| 00:07.00 | *** join/#asterisk mindCrime_ (n=chatzill@cpe-065-190-188-124.nc.res.rr.com) |
| 00:27.31 | *** join/#asterisk ingenio (n=ingenio@12-216-99-16.client.mchsi.com) |
| 00:28.29 | ingenio | so i'm looking at a tdm02b with a linksys spa942 for my soho.. but i'm confused about the PoE. does the tdm output or do I need to buy an additional switch/router/hub ? |
| 00:30.07 | xuser | anyone used a Orion gsm gateway? |
| 00:30.15 | *** join/#asterisk ThOr101 (n=bthorson@pool-71-126-163-76.washdc.fios.verizon.net) |
| 00:31.18 | ThOr101 | As per the "~book" I tried to setup * with FreeWorldDialup. I'm using 1.4.4 is this doomed to failure? I don't really need this to work, I just wanted to go step by step through the book. I get a Registration Refused but my number & password are correct? Dig deeper or move on and learn FWD is junk? |
| 00:32.51 | [TK]D-Fender | ingenio, those 2 items have nothing to do with each other |
| 00:33.20 | [TK]D-Fender | ThOr101, Nothing "wrong" with FWD,a nd yes, it can work fine. |
| 00:33.39 | [TK]D-Fender | ThOr101, Good for calling people on it (if you know anyone). |
| 00:35.36 | ThOr101 | http://paste.debian.net/28873 |
| 00:36.13 | stridernzl | [TK}D-Fender: I played around a bit more ... went through tuning wizard in Eyebeam, managed to cancel out the radio :) , didn't really like or want to touch the Video Tuning .. but had to :( .. made another 1 call and seemed to have a bit of echo ? But it worked prob 80% .. so thats kind of good and ongoing! |
| 00:36.46 | [TK]D-Fender | stridernzl, Many factors to consider |
| 00:37.33 | *** join/#asterisk CrazyTux[m] (n=CrazyTux@cpe-24-162-48-94.houston.res.rr.com) |
| 00:37.38 | [TK]D-Fender | ThOr101, I've never used IAX to connect to FWD before... |
| 00:37.57 | *** part/#asterisk csd-199 (n=adsf@189.158.190.64) |
| 00:38.13 | ThOr101 | Can you suggest another free service that could potentially be used to call back into my box from a softphone? |
| 00:38.53 | stridernzl | [TK]D-Fender: oh yeah for sure :), but still I feel I'm on the right track .. no thanks to you as well. |
| 00:38.56 | ThOr101 | Hmm, I suppose I really don't need a service, if I forward the appropriate ports on the firewall |
| 00:40.45 | [TK]D-Fender | ThOr101, the error does say 1 of your suer/pass is wrong |
| 00:40.49 | [TK]D-Fender | user* |
| 00:40.59 | ThOr101 | Yeah, I checked that, and changed it several times. |
| 00:41.15 | ThOr101 | Even took a break (bite to eat) came back and checked it again. |
| 00:42.22 | [TK]D-Fender | ThOr101, make sure you also have "allow=ulaw" in [general] as well... |
| 00:42.34 | [TK]D-Fender | ThOr101, "reload iax2" |
| 00:42.53 | ingenio | [TK]D-Fender: i can't use a tdm02b with a linksys spa942 phone?! |
| 00:43.04 | ThOr101 | yeah got that, and with my problems from before, I've just started killing and restarting all of asterisk |
| 00:43.32 | ThOr101 | reload chan_iax2 |
| 00:44.03 | [TK]D-Fender | ingenio, Sure, you can use both, but the 2 have nothing to do with one another. PoE is 1 way to power the phone (if you have an injector or a PoE switch), the other is a tyipcal Power brick. |
| 00:44.14 | CrazyTux | [TK]D-Fender, do you happen to know if you can store the .txt files in /var/spool/asterisk/voicemail/INBOX/DID in a more realtime manner? I.e. database. |
| 00:44.20 | [TK]D-Fender | ingenio, Where are you located? |
| 00:45.19 | [TK]D-Fender | CrazyTux[m], Look at realtime voicemail on the WIKI |
| 00:45.19 | CrazyTux | [TK]D-Fender, I have the mailboxe's etc realtime, just not the message information |
| 00:46.33 | [TK]D-Fender | CrazyTux[m], Ok, well I can't advise more on that front as my installs are using standard storage |
| 00:47.55 | *** join/#asterisk caio1982 (i=caio1982@CAcert-br/caio1982) |
| 00:50.34 | ingenio | iowa, you? |
| 00:50.34 | ingenio | brb |
| 00:52.48 | [TK]D-Fender | ingenio, In that case, forget the Linksys phones and get a Polycom instead, much better option. |
| 00:56.44 | *** join/#asterisk samy_b1 (n=baind@2001:49f0:1000:0:0:0:0:7) |
| 00:56.52 | samy_b1 | hey all |
| 00:57.49 | samy_b1 | can some one tell me way i'm geting "Got SIP response 302 "Moved Temporarily" back from" msg? |
| 00:58.21 | samy_b1 | when i try to dial out true my sip trunk |
| 00:58.45 | bkw_ | the sip trunk sent a 302 redirect |
| 01:02.05 | *** join/#asterisk Zipper_32 (n=None@s207-6-25-182.bc.hsia.telus.net) |
| 01:04.13 | Zipper_32 | Is anybody familiar with, or know why transfering between my new Polycom IP430's results in no ringing tone to the transferred caller to the destination? If I transfer with a different brand phone, the caller hears a ringing tone when he is being transfered, however, transferring with an IP430, the caller does not hear the ring tone while the transferred phone is ringing. |
| 01:04.30 | samy_b1 | bkw_: is it somting that i did |
| 01:04.54 | samy_b1 | or is from my DID sip |
| 01:04.56 | samy_b1 | ? |
| 01:06.05 | samy_b1 | i'm just trying to solve that problem |
| 01:07.35 | *** part/#asterisk mog (n=mog@c-71-207-214-192.hsd1.al.comcast.net) |
| 01:11.25 | *** join/#asterisk SirThomas_Home (n=tomc@209-169-199-174.us.warpdriveonline.com) |
| 01:24.58 | *** part/#asterisk stevej (n=stevej@mail.joneslinux.com) |
| 01:28.41 | *** join/#asterisk Lloydie-t (n=chatzill@thomasclan.plus.com) |
| 01:30.00 | *** join/#asterisk mvand (n=mvand@CPE-65-28-181-127.neb.res.rr.com) |
| 01:30.42 | Lloydie-t | Does anyone know of a problem with * CDR showing the wrong duration for outgoing calls on IAX2 channels? |
| 01:38.44 | *** join/#asterisk mightnare (n=mike@s230165.ppp.asahi-net.or.jp) |
| 01:38.55 | *** join/#asterisk mog (n=mog@c-71-207-214-192.hsd1.al.comcast.net) |
| 01:38.55 | *** mode/#asterisk [+o mog] by ChanServ |
| 01:42.45 | *** join/#asterisk n00dle (n=ccraft@ip-249-27.springsips.com) |
| 01:42.50 | *** part/#asterisk mog (n=mog@c-71-207-214-192.hsd1.al.comcast.net) |
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| 01:44.48 | *** mode/#asterisk [+o mog] by ChanServ |
| 01:45.46 | *** join/#asterisk sakic (n=sakic@cpe-071-075-118-121.carolina.res.rr.com) |
| 01:46.10 | sakic | I can't seem to get asterisk installed on ubuntu, should I use a different linux? |
| 01:46.54 | *** join/#asterisk steliosk (n=Stelios@ipa226.211.tellas.gr) |
| 01:47.00 | Nugget | that's not likely to help. |
| 01:47.19 | Nugget | installing asterisk is pretty much exactly the same no matter what unix you're using |
| 01:47.26 | sakic | hmm |
| 01:47.37 | sakic | I was following a tutorial: http://blog.thegoldfish.net/asterisk-with-freepbx-on-ubuntu-704-desktop-tutorial/ |
| 01:48.04 | Nugget | freepbx is something entirely different. we can't really help with that here. |
| 01:48.15 | Nugget | I think there's a #freepbx if that's really the route you want to go |
| 01:48.17 | n00dle | sakic, I installed asterisk on ubuntu just fine. |
| 01:48.24 | n00dle | ...no freepbx, though. |
| 01:48.29 | sakic | well I am cool w/o freepbx |
| 01:48.39 | sakic | n00dle where did you get instructions? |
| 01:49.04 | *** join/#asterisk kuku5 (n=kuku5@c-71-201-219-72.hsd1.il.comcast.net) |
| 01:49.11 | kuku5 | May 25 20:47:32 WARNING[21780]: chan_iax2.c:7118 socket_read: Call rejected by 64.61.93.109: No such context/extension << someone please help |
| 01:49.23 | sakic | make clean; make install gives me error :( |
| 01:49.39 | Zipper_32 | kuku5: I'd probably focus on the "No such context/extension" part of that error. |
| 01:49.46 | n00dle | I googled and found some hints on ubuntu forums. Some hints: you have to install kernel source, kernel headers and such. |
| 01:50.04 | Nugget | that's really only if you need zaptel. |
| 01:50.07 | mightnare | sakic: what error did you get exactly? |
| 01:50.22 | Nugget | obviously asterisk doesn't need kernel source or headers. |
| 01:50.25 | kuku5 | Zipper_32: Executing Dial("SIP/saturn110-b7a096e0", "IAX2/voicepulse03/7735445655") in new stack |
| 01:50.39 | kuku5 | Zipper_32: Its weird. I get incomign calls from them fine. |
| 01:50.39 | sakic | configure script must be executed before running make |
| 01:50.53 | kuku5 | Zipper_32: what does it mean "no context/extension" ? |
| 01:50.56 | sakic | FYI: this is my first attempt w/ linux too |
| 01:50.57 | Zipper_32 | kuku5: it can't find: 7735445655 when trying to dial out. |
| 01:51.08 | n00dle | sakic, "./configure && make && make install" |
| 01:51.14 | *** part/#asterisk BSD_Tech (n=BSDTech@adsl-69-230-174-37.dsl.irvnca.pacbell.net) |
| 01:51.28 | kuku5 | exten => _NXXNXXXXXX,1,Dial(IAX2/${VOICEPULSE_GATEWAY_OUT_A}/${EXTEN}) |
| 01:51.55 | n00dle | Nugget, er... you're right, but * without zap? Seems weird to me. ;) |
| 01:52.14 | Zipper_32 | kuku5: The last part is the extension. It's sending the extension out, but the number is being rejected. |
| 01:52.21 | Nugget | works fine for me :) |
| 01:52.39 | sakic | man I like windows installers :P |
| 01:52.43 | kuku5 | Zipper_32: what does that mean ? |
| 01:52.50 | sakic | easier for morons like me |
| 01:53.00 | Zipper_32 | kuku5: Did you specify the proper values in your iax.conf ? Specifically the 'context' parameter? |
| 01:53.04 | Nugget | Zipper_32: you should ask voicepulse why they're rejecting it. |
| 01:53.27 | Nugget | I have voicepulse and my config doesn't even remotely resemble what you're doing, so perhaps you're taking the wrong approach. |
| 01:53.42 | Nugget | is that what voicepulse's sample configs told you to do? |
| 01:54.19 | kuku5 | Nugget: yes |
| 01:54.35 | Nugget | bizarre |
| 01:54.37 | Nugget | I dunno then |
| 01:55.05 | kuku5 | Zipper_32: they didnt request a context in iax.conf |
| 01:55.09 | sakic | once installed is this hard to setup? |
| 01:55.21 | Zipper_32 | sakic: It all depends what kind of features you want. |
| 01:55.34 | Zipper_32 | kuku5: Ask them why the call is being rejected then, that's all I can really say... |
| 01:55.36 | kuku5 | Nugget: how do you have it call out then ? |
| 01:55.44 | sakic | hold, call waiting, conference |
| 01:55.57 | kuku5 | They closed at 7 eastern. Open probably on tuesday. Need to get it done today :( |
| 01:56.11 | kuku5 | Nugget: what type of context do you have ? |
| 01:56.32 | Zipper_32 | sakic: Analog channels? |
| 01:56.50 | sakic | nah voip phones |
| 01:56.56 | sakic | and voip provider |
| 01:57.18 | Zipper_32 | It all depends on your learning curve. |
| 01:57.27 | Zipper_32 | Shouldn't be too difficulot |
| 01:57.33 | sakic | don't have tons of time for this |
| 01:57.33 | Zipper_32 | difficult*. Those are basic features. |
| 01:57.41 | sakic | good enough |
| 01:57.42 | *** join/#asterisk lwh (n=lwh192@rdsl-0469.tor.pathcom.com) |
| 01:58.05 | kuku5 | Nugget: Do you have a context fox voicepulse ? |
| 01:58.15 | n00dle | sakic, try looking around www.voip-wiki.org for examples. It helped me immensely. |
| 01:58.26 | sakic | excellent |
| 01:58.59 | Nugget | nope, sorry. |
| 02:00.18 | *** part/#asterisk mog (n=mog@c-71-207-214-192.hsd1.al.comcast.net) |
| 02:00.48 | kuku5 | Nugget: can you do me a huge favor and paste your dial line ? |
| 02:02.40 | sakic | I think I may start over from scratch :P |
| 02:02.58 | ingenio | can anyone confirm whether it is difficult to disable VAD/silence surpression on the Polycom 501? |
| 02:03.27 | sakic | I found a fedora step by step |
| 02:06.12 | *** join/#asterisk Cresl1n (n=matt@c-68-62-219-187.hsd1.al.comcast.net) |
| 02:06.13 | *** mode/#asterisk [+o Cresl1n] by ChanServ |
| 02:06.29 | sakic | is there a gui or is it all commandline? |
| 02:08.07 | Zipper_32 | command line. |
| 02:08.08 | n00dle | sakic, If you want GUI try trixbox first... study what it does, play with it, then format and install something you have more control over. :) |
| 02:08.13 | mvand | sakic: try asteriskNOW |
| 02:08.15 | n00dle | ...like I did. |
| 02:09.11 | sakic | I think I have it running |
| 02:10.58 | Corydon76-home | Yeah, I'm rather impressed by asteriskNOW |
| 02:11.52 | n00dle | I didn't try that one... hm... |
| 02:12.01 | *** join/#asterisk BullWeivel (n=bullweiv@12.165.69.254) |
| 02:12.33 | mvand | asteriskNOW looks great, but I need mgcp support |
| 02:12.38 | Corydon76-home | the gui is an interface to Asterisk itself, whereas trixbox is an interface to a pbx implementation |
| 02:12.40 | sakic | mgcp? |
| 02:12.49 | BullWeivel | uhhhh... MGCP ??? |
| 02:13.05 | BullWeivel | isnt MCGP proprietary ?? |
| 02:13.19 | Corydon76-home | No, it's an early IETF spec |
| 02:13.46 | BullWeivel | didnt realize that... i thought Cisco was the only one using it and it was proprietary. |
| 02:13.53 | mvand | I've been using a D-Link DG-104S and DG-102S for approx 2 years with asterisk |
| 02:15.01 | BullWeivel | im just curious..what is the best way to accomplish this. I want a phone system that can have a prompting system for when people dial in (ie press 1 for support) and also records the phone calls. Plus voicemail can be emailed to whoever. |
| 02:15.22 | mvand | also a clarent CPG-201, but it only worked with version 1.2.x (I am upgrading to 1.4.4, and the clarent box no longer works) |
| 02:16.16 | BullWeivel | wow... i feel stupid for always thinking MGCP was proprietary. :) |
| 02:16.53 | Corydon76-home | It's not proprietary, just crappy |
| 02:16.53 | mvand | well, not exactly proprietary, mu certainly not mainstream. |
| 02:17.31 | `Sean | g729 is not kinda propietry either i saw a widow long time ago on how to obtain a g729 licence, for personal use err not obtain a licene well use g729 for personal use |
| 02:17.38 | mvand | very crappy, but the boxes were cheap. |
| 02:17.56 | `Sean | then i ended up getting cracked copys of the modules |
| 02:18.04 | Zipper_32 | BullWeivel: To make a prompting system, it's as easy as http://www.voip-info.org/wiki/view/IVR . The voicemail is just using voicemail.conf, and the recording I can't answer for since I've never done it. |
| 02:18.47 | BullWeivel | Zipper thanks... that looks like a very good starting point. |
| 02:19.01 | Zipper_32 | BullWeivel: I hope it helps. |
| 02:19.03 | BullWeivel | yeah i want to record all phone calls coming/going. employees piss me off ;) |
| 02:21.36 | *** join/#asterisk flashnet (n=flashnet@208.8.160.110) |
| 02:22.26 | Zipper_32 | BullWeivel: Asterisk creates a call detail record by default so that you can see exactly 'who' people are calling, it's in /var/log/asterisk/cdr-csv/ |
| 02:22.58 | BullWeivel | well that is cool. I want to record their phone converstations. I own a small business and have customers always saying shit so. |
| 02:23.00 | Zipper_32 | I know i would want to see the numbers first before I start to listen to calls. I would hate to listen to 'everything' |
| 02:23.41 | mvand | sakic: I've been running asterisk on fedora systems for quite a while, and redhat systems before that. If you're new to linux, however, I would highly recommend asterisknow or trixbox |
| 02:23.45 | BullWeivel | yeah that is true too. mainly need it to just watch over things. Owning a business and having a full time job kinda sucks. |
| 02:24.11 | sakic | asterisknow didn't work well for me |
| 02:24.27 | BullWeivel | should i use the asterisk packages in Debian if I am running that distro? or best to go download it and install ?? |
| 02:24.37 | BullWeivel | just curious what others think of the debian package. |
| 02:25.58 | sakic | trixbox goes over top linux? |
| 02:26.15 | Corydon76-home | ~trixbox |
| 02:26.25 | jbot | Trixbox is a full linux distro that includes , FreePBX, and other 3rd party add-ons. It is these things on top of which make it seriously painful to support and hence you will find little help here for it. Try asking in #trixbox , or their forums & WIKI at http://www.trixbox.org |
| 02:26.32 | ez` | BullWeivel, i use click2call using with siptapi in windows ;) |
| 02:27.07 | sakic | trixbox it is :) |
| 02:28.50 | BullWeivel | what is the best gui for asterisk? |
| 02:29.21 | BullWeivel | sorry for a lot of the n00b questions :) |
| 02:31.29 | *** join/#asterisk codefreeze (n=steve_mu@65-182-39-254.cre.bil.biltmorecommunications.net) |
| 02:31.57 | crimethinker | that's sort of like asking which breed of dog is best to drive a backhoe. |
| 02:32.20 | ez` | ehhe |
| 02:32.40 | ez` | BullWeivel, none |
| 02:33.02 | BullWeivel | ez i figured the first answer was console lol |
| 02:33.06 | crimethinker | I haven't seen an asterisk gui that actually works well |
| 02:33.25 | BullWeivel | http://users.pandora.be/Asterisk-PBX/InstallAsterisk.htm this looks interesting..im probably goingto start with it and see where it lands me lol |
| 02:33.27 | crimethinker | unless your dialplan is basic basic |
| 02:33.30 | *** join/#asterisk voipnet-tech (n=owned@cpe-72-227-71-117.maine.res.rr.com) |
| 02:33.34 | ez` | crimethinker, some work fine ; use standard option , function ... |
| 02:34.47 | *** join/#asterisk phobus (n=phobus@crlspr-69.65.75.232.myacc.net) |
| 02:34.49 | crimethinker | or they assume you always want a ringing extension to do the same thing |
| 02:34.54 | crimethinker | etc etc |
| 02:34.59 | voipnet-tech | hi all, i just setup a new SIP trunk to a broadsoft switch... when i make a call into asterisk, asterisk appears to be rejecting the call with a 407 (proxy authentication problem) in response to the invite and a 603 in response to the notify... |
| 02:35.17 | voipnet-tech | what's the CLI command to make it show me everything that's going on in real time (all the pretty colored scrolling text...) |
| 02:35.52 | n00dle | er... "asterisk -vvvvvvvvvvvvvvvvvr" ? |
| 02:35.53 | ez` | maybe tail -f /var/log/asterisk/full |
| 02:36.19 | ez` | asterisk -rvvvvvvvvvvvvvvvvvvvvvvvvvT |
| 02:38.41 | voipnet-tech | cool thx, any ideas why i'm getting those errors? |
| 02:39.04 | *** join/#asterisk rpm (n=rpm@S010600111155e117.vc.shawcable.net) |
| 02:40.48 | *** join/#asterisk osiris (n=osiris@c-71-205-27-131.hsd1.mi.comcast.net) |
| 02:41.14 | Zipper_32 | Is anybody familiar with, or know why transfering between my new Polycom IP430's results in no ringing tone to the transferred caller to the destination? If I transfer with a different brand phone, the caller hears a ringing tone when he is being transfered, however, transferring with an IP430, the caller does not hear the ring tone while the transferred phone is ringing. |
| 02:47.45 | ez` | Zipper_32, i beleiev it something about you sip.cfg |
| 02:50.39 | Zipper_32 | Thanks ez`, I'll take a look at that. You're the only person who seems to have any sort of idea so far. |
| 02:51.12 | Zipper_32 | Although, to add a twist to it, my IP501's work fine with the same sip.cfg |
| 02:54.36 | ez` | Zipper_32, maybe you could creat a second sip.cfg and tweak it ... i did not use often ip430 ; |
| 02:55.09 | ez` | 300,301,500,501 use almost same option ; ; dunno about yours |
| 02:55.51 | Zipper_32 | Alright, thanks. I just got these 430's today and they don't seem to be cooperating with Asterisk. |
| 03:06.45 | voipnet-tech | i think the reason asterisk is rejecting the incomming calls is because there is a SIP proxy (edgemarc) proxying between the asterisk and the broadsoft switch... asterisk isn't seeing the calls as coming from broadsoft, but the edgemarc instead... Is this possibly why they'd get rejected, and if so, how would i fix it so that asterisk allows this ? |
| 03:08.50 | *** join/#asterisk Qwell (n=north@pdpc/sponsor/digium/Qwell) |
| 03:08.50 | *** mode/#asterisk [+o Qwell] by ChanServ |
| 03:09.48 | *** join/#asterisk _PauloS_ (n=_PauloS_@200-207-62-43.dsl.telesp.net.br) |
| 03:09.56 | _PauloS_ | Hi all... |
| 03:12.39 | *** join/#asterisk stkn_ (n=stkn@gentoo/developer/pdpc.active.stkn) |
| 03:13.39 | _PauloS_ | I have * 1,2 instaled on a Ubuntu server, pretty iddle. But its eating 100% from 1 cpu... |
| 03:14.51 | _PauloS_ | I have the same version on other single processor machines and they do not show this problem. |
| 03:15.01 | justdave | doing a lot of transcoding? |
| 03:15.19 | _PauloS_ | no, the server is idle. |
| 03:15.23 | _PauloS_ | no calls |
| 03:15.31 | justdave | ok, how about music-on-hold? |
| 03:15.55 | justdave | if you have it set to use an external player (like mpg123) it usually leaves it running in the background and just ignores the stream until it needs it |
| 03:16.16 | voipnet-tech | what's it mean when asterisk returns a "407" error on a SIP trunk when there's a new incomming call ? |
| 03:16.27 | justdave | and I've also seen it have trouble killing those off so if * has been restarted a few times there could be a bunch of them |
| 03:18.10 | _PauloS_ | justdave, in this case top would show mpg123 eating cpu, not asterisk right? |
| 03:19.03 | justdave | yeah, that's true |
| 03:19.32 | justdave | "show channels" show anything? |
| 03:20.12 | _PauloS_ | nothing. |
| 03:20.27 | _PauloS_ | I think it's related to interrupt sharing: |
| 03:20.33 | _PauloS_ | <PROTECTED> |
| 03:20.56 | _PauloS_ | HDA is something related to sound, isnt it? |
| 03:22.05 | voipnet-tech | here's what shows up in the log when i make a call to asterisk: http://rafb.net/p/2sNPhu89.html |
| 03:30.14 | *** join/#asterisk Cabal_ (n=Cabal@unaffiliated/cabal) |
| 03:32.08 | *** join/#asterisk Deeewayne (i=dwayne@nat/digium/x-3efcebf3da96a0d7) |
| 03:32.16 | *** join/#asterisk Qwell[] (i=qwell@pdpc/sponsor/digium/Qwell) |
| 03:32.17 | *** mode/#asterisk [+o Qwell[]] by ChanServ |
| 03:34.34 | *** join/#asterisk alrs (n=lars@170.206.224.58) |
| 03:36.16 | voipnet-tech | http://rafb.net/p/RDGLfd64.html |
| 03:36.27 | voipnet-tech | 407 Proxy Authentication Required |
| 03:37.00 | voipnet-tech | i can't believe no one here will help me figure this out :-( |
| 03:38.27 | shido6 | ? |
| 03:39.11 | shido6 | ur behind nat and not registering properly? |
| 03:41.17 | voipnet-tech | broadworks via internet -> edgemark(router + sip proxy) via LAN -> asterisk |
| 03:41.33 | voipnet-tech | asterisk should be communicating with the edgemarc directly on the lan side |
| 03:42.03 | voipnet-tech | all the phones in the building already point to the edgemarc as their sip server, edgemarc proxies to broadworks |
| 03:42.03 | *** join/#asterisk bkw_ (i=brian@70.128.114.89) |
| 03:42.23 | voipnet-tech | phones work fine, but asterisk is rejecting the incomming calls from the edgemarc with that 407 error message |
| 03:42.39 | voipnet-tech | i found like 3 threads already on various forums regarding this error, and none have replies |
| 03:46.44 | shido6 | lord. |
| 03:47.07 | shido6 | im going to bed, catch me in the morning or drop me a message |
| 03:47.08 | *** join/#asterisk woolbeo (n=woolbeo@75-162-219-95.slkc.qwest.net) |
| 03:47.28 | *** part/#asterisk x86_ (n=x86@p3m/member/x86) |
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| 04:07.51 | rpm | voipnet-tech, trying to get asterisk working with broadworks? |
| 04:07.56 | rpm | r13 or r14? |
| 04:10.43 | *** join/#asterisk CrazyTux (n=CrazyTux@216-110-94-230.static.twtelecom.net) |
| 04:13.05 | *** join/#asterisk anthm (n=anthm@CPE-72-131-113-50.wi.res.rr.com) |
| 04:13.05 | *** mode/#asterisk [+o anthm] by ChanServ |
| 04:13.07 | *** join/#asterisk Daejeo1 (n=chatzill@124.62.150.49) |
| 04:13.23 | Daejeo1 | i created the two IAX2 extensions. when both the extensions are online I am able to communicate. But if one extension is offline and i try it call it then call is dropping. I am unable to reach mail box |
| 04:14.05 | Zipper_32 | Can you show us your extensions.conf using pastebin.com? |
| 04:14.10 | Zipper_32 | ~pastebin |
| 04:14.21 | jbot | methinks pastebin is a place to paste your stuff without flooding the channel - try http://pastebin.ca, or http://channels.debian.net/paste, or http://rafb.net/paste/, or http://pastebin.com is usually painfully too slow and unresponsive to use, use one of the other pastebin sites, or dpaste.com is a very nice pastebin as well |
| 04:15.40 | Zipper_32 | It sounds like your priority is off, that's all. |
| 04:18.16 | Daejeo1 | http://pastebin.ca/510917 |
| 04:18.45 | Daejeo1 | how do I set the priority? |
| 04:19.48 | Zipper_32 | You're sending your iax.conf, send your extensions.conf |
| 04:20.14 | Zipper_32 | Rather, just the part that pertains to your dialing of 200 and 207 |
| 04:20.23 | Daejeo1 | oh |
| 04:21.37 | *** join/#asterisk rmayorga (n=rmayorga@unaffiliated/rmayorga) |
| 04:22.54 | Daejeo1 | I am using freepbx |
| 04:24.12 | Zipper_32 | ~freepbx |
| 04:24.39 | jbot | [freepbx] unable to be supported here. It's a complex piece of dialplan and GUI, and can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking there |
| 04:24.40 | *** part/#asterisk n00dle (n=ccraft@ip-249-27.springsips.com) |
| 04:27.07 | Zipper_32 | I'm having a bit of a problem with call parking. Is anybody around to tell me what I may be doing wrong? My Dial option has the 't' on it, the include => parkedcalls is in the correct context, and features.conf is ripped from a previously existing setup (so I know it works), sip.conf users have canreinvite=no, and I think that should be it. |
| 04:27.44 | Zipper_32 | My problem is that pressing the # does nothing during the call. It simply doesn't register, even with the 't' in the Dial command. |
| 04:29.24 | BullWeivel | anyone know if a Cisco WIfi 7920 phone can be flashed to work with SIP on a Asterisk box ? |
| 04:30.49 | Zipper_32 | nevermind... spelling error. |
| 04:39.23 | voipnet-tech | rpm, r11 actually, r13 will be up next week |
| 04:40.32 | voipnet-tech | i just managed to get it to work, the problem was that i was trying to make asterisk talk to broadworks through an edgemarc as proxy and asterisk was denying the calls because proxy authentication is required |
| 04:40.35 | BullWeivel | cvs.digium.com no longer exists i take it ??? |
| 04:40.48 | voipnet-tech | i'd still like to be able to get asterisk to work with broadworks thru the proxy |
| 04:40.55 | voipnet-tech | so that asterisk doesn't need a public IP |
| 04:42.37 | BullWeivel | does cvs.digium.com resolve for you guys at all ????? |
| 04:43.09 | *** join/#asterisk joat (n=joat@ip70-160-147-169.hr.hr.cox.net) |
| 04:43.29 | Zipper_32 | Not for me. |
| 04:43.58 | BullWeivel | i am doing this right... right?? go to cvs.digium.com to download the latest and greatest using the CVS???? |
| 04:44.55 | BullWeivel | oh well...screw it im going to bed....gnight all |
| 04:49.20 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 04:51.48 | Zipper_32 | <PROTECTED> |
| 04:52.24 | Zipper_32 | BullWeivel: This should do you fine: http://ftp.digium.com/pub/asterisk/releases/asterisk-1.2.18.tar.gz |
| 04:53.31 | Zipper_32 | Is anybody familiar with, or know why transfering between my new Polycom IP430's results in no ringing tone to the transferred caller to the destination? If I transfer with a different brand phone, the caller hears a ringing tone when he is being transfered, however, transferring with an IP430, the caller does not hear the ring tone while the transferred phone is ringing. |
| 04:59.14 | *** join/#asterisk bintut (n=bintut@cm209.gamma182.maxonline.com.sg) |
| 04:59.20 | bintut | hello all.. |
| 04:59.25 | *** join/#asterisk kiscokid (n=Ron@adsl-216-101-109-187.dsl.snfc21.pacbell.net) |
| 05:00.14 | *** part/#asterisk kiscokid (n=Ron@adsl-216-101-109-187.dsl.snfc21.pacbell.net) |
| 05:02.08 | bintut | i just got my digium tdm dev kit here.. my asterisk-1.2.13.x on my debian etch is working already.. now, i'm testing it.. |
| 05:02.20 | *** join/#asterisk Cresl1n (n=matt@c-68-62-219-187.hsd1.al.comcast.net) |
| 05:02.20 | *** mode/#asterisk [+o Cresl1n] by ChanServ |
| 05:02.57 | bintut | but i am leaving the home in a few minutes.. how can i forward the incoming call from the trunk to my mobile number? |
| 05:04.05 | *** join/#asterisk vAd0r (n=IceChat7@65.67.210.121) |
| 05:04.07 | vAd0r | can someone tell me the diff between eyebeam v1.5 and xlite? |
| 05:04.18 | *** join/#asterisk treeshoo (n=t@bas2-toronto12-1168024427.dsl.bell.ca) |
| 05:05.32 | bintut | vAd0r: eyebeam comes with a video support |
| 05:05.49 | vAd0r | xlite shows my video to i though |
| 05:05.55 | vAd0r | i see it on the side when i have a webcam |
| 05:07.53 | *** join/#asterisk nick125_lappy (n=nick@atarack/staff/nick125) |
| 05:08.13 | nick125_lappy | Has anyone here played with any of the open source voice recognition tools and asterisk? |
| 05:10.50 | nick125_lappy | (mainly sphinx) |
| 05:17.03 | *** join/#asterisk jbot (i=ibot@pdpc/supporter/active/TimRiker/bot/apt) |
| 05:17.04 | *** topic/#asterisk is Asterisk: The Open Source PBX -=- Asterisk 1.4.4 (April 27, 2007) Asterisk 1.2.18 (April 24, 2007), Zaptel 1.2.17.1, 1.4.2.1 (April 25, 2007) -=- Other fun channels: #asterisk-gui, #asterisknow, #asterisk-commits, #astridevcon -=- Join #freepbx for freepbx/#trixbox for trixbox support. |
| 05:21.56 | SwK | since when does 1.2 407 calls when the peer is insecure=very? |
| 05:24.03 | [TK]D-Fender | SwK, because peers are used to place calls, and user is used to receive calls. |
| 05:25.34 | SwK | yeah whatever |
| 05:25.46 | SwK | I never explicity use user or peer... i always use friend |
| 05:26.02 | SwK | and its set insecure=very with a host and permit/deny |
| 05:26.20 | SwK | and its matching it after the 407 is being sent |
| 05:26.48 | SwK | which is about bass ackwards |
| 05:26.49 | [TK]D-Fender | hrm |
| 05:27.13 | SwK | anyone that work for agile in here? |
| 05:27.23 | SwK | i gotta few comments about that crap |
| 05:29.43 | Zipper_32 | Fender, are you familiar with any transfer problems with the IP430? |
| 05:31.12 | [TK]D-Fender | Zipper_32, nope |
| 05:33.14 | ohadz | hi, anyone knows the svn path for stable version 1.4.4? |
| 05:35.53 | Zipper_32 | I have a problem transfering between my Polycom IP430's. Transferring results in no ringing tone being heard by the transferred caller. If I transfer with a different brand phone, the caller hears a ringing tone when he is being transfered, however, transferring with an IP430, the caller does not hear the ring tone while the transferred phone is ringing. |
| 05:37.12 | mcab | Zipper_32: get a packet capture of the IP430 doing a transfer, and the 501 doing the transfer, and compare the traces |
| 05:37.32 | mcab | Zipper_32: AFAIK the IP430 and IP501 use the same SIP stack, so their behavior should be the same |
| 05:38.04 | mcab | Zipper_32: the other things to check are - are you using the s/w version you think you're using, and are you using the sip.cfg you think you're using |
| 05:38.27 | Zipper_32 | They're all using the newest sip.cfg and firmware. No custom modifications. |
| 05:38.46 | mcab | Zipper_32: right, are you *sure*? :-) |
| 05:38.50 | SwK | ok |
| 05:41.20 | Zipper_32 | mcab: I reloaded them all today. Yes I'm sure. =) |
| 05:41.31 | Zipper_32 | Rather, I reloaded them tonight. |
| 05:42.08 | *** join/#asterisk bugsy448 (n=rickw@ppp-70-128-124-134.dsl.tulsok.swbell.net) |
| 05:42.48 | Zipper_32 | They weren't able to transfer with the 1.6.7 that they had on when I got them today, so I put the latest 2.1.0 on tonight, with no improvement |
| 05:43.26 | Zipper_32 | I'm going to sleep on it and give it another whirl tomorrow. Thanks for your help. |
| 05:44.03 | [TK]D-Fender | <PROTECTED> |
| 05:45.54 | Zipper_32 | The thing is... I transfer with a different phone, and everything works, |
| 05:46.06 | Zipper_32 | The dial commands to each phone is identical. |
| 05:47.02 | [TK]D-Fender | Zipper_32, would help if you pastebined all the configs and CLI output... |
| 05:47.37 | Zipper_32 | Fender, I have to get going now, but I'll send them tomorrow morning if you're around. |
| 05:47.46 | Zipper_32 | Thanks for your help so far. I appreciate it. =) |
| 05:47.48 | [TK]D-Fender | we'll see.... |
| 05:47.50 | [TK]D-Fender | later |
| 05:48.03 | *** part/#asterisk bugsy448 (n=rickw@ppp-70-128-124-134.dsl.tulsok.swbell.net) |
| 05:48.10 | nick125_lappy | I really wish there was a decent free voice recognition tool that would work well with asterisk |
| 05:49.11 | nick125_lappy | (I had some interesting ideas for a "random" selector, when you have problems making real decisions, you can just call it and ask it a question, and it would answer it randomly) |
| 05:52.45 | SwK | and what kinda fucktard from a 'billing package company" sets ALL your sip friends for customers to 'insecure=very' |
| 05:52.54 | SwK | along w/ host=dynamic |
| 05:52.58 | nick125_lappy | ... |
| 05:53.03 | SwK | thats like saying please assrape my terminate |
| 05:53.06 | SwK | err termination |
| 05:53.57 | nick125_lappy | ...yeah |
| 05:56.02 | nick125_lappy | Please tell me who did that so I can slap them for stupidity |
| 05:59.32 | *** join/#asterisk rickw (n=rickw@ppp-70-128-124-134.dsl.tulsok.swbell.net) |
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| 07:44.31 | linagee | can someone explain trellis code modulation? |
| 07:44.42 | linagee | http://en.wikipedia.org/wiki/Trellis_modulation |
| 07:45.37 | L|NUX | is there any way to change voicemail file names ? |
| 07:47.22 | *** join/#asterisk tzafrir_laptop (i=tzafrir@69-94-197-125.biltmorecomm.com) |
| 07:53.50 | *** join/#asterisk stoffell_h (n=stoffell@d5153F90F.access.telenet.be) |
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| 08:05.44 | L|NUX | is there any way to change voicemail file names ? |
| 08:10.05 | mightnare | L|NUX: edit apps/app_voicemail.c |
| 08:10.33 | L|NUX | mightnare : can't we do this using VARIABLES ? |
| 08:11.14 | mightnare | if you take a look at app_voicemail.c, they're all hard coded... |
| 08:16.37 | L|NUX | ok |
| 08:24.40 | *** join/#asterisk Star568 (i=Star@cpe-75-84-29-38.socal.res.rr.com) |
| 08:25.51 | Star568 | dear all, i am new to asterisk. i have download 1.4.4 and try to do "make" , it tells me i need to do some configure script first, how can i do it |
| 08:26.43 | Star568 | make clean, make install? |
| 08:30.17 | Mavvie | ./configure |
| 08:31.27 | Star568 | i tried ./configure, then make, but still error message |
| 08:31.51 | Star568 | should i add ./configure options? |
| 08:32.00 | mightnare | what error did you get from ./configure ? |
| 08:34.34 | Star568 | configure: error: C++ preprocessor "/lib/cpp" fails sanity check |
| 08:36.14 | mightnare | are you using an rpm based distro? what does "rpm -q cpp" say |
| 08:37.41 | mightnare | rpm -q gcc-c++ |
| 08:38.09 | Star568 | i use yum |
| 08:38.57 | Star568 | gcc.i386 3.4.6-8 installed |
| 08:40.32 | Star568 | i should install gcc-c++.i386 too |
| 08:44.41 | Star568 | how to install asterisk without zaptel |
| 08:44.52 | Star568 | i got error like |
| 08:44.55 | Star568 | checking for ZT_DIAL_OP_CANCEL in zaptel/zaptel.h... no |
| 09:11.59 | *** join/#asterisk yannj_fr (n=yannj@vpn.intelunix.fr) |
| 09:16.49 | *** join/#asterisk sasch (n=sasch@host102-30-static.107-82-b.business.telecomitalia.it) |
| 09:16.53 | sasch | hi all |
| 09:17.58 | yannj_fr | does anyone knows the differences between users.conf and sip.conf |
| 09:18.42 | yannj_fr | ? |
| 09:52.46 | *** join/#asterisk friedrich| (n=friedric@e177248183.adsl.alicedsl.de) |
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| 09:56.33 | *** join/#asterisk nickzxcv (i=nick@nick.xen.prgmr.com) |
| 09:57.34 | nickzxcv | is it possible, with asterisk, if something calls up and does an access code or something, that asterisk will turn into a modem that a pppd could be connected to in software? |
| 09:58.19 | nickzxcv | or a getty |
| 10:01.40 | yannj_fr | kind of callback? |
| 10:09.32 | nickzxcv | no, so that you wouldn't have to have an extra line that a real modem would be connected to |
| 10:10.51 | jql | yeah, no, neither asterisk nor any other software component will act as a modem by itself |
| 10:10.56 | nickzxcv | asterisk can already receive faxes right? kind of like that except |
| 10:11.24 | nickzxcv | with pppd or getty |
| 10:11.50 | jql | http://forums.digium.com/viewtopic.php?p=47009&sid=0fa6111d8a4d9202ab8154d1166e437b if you're curious |
| 10:13.17 | nickzxcv | I'm thinking, if somebody has only one phone line, maybe they use asterisk for their voicemail, and they also want dialin to their home lan or shell |
| 10:13.44 | nickzxcv | not as an isp, you would need plenty of lines for an isp anyway |
| 10:13.53 | nickzxcv | but the same idea |
| 10:14.38 | jql | and you read the part about how nobody's written the software decoders to do that? |
| 10:15.42 | nickzxcv | yeah, but it says the beginning parts are there, anyway that answers my question. Thanks! |
| 10:16.02 | jql | it's just a matter of time |
| 10:34.49 | *** join/#asterisk jm|laptop (n=jm|home@dilbert.jamiem.com) |
| 10:36.12 | yannj_fr | Does anyone knows why the presence (hint) doesnt work if the user is only in users.conf |
| 10:38.12 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 11:07.19 | *** join/#asterisk stoffell_h (n=stoffell@d51A4D08A.access.telenet.be) |
| 11:17.24 | *** join/#asterisk quitte (n=quitte@stgt-d9beb6d9.pool.mediaWays.net) |
| 11:18.29 | quitte | hi. is it possible to build asterisk against readline instead of using libedit? i'm having trouble crosscompiling libedit for mipsel-linux-uclibc |
| 11:22.55 | *** join/#asterisk zotz (n=zotz@24.244.163.157) |
| 11:29.27 | quitte | hmm what's the difference between 1.4 and 1.2? |
| 11:31.15 | *** join/#asterisk coppice (n=chatzill@10.198.17.210.dyn.pacific.net.hk) |
| 11:44.13 | *** join/#asterisk berktr (n=canberk@teknopet.com) |
| 11:44.16 | berktr | hi |
| 11:44.19 | berktr | who's there |
| 11:47.05 | *** join/#asterisk kombi (n=kombi@213.160.14.18) |
| 11:47.56 | kombi | chan_sip.c:3487 sip_write: Asked to transmit frame type 4, while native formats is 0x100 (g729)(256) read/write = 0x4 (ulaw)(4)/0x2 (gsm)(2) ?? |
| 11:48.21 | kombi | anyone? |
| 11:48.50 | kombi | wrong codec? |
| 11:49.40 | berktr | kombi, i am getting the same error |
| 11:49.52 | berktr | this is because of wrong frame count |
| 11:50.08 | berktr | yours is most probably 20ms and your termination company is asking for 10ms... |
| 11:50.57 | kombi | berktr: strange thing is: it happens when calling voicemail from a hardphone, not from softphone. Company is not involved so far (no connection yet) |
| 11:51.30 | berktr | yeah, i mean this does not affect your calls though |
| 11:52.05 | kombi | berktr: the hardphone does not connect to voicemail and the error is thrown instead.. |
| 11:52.25 | kombi | could it be due to the codec of the phone? |
| 11:52.31 | *** join/#asterisk SuperID (n=gary@c-65-96-225-97.hsd1.ma.comcast.net) |
| 11:52.49 | berktr | i guess so |
| 11:53.07 | kombi | got to look up what codecs there are for 7941 |
| 11:57.43 | berktr | is there a termination company that you are working with? |
| 11:57.48 | berktr | that you may recommend |
| 11:58.23 | kombi | berktr: you mean like a public phone net company? |
| 11:58.49 | berktr | if you are calling it that way yeah |
| 11:59.41 | kombi | I havn't put in the bri card yet, so it would be too early for recommendations. We us a german company called QSC though |
| 11:59.50 | berktr | qsc |
| 11:59.51 | kombi | s/us/use/ |
| 12:00.04 | kombi | scary bot... |
| 12:00.11 | berktr | lol |
| 12:00.19 | kombi | are you familiar with qsc? |
| 12:00.34 | berktr | no this is the first time i'm hearing |
| 12:01.15 | kombi | berktr: doesn't make much sense if you're not in the country I suppose, we use an atm based line there |
| 12:02.00 | berktr | i see |
| 12:02.05 | berktr | are they cheap or expensive |
| 12:03.15 | kombi | inexpensive I'd say, some 15 Euro for an S0 bus, prices per minute almost unmeasurable |
| 12:03.27 | *** join/#asterisk stoffell_h (n=stoffell@d51A4D08A.access.telenet.be) |
| 12:03.43 | berktr | can you send a rate list if you have one |
| 12:04.10 | kombi | that would be best found on their website at qsc.de |
| 12:04.39 | berktr | i can't speak german :) |
| 12:05.32 | kombi | most people can't.. |
| 12:05.49 | berktr | 0.12 eur/minute for turkey |
| 12:05.51 | berktr | jeez |
| 12:06.12 | berktr | my rate for turkey is 0.014usd |
| 12:06.15 | berktr | per minute |
| 12:06.42 | kombi | sounds much better, we almost never call turkey though. What's your company then? |
| 12:07.59 | *** join/#asterisk cspot (i=cspot@ip68-1-63-100.pn.at.cox.net) |
| 12:08.50 | coppice | most people sell turkey by weight rather than time :-) |
| 12:09.07 | kombi | good one coppice.. |
| 12:09.46 | kombi | if you connect entirely over ip there is no real need to use a local company, billing maybe |
| 12:11.04 | kombi | whoops, the G711ulaw does it, and the thing speaks to me in... german (that hardly anyone speaks though..;) |
| 12:11.45 | kombi | I love being a nerd.. |
| 12:12.25 | *** join/#asterisk DaveCanoe (n=Dave@adsl-70-235-73-216.dsl.mrdnct.sbcglobal.net) |
| 12:13.01 | *** join/#asterisk Prato (n=Prato@dslb-088-073-067-158.pools.arcor-ip.net) |
| 12:13.56 | kombi | just how do I tell my girlfriend? "I spend three day getting that cisco phone to work" "so what does it do?" "well, dial a number, make a phone call, hang up.." "kombi, you are *so* exciting.." |
| 12:17.56 | macTijn | kombi: lol |
| 12:18.03 | Prato | Hello to all, it's my first post and i just got startet with asterisk. So i think it's a real newbee question. I have to realize an atxfer for 3 phones in my institute but have trouble with the hook/flash button |
| 12:18.44 | Prato | Can someone give me a hint how to use the hook/flash in asterisk to realize an atxfer. I search for 3 days right now |
| 12:19.26 | kombi | I freely confess that I have not the slightest idea, prato |
| 12:20.40 | Prato | So this is a pity. I know that there is an option in features conf called atxfer where you can enter a dtmf code, But the telephone only sends a small break and no dtmf |
| 12:25.46 | stoffell_h | i just configured an astribank 8 but the led H/W OK keeps flickering constantly and syslog says: ERR-xpp_usb: xusb_xframe_send: XBUS-00: more than 100 pending writes (65001). Dropping |
| 12:25.50 | stoffell_h | is that normal ? |
| 12:27.55 | Qwell | stoffell_h: probably not |
| 12:28.25 | stoffell_h | i remember from a previous astribank that the sync should light up |
| 12:29.22 | stoffell_h | now i was looking for a way to use the ATU (test utility) but it's not on the cd anymore it seems :-( |
| 12:35.12 | kombi | anything on asterisk side I need to configure to get the messagesWaiting prompt working on the phone? |
| 12:36.08 | tzafrir_laptop | stoffell, what model of the astribank is this? |
| 12:36.44 | stoffell_h | tzafrir_laptop, it's a fresh astribank 8, just got it yesterday from our friends in paris :) |
| 12:36.46 | berktr | is there a good billing system for asterisk |
| 12:37.20 | tzafrir_laptop | the ATU has been discontinued: replaced with http://updates.xorcom.com/iso/live-1.0.0.iso |
| 12:37.30 | tzafrir_laptop | maybe you got that in the package |
| 12:38.08 | tzafrir_laptop | stoffell, cat /proc/xpp/sync |
| 12:38.37 | tzafrir_laptop | also: do you get those messages immedietly or after a few minutes? |
| 12:42.00 | stoffell_h | i get them pretty fast, after doing ztcfg (i generated a zaptel conf with genzaptelconf) |
| 12:42.00 | berktr | is there a good billing system for asterisk |
| 12:42.00 | stoffell_h | tzafrir_laptop, http://pastebin.ca/511413 |
| 12:42.00 | stoffell_h | tzafrir_laptop, i'll download the live iso to try later today |
| 12:42.00 | tzafrir_laptop | it's in host sync right now.... |
| 12:42.07 | stoffell_h | so i'll echo 0 0 to the sync ? |
| 12:42.11 | tzafrir_laptop | a bit of a long shot: try: echo 1 >/sys/module/xpp/pcm_tasklet |
| 12:43.01 | stoffell_h | okay, did that |
| 12:43.03 | tzafrir_laptop | right: /bin/echo SYNC=0 >/proc/xpp/sync |
| 12:43.17 | tzafrir_laptop | (/bin/sco gives explicit errors) |
| 12:43.58 | stoffell_h | okay: # QUERY=nn - Query XBUS-nn for sync information (DEBUG) |
| 12:43.58 | stoffell_h | HOST |
| 12:43.58 | stoffell_h | tick: #914122 |
| 12:43.58 | stoffell_h | tick rate: 250/second (SAMPLE_TICKS=10000) |
| 12:43.58 | stoffell_h | TASKLETS: missed_ticks: 0 |
| 12:44.25 | berktr | is there a good billing system for asterisk? |
| 12:44.50 | stoffell_h | berktr, try voip-info.org, i have only heard of astbilling |
| 12:45.26 | stoffell_h | tzafrir_laptop, the echo gives a message in syslog: |
| 12:45.27 | stoffell_h | ay 26 14:44:49 hpdv kernel: [ 3942.840000] NOTICE-xpp: Using deprecated syntax to update /proc/xpp/sync |
| 12:45.27 | stoffell_h | May 26 14:44:49 hpdv kernel: [ 3942.840000] NOTICE-xpp: Cannot set sync master to XBUS-00 (has no XPD #0). |
| 12:45.51 | Prato | Does really noone has an idea how to enable hook-flash with atxfer? I would be so lucky for an answer. |
| 12:46.52 | tzafrir_laptop | hmm... there was already a usb-level flood. can you try disconnecting the astribank and reconnecting it? |
| 12:47.13 | stoffell_h | okay |
| 12:47.41 | stoffell_h | did that |
| 12:47.53 | stoffell_h | <PROTECTED> |
| 12:47.54 | stoffell_h | May 26 14:47:04 hpdv kernel: [ 4077.280000] INFO-xpp: New xbus: XBUS-00 |
| 12:47.54 | stoffell_h | May 26 14:47:04 hpdv kernel: [ 4077.868000] ERR-xpp: XBUS-00: Poll timeout. Continuing anyway. |
| 12:47.54 | stoffell_h | May 26 14:47:04 hpdv kernel: [ 4077.868000] INFO-xpp: XBUS-00: Poll results: removals=0 additions=0 |
| 12:48.44 | stoffell_h | i did not do a ztcfg -v yet |
| 12:50.59 | *** join/#asterisk Corydon76-home (i=ten@pdpc/supporter/sustaining/Corydon76-home) |
| 12:50.59 | *** mode/#asterisk [+o Corydon76-home] by ChanServ |
| 12:51.08 | tzafrir_laptop | I see a bunch of things which may be hardware-related and may be driver related. The live CD includes the version that is also in current SVN branches of zaptel. I'd recommend that you either try the live CD or zaptel svn (1.2, 1.4 or trunk: all of them have it) |
| 12:51.13 | Qwell | Corydon76-home: !!! |
| 12:51.28 | Corydon76-home | Yuck. Crapcast's service is just abysmal |
| 12:51.55 | Corydon76-home | Qwell: when'd you make it back? |
| 12:52.01 | Qwell | about 10 |
| 12:52.07 | Corydon76-home | Ah... |
| 12:52.16 | stoffell_h | tzafrir_laptop, i was now using 1.2.17.1, so i best try the live cd to test it now, right ? |
| 12:52.25 | Corydon76-home | I dropped off Cresl1n around 6 |
| 12:52.43 | tzafrir_laptop | I guess |
| 12:52.48 | Qwell | I worked on ast_storage stuff on the way back, heh |
| 12:52.59 | Qwell | ...and wardriving |
| 12:53.04 | Corydon76-home | Cool, your ball... ;-) |
| 12:53.07 | tzafrir_laptop | you don't use packages there (that is ubuntu kernel there?) |
| 12:53.17 | Qwell | and then I DoSed Russell's mac |
| 12:53.22 | Corydon76-home | lol |
| 12:53.26 | Qwell | then he DoSed me back, and my laptop crashed ;/ |
| 12:53.30 | Qwell | probably unrelated |
| 12:53.40 | Qwell | although, with ndiswrapper, it wouldn't surprise me |
| 12:53.42 | stoffell_h | yeah, i am testing it on ubuntu (on my laptop) right now because i had it on an HP server yesterday (with debian) and had the same issues |
| 12:54.00 | Corydon76-home | Yeah, ndiswrapper just barely works as it is |
| 12:54.03 | Qwell | Corydon76-home: You'd be surprised just how many wifi networks we found on the way, in random unpopulated places |
| 12:54.09 | Corydon76-home | Heh |
| 12:54.23 | Qwell | a good 75% were unsecure... |
| 12:54.27 | Qwell | insecure..whatever |
| 12:54.43 | Corydon76-home | Odd |
| 12:55.19 | Corydon76-home | I'm less surprised in back out-of-the-way places, but networks accessible from US-72? |
| 12:55.36 | Qwell | we took 20, or whatever |
| 12:55.49 | Qwell | but yeah, all up that, and 75/85 |
| 12:55.57 | Corydon76-home | Oh, okay |
| 12:56.07 | Qwell | 431 actually had an AP with an ESSID that was their phone number.. |
| 12:56.14 | Corydon76-home | We went up to CHA, then cut down US-72 |
| 12:56.17 | Qwell | (I was gonna call, but that was when my box crashed) |
| 12:56.24 | Corydon76-home | Heh |
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| 13:08.41 | *** part/#asterisk SuperID (n=gary@c-65-96-225-97.hsd1.ma.comcast.net) |
| 13:09.20 | stoffell_h | tzafrir_laptop, i will download the live cd and let you know how it goes.. bbl ;. thanks! |
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| 13:26.21 | berktr | i need a good termination company for terminating to Turkey |
| 13:28.52 | *** join/#asterisk s3047276 (n=erick@60-241-9-126.static.tpgi.com.au) |
| 13:29.10 | s3047276 | hi all |
| 13:29.12 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 13:30.42 | s3047276 | test |
| 13:30.51 | s3047276 | can you read 'this'? iam testing |
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| 13:43.29 | festr__ | anyone experiencing doubled cdr when no_answer? |
| 13:43.31 | festr__ | 1.4.4 |
| 13:44.00 | codefreeze | festr__: yep. Matter of fact, I made the change |
| 13:44.13 | festr__ | codefreeze: how change? |
| 13:44.34 | festr__ | codefreeze: can you share? |
| 13:44.39 | codefreeze | I fixed some of the CDR bugs, but some differences in behavior result. |
| 13:44.44 | s3047276 | hi, do u know something about NAT (Network Address Translator)? can you help me? |
| 13:45.01 | festr__ | codefreeze: and is it at bugs.digium? |
| 13:45.12 | codefreeze | 8221 was one of them |
| 13:46.03 | codefreeze | I have immediate plans to fix a bunch of CDR problems-- I'm in the middle of it now |
| 13:46.35 | festr__ | codefreeze: i will check this in 8221 |
| 13:46.53 | festr__ | i've also issues with latest 1.4 svn, deadlocks in chan_sip |
| 13:47.26 | *** join/#asterisk [Airwolf] (n=airwolf@voip.ymav.nl) |
| 13:47.47 | festr__ | i've some backtraces but no clue where this deadlock come from. 1.4.4 is ok. i've checked changelog and there is huge changes in res_features |
| 13:50.39 | codefreeze | oh, better get going. Gotta check out of the hotel. Bye! |
| 13:54.34 | *** join/#asterisk ApEtc (n=apetc@ip70-162-197-214.ph.ph.cox.net) |
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| 14:22.56 | berktr | do you guys know a good a-z termination company |
| 14:22.59 | berktr | with high quality |
| 14:23.07 | berktr | or can you tell which one are you working with? |
| 14:29.15 | *** join/#asterisk SirThomas_Home (n=tomc@209-169-199-174.us.warpdriveonline.com) |
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| 14:32.31 | *** join/#asterisk ThOr101 (n=bthorson@pool-71-126-163-76.washdc.fios.verizon.net) |
| 14:36.27 | ThOr101 | Busy extensions are supposed to go to DialPriority+101 right? http://paste.debian.net/28914 |
| 14:37.26 | ThOr101 | * 1.4.4 |
| 14:38.11 | Strom_M | no no and no. |
| 14:38.47 | ThOr101 | From the "~book" If, however, the destination channel is busy, Dial( ) will go to |
| 14:38.47 | ThOr101 | priority n+101, if it exists (where n is the priority where the Dial( ) application was |
| 14:38.48 | ThOr101 | called). |
| 14:38.56 | Strom_M | yeah, for version 1.0 |
| 14:39.02 | Strom_M | that's old deprecated syntax |
| 14:39.08 | Strom_M | don't use it. |
| 14:39.11 | ThOr101 | Bah! So I need to find myself some current docs. |
| 14:39.33 | Strom_M | look up the "n" priority, priority labeling, and GotoIf() |
| 14:39.33 | ThOr101 | This orielly book is really screwing me over |
| 14:39.47 | ThOr101 | ok thanks |
| 14:39.47 | Strom_M | a new edition will be out soon |
| 14:39.48 | *** join/#asterisk Daejeo1 (n=chatzill@124.62.150.49) |
| 14:40.29 | Daejeo1 | link for H.323 video codec? |
| 14:40.41 | Daejeo1 | anyone knows |
| 14:40.50 | ThOr101 | Can you suggest a good source for documentation? Is it the docs folder in the source code? Is that what I should be using for my references? |
| 14:41.08 | Strom_M | ThOr101: take a look there, and also on voip-info.org |
| 14:41.19 | ThOr101 | Ack, thanks. |
| 14:41.33 | Daejeo1 | does asterisk support for h.323 codec? |
| 14:41.52 | Strom_M | Daejeo1: what are you talking about? H.323 is an ITU-T spec for signaling and media transport. it's not a codec. |
| 14:43.19 | Daejeo1 | sorry man it is a proto |
| 14:43.31 | Daejeo1 | misunderstood |
| 14:43.55 | Strom_M | there is support for H.323, but it's not built into asterisk IIRC |
| 14:43.59 | *** join/#asterisk steliosk (n=Stelios@ipa226.211.tellas.gr) |
| 14:46.31 | Daejeo1 | can i configure on asterisk? |
| 14:46.37 | Strom_M | yes |
| 14:46.43 | Strom_M | that's what I just said. |
| 14:47.08 | Daejeo1 | can u direct me to documentation and source |
| 14:47.37 | Strom_M | http://www.voip-info.org/wiki/index.php?page=Asterisk+H323+channels |
| 14:48.36 | Daejeo1 | is there any way to get a eyebeam softphone? |
| 14:49.02 | Daejeo1 | torrentbox has only one seeder |
| 14:49.39 | Daejeo1 | I am trying from last week, so far no success |
| 14:49.43 | Strom_M | uh |
| 14:49.44 | Strom_M | yeah |
| 14:49.47 | Strom_M | you PAY for it |
| 14:49.51 | Strom_M | or download something else that's free |
| 14:51.35 | ThOr101 | This is the Internet. Everything is free! (or it's on ebay) |
| 14:58.24 | ThOr101 | Ah HA! The ael.txt, a whole document that talks about extensions. Now we're getting somewhere. |
| 14:58.33 | Strom_M | thats not extensions.conf |
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| 14:58.46 | Strom_M | that's AEL, which is a completely different way of configuring extensions |
| 14:59.37 | ThOr101 | Oh, darn it. I wasn't able to find in the source docs discussion about priority labeling. I found an example, but it wasn't all that clear. The search continues. |
| 14:59.53 | ThOr101 | is AEL the future? |
| 15:00.16 | Strom_M | gee, I typed "asterisk priority labeling" into a search engine, and look what comes up as the FIRST HIT: |
| 15:00.17 | Strom_M | http://www.voip-info.org/wiki/index.php?page=Asterisk+priorities |
| 15:01.07 | *** join/#asterisk fakhir (n=Fakhir@unaffiliated/fakhir) |
| 15:01.34 | ThOr101 | Yeah, I know. I wanted to find docs that supported that. The same site lead me down a 2 day hole that ended up with digium bugs 9806, 9808 and 9809. |
| 15:01.59 | *** join/#asterisk bintut (n=bintut@cm13.gamma188.maxonline.com.sg) |
| 15:02.19 | Strom_M | um, ok? |
| 15:02.39 | Strom_M | you can also just try these things and see if they work, you know |
| 15:04.43 | bintut | how do you configure your asterisk box when someone calls you from your trunk number and after 10 seconds of ringing state with no one answers the call it will be forwarded to (or dial my) mobile phone? |
| 15:05.00 | Strom_M | Ringing() |
| 15:05.02 | Strom_M | Wait(10) |
| 15:05.11 | Strom_M | Dial(mobile phone) |
| 15:05.13 | Strom_M | or alternatively |
| 15:05.18 | Strom_M | Dial(deskphone,10) |
| 15:05.26 | Strom_M | Dial(mobilephone) |
| 15:06.03 | bintut | Strom_M: shall i put it on my extensions.conf? |
| 15:06.34 | *** join/#asterisk [TK]D-Fender (n=Joe@64.235.216.2) |
| 15:06.37 | Strom_M | no, I'd put it on a piece of paper and then dance around it to a Scritti Politti song :) |
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| 15:08.13 | ThOr101 | You spin me right round... NM |
| 15:08.24 | Strom_M | no, that's Dead or Alive |
| 15:08.59 | ThOr101 | exten => 123,1,Dial(Zap/3,10) So I take this line and change it to exten => 123,n(CallWhite),Dial(Zap/3,10) but it doesn't seem to like it. I need a number to start the config for the extension, no? |
| 15:09.13 | ThOr101 | I've got the perfect way to make the girls go crazy? |
| 15:09.19 | Strom_M | you need to start your priority numbering at 1 |
| 15:09.26 | Strom_M | then the rest of them are "n" |
| 15:09.27 | ThOr101 | So then how do I name it? |
| 15:09.29 | Strom_M | ThOr101: yes! :) |
| 15:09.44 | ThOr101 | so that I can later do name+101 |
| 15:10.07 | Strom_M | no no no |
| 15:10.12 | Strom_M | first is "1" |
| 15:10.21 | Strom_M | the rest are "n" or "n(label) |
| 15:10.37 | Strom_M | +101 is not something you should use |
| 15:11.19 | ThOr101 | At all ever? All the docs that I see (are they all old and should be disregarded) say that is how to handle the busy extension. |
| 15:11.25 | [TK]D-Fender | Hard numbering FTW! Down with "n"! |
| 15:11.36 | Strom_M | ThOr101: that's all old crap |
| 15:11.53 | ThOr101 | Ahhh ok. So I need a new base set of knowledge for this numbering / extensions.conf |
| 15:11.57 | Strom_M | use GotoIf() and do things based on the contents of the DIALSTATUS variable |
| 15:12.00 | *** join/#asterisk blitzrage (n=blitzrag@asterisk/documenteur-extraordinaire/blitzrage) |
| 15:12.08 | ThOr101 | even the 1.4.4 extensions.conf.sample had that 101 stuff in there. |
| 15:12.22 | ThOr101 | ok, let me go search on that bit of knowledge |
| 15:16.35 | ThOr101 | it isn't gotoif, but am I in the right neighborhood: exten => s,2,Goto(s-${DIALSTATUS},1) |
| 15:17.04 | [TK]D-Fender | ThOr101, That is using the DIALSTATUS variable which is set after a dial indicating why it left. |
| 15:18.04 | [TK]D-Fender | ThOr101, it is making a new exten to jump to by concatenating "s-" withthat status; eg : s-NOANSWER |
| 15:18.21 | ThOr101 | Ahh, new extension. That's not the direction I want to go in. |
| 15:18.59 | [TK]D-Fender | ThOr101, that exten is not a "standard extension", nor a numbered exten which pretty solidly says the only way you'd get there is by a purposeful jump. |
| 15:19.08 | ThOr101 | Ahhh |
| 15:22.02 | bintut | i notice that when calling my trunk/telephone number, it takes 3 rings from the caller before the call is forwarded to an extension phone (zap). i just setup a NoOp here and directly forwarded to my analog phone. is this normal that it takes 3 rings first? |
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| 15:22.25 | Strom_M | bintut: what kind of trunk is it? |
| 15:22.27 | [TK]D-Fender | bintut, Thats because * is waiting for CallerID |
| 15:24.10 | bintut | Strom_M: i have a dev kit tdm card from digium here with 1 fxo and 1 fxs.. my fxo is connected directly from my telephone service subscription and my fxs is on my analog phone as an extension number.. it's a pstn trunk |
| 15:24.24 | Strom_M | then yes, what [TK]D-Fender said |
| 15:24.27 | [TK]D-Fender | bintut, --^ |
| 15:25.04 | [TK]D-Fender | Strom_M, I'd hat to imagine anyone buying a PCI TDM card just for FXS *shudder* |
| 15:25.18 | bintut | [TK]D-Fender: so, it's normal.. actually, i can't get a caller id maybe because i didn't add this feature when i subscribe a telephone line from my telco |
| 15:25.37 | [TK]D-Fender | bintut, Well * is still waiting for it |
| 15:25.45 | ThOr101 | So I think I am getting closer http://paste.debian.net/28922 |
| 15:25.53 | [TK]D-Fender | bintut, set "usecallerid=no" in zapata.conf for those channels |
| 15:26.04 | Strom_M | ThOr101: no no no no no. |
| 15:26.11 | bintut | [TK]D-Fender: ok.. |
| 15:26.31 | ThOr101 | Wow, 5 no(s) that time. I thought I was doing so well. |
| 15:27.11 | ThOr101 | I copied an example out of the extensions.conf |
| 15:27.21 | ThOr101 | embellished a bit |
| 15:27.22 | Strom_M | ThOr101: no, you kluged a macro |
| 15:27.25 | [TK]D-Fender | ThOr101, CLOSER. exten => s,n,Goto(s-${DIALSTATUS},1) <- this is what you have following *123* |
| 15:27.27 | bintut | [TK]D-Fender: how can i set my asterisk to forward all calls from my pstn trunk (fxo) number when no one answers after 10 seconds/rings to my mobile phone number? |
| 15:27.58 | [TK]D-Fender | ThOr101, change that "s,n" for "123,n" and it will work |
| 15:28.16 | bintut | where can i find the complete description and usage for the extensions.conf file? |
| 15:28.27 | [TK]D-Fender | ThOr101, you HAVE still bastardized what was done in a macro into an in-line normal dialable exten however. |
| 15:29.16 | [TK]D-Fender | bintut, how would you do it if you had a normal analog phone puleed onto the line? |
| 15:29.59 | ThOr101 | Monkeys. Hilarious! |
| 15:30.05 | bintut | what do you mean? |
| 15:30.31 | [TK]D-Fender | bintut, You asked how to have * "transfer" the call. To where? |
| 15:30.51 | ThOr101 | So I got it to work, but the bastardization of the MACRO is over my head. Probably because I know nothing about macros |
| 15:30.56 | ThOr101 | Thanks. |
| 15:31.17 | bintut | [TK]D-Fender: this is the scenario i want to happen here.. |
| 15:32.04 | bintut | [TK]D-Fender: as i've said earlier, i have a digium dev kit with 1 fxo and 1 fxs pci card.. my fxo is connected to my telco.. |
| 15:32.40 | [TK]D-Fender | ThOr101, s-**whatever** is where you jump in the sample your hacked up because it was CALLED from "s". that gave the feeling that its like a SUBSET named series of extens like "children". |
| 15:33.03 | [TK]D-Fender | ThOr101, Sure a name is a name and a number is a number, but it logically FELT like a subset. |
| 15:33.04 | bintut | [TK]D-Fender: basically, i set my trunks context in my extensions.conf to NoOp so that all calls intended for the fxo goes directly to my extension number connected to the fxs analog phone |
| 15:33.28 | [TK]D-Fender | bintut, I'm asking about what exactly you are expecting it to do on NO ANSWER. |
| 15:33.39 | [TK]D-Fender | bintut, Transfer to WHERE? How?! |
| 15:34.21 | bintut | now, someone (externel) calls my telephone pstn number and basically my extension number (analog) will ring but i want that after 10 seconds/rings when no one picks up my analog phone, the call will be forwarded to my mobile phone |
| 15:34.24 | [TK]D-Fender | ThOr101, It might have been more logical to do : Goto(123-${DIALSTATUS},1) |
| 15:34.44 | BullWeivel | anyone know where the CVS is located for Asterisk ????? |
| 15:35.01 | [TK]D-Fender | bintut, Lets say you didn't HAVE * at all. If you had a normal analog phone plugged onto tat line instead, how would this request of your be possible? |
| 15:35.34 | ThOr101 | Ahh and then rename them 123-NOANSWER so I would associate them with that extension rather than a generic s-** which would most likely be used as a subset for all extensions in the context. |
| 15:35.41 | [TK]D-Fender | BullWeivel, www.asterisk.org <- go read the SVN instruction. SVN replaced CVS years ago |
| 15:36.05 | [TK]D-Fender | ThOr101, Were you to have coded their jumpts the same, yes |
| 15:36.11 | bintut | [TK]D-Fender: because you're saying that my fxo/trunk number is currently busy already? |
| 15:36.18 | ThOr101 | That makes sense. |
| 15:36.59 | [TK]D-Fender | ThOr101, See the sample uses a macro because it allows you to pass some parameters (to fill in the blanks of who to actually ring) and then process them in the same uniform way changing only things like the actual tech to dial, VM boxes to leave messages in if applicable ,etc |
| 15:37.37 | [TK]D-Fender | bintut, Before I can answer as to how * would do it I need to know how YOU YOURSELF with an analog phone IN YOUR HAND can do it. |
| 15:38.04 | [TK]D-Fender | bintut, I'm betting this hypotheitcal request of yours is not possible on a single line where you are period. |
| 15:38.36 | bintut | [TK]D-Fender: if i added this kind of service to my telco, it will be possible that the call will be forwarded to my mobile phone |
| 15:39.12 | [TK]D-Fender | bintut, well clearly its your TELCO that does this. There is nothing YOU can do to your line to make * decide for this to happen |
| 15:39.50 | [TK]D-Fender | hoummmmmmmmmmmmmm |
| 15:40.20 | ThOr101 | Namaste |
| 15:42.02 | *** join/#asterisk sharp (n=sharp@dsl092-234-217.phl1.dsl.speakeasy.net) |
| 15:46.29 | tzanger | [TK]D-Fender: did you see the steampunk strat? |
| 15:46.46 | tzanger | [TK]D-Fender: http://steampunkworkshop.com/steampunk-strat.shtml |
| 15:47.22 | bintut | [TK]D-Fender: ok.. |
| 15:49.45 | [TK]D-Fender | tzanger, funky... |
| 15:53.05 | berktr | and yes asterisk does support h323 |
| 15:54.28 | [TK]D-Fender | berktr = smrt |
| 15:54.43 | [TK]D-Fender | You'd almost think he read that in the BIG PRINT in the BOOK! |
| 15:55.43 | ThOr101 | or h323.txt in the doc subdirectory |
| 15:56.12 | [TK]D-Fender | *woah* |
| 16:00.12 | *** join/#asterisk A[s]H (n=TnT@host117-192-static.53-88-b.business.telecomitalia.it) |
| 16:00.37 | A[s]H | please this codec IS FOR CELERON ??http://ftp.digium.com/pub/telephony/codec_g729/asterisk-1.2/x86-32/codec_g729a_v31_c3.tar.gz |
| 16:00.49 | A[s]H | from official site of digium |
| 16:00.53 | *** join/#asterisk Strom_M (n=strom@ip70-170-60-8.lv.lv.cox.net) |
| 16:00.55 | A[s]H | someone can help me? |
| 16:01.30 | [TK]D-Fender | A[s]H, looks right. |
| 16:01.50 | A[s]H | there are for pentium, atlon, k6 |
| 16:01.57 | A[s]H | and for celeron ?? is right ? this |
| 16:02.14 | A[s]H | http://ftp.digium.com/pub/telephony/codec_g729/asterisk-1.2/x86-32/ this is the directory |
| 16:02.14 | [TK]D-Fender | A[s]H, Celeron = slow Pentium |
| 16:02.26 | A[s]H | uhm... ok |
| 16:02.45 | shido6 | :) |
| 16:02.45 | A[s]H | this http://ftp.digium.com/pub/telephony/codec_g729/asterisk-1.2/x86-32/codec_g729a_v31_pentium-m.tar.gz ???? |
| 16:03.00 | [TK]D-Fender | A[s]H, No, use the first one |
| 16:03.12 | [TK]D-Fender | A[s]H, P-M is a different arch |
| 16:03.16 | A[s]H | grrrr |
| 16:03.33 | A[s]H | i must buy i license, i dont want to chose wrong file |
| 16:04.01 | A[s]H | u confirm me? the first one? |
| 16:04.17 | [TK]D-Fender | A[s]H, Go call digium and give them your motherboad model # |
| 16:04.25 | A[s]H | not answer me |
| 16:04.47 | A[s]H | im italian |
| 16:04.55 | *** join/#asterisk anthm (n=anthm@CPE-72-131-113-50.wi.res.rr.com) |
| 16:04.55 | *** mode/#asterisk [+o anthm] by ChanServ |
| 16:05.27 | ThOr101 | Yeah, those digium people won't answer the phone if there is an italian on the other end. Darn CallerNationalityID |
| 16:05.57 | [TK]D-Fender | A[s]H, e-mail support then |
| 16:06.05 | A[s]H | not answer |
| 16:06.13 | A[s]H | only automatic answer grrrr |
| 16:06.25 | *** join/#asterisk Fieldy (i=iPUIoeIT@gentoo/contributor/Fieldy) |
| 16:06.43 | ThOr101 | Ash, it is a holiday weekend over here. So if you sent the e-mail Friday afternoon, it may not have been answerd. Just a though. |
| 16:09.16 | ThOr101 | So is AEL something I should learn? Is it old on the way out, or new on the way in? |
| 16:09.43 | russellb | new way in |
| 16:21.17 | *** join/#asterisk Cresl1n (n=matt@c-68-62-219-187.hsd1.al.comcast.net) |
| 16:21.17 | *** mode/#asterisk [+o Cresl1n] by ChanServ |
| 16:23.34 | *** join/#asterisk brussel (n=brussel@cpe-72-130-172-229.san.res.rr.com) |
| 16:23.44 | ThOr101 | what is the maximum number of digits a phone number can have (assuming the worst case scenario for international calls?) |
| 16:27.39 | A[s]H | i have buyed codec, payed whit paypal, but??? where i must receive che key ID ??? |
| 16:30.51 | *** join/#asterisk john-eman0n (n=eman0n@cpe-76-181-165-26.columbus.res.rr.com) |
| 16:33.34 | [TK]D-Fender | ThOr101, AEL = nobody really cares |
| 16:33.59 | *** join/#asterisk holiday42 (n=me@70-57-197-218.farg.qwest.net) |
| 16:34.02 | Qwell | A[s]H: it gets emailed to you - I think you have to wait until Tuesday |
| 16:34.13 | A[s]H | ma porca troia |
| 16:34.45 | A[s]H | ma che cazzo , ma la serietà dov'è finita? |
| 16:34.47 | ThOr101 | I've been hungry for porca troia lately. |
| 16:35.28 | ThOr101 | [TK]D-Fender : Nobody cares about AEL yet? Or it will never replace exten => ? |
| 16:37.36 | holiday42 | me: asterisk 1.2, a la asterisknow. softphone works fine (x-lite), need help configuring a linksys pap2 ATA.... |
| 16:37.51 | holiday42 | sip debug ip:x.x.x.x here: http://pastebin.ca/511726 |
| 16:42.39 | *** join/#asterisk ThOr101 (n=Thor@pool-71-126-163-76.washdc.fios.verizon.net) |
| 16:45.17 | *** join/#asterisk pejo_ (n=peter@1-1-5-39a.lio.sth.bostream.se) |
| 16:47.06 | pejo_ | Is asterisk full functional implementation of a SIP-server? |
| 16:47.06 | *** join/#asterisk stoffell_h (n=stoffell@d51A4D08A.access.telenet.be) |
| 16:47.06 | A[s]H | no |
| 16:47.06 | [TK]D-Fender | ThOr101, No. AEL is a hack module that reads .ael source and GENERATES standard dialplan logic out of it. It in and of itself does NOTHING you can't do with extensions.conf yourself. |
| 16:47.50 | [TK]D-Fender | ThOr101, the standard has changed twice and there have been conversion bugs. All of this adds up to "wate of my time" |
| 16:48.37 | [TK]D-Fender | pejo_, Go read ... THE BOOK |
| 16:48.39 | [TK]D-Fender | ~book |
| 16:48.59 | jbot | rumour has it, book is a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 16:49.00 | *** join/#asterisk irule (n=irule@189.164.43.19) |
| 16:49.29 | ThOr101 | Got it. Thanks. |
| 16:49.37 | stoffell_h | on loading the fpga firmware on my astribank; the h/w ok light begins to flicker and doesn't stop, not even after loading the modules |
| 16:49.55 | [TK]D-Fender | ok, off to martial arts... back in several hours... |
| 16:50.16 | kombi | anything special about MWI on cisco phones? can't seem to get it working.. |
| 16:51.41 | stoffell_h | and still these errors in syslog about my astribank: ERR-xpp_usb: xusb_xframe_send: XBUS-00: more than 100 pending writes (30001). Dropping. |
| 16:53.17 | *** join/#asterisk vAd0r (n=IceChat7@65.67.210.121) |
| 16:54.18 | vAd0r | anyone got a tutorial for a linksys pap2 |
| 16:55.03 | *** join/#asterisk ohadz (n=ohad@cpe-69-203-27-50.nyc.res.rr.com) |
| 16:57.33 | BullWeivel | what is a decent free softphone you guys like to use? doing my first install of * and figure for now i will try it with a softphone or two...then convert my Cisco 7960's over to SIP after I do that. |
| 16:58.19 | ThOr101 | All the documentation I have seen in the past 5 days of setting up * have pointed to X-Lite |
| 16:58.51 | ThOr101 | ~xlite |
| 16:58.54 | jbot | rumour has it, xlite is at download xlite at: http://snipurl.com/5tgi | and see sample configs at http://snipurl.com/5tgj, or xlite is a free SoftPhone (software phone, requires no hardware) from xten inc, |
| 16:59.43 | BullWeivel | cool...will have to try that out. now to get debian to behave |
| 17:02.45 | holiday42 | vAd0r: linksys pap2 info here: http://www.bargainshare.com/index.php?showtopic=87504&st=120 |
| 17:02.52 | kombi | BullWeivel: do unstable, it has 1.4 |
| 17:05.20 | kombi | for editing voice prompts, is there a good free wave editor for mac ox x? |
| 17:08.05 | vAd0r | i found a video tutorial for the 7970 cisco phone. can i use the same config out of it and just a firmware for the 7950 |
| 17:08.10 | vAd0r | 7940 |
| 17:08.42 | BullWeivel | kombi: do you remember the command to upgrade to unstable ? (isnt it like apt-get disto-upgrade) |
| 17:09.20 | kombi | BullWeivel: change /etc/apt/sources.list and do apt-get dist-upgrade, do one version step at a time though |
| 17:09.27 | *** join/#asterisk drako (n=ljd@nelug/coreteam/luisjose) |
| 17:09.35 | *** join/#asterisk putnopvut (n=putnopvu@user-24-214-124-177.knology.net) |
| 17:10.06 | BullWeivel | kombi: yeah im using stable right now. need to upgrade my kernel to 2.6 as well i guess. im running 2.4.27 |
| 17:10.27 | BullWeivel | any advantage to running 2.6 compared to 2.4 when it comes to * ?? |
| 17:10.36 | kombi | BullWeivel: not necessarily, depends on the hardware you want to use |
| 17:11.12 | BullWeivel | well im thinking of going with no hardware :) just using an external gateway like... uhh broadwing i think is one of them. |
| 17:11.36 | kombi | no desperate need for a new kernel then |
| 17:11.46 | *** join/#asterisk wunderkin (i=wunderki@ip68-108-204-139.ph.ph.cox.net) |
| 17:12.33 | BullWeivel | yeah the box im working on is 3 hours from my house...so no rebooting would be best for now :) |
| 17:12.33 | BullWeivel | my overall goal is to make a phone system that i can record incoming/outgoing calls, and also to have a somewhat advanced IVR menu system. |
| 17:12.55 | kombi | You could even stay with stable then |
| 17:13.25 | *** join/#asterisk marc\cba (n=marc@cpc2-whit2-0-0-cust200.cdif.cable.ntl.com) |
| 17:14.03 | BullWeivel | well you have what... stable (fairly old but works great) unstable (not all the kinks are out of it yet) and testing (hrmm... riiiiigggt) |
| 17:15.30 | kombi | BussWeivel: considering the box being off site and requirements not too demanding, stay with stable and install *1.2 |
| 17:15.53 | *** join/#asterisk ingenio (n=ingenio@12-216-99-16.client.mchsi.com) |
| 17:15.57 | ingenio | can anyone confirm whether it is difficult to disable VAD/silence surpression on the Polycom 501? |
| 17:18.47 | *** join/#asterisk ToyMan (n=Stuart@10062.webjogger.net) |
| 17:19.08 | *** join/#asterisk nick125_lappy (n=nick@atarack/staff/nick125) |
| 17:20.48 | nick125_lappy | What would be the best way to block calls on my asterisk system? I keep getting telemarketing calls, even though my numbers are listed on do not call. The only issue with doing it in extensions.conf would be that it might grow to a quite large list.. |
| 17:21.57 | kombi | nick: do they call with caller id? |
| 17:22.26 | nick125_lappy | kombi: Yes |
| 17:22.29 | ingenio | I'm looking to purchase a number of Polycom 501s, but I've read a forum post where a user is unable to disable the silence suppression/voice activity detection making the phone sound like it is half duplex. |
| 17:22.59 | nick125_lappy | So, I could block by caller ID...but, I'm worried about getting a huge unmanagable list of these numbers in my extensions.conf (or rather, extensions/blacklist.conf) |
| 17:23.57 | kombi | nick: you could store them in AstDB |
| 17:24.11 | nick125_lappy | kombi: Ah, true |
| 17:26.43 | nick125_lappy | There was some kind of script on voip-info, but, I can't find it |
| 17:26.56 | nick125_lappy | I also would like something that isn't too resource-intensive... |
| 17:26.57 | nick125_lappy | Should I just Hangup() the calls or..? |
| 17:28.31 | nick125_lappy | Or should I try something like Zapateller |
| 17:30.30 | ingenio | :( |
| 17:33.01 | *** join/#asterisk jbot (i=ibot@pdpc/supporter/active/TimRiker/bot/apt) |
| 17:33.01 | *** topic/#asterisk is Asterisk: The Open Source PBX -=- Asterisk 1.4.4 (April 27, 2007) Asterisk 1.2.18 (April 24, 2007), Zaptel 1.2.17.1, 1.4.2.1 (April 25, 2007) -=- Other fun channels: #asterisk-gui, #asterisknow, #asterisk-commits, #astridevcon -=- Join #freepbx for freepbx/#trixbox for trixbox support. |
| 17:36.25 | BullWeivel | any of you guys using exteral gateways for your phone calls?? curious who your using and what you think of their service. |
| 17:37.19 | JunK-Y | mediatrix |
| 17:38.44 | BullWeivel | i should have stated that better... gateway providers. so that for example my pbx will connect to theirs to make phone calls going out to the world. |
| 17:39.03 | *** join/#asterisk mindCrime (n=chatzill@cpe-065-190-188-124.nc.res.rr.com) |
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| 17:41.58 | *** mode/#asterisk [+o anthm] by ChanServ |
| 17:43.49 | nick125_lappy | Sweet, now I can block these stupid idiots from calling! |
| 17:46.48 | *** join/#asterisk putnopvut (n=putnopvu@user-24-214-124-177.knology.net) |
| 17:47.18 | BullWeivel | dude...dont block them. |
| 17:47.23 | BullWeivel | send them to a recording. |
| 17:48.01 | BullWeivel | Say something like. Hello. Hi how can i help you. <10 second pause> that sounds interesting... how do i buy? |
| 17:48.52 | nick125_lappy | haha |
| 17:49.15 | nick125_lappy | but, I don't want to waste my money answering those calls |
| 17:52.25 | nick125_lappy | or maybe use voice recognition and festival :-) |
| 17:52.51 | *** join/#asterisk iBuMp (n=iBuMp@rrcs-71-42-227-6.sw.biz.rr.com) |
| 17:53.32 | iBuMp | hello |
| 17:53.58 | iBuMp | needing help with error 488 after g729a install |
| 17:54.31 | nick125_lappy | Or maybe fiddling around with sphinx and flite |
| 17:54.32 | nick125_lappy | I don't know |
| 17:54.36 | nick125_lappy | It would be great though |
| 17:58.40 | ThOr101 | sound tt-allbusy at least |
| 18:00.05 | nick125_lappy | ThOr101: I'm using the blacklisted context on voip-info (http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+LookupBlacklist) |
| 18:01.01 | nick125_lappy | BullWeivel: I really like your idea though |
| 18:06.50 | wunderkin | ingenio, it is off by default, check the admin manual, the guy is a nubb |
| 18:07.03 | BullWeivel | how does it waste your money? they called you... just curious. if you keep them on the phone, well that waists their money...because to them..time is money. |
| 18:07.54 | nick125_lappy | BullWeivel: well, paying for call origination.. |
| 18:07.59 | *** join/#asterisk badcfe (n=cso@d80-170-243-199.cust.tele2.fr) |
| 18:13.32 | *** join/#asterisk steliosk (n=Stelios@ipa226.211.tellas.gr) |
| 18:17.37 | BullWeivel | anyone using Broadvoice??? what do you think of their service ???? |
| 18:17.47 | voipnet-tech | their phone service or customer service? |
| 18:17.55 | BullWeivel | well both of course :) |
| 18:18.07 | voipnet-tech | phone service was OK, customer service sucked big donkeyballs |
| 18:18.32 | voipnet-tech | worst customer support from a telephony company i've ever seen |
| 18:18.46 | voipnet-tech | had to deal with them last year for a simple single number Local number port |
| 18:18.57 | voipnet-tech | took them 9 months of trying to port it before i gave up |
| 18:19.07 | voipnet-tech | meanwhile i had to pay 2 bills to keep my other number active |
| 18:19.23 | voipnet-tech | they would never call me back or respond to emails either |
| 18:19.27 | BullWeivel | yeah im not going to port any numbers... i have heard many a nightmares over that. |
| 18:19.33 | BullWeivel | so you still using them? |
| 18:19.36 | voipnet-tech | hellllll no |
| 18:20.16 | *** join/#asterisk CrazyTux[m] (n=CrazyTux@cpe-24-162-48-94.houston.res.rr.com) |
| 18:20.25 | voipnet-tech | and after all that BS i went thru with them, they still charged me a $40 fee for service cancellation |
| 18:21.11 | voipnet-tech | i filed complaints with the FCC and PUC, also thought about trying lawsuit to recover 9 months of service charges for service i could never use because they couldn't manage to get my # ported |
| 18:21.12 | nick125_lappy | boo for bullcrap service providers |
| 18:21.12 | BullWeivel | so who did you change to? |
| 18:21.30 | voipnet-tech | well i went back to a verizon landline after that lol |
| 18:21.48 | BullWeivel | lol DUDE...dont get me started on flipping Verizon and their FIOS bullshit! |
| 18:22.01 | nick125_lappy | I've heard some pretty bad things about broadvoice, nothing first hand though |
| 18:22.30 | BullWeivel | well other then broadvoice..what are the other options that support/allow * ? |
| 18:23.21 | nick125_lappy | What kind of provider? per-minute or "unlimited" (which is a total lie, by the way) |
| 18:24.23 | BullWeivel | im looking for unlimited that can service 580 area code. |
| 18:24.41 | voipnet-tech | now i work for an ITSP, so i have my own sip service |
| 18:24.42 | nick125_lappy | I've been using vitelity here for a while, and it has worked fine for me, but, it's pay-minute |
| 18:24.52 | findlay | hey nick |
| 18:24.53 | voipnet-tech | i've got a carrier grade broadsoftswitch |
| 18:25.01 | nick125_lappy | findlay: hiya |
| 18:27.38 | ingenio | I'm looking to purchase a number of Polycom 501s, but I've read a forum post where a user is unable to disable the silence suppression/voice activity detection making the phone sound like it is half duplex. Can anyone with a 501 comment? |
| 18:28.06 | ingenio | wunderkin: gracias.. i've heard the suppression in other phones and it bothers me. |
| 18:28.32 | ThOr101 | BullWeivel, What's wrong with Verizon Fios? I've got it. Not trying to do any crazy * over it yet, but I'm curious as to your opinion. |
| 18:30.41 | *** join/#asterisk CVirus (n=GoD@196.219.179.75) |
| 18:31.38 | BullWeivel | my opinion is that FIOS TV and internet rock... their billing department.... they are a bunch of #*N#JKD |
| 18:32.03 | voipnet-tech | i hate silence suppression :/ personal preference... it screws with your head |
| 18:32.30 | BullWeivel | they have my credit card number...they bill me monthly whatever they need to bill me. Last week they cut my service off saying i hadn't paid something like $370. The last bill they sent me showed they owed me $10 |
| 18:32.48 | BullWeivel | so they cut my service off even though they sent me a bill saying they owed me $10. yeah that pissed me off. |
| 18:33.03 | voipnet-tech | i've been using aastra phones recently, they've been great |
| 18:33.10 | BullWeivel | it took me roughly 3 hours to get their their bullshit and get it turned back on. |
| 18:33.11 | coppice | silence suppression != VAD |
| 18:33.27 | ThOr101 | I agree with you on that one. Their billing service is pretty awful (their customer service). The only company that I have found to be worse was Cox. |
| 18:33.37 | voipnet-tech | coppice, i knew that, i don't like either of them |
| 18:33.56 | coppice | if the VAD is any good its hard to detect its there |
| 18:33.59 | ingenio | coppice: :/ i was told they were the same |
| 18:34.11 | ingenio | can the VAD be disabled though? |
| 18:34.33 | ThOr101 | Even after I got FIOS to my house, they told me I couldn't have it (when I asked for consolidated billing) because one of their maps was off. Interesting to hear one person say. We can't combine your Fios bill and your phone becuase fios isn't available to your house. Then to have them look at my fios bill. |
| 18:35.05 | *** join/#asterisk kiscokid (n=Ron@216.101.109.187) |
| 18:36.15 | ThOr101 | I've been thinking about their FiosTV, but I don't have any TV at the moment, and it is rather nice. The world is such a happier place without the daily news. |
| 18:38.14 | Daejeo1 | http://thepiratebay.org/tor/3667464/Xten-eyeBeam-comersial |
| 18:40.32 | shido6 | Th0r101 that depends on how you look at the news. If you're constructing the news its great entertainment to watch how ppl "report" on that news and watch ppls reactions |
| 18:40.38 | *** part/#asterisk killfill_ (n=killfill@201.238.233.3) |
| 18:41.55 | shido6 | but its nice to hear you stopped your programing |
| 18:42.06 | ThOr101 | Yeah, that's the problem. Broadcast news is pure entertainment. At least the comedy channel owns up to it. Even the BBC (which used to be at least a little less sensational) is headed in that direction. Daily Show and Colbert report are (sadly, and ironically) more "honest" than these broadcasters. |
| 18:42.08 | shido6 | +m |
| 18:42.14 | *** join/#asterisk shtoom (n=godson@59.93.115.88) |
| 18:42.57 | shido6 | hehe - you might watch tv again when you get a few commercials on different stations running. |
| 18:43.10 | shido6 | then turn it back off :) |
| 18:43.39 | ThOr101 | True. I suppose if I had commercials on the air, I'd be tempted to have TV to watch them :-) |
| 18:43.56 | shido6 | hows your ivrs coming along? |
| 18:44.49 | ThOr101 | Pretty well I think. I've got the basics down. * isn't too bad when I don't have to worry about outbound calls, etc. The examples are pretty good, and the orielly book helped a little (except for the parts where it is outdated and really sent me for a spin) |
| 18:45.17 | ThOr101 | I've got the main menu, and all the voicemail boxes setup. Now I just need to write the extensions.conf to put it all together. |
| 18:45.41 | shido6 | cool, well on your way |
| 18:45.44 | *** join/#asterisk mihinomenest (i=Hrge@cerebus.clandestineresearch.com) |
| 18:45.55 | ThOr101 | with a lot of help from this channel :-) |
| 18:46.10 | shido6 | im working on a new project here with lumenvox |
| 18:46.16 | shido6 | i get bored travelling |
| 18:46.21 | ThOr101 | I applied to get into the forum for FWD so I could post my issues with FWD. |
| 18:46.30 | shido6 | so i call into asterisk and ask it to download a movie for me |
| 18:46.40 | ThOr101 | wow, cool. |
| 18:46.41 | shido6 | it searches various torrent sites and downloads them to the xbox |
| 18:47.08 | shido6 | to make a long story short. its broken right now :) |
| 18:48.19 | shido6 | im also working on recording live tv |
| 18:48.44 | shido6 | if im on a trip and want to record "the game" but forgot to record it before i left... i can call into asterisk and have it record it |
| 18:49.07 | ThOr101 | I had MythTV setup for a while. It was pretty darn cool. I liked the fact that I could watch this old house (great show, but they talk slow) at 1.25X speed and it compensated for the change in frequency. |
| 18:49.42 | *** part/#asterisk Daejeo1 (n=chatzill@124.62.150.49) |
| 18:50.39 | shido6 | yeah |
| 18:50.52 | shido6 | almost the same idea here only with asterisk running a script to record the show |
| 18:51.10 | shido6 | lumenvox just translates the words and looks at a tv guide |
| 18:51.31 | shido6 | well .... im leaving out a lot of apps that make this work but u get the idea |
| 18:52.06 | shido6 | i'll drop it on our website when im satisfied that it works properly and has a decent interface |
| 18:52.19 | ThOr101 | What website? |
| 18:52.32 | ThOr101 | lumenvox? |
| 18:52.52 | shido6 | www.nufone.net |
| 18:53.00 | shido6 | or my project page |
| 18:53.45 | ThOr101 | Jeremy? |
| 18:54.08 | shido6 | ?? I'm not Jeremy |
| 18:54.17 | shido6 | and more than 1 person works at NuFone :) |
| 18:54.30 | ThOr101 | Oh. I wasn't sure. He's the only guy I know that works there :-) |
| 18:54.55 | ThOr101 | Now I guess I know 2 :-) |
| 18:58.12 | *** join/#asterisk e-milioX (n=emilio@adsl-225-148-191.mia.bellsouth.net) |
| 19:06.50 | BullWeivel | anyone know if a Cisco 7920 can be flashed and used with SIP ??? wanting to connect to my new * box :) |
| 19:07.06 | *** join/#asterisk zotz (n=zotz@24.244.163.157) |
| 19:08.59 | *** join/#asterisk cr4z3d (n=cr4z3d@ip70-162-96-242.ph.ph.cox.net) |
| 19:12.27 | shtoom | i |
| 19:20.34 | ThOr101 | Hmm, the phone plugged into the FXS port rings when called, but no longer has a dialtone. |
| 19:21.00 | CVirus | immediate=no ? |
| 19:21.18 | CVirus | was just reading about it |
| 19:21.21 | CVirus | <--- is a n00b |
| 19:21.28 | *** join/#asterisk frenzy (n=frenzy@unaffiliated/frenzy) |
| 19:23.09 | ThOr101 | Hmm, rmmod the kernel modules, modprobe, restart * and it all works again. Weird. |
| 19:28.08 | shido6 | heh |
| 19:29.14 | *** join/#asterisk Strom_M (n=strom@ip70-170-60-8.lv.lv.cox.net) |
| 19:38.09 | *** join/#asterisk lowlevel (n=Stuart@CPE0017f2e2a3e8-CM000f9f7d6742.cpe.net.cable.rogers.com) |
| 19:40.23 | *** join/#asterisk marcan (i=1337@198.Red-83-54-248.dynamicIP.rima-tde.net) |
| 19:48.13 | *** join/#asterisk frenzy_ (n=frenzy@unaffiliated/frenzy) |
| 19:51.17 | ThOr101 | Would I really screw up * if I just started delete wav/txt/gsm files from the /var/spool/asterisk directories? |
| 19:51.35 | ThOr101 | I'm testing voicemail, and would rather blow away the files than delete them through the VM system |
| 19:55.59 | Strom_M | ThOr101: that wont mess up asterisk, no |
| 19:56.15 | Pagautas | hi |
| 19:56.34 | Pagautas | is it able to stop res_mysql logging into console |
| 19:56.44 | Pagautas | like res_odbc do |
| 19:58.31 | *** join/#asterisk Daejeo1 (n=chatzill@124.62.150.49) |
| 19:59.43 | Daejeo1 | is it possible to enable an instant message on asterisk? |
| 19:59.54 | Strom_M | what do you mean "instant message" |
| 20:00.01 | ThOr101 | Strom_M, Thanks |
| 20:00.16 | Daejeo1 | yahoo messesge |
| 20:00.27 | Daejeo1 | chatting |
| 20:00.29 | Strom_M | Daejeo1: what are you trying to do, exactly |
| 20:00.35 | Strom_M | asterisk is not a chat server |
| 20:00.39 | Strom_M | asterisk is a PBX |
| 20:01.07 | file | Friday night and the lights are low |
| 20:01.07 | Daejeo1 | but it is possible to integrate chat server |
| 20:01.10 | file | looking out for a place to go |
| 20:01.16 | Strom_M | dialing queen |
| 20:01.21 | Strom_M | something clean |
| 20:01.24 | Strom_M | only seventeen |
| 20:01.37 | Strom_M | dialing queen |
| 20:01.47 | Strom_M | feel the pulse of the rotary, oh yeah |
| 20:09.01 | *** join/#asterisk bmg505 (n=leon@196.209.177.251) |
| 20:09.28 | *** join/#asterisk xpot (n=jim@c-71-195-241-115.hsd1.ma.comcast.net) |
| 20:09.28 | kiscokid | I thought there was a jabber module in Asterisk |
| 20:15.08 | Daejeo1 | Strom_M http://www.voip-forum.com/index.php?p=184 |
| 20:18.14 | JunK-Y | kiscokid: res_jabber |
| 20:20.45 | ThOr101 | This IVRS is going to rock! |
| 20:21.06 | Daejeo1 | ? IVRS |
| 20:22.09 | ThOr101 | Yeah, I am setting up * to be an IVRS for my biz. |
| 20:22.13 | *** join/#asterisk SirThomas_Home (n=tomc@209-169-199-174.us.warpdriveonline.com) |
| 20:22.16 | ThOr101 | I'm recording all the prompts |
| 20:22.57 | *** join/#asterisk badcfe_ (n=cso@d80-170-243-199.cust.tele2.fr) |
| 20:24.02 | ThOr101 | It includes the black eyed peas |
| 20:24.30 | *** join/#asterisk rmayorga (n=rmayorga@unaffiliated/rmayorga) |
| 20:30.20 | *** join/#asterisk bmg505 (n=leon@196.209.177.251) |
| 20:31.48 | *** join/#asterisk ToyMan (n=Stuart@user-12lcqol.cable.mindspring.com) |
| 20:34.07 | *** join/#asterisk adorah (n=Michael@87.69.57.73.cable.012.net.il) |
| 20:34.09 | *** join/#asterisk ploieel (n=manni@Fb36e.f.ppp-pool.de) |
| 20:37.24 | *** join/#asterisk shido6 (i=shido6@d221-68-200.commercial.cgocable.net) |
| 20:39.21 | tzafrir_laptop | anybody here remembers the ps command to show CPU consumption of different threads? |
| 20:39.53 | tzafrir_laptop | I have asterisk in 100% CPU, and I'd like to know in which thread this is without stracing them all |
| 20:40.53 | ThOr101 | why not just use top? |
| 20:41.00 | *** join/#asterisk matsk (i=matsk@h110n2fls32o882.telia.com) |
| 20:41.39 | tzafrir_laptop | because it does not show different threads (on a decent kernel) |
| 20:41.41 | ThOr101 | shido6, the graphics on that site are really cool. |
| 20:42.22 | shido6 | ? |
| 20:42.26 | tzafrir_laptop | 'ps axmu' is a good start, but is does not show the pids of the different threads |
| 20:42.39 | ThOr101 | *theory |
| 20:42.44 | shido6 | yeah its not done yet |
| 20:44.13 | ThOr101 | If I send a caller off to VoiceMail(blah) will they ever return to the dialplan from that, or do they terminate in voicemail? |
| 20:44.55 | tzafrir_laptop | ps -eLf seems to be the closest. Seems actually useful |
| 20:45.57 | tzafrir_laptop | bah. strace showed nothinng. tried ltrace and it killed asterisk |
| 20:46.58 | *** join/#asterisk sakic (n=sakic@cpe-071-075-118-121.carolina.res.rr.com) |
| 20:47.27 | tzafrir_laptop | didn't kill asterisk. but seems to have killed the relevant thread |
| 20:47.44 | sakic | anyone know are the Grandstream GXP-2000 any good? |
| 20:48.32 | SirThomas_Home | sakic: its a nice phone, feels kinda toy-ish, cheep... but it works well. My nephew has one that doesn't work very well, but mine works fine. |
| 20:49.03 | SirThomas_Home | sakic: The biggest complaint *I* had about it is that you can't use the web interface while using the phone. |
| 20:49.22 | sakic | I am putting it in a business |
| 20:49.36 | sakic | maybe I should go polycom then... the 330 is my thought |
| 20:50.20 | SirThomas_Home | its hard to beat the price -vs- performance of the grandstreams. I have no experience with polycom. |
| 20:50.41 | sakic | I want something that is reliable |
| 20:50.51 | sakic | $25 a phone isn't a big deal, even $50 |
| 20:50.54 | adorah | price is all about GXP-2000 but there is an even cheaper Chinese clone |
| 20:51.11 | sakic | what is best in the $100-$125 range? |
| 20:52.22 | sakic | when they say 2 lines, what does that mean... one can be on hold and one can be talking on? |
| 20:52.32 | SirThomas_Home | yup, I think so |
| 20:52.35 | sakic | I will have 4-5 lines going into the biz |
| 20:52.47 | Nugget | heh: It says a lot about this industry that Steve Jobs can generate hype by claiming the "killer app" on the iPhone is "making calls". |
| 20:53.01 | sakic | maybe for the secretary I will need more "lines" |
| 20:53.09 | SirThomas_Home | sakic: will each person need to talk to 4-5 people at a time? Maybe just the receptionist phone needs more lines... |
| 20:53.19 | sakic | what about conference? |
| 20:53.34 | sakic | can I conference 3-4 or is that part of asterisk? |
| 20:53.49 | *** join/#asterisk Igbothom_III (n=Hilton@office.quarkit.com.au) |
| 20:53.57 | SirThomas_Home | yes |
| 20:54.17 | sakic | I think I will go w/ the polycom, I have read they have good sound |
| 20:54.24 | *** join/#asterisk tuxd00d (n=tuxinato@128.187.169.195) |
| 20:54.25 | ThOr101 | Ahh, you do get returned back to the dialplan after hitting # out of voicemail. |
| 20:54.55 | sakic | these phones don't need outlets right, they get powered off the cat5 right? |
| 20:59.49 | *** join/#asterisk abrdeco (n=tbmjrf@c9113931.rjo.virtua.com.br) |
| 20:59.50 | *** join/#asterisk kclaussen (n=kclausse@204.13.224.242) |
| 20:59.57 | sakic | do you just plug the linux machine into the same network that the voip is coming in on? |
| 21:01.52 | [TK]D-Fender | ~gs |
| 21:02.07 | jbot | GrandSuck phones & gateways are cheap junk which should be avoided with extreme prejudice. |
| 21:02.22 | abrdeco | Hi all, i'm needing some help. I do have an TE210P and using MFC/R2, some troubles of echo are ocurring and I take a like look at ztmonitor audio levels. |
| 21:03.22 | abrdeco | The audio level is too louder at TX of telephone company and when I try change rxgain/txgain to anything less than 0.0 default settings other problems are happening |
| 21:03.28 | Cresl1n | abrdeco: what version of zaptel are you running? |
| 21:04.12 | abrdeco | Zaptel 1.2 |
| 21:05.01 | abrdeco | When i change rxgain/txgain troubles the channel drops calls/doesn't complete another/caller id gets wrong numbers and echo at digits too. |
| 21:05.24 | Cresl1n | try zaptel 1.4 |
| 21:05.30 | abrdeco | I'm changing rxgain/txgain in unicall.conf |
| 21:05.32 | Cresl1n | it has a newer echo canceler in it |
| 21:05.35 | [TK]D-Fender | sakic, you need a PoE switch to power them that way |
| 21:06.20 | sakic | anyone use the asterisk with a voip provider? |
| 21:06.33 | Cresl1n | abrdeco: or try using MG2 from 1.4 |
| 21:06.35 | abrdeco | Cres1l1n - I think my first trouble is about the audio level, do you know if is possible change audio level at zaptel too ? |
| 21:06.35 | [TK]D-Fender | sakic, most, yes |
| 21:06.37 | Cresl1n | it's with 1.2 |
| 21:06.43 | sakic | any recommended? |
| 21:06.48 | [TK]D-Fender | sakic, Kind of a silly question... |
| 21:06.50 | Cresl1n | use the new echo canceler first |
| 21:06.56 | [TK]D-Fender | sakic, Depends where yuo are. |
| 21:06.59 | Cresl1n | then adjust audio levels |
| 21:07.04 | sakic | NC |
| 21:07.18 | Cresl1n | make sure you set them back to 0 before you play with the new echo canceler from 1.4 though |
| 21:07.19 | abrdeco | Do you know if will be necessary recompile all libraries of unicall/libpri/asterisk if I change the zaptel version ? |
| 21:07.36 | [TK]D-Fender | sakic, Several of my clients have been happy with VoicePulse, others have spoken well of Iaxtel |
| 21:07.53 | Cresl1n | abrdeco: probably |
| 21:08.09 | Cresl1n | unless you just copy the mg2ec.h file from 1.4 into 1.2 zaptel and recompile zaptel 1.2 |
| 21:08.13 | ThOr101 | Well I'm done with my IVRS. If anyone wants to poke at it msg me, and I'll give you the POTS number (703) area code. Cheers! |
| 21:08.22 | *** join/#asterisk putnopvut (n=putnopvu@user-24-214-124-177.knology.net) |
| 21:08.25 | Cresl1n | you'll have to switch it to use mg2ec.h in zconfig.h though |
| 21:08.30 | sakic | [TK] I am just trying to figure this stuff out, don't see how it comes in off the cable modem into the asterisk computer |
| 21:08.42 | abrdeco | And to enable the MG2 is at compile time or in configurations file ? |
| 21:08.51 | sakic | without some sort of adapter, or the asterisk needs to be programmed for the voip provider |
| 21:08.54 | Cresl1n | abrdeco: it's in zconfig.h |
| 21:09.00 | Cresl1n | compile time |
| 21:09.05 | Cresl1n | to get it |
| 21:09.06 | [TK]D-Fender | sakic, its all just UDP over the internet. Are you using a router in front of your modem? |
| 21:09.08 | Cresl1n | then runtime to use it |
| 21:09.17 | *** join/#asterisk bmg505 (n=leon@196.209.177.251) |
| 21:09.26 | sakic | it will be behind (the router of course) |
| 21:09.37 | sakic | cable modem ---> router --->asterisk |
| 21:09.47 | [TK]D-Fender | sakic, then you just forward a handful of ports to your * box |
| 21:10.01 | sakic | ahh it comes in on a certain port, didn't know that |
| 21:10.10 | abrdeco | Well I will do it now, thanks a lot Cresl1n |
| 21:10.13 | [TK]D-Fender | sakic, * would normally register with your ITSP (VoIP provider) and then you're in business |
| 21:10.33 | [TK]D-Fender | sakic, first start with.. THE BOOK |
| 21:10.33 | [TK]D-Fender | ~book |
| 21:10.38 | jbot | book is probably a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 21:10.38 | sakic | so I have to tell them my static ip |
| 21:10.38 | sakic | url? |
| 21:10.44 | [TK]D-Fender | sakic, You need to learn the fundamental technologies involved. |
| 21:11.19 | sakic | I just need it for my biz :) |
| 21:11.24 | [TK]D-Fender | ThOr101 if you want a proof-read, pastebin your dialplan |
| 21:12.51 | sakic | well I got trixbox setup, now need the phones, and I am thinking of using viatalk |
| 21:13.16 | [TK]D-Fender | sakic, in that case.... |
| 21:13.19 | [TK]D-Fender | ~trixbox |
| 21:13.26 | jbot | Trixbox is a full linux distro that includes , FreePBX, and other 3rd party add-ons. It is these things on top of which make it seriously painful to support and hence you will find little help here for it. Try asking in #trixbox , or their forums & WIKI at http://www.trixbox.org |
| 21:13.26 | ThOr101 | http://paste.debian.net/28969 |
| 21:13.55 | *** join/#asterisk witwolf (n=carel@196.30.118.76) |
| 21:14.33 | witwolf | Hi, Does anyone here know the inner workings of zaptel? |
| 21:15.12 | sakic | ok maybe I should hire someone to set it up :P |
| 21:15.23 | [TK]D-Fender | ThOr101, a start... you now REALLY need to implement the "t" and "i" standard extensions to handle people taking too long or entering something invalid |
| 21:15.26 | sakic | what would I pay someone to install, set up? |
| 21:15.42 | [TK]D-Fender | sakic, depends on your needs |
| 21:15.47 | ThOr101 | Oh yeah. I read about that, tinkered with it, and forgot to add it. I'll go toss it in there. Thanks. |
| 21:16.08 | witwolf | I am trying to trace a problem I've had for 4 months now. |
| 21:16.11 | sakic | nothing fancy, voicemail, caller id, forwarding |
| 21:16.15 | sakic | basic features |
| 21:16.21 | [TK]D-Fender | sakic, PM |
| 21:17.29 | witwolf | My box works fine, MOH turns to noise on only the FXO when it gets soft. Anyone know what the problem might be? |
| 21:19.32 | witwolf | Have looked everywhere in the net. Even started to go thru code to try and find the source |
| 21:21.50 | witwolf | If no one can help me here, please direct me in the right direction! |
| 21:22.05 | kombi | speaking of moh, what format does that need to be in? gsm? mp3? |
| 21:22.36 | [TK]D-Fender | witwolf, First I'd suspect its your echo cancellation. Turn it off and see if its clear (ignore the echo). If you can, there it is... gain issues |
| 21:22.39 | ThOr101 | http://paste.debian.net/28970 now with t&i |
| 21:23.07 | [TK]D-Fender | kombi, * will transcode from whatever format the files are in. if you have MoH encoded in the same codec as the call you will reduce server load during MoH |
| 21:23.17 | [TK]D-Fender | t&a ... my favourite! |
| 21:23.21 | kombi | thank Fender! |
| 21:23.25 | kombi | s.. |
| 21:23.25 | ThOr101 | Ba ha! |
| 21:23.37 | [TK]D-Fender | ThOr101, Yup, basic, and better... |
| 21:24.01 | *** join/#asterisk drako (n=ljd@nelug/coreteam/luisjose) |
| 21:24.18 | [TK]D-Fender | ThOr101, Though on timeout I might think that you'd want it to fall toa default box of some kind... for those with DTMF, etc... |
| 21:25.06 | ThOr101 | Would it be best to push them to voicemail, and then hangup on them? Rather than VM and back to the main menu? |
| 21:26.00 | *** join/#asterisk `Sean (i=Un1x@CPE000c248d137c-CM00111ae601f8.cpe.net.cable.rogers.com) |
| 21:26.46 | kombi | how does one best debug MWI issues? Is there some low level testing method? |
| 21:27.04 | [TK]D-Fender | ThOr101, if they fell to timeout one could assume they can't handle the menu for whatever reason, so best to hangup I would think. |
| 21:27.19 | *** join/#asterisk phobus (n=phobus@crlspr-69.65.75.232.myacc.net) |
| 21:27.21 | [TK]D-Fender | kombi, depends what the issues ARE clearly ;) |
| 21:27.24 | ThOr101 | Concur, thanks. |
| 21:28.49 | kombi | Fender: no indication of waiting messages, is it a standard in the sip protocol? |
| 21:29.01 | [TK]D-Fender | kombi, yes. |
| 21:29.42 | kombi | hmm, would x-lite indicate waiting messages? |
| 21:30.31 | kombi | if it does, it must be something in my conf |
| 21:30.38 | [TK]D-Fender | kombi, yes, normally via a little folder icon. |
| 21:30.52 | [TK]D-Fender | ont he bottom edge of the "display" area |
| 21:31.41 | kombi | I see an envelope icon there, that is not it though? |
| 21:31.41 | *** join/#asterisk TJ` (n=tj@host81-154-60-31.range81-154.btcentralplus.com) |
| 21:31.41 | TJ` | hi... |
| 21:31.41 | [TK]D-Fender | kombi, Should be... |
| 21:32.55 | kombi | Fender: yip, that is it, so it must be the phone |
| 21:32.57 | TJ` | was wondering if anyone could help me out? i can successfully connect to my box from an external connection, but it wont pass sound, though all the phones on the same network do... |
| 21:33.06 | [TK]D-Fender | TJ`, ... |
| 21:33.08 | [TK]D-Fender | ~sipnat |
| 21:33.19 | jbot | hmm... sipnat is for for more information about configurtion of Asterisk with SIP behind NAT, see http://voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
| 21:33.20 | TJ` | im a noob |
| 21:34.01 | TJ` | thanks [TK]D-Fender ill look into it |
| 21:35.38 | TJ` | is there a list of porst that can be opened? |
| 21:36.25 | TJ` | ouch everything between 10,000 and 20,000 |
| 21:36.44 | [TK]D-Fender | TJ`, SIP normally should have 5060, 10000-20000 all UDP. * needs OTHER settings in sip.conf as well. |
| 21:36.44 | *** join/#asterisk yannj_fr (n=yannj@eberkut.net1.nerim.net) |
| 21:37.20 | TJ` | odds of a voip server being hacked.... |
| 21:37.52 | TJ` | hell ill just open everything for that machine i think |
| 21:38.19 | [TK]D-Fender | TJ`, you're call.... |
| 21:38.51 | TJ` | theres nothing on that network anyway :) |
| 21:38.52 | TJ` | tho |
| 21:38.54 | TJ` | i found this |
| 21:39.00 | TJ` | http://www.bcr.com/architecture/briefing/voip_v_firewalls_20020719288.htm |
| 21:39.08 | TJ` | those ports alone wouldnt work would they |
| 21:39.40 | yannj_fr | does anyone knows differences between user.conf and sip.conf |
| 21:41.03 | yannj_fr | and why I cannot make work the presence led (hint) if the user is only defined in user.conf and not in sip.conf |
| 21:41.24 | [TK]D-Fender | TJ`, The ones I specified are the ones you need. Thats all. |
| 21:41.26 | *** join/#asterisk blitzrage (n=blitzrag@asterisk/documenteur-extraordinaire/blitzrage) |
| 21:41.28 | JunK-Y | yannj_fr: sip.conf is only for sip, users contains a much more general architecture, for sip,iax, voicemail, etc |
| 21:41.38 | TJ` | [TK]D-Fender ill just open those then, thanks for your help :) |
| 21:42.07 | yannj_fr | but even when I configure hassip for a user, the hints dont work |
| 21:42.10 | abrdeco | NAobody here uses libunicall 0.3 ? |
| 21:42.35 | *** join/#asterisk SECGOD (i=SECGOD@c-71-57-36-106.hsd1.il.comcast.net) |
| 21:43.09 | [TK]D-Fender | yannj_fr, pastebin all your applicable configs. |
| 21:43.10 | [TK]D-Fender | ~pb |
| 21:43.24 | jbot | pb is probably a place to paste your stuff without flooding the channel - try http://pastebin.ca, or http://channels.debian.net/paste |
| 21:43.39 | abrdeco | Anybody uses rxgain/txgain to lower audio on E1 channels ? |
| 21:43.59 | yannj_fr | [6009] |
| 21:43.59 | yannj_fr | callwaiting = yes |
| 21:43.59 | yannj_fr | cid_number = 6009 |
| 21:43.59 | yannj_fr | context = |
| 21:43.59 | yannj_fr | email = |
| 21:43.59 | yannj_fr | fullname = New User |
| 21:44.01 | yannj_fr | group = |
| 21:44.03 | yannj_fr | hasagent = yes |
| 21:44.05 | yannj_fr | hasdirectory = no |
| 21:44.07 | yannj_fr | hasiax = yes |
| 21:44.09 | yannj_fr | hasmanager = no |
| 21:44.11 | yannj_fr | hassip = yes |
| 21:44.13 | yannj_fr | hasvoicemail = yes |
| 21:44.15 | yannj_fr | host = dynamic |
| 21:44.17 | yannj_fr | mailbox = 6009 |
| 21:44.19 | yannj_fr | secret = |
| 21:44.21 | yannj_fr | threewaycalling = yes |
| 21:44.23 | yannj_fr | vmsecret = 1234 |
| 21:44.25 | yannj_fr | zapchan = |
| 21:44.27 | yannj_fr | registeriax = yes |
| 21:44.29 | yannj_fr | registersip = yes |
| 21:44.31 | yannj_fr | example for user.conf |
| 21:44.34 | ThOr101 | I guess he didn't catch that pastebin line ;-) |
| 21:45.18 | yannj_fr | pastebin? |
| 21:45.41 | ThOr101 | scroll up |
| 21:45.43 | yannj_fr | ok |
| 21:45.48 | yannj_fr | sorrrrry |
| 21:45.56 | JunK-Y | ~pb |
| 21:46.10 | jbot | pb is, like, a place to paste your stuff without flooding the channel - try http://pastebin.ca, or http://channels.debian.net/paste |
| 21:46.15 | ThOr101 | no problem, just fill out a TPS report and everything will be ok |
| 21:46.48 | *** join/#asterisk Ciber311 (n=Ciber311@user-12ld42j.cable.mindspring.com) |
| 21:47.41 | abrdeco | I getting one issue with Asterisk Queues. I have 10 agents logged on, and when I got more than 1 call waiting to be answered the queue waits the first one be answered to start riging the second agent. Does anybody know how fo fix it ? |
| 21:49.32 | yannj_fr | http://www.intelunix.fr/asterisk |
| 21:49.51 | yannj_fr | http://www.intelunix.fr/asterisk/chan.txt |
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| 21:51.51 | blitzrage | abrdeco: see the queues.conf.sample file -- yes, it is an option in queues.conf that you will need to enable |
| 21:53.56 | [TK]D-Fender | abrdeco, There is an option in later 1.2 & 1.4 releases that will fill all open agents. |
| 21:55.06 | abrdeco | Thanks FTKDender |
| 21:55.23 | yannj_fr | strategy = ringall |
| 21:55.26 | yannj_fr | isnt it? |
| 21:55.46 | abrdeco | No, ringall will ring the same first call on the queue to all. |
| 21:55.49 | yannj_fr | autofill = yes |
| 21:56.10 | abrdeco | Sure I will change my Asterisk version and try autofill right now. |
| 21:56.53 | yannj_fr | any idea for my problem with users.conf and hits |
| 21:56.55 | yannj_fr | hints* |
| 21:57.43 | abrdeco | The audio level showed by ztmonitor is too louder but the config files like unicall.conf when I try rxgain/txgain lower than 0 strange trings happens like: |
| 21:57.52 | abrdeco | Caller ID with duplicated digis, missing digits, calls drops, calls aren't received and so. Can that be a bug in zapata or Libunicall ? |
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| 22:02.27 | TJ` | [TK]D-Fender ive opened all the ports and still no sound |
| 22:03.09 | TJ` | :( |
| 22:04.11 | ThOr101 | Any thought to opening up ethereal/wireshark and checking to see what packets are getting whacked? |
| 22:04.21 | [TK]D-Fender | TJ`, I did say there were a whole whack of other setting to do in sip.conf for * to work... |
| 22:04.23 | [TK]D-Fender | ~sipnat |
| 22:04.35 | jbot | it has been said that sipnat is for for more information about configurtion of Asterisk with SIP behind NAT, see http://voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
| 22:04.36 | [TK]D-Fender | ThOr101, he hasn't fixed the * side yet. |
| 22:04.49 | ThOr101 | Aiight. |
| 22:05.55 | SirThomas_Home | I'm looking for a ENUM test number... |
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| 22:06.13 | *** mode/#asterisk [+o mog] by ChanServ |
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| 22:08.28 | john-eman0n | Digium TDM11B is what i need right? |
| 22:08.34 | john-eman0n | to spoof etc |
| 22:09.15 | [TK]D-Fender | john-eman0n, no. |
| 22:09.20 | carrar | to spoof? |
| 22:09.20 | Cresl1n | john-eman0n: no |
| 22:09.25 | Cresl1n | you need a a PRI |
| 22:09.31 | [TK]D-Fender | john-eman0n, To set your own callerid you need a digital link to the telco |
| 22:09.38 | john-eman0n | look |
| 22:09.43 | john-eman0n | iv done it befor |
| 22:09.43 | john-eman0n | lol |
| 22:10.06 | kombi | buggymwi=yes and off you go, I love this software! |
| 22:10.09 | [TK]D-Fender | john-eman0n, not on ANALOG you haven't |
| 22:10.16 | Cresl1n | it only works if you plug your own phone into it |
| 22:10.23 | john-eman0n | yes |
| 22:10.23 | john-eman0n | i know |
| 22:10.27 | Cresl1n | you can't send it across the phone network |
| 22:10.27 | john-eman0n | i do not want soft phone |
| 22:10.34 | [TK]D-Fender | john-eman0n, At least not without using your line to call a 3rd party service to do the actual dirty work for you |
| 22:10.43 | john-eman0n | voipstreet.com + that car + phone pluged in |
| 22:10.44 | john-eman0n | works |
| 22:10.45 | john-eman0n | right? |
| 22:10.49 | john-eman0n | card* |
| 22:10.49 | [TK]D-Fender | john-eman0n, Nobody said anything about a soft-phone. |
| 22:11.01 | john-eman0n | <Cresl1n> it only works if you plug your own phone into it |
| 22:11.21 | [TK]D-Fender | john-eman0n, you don't need that card to sue a regular phone. ATA's do the job better & cheaper for the mostpart |
| 22:11.33 | john-eman0n | i mean |
| 22:11.37 | john-eman0n | i have the extra money |
| 22:11.42 | john-eman0n | i just want my old setup back |
| 22:11.42 | [TK]D-Fender | john-eman0n, and the card has nothing to do with rigging the callerid... thats jsut you & your provider |
| 22:11.44 | john-eman0n | like im use to |
| 22:11.51 | yannj_fr | [TK]D-Fender : Did you see anyproblem on http://www.intelunix.fr/asterisk/chan.txt |
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| 22:12.04 | [TK]D-Fender | yannj_fr, Soory, I can't comment on users.conf. |
| 22:13.44 | yannj_fr | in fact as soon as I added the phone in sip.conf, it works |
| 22:14.13 | [TK]D-Fender | yannj_fr, I so suspect that you should have it as "mailbox = 6000@[somecontext]" however |
| 22:14.56 | yannj_fr | why? |
| 22:15.25 | *** join/#asterisk Vec (n=Vec@dsl-243-125-65.telkomadsl.co.za) |
| 22:15.27 | [TK]D-Fender | yannj_fr, just try it..... its the proper way of referencing a box and used to screw up Polycom's and a few other when you weren't explicit |
| 22:15.38 | yannj_fr | ok |
| 22:16.00 | yannj_fr | <PROTECTED> |
| 22:16.18 | yannj_fr | snom, grandstream, cisco, thomson... |
| 22:23.10 | [TK]D-Fender | yannj_fr, it is my stance that users.conf = useless... |
| 22:23.10 | *** join/#asterisk vAd0r (n=IceChat7@65.67.210.121) |
| 22:24.35 | yannj_fr | useless? |
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| 22:28.15 | [TK]D-Fender | yannj_fr, echocancel=yes |
| 22:29.34 | vAd0r | can someone assist me w/ unlocking my linksys pap2 |
| 22:30.05 | yannj_fr | ... |
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| 23:36.33 | mosty | i'm having trouble getting an E1 pri line working, the D channel is down |
| 23:37.29 | mosty | what could be wrong? the card is a te412p, the light on that port is green when i plug it in, pri show span 1 has this status line: Status: Provisioned, Down, Active |
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| 23:54.31 | EvilDeshi | anyone here use sip broker? |
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| 23:59.26 | juliusspencer | hi, has anyone here returned a digium card? |
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