| 00:03.22 | *** join/#asterisk davidcsi (n=david@45.Red-79-145-214.dynamicIP.rima-tde.net) |
| 00:03.40 | davidcsi | hello, anybody got experience with ss7? |
| 00:05.49 | ghento | Hmm it looks like it was something to do with the 'w' option that I include. When I remove it, the audio is present. I do need the call recording capability though so the debugging continues heh |
| 00:08.17 | *** join/#asterisk jer (n=jtregunn@unaffiliated/jer) |
| 00:11.15 | davidcsi | does anybody know if it is possible to "bridge" 2 SS7 links? i.e. connect in the middle of a link and just do a bridge in order to see messages going through? |
| 00:12.53 | thebbxx2000 | so when I am running in verbose mode, where is all the output supposed to show up? |
| 00:13.31 | mmlj4 | thebbxx2000: in your console |
| 00:13.38 | thebbxx2000 | ok |
| 00:13.53 | AsteriskDom | did you configured your phone? |
| 00:14.25 | thebbxx2000 | working on it... there are a few options listed under "Basic SIP Network settings" |
| 00:14.30 | thebbxx2000 | and i'm not sure which one to fill out |
| 00:14.32 | thebbxx2000 | there is |
| 00:14.41 | thebbxx2000 | Proxy Server, Outbound Proxy Server |
| 00:14.45 | thebbxx2000 | Registrar Server |
| 00:14.56 | AsteriskDom | proxy server |
| 00:14.59 | thebbxx2000 | ok |
| 00:15.03 | thebbxx2000 | port? |
| 00:15.13 | AsteriskDom | default |
| 00:15.32 | thebbxx2000 | what is the default port? |
| 00:15.36 | Fabian- | 5060 |
| 00:15.39 | thebbxx2000 | thanks |
| 00:17.25 | thebbxx2000 | should it show some output if the phone registers with the server? |
| 00:17.33 | thebbxx2000 | i restarted the phone but don't see any output |
| 00:19.11 | AsteriskDom | sip show peers |
| 00:19.23 | AsteriskDom | and verify if it is registered |
| 00:19.27 | mmlj4 | thebbxx2000: have you tried any of this? you're asking questions that indicate you |
| 00:19.45 | mmlj4 | thebbxx2000: have you tried any of this? you're asking questions that indicate you've never even started up asterisk before |
| 00:20.00 | mmlj4 | try "asterisk -rvvvvv" and see what happens |
| 00:20.36 | thebbxx2000 | mmlj4: i haven't setup asterisk before |
| 00:20.47 | mmlj4 | then try it, see what happens |
| 00:20.51 | thebbxx2000 | k, hang on |
| 00:21.09 | mmlj4 | half of what you've asked answers itself if you run it |
| 00:21.31 | thebbxx2000 | thanks, what does the r stand for |
| 00:21.51 | thebbxx2000 | and how do i close asterisk from the CLI |
| 00:21.53 | mmlj4 | something or other, I've forgotten, I'd have to read the docs again |
| 00:22.02 | mmlj4 | ~book |
| 00:22.03 | infobot | [~book] Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://astbook.asteriskdocs.org or see ~buybook |
| 00:22.37 | mmlj4 | we get tired of spoon-feeding people after a while |
| 00:23.17 | thebbxx2000 | i understand |
| 00:23.20 | *** join/#asterisk umpc (n=Justin@unaffiliated/umpc) |
| 00:23.36 | thebbxx2000 | figured it out though - stop now |
| 00:24.28 | thebbxx2000 | so i ran the command you mentioned with -rvvvvv and it says it's unable to connect to a remote asterisk |
| 00:24.59 | *** part/#asterisk davidcsi (n=david@45.Red-79-145-214.dynamicIP.rima-tde.net) |
| 00:26.13 | *** join/#asterisk Magicblaze0071 (n=sony@fl-67-233-199-108.dhcp.embarqhsd.net) |
| 00:26.17 | *** join/#asterisk tfrew (n=tfrew@c-68-57-89-103.hsd1.va.comcast.net) |
| 00:27.07 | ghento | Has anyone got automon call recording working with asterisk 1.4.x? I have 'wW' as options in my Dial() however whenever they are added, no audio comes through. I have 'automon => *1' in my features.conf, and make sure to Set(DYNAMIC_FEATURES=automon) prior to my Dial() but no luck. Any ideas? |
| 00:29.44 | *** join/#asterisk [intra]lanman (n=lanman@freeswitch/developer/intralanman) |
| 00:30.09 | *** join/#asterisk AsteriskDom (n=munoz@190.94.5.57) |
| 00:30.24 | *** join/#asterisk joobie (n=joobie@mx01.anric.com.au) |
| 00:30.52 | *** join/#asterisk bbkt-trix (n=bbkt-tri@unaffiliated/bbkt-trix) |
| 00:36.21 | *** join/#asterisk rue_mohr (n=rue@h24-207-90-17.cst.dccnet.com) |
| 00:36.57 | rue_mohr | so, the managemnt has just about had it with the asterisk phone system, and what sucks is that most of it is over issues they didn't tell me about, their looking into cisco |
| 00:37.42 | rue_mohr | right now I'm fighting with "we cant hear people on our phones, even when their turned all the way up" |
| 00:38.32 | rue_mohr | today the system started dropping bits of audio, the only things i had changed were the fax detect for the fax line (now set back to default) and the persistant volume settings on the polycom phones |
| 00:38.33 | tfrew | hmm |
| 00:38.36 | jaytee | rue_mohr, on sip to sip calls or calls from a PSTN? |
| 00:38.37 | rue_mohr | (also now set back) |
| 00:38.41 | tfrew | make them look at polycoms instead |
| 00:38.44 | rue_mohr | polycom phones suck :) |
| 00:38.52 | tfrew | so |
| 00:38.59 | rue_mohr | calls via asterisk from the digium card |
| 00:38.59 | jaytee | our Polycoms works just fine |
| 00:39.03 | tfrew | they cost almost as much as cisco |
| 00:39.16 | tfrew | and that means they are better |
| 00:39.18 | tfrew | right?? |
| 00:39.28 | jaytee | have you tried boosting the RX gain setting in your zaptel/dahdi configuration? |
| 00:39.30 | rue_mohr | I swear that I cant adjust the dahdi gains cause of the echo canceler, I tried all the way up to 6db and there was no difference |
| 00:39.57 | rue_mohr | well I dont know what the point of view settings are for "rx" and "tx" so I'v tried it all |
| 00:40.23 | rue_mohr | there are two audio paths, there are two tx's and two rx's for every device |
| 00:40.42 | jaytee | if you're at the end with the card the RX is for audio from the PSTN and TX is audio to the PSTN |
| 00:40.50 | rue_mohr | rx from one side, rx from the other side, same with tx |
| 00:41.24 | rue_mohr | so for the dahdi config, if the person in our office wants more volume ont eh speaker I shoudl dial up rx |
| 00:41.26 | rue_mohr | yes? |
| 00:41.35 | jaytee | yes |
| 00:41.38 | rue_mohr | anyhow I'v done that, to 6db and there was no difference |
| 00:41.40 | thebbxx2000 | if i want to dial an extension to another phone, do i have to do anything special or just punch in the extension and press dial? |
| 00:41.48 | rue_mohr | I should notice a 6db difference |
| 00:43.55 | *** join/#asterisk MaliutaLap (n=biteme@kiev.lusan.id.au) |
| 00:44.11 | *** part/#asterisk thebbxx2000 (n=bbxx@c-67-182-169-155.hsd1.ca.comcast.net) |
| 00:44.11 | jaytee | thebbxx2000, if you want to dial an extension in asterisk it has to exist in your dialplan for asterisk to "find" it. |
| 00:44.25 | tfrew | this might be a dumb question, but did you plug a molex into the digium card? |
| 00:44.33 | jaytee | another friggin newb with no patience |
| 00:44.48 | tfrew | thebbxx2000, just dial the extension |
| 00:44.57 | jaytee | tfrew, too late he quit |
| 00:45.04 | tfrew | oh |
| 00:45.09 | rue_mohr | I'm trying to set the gains up on the polycom phones, it really sucks you have to wait 5 minutes for every reboot to see if you have the right setting cause of lack of ability to set anythingfrom their web interface |
| 00:45.39 | rue_mohr | it also sucks that the company that sold me the polycom phones cant help descently with tech support |
| 00:45.40 | tfrew | rue_mohr are you centrally provisioning them? |
| 00:45.46 | rue_mohr | yes |
| 00:46.02 | jaytee | mine are provisioned via FTP with DHCP and only take about a minute and half to be fully operational |
| 00:46.02 | tfrew | the gain should not be on the polycom side |
| 00:46.16 | tfrew | jaytee: you must have 330's and 650's |
| 00:46.19 | tfrew | i have a couple of those |
| 00:46.22 | jaytee | yes |
| 00:46.30 | tfrew | but my 601 is slow as hell :P |
| 00:46.33 | rue_mohr | well I have to dial it up somewhere I dont CARE where, I need to make this system APPEAR to work |
| 00:46.39 | tfrew | i'm going to replace it, when sip 3.2 comes out |
| 00:46.40 | rue_mohr | yea, mine are all 601 |
| 00:46.56 | jaytee | and I'm using the small footprint sip.ld file specific to each model, not the generic one size fits all bloated sip.ld file |
| 00:46.58 | tfrew | rue_mohr: it should be the diguim card that is the problem |
| 00:46.59 | rue_mohr | tk recomended them |
| 00:47.29 | rue_mohr | I CANT ADJUST THE GAIN ON TEH DIGUIM CARD BECASUE THE HARDWARE ECHO CANCELER HAS CONTROL OF THAT <<<<<---- please prove me wrong? |
| 00:47.40 | rue_mohr | I'v adjusted the gains by 6db and had _NO_ difference |
| 00:47.54 | tfrew | do you have the molex power adaptor plugged in? |
| 00:48.04 | rue_mohr | sorry, I'm frustrated as hell, cause of nothing less than them thinking that asterisk sucks |
| 00:48.08 | rue_mohr | yes, its plugged in |
| 00:48.41 | jaytee | I used Grandstreams with great pain because my boss is a cheapo. I finally convinced him to go in another direction. I tested Aastras and didn't care for them. too clunky. I tested 2 Polycoms and while the configuration can be a pain in the ass to setup once you've got it it's a breeze and the audio quality from my point of view is unsurpassed. |
| 00:49.15 | jaytee | rue_mohr, you have a mix of FXO and FXS modules on an analog card from Digium? |
| 00:49.29 | rue_mohr | I was going to get aastra and I REALLY wish I had, the documentation, support, and setting access is FAR FAR FAR supperior to polycom |
| 00:49.35 | rue_mohr | yes |
| 00:50.05 | jaytee | ok, cuz you only need the molex connector for FXS modules for analog phone sets, not for FXO modules with analog lines from the PSTN |
| 00:50.12 | Octothorpe | Does anyone have any recommendations for SIP softphones for Mac OS X? |
| 00:50.29 | rue_mohr | yea, we have two analog stations, a cordless and a fax machine |
| 00:50.43 | rue_mohr | which is another issue, SOME clients we cant fax |
| 00:50.47 | jaytee | and the rest of the modules on the card are FXO then |
| 00:50.49 | AsteriskDom | xlite for mac |
| 00:50.52 | rue_mohr | it tries but cant connect |
| 00:51.10 | rue_mohr | yes, its a TDM800 with 4 fxo and 2 fxs |
| 00:51.18 | rue_mohr | and the hwec |
| 00:51.26 | *** join/#asterisk pl0sh (i=pyr0@200.92.44.44) |
| 00:51.29 | joobie | guys i use pennytel as an uplink for sip calls... i need to "relogin" to their system so that some new settings can apply.. how can i force asterisk to do this? |
| 00:51.55 | KyleK | is the HandyTone 503 a good piece of hardware? |
| 00:51.59 | jaytee | rue_mohr, echocancel=yes and echocancelwhenbridged=yes are set? |
| 00:52.04 | tfrew | joobie: restart your asterisk service |
| 00:52.08 | rue_mohr | yes |
| 00:52.13 | joobie | only way to do it tfrew ? |
| 00:52.18 | rue_mohr | jaytee, |
| 00:52.26 | tfrew | you can do it in the cli to |
| 00:52.29 | rue_mohr | when bridged might be off |
| 00:52.34 | joobie | how tfrew ? |
| 00:52.34 | tfrew | but i don't know the command |
| 00:52.39 | pl0sh | hey there, i am using kiax to connect to my asterisk, i can do it by local, but if i try to connect remotedly it fails..? |
| 00:52.40 | rue_mohr | I'm at home, I cant access the system from here |
| 00:52.43 | pl0sh | any idea? |
| 00:52.46 | tfrew | i think it's sip reregister somthing |
| 00:52.56 | jaytee | module reload chan_sip.so |
| 00:53.01 | jaytee | that's for joobie |
| 00:53.09 | tfrew | there you go |
| 00:53.28 | rue_mohr | jaytee, changing rxgain to 6 in chan_dahdi.conf shoudl make one hell of a difference, right? |
| 00:53.33 | tfrew | i use amportal so i wouldn't know :P |
| 00:53.39 | ghento | argh Dial() shouldn't be this frustrating heh |
| 00:53.42 | jaytee | rue_mohr, 6db isn't huge |
| 00:53.44 | joobie | i ended up doin an /etc/init.d/asterisk restart |
| 00:53.49 | joobie | that would have reloaded the module right jay? |
| 00:53.52 | rue_mohr | 6db is 4xyesno? |
| 00:53.56 | jaytee | joobie, yes |
| 00:53.56 | tfrew | yes |
| 00:53.57 | rue_mohr | 6db is 4x yesno? |
| 00:53.58 | joobie | cheers |
| 00:54.02 | joobie | thanks tf / jay |
| 00:54.17 | rue_mohr | 3db is 2x |
| 00:54.33 | rue_mohr | whats the max gain for a dahdi ? |
| 00:54.45 | rue_mohr | 6 shoudl be stupidly high |
| 00:55.14 | rue_mohr | 9 shoudl be outrageos (8x iirc) |
| 00:55.39 | rue_mohr | trims back the timing on the d key |
| 00:56.32 | *** join/#asterisk thebbxx2000 (n=bbxx@c-67-182-169-155.hsd1.ca.comcast.net) |
| 00:56.35 | rue_mohr | I dunno, I feel I'v failed and only managed to make the word asterisk suck in the minds of 6 more people |
| 00:56.48 | kc8pxy | i recently came by a linksys rtp300. it seems someone may have dumped me their ATA. do i simply need to unlock the provisioning, for me to be able to program it for my asterisk server? |
| 00:57.04 | jaytee | rue_mohr, you keep using db but what is the actual value of rxgain= in your config? |
| 00:57.15 | rue_mohr | I tried 6db |
| 00:57.23 | rue_mohr | aka 6.0 |
| 00:57.26 | thebbxx2000 | My phones are connected now... but when i try to dial the other one I just get a busy signal and the CLI says unable to create channel of type 'SIP' |
| 00:57.39 | rue_mohr | the gain IS in db, yesno? |
| 00:57.56 | tfrew | thebbxx2000: how is your outbound route setup? |
| 00:58.06 | rue_mohr | I shoudl try setting is to 10, I'm gonna notice that! |
| 00:58.11 | rue_mohr | even if it isn't db |
| 00:58.23 | thebbxx2000 | i only have added lines to extensions.conf and sip.conf |
| 00:58.26 | thebbxx2000 | in extenstions I have |
| 00:58.38 | thebbxx2000 | exten => 100,1,Dial(SIP/100) |
| 00:58.55 | thebbxx2000 | exten => 101,1,Dial(SIP/101) |
| 00:59.03 | joat | thebbxx2000: you configured the phones too? |
| 00:59.07 | thebbxx2000 | yeah |
| 00:59.13 | rue_mohr | but I have it engrained in my head that when the echo can is on, it controls the channel gains as part of its anti-echo behaviour |
| 00:59.18 | tfrew | you can dial 7777 on the phone |
| 00:59.24 | thebbxx2000 | they show up when i do sip show peers |
| 00:59.27 | thebbxx2000 | hang on |
| 00:59.33 | tfrew | can you dial 7777 on the phones |
| 00:59.36 | jaytee | rue_mohr, yes the setting is in db |
| 00:59.38 | tfrew | or dial each other |
| 00:59.53 | AsteriskDom | but they show up in OK status |
| 00:59.55 | AsteriskDom | ? |
| 01:00.06 | jaytee | so rxgain=3.0 would give you 3db of amplification or roughly 5mw increase |
| 01:00.21 | thebbxx2000 | 7777 says rejected because extension not found |
| 01:00.47 | jaytee | thebbxx2000, do you have extension 7777 in your dialplan in extensions.conf? |
| 01:00.48 | thebbxx2000 | status says "unmonitored" for the phones when i do show peers |
| 01:00.53 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 01:00.58 | thebbxx2000 | i don't think so |
| 01:01.13 | thebbxx2000 | tfrew asked me to do that |
| 01:01.13 | jaytee | then perhaps it's time for you to actually read the book? |
| 01:01.24 | AsteriskDom | dial 8500 from one of the phones |
| 01:01.46 | tfrew | sorry, i'm thinking premade asterisk distros |
| 01:01.59 | tfrew | they have 7777 for testing |
| 01:02.01 | thebbxx2000 | 8500 - same thing .. u joking with me? |
| 01:02.07 | thebbxx2000 | ah |
| 01:02.09 | AsteriskDom | nope |
| 01:02.17 | KyleK | is the Grandstream HandyTone 503 a good piece of hardware? |
| 01:02.22 | jaytee | ~gs |
| 01:02.23 | infobot | GrandSuck phones & gateways are cheap junk which should be avoided with extreme prejudice. |
| 01:02.26 | tfrew | they should be monitored |
| 01:02.27 | AsteriskDom | 8500 its the default VoicemailMain extension |
| 01:02.30 | tfrew | for an internal phone |
| 01:02.33 | thebbxx2000 | oh |
| 01:02.38 | rue_mohr | http://www.sengpielaudio.com/calculator-gainloss.htm 6db is a gain of 2.... what... |
| 01:02.42 | thebbxx2000 | says extension not found |
| 01:03.10 | rue_mohr | so I'll try 12db |
| 01:04.06 | tfrew | thebbxx2000: is your phone using your extension as the user and your defined secret to register? |
| 01:04.06 | *** join/#asterisk fnordus (n=dnall@70.71.225.48) |
| 01:04.13 | jaytee | thebbxx2000, you're just gonna be beating yourself over the head if you don't take advantage of the materials available. read the damn book and specifically pay attention to Chapter 5 |
| 01:04.41 | thebbxx2000 | is there a link? |
| 01:04.49 | jaytee | yes, for the second time |
| 01:04.52 | jaytee | ~book |
| 01:04.53 | infobot | [~book] Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://astbook.asteriskdocs.org or see ~buybook |
| 01:04.58 | thebbxx2000 | thanks |
| 01:05.02 | jaytee | you can download the PDF |
| 01:05.13 | *** join/#asterisk moy (n=moy@CPE001cdfec4cee-CM00080dab8485.cpe.net.cable.rogers.com) |
| 01:05.26 | rue_mohr | ah but this is power, so 3db is 2x |
| 01:05.44 | tfrew | ~book |
| 01:05.44 | infobot | [~book] Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://astbook.asteriskdocs.org or see ~buybook |
| 01:06.15 | jaytee | thebbxx2000, note that the book covers up to 1.4 and there have been some changes in 1.6 where some features were deprecated. Those are covered in the text files included in the tarballs i.e. upgrade.txt |
| 01:06.28 | AsteriskDom | exten => 8500,1,VoicemailMain |
| 01:06.30 | rue_mohr | so , I' should have noticed 6db, therefore, the rxgain setting in chan_dahdi.conf had no effect |
| 01:07.08 | jaytee | you should read those as well after reading up to Chapter 6 at least of the book to give you a better understanding of * and what needs to be configured. It's a toolkit, not a fully grown PBX. You build it yourself. |
| 01:08.06 | jaytee | is the rxgain statement before or after the channel => statement? |
| 01:08.45 | jaytee | and did you restart dahdi after making changes? |
| 01:14.28 | ghento | Hmm when I add the 'wW' option into Dial(), debug does not show 'Native bridging' between the two SIP calls, and there is no audio. However when the options are removed, there is native bridging and audio. Any idea why 'wW' would be causing this? |
| 01:15.27 | joobie | guys i have static queue members setup to point to agents |
| 01:15.40 | joobie | how can i get a SIP user to login and register as an agent so they can accept the calls from that queu? |
| 01:16.18 | jaytee | joobie, what version of *? |
| 01:16.22 | joobie | 1.4 |
| 01:16.42 | jaytee | AgentLogin in the dialplan or the equivalent function from an AGI script |
| 01:17.02 | joobie | thanks jay |
| 01:17.17 | joobie | jay do you have any decent tutorial / docuemntations on setting up dynamic mmebers for the queue? |
| 01:17.20 | jaytee | or AgentCallBackLogin which logs in the member but doesn't require them to stay on the line to wait for calls, the phone will ring when a call is delivered to it. |
| 01:17.24 | joobie | i found a write up but it's a bit too complex for me to grasp |
| 01:17.56 | jaytee | but be advised both of those functions are deprecated in 1.6 in favor of a very (IMHO) nasty, fugly and messy implementation using AEL |
| 01:18.05 | joobie | http://www.voip-info.org/wiki/view/Asterisk+cmd+AgentCallbackLogin says the function is depreciated for 1.4 |
| 01:18.18 | joobie | ahh |
| 01:18.19 | jaytee | it works in 1.4. I've used it |
| 01:18.34 | jaytee | it's deprecated for 1.6 as it won't work there |
| 01:18.35 | joobie | can i use AEL in 1.4 instead to do the same as AgentCallbackLogin ? |
| 01:18.49 | ghento | I tried the 'h' option as well actually with the same no native bridging and no audio. In the Dial() docs I notice: "When options t, T", "h", "H", "w", "W" or "L" (with multiple arguments) are applied, Asterisk will remain in the media path, even if canreinvite=yes'' (a SIP channel option) has been specified. ". I have canreinvite=nonat , would that conflict with this? |
| 01:19.23 | jaytee | joobie, I believe so. there's some documentation for the AEL solution out there on the WIKI. I don't have a link handy at the moment. |
| 01:19.26 | leifmadsen | asterisk will NOT reinvite |
| 01:19.29 | leifmadsen | with those options |
| 01:19.36 | joobie | thanks jay |
| 01:19.37 | leifmadsen | so it doesn't matter what you set it to |
| 01:19.42 | joobie | ill see if i can find it |
| 01:19.57 | leifmadsen | joobie: it's in doc/ of your asterisk source |
| 01:20.05 | thebbxx2000 | so i read through the section of the book about dialplans, and as far as I can see, mine should work |
| 01:20.07 | thebbxx2000 | extensions.conf |
| 01:20.07 | joobie | it's in /home/richardb/asterisk-1.4.22.1/doc/queues-with-callback-members.txt |
| 01:20.08 | thebbxx2000 | ------------------ |
| 01:20.08 | thebbxx2000 | [internal-phones] |
| 01:20.08 | thebbxx2000 | exten => 100,1,Dial(SIP/100) |
| 01:20.08 | thebbxx2000 | exten => 101,1,Dial(SIP/101) |
| 01:20.08 | thebbxx2000 | sip.conf |
| 01:20.10 | thebbxx2000 | --------- |
| 01:20.12 | thebbxx2000 | [100] |
| 01:20.14 | thebbxx2000 | type=friend |
| 01:20.16 | thebbxx2000 | context=internal-phones |
| 01:20.16 | joobie | ok, forget the richardb part:P |
| 01:20.18 | thebbxx2000 | username=100 |
| 01:20.18 | *** kick/#asterisk [thebbxx2000!n=Leif@asterisk/documenteur-extraordinaire/blitzrage] by leifmadsen (use pastebin!) |
| 01:20.27 | jaytee | yay!!! |
| 01:20.28 | thehar | haw |
| 01:20.34 | joobie | lol |
| 01:20.36 | thehar | go leif! |
| 01:20.38 | *** join/#asterisk thebbxx2000 (n=bbxx@c-67-182-169-155.hsd1.ca.comcast.net) |
| 01:20.47 | leifmadsen | thebbxx2000: see ~pb |
| 01:20.49 | leifmadsen | ~pb |
| 01:20.50 | infobot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste , or , http://bin.cakephp.org/ |
| 01:21.01 | jaytee | leifmadsen, after all the trouble you and Jared went to writing the book that newb refuses to take the time to actually read it. |
| 01:21.16 | *** kick/#asterisk [jaytee!n=Leif@asterisk/documenteur-extraordinaire/blitzrage] by leifmadsen (don't be a dick) |
| 01:21.17 | thebbxx2000 | i did read it |
| 01:21.18 | thehar | when will 1.6.x asterisk book be out! |
| 01:21.26 | thehar | jumps up and down in excitement =D |
| 01:21.36 | ghento | leifmadsen: ok thanks. any idea why when i add one of those options in it causes the no-audio problems? |
| 01:21.46 | leifmadsen | no idea |
| 01:21.52 | leifmadsen | sounds like a NAT issue |
| 01:22.04 | *** join/#asterisk ingenius (n=alektro@host57.190-138-60.telecom.net.ar) |
| 01:22.30 | leifmadsen | use 'sip debug' and 'rtp debug' as appropriate to determine how the call is being setup, and where audio is going to (and coming from) |
| 01:22.50 | leifmadsen | thehar: not for a bit, but we've started a little bit on it. I plan on getting more time in the next 2 weeks to actually start on it. |
| 01:23.08 | thehar | leifmadsen: tilghman told me already.. i'm just playing. |
| 01:23.13 | leifmadsen | k |
| 01:23.21 | thehar | but awesome. good to hear. |
| 01:23.30 | thebbxx2000 | ok, sorry about that... i didn't know about pastebin |
| 01:23.30 | thebbxx2000 | http://pastebin.com/d3fa18401 |
| 01:23.56 | leifmadsen | thebbxx2000: you didn't show any output from the console the shows what is not working |
| 01:24.10 | ghento | leifmadsen: cheers |
| 01:24.40 | thebbxx2000 | it says Unable to create channel of type 'SIP' (cause 20- unknown) |
| 01:25.04 | leifmadsen | thebbxx2000: sound like the peer is not registered |
| 01:25.42 | thebbxx2000 | but when i do sip show peers, both phones show up |
| 01:25.47 | thebbxx2000 | as unmonitored:online |
| 01:26.53 | tfrew | does your sip debug show 200 ok or 403 ? |
| 01:27.04 | tfrew | as far as the phone registering |
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| 02:48.43 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
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| 03:27.35 | ambush276 | hey guys i need some help |
| 03:27.39 | ambush276 | im getting this error when running /usr/sbin/asterisk -vvvgc |
| 03:27.48 | ambush276 | http://pastebin.com/m379a09a5 |
| 03:28.17 | ambush276 | any help? |
| 03:28.22 | ambush276 | im on an ubuntu machine |
| 03:28.30 | mmlj4 | only if you give us time to read your paste |
| 03:28.40 | mmlj4 | ubuntu? there's your problem ;-) |
| 03:28.45 | ambush276 | well |
| 03:28.56 | ambush276 | here i had it installed, then my machine crashed for other reasons... |
| 03:29.01 | ambush276 | so i had to reinstall |
| 03:29.09 | drmessano | It crashed because it was Ubuntu |
| 03:29.14 | ambush276 | did it the same way but everythings starts up fine but i get it up ... |
| 03:29.17 | mmlj4 | is asterisk already running, perhaps? |
| 03:29.19 | ambush276 | and then that area |
| 03:29.26 | ambush276 | i dont think so... |
| 03:29.51 | mmlj4 | drmessano: my thoughts exactly |
| 03:30.03 | ambush276 | its not running |
| 03:30.07 | russellb | ambush276: $ ls /usr/lib/asterisk/modules/*voicemail* |
| 03:30.14 | russellb | do you have more than one? |
| 03:30.39 | ambush276 | <PROTECTED> |
| 03:30.41 | ambush276 | /usr/lib/asterisk/modules/app_voicemail_imap.so |
| 03:30.42 | ambush276 | /usr/lib/asterisk/modules/app_voicemail_odbc.so |
| 03:30.43 | ambush276 | /usr/lib/asterisk/modules/app_voicemail.so |
| 03:31.03 | russellb | Ok, so Asterisk is loading more than one of the app_voicemail* variants, and that is your problem. |
| 03:31.15 | russellb | You will need to modify /etc/asterisk/modules.conf to ensure that it only loads one of the 3 |
| 03:31.31 | ambush276 | can i delete the other 2? |
| 03:31.35 | thehar | choose your poison! |
| 03:31.35 | ambush276 | or w/e? |
| 03:31.37 | russellb | if you want ... |
| 03:31.45 | russellb | or just put noload => statements in modules.conf |
| 03:31.46 | ambush276 | so do that for all three of those errors? |
| 03:31.53 | ambush276 | kk |
| 03:31.55 | thehar | russellb: finish writing my job description, please? |
| 03:31.57 | thehar | =) |
| 03:31.59 | mmlj4 | ambush276: don't delete anything |
| 03:32.06 | russellb | thehar: I have no idea what you're talking about, heh |
| 03:32.08 | mmlj4 | use modules.conf, as noted |
| 03:32.10 | eppigy | haha |
| 03:32.10 | ambush276 | ok give me a few mins. |
| 03:32.34 | thehar | russellb: every quarter we must revise our job descriptions.. mine has changed dramatically :( |
| 03:32.44 | russellb | thehar: put that you rock the house |
| 03:33.02 | thehar | haha |
| 03:33.05 | russellb | and that you raise the roof |
| 03:33.06 | thehar | russellb: should i go to astricon this year? |
| 03:33.10 | russellb | of course. |
| 03:33.15 | thehar | i thought about speaking but i'm not sure what about |
| 03:33.27 | russellb | same :-) |
| 03:33.47 | russellb | I'll probably end up doing a generic "Asterisk project update" type of thing |
| 03:34.00 | russellb | what new stuff has happened, what new stuff is planned .. |
| 03:34.08 | *** join/#asterisk [TK]D-Fender (n=joe@64.235.218.194) |
| 03:34.13 | thehar | i was thinking of talking about our unique * pbx @ xm |
| 03:34.23 | russellb | that would be great |
| 03:34.27 | thehar | jsmith knows what i'm talking about.. he wanted a white paper but i have yet to respond to marketing |
| 03:34.42 | russellb | case studies are full of win |
| 03:34.47 | ambush276 | sweeet guys thanks! |
| 03:34.52 | russellb | ambush276: you're welcome |
| 03:34.55 | ambush276 | got it working |
| 03:34.57 | thehar | yes yes they are |
| 03:35.02 | russellb | that will be $9.99 plus taxes and fees |
| 03:35.13 | drmessano | $37.50 rounded up to the first hour |
| 03:35.18 | ambush276 | well i have a few more questions but i will save them for another day.. i have to explore around a little bit more |
| 03:35.19 | thehar | i still amazing amdtech with the app_queue things that are written into my pbx01 asterisk 1.0 cvs code |
| 03:35.31 | thehar | as i'm move along in the migration to 1.4.x |
| 03:36.23 | thehar | also.. russellb we are starting a mirror of downloads.digium.. =) |
| 03:37.38 | ambush276 | well lol i guess i will ask |
| 03:37.40 | ambush276 | .. |
| 03:38.05 | russellb | thehar: orly? |
| 03:38.15 | thehar | russellb: yes |
| 03:38.17 | russellb | thehar: whyfor? |
| 03:38.26 | thehar | russellb: if brad in IT would fix rsync ;) |
| 03:38.39 | thehar | russellb: because we are nice people at XMission. |
| 03:38.41 | ambush276 | ok basically i have an SIP provider that i can recieve incoming calls great!, but i want to make an outgoing extension (like on a 10 digit number). This is on an SIP line, do i need a trunk, or how do i when someone dials lets say extension "9" (and have transfering setup etc..) it goes to an outside line or cell phone etc.. |
| 03:38.52 | thehar | russellb: http://mirrors.xmission.com is where it will be when we get it fully up and going |
| 03:39.54 | russellb | pokes thehar with a priv msg |
| 03:44.44 | ambush276 | nybody? |
| 03:44.48 | leifmadsen | pokes russellb just for fun |
| 03:45.31 | thehar | lulz |
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| 05:22.47 | jplank | anyone play around with the vvx 1500 yet? |
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| 05:47.53 | ambush276 | so im assuming its a now? |
| 05:47.56 | ambush276 | no. |
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| 06:35.16 | KyleK | ambush276: i dont get it, you want to dial in with voip and then dial back in again? |
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| 07:22.22 | nikpakar | hello friends |
| 07:22.33 | nikpakar | im new to asterisk :( |
| 07:23.03 | nikpakar | i want to build a sip -> ss7 gateway using asterisk |
| 07:23.19 | KyleK | lol im new to this too, whats ss7? |
| 07:23.35 | nikpakar | ss7 is signalling system 7 |
| 07:23.38 | nikpakar | on pstn |
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| 07:24.16 | KyleK | oh neat |
| 07:26.22 | nikpakar | :( |
| 07:26.28 | KyleK | so thats totally different from a e1? |
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| 07:28.12 | nikpakar | E1 is the physical connectivity |
| 07:28.23 | nikpakar | there can be many ways of signalling on E1 |
| 07:28.33 | nikpakar | i.e. CAS, CCS |
| 07:28.41 | nikpakar | CAS - Eg.R2 |
| 07:28.54 | nikpakar | CCS - Eg.ISDN Q.931, SS7 ISUP |
| 07:29.21 | KyleK | oh so the hardware is available just not software support? |
| 07:29.44 | nikpakar | pretty much |
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| 08:23.43 | kchehab | good morning All |
| 08:24.14 | kchehab | Why i am having this Status: 420 Bad extension (unsupported) when Client GW contact Asterisk |
| 08:45.28 | sach | good morning |
| 08:45.34 | sach | awefully quiet isn't it? |
| 08:45.36 | *** join/#asterisk ITguru (n=ITGuru@ipguys-adsl.demon.co.uk) |
| 08:45.51 | sach | I don't recognize that particular error. is that from the log? |
| 08:48.23 | iamy_china | I think maybe there were some errors in your dial plan. |
| 10:17.10 | *** join/#asterisk infobot (i=ibot@rikers.org) |
| 10:17.10 | *** topic/#asterisk is Asterisk: The Open Source PBX and Telephony Platform (asterisk.org) -=- Asterisk 1.6.0.9 (2009/04/06), 1.4.24.1 (2009/04/02), *-Addons 1.6.0.1 (2008/12/02), 1.4.7 (2008/06/04), dahdi-linux 2.1.0.4, dahdi-tools 2.1.0.3 (2009/02/03), Libpri 1.4.10 (2009/04/18) -=- Related channels: #asterisknow #asterisk-gui #switchvox #freepbx #asterisk-bugs #asterisk-dev #asterisk-commits |
| 10:20.48 | *** join/#asterisk faltu (i=3ba0e012@gateway/web/ajax/mibbit.com/x-e2aae140332b0c7b) |
| 10:21.23 | faltu | hello |
| 10:23.05 | faltu | need help with dahdi.. |
| 10:23.13 | jblack | ask your mahmi |
| 10:23.38 | faltu | do i have to explicitly call Answer in dialplan befroe calling Dahdi trunk |
| 10:23.58 | jblack | No, you don't need to answer before a Dial() |
| 10:24.02 | jblack | In fact, you shouldn't. |
| 10:24.07 | faltu | yeah |
| 10:24.12 | faltu | i suppose so too |
| 10:24.43 | faltu | But in my case the sip client doesnt change its status to answered from Calling |
| 10:24.52 | faltu | even when i answer the call from other end |
| 10:25.06 | jblack | shrug. |
| 10:28.17 | faltu | asterisk doesnt seem to send back the 180 answered packet |
| 10:30.15 | faltu | though i can hear and chat normally over the channel |
| 10:30.33 | *** join/#asterisk angryuser (n=angryuse@LPuteaux-151-42-35-99.w193-251.abo.wanadoo.fr) |
| 10:30.38 | angryuser | hello |
| 10:31.53 | angryuser | someone with a good experience with sangoma card here ? |
| 10:32.41 | angryuser | i have sometimes my system spammed with messages like this,Apr 20 17:42:18 NOTICE[23361]: channel.c:2200 ast_read: No/unknown event '0' on timer for 'Local/#4030401@neoagent-283d,2'? |
| 10:32.41 | angryuser | Apr 20 17:42:18 NOTICE[23361]: channel.c:2200 ast_read: No/unknown event '0' on timer for 'Local/#4030401@neoagent-b3e5,2'? and i wonder if it is a harwared or software failure ? sangoma 6x T0 bri card |
| 10:33.06 | angryuser | anyone encountered this on his system ? |
| 10:33.17 | angryuser | i am using 3.3.15 version of wanpipe driver |
| 10:33.32 | joobie | guys if i have quque members (dynamic) . how can i remove tem? |
| 10:33.41 | joobie | like from asterisk console can i boot a dynamic member out? |
| 10:40.58 | *** join/#asterisk freh (n=freh@198.0-66-87.adsl-static.isp.belgacom.be) |
| 10:42.44 | KrisWillis | If I run dahdi_monitor and dial my PSTN number and it detects the ringing can I assume that my card is configured correctly to receive incoming calls? I'm having troubles getting Asterisk to answer it... |
| 10:49.30 | joobie | guys anyone decent with queues? |
| 10:49.35 | joobie | i keep getting .. [Apr 22 20:43:45] WARNING[8937]: app_queue.c:4077 queue_exec: Unable to join queue 'customerserv' |
| 10:49.47 | joobie | i can add a member tothe customerserv queue, so i know it's created |
| 10:49.58 | joobie | just when i use queue() to join someone to the queue, it gives me that error |
| 10:49.58 | *** join/#asterisk ccesario (n=ccesario@189-19-9-100.dsl.telesp.net.br) |
| 11:00.11 | freh | joobie, you are sure there is a member in the queue? Do a "queue show <queuename>" on the console |
| 11:06.16 | joobie | freh, Local/1000@agents with penalty 10 (dynamic) (Invalid) has taken no calls yet |
| 11:06.20 | joobie | that's what it says as a member |
| 11:06.26 | joobie | duno what invalid means |
| 11:09.53 | joobie | freh, is there any reason why i should be using LOCAL as extension there to dial? |
| 11:10.00 | joobie | i changed to my SIP extenion and its fine |
| 11:12.57 | freh | joobie, so it's working now? |
| 11:13.09 | joobie | when i specify SIP |
| 11:13.24 | joobie | but i read somewhere that i should be using local |
| 11:13.33 | joobie | i duno why, because they are SIP phones i want to call |
| 11:13.50 | freh | joobie, you happen to have joinempty=strict set in queues.conf? |
| 11:14.15 | joobie | i do |
| 11:14.17 | freh | or leavewhenempty=strict? |
| 11:14.26 | joobie | i have both |
| 11:14.32 | freh | indeed |
| 11:14.45 | freh | You are working with agents? |
| 11:15.02 | *** join/#asterisk nicko_ast (n=nicko_as@124.158.139.170) |
| 11:15.12 | freh | and agent logins? |
| 11:15.19 | joobie | what's the difference between an agent and a queue member? |
| 11:15.32 | joobie | because i use addqueuemember to populate the queue.. but i dont think i define agents |
| 11:15.36 | nicko_ast | hi all |
| 11:15.53 | freh | then you're not using agents |
| 11:15.55 | joobie | i just have like a number sequence users dial on their phones.. then that triggers addqueuemembers |
| 11:16.06 | joobie | what's the diff freh? |
| 11:16.43 | freh | with addqueuemember you can dynamically add members to the queue from the dialplan |
| 11:17.06 | nicko_ast | i have problem, maybe somebody here can help me, how to running AGI script in the middle of conversation ? thanks |
| 11:17.35 | freh | with agents, you can have people log in to a queue as member |
| 11:18.13 | freh | you can also statically define queuemembers in queues.conf |
| 11:18.27 | joobie | ya i first did the static thing |
| 11:18.28 | freh | member => SIP/1000 |
| 11:18.46 | joobie | yaa |
| 11:18.48 | joobie | i did that first freh |
| 11:18.51 | joobie | but see |
| 11:19.18 | joobie | now what ive done is, i dial 999 or 888 on a handset.. depending on which one i dial, it uses addqueuemember and specifies a SIP extension to add to the queue |
| 11:19.32 | freh | I use the same |
| 11:19.33 | joobie | so it's kinda like "people logging themselves into the queue as a member" |
| 11:19.44 | joobie | but what's the difference between that and agents? |
| 11:19.52 | joobie | sounds very much like an agent to me?? |
| 11:20.30 | freh | it's very alike. An agent can log in from any SIP phone. |
| 11:20.47 | freh | that's the difference i believe |
| 11:21.04 | joobie | ahh |
| 11:21.10 | joobie | i mean you can set your dialplan to do the same |
| 11:21.18 | joobie | so an agent can login using the above method from any sip phone |
| 11:21.26 | joobie | freh i did notice agents.conf you can specify a pin |
| 11:21.37 | joobie | maybe it's more for username/password adding to queues |
| 11:21.45 | freh | indeed |
| 11:22.03 | joobie | freh |
| 11:22.12 | joobie | got a few mins for me to brain dump some info from you?:P |
| 11:22.27 | joobie | i have a few things i want to achieve with the queue.. if yuou can sorta point me int he right direction it will save a bit of time for me |
| 11:22.46 | joobie | first time im playingwith the queue system.. really only done basic dialplan stuff before |
| 11:23.03 | freh | I'm just getting to know it myself but I'll help where i can. Just started using queues yesterday |
| 11:23.11 | joobie | ahh cool |
| 11:23.21 | joobie | they are pretty cool when you get your head around em yer? |
| 11:23.33 | freh | kinda |
| 11:23.37 | *** join/#asterisk Victor_Yure (n=victor@unaffiliated/victoryure/x-837844) |
| 11:24.04 | *** join/#asterisk Pazzo (n=ugelt@host156-36-static.14-79-b.business.telecomitalia.it) |
| 11:24.21 | joobie | well one thing im trying to achieve with the dialplan is .. when teh user joins the queue, play an mp3 for like 1 min.. then break and play another mp3.. then ctoninue the mp3 for another min.. then break and play another mp3 |
| 11:24.23 | joobie | and so on |
| 11:24.41 | joobie | sorta like having music playing and a regular interruption with a message saying 'press 1 to leave a voicemail' |
| 11:25.38 | joobie | currently when the user joins the queu, i use background() to play some music.. then queue() to put them in the queue |
| 11:25.56 | joobie | but not sure how to structure so i can have the music play, add them to the queue, and have that interruption message every 1 min |
| 11:26.33 | freh | the regular interruption you can do by setting periodic announce-frequency=60 in queues.conf |
| 11:27.17 | *** part/#asterisk nicko_ast (n=nicko_as@124.158.139.170) |
| 11:27.20 | freh | then also set periodic-announce = <filename> for the sound file |
| 11:27.31 | joobie | ahhh |
| 11:27.57 | joobie | should i be setting the queue hold music with background? |
| 11:28.20 | freh | no |
| 11:29.10 | freh | http://www.orderlyq.com/asteriskqueues.html |
| 11:29.27 | freh | go there for the configuration. Just skip the agents part |
| 11:31.00 | joobie | thanks freh |
| 11:32.08 | freh | joobie, np |
| 11:34.32 | *** join/#asterisk The-Bat (n=karthik@203.199.114.33) |
| 11:37.43 | *** join/#asterisk esaym (n=user@cpe-70-120-89-6.satx.res.rr.com) |
| 11:42.33 | joobie | hey freh .. havent read that doc yet.. but i set periodic-announce-frequency=20 and periodic-announce=tt-weasels in queues.conf and restarted asterisk |
| 11:42.38 | joobie | it doesnt play that file every 20 seconds |
| 11:45.29 | MatBoy | what can cause no sound on a misc destinatino ? |
| 11:55.42 | *** join/#asterisk ITguru (n=ITGuru@ipguys-adsl.demon.co.uk) |
| 11:55.56 | joobie | MatBoy, nat |
| 11:58.41 | *** join/#asterisk guax (n=guaxinim@unaffiliated/guaxinim) |
| 12:13.35 | MatBoy | joobie: true, but NAT is OK |
| 12:13.44 | *** join/#asterisk jer (n=jtregunn@unaffiliated/jer) |
| 12:14.57 | *** join/#asterisk [TK]D-Fender (n=joe@216.191.106.163) |
| 12:17.56 | *** join/#asterisk degrade (n=degrade@unaffiliated/degrade) |
| 12:18.05 | degrade | O_o |
| 12:18.37 | MatBoy | woei |
| 12:18.46 | MatBoy | yes annoying.. ports are open, so that is ok |
| 12:18.59 | MatBoy | allow ia ulaw, alaw and gsm |
| 12:19.02 | MatBoy | so should be ok |
| 12:19.22 | beek | Morning [TK]D-Fender |
| 12:19.38 | [TK]D-Fender | *yawn* |
| 12:21.48 | *** join/#asterisk blackest_mamba (n=blackest@71.239.160.143) |
| 12:22.11 | creativx | hi [TK]D-Fender |
| 12:22.24 | *** join/#asterisk zaafouri (n=zaafouri@196.203.51.238) |
| 12:23.12 | *** join/#asterisk trif_ (n=trif@zoro.loria.fr) |
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| 12:23.25 | MatBoy | [TK]D-Fender: that was in my neck... tissue please |
| 12:25.29 | trif_ | hello all, i would like to use cdr with sqlite. But in the 'make menuselect' if i want to add cdr_sqlite, it says that i'm missing the dependancy "sqlite(E)" |
| 12:26.21 | trif_ | where is that depandency ? |
| 12:26.40 | trif_ | *dependency sorry |
| 12:26.53 | *** join/#asterisk Victor_Yure (n=victor@unaffiliated/victoryure/x-837844) |
| 12:27.38 | *** join/#asterisk tiav (n=tiav@91.197.165.222) |
| 12:27.48 | [TK]D-Fender | trif_: Something like "libsqlite" |
| 12:28.18 | trif_ | it's in the menuselect ? |
| 12:29.02 | trif_ | i don't think so |
| 12:29.07 | trif_ | i've just done |
| 12:29.08 | trif_ | ./configure | grep sqlite |
| 12:29.08 | trif_ | checking for sqlite_exec in -lsqlite... no |
| 12:29.08 | trif_ | checking for sqlite3_open in -lsqlite3... no |
| 12:29.26 | tzafrir_laptop | trif_, what distro do you have? |
| 12:29.26 | kchehab | Why i am having this Status: 420 Bad extension (unsupported) when Client GW contact Asterisk |
| 12:29.33 | trif_ | ubuntu |
| 12:30.06 | tzafrir_laptop | -lfooN is normally libfooN-dev or libfoo-dev |
| 12:31.17 | tzafrir_laptop | sqlite (2) and sqlite3 are different . sqlite2 is libsqlite0.so, actually |
| 12:31.44 | trif_ | ok |
| 12:31.54 | trif_ | but at which step can i include |
| 12:31.57 | *** join/#asterisk Tar-Get (n=Tar-Get@83.101.36.29) |
| 12:31.59 | trif_ | that lib |
| 12:32.17 | MatBoy | weird, no sound at all.. nat=yes is set |
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| 12:34.30 | [TK]D-Fender | MatBoy: takes a lot more... read the guide |
| 12:34.31 | [TK]D-Fender | ~sipnat |
| 12:34.32 | infobot | [~sipnat] Quick guide on configuring * + SIP behind NAT : http://www.aocomputing.net/?p=3 , otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
| 12:35.31 | MatBoy | [TK]D-Fender: true, ulaw, alaw so on... but it worked before |
| 12:35.33 | MatBoy | let me read |
| 12:36.59 | *** join/#asterisk gego (n=rick@b238085.customer.hansenet.de) |
| 12:40.06 | *** join/#asterisk esaym (n=user@cpe-70-120-89-6.satx.res.rr.com) |
| 12:41.00 | *** join/#asterisk maximo (n=maximo@CPE001217b1920e-CM00111ade9528.cpe.net.cable.rogers.com) |
| 12:42.12 | *** join/#asterisk DavidR2008 (n=chatzill@fw1.safedataisp.net) |
| 12:44.40 | MatBoy | [TK]D-Fender: can there be something else wrong ? type=peer so that should be ok |
| 12:44.56 | *** join/#asterisk esaym (n=user@cpe-70-120-89-6.satx.res.rr.com) |
| 12:45.02 | *** join/#asterisk coppice (n=chatzill@46.166.17.210.dyn.pacific.net.hk) |
| 12:45.06 | [TK]D-Fender | MatBoy: I'm not going to comment on tiny single parms when I don't see the big picture. |
| 12:45.23 | *** part/#asterisk Tar-Get (n=Tar-Get@83.101.36.29) |
| 12:46.01 | *** join/#asterisk Wombles (n=wombles@209.40.204.23) |
| 12:46.52 | The-Bat | Hie |
| 12:46.55 | The-Bat | How can i take an input(numeric) from the caller and store it in a variable |
| 12:47.00 | MatBoy | [TK]D-Fender: yes I understand that |
| 12:47.15 | [TK]D-Fender | The-Bat: "core show application read" |
| 12:47.51 | DavidR2008 | [TK]D-Fender: I can't find SIP_PEER. core show application SIP_PEER returns "Your application(s) is (are) not registered" so I looked in the apps directory in source and I don't see a module for it. Is there something I need to do to enable it? * 1.4.23.1 |
| 12:48.09 | [TK]D-Fender | DavidR2008: function, not application |
| 12:48.18 | DavidR2008 | ah |
| 12:49.14 | The-Bat | [TK]D-Fender, thanks |
| 12:50.06 | *** join/#asterisk esaym (n=user@cpe-70-120-89-6.satx.res.rr.com) |
| 12:50.46 | *** join/#asterisk bbkt-trix (n=bbkt-tri@unaffiliated/bbkt-trix) |
| 12:51.40 | DavidR2008 | I assume this would be correct then? asteriskpbx0*CLI> core show function SIP_PEER if so, I get this: No function by that name registered. |
| 12:52.55 | [TK]D-Fender | DavidR2008: no "_" |
| 12:54.03 | DavidR2008 | ok, I give up. I know you said core show .... SIP_PEER yesterday, but I forgot what the ... was. Would you share again please |
| 12:55.34 | trif_ | tzafrir_laptop : thank you for pointing me out that it was just an apt-get issue :) |
| 12:55.36 | *** join/#asterisk Tar-Get (n=Tar-Get@83.101.36.29) |
| 12:56.44 | *** join/#asterisk ingenius (n=alektro@69.90.72.173) |
| 12:56.56 | *** join/#asterisk leifmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 12:56.56 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 12:56.58 | *** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 12:58.37 | DavidR2008 | I did a a core show functions and found a SIPPEER is that the same thing? |
| 12:58.53 | [TK]D-Fender | [08:52]<[TK]D-Fender>DavidR2008: no "_" <-------- |
| 12:59.53 | leifmadsen | pokes Katty |
| 12:59.57 | DavidR2008 | oh, duh. I that was some sort of emoticon |
| 13:00.57 | *** join/#asterisk cfh (n=luca@host196-137-static.90-82-b.business.telecomitalia.it) |
| 13:01.32 | DavidR2008 | thx! |
| 13:03.10 | cfh | hi all, is possibile monitor with BLF hint + devstate function when DND is on/off of a phone (Polycom) ? |
| 13:03.19 | *** join/#asterisk pagec (n=pagec@96.57.210.34) |
| 13:04.39 | [TK]D-Fender | cfh: Poylcom's DND is not trackable. You can use BLF as an indicator to the phone of a dialplan-based DND however |
| 13:04.43 | Katty | squeaks at leifmadsen |
| 13:04.57 | Katty | is already on the phone with bandwidth this morning :< |
| 13:05.57 | cfh | D-Fender ok, i have tried to crate a custom extension to mantain the phone busy |
| 13:06.24 | leifmadsen | Katty: oh nice |
| 13:06.31 | Katty | :> |
| 13:06.41 | Katty | PROBLEM SOLVED! |
| 13:06.49 | Katty | record speed! 6minutes! |
| 13:07.25 | pagec | Using Asterisk 1.6 with no diguim cards, what drivers do I need to load for timing (I know on 1.4 I needed to load a usb driver, but not sure about 1.6) |
| 13:07.26 | Katty | leifmadsen: whyfordidyoupokesir |
| 13:08.30 | leifmadsen | Katty: because I'm that kinda guy |
| 13:08.49 | [TK]D-Fender | pagec: No, don't need USB for 1.4 (thats only the oldest Kernel 2.4 rerquirement). You need dahdi_dummy for 1.6 |
| 13:08.51 | leifmadsen | pagec: just install DAHDI as it'll install the DAHDI-dummy driver |
| 13:09.07 | leifmadsen | same thing for 1.4 |
| 13:09.08 | Katty | leifmadsen: mmmkay |
| 13:09.18 | *** join/#asterisk tcseke (n=chatzill@217.20.134.239) |
| 13:09.38 | leifmadsen | ok, and now I'm gonna head to the grocery store as was my plan 10 mins ago |
| 13:10.15 | Katty | eppigy: riddick tried to get me out of bed this morning, and succeeded, i got the whole thing on my phone. |
| 13:10.53 | cfh | D-Fender : if i create a custom ext the watcher cant recognize if the phone is on DND or is in call comunication are there some workaround to do this ? |
| 13:11.08 | eppigy | Katty: haha SWEET |
| 13:11.21 | Katty | eppigy: it's uploading to youtube |
| 13:11.24 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 13:11.25 | Katty | eppigy: RIGHT NOW |
| 13:11.28 | eppigy | :D |
| 13:11.33 | eppigy | INSTANT GRATIFICATION |
| 13:12.58 | Katty | it's processing |
| 13:13.10 | Katty | http://www.youtube.com/watch?v=N2d_QN16BLE |
| 13:13.59 | eppigy | haha |
| 13:14.04 | eppigy | im training some dude in here |
| 13:14.12 | Katty | HI DUDE |
| 13:14.12 | eppigy | i am not sure what the opening audio sounds like |
| 13:14.26 | eppigy | well he is a cubicle over |
| 13:14.28 | Katty | there's some doggy noises. |
| 13:14.30 | eppigy | reading a manual |
| 13:14.33 | Katty | and then giggling |
| 13:14.36 | eppigy | hahaha |
| 13:14.37 | eppigy | great |
| 13:14.38 | Katty | and then me complaining about having to get up |
| 13:15.07 | eppigy | i didnt think it was done processing so i didnt bother to mute it |
| 13:15.07 | [TK]D-Fender | cfh: I jsut told you what you can do with this. Don't make me repeat myself |
| 13:15.13 | eppigy | and my ipod is cranked |
| 13:15.19 | Katty | oh noes! |
| 13:15.35 | eppigy | he already knows im crazy |
| 13:15.45 | eppigy | the other day the head of accounting |
| 13:15.49 | eppigy | my dircet boss |
| 13:15.52 | *** join/#asterisk jaytee (n=jforde05@unaffiliated/jaytee) |
| 13:15.53 | eppigy | and our dba |
| 13:16.00 | Katty | jaytee: http://www.youtube.com/watch?v=N2d_QN16BLE |
| 13:16.04 | eppigy | were talking and I came in to introduce the new dude |
| 13:16.17 | eppigy | and theyre like oh i know you from blah blah, small world |
| 13:16.29 | eppigy | and Im like yeah man I;ve got these nude photos floating around |
| 13:16.34 | eppigy | that just wont go away |
| 13:16.51 | eppigy | the tension was palpable |
| 13:16.58 | Katty | :> |
| 13:17.14 | eppigy | haha |
| 13:17.26 | eppigy | riddick is the man |
| 13:18.00 | jaytee | I'd be chewing on her leg and blankets too if I had a full bladder and wanted out. |
| 13:18.09 | Katty | lol |
| 13:18.12 | eppigy | man |
| 13:19.21 | jaytee | and that's not to say he doesn't love you and want attention just that it's morning |
| 13:19.32 | creativx | search of the day.. http://similar-images.googlelabs.com/images?q=jenna+haze&prev=%2Fimages%3Fq%3Djenna%2Bhaze&qtype=similar&tprev=%2Fimages%3Fq%3Djenna%2Bhaze&tbnid=JY4dr-qmlghFjM&start=21&ndsp=21 |
| 13:19.40 | creativx | SFW |
| 13:21.04 | Katty | Jenna Haze? |
| 13:21.39 | jaytee | ok, so we've got multiple perv photos montage with misc cleavage and girl on girl shots interspersed and a doctor looking at the back of a man's head. It all makes perfectly twisted sense except that last one. |
| 13:22.35 | Katty | weird. |
| 13:23.04 | jaytee | my thoughts exactly along with, "and this relates to Asterisk how?" |
| 13:23.20 | creativx | i tought you needed a break |
| 13:23.21 | creativx | ;) |
| 13:23.48 | eppigy | jenna haze |
| 13:23.50 | eppigy | is kinda cute |
| 13:23.57 | eppigy | but good god someone shut her up |
| 13:23.58 | Katty | i think stacie orrico is cuter |
| 13:24.51 | eppigy | OFENDED QUIT |
| 13:25.56 | Katty | http://www.mp3parati.com.ar/images/S/stacie-orrico-f.jpg <- how can a face like that offend you. |
| 13:26.21 | eppigy | no it offende awk_r |
| 13:26.26 | eppigy | and he rage quit |
| 13:26.49 | Katty | ohisee :< |
| 13:26.53 | eppigy | but yeah id hit that |
| 13:26.56 | eppigy | 4 certain |
| 13:27.05 | mmlj4 | aww, da cute widdle marilyn spot |
| 13:27.18 | eppigy | MALIGNANT |
| 13:27.23 | mmlj4 | eppigy++ |
| 13:27.55 | Katty | she sings nicely, too. |
| 13:28.57 | eppigy | i want a girl that plays the violin |
| 13:28.59 | eppigy | that is all |
| 13:29.41 | Katty | wai |
| 13:30.08 | eppigy | it is awesome |
| 13:30.31 | *** join/#asterisk tfrew (n=tfrew@office.neteasyinc.com) |
| 13:31.15 | *** join/#asterisk awk_r (n=awk_r@nat/digium/x-276b8de90ea8cc6b) |
| 13:34.59 | Katty | eppigy: three little birds, sat on your window and they told you you don't need to worry. |
| 13:35.06 | Katty | eppigy: summer came like cinnamon, so sweet |
| 13:35.10 | *** join/#asterisk anthm (n=anthm@freeswitch/developer/anthm) |
| 13:35.23 | MatBoy | I wonder if something goes wrong here with no sound Native bridging SIP/ |
| 13:35.43 | Katty | rtp immediately comes to mind. |
| 13:35.53 | Katty | which may or may not apply. |
| 13:36.37 | eppigy | i like the summer sun ^_____^ |
| 13:36.49 | *** join/#asterisk blkry (n=blkry@64.147.222.130) |
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| 13:42.17 | *** join/#asterisk [intra]lanman (n=lanman@freeswitch/developer/intralanman) |
| 13:42.31 | Katty | eppigy: i made tacos last night. they were super. what did you have |
| 13:43.09 | *** join/#asterisk mikkel (n=mikkel@84-238-113-66.u.parknet.dk) |
| 13:45.24 | Katty | today is picnic material |
| 13:45.46 | coppice | russellb: is there anybody there? |
| 13:45.54 | eppigy | I ate at piccadilly |
| 13:46.02 | eppigy | i had monterey chicken |
| 13:46.03 | russellb | coppice: here, yep |
| 13:46.03 | eppigy | :D |
| 13:46.19 | Katty | monterey chicken? |
| 13:46.28 | Katty | tell me it's more than a chicken breast with some cheese on top |
| 13:46.30 | Katty | DETAILS |
| 13:46.41 | coppice | russellb: try http://www.soft-switch.org/downloads/snapshots/spandsp/spandsp-20090422.tar.gz |
| 13:46.56 | eppigy | well |
| 13:47.02 | eppigy | it had bacon too |
| 13:47.03 | russellb | coppice: will do! |
| 13:47.04 | eppigy | and it was breaded |
| 13:47.05 | *** part/#asterisk Tar-Get (n=Tar-Get@83.101.36.29) |
| 13:47.23 | coppice | russellb: there is something odd looking and sounding about your alaw prompts, but it looks like the G.722 codec was getting things 6dB out |
| 13:48.31 | *** join/#asterisk AsteriskDom (n=Asterisk@75.112.88.200.m.sta.codetel.net.do) |
| 13:48.36 | russellb | Hm, I wonder what could be odd. The one different thing that was done recently was that someone suggested that we run our prompts through 50 Hz high pass filter before converting them into the various other codecs. |
| 13:48.59 | russellb | As it makes them sound better when encoded into low bitrate codecs, supposedly |
| 13:49.18 | russellb | Anyway, I'll work on testing your update now |
| 13:49.21 | russellb | thank you! |
| 13:50.47 | Katty | eppigy: post gifs? |
| 13:51.24 | *** join/#asterisk putnopvut (n=putnopvu@asterisk/master-of-queues/mmichelson) |
| 13:51.24 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 13:52.22 | [TK]D-Fender | Katty: http://www.recipezaar.com/Chilis-Monterey-Chicken-59633 |
| 13:52.45 | [TK]D-Fender | Katty: http://allrecipes.com/Recipe/Monterey-Chicken/Detail.aspx |
| 13:52.56 | coppice | russellb: cleaning off the bass like Jean-Marc suggested is good for mitigating the lossiness of alaw/ulaw and helps avoid confusing echo cancellers. however, when I downsample and filter your sln16 files and converted them to alaw and back to linear they look balanced around zero. try looking at yours in an audio editor. the positive side is much spikier than the negative |
| 13:53.01 | [TK]D-Fender | Katty: not just cheese... BACON! |
| 13:53.57 | russellb | coppice: interesting ... will take a look |
| 13:54.09 | Katty | hmm |
| 13:54.28 | Katty | [TK]D-Fender: i'm oddly not impressed. |
| 13:54.38 | Katty | :< |
| 13:54.40 | [TK]D-Fender | Katty: myocardial infarction risk detected... |
| 13:54.49 | Katty | :P |
| 13:54.57 | Katty | [TK]D-Fender: did you see the vid of riddick this morning? |
| 13:55.25 | [TK]D-Fender | Katty: Nope, but I did see yesterdays pic of him in a t-shirt... poor dog... |
| 13:55.33 | Katty | lol |
| 13:56.34 | Katty | [TK]D-Fender: http://www.youtube.com/watch?v=N2d_QN16BLE |
| 13:57.51 | eppigy | Katty: haha I didnt take any pictures of it at piccadilly before i ate it |
| 13:58.10 | Katty | eppigy: WHAT?! |
| 13:58.13 | Katty | eppigy: ;> |
| 13:58.19 | eppigy | D: |
| 14:00.30 | dni | hello all,. im having a weird issue/error,. i created a simple IVR, but when i hit '1 for support' the call gets dropped with a error msg saying "unable to aprse message from, etc.. unsupported media type" the error message is on line 135 of this http://pastebin.com/m7d20fb1d ,.. anyone have any idea as to why im getting this,. everything should be ulaw |
| 14:00.36 | [TK]D-Fender | Katty: Should ahve seen what my sisters servals did... they'd leap up like 6' and bore down with both front paws close together and wind her :) |
| 14:01.05 | Katty | [TK]D-Fender: aww lol |
| 14:02.30 | [TK]D-Fender | tomorrow = Happy Ubuntu 9.04 day :) |
| 14:03.44 | pagec | Asterisk 1.6 commandline `dahdi show status` shows DAHDI_DUMMY/1 as unconfigured, is that ok? I am thinking it is because there isn't anything to configure as far as i can tell, but wanted to make sure. |
| 14:06.42 | *** part/#asterisk cfh (n=luca@host196-137-static.90-82-b.business.telecomitalia.it) |
| 14:08.08 | *** join/#asterisk apeiron (n=Chris@32.164.134.103) |
| 14:09.47 | *** join/#asterisk ingenius (n=alektro@69.90.72.173) |
| 14:09.53 | [TK]D-Fender | pagec: Probably fine. go test it |
| 14:11.56 | *** join/#asterisk qdk (n=qdk@81.7.168.130) |
| 14:13.35 | *** join/#asterisk wonderworld (n=ww@ip-62-143-16-28.unitymediagroup.de) |
| 14:17.57 | freh | Anyone knows if it's possible to configure DAHDI for sending and receiving SMS? |
| 14:18.33 | MatBoy | mhh, to weird... calls are being forwarded but no sound... but the fax... I can hear that one when I call it |
| 14:18.35 | Juggie | anyone have any recomendations for any asterisk cdr stats packages besides asterisk-stat? |
| 14:20.09 | MatBoy | so I wonder what this can be... [TK]D-Fender when you hear a fax tone.. is this sound too ? |
| 14:22.51 | [TK]D-Fender | MatBoy: Forwarded? Fax? HUH!? |
| 14:23.16 | [TK]D-Fender | MatBoy: First we're talking NAT, what are you going on about now? You aren't making any sense |
| 14:24.28 | JerJer | ok what does one need to install on Fedora Core 7 to use 'make menuselect' ? |
| 14:24.43 | JerJer | we have every ncurses package we can find installed - still tells us to install ncurses |
| 14:24.47 | eppigy | ncrses-devel |
| 14:24.52 | eppigy | ncurses-devel |
| 14:24.53 | JerJer | its there |
| 14:24.57 | eppigy | no its not |
| 14:25.17 | eppigy | yum install ncurses ncurses-devel |
| 14:25.22 | rob0 | I ate it. |
| 14:25.59 | JerJer | Package ncurses - 5.6-9.20070812.fc7.i386 is already installed. |
| 14:26.00 | JerJer | Package ncurses-devel - 5.6-9.20070812.fc7.i386 is already installed. |
| 14:26.01 | JerJer | eat that |
| 14:26.18 | JerJer | re-ran configure |
| 14:26.26 | JerJer | it even says it finds ncurses |
| 14:26.30 | eppigy | wow bro |
| 14:26.52 | Corydon76-dig | JerJer: try libncurses-devel |
| 14:26.57 | eppigy | good call |
| 14:27.39 | JerJer | tried that too: No package libncurses-devel available. |
| 14:27.59 | eppigy | yeha i think libncurses is apt |
| 14:28.12 | JerJer | + 5-dev on apt |
| 14:28.13 | Corydon76-dig | Did it install libslang-devel at the same time? |
| 14:28.25 | *** join/#asterisk jdnwest (n=jdnwest4@mx1.westparkcom.net) |
| 14:28.42 | JerJer | No package libslang-devel available. |
| 14:28.43 | Corydon76-dig | JerJer: ah, I suspect I know what it is: make -C menuselect clean |
| 14:28.53 | Corydon76-dig | Then try again |
| 14:29.07 | Corydon76-dig | or make -C menuselect distclen |
| 14:29.09 | *** join/#asterisk KavanS (n=KavanS@static-71-117-242-28.ptldor.dsl-w.verizon.net) |
| 14:29.14 | Corydon76-dig | s/clen/clean/ |
| 14:29.25 | rob0 | eats more ncurses |
| 14:29.33 | JerJer | then ./configure |
| 14:29.36 | JerJer | trying now |
| 14:29.42 | Corydon76-dig | Nope, no need to rerun configure |
| 14:29.53 | Corydon76-dig | The issue is one of two configure scripts |
| 14:29.54 | JerJer | ok doing it anyways :) |
| 14:29.59 | eppigy | haha |
| 14:30.23 | JerJer | works now - thank you |
| 14:30.34 | eppigy | NICE |
| 14:30.36 | Corydon76-dig | You're welcome |
| 14:31.04 | russellb | coppice: I have tested the changes and it does seem better. I'm going to let some other folks here play with it and see what they think. I'll get back to you with an email with the results. Thanks again! |
| 14:32.22 | jdnwest | I'm having weird issues. My * server is in a colo with a public IP and I have 2 phones behind different NAT'D networks. The phones can call each other fine, but I can not get a DID from vitelity to ring to the phones (It can hit the * just fine and go to voicemail though) |
| 14:32.49 | [TK]D-Fender | jdnwest: yes it matters if your phones are behind NAT's of their own. |
| 14:32.55 | [TK]D-Fender | ~sipnat |
| 14:32.56 | infobot | [~sipnat] Quick guide on configuring * + SIP behind NAT : http://www.aocomputing.net/?p=3 , otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
| 14:32.57 | [TK]D-Fender | ^^^^^^^^ |
| 14:33.41 | coppice | russellb: I've never actually had another G.722 implementation here to test against |
| 14:34.06 | [TK]D-Fender | jdnwest: reinvites will fail hard, and if you don't have a NAT keep-alive the port will close up and phones will become uncontactible |
| 14:34.13 | russellb | coppice: I wish I could help you out by getting you something ... but I'm not sure how I'd do that, heh |
| 14:34.15 | Katty | omgpolycomsarecomingoutofthewoodwork |
| 14:34.25 | Katty | i found 4 in my office! |
| 14:34.30 | [TK]D-Fender | Katty: omgwtfbbq! |
| 14:34.31 | *** join/#asterisk defsmac (n=andy@213.128.237.170) |
| 14:34.49 | *** join/#asterisk astassistant (n=skip@65-126-63-1.dia.static.qwest.net) |
| 14:34.49 | rob0 | Poly want a com! |
| 14:34.51 | Katty | omgyes! |
| 14:34.53 | defsmac | anyone able to help me configure a polycom soundstation ? |
| 14:34.59 | *** join/#asterisk Wulfy814 (n=AskMe@dynamic-acs-24-154-119-148.zoominternet.net) |
| 14:35.12 | *** join/#asterisk iratik (n=root@74-84-99-12.client.mchsi.com) |
| 14:35.23 | Wulfy814 | I had to recompile after upgrading from Ubuntu 6.06 LTS to 8.04 LTS and now all of the system prompts are scratchy, but working |
| 14:35.26 | Wulfy814 | what did I do wrong? |
| 14:35.47 | [TK]D-Fender | Wulfy814: ... |
| 14:35.48 | Wulfy814 | running asterisk 1.4.18 |
| 14:35.49 | [TK]D-Fender | ~gsmbug |
| 14:35.50 | infobot | [~gsmbug] there is a bug compiling Asterisk with GCC 4.2 where optimization errors cause GSM transcoding to distort heavily. Compile with GCC 4.1 or lower, or follow this patch : http://bugs.digium.com/view.php?id=11243 Also please read : http://forums.digium.com/viewtopic.php?p=116731&sid=3c65e49ec840380ed20f1c8426382b39 |
| 14:35.51 | [TK]D-Fender | ^^^^^ |
| 14:36.02 | Wulfy814 | great |
| 14:36.42 | jdnwest | Fender: I have no problem calling from extension to extension (thats passing through the NAT as each extension is in a different office) |
| 14:36.54 | jdnwest | I figured that would be harder than passing a call from the server to the phone |
| 14:36.59 | *** join/#asterisk The_Boy_Wonder (n=davidvos@asterisk/batman-developer/dvossel) |
| 14:37.16 | KrisWillis | If I run dahdi_monitor and dial my PSTN number and it detects the ringing can I assume that my card is configured correctly to receive incoming calls? I'm having troubles getting Asterisk to answer it... |
| 14:38.09 | [TK]D-Fender | jdnwest: You'd have to actually LOOK at the SIP debug of a call attempt and check *'s status on the peer |
| 14:38.51 | [TK]D-Fender | KrisWillis: Just because the device is configured doesn't mean your dialplan is. |
| 14:39.07 | [TK]D-Fender | KrisWillis: jsut like the fact I put a functioning phone at my CSR's desk doesn't mean they're going to get off their ass to answer it |
| 14:39.26 | [TK]D-Fender | KrisWillis: Go look at the CLI output of your inbound call attempt and see where it's looking. |
| 14:39.39 | coppice | [TK]D-Fender: isn't that GSM bug thing just ancient history? |
| 14:39.54 | [TK]D-Fender | coppice: So is Ubuntu 8.04 :) |
| 14:40.06 | [TK]D-Fender | coppice: and, its within this past year..... just like 8.0.4 :) |
| 14:40.28 | [TK]D-Fender | coppice: and the fact he's still on 1.4, and an older version at that. |
| 14:40.43 | KrisWillis | [TK]D-Fender: Understood, I'm just trying to pinpoint where my problem is, in my dial plan, my dahdi config or my card. I was hoping to rule out the card and dahdi config having dahdi_monitor see the line ringing - I'll check the CLI though, thanks, do I need to run anything on the CLI or just log into it and dial my number? |
| 14:41.12 | [TK]D-Fender | KrisWillis: versbose 10, sit and watch |
| 14:41.23 | KrisWillis | [TK]D-Fender: Thanks, I'll give it a go |
| 14:41.39 | [TK]D-Fender | KrisWillis: Also assumes that you compiled * AFTER DAHDI and that * has the compiled channel driver loaded |
| 14:43.08 | *** join/#asterisk moy (n=chatzill@74.12.124.89) |
| 14:43.32 | *** join/#asterisk mweichert (n=mweicher@216.13.154.21) |
| 14:43.39 | mweichert | hello |
| 14:43.59 | mweichert | in my dialplan, how do I determine if a variable is defined or is null? |
| 14:44.25 | KrisWillis | [TK]D-Fender: Hmmm, nothing appearing in the CLI - How can I confirm that my channel driver is loaded? |
| 14:44.41 | iratik | What should i be looking at as far as figuring why i might be getting alot of dropped calls? Here is what i have so far, there is a recording, extensions.conf and asterisk log output. http://pastie.org/454562 |
| 14:44.51 | [TK]D-Fender | mweichert: "core show function ISNULL" |
| 14:45.13 | [TK]D-Fender | mweichert: Or just compare it in an expression. |
| 14:45.14 | mweichert | [TK]D-Fender, thanks again! ;) |
| 14:45.27 | *** join/#asterisk martyn-dev (i=be18869a@gateway/web/ajax/mibbit.com/x-20ea848adc70179f) |
| 14:45.32 | martyn-dev | hi |
| 14:45.34 | martyn-dev | :D |
| 14:45.48 | [TK]D-Fender | KrisWillis: "dahdi show channels" if that fails completely : "module load chan_dahdi.so" |
| 14:46.14 | martyn-dev | I have a cisco 7940 and i need that work with asterisk across a nat .. i dont know why not register with asterisk :( |
| 14:46.38 | martyn-dev | do you know some special settings to this phone to work with asterisk when i have a nat environment |
| 14:46.44 | mweichert | [TK]D-Fender, so if I'm testing if a variable is defined or not, should I use EXISTS instead of IFNULL? |
| 14:47.07 | [TK]D-Fender | iratik: Your dialplan clearly doesn't match your CLI |
| 14:47.13 | [TK]D-Fender | iratik: No extens match |
| 14:47.40 | KrisWillis | [TK]D-Fender: Failed to load the module, no such file. Should I re-install dahdi, then re-install asterisk in that order, I have since re-installed dahdi after my original installation of asterisk |
| 14:47.44 | iratik | the 91NXXNXXXXXX matches, you must be referring to the 8600061 right? |
| 14:47.52 | [TK]D-Fender | mweichert: no. |
| 14:47.56 | iratik | [TK]D-Fender:^^ |
| 14:48.11 | iratik | err... 8600058 |
| 14:48.11 | *** join/#asterisk aryonoco (n=chatzill@123-243-106-166.static.tpgi.com.au) |
| 14:48.16 | [TK]D-Fender | KrisWillis: no such file means you did not compile * AFTER you did DAHDI and the channel driver wasn't built |
| 14:48.39 | [TK]D-Fender | iratik: Clearly doesn't match. trust factor rapidly dropping towards 0... |
| 14:48.52 | KrisWillis | I see, so a recompile of asterisk is in order then... Will give it a go |
| 14:49.13 | aryonoco | hi everyone |
| 14:49.22 | [TK]D-Fender | mweichert: And there is no normal way to see if a variable is defined because they are used on the fly |
| 14:49.25 | iratik | [TK]D-Fender: you re right... its not matching... thats so obvious to me now |
| 14:49.49 | aryonoco | I'm having an issue using asterisk-gui 2.0 I checked out the version today, it is revision 4757. |
| 14:49.52 | iratik | [TK]D-Fender: Thank you, i will look into that and see if that path leads me closer to a resolution... thank you |
| 14:50.15 | [TK]D-Fender | iratik: Hope you find whatever it is... because right now we certainly can't :) |
| 14:50.29 | aryonoco | I've setup the http.conf and manager.conf files as described in the howto |
| 14:50.42 | [TK]D-Fender | aryonoco: GUI's are not supported here. go ask in #asterisk-gui |
| 14:50.49 | aryonoco | ok, thanks |
| 14:50.52 | *** part/#asterisk defsmac (n=andy@213.128.237.170) |
| 14:50.55 | *** join/#asterisk juanjoc (n=jcomella@200.69.219.113) |
| 14:50.56 | *** part/#asterisk aryonoco (n=chatzill@123-243-106-166.static.tpgi.com.au) |
| 14:55.20 | *** join/#asterisk haib (n=haib@207.157.76.1) |
| 14:57.02 | haib | Can you propagate smdi over serial with asterisk, or can it only receive? Looking at the smdi.conf in 1.4 it seems like it can only receive, it can't send smdi over serial, correct? |
| 14:59.24 | mike345 | can any callerid info be passed on non-PRI T1 that is using e&m signalling? or is callerid PRI only? |
| 14:59.46 | *** join/#asterisk mort_gib (n=mjensen@16.Red-83-36-63.staticIP.rima-tde.net) |
| 15:02.05 | mike345 | I want to log calls coming from my NEC PBX -> Asterisk -> Telco Long Distance T1 but Asterisk only sees the destination #. Is there any way to see the origination #? |
| 15:05.24 | *** join/#asterisk proxium (n=bmr@196.203.51.238) |
| 15:05.47 | *** join/#asterisk bl4__ (n=kim@64.0.29.254.ptr.us.xo.net) |
| 15:06.47 | *** join/#asterisk e` (n=IceChat7@38.102.196.202) |
| 15:07.22 | proxium | Hello everybody, can I have a separate DialGUI as "vicidial" and Asterisk in separate server? |
| 15:07.57 | e` | Hi all, can anyone take a look at my thread? I'm having trouble installing chan_unitstum in 1.4.2 http://forums.digium.com/viewtopic.php?p=128985#128985 |
| 15:10.46 | [TK]D-Fender | proxium: what does VICIDIAL say? |
| 15:10.56 | *** part/#asterisk gego (n=rick@b238085.customer.hansenet.de) |
| 15:11.50 | thedonvaughn | proxium: come again? You mean you want the web server for vicidial not on the same server as asterisk? of course. I recommend having your web server for vici on one server, DB on another, and astersk on another. |
| 15:11.54 | thedonvaughn | if that's what you're asking |
| 15:12.57 | proxium | thedonvaughn: yes of course that's what I need to do! |
| 15:14.14 | *** join/#asterisk n3hxs (n=HAMming@63.68.135.4) |
| 15:14.38 | iratik | lol |
| 15:15.07 | thedonvaughn | proxium: yes you can do that. :) Simple just copy the web files to your webserver and it all should work. You do need the perl scripts tho running on the asterisk box i believe |
| 15:15.12 | proxium | but I have a big problem: the Vicidial server did not communicate with my Asterisk server, I didn't see any registration phone on CLI or any opened socket between Vici & Asterisk |
| 15:15.37 | thedonvaughn | proxium: for detailed vicidialer problems, i'd suggest going to vicidial's forums. great help for me there when i first started with vici |
| 15:15.49 | thedonvaughn | usually mflorrell will answer himself (author of vici) |
| 15:16.18 | iratik | proxium: you might get started with vicidialnow, its a great easy way to get running fast |
| 15:17.25 | proxium | I need some basic instruction :S is the perl script (FastAgi.pl ...) - those in the cron - need to be run on the Asterisk Server not on the Vidial side?? |
| 15:21.06 | *** join/#asterisk grEvenX (n=even@cB78D5AC1.dhcp.bluecom.no) |
| 15:21.11 | KrisWillis | [TK]D-Fender: Ok, I have now recompiled both DAHDI and Asterisk in the right order and the channel driver is now built and loaded, "dahdi show channels" returns one line, a "psuedo" chan extension, asterisk -rvvvvvvvvvv doesn't display anything when I call the PSTN |
| 15:21.42 | [TK]D-Fender | KrisWillis: If it only lists pseudo then you didn't define your channel |
| 15:23.04 | *** join/#asterisk apeiron (n=Chris@c-76-124-252-61.hsd1.pa.comcast.net) |
| 15:24.06 | *** join/#asterisk jmacz (n=jmacz@190.144.75.22) |
| 15:24.14 | KrisWillis | [TK]D-Fender: Is that set-up with dahdi_cfg ? |
| 15:24.34 | [TK]D-Fender | KrisWillis: doubt it. go LOOK |
| 15:25.06 | *** join/#asterisk Deeewayne (n=dwayne@95.104.105.2) |
| 15:25.06 | *** mode/#asterisk [+o Deeewayne] by ChanServ |
| 15:26.00 | *** join/#asterisk marv[work] (n=timr@router.asteriasgi.com) |
| 15:26.03 | proxium | iratik: Vicidial now doesn't separate Vicidial & Asterisk |
| 15:26.08 | *** join/#asterisk Sargun (n=Sargun@atarack/Staff/Sargun) |
| 15:26.56 | *** join/#asterisk Serge8888 (n=Serge@209.47.87.102) |
| 15:27.46 | *** join/#asterisk outtolunc (n=me@adsl-66-218-53-172.dslextreme.com) |
| 15:28.24 | *** join/#asterisk CunningPike (n=CunningP@vpn.dnv.org) |
| 15:31.53 | iratik | proxium: exactly |
| 15:33.07 | proxium | iratik: plz can you tell if the cronjob should be in the Asterisk side or Vicidial |
| 15:33.12 | iratik | proxium: if you are considering integrating an existing asterisk system with vicidial, i wish you luck. Because that is pretty difficult. |
| 15:33.53 | proxium | iratik: no this is a new Install based on Lenny Debian |
| 15:33.55 | haib | Does anyone know if you can send smdi over serial? It looks in smdi.conf that it only receives it. Is that true or is it send and receive? |
| 15:37.14 | *** join/#asterisk oej_ (n=olle@ns.webway.se) |
| 15:45.21 | *** join/#asterisk moy (n=moy@74.12.124.89) |
| 15:47.39 | *** join/#asterisk arpu (n=arpu@chello062178159144.10.14.univie.teleweb.at) |
| 15:57.06 | *** join/#asterisk arpu (n=arpu@chello062178159144.10.14.univie.teleweb.at) |
| 15:58.20 | *** join/#asterisk jplank (n=gbove@cpe-075-181-097-208.carolina.res.rr.com) |
| 16:04.06 | Katty | mmmmmmmmmmmtacosalad |
| 16:04.18 | Katty | omnomnomnomnom |
| 16:05.00 | wpbrown504 | lol |
| 16:05.13 | jplank | I'm having a brain fart, and because I dont know the name, its making it hard to google, what's the name of the zaptel tool used to monitor a PRI? |
| 16:05.48 | wpbrown504 | i forget too |
| 16:05.58 | wpbrown504 | dahdi_tool is the current version |
| 16:06.14 | wpbrown504 | zaptool? |
| 16:06.20 | wpbrown504 | i forget look at the wiki |
| 16:06.47 | wpbrown504 | or look in /usr/sbin |
| 16:06.54 | *** join/#asterisk oglynn (n=me@173.9.102.81) |
| 16:07.59 | oglynn | i have an odd issue with MeetMe, MeetMe(9000) works MeetMe(9000|M) fails i tested my music on hold and it does work from MusicOnHold() |
| 16:08.38 | jplank | <-idiot zttool |
| 16:08.50 | jplank | I thought there was a seperate tool for PRI monitoring |
| 16:09.14 | *** join/#asterisk jjshoe (n=jjshoe@h69-129-142-83.mdsnwi.tisp.static.tds.net) |
| 16:09.31 | jjshoe | any thoughts on how to trouble shoot: dial_exec_full Unable to create channel of type dahdi |
| 16:10.15 | jjshoe | dial is going out group 0, which is channels 1-4 |
| 16:13.06 | Qwell | group 0? |
| 16:13.09 | bpgoldsb | Anyone here familiar with CDR who can help me understand it a little better? |
| 16:13.20 | Katty | i have a reese's mini |
| 16:13.55 | bpgoldsb | All my cdr records are logging a dst of 's', while NoOp($CDR{dst}); shows the correct exten |
| 16:14.08 | Katty | mmm |
| 16:14.17 | Katty | reese's mini no more :< |
| 16:16.45 | *** join/#asterisk lordvadr (n=lordvadr@otis.ctc.biz) |
| 16:17.34 | *** join/#asterisk jer (n=jtregunn@unaffiliated/jer) |
| 16:21.05 | *** join/#asterisk juanIMP (n=Juancho@190.66.8.212) |
| 16:21.43 | eppigy | Katty: :< |
| 16:22.15 | jjshoe | Qwell is defining group 0 a bad idea? |
| 16:23.36 | Corydon76-dig | jjshoe: why would it be a bad idea? |
| 16:24.07 | *** join/#asterisk dcr226 (n=dcr226uk@80.175.10.234) |
| 16:24.25 | Corydon76-dig | bpgoldsb: are you doing a Goto to the 's' extension in your 'h' extension? |
| 16:24.39 | jjshoe | Corydon76-dig no idea: [11:12] <Qwell> group 0? |
| 16:24.53 | dcr226 | hi all. does anyone have any experience in limiting the number of minutes through a trunk on a monthly basis, then switching to the next trunk. (am pulling my hair out) |
| 16:25.02 | Qwell | jjshoe: dunno, just seems wrong |
| 16:25.13 | Qwell | it very well might be fine |
| 16:25.15 | Corydon76-dig | Qwell: nope, groups are 0-63 |
| 16:25.20 | Qwell | Corydon76-dig: stupid c |
| 16:25.36 | Corydon76-dig | Think 2 ^ n |
| 16:25.55 | Qwell | I'm familiar with how it's coded |
| 16:26.05 | Corydon76-dig | and not EOR |
| 16:26.29 | *** join/#asterisk juanIMP (n=Juancho@190.66.8.212) |
| 16:26.33 | dcr226 | was that for me Qwell? |
| 16:26.43 | Corydon76-dig | BTW, if you see EOR, he's lost his tail |
| 16:26.55 | *** join/#asterisk oej (n=olle@ns.webway.se) |
| 16:26.56 | Qwell | dcr226: I would use your name if I was to address you. |
| 16:27.41 | dcr226 | sorry |
| 16:27.57 | SuPrSluG | proxium:http://www.vicidial.org/VICIDIALwiki/tiki-index.php?page=MultiServer |
| 16:28.06 | jjshoe | Qwell I'm not a big fan of it, but it's used all over the place by lots of people :\ |
| 16:28.23 | jjshoe | app_dial.c:1237 dial_exec_full: Unable to create channel of type 'DAHDI' (cause 0 - Unknown) |
| 16:28.28 | jjshoe | I just love Unkonwn as cause :P |
| 16:28.32 | jjshoe | makes troubleshooting easy :P |
| 16:29.54 | proxium | SuPrSluG: Thx a lot, I'll take a look @ that URL |
| 16:30.20 | *** join/#asterisk rue_mohr (n=rue@24.207.122.10) |
| 16:30.30 | jjshoe | dahdi show channels or whatever showed them as psuedo channels too, whatever the heck that is |
| 16:30.57 | rue_mohr | am I right to say that polycom phones are incapable of showing you digits dialed during a call? |
| 16:32.06 | Katty | eppigy: i figured out why i was starving. it was that vault. stopped drinking it and i'm no longer eating everything in sight! |
| 16:32.40 | *** join/#asterisk ttl- (n=patrick@d5153AE82.access.telenet.be) |
| 16:33.20 | rue_mohr | and does anyone know how I can get asterisk to tell a sip phone to use a particular ring tone for a call? |
| 16:33.39 | *** join/#asterisk freakazoid0223 (n=matt@pool-71-246-17-74.phlapa.fios.verizon.net) |
| 16:34.59 | eppigy | Katty: o. sacrificing caffeine sucks |
| 16:35.14 | *** part/#asterisk dcr226 (n=dcr226uk@80.175.10.234) |
| 16:36.20 | MatBoy | weird, my nat is always N |
| 16:36.24 | MatBoy | no matter what I set |
| 16:36.41 | Katty | eppigy: well not exactly. i'm still drinking dr pepper and tea. |
| 16:36.59 | Katty | eppigy: peach tea! |
| 16:38.55 | MatBoy | lol |
| 16:38.58 | MatBoy | nespresso !! |
| 16:39.29 | rue_mohr | our incomming audio is cutting out when we call one of our suppliers, only seems to happen with their phone system |
| 16:39.37 | rue_mohr | its a dahdi interface |
| 16:39.52 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 16:40.02 | rue_mohr | I need asterisk to tell the sip phone what ring tone to use for incomming calls |
| 16:44.28 | *** join/#asterisk ccesario (n=ccesario@189-19-9-100.dsl.telesp.net.br) |
| 16:45.03 | *** join/#asterisk anonymouz666 (n=anonymou@189.24.68.173) |
| 16:45.38 | eppigy | Katty: peach tea is awesome |
| 16:46.17 | oglynn | any ideas on MeetMe(XXXX) working but MeetMe(XXXX|M) failing |
| 16:46.34 | oglynn | * 1.6.0.9 |
| 16:46.40 | seanbright | rue_mohr: what kind of phone? |
| 16:47.04 | SuPrSluG | oglynn:use commas w/ 1.6 |
| 16:47.24 | SuPrSluG | not pipes |
| 16:47.42 | oglynn | SuprSlug: thanks i keep reading old examples. |
| 16:48.12 | SuPrSluG | pulled a hair or two out figuring that out myself |
| 16:48.14 | AsteriskDom | but its not the same for asteris commas and pipes |
| 16:48.18 | AsteriskDom | ? |
| 16:48.30 | AsteriskDom | and that case |
| 16:48.38 | *** join/#asterisk ruben23 (n=AGENT@124.107.3.178) |
| 16:48.48 | *** join/#asterisk telnettech (i=telnette@gw.percipia.com) |
| 16:49.42 | ruben23 | hi...how many concurrent calls can an asterisk box server handles with a xeon dual core processors 4 GB RAM.. |
| 16:50.08 | Katty | a lot |
| 16:50.24 | telnettech | need some help.......i am loading the asterisk software and I get this error info and cant get it to clear......any idea what the problem may be? |
| 16:50.29 | telnettech | http://pastebin.com/d6dfad519 |
| 16:50.32 | ruben23 | 25 callers..? |
| 16:50.33 | SuPrSluG | reben23: a lot of variables to that question. transcoding or not? etc... |
| 16:50.44 | Fabian- | 100+ callers ? |
| 16:50.57 | ruben23 | g729 codec trqanscoding |
| 16:51.02 | Katty | ruben23: there's not really a formula anywhere that i know of. |
| 16:52.25 | ruben23 | Katty: i have talk to a consultant..saying that 25 callers are the max for a quad core asterisk server...or else it will crash |
| 16:52.28 | seanbright | rue_mohr: search for 'polycom distinctive alert-info' on google. you'll need to use SIPAddHeader most likely. |
| 16:52.34 | ruben23 | using g729 codec.. |
| 16:52.49 | Katty | ruben23: did that consultant give you a formula to follow? |
| 16:53.05 | Qwell | ruben23: Fire him. Now. |
| 16:53.43 | ruben23 | Katty: no...he just said that its based on his experience |
| 16:53.58 | Katty | ruben23: so you have no real, hard data, to go on. |
| 16:54.00 | Qwell | ruben23: can you msg me his name? |
| 16:54.01 | Katty | ruben23: shucks |
| 16:54.31 | wpbrown504 | ruben23:crikey |
| 16:54.51 | ruben23 | his name is cordova |
| 16:55.42 | *** join/#asterisk MmixX (n=mix@61.14.191.137) |
| 16:58.09 | *** join/#asterisk b0bu5 (n=chatzill@201.36.214.116) |
| 17:00.45 | ruben23 | he said the cause of caller limits..because of the agi scripts..and meetme |
| 17:01.02 | ruben23 | which takes the load of the asterisk box |
| 17:01.10 | Qwell | You should find a better consultant. |
| 17:01.22 | Qwell | one who knows what he's talking about. |
| 17:01.27 | eppigy | 4sure |
| 17:01.48 | *** join/#asterisk jtodd (n=jtodd@blob.fox-den.com) |
| 17:01.49 | *** mode/#asterisk [+o jtodd] by ChanServ |
| 17:03.23 | ruben23 | actually it a vicidial project....with asterisk |
| 17:03.28 | *** join/#asterisk defsmac (n=andy@213.128.237.170) |
| 17:03.52 | *** part/#asterisk mike345 (n=mike_sim@64.74.198.10) |
| 17:04.05 | defsmac | Hi - I 'm using 1.4.24 and dahdi with freepbx and don't get sound on incoming and outgoing calls over my PRI - any suggestions ? |
| 17:04.40 | Qwell | defsmac: NAT |
| 17:04.51 | defsmac | Qwell, PRI |
| 17:04.52 | Qwell | (You're using SIP phones, right?) |
| 17:04.59 | rue_mohr | hah so, I'm being told asterisk sucks because we dont have a central set all the calls are ringing to, at the same time, the same person says all lines have to ring to all the sets |
| 17:05.13 | defsmac | Qwell, sip locally - all works ok - all outoing calls are over PRI - there is no NAT |
| 17:05.22 | [TK]D-Fender | rue_mohr: All solved in the dialplan |
| 17:05.38 | rob327 | does anyone know of a decently priced auto dialer service? it would be for only one person (me) to sit on for a few hours a day |
| 17:05.38 | [TK]D-Fender | rue_mohr: and s/we/I/ |
| 17:05.55 | rob327 | web based auto dialer service, just one agent |
| 17:05.58 | rue_mohr | tell me how a dialplan is supposed to ring all the phones for all the lines and have only 1 central phone ring |
| 17:06.02 | SuPrSluG | reben23: vicidial is a different animal than asterisk alone. since it uses conferences for each call the load will be higher |
| 17:06.32 | Qwell | SuPrSluG: but 25 calls max on a quad core system? that's complete BS |
| 17:06.42 | [TK]D-Fender | rue_mohr: If all phones ring for all lines, and central set has to ring for all as well, all phones are treated the same |
| 17:07.18 | ruben23 | SuPrSluG:so based on what he says....its true..? |
| 17:07.32 | Qwell | ruben23: No, it's complete BS |
| 17:07.34 | SuPrSluG | asking in the vicidial forums would be the thing to do. |
| 17:07.50 | Qwell | find a new consultant. |
| 17:07.50 | SuPrSluG | it should handle more than that. |
| 17:08.49 | SuPrSluG | why do you want to transcode 729 in a call center environment? that's the question! |
| 17:09.43 | ruben23 | SuPrSluG: i use g729 codec form asterisk to voip provider...on my local..i use ulaw/alaw... |
| 17:10.38 | defsmac | anyone got any suggestions ? I'm on site and a little stuck :o |
| 17:10.46 | oglynn | @Quell: SuPrSlug: We had an install where we were using MeetMe for every inbound call and not using it for outbound. All G711 we had a bunch of complaints about voice quality on the inbound calls. Chnaged it so as no more MeetMe overuse and they are very happy. It was extreemly low usage like 4 concurrent calls max |
| 17:11.59 | oglynn | 4 calls = 8 legs |
| 17:12.11 | *** join/#asterisk StephenF (n=none@198.144.201.106) |
| 17:13.21 | MatBoy | mhh config issues can be difficult :) |
| 17:13.48 | Qwell | MatBoy: referring to the above? I agree. |
| 17:16.16 | rue_mohr | wholy gazoo, TX is off the scale on the dahdhi_monitor! |
| 17:16.49 | rue_mohr | txgain is tx in chan_dahdi.conf right? |
| 17:17.23 | defsmac | DEBUG[3098] chan_dahdi.c: Set option AUDIO MODE, value: ON(1) on DAHDI/1-1 |
| 17:17.32 | defsmac | I get that and a click when I hang up a call |
| 17:17.51 | defsmac | wonders if it should do that at the start not the end :) |
| 17:18.19 | [TK]D-Fender | rue_mohr: I recall all of the psycho gains you've been applying as you built that system up... |
| 17:19.52 | *** join/#asterisk denovaldo (n=chatzill@201.36.214.116) |
| 17:21.39 | *** join/#asterisk bbryant (n=ssddsasd@m445a36d0.tmodns.net) |
| 17:21.39 | *** join/#asterisk Mark21 (n=mark@freenode/sponsor/mark17) |
| 17:21.53 | MatBoy | Qwell: to what above ? |
| 17:22.36 | rue_mohr | right now I have to take the dahdi tx down by 8db and the rx up by 8db |
| 17:22.47 | rue_mohr | I'm in a try and adjust loop here... |
| 17:23.25 | Mark21 | hello, i have a dialplan that first answer the incoming call, it starts recording the call (using MixMonitor), it start dial (2 sip number + mobile phone number, this last one is done via a trunk) and repeats this a few times |
| 17:23.47 | MatBoy | someone interrested in in 2x 410P cards ? I don't need them anymore |
| 17:24.00 | MatBoy | I actually never needed them :) |
| 17:24.37 | Mark21 | but here is the problem: today i did get a call via this dialplan and i did answer it on the mobile phone (because i was outside) and the call isn't recorded (but it was important), why can this be? and how can i prevent this in the future? |
| 17:24.58 | Mark21 | the caller id is faked (0696*0001, this is not a correct phone number) |
| 17:25.58 | rue_mohr | what are the gian limits for the chan_dahdi.conf!?!?! |
| 17:27.39 | *** join/#asterisk captiancrash (n=jonmoore@70.159.118.70) |
| 17:27.47 | [TK]D-Fender | Mark21: Show us a broken call. There is nothing to work with to fix anything. |
| 17:28.26 | rue_mohr | I'm getting close at rx of 10 and a tx of -15 |
| 17:28.55 | Qwell | rue_mohr: Use *MUCH* lower numbers. |
| 17:30.03 | rue_mohr | I'm watching for saturation on the chan_dahdi with dahdi_monitor |
| 17:30.14 | rue_mohr | these are whats making it come in right |
| 17:30.29 | rue_mohr | I need to take the tx even further, but I'm gonna wait and see |
| 17:30.55 | Mark21 | [TK]D-Fender: moment, i will look for all info i have @ logs and config that i can give via a pastebin |
| 17:30.57 | rue_mohr | those are *CRAZY* numbers, I know |
| 17:31.04 | rue_mohr | I should be under 6db |
| 17:31.08 | rue_mohr | either way |
| 17:31.21 | [TK]D-Fender | rue_mohr: How much time have you wasted tweaking that card for gains, echo, etc? |
| 17:34.32 | kc8pxy | [TK]D-Fender: too much? |
| 17:34.51 | [TK]D-Fender | kc8pxy: I thought that a long while back :) |
| 17:35.04 | [TK]D-Fender | kc8pxy: And the show keeps getting better |
| 17:35.09 | rue_mohr | [TK]D-Fender, on the dahdi_monitor, alsmost everything said from the office was off the scale |
| 17:35.28 | [TK]D-Fender | rue_mohr: You clearly need Scalr 2.0! |
| 17:35.34 | kc8pxy | [TK]D-Fender: when i get a dahdi card, is there any reason i should not just use the OOB settings? |
| 17:35.44 | rue_mohr | rxgain=10.0 |
| 17:35.45 | rue_mohr | txgain=-10.0 |
| 17:35.48 | [TK]D-Fender | kc8pxy: ...huh? |
| 17:36.02 | [TK]D-Fender | rue_mohr: Yup... psycho.. |
| 17:36.06 | rue_mohr | is that db, or scalar? |
| 17:36.15 | [TK]D-Fender | rue_mohr: db last I checked |
| 17:36.24 | rue_mohr | why do I have to adjust these gains by so much!? |
| 17:36.28 | Mark21 | [TK]D-Fender: a part from my logfile: http://yourpaste.net/1883/ |
| 17:36.33 | Qwell | rue_mohr: you...don't. |
| 17:36.36 | kc8pxy | [TK]D-Fender: i think I'm again showing my asterisk noobness. |
| 17:36.55 | rue_mohr | the echo can is freaking out and blanking incomming audio during calls |
| 17:37.02 | Qwell | yeah, duh |
| 17:37.10 | Qwell | turn them DOWN. A LOT |
| 17:37.26 | rue_mohr | I had them set to 0 and 0 when it was doing it |
| 17:37.56 | [TK]D-Fender | Mark21: "/var/log/record/record-1200-0696*0001_1240416444.wav" <- last I checked, "*" was an illegal character |
| 17:38.01 | rue_mohr | now its being more consistant and just supressing background static, which people dont like cause they dont know if the call has been droped |
| 17:38.45 | rue_mohr | I dont believe it is if you escape it, under linux you can use any character... |
| 17:38.59 | rue_mohr | deleting files called - is fun |
| 17:39.14 | *** join/#asterisk jeffgus (n=jeffgus@green.zimage.com) |
| 17:39.25 | Mark21 | [TK]D-Fender: you can see how i create 0696*0001 part, but how can i change illegal chars to a legal character (like _ or -)? |
| 17:39.30 | kc8pxy | rue_mohr: files called - ??? why do you have one in the first place? |
| 17:39.38 | rue_mohr | Qwell, tell me this, on the dahdi_monitor, where shoudl the audio peaks land? |
| 17:39.48 | Mark21 | so next time it can be recorded (and yes i need to record the phone number in the file name) |
| 17:40.03 | Qwell | rue_mohr: whereever they land when it *SOUNDS* correct. |
| 17:40.03 | rue_mohr | kc8pxy, they are usefull for stopping recursive directory deletes |
| 17:40.28 | [TK]D-Fender | Mark21: "core show function CUT" |
| 17:40.33 | defsmac | does anyone think that my problem could be to do with this SET AUDIO MODE |
| 17:40.35 | defsmac | option ? |
| 17:41.28 | [TK]D-Fender | Qwell: PH34R his Polycom gain settings ;) |
| 17:41.40 | kc8pxy | rue_mohr: how does - stop `sudo rm ./* -rf` ?? |
| 17:42.42 | Qwell | that's nonsense |
| 17:42.50 | *** join/#asterisk voxter (n=voxter@76.77.91.250) |
| 17:42.57 | Qwell | having a file named - isn't going to stop you from deleting anything |
| 17:43.03 | captiancrash | when using followme app, if i pick up a call the audio appears to be cut off by a few seconds. .. i hear "press 2 to .." instead of starting with press 1. |
| 17:43.35 | SuPrSluG | kc8pxy:do you have x permission on the directory? |
| 17:44.23 | kc8pxy | SuPrSluG: with sudo i do :) |
| 17:44.41 | Qwell | with sudo, all things are possible. |
| 17:45.08 | haib | Does anyone know if you can send smdi over serial? It looks in smdi.conf that it only receives it. Is that true or is it send and receive? |
| 17:45.36 | ipstatic | Would someone mind helping me troubleshoot the following error: |
| 17:45.37 | ipstatic | Call failed to go through, reason (0) Call Failure (not BUSY, and not NO_ANSWER, maybe Circuit busy or down?) |
| 17:45.46 | [TK]D-Fender | ipstatic: that proves absolutely nothing |
| 17:45.55 | [TK]D-Fender | ipstatic: Go look at REAL debug (SIP / PRI, etc) |
| 17:46.14 | ipstatic | [TK]D-Fender: I have them, but all I see is CSeq: 102 CANCEL |
| 17:46.20 | ipstatic | that has any real importance |
| 17:46.25 | ipstatic | [TK]D-Fender: I fixed NAT |
| 17:46.29 | ipstatic | however I still have issues |
| 17:46.42 | ipstatic | could it be the SIP provider? |
| 17:47.00 | [TK]D-Fender | ipstatic: However you still haven't learned to PASTEBIN all of your backup and not make us ask for every detail you should already be providing immediately |
| 17:47.09 | kc8pxy | sits back, waiting for the next bit of insanity spewed into the channel :) |
| 17:47.42 | ipstatic | [TK]D-Fender: I am sorry, I have pastebined what I think you might need here: http://pastebin.com/d10b15a4a |
| 17:47.57 | ipstatic | [TK]D-Fender: Please tell me what else you need |
| 17:48.44 | [TK]D-Fender | ipstatic: and you only show it at the point of failure, not what leads to it. THE ENTIRe DAMN CALL |
| 17:49.25 | *** join/#asterisk dethstar (n=Obi-Wan@unaffiliated/dethstar) |
| 17:49.44 | ipstatic | [TK]D-Fender: I tried finding the entire call, its hard while 40 other calls arein the output as well |
| 17:49.56 | *** join/#asterisk MaliutaLap (n=biteme@kiev.lusan.id.au) |
| 17:50.09 | [TK]D-Fender | ipstatic: Doesn't help us much.. |
| 17:50.20 | ipstatic | [TK]D-Fender: ok, I will try and get more |
| 17:50.24 | ipstatic | [TK]D-Fender: thank you for you time |
| 17:50.26 | defsmac | is having a bad day |
| 17:50.50 | [TK]D-Fender | defsmac: Do like Qwell and wait 5 minute |
| 17:51.03 | Qwell | heh |
| 17:51.14 | Qwell | my day was at least acceptable before those 5 minutes |
| 17:51.19 | defsmac | :) |
| 17:51.30 | defsmac | I'm 400 miles from home and have to drive back tonight |
| 17:52.33 | [TK]D-Fender | defsmac: Pinto w/ full tank rear-ended by a Mack truck, home in 5.2 seconds |
| 17:53.01 | [TK]D-Fender | defsmac: At least.. little bits of you... |
| 17:53.18 | russellb | depending on your definition of "home" ... |
| 17:53.19 | defsmac | this is going to be something stupid I have overlooked - I am sure |
| 17:53.41 | defsmac | 1.4.24 with dahdi and sangoma a101 - no audio on PRI calls in or out |
| 17:56.37 | *** join/#asterisk columbo (n=PeterFal@pool-71-177-208-38.lsanca.dsl-w.verizon.net) |
| 17:58.08 | rue_mohr | I believe in using numbers to adjust these gains, the voip sets seem to have agc, which means when i take hte dahdi drivers from 0db to 15 db, there is almost NO difference at the sets, however, it totally changes the dahdi echo can behaviour |
| 17:58.10 | kc8pxy | ipstatic: grep is your friend :) |
| 17:58.18 | ipstatic | kc8pxy: yeah thats what I used |
| 17:58.21 | ipstatic | based on the call-id |
| 17:58.26 | ipstatic | and that is all that came up with it |
| 17:58.37 | ipstatic | also these calls are orginated by call files, if that makes a difference |
| 17:59.47 | [TK]D-Fender | kc8pxy: can't cleanly grep for multi-line packets... |
| 18:00.24 | Qwell | rue_mohr: 15 is *FAR* too high. |
| 18:00.33 | Qwell | I've said that several times now. |
| 18:01.02 | ipstatic | [TK]D-Fender: can you give me any hints on what I should be looking for? these calls do not have any rtp stream that i can see, only those sip headers |
| 18:01.05 | *** join/#asterisk ChkDigit (n=mike@static24-72-71-175.regina.accesscomm.ca) |
| 18:01.08 | *** join/#asterisk Amazinzay (n=ijensen@67.108.187.186.ptr.us.xo.net) |
| 18:01.31 | [TK]D-Fender | ipstatic: I'm not going to waste my time "guessing" |
| 18:01.50 | ipstatic | [TK]D-Fender: I meant what I should search for, to find what you need |
| 18:02.19 | ipstatic | If those sip headers don't have all you need, what would the ones you looking for have so I can find them |
| 18:02.34 | [TK]D-Fender | ipstatic: Entire call. |
| 18:02.55 | ipstatic | [TK]D-Fender: ok, and does each part of that call use the call-id? |
| 18:03.10 | ipstatic | and can I see this call from sip set debug output? |
| 18:04.15 | [TK]D-Fender | ipstatic: verbose 10, sip debug enabled |
| 18:05.27 | ipstatic | [TK]D-Fender: I had verbose set to 24 and sip debug was enabled. Would the other parts of the call use the call-id so that I can find it? |
| 18:05.49 | rue_mohr | Qwell, how do I find the right values? |
| 18:06.15 | [TK]D-Fender | ipstatic: thats it. |
| 18:06.19 | *** join/#asterisk bijit (n=chatzill@190.10.115.50) |
| 18:06.26 | ipstatic | [TK]D-Fender: then that is all of the call |
| 18:06.33 | [TK]D-Fender | ipstatic: BS |
| 18:06.36 | bijit | can I use zaptel 1.4 with asterisk 1.2? |
| 18:06.37 | ipstatic | I swear |
| 18:06.38 | [TK]D-Fender | ipstatic: you are showing a CANCEL |
| 18:06.52 | [TK]D-Fender | ipstatic: How the hell can you CANCEL something you didn't START? |
| 18:06.52 | MmixX | hi... |
| 18:06.52 | Qwell | rue_mohr: by listening. |
| 18:06.55 | defsmac | does anyone not have even the slighest suggestion for me ? |
| 18:07.00 | [TK]D-Fender | bijit: NO |
| 18:07.20 | eppigy | i am dave |
| 18:07.22 | eppigy | hello |
| 18:07.29 | jaytee | hi dave |
| 18:07.31 | bijit | [TK]D-Fender: thanks no wonder its not loading chan_zap |
| 18:07.38 | rue_mohr | Qwell, |
| 18:07.41 | rue_mohr | I believe in using numbers to adjust these gains, the voip sets seem to have agc, which means when i take hte dahdi drivers from 0db to 15 db, there is almost NO difference at the sets, however, it totally changes the dahdi echo can behaviour |
| 18:08.01 | MmixX | is there a way to change the bitrate of the recordings? our recordings currently set to 8bit... we want to change it to 16bit |
| 18:08.17 | rue_mohr | I'v been from -3 to 15, it sounds the same on the sets |
| 18:08.25 | [TK]D-Fender | rue_mohr: You shouldn't be USING the DAHDI echocan. |
| 18:08.27 | rue_mohr | but it makes a HUGE difference to the echo can |
| 18:09.06 | rue_mohr | is there not a silence mute option somewhere? |
| 18:10.42 | rue_mohr | phone.conf:;silencesupression=yes |
| 18:10.48 | rue_mohr | whats the default? |
| 18:12.38 | juanIMP | good day every body :D |
| 18:13.33 | *** join/#asterisk thansen (n=thansen@c-76-27-110-194.hsd1.ut.comcast.net) |
| 18:14.19 | juanIMP | I have the next query, can i set sip trunking between hybrid tda600 and Asterisk? |
| 18:14.57 | [TK]D-Fender | rue_mohr: No. phone.conf = ignore |
| 18:15.09 | rue_mohr | oh |
| 18:15.15 | [TK]D-Fender | juanIMP: Sure |
| 18:16.31 | juanIMP | ummm, great :D the same model * to * [TK]D-Fender ?? |
| 18:16.37 | rue_mohr | ;transmit_silence = yes <-- is that what causes dtmf to be muted out |
| 18:17.51 | [TK]D-Fender | juanIMP: huh? |
| 18:18.10 | rue_mohr | ok, so now I shoudl check and see if its the polycom sets doing it |
| 18:18.16 | *** join/#asterisk lanning (n=lanning@nat/yahoo/x-3bb4c0604666f805) |
| 18:23.54 | *** join/#asterisk sah-work (n=Bawbatos@65-119-47-100.dia.static.qwest.net) |
| 18:24.00 | rue_mohr | no it seems to be turned off |
| 18:24.02 | rue_mohr | hmm |
| 18:24.17 | rue_mohr | anything else that might do silence supression? |
| 18:24.55 | [TK]D-Fender | rue_mohr: only EC on crack |
| 18:27.53 | *** join/#asterisk jameswf (n=james@unaffiliated/jameswf-home) |
| 18:28.08 | rue_mohr | so when both audio channels fall to ambient noise the echo canceler freaks out and mutes the audio from the pots? |
| 18:28.26 | juanIMP | [TK]D-Fender: could i use the same steps that asterisk to asterisk trunking uses? |
| 18:29.56 | Qwell | ~ecfo |
| 18:29.56 | infobot | Echo Canceler Freak Out, this happens when the rxgain is too high and the echo canceler freaks out. Some users describe it as "screeching", "feedback", "static", or other useless terms. If users report "static" on a system where there cannot be static (all digital, PRI, SIP, etc), you might be experiencing ECFO. what happens when the echo canceller suddenly realises its a crappy design based on a half baked 20 year old apps note. |
| 18:30.10 | Qwell | hey look, a technical term |
| 18:30.43 | [TK]D-Fender | juanIMP: SIP is SIP... it isn't "magic". Settings depends on whats on the other side. |
| 18:30.45 | rue_mohr | this is when both sides aren't talking, just background static |
| 18:30.47 | Corydon76-dig | What, "crappy"? |
| 18:30.53 | Qwell | Corydon76-dig: ecfo |
| 18:31.15 | rue_mohr | on the office side, the static drops out and the receptionist dosn't know if the call was dropped |
| 18:32.20 | rue_mohr | yesterday I had the tx gain at 3 and it was causing the audio to go mute when the incomming callers were talking |
| 18:32.36 | rue_mohr | right now, the tx gain is about -13 and its ok |
| 18:33.04 | rue_mohr | whats amd.conf? |
| 18:33.15 | rue_mohr | oh k |
| 18:34.27 | juanIMP | ok [TK]D-Fender] , thanks a lot |
| 18:38.18 | *** part/#asterisk Amazinzay (n=ijensen@67.108.187.186.ptr.us.xo.net) |
| 18:40.00 | *** join/#asterisk dominicl (n=blah@ri-sw-metro.ptgrey.com) |
| 18:43.09 | Katty | i think it's snacktime |
| 18:44.24 | eppigy | i am too sleepy to snack :< |
| 18:44.51 | Katty | :< |
| 18:44.52 | Katty | go nap |
| 18:45.02 | *** join/#asterisk Erol_ (n=x@88.235.144.246) |
| 18:45.28 | jeffgus | could anyone here offer any suggestions on why a sangoma a200 card won't pass audio? |
| 18:45.40 | jeffgus | i have installed pci ones before never had trouble |
| 18:45.49 | jeffgus | this is a pci-e |
| 18:45.55 | jeffgus | if that makes any difference |
| 18:45.55 | Erol_ | hi, is it possible to make outside calls by using an internet connection rather than using PSTN lines? |
| 18:46.03 | Katty | Erol_: yes |
| 18:46.13 | eppigy | i cannot nap |
| 18:46.16 | eppigy | it is work tyme |
| 18:46.18 | Katty | Erol_: :< |
| 18:46.20 | Katty | err |
| 18:46.22 | Katty | eppigy: :< |
| 18:46.29 | eppigy | yesh |
| 18:46.35 | Erol_ | Katty: what do i need to do so? |
| 18:46.44 | Katty | Erol_: a 'sip trunk' provider |
| 18:46.54 | eppigy | or iax |
| 18:47.02 | eppigy | IF YOU SO CHOOSE |
| 18:47.04 | Erol_ | Katty: is this different than ip trunk provider? |
| 18:47.14 | Erol_ | eppigy: what is iax? |
| 18:47.16 | Katty | i've never heard of 'ip trunk' before, so i dont' know. |
| 18:47.19 | eppigy | Erol_: i think some due diligence is in order |
| 18:47.25 | eppigy | YOU KNOW WHAT I MEAN |
| 18:47.32 | eppigy | dilligence |
| 18:47.33 | eppigy | ? |
| 18:47.40 | Katty | eppigy: probably the later |
| 18:47.45 | Katty | latter? |
| 18:47.48 | eppigy | yesh |
| 18:47.50 | eppigy | haha |
| 18:48.02 | Katty | we can't spell today. |
| 18:48.08 | eppigy | i know |
| 18:48.16 | eppigy | but we make it look cool |
| 18:48.17 | Erol_ | eppigy: what is the difference between IAX and SIP? |
| 18:48.24 | eppigy | bro please read up |
| 18:48.30 | Katty | infobot: iax |
| 18:48.31 | infobot | i guess iax is port 5036 for the original (deprecated) IAX protocol. Port 4569 is for the the current IAX2 protocol. IAX is pronounced "Eeks". stands for Inter-Asterisk Exchange |
| 18:48.32 | eppigy | "due diligence" |
| 18:48.44 | Katty | infobot: sip |
| 18:48.44 | infobot | somebody said sip was http://www.cs.columbia.edu/sip/ X11 PPP dialer interface written in gtk+. URL: http://www.geocities.com/SiliconValley/Campus/3104/sip/ Session Initiation Protocol (see RFC 3261) |
| 18:48.45 | eppigy | I had to learn all this stuff |
| 18:48.49 | eppigy | as did everyone else |
| 18:48.54 | Katty | we did?! |
| 18:48.59 | Katty | i thought i was refusing to learn this week. |
| 18:49.01 | eppigy | ok I am lying |
| 18:49.06 | eppigy | but I mean |
| 18:49.25 | Erol_ | i got it thanks |
| 18:49.27 | Erol_ | and lastly.. |
| 18:49.30 | AsteriskDom | hey thats nice |
| 18:49.31 | Katty | [TK]D-Fender: what's the infobot thingy for the bandwidth.com type provider thingies |
| 18:49.42 | AsteriskDom | infobot: nat |
| 18:49.43 | infobot | extra, extra, read all about it, nat is Network Address Translation Usable in Asterisk sip.conf file with externip, localnet, and localmask setup properly. See docs. |
| 18:49.44 | Katty | [TK]D-Fender: what's the thingy for thingies |
| 18:49.57 | Katty | wow look at technical me today |
| 18:50.11 | [TK]D-Fender | ~itsplist-us |
| 18:50.12 | infobot | [~itsplist-us] Here are some popular ITSPs (USA) starting with the more respected ones : http://www.teliax.com , http://connect.voicepulse.com , http://www.nufone.net , http://www.broadvoice.com, http://www.jnctn.com , http://www.bandwidth.com , http://vitelity.net |
| 18:50.13 | [TK]D-Fender | ~itsp |
| 18:50.14 | infobot | [~itsp] An ITSP is an Internet telephony Service Provider or "VoIP Telephone company". The allow you to either PLACE calls to the PSTN (called Termination), RECEIVE calls from the PSTN (called Origination), or both. Some offer fixed rates, others $/min. Enter ~itsplist-us (USA) or ~itsplist-ca (Canada) for a listing of popular ITSPs |
| 18:50.26 | Katty | commits itsp to memory |
| 18:50.31 | Katty | POP QUIZ |
| 18:50.38 | Katty | what's the difference between effect and affect |
| 18:50.49 | *** join/#asterisk nhaines (n=nathan@ubuntu/member/nhaines) |
| 18:50.50 | haib | one is a noun |
| 18:50.52 | n3hxs | effect will affect you |
| 18:51.05 | *** join/#asterisk tobias (n=tobias@cpe-071-070-219-040.nc.res.rr.com) |
| 18:51.10 | n3hxs | OR a/e ;) |
| 18:51.13 | Erol_ | How does wi-fi users solve roaming problem?_ |
| 18:51.28 | Katty | what the ^_- |
| 18:52.04 | eppigy | i am not good with words |
| 18:52.29 | Katty | your affect is not very affective. |
| 18:52.34 | Katty | oh |
| 18:52.38 | eppigy | haha |
| 18:52.39 | Katty | s/affect/effect/ |
| 18:52.58 | Katty | my typing is doom :< |
| 18:53.00 | haib | self pwn |
| 18:53.01 | Erol_ | Katty: ever used wi-fi phones with asterisk? |
| 18:53.16 | Katty | Erol_: if by wifi phone you mean a laptop connected to a wap or wireless router, then sure. |
| 18:53.18 | eppigy | what do you mean by roaming problem |
| 18:53.27 | eppigy | is you set up an ESS wlan scheme |
| 18:53.33 | eppigy | with non-overlapping channels |
| 18:53.37 | Katty | is you set up?! |
| 18:53.38 | eppigy | there is no need to switch IP's |
| 18:53.42 | eppigy | *IF |
| 18:53.43 | Erol_ | There are wi-fi ip phones like linksys etc. |
| 18:53.52 | Katty | doom :< |
| 18:53.54 | eppigy | yes I understand that portion Erol_, thank you |
| 18:54.08 | eppigy | move on to the IQ80+ part |
| 18:54.10 | eppigy | THANKS |
| 18:54.25 | Katty | POP QUIZ |
| 18:54.29 | Erol_ | eppigy: when using wi-fi phones and when you move with the phone and get out of the ap range.. |
| 18:54.30 | Katty | Olestra is used for WHAT type of industrial product |
| 18:54.43 | eppigy | Erol_: ok |
| 18:54.44 | nhaines | Katty: snack foods. |
| 18:54.50 | eppigy | then you are out of AP range |
| 18:54.53 | Katty | nhaines: keyword INDUSTRIAL |
| 18:54.54 | eppigy | simple enough |
| 18:54.54 | Erol_ | eppigy: it should automaticaly change the ap |
| 18:54.58 | eppigy | no |
| 18:55.02 | nhaines | Katty: you've never seen my coworkers eat. :) |
| 18:55.10 | Katty | i giggled. |
| 18:55.11 | eppigy | if you have ESS set up corrcetly yes |
| 18:55.21 | eppigy | it will change AP without changing IP |
| 18:55.27 | Carlos_PHX | If you want to roam with a wi-fi phone, set up a single SSID with non-overlapping channel allocation. Use APs, not routers, so the phone keeps a single IP. Pretty simple really. |
| 18:55.38 | eppigy | yes |
| 18:55.48 | Katty | nhaines: Industrial LUBRICANT! |
| 18:55.50 | Carlos_PHX | On the other hand, I've never found a wi-fi phone that isn't total crap. |
| 18:55.54 | eppigy | Katty: hawt |
| 18:55.57 | Katty | nhaines: is in those fat free pringles. |
| 18:55.58 | Carlos_PHX | And I have piles of them here. |
| 18:56.06 | eppigy | olestra |
| 18:56.12 | Katty | and olean |
| 18:56.15 | Katty | and sufrose |
| 18:56.16 | Katty | sefrose? |
| 18:56.19 | Katty | something like that. |
| 18:56.36 | Erol_ | Carlos_PHX: what do you mean by non-overlaping channel allocation? |
| 18:56.43 | Katty | oh dear |
| 18:56.55 | eppigy | Erol_: BRO |
| 18:57.18 | eppigy | channels 1,6,11 |
| 18:57.22 | eppigy | for 802.11g |
| 18:57.51 | eppigy | what would you like to know abotu quantum ambiguety? |
| 18:57.56 | Katty | eppigy: song of the day http://www.youtube.com/watch?v=GdKrAlXE8w4 |
| 18:58.07 | Katty | oooh!!! quantum physics is next?! |
| 18:58.12 | Katty | can we talk about spookey entanglement?! |
| 18:58.42 | eppigy | we can talk entanglement |
| 18:58.46 | eppigy | for certain |
| 18:58.47 | Qwell | spooky action at a distance FTW |
| 18:58.50 | eppigy | YOU KNWO WHAT I MEAN |
| 18:59.36 | Erol_ | Carlos_PHX: you there? |
| 18:59.49 | eppigy | haha did Erol_ ignore me |
| 19:00.05 | Erol_ | eppigy: nope why would i do that |
| 19:00.10 | Katty | quantum entanglement is like... |
| 19:00.14 | Katty | Qwell and russellb |
| 19:00.17 | Erol_ | eppigy: i read it all and learned what it is |
| 19:00.31 | eppigy | well those are the non-overlapping channels |
| 19:00.32 | Katty | they can't be adequately described without mention of their counterpart |
| 19:00.36 | eppigy | for 802.11g |
| 19:00.41 | Katty | despite their spatial seperation |
| 19:00.46 | Katty | spacial? |
| 19:01.01 | Katty | spatially separated |
| 19:01.09 | Katty | nonlocal. |
| 19:01.16 | Katty | i can spell that one! |
| 19:01.19 | Erol_ | eppigy: so with simple logic, i will hang access points with non overlapping channels close to each other.. |
| 19:01.40 | Erol_ | eppigy: and configure them with the same ssid |
| 19:02.03 | Erol_ | eppigy: since they wont be able to sense each other this wont be a problem |
| 19:02.15 | Erol_ | eppigy: did i get it right? |
| 19:03.31 | AsteriskDom | what king of access points he has maybe they haves proxy mobile features |
| 19:03.38 | Katty | oh neat. physicist have 'teleported' information between two atoms separated by a large ammount of distance |
| 19:03.58 | Katty | by the use of entanglement |
| 19:04.20 | Erol_ | astassistant: i have cisco 1200 series |
| 19:04.20 | mmlj4 | old news, actually |
| 19:04.31 | Katty | i wonder how long it will take for someone to patent quantum entangled bank transactions |
| 19:05.00 | bpgoldsb | Anyone here familiar with CDR who can help me understand it a little better? All my cdr records are logging a dst of 's', while NoOp($CDR{dst}); shows the correct exten |
| 19:05.05 | Katty | technically, it would be faster than the speed of light |
| 19:05.12 | Katty | or, maybe not... it'd be more like point to point folding |
| 19:05.26 | Katty | i'm not sure how it would really work. |
| 19:07.47 | eppigy | you just blew my mind |
| 19:07.51 | Katty | does that mean a black hole spews all over the theory of special relativity? |
| 19:07.57 | eppigy | Erol_: yes if each ap is configured correctly |
| 19:07.59 | Katty | because it eats light |
| 19:08.04 | eppigy | man |
| 19:08.09 | eppigy | that is really freaky |
| 19:08.11 | Katty | isn't that why you can't see a black hole? no light escapes? |
| 19:08.15 | eppigy | cause I was thinking abotu that onlunch |
| 19:08.24 | eppigy | yes past the event horizin |
| 19:08.27 | eppigy | horizon |
| 19:08.31 | eppigy | light photons |
| 19:08.33 | eppigy | are pulled in |
| 19:08.34 | Katty | welll |
| 19:08.37 | Katty | so then |
| 19:08.44 | Katty | black holes suck faster than the speed of light |
| 19:08.49 | eppigy | well |
| 19:08.53 | eppigy | I am not sure if speed |
| 19:08.57 | eppigy | is the same as force |
| 19:09.06 | eppigy | is what I was thinking |
| 19:09.09 | Katty | i'm sure gravity has something to do with bending it |
| 19:09.28 | Katty | i'd liek to see gravity bend light |
| 19:09.45 | wpbrown504 | Black holes and stars bend the fabric of space time.. so anything can be possible |
| 19:09.50 | [TK]D-Fender | Katty: It does, even within the scope of Earth's gravity |
| 19:09.55 | Qwell | Katty: Hawking radiation |
| 19:10.11 | Katty | googles hawking radiation |
| 19:10.13 | *** part/#asterisk martyn-dev (i=be18869a@gateway/web/ajax/mibbit.com/x-20ea848adc70179f) |
| 19:11.41 | seanbright | Qwell: have you gotten your ecigareet yet? |
| 19:11.45 | seanbright | wow... |
| 19:11.49 | Qwell | seanbright: arrived in the mail this morning |
| 19:11.50 | seanbright | cigarette |
| 19:11.54 | wpbrown504 | I have a question for you guru's. 1 have 1 asterisk box with a pri inbound and I direct my fax lines across a fxs card to the fax machines. I am working on another box which has a 4 port fxo card in it. Do you think if I provisioned a line on the fxs card and fed a line to the fxo card on the other box would the fxo channel be able to pick up? |
| 19:11.57 | Qwell | we'll see how it goes when I get home |
| 19:12.11 | seanbright | excellent. keep me posted. |
| 19:12.32 | *** join/#asterisk DrAk0 (n=luisjose@nelug/coreteam/luisjose) |
| 19:12.44 | mmlj4 | wpbrown504: that's affirmative |
| 19:12.46 | Katty | i wonder if that means white holes are no longer a theory |
| 19:13.11 | wpbrown504 | thanks mmlj4. I am about to go test a couple of configs. |
| 19:14.27 | mmlj4 | wpbrown504: but why tie up a couple of POTS channels? use sip or iax to tie the two boxen together |
| 19:14.45 | *** join/#asterisk Urthwhyte (n=urthwhyt@0x5da320aa.cpe.ge-1-1-0-1101.oebrnqu2.customer.tele.dk) |
| 19:14.46 | Erol_ | eppigy: what did he mean by 'Use APs, not routers, so the phone keeps a single IP'? |
| 19:14.50 | *** join/#asterisk Hazuki (n=Hazuki@cpe-72-229-137-30.nyc.res.rr.com) |
| 19:15.19 | eppigy | Erol_: he just mean dont use a bunch of wireless routers |
| 19:15.20 | Katty | i wonder what would happen if a photon somehow got split in a blackhole |
| 19:15.20 | Hazuki | Question: I'm completely new to Asterisk, and reading the O'Reilley book on it. My boss wants a small office network |
| 19:15.22 | wpbrown504 | mmlj4: it is just a test. It is hurricaine season in nawleans. I am sending a box to memphis and everything is analog. Just trying to do a little bit of hurricaine prepardness. |
| 19:15.28 | Hazuki | one line in, multiplexed among 10 real estate agents |
| 19:15.30 | Katty | and half of it went spewing out the back of the white hole |
| 19:15.36 | Katty | it would already exist in the other dimension... |
| 19:15.38 | Urthwhyte | Hazuki: Does it need DID and Outbound trunking, or internal only? |
| 19:15.39 | mmlj4 | um, you're in floody new orleans? |
| 19:15.41 | Hazuki | what would I need for this? I won't need 10 FXS and one FXO card will I? |
| 19:15.52 | Katty | does that mean dark matter would suddenly popup to compensate for it? |
| 19:15.57 | Hazuki | Urthwhyte, I think he's not sure how this works *rolls her eyes* |
| 19:15.59 | eppigy | n some alternate reality i am not dave |
| 19:16.01 | eppigy | in |
| 19:16.06 | Erol_ | eppigy: i just have a single network in my LAN, so I guess this isnt a problem for me, right? |
| 19:16.07 | wpbrown504 | Yup. Have a office in memphis. Just getting a box set up there so I can call forward if needs be. |
| 19:16.08 | Urthwhyte | Hazuki: What're you currently using for voice? |
| 19:16.11 | Hazuki | He wants to be able to multiplex one line from a telco among 10 agents inside the building |
| 19:16.11 | Katty | and what would happen to its other engtangled half |
| 19:16.16 | Katty | because... they're connected. always. |
| 19:16.17 | eppigy | Erol_: that is correct |
| 19:16.19 | Hazuki | as of now nothing |
| 19:16.25 | Katty | it would experience the same effect. |
| 19:16.32 | Hazuki | skype intra-office and a plain phone for handling calls from the outside |
| 19:16.34 | mmlj4 | to plagiarize that dog on foghorn leghorn... |
| 19:16.47 | wpbrown504 | lol |
| 19:16.51 | Urthwhyte | Katty: I finally got RTP working correctly, was a weird bug with my router and * config |
| 19:17.01 | Hazuki | what he wants to do it make it so someone can call the line that goes into the Asterisk server plus and extension, and the server will redirect to an agent's headset depending on extension |
| 19:17.08 | Urthwhyte | Hazuki: Mind if I PM you, a bit busy in here ATM |
| 19:17.08 | mmlj4 | well shet ma mouth and stuff me with crawfish, I'm from the flood too! |
| 19:17.10 | Katty | Urthwhyte: oh neat. next you can work on entangled bank transactions. |
| 19:17.14 | Hazuki | Urthwhyte, yes, thank you very much :) |
| 19:17.20 | Katty | i should invent it. |
| 19:17.21 | Katty | i'd be rich. |
| 19:17.22 | Erol_ | eppigy: do people with subnets need proxy mobile IP and IP Mobile features? |
| 19:17.24 | wpbrown504 | lol |
| 19:17.33 | mmlj4 | wpbrown504: seriously, I'm in metairie |
| 19:17.34 | [TK]D-Fender | mmlj4: ANTI-DELUVIAN |
| 19:17.36 | Urthwhyte | Entangled as in QMT? |
| 19:17.42 | wpbrown504 | cool im in Mandeville |
| 19:17.48 | mmlj4 | [TK]D-Fender: AND PROUD OF IT |
| 19:18.03 | eppigy | Erol_: brother, that I do not know |
| 19:18.06 | Katty | oh crap |
| 19:18.10 | Katty | someone already patented the idea |
| 19:18.18 | Urthwhyte | I've got an interesting bug though. I've currently three extensions configured, my own, one for a friend, and a testing extension |
| 19:18.20 | Erol_ | eppigy: thanks |
| 19:18.21 | Katty | AND posted an aritcle in the new york times |
| 19:18.36 | wpbrown504 | I am about a 30 mile swim from you mmlj4 |
| 19:18.44 | mmlj4 | 24 miles |
| 19:18.48 | Katty | maybe i can patent space folding travel |
| 19:18.55 | dni | [TK]D-Fender, the other day you posted a link of a company that offered free DID's can you link me again ? |
| 19:19.01 | mmlj4 | 24 of the most boring ever paved miles |
| 19:19.02 | Katty | that would be hard considering we're not made of photons |
| 19:19.04 | Katty | nevermind :< |
| 19:19.06 | Urthwhyte | Using a hardphone for my own extension, and a softphone for the testing, extension #2 (the friend) can call 100, and 100 can call the test, but the test won't connect to 100 |
| 19:19.08 | wpbrown504 | lmao |
| 19:19.15 | Urthwhyte | I get a "Not in Service" message |
| 19:19.17 | [TK]D-Fender | dni: www.ipkall.com |
| 19:19.19 | dni | thanks |
| 19:19.27 | Katty | eppigy: dude |
| 19:19.31 | Katty | eppigy: you could call MARS in seconds! |
| 19:19.40 | mmlj4 | wpbrown504: you heard of nolug.org, our users group? |
| 19:19.43 | Katty | eppigy: quantum entangled phone call on photon backs! |
| 19:19.53 | wpbrown504 | No I have not. |
| 19:19.59 | wpbrown504 | I will check it out. |
| 19:20.20 | mmlj4 | there's a few linux guys on the northshore |
| 19:20.32 | mmlj4 | how much do you do with asterisk? just starting? |
| 19:20.38 | Carlos_PHX | Erol_: Sorry, got called away. |
| 19:20.52 | Carlos_PHX | Keep all your channels 3 apart if the APs are in range of each other. |
| 19:21.02 | Katty | eppigy: asterisk will be soooo obsolete |
| 19:21.14 | mmlj4 | i gotta go to mandeville in a day or so... if you're not a stalker, let's do lunch |
| 19:21.24 | Katty | eppigy: chan_photon.so |
| 19:21.27 | haib | Does anyone know if you can send smdi over serial? It looks in smdi.conf that it only receives it. Is that true or is it send and receive? |
| 19:21.34 | ipstatic | [TK]D-Fender: ok, i think I have the whole call now: http://pastebin.com/m2f58a38e |
| 19:21.34 | Erol_ | Carlos_PHX: but how could they not be in range of each other? |
| 19:21.39 | wpbrown504 | I know a little bit about asterisk. Haven't messed with it since 2005 before katrina. |
| 19:21.55 | eppigy | Katty: haha that would be awesome |
| 19:21.58 | Katty | wpbrown504: it's changed a lot |
| 19:22.01 | Erol_ | Carlos_PHX: they make spherical broadcast |
| 19:22.19 | wpbrown504 | Aye, i am laboring though the changes as we speak Katty. |
| 19:22.28 | wpbrown504 | Having fun though. |
| 19:22.37 | Katty | wpbrown504: i think it's much more stable and lots of new fun stuff. |
| 19:22.45 | Katty | well. as fun as a ball of foil can be. |
| 19:22.49 | [TK]D-Fender | ipstatic: Scheduling destruction of SIP dialog '482ce706488d011d5daf0eb868b6e656@72.15.246.164' in 6400 ms (Method: INVITE) -- 6.4 seconds till hangup? odd |
| 19:22.57 | mmlj4 | that's chan_scotty, no? |
| 19:23.03 | Erol_ | Carlos_PHX: you there? |
| 19:23.07 | Carlos_PHX | Yes |
| 19:23.28 | Erol_ | Carlos_PHX: it is impossible to get them out of their range |
| 19:23.38 | Carlos_PHX | How many do you have? |
| 19:23.56 | Erol_ | Carlos_PHX: in 3 sepetae buildings 12 |
| 19:23.58 | [TK]D-Fender | steals mmlj4's sub-space transceiver |
| 19:24.04 | ipstatic | [TK]D-Fender: This only happens when the box starts to make a lot of outgoing calls |
| 19:24.15 | Carlos_PHX | You're saying all those APs can see each other? Then you have too many. |
| 19:24.18 | *** join/#asterisk voxter (n=voxter@76.77.95.2) |
| 19:24.44 | Carlos_PHX | If you have a phone that sees two APs on the same channel, you're going to have issues. Even one channel away will be problematic. |
| 19:24.44 | ipstatic | [TK]D-Fender: basically an application makes alot of call files and moves them to the outgoing dir, and the customer is complaining of delayed playback, or even some numbers not being called |
| 19:24.48 | Erol_ | Carlos_PHX: no but each see at least 2 of them |
| 19:25.02 | Carlos_PHX | That's fine, just make sure they are three channels away. |
| 19:25.19 | Katty | cars should run off photon gas. |
| 19:25.31 | Erol_ | Carlos_PHX: 3 channels away? |
| 19:25.35 | Katty | it would be like a limitless source of energy to run cars. |
| 19:25.40 | Katty | and no by-product |
| 19:25.54 | Carlos_PHX | Erol_: You are aware that APs have 11 channels? |
| 19:26.02 | Erol_ | Carlos_PHX: yes |
| 19:26.22 | eppigy | 1,6,11 are non overlapping |
| 19:26.23 | Carlos_PHX | So any AP within range of another AP should be on a channel that is at least three channels away from the others. |
| 19:26.27 | eppigy | strategize accordingly |
| 19:26.43 | Katty | eppigy: let's build a car that works on photon gas. |
| 19:26.46 | Carlos_PHX | And I re-use them closer with the farther APs that might see each other but are farther away. |
| 19:26.53 | eppigy | Katty: im down |
| 19:27.07 | Katty | Qwell can help. |
| 19:27.09 | eppigy | quantum VROOOM |
| 19:27.14 | Erol_ | Carlos_PHX: for example if one is in channel 1, the other should be 4? |
| 19:28.05 | eppigy | IT WOULD HAVE TO BE AT LEAT CHANNEL 5 |
| 19:28.08 | eppigy | whoops |
| 19:28.12 | eppigy | cap lock |
| 19:28.27 | Erol_ | i felt like a dumb, i couldnt get it |
| 19:28.44 | Erol_ | what does 3 channels away mean? |
| 19:28.51 | eppigy | oly cow |
| 19:28.57 | Carlos_PHX | Erol_: Ideally more, but I have found that a difference of 3 works well. |
| 19:29.04 | Katty | well dangit |
| 19:29.15 | Katty | someone already has that patented too |
| 19:29.18 | Carlos_PHX | The details depend on the shape of your space and materials in the building. |
| 19:29.38 | Erol_ | Carlos_PHX: so i dont need to set them up with non overlapping channels? |
| 19:29.44 | Carlos_PHX | But I try to use 1/6/11 for the three closest, then go in-between for others, etc. |
| 19:29.53 | whys | Is it possible to put speed dial numbers in the 7960 phone configs (tftp) |
| 19:30.16 | bmoraca | Erol_: your "channel" is a modulation of + or - a few mhz from 2400mhz. Because the width of a "channel" is greater than the difference between the starting frequencies of each channel, you can get interference if two devices using close channels are too close together. |
| 19:30.58 | Erol_ | bmoraca: oh ok |
| 19:31.10 | *** join/#asterisk MooingLemur (n=troy@unaffiliated/mooinglemur) |
| 19:31.13 | bmoraca | here's a graphical representation: http://en.wikipedia.org/wiki/File:2.4_GHz_Wi-Fi_channels_(802.11b,g_WLAN).png |
| 19:31.24 | eppigy | that is where the term bandwidth comes from kids |
| 19:31.34 | eppigy | band-width |
| 19:31.36 | eppigy | GET IT |
| 19:32.06 | *** join/#asterisk Urthwhyte_ (n=urthwhyt@0x5da320aa.cpe.ge-1-1-0-1101.oebrnqu2.customer.tele.dk) |
| 19:32.36 | Erol_ | Carlos_PHX: and you say using 3 channel different doesn not interfere as if you are using 1-6-11 |
| 19:32.42 | Erol_ | ah shit.. |
| 19:32.47 | Carlos_PHX | Interesting graphical representation, pretty good. |
| 19:32.50 | mmlj4 | 1, 6 and 11... the only non-overlapping channels |
| 19:33.16 | Erol_ | 1-6-11 is 1-6-11 because the difference is at max for 11 channels, ahh |
| 19:33.18 | Carlos_PHX | What I am saying is that if you have an AP that is barely in range at one, and you have a nearby AP at 3, then the far AP won't interfere. |
| 19:33.18 | bmoraca | 1, 6, and 11 are the only non-overlapping channels. depending on the proximity of the APs and the people connecting to them, they could be 1 channel away and not have interference. |
| 19:33.20 | mmlj4 | but WHY would they come up with overlapping channels? sense make, it does not |
| 19:33.35 | Carlos_PHX | What I am saying is that this is complex and if you don't understand radio theory at all, you may not get this. |
| 19:34.08 | Carlos_PHX | mmlj4: If you want to talk theory, ask the Wi-fi committe. |
| 19:34.18 | Erol_ | Carlos_PHX: one sec.. |
| 19:34.21 | mmlj4 | apparently someone needs to talk theory to them |
| 19:34.38 | bmoraca | Erol_: this is why there are people paid to do nothing but wireless surveys to design a proper layout |
| 19:34.51 | eppigy | Katty: we will buy the rights from them |
| 19:35.00 | *** join/#asterisk Deeewayne (n=dwayne@95.104.105.2) |
| 19:35.00 | *** mode/#asterisk [+o Deeewayne] by ChanServ |
| 19:35.25 | Erol_ | Carlos_PHX: what i didint get sure is, you mean 3 difference as like channel 1 for an ap and 1+3=4 for the other ap? |
| 19:35.29 | Carlos_PHX | Reminds me I gotta go see if my box of Meraki APs arrived for my job tomorrow. |
| 19:35.45 | Carlos_PHX | Erol_: Yes |
| 19:35.48 | Erol_ | by the way this graph shows 14channes |
| 19:35.59 | Carlos_PHX | There are 14, but only 11 allowed in the US. |
| 19:35.59 | bmoraca | Erol_: in EU and Asia, they have 14 channels |
| 19:36.13 | Erol_ | ?!??! |
| 19:36.17 | Erol_ | really? |
| 19:36.21 | bmoraca | yes |
| 19:36.25 | Erol_ | lol |
| 19:36.28 | kc8pxy | Erol_: really |
| 19:36.31 | eppigy | DAMN YOU FCC |
| 19:36.35 | Erol_ | then it should be bad for health |
| 19:36.55 | Erol_ | thats like pumping cigarettes, lol |
| 19:37.02 | Carlos_PHX | Erol_: Look at that graph. Notice the overlap and the fall-off in signal? So what you want is for distance + channel separation to equal no interference. |
| 19:37.13 | Carlos_PHX | If distance is low, channel separation must be large. |
| 19:37.21 | Carlos_PHX | Large distance allows for less separation. |
| 19:37.48 | Carlos_PHX | My experience is that the closest APs go on 1/6/11 and then the next set on something like 3/9, etc. |
| 19:38.06 | bmoraca | also position of connected devices and your antenna type/gain can affect it as well |
| 19:38.09 | kc8pxy | Erol_: IMHO, all wifi is bad for your health. 2.4ghz is very close to what is used in microwaves. the only major difference is power dissipation. with everyone and their brother using wifi, that mitigates the difference :) |
| 19:38.09 | Carlos_PHX | It varies greatly by the space. |
| 19:38.20 | Carlos_PHX | Yes, antennas vary this hugely. |
| 19:38.49 | Carlos_PHX | Yes, a 70mw AP is just like a 1200 watt microwave. |
| 19:39.03 | [TK]D-Fender | only different! |
| 19:39.47 | Carlos_PHX | is going to put a bag of popcorn on his AP. |
| 19:40.08 | kc8pxy | dbi gain changes the ERP. is very simple to create :) |
| 19:40.24 | *** join/#asterisk erol___ (n=x@88.235.144.246) |
| 19:40.37 | erol___ | out of battery, lol |
| 19:40.55 | erol___ | Carlos_PHX: yeah its quiet simple, i thought it was more complicated |
| 19:41.00 | kc8pxy | as i said in my original comment IMHO. |
| 19:41.05 | erol___ | Carlos_PHX: i thought 1-6-11 is some kind special |
| 19:41.08 | erol___ | lol |
| 19:41.15 | *** join/#asterisk SuPrSluG (n=SuPrSluG@firewall-a.buf.ny.i-evolve.net) |
| 19:42.28 | bmoraca | networking is fun...but i hate wireless...people expect it to be a cheap alternative to everything, yet it often causes more problems than it helps...at least in my experience |
| 19:43.01 | leifmadsen | bmoraca: amen |
| 19:43.09 | *** join/#asterisk netvulture (n=root@dns1.netvulture.com) |
| 19:43.39 | leifmadsen | I used to be a fully wireless office, until there were too many APs in my area (downtown Toronto), and SSH got very laggy, so I switched to an entirely cabled network except for cell phone (which has wifi+SIP on it) |
| 19:43.48 | leifmadsen | and wifi for guests who visit |
| 19:43.57 | whys | Is it possible to put speed dial numbers in the 7960 phone configs (tftp) |
| 19:43.59 | whys | anyone? |
| 19:43.59 | netvulture | got a problem - need my transcoder tc400b to work in 1.2 on 2.6.33.1 kernel |
| 19:44.42 | *** join/#asterisk telecos (n=sergio@19.166.219.87.dynamic.jazztel.es) |
| 19:44.55 | netvulture | i can get it to compile by adding a NULL argument to the device_create - module loads and finds card - but asterisk show transcoders shows nothing |
| 19:45.29 | kc8pxy | is guessing he's the only one putting 16dbi antennas on a 100mW WAP. |
| 19:45.31 | *** join/#asterisk theHub (n=theHub@69.177.93.21) |
| 19:45.56 | Carlos_PHX | leifmadsen: The right solution is 802.11a or n on the a frequencies. |
| 19:46.20 | bmoraca | kc8pxy: I put 28dbi dish antennae on 100mW radios...7 mile wireless bridge |
| 19:48.01 | ipstatic | [TK]D-Fender: Do you know of anyone having issues with Broadvox? |
| 19:48.27 | [TK]D-Fender | ipstatic: nope |
| 19:50.03 | kc8pxy | bmoraca: cool.. similar idea for me, but mine is a large-event area hotspot. omni. |
| 19:50.16 | Urthwhyte_ | Any one know of a ten port ATA? |
| 19:50.27 | Urthwhyte_ | with individual configuration |
| 19:50.36 | *** join/#asterisk esaym (n=user@cpe-70-120-89-6.satx.res.rr.com) |
| 19:50.55 | bmoraca | kc8pxy: haven't done much with omnis. i almost got to set up a metro wifi for a small city...papers were signed and money budgetted...but public outcry caused them to nix the project :P |
| 19:51.03 | erol___ | lastly, which would you guys suggest? wi-fi ip phones or ip phones with GAP and using GAP Access points . |
| 19:51.08 | bmoraca | Urthwhyte_: mediatrix? |
| 19:51.20 | *** join/#asterisk wierdo (n=jimmy@wifi-traf5.networx-bg.com) |
| 19:51.43 | ipstatic | [TK]D-Fender: Is there something I can change to fix that error, or is that on the carrier side? |
| 19:51.52 | Urthwhyte_ | I'm assuming you're not referring to Catholic theologoy |
| 19:51.53 | kc8pxy | bmoraca: outcry?? not much of a metro area? who doesn't want wifi nowadays? |
| 19:51.55 | [TK]D-Fender | ipstatic: dunno |
| 19:52.33 | Urthwhyte_ | kc8pxy: 3G or WiMAX are far better suited than WiFi |
| 19:53.22 | bmoraca | kc8pxy: some of the local businesses offer free wifi as an attractor for customers and didn't want the city wifi eating in to that...and it was a very expensive product with money that could have and should have been spent on other things. |
| 19:53.29 | kc8pxy | Urthwhyte_: do they have sip-capable 3G or WIMZX phones, though? |
| 19:53.59 | bmoraca | erol___: neither. use an ATA and a cheap analog cordless phone...DECT or 900mhz, your choice |
| 19:54.15 | Urthwhyte_ | kc8pxy: Any smartphone should do |
| 19:55.29 | erol___ | bmoraca: uhm, what is ATA? |
| 19:55.43 | bmoraca | analog telephony adapter...or sip-to-analog gateway |
| 19:56.22 | bmoraca | Urthwhyte_: http://www.voiplink.com/Mediatrix_Analog_Telephone_Adapters_s/55.htm |
| 19:56.27 | erol___ | bmoraca: oh got it |
| 19:56.27 | bmoraca | er |
| 19:56.31 | bmoraca | wrong link |
| 19:56.46 | bmoraca | Urthwhyte_: http://www.voiplink.com/Mediatrix_IP_Gateways_s/67.htm |
| 19:56.46 | erol___ | bmoraca: yeah i know that |
| 19:59.22 | erol___ | bmoraca: for people who need digital phones? |
| 20:00.07 | bmoraca | erol___: what do they need that they can't do with an analog phone? wifi phones and the sip phones with cordless handsets do not work very well...i'm speaking from experience here |
| 20:00.52 | haib | Does anyone know if you can send smdi over serial? It looks in smdi.conf that it only receives it. Is that true or is it send and receive? |
| 20:01.22 | erol___ | bmoraca: thanks man, i really appriciate that. and did you ever use GAP phones? |
| 20:01.40 | mweichert | any sox experts here? I'm having trouble converting a .wav file to .ulaw: sox hello.wav -r 8000 -t ul hello.ulaw |
| 20:01.53 | mweichert | the result is distorted |
| 20:02.25 | *** join/#asterisk matsk (n=matkar@c-7184e253.174-6-64736c10.cust.bredbandsbolaget.se) |
| 20:03.13 | bmoraca | erol___: if you're talking about those multi-phone base stations and things like that, you could probably use them, but I'm not familiar with them enough to say that each phone could be a different extension...i don't believe they could be |
| 20:03.14 | Hazuki | If I wanted to connect a business-class SIP line from outside for my Astrisk server to use, would I need an FXO card? |
| 20:03.40 | bmoraca | Hazuki: no, but it'd probably be a good idea for failover purposes |
| 20:03.50 | [TK]D-Fender | Hazuki: No. |
| 20:03.59 | Carlos_PHX | What is a "SIP line?" You don't need anything to connect two SIP servers together. |
| 20:04.02 | Hazuki | bmoraca, sorry for the dumb question...I am so overwhelmed now >< |
| 20:04.10 | Hazuki | I've never dealt with telephony before |
| 20:04.14 | Carlos_PHX | Unless you have a provider doing a SIP to POTS handoff. |
| 20:04.38 | rob0 | ~fxo |
| 20:04.38 | infobot | foreign exchange office - type of port you need to connect a POTS (Plain Old Telephone Service) line from your telco to a pbx http://www.digium.com/index.php?menu=fxsvfxo |
| 20:05.08 | Hazuki | okay, would this work? Outside POTS line --> FXO card --> Asterisk server --> ATA --> 10 x POTS phones inside office? |
| 20:05.16 | erol___ | bmoraca: there are some special access points which use GAP and change it to ip |
| 20:05.30 | *** join/#asterisk grx0 (n=root@70.94.220.114) |
| 20:05.42 | jplank | grrrrr polycom's new vvx1500 is driving me crazy |
| 20:05.46 | Hazuki | basically he wants to buy one business line, configure 10 extensions on it through Asterisk, and have it so whenever that number + an extension is called, the phone at the desk of the relevant agent rings |
| 20:05.54 | erol___ | bmoraca: gap is different than wi-fi |
| 20:05.58 | jplank | the coolest video phone I've ever used, but it doesn't work! |
| 20:06.15 | *** part/#asterisk matsk (n=matkar@c-7184e253.174-6-64736c10.cust.bredbandsbolaget.se) |
| 20:06.15 | jplank | its the most expensive digital picture frame I have |
| 20:06.40 | Urthwhyte_ | jplank: If it doesn't work how cool of a vid phone can it be? |
| 20:06.56 | *** join/#asterisk grx0 (n=root@70.94.220.114) |
| 20:07.03 | jplank | nice looking phone, a lot of cool options, etc etc |
| 20:07.18 | jplank | I just don't understand why it doesn't work |
| 20:07.20 | jplank | voice works |
| 20:07.31 | jplank | it looks like its negotiating the video codec correctly |
| 20:07.39 | jplank | but no video |
| 20:07.54 | bpgoldsb | Anyone here familiar with CDR who can help me understand it a little better? All my cdr records are logging a dst of 's', while NoOp($CDR{dst}); shows the correct exten. |
| 20:07.55 | [TK]D-Fender | Hazuki: thats fine |
| 20:08.33 | Hazuki | [TK]D-Fender, okay, how hard will that be to configure? Are dialplans very hard? I read about 200 pages of the O'Reilley book but the diaplan stuff didn't stick |
| 20:08.51 | Urthwhyte_ | jplank: Are you using two of the same phone? |
| 20:08.59 | Urthwhyte_ | jplank: Or connecting to a softphone of some sort? |
| 20:09.04 | [TK]D-Fender | Hazuki: depends on you. |
| 20:09.20 | [TK]D-Fender | Hazuki: Here's some "inspiration" for you : |
| 20:09.22 | [TK]D-Fender | ~jerjerguide |
| 20:09.23 | infobot | [~jerjerguide ] Jeremy McNamara's quick sample guide to setting up * : http://www.jeremy-mcnamara.com/2007/02/26/how-to-configure-asterisk-your-first-installation/ |
| 20:09.33 | Hazuki | [TK]D-Fender, I've never done or seen any of this before but I learn very, very fast and am extremely motivated |
| 20:09.37 | Urthwhyte_ | [TK]D-Fender: 404 |
| 20:09.41 | jplank | Urthwhyte_: I'm trying to call from the VVX to Polycom's HDX |
| 20:09.46 | Urthwhyte_ | Hazuki: You're not applying for a job here :p |
| 20:09.53 | Hazuki | Urthwhyte_, I alreayd have one |
| 20:09.54 | bmoraca | erol___: I realize that. my concern is that every gateway I've used that doesn't go to analog, such as the Snom M3 which is a SIP-to-DECT gateway as well, isn't very good. I like the Philips CD1 phone. |
| 20:09.55 | Hazuki | sort of |
| 20:09.56 | grx0 | lol |
| 20:09.58 | jplank | We ordered two VVX's but one is back ordered |
| 20:10.02 | [TK]D-Fender | http://74.125.95.132/search?q=cache:x-SVjZ-02u8J:www.jeremy-mcnamara.com/2007/02/26/how-to-configure-asterisk-your-first-installation/+http://www.jeremy-mcnamara.com/2007/02/26/how-to-configure-asterisk-your-first-installation/&cd=1&hl=en&ct=clnk&gl=ca |
| 20:10.09 | Hazuki | I'm on a per-diem basis for the 3rd largest real estate broker in Manhattan as a tech support girl |
| 20:10.20 | Hazuki | he's been talking about partnering with me for real estate too |
| 20:10.25 | Hazuki | and even offered to put me through the school for it |
| 20:10.34 | Hazuki | but I need to make the best impression I can |
| 20:10.46 | *** join/#asterisk ukine (n=ukine@6532200hfc143.tampabay.res.rr.com) |
| 20:10.49 | Hazuki | which means doing as much of what he dreams of as I can, even if he doesn't know what work goes into it |
| 20:10.49 | [TK]D-Fender | Hazuki: I'm already scared for your clients... |
| 20:10.56 | Hazuki | Why? |
| 20:11.16 | Urthwhyte_ | Hazuki: How long have you been in IT? |
| 20:11.18 | [TK]D-Fender | Hazuki: the thought of you selling * and not having a basic understanding of it |
| 20:11.33 | Hazuki | [TK]D-Fender, why would I sell Asterisk? he's in real estate |
| 20:11.44 | Hazuki | and Urthwhyte_ I've been freelancing since age 15...so, 8 years |
| 20:12.34 | Carlos_PHX | You don't deploy Asterisk for your first time in a live situation. |
| 20:12.51 | Carlos_PHX | Build one with a random SIP account and a couple of phones and spend a few weeks learning. |
| 20:13.17 | Hazuki | well, yes, we're going to test it first >>; |
| 20:13.19 | Carlos_PHX | Unless you plan to do MUCH more with Asterisk, I'd use one of the GUI versions instead of Asterisk itself. |
| 20:13.26 | erol___ | bmoraca: thanks for the info man |
| 20:13.39 | erol___ | thanks all guys for the info |
| 20:13.41 | Urthwhyte_ | Trixbox or A&H are both decent |
| 20:13.43 | Urthwhyte_ | Trixbox or A@H are both decent |
| 20:13.51 | bmoraca | i got lucky...our telco gave me 30 days of PRI service to test with for my first deployment |
| 20:13.54 | Urthwhyte_ | double post |
| 20:14.07 | rob0 | Indecent! |
| 20:14.15 | Hazuki | Carlos_PHX, there's a GUI version? |
| 20:14.24 | Hazuki | I know about AstLinux but that's for embedded things.. |
| 20:14.33 | Urthwhyte_ | <PROTECTED> |
| 20:14.45 | bmoraca | Hazuki: I like Elastix |
| 20:14.52 | *** part/#asterisk erol___ (n=x@88.235.144.246) |
| 20:14.55 | Hazuki | memorizes the names |
| 20:14.56 | Qwell | Hazuki: AsteriskNOW. |
| 20:15.02 | Qwell | clearly the better option |
| 20:15.11 | rob0 | A ster is know. |
| 20:15.25 | Qwell | rob0: nice |
| 20:15.28 | [TK]D-Fender | Urthwhyte_ : Asterisk@Home <-- got to be kidding me.. |
| 20:15.29 | Hazuki | ...wow. This looks much easier |
| 20:15.41 | Qwell | Hazuki: trust me on this - steer clear of trixbox. |
| 20:15.50 | Hazuki | nodnod and pays attention |
| 20:16.25 | Urthwhyte_ | [TK]D-Fender: Haven't played with it my self, but for 10 extensions I can't see any major issues |
| 20:16.42 | Urthwhyte_ | Qwell: What bad experience did you have with trixbox?\ |
| 20:16.48 | Qwell | Urthwhyte_: all of it. |
| 20:16.58 | grx0 | Trixbox ate my puppy :( |
| 20:17.04 | Qwell | Urthwhyte_: and, as for A@H being a joke... it hasn't existed in 3+ years. |
| 20:17.25 | RypPn | lol |
| 20:17.28 | blkry | <PROTECTED> |
| 20:17.34 | Hazuki | okay...so once more: if he buys a business-class POTS line, we need an FXO card from, e.g., Digium to plug it into right? |
| 20:17.39 | Qwell | blkry: congrats on seeing the light. |
| 20:17.40 | *** join/#asterisk Urthwhyte (n=urthwhyt@0x5da320aa.cpe.ge-1-1-0-1101.oebrnqu2.customer.tele.dk) |
| 20:17.41 | blkry | its kind of like training wheels |
| 20:17.44 | Qwell | Hazuki: yes |
| 20:17.55 | Urthwhyte | Woops, got Now and @Home confused |
| 20:18.07 | Urthwhyte | In my defense, I haven't had my coffee yet tonight |
| 20:18.07 | Qwell | blkry: shoddy ones that keep getting too loose and falling off? |
| 20:18.07 | Hazuki | Okay, so it goes POTS Line --> FXO --> Asterisk --> ATA --> LOLPhones |
| 20:18.16 | Qwell | LOLPhones? I like you. |
| 20:18.21 | Hazuki | hehe, thanks ^^ |
| 20:18.34 | [TK]D-Fender | Analog phone... ew. |
| 20:18.45 | haib | Hence the lolphone |
| 20:19.58 | haib | Can you propagate smdi over serial with asterisk, or can it only receive? Looking at the smdi.conf it seems like it can only receive, it can't send smdi over serial, correct? |
| 20:20.58 | grx0 | Hazuki: |
| 20:21.28 | grx0 | whoops I meant to type a msg with that.... whys he getting a 'business class pots line' lol... sip provider seems much cheaper...no high speed interwebs? |
| 20:21.41 | Urthwhyte | grx0: That's what I said |
| 20:21.56 | Qwell | ~cheap |
| 20:21.57 | infobot | i heard cheap is a bad idea. If you're setting up an Asterisk system, one of the wisest pieces of advice you can take is this: DON'T BE A CHEAPSKATE. |
| 20:21.57 | [TK]D-Fender | Single POTS line for 10 people? Ew. |
| 20:21.57 | Urthwhyte | not to mention SIP trunking is far easier for someone not familar with * |
| 20:22.18 | Qwell | Urthwhyte: ...until you have to deal with NAT |
| 20:22.22 | Qwell | and codecs, and... |
| 20:22.43 | grx0 | right...I setup a business with a magicjack account for 40$ and a little research (and praying they dont change up their code)....granted its not 10 lines but works better than a pots and is hella cheaper |
| 20:22.55 | Qwell | better? I beg to differ. |
| 20:23.07 | Qwell | How much is your phone service worth to you? |
| 20:23.08 | grx0 | magicjack is setup on the asterisk server that is...not that thats supported, but it works...and works well for them... |
| 20:23.08 | Urthwhyte | Qwell: I just spent 8 days dealing with SIP over NAT |
| 20:23.15 | Qwell | Can you afford it going down for a week? |
| 20:23.45 | [TK]D-Fender | [16:23]<Urthwhyte>Qwell: I just spent 8 days dealing with SIP over NAT <- should take 2 minutes |
| 20:23.47 | [TK]D-Fender | ~sipnat |
| 20:23.48 | infobot | [~sipnat] Quick guide on configuring * + SIP behind NAT : http://www.aocomputing.net/?p=3 , otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
| 20:23.57 | [TK]D-Fender | Read the &^$#ing guide people! |
| 20:24.14 | Urthwhyte | [TK]D-Fender: What should be and what actually works are often two different things |
| 20:24.36 | Urthwhyte | and I did read the guide, wasn't of too much help |
| 20:24.38 | [TK]D-Fender | Urthwhyte: Yes, 99% of the time with it falling under the category of PEBKAC |
| 20:25.06 | [TK]D-Fender | Urthwhyte: And what was your solution at the end of it all? |
| 20:25.24 | bmoraca | the only time i've ever had a problem with SIP and NAT was when the router a customer was using didn't support SIP Passthrough...in which case, it wasn't going to work regardless. |
| 20:25.25 | grx0 | almost everyone I've seen have nat issues is due to ports not being correctly opened. |
| 20:25.50 | grx0 | specificially RTP...everyone remembers to open 5060... |
| 20:25.51 | Urthwhyte | [TK]D-Fender: Replace the router with one that actually port forwards correctly and doesn't barf on RTP, even with the proper ports opened |
| 20:26.11 | grx0 | I didnt know that RTP induced vomiting :) |
| 20:26.19 | [TK]D-Fender | Urthwhyte: Yup, BS router can do that. Of course finding out should take about 5 minutes... |
| 20:26.49 | [TK]D-Fender | Urthwhyte: If replacing it takes the remaining 7 days, 23 hours, and 55 minutes...... well thats your problem :) |
| 20:27.00 | grx0 | hah |
| 20:27.24 | Urthwhyte | Meh, outside of not liking SIP, it was a decent router |
| 20:27.41 | Urthwhyte | Rather try and come up with a workaround then replace something that's been working well for several years |
| 20:28.23 | Qwell | was it a $40 router? |
| 20:28.32 | Qwell | refers back to the bots "cheap" comment. |
| 20:29.07 | Urthwhyte | It was a nice Cisco router from a dotcom that went under |
| 20:29.11 | grx0 | cheap is great when you dont expect much (eg if you're one of the people who freaks out when gmail goes down, cheap is not for you probably) |
| 20:29.25 | [TK]D-Fender | PIX = crap |
| 20:29.26 | [TK]D-Fender | :) |
| 20:29.31 | grx0 | maybe if they bought routers that werent hinky they'd have had more luck :) |
| 20:29.44 | [TK]D-Fender | checkout time, BBIAB |
| 20:30.02 | *** join/#asterisk thansen (n=thansen@c-76-27-110-194.hsd1.ut.comcast.net) |
| 20:30.24 | Urthwhyte | grx0: I don't think SIP was quite as prevalent around 2000. and since the company was Webvan, I don't think their problem was screwed up routers |
| 20:31.13 | *** join/#asterisk umpc (n=Justin@unaffiliated/umpc) |
| 20:31.13 | grx0 | Urthwhyte: lol probably not...wow 30 minute window of delivery, thats hot... |
| 20:31.25 | Urthwhyte | so much as massive overspending and complete failure at supply chain management |
| 20:31.45 | grx0 | MGCP was the shiz back then, I still enjoy dealing with all my avaya MCP junk...::gag:: |
| 20:32.19 | Urthwhyte | and 8 days != 8 days straight working on the problem |
| 20:33.03 | *** join/#asterisk cesar_CR (n=cesar@201.195.239.11) |
| 20:33.17 | grx0 | they spent around 92,000$ on chairs...wow... |
| 20:33.35 | Urthwhyte | but...but...but....Aerons are comfy ! |
| 20:33.40 | grx0 | which convienently is the same chair Im sitting in at the moment lol |
| 20:34.23 | haib | fahq! |
| 20:34.27 | haib | m/t |
| 20:34.29 | Urthwhyte | Either way it works now, and all it cost me was some time futzing around that I would've spent unproductively any ways |
| 20:35.06 | grx0 | Urthwhyte: fixing stuff thats broke is always an effective way to learn and learn quick lol |
| 20:35.07 | haib | Can you propagate smdi over serial with asterisk, or can it only receive? Looking at the smdi.conf it seems like it can only receive, it can't send smdi over serial, correct? |
| 20:35.37 | Urthwhyte | grx0: I usually start by getting things working and then giving them to a friend to break, remarkably effective |
| 20:35.43 | Katty | hmmmmmm. |
| 20:35.50 | Urthwhyte | Except for the time they rm -r / ... |
| 20:35.54 | Katty | dave |
| 20:36.13 | Katty | eppigy: ping |
| 20:36.56 | grx0 | Urthwhyte: well yeah or clients, when they insist on having access to make changes and of course break it...its call job security at that point though :) |
| 20:37.27 | Urthwhyte | Consulting: If you're not part of the problem, there's good money to be made in prolonging the solution |
| 20:37.30 | grx0 | its also called being able to use complete words, which I apparently can not. |
| 20:37.33 | grx0 | amen :) |
| 20:37.52 | grx0 | especially when that solution requires you to be in a costal state for the duration lol |
| 20:38.19 | Urthwhyte | My favorite was the client who insisted on deploying Sendmail in an all Windows+Outlook environment, when they already had a functioning Exchange server that met all of their needs. |
| 20:39.07 | eppigy | Katty: hiya |
| 20:39.41 | Urthwhyte | They wanted to pay me so they could have less features that was completely different from the rest of their environment. The owners response: "But some said it was free..." |
| 20:40.00 | Katty | eppigy: what's for dinners |
| 20:40.09 | grx0 | It is free...however supporting it, integrating it, and the cost of the software they already have...not so much lol |
| 20:40.17 | eppigy | man I dont know but i am starving |
| 20:40.31 | eppigy | im tryna think |
| 20:40.34 | Urthwhyte | grx0: I ended up turning them down, but it took a presentation and significant persuasion |
| 20:40.49 | *** join/#asterisk juanIMP (n=Juancho@200.71.41.22) |
| 20:41.29 | grx0 | Urthwhyte: that was probably wise :D |
| 20:41.51 | eppigy | Katty: i am almost thinking taco bell |
| 20:41.52 | grx0 | hates to think on an empty stomach |
| 20:42.03 | Qwell | oh, speaking of... |
| 20:42.07 | Qwell | Katty: how were the tacos?! |
| 20:42.16 | Katty | Qwell: delicious |
| 20:42.18 | Katty | Qwell: also gone |
| 20:42.22 | Qwell | :( |
| 20:42.27 | Katty | i know :< |
| 20:43.15 | Katty | and i didn't thaw anything out from the freezer either |
| 20:43.24 | eppigy | :< |
| 20:43.32 | eppigy | I really want a steak |
| 20:43.37 | Katty | IT WILL BE THE END OF THE WORLD |
| 20:43.43 | eppigy | MAKE THEM PAY |
| 20:43.44 | Katty | sobs. |
| 20:43.45 | grx0 | mmmmm steak |
| 20:43.57 | Katty | not in the mood for steak. |
| 20:44.04 | Katty | been getting baddd steaks here lately that suck. |
| 20:44.13 | eppigy | man a nice fresh ribeye |
| 20:44.14 | grx0 | has anyone here dealt with licensing realspeak for linux and have an idea of the pricing? |
| 20:44.22 | eppigy | cooked for 1 minute on each side |
| 20:44.31 | Katty | mooooooo |
| 20:44.32 | eppigy | with light salt and fresh ground blakc pepper |
| 20:44.38 | Katty | moooooooooooooooooooooo |
| 20:44.41 | eppigy | yesh |
| 20:44.47 | eppigy | I enjoy its pleas for mercy |
| 20:44.55 | Katty | tuna melt sounds pretty good, actually |
| 20:44.59 | eppigy | oh man |
| 20:45.03 | eppigy | that would be good too |
| 20:45.05 | eppigy | a tuna steak |
| 20:45.06 | eppigy | rare |
| 20:45.18 | Katty | broil it with some cheeseeeee |
| 20:45.27 | Katty | and lettuce shreddies on top |
| 20:45.32 | Katty | k, panic diverted. |
| 20:45.53 | *** join/#asterisk bbryant (n=ssddsasd@m445a36d0.tmodns.net) |
| 20:46.23 | grx0 | or any reasonable tts that is...I looked at cepstral, realspeak, and att ...suggestions? |
| 20:47.11 | *** join/#asterisk MrEd7777 (n=Mr@c-24-22-129-17.hsd1.wa.comcast.net) |
| 20:48.15 | MrEd7777 | where is a good place to find cheap asterisk compatible voice/data modems online? |
| 20:50.29 | pmhaddad-work | how would i set up a call with an outbound DID for an internal extension for just one call session (this is for 911) |
| 20:50.31 | Katty | modems? |
| 20:50.43 | Katty | pmhaddad-work: that does not parse. |
| 20:50.46 | Katty | pmhaddad-work: please try again. |
| 20:50.55 | pmhaddad-work | ok |
| 20:51.11 | bmoraca | Set(Callerid(number)=XXXXXXXXXX) |
| 20:52.03 | pmhaddad-work | so i have an extension. it dials 911. according this: http://www.asteriskvoipnews.com/asterisk_help/_e911_on_the_asterisk_open_source_pbx.html i need a unique DID for each extension to dial 911 for E911 to work properly. |
| 20:52.28 | pmhaddad-work | basically how do we give it that unique DID for each extension |
| 20:52.33 | pmhaddad-work | for that 911 call session |
| 20:52.41 | *** join/#asterisk bbryant (n=ssddsasd@m445a36d0.tmodns.net) |
| 20:52.47 | Katty | are you using a pri? |
| 20:53.24 | pmhaddad-work | Katty, well... no. |
| 20:53.36 | Katty | uhoh |
| 20:53.50 | pmhaddad-work | we need to do this over SIP |
| 20:53.51 | pmhaddad-work | well |
| 20:53.53 | pmhaddad-work | it mgiht be a pri |
| 20:54.05 | pmhaddad-work | is scrambling to that out |
| 20:54.07 | bmoraca | if it's over SIP, then it's not a PRI |
| 20:54.10 | pmhaddad-work | *figure |
| 20:54.12 | pmhaddad-work | ok |
| 20:54.14 | pmhaddad-work | tahts what i though |
| 20:54.15 | pmhaddad-work | but |
| 20:54.20 | pmhaddad-work | we do have 4 POTS lines |
| 20:54.28 | pmhaddad-work | they are mutually exclusive |
| 20:54.32 | bmoraca | if it's over SIP, then your SIP provider needs to support E911 |
| 20:54.33 | Katty | you can't change callerid information on pots lines |
| 20:54.39 | pmhaddad-work | (ie not rolled into a T1) |
| 20:54.46 | Katty | which means when you dial 911 through a pots line, all it will know is the number's 911 location |
| 20:54.54 | pmhaddad-work | that's what i figured |
| 20:55.09 | pmhaddad-work | so we need to do E911 through SIp without a PRI then |
| 20:55.13 | pmhaddad-work | possible? |
| 20:55.32 | bmoraca | only if your SIP provider supports it or you contract with a third-party E911 provider |
| 20:55.37 | MrEd7777 | Katty: according to a forum I found (http://www.voipuser.org/forum_topic_806.html) you can use intel v92 voice/data modems as FXO cards? so I'm trying to figure out where I can find the intel modems that work as FXOs... |
| 20:55.45 | pmhaddad-work | bmoraca, yeah our SIP provider supports it |
| 20:55.52 | Katty | pmhaddad-work: well that makes it easier |
| 20:55.59 | Katty | pmhaddad-work: just change your callid info in sip.conf then |
| 20:56.08 | Urthwhyte | Oh right, any one have suggestions for an IAX or SIP outbound provider? |
| 20:56.11 | Katty | pmhaddad-work: callerid=pmhadad-work <mydidnumber> |
| 20:56.28 | Katty | pmhaddad-work: and when you dial 911, do NOT overwrite the callerid |
| 20:56.31 | bmoraca | then you don't need to worry about it. your sip provider likely doesn't allow you to change your outbound callerid and thus will have the correct LIDB information already in their E911 database |
| 20:56.33 | Katty | pmhaddad-work: so that the did number shows. |
| 20:56.38 | pmhaddad-work | right, but we only want that CID when we call 911 |
| 20:57.10 | pmhaddad-work | hmm |
| 20:57.27 | Katty | you should probably set your primary number for outbound calls in general |
| 20:57.57 | pmhaddad-work | Katty, these "emergency DIDs" aren't going to be published |
| 20:58.10 | pmhaddad-work | like we are planning on getting one for every extension (i know its crazy) |
| 20:58.20 | bmoraca | pmhaddad-work: are all of your phones in the same location? |
| 20:58.34 | pmhaddad-work | but we need to be able to provide the PSAP with the room and ext that the call to 911 came from |
| 20:58.50 | grx0 | pmhaddad-work: no callerid before just throw in a SetCallerPres(prohib) before you dial... |
| 20:58.55 | pmhaddad-work | bmoraca, roughly? 4 different buildings in the same rough area |
| 20:58.56 | grx0 | that only works on pri or sip though |
| 20:59.07 | pmhaddad-work | (its a school) |
| 20:59.09 | Katty | MrEd7777: you can NOT use modems |
| 20:59.15 | bmoraca | pmhaddad-work: clear it with your provider that they allow you to specify multiple outbound callerids. |
| 20:59.20 | Katty | MrEd7777: put that thought out of your head immediately. |
| 20:59.45 | bmoraca | pmhaddad-work: if they do, then all you have to do is set the caller id in your 911 dialplan before the Dial() command is executed |
| 20:59.47 | Katty | MrEd7777: don't be cheap. |
| 20:59.50 | Katty | infobot: cheap |
| 20:59.51 | infobot | from memory, cheap is a bad idea. If you're setting up an Asterisk system, one of the wisest pieces of advice you can take is this: DON'T BE A CHEAPSKATE. |
| 20:59.58 | Katty | MrEd7777: ^- cheap is a very BAD idea |
| 21:00.14 | *** join/#asterisk ccesario (n=ccesario@201-42-147-249.dsl.telesp.net.br) |
| 21:00.19 | Katty | MrEd7777: that's like buying CHEAP eye surgery. |
| 21:00.54 | Katty | MrEd7777: if you want cheap, invest in tin cans and string. |
| 21:01.05 | Katty | MrEd7777: you will have the same quality. |
| 21:01.16 | haib | Hey I got decent range with some tin cans and string before. |
| 21:01.22 | Katty | MrEd7777: otherwise, please visit voipsupply they are very competative. |
| 21:01.25 | haib | don't knock it |
| 21:02.05 | MrEd7777 | Katty: haha ok ok... no cheap stuff I will just get the good stuff from digium or wherever |
| 21:02.28 | bmoraca | Sangoma ftw |
| 21:02.37 | Katty | sangoma is very nice. |
| 21:02.52 | Katty | and they have very helpful support staff. |
| 21:03.17 | pmhaddad-work | bmoraca, what do you mean exactly? |
| 21:03.33 | bmoraca | pmhaddad-work: are you familiar with Asterisk dialplan scripting at all? |
| 21:04.13 | pmhaddad-work | bmoraca, nvm i misunderstood what you were saying |
| 21:04.19 | pmhaddad-work | sorry |
| 21:06.00 | pmhaddad-work | ok, bmoraca when you say set the callerid before you dial do you mean the local extension and then our sip provider will translate that to the right DID somehow? basically, how does the sip provider know how to map it? |
| 21:06.07 | pmhaddad-work | i guess that's what i'm hung on |
| 21:06.35 | bmoraca | pmhaddad-work: if you're not familiar with Asterisk dialplan scripting, hire someone else to do it for you...or read the book |
| 21:07.25 | Katty | bmoraca: yeah i think that was obvious earlier. |
| 21:07.28 | Katty | bmoraca: he's not 4 (= |
| 21:08.11 | pmhaddad-work | :/ |
| 21:08.33 | bmoraca | pmhaddad-work: this isn't something I can teach over the interwebs. i told you what needs to be done |
| 21:08.34 | eppigy | i am 10.5 |
| 21:08.49 | Katty | orly |
| 21:09.07 | eppigy | negative |
| 21:09.11 | Katty | i am fishing. |
| 21:09.18 | pmhaddad-work | bmoraca, i can script it in, i'm just having trouble understanding it exactly |
| 21:09.30 | pmhaddad-work | like how it works at the provider level |
| 21:10.05 | bmoraca | pmhaddad-work: you don't need to care how it works at the provider level. all you need to do is set your caller ID before you dial 911 to the provider (if it needs to be said, this happens AFTER the user dials 911 on the quone) |
| 21:10.13 | bmoraca | s/quone/phone/ |
| 21:10.16 | pmhaddad-work | right |
| 21:10.18 | pmhaddad-work | ok |
| 21:10.37 | pmhaddad-work | well i'll get in touch with the provider and see if they let us do multiple outbound cids |
| 21:10.40 | pmhaddad-work | my thought is yes |
| 21:10.51 | Katty | that sounds smashing. |
| 21:11.05 | pmhaddad-work | thanks guys. sorry for being a little slow :/ |
| 21:11.14 | Katty | don't worry about bmoraca |
| 21:11.22 | Katty | some people are just rude. |
| 21:12.21 | bmoraca | hey now |
| 21:26.26 | *** join/#asterisk Flashtek (n=neil@flashtek-uk.com) |
| 21:26.30 | ambush276 | hey can anyone help me or point me in the right direction |
| 21:27.03 | Flashtek | I can't get dahdi to compile properly, is this the place to ask for pointers ? |
| 21:27.04 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 21:27.06 | ambush276 | i have my SIP registered in the settings, but i cant figure out how to dial out with it.. Basically i want extension "11" lets say to be my cell phone, how do i set up the SIP in extensions to do this |
| 21:27.26 | ambush276 | im not sure about TRUNKS, zaptel? |
| 21:27.31 | *** join/#asterisk [TK]D-Fender (n=joe@64.235.218.194) |
| 21:27.35 | *** join/#asterisk Joe_CoT (n=joecot@ubuntu/member/pdpc.bronze.joeterranova) |
| 21:28.27 | Joe_CoT | if I have two servers configured the same way, is there any problem with copying asterisk's Berkeley db from one server to another? I'm switching our asterisk from one machine to another |
| 21:31.53 | *** join/#asterisk tessier_ (n=treed@kernel-panic/sex-machines) |
| 21:32.34 | tessier_ | Hello all! Anyone using frame relay with asterisk/zaptel? I've got a frame circuit which is used for data and voice and some Micom units handling them. But they are ancient and we need to replace them and replace the Altigen phone system as well. |
| 21:33.10 | *** join/#asterisk mltlnx (n=mltlnx@cpe-24-193-11-14.nyc.res.rr.com) |
| 21:34.06 | *** join/#asterisk ccesario (n=ccesario@189.111.1.40) |
| 21:34.25 | mltlnx | is there anyway to receive a call on a pri or sip trunk that only rings never answers? |
| 21:34.53 | [TK]D-Fender | mltlnx: huh? |
| 21:35.40 | mltlnx | sorry. I would like to have a number that never answers only rings. |
| 21:35.54 | grx0 | soooo dont have anyone answer it? |
| 21:36.09 | *** join/#asterisk BadHAL (n=nn@66.194.174.11) |
| 21:36.13 | [TK]D-Fender | </hard> |
| 21:36.30 | Flashtek | i'm guessing this isn't the place to ask then.. |
| 21:37.27 | seanbright | Flashtek: just ask your question |
| 21:37.41 | Flashtek | 22:27 < Flashtek> I can't get dahdi to compile properly, is this the place to ask for pointers ? |
| 21:37.47 | Flashtek | seanbright: i did already.. |
| 21:37.58 | seanbright | ... |
| 21:37.59 | mltlnx | Yeah but the DID does not have a phone on it. |
| 21:38.19 | Flashtek | seanbright: it compiles, but the modules refuse to load... |
| 21:38.23 | seanbright | Flashtek: the REAL question. |
| 21:38.30 | seanbright | ah. |
| 21:38.35 | Flashtek | why might that be ? |
| 21:38.37 | seanbright | what have you tried? |
| 21:38.48 | Flashtek | i have a crowbar.. |
| 21:38.54 | Flashtek | i've done a few recompiles.. |
| 21:39.03 | Flashtek | after removing the original files.. |
| 21:39.09 | Flashtek | used the svn copy.. |
| 21:39.17 | Flashtek | used the tgz bundle.. |
| 21:39.20 | [TK]D-Fender | Flashtek: And where are you showing us your problem? |
| 21:39.52 | seanbright | you should pastebin some relevant console output |
| 21:39.58 | seanbright | showing errors and such |
| 21:39.59 | seanbright | ~pb |
| 21:40.00 | infobot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste , or , http://bin.cakephp.org/ |
| 21:40.07 | Flashtek | [TK]D-Fender: I was asking if this was the right place to ask first of all.. |
| 21:40.16 | seanbright | Flashtek: YES |
| 21:40.21 | ambush276 | anyone answer? |
| 21:40.22 | [TK]D-Fender | Flashtek: So far yes. |
| 21:41.40 | Flashtek | FATAL: Error inserting dahdi (/lib/modules/2.6.26-1-686/asterisk/dahdi/dahdi.ko): Invalid module format |
| 21:41.58 | [TK]D-Fender | Flashtek: Usually means it wasn't compiled for your current kernel |
| 21:44.41 | Flashtek | hmm... let me kick it about a bit.. |
| 21:45.30 | grx0 | laters |
| 21:45.43 | seanbright | laters? |
| 21:45.46 | seanbright | you're fired. |
| 21:47.40 | Flashtek | i've installed "linux-modules-2.6.26-1-686" and "linux-headers-2.6.26-1-686" |
| 21:47.42 | bpgoldsb | seanbright, I never got a chance to thank you regarding that AEL fix... for my inbox being full the next morning. |
| 21:47.50 | *** join/#asterisk Chainsaw (n=chainsaw@gentoo/developer/atheme.member.chainsaw) |
| 21:47.54 | Flashtek | looks like the compile is taking a little longer this time.. |
| 21:48.15 | seanbright | bpgoldsb: not a problem. minor fix, but it took me 4 times to get right :) |
| 21:48.35 | bpgoldsb | :) |
| 21:54.38 | *** join/#asterisk bradlis7 (n=bradlis7@dsl-76-77-16-197.networklubbock.net) |
| 22:01.20 | *** join/#asterisk xpot (n=james@70.91.210.233) |
| 22:01.48 | *** join/#asterisk qdk (n=qdk@0x573d8dd1.bynqu1.dynamic.dsl.tele.dk) |
| 22:04.04 | Flashtek | huzzah, it's worked.. |
| 22:09.42 | Flashtek | amazing, it didn't noticeably whinge... but works now.. |
| 22:10.43 | Flashtek | runs off and hides under a rock.. |
| 22:10.50 | Flashtek | thanks for the pointers.. |
| 22:10.52 | *** part/#asterisk Flashtek (n=neil@flashtek-uk.com) |
| 22:14.12 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 22:20.18 | *** join/#asterisk Tuxguy (n=jimi@cpe-065-184-197-243.ec.res.rr.com) |
| 22:20.44 | Tuxguy | I want to connect a hard phone through my laptop's ethernet, and have my linksys assign it an IP. How can I do this? |
| 22:20.47 | *** join/#asterisk AsteriskDom (n=munoz@190.94.5.57) |
| 22:21.33 | [TK]D-Fender | Tuxguy: Your "Linksys"? Could you be a little more vague about your network layout please... |
| 22:21.47 | Tuxguy | Right, linksys router |
| 22:21.53 | Tuxguy | Here is my layout |
| 22:22.09 | Tuxguy | Cable modem -> Linksys Router -> Rest of computers on my nework |
| 22:22.42 | Tuxguy | I want to plug a hardphone into my laptop, so I do not have to be near the router, and have my laptop pass a dhcp address to my hardphone |
| 22:24.52 | AsteriskDom | but this hardphone is a usb or what? |
| 22:25.25 | *** join/#asterisk pmhaddad-work (n=pmhaddad@141.219.87.43) |
| 22:25.42 | *** join/#asterisk Valect (n=Valect@unaffiliated/valect) |
| 22:25.51 | [TK]D-Fender | Tuxguy: Depends what your Laptop is running |
| 22:25.55 | *** part/#asterisk Valect (n=Valect@unaffiliated/valect) |
| 22:29.20 | *** join/#asterisk H3XiL3D (n=H3X@unaffiliated/hexiled) |
| 22:30.30 | AsteriskDom | infobot: ani |
| 22:30.31 | infobot | extra, extra, read all about it, ani is Automatic Number Identification Systems |
| 22:38.13 | AsteriskDom | infobot: dsp |
| 22:38.14 | infobot | dsp is probably (Digital Signal Processor) A DSP is a microprocessor designed to work with analog signals such as video or audio that have been digitally encoded. The DSP then takes these digital representations and performs operations on them. DSPs are used in video, sound, and modem technology. Intel's MMX instruction set is basically an attempt to make the Pentium processor line capable of DSP operations. This follows Intel's theory of putting all ... |
| 22:40.43 | *** join/#asterisk CrazyTux (n=brandon@ip98-164-236-47.oc.oc.cox.net) |
| 22:40.51 | *** join/#asterisk ingenius (n=alektro@69.90.72.173) |
| 22:44.02 | Tuxguy | [TK]D-Fender, Linux |
| 22:44.08 | Tuxguy | astassistant, ethernet |
| 22:44.09 | Tuxguy | er |
| 22:44.12 | Tuxguy | AsteriskDom, ethernet |
| 22:44.40 | [TK]D-Fender | Tuxguy: well this is basic routing. |
| 22:44.55 | [TK]D-Fender | Tuxguy: Go ask in ##linux or #(yourdistro) |
| 22:45.14 | Tuxguy | they said to go to #asterisk |
| 22:45.15 | Tuxguy | :( |
| 22:49.46 | [TK]D-Fender | Tuxguy: .... then ask differently |
| 22:50.04 | Tuxguy | so, should i install a dhcp server on the laptop? |
| 22:50.23 | AsteriskDom | you want to pass dhcp address from your wi fi card to your ethernet card to the phone |
| 22:50.24 | AsteriskDom | ? |
| 22:50.39 | Tuxguy | correct |
| 22:50.49 | Tuxguy | linksys -> wifi -> ethernet -> phone |
| 22:50.53 | AsteriskDom | this is linux work |
| 22:50.58 | AsteriskDom | no asterisk |
| 22:51.30 | Tuxguy | what should igoogle for? |
| 22:51.37 | AsteriskDom | like <[TK]D-Fender> wrote thid is routing |
| 22:51.51 | AsteriskDom | in this case bridging |
| 22:52.21 | AsteriskDom | you have to bridge your wifi and ethernet card |
| 22:52.57 | Tuxguy | oh ok |
| 22:52.57 | [TK]D-Fender | AsteriskDom: Depends how you do it. |
| 22:53.12 | Tuxguy | i thought i had to install dhcp server or dhdcp-forward |
| 22:54.18 | AsteriskDom | you need a dhcp relay or something like that |
| 22:54.58 | Tuxguy | ah ok ty |
| 22:56.25 | dominicl | If I've got an iaxmodem registered as an IAX extension in asterisk (ext 9001, in this case), can I just route a given DID straight to it, disabling any fax detection or other features, and that should work for passing faxes to the iaxmodem (with hylafax+) to receive the fax? |
| 22:57.01 | Tuxguy | i am going to try the bridged method |
| 22:59.06 | *** join/#asterisk joobie (n=joobie@mx01.anric.com.au) |
| 23:00.29 | elitecoder | [Apr 22 15:57:18] NOTICE[4542]: chan_sip.c:14035 handle_request_invite: Call from '' to extension '+17628204324' rejected because extension not found. |
| 23:00.38 | elitecoder | How would I match that? literally? |
| 23:00.45 | elitecoder | I'm trying with exten => 17608204124, 1, Answer() |
| 23:00.52 | elitecoder | rr except the number is right |
| 23:01.08 | rob0 | extension '+17628204324' ? |
| 23:01.16 | elitecoder | you can put + signs eh? |
| 23:01.29 | rob0 | Looks like you did.' |
| 23:01.38 | elitecoder | I didn't! that was an inbound call |
| 23:01.46 | elitecoder | trying it now tho |
| 23:02.06 | rob0 | strip the first character, or make an extension with the + |
| 23:02.37 | elitecoder | ;o |
| 23:04.44 | elitecoder | ok got it figured out I was just in the wrong context sorry. |
| 23:04.50 | elitecoder | plus signs work fine |
| 23:08.20 | *** join/#asterisk Tuxguy (n=jimi@cpe-065-184-197-243.ec.res.rr.com) |
| 23:11.53 | *** part/#asterisk mltlnx (n=mltlnx@cpe-24-193-11-14.nyc.res.rr.com) |
| 23:18.17 | *** join/#asterisk telecos (n=sergio@19.166.219.87.dynamic.jazztel.es) |
| 23:27.27 | *** join/#asterisk jinxed (n=chatzill@64.235.108.210) |
| 23:33.17 | joobie | guys got an issue with my queue |
| 23:33.37 | joobie | when I do `queue show`, i see "Local/1000@agents with penalty 10 (dynamic) (Invalid) has taken no calls yet" .. |
| 23:33.41 | joobie | i can't seem to get rid of it |
| 23:33.54 | joobie | ive tried reloading app_queue, retstarting asterisk.. it's still there |
| 23:34.46 | *** join/#asterisk [intra]lanman (n=lanman@freeswitch/developer/intralanman) |
| 23:37.31 | [TK]D-Fender | joobie: look at how you added them, and think for a minute... |
| 23:38.40 | joobie | originally i registered them in the queue as Local/1000 |
| 23:38.50 | joobie | ive removed this though.. so need to "flush" the queue |
| 23:39.55 | [TK]D-Fender | joobie: "registered"? Perhaps you could be a LOT more specific about how that became a member of the queue... |
| 23:40.09 | *** join/#asterisk aaroneous (n=aaroneou@p86-64.acedsl.com) |
| 23:40.19 | joobie | hmm.. i could |
| 23:40.19 | joobie | :) |
| 23:40.27 | joobie | I added Local/1000 via AddQueueMember |
| 23:41.03 | joobie | I tried removing it via RemoveQueueMember, it didn't remove.. i could try again as I wasn't 100% comprehensive with this |
| 23:41.21 | joobie | though i'm wondering, is there a way via the CLI that i can force the queue to flush.. would rather do it this way |
| 23:41.58 | [TK]D-Fender | joobie: s/comprehensive/trying at all or backing it up, yup, I'm just plain lazy/ |
| 23:42.24 | [TK]D-Fender | joobie: there is no quick flush otherwise any little think might wipe out call centers |
| 23:42.27 | *** join/#asterisk juanIMP (n=Juancho@200.26.152.222) |
| 23:42.50 | [TK]D-Fender | joobie: there is AQM and RQM for a reason. To think that one isn't meant to undo the other is absurd |
| 23:44.53 | KyleK | any canadians in here? trying to find a good online retailer to buy a Grandstream Handytone 503 |
| 23:48.12 | [TK]D-Fender | ~gs |
| 23:48.13 | infobot | GrandSuck phones & gateways are cheap junk which should be avoided with extreme prejudice. |
| 23:49.16 | aaroneous | ~linkydink |
| 23:49.32 | aaroneous | damn :< |
| 23:51.44 | KyleK | well I don't think I can get a non-junk FXO for $80usd or less |
| 23:52.19 | aaroneous | nobody reputable seems to make ATAs anymore |
| 23:52.21 | *** join/#asterisk columbo (n=PeterFal@pool-71-177-208-38.lsanca.dsl-w.verizon.net) |
| 23:52.36 | pfn | doesn't linksys? |
| 23:52.41 | aaroneous | I think I was recommended a linksys the last time I was here.. |
| 23:52.55 | aaroneous | doesn't do linkydinks |
| 23:53.19 | KyleK | I've been hearing that linksys makes crap these days :-( |
| 23:53.37 | aaroneous | since when were they not making crap.. |
| 23:54.02 | aaroneous | linksys is the antithesis of reputable |
| 23:55.08 | pfn | since cisco bought 'em |
| 23:55.11 | pfn | linksys isn't horrid |
| 23:55.16 | pfn | and rather decent |
| 23:55.16 | aaroneous | that said, I too need to decide on an ATA.. need a few FXS ATAs to be exact |
| 23:55.44 | *** join/#asterisk bbryant (n=brett@c-68-59-20-153.hsd1.sc.comcast.net) |
| 23:55.51 | aaroneous | pfn: they're still doing the same old BS of slapping their worthless name on lowest-bid ODM devices.. |
| 23:56.28 | joobie | [TK]D-Fender, sounds like it's a good feature to have |
| 23:56.37 | joobie | that's like saying don't give administrator access because you can fuk things up |
| 23:56.53 | joobie | i can unload the app_queue module, and fuk up the queue.. but i cant flush the queue.... |
| 23:56.53 | aaroneous | and they're still doing the same old BS of putting technologically unrelated revisions under the same model numbers/SKUs |
| 23:58.02 | [TK]D-Fender | joobie: use RQM like the obvoius instructions tell you to |
| 23:58.04 | aaroneous | such that WBS54G might be a broadcom card one day and an atheros the next.. |
| 23:58.09 | KyleK | aaroneous: been looking an any fxo hardware? |
| 23:58.20 | joobie | dont have a choice TK.. will have to |
| 23:58.47 | aaroneous | KyleK: nah.. we have a PRI attached to a cisco 3640 for our pstn connectivity |
| 23:59.39 | aaroneous | I need FXS ATAs to VoIP-enable legacy devices |
| 23:59.52 | KyleK | ah |
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