| 00:01.47 | illizit | anyone else have any ideas? :) |
| 00:02.25 | [TK]D-Fender | illizit: It calls prior to bridging. Thats jsut teh way its meant to work. If you don't like it feelf ree to mod the source |
| 00:02.41 | illizit | lol |
| 00:02.44 | illizit | ok |
| 00:02.58 | illizit | why im trying to do is have a script find out what agent answered the call |
| 00:03.23 | illizit | the software needs to know during the call not after |
| 00:03.43 | illizit | so i cant use cdr, since that does not get written after |
| 00:04.03 | illizit | i tried parsing the queue_log file but that does not have the DID dialed among other things. |
| 00:05.29 | *** join/#asterisk path_ (n=path_@pc-15-190-86-200.cm.vtr.net) |
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| 00:18.00 | logicwrath | can anyone suggest a softphone that supports alert_info |
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| 00:30.01 | *** mode/#asterisk [+o mog] by ChanServ |
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| 00:43.00 | Darky2 | what are the common technologys used to have two or more users let view each others webcam? anyone knows that? |
| 00:49.20 | rue_mohr | how about jpeg stream |
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| 00:50.55 | *** mode/#asterisk [+o Deeewayne] by ChanServ |
| 00:51.47 | Darky2 | should be close to real-time. I found "Flash Media Server" now though |
| 00:58.24 | *** join/#asterisk ograce (n=ograce@hlfxns0163w-142068143117.pppoe-dynamic.ns.aliant.net) |
| 01:00.44 | ograce | hello, I'm new to asterisk and I think I have it conf but I wanted to know if there is a good how to get a softphone up and going to see if I got it right. I have installed asterisk on dd-wrt running on a linksys wrt300 |
| 01:01.06 | pfn | asterisk on a dd-wrt? people do that? |
| 01:01.07 | pfn | boggles |
| 01:01.24 | ograce | well that could be half my problem |
| 01:01.46 | ograce | I was reading and saw that it could be done so I tried it out. |
| 01:02.11 | ograce | I have a process running but don't know how to test it |
| 01:03.45 | ograce | it was easier to setup a linksys running dd-wrt then upgrading a server to a current version. :) |
| 01:05.35 | pfn | who needs to upgrade to a current version |
| 01:05.39 | pfn | just get the source and build |
| 01:06.46 | ograce | yah but I have been lazy ... very lazy I'm almost 3 versions back. |
| 01:07.16 | ograce | and all I use the server for is a webserver and a nas |
| 01:08.13 | ograce | but when I went to try out asterisk ... it said that I had a version problem |
| 01:08.25 | ograce | I has an extra router so what the hell |
| 01:08.35 | pfn | I was running asterisk on redhat7 or some old shit |
| 01:08.42 | pfn | and that was fine up until 1.2 or so |
| 01:09.01 | ograce | dd-wrt works great ... and the wifi speed rocks |
| 01:09.50 | ograce | I'm impressed with just the dd-wrt upgrades |
| 01:10.08 | ograce | but I'm here to learn more about asterisk. |
| 01:10.50 | [TK]D-Fender | ~book |
| 01:10.51 | 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 |
| 01:10.52 | ograce | so once I have everything set up what is the next step to test out asterisk |
| 01:10.53 | [TK]D-Fender | ^^^^^^^ |
| 01:13.13 | ograce | K ... I was just hopiong there was a good and easy test or a how to. |
| 01:13.22 | ograce | thanks fender |
| 01:15.20 | [TK]D-Fender | ograce: Here's a little something for "inspiration" in your first install : |
| 01:15.22 | [TK]D-Fender | ~jerjerguide |
| 01:15.23 | 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/ |
| 01:18.12 | ograce | fender I think i have all the conf files updated with everything I know sofar but I think I'm missing the ends |
| 01:19.11 | ograce | what goes on the computer and who can I connect the asterisk box to for testing |
| 01:19.45 | ograce | before I go buy a voip service and a did number |
| 01:20.06 | [TK]D-Fender | ograce: Everything depends on what you want to do. |
| 01:20.28 | [TK]D-Fender | ograce: And to test * you can do all of this with a simple softphone |
| 01:20.31 | [TK]D-Fender | ~softphone |
| 01:20.32 | infobot | [~softphone] A soft-phone is a program that lets your computer use a compatible audio interface (usually a speaker&mic, or headset), and optionally a webcam, to place VoIP calls with a given protocol (SIP/IAX/H.323/MGCP/etc). For links enter ~zoiper , ~xlite , ~twinkle , ~bria , ~eyebeam , ~ekiga |
| 01:20.42 | ograce | well I would really like to replace my pots with what ever I build |
| 01:21.20 | ograce | so in the end I will need a DID, and an ata of some type. |
| 01:21.44 | ograce | so I thought I would start with asterisk and a test to see if I can pull it off. |
| 01:22.23 | [TK]D-Fender | ograce: And can you present any reason you shouldn't be able to? |
| 01:22.45 | [TK]D-Fender | ograce: ograce Becuase thousands of * users seem to do just fine |
| 01:26.21 | ograce | I hope to be one of those in the future. do I need a DID and IAX service before I can test out a softphone connection running through asterisk running on this linksys. |
| 01:26.50 | [TK]D-Fender | ograce: Nope |
| 01:27.42 | [TK]D-Fender | ograce: thing to know about *. In general it just processes calls. Whatever you've heard of other people doing with it you can do as well. not much to say with that in mind |
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| 04:18.08 | jostua | Hi, I'm running Asterisk 1.6.0.6 on CentOS 5. Meetme is not passing audio when user agents connect. Dahdi_dummy is loaded. Any ideas? |
| 04:19.14 | [TK]D-Fender | <PROTECTED> |
| 04:20.20 | jostua | i can make SIP calls just fine |
| 04:22.09 | [TK]D-Fender | jostua: Confirm that device specifically to Record & Playback. Then again with Meetme. Pastebin the failed attempt with MeetMe with SIP debug enabled. |
| 04:22.52 | [TK]D-Fender | jostua: And please be more specific about the device in question, what kind, networking, etc |
| 04:24.37 | [TK]D-Fender | ~pb |
| 04:24.38 | 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/ |
| 04:26.41 | jostua | [TK]D-Fender: I'm using Twinkle as the user agent for tests. This is only a SIP environment. UA is connected to the asterisk box over the internet. |
| 04:27.21 | [TK]D-Fender | jostua: NAT misconfiguration is the #1 reason for 1-way audio |
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| 04:32.08 | jostua | [TK]D-Fender: pastebin: http://pastebin.com/m2543137f |
| 04:32.39 | jostua | [TK]D-Fender: I've made sure NAT isn't the problem. I can make a call from one Twinkle UA to an Xlite UA on a different machine. |
| 04:33.25 | jostua | [TK]D-Fender: I've also tried isolating the codec to PCMU for all SIP peers |
| 04:33.42 | *** part/#asterisk rue_mohr (n=rue@h24-207-90-17.cst.dccnet.com) |
| 04:35.45 | [TK]D-Fender | jostua: answer the call first and play some audio, even 1s silence |
| 04:35.58 | [TK]D-Fender | jostua: before dumping into meetme. |
| 04:40.30 | jostua | [TK]D-Fender: My wife just called and needs me to go pick her up... I'll try what you've suggested to see where the audio drops and be back when I can |
| 04:41.03 | jostua | [TK]D-Fender: Thank you for the help, what little time I had. |
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| 05:40.44 | IDXRob | Does anyone know how to resolve this error?: *CLI> -- Got SIP response 500 "Server error occurred (19/SL)" back from 198.65.166.131 |
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| 05:46.36 | [TK]D-Fender | IDXRob: Depends what its in response to. |
| 05:47.39 | IDXRob | I'm just trying to dial out on SIP using a gizmo account |
| 05:48.08 | IDXRob | I think it might be an authentication problem, however I can receive calls |
| 05:50.14 | Doc | heh.. gizmo |
| 05:50.27 | jblack | check out sip debug. |
| 05:51.58 | IDXRob | Doc: Doing this as a hobby.. |
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| 06:22.46 | *** join/#asterisk ruben23 (n=ITadmin@203.177.91.163) |
| 06:22.53 | ruben23 | hi |
| 06:24.25 | ruben23 | if i setup asterisk on a single server with 2 ethernet ports.. like this internet==>modem==>(eth0)asterisk server(eth1)==>switch==>SIP client |
| 06:25.09 | ruben23 | do i have to create a NAT and dhcp service on it..including the ipforward... |
| 06:27.02 | lanning | DHCP can be served from any host on the network, does not have to be the gateway. |
| 06:28.02 | lanning | You will have to configure NAT and other firewall rules, if the asterisk box is the gateway. (also turn on ipforward, if it is the gateway.) |
| 06:28.41 | lanning | if it is parallel to the gateway (ie. you have two internet addresses from your ISP), then you don't have to do anything. |
| 06:29.42 | lanning | all the special configs (NAT, firewall, ipforward, etc...) are only to support other protocols routing through the asterisk box. |
| 06:30.04 | [TK]D-Fender | ruben23: You only have to have your * box do NAT if you are not routing a public IP through it and want your devices behind it to be able to access the internet |
| 06:35.21 | *** join/#asterisk ruben23 (n=ITadmin@203.177.91.163) |
| 06:35.59 | [TK]D-Fender | ruben23: You only have to have your * box do NAT if you are not routing a public IP through it and want your devices behind it to be able to access the internet |
| 06:36.07 | ruben23 | lanning: thanks |
| 06:37.37 | ruben23 | [TK]D-Fender: is that consider as a good set for asterisk server |
| 06:38.08 | ruben23 | or having my asterisk server behind a Nat. |
| 06:38.10 | [TK]D-Fender | ruben23: Doesn't really matter. As long as packets get where they need to be. |
| 06:38.38 | [TK]D-Fender | ruben23: Others would say its better to use a fixed router so that your global internet access doesn't go down if your * server does |
| 06:38.58 | [TK]D-Fender | ruben23: You'll have to judge for yourself based on your needs |
| 06:39.45 | ruben23 | [TK]D-Fender:how about issue with SIP to NAT... |
| 06:40.02 | [TK]D-Fender | ruben23: Standard rules apply |
| 06:42.04 | ruben23 | [TK]D-Fender:you have link for that standard rules...so i can view and study it... |
| 06:42.23 | [TK]D-Fender | ~sipnat |
| 06:42.23 | 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 |
| 06:51.19 | [TK]D-Fender | checkout time, later all |
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| 08:41.24 | key2 | hi |
| 08:41.34 | key2 | is it possible to send sms using chan DAHDI ? |
| 08:55.40 | *** join/#asterisk nighty^ (n=nighty@x122091.ppp.asahi-net.or.jp) |
| 09:04.38 | *** join/#asterisk esperegu (n=esperegu@145.116.15.244) |
| 09:05.40 | esperegu | anyone knows how I can see which line in a xml config file for a Cisco 7970 is causing the phone to give an "Error Verifying Config Info" |
| 09:08.05 | *** join/#asterisk mikkel (n=mikkel@84-238-113-66.u.parknet.dk) |
| 09:09.38 | tzafrir_laptop | key2, depends on your provider |
| 09:10.08 | tzafrir_laptop | in most eurepean countries you can |
| 09:10.35 | jql | esperegu: have you at least run it through an XML parser? |
| 09:12.22 | esperegu | jql: yeah. I used an online validator |
| 09:13.46 | jql | if it's valid xml, then if I were in your shoes I'd have to bisect the difference between a known-good config and the problem config. sucky |
| 09:17.45 | dzup | hi, is there a way to uninstall asterisk and the zap driver? i recompile my kenel but zap drive is not letting me finish with the compilation of my new kernel, i want to manually remove it, any one? |
| 09:20.30 | tzafrir_laptop | dzup, hmm... what error do you get when building the new kernel? |
| 09:21.10 | dzup | zap module dont accept root something |
| 09:21.10 | esperegu | jql: if I only had a known-good workinc config :-( |
| 09:22.17 | dzup | then the module compile broke to exit with error, if i can momentarily disable that module and i can ecompile later. |
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| 09:24.30 | dzup | tzafrir_laptop: if i can momentarily disable zap driver i think i be okay, then when i got my kernel ready i can recompile the drive. |
| 09:25.01 | tzafrir_laptop | dzup, I really can't see how the presence of zaptel/dahdi affects building a kernel |
| 09:25.15 | dzup | it does |
| 09:25.37 | tzafrir_laptop | do you include zaptel inside your kernel tree? |
| 09:25.45 | tzafrir_laptop | what error do you actually get? |
| 09:26.10 | dzup | becouse fails to compile, when the kernel compilation in debian reaches the /usr/src/linux/module/zaptel the compilation halts |
| 09:27.09 | tzafrir_laptop | dzup, /usr/src/module/zaptel is normally something you can safely delete, as it is extracted from /usr/src/zaptel.bz2 |
| 09:27.32 | tzafrir_laptop | but I wonder why you build the kernel directly under /usr/src/linux |
| 09:27.56 | tzafrir_laptop | Generally just build Zaptel / dahdi after the kernel has been built and installed |
| 09:28.27 | dzup | i already did remove all this: rm -rf /etc/asterisk; rm -rf /usr/lib/asterisk/modules; rm -rf /var/lib/asterisk;rm /etc/zaptel.conf;rm /var/spool/asterisk/;rm -rf /var/spool/asterisk/;rm -rf /var/log/asterisk; |
| 09:28.29 | tzafrir_laptop | alternatively, the files of dahdi can be dropped into a kernel tree and will also be configured by Kconfig |
| 09:28.46 | tzafrir_laptop | dzup, why do you remove configuration? |
| 09:30.13 | dzup | tzafrir_laptop: 'cuz i got backups, and i dont need that since am moving from one etch to gutsi (or somethink like that) and is a custom kernel |
| 09:30.58 | tzafrir_laptop | you're installing gutsi now??? |
| 09:31.14 | dzup | am upgrading the kernel |
| 09:31.35 | dzup | to that yes, but am only migration from this into that |
| 09:32.20 | dzup | what i need is the zap driver not to be load in memory |
| 09:32.32 | dzup | when i restart |
| 09:33.04 | dzup | so am guessing that will be on /etc/rc.d ? |
| 09:33.49 | dzup | or /etc/init.d ? |
| 09:33.56 | dzup | am lost here |
| 09:41.13 | jblack | check your /etc/modules.conf |
| 09:41.27 | jblack | pardon, /etc/modules |
| 09:43.34 | tzafrir_laptop | dzup, do you have an actual hardware? |
| 09:43.51 | tzafrir_laptop | also, why does it bother you that the module is loaded? |
| 09:45.28 | *** join/#asterisk oej (n=olle@ns.webway.se) |
| 09:49.43 | dzup | tzafrir_laptop: no i dont |
| 09:50.41 | dzup | jblack: is not there either on /etc/modules |
| 09:50.59 | dzup | i may reboot now to see if still been load in memory. |
| 09:51.47 | dzup | tzafrir_laptop: bother my kernel custom compilation, looks like is not compatible with kernel 2.6.29 |
| 09:52.26 | tzafrir_laptop | Fixed in zaptel svn |
| 09:53.31 | dzup | hmm so is a bug? |
| 09:53.52 | tzafrir_laptop | yes |
| 09:55.06 | dzup | ok, thanks, i will reboot my box now to see if the zaptel drive still trying to load =) |
| 09:57.04 | *** join/#asterisk war9407 (i=war@liquidswords.org) |
| 10:04.48 | noisewaterphd | my asterisk will not save voicemail greetings when you call in and record them. it also will not save voicemail messages after emailing them, even though delete=no. folder permissions are fine, no errors in asterisk cli. If i go into /var/spool/asterisk/voicemail/default/200 and touch unavail.wav, then call in and record new unavail message, unavail.wav disappears. asterisk and freepbx source installs on centos 5.4 |
| 10:05.49 | noisewaterphd | can anyone help point me towards any possible info to fix this, i'm stumped and losing hair |
| 10:17.07 | *** join/#asterisk [TK]D-Fender (n=joe@64.235.218.194) |
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| 10:27.29 | war9407 | I have a question, with an ATA adapter, can I receive calls to my regular phone line and send them over ethernet to a host running asterisk so I can e.g., send voice mail via email? |
| 10:31.43 | [TK]D-Fender | war9407: You plug phones into an ATA, not a telephone line |
| 10:32.28 | [TK]D-Fender | war9407: ATA's are FXS interfaces, you need FXO. |
| 10:32.48 | war9407 | thanks, will go look up FXOs now |
| 10:33.56 | [TK]D-Fender | war9407: There are cards by several makers including Digium & Sangoma, as well as SIP gateways from others. |
| 10:34.52 | war9407 | [TK]D-Fender: The last one I used/tried was a zaptel card, it requires kernel drives to be built not in the kernel, (pia), are their any 'recommended' brand-names to look for that produce external (usb or ethernet-based) FXOs? I will check Digium & Sangoma, thx. |
| 10:34.56 | [TK]D-Fender | war9407: the least expensive largely usable option would probably be : http://www.telephonydepot.com/Catalog/Linksys-Analog-Adapters/Linksys-SPA3102 |
| 10:35.12 | war9407 | is reading |
| 10:35.13 | [TK]D-Fender | war9407: this unit fits that need |
| 10:36.21 | war9407 | that one looks very nice, good reviews too, thx for the advice, I need to read up on it now ;) |
| 11:16.01 | *** join/#asterisk esperegu (n=esperegu@145.116.15.244) |
| 11:21.18 | war9407 | [TK]D-Fender: read up on it, seems like a very good model, etc, it can be used for voice -> out via skype or other VoiP provider, etc |
| 11:21.38 | war9407 | [TK]D-Fender: it will also let me receive calls from [pots/phone line] =====> device =====> forward it to asterisk right? :) |
| 11:21.43 | war9407 | checking atm as well |
| 11:21.59 | [TK]D-Fender | war9407: Not Skype |
| 11:22.32 | [TK]D-Fender | war9407: It can let you use 1 line and 1 phone independently with * or interactively by itself |
| 11:23.33 | war9407 | [TK]D-Fender: cool, so from comcast voip (is what I have) that plugs into a regular phone line, I will plug it into the SPA3102 and then the SPA3102 via ethernet on the network so it can 'forward' the calls to asterisk, from there, I can do a lot of things? oh, one other question |
| 11:23.40 | war9407 | [TK]D-Fender: I have 3-way calling with comcast VOiP |
| 11:23.47 | war9407 | [TK]D-Fender: is it possible with asterisk to forward phone calls? |
| 11:23.55 | war9407 | [TK]D-Fender: in my configuration |
| 11:24.04 | [TK]D-Fender | war9407: * can do more with your resources than you can. |
| 11:24.40 | [TK]D-Fender | war9407: If you have lines, * can typically do whatever you can with those same links yourself. |
| 11:24.51 | war9407 | cool |
| 11:24.55 | war9407 | [TK]D-Fender: ok, just to confirm, inbound call to phone line => spa3102 can forward to asterisk, from there I can e-mail the voice mail to another address, correct? |
| 11:25.00 | [TK]D-Fender | war9407: the concept of "forward" is largely inappropriate |
| 11:25.05 | war9407 | true |
| 11:25.28 | [TK]D-Fender | war9407: If you have 2 separate channels * can receive on one, and call out the other, bridging the two |
| 11:25.56 | [TK]D-Fender | war9407: sure, it can take VM and e-mail it |
| 11:25.59 | war9407 | but what if one one channel and it has '3 way support', e.g. you press the on button, dial and press on again |
| 11:26.04 | war9407 | [TK]D-Fender: excellent (voice mail) |
| 11:26.27 | war9407 | [TK]D-Fender: sounds like a better idea to go with a VoIP provider to do native VoIP but my initial plan is for voice mail only |
| 11:27.39 | [TK]D-Fender | war9407: "on" button? |
| 11:27.50 | war9407 | aka 'flash' |
| 11:27.58 | [TK]D-Fender | war9407: Sounds like you're trying to refer to an analog phone's "flash" key |
| 11:28.15 | war9407 | yeah, doesnt sound like Ill be able to do that with asterisk(?) |
| 11:28.26 | war9407 | and the SPA3102, but it would be neat. |
| 11:28.33 | [TK]D-Fender | war9407: Usually possible, just a little tricky |
| 11:28.39 | [TK]D-Fender | war9407: but not with that device |
| 11:28.58 | [TK]D-Fender | war9407: the PCI solutions would be capable of this |
| 11:29.16 | war9407 | [TK]D-Fender: ah.., so if one wants selective forwarding, etc you'd just go with a VoIP provider and asterisk/PCI solution |
| 11:29.49 | [TK]D-Fender | war9407: You are mixing a few things in that statement without context as to their use |
| 11:30.36 | war9407 | [TK]D-Fender: if($phone_num == "123-456-7890") { fwd_phone to "123-444-5675" } |
| 11:30.37 | [TK]D-Fender | war9407: You'd get an FXO if you wanted to use an analog line. Typically I'd prefer using a sing tech for everything |
| 11:30.47 | war9407 | "sing tech"? |
| 11:31.05 | [TK]D-Fender | war9407: again, thats just looking at the CID and dialing out another resource you have. Type doesn't matter |
| 11:31.09 | [TK]D-Fender | single* |
| 11:34.27 | mort_gib | [TK]D-Fender: Sing tech?? Eh, you all right?? |
| 11:35.33 | war9407 | [TK]D-Fender: so for a simple external device that will let me forward an inbound phone call to asterisk, to do voice mail, or have call trees, or to filter out telemarketers, the SPA3102 looks like the device to get, correct? |
| 11:36.04 | [TK]D-Fender | war9407: it'll do. |
| 11:36.37 | war9407 | thanks! |
| 12:00.24 | *** join/#asterisk Ibrar (n=ibrar@mbl-82-54-34.dsl.net.pk) |
| 12:00.33 | Ibrar | Hi all |
| 12:01.53 | Ibrar | First of all I am a newbie of asterisk, so pardon me on my stupid questions. I have installed a asterisk on public Ip address, and I have a softphone (SIP) behind a firewall. Can I connect to that asterisk |
| 12:02.18 | Ibrar | If yes the what ip address I have to give in sip configuration of asterisk |
| 12:05.21 | *** join/#asterisk Urthwhyte (n=urthwhyt@0x5da320aa.cpe.ge-1-1-0-1101.oebrnqu2.customer.tele.dk) |
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| 12:29.01 | whitehat | hello group. looks like the asterisk-gui is still asleep. i've just download the svn version of the asterisk-gui and when I attempt to run it it continues in a loop. ideas? thank you. |
| 12:32.46 | key2 | is it possible to send sms using chan DAHDI ? |
| 12:51.36 | *** join/#asterisk esaym (n=user@cpe-70-120-89-6.satx.res.rr.com) |
| 12:57.43 | tzafrir_laptop | key2, depends on your provider |
| 12:58.05 | tzafrir_laptop | in most eurepean countries you can |
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| 13:44.22 | Steve_J-obs | hello everybody!!! |
| 13:44.39 | Steve_J-obs | anybody here? |
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| 16:02.30 | *** join/#asterisk Wired_Life (n=Chatzill@mgdb-4db814e4.pool.einsundeins.de) |
| 16:03.19 | Wired_Life | hello i have a question... its possible to check that a sip acc is online beforce calling them? |
| 16:03.46 | jaytee | at the CLI, type "sip show peers" |
| 16:03.58 | *** join/#asterisk andresmujica (n=andresmu@ubuntu/member/andresmujica) |
| 16:04.10 | *** join/#asterisk jstoker (n=jstoker@80-41-183-84.dynamic.dsl.as9105.com) |
| 16:04.12 | jaytee | or are you asking if they are on a call already? |
| 16:04.35 | Wired_Life | i search a function for dialplan |
| 16:05.52 | Wired_Life | before that "exten = mynumber,20,Dial(SIP/garage)" i want to check that garage is online because when the person is offline i get a busy congestion error |
| 16:06.42 | *** part/#asterisk jstoker (n=jstoker@80-41-183-84.dynamic.dsl.as9105.com) |
| 16:07.38 | [TK]D-Fender | Wired_Life: Which you should react to. |
| 16:07.57 | [TK]D-Fender | Wired_Life: thats what DIALSTATUS is for |
| 16:07.59 | eppigy | YES |
| 16:08.32 | eppigy | ${DIALSTATUS} |
| 16:09.17 | eppigy | whoops |
| 16:09.23 | eppigy | you a;ready gave the variable |
| 16:09.37 | Wired_Life | with DIALSTATUS i can check that garage is online? |
| 16:10.00 | [TK]D-Fender | Wired_Life: No, you can check that it failed |
| 16:10.33 | [TK]D-Fender | Wired_Life: At least more importantly the reason why |
| 16:10.34 | jaytee | why would it go "offline" if it's setup properly? my sip phones stay registered unless someone unplugs one |
| 16:10.46 | Wired_Life | ok also i must first call the person than i check DIALSTATUS if it failed? |
| 16:11.03 | Wired_Life | sry for my bad english |
| 16:11.44 | Wired_Life | ah ive find it on google thanks |
| 16:11.56 | [TK]D-Fender | Wired_Life: DIALSTATUS. The status of your dial attempt . |
| 16:11.57 | Wired_Life | asterisk ownz |
| 16:12.08 | eppigy | haha |
| 16:12.16 | eppigy | indeed |
| 16:12.42 | [TK]D-Fender | eppigy: does FAILREASON mean I have to try first? DAMMIT! |
| 16:16.34 | Wired_Life | one question what for a error shows when a sip acc is offline? chanunavail? |
| 16:17.00 | Wired_Life | i have only softphones thats why i must check |
| 16:21.34 | eppigy | well it couldnt be busy of no answer |
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| 16:25.26 | *** part/#asterisk jstoker (n=jstoker@80-41-183-84.dynamic.dsl.as9105.com) |
| 16:27.43 | [TK]D-Fender | Wired_Life: Why do you have to check first? What is the consequence of dialing and failing? |
| 16:30.03 | ruben23 | hi how do i upgrade my version of asterisk without reloading the whole system.... |
| 16:31.30 | [TK]D-Fender | ruben23: Trash your modules folder, recompile and install |
| 16:33.25 | ruben23 | [TK]D-Fender:would it be also be reliable using svn....can i have that options |
| 16:34.21 | [TK]D-Fender | ruben23: SVN depends on the state its in, and you will not have anywhere near as much information on bugs, etc for a given version that way |
| 16:34.42 | Wired_Life | [TK]D-Fender http://pastebin.com/d67294a59 |
| 16:35.19 | [TK]D-Fender | Wired_Life: and...? |
| 16:35.35 | ruben23 | [TK]D-Fender:ok ill do the manual method ...thanks |
| 16:35.38 | Wired_Life | the second call fails but the softphone is online |
| 16:36.31 | [TK]D-Fender | Wired_Life: And you aren't looking at teh SIP debug. Just becaUSE THE SOFTPHONE THINKS EVERYTHING IS ok DOESN'T MEAN IT IS. |
| 16:37.10 | jaytee | stop shouting, i'm trying to take a nap here! |
| 16:37.51 | Wired_Life | garage is only online when im in the garage and oli is online everytime so in the dialplan i first call garage and if that fails i call oli but i dosent work sometimes |
| 16:39.52 | Wired_Life | exten = 6629xxx,20,Dial(SIP/garage) |
| 16:39.52 | Wired_Life | exten = 6629xxx,21,Dial(SIP/oli) |
| 16:39.57 | [TK]D-Fender | Wired_Life: Yes, and how does checking make it WORK? |
| 16:40.10 | [TK]D-Fender | Wired_Life: You seem to have a serious logic flaw |
| 16:40.27 | [TK]D-Fender | Wired_Life: Your softphone has issues and youa ren't even looking at SIP DEBUG for your call attempt |
| 16:45.18 | Wired_Life | hm |
| 16:47.46 | Wired_Life | you think its the softphone that dosnt work? |
| 16:48.37 | Wired_Life | i think asterisk is to fast with calling |
| 16:48.45 | [TK]D-Fender | Wired_Life: I think you aren't looking at the SIP DEBUG like you should. |
| 16:48.59 | Wired_Life | where is the sip debug? |
| 16:49.03 | Urthwhyte | !recap |
| 16:49.07 | [TK]D-Fender | Wired_Life: you have to enable it at * CLI |
| 16:49.17 | [TK]D-Fender | Wired_Life: "sip set debug on" |
| 16:49.42 | *** join/#asterisk chandoo (n=chandoo@ool-4353b978.dyn.optonline.net) |
| 16:50.12 | Wired_Life | ok i enabled that now i test again |
| 16:50.36 | Wired_Life | hey cool there are sip reads from the softphones ^^ |
| 16:50.52 | Wired_Life | every 10 seconds or so |
| 16:54.01 | Wired_Life | [TK]D-Fender you are great |
| 16:54.19 | *** join/#asterisk JuStIcIa_ (i=john@cbl-sd-74-96.aster.com.do) |
| 16:54.24 | Wired_Life | thx for help and happy easter |
| 17:04.41 | *** join/#asterisk hi365 (n=hi365@213.151.52.163) |
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| 17:22.38 | *** join/#asterisk mazpe (n=mazpe@c-71-196-32-22.hsd1.fl.comcast.net) |
| 17:23.59 | mazpe | is there an option for voicemail retreival without using the ivr or dedicated extension approach? |
| 17:24.06 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 17:24.10 | mazpe | like while listening to the voicemail recording, press * |
| 17:25.51 | xheliox | mazpe: Sure. |
| 17:26.05 | mazpe | background() ? |
| 17:26.12 | xheliox | No? |
| 17:26.54 | xheliox | exten => a,1,VoiceMailMain(${CLIENTVM}@default) -- put something like that in the context of however they're accessing the VM. |
| 17:27.01 | mazpe | s,1,Background(voicemail-recording) s,2,Voicemail(100) *,1,Voicemail |
| 17:27.15 | mazpe | ok |
| 17:27.20 | *** join/#asterisk juanIMP (n=juan@201.244.250.91) |
| 17:27.53 | juanIMP | Good day everybody :D |
| 17:29.16 | juanIMP | I want to upgrade asterisk 1.2.24 to asterisk 1.2.30, the server has a sangoma with PRI, Which is the best way to upgrade? |
| 17:30.38 | [TK]D-Fender | juanIMP: Try upgrading just *. If it works then you're fine. Otherwise you'll have to upgrade Zaptel & libpri and recompile wanpipe after |
| 17:31.40 | mazpe | xheliox: this is how i have it setup now: http://rafb.net/p/8Pu77L42.html |
| 17:31.54 | mazpe | in my [inbound] content |
| 17:31.56 | [TK]D-Fender | mazpe: Why are you making an IVR to go to VM and not using the box's normal message recording system? |
| 17:31.58 | mazpe | context+ |
| 17:33.00 | [TK]D-Fender | mazpe: And you were told for this, you use the "a" Asterisk Standard Extension to exit out of Voicemail via "*" while its native announcements are playing |
| 17:33.49 | juanIMP | thanks a lot [TK]D-Fender |
| 17:34.37 | mazpe | [TK]D-Fender: as a work around.. but if there is no need. |
| 17:34.52 | mazpe | i rather use, the native procedure |
| 17:35.28 | mazpe | trying to figure out where in the context a,1,.... goes |
| 17:36.08 | mazpe | tesing |
| 17:36.13 | mazpe | testing+ |
| 17:36.41 | xheliox | Anywhere inside that context. However. Since your VM box doesn't at all match your extension, I'd just put a,1,VoiceMailMain() |
| 17:36.47 | xheliox | but it'll prompt you for the mailbox number again. |
| 17:37.53 | mazpe | thats fine |
| 17:38.16 | xheliox | then just add a,1,VoiceMailMain() inside that same context and it should allow you to hit * when listening to the greeting |
| 17:39.38 | [TK]D-Fender | xheliox: Or maybe he only uses 1 box and can just go ahead and specify it. |
| 17:39.52 | xheliox | [TK]D-Fender: True enough. |
| 17:40.18 | xheliox | My advice is free, but not always sound. :) |
| 17:40.32 | baliktad | I'm trying to get BLF working by assigning a hint (associated with a SIP device) to an extension: exten => 7245,hint,SIP/247 |
| 17:40.33 | mazpe | its a multi-tenant box and i have several clients in the same inbound context |
| 17:40.40 | *** join/#asterisk bbkt-trix (n=bbkt-tri@unaffiliated/bbkt-trix) |
| 17:40.48 | baliktad | the hint is correct for calls placed TO the SIP device |
| 17:41.03 | mazpe | the way i seperate them is by the DID |
| 17:41.11 | baliktad | but the state never changes from IDLE when any call is placed FROM the SIP device |
| 17:41.20 | xheliox | well, your example being 101 wouldn't have indicated that. |
| 17:41.57 | mazpe | yeah i know... i just realized that |
| 17:42.03 | mazpe | sorry :) |
| 17:42.05 | [TK]D-Fender | baliktad: pastebin your peer entry minus PW |
| 17:42.22 | xheliox | What I do is set a variable of "clientvm" and pass that to the voicemailmain app |
| 17:43.06 | *** join/#asterisk Franck78 (n=fbo@204.26.72-86.rev.gaoland.net) |
| 17:43.27 | xheliox | e.g. exten => +3334445555,n,Set(CLIENTVM=${EXTEN}) and then a,1,VoiceMailMain(${CLIENTVM}@default) -- then it'll just prompt for the password, not the mailbox #. |
| 17:43.51 | baliktad | http://pastebin.ca/1388976 |
| 17:44.06 | mazpe | but wouldnt it require the context for that client? |
| 17:44.24 | [TK]D-Fender | baliktad: limitonpeers=yes <- would help if you actually set "type=peer" :) |
| 17:44.26 | mazpe | as each client have a voicemail in their own context.. so you might have 101@client1, 101@client2 |
| 17:44.42 | xheliox | why the hell would do you have each client in their own vm context? :) |
| 17:44.51 | *** part/#asterisk xheliox (n=jeff@178.199.8.67.cfl.res.rr.com) |
| 17:44.55 | *** join/#asterisk xheliox (n=jeff@178.199.8.67.cfl.res.rr.com) |
| 17:45.02 | xheliox | whoops, one too many ctrl-f4's. |
| 17:45.08 | [TK]D-Fender | xheliox: Only if each client had multiple boxes, which we don't have any evidence of of course... |
| 17:45.15 | xheliox | I'm presuming too much about your setup. |
| 17:45.32 | xheliox | But you got the basics, go forth and practice. :) |
| 17:45.51 | mazpe | well, so that i can 101@client1 and 101@client2, etc. |
| 17:45.58 | mazpe | does that not makes sense? |
| 17:46.01 | baliktad | they are all type=friend because the phones place and receive calls |
| 17:46.09 | mazpe | or is there a better way to achieve that |
| 17:46.12 | xheliox | mazpe: Considering you just said you had your VMs assigned by DID, not really. |
| 17:46.21 | mazpe | no vm |
| 17:46.42 | mazpe | i said, that i have one inbound context |
| 17:47.16 | mazpe | nevermind |
| 17:47.19 | mazpe | got an idea |
| 17:47.22 | xheliox | Neverminded. |
| 17:47.23 | mazpe | you are genius xheliox |
| 17:47.26 | mazpe | :) |
| 17:47.33 | *** join/#asterisk path_ (n=path_@pc-15-190-86-200.cm.vtr.net) |
| 17:48.01 | xheliox | belches |
| 17:48.16 | mazpe | sorry mate.. i'm slow thinker.. but a thinker :) |
| 17:48.36 | xheliox | I see you're from Florida, it's something in the water. |
| 17:48.52 | [TK]D-Fender | baliktad: * doesn't work that way since 1.4 |
| 17:49.03 | mazpe | its worse. I'm from Spain ;) living in florida now |
| 17:49.27 | xheliox | Geesh, another foreigner stealing my Asterisk gigs. The nerve. ;) |
| 17:49.35 | mazpe | hahaha |
| 17:49.44 | baliktad | this is on a 1.6 box but most of the configs are migrated from 1.4 and older |
| 17:50.06 | [TK]D-Fender | baliktad: type=peer has been appropriate since 1.4 |
| 17:50.45 | [TK]D-Fender | baliktad: Change it and things should start working |
| 17:52.01 | baliktad | brilliant, I never would have figured that out myself TK |
| 17:52.04 | baliktad | thank you very much |
| 17:52.31 | [TK]D-Fender | baliktad: You're welcome |
| 17:57.34 | xheliox | I am so freaking bored. |
| 17:57.43 | xheliox | Nothing worse than a holiday weekend with a car in the shop. |
| 17:58.23 | xheliox | Well, I suppose there's a lot worse, but I'm going to whine about how bad it is regardless. |
| 18:02.09 | eppigy | tunes his violin |
| 18:02.59 | xheliox | haha |
| 18:03.11 | [TK]D-Fender | <PROTECTED> |
| 18:03.15 | [TK]D-Fender | eppigy: Now its a duet! |
| 18:03.18 | eppigy | yes |
| 18:03.25 | eppigy | now we need a cello |
| 18:03.32 | eppigy | and we'll be set |
| 18:03.35 | xheliox | I play. |
| 18:03.53 | eppigy | xheliox: you will just weep alligtor tears while we play |
| 18:04.00 | eppigy | *alligator |
| 18:04.04 | xheliox | :( |
| 18:05.27 | *** join/#asterisk lowlevel (n=Stuart@lowlevel.ca) |
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| 18:14.49 | *** mode/#asterisk [+o bkruse] by ChanServ |
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| 19:00.20 | keith4_ | are the units on externrefresh... minutes? |
| 19:00.22 | keith4_ | seconds? |
| 19:00.40 | *** join/#asterisk eric256 (n=Administ@c-75-71-188-88.hsd1.co.comcast.net) |
| 19:01.10 | keith4_ | oh, it's seconds. the sample file has it as 10 seconds. yikes |
| 19:03.10 | eric256 | is there anyway to make sure that when the incoming caller hangs up all extensions that it was ringing will stop ringing? |
| 19:03.37 | [TK]D-Fender | eric256: This is what * already does |
| 19:04.11 | eric256 | then <insert dirty word for a specific version of asterisk here> is doing something annoying |
| 19:04.14 | eric256 | :) |
| 19:04.45 | eric256 | confesses to using trixbox :( |
| 19:05.50 | eppigy | LOL |
| 19:06.34 | eric256 | asked for help in that chat room, but they never seem to answer in there |
| 19:06.54 | keith4_ | shocking |
| 19:07.33 | lanning | what technology does the inbound call come from? |
| 19:07.35 | rob0 | The problem is that the blind tend to do a poor job in leading the blind. |
| 19:08.02 | *** join/#asterisk asterisk` (n=asterisk@unaffiliated/adrianxxx) |
| 19:08.38 | xheliox | rob0: Truth. |
| 19:10.23 | eric256 | yea i should build a scratch box, but i've been having a fair bit of fun with trix box and it is easy to nuke the machine and rebuild fast ;) (i'm lazy also) |
| 19:10.38 | eric256 | its an inbound call from vitelity (SIP) |
| 19:10.45 | xheliox | And look at where laziness gets you. |
| 19:11.23 | lanning | well, SIP is nice and explicit, so, it is not the detection of the hangup. |
| 19:12.04 | eric256 | that was odd, i just set verbose to 10 to see what was happening, and now it disconnected fine.../me scratches his head |
| 19:12.59 | *** join/#asterisk seb- (n=seb@li30-51.members.linode.com) |
| 19:13.16 | seb- | [TK]D-Fender: are you there? |
| 19:15.31 | eppigy | eric256: if you are just tinkering, ESXi and a snapshot would server your laziness better |
| 19:15.39 | eppigy | *serve |
| 19:15.53 | eppigy | with less undesirable side effects |
| 19:16.21 | rob0 | To be precise, he is always "there" unless he is somewhere else, in which case it's perfectly acceptable to refer to the other place as "there". So he is always "there," in a manner of speaking. |
| 19:16.55 | *** join/#asterisk |Cybex| (n=John@80.100.126.176) |
| 19:17.31 | rob0 | We'll have an instant replay just after the commercial break. Stay tuned. |
| 19:20.13 | *** join/#asterisk juanIMP (n=juan@201.244.250.91) |
| 19:20.13 | eric256 | or my internet just died...that makes more sense...frakin comcast |
| 19:23.34 | Urthwhyte | heh |
| 19:23.58 | Urthwhyte | Comcast FTL |
| 19:24.03 | Urthwhyte | Atleast they don't block ports |
| 19:24.28 | lanning | only if you have a business account. |
| 19:24.29 | eric256 | FTL? |
| 19:24.52 | xheliox | I think cable broadband is fundamentally unsustainable, regardless of the carrier. |
| 19:25.45 | eric256 | yea this one doesn't soo much go down, as just dropping 20% of the packets i send out...which gets pretty annoying |
| 19:25.47 | Urthwhyte | lanning: I was with them only a few months ago and they didn't block anything |
| 19:26.25 | eric256 | yea they arn't blocking anything here, though they do moan when i tell them i have my own router etc. |
| 19:26.49 | lanning | they block DNS |
| 19:27.03 | lanning | I have them right now, had to switch to business |
| 19:27.29 | *** part/#asterisk baliktad (i=baliktad@c-24-16-23-12.hsd1.wa.comcast.net) |
| 19:27.34 | rob0 | I have suspected them of blocking my DNS before, but I'm not sure. Last time it turned out my suspicion was wrong. |
| 19:28.09 | rob0 | I would gladly upgrade to business service if they had the least bit of clue about how to provide it. |
| 19:28.40 | eric256 | just found out today i could exchange the modem i got from them (current one is like 5 years old, doubt it was new when they gave it to me..) |
| 19:28.45 | lanning | I run a DNS server, and I support Seagate and Yahoo DNS. I could only query Comcast DNS resolvers, could not directly test any other servers. |
| 19:28.48 | eric256 | crosses his fingers and hopes that helps |
| 19:30.06 | rob0 | Oh you have to have a DNS server in a static IP netblock, else people might think you're a spammer. |
| 19:30.12 | eppigy | "malware propogation mitigation" |
| 19:30.14 | *** join/#asterisk propellerhead (n=yogurt2u@200.43.87.65) |
| 19:30.41 | rob0 | but, I find that on my residential Comcast, I can query specific DNS servers just fine. |
| 19:31.21 | lanning | what region are you in? |
| 19:31.29 | rob0 | north Alabama |
| 19:31.58 | lanning | I am in Silly-Con Valley :) |
| 19:32.53 | lanning | They might have different rules here. Being that everyone and his brother probably runs a server out of their house, here. |
| 19:33.54 | rob0 | I'm rural, and my Catcosm upstream node dies every time the power flickers. No power protection on their equipment. That's why I won't consider them for business service. They'll collect more money for the same lousy service. |
| 19:34.56 | rob0 | Mind you, I *have* been blocked resolving zen.spamhaus.org. in the past, so they might indeed do some experimenting. |
| 19:35.32 | lanning | ya, that would be crap. I have only seen upstream routing issues. Which, I did not see, when I was on their residential service. |
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| 19:58.47 | guilherme-jorge | hello all, I've a server running asterisk 1.6.0.9, and I've problems to use call files in asterisk. When I move call file to outgoing directory, and the call is started, I get the following error in asterisk CLI: "No audio format found to offer". Cancelling call to xxxxxxxx". Here (http://pastebin.com/m7462a894) is my sip.conf, and the call file. Someone can help me? |
| 19:59.34 | *** join/#asterisk hi365 (n=hi365@213.151.52.163) |
| 20:01.17 | guilherme-jorge | When I make a call using some extension with the same destiny (SIP/telintel/1111111111) everything works fine... |
| 20:01.23 | [TK]D-Fender | guilherme-jorge: Enable SIP debug and PB a failed attempt in full. |
| 20:01.41 | guilherme-jorge | [TK]D-Fender: ok |
| 20:06.39 | guilherme-jorge | [TK]D-Fender: http://pastebin.com/m1eab3e9f |
| 20:07.20 | |Cybex| | Hello all, is it possible to get a analog trunk (TDMoE) configured so it shows in GUI 2.0 as analog hardware? |
| 20:08.22 | [TK]D-Fender | guilherme-jorge: You seem to be offering ulaw/alaw. They don't even list what is supported and they are rejecting you |
| 20:08.56 | guilherme-jorge | [TK]D-Fender: I was doing some tests using ulaw and alaw, so dont consider this SIP debug... Ill post a new one... sorry |
| 20:09.09 | [TK]D-Fender | |Cybex|: GUI's not supported in this channel, they can be found at #asterisk-gui |
| 20:09.15 | |Cybex| | oops sorry |
| 20:09.32 | [TK]D-Fender | guilherme-jorge: Well I am surprised they doesn't accept either. |
| 20:22.15 | *** part/#asterisk |Cybex| (n=John@80.100.126.176) |
| 20:25.34 | guilherme-jorge | [TK]D-Fender: Now I've a strange problem.. If I let telintel user with option allow=g729 I cannot get anything in SIP debug, but if I configure allow=g729,ulaw I get data in SIP debug. Does it make sense for you? |
| 20:29.43 | [TK]D-Fender | guilherme-jorge: nope |
| 20:34.25 | *** join/#asterisk tamiel (n=tamiel@ip-176.net-89-2-117.rev.numericable.fr) |
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| 20:42.25 | nn | anyone unlucky enough to deal with voip.com? |
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| 20:48.10 | *** part/#asterisk jstoker (n=jstoker@80-41-183-84.dynamic.dsl.as9105.com) |
| 20:49.28 | keith4_ | hmmmm. any chance of using google voice + asterisk? |
| 20:51.46 | drmessano | Theres plenty of docs on it |
| 20:51.53 | drmessano | For inbound via Gizmo |
| 20:52.21 | keith4_ | how about outbound? |
| 20:52.23 | [TK]D-Fender | drmessano: But that might imply having done even a a basic Google search! Heaven forbid! |
| 20:52.32 | drmessano | No kiddin |
| 20:52.58 | keith4_ | [TK]D-Fender, you're such a better google proxy, though |
| 20:53.32 | [TK]D-Fender | keith4 : True, but so far I don't think it'll get tired of you :p |
| 20:53.41 | jaytee | hehe |
| 20:57.49 | keith4_ | I widh their was a page on Gizmo's website that explained their software/service in non-idiot terms. does using Gizmo get me anything that VoicePulse doesn't? |
| 20:59.22 | drmessano | I thought this was about Google Voice? |
| 20:59.42 | keith4_ | It is. But apparently that also includes using Gizmo |
| 21:00.20 | drmessano | It includes setting up a Gizmo account, not making them your ITSP |
| 21:00.31 | drmessano | Just set up and account, add it to Google Voice and be done with it |
| 21:00.36 | drmessano | an* |
| 21:03.21 | keith4_ | is Gizmo any good, as an ITSP? |
| 21:04.49 | keith4_ | it looks like their "Call In" numbers are pretty cheap |
| 21:09.59 | *** join/#asterisk Steve_J-obs (i=Steve_J-@pool-71-190-74-157.nycmny.east.verizon.net) |
| 21:10.32 | Steve_J-obs | hello everybody! |
| 21:13.31 | drmessano | keith4_: Dunno |
| 21:13.59 | keith4_ | looks like their outboung rates aren't great |
| 21:14.11 | drmessano | No, they suck |
| 21:14.24 | *** join/#asterisk NirS (n=NirS@77.127.107.162) |
| 21:14.25 | drmessano | I would rather use flowroute |
| 21:14.30 | NirS | Hi all |
| 21:14.38 | NirS | Hi Brandon |
| 21:14.44 | NirS | Hi Tilghman |
| 21:15.15 | NirS | is looking for someone who is familiar with main/pbx.c |
| 21:15.25 | keith4_ | drmessano, right. so, i'll stick with voicepulse for outbound, which includes unlimited free inbound... which then makes free inbound via Google Voice DID much less appealing |
| 21:15.34 | keith4_ | sighs |
| 21:16.35 | NirS | anybody home ? |
| 21:17.39 | *** join/#asterisk oej (n=olle@ns.webway.se) |
| 21:17.59 | drmessano | keith4_:First off, its forever.. Second, its free, and third, you can just get a FREE GIZMO ACCOUNT and point it there |
| 21:18.00 | jaytee | "We're sorry, but no one is available to take your chat at the moment. Please leave a message at the beep." exten = _X.,1,Playback(beep) |
| 21:18.05 | drmessano | Which is what I said earlier |
| 21:18.55 | NirS | is beeping |
| 21:21.12 | keith4_ | drmessano, is that any better than getting a DID from an ITSP and just point Google Voice at that? |
| 21:21.42 | keith4_ | I guess I don't see what benefit comes from throwing Gizmo into the mix |
| 21:22.31 | drmessano | I guess I don't make enough money from Google or Gizmo to continue to waste time trying to explain it. |
| 21:22.49 | drmessano | goes back to talking 4yr olds out of putting square pegs in round holes |
| 21:24.47 | keith4_ | I hope you're better at explaining pegs and holes than you are with Gizmo |
| 21:24.50 | *** join/#asterisk inv_arp (n=junya@b07s03mr.corenetworks.net) |
| 21:25.08 | drmessano | Its not really that damn hard |
| 21:25.25 | drmessano | Theres really no explaining needed.. maybe some prerequisite common sense |
| 21:25.53 | drmessano | Anyway.. pegs.. holes |
| 21:25.59 | drmessano | BBL |
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| 23:04.26 | CrazyTux | Hey guys I just installed asterisk addon for native mysql support, I did a load =>, and tried preload => as well for the addon and I do not see it when i restart asterisk and do 'module show like sql' |
| 23:04.31 | CrazyTux | any thoughts / ideas? |
| 23:31.40 | jaytee | try doing a module load and see what errors show in the CLI |
| 23:37.43 | CrazyTux | jaytee: good idea let me try |
| 23:41.16 | *** join/#asterisk smbrienz (n=guest@host202-204-dynamic.1-87-r.retail.telecomitalia.it) |
| 23:41.24 | CrazyTux | jaytee: libmysqlclient linked library was not in the path :) -- corrected it, and now all is well, thanks! |
| 23:41.52 | jaytee | CrazyTux, what distro you running? |
| 23:43.00 | CrazyTux | jaytee: centos |
| 23:43.18 | CrazyTux | in this particular case, I much prefer gentoo |
| 23:43.54 | jaytee | excellent howto that covers * install plus mysql setup and config for * on centos, p://www.voip-info.org/wiki/view/CentOS+5+and+Asterisk+1.4.x+installation |
| 23:44.08 | jaytee | oops, the link got hosed |
| 23:44.19 | jaytee | http://www.voip-info.org/wiki/view/CentOS+5+and+Asterisk+1.4.x+installation |
| 23:46.38 | *** join/#asterisk Bart[mdv] (n=bart@i.love.freenode.and.solarnet.ru) |
| 23:47.44 | Steve_J-obs | hi guys, anybody knows if it is possible to detect when a caller presses ## in the middle of a conversation? |
| 23:51.57 | *** join/#asterisk Ryushin (i=proxy@windwalker.openinnovations.com) |
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| 23:53.00 | [TK]D-Fender | Steve_J-obs I figure yes, if you setup a dynamic application map for it |
| 23:55.11 | jaytee | steals [TK]D-Fender's brain with all it's invaluable knowledge and sneaks away to his hidden lair to plot world domination {insert maniacal cackling here} |
| 23:57.27 | xheliox | jaytee: I'm sure as smart as he is, he's using an encryption partition. |
| 23:57.31 | xheliox | encrypted* |