| 00:01.16 | Peggerr | humm how shoudl i buy my asterik hardware from, sun or hp, HP is cheep er buy sun makes nicer equpemnt but is more expensive |
| 00:02.55 | rene- | i cant find any references to realtime configuration for agents, i thought it existed... was i dreaming? |
| 00:06.08 | tehdely | mattwj2005: i am using app_conference from cvs as of about a month ago |
| 00:06.18 | tehdely | there really is no configuration. it's just a dialplan application |
| 00:06.30 | tehdely | exten => 777,1,Conference(meatwhore/SVD/1) |
| 00:06.44 | tehdely | to login as a conference manager (which is currently just a stub and does nothing) |
| 00:06.47 | tehdely | replace SVD with LVD |
| 00:06.53 | tehdely | exten => 778,1,Conference(meatwhore/LVD/1) |
| 00:07.32 | mattwj2005 | cool anything else? |
| 00:09.02 | *** part/#asterisk g__ (n=g@itd01fw-fibre.itdepartment.com) |
| 00:09.19 | tehdely | nah that's pretty much it |
| 00:09.23 | tehdely | let me know if your asterisk randomly segfaults |
| 00:09.35 | tehdely | mine does occasionally, and it happens in some app_conference code |
| 00:09.48 | tehdely | i believe it has something to do with misconfigured codecs one of my callers may be using |
| 00:09.50 | tehdely | YMMV |
| 00:11.50 | marcus2 | so i've got a CAC AB-II plugged into my asterisk server |
| 00:12.09 | marcus2 | and when people use channels on that bank to make outbound calls, after the remote end hangs up, the local end immediately gets fast busy |
| 00:12.25 | marcus2 | the asterisk console says it is hanging up the zap channel when this happens |
| 00:12.55 | mattwj2005 | Illegal instruction error |
| 00:13.05 | *** join/#asterisk VoIPMasta (n=John@201.160.17.234.cableonline.com.mx) |
| 00:13.13 | VoIPMasta | Hi there |
| 00:13.23 | marcus2 | is this the only possible behavior for this channel bank? |
| 00:13.30 | VoIPMasta | A quick question: does anyone know why it is that when I'm using g.729 the caller ID doesn't work? |
| 00:14.56 | *** join/#asterisk Jaxxan (n=jaxxan@202.70.125.124) |
| 00:15.14 | Jaxxan | hey guys |
| 00:15.58 | x86 | note to self.... NEVER use ilbc ;) |
| 00:16.00 | *** join/#asterisk Strom_C (n=strom@gateway.digium.com) |
| 00:16.01 | *** join/#asterisk gniretar_work (n=mark@gateway.meteor-web.com) |
| 00:16.03 | gniretar_work | hi all |
| 00:16.10 | gniretar_work | anyone done any programming with the manager API? |
| 00:16.13 | timscott | :) |
| 00:16.19 | x86 | gniretar_work: i have, with Perl |
| 00:16.44 | gniretar_work | well, i'm doing it with PHP and i'm encountering an interesting phenominon regarding arents |
| 00:16.45 | Jaxxan | so after i upgraded from zaptel-1.0.9.1 to zaptel-trunk, whenever i make a call, the ringing sounds rather strange and i get voltage increases/decreases stamped to my /var/log/messages file which kinda match the sound of the ringing. |
| 00:17.03 | Jaxxan | what can i do to make the ring one specific ummm... voltage again. |
| 00:17.26 | gniretar_work | when an agent is logged in but not on the phone i get this for the commands 'action: Agents' then 'ActionID: 1' |
| 00:18.04 | gniretar_work | 1003 |
| 00:18.04 | gniretar_work | Name - Randy |
| 00:18.04 | gniretar_work | Status - AGENT_IDLE |
| 00:18.04 | gniretar_work | Channel - 703@employees |
| 00:18.14 | gniretar_work | then then the agent is on the phone i get this: |
| 00:18.15 | gniretar_work | 1003 |
| 00:18.15 | gniretar_work | Name - Randy |
| 00:18.15 | gniretar_work | Status - AGENT_IDLE |
| 00:18.15 | gniretar_work | Channel - 703@employees (Confirmed) |
| 00:18.23 | gniretar_work | why is his status still idle?? |
| 00:18.28 | Strom_C | gniretar_work: dont flood the channel, please |
| 00:18.30 | *** part/#asterisk TTT_Travis (n=Travis@bal-broadband2-ws-14.dsl.airstreamcomm.net) |
| 00:18.31 | Strom_C | ~pb |
| 00:18.32 | jbot | pb is, like, a place to paste your stuff without flooding the channel - try http://pastebin.com/, or http://pastebin.ca |
| 00:18.33 | *** join/#asterisk SplasPood (n=jwb@ool-18b935fd.dyn.optonline.net) |
| 00:18.35 | *** join/#asterisk MGSsancho (n=user@ppp-67-126-243-88.dsl.irvnca.pacbell.net) |
| 00:18.52 | gniretar_work | i aplologise |
| 00:20.32 | gniretar_work | x86: is that the way its supposto be? |
| 00:21.18 | *** join/#asterisk xachen (i=justin@pdpc/supporter/student/xachen) |
| 00:22.02 | *** join/#asterisk pagec (n=cpage@64-252-98-136.adsl.snet.net) |
| 00:23.03 | Jaxxan | what does this log entry in /var/log/messages mean? i know it has to do with zaptel. -- Apr 13 13:22:07 asterisk kernel: EC: DC bias calculated: 5 V |
| 00:26.20 | *** join/#asterisk doughecka (n=Miranda@unaffiliated/doughecka) |
| 00:27.00 | *** join/#asterisk lokkju (n=lokkju@unaffiliated/lokkju) |
| 00:28.11 | x86 | gniretar_work: i have no idea... i use the perl module to handle all the communications ;) |
| 00:28.21 | x86 | gniretar_work: Asterisk::Manager |
| 00:28.30 | gniretar_work | >.< |
| 00:28.33 | mattwj2005 | tehdely you still here? |
| 00:29.14 | tehdely | yes |
| 00:29.26 | mattwj2005 | where do you get your source from? |
| 00:29.50 | tehdely | iaxclient cvs |
| 00:30.07 | mattwj2005 | darn it |
| 00:30.08 | tehdely | http://sourceforge.net/projects/iaxclient/ |
| 00:31.35 | mattwj2005 | what distro are you using |
| 00:31.36 | mattwj2005 | ? |
| 00:33.09 | x86 | gniretar_work: use perl... you know... a _real_ language ;) |
| 00:33.30 | gniretar_work | yea, its nice. This is web based tho |
| 00:33.36 | x86 | and? |
| 00:33.45 | x86 | perl can be web based ;) |
| 00:33.54 | gniretar_work | i dont wanna set up CGI |
| 00:33.56 | *** join/#asterisk OMFGICBTS (i=ray@cpe-65-189-198-222.neo.res.rr.com) |
| 00:34.06 | gniretar_work | besides, Asterisk shouldnt act likt hat |
| 00:34.11 | gniretar_work | an agent thats on the phone isnt idle |
| 00:34.13 | *** join/#asterisk ramo (n=ramo@59.92.137.57) |
| 00:35.10 | Jaxxan | an agent on the phone should be unavailable |
| 00:35.40 | tehdely | mattwj2005: i'm using openbsd actually |
| 00:35.58 | mattwj2005 | oh okay....I am tried to get this installed on Debian |
| 00:36.03 | tehdely | shouldn't be any different |
| 00:36.04 | x86 | gniretar_work: "set up CGI" ? |
| 00:36.08 | tehdely | do you have asterisk installed globally |
| 00:36.12 | tehdely | or is it running out of a user account somewhere |
| 00:36.24 | x86 | gniretar_work: it's ready to go by default with Apache ;) |
| 00:36.28 | mattwj2005 | what do you mean globally? |
| 00:36.33 | tehdely | as in, did you install asterisk over / |
| 00:36.37 | tehdely | is it available to all users |
| 00:36.39 | tehdely | and in its usual place |
| 00:36.48 | tehdely | /var/lib/asterisk, /usr/bin/asterisk,e tc. |
| 00:36.57 | mattwj2005 | no I didn't install asterisk again? |
| 00:37.05 | mattwj2005 | should I? |
| 00:37.05 | tehdely | you misunderstand |
| 00:37.06 | tehdely | my question is |
| 00:37.08 | tehdely | where is asterisk installed |
| 00:37.15 | tehdely | did you install from the debian packages? did you install from source? |
| 00:37.29 | mattwj2005 | on my asterisk server |
| 00:37.32 | mattwj2005 | source |
| 00:37.40 | tehdely | and you just did make install |
| 00:37.43 | *** join/#asterisk bkw_ (n=brian@adsl-70-142-62-199.dsl.tul2ok.sbcglobal.net) |
| 00:37.45 | tehdely | no special INSTALL_PREFIX or anything |
| 00:37.46 | tehdely | right? |
| 00:37.52 | mattwj2005 | correct |
| 00:37.58 | tehdely | then app_conference should build right off the bat |
| 00:38.03 | tehdely | what happens when you try to build it? |
| 00:38.11 | mattwj2005 | it makes fine |
| 00:38.24 | mattwj2005 | the make install is where I am having trouble |
| 00:38.34 | tehdely | ah |
| 00:38.37 | tehdely | that is because in his Makefile |
| 00:38.40 | tehdely | he has some iaxclient-specific stuff |
| 00:38.42 | tehdely | the make install will fail on this line |
| 00:38.45 | tehdely | <PROTECTED> |
| 00:38.50 | tehdely | that is ok, because it is irrelevant to app_conference |
| 00:38.52 | tehdely | ignore the error |
| 00:38.57 | tehdely | app_conference is installed :) |
| 00:39.17 | *** join/#asterisk danlane (n=dan@invalid.name) |
| 00:39.23 | mattwj2005 | yeah after the /usr/sbin/asterisk -rx "restart now" it dies |
| 00:39.37 | tehdely | so asterisk is not starting now? |
| 00:39.49 | mattwj2005 | nope asterisk installs fine |
| 00:39.52 | tehdely | not install |
| 00:39.53 | tehdely | start |
| 00:39.54 | lokkju | does app_conference work better then meetme, or is it sorta a tossup? |
| 00:40.02 | tehdely | it's not that it works better5 |
| 00:40.04 | mattwj2005 | it even show the mod being loaded |
| 00:40.07 | tehdely | it's that it has a lot less requirements |
| 00:40.12 | tehdely | no need for a timing source, etc. |
| 00:40.18 | tehdely | you can run app_conference in lots of places where meetme simply won't work |
| 00:40.42 | lokkju | k, I am having meetme run just fine with ztdummy, but I had read about app_conference, so wasn't sure |
| 00:40.46 | tehdely | it's very primitive; no commands, no announce, etc. |
| 00:40.56 | tehdely | if you need a basic bridge, app_conference is great. i think of it as app_mixchannels |
| 00:41.00 | tehdely | because it really doens't do much more than that :) |
| 00:41.07 | lokkju | great for paging or something though, then |
| 00:41.13 | tehdely | yeah |
| 00:41.25 | tehdely | mattwj2005: so where is the problem? |
| 00:41.32 | tehdely | is it when you try to execute Conference in your dialplan? |
| 00:41.43 | mattwj2005 | when I actually die the extension...it kills asterisk |
| 00:41.47 | tehdely | ah |
| 00:41.49 | mattwj2005 | *dial |
| 00:42.02 | tehdely | next time you start asterisk |
| 00:42.06 | tehdely | start it with -g flag |
| 00:42.09 | tehdely | it will dump core when it crashes |
| 00:42.37 | danlane | Does anyone know why addons 1.2.2 (res_config_mysql etc) won't compile against recent SVN checkouts (like oej's RTCP branch)? |
| 00:42.41 | tehdely | load the core file into gdb and see where it is failing |
| 00:42.51 | tehdely | what version asterisk? |
| 00:43.28 | Luhiwu | does anyone knows why could it be possible to get '1 active channel 9 active calls' when doing a 'show channels'? |
| 00:43.44 | mattwj2005 | okay I got a core dump |
| 00:44.13 | tehdely | do you know how to use gdb? |
| 00:44.58 | mattwj2005 | now what? |
| 00:45.07 | tehdely | whatt is the name of the core file |
| 00:45.14 | tehdely | should be core.some number |
| 00:45.21 | mattwj2005 | core.2726 |
| 00:45.28 | tehdely | $ gdb -c core.2726 `which asterisk` |
| 00:45.32 | tehdely | at gdb prompt, type 'bt' |
| 00:45.35 | tehdely | paste the output to the pastebin |
| 00:45.41 | mattwj2005 | okay |
| 00:47.17 | *** join/#asterisk ManxPower (n=ewieling@dpc6745150107.direcpc.com) |
| 00:48.29 | danlane | so nobody knows how to get addons to compile against bleeding-edge svn checkouts? I can't find anything in svn newer than addons 1.2.2 |
| 00:49.02 | tehdely | what error are you getting when compiling? |
| 00:49.55 | *** part/#asterisk rene- (n=rene-@dsl-201-128-115-107.prod-infinitum.com.mx) |
| 00:50.01 | mattwj2005 | how much of this should I get? |
| 00:50.09 | mattwj2005 | it goes on for a long time |
| 00:51.02 | danlane | each addon (format_mp3, res_config_mysql etc) throws up errors about "conflicting types for `description'" |
| 00:52.11 | *** join/#asterisk Vyeperman (n=Vye@ip68-6-130-59.sd.sd.cox.net) |
| 00:52.17 | mattwj2005 | http://pastebin.ca/49309 |
| 00:52.19 | danlane | I figured it might be a known thing since I've tried a number of different SVN branches and addons 1.2.2 fails in the same way on each one while it compiles fine against 1.2.6/7 for me :/ |
| 00:54.37 | danlane | oh well, guess it's not a known thing... I'll hack on it some more |
| 00:54.39 | *** join/#asterisk Qwell (n=north@unaffiliated/qwell) |
| 00:54.47 | mattwj2005 | <PROTECTED> |
| 00:54.47 | mattwj2005 | <PROTECTED> |
| 00:56.49 | mattwj2005 | brb |
| 00:57.19 | tehdely | mattwj2005: strange |
| 00:57.41 | tehdely | what signal does it die with? |
| 00:58.50 | *** join/#asterisk hans0lo (n=hans0lo@64.123.97.58) |
| 00:59.00 | *** part/#asterisk hans0lo (n=hans0lo@64.123.97.58) |
| 00:59.27 | marcus2 | hrm |
| 00:59.35 | marcus2 | i wonder how i get ground start working on a CAC accessbank-ii |
| 01:00.06 | Renacor | whats the difference between openpbx and asterisk?? |
| 01:02.22 | OMFGICBTS | Is there a way to control how long Asterik lets an incomming call on an X100P clone ring before answering it ? I want to extend this time, not shorten it as most seem to. |
| 01:06.57 | Qwell | BUAHAHAHAHAHA! |
| 01:06.59 | Qwell | It lives! |
| 01:07.29 | [hC] | your cable came?? |
| 01:07.49 | *** join/#asterisk Ariel_ (n=Ariel@dsl-20-177.cofs.net) |
| 01:07.49 | Qwell | no |
| 01:07.53 | Qwell | I fixed mine |
| 01:07.57 | Peggerr | Qwell, then what lives? |
| 01:08.03 | Ariel_ | evening everyone |
| 01:08.09 | Peggerr | Qwell, the cable you made? |
| 01:08.11 | Qwell | I kinda...had it...reversed |
| 01:08.55 | Peggerr | Qwell, how ? |
| 01:09.13 | Qwell | I had a db9-rj45 converter, but I had the colors completely backwards |
| 01:11.35 | OMFGICBTS | Those colors are not all that standard. |
| 01:12.26 | *** join/#asterisk Smokes (i=SMOKEY@modemcable075.195-131-66.mc.videotron.ca) |
| 01:12.39 | Hmmhesays | got my new guitarone today |
| 01:12.40 | Hmmhesays | weee |
| 01:14.56 | OMFGICBTS | So no ideas on slowing down Asterisks answering of incomming calls ? |
| 01:15.21 | Hmmhesays | Wait() |
| 01:15.48 | Qwell | okay...that's loud |
| 01:16.41 | OMFGICBTS | Hmm |
| 01:17.03 | Ariel_ | slowing down asterisk most of the time people want to speed it up. But you can put wait(5) or more in a rule before you answer the line |
| 01:18.35 | OMFGICBTS | I can put a wait() in for a zap trunk ? |
| 01:18.49 | mattwj2005 | what do you mean signal? |
| 01:19.12 | mmlj4 | OMFGICBTS: certainly |
| 01:19.33 | [hC] | Qwell: so what was the error with it, to begin with? |
| 01:19.38 | [hC] | Qwell: or was it just that it wasnt configured yet? |
| 01:19.46 | Qwell | it wasn't configured |
| 01:19.50 | [hC] | ahh I see. |
| 01:19.54 | [hC] | how did you get it for free, anyways? |
| 01:20.02 | Qwell | sun.com |
| 01:20.38 | mattwj2005 | any idea tehdely? |
| 01:20.57 | tehdely | mattwj2005: it's a strange crash |
| 01:21.06 | tehdely | what is the line in your dialplan |
| 01:21.06 | [hC] | Qwell: do tell? |
| 01:21.09 | tehdely | also what version nof asterisk |
| 01:21.09 | [hC] | Qwell: I want one! |
| 01:21.35 | *** join/#asterisk lunaphyte (n=lunaphyt@pool-71-115-135-44.gdrpmi.dsl-w.verizon.net) |
| 01:21.54 | mattwj2005 | exten => 4000,1,Conference(meatwhore/SVD/1) |
| 01:21.55 | mattwj2005 | exten => 4001,1,Conference(meatwhore/LVD/1) |
| 01:22.10 | mattwj2005 | in the default context |
| 01:22.45 | mattwj2005 | 1.2.6 |
| 01:24.06 | mattwj2005 | here is the core dump thing |
| 01:24.07 | mattwj2005 | <PROTECTED> |
| 01:27.20 | *** join/#asterisk pauldy (n=pauldy@24-155-86-154.ip.grandenetworks.net) |
| 01:28.41 | jeebusroxors | anyone know if FWD is down? |
| 01:28.58 | OMFGICBTS | Anybody have recomendation for reading which files affect call flow ? |
| 01:30.39 | OMFGICBTS | I've read quite a bit of documentation and much of it seems contradictory. Time to focus on one set of docs maybe ? |
| 01:32.18 | *** join/#asterisk tzafrir_laptop (n=tzafrir@bzq-179-75-202.cust.bezeqint.net) |
| 01:32.48 | Ariel_ | OMFGICBTS, you would do like exten => s,1,Wait(5) exten => s,2,Answer then what every you want in the context your send the zap cannel to |
| 01:34.42 | Ariel_ | ~docs |
| 01:34.44 | jbot | docs is, like, probably Documentation can be found at http://www.digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or http://www.voip-info.org/wiki-Asterisk or http://www.asteriskdocs.org or http://www.oreilly.com/catalog/asterisk or http://www.asteriskguru.com, or http://www.astmasters.net/howtos.html |
| 01:37.57 | OMFGICBTS | Thanks Ariel |
| 01:38.18 | mmlj4 | OMFGICBTS: you want exensions.conf... check the wiki for a good explanation |
| 01:39.12 | OMFGICBTS | Yea...I was looking in zapata.conf |
| 01:39.16 | Ariel_ | OMFGICBTS, are you running something like asterisk@home |
| 01:39.39 | Ariel_ | zapata.conf will set the context you drop the call into in the extensions.conf |
| 01:40.17 | mattwj2005 | any ideas? |
| 01:40.26 | OMFGICBTS | Yes I am Ariel. Right now I'm trying to use it to bridge some remote users into our PBX. The long term goal is to completely replace our PBX with Asterisk. |
| 01:40.34 | mmlj4 | many ideas |
| 01:40.44 | mattwj2005 | about my problem |
| 01:41.29 | *** join/#asterisk MacDome (n=eseidel@A17-255-96-116.apple.com) |
| 01:43.06 | mattwj2005 | tehdely do I need to rebuild asterisk after the install? |
| 01:43.18 | Ariel_ | OMFGICBTS, asterisk@home uses amp or Freepbx to do the routing. |
| 01:43.34 | Ariel_ | it's conf files are based on the gui setup but they can be edited |
| 01:44.00 | Ariel_ | if your trying to put it between the pbx of yours what type of ports do you have. |
| 01:45.18 | OMFGICBTS | Ariel, I think I understand how to edit the conf files manually. Still working on what edits to make but that will come along. |
| 01:45.53 | Ariel_ | mattwj2005, you have either a bad conf file which is making your setup restart over and over again |
| 01:46.21 | Ariel_ | OMFGICBTS, in a@h you can only work with the _custom.conf files |
| 01:46.38 | mattwj2005 | or? |
| 01:46.43 | Ariel_ | also the zapata-auto.conf |
| 01:46.54 | Ariel_ | mattwj2005, a really messed up system |
| 01:47.00 | marcus2 | hmm |
| 01:47.08 | mattwj2005 | lol....geez thanks :P |
| 01:47.26 | OMFGICBTS | Ariel, I plan to parallel an X100P with an existing analog phone for two users. |
| 01:47.30 | Ariel_ | it's failing just as it gets to the queues |
| 01:47.46 | Ariel_ | parallel |
| 01:47.52 | Ariel_ | hummmm |
| 01:48.01 | tainted- | OMFGICBTS what does your nick stand for |
| 01:48.54 | OMFGICBTS | tainted, My current nick represents some frustration about forgetting the password for my other nick... |
| 01:49.16 | tainted- | oh my fucking god i can't believe that shit? |
| 01:50.29 | tehdely | mattwj2005: no |
| 01:50.32 | tehdely | you do not need to rebuild it |
| 01:50.59 | mattwj2005 | okay |
| 01:51.10 | mattwj2005 | what about my extensions.conf |
| 01:51.13 | tehdely | i am not sure why you are getting that error |
| 01:51.16 | tehdely | your extensions look proper |
| 01:51.22 | tehdely | i think i am on ast 1.2.5, i have not tried witha newerone |
| 01:51.26 | tehdely | perhaps it is a recent incompatibility? |
| 01:51.34 | mattwj2005 | don't I need a hangup or anything? |
| 01:51.38 | tehdely | i will try upgrading to 1.2.7 tongiht,install latestapp_conference, and see if i have an issue |
| 01:51.41 | OMFGICBTS | Ariel, I think the parallel arangement should work. It means the users involved don't have to do much different from home. The only problem I've run into with it is Asterisk answering and recording voicemail. They still want that on the older PBX. |
| 01:51.51 | tehdely | i suppose you can add a hangup, but the absence is not responsible for your crash |
| 01:54.43 | mattwj2005 | do we need an answer or anything before? |
| 01:56.40 | tehdely | hmm |
| 01:56.45 | tehdely | i do |
| 01:56.45 | tehdely | exten => 791,1,Answer |
| 01:56.46 | tehdely | exten => 791,2,Wait,1 |
| 01:56.46 | tehdely | exten => 791,3,Conference(meatwhore/SVD/1) |
| 01:56.50 | tehdely | give it a shot |
| 01:59.02 | *** join/#asterisk Jaxxan (n=jaxxan@leone-canopy05.bluelink.as) |
| 02:00.35 | OMFGICBTS | AMP/FreePBX only mucks with the .conf files when I hit the "save" button, correct ? |
| 02:05.24 | Ariel_ | OMFGICBTS, yes but it only touches the ones that have additional.conf |
| 02:05.44 | Ariel_ | it does not unless you upgrade the extensions.conf nor sip.conf iax2.conf |
| 02:10.21 | *** join/#asterisk tecnico (n=tecnico@user-24-236-120-2.knology.net) |
| 02:12.42 | ManxPower | Is anyone here anywhere near Birmingham AL and qualified to put ends on fiber cables? |
| 02:15.40 | CpuID2 | lol :) |
| 02:17.19 | *** part/#asterisk OMFGICBTS (i=ray@cpe-65-189-198-222.neo.res.rr.com) |
| 02:18.08 | ManxPower | CpuID2, Will have. |
| 02:18.19 | ManxPower | only local, however. |
| 02:18.23 | CpuID2 | hehe :) |
| 02:18.30 | LostFrog | Has anyone used x-lite recently? I don't see a way to change the configuration.. |
| 02:18.34 | CpuID2 | anything is better than nothing :) |
| 02:18.37 | ManxPower | I'd give my left nut for some dark fiber to here. |
| 02:18.45 | CpuID2 | lol |
| 02:18.47 | CpuID2 | :) |
| 02:18.54 | CpuID2 | i probly would too |
| 02:19.04 | CpuID2 | theres no dark fibre that im aware of down our street atm |
| 02:19.19 | *** join/#asterisk tekati (n=captain@cpe-66-75-215-63.bak.res.rr.com) |
| 02:19.42 | ManxPower | CpuID2, the telco had to install new cables because they didn't have enough pairs coming into the area |
| 02:19.47 | Sedorox | ahhh... but where would you terminate the dark fiber? :p |
| 02:19.50 | ManxPower | I ordered two POTS lines |
| 02:19.57 | jeebusroxors | anyone use FWD in here? |
| 02:20.00 | ManxPower | Sedorox, a local ISP 8-) |
| 02:20.04 | Sedorox | lol |
| 02:20.12 | ManxPower | What I SHOULD do is just take training on it. |
| 02:20.23 | tekati | Anyone have any luck with zaptel and udev on Fedora Core 4? For some reason I can not get the drivers to load at all. Did the make clean, make linux26, make install-udev stuff. |
| 02:21.07 | Sedorox | I wanna learn how to term fiber.. but I gotta check with the school.. to get in the class I would need a lotta DC-AC and other electronics classes.. which I don't think is needed.. so I think I could wave them |
| 02:21.08 | LostFrog | nm |
| 02:21.28 | Sedorox | for networking that I'm going into.. I should it would be handy to know how and be skilled in doing it |
| 02:21.32 | LostFrog | I don't see why you would need electronics classes to term fiber.. |
| 02:21.38 | LostFrog | It is a skilled labor thing. |
| 02:21.50 | Sedorox | its a degree.. from the electronics department in the school |
| 02:22.01 | Sedorox | they don't have a cross-degree for the IT students.... yet.. I'm hoping to change that |
| 02:22.07 | Sedorox | they have a cross for the Cisco classes... |
| 02:22.19 | Sedorox | so like pre-req's and other classes in the degree you would be taking anyway |
| 02:24.10 | tecnico | http://www.lanshack.com/fiber-optic-tutorial-termination.aspx I've seen a better "howto" somewhere else, but I can't find it.. |
| 02:24.54 | Sedorox | hehe... I had the teacher that does it for another class.. he showed a me and a few others how to do it.. .I've done it once.. hehe |
| 02:25.04 | Sedorox | isn't that hard... (well I didn't do epoxy-polish tho) |
| 02:27.20 | *** join/#asterisk cced2 (n=dev2003@222.33.36.205) |
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| 02:40.16 | cced | :) |
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| 02:47.48 | cced | :_ |
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| 03:12.03 | tekati | Okay I give up trying to make * work with Fedora Core 4. What is the best OS to use that I can make a firewall/* server? |
| 03:12.15 | Sedorox | FC sucks! |
| 03:12.21 | tekati | I agree. |
| 03:12.25 | tekati | Suggestions? |
| 03:12.33 | Sedorox | firewall == fbsd... server == gentoo (imho) |
| 03:12.37 | Sedorox | :p |
| 03:12.54 | Sedorox | (can't believe I'm saying this) but ubuntu seems to have good luck with people.... |
| 03:13.30 | ManxPower | tekati, use any distro you want. |
| 03:13.43 | ManxPower | FC should work just fine. |
| 03:13.54 | ManxPower | of course, you checked the Wiki for info about FC4, right? |
| 03:14.18 | tekati | No matter what I do I can not get the zap drivers to work in udev. Yes I followed that and anything else I could find on google with no luck at all. |
| 03:14.21 | hinckc | asterisk 1.2.6 is working fine for me on FC4 |
| 03:14.45 | LostFrog | I can't recreate the problem I have at work in my home lab. |
| 03:15.06 | tekati | hinckc: if you do a ls /dev or ls /dev/.udevdb do you see the zaptel stuff in there? |
| 03:15.15 | LostFrog | Maybe I will upgrade both servers at work to 1.2.7. |
| 03:15.23 | hinckc | I'm not using zaptel... sip only... :( |
| 03:15.34 | hinckc | cd /dev |
| 03:15.36 | tekati | Ah okay that is probably why then. Thanks. |
| 03:15.52 | hinckc | yeah, no zap |
| 03:16.01 | tekati | I use SIP with a TDM400 card for my phones. |
| 03:16.32 | tekati | Any udev guru's in here? |
| 03:21.29 | *** join/#asterisk isamar (n=isamar@202.95.220.92) |
| 03:21.33 | isamar | hi folks |
| 03:22.13 | isamar | any1 using a2billing_? |
| 03:22.22 | isamar | or better.. any1 hacking a2billing? |
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| 03:32.25 | *** join/#asterisk chris_ast (n=Administ@59.93.56.163) |
| 03:32.51 | chris_ast | Need help on Asterisk java manager API, please help |
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| 04:01.29 | Ridgeback | hello... |
| 04:01.54 | Ridgeback | anyone on here work with "hint" extensions" |
| 04:02.37 | *** join/#asterisk SplasPood (n=jwb@brooklyn.paravolve.net) |
| 04:03.17 | chris_ast | Need help on Asterisk java manager API, please help |
| 04:03.33 | Ridgeback | hmmm dont know muc habout that..whats wrong? |
| 04:04.14 | chris_ast | I am initiating a call but later after acceptong call there is no connection with * |
| 04:04.36 | Ridgeback | so you can build a call, but it fails to answer? |
| 04:05.03 | chris_ast | I can answer, later no connection with * |
| 04:05.26 | Ridgeback | oh ok so the call is built but there is no actual conection... |
| 04:05.43 | Ridgeback | sonds like SIP signaling isworking, but perhaps there is no RTP stream being built |
| 04:05.44 | chris_ast | exactly |
| 04:06.00 | chris_ast | yep, I conformed with ethreal |
| 04:06.26 | Ridgeback | hmmm wiht Java do you have to setup an RTP class, then attach that class to some sort of sip handler? |
| 04:07.06 | chris_ast | nope, we have java package for manager api |
| 04:07.13 | chris_ast | from * |
| 04:07.42 | chris_ast | using those classes I initiated a call |
| 04:08.13 | Ridgeback | hmm it seems the RTP portion just isnt getting setup.... |
| 04:08.28 | chris_ast | yep, what cud I do? |
| 04:08.48 | Ridgeback | geez i dont know.. .let me look at the java api real quick |
| 04:09.50 | Ridgeback | hmm dont see it on voip-info... where is the documentation? |
| 04:10.16 | Ridgeback | did you see this? ist for asterisk java http://www.simitel.com/resources/booklet1/ |
| 04:10.38 | chris_ast | http://www.voip-info.org/wiki/view/Asterisk-java |
| 04:11.38 | Ridgeback | ah it says the Manager API is only for sending actions or monitoring events. |
| 04:11.43 | chris_ast | http://www.asterisk-java.org/latest/tutorial.html |
| 04:12.33 | chris_ast | I used telnet and via manager api everything was working fine. I have problem only with java |
| 04:13.06 | chris_ast | actually u cud find two examples at the end of link, I clubbed it to one and originated a call |
| 04:13.07 | Ridgeback | hmmm... that wierd. perhaps the commands the java api comapred to the ones you send via telnet are modified or broken? |
| 04:13.21 | Ridgeback | you could ethereal the two and compare |
| 04:13.36 | Ridgeback | could you get the basic hello world.java to work? |
| 04:13.41 | lokkju | any idea how to get the full param output when I do the show functions command from the cli? it cuts off a lot of the params when it displays the functions |
| 04:13.46 | *** join/#asterisk kemp (n=10330405@58.48.212.28) |
| 04:13.53 | chris_ast | I did that, it was one way only with java |
| 04:14.22 | chris_ast | clubbing examples in that might also create problem, may be using java wrongly |
| 04:14.40 | Ridgeback | the paramerters are mangled... there is no way to show them fully via the cli |
| 04:14.54 | *** join/#asterisk h3x0r (i=hex@ip68-96-175-172.lv.lv.cox.net) |
| 04:15.31 | chris_ast | :( there is no complete example even |
| 04:15.34 | Ridgeback | you could do show function sort and grep out the syntax line |
| 04:15.58 | Ridgeback | but it would not be one command ,you would have to do that for all of them |
| 04:16.03 | chris_ast | Ridgeback: I am sure many got this to work |
| 04:16.15 | Ridgeback | the java api? yes i'm sure too |
| 04:16.19 | lokkju | Ridgeback, yeah, hmf |
| 04:16.36 | chris_ast | Ridgeback: both manager api and java |
| 04:17.03 | Ridgeback | unfortunatly my JAva-kung-fu is really poor... ;) |
| 04:17.15 | marcus2 | anyone here using a CAC AB-II ? |
| 04:17.28 | Ridgeback | wtf is cac ab-ii? |
| 04:17.33 | marcus2 | a channel bank |
| 04:17.47 | Ridgeback | oh ok, like an adnx/24? |
| 04:18.16 | Ridgeback | guys, its been fun..but time for bed |
| 04:18.19 | Ridgeback | see yas! |
| 04:18.32 | *** join/#asterisk litage (n=nick@203.220.55.70) |
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| 04:48.05 | terrapen | anyone use a Blackberry? |
| 04:49.02 | *** join/#asterisk amdtech (n=root@ip70-191-164-238.fv.dl.cox.net) |
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| 04:57.36 | dlynes | sharp zaurus |
| 04:58.56 | *** join/#asterisk asterboy (n=kevin@S010600485480f4be.ed.shawcable.net) |
| 05:01.52 | SplasPood | If I'm doing a dial to 3 different sip users, via the '&' method... Any way to suppress the No Route To Host error when certain users are offline? |
| 05:06.09 | LostFrog | ok.. * hates me. |
| 05:06.22 | LostFrog | file: I am confused. |
| 05:07.14 | LostFrog | I guess everyone is dead. |
| 05:07.37 | LostFrog | SplasPood: I see the same problem, especially when a phone hasn't registered. |
| 05:09.05 | terrapen | lost, maybe you need to check somehow to see which are online |
| 05:09.11 | terrapen | make the dialplan smrt or something |
| 05:09.22 | *** part/#asterisk bkw__ (n=bkw_@adsl-70-142-62-199.dsl.tul2ok.sbcglobal.net) |
| 05:09.27 | LostFrog | I am going to queues soon anyways. |
| 05:09.59 | *** join/#asterisk litage (n=nick@203.220.55.70) |
| 05:10.07 | terrapen | smart move |
| 05:10.22 | terrapen | sheesh, finally my Polycom's clock is set and syncing |
| 05:10.36 | LostFrog | YEah.. except for the message before connecting to the agent thing. |
| 05:10.36 | SplasPood | terrapen: I thought about that.. but it seemed like a lot of code simply to parse out the string blah&blah&blah and then do a chanisavail or whatever on each.. |
| 05:11.05 | SplasPood | LostFrog: ackcall = no ? |
| 05:11.05 | terrapen | anybody know what to do about the problem with the first half-second or so of a call being silent? |
| 05:11.12 | SplasPood | Wait(1) |
| 05:11.13 | SplasPood | ? |
| 05:11.17 | terrapen | i've had this problem with asterisk forever |
| 05:11.18 | LostFrog | SplasPood: really? |
| 05:11.25 | terrapen | splas, this is in the VoiceMailMain app |
| 05:11.26 | SplasPood | LostFrog: yea... |
| 05:11.33 | terrapen | or anything else, really |
| 05:11.44 | LostFrog | Thank you. |
| 05:11.48 | SplasPood | terrapen: does doing a Wait(1) before calling it change anything? and have you Answer()'d |
| 05:11.54 | terrapen | probably |
| 05:11.56 | terrapen | that's so lame |
| 05:11.57 | terrapen | :) |
| 05:12.12 | SplasPood | LostFrog: I've been playing with queues all day, but with ackcall = yes on purpose |
| 05:12.25 | lokkju | does the -n option not work with -r? |
| 05:12.27 | LostFrog | We have a law firm and that would piss our clients off. |
| 05:12.40 | LostFrog | "I want to be connected immediately!" |
| 05:12.49 | SplasPood | LostFrog: wait.. piss your .. they'd have no idea |
| 05:12.57 | terrapen | Wait() did not fix it |
| 05:12.59 | SplasPood | your clients calling you |
| 05:12.59 | terrapen | err does |
| 05:13.18 | SplasPood | or you providing some sorta queue as a service to your clients? |
| 05:13.21 | terrapen | it just waits a second and then still silences the first half-second or so of VoiceMailMain |
| 05:13.42 | SplasPood | terrapen: what about doing an Answer() (if you're not already) |
| 05:13.46 | LostFrog | SplasPood: if it says "Please wait" before delivering the call, they would. |
| 05:13.50 | terrapen | lemme try |
| 05:14.03 | LostFrog | Or "Thank you.. blahblahblah" |
| 05:14.29 | SplasPood | LostFrog: So you're talking about announcements played to the 'caller' not the 'callee' (callee being the agent in the queue) |
| 05:14.41 | terrapen | Answer() did not help, either :( |
| 05:14.48 | SplasPood | terrapen: hrm.. |
| 05:15.01 | LostFrog | SplasPood: yes. |
| 05:15.06 | terrapen | heh, i've had this problem for years, through many different asterisk installations |
| 05:15.32 | SplasPood | LostFrog: Oh.. well you can turn off all announcements to the caller, and make it ring so they have no idea |
| 05:15.38 | SplasPood | ackcall is something else |
| 05:15.43 | terrapen | strangely, it does not happen on the SNOM |
| 05:15.45 | terrapen | only the polycom |
| 05:15.53 | LostFrog | ackcall is "You have a call, press #"? |
| 05:16.02 | SplasPood | LostFrog: yea |
| 05:16.13 | SplasPood | although mine doesn't seem to say anything, just waits for '#' |
| 05:16.23 | LostFrog | ok.. well, I will be playing around with queues. |
| 05:16.35 | LostFrog | tommorow or this weekend. |
| 05:16.40 | SplasPood | LostFrog: hang on, I have the config.. |
| 05:16.43 | terrapen | so strange. The Linksys SPA942 and SNOM 320 do not suffer from this problem |
| 05:16.51 | terrapen | it must be some kind of obscure polycom thing |
| 05:17.10 | SplasPood | announce-frequency = 0 |
| 05:17.10 | SplasPood | announce-holdtime = no |
| 05:17.21 | SplasPood | and you can pass an option to Queue() to make it ring rather than moh |
| 05:17.40 | SplasPood | terrapen: what polycom firmware rev? |
| 05:18.13 | *** join/#asterisk Thock (i=Landon@ip70-162-111-39.ph.ph.cox.net) |
| 05:18.15 | Thock | Howdy all |
| 05:18.20 | LostFrog | Can you have two layers in the queue? Like if it doesn't get answered in 30 seconds, start announcing.. |
| 05:18.48 | SplasPood | Lost: I think you'd need two queues, one without, one with, and then you'd make the first queue timeout into the 2nd |
| 05:19.03 | SplasPood | although... |
| 05:19.12 | SplasPood | announce-frequency could be set high |
| 05:19.18 | SplasPood | I don't think it says anything till that is up.. |
| 05:19.41 | LostFrog | I will have to play with it. |
| 05:19.45 | SplasPood | but then it'll ONLY say it that often |
| 05:20.17 | terrapen | splas, lemme check |
| 05:20.35 | terrapen | 2.6.0 |
| 05:20.39 | terrapen | does that sound right? |
| 05:20.45 | LostFrog | Maybe you can help me with my other problem. |
| 05:20.48 | terrapen | bootrom 3.1.0.0269 |
| 05:21.18 | terrapen | SIP application is 1.6.3.0067 |
| 05:21.35 | LostFrog | When I get a sip call into server1 it dials server2 via IAX and server2 passes the call to a SIP phone |
| 05:22.02 | LostFrog | Sometimes the user of the SIP phone transfers the call back to a SIP phone attached to server1. |
| 05:22.18 | LostFrog | My problem is that the IAX doesn't use native bridge. |
| 05:24.43 | terrapen | ok, go home time |
| 05:24.44 | terrapen | bbiab |
| 05:29.39 | lokkju | whee |
| 05:29.42 | lokkju | http://www.lokkju.com/blog/index.php/2006/04/13/bashperl-command-to-get-all-asterisk-functions/ |
| 05:29.52 | lokkju | talk about a crock, to get that nice output |
| 05:38.11 | *** join/#asterisk MikeJ[Laptop] (n=vircuser@d14-69-8-30.try.wideopenwest.com) |
| 05:47.15 | Qwell | Netgeeks: ping |
| 06:00.58 | *** join/#asterisk krisguy (n=krisguy@h216-170-039-057.adsl.navix.net) |
| 06:02.11 | kemp | Does anybody know if the callerid number can be getted from x100p clone card? |
| 06:02.47 | MikeJ[Laptop] | kemp, definate sometimes |
| 06:02.56 | MikeJ[Laptop] | depends on the card |
| 06:03.54 | asterboy | lokkju, love your work. |
| 06:04.37 | lokkju | asterboy, oh? |
| 06:05.46 | kemp | hi,MikeJ, it's very nice to meet you,can you tell me how to config it in detail? |
| 06:06.01 | MikeJ[Laptop] | ummm |
| 06:06.02 | MikeJ[Laptop] | I could |
| 06:06.09 | MikeJ[Laptop] | but the wiki has good docs on that |
| 06:06.36 | lokkju | (I'm going to be doing the same thing for the operators - Playback, etc - and the built in variables -- this is all in preperation for a php/ajax/javascript based IVR/exten advanced generation tool |
| 06:08.25 | kemp | ok,I had asked the same question on the Digium's forum,but I still do not solve it.Bythe way,I come from China. |
| 06:08.57 | asterboy | lokkju, just going through your site...you have some cool projects |
| 06:09.08 | asterboy | gmprice is great |
| 06:09.51 | lokkju | heh |
| 06:09.54 | *** part/#asterisk krisguy (n=krisguy@h216-170-039-057.adsl.navix.net) |
| 06:10.01 | asterboy | null modem cable from cat5, excellent! |
| 06:10.02 | lokkju | needs a lot of work, most people do not understand gmprice at first |
| 06:11.06 | lokkju | gmprice needs a logo, and then styling, and then a nice consice "about" blurb, to explain the concept - most people do not understand that yes, it only searches completed auctions, and yes, it only searches completed autions that successfully *sold* |
| 06:11.47 | lokkju | asterboy, check out the dnsEditor: Ajaxified - one of my best web control GUIs, so far |
| 06:11.48 | asterboy | perfect |
| 06:11.58 | asterboy | ya, I really like it. |
| 06:12.09 | asterboy | sure beats vi |
| 06:12.17 | lokkju | so, got any paying project for me? *grin* |
| 06:12.19 | lokkju | hehe |
| 06:12.38 | asterboy | I do all my DNS manually and worse I need to do it often cause I run my sites on dynamic IP. |
| 06:12.55 | lokkju | asterboy, yeah, but it does only work with mysql, and bind-dlz, right now - next step for it is to put in a DB abstraction layer |
| 06:13.00 | asterboy | ya I know about the freee DNS services for dynamic, but I like to be in control of my own |
| 06:13.12 | lokkju | asterboy, oh, shit, get bind-dlz, *now* |
| 06:13.23 | lokkju | so sweet |
| 06:14.07 | lokkju | we currently run over 400 zones, with an average of 20 entries per zone (max of 300, min of 1), all replicating to three server, and all *live* updates |
| 06:14.27 | asterboy | thats heavy |
| 06:14.33 | lokkju | bind-dlz.sourceforge.net |
| 06:14.54 | lokkju | it's nothing compared with what bind-dlz can handle |
| 06:15.38 | lokkju | if you do decide to implement bind-dlz though, do not use their example table structure, it has some issues - like no unique id :) |
| 06:16.24 | asterboy | dlz looks great |
| 06:17.34 | lokkju | it is - can't wait till it gets rolled into mainstream bind, which it eventually will |
| 06:18.00 | asterboy | lokkju, gmprice has some serious potential |
| 06:18.31 | asterboy | ya, bind has be stegnant for a long time. |
| 06:19.53 | lokkju | dlz 100% stable, as long as your database does not crash - which is why it is recommended (though I have not implented) that your actual database for dlz be db4, and you update db4 from mysql, or whatever |
| 06:19.56 | asterboy | lol, i cant care about spelling this late |
| 06:20.26 | lokkju | asterboy, well, if you have ideas for a logo, or styling, or anything else for gmprice, do tell |
| 06:20.32 | LostFrog | ok.. Don't Dial(IAX2/blah/blah,,t) if you want native transfer. |
| 06:20.40 | Corydon76-home | Hmmm, perhaps if I spell it asterboytoy |
| 06:20.43 | lokkju | (gmprice has one major bug right now - a .6MB memory leak per query) |
| 06:21.07 | asterboy | memory leaks are so annoying. |
| 06:22.48 | lokkju | yeah, well, it is a perl service on the backend, the php is only a xmlrpc frontend to my actual service that does all the hard work, and keeps itself logged into ebay |
| 06:23.41 | *** join/#asterisk [Airwolf] (n=airwolf@83.98.235.222) |
| 06:23.47 | lokkju | seriously though, if like my work, please recommend me for programming work if you know anyone who needs it - if it is a programming language, I have probably worked with it |
| 06:23.56 | lokkju | (yeah, that includes AGI) |
| 06:24.09 | MikeJ[Laptop] | lokkju, RPG4? |
| 06:24.45 | asterboy | stagnans - latin for stagnant |
| 06:25.04 | lokkju | MikeJ[Laptop], once, and it is hell |
| 06:25.25 | MikeJ[Laptop] | heh.. my first real job was RPG4 |
| 06:25.31 | asterboy | lokkju, I'm keeping you in mind. |
| 06:25.36 | asterboy | for programming. |
| 06:26.01 | lokkju | MikeJ[Laptop], if you know someone who is desperate, I could try again, but otherwise I prefer to not touch it - the three worst mainstream languages, in terms of me not liking to use them: RPG, COBAL, and VFP |
| 06:26.08 | lokkju | ah |
| 06:26.09 | lokkju | poor you |
| 06:26.31 | MikeJ[Laptop] | hehe.. |
| 06:26.33 | MikeJ[Laptop] | long time ago |
| 06:26.50 | MikeJ[Laptop] | 1/2 a lifetime ago almost now |
| 06:26.56 | lokkju | course, bf is just nasty too, but worth using once in a while just to be able to tell a client I am considering using a brainfuck program on his server :) |
| 06:27.15 | *** join/#asterisk cybergypsy (n=cybergyp@APoitiers-156-1-7-145.w86-207.abo.wanadoo.fr) |
| 06:27.23 | asterboy | cobol...LOL...a language created by a committee |
| 06:27.34 | asterboy | nothing good ever comes from committees |
| 06:28.22 | lokkju | hehe, or LISP - Lost in Stupid Parenthesis |
| 06:28.39 | MikeJ[Laptop] | I like lisp actually. |
| 06:28.51 | MikeJ[Laptop] | all my formal theory classes were in lisp. |
| 06:29.02 | MikeJ[Laptop] | it is a good teaching language.. |
| 06:29.10 | Qwell | Pleathe don't make fun of my lithp |
| 06:29.15 | lokkju | I've really only had to use it as a scripting language in autocad - it isn't bad, but not my prefered language, by a long shot |
| 06:29.42 | asterboy | ya, that's all my experience with LISP...autocad |
| 06:29.47 | MikeJ[Laptop] | Qwell, you just said you were going to bed.. LIAR! |
| 06:30.07 | Qwell | umm...yeah |
| 06:30.09 | Qwell | bed! :P |
| 06:30.21 | Shaun2222 | when one extention dials another, it just rings forever... |
| 06:30.27 | asterboy | ya, I gotta get there before my spelling gets real bad |
| 06:30.29 | Shaun2222 | what happened to voicemail, am i missing somthing? |
| 06:30.56 | asterboy | Shaun2222, SIP phones? |
| 06:31.01 | Shaun2222 | ya |
| 06:31.15 | asterboy | watch your CLI with verbosity at 345 |
| 06:31.28 | asterboy | better make that 347 |
| 06:31.35 | Qwell | asterboy: newb, everybody knows you need to use 1337 |
| 06:31.40 | Qwell | that, or 42 |
| 06:31.42 | asterboy | lol |
| 06:31.44 | Qwell | damnit |
| 06:31.45 | Qwell | bed |
| 06:31.47 | asterboy | oh ya 42 |
| 06:31.54 | asterboy | that's from monty |
| 06:31.55 | Qwell | and this time...I mean it |
| 06:31.57 | Shaun2222 | 345? 347... not sure what your talking about |
| 06:32.00 | asterboy | night |
| 06:32.06 | Qwell | asterboy: and no, you're way off |
| 06:32.07 | asterboy | just a joke |
| 06:32.11 | lokkju | have to be 43 |
| 06:32.13 | lokkju | ack |
| 06:32.13 | lokkju | 42 |
| 06:32.21 | lokkju | answer to everything, dontcha know? |
| 06:33.03 | asterboy | "I have three tablets...", drops 1, "I have two tablets |
| 06:33.48 | drray | I thought it was "I give you these 15 (drops 1) 10 commandments." |
| 06:34.06 | Strom_C | drray wins the line accuracy contest |
| 06:34.22 | asterboy | oh ya...that's it. |
| 06:34.32 | asterboy | been a while since I seen it. |
| 06:35.03 | asterboy | Shaun2222, start * with -cvvvvvvvvvvvvvvvvvvvvvvvvvvvv |
| 06:35.18 | asterboy | or howevermany "V" it feels good to press. |
| 06:35.24 | asterboy | max is 10 anyway. |
| 06:35.32 | Shaun2222 | i have it running with like 6 or 7 or somthing |
| 06:35.34 | asterboy | then place your calls and watch what CLI gives |
| 06:35.39 | asterboy | good enough. |
| 06:35.57 | Shaun2222 | cli doesnt do anything, all i see if the sip phoens registering every 40 seconds... |
| 06:36.03 | *** join/#asterisk dlynes (n=dlynes@S010600c09f9a0fc4.vc.shawcable.net) |
| 06:36.04 | lokkju | eh |
| 06:36.06 | Shaun2222 | i see the call placed obviously |
| 06:36.12 | lokkju | just tail -f /var/log/asterisk/full |
| 06:36.19 | lokkju | so much more info... |
| 06:36.29 | asterboy | nice...did'nt know that one |
| 06:36.36 | asterboy | always learning |
| 06:37.03 | Shaun2222 | i dont have a /var/log/asterisk/full |
| 06:37.03 | LostFrog | vvv |
| 06:37.27 | lokkju | shaun222, if it ain't there, it is either not enabled, or somewhere else - do you even have a /var/log/asterisk? |
| 06:37.32 | dlynes | shaun222: check your /etc/asterisk/logger.conf file to see what log files you ahve, and what's in them |
| 06:37.53 | LostFrog | ewww |
| 06:38.00 | LostFrog | It's so easy to compile it yourself. |
| 06:38.02 | dlynes | lokkju: it's not necessarily called full, and it doesn't necessarily exist |
| 06:38.43 | lokkju | LostFrog, yes, it is, and I have for some other things (asterisk-addons, zaptel) but when a package is available, I like to use it |
| 06:39.07 | lokkju | usually gets all sorts of nice tweaks specifically to compile and run well on the target distro |
| 06:39.09 | Shaun2222 | dlynes: was commented out, now it's loggin :) |
| 06:39.29 | asterboy | Shaun2222, you could add an entry in your dial plan, (extensions.conf) to dial the extension for you. |
| 06:39.51 | asterboy | something like: exten => 1234,1,Dial(SIP/exten) |
| 06:40.12 | asterboy | just to test |
| 06:40.14 | Shaun2222 | i have this under default... exten => 1001,1,Dial(SIP/1001) |
| 06:40.17 | LostFrog | or exten => _12XX,1,Dial(SIP/${EXTEN})) |
| 06:40.21 | lokkju | course, the only bug I ran across was actually a kernel bug - freaking linux 2.6, rtc, aspi, and ztdummy on a dell server == playback just hangs the call until the client inits a hangup |
| 06:40.37 | lokkju | shaun222, don't forget to reload |
| 06:40.57 | Shaun2222 | lokkju: thats been in my config since the beggining, it's in their... |
| 06:41.02 | asterboy | and what's with 1001 --> 1001 |
| 06:41.03 | asterboy | ? |
| 06:41.19 | Shaun2222 | what do you mean? |
| 06:41.33 | asterboy | shouldn't that be at least different? like 222,1,Dial(SIP/1001) |
| 06:41.46 | h3x0r | dell sux |
| 06:41.56 | asterboy | so does google |
| 06:41.57 | LostFrog | What's wrong with extensions matching users? |
| 06:41.57 | Shaun2222 | exten 1001 points to sip context 1001, whats wrong with that? |
| 06:41.58 | dlynes | lokkju: you've only run across one bug in asterisk? How often do you use it? |
| 06:42.13 | lokkju | dlynes, rofl - no, only one that had me totally stumped |
| 06:42.17 | asterboy | looks like your calling yourself. |
| 06:42.21 | dlynes | ah...hehe |
| 06:42.30 | DoktorGreg | http://www.firefoxflicks.com/flick/index.php?id=19542&c=false |
| 06:42.47 | Shaun2222 | asterboy: how so? |
| 06:43.16 | Shaun2222 | phoneA was given extention 1001 and the username in the sip conf is 1001 |
| 06:43.17 | asterboy | nvr mind, it just seems for a second it would call the number |
| 06:43.22 | *** join/#asterisk SuperLag (n=aaron@gentoo/developer/SuperLag) |
| 06:43.25 | asterboy | it's late. |
| 06:43.28 | asterboy | brain fried |
| 06:43.29 | Shaun2222 | ok.. |
| 06:43.39 | lokkju | dlynes, that one bug took me 4 days to find even what was causing it, and in the end it was only sort of asterisk - asterisk will using the ztdummy timing source, if it is available, even if it does not respond with any timing info - that is the extant of asterisk's fault - the rest of it is linux kernel 2.6 and dell motherboards |
| 06:43.42 | Shaun2222 | do i need a timeout or somthing |
| 06:44.07 | asterboy | no, I need one :P |
| 06:44.13 | DoktorGreg | if brain fried try this |
| 06:44.14 | dlynes | lokkju: i thought the zaptel drivers didn't even talk to the kernel? |
| 06:44.15 | DoktorGreg | http://www.firefoxflicks.com/flick/index.php?id=19542&c=false |
| 06:44.17 | lokkju | really, that is something in asterisk that should be fixed - just something that test if the ztdummy is even working, and if not, wanr and disable |
| 06:44.32 | *** part/#asterisk freat (n=ron@h-72-244-84-43.chcgilgm.covad.net) |
| 06:45.06 | asterboy | lol, I like the Explorer "weeeeeee" |
| 06:45.06 | lokkju | dlynes, ztdummy talks to /dev/rtc - on dell servers, running the 2.6 kernel, rtc does not give timing when acpi is enabled, which it is by default |
| 06:45.33 | dlynes | ah...it only talks to /dev/rtc on dell servers? no other machines? |
| 06:45.51 | lokkju | dlynes, no, it talks to /dev/rtc on ALL machines, that is the timing source for ztdummy |
| 06:46.14 | Foxtro | hi! |
| 06:46.16 | Shaun2222 | whats the best way to check if a phone is logged into a extension... |
| 06:46.21 | lokkju | but on dell servers, the rtc timing does not work - it never responds to timing requests, unless acpi is turned off at the kernel level |
| 06:46.23 | dlynes | ah...so it sounds like a bug on acpi machines then |
| 06:46.33 | Foxtro | how can configure voicemail.conf for mailing with exim4 ? (no sendmail) |
| 06:46.34 | Foxtro | :( |
| 06:46.43 | lokkju | dlynes, no, it is a bug with the timing chips on dell motherboards |
| 06:46.50 | dlynes | ah |
| 06:46.51 | dlynes | suckage |
| 06:46.57 | asterboy | sip show registry? |
| 06:47.15 | lokkju | asterisk just does not handle it well - if it can not get timing from ztdummy, it should warn and disable, or something |
| 06:47.16 | asterboy | or do you mean buddy watch/* presense |
| 06:47.22 | dlynes | asterisk is hard coded to use sendmail? |
| 06:47.23 | Corydon76-home | exim doesn't have a sendmail binary? |
| 06:47.35 | Shaun2222 | asterboy: i mean to use in the dialplan |
| 06:47.38 | dlynes | that i have a hard time believing....I would imagine most people are using postfix |
| 06:47.54 | Foxtro | how can configure voicemail.conf for mailing with exim4 ? (no sendmail) |
| 06:47.57 | asterboy | ah |
| 06:48.04 | *** join/#asterisk Z-Knight (n=Z-Knight@cpe-67-10-28-100.houston.res.rr.com) |
| 06:48.27 | Shaun2222 | Foxtro: ln -s /usr/sbin/exim /usr/sbin/sendmail |
| 06:48.32 | Shaun2222 | that should take care of it... |
| 06:48.35 | lokkju | heh |
| 06:48.43 | asterboy | ya that's what I did with nail |
| 06:48.58 | dlynes | that's stupid...is asterisk seriously hardcoded to launch sendmail to send an email? |
| 06:48.58 | Shaun2222 | just about every linux MTA is sendmail compliant... |
| 06:49.03 | Shaun2222 | otherwise nobody would use it |
| 06:49.04 | Z-Knight | hi...can someone give me advice....I installed Asterisk on CentOS and trying to connect with XLITE I get this Xlite error: Discovered Port Restricted Cone NAT Firewall And nothing appears in asterisk or any logs |
| 06:49.09 | Foxtro | shaun222: ;mailcmd=/usr/sbin/sendmail -t |
| 06:49.18 | Foxtro | how remplace with exim |
| 06:49.22 | Shaun2222 | in fact... most MTA installs symlink /usr/sbin/sendmail to them selfs.. |
| 06:49.32 | *** join/#asterisk Creperum (n=ilya@mail.tex.kiev.ua) |
| 06:49.33 | Shaun2222 | Foxtro: just symlink it! |
| 06:50.00 | dlynes | shaun222: why symlink at all? smtp code is pretty simple; you'd think asterisk would implement it |
| 06:50.04 | Corydon76-home | I'm guessing he doesn't know what symlink means. |
| 06:50.06 | asterboy | night guys |
| 06:50.18 | Corydon76-home | Night, asterboytoy |
| 06:50.36 | dlynes | Corydon-w: probably does...he just doesn't pay attention some times |
| 06:50.50 | dlynes | right, Foxtro ? |
| 06:50.59 | Shaun2222 | dlynes: it's pretty commen to symlink it anyways... you expecially in my type of business (hosting) you know how many programs and scripts are written to use /usr/sbin/sendmail or /usr/lib/sendmail |
| 06:51.03 | Corydon76-home | With some of the people in this channel, I'm never sure |
| 06:51.10 | *** join/#asterisk somegeek (i=levin@unaffiliated/somegeek) |
| 06:51.11 | Shaun2222 | it's always symlinked or some type of wrapper exists... |
| 06:51.16 | *** join/#asterisk MacDome (n=eseidel@c-71-198-177-144.hsd1.ca.comcast.net) |
| 06:51.19 | *** join/#asterisk UrielS (n=u_stettn@TLV62-0-121-254.bb.netvision.net.il) |
| 06:51.31 | dlynes | shaun222: it's still a pretty dumb way to do it |
| 06:51.35 | Foxtro | thanks |
| 06:51.39 | Foxtro | now its working |
| 06:51.40 | dlynes | regardless of whether everyone else is doing it, or not |
| 06:51.40 | Foxtro | :) |
| 06:51.46 | Z-Knight | <PROTECTED> |
| 06:51.50 | *** join/#asterisk MacDome (n=eseidel@c-71-198-177-144.hsd1.ca.comcast.net) |
| 06:52.11 | Shaun2222 | dlynes: php even uses /usr/sbin/sendmail by default |
| 06:52.17 | Shaun2222 | nobody even uses sendmail these days... |
| 06:52.20 | dlynes | Z-Knight: try checking your firewall log |
| 06:52.25 | dlynes | shaun222: I do! |
| 06:52.26 | Corydon76-home | I use sendmail |
| 06:52.27 | Shaun2222 | but the links still exist for backward compat. |
| 06:52.34 | Shaun2222 | ewww |
| 06:52.45 | dlynes | I know sendmail quite well, and i've got it well secured |
| 06:52.54 | Z-Knight | dlynes: I disabled the firewall afterwards and did a service network restart |
| 06:53.00 | Z-Knight | do I need to do a reboot? |
| 06:53.03 | Shaun2222 | dlynes: cooool |
| 06:53.09 | Corydon76-home | I hope nobody here uses that bastardization of RFC 2822, qmail |
| 06:53.28 | dlynes | But then again, I started out on slackware back when postfix and exim didn't exist |
| 06:53.33 | Shaun2222 | Z-Knight: how did you disable the firewall? |
| 06:53.45 | dlynes | The only other alternatives around at the time were qmail and smail |
| 06:53.47 | Z-Knight | Shaun: I did it via the CentOS menu |
| 06:54.03 | Shaun2222 | Z-Knight: during the install your talking about? or in the setup program.. |
| 06:54.16 | Corydon76-home | Z-Knight: I bet that your policy is still set to DROP |
| 06:54.36 | Z-Knight | Shaun: No I mean I have X running so I goto start menu and go to "security level" gui and disable it there |
| 06:55.16 | Corydon76-home | Z-Knight: if you run 'iptables -L', do you see a policy of ACCEPT or DROP? |
| 06:55.19 | Shaun2222 | Z-Knight: cat /etc/sysconfig/iptables |
| 06:55.22 | Z-Knight | one sec |
| 06:55.34 | Shaun2222 | actually just echo > /etc/sysconfig/iptables |
| 06:55.36 | Z-Knight | I see ACCEPT |
| 06:55.39 | Shaun2222 | that way you know it's disabled... |
| 06:55.48 | Shaun2222 | then run service iptables restart |
| 06:55.51 | Z-Knight | wait a minute |
| 06:55.53 | Z-Knight | what the hey |
| 06:55.58 | Z-Knight | not it seems to be running |
| 06:56.09 | Z-Knight | i was waiting for over 10 mintutes |
| 06:56.21 | Z-Knight | I figured I could do a simple service network restart |
| 06:56.33 | Z-Knight | but I guess it takes a while to go into effect? |
| 06:56.36 | Corydon76-home | iptables isn't the network |
| 06:56.51 | Z-Knight | how do you "restart" iptables? |
| 06:56.52 | *** join/#asterisk CpuID2 (n=none@gentoo/contributor/cpuid) |
| 06:57.03 | Corydon76-home | All a network restart does is to reload the interface configs |
| 06:57.04 | Z-Knight | or do they just refresh eventually |
| 06:57.06 | Shaun2222 | Z-Knight: i just told you... |
| 06:57.11 | Shaun2222 | service iptables restart |
| 06:57.20 | Corydon76-home | Or service iptables stop |
| 06:57.26 | Z-Knight | ahhh |
| 06:57.32 | Z-Knight | did not realize it was a service |
| 06:57.36 | Z-Knight | thank you very much |
| 06:57.46 | CpuID2 | most of the time iptables has a set of init scripts |
| 06:57.52 | CpuID2 | since it has to load the rules on startup |
| 06:57.55 | Z-Knight | yeah, I did not realize this |
| 06:58.02 | UrielS | Hi all, can anyone tell me how to gain read access permission for the SVN? |
| 06:58.02 | Corydon76-home | It's not a service... it's just kernel settings |
| 06:58.04 | CpuID2 | s/set of init scripts/an init script |
| 06:58.20 | Corydon76-home | but it does have an interface into /etc/init.d/ |
| 06:58.35 | Corydon76-home | so you can control it LIKE a service |
| 06:58.43 | CpuID2 | it depends on the distro, but gentoo for example, allows you to /etc/init.d/iptables save |
| 06:58.46 | Z-Knight | yeah I see it |
| 06:58.52 | CpuID2 | and then on startup, it will load your ruleset thats saved |
| 06:59.33 | Z-Knight | I'm still getting the initial error in XLITE: Discovered Port Restricted Cone NAT Firewall and the Asterisk Console does not seem to be recording a SIP signup....but at least it works |
| 07:00.00 | Z-Knight | i wonder if I have enough verbosity |
| 07:00.20 | Z-Knight | yup |
| 07:00.47 | Z-Knight | that was it....I did not have iptables restart (so eventually it did) and I did nothave the asterisk verbosity set high enough. |
| 07:01.04 | Z-Knight | Thank you again for all of your comments and help....you guys/gals/etc rock! |
| 07:01.49 | dlynes | it |
| 07:02.25 | Corydon76-home | There are girls on here? <shock> |
| 07:02.31 | Z-Knight | LOL |
| 07:02.38 | dlynes | yeah |
| 07:02.41 | dlynes | Katty |
| 07:02.52 | dlynes | and linuxchik |
| 07:03.06 | Corydon76-home | I've never seen linuxchik in here |
| 07:03.27 | dlynes | well, maybe she's not on here...maybe she's on ##slackware |
| 07:03.46 | dlynes | i'm in here, slackware, and freeswitch all the time |
| 07:03.47 | *** join/#asterisk CrummyGummy (n=wayne@dsl-145-70-182.telkomadsl.co.za) |
| 07:03.49 | dlynes | hard to keep track |
| 07:04.29 | dlynes | but Katty's definitely in here |
| 07:04.33 | Corydon76-home | There's a few of us in here who half count as girls... |
| 07:04.38 | Z-Knight | lol |
| 07:04.51 | dlynes | speak for yourself :) |
| 07:04.58 | Corydon76-home | Oh, I do... |
| 07:05.31 | Corydon76-home | Men are, like, so totally yum |
| 07:05.47 | dlynes | can you say ewwwwwwwwww? |
| 07:05.54 | Corydon76-home | Nope |
| 07:06.14 | Z-Knight | man...one simple comment and I start all this?! ;) |
| 07:06.33 | Corydon76-home | How can you say men are yucky and expect women to think the opposite? |
| 07:06.39 | dlynes | dood....cute chinese chicks are the only way to go :) |
| 07:06.59 | Z-Knight | for some of us we have to be happy just with chicks...we can't be very picky |
| 07:07.17 | dlynes | um....god made us different for a reason |
| 07:07.19 | Corydon76-home | Speak for yourself |
| 07:07.26 | dlynes | lol |
| 07:07.41 | Corydon76-home | dlynes: yes, she does have a sense of humor |
| 07:08.22 | dlynes | lol |
| 07:08.38 | *** join/#asterisk The_ritz (n=The_ritz@220.225.34.210) |
| 07:08.55 | The_ritz | hi |
| 07:09.10 | The_ritz | i need to connect cisco 7940 to asterisk |
| 07:09.13 | The_ritz | please guide me |
| 07:09.40 | dlynes | not many people on right now, ritz |
| 07:09.50 | Z-Knight | The_ritz: using SIP? If so, is the phone SIP configured? |
| 07:09.51 | dlynes | you might have trouble trying to find someone that uses cisco phones |
| 07:09.58 | dlynes | or not :) |
| 07:09.59 | Z-Knight | I have the 7960 cisco |
| 07:10.05 | *** join/#asterisk nagl (n=nagl@rih.zid-nw.wu-wien.ac.at) |
| 07:10.06 | Z-Knight | close enough to 7940 |
| 07:10.09 | The_ritz | yes i am using SIP |
| 07:10.28 | Z-Knight | I would recommend using the tftpboot server |
| 07:10.32 | The_ritz | so i need to check on phone first if that is SIP configured...right? |
| 07:10.33 | Z-Knight | it makes it easy |
| 07:10.51 | Z-Knight | yeah...you should see a little SIP symbol in upper right |
| 07:11.23 | Z-Knight | also you can check to see if you have a "SIP Configuration" option |
| 07:11.35 | Z-Knight | if not, then you will need to install the SIP firmware |
| 07:11.38 | The_ritz | well i have just switched on the phone....it has got an IP ....but there is no SIP symbol as u say |
| 07:11.52 | Z-Knight | go into the settings |
| 07:11.54 | The_ritz | well let me check |
| 07:11.56 | The_ritz | ok |
| 07:12.15 | Z-Knight | then status |
| 07:12.24 | Z-Knight | then firmware versions |
| 07:12.31 | *** join/#asterisk psk (n=psk@golia.caltanet.it) |
| 07:12.45 | Z-Knight | what does the Boot load ID or application LOAD ID say |
| 07:13.06 | Z-Knight | actually what does the Application Load ID say |
| 07:13.19 | The_ritz | i am in firmware versions ...wait |
| 07:13.54 | The_ritz | app load ID : P0030301MFG2 |
| 07:14.04 | Z-Knight | hmmm...that does not look like SIP |
| 07:14.05 | Z-Knight | let me check |
| 07:14.07 | Z-Knight | one sec |
| 07:14.09 | The_ritz | Boot load ID: PC0303010200 |
| 07:14.25 | The_ritz | where do you check from? :) |
| 07:14.55 | Z-Knight | I wrote myself a little (long) tutorial...got stuff from multiple sources...I'll check that first to see if I wrote the firmware info down or not |
| 07:15.00 | The_ritz | Version: 3.1 (MF.G2) |
| 07:15.13 | *** join/#asterisk TonyM (n=TonyM@adsl-solo-80-168-224-84.claranet.co.uk) |
| 07:15.14 | The_ritz | ok |
| 07:15.34 | Z-Knight | <PROTECTED> |
| 07:15.41 | The_ritz | ok |
| 07:15.57 | *** join/#asterisk Vyeperman (n=Vye@ip68-6-130-59.sd.sd.cox.net) |
| 07:15.58 | The_ritz | so do i really need SIP for asterisk? |
| 07:16.02 | Z-Knight | you will need to provision it with the SIP .... or I think there is something called chan_sccp that can be used |
| 07:16.07 | *** join/#asterisk angom_h (n=angom@red-corp-201.130.165.246.telnor.net) |
| 07:16.08 | The_ritz | and if yes how do i install it? |
| 07:16.15 | Z-Knight | you can do the chan_sccp....but that I have no clue about |
| 07:16.27 | Z-Knight | I think that is what it is called....you'd have to yahoo/google it |
| 07:16.38 | The_ritz | i'll do that thanks |
| 07:16.42 | Z-Knight | I only know a little about the SIP...spent a long night working on that |
| 07:16.54 | Z-Knight | if you want I can send you my tutorial for doing the SIP |
| 07:16.56 | Z-Knight | it may be of help |
| 07:17.01 | The_ritz | would you please guide me |
| 07:17.06 | dlynes | The_ritz: it's also called cisco skinny protocol (chan_sccp.so) |
| 07:17.10 | The_ritz | yes |
| 07:17.24 | The_ritz | email me the tutorial on riturajb@gmail.com |
| 07:17.29 | The_ritz | please... |
| 07:17.34 | Z-Knight | mind you the tutorial is not complete |
| 07:17.45 | Z-Knight | and it has just a bunch of info |
| 07:17.53 | The_ritz | i c. ok |
| 07:18.05 | Z-Knight | if you do want to do SIP then contact me via email after a send you the tutorial/notes I have and I can help you out tomorrow |
| 07:18.32 | Z-Knight | also...what I have is for the 7960 phone ,but the 7940 should be the exact same |
| 07:18.37 | The_ritz | ok thanks a lot :) |
| 07:18.50 | The_ritz | and one more thing.... |
| 07:18.54 | Z-Knight | yeah |
| 07:18.54 | *** join/#asterisk tuxd00d (n=tuxinato@netblock-68-183-112-100.dslextreme.com) |
| 07:19.30 | The_ritz | after configuring SIP firmware on my cisco....where do i setup extensions/ what config is needed on asterisk side...i have no idea |
| 07:19.46 | Z-Knight | ahh...multiple ways I think |
| 07:19.51 | Z-Knight | you can do it via the menu |
| 07:20.02 | The_ritz | ok fine |
| 07:20.03 | Z-Knight | but the best way is to do it via a tftpboot server |
| 07:20.20 | Z-Knight | I've not had much success with the menu because I jumped right onto using the tftpboot server |
| 07:20.28 | The_ritz | i c |
| 07:20.39 | Z-Knight | it has a nice file you edit and then you can set it up easily |
| 07:20.47 | Z-Knight | there might be a web interface as well..i've not tried |
| 07:20.48 | The_ritz | ok |
| 07:21.09 | Z-Knight | ok...I'm going to zip up my tutorial and send it off...give me one minute |
| 07:21.29 | *** join/#asterisk Uberbot (n=Uberbot@c-69-252-219-76.hsd1.nm.comcast.net) |
| 07:21.35 | The_ritz | ok |
| 07:21.37 | Uberbot | Hi all. |
| 07:21.44 | lokkju | mmm |
| 07:22.39 | Uberbot | I'm trying to override the CID for an outgoing call using exten => s, 8,set(CALLERIDNAME="${name}") |
| 07:22.50 | Uberbot | And it's still sending the CID defined in sip.conf. |
| 07:23.04 | Uberbot | What am I missing? |
| 07:23.14 | dlynes | Is it an analog line? |
| 07:23.17 | Corydon76-home | You're missing the function |
| 07:23.22 | Uberbot | Sip to Sip. |
| 07:23.25 | *** join/#asterisk Vyeperman (n=Vye@ip68-6-130-59.sd.sd.cox.net) |
| 07:23.26 | Uberbot | I'm all ears. |
| 07:23.29 | Corydon76-home | Set(CALLERID(name)=foo) |
| 07:23.48 | Uberbot | Got it. Thanx. |
| 07:23.48 | Corydon76-home | Please note that CALLERID must be all caps |
| 07:24.00 | Uberbot | Good to know. |
| 07:24.11 | Corydon76-home | All functions are ALL CAPS |
| 07:24.14 | *** join/#asterisk TonyM (n=TonyM@adsl-solo-80-168-224-84.claranet.co.uk) |
| 07:24.33 | Uberbot | I didn't know that. Thanx. |
| 07:24.37 | Corydon76-home | for more info, please type 'show function CALLERID' |
| 07:24.56 | Uberbot | You've given me all I need. |
| 07:25.11 | Corydon76-home | Or just 'show functions' |
| 07:27.20 | The_ritz | <PROTECTED> |
| 07:27.34 | The_ritz | thanks a lot |
| 07:28.44 | Z-Knight | The_Ritz....yeah few more minutes...I need to include the files that I have on my tftpboot server |
| 07:29.45 | The_ritz | anyone can send me more info on chan_sscp |
| 07:29.59 | The_ritz | Z-Knight: take ur time |
| 07:30.38 | dlynes | The_ritz: there's plenty of info on it on voip-info |
| 07:30.48 | dlynes | it's chan_sccp though, not chan_sscp |
| 07:31.03 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 07:31.04 | The_ritz | oh ok |
| 07:31.31 | dlynes | just do a search on there for sccp |
| 07:31.47 | dlynes | or for skinny |
| 07:32.59 | Uberbot | 'fraid its still not working: exten => s, 8, Set(CALLERID(name)="${name}") |
| 07:33.09 | Uberbot | This causes the call to simply not go through. |
| 07:33.26 | Corydon76-home | Are those spaces in here? |
| 07:33.41 | Uberbot | If I comment the line out and reload extensions, the call goes through. |
| 07:33.44 | Corydon76-home | s,8,Set <-- no spaces |
| 07:34.05 | Uberbot | Never had that problem before. |
| 07:34.33 | Uberbot | I've got spaces between each field in the line. |
| 07:34.52 | Corydon76-home | Yeah, no spaces |
| 07:35.28 | Corydon76-home | You don't have another priority 8, do you? |
| 07:35.39 | Corydon76-home | ~pb |
| 07:35.41 | jbot | extra, extra, read all about it, pb is a place to paste your stuff without flooding the channel - try http://pastebin.com/, or http://pastebin.ca |
| 07:35.49 | Uberbot | No. Only one. |
| 07:36.13 | dlynes | Uberbot: try Set(CALLERIDNAME("name")) |
| 07:36.26 | Gamercjm | VoIPMasta: you here? |
| 07:36.47 | Corydon76-home | Pastebin your extensions.conf |
| 07:37.57 | Uberbot | Unfortunately, its in a macro. I'll paste the macro and the console log. Good enough? |
| 07:38.11 | *** join/#asterisk chapeaurouge (n=chap@vilhost1.vision.lu) |
| 07:38.12 | Corydon76-home | Sure |
| 07:38.14 | Uberbot | This lets the call go through, but doesn't override the CID. |
| 07:38.34 | Corydon76-home | Please use pastebin, though |
| 07:38.50 | Uberbot | No prob. |
| 07:40.17 | Z-Knight | The_ritz....you should have the email now |
| 07:41.43 | Uberbot | http://pastebin.com/659138 |
| 07:42.01 | Uberbot | But I warn you. It's ugly. |
| 07:42.43 | Uberbot | Priorities 6-8 setup the CID. |
| 07:44.22 | dlynes | Uberbot: don't enclose 'name' in quotes |
| 07:44.40 | The_ritz | yeah |
| 07:44.46 | The_ritz | i got the mail...thanks a lot |
| 07:44.49 | Uberbot | You mean the one inside the call to CALLERID? Ok. Testing. |
| 07:44.53 | dlynes | correct |
| 07:45.02 | The_ritz | i will try it and let you know |
| 07:45.06 | Z-Knight | The_ritz: open the html file to see the notes and some instructions |
| 07:45.10 | Z-Knight | hope it helps |
| 07:45.13 | The_ritz | ok |
| 07:45.48 | dlynes | Uberbot: http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+SetCIDName |
| 07:46.08 | CpuID2 | Set() |
| 07:46.08 | *** join/#asterisk CrummyGummy (n=wayne@dsl-145-70-182.telkomadsl.co.za) |
| 07:46.15 | dlynes | ~set |
| 07:46.17 | jbot | extra, extra, read all about it, set is xbmodder is a world record setting 14 year old awesomely great guy |
| 07:46.29 | CpuID2 | Set(CALLERID(NAME)=moo) |
| 07:46.34 | CpuID2 | from memory |
| 07:46.34 | CpuID2 | :) |
| 07:46.46 | dlynes | Yeah...I just finished telling him that :) |
| 07:47.31 | Uberbot | Without the quotes, the call doesn't go through. I'm calling a softphone on the laptop beside me. |
| 07:47.43 | Uberbot | With the quotes, the softphone rings. |
| 07:48.47 | Corydon76-home | Uberbot: No quotes on "name" |
| 07:49.03 | Corydon76-home | It's just CALLERID(name), not CALLERID("name") |
| 07:49.08 | Uberbot | Then I've got other problems, then. |
| 07:49.22 | dlynes | Uberbot: Try Set(CALLERID(name)="test") |
| 07:49.38 | dlynes | And test it that way...if you're still getting problems, it's something outside that statement |
| 07:49.40 | Corydon76-home | And you don't need to quote the value, either, unless you want literal quotes in the callerid |
| 07:49.58 | Corydon76-home | So, Set(CALLERID(name)=${name}) |
| 07:50.19 | Uberbot | Even if there are spaces in ${name} ? |
| 07:50.24 | Corydon76-home | Correct |
| 07:50.30 | Uberbot | Ok. |
| 07:50.40 | Corydon76-home | You only ever need to quote a value if you're using it inside an expression |
| 07:50.54 | Uberbot | Good to know. |
| 07:51.29 | Uberbot | Looks like I've got other problems..... |
| 07:51.47 | Uberbot | exten => s,8,Set(CALLERID(name)="Test") Doesn't work. |
| 07:51.57 | Corydon76-home | oh, and ${CALLERIDNUM} is deprecated. You should be using ${CALLERID(num)} instead |
| 07:52.27 | Uberbot | I'll make a note of that and change it once I've got this working. Thanx. |
| 07:52.32 | Corydon76-home | Sure it does. It works fine |
| 07:52.42 | Corydon76-home | Perhaps you're not actually getting there? |
| 07:52.55 | Uberbot | <PROTECTED> |
| 07:53.31 | Uberbot | I didn't mean it didn't work IN GERNERL, just not for me. :-D |
| 07:53.37 | Corydon76-home | What makes you think it's not working? |
| 07:54.50 | Uberbot | Hold on. |
| 07:56.02 | Uberbot | It's working now. Feeling kinda silly. Must have forgotten to reload extensions. Thanx for your time. ;-) |
| 07:56.22 | dlynes | lol |
| 07:56.22 | Corydon76-home | yw |
| 07:56.30 | Uberbot | yw? |
| 07:56.36 | dlynes | you're welcome :) |
| 07:56.44 | Uberbot | :-D |
| 07:57.37 | Uberbot | Working pretty slick. I store most of the "fun" stuff in a SQL database. Query it just before I need to use it. |
| 07:58.03 | dlynes | yeah...i've got my own box of fun right now |
| 07:58.08 | Corydon76-home | There's more elegant ways to do that. |
| 07:58.12 | dlynes | trying to finish off my billing system |
| 07:58.18 | Uberbot | Oh? |
| 07:58.24 | Corydon76-home | Wait until you see func_odbc in trunk |
| 07:59.40 | Corydon76-home | Fully templated SQL, for both read and write operations |
| 07:59.59 | *** join/#asterisk Gamercjm (n=chris@pool-71-254-176-82.lsanca.fios.verizon.net) |
| 08:00.00 | Uberbot | Is it available now? |
| 08:00.11 | Corydon76-home | Yes, but only in trunk |
| 08:00.18 | Corydon76-home | Or you could backport it to 1.2 |
| 08:00.45 | Uberbot | Only in "trunk?" Is this a prerelase version? |
| 08:00.47 | Corydon76-home | The second revision in trunk is suitable for use in 1.2 |
| 08:00.58 | Corydon76-home | Trunk is the development tree |
| 08:01.16 | Uberbot | Quite a pun, no? |
| 08:01.45 | Corydon76-home | If you say so |
| 08:02.03 | Corydon76-home | It's more of a practical metaphor |
| 08:02.14 | Uberbot | :-D |
| 08:02.18 | Corydon76-home | 1.2 was a branch off of trunk |
| 08:02.26 | Corydon76-home | 1.4 will also be a branch off of trunk |
| 08:02.44 | Shaun2222 | whats the deal with agentlogin, if i wanted to do this do i really have to sit their with the line open.. what if i wanted to log into multiple queues... |
| 08:02.53 | Uberbot | Any other new features I should get excited about? |
| 08:03.04 | Shaun2222 | i want the agent to hit a button, login and be done, when a call comes in ring him... |
| 08:03.09 | Shaun2222 | not for the phone to sit their... |
| 08:03.18 | Shaun2222 | playing hold music. |
| 08:04.42 | Shaun2222 | hmm, looks like i may have found somthing better.. AgentCallbackLogin |
| 08:08.28 | dlynes | Uberbot: a complete rewrite of app_dial.c is in the works |
| 08:09.01 | dlynes | Uberbot: that's in the rollercoaster branch |
| 08:09.02 | Uberbot | Interesting. Any work being done on voicemail? |
| 08:09.31 | *** join/#asterisk bulibuta (n=bulibuta@80.97.12.10) |
| 08:09.50 | *** join/#asterisk nextime (n=nextime@213-140-6-103.ip.fastwebnet.it) |
| 08:10.05 | tainted- | dlynes rollercoaster branch? |
| 08:10.05 | dlynes | no idea, offhand...i just heard about the app_dial today, myself |
| 08:10.12 | dlynes | tainted-: the oej branch :) |
| 08:10.34 | dlynes | surprises around every bend :) |
| 08:10.44 | tainted- | where can i read about that? |
| 08:10.53 | dlynes | irc, of course |
| 08:11.42 | dlynes | one sec...I have the svn branch logged...just have to look back through my logs |
| 08:11.43 | tainted- | lol |
| 08:12.15 | b4ka | bulibuta ;) |
| 08:13.34 | dlynes | <PROTECTED> |
| 08:14.25 | dlynes | It's the development branch that may or may not even compile; it includes code that gets considered for trunk |
| 08:14.32 | dlynes | trunk is code that gets considered for releases |
| 08:14.40 | tainted- | interesting |
| 08:15.21 | bulibuta | b4ka, hello:) |
| 08:16.37 | The_ritz | anyone had success with asterisk + chan_sccp cisco phone? |
| 08:18.53 | thx2000 | ne1 have sip workin w/ teliax? |
| 08:21.03 | Shaun2222 | anybody figured out how to program the keys on the 7960 cisco phones... |
| 08:21.14 | Shaun2222 | where redial/newcall/cfwdall is |
| 08:21.21 | Shaun2222 | figured their is a way |
| 08:21.33 | Uberbot | Lagers, all. |
| 08:21.52 | *** join/#asterisk mexuar-tim (n=mexuar-t@host-212-158-206-61.bulldogdsl.com) |
| 08:22.37 | dlynes | Where? |
| 08:24.31 | *** part/#asterisk Uberbot (n=Uberbot@c-69-252-219-76.hsd1.nm.comcast.net) |
| 08:27.37 | *** join/#asterisk dcmwai (n=dcmwai@219.93.241.13) |
| 08:27.42 | Shaun2222 | how can i see what agents exist for what queues? |
| 08:27.58 | dcmwai | hello all |
| 08:28.07 | dcmwai | anyone have a good voip info page? |
| 08:28.12 | Shaun2222 | n/m show agents looks to do what i wanted. |
| 08:28.22 | dcmwai | http://www.voip-info.org/ is very slow to me |
| 08:28.36 | *** join/#asterisk hfb (n=hfb@adsl-69-231-53-173.dsl.irvnca.pacbell.net) |
| 08:28.48 | kamileon | hello |
| 08:29.02 | kamileon | can anyone help me with this dial error: Rejected call to 192.168.0.150, format 0x4 incompatible with our capability 0xff03. |
| 08:31.54 | dlynes | one side is saying it takes one codec and the other side is pretty much saying it takes every codec except for that one |
| 08:32.07 | *** join/#asterisk stoffell (n=stoffell@fw.catsanddogs.com) |
| 08:32.14 | dlynes | the format 0xnnnn is a bitmask describing which codecs it supports |
| 08:32.52 | dlynes | kamileon: sorry...the above two lines were meant for you |
| 08:33.17 | *** join/#asterisk fr00d (n=andi@zockt.normalerweise.net) |
| 08:33.25 | fr00d | Hello! |
| 08:34.25 | fr00d | I'm trying to set up asterisk with chan_bluetooth. Can somebody help me with my extensions.conf? |
| 08:34.33 | *** join/#asterisk MacDome (n=eseidel@c-71-198-177-144.hsd1.ca.comcast.net) |
| 08:34.51 | fr00d | I've no idea what to to when my phone and my headset is connected to asterisk. |
| 08:35.26 | *** join/#asterisk mkl1525 (n=daniel@93.236.80.212.versanetonline.de) |
| 08:40.16 | mkl1525 | Hi, when I use "exten => 73099441,1,Macro(pmx2sip,${EXTEN:5})" in my extension.conf to call a pmx2sip-macro and access $MACRO_EXTEN I get the 73099441 and not like I expected 441 - so did I get something wrong? |
| 08:42.11 | *** join/#asterisk nextime (n=nextime@213-140-6-103.ip.fastwebnet.it) |
| 08:42.31 | Shaun2222 | mkl1525: i belevie the 441 your looking for is set as ${ARG1} |
| 08:42.38 | Shaun2222 | Macro(macroname,arg1,arg2...) |
| 08:42.47 | Shaun2222 | http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+Macro |
| 08:44.59 | mkl1525 | thanks works! |
| 08:55.02 | *** join/#asterisk mexuar-tim (n=mexuar-t@host-212-158-206-61.bulldogdsl.com) |
| 08:55.05 | kamileon | dlynes: thanks. |
| 08:59.42 | *** join/#asterisk backblue (n=igor@82.102.1.42) |
| 08:59.45 | backblue | morning all |
| 09:01.51 | *** join/#asterisk abatista (n=Ariel@dsl-20-177.cofs.net) |
| 09:02.45 | fr00d | Moin backblue |
| 09:06.44 | The_ritz | i am getting compilation error on chan_sccp compilation |
| 09:06.48 | The_ritz | can anyone help |
| 09:09.44 | backblue | The_ritz: pastebin.com |
| 09:12.11 | *** join/#asterisk speedwagon (n=Ariel@dsl-20-177.cofs.net) |
| 09:15.25 | rkr245 | hi |
| 09:16.05 | rkr245 | hi |
| 09:19.25 | rkr245 | shaun2222:can you solve this for me ,i got error in dialing xlite soft phone when dialled to grandstream handy tone adapter phone it is ringing but when i lif the phone i got this message on asterisk server translation codecs not found for ulaw to g723 |
| 09:21.09 | rkr245 | uanble to find translation path from g723 to ulaw |
| 09:22.07 | *** join/#asterisk kainam (n=Jake@202.137.160.110) |
| 09:23.41 | dlynes | rkr245: asterisk only does passthrough for g723....if you're wanting to translate to or from g723, you're up the creek without a paddle |
| 09:24.16 | *** join/#asterisk Dibbler (n=Dibbler@zidane.pi-net.net) |
| 09:24.25 | *** join/#asterisk L0g0ff (n=thomas@pix89.global-e.nl) |
| 09:24.31 | L0g0ff | hi, all |
| 09:24.54 | The_ritz | what is the link of pastebin? |
| 09:25.42 | fr00d | The_ritz: Google is your friend.. |
| 09:26.04 | *** join/#asterisk austinnichols101 (n=austinni@dsl-10-169.cofs.net) |
| 09:26.31 | fr00d | How can I send the soundoutput of the asterisk on my notebook to my bt headset instead of OSS? |
| 09:26.43 | L0g0ff | I have create a asterisk queue with a silence.mp3 sound file. I want to change that file with a "default beep" sound file. The same file when you call somebody and the same when you call a ringgroup. Does domebody have that fie or what is the name in /var/lib/asterisk/sounds ? |
| 09:27.15 | dlynes | It's in /var/lib/asterisk/sounds |
| 09:27.29 | L0g0ff | do you know the name ? |
| 09:27.36 | dlynes | beep.gsm? |
| 09:27.53 | dlynes | naaaaaaaaaaaah....that would be too simple, wouldn't it? :) |
| 09:27.55 | L0g0ff | no, iĺl try that but thats not tht file |
| 09:28.08 | L0g0ff | hehe ;) |
| 09:28.08 | dlynes | No, that is the file |
| 09:28.26 | dlynes | At least it's the beep fro when it says please leave your message at the beep |
| 09:29.43 | *** join/#asterisk ramo (i=ramo@219.65.128.30) |
| 09:30.59 | rkr245 | dlyne: i dint understan clearly can you please explain me clearly |
| 09:31.19 | dlynes | asterisk does not fully support g723 |
| 09:31.26 | rkr245 | o.k |
| 09:31.29 | rkr245 | then? |
| 09:31.48 | dlynes | rkr245: so try to avoid it if you possibly can, because you won't be able to convert it to another codec |
| 09:32.00 | rkr245 | o.k |
| 09:32.05 | dlynes | or convert another codec to g723 for that matter |
| 09:32.18 | L0g0ff | no, i mean the beep when you call somebody and you must wait till he (or she) pickup the phone |
| 09:32.29 | dlynes | you mean the ringing? |
| 09:32.44 | rkr245 | grandstream can support other than g723? |
| 09:32.51 | dlynes | yes |
| 09:32.54 | rkr245 | like gsm ulaw |
| 09:33.10 | *** join/#asterisk tessier_ (n=treed@adsl-75-5-99-178.dsl.sndg02.sbcglobal.net) |
| 09:33.16 | rkr245 | o.k then i will take off g723 from the list |
| 09:33.23 | dlynes | rkr245: grandstream budgetone (don't know about other grandstreams) can support g729, g723, ulaw, alaw, gsm, speex, and i think ilbc |
| 09:33.40 | rkr245 | o.k |
| 09:34.03 | dlynes | g729 and ilbc are better anyways |
| 09:34.04 | rkr245 | i have here now grandstream handytone-496 adapter |
| 09:34.20 | dlynes | g729 takes less processing power than ilbc, but ilbc is patent free; g729 isn't |
| 09:34.30 | rkr245 | which are best and commonly suit for all phones |
| 09:34.36 | dlynes | so therefore, asterisk out of the box treats g729 the same way as g723 |
| 09:34.37 | rkr245 | i mean the codecs |
| 09:34.48 | L0g0ff | i mean the sound that you hear when you call somebody |
| 09:34.51 | dlynes | but you can purchase g729 licenses so g729 works the same as every other codec |
| 09:34.51 | *** join/#asterisk CMike (i=daemon@c-544171d5.116-1-64736c10.cust.bredbandsbolaget.se) |
| 09:35.11 | rkr245 | purchase? |
| 09:35.15 | dlynes | rkr245: g729, ulaw, g723, g726 |
| 09:35.17 | rkr245 | how much ti costs |
| 09:35.31 | dlynes | rkr245: $10 per call leg |
| 09:35.44 | rkr245 | call leg? |
| 09:36.06 | dlynes | rkr245: yeah...if you need to decode g729, that's one leg, if you need to encode, that's one leg |
| 09:36.37 | dlynes | rkr245: So, if you purchase ten licenses, you can encode or decode up to ten call legs simultaneously |
| 09:36.50 | rkr245 | o.k |
| 09:36.56 | dlynes | rkr245: so say user a is using gsm and user b is using g729, you'd need one license |
| 09:37.10 | rkr245 | o.k |
| 09:37.14 | dlynes | rkr245: if user A is using g729 and he wants to leave voicemail (gsm) he'd need one channel |
| 09:37.21 | dlynes | erm one license |
| 09:37.28 | rkr245 | o.k |
| 09:37.51 | zoa | ilbc is not patent free |
| 09:38.00 | dlynes | it isn't? |
| 09:38.01 | zoa | you just dont need to pay for it |
| 09:38.04 | dlynes | ah |
| 09:38.07 | dlynes | lol |
| 09:43.53 | rkr245 | dlynes: its working now |
| 09:44.09 | rkr245 | thankyou very much for your information |
| 09:46.30 | kamileon | what else is similar to asterisk@home |
| 09:46.43 | fr00d | Does anybody know a good howto to setup asterisk with a bluetooth headset? |
| 09:49.03 | stoffell | fr00d, i think there's an article on that at nerdvittles website |
| 09:50.32 | fr00d | stoffell: That's just a article about connecting a cellphone. |
| 09:50.59 | stoffell | fr00d, oh, k, sorry ;) |
| 09:51.19 | fr00d | np! thnx for answer.. ;) |
| 09:51.21 | austinnichols101 | fr00d: not sure that asterisk cares about a bt headset at all. |
| 09:51.50 | fr00d | austinnichols101: I think there are addons to compile in that it works. |
| 09:51.54 | austinnichols101 | set up your machine to use the headset as speakers/mic and use a softphone - I've definitely made it work |
| 09:52.13 | austinnichols101 | what are you trying to get it to do? |
| 09:52.49 | fr00d | I thought chan_bluetooth is what i'm searching for. |
| 09:53.39 | fr00d | I compiled asterisk with it and tried per multipeer to connect my headset and cellphone (works), but I do not get any sound to my headset. |
| 09:54.05 | austinnichols101 | chan_bluetooth allows you to use a bluetooth compatible cell phone to connect to your Asterisk box |
| 09:54.26 | fr00d | I'm not so familiar with asterisk and so the demo breaks every incoming call to my cellphone after 90 secs. |
| 09:55.48 | fr00d | My scope is to use my cellphone and my headset via an asterisk. So I can use IAX2 and the standard cellphone functions. |
| 09:56.30 | austinnichols101 | is the cellphone a smartphone? |
| 09:57.12 | fr00d | What's a smartphone? It's a Nokia 6230i. |
| 09:57.41 | fr00d | There should be anywhere here also a 6310i.. |
| 09:58.03 | austinnichols101 | smartphone = running windows mobile, etc |
| 09:58.15 | austinnichols101 | although I'm not sure how 'smart' that is |
| 09:58.17 | fr00d | No! |
| 09:58.42 | fr00d | It's not a smartphone. |
| 10:00.03 | *** join/#asterisk zepmantra (i=waaa@203.76.202.78) |
| 10:00.07 | *** join/#asterisk cybergypsy (n=cybergyp@APoitiers-156-1-41-221.w86-213.abo.wanadoo.fr) |
| 10:00.34 | fr00d | austinnichols101: Should it be one? |
| 10:00.40 | *** part/#asterisk zepmantra (i=waaa@203.76.202.78) |
| 10:00.53 | mkl1525 | is there an option to always show date + time in the log output? |
| 10:01.09 | bulibuta | I installed with make samples, where can I find what the default auth user is and his pass? |
| 10:01.09 | tzafrir_laptop | ilbc has some minor licensing issues. It's basically being removed from Debian due to "strange" limitations in its usage license |
| 10:02.25 | austinnichols101 | fr00d: I don't think so. I was just thinking of other ways to solve the problem |
| 10:02.51 | tzafrir_laptop | bulibuta, what user? |
| 10:03.29 | bulibuta | well that's my question, with the sample files as my /etc/asterisk/ confs. what user is good for auth? |
| 10:04.25 | bulibuta | gtg bbl |
| 10:06.00 | cybergypsy | anyone else had this ? if i use exten => _00XXXXX.,1,dial(SIP/sipprovider,60,r) it thinks its an internal call , whereas exten => _900XXXXX.,1,dial(SIP/sipprovider/${EXTEN:1}.60,r) works ? |
| 10:06.29 | *** join/#asterisk vlrk (n=vlrk@202.65.134.119) |
| 10:08.19 | fr00d | I'll be happy when my headset works with asterisk. |
| 10:12.04 | *** join/#asterisk UlbabraB (n=caplaz@host241-43.pool8172.interbusiness.it) |
| 10:15.21 | *** join/#asterisk astra^^ (n=Im@59.145.104.74) |
| 10:15.26 | astra^^ | hello all |
| 10:16.06 | astra^^ | i need some help in configuring my new ATA for net2phone.. |
| 10:17.33 | vlrk | which ata u are using ? |
| 10:23.52 | astra^^ | tiger netcom |
| 10:24.16 | astra^^ | can u please help me conf it... :) |
| 10:24.49 | vlrk | sorry i donot have any idea on that tiger netcom |
| 10:25.22 | astra^^ | will u be able to figure it out .. if i can give u the access to it .. please.. |
| 10:26.30 | *** join/#asterisk Vyeperman (n=Vye@ip68-6-130-59.sd.sd.cox.net) |
| 10:30.01 | astra^^ | helloooooo |
| 10:30.05 | *** join/#asterisk apardo (n=apardo@62-15-114-125.inversas.jazztel.es) |
| 10:33.03 | chris_ast | Can we connect to a localhost DB from Asterisk PHP AGI? I am unable to connect and get data from it, please help. whereas that php connects to db and gets data if executed outside |
| 10:33.47 | chris_ast | astra,fr00d,vlrk: any ideas |
| 10:35.58 | *** join/#asterisk Hali_303 (n=surfk@dsl5402AC0D.pool.t-online.hu) |
| 10:36.09 | Hali_303 | hi! |
| 10:37.43 | Hali_303 | is the caller ID patent still active? for example, could caller-id be implemented in software? (I'm not sure if this is already in *) |
| 10:40.32 | *** join/#asterisk zotz (n=zotz@24.231.32.85) |
| 10:55.32 | mutilator | http://www.blu-haze.net/cgi-bin/schlabo/potd.pl?day=14&month=4&year=2006 |
| 11:06.09 | *** part/#asterisk fr00d (n=andi@zockt.normalerweise.net) |
| 11:14.19 | mkl1525 | I get "Got SUBSCRIBE for extensions without hint. Please add hint to 805066603621 in context from-sip" shown in the cli. I've got a snom360 - the number is attached to one of my snom keys and the snom tries to get the status of this number, but this message clutters the log file so is there a way to prevent this messages in the cli output? |
| 11:17.58 | Rawplayer | OMG! PONIES! |
| 11:19.50 | *** join/#asterisk zeedo (n=zeedo@reboot-robot.net) |
| 11:19.52 | *** join/#asterisk spunz_ (n=spunz@h081217096096.dyn.cm.kabsi.at) |
| 11:21.44 | *** join/#asterisk Vyeperman (n=Vye@ip68-6-130-59.sd.sd.cox.net) |
| 11:26.28 | drray | set verbose 0 |
| 11:26.33 | drray | er |
| 11:33.50 | *** join/#asterisk RippPPppE (n=ripppppp@203.115.71.253) |
| 11:33.51 | Hali_303 | I've got an FXS on span2, channel 1. how to configure this into zaptel.conf? |
| 11:34.21 | *** join/#asterisk cybergypsy (n=cybergyp@APoitiers-156-1-48-4.w86-213.abo.wanadoo.fr) |
| 11:34.33 | Hali_303 | I've tried signalling=fxso_ks group=2 context=internal channel => 1 |
| 11:35.01 | Hali_303 | but on asterisk startup it says it cannot load chan_zap, because there is no such as channel 1 |
| 11:35.55 | Hali_303 | in zttool, on my fxs device it says 1 total channels, 0 configured |
| 11:36.10 | *** join/#asterisk X-Rob (n=rob-x@dsl-202-173-151-24.qld.westnet.com.au) |
| 11:36.32 | drray | ztcfg -vv? |
| 11:36.40 | RippPPppE | Hali-have you done modprobe zaptel |
| 11:36.42 | RippPPppE | wcfxo |
| 11:36.45 | RippPPppE | wcfxs |
| 11:38.36 | Hali_303 | ztcfg -vv says: Channel 01: FXO Kewlstart (Default) (Slaves: 01) 1 channels configured. |
| 11:38.49 | Hali_303 | hmm this is strange |
| 11:39.12 | Hali_303 | RippPPppE: yes, zaptel and wcusb is loaded |
| 11:39.45 | Hali_303 | when I make changes to /etc/zapata.conf, do I have to reload the zaptel modules? |
| 11:39.57 | RippPPppE | nope |
| 11:40.05 | RippPPppE | you have to restart (*) |
| 11:40.06 | drray | ztcfg just fixed it |
| 11:40.25 | drray | if you run zttool now |
| 11:40.27 | RippPPppE | stop now |
| 11:40.34 | RippPPppE | and then restart * |
| 11:40.38 | drray | it should say 1 configured |
| 11:41.23 | Hali_303 | hm yes it is configured now! |
| 11:41.25 | Hali_303 | thx |
| 11:41.34 | RippPPppE | cool |
| 11:41.34 | drray | sure, help someone else later |
| 11:41.50 | RippPPppE | guys, i also have a small question |
| 11:42.06 | drray | you always have to run ztcfg -vv after modprobing your modules |
| 11:42.10 | RippPPppE | how does one configure a custom filename for Agent call recording |
| 11:42.18 | X-Rob | Hali_303, when you change zapata.conf you need to run 'ztcfg -vv' |
| 11:42.22 | X-Rob | ignore anyone who says otherwise. |
| 11:42.27 | RippPPppE | MONITOR_FILENAME=XXX does not do it |
| 11:43.06 | drray | X-Rob is more than likely correct |
| 11:43.16 | Hali_303 | any since my device is an FXS, ztconfig telling "Channel 01: FXO Kewlstart" is OK, right? |
| 11:43.17 | drray | like 93% certain |
| 11:43.35 | X-Rob | Hali_303, a FXO device uses FXS signalling, and vice versa |
| 11:43.36 | Hali_303 | since I have to use fxo signalling with an fxs |
| 11:44.28 | Hali_303 | or in zaptel.cong I have to set it up as FXS and only FXO in zapata.conf? |
| 11:46.28 | chris_ast | Can we connect to a localhost DB from Asterisk PHP AGI? I am unable to connect and get data from it, please help. whereas that php connects to db and gets data if executed outside |
| 11:46.50 | chris_ast | X-Rob,Hali_303,drray,RippPPppE: any ideas |
| 11:48.44 | RippPPppE | have you followed the basics of phpAGI |
| 11:48.49 | RippPPppE | http://www.voip-info.org/wiki-Asterisk+AGI+php |
| 11:49.21 | chris_ast | I have few agi's working in php, only this agi php has localhost db connection and it just fails |
| 11:51.09 | *** join/#asterisk X-Rob (n=rob-x@dsl-202-173-151-24.qld.westnet.com.au) |
| 11:51.14 | *** join/#asterisk apardo (n=apardo@62-15-114-125.inversas.jazztel.es) |
| 11:53.22 | *** join/#asterisk [SzYnA] (n=adam@bxj178.neoplus.adsl.tpnet.pl) |
| 11:53.27 | [SzYnA] | hello :) |
| 11:53.44 | [SzYnA] | somebody known how to register a conversation on asterisk ? |
| 11:54.00 | [SzYnA] | tfu.. |
| 11:54.05 | [SzYnA] | no register.. record :-) |
| 12:00.56 | drray | record application |
| 12:05.28 | tzafrir_laptop | [SzYnA], in Asterisk it is called "Monitor" |
| 12:05.45 | tzafrir_laptop | Try Monitor or MixMonitor |
| 12:08.28 | *** join/#asterisk tdonahue (n=tdonahue@208.51.101.201) |
| 12:09.34 | *** join/#asterisk doughecka_ (n=Miranda@unaffiliated/doughecka) |
| 12:10.24 | *** join/#asterisk PakiPenguin (n=uppal@linuxpakistan/admin/pakipenguin) |
| 12:14.57 | *** join/#asterisk seong (n=seong@218.111.64.209) |
| 12:16.34 | [SzYnA] | thank you :) |
| 12:17.28 | jsharp | Glorp |
| 12:22.08 | *** join/#asterisk cced2 (n=dev2003@222.33.36.205) |
| 12:22.12 | cced2 | :) |
| 12:26.06 | *** join/#asterisk pauldy (n=pauldy@24-155-86-154.ip.grandenetworks.net) |
| 12:31.46 | cced2 | :) |
| 12:31.52 | cced2 | who is online~ |
| 12:33.13 | *** join/#asterisk pif (n=ldm@zenon.apartia.fr) |
| 12:33.23 | *** part/#asterisk pif (n=ldm@zenon.apartia.fr) |
| 12:34.26 | cced2 | who is online~ |
| 12:36.10 | *** join/#asterisk pif (n=ldm@zenon.apartia.fr) |
| 12:41.07 | docelm0 | Hay anyone from Digium in here? |
| 12:41.33 | *** part/#asterisk chris_ast (n=Administ@59.93.56.163) |
| 12:43.27 | *** join/#asterisk Luhiwu (n=marsosa@200-127-3-20.cab.prima.net.ar) |
| 12:45.10 | docelm0 | Lots are online.. Why? |
| 12:48.18 | *** join/#asterisk cced (n=dev2003@222.33.36.205) |
| 12:49.18 | *** join/#asterisk nfi|ermes (n=ermsewrk@217.220.121.62) |
| 12:52.37 | isamar | hi |
| 12:53.32 | tzafrir_laptop | cced2, if you have a question, ask |
| 12:54.22 | cced | about asterisk-dev |
| 12:54.45 | cced | <PROTECTED> |
| 12:54.45 | cced | <PROTECTED> |
| 12:54.45 | cced | <cced2> what means of callback? this code in libpri |
| 12:55.02 | *** join/#asterisk X-Rob (n=rob-x@dsl-202-173-151-24.qld.westnet.com.au) |
| 12:55.42 | isamar | events? |
| 12:57.33 | cced | events? no .Sometime callback is call .I think |
| 12:57.59 | jsharp | A callback is a function that is called when another function is finished processing data. |
| 12:59.31 | cced | why use callback? puzzle.in Sip ,I see it |
| 13:02.13 | cced | callback is init call? |
| 13:02.21 | austinnichols101 | cced: no |
| 13:03.50 | austinnichols101 | cced: http://en.wikipedia.org/wiki/Callback_%28computer_science%29 |
| 13:04.02 | cced | faint. |
| 13:05.24 | cced | <PROTECTED> |
| 13:05.25 | cced | <PROTECTED> |
| 13:05.47 | cced | pri_io_cb deal call some . I confuse it |
| 13:08.37 | Katty | hihi. |
| 13:09.41 | cced | callback in cs,one function run finish,then pop stack,next function run |
| 13:19.54 | *** join/#asterisk cced2 (n=dev2003@222.33.36.205) |
| 13:21.23 | docelm0 | MEW MEW MEW MEW MEW MEW MEW MEW MEW MEW MEW MEW MEW MEW MEW MEW MEW MEW MEW MEW MEW MEW MEW MEW MEW MEW MEW MEW MEW MEW MEW MEW MEW MEW MEW MEW MEW MEW MEW MEW MEW MEW MEW MEW MEW MEW MEW MEW |
| 13:21.33 | cced2 | <austinnichols101> .thanks got |
| 13:22.14 | russellb | docelm0: stop doing that, sheesh |
| 13:23.13 | X-Rob | heh |
| 13:23.22 | X-Rob | I just realised, he even had the right number of mew's. |
| 13:24.01 | tzafrir_laptop | What's special about 48? |
| 13:24.11 | X-Rob | tzafrir, you have to sing along with it. |
| 13:24.24 | syle | rhymes with masterbate |
| 13:24.34 | docelm0 | hehe |
| 13:24.36 | docelm0 | ok ok |
| 13:24.41 | cced2 | One phone call is much the same as another |
| 13:24.41 | cced2 | A large (some would say ludicrous) number of signalling protocols have existed over the life of the public telephone network, and across the world's administrations. The features they offer, and the call model on which they are based, varies considerably. However, these days everything has to be squeezable through an SS7 or ISDN channel. The call model used by these modern protocols is, essentially, a superset of all other call models. Basing an API around |
| 13:24.41 | cced2 | // What is ISDN call model? |
| 13:25.36 | *** join/#asterisk hinckc (n=hinckc@ool-4354c627.dyn.optonline.net) |
| 13:25.38 | tzafrir_laptop | cced, have you looked up in wikipedia before asking? |
| 13:26.20 | cced2 | No . I should read.. |
| 13:29.33 | isamar | what is the best wholesale prepaid opensource system ? |
| 13:29.52 | blitzrage | russellb: !!! |
| 13:32.09 | X-Rob | blitzrage, he's hiding. |
| 13:32.14 | X-Rob | I think I scared him off. |
| 13:33.01 | blitzrage | aye |
| 13:33.34 | X-Rob | I am an ugly bastard, I know. |
| 13:35.13 | *** join/#asterisk boddy (n=e@212.58.24.138) |
| 13:35.16 | blitzrage | hmmm... isn't there an app or function that allows you to do 302 redirects from the dialplan? What provided that? Its not obvious in the list of modules in /usr/lib/asterisk/modules.... |
| 13:35.55 | boddy | hii could I use asterisk as sip gateway ? |
| 13:35.59 | russellb | blitzrage: Transfer() |
| 13:36.16 | blitzrage | russellb: oh it just got added into transfer? thought I saw in the bug tracker that it was a full other app |
| 13:36.28 | russellb | i'm pretty sure that's it ... |
| 13:36.29 | blitzrage | like app_siptransfer or something by jtodd |
| 13:36.35 | blitzrage | ok -- will check it, thx |
| 13:36.38 | russellb | yeah, it's just in Transfer |
| 13:36.47 | blitzrage | makes sense |
| 13:36.49 | X-Rob | Note that for SIP, if you transfer before call is setup, a 302 redirect |
| 13:36.49 | X-Rob | SIP message will be returned to the caller. |
| 13:36.56 | X-Rob | blitzrage, 'show application transfer' 8) |
| 13:37.15 | X-Rob | russellb, *poke* msgs. |
| 13:37.27 | russellb | X-Rob: i got them. I'm going to leave it open, and I'll fix up the wording later on |
| 13:37.36 | X-Rob | ok |
| 13:37.37 | blitzrage | X-Rob: thanks Mr. Obvious! :) |
| 13:37.39 | boddy | I am planing install asterisk and make connection between asterisk and Nortel meridian 1c over pri is this possible ? |
| 13:37.42 | russellb | X-Rob: thanks |
| 13:38.00 | X-Rob | russellb, np. Wasn't sure wether I should open a bug about it, considering it's such a trivial thing |
| 13:38.50 | boddy | ? |
| 13:40.45 | blitzrage | hrmmm... i think I can use transfer to control the number of calls a system handles... if I do a groupcount check, and the max calls limit for that box is reached, then I just do a 302 to another box, which then checks if it can handle the call, and so on down the line until either someone can handle the call, or the call needs to be dropped.... |
| 13:41.11 | blitzrage | should be interesting to try anyways -- oh -- and func_odbc r0xerz s0xerz |
| 13:42.20 | boddy | anybody help me ? |
| 13:42.46 | boddy | :D |
| 13:43.38 | *** join/#asterisk RippPPppE (n=ripppppp@203.115.71.253) |
| 13:44.28 | SwK | boddy yes its possible |
| 13:44.39 | SwK | depending on what exactly you plan to accomplish |
| 13:44.52 | syle | why not use SER? |
| 13:45.41 | SwK | syle: cause blitz like to reload every 5 minutes |
| 13:45.56 | SwK | heh |
| 13:46.05 | syle | lol |
| 13:46.06 | blitzrage | SwK: I'm rebuilding the network to stop doing that :) |
| 13:46.29 | blitzrage | SwK: everything will be on the fly from a local DB |
| 13:46.32 | boddy | I am planing client on internet connetct to sip server(asterisk) over adsl and call user over Meridan |
| 13:47.10 | SwK | blitzrage: :P |
| 13:47.11 | SwK | hahah |
| 13:47.17 | jsharp | You want to connect Asterisk to your Meridian by PRI? Yes, you can do that. |
| 13:47.18 | boddy | SwK |
| 13:47.33 | SwK | boddy: tie line style? |
| 13:48.10 | boddy | tie line style ? |
| 13:48.18 | SwK | just set up asterisk with what you need on it with a zap t1 card and set up the meridian as whatever you need... and go for it |
| 13:48.50 | boddy | which card that you advise me ? |
| 13:49.05 | SwK | how many T1s do you need in the asterisk box? |
| 13:49.09 | boddy | 1 |
| 13:49.16 | *** join/#asterisk Luhiwu (n=marsosa@200.63.89.242) |
| 13:49.40 | SwK | jsut get the TE110P (or similar) from digium |
| 13:49.59 | SwK | if you are in the states I'll sell you one |
| 13:51.05 | boddy | I am not which trademark |
| 13:51.34 | SwK | Digium... |
| 13:51.39 | SwK | www.digium.com |
| 13:52.16 | boddy | ok thanks |
| 13:52.19 | boddy | alot |
| 13:52.35 | SwK | http://www.digium.com/en/products/hardware/te110p.php |
| 13:58.25 | *** join/#asterisk umay (n=chris@70-101-61-50.dsl2-plymouth.roc.ny.frontiernet.net) |
| 14:00.45 | *** part/#asterisk RippPPppE (n=ripppppp@203.115.71.253) |
| 14:01.43 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 14:05.07 | *** join/#asterisk lokkju (n=lokkju@unaffiliated/lokkju) |
| 14:05.32 | *** join/#asterisk anthm (n=anthmct@CPE-69-76-83-52.wi.res.rr.com) |
| 14:05.32 | *** mode/#asterisk [+o anthm] by ChanServ |
| 14:06.37 | *** join/#asterisk malcolmd (n=malcolmd@pdpc/sponsor/digium/malcolmd) |
| 14:08.16 | grem_lin | Hi, does anybody have any knowledge of SIP VoIP providers returning response 476 "We dont accept private IP contacts", and how I would go about overcoming this problem? Thanks in advance... |
| 14:09.20 | docelm0 | grem_lin, who is giving you that back? |
| 14:09.29 | *** part/#asterisk ghento (n=ghento@CPE0014bfaf7d7a-CM00111ae4684c.cpe.net.cable.rogers.com) |
| 14:10.11 | docelm0 | grem_lin, and yes.. If your using asterisk as your UA you can force it to be a public IP in the contact if its local IP is private |
| 14:10.11 | grem_lin | PlusTalk, my ISP's VoIP service - I have SipGate (uk) running perfectly well though... and to correct what I said, it was response 479 - sorry |
| 14:10.42 | docelm0 | You could use Plainvoip.com Im sure they are overall cheaper and work weel |
| 14:10.45 | docelm0 | *well |
| 14:10.57 | grem_lin | I have nat=yes, qualify=yes for plustalk and then in general externip=myip |
| 14:11.10 | docelm0 | you have what you need then |
| 14:11.11 | grem_lin | docelm0, I'm only using them because I get so many free outbound calls per month :) |
| 14:11.20 | docelm0 | ohh |
| 14:11.27 | grem_lin | Or rather, trying to use them |
| 14:11.28 | docelm0 | Call em up |
| 14:11.42 | grem_lin | Yeah, I might just do that - thanks for your help |
| 14:13.42 | *** join/#asterisk tdonahue-laptop (n=tdonahue@208.51.101.201) |
| 14:13.59 | *** join/#asterisk ]expic (i=xuy@217.27.35.139) |
| 14:14.28 | ]expic | anybody knows how to get Openser support? |
| 14:14.36 | docelm0 | Plainvoip is still kickass tho.. :) .9c termination flat. |
| 14:14.44 | docelm0 | ]expic, PRAY! |
| 14:15.09 | *** join/#asterisk xermesx (n=ermsewrk@217.220.121.62) |
| 14:15.15 | *** join/#asterisk bkw_ (n=bkw_@adsl-70-142-62-199.dsl.tul2ok.sbcglobal.net) |
| 14:15.57 | *** join/#asterisk Cybertoy (n=maxim@dsl254-123-241.nyc1.dsl.speakeasy.net) |
| 14:16.04 | ]expic | i need something like prefix but for callerid in SER |
| 14:16.28 | ]expic | uac_replace_from but now idea how to add prefix and save current ID |
| 14:16.34 | ]expic | maybe somebody can help me |
| 14:17.25 | Luhiwu | is it there any way to log the used codec in cdr? |
| 14:20.20 | *** join/#asterisk x86 (n=x86@p3m/member/x86) |
| 14:22.03 | Katty | SwK: (= |
| 14:26.49 | docelm0 | Luhiwu, yes.. Do you know how to program in C? |
| 14:39.44 | *** join/#asterisk TripleF555 (n=TripleF5@modemcable084.12-201-24.mc.videotron.ca) |
| 14:39.49 | TripleF555 | ello |
| 14:39.49 | *** join/#asterisk inv_arp[work] (i=junya@adsl-11-225-195.mia.bellsouth.net) |
| 14:40.33 | TripleF555 | as i ssaid.. while unreged lol , my parkign does not seem to work .. its enabled in features.conf .. i added the include-> in the context of sip phone.. but #700 does nada |
| 14:42.14 | *** join/#asterisk wiseguy_ (n=chivilis@infospalvos.lt) |
| 14:42.15 | wiseguy_ | ;-) |
| 14:42.31 | wiseguy_ | CAPI INFO 0x349a: Non-selected user clearing |
| 14:42.40 | wiseguy_ | what does it mean in human language? |
| 14:43.43 | TripleF555 | no idea |
| 14:43.48 | TripleF555 | !google it ? |
| 14:43.57 | TripleF555 | , my parkign does not seem to work .. its enabled in features.conf .. i added the include-> in the context of sip phone.. but #700 does nada |
| 14:44.08 | TripleF555 | as does the *1, as does ## and # |
| 14:44.10 | TripleF555 | anyidea ? |
| 14:44.13 | TripleF555 | i need osmething else ? |
| 14:47.23 | *** join/#asterisk brodiem (i=1000@cpe-66-69-222-36.austin.res.rr.com) |
| 14:47.42 | brodiem | anyone have any issues running fax over the digium analog cards? |
| 14:48.17 | TripleF555 | ulaw ? |
| 14:48.18 | tzafrir_laptop | brodiem, where are faxes coming from? what's the trunk? |
| 14:49.12 | brodiem | tzafrir_laptop, right now a channelized T1, but will probably be switching to PRI shortly |
| 14:49.41 | tzafrir_laptop | brodiem, what's the zap timing source? |
| 14:49.53 | brodiem | It's configured for the telco side |
| 14:50.14 | *** join/#asterisk dapatrick (n=ubuntu@pool-70-110-137-116.phil.east.verizon.net) |
| 14:50.54 | brodiem | span=1,1,0,esf,b8zs |
| 14:51.39 | wiseguy_ | [incoming] |
| 14:51.39 | wiseguy_ | exten => s,1,Answer |
| 14:51.39 | wiseguy_ | exten => s,2,Background(Sound-file) |
| 14:51.46 | wiseguy_ | oh sorry |
| 14:52.26 | TripleF555 | someone ? |
| 15:02.20 | *** join/#asterisk Strom_M (n=strom@gateway.digium.com) |
| 15:02.53 | *** join/#asterisk littleball (n=littleba@cm55.epsilon171.maxonline.com.sg) |
| 15:03.55 | littleball | hello, how to monitor the status of the asterisk server? CLI is good, but is it possible to acess such info from external program? |
| 15:05.08 | Strom_M | littleball: manager interface |
| 15:06.16 | *** join/#asterisk bulibuta (n=bulibuta@80.97.12.10) |
| 15:06.35 | bulibuta | how can I disable auth? |
| 15:06.56 | bulibuta | I want a blind connection to asterisk, is that possible? |
| 15:07.28 | dapatrick | Is it possible to determine order of trunk use in a zaptel group? |
| 15:07.30 | littleball | Strom_M, thanks. let me read. anyway, any GUI tools available so that it is easy to admin/monitor the statusof asterisk system? |
| 15:07.52 | Strom_C | littleball: FOP is the only half-decent one |
| 15:08.02 | Strom_C | better to write, say, a plugin for nagios or something |
| 15:08.27 | littleball | Strom_M, what is FOP? |
| 15:08.35 | Strom_C | flash operator panel |
| 15:08.40 | file | my brother ran off the road lastnight, I am oddly not surprised |
| 15:09.00 | wiseguy_ | help me with - Non-selected user clearing? |
| 15:09.47 | TripleF555 | no |
| 15:09.55 | wiseguy_ | thanks |
| 15:09.55 | wiseguy_ | ;-) |
| 15:09.56 | TripleF555 | oh well |
| 15:10.01 | TripleF555 | ill wait for @home |
| 15:10.05 | TripleF555 | since parking not working |
| 15:10.35 | *** join/#asterisk Nodren (n=nodren@64.193.95.10) |
| 15:10.43 | blitzrage | file: back to work! |
| 15:10.54 | file | blitzrage: you are NOT my boss :P |
| 15:10.54 | blitzrage | heh :) |
| 15:11.00 | blitzrage | file: but I'm still in charge! |
| 15:11.00 | TripleF555 | i am |
| 15:11.00 | TripleF555 | ;) |
| 15:11.05 | file | blitzrage: pfft you wish |
| 15:11.21 | blitzrage | file: I don't need to wish for that which is true! |
| 15:11.55 | *** join/#asterisk brif8 (n=chatzill@rrcs-71-41-50-162.se.biz.rr.com) |
| 15:12.50 | brif8 | anyone use IPerf to test their network? I'm running with -c Asterisk* -l 33 and getting 1.25 MBytes 1.05 Mbits/sec is this good or bad ? |
| 15:13.37 | TripleF555 | so can one help me figure out why parking lot not woring on latest ? |
| 15:13.38 | x86 | brif8: on a 10mbps network, 1.25MB/sec is the theoretical max |
| 15:13.52 | x86 | brif8: on a 100mbps network, 12.5MB/sec is the theoretical max |
| 15:13.58 | TripleF555 | more like 80% since overhead |
| 15:14.01 | blitzrage | file: oh no you didn't! |
| 15:14.02 | *** join/#asterisk tuxd00d (n=tuxinato@adsl-63-205-99-182.dsl.lsan03.pacbell.net) |
| 15:14.12 | file | blitzrage: :D |
| 15:14.14 | blitzrage | file: I'm selling the laptop today |
| 15:14.21 | x86 | TripleF555: more like, the "theoretical max" ;) |
| 15:14.21 | file | blitzrage: excellent |
| 15:14.24 | file | blitzrage: to whom? |
| 15:14.27 | brif8 | x86: yes it's a 10mbps network switch, what about the 1.05 Mbits/sec |
| 15:14.30 | blitzrage | file: some random |
| 15:14.43 | file | blitzrage: exciting - prospect of getting the bike? |
| 15:14.44 | x86 | brif8: no idea what that is :P |
| 15:14.45 | blitzrage | file: also think I may have sold my car as well.... |
| 15:14.58 | brif8 | x86: thanks |
| 15:15.00 | blitzrage | file: I might be able to get a bicycle.... not sure if a motorcycle is in the cards this year :( |
| 15:15.05 | x86 | brif8: i'm just telling you 1.25MB/sec is about the absolute max you can push over a 10mbps network |
| 15:15.08 | file | blitzrage: awwwww |
| 15:15.17 | blitzrage | file: really should focus on that school loan unfortunately :( |
| 15:15.42 | brif8 | x86: ok so that would say to me I have most of the bandwidth available to me to use |
| 15:15.49 | blitzrage | file: think I might get an IBM laptop though.... |
| 15:16.01 | TripleF555 | file heard abotu parkign lot or im paranoid ? |
| 15:16.02 | file | pesky school loans |
| 15:16.19 | file | TripleF555: rephrase that so it makes sense to me, and maybe |
| 15:16.26 | file | blitzrage: how much do you owe... dare I ask |
| 15:16.35 | blitzrage | file: don't ask -- it'll scare you :) |
| 15:16.46 | file | yuck |
| 15:16.47 | x86 | brif8: if you're pulling 1.25MB/sec on a 10mbps network, you have ALL of the bandwidth ;) |
| 15:17.03 | *** join/#asterisk nextime (n=nextime@213-140-6-103.ip.fastwebnet.it) |
| 15:17.04 | brif8 | x86: right |
| 15:17.25 | TripleF555 | my parking not working |
| 15:17.37 | TripleF555 | <PROTECTED> |
| 15:17.59 | blitzrage | TripleF555: you still need to enable transfers with 't' or 'T' in Dial() |
| 15:18.22 | blitzrage | TripleF555: include => parking only allows you to get the calls back out of parking |
| 15:18.41 | file | achoo! |
| 15:18.50 | blitzrage | errr parkedcalls* :) |
| 15:18.56 | blitzrage | couldn't remember the default context name -- that's bad |
| 15:19.15 | file | eh er uh... bah |
| 15:19.17 | TripleF555 | oh |
| 15:19.26 | TripleF555 | lol |
| 15:19.28 | TripleF555 | ok |
| 15:19.28 | file | iaxtel has been up for 2 days and 1 hour, and this bug I'm trying to track down has not come up |
| 15:19.38 | Strom_C | blitzrage: the dCAP plaque monitors the mistakes you make and then self-destructs if you screw up too many times :) |
| 15:19.50 | *** join/#asterisk southtel (n=slester@c-69-180-24-164.hsd1.ga.comcast.net) |
| 15:19.58 | file | know what? I should become dCAP certified ... |
| 15:19.59 | blitzrage | Strom_C: lol |
| 15:20.16 | blitzrage | file: aye! take the test at Astricon in Dallas |
| 15:20.28 | blitzrage | file: assuming Digium sends you of course |
| 15:20.29 | *** join/#asterisk SplasPood (n=jwb@206.252.198.100) |
| 15:20.35 | Strom_C | I'm going to take it in San Jose in July |
| 15:20.36 | southtel | ALL: I'm having trouble configuring an inbound SIP trunk with asterisk 1.2. |
| 15:20.37 | file | I can use it to hit people over the head |
| 15:20.42 | PakiPenguin | evening |
| 15:20.47 | file | blitzrage: I wouldn't doubt it... |
| 15:20.50 | blitzrage | southtel: welcome to the club |
| 15:20.57 | Strom_C | southtel: just ask your question |
| 15:21.02 | Strom_C | blitzrage: the nub club? |
| 15:21.15 | blitzrage | Strom_C: :D |
| 15:21.35 | Strom_C | i think brookshire bought nubclub.com yesterday |
| 15:21.39 | file | Here at nub club we specialize in nubish activities, for all levels of nub! |
| 15:21.46 | southtel | blitzrage: I keep getting "Failed to authenticate user "+NXXNXXXXXX" |
| 15:22.21 | blitzrage | southtel: wow -- you've got serious issues then :) |
| 15:22.28 | southtel | blitzrage: where the NXXNXXXXXX is the from number. I'm on 1.2 and I've tried various flavors of "insecure". |
| 15:22.53 | blitzrage | southtel: I'm going to take a wild shot in the dark that NXXNXXXXXX is not a valid user in your sip.conf file |
| 15:23.12 | Strom_C | blitzrage: it appears that you have good night vision |
| 15:23.13 | southtel | No, and I don't feel like adding every number out there as users. |
| 15:23.33 | blitzrage | southtel: I'm also going to take a shot that the far end isn't doing pattern matching correctly and the DIal() app isn't using the right syntax |
| 15:23.37 | file | so add a peer entry, make sure the host is the absolute IP address the packets will be coming from, and make it insecure=very |
| 15:23.45 | file | and send it to a context |
| 15:23.45 | blitzrage | southtel: what file just said |
| 15:23.52 | *** join/#asterisk xphreak (n=zsolti@ns1.zrlocal.net) |
| 15:23.58 | file | if that's not working, then you look at sip debug to see if it's matching a user entry or what it's doing... |
| 15:24.15 | Strom_C | file: we need an even more insecure setting.... insecure=holyohmygod |
| 15:24.26 | xphreak | hello everyone |
| 15:24.29 | file | actually, very is deprecated... |
| 15:24.29 | blitzrage | user: matches on name in From: header || peer: matches on IP address of far end || friend: matches on name first, then IP address 2nd (if I remember correctly) |
| 15:24.32 | file | sort of... |
| 15:24.37 | jsharp | Or the lowest security setting: insecure=windows |
| 15:24.43 | Strom_C | HAHAH |
| 15:24.49 | southtel | file: Thanks, I'll try that. |
| 15:24.52 | blitzrage | insecure=invite,port == insecure=very |
| 15:24.57 | *** join/#asterisk bkw__ (n=brian@adsl-70-142-62-199.dsl.tul2ok.sbcglobal.net) |
| 15:25.36 | xphreak | have one question for you all if you are well informed about MeetMe ????? |
| 15:25.49 | file | just ask thy question |
| 15:25.52 | blitzrage | xphreak: no one is |
| 15:26.00 | blitzrage | xphreak: in fact, MeetMe doesn't even really exist |
| 15:26.03 | *** join/#asterisk klasstek (n=nunyobiz@sta-206-168-96-70.rockynet.com) |
| 15:26.18 | xphreak | I initiate calls using the Manager API and put them in a conference room |
| 15:26.21 | Strom_C | ~ask |
| 15:26.22 | jbot | Questions in the channel should be specific, informative, complete, concise, and on-topic. Don't ask if you can ask a quesiton first. Don't ask if a person is there, just ask what you intended to ask them. Better questions more frequently yield better answers. We are all here voluntarily. See also http://catb.org/~esr/faqs/smart-questions.html |
| 15:26.22 | xphreak | after 10 seconds |
| 15:26.26 | TripleF555 | -- Playing 'pbx-transfer' (language 'en') |
| 15:26.26 | TripleF555 | Apr 14 11:25:58 WARNING[4428]: res_features.c:824 builtin_atxfer: Did not read data. |
| 15:26.26 | TripleF555 | <PROTECTED> |
| 15:26.27 | xphreak | the calls just get hanged up |
| 15:26.31 | xphreak | why ??? |
| 15:27.00 | blitzrage | xphreak: more info needed |
| 15:27.04 | file | xphreak: you have to provide details... configuration examples... console output |
| 15:27.19 | xphreak | do you want an extract from the extensions.conf ? |
| 15:27.21 | file | there's tons of reasons why I could list off :P |
| 15:27.29 | file | console output would be best... |
| 15:27.31 | blitzrage | file: list them all! |
| 15:27.34 | *** join/#asterisk zepmantra (i=waaa@203.76.217.159) |
| 15:27.35 | *** join/#asterisk ast_freak|Laptop (n=jesse@68-112-130-237.dhcp.stcd.mn.charter.com) |
| 15:27.37 | Strom_C | file: you forgot "first born child" and "cookie platter" |
| 15:27.41 | file | blitzrage: I refuse!!! |
| 15:27.41 | xphreak | ok just a second please |
| 15:27.48 | file | xphreak: and DO NOT paste it directly in here |
| 15:27.54 | Strom_C | ~pb |
| 15:27.55 | jbot | well, pb is a place to paste your stuff without flooding the channel - try http://pastebin.com/, or http://pastebin.ca |
| 15:28.05 | *** join/#asterisk MikeJ[Laptop] (n=vircuser@d14-69-8-30.try.wideopenwest.com) |
| 15:28.19 | xphreak | Manager 'xphreak' logged on from 127.0.0.1 |
| 15:28.19 | xphreak | <PROTECTED> |
| 15:28.19 | xphreak | <PROTECTED> |
| 15:28.19 | xphreak | <PROTECTED> |
| 15:28.19 | xphreak | <PROTECTED> |
| 15:28.20 | xphreak | <PROTECTED> |
| 15:28.22 | xphreak | <PROTECTED> |
| 15:28.24 | xphreak | <PROTECTED> |
| 15:28.26 | xphreak | <PROTECTED> |
| 15:28.26 | file | he did it anyway |
| 15:28.28 | xphreak | <PROTECTED> |
| 15:28.29 | Strom_C | oh for god's sake |
| 15:28.30 | TripleF555 | ok |
| 15:28.30 | xphreak | <PROTECTED> |
| 15:28.32 | xphreak | <PROTECTED> |
| 15:28.34 | xphreak | <PROTECTED> |
| 15:28.36 | xphreak | <PROTECTED> |
| 15:28.38 | xphreak | <PROTECTED> |
| 15:28.40 | xphreak | <PROTECTED> |
| 15:28.42 | xphreak | this is what I get from asterisk console |
| 15:28.42 | Strom_C | AAAAAAAAAAAAAAAAAAAAGHHHHHHHHHHHHHHHHHHHH |
| 15:28.46 | xphreak | sorry |
| 15:28.46 | Strom_C | xphreak: |
| 15:28.48 | Strom_C | ~ |
| 15:28.49 | file | xphreak: we warned you not to do that. |
| 15:28.52 | Nodren | ~pastebin |
| 15:28.53 | jbot | from memory, pastebin is a place to paste your stuff without flooding the channel - try http://pastebin.com/, or http://pastebin.ca, or http://channels.debian.net/paste |
| 15:28.53 | Strom_C | ~pb |
| 15:28.55 | jbot | methinks pb is a place to paste your stuff without flooding the channel - try http://pastebin.com/, or http://pastebin.ca |
| 15:29.14 | xphreak | sorry haven't see the post |
| 15:29.18 | xphreak | wont happen again |
| 15:29.40 | file | xphreak: now grab a sip debug to see which side hung up, and pastebin it |
| 15:29.49 | xphreak | OK |
| 15:29.52 | file | this way we can verify that it was Asterisk that hung up |
| 15:30.07 | *** join/#asterisk rene- (n=rene-@dsl-201-128-115-107.prod-infinitum.com.mx) |
| 15:31.05 | TripleF555 | ok so i added the T and t's |
| 15:31.13 | TripleF555 | now i see transfer but exten no exist |
| 15:31.19 | TripleF555 | so i need an actual 700 extension ? |
| 15:31.22 | TripleF555 | what i make of it ? |
| 15:31.57 | *** join/#asterisk justinu|laptop (n=Justin@66.209.15.235) |
| 15:32.36 | rene- | hello, iam using realtime queues,,, what differences can i expect from static defined ones? i want to make sure callers dont get into an empty queue, and i dont want to send calls to agents that have been removed from the queue using realtime, i think that agent login doesntwork, and that asterisk only refreshes from database when a join occurs, so for logins do i need to use the addqueuemember app? and in 1.2.5 can i expect that |
| 15:33.34 | blitzrage | TripleF555: if you dial extension 700, and that is the default parking exten in features.conf, Asterisk will tell you which extension the call is parked on |
| 15:34.01 | TripleF555 | res_features.c:814 builtin_atxfer: Extension 700 does not exist in context cisco-out |
| 15:34.24 | *** join/#asterisk Foxtro (i=foxtro@251-79-246-201.adsl.terra.cl) |
| 15:34.26 | Foxtro | hi |
| 15:34.26 | DoktorGreg | ok, |
| 15:34.30 | DoktorGreg | hey all |
| 15:34.38 | TripleF555 | that the context i dial out to my cell cisco-out |
| 15:34.50 | Foxtro | how configure a sip cliente connection from nat to internet ? |
| 15:34.50 | xphreak | here it is |
| 15:34.52 | xphreak | http://pastebin.com/659696 |
| 15:34.59 | TripleF555 | 6 pbx_load_config: Unable to include context 'parkedcalls' in context 'cisco-out' |
| 15:35.00 | TripleF555 | also |
| 15:35.12 | xphreak | 1640 lines I'm afraid so |
| 15:35.31 | DoktorGreg | when a want to use my key systems features from an analog line, it tells me to flash, what is flash? |
| 15:35.59 | xphreak | sorry for the last post into the forum |
| 15:36.11 | Strom_C | DoktorGreg: open the line very briefly |
| 15:36.19 | xphreak | can anyone take a look at the pastebin I have posted ? |
| 15:36.20 | Strom_C | i.e. hang up then stop hanging up real quick |
| 15:36.38 | TripleF555 | ok i found it |
| 15:36.41 | TripleF555 | - Executing Park("Local/700@cisco-out-a8fe,2", "") in new stack |
| 15:36.41 | TripleF555 | <PROTECTED> |
| 15:36.49 | TripleF555 | Local ? why that default |
| 15:37.17 | DoktorGreg | um, how do i flash xlite? any ideas? |
| 15:37.29 | TripleF555 | [cisco-out] |
| 15:37.29 | TripleF555 | <PROTECTED> |
| 15:37.29 | TripleF555 | <PROTECTED> |
| 15:37.31 | TripleF555 | AND |
| 15:37.36 | TripleF555 | 700 => i,1,Playback(pbx-invalidpark) |
| 15:37.36 | TripleF555 | <PROTECTED> |
| 15:37.44 | DoktorGreg | plz use pastebin |
| 15:37.46 | Strom_C | what the hell |
| 15:37.47 | TripleF555 | hmmm wondering how the hell this works |
| 15:37.48 | TripleF555 | sorry |
| 15:37.50 | DoktorGreg | ~pastebin |
| 15:37.51 | jbot | rumour has it, pastebin is a place to paste your stuff without flooding the channel - try http://pastebin.com/, or http://pastebin.ca, or http://channels.debian.net/paste |
| 15:37.58 | Strom_C | TripleF555: exten => 700,s,1 |
| 15:38.08 | Strom_C | you've totally mangled the extension format |
| 15:38.18 | Strom_C | ALWAYS ALWAYS start with exten => |
| 15:38.28 | TripleF555 | lol didnt realize |
| 15:38.29 | TripleF555 | lol |
| 15:39.06 | TripleF555 | thats wha happens when cxopy paste wiki without checkng |
| 15:39.23 | Strom_C | TripleF555: yes...turning your brain on generally helps |
| 15:40.03 | TripleF555 | - Executing Park("Local/700@cisco-out-5208,2", "") in new stack AND Spawn extension (cisco-out, s, 1) exited non-zero on 'Local/700@cisco-out-5208,2' |
| 15:40.05 | TripleF555 | weir |
| 15:40.18 | TripleF555 | i need a locl context ? |
| 15:40.20 | TripleF555 | local ? |
| 15:40.47 | TripleF555 | trying that |
| 15:41.01 | TripleF555 | same |
| 15:41.02 | TripleF555 | darn |
| 15:41.19 | TripleF555 | context => parkedcalls |
| 15:41.22 | TripleF555 | from features |
| 15:41.34 | TripleF555 | that means it pushed them there.. whjy local then |
| 15:41.40 | Strom_C | TripleF555: pastebin your extensions.conf please |
| 15:41.47 | Strom_C | your entire extensions.conf |
| 15:43.02 | *** join/#asterisk Cardoe_work (n=dougg@gentoo/developer/Cardoe) |
| 15:43.13 | xphreak | hello people ? |
| 15:43.13 | xphreak | could anyone look at the http://pastebin.com/659696 ??? |
| 15:43.20 | xphreak | since I'm clueless |
| 15:43.25 | TripleF555 | http://pastebin.ca/49341 |
| 15:44.16 | Strom_C | TripleF555: remove the spaces after 700, |
| 15:44.22 | TripleF555 | ok |
| 15:44.27 | TripleF555 | exten 201 works |
| 15:44.28 | TripleF555 | lol |
| 15:44.39 | TripleF555 | can i repick it up from sourc e? |
| 15:44.46 | Strom_C | ?? |
| 15:45.51 | TripleF555 | let me check 700 now |
| 15:46.58 | *** join/#asterisk juanjoc (n=juanjoc@248-32-235-201.fibertel.com.ar) |
| 15:47.02 | Strom_C | TripleF555: and you've made the appropriate changes to features.conf right? |
| 15:47.22 | TripleF555 | y |
| 15:47.36 | TripleF555 | <PROTECTED> |
| 15:47.50 | TripleF555 | oh |
| 15:47.53 | TripleF555 | i need a park app |
| 15:47.55 | TripleF555 | lol |
| 15:48.17 | TripleF555 | no ? |
| 15:48.20 | *** join/#asterisk digime (n=digime@user-0cdf0g7.cable.mindspring.com) |
| 15:48.26 | TripleF555 | 700,1,parkk() |
| 15:48.27 | TripleF555 | ? |
| 15:48.30 | Strom_C | also, you can't have such a thing as exten => 700,s,1,whatever |
| 15:48.32 | *** part/#asterisk digime (n=digime@user-0cdf0g7.cable.mindspring.com) |
| 15:48.38 | TripleF555 | true |
| 15:48.43 | Strom_C | s and i are extension names, not priorities |
| 15:48.48 | TripleF555 | http://www.voip-info.org/wiki/view/Asterisk+call+parking |
| 15:50.06 | TripleF555 | take 43 |
| 15:50.53 | TripleF555 | <PROTECTED> |
| 15:50.53 | TripleF555 | <PROTECTED> |
| 15:50.56 | TripleF555 | ok this is bad |
| 15:51.01 | TripleF555 | you have a working ewxample of it ? |
| 15:52.07 | Strom_C | TripleF555: pastebin your features.conf |
| 15:52.21 | *** join/#asterisk lokkju (n=lokkju@unaffiliated/lokkju) |
| 15:52.29 | Strom_C | and take the exten => 700 garbage out of your extensions.conf |
| 15:53.16 | xphreak | please people if you can help me it would be good |
| 15:53.16 | xphreak | if you're busy with something else please let me know cause my time is ticking OUT :-) |
| 15:53.18 | xphreak | http://pastebin.com/659696 |
| 15:53.20 | TripleF555 | all 700 stuff ? |
| 15:53.59 | Strom_C | TripleF555: yes, your features.conf should specify the parking extensions, not your extensions.conf |
| 15:54.09 | Strom_C | xphreak: ok, stop whining and i'll look |
| 15:54.24 | TripleF555 | ok |
| 15:54.24 | xphreak | thanks :-D |
| 15:54.26 | TripleF555 | removed all |
| 15:54.29 | TripleF555 | now i get res_features.c:814 builtin_atxfer: Extension 700 does not exist in context cisco-out-cname |
| 15:54.44 | Strom_C | TripleF555: pastebin your features.conf |
| 15:55.01 | Strom_C | xphreak: what am i looking at? what is your problem? |
| 15:55.02 | TripleF555 | http://pastebin.ca/49343 |
| 15:55.40 | Strom_C | TripleF555: do a "reload" at the console |
| 15:56.04 | *** join/#asterisk saftsack (n=oliver@p54A7FEE0.dip.t-dialin.net) |
| 15:56.13 | *** join/#asterisk lzhang (n=rjrae@adsl-69-153-39-209.dsl.snantx.swbell.net) |
| 15:56.39 | TripleF555 | <PROTECTED> |
| 15:57.12 | saftsack | hi are there some news from digiums b410p isdn card? |
| 15:57.42 | Strom_C | saftsack: still in beta AFAIK |
| 15:57.49 | TripleF555 | .still no luck |
| 15:58.12 | TripleF555 | Parking context : parkedcalls |
| 15:58.16 | TripleF555 | that doesn exist |
| 15:58.16 | Strom_C | TripleF555: are you dialing 700 directly from the phone? |
| 15:58.19 | TripleF555 | yes |
| 15:58.24 | saftsack | Strom_C, is there any release date or are there any beta blogs? |
| 15:58.24 | Strom_C | um |
| 15:58.34 | Strom_C | you realize you're supposed to TRANSFER calls to 700, right? |
| 15:58.45 | TripleF555 | yes |
| 15:58.50 | TripleF555 | 201 works |
| 15:58.53 | xphreak | that's the SIP DEBUG turned on |
| 15:58.55 | *** join/#asterisk TiKiTaKi_ (n=Heaven@acwr75.neoplus.adsl.tpnet.pl) |
| 15:59.01 | TripleF555 | 700 is a phone simulator or virtual phone with moh |
| 15:59.03 | TiKiTaKi_ | hello |
| 15:59.04 | Strom_C | TripleF555: try transferring a call to 700 |
| 15:59.09 | xphreak | Strom_C: two calls made |
| 15:59.13 | lokkju | are there any web guis that make use of asterisk's realtime configuration database options? |
| 15:59.15 | TripleF555 | i can transfer to 201 |
| 15:59.16 | TiKiTaKi_ | does anyone has experience with asterisk + misdn card like AVMfritz? |
| 15:59.22 | TripleF555 | with is spa841 in front desk |
| 15:59.26 | xphreak | Strom_C: and put into a conference room |
| 15:59.27 | TripleF555 | but not 700 |
| 15:59.28 | TripleF555 | weird |
| 15:59.32 | xphreak | Strom_C: using MeetMe |
| 16:00.01 | saftsack | Strom_C, do you have any ideas? |
| 16:00.21 | Strom_C | xphreak: next time please put everything on one line...i dont need to go picking through the chat log to assemble the description of your problem |
| 16:00.39 | xphreak | Strom_C: ok |
| 16:00.50 | Strom_C | TripleF555: just for kicks, try restarting asterisk |
| 16:01.12 | TripleF555 | i did |
| 16:01.21 | TripleF555 | i always restart from scratch |
| 16:01.36 | Strom_C | TripleF555: that's usually unnecessary |
| 16:01.46 | DoktorGreg | how do i issue a loop back dial? |
| 16:01.59 | *** join/#asterisk justinu|laptop (n=Justin@wirelessdata-031-029.mycingular.net) |
| 16:02.23 | TiKiTaKi_ | does anyone knows any resource concerns isdn cards + asterisk , configuration etc..? |
| 16:02.33 | TripleF555 | http://pastebin.ca/49345 |
| 16:02.38 | xphreak | Strom_C: do you want me to write it now in one line or did you understand the problem ? |
| 16:02.50 | DoktorGreg | also restarting asterisk is potentially bad habit and inviolation of best practices |
| 16:02.53 | Strom_C | xphreak: I'm looking through your sip debug |
| 16:03.01 | TripleF555 | so |
| 16:03.04 | TripleF555 | #1 |
| 16:03.12 | TripleF555 | where in extensions do i include parkedcalls |
| 16:04.30 | Strom_C | xphreak: i really am not up for picking through 1500 lines of sp debug...you know your system better than I do - figure out which side is hanging up |
| 16:04.59 | *** join/#asterisk stoffell_h (n=stoffell@d51A5811B.access.telenet.be) |
| 16:05.01 | Strom_C | TripleF555: you're transferring to 700 in this example, right? |
| 16:05.03 | southtel | blitzrage: I'm still having issues getting that inbound sip working. I've tried what you and file suggested. |
| 16:05.07 | xphreak | Strom_C: I think I know which side is hanging up |
| 16:05.08 | TripleF555 | yes |
| 16:05.15 | southtel | blitzrage: does the choice of username matter? |
| 16:05.17 | xphreak | but don't know how to fix the problem |
| 16:05.19 | TripleF555 | # says trasnfer.. dialtone i type 700 and hangup |
| 16:05.27 | *** join/#asterisk alephcom (n=alephcom@host75.net14.mcsnet.ca) |
| 16:05.37 | Strom_C | are you waiting for it to tell you which extension it parked the call on? |
| 16:05.49 | TripleF555 | lol i just found it out |
| 16:05.51 | TripleF555 | lol |
| 16:05.57 | TripleF555 | i need #700# |
| 16:05.59 | southtel | blitzrage: Also, the sip trunk is using register...can that affect anything? |
| 16:06.00 | TripleF555 | shit this sucks |
| 16:06.05 | TripleF555 | thanks for help |
| 16:06.08 | Strom_C | TripleF555: |
| 16:06.19 | Strom_C | or you could just dial 700 and wait a few seconds |
| 16:06.37 | *** join/#asterisk MattH (n=MattH@63.174.244.195) |
| 16:07.14 | TripleF555 | works like a charm now |
| 16:07.19 | TripleF555 | well |
| 16:07.21 | TripleF555 | yeah |
| 16:07.22 | Strom_C | TripleF555: see, i assumed you already knew how to use the park application. I mean, really, what good is parking a call if you dont wait for it to tell you where it parked the call? |
| 16:07.25 | TripleF555 | nice stom |
| 16:07.33 | TripleF555 | i tought was defaulting to 701 |
| 16:07.47 | Strom_C | it does, but you have to wait for it to tell you |
| 16:07.51 | TripleF555 | yeah |
| 16:07.52 | TripleF555 | i see |
| 16:07.53 | TripleF555 | lol |
| 16:07.57 | Strom_C | ugh |
| 16:07.59 | Strom_C | user error |
| 16:08.17 | xphreak | Strom_C: look these calls are places using JMS messages, informations are extracted from it like extensions, context and so on and the call is made, and afterwards the calls are disconnected when 10 seconds pass. When I put two channels into conference room by calling an hardcoded extension that is using MeetMe the calls are not hanged up |
| 16:08.31 | xphreak | Strom_C: I'm just interested what could cause this |
| 16:08.43 | xphreak | Strom_C: to be able to know where to search the cause of the problem |
| 16:08.51 | Strom_C | JMS? |
| 16:09.02 | xphreak | Strom_C: JAVA MESSAGING SYSTEM |
| 16:09.10 | Strom_C | xphreak: don't shout at me |
| 16:09.11 | xphreak | Strom_C: trough JBOSS application server |
| 16:09.24 | TripleF555 | oh |
| 16:09.26 | TripleF555 | 10 seconds |
| 16:09.26 | *** join/#asterisk Whisk (n=whisk@whisk.gotadsl.co.uk) |
| 16:09.29 | xphreak | Strom_C: I'm not shouting :-) |
| 16:09.32 | TripleF555 | that my magic # |
| 16:09.43 | Strom_C | xphreak: typing in all caps == shouting |
| 16:09.44 | TripleF555 | i had that prob with double natted clients |
| 16:09.46 | *** join/#asterisk nextime (n=nextime@213-140-6-103.ip.fastwebnet.it) |
| 16:09.53 | xphreak | Strom_C: sorry did not know that |
| 16:09.54 | TripleF555 | put your app directly on the net |
| 16:10.01 | xphreak | Strom_C: I'm not often on forums |
| 16:10.06 | TripleF555 | 3-4 seconds latency is connection answer etc.. |
| 16:10.11 | TripleF555 | then 3-4 of rigning |
| 16:10.17 | *** part/#asterisk southtel (n=slester@c-69-180-24-164.hsd1.ga.comcast.net) |
| 16:10.18 | TripleF555 | plus 2 to release channel |
| 16:10.21 | xphreak | Strom_C: that's why I have pasted the stupid extensions.conf extract into the forum window |
| 16:10.21 | TripleF555 | that wehn it happens |
| 16:10.26 | Strom_C | xphreak: all-caps is universally considered shouting everywhere on the internet |
| 16:10.30 | TripleF555 | since next packets use the ip of internal etc etc |
| 16:10.36 | Strom_C | email, forums, IRC, everywhere |
| 16:10.41 | TripleF555 | BIGIP 5 caused that to us.. so we shipped it back |
| 16:10.53 | TripleF555 | try canreinvite=yes |
| 16:10.57 | xphreak | Strom_C: I really did not know that, If I have offended you then I'm sorry |
| 16:10.57 | TripleF555 | so asterisk keeps it |
| 16:11.01 | TripleF555 | and see if it works |
| 16:11.10 | TripleF555 | or out of nat |
| 16:11.17 | xphreak | Strom_C: I'm using all caps to point out to things |
| 16:11.25 | *** part/#asterisk a1fa (n=a1fa@207.210.210.202) |
| 16:12.19 | Qwell | Strom_C: And Capitalizing Every Word Makes One Look Like An Idiot |
| 16:12.26 | xphreak | Strom_C: accept the apology ? |
| 16:12.47 | Strom_C | Qwell: haha |
| 16:13.00 | Strom_C | xphreak: ok, i accept, but I don't know how to fix your problem |
| 16:13.10 | Qwell | Strom_C: Do you want 7.x or 8.x? |
| 16:13.20 | Strom_C | Qwell: which is better? |
| 16:13.26 | Qwell | 8.x has a few random bugs |
| 16:13.32 | xphreak | Strom_C: ok, no problem thanks anyways |
| 16:13.51 | TripleF555 | anyone have BOT that used to be here. .. note the capitalization is desbiing an acronym here |
| 16:13.52 | TripleF555 | ;) |
| 16:14.08 | Strom_C | Qwell: well then lets do 7.x |
| 16:14.22 | Qwell | okay, I'll send it this afternoon |
| 16:14.27 | Strom_C | <3 |
| 16:15.02 | Qwell | off to work |
| 16:16.14 | file | <PROTECTED> |
| 16:17.21 | Nodren | is there any way to tell if a person is already using their sip phone? |
| 16:17.30 | *** join/#asterisk Renacor (n=kvirc@ip21.farheap.net) |
| 16:17.38 | *** part/#asterisk xphreak (n=zsolti@ns1.zrlocal.net) |
| 16:17.49 | Strom_C | file: how poignant |
| 16:19.58 | *** join/#asterisk dsfr (n=dsfr@pdpc/sponsor/digium/dsfr) |
| 16:20.08 | *** join/#asterisk Samoied (n=Samoied@200-193-76-104.fnsce7006.dsl.brasiltelecom.net.br) |
| 16:22.30 | Nodren | am i really the first person to ever want to find out if someone is using their sip phone? i dont understand how there isnt a feature in asterisk to check this. |
| 16:24.51 | *** join/#asterisk spatulamaan (n=ggilmore@ip66-107-33-196.z33-107-66.customer.algx.net) |
| 16:24.57 | file | darn it I missed xphreak |
| 16:25.01 | file | his SIP stuff was on crack |
| 16:25.59 | *** join/#asterisk zepmantra (i=waaa@203.76.203.44) |
| 16:27.21 | *** join/#asterisk zeppelin_ (n=zeppelin@201.66.149.106) |
| 16:28.06 | littleball | hello, i am planing to design a web based interface to monitor/control the asterisk. In my mind, i am planing to use java-asterisk and java portlet. Who can give me some suggestions aboout java-asterisk and whether/how to implement this program? |
| 16:28.33 | Nodren | there already are some interfaces that do what your talking about |
| 16:28.38 | Nodren | should try those and save yourself some time |
| 16:28.45 | Nodren | freepbx is pretty well known |
| 16:28.58 | Nodren | its php/mysql driven |
| 16:29.07 | Strom_C | freepbx blows donkeys for quarters |
| 16:29.16 | Nodren | didnt say it was good |
| 16:29.37 | Nodren | but its a bigger start then a blank document to code in :P |
| 16:30.15 | littleball | (1)I want to do that iss because there should be a module for J2EE portlet, which itself is very good technology. (2)It will integrate better within my own project. |
| 16:30.53 | file | Katty: what kind? |
| 16:31.11 | Katty | file: devils food chocolate with peanut butter chips. |
| 16:31.52 | file | oooh ok! |
| 16:31.57 | file | thankies Miss Kitty Katty |
| 16:32.04 | terrapen | anybody know how to stop this? |
| 16:32.04 | terrapen | Apr 14 08:47:22 WARNING[18562]: db.c:67 dbinit: Unable to open Asterisk database |
| 16:32.10 | Katty | terrapen: yes! |
| 16:32.13 | Katty | terrapen: turn the machine off. |
| 16:32.17 | terrapen | please, do tell |
| 16:32.17 | terrapen | heh |
| 16:32.48 | Katty | couldn't resist ;) |
| 16:32.53 | littleball | I am monitoring the events emits by asterisk and logged by java-asterisk, i am thinking what should be the pivot of the design. Channel? Context? or somethign else |
| 16:34.15 | terrapen | what's the linux equivalent of ktrace/truss? |
| 16:34.23 | jsharp | strace |
| 16:34.26 | terrapen | ah,t hx |
| 16:34.54 | terrapen | haha, RHEL does not install it by default! |
| 16:34.58 | terrapen | what a shame! |
| 16:37.09 | terrapen | great, this message is completely random |
| 16:38.07 | Katty | Ooo! |
| 16:38.11 | Katty | hugs++ |
| 16:38.38 | zoa | ela |
| 16:38.40 | zoa | bastard! |
| 16:38.41 | terrapen | open("/usr/local/share/asterisk/astdb", O_RDWR|O_CREAT, 0664) = -1 EACCES (Permission denied) |
| 16:38.44 | terrapen | bingo. |
| 16:39.05 | jsharp | Not running Asterisk as root, then? |
| 16:39.16 | terrapen | hell no |
| 16:39.33 | DoktorGreg | will _X. capture every call coming in on that context? |
| 16:39.42 | jsharp | Yup. |
| 16:41.07 | terrapen | open("/usr/local/share/asterisk/astdb", O_RDWR|O_CREAT, 0664) = -1 EISDIR (Is a directory) |
| 16:41.09 | terrapen | doh doh! |
| 16:41.34 | terrapen | f |
| 16:41.38 | terrapen | err fixed! |
| 16:43.01 | *** part/#asterisk TripleF555 (n=TripleF5@modemcable084.12-201-24.mc.videotron.ca) |
| 16:45.35 | Nodren | i'm looking for a way to determine if a sip phone is in use.. does anyone know of anything that might be useful for me? |
| 16:45.51 | lzhang | Nodren, check hints |
| 16:47.47 | *** join/#asterisk Qwell[] (i=north@unaffiliated/qwell) |
| 16:49.57 | *** join/#asterisk elg (n=fugalh@falcon.fugal.net) |
| 16:50.36 | Foxtro | who can helpme ? |
| 16:50.37 | Foxtro | http://pastebin.com/659835 |
| 16:51.16 | terrapen | your bluetooth is teh sux0rs |
| 16:51.38 | terrapen | looks like you're missing some includes |
| 16:52.49 | Foxtro | like as? |
| 16:52.51 | Foxtro | :( |
| 16:55.47 | *** join/#asterisk shido6 (n=shido6@d38-45-81.commercial1.cgocable.net) |
| 16:56.19 | *** part/#asterisk elg (n=fugalh@falcon.fugal.net) |
| 17:04.04 | DoktorGreg | what is the history of the "dial 9 to get a line out" convention? |
| 17:04.52 | *** part/#asterisk Cardoe_work (n=dougg@gentoo/developer/Cardoe) |
| 17:04.59 | grem_lin | because when you're about to die it's easier to press '9' and extra time than another (string of) number(s)... I guess |
| 17:06.16 | Cybertoy | in Europe and Asia people use "0" instead of 9 |
| 17:06.50 | *** join/#asterisk trimi` (i=Whatt@62.162.243.210) |
| 17:11.38 | terrapen | anybody else see a lot of stutter when calling out with Teliax? |
| 17:13.35 | lzhang | teliax quality has been decent in my experience, but I primarily use them for inbound only now |
| 17:13.58 | terrapen | i've always had stutter with them, even at my previous employer |
| 17:14.09 | terrapen | im using voip-co4 |
| 17:14.13 | terrapen | (for outbound) |
| 17:14.39 | *** join/#asterisk DaveHope (n=dave@internal.davehope.co.uk) |
| 17:17.03 | DaveHope | Quick question. At times asterisk seems to omit the first second or two from output, (music, voice, etc). I've tried adding a Wait(2) but still loose a second or two of output, is there a workaround anyone can think of ? |
| 17:17.11 | *** join/#asterisk darkskiez (n=darkskie@bb-87-81-62-203.ukonline.co.uk) |
| 17:17.23 | *** join/#asterisk sandos (n=sandos@me-0-50-da-e0-bb-36.2.cust.bredband2.com) |
| 17:17.49 | Foxtro | hi any can helpme for compile bluetooth chan support for asterisk please |
| 17:17.50 | Foxtro | :( |
| 17:18.12 | file | DaveHope: Answer, Wait, then do whatever... it can take a second or two for the audio stream to be setup if you're using SIP for example |
| 17:18.57 | DaveHope | file: That's what I'm doing: Answer, Wair(2), Whatever(). Perhaps I just need more than 2 seconds, will try using 5 :) Thanks. |
| 17:19.08 | file | what technology? |
| 17:19.17 | DaveHope | file: SIP :) |
| 17:19.17 | lzhang | terrapen, in my experience using teliax or any other voip provider really is a crapshoot with regards to call quality |
| 17:19.27 | file | DaveHope: behind NAT? |
| 17:19.35 | DaveHope | file: Indeed. |
| 17:19.50 | file | ah |
| 17:19.57 | file | and you're using nat=yes - right? |
| 17:20.28 | DaveHope | file: I wasn't, no :) |
| 17:20.34 | DaveHope | file: Will do now though :) |
| 17:21.02 | file | well, it could explain why it took so long... because nat=yes will send to the internal LAN IP, and once it gets a packet from the device it'll switch over to the right IP address and port |
| 17:21.06 | file | but still, odd |
| 17:21.12 | file | done an rtp debug to see the audio flowing? |
| 17:21.59 | DaveHope | file: Bingo. That seems to have done it :) |
| 17:22.23 | file | :) |
| 17:23.41 | lokkju | sudo -u www-data asterisk -rx 'show applications'|grep :|awk -F : '{gsub( /^ +| +$/, "", $1 ) ; print $1}'|sort|while read line; do sudo -u www-data asterisk -r -n -q -x 'show application '$line''|perl -0777 -ne "\$f = \$_;\$f = s/\033\[(?:\d+(?:;\d+)*)*m//go;\$f = s/\</</go;\$f = s/\>/>/go;if (/application\s'(.*?)'.*?\[Synopsis\]\n(.*?)\n.*?\[Description\]\n(.*?)\n(.*?)\n\n/sg) {print '<application><name>' . \$1 . '</name><synopsis>' . \$2 . |
| 17:23.41 | lokkju | <PROTECTED> |
| 17:23.59 | lokkju | if anyone wants an xml list of all applications, their usage, etc.... |
| 17:24.06 | Qwell[] | umm |
| 17:24.06 | lokkju | that command will get em all for you |
| 17:24.14 | lokkju | (yes, one line |
| 17:24.33 | Foxtro | lokkju: this code is of freepbx? |
| 17:24.41 | lokkju | Foxtro, hell no |
| 17:24.47 | Foxtro | ahh |
| 17:24.47 | Foxtro | ok |
| 17:24.50 | *** join/#asterisk [TK]D-Fender (n=joe@66.11.164.239) |
| 17:25.19 | lokkju | something I wrote to extract the info, so I can feed it into intellisense |
| 17:25.35 | Foxtro | ahhh |
| 17:25.37 | lokkju | one long ass command - I have another that will do essentially the same thing for functions |
| 17:25.40 | Foxtro | cli to xml |
| 17:25.40 | Foxtro | :D |
| 17:26.10 | Foxtro | how can make chan_btp ? |
| 17:26.49 | lokkju | there was no way I was going to go through and manually put together a functions and applications list, specially when they could easily change depending on the madule you have installed, so.... dynamic seemed best |
| 17:28.07 | lokkju | eventualy goal though is an intellisense dialplan (well, more specifically IVR, for now) designer, in javascript |
| 17:29.47 | *** part/#asterisk mtaht3 (n=m@c-71-198-23-124.hsd1.ca.comcast.net) |
| 17:29.51 | lzhang | so I have asterisk set up with some zap lines and polycom 601 phones, and I'm occasionally getting one way communication "one party can hear the other but not vice versa"... how do I go about debugging this |
| 17:30.16 | lokkju | NAT involved anywhere? |
| 17:30.39 | Foxtro | [TK]D-Fender: helpmeeeeee :( |
| 17:30.40 | lzhang | lokkju, no NAT, Polycoms are on the same network as PBX |
| 17:30.42 | *** join/#asterisk thock (n=thock@216.119.93.253) |
| 17:31.37 | thock | Quick question: if i want to connect 2 plain phone lines to asterisk, and then out to two plain old phone lines via plain copper wire, i'll need a TDM400P with 2 FXO's and 2FXS's, right? |
| 17:31.50 | thock | those are all RJ-11 jacks on that card? |
| 17:32.08 | thock | i.e, https://shop.resv.net/Shops/ViewItem.aspx/27934028032-35768195584.htm this |
| 17:32.22 | *** join/#asterisk UrielS (i=Uriel@bzq-219-223-87.pop.bezeqint.net) |
| 17:32.24 | lzhang | don't know about the FXO/FXS stuff but yes, those are RJ11 |
| 17:32.41 | thock | That's what's confusing me |
| 17:32.57 | thock | the FXO is the phone-in, and the FXS is phone-out? |
| 17:33.07 | Nodren | do you need to make agents for queues? |
| 17:33.07 | lzhang | Red ones are phones in |
| 17:33.14 | lzhang | green should be phones out |
| 17:33.14 | Strom_C | thock: the new cards have RJ-11 jacks |
| 17:33.15 | Nodren | or can you just set the members to be sip extensions? |
| 17:33.21 | lokkju | thock, O is "from office" and S is "from station" |
| 17:33.23 | Strom_C | phone lines plug into FXO ports |
| 17:33.31 | Strom_C | telephone sets plug into FXS ports |
| 17:33.55 | lokkju | (office being the local phone provider, and station being each phone) |
| 17:34.43 | thock | okay |
| 17:34.50 | thock | https://shop.resv.net/Shops/ViewItem.aspx/27934028032-35768195584.htm so that is exactly what i need |
| 17:35.01 | *** join/#asterisk bjohnson (n=bjohnson@jecinc.tor.istop.com) |
| 17:35.05 | Strom_C | thock: we went over this last night |
| 17:35.19 | thock | i realize that |
| 17:35.27 | thock | but i'm still not 100% sure, that's all :/ |
| 17:35.43 | Strom_C | thock: if you're using IP phones you don't need FXS ports |
| 17:35.53 | Strom_C | FXS ports are for ANALOG TELEPHONES |
| 17:36.08 | Strom_C | IP phones plug into your data network |
| 17:36.44 | lzhang | FXS is useful for stuff like forwarding out to a fax machine |
| 17:37.00 | thock | i'll be using normal telephones and soft SIP phones |
| 17:37.16 | Strom_C | time for me to shower |
| 17:38.47 | tzafrir_laptop | ~fxsfxo |
| 17:38.49 | jbot | i guess fxsfxo is An FXO port expects to receive dialtone and receive ring voltage. An FXS port expects to provide dialtone and provide ring voltage. |
| 17:39.04 | tzafrir_laptop | bah, not clear enough |
| 17:41.12 | tzafrir_laptop | jbot, no, fxsfxo is An FXO port expects to rece fxsfxo is An FXO port expects to receive dialtone and receive ring voltage. An FXS port expects to provide dialtone and provide ring voltage.ive dialtone and receive ring voltage. You can connect it to a PSTN line from the Telco. An FXS port expects to provide dialtone and provide ring voltage. You can connect a phone or a fax to it. |
| 17:41.13 | jbot | okay, tzafrir_laptop |
| 17:42.37 | *** join/#asterisk TonyM (n=TonyM@adsl-solo-80-168-224-84.claranet.co.uk) |
| 17:43.04 | tzafrir_laptop | lokkju, the command you suggested will work if asterisk is not verbose |
| 17:44.31 | lokkju | tzafrir_laptop, which command? my functions/applications command? |
| 17:45.31 | lzhang | is there any way to get these polycoms to ring with a different tone for internal as opposed to external calls? |
| 17:45.42 | tzafrir_laptop | something that produces xml (or html?) from the output of asterisk -rx something |
| 17:46.11 | lokkju | tzafrir_laptop, should work in verbose mode or not, since it strips out anything that does not match what it is expecting |
| 17:46.16 | Qwell[] | tzafrir_laptop: You broke jbot |
| 17:46.31 | *** join/#asterisk shido6 (n=shido6@d38-45-81.commercial1.cgocable.net) |
| 17:46.42 | lokkju | tzafrir_laptop, but remember, the perl regex has to customized for each thing - applications, functions, etc all use a diff regex, a diff awk, etc |
| 17:46.54 | tzafrir_laptop | lokkju, hmm, you have a grep after that. Better make that grep 'grep -a' . |
| 17:47.12 | tzafrir_laptop | Qwell[], me? how? |
| 17:47.23 | Qwell[] | tzafrir_laptop: by feeding him garbage in the front :p |
| 17:47.31 | Qwell[] | jbot, no, fxsfxo is An FXO port expects to rece fxsfxo |
| 17:47.33 | jbot | Qwell[]: okay |
| 17:47.35 | Qwell[] | eh? |
| 17:47.45 | Qwell[] | well, now I broke him |
| 17:47.45 | lokkju | tzafrir_laptop, does not need -a, it works fine without it |
| 17:48.16 | Katty | paging file! |
| 17:48.19 | tzafrir_laptop | lokkju, I'm trying to remember what caused non-text chars to appear there. |
| 17:48.24 | Katty | file to the front desk plskthx! |
| 17:48.38 | lokkju | tzafrir_laptop, that is the ascii colors |
| 17:48.47 | tzafrir_laptop | But I do remember cases where the output was suddenly "binary" |
| 17:48.47 | file | HELLO |
| 17:48.53 | file | Katty: How may I help you today? |
| 17:49.14 | Katty | do you still have your conference setup with iax? |
| 17:49.36 | Katty | i presume that was yours, anyway. |
| 17:49.45 | file | wasn't mine, but I have a few on there :) |
| 17:49.52 | tzafrir_laptop | jbot, no, fxsfxo is An FXO port expects to receive dialtone and receive ring voltage. An FXS port expects to provide dialtone and provide ring voltage.ive dialtone and receive ring voltage. You can connect it to a PSTN line from the Telco. An FXS port expects to provide dialtone and provide ring voltage. You can connect a phone or a fax to it. |
| 17:49.54 | jbot | tzafrir_laptop: okay |
| 17:49.55 | file | IAX2/guest@neutrino.file-radio.com/300 or SIP/300@neutrino.file-radio.com |
| 17:50.00 | tzafrir_laptop | is that OK? |
| 17:50.02 | Qwell[] | tzafrir_laptop: Still b0rked :p |
| 17:50.04 | Qwell[] | jbot, no, fxsfxo is an FXO port expects to receive dialtone and receive ring voltage. You can connect it to a PSTN line from the telco. An FXS port expects to provide dialtone and provide ring voltage. You can connect a phone or a fax to it. |
| 17:50.05 | jbot | okay, Qwell[] |
| 17:50.05 | Katty | file: are you always in them? |
| 17:50.10 | file | Katty: nope |
| 17:50.15 | file | I'm rarely on the phone |
| 17:50.15 | Katty | :< |
| 17:50.18 | lokkju | tzafrir_laptop, the tr command strips out the ascii colors |
| 17:50.18 | Qwell[] | better. |
| 17:50.24 | Katty | file: is that your extension? |
| 17:50.25 | file | phones are silly |
| 17:50.33 | file | 145 is my extension |
| 17:50.42 | Katty | oh ah. |
| 17:50.43 | lokkju | (since the -n command, which is supposed to disable colors, does not for some reason on my install) |
| 17:50.57 | *** join/#asterisk Qwell[] (i=north@unaffiliated/qwell) |
| 17:51.05 | Qwell[] | woops |
| 17:51.36 | Katty | file: ringring? |
| 17:51.48 | tzafrir_laptop | I don't think that those are the colors, because I always run asterisk without a controlling terminal, which implies no colors |
| 17:51.49 | file | Katty: I've got nothing |
| 17:51.55 | Katty | file: oh. that's not what i meant. |
| 17:51.55 | lokkju | tzafrir_laptop, whoops, I was using tr, I forgot, I switched to doing it in perl - look at the s/// in the perl section |
| 17:51.59 | *** join/#asterisk mitcheloc (n=mitchelo@70-32-188-167.lmdaca.adelphia.net) |
| 17:51.59 | Katty | file: that's a general request. |
| 17:52.07 | Katty | file: obviously. |
| 17:52.18 | Katty | *hee* |
| 17:52.54 | thx2000 | X-Asterisk-HangupCause: Unallocated (unassigned) number <==anyone know why im getting that when trying to connect to teliax via sip? |
| 17:54.11 | *** join/#asterisk crochat (i=crochat@84-74-158-130.dclient.hispeed.ch) |
| 17:58.15 | xachen | a dizzy file? :O |
| 17:58.19 | file | ^_^ |
| 17:59.06 | DoktorGreg | darn nick server |
| 17:59.08 | xachen | I wish Les from Wy-com would just get out here and install my nw net :( |
| 18:03.25 | *** join/#asterisk Gamercjm (n=chris@pool-71-254-176-82.lsanca.fios.verizon.net) |
| 18:03.59 | Netgeeks | A timing source (zap card, zt dummy, etc.) is required for meetme and what else? I forget... |
| 18:04.15 | Katty | but! |
| 18:04.20 | Katty | not before hugging Netgeeks. |
| 18:04.26 | file | Netgeeks: IAX2 trunking |
| 18:04.33 | *** join/#asterisk annonimous (n=annonimo@201.137.44.113) |
| 18:04.37 | annonimous | good day |
| 18:04.43 | Netgeeks | Thanks file! |
| 18:04.44 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 18:04.49 | Katty | file: :< |
| 18:04.56 | file | eep |
| 18:04.57 | Katty | file: :<<< |
| 18:05.00 | file | evil face! |
| 18:05.11 | Katty | :> |
| 18:05.16 | file | a... |
| 18:05.16 | Katty | k, all better. |
| 18:05.18 | file | blueberry muffin! |
| 18:05.21 | Katty | !!! |
| 18:05.24 | file | %%% |
| 18:06.00 | *** join/#asterisk Dr-Linux (n=Nothing@202.125.141.6) |
| 18:06.53 | Dr-Linux | question, what module should i reload for "agents.conf" i only modified agents.conf and i wanna load it, i don't want whole reload |
| 18:06.55 | Dr-Linux | any clue? |
| 18:07.26 | [TK]D-Fender | "reload agents" |
| 18:08.12 | Dr-Linux | Thanks [TK]D-Fender |
| 18:08.45 | crochat | Hello ! |
| 18:09.15 | crochat | I have a problem of stability with my Asterisk configuration... |
| 18:09.30 | Dr-Linux | [TK]D-Fender: no luck! |
| 18:09.30 | Dr-Linux | LHR-PBX*CLI> reload agents |
| 18:09.31 | Dr-Linux | No such module 'agents' |
| 18:09.37 | annonimous | question, anybody knows why appears the "Got SIP response 481 "Call/Transaction Does Not Exist" back from 10.0.0.2"? |
| 18:09.46 | Netgeeks | try reload chan_agent.so |
| 18:09.57 | [TK]D-Fender | Dr-Linux : reaload the whole queue system then |
| 18:10.24 | Dr-Linux | [TK]D-Fender: i reloaded queues.conf |
| 18:10.54 | Dr-Linux | [TK]D-Fender: but i also wanna reload agents.conf module, i tried much but didn't find any help, that's why asking |
| 18:11.05 | crochat | After each call (with the SIP provider), after the hangup, Asterisk crashes !! |
| 18:11.14 | *** join/#asterisk apardo (n=apardo@62-15-114-125.inversas.jazztel.es) |
| 18:11.26 | PakiPenguin | ~seen hanben |
| 18:11.31 | jbot | PakiPenguin: i haven't seen 'hanben' |
| 18:11.41 | Dr-Linux | Netgeeks: thanks man, that works :) |
| 18:11.45 | Katty | Nivex: :< |
| 18:11.52 | Katty | Nivex: i don't appreciate reparsing. |
| 18:12.11 | Nivex | Katty: no I reparsed it to get it right. that's the problem with having too many channels open |
| 18:12.24 | Katty | k |
| 18:13.09 | Katty | yay 45 minutes! |
| 18:13.16 | Nivex | until? |
| 18:13.20 | Katty | i poof. |
| 18:13.29 | Qwell[] | to? |
| 18:13.40 | Katty | university of il |
| 18:13.47 | Qwell[] | for? |
| 18:13.55 | Qwell[] | </nosy> |
| 18:13.55 | Katty | the weekend |
| 18:13.55 | *** join/#asterisk sulex (n=sulex@pdpc/supporter/active/sulex) |
| 18:13.56 | Netgeeks | Champaign-Urbana! |
| 18:14.04 | Katty | chambana |
| 18:14.15 | Katty | i must say bye :< |
| 18:14.22 | Netgeeks | bye Katty |
| 18:14.23 | Katty | friend is leafing teh country for studying abroad, etc. |
| 18:14.29 | Katty | Netgeeks: not me, silly rabbit. |
| 18:14.35 | Netgeeks | oh, you are going there to say 'bye' |
| 18:14.38 | Katty | i am. |
| 18:14.40 | Qwell[] | Netgeeks: hey...do you still have your configs from Solaris *? |
| 18:14.42 | Netgeeks | I only went there for football games |
| 18:14.46 | Nivex | ahh the north, I do miss it from time to time |
| 18:14.49 | Katty | Netgeeks: that's silly. |
| 18:14.59 | Katty | Netgeeks: you should go there to frolic about the parks too. |
| 18:15.06 | Katty | Netgeeks: and watch pretty university girls run around in track suits. |
| 18:15.22 | Katty | Netgeeks: tonight i shall join in running about, and then frisby |
| 18:15.23 | Netgeeks | Qwell[] yep, alas, that preformance project is sitting someone on a back burner behind a few more back burners |
| 18:15.46 | Qwell[] | Netgeeks: any chance I can get those from you? I want to do some testing on my sunfire... |
| 18:15.59 | Qwell[] | and you used sipp, right? |
| 18:16.20 | Netgeeks | Ah, watching pretty university girls run around in track suits.... somewhere in you is a devil... |
| 18:16.32 | Katty | pfft. |
| 18:16.34 | Netgeeks | Qwell[] correct. |
| 18:16.37 | Katty | my halo still has the price tag on it. |
| 18:16.40 | Katty | thankyouVERYmuch. |
| 18:16.52 | Qwell[] | Netgeeks: cool, I need to learn about sipp this weekend then |
| 18:16.57 | Netgeeks | you know you can remove that tag if you are the final owner... |
| 18:16.59 | Qwell[] | gonna hammer the hell out of it |
| 18:17.02 | Katty | oh, right. |
| 18:17.04 | Katty | k. |
| 18:17.29 | Qwell[] | I have a feeling I'm going to very quickly max out my amd64... |
| 18:17.49 | Katty | :< |
| 18:17.53 | Netgeeks | You want my very preliminary thoughts on asterisk performance? |
| 18:18.01 | Qwell[] | Netgeeks: sure |
| 18:18.03 | Katty | is it quirky? |
| 18:18.09 | Katty | cause i thought it was quirky. |
| 18:18.56 | Netgeeks | There are two major bottlenecks in any asterisk implementation... transcoding which everyone knows about and kernel interrupt handling due to packet per second rates which many folks don't seem to know about |
| 18:19.17 | Netgeeks | if you use transcoding, on pretty much any PC platform, you won't need to worry about pps issues.... |
| 18:19.32 | Qwell[] | heh, transcoding on the sunfire is insanely slow |
| 18:19.38 | Netgeeks | the transcoding will overshadow it significantly |
| 18:19.40 | Qwell[] | 433 for lpc10 to ilbc |
| 18:19.52 | Qwell[] | it's like 9 for ulaw <> gsm |
| 18:20.04 | Netgeeks | However, if you are doing no transcoding, then the pps interrupt issue will become the limiting factory |
| 18:20.06 | Netgeeks | factor |
| 18:20.15 | Netgeeks | where did that y come from? Katty? |
| 18:21.07 | Katty | mew? |
| 18:21.12 | Katty | quirky is a verb. |
| 18:21.14 | *** join/#asterisk timscott (n=a@d198-53-19-216.abhsia.telus.net) |
| 18:21.18 | *** join/#asterisk PakiPenguin_ (n=uppal@linuxpakistan/admin/pakipenguin) |
| 18:21.31 | Katty | err, adjective |
| 18:21.51 | Katty | quirk being a noun, etc. |
| 18:22.14 | SplasPood | hrm, does grandstream make their firmware actually downloadable anywhere rather than pointing to their tftp (and yes, I'm too lazy to snifff for the filenames) |
| 18:22.28 | Netgeeks | Linux (even on solaris - I used Aurora) supports the 'New API' interface for certain ethernet interfaces. This support is a must if you plan to run more than a few hundred concurrent calls. so far the Broadcom BCM57XX interfaces as well as the intel 1000 Pro are supported |
| 18:22.50 | *** join/#asterisk sandos (n=sandos@me-0-50-da-e0-bb-36.2.cust.bredband2.com) |
| 18:23.00 | Qwell[] | I wonder what my sunfire has |
| 18:23.15 | Netgeeks | Under solaris, there is no support for dynamic interrupt coalescence like under linux, however, you can manually set interrupt coalescance under solari |
| 18:23.20 | timscott | bcm57xx: does that include Tigon3 cards? |
| 18:23.26 | Netgeeks | gah, now someone stole my s |
| 18:23.36 | Qwell[] | Netgeeks: swap your y and s keys |
| 18:23.50 | Netgeeks | timscott: the tg3 driver suppors NAPI (i.e. dynamic interrupt coalescance) |
| 18:23.50 | *** join/#asterisk justinu|laptop (n=Justin@cpe-69-204-225-187.nyc.res.rr.com) |
| 18:24.14 | annonimous | question, anybody here have experience with the gateway audiocodes?? |
| 18:24.18 | Qwell[] | Netgeeks: any idea if sipp works on solaris? |
| 18:24.19 | timscott | thought so |
| 18:24.35 | Netgeeks | Qwell, I see no reason why it wouldn't, but I didn't try |
| 18:24.48 | Katty | sipp is not sip i take it. |
| 18:25.05 | Qwell[] | I'm thinking about testing on loopback also |
| 18:25.10 | Qwell[] | just to see |
| 18:25.15 | Katty | what happened to sip? |
| 18:25.21 | Netgeeks | So to top it off, on the E4500. without NAPI active, I was able to get about 800 concurrent calls through the system before things started going downhill (no transcoding) |
| 18:25.22 | Katty | is it not all the rage this season? |
| 18:25.55 | Netgeeks | sipp is a program developed by some guys at HP that allows you to create hich sip call volumes and high call rates for testing |
| 18:26.05 | Netgeeks | ~sipp |
| 18:26.06 | jbot | Single In-Line Pin Package: The last "standard" PC RAM configuration before they started making SIMMsA lot like SIMMs, but they have little pins instead of contacts. SIPPs are to VLB what SIMMs are to PCI.. A suicide tool for geeks |
| 18:26.21 | Netgeeks | it's also that, but thats not what we are talking about |
| 18:26.55 | Qwell[] | is it fairly easy to config? |
| 18:27.12 | Katty | oh... /those/ sipps. |
| 18:27.17 | Netgeeks | With NAPI active (using a BCM5704 card), I ran out of juice on my testing systems before the sun showed any signs of problems |
| 18:27.45 | Katty | Netgeeks: hich? |
| 18:27.48 | Netgeeks | Qwell: it took me about 4 hours to figure out how to use sipp and create a source file for a 4 minute rtp stream |
| 18:27.50 | Qwell[] | Netgeeks: That's why I want to try sipp from the same box. It won't be a great estimate, but... |
| 18:28.03 | Katty | Netgeeks: oh, high. |
| 18:28.10 | Qwell[] | I know my amd64 won't be able to handle it |
| 18:28.14 | Netgeeks | :s/hich/high/g |
| 18:28.30 | Qwell[] | especially with only 1x gbit and 1x 100mbit |
| 18:28.36 | Katty | Netgeeks: what makes sipp easier for higher call volumes? |
| 18:29.00 | Katty | Netgeeks: i figured that was just the encoding or something. |
| 18:29.12 | Qwell[] | Netgeeks: and how much bandwidth did those 800 calls end up taking? |
| 18:29.32 | Katty | less port hoggy maybe? |
| 18:29.35 | Netgeeks | Katty: sipp just creates fake calls using a script you write.. for example, say I want to test asterisk running on a pentium 4 and figure out how many sip to sip calls it can handle |
| 18:29.41 | Katty | oooh. |
| 18:29.43 | Katty | k |
| 18:30.22 | Netgeeks | I would tell sipp to make 1000 calls at 10 calls per second and have them last for 5 minutes... then I would 'watch' the asterisk box by whatever means I feel is appropriate to determine when it starts to fail... |
| 18:31.05 | Netgeeks | and it comsumes maybe a 5th of the resources asterisk does to create a call. |
| 18:31.37 | Netgeeks | 800 calls too up about 70Mbps |
| 18:31.46 | Netgeeks | they were all ulaw |
| 18:31.52 | Qwell[] | oh, that |
| 18:31.54 | Qwell[] | s nothing |
| 18:32.20 | annonimous | question, anybody knows how to make a gw audiocode register to asterisk? cause he can receive calls but dont make =/ |
| 18:32.32 | Netgeeks | ~88 kbps per call was finally what I measured off the switch port |
| 18:33.02 | Qwell[] | brb |
| 18:33.18 | Katty | i don't think 13 phones and 8 lines are going to start overloading our asterisk box anytime soon. |
| 18:33.22 | Katty | however! sipp would still be fun to play with. |
| 18:33.29 | Katty | just for statistics. |
| 18:33.53 | Netgeeks | I think your system is safe from overloads, yes! |
| 18:34.00 | Katty | *grin* |
| 18:34.56 | Katty | i wonder how iax compares to sip, resourcy wise. |
| 18:35.06 | Katty | 10:1 |
| 18:35.34 | Netgeeks | I'm not sure to be honest. I've had too many issues with IAX that I gave up using it |
| 18:35.43 | Katty | iax is nice for port factor. |
| 18:36.01 | *** join/#asterisk skyboy (n=skyboy@72.18.13.34) |
| 18:36.05 | Katty | we don't use it often. |
| 18:36.31 | Katty | in fact, i think i'm the only one who does... |
| 18:36.45 | Netgeeks | it definately has some nice features, but the single-threaded port handler issue really hurts in high volume scenerios |
| 18:37.57 | jsharp | I did some IAX links over satellite and they kinda flopped, whereas the same Asterisk servers connected with SIP worked without a problem. |
| 18:38.07 | Qwell[] | Netgeeks: iax can be multithreaded now... |
| 18:38.15 | Netgeeks | I did hear that mark was going to rewrite.. ah, he already did? |
| 18:38.33 | Qwell[] | and file made it more dynamicly allocated |
| 18:38.47 | *** join/#asterisk Eggplant (i=No@dsl-731.cascadeaccess.com) |
| 18:38.49 | Qwell[] | dynamically |
| 18:39.24 | file | fancy buzzwords! |
| 18:39.38 | tainted- | hey what distro do u guys run |
| 18:39.55 | Netgeeks | I had another weird issue with iax that I never did debug... I wrote a 'clustering' system that used iax for inter-system communications between nodes (i.e. get a call on node a and destination was on node b) but for some reason when using iax in that scenerio I kept running into one way audio issues, switched to sip, and it just worked |
| 18:39.59 | Qwell[] | tainted-: I use astwin32 |
| 18:40.13 | tainted- | really? |
| 18:40.17 | Qwell[] | no |
| 18:40.17 | Netgeeks | I've been to lazy to debug that problem |
| 18:40.20 | tainted- | that's pretty crazy |
| 18:40.52 | tainted- | i was thinking about getting back into debian.. maybe even trying ubuntu |
| 18:41.10 | mitcheloc | does anyone know what the difference between Zap/1 and Zap/1-1 is? |
| 18:41.33 | mitcheloc | i.e. what does the dash and the data following it symbolize? |
| 18:44.24 | [TK]D-Fender | mitcheloc : the first is a tech/port combo, the second indicates a CANNEL running on it. |
| 18:44.29 | [TK]D-Fender | CHANNEL* |
| 18:45.31 | mitcheloc | so it is possible to have Zap/1-2? would that be the channel someone is on when you have call waiting? or "flash" over? |
| 18:48.17 | Katty | welllll, time to go. |
| 18:48.23 | Katty | wish me luck :< |
| 18:48.31 | Netgeeks | have fun! |
| 18:49.23 | [TK]D-Fender | mitcheloc : No, * doesn't have real control over flashing a line and they aren't seperate channels really. Mostly to conform to techs that support multipl channels like SIP. |
| 18:50.24 | mitcheloc | [TK]D-Fender: okay, i have a place to start now, thank you |
| 18:50.38 | jsharp | Buh. I hate Indian telemarketers and their crappy long delay, non echo-cancelled lines. |
| 18:50.58 | tzafrir_laptop | mitcheloc, Zap/1-2 is a temporary name. It is basically one leg of a call |
| 18:51.29 | tzafrir_laptop | Zap/1 is the "physical" channel on which that call leg passes |
| 18:51.44 | blitzrage | Qwell[]: lol |
| 18:51.48 | Qwell[] | blitzrage: ? |
| 18:51.58 | blitzrage | 12:39 < Qwell[]> tainted-: I use astwin32 |
| 18:51.58 | blitzrage | 12:40 < tainted-> really? |
| 18:51.58 | blitzrage | 12:40 < Qwell[]> no |
| 18:52.01 | Qwell[] | ahh, yes |
| 18:53.29 | mitcheloc | tzafrir: I understand, I'm only trying to get only the channel name, so I'm going to split the channel on the dash then and take the first index: string[] channel = "Zap/1-1".Split("-")" should get me "Zap/1" |
| 18:54.22 | lzhang | guys is there any way to choose which parking spot I would like a call to be parked to? |
| 18:54.34 | mitcheloc | however this won't work if dashes are used in the channel name, that's possible isn't it? |
| 18:56.20 | blitzrage | lzhang: nope |
| 18:56.30 | [TK]D-Fender | mitcheloc : Zap/1-1 *is* the channel name, in its entirety |
| 18:56.35 | blitzrage | lzhang: there is a new parking lot thing in the oej branch in SVN |
| 18:56.52 | mitcheloc | [TK]D-Fender: i understand, i want the physical channel name, not the temporary |
| 18:57.18 | blitzrage | isn't Zap/1-1 mean span 1, channel 1 -- thus needing both? |
| 18:57.20 | [TK]D-Fender | mitcheloc : If by that you mean caring about which physical Zap port, then sure... |
| 18:58.02 | mitcheloc | [TK]D-Fender: yes, but channel names can have dashes in them (as i'm applying this to sip as well), so that causes a parsing problem for me |
| 18:58.46 | Netgeeks | span 1 channel 1 is still Zap/1, span 1 channel 2 is Zap/2 Span 2 channel 1 on a system with multiple T1 pri's would be Zap/25 |
| 19:00.03 | *** join/#asterisk TripleF555 (n=TripleF5@modemcable084.12-201-24.mc.videotron.ca) |
| 19:00.04 | *** join/#asterisk ramo (n=ramo@59.92.196.130) |
| 19:00.06 | TripleF555 | on freebsd |
| 19:00.26 | TripleF555 | i get checking for TIFFOpen in -ltiff... no but i know i installed it.. that for the spandsp configure.. so i dont know how to fix |
| 19:02.01 | mitcheloc | Netgeeks: I understand, my example isn't clear enough, but this needs to apply to sip..i.e. when I use my broadvoice account i get something like this: SIP/broadvoice-home-572f |
| 19:02.26 | mitcheloc | i suppose then my question is, will the temporary channel name (572f) ever contain dashes? |
| 19:04.37 | Foxtro | [TK]D-Fender: can helpme for configure my intel 537 ? now i have a line connected fro testing |
| 19:05.32 | Foxtro | i have |
| 19:05.36 | Foxtro | *CLI> zap show status |
| 19:05.39 | Foxtro | Generic Clone Board 1 OK 0 0 0 |
| 19:07.07 | *** part/#asterisk TripleF555 (n=TripleF5@modemcable084.12-201-24.mc.videotron.ca) |
| 19:10.06 | [TK]D-Fender | Foxtro : What is there to help you with? You haven't described a PROBLEM! Whats wrong with it? |
| 19:10.33 | Foxtro | i need help from 0, for configure this |
| 19:13.10 | *** join/#asterisk b00mer_ (i=fwuser@blackhole.c5i.com) |
| 19:13.51 | *** join/#asterisk kainam (n=Jake@202.137.160.110) |
| 19:14.47 | annonimous | prbolem, got a gateway audiocode that doesnt register to the asterisk, tried everithing and doesnt connect, any idea?? =( (please) |
| 19:16.36 | Corydon76-home | I love people who say they've tried everything and expect you to be able to suggest something they didn't try |
| 19:16.48 | annonimous | Corydon76-home, lol |
| 19:17.34 | annonimous | Corydon76-home, well, i tried everything in the way to change codecs, change configurations tried other confs, restarted the gateway in factory defaylts (lol) what else can i try? |
| 19:17.54 | Corydon76-home | Dunno, I don't have an Audiocodes gateway to try |
| 19:18.00 | annonimous | for handle a Got SIP response 481 "Call/Transaction Does Not Exist" back from 10.0.0.2?? |
| 19:18.11 | annonimous | Corydon76-home, ah i see |
| 19:18.33 | Corydon76-home | What extension did it try to send? |
| 19:18.53 | annonimous | 101, 102, 103 104 (gateway with 4 ports fxs) |
| 19:19.07 | Corydon76-home | Do you have a context= for that sip user? |
| 19:19.16 | annonimous | yep |
| 19:19.25 | annonimous | cotext = from-internal |
| 19:19.29 | annonimous | */context |
| 19:19.31 | Corydon76-home | And those extensions exist in that context? |
| 19:19.35 | annonimous | yep |
| 19:20.01 | Corydon76-home | Might try insecure=very |
| 19:20.14 | annonimous | insecure=very? |
| 19:20.16 | annonimous | ok |
| 19:20.24 | annonimous | cause i tried with secure= invite |
| 19:20.28 | annonimous | ok hold on |
| 19:22.35 | x86 | is it possible to use DUNDi with multiple origination providers, to allow fail-over if one of them is down? |
| 19:22.47 | annonimous | Corydon76-home, nop, the same error 0/ |
| 19:22.50 | x86 | kind of like BGP in the networking world |
| 19:22.59 | Corydon76-home | Dunno then |
| 19:23.15 | *** join/#asterisk MacDome (n=eseidel@c-71-198-177-144.hsd1.ca.comcast.net) |
| 19:23.16 | annonimous | h thanks anyway |
| 19:23.18 | annonimous | =) |
| 19:24.32 | *** join/#asterisk MacDome (n=eseidel@c-71-198-177-144.hsd1.ca.comcast.net) |
| 19:24.32 | marcus2 | anyone here using CAC AB-II channel banks with asterisk? |
| 19:26.07 | [TK]D-Fender | Foxtro : What do you mean help from 0? When we last left off you had what seemed to be a pretty much ready setup |
| 19:26.57 | jsharp | marcus2: Not currently, but I've used them. Whatcha need? |
| 19:29.30 | *** join/#asterisk zamsler (n=zamsler@c-67-184-240-80.hsd1.il.comcast.net) |
| 19:33.57 | Foxtro | [TK]D-Fender: i cant call to external number with the line connected to intel 537 :( |
| 19:34.42 | marcus2 | i'm trying to figure out how to either (a) make disconnect supervision work in loopstart mode on the fxs ports |
| 19:34.51 | marcus2 | or (b) make groundstart work on the fxs ports |
| 19:35.38 | *** join/#asterisk angom_h (n=angom@red-corp-201.130.165.246.telnor.net) |
| 19:37.12 | [TK]D-Fender | Foxtro : SHOW us the error! And your configuration. How are we supposed to help you if there isn't anything to SEE? |
| 19:37.15 | [TK]D-Fender | ~pb |
| 19:37.16 | jbot | [pb] a place to paste your stuff without flooding the channel - try http://pastebin.com/, or http://pastebin.ca |
| 19:38.01 | Foxtro | u can help me step by step for configure ? |
| 19:42.39 | [TK]D-Fender | Foxtro : We already did all of that last time. what is there to start over? |
| 19:43.20 | Foxtro | sorry, but nothing work.. :( |
| 19:43.31 | Foxtro | i try, try, try, try. but dont work :( |
| 19:43.35 | [TK]D-Fender | Foxtro : SHOW US THE ERROR. |
| 19:43.51 | Foxtro | i delete the configuration... |
| 19:43.55 | Foxtro | now have all from 0 |
| 19:43.58 | marcus2 | heh |
| 19:44.19 | [TK]D-Fender | Foxtro : Then get reading the WIKI, I'm not going through the whole thing all over again from scratch. |
| 19:44.36 | DoktorGreg | well here i go to install the pri line... |
| 19:44.53 | Foxtro | ok... |
| 19:44.59 | Foxtro | thanks for all |
| 19:45.29 | DoktorGreg | I should get, if i have everything set correctly |
| 19:45.40 | DoktorGreg | "just works" tm status |
| 19:49.44 | *** part/#asterisk sandos (n=sandos@me-0-50-da-e0-bb-36.2.cust.bredband2.com) |
| 19:54.44 | *** join/#asterisk glm2k (n=glm@rrcs-24-199-11-46.west.biz.rr.com) |
| 19:57.04 | mikey2600 | hey everyone |
| 19:57.12 | *** join/#asterisk MikeJ[Laptop] (n=vircuser@adsl-69-208-115-227.dsl.sfldmi.ameritech.net) |
| 20:04.30 | *** join/#asterisk dager (n=dager@c-69-251-68-26.hsd1.md.comcast.net) |
| 20:04.44 | *** join/#asterisk Skinzy (n=tom@81-179-68-232.dsl.pipex.com) |
| 20:11.42 | tainted- | is there a gui for editing extensions.conf/sip.conf/iax.conf out there? i need some ideas |
| 20:12.57 | [TK]D-Fender | Forget about them - there's an idea ;) |
| 20:13.29 | [TK]D-Fender | maybe gedet, kedet, OOo Write would be more appropriate? ;) |
| 20:13.48 | [TK]D-Fender | (wow, a lot of typos in that one!) |
| 20:13.56 | [av]bani | \o/ |
| 20:14.09 | tainted- | gedet, kedet? |
| 20:14.12 | LostFrog | vi, emacs, nano, pico |
| 20:14.34 | tainted- | i need to create a CSR friendly gui for managing user definitions |
| 20:14.38 | LostFrog | ed, sed, awk |
| 20:14.38 | tainted- | not necessarily dialplans |
| 20:15.14 | [TK]D-Fender | tainted- : GEDIT, KEDIT. |
| 20:15.37 | tainted- | they don't get to use those |
| 20:15.39 | [TK]D-Fender | tainted- : How badly do they think they want it? |
| 20:16.12 | [TK]D-Fender | tainted- : because once you go down the GUI route, your soul is sold... |
| 20:16.28 | tainted- | it's just to add / remove users |
| 20:16.35 | tainted- | add / edit / remove users |
| 20:16.38 | tainted- | in a cleaner format |
| 20:16.42 | [TK]D-Fender | tainted- : I don't know of any that modularize to that leve... |
| 20:16.50 | tainted- | no need for them to ssh into the boxen and mod conf files |
| 20:16.51 | mitcheloc | shh [TK]D-Fender: thats not true X( |
| 20:16.53 | [TK]D-Fender | tainted- : best to write it yourself. |
| 20:17.11 | tainted- | i can write it no problem.. just looking for design ideas |
| 20:17.19 | [TK]D-Fender | mitcheloc : I said *I* don't know of any that modularize to that level.... |
| 20:17.23 | *** part/#asterisk dager (n=dager@c-69-251-68-26.hsd1.md.comcast.net) |
| 20:17.41 | [TK]D-Fender | tainted- : What in IAX would you define? |
| 20:17.43 | tainted- | implementing the [context] foo = bar structure in a cli |
| 20:17.48 | tainted- | in a gui i mean |
| 20:17.55 | *** join/#asterisk grem_lin (n=gremlin@your-face.scares.me.uk) |
| 20:17.59 | mitcheloc | tainted-: realtime? |
| 20:18.08 | tainted- | yea |
| 20:18.21 | mitcheloc | ~asterisk realtime |
| 20:18.31 | tainted- | right now i've got a web-based textbox that upload sip.conf |
| 20:18.36 | tainted- | err |
| 20:18.37 | mitcheloc | ~jbot asterisk realtime |
| 20:18.39 | tainted- | textarea |
| 20:18.46 | mitcheloc | i dunno, heh, how do you use jbot again? |
| 20:20.01 | tainted- | i love it when end users say 'doesn't work' |
| 20:20.11 | tainted- | 'just doesn't work.. i dunno' |
| 20:20.19 | [TK]D-Fender | tainted- : Maybe make a standardized form and have it general a "users" file that gets "included" my sip.conf and the rest.... |
| 20:20.31 | [TK]D-Fender | by* |
| 20:20.38 | mitcheloc | can't you use http://www.voip-info.org/wiki-Asterisk+RealTime ??? |
| 20:20.44 | mitcheloc | and then write a script to manage the database |
| 20:21.02 | tainted- | yea that's what built |
| 20:21.10 | tainted- | what i built in db tables |
| 20:21.19 | tainted- | but i'm looking for gui ideas |
| 20:21.27 | mitcheloc | then point asterisk @ your tables with realtime |
| 20:21.41 | tainted- | i'm shocked that no one uses gui to provision user accts |
| 20:22.22 | jsharp | A lot of people use their own custom gui. |
| 20:22.46 | tainted- | jsharp what does yours look like |
| 20:24.27 | Shaun2222 | how can i make the queue stop disconnecting the caller when all agents are busy? |
| 20:24.54 | jsharp | Its fairly simple on the front end. It asks for the phone MAC address, phone type, has a button for "assign next number", and a drop down menu for customer and customer site. |
| 20:26.53 | tainted- | and that just hardcodes a [foo] settings=something context using the values? |
| 20:27.09 | *** join/#asterisk hfb (n=hfb@pool-71-116-252-188.lsanca.dsl-w.verizon.net) |
| 20:28.34 | *** part/#asterisk mitcheloc (n=mitchelo@70-32-188-167.lmdaca.adelphia.net) |
| 20:29.16 | jsharp | I use the realtime SIP stuff to keep track of the phones, and I have very generic dialplans to acutally distribute the calls. I don't have to change my dialplans at all. |
| 20:31.32 | *** join/#asterisk a1fa (n=a1fa@207.210.210.202) |
| 20:31.33 | a1fa | hehe |
| 20:31.34 | a1fa | Connected to Asterisk SVN-tag-1.2.7.1-r19816M |
| 20:31.41 | a1fa | *.code is becoming sloppy |
| 20:31.54 | Qwell[] | a1fa: How is that sloppy? |
| 20:31.58 | a1fa | hehee |
| 20:32.01 | *** join/#asterisk Defraz (n=t0tal@fw.centrisys.com) |
| 20:32.10 | a1fa | bug release 1.2.7.1? |
| 20:32.26 | a1fa | you think they would make sure Page() works before releasing 1.2.7, but no |
| 20:32.27 | Qwell[] | What, you've never seen kernel versions like that? |
| 20:32.32 | a1fa | yeah |
| 20:32.32 | a1fa | :) |
| 20:32.36 | a1fa | but kernel is different |
| 20:32.40 | Qwell[] | how? |
| 20:32.54 | a1fa | bcos you dont go outthere and update your kernel that often |
| 20:33.01 | a1fa | at least i dont |
| 20:33.05 | Qwell[] | You don't need to upgrade asterisk that often either |
| 20:33.11 | a1fa | tru |
| 20:33.12 | a1fa | you got me |
| 20:33.25 | a1fa | hehe |
| 20:33.28 | a1fa | at least it is working, ya know |
| 20:35.06 | a1fa | hexe hexe hexe |
| 20:35.26 | a1fa | Qwell[] : are you running that remote manager? |
| 20:35.33 | Qwell[] | what remote manager? |
| 20:35.41 | a1fa | ARM :) |
| 20:35.47 | a1fa | * remote manager |
| 20:36.57 | a1fa | i guessnot |
| 20:40.50 | lzhang | so if my phones are on the same network as asterisk, I should set nat=no? |
| 20:41.07 | a1fa | yes |
| 20:46.04 | jsharp | Does setting nat=yes actually break anything? |
| 20:46.12 | Qwell[] | jsharp: yes, it can |
| 20:46.22 | jsharp | Oh. |
| 20:46.26 | Qwell[] | nat=yes on a lan, will send the devices your externip |
| 20:46.27 | timscott | :) |
| 20:46.32 | Qwell[] | which...obviously won't work |
| 20:47.22 | jsharp | Oh. So nat=yes with * behind the NAT breaks local phones. |
| 20:47.40 | *** join/#asterisk MacDome (n=eseidel@A17-255-96-116.apple.com) |
| 20:49.03 | *** join/#asterisk MacDome (n=eseidel@A17-255-96-116.apple.com) |
| 20:50.17 | Qwell[] | jsharp: So, you would never put * on a corporate LAN? |
| 20:50.29 | *** join/#asterisk MacDome (n=eseidel@A17-255-96-116.apple.com) |
| 20:50.31 | Qwell[] | it never needs to talk to the outside, so giving it a public IP is incredibly stupid |
| 20:50.54 | *** join/#asterisk Timmerman (n=Lucas@201-34-213-117.gnace703.dsl.brasiltelecom.net.br) |
| 20:50.55 | jsharp | If it never needs to talk to the outside, you'd never give it an externip. |
| 20:51.02 | Qwell[] | touche |
| 20:52.38 | Timmerman | Asterisk can call for phone numbers in directly way using IAX VoIP? Or it just to build PBX Centrals? |
| 20:52.48 | *** join/#asterisk websae (n=icechat5@CPE-24-167-204-30.wi.res.rr.com) |
| 20:52.54 | Qwell[] | Timmerman: It can do whatever you like |
| 20:53.38 | websae | a friend of mine owns a business--he asked me to price a PBX system for him and all new phones, he needs about 25 phones--any suggestions anyone? |
| 20:53.42 | *** join/#asterisk trimi` (i=Whatt@62.162.243.54) |
| 20:53.45 | Timmerman | Qwell[], in the true, I want use a software like Skype but free to call from telephone numbers, it is possible? |
| 20:53.51 | Qwell[] | websae: hire a consultant? |
| 20:53.57 | Qwell[] | Timmerman: no |
| 20:54.01 | Qwell[] | everything is possible...except skype |
| 20:54.08 | websae | Qwell: I am quite familiar with Asterisk |
| 20:54.11 | trimi` | hey any1 can tell my why asterisk doesnt detect bussy signal on my zap x100p ? |
| 20:54.16 | Qwell[] | websae: Then you should know how to price phones :P |
| 20:54.17 | trimi` | i have bussydetect=yes |
| 20:54.28 | websae | what would be good phones to use in this situation |
| 20:54.28 | Qwell[] | trimi`: because the x100p is pretty much junk |
| 20:54.37 | Qwell[] | websae: How much does he want to spend? |
| 20:54.48 | blitzrage | I like the Linksys SPA-942's and Polycom IP501s |
| 20:54.53 | trimi` | Qwell[] what other zap config file is there ????? |
| 20:54.57 | websae | he didn't give me a budget |
| 20:54.59 | trimi` | i remember they were 2 |
| 20:55.06 | Qwell[] | zaptel.conf and zapata.conf |
| 20:55.12 | websae | i think we should stick with middle of the road |
| 20:55.17 | trimi` | where is zaptel.conf located ? |
| 20:55.20 | Qwell[] | websae: So midgrade polycoms |
| 20:55.22 | Qwell[] | ala 501 |
| 20:55.32 | *** join/#asterisk MikeJ__ (n=vircuser@adsl-69-208-116-212.dsl.sfldmi.ameritech.net) |
| 20:55.35 | trimi` | i found it |
| 20:55.36 | trimi` | thnx |
| 20:56.09 | *** join/#asterisk MikeJ[Laptop] (n=vircuser@adsl-69-208-116-212.dsl.sfldmi.ameritech.net) |
| 20:56.41 | Timmerman | Qwell[], in another words, I can build a PBX center in my home, but to call from another tel, ASterisk will use the telephon line and not the TCP Protocol (connected on internet) |
| 20:56.42 | Timmerman | ? |
| 20:56.45 | blitzrage | speaking of IP501's, I'm looking to replace one I had to give away -- anyone got a used one they want to sell? |
| 20:56.57 | Qwell[] | Timmerman: huh? |
| 20:57.03 | Qwell[] | It can use whatever you want |
| 20:57.13 | blitzrage | Timmerman: you can route calls from whichever technology to whichever OTHER technology you want |
| 20:57.34 | Timmerman | yeah, man, but I want OpenSource softwares and technologies... |
| 20:57.37 | blitzrage | SIP -> ZAP, ZAP -> IAX2, IAX2 -> SIP, etc.... |
| 20:58.06 | blitzrage | Timmerman: then use Asterisk -- Asterisk does not give you free telephone calls -- PSTN is a separate network, and you need to pay for termination to it |
| 20:58.11 | Qwell[] | Timmerman: then use it |
| 20:58.16 | trimi` | any unlimited plan on IAX, all i see are in SIP, does any1 know |
| 20:58.40 | SplasPood | anyone happen to have the latest grandstream BT-101 firmware in a zip? |
| 20:58.41 | Timmerman | but what I really wish know if building a PBX, with Asterisk I will call telephon numbers using internet |
| 20:58.46 | *** join/#asterisk Gamercjm (n=chris@pool-71-254-176-82.lsanca.fios.verizon.net) |
| 20:59.00 | tainted- | trimi` what are u using it for |
| 20:59.03 | Qwell[] | Timmerman: If you pay for it, sure |
| 20:59.06 | websae | tainted: how are you? |
| 20:59.07 | blitzrage | Timmerman: no -- unless you terminate to the PSTN via a service provider |
| 20:59.12 | trimi` | <tainted-> termination to USA |
| 20:59.17 | trimi` | i need unlimited plans |
| 20:59.19 | mog_work | or have everyones ip |
| 20:59.25 | mog_work | to do direct uri to uri calling |
| 20:59.25 | tainted- | websae hey dude.. how did the meeting go |
| 20:59.25 | blitzrage | Timmerman: Asterisk does not equal free phone calls |
| 20:59.32 | Qwell[] | mog_work: or at least, one persons IP...who didn't read REAME.security :) |
| 20:59.33 | file | mine is 127.0.0.1 |
| 20:59.35 | Timmerman | Qwell[], :D |
| 20:59.41 | tainted- | trimi` unlimited meaning you're running a business off of it? |
| 20:59.47 | mog_work | echo echo echo |
| 20:59.48 | Timmerman | Qwell[], Is what I need know! |
| 20:59.53 | Timmerman | thanx!! |
| 20:59.55 | MikeJ[Laptop] | File not found! |
| 20:59.55 | websae | tainted: eventually i'll get some business out of him, he wants me to price a new PBX system and 25 phones (wants to get a new one) eeeks |
| 21:00.15 | tainted- | websae that's great! |
| 21:00.22 | tainted- | good job |
| 21:00.39 | websae | so I'll have to see what I can come up with for that |
| 21:00.43 | tainted- | was their cable modem sufficient for call quality |
| 21:00.53 | SplasPood | Apr 14 17:00:53 WARNING[9474]: channel.c:2051 ast_indicate: Unable to handle indication 3 for 'SIP/69.9.166.254-09e28ab8' |
| 21:01.04 | websae | I didn't do the test yesterday---I will be doing that this coming week |
| 21:01.09 | SplasPood | Anyone have any idea what causes that? Sending outgoing calls via a Dial() to an AS5300 |
| 21:01.16 | websae | how's your termination carrier working out? |
| 21:02.04 | tainted- | websae oh man.. working on a deal to provide service to an entire building with service in downtown LA |
| 21:02.27 | tainted- | taking up all my time atm |
| 21:02.36 | websae | anything i can do to help? |
| 21:02.58 | bkw__ | tainted-, wasabi |
| 21:03.25 | bkw__ | SplasPood, if you do that you have to turn some stuff off on the 5300 |
| 21:03.41 | bkw__ | tone ringback alert-no-PI |
| 21:03.42 | *** join/#asterisk shido6 (n=shido6@d38-45-81.commercial1.cgocable.net) |
| 21:03.44 | bkw__ | on your dial peer |
| 21:03.45 | tainted- | bkw_ just trying to put together some stuff for the CSR |
| 21:03.51 | *** join/#asterisk sevard (n=kynan@198.174.233.135) |
| 21:04.02 | tainted- | posted some digg articles on freeswitch |
| 21:04.05 | tainted- | want the url? |
| 21:04.07 | SplasPood | bkw: if I do what? |
| 21:04.08 | bkw__ | sure |
| 21:04.18 | bkw__ | SplasPood, put that in the dial peer |
| 21:04.25 | sevard | Does * have a S.T.U.N. server built in? If not.. could somebody recommend one? |
| 21:04.31 | bkw__ | sevard, no |
| 21:04.35 | SplasPood | bkw: if I put that in my dial peer that will "fix" the problem, or... |
| 21:04.44 | bkw__ | SplasPood, on the 5300 yes |
| 21:04.45 | tainted- | http://digg.com/linux_unix/New_Open_Source_Softswitch_Speaks_GoogleTalk |
| 21:04.46 | SplasPood | or that'd be the cause OF the problem.. |
| 21:04.46 | bkw__ | it should fix it |
| 21:04.46 | SplasPood | ok |
| 21:04.50 | sevard | bkw__: Can you recommend one? |
| 21:05.01 | SplasPood | bkw: other than the error message, what issues (if any) would that cause? |
| 21:05.07 | tainted- | sevard just roll your own |
| 21:05.13 | MikeJ[Laptop] | bkw__!! |
| 21:05.36 | bkw__ | SplasPood, not sure |
| 21:05.39 | sevard | tainted-: it's that simple of a server? |
| 21:05.41 | bkw__ | Mike yes? |
| 21:05.57 | SplasPood | bkw: I'll see what happens... Thanks man |
| 21:05.58 | MikeJ[Laptop] | hi |
| 21:06.02 | tainted- | sevard umm.. |
| 21:06.10 | tainted- | sevard http://sourceforge.net/projects/stun/ |
| 21:06.59 | bkw__ | stun is kewl |
| 21:07.09 | sevard | 'kewl' |
| 21:07.19 | SplasPood | bkw: hrm... the voip dialpeer already has that... |
| 21:07.38 | trimi` | hey which is the best VOIP ATA BOX |
| 21:07.48 | trimi` | what you say about linksys pap2 |
| 21:08.02 | trimi` | any1 had experiance with ATAs ? |
| 21:09.06 | tainted- | trimi` grandstream makes good, cheap atas |
| 21:09.31 | trimi` | <tainted-> does they support low bandwidth codecs |
| 21:09.38 | trimi` | i think linksys its cheapest |
| 21:10.09 | tainted- | ur wrong |
| 21:10.13 | tainted- | grandstream is cheapest |
| 21:10.19 | bkw__ | SplasPood, then ignore it |
| 21:10.29 | tainted- | and yes - they support low bandwidth codecs like g729 |
| 21:11.08 | SplasPood | bkw: I'm having an odd issue where one of my DIDs just has a Dial() out to another number.. when that number returns BUSY even tho I see it jumping properly based upon my ${DIALSTATUS} it seems to loop over and over and over |
| 21:11.49 | VoIPMasta | trimi`: I would go for Cisco's ATA 186 |
| 21:12.31 | trimi` | <VoIPMasta> they are 2 expensive |
| 21:12.43 | VoIPMasta | trimi`: you asked about "the best" not "the cheapest" |
| 21:12.54 | VoIPMasta | trimi`: "the cheapest" will never be "the best" |
| 21:13.05 | trimi` | greandstream had 1 FXO + 1FXS for the half price of the cisco |
| 21:13.31 | tainted- | SplasPood paste your dialplan.. sounds like u have issues |
| 21:13.48 | SplasPood | tainted: heh def seems like it... whats the pastebin url again |
| 21:13.52 | VoIPMasta | trimi`: I'm talking about hardware quality, multi-protocol support, highly customizable, reliability, manufacturer support |
| 21:14.15 | VoIPMasta | trimi`: that's what makes something "the best" for me |
| 21:14.34 | tainted- | ~pb |
| 21:14.35 | jbot | well, pb is a place to paste your stuff without flooding the channel - try http://pastebin.com/, or http://pastebin.ca |
| 21:14.42 | trimi` | VoIPMasta are you cisco's sales representative ;) |
| 21:14.47 | trimi` | your talking like them |
| 21:14.53 | Qwell[] | cisco shill? |
| 21:15.04 | Qwell[] | VoIPMasta: and the cheapest most certainly can be the best |
| 21:15.23 | VoIPMasta | trimi`: no, I'm a very happy Cisco user for over 7 years, I've used routers, atas, switches |
| 21:15.42 | Qwell[] | VoIPMasta: good, go test my chan_skinny patches |
| 21:15.43 | VoIPMasta | Qwell[]: Name one single "cheap" ata that is capable of handing 2 or more protocols |
| 21:15.50 | Qwell[] | You didn't say ATA |
| 21:15.57 | VoIPMasta | [16:07] <trimi`> hey which is the best VOIP ATA BOX |
| 21:16.04 | trimi` | VoIPMasta come on i had a ISDN router i bought it 600$, i wish i wasnt that dump |
| 21:16.07 | Qwell[] | < VoIPMasta> trimi`: "the cheapest" will never be "the best" |
| 21:16.19 | trimi` | cisco its too expensive |
| 21:16.21 | VoIPMasta | trimi` was asking about ATAs |
| 21:16.32 | Qwell[] | anyhow...since you like cisco gear so much...test my chan_skinny patch, so you can use skinny with * |
| 21:16.44 | tainted- | VoIPMasta why would u need to handle two or more protocols |
| 21:16.52 | Qwell[] | Would be nice to figure out the right line settings for an ata186 |
| 21:17.00 | *** join/#asterisk stkn (i=nobody@gentoo/developer/pdpc.active.stkn) |
| 21:17.00 | VoIPMasta | Qwell[]: is your patch in *'s release? |
| 21:17.06 | Qwell[] | no, it's on the bug tracker |
| 21:17.28 | VoIPMasta | tainted-: portability, sometimes I need a h.323 client and sometimes a SIP client |
| 21:17.29 | Qwell[] | 6859 |
| 21:17.37 | VoIPMasta | Qwell[]: let me look at it |
| 21:17.46 | [TK]D-Fender | ATA186 is overpriced... SPA-2002 is good for most uses, and Mediatrix' 1021 si really nifty though less friendly. |
| 21:17.57 | Qwell[] | right now, I only have one line for the ata186 I think. If you could test that, and tell me if it's right or not... |
| 21:17.58 | VoIPMasta | [TK]D-Fender: but those do SIP only |
| 21:18.36 | [TK]D-Fender | I believe the Mediatrix may be flash-able to H.323 |
| 21:18.59 | [TK]D-Fender | I know the craptastic PS168X ones can, but well... craptastic... |
| 21:19.43 | VoIPMasta | I started working with 186 instead of sipura/linksys because I needed a single way to remotely configure them and lock them (so my customers won't mess them up) and I couldn't find any reliable documentation on sipura/linksys, but Cisco even offered a tech to teach us how to do it |
| 21:19.55 | VoIPMasta | that kind of support is always valuable |
| 21:20.19 | [TK]D-Fender | VoIPMasta : You do get what you pay for, and I'm sure it didn't come cheap |
| 21:20.19 | tzanger | VoIPMasta: probably cost was a concern, but why not polycom + lockdown |
| 21:20.37 | VoIPMasta | tzanger: I haven't used polycom |
| 21:20.38 | Qwell[] | tzanger: does polycom make an ata? |
| 21:20.46 | [TK]D-Fender | tzanger : well he IS talking ATA's, and H.323 |
| 21:20.54 | tzanger | ohhhhhh.. ATAs... my mistake |
| 21:21.02 | VoIPMasta | [TK]D-Fender: sure, as you said, you get what you pay for |
| 21:21.05 | [TK]D-Fender | tzanger : And don't forget H.323 ;) |
| 21:21.10 | tzanger | [TK]D-Fender: ewwwwwwwwww |
| 21:21.15 | VoIPMasta | [TK]D-Fender: that's what I was telling trimi` |
| 21:21.26 | tzanger | [TK]D-Fender: oh I have a list of DHCP option codes... not sure if they're all supported by polycoms |
| 21:21.33 | tzanger | [TK]D-Fender: are you on the asterisk-ontario mailing list? |
| 21:21.33 | [TK]D-Fender | VoIPMasta : your situation sounds somewhat rare and I guess there is a real limit to your options |
| 21:21.33 | VoIPMasta | You can't expect to have the best ata for US$75 |
| 21:21.53 | [TK]D-Fender | tzanger : Being in QC, nope ;) |
| 21:21.56 | VoIPMasta | [TK]D-Fender: rare? |
| 21:21.59 | tzanger | [TK]D-Fender: ha |
| 21:22.10 | tzanger | [TK]D-Fender: shoot me your email address, I'll forward the message over |
| 21:22.13 | tzanger | you might find use for it |
| 21:22.43 | [TK]D-Fender | VoIPMasta : in HERE anyways, and as far as commodity equipment goes. You are looking a little more "old-school" integrator style trying to make too many other peoples things work. |
| 21:23.23 | tainted- | VoIPMasta u can lockdown a grandstream ATA easily.. i still don't get your multiple protocols argument |
| 21:23.42 | [TK]D-Fender | tainted- : Its a need for him and I'd accept it at that. |
| 21:23.59 | VoIPMasta | tainted-: We have SIP, IAX and H.323 gateways, sometimes we just NEED to connect to a H.323 gateway, so we need a H.323 client |
| 21:24.46 | Qwell[] | VoIPMasta: So, yeah...when you test that, post a note on the bug with anything you find |
| 21:25.04 | VoIPMasta | [TK]D-Fender: Most of our users are on the lame side, if they read somewhere that they can modify their ATAs to "do whatever" they'll try it... so they keep sending them to be reconfigured |
| 21:25.20 | VoIPMasta | [TK]D-Fender: It's easier to just lock them and have a "backdoor" to update configs remotely |
| 21:25.36 | *** join/#asterisk justinu|laptop (n=Justin@wirelessdata-031-029.mycingular.net) |
| 21:25.38 | VoIPMasta | Qwell[]: can you pm me your email address to send you any comments? |
| 21:25.42 | [TK]D-Fender | VoIPMasta : that factor can be done in the Linksys ATA |
| 21:25.52 | Qwell[] | VoIPMasta: anything that gets added to the bug, will get emailed to me |
| 21:26.03 | VoIPMasta | [TK]D-Fender: You can partially lock them, but the "dial in reset" will still work |
| 21:26.36 | VoIPMasta | [TK]D-Fender: Linksys will sell "pre-locked" ATAs if you commit to large scale buyouts |
| 21:26.42 | VoIPMasta | [TK]D-Fender: something like what Vonage did |
| 21:26.52 | [TK]D-Fender | VoIPMasta : Yeah, I haven't seen anyone bypass that, however it will effectively FLUSH the box... |
| 21:27.15 | VoIPMasta | [TK]D-Fender: with Cisco 186 I can lock them and there's no way to unlock them |
| 21:27.20 | *** join/#asterisk hinckc (n=hinckc@ool-4354c627.dyn.optonline.net) |
| 21:27.55 | VoIPMasta | [TK]D-Fender: Cisco also provided a modified firmware that takes out the setup options that we don't use, so when we log in to the web interface we just see what we need |
| 21:27.55 | [TK]D-Fender | VoIPMasta : well if was a train-once scenario on the ATA 186 and you get a good deal on bulk pricing, why not? |
| 21:28.25 | VoIPMasta | There's one more good thing about Cisco's 186 |
| 21:28.29 | VoIPMasta | it supports GSM :) |
| 21:28.42 | [TK]D-Fender | VoIPMasta : Its hard to protect yourself against the dumb people... especially the creative kind :) |
| 21:28.47 | VoIPMasta | Linksys ATAs support G.729 but not in both channels at the same time, due to hardware limitations |
| 21:29.01 | [TK]D-Fender | VoIPMasta : I know, that annoys me... |
| 21:29.19 | [TK]D-Fender | I hate it when companies cut STUPID corners... |
| 21:29.56 | [TK]D-Fender | like Polycom's low-end PoE concept. add it and charge jsut a little more and save creating a dozen new sku's |
| 21:30.52 | [TK]D-Fender | Speaking of which, it looks like another few weeks before my friggen IP 501 arrives... |
| 21:31.18 | VoIPMasta | .brb coffee time |
| 21:35.19 | *** join/#asterisk pagec (n=cpage@64-252-98-136.adsl.snet.net) |
| 21:37.19 | *** join/#asterisk gbodemantv (n=gbodeman@216.142.38.154) |
| 21:39.28 | mikey2600 | anyone have any idea what the syntax in extensions*.conf would look like for an h323 gateway |
| 21:40.39 | Netgeeks | Dial(OH323/<gateway ip>/<argument>) |
| 21:40.40 | *** join/#asterisk TedC (n=ted@gray.impulse.net) |
| 21:41.01 | Netgeeks | or switch OH323 for H323 depending on the app you will be using |
| 21:41.52 | *** part/#asterisk skyboy (n=skyboy@72.18.13.34) |
| 21:42.54 | mikey2600 | omg i hate h323 :P it has been nothing but a pain in the butt to get working. plus all AMP doesnt know wtf it is in config file. |
| 21:51.37 | websae | mikey2600: have you ever gotten it to work yet? |
| 21:51.41 | websae | the h323 |
| 21:53.02 | justinu|laptop | anyone know the name of the flight tracking website Nugget is working on? |
| 21:53.08 | file[laptop] | flightaware |
| 21:53.34 | *** join/#asterisk sigwerk (n=sigwerk@cyclone.sigterm.net) |
| 21:53.37 | justinu|laptop | ty |
| 21:53.58 | *** part/#asterisk sigwerk (n=sigwerk@cyclone.sigterm.net) |
| 21:54.13 | *** join/#asterisk surfdue (n=tyler@unaffiliated/surfdue) |
| 21:54.19 | *** part/#asterisk surfdue (n=tyler@unaffiliated/surfdue) |
| 21:55.56 | docelmo | oi! |
| 21:56.17 | kamileon | hello, i have a problem where if someone calls into my pbx through my iax they cant hear me, if they call in via sip they can, and can hear me if i initiate on either, any suggestions? |
| 21:59.51 | *** join/#asterisk stkn (n=stkn@gentoo/developer/pdpc.active.stkn) |
| 21:59.55 | rollergrrl | Is it possible to run different hold music for each queue? |
| 22:00.22 | Nugget | of course. |
| 22:00.48 | rollergrrl | ok |
| 22:02.04 | rollergrrl | SetMusicOnHold(myclass) right? |
| 22:02.25 | [TK]D-Fender | rollergrrl : its also in the queu definition itself... |
| 22:02.36 | rollergrrl | cool thanks |
| 22:05.37 | *** join/#asterisk dougster (n=doug@bil.oneeighty.com) |
| 22:05.58 | Cybertoy | anyone with a 7970 phone on the east coast that have daylight savings time on the phone now? |
| 22:06.29 | dougster | Oooo asterisk talk. *poos pants* |
| 22:07.00 | timscott | dude, of course asterisk talk. |
| 22:07.02 | timscott | it's #asterisk |
| 22:07.16 | Cybertoy | well .. I have the 7970 on asterisk ... ;) |
| 22:07.28 | Cybertoy | but yeah .. a bit off topic |
| 22:07.28 | dougster | timescott: eez my first time here. :) the glary eyed amazement will disappear soon enough |
| 22:08.18 | gbodemantv | hi all |
| 22:08.56 | gbodemantv | who knows macro's |
| 22:09.38 | gbodemantv | I need a meemte macro |
| 22:10.00 | dougster | meemte? is that like meetme? |
| 22:10.22 | gbodemantv | yeah |
| 22:10.24 | gbodemantv | sorry |
| 22:14.28 | gbodemantv | trying to make a meetme admin authenticate against vm password |
| 22:14.43 | *** join/#asterisk justinu|laptop (n=Justin@wirelessdata-031-031.mycingular.net) |
| 22:15.28 | *** join/#asterisk pagec (n=cpage@64-252-98-136.adsl.snet.net) |
| 22:16.27 | Hmmhesays | nothing like albuterol to give you that nice high feeling |
| 22:17.57 | Hmmhesays | god thats good shit |
| 22:18.02 | Hmmhesays | now I can have a cigarette |
| 22:18.02 | *** join/#asterisk naturalblue (n=Administ@87.192.100.109) |
| 22:18.07 | theorem_ | .. |
| 22:18.17 | Hmmhesays | yeah it a grey goose on the rocks night for me |
| 22:18.28 | theorem_ | Hmmhesays - yeah |
| 22:18.37 | theorem_ | oddly, that's exactly what I poured |
| 22:18.51 | Hmmhesays | only premium vodka should be drank on the rocks |
| 22:18.59 | theorem_ | drunk ? |
| 22:19.09 | theorem_ | yeah |
| 22:19.13 | theorem_ | also .. |
| 22:19.22 | theorem_ | a good combo is vanilla vodka + coke |
| 22:19.28 | theorem_ | kinda girly, but nice. |
| 22:20.03 | theorem_ | THreee Olives Vodka, Cherry or their Vanilla with the coke is excellent. |
| 22:27.49 | LostFrog | file: you here? |
| 22:29.53 | *** join/#asterisk NewSole (n=dave@d226-108-46.home.cgocable.net) |
| 22:30.14 | NewSole | Question Any one hear of VegaStream |
| 22:38.08 | *** join/#asterisk skyboy (n=skyboy@72.18.13.34) |
| 22:40.15 | NewSole | anyone alive |
| 22:41.06 | Hmmhesays | locomotive by guns n roses is fucking impossible to play |
| 22:45.58 | *** join/#asterisk Eggplants (i=No@dsl-731.cascadeaccess.com) |
| 22:47.39 | tzanger | I haven't tried |
| 22:47.45 | gbodemantv | hello all |
| 22:48.06 | gbodemantv | I am implementing the "marked user" function in Meetme |
| 22:48.17 | gbodemantv | but I want to limit who can call as the marked user |
| 22:48.33 | gbodemantv | I am already capturing VM password in Mysql |
| 22:48.39 | *** part/#asterisk naturalblue (n=Administ@87.192.100.109) |
| 22:48.47 | gbodemantv | Can anyone help wiyth the macro |
| 22:49.13 | skyboy | A quick question about codecs.. I saw an interesting report on xorcom's site with regards to load testing. Specically each codec was tested and shows how it was cpu bound. Has anybody done testing to see what bound I/O has? |
| 22:49.41 | skyboy | http://www.xorcom.com/ts-1/test-results.html |
| 22:50.39 | gbodemantv | http://pastebin.ca/49385 |
| 22:50.49 | gbodemantv | is the current macro I am using |
| 22:50.58 | gbodemantv | need to add 2 more lines |
| 22:51.13 | gbodemantv | one to check if the user has voicemail |
| 22:51.23 | gbodemantv | if not it hangs up |
| 22:51.28 | *** join/#asterisk ldnblk (n=Just@212.183.128.185) |
| 22:51.29 | gbodemantv | if so it goes to the next line |
| 22:51.45 | gbodemantv | which has them enter their vm password |
| 22:51.59 | gbodemantv | any idea how I do a DB lookup for these items? |
| 22:54.02 | *** join/#asterisk darkskiez (n=darkskie@bb-87-81-62-203.ukonline.co.uk) |
| 23:26.58 | *** join/#asterisk needasteriskcon (n=johnb@ip24-251-151-16.ph.ph.cox.net) |
| 23:27.14 | needasteriskcon | Anybody here able to help me with a ring group a@h problem? Willing to pay for help... |
| 23:27.47 | *** join/#asterisk justinu|laptop (n=Justin@wirelessdata-031-030.mycingular.net) |
| 23:27.59 | [TK]D-Fender | please read the channel topic... |
| 23:28.06 | needasteriskcon | got it... sorry... |
| 23:33.38 | *** join/#asterisk shido6 (n=shido6@d38-45-81.commercial1.cgocable.net) |
| 23:37.20 | Hmmhesays | hey [TK]D-Fender |
| 23:38.04 | [TK]D-Fender | y0 |
| 23:38.10 | Hmmhesays | whats up? |
| 23:38.22 | [TK]D-Fender | just grabbing a bit before I'm out to play some pool. |
| 23:38.26 | [TK]D-Fender | bite* |
| 23:38.36 | Hmmhesays | yeah i'm playing some guitar before going to play some pool |
| 23:38.49 | FuriousGeorge | join #api |
| 23:38.51 | FuriousGeorge | oops |
| 23:38.53 | Hmmhesays | you ever checked out the rhythm section to insterstate lovesong? |
| 23:39.04 | [TK]D-Fender | Don't know the piece, who's it by? |
| 23:39.10 | Hmmhesays | stone temple pilots |
| 23:39.21 | Hmmhesays | its so.... odd |
| 23:39.25 | [TK]D-Fender | Definately don't.... |
| 23:39.40 | Hmmhesays | you want it? |
| 23:39.46 | [TK]D-Fender | sure, why not. |
| 23:39.51 | Hmmhesays | dcc? |
| 23:40.31 | [TK]D-Fender | sure |
| 23:40.49 | [av]bani | needasteriskcon: whats cheep? |
| 23:40.51 | [av]bani | $5000? |
| 23:41.05 | Hmmhesays | has some crazy chords in it |
| 23:42.50 | [TK]D-Fender | ok, I've head it before... |
| 23:43.51 | Hmmhesays | i've been working on more solid rhythm playing lately, listening to stuff like this |
| 23:44.05 | DoktorGreg | well here i gp |
| 23:44.17 | DoktorGreg | in 20 minutes this location will be closed for business |
| 23:44.32 | DoktorGreg | and i start installing away |
| 23:44.52 | [TK]D-Fender | Hmmhesays : I'm picking it up now.... |
| 23:45.22 | Hmmhesays | the chord progression is basic |
| 23:45.31 | Hmmhesays | if you play all just major/minor chords |
| 23:46.09 | [TK]D-Fender | yup, in e-major. Thats where 90% of my material is in :) |
| 23:46.45 | *** join/#asterisk incontwin (n=FreePBX5@phone.linuxsys.com) |
| 23:48.17 | Hmmhesays | it is a funky chord progression too, but it works |
| 23:48.23 | *** join/#asterisk incontwin (n=FreePBX3@phone.linuxsys.com) |
| 23:48.49 | [TK]D-Fender | Hmmhesays : Not sure how "funky". Pretty basic, I think I just go enough to faske it in the first 54 mins :) |
| 23:48.53 | [TK]D-Fender | *5 |
| 23:49.26 | Hmmhesays | yeah, not very... sorry i've been playing a lot of ac/dc greenday, shit like that to get pick hand more developed |
| 23:49.45 | [TK]D-Fender | Try : "Inside out" by Eve6 |
| 23:49.53 | Hmmhesays | yeah i've known that one for years |
| 23:50.23 | [TK]D-Fender | great one to learn tempo from and is very interesting in their use of off-time lyrically. |
| 23:50.39 | Hmmhesays | yes indeed |
| 23:50.41 | Hmmhesays | and its fun to play |
| 23:51.25 | [TK]D-Fender | I did send you "I'm Alright" about 2 weeks ago right? |
| 23:51.41 | Hmmhesays | hmm its not on this pc |
| 23:51.52 | twisla | The more time you spend with *, to more you love it. |
| 23:51.55 | [TK]D-Fender | ok, there's one to pick up ;) |
| 23:52.01 | twisla | Just my tought of this night |
| 23:52.09 | Hmmhesays | twisla: wtf planet are you on |
| 23:52.10 | twisla | s/to/the/ |
| 23:52.34 | twisla | Hmmhesays: mmh, planet belgium. |
| 23:52.42 | twisla | where the beer is fine. |
| 23:52.50 | [TK]D-Fender | WAFFLES! |
| 23:55.00 | *** part/#asterisk Foxtro (i=foxtro@251-79-246-201.adsl.terra.cl) |
| 23:55.05 | [TK]D-Fender | ok, I'm out. Might be back later. |