| 00:10.06 | hamush | i have a very simple sip only setup. i want to be able to have two sip phones call each other. to do this, i understand i need to define two extensions in the dialplan. how should these extensions be defined? Answer(),Dial(),Hangup()? where does Ringing() fit in? |
| 00:10.23 | [TK]D-Fender | hamush: All you really need is Dial |
| 00:12.22 | hamush | [TK]D-Fender: i tried just using Dial, but my voip phone immediately disconnects... only by putting Answer() in the mix was i able to get a call established |
| 00:13.26 | [TK]D-Fender | hamush: Show us the problem an maybe we can do something about it |
| 00:21.00 | hamush | [TK]D-Fender: http://pastebin.com/m4f5df9f8 it looks like i'm getting a 404 unless i put Answer in the dialplan |
| 00:21.40 | [TK]D-Fender | hamush: That isn't the problem, pastebin the dialplan without it |
| 00:22.25 | hamush | [TK]D-Fender: http://pastebin.com/mbb2e4a1 |
| 00:22.49 | [TK]D-Fender | hamush: Problem is you have no **1** priority |
| 00:22.59 | hamush | [TK]D-Fender: doh! |
| 00:23.01 | [TK]D-Fender | hamush: You can't just have "n" like that, you need a "step 1" |
| 00:25.04 | hamush | [TK]D-Fender: thanks, that did it. i should have caught that :-/ |
| 00:25.22 | [TK]D-Fender | hamush: You seem to be learning. Keep it up |
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| 00:37.02 | *** join/#asterisk [T]ank (n=ckwall@c-71-199-25-239.hsd1.ut.comcast.net) |
| 00:37.34 | [T]ank | is there a way in extensions.conf to dial a sip channel using the peer username and pass instead of having to have a user set up in the sip.conf? |
| 00:37.58 | [T]ank | something like dial(SIP/user:pass@host/${EXTEN}) |
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| 00:39.01 | [TK]D-Fender | [T]ank: Exactly like that |
| 00:39.16 | dshap | hey all im trying to use pattern matching to allow a user to enter a time of day |
| 00:39.22 | dshap | so they would type "230" for 2:30 |
| 00:39.23 | [T]ank | hmmm. i must be doing something wrong then... gotta double check... thanks TK |
| 00:39.29 | dshap | or "1258" for 12:58 |
| 00:39.41 | dshap | i have pattern matching extensions for both 3 and 4 number inputs |
| 00:40.04 | dshap | but the issue is that if someone is slow typing in a 3 digit time |
| 00:40.07 | dshap | err |
| 00:40.08 | dshap | i mean a 4 digit time |
| 00:40.13 | dshap | then the 3 digit extension will kick in |
| 00:40.18 | dshap | so if someone SLOWLY types "1258" |
| 00:40.28 | dshap | it will go to the 3-digit "125" |
| 00:40.29 | dshap | get it? |
| 00:40.32 | dshap | what should ido? |
| 00:41.24 | [TK]D-Fender | dshap: Not accept 3-digit time |
| 00:41.43 | dshap | i dont wanna make people type "0230" for 2:30 though :-\ |
| 00:42.25 | dshap | i was thinking about using a wildcard, but then there are issues with the allowed digit |
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| 00:42.52 | dshap | the 2nd to last digit can only be [1-5] |
| 00:43.05 | dshap | the 2nd to last digit can only be [1-5] |
| 00:43.19 | dshap | so i cant just tack on a wildcard since the last digit can be [1-9] |
| 00:43.26 | dshap | (woah wtf sry dunno why i double posted that) |
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| 00:46.19 | dshap | maybe i could use the wait for digit AGI command |
| 00:46.26 | dshap | although that sounds like it will probably be more of a hassle than it is worth |
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| 00:49.39 | [T]ank | dshap: I missed what you were asking... are you trying to do pattern matching? |
| 00:51.22 | [TK]D-Fender | dshap: what is someone who wants to enter1:25 supposed to do in your mind? |
| 00:51.42 | dshap | [T]ank: yea just to accept time inputs |
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| 00:52.07 | dshap | if it's 11 AM, for example, someone who enters 125 would be specifying 1:25 PM that day |
| 00:52.18 | dshap | (i'm writing AGI logic that will handle this part of it, right now i'm just working on getting the input) |
| 00:52.39 | dshap | if it's 1:30 PM, then 125 would mean 1:25 AM later that night/next day |
| 00:53.36 | dshap | i don't think there is any good way to do this with pattern matching |
| 00:53.47 | dshap | since the first 3 digits of a 4-digit input are valid also for a 3-digit input |
| 00:53.52 | dshap | i.e. 1250 and 125 |
| 00:54.22 | [TK]D-Fender | dshap: So you're saying that it should take the 125 as 1:25 immediately? |
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| 00:55.13 | [TK]D-Fender | dshap: Because how long SHOULD * wait for a 4th digit? |
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| 00:55.48 | Borai | Hello, anyone using lumenvox on 64bit os? |
| 00:56.16 | dshap | [TK]D-Fender: well what if i want them to press pound when they are done? |
| 00:56.24 | dshap | enter time and press # |
| 00:56.30 | dshap | can i put # in the extension? |
| 00:56.57 | [TK]D-Fender | dshap: yes |
| 00:57.03 | dshap | i think that would solve my problem |
| 00:57.09 | dshap | i could put the rules for 3-digit followed by # |
| 00:57.20 | dshap | and the same for 4 |
| 00:57.22 | dshap | ok this is cool |
| 00:57.23 | dshap | thanks |
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| 00:58.11 | [TK]D-Fender | dshap: Only problems you have are the ones you're creating |
| 00:58.18 | dshap | lol |
| 00:58.20 | dshap | story of my life |
| 00:58.32 | dshap | my project is gonna be sweet when it's done though |
| 00:58.33 | dshap | trust me |
| 00:58.53 | dshap | i think ppl are really gonna use it |
| 00:58.59 | dshap | maybe |
| 01:00.27 | dshap | [TK]D-Fender: is there any way to increase the amount of time asterisk waits before it goes to the invalid extension? |
| 01:00.46 | [TK]D-Fender | dshap: "core show function TIMEOUT" |
| 01:00.46 | dshap | right now if i press SLOWLY 1250#, then it will think i'm just pressing 1250 which is invalid |
| 01:00.58 | dshap | it's not the timout extension though, it's the invalid extension |
| 01:01.00 | Borai | dshap: what are you creating? |
| 01:01.07 | dshap | Borai: secret :) |
| 01:01.34 | Borai | lol then nobody will use it |
| 01:02.00 | Borai | WARNING[19762]: loader.c:375 load_dynamic_module: Error loading module 'res_speech_lumenvox': /usr/lib/asterisk/modules/res_speech_lumenvox.so: wrong ELF class: ELFCLASS32 - what can I do about this? |
| 01:02.01 | dshap | Borai: well it won't be secret once it's done, obviously lol |
| 01:02.13 | dshap | Borai: i just don't want to give anyone else the idea |
| 01:02.30 | dshap | Borai: i'm a huge asterisk n00b and so if someone actually knows what they are doing and got the idea, they could easily beat me to the chase |
| 01:02.55 | Borai | oh |
| 01:02.59 | dshap | Borai: but the basic plan is that i develop an amateur yet functional prototype and if it gets popular then i hire a pro to make it legit |
| 01:03.07 | dshap | and scalable |
| 01:04.16 | dshap | [TK]D-Fender: is there any way to have a different timeout for different contexts? |
| 01:05.36 | Borai | Is there a way to make asterisk that is compiled on a 64bit os as a 64bit binary, load a 32bit module? |
| 01:06.29 | dshap | actually |
| 01:06.30 | dshap | nevermind |
| 01:06.32 | dshap | this is working fine |
| 01:07.33 | [TK]D-Fender | dshap: it isn't context based |
| 01:09.42 | dshap | [TK]D-Fender: got it |
| 01:10.45 | dshap | [TK]D-Fender: ok, thanks for the help. im goin out to a bar with some kids to party - it's the eve of the 4th of july! |
| 01:10.53 | dshap | ttyl |
| 01:13.44 | Borai | there must be a way, other than recompiling asterisk :( |
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| 01:44.52 | Olobola | after installing speex via yum and asterisk make install, codec_speex.so isn't anywhere to be found. I found codec_speex.c in /codecs, no .so. Where do I go from here? |
| 01:50.46 | Borai | make menuconfig have you selected it there? |
| 01:52.21 | Olobola | under 12. Core Sound Packages ? |
| 01:52.40 | Olobola | I don't see it in there anywhere |
| 01:52.46 | jaytee | I'd look under codecs since it's not a sound package |
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| 01:54.25 | Olobola | ohh.. XXX Depends on: speex(E), speex_preprocess(E) ? |
| 01:54.30 | Olobola | can't select it |
| 01:54.36 | Borai | install the libs |
| 01:54.59 | Borai | doesn't anyone know a workaround for my issue? |
| 02:01.19 | Olobola | Borai: do you mean speex devel? |
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| 02:21.50 | Borai | yes u need the devel and lib packages |
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| 02:25.33 | Olobola | I yum installed speex and speex-devel, still not building codec_speex.so for some reason |
| 02:30.45 | Borai | libspeex? |
| 02:31.02 | drmessano | libogg-devel |
| 02:31.41 | drmessano | sorry |
| 02:31.58 | drmessano | libvorbis and libvorbis-devel |
| 02:31.58 | drmessano | brb, dinner |
| 02:36.38 | russellb | make sure you re-run configure after installing the packages |
| 02:36.49 | russellb | also, check to see if there is a separate speexdsp and speexdsp-devel package. |
| 02:40.22 | Olobola | @russellb: I did not run ./configure several times while trying to get it to install.. just make clean, make, make install. |
| 02:40.42 | Olobola | I will try again |
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| 02:44.05 | Borai | what should I do? I couldn't figure out a way to get it running |
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| 02:55.55 | kyoshi | If anyone can help with an ENUM problem with asterisk, I would deeply appreciate it... http://pastebin.com/m15f92909 |
| 02:56.00 | kyoshi | thanks |
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| 02:57.22 | Borai | why is it that asterisk cant cant load a 32 bit module when it runs as 64bit |
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| 03:01.08 | Olobola | @russellb: wahoo! I removed my yum installation of speex and built from source, it's working now. Thanks to all :) |
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| 03:17.25 | dwschool | anyone here familier with faxmail script |
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| 03:32.11 | dwschool | is there a way to get the subject value from faxmail for faxing? |
| 03:38.24 | [TK]D-Fender | kyoshi: "The problem is that it checks e164.arpa" |
| 03:38.36 | [TK]D-Fender | kyoshi: [Jul 3 17:35:59] DEBUG[23514]: enum.c:426 ast_get_enum: ast_get_enum(): n='+13015611020,sip,,1,freenum.org', tech='sip', suffix='e164.arpa', options='0', record='1' |
| 03:38.47 | [TK]D-Fender | kyoshi: That isn't ARPA, thats the FIRST one. |
| 03:39.23 | [TK]D-Fender | kyoshi: And of course it isn't going to look at another |
| 03:40.26 | [TK]D-Fender | kyoshi: exten => _X.,2,GotoIf($["${eres}"=""]?11:14) <-- you immeditely SKIP looking up any of the others with this (probably sloppy copy&pasted) GTOT |
| 03:40.29 | [TK]D-Fender | GOTO |
| 03:40.54 | [TK]D-Fender | kyoshi: 3-10 will never get executed |
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| 03:44.10 | Borai | :( |
| 03:49.48 | kyoshi | tkd fender, even if only 1 was in there, it fails. i tried building it step by step |
| 03:50.08 | kyoshi | freenum.org still it tries to check e164.arpa and thats the problem im having |
| 03:50.40 | kyoshi | so even if i specify which registrar to us and i only have 1 line, it still checks e164.arpa and not the registrar specified |
| 03:50.49 | *** part/#asterisk [T]ank (n=ckwall@c-71-199-25-239.hsd1.ut.comcast.net) |
| 03:50.56 | kyoshi | lemme show ya something else i tried |
| 03:52.22 | kyoshi | http://pastebin.com/m16bcd3b9 |
| 03:52.35 | kyoshi | same results, it tries to connect with e164.arpa not freenum.org |
| 03:54.51 | kyoshi | i love coconut |
| 03:57.34 | kyoshi | [23:38] <[TK]D-Fender> kyoshi: exten => _X.,2,GotoIf($["${eres}"=""]?11:14) <-- you immeditely SKIP looking up any of the others with this (probably sloppy copy&pasted) |
| 03:57.58 | kyoshi | actually thats very good coding, minimizes having tons of DIAL commands in there and wraps it together in a single "sub" if you will |
| 03:58.21 | kyoshi | they all use 11:14 cause 11 if true, 14 if false. |
| 03:58.24 | [TK]D-Fender | kyoshi: The priorities 3-10 will never get called. |
| 03:58.40 | [TK]D-Fender | kyoshi: Su unless you intentially crippled them I don't see the point |
| 03:58.42 | kyoshi | no need to if ENUM worked :-p |
| 03:58.56 | [TK]D-Fender | kyoshi: You jump away from those other extens REGARDLESS |
| 03:59.00 | kyoshi | i dont want it to go thru each one, i wanted to know WHY enum was not using what I wanted |
| 03:59.04 | kyoshi | i know what you're saying |
| 03:59.08 | [TK]D-Fender | kyoshi: Doesn't matter what, 3-10 will never get called |
| 03:59.15 | kyoshi | it should be 11:3 then 11:5 and so on |
| 03:59.30 | kyoshi | but for testing why its not calling freenum.org i rewrote it |
| 04:00.00 | kyoshi | as i said...http://pastebin.com/m16bcd3b9 is the simpler version and it still calls e164.arpa not freenum.org |
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| 04:08.25 | [TK]D-Fender | kyoshi: Ok, not having used ENUM myself I have hit the limits of what I can advise based on a read of the docs & configs & your dialplan |
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| 04:13.24 | kyoshi | hehehe |
| 04:14.02 | kyoshi | its ok, i tried. you've always been good with helping me. i know that dialplan i showed you was messy thats why i showed you the one im using for testing. i appreciate the effort though. |
| 04:14.30 | kyoshi | enum "would be" cool if providers actually registered their DID's with ripe but of course, they dont cause they want everyone to use their pstn's |
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| 04:35.13 | dwschool | anyone out there that uses hylafax-could you give me a hint on how i could setup my email to fax gateway so that the dest fax number is retrieved from the subject line in the email? |
| 04:37.16 | [TK]D-Fender | dwschool: This isn't #hylafax. |
| 04:37.52 | dwschool | i know i thought maybe someone here could help me |
| 04:37.58 | dwschool | sorry |
| 04:38.20 | dwschool | maybe you know where is should be looking/what docs i should be reading :D |
| 04:38.31 | coppice | don't be sorry. be bloody bold and resolute |
| 04:39.29 | dwschool | can't hurt to try |
| 04:39.37 | [TK]D-Fender | dwschool: www.hylafax.org |
| 04:40.02 | dwschool | [TK]D-Fender - been over and over it |
| 04:40.26 | dwschool | i just need a way to get the subject line of the email to pass to sendfax or faxmail |
| 04:40.33 | kyoshi | dwschool, the problem with open source is that most of the time is spendt coding, not documenting :( |
| 04:40.53 | dwschool | well that's why i'm asking |
| 04:41.07 | kyoshi | check to see if there is a #hylafax |
| 04:41.37 | dwschool | there is - i'm on it but no one's every there - there are 10 members |
| 04:42.07 | dwschool | there where a few members here yesterdau who said they where running hylafax |
| 04:43.40 | [TK]D-Fender | dwschool: Yes and just because I use FireFox doesn't make this a FireFox support channel. |
| 04:44.07 | dwschool | ok point taken |
| 04:44.38 | coppice | asterisk + iaxmodem + hylafax is a very popular combination. saying questions related to setting that up are inappropriate is dumb |
| 04:46.22 | dwschool | well that's why i figured someone would help me |
| 04:46.35 | dwschool | would/could |
| 04:47.01 | dwschool | my setup is working awesome is just a final touch i'm missing |
| 04:47.11 | [TK]D-Fender | coppice: Yes, but this takes "transitive" well outside the scope of *'s involvelemnt. |
| 04:48.24 | dwschool | yeaj but considerins asterisks broad usage and open sources huge envelope one could argue relevance. |
| 04:54.54 | [TK]D-Fender | dwschool: And apparently one could parkan 18 wheeler in your envelope... |
| 04:56.09 | dwschool | yup - i can back a super-c tractor trailer into a envelope - had to back one for a mile yesterday |
| 04:57.49 | coppice | anyone who drives an artic backwards for a mile lacks a sense of direction |
| 05:10.37 | Olobola | after installing speex: "Unable to find a codec translation path from 0x200 (speex) to 0x2 (gsm)". This happens after calling voicemail. |
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| 05:16.22 | dwschool | well u've never been in tight corners with a big rig |
| 05:22.28 | [TK]D-Fender | Olobola: Did you recompile & install *? |
| 05:22.44 | Jumpie | dwschool ugh f that |
| 05:22.49 | Jumpie | i'd hate driving trucks that long |
| 05:23.02 | Jumpie | i was a nervous wreck even in the military in drivers school with the hitch trailer on the hummer |
| 05:23.03 | Jumpie | :D |
| 05:23.21 | Jumpie | and now i see these truckers with 2, sometimes 3 trailers on it..ffs thats skill |
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| 05:41.03 | kyoshi | tkd-fender, i found out why it wasnt working and i aint gonna fix it |
| 05:41.57 | kyoshi | enum.c is written half ass and its gonna be hell figuring out why the functions werent written complete |
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| 06:21.57 | Borai | ugh ugh ugh |
| 06:25.02 | Borai | so I have to recompile in 32bit mode :S |
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| 07:36.58 | Borai | <PROTECTED> |
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| 08:06.23 | Borai | :S cool now cflags=-m32 doesnt work :S |
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| 09:39.37 | b14ck | on a clean asterisk 1.6 install, there is no mysql db dependencies, correct? |
| 09:40.07 | russellb | correct |
| 09:40.13 | russellb | unless you want to use some of the mysql addons :-) |
| 09:40.20 | russellb | or mysql via odbc |
| 09:41.29 | b14ck | I'm still such an asterisk noob =p |
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| 09:41.40 | russellb | it's all good |
| 09:41.43 | russellb | everyone starts there. |
| 09:41.56 | russellb | ~thebook |
| 09:41.57 | infobot | extra, extra, read all about it, thebook is Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://astbook.asteriskdocs.org |
| 09:42.00 | russellb | have that? |
| 09:42.07 | b14ck | yea, read through it fully |
| 09:42.12 | russellb | excellent |
| 09:42.31 | b14ck | im just getting into the internals more now, and learning a lotta nwe things |
| 09:42.43 | russellb | having fun? :-) |
| 09:42.49 | b14ck | my first experience with asterisk was through trixbox. so i missed out on a lot |
| 09:42.58 | b14ck | oh ya, <3 asterisk, you guys did a great job ^^ |
| 09:43.09 | russellb | thanks! |
| 09:43.48 | b14ck | atm i have a 64 bit centos box with asterisk 1.6 installed |
| 09:43.59 | b14ck | and im writing my own dialplans, configuring everything from scratch |
| 09:44.16 | b14ck | trying to get the AMI up and running using tls |
| 09:48.04 | Borai | i have a question |
| 09:48.16 | Borai | I am trying to compile 32bit asterisk in 64bit everything went fine with dahdi and stuff |
| 09:48.18 | russellb | i have an answer |
| 09:48.44 | Borai | but when i run CFLAGS="-m32" make i get this |
| 09:48.58 | Borai | CC="cc" CXX="g++" LD="" AR="" RANLIB="" CFLAGS="" make -C menuselect CONFIGURE_SILENT="--silent" makeopts |
| 09:49.07 | Borai | the CFLAGS does not append to it |
| 09:49.16 | russellb | that's just menuselect though |
| 09:49.20 | Borai | and chan_dahdi.so doesnt compile |
| 09:49.50 | Borai | http://www.pastebin.ca/1483725 |
| 09:50.08 | russellb | ASTCFLAGS="-m32" make |
| 09:50.09 | russellb | try that |
| 09:50.12 | Borai | that one was just a make try without the CFLAGS but its the same output |
| 09:50.24 | Borai | russellb: same |
| 09:50.31 | Borai | I even tried CFLAGS and ASTCFLAGS |
| 09:50.43 | russellb | ASTCFLAGS="-m32" make NOISY_BUILD=yes |
| 09:50.46 | russellb | and see if -m32 is in there |
| 09:51.07 | Borai | no its still not there :S |
| 09:51.13 | russellb | fail! |
| 09:51.15 | russellb | k, let me look ... |
| 09:51.47 | russellb | make COPTS="-m32" NOISY_BUILD=yes |
| 09:51.51 | russellb | see if that does it ... |
| 09:52.38 | Borai | FAIL! |
| 09:52.41 | russellb | hehe |
| 09:52.49 | russellb | i hate build system stuff |
| 09:53.02 | Borai | im going through this just because lumenvox techs are lazy bums |
| 09:53.05 | Borai | and dont create a 64bit connector |
| 09:53.20 | russellb | or you could install a 32-bit OS |
| 09:53.29 | russellb | and then this problem goes away ^_^ |
| 09:53.42 | Borai | no! |
| 09:53.53 | Borai | how about 128bit? lol |
| 09:54.04 | russellb | what version btw ... |
| 09:54.13 | russellb | meh, should be the same regardless |
| 09:54.24 | russellb | i give up for now. |
| 09:54.24 | Borai | 1.6.1.1 |
| 09:54.25 | Borai | i think |
| 09:54.29 | Borai | centos 5.3 final |
| 09:54.58 | Borai | :( |
| 09:55.13 | russellb | well, try just hacking the top level Makefile |
| 09:55.20 | russellb | find an ASTCFLAGS+= line |
| 09:55.22 | russellb | and add it there |
| 09:55.32 | russellb | maybe we're doing something that is messing up passing in those values ... |
| 09:55.41 | russellb | (and do a make clean btw) |
| 09:56.07 | Borai | make clean doesnt clean the menuselect |
| 09:56.16 | russellb | who cares about menuselect |
| 09:56.17 | Borai | i figured that out recently |
| 09:56.20 | russellb | you just need 32 bit asterisk |
| 09:56.20 | Borai | oh just saying |
| 09:56.24 | russellb | ah |
| 09:56.26 | russellb | it should ... |
| 09:56.36 | russellb | it does, i see it! |
| 09:56.40 | russellb | or at least it tries. |
| 09:57.04 | Borai | ok lets try |
| 09:57.48 | Borai | omg am i stupid |
| 09:58.22 | russellb | no, just special |
| 09:58.25 | Borai | :D |
| 09:58.36 | Borai | ok i did that too and its not working either im really stupid or |
| 09:58.39 | Borai | really special :P |
| 09:59.03 | russellb | It's not there in the NOISY_BUILD output? |
| 09:59.17 | Borai | NOISY_BUILD output? |
| 09:59.25 | russellb | make NOISY_BUILD=ye |
| 09:59.32 | russellb | so you can see if your CFLAG is being used |
| 10:00.30 | russellb | is surprised how many vendors still haven't gotten with the program on 64-bit support ... |
| 10:00.41 | russellb | it's not like 64-bit is brand new anymore |
| 10:01.47 | Borai | :D |
| 10:01.47 | Borai | and im mad |
| 10:01.47 | Borai | because today I paid my reneweal fees |
| 10:01.47 | Borai | :@ |
| 10:02.03 | Borai | nwhat does make NOISY_BUILD=ye actually do? |
| 10:02.08 | Borai | thats =yes i think |
| 10:02.21 | russellb | same thing actually |
| 10:02.24 | russellb | just needs to be non-empty |
| 10:02.34 | russellb | make NOISY_BUILD=YESNOWWORKDAMNIT |
| 10:02.38 | russellb | that works, too |
| 10:03.19 | Borai | no:S |
| 10:03.52 | russellb | blinks |
| 10:04.03 | russellb | pastebin "svn diff" or something that shows where you put -m32 |
| 10:04.08 | russellb | and also some of the build output |
| 10:04.32 | Borai | svn diff its not svn |
| 10:04.44 | russellb | ok, just copy a few lines from the Makefile where you put it |
| 10:04.51 | Borai | ok |
| 10:07.20 | Borai | http://pastebin.ca/1483735 thats the CFLAGS="-m32" ./configure |
| 10:09.42 | Borai | ASTCFLAGS+=$(COPTS) I change this line to |
| 10:09.52 | Borai | ASTCFLAGS+="-m32" right? |
| 10:11.35 | russellb | sure |
| 10:11.49 | Borai | then I run CFLAGS="-m32" make right |
| 10:11.58 | russellb | or just "make" |
| 10:12.07 | russellb | and preferably "make NOISY_BUILD=yes" so we can see what is happening |
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| 10:16.34 | Borai | http://pastebin.ca/1483741 |
| 10:17.05 | wdoekes | CXXFLAGS="-m32" ? |
| 10:17.38 | Borai | should I add that too? |
| 10:17.43 | Borai | in the command line or makefile? |
| 10:17.58 | wdoekes | before configure |
| 10:18.08 | Borai | oh |
| 10:19.53 | Borai | again at dahdi its exitintg |
| 10:20.09 | Borai | /usr/bin/ld: skipping incompatible /usr/lib/libtonezone.so when searching for -ltonezone |
| 10:22.32 | russellb | maybe -m32 is supposed to go in ASTLDFLAGS, too |
| 10:22.33 | russellb | probably is |
| 10:23.08 | Borai | ok |
| 10:23.35 | Borai | ASTCFLAGS="-m32" ASTLDFLAGS="-m32" CXXFLAGS="-m32" CFLAGS="-m32" make? |
| 10:26.34 | Borai | same warnings appeared again |
| 10:26.59 | russellb | can you paste the output again? |
| 10:27.02 | Borai | yes |
| 10:27.46 | Borai | ops nevermind :S I accidentally wrote make cleean thats why it happened i think its compiling |
| 10:28.03 | Borai | if i dont do menuselect will it compile everything? |
| 10:28.09 | Borai | I wanted to enable gtalk |
| 10:28.30 | russellb | it will compile everything that dependencies were met for (and is set to build by default) |
| 10:28.35 | russellb | very few things are not set to build by default |
| 10:28.46 | Borai | usr/bin/ld: skipping incompatible /usr/lib64/libgssapi_krb5.so when searching for -lgssapi_krb5 |
| 10:28.53 | Borai | i think krb5 32bit libraries are missing |
| 10:28.56 | russellb | yep |
| 10:29.28 | russellb | # |
| 10:29.28 | russellb | /usr/bin/ld: warning: i386 architecture of input file `md5.o' is incompatible with i386:x86-64 output |
| 10:29.33 | russellb | are you still getting those warnings, though? |
| 10:29.47 | russellb | hopefully not, and you're incredibly close. |
| 10:30.02 | Borai | no i dont think so |
| 10:30.07 | russellb | xlnt. |
| 10:30.15 | Borai | ill post the log once done |
| 10:33.44 | Borai | Error: Missing Dependency: krb5-libs = 1.6.1-31.el5 is needed by package krb5-devel-1.6.1-31.el5.i386 (base) ...... Package krb5-libs-1.6.1-31.el5_3.3.i386 already installed and latest version |
| 10:33.54 | Borai | never seen a el5_3.3 :D |
| 10:34.53 | russellb | enterprise linux 5.3 |
| 10:34.58 | russellb | RHEL |
| 10:35.18 | Borai | so i remove that package and install el5.i386? rather than el5_3.3 |
| 10:35.19 | russellb | i need to try to get some sleep ... |
| 10:35.25 | Borai | thanks russellb |
| 10:35.30 | russellb | yeah |
| 10:35.32 | russellb | you're welcome |
| 10:35.36 | russellb | good luck to you |
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| 10:48.31 | Borai | erm it wont like the libs that are installed |
| 10:57.30 | Borai | its looking for /lib64 first how can i change that |
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| 11:18.09 | joako | Borai: If you are trying to compile 32-bit dahdi on 64-bit system you need 32-bit devel/source packages |
| 11:18.57 | Borai | i have all packages |
| 11:18.59 | Borai | its not dahdi |
| 11:19.02 | Borai | its the asterisk |
| 11:19.11 | Borai | and i got everything to work the files are there |
| 11:19.22 | Borai | the dev packages are installed but it searches /usr/lib64 first |
| 11:23.53 | tzafrir_laptop | Borai, what exactly are you trying to do? |
| 11:24.28 | Borai | compile asterisk in 32bit mode on a 64bit system |
| 11:24.33 | Borai | so that I can use lumenvox |
| 11:24.41 | Borai | lumenvox doesnt have a 64bit connector :( |
| 11:25.10 | tzafrir_laptop | that system is centos? |
| 11:25.20 | Borai | yes 5.3 64bit |
| 11:36.03 | coppice | the lack of 64 bit libraries for various things is weird. who runs 32 bit on a server these days? |
| 11:36.19 | Borai | stupid people |
| 11:36.28 | Borai | or those who still have old hardware |
| 11:36.31 | Borai | that doesnt support 64 |
| 11:37.07 | Borai | just because I Want speech recognition i have to compile 32bit version of asterisk how fun eh |
| 11:39.33 | Borai | bah need to sleep |
| 11:39.33 | joako | Borai: Have you tried a 32-bit RPM of Asterisk? |
| 11:39.40 | Borai | no |
| 11:39.50 | Borai | i will try that in the worst case |
| 11:39.57 | Borai | but i need some sleep or i will do a rm -rf / |
| 11:40.10 | joako | cat /dev/zero > /dev/sda is funner |
| 11:40.21 | Borai | oh |
| 11:40.30 | Borai | is that the 2k10 version of rm -rf / |
| 11:40.46 | joako | it wipes your partition table too |
| 11:41.03 | Borai | not really |
| 11:41.06 | Borai | i still have IDE :P |
| 11:41.29 | Borai | good luck trying to wipe /dev/sda i think thats my sata cdrom drive :P |
| 11:41.31 | Borai | lol |
| 11:41.43 | Borai | writing all zeroes to a drive thats fun |
| 11:42.33 | *** join/#asterisk degrade (n=degrade@unaffiliated/degrade) |
| 11:43.06 | Borai | night/morning whatever;wherever you guys are |
| 11:43.11 | Borai | thanks for all the help |
| 11:45.27 | tzafrir_laptop | coppice, 32bit libraries take less download space :-) |
| 11:49.35 | coppice | but they run slower |
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| 13:36.20 | joako | Anyone know how I can set a hostname for Polycom IP phones to better manage on the DHCP server? Right now they send DHCP no hostname |
| 13:38.20 | [TK]D-Fender | joako: MAC not good enough? |
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| 13:40.53 | joako | [TK]D-Fender: It would be nice to set the hostname e.g to extension Nr. |
| 13:41.09 | joako | Polycom-501-EXT300 or something like that... |
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| 13:54.17 | *** mode/#asterisk [+o Deeewayne] by ChanServ |
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| 15:01.47 | Stdht | Hi |
| 15:02.29 | Stdht | Does SVN http://svn.digium.com/svn/asterisk/branches/1.4 corresponds to Asterisk 1.4.26-rc5 released?? |
| 15:03.48 | Stdht | ??? |
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| 15:24.46 | LemensTS | if someone calls my att cell phone from spain, would that cost me international rates for receiving it? |
| 15:25.02 | LemensTS | It is a weekend of course for me |
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| 15:45.57 | beek | Does anyone have a clever way to tell if a device exists? e.g. If someone dials 5555 how can I tell if 5555 exists without dialing? DEVICE_STATE returns NOT_INUSE for an invalid device. |
| 15:48.24 | beek | Ahhh... just found ChanIsAvail.... testing now. |
| 15:49.40 | jaytee | that's what I was thinking but since it's the channel and not specifically the device I wasn't sure if it would give you what you need |
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| 16:04.47 | beek | Hi jaytee ... |
| 16:04.58 | Olobola | age old question: no dtmf tones when calling real numbers. Is this a provider issue? |
| 16:05.37 | russellb | maybe. :-) |
| 16:17.16 | dwery | Anyone found a way to programmatically load a phonebook to a Siemens C470IP? |
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| 16:19.47 | *** join/#asterisk Kobaz (n=kobaz@its.kobaz.net) |
| 16:20.13 | Kobaz | every time i have nameserver issues, asterisk completely freaks out and drops local phone registrations |
| 16:20.30 | Kobaz | even though all the phones aren't set up to even use dns |
| 16:23.32 | russellb | it's because Asterisk will hang on a DNS lookup |
| 16:23.52 | russellb | that situation has been improved in 1.6 |
| 16:27.09 | *** join/#asterisk SkywaIker (n=pirch@58.147.17.166) |
| 16:27.37 | *** join/#asterisk styelz (n=yoohoo@m0o0.mooo.com) |
| 16:30.08 | Kobaz | yeah i'm running 1.6.0.6 |
| 16:32.58 | *** join/#asterisk moy (n=moy@CPE001cdfec4cee-CM00080dab8485.cpe.net.cable.rogers.com) |
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| 16:41.27 | russellb | Kobaz: do you have the dns manager enabled? /etc/asterisk/dnsmgr.conf |
| 16:44.08 | Kobaz | hmm |
| 16:44.12 | Kobaz | probalby not |
| 16:44.25 | Kobaz | nope |
| 16:45.10 | Kobaz | hmm |
| 16:45.22 | Kobaz | will this also fix timing issues on dns lookups? |
| 16:45.29 | russellb | timing issues? |
| 16:45.38 | Kobaz | ie: system starts, and asterisk starts up, and can't look up hosts with dns |
| 16:45.55 | russellb | well. for some modules it will, yes |
| 16:45.57 | Kobaz | and it will never retry, so all your hosts by dns are unreachable |
| 16:46.00 | Kobaz | sip/iax |
| 16:46.01 | russellb | not all of asterisk uses it today |
| 16:46.07 | russellb | sip/iax both use it |
| 16:46.30 | Kobaz | but if asterisk was started like, a minute later, it would have worked fine, since dns was up and going |
| 16:46.33 | Kobaz | nifty |
| 16:46.49 | russellb | though both still have to do direct lookups at times, but during call processing i think |
| 16:47.14 | Kobaz | i start my nameserver before asterisk, but half the time the first several queries will time out |
| 16:47.58 | Kobaz | we lost internet for a bit the other day, and all the phones lost their registrations |
| 16:49.00 | russellb | hopefully this will improve that .. |
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| 16:49.06 | Kobaz | ah yeah, i see.. the dns manager does periodic refreshes |
| 16:49.51 | Kobaz | so if the dns lookup fails at first, it should theoretically look it up again |
| 16:50.52 | *** join/#asterisk ingenius (n=alektro@186.136.6.218) |
| 16:52.05 | Kobaz | i've always set up a script in cron to run every so often to do an iax/sip reload |
| 16:52.09 | Kobaz | to fix any dns issues |
| 16:54.14 | *** join/#asterisk lesouvage (n=lesouvag@82.73.69.76) |
| 16:57.26 | rob0 | speaking of DNS, I'm seeing a lot of strange queries being rejected, which I think came from my new * system. I thought I disabled dundi, not sure where else to look. |
| 16:58.25 | rob0 | The reason I suspect asterisk, apart from it being the only thing running, is that queries seem to be associated with Mark/Digium. |
| 16:58.37 | rob0 | (marko.net being one) |
| 16:59.30 | lesouvage | I'm looking for sip eneabled hardware not being a phone to experiment with devstate to switch f.i. lights by dialing a number . Can somebody please point me to info about the hardware needed for this. |
| 17:00.19 | *** join/#asterisk tAnk (i=tank@the.matrix.has-you.net) |
| 17:00.51 | russellb | rob0: grep /etc/asterisk |
| 17:01.52 | rob0 | voicemail.conf:;4310 => -5432,Sales,sales@marko.net |
| 17:02.00 | rob0 | thanks |
| 17:02.23 | rob0 | that was the only *.conf I hadn't gotten to yet |
| 17:02.52 | russellb | that's still weird though, a commented out line shouldn't do a lookup of marko.net |
| 17:03.02 | russellb | further, marko.net isn't in the code anywhere |
| 17:04.01 | rob0 | oh hmm, I missed the ; on that |
| 17:04.34 | rob0 | I have a feeling it might be some NS queries, those would be hard to find. |
| 17:08.31 | *** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 17:13.12 | rob0 | other suspicious queries look like they're to alabama.{edu,gov} sites. I might have inadvertently left a browser running, but I don't remember using it to go to Digium. I will kill the browser and see if the queries stop. |
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| 17:37.26 | *** join/#asterisk ghatak (n=dontknow@5ac72e01.bb.sky.com) |
| 17:37.30 | ghatak | Hi, I need little bit of help with SIP trunk and Conferencing |
| 17:37.32 | ghatak | I want an incomming call to be forwarded to a conference room that I have setup, the bit that works is = 1. Trunk is registered and calls come in fine, the conference room works for locally registered users. I want to forward all call coming in on the SIP trunk to conference room. That bit does not work |
| 17:38.49 | ghatak | How do I achieve this, ? |
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| 17:48.41 | *** join/#asterisk tarapuez1 (n=tarapuez@190-14-254-194.ip.mediacommerce.com.co) |
| 17:49.12 | tarapuez1 | Hey everybody |
| 17:49.15 | *** join/#asterisk lesouvage (n=lesouvag@82.73.69.76) |
| 17:49.59 | tarapuez1 | hi |
| 17:51.42 | *** join/#asterisk juanIMP (n=juan@200.71.41.254) |
| 17:52.40 | tarapuez1 | hey we made a asterisk - panasonic integration - we are connecting panasonic and asterisk using a digium te110p card. |
| 17:55.16 | tarapuez1 | asterisk connect to pstn using a sip trunk. the problem is that dtmf tones in outgoing calls from panasonic no arriving to the sip chanel |
| 17:55.54 | *** join/#asterisk errotan (n=errotan@5403E569.catv.pool.telekom.hu) |
| 17:57.00 | tarapuez1 | we made a debug and we can see received dtmf tones from panasonic in Dahdi channel but we dont see dtmf tones in the Sip trunk. |
| 17:58.41 | *** join/#asterisk selinuxium (n=james@unaffiliated/selinuxium) |
| 17:59.49 | tarapuez1 | hi |
| 18:00.07 | selinuxium | Hi all, I am new to asterisk and would just like to ask some questions... Firstly... 1.4 or 1.6... I use ubuntu and 1.4 is in the repo, though I don't mind compiling... |
| 18:00.53 | *** join/#asterisk mltlnx (n=mltlnx@cpe-24-193-11-14.nyc.res.rr.com) |
| 18:06.49 | tarapuez1 | any suggestions? |
| 18:08.02 | *** join/#asterisk mgarfias (n=mike@scio-fe2-67-43-74-192.smt-net.com) |
| 18:09.23 | rob0 | selinuxium might want to look at the release notes and see if any new 1.6 features sound appealing. Generally it's probably best to start out with the newer version, that way when you get up to speed on what you're doing, you're not stuck with ancient software. |
| 18:09.52 | rob0 | otoh, there's also appeal in apt-get(8) |
| 18:10.45 | selinuxium | rob0: Cheers! ok... next as a complete newbie with asterisk, should I go down the AsteriskNow route? Again, I don't mind taking the leap... |
| 18:11.54 | tarapuez1 | any suggestions? |
| 18:12.14 | rob0 | I think AsteriskNow and others are more targeted at Linux/Unix beginners, not * beginners. You will learn more about * and have more configurability by using * on your preferred distro. |
| 18:12.23 | tarapuez1 | hey we made a asterisk - panasonic integration - we are connecting panasonic and asterisk using a digium te110p card. asterisk connect to pstn using a sip trunk. the problem is that dtmf tones in outgoing calls from panasonic no arriving to the sip chanel.we made a debug and we can see received dtmf tones from panasonic in Dahdi channel but we dont see dtmf tones in the Sip trunk. |
| 18:13.17 | selinuxium | rob0: Thanks for this! :) |
| 18:13.44 | selinuxium | rob0: OK, final one... I am thinking of running it in a virtual machine... is this an issue? |
| 18:15.23 | rob0 | If you need hardware like DAHDI (called zaptel for 1.4), sure, it's an issue, unless your VM has some workaround. Otherwise it probably only introduces a tiny amount of latency. |
| 18:22.36 | *** join/#asterisk frk2 (n=frk2@zivios/member/fkhan) |
| 18:25.21 | tarapuez1 | hey we made a asterisk - panasonic integration - we are connecting panasonic and asterisk using a digium te110p card. asterisk connect to pstn using a sip trunk. the problem is that dtmf tones in outgoing calls from panasonic no arriving to the sip chanel.we made a debug and we can see received dtmf tones from panasonic in Dahdi channel but we dont see dtmf tones in the Sip trunk. |
| 18:25.24 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 18:25.27 | tarapuez1 | Any sugggesions? |
| 18:30.33 | rob0 | Yes. Learn IRC etiquette. There were two /joins before your first repeat, and only one before the second. Do some debugging yourself, don't expect to be spoonfed with questions, "did you check $FOO". Check it on your own. Make a pastebin with debugging AND a complete problem description, so your repeats are not as obnoxious. |
| 18:30.36 | *** join/#asterisk jeffgus (n=jeffgus@2002:ad33:b504:0:0:0:0:1) |
| 18:32.29 | rob0 | Also, your issue seems to fail on the "smart questions" tests, http://www.catb.org/~esr/faqs/smart-questions.html |
| 18:33.21 | rob0 | I use a Panasonic phone on my own ATA, formerly used it on a Digium TDM400 FXS. Nothing unusual about it. |
| 18:36.32 | rob0 | So long, and thanks for all the coffee. |
| 18:37.30 | ISO9001 | I don't think he appreciated your constructive criticism ;) |
| 18:37.49 | rob0 | :) |
| 18:38.02 | mgarfias | hah |
| 18:38.15 | rob0 | Hey, it WAS constructive. I didn't call him names, gave good advice. |
| 18:40.33 | ISO9001 | I know. I'm just amused by the fleeing. |
| 18:41.50 | rob0 | Perhaps he didn't even see it. A lot of them expect to be highlighted on all replies. |
| 18:43.55 | dwschool | i have PSTN -> Callmanager -> ATA -> FXO -> asterisk. is there any way i can improve dtmp recognition on incoming calls |
| 18:44.39 | selinuxium | rob0: Sorry, had to start the dinner... Thank you for all your help. :) |
| 18:45.38 | rob0 | np. You asked questions I could answer, so I did. It's nothing personal against Colombians and other LACNIC folk. :) |
| 18:55.18 | degrade | Does anybody using sipp with uac_pcap? |
| 19:00.47 | *** join/#asterisk Iamnach0 (i=Iamnacho@ip174-70-137-120.ks.ks.cox.net) |
| 19:15.43 | *** join/#asterisk fakhir_ (n=fakhir@unaffiliated/fakhir) |
| 19:20.11 | drmessano | So what is the standard softphone for linux? |
| 19:21.35 | *** join/#asterisk riddlebox (n=james@75-132-225-75.dhcp.stls.mo.charter.com) |
| 19:23.06 | mchou | drmessano: so you heard? Google voice is going wide :) |
| 19:23.37 | drmessano | When? |
| 19:23.50 | mchou | it happened last week, I think |
| 19:24.23 | mchou | no longer need to be a member of Grand Central..... |
| 19:24.35 | drmessano | Its still invite only |
| 19:24.47 | mchou | yeah.... |
| 19:25.06 | mchou | but I think that's supposed to change soon |
| 19:26.18 | mchou | and something got fixed so they can accept programmatic DTMF "1" (from *) when you pick up an inbound call |
| 19:26.36 | mchou | it's all gravy now |
| 19:30.07 | *** join/#asterisk fakhir_ (n=fakhir@unaffiliated/fakhir) |
| 19:30.47 | *** join/#asterisk ingenius (n=alektro@host50.190-229-165.telecom.net.ar) |
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| 20:15.16 | carrar | w00t |
| 20:15.19 | carrar | ! |
| 20:22.51 | *** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 20:32.16 | *** join/#asterisk lesouvage (n=lesouvag@82.73.69.76) |
| 20:41.22 | JackTheNipple | good evening! |
| 20:41.46 | JackTheNipple | mchou: do you know anything about google voice comming to europe? |
| 20:47.27 | *** join/#asterisk xpot-mobile (n=james@70-91-210-233-BusName-Utah.hfc.comcastbusiness.net) |
| 20:47.29 | *** join/#asterisk [TK]D-Fender (n=joe@64.235.218.194) |
| 20:59.14 | *** join/#asterisk Chainsaw (n=chainsaw@gentoo/developer/atheme.member.chainsaw) |
| 21:03.45 | JackTheNipple | looks really interesting, I wonder if they use asterisks in background ;-) |
| 21:04.47 | JackTheNipple | I like the hand-over feature and the mailbox stuff. pretty nice :-) |
| 21:05.02 | JackTheNipple | nice ideas for *projects! |
| 21:05.32 | lesouvage | Does any of you know of sip devices that can be used for houshold automation, f.i. for switching light or turn on the heating by picking the proper choice from a voicemenu. |
| 21:05.52 | tzafrir_laptop | any idea why http://code.google.com/p/altalena/ is called that way? |
| 21:06.16 | [TK]D-Fender | lesouvage: SIP no, -> www.x10.com |
| 21:06.23 | tzafrir_laptop | seems basically like reinventing the wheel of res_lua |
| 21:17.16 | lesouvage | [TK]D-Fender: This really surprises me, I think using SIP for controlling all kind of devices should be very useful. |
| 21:18.15 | [TK]D-Fender | lesouvage: And what gives you the impression that SIP is a "do anything at all" communication protocol? |
| 21:21.26 | carrar | Happy Independence Day!! |
| 21:23.26 | lesouvage | [TK]D-Fender: I have done some reading and I think I understand the concept of SIP as a protocol that triggers sessions (not only phonecalls but all kind of sessions) But I might be wrong. |
| 21:23.35 | jblack | I read the sip standard once. It has the potential to do that sort of thing |
| 21:26.41 | jblack | I don't know of a single device that talks sip though... |
| 21:28.09 | lesouvage | I have this ehternet over power devices at home and they work pretty well. It must technically be possible to add SIP support. |
| 21:30.45 | jblack | Theoretically possible, at least. Like I said, I don't know of any equipment that uses the sip protocol. It's a bit heavy of a protocol for that sort of thing |
| 21:31.45 | jblack | A couple more standards, someone willing to sell for chips $0.50 that talk the right protocols, and customer demand, and you'd be set. |
| 21:32.48 | jblack | A billionaire could make it happen. |
| 21:33.01 | *** join/#asterisk Olobola (i=Olobola@202.sub-70-210-238.myvzw.com) |
| 21:33.08 | *** join/#asterisk JackTheNipple (n=JackTheN@static-87-79-237-194.netcologne.de) |
| 21:38.02 | lesouvage | jblack: I had this vision of asterisk calling me with the pre recorded message that my orange tree is in need of water because the zigbee/sip device in the pot of this tree sends this status to my asterisk server. |
| 21:46.59 | jblack | Ok. Get on it. ;) |
| 21:52.00 | lesouvage | [TK]D-Fender: What x10 software should I install to make it work. I did some googling but still didn't find the answer. On http://lorance.freeshell.org/asterisk/ I come the closest to what I think is the answer. |
| 21:53.34 | *** join/#asterisk hfb (n=hfb@cpe-98-151-255-13.socal.res.rr.com) |
| 21:56.45 | JackTheNipple | jblack: why not use nagios for this? |
| 21:57.16 | JackTheNipple | jblack: was ment for lesouvage: ;-) |
| 21:58.04 | *** join/#asterisk neurosys (n=vinix@c-71-196-19-254.hsd1.fl.comcast.net) |
| 21:58.36 | JackTheNipple | lesouvage: your can use nagios for checking your enviroment/information sources and than have asterisk notify you |
| 22:01.01 | lesouvage | JackTheNipple: Thanks, I'm in reading phaseand you give me more to read :-) |
| 22:01.33 | JackTheNipple | ;-) |
| 22:02.18 | JackTheNipple | lesouvage: oif you don't know Nagios and you want to do things like described, this might be something you want to know about ;-) |
| 22:03.07 | *** join/#asterisk serph (n=serph@70.49.145.76) |
| 22:03.31 | *** part/#asterisk neurosys (n=vinix@c-71-196-19-254.hsd1.fl.comcast.net) |
| 22:05.38 | *** join/#asterisk riddlebox (n=user@75-132-244-219.dhcp.stls.mo.charter.com) |
| 22:16.25 | lesouvage | JackTheNipple: Isn't Nagios a little bit overdone for switching a handful of devices like lamps. Nagios seems to be a very interesting project. The example I found was just exten => _2XX,1,System(/usr/local/bin/x10 off ${EXTEN:1}) and with x10 in place that seems pretty simple. |
| 22:28.02 | JackTheNipple | maybe you're right. I know nagios quite well and I agree, it complex. But for a whole house-automation I'd use it in my own house ;-) |
| 22:37.44 | carrar | lesouvage, You can use x10 with Asterisk using BottleRocket & X10 FireCracker kit |
| 22:38.08 | carrar | http://kbase.x10.com/wiki/Firecracker |
| 22:38.17 | carrar | http://www.linuxha.com/bottlerocket/ |
| 22:38.35 | carrar | go forth and teach thyself |
| 22:38.45 | lesouvage | carrar: thanks for the info. I look into it. |
| 22:39.10 | lesouvage | JackTheNipple: installing nagios from soure at this moment. |
| 22:39.20 | carrar | hahah |
| 22:41.43 | *** join/#asterisk SkramX (n=SkramX@cpe-70-112-25-138.austin.res.rr.com) |
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| 23:14.09 | SkramX | anyone familiar with downgrading a Cisco 7940 from a SIP firmware to a Cisco-supplied SCCP firmware? |
| 23:22.01 | *** join/#asterisk [T]ank (n=ckwall@c-76-23-32-201.hsd1.ut.comcast.net) |
| 23:23.26 | [T]ank | just picked up a grandstream handytone 286. I have it connected to a host over the internet. I have other ATAs that are connected, but do not experience issues... this one I must have a setting wrong. I have hearing about every 1 second where the sound cuts in and out. Very regular interval. any ideas? |
| 23:23.34 | [T]ank | using ulaw |
| 23:23.41 | [T]ank | over sip |
| 23:24.41 | mgarfias | Is there any thing out there that'll give me pointers on debugging dahdi? I'm having problems, and the logs aren't saying anything but 'unable to request channel' |
| 23:24.59 | *** join/#asterisk kamanashisroy (n=kamanash@202.56.7.216) |
| 23:25.25 | b14ck | sup everyone |
| 23:25.34 | b14ck | =) |
| 23:26.24 | [TK]D-Fender | lesouvage: I use heyu2 |
| 23:31.17 | [T]ank | anyone know why the sound on my sip peer would cut out at a random 1 second interval |
| 23:31.31 | b14ck | nat maybe? |
| 23:31.44 | [T]ank | it is behind nat... i do have it set to take care of it. |
| 23:31.44 | b14ck | do you have nat configured? |
| 23:31.59 | [T]ank | yeah |
| 23:32.01 | b14ck | can you turn on sip debugging for that extensions? |
| 23:32.08 | b14ck | and then get a sip log and pastbin.com it |
| 23:32.13 | b14ck | then we can check it out in detail |
| 23:32.28 | b14ck | sip set debug ip <iphere> |
| 23:37.42 | *** join/#asterisk lesouvage (n=lesouvag@82.73.69.76) |
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