| 00:00.28 | rhombus | ManxPower: Stuff that works tends to be expensive. |
| 00:01.04 | rhombus | ManxPower: Have you tried the Digium HPEC? |
| 00:01.25 | *** join/#asterisk zeedo (n=zeedo@reboot-robot.net) |
| 00:01.46 | rhombus | For that matter, as anybody tried the Digium HPEC for echo cancellation? |
| 00:03.00 | flenders | ManxPower: do you work at digium? |
| 00:07.20 | flenders | damn it, just tried digium's sales dept and no one is available |
| 00:07.24 | bkruse_home | rhombus: yes i have |
| 00:07.29 | bkruse_home | flenders: its after hours fool |
| 00:07.34 | *** join/#asterisk fbffff (n=fbffff@r74-193-79-81.pfvlcmta01.grtntx.tl.dh.suddenlink.net) |
| 00:08.08 | flenders | what's the time there? |
| 00:08.58 | rhombus | bkruse_home: and, how were the results? |
| 00:08.58 | Qwell | after 7 |
| 00:08.58 | interworx | s/7/19/ |
| 00:11.27 | DrukenLPY | anyone know about payphones in here? |
| 00:12.15 | *** join/#asterisk mercestes (n=merceste@inet.hou.devry.net) |
| 00:16.26 | *** join/#asterisk Gunnar (n=gunnar@nat.sigmasoft.com) |
| 00:16.44 | rhombus | bkruse_home: what has your experience with the Digium HPEC been so far? |
| 00:19.48 | *** join/#asterisk netlouis (n=netlouis@a213-22-64-193.cpe.netcabo.pt) |
| 00:20.41 | bkruse_home | great |
| 00:20.43 | bkruse_home | sounds awesome |
| 00:22.11 | JT | flenders: ping |
| 00:23.22 | flenders | JT: pong |
| 00:24.47 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 00:25.44 | rhombus | HELLO |
| 00:25.55 | flenders | hi |
| 00:26.00 | rhombus | oh, hi. |
| 00:26.02 | rhombus | :D |
| 00:26.07 | mercestes | rhombus: not a good first impression. :P |
| 00:26.30 | rhombus | mercestes: then you missed me last time because I wasn't loud enough :) |
| 00:26.39 | rhombus | (I've seen you around before) |
| 00:26.43 | mercestes | rhombus: LMAO I caught you then too. |
| 00:27.11 | rhombus | where did bkruse_home go? |
| 00:27.32 | flenders | home, I guess |
| 00:27.35 | flenders | :D |
| 00:27.48 | rhombus | mmm. perogies. |
| 00:27.49 | JT | flenders: going to cebit? |
| 00:28.00 | flenders | JT: dunno, last 2 years it was shit |
| 00:28.06 | flenders | JT: you? |
| 00:28.11 | JT | heh yeah |
| 00:28.30 | JT | there's 2 days left to register for free... supposedly it's not bs this year |
| 00:28.42 | flenders | I'll register then! |
| 00:29.25 | JT | the voip conference is a bit of a hike |
| 00:29.28 | JT | $445 + gst |
| 00:30.19 | flenders | wow |
| 00:30.26 | flenders | you're going to that one? |
| 00:30.42 | JT | no |
| 00:30.48 | JT | free exhibition for me |
| 00:30.58 | JT | i'm sure it'd be interesting to hear mark spencer to talk |
| 00:31.19 | flenders | no doubt about it |
| 00:31.21 | JT | but i'm not spending 500bucks on a sales pitch from half a dozen speakers :) |
| 00:32.41 | tzanger | [TK]D-Fender: polycom question for you -- any way to have the phones handle daylight savings time? Right now I have the gmt-offset in dhcp but that's a real pain in the ass |
| 00:33.29 | mercestes | tzafrir: 2.1.0 fixes the new DST crap. and polycom does handle DST with the NTP server. Just set the offset to whatever it is in seconds. |
| 00:34.03 | mercestes | tzafrir: And set "override dhcp" = 1 in the multiple places it shows up, for NTP, and DST. |
| 00:34.03 | tzanger | mercestes: but the DST offset changes by 3600 seconds for daylight savings... |
| 00:34.04 | tzanger | mercestes: ahhhhhh |
| 00:34.06 | tzanger | that's it |
| 00:34.09 | mercestes | :) |
| 00:34.47 | mishehu | hmm hmm looks like there's a conversation going on about polly coms *sqwak* and DST... what's the fix? can't use sip 1.6.7 I suppose? |
| 00:34.57 | mercestes | Sip 2.1.0 fixes it |
| 00:35.20 | mishehu | mercestes: is 2.1.0 too large for the ip500's? I still have some of those in use. |
| 00:35.25 | flenders | JT: oh, just remembered, I'll be in brazil in may |
| 00:35.32 | mercestes | Not that I've heard of. |
| 00:35.35 | tzanger | mercestes: so you set the offset or the time zone in the polycom config |
| 00:35.39 | mercestes | but I am not educated on IP500's |
| 00:35.46 | JT | flenders: the whole of may? |
| 00:35.54 | mcab | fix the DST settings, then you don't have to worry about it again |
| 00:35.56 | mercestes | tzazanger: you set the offset and the default DST settings work. |
| 00:36.00 | flenders | JT: until the 23rd |
| 00:36.12 | JT | flenders: i guess that's a problem then |
| 00:36.50 | creature1 | if i set a language in sip.conf asterisk searches for that file in lang/ folder.. right? and if it's a digit then in lang/digit folder? my problem is that the files in lang dir are played correctly but for digits, letters etc asterisk chooses the default lang en.. |
| 00:37.10 | flenders | JT: :D |
| 00:37.35 | flenders | JT: haven't gone home for over a year |
| 00:39.14 | JT | flenders: you're away right now? |
| 00:39.44 | flenders | JT: away in .au |
| 00:39.49 | flenders | :o) |
| 00:39.54 | JT | you're from sydney aren't you? |
| 00:39.57 | flenders | been over here for 5 years |
| 00:40.03 | *** join/#asterisk droops (n=droops@adsl-065-005-212-128.sip.jan.bellsouth.net) |
| 00:40.08 | flenders | nope, I'm a brazo |
| 00:40.14 | JT | ah ok |
| 00:40.19 | JT | but you're in sydney now? |
| 00:40.20 | flenders | I live in syd |
| 00:40.27 | creature1 | hmm seems like asterisk are looking for the files in lang/digits/lang folder... |
| 00:40.29 | flenders | yeah, work in north sydney |
| 00:40.33 | creature1 | if its a digit.. |
| 00:40.39 | JT | cool |
| 00:40.45 | JT | we should catch up some time |
| 00:40.54 | sivana | anyone know what chan_phone is for? |
| 00:40.54 | flenders | JT: definetely mate |
| 00:41.45 | JT | sivana: something very old iirc |
| 00:41.47 | DrukenLPY | hey sivana |
| 00:44.47 | flenders | JT: you're next to the greenwood pl? |
| 00:45.07 | JT | not quite, but it's not that far away i guess |
| 00:46.09 | flenders | I'm up on pacific hwy, probably a 15 minute walk from there |
| 00:48.24 | mishehu | blargh, freedomphones doesn't seem to have polly com sip 2.1.x on it. |
| 00:50.36 | sivana | hey |
| 00:52.14 | JT | flenders: sounds like closer to crows nest? |
| 00:52.28 | boch | can i use file:// in CURL() func ? |
| 00:52.47 | flenders | yeah, it is a short walk to crows nest |
| 00:52.57 | JT | ah ok |
| 00:53.53 | *** join/#asterisk topping (n=topping@ppp-68-122-72-235.dsl.pltn13.pacbell.net) |
| 00:54.53 | *** join/#asterisk sudhir492 (n=sudhir@c-71-63-59-45.hsd1.va.comcast.net) |
| 00:55.48 | sudhir492 | Is anyone here using Cisco phones, 7940 or 7960 in particular? |
| 00:56.16 | Qwell | yeah, with chan_skinny |
| 00:56.45 | sudhir492 | which version of firmware? |
| 00:57.24 | Qwell | dunno, 7.x |
| 00:59.11 | *** join/#asterisk mog (i=ejabberd@71.207.215.93) |
| 00:59.11 | *** mode/#asterisk [+o mog] by ChanServ |
| 00:59.31 | *** join/#asterisk RoyK (n=roy@217-175-152.100710.adsl.tele2.no) |
| 00:59.59 | sudhir492 | Qwell: Are you registered with Cisco? |
| 01:00.04 | Qwell | nope |
| 01:00.29 | sudhir492 | ok |
| 01:00.54 | sudhir492 | I need an example SIPdefault.cnf file |
| 01:01.11 | Qwell | there are some on the wiki |
| 01:01.12 | Qwell | ~wikis |
| 01:01.23 | jbot | from memory, wikis is http://www.voip-info.org |
| 01:02.01 | sudhir492 | I tried finding one last night. Going to look more diligently again. |
| 01:04.18 | *** join/#asterisk sav_mcfly (n=R00T@pergamo.zonaz.net) |
| 01:04.53 | *** part/#asterisk sav_mcfly (n=R00T@pergamo.zonaz.net) |
| 01:09.59 | *** join/#asterisk RoyK (n=roy@217-175-152.100710.adsl.tele2.no) |
| 01:14.58 | *** join/#asterisk juanjoc (n=juanjoc@248-32-235-201.fibertel.com.ar) |
| 01:18.15 | *** join/#asterisk josef_k (n=josef_k@201009235147.user.veloxzone.com.br) |
| 01:21.55 | sudhir492 | very quiet her today |
| 01:22.31 | mercestes | shh. I'm trying to sleep |
| 01:23.44 | xheliox | We noticed. The rest of us can't get a moments peace with your snoring. |
| 01:23.51 | xheliox | You really should have that looked that. |
| 01:23.56 | mercestes | >.< |
| 01:24.27 | mercestes | Between my narcolepsi and my insomnia and my night terrors I figured I am psychologically screwed in the sleep department. Atleast if I can hear myself snoring I know I'm breathing. |
| 01:26.26 | *** join/#asterisk Kumbang (n=kumbang@router-wmi.paume.ITB.ac.id) |
| 01:27.29 | mercestes | yay |
| 01:28.38 | *** join/#asterisk jcool (n=zoro@124.106.206.247) |
| 01:29.17 | jcool | good day guys, has anyone notice that if you enable jitterbuffer on zap you will see a lot of this on your console "Recieved frame with invalid timing info" |
| 01:29.26 | josef_k | can you help me with a philosophical question? Why asterisk 1.4.0 sends the reinvite AFTER all the session? It is obvious in 1.2.x, but I dont understand why it needs to send a invite before the bye message |
| 01:30.38 | mercestes | josef_k: Philosphically? Probably just to make sure the device is still there and listening before sending the "bye." |
| 01:30.52 | creature1 | josef_k: when you give instructions to a person you usually give him all of the instructions before saying goodbye. i have _no idea_ if thats the case here but well, seems logical :D |
| 01:31.26 | flenders | why is I have a good one: why is asterisk-1.4.0 tarball 11MB in size and 1.4.1 is 17MB? |
| 01:31.30 | mercestes | jcool: It's probably letting you know it recieved a stale nonce packet. |
| 01:32.10 | mercestes | flenders: Because 1.4.1 has 6MB more of compressed fixes, feature enhancements, and easter eggs...mostly easter eggs. |
| 01:32.57 | flenders | never seen such an increase in size from one release to another |
| 01:32.57 | josef_k | mercestes: I dont know if it is the answer. Maybe... it sends the invite changing the C header, asking the rtp stream. After that, it shut down. |
| 01:33.10 | josef_k | but you should be right |
| 01:34.28 | mercestes | josef_k: Well, at one point * waited for an ack to a "bye" before closing a stream, and if you unplugged a phone you had a channel that never went away. |
| 01:34.28 | jcool | mercestes: well, since i don't know much on how jitterbuffer will work, i think it |
| 01:34.28 | jcool | safe to ignore it then remove warning on the console |
| 01:34.28 | mercestes | josef_k: Maybe sending a "invite" before sending a "bye" makes certain that the device will respond before sending the bye, reestablishing hte channel if it closed before or giving it a chance to error out and handle it differently. |
| 01:34.51 | mercestes | jcool: I think it is acknowledging that it did recieved a jittered packet for informational reasons. Try googling hte exact text of the error, - any context specific #'s. |
| 01:35.22 | mercestes | jcool: jitterbuffer just creates a queue of so many seconds that it uses to collect and reorganize the data before it begins playback. |
| 01:35.43 | mercestes | flenders: Then you will be rewarded as you never have before. |
| 01:35.48 | josef_k | yes... |
| 01:35.59 | josef_k | cool, cool. :) Thanks, mercestes |
| 01:36.18 | mercestes | josef_k: I'm not a developer tho. They know better than I do. That's my educated guess tho. |
| 01:36.46 | josef_k | i agree with you. I think it is the answer |
| 01:36.50 | jcool | mercestes: thanks man, to be honest i don't see any meaning full input from google except from you !!" |
| 01:37.04 | *** join/#asterisk visba (n=dca[lapt@c-24-8-53-17.hsd1.co.comcast.net) |
| 01:37.06 | jcool | Recieved frame with invalid timing info <-- this is the exact error |
| 01:37.12 | jcool | oopss warning rather |
| 01:37.19 | mercestes | jcool: Yea, then it probably means it's info only. Google is pretty verbose about "errors." |
| 01:37.29 | mercestes | yea, warnings are more..."FYI" than "warning." |
| 01:37.37 | mercestes | it should just say "hey, you:" then the same text. |
| 01:38.10 | *** join/#asterisk anthony] (n=anthony@175.21.188.72.cfl.res.rr.com) |
| 01:46.19 | *** join/#asterisk riddlebox (n=james@75-132-205-166.dhcp.stls.mo.charter.com) |
| 01:47.01 | JacksLivr | WHOOOOOOOOOOOO HOOOOOOOOOOOOO!!!!!!!! 7910 working. the upgrade to 1.4 worked!!! Qwell is the MAN! |
| 01:47.30 | JacksLivr | all hale teh Qwell! |
| 01:47.46 | *** join/#asterisk shmaltz (n=mybox@mail.dmaven.com) |
| 01:48.56 | shmaltz | OT:what database uses *.xd and *.nx file types? |
| 01:49.10 | sivana | heh |
| 01:49.20 | mercestes | shmaltz: google it. |
| 01:49.34 | shmaltz | mercestes, tried, but not much luck |
| 01:49.38 | shmaltz | thanks though |
| 01:50.07 | mercestes | shmaltz: Do a google search for "file extensions" There are some online databases dedicated to the subject. |
| 01:50.44 | rhombus | I wonder if ManxPower would ever let me visit him and give a guided tour of his setup. |
| 01:51.11 | rhombus | I am betting it is pretty sweet. I could learn a lot. |
| 01:51.45 | mercestes | Or we would stare in awe and understand nothing. Manx's stuff is pretty 133t. |
| 01:52.50 | mercestes | I didn't like him too much until I saw him slip out some of his stuff and man.....he's way beyond anything I'd even suggest could be done. He's put alot of work into it. |
| 01:53.16 | mercestes | So, looking at his stuff v/s what's "readily available" I can see why he's "psh, I'm not just going to spoon feed you the answer." becuase he definately worked for his answer. |
| 01:55.13 | bulle | mercestes: well, everyone needs spoon feeding in the beginning |
| 01:55.26 | mercestes | bulle: That's what the wiki is for. |
| 01:55.37 | mercestes | and "the book" |
| 01:55.42 | sivana | too bad the wiki is outdated |
| 01:55.51 | bulle | mercestes: sadly the wiki isnt that great, its pretty confusing from time to time |
| 01:55.53 | mercestes | sivana: So fix it. It's a wiki |
| 01:56.01 | bulle | mercestes: "the book" is good though |
| 01:56.09 | sivana | yea, I'll do that in my other spare time |
| 01:56.23 | mercestes | bulle: If you read the book, and reference the wiki, and demonstrate you did research, and are still confused, most of us are more than happy to spoon feed you as long as you demonstrate that you tried beforehand. |
| 01:57.03 | bulle | mercestes: no worries, i have gotten nice support here for all i have asked, except the one question about why macros are called macros, when they behave more like functions =D |
| 01:57.34 | mercestes | bulle: Because whoever named it was more a Microsoft person than a linux person. |
| 01:58.06 | SwK | incase anyone is still fighting with polycom time issues.... |
| 01:58.06 | SwK | http://knowledgebase.polycom.com/kb/search.do?cmd=displayKC&docType=kc&externalId=10627&sliceId=SAL_PUBLIC_1_2&dialogID=1890871&stateId=1%200%201886835 |
| 01:59.06 | mishehu | only thing I must say about that config is that it needs to be updated every year since the start date is set as a date and not as a week and day of week. |
| 01:59.08 | mercestes | nice, thanks SwK. |
| 01:59.22 | mercestes | too bad the link is too long to /topic. =/ |
| 01:59.41 | mishehu | tinyurl anybody? |
| 01:59.44 | mercestes | It's "fixed" in Sip 2.1.0 by default as I understand it. |
| 01:59.50 | *** join/#asterisk intralanman (n=lanman@pool-71-253-253-149.nrflva.east.verizon.net) |
| 01:59.54 | *** join/#asterisk TedNJ37 (n=HungLad@ool-4573adc7.dyn.optonline.net) |
| 02:00.26 | mercestes | I'd rather just post the Sip 2.1.0 and bootroms. |
| 02:01.35 | *** join/#asterisk Uberbot (n=Uberbot@c-76-18-87-61.hsd1.nm.comcast.net) |
| 02:01.44 | Uberbot | Hi all. |
| 02:02.12 | mercestes | Greetings |
| 02:02.13 | SwK | mercestes its not really a "fix" in the way of a programming change its a default configs parameter change |
| 02:02.29 | mercestes | SwK: indeed. I agree. |
| 02:02.56 | Uberbot | I'm trying to get an IVR working and I'm having difficulty getting * to recognize keypresses... |
| 02:03.00 | SwK | these settings are already in ipmid.cfg or sip.cfg depending on what style of configs you are using (i still use the old style as I think the new style is just to much crap to deal with in 1 file) |
| 02:03.00 | TedNJ37 | I have a problem. I have installed hudlite-server but freepbx is not interfacing with it, it shows a blank page with a language selecting box. There are no more options in that page. And when I put the mouse over the option of Asterisk, I don't see it listed in the drop down list. But if I click on Asterisk, I see HudLite Admin listed in the page. When I select it, it shows a blank page. I have installed using the package manager o |
| 02:03.58 | Uberbot | This should work, right? exten => _X, 1, noop(The caller pressed the ${EXTEN} key) |
| 02:04.19 | mercestes | Uberbot: That matches 1 character. Call an Answer() first. |
| 02:04.26 | mercestes | and read the book |
| 02:04.28 | mercestes | ~thebook |
| 02:04.39 | jbot | somebody said thebook was a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 02:04.39 | mercestes | ~book |
| 02:04.42 | jbot | extra, extra, read all about it, book is a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 02:04.43 | Uberbot | I've read it. |
| 02:04.52 | mercestes | Ok, well, IVR is covered in there with good examples. |
| 02:05.03 | Uberbot | The problem is that nothing is being matched at all and not being logged to the console. |
| 02:05.34 | JT | don't put spaces between arguments in exten => |
| 02:05.41 | Uberbot | I'm playing a message with background(). |
| 02:05.41 | JT | that's not the right way to build an ivr |
| 02:05.50 | JT | check the wiki ivr pages |
| 02:05.55 | Uberbot | exten => _X,1, noop(The caller pressed the ${EXTEN} key) |
| 02:06.03 | Uberbot | With no spaces... I'll try that. |
| 02:06.06 | JT | _X,1,NoOp |
| 02:06.20 | JT | you're still doing it the wrong way |
| 02:06.27 | JT | you don't usually pattern match in an ivr |
| 02:06.38 | JT | you need to start at extension s |
| 02:06.49 | Uberbot | I'm using the extension as a lookup into a database. the db tells me what to do next. |
| 02:07.41 | mercestes | Uberbot: You have to call an Answer() or Asterisk won't "hear" your key presses at all. |
| 02:08.23 | Uberbot | I call answer, then do the background(). |
| 02:08.38 | mercestes | Uberbot: then set your timeout values. |
| 02:09.31 | Uberbot | Like this: WaitExten(seconds) |
| 02:10.03 | mercestes | According to teh book and the wiki as it pertains to your version. |
| 02:10.18 | mercestes | Mainly "digittimeout" or Timeout(digit) |
| 02:10.37 | *** join/#asterisk jellyfishnetwork (n=admin@67.159.5.246) |
| 02:10.41 | jellyfishnetwork | hello |
| 02:11.29 | mercestes | hi |
| 02:11.54 | mercestes | and what JT says, he's giving good advice. |
| 02:11.56 | jellyfishnetwork | I am new to asterisk but from what I have seen, its VERY nice |
| 02:11.59 | mercestes | tho you don't *have* to start at s. |
| 02:12.07 | mercestes | and nooping on _X is.....not a good test. |
| 02:12.12 | mercestes | use 1,2,3,4,5 etc. |
| 02:12.33 | TedNJ37 | Can someone help me please? |
| 02:12.34 | TedNJ37 | I have a problem. I have installed hudlite-server but freepbx is not interfacing with it, it shows a blank page with a language selecting box. There are no more options in that page. And when I put the mouse over the option of Asterisk, I don't see it listed in the drop down list. But if I click on Asterisk, I see HudLite Admin listed in the page. When I select it, it shows a blank page. I have installed using the package manager o |
| 02:12.46 | sivana | or 1,n,n.... |
| 02:12.57 | jellyfishnetwork | that being said.. I cannot get it working. I can get my softphone to connect but when I try to call I get the "call not approved" message |
| 02:12.58 | JT | TedNJ37: i swear i saw that exact same question a few minutes ago |
| 02:14.40 | mercestes | ~freepbx |
| 02:14.52 | jbot | somebody said freepbx was unable to be supported here. It's a complex piece of dialplan and GUI, and can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking there |
| 02:14.55 | jellyfishnetwork | so far, I have edited sip.conf and extensions.conf |
| 02:15.18 | mercestes | jellyfishnetwork: Sounds like a config error. |
| 02:15.21 | TedNJ37 | But nobody told me if they have encountered that problem. |
| 02:15.32 | *** join/#asterisk eltech (i=G00Ds@ool-457c93b6.dyn.optonline.net) |
| 02:15.38 | Uberbot | exten => robo, 3, Set(TIMEOUT(digit)=20) fixed it. Thank you. |
| 02:15.47 | JT | TedNJ37: maybe no-one knows. it's rude to repeat. especially 7 lines |
| 02:16.09 | jellyfishnetwork | mercestes: yeah that is pretty much what I figured.. unfortunately I have no idea how to troubleshoot it |
| 02:16.38 | mercestes | jellyfishnetwork: Sounds like your call is not registering. focus on sip.conf examples and putting the apprpriate fields in the softphone to match sip.conf |
| 02:16.50 | TedNJ37 | Sorry. |
| 02:16.51 | mercestes | jellyfishnetwork: just make your username/authname all the same. |
| 02:17.13 | mercestes | Uberbot: NP. Go well asterisk warrior |
| 02:17.47 | JT | Uberbot: you should drop the spaces, they're non-standard :) |
| 02:18.48 | Uberbot | Noted. |
| 02:18.58 | jellyfishnetwork | mercestes do I have to have register syntax for my softphone? I thought it was for registering with my provider |
| 02:19.38 | mercestes | jellyfishnetwork: No.... Just a [username] and the stuff there. adn then the appropriate fields in your settings for the softphone |
| 02:20.02 | mercestes | qualify=yes , etc. |
| 02:20.17 | mercestes | nothing for "register => " ignore that stuff. That's for a remote pbx. |
| 02:20.56 | *** join/#asterisk orlock (i=jwr@202.44.174.4.static.nexnet.net.au) |
| 02:21.12 | orlock | Has anybody here used reinvite? |
| 02:21.21 | mercestes | Yes. |
| 02:21.21 | *** part/#asterisk rhombus (n=sfbosch@dsl-cap-66-18-218-36-cgy.nucleus.com) |
| 02:21.41 | jellyfishnetwork | I didn't have qualify=yes |
| 02:21.49 | orlock | mercestes: was that a yes to me? |
| 02:22.48 | *** join/#asterisk ltdwk (n=z@203-173-10-9.perm.iinet.net.au) |
| 02:24.21 | jellyfishnetwork | mercestes my softphone was dialing 1 first.. Now I am getting 404 not found |
| 02:25.02 | mercestes | jellyfishnetwork: YOu still have a register statement. |
| 02:25.11 | mercestes | jellyfishnetwork: Or your PBX is defined as a hostname and not an IP address. |
| 02:25.19 | mercestes | jellyfishnetwork: *or* your softphone cannot reach your pbx. |
| 02:25.29 | mercestes | orlock: Yes. and Yes. |
| 02:26.31 | mercestes | jellyfishnetwork: You can use hostnames, just make certain the dns resolves from the softphone perspective. |
| 02:27.03 | *** join/#asterisk sysreq (n=sysreq@modemcable171.134-81-70.mc.videotron.ca) |
| 02:27.05 | jellyfishnetwork | even if my softphone says logged in extension 2001? |
| 02:27.21 | mercestes | jellyfishnetwork: What does sip show peers show for your phone?? |
| 02:28.17 | jellyfishnetwork | 2001/2001 68.29.40.253 D 5060 OK (279 ms) |
| 02:28.29 | mercestes | what's cli output of an attempted call? |
| 02:28.38 | mercestes | pastebin it if it's more than 3 lines |
| 02:28.52 | jellyfishnetwork | i don't know? |
| 02:28.54 | JacksLivr | pwd |
| 02:29.00 | JacksLivr | whoops, srry |
| 02:29.09 | mercestes | jellyfishnetwork: What's your * version? |
| 02:29.55 | jellyfishnetwork | Asterisk 1.2.13 |
| 02:30.20 | orlock | mercestes: I think that the cause of some echo issues i am having is due to excessive latency between the Asterisk server and the handsets. enabling reinvite should allow the RTP stream to go directly between the handsets and the upstream sip provider, bypassing Asterisk, correct? |
| 02:30.26 | mercestes | jellyfishnetwork: set verbose 6 and then try to make a call, Pastebin what asterisk spits out when you try to make a call |
| 02:30.55 | mercestes | orlock: canreinvite=yes yes. but...Echo is a complex beast that can be caused by many things. |
| 02:31.24 | mercestes | orlock: otherwise, yes, you are correct, * will hand off if it is able. |
| 02:31.33 | orlock | cool. |
| 02:31.46 | orlock | mercestes: and if the SIP handsets are behind NAT? |
| 02:31.59 | jellyfishnetwork | Mar 13 02:31:21 NOTICE[56405]: pbx.c:1741 pbx_extension_helper: Cannot find extension context 'from-sip-internal' |
| 02:32.19 | mercestes | jellyfishnetwork: There is your problem. You do not have a context called "from-sip'internal." |
| 02:32.33 | mercestes | jellyfishnetwork: Write your outgoing dial rules under [from-sip-internal] and you should be good to go. |
| 02:32.56 | mercestes | orlock: It is quite likely that * will not be able to hand off the RTP to two handsets behind NAT unless those handsets are handling their own NAT translations. |
| 02:33.16 | mercestes | orlock: In that case, "reinvite" handoff will not happen. |
| 02:33.19 | orlock | hmm. |
| 02:34.49 | FuriousGeorge | how much latency is there between the phone and asterisk that you think that is the cause for the echo |
| 02:35.07 | jellyfishnetwork | does voip over evdo even work? |
| 02:35.22 | mercestes | jellyfishnetwork: what is evdo? |
| 02:35.26 | *** part/#asterisk mog (i=ejabberd@71.207.215.93) |
| 02:35.32 | creature1 | this is probably .. like one of the most useful things to use asterisk for ;) -> http://janson.naiv.us/asterisk/cnid_xbox.png |
| 02:35.43 | FuriousGeorge | in my experience, the cause of echo is almost always some analog leg of the conversation, like pots |
| 02:35.49 | jellyfishnetwork | sprint's wireless product.. the latency might be a little too high |
| 02:36.07 | mercestes | FuriousGeorge: I saw it happen under SIP with some freaky routing errors. Circular loop logic. |
| 02:37.15 | mercestes | jellyfishnetwork: Hrm. Those wireless cards are routers in which Sprint can do any level of natting and port blocking. |
| 02:37.32 | FuriousGeorge | mercestes: i guess anything is possible, but under normal circumstances with asterisk and sip, i think you should be ok. i use remote clients with two NATs between me and server and latency is higher but no echo |
| 02:37.39 | mercestes | jellyfishnetwork: They can VPN them tho and set them up as wireless vpn access points. |
| 02:37.43 | orlock | FuriousGeorge: this is 100% sip |
| 02:37.54 | mercestes | FuriousGeorge: Normally, SIP does not cause echo, irregardless of lag, correct. |
| 02:37.59 | orlock | goddamn |
| 02:38.39 | mercestes | orlock: Yea. Pretty much. |
| 02:38.45 | FuriousGeorge | mercestes: yeah, that's how i always pictured it. its when you add voip latency + pstn side-tone that you normally need to worry about cancelling echo |
| 02:39.20 | mercestes | phones can cause echo within themselves (via speaker phone) via SIP, but..that's about it. |
| 02:39.35 | FuriousGeorge | iow, you always echo on the pstn, but its so fast you dont hear it, but the added latency of voip causes the echo to "separate" from the communication (so to speak) |
| 02:39.40 | *** join/#asterisk stridernzl (n=neville@222-155-178-102.jetstream.xtra.co.nz) |
| 02:39.41 | FuriousGeorge | orlock: have you tried another device |
| 02:39.49 | mercestes | or some really freaky jitterbuffering. =/ |
| 02:40.00 | mercestes | I guess if you jitterbuffer on both the endpoint and the PBX it'd do the same thing. |
| 02:40.21 | jellyfishnetwork | mercestes http://iraq.jellyfishnetwork.com/extensions.conf not sure why it doesnt work |
| 02:41.01 | mercestes | jellyfishnetwork: write that file and do an extensions reload. |
| 02:41.10 | mercestes | jellyfishnetwork: it will ring the softphone but......that's it. Everythign else is broken. |
| 02:41.24 | mercestes | and it wil only match 2001 and 2000. |
| 02:41.39 | jellyfishnetwork | Mar 13 02:41:27 WARNING[56741]: config.c:499 process_text_line: parse error: No category context for line 1 of extensions.conf |
| 02:41.54 | FuriousGeorge | orlock: try x-lite rather than whatever that device is, just to make sure its not not some issue with the device causing it |
| 02:42.02 | mercestes | delete everything above [local-extensions] |
| 02:42.11 | mercestes | Ok, I gtg. l8s |
| 02:42.16 | orlock | FuriousGeorge: we have the same setup at 3 or 4 sites.. |
| 02:42.24 | orlock | hmm. the sites with the issues are sing internal DSL modems |
| 02:42.32 | orlock | but i am using one at home with no issues either |
| 02:42.49 | orlock | the only single difference they have is the latency due to physical distance |
| 02:42.59 | FuriousGeorge | orlock: whats the device? |
| 02:43.36 | jellyfishnetwork | ok now it works |
| 02:44.06 | jellyfishnetwork | now i just have to get the rest working |
| 02:44.20 | orlock | FuriousGeorge: Sipura 941's |
| 02:45.49 | jellyfishnetwork | thanks |
| 02:47.01 | orlock | grrr |
| 02:48.03 | *** join/#asterisk `mw (n=mike@graffiti.corp.ef.net) |
| 02:48.45 | `mw | anyone use iax2 with les.net? im having problems authenticating i think, all connections keep going to the default no-auth context |
| 02:50.10 | FuriousGeorge | <PROTECTED> |
| 02:50.19 | FuriousGeorge | orlock: my suggestion remains to swap the device just to be sure |
| 03:00.58 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 03:03.33 | jellyfishnetwork | http://iraq.jellyfishnetwork.com/sip.conf is this right to accept calls? |
| 03:03.38 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 03:04.07 | *** join/#asterisk nDuff (n=ccd@204.57.75.200) |
| 03:04.26 | *** join/#asterisk tessier_ (n=treed@kernel-panic/sex-machines) |
| 03:04.29 | tessier_ | Hello all! |
| 03:04.36 | nDuff | Does anyone have the source to NVBackgroundDetect v1.0.6? newmantelecom.com appears to be down, and backgrounddetect (unlike faxdetect) isn't in Google's cache. |
| 03:04.48 | tessier_ | Anyone got a Snom 320 phone? Does yours refuse to handle the time change also? |
| 03:05.20 | nDuff | tessier_: if you wait for a while (ie. until I've worked out my more immediate issues), I don't mind checking. |
| 03:05.42 | tessier_ | nDuff: Sure, thanks |
| 03:07.28 | nDuff | hrm... no copy of NVBackgroundDetect in Corel Cache either. |
| 03:07.34 | *** join/#asterisk CunningPike (n=CunningP@S010600095b33697f.vc.shawcable.net) |
| 03:08.04 | JT | what is nvbackgrounddetect? |
| 03:10.27 | nDuff | JT, an asterisk application providing background fax detection on non-zap channels. |
| 03:10.41 | *** join/#asterisk Braxus (n=braxus@66.147.214.164) |
| 03:11.02 | *** join/#asterisk |dennis| (n=dennis@200.32.233.82) |
| 03:11.07 | JT | hmm, why would you want to detect a fax in the background? |
| 03:11.21 | JT | like a fax starting up half way through a call? |
| 03:11.32 | nDuff | JT, so I can be playing a voice menu to non-fax users. |
| 03:11.48 | *** join/#asterisk florz (i=nobody@2001:1a50:503c:0:0:0:0:1) |
| 03:12.00 | JT | what does it do when it detects a fax? |
| 03:12.11 | nDuff | JT, http://www.voip.cc/wiki/view/NVBackgroundDetect |
| 03:13.13 | *** join/#asterisk SECGOD (i=SECGOD@71.57.36.106) |
| 03:20.17 | tessier_ | This is rather weak. Apparently the Snom phones do not have an actual timezone data file. |
| 03:20.18 | tessier_ | You hard code a GMT offset into them |
| 03:20.18 | tessier_ | So they will not change time on their own. |
| 03:20.18 | tessier_ | You have to manually go in and change your GMT offset. |
| 03:20.52 | *** part/#asterisk Uberbot (n=Uberbot@c-76-18-87-61.hsd1.nm.comcast.net) |
| 03:21.27 | *** join/#asterisk Mercestes (n=Merceste@cpe-24-175-82-3.houston.res.rr.com) |
| 03:21.44 | Mercestes | back for a bit |
| 03:23.36 | JT | heh, add that as another negative for the snom |
| 03:25.34 | ltdwk | I told you that when I was talking about them |
| 03:25.43 | ltdwk | the only thing I've found annoying about them is their DST adjustment |
| 03:26.31 | ltdwk | they do it, they just don't tell you how it works |
| 03:27.50 | ltdwk | there is a config item called "dst" but buggered if I know what the values mean in it |
| 03:28.34 | masked | does anyone have the nuance realspeak demo? it's 'temporarily unavailable' |
| 03:28.38 | ltdwk | dst!: 3600 10.01.07 02:00:00 03.05.07 03:00:00 |
| 03:28.42 | masked | on their website. |
| 03:29.09 | ltdwk | guessing some of those are to do with when to switch time |
| 03:29.34 | sivana | 03.05.07 month to spring ahead |
| 03:29.44 | tessier_ | JT: Actually we really like the snom phones |
| 03:29.45 | sivana | 10.01.07 month to fall back |
| 03:29.56 | tessier_ | JT: I have a 30 seat call center which use these phones 16 hours a day 7 days a week. |
| 03:30.03 | tessier_ | JT: Been using them for 9 months now. Work great. |
| 03:30.15 | ltdwk | tessier_: they don't like them here for some reason... I also really like them |
| 03:30.24 | tessier_ | JT: I really like how they are easily provisioned via http. |
| 03:30.38 | tessier_ | Best business class VOIP phone I have used so far. |
| 03:31.35 | Mercestes | ltwk: 3600 is the amount of adjustment: 10.01.07 is the stop date, 02:00:00 is the stop time to STOP daylight savings time, 03.05.07 is the start date, and 03:00:00 is the start time to START daylight savings time. |
| 03:31.58 | Mercestes | 3600 is one hour in seconds. |
| 03:31.59 | ltdwk | mercestes: figures |
| 03:32.12 | Mercestes | ltdwk: See? perfectly clear. :D |
| 03:32.26 | ltdwk | mercestes: Clear as in undocumented and nowhere to configure it |
| 03:32.49 | ltdwk | except using the mass deployment config |
| 03:33.04 | ltdwk | Those values are wrong for the timezone that is currently set I believe |
| 03:33.50 | ltdwk | I'm guessing that i'm going to have to configure this value every year |
| 03:35.16 | ltdwk | tessier_: So, to answer your question - yes they do change for DST (I've experienced it myself) however the dst field is generally set incorrectly. |
| 03:36.00 | tessier_ | ltdwk: huh? You are saying the Snom 320 phones will automatically change for DST? |
| 03:36.11 | ltdwk | Yes |
| 03:36.24 | tessier_ | ltdwk: What do you set your timezone setting to? |
| 03:36.27 | *** join/#asterisk tengulre (n=tengulre@222.90.66.10) |
| 03:36.29 | tessier_ | The only options seem to be offsets from GMT. |
| 03:36.43 | tessier_ | timezone$: USA-8 |
| 03:36.45 | ltdwk | AUS+10 |
| 03:37.02 | tessier_ | ltdwk: And if your phone is always GMT+10 how will it change for DST? |
| 03:37.20 | ltdwk | did you just read any of what i was talking about above? |
| 03:37.24 | ltdwk | dst!: 3600 10.01.07 02:00:00 03.05.07 03:00:00 |
| 03:37.30 | ltdwk | there is a field inside the configuration |
| 03:37.48 | ltdwk | that shows how much to adjust, and when to turn it on and off |
| 03:37.53 | tessier_ | ah, I missed your comments above. |
| 03:38.12 | ltdwk | there's nothing in the actual web interface to configure that field though |
| 03:38.23 | ltdwk | so unless you look in the "Settings" page you won't see what it's set to |
| 03:39.22 | ltdwk | I'm guessing when ou set the "timezone" field, it populates dst from some internal table, but the problem is phone manufacturers (Snom included) don't understand DST properly) and don't understand those values change |
| 03:43.23 | tessier_ | dst$: 3600 11.04.07 02:00:00 03.11.07 02:00:00 |
| 03:43.32 | tessier_ | That looks like it should be appropriate for here in the US. |
| 03:43.39 | ltdwk | kewl |
| 03:43.47 | tessier_ | ltdwk: Thanks for pointing that out! :) |
| 03:43.52 | ltdwk | no worries |
| 03:44.08 | ltdwk | keep up the snomming |
| 03:44.15 | tessier_ | Will do. :) |
| 03:44.17 | ltdwk | smite all these polycom using fools |
| 03:44.36 | tessier_ | I've been using asterisk for 3 years now. Polycom was anti-asterisk when I had to make a decision on what phones I would support. |
| 03:44.45 | *** join/#asterisk InHisName (n=Administ@68.38.105.1) |
| 03:45.05 | ltdwk | aye. Snom's a bit more community oriented... plus it's cool to run linux on your phones |
| 03:45.10 | tessier_ | Indeed. |
| 03:47.46 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 03:48.32 | [TK]D-Fender | Polycom > ALL |
| 03:48.51 | ltdwk | Prove it |
| 03:48.53 | Nugget | I'd rather use tin cans and twine than run linux on a phone. |
| 03:49.09 | [TK]D-Fender | Snom means well but I like my phonesnot to crash so much, and polycom wins on LCD usability & audio quality. |
| 03:50.00 | ltdwk | If one crashes I send it back, not that it's really happened |
| 03:50.30 | riddlebox | I know this isnt really topic, but is anyone familiar with the avaya Partner system? |
| 03:51.18 | ltdwk | [TK]: My experience with audio quality is exceptional with the snom's I've used. Unless you were expecting a studio monitor on your phone I don't see how you could complain |
| 03:52.17 | tessier_ | I did have a snom crash a few days ago. I think that is the only thing more I would ask of them. The audio quality is great. |
| 03:52.34 | *** join/#asterisk thoughtpolice (n=austin@ip68-98-250-69.lu.dl.cox.net) |
| 03:52.56 | ltdwk | i have a lot of stock of phones, if i have one that causes issues i've got plenty more |
| 03:53.43 | ltdwk | one guy killed one by plugging in the 220 into a 190 power adapter |
| 03:54.10 | ltdwk | BAM. that was the end of that |
| 03:57.31 | nDuff | ...and he's not really the sour-grapes type, so I don't think it's just general-purpose former-employer unhappiness. |
| 03:57.32 | ltdwk | polycom must be a fairly new company with respect to phones? |
| 03:57.38 | gambolputty | lol |
| 03:57.48 | sivana | ltdwk: no! |
| 03:57.50 | nDuff | ltdwk: They've been making high-end conference phones for a long time. |
| 03:57.52 | gambolputty | yes!! |
| 03:57.53 | sivana | ltdwk: they're like phone/audio gods |
| 03:58.26 | ltdwk | when we bought all our snom gear, i didn't read anything anywhere about polycom |
| 03:58.37 | *** join/#asterisk shido6 (n=shido6@d221-68-200.commercial.cgocable.net) |
| 03:59.08 | tengulre | how to using jabber in asterisk? |
| 03:59.08 | tessier_ | Depends on what you consider "high-end" |
| 03:59.09 | tessier_ | If you mean having 3 speakers/mics pointing in different directions, sure. |
| 03:59.13 | sivana | ltdwk: they've been dealing with phones for years and years |
| 03:59.19 | ltdwk | sivana: and IP phones? |
| 03:59.22 | sivana | conference phones, etc.. |
| 03:59.26 | tessier_ | But if you are the poor guy on the other end of that conference call that has to listen to a whole room it still sounds like crap. |
| 03:59.37 | sivana | yep... |
| 03:59.55 | *** join/#asterisk L|NUX (n=linux@unaffiliated/lnux/x-10290) |
| 04:01.02 | ltdwk | At the end of the day you use what works... |
| 04:01.30 | sivana | yea, I love Polycom |
| 04:02.10 | ltdwk | At the time I chose to use snom, it was considered the highest quality business voip handset you could get |
| 04:02.32 | gambolputty | I will give credit to Polycom for having DST change ability at a fine grain level without a firmware upgrade. |
| 04:02.40 | *** join/#asterisk DocHolliday (i=RgRabbit@gateway/gpg-tor/key-0x0E4F6D6C) |
| 04:03.24 | DocHolliday | is it possible to dial an exten to reach the contents of a context? |
| 04:03.44 | ltdwk | using IAX you can dial into certain contexts |
| 04:04.32 | DocHolliday | ltdwk, yeah i want to dial fron default into [example] |
| 04:04.42 | DocHolliday | an IAX connection shouldn't be necessary |
| 04:05.00 | ltdwk | huh? you're not making much sense |
| 04:05.08 | ltdwk | explain in detail what you want to do |
| 04:07.19 | DocHolliday | i want to dial an extension and have it play the contents of a context |
| 04:07.58 | tessier_ | Finally! All of the phones are rebooted and checked. Now I get to go home. Night all! |
| 04:08.12 | ltdwk | you mean you want to have a handset which has context=default match an extension, and have it Goto() another context? |
| 04:12.12 | ltdwk | http://pastebin.ca/392883 - is that what you mean? |
| 04:14.51 | JacksLivr | help with 7920? |
| 04:16.38 | *** join/#asterisk ShadowHntr (i=sentinel@wikipedia/Shadowhntr) |
| 04:17.06 | JT | ltdwk: i've noticed most people in australia haven't heard of polycom |
| 04:18.04 | ltdwk | JT: probably not that common.... I'd take a german product over a US product any day |
| 04:20.09 | *** join/#asterisk gmaxwell (n=gmaxwell@wikimedia/KatWalsh/x-0001) |
| 04:20.51 | *** join/#asterisk stridernzl (n=neville@125-237-103-112.jetstream.xtra.co.nz) |
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| 04:23.21 | nDuff | Does anyone know what the "lineprobe" tool referenced by Sangoma's wanrouter is? |
| 04:27.37 | *** join/#asterisk maverickbna (i=sentinel@wikipedia/Shadowhntr) |
| 04:30.24 | JT | ltdwk: all the vendors in australia keep trying to push grandstream, linksys and snom |
| 04:30.39 | JT | ltdwk: i don't care where a product is made really |
| 04:30.44 | JacksLivr | http://pastebin.ca/392897 |
| 04:31.08 | DocHolliday | ltdwk, yes |
| 04:31.19 | DocHolliday | sorry when you didnt reference my nick i didn't notice your response |
| 04:32.22 | DocHolliday | is it possible to slow down the beginning of an IVR? it just starts wayy too fast (beginning gets cut off) |
| 04:32.32 | JacksLivr | wait(1) |
| 04:33.07 | DocHolliday | JacksLivr, already tried that |
| 04:33.21 | JacksLivr | hmmm, works for me |
| 04:33.35 | DocHolliday | the sounds file just begins to abruptly |
| 04:34.13 | JacksLivr | i dunno, i had the same prob and that fixed it for me |
| 04:34.30 | JacksLivr | ok, this 7920 will have to rot here tonight. i have to go to bed now |
| 04:34.31 | ltdwk | yeah i always do a Wait(1) after I Answer as there seems to be a gap before the zap channel is forwarding audio |
| 04:34.31 | JacksLivr | night |
| 04:36.10 | ltdwk | on analog Zap's that is |
| 04:36.22 | ltdwk | don't have the same problem on isdn |
| 04:36.52 | DocHolliday | ltdwk, yea |
| 04:36.54 | DocHolliday | well i'm using a DID provider |
| 04:38.29 | DocHolliday | ltdwk, its almost like the first 1/2 second of the sound file is being cut off |
| 04:38.33 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 04:38.46 | ltdwk | yes i experience this. Wait(1) always works for me |
| 04:38.56 | ltdwk | as long as it's after the Answer call |
| 04:39.32 | ltdwk | exten => _X.,n,Answer, exten => _X.,n,Wait(1), exten => _X.,n,Voicemail(${EXTEN:-4},${SMDI_VM_TYPE}) |
| 04:39.37 | ltdwk | that's what I use on my voicemail system |
| 04:39.47 | DocHolliday | i've got exten => 1,1,Answer() exten => s,1,Wait(1) exten => s,2,Playback(ext-or-zero) |
| 04:40.41 | ltdwk | Not sure man. Works for me :-) |
| 04:40.49 | DocHolliday | heh |
| 04:40.54 | ltdwk | Could be due to echo cancel training? |
| 04:40.59 | DocHolliday | very kind of asterisk to work for you and not me :P |
| 04:41.00 | ltdwk | Badly configured |
| 04:41.10 | DocHolliday | possible? |
| 04:41.29 | DocHolliday | what would be the best way to investigate that.. there are no hardware cards being used.. all IP |
| 04:41.51 | ltdwk | it would be on the providers' end, but it's unlikely |
| 04:42.39 | DocHolliday | ah |
| 04:44.13 | DocHolliday | ltdwk, its unusual because after that 1/2 second everything is perfect.. |
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| 05:12.22 | *** join/#asterisk hads (n=hads@reef80.anchor.net.au) [NETSPLIT VICTIM] |
| 05:12.22 | *** join/#asterisk flip2 (i=azbmw@azeuros.org) [NETSPLIT VICTIM] |
| 05:12.22 | *** join/#asterisk nitram (i=foo@superblob.com) [NETSPLIT VICTIM] |
| 05:14.04 | *** join/#asterisk Mahmoud (n=fake@unaffiliated/mahmoud) |
| 05:14.08 | Mahmoud | hello |
| 05:14.35 | Mahmoud | my server uses private ip addresses, but it's placed in a DMZ |
| 05:14.41 | Mahmoud | so, all traffic is forwared to the server |
| 05:14.47 | Mahmoud | server=asterisk 1.4.1 |
| 05:15.13 | Mahmoud | the problem is that my SIP phones send INFO DTMF message destined to my server's private Ip addres |
| 05:15.26 | Mahmoud | the way SIP phones learn it is via SIP messages sent from the Asterisk.. |
| 05:15.44 | Mahmoud | how to not let the Asterisk to advertise its local private IP addresses? |
| 05:19.09 | nDuff | Mahmoud: I think having "DMZ" systems have private address is bad practice in the first place. |
| 05:19.28 | Mahmoud | my router just forwards anything to my asterisk |
| 05:19.30 | nDuff | Mahmoud: Directly assign the system its public IP, and then the whole mess is moot. |
| 05:19.48 | Mahmoud | the ip address is dynamic, i can't.. |
| 05:19.58 | Mahmoud | phones can register and hear voices |
| 05:20.13 | Mahmoud | but the problem is when phones sends DTMF INFO messages, they send it to server's private IP address |
| 05:20.33 | Mahmoud | i wonder where did they learn it? by sniffing packets, i noticed the server is advertising its private ip address.. |
| 05:22.09 | *** join/#asterisk `p4r14h (n=j0sh@69.92.145.178) |
| 05:22.41 | Mahmoud | this is weird.. |
| 05:22.53 | Mahmoud | any idea where do clients learn about server's private ip address? |
| 05:25.21 | DocHolliday | whats the command that gets rid of all the commented stuff in the config files? |
| 05:26.01 | nDuff | DocHolliday: sed? |
| 05:26.40 | nDuff | DocHolliday: or grep -v '^;' |
| 05:26.43 | DocHolliday | nDuff, i want to get rid of everything thats commented in my config files too messy |
| 05:26.44 | nDuff | DocHolliday: or grep -v '^[#;]' |
| 05:26.51 | `p4r14h | grep -v ";" extensions.conf |
| 05:27.00 | DocHolliday | there used to be a way to do it in the asterisk CLLI |
| 05:27.05 | JT | Mahmoud: externip= |
| 05:27.08 | JT | Mahmoud: localnet= |
| 05:27.19 | Mahmoud | JT, isn't this only for registering to proxies? |
| 05:27.30 | nDuff | for FILE in *.conf; do egrep -v '^[#;]' < $FILE > $FILE.new && mv $FILE.new $FILE; done |
| 05:28.40 | DocHolliday | heh |
| 05:33.31 | JT | Mahmoud: no. |
| 05:34.39 | *** join/#asterisk Ryushin (i=proxy@windwalker.openinnovations.com) |
| 05:34.42 | Mahmoud | JT, the problem is that i'm using split DNS |
| 05:34.42 | DocHolliday | whats the best codec for outbound calls? i have the choice of: ulaw,alaw,gsm,libc,g726,adpcm,lpc10 |
| 05:38.29 | Mahmoud | JT, hmmm the issue is that asterisk tries to resolve the name into IP, and i'm using split DNS internally lol |
| 05:39.02 | *** join/#asterisk Faithful (n=Faithful@ns.linuxterminal.com) |
| 05:48.18 | nDuff | If I want to use a channel bank to communicate with a bunch of analog phones, I'd use fxs_ls to communicate with it (as opposed to fxo_ls) -- right? |
| 05:48.25 | Mahmoud | JT, any solution? |
| 05:49.09 | *** join/#asterisk L|NUX (n=linux@unaffiliated/lnux/x-10290) |
| 05:50.45 | DocHolliday | is it possible to get a fresh extensions.conf? |
| 05:51.10 | *** join/#asterisk sumasuma (n=kurukko@61.14.86.23) |
| 05:51.46 | Mahmoud | DocHolliday, extensions.conf.examples? |
| 05:52.02 | DocHolliday | got it :) |
| 05:53.26 | *** join/#asterisk kezza491 (n=opera@c211-28-159-50.brasd1.vic.optusnet.com.au) |
| 05:53.42 | *** join/#asterisk Tili (n=tili@cm109.gamma248.maxonline.com.sg) |
| 05:54.35 | Tili | I have one TDM04B card. I can receive inbound calls. But for outbound the zap thinks channel is answered even though it is not answered and then zap hangsup instantly |
| 05:54.55 | Tili | one thing i must mention is that i see a lot of chan_zap.c:4502 __zt_exception: Exception on 13, channel 1 |
| 05:56.08 | *** join/#asterisk creativx (n=creadure@226.62-97-205.bkkb.no) |
| 05:57.25 | *** join/#asterisk marcan (n=marcanso@160.10.7.121) |
| 05:57.32 | *** part/#asterisk kezza491 (n=opera@c211-28-159-50.brasd1.vic.optusnet.com.au) |
| 06:09.27 | *** join/#asterisk IgorG (n=FeedomPa@195.162.32.126) |
| 06:20.39 | *** join/#asterisk [Airwolf] (n=airwolf@voip.ymav.nl) |
| 06:23.05 | *** join/#asterisk |dennis| (n=dennis@200.32.233.82) |
| 06:25.35 | *** join/#asterisk phpboy (n=shane@196.211.17.202) |
| 06:26.22 | phpboy | hey all, is there any specific hardware that would work VERY well with asterisk? |
| 06:32.34 | *** join/#asterisk reber (n=reber@cl-158.dub-01.ie.sixxs.net) |
| 06:33.41 | *** join/#asterisk jpe-nyc (n=jpe-nyc@p77-37.acedsl.com) |
| 06:34.22 | Tili | phpboy: u want for analog/ISDN BRI/ISDN PRI/SS7? |
| 06:34.23 | *** join/#asterisk abuyazan (n=khaled@dogbert.palnet.com) |
| 06:34.54 | phpboy | ISDN |
| 06:35.05 | phpboy | Quad junghannes card |
| 06:35.15 | phpboy | it'll be doing recording, etc etc |
| 06:39.10 | *** join/#asterisk mtaht4 (n=m@c-71-198-23-124.hsd1.ca.comcast.net) |
| 06:41.19 | DocHolliday | what does this mean? : The option 'notransfer' is deprecated in favor of 'transfer' which has options 'yes', 'no', and 'mediaonly' |
| 06:41.56 | tengulre | anybody come from china? |
| 06:42.41 | DocHolliday | heh, nope |
| 06:42.47 | tengulre | ;( |
| 06:43.26 | tengulre | DocHolliday: what do you doing in company? |
| 06:44.09 | DocHolliday | i dont understand |
| 06:45.07 | *** join/#asterisk Braxus (n=braxus@66.147.214.164) |
| 06:45.09 | tengulre | DH: oh!!! |
| 06:45.35 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 06:45.56 | tengulre | DH: sorry! I known only a litte english! ;( |
| 06:46.32 | tengulre | DH: what work are you doing in your company? |
| 06:46.57 | *** join/#asterisk saftsack (n=oliver@p54A7E02C.dip.t-dialin.net) |
| 06:47.07 | DocHolliday | infrastructure consulting |
| 06:47.08 | *** join/#asterisk yonahw-work (n=yonahw-w@genie03-173-74.inter.net.il) |
| 06:49.09 | tengulre | DocHolliday: what kind of the asterisk's services for your working? |
| 06:50.04 | DocHolliday | mostly do the presales end of things :) helping customers determine what they need and creating the proposals. |
| 06:51.17 | tengulre | COOL! |
| 06:52.43 | tengulre | I using it in my home. provide voicemail service, my name is William Zhang, my chinese name is Zhang Teng Hong. Nice to meet u! |
| 06:53.16 | DocHolliday | likewise. |
| 06:53.44 | DocHolliday | i'm afraid the most personal information i can give you is 'Doc Holliday' :) |
| 06:54.23 | *** join/#asterisk Mahmoud (n=fake@unaffiliated/mahmoud) |
| 06:54.25 | Mahmoud | man.. |
| 06:54.27 | Mahmoud | i don't understand |
| 06:54.43 | tengulre | Yeah! I known. It's doesnt matter! |
| 06:54.46 | Mahmoud | why on earth asterisk is sending it's local ip address in sip message |
| 06:54.54 | Mahmoud | messages* |
| 06:55.12 | DocHolliday | Mahmoud, i have something even weirder happening |
| 06:55.33 | DocHolliday | after my IVR passes the call it still continues the IVR even though the conversation is active |
| 06:55.34 | yonahw-work | Mahmoud, I have it sending an old ip address for the same computer in some SIP messages |
| 06:55.49 | Mahmoud | yonahw-work, just clear you dns cache |
| 06:55.58 | yonahw-work | how do i do that? |
| 06:56.18 | Mahmoud | restart named |
| 06:56.28 | Mahmoud | for windows (ipconfig /flushdns) in cmd |
| 06:56.39 | Mahmoud | or, it could be you sip.conf has externip = 200.201.202.203. so remove this line |
| 06:56.44 | yonahw-work | its centos 4.4 |
| 06:57.02 | yonahw-work | i dont understand what you mean by restart named? |
| 06:57.04 | nDuff | hrm. "wanrouter status" shows the channel bank as "connected" -- but looking at the channel bank itself, it has a light blinking indicating a framing error. |
| 06:57.05 | DocHolliday | yay $50 in voip credits remaining |
| 06:57.17 | Mahmoud | yonahw-work, named (aka bind) |
| 06:57.48 | nDuff | Except that the framing is set to D4 on both the dip switches on the channel bank, the /etc/wanrouter/wanpipe2.conf file and /etc/zaptel.conf. |
| 06:58.44 | Mahmoud | bbiab dears |
| 07:06.05 | *** join/#asterisk nuonguy (n=john@24.6.175.26) |
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| 07:12.55 | *** join/#asterisk topping (n=topping@204.152.96.238) |
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| 07:21.46 | *** join/#asterisk saftsack (n=oliver@p54A7E83F.dip.t-dialin.net) |
| 07:23.51 | *** part/#asterisk [Airwolf] (n=airwolf@voip.ymav.nl) |
| 07:27.34 | *** join/#asterisk dseeb_ (n=dcb@CPE-58-169-152-56.vic.bigpond.net.au) |
| 07:28.57 | *** join/#asterisk Shaun2222 (n=shaun@ip68-4-212-221.oc.oc.cox.net) |
| 07:29.05 | Shaun2222 | is there a way to have all phones setup to use the same line? |
| 07:29.25 | Shaun2222 | for example if i wanted to have line1, line2, line3 displayed on the unit... |
| 07:30.19 | *** join/#asterisk scardinal (n=supreme@port816.ds1-suoe.adsl.cybercity.dk) |
| 07:35.41 | gfraysse | <PROTECTED> |
| 07:37.29 | *** join/#asterisk ShadowHntr (i=sentinel@wikipedia/Shadowhntr) |
| 07:45.35 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 07:46.30 | *** join/#asterisk tzafrir_laptop (n=tzafrir@62.90.10.53) |
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| 07:52.01 | *** join/#asterisk lesouvage (n=lesouvag@cc341200-a.assen1.dr.home.nl) |
| 07:55.35 | nDuff | What's the distinction between "PRI CPE" and "PRI NET"? |
| 08:00.13 | tzafrir_laptop | CPE: TE, NET: NT |
| 08:00.46 | tzafrir_laptop | nDuff, are you trying to connect to a telco? |
| 08:06.06 | Shaun2222 | is there a way to get to voicemail admin area by calling your own voicemail? |
| 08:06.16 | Shaun2222 | basically like presss * and then it asks for password |
| 08:09.08 | tzafrir_laptop | what do you mean by "voicemail admin area"? |
| 08:09.24 | tzafrir_laptop | VoicemailMain(${EXTEN})? |
| 08:09.26 | Shaun2222 | guess it wouldnt be admin, but the employee area... |
| 08:09.28 | Shaun2222 | ya |
| 08:09.33 | *** join/#asterisk inspired (n=mikael@85.221.7.59) |
| 08:09.54 | tzafrir_laptop | actually, more of the sort of: |
| 08:10.04 | tzafrir_laptop | VoicemailMain(${CALLERID(number)})? |
| 08:10.46 | *** join/#asterisk zeach (n=no@62.61.128.141.generic-hostname.arrownet.dk) |
| 08:12.07 | Shaun2222 | exten => _220,1,VoiceMailMain(222) |
| 08:12.17 | Shaun2222 | when i dial that exten, it kills asterisk... |
| 08:12.19 | Shaun2222 | weird. |
| 08:12.25 | Shaun2222 | asterisk ver 1.4.1 |
| 08:13.09 | tzafrir_laptop | that is generally worthy of a bug report, if it is reproducable. |
| 08:13.40 | tzafrir_laptop | Voicemail() also kills Asterisk? |
| 08:13.45 | Shaun2222 | asterisk: symbol lookup error: /usr/lib/asterisk/modules/app_voicemail.so: undefined symbol: ast_adsi_available |
| 08:14.11 | Shaun2222 | voicemail() looks ok... i just called into it... |
| 08:14.21 | zeach | Anybody know if it is possible replace the SIP "to:" header, but keep the original recipient of SIP package? |
| 08:14.30 | tzafrir_laptop | look at the date of app_voicemail.so . Is it a leftover from an older compilation? |
| 08:14.52 | tzafrir_laptop | ls -lS /usr/lib/asterisk/modules |
| 08:15.08 | zeach | So that I can sent an SIP "To" header with sip:user@own-domain ? |
| 08:15.45 | zeach | I know it sems dirty, but it is actually compliant with rfc3261 |
| 08:16.06 | Shaun2222 | tzafrir: looks like the prblem was that i didnt load res_adsi.so |
| 08:16.12 | Shaun2222 | i just added it and it looks to be working now.. |
| 08:20.46 | *** join/#asterisk Chris-NB (n=chris@ng1.kurtkrenn.com) |
| 08:20.50 | Chris-NB | hi |
| 08:20.58 | Chris-NB | anyone got this warning on asterisk cli? |
| 08:21.02 | Chris-NB | codec_ilbc.c:175 ilbctolin_framein: Huh? An ilbc frame that isn't a multiple of 50 bytes long from RTP (4)? |
| 08:21.24 | Chris-NB | i can imagine what it means, but where does it come from? |
| 08:21.54 | Chris-NB | it appears, when I hit the pound key to initiate an attended transfer |
| 08:23.07 | *** join/#asterisk shinux__ (n=shinux@196.201.152.228) |
| 08:24.25 | fetcher | Chris-NB: is the # key properly recognized? |
| 08:24.28 | *** join/#asterisk shinux__ (n=shinux@196.201.152.228) |
| 08:24.46 | Chris-NB | fetcher, think so. login into mailboxmenu works |
| 08:25.04 | Chris-NB | fetcher, so the chars should be recognized correct |
| 08:25.34 | fetcher | Chris-NB: that "short" frame is probably out-of-band DTMF signaling (rfc2833), which Asterisk is treating as a voice frame for some reason |
| 08:26.06 | fetcher | I've never tried iLBC over SIP/RTP, though. That's probably an uncommon pairing |
| 08:26.27 | fetcher | since most things supporting iLBC will also do IAX... |
| 08:27.06 | *** join/#asterisk shinux__ (n=shinux@196.201.152.228) |
| 08:29.56 | *** join/#asterisk shinux__ (n=shinux@196.201.152.228) |
| 08:30.16 | Shaun2222 | anybody know why voicemail(num@default) just plays a beep and not the greeting? |
| 08:30.37 | Shaun2222 | do i need to set the temp greeting? seams weird being named temp |
| 08:31.11 | fetcher | Shaun2222: yeah, "temp" is a bad name... "default" would be more appropriate |
| 08:31.40 | fetcher | temp is used if you don't specify a context (uXXX or bXXX for Busy / Unavailable) when the call's sent to voicemail |
| 08:31.56 | Shaun2222 | i see |
| 08:32.04 | Shaun2222 | ya should be renamed to default.. |
| 08:32.20 | fetcher | I edited the voice prompt to remove the word "temporary", after some of our users were confused by that |
| 08:32.25 | *** join/#asterisk shinux__ (n=shinux@196.201.152.228) |
| 08:38.01 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 08:38.10 | *** join/#asterisk r0d3nt (i=nobody@foster.stonedcoder.org) |
| 08:38.31 | Chris-NB | fetcher, I'v connected asterisk to a alcatel 4400 pbx |
| 08:38.44 | Chris-NB | fetcher, the ilbc calls come from there, i think |
| 08:39.09 | fetcher | Chris-NB: so Alcatel's supporting iLBC now? Cool... |
| 08:39.29 | fetcher | I wish the major softphone vendors would add it |
| 08:39.40 | fetcher | s/softphone/hardphone/ |
| 08:40.15 | *** join/#asterisk modulus_ (n=modulus@shell.blacksun.net) |
| 08:40.20 | modulus_ | yo |
| 08:40.51 | Chris-NB | fetcher, I'm not exactly sure where the iLBC thing comes from |
| 08:40.53 | modulus_ | wow there's afuckload of ppl here |
| 08:40.55 | *** join/#asterisk ComaVN (n=blaargh@unaffiliated/comavn) |
| 08:40.58 | Chris-NB | fetcher, I wanna get rid of it : D |
| 08:41.16 | modulus_ | spa 941 error msg: Got SIP response 486 "Busy Here" |
| 08:41.25 | modulus_ | anyone know what i have to change in sip.conf? |
| 08:41.37 | fetcher | ya, I seem to be one of the only iLBC fans here :) |
| 08:41.47 | modulus_ | i set all the settings just like my other sipura |
| 08:41.52 | modulus_ | sip registers fine too |
| 08:42.18 | *** join/#asterisk Mahmoud (n=fake@unaffiliated/mahmoud) |
| 08:42.19 | Mahmoud | hello |
| 08:42.28 | Mahmoud | asterisk is great. my problem is solved |
| 08:42.50 | fetcher | modulus_: could the SPA be in a do-not-disturb mode? Polycom's response with that "Busy here" msg when they are |
| 08:42.52 | Mahmoud | SIP works perfectly over my damn monopoly analogish anti-viop ISP |
| 08:43.22 | modulus_ | fetcher, do you use spa 941? |
| 08:43.32 | fetcher | Mahmoud: anti-VoIP? Do they try to block/interfere with it? |
| 08:43.45 | Mahmoud | fetcher, oh dear.. you have no clue |
| 08:43.59 | Mahmoud | fetcher, they block SIP based on application layer (not only port number) |
| 08:44.57 | fetcher | Mahmoud: nasty... where is this? I heard about that happening in a south/central American country recently.. Belize? |
| 08:45.00 | Mahmoud | after modifying my asterisk + sip clients, I had to add localnet and externip and that's it.. |
| 08:45.09 | *** join/#asterisk olivierb (n=olivierb@LSt-Amand-152-31-23-31.w82-127.abo.wanadoo.fr) |
| 08:45.10 | Mahmoud | fetcher, UAE |
| 08:45.42 | fetcher | interesting. I wonder if they bother IAX at all |
| 08:45.44 | Mahmoud | fetcher, the ISP is rich, with lots of b/w, but they want more money since they provide analog phones too |
| 08:46.09 | Mahmoud | fetcher, not IAX, but IAX is less efficent when it comes to transfering voice |
| 08:46.26 | Mahmoud | IAX is better than SIP for signalling and (as i heard) for trunk links |
| 08:47.09 | fetcher | Mahmoud: actually IAX has less overhead per speech frame (4 bytes, vs. 12 for RTP). |
| 08:47.26 | Mahmoud | fetcher, oh.. you may be right |
| 08:47.42 | Mahmoud | fetcher, I run ethereal once, and didn't notice that IAX doesn't use RTP! |
| 08:48.17 | fetcher | Mahmoud: yup, everything over a single udp/4569 socket... voice and signaling all muxed together, which is nice when dealing with firewalls |
| 08:50.13 | Mahmoud | fetcher, you are 100% right! |
| 08:50.28 | Mahmoud | just sniffed packets with ethereal, and IAX doesn't use RTP |
| 08:50.52 | Mahmoud | so, IAX is just better |
| 08:51.10 | Mahmoud | except that IAX doesn't have much soft or hard phones.. |
| 08:51.42 | fetcher | Mahmoud: yeah, that's the main drawback. It's a simpler standard than SIP, though, so hopefully more vendors will pick it up over time |
| 08:52.23 | Mahmoud | yeah, although it's not a standard protocol, companies are still making phones working with it.. |
| 08:52.34 | Mahmoud | what about Skinny? something that cisco made, didn't read about it yet.. |
| 08:53.25 | *** join/#asterisk hieunm_vips (n=hieunm_v@210.245.57.162) |
| 08:53.37 | fetcher | Mahmoud: Skinny, aka SCCP is mosly on the way out... no reason to use it unless you have to for compatibility |
| 08:53.55 | Mahmoud | is it older than SIP? |
| 08:54.01 | hieunm_vips | Hi all, Could I ask a question about Playback application |
| 08:54.02 | fetcher | Mahmoud: yup. |
| 08:54.12 | Mahmoud | fetcher, does cisco rely on SIP at the moment? |
| 08:54.18 | fetcher | Mahmoud: MGCP is another one, and of course H.323. All use RTP for the actual voice transport |
| 08:54.29 | hieunm_vips | How can I playback a random length sound file in specified period of time |
| 08:54.47 | hieunm_vips | For ex, playback hello-world for 3 minutes |
| 08:54.59 | fetcher | RTP is an extremely old protocol. I've seen references to it in RFCs written in 1981 |
| 08:55.00 | Mahmoud | fetcher.. i see |
| 09:01.52 | *** join/#asterisk litnimax (n=chatzill@host-86-106-208-182.moldtelecom.md) |
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| 09:09.31 | *** join/#asterisk Dovid (n=Dovid@l192-117-114-1.broadband.actcom.net.il) |
| 09:09.54 | Dovid | morning all |
| 09:09.59 | Dovid | can anyone tell me what this error means ? |
| 09:10.01 | Dovid | Leaving directory `/usr/src/asterisk-1.2.16/codecs' |
| 09:12.41 | sbingner | uih |
| 09:12.45 | sbingner | that it's not an error? |
| 09:12.59 | Dovid | yes |
| 09:13.00 | *** join/#asterisk shinux__ (n=shinux@196.201.152.228) |
| 09:13.11 | Dovid | make[1]: *** [codec_zap.o] Error 1 |
| 09:13.12 | Dovid | make[1]: Leaving directory `/usr/src/asterisk-1.2.16/codecs' |
| 09:13.12 | Dovid | make: *** [subdirs] Error 1 |
| 09:13.31 | Dovid | is the exact errors - tryin to install asterisk 1.2.16 with zaptel 12.8 |
| 09:13.38 | sbingner | no, no those are not the errors |
| 09:13.45 | sbingner | those are the notes after the errors |
| 09:14.04 | Dovid | hmm |
| 09:14.09 | Dovid | gona pb it all |
| 09:15.15 | tzafrir_laptop | Dovid, do you really need codec_zap? |
| 09:15.34 | Dovid | i believe so |
| 09:15.35 | tzafrir_laptop | (g729 codec in a Digium card?) |
| 09:15.42 | Dovid | using sangoma |
| 09:15.48 | Dovid | lol |
| 09:15.50 | Dovid | guess not |
| 09:16.02 | sbingner | that would probably be chan_zap |
| 09:16.06 | Dovid | but y would it spit out the error ? |
| 09:16.21 | Dovid | can it be cause i am tryin to use zaptel 1.2.8 with asterisk 1.2.16 ? |
| 09:16.29 | tzafrir_laptop | you need a newer zaptel to support codec_zap . Or just don't build codec_zap |
| 09:16.43 | Dovid | how would i disable it ? |
| 09:16.59 | tzafrir_laptop | codecs/Makefile somewhere |
| 09:17.12 | Dovid | so codecs in Makefile? |
| 09:18.24 | Dovid | this is the only word codecs in the Makefile |
| 09:18.33 | Dovid | SUBDIRS=res channels pbx apps codecs formats agi cdr funcs utils stdtime |
| 09:18.44 | sbingner | Dovid, no edit Makefile in the "codecs" directory |
| 09:19.01 | sbingner | or, "codecs/Makefile" |
| 09:19.07 | *** join/#asterisk shinux__ (n=shinux@196.201.152.228) |
| 09:19.07 | Dovid | kk |
| 09:19.30 | sbingner | or just update your zaptel like he said |
| 09:19.36 | *** join/#asterisk active_si (n=AcTiVe@84-255-238-137.static.dsl.t-2.net) |
| 09:19.39 | Dovid | i think thats the better idea ;) |
| 09:19.44 | *** join/#asterisk psk (n=psk@golia.caltanet.it) |
| 09:19.55 | *** join/#asterisk shinux__ (n=shinux@196.201.152.228) |
| 09:19.55 | sbingner | I expect your actual errors would have pointed to something along those lines |
| 09:20.45 | *** join/#asterisk Aurs (n=Aurs@c6870BF51.dhcp.bluecom.no) |
| 09:20.56 | *** join/#asterisk shinux__ (n=shinux@196.201.152.228) |
| 09:21.19 | *** join/#asterisk tcseke (n=chatzill@22-36.adsl.etel.hu) |
| 09:22.49 | Dovid | anyone know the most recent wanpipe version ? |
| 09:25.40 | *** join/#asterisk shinux__ (n=shinux@196.201.152.228) |
| 09:28.12 | tzafrir_laptop | the sangoma wiki? |
| 09:30.03 | *** join/#asterisk MrChimpy (n=MrChimpy@212.158.8.162) |
| 09:30.38 | Dovid | kk |
| 09:30.43 | Dovid | a bit tired. sorry |
| 09:31.00 | yonahw-work | tzafrir: may I pm you? |
| 09:33.35 | *** join/#asterisk shinux__ (n=shinux@196.201.152.228) |
| 09:34.41 | *** join/#asterisk RoyK (n=roy@217-175-152.100710.adsl.tele2.no) |
| 09:34.53 | Dovid | hmm |
| 09:35.01 | Dovid | thier wiki isnt the most forthcoming |
| 09:35.03 | Dovid | a bit confusing |
| 09:35.06 | Dovid | argh !! |
| 09:35.33 | Dovid | for now i got asterisk working wby comenting out chan_zap |
| 09:36.11 | *** part/#asterisk litnimax (n=chatzill@host-86-106-208-182.moldtelecom.md) |
| 09:36.18 | Dovid | tzafrir: even if they are using a sangoma card it shouldnt be an issue if i commented out codec_zap |
| 09:36.51 | *** join/#asterisk shinux__ (n=shinux@196.201.152.228) |
| 09:38.43 | *** join/#asterisk shinux__ (n=shinux@196.201.152.228) |
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| 09:40.46 | *** join/#asterisk geoaxis (n=geoaxis@58-65-160-140.nayatel.pk) |
| 09:40.50 | geoaxis | hello people |
| 09:41.19 | yonahw-work | hi |
| 09:41.23 | *** join/#asterisk shinux__ (n=shinux@196.201.152.228) |
| 09:41.26 | geoaxis | can any one recomend me a tutorial for setting up asterisk with SIP |
| 09:41.38 | geoaxis | i just pulled code from svn and installed it |
| 09:42.22 | yonahw-work | geoaxis: check out asteriskguru.com |
| 09:42.24 | *** join/#asterisk shinux__ (n=shinux@196.201.152.228) |
| 09:43.00 | Dovid | geoaxis: have a look at the wiki, also have you read the book ? |
| 09:43.02 | Dovid | ~wiki |
| 09:43.05 | *** join/#asterisk shinux__ (n=shinux@196.201.152.228) |
| 09:43.11 | Dovid | ~book |
| 09:43.15 | jbot | [book] a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 09:43.15 | Dovid | ~wiki |
| 09:43.31 | Dovid | have a look at voip-info.org |
| 09:44.22 | *** join/#asterisk shinux__ (n=shinux@196.201.152.228) |
| 09:44.26 | tzafrir | Dovid, well, is it an issue if you comment-out codec_zap? |
| 09:45.14 | *** join/#asterisk shinux__ (n=shinux@196.201.152.228) |
| 09:45.26 | Dovid | tzafrir: i want to just understand what it does |
| 09:45.35 | Dovid | tzafrir: when i comented it out it worked fine |
| 09:46.38 | *** join/#asterisk shinux__ (n=shinux@196.201.152.228) |
| 09:47.10 | geoaxis | is there a cleaner way to shutdown asterisk other than killing the process |
| 09:47.30 | Dovid | geoaxis: yes |
| 09:47.40 | Dovid | geoaxis: from cli type in help |
| 09:47.48 | *** join/#asterisk topping (n=topping@204.152.96.238) |
| 09:47.57 | Dovid | geoaxis: one of the options is: shut down when convenient |
| 09:48.49 | geoaxis | is there grep facility with in the asterisk CLI |
| 09:49.21 | Dovid | not that i know of but you can do this |
| 09:49.31 | Dovid | asterisk -rx 'insert command here | grep whatever |
| 09:49.53 | Dovid | asterisk -rx 'insert command here' | grep whatever |
| 09:49.58 | Dovid | do this from the CLI |
| 09:50.32 | e-ddie | asterisk -rx 'insert command here' | grep 'exclude command here' |
| 09:51.56 | Dovid | thanx e |
| 09:53.34 | geoaxis | I better read the book :) |
| 09:54.32 | Dovid | hehe |
| 09:54.35 | Dovid | its pretty god |
| 09:54.37 | Dovid | good* |
| 09:54.39 | geoaxis | btw, any one from dev who can tell me if asterisk would be in google summer of code this year |
| 09:55.02 | geoaxis | Asterisk, was part in 2005 but not in 2006 |
| 10:01.59 | FreezeS | is voip-info.org down, or it's just my local problem ? |
| 10:02.02 | Dovid | argh !! i just crashed the box ;) tried rmmod wanpipe and now i cant get it |
| 10:02.35 | Dovid | Freezes: not working for me either |
| 10:02.58 | *** join/#asterisk arcanine (n=arcanine@203.82.44.179) |
| 10:16.52 | *** join/#asterisk fbffff (n=fbffff@r74-193-79-81.pfvlcmta01.grtntx.tl.dh.suddenlink.net) |
| 10:18.20 | *** join/#asterisk [Airwolf] (n=airwolf@89.205.159.86) |
| 10:21.47 | *** join/#asterisk jm|work (n=jm@sentry.flags.co.uk) |
| 10:24.00 | Mahmoud | any good music on hold samples? |
| 10:24.13 | Mahmoud | i don't want music, it's annoying most of the time, just some electronic sound |
| 10:24.18 | Mahmoud | any good site for these kind of stuff? |
| 10:25.17 | Dovid | eh. not really |
| 10:25.27 | Dovid | atleast not that i know of |
| 10:25.54 | Dovid | liscence some stuff out and then u can use it ;) |
| 10:26.15 | Mahmoud | yeah, but it's stealing :P |
| 10:26.24 | Dovid | not if u get a liscence |
| 10:26.52 | Dovid | hence liscnce some stuff ;) |
| 10:27.28 | Mahmoud | this one is cool http://www.ilaudio.com/mpeg/Int_6.mp3 |
| 10:28.04 | *** join/#asterisk backblue (n=igor@82.102.1.42) |
| 10:28.24 | backblue | hi, there is any web interface for queues, opensource for asterisk? |
| 10:28.56 | Dovid | backblue: pick ur fav. |
| 10:29.04 | Dovid | www.asternic.org |
| 10:29.10 | Dovid | (i think thats the URL) |
| 10:29.13 | Dovid | quemetrics |
| 10:29.26 | Dovid | have a look on the wiki (i think its down at the moment) |
| 10:29.32 | Dovid | www.voip-info.org |
| 10:31.45 | geoaxis | you can always use stuff (music etc) which is licences under creative commons |
| 10:33.22 | backblue | queumetrics it's not opensource |
| 10:33.30 | Dovid | sorry - thought it was |
| 10:33.52 | Dovid | wait till voip-info comes back up (first time i have ever seen it down) |
| 10:34.06 | backblue | Dovid: i only want statistics |
| 10:34.32 | *** part/#asterisk hieunm_vips (n=hieunm_v@210.245.57.162) |
| 10:36.53 | Dovid | try astrinic |
| 10:37.23 | Dovid | i know i am not spelling it correctly |
| 10:37.32 | Mahmoud | hmmmm any way with asterisk to play MOH when the analog phone places the phone on-hock but the callee didn't go on-hock yet? |
| 10:37.47 | Dovid | hwre u go |
| 10:37.47 | Dovid | http://www.asternic.org/ |
| 10:37.57 | Dovid | here* |
| 10:38.14 | Mahmoud | dive in? |
| 10:38.20 | Dovid | Mahmoud: u want to play MOH when ? |
| 10:38.26 | Dovid | when the phone is on hook ? |
| 10:39.04 | Mahmoud | let's say you called me, and my phone is an analog one connected to my Asterisk via a FXS card that's installed on Asterisk |
| 10:39.18 | Mahmoud | my phone doesn't support on-hold, so ill just go on-hock |
| 10:39.28 | Mahmoud | normally, if i go off-hock, i'll see you listening on the line |
| 10:39.37 | Mahmoud | so, in the mean time, hear some MOH, possible? |
| 10:39.58 | Dovid | meaning u flash the line ? |
| 10:40.28 | Dovid | i dont know zap to well but from what i remember if u flash hook then it should play MOH |
| 10:40.34 | Dovid | also u can look at parking the clal |
| 10:40.35 | Dovid | call* |
| 10:40.57 | Mahmoud | flash? |
| 10:41.02 | Mahmoud | i meant, on-hook, sorry |
| 10:41.06 | Mahmoud | horrible typo |
| 10:42.19 | Dovid | i am a bit tired |
| 10:42.48 | Dovid | i believe that if u leave the phone on hook after a bit it will dump the call |
| 10:42.57 | Dovid | if u flash hook then it should give the user MOH |
| 10:43.04 | Dovid | u can also park the call |
| 10:44.12 | Mahmoud | i see |
| 10:44.25 | Mahmoud | i'll try it once i buy a TD411P card |
| 10:44.35 | Mahmoud | TDM411P rather |
| 10:45.13 | Dovid | y not go with a voip phone ? |
| 10:45.35 | Mahmoud | soft voip phones |
| 10:45.37 | Dovid | with the prices per port on an fxs card its not much more to get a basic voip phone such as the spa-841 |
| 10:46.17 | Dovid | http://search.ebay.com/search/search.dll?cgiurl=http%3A%2F%2Fcgi.ebay.com%2Fws%2F&fkr=1&from=R8&satitle=spa-841&category0=&submitSearch=Search |
| 10:46.20 | Mahmoud | i need an analog interface any way, i should be connecting to pots |
| 10:46.23 | Dovid | or soft |
| 10:46.36 | Dovid | well that will be FXO (if u r connecting to a telco line) |
| 10:46.44 | Dovid | Mahmoud: where are you located ? |
| 10:46.56 | Mahmoud | yeah, and i have an old analog phone already, so thought of buying an FXS module too |
| 10:46.59 | Mahmoud | UAE |
| 10:47.22 | Dovid | if u can get stuff from ebay get the spa-841 |
| 10:47.25 | Mahmoud | which one is more expensive, FXS module, or a voip phone that supports SIP? |
| 10:47.31 | Dovid | i posted a link on ebay a moment ago |
| 10:47.46 | Dovid | depends. if u go with used SIP u can get it a lot cheaper |
| 10:47.58 | *** join/#asterisk xheliox (n=jeff@pdpc/supporter/active/xheliox) |
| 10:48.06 | Mahmoud | heh used.. tough to find used voip phones here |
| 10:48.15 | Dovid | though u would want a card for timing (even though u can use ztdummy - u get slightly better timing with a card) |
| 10:48.24 | Mahmoud | my house will be the 1st house to deploy voip in my city i guess |
| 10:48.30 | Dovid | u guys cant get stuff from ebay? |
| 10:48.31 | Dovid | lo |
| 10:48.32 | Dovid | lol* |
| 10:48.53 | Mahmoud | dunno.. paypal doesnt work for my country |
| 10:49.14 | Dovid | oh well. so get one remotely ;) |
| 10:49.23 | Mahmoud | that's expensive |
| 10:49.31 | masked | bugger |
| 10:49.38 | masked | how do i get wav to g711? |
| 10:49.45 | Mahmoud | $24.00 for a used voip |
| 10:50.13 | Dovid | hmm |
| 10:50.17 | Mahmoud | masked, use cool edit pro, or adobe audition, then record the voice with 8000 samples per second, and 32 bit |
| 10:50.20 | Dovid | masked: how r u recording it ? |
| 10:50.25 | masked | uhmma |
| 10:50.35 | masked | any nix based stuff? |
| 10:50.56 | florz | masked: Why not 512 bit? |
| 10:50.58 | florz | hmpf |
| 10:51.03 | florz | Mahmoud: Why not 512 bit? |
| 10:51.09 | masked | sorry i deleted windows back in 95 |
| 10:51.11 | Mahmoud | florz, does it work? |
| 10:51.26 | florz | masked: sox should do |
| 10:51.30 | Mahmoud | florz, dunno, i tried other bits and didn't work, only 32 worked with adobe, others caused an error |
| 10:51.52 | Mahmoud | sorry, actually 16 bit worked |
| 10:51.57 | Mahmoud | 32 bits, or 8 bits didn't work |
| 10:52.26 | Mahmoud | cool edit pro gives 3 options, 8bit, 16bit and 32bit.. 16bit and 8000 samples is the only thing that worked with me |
| 10:52.32 | *** join/#asterisk kezza491 (n=opera@c211-28-159-50.brasd1.vic.optusnet.com.au) |
| 10:52.36 | Mahmoud | other combinations didn't work with me.. |
| 10:52.47 | florz | Mahmoud: IC. For, 32 bit really doesn't make much sense if you wanna use it for something that effectively does have only 12 bits of resolution. |
| 10:53.02 | reber | Dovid, what are the consequenses of a "bad timing" ? |
| 10:53.47 | florz | Mahmoud: And only very few sound cards (any, actually?) are likely to ever deliver 32 bit samples where the LSB(s) is(are) not just noise ... |
| 10:54.06 | Dovid | reber: yes - if u want confrencing |
| 10:54.10 | Dovid | and sometimes for MOH |
| 10:54.19 | Mahmoud | MGH? |
| 10:54.33 | Mahmoud | oh, MOH, my eyes, too tired |
| 10:55.02 | Dovid | hehe |
| 10:55.12 | Dovid | it happens to the best of us. i got 3 hours in last night |
| 10:55.26 | Mahmoud | only? :/ |
| 10:55.33 | Dovid | teehee |
| 10:55.44 | Mahmoud | i'm more than 12 hours awake |
| 10:55.55 | Mahmoud | but i slept alot |
| 10:55.57 | Dovid | what time is it by u ? |
| 10:56.19 | Mahmoud | 2:56PM |
| 10:56.27 | Dovid | 12:56 here |
| 10:56.39 | reber | 11:56 here |
| 10:56.48 | Mahmoud | where do you live Dovid |
| 10:57.00 | Mahmoud | austeria? |
| 10:57.01 | Dovid | Israel |
| 10:57.09 | Dovid | well I live in the US. In israel now |
| 10:57.18 | Mahmoud | jews? =P |
| 10:57.24 | Dovid | yea yea. |
| 10:57.29 | Dovid | -P |
| 10:57.41 | Mahmoud | with all the religious bits around you? |
| 10:57.50 | Dovid | bits = ? |
| 10:57.57 | Mahmoud | clothing stuff |
| 10:58.08 | Mahmoud | not bits as 0 or 1 heh |
| 10:58.15 | Dovid | hehe |
| 10:58.22 | Mahmoud | bits = small pieces |
| 10:58.43 | Mahmoud | most jews i meet in irc are good, better than others, wonder why we fight |
| 10:58.57 | Dovid | i dont think WE ALL fight |
| 10:59.00 | Dovid | its a select few |
| 10:59.13 | reber | "make love ..." |
| 10:59.21 | Dovid | i knew u were muslim from ur name - didnt stop me from helping u |
| 10:59.40 | Mahmoud | nah, you helped me to just say that jews people are good =P |
| 11:00.01 | Mahmoud | joking, never mind.. |
| 11:00.42 | Dovid | bbs |
| 11:00.58 | Mahmoud | take care |
| 11:02.04 | Mahmoud | special camels with extra camel fleas to protect you from enemies |
| 11:03.38 | Mahmoud | florz, and, for sure, it should be mono =] |
| 11:03.48 | *** part/#asterisk backblue (n=igor@82.102.1.42) |
| 11:03.59 | Dovid | camels for marrige ? (sorry i had to) |
| 11:04.01 | florz | unless you do have a stereo telephone. Erm, whatever ... ;-) |
| 11:04.30 | Mahmoud | Dovid, you married them? |
| 11:04.59 | Mahmoud | Dovid, jews don't even marry non-jew, how come you married a camel? lol |
| 11:05.22 | florz | it was probably a jewish camel, I guess? *duck* |
| 11:05.33 | *** join/#asterisk lnx (n=lnx@mail.oefi.hu) |
| 11:05.36 | lnx | hi all |
| 11:06.05 | *** join/#asterisk zotz (n=zotz@24.244.163.157) |
| 11:06.13 | *** join/#asterisk cpm (n=Chip@pdpc/supporter/active/cpm) |
| 11:06.37 | Mahmoud | a camel that managed to scape The Holocaust |
| 11:09.23 | *** join/#asterisk FaithX (n=faithful@ns.linuxterminal.com) |
| 11:09.35 | lnx | i have voipnow-asterisk-1.2.13-070124.11.rhel4 in CentOS, is it possible to contains h263+ ? I set videosupport=yes in sip.conf and output of asterisk -d does not say about h263+ anything. |
| 11:10.08 | florz | You mean, like, a camel from Germany? I doubt it =:-) |
| 11:10.53 | Mahmoud | at least they have two ^s rather than one ^ |
| 11:11.27 | florz | How ya mean? Don't get it ... |
| 11:11.41 | Mahmoud | ~^^o |
| 11:11.56 | Mahmoud | ^^ on their back |
| 11:12.19 | florz | Yeah, I guess I got that, but not how that fit into the context ... |
| 11:12.26 | Mahmoud | while arabic camels have only one, tough to set on! http://camelphotos.com/photos/breeds-3.jpg |
| 11:12.38 | Mahmoud | arabian* |
| 11:13.17 | Mahmoud | context=camels |
| 11:13.18 | Mahmoud | done |
| 11:14.30 | *** join/#asterisk qdk (n=qdk@213.150.62.32) |
| 11:15.45 | Mahmoud | sleep time |
| 11:16.05 | Mahmoud | night all =] |
| 11:21.55 | yonahw-work | what was all that about camels surviving the holocaust? Are there many European camels? |
| 11:26.26 | *** join/#asterisk walld (n=walld3@212.248.241.2) |
| 11:27.24 | walld | anyone can help with a trunk selection problem? |
| 11:30.19 | *** join/#asterisk rrobert (n=rrobert@mbl-82-51-38.dsl.net.pk) |
| 11:30.23 | Dovid | walld: whats the issue (not to good with trunks but I can give it a shot) |
| 11:30.39 | JT | sounds like walld is using freepbx |
| 11:30.45 | Dovid | hehe |
| 11:30.56 | Dovid | walld: r u using freepbx ? |
| 11:30.59 | Dovid | or trixbox? |
| 11:33.48 | Dovid | i know this is OT but does anyone know of a good device that will allow me to get console access to a box over IP ? |
| 11:34.40 | JT | ssh |
| 11:34.57 | walld | yes I'm using freepbx and when the dialout-trunk macro is used it returns busy and doesn't attempt to use any of the other trunks. |
| 11:35.59 | JT | ~freepbx |
| 11:36.08 | jbot | rumour has it, freepbx is unable to be supported here. It's a complex piece of dialplan and GUI, and can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking there |
| 11:36.46 | walld | many thanks |
| 11:41.24 | walld | BTW is this a known issue with freepbx/trixbox? |
| 11:42.00 | *** join/#asterisk rrrobert (n=rrobert@mbl-82-51-38.dsl.net.pk) |
| 11:42.04 | rrrobert | ? |
| 11:42.10 | JT | no idea |
| 11:42.10 | *** join/#asterisk Mavvie (n=edwin@ppp23-199.lns2.syd7.internode.on.net) |
| 11:42.56 | rrrobert | aaahh... then where could i get the idea:) |
| 11:50.12 | *** join/#asterisk coppice (n=chatzill@90.203.17.210.dyn.pacific.net.hk) |
| 11:50.56 | *** join/#asterisk Vec (n=Vec@dsl-241-146-98.telkomadsl.co.za) |
| 11:51.54 | Vec | Hi could someone please tell me what pins on the RJ45 connector are the TX and RX for a digium card? |
| 11:54.07 | Aurs | Which pins of the crossover cable are for TX- TX+ RX- RX+? |
| 11:54.07 | Aurs | 1: RX-, 2: RX+, 4: TX-, 5: TX+ |
| 11:54.09 | *** part/#asterisk kezza491 (n=opera@c211-28-159-50.brasd1.vic.optusnet.com.au) |
| 11:54.34 | Vec | Aurs : yeh thanks! |
| 11:54.39 | Aurs | Vec: http://kb.digium.com/entry/1/17/ |
| 12:01.43 | JT | "a digium card" |
| 12:01.56 | JT | i'm going to take a wild guess that you mean a t1/e1 card |
| 12:02.01 | *** join/#asterisk Skaag (n=skaag@212.199.180.157.static.012.net.il) |
| 12:02.13 | Skaag | can someone please remind me the name of the zapata timing interface kernel module? |
| 12:02.31 | Merlin83b | ztdummy ? |
| 12:02.40 | Skaag | thanks!! |
| 12:02.42 | Skaag | :-) |
| 12:03.27 | Skaag | ok the reason it didn't load is that my kernel was upgraded from 2.6.15-27 to 2.6.15-28 |
| 12:03.32 | Skaag | guess I have to recompile the module |
| 12:04.09 | Vec | JT : yeh |
| 12:04.25 | JT | Vec: search voip-info for t1 crossover cable |
| 12:04.45 | Vec | When a E1 line is connected to a digium E1 card correctly, does the led on the card light a perticular color, at the moment its flashing red ? |
| 12:05.05 | *** join/#asterisk caio1982 (i=caio1982@CAcert-br/caio1982) |
| 12:05.46 | JT | yes it should go green |
| 12:06.32 | Vec | any idea how long it takes ? |
| 12:06.42 | JT | is asterisk running? |
| 12:07.05 | Vec | yeh |
| 12:07.17 | JT | Vec: is the unboard jumper set to E1? |
| 12:07.21 | JT | onboard |
| 12:07.26 | Vec | yup |
| 12:07.36 | JT | what is it connected to? |
| 12:07.53 | geoaxis | any one using asterisk on gentoo here |
| 12:08.42 | Vec | to a crone block from the telco, the telco is not sure which one is TX- or TX+ |
| 12:08.42 | Vec | so need to play around |
| 12:08.43 | JT | krone :) |
| 12:08.43 | JT | lol stupid telco |
| 12:08.43 | JT | it's easy to find out with an oscilloscope |
| 12:08.45 | *** join/#asterisk vasquez (n=vasquez@85.183.64.6) |
| 12:10.00 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 12:10.17 | Vec | JT : don't have one on me :) just a multimeter |
| 12:10.30 | coppice | is this the old krone from hansel and gretel? |
| 12:10.42 | JT | hmm, even that might show something, never know |
| 12:11.32 | Vec | no, krone from block |
| 12:12.26 | Vec | Does zapata.conf need to be configured correctly for the light to go green or only zaptel.conf ? |
| 12:13.05 | *** join/#asterisk msetim (n=marcos@200.195.161.164) |
| 12:18.19 | Aurs | Vec: should only be necessary to start zaptel, not asterisk, ergo: zaptel.conf |
| 12:18.45 | Aurs | zttool might give you some info |
| 12:21.40 | *** join/#asterisk FreezeS (n=bla@82.208.157.125) |
| 12:22.43 | *** join/#asterisk |dennis| (n=dennis@200.32.233.82) |
| 12:29.15 | *** join/#asterisk TaiSHi (n=juanma@zion.dattaweb.com) |
| 12:29.19 | TaiSHi | Hello everyone |
| 12:30.37 | masked | Heaveno TaiSHi |
| 12:30.53 | E-bola | do any1 knwo what the default password is for grandstream phones? |
| 12:31.05 | E-bola | nevermind |
| 12:31.13 | TaiSHi | Heaveno ? |
| 12:31.29 | Skaag | what could this be: install: cannot stat `udev/zaptel.rules-combined': No such file or directory |
| 12:31.29 | Skaag | make: *** [devices] Error 1 |
| 12:31.50 | TaiSHi | I can't find that file |
| 12:31.57 | TaiSHi | ls udev/ |
| 12:32.42 | masked | TaiSHi: yer iono it was some crazy proposal by some churchies last year that we should change our greetings |
| 12:33.13 | *** join/#asterisk Vec (n=Vec@dsl-243-102-225.telkomadsl.co.za) |
| 12:34.15 | DrukenLPY | figures... some religious cult thing |
| 12:34.25 | rrrobert | i am using Asterisk1.4, there is a snmp module which comes with asterisk internally. How can i configure that module. |
| 12:37.49 | *** join/#asterisk Turt|e (n=danny@0x55532532.adsl.cybercity.dk) |
| 12:37.55 | *** join/#asterisk myiagy (n=myiagy@200.175.61.250.static.gvt.net.br) |
| 12:38.43 | *** join/#asterisk gambolputty (n=gambolpu@cblmdm72-240-185-4.buckeyecom.net) |
| 12:42.18 | Turt|e | Hi, i run openbsd4.1 and asterisk 1.2.15, and i, having troubles with moh and mp3s i have installed the mpg123(not mpg321), and converted the mp3 to 128bit and removed the id3 tags. In the cli i starts fine by telling me that the music on hold is playing however there isnt a sound, and after i have printed the lines with the numbers to call i says that music on hold stopped. Is this a know issue? What om i overseeing? |
| 12:43.46 | Vec | I dont know whats wrong with "switchtype=euroisdn" I keep getting warning ignoring switchtype ? |
| 12:46.37 | *** join/#asterisk wunderkin (n=kev@ip72-208-1-190.ph.ph.cox.net) |
| 12:50.46 | JT | pastebin.ca the relevant configs |
| 12:53.22 | *** join/#asterisk fakhir (i=fakhir@unaffiliated/fakhir) |
| 12:53.26 | *** join/#asterisk ToyMan (n=Stuart@12.23.30.130) |
| 12:58.12 | wunderkin | abovenet in phoenix is fucked... err |
| 12:59.53 | DrukenLPY | why? |
| 13:01.52 | *** join/#asterisk JunK-Y (n=junky@modemcable140.185-70-69.mc.videotron.ca) |
| 13:04.10 | TaiSHi | Is autopause = yes / ringinuse = no problem still up ? |
| 13:05.19 | Vec | http://pastebin.ca/393294 < can someone check whats going on, keep getting Unable to reconfigure channel and ignoring switchtype ? |
| 13:06.10 | kFuQ | voip-info.org down? |
| 13:06.55 | wunderkin | DrukenLPY, major packet loss |
| 13:07.03 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 13:08.09 | DrukenLPY | wunderkin: oh... shitty deal |
| 13:08.33 | DrukenLPY | although, that oculd just be a router on the fritz |
| 13:09.08 | wunderkin | that = fucked to me :D |
| 13:09.21 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 13:09.56 | DrukenLPY | yeah, but could also be lixed in 15 mins provinding the right guy notices |
| 13:10.11 | wunderkin | yeah.. |
| 13:11.00 | *** join/#asterisk lorinc (n=ang@pool-2896.adsl.interware.hu) |
| 13:22.29 | JunK-Y | some1 knows how to listen iax2 mini-frames via ethereal? |
| 13:24.27 | JerJer | i saw a perl script somewhere that claimed to write iax2 mini-frames to disk - then you could play them back with whatever player can deal with the codec used |
| 13:26.37 | JunK-Y | link? |
| 13:27.11 | Vec | Why would asterisk ignore singalling and switchtype in zapata.conf (it says its ignoring it) and then it says unable to reconfigure channels ? |
| 13:27.25 | JunK-Y | cause they're invalid? |
| 13:27.27 | Skaag | how do I get asterisk to forget peers that are no longer in the configuration? |
| 13:27.40 | JunK-Y | vec: you have to configure the zaptel.conf 1st |
| 13:27.42 | Skaag | I added some peer a few days ago and it's still trying to iax2 to that peer...! |
| 13:27.57 | Skaag | even though I removed it from iax.conf a long time ago |
| 13:28.00 | Skaag | and restarted the box even |
| 13:28.12 | Vec | JunK-Y : I have check http://pastebin.ca/393294 |
| 13:28.21 | Vec | Juggie-Y : I will try a reboot |
| 13:28.38 | JunK-Y | ran ztcfg -vvvv? |
| 13:30.34 | *** join/#asterisk Strom_M (i=strom@nat/digium/x-a3d16b54ad123a6e) |
| 13:31.01 | Vec | JunK-Y : cool, neve knew about that, I get Channel 01: Clear channel (Default) (Slaves: 01) for each channel |
| 13:31.45 | Vec | JunK-Y : still get the same error though |
| 13:32.19 | *** join/#asterisk af_ (n=getsmart@ip-202-133.sn2.eutelia.it) |
| 13:32.35 | JunK-Y | ur missing stuff in ur zaptel.comnf |
| 13:33.48 | *** join/#asterisk wunderkin (n=kev@ip72-208-1-190.ph.ph.cox.net) |
| 13:33.49 | TaiSHi | Mmm |
| 13:33.49 | TaiSHi | I have 3 agents, and there will probably get more calls than 3 |
| 13:33.49 | Vec | JunK-Y : any idea whats missing ? |
| 13:33.49 | JunK-Y | vec: signalling |
| 13:33.49 | TaiSHi | I've heard that there was a problem with autopause = yes / ringinuse = no |
| 13:33.50 | JunK-Y | pastebin all ur ztcfg output. |
| 13:34.04 | Vec | ok 1 sec, thanks for the help by the way |
| 13:34.29 | JunK-Y | just paypal me a good heineken! |
| 13:34.31 | JunK-Y | :P |
| 13:34.42 | JunK-Y | beer at 9am? not a good idea. |
| 13:35.26 | infi | there is no good heineken! |
| 13:35.54 | *** join/#asterisk puzzled (n=patrick@puzzled.xs4all.nl) |
| 13:37.44 | Vec | This is everything, http://www.pastebin.ca/393325. |
| 13:37.46 | puzzled | hi |
| 13:38.08 | *** join/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net) |
| 13:38.08 | *** mode/#asterisk [+o mog] by ChanServ |
| 13:38.28 | Vec | JunK-Y : unfortunately where I live paypal don't accept our money. |
| 13:38.39 | wunderkin | mawwwg |
| 13:38.41 | JunK-Y | pastebin ur stuff then :) |
| 13:39.39 | Vec | Sorry forgot http://www.pastebin.ca/393327 |
| 13:41.14 | JunK-Y | output when starting * ? |
| 13:41.15 | *** join/#asterisk jeedi (n=jeedi@t42.de) |
| 13:41.45 | jeedi | good morning :) |
| 13:42.34 | *** join/#asterisk genz (n=chatzill@im.jobdig.com) |
| 13:43.13 | genz | Anyone know how to delete a temporary greeting? |
| 13:43.43 | jeedi | uhm, maybe.. if you explain in what context you mean that.. |
| 13:45.01 | genz | jeedi: When you record a "temporary greeting", how does the user de-activate it? |
| 13:45.32 | JunK-Y | genz: just go in that menu |
| 13:45.37 | JunK-Y | option 2 is to erase it. |
| 13:46.33 | *** join/#asterisk juanjoc (n=juanjoc@200.69.219.113) |
| 13:46.40 | genz | *embarrassed* |
| 13:46.42 | *** part/#asterisk jeedi (n=jeedi@t42.de) |
| 13:47.02 | *** join/#asterisk jeedi (n=jeedi@t42.de) |
| 13:47.06 | jeedi | oops?! |
| 13:47.41 | genz | i thought it'd be on the message before it |
| 13:48.27 | Vec | JunK-Y : here is patebin of asterisk on startup http://www.pastebin.ca/393333 |
| 13:48.28 | genz | system says "press 4 to record your temporary message". going in there to erase it seems like hitting start to shutdown |
| 13:49.22 | JunK-Y | genz: you've your answer, thats another point. |
| 13:49.39 | genz | JunkK-Y: got it. thanks. |
| 13:49.56 | JunK-Y | Vec: so ur zap works now, no? |
| 13:50.12 | jeedi | i got a question a little more technical in nature.. i haven't found any instructions or hints on how to make multiple asterisk boxes write their CDRs to the same table on the same mysql database.. and neither have i been able to locate info on how to make the asterisk boxes "fail over" to a secondary database in case the connection to the primary db fails.. any ideas? |
| 13:50.13 | JunK-Y | # |
| 13:50.13 | JunK-Y | <PROTECTED> |
| 13:50.13 | JunK-Y | # |
| 13:50.14 | JunK-Y | <PROTECTED> |
| 13:50.45 | JunK-Y | type: zap show status |
| 13:51.24 | *** join/#asterisk fakhir (i=fakhir@unaffiliated/fakhir) |
| 13:51.46 | Turt|e | Is there a way to implement so there will be played an sound when i hit the hold button om my phone ? |
| 13:53.00 | Vec | JunK-Y : not sure when I type [Mar 13 15:52:23] WARNING[11713]: chan_zap.c:11067 process_zap: Ignoring signalling |
| 13:53.01 | Vec | [Mar 13 15:52:23] WARNING[11713]: chan_zap.c:11067 process_zap: Ignoring switchtype |
| 13:53.04 | Vec | sorry |
| 13:53.09 | Vec | I get that not sure why |
| 13:53.47 | *** join/#asterisk Cyber-Dogg (i=Cyber-Do@24-178-240-97.dhcp.stls.mo.charter.com) |
| 13:53.58 | JunK-Y | just dont pay attention for now. |
| 13:54.04 | JunK-Y | your zap works no? |
| 13:54.12 | *** join/#asterisk iamnowonmai (n=iamnowon@unaffiliated/iamnowonmai) |
| 13:54.26 | JunK-Y | Turt|e: u mean the music on hold? |
| 13:54.34 | JunK-Y | read configs/musiconhold.conf.sample. |
| 13:55.15 | *** join/#asterisk b11d (n=no@234-200-29-134.hcc.mnscu.edu) |
| 13:55.24 | b11d | hey all |
| 13:56.46 | *** join/#asterisk penguinFunk (n=penguin@87.224.86.46) |
| 13:57.21 | Vec | JunK-Y : looks like it doing some call tests will tell u what happens |
| 13:57.53 | JunK-Y | whats ur output of zap show status? |
| 13:58.27 | *** join/#asterisk jm|work (n=jm@sentry.flags.co.uk) |
| 14:00.00 | Turt|e | Junk-Y: yeah thats what i mean, i got this working as i playing music, but not when i hit the hold button |
| 14:01.24 | *** join/#asterisk Assid (n=assid@202.88.132.238) |
| 14:02.33 | Turt|e | Sorry, i messed up .. its working now |
| 14:03.07 | *** join/#asterisk mivck (i=1000@ip-70-228.telesat.com.co) |
| 14:05.06 | jeremy_g | hi |
| 14:05.07 | jeremy_g | :0 |
| 14:05.15 | jeremy_g | is it possible to use variables in sip.conf |
| 14:05.50 | jeremy_g | just like we use dialplan variables in extensions.conf |
| 14:08.12 | jeedi | yes. |
| 14:09.35 | jeedi | try something like "setvar=FOOBAR=123" |
| 14:10.26 | *** join/#asterisk tsurko (n=tsurko@vpn-pppoe-213-240-242-237.megalan.bg) |
| 14:11.02 | *** join/#asterisk quidpro (n=quid@CPE00131078ba5d-CM000f9f7eff1e.cpe.net.cable.rogers.com) |
| 14:12.17 | *** join/#asterisk codazoda (n=chatzill@mail.hurdmanivr.com) |
| 14:13.34 | *** join/#asterisk quidpro (n=quid@CPE00131078ba5d-CM000f9f7eff1e.cpe.net.cable.rogers.com) |
| 14:15.04 | codazoda | Oy, I'm lost without voip-info.org (which appears to be down). |
| 14:16.22 | codazoda | Does the TDM800P use the wctdm24xxp driver? |
| 14:16.33 | Vec | I confirm it looks like voip-info is down |
| 14:16.55 | *** join/#asterisk mercestes (n=merceste@rrcs-71-41-157-70.sw.biz.rr.com) |
| 14:18.32 | *** join/#asterisk anthm (n=anthm@CPE-72-131-113-50.wi.res.rr.com) |
| 14:18.32 | *** mode/#asterisk [+o anthm] by ChanServ |
| 14:20.05 | *** part/#asterisk jart (n=user@ool-43509aa5.dyn.optonline.net) |
| 14:21.33 | *** join/#asterisk marv[work] (n=timr@24.214.206.254) |
| 14:22.16 | codazoda | When configuring an FXO port, you specify "fxsks=1-8" (FXS Signaling) in zaptel.conf, correct? |
| 14:22.54 | Vec | I have a PRI line, and for some reason whatever number I dial my telco says does not exist, but I can recieve calls. |
| 14:23.30 | jeedi | Vec: hmm.. that sounds similar to a problem i had half a year ago |
| 14:23.42 | jeedi | Vec: what kind of PRI? e1/t1? what carrier? |
| 14:24.54 | JunK-Y | vec: pastebin a pri debug |
| 14:25.15 | JunK-Y | and verify with ur telco that u can outounds. |
| 14:26.05 | Vec | JunK-Y : can u tell me how do a pri debug :P |
| 14:26.31 | jeedi | Vec: on the asterisk prompt, set pri debug span <span-number> |
| 14:26.41 | Vec | Its an e1, carrier is telkom www.telkom.co.za :), they said that I should be able to do outbound calls, inbound works |
| 14:27.08 | jeedi | hmm, what hardware do you use for that? |
| 14:27.57 | *** join/#asterisk IPmonger (n=ipmonger@c-68-84-208-206.hsd1.pa.comcast.net) |
| 14:28.17 | mercestes | Vec: Outbound CallerID does not match within the block your telco gave you. Set your outbound callerID to something within their system. They'll block anything that dosen't match as an account in their system. |
| 14:29.57 | TaiSHi | I've heard that there was a problem with autopause = yes / ringinuse = no, anyone heard of it ? |
| 14:29.59 | *** join/#asterisk xtr-II (n=94752345@S0106000c41ed11e1.vf.shawcable.net) |
| 14:30.15 | b11d | hey Vec.. are you passing a 9 out to the telco by accident? |
| 14:30.18 | b11d | i had that " |
| 14:30.23 | b11d | issue" with my PRI at first :) |
| 14:30.29 | b11d | couldnt dial shit.. but could receive clals |
| 14:30.31 | *** join/#asterisk stkn|work (n=stkn|wor@gentoo/developer/pdpc.active.stkn) |
| 14:31.03 | jeedi | here (e1 from deutsche telekom) it was a problem with the wrong "pridialplan" variable in zapata.conf ;) |
| 14:31.24 | Vec | Here is my pastebin http://www.pastebin.ca/393366 |
| 14:34.49 | jeedi | Vec: line 18 looks suspicious.. maybe mercestes was right, and you need to set a correct CallerID(num) before dialing out |
| 14:34.51 | *** join/#asterisk ellisdee (n=ellisdee@mail.globalgeophysical.com) |
| 14:35.25 | mercestes | It's making it all the way to your telco and the call is initiated and the telco is hanging up on you. |
| 14:35.32 | mercestes | so your telco is rejecting the call. |
| 14:35.48 | *** join/#asterisk hohum (n=dcorbe@mercury.sunrocket.com) |
| 14:36.34 | b11d | you ARE doing a 'pri intense debug' right? |
| 14:36.39 | b11d | are you 100% sure you're sending a properly formatted number to the telco? |
| 14:37.09 | TaiSHi | mercestes: After all the blame and such |
| 14:37.21 | TaiSHi | I told another supervisor to tell the boss to get 3 G729 licenses |
| 14:37.23 | TaiSHi | Guess what |
| 14:37.33 | jeedi | b11d: looks like * took the "2116" part from his sip channel and passed it to the telco as the outbound clid.. definitely a reason to go BOINK |
| 14:37.52 | gambolputty | "Why don't we take the $30 g729 fee out of your paycheck?" |
| 14:38.18 | mercestes | TaiSHi: he fired you? |
| 14:38.21 | b11d | I guess it depends on the situation.... my telco just strips all CID info I send to them :( |
| 14:39.05 | jeedi | b11d: deutsche telekom simply ignores whatever you send them, and sets the clid on their own systems. |
| 14:39.13 | b11d | yeah, thats the same as here.. |
| 14:39.24 | b11d | jeedi.. let me come there and live.. Holland rocks |
| 14:39.30 | TaiSHi | mercestes: Nah, he did that yesterday |
| 14:39.36 | TaiSHi | He's gonna buy them today |
| 14:40.02 | b11d | oh wait, you're .de |
| 14:40.05 | b11d | still.. :) |
| 14:40.36 | jeedi | b11d: got java skills? linux admin experience? i assume you know your way around with asterisk stuff, so... send me your resume and CV ;) |
| 14:40.43 | b11d | :) mint! |
| 14:40.49 | quidpro | Speaking of G729 is there a way to get * to pass-through G729 *and* G711? From what i've read... you can only pass through properly if G729 is the only allowed codec |
| 14:41.00 | b11d | well.. I used to program in JAVA like, 7 years ago.. but im a BSD Man, not Linux :( |
| 14:41.27 | mercestes | TaiSHi: Great! |
| 14:42.10 | mercestes | jeedi: oooo...can I go too? I got everythign but Java skills...but I make up for it with Microsoft 133tn3$$. And awesome 133t$p3@k. |
| 14:42.28 | b11d | yeah and mercestes and I already have a strong rapport |
| 14:42.41 | codazoda | I'm getting the following error: "line 0: Unable to open master device '/dev/zap/ctl'" I believe this has something to do with udev, but I can't recall the fix and voip-info.org (where the fix is listed) is down. |
| 14:42.44 | codazoda | Any ideas? |
| 14:42.45 | mercestes | yea and....we have a good history too. |
| 14:43.19 | jeedi | mercestes: oh, M3y3kr0s0f7 1337N355? 5hw33t! |
| 14:43.20 | mercestes | quidpro: I don't think that's exactly right. By passthrough, do you mean passthrough with no transcoding?? |
| 14:43.23 | *** join/#asterisk giasai68 (n=giasai@ip-240-130.sn2.eutelia.it) |
| 14:43.28 | *** join/#asterisk MrTelephone (n=MrTeleph@h697179-171.picriverisp.net) |
| 14:43.28 | codazoda | I get that error when running "modprobe wctdm24xxp". I should also mention this is a tdm800p, which I believe uses the wctdm24xxp driver. |
| 14:43.36 | b11d | i just hit 100 days of running Asterisk problem-free.. |
| 14:43.55 | b11d | and i've only been running it 100 days.. so.. so far, no problems :) |
| 14:43.56 | jeedi | but for our current job opening, it's no java, no job. |
| 14:44.00 | mercestes | codazoda: Appears your zaptel drivers are not installed/loaded. Try a modprobe zaptel |
| 14:44.18 | MrTelephone | does someone have a tip on how to get polycom 501 working with the common linux ntp package? The phones seem to send packets but ntp doesn't respond.. I tried msntp and it seems to work more reliably but I don't really want to use it as it is too simple.. |
| 14:44.21 | b11d | jeedi.. is java a dutch word for marijuana? becuase in that case.. |
| 14:44.28 | b11d | i have a Ph.D in "java" |
| 14:44.29 | b11d | :) |
| 14:44.34 | jeedi | b11d: uhm, no. it's a dutch word for coffee ;) |
| 14:44.40 | b11d | ok well im down with that :) |
| 14:44.42 | b11d | i like coffee |
| 14:45.14 | giasai68 | hello |
| 14:45.19 | jeedi | you'd have to "suffer" through a whole day at the coffee museum here in hamburg to be only accepted for an interview ;) |
| 14:45.19 | b11d | hi |
| 14:45.31 | giasai68 | I'm using asterisk 1.4 with zapata 1.4 |
| 14:45.32 | b11d | yeah that'd be "rough" |
| 14:45.33 | b11d | :) |
| 14:45.56 | b11d | i'd love to be over that way though.. for real.. I'm getting a sour taste for North America. |
| 14:46.26 | codazoda | Okay, now when running modprobe wctdm24xxp I get "ZT_CHANCONFIG failed on channel 1: No such device or address (6)". |
| 14:46.33 | TaiSHi | mercestes: Now I will leech you for help (6) (msn emoticon) |
| 14:46.44 | giasai68 | I have configured digium te205p card and all work fine, but same calls dont have success, I have this error: Ext: 1 Cause: Temporary failure (41), class = Network Congestion (resource unavailable) (2) |
| 14:47.03 | giasai68 | error is: Ext: 1 Cause: Temporary failure (41), class = Network Congestion (resource unavailable) (2) |
| 14:47.13 | giasai68 | any idea how I can fix it? |
| 14:47.18 | giasai68 | pls, let me know |
| 14:47.27 | giasai68 | thanks in advance |
| 14:48.12 | b11d | brb all.. gotta go watch some gay web seminar bullshit.. |
| 14:48.13 | JunK-Y | ur telco cant route the call? |
| 14:48.41 | mercestes | TaiSHi: I'm really on MSN. |
| 14:48.53 | De_Mon | how would I setup a sip trunking provider with multiple sip gateways in asterisk? |
| 14:48.54 | jeedi | b11d|bbl: gay web seminar? |
| 14:48.55 | *** join/#asterisk telmich (i=telmich@gpm/telmich) |
| 14:49.09 | b11d|bbl | hahaha |
| 14:49.10 | jeedi | b11d|bbl: like "how to choo-choo on the man-train"? |
| 14:49.12 | b11d|bbl | ok.. no |
| 14:49.15 | b11d|bbl | HAHA |
| 14:49.36 | telmich | what is the best thing to read, if one wants to know more about asterisk routing (how to put which number where) and the digium e1 card? |
| 14:49.47 | jeedi | telmich? |
| 14:49.52 | b11d|bbl | nah, it's this web broadcast of some asshole trying to get people to "smarten uo |
| 14:49.53 | mercestes | codazoda: config issue. :P Read your directions. |
| 14:49.55 | b11d|bbl | "smarten up |
| 14:49.58 | b11d|bbl | dammit |
| 14:50.05 | b11d|bbl | "smarten up" with their passwords and the like.. |
| 14:50.05 | telmich | jeedi: yes? |
| 14:50.15 | *** join/#asterisk Cyon (n=cyon@216.179.31.170) |
| 14:50.21 | b11d|bbl | the message is good, the way they are doing it is bad.. |
| 14:50.39 | jeedi | telmich: if you are "the" telmich i'm thinking you are, the best thing you can do is come along for my asterisk workshop at easterhegg ;) |
| 14:50.41 | mercestes | giasai68: Sounds like you rrunning too many calls at once or your failing to hang up your channels. |
| 14:50.48 | codazoda | See my config at: http://www.pastebin.ca/393375. I didn't get any instructions with the card and voip-info.org seems to be down. I'm lost. |
| 14:51.00 | *** join/#asterisk Turt|e (n=danny@0x55532532.adsl.cybercity.dk) |
| 14:51.04 | mercestes | telmich: the book. |
| 14:51.04 | telmich | jeedi: if you are the jeedi I think you are, you should take care about your drinks |
| 14:51.07 | mercestes | ~bok |
| 14:51.09 | mercestes | ~book |
| 14:51.10 | jbot | from memory, book is a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 14:51.30 | jeedi | telmich: workshop during the day.. the bar opens at 22:00 |
| 14:51.50 | telmich | jeedi: will be there |
| 14:51.55 | jeedi | telmich: alright. |
| 14:51.57 | Vec | b11d : was not doing an intense debug, will do one and check, the telco is hanging up after a msg is played saying the number does not exist |
| 14:52.16 | mercestes | Vec: set your callerID to something valid. |
| 14:52.25 | telmich | jeedi: btw, personally I like yate more than asterisk, but I shouldn't say that in here, I guess :-) |
| 14:52.25 | giasai68 | mercestes: what's do u mean? |
| 14:52.40 | telmich | hehe |
| 14:53.10 | Turt|e | Can someone set an variable with the content af a system calls return data ? |
| 14:53.40 | mercestes | giasai68: say you can handle 5 calls simultaniously. It *looks* like you are trying to setup your 6th call so you are getting a congestion, network unavailable error. That can be done by running 6 simultanious calls, *or* by your channels not hanging up leaving them all falsely "open." |
| 14:53.53 | mercestes | giasai68: in my example, 6 = your max number of concurrent connections +1 of course. |
| 14:54.17 | mercestes | Turt|e: I think you have to use AGI for that sort of magic. |
| 14:54.48 | *** join/#asterisk ZaVoid (n=colin@65.244.210.46) |
| 14:55.01 | Turt|e | hmm |
| 14:55.04 | mercestes | Turt|e: I'm googling asterisk cmd system however to see what the wiki says |
| 14:55.05 | ZaVoid | anyone ever use a DUV 1000? |
| 14:55.07 | JunK-Y | define 'system calls return data' |
| 14:55.38 | *** join/#asterisk Gunnar (n=gunnar@nat.sigmasoft.com) |
| 14:55.51 | mercestes | JunK-Y: I think he wants to do a Set(dir=System(ls -1)) or something like that. |
| 14:56.05 | active_si | why is it better to have one BRI card with 4 S0 (8 channels) (HFC4S) than four cards with each one S0 (2 channels) (HFCS)? other than it takes only one PCI slot instead of four. |
| 14:56.34 | mercestes | ok is voip-info.org down? I might have to go back to being a stripper. |
| 14:56.39 | Turt|e | Junk-y: yeah like mercestes says |
| 14:56.40 | giasai68 | mercestes: thanks, but I'm sending just 1 call, and I can receive till 30 calls |
| 14:56.56 | JunK-Y | Turt|e: just drop the output in a file? |
| 14:57.04 | mercestes | active_si: It doesn't rape your system timing and kidnap all your available IRQs, ship them overseas, and sell them to druglords. |
| 14:57.23 | Turt|e | Junk-y no i need to get the output of a cmd into asterisk |
| 14:57.36 | ZaVoid | meh http://www.voip-info.org/ is down |
| 14:58.01 | JunK-Y | Turt|e: i dont think ya can do multi-lines vars. |
| 14:58.03 | ZaVoid | these duv 1000's are junk |
| 14:58.16 | mercestes | I will be answering no more questions today....as I really know nothing about this crap. I just google faster than everyone else. |
| 14:58.23 | _charly_ | hi, has anyone here connected an asterisk to a siemens hipath? |
| 14:58.25 | *** join/#asterisk Exhar (n=Roy@84-105-192-215.cable.quicknet.nl) |
| 14:58.39 | Turt|e | Junk-y: But if the cmd returns data of one line |
| 14:58.53 | JunK-Y | so drop in a file, then read the file back to insert with function ARRAY |
| 14:58.58 | JunK-Y | then ur fine |
| 14:59.18 | mercestes | Turt|e: in agi you can assign output to variables and then expose that variable to *. |
| 14:59.44 | Turt|e | i did the set(abc=system(onelinereturn)) but didnt really look like i worked .. ille try agin then |
| 14:59.57 | *** join/#asterisk reber (n=reber@cl-158.dub-01.ie.sixxs.net) |
| 15:00.33 | mercestes | Turt|e: ...Uh, I never said that would *work* only that was basically what you wanted to do. |
| 15:01.22 | *** join/#asterisk Gunnar (n=gunnar@nat.sigmasoft.com) |
| 15:02.37 | Turt|e | mercestes: yeah i know .. =) |
| 15:02.44 | mercestes | Turt|e: show application system does not show any way/method/switch to provide feedback save for the specified "success/failure" status message. |
| 15:04.23 | JunK-Y | http://svn.digium.com/view/asterisk/trunk/funcs/func_shell.c?view=markup |
| 15:04.27 | JunK-Y | use function SHELL |
| 15:05.04 | JunK-Y | Set(foo=${SHELL(echo \"bar\")}) |
| 15:05.37 | Turt|e | shell!! yeah baby |
| 15:05.51 | JunK-Y | enjoy! |
| 15:05.57 | Turt|e | thanks alot |
| 15:06.27 | mercestes | Go JunK-Y, go JunK-Y, go JunK-Y. nice. |
| 15:07.00 | *** join/#asterisk L|NUX (n=linux@unaffiliated/lnux/x-10290) |
| 15:07.22 | JunK-Y | i did nothing, just giving a link, relax. |
| 15:07.23 | *** join/#asterisk codefreeze (n=steve_mu@216.166.159.235) |
| 15:07.23 | *** join/#asterisk shido6 (n=shido6@d221-68-200.commercial.cgocable.net) |
| 15:07.23 | JunK-Y | :) |
| 15:07.51 | telmich | if someone _is_ connected and I type 'sip show peers', it should show all peers connected via sip, shouldn't it? |
| 15:08.33 | JunK-Y | if its a peer, ya |
| 15:10.45 | *** join/#asterisk mut (n=ana@65.111.222.120) |
| 15:10.49 | mut | anyone know of any billing software that breaks down different kinds of emi/cdr records? |
| 15:11.11 | *** join/#asterisk eltech (i=G00Ds@ool-457c93b6.dyn.optonline.net) |
| 15:11.12 | giasai68 | I have this error in debug: Ext: 1 Cause: Temporary failure (41), class = Network Congestion (resource unavailable) (2) |
| 15:12.10 | giasai68 | is it possible to fix? |
| 15:12.20 | JunK-Y | sure, talk to ur telco. |
| 15:12.33 | *** join/#asterisk Exhar (n=Roy@84-105-192-215.cable.quicknet.nl) |
| 15:13.06 | giasai68 | junk-y: only telco can fix? |
| 15:13.32 | *** join/#asterisk hyphen (n=hyphen@c-71-224-213-97.hsd1.pa.comcast.net) |
| 15:13.56 | JunK-Y | Network Congestion |
| 15:14.25 | giasai68 | junk-y: their can fix? |
| 15:14.51 | JunK-Y | yes, they can. |
| 15:15.14 | Dr-Linux | anybody is doing reporting from asterisk? |
| 15:15.39 | Dr-Linux | what i can grab to the db as unique ID |
| 15:16.05 | Dr-Linux | as zap channel often same in next call |
| 15:16.33 | *** join/#asterisk kanelbullar (n=kanelbul@83.240.200.92) |
| 15:16.52 | Assid | Dr-Linux: make a timestamp.. or add a field as auto increment |
| 15:18.38 | Dr-Linux | i also got the unixtime.id suggestion for another dude |
| 15:18.49 | Dr-Linux | Assid: so timestamp is always unique? |
| 15:18.58 | jeedi | hmm.. |
| 15:19.01 | jeedi | no. |
| 15:19.04 | Assid | nope |
| 15:19.07 | jeedi | timestamps are never unique. |
| 15:19.09 | Assid | auto increment id |
| 15:19.21 | jeedi | timestamp+channel+server-id would be unique. |
| 15:19.33 | tzanger | directed pickup with sip phones... where is the directed pickup "key sequence" defined? features.conf would be my first guess but that doesn't appear to be right |
| 15:19.49 | Dr-Linux | aww |
| 15:19.51 | jeedi | Assid: any experience with cdr storage in mysql? |
| 15:20.05 | Dr-Linux | jeedi: how can i grab that at in one variable? |
| 15:20.12 | Assid | jeedi: a little on pgsql |
| 15:20.20 | JunK-Y | channel+server-id would be no? |
| 15:20.20 | Dr-Linux | jeedi: is there any function or something available? |
| 15:21.07 | Dr-Linux | as this is only issue our 8 months worked domain is not being deployed |
| 15:21.17 | Dr-Linux | s/domain/application |
| 15:21.21 | jeedi | Dr-Linux: well.. you'd have to play around with the "userfield", i think. |
| 15:21.23 | *** join/#asterisk gr0mit (n=w10277@dhcp4.zuk40.mot-tools.co.uk) |
| 15:21.24 | *** join/#asterisk pagec (n=pagec@cpe-66-65-97-42.nyc.res.rr.com) |
| 15:21.36 | jeedi | i'm actually facing a similar problem.. |
| 15:21.37 | giasai68 | junk-y: can I try from my end to fix it? |
| 15:21.45 | Dr-Linux | i see |
| 15:21.47 | gr0mit | any idea what has happened to voip-info.org? |
| 15:21.59 | Dr-Linux | jeedi: so what you got an appropriate idea? |
| 15:22.02 | pagec | is there a way with Asterisk 1.4 to drop specific Zap channels? (i have calls on those channels that are going forever and i don't want to drop all calls) |
| 15:22.03 | JunK-Y | giasai68: no, call ur telco, like i told ya 3 times. |
| 15:22.06 | jeedi | Assid: any way to make one of the cdr->db modules use multiple databases? |
| 15:22.21 | jeedi | pagec: "soft hangup" ;) |
| 15:22.22 | JunK-Y | pagec: soft hangup zap/20-1 |
| 15:22.32 | pagec | ty |
| 15:22.37 | jeedi | np |
| 15:22.47 | *** join/#asterisk stoffell (n=stoffell@fw.catsanddogs.com) |
| 15:22.59 | Assid | jeedi: not sure |
| 15:23.13 | JunK-Y | jeedi: yes ya can have n dbs for cdrs. |
| 15:23.15 | Assid | jeedi: you could always just replicate your db |
| 15:23.18 | JunK-Y | type: cdr status |
| 15:23.41 | Assid | i wish someone would make my db structures for me |
| 15:23.45 | jeedi | Assid: i got two mysql-servers doing circular replication (master-master setup) |
| 15:23.56 | Assid | circular replication? |
| 15:24.00 | Assid | rsync? |
| 15:24.03 | jeedi | no. |
| 15:24.18 | JunK-Y | Assid: struct are all available. |
| 15:24.22 | jeedi | mysql5 allows two masters to sync each other.. |
| 15:24.38 | Assid | aah |
| 15:24.43 | Assid | need to try it one day |
| 15:24.47 | Assid | never needed it tho |
| 15:24.54 | jeedi | but i'd hate to have to find/build/write a mysql load balancer/fail-over system before deployment. |
| 15:25.03 | Assid | i just take daily snapshots |
| 15:25.26 | Assid | jeedi: load balancing if you are using php is pretty easy |
| 15:25.31 | ZaVoid | duv1000= junk |
| 15:25.35 | Assid | or if you want fail over |
| 15:25.44 | jeremy_g | Assid:how is that so with php? |
| 15:25.55 | jeremy_g | Assid: die(go to other backup server) |
| 15:25.57 | jeremy_g | ? |
| 15:26.00 | Assid | if it connects.. you use this connection.. ELSE .. connect to other server |
| 15:26.06 | Assid | add the server ips into an array |
| 15:26.07 | codazoda | I'm getting the error "ZT_CHANCONFIG failed on channel 1: No such device or address (6)" when I run 'modprobe wctdm24xxp'. I don't see any issues in my config. Here's a dump of my zaptel.conf and zapata.conf and modprobe commands: http://www.pastebin.ca/393397 |
| 15:26.26 | Assid | just +1 every time it fails a connection |
| 15:26.43 | *** join/#asterisk funxion (n=nunya@63.214.236.169) |
| 15:26.44 | pagec | and why wouldn't zap detect hang ups all the time. is there some setting that i am messing up? |
| 15:27.03 | Assid | make a dummy text file where it writes which server/array key it connected to last.. so it doesnt jump back every script |
| 15:27.38 | Assid | then when you rectify the problem.. you edit the text file and put the pointer back to 0/1 or whatever you want your default array key/server to be |
| 15:28.07 | Assid | if (!mysql_connect(.. |
| 15:28.31 | jeedi | Assid: uhm, this is not #php ;) |
| 15:28.44 | Assid | if you want to do load balancing.. thats pretty simple too. just use either round robin dns.. OR a rand() |
| 15:28.48 | Assid | oh yeah.. hehe |
| 15:29.13 | mercestes | mut: I do custom billing apps in .net. :) |
| 15:29.27 | jeedi | round-robin-dns? wtf? i don't use dns at all for my production setups.. i hate having dns fail and break all my stuff. |
| 15:29.51 | mut | mercestes: yea me too |
| 15:29.55 | mut | and its annoying |
| 15:29.59 | HarryR | Assid, round robin RAND() stuff is pretty crap for load balancing |
| 15:30.02 | mut | cause we keep changing ld carriers |
| 15:30.11 | Assid | you still have code to handle a non connecting servers |
| 15:30.20 | mercestes | mut: I love doing it. |
| 15:30.21 | Assid | HarryR: round robin OR rand |
| 15:30.29 | HarryR | I missed the OR out :) |
| 15:30.33 | mut | heh |
| 15:30.37 | Assid | balancing at script part |
| 15:30.43 | HarryR | Assid, linux vserver though (through Pirahna or similar) does a much better job through |
| 15:30.45 | mut | yea, i just finally started getting my qwest records |
| 15:30.46 | Assid | if you have the money.. get a real load balanser |
| 15:30.48 | funxion | anyone good with TDM channels? |
| 15:30.54 | mut | so i gotta get this system done by thursday |
| 15:31.57 | Assid | HarryR: was referring to script level if you have a hardware load balancer with heartbeat etc.. nothing likke it |
| 15:32.06 | HarryR | ah fair enough |
| 15:32.25 | HarryR | but still... AEL or dialplan is a nightmare |
| 15:33.13 | mercestes | mut: lol nice. |
| 15:33.34 | mercestes | Gotta love the "we sold this last week and need to bill in two weeks" billing system projects. |
| 15:33.47 | mut | its always like tath |
| 15:33.49 | mut | that* |
| 15:34.25 | mercestes | yea, same with the company I was at once upon a time. |
| 15:34.33 | jeremy_g | is it possible to have a backup gateway to register with in case one fails |
| 15:34.38 | jeremy_g | how in sip.conf can i specify |
| 15:34.49 | Assid | mercestes: can i have that |
| 15:34.52 | jeremy_g | i wish register => had fall back stuff |
| 15:34.52 | mercestes | jeremy_g: Just two register => statements I believe and then chain through your options. |
| 15:34.53 | Assid | i get tons of thse |
| 15:34.53 | mut | i'm hoping i can say that here soon |
| 15:35.06 | Assid | mercestes: sometimes the payment comes 2 months late |
| 15:35.12 | mut | my work day = last minute projects because of no foresight or planning |
| 15:35.18 | jeremy_g | mercestes:what do you mean by chaing through one's options |
| 15:36.03 | jeremy_g | my phones register with my sip server but what if it dies, i want another server to act as its hot backup |
| 15:39.45 | jeremy_g | obsolete i guess |
| 15:41.32 | funxion | does anyone know how to detect if a line is present on a TDM line? |
| 15:42.13 | *** join/#asterisk Igbothom_3rd (n=Hilton@office.quarkit.com.au) |
| 15:42.15 | *** part/#asterisk ZaVoid (n=colin@65.244.210.46) |
| 15:42.22 | JT | present? |
| 15:42.57 | *** join/#asterisk martineyles_ (n=martiney@adsl-w-234.as15758.net) |
| 15:43.10 | martineyles_ | hello |
| 15:43.10 | funxion | meaning if for some reason the carrier shut off the line it would then not be present |
| 15:43.23 | *** join/#asterisk ru_wing (i=wing@mars.tversu.ru) |
| 15:43.38 | martineyles_ | do you guys read the digium forums? |
| 15:44.02 | *** join/#asterisk supjigatr (n=syslod@152.53.16.10) |
| 15:44.06 | martineyles_ | I put a question up there |
| 15:44.07 | martineyles_ | http://forums.digium.com/viewtopic.php?t=13652 |
| 15:44.15 | funxion | so if I have a tdm400p with four separate carriers connected and one line goes dead how do i detect that it is dead and dial the call on the next available channel in the group |
| 15:44.26 | martineyles_ | but couldn't solve my problem |
| 15:44.53 | martineyles_ | so was hoping someone here might be able to help |
| 15:45.04 | supjigatr | Anyone have a quick fix for time on polycom soundpoints. There web seems to be down for that link. |
| 15:45.44 | funxion | martineyles_ i think our problems are similiar |
| 15:46.28 | JT | funxion: not sure if there's an easy method |
| 15:46.53 | funxion | doesnt have to be easy |
| 15:46.53 | E-bola | is there an installation howto/guide for asterisk tarballs*? |
| 15:47.06 | *** join/#asterisk jm|work (n=jm@sentry.flags.co.uk) |
| 15:47.51 | Dr-Linux | why one of my cisco phone doesn't ring? all works fine |
| 15:48.24 | funxion | JT I've been playing with all kinds of dialplan tricks but nothing seems to get it to roll over to the next channel |
| 15:48.44 | funxion | shouldnt it just detect is there is power on the line before it dials |
| 15:51.32 | *** join/#asterisk chiang_sg (i=chiang_s@121.7.131.44) |
| 15:52.26 | JT | it assumes lines aren't stuffed i guess |
| 15:52.43 | *** topic/#asterisk by Qwell[] -> Asterisk: The Open Source PBX -=- Asterisk 1.4.1 (Mar. 2, 2007), Asterisk 1.2.16 (Mar. 2, 2007), Zaptel 1.2.15 (Mar. 2, 2007) -=- Other fun channels: #asterisk-gui, #asterisknow, #asterisk-commits -=- Join #freepbx for freepbx/trixbox support. |
| 15:53.14 | chiang_sg | hi, i'm using asterisk 1.4 with OOH323 addons, when i try to make a call through this trunk, i got error "Comfort noise support incomplete in Asterisk" how do i disable this confort noise? i have no right to touch the h323 device |
| 15:53.18 | *** join/#asterisk elriah (i=elriah@adsl-072-149-159-016.sip.bhm.bellsouth.net) |
| 15:53.30 | elriah | Hi all. Does asterisk use sendmail to send voicemails? |
| 15:54.18 | funxion | how about voltage detection on a tdm card? |
| 15:55.36 | JT | chiang_sg: if you can't touch the h.323 device, there's nothing you can do |
| 15:56.01 | chiang_sg | jt: are there any effect with the call quality ? |
| 15:56.08 | sudhir492 | elriah: yes, asterisk uses sendmail client to send voicemails |
| 15:56.29 | sudhir492 | however, you can have any mailserver running there. |
| 15:56.33 | JT | chiang_sg: the fact that the other end is already doing silence supression means quality is being affected |
| 15:56.38 | Vec | Please can someonw help me out, whenever I try make an outgoing call on an E1 my telco plays a msg saying the number u have dialed does not exist, but everything looks fine, here is my pri debug of a call I made to 0118849751 - http://www.pastebin.ca/393453? |
| 15:56.39 | elriah | Got ya. Thanks. |
| 15:56.41 | sudhir492 | e.g. I have qmail instead of sendmail |
| 15:56.42 | mercestes | jeremy_g: Just set up two sippeers and register them both, say teliax and c-beyond, just hypothetically. |
| 15:56.54 | JunK-Y | elriah: you choose it with mailcmd= in voicemail.conf |
| 15:57.00 | JunK-Y | but by default, yes its sendmail |
| 15:57.06 | elriah | Ahh. Thanks. |
| 15:57.07 | JT | chiang_sg: however it may unnerve people a bit, as it will go completely silent when the other end detects silence or background noise |
| 15:57.16 | chiang_sg | i c |
| 15:57.20 | mercestes | jeremy_g: Then call a Dial(Sip/1234@teliax|30) on exten ,1, and a Dial(Sip/1234@cbeyond|30) on exten ,2, and it should facilitate the "hotswap" you are looking for. |
| 15:57.22 | elriah | Can you specify an SMTP server that may not be localhost? |
| 15:57.28 | chiang_sg | JT: thanks for the clarification |
| 15:58.03 | *** join/#asterisk qdk (n=qdk@80.243.125.204) |
| 16:01.38 | funxion | does anyone know where i can find the code for nvlinedetect |
| 16:02.06 | chiang_sg | how to set OOH323 to be able to handle more than 1 channel? |
| 16:03.05 | chiang_sg | say my dialplan is : _8.,1,Dial(OOH323/${EXTEN:1}@x.x.x.x,30,,Ttrf) |
| 16:04.02 | JunK-Y | some1 knows how to listen to ethereal iax2 mini-frame? |
| 16:06.11 | codazoda | I'm getting the error 'ZT_CHANCONFIG failed on channel 1: No such device or address (6)'. I believe that I recall the solution on voip-info.org, but that's down. I've got my configs, errors, /var/log/messages/ and ztcfg output here: http://www.pastebin.ca/393461. Any ideas? |
| 16:07.16 | Sweeper | Fast And Responsive Telephony System :D |
| 16:07.51 | cpm | good one! |
| 16:07.53 | bmd | FARTS? |
| 16:08.08 | cpm | <PROTECTED> |
| 16:08.12 | martineyles_ | Minesweeper ;-) It'll sell out in no time |
| 16:08.15 | Sweeper | I'm tempted, but I know it'd get shot down :( |
| 16:08.51 | martineyles_ | Spider solitaire for web 2.0 should sell well too |
| 16:09.12 | DrukenLPY | i like spider solitaire..... |
| 16:09.35 | DrukenLPY | can't beat it above 2 decks... but it passes time.... :) |
| 16:10.19 | codazoda | I can see your context names now. [farts-incoming], [farts-outgoing]... |
| 16:11.41 | Sweeper | funny thing is, all this was started by my quandry over what to call the svn repo XD |
| 16:11.44 | jeedi | Sweeper: what kind of project is it? |
| 16:12.05 | tzafrir | codazoda, what do you see on /proc/zaptel/* |
| 16:12.18 | tzafrir | does it match your /etc/zaptel.conf ? |
| 16:12.26 | codazoda | Nothing. /proc/zaptel is empty. |
| 16:12.43 | gr0mit | Vec: can you recommend a good isp in za with good performance on voip for connection to a UK server? |
| 16:12.52 | tzafrir | any gentoo user in the crowd? Can anybody give me a one-liner to build asterisk with bristuff support on gentoo? |
| 16:13.07 | mercestes | tzafrir: Sure |
| 16:13.08 | tzafrir | codazoda, so the drivers have not loaded |
| 16:13.18 | Sweeper | jeedi: I'm using asterisk + rails + realtime to create a redundnat, multi-site, fault-tolerant pbx network with advanced services such as call following and personnel-on-call |
| 16:13.45 | mercestes | tzafrir: USE="bri asterisk zaptel" emerge -av asterisk asterisk-addons bristuff zaptel asterisk-sounds |
| 16:13.50 | *** part/#asterisk chiang_sg (i=chiang_s@121.7.131.44) |
| 16:13.51 | Vec | gr0mit : u can, not sure how that will help me |
| 16:13.51 | codazoda | tzafrir, right. modprobe loads them, right? But, I get errors for some reason. |
| 16:13.53 | Sweeper | we're in southern luisiana, so "office X is gone" is something we've got to plan for ;) |
| 16:13.59 | mercestes | tzafrir: Something along those lines. |
| 16:14.05 | jeedi | Sweeper: what about "das telefon" ;) |
| 16:14.17 | Sweeper | blitzfon |
| 16:14.24 | cpm | Sweeper, where in so la? |
| 16:14.33 | jeedi | "i need to talk to the on-call admin in bumfuck, idaho" - "well, use DAS TELEFON, dammit." |
| 16:14.35 | Sweeper | cpm: all over it, and in texas too ;) |
| 16:14.41 | *** join/#asterisk phillipk (n=pkey@216.248.143.77) |
| 16:14.49 | gr0mit | Vec: i mean i am looking for a recommendation for an ISP in za to help out a friend who wants to make voip calls back to my server in .uk |
| 16:15.04 | Sweeper | specifically, lafayette, la rose, new orleans, and a few smaller satellite offfices |
| 16:15.08 | cpm | didn't know they had the intarweb in louisiana |
| 16:15.13 | jeedi | Sweeper: just kidding.. i really like your FARTS idea, tho ;) |
| 16:15.18 | Sweeper | :D |
| 16:15.21 | codazoda | tzafrir, I remember that this might have something to do with udev. But, I did run 'make install-udev' when I compiled zaptel. |
| 16:15.40 | tzafrir | codazoda, unrelated |
| 16:15.46 | Sweeper | I think I'll roll with it, and change it if anyone bitches |
| 16:15.52 | tzafrir | /proc/zaptel/* does not need udev |
| 16:16.08 | Sweeper | s/anyone/anyone\ that\ matters/ |
| 16:16.32 | codazoda | tzafrir, k, good to know. I'm also running CentOS 4.4. Could I be missing some packages? If so, I imagine the zaptel driver wouldn't have compiled. |
| 16:16.34 | Sweeper | jbot has bad syntax :( |
| 16:16.51 | tzafrir | so jbot does not escape very much Not a true sed |
| 16:17.02 | jeedi | of course, "DAS TELEFON" would spell out as "Distributed Asterisk System - Telephony Environment Locating Emergency Freaks On Nightshift" ;) |
| 16:17.17 | gr0mit | Vec: looking at your e1 issue, please can you pastebin your zaptel.conf and your zapata.conf files for me to look at |
| 16:17.51 | Sweeper | XD |
| 16:18.01 | gr0mit | Vec: and also pls disable intense debugging- this is only useful for layer 2 stuiff and your layer 2 is fine |
| 16:20.13 | Vec | gr0mit : I think I may have solved my prob |
| 16:20.25 | Vec | gr0mit : my telco was putting a 0 infront of the number for me |
| 16:21.12 | codazoda | Maybe my TDM808B is DOA. I'll drop in a TDM04B and see if that works. |
| 16:22.02 | gr0mit | well, you need to look at the zapata.conf and see what you have set for dial plan |
| 16:22.12 | gr0mit | i normally use 'unknown' |
| 16:22.18 | gr0mit | at least in uk that works |
| 16:24.07 | jeedi | when your telco prefixes a 0 for you automagically, try "pridialplan=local" |
| 16:25.15 | *** part/#asterisk ru_wing (i=wing@mars.tversu.ru) |
| 16:26.08 | *** join/#asterisk myiagy (n=myiagy@200.175.61.250.static.gvt.net.br) |
| 16:27.23 | *** join/#asterisk reber (n=reber@cl-158.dub-01.ie.sixxs.net) |
| 16:29.16 | jeremy_g | mercestes:i dont want to tell the user that we have different extensions |
| 16:29.21 | jeremy_g | mercestes:user should not know |
| 16:29.49 | funxion | does anyone use fxsgs on TDM? |
| 16:30.19 | mercestes | jeremy_g: I don't think the user would know. |
| 16:30.23 | tzafrir | what's gs anyway? |
| 16:30.25 | mercestes | jeremy_g: The "rollover" would be blind to the user. |
| 16:30.40 | funxion | groundstart |
| 16:30.48 | jeremy_g | mercestes:wont it be the user who has to press 2 |
| 16:30.52 | tzafrir | I know that name, but what does it mean? |
| 16:30.57 | mercestes | jeremy_g: it'd be priority 1 and 2 of the same extension to dial SIP@teliax and SIP@cbeyond respectively. |
| 16:31.06 | jeremy_g | mercestes:oh then its ok |
| 16:31.14 | codazoda | Does anyone here know if the TDM800P uses the wctdm24xxp drivers for sure? I replaced the TDM808B with a TDM04B and that one works fine (using the wctdm driver). |
| 16:31.17 | Skaag | would could cause REGAUTH problems with my iax connection? |
| 16:31.24 | Skaag | it's something that used to work fine for me |
| 16:31.31 | funxion | A method for seizing a phone line. Widely used by PBXs, the ring lead of the line (tip and ring) is momentarily connected to ground, and the CO detects the current. The CO grounds the tip lead of dedicated lines, which the PBX can test to determine if the line is dedicated for use. |
| 16:31.32 | tzafrir | codazoda, yes, it uses the same driver |
| 16:31.34 | Skaag | now i can no longer connect to my iax provider |
| 16:32.06 | mercestes | jeremy_g: exten => 7131234567,1,Dial(SIP/${EXTEN}@teliax) exten => 7131234567,2,Dial(SIP/${EXTEN}@cbeyond) |
| 16:32.15 | tzafrir | codazoda, you need a rather latest version (not sure if zaptel 1.2.14 supports it. 1.2.15 surely does) |
| 16:32.28 | jeremy_g | mercestes:yep yep, thanks man, um not that new to asterisk |
| 16:32.37 | jeremy_g | mercestes:rather um not at all new to asterisk |
| 16:32.52 | jeremy_g | :) |
| 16:33.30 | codazoda | I'm running zaptel-1.4.0 |
| 16:33.38 | funxion | anyone know why when I try to set my tdm trunk to fxsgs and fxs_gs asterisk wont start fails to load zap_tel.so |
| 16:33.44 | mercestes | jeremy_g: I know, in fact, I was kind of suprised at your question. But, somehow we weren't communicating so I figured a specific example would save some time. ;) |
| 16:33.59 | funxion | chan_zap |
| 16:34.00 | *** join/#asterisk [TK]D-Fender (n=joe@MTRLPQ02-1177745839.sdsl.bell.ca) |
| 16:34.28 | codazoda | My card must be DOA. |
| 16:34.42 | Corydon-w | funxion: probably because your zaptel.conf is not set up the same way |
| 16:34.51 | funxion | it is |
| 16:34.57 | *** join/#asterisk rkeels (n=chatzill@99.eedinc.com) |
| 16:35.00 | funxion | I set fxsgs is zaptel and zapata |
| 16:35.00 | Corydon-w | funxion: and it's highly unlikely that you have groundstart equipment |
| 16:35.08 | funxion | how so |
| 16:35.17 | funxion | Im connect to an inmarsat mini m |
| 16:35.28 | Corydon-w | loopstart has been ubiquitous since the 1950s |
| 16:35.41 | Corydon-w | funxion: what hardware are you running? |
| 16:35.53 | funxion | tdm405p |
| 16:36.03 | funxion | or something like that |
| 16:36.06 | Corydon-w | No such card |
| 16:36.08 | funxion | 400p |
| 16:36.18 | Corydon-w | which one? |
| 16:36.23 | funxion | 4 port fxo |
| 16:36.32 | jeremy_g | mercestes:you got it ;) |
| 16:36.35 | Corydon-w | That card does not support groundstart |
| 16:36.40 | funxion | o |
| 16:36.43 | funxion | that explains it |
| 16:36.46 | funxion | thnx |
| 16:37.12 | Corydon-w | About the only way you're going to get groundstart is with a TE110P to a channel bank |
| 16:37.23 | jeremy_g | In sip.conf, [general] setvar=FOOBAR=1234 |
| 16:37.24 | *** join/#asterisk markit (n=konversa@host119-245-static.72-81-b.business.telecomitalia.it) |
| 16:37.26 | funxion | I was afraid you'd say that |
| 16:37.38 | jeremy_g | and then referencing ${FOOBAR} later in sip.conf would work? |
| 16:37.42 | jeremy_g | asterisk 1.2 |
| 16:37.56 | jeremy_g | jeedi:are you sure it would work on 1.2 |
| 16:38.02 | [TK]D-Fender | jeremy_g: No, in the DIALPLAN |
| 16:38.11 | markit | any mISDN guru? I wouuld like to obtain a certain behaviour of my 1 port bri |
| 16:38.27 | Skaag | how do I debug an iax gateway that gives me UNREACHABLE? |
| 16:38.35 | jeremy_g | [TK]D-Fender: i want to use variables in sip.conf and jeedi told me that it would work that way |
| 16:38.37 | [TK]D-Fender | jeremy_g: Actually I don't think it applies under [general], only under a peer entry |
| 16:38.52 | Corydon-w | Skaag: iax2 debug |
| 16:38.57 | [TK]D-Fender | jeremy_g: And what else do your rice crispies say to you? :) |
| 16:39.09 | [TK]D-Fender | jeremy_g: There is nothing "variable" about sip.conf |
| 16:39.12 | Corydon-w | Skaag: it's probably a firewall issue |
| 16:39.20 | Skaag | Corydon-w: I do that, I see my REGREQ, REGAUTH, and the INVAL |
| 16:39.23 | *** part/#asterisk martineyles_ (n=martiney@adsl-w-234.as15758.net) |
| 16:39.27 | jeremy_g | [TK]D-Fender:in general section, i am just truing to initialize the variable. i ll use that in the peer or friends sections |
| 16:39.30 | Skaag | what ports do I need to open on which side, for IAX2 to work? |
| 16:39.36 | Skaag | I already opened 4569 tcp/udp |
| 16:39.37 | shido6 | 4569 |
| 16:39.41 | shido6 | or 5036 |
| 16:39.41 | [TK]D-Fender | jeremy_g: Not happening |
| 16:39.48 | Skaag | what's 5036? |
| 16:40.02 | shido6 | or whatever port you want to use, really.... 4569 is the default port for IAX2 |
| 16:40.40 | shido6 | 5036 was the old iax port |
| 16:40.52 | shido6 | 4569 UDP |
| 16:41.15 | Corydon-w | Skaag: if you're getting INVAL, sounds like your password is wrong |
| 16:41.30 | jeremy_g | [TK]D-Fender:its basically when i have 50 ips to register with and i have them as blocks of 5. Total 50 lines of register statements with 5 lines having same ip being registered. so having a variable would just imply changing 5 variables rather than those 50 :( lines |
| 16:41.32 | Skaag | it works fine on another asterisk setup |
| 16:41.41 | Skaag | my provider says the password is ok :( |
| 16:41.55 | Skaag | he says it's definitely something on my side |
| 16:42.08 | jeremy_g | why am i registering 5 times with same ip? |
| 16:42.15 | [TK]D-Fender | jeremy_g: Search & replace. |
| 16:42.30 | *** join/#asterisk CunningPike (n=CunningP@dhcp-10-234.district.north-van.bc.ca) |
| 16:42.30 | jeremy_g | basically 5 different user names |
| 16:42.36 | jeremy_g | for different service agreements |
| 16:42.48 | jeremy_g | [TK]D-Fender:ok |
| 16:42.59 | *** join/#asterisk thekidrio (n=thekidri@66.107.42.13) |
| 16:43.46 | *** join/#asterisk gbodemantv (n=gbodeman@corpex.pivotal.televerde.com) |
| 16:43.49 | gbodemantv | hello all |
| 16:43.57 | gbodemantv | I am having a Music on hold issue |
| 16:44.46 | Skaag | 070313-184421 NOTICE[6194]: chan_iax2.c:7791 iax2_poke_noanswer: Peer '4021' is now UNREACHABLE! Time: 0 |
| 16:44.54 | Skaag | gbodemantv: what kind? |
| 16:45.04 | gbodemantv | I have one box answer the calls and then IAX2 trunk to or main server in wich everyone is registered |
| 16:45.10 | *** join/#asterisk af_ (n=getsmart@ip-202-133.sn2.eutelia.it) |
| 16:45.16 | gbodemantv | up until last week it was all one box |
| 16:45.40 | gbodemantv | when a call is sent to MOH, it seems to want to play the music on the Iax2 channel |
| 16:45.43 | *** join/#asterisk Ahrimanes (n=ma@x1-6-00-0a-e4-2e-90-43.k707.webspeed.dk) |
| 16:46.57 | *** join/#asterisk RoyK (n=roy@ti211310a080-6495.bb.online.no) |
| 16:47.11 | gbodemantv | so no one hears it now |
| 16:47.43 | gbodemantv | it hought it was maube that I did not have asterisk addons and sounds compiled on the system that was answering calls |
| 16:47.46 | gbodemantv | but that did not work |
| 16:48.20 | *** join/#asterisk Ryushin (i=proxy@windwalker.openinnovations.com) |
| 16:50.10 | gbodemantv | calls goes PSTN - Server 1 (digium t1 card) -Iax2 to another box and sent to proper extension |
| 16:50.12 | gbodemantv | but none of the hold music works on queues or zpatel calls |
| 16:51.32 | af_ | indications.conf: it has effects only on directly connected to * lines ? like zaptel channels? |
| 16:52.00 | Vec | Anyone know what is up with voip-info ? |
| 16:52.33 | b11d|bbl | its down |
| 16:52.41 | *** join/#asterisk Assid (n=assid@59.183.30.78) |
| 16:52.43 | b11d|bbl | which isnt entirely unusual |
| 16:53.13 | gbodemantv | when I was going zap to sip on the same server I had no problems |
| 16:53.20 | b11d|bbl | for such a great site.. you'd think there'd be mirrors or something |
| 16:53.32 | gbodemantv | not that I am answering/calling with one and regostering with the other it does not want to work |
| 16:57.18 | *** part/#asterisk supjigatr (n=syslod@152.53.16.10) |
| 16:58.09 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 16:58.11 | *** join/#asterisk vooduhal (n=vooduhal@tc-proxy2.catt.com) |
| 16:58.51 | RoyK | are there any ipv6 support in asterisk yet? |
| 16:59.28 | b11d|bbl | i heard in 1.4 there was.. |
| 16:59.33 | b11d|bbl | no experience with it though |
| 16:59.48 | JunK-Y | RoyK: nope, in progress like the last time you ask :) |
| 17:00.08 | vooduhal | Does the Status (xx ms) column from "sip show peers" have any real significance? We have a phone we are testing for deployment, but everyone of these models show's 400ms on the same LAN segment while a SPA922 Linksys shows 20ms. |
| 17:00.18 | RoyK | JunK-Y: I beleive that was half a year ago or something :) |
| 17:00.36 | RoyK | I don't need it yet, though. just curious |
| 17:00.40 | *** join/#asterisk puzzled (n=patrick@puzzled.xs4all.nl) |
| 17:01.29 | *** join/#asterisk ChkDigit (n=mrw@static24-89-65-166.regina.accesscomm.ca) |
| 17:01.29 | *** join/#asterisk topping (n=topping@204.152.96.238) |
| 17:02.21 | [TK]D-Fender | vooduhal: Could be a crappy CPU or stack implementation on the phone. What model? |
| 17:03.47 | *** join/#asterisk angryuser (n=Miranda@i03v-213-44-169-43.d4.club-internet.fr) |
| 17:04.02 | angryuser | gooe evening |
| 17:04.06 | angryuser | ;) good |
| 17:04.45 | angryuser | i have a question bout asterisk ports, by defaul asterisk use 5060 tcp + 1000-2000 udp? |
| 17:04.59 | angryuser | 10000-20000 |
| 17:05.18 | vooduhal | That's what I was thinking, but I didn't know for sure what that delay really represented. |
| 17:07.01 | [TK]D-Fender | angryuser: 5060,10000-20000 ALL UDP |
| 17:07.10 | [TK]D-Fender | angryuser: * doesn't do SIP over TCP |
| 17:07.32 | DrukenLPY | [TK]D-Fender: can sip be done over tcp ? |
| 17:07.44 | angryuser | [TK]D-Fender: why so many udp ports? |
| 17:07.49 | [TK]D-Fender | DrukenLPY: Yes, and is so on other implementations. |
| 17:08.09 | [TK]D-Fender | angryuser: 1 port per simultaneous call. Just for breathing room. You can reduce this. |
| 17:08.11 | *** join/#asterisk drfreeze (n=Jim@www.freeze.org) |
| 17:08.17 | DrukenLPY | wouldn't that make sip better with nats? or am i just dreaming? |
| 17:08.34 | [TK]D-Fender | DrukenLPY: SIP isn't so much the problem as RTP <- |
| 17:08.40 | drfreeze | Hi |
| 17:09.01 | drfreeze | Anyone have a quick fix for the time change for Polycom 501 phones? |
| 17:09.06 | DrukenLPY | true... however isn't the rtp the same with sip and iax ? |
| 17:09.10 | [TK]D-Fender | DrukenLPY: Now IAX's strength is that the signalling & all voice is on the SAME port. |
| 17:09.12 | angryuser | [TK]D-Fender: lets say i have 10 maximum to outbound, can i fix this in rtp.conf like 10000-10010? |
| 17:09.16 | drfreeze | Do I have to update the firmware? |
| 17:09.23 | gambolputty | http://knowledgebase.polycom.com/kb/search.do?cmd=displayKC&docType=kc&externalId=10627&sliceId=SAL_PUBLIC_1_2&dialogID=1890871&stateId=1%200%201886835 |
| 17:09.25 | gambolputty | no |
| 17:09.46 | [TK]D-Fender | DrukenLPY: No, RTP is a completely seperate protocol. SIP is only a channel setup/teardown mechanism. IAX2 has its own audio encoding in its spec |
| 17:10.05 | [TK]D-Fender | angryuser: Sure, but I'll allow a few more just in case |
| 17:10.17 | DrukenLPY | which is why it's much better with nats |
| 17:10.27 | [TK]D-Fender | DrukenLPY: And does not use RTP |
| 17:10.30 | DrukenLPY | makes sence |
| 17:10.37 | [TK]D-Fender | DrukenLPY: Correct. |
| 17:10.44 | angryuser | [TK]D-Fender:ok thank you, it was for traffic shafting, i am unable to set a range;( |
| 17:11.25 | drfreeze | gambolputty: thanks for that bulletin |
| 17:13.00 | telmich | I am wondering, where to find out the correct values for zaptel.conf |
| 17:13.30 | telmich | I am having a te110p here, but I am not sure, which settings I have to set |
| 17:13.43 | *** join/#asterisk Gido-E (n=gido@lounge.datux.nl) |
| 17:13.56 | drfreeze | gambolputty: I'm not sure how to update the sip.cfg or ipmid.cfg files. Does this process assume I have tftp setup and running? |
| 17:14.33 | Gido-E | is there a way to retrieve sip status or other statusses with a default make-up? |
| 17:15.26 | [TK]D-Fender | drfreeze: It assumes you are running a provisioning server of SOME kind, and IPMID is only for MGCP and ancient SIP releases |
| 17:19.24 | Vec | Is the way I can use pattern matching 21XX = Dial(blah blah) for incoming calls for DID ? |
| 17:21.31 | drfreeze | [TK]D-Fender: is there another way to get the phones to display the right time? |
| 17:21.42 | drfreeze | other than changing their gmt offset. :) |
| 17:22.40 | JunK-Y | vec: _21XX,1,Dial(blahlbah) |
| 17:23.34 | [TK]D-Fender | drfreeze: typicall no, you have to fix the timezone settings, and thats where |
| 17:23.56 | b11d|bbl | hahaha.. it's like no one knew that the DST stuff was changing.. |
| 17:24.09 | [TK]D-Fender | drfreeze: Unless you want to cheat and change your time-zone temporarily or run an alternative time server..... |
| 17:24.43 | [TK]D-Fender | b11d|bbl: load chan_ignorance.so |
| 17:24.48 | b11d|bbl | :) |
| 17:25.15 | [TK]D-Fender | Error: Module already loaded |
| 17:25.30 | [TK]D-Fender | set stupid=very |
| 17:25.34 | [TK]D-Fender | OK |
| 17:25.39 | b11d|bbl | lol |
| 17:25.53 | [TK]D-Fender | </sarcasm> |
| 17:25.58 | *** join/#asterisk xtr-II (i=01928375@S0106000c41ed11e1.vf.shawcable.net) |
| 17:28.19 | *** join/#asterisk _Vile (n=vile@bc182112.bendcable.com) |
| 17:28.44 | gr0mit | vec: exten=> _21XX,1,Dial(SIP/${EXTEN:2} or whatever |
| 17:29.07 | *** join/#asterisk puzzled (n=patrick@puzzled.xs4all.nl) |
| 17:31.07 | b11d|bbl | ayoohhhh |
| 17:33.30 | Vec | unfortunately that does not work |
| 17:33.30 | Vec | it says 2122 does not exist in context blah |
| 17:33.30 | JunK-Y | vec: read the doc a bit and you'll be able. |
| 17:33.30 | gr0mit | are you sure your telco is sending you 4 digits? |
| 17:33.30 | JunK-Y | add this in extensions.conf (in ur context blah) exten => 2122,1,Playback(tt-monkeys); |
| 17:33.31 | JunK-Y | and reload |
| 17:33.54 | Vec | it works if I use a hard number like 2122 but not if I use _21XX |
| 17:34.42 | JunK-Y | output of: dialplan show 2122@blah |
| 17:34.58 | JunK-Y | probably cause you're in the wrong cotnext |
| 17:35.19 | mafkees | hey JunK-Y :) |
| 17:35.48 | JunK-Y | lunch break's over. |
| 17:36.39 | [TK]D-Fender | Vec: Pastebin your entire context please.... |
| 17:36.41 | [TK]D-Fender | ~pb |
| 17:36.50 | jbot | rumour has it, pb is a place to paste your stuff without flooding the channel - try http://pastebin.ca, or http://channels.debian.net/paste |
| 17:41.52 | mihinomenest | is there a way to set the "Expires:" SIP header value when I register with my provider? |
| 17:42.41 | gr0mit | Vec: paste the line in question into www.pastebin.ca |
| 17:42.49 | gr0mit | we can take a look |
| 17:43.03 | mercestes | I didn't hear about the DST crap until about 2 days before it happened. |
| 17:44.49 | drfreeze | Is there a way to query the time a Polycom phone is displaying from within asterisk or from the commandline? |
| 17:45.53 | *** join/#asterisk ToyMan (n=Stuart@mcha-aj-74-209-16-160.taconic.net) |
| 17:50.58 | JunK-Y | drfreeze: not that i know, if ya find way, msg me that way ;) |
| 17:51.05 | drfreeze | JunK-Y: sure |
| 17:54.20 | *** join/#asterisk Ebola (n=Ebola@host86-143-156-147.range86-143.btcentralplus.com) |
| 17:54.48 | *** join/#asterisk awannabe (n=gti@ip24-251-135-202.ph.ph.cox.net) |
| 17:55.07 | [TK]D-Fender | drfreeze: No. |
| 17:56.53 | *** join/#asterisk Vec (n=Vec@dsl-242-252-39.telkomadsl.co.za) |
| 17:57.08 | awannabe | does anyone know why call parking stop works? we dial the DTMF tones, but * wont park the call, and no messages or anything on the console appear |
| 17:57.37 | drfreeze | [TK]D-Fender: bummer |
| 17:58.04 | *** join/#asterisk diclophis-work (n=jbardin@65.203.37.58) |
| 17:58.06 | diclophis-work | hello all |
| 17:58.10 | *** join/#asterisk caio1982 (i=caio1982@CAcert-br/caio1982) |
| 17:58.23 | diclophis-work | i am wondering what type of "pid check" i need to do to ensure asterisk is running on a system? |
| 18:00.20 | *** join/#asterisk nextime (n=nextime@unaffiliated/nextime) |
| 18:00.26 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 18:00.38 | nextime | hi. Where i can find a detailed doc about user.conf file? |
| 18:00.58 | awannabe | is it bad to have to do hourly reloads of asterisk? lol |
| 18:03.05 | *** join/#asterisk awk (n=phil@vc-196-207-45-253.3g.vodacom.co.za) |
| 18:04.41 | Vec | diclophis-work : take a look at daemontools |
| 18:05.08 | *** join/#asterisk FinboySlick (n=Miranda@207.134.8.202) |
| 18:06.01 | diclophis-work | daemontools? |
| 18:06.03 | diclophis-work | is that a linux thing?> |
| 18:09.06 | FinboySlick | Hello gang. Can zaptel groups overlap? I have an fxo card with three lines. One number we try to keep free all the time for incoming, one is the fax line which we mostly use for outgoing (but we'd want to pick up with asterfax), and the third is a home line we want in the same 'outgoing' group as the fax line in case the fax line is busy, but we don't want asterisk to touch it for incoming. Problem is that with lines 2 and 3 in the outgoing group |
| 18:09.06 | FinboySlick | <PROTECTED> |
| 18:10.35 | *** part/#asterisk nextime (n=nextime@unaffiliated/nextime) |
| 18:10.41 | [TK]D-Fender | drfreeze: Time to get off your ass and provision them like you're supposed to... |
| 18:10.50 | vooduhal | Could someone point in the direction of documentation of what the "sip show peers" status value actually means? |
| 18:13.54 | *** join/#asterisk ChkDigit (n=mrw@static24-89-65-166.regina.accesscomm.ca) |
| 18:14.01 | Assid | damn.. i cant connect to thjis box.. it just keeps sending register and nothing happens |
| 18:14.57 | *** join/#asterisk boch (n=fran@190.48.211.170) |
| 18:14.58 | [TK]D-Fender | Assid: Sounds like NAT issues.... |
| 18:15.01 | FinboySlick | Actually, to simplify my question: Can the same zaptel line be in two different contexts? |
| 18:15.16 | boch | is the voip-info.org wiki down ? |
| 18:15.29 | [TK]D-Fender | boch: Looks like |
| 18:15.50 | boch | damn, im addict to that site |
| 18:17.45 | mercestes | [TK]D-Fender: Yea, I had to put in my two weeks because I don't actually know anything.....except how to google the wiki |
| 18:17.51 | vooduhal | Ok, if you can't point me to the docs, doesn't anyone know what the status delay (xx ms) from "sip show peers" actually means specifically? |
| 18:17.55 | Qwell[] | mercestes: :P |
| 18:18.12 | Assid | [TK]D-Fender: works 1 day .. doesnt the next? |
| 18:18.21 | [TK]D-Fender | mercestes: When/where was this? And how bad were their problems? |
| 18:18.36 | [TK]D-Fender | Assid: Sure.. if an IP changes, etc... |
| 18:18.47 | *** join/#asterisk spatulamaan (n=ggilmore@207.188.8.252) |
| 18:18.56 | mercestes | [TK]D-Fender: Thinking of putting in yoru resume? :D |
| 18:18.56 | Assid | have a sip reload and evertything |
| 18:20.34 | *** join/#asterisk awk (n=phil@vc-196-207-45-253.3g.vodacom.co.za) |
| 18:23.59 | *** join/#asterisk robl^ (n=robl@pdpc/supporter/monthlybyte/robl) |
| 18:27.08 | awk | tell me something, I have an asterisk box for a local company, that box has a dial prefix of say 1 that will have an outbound sip call to a carrier than will then link up to the pstn again in the same country |
| 18:27.19 | awk | should there be a lagg in call |
| 18:27.28 | awk | like hello, wait a couple seconds and then I hear it? |
| 18:38.13 | *** join/#asterisk eltech (i=G00Ds@ool-457c93b6.dyn.optonline.net) |
| 18:39.58 | [TK]D-Fender | mercestes: I have no plans of crossing your border, let alone becoming a citizen thank-you :) |
| 18:40.10 | [TK]D-Fender | mercestes: Just wondering ho far things had gotten.... |
| 18:40.21 | *** join/#asterisk DocHolliday (i=RgRabbit@gateway/gpg-tor/key-0x0E4F6D6C) |
| 18:40.45 | DocHolliday | is it possible to dial an extension and have asterisk record something in the GSM format? |
| 18:41.14 | Strom_M | DocHolliday: yes...and in wav or ulaw too |
| 18:41.30 | [TK]D-Fender | DocHolliday: "show application record" or in 1.4 "core show application record" |
| 18:42.00 | awk | anyone had this issue, I have a trixbox, and added a sip trunk to another machine that has access to the pstn, it passes the call I can see it, my cell phone i'm phoning rings, I answer, yet the system keeps ringing, and doesn't actually connect the call |
| 18:42.25 | bkruse | awk: #trixbox |
| 18:42.26 | awk | the other machine is an asterisk box, I see the sip connection taking place and it is dialing out from the asterisk box, it just isn't connecting the calls? |
| 18:42.39 | awk | well what makes you think its the trixbox causing the issue? |
| 18:42.45 | DocHolliday | [TK]D-Fender, how can i execute the record function from my dial plan? |
| 18:42.48 | bkruse | its trixbox |
| 18:42.51 | bkruse | do i really need to say more? |
| 18:43.18 | awk | you have no proof its the trixbox, i'm asking for an explenation of what might be the problem? |
| 18:43.30 | awk | lets pretend its 2 asterisk boxes |
| 18:43.40 | bkruse | i do have proof. its trixbox |
| 18:43.41 | bkruse | no |
| 18:43.43 | bkruse | lets not pretend |
| 18:43.45 | bkruse | download asterisk. |
| 18:43.52 | bkruse | what version of asterisk is it anyways? |
| 18:43.57 | awk | what could I have a look at to see why it connecting the call |
| 18:44.00 | awk | 1.2 |
| 18:44.05 | bkruse | 1.2............ |
| 18:44.05 | bkruse | what |
| 18:44.07 | [TK]D-Fender | awk: Probably NAT issues |
| 18:44.11 | bkruse | 1.2.-9? |
| 18:44.30 | bkruse | awk: sip.conf nat=yes.......or are you doing this all through the "webgui" |
| 18:44.52 | [TK]D-Fender | bkruse: DUH its being done through the GUI (at least on the FreePBX side...) |
| 18:44.59 | [TK]D-Fender | :) |
| 18:45.00 | bkruse | [TK]D-Fender: lame... |
| 18:45.09 | awk | no gui, I have around 2000 minutes routing through the box a night |
| 18:45.11 | bkruse | teh gui clicky! yay! |
| 18:45.29 | awk | international traffic works perfectly |
| 18:45.45 | bkruse | im supposed to believe you have 2000 minutes(wtf kind of measurement is that?) and you dont know how to configure nat problem?? |
| 18:45.50 | awk | its just this sip trunk from the trixbox that is causing issues |
| 18:46.01 | *** join/#asterisk tzafrir_laptop (n=tzafrir@bzq-88-154-8-247.red.bezeqint.net) |
| 18:46.16 | [TK]D-Fender | awk: Check your nat settings on both sides, and your firwall, etc.... |
| 18:46.18 | awk | firstly you say its trixbox now you saying its nat? |
| 18:46.40 | awk | no firewall rules that are cuausing issues, I just told you its routing through and the call is being dialed out |
| 18:46.47 | bkruse | awk: actually, its a combination of the two. |
| 18:46.50 | awk | its just recently that it isn't allowing pickup |
| 18:47.03 | [TK]D-Fender | awk: *I'M* saying its likely a NAT issue. Trixbox doesn't CAUSE the problem, it just lets you be ignorant about what BELONGS in there. |
| 18:47.18 | [TK]D-Fender | awk: That isn't enough to rule it out. |
| 18:47.19 | bkruse | [TK]D-Fender: thats good, i like it |
| 18:47.41 | awk | bkruse: you don't seem to know what you talking about, as you seem to speculate and not give me any actual proof to what you are saying. |
| 18:48.04 | awk | [TK]D-Fender: I understand. let me monitor the traffic a bit further and see where its breaking |
| 18:48.12 | bkruse | awk: thats right, i have no idea what im talking about.... |
| 18:48.17 | bkruse | but im not using trixbox. |
| 18:48.23 | [TK]D-Fender | awk: Pastebin the [general] sections (ALL LINKED BITS TOO) for BOTH sides please, and writeup the path betweent he two boxes. |
| 18:48.44 | [TK]D-Fender | bkruse: Me neither :) (as far ass using FreePBX that is...) |
| 18:48.49 | bkruse | agreed. |
| 18:48.50 | bkruse | ~pb |
| 18:48.51 | jbot | i heard pb is a place to paste your stuff without flooding the channel - try http://pastebin.ca, or http://channels.debian.net/paste |
| 18:48.56 | [TK]D-Fender | bkruse: I just know what it DOES, and the kinds of people using it. |
| 18:49.10 | bkruse | [TK]D-Fender: same, thats why i make crazy generalizations. |
| 18:49.15 | bkruse | awk: ill help, i just hate trixbox |
| 18:49.25 | awk | bkruse: well I sell VOIP services, you sell what the client wants |
| 18:49.32 | bkruse | awk: true |
| 18:49.37 | bkruse | asterisk has a gui, if you didnt know |
| 18:49.41 | awk | they want a gui interface, I suppose I could stick 1.4 and use the new GUI |
| 18:49.55 | bkruse | awk: true, the way the gui works, its impossible to port to 1.2 |
| 18:49.58 | awk | but its all the reporting, etc that trixbox does they want. |
| 18:49.59 | bkruse | so sure, i see where your coming from |
| 18:50.10 | [TK]D-Fender | awk: What I expect for the network pasth is something like "Asterisk1 (Trixbox) -> D-Link 12345 NAT router -> internet -> Someother NAT router -> othernormal Asterisk, etc... |
| 18:50.13 | awk | and they like the crm package it ofers |
| 18:50.14 | bkruse | awk: our gui uses just javascript, its light, and is not server dependent. |
| 18:50.27 | bkruse | if they gave me php and mysql and let me loose, of course we could easily do much more. |
| 18:50.39 | bkruse | but it would take away from the whole profile we started with, and intend on keeping |
| 18:50.42 | bkruse | = |
| 18:50.51 | [TK]D-Fender | awk: Ok, lets drop the whole FreePBX thing ok? If you're going to be/get any help, please provide the information I have requested/. |
| 18:51.07 | bkruse | http://asterisknow.org/image |
| 18:51.10 | awk | [TK]D-Fender: how it works, natted asterisk box, dynamic dns (passing out from the trixbox) to a dedicated no natted server |
| 18:51.21 | awk | err natted freepbx box |
| 18:51.34 | [TK]D-Fender | awk: Very well, now the pastebin please. |
| 18:51.55 | bkruse | ~pb |
| 18:51.56 | jbot | it has been said that pb is a place to paste your stuff without flooding the channel - try http://pastebin.ca, or http://channels.debian.net/paste |
| 18:51.59 | bkruse | ;] |
| 18:52.04 | awk | ok, let me read what I must pastebin |
| 18:55.50 | *** join/#asterisk Malph (n=chatzill@66-231-0-194.hosts.sdnet.net) |
| 18:56.26 | [TK]D-Fender | bkruse: Every now and again I think of making little GUI bits for * configuration, but nothing "complete". |
| 18:56.44 | bkruse | [TK]D-Fender: ive got a lil something something in the works |
| 18:56.46 | [TK]D-Fender | bkruse: Because "complete" is often followed by "garbage" :) |
| 18:56.51 | bkruse | a couple additions to our gui, but youll have to wait and see ;] |
| 18:57.06 | [TK]D-Fender | bkruse: I haven't really seen it at all yet ;) |
| 18:57.18 | bkruse | [TK]D-Fender: maybe you can beta it or something |
| 18:57.19 | bkruse | :D |
| 18:57.20 | bkruse | [TK]D-Fender: exactly, when you try to mimic ALL of asterisk features in a single, tab filled, website?? |
| 18:57.23 | [TK]D-Fender | bkruse: Logged in ONCE, and have seen a few screen-shots. users.conf = asjhgdkjhsdagjksdhgf |
| 18:57.43 | [TK]D-Fender | bkruse: You mean like where I work? ;) |
| 18:57.47 | bkruse | [TK]D-Fender: i have to say i agree, it goes against all convention i learned growing up on asterisk, but i guess its useful to some |
| 18:58.09 | [TK]D-Fender | bkruse: Yeah if you don't have in-house staff and change things often enough... |
| 18:58.11 | awk | nearlly done |
| 18:58.18 | bkruse | awk: yay |
| 18:58.46 | bkruse | [TK]D-Fender: agreed....the gui is useful for a good bit of things, but i cannot see an professional using it on an enterprise level, but who knows, i am a nub ;] |
| 18:59.14 | [TK]D-Fender | bkruse: Well then again, few entrprises would use * as a straight PBX w/o a GUI.... |
| 19:01.57 | robl^ | hrmmm. the new SLA features look nice! I might try to add them to my setup |
| 19:02.19 | bkruse | [TK]D-Fender: think so? |
| 19:02.25 | *** part/#asterisk Burgwork (n=corey@ubuntu/member/burgundavia) |
| 19:02.32 | awk | i'm enjoying my new realtime setup |
| 19:02.43 | [TK]D-Fender | bkruse: Those without in-house competant Linux/Asterisk people that is. |
| 19:02.45 | awk | iax, sip, voicemail, and cdr |
| 19:03.01 | bkruse | [TK]D-Fender: oh ya, well, i would think enterprise solutions WOULD have in-house technicians for that, but maybe not |
| 19:03.04 | awk | http://channels.debian.net/paste/5708. |
| 19:03.04 | [TK]D-Fender | awk: And concentrating HARD on that pastebin as well I see :) |
| 19:03.08 | bkruse | [TK]D-Fender: i would think it would be pure sip, and call balancing mostly, openser, yay |
| 19:03.50 | bkruse | canreinvite=no, wewt |
| 19:03.56 | [TK]D-Fender | awk: From what you showed me your FreePBX side has NONE of the NAT settings required to operate. |
| 19:04.17 | bkruse | [TK]D-Fender: yep, that would explain the half-workingness |
| 19:04.38 | [TK]D-Fender | awk: and I am NOT looking at the [general] section as requested. |
| 19:04.40 | awk | ok, let me look at how to do that with FREEPBX :P |
| 19:04.46 | awk | everyime I modify the dam sip.conf it gets overwriten with freepbx |
| 19:05.07 | [TK]D-Fender | awk: Look at HOW?! you grab 2 silly config files and you PASTE THEM |
| 19:05.11 | bkruse | awk: ya, that doesnt surprise me, its the only way (really) to write configurations without stacking up |
| 19:05.14 | [TK]D-Fender | :) |
| 19:05.18 | awk | 1 sec |
| 19:05.18 | bkruse | [TK]D-Fender: yep ;] |
| 19:05.35 | bkruse | [TK]D-Fender: hes setting tons of variables lol |
| 19:05.40 | [TK]D-Fender | unload chan_bile.so |
| 19:06.08 | [TK]D-Fender | bkruse: And not even the likely incriminating SIP debug enabled ;) |
| 19:06.23 | bkruse | [TK]D-Fender: never |
| 19:06.29 | bkruse | does freepbx have sip debugging in the gui!? |
| 19:06.30 | bkruse | :P |
| 19:06.33 | bkruse | woah, port 55001 |
| 19:06.33 | [TK]D-Fender | bkruse: Heaven forbid... |
| 19:06.34 | awk | http://channels.debian.net/paste/5709. |
| 19:06.48 | awk | bkruse: hmf, not that i've come across |
| 19:07.08 | bkruse | awk: it doesnt |
| 19:07.13 | *** join/#asterisk ping2921 (n=marc3234@206-248-134-179.dsl.teksavvy.com) |
| 19:07.28 | bkruse | are you really doing ilbc? brave man you are. |
| 19:07.32 | ping2921 | is there a text2speech function in asterisk? |
| 19:07.41 | bkruse | ping2921: festival |
| 19:07.48 | Corydon-w | No, but there's Cepstral |
| 19:07.56 | bkruse | or its proprietary equivilent(by the same person...) ^^ |
| 19:08.17 | bkruse | fyi, if you have the cash, cepstral sounds WAY better....obviously |
| 19:08.28 | Corydon-w | the cash == $30 |
| 19:08.35 | bkruse | Corydon-w: thats it? not bad |
| 19:08.38 | awk | no, using gsm for that trixbox at the moment, but getting a huge delaying passing through the box when I had it working |
| 19:08.47 | awk | it passes the call, I talk wait, and wait then it comes through |
| 19:08.47 | bkruse | awk: hmm |
| 19:08.51 | awk | going to try g729 |
| 19:08.53 | Corydon-w | bkruse: Dog is $7 |
| 19:08.54 | ping2921 | I would like to hear the incoming callerid. |
| 19:09.08 | bkruse | Corydon-w: nice |
| 19:09.29 | awk | as that freepbx box is on a 4mb link, but something like 256kb downstream |
| 19:09.35 | [TK]D-Fender | awk: Congratulations if thats it you're DOA. Go read a guide on how to set that up |
| 19:09.42 | bkruse | ping2921: hear it? sweet....ya, exten => omg,n,saytext(${CALLERID(num)}) |
| 19:09.51 | awk | so not sure if thats causing the issue |
| 19:09.53 | bkruse | eww! |
| 19:10.04 | bkruse | LOL |
| 19:10.08 | *** join/#asterisk Seba_soy (n=s@200.110.218.146) |
| 19:10.12 | Seba_soy | hello :) |
| 19:10.14 | bkruse | awk: dont think its going to help, i think its your crazy dialplan thats taking the call SO long to setup...... |
| 19:10.17 | bkruse | Seba_soy: sup |
| 19:10.17 | [TK]D-Fender | awk: 256kbit downsteam is BAD. 2-3 calls tops |
| 19:10.21 | Seba_soy | I will post my question: |
| 19:10.24 | [TK]D-Fender | awk: Assuming G.729 |
| 19:10.30 | awk | and gsm ? |
| 19:10.35 | awk | 1? |
| 19:10.35 | bkruse | awk: worst. |
| 19:10.39 | [TK]D-Fender | awk: Yeah, maybe GSM |
| 19:10.50 | awk | still the quality is good |
| 19:10.53 | awk | just the delay |
| 19:10.54 | bkruse | g729 sound quality way > than gsm |
| 19:10.59 | bkruse | but sucks for MOH |
| 19:11.01 | [TK]D-Fender | bkruse: No.... for wosrt we still have LPC10. Domo Arigato! |
| 19:11.02 | awk | so the compression shoulnt speed up the delay |
| 19:11.14 | bkruse | [TK]D-Fender: LOL, i know exactly what you mean |
| 19:11.19 | [TK]D-Fender | bkruse: I've heard mixed reviews between G.729 & GSM |
| 19:11.22 | awk | i'm not sure whats causing the delay |
| 19:11.34 | awk | the box is only 7 hops from the next box |
| 19:11.38 | bkruse | [TK]D-Fender: g729 is AMAZING, seriously |
| 19:11.50 | Seba_soy | I can't HEAR ANY announcement over my ISDN-PRI, like "line is not available" and all that.. (I am from Argentina) |
| 19:11.52 | Seba_soy | some clue? |
| 19:11.52 | [TK]D-Fender | bkruse: Well.... I use ULAW. |
| 19:11.55 | [TK]D-Fender | :D |
| 19:12.04 | bkruse | [TK]D-Fender: ulaw sound quality is AMAZING |
| 19:12.24 | bkruse | but, obviously, 80kbps(with ip overhead) |
| 19:12.39 | bkruse | but if you have the bandwidth, amazing |
| 19:12.52 | awk | so guys, you saying it could be the nat enable = no |
| 19:12.52 | Qwell[] | g722 > ulaw :D |
| 19:12.55 | awk | and what else? |
| 19:13.01 | bkruse | Qwell[]: woah, NO! :P |
| 19:13.17 | Qwell[] | g722 WAY > ulaw :P |
| 19:13.23 | bkruse | haha |
| 19:13.40 | bkruse | lpc10 transcoded to ilbc, and back to lpc10, trying to fax |
| 19:13.47 | bkruse | why is my fax machine no worky?! |
| 19:13.57 | awk | you guys using asterisk realtime at all? |
| 19:14.15 | awk | i think its brilliant, working so well on 1 of my networks |
| 19:14.15 | bkruse | awk: ive tried it, and it is brilliant |
| 19:14.16 | bkruse | very useful |
| 19:14.45 | Seba_soy | Any clue about I cant hear any announcement on my ISDN-PRi... Line is disonnected on zaptel when it received out of order and it does not play announcemente |
| 19:15.03 | bkruse | Seba_soy: but will the call eventually setup? |
| 19:15.05 | awk | ye, specially that I have 5 boxes trying to read sip,iax aswell as the dialplan |
| 19:15.29 | bkruse | awk: dundi is the bomb |
| 19:15.35 | bkruse | you messed with it at all? |
| 19:16.21 | awk | no |
| 19:16.22 | awk | ? |
| 19:16.23 | Seba_soy | I send the call to a number I know is not working |
| 19:16.39 | bkruse | Seba_soy: that made no sense |
| 19:16.42 | Seba_soy | and I not hear "number is disconnected", asterisk just hangup |
| 19:16.43 | bkruse | awk: it rocks |
| 19:17.04 | bkruse | Seba_soy: than have h,1,Playback(file) |
| 19:17.10 | awk | I still got a lot to learn, only been using asterisk for around 1 year, i've been on a few courses. |
| 19:17.17 | Seba_soy | I should hear for example (in spanish) "El numero q ha discado se encuentra momentaneamente desconectado" from my telco |
| 19:17.24 | awk | but kind of stuck to what im using and not really expanding |
| 19:17.41 | *** join/#asterisk froguz (n=alvaro@pc-69-217-46-190.cm.vtr.net) |
| 19:17.42 | Seba_soy | I send the call thorug E1 |
| 19:17.50 | bkruse | awk: cool, well, voip-info.org is your friend, and just start looking at all those config files |
| 19:18.04 | froguz | hi! |
| 19:18.09 | bkruse | getting away from http://asteriskbox and getting into file:///etc/asterisk is a huge difference, and surprising |
| 19:18.11 | bkruse | froguz: wuts up! |
| 19:18.32 | froguz | is www.voip-info.org down? or should i complain my ISP? |
| 19:18.44 | mut | it gets lagged |
| 19:18.44 | bkruse | no |
| 19:18.47 | bkruse | its down ;[ |
| 19:18.50 | bkruse | mut: is it up? just slow? |
| 19:19.01 | bkruse | i thought tzafrir had something to do with it :X |
| 19:19.07 | *** join/#asterisk DocHolliday (i=RgRabbit@gateway/gpg-tor/key-0x0E4F6D6C) |
| 19:19.13 | froguz | i can't load a single page from that site |
| 19:19.20 | DocHolliday | is it possible to ring multiple phones at the same time? |
| 19:19.59 | froguz | maybe they're making some improvements to the wiki. |
| 19:20.37 | froguz | DocHolliday, yes. just add how many SIP extensions you want to ring in Dial app (comma separated) |
| 19:22.02 | toot | anyone know of a technical author? that writes help manuals for voip stuff? :) |
| 19:22.02 | bkruse | DocHolliday: dial(sip/omg&sip/omg2|30) |
| 19:22.39 | froguz | jared smith? |
| 19:22.41 | bkruse | toot: depends on how much it pays :D |
| 19:22.44 | [TK]D-Fender | DocHolliday: You definately need to stop and read THE BOOK |
| 19:22.46 | [TK]D-Fender | ~book |
| 19:22.48 | jbot | [book] a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 19:22.52 | bkruse | [TK]D-Fender: ya |
| 19:23.02 | giesen | DocHolliday: the wiki is your friend |
| 19:23.19 | giesen | www.voip-info.org/wiki |
| 19:23.25 | bkruse | giesen: its down ;[ |
| 19:23.35 | froguz | the wiki is down :'( |
| 19:23.39 | giesen | gah |
| 19:23.46 | bkruse | lol |
| 19:23.52 | bkruse | its a sad day, now we are getting more nub questions that usual |
| 19:23.58 | toot | how much it pays would depend on the quality of the work :) but rates would be fair |
| 19:23.59 | giesen | how can I charge $150/hour to clients |
| 19:24.06 | giesen | when I dont have the wiki to gimme the answers |
| 19:24.12 | jeedi | *lol* |
| 19:24.17 | bkruse | toot: hmm, i wonder if im alowed to do that, what is it on? |
| 19:24.21 | mercestes | lmao. Exactly. |
| 19:24.29 | toot | can i msg ya bkruse? |
| 19:24.33 | bkruse | sure |
| 19:24.37 | mercestes | aw. |
| 19:24.54 | mercestes | lol |
| 19:24.56 | mercestes | Sup ruse? :D |
| 19:25.02 | giesen | I guess it's a good thing my gmail account is a giant mailing list archive |
| 19:25.11 | giesen | searchable by google =) |
| 19:25.13 | mercestes | Chapter 1: You are an idiot. |
| 19:25.32 | giesen | Chapter 2: Read Chapter 1 again. |
| 19:25.40 | jeedi | giesen: for 150$/h you should be smart enough to mirror the wiki onto your laptop.. or use the google cache, ferchrissakes. |
| 19:25.51 | giesen | jeedi: I was joking |
| 19:25.55 | jeedi | ah, okay |
| 19:25.58 | [TK]D-Fender | giesen: Appendix : You thought you could just skip to the end. You're STILL an IDIOT. |
| 19:25.58 | jeedi | sorry, then. |
| 19:25.59 | giesen | I only charge $85/hour |
| 19:26.05 | jeedi | haha |
| 19:26.06 | mercestes | wget http://voipinfo.org/wiki :D |
| 19:26.55 | jeedi | mercestes: wget -mrk ;) |
| 19:26.59 | giesen | [TK]D-Fender: I think we could get a nice fat co-write deal |
| 19:27.17 | froguz | google's cache isn't working neither |
| 19:27.42 | DocHolliday | [TK]D-Fender, any idea why this isnt working? exten => 0,1,Dial(SIP/240,SIP/250,SIP/260|20) |
| 19:27.49 | ping2921 | whats the recommended mysql connection, unixodbc or mysql-addons ? |
| 19:27.49 | DocHolliday | it just goes right to voicemail |
| 19:27.50 | giesen | you cant use , |
| 19:27.51 | giesen | use & |
| 19:27.57 | DocHolliday | ahh |
| 19:28.06 | giesen | SIP/240&SIP/250&SIP/260 |
| 19:29.48 | DocHolliday | :) |
| 19:29.51 | DocHolliday | thanks!! |
| 19:30.15 | [TK]D-Fender | DocHolliday: "show application dial" |
| 19:30.39 | DocHolliday | [TK]D-Fender, i figured it out :P |
| 19:30.51 | mercestes | jeedi: bwahahaha |
| 19:30.57 | [TK]D-Fender | Just goes to show again that you can lead a horse to water, but the SPCA won't let you hold its head under... |
| 19:31.11 | *** join/#asterisk gr0mit (n=tim@extrt.txrx.org.uk) |
| 19:31.12 | *** join/#asterisk shinux__ (n=shinux@196.220.25.58) |
| 19:32.14 | DocHolliday | hah |
| 19:32.29 | jeedi | ha ha |
| 19:32.48 | DocHolliday | everything works now, just wish you could dial an extension before the IVR finished |
| 19:35.30 | *** join/#asterisk _deg_ (n=deg@200.195.161.164) |
| 19:35.49 | awannabe | bastard asterisk and call parking! |
| 19:35.49 | bkruse | jeedi: nice... |
| 19:35.53 | [TK]D-Fender | DocHolliday: KEEP READING |
| 19:35.55 | _deg_ | Hi all... Is there a way to change de callerid behavior on blind/attended transfers? |
| 19:36.00 | JacksLivr | bkruse: wassup? |
| 19:36.01 | bkruse | [TK]D-Fender: exactly |
| 19:36.04 | bkruse | JacksLivr: wuts up man! |
| 19:36.22 | [TK]D-Fender | _deg_: Sure, you've got... THE SOURCE :) |
| 19:36.52 | _deg_ | What I want is to receive the callerid of the extension that trasnfered the call and not the callerid of the calling. |
| 19:36.54 | [TK]D-Fender | _deg_: To * anyways :) And then there is the issu of the PHONES you are using. |
| 19:37.07 | [TK]D-Fender | _deg_: Then Attended Transfer it is... |
| 19:37.11 | _deg_ | [TK]D-Fender, of course we have, but is there a Dial flag/option to do that? |
| 19:37.18 | mercestes | DocHolliday: use background instead of playback |
| 19:37.23 | [TK]D-Fender | _deg_: There is no option. |
| 19:37.33 | _deg_ | hmmm |
| 19:37.55 | _deg_ | [TK]D-Fender, In brazil we have some diferences from USA. CAllerid on ytransfers is one of them.... |
| 19:38.03 | JacksLivr | bkruse: no training for me. TX here i come. |
| 19:38.16 | *** join/#asterisk yonahw (n=yonahw@84.229.143.162) |
| 19:38.23 | mercestes | JacksLivr: What part of Tx. |
| 19:38.28 | _deg_ | [TK]D-Fender, Our expected behaviour is to receive the callerid of the extensios that do the transfer.... |
| 19:38.29 | [TK]D-Fender | _deg_: "That's nice", but these systems don't care about what you ight consider "standard". |
| 19:38.30 | JacksLivr | Dallas |
| 19:38.44 | mercestes | JacksLivr: I'm in h-town. |
| 19:38.50 | [TK]D-Fender | _deg_: Then only use attended Transfers, not BLIND |
| 19:39.00 | _deg_ | attended transfers do the same. |
| 19:39.11 | froguz | maybe the wiki has been hacked by some CallManager fan :-o |
| 19:39.13 | [TK]D-Fender | _deg_: What phones? |
| 19:39.14 | _deg_ | [TK]D-Fender, What about the "o" option in Dial cmd? |
| 19:39.18 | _deg_ | Polycom |
| 19:39.21 | _deg_ | 501 |
| 19:39.29 | *** join/#asterisk zotz (n=zotz@24.244.163.157) |
| 19:39.35 | _deg_ | Asterisk 1.4.1 here |
| 19:40.05 | [TK]D-Fender | _deg_: * 1.4.X is still somewhat proken with regards to Polycoms from what I hear. This is NOT normal behaviour |
| 19:40.18 | _deg_ | [TK]D-Fender, hmmmm |
| 19:40.26 | _deg_ | gona try with audiocodes ATA |
| 19:40.39 | *** join/#asterisk [Airwolf] (n=airwolf@voip.ymav.nl) |
| 19:42.51 | mercestes | 1.4.X is broken with polycoms? saywha? |
| 19:43.29 | [TK]D-Fender | mercestes: I've heard a few reports in here from those who've tried.... wierdness ensued.... |
| 19:43.55 | [TK]D-Fender | _deg_: I'd also get a firsthand look at these attempted transfers if I were you. |
| 19:45.12 | _deg_ | [TK]D-Fender, thank Fender! |
| 19:45.13 | mercestes | =/ Half my system is polycoms and I've a scheduled upgrade cmoing tomorrow. :( |
| 19:45.26 | _deg_ | [TK]D-Fender, on a meetding now... |
| 19:45.27 | [TK]D-Fender | _deg_: and? |
| 19:45.53 | [TK]D-Fender | mercestes: What kind of environment & upgrade? |
| 19:48.16 | jeedi | this is exactly the kind of moment where i _love_ working for a company that only does pstn. no achy breaky voip-stuff ;) |
| 19:48.57 | mercestes | [TK]D-Fender: Gentoo and upgrade to asterisk 1.4 |
| 19:49.08 | mercestes | Running Sip-2.1.0 |
| 19:49.12 | [TK]D-Fender | mercestes: ..... |
| 19:49.16 | [TK]D-Fender | ~wglwat |
| 19:49.17 | jbot | from memory, wglwat is well, good luck with all that |
| 19:49.19 | [TK]D-Fender | :D |
| 19:49.20 | mercestes | lol |
| 19:49.26 | mercestes | thanks. |
| 19:49.46 | [TK]D-Fender | mercestes: You are clearly either well versed with masochism... or ABOUT TO BE. |
| 19:49.52 | robl^ | yay!! polycoms!! |
| 19:49.56 | thekidrio | hhahahaha tk |
| 19:50.06 | mercestes | .....Ask me in another channel....and....why? |
| 19:50.31 | [TK]D-Fender | jeedi: And what do you run? |
| 19:51.09 | [TK]D-Fender | mercestes: No thanks, I'm taking a break before finding a new "top" ;) |
| 19:51.15 | mercestes | [TK]D-Fender: is it the gentoo, the asterisk 1.4, or the polycoms? ...or all three? |
| 19:51.23 | [TK]D-Fender | mercestes: yes |
| 19:51.29 | Cyon | Is there any known reason why an outside call coming into an extension that has the screening option (p) would work, while a call transferred from another extension gets dropped? |
| 19:51.36 | jeedi | [TK]D-Fender: a bunch of asterisk boxes with sangoma a108d (octal-pri) - 240 lines per box ;) |
| 19:51.41 | mercestes | [TK]D-Fender: Ok, well call me when your done with your vacation. |
| 19:51.49 | [TK]D-Fender | jeedi: And for phones? |
| 19:52.18 | jeedi | [TK]D-Fender: phones? we don't use no steenkin' fones 'ere ;) |
| 19:52.44 | [TK]D-Fender | jeedi: Call in / out / redirect only? |
| 19:52.55 | jeedi | on my desk, there's two alcatel phones.. connected to an OmniPCX. |
| 19:53.36 | jeedi | [TK]D-Fender: mostly IVR stuff, redirection and call-through. |
| 19:53.53 | *** join/#asterisk fakhir (i=fakhir@unaffiliated/fakhir) |
| 19:53.57 | [TK]D-Fender | jeedi: cool. |
| 19:54.13 | jeedi | we got plans to throw out the omnipcx and replace it with a small asterisk box (2xPRI only) |
| 19:54.37 | pigpen | anyone know why the page app in 1.4.1 is blowing up asterisk? |
| 19:54.38 | *** join/#asterisk kgx (n=kgx@60.234.20.178) |
| 19:54.49 | bkruse | pigpen: write a bug report, that should have just been fixed. |
| 19:54.56 | bkruse | file: any comments? |
| 19:55.24 | pigpen | I ran across it last week...sick wife and kid kept me from getting around to it. |
| 19:55.40 | pigpen | still sick... |
| 19:56.02 | JacksLivr | to live? |
| 19:56.06 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 19:56.14 | pigpen | I have something else killing 1.4.1 ....trying to figure out what however.... |
| 19:56.17 | bkruse | pigpen: that would be great, any output, does asterisk blow up? |
| 19:56.35 | *** join/#asterisk Stridernzl (n=neville@222-153-62-109.jetstream.xtra.co.nz) |
| 19:56.35 | file | there is already a bug open that I am in the process of looking at |
| 19:56.41 | pigpen | yeah...when I do page app....I also get grsec output as well. |
| 19:56.58 | pigpen | file, tks....worked fine in 1.4.0 |
| 19:57.04 | jeedi | [TK]D-Fender: and we're running two asterisk boxes with two quad-pri tormenta cards in them.. for SS7 stuff. |
| 19:57.07 | file | as the bug says. |
| 19:57.27 | jeedi | two cards per machine, that is. |
| 19:57.27 | [TK]D-Fender | jeedi: Cooler still.... |
| 19:57.27 | pigpen | I will try to figure out the other item...kinda tough as I am running it production for about 200 phones. |
| 19:57.28 | *** join/#asterisk ManxPower (n=manxpowe@71-8-56-64.dhcp.leds.al.charter.com) |
| 19:57.31 | *** join/#asterisk saftsack (n=oliver@pD9E07BE9.dip.t-dialin.net) |
| 19:57.56 | saftsack | today i got my first embedded * working. openwrt on an asus wl-500gp + patton gateway to BRI. works quite NICE :) |
| 19:57.58 | jeedi | it was a major pain in the lower back to find a carrier that would let us connect asterisk+chan_ss7 to his ss7 switch ;) |
| 19:58.06 | [TK]D-Fender | bkruse: Sounds like I'm on to waiting for 1.4.2 ;) |
| 19:58.44 | bkruse | [TK]D-Fender: possibly |
| 19:58.50 | bkruse | 1.4.1 was "rushed" out i guess you could say |
| 19:59.27 | pigpen | bkruse, thanks for the info..I won't kill myself trying to figure out why I have the services stop now and then. |
| 19:59.36 | pigpen | ie: once every 4 days or so... |
| 20:01.14 | gr0mit | anyone here from south africa who has experience running sip or iax to a uk-hosted asterisk box? |
| 20:01.39 | ManxPower | bkruse: I figured that when some critical (I don't recall which one, maybe Polycom parking) fix was not included in 1.4.1 |
| 20:02.27 | JacksLivr | gr0mit: that is the most specific request I have ever seen. |
| 20:03.03 | pigpen | gr0mit, that reminds me, can personal users use something like skype or "out of country voip" provider when in Ethiopia? I am helping out a missionary.... |
| 20:03.40 | JacksLivr | you dont hear, "Could you be more specific?" often, i bet. |
| 20:03.40 | pigpen | I heard the government out-laws this.... |
| 20:03.57 | [TK]D-Fender | ManxPower: You mean as in bweske's tree for integration? |
| 20:04.19 | gr0mit | jackslivr, if you look at the south african adsl services, you will see that there are a plethora of capped vs uncapped offerings, some go via satelite for int service, others via the SAT-3 cable |
| 20:05.10 | ManxPower | [TK]D-Fender: it was on bugs.digium.com. no idea what branch/tree |
| 20:05.11 | gr0mit | they also shape most traffic. i need to know if anyone has encountered good results running iax2 or sip/RTP via one of the traffic-shaped services al |
| 20:05.49 | [TK]D-Fender | ManxPower: It was "in progress" forever... didn't knwo it was anywhere near being ready for full-merge. |
| 20:05.54 | [TK]D-Fender | Can anyone corroborate? |
| 20:06.02 | gr0mit | i.e. do i need to pay for an unshaped, guaranteed fibre connections to get reasonable voip performance. |
| 20:06.20 | Corydon-w | It couldn't hoit |
| 20:06.36 | gr0mit | pigpen - i have no idea about ethiopia. |
| 20:07.26 | gr0mit | i am just astonished that the the south african telcos seem to limit both bandwith and relative bandwith by imposing these limits |
| 20:08.25 | rkeels | is voip-info.org down? |
| 20:08.38 | robl^ | yup! |
| 20:08.47 | rkeels | Grrrr |
| 20:09.04 | mercestes | Crapsters |
| 20:09.26 | gr0mit | jackslivr are you in south africa then? |
| 20:09.28 | *** join/#asterisk ShadowHntr (i=sentinel@wikipedia/Shadowhntr) |
| 20:09.33 | JacksLivr | im not |
| 20:09.40 | mercestes | Sounds like 1.4 is lemonified. |
| 20:09.44 | JacksLivr | us and a |
| 20:09.46 | gr0mit | lemme guess, US? |
| 20:09.51 | JacksLivr | lol |
| 20:10.13 | *** join/#asterisk emiquelito (n=evandro@200-155-185-1.static.spo.ifx.net.br) |
| 20:10.21 | bkruse | someone change the topic of the room to "Yes, Voip-info is down." |
| 20:10.23 | bkruse | lol |
| 20:10.26 | ManxPower | [TK]D-Fender: The bug did not exist in 1.2.x |
| 20:10.26 | gr0mit | ok, well if you have not tried to work out ADSL in SA then you would not understand the question! |
| 20:10.36 | Qwell[] | bkruse: nobody'll read it anyways |
| 20:10.41 | gr0mit | I did not realise there might even be a prob til today! |
| 20:11.12 | *** join/#asterisk Ryushin (i=proxy@windwalker.openinnovations.com) |
| 20:11.17 | ManxPower | mercestes: it seems the 1.0 and 1.2 were just as buggy during the initial releases |
| 20:12.53 | rkeels | my queues.conf blows asterisk up too once in a while. especially if I module unload app_queue.so |
| 20:13.01 | *** join/#asterisk wwalker (n=wwalker@pdpc/supporter/sustaining/wwalker) |
| 20:13.16 | JoNate | mercestes: I'm having problems too...I'm totaly switching over to that new system... |
| 20:13.42 | JoNate | mercestes: I just don't know what kind of signal strength Telepathy can give me... |
| 20:13.56 | gr0mit | so no-one in South Africa able to give me any hints on running a voip conection over telkom's encumbered ADSL offerings? |
| 20:14.35 | [TK]D-Fender | ManxPower: What bug? |
| 20:15.06 | ManxPower | [TK]D-Fender: polycoms not being able to transfer or something like that. It was in specific situations like transfering after a park |
| 20:15.36 | wwalker | Anyone know what setting controls what # is dialed by Polycom IP500 phones when the "Messages" button is hit? voip-info.org appears down so the relevant google results don't come up... |
| 20:15.45 | [TK]D-Fender | ManxPower: OH, I thought you were talking about the SVN bracnh for enabling Polycom's internal call-parking feature... |
| 20:16.06 | ManxPower | no, I'm taking about 1.4.1 release |
| 20:16.15 | [TK]D-Fender | wwalker: in the MWI tag near the bottom of sip.cfg |
| 20:16.24 | ManxPower | wwalker: whatever it is configured for |
| 20:16.33 | [TK]D-Fender | wwalker: Actually I think that'd be in your phones specific config file.. |
| 20:16.42 | ManxPower | [TK]D-Fender: it can be either place. |
| 20:16.50 | ManxPower | in sip.cfg or in mac-phone.cfg |
| 20:16.56 | [TK]D-Fender | ManxPower: Yeah, basic parking bug... I hadn't heard of that one actually... |
| 20:17.23 | *** join/#asterisk eltech (i=G00Ds@ool-457c93b6.dyn.optonline.net) |
| 20:18.23 | mercestes | Ugh. |
| 20:18.26 | wwalker | ManxPower - yes, which setting configures that button? |
| 20:18.34 | mercestes | Maybe I should just compile 1.2.16 from source. |
| 20:19.01 | *** join/#asterisk ToyMan (n=Stuart@user-12lcqut.cable.mindspring.com) |
| 20:19.21 | mercestes | what's a good version to fix the voicemail forwards from places other than INBOX or is there a patch to fix that?? |
| 20:22.43 | [TK]D-Fender | wwalker: under the MWI tab. Go download the Admin Guide |
| 20:24.00 | *** join/#asterisk gmfm (n=gmfm@67.60.56.115) |
| 20:29.54 | *** join/#asterisk Braxus (n=braxus@66.147.214.164) |
| 20:31.35 | [TK]D-Fender | BBIAB |
| 20:32.47 | *** join/#asterisk tsurko (n=tsurko@vpn-pppoe-213-240-242-237.megalan.bg) |
| 20:33.57 | mercestes | gr0mit: My advice is ...... don't do it. |
| 20:34.18 | rkeels | I am confused |
| 20:34.20 | rkeels | Can someone tell me which part of this is family tree and key |
| 20:34.21 | rkeels | //Queue/PersistentMembers/ProdHunt |
| 20:34.38 | JunK-Y | family/key/value |
| 20:34.48 | bkruse | JunK-Y: nice |
| 20:35.06 | rkeels | thx |
| 20:35.08 | rkeels | so what is the tree |
| 20:36.01 | gr0mit | mercestes - can you elaborate on this advice please?! |
| 20:36.45 | *** join/#asterisk fbffff (n=fbffff@r74-193-79-81.pfvlcmta01.grtntx.tl.dh.suddenlink.net) |
| 20:38.17 | JunK-Y | the tree is the general concept. |
| 20:38.27 | brian | can anyone here get me a free DID for development purposes? |
| 20:39.23 | rkeels | kew |
| 20:39.37 | brad_mssw | brian: freeworlddialup and/or iaxtel are free |
| 20:40.48 | *** join/#asterisk codazoda (n=chatzill@mail.hurdmanivr.com) |
| 20:42.19 | codazoda | I have 2 TDM404B cards. I've written an IVR system in PHP using the AGI. When I dial one of my ZAP ports, it has a hard time hearing the touch tones. They work, but I have to press them for about half a second. With a SIP call that goes to the AGI, the touch tones work better. Ideas? |
| 20:42.50 | mvanbaak | a tree is one of those things in my garden |
| 20:42.54 | mvanbaak | with leaves and stuff |
| 20:44.03 | rkeels | Is anyone able to get a working production call center with queues working on 1.4 or any other for that matter |
| 20:44.29 | mvanbaak | rkeels: what is the problem ? |
| 20:47.44 | *** part/#asterisk mtaht4 (n=m@c-71-198-23-124.hsd1.ca.comcast.net) |
| 20:49.14 | toombaloomba | argh wiki is down :( |
| 20:49.34 | codazoda | voip-info.org has been down all day. |
| 20:50.00 | gr0mit | needs a mirror, methinks. |
| 20:50.05 | toombaloomba | thats coz they run off some shitty line in hawaii |
| 20:50.05 | marv[work] | does asterisk do any kind of volume scaling behind my back? |
| 20:50.08 | codazoda | The DNS on that thing is DYNDNS. Does that mean it's run on some guys DSL line? |
| 20:50.34 | robl^ | I think he runs it on dial-up ppp |
| 20:50.47 | codazoda | Probably. |
| 20:51.04 | marv[work] | I'm recording a call using Monitor() in the dialplan, and Playback()ing a sound file |
| 20:51.16 | marv[work] | the recording of the sound file is a lot louder than the sound file |
| 20:51.50 | DrukenLPY | i'd mirror it... |
| 20:51.51 | marv[work] | as in the recording is at full scale and the original is at 2.4 of scale |
| 20:51.53 | mercestes | gr0mit: The "don't do it" part? |
| 20:52.09 | codazoda | There is a lot of info on that site. Perhaps Digium should mirror it. I'd mirror it as well. |
| 20:52.52 | gr0mit | yes - i might even mirror it - what b/w would mirroring it take, d'you reckon? |
| 20:53.08 | gr0mit | mercesetes - yup. that part! |
| 20:53.09 | codazoda | Permission. ;-) |
| 20:53.46 | codazoda | Google's Cache has helped me lots today... |
| 20:53.59 | bkruse | google cache rocks... |
| 20:53.59 | gr0mit | marv[work] this is a long known issue |
| 20:54.33 | marv[work] | gr0mit: is it? could you fill me in, or give me a url, or the proper google search term? |
| 20:54.51 | gr0mit | there is a gain setting that amplifes the recordings. we have to change the setting. let me look for the bug . 1 sec. |
| 20:55.19 | marv[work] | gr0mit: is this just in the recordings, or the actual audio being sent? |
| 20:56.29 | *** join/#asterisk flenders (n=fserto@unaffiliated/flenders) |
| 20:56.42 | *** join/#asterisk |Vulture| (n=_Vulture@101.222.121.70.cfl.res.rr.com) |
| 20:56.54 | flenders | is anyone else having problems to access the wiki? |
| 20:57.06 | |Vulture| | Anyone here have an updated line for Daylight Savings Time for the Polycom IP-XXX series of phones? |
| 20:57.07 | gr0mit | it is just the recordings that are broken |
| 20:57.52 | gr0mit | http://bugs.digium.com/view.php?id=5823 |
| 20:58.01 | marv[work] | thanks |
| 20:58.08 | bkruse | 5823, wow |
| 20:58.43 | *** part/#asterisk wwalker (n=wwalker@pdpc/supporter/sustaining/wwalker) |
| 20:58.55 | flenders | damn it, I got in early today to try the fxotune thingy, and the bloody wiki is down |
| 20:59.11 | bkruse | flenders: google cache! |
| 20:59.12 | jeedi | google cache is the way to go, man.. |
| 20:59.32 | bkruse | ;] |
| 20:59.33 | mcab | |Vulture|: http://knowledgebase.polycom.com/kb/search.do?cmd=displayKC&docType=kc&externalId=10627&sliceId=SAL_PUBLIC_1_2&dialogID=1890871&stateId=1%200%201886835 |
| 20:59.42 | *** join/#asterisk [TK]D-Fender (n=chatzill@64.235.216.2) |
| 20:59.59 | gr0mit | marv[work] i just pasted the URL. Digium bug 5823 |
| 21:00.09 | |Vulture| | mcab: GRACIAS! |
| 21:00.10 | marv[work] | gr0mit: thanks |
| 21:00.18 | gr0mit | it is aright PIA |
| 21:00.27 | mcab | |Vulture|: no problem |
| 21:00.58 | gr0mit | what needs to be done to mirror the voip-info wiki? what is the nettiquete here? |
| 21:01.21 | gr0mit | it is like having my arms and legs chopped off |
| 21:01.50 | flenders | just tried google cache, and it's no good either |
| 21:02.41 | b11d|bbl | isnt voip-info.org archived on web.archive.org ? |
| 21:03.29 | *** join/#asterisk mivck (i=1000@ip-70-228.telesat.com.co) |
| 21:04.10 | bkruse | dunno |
| 21:04.15 | bkruse | i would just use google |
| 21:04.34 | bkruse | doesnt the person taht runs voip-info.org local #asterisk? |
| 21:04.42 | marv[work] | i've found reading the asterisk sample config files and using 'show application xx' 'show agi xx' and 'show manager xx' and mostly replaced my voip-info usage for asterisk |
| 21:04.53 | bkruse | marv[work]: true. |
| 21:06.52 | *** join/#asterisk MrTelephone (n=MrTeleph@h697179-171.picriverisp.net) |
| 21:07.01 | MrTelephone | does anyone else get errors with enum.c when compiling asterisK? |
| 21:07.23 | JunK-Y | MrTelephone: which version? |
| 21:07.26 | *** join/#asterisk Assid (n=assid@59.183.47.77) |
| 21:07.27 | bkruse | errors, as in errors out? |
| 21:07.31 | bkruse | and what are the errors? |
| 21:07.35 | MrTelephone | unsignedness errors |
| 21:07.40 | bkruse | mm |
| 21:08.10 | MrTelephone | do you guys get that? |
| 21:08.18 | bkruse | what version, trunk? |
| 21:08.21 | bkruse | what rev, also |
| 21:08.31 | Assid | russellb you around? |
| 21:08.56 | MrTelephone | release 1.4.16 |
| 21:09.00 | MrTelephone | i mean 1.2.16 |
| 21:09.03 | MrTelephone | from the website |
| 21:10.55 | MrTelephone | RANT -D_GNU_SOURCE -O6 -march=i686 -DZAPTEL_OPTIMIZATIONS -fomit-frame-pointer -c -o enum.o enum.c |
| 21:10.55 | MrTelephone | enum.c: In function âparse_naptrâ: |
| 21:10.55 | MrTelephone | enum.c:107: warning: pointer targets in initialization differ in signedness |
| 21:10.55 | MrTelephone | enum.c:133: warning: pointer targets in passing argument 1 of âparse_ieâ differ in signedness |
| 21:11.09 | bkruse | hmm |
| 21:11.25 | russellb | Assid: Yes, but busy |
| 21:11.29 | russellb | MrTelephone: warning is not an error |
| 21:11.33 | MrTelephone | ok |
| 21:11.36 | russellb | you can ignore it |
| 21:11.42 | MrTelephone | as long as I can compile then? |
| 21:11.48 | denon | right |
| 21:12.29 | b11d|bbl | when is asterisk 2.0 coming out? |
| 21:12.32 | b11d|bbl | how can I fix my DST issues? |
| 21:12.36 | b11d|bbl | where is IPv6 in Asterisk ? |
| 21:12.38 | b11d|bbl | etc.. |
| 21:12.39 | b11d|bbl | :P |
| 21:12.48 | b11d|bbl | hehe |
| 21:13.04 | *** join/#asterisk elriah (i=elriah@adsl-072-149-159-016.sip.bhm.bellsouth.net) |
| 21:13.27 | elriah | Greets. We got asterisk 1.2 compiled and running on NT4 Embedded. |
| 21:13.29 | Assid | russellb: okay, let me know when you free we can do this |
| 21:13.43 | denon | b11d|bbl: set it to universal time, 2.0 will be out next week (russellb promises), and ipv6 is in 2.0 |
| 21:13.45 | *** join/#asterisk gambolputty (n=gambolpu@cblmdm72-240-185-4.buckeyecom.net) |
| 21:13.45 | MrTelephone | whats with this RANT -D_GNU_SOURCE -O6 -march=i686 -DZAPTEL_OPTIMIZATIONS -fomit-frame-pointer -c -o enum.o enum.c |
| 21:13.45 | MrTelephone | enum.c: In function âparse_naptrâ: |
| 21:13.45 | MrTelephone | enum.c:107: warning: pointer targets in initialization differ in signedness |
| 21:13.45 | MrTelephone | enum.c:133: warning: pointer targets in passing argument 1 of âparse_ieâ differ in signedness |
| 21:13.45 | russellb | Assid: there are some others in #asterisk-bugs who can help you ... |
| 21:13.49 | MrTelephone | sorry didn't mean to do that |
| 21:13.52 | MrTelephone | yikes |
| 21:14.02 | bkruse | MrTelephone: its all good ;] |
| 21:14.21 | b11d|bbl | denon.. nice :) |
| 21:14.38 | denon | b11d|bbl: 2.1 will support IPv8 |
| 21:14.43 | denon | may be worth holding off for that |
| 21:14.45 | b11d|bbl | it BETTER. |
| 21:14.47 | b11d|bbl | :) |
| 21:14.51 | MrTelephone | why do the warnings happen? is it so it will compile on other systems? |
| 21:15.04 | b11d|bbl | ttyl lads |
| 21:15.35 | bkruse | DrukenLPY: im only mean to the nubs |
| 21:15.35 | elriah | Does anyone actually have a production IPv6 network yet? |
| 21:15.51 | denon | lots of people do |
| 21:15.57 | denon | but most dont really *need* it |
| 21:16.01 | elriah | lol |
| 21:16.04 | bkruse | DrukenLPY: i am feeling rather peppy also :D |
| 21:16.15 | DrukenLPY | bkruse: peppy? hehehe |
| 21:16.27 | bkruse | is that the word? |
| 21:16.36 | bkruse | i cannot spell, i will be honest. |
| 21:16.48 | DrukenLPY | not sure... this is irc... can anyone spell? |
| 21:16.58 | bkruse | DrukenLPY: very true. |
| 21:17.17 | DrukenLPY | god knows i can't |
| 21:17.32 | nDuff | not a good day for useful *-related resources; from where I'm at, voip-info.org and newmantelecom.com both look down. |
| 21:18.06 | denon | seems like every time it's down, people go into shock |
| 21:18.25 | mut | heh |
| 21:18.31 | DrukenLPY | hey mut |
| 21:18.33 | mut | last i checked he didnt want mirroring |
| 21:18.33 | bkruse | denon: a mirror with advertising |
| 21:18.40 | denon | bkruse: shrugs .. just a mirror |
| 21:18.43 | mut | hello |
| 21:18.47 | bkruse | denon: nvm. |
| 21:18.49 | bkruse | ;] |
| 21:18.50 | bkruse | $$$ |
| 21:18.58 | denon | yeah |
| 21:19.39 | DrukenLPY | you got any experince with a Millennium multipay ?? |
| 21:20.13 | DrukenLPY | oh.. yeah that was for mut btw :) my bad |
| 21:21.18 | mut | nope |
| 21:21.20 | flenders | wow, google cache is REALLY slow, isn't it? about 15 minutes to show the page |
| 21:21.39 | denon | it pulls it off a very low priority storage store, I believe |
| 21:21.41 | bkruse | slow, ya, but usually not 15 minutes |
| 21:21.44 | denon | and reconstructs it from high compression |
| 21:21.49 | toombaloomba | yea :( |
| 21:21.53 | denon | or at least that's how it was explained to me |
| 21:22.01 | bkruse | denon: that sounds about right |
| 21:22.01 | DrukenLPY | mut: damn.... |
| 21:22.18 | denon | but yeah, not 15min slow |
| 21:23.17 | DrukenLPY | that's gotta cost a fortune in storage.... |
| 21:23.46 | denon | actually, google's BigTable design is pretty slick |
| 21:24.04 | denon | large logical storage devices on commodotity hardware |
| 21:24.11 | bkruse | googles slick all together, they own |
| 21:24.19 | denon | eh, I wouldn't go that far |
| 21:24.23 | bkruse | no? |
| 21:24.33 | denon | they have some nice technical implementations, but their business ethics may be bordering on .. not so nice |
| 21:24.57 | bkruse | ahh, i gotcha |
| 21:25.07 | bkruse | i was speaking on behalf of the technical aside from all else |
| 21:25.12 | bkruse | but i see what you mean |
| 21:25.21 | denon | nod .. though lately their search algorithms have been less than ideal |
| 21:25.30 | denon | you may have noticed, it takes longer to find what you're looking for lately .. |
| 21:25.32 | bkruse | ohrly? |
| 21:25.37 | bkruse | hmmmm |
| 21:25.37 | denon | and the results seem to be watered down a bit |
| 21:25.47 | bkruse | interesting......right, i have noticed that, oddly enough |
| 21:25.50 | denon | especially the past 3-4 months from our experiences |
| 21:25.53 | MrTelephone | i just upgraded my pri card drivers and now I can't make calls out my pri :( |
| 21:25.55 | denon | I dont recall which updates seemed the worst |
| 21:26.05 | bkruse | MrTelephone: ztcfg -vv |
| 21:26.06 | MrTelephone | I keep getting messages that dchan is up |
| 21:26.07 | mercestes | so what's up with Google's business ethics? </troll> |
| 21:26.08 | MrTelephone | oh |
| 21:26.18 | denon | we kinda monitor their algorithm updates, at least the ones they publish to the world |
| 21:26.20 | *** join/#asterisk thoughtpolice (n=austin@ip68-98-250-69.lu.dl.cox.net) |
| 21:26.20 | MrTelephone | I rebooted anyways |
| 21:26.21 | elriah | Ayone tried to port a number FROM Vonage? I'm having a hell of a time. It's been about 20 business days and no results yet. |
| 21:26.33 | bkruse | i bet |
| 21:26.43 | bkruse | vonage is crazy retarded (at least from all the stories i hear) |
| 21:26.46 | mercestes | elriah: PUC is usually a good resource for that. :D |
| 21:26.49 | *** join/#asterisk danalien (n=danalien@unaffiliated/danalien) |
| 21:26.59 | elriah | PUC? |
| 21:27.05 | denon | public utilities commission |
| 21:27.05 | mercestes | Public Utilities Comission. |
| 21:27.09 | denon | the organization that governs these things |
| 21:27.16 | denon | well, govt branch |
| 21:27.36 | *** join/#asterisk sudhir492 (n=sudhir@c-71-63-59-45.hsd1.va.comcast.net) |
| 21:27.40 | sudhir492 | Hi all |
| 21:27.45 | denon | vonage is in NJ, so you'll want to talk to the NJ PUC |
| 21:28.16 | elriah | Thanks. So you're saying call them and complain? |
| 21:28.31 | denon | yes, sometimes they'll talk to the carrier on your behalf |
| 21:28.37 | denon | which .. is usually very effective |
| 21:29.03 | flenders | after running fxotune, do I need to re-tune my rx/tx gains? |
| 21:29.12 | denon | elriah: http://www.bpu.state.nj.us/home/home.shtml |
| 21:33.36 | MrTelephone | hmmm |
| 21:33.43 | MrTelephone | pri's are not too good up here |
| 21:34.13 | bkruse | MrTelephone: pri intense debug span 1 |
| 21:34.16 | bkruse | :D |
| 21:35.14 | MrTelephone | > Unnumbered frame: |
| 21:35.14 | MrTelephone | > SAPI: 00 C/R: 0 EA: 0 |
| 21:35.14 | MrTelephone | > TEI: 000 EA: 1 |
| 21:35.14 | MrTelephone | > M3: 3 P/F: 1 M2: 3 11: 3 [ SABME (set asynchronous balanced mode extended) ] |
| 21:35.14 | MrTelephone | > 0 bytes of data |
| 21:35.29 | MrTelephone | alarms are ok |
| 21:35.31 | *** join/#asterisk drako (n=ljd@unaffiliated/luisjose) |
| 21:35.58 | MrTelephone | I get an error no d channel available after it says dchannel is up |
| 21:35.59 | MrTelephone | hmm |
| 21:36.07 | bkruse | hardhdlc=dchan |
| 21:36.14 | bkruse | you got cpe and net set correctly? |
| 21:36.36 | MrTelephone | if I set it to net then the t1 card relay clicks continuously |
| 21:36.51 | JoNate | anyone know a good bandwidth calculator? |
| 21:37.03 | MrTelephone | the main office runs off a dms100 should I set switchtype to dms100? because it works on national too |
| 21:37.44 | DrukenLPY | i've heard that vonage doesn't allow ANY number to be ported from them.... |
| 21:38.02 | bkruse | lame |
| 21:38.15 | bkruse | MrTelephone: national is yur best bet |
| 21:38.23 | bkruse | JoNate: uh, there is one on asteriskguru |
| 21:38.26 | bkruse | search it on google |
| 21:38.27 | bkruse | its great |
| 21:39.21 | JoNate | thanks |
| 21:39.25 | MrTelephone | > M3: 3 P/F: 1 M2: 0 11: 3 [ UA (unnumbered acknowledgement) ] |
| 21:39.55 | *** join/#asterisk |kahless| (n=kvirc@i577ACFAB.versanet.de) |
| 21:39.57 | |kahless| | hi |
| 21:39.58 | *** join/#asterisk [Airwolf] (n=airwolf@voip.ymav.nl) |
| 21:40.08 | *** join/#asterisk HockeyInJune (n=HockeyIn@pool-68-161-174-162.ny325.east.verizon.net) |
| 21:40.41 | MrTelephone | it's not synchronizing or something |
| 21:40.47 | MrTelephone | once in a while dchan goes up then goes down |
| 21:42.13 | JacksLivr | i am wanting to have a sip address like sip:JacksLivr@domain.com that people without voip service of any kind can call from a computer softphone and it ring into my asterisk server. this is possible, right? |
| 21:42.34 | bkruse | JacksLivr: yes, as long as their codec is supported |
| 21:42.53 | bkruse | if its a SIP call, itll work(with a sip softphone) |
| 21:44.39 | flenders | after running fxotune, do I need to re-tune my rx/tx gains? does anyone know that? |
| 21:45.48 | MrTelephone | hmmm |
| 21:45.55 | MrTelephone | reinstall the drivers i guess |
| 21:47.41 | MrTelephone | what happens if I set this to no? Enable TDMV DCHAN Native HDLC Support & Patch Zaptel ? (y/n) |
| 21:48.09 | *** join/#asterisk jart (n=user@ool-43509aa5.dyn.optonline.net) |
| 21:48.34 | jart | does anyone know why an asterisk call sent to a perl agi app might freeze and never exit properly? |
| 21:50.05 | telmich | jart: perhaps blocking in/output? |
| 21:50.11 | telmich | jart: does it react on sigstop? |
| 21:50.42 | jart | it must be blocking on /some/ i/o |
| 21:51.02 | telmich | well, give it an strace before |
| 21:51.44 | jart | i was coming here to hope i wouldn't have to get dirty |
| 21:52.35 | MrTelephone | hmmm |
| 21:52.40 | toombaloomba | damnit, anyone know on cisco 7941/7961 phones how to get to the network settings after doing a factory reset? |
| 21:52.48 | MrTelephone | sucks upgrading to the newest verson of things |
| 21:53.43 | flenders | WOW! fxotune is awesome! |
| 21:53.56 | giesen | toombaloomba: the settings button doesnt work? |
| 21:54.12 | flenders | one of our lines went from 38% of echo to 1% |
| 21:54.16 | MrTelephone | fxotune didn't work for me too well when I had analog lines |
| 21:54.27 | toombaloomba | giesen: apparently not, its stuck on Upgrading screen for some reason, but theres no dhcp/tftp/anything available for it to use |
| 21:54.31 | flenders | now the echotraining and echocancel work perfectly! |
| 21:54.40 | giesen | toombaloomba: then you need to provide it |
| 21:54.41 | jart | the thing is |
| 21:54.53 | jart | it happens intermittently |
| 21:54.59 | giesen | your only option might be to hook a console cable to the phone |
| 21:55.02 | toombaloomba | giesen yea i know but its dumb that I cannot enter network settings, stupid cisco |
| 21:55.15 | giesen | I agree |
| 21:55.20 | giesen | but I think you're boned |
| 21:55.33 | giesen | just setup a dhcp/tftp server on your desktop |
| 21:55.35 | toombaloomba | not me, the customer on the phone hahahaha :P |
| 21:55.45 | toombaloomba | yea I've done it many times |
| 21:55.58 | toombaloomba | but now I have to explain to someone remotely how to do it :( |
| 21:56.07 | giesen | ouch |
| 21:56.13 | giesen | Im glad Im not you =) |
| 21:56.18 | toombaloomba | lol :( |
| 21:56.21 | toombaloomba | thanks |
| 21:56.24 | giesen | they dont have dhcp? |
| 21:56.28 | giesen | what kind of heathens are they? |
| 21:57.04 | toombaloomba | dhcp wouldnt help anyway, most broadband routers cannot tell the phones where to look for tftp |
| 21:57.05 | giesen | if a 7961 is anything like a 7960 or a 7970 |
| 21:57.09 | toombaloomba | it has to be tftpd32 on a desktop :( |
| 21:57.19 | giesen | you should be able to hit settings anyways |
| 21:57.26 | giesen | even when it's trying to grab an Ip |
| 21:57.33 | giesen | as long as the network cable is plugged in |
| 21:57.37 | toombaloomba | well you cant hit settings on a 7940 or 7960 if you get protocol application invalid |
| 21:57.38 | giesen | but if it's stuck on upgrading |
| 21:57.47 | giesen | you might be boned. |
| 21:57.56 | toombaloomba | 7941 7961 run same OS as 7970/7971 |
| 21:58.02 | toombaloomba | nothing like 40s/60s |
| 21:58.07 | giesen | yeah |
| 21:58.13 | giesen | Ive played with 70/71 |
| 21:58.18 | giesen | had a few at the office |
| 21:58.20 | giesen | and one here |
| 21:58.23 | toombaloomba | yea me too, i hate them |
| 21:58.30 | giesen | I loved em |
| 21:58.34 | toombaloomba | never got a 70 to work as it should, 71 worked fine |
| 21:58.35 | giesen | other than trying to write the config file |
| 21:58.37 | toombaloomba | (with asterisk) of course |
| 21:58.39 | giesen | make one typo |
| 21:58.45 | giesen | and the whole config is boned. |
| 21:58.48 | toombaloomba | haha yea |
| 21:58.52 | elriah | Unless of course, you have NAT in the picture, then the Cisco's are paper weights. |
| 21:58.56 | giesen | yeah |
| 21:59.06 | giesen | 7970/71 are useless for nat |
| 21:59.14 | giesen | the 40/60 work fine |
| 21:59.14 | elriah | 79x1's |
| 21:59.16 | toombaloomba | 7941/7961 also |
| 21:59.17 | giesen | with the right code. |
| 21:59.23 | toombaloomba | yup, elriah knows :P |
| 21:59.41 | elriah | We ended up doing ipsec tunnels to solve the tftp and nat issues. |
| 21:59.43 | giesen | yeah Ive had my share of cisco nat fun |
| 21:59.48 | giesen | elriah: same |
| 22:00.30 | toombaloomba | thanks cisco!! :) |
| 22:00.34 | elriah | lol |
| 22:00.43 | toombaloomba | and no, I will still not use your damn CCM |
| 22:00.49 | giesen | hahahaha |
| 22:01.16 | mercestes | Funny, A cisco phone, on a cisco switch, running through a cisco router, causes cascading nat translation port assignments that slowly increment until the router memory is filled and it bones yoru nat translations, destroying the router's ability to even route itself. |
| 22:01.44 | mercestes | seen it twice. |
| 22:02.19 | giesen | mercestes: nice |
| 22:02.26 | giesen | there a bug report anywhere on that? |
| 22:03.33 | |kahless| | does anyone use a fritz isdn card? |
| 22:04.38 | JacksLivr | bkruse: still here? |
| 22:04.38 | toombaloomba | all this mess (for me) is thanks to DST, weee! |
| 22:05.03 | mercestes | giesen: not publicly, but if your entire network dies and the router can't even translate it's own ntp..that's probably why. |
| 22:05.12 | mercestes | giesen: We fixed it by putting all the phones of public static IPs. |
| 22:05.50 | mercestes | giesen: But we proably could have fixed it using static internal IPs and static nat routes. |
| 22:05.50 | mercestes | giesen: We had a CCIE come back with some updated firmware that was supposed to fix it but it didn't work. |
| 22:05.53 | giesen | toombaloomba: you figure out how to fix dst on a 79x1? |
| 22:05.53 | bkruse | JacksLivr: ya, sorry, working |
| 22:06.01 | bkruse | wuts up |
| 22:06.07 | JacksLivr | np, can you try and call me on sip? |
| 22:06.10 | bkruse | sure |
| 22:06.15 | giesen | one lovely asterisk bug |
| 22:06.18 | JacksLivr | i have, i think, set all this up |
| 22:06.19 | giesen | I discovered |
| 22:06.24 | bkruse | \/msg me |
| 22:07.17 | MrTelephone | I got my pri working.. zaptel driver had to have a dchan patch |
| 22:07.26 | giesen | is if you have it bound to multiple ips |
| 22:07.28 | bkruse | JacksLivr: /msg bkruse heres my #: blah |
| 22:07.35 | giesen | it will only send stuff out on the src ip |
| 22:07.37 | giesen | erm |
| 22:07.40 | giesen | main ip |
| 22:07.45 | giesen | not the IP it received the traffic on |
| 22:07.53 | giesen | which works wonders for nat |
| 22:07.55 | giesen | *cough* |
| 22:08.30 | mercestes | lol |
| 22:08.31 | ManxPower | giesen: don't bind it to specific IPs |
| 22:08.58 | toombaloomba | giesen: yea, upgrade it to newest firmware, released February 28th (thanks cisco!) |
| 22:09.11 | giesen | toombaloomba: does it actually work? |
| 22:09.15 | toombaloomba | giesen unfortunately its not like the 7940/7960 where you can set the DST start and end in the conf file |
| 22:09.19 | toombaloomba | yup it does |
| 22:09.21 | giesen | okay |
| 22:09.29 | toombaloomba | but it got me into this mess now :( |
| 22:09.36 | toombaloomba | luckily dont have many 79x1 out there |
| 22:09.36 | giesen | my boss is the only one with a 7970 |
| 22:09.41 | giesen | so if I break it |
| 22:09.52 | toombaloomba | heh |
| 22:09.53 | giesen | it might be bad news |
| 22:10.11 | giesen | it's kinda stupid that it's not like the 7960 in that respect |
| 22:10.16 | giesen | being able to set DST in the config |
| 22:10.18 | giesen | is uber handy |
| 22:10.25 | toombaloomba | yea that was an easy find & replace |
| 22:10.31 | giesen | hell |
| 22:10.39 | giesen | edit SIPDefault.cnf, and you're off to the races |
| 22:10.42 | toombaloomba | linksys 941/942 also need new firmware |
| 22:11.01 | giesen | the funny thing is |
| 22:11.06 | giesen | we may go through this all again |
| 22:11.09 | giesen | if they change it back |
| 22:11.15 | toombaloomba | they better not, idiots |
| 22:11.32 | giesen | well all these patches they've written |
| 22:11.36 | giesen | they would have been much better off |
| 22:11.41 | giesen | if they just made it user configurable |
| 22:12.32 | giesen | the 7960s are a shining example of how to do it right |
| 22:12.46 | nDuff | I vaguely recall there being a prefix to use for a channel name to make it be evaluated through the extensions list (so I can always transfer to foo/91234567 and have the call go out through Zap/* or SIP/* or whatever is appropriate). What would that be? |
| 22:13.57 | ManxPower | nDuff: you must be using all softphones |
| 22:14.18 | elriah | The 7941 is a great phone, aside from the NAT issues. I thin |
| 22:14.24 | elriah | Oops... |
| 22:14.37 | elriah | ... If Cisco would fix the firmware and drop the price, they would clean up. |
| 22:14.39 | *** join/#asterisk Euhll (n=Tarnsman@63.245.178.186) |
| 22:14.59 | ManxPower | elriah: Cisco's policies are one of the reasons Polycom is so popular. |
| 22:15.27 | elriah | Yea. Our Polycom (and even Aastra) phones work great. |
| 22:15.45 | data23 | erm, stupid question, but what time is it in america (east coast atm?) just having a debate with someone over daylight savings... did you folks change your clocks already? like 2 weeks early? |
| 22:15.55 | giesen | data23: 18:14 |
| 22:16.02 | giesen | data23: 3 weeks early |
| 22:16.07 | giesen | we did it on 2am sunday |
| 22:16.09 | bkruse | data23: we changed on saturday night (of this past week) |
| 22:16.25 | bkruse | Euhll: all |
| 22:16.29 | data23 | ta |
| 22:16.34 | giesen | data23: who won? |
| 22:16.35 | bkruse | well, ive run it with alot of them, anyways |
| 22:16.39 | data23 | giesen: me :) |
| 22:16.39 | bkruse | theres not a reason is SHOULDNT |
| 22:16.40 | Igbothom_3rd | Queensland (Australia) still doesn't have DST at all - our bloody farmers are scared our cows will fade or something |
| 22:16.48 | giesen | haha |
| 22:16.55 | bkruse | Igbothom_3rd: wtf? lol |
| 22:17.06 | giesen | what state is Sydney in? |
| 22:17.10 | Igbothom_3rd | NSW |
| 22:17.17 | giesen | they did DST for the games, didnt they? |
| 22:17.20 | Igbothom_3rd | no |
| 22:17.22 | giesen | in 2000 |
| 22:17.23 | Euhll | I tried 1.2.5 but got an error in the make process "termcap support not found". Any ideas? |
| 22:17.26 | Igbothom_3rd | they changed it just for the games |
| 22:17.47 | bkruse | Euhll: seriously? thats your fault. |
| 22:17.47 | Igbothom_3rd | by a few weeks or so |
| 22:17.53 | bkruse | yum install libtermcap or termcap |
| 22:17.58 | giesen | and found they actually used more energy |
| 22:18.00 | bkruse | its because YOU dont have the proper dependencies |
| 22:18.08 | giesen | since people were turning on lights in the morning |
| 22:18.29 | Euhll | I'm sure it's my fault, the question is how to fix it. install libtermcap then? |
| 22:18.35 | giesen | what's the stability like on 1.4 nowadays |
| 22:18.41 | giesen | wondering if I should upgrade |
| 22:18.52 | bkruse | Euhll: yep |
| 22:18.59 | Euhll | thanks!! |
| 22:19.08 | ManxPower | Euhll: install termcap-devel |
| 22:19.27 | bkruse | ManxPower: thanks, i couldnt remember the name for the yum/fc package |
| 22:19.31 | Euhll | thanks |
| 22:19.32 | bkruse | i was about to say apt-get :D |
| 22:19.39 | bkruse | Euhll: np, thansk ManxPower |
| 22:19.42 | bkruse | jbot: ManxPower++ |
| 22:20.00 | ManxPower | bkruse: I assumed Mandriva RPM, since I didn't see a distro listed. |
| 22:20.10 | bkruse | ManxPower: ha, good idea |
| 22:20.16 | bkruse | i just love my package manager |
| 22:20.20 | ManxPower | it is prolly libtermcap-devel actually |
| 22:20.28 | bkruse | who knows |
| 22:20.34 | Euhll | ManxPower, termcap-devel is NOT part of the install on FC6 then? |
| 22:20.41 | sudhir492 | I need to program 3 Cisco 7940G (from MGCP 6.4.0 to SIP 8.4.00). Can anyone help me with that. I am offering $50 bountyfor this |
| 22:20.52 | mercestes | ManxPower: your a genius. At what point did Voicemail forwarding work in *? 1.2.16? 1.4? The bug where you try to foward a voicemail that's not in the INBOX. Remember? |
| 22:20.53 | ManxPower | bkruse: URPMI is for RPM what apt-get is for DEB |
| 22:20.55 | elriah | sudhir492: It's easy. |
| 22:21.03 | bkruse | ManxPower: i now, useful too |
| 22:21.05 | ManxPower | Euhll: No idea. |
| 22:21.11 | bkruse | know* |
| 22:21.12 | mercestes | sudhir492: It'll cost you more than $50 I'm certain. |
| 22:21.28 | ManxPower | Euhll: We assume you know how to resolve dependencies |
| 22:21.34 | bkruse | sudhir492: make it 150k and you got a deal. |
| 22:21.35 | giesen | mercestes: I loved when you had a call come into an agent |
| 22:21.38 | mercestes | sudhir492: Anyone with the Cisco certs /access required to get those files to begin with are a bit more expensive than the $50 your offering. |
| 22:21.42 | giesen | through a queue, and then transfered it |
| 22:21.45 | giesen | and it blew up asterisk. |
| 22:21.46 | bkruse | mercestes: yep |
| 22:21.50 | ManxPower | mercestes: no clue. My users have not complained about forwarding to other than INBOX |
| 22:22.00 | bkruse | mercestes: what cisco certs? voip related? or just any? |
| 22:22.02 | mercestes | giesen: yea...Good times...good times. |
| 22:22.08 | giesen | has that been fixed? |
| 22:22.12 | *** join/#asterisk sjobeck (n=sjobeck@199.72.56.200) |
| 22:22.13 | ManxPower | sudhir492: do you have the Cisco SIP firmware? |
| 22:22.15 | mercestes | giesen: I think so. |
| 22:22.26 | giesen | if it has I'll upgrade for that alone |
| 22:22.27 | elriah | sudhir492: Do you have the 8.4 firmware files? |
| 22:22.34 | elriah | Oh, sorry, what ManxPower said. lol |
| 22:22.48 | sudhir492 | I have Cisco 8.4 SIP firmware. But I do not have SIP 6.4 if you need that too |
| 22:22.56 | mercestes | giesen: Actually< i know it has as of 1.2.13 because we transfer out of queues all the time. |
| 22:23.24 | mercestes | sudhir492: you need the P0S3-07-*-* firmware to go to that first, then to P0S3-8-*-* |
| 22:23.39 | giesen | mercestes: do you use agents? |
| 22:23.55 | giesen | it only manifests itself when an *agent* transfers a call out of a queue |
| 22:24.00 | bkruse | giesen: agents roxor, but there will be a new way to do agents and queues(if there isnt already) |
| 22:24.01 | sudhir492 | mercestes: I do not have P0S3-07-* |
| 22:24.04 | mercestes | giesen: Members. |
| 22:24.30 | giesen | yeah |
| 22:24.33 | giesen | you're not using agents |
| 22:24.40 | giesen | we cant use them either |
| 22:24.43 | giesen | because of that bug. |
| 22:24.44 | Euhll | bkruse and ManxPower, I installed termcap-devel and ran "make clean" and "make" again. the make process still tells me "configure:error: termcap support not found. what am I missing? |
| 22:25.01 | giesen | Im talking about agents taht can log on and off |
| 22:26.08 | bkruse | Euhll: sh configure, or ./configure |
| 22:26.34 | bkruse | giesen: http://bugs.digium.com |
| 22:27.18 | giesen | yeah I have the bugid written down somewhere |
| 22:27.23 | giesen | havent followed up on it in a while |
| 22:27.25 | *** part/#asterisk Euhll (n=Tarnsman@63.245.178.186) |
| 22:27.28 | ManxPower | Euhll: and "rpm -qa | grep termcap" shows it as installed. |
| 22:27.31 | sudhir492 | mercestes: do you have P0S3-07-*-*? |
| 22:27.43 | ManxPower | I don't see any other reports of finding termcap for asterisk |
| 22:27.56 | *** join/#asterisk Euhll (n=Tarnsman@63.245.178.186) |
| 22:27.56 | bkruse | ManxPower: its probably because he has to reconfigure to look for termcap |
| 22:28.04 | ManxPower | Euhll: what version of asterisk again? |
| 22:28.14 | Euhll | ok, what did I miss? |
| 22:28.16 | sudhir492 | is wiki down? |
| 22:28.21 | ManxPower | bkruse: "make" always ran configure for me |
| 22:28.37 | ManxPower | (17:27:18) ManxPower: Euhll: and "rpm -qa | grep termcap" shows it as installed. |
| 22:28.43 | ManxPower | (17:27:33) ManxPower: I don't see any other reports of finding termcap for asterisk |
| 22:29.23 | ManxPower | sudhir492: Yes! |
| 22:29.42 | sudhir492 | ManxPower: Can you send it to me? |
| 22:30.04 | Euhll | I get back: libtermcap-2.0.8-46.1 and termcap -5.5-1.12006...... |
| 22:30.24 | bkruse | ManxPower: in 1.4? |
| 22:30.30 | ManxPower | Euhll: See http://lists.digium.com/pipermail/asterisk-users/2003-May/003970.html |
| 22:30.36 | bkruse | not after you have already run it and it makes makefile.ops or w/e its called |
| 22:30.47 | ManxPower | bkruse: I would have to insane to run 1.4 |
| 22:31.04 | ManxPower | ..er.. |
| 22:31.07 | ManxPower | I would have to |
| 22:31.11 | ManxPower | BE insane to run 1.4 |
| 22:31.23 | Euhll | ManxPower thanks for the url, I'll check it out! What the current version of * to be running? the last 1.12 or the new 1.4? |
| 22:31.40 | mercestes | sudhir492: I can get it |
| 22:31.51 | ManxPower | Euhll: I recommend 1.2.x for production servers. |
| 22:32.02 | Euhll | 1.2.x it is then! THANKS |
| 22:32.02 | ManxPower | 1.4 has not been out long enough for me to put it on a production server |
| 22:32.22 | mercestes | sudhir492: Will have it latre today as a matter of fact. |
| 22:32.59 | bkruse | ManxPower: gotcha, 1.2 doesnt have a configure script |
| 22:33.45 | *** join/#asterisk mmartinn (n=martin@adsl-065-005-200-225.sip.gnv.bellsouth.net) |
| 22:34.19 | giesen | ooh |
| 22:34.25 | giesen | it was fixed as of 1.2.12 |
| 22:34.33 | giesen | http://bugs.digium.com/view.php?id=7458 |
| 22:35.26 | toombaloomba | i really wish using different MOH for different channels/calls/whatever worked in 1.2.xx |
| 22:36.12 | giesen | it does for queues |
| 22:36.15 | ManxPower | bkruse: thew included editline DOES have a configure script, which what confused me |
| 22:36.39 | ManxPower | toombaloomba: in what way does it not work? |
| 22:36.45 | giesen | anyone actually used SLA in 1.4? |
| 22:37.10 | nDuff | Hmm. I used to rely on the behavior of DBget() jumping based on whether the item was found; how do I replicate that with Set(foo=${DB(bar/baz)})? |
| 22:37.16 | ManxPower | giesen: the broken SLA in 1.4.0 or the rewritten SLA in 1.4.1? |
| 22:37.29 | ManxPower | nDuff: no mention in UPGRADE.txt? |
| 22:37.32 | nDuff | ManxPower: it appears that what I was looking for was Local/* |
| 22:37.46 | nDuff | [previously, not just now] |
| 22:37.48 | giesen | 1.4.1 |
| 22:38.17 | toombaloomba | ManxPower: I cant have different MOH classes in the .conf and then musicclass=blah in sip.conf or even SetMusicOnHold(blah) to get it to play, I've tried all sorts of things and it always plays the default one |
| 22:38.41 | nDuff | ManxPower: not anything answering that question in UPGRADE.txt, no; it just says that DBGet is gone. |
| 22:38.58 | ManxPower | nDuff: How terribly helpful of it. |
| 22:39.34 | ManxPower | toombaloomba: if it was a problem one would think more than just you would have experienced it. |
| 22:40.18 | toombaloomba | ManxPower yea i've spoken about it here before, and some people said it worked while others agreed it didnt, no idea what the deal is |
| 22:40.18 | mercestes | Does\ someone have time to help me with my zaptel config? I have 3 PRIs configured thusly. http://pastebin.ca/393929 |
| 22:40.24 | *** join/#asterisk dseeb_ (n=dcb@58.169.130.113) |
| 22:40.44 | giesen | homey home |
| 22:42.36 | ManxPower | nDuff: DB_EXISTS DB_EXISTS(<family>/<key>) Check to see if a key exists in the Asterisk database |
| 22:42.36 | ManxPower | DB DB(<family>/<key>) Read from or write to the Asterisk database |
| 22:43.21 | ManxPower | nDuff: or better yet "pbx-1*CLI> show function DB" |
| 22:43.25 | *** join/#asterisk Euhll (n=Tarnsman@63.245.178.186) |
| 22:43.39 | Euhll | Sorry, keep getting kicked off |
| 22:45.34 | nDuff | hrm. |
| 22:45.51 | nDuff | right now, I'm trying to figure out why exten => _7676,2,GotoIf($["${fwd}"=""], 50) doesn't jump to 50 when fwd is empty. |
| 22:46.09 | ManxPower | nDuff: syntax error |
| 22:46.36 | Euhll | ManxPower, can you assist with my "cannot find -lssl" error? |
| 22:46.47 | Euhll | or recommend someone who can? |
| 22:46.51 | ManxPower | exten => _7676,2,GotoIf($["${fwd}"=""]?50) would jump to priority 50 |
| 22:46.51 | Qwell[] | Euhll: what distro? |
| 22:46.57 | Euhll | FC6 |
| 22:47.03 | Qwell[] | yum install openssl-dev |
| 22:47.04 | ManxPower | Euhll: reread that URL I sent you |
| 22:47.07 | Qwell[] | or devel...I forget |
| 22:47.56 | nDuff | ahh. |
| 22:48.25 | ManxPower | nDuff: But I would use exten => _7676,2,GotoIf(($[${LEN(${fwd})} != 0]?50) |
| 22:48.31 | ManxPower | or = 0 at least |
| 22:49.06 | ManxPower | I also have two (( where there should be 1 (. That's what I get for pasting part and writing part |
| 22:49.18 | *** join/#asterisk tsurko (n=tsurko@vpn-pppoe-213-240-242-237.megalan.bg) |
| 22:49.27 | ManxPower | This should be correct: exten => _7676,2,GotoIf($[${LEN(${fwd})} = 0]?50) |
| 22:52.38 | ManxPower | nDuff: here's come complex macros and contexts if you want to use them as an example of doing some stuff |
| 22:52.48 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 22:52.56 | *** join/#asterisk kFuQ (n=somedude@c-67-185-123-34.hsd1.wa.comcast.net) |
| 22:53.11 | JacksLivr | im not sure what happened. [s@incoming:1] Answer("Zap/3-1", "") all of a sudden, it's not getting caller id |
| 22:53.27 | *** join/#asterisk bartpbx (n=bartpbx@pD9E88736.dip0.t-ipconnect.de) |
| 22:53.33 | Euhll | Qwell I installed openssl-dev like you said and it came back "nothing to do" |
| 22:53.45 | JacksLivr | stopped working about an hour ago. i put things back like they were and stopped and started zaptel and * |
| 22:54.15 | ManxPower | JacksLivr: you must be on analog ports |
| 22:54.20 | gmfm | is anyone else having problems with phpagi scripts after the DST change? |
| 22:54.43 | Qwell[] | gmfm: Only people who didn't update their systems timezone settings properly |
| 22:55.07 | JacksLivr | yeah, analog port. was working an hour ago though |
| 22:55.25 | ManxPower | JacksLivr: playing with rxgain and txgain can cause issues. |
| 22:55.27 | JacksLivr | i plugged a phone into the wall and the caller is getting sent |
| 22:55.57 | gmfm | Qwell[]: my system got the timezone update... but for some odd reason it sends "Invalid or unknown command" when using get_variable on certain things |
| 22:56.19 | Qwell[] | I highly doubt that has anything at all to do with timezones |
| 22:56.53 | Euhll | Qwell now I'm trying openssl-devel and that IS getting somewhere! thanks!! |
| 22:57.22 | gmfm | Qwell[]: that's basically what I thought, but this script has been untouched since october and was working up until last weekend... oh well |
| 22:57.56 | JacksLivr | ManxPower: i did have rxgain set. i undid that and started over. did not fix the problem |
| 22:58.09 | ManxPower | JacksLivr: It sucks to be you |
| 22:58.20 | JacksLivr | stopped and started both zaptel and asterisk |
| 22:58.27 | JacksLivr | ManxPower: some days it does |
| 22:58.46 | Euhll | Qwell[] Sorry about that! I wasn't watching your full nick! |
| 23:01.24 | nDuff | ManxPower: I appreciate examples to look at -- but I think I missed your link. |
| 23:02.34 | *** join/#asterisk carrar (i=tim@osburn.com) |
| 23:03.34 | saftsack | has someone else asterisk running on an embedded router? |
| 23:04.15 | JunK-Y | saftsack: yes |
| 23:04.16 | Euhll | Qwell[] that got me farther, but I have a new error. Linux/compiler.h: No such file or directory. What am I missing now? |
| 23:04.52 | saftsack | JunK-Y, openwrt? |
| 23:04.56 | JunK-Y | yes |
| 23:05.13 | JunK-Y | it rocks! |
| 23:05.52 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 23:07.04 | quidpro | Is there a way to get * to pass-through G729 *and* G711? From what i've read in the wiki... you can only pass through properly if G729 is the only allowed codec. (ie. disallow=all, allow=g729) |
| 23:08.16 | JacksLivr | what can i do to restore callerid. to normalize everything, i rebooted the server. still no callerid. |
| 23:10.57 | *** join/#asterisk CrazyTux[m] (n=CrazyTux@c-67-188-233-116.hsd1.ca.comcast.net) |
| 23:12.11 | JacksLivr | ahhhh, i think callerid broke when i upgraded from 1.2 to 1.4 |
| 23:13.08 | *** part/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net) |
| 23:13.10 | JacksLivr | before upgrade "-- Executing NoOp("Zap/3-1", "2565551212")" after "Executing [s@incoming:2] NoOp("Zap/3-1", "")" |
| 23:13.33 | flenders | hey, does fxotune work with TE110P cards? |
| 23:14.27 | *** join/#asterisk Ahrimanes (n=ma@x1-6-00-0a-e4-2e-90-43.k707.webspeed.dk) |
| 23:14.57 | Ahrimanes | any 1.4.1 users using ubuntu ? |
| 23:15.40 | russellb | my dev machine is running ubuntu ... |
| 23:15.43 | russellb | what's the problem? |
| 23:16.01 | Ahrimanes | ./configure fails on termcap support |
| 23:16.10 | russellb | install libncurses-dev |
| 23:16.17 | Ahrimanes | ah thx |
| 23:16.24 | russellb | you're welcome |
| 23:16.29 | Qwell[] | on ubuntu it's libncurses5-dev |
| 23:16.37 | russellb | whatever :-p |
| 23:16.37 | Qwell[] | silly debian :P |
| 23:16.51 | |kahless| | does anyone use a fritz isnd card with the misdn drivers? |
| 23:16.51 | Ahrimanes | hehe |
| 23:17.06 | russellb | Qwell[]: libncurses-dev is an alias for it :-p |
| 23:17.07 | Ahrimanes | Qwell[]: it figured that part out on it's own :D |
| 23:17.16 | Qwell[] | silly debian! |
| 23:17.22 | russellb | debian pwns |
| 23:17.24 | russellb | don't lie. |
| 23:17.27 | Qwell[] | I tried to dpkg -s it, heh |
| 23:17.42 | Qwell[] | Package `libncurses-dev' is not installed and no info is available. |
| 23:17.44 | Ahrimanes | hehe |
| 23:18.00 | Ahrimanes | russellb: i really should get around to testing the devstate stuff you did |
| 23:18.11 | Qwell[] | You really should |
| 23:18.14 | russellb | Ahrimanes: yes you should! |
| 23:18.19 | russellb | Ahrimanes: you see my blog on *.org about it? |
| 23:18.49 | Ahrimanes | russellb: hm dont think so, link? |
| 23:18.51 | russellb | i made an example for controlling a light :) |
| 23:18.54 | russellb | it's on the main page of *.org |
| 23:18.58 | russellb | "Custom Device State" |
| 23:19.18 | Ahrimanes | you could have pasted a link.. now i had to type myself ;) |
| 23:19.31 | russellb | but then i would have had to type it |
| 23:19.34 | russellb | and i didn't feel like it |
| 23:19.53 | russellb | because i'm trying to fix other things right now :) |
| 23:20.28 | Ahrimanes | hehe |
| 23:20.37 | mercestes | I'm trying to fix my pris |
| 23:21.04 | saftsack | JunK-Y, yes same opinion here :) |
| 23:21.18 | saftsack | do you have the wl-500gp from asus too? |
| 23:22.03 | Ahrimanes | russellb: did you talk to oej about devstate? |
| 23:22.55 | russellb | nope |
| 23:23.05 | Ahrimanes | ok |
| 23:23.09 | Euhll | Guys (and gals) I have installed * on FC4 with NO problems, am I better off going back to FC4 as opposed to messing around with FC6? |
| 23:23.10 | russellb | was i supposed to? |
| 23:23.22 | Ahrimanes | he has some comments on memory allocation, but not sure how important it is, hehe |
| 23:23.26 | mercestes | what does a signalling=pri_cpe connect to? |
| 23:23.34 | russellb | Ahrimanes: memory allocation in my code, or in general? |
| 23:23.42 | mercestes | I get a PRI Error: We think we're the CPE but they think they're the CPE too. |
| 23:23.42 | Ahrimanes | russellb: in yours i think |
| 23:23.50 | mercestes | what signalling type should I use? |
| 23:23.58 | Ahrimanes | mercestes: pri_net |
| 23:23.59 | Ahrimanes | i guess |
| 23:24.00 | russellb | well, I would be happy to hear about it ... he hasn't said anything to me, though |
| 23:24.11 | ManxPower | mercestes: that can be caused by 2 things. you need to be pri_net OR the telco has a loopback on the line |
| 23:24.19 | Ahrimanes | russellb: nah, he's way busy travelling unfortunately |
| 23:24.45 | *** join/#asterisk testdriver12500 (n=pete@adsl-072-151-080-066.sip.rmo.bellsouth.net) |
| 23:25.10 | mercestes | ManxPower: Bingo |
| 23:25.22 | mercestes | ManxPower: "We think we're the network, but they think they're the network too." |
| 23:25.23 | mercestes | bastards. |
| 23:25.28 | shido6 | lol |
| 23:25.43 | ManxPower | mercestes: Damn I'm good. |
| 23:25.56 | Ahrimanes | haha |
| 23:26.19 | Ahrimanes | "Here's your E1 link.. you _MAY_ have trouble calling out.." |
| 23:26.37 | ManxPower | Everytime I make a call someone calls me! |
| 23:28.07 | ManxPower | Euhll: I recommend that you stop trying to use Asterisk and learn Linux first. |
| 23:28.09 | xheliox | Euhll: You're better off dumping Fedora and going with Centos. |
| 23:28.10 | Ahrimanes | Euhll need to type /me less |
| 23:28.39 | mercestes | ManxPower: I still worship you. :) and yea, damn your good. ;) |
| 23:28.39 | xheliox | Euhll: Centos is a clone of RHEL, where as Fedora is more of a bleeding edge, short life span development distribution. |
| 23:29.03 | Euhll | Well i appreciate all the advice. |
| 23:29.54 | *** join/#asterisk sharp (i=sharp@outbound.silenceisdefeat.org) |
| 23:32.19 | JacksLivr | AHA! figired it out. $CALLERID was the variable that i had set up in 1.2 and it worked there. i changed all that t$CALLERID(NUM) and it is happy |
| 23:34.13 | mercestes | oh well, gtg now. Thanks Manx. Byes |
| 23:34.54 | testdriver12500 | I have a challenge. Running 1.4.1 on Ubuntu 6.06 LTS. asterisk and zaptel compiled fine. zaptel loads no problem. zapata barfs, however. When I try do a module reload chan_zap, I get an error "unable to reconfigure channel". http://pastebin.com/898532 has the error and configs. |
| 23:35.04 | testdriver12500 | Any takers?? |
| 23:35.24 | testdriver12500 | chan_zap.so |
| 23:37.05 | Ahrimanes | russellb: devstate stuff is just an addon to the 1.4 source right? |
| 23:37.30 | russellb | eh? |
| 23:37.34 | russellb | which part of it |
| 23:41.00 | Ahrimanes | russellb: well wanted to throw it into this 1.4.1 install to test... |
| 23:41.59 | russellb | oh, ok, the module. Yes, you can just copy it into 1.4 and it will work |
| 23:42.11 | russellb | you have to actually build it against 1.4, though |
| 23:42.21 | russellb | so ... cp trunk/funcs/func_devstate.c 1.4/funcs/ |
| 23:42.27 | russellb | and then it will be automagically compiled and installed |
| 23:43.29 | Ahrimanes | ah |
| 23:43.30 | *** join/#asterisk Ryushin (i=proxy@windwalker.openinnovations.com) |
| 23:43.33 | Ahrimanes | just need a checkout |
| 23:44.06 | *** join/#asterisk coppice (n=chatzill@90.203.17.210.dyn.pacific.net.hk) |
| 23:44.15 | Ahrimanes | hm a via 1 ghz isnt lightning fast to compile on |
| 23:46.00 | JunK-Y | what do ya expect? |
| 23:46.21 | mihinomenest | where's [TK]D-Fender ? |
| 23:46.21 | Ahrimanes | JIT :P |
| 23:51.24 | *** join/#asterisk Moobius (i=Moobius@www2.techcavalry.com) |
| 23:53.34 | quidpro | Is there a way to get * to pass-through G729 *and* G711? From what i've read in the wiki... you can only pass through properly if G729 is the only allowed codec. (ie. disallow=all, allow=g729) |
| 23:55.46 | ManxPower | quidpro: both legs of the call has to be the same codec or asterisk will have to transcode. Also if asterisk has to do anything with the audio it will have to transcode G729-SLIN-G729 to listen to the audio |
| 23:57.25 | Ahrimanes | hm russellb a copy and then make doesnt go to well with 1.4.1 |
| 23:57.25 | quidpro | Hmm, when I tried using G729 on passthrough... I tended to get the CLI flooded with warnings of "can't find a path from x to y" |
| 23:57.38 | quidpro | Although both legs were in G729 |
| 23:57.50 | ManxPower | quidpro: what were the DIAL options? |
| 23:57.55 | *** join/#asterisk rhombus (n=rhombus@S01060006257edf62.cg.shawcable.net) |
| 23:58.13 | rhombus | has anybody noticed that the wiki is down? |
| 23:58.18 | ManxPower | t,T,w,W,r, Monitor, ChanSpy, and a zillion other things would cause asterisk to stay in the audio path |
| 23:58.22 | ManxPower | rhombus: yes the wiki is down |
| 23:58.24 | Moobius | rhombus: yea. |
| 23:58.55 | rhombus | have there been any announcements with more information? |
| 23:58.56 | ManxPower | quidpro: cant' find path from x to y would tell you what the two legs of the call are unless it is SLIN in which case asterisk wants to be in the audio path |
| 23:59.18 | quidpro | Manx: Hmm, let me look |
| 23:59.35 | rhombus | What format should music on hold files be in? |
