| 00:02.49 | hmmhesays | yeah but they book quickly |
| 00:03.24 | hmmhesays | *boot even |
| 00:03.24 | dan__t | Hrm. |
| 00:03.34 | hmmhesays | why isn't asterisk finding my unixODBC libs! |
| 00:03.38 | dan__t | What would be neat is a live Asterisk distribution that can mount its configs via NFS or some such |
| 00:03.42 | dan__t | Anyone heard anything about that? |
| 00:03.46 | JT | nfs |
| 00:03.52 | JT | you mean an sql database? |
| 00:04.16 | elixer | no, he means network file system |
| 00:04.35 | dan__t | sql would be rad, too. |
| 00:04.42 | JT | why would you want to do that, elixer ? |
| 00:04.44 | JT | illogical |
| 00:04.50 | dan__t | I don't have any more disks to throw in this * box that I want to set up, so i thought I'd learn something here |
| 00:05.04 | elixer | dan__t: why can't you just do /sbin/asterisk -C /some/mount/point/asterisk.conf |
| 00:05.04 | JT | voicemail by nfs perhaps, but not configs |
| 00:05.52 | dan__t | Noted. |
| 00:05.53 | *** join/#asterisk Ebola (n=Ebola@host86-143-7-120.range86-143.btcentralplus.com) |
| 00:05.58 | dan__t | elixer, diskless workstation. |
| 00:06.04 | dan__t | Suppose I could spin and boot a livecd |
| 00:06.10 | elixer | ohhh |
| 00:06.23 | elixer | yeah, that i dunno |
| 00:06.31 | JT | you could do it, just not sure if it has benefits over using sql |
| 00:06.50 | dan__t | What can I store in SQL? |
| 00:06.58 | dan__t | Well, aside from the obvious - what relating to * can I store in SQL? |
| 00:07.00 | dan__t | That sounds interesting. |
| 00:08.09 | hmmhesays | anything you want |
| 00:08.15 | hmmhesays | especially with func_odbc |
| 00:08.20 | JT | configs |
| 00:08.24 | JT | ARA |
| 00:08.35 | hmmhesays | naked pictures of your girlfriend |
| 00:08.47 | hmmhesays | that you can have mms's to your friends phones when you find her on a date at your favorite bar |
| 00:08.56 | hmmhesays | er.. |
| 00:09.12 | wothinn | Maybe it shows my age, but I prefer naked pictures of his mother. |
| 00:09.32 | hmmhesays | sure you can store those too |
| 00:10.01 | TJNII | WTH. I chance screens for 10 seconds and the conversation goes from NFS mounts to naked pictures of hmmhesays's mom |
| 00:10.25 | hmmhesays | i'm good like that |
| 00:10.42 | elixer | it was a quick segue. only took one message. |
| 00:10.50 | TJNII | I notices |
| 00:10.58 | TJNII | s/notices/noticed |
| 00:11.18 | elixer | jbot? |
| 00:11.28 | elixer | s/jbot/alive/ |
| 00:11.32 | elixer | ah |
| 00:11.37 | elixer | its all about the trailing slash |
| 00:11.53 | Qwell | // |
| 00:11.59 | Qwell | s//// |
| 00:12.03 | TJNII | s/// |
| 00:12.05 | Qwell | s/\/// |
| 00:12.07 | Qwell | damn bot |
| 00:12.18 | elixer | s/.*/f/ |
| 00:12.28 | elixer | damn you jbot |
| 00:12.46 | elixer | s/damn/love/ |
| 00:12.57 | Qwell | s/ / you j/ |
| 00:13.01 | Qwell | :D |
| 00:13.11 | *** join/#asterisk ming_zym (i=ming_zym@nat/yahoo/x-3ca2e65e64319fda) |
| 00:13.18 | elixer | very nice |
| 00:13.24 | elixer | s/nice/lame/ |
| 00:13.27 | elixer | ;-) |
| 00:14.00 | elixer | he doesn't do regex. that's upsetting. |
| 00:14.06 | Qwell | jbot: sedtest is <reply> s/lame/slick/ |
| 00:14.07 | jbot | okay, Qwell |
| 00:14.08 | elixer | s/./\./ |
| 00:14.15 | Qwell | you broke it ;p |
| 00:14.27 | elixer | did i? |
| 00:14.30 | Qwell | mmhmm |
| 00:14.32 | elixer | how? |
| 00:14.41 | Qwell | jbot: sedtest is <reply> s/upsetting/lame/ |
| 00:14.41 | jbot | ...but sedtest is already something else... |
| 00:14.46 | Qwell | jbot: no, sedtest is <reply> s/upsetting/lame/ |
| 00:14.46 | jbot | okay, Qwell |
| 00:14.58 | Qwell | jbot: no, sedtest is <reply> s/okay/no thanks/ |
| 00:14.58 | jbot | okay, Qwell |
| 00:15.00 | Qwell | ~sedtest |
| 00:15.01 | jbot | s/okay/no thanks/ |
| 00:15.07 | Qwell | meh |
| 00:15.27 | Qwell | no internal infinite loops, heh |
| 00:15.30 | elixer | jbot: i love you |
| 00:15.30 | jbot | You love you? |
| 00:15.38 | elixer | jbot: no, i love YOU |
| 00:16.47 | elixer | i still don't see how i broke him |
| 00:17.14 | TJNII | Your love of him was probably too rough. |
| 00:17.20 | elixer | usually is |
| 00:18.32 | *** join/#asterisk WilliamK (n=noc@static-71-170-144-3.dllstx.fios.verizon.net) |
| 00:19.53 | WilliamK | anyone else see NAT = N on the column (CLI) when doing a sip show peers... (phone is really behind NAT) |
| 00:21.44 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 00:22.15 | hmmhesays | thats because N says its using nat |
| 00:22.26 | WilliamK | I thought it was supposed to be Y |
| 00:22.29 | WilliamK | and N was no |
| 00:23.11 | *** join/#asterisk grndslm (n=grndslm@24-116-87-97.cpe.cableone.net) |
| 00:23.25 | hmmhesays | blank is no |
| 00:23.39 | hmmhesays | like D is dynamic, N is nat |
| 00:24.01 | WilliamK | ah |
| 00:24.46 | WilliamK | know where to find templates for the cisco 7940/60s? |
| 00:27.17 | TJNII | Asterisk reads the dialplan in order, correct? So if I have a number that matches two extensions it will ring the first match, correct? |
| 00:27.43 | hmmhesays | that is completely wrong |
| 00:27.59 | TJNII | Okay. Give me a minute to reword my question, then. |
| 00:30.49 | TJNII | Allright, I have a plan that is free for in state calls. So in my dialplan I have a couple extensions that match 9, an allowed area code, and the rest of the number (_91319XXXXXXX for 319). If none of the extensions are matched and the extension dialed starts with 9 I want a "no long distance" message to play. How would I do that? |
| 00:34.27 | *** join/#asterisk PepOSX (n=pepOSX@190.72.152.245) |
| 00:35.47 | TJNII | I was thinking that maybe I could put _9. after all the other extensions, but if it doesn't parse the context in order I can't. |
| 00:36.08 | *** join/#asterisk jablko (n=jablko@gallery/soc/jablko) |
| 00:42.34 | *** join/#asterisk saftsack (n=saftsack@pD9E05496.dip.t-dialin.net) |
| 00:47.42 | *** join/#asterisk rmayorga (n=rmayorga@unaffiliated/rmayorga) |
| 00:48.51 | TJNII | Well I tried my idea and it seems to work. |
| 00:49.20 | TJNII | hmmhesays: If you take my first question and replace dialplan with context, is it then correct? |
| 00:49.51 | *** join/#asterisk coppice (n=chatzill@234.155.17.210.dyn.pacific.net.hk) |
| 00:52.35 | *** join/#asterisk red9012 (n=marc3234@76-10-149-62.dsl.teksavvy.com) |
| 00:52.58 | red9012 | so far I havent been able to find a single reliable itsp. anyone has any suggestions? |
| 00:53.59 | TJNII | Me and my friend Scott are using Broadvoice. I've only had it a day, but he's been running it for a month and hasn't complained. |
| 00:54.23 | *** join/#asterisk mitcheloc (n=mitchel@adsl-68-120-230-95.dsl.irvnca.pacbell.net) |
| 00:55.31 | dijungal | hi does asterisk have a "call-limit" option in SIP, to restrict only one call at a time to an extension? |
| 01:00.19 | *** join/#asterisk wunderkin (n=wunderki@ip72-223-86-126.ph.ph.cox.net) |
| 01:02.26 | coppice | wouldn't there need to be a reliable internet before there could be a reliable provider of service over it? :-\ |
| 01:03.10 | dijungal | yes |
| 01:03.11 | *** join/#asterisk blinky42 (n=me@c-71-230-47-244.hsd1.pa.comcast.net) |
| 01:03.26 | dijungal | how can i limit the qty of calls going to an extension? |
| 01:03.41 | dijungal | if the extension is off hook a call should not go to it |
| 01:06.01 | TJNII | dijungal: My sip phones have a no call waiting option for that. |
| 01:15.01 | giesen | I've tried everything |
| 01:15.13 | giesen | but I cant for the life of me get this sangoma a101d to work properly |
| 01:15.18 | giesen | any one have any experience with them? |
| 01:16.32 | *** join/#asterisk MaliutaWrk (n=nikolai@fw.hitwise.com) |
| 01:21.25 | coppice | you might be more response if you said specifically what problem you have |
| 01:22.20 | giesen | my pri card goes up and down |
| 01:22.25 | giesen | and when it actually works |
| 01:22.30 | giesen | it crackles |
| 01:22.47 | *** join/#asterisk famicon (i=pastry@c51447ddc.cable.wanadoo.nl) |
| 01:23.09 | *** join/#asterisk rmayorga (n=rmayorga@unaffiliated/rmayorga) |
| 01:23.31 | coppice | crackes, or clicks? |
| 01:23.45 | giesen | well first I was getting clicks |
| 01:23.47 | giesen | and no sound at all |
| 01:23.58 | giesen | and I've messed around with the drivers a bit |
| 01:24.00 | giesen | and now I get sound |
| 01:24.04 | giesen | but it randomly drops calls |
| 01:24.13 | giesen | and there's crackling on the line |
| 01:24.48 | coppice | crackle is strange, but clicks are probably caused by selecting the sync source incorrectly |
| 01:24.57 | giesen | it's set to normal |
| 01:25.29 | giesen | Write to 36 failed: Unknown error 500 |
| 01:25.29 | giesen | Short write: 0/15 (Unknown error 500) |
| 01:25.37 | giesen | that's what happens when it drops the call |
| 01:25.47 | giesen | I get that on the asterisk console |
| 01:29.45 | *** join/#asterisk chendy (n=chendy@218.242.110.26) |
| 01:30.32 | *** join/#asterisk Strom_M (n=strom@216.64.24.250) |
| 01:31.55 | TJNII | So Asterisk reads a context in the dialplan in order, correct? So if I have a number that matches two extensions it will ring the first match, correct? |
| 01:32.23 | TJNII | I asked a variation of this above, and I think I know why I was wrong before. If I'm still wrong I don't know why. :P |
| 01:35.30 | viKing78 | What methods are available to create a distributed Voicemail implementation? I like for branch Asterisk servers to be able to transfer messages to other branch servers, is that possible? |
| 01:38.53 | TJNII | A distributed asterisk system? Boxes a, b, and c can all handle mailbox x? |
| 01:39.02 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 01:39.55 | viKing78 | Yes, but also if user x on server b want's to transfer a message to user a that is on server a. |
| 01:41.15 | TJNII | I don't know. I'm still pretty green, but sharing the mailbox directory between a, b, and c might work. |
| 01:41.23 | TJNII | Don't hold me to that. |
| 01:43.21 | viKing78 | Yeah, I've thought about that but I have seprate sites. NFS over a WAN is not something I want. |
| 01:43.51 | TJNII | Heh. I can see that. |
| 01:44.00 | viKing78 | Also my idea is to try and have a survivable system in each site if something happens to the internet connection in a branch office. |
| 01:44.26 | viKing78 | Each site has it's own Asterisk and PSTN connection |
| 01:44.58 | TJNII | I think I have read about a file system like this before, but I can't remember where. |
| 01:45.43 | viKing78 | I'm interested in the IMAP backend in 1.4. Might be able to use something like Cyrus IMAP with it's Murder system for replication. |
| 01:46.12 | WilliamK | viKing78, why not just use a cellular side for backup |
| 01:46.12 | WilliamK | =) |
| 01:46.49 | WilliamK | in most cases, if internet goes down at a site it's also going to loose dialtone (cable cuts) |
| 01:48.18 | TJNII | He'd lose outgoing lines, but his internal PBX would still work off the local asterisk server. |
| 01:49.05 | viKing78 | True, but most cable cuts don't take down POTS lines in my experience. Also some of these sites are using DSL/Cable which is not exactly a stable platform even when the cable is intact! :) |
| 01:50.12 | TJNII | So you want this to work over POTS, then? |
| 01:50.34 | viKing78 | No POTS is just for incoming |
| 01:50.39 | TJNII | oh |
| 01:50.45 | *** join/#asterisk los415 (n=los415@cpe-76-171-125-207.socal.res.rr.com) |
| 01:51.17 | TJNII | I was going to say, what you are talking about is going to be a trick, but over POTS? Damn. |
| 01:51.57 | viKing78 | No most of the sites are using a T1/FracT1/DSL/Cable Inet connection |
| 01:52.57 | *** join/#asterisk zeedo (n=zeedo@reboot-robot.net) |
| 01:53.03 | viKing78 | I do have two main sites that are using redundant T3+ connection. I'm thinking about probably just setting up a replicated centralized VM server and have all the branches use that. |
| 01:53.47 | viKing78 | That way if the branches loose inet they still can call in and out just no VM. |
| 02:05.01 | *** join/#asterisk r0d3nt (i=nobody@foster.stonedcoder.org) |
| 02:09.19 | J4k3 | centralized pbx.... sounds like hell |
| 02:10.04 | *** join/#asterisk dug (n=chatzill@c-76-102-23-25.hsd1.ca.comcast.net) |
| 02:12.11 | dug | I have a playback problem and I am sure it is a zaptel timing issue, how can I be sure that asterisk is pointed to my card for hw timing and not ztdummy? |
| 02:18.03 | *** join/#asterisk tengulre (n=tengulre@222.90.66.10) |
| 02:30.47 | ManxPower | dug: do you have ztdummy loaded? See the ourput of "lsmod" |
| 02:31.45 | *** join/#asterisk ZX81 (n=matt@202.20.97.200) |
| 02:31.46 | dug | ManxPower: yes |
| 02:32.18 | *** join/#asterisk fujin (n=aj@unaffiliated/fujin) |
| 02:32.19 | ManxPower | don't load ztdummy if you have a zaptel or compatible card. |
| 02:32.22 | ZX81 | hi all, anyone know if g726 works? I'm trying to make a call between two asterisk boxes via iax - all codecs except for g726 seem to work (it results in static) |
| 02:32.40 | ZX81 | one is asterisk 1.2 and the other is 1.4 |
| 02:32.51 | ZX81 | do I need to set a packetisation value on the 1.4 box? |
| 02:32.56 | ZX81 | or are both just 20ms? |
| 02:33.17 | ManxPower | I don't use 1.4 |
| 02:33.20 | ZX81 | :) |
| 02:33.50 | dug | ManxPower: ok ... also I have zttranscode loading... do I need this? |
| 02:34.22 | ManxPower | dug: no, but it doesn't seem to hurt. See the README in the Zaptel source dir for a description of the various kernel modules. |
| 02:35.29 | ZX81 | ooh customer here brb |
| 02:44.19 | *** join/#asterisk _10nix_ (n=hyjnx@user-160u96o.cable.mindspring.com) |
| 02:50.55 | *** join/#asterisk lowlevel (n=Stuart@CPE0017f2e2a3e8-CM000f9f7d6742.cpe.net.cable.rogers.com) |
| 02:57.42 | *** join/#asterisk Krurst (n=icr_lol_@eth244.wa.adsl.internode.on.net) |
| 02:58.42 | *** join/#asterisk Quintana (n=sylvain@office.proformatique.com) |
| 02:58.45 | Krurst | anyone know what causes the error message "Incoming call: Got SIP response 400 "Out Of Order" back from <ipaddress>"? |
| 02:59.12 | Nugget | Got SIP response "Insert coin" back from payphone. |
| 03:00.09 | Quintana | Hi all, is there is people here who have digium TE120P working ? |
| 03:00.50 | WilliamK | Quintana, I do |
| 03:01.21 | Quintana | ok have you already have IRQ misses ? |
| 03:01.23 | *** join/#asterisk bkw__ (n=brian@adsl-64-149-35-171.dsl.tul2ok.sbcglobal.net) |
| 03:01.36 | WilliamK | yeah I've seen some |
| 03:01.46 | WilliamK | but no affects on the card/performance |
| 03:01.59 | Quintana | for me it doesn't working |
| 03:02.08 | WilliamK | and I haven't had time to lock the IRQ for the PCI slot |
| 03:02.14 | Quintana | i have 3 cards and all card have same problem |
| 03:02.28 | WilliamK | got the latest zaptel build? |
| 03:02.34 | Quintana | with TE11OP i haven't any problem |
| 03:02.44 | Quintana | yes with 1.2.20.1 and 1.4.5.1 |
| 03:02.54 | WilliamK | it's probably a module issue |
| 03:03.05 | WilliamK | I know the svn copies got fixed this week |
| 03:04.04 | Quintana | yes we looking the code from this module since 4/5 hours but it a little complicated, it missing some doc on the code :-( |
| 03:04.18 | *** join/#asterisk linagee (n=linagee@about/linux/staff/linagee) |
| 03:04.24 | Quintana | hum ok i going to test the svn version now |
| 03:04.28 | WilliamK | 4/5? |
| 03:04.31 | linagee | avaya call manager sucks. :) |
| 03:04.44 | Quintana | 4 hours |
| 03:04.59 | WilliamK | svn branches 1.4 |
| 03:05.05 | WilliamK | I know that works |
| 03:07.26 | WilliamK | earlier this week the zaptel file that calls the modules wasn't calling the te120p at all (svn copy in 1.4) and tzafrir gets the credit for fixing the svn :) |
| 03:08.15 | bkw__ | now the hotel for astricon is 35 miles from the airport |
| 03:08.18 | bkw__ | man thats retarde |
| 03:11.49 | Quintana | WilliamK, i have the same problem with SVN-branch-1.4-r3012 |
| 03:12.01 | SwK | moo |
| 03:12.07 | Nugget | ]:8) |
| 03:12.10 | SwK | heh |
| 03:13.24 | *** join/#asterisk dug (n=chatzill@adsl-71-131-39-119.dsl.sntc01.pacbell.net) |
| 03:13.54 | *** join/#asterisk corydon76-home (i=white@pdpc/supporter/sustaining/Corydon76-home) |
| 03:13.54 | *** mode/#asterisk [+o corydon76-home] by ChanServ |
| 03:14.35 | *** join/#asterisk ZaVoid (n=zavoid@c-67-165-25-195.hsd1.ct.comcast.net) |
| 03:14.40 | ZaVoid | hey all |
| 03:15.33 | *** join/#asterisk lowlevel (n=Stuart@CPE0017f2e2a3e8-CM000f9f7d6742.cpe.net.cable.rogers.com) |
| 03:16.10 | *** join/#asterisk Corydon76-dig (i=purple@pdpc/supporter/sustaining/Corydon76-home) |
| 03:16.10 | *** mode/#asterisk [+o Corydon76-dig] by ChanServ |
| 03:16.16 | dan__t | Anyone familiar with the PolyCom phones? I'm trying to default one, and I'm not having much luck. I read on a wiki somewhere that holding down 4, 6, 7, 8, and * would do it - but I'm having no such luck. |
| 03:19.04 | ZaVoid | think you can get the reset codes on the polycom site |
| 03:19.08 | *** join/#asterisk lowlevel (n=Stuart@CPE0017f2e2a3e8-CM000f9f7d6742.cpe.net.cable.rogers.com) |
| 03:19.48 | *** join/#asterisk ZX81 (n=matt@202.20.97.200) |
| 03:19.55 | dan__t | Yeap, been looking. |
| 03:20.19 | ZX81 | hi all, anyone remember the link that murf wrote up about how to redo agentcallbacklogin using dialplan logic in AEL? |
| 03:20.28 | SwK | dan__t, advanced admin menu, reset user settings, or format the file system |
| 03:20.29 | ZX81 | I thought it was on the blog thingy but can't find it now |
| 03:20.56 | Corydon76-dig | dan__t: Menu, Settings, Advanced, 456, Admin Settings, Reset to Default, Format File System |
| 03:20.58 | SwK | dan__t, depends on firmware version... the 4568* thing only resets the boot params |
| 03:21.44 | SwK | Corydon76-dig, thats only if he's using the the 1.6 or later firmware FYI before that you have to do the "reset user settings" thing |
| 03:22.18 | Corydon76-dig | SwK: I figure that if he doesn't know yet, he probably has a newer phone with the newer firmware |
| 03:22.35 | SwK | you'ld be suprised heh |
| 03:22.44 | SwK | s/you'ld/you'd/ |
| 03:22.53 | SwK | thanks jbot |
| 03:22.56 | Corydon76-dig | Well, unless it's a second market phone |
| 03:23.02 | SwK | yeah |
| 03:23.15 | ZaVoid | anyone semi decent with php? |
| 03:23.28 | SwK | ZaVoid, doing what? |
| 03:23.40 | ZaVoid | well i'm setting a variable to pass back to asterisk |
| 03:24.18 | ZaVoid | i got a php script that does this: $limit = 4500; and then $limit2 = $limit / 60; 4500 can really be any # |
| 03:24.48 | SwK | ok |
| 03:24.56 | ZaVoid | then if i do this: |
| 03:24.56 | ZaVoid | $agi->set_variable( "LIMIT2", $limit2 ); |
| 03:25.14 | ZaVoid | i can't NoOp $limit2 and see it or use the variable in saydigits command |
| 03:25.15 | ZaVoid | however |
| 03:25.29 | ZaVoid | this does work |
| 03:25.30 | ZaVoid | $agi->set_variable( "LIMIT2", sprintf("%.02f" , $limit2) ); |
| 03:25.38 | ZaVoid | but that gives me 75.00 in this sample |
| 03:25.52 | ZaVoid | so i assume my php just isn't right to return just 75 and not the .00 |
| 03:26.06 | ZaVoid | and i been looking at http://us.php.net/manual/en/function.sprintf.php but thats making my head spin worse |
| 03:26.09 | SwK | enable agi debugging on the console and see what its spitting out |
| 03:26.34 | ZaVoid | well is there a better way i can craft the %.02f to spit out 75 instead of 75.00 |
| 03:26.48 | ZaVoid | i'm guessing 02 is spitting out 2 to the right of the decimal? |
| 03:27.07 | SwK | well the .00 is because of the .02 in front of the f |
| 03:27.25 | ZaVoid | right so... 02 = 2 decimal places basically? |
| 03:27.33 | SwK | try int(limit2) or floor($limit2) |
| 03:27.37 | SwK | yeah |
| 03:27.43 | ZaVoid | ok cool let me try somthing then |
| 03:28.08 | SwK | $agi->set_variable( "LIMIT2", floor($limit2) ); |
| 03:28.10 | SwK | try that |
| 03:28.27 | ZaVoid | floor? |
| 03:29.40 | WilliamK | Quintana, what OS are you using? |
| 03:29.47 | WilliamK | (distribution) |
| 03:29.47 | ZaVoid | whats floor do sorry? |
| 03:29.59 | SwK | floor rounds floats down to an int |
| 03:30.14 | SwK | so 123 = floor(123.45) |
| 03:30.15 | *** join/#asterisk Defraz (n=t0tal@24-116-152-177.cpe.cableone.net) |
| 03:30.36 | ZaVoid | ok let me try that |
| 03:30.37 | SwK | as opposed to 124 = ceil(123.45) |
| 03:30.37 | ZaVoid | thanks |
| 03:30.44 | dan__t | Hah. There's a password on it. |
| 03:30.46 | dan__t | Awesome. |
| 03:31.18 | SwK | ZaVoid, paypal thanks gladly accepted :P |
| 03:31.19 | SwK | heh |
| 03:31.22 | dan__t | 2.1.1.0052 |
| 03:31.54 | SwK | dan__t, follow the instructions Corydon76-dig gave and the password default is 456 |
| 03:33.23 | ZaVoid | that did't work |
| 03:33.38 | ZaVoid | for some reason it will NoOp correctly but won't work with say digits |
| 03:33.46 | ZaVoid | but sprintf will |
| 03:35.43 | *** join/#asterisk mtaht4 (n=m@219-106-62-200.enitel.net.ni) |
| 03:37.21 | WilliamK | hey dan__t, just curious... what Polycom phone do you have? |
| 03:37.39 | *** join/#asterisk stkn__ (n=stkn@gentoo/developer/pdpc.active.stkn) |
| 03:39.15 | dan__t | Awesome, thanks, Corydon76-dig. |
| 03:39.20 | dan__t | 601 |
| 03:39.51 | SwK | ZaVoid, look at the agi debug and see whats different about them |
| 03:40.19 | ZaVoid | k |
| 03:41.14 | *** join/#asterisk techman97_andy (n=me@75-134-128-138.dhcp.roch.mn.charter.com) |
| 03:41.30 | ZaVoid | ok so 1st the sprintf way |
| 03:41.31 | ZaVoid | AGI Rx << SET VARIABLE LIMIT2 "75.00" |
| 03:41.31 | ZaVoid | AGI Tx >> 200 result=1 |
| 03:41.38 | techman97_andy | evening everyone - I realized that the timestamps on my voicemails are off, however the Linux box itself seems to be showing the right time...anyone have any hints? |
| 03:41.47 | *** join/#asterisk drfreeze (n=Jim@www.freeze.org) |
| 03:42.06 | *** join/#asterisk mog (n=mog@c-71-207-200-130.hsd1.al.comcast.net) |
| 03:42.06 | *** mode/#asterisk [+o mog] by ChanServ |
| 03:42.09 | WilliamK | check both the hw clock and the system clock? |
| 03:42.19 | techman97_andy | hmmm - I did not check the hw clock... |
| 03:42.45 | techman97_andy | I wouldn't think Asterisk would pull from the hw clock instead of the system clock tho? |
| 03:42.54 | techman97_andy | but I'm up for just about anything. |
| 03:43.33 | ZaVoid | with floor: AGI Rx << SET VARIABLE LIMIT2 "75" |
| 03:43.33 | ZaVoid | AGI Tx >> 200 result=1 |
| 03:44.05 | ZaVoid | hmm |
| 03:44.07 | ZaVoid | i got an idea |
| 03:44.09 | drfreeze | Hi |
| 03:44.39 | drfreeze | After installing 1.4.2, I just noticed that I don't have a Master.csv call log file being generated any more |
| 03:45.19 | WilliamK | =) |
| 03:45.37 | drfreeze | WilliamK: Hey, I resemble that remark! :) |
| 03:45.50 | ZaVoid | i think this might work swk exten => s,n,SayNumber(${LIMIT2},m) |
| 03:45.53 | ZaVoid | testingit now |
| 03:45.58 | *** join/#asterisk bintut (n=bintut@203.125.63.150) |
| 03:46.14 | WilliamK | cept the last time I went it was like mr. blowtorch in the face |
| 03:46.15 | drfreeze | What's the trick to getting call detail records now |
| 03:46.20 | WilliamK | during the heat of the summer |
| 03:47.00 | codefreeze | ZX81: I dimly remember leaving something in the doc/ dir....? |
| 03:47.15 | ZaVoid | ok sweet that worked |
| 03:47.31 | ZaVoid | i guess the problem was with saydigits was that it was not a digit but a double number maybe.. |
| 03:47.33 | ZaVoid | thank swk |
| 03:47.41 | WilliamK | hmmm, must be getting closer to winter with all the freezing around... |
| 03:47.43 | WilliamK | :) |
| 03:49.39 | drfreeze | Hey, anyone know how to activate call logging? |
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| 04:04.18 | Blackthorn | I've been having my service crash out from time to time with a unable to forward frame. but tonight while i was monitoring it i finally caugt the first part of it |
| 04:05.05 | Blackthorn | http://pastebin.com/db075f32 |
| 04:05.45 | Blackthorn | says call was rejected from an ip (unkown ip to me btw) did some reserach not a user of mine. and it says no such context/extenion. |
| 04:06.16 | Blackthorn | then jumps to destory command avoiding iax destroy dread lock. then gotes to the message i've been posting in here for a few days that goes continues |
| 04:06.20 | Blackthorn | on the server untill it's reset. |
| 04:06.42 | Qwell | Blackthorn: are you running the latest version of asterisk? |
| 04:07.07 | Blackthorn | 1.4.11 |
| 04:07.34 | Qwell | I'd like to see the output of an iax2 debug when that happens... |
| 04:08.14 | Blackthorn | i'ts very unpredictable.. server runs 2 to 3 days before it bombs out. |
| 04:08.26 | Qwell | but it's always when a call is rejected? |
| 04:09.04 | Blackthorn | well see i donno. this was the first time i got that part of the message. it's not logged in the logging file. I just happed to be logged in thorugh ssh and saw it scrool past and grabbed it |
| 04:09.24 | Blackthorn | Sep 12 23:58:22] WARNING[20263]: app_dial.c:738 wait_for_answer: Unable to forward voice or dtmf is the only thing that shows up inthe logs |
| 04:09.26 | Qwell | please do start logging what you can |
| 04:09.32 | Blackthorn | not the rejection |
| 04:09.48 | Qwell | if you can get an iax2 debug, I'd be interested in seeing that |
| 04:09.57 | Blackthorn | sure. what do you suggest as a debug |
| 04:10.18 | Blackthorn | is there a way to set that to debug as a file? |
| 04:10.31 | Qwell | yeah, check logger.conf |
| 04:11.04 | Qwell | enable everything on the full => line, and it'll go to /var/log/asterisk/full |
| 04:11.36 | Qwell | then turn on debug and iax2 debug from the CLI |
| 04:11.46 | Qwell | brb, rebooting |
| 04:11.52 | Qwell | ..maybe brb |
| 04:11.59 | Blackthorn | ok done |
| 04:12.01 | Qwell | assuming this kernel works |
| 04:13.07 | *** join/#asterisk mightnare (n=mike@202.164.181.222) |
| 04:17.05 | mightnare | are there known voice quality issues with chanspy when there is alaw to g729 transcoding? |
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| 04:49.14 | Daviey | Does * Trunk use 'includes' differently? |
| 04:51.50 | *** join/#asterisk AJaymn (n=mypocket@71-82-218-158.dhcp.mdsn.wi.charter.com) |
| 04:52.16 | AJaymn | Is it possible to txt message from a Wifi SIP phone to a Cellular phone in the US? |
| 04:52.20 | [TK]D-Fender | Daviey, can't imagine why. pastebin your dialplan and "dialplan show" so we can see the difference |
| 04:52.44 | [TK]D-Fender | AJaymn, * SIP does not support SIP IM's |
| 04:53.53 | AJaymn | [TK]D-Fender what is the Asterisk cmd SMS for then? |
| 04:54.36 | Daviey | [TK]D-Fender: Same settings work on 1.4 but not trunk :S.. The dialplan is pretty standard |
| 04:55.00 | Daviey | [TK]D-Fender: It ignores GOTO a ringgroup, but works fine in 1.4 :S |
| 04:55.40 | [TK]D-Fender | AJaymn, just because * can SEND one via a dialplan app doesn't mean the SIP CHANNEL DRIVER can receive one. |
| 04:55.54 | [TK]D-Fender | Daviey, PASTEBIN <---------- |
| 04:56.05 | AJaymn | gotcha |
| 04:56.22 | Daviey | There's lot of clutter.. I'll have to clean it up and provide later |
| 04:56.23 | Daviey | thanks |
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| 04:57.18 | [TK]D-Fender | Daviey, well that was productive..... |
| 04:58.01 | Daviey | well it was mean't to be a general question... ie known issue? |
| 04:59.27 | [TK]D-Fender | Daviey, You know you couldn't just pastebinned it all by now so we can SEE whats going on but you just keep TALKING about it.... |
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| 05:05.49 | CCFL_Man2 | viva viagra |
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| 05:08.52 | jiboneus | hello,.... I've a silly question,... do Asterisk detect voice signals? |
| 05:09.40 | jiboneus | or does it just connect the channels? |
| 05:10.08 | [TK]D-Fender | jiboneus, ....HUH? |
| 05:11.29 | jiboneus | hi,.. just wanted to know if asterisks detect voice signals |
| 05:13.02 | jiboneus | or can you record conversations with asterisk? |
| 05:13.22 | kaldemar | "or"? |
| 05:14.16 | denon | that's a weird question |
| 05:14.21 | denon | but yes, you can record calls with * |
| 05:14.36 | denon | search the wiki for Monitor |
| 05:14.44 | denon | or do a show application monitor |
| 05:15.03 | kaldemar | and if by voice signals you mean DTMF, yes, Asterisk does detect DMTF. |
| 05:16.27 | jiboneus | Is asterisk process the DTMF or the fxo/fxs card that process it and send asterisk the code? |
| 05:18.36 | [TK]D-Fender | jiboneus, just come out and ask what it is you want to DO EXACTLY |
| 05:19.31 | jiboneus | hehe,... my company wants to integrate * with it's comms which have radio HF, VHF and all,... |
| 05:21.02 | jiboneus | we are able to do that but,.. on the phone we have to press *(some no) to send the PTT (push to talk) to the radio,... |
| 05:28.07 | *** join/#asterisk jiboneus (i=db5d3912@gateway/web/cgi-irc/ircatwork.com/x-e30b0f82b595e4a2) |
| 05:28.21 | jiboneus | sorry i got cut off,... |
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| 06:34.30 | JT | anyone with a cisco ip phone here? |
| 06:35.02 | JT | anyone in the USA/Canada that is |
| 06:35.34 | jarod14 | JT : I have linksys phones, not cisco. they should be close, |
| 06:35.46 | JT | nah, specifically interested in cisco |
| 06:35.53 | JT | eg 7940 or 7905 |
| 06:36.12 | jarod14 | yep I already use them 1 year ago |
| 06:36.37 | jarod14 | what is your question |
| 06:36.53 | JT | under flourescent light, do the screens appear to flicker? |
| 06:37.17 | flenders | JT: why you reckon this wouldn't help there? |
| 06:37.24 | jarod14 | I did not notice this when I use them |
| 06:37.25 | JT | ? |
| 06:37.25 | flenders | s/help/happen |
| 06:37.34 | JT | different mains frequency |
| 06:37.47 | JT | different pulse rate coming from floursecent light |
| 06:37.58 | jarod14 | In France , it's 50Hz |
| 06:38.15 | JT | may be designed to not flicker with 60Hz mains |
| 06:38.18 | jarod14 | don't know the fluorescent light frequency |
| 06:38.31 | JT | jarod14: what country did you view the ciscos in? |
| 06:38.36 | jarod14 | France |
| 06:38.50 | JT | i did ask for usa/canada... |
| 06:38.56 | JT | but interesting you dont notice it |
| 06:39.05 | *** join/#asterisk Strom_M (n=strom@216.64.24.250) |
| 06:39.27 | jarod14 | oki |
| 06:42.59 | *** join/#asterisk steliosk (n=Stelios@ipa226.211.tellas.gr) |
| 06:44.52 | awk | hmm, anyone have any docs to get halifax working with asterisk |
| 06:45.14 | JT | Strom_M: do you have any cisco ip phones there? |
| 06:48.54 | *** join/#asterisk yannj_fr (n=yannj@chilli.esiee.fr) |
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| 07:00.41 | Strom_M | JT: not right here, no |
| 07:00.41 | Strom_M | back home I do though |
| 07:00.51 | JT | hrm |
| 07:01.06 | JT | have you ever noticed their display flickering under flourescent light? |
| 07:02.37 | awk | nobody used halifax with asterisk |
| 07:02.55 | SwK | halifax? is that supposed to be hylafax? |
| 07:03.30 | awk | possibly |
| 07:03.42 | SwK | i use hylafax all the time |
| 07:03.56 | awk | no wonder i never got any google help :) |
| 07:04.16 | SwK | http://hylafax.sourceforge.net/ |
| 07:04.55 | awk | thanks |
| 07:08.46 | awk | you got any install documentation |
| 07:08.50 | awk | or wiki's, etc for it? |
| 07:16.21 | *** join/#asterisk luka1 (n=luka1@mar35.ttc.cz) |
| 07:18.34 | SwK | awk; theres docs there on the website |
| 07:18.45 | SwK | any hylafax install howto will work for you |
| 07:20.01 | *** join/#asterisk LukinoVoip (n=LukinoVo@gw.abanet.it) |
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| 07:23.19 | LukinoVoip | hi all, is it possible to create a t.38 termination for sending fax from IP network over PSTN by a combination of ast+hylafax+t38modem? |
| 07:23.30 | LukinoVoip | any ideas? |
| 07:23.31 | *** part/#asterisk dominic1 (n=dob@213.221.82.242) |
| 07:25.14 | *** join/#asterisk enioh (n=enioreh@core.kahmm.net) |
| 07:25.18 | enioh | Hi ppl |
| 07:25.25 | SwK | LukinoVoip, whats going to do the T38 to TDM conversion? |
| 07:25.42 | *** join/#asterisk yannj_fr (n=yannj@chilli.esiee.fr) |
| 07:26.02 | LukinoVoip | uhm...i'm confused |
| 07:27.16 | SwK | LukinoVoip, T38 is pure Digital... it it only works over IP (either SIP or H323 at this time) so where are you planning to conver from T38 to PSTN? are you operating the SIP or H323 to TDM gateways or is your carrier doing that? |
| 07:30.44 | LukinoVoip | FAX =>ATA =>AST =>h323 =>t38modem >hylafax |
| 07:30.51 | LukinoVoip | => analog modem |
| 07:31.49 | SwK | thats over complicating things |
| 07:32.03 | SwK | in that configuration you dont need hylafax |
| 07:32.33 | JT | that's insanity |
| 07:32.35 | SwK | fax -> ATA (must be t38 enabled) -> ast -> another t38 gateway -> analog line |
| 07:32.44 | JT | and h.323, do you really need it |
| 07:33.06 | LukinoVoip | for t38modem to communicate with asterisk |
| 07:33.09 | SwK | or hylafax -> t38modem -> SIP -> to whatever -> t38 gateway -> TDM |
| 07:33.18 | SwK | t38modem does sip now you know |
| 07:33.37 | LukinoVoip | wow |
| 07:34.23 | LukinoVoip | asterisk must be configured for t38 passtrhu |
| 07:34.39 | JT | can t38modem talk to a pri? |
| 07:34.39 | LukinoVoip | over t38modem, is it true? |
| 07:34.44 | *** join/#asterisk yannj_fr (n=yannj@chilli.esiee.fr) |
| 07:34.53 | SwK | t38modem does not do analog at all |
| 07:34.57 | SwK | it just does T38 |
| 07:35.06 | JT | what does it actually do |
| 07:35.13 | SwK | if you want to talk to a PRI via asterisk use iaxmodem |
| 07:35.18 | LukinoVoip | nothing, i'm just planning |
| 07:35.19 | SwK | T38modem is a software t38modem |
| 07:35.38 | *** join/#asterisk fujin (n=aj@unaffiliated/fujin) |
| 07:35.54 | JT | t.38 protocol to tiff? |
| 07:36.06 | SwK | no Class1 fax modem to t38 |
| 07:36.15 | SwK | jt: ever use IAXModem? |
| 07:36.31 | JT | nup |
| 07:37.30 | LukinoVoip | what's the difference between iaxmodem and t38modem? |
| 07:37.32 | SwK | t38modem emulates a class1 faxmodem for doing faxes only and takes fax commands from fax software like hylafax (which things t38modem is jsut another hayse compat faxmodem) and on the 'PSTN side' it talks T38 over SIP or h323 |
| 07:37.48 | JT | ah ok |
| 07:38.05 | SwK | iaxmodem doesnt do t38 it has DSPs the encode the digital fax commands into an analog g711 signal and sends it out over iax |
| 07:38.10 | JT | so the other side if usually .tif, in hylafax |
| 07:38.14 | SwK | with t38 there is no analog component |
| 07:38.17 | SwK | no |
| 07:38.26 | SwK | its t.30 commands |
| 07:38.47 | *** join/#asterisk dikdust (n=dikdust@2001:1418:1c7:0:211:11ff:fed3:ecc7) |
| 07:38.50 | enioh | is there a way to disable asterisk sending a 200 OK with autocreatepeer=yes ? |
| 07:39.11 | *** join/#asterisk appelza (n=d@196.35.241.75) |
| 07:39.28 | SwK | oh well screw it i'm crashing its 0250 here |
| 07:39.42 | JT | alright |
| 07:40.09 | LukinoVoip | i have a fax behind an IP network...haw can i solve faxes troubles? |
| 07:40.36 | SwK | LukinoVoip, quit using G711 passthru or order an analog line for it |
| 07:40.53 | fujin | analogue line for the win! |
| 07:40.55 | appelza | how do I create a sip-sip extentions? I have a couple of sip phones registered, they call all dial out but none call dial eachother |
| 07:41.37 | fujin | appelza, create a context which handles your extensions |
| 07:41.45 | fujin | include it into the same extension that the sip.conf points the phones to |
| 07:41.50 | fujin | and then create a macro to dial |
| 07:42.06 | fujin | macro dial { Dial(SIP/${MACRO_EXTEN}) }; |
| 07:42.48 | LukinoVoip | i could use a fax server connected to n analog lines...but i don't know how to connect to it remote faxes... |
| 07:42.50 | fujin | context phones { XXX => &dial(); } |
| 07:42.58 | appelza | cool |
| 07:42.59 | appelza | tnx |
| 07:43.21 | fujin | of course you probably want to handle voicemail/inuse detection in your macro |
| 07:43.48 | *** join/#asterisk lbow (n=lbow@dsl-146-6-152.telkomadsl.co.za) |
| 07:43.48 | SwK | LukinoVoip, use T38 enabled ATA's not all ATAs do T38... PAP2, SPA2000 series etc dont do t38 |
| 07:44.37 | LukinoVoip | SPA3102? |
| 07:45.09 | LukinoVoip | LInksys claims t38 to work in that |
| 07:45.43 | SwK | that one might work |
| 07:45.57 | SwK | but you have to have t38 enabled gateways on both ends of the IP link |
| 07:46.05 | SwK | asterisk will pass it thru but it wont convert it |
| 07:46.20 | dan__t | I *want* to say that I read somewhere that there were special ROMs that you could load on these Polycom phones that have extra features and such. |
| 07:46.26 | dan__t | Am I just too tired to make sense of most anything? |
| 07:46.27 | LukinoVoip | t38modem can act as a t38 endpoint? |
| 07:46.43 | SwK | yes |
| 07:47.10 | SwK | but using it in the configuration you described earlier is going to cause more problems then its worth |
| 07:47.45 | SwK | if you really want to do it that way and have something like E1s coming in, get a Cisco AS5300 or something that does proper T38 to E1 |
| 07:48.01 | appelza | fujin, was that a complete dial macro? |
| 07:48.03 | SwK | or put another ATA out there to do the T38 -> TSM |
| 07:48.25 | fujin | appelza, lol, yep. it will work :) |
| 07:48.31 | SwK | appelza, look at the dialplan that comes with the asterisk source theres a good example of a macro that calls local extensions there |
| 07:48.35 | fujin | that's AEL by the way, not standard extensions.conf |
| 07:49.36 | LukinoVoip | SwK, thanks |
| 07:54.49 | *** join/#asterisk qdk (n=qdk@213.150.62.32) |
| 07:59.43 | appelza | fujin: http://pastie.caboo.se/96759 am I doing anything wrong? |
| 08:00.07 | fujin | probably |
| 08:00.08 | appelza | (anyone else is also welcome to check ofcourse) |
| 08:00.13 | fujin | can you not work that out for yourself? |
| 08:00.19 | fujin | logic flow would indicate to me |
| 08:00.24 | fujin | that if you've done it correctly (i.e; not wrong) |
| 08:00.26 | fujin | it would work |
| 08:00.38 | fujin | lol |
| 08:00.41 | fujin | yes, you're doing it wrong |
| 08:00.43 | fujin | like I said before |
| 08:00.50 | fujin | that macro isn't standard extensions.conf |
| 08:00.55 | fujin | it's extensions.ael material |
| 08:01.01 | fujin | pbx_ael parser, even |
| 08:01.40 | fujin | in sip-sip |
| 08:01.41 | fujin | you want |
| 08:01.46 | fujin | err |
| 08:01.47 | fujin | no |
| 08:01.48 | appelza | :] |
| 08:01.50 | fujin | that's not what you want |
| 08:01.57 | fujin | in [sip-sip] |
| 08:01.58 | fujin | you want |
| 08:02.22 | fujin | exten => XXX,1,Macro(Dial,${MACRO_EXTEN}) |
| 08:02.29 | fujin | and uh |
| 08:02.34 | fujin | I forget the format for macros in extensions.conf |
| 08:02.46 | fujin | I use AEL. That's how I roll. |
| 08:02.50 | JT | missing an underscore |
| 08:03.08 | fujin | yeah |
| 08:03.09 | fujin | right ;p |
| 08:03.15 | fujin | _XXX |
| 08:03.15 | appelza | not Dial/Sip${MACRO_EXTEN}) ? |
| 08:03.26 | fujin | no, the Macro that you haven't written should be called Dial |
| 08:03.28 | fujin | or Localdial |
| 08:03.30 | fujin | or stdexten |
| 08:03.32 | fujin | or something like that |
| 08:03.39 | fujin | Macro just starts it up |
| 08:03.49 | fujin | JT, what's the format for macro's in extension.conf? |
| 08:03.54 | fujin | [macro blah]? |
| 08:04.03 | JT | [macro-poo] |
| 08:04.08 | fujin | ah. |
| 08:04.13 | appelza | Ah ok |
| 08:04.18 | appelza | So, now I need a [macro-dial] |
| 08:04.38 | fujin | appelza, [macro-dial]\n exten=>s,1,Dial(SIP/${MACRO_EXTEN})\n exten=>s,n,Hangup |
| 08:04.40 | fujin | yessir |
| 08:04.49 | fujin | and in [sip-sip] you'll have |
| 08:05.01 | fujin | exten => _XXX,1,Macro(Dial,${EXTEN}) |
| 08:05.04 | fujin | not macro_exten, myb ad ;] |
| 08:05.13 | fujin | yucky extensions.conf |
| 08:05.17 | fujin | anyway, I'm off. going to makie food. |
| 08:05.21 | appelza | :] |
| 08:05.28 | appelza | ktnx :D |
| 08:11.16 | *** join/#asterisk ming_zym (i=ming_zym@nat/yahoo/x-1ebc68074b10f5f8) |
| 08:18.06 | *** join/#asterisk Loke-pt (n=kumar@mail.netvita.com) |
| 08:23.18 | *** join/#asterisk bintut (n=bintut@203.125.63.150) |
| 08:34.15 | dan__t | I love these phones. |
| 08:34.16 | dan__t | So much. |
| 08:34.35 | Uatec | What phones are they? |
| 08:39.18 | dan__t | PolyCom 601's |
| 08:40.20 | Daviey | dan__t: v. expensive tho.... What so good about it? |
| 08:40.47 | *** join/#asterisk yang (n=jan@212.103.134.208) |
| 08:41.00 | dan__t | They just work. |
| 08:41.06 | dan__t | I picked up 5 of them for $125 a piece |
| 08:41.13 | Daviey | good price.. |
| 08:41.35 | yang | where does asterisk store config files /etc/asterisk ? |
| 08:42.12 | yang | So only this directory is suitable for a backup...any other files too? |
| 08:42.12 | JT | bingo |
| 08:42.23 | JT | /etc/zaptel.conf |
| 08:42.24 | Daviey | dan__t: I've not had problems with Linksys (nee Sipura) and Aastra |
| 08:42.48 | JT | polycom are by far the best |
| 08:42.52 | JT | especially over linksys |
| 08:43.29 | Daviey | possibly improved sound quality, but what makes it worth 2-3 times more? |
| 08:43.31 | yang | having 1.2.13 version of asterisk is preety much out of date heh? |
| 08:43.50 | JT | Daviey: you don't need to buy a 601. |
| 08:44.06 | Daviey | JT: Did i say i did? |
| 08:44.15 | JT | you said 2-3 times more |
| 08:44.19 | JT | not all of them would be |
| 08:44.23 | *** join/#asterisk IgorG (n=FeedomPa@195.162.32.126) |
| 08:44.26 | dan__t | Eh I've got a little consulting company, a little hosting company, and i work from home for my main employer |
| 08:44.39 | dan__t | Cell phone bills were approaching $5-600/mo |
| 08:44.46 | *** join/#asterisk d00gster (n=doughant@bas1-toronto12-1128666903.dsl.bell.ca) |
| 08:44.57 | dan__t | I thought this would be a good way, with the six lines, to separate each queue and mailbox and blah blah |
| 08:45.12 | yang | I wonder if these errors have been fixed in the last versions of asterisk... |
| 08:45.13 | yang | Sep 13 05:50:14 NOTICE[2187] chan_zap.c: PRI got event: HDLC Bad FCS (8) on Primary D-channel of span 1 |
| 08:45.16 | yang | Sep 13 05:51:12 NOTICE[2187] chan_zap.c: PRI got event: HDLC Abort (6) on Primary D-channel of span 1 |
| 08:45.23 | dan__t | Anyone seen a 1.4 RPM for CentOS5 by chance? |
| 08:45.39 | tzafrir_laptop | there are some at atrpms |
| 08:45.49 | dan__t | Cool, thanks. |
| 08:46.12 | tzafrir_laptop | there's a pending package for Fedora |
| 08:46.29 | dan__t | That'll work, too. |
| 08:47.02 | Daviey | dan__t: build from src ;) |
| 08:47.08 | dan__t | rpm owns my soul. |
| 08:47.19 | dan__t | nothing bugs me more than files in /$allover |
| 08:47.24 | dan__t | without any kind of accounting heh |
| 08:52.06 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 08:52.27 | *** join/#asterisk Strom_M (n=strom@216.64.24.250) |
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| 08:52.30 | *** mode/#asterisk [+o Corydon76-home] by ChanServ |
| 08:57.29 | appelza | can someone please tell me why I can't make sip to sip calls? : http://pastie.caboo.se/96766 |
| 09:06.14 | *** join/#asterisk fordfrog (n=fordfrog@gentoo/developer/fordfrog) |
| 09:06.26 | ai-a[afk] | appelza: you need to tell us what is not working, where is the CLi log,, where is sip_accounts.conf" do you authenticate ? do they have a dial tone? .... |
| 09:07.11 | ai-a[afk] | guessing your sips are all 3 digit ext, and you dial them using the [sip-sip] context |
| 09:07.23 | appelza | the sip phones do register, and have dial tones, they can phone 'out' but they can't phone eachother, I get no error in asterisk, but my xlite says not found |
| 09:07.28 | ai-a[afk] | where is [ext-local] |
| 09:07.37 | ai-a[afk] | ok. |
| 09:07.47 | ai-a[afk] | not-found means there is log on the CLI.. so show us. |
| 09:07.55 | *** join/#asterisk mvanbaak (i=michiel@vanbaak.xs4all.nl) |
| 09:07.57 | ai-a[afk] | asterisk -rvvvvvvvvvvvvvvvv and capture the log of a call. |
| 09:08.22 | appelza | I'll paste my ext-local aswel |
| 09:08.49 | appelza | ext-local: http://pastie.caboo.se/96768 |
| 09:08.52 | appelza | and sec for the log |
| 09:08.53 | ai-a[afk] | dont think i need that. |
| 09:09.10 | ai-a[afk] | maybe we do.. ok. and the log. |
| 09:10.15 | tzafrir_laptop | dan__t: rms are a powerful tool, when properly used |
| 09:10.37 | tzafrir_laptop | People under-estimate them |
| 09:10.41 | appelza | nothing show son the console ai-a[afk] |
| 09:10.55 | dan__t | They sure do. |
| 09:11.03 | appelza | (dialing from a three digit sip ext to another three digit sip ext) |
| 09:11.09 | ai-a[afk] | appelza: the console must show the authentacation... attempt to call.. |
| 09:11.20 | appelza | it doesnt |
| 09:11.26 | ai-a[afk] | eh ? |
| 09:11.32 | appelza | check |
| 09:11.37 | ai-a[afk] | your doing something wrong. |
| 09:11.47 | appelza | Connected to Asterisk 1.2.23-BRIstuffed-0.3.0-PRE-1y-j currently running on asterisk (pid = 3654) |
| 09:11.50 | appelza | Verbosity is at least 25 |
| 09:11.50 | appelza | asterisk*CLI> |
| 09:11.55 | appelza | now I'll try dial |
| 09:11.55 | ai-a[afk] | dont paste in here. |
| 09:12.17 | appelza | I get the busy tone but nothing happened on the console |
| 09:12.34 | appelza | I am doing something wrong, thats why im here :< |
| 09:12.43 | ai-a[afk] | include => ext-local-custom |
| 09:13.04 | ai-a | appelza: are you going to show us everything, or keep hiding things until i ask you for it ? |
| 09:13.19 | appelza | sec, sorry |
| 09:13.34 | ai-a | appelza: uplocate the whole sip.conf extention.conf and whateverelse is needed.. dont chop small bits that you are viewing... |
| 09:13.41 | ai-a | if we see what you see, we dont find the bug either. |
| 09:13.49 | ai-a | its what your not seeing thats the problem. |
| 09:14.36 | appelza | ok, but its a mess |
| 09:15.29 | dan__t | Huh. |
| 09:15.33 | ai-a | then that might be your problem. |
| 09:15.42 | dan__t | Would be neato if someone rolled some open roms for these polycom phones |
| 09:15.53 | dan__t | something akin to the whole wrt54g thing |
| 09:15.57 | appelza | some of it is custom/handedited and other stuff is from freepbx |
| 09:15.59 | appelza | ;/ |
| 09:16.02 | appelza | its gross actually |
| 09:16.13 | appelza | that is probably also my prob ye |
| 09:16.14 | yang | I wonder if these errors have been fixed in the last versions of asterisk... |
| 09:16.16 | appelza | ill try soemthing else |
| 09:16.17 | yang | Sep 13 05:50:14 NOTICE[2187] chan_zap.c: PRI got event: HDLC Bad FCS (8) on Primary D-channel of span 1 |
| 09:16.26 | yang | Its shared IRQ |
| 09:17.02 | JT | open roms for polycoms would likely suck |
| 09:17.20 | JT | their firmware has tonnes of proprietary algorithms that make them sound so good |
| 09:18.59 | *** join/#asterisk qdk_ (n=qdk@213.150.62.32) |
| 09:20.24 | tzafrir_laptop | yang: shared IRQs can be a problem, but not necessarily |
| 09:21.40 | yang | i think there are 2 slots |
| 09:21.44 | yang | and there are 2 cards inside |
| 09:21.50 | yang | no free slots |
| 09:22.15 | yang | but i am using asterisk 1.2.x i wonder maybe with an asterisk update the problem would be solved |
| 09:22.16 | *** join/#asterisk syneus (n=syneus@syneus.aemcom.net) |
| 09:23.25 | arcanine | how many users does asterisk is tested? |
| 09:23.41 | *** join/#asterisk KpoH (n=AID@host-86-106-208-182.moldtelecom.md) |
| 09:23.44 | KpoH | how to compile asterisk with debug symbols? |
| 09:24.00 | arcanine | can asterisk accomodate 250 users simultaneously? |
| 09:24.42 | KpoH | arcanine: without transcoding, I think yes |
| 09:25.19 | tzafrir_laptop | yang: if you suspect anything related to interrupts and such, then it is Zaptel you should consider upgrading |
| 09:28.10 | ai-a | CoaxD: its "shut up, i want to talk now please..." |
| 09:28.42 | CoaxD | uh huh... |
| 09:29.18 | CoaxD | man, i could be sleeping. but no. Stupid database feed blows up in europe and deletes the entire database. yeah. i'm NOT gonna sleep tonight. |
| 09:30.15 | Uatec | hi |
| 09:30.23 | Uatec | i'm thinking of setting up a queue system on my asterisk box |
| 09:30.31 | yang | tzafrir_laptop: thanks |
| 09:30.44 | Uatec | is there a way that i can make the system wait 30second to a minute before ringing an agent again? |
| 09:31.11 | KpoH | so how about debugging symbols in asteisk? in menuselect I don |
| 09:31.12 | Uatec | i want my engineers to have the opportunity to complete notes and things before they get pestered by another call |
| 09:31.25 | *** join/#asterisk Strom_M (n=strom@216.64.24.250) |
| 09:31.29 | KpoH | i don't see "debug symbols" option |
| 09:31.46 | *** join/#asterisk Strom_M (n=strom@216.64.24.250) |
| 09:31.57 | yang | tzafrir_laptop: actually from what i see is that there is no "zaptel" package installed at all....however /etc/zaptel.conf exists |
| 09:33.40 | ai-a | Uatec: cant they ignore a ringing phone ? i always ignore mine :) |
| 09:33.52 | tzafrir_laptop | yang: you probably installed from source. modinfo zaptel | grep ^version |
| 09:35.29 | yang | yes 1.2.11 |
| 09:37.17 | tzafrir_laptop | yang: for some cards I think that they god some major improvements around zaptel 1.2.4-16 or so |
| 09:37.45 | tzafrir_laptop | I mean: in handling shared IRQs. |
| 09:37.52 | tzafrir_laptop | But I don't really know this well |
| 09:39.45 | yang | ok |
| 09:41.40 | tzafrir_laptop | again, I'm really not sure that shared interrupt lines are the issues here |
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| 09:52.58 | *** join/#asterisk chris_1 (n=chris@ng1.kurtkrenn.com) |
| 09:53.10 | chris_1 | hi folks |
| 09:53.56 | chris_1 | how to solve "channel.c:787 channel_find_locked: Avoided initial deadlock for '0xb7865700', 10 retries!" |
| 09:57.04 | *** part/#asterisk pks (n=pks@elit.pl) |
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| 09:59.18 | puzzled | hi |
| 10:01.15 | *** join/#asterisk ghenry (n=ghenry@suretecsystems.plus.com) |
| 10:01.58 | ai-a | chris_1: as an initial install of asterisk that dosnt work, of consider we might need to know what you have done. |
| 10:05.13 | *** part/#asterisk ming_zym (i=ming_zym@nat/yahoo/x-1ebc68074b10f5f8) |
| 10:06.14 | chris_1 | the * has worked now for months (v.1.2.13 with sangoma A104d). Yesterday changed implemented a queue / agent and suddenly it crashes... this is the last logged msg |
| 10:11.22 | ai-a | hunt it down and issue a bug request. |
| 10:21.26 | JT | i don't think 1.2.13 is the latest 1.2 |
| 10:24.31 | *** join/#asterisk mohsen (n=chatzill@81.31.160.140) |
| 10:26.44 | Uatec | ai-a, yes, but how many phones do you have ringing |
| 10:26.51 | Uatec | and if you spend the entire day sitting there with phones ringing |
| 10:27.02 | Uatec | we're trying to find a way of keeping the office quiet |
| 10:27.07 | Uatec | if noones going to answer the phone |
| 10:27.10 | Uatec | why ring it? |
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| 10:36.22 | *** join/#asterisk remi____ (n=reber@LNeuilly-152-21-126-73.w193-253.abo.wanadoo.fr) |
| 10:36.27 | remi____ | hi all |
| 10:36.58 | remi____ | What distribution would you use to have conferencing whithout pain ? And latest asterisk without pain at all too. |
| 10:40.16 | puzzled | remi____: depends on your personal preference. fedora, centos/rhel, debian etc. all work fine |
| 10:40.50 | dan__t | Hell yeah. |
| 10:40.52 | yang | asteriskdoc.com website has difficulties |
| 10:40.55 | dan__t | Got this phone to boot one of the roms. |
| 10:41.02 | remi____ | puzzled, wich ones have the latests asterisk releases, and wich one follows * dev ? |
| 10:41.22 | puzzled | remi____: there may be an advantage in using a distro with a kernel that supports high resolution timers. I know Fedora 7 supports it and I think centos/rhel don't. I don't know about other distro's |
| 10:41.58 | *** join/#asterisk clandmeter (n=Carlo@81.175.82.2) |
| 10:42.24 | puzzled | remi____: atrpms.net has asterisk 1.4 rpms for fedora/centos/rhel. I don't know about others. * dev is something you currently want to stay far away from |
| 10:46.10 | remi____ | i was using gentoo until now, but have issues about stability, wonder if i should change ... |
| 10:50.14 | *** join/#asterisk saftsack (n=saftsack@pD9E07E2D.dip.t-dialin.net) |
| 10:52.50 | JT | remi____: just compile asterisk |
| 10:52.56 | JT | it's what everyone else does |
| 10:58.23 | awk | dont touch me |
| 10:58.40 | JT | awk: ? |
| 10:59.00 | *** join/#asterisk zotz (n=zotz@24.244.163.157) |
| 10:59.29 | awk | *shrug* |
| 10:59.57 | *** join/#asterisk snowchyld (n=snowchyl@naku.org) |
| 11:00.39 | JT | ... |
| 11:00.40 | snowchyld | anyone have issues with a linksys SPA2102 sipura adapter ? |
| 11:01.33 | *** join/#asterisk mightnare (n=mightnar@202.164.181.222) |
| 11:04.31 | Uatec | hello, i'm trying to dial my other asterisk box using sip |
| 11:04.50 | Uatec | my dial string goes Dial(SIP/username:password@12.34.56.78) |
| 11:05.15 | Uatec | i'm just getting: -- SIP/12.34.56.78-234234 is circuit-busy |
| 11:05.18 | Uatec | WTF does that mean |
| 11:05.21 | Uatec | i looked at the other asterisk box |
| 11:05.29 | Uatec | and it didn't record anything |
| 11:06.00 | JT | probably can't contact it |
| 11:06.10 | JT | why don't you dial using a sip.conf entry |
| 11:06.11 | Uatec | that's what i thought |
| 11:06.17 | Uatec | but i can ping the machine |
| 11:06.18 | Uatec | and ssh in to it |
| 11:06.27 | Uatec | i ahve tried using sip.conf |
| 11:06.33 | JT | imho using usernames and passwords and ips in the dialstring is silly |
| 11:07.28 | Uatec | oh great |
| 11:07.32 | Uatec | i modify my extensions.conf |
| 11:07.40 | Uatec | to use an entry from sip.conf |
| 11:07.46 | Uatec | and now whenever i type anything it does nothing |
| 11:08.13 | Uatec | oh fixed |
| 11:09.21 | Uatec | now i get: Unable to create channel of type 'SIP' (cause 3 - No route to destination) |
| 11:09.43 | Uatec | but i can ssh in to the destination server, and ping it |
| 11:10.10 | jwh | is it registered? |
| 11:10.25 | mightnare | is there a known problem when using chanspy with a bridged call uing g729 and g711? i get choppy voices... |
| 11:12.11 | Uatec | OMFG, IT WORKS |
| 11:13.09 | snowchyld | grats |
| 11:13.35 | snowchyld | my asterisk looks ok it seems, its the linksys adapter which is not... sending to the phone output or something |
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| 11:49.30 | ectospasm | I've got a problem. I've got an IAXy, and a connection to an ITSP over IAX. Whenever a call comes in over the IAX provider, it tries to authenticate as the IAXy user, not the ITSP user. The only workaround I've gotten to work is setting the IAXy user password to be the same as the ITSP password, which I understand is an undesirable situation... |
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| 12:00.53 | thewiizle | ello |
| 12:01.02 | thewiizle | asterisknow |
| 12:01.06 | thewiizle | is the root password locked out? |
| 12:02.04 | JT | wrong channel |
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| 12:20.18 | reber | JT, oka |
| 12:20.19 | reber | y |
| 12:20.33 | masus | hi all , is it possible that macro is now work with realtime together |
| 12:20.38 | masus | or am i missing something |
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| 12:23.47 | *** mode/#asterisk [+o Corydon76-dig] by ChanServ |
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| 12:31.18 | JT | reber: ? |
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| 12:32.28 | Shido6 | anyone have a sonicwall for sale? |
| 12:34.00 | coppice | some people say teh PAP2 does T.38. The data sheet says it doesn't. doe anyone know the real story? |
| 12:38.31 | *** join/#asterisk bintut (n=bintut@cm47.gamma178.maxonline.com.sg) |
| 12:42.33 | masus | <PROTECTED> |
| 12:42.38 | JT | coppice: i swear i saw someone earlier today that it does not do t.38 |
| 12:43.30 | coppice | the data sheet definitely says the PAP2T does not do T.38, but I've not clear if there are other variants that might do it |
| 12:45.34 | file | the PAP2(no T) does not |
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| 13:20.31 | rantsh | Hi all |
| 13:20.56 | rantsh | if you remember from yesterday I have a problem with my call queue |
| 13:21.13 | nicchap | Any experienced Sangoma card (A102D) users out there that have been able to do RLT calls on a DMS100 pri? |
| 13:21.37 | *** join/#asterisk caio1982 (i=caio1982@CAcert-br/caio1982) |
| 13:21.43 | rantsh | not only it accepts calls from callers when there's no agent logged in, it calls on of them persistently even if he's logged off |
| 13:21.59 | nicchap | RLT works fine on Digium based cards, but doesn't seem to work on Sangoma based systems |
| 13:22.04 | rantsh | if anyone can help me I've put my configuration files in pastebin |
| 13:22.16 | rantsh | the URL is http://pastebin.com/m20ed3434 |
| 13:22.44 | *** join/#asterisk kwame (n=kwame@208.66.168.243) |
| 13:23.06 | rantsh | I'd really (REALLY) appreciate any help you can provide me with |
| 13:23.49 | nicchap | I have the pastebin opened and will look into it. I have to leave for 10-20 minutes |
| 13:24.28 | *** join/#asterisk stkn|work (n=stkn|wor@gentoo/developer/pdpc.active.stkn) |
| 13:24.46 | rantsh | thanks |
| 13:25.24 | kwame | Hi, does anyone here knows if there is a repo from where I can get the asterisk rpms for CentOS 5? |
| 13:27.37 | *** join/#asterisk anonymouz666 (n=anonymou@189.25.74.159) |
| 13:28.17 | Teln1100A | http://www.laimbock.com/asterisk/ |
| 13:28.31 | Teln1100A | Asterisk 1.2.24 RPMs for CentOS 5 |
| 13:29.09 | *** part/#asterisk ShakaGoldSaint (n=eleazar@190.142.2.250) |
| 13:30.19 | kwame | I would like to go for asterisk 1.4 instead of 1.2 |
| 13:31.20 | Teln1100A | have you tried compile from source? its not that bad |
| 13:32.26 | *** join/#asterisk arekm (i=arekm@pld-linux/arekm) |
| 13:33.03 | kwame | Teln1100A: you might say I have tried, I haven't tried that hard or invested a lot of time though |
| 13:33.05 | arekm | should I set fxoks or fxsls in zaptel.conf? |
| 13:33.29 | *** join/#asterisk bbryant (n=12243@c-68-59-20-153.hsd1.sc.comcast.net) |
| 13:33.35 | Teln1100A | try getting yum to work |
| 13:33.45 | Teln1100A | then you can just do yum install asterisk |
| 13:33.47 | *** join/#asterisk [TK]D-Fender (n=joe@MTRLPQ02-1177745839.sdsl.bell.ca) |
| 13:34.00 | kwame | Teln1100A: better said, I've tried to create my own rpms, you know, I like to keep my rpm database the most accurate possible |
| 13:34.08 | *** join/#asterisk lirakis (n=lirakis@65.200.191.253) |
| 13:34.20 | kwame | Teln1100A: try getting yum to work? |
| 13:34.31 | Teln1100A | with the right repositories |
| 13:35.10 | Teln1100A | Centos is compatible with RHEL 5 so maybe you could try : http://atrpms.net/dist/el5/asterisk/ |
| 13:37.26 | mohsen | any one can advise me about some good FOSS billing software for asterisk? |
| 13:38.10 | ManxPower | Just remember, not many people here can help you if you are using a prepackaged Asterisk |
| 13:39.37 | *** join/#asterisk Corydon76-home (i=indigo@pdpc/supporter/sustaining/Corydon76-home) |
| 13:39.37 | *** mode/#asterisk [+o Corydon76-home] by ChanServ |
| 13:39.43 | [TK]D-Fender | Just download the source like everyone else. Source works, packages fail. |
| 13:39.53 | awk | pft |
| 13:40.00 | awk | anyone know how to create a account for hylafax |
| 13:40.02 | Teln1100A | Hi [TK]D-Fender |
| 13:40.59 | *** part/#asterisk masus (n=tet@88.248.73.2) |
| 13:42.36 | yang | is asterisk-users list only for subscribers? |
| 13:42.47 | yang | to post mails tehre |
| 13:42.47 | ManxPower | yang: Correct. |
| 13:42.52 | ManxPower | So subscribe to it. |
| 13:42.57 | yang | ok |
| 13:43.29 | *** join/#asterisk Strom_M (i=strom@nat/digium/x-eb42f5c1a958dcd5) |
| 13:43.31 | ManxPower | it is the must useful thing you can do when working with Asterisk |
| 13:44.59 | ManxPower | arekm: that is because configs are very specific to each installation |
| 13:45.42 | ManxPower | Now if you have a specific question, feel free to ask it. |
| 13:46.27 | yang | yeah |
| 13:46.51 | arekm | 24 FXS x channel bank (zhone z-plex 10) and some 4 port E1 card. Trying to configure it so asterisk will see channel bank. Now should I configure it as FXO or FXS at asterisk side? |
| 13:47.38 | arekm | channel bank is connected via T1/E1 to card of course |
| 13:47.46 | [TK]D-Fender | arekm: fxols=1-24 in zaptel. signalling=fxo_ls in zapata.conf |
| 13:48.08 | [TK]D-Fender | arekm: You use FXO signalling for FXS ports. |
| 13:48.11 | *** join/#asterisk jtexter3 (n=jtexter3@12.159.220.114) |
| 13:48.48 | [TK]D-Fender | arekm: so * is the FXO (Foreign eXchange Office), andd the phone attached to your CB's port is the Foreign eXchange Station |
| 13:50.30 | *** join/#asterisk tsurko (n=tsurko@213.91.216.130) |
| 13:50.44 | *** join/#asterisk Digitaldaz21 (n=me@87-194-8-193.bethere.co.uk) |
| 13:50.46 | arekm | and what about span=1,1,0,esf,b8zs ? is this correct? |
| 13:50.47 | ManxPower | you would also want esf b8zs in /etc/zaptel.conf for the span that connects to the channel bank. I am assuming the channel bank is a T-1 channel bank (they are MUCH cheaper that E-1) |
| 13:51.00 | nicchap | rantsh: What version of * are you using? |
| 13:51.15 | coppice | so obviously from its name FXO is a place that changes currencies :-) |
| 13:51.50 | ManxPower | the 2nd 1 should be a 0 |
| 13:52.15 | ManxPower | you only want a 1 in the 2nd field for your primary telco span |
| 13:53.56 | *** join/#asterisk Corydon76-dig (i=indigo@pdpc/supporter/sustaining/Corydon76-home) |
| 13:53.56 | *** mode/#asterisk [+o Corydon76-dig] by ChanServ |
| 13:54.27 | [TK]D-Fender | arekm: yes, that span definition is typical |
| 13:55.11 | [TK]D-Fender | arekm: so fist the snap line, then the fxols=1-24 |
| 13:55.14 | [TK]D-Fender | span* |
| 13:55.34 | *** join/#asterisk etfonhomey (n=chatzill@mail.advancmed.org) |
| 13:56.01 | arekm | ok, I have span=1,0,0,esf,b8zs |
| 13:56.02 | arekm | fxols=1-24 |
| 13:56.26 | [TK]D-Fender | arekm: that looks about right |
| 13:56.46 | [TK]D-Fender | arekm: 1,0,0 is indeed the better choice so * provides the clocking to your CB |
| 13:57.26 | *** join/#asterisk mindCrime (n=chatzill@66.83.208.218.nw.nuvox.net) |
| 13:57.45 | etfonhomey | [TK]D-Fender, where is the most appropriate place to advertise the need for an * consultant? Would it be the asterisk-users mailing list? |
| 13:58.31 | arekm | [TK]D-Fender: fxols=1-24 in zaptel.conf? or 1-4? (this is 4 port T1/E1 card connected to 24 FXS channel bank using 1 of T1/E1 ports) |
| 13:59.17 | ManxPower | arekm: there are 24 channels on a T-1, therefore you need to configure 24 channels |
| 13:59.44 | coppice | the only ATAs now listed on the Linksys site are the PAP2 and the SPA2002. neither do T.38. progress - they've heard of it :-\ |
| 13:59.45 | ManxPower | etfonhomey: asterisk-giz |
| 13:59.49 | arekm | where I tell asterisk that there are 4 T1 ports in my card? |
| 14:00.04 | etfonhomey | Gracias, ManxPower. |
| 14:00.09 | `Sean | lol at astterisk-giz |
| 14:00.09 | ManxPower | etfonhomey: where are you located? I occasionally accept new customers. |
| 14:00.15 | ManxPower | asterisk-biz |
| 14:00.26 | ManxPower | I'm not done with my first cup of Mega ManxPower Coffee |
| 14:00.32 | `Sean | haha |
| 14:00.36 | *** join/#asterisk ruied_ (n=ruied@89-180-225-202.net.novis.pt) |
| 14:00.41 | awk | hrm |
| 14:00.52 | awk | anyone know whre in hylafax you add a *%#(#%(% user/password |
| 14:01.11 | ManxPower | awk: perhaps you can try a place where people use hylafax? |
| 14:01.20 | _x86_ | haha |
| 14:01.32 | awk | ManxPower err allot of people should with the intergration to asterisk |
| 14:01.48 | ManxPower | I use RxFax/TxFax and I suspect most people here do. |
| 14:01.58 | awk | well thats you |
| 14:02.03 | _x86_ | most people use iaxmodem/hylafax, actually |
| 14:02.05 | coppice | awk: I use hylafax for testing, bu I don't remember ever setting a user ID or password |
| 14:02.09 | *** join/#asterisk ZaVoid (n=zavoid@host-24-225-239-34.patmedia.net) |
| 14:02.12 | ZaVoid | hey all |
| 14:02.25 | awk | coppice why I say this, is I have a windows client that references a hylafax server |
| 14:02.25 | ZaVoid | when i use a SayNumber command for 16 |
| 14:02.31 | awk | but in the port setting it says, user/pass |
| 14:02.36 | awk | some sort of auth |
| 14:02.39 | ZaVoid | it says sixteen... but i don't see a sixteen soundfile.. just a sixteenth |
| 14:02.45 | ZaVoid | is that the same i guess? |
| 14:02.56 | awk | I know you have faxadduser, etc but i dont see what file it passes to |
| 14:03.04 | awk | maybe then i could understart it better |
| 14:03.16 | coppice | awk: oh, for that you need to create an ordinary unix account on the machien running hylafax |
| 14:03.33 | awk | coppice ahhh, so any user account should work |
| 14:04.00 | awk | with correct perm associated to hylafax that is |
| 14:04.05 | coppice | if you have an ID and password that can log in, it should work for passing those hylafax jobs |
| 14:04.24 | ManxPower | Louisiana must be freaking. A small tropical storm turned into a hurricane overnight and made landfall |
| 14:04.36 | awk | _x86_ hmm, leet.org doesn't jakes own that domain ? |
| 14:04.59 | ruied_ | does anyone have an 26992pci isdn car working? my L2 doesn't goes up, I also have a digium t410p and it works fine... I would like to set w6692pci, but I don't know if it is a mISDN problem, since mISDN is supposed to work with this card, and it detects fine.... |
| 14:05.08 | ruied_ | car=card |
| 14:05.15 | *** join/#asterisk jfitzgibbon (n=chatzill@64.72.237.187) |
| 14:06.30 | awk | coppice hrm, nope I keep getting login incorrect |
| 14:06.37 | awk | even with root account |
| 14:06.56 | awk | Sep 13 16:06:13 udi HylaFAX[14318]: pam_authenticate failed in pamCheck with 0x7: Authentication fail ..... |
| 14:07.22 | arekm | ManxPower: the channel bank has "clock-source-eligible" option, should this be enabled? I assume that my 4xT1 card is providing clock to channel nak right? |
| 14:07.31 | [TK]D-Fender | etfonhomey: I suppose that might do |
| 14:07.38 | *** join/#asterisk mrbond82 (n=bondo@d221-91-164.commercial.cgocable.net) |
| 14:07.40 | ManxPower | arekm: I have no idea. |
| 14:08.18 | rantsh | nicchap: Asterisk 1.2.3 |
| 14:08.24 | [TK]D-Fender | arekm: what we showed you configures your FIRST span. You'll have to add span declarationsfor the rest depending on how you deploy them. |
| 14:08.39 | mrbond82 | What's the likely candidate with being able to call another asterisk server and ring an extension there and after the person answers there is no voice being transmitted in either direction. If I hang up or I hang up, my extension knows about it (shows that the call has ended) |
| 14:09.03 | *** join/#asterisk barros (n=barros@189-19-24-162.dsl.telesp.net.br) |
| 14:09.44 | barros | Hi guys. I was wondering, is there a way to detect if a peer is unreachable via dial plan? |
| 14:10.06 | [TK]D-Fender | mrbond82: typcally NAT & reinvite settings |
| 14:10.08 | [TK]D-Fender | ~sipnat |
| 14:10.08 | jbot | [~sipnat] Quick guide on configuring * + SIP behind NAT : http://www.aocomputing.net/?p=3 otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
| 14:10.22 | arekm | [TK]D-Fender: ok, I understand |
| 14:10.36 | Digitaldaz21 | Could anyone tell me why in an agi script the following: $hint = $agi->database_get('DEVICE', '501/dial'); puts the value 'Array' in $hint? |
| 14:10.58 | mrbond82 | D-Fender would I need to worry about nat if the asterisk box is also the gateway (and thus has a direct connection to the internet) |
| 14:11.19 | [TK]D-Fender | barros: to get info like that You could do it in AGI where you'd have access to AMI/ * CLI type options. Certainly nothing "simple" or "easy" for this though. |
| 14:12.07 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 14:13.04 | ruied_ | how or where can I get help about the w6692pci, I don't know if this isdn card/modem is fully supported by mISDN, because it doesn't work fine with *... I don't know if just quit of the card or if there is any special configuration missing... I don't know what path I should take.... |
| 14:13.12 | Uatec | hi |
| 14:13.20 | Uatec | i'm making this call using SIP, to my other asterisk box |
| 14:13.27 | Uatec | but when making the call i get to: |
| 14:13.28 | barros | [TK]D-Fender, hmm.. maybe it is time to move to AGI. So, here it goes another question: my dialplan is generated on-the-fly via an external module that fetch data from the database (not realtime). The dialplan make calls do macros I have defined in a static file. These macro is the core of my dialplan. So, can I call AGI scripts from the dialplan? |
| 14:13.39 | Uatec | <PROTECTED> |
| 14:13.40 | Uatec | <PROTECTED> |
| 14:13.42 | Uatec | and that's it |
| 14:13.47 | Uatec | nothing happens from there |
| 14:13.48 | Uatec | it just waits |
| 14:13.51 | [TK]D-Fender | mrbond82: Well you've only described ONE of the servers... what about the OTHER? |
| 14:13.53 | nicchap | rantsh: My first response would definetaly be to at least upgrade to version 1.2.24. That could very well fix your issues. |
| 14:14.28 | [TK]D-Fender | barros: AGI *is* a dialplan app... |
| 14:14.30 | *** join/#asterisk javar (n=javar@69.79.134.24) |
| 14:14.33 | nicchap | rantsh: Queues have been known to be touchy in earlier versions. |
| 14:14.58 | ZaVoid | ahh i see 16 is in digits |
| 14:15.11 | [TK]D-Fender | Uatec: PASTEBIN is your friend and MAYBE it would help if you did so with SIP DEBUG enabled so you can actually see whats going on.... |
| 14:15.28 | barros | [TK]D-Fender, nice.. so maybe I can create an AGI just to fetch if the peer is reachable or not. I'm completely new to AGI, can you point me some links about it? |
| 14:15.29 | *** join/#asterisk elixer (i=elixer@65.207.74.18) |
| 14:15.29 | Uatec | i am doing that now |
| 14:15.30 | Uatec | http://rafb.net/p/zqEhHG21.html |
| 14:15.36 | Uatec | this is what i get if i turn on sip debug |
| 14:15.54 | mrbond82 | [TK]D-Fender They are both connected directly to the internet |
| 14:15.59 | [TK]D-Fender | Uatec: Retransmitting #4 (NAT) to xx.xx.xx.xx:45860: <--- sure looks like a networking path porblem to me.... |
| 14:16.01 | Uatec | it just keeps retransmitting |
| 14:16.06 | Uatec | really? |
| 14:16.29 | Uatec | that's weird becuase it's not complaining that it cant find a route |
| 14:16.30 | [TK]D-Fender | Uatec: its not even getting answered, positive OR negative. So yeah its clearly networking. |
| 14:16.35 | rantsh | nicchap: problem is it's a production server and the company's upgrade policy is very poor/slow (even though I've warned them on the fact that 1.2 branch is going to be unsupported next year) |
| 14:16.53 | *** join/#asterisk mog (i=mog@nat/digium/x-63525e74b7759a12) |
| 14:16.53 | *** mode/#asterisk [+o mog] by ChanServ |
| 14:17.05 | Uatec | i'm looking on the receiving asterisk box |
| 14:17.11 | Uatec | an it's getting nothing on the screen |
| 14:17.12 | Uatec | not a job |
| 14:17.14 | Uatec | not a jot |
| 14:17.19 | rantsh | nicchap: have you seen the conf files in the pastebin? do you think there might be something wrong there? |
| 14:17.22 | [TK]D-Fender | barros: in AGI go do an AMI call to "sip show peer [whatever]" and parse its output. |
| 14:17.35 | nicchap | rantsh: Have you checked the changelogs to see if such an issue was fixed? I can look them up for you. If so, that might be your only option. |
| 14:17.43 | [TK]D-Fender | Uatec: Of course its not getting it, thats the PROBLEM. |
| 14:17.45 | Uatec | what kind networking issue blocks SIP like that, but doesn't cause it to actively fail |
| 14:18.06 | [TK]D-Fender | Uatec: It can't fail if its continuing to TRY <--- |
| 14:18.22 | [TK]D-Fender | Uatec: It'd probably bomb out at some later point if you waited long enough |
| 14:18.28 | barros | [TK]D-Fender, nice.. I'll try it. thanks |
| 14:18.36 | rantsh | nicchap: that'd be nice, do you have the URL for the changelogs? |
| 14:18.38 | [TK]D-Fender | barros: You're welcome. |
| 14:18.42 | *** join/#asterisk dijungal (n=kdaniel@64.86.52.254) |
| 14:18.44 | *** join/#asterisk mbranca (n=matteo@host-210-mi.linuxserver.it) |
| 14:19.00 | Uatec | yes, it does fail eventually |
| 14:19.03 | nicchap | rantsh: lemme look the logs for you... standbye. |
| 14:19.10 | Uatec | but usually if there's a networking problem, i.e. it can't connect, it fails immediately |
| 14:19.14 | Uatec | that's why this is confusing me |
| 14:19.17 | mrbond82 | What is canreinvite for ? |
| 14:19.21 | rantsh | nicchap: thanks |
| 14:19.26 | [TK]D-Fender | Uatec: the packet isn't being responded to..... |
| 14:19.31 | Uatec | i don't know where a problem like this could be coming from |
| 14:19.48 | *** join/#asterisk Dan0maN_Work (n=dschuh@64.149.174.136) |
| 14:19.49 | [TK]D-Fender | mrbond82: go read up on that on the WIKI |
| 14:20.03 | mrbond82 | [TK]D-Fender you are much faster than the wiki |
| 14:20.13 | [TK]D-Fender | Uatec: pastebin the ENTIRE call, and don't mask anything except PASSWORDS |
| 14:20.35 | [TK]D-Fender | mrbond82: Yes, but I get tired of your lazyness, the WIKI is far more patient. |
| 14:21.14 | *** join/#asterisk elriah (n=null@adsl-070-148-053-190.sip.bhm.bellsouth.net) |
| 14:22.38 | elriah | Hi all. I want to run a cronjob every minute to basically do a "asterisk -rx 'sip show peers' >somefile.txt" so I can parse it out with another tool that doesn't have permission to asterisk directly. Would this action cause any performance issues in a medium volume system say that has on average 25-50 active calls (4 Cores/4GB)? |
| 14:23.32 | snowchyld | IT WAS THE PINS! |
| 14:23.35 | *** part/#asterisk snowchyld (n=snowchyl@naku.org) |
| 14:23.40 | dijungal | i have a problem where agent's in a queue, will get another call to their phone, even if they're already on a call. I am using asterisk 1.2.19, how do i solve this? |
| 14:23.43 | Uatec | this is the problem with your development system being a live system as well |
| 14:24.03 | *** join/#asterisk anthm (n=anthm@CPE-72-131-113-50.wi.res.rr.com) |
| 14:24.03 | *** mode/#asterisk [+o anthm] by ChanServ |
| 14:25.09 | *** join/#asterisk wltjr (n=wltjr@gentoo/developer/wltjr) |
| 14:25.46 | wltjr | greetings telco peeps, anyone got any info on Aastra products, thumbs up/down, any experience horror stories you care to share? |
| 14:26.19 | mrbond82 | [TK]D-Fender this wiki looks like you wrote it. |
| 14:26.42 | [TK]D-Fender | elriah: You're better off running a daemon that does this via AMI as a single process. Less overhead and no auth problems. |
| 14:26.58 | [TK]D-Fender | mrbond82: I have never edited anything on it. |
| 14:27.29 | [TK]D-Fender | elriah: And this way would not pose any load issues whatsoever |
| 14:27.32 | alrs | wltjr: Aastra firmware can be mildly flaky, but they're well put together. |
| 14:27.49 | alrs | wltjr: come to think of it, all SIP phones have flaky firmware |
| 14:28.00 | [TK]D-Fender | wltjr: Aastra works, but is a runner up to Polycom who wins first place on the suggest list, hands down. |
| 14:28.17 | *** join/#asterisk blitzrage (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 14:28.17 | *** mode/#asterisk [+o blitzrage] by ChanServ |
| 14:28.27 | *** join/#asterisk jmesquita (n=jmesquit@c915230f.virtua.com.br) |
| 14:28.38 | wltjr | alrs: would you recommend over polycoms? client is kinda fixated on wireless |
| 14:29.02 | arekm | maybe I have wrong T1 cable? |
| 14:29.46 | elriah | [TK]D-Fender: Thanks for the help. That makes sense, thanks. I had a problem with early 1.2 builds that just core dumped whenusing the Asterisk Manager Interface but if I remember I was trying to make a bunch of seperate connections. |
| 14:30.41 | alrs | wltjr: I think Polycom recently bought some company to have a wireless offering, but I don't know much about it. I have coworkers that use the Aastra wireless stuff and they haven't complained |
| 14:30.55 | Uatec | http://rafb.net/p/RJUmZ974.html <-- this is the entire sip debug data generated when i make the call |
| 14:31.14 | wltjr | [TK]D-Fender: yeah I am not really pleased client purchased gear other than recommended, and of course they want me to be as affordable (cheap) on pricing to install and deal with them, which I can with the Polycoms, but not familiar with Aastra, so not sure what I would be getting into |
| 14:31.29 | Uatec | my dial line is: exten => s,1,Dial(SIP/lucifer/${ARG1}) |
| 14:31.43 | alrs | wltjr: Polycoms are easier to deal with configuration from handset, Aastras can get more done in web interface |
| 14:32.36 | wltjr | I am looking for remote hands off install. I want to know like a mac address, and be able to do the config and phone specific stuff in dhcp and tftp ideally, like I am for existing Ciscos now |
| 14:33.01 | Uatec | and my lines in sip.conf are: http://rafb.net/p/FeL00M36.html |
| 14:33.23 | wltjr | I did not like having to manually setup my grandstream, and I have seen ways to push configs to them, but didn't get to mess with that myself, I bought it mostly to see what the quality was like at the low end, not to bad, only compliant is lack of volumne |
| 14:34.22 | wltjr | the aastras seem to have a config file so I will look into that. another thing I hear about polycoms was you can easily reboot them remotely :) |
| 14:35.06 | Uatec | [TK]D-Fender, is any of that information useful? |
| 14:36.27 | jfitzgibbon | wltjr: you can remotely tell an Aastra to reload it's config. You just need "Event => check-sync" and "Content-Length => 0" in sip_notify.conf |
| 14:37.01 | wltjr | cool, I think client might have bought them already, so might be inevitable |
| 14:37.36 | nicchap | rantsh: According to the changelog, there have been many fixes and changes the the app_queue app. There is no specific mention of your problem, but it could very well have been fixed. |
| 14:37.54 | *** join/#asterisk saftsack (n=saftsack@pD9E07E2D.dip.t-dialin.net) |
| 14:38.03 | jfitzgibbon | wltjr: I only have a single Polycom, and overall I wouldn't say the provisioning is worse for Aastra. The slight downside is that aastras will just pull two config files - aastra.cfg and $MAC.cfg, whereas polycom can be told what additional files to pull in |
| 14:38.19 | jfitzgibbon | wltjr: makes it a bit easier if you need fine-grained multi-layered config templates I suppose |
| 14:38.35 | jfitzgibbon | wltjr: but if you create $MAC.cfg using some homegrown tool, the difference is moot |
| 14:38.43 | nicchap | rantsh: I would recommend a test environment for you to check in. Especially if you need to update your production env. one day. |
| 14:38.58 | wltjr | <PROTECTED> |
| 14:39.00 | nicchap | rantsh: But I'm sure you already know that |
| 14:39.24 | rantsh | nicchap: mmm... I see, thanks for takind the time to read the changelog for me, I really appreciate the fact that you're going throuhg all that trouble to help me out... thanks |
| 14:40.00 | nicchap | rantsh: Wish I could do more then just tell you to update, but it is the obvous choice. GL. |
| 14:40.05 | rantsh | nicchap: yup, we do have to test our B2BUA application in * 1.4 to see how it goes |
| 14:41.05 | nicchap | So, any Sangoma users out there? |
| 14:41.19 | rantsh | nicchap: thanks a lot for all the help and trouble, I'll try to upgrade my test server and see what happens |
| 14:41.30 | nicchap | OK GL. |
| 14:41.47 | nicchap | rantsh: Start with 1.2.24 first. |
| 14:41.55 | rantsh | rantsh: maybe one day I'll be able to help you... |
| 14:42.03 | nicchap | :) |
| 14:42.10 | rantsh | nicchap: sure thing |
| 14:43.44 | wltjr | alrs: [TK]D-Fender: jfitzgibbon: ty all very much for the info, sounds like I might be ok with the aastra, just never heard of the brand :) always skeptical on those scenarios |
| 14:43.51 | elriah | Hey guys how do you 'reload' the asterisk manager interface config without doing a full reload? I can't seem to find the module ... |
| 14:44.03 | elriah | In 1.2.x |
| 14:44.53 | *** join/#asterisk Shazoo (i=Shazoo@60.232.180.198) |
| 14:46.35 | *** join/#asterisk yangvnc (n=yang@cpe-212-18-59-220.dynamic.amis.net) |
| 14:47.21 | yangvnc | I am wondering where are stored usernames and password for SIP client connections, and does asterisk needs to be restarted each time those files are modified? |
| 14:49.35 | enioh | yangvnc: sip reload on asterisk console |
| 14:49.50 | enioh | yangvnc: and stored in sip.conf, or in database with realtime |
| 14:50.07 | yangvnc | enioh: thank you...i need to enter into asterisk with asterisk -r ? |
| 14:50.16 | enioh | yep |
| 14:50.38 | *** part/#asterisk mohsen (n=chatzill@81.31.160.140) |
| 14:51.29 | nicchap | last for for Sangoma users. Need to pick your brain. |
| 14:53.20 | Shazoo | hi , I had made a sip2qsig gateway , can i use it in asterisk ? |
| 14:54.00 | *** join/#asterisk putnopvu1 (i=putnopvu@nat/digium/x-8c5ad60de63b689d) |
| 14:54.08 | *** join/#asterisk NOT_guru (n=chatzill@24-241-103-142.static.stls.mo.charter.com) |
| 14:54.23 | *** join/#asterisk hypa7ia (i=hypatia@judecca.aculei.net) |
| 14:54.46 | *** join/#asterisk putnopvut (i=putnopvu@nat/digium/x-892e73c0f66c706a) |
| 14:56.43 | yangvnc | enioh: i get username/auth missmatch |
| 14:56.56 | [TK]D-Fender | wltjr: Polycom in North America costs on par with Aastra for a better phone... |
| 14:57.02 | awk | yes fax working with the pri |
| 14:57.09 | yangvnc | i dont know if its SIP supported, but i saw someone used softphone on it |
| 14:57.44 | viKing78 | [TK]D-Fender: whay do you like the Polycom better than Aastra? |
| 14:57.51 | *** join/#asterisk rmayorga (n=rmayorga@unaffiliated/rmayorga) |
| 14:58.08 | *** join/#asterisk [T]ank (n=ckwall@206.71.78.172) |
| 14:58.25 | [T]ank | does anyone know if there is a channel for discussing asterisk on OpenWRT? |
| 14:58.28 | wltjr | <PROTECTED> |
| 14:59.16 | ruied_ | what does it means 'capi' ? |
| 14:59.30 | Uatec | [TK]D-Fender,did you look at the data i posted? |
| 14:59.32 | Uatec | http://rafb.net/p/RJUmZ974.html <-- this is the entire sip debug data generated when i make the call |
| 14:59.35 | Uatec | my dial line is: exten => s,1,Dial(SIP/lucifer/${ARG1}) |
| 14:59.38 | Uatec | and my lines in sip.conf are: http://rafb.net/p/FeL00M36.html |
| 15:00.40 | [TK]D-Fender | viKing78: Superior call handling, more flexible use of lines, superior audio quality. |
| 15:02.06 | viKing78 | [TK]D-Fender: I have about a dozen Polycom phones (650s and 330s). Would you reccomend a Cisco phone over these? |
| 15:02.26 | ManxPower | Uatec: your calls are coming in on analog or digital FXO ports? |
| 15:02.49 | [TK]D-Fender | viKing78: Never. Frankly as far as SIP goes, Cisco is THRID or worse on the suggest list. |
| 15:02.54 | alrs | wltjr: Cisco phones blow |
| 15:03.07 | ManxPower | Uatec: and your Dial line is screwed up |
| 15:03.12 | ManxPower | Dial(SIP/number@peer) |
| 15:03.42 | viKing78 | wltjr: Why do you think Cisco phones are better that Polycom? |
| 15:03.42 | [TK]D-Fender | ManxPower: Thats not screwed up |
| 15:04.22 | coppice | the cisco handsets are the best if you want to hit someone. very heavy and solid :-) |
| 15:04.30 | [TK]D-Fender | Uatec: Retransmitting #1 (NAT) to 83.148.178.177:45860: <--- this shouldn't think its NAT. |
| 15:04.41 | wltjr | viKing78: just my experience with them |
| 15:05.07 | wltjr | only thing I don't like about Cisco phones is the whole smart net requirement aspect to get firmware and etc |
| 15:05.28 | wltjr | it's gotten harder and harder to get firmware over the years, so I am reduce use of all cisco gear, short of like switches |
| 15:07.40 | [TK]D-Fender | Uatec: And you never described Lucifer's setup |
| 15:09.51 | *** join/#asterisk disposable (n=michal@82-69-60-92.dsl.in-addr.zen.co.uk) |
| 15:11.33 | *** join/#asterisk clayeastwood (n=clayeast@69-215-202-195.ded.ameritech.net) |
| 15:11.36 | *** join/#asterisk flujan (n=flujan@200-160-115-020.static.spo.ctbc.com.br) |
| 15:12.44 | *** join/#asterisk brodiem (n=brodiem@67.110.68.66.ptr.us.xo.net) |
| 15:13.07 | brodiem | Anyone know if compiling asterisk for x86-64 will have any noticeable gains over IA32? |
| 15:15.47 | [TK]D-Fender | brodiem: Sure, you don't have to wait half as long to core-dump ;) |
| 15:16.02 | brodiem | haha |
| 15:16.41 | coppice | the modems in spandsp run a lot faster compiled for x86_64 |
| 15:16.42 | [TK]D-Fender | brodiem: But I haven't heard anything positive form anyone who has tried it, only negative... |
| 15:16.45 | brodiem | so I take it that it generally won't be as stable? I assume probably because of modules that won't compile as 64? |
| 15:17.38 | coppice | things have run just fine with x86_64 for a couple of years |
| 15:17.46 | brodiem | coppice: what about transcoding? anything noticeable? |
| 15:18.38 | coppice | i've never used the G.729 codec. it won't make much difference to the others |
| 15:20.05 | disposable | i have an asterisk server with sip phones (gxp-2000). i want to buy a voip package that works through IAX. does it make any difference (to my phones) what connection my asterisk uses with the provider? |
| 15:21.11 | clayeastwood | disposable: probably doesn't matter |
| 15:22.13 | disposable | clayeastwood, unfortunately i cannot buy a package that "might" work |
| 15:23.29 | [TK]D-Fender | disposable: No, it doesn't matter. |
| 15:23.50 | [TK]D-Fender | disposable: * is a B@BUA and translates between ends of each call regardless. |
| 15:25.11 | disposable | [TK]D-Fender, although i don't understand the second sentence (was it for me?), thank you |
| 15:25.24 | *** join/#asterisk Tili (n=tili@cm48.gamma244.maxonline.com.sg) |
| 15:27.50 | Tili | I have an issue. I create an outgoing call in spool dir and when the user picks up i dial to another number to connect him. sort of auto callback |
| 15:28.16 | Tili | but asterisk makes another call to first user(A) while calling the other party (B) |
| 15:28.29 | Tili | (A) has already picked up on callback. why asterisk wud call A again |
| 15:30.29 | [TK]D-Fender | disposable: Yes, it was. When your IAX2 ITSP sends you a call, its a call between * and your ITSP over IAX2. If your * setup decides to ring one of your SIP phones, the cals is between * and your phone and the audio is bridged and translated by *. |
| 15:31.10 | [TK]D-Fender | Tili: I guess you'd better pastebin your call file, any dialplan it uses, and the CLI output of your failed call.... |
| 15:31.12 | [TK]D-Fender | ~pb |
| 15:31.13 | jbot | A Pastebin is a place to paste your stuff without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste |
| 15:31.14 | [TK]D-Fender | ^^^^^^^^^^^^^^^^ |
| 15:31.31 | Tili | [TK]D-Fender: will do |
| 15:31.44 | *** join/#asterisk BockBilbo (n=BockBilb@eu85-84-62-227.clientes.euskaltel.es) |
| 15:31.46 | BockBilbo | helo |
| 15:32.13 | lirakis | anyone been to "asterisk world?" before? |
| 15:33.11 | *** join/#asterisk drako (n=ljd@nelug/coreteam/luisjose) |
| 15:35.07 | Tili | [TK]D-Fender: i found it. it is some weird difference between 1.2 and 1.4 implementation of IAX2. my iax2 softphone doesnt work well with 1.4 |
| 15:35.34 | [TK]D-Fender | lirakis: Asterisk World! Party time! Excellent!!! SCHWING! |
| 15:36.40 | *** join/#asterisk henkoegema (n=henkoege@d54C552E4.access.telenet.be) |
| 15:36.52 | henkoegema | <PROTECTED> |
| 15:37.06 | lirakis | [TK]D-Fender: lol |
| 15:37.23 | [TK]D-Fender | henkoegema: You don't say! |
| 15:37.37 | lirakis | [TK]D-Fender: .. im getting a 2 day conf. trip.. and i have to choose between 2 days of von.. or 2 days of * world |
| 15:37.37 | *** join/#asterisk Peaceful (n=peaceful@70.98.162.62) |
| 15:37.47 | henkoegema | YEAH I do :-) |
| 15:38.06 | lirakis | .. ive never been to *world .. so im not sure if its as party time! excellent! as you seem to imply.. ha ha... |
| 15:38.09 | Peaceful | Will the new "Set(TIMEOUT(digit)=3)" syntax work in 1.2.24? I know that that's the syntax you're supposed to use in 1.4... |
| 15:38.12 | [TK]D-Fender | lirakis: hate to say I'd go with VON. Much bigger show, and its teh TOYS you want to see.... |
| 15:38.27 | Qwell | is't it the same thing? :p |
| 15:38.28 | [TK]D-Fender | Peaceful: thats 1.2 syntax. |
| 15:38.32 | lirakis | [TK]D-Fender: .. well if you go to *world.. i believe they give you an expo pass |
| 15:38.40 | lirakis | so you can still see toys.. :D |
| 15:38.47 | lirakis | like the new gxp1200 i bet! whoo! |
| 15:38.48 | lirakis | lol |
| 15:38.49 | Peaceful | oh? I've been using "DigitTimeout(3)" in 1.2 |
| 15:38.54 | dlynes_laptop | Is it possible to intercom more than one phone simultaneously? |
| 15:38.57 | [TK]D-Fender | lirakis: But remember, its the MOST toys that win! ;) |
| 15:39.03 | dlynes_laptop | i.e. internal page? |
| 15:39.09 | [TK]D-Fender | dlynes_laptop: Sure, welcome to app_page |
| 15:39.25 | dlynes_laptop | [TK]D-Fender: so that's not external page, then? |
| 15:39.33 | lirakis | [TK]D-Fender: .. yeah.. the "sip tutorial" looks pretty nice at von.. but the asterisk+openser workshop looks nice at *world.. hrmm.. :\ |
| 15:39.46 | [TK]D-Fender | dlynes_laptop: "internal" and "external" are wonderfully vague and useless terms :) |
| 15:39.50 | BockBilbo | hey again, im about to start learning asterisk. Im checking some web interfaces, which should i try, asterisk GUI or FreePBX? |
| 15:40.17 | lirakis | BockBilbo: neither! muwhahaha! |
| 15:40.21 | [TK]D-Fender | lirakis: if * World covers some specific points that are of real interest weight it all in accordingly. |
| 15:40.21 | BockBilbo | xD |
| 15:40.25 | BockBilbo | haha |
| 15:40.25 | BockBilbo | xD |
| 15:40.29 | dlynes_laptop | [TK]D-Fender: well, 'external' would be through the soundcard's line out/speaker out jacks, and internal would be through the phones |
| 15:40.30 | [TK]D-Fender | BockBilbo: none of the above |
| 15:40.43 | BockBilbo | why not lirakis , [TK]D-Fender ? |
| 15:40.44 | [TK]D-Fender | dlynes_laptop: EITHER |
| 15:40.45 | lirakis | BockBilbo: seriously though.. dont.. ha ha.. learn to use the configs |
| 15:40.52 | dlynes_laptop | [TK]D-Fender: ok, thanks muchly |
| 15:41.03 | BockBilbo | right... ive been trying to configure some sip users... |
| 15:41.11 | *** join/#asterisk t3rror (n=t3rror@gateway.sscgp.com) |
| 15:41.11 | BockBilbo | but cant manage to log in |
| 15:41.14 | lirakis | BockBilbo: you will get infinite flexibility from the conf files... NOT from a web interface |
| 15:41.16 | [TK]D-Fender | BockBilbo: because by trying to rely on a GUI you won't learn ANYTHING and when things go wrong you'll have no experience to debug or any support here. |
| 15:41.26 | BockBilbo | i know that... |
| 15:41.29 | BockBilbo | true |
| 15:41.38 | [TK]D-Fender | BockBilbo: And clarify "can't log in" please. |
| 15:41.42 | BockBilbo | well |
| 15:41.53 | BockBilbo | nmap says that the sip port is closed |
| 15:42.06 | lirakis | BockBilbo: psst.. dont tell any one.. but the first * box i had was trixbox... it was a revalation to get off that thing and onto a proper * system |
| 15:42.10 | BockBilbo | and even though ive added the user to sip.conf, and the extension on extension.conf |
| 15:42.26 | BockBilbo | hehe |
| 15:42.27 | BockBilbo | xD |
| 15:42.35 | lirakis | BockBilbo: you will be much better off in the long run learning the stuff.. rather than having a pretty gui abastract what is really going on from you |
| 15:42.57 | [TK]D-Fender | BockBilbo: And SIP uses UDP, not TCP so stop trying to nmap it. |
| 15:43.02 | BockBilbo | mmm |
| 15:43.06 | BockBilbo | :$ |
| 15:43.11 | [TK]D-Fender | BockBilbo: wake up time. |
| 15:43.19 | BockBilbo | i though nmap showed both udp and tcp ports |
| 15:43.19 | BockBilbo | my bad |
| 15:43.25 | [TK]D-Fender | BockBilbo: So that in mind enable SIP debug at * CLI and pastebin an incoming call. |
| 15:43.27 | blitzrage | you have to tell it to do UDP |
| 15:43.53 | [TK]D-Fender | BockBilbo: Actually you need to stop chasing wild geese and start higher up |
| 15:44.05 | [TK]D-Fender | BockBilbo: PASTEBIN is your friend. |
| 15:44.06 | ManxPower | [TK]D-Fender: wild asteroids? |
| 15:44.07 | [TK]D-Fender | ~pb |
| 15:44.07 | jbot | A Pastebin is a place to paste your stuff without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste |
| 15:44.17 | BockBilbo | mmm ok |
| 15:44.22 | BockBilbo | well, hold on [TK]D-Fender |
| 15:44.28 | BockBilbo | i have to start all over again |
| 15:44.34 | BockBilbo | and ill paste that |
| 15:44.37 | BockBilbo | one sec |
| 15:44.47 | [TK]D-Fender | ManxPower: Yes, it is my sincere belief that there is a giant piece of space rock jsut waiting to crash down and obliterate the stupid...... |
| 15:45.01 | BockBilbo | (i was trying to log in using sip from ekiga by the way) |
| 15:45.30 | ManxPower | [TK]D-Fender: if it only killed the stupid there would not be many people left. |
| 15:45.49 | ManxPower | But it sure would be a nice world after the dust settled. |
| 15:46.08 | *** join/#asterisk [hC] (n=hardcore@S01060016b61c8983.vf.shawcable.net) |
| 15:46.37 | *** join/#asterisk jcmoore (n=jcmoore@unaffiliated/tgrman) |
| 15:47.43 | [TK]D-Fender | ManxPower: don't worry, atmospheric friction will whittle each piece to the exact size required for tidy lethality leaving the worthy to inherit the earth :p |
| 15:47.53 | BockBilbo | [TK]D-Fender , lirakis im using the asterisk binary from ubuntu gutsy, which is 1.4, that's ok |
| 15:47.53 | BockBilbo | right? |
| 15:48.06 | jfitzgibbon | all non-stupid people who want to be gabage collectors, put your hands up |
| 15:48.17 | jfitzgibbon | it would be a nice world for a week, maybe two |
| 15:48.19 | [TK]D-Fender | BockBilbo: Oh yeah, you're labelling yourself perfectly.... |
| 15:48.27 | BockBilbo | xDD |
| 15:48.31 | *** join/#asterisk CunningPike (n=arodgers@204.239.12.183) |
| 15:48.35 | BockBilbo | or.. you are xD |
| 15:48.42 | BockBilbo | ok |
| 15:48.43 | BockBilbo | .. |
| 15:48.43 | [TK]D-Fender | jfitzgibbon: Its a dirty job, but it pays remarkably well |
| 15:48.45 | BockBilbo | ill compile the source |
| 15:48.46 | BockBilbo | xD |
| 15:48.46 | lirakis | BockBilbo: ive had no luck with ekiga |
| 15:48.49 | ManxPower | jfitzgibbon: Um, you don't have to be STUPID to be a garbage collector - just not very ambitious |
| 15:48.55 | *** join/#asterisk tru_`z24 (n=truz_`24@74-129-166-232.dhcp.insightbb.com) |
| 15:48.57 | [TK]D-Fender | bock if its installed, then its fine. |
| 15:48.59 | lirakis | BockBilbo: i generally use x-ten |
| 15:49.12 | BockBilbo | ok [TK]D-Fender |
| 15:49.12 | [TK]D-Fender | bock here's the big question : is your Ekiga *ON* your * server PC? |
| 15:49.22 | BockBilbo | yes [TK]D-Fender |
| 15:49.26 | BockBilbo | thoigh |
| 15:49.30 | [TK]D-Fender | BockBilbo: There's a nearly immediate problem |
| 15:49.51 | BockBilbo | I've also tried to log in via sip with a linksys PAP2-NA |
| 15:49.54 | [TK]D-Fender | BockBilbo: They will FIGHT for the SIP port (5060) and chances are * won't LOAD SIP support because of it. |
| 15:50.09 | [TK]D-Fender | BockBilbo: hence no SIP ANYWHERE |
| 15:50.13 | BockBilbo | ok.. i see |
| 15:50.14 | BockBilbo | xD |
| 15:50.15 | BockBilbo | well |
| 15:50.22 | [TK]D-Fender | BockBilbo: Set your Ekiga to use 5061 |
| 15:50.23 | BockBilbo | i was trying all of this on my laptop |
| 15:50.38 | [TK]D-Fender | BockBilbo: Would you like fries with that, sir? |
| 15:50.40 | BockBilbo | i was going to tell you that im going to install asterisk on my srver |
| 15:50.45 | BockBilbo | xDDDDD |
| 15:51.26 | BockBilbo | ok |
| 15:51.27 | [TK]D-Fender | BockBilbo: So go change Ekiga, or stop it entirely and restart * and test with your ATA only |
| 15:51.42 | BockBilbo | ok |
| 15:51.47 | [TK]D-Fender | BockBilbo: And then provide that pastebin. |
| 15:52.03 | BockBilbo | alrite |
| 15:52.07 | *** join/#asterisk _ShrikE (n=ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 15:52.49 | *** join/#asterisk drako (n=ljd@nelug/coreteam/luisjose) |
| 15:54.14 | drako | 2 message not nice in my CLI |
| 15:54.16 | drako | [Sep 13 17:32:00] NOTICE[3962]: res_features.c:1241 ast_feature_request_and_dial: Don't know what to do about control frame: -1 |
| 15:54.18 | drako | [Sep 13 17:32:00] WARNING[3962]: cdr.c:831 ast_cdr_init: CDR already initialized on 'Local/10@local-ext-b4f2,1' |
| 15:54.27 | drako | any idea? |
| 15:54.58 | *** join/#asterisk qdk_ (n=qdk@0x50c627be.bynxx11.adsl-dhcp.tele.dk) |
| 15:55.31 | *** join/#asterisk yannj_fr (n=yannj_fr@vpn.intelunix.fr) |
| 15:57.52 | henkoegema | Cant get my ENUM working. Who wants to test for me :) see http://www.pastebin.com , |
| 16:00.42 | codefreeze | drako: it's ok; in 1.4, all channels automatically get a cdr initialized when they are created. I think I got rid of the message in the latest rel... |
| 16:03.40 | drako | codefreeze, Asterisk 1.4.11 |
| 16:03.50 | drako | codefreeze, this is the version im using |
| 16:04.20 | *** join/#asterisk VoicePulse (n=contact@unaffiliated/voicepulse) |
| 16:05.37 | codefreeze | drako: hmmm. maybe in 1.4.12 then... ;) |
| 16:09.16 | BockBilbo | [TK]D-Fender should i open the sip port on iptables? |
| 16:09.38 | [TK]D-Fender | BockBilbo: Gee, do you THINK you should be blocking it? How about RTP to while you're at it? |
| 16:10.06 | BockBilbo | mmm let me check |
| 16:10.22 | drako | [TK]D-Fender, any idea about [Sep 13 17:32:00] NOTICE[3962]: res_features.c:1241 ast_feature_request_and_dial: Don't know what to do about control frame: -1 |
| 16:10.57 | |Vulture| | Anyone have a recommendation for a good IAX2/SIP Toll-Free provider? |
| 16:11.12 | [TK]D-Fender | drako: I heard you the first time... if I had anything to comment, you'd have gotten it already :) |
| 16:11.22 | drako | :P |
| 16:11.25 | [TK]D-Fender | |Vulture|: ... |
| 16:11.27 | [TK]D-Fender | ~itsp |
| 16:11.28 | jbot | An ITSP (Internet Telephony Service Provider.) is a "VoIP Phone Company". Example : Vonage, Broadvoice, Teliax, VoicePulse, etc. "All ITSPs suck. Some suck less than others. Teliax seems to suck less than most.." (tm) (c) 2007 ManxPower |
| 16:12.11 | ManxPower | I'm famous! |
| 16:12.25 | |Vulture| | lol |
| 16:12.39 | |Vulture| | man I remeber Manx and D-Fender from way back when lol |
| 16:12.48 | *** join/#asterisk mistermocha (n=chef@adsl-75-40-192-38.dsl.irvnca.sbcglobal.net) |
| 16:12.53 | mistermocha | haroo |
| 16:13.08 | |Vulture| | Ill go with Teliax since they suck less |
| 16:13.21 | ManxPower | drako: most "I don't know what to do with X" messages in Asterisk are because you don't have a /etc/asterisk/indications.conf I doubt that is YOUR problem, but it is easy to setup indications.conf Just use the sample one. |
| 16:13.28 | mistermocha | question for you all... what might cause * to stop producing master.csv files? |
| 16:13.52 | ManxPower | mishehu: if it is bigger than 2GB or you ran out of disk space on that partition |
| 16:14.09 | ManxPower | or you disabled loading of cdr_csv |
| 16:14.49 | *** join/#asterisk outtolunc (n=me@adsl-66-218-53-172.dslextreme.com) |
| 16:15.14 | mistermocha | hmm... the former is not the case |
| 16:15.29 | mistermocha | now the latter... I'll have to look into it |
| 16:20.11 | ManxPower | df -h |
| 16:20.17 | ManxPower | that would be the way to look at it |
| 16:21.02 | henkoegema | how can I find my post back in: http://www.pastebin.com ? can only see 8 recent posts? where are the rest ? |
| 16:21.43 | *** join/#asterisk dijungal (n=kdaniel@209.59.110.15) |
| 16:22.56 | [TK]D-Fender | henkoegema: You should try posting the link to YOUR paste. |
| 16:23.37 | *** join/#asterisk flujan (n=flujan@200-160-115-020.static.spo.ctbc.com.br) |
| 16:27.21 | *** join/#asterisk fordfrog (n=fordfrog@gentoo/developer/fordfrog) |
| 16:30.27 | keith4_ | wow |
| 16:30.28 | *** part/#asterisk luka1 (n=luka1@mar35.ttc.cz) |
| 16:31.58 | thewiizle | hi, whats the most recommended os for an asterisk installation |
| 16:34.59 | russellb | windows vista home premium office edition |
| 16:35.03 | Strom_M | whichever linux you're most comfortable with |
| 16:35.07 | Strom_M | hahahaahah russellb |
| 16:35.09 | russellb | jk ... linux, generally. |
| 16:35.27 | russellb | ;) |
| 16:36.17 | tru_`z24 | with the tdm400p, can you transfer calls from the fxo to the fxs module? And can you transfer from the fxo to an outside phone number? |
| 16:36.43 | thewiizle | any particular flavour |
| 16:36.48 | *** part/#asterisk nicchap (n=nicchap@204.101.222.130) |
| 16:36.57 | russellb | thewiizle: nope, whatever you are most comfortable with. |
| 16:37.01 | ManxPower | tru_`z24: 1) Yes. 2) Yes, but it will take up two lines |
| 16:37.03 | thewiizle | ok sweet |
| 16:37.14 | russellb | preferably something recent |
| 16:37.22 | thewiizle | core6 |
| 16:37.29 | tru_`z24 | ManxPower: so once the outside line picks up, I can't complete the conference call by hanging up? |
| 16:37.30 | thewiizle | well no |
| 16:37.30 | russellb | some people try to run asterisk on redhat 8 and wonder why they have problems |
| 16:37.31 | thewiizle | sod that |
| 16:37.35 | thewiizle | core 2 |
| 16:37.38 | thewiizle | at best |
| 16:37.45 | thewiizle | maybe 3 or 4 |
| 16:37.50 | thewiizle | just bloated |
| 16:37.59 | ManxPower | tru_`z24: the problem is sending a flash to the telco line. |
| 16:38.05 | russellb | heh ... if you're going to use fedora, you should really use the latest release |
| 16:38.06 | BockBilbo | [TK]D-Fender how do i know from the * cli if a sip user is logged in? ¿sip show peers?? |
| 16:38.19 | russellb | for the sake of getting supported packages ... |
| 16:38.20 | ManxPower | Also, the telco must permit transfers (rather than just a three-way call) |
| 16:38.40 | ManxPower | BockBilbo: yes, "sip show peers" |
| 16:38.44 | thewiizle | hmm just thought that with core6 coming bundled with x an older might be better |
| 16:38.49 | [TK]D-Fender | BockBilbo: thats one way. Then you could also do what I said earlier and ENABLE SIP DEBUG |
| 16:38.51 | tru_`z24 | Which line would it tie up, the fxo or fxs line? |
| 16:38.52 | tru_`z24 | or both? |
| 16:38.59 | tru_`z24 | i'd assume the fxo? |
| 16:39.08 | tru_`z24 | meaning the agent could take another call |
| 16:39.11 | ManxPower | tru_`z24: it would tie up 2 FXOs, one for the incoming call, one for the outgoing call. |
| 16:39.15 | tru_`z24 | ouch |
| 16:39.28 | russellb | thewiizle: surely you can do a minimal install without x, right? |
| 16:39.30 | *** join/#asterisk saftsack (n=saftsack@pD9E07E2D.dip.t-dialin.net) |
| 16:39.39 | thewiizle | yeh right ;) |
| 16:39.52 | ManxPower | you would prolly spend a week or two trying to get Conference Drop and Transfer service from your telco, then much more time trying to make Asterisk send a flash to the FXO. |
| 16:40.13 | BockBilbo | great!!! |
| 16:40.14 | russellb | thewiizle: ha, then i would advise going with something else, then, that is up to date and doesn't suck |
| 16:40.17 | BockBilbo | sip works! |
| 16:40.26 | ManxPower | I don't know of a telco that has CDT for residential lines, only business lines, but our telco might offer it |
| 16:41.11 | tru_`z24 | well this is a business line |
| 16:41.11 | ManxPower | tru_`z24: the real challange is to get Asterisk to send a FLASH to the telco. |
| 16:41.11 | thewiizle | heh, well im just loading the installer now to see if i can stop X |
| 16:41.11 | tru_`z24 | Right |
| 16:41.11 | thewiizle | other than that i might just go centos5 |
| 16:41.12 | ManxPower | Asterisk was never designed to do that. |
| 16:41.13 | thewiizle | as its basically RHEL4 |
| 16:41.54 | ManxPower | BTW, the Flash() app in Asterisk would normally send the flash to the FXS |
| 16:41.55 | tru_`z24 | Is it possible to use asterisk at a lower level and not use it's scripting language? |
| 16:42.00 | ManxPower | tru_`z24: no |
| 16:42.21 | ManxPower | But you could write a PBX to use the ZAPTEL API |
| 16:42.37 | tru_`z24 | We have dialogic boards now using an in-house written software package which uses a software library called "Multi-vox" |
| 16:42.51 | tru_`z24 | And im trying to evaluate if asterisk is flexible enough to be used |
| 16:43.05 | ManxPower | Asterisk is a PBX. |
| 16:43.08 | tru_`z24 | because I need to take data and write it to our database, and I want to make sure asterisk provides hooks or access to the data |
| 16:43.17 | ManxPower | Well, it is more of a PBX Toolkit that allows you to build your own PBX |
| 16:43.45 | [TK]D-Fender | tru_`z24: Its can do all that, but "funcky analog line manipulation" may be tricky depending. |
| 16:46.21 | tru_`z24 | I don't know if i'd classify it as funky, a customer calls our phone, auto attendant asks them a question, they press 1 or 2, if they press 2, it transfers them to live customer support... if the rep can't help them they might transfer to an outside party |
| 16:47.22 | *** join/#asterisk zeedo (n=zeedo@reboot-robot.net) |
| 16:47.33 | *** join/#asterisk deeperror (n=deeperro@69-215-202-195.ded.ameritech.net) |
| 16:47.53 | deeperror | I'm having a zt_exception that gets thrown and causes an endless loop. Any clues? http://pastebin.ca/695949 |
| 16:48.14 | dijungal | i'm running * 1.2.19, and i'm having an issue where agents are getting multiple calls to their phones, while on a call... |
| 16:48.33 | *** join/#asterisk jfg (n=jfg@dyn-83-152-249-84.ppp.tiscali.fr) |
| 16:48.35 | jfg | hi |
| 16:48.45 | [TK]D-Fender | tru_`z24: that isn't the tricky idea. getting * to take that exacty channel and DIRECTLY tell the telco to take it back and trasfer out and LIBERATE the line it was on is tricky. |
| 16:49.06 | [TK]D-Fender | tru_`z24: Just having * dial out ANOTHER port is no difficulty at all... |
| 16:49.06 | thewiizle | dijungal, call waiting |
| 16:49.14 | tru_`z24 | Well, but it is possible if i'm willing to tie up both fxo's tho right? |
| 16:49.14 | dijungal | call-limit=1 works for awhile then "sip show inuse" says they're on calls but they're not |
| 16:49.27 | dijungal | and they don't get any calls after that |
| 16:49.27 | tru_`z24 | But dropping the call on my end is what is tricky? |
| 16:49.33 | dijungal | is this a bug in 1.2? |
| 16:49.51 | thewiizle | does the channel show in use |
| 16:49.56 | thewiizle | or just the extension |
| 16:50.02 | [TK]D-Fender | tru_`z24: only if you want the TELCO to take back the call on that single line so that your system is no longer implicated with its handling. |
| 16:50.03 | dijungal | thewizzle: call waiting would not cut it, when there's 20 persons in a queue |
| 16:50.17 | thewiizle | no i mean, CallWaiting disabled |
| 16:50.22 | thewiizle | ie. multiplecalls=no |
| 16:50.30 | jfg | i'd like to work on adding MGCP client feature to asterisk, i read asterisk code practices, and a part of MGCP RFC, can someone help me now to really start? |
| 16:50.37 | dijungal | on the sip profiles? |
| 16:50.38 | tru_`z24 | so basically, if i have 2 fxo's and an incoming call is transferred out, i'm going to be busy until that transfer is done |
| 16:50.38 | thewiizle | dont quote me on that, just follow my trail of thought :) |
| 16:50.54 | thewiizle | i would have imagined an option to disable secondary call |
| 16:51.01 | [TK]D-Fender | jfg: I'd suggest looking to the asterisk-dev mailing lists and bug tracker to start this. |
| 16:51.02 | thewiizle | many other pbx boast this feature |
| 16:51.20 | *** join/#asterisk Rtx (n=xtr@bas1-quebec09-1178041373.dsl.bell.ca) |
| 16:51.26 | [TK]D-Fender | tru_`z24: until that call terminates, yes |
| 16:51.44 | dijungal | lol |
| 16:51.44 | Rtx | someone know if Openvox card are good ? |
| 16:51.55 | [TK]D-Fender | tru_`z24: but, as we've mentioned it may be doable to to a flash-transfer if your line supports it. |
| 16:51.59 | *** join/#asterisk riddlebox (n=JamesMid@75-128-170-26.static.stls.mo.charter.com) |
| 16:52.28 | alrs | Rtx: better than Digium, I hear. |
| 16:52.29 | tru_`z24 | [TK]D-Fender: where can i read more details about what i need to do? So that I know my options? is it in the asterisk manual? |
| 16:52.32 | coppice | Rtx: of course (feels he should support local businesses :-) ) |
| 16:52.35 | alrs | Rtx: which isn't a major accomplishment |
| 16:52.35 | jfg | [TK]D-Fender: ok, but i hate using mailing lists.. is it the only way ? |
| 16:52.44 | tru_`z24 | [TK]D-Fender: like documentation on flash transfers |
| 16:52.56 | tru_`z24 | Or are these undocumented api's |
| 16:53.05 | riddlebox | if I buy a TDM card with 1 FX0 and 2 FXS, how much of a power supply would I need, I also have a tv tuner card in the pc? |
| 16:53.11 | Rtx | ok thks for your info |
| 16:53.12 | Rtx | :) |
| 16:53.18 | Rtx | i will buy one then heh |
| 16:53.40 | Rtx | i have nothing about td400 exept price :\ |
| 16:53.43 | dijungal | i might have to call digium |
| 16:53.46 | Rtx | im low budget |
| 16:53.51 | [TK]D-Fender | tru_`z24: there is no "manual". Just ask your telco what kind of analog "transfer" capabilities they offer and how they are accessed. If you can describe it like "I get a call. I then [flash] and dial another number and then hangup and the call gets thrown ofer there and my line is freed up" and they say "yes", then there you have it |
| 16:54.19 | tru_`z24 | right, but is there a specifc call in asterisk to request this? |
| 16:54.27 | tru_`z24 | I mean, what if i wanted to choose what happens after i flash |
| 16:54.32 | [TK]D-Fender | riddlebox: I advise against PCI FXS unless you're planning to put a modem device on it. And whats the TV card for? |
| 16:54.52 | dijungal | anyone know's the digium support number? |
| 16:55.06 | dijungal | lol |
| 16:55.07 | eldon | dijungal, 1877LINUXME |
| 16:55.08 | deeperror | I seem to be having some odd issues when flashing as well any clues? http://pastebin.ca/695949 |
| 16:55.11 | [TK]D-Fender | tru_`z24: yes, you would call "Flash" from your dialplan and then "SendDTMF(1234567)" etc to do the rest. |
| 16:55.15 | *** join/#asterisk fakhir (n=Fakhir@unaffiliated/fakhir) |
| 16:55.22 | *** part/#asterisk jfg (n=jfg@dyn-83-152-249-84.ppp.tiscali.fr) |
| 16:55.37 | tru_`z24 | [TK]D-Fender: roger. Now why do you recommend against the tdm400p with fxs ? |
| 16:55.47 | dijungal | thnks |
| 16:55.50 | tru_`z24 | That is exactly what i was going to get |
| 16:55.56 | tru_`z24 | but with 2 fxs and 2 fxo |
| 16:56.02 | tru_`z24 | so that i can transfer the call to a rep |
| 16:56.29 | mistermocha | ManxPower: wasn't disk usage |
| 16:56.41 | [TK]D-Fender | tru_`z24: Zaptel FXS for normal analog phone usage is ASS. SIP ATA's are cheaper, more flexible, and rpesent a lower load to your system while maintaining more ports available for FXO expansion without adding cards. |
| 16:56.44 | mistermocha | for some odd reason, the last reload didn't load cdr_csv.so properly |
| 16:57.05 | mistermocha | I just did a reload again and it appears to be working |
| 16:57.14 | riddlebox | [TK]D-Fender: so is it better to have the PCI FX0 card from Digium, and use ATA devices for any analog phones? |
| 16:57.15 | mistermocha | thanks for the tidbits |
| 16:57.27 | *** join/#asterisk `Sean (i=Un1x@CPE000c256d416d-CM000a73a1559d.cpe.net.cable.rogers.com) |
| 16:57.27 | ManxPower | tru_`z24: also the telco will frequently limit the number of calls you can transfer. |
| 16:57.52 | [TK]D-Fender | tru_`z24: And you need to stop thinking of it as "transferring a call to a rep". thats just * CALLING the rep ( on the FXS port), and BRIDGING the call. The word "transfer" is a terrible misnomer. |
| 16:57.53 | ManxPower | riddlebox: no, it is better to use a PRI and IP phones. |
| 16:58.00 | [TK]D-Fender | riddlebox: yes |
| 16:58.14 | thewiizle | bah PRI |
| 16:58.37 | thewiizle | SIP > *RI |
| 16:58.41 | tru_`z24 | [TK]D-Fender: Roger that, so SIP ATA requires what board and what services from the telco? |
| 16:58.49 | eldon | SIP sucks |
| 16:59.02 | tru_`z24 | SIP is just a protocol right? |
| 16:59.02 | [TK]D-Fender | riddlebox: PCI FXS = ass. Polycom > All other SIP phones & ATA's. PRI > FXO |
| 16:59.11 | [TK]D-Fender | tru_`z24: None. |
| 16:59.24 | mistermocha | SIP sucks if your internets and networks don't like sip |
| 16:59.31 | [TK]D-Fender | tru_`z24: Correct, a VoIP protocal for call setup and RTP negotiation. |
| 16:59.51 | tru_`z24 | Right, so at what point does the phone call hit the telco's pbx if you're using sip |
| 16:59.56 | riddlebox | ManxPower: well in my home I really dont see the need for PRI, and I am using SIP phones, I just have one cordless phone that the girlfriend wants to use, I want to get a sip cordless phone but they are expensive |
| 16:59.56 | tru_`z24 | sip ata that is |
| 17:00.52 | outtolunc | in my home i have a 4 port t1 card and 3 channel banks <G> |
| 17:00.58 | Uatec | hi |
| 17:01.11 | tru_`z24 | you have a t1 running to your house? |
| 17:01.16 | Uatec | i've downloaded misdn using conary |
| 17:01.37 | BockBilbo | is there any iax client appart from kiax for gnu/linux? |
| 17:02.03 | outtolunc | i used too.. but i just use the 4port t1 card to talk to the 3 channel banks for dev |
| 17:04.23 | Uatec | but it's installed into asterisk |
| 17:04.28 | alrs | BockBilbo: iaxcomm |
| 17:04.31 | Uatec | although it is starting when i i start the OS |
| 17:04.48 | *** join/#asterisk r0d3nt (n=astrutt@foster.stonedcoder.org) |
| 17:06.27 | BockBilbo | thanks alrs |
| 17:07.59 | *** part/#asterisk riddlebox (n=JamesMid@75-128-170-26.static.stls.mo.charter.com) |
| 17:09.05 | Uatec | ARGH |
| 17:09.15 | Uatec | i don't understand why asterisk isn't picking up mISDN |
| 17:09.16 | BockBilbo | great |
| 17:09.20 | BockBilbo | iax works too |
| 17:09.36 | BockBilbo | now.. i dont see why ekiga wont login |
| 17:10.56 | *** join/#asterisk NightMonkey (n=NightMon@pdpc/supporter/sustaining/NightMonkey) |
| 17:11.32 | BockBilbo | well |
| 17:11.33 | BockBilbo | thanks |
| 17:11.45 | BockBilbo | i've got to restart my computer now |
| 17:12.13 | *** part/#asterisk BockBilbo (n=BockBilb@eu85-84-62-227.clientes.euskaltel.es) |
| 17:13.24 | [TK]D-Fender | tru_`z24: ATA talks SIP to *, * lets you use your FXO's |
| 17:13.39 | [TK]D-Fender | tru_`z24: * sits BETWEEN calls in/out your various channels. |
| 17:18.31 | henkoegema | Why is this NOT working ?? http://pastebin.com/d7ec72383 |
| 17:19.08 | [TK]D-Fender | henkoegema: Pastebin your failed CALL so we can see whats actually happening. |
| 17:19.44 | henkoegema | ok |
| 17:22.48 | *** part/#asterisk jarod14 (n=jarod14@mail.viatelecom.com) |
| 17:23.13 | henkoegema | http://pastebin.com/d608b999 |
| 17:24.27 | *** join/#asterisk los415 (i=los415@LAX-DHCP-64-201-109-227.race.com) |
| 17:26.04 | *** join/#asterisk generalhan (n=asd@ip67-90-64-2.z64-90-67.customer.algx.net) |
| 17:28.46 | ZaVoid | i think asterisk 1.4.9 is the most amazing asterisk build yet |
| 17:28.54 | ZaVoid | it has not crashed in 4 days now on any of my asterisk boxes |
| 17:29.03 | ZaVoid | not one single core dump |
| 17:29.29 | ZaVoid | 1.4.11 crashed 10-20 times a day |
| 17:31.24 | ManxPower | henkoegema: what is the value of ${result} ? |
| 17:31.34 | ManxPower | Most enum issues are incorrectly set up DNS |
| 17:31.42 | ManxPower | or incorrect enum.conf |
| 17:32.14 | ManxPower | henkoegema: unless you are using 1.0.x or 1.2 with priorityjumping=yes, it will NEVER jump to n+101 |
| 17:32.26 | *** join/#asterisk hermuli (n=Eladamri@xdsl-83-145-207-63.nebulazone.fi) |
| 17:32.32 | *** join/#asterisk caio1982 (i=caio1982@CAcert-br/caio1982) |
| 17:32.38 | ManxPower | I'll bet you are using an example from the always outdated and frequently outright wrong voip-info Wiki, aren't you. |
| 17:32.57 | *** join/#asterisk melbert (n=IceChat7@66.179.79.70) |
| 17:33.14 | *** part/#asterisk melbert (n=IceChat7@66.179.79.70) |
| 17:34.24 | *** part/#asterisk jtexter3 (n=jtexter3@12.159.220.114) |
| 17:34.25 | henkoegema | the value of ${result} = EMPTY |
| 17:34.57 | henkoegema | ManxPower: I use 1.4.11 |
| 17:35.13 | *** join/#asterisk flujan (n=flujan@200-160-115-020.static.spo.ctbc.com.br) |
| 17:36.01 | *** join/#asterisk r0d3nt (i=nobody@foster.stonedcoder.org) |
| 17:36.26 | [TK]D-Fender | henkoegema: the format for yoru function call does not match anything in the DOC samples |
| 17:36.29 | henkoegema | ManxPower: Most enum issues are incorrectly set up DNS ?? What else do I need to set up ? |
| 17:36.50 | ManxPower | henkoegema: 1.4 never jumps to priority n+101 |
| 17:36.55 | [TK]D-Fender | henkoegema: go read the DOCS on it |
| 17:37.00 | ManxPower | henkoegema: ENUM uses DNS to route calls |
| 17:37.01 | henkoegema | <PROTECTED> |
| 17:37.14 | ManxPower | ENUM is just a way to use DNS to lookup routes for calls. |
| 17:37.29 | ManxPower | you must have very specific and correct DNS entries for ENUM to work. |
| 17:37.50 | ManxPower | henkoegema: I don't care if you have a monkey that can fly the space shuttle, don't use 103 104, etc priorities in 1.4.x |
| 17:37.55 | henkoegema | i Know. I have been using ENUM successfully in version 1.2.x but now it doesnt work |
| 17:38.55 | henkoegema | the problem is that ENUMLOOKUP doesnt return a correct value |
| 17:39.52 | *** join/#asterisk PC_Clone (n=pc_clone@c-76-16-79-154.hsd1.il.comcast.net) |
| 17:39.59 | henkoegema | sorry I have to leave now, but will return another time. Greetings |
| 17:40.29 | PC_Clone | hello all...does anyone know where I can find a list of milliwatt test lines? |
| 17:40.40 | Strom_M | PC_Clone: which telephone company serves your local loop? |
| 17:40.48 | ManxPower | exten=>3008,3,NoOp(${result}) should NEVER EVER print Noop(result) on the cli. |
| 17:40.53 | ManxPower | It should print the VALUE of result |
| 17:41.04 | PC_Clone | SBC/ATT/Ameritech/IL Bell/Whatever they are next week |
| 17:41.12 | Strom_M | PC_Clone: i'll PM you a URL |
| 17:41.23 | PC_Clone | Strom_M: thanks |
| 17:41.26 | outtolunc | Milliwatt test tones are often found at NPA-NXX-0010. |
| 17:41.31 | ManxPower | next week they will be Walmart-Utami (I think that is how you spell it) aka The Company |
| 17:41.33 | Strom_M | outtolunc: hahahaha no. |
| 17:41.36 | PC_Clone | outtolunc: tried that |
| 17:41.41 | Strom_M | outtolunc: depends on the telephone company. |
| 17:41.50 | *** join/#asterisk dijungal (n=kdaniel@64.86.52.254) |
| 17:42.04 | PC_Clone | outtolunc: actually it listed the number as changed....and it was a friend of my brothers |
| 17:42.07 | PC_Clone | weird |
| 17:42.24 | *** part/#asterisk deeperror (n=deeperro@69-215-202-195.ded.ameritech.net) |
| 17:43.09 | Strom_M | outtolunc: for example, on my switch in los angeles, it's NXX-0002 |
| 17:43.32 | outtolunc | yeah i'm sure they are all over the place now |
| 17:43.42 | outtolunc | just hoping his was still that |
| 17:43.59 | ManxPower | you can't figure out the number most of the time. You need to find a list from the telco. |
| 17:44.12 | ManxPower | or a list that was taken from a telco list |
| 17:44.23 | ManxPower | I used (might still) have the list for BellSouth areas |
| 17:47.29 | *** join/#asterisk jer (n=jtregunn@unaffiliated/jer) |
| 17:48.58 | Strom_M | spliceorama! |
| 17:49.12 | Yourname` | Myth: When a dialer is running more than 300 calls, etc. Can I look at top to see if it's having any issues doing so, so I can turn up or down the channels? |
| 17:49.27 | ManxPower | Strom_M: actually taking 2 wifi APs, a satellite dish and a long range vhf/uhf antenna down |
| 17:50.19 | *** join/#asterisk denon (n=denon@208.122.43.201) |
| 17:50.19 | *** mode/#asterisk [+o denon] by ChanServ |
| 17:50.19 | Strom_M | ManxPower: ah |
| 17:50.47 | *** join/#asterisk yannj_fr (n=yannj_fr@vpn.intelunix.fr) |
| 17:51.13 | TJNII | The zaptel timing dumyy driver is ztdummy, correct? |
| 17:51.22 | TJNII | s/dumyy/dummy/ |
| 17:51.31 | Strom_M | correct |
| 17:56.23 | *** join/#asterisk saftsack (n=saftsack@pD9E07E2D.dip.t-dialin.net) |
| 18:00.27 | *** join/#asterisk mrbond82 (n=bondo@d221-91-164.commercial.cgocable.net) |
| 18:03.40 | *** join/#asterisk FlyboySR22 (n=rsears@206.251.251.2) |
| 18:04.41 | mrbond82 | What does it mean if I can call out from server A to server B (both un-natted, on the internet directly) I can call extension 1 at server B from extension 2 on server A, he picks up, I can hear him, he cannot hear me. If I dial extension 2 at server B (which is echo only) I can speak and it echos back to me (both ways)... what is going on ? |
| 18:07.39 | dlynes_laptop | Is there presence detection for zaptel channels? |
| 18:07.48 | [TK]D-Fender | dlynes_laptop: yup |
| 18:07.52 | [TK]D-Fender | dlynes_laptop: Always was |
| 18:08.26 | [TK]D-Fender | dlynes_laptop: if you mean being able to detect presence FOR that chan, not OVER it. |
| 18:08.27 | dlynes_laptop | [TK]D-Fender: ah, ok...can't seem to find anything about it on voip-info.org |
| 18:08.42 | dlynes_laptop | [TK]D-Fender: as in being able to blf a zaptel channel |
| 18:08.54 | [TK]D-Fender | dlynes_laptop: it just says "tech", so exten => 1,hint,Zap/1 |
| 18:09.02 | mrbond82 | Is there a limit to how long IRC message can be in this channel, sometimes they dont' get through for me? |
| 18:09.16 | [TK]D-Fender | mrbond82: Possibly |
| 18:09.32 | dlynes_laptop | [TK]D-Fender: and i'm guessing that'll work for fxo as well as fxs, right? |
| 18:09.58 | [TK]D-Fender | mrbond82: your phones must not be allowed to reinvite |
| 18:10.03 | [TK]D-Fender | dlynes_laptop: yup |
| 18:10.11 | dlynes_laptop | [TK]D-Fender: thanks muchly |
| 18:10.34 | *** join/#asterisk sysreq (n=sysreq@unaffiliated/sysreq) |
| 18:10.35 | mrbond82 | [TK]D-Fender put canreinvite=no in each of the phone configs in sip.conf ? |
| 18:10.41 | [TK]D-Fender | yup |
| 18:11.24 | mrbond82 | sweet. |
| 18:11.52 | [TK]D-Fender | mrbond82: is that a statement of success? |
| 18:12.00 | mrbond82 | yes |
| 18:12.01 | *** join/#asterisk neverblue2 (n=neverblu@unaffiliated/neverblue) |
| 18:12.02 | mrbond82 | that was easy |
| 18:13.12 | [TK]D-Fender | mrbond82: Good to hear. What happens is your first call goes from PHONE>A>B, and then tries to put PHONE>B direct. then B brings PHONE2 into the picture and tries to get PHONE1 & PHONE2 going directly and neither of them can handle all this NAT-ing |
| 18:13.27 | *** join/#asterisk t3rror (n=t3rror@gateway.sscgp.com) |
| 18:13.39 | [TK]D-Fender | mrbond82: So we are (and should) force all traffic going through A & B |
| 18:15.49 | *** join/#asterisk BockBilbo (n=BockBilb@eu85-84-62-227.clientes.euskaltel.es) |
| 18:19.51 | dlynes_laptop | [TK]D-Fender: so can you do hints on trunked iax channels, too? i.e. so that it can show each call, individually, using blf? |
| 18:23.25 | Shido6 | damnit the vending machine should take paypal |
| 18:28.13 | *** join/#asterisk anonymouz666 (n=anonymou@189.25.190.36) |
| 18:28.37 | *** join/#asterisk rob0 (i=rob0@sorry.no-ip-here.net) |
| 18:28.58 | rob0 | hiya * |
| 18:29.57 | juuva | Shido6: just call to it |
| 18:32.02 | rob0 | I'm a small-timer, just have * at 2 sites for personal/family use. I have some free dial-in numbers (stanaphone and ipkall). What's the easiest way to move those numbers from one place to the other? Just maintain multiple sip.conf and extensions.conf files at each place? |
| 18:35.25 | BockBilbo | ive got one little question... my ATA is connected to my server using eth1, and im trying to connect to a phone on the ata from a machine on a network to wich the server is connected using eth0. The call signaling process works correctly, and it's stablished, though there is no sound neither in the laptop or in the phone... |
| 18:35.59 | BockBilbo | i guess that it's because neither the ata or my laptop know the route to each other |
| 18:36.53 | *** join/#asterisk drwelby (n=mpfister@mail.enplan.com) |
| 18:37.02 | BockBilbo | ive tried to configure the ip routers on both the server and the laptop, but it's still not working... |
| 18:37.04 | BockBilbo | any idea? |
| 18:38.47 | dlynes_laptop | BockBilbo: and probably because one or the other end, or both ends are behind firewalls |
| 18:39.36 | BockBilbo | dlynes_laptop ive configured iptables on the server to accept rtp connections |
| 18:39.41 | dlynes_laptop | BockBilbo: try the following link: http://www.voip-info.org/wiki-Asterisk+SIP+NAT+solutions |
| 18:40.01 | dlynes_laptop | BockBilbo: your laptop probably has a firewall |
| 18:40.22 | dlynes_laptop | BockBilbo: it's probably running vista or xp with windows firewall |
| 18:40.32 | BockBilbo | i dont use windows |
| 18:40.32 | BockBilbo | xD |
| 18:40.37 | dlynes_laptop | BockBilbo: or norton 360, or norton internet security, or zone alarm, or whatever else |
| 18:40.40 | BockBilbo | xD |
| 18:40.41 | BockBilbo | haha |
| 18:40.42 | dlynes_laptop | BockBilbo: xd? |
| 18:40.43 | BockBilbo | i use ubuntu |
| 18:40.55 | dlynes_laptop | BockBilbo: ok, maybe ubuntu's got a firewall then |
| 18:41.01 | BockBilbo | it does, iptables |
| 18:41.09 | dlynes_laptop | BockBilbo: and the server does, too? |
| 18:41.12 | rob0 | IP forwarding enabled? |
| 18:41.27 | BockBilbo | rob0, you mean in the server? |
| 18:41.30 | rob0 | yes |
| 18:41.47 | BockBilbo | how do you do IP forwarding? using the route comand? |
| 18:42.07 | ManxPower | BockBilbo: that is really beyond the scope of this channel |
| 18:42.17 | BockBilbo | ok ManxPower |
| 18:42.18 | BockBilbo | sorry |
| 18:42.19 | BockBilbo | :( |
| 18:42.25 | dlynes_laptop | BockBilbo: echo 1 > /proc/sys/net/ipv4/ip_forward |
| 18:42.31 | dlynes_laptop | BockBilbo: but that's assuming you're using ipv4 |
| 18:42.38 | BockBilbo | yep |
| 18:42.39 | BockBilbo | thanks |
| 18:42.42 | rob0 | Your distro probably provides a way to make it permanent, but no, it's merely "echo 1 > /proc/sys/net/ipv4/ip_forward" |
| 18:42.49 | rob0 | dang lag |
| 18:44.06 | BockBilbo | great |
| 18:44.08 | BockBilbo | it works |
| 18:44.09 | BockBilbo | :) |
| 18:44.30 | BockBilbo | thanks a lot rob0 |
| 18:44.30 | dlynes_laptop | Is it possible to do presence detection on each individual call within an iax2 line that has say calllimit=8? |
| 18:46.41 | [TK]D-Fender | dlynes_laptop: No |
| 18:47.04 | *** join/#asterisk jer (n=jtregunn@unaffiliated/jer) |
| 18:48.09 | *** join/#asterisk MaXpEr (n=maxper@host32-218-dynamic.17-87-r.retail.telecomitalia.it) |
| 18:48.13 | MaXpEr | Hi! |
| 18:48.43 | MaXpEr | I have some timeout problems and would like to understand better the ast_safe_sleep function... |
| 18:48.57 | MaXpEr | What does this function exactly do? |
| 18:50.10 | *** join/#asterisk rnovotny22 (n=ro085181@h460dee33.area2.spcsdns.net) |
| 18:50.19 | ManxPower | MaXpEr: try #asterisk-dev |
| 18:50.27 | MaXpEr | grazie :) |
| 18:50.27 | ManxPower | there is also the doxygen docs. build them |
| 18:50.41 | Qwell | it sleeps |
| 18:50.42 | Qwell | safely |
| 18:50.52 | MaXpEr | I read those comments but they are not that clear |
| 18:51.03 | MaXpEr | too brief... |
| 18:51.20 | russellb | it sleeps, but services the channel while sleeping |
| 18:51.24 | russellb | reading frames that come in |
| 18:51.33 | russellb | and throwing them away |
| 18:51.39 | MaXpEr | May I explain the problem? |
| 18:51.54 | MaXpEr | chan_mgcp seems not to sleep properly... |
| 18:52.10 | MaXpEr | That should sleep (by default) 16 seconds, but it doesn't... |
| 18:52.30 | MaXpEr | It sleeps, on my PC, just 5 seconds... |
| 18:53.32 | waKKu | folks.. i'm having a problem with SIP + Linksys PAP2na ... when I call and get some moh message, if I hangup the phone it rings back .. (I already disable Call Return and Call Back services on ATA) |
| 18:56.39 | *** join/#asterisk Blackthorn (i=blacktho@76.77.160.10) |
| 18:58.37 | *** join/#asterisk fadumpt (n=john@adsl-070-154-035-081.sip.gsp.bellsouth.net) |
| 18:58.51 | fadumpt | hi |
| 18:59.46 | fadumpt | 3 of my extensions hang up before/instead of going to voicemail...I created a new extension and it worked fine on it, and I tried an existing extension that also worked fine |
| 18:59.59 | fadumpt | I tried recreating one of the non working extensions and it does the same thing |
| 19:00.49 | *** join/#asterisk infernix (i=nix@unaffiliated/infernix) |
| 19:02.14 | *** join/#asterisk Winkie (i=sd@87-194-8-125.bethere.co.uk) |
| 19:02.23 | lirakis | fadumpt: could be your dialplan.. could be that they ext's dont have vm properly setup |
| 19:03.03 | infernix | i just read Asterisk with IPv6: Seamless and Ubiquitous VoIP and I was wondering what the current state for ipv6 is. With current code, is it possible to have an asterisk setup with an ipv4 SIP trunk to $world, and ipv6 SIP to phones? |
| 19:03.41 | infernix | asterisk would have to do the rerouting to ipv6, obviously |
| 19:03.54 | fadumpt | (I realized there was a freepbx channel after asking the question) freepbx shows the vm to be the same as the working ext's...I guess my next step is to check the config file and see if there is something odd there... |
| 19:04.12 | *** part/#asterisk MaXpEr (n=maxper@host32-218-dynamic.17-87-r.retail.telecomitalia.it) |
| 19:04.25 | lirakis | fadumpt:smart to take it to freepbx |
| 19:05.00 | dlynes_laptop | [TK]D-Fender: well, if you define multiple iax channels going to the same destination and then you trunk them, will all those individual lines share bandwidth, or will they consume the same amount of bandwidth? |
| 19:05.19 | dlynes_laptop | [TK]D-Fender: i.e. all with call-limit=1? |
| 19:11.18 | *** join/#asterisk metfan2007 (n=metfan20@201.104.20.25) |
| 19:12.19 | metfan2007 | Hi all, how do I define the audio record format to use when the Voicemail record the greetings prompts? |
| 19:12.38 | *** part/#asterisk wltjr (n=wltjr@gentoo/developer/wltjr) |
| 19:12.50 | hmmhesays | i'm a roller, I'm a rider |
| 19:13.36 | *** join/#asterisk MaXpEr (n=maxper@host32-218-dynamic.17-87-r.retail.telecomitalia.it) |
| 19:13.43 | MaXpEr | One more thing |
| 19:13.44 | MaXpEr | ... |
| 19:13.53 | metfan2007 | any one??? |
| 19:13.57 | MaXpEr | How can I show the correct callerid of a transfer? |
| 19:16.11 | [TK]D-Fender | dlynes_laptop: its trunked by peer entry so no, you can't get a granular view of them. |
| 19:16.36 | [TK]D-Fender | metfan2007: Looks at the voicemail.conf sample config |
| 19:17.15 | dlynes_laptop | [TK]D-Fender: hrm...so i guess if i want blf'ing of iax2 channels, i have to untrunk, and define multiple peers |
| 19:17.27 | dlynes_laptop | [TK]D-Fender: but it won't blf the users then either, right? only the peers? |
| 19:17.48 | [TK]D-Fender | dlynes_laptop: same thing. its counts the CHANNEL as being in-use |
| 19:18.03 | dlynes_laptop | [TK]D-Fender: ok, thanks |
| 19:19.28 | *** join/#asterisk saftsack (n=saftsack@pD9E07E2D.dip.t-dialin.net) |
| 19:20.33 | *** part/#asterisk MaXpEr (n=maxper@host32-218-dynamic.17-87-r.retail.telecomitalia.it) |
| 19:21.28 | *** join/#asterisk kwame (n=kwame@208.66.168.243) |
| 19:22.44 | *** join/#asterisk McDouglas (n=mcd@wxdna0kdgg.adsl.datanet.hu) |
| 19:23.13 | *** join/#asterisk JimVanM (n=jimvanm@bas1-toronto63-1096579178.dsl.bell.ca) |
| 19:23.51 | McDouglas | if i have 20 extensions and 4 outgoing lines, how can i restrict some of the users to only be able to use line 1 and 2 ? (but still be able to dial each other internaly) |
| 19:25.51 | [TK]D-Fender | McDouglas: contexts <- |
| 19:26.02 | McDouglas | well, i'm awarwe of them |
| 19:26.09 | McDouglas | but not exactly sure how |
| 19:26.17 | *** join/#asterisk rmayorga (n=rmayorga@unaffiliated/rmayorga) |
| 19:26.24 | McDouglas | i have 3 right now: incoming, internal, outgoing |
| 19:26.33 | McDouglas | internal includes outgoing |
| 19:26.36 | [TK]D-Fender | McDouglas: Set those phones into a context where the oubound dials only have access to lines 1 & 2 |
| 19:26.37 | McDouglas | so everyone can dial out |
| 19:26.57 | [TK]D-Fender | McDouglas: Clearly you're going to have to make *2* sets of outbound contexts |
| 19:27.13 | McDouglas | you mean internal? |
| 19:27.24 | McDouglas | well, or both? |
| 19:28.54 | McDouglas | so, i should make an internal-full and internal-restricted contexts and an outgoing-4 outgoing-2 and link them together? |
| 19:28.55 | [TK]D-Fender | McDouglas: internal should have your PHONE extens in them. outbound1 & outbound 2 should have those 2 different groups worth of extens. phones1 & phones2 should include the appropriate outbound + internal. Your phones should use either phones1 or phones2 as appropriate |
| 19:29.28 | [TK]D-Fender | McDouglas: With differnt names you have jsut said the same thing, yes |
| 19:29.37 | McDouglas | hmm, i think i get it then |
| 19:29.43 | *** join/#asterisk heartones (n=Omar@196.218.146.36) |
| 19:30.13 | McDouglas | oh, but with your solution (having a global internal) i can also solve the problem of calling each other |
| 19:30.17 | McDouglas | nice, thanks |
| 19:31.10 | [TK]D-Fender | McDouglas: no problem. |
| 19:31.32 | [TK]D-Fender | McDouglas: I usually have to beat this concept over your typical visitors head for at least 20-30 minutes before they "get it" :) |
| 19:31.42 | McDouglas | :P |
| 19:31.48 | heartones | what's the difference between installing the modules from CLI and installing from the package manager in trixbox GUI |
| 19:31.56 | McDouglas | i have been experimenting with the dialplans for a while now :P |
| 19:32.14 | [TK]D-Fender | heartones: you are in the wrong channel to ask that.... |
| 19:32.15 | heartones | are they fairly identical |
| 19:32.16 | heartones | ' |
| 19:32.25 | heartones | sorry |
| 19:32.38 | heartones | I didn't realise I am on asterisk lol |
| 19:33.01 | heartones | I love CLI |
| 19:33.29 | McDouglas | one more thing: i have read that you can use callgroup and pickgroup to set up who can pick up other calls using the keys *8 |
| 19:33.42 | McDouglas | but what if i want to pick up with "40" ? |
| 19:33.44 | [TK]D-Fender | McDouglas: yeah, something like that. |
| 19:33.47 | McDouglas | (our current system works like this) |
| 19:34.01 | [TK]D-Fender | McDouglas: I think there is a dialplan app that lets yuo pickup as well. |
| 19:34.16 | McDouglas | hmm, have to lok into that |
| 19:34.18 | [TK]D-Fender | McDouglas: which you could theoretically use and exten you wanted |
| 19:34.19 | McDouglas | +o |
| 19:34.26 | [TK]D-Fender | any* |
| 19:35.03 | *** join/#asterisk neuwald (n=neuwald@200.199.198.61) |
| 19:36.48 | neuwald | Hi all. I have some X-Lite SIP clients connected to one asterisk server. Sometimes, when I try to call some user, I got a busy signal, but the user isn't busy. Does anybody knows why it's happening? Asterisk 1.2.24 |
| 19:37.55 | [TK]D-Fender | neuwald: Without seeing the CLI output of a failed call with SIP debug enabled.... NO. |
| 19:38.03 | [TK]D-Fender | neuwald: PASTEBIN is your friend |
| 19:38.05 | [TK]D-Fender | ~pb |
| 19:38.05 | jbot | A Pastebin is a place to paste your stuff without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste |
| 19:38.06 | [TK]D-Fender | ^^^^^^^^^^^^^^^^^ |
| 19:39.06 | neuwald | [TK]D-Fender http://www.pastebin.ca/696209 - A call from 8218 to 8227 |
| 19:39.44 | [TK]D-Fender | Got SIP response 486 "Busy Here" back from 192.168.200.32 <- it IS busy. |
| 19:40.24 | [TK]D-Fender | neuwald: that is a SIP response, and not an assumption made by *, so its real. If they aren't on the phone, then they either have DND or selective rejection on. |
| 19:40.29 | neuwald | [TK]D-Fender it isn't busy. I'm sure |
| 19:40.35 | [TK]D-Fender | DND <----- |
| 19:41.21 | neuwald | look here: http://www.pastebin.ca/696212 |
| 19:42.50 | DrAk0 | i can jump from a dialplan to another with Dial(SIP/XX@new_context) ? |
| 19:42.55 | DrAk0 | or do i need to use DISA ? |
| 19:44.04 | [TK]D-Fender | neuwald: Means nothing. I told you what to look for. |
| 19:44.45 | keulin | hi everybody |
| 19:44.54 | [TK]D-Fender | DrAk0: ummm... no. You want to just continue processing the call from another exten in another context? |
| 19:45.37 | DrAk0 | [TK]D-Fender, yes, i have the context agents for the queue but i want the agent to be able to transfer calls to the users on another context |
| 19:45.42 | keulin | I'm testing a trixbox configuration. |
| 19:45.54 | [TK]D-Fender | keulin: Then you are in the WRONG place. |
| 19:46.11 | DrAk0 | keulin, #freepbx |
| 19:46.17 | [TK]D-Fender | DrAk0: Then you need your agents context to include that other context. |
| 19:46.33 | keulin | thanks [TK]D-Fender and DrAk0 |
| 19:46.39 | McDouglas | [TK]D-Fender: problem with the dialplan cmd pickup is that you have to specify the extension you want to pick up. i want to pick up any of the ringing ones |
| 19:47.07 | [TK]D-Fender | McDouglas: I thought there was another app than just that one based on groupings. I may be mistaken, but take a quick look |
| 19:47.17 | DrAk0 | [TK]D-Fender, but then its a mess because the agents are on this context with a macro with voicemail, then when they are not avaliable the queue jump to the voicemail |
| 19:47.51 | [TK]D-Fender | DrAk0: You basically don't want it falling to agent's VM, right? |
| 19:47.58 | DrAk0 | yes |
| 19:48.14 | [TK]D-Fender | DrAk0: then stop telling your queue/agent login to USE extens that have it! |
| 19:48.39 | neuwald | [TK]D-Fender so... where to start? the SIP client really isn't busy. Ok, I still got a busy signal... |
| 19:48.48 | DrAk0 | [TK]D-Fender, http://pastie.caboo.se/96926 |
| 19:49.03 | [TK]D-Fender | neuwald: that device is SAYING its busy, so go look at the DEVICE. |
| 19:49.16 | *** join/#asterisk zotz (n=zotz@24.244.163.157) |
| 19:49.17 | DrAk0 | [TK]D-Fender, jeez... they are tooo dump, im trying to make stuff easier for them hehe |
| 19:49.27 | DrAk0 | dumb* |
| 19:50.05 | [TK]D-Fender | DrAk0: that doesn't show me the source of your screw-up |
| 19:51.26 | DrAk0 | [TK]D-Fender, as you said, I have 2 options, tell the agents to use another extension without VM or try to make things easier for them with for example jump to another context in a transparent way for them |
| 19:51.39 | neuwald | [TK]D-Fender there is someway that asterisk can check the status of the device every time interval ? |
| 19:51.50 | *** join/#asterisk michaeljoser (n=aldfj@ADSL-TPLUS-90-146.telecomplus.net) |
| 19:52.35 | [TK]D-Fender | neuwald: Not a "special way", but you can make a script to do this, yes. |
| 19:52.38 | michaeljoser | hi wanted to know how i can have 1 outbound call via my T100p and 3 sip phones talk in the same conference... is that possible? |
| 19:53.25 | *** join/#asterisk Daviey (n=dave@ubuntu/member/daviey) |
| 19:53.43 | neuwald | [TK]D-Fender more one thing. If i have no call-limit on my sip.conf, I got the busy response. If I have call-limit=2, 100, 200, or other value, I don't got the busy response |
| 19:53.47 | [TK]D-Fender | michaeljoser: Yes. "show application meetme" |
| 19:54.26 | [TK]D-Fender | neuwald: do another call with sip debug enabled |
| 19:55.30 | [TK]D-Fender | DrAk0: When for calling agents, it should use the [agentes] context, and your phones should include the [local-ext] context, and NOT [agentes] |
| 19:55.30 | michaeljoser | <[TK]D-Fender>: i got meetme working so that i can have 3 sip phones in the conference with ext 6000 but how do i make a call throught pstn go to the conference |
| 19:55.49 | [TK]D-Fender | michaeljoser: You want * to call out and dump them into the conference? |
| 19:57.18 | neuwald | [TK]D-Fender now I can talk to the SIP client... no changes on my configuration? |
| 19:57.19 | michaeljoser | <[TK]D-Fender>: i guess i want internal extension 2000 to ring the outside number and then extension 2000 and the outside call join extension 2001 and 2002 in conference 6000.. |
| 19:57.22 | neuwald | no changes on my configuration. |
| 19:57.52 | michaeljoser | i hope that makes sense :/ |
| 19:57.55 | *** join/#asterisk rabelais (n=blank@hpolaris.Stanford.EDU) |
| 19:58.25 | DrAk0 | [TK]D-Fender, again? |
| 19:58.30 | DrAk0 | [TK]D-Fender, didn't get it |
| 19:58.38 | McDouglas | [TK]D-Fender: cant find anything :\ could you look around too, if i may ask? |
| 20:00.39 | DrAk0 | [TK]D-Fender, i think i have a workaround |
| 20:00.58 | DrAk0 | ill try it later, thanks |
| 20:02.42 | [TK]D-Fender | McDouglas: You can ask, but I'mm a little busy now... |
| 20:02.51 | McDouglas | okay ;) |
| 20:02.55 | jcanfield | i'm a noob who need enlightened...any idea why I might not be able to receive calls on an X-lite softphone? I'm getting no route to destination in the logs. |
| 20:04.47 | [TK]D-Fender | jcanfield: pastebin a failed call at verbose 10 & sip debug enabled |
| 20:04.49 | [TK]D-Fender | ~pb |
| 20:04.49 | jbot | A Pastebin is a place to paste your stuff without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste |
| 20:04.50 | [TK]D-Fender | ^^^^^^^^^^^^^^^^^^^^^^^^^ |
| 20:06.17 | *** join/#asterisk guillote_GNU (n=bancaria@host73.201-253-20.telecom.net.ar) |
| 20:08.21 | michaeljoser | <[TK]D-Fender>: any tips for me on where i should look... i will try to get ext 2001 & 2002 in a conference then i use ext 2000 to ring the outside number then i transfer the call to the conference and then i join the conference... can that work this way? :S |
| 20:08.56 | ManxPower | michaeljoser: what you want to do is complex and will require hours for you to learn enough to do it. |
| 20:09.03 | ManxPower | actually more like days |
| 20:09.34 | [TK]D-Fender | michaeljoser: Call this outside number and TRANSFER them to the conference exten. Then join it yourself along with everybody else who wants in. |
| 20:10.01 | michaeljoser | haha that's what i thought...!! |
| 20:10.02 | [TK]D-Fender | michaeljoser: this requires NO effort. |
| 20:10.05 | ManxPower | [TK]D-Fender: I think he wants the Magical Phone Faerie to do it for him. |
| 20:10.12 | michaeljoser | boooo manxpower |
| 20:10.23 | [TK]D-Fender | ManxPower: No, he's happy with my suggestion, which is all I'd advise to someone anyways. |
| 20:14.26 | [TK]D-Fender | ManxPower: Or he could spend a few hours learning about call-files and stuip-level automation :) |
| 20:14.37 | [TK]D-Fender | stupid* |
| 20:14.48 | [TK]D-Fender | ManxPower: Which well... just isn't worth it :) |
| 20:15.06 | jcanfield | What is the default *now root password? Or is root logon over ssh even allowed? |
| 20:15.08 | ManxPower | it isn't |
| 20:15.20 | ManxPower | jcanfield: Asterisk does not have a root password. |
| 20:15.45 | ManxPower | But if you are asking about AsteriskNOW, perhaps a trip to the #asterisknow channel would be in order. |
| 20:16.26 | jcanfield | ManxPower, yes I am...have been there quite a while...it's a ghosttown. |
| 20:16.52 | outtolunc | http://asterisknow.org/files/downloads/quickstart_asterisknow.pdf |
| 20:17.16 | *** join/#asterisk caio1982 (i=caio1982@CAcert-br/caio1982) |
| 20:17.41 | jcanfield | outtolunc, Thanks! |
| 20:18.02 | *** part/#asterisk michaeljoser (n=aldfj@ADSL-TPLUS-90-146.telecomplus.net) |
| 20:18.10 | ManxPower | jcanfield: then I guess not many people use it. I wonder why.... |
| 20:19.25 | jcanfield | ManxPower, seems most of the doc links are too asteriskdocs.org on the site. ...obviously down. |
| 20:19.25 | [TK]D-Fender | ManxPower: Oh.. you forgot the </sarcasm> closing tag! you wouldn't want to be outputting invalid commentary now would you? :p |
| 20:20.08 | [TK]D-Fender | jcanfield: 10 PRINT "I AM GOING CRAZY!!!!" 20 GOTO 10 |
| 20:21.43 | ManxPower | [TK]D-Fender: I was being serious. |
| 20:22.15 | jcanfield | [TK]D-Fender, LOL...I had a feeling you would not like that one! |
| 20:22.24 | outtolunc | probably due to it still (iirc) being in beta, and the lack of card support |
| 20:22.38 | [TK]D-Fender | ManxPower: you mean you feel there should be MORE people sticking with the gui? |
| 20:23.21 | *** part/#asterisk heartones (n=Omar@196.218.146.36) |
| 20:23.37 | *** join/#asterisk rmayorga (n=rmayorga@unaffiliated/rmayorga) |
| 20:23.59 | *** join/#asterisk nighty^ (n=nighty@65.78.203.78) |
| 20:25.20 | *** join/#asterisk steliosk (n=Stelios@ipa226.211.tellas.gr) |
| 20:27.07 | hmmhesays | any func_odbc gurus in here? |
| 20:28.20 | hmmhesays | I'm wondering if you can return more than one field from a query |
| 20:29.18 | hmmhesays | all the examples show a SELECT field1 FROM table |
| 20:30.18 | hmmhesays | i'd like to SELECT field1 field2 FROM table |
| 20:30.25 | [TK]D-Fender | BBIAB |
| 20:38.13 | *** join/#asterisk anonymouz666 (n=anonymou@189.25.132.243) |
| 20:38.44 | McDouglas | i'd like to receive fax from an isdn (misdn) channel and put it through a zap channel to a fax machine |
| 20:38.53 | McDouglas | is there any possible problems with this? |
| 20:43.00 | *** join/#asterisk hspaans (n=hspaans@nullrouted.xs4all.nl) |
| 20:47.43 | hspaans | g'day all, are the more people working on getting asterisk to compile/work on solaris? |
| 20:48.10 | *** join/#asterisk exvito (n=exvito@89-180-155-58.net.novis.pt) |
| 20:49.51 | Peaceful | Can anyone recommend a good headset for a Polycom 550? |
| 20:50.11 | *** part/#asterisk exvito (n=exvito@89-180-155-58.net.novis.pt) |
| 20:51.12 | *** join/#asterisk [TK]D-Fender (n=Joe@64.235.216.2) |
| 20:51.40 | Peaceful | We've only ever used a bunch of Plantronics headsets that we inherited with the phone system several years ago, but apparently they need an amplified cord for the Polycom 550 |
| 20:52.41 | [TK]D-Fender | Peaceful, highly recommended |
| 20:53.26 | [TK]D-Fender | Peaceful, just imagine that you could have bought a lower more appropriate & space saving Polycom model AND the amp for less than the price of your 550 :) |
| 20:58.11 | *** join/#asterisk [hC] (n=hardcore@S01060016b61c8983.vf.shawcable.net) |
| 20:58.55 | *** join/#asterisk jfg (n=jfg@dyn-83-152-249-84.ppp.tiscali.fr) |
| 20:59.53 | jfg | hi |
| 21:02.08 | McDouglas | is there a recommended dialer for outlook/thunderbird ? |
| 21:09.50 | *** join/#asterisk bkruse (i=bkruse@nat/digium/x-79861d6cd699fa1a) |
| 21:09.59 | McDouglas | [TK]D-Fender: any info about receiving a fax on misdn channel and putting it through on a zap channel? does it work? |
| 21:10.47 | [TK]D-Fender | McDouglas, go try. If you're looking to DETECT the fax and if thats what it is, then go out to that zap chan, then use the "fax" Standard Extension. |
| 21:11.10 | McDouglas | nah, i have direct number for the fax |
| 21:11.43 | [TK]D-Fender | McDouglas, then on that specific MSN you can just toss it right over. |
| 21:11.47 | [TK]D-Fender | McDouglas, YMMV |
| 21:18.23 | *** join/#asterisk tomcontr3 (n=tomcontr@171-161-246-201.adsl.terra.cl) |
| 21:19.18 | tomcontr3 | Im having a little problem with my TDM400P card, I have 2 FXO modules, and when I make calls out, I hear little noise, that is anoying |
| 21:19.46 | tomcontr3 | Its like if the line was not clean or something |
| 21:20.34 | *** join/#asterisk syzygyBSD (n=chatzill@66.226.228.204.cpe.speedyquick.net) |
| 21:28.41 | *** join/#asterisk fujin (n=aj@unaffiliated/fujin) |
| 21:33.13 | *** join/#asterisk groogs (n=gregmac@d38-54-164.commercial1.cgocable.net) |
| 21:34.45 | *** join/#asterisk angler (i=angler@pdpc/sponsor/digium/angler) |
| 21:34.45 | *** mode/#asterisk [+o angler] by ChanServ |
| 21:37.02 | *** join/#asterisk james4765 (n=james476@pool-72-84-206-122.rcmdva.east.verizon.net) |
| 21:37.41 | james4765 | anyone know how to get the leds on a Polycom 601 to show other extension's status? |
| 21:38.07 | james4765 | I'm provisioning them via tftp |
| 21:40.27 | [TK]D-Fender | james4765, enable presence in your provisioning, then you'll have a "buddies" soft-key. once you do, add a contact and enable buddy watch in the directory entry. make sure to have a matching SIP hint in your dialplan |
| 21:41.11 | james4765 | cool |
| 21:41.55 | james4765 | so it'll show if someone is on a call, or DND, or whatnot? |
| 21:42.07 | fujin | [TK]D-Fender, know if you can do the hint priority in AEL yet? |
| 21:42.21 | [TK]D-Fender | james4765, only (in-use) while ringing or actually on a call |
| 21:42.30 | james4765 | that'll work |
| 21:42.31 | [TK]D-Fender | fujin, You always could. |
| 21:42.37 | fujin | oh? how..? |
| 21:42.41 | james4765 | awesome |
| 21:43.19 | james4765 | thx |
| 21:45.06 | [TK]D-Fender | fujin, I don't use it personally so I can't give you the exact syntax. Go WIKI it |
| 21:45.37 | fujin | ughh @ wiki |
| 21:46.24 | james4765 | So I should be able to push it through <mac>-directory.xml? |
| 21:47.31 | [TK]D-Fender | fujin, there's even a sample in the sample extension.ael file... |
| 21:47.39 | [TK]D-Fender | james4765, sure. |
| 21:48.20 | james4765 | cool - this is the first time I've had to centrally provision a system, and it's been a bit of a trial |
| 21:49.58 | *** join/#asterisk t3rror (n=t3rror@gateway.sscgp.com) |
| 21:53.27 | *** part/#asterisk brodiem (n=brodiem@67.110.68.66.ptr.us.xo.net) |
| 21:53.59 | tomcontr3 | can anyone help me to clean up my zapata.conf configuration? Im getting too much noise when I call out |
| 21:54.07 | tomcontr3 | using my TDM400 with FXO |
| 21:56.58 | tomcontr3 | anyone? |
| 21:58.11 | tzafrir_laptop | what type of noise? echo? |
| 21:58.19 | tomcontr3 | like static |
| 21:58.19 | [TK]D-Fender | tomcontr3, if its noisy, its noisy. There is no zapata option for that if your gains sound OK on both sides and its just static |
| 21:58.37 | tomcontr3 | how can I fix that static? |
| 21:58.44 | [TK]D-Fender | tomcontr3, Another of the many reasons I avoid the TDM400P like the plague |
| 21:58.58 | tzafrir_laptop | tomcontr3: have you set gains |
| 21:59.28 | tomcontr3 | rxgain=3.0 |
| 21:59.28 | tomcontr3 | txgain=8.0 |
| 22:00.48 | tzafrir_laptop | tomcontr3: as a sanity check, how does it compare to a simple analog phone connected to that same line instead of the TDM400P card? |
| 22:00.58 | tomcontr3 | very bad |
| 22:01.13 | tomcontr3 | a normal phone, does has static, but this is way too bad |
| 22:01.24 | tzafrir_laptop | you mean: with a simple phone it sounds well? |
| 22:01.29 | tomcontr3 | yep |
| 22:02.19 | tomcontr3 | this is my zapata.conf config http://pastebin.ca/696434 |
| 22:06.43 | tomcontr3 | any idea? |
| 22:08.13 | *** part/#asterisk cellphone (i=lysol@monoperative.net) |
| 22:18.23 | *** join/#asterisk magic_hat (n=geoffdou@h-74-2-87-16.chcgilgm.covad.net) |
| 22:19.14 | magic_hat | hey everyone. I'm trying to get meetme set up, and I'm getting this when I dial the extension I'm trying to conference on: RNING[5971]: pbx.c:1797 pbx_extension_helper: No application 'Meetme' for extension (default, 99, 1) |
| 22:19.14 | magic_hat | ... help? |
| 22:23.26 | bkruse | magic_hat: zaptel installed? |
| 22:23.28 | bkruse | make menuselect |
| 22:23.29 | *** join/#asterisk CrazyTux[m] (n=CrazyTux@h460eb882.area3.spcsdns.net) |
| 22:24.01 | magic_hat | bkruse: I'm not sure... is it part of the default install? |
| 22:24.15 | bkruse | magic_hat: nope |
| 22:24.22 | bkruse | http://svn.digium.com/svn/zaptel |
| 22:24.38 | magic_hat | ah, crap! |
| 22:25.02 | bkruse | magic_hat: it provides the timing source for meetme (module ztdummy) |
| 22:25.33 | magic_hat | gotcha. lol, for the volume of conferencing we do it's probably easier to just go with a 3rd party dialup host. |
| 22:29.37 | *** join/#asterisk mitcheloc (n=mitchel@adsl-68-120-230-95.dsl.irvnca.pacbell.net) |
| 22:33.09 | *** join/#asterisk riddlebox (n=james@75-132-225-75.dhcp.stls.mo.charter.com) |
| 22:38.22 | luke-jr | When using a '#' transfer, how do I determine the transferring party? |
| 22:39.24 | *** join/#asterisk tsurko (n=tsurko@78.90.100.214) |
| 22:40.42 | *** join/#asterisk jfitzgibbon (n=chatzill@64.72.237.187) |
| 22:43.21 | riddlebox | does anyone have a good website to buy the TDM cards from? |
| 22:44.28 | *** join/#asterisk Math` (n=math@bas4-montreal19-1242357874.dsl.bell.ca) |
| 22:44.58 | tomcontr3 | does anyone knows how to fix the static problem with the TDM400P FXO card? |
| 22:45.18 | *** join/#asterisk yannj_fr (n=yannj_fr@vpn.intelunix.fr) |
| 22:45.59 | JT | get rid of analogue? ;) |
| 22:46.17 | JT | riddlebox: telephonydepot.com |
| 22:46.35 | Qwell | tomcontr3: You should try calling support - they should be able to help you with that |
| 22:46.50 | riddlebox | JT, I will get rid of analogue, only if you pay for the circuit |
| 22:47.17 | Qwell | riddlebox: digium.com sells most of the stuff directly |
| 22:47.25 | JT | riddlebox: eh |
| 22:47.57 | riddlebox | JT, I was joking about your answer to tomcontr3 |
| 22:48.23 | *** join/#asterisk Keltus (i=Keltus@about/cooking/nakedchef/beefstew/Keltus) |
| 22:48.40 | JT | yeah i was confused how it related to me paying for your line :) |
| 22:48.45 | *** join/#asterisk tsurko (n=tsurko@78.90.100.214) |
| 22:48.51 | JT | realisticly, tomcontr3 hasn't said exactly what the issue is |
| 22:49.02 | JT | so we don't have a hope of helping him |
| 22:49.16 | JT | but analogue doesn't actually sound that good... |
| 22:54.25 | *** join/#asterisk jedaustin (n=chatzill@austin-j.its.dist.maricopa.edu) |
| 22:55.19 | jedaustin | Question: does this mean the network connection is dropping? "channel.c: Didn't get a frame from channel: SIP/100-081b9750" |
| 22:58.43 | riddlebox | JT, do you know any coupon codes for telephonydirect.com? |
| 23:04.49 | Math` | any open bugs that would make asterisk ignore call files |
| 23:05.35 | Qwell | Math`: make sure pbx_spool is loaded, and that they aren't dated in the future |
| 23:06.07 | Math` | well when I restart asterisk it picks it up right away |
| 23:06.38 | Math` | running 1.4.7.1 |
| 23:06.43 | Qwell | upgrade |
| 23:07.04 | Math` | ok |
| 23:07.48 | *** join/#asterisk Burgerlicious (n=none@202.83.84.15) |
| 23:08.24 | *** join/#asterisk jebba (n=jebba@unaffiliated/jebba) |
| 23:08.56 | jebba | hey, i'm doing callfiles with "RetryTime: 900" but it doesn't appear to be waiting 15 minutes, but retrying much more quickly. Any ideas why? |
| 23:10.16 | jebba | "Unable to set utime" I think i may have just answered my own Q ;) |
| 23:11.47 | *** join/#asterisk rmayorga (n=rmayorga@unaffiliated/rmayorga) |
| 23:20.26 | JT | riddlebox: nup |
| 23:20.34 | *** part/#asterisk jebba (n=jebba@unaffiliated/jebba) |
| 23:20.48 | sparq | Hey, does it make sense to use Asterisk for classroom video conferencing? |
| 23:21.11 | sparq | (i.e., several video users in a conference) |
| 23:21.52 | JT | not really |
| 23:22.11 | russellb | sure it does |
| 23:22.16 | jedaustin | Question: does this mean the network connection is dropping? "channel.c: Didn't get a frame from channel: SIP/100-081b9750" |
| 23:22.16 | russellb | someone has even made a business out of it |
| 23:22.21 | russellb | see www.wimba.com |
| 23:22.36 | russellb | the client side is actually the more tricky part |
| 23:22.43 | russellb | because asterisk doesn't trascode video |
| 23:22.50 | russellb | and there aren't many good video clients out there |
| 23:22.52 | JT | it multiplexes multiples users into one stream? |
| 23:22.57 | JT | for video |
| 23:23.07 | russellb | switches based on who is speaking i think |
| 23:23.21 | russellb | i'm not exactly sure |
| 23:24.17 | sparq | Basically, I'd like to allow a couple of remote users to view a single video feed (from the lecture) and be able to ask questions over a speaker phone. |
| 23:24.34 | russellb | i think you can do that with app_conference |
| 23:24.42 | russellb | app_meetme won't switch video |
| 23:24.50 | sparq | Nifty! |
| 23:25.15 | russellb | keep in mind all video users must be using the same video codec |
| 23:25.20 | russellb | not necessarily the same audio codec, though |
| 23:25.22 | russellb | (for now ...) |
| 23:25.27 | russellb | it's one area we need to improve on, for sure |
| 23:25.30 | russellb | video in general .. |
| 23:25.47 | sparq | That's allright. I'm pretty sure everyone will be using either Ekiga or XMeeting. |
| 23:25.56 | russellb | cool |
| 23:26.02 | russellb | in theory, that should be fine, heh |
| 23:26.57 | sparq | my research group is spending a completely silly amount of money on an ISDN-based hardware solution from the early 1990s, and it sucks ass. |
| 23:27.13 | jedaustin | Is there somewhere where the zap exceptions are documented? chan_zap.c: Exception on 21, channel 5 |
| 23:27.17 | Qwell | silly as in $3, or silly as in $3,000,000? |
| 23:27.28 | JT | jedaustin: in chan_zap.c |
| 23:28.03 | sparq | Qwell: somewhere between the those numbers ^_^ |
| 23:28.05 | russellb | sparq: ha ... we've been having an argumen... er, discussion on the asterisk-dev mailing list about ISDN video for the past week or so |
| 23:28.31 | sparq | russellb: Oh? |
| 23:28.50 | russellb | specifically how to integrate it into asterisk so you can connect it to everything asterisk can do ... monitoring, recording, connecting it to IP ... |
| 23:28.54 | russellb | yup. |
| 23:29.42 | putnopvut | That's a fun thread to follow too. |
| 23:29.47 | sparq | I think most ISDN video users would be *delighted* to scrap their hardware for something simpler that just works over SIP. |
| 23:29.49 | russellb | heh, it has been stressful for me |
| 23:29.59 | russellb | SIP and "just works" ... that's dangerous :) |
| 23:30.38 | russellb | once you get it working, step back and don't touch it |
| 23:30.40 | russellb | :-p |
| 23:30.53 | russellb | it's always fun to pick on SIP ... |
| 23:31.10 | jedaustin | russelb: SIP can be a real joy when NAT/firewalls are involved :) |
| 23:31.21 | sparq | Beats the stuffing out of ISDN tunneled over frame relay encapsulated on an IPSEC link... |
| 23:31.29 | russellb | sparq: lol ... |
| 23:31.39 | russellb | jedaustin: oh yes |
| 23:31.45 | russellb | and i don't think SIP has enough ways to pass DTMF yet |
| 23:31.47 | russellb | 5 just isn't enough |
| 23:31.53 | JT | 5? |
| 23:32.39 | russellb | inband, rfc2833, SIP INFO, SUBSCRIBE/NOTIFY, and some other weird XML encoded thing we don't support |
| 23:33.21 | sparq | For classroom video conferencing, I think it's reasonable to require that the host (at least) be on a routable IP. Anyone who's going to invest the money and time to build something like that can get a real freaking IP address. |
| 23:34.01 | file | russellb: separate signalling port just for DTMF! |
| 23:34.08 | russellb | file: GOOD IDEA |
| 23:34.21 | JT | i still think sending dtmf tones over email is king |
| 23:35.07 | russellb | DTMF over Fax over T.38 over IP with SIP signalling |
| 23:35.18 | russellb | obviously. |
| 23:35.25 | ManxPower | sparq: trust me, people will want to run it behind NAT |
| 23:35.36 | sparq | ManxPower: Then they should be shot. |
| 23:35.39 | jedaustin | Question: does this mean the network connection is dropping? "channel.c: Didn't get a frame from channel: SIP/100-081b9750" |
| 23:35.44 | ManxPower | more often than you can possibly imagine. |
| 23:35.52 | JT | sparq: sip works fine behind nat |
| 23:35.57 | ManxPower | jedaustin: that message does not give enough information to know. |
| 23:36.16 | russellb | jedaustin: it can probably be ignored |
| 23:36.26 | russellb | unless you get a flood of them or something |
| 23:36.33 | russellb | should probably be a debug message ... |
| 23:37.13 | sparq | JT: I would say that it can be made to work from behind a NAT, but I wouldn't go quite so far as to say "just fine." ^_^ |
| 23:37.39 | JT | sparq: works fine when the server is set up right |
| 23:38.28 | *** join/#asterisk blinky42 (n=me@c-71-230-47-244.hsd1.pa.comcast.net) |
| 23:38.29 | jedaustin | The message right after that is channel.c: Bridge stops bridging channels Zap/5-1 and SIP/100-081b9750 |
| 23:39.12 | JT | jedaustin: if you can ignore it, it's all good, right? :) |
| 23:39.41 | tru_`z24 | With the digium cards, do they have hardware support for answering machine detection like the dialogic ones do? |
| 23:39.45 | jedaustin | JT: Its the time when they issue the client was having (transfer to extension that goes to a cell phone) |
| 23:39.54 | russellb | tru_`z24: not on the hardware, but there is software support for it |
| 23:40.10 | tru_`z24 | So is that a key reason why the dialogic cards are so much more expensive? |
| 23:40.10 | jedaustin | JT: Its the time when the issue the client was having ocurred (transfer to extension that goes to a cell phone) |
| 23:40.12 | JT | jedaustin: ...and that issue would be? |
| 23:40.14 | russellb | (keep in mind the whole idea behind low cost telephony hardware is pushing computation to the host CPU) |
| 23:40.21 | jedaustin | call dropped |
| 23:40.33 | russellb | tru_`z24: right, because processing is done on high cost specialty hardware |
| 23:40.50 | JT | ah |
| 23:40.51 | tru_`z24 | Right. |
| 23:40.52 | russellb | tru_`z24: where with asterisk and digium cards, you get to take advatage of doing processing on low cost general purpose hardware on the host CPU |
| 23:40.55 | ManxPower | jedaustin: we had dropped calls if the callerid was not valid when the call goes to the telco |
| 23:41.25 | jedaustin | ManxPower: this is with an Astribank. What hardware were you using? |
| 23:41.35 | ManxPower | Asterisk |
| 23:42.13 | *** join/#asterisk DarkRift (i=dark@bas10-montreal02-1177582690.dsl.bell.ca) |
| 23:43.05 | JT | jedaustin: hell, what type of interface is it? you haven't provided much info |
| 23:43.18 | jedaustin | JT: Sip to Zaptel |
| 23:43.26 | JT | zaptel... |
| 23:43.33 | JT | WHAT interface? |
| 23:44.56 | ManxPower | jedaustin: Analog FXO, Channelized T-1 FXO, PRI, E&M, E&M/Wink, etc, etc, etc |
| 23:44.59 | *** join/#asterisk Schreiber1337 (i=d8a9a5b2@gateway/web/cgi-irc/ircatwork.com/x-d78147bd9dbc1365) |
| 23:45.12 | jedaustin | JT: They're using a dial line in extensions.conf to send extension 107 calls to a cell phone number (dialed over zap). It's an 8 channel FXO Astribank |
| 23:45.18 | ManxPower | There are more types of interfaces between heaven and earth than you can dream of Horatio. |
| 23:45.20 | JT | jedaustin: OH AN FXO, THANKS |
| 23:45.45 | JT | sounds like an astribank issue |
| 23:45.52 | ManxPower | jedaustin: well you don't have to worry about callerid on analog ports |
| 23:46.44 | jedaustin | My gut feeling is the custom built dialplan is the problem. I inherited this thing and it's not built in a normal fashion. If it wasn't a production box I'd just tear it down and rebuild it. Why someone would build it on Ubuntu and write their own ruby app to create the dialplan is beyond me. |
| 23:47.43 | JT | i don't see why that'd be an issue, but okay |
| 23:47.53 | JT | you should pastebin the relevant bits |
| 23:48.10 | Schreiber1337 | Anyone ever have problems with MWI on Asterisk 1.2.16? I see a lot of posts about 1.4 but nothing like the problem that I'm having... |
| 23:48.51 | russellb | Schreiber1337: you could start by not being 10 releases behind :) |
| 23:49.15 | russellb | or 8 ... |
| 23:49.16 | russellb | whatever |
| 23:49.40 | Schreiber1337 | Well, my sytem can only have 2 hours of downtime a week.... blah... blah... |
| 23:50.02 | JT | it takes 5mins to upgrade, max |
| 23:50.13 | JT | 2 hours is plenty |
| 23:50.21 | Schreiber1337 | It was stable for 8 months with no intervention... then after a reboot.. MWI breaks... |
| 23:50.21 | _ShrikE | yeah and you spend the next month paying for it :) |
| 23:50.34 | russellb | Schreiber1337: you might want to try rebooting the phones |
| 23:51.11 | Schreiber1337 | Tried that.. It works sort of for 5 seconds and then clears... Every message, every phone... |
| 23:52.54 | Schreiber1337 | If I understand MWI, if the MWI gets set, it should remain set unless told otherwise... Right? |
| 23:53.21 | sparq | neat. app_conference is exactly what I'm looking for. That should save us about $4000 a year. |
| 23:53.29 | russellb | sparq: nice! |
| 23:53.41 | jedaustin | JT: By the time I got this machine it had been there 2 years without any updates, the apt repository was no longer working, etc. |
| 23:57.35 | jedaustin | The dialplan has no exception handling (t, i, etc) :( |
| 23:58.46 | riddlebox | does anyone know of a howto, to setup a TDM01B card? |