING while the promoters rake in all the dough |
| 08:03.26 | mchou | what else is new |
| 08:03.31 | TrentCreek | hehe |
| 08:04.15 | TrentCreek | i think the guy who invented the Thighmaster got around $3000, while the promoters rakes in millions. |
| 08:07.37 | TrentCreek | who besides VoxBone offers SMS? |
| 08:09.09 | mchou | SMS kills |
| 08:09.39 | *** join/#asterisk PaulQ (n=PaulTech@dc2.ssh.dimenoc.com) |
| 08:09.39 | TrentCreek | why |
| 08:09.59 | PaulQ | Quick question I think.... Asterisk 1.4.22, Getting 401 on any RealTime SIP peer |
| 08:10.05 | PaulQ | Normal flat file loads fine |
| 08:10.11 | PaulQ | But any peer in realtime, gets 401 |
| 08:10.18 | PaulQ | Am I over looking anything |
| 08:13.05 | TrentCreek | i dotn know what 401 is |
| 08:13.42 | PaulQ | Unauthorized |
| 08:15.21 | TrentCreek | then authentication |
| 08:16.05 | TrentCreek | on the SIP clients |
| 08:16.38 | PaulQ | Huh |
| 08:19.30 | PaulQ | yeaaa |
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| 08:36.29 | *** join/#asterisk baxtas (n=buzukeli@ip-195-14-171-9.bnk.lt) |
| 08:38.38 | baxtas | Hi all, does anybody know how : call multiple extensions, but when one is answered, other stil ring ??? |
| 08:39.17 | baxtas | is there dialplan function can do that ?? |
| 08:41.12 | *** join/#asterisk r0land (n=roland@193.227.191.91) |
| 08:41.14 | r0land | hi all |
| 08:41.38 | baxtas | hi |
| 08:41.47 | r0land | im facing a certain prob with my dialplan |
| 08:41.50 | r0land | http://pastebin.com/m1c0afffa |
| 08:41.53 | r0land | heres the CLI output |
| 08:41.57 | r0land | any suggestion about wht im doing wrong ?! |
| 08:45.34 | baxtas | do you know how : call multiple extensions, but when one is answered, other stil ring ??? |
| 08:47.01 | r0land | tht could b done with priorities! |
| 08:49.25 | *** join/#asterisk Corydon76-dig (i=peuce@pdpc/supporter/bronze/Corydon76-home) |
| 08:49.25 | *** mode/#asterisk [+o Corydon76-dig] by ChanServ |
| 08:51.43 | baxtas | same priorit number? it will ring same time? i need conference. but meetme application doesn't what i need. |
| 08:54.48 | PaulQ | What do you want to be on the other end of the channel? |
| 08:54.52 | bkw_ | baxtas: are you saying you want the other phones to keep ringing after its answered? |
| 08:55.06 | bkw_ | so you call 10 phones.... one answers.. you want the other 9 to keep ringing? |
| 08:55.28 | PaulQ | and what happens when the other 9 answer |
| 08:55.35 | PaulQ | Are they going into a MeetMe? |
| 08:55.36 | bkw_ | well the request is not very logical |
| 08:55.44 | bkw_ | their would be no reason to actually do that |
| 08:55.45 | PaulQ | I think we are talking to each other |
| 08:55.47 | bkw_ | because at some point the invite will time out |
| 08:55.52 | PaulQ | si |
| 08:56.07 | PaulQ | bkw, You work with realtime much? |
| 08:56.33 | bkw_ | PaulQ: we wrote most of what was realtime in the start. |
| 08:56.45 | bkw_ | but I don't use it anymore |
| 08:57.08 | PaulQ | Ahh |
| 08:57.13 | bkw_ | PaulQ: I work pretty much full time on FreeSWITCH now... anyway I have to go back to bed.. me tired |
| 08:57.23 | PaulQ | Ohhh |
| 08:57.24 | baxtas | then how call to meetme and make other extension call to meetme too??? |
| 08:57.24 | PaulQ | Enjoy |
| 08:57.26 | r0land | hey every1 |
| 08:57.33 | r0land | coudl any1 advise me whts up with my dial plan! |
| 08:57.36 | r0land | heres my CLI output |
| 08:57.40 | r0land | http://pastebin.com/m1c0afffa |
| 09:00.42 | PaulQ | Grrr |
| 09:00.44 | PaulQ | Fixed my problem |
| 09:00.49 | PaulQ | but im pissed at why it was happening |
| 09:00.54 | PaulQ | let me look r0land |
| 09:03.44 | r0land | PaulQ thank you |
| 09:04.19 | r0land | PaulQ if u notice, it says tht channel 300 is busy, though when i call channel 300 directly it rings and answrs..! |
| 09:10.02 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 09:11.03 | baxtas | I fixed my problem. solution was : n-way call. |
| 09:12.30 | *** join/#asterisk fiddur (n=fiddur@c042.rit.se) |
| 09:26.08 | r0land | PaulQ any advice? |
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| 09:54.57 | Touns | hi :) |
| 09:56.11 | Touns | please, I have a modem an olitec self memory 2000. I try to use it with hylafax to become fax but i was not able to configure my modem with hylafax. Are asterisk able to receive fax and handle V90 modem ? |
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| 11:34.43 | fiddur | Hi. Is there any page with gotch'as for upgrading 1.4 -> 1.6.0-rc6? |
| 11:35.34 | russellb | fiddur: read the UPGRADE.txt file in 1.6 |
| 11:36.04 | fiddur | http://svn.digium.com/view/asterisk/branches/1.6.0/UPGRADE-1.4.txt?view=markup ? thanks |
| 11:36.31 | russellb | no |
| 11:36.35 | russellb | just UPGRADE.txt |
| 11:36.39 | russellb | not UPGRADE-1.4.txt |
| 11:37.02 | fiddur | ahh, thought 1.4 was about upgrading from 1.4... |
| 11:37.35 | russellb | nah, that's the file from 1.4 which documents stuff from 1.2->1.4 |
| 11:37.55 | fiddur | Yeah, I saw that once I opened UPGRADE.txt :) |
| 11:38.04 | russellb | cool |
| 11:38.22 | fiddur | I'm getting tired of getting backports of the stuff I want to use :P |
| 11:38.35 | fiddur | Would you say 1.6.0 is getting stable enough for production systems? |
| 11:38.53 | russellb | I will not officially say that until it's officially released :) |
| 11:39.04 | russellb | but i can say that we're ready to release 1.6.0 |
| 11:39.11 | russellb | we're just waiting on DAHDI to get released first ... |
| 11:40.05 | lmadsen | russellb: omghi! |
| 11:40.17 | russellb | lmadsen: omghi2u2 |
| 11:40.21 | russellb | why are we up? :( |
| 11:40.26 | lmadsen | russellb: happy phoenix day! :) |
| 11:40.32 | fiddur | Hmm, but I can use Zaptel for 1.6.0 I hope; I don't have to wait for dahdi? |
| 11:40.37 | lmadsen | I will be seeing you in TWO DAYS! |
| 11:40.51 | russellb | fiddur: nope. 1.6.0 only supports DAHDI |
| 11:41.03 | russellb | lmadsen: zomg |
| 11:41.22 | fiddur | russellb: Ah... sounds like a big gotcha :) |
| 11:41.24 | lmadsen | there are DAHDI release candidates you can use to play with / test your system on in order to make sure things at least load and do nice things |
| 11:41.33 | lmadsen | russellb: zomg indeed! |
| 11:41.48 | lmadsen | russellb: you are up VERY early (or you're still up) |
| 11:41.58 | lmadsen | must still be a bit jet lagged |
| 11:41.59 | russellb | woke up and couldn't get back to sleep |
| 11:42.04 | russellb | jet lagged? |
| 11:42.08 | russellb | it's only 2 hours off, heh |
| 11:42.12 | russellb | and i've been here since last sunday |
| 11:42.16 | lmadsen | ya, same here. Sent the lady friend off to work, and decided to stay up |
| 11:42.24 | lmadsen | russellb: shhhhhhhhh, I'm giving you an our |
| 11:42.26 | lmadsen | out* |
| 11:42.38 | russellb | excited about the weekend is part of it |
| 11:42.43 | lmadsen | fair enough |
| 11:42.50 | lmadsen | I'm excited for astricon and astridevcon |
| 11:42.58 | lmadsen | russellb: this will be my FIRST astridevcon! |
| 11:43.00 | russellb | yes, that too :-p |
| 11:43.04 | russellb | lmadsen: w00t |
| 11:43.13 | lmadsen | which is funny since i"ve been to every single astricon :) |
| 11:44.02 | lmadsen | I'm really wishing I could turn my speakers on remotely now... |
| 12:09.28 | *** join/#asterisk Laureano (n=Laureano@190.245.108.2) |
| 12:09.44 | henk | there is probably something wrong with my config if i see this regularly: "Spawn extension (SOMETHING) exited non-zero on 'SOMETHINGELSE'" or is that normal? |
| 12:11.25 | *** join/#asterisk [TK]D-Fender (n=joe@216.191.106.163) |
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| 12:24.06 | EmleyMoor | Using 1.4.21.2 on Debian, is there any way I can install a low bandwidth codec that the N95 also supports? |
| 12:24.49 | [TK]D-Fender | EmleyMoor: And what codecs does the N95 support? |
| 12:25.07 | EmleyMoor | In addition to G.711, AMR, iLBC, G729 and CN |
| 12:25.33 | [TK]D-Fender | EmleyMoor: And ILBC & G.729 is it |
| 12:26.14 | EmleyMoor | They look like the only really viable choices |
| 12:26.47 | henk | there is probably something wrong with my config if i see this regularly: "Spawn extension (SOMETHING) exited non-zero on 'SOMETHINGELSE'" or is that normal? |
| 12:26.49 | merlinn | no matter which codec I use I don't see a massive dip in network bandwidth utilization |
| 12:27.07 | merlinn | not without using custom doohickeys to compress and rip out control traffic etc. |
| 12:27.21 | EmleyMoor | In this case, I'm actually looking more to hear the difference |
| 12:27.33 | merlinn | in what way? |
| 12:27.41 | EmleyMoor | Less dropped audio |
| 12:27.53 | merlinn | by dropped |
| 12:28.08 | merlinn | you mean you want the traffic to F up less |
| 12:28.13 | EmleyMoor | Yes |
| 12:28.25 | merlinn | is this wifi |
| 12:28.28 | [TK]D-Fender | henk: Just says your call is ending. |
| 12:28.30 | merlinn | or 3g or something |
| 12:28.51 | EmleyMoor | 3G. Over WiFi it's mostly fine |
| 12:29.15 | henk | [TK]D-Fender: ok, why is that 'non-zero' then? isn't non-zero usually a return code to indicate an error? |
| 12:29.21 | merlinn | I think you're fighting a losing battle to be honest |
| 12:29.47 | [TK]D-Fender | henk: go look at everything else thats happening in your call. Is it doing everything it should? |
| 12:29.48 | merlinn | I've given up on 3g completely |
| 12:29.55 | merlinn | not because of raw bandwidth throughput |
| 12:30.01 | *** join/#asterisk esaym (n=user@cpe-70-120-89-6.satx.res.rr.com) |
| 12:30.04 | merlinn | but more because of the typical latency experienced |
| 12:31.40 | henk | [TK]D-Fender: it is. i just wanted to know whether that means that the extension didn't exit normally and didn't clean up after itself or something... |
| 12:32.02 | [TK]D-Fender | henk: No, non-zero exit just indicates a hangup IIRC |
| 12:32.33 | henk | [TK]D-Fender: alright, thanks :) |
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| 12:40.46 | EmleyMoor | Dial *always* exits non-zero AIUI |
| 12:43.26 | [TK]D-Fender | EmleyMoor: "g" <- exception |
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| 12:43.42 | [TK]D-Fender | EmleyMoor: Or any failure to set up the call. |
| 12:48.07 | EmleyMoor | Any hints on what to do to recompile asterisk for Debian with ilbc added>? |
| 12:49.19 | [TK]D-Fender | EmleyMoor: There are instruction in your source tarball. Follow them |
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| 13:23.03 | pif | tzafrir_laptop: hi, |
| 13:23.40 | pif | tzafrir_laptop: debian/unstable's latest * packs fail with: 1. You only have to compile Zaptel support into Asterisk if you need it. One option is to recompile without Zaptel support." |
| 13:27.02 | *** join/#asterisk soulfreshner (n=derick@dsl-243-53-129.telkomadsl.co.za) |
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| 13:28.14 | soulfreshner | is there some script that can process an asterisk cdr csv file into a usage summary? |
| 13:28.29 | soulfreshner | using a db would be overkill in this case |
| 13:29.02 | *** join/#asterisk AndyML (n=alauppe@pool-71-185-74-86.phlapa.fios.verizon.net) |
| 13:29.22 | [TK]D-Fender | soulfreshner: I don't know of any offhand, but go look on the WIKI GUI list to see if one there supports CSV |
| 13:30.00 | AndyML | i'm running asterisk 1.4.21.2 and zaptel 1.4.11 on ubuntu 8.04.1 - i don't have any zap hardware, and I can't get ztdummy to work |
| 13:30.20 | soulfreshner | thanks [TK]D-Fender |
| 13:30.33 | AndyML | when i "zap show channels" |
| 13:30.48 | AndyML | it shows the pseudo channel |
| 13:31.00 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 13:31.14 | [TK]D-Fender | AndyML: that should be ztdummy... |
| 13:31.16 | AndyML | but when i run zap show status, it doesn't give me anything |
| 13:31.34 | [TK]D-Fender | AndyML: What happens when you actually try to use it? |
| 13:31.55 | AndyML | i'm trying to use it as a timing source for conferences. it doesn't let me start meetme |
| 13:32.30 | [TK]D-Fender | soulfreshner: An alternative it to use ODBC and use a CSV wrapper, or import it into something else that has an easy connector |
| 13:32.41 | [TK]D-Fender | AndyML: Did you recompile * after installing Zaptel? |
| 13:32.59 | AndyML | i didn't compile asterisk until after zaptel was compiled. |
| 13:33.08 | *** join/#asterisk overworked554 (n=kenwiesn@12-201-81-231.client.mchsi.com) |
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| 13:34.26 | jameswf-home | agi_extension = ${EXTEN}? |
| 13:34.54 | soulfreshner | [TK]D-Fender, I think the easiest with the tools I have available would probably be to use the sqlite addon |
| 13:35.12 | overworked554 | hello, does anyone know what would cause ast 1.6 not to record a cdr when using a callfile? i am using context/exten/prio instead of app/data |
| 13:35.17 | soulfreshner | but I was hoping to get around that :( |
| 13:35.36 | soulfreshner | for this client it would be more than adequate to just mail usage stats once a week and rotate the cdr log |
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| 13:40.49 | [TK]D-Fender | soulfreshner: Its your solution and you seem to know the options. I'm sure you'll find an acceptably simple comprimise. Mind you you might seriously want to offer them one of those WEB CDR viewers.... |
| 13:42.16 | AndyML | I needed to do 'modprobe zaptel' and 'modprobe ztdummy'.... |
| 13:44.32 | tzafrir_laptop | pif, what packages exactly? A new version of zaptel has been uploaded to unstable yesterday |
| 13:44.32 | *** join/#asterisk RediXe (n=steven@pool-71-114-236-60.austtx.dsl-w.verizon.net) |
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| 13:46.05 | pif | tzafrir_laptop: yea, I've installed it, and with * 1:1.4.21.2~dfsg-1+b1 it fails |
| 13:46.24 | [TK]D-Fender | AndyML: Good idea |
| 13:46.47 | tzafrir_laptop | pif, could you please be more specific about "fail"? What exactly fails? |
| 13:47.02 | pif | bug #491310 is still there |
| 13:47.31 | pif | which I reported on 18th july |
| 13:47.41 | *** join/#asterisk stencil (n=stencil@206-248-152-232.dsl.teksavvy.com) |
| 13:48.27 | RediXe | Any idea if Asterisk can turn blackberry's into office phone's? Basically, everyone in our office is given a Blackberry and we want it to function like an office phone. There are a few commercial solutions I am just curious if Asterisk could do it or not. |
| 13:49.04 | merlinn | it's not really a case of making asterisk do it |
| 13:49.19 | pif | blackberrys can't be 'turned' into anything by asterisk |
| 13:49.24 | merlinn | asterisk is just a piece of software that routes calls |
| 13:49.26 | tzafrir_laptop | pif, it's a bug of asterisk and not of zaptel |
| 13:49.42 | tzafrir_laptop | pif, do you have any zaptel card? |
| 13:49.48 | pif | yes, the new zaptel-source works fine, and compiles with 2.6.26 |
| 13:49.54 | merlinn | you ucan get software for symbian that will install an application on your blackberry |
| 13:50.03 | merlinn | that converse with a third party sip proxy |
| 13:50.07 | pif | tzafrir_laptop: I do, vzaphfc |
| 13:50.21 | merlinn | and will turn the blackberry into an extension on whatever voip system you're using |
| 13:50.51 | merlinn | it will also route calls over wireless/gprs seamlessly as you roam in/out of signal |
| 13:51.27 | RediXe | merlinn, hmmmm I really have no idea as I'm new to this but I'll look into symbian. |
| 13:51.34 | merlinn | dude |
| 13:51.40 | merlinn | symbian is the software that runs on your blackberry |
| 13:51.45 | merlinn | it's the OS |
| 13:52.03 | pif | tzafrir_laptop: with asterisk 1:1.4.19.1~dfsg-1 works fine |
| 13:52.15 | RediXe | merlinn, lol really? never heard of it. |
| 13:52.34 | merlinn | http://www.vcomm.co.uk/comdasys.php |
| 13:52.35 | merlinn | look at that |
| 13:52.39 | RediXe | merlinn, but I haven't had my BB but a few days |
| 13:52.43 | *** join/#asterisk Aquahallic (n=rich@CPE001310455a97-CM001ac3152b2a.cpe.net.cable.rogers.com) |
| 13:52.48 | [TK]D-Fender | RediXe: Got WiFi or unlimited data on those BB's? |
| 13:52.59 | *** join/#asterisk ToTo (n=ToTo@209.8.41.42) |
| 13:53.02 | Aquahallic | Mornin' folks |
| 13:53.12 | RediXe | [TK]D-Fender, They have WiFi and unlimited data plan. |
| 13:53.13 | pif | tzafrir_laptop: could it be the bristuff patch ? |
| 13:53.17 | overworked554 | RediXe, if you are talking about the 7270s which are sip based, you could use asterisk as your pbx. if not then what you are looking for is a virtual pbx. |
| 13:53.26 | [TK]D-Fender | RediXe: ok, go look for a BB SIP client |
| 13:53.26 | tzafrir_laptop | pif, no |
| 13:53.33 | Aquahallic | if I'm building an asterisk server that's purely IP based... do I need Zaptel and Libpri? |
| 13:53.36 | pif | tzafrir_laptop: you still apply it? |
| 13:53.49 | tzafrir_laptop | pif, I can disable this test at startup |
| 13:54.00 | [TK]D-Fender | Aquahallic: Zaptel yes, if you want MeetMe or IAX2 trunking. Otherwise no |
| 13:54.09 | [TK]D-Fender | Aquahallic: No need period for libpri |
| 13:54.19 | pif | tzafrir_laptop: because vzaphfc is bristuff material |
| 13:54.42 | Aquahallic | libpri is just for my PRI's like if I was using some T1's or some other h/w based PRI.. correct? |
| 13:54.46 | soulfreshner | [TK]D-Fender, which web cdr viewer would you recommend? |
| 13:55.12 | [TK]D-Fender | soulfreshner: Don't use any of the common ones. I heard Areski is pretty good |
| 13:55.21 | [TK]D-Fender | Aquahallic: yup |
| 13:55.28 | Aquahallic | ty...;) |
| 13:55.39 | [TK]D-Fender | soulfreshner: **I** Don't use any of the common ones. I heard Areski is pretty good |
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| 13:59.01 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 13:59.54 | tzafrir_laptop | pif, what architechture do you use? |
| 13:59.57 | EmleyMoor | It looks like Debian meant it when they removed ilbc - there isn't even a way, having fetched the source code, to use it. |
| 14:00.14 | pif | tzafrir_laptop: amd64 |
| 14:02.13 | *** join/#asterisk fordfrog (n=fordfrog@gentoo/developer/fordfrog) |
| 14:04.35 | tzafrir_laptop | EmleyMoor, you don't have the asterisk source tree, but you have the includes, and that should mostly do it |
| 14:05.07 | soulfreshner | I was just checking out Areski earlier :) it will be my next choice - but I need to run mysql PHP and a web server to use it |
| 14:05.25 | *** join/#asterisk l2trace99 (n=jr@static-71-251-65-16.tampfl.fios.verizon.net) |
| 14:05.26 | soulfreshner | and the PC is not very powerful |
| 14:05.37 | EmleyMoor | tzafrir_laptop: Agreed it should - the little bit it is doing amounts to it doing nothing though :-( |
| 14:05.52 | soulfreshner | thanks for the suggestion, though - I might end up using it anyway |
| 14:06.50 | tzafrir_laptop | basically try building it with: gcc -shared -Wall -DAST_MODULE=\"codec_ilbc\" -o codec_ilbc.so codec_ilbc.c |
| 14:07.48 | tzafrir_laptop | there should be a script in the asterisk source directory to fetch it. I wonder if it would work stand-alone |
| 14:08.21 | EmleyMoor | I've fetched the tree. Having done so, you are supposed to bring it in with menuselect - it's not in there to be brought in |
| 14:10.04 | tzafrir_laptop | hmmm... that ilbc/ subdirectory will take some extra tweaking to build out of the asterisk tree |
| 14:10.18 | *** join/#asterisk Zeeek (n=IceChat7@pdpc/supporter/active/Zeeek) |
| 14:11.42 | EmleyMoor | I'm tempted to abandon this and just put up with alaw |
| 14:11.42 | tzafrir_laptop | gcc -shared -Wall -DAST_MODULE=\"codec_ilbc\" -o codec_ilbc.so codec_ilbc.c ilbc/libilbc.a |
| 14:11.51 | tzafrir_laptop | speex! |
| 14:11.58 | *** join/#asterisk jpeeler (n=jpeeler@asterisk/digium-software-dev/jpeeler) |
| 14:12.09 | EmleyMoor | The phone I want it for doesn't support speex |
| 14:12.22 | Zeeek | Friday! |
| 14:13.52 | EmleyMoor | I just can't see any way to do it |
| 14:13.52 | overworked554 | is there a way to force a cdr to be generated when you place a call with a callfile? |
| 14:14.45 | *** join/#asterisk arpu (n=arpu@chello084113208186.3.14.vie.surfer.at) |
| 14:17.33 | *** join/#asterisk anonymouz666 (n=anonymou@201.19.95.130) |
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| 14:21.02 | anonymouz666 | if app_queue.so uses mixmonitor to record calls, what happens if I use monitor-format = g729 in queues.conf? |
| 14:24.49 | *** join/#asterisk propellerhead (n=yogurt2u@host38.190-137-2.telecom.net.ar) |
| 14:24.58 | RediXe | [TK]D-Fender, Well, guess I'll figure something else out. Doesn't appear to be a SIP client for the Blackberry 8320. |
| 14:25.23 | bkw_ | anonymouz666: you'll use the hell out of the g729 licenses :P |
| 14:27.14 | anonymouz666 | bkw_: that's correct, but if it works I guess the queues.conf.samples should be something like: ; monitor-format = gsm|wav|wav49|g729|g726|alaw|ulaw|etc |
| 14:27.25 | anonymouz666 | to pass the idea you can use whatever format you want. |
| 14:27.38 | bkw_ | well mix monitor will have to convert to slin to mux the call first |
| 14:27.54 | bkw_ | you can't mux g729 native |
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| 14:29.32 | Zeeek | Moosebreath |
| 14:29.50 | Zeeek | the prophecy has come true |
| 14:29.56 | Aquahallic | A while back I installed AsteriskNow and noticed that some of the HowTo docs I read, which spec'd to modify certain config files, that the AsteriskNow box seemed to be editing different config files... this time I installed a base server OS and installed asterisk by compiling (not AsteriskNow) and then installed the Asterisk Gui 2.0.... does this gui modify the normal config files... or will it modify the same "d |
| 14:29.57 | anonymouz666 | if the call is established as ulaw and the monitor-format is g726 then the process will be: g711 -> slin -> g726 ? |
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| 14:30.40 | *** mode/#asterisk [+o Corydon76-dig] by ChanServ |
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| 14:41.11 | anonymouz666 | if the call is established as g729 and I am recording as g729 so there's no need to convert to slin first, is this correct? |
| 14:41.50 | *** join/#asterisk RB2 (n=RB2@pool-71-125-95-246.nwrknj.east.verizon.net) |
| 14:42.37 | [TK]D-Fender | anonymouz666: Anything spying on the channel will, and I'm not certain on recording. I doubt it... |
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| 14:46.19 | anonymouz666 | I am not certain either |
| 14:47.54 | BrianR___ | Hmm. Aparently there's no answer confirmation with agentcallbacklogin :( |
| 14:48.04 | BrianR___ | Did I misconfigure something? |
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| 14:52.12 | EmleyMoor | I've stepped back my asterisk now - chances are, if I can do what I want, I won't need a full recompile |
| 14:53.10 | Zeeek | I called a friend on his asterisk box. He surprised me by answering on his Nokia N95 from another country! Sounded great |
| 14:53.28 | EmleyMoor | Zeeek: In theory possible |
| 14:53.38 | EmleyMoor | At least on WiFi... |
| 14:53.40 | Zeeek | it happened! |
| 14:53.51 | Zeeek | I was impressed by what I had always heard was flaky |
| 14:54.14 | EmleyMoor | Not great on 3G but sometimes usable |
| 14:54.32 | Zeeek | it was wifi athis dad's place in France. Sounded like next door |
| 14:54.37 | brodiem | I've been looking for feedback on nokia's SIP stack for a while |
| 14:54.50 | brodiem | has anyone tested it over the 3g network as opposed to WLAN? |
| 14:55.19 | Zeeek | my problem is I don't want to pay for 3g or any other cell provider. However, wifi is free (mostly) |
| 14:55.42 | Zeeek | I may consider buying his N95 as he's getting some newer toy |
| 14:55.54 | brodiem | EmleyMoor: oh, you have? Was it US (AT&T)? |
| 14:57.34 | *** join/#asterisk bbryant (n=brett@68.208.65.50) |
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| 15:02.54 | [TK]D-Fender | BrianR___: Agentlogin asks for confirmation because you're on the call. the other CALLS your phone. The fact you picked up = confirmation. |
| 15:03.40 | *** join/#asterisk meuserj (n=meuserj@indianalifesciences.com) |
| 15:04.00 | *** join/#asterisk neurosys (n=neurosys@adsl-067-032-132-060.sip.bct.bellsouth.net) |
| 15:04.08 | *** join/#asterisk tobias (n=tobias@user-0c2hj2f.cable.mindspring.com) |
| 15:04.14 | BrianR___ | [TK]D-Fender: No... I was talking about agentcallbacklogin. Turns out I fat-fingered the ackcall option in agents.conf |
| 15:05.20 | BrianR___ | [TK]D-Fender: sometimes folks log into the queue from their cell phones or whatever and I want to make sure the calls don't go to voicemail |
| 15:06.06 | meuserj | ok.. I'm trying to write an AGI script using ruby and batphone (http://hans.fugal.net/src/batphone/doc/files/README.html) and it doesn't seem to be executing... what user are agi scripts typically run as? |
| 15:06.16 | [TK]D-Fender | BrianR___: taht you'll have to control in the way your dial them. Make sure to REALLY read Dial's insructions. |
| 15:06.39 | [TK]D-Fender | meuserj: The user * is running as |
| 15:07.25 | *** join/#asterisk ddunavant (n=David@75.145.240.14) |
| 15:07.37 | [TK]D-Fender | meuserj: What are you planning to9 do in AGI? |
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| 15:10.20 | meuserj | [TK]D-Fender: well, specifically for this script, I plan to have it download an iCal file and check the date to see if it is a day off for the company and play a message for incoming calls.... but I'm mostly just wanting to get ruby agi scripting working so that I can do more complicated stuff in the future... and ruby is my language of choice |
| 15:11.22 | [TK]D-Fender | meuserj: Ok, yup, this is definitely the place for AGI... |
| 15:13.55 | Zeeek | oh yes |
| 15:14.25 | Zeeek | dig voipusersconference.org |
| 15:14.26 | meuserj | [TK]D-Fender: I did a bash agi script yesterday, which was just a voice prompt for users to set a call forward number, and it works fine.. when I execute the ruby script from the command line it also works fine, but I'm getting a response SIP response 400 when I do the ruby script, which in my experience with the bash script is what happens when the script fails to execute for some reason. |
| 15:14.43 | meuserj | sorry for the long message. |
| 15:14.43 | BrianR___ | [TK]D-Fender: The ackcall option in agents.conf does the right thing for agentcallbacklogins. It's like using the "c" option when dialing on a zap channel. |
| 15:14.48 | Zeeek | ~dig voipusersconference.org |
| 15:15.03 | RB2 | Hey all, had a quick question. I've implemented asterisk before, but has anyone had any experience with a hosted asterisk system? I've read some complaints about dropped calls, etc. but don't have any first-hand experience. |
| 15:15.47 | BrianR___ | [TK]D-Fender: User needs to press "#" or the channel isn't really considered answered. It wasn't working only because I had fat-fingered the directive in the agents.conf file. |
| 15:17.40 | *** join/#asterisk af_ (n=getsmart@88-149-241-240.dynamic.ngi.it) |
| 15:18.11 | [TK]D-Fender | meuserj: This should have no impact on SIP responses. make sure to ANSWER the call before entring the AGI |
| 15:18.16 | *** join/#asterisk riccyb (n=rnbrady@193.82.139.121) |
| 15:18.29 | [TK]D-Fender | BrianR___: OK |
| 15:18.42 | *** join/#asterisk knobo` (n=user@noosbohknu.freecode.no) |
| 15:19.00 | knobo` | Is it possible to debug which steps a call goes throug a dialplan? |
| 15:19.29 | knobo` | someting like "sip set debug" for extensions? |
| 15:19.38 | *** join/#asterisk riccyb (n=rnbrady@193.82.139.121) |
| 15:19.41 | jameswf-home | is writing up a gizmo agi that doesnt require freepbx... |
| 15:19.57 | [TK]D-Fender | knobo : You cann't debug the dialplan in a targeted way. All or nothing |
| 15:20.23 | knobo` | [TK]D-Fender: how to debug all then? |
| 15:20.58 | [TK]D-Fender | knoboJust go look in CLI. Do "set verbose 10" and you'll see how your calls are processed |
| 15:21.56 | [TK]D-Fender | RB2: there is no answer for a question as vague as that. We all run * just fine. To address a certain scenario we'd have to be able to see it |
| 15:22.53 | meuserj | [TK]D-Fender: I am: Answer...AGI(batphone.bat)...Hangup |
| 15:24.20 | EmleyMoor | brodiem: No - T-Mobile UK |
| 15:26.50 | knobo` | Could this be a codec issue?: chan_sip.c:12322 handle_response_invite: Failed to authenticate on INVITE to '<sip:100@192.168.0.132>;tag=as298b296a' |
| 15:27.18 | knobo` | There is more, but I don't want to paste it all.. |
| 15:27.29 | [TK]D-Fender | knobo : no, that is a striaght SIP auth failure |
| 15:27.37 | [TK]D-Fender | knobo`: PASTEBIN |
| 15:27.38 | [TK]D-Fender | ~pb |
| 15:27.39 | jbot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste |
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| 15:30.10 | knobo` | Could it be the fact that I have ekiga and asterisk on the same computer. I have a virtal IP for ekiga, and bind only to the primary-ip with asterisk |
| 15:32.25 | RB2 | [TK]D-Fender, sorry. I was just looking for general feedback on the hosted asterisk solutions that are out there. I'm looking at 5 extensions in one office and 5-10 other extensions scattered at home offices. So, I'm trying to figure out if the hosted asterisk is a possible reliable solution instead of running it in-house. |
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| 15:33.40 | [TK]D-Fender | RB2: Its all the same. |
| 15:34.24 | [TK]D-Fender | knobo`: Its says "auth" so I'm guessing you jsut didn't set it up right. Go look at your SIP config in * and on the phone |
| 15:34.35 | meuserj | ok.. I'm completely stumped.. I can't figure out why it isn't running: http://pastebin.com/m15a00b8b |
| 15:34.44 | meuserj | it works fine from the command line... |
| 15:35.14 | Zeeek | join us now and every friday at 12 noon EDT : http://voipusersconference.org |
| 15:35.28 | Zeeek | sometimes there are even... gasp... females there! |
| 15:35.37 | Zeeek | and beer :) |
| 15:35.43 | RB2 | [TK]D-Fender, So you've not heard of any general problems utilizing a hosted asterisk service? dropped calls, etc.? |
| 15:35.56 | Zeeek | and [TK]D-Fender |
| 15:36.19 | meuserj | the only thing could think is that the agi user can't execute /usr/bin/ruby1.8 or can't find the agi library |
| 15:36.54 | [TK]D-Fender | RB2: * is *, doesn't matter who's physical box its on. |
| 15:39.25 | RB2 | [TK]D-Fender, ok, I thought perhaps there were service quality issues (unrelated to asterisk itself) that may manifest themselves on a hosted solution. Thanks! |
| 15:39.51 | [TK]D-Fender | RB2: No such thing as "magic" here... |
| 15:40.26 | jameswf-home | So i wonder if i publih this agi if gizmo will break it ? |
| 15:40.42 | jameswf-home | *publish |
| 15:41.19 | knobo` | http://pastebin.com/m409a48f3 |
| 15:41.29 | RB2 | [TK]D-Fender, ;) I was thinking more along the lines of issues because of network congestion between the end user and the box, NAT transversal issues, etc. |
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| 15:43.07 | s0ck | afternoon |
| 15:43.23 | knobo` | [TK]D-Fender: are you able to open the url? |
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| 15:44.46 | [TK]D-Fender | knobo`: You might want to actually spell "secret" PROPERLY if you want things to work. |
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| 15:47.52 | knobo` | [TK]D-Fender: That would probably help :) Sorry for misusing a genius as a spell corrector ;) |
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| 15:50.25 | anonymouz666 | I think the app_queue should have some option like jump_if_everybody_is_inuseorunavailable=yes :D |
| 15:51.03 | *** join/#asterisk pa (n=pa@unaffiliated/pa) |
| 15:51.04 | knobo` | But, now I get X-Asterisk-HangupCause: Call Rejected |
| 15:51.04 | anonymouz666 | I am using devstate for that and the code is becoming a beautiful spaguetti |
| 15:51.33 | *** join/#asterisk CanWood (n=chatzill@24.108.64.80) |
| 15:53.10 | knobo` | I have the sip messages here http://pastebin.com/d1357d906 |
| 15:53.31 | tzafrir_laptop | pif, here? |
| 15:54.29 | meuserj | well.. I figured it out.. kinda... decided that I needed to run asterisk in a console so that I can see the stderr of the script... and restarting asterisk fixed whatever the problem was.... which makes me think the problem has to do with the environment in which asterisk was initially started. |
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| 15:56.02 | knobo` | Could it be that it is trying to use md5? I found " .. algorithm=md5, ..." |
| 15:56.13 | knobo` | I have not md5 encrypted the passwords. |
| 15:56.26 | *** join/#asterisk af_ (n=getsmart@88-149-241-240.dynamic.ngi.it) |
| 15:56.46 | knobo` | # |
| 15:56.46 | knobo` | [Sep 19 17:51:59] NOTICE[20861]: chan_sip.c:12322 handle_response_invite: Failed to authenticate on INVITE to '"knobo| knobo" <sip:100@192.168.0.132>;tag=as6fb456ed' |
| 15:56.46 | knobo` | # |
| 15:56.52 | knobo` | <PROTECTED> |
| 15:56.57 | knobo` | # |
| 15:57.03 | knobo` | <PROTECTED> |
| 15:57.10 | knobo` | # |
| 15:57.15 | knobo` | <PROTECTED> |
| 15:57.18 | meuserj | dude, pastebin |
| 15:57.21 | knobo` | sorry that was not suppose to happen |
| 15:57.26 | knobo` | sorry |
| 15:57.44 | knobo` | terminal trubble.. Running emacs in a terminal. not use to do that. |
| 15:57.59 | meuserj | ah |
| 15:58.41 | *** join/#asterisk wacky_ (n=abourget@mtl.savoirfairelinux.net) |
| 15:58.42 | nr4q | kicks his asterisk box |
| 15:58.54 | knobo` | what I ment to do, was asking what is: -- SIP/knobo-081e61d0 is circuit-busy |
| 15:58.59 | wacky_ | Does anyone know why I lose the colored output when connecting to a *remote* asterisk box (via SSH) ? |
| 15:59.28 | *** join/#asterisk imcdona (i=imcdona@imcdona.broker.freenet6.net) |
| 15:59.32 | nr4q | wacky: because your ssh client doesnt support color |
| 15:59.55 | ManxPower | wacky_: that question has been discussed on the mailing lists |
| 16:00.10 | wacky_ | ManxPower: have any keywords that would point me there ?? |
| 16:00.14 | nr4q | wacky: or the term set on the remote side doesn't support color |
| 16:00.33 | wacky_ | nr4q: like some color-xterm stuff or packages ?! |
| 16:01.02 | ManxPower | wacky_: how about "color" "terminal" "site:lists.digium.com" |
| 16:01.05 | nr4q | wacky: er.. no not really. like vt100 |
| 16:01.38 | ManxPower | knobo`: Circuit busy means pretty much what is says. |
| 16:01.54 | wacky_ | heh.. thanks :P |
| 16:02.22 | *** join/#asterisk grEvenX (n=even@pc107-130.ktv.no) |
| 16:02.55 | nr4q | with iax2 behind a firewall.... (win2k3 server) all i should have to do is forward 4569 UDP ? |
| 16:04.02 | knobo` | ManxPower: but if nobody other then me is using the servers |
| 16:04.08 | knobo` | then why? |
| 16:04.45 | ManxPower | knobo`: Well what are you calling? |
| 16:05.26 | knobo` | I have two servers. Both are asterisk |
| 16:05.40 | ManxPower | The thing is with all the OTHER problems you are having, I think it is silly to try to diagnose the circuit busy issue before you try to fix the other issues. |
| 16:06.15 | knobo` | I'm able to call from Ekiga on server-2 to asterisk on server-2 |
| 16:06.22 | knobo` | But not from server-1 to server-2 |
| 16:06.30 | knobo` | To play back a simple message |
| 16:07.04 | *** part/#asterisk meuserj (n=meuserj@indianalifesciences.com) |
| 16:08.15 | [TK]D-Fender | knobo`: does Ekiga to #1 work by itself? |
| 16:08.50 | knobo` | by calling local asterisk? |
| 16:09.03 | knobo` | I get a dial tone when I'm forwarded to the server-2 |
| 16:09.18 | knobo` | So Asterisk is delivering this tone. |
| 16:09.27 | knobo` | Sorry, it is not dial tone, it is ring-tone |
| 16:09.52 | ManxPower | knobo`: A simple mistake like that could open up your server to hackers. Be more careful. |
| 16:10.57 | *** join/#asterisk casix (n=casix@pbxedifici.adamvozip.es) |
| 16:10.59 | casix | hello |
| 16:11.05 | knobo` | that's ok. Now I just whant it to work |
| 16:11.36 | knobo` | If I could remove all security, and just make it work |
| 16:11.42 | knobo` | Then I could add security after |
| 16:13.44 | casix | I have a simple trouble. I've reinstalet the system of a computer and reinstall asterisk. I put the codec_g729a.so into /usr/lib/asterisk/modules, and the licenses file in /var/lib/asterisk/licenses/ but when it start it found 0 licenses. The hardware is the same. The host-id of the file is the same that show asterisk when trying to load the module. any ideas? |
| 16:14.55 | *** join/#asterisk af_ (n=getsmart@88-149-241-240.dynamic.ngi.it) |
| 16:17.03 | knobo` | ManxPower: or, do you disagre? Isn't it better to make it work first? |
| 16:19.43 | knobo` | I get sip messages on both servers when I call. So i obviusly get throug. If I paste the sip messages in the pastebin, would anyone be able to help me? |
| 16:20.34 | knobo` | Should it be the messages from the calling side, or the called side. Or are they equal? |
| 16:20.49 | *** join/#asterisk Deeewayne (n=Deeewayn@nat/digium/x-487904ba15f42277) |
| 16:20.49 | *** mode/#asterisk [+o Deeewayne] by ChanServ |
| 16:20.58 | *** join/#asterisk russellb (n=russell@asterisk/digium-open-source-team-lead/russellb) |
| 16:20.58 | *** mode/#asterisk [+o russellb] by ChanServ |
| 16:21.36 | *** join/#asterisk sah-work (n=Bawbatos@65-119-47-100.dia.static.qwest.net) |
| 16:21.41 | Aquahallic | what's the latest "stable" release? |
| 16:22.54 | *** join/#asterisk simNIX (n=simNIX@82-204-21-111.dsl.bbeyond.nl) |
| 16:23.16 | casix | Aquahallic: Asterisk 1.4.21.2 |
| 16:23.20 | hardwire | ARR! |
| 16:23.27 | Aquahallic | ty |
| 16:24.00 | knobo` | It looks like it is not able to handle the ekiga on the same server as the asterisk server, even though I have a virtual IP for it. Is there a way around that? |
| 16:24.56 | [TK]D-Fender | knobo`: not the case. your auth is wrong. This is a misconfiguration problem. |
| 16:26.25 | knobo` | [TK]D-Fender: does there exsist a minimum.sip.conf example out there, that is easy to understand for a simple user like me? |
| 16:27.05 | [TK]D-Fender | knobo' : first stop putting IP's for your SIP phones. do "host=dynamic" and let them register |
| 16:27.07 | *** join/#asterisk grEvenX (n=even@pc107-130.ktv.no) |
| 16:27.34 | knobo` | ok, done. |
| 16:27.47 | [TK]D-Fender | knoboNext forget about the bridge between 2 servers until you get the phone working to the first. when testing do "sip debug" from * LCI and look at the complete communication. |
| 16:28.37 | knobo` | [TK]D-Fender: so, you mean from ekiga-1 to ekiga-2? |
| 16:28.52 | casix | knobo: you can do a make samples in the source code and it generate a config files. Be carefully it will overwrite what you have in /etc/asterisk |
| 16:29.11 | knobo` | [TK]D-Fender: or ekiga-1 to asterisk-1 |
| 16:29.17 | [TK]D-Fender | knobo`: you are mixing thing up between connecting to ASTERISK boxes and 2 SOFTPHONE. Please clarify this. |
| 16:29.24 | [TK]D-Fender | 2* |
| 16:29.53 | knobo` | I have ekiga-1 -> asterisk-1 -> asterisk-2 -> ekiga-2 |
| 16:30.29 | knobo` | Calling from ekiga-2 to asterisk-2 gives expected result. I can here a playback sound |
| 16:30.29 | [TK]D-Fender | knobo`: deal with ekiga-1 -> asterisk-1 first |
| 16:31.00 | knobo` | ekiga-1 to asterisk-1 worked a minute ago, I will double check now. |
| 16:31.25 | *** join/#asterisk imcdona (i=imcdona@imcdona.broker.freenet6.net) |
| 16:31.29 | knobo` | Yes I can here the monkey screeming |
| 16:31.44 | *** join/#asterisk mort_gib (n=mjensen@16.Red-83-36-63.staticIP.rima-tde.net) |
| 16:33.32 | wacky_ | I can't find anything that fixes my remote coloring problem on the lists.. |
| 16:33.32 | knobo` | writing "dial 8556@my-config" on server-1 (which should then direct me to server-2) gives: channel.c:2125 __ast_read: Dropping deferred DTMF digits on Console/dsp1 |
| 16:34.18 | [TK]D-Fender | steps out for a while |
| 16:34.55 | knobo` | '_8XXX' => 1. Dial(SIP/knobo/${EXTEN:1}|30|r) [pbx_config] |
| 16:35.13 | knobo` | something missing in the dialplan maybe. |
| 16:36.51 | *** join/#asterisk questprojects (n=mg@82-71-8-45.dsl.in-addr.zen.co.uk) |
| 16:37.15 | qp | afternoon |
| 16:38.37 | knobo` | Conecting from ekiga-1 to ekiga-2 looks like it is working |
| 16:38.42 | knobo` | We have gsm both ways |
| 16:39.09 | nr4q | where would you start troubleshooting if you saw "REGREQ" state on peer registries (iax2) and that state never changes |
| 16:39.35 | knobo` | And we have also sound both ways. |
| 16:40.31 | knobo` | Connecting from ekiga-2 to ekiga-1 also works. |
| 16:41.29 | jameswf-home | russellb: my abuse of your function :) http://dontcallmyboss.blogspot.com/2008/09/asterisk-volume-function-dialplan.html |
| 16:41.36 | qp | how can I connect 2 calls using the AMI ? ie, 2 calls to cell phones from a softphone, and I want to use the AMI to connect them together, and then free up my lines |
| 16:42.03 | knobo` | Both asterisk servers use host=IPADDRESS, not registre |
| 16:42.19 | knobo` | Should I use register from one of the servers? |
| 16:42.26 | knobo` | Or, is IP both ways good? |
| 16:44.17 | russellb | jameswf-home: glad i could help :) |
| 16:44.40 | knobo` | putting exten => _8XXX,1,Dial(SIP/192.168.0.83/${EXTEN:1},30,r) in the extensions.conf does not contact the other ekiga. |
| 16:45.00 | knobo` | 192.168.0.83 is the IP-address of the other ekiga. |
| 16:45.02 | qp | hmm ... eyebeam (xlite) seems to be able to connect 2 lines without using the AMI |
| 16:45.25 | qp | can you "join" to calls from the CLI? |
| 16:45.29 | qp | two |
| 16:47.20 | [TK]D-Fender | qp: Redirect each party to a Meetme |
| 16:47.50 | *** join/#asterisk wiscados (n=mint@81.25.184.155) |
| 16:47.53 | [TK]D-Fender | knobo : when yuo dial by IP like that you are not passing ANY auth to the other server. Use a proper peer entry |
| 16:48.25 | jameswf-home | I need to adjust it and do lik the parrot thing but i was to lazy to workout the loop |
| 16:48.53 | *** mode/#asterisk [-b *!email@*unaffiliated/jeev] by russellb |
| 16:48.54 | knobo` | [TK]D-Fender: could it be, that it expects md5? |
| 16:49.15 | [TK]D-Fender | knobo : right now youa ren't even passing a password. |
| 16:49.23 | nr4q | do IAX2 ackknowledgements occur over port 4569 ? |
| 16:49.23 | qp | fender: will try that, but dont you need to know the meetme number? or can you have a "please join the next free one" so that two softphones dont end up putting 4 people in one room, rather than 2 in a room each |
| 16:49.42 | knobo` | [TK]D-Fender: could it be that exten => _8XXX,1,Dial(SIP/knobo/${EXTEN:1},30,r) is not correct in extensions.conf? |
| 16:49.45 | [TK]D-Fender | qp: What are you trying to actually accomplish? |
| 16:49.50 | *** mode/#asterisk [-b *!*n=email@*unaffiliated/jeev] by russellb |
| 16:50.10 | [TK]D-Fender | knobo`: putting exten => _8XXX,1,Dial(SIP/192.168.0.83/${EXTEN:1},30,r) <-- specifying by IP here = bad |
| 16:50.15 | *** join/#asterisk jeev (n=email@unaffiliated/jeev) |
| 16:50.16 | qp | we have a softphone, 2 lines, and want an operator to be able to click a button and have the 2 lines "connect" and then the operator's lines are freed |
| 16:50.19 | jeev | damn |
| 16:50.32 | [TK]D-Fender | qp: then just transfer the call. |
| 16:50.50 | [TK]D-Fender | qp: Or do a 3-way confernce and hang up |
| 16:51.09 | jeev | -- Executing [8188230422@default:2] Dial("SIP/64.34.251.201-08815000", "SIP/viatalkhome/*671818230422|60|t") in new stack |
| 16:51.09 | jeev | -- Called viatalkhome/*671818230422 |
| 16:51.20 | jeev | any idea what the hell is causing that? it seems to be the only number and it's cutting it off ALL THE TIME! |
| 16:51.31 | knobo` | [TK]D-Fender: on this local network we don't have dns. |
| 16:51.46 | knobo` | [TK]D-Fender: would using the hosts file be better? |
| 16:51.47 | qp | it cant be a blind transfer, has to be "hey, you want to chat tothis guy?" "yeah" "ok, connecting him" ... then press a button and they are chatting |
| 16:52.10 | [TK]D-Fender | knobo`: you need to specifiy the sip.conf peer to use or you won't be passing the auth along with the call. |
| 16:52.11 | qp | I will try and packet sniff what eyebeam does too |
| 16:52.19 | [TK]D-Fender | knobothis has absolutely nothing to do with DNS |
| 16:52.34 | knobo` | [TK]D-Fender: I see. |
| 16:52.37 | [TK]D-Fender | qp: that would be an ATTENDED transfer |
| 16:52.50 | [TK]D-Fender | qp: Which it should support. |
| 16:52.58 | [TK]D-Fender | qp: Go check its manual |
| 16:53.08 | qp | the API we use doesn't, its our own phone |
| 16:53.15 | qp | they pointed us at AMI |
| 16:53.19 | *** join/#asterisk FlyboySR22 (n=rsears@hq.fw.americanis.net) |
| 16:53.25 | qp | I will do some digging, ta |
| 16:53.50 | [TK]D-Fender | qp: you should be doing this IN eyebeam, not on anything external |
| 16:54.06 | knobo` | [TK]D-Fender: would it be a mistake to have the same section name in both sip.conf files (on both servers)?. |
| 16:54.20 | qp | sorry, its not eyebeam, I was using that as an exampe. we have written a softphone from scratch |
| 16:54.30 | knobo` | so the sip section is [knobo] on both sides. |
| 16:54.45 | [TK]D-Fender | knobo`Should have 2 entries on each server. One to go out to the other, the other for incoming from the other side\ |
| 16:55.03 | qp | if he AMI cant do it, and the CLI cant do it, then I will have to nag the API developers |
| 16:55.18 | knobo` | [TK]D-Fender: So, I can not use type=friend |
| 16:55.53 | [TK]D-Fender | knobo`: I'm going to leave you to play around with this for a while. |
| 16:56.16 | knobo` | [TK]D-Fender: ok, so long, and thanks for all the fish :) |
| 16:56.18 | [TK]D-Fender | qp: Why is it you aren't just doing a transfer like normal on the phone itself like you're supposed to? |
| 16:56.30 | [TK]D-Fender | knobo`: DON'T PANIC |
| 16:57.11 | qp | [TK]D-Fender: its a managed service. so a call comes in, we stick it on hold, call a cell phone, see if they want to talk to this caller, if so connect. if not, hang up 2nd line, get 1st line back, take a message |
| 16:57.59 | [TK]D-Fender | qp: With a human operator doing the leg-work? |
| 16:58.20 | qp | yes, as its many customers calling for many companies, like a virtual PA service |
| 16:58.58 | [TK]D-Fender | qp: then let your "operator" just do an attended transfer like the rest of us already do. |
| 16:58.59 | qp | on Eyebeam, you press transfer, then click the line you want to connect, and it all happens. so I know its possible in plain old sip. just need to work out "how" |
| 16:59.23 | casix | by |
| 16:59.24 | casix | bye |
| 16:59.24 | *** join/#asterisk jplank (n=GBove@reports.nyigc.net) |
| 16:59.38 | qp | yeah, could do. just easier to click a button and lines 1 and 2 are connected and then go away, so to speak |
| 16:59.38 | jplank | anyone in here going to astricon next week? |
| 17:00.07 | *** part/#asterisk dandre (n=daniel@was59-3-82-236-48-30.fbx.proxad.net) |
| 17:00.07 | [TK]D-Fender | qp: as I said, go read its manual. It should support this. |
| 17:03.41 | jplank | no one is going? |
| 17:04.05 | jeev | maybe |
| 17:04.15 | [TK]D-Fender | jplank: No, its that no-one is who is is in any hurry to answer you. |
| 17:04.56 | qp | I have been through the sdk manual, and emailed the devs, nothing obvious. will carry on researching tho. thanks for your help Fender |
| 17:04.56 | jplank | I hope your going fender :) |
| 17:04.58 | *** join/#asterisk mog (n=mog@nat/digium/x-cef6c377c17bae0f) |
| 17:04.58 | *** mode/#asterisk [+o mog] by ChanServ |
| 17:05.01 | [TK]D-Fender | jplank: You did wait an entire 4 minutes. |
| 17:05.07 | jplank | exactly |
| 17:05.09 | [TK]D-Fender | jplank: And no, I'm not going. |
| 17:05.17 | jplank | in IRC land, thats an eternity |
| 17:05.38 | [TK]D-Fender | I'd demand instant gratification, but I can't wait that long.... |
| 17:05.47 | jplank | lol |
| 17:05.55 | jeev | if fender isn't going, i'm not going.. no point, i just wanted to kick his ass for yelling at me all the time |
| 17:06.05 | jplank | I was just thinking the same thing |
| 17:06.26 | jeev | "YOU DOOFUS, YOU DONT KNOW 1+1?" |
| 17:06.28 | jeev | that's all i hear |
| 17:06.35 | jplank | I'm sure most of the times it was warranted though |
| 17:06.41 | jameswf-home | okay astriconers I threw together a list of food places locals like so you dont have to rely on travel websites for tourist traps http://dontcallmyboss.blogspot.com/2008/09/coming-to-astricon-need-food.html |
| 17:06.49 | [TK]D-Fender | jeev: Threat duly noted. |
| 17:06.53 | jeev | lol |
| 17:07.02 | jplank | the only reason I ask is I'm curious of dress attire. |
| 17:07.14 | jplank | my boss seems to think it will be like comptel and the like |
| 17:07.18 | jplank | and dressy |
| 17:07.24 | jplank | I think its more casual |
| 17:07.30 | fiddur | I have an emergency here :-/ After running smoothly a few days with 100 new sip lines (sip registry), it suddenly started getting timeout on registry and being quite unresponsive. We haven't changed anything, and network to the provider seems ok; they say there's nothing wrong on their side. Now, asterisk hangs on a sip reload! |
| 17:07.55 | [TK]D-Fender | fiddur: check your DNS |
| 17:09.24 | *** part/#asterisk Zeeek (n=IceChat7@pdpc/supporter/active/Zeeek) |
| 17:09.38 | fiddur | [TK]D-Fender: a dig on providers host quickly gives me the correct ip... what else can I check? |
| 17:10.13 | fiddur | After quite a few timeout messages I see "sched.c: Request to schedule in the past?!?" in the debug-log |
| 17:10.31 | *** join/#asterisk snocetti (n=sebastia@190.144.113.58) |
| 17:10.40 | snocetti | Hi everybody |
| 17:10.43 | [TK]D-Fender | fiddur: thats a sign foa zaptel timer issue primarily |
| 17:10.44 | jeev | http://ggasd.pastebin.com/d2722b11a fender, does that look good for a failover? i'm having doubts |
| 17:11.36 | snocetti | I am having a problem with a quintum tenor AF 8 ports FXO, this is not sending to asterisk callerid information, someone has configured it? |
| 17:11.41 | jeev | jameswf-home, you're missing a comma in your title! |
| 17:11.56 | fiddur | [TK]D-Fender: With only 15 lines (sip regs) active, it works, with a few timeouts in a 10 minute period... With more, it seems to cough up on itself when the timeout starts coming. |
| 17:13.07 | fiddur | I'm not sure I trust that all is well at the providers side, but I don't know how to get "evidence" of that either. |
| 17:13.10 | [TK]D-Fender | fiddur: check the rest of your network settings. routing, IP's, the works |
| 17:13.19 | jplank | fiddur: do they all initially register? |
| 17:13.31 | jplank | are they all registering to the same sip gateway? |
| 17:13.43 | fiddur | [TK]D-Fender: It goes out through the firewall without any problem... |
| 17:15.06 | fiddur | jplank: 88 of them are on one ip, 12 on another... Right now I only have 25 active, as they go up and down. The initial registration is much slower now than when I first added the accounts. |
| 17:15.45 | jeev | http://ggasd.pastebin.com/d2722b11a anyone know if that'll survive as a failover if the first ITSP reports an error ? |
| 17:15.58 | fiddur | The problem is that this is in active use... We moved these 100 numbers from isdn pri to a sip provider 2 days ago... they're supposed to take quite a lot of calls right now and a couple of hours more |
| 17:17.37 | fiddur | I have rtcp debug on the providers ip... I have nothing to compare to, but I can see the registration sometimes have about 33 packets lost... |
| 17:18.57 | fiddur | What is DLSR and RTT in the rtcp debug? round trip time? |
| 17:21.51 | knobo` | I can connect now, but I have very bad sound quality. I have reinvite=no |
| 17:23.40 | knobo` | When I had reinvite=yes, only one of us could hear anything |
| 17:23.48 | knobo` | Then the quality was ok. |
| 17:23.50 | fiddur | [TK]D-Fender: The "Request to schedule in the past?!?!" comes from the same process and shortly after "chan_sip.c: Scheduled a registration timeout for cx-cust-shg02.rixtelecom.se id #3746" |
| 17:24.33 | jeev | http://ggasd.pastebin.com/d2722b11a anyone know if that'll survive as a failover if the first ITSP reports an error ? |
| 17:24.49 | knobo` | How can I improve the quality on the sound. I am on a lan, so there is no problem with the network capability |
| 17:26.22 | fiddur | knobo`: Check what codec is being used on your call |
| 17:29.43 | knobo` | fiddur: gsm in at both ends. |
| 17:29.50 | *** join/#asterisk ManxPower (n=manxpowe@89.sub-75-251-68.myvzw.com) |
| 17:30.31 | knobo` | what is: "Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)"? |
| 17:30.35 | jameswf-home | was looking at the marksters booty call deal and was thinking anyone with the time to write that or who would use it probably has no need as they are not getting laid |
| 17:30.49 | *** join/#asterisk maxxim (n=maxxxim@93-96-96-106.zone4.bethere.co.uk) |
| 17:31.13 | jeev | jameswf-home, do you plan on getting laid at astricon |
| 17:31.31 | knobo` | does this mean gsm?: "m=audio 5028 RTP/AVP 3 101" |
| 17:31.32 | fiddur | knobo`: That's a low bandwidth codec. You could e.g. use alaw instead... add disallow = all and allow = alaw if applicable (if your phones can handle it...) otherwise you'll have to check what the phones handles etc |
| 17:31.35 | ManxPower | knobo`: that is non-codec stuff like DTMF |
| 17:31.42 | jameswf-home | heh I am married I havent been laid in years |
| 17:31.48 | maxxim | hi. could anyone to help me to setup connection betweek GNUGK and Asterisk? Asterisk is registering successeful to GNUGK. I have the problems with calling asterisk from the clients registered on GNUGK. |
| 17:31.54 | jeev | http://ggasd.pastebin.com/d2722b11a anyone know if that'll survive as a failover if the first ITSP reports an error ? |
| 17:31.57 | jeev | hahahah jameswf-home, serious ? |
| 17:31.58 | *** join/#asterisk Bananaskin (n=Banana@93-97-226-229.zone5.bethere.co.uk) |
| 17:32.24 | jameswf-home | did i add an lol :) |
| 17:32.29 | jeev | heh |
| 17:32.38 | jeev | eww, someone mms'd me a pic of someone's dong with glasses over it |
| 17:32.38 | jeev | lol |
| 17:32.41 | ManxPower | jeev: nope, not going to work |
| 17:33.04 | ManxPower | mixing extensions does not work |
| 17:33.24 | jeev | hm |
| 17:33.39 | maxxim | ManxPower> i've set the 'demo' context. it show that asterisk plays the demo, but i can't hear anythin on NetMeeting client |
| 17:34.12 | maxxim | if you have experiense, i can paste somewhere the log |
| 17:34.35 | ManxPower | jeev: http://ggasd.pastebin.com/m4c62813e |
| 17:34.37 | maxxim | i have a lot of the following lines in the log: |
| 17:34.38 | maxxim | 10:46:51:489 H.245 Listerner socket being monitored (incoming, ooh323c_1) |
| 17:34.39 | jeev | ManxPower, i know, i tried to do it the right way but |
| 17:34.42 | jeev | hm |
| 17:34.43 | maxxim | is it ok, or not? |
| 17:34.50 | ManxPower | maxxim: I'm sorry to hear that |
| 17:35.04 | knobo` | PCMA=alaw? |
| 17:35.04 | jeev | yea ManxPower, uh.. how do i do the rest though? |
| 17:35.04 | jeev | exten => _XXX-1NXXNXXXXXX,n,Goto(s-${DIALSTATUS},1) |
| 17:35.05 | jeev | exten => s-NOANSWER,1,Hangup |
| 17:35.06 | jeev | do i do |
| 17:35.11 | ManxPower | But as I NEVER EVER do H323 I guess yoll have to find some other poor sod to help you. |
| 17:35.16 | jeev | exten => _XXX-1NXXNXXXXXX,s,NOANSWER,1,Hangup |
| 17:35.26 | jeev | or it's fine how you gave it ? |
| 17:35.58 | knobo` | Ekiga does not have alaw. |
| 17:36.03 | ManxPower | jeev: why don't you just try it like I have it to you |
| 17:36.28 | jeev | because i tried it the way i did it and i couldn't reproduce an error, sucks. ok ManxPower, thanks! |
| 17:36.33 | jeev | manx, thanx! |
| 17:36.41 | ManxPower | jeev: you had a LOGIC error |
| 17:36.56 | ManxPower | You don't understand how extensions.conf works yet. |
| 17:37.04 | fiddur | knobo`: so what does it have? |
| 17:37.23 | knobo` | pcma |
| 17:37.33 | knobo` | is that the same? |
| 17:37.39 | ManxPower | PCMA is G.711 Alaw |
| 17:37.57 | ManxPower | knobo`: a 15 second google search would have given you the answer |
| 17:37.59 | fiddur | knobo`: ekiga has alaw according to http://www.gnomemeeting.org/index.php?rub=2 |
| 17:38.37 | coppice | there's alaw for the rich and alaw for the poor. |
| 17:38.47 | knobo` | so, pcma, should be ok. |
| 17:38.58 | knobo` | But I only get sound one way at the time |
| 17:39.01 | maxxim | ManxPower> here is the log: http://ggasd.pastebin.com/m10502719 Could you plese have a look |
| 17:39.04 | anonymouz666 | coppice: you mean polycom and grandstream? |
| 17:39.07 | anonymouz666 | :) |
| 17:39.21 | knobo` | I can hear the other phone, but the other can no hear me |
| 17:39.22 | ManxPower | maxxim: You understand that I am not paid to be here and do not work for Digium, right? |
| 17:39.31 | *** join/#asterisk mascool (n=george@12.188.136.66) |
| 17:40.00 | ManxPower | maxxim: I'm not going to tell you again. I don't use H323 and I will not help people using H323. |
| 17:40.10 | mascool | does anyone know if it's possible to set mohsuggest in users.conf ? (asterisk 1.4.19) ? |
| 17:40.27 | *** join/#asterisk oilinki (n=oil@ppp-124-120-1-52.revip2.asianet.co.th) |
| 17:40.41 | ManxPower | mascool: users.conf is primarily for GUIs. Almost nobody here uses it. |
| 17:41.08 | mascool | I know, but this customer wants to be able to use the GUI |
| 17:41.33 | ManxPower | mascool: maybe you should go to the channel that is dedicated to whatever GUI you are using. |
| 17:41.54 | maxxim | ManxPower> is it possible to register SIP users using sql queries? |
| 17:42.00 | *** join/#asterisk grEvenX (n=even@pc107-130.ktv.no) |
| 17:42.03 | mascool | ManxPower, thanks |
| 17:42.05 | ManxPower | maxxim: I don't use Realtime. |
| 17:43.02 | knobo` | The sip message is "Found audio description format PCMA for ID 8" from astersik. But in asterisk I pecified alaw. Does astersik use both names? |
| 17:43.07 | knobo` | at random choise? |
| 17:43.09 | jeev | thanks ManxPower. i'll have to watch the logs |
| 17:43.23 | jeev | -- Executing [8188230422@default:2] Dial("SIP/64.34.251.201-08815000", "SIP/viatalkhome/*671818230422|60|t") in new stack |
| 17:43.23 | jeev | -- Called viatalkhome/*671818230422 |
| 17:43.35 | jeev | ManxPower, someone calls that specific number and that's what it does, it cuts off a digit.. |
| 17:43.44 | jeev | she hasn't complained cause i think she's calling home lol but tha's weird |
| 17:43.45 | ManxPower | knobo`: PCMA is the SAME AS ALAW. |
| 17:43.47 | jeev | it's missing an 8 |
| 17:44.28 | [TK]D-Fender | jeev: your phone # seems to be missing a digit, and you aren't showing the complete call |
| 17:44.28 | ManxPower | jeev: what is the number SUPPOSED to be? |
| 17:44.42 | jeev | Executing [8188230422@default:2] |
| 17:44.51 | jeev | see how it's 818 823 0422 (which i've changed btw so that's random) |
| 17:44.56 | jeev | then it executes 818 230 422 |
| 17:45.09 | knobo` | It worked both ways with reinvite=no, but only one way with reinvite=yes |
| 17:45.11 | ManxPower | jeev: looks like you have some very very serious problem, as SIP phones send the entire number at once once the user is done dialing |
| 17:45.19 | [TK]D-Fender | jeev: and where's your code? |
| 17:45.52 | *** join/#asterisk cgardner (n=cgardner@64.239.154.240) |
| 17:46.09 | jeev | that's the only number it does it to |
| 17:46.11 | ManxPower | knobo`: sounds like you are using NAT |
| 17:46.16 | [TK]D-Fender | jeev: exten => _XXX-1NXXNXXXXXX,2,Dial(SIP/viatalkhome/*67${EXTEN:4:11},60,t) <-- your chop up job is clearly not being done right. |
| 17:46.25 | ManxPower | jeev: Asterisk does not just randomly drop digits |
| 17:46.31 | knobo` | ManxPower: no, no nat. |
| 17:46.41 | ManxPower | knobo` then a firewall issue |
| 17:46.49 | jeev | i know ManxPower |
| 17:46.58 | cgardner | Quick hardware question. I have a TDM400P card with 3 X100M REVC Daughter FXO cards in it and would like to add another FXO. Is the X400M FXO Module compatible with this setup? |
| 17:46.59 | [TK]D-Fender | jeev: (assuming you hybridized off that one) |
| 17:47.02 | jeev | fender, what i do is i want the extension of the user chopped off, then sent |
| 17:47.03 | knobo` | We don't have a firewall here |
| 17:47.11 | [TK]D-Fender | jeev: go look at your dial command and what you're passing it. |
| 17:47.34 | ManxPower | knobo`: then you have a problem we've never heard of before. |
| 17:47.35 | knobo` | all computers are connected to one swithc |
| 17:47.36 | [TK]D-Fender | jeev: Looks like youa ren't paying attention to WHAT you are chopping off very much. |
| 17:47.41 | jeev | exten =>_X.,1,Dial(SIP/jupiter/${CHANNEL:4:4}${EXTEN}) |
| 17:47.49 | jeev | that's what is sent from the internal asterisk box |
| 17:47.56 | jeev | jupiter is the gre tunnel to the main server |
| 17:47.58 | [TK]D-Fender | jeev: feel free to count what you're doing there... |
| 17:48.48 | knobo` | ManxPower: that's nice :( |
| 17:48.49 | [TK]D-Fender | jeev>-- Executing [8188230422@default:2] Dial("SIP/64.34.251.201-08815000", "SIP/viatalkhome/*671818230422|60|t") in new stack <- and this doesn;'t match the exten you jsut showed us failing |
| 17:48.57 | [TK]D-Fender | jeev: stop showing us apples & oranges |
| 17:50.00 | jeev | fender, are you talking about 'exten =>_X.,1,Dial(SIP/jupiter/${CHANNEL:4:4}${EXTEN})' this ? |
| 17:50.56 | [TK]D-Fender | jeev: Show us the exactly CLI output at the point is screws up along with your dialplan code. |
| 17:51.10 | jeev | that was the CLI output. |
| 17:51.11 | [TK]D-Fender | jeev: in a nice single pasetbin |
| 17:51.14 | jeev | do you awnt the dial plan from the internal |
| 17:51.14 | jeev | ok |
| 17:51.22 | jeev | do you want the dial plan from the internal asterisk also? which is the one i pasted. |
| 17:51.48 | ManxPower | jeev: one like does not provide us with enough context. |
| 17:51.48 | [TK]D-Fender | jeev: When you learn how to be coherent maybe I can help. |
| 17:52.17 | ManxPower | ..er... one line... |
| 17:54.16 | *** part/#asterisk bradleyprice861 (n=bradleyp@fw.datafax.net) |
| 17:54.54 | *** join/#asterisk bradleyprice86 (n=bprice@fw.datafax.net) |
| 17:57.09 | ManxPower | It is IMPOSSIBLE for this line "-- Executing [8188230422@default:2] Dial("SIP/64.34.251.201-0881500" to have been generated by this line in the dialplan "exten =>_X.,1,Dial(SIP/jupiter/${CHANNEL:4:4}${EXTEN})". |
| 17:57.16 | ManxPower | We get REALLY pissed off when people do that to us. |
| 17:57.56 | jeev | yea |
| 17:57.58 | jeev | sorry, getting it |
| 17:58.06 | jeev | ManxPower, i guess she made a call and the redundancy thing had an issue too so lets see |
| 17:58.15 | ManxPower | It's about as logical as telling a computer person you need to have your PC w/Vista fixed, and then you bring in a Mac instead. |
| 17:58.47 | anonymouz666 | app_queue.so is strange... sometimes report the member as INVALID sometimes UNAVAILABLE when both are static members and also unregistered...why? |
| 17:59.14 | ManxPower | anonymouz666: I have up on Queues in Asterisk YEARS ago. |
| 18:00.34 | jeev | http://issues.pastebin.com/d6df07fb6 |
| 18:00.45 | jeev | ManxPower, i'm thinking it's easy to fix that BUSY thing right? set the dial command after BUSY |
| 18:00.59 | ManxPower | Yesterday we had someone forward the office phones to an invalid cellphone number. |
| 18:01.08 | ManxPower | jeev: Why would you do that. If the number is busy then it is busy. |
| 18:01.16 | jeev | true |
| 18:01.17 | ManxPower | BUSY is not an error condition |
| 18:01.22 | jeev | yea, i meant the other error |
| 18:01.33 | tzanger | ManxPower: depends on whether you really need to speak with them or not :-) |
| 18:01.41 | jeev | chanavail will take over if it shows CONGESTION ? |
| 18:01.45 | tvirus | Under my incoming call rule, how can I send channel 2 on my zap card (emergency line) to voicemail if it rings? Our analog company will hunt the lines automatically and send calls to the appropriate lines, I just don't know how to detect what line it's on. Would it be something like exten = _NXXNXXXXXX,1,GotoIf($["${CHANNEL}" = "ZAP/2"]?700) 700 being the extension of the VM. |
| 18:02.04 | *** join/#asterisk Tako-san (n=Tako-san@24.108.192.144) |
| 18:02.22 | [TK]D-Fender | tvirus: Just put that channel into a separate context as specified in zapata.conf |
| 18:02.27 | ManxPower | tvirus: easiest is to put that channel is a totally separate context. |
| 18:02.54 | tvirus | Well, the other issue is that we use it for faxes :| |
| 18:03.19 | fiddur | Can lots of sip register timeouts make it hard for other sip devices to get through to register themselves, or is it proof of that the problem is on our side when my sip-phone can't reach our asterisk when it's clogged up with register timeouts? |
| 18:04.55 | *** join/#asterisk Defraz (n=T0tal@fw.fuzecore.com) |
| 18:07.35 | ManxPower | tvirus: you just said the line is only used for incoming calls to voicemail |
| 18:07.39 | jeev | ManxPower [TK]D-Fender http://issues.pastebin.com/d6df07fb6 |
| 18:08.38 | tvirus | It's shared with a fax, I forgot about that until you mentioned the zapata.conf file. I guess I'd have to do a fax detect on the context (defined as fax in zapata.conf) and if it's not a fax, goto my voicemail. |
| 18:08.38 | ManxPower | jeev: I see nothing wrong with that |
| 18:09.26 | [TK]D-Fender | I do |
| 18:09.28 | ManxPower | jeev: you understand that most providers do not support *67, right? |
| 18:10.06 | [TK]D-Fender | jeev: exten => _XXX-1NXXNXXXXXX,1,Set(CALLERID(number)=2132222222) <---- this has a DASH in the exten pattern |
| 18:10.28 | [TK]D-Fender | jeev: Executing [8182331233@default:2] Dial("SIP/64.34.251.201-08815000", "SIP/viatalkhome/*67331233|60|t") in new stack <--- THIS clearly should not be matching the above lin |
| 18:10.30 | [TK]D-Fender | lien* |
| 18:10.35 | ManxPower | [TK]D-Fender: as I undertand it - is ignored in an exten |
| 18:10.51 | *** join/#asterisk ball (n=ball@adsl-99-144-221-253.dsl.emhril.sbcglobal.net) |
| 18:11.01 | [TK]D-Fender | ManxPower: Should be a literal char. |
| 18:11.11 | ball | Do I need some sort of "SIP account" to be able to call another SIP phone via the Internet? |
| 18:11.18 | ManxPower | [TK]D-Fender: it SHOULD be, but isn't. |
| 18:11.23 | ManxPower | ball: no. |
| 18:11.37 | ManxPower | ball: but it's a LOT easier if you have a provider |
| 18:11.47 | [TK]D-Fender | jeev: either way, you should not have a "-" in there, and its staring to look like you're adding stuff that shouldn't be there |
| 18:11.58 | ball | ManxPower: I would like one, but I'm not sure where to look. |
| 18:12.00 | jeev | you know fender |
| 18:12.06 | jeev | i can't find anything in the log of ALPHA, the internal asterisk server |
| 18:12.13 | jeev | could this be a failed incoming call instead? |
| 18:12.16 | ball | The SIP terminators I've found so far seem to be after corporate customers |
| 18:12.29 | jeev | fender, i dont see that call being sent from the internal asterisk server. i see EVERY other call |
| 18:12.37 | ManxPower | ~itsp |
| 18:12.37 | jbot | [~itsp] An ITSP is an Internet telephony Service Provider or "VoIP Telephone company". The allow you to either PLACE calls to the PSTN (called Termination), RECEIVE calls from the PSTN (called Origination), or both. Some offer fixed rates, others $/min. Enter ~itsplist-us (USA) or ~itsplist-ca (Canada) for a listing of popular ITSPs |
| 18:12.46 | [TK]D-Fender | jeev: you added a DASH to your pattern. Why? then look at how you chop the number that you do see clearly coming in. |
| 18:13.08 | jeev | fender, the dash is there to send extension to the database to determine whether or not they prefer blocked caller id or not. |
| 18:13.20 | [TK]D-Fender | ball: What is this "sip phone" connected to randomly over the internet? |
| 18:13.33 | [TK]D-Fender | jeev: You aren't dialing a dash anywhere |
| 18:13.42 | lmadsen | anyone know a good place that will provide London, UK DIDs who are Asterisk friendly? |
| 18:13.58 | [TK]D-Fender | lmadsen: gradwell |
| 18:14.04 | lmadsen | [TK]D-Fender: merci |
| 18:14.13 | [TK]D-Fender | lmadsen: Had happy customers of mine use them |
| 18:14.24 | lmadsen | excellent. Friend of mine was asking me for a recommendation |
| 18:14.33 | ball | [TK]D-Fender At the moment, nothing. If I can test it and I like it, I may buy termination service and use it to call telephones. I would also like to use it to call any friends I can persuade to use SIP. |
| 18:14.41 | maxxim | who is supporting H323? |
| 18:14.44 | ball | there's also a small office where I may use it. |
| 18:14.51 | ball | (perhaps with Asterisk) |
| 18:15.20 | [TK]D-Fender | ball: You asked about connecting to ANOTHER phone "out on the internet". Now that phone doesn't even exist. I don't seem to be able to figure what it is you're looking to do in the big picture here... |
| 18:15.46 | ball | [TK]D-Fender: well, first off, I'd like to test X-Lite |
| 18:16.01 | [TK]D-Fender | ball: If you jsut want to use SIP phones between you & friends, then install * and thats it. You ARE the service provider |
| 18:16.15 | [TK]D-Fender | ball: they would connect to your * for instance. |
| 18:16.26 | [TK]D-Fender | ball: no extra parties involved |
| 18:16.31 | ball | Is Asterisk horribly complex to set up though? |
| 18:16.34 | *** join/#asterisk Alan_Hicks (n=alan@cardinal.lizella.net) |
| 18:16.45 | [TK]D-Fender | ball: Not really. Go read the book, and get started |
| 18:16.48 | [TK]D-Fender | ~book |
| 18:16.49 | jbot | extra, extra, read all about it, book is Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com or see ~buybook |
| 18:16.50 | [TK]D-Fender | ^^^^^^^^^ |
| 18:17.08 | [TK]D-Fender | ball: for a sample of how simple a setup could be : |
| 18:17.10 | [TK]D-Fender | ~jerjerguide |
| 18:17.11 | jbot | [~jerjerguide ] Jeremy McNamara's quick sample guide to setting up * : http://www.jeremy-mcnamara.com/2007/02/26/how-to-configure-asterisk-your-first-installation/ |
| 18:17.13 | [TK]D-Fender | ^^^^ |
| 18:17.27 | [TK]D-Fender | ball: use that last one as "inspiration" only |
| 18:17.31 | jeev | fender, i know what you're saying. let me look at it again. |
| 18:18.08 | ball | [TK]D-Fender: I should probably test this (soft)phone - to -phone first though. |
| 18:18.26 | ball | ...although Asterisk is on my list of things to learn, I doubt that'll happen quickly. |
| 18:18.33 | jeev | exten =>_X.,1,Dial(SIP/jupiter/${CHANNEL:4:4}${EXTEN}), there is no dash. but when it's accepted, it does this exten => _XXX-1NXXNXXXXXX,1,Set(CALLERID(number)=2132222222) looks for the dash |
| 18:18.35 | jeev | correct? |
| 18:19.56 | jeev | ahh, fender, what i thought my exten => _XXX-1NXXNXXXXXX,n,Dial(SIP/viatalkhome/*67${EXTEN:4:11},60,t) removes the first 4 which would be the extension and dash.. |
| 18:20.18 | ManxPower | jeev: unless the - is not counted. JUST DON'T USE A - |
| 18:20.27 | [TK]D-Fender | jeev: I'm going to leave you to look at this for a while. |
| 18:20.43 | lmadsen | _X. and CUT() are good methods |
| 18:20.51 | [TK]D-Fender | EW |
| 18:21.11 | [TK]D-Fender | uber-catch-all's.... *shudder* |
| 18:21.14 | lmadsen | I don't even know what you're trying to do though, but it looks like using ${CHANNEL:X:Y} is dumb |
| 18:21.19 | [TK]D-Fender | runs in circles |
| 18:21.29 | lmadsen | shrugs his shoulders and goes back to paid work then |
| 18:21.41 | ball | Well, at least X-Lite doesn't just crap out like Skype did. It's supposedly SIP too, which I consider a huge plus. |
| 18:21.58 | [TK]D-Fender | ball: No, skype is its own proprietary protocol. |
| 18:22.06 | ball | nods |
| 18:22.37 | ball | [TK]D-Fender: that's one of the reasons I would prefer SIP. |
| 18:22.41 | [TK]D-Fender | ball: You certainly shold start learning with soft-phone's first and then when you're ready to look at some hardware we will help you pick out the best choice for your specific scenario. |
| 18:22.51 | [TK]D-Fender | ball: It's my rason. |
| 18:22.54 | jeev | lmadsen, i'm gonna buy you one day |
| 18:22.56 | [TK]D-Fender | reason* |
| 18:22.57 | ball | I have a soft phone in front of me now, but I don't have anyone to call. |
| 18:23.04 | lmadsen | jeev: ...I'm.... not for sale? |
| 18:23.13 | lmadsen | is not a russian mail order bride |
| 18:23.18 | russellb | lmadsen: that's not what i heard from the last astricon |
| 18:23.20 | jeev | lol |
| 18:23.23 | ball | Not sure whether I need to start opening ports on my router for this to work |
| 18:23.24 | lmadsen | russellb: oh burn |
| 18:23.26 | jeev | infact |
| 18:23.29 | [TK]D-Fender | lmadsen: probably that "buy your friends" Facebook app |
| 18:23.32 | jeev | i'm gonna hire leif for 500k/year |
| 18:23.34 | jeev | and make him mowe my lawn |
| 18:23.35 | jeev | hahahahaha |
| 18:23.41 | [TK]D-Fender | ball: bookmark this : |
| 18:23.42 | [TK]D-Fender | ~sipnat |
| 18:23.43 | 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 |
| 18:23.43 | jeev | so his asterisk skills are wasted |
| 18:23.49 | *** join/#asterisk drako (n=ljd@nelug/coreteam/luisjose) |
| 18:23.51 | [TK]D-Fender | ball: Start learning * LOCALLY first |
| 18:23.57 | lmadsen | jeev: I'd totally mow your lawn with scissors and cut each blade individually for $500k/year |
| 18:24.01 | jeev | lol |
| 18:24.14 | ball | [TK]D-Fender short of talking to the guinea pig... ;-) |
| 18:24.20 | *** part/#asterisk cgardner (n=cgardner@64.239.154.240) |
| 18:24.33 | jeev | leif, how far would you go for 500k/year |
| 18:25.12 | [TK]D-Fender | lmadsen: Don't do it!!!! |
| 18:25.14 | ball | I'm not having much luck copying these URLs. I think MacOS X has two clipboards |
| 18:25.22 | lmadsen | jeev: not that far |
| 18:25.26 | jeev | damn |
| 18:25.31 | lmadsen | probably Japan |
| 18:25.35 | [TK]D-Fender | ball: try pen & paper. Worked for centuries |
| 18:25.36 | jeev | not distance |
| 18:25.37 | lmadsen | maybe New Zealand |
| 18:25.38 | jeev | sexually |
| 18:25.39 | jeev | just kidding |
| 18:25.40 | jeev | lol |
| 18:25.44 | lmadsen | o.O |
| 18:25.50 | *** mode/#asterisk [+b %jeev!*@*] by russellb |
| 18:25.56 | jameswf-home | oh snap |
| 18:25.59 | lmadsen | indeed |
| 18:26.02 | ball | [TK]D-Fender: you chaps are typing too fast, it's all scrolling off my screen |
| 18:26.06 | russellb | jeev: once again, you have pushed the limit and you have lost your talk priviliges for another day |
| 18:26.32 | jameswf-home | lmadsen: we shall have to meet wed |
| 18:26.43 | lmadsen | jameswf-home: oh! you're going to astricon? |
| 18:26.46 | lmadsen | come see me talk at 11am |
| 18:26.47 | Alan_Hicks | I've got a perplexing problem. Even though I've added an extension to my dialplan, asterisk is telling me it can't find it. The exact error message is this... |
| 18:26.50 | Alan_Hicks | [Sep 19 14:25:47] NOTICE[7485]: chan_sip.c:14035 handle_request_invite: Call from 'cts' to extension '600' rejected because extension not found. |
| 18:27.01 | Alan_Hicks | My extensions.conf can be found here: http://lizella.net/asterisk/extensions.conf |
| 18:27.28 | jameswf-home | For a day I have to clear my calander for that wendesday seemed to be the day i could piss off the least amount of people by not existing :) |
| 18:27.50 | lmadsen | Alan_Hicks: sip set debug peer cts, then see what context it is trying to come in on, and then you can do *CLI> dialplan show 600@<context> to see if it would match |
| 18:27.53 | Alan_Hicks | I don't think this is related to my voicemail setup either. If I add an extension say... "exten => 400,s,Echo()" it will fail similarly, even after reloading the dialplan. |
| 18:27.59 | lmadsen | then you can check your diaplan to determine what you have wrong |
| 18:28.09 | Alan_Hicks | lmadsen: Ah! Good idea. I bet that's my problem... |
| 18:28.14 | lmadsen | Alan_Hicks: your phone is not matching in the right context that you expect |
| 18:28.34 | lmadsen | jameswf-home: excellent! well come to the best talk that day and see me at 11am :D |
| 18:28.34 | Alan_Hicks | lmadsen: I bet that's the case. Thanks. I'll check that. |
| 18:28.44 | ball | Any other VoIP channels on Freenode, perhaps for someone who's not quite ready for Asterisk? |
| 18:29.18 | lmadsen | dive right in! :) |
| 18:29.25 | *** join/#asterisk imcdona (i=imcdona@imcdona.broker.freenet6.net) |
| 18:29.43 | lmadsen | ball: a good place is actually to go and read the first few pages of the SIP RFC |
| 18:29.58 | Alan_Hicks | Hmmm.... No. context=internal. |
| 18:30.48 | *** join/#asterisk bbryant (n=brett@c-68-59-20-153.hsd1.sc.comcast.net) |
| 18:30.48 | ManxPower | Alan_Hicks: "s" is not a vlid priority |
| 18:32.42 | Alan_Hicks | ManxPower: Thank you. I'm a bit confused as to why it's a valid priority in some cases and not in others. |
| 18:33.01 | ball | lmadsen: thanks, I'll have a look at that now. |
| 18:34.48 | jaytee | ManxPower, what's your preferred editor for the xml config files for Polycom? |
| 18:35.59 | Alan_Hicks | ManxPower: Ah! Now I see where I was confused. "s" is a valid _extension_, not a valid _priority_. |
| 18:37.28 | lmadsen | Alan_Hicks: technically 's' is also a valid priority :) |
| 18:37.39 | lmadsen | 's' means 'same' but is used VERY rarely |
| 18:38.11 | Alan_Hicks | lmadsen: In other words, n00bs like me shouldn't ever use it. :-) |
| 18:38.32 | lmadsen | Alan_Hicks: I think I've only used it once in my lifetime :) |
| 18:39.02 | lmadsen | Alan_Hicks: it just isn't necessary except for very special circumstances, and I can't even remember what I used it for, so I can't provide you an example :) |
| 18:39.26 | [TK]D-Fender | lmadsen: I didn't know that one... |
| 18:39.32 | Alan_Hicks | No problem. Thanks anyway. |
| 18:39.54 | lmadsen | [TK]D-Fender: ya, I'm not even sure its documented anywhere really... I used it once, and really wish I could remember what for, because it was neat |
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| 18:43.52 | *** join/#asterisk Specialist1 (n=me@119.160.105.172) |
| 18:43.55 | Specialist1 | hi everyone |
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| 19:10.49 | *** join/#asterisk JakBeatZ (n=jircii@206.223.182.67) |
| 19:14.21 | JakBeatZ | Looking for some perspectives on a weird startup error. Asterisk 1.2.26.1 complains about permission denied writing to the /var/log/asterisk/full logfile. asterisk is starting with -U asterisk -G asterisk and /var/log/asterisk is asterisk:asterisk, mode 770 so I have no idea where the permission denied is coming from. |
| 19:16.01 | *** join/#asterisk mvanbaak (n=michiel@asterisk/contributor-and-bug-marshal/mvanbaak) |
| 19:16.11 | maxxim | is it possible to register dynamicly sip users in asterisk, via quering an existing mysql database using custom sql selects? |
| 19:16.52 | *** join/#asterisk tobias (n=tobias@rrcs-70-62-101-155.midsouth.biz.rr.com) |
| 19:17.14 | ManxPower | maxxim: it is called ARA, Asterisk Realtime Architecture. Look it up. |
| 19:17.46 | ManxPower | JakBeatZ: what is the permissions of var/log/asterisk/full |
| 19:18.07 | JakBeatZ | ManxPower: 660 asterisk:asterisk |
| 19:18.12 | maxxim | i'm looking at it, but as i read ARA is working just with specifict asterisk tables... but i need to make an custom select on already existing datavase |
| 19:18.16 | maxxim | is it possible? |
| 19:18.40 | ManxPower | maxxim: you would have to change the source code |
| 19:18.50 | maxxim | good point |
| 19:18.52 | maxxim | thanks |
| 19:18.53 | [TK]D-Fender | maxxim: "host=dynamic" |
| 19:19.15 | ManxPower | [TK]D-Fender: how does that let him use a custom database as the Realtime backend? |
| 19:19.27 | maxxim | [TK]D-Fender> i need the similar like in gnugk. Query=Select ....... |
| 19:19.51 | *** part/#asterisk eliel (n=eliel@200.61.172.61) |
| 19:19.53 | maxxim | this is why i'm forced to use gnugk with h323 :( |
| 19:20.02 | JakBeatZ | ManxPower: Actually, I deleted the full log in /var/log/asterisk hopeing the startup would recreate it, but it doesn't. Even if I touch /var/log/asterisk/full ; chmod 660 /var/log/astrisk/full ; chown asterisk:asterisk /var/log/asterisk/full and restart, I still get the permissions error |
| 19:20.08 | maxxim | i would like to use SIP, if i could make something like this.... |
| 19:20.13 | [TK]D-Fender | Maybe some people should be using a a database that supports views & triggers & stored procedures... |
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| 19:25.55 | maxxim | what is the differense between: sipusers and sippeers? |
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| 19:52.07 | neurosys | can someone tell me what this extra crap on the end of this variable represents? ${EXTEN:@$} |
| 19:53.25 | [TK]D-Fender | neurosys: that is invalid. for proper use go read CHANNELVARIABLES.TXT |
| 19:54.01 | neurosys | [TK]D-Fender: Thats the crap the ITSP gave me to dial out with in thier Dial statement :( |
| 19:54.56 | [TK]D-Fender | neurosys: well this is Asterisk variables 101. Go give that doc a good read to understand what you can do and think abot what you want to do. |
| 19:54.57 | [intra]lanman | someone was probably holding the shift when they shouldn't have been |
| 19:55.10 | neurosys | heh |
| 19:55.10 | neurosys | exten => _x.,1,Dial(SIP/${EXTEN:@$}sip.quivoice.it,,tt) |
| 19:56.29 | [TK]D-Fender | neurosys: Please go read the doc I referred you to. |
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| 20:05.36 | smth | is there anyway to improve the asterisk's DTMF detection? Sometimes entering DTMF digits quickly would cause that asterisk read them in a wrong order when calling from sip phone out through asterisk. |
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| 20:05.59 | Katty | does call parking play the same thing as on hold music? |
| 20:06.03 | [TK]D-Fender | smth: what version are you using? |
| 20:06.11 | [TK]D-Fender | Katty: Yes |
| 20:06.15 | Katty | for some weird reason, call parking plays the music--but on hold doesn't play anything |
| 20:06.21 | smth | 1.4.20.1 |
| 20:06.34 | [TK]D-Fender | smth: you can try "relaxdtmf=yes |
| 20:06.41 | smth | I did |
| 20:06.54 | [TK]D-Fender | kayyProbably a question of what class is specified |
| 20:07.14 | [TK]D-Fender | smth: What phone are you using? |
| 20:07.19 | Katty | [TK]D-Fender: hmm. |
| 20:07.23 | Katty | [TK]D-Fender: you mean class=default |
| 20:07.46 | smth | I tried on linksys and a analog phone |
| 20:08.09 | [TK]D-Fender | smth: And how was the analog phone connected to *? |
| 20:08.38 | [TK]D-Fender | smth: Because a Linksys hard phone should be using OOB DTMF which is not "detected". |
| 20:09.27 | smth | analog connect to fxs . call out through fxo |
| 20:10.15 | smth | I use analog port as a peer |
| 20:11.59 | [TK]D-Fender | smth: what FXS? what FXO? |
| 20:12.52 | smth | s for analog phone connection . o for phone line |
| 20:13.19 | [TK]D-Fender | smth: No, what piece of HARDWARE do you have to phone plugged into. The model is IMPORTNAT. |
| 20:13.27 | [TK]D-Fender | IMPORTANT even |
| 20:13.54 | smth | sangoma |
| 20:14.27 | *** part/#asterisk Deeewayne (n=Deeewayn@nat/digium/x-487904ba15f42277) |
| 20:15.07 | jblack | Does anyone know of a bulk provider they'd care to recommend? |
| 20:16.37 | [TK]D-Fender | smth: better... ok, you might want to try playing a little with your gains. thats about all aI can advise for that one since you've tried "relaxdtmf" |
| 20:16.55 | smth | actually I found that asterisk read a wrong-order dtmf digits from debug messages if I dial fast. |
| 20:17.07 | jblack | I'm looking at 1400-2000 minutes a day. |
| 20:20.30 | jaytee | if I'm trying to enter the address of my FTP server manually on a Polycom what do I press to get a period between the octets? |
| 20:21.09 | [TK]D-Fender | * |
| 20:21.56 | jaytee | that's what I thought but I couldn't find the section of in the Admin guide where I'd read that before. |
| 20:22.03 | jaytee | thanks, [TK]D-Fender |
| 20:23.05 | [TK]D-Fender | Single-char help. The bestest. |
| 20:27.22 | [TK]D-Fender | Checkout tie... heading home... |
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| 20:39.36 | *** join/#asterisk elkatoxx (n=pidgin@190.2.33.4) |
| 20:40.11 | elkatoxx | Hi, can anyone help with no audio in one way problem? |
| 20:40.58 | elkatoxx | when i call anybody out, i don't listen to the other side. The other side listen to me perfectly. |
| 20:41.31 | elkatoxx | i checked and rechecked NAT and RTP, i really don't know it could be |
| 20:42.44 | smth | [TK]D-Fender, my sip phone work with 2833 dtmf. it works ok if dialing in normal speed. but sometimes I dial the numbers at a fast speed. astereisk pass by wrong order to destination. |
| 20:43.23 | smth | even I used sip trunk . sip issues occured |
| 20:44.09 | jameswf-home | elkatoxx: canreinvite=no |
| 20:45.44 | smth | 'sip' is same |
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| 20:49.16 | jameswf-home | I should get a couple unlimited sip channels and let people do irc originates from them.... |
| 20:49.27 | lmadsen | eh? |
| 20:49.46 | lmadsen | like you mean to place a test call? |
| 20:49.52 | jameswf-home | badidabot: call userA and userB |
| 20:50.18 | jameswf-home | liks a FFA click to call from an irc bot |
| 20:50.52 | lmadsen | oh neat idea |
| 20:51.23 | lmadsen | jbot: call jameswf-home at sip:ivr@pbx.leifmadsen.com |
| 20:51.23 | jbot | ACTION looks around and then screams out jameswf-home at sip:ivr@pbx.leifmadsen.com as loudly as possible |
| 20:51.31 | lmadsen | heh |
| 20:51.46 | jameswf-home | I have irc bots on all of our testlab boxes so I can pop open an irc window and ask each their status and a few other things wouldnt take much to add click to call |
| 20:51.57 | lmadsen | hawtness |
| 20:51.59 | jameswf-home | heh |
| 20:52.15 | lmadsen | yay! I'm calling it a work day done |
| 20:52.26 | lmadsen | 10 mins early none the less |
| 20:52.34 | lmadsen | however, I'm the boss, so what's he gonna do? |
| 20:52.39 | jameswf-home | its only 2 here |
| 20:53.07 | lmadsen | ouch |
| 20:53.09 | lmadsen | almost 5 here |
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| 21:02.10 | elkatoxx | jameswf, thanks, i have that configuration, but doesn't work |
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| 21:03.11 | m2oluf | anyone who can help get my hfc based card up running? |
| 21:04.27 | elkatoxx | jameswf, and it doesn't happend always. some calls yes, some not |
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| 21:04.46 | *** mode/#asterisk [+o russellb] by ChanServ |
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| 21:23.13 | henkoegema | has asterisk still not released official version 1.6.0 ? |
| 21:23.30 | russellb | correct. |
| 21:23.38 | russellb | it's ready, just waiting on DAHDI 2.0.0 ... |
| 21:24.13 | henkoegema | i'm still using 1.4.11 and want to try a new one |
| 21:24.59 | jameswf-home | henkoegema: is your asterisk broke? |
| 21:25.43 | russellb | it is most likely vulnerable to some security issues that have been fixed since 1.4.11 ... |
| 21:25.48 | russellb | if exposed to the internet |
| 21:25.53 | [TK]D-Fender | russellb: so the latest SVN over the latest RC is teetering on release-ready? |
| 21:26.08 | henkoegema | no, but I like to play around with asterisk. have been using the old version for a long time now |
| 21:26.35 | russellb | [TK]D-Fender: yeah. the latest svn doesn't have much over the latest RC. So, yeah, it's ready to roll |
| 21:26.59 | russellb | stupid ... zaptel dahdi crap ... |
| 21:27.03 | [TK]D-Fender | russellb: As usual I won't hold my breath, but Will keep an ey open |
| 21:27.06 | russellb | nods |
| 21:27.07 | russellb | :) |
| 21:28.44 | jameswf-home | russellb: do you know if 11243 was fixed 3 months ago why it would still be an issue? |
| 21:29.12 | russellb | eh? |
| 21:29.33 | russellb | jameswf-home: using a version that includes the fix? |
| 21:29.39 | jameswf-home | on dbian i have to compile with do not optimize or gsm is total crap |
| 21:29.47 | jameswf-home | 1.4.21.2 |
| 21:29.54 | russellb | it's quite possible that version does not have the fix |
| 21:29.56 | russellb | try 1.4.22 |
| 21:29.58 | russellb | and let me know |
| 21:30.59 | *** part/#asterisk asteriskmonkey (n=asterisk@69.77.169.14) |
| 21:31.21 | Dr-Linux|home | russellb: Is there any chance to introduce something alternative to chan_agent? |
| 21:31.28 | jameswf-home | have to do it when the wife isnt looking so she doesnt give me the uh its down again look |
| 21:33.08 | *** join/#asterisk mvanbaak (n=michiel@asterisk/contributor-and-bug-marshal/mvanbaak) |
| 21:35.14 | jameswf-home | the wife doesnt understand her phone is my toy |
| 21:35.44 | russellb | Dr-Linux|home: file is working on a replacement |
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| 21:36.45 | Dr-Linux|home | how long it will take .. maybe i'll discuss with file .. incuding Lumenvox SR |
| 21:37.05 | Dr-Linux|home | we are very fedup with chan_agent |
| 21:37.55 | x86 | what does chan_agent do? |
| 21:39.21 | jjshoe | what does teh google doo? |
| 21:39.34 | elkatoxx | hehe... |
| 21:39.39 | LemensTS | http://www.voip-info.org/wiki/view/Agents+without+agent+channel |
| 21:40.35 | russellb | Dr-Linux|home: use dynamic queue members, then ... |
| 21:40.41 | x86 | jjshoe: it makes people like me ask questions, and people like you point at it ;) |
| 21:40.44 | russellb | it's documented in the doc dir of 1.4 .. |
| 21:42.09 | seanbright | documented schmocumented |
| 21:42.23 | Dr-Linux|home | russellb: since we are using web application with chan_agent, so AddQueueMember won't help for events etc |
| 21:43.38 | russellb | so you want something completely different, as long as it requires no work on your part to switch to it? |
| 21:43.41 | russellb | picky picky! |
| 21:43.42 | russellb | :-p |
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| 21:49.02 | *** join/#asterisk rigel (n=rigel@c-98-210-196-128.hsd1.ca.comcast.net) |
| 21:49.37 | rigel | i'm interested in setting up voice recognition |
| 21:49.44 | rigel | and was told to start here |
| 21:49.50 | rigel | but i dont have a landline |
| 21:49.56 | rigel | so i dont know why i would use asterisk |
| 21:50.08 | jblack | sounds like the wrong channel to me. |
| 21:50.10 | russellb | voice recognition in combination with telephony? |
| 21:50.15 | russellb | or just voice recognition stuff in general ... |
| 21:50.28 | russellb | if you're looking for something in general, check out speex for something open source ... |
| 21:50.36 | russellb | lumenvox is the way to for something with asterisk |
| 21:50.51 | rigel | well, it was suggested that i start with the particular case, setting up voice recognition with asterisk, and then proceed to the general case of voice recognition |
| 21:50.58 | russellb | i see. |
| 21:51.01 | rigel | because the sphinx tutorials, etc, i cant make heads or tails of |
| 21:51.19 | rigel | and apparently the asterisk howtos and information is more geared toward the end user |
| 21:51.42 | rigel | novice, whatever |
| 21:51.49 | russellb | the asterisk stuff is very tied to asterisk |
| 21:52.07 | rigel | hrm |
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| 22:02.11 | mchou | there's a big difference between voice rcognition and natural language processing |
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| 22:14.58 | elkatoxx | any can help me with NAT and RTP? |
| 22:16.28 | seanbright | ~sipnat |
| 22:16.29 | 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 |
| 22:16.40 | seanbright | that'll be $125 |
| 22:18.31 | elkatoxx | thanks seanbright, but i have another model |
| 22:19.20 | elkatoxx | i have this: [SIP provider] ---------- inet ------ > (my building) [iptables FW] ----- * ------ ATAs --- phone |
| 22:20.12 | elkatoxx | i have configured just like tutorials, but i'm having calls without audio in one ways sometimes |
| 22:21.07 | elkatoxx | at FW i configured letting in IP Provider udp 5060 and the range of ports RTP (10000:10100) |
| 22:21.31 | elkatoxx | at * i configured, nat=yes, canreinvite=no |
| 22:22.10 | [TK]D-Fender | ~sipnat |
| 22:22.11 | 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 |
| 22:22.25 | elkatoxx | ok, thanks for listening.. |
| 22:24.11 | [TK]D-Fender | elkatoxx: Is * on a public IP, or is that IP tables doing NAT and on another box? |
| 22:24.37 | elkatoxx | * have private IP, FW is masquerading |
| 22:25.20 | elkatoxx | *, ATAs, and siphones are in the same LAN |
| 22:25.22 | mchou | elkatoxx: that's a standard setup I'm sure virtually everyone here uses |
| 22:25.47 | [TK]D-Fender | elkatoxx: then pastebin your sip.conf masking only passwords. |
| 22:25.49 | [TK]D-Fender | ~pb |
| 22:25.49 | jbot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste |
| 22:25.54 | [TK]D-Fender | ^^^ |
| 22:26.12 | elkatoxx | i'm doing... |
| 22:26.13 | mchou | elkatoxx: make sure canreinvite=no, externip and locatnet are set |
| 22:26.24 | mchou | localnet* |
| 22:26.37 | nosbig | elkatoxx, You have to verify that the RTP ports being forwarded are the ones the provider is sending... |
| 22:26.39 | seanbright | mchou: you mean just like described at ~sipnat? |
| 22:26.44 | seanbright | mchou: yes, that is something to make sure of. |
| 22:27.35 | mchou | seanbright: there is no need to be patronizing |
| 22:27.44 | elkatoxx | i have configures externalip and localnet |
| 22:27.58 | seanbright | mchou: sure there is, it's friday. |
| 22:28.07 | nosbig | elkatoxx, For example, bandwidth.com (who actually recommend putting your phone system on a public, non-NATted IP, state that incoming RTP could be on any port from 1024-64000, requiring massive holes in my firewall... |
| 22:28.16 | mchou | seanbright: stfu |
| 22:28.56 | [TK]D-Fender | elkatoxx: ...pastebin please... |
| 22:28.58 | nosbig | elkatoxx, The parameter is called externip... * will not know what to do with an option called externalip... ;-) |
| 22:29.40 | elkatoxx | http://pastebin.com/m72391038 |
| 22:29.42 | seanbright | and don't forget about everything else listed at http://www.aocomputing.net/?p=3 |
| 22:30.00 | elkatoxx | sorry nosbig, it's externip i confirm |
| 22:30.55 | [TK]D-Fender | elkatoxx: your "register" must come AFTER everything else in [general] |
| 22:31.01 | [TK]D-Fender | elkatoxx: Change the order |
| 22:31.07 | elkatoxx | ok, but may be i don't understand. I need to forward incoming UDP in range of 1024:64000? all that? |
| 22:31.21 | elkatoxx | i'm changing |
| 22:31.37 | [TK]D-Fender | elkatoxx: and ITSP peer entries should be listed as nat=no" |
| 22:32.23 | nosbig | np, just throwing that out there... |
| 22:32.45 | elkatoxx | Fender, the sipphones? like 201? |
| 22:33.02 | elkatoxx | oh ok.. the provider? |
| 22:33.15 | nosbig | elkatoxx, every SIP provider out there will have different port ranges... Ask your vendor... For my vendor, it is 1024-64000... |
| 22:33.45 | elkatoxx | ok, but it's not what i configure in rtp.conf, right? so it need to be wide.. |
| 22:33.58 | mchou | nosbig: I think you're mistaken about how RTP gets negotiated. you dont need to open "everything" |
| 22:34.57 | [TK]D-Fender | elkatoxx: just your ITSP peers. remote phones you typically leave as "nat=yes". this was all in the guide. |
| 22:35.00 | mchou | nosbig: in fact it doesnt matter what range the ITSP uses (as long as your asterisk box uses a subset in that range) |
| 22:35.03 | [TK]D-Fender | elkatoxx: Test & report |
| 22:35.38 | elkatoxx | i;m changing... |
| 22:35.49 | mchou | nosbig: during SIP call setup RTP ports have to be negotiated and agreed up |
| 22:35.54 | mchou | upon* |
| 22:36.48 | mchou | nosbig: so if you're opening all udp ports on your firewall from 1024-64000 you're doing it wrong |
| 22:37.18 | elkatoxx | Fender... u r the man... |
| 22:37.28 | elkatoxx | i test it, and working |
| 22:37.45 | elkatoxx | just change nat=yes to no at the peer |
| 22:39.13 | elkatoxx | i'll be testing |
| 22:39.37 | elkatoxx | thanks to everyone! |
| 22:43.03 | elkatoxx | testing, i have the problem again. Some outgoing calls, doesn't receive audio. But the other end listen to me perfectly |
| 22:43.58 | nosbig | mchou: Really??? Pardon my ignorance, but I have not studied the details of SIP and RTP closely enough... I guess that would be the point of SIP, to negotiate all aspects of the upcoming RTP stream(s)... |
| 22:44.14 | mchou | nosbig: bingo |
| 22:45.23 | *** join/#asterisk jaytee (n=jforde05@unaffiliated/jaytee) |
| 22:45.37 | elkatoxx | true.. |
| 22:45.49 | elkatoxx | this it my last config: http://pastebin.com/d10532846 |
| 22:45.55 | elkatoxx | sip.conf |
| 22:46.24 | *** join/#asterisk d3wayne (n=dwayne@76.29.245.9) |
| 22:46.24 | *** mode/#asterisk [+o d3wayne] by ChanServ |
| 22:47.05 | nosbig | mchou, I will have to verify using some packet captures.... Just my scientific nature... But I bet I will get to restrict that range significantly... I do believe those holes have been causing some intermittent issues with my Internet connectivity due to random port scanning and such... ;-) My CPU utilization on my router has been at 16%, which is much higher than it was before the change to the firewall... |
| 22:47.14 | [TK]D-Fender | elkatoxx: enable sip debug and pastebin a failed call. |
| 22:47.33 | elkatoxx | ok... |
| 22:48.15 | mchou | nosbig: you dont need a packet capture. just turn sip debug on during a call and you'll see rtp ports being negotiated |
| 22:48.25 | mchou | nosbig: along with the codec |
| 22:48.49 | mchou | it's all part of the same process |
| 22:49.14 | mchou | I mean right _before_ a call |
| 22:49.21 | mchou | not during :) |
| 22:49.44 | nosbig | I guess that is true.... lol I assumed as much... Kinda pointless to negotiate a call if it is already set up... ;-) |
| 22:53.12 | elkatoxx | ok, i needed to test 6 calls. the 7th with no audio i will paste |
| 22:53.23 | *** join/#asterisk [intra]lanman (n=lanman@freeswitch/developer/intralanman) |
| 22:55.24 | Defraz | so has anyone successfully compiled asterisk 1.4 and have rxfax working with any version of spandsp. |
| 22:55.28 | Defraz | mine crashes asterisk. |
| 22:56.36 | *** join/#asterisk mcab (n=mb@mostly-harmless.ca) |
| 22:59.19 | elkatoxx | D-Fender, here is the sip debug about failed call: http://pastebin.com/d44f76ab1 |
| 23:00.31 | nosbig | See, this is why I love this chat room... I assert my arrogance and say something stupid, and someone calls me on it, politely, of course... ;-) |
| 23:01.36 | *** join/#asterisk [gquit]bombadil (n=dana@CPE-70-94-44-157.wi.res.rr.com) |
| 23:03.39 | *** join/#asterisk mvanbaak (n=michiel@asterisk/contributor-and-bug-marshal/mvanbaak) |
| 23:06.56 | jaytee | [TK]D-Fender, I got the provisioning setup working with DHCP in our Windows domain using DHCP from our domain controllers and got it to download the configs from one of my asterisk servers I setup as the provisioning server. |
| 23:07.18 | *** join/#asterisk coolthreads (n=shane@203-97-238-71.cable.telstraclear.net) |
| 23:07.41 | jaytee | and with the new configs I got ring-answer working from my dialplan if I add an 8 to any extension. |
| 23:07.45 | [TK]D-Fender | elkatoxx: Ok, I can't readily see the prblem. |
| 23:08.00 | [TK]D-Fender | jaytee: Most excellent |
| 23:08.04 | elkatoxx | me nether |
| 23:08.58 | jaytee | [TK]D-Fender, one quick question, if I don't modify the configs and I reboot the phone does it take less time to boot the second time around since I'm using FTP? |
| 23:09.15 | elkatoxx | may be this line: Got unsupported a:fmtp in SDP offer ? |
| 23:09.41 | [TK]D-Fender | jaytee: only time diff is if you're loading a new ROM. settings is the same |
| 23:09.57 | jaytee | [TK]D-Fender, cool |
| 23:10.38 | elkatoxx | i have defined int rtp.conf start/end 10600/10650 |
| 23:10.52 | jaytee | [TK]D-Fender, I don't think I need the FTP server address setup on the phone though since I'm passing the boot server address in Option 66. |
| 23:12.22 | [TK]D-Fender | elkatoxx: try leaving the range the natural 10000-20000 and configure to match and test from there |
| 23:12.41 | [TK]D-Fender | jaytee: to grab from 66 you need the boot server to be 0.0.0.0 |
| 23:12.52 | [TK]D-Fender | jaytee: and the use of 66 selected on the phone |
| 23:13.37 | *** join/#asterisk Gary (i=gary@freenode/staff/colchester-lug.gary) |
| 23:13.51 | jaytee | maybe if you're using the asterisk server as the DHCP server but I'm not. |
| 23:14.09 | *** join/#asterisk voxter (n=voxter@S01060016b6b53c0c.vc.shawcable.net) |
| 23:14.14 | elkatoxx | D-Fender, the line 175. it is correct? don't need to be NAT? |
| 23:14.42 | *** join/#asterisk CunningPike (n=arodgers@S010600095b33697f.vc.shawcable.net) |
| 23:14.43 | voxter | so, trying to do a chanspy, i'm being told "cannot provide audio mixing on 'ulaw'-'ulaw' format spies. Doesnt make sense. |
| 23:14.48 | voxter | anyone have insight on that? |
| 23:16.04 | *** join/#asterisk CunningPike (n=arodgers@vpn.dnv.org) |
| 23:17.38 | [TK]D-Fender | elkatoxx: no, thats proper. you need to trust the IP they tell you to use |
| 23:18.01 | [TK]D-Fender | elkatoxx: if your failure is inconsistent it might be your providers fault. |
| 23:18.38 | *** join/#asterisk tobias (n=tobias@user-0c2hj2f.cable.mindspring.com) |
| 23:19.09 | *** join/#asterisk imcdona (i=imcdona@imcdona.broker.freenet6.net) |
| 23:19.48 | *** join/#asterisk CrazyTux[m] (n=CrazyTux@ip68-111-67-4.oc.oc.cox.net) |
| 23:19.49 | elkatoxx | the service was originally with a SIPURA ATA, but i changed it with the asterisk. Not support from the provider :( |
| 23:23.49 | *** join/#asterisk dramman (i=7a6f389e@gateway/web/ajax/mibbit.com/x-56fb8fde828201d1) |
| 23:25.13 | elkatoxx | well D-Fender and rest of you, i need to go. I will continue with this on monday. Thanks to all of you for the support! ;) |
| 23:25.15 | elkatoxx | c ya |
| 23:26.11 | dramman | When I try to connect to my local asterisk server (which i'm trying to configure) from a softphone asterisk says "not a local domain" - how do I fix this? |
| 23:26.33 | dramman | registration is successful only if I don't subscribe for incoming calls |
| 23:33.37 | dramman | can't figure this out |
| 23:36.09 | jblack | Bleh. future-nine nine not only supports just SIP, but they send email in html format and their only US POP is in the bottom right of the US. |
| 23:36.39 | jblack | I can understand New York or Los Angeles, but Miami? |
| 23:37.29 | jblack | Great latency for Mickey Mouse and aligators, I suppose. |
| 23:41.28 | dramman | How do I best configure local extension/users with a username like "bob" and an extension "10"? |
| 23:41.55 | *** join/#asterisk Daviey (n=dave@ubuntu/member/daviey) |
| 23:55.13 | *** join/#asterisk kamanashisroy (n=kamanash@119.30.34.3) |