| 00:00.05 | zmauve | harryvv, do you have any pointer what I should search for? |
| 00:00.31 | *** join/#asterisk |dennis| (i=dennis@200.32.215.84) |
| 00:00.48 | znoG | harryvv: right, but a working configuration would help. Are you able to show me your zapata.conf? |
| 00:01.40 | znoG | harryvv: www.pastebin.com ? |
| 00:01.52 | harryvv | zmauve what does your company do? |
| 00:02.02 | harryvv | znoG I dont have your card. |
| 00:02.12 | znoG | harryvv: you have a Zaptel card? |
| 00:02.56 | *** join/#asterisk cp5 (n=samy@69.111.14.189) |
| 00:03.02 | cp5 | hello |
| 00:03.11 | *** join/#asterisk kiwnix (n=kiwnix@175.red-82-158-153.user.auna.net) |
| 00:03.43 | zmauve | we develop eyetrackers |
| 00:04.19 | harryvv | eyetrackers? |
| 00:04.54 | *** join/#asterisk docE (n=docelmo@static-71-251-95-4.tampfl.fios.verizon.net) |
| 00:04.59 | docE | ~seen damin |
| 00:05.05 | jbot | damin is currently on #asterisk (3d 4h 14m 2s). Has said a total of 10 messages. Is idling for 1d 22h 36m 25s |
| 00:05.12 | docE | sigh |
| 00:05.21 | docE | My DTMF issue is fixed.. |
| 00:05.23 | docE | sigh |
| 00:05.48 | zmauve | harryvv, a tft screen which sees where you're watching we (among others) develop products for people with severe disabilities |
| 00:06.04 | *** join/#asterisk bugz (n=bugz@cpe-24-27-67-66.houston.res.rr.com) |
| 00:06.42 | harryvv | thats great |
| 00:08.05 | zmauve | I have only worked there for about 5 months but it's cool, I mean it looks like a totally normal TFT screen, a normal user can't tell the difference between our eyetrackers and a normal monitor :) |
| 00:08.10 | heroine | Hi |
| 00:08.59 | harryvv | zmauve that can also help to free up hands |
| 00:09.34 | harryvv | say a officer is looking at a licence plate it would display the RO on the computer screen without typing it in. |
| 00:11.01 | zmauve | at work we play a lot of quake3 with our eyes, you aim pretty quick with an eyetracker :) |
| 00:11.38 | harryvv | no kidding |
| 00:11.39 | bwzb | anyone can tell me what the heck is this annoying warning message in my *? I just build 1.2.1, and I disable chan_modem.so, as it does not seem to build it. |
| 00:11.39 | bwzb | Dec 8 16:10:24 WARNING[32300]: format_wav.c:247 update_header: Unable to find our position |
| 00:11.40 | bwzb | Dec 8 16:10:24 WARNING[32300]: format_wav_gsm.c:243 update_header: Unable to find our position |
| 00:11.53 | harryvv | zmauve, does this work on linux? |
| 00:12.23 | harryvv | Could it be possible to guide a robotic arm with this software? |
| 00:12.34 | zmauve | harryvv, we are working on embedded all the logic into the monitor and simulate a USB HID device, then it would work :) |
| 00:13.35 | harryvv | if you can make a robotic arm say pick up a glass with eye movement then you have the potential for a great device to help a quad |
| 00:13.42 | zmauve | harryvv, the eyetracker just computes screen coordinates, eye coordinates (rel to screen in x,y,z) as well as an 3d-angle, you can then do whatever you want with this :) |
| 00:13.58 | *** join/#asterisk MikeJ[Laptop] (n=ircatjer@64.241.37.140) |
| 00:14.00 | harryvv | I see |
| 00:14.10 | zmauve | the problem today is the price tag |
| 00:14.20 | harryvv | for the software? |
| 00:14.25 | *** join/#asterisk alephcom (n=alephcom@207.34.97.130) |
| 00:14.38 | harryvv | so the eye can act kind like a joystick in a way.. |
| 00:15.35 | zmauve | the price for the hardware |
| 00:15.42 | harryvv | yea |
| 00:15.45 | zmauve | is about $30k |
| 00:15.49 | harryvv | for the robotic arm you mean |
| 00:16.09 | zmauve | the eyetracker costs about $30k |
| 00:16.27 | bugz | zmauve: you could almost hire someone to do that job for less |
| 00:16.59 | harryvv | ahh |
| 00:17.17 | harryvv | so lots of development time went into this then. |
| 00:18.09 | zmauve | the biggest use for eyetracking is neurological analisis, disability stuff, psycho analisis and of course software and web interface validation as well as for marketing research (does this ad really work?) |
| 00:18.38 | bugz | it would work wonders for the porn industry |
| 00:19.26 | zmauve | we showed a couple people some very well known ads (from Helly Hanses e.g.) and could pretty much conclude that while they (the ads) all caught the attention almost nobody looked at the logo of the company and therefore most ads were useless |
| 00:19.36 | nextime | harryvv : do you know "openeeg"? |
| 00:20.01 | nextime | it is better than moving your eye |
| 00:20.32 | zmauve | harryvv, when I add s,n,NoOp(${CALLERID}) to my dial plan I get this output in the asterisk console "-- Executing NoOp("Zap/4-1", "") in new stack", why is it empty? |
| 00:21.39 | harryvv | do you have cid on your line? |
| 00:21.50 | harryvv | is that a part of your phone service? |
| 00:22.07 | harryvv | openegg? |
| 00:22.07 | harryvv | no |
| 00:22.50 | zmauve | I don't know if I have callerid on my phone line, I kind of hoped everybody has this, I mean were in 2005 (soon 2006) aren't we? ;) |
| 00:24.30 | harryvv | zm, look at your phone bill |
| 00:24.42 | harryvv | if its not on there then no point troubelshooting this. |
| 00:24.47 | harryvv | brb |
| 00:25.35 | bugz | anyone know what the cause of garbling and chirping is when on a call? |
| 00:25.46 | bugz | it seems like any time the cpu gets interrupted theres a chirp |
| 00:26.01 | bugz | so for instance, if you are on a call and you do, say, ls -lR / |
| 00:26.07 | bugz | you can hear anything but f490 j9j4-9snanw~-9g~jgjgw0gj |
| 00:26.11 | bugz | its wierd |
| 00:26.17 | bugz | you might have heard it in some mp3's |
| 00:26.34 | bugz | it seems like a problem with voltage on the board itself |
| 00:32.02 | zmauve | how do I get the number which is being called on Zap channel when in the "s" extension? |
| 00:32.52 | *** join/#asterisk Igbothom_III (n=HiltonT@static-84.217.240.220.dsl.comindico.com.au) |
| 00:35.01 | *** join/#asterisk }btorch{ (n=kvirc@208.63.19.172) |
| 00:36.04 | }btorch{ | hey guys i'm trying to record some audio using the Record command and i have set the format to be gsm but when i try to playback that file on another context i get an error saying its not the correct format |
| 00:36.56 | zmauve | harryvv, have a nice day! |
| 00:38.46 | *** join/#asterisk anthm (n=anthm@adsl-68-254-173-186.dsl.milwwi.ameritech.net) |
| 00:38.46 | *** mode/#asterisk [+o anthm] by ChanServ |
| 00:46.01 | nextime | harryvv openeeg.sf.net |
| 00:47.02 | nextime | anyone can tell me how good is ooh323 and g729 on HEAD? |
| 00:47.13 | harryvv | yea read on it |
| 00:49.57 | *** join/#asterisk bugz (n=bugz@cpe-24-27-67-66.houston.res.rr.com) |
| 00:50.06 | bugz | did anyone get that q about the chirping? |
| 00:50.47 | tessier | How much are Cisco 7960's going for these days? Don't seem to be many on ebay. |
| 00:50.53 | tessier | I have a couple I am thinking about unloading. |
| 00:51.25 | tessier | Oh...scroll down. I do see quite a few now. Looks like $200 used. |
| 00:51.26 | tessier | Hrm.. |
| 00:53.12 | nextime | uh oh, but now g729 is included in * ? |
| 00:54.01 | nextime | ok, no, isn't included |
| 00:54.15 | nextime | show codecs lie |
| 00:54.41 | Corydon-w | nextime: what the fuck are you talking about? |
| 00:55.06 | Corydon-w | G729 has NEVER been included in Asterisk |
| 00:55.13 | Corydon-w | It's a purchaseable addon |
| 00:55.52 | Corydon-w | and it won't be included in Asterisk, either, until at least 2015 |
| 00:56.05 | mog_work | yup |
| 00:56.19 | nextime | Corydon-w: i know, show codecs show it and for a seconds i think to a false new better world |
| 00:56.43 | Corydon-w | show codecs is informational only |
| 00:57.24 | nextime | Corydon-w: yes, i've read the first line on the command output |
| 00:57.27 | Corydon-w | Note the disclaimer at the top of 'show codecs' which I inserted exactly because idiots thought it meant something. |
| 00:57.44 | nextime | thanks for idiot :) |
| 00:59.08 | Corydon-w | Note that you can turn off the disclaimer with export I_AM_NOT_AN_IDIOT=1 |
| 01:02.05 | *** join/#asterisk hhoffman (n=hhoffman@port-212-202-184-91.dynamic.qsc.de) |
| 01:02.53 | *** join/#asterisk jahani (n=k@adsl-155-41-192-81.adsl.iam.net.ma) |
| 01:04.23 | *** join/#asterisk TurboBuG (n=BadBug@c-24-61-4-191.hsd1.ma.comcast.net) |
| 01:04.32 | TurboBuG | hello everyone |
| 01:04.55 | *** join/#asterisk kiwnix (n=egarcia@175.red-82-158-153.user.auna.net) |
| 01:06.35 | *** join/#asterisk crash3m (i=crash3m@unaffiliated/crash3m) |
| 01:07.34 | *** part/#asterisk crash3m (i=crash3m@unaffiliated/crash3m) |
| 01:08.41 | TurboBuG | I have a spa3k in a remote location nat is working fine for line 1and the also the pstn line. I have one problem though the remote location has dyndns address. I recive calls perefectly fine and I can call fine, the one problem is when the ip changes in the remote location I have to reload so it can change the IP. |
| 01:09.20 | *** join/#asterisk nesys (n=nesys@2001:1418:1a6:0:20d:93ff:fe28:3ef8) |
| 01:09.35 | *** part/#asterisk nesys (n=nesys@2001:1418:1a6:0:20d:93ff:fe28:3ef8) |
| 01:20.53 | twisted[asteria] | TurboBuG, make the spa3k REGISTER to asterisk. |
| 01:21.46 | TurboBuG | twisted[asteria] yeah line 1 regesters |
| 01:22.07 | TurboBuG | are you refering to the pstn? |
| 01:22.09 | twisted[asteria] | if it registers then you don't have to reload to change the IP. |
| 01:23.52 | znoG | is 1.2.1 a complete rewrite of Asterisk or just some parts? |
| 01:24.22 | TurboBuG | twisted[asteria], I am trying it right now.. by the way does it matter if i am using the http athentication? |
| 01:24.28 | JunK-Y | znoG: just fixed bugs from 1.2.0 |
| 01:24.34 | bkw__ | yo yo yo twisted |
| 01:24.36 | bkw__ | ltns |
| 01:24.40 | twisted[asteria] | hah |
| 01:24.45 | twisted[asteria] | bkw__, no kidding |
| 01:24.45 | bkw__ | :P |
| 01:24.53 | bkw__ | we have both been busy |
| 01:24.58 | twisted[asteria] | yep |
| 01:25.08 | twisted[asteria] | still am... taking a breather |
| 01:25.17 | bkw__ | same here |
| 01:25.22 | bkw__ | i'm in Maryland right now |
| 01:25.23 | bwzb | anyone knows what this error about?: format_wav_gsm.c:243 update_header: Unable to find our position |
| 01:25.31 | bwzb | I just build my * with 1.2.1 |
| 01:25.33 | bkw__ | chances are its a WARNING |
| 01:25.34 | twisted[asteria] | wtf are you in MD? |
| 01:25.35 | bkw__ | and not an ERROR |
| 01:25.42 | bwzb | yes |
| 01:25.58 | bkw__ | twisted about an hour away from our datacenter |
| 01:26.01 | bkw__ | we are installing a new DB server |
| 01:26.03 | twisted[asteria] | nono |
| 01:26.05 | bwzb | but it's annoying, and only happened when people leaving voicemail |
| 01:26.08 | twisted[asteria] | i was asking why, but you answered that too |
| 01:26.28 | bkw__ | quad, Dual Core Opteron, 5tb array and 16 gigs of ram |
| 01:26.36 | twisted[asteria] | dude |
| 01:26.42 | twisted[asteria] | i could SOOOO play unreal on that thing nicely. |
| 01:26.46 | bkw__ | haha |
| 01:26.52 | bkw__ | our control panel will fly after this |
| 01:26.56 | justinu | battlefield2 |
| 01:27.17 | denon | bkw_: you're doing 4 of those in a fault-tolerant cluster .. right? |
| 01:27.17 | denon | <G> |
| 01:27.18 | twisted[asteria] | not to mention store all my pr0n |
| 01:27.18 | bkw__ | it has 4x1000watt power supplies |
| 01:27.18 | bwzb | bkw__: not sure you know this error, it's causing me headache. |
| 01:27.23 | twisted[asteria] | and all the pr0n on the net |
| 01:27.39 | bkw__ | bwzb, sox infile.wav -c 1 -r 8000 out.gsm |
| 01:27.41 | bkw__ | live happy |
| 01:28.09 | bwzb | bkw__: what does this do? |
| 01:28.18 | twisted[asteria] | lets the magic smoke out |
| 01:28.22 | twisted[asteria] | speaking of smoke... |
| 01:28.23 | twisted[asteria] | brb |
| 01:28.54 | bwzb | bkw__: Do I run this program anywhere? |
| 01:30.31 | bwzb | bkw__: is infile.wav those input wav files that I need to convert to gsm? |
| 01:31.54 | TurboBuG | twisted, do set "make call without reg"? currently set to yes |
| 01:32.47 | znoG | JunK-Y: i mean the 1.2.x series in general |
| 01:34.50 | bkw__ | oh file |
| 01:34.52 | bkw__ | where are you |
| 01:36.33 | *** join/#asterisk tengulre (n=tengulre@222.90.66.4) |
| 01:41.40 | *** join/#asterisk Dr-Linux (n=Nothing@202.125.141.9) |
| 01:42.23 | tengulre | hi,all |
| 01:42.27 | tengulre | anybody active? |
| 01:44.58 | harryvv | sorta |
| 01:44.59 | harryvv | :) |
| 01:45.59 | tengulre | are you use digium cards? |
| 01:52.00 | harryvv | i have |
| 01:52.48 | tengulre | analog or digital? |
| 01:53.12 | *** join/#asterisk bjohnson (n=bjohnson@i216-58-63-114.cybersurf.com) |
| 01:53.22 | harryvv | well, asterisk does not work with normal digital phones. only tdm and voip |
| 01:57.23 | *** part/#asterisk santiago (n=santiago@208.195.215.160) |
| 01:59.28 | hhoffman | do I need to worry about "Operating with different codecs 2[0x2 (gsm)] 4[0x4 (ulaw)] , can't native bridge..." |
| 01:59.39 | fugitivo | no |
| 02:00.17 | *** join/#asterisk javar (n=javar@69.79.133.185) |
| 02:00.33 | hhoffman | ktnx |
| 02:00.44 | JT | does anyone make adapters to connect a mobile telephone to an asterisk server? |
| 02:01.10 | *** join/#asterisk jsolares (n=jsolares@200.12.44.221) |
| 02:01.13 | zemmad | are u referring to a SIP/GSM phone?/ |
| 02:01.24 | JT | GSM |
| 02:01.55 | jsolares | i'm having a weird problem with sip and chanisavail, i have it in sip.conf to only allow one call at a time, however chanisavail says the channel is available even when there's a call in progress. any ideas? |
| 02:04.07 | *** join/#asterisk kFuQ (n=somedude@c-67-185-114-199.hsd1.wa.comcast.net) |
| 02:06.21 | *** join/#asterisk klictel (n=klictel@modemcable185.108-200-24.mc.videotron.ca) |
| 02:06.40 | flashbac1 | hey guys |
| 02:06.53 | JT | zemmad: no idea? |
| 02:06.53 | flashbac1 | i really need some help... |
| 02:07.04 | flashbac1 | i have asterisk running with realtime |
| 02:07.29 | flashbac1 | and i keep getting this error msg: |
| 02:07.30 | flashbac1 | Dec 8 21:03:05 WARNING[11188]: res_odbc.c:166 odbc_smart_execute: stmtDec 8 21:03:05 WARNING[11188]: res_odbc.c:172 odbc_smart_execute: SQL Execute returned an error -1: 42000: [Sybase][ODBC Driver][Adaptive Server Enterprise]Implicit conversion from datatype 'CHAR' to 'INT' is not allowed. Use the CONVERT function to run this query. |
| 02:07.30 | flashbac1 | <PROTECTED> |
| 02:07.47 | JT | sounds straightforward |
| 02:08.06 | flashbac1 | any ideas? |
| 02:08.07 | JT | i assume you'll need to enclose the query in CONVERT() |
| 02:08.08 | fugitivo | JT: www.2n.cz |
| 02:08.13 | flashbac1 | where? |
| 02:08.21 | JT | but read the sybase ASA reference manual |
| 02:08.23 | flashbac1 | how do i do that in res_odbc.c? |
| 02:08.29 | JT | oh umm |
| 02:08.40 | JT | are you sure what you're using supports sybase? |
| 02:08.57 | flashbac1 | i'm using Sybase native odbc drivers |
| 02:09.22 | flashbac1 | i just need to find where the stmt gets set so that i can do the CONVERT() thing |
| 02:09.22 | fugitivo | the message is clear |
| 02:09.30 | fugitivo | the field is int, and you're sending a char value |
| 02:09.42 | flashbac1 | not me...asterisk is outputting this on the CLI |
| 02:09.45 | fugitivo | change the field to char type :) |
| 02:09.55 | flashbac1 | but which field is that? |
| 02:10.14 | fugitivo | what is that? |
| 02:10.15 | flashbac1 | i keep getting that error msg over and over in the asterisk CLI |
| 02:10.17 | fugitivo | voicemail? |
| 02:10.23 | fugitivo | cdr? |
| 02:10.24 | jsolares | anyone know how can i send a progress indicator of 8 message with chan_zap with E1's? |
| 02:10.29 | fugitivo | do you use realtime? |
| 02:10.30 | JT | packetsniff it |
| 02:10.35 | flashbac1 | yeah |
| 02:10.37 | JT | read the sql |
| 02:11.00 | *** join/#asterisk areski (n=areski@145.Red-83-60-96.dynamicIP.rima-tde.net) |
| 02:11.13 | fugitivo | flashbac1: what do you do when you get that error? |
| 02:11.14 | flashbac1 | its trying to update sipfriends table...but the only column there that's INT is the id column... |
| 02:11.18 | *** join/#asterisk kFuQ (n=somedude@c-67-185-114-199.hsd1.wa.comcast.net) |
| 02:11.26 | flashbac1 | nothing...phones are registering... |
| 02:11.27 | docE | Say can someone tell me what dtmfmode=auto exactly does? |
| 02:11.47 | flashbac1 | chooses rfc2833 if not available uses inband |
| 02:11.53 | fugitivo | flashbac1: change it to char |
| 02:12.00 | flashbac1 | the id column? |
| 02:12.09 | flashbac1 | cant...has to be INT is AUTO_INCREMENT |
| 02:12.11 | docE | Cause I had one leg set to auto and the other to inband and its bitching about inband and rfc2833 |
| 02:12.14 | fugitivo | oh |
| 02:12.16 | flashbac1 | its the primary key in the table |
| 02:12.55 | flashbac1 | how can i make it output to the log the actual SQL statement? |
| 02:13.04 | javar | hi flashbacl |
| 02:13.18 | flashbac1 | hello |
| 02:13.24 | javar | what version of asterisk, are you working? |
| 02:13.27 | flashbac1 | 1.2 |
| 02:13.37 | javar | MySQL? |
| 02:13.42 | flashbac1 | no |
| 02:13.43 | JT | sybase |
| 02:13.46 | flashbac1 | Sybase 12.5 ASE |
| 02:13.46 | JT | see the error |
| 02:14.04 | javar | you installed unixODBC? |
| 02:14.09 | flashbac1 | and i'm using the native Sybase ODBC drivers for unixODBC |
| 02:14.18 | javar | ok |
| 02:14.27 | javar | at CLI> odbc show |
| 02:14.59 | flashbac1 | Name: sybase> |
| 02:14.59 | flashbac1 | DSN: SYBASE |
| 02:14.59 | flashbac1 | Connected: yes |
| 02:15.06 | javar | good |
| 02:15.56 | flashbac1 | also this msg comes up right after the other one: |
| 02:15.57 | flashbac1 | ec 8 21:15:14 WARNING[11283]: res_config_odbc.c:399 update_odbc: SQL Execute error! |
| 02:15.57 | flashbac1 | [UPDATE sipfriends SET ipaddr=?, port=?, regseconds=?, username=?, fullcontact=? WHERE name=?] |
| 02:16.27 | docE | is that the syntax for sybase? |
| 02:16.33 | *** join/#asterisk puzzled (n=patrick@puzzled.xs4all.nl) |
| 02:16.36 | flashbac1 | yes |
| 02:16.46 | flashbac1 | syntax is fine... |
| 02:16.51 | puzzled | evening |
| 02:17.51 | *** join/#asterisk jahani (n=k@adsl-7-47-192-81.adsl.iam.net.ma) |
| 02:18.39 | flashbac1 | i think asterisk is inserting something in '' when is not supposed to? |
| 02:21.08 | JunK-Y | yay, we have a new cat now! |
| 02:21.13 | *** join/#asterisk znoG (n=gs@OL101-122.fibertel.com.ar) |
| 02:21.50 | brookshire | yay! |
| 02:22.02 | docE | huh? |
| 02:22.05 | docE | new cat? |
| 02:22.19 | brookshire | how many do you got? |
| 02:22.41 | docE | I got 1 cat and dog and my cat beats up my dog |
| 02:22.45 | JunK-Y | just 1! |
| 02:23.26 | JunK-Y | hes affraid like hell and hes above the bed now |
| 02:23.40 | JunK-Y | so his name: microsoft is a perfect fit for him! |
| 02:23.47 | docE | hehe |
| 02:23.52 | docE | I named mine Asterisk |
| 02:24.03 | JunK-Y | asterisk sounds much better then a cat :) |
| 02:24.03 | brookshire | awh that's cute |
| 02:24.33 | docE | Well I didnt have any funky penguins around.. so I named the cat it.. |
| 02:25.41 | docE | Does CVS work for Asterisk 1.2 and Addons? |
| 02:26.37 | brookshire | svn? |
| 02:26.51 | brookshire | svn would be a great name for a kitty |
| 02:26.52 | brookshire | lol |
| 02:27.11 | docE | I couldnt get svn to download kept bitching about sighup's and I didnt send any to it.. |
| 02:27.29 | *** join/#asterisk ctooley (n=ctooley@rrcs-24-227-212-162.sw.biz.rr.com) |
| 02:27.34 | docE | This sucks.. I could only get about 9.5Mb worth of bandwidth from Digiums ftp site.. |
| 02:27.47 | brookshire | there are two of them |
| 02:27.51 | orlok | my cat is called grep |
| 02:27.53 | orlok | <PROTECTED> |
| 02:27.53 | brookshire | ftp1.digium.com and ftp2.digium.com |
| 02:28.32 | docE | hmmm |
| 02:28.35 | *** join/#asterisk Druken (n=druken@CPE00121716da99-CM000e5cde4ca2.cpe.net.cable.rogers.com) |
| 02:28.44 | docE | brookshire you work @ digium right? |
| 02:29.32 | brookshire | yes |
| 02:31.25 | hhoffman | orlok: that's great :-) |
| 02:31.38 | orlok | hmm |
| 02:31.43 | brookshire | g2g.. lates :) |
| 02:31.44 | orlok | i think our sip provider has just crapped out |
| 02:31.48 | docE | Have you heard anytthing about dCAP? I passed but have nothing to show for it.. |
| 02:31.52 | orlok | i can see port 5060udp reaching out one way only |
| 02:31.53 | docE | Who's your provider? |
| 02:32.07 | orlok | a pretty new one, you woulent have heard of them i'd say :) |
| 02:32.09 | *** join/#asterisk jimmy_deanPB (n=jhodapp@cpe-65-26-179-224.indy.res.rr.com) |
| 02:32.14 | docE | Try me |
| 02:32.28 | orlok | its in another country too |
| 02:32.31 | orlok | nextep |
| 02:32.48 | docE | ok in that case.. Try plainvoip.com free .25c for trying it out.. good clear calls |
| 02:32.50 | jimmy_deanPB | Can anyone recommend the best VOIP provider for cheapest to best performance ratio in the U.S. that supports Asterisk? |
| 02:33.01 | docE | Plainvoip.com |
| 02:33.09 | docE | .0092 domestic termination US48 |
| 02:33.42 | jimmy_deanPB | docE: you suggesting that to me? |
| 02:33.52 | justinu | what about origination? |
| 02:34.21 | docE | Soon.. Having some problems with origination provider.. Will be worked out tomorrow I hope |
| 02:35.39 | Druken | docE: only NYC did's... |
| 02:35.51 | docE | and yes.. We do termination right now.. Will have more.. Not programmed yet until I get issues worked out |
| 02:35.55 | docE | They also come with 911 |
| 02:36.05 | docE | But still working on the 911 routines |
| 02:36.11 | jimmy_deanPB | docE: only New York terminations? Too bad I'm in Indy |
| 02:36.21 | docE | Anyone have a SPA3000 they could help me configure the FXO side? |
| 02:36.34 | docE | I have A-Z termination |
| 02:36.38 | docE | Just NYC DID's right now |
| 02:36.54 | docE | check out www.plainvoip.com/?action=rates |
| 02:36.57 | Druken | how much for canadian termination ?? :) |
| 02:36.58 | docE | I think thats the URL |
| 02:37.00 | *** join/#asterisk javar (n=javar@69.79.133.185) |
| 02:37.18 | Druken | ?page=showrates :) |
| 02:37.19 | *** join/#asterisk wunderkin (i=kev@ip68-226-113-228.ph.ph.cox.net) |
| 02:37.19 | docE | Would have to look I dont know off the top of my head |
| 02:37.22 | docE | sorry |
| 02:37.30 | docE | I coded it like 4 months ago |
| 02:37.48 | Druken | uhg... |
| 02:37.49 | javar | hi |
| 02:38.24 | Druken | ya could have at least a sort by, so it's in alphabetical order |
| 02:39.02 | docE | Should be ordered by code |
| 02:39.09 | Druken | it is... |
| 02:39.10 | docE | and I plan to.. working on too many other projects |
| 02:39.33 | Druken | or have it like links at the top, to specify what you want to order by :) |
| 02:39.47 | docE | Like I said.. I plan to.. Just havent done it yet.. |
| 02:39.55 | *** join/#asterisk test34 (n=test34@unaffiliated/test34) |
| 02:39.59 | docE | Maybe over my 2 week christmas vacation |
| 02:40.15 | test34 | anyone ever tried the Arris TM402P MTA or the Scientific Atlanta WebSTAR DPX2203 ? |
| 02:40.25 | *** join/#asterisk javar (n=javar@69.79.133.185) |
| 02:40.30 | docE | Nope.. Not me.. |
| 02:40.48 | docE | anyone got a SPA3000 they care to share config information with me? |
| 02:41.05 | jimmy_deanPB | I've got a 2002 |
| 02:41.12 | jimmy_deanPB | but haven't configured it yet |
| 02:41.25 | javar | copy the extensions? |
| 02:41.31 | flashbac1 | yeah |
| 02:41.43 | docE | I need to configure the FXO (line) port |
| 02:41.48 | docE | cant figure it out for asterisk |
| 02:42.03 | docE | I wanna buy a TDM 1FXO/FXS card.. but cant afford it now.. |
| 02:42.11 | docE | Maybe next month or something |
| 02:42.23 | zemmad | has anyone connected asterisk to sip phone?? |
| 02:42.35 | zemmad | could i get some help in doing this |
| 02:43.05 | docE | um, dude.. everyone but jbot and you has done that |
| 02:43.22 | Druken | tdm is crap :( |
| 02:43.35 | docE | Well shit then.. someone help me with this one.. :) |
| 02:43.44 | Druken | jbot: have you got a sip phone working yet? |
| 02:44.12 | Druken | jbot, your dumb |
| 02:44.17 | zemmad | docE, i'm tryin to get this done....i'm not seeing my client even connecting to sip phone |
| 02:44.43 | Druken | docE: did you google the configs for a SPA3003 ? |
| 02:44.50 | Druken | er.. 3000 |
| 02:44.53 | *** join/#asterisk Insanity5 (n=feaw@ip68-111-5-23.sv.om.cox.net) |
| 02:45.01 | docE | yes nada.. worth while |
| 02:45.16 | Druken | wut can't ya figure out? |
| 02:45.31 | Druken | i've never even laid eyes on one, but i'm willing to give it a whirl :) |
| 02:45.52 | docE | How to configure FXO port to ring to asterisk |
| 02:46.30 | Druken | ok, what kinda settings are there for the FXO ? |
| 02:47.48 | docE | found a thing on voxiolla |
| 02:47.58 | docE | lemme try this.. thanks drunk tho |
| 02:53.36 | harryvv | http://www.canada.com/vancouversun/story.html?id=1a45473e-ee19-47ff-b1eb-6ecb90b5dc37&k=95504 |
| 02:53.49 | harryvv | Canadians ringing up high-tech phones |
| 02:54.06 | harryvv | dont use voip in your advertising |
| 02:54.07 | harryvv | ;) |
| 02:54.23 | harryvv | seems the word is getting associated with poor sound quality. |
| 02:54.38 | harryvv | heard that complaint from two other people. |
| 02:54.46 | Druken | hmmm.... |
| 02:54.53 | Druken | i use Broadband telephone service :) |
| 02:55.02 | Druken | must be vonage that's killing the voip :) |
| 02:55.50 | harryvv | no |
| 02:56.07 | harryvv | Rogers communications in canada is not including the term voip in its advertising. |
| 02:56.43 | Druken | shit... i wish i could only spend 207 a month |
| 02:56.44 | JT | heh it's no wonder |
| 02:56.53 | JT | people use stingey codecs and wonder why it sounds like shit |
| 02:56.59 | Druken | uhmm... rogers isn't even a voip player yet |
| 02:57.23 | Druken | they are doing "home phone" which is sprint's local telephone service |
| 02:58.56 | *** part/#asterisk javar (n=javar@69.79.133.185) |
| 02:59.23 | jimmy_deanPB | crap, stupid US government and the E911 ruling...why can't they stay out of people's lives? |
| 02:59.39 | Qwell | jimmy_deanPB: because the phone companies pay them not to? |
| 02:59.41 | jimmy_deanPB | I can't sign up for a VOIP provider because they don't offer E911 service in my area yet |
| 02:59.56 | jimmy_deanPB | I have 911 on my cell phone! |
| 03:00.02 | denon | yeah .. and that stupid Interstate Highway Syste.. why cant they keep off my land! |
| 03:00.07 | denon | system |
| 03:00.09 | *** join/#asterisk javar (n=javar@69.79.133.185) |
| 03:00.21 | jimmy_deanPB | why are they forcing me to have 911 or no service at all |
| 03:00.33 | Qwell | kinda funny if you think about it |
| 03:00.39 | Druken | :) |
| 03:00.39 | Qwell | without service, you've not got 911 anyhow |
| 03:00.43 | jimmy_deanPB | it is!~ |
| 03:00.48 | jimmy_deanPB | exactly! |
| 03:01.00 | denon | jimmy_deanPB: because if they dont force companies, nobody will do it .. and our emergency system will be worthless |
| 03:01.12 | denon | nobody thinks about emergency services until they need em |
| 03:01.19 | denon | so its the govt's job to mandate it for our own good |
| 03:01.28 | Druken | ya know.. if they really took 911 seriously, no circuit would ever to cut, even if you had no phone, you would still get a dialtone and be able to dial 911 |
| 03:01.33 | Druken | but i notice that doesn't happen |
| 03:01.53 | denon | that didnt even make sense .. |
| 03:02.11 | Druken | no circuit would ever BE cut |
| 03:02.15 | Druken | that better? |
| 03:02.31 | denon | the govt can't stop you from renting a backhoe and wiping out ma bell |
| 03:02.41 | denon | though they do have a service you can call to flag stuff out |
| 03:02.46 | jimmy_deanPB | so who do I sign up for? |
| 03:02.52 | Druken | i don't mean that way.. i mean if you cancel your phone |
| 03:02.53 | jimmy_deanPB | denon, worthless, hardly |
| 03:02.59 | Druken | they wouldn't disconnect the loop |
| 03:03.03 | harryvv | make it simple and just include one pstn line in the pbx |
| 03:03.18 | denon | Druken: dunno, most residential loops still allow 911 even when they're disconnected |
| 03:03.23 | denon | 911, operator, etc |
| 03:03.33 | Druken | well, not here... |
| 03:03.43 | Druken | they disconnect the loop if you don't have service |
| 03:04.17 | denon | hmm, qwest among others will give you a tone, and a message on how to get it reconnected |
| 03:04.22 | denon | dial a number for customer service or something |
| 03:04.45 | Druken | apparently bell canada plays by a diffrent set of rules |
| 03:04.52 | denon | oh, canada .. heh |
| 03:05.50 | harryvv | my bell cell phone sucks |
| 03:05.51 | harryvv | :) |
| 03:06.24 | Druken | cell phones always allow 911, i know that much |
| 03:06.30 | Druken | service or no, you can dial 911 |
| 03:06.34 | denon | right .. and why? |
| 03:06.39 | denon | because the cell companies are generous? |
| 03:06.40 | harryvv | all use gsm codec right? |
| 03:06.42 | jimmy_deanPB | so, I can't use www.voicepulse.com it looks like, any other providers someone can suggest to me? |
| 03:06.43 | Druken | fuct if i know |
| 03:06.43 | denon | or because it's mandated? |
| 03:06.50 | denon | it's mandated. |
| 03:06.55 | denon | I'm saying rules aren't a bad thing |
| 03:07.06 | Druken | i never said they were... |
| 03:07.15 | denon | well, someone was .. |
| 03:07.24 | denon | jimmy_whine I think |
| 03:07.29 | Druken | i'm just saying, residential circuits should play by the same rules |
| 03:07.36 | denon | move south :) |
| 03:07.50 | jimmy_deanPB | denon, that's not my name :) |
| 03:07.55 | Druken | uhmm.. no that's ok... i prefer canada :) |
| 03:07.59 | jimmy_deanPB | denon, just because I'm mad at our government |
| 03:08.27 | jimmy_deanPB | because they don't like to defend individual freedom anymore |
| 03:08.50 | denon | to each his own |
| 03:12.26 | *** join/#asterisk mog_home (n=mogorman@user-24-236-84-48.knology.net) |
| 03:12.41 | ctooley | jimmy_deanPB, you might try Asterlink |
| 03:12.53 | mog_home | or nufone ^_^ |
| 03:13.59 | *** join/#asterisk alephcom (n=alephcom@207.34.97.130) |
| 03:18.34 | *** join/#asterisk shmaltz (n=chatzill@mail.dmaven.com) |
| 03:20.18 | harryvv | u purchaced it? |
| 03:20.28 | harryvv | okay what does that unit cost? |
| 03:20.46 | jimmy_deanPB | ctooley, thanks! |
| 03:20.54 | shmaltz | harryvv, it costed more then an asterisk mini itx |
| 03:21.06 | harryvv | How much per seat? |
| 03:21.25 | shmaltz | it's not per seat it's per port ~$500 for 2 ports |
| 03:21.30 | harryvv | per port |
| 03:21.43 | harryvv | that does not include the phones? |
| 03:21.43 | SkramX | hi, harryvv |
| 03:21.48 | harryvv | hi sk |
| 03:21.50 | shmaltz | the base system inclues 2 ports and costs ~$500 |
| 03:21.59 | CoaxD | jesus christ $500 2 ports |
| 03:22.03 | shmaltz | harrryvv, of course not, not even the PBX |
| 03:22.11 | ctooley | jimmy_deanPB, #asterlink has the guys that can help you out. bkw_ and file[desk] are your friends. :) |
| 03:22.11 | shmaltz | it's just the VM |
| 03:22.16 | harryvv | Soo asterisk beats it out then. |
| 03:22.23 | CoaxD | um yea |
| 03:22.27 | CoaxD | @ $0.00 |
| 03:22.37 | harryvv | heheh |
| 03:22.38 | *** join/#asterisk Syrus_ (n=pascal@tahiti.mpl.rullier.net) [NETSPLIT VICTIM] |
| 03:22.53 | shmaltz | harryvv, of course, but to intergrate with an existing panasonic, this is much cheaper, and easier |
| 03:23.19 | jimmy_deanPB | ctooley, like how? |
| 03:23.40 | *** join/#asterisk Syrus_ (n=pascal@tahiti.mpl.rullier.net) [NETSPLIT VICTIM] |
| 03:24.02 | ctooley | bkw_, and file[desk] work at Asterlink and #asterlink is a channel where you can ask questions. |
| 03:24.10 | ctooley | if you went with Asterlink anyway |
| 03:24.28 | jimmy_deanPB | it doesn't seem like they have residential termination services |
| 03:24.45 | ctooley | The don't provide local DID's, no |
| 03:24.46 | *** join/#asterisk Astinus (i=iBook@freenode/staff/gentoo.astinus) |
| 03:25.04 | jimmy_deanPB | bummer |
| 03:25.09 | jimmy_deanPB | that's what I'm looking for |
| 03:25.11 | ctooley | You might email the folks at TXLink or CommPartners if you need someone with a large supply of local DIDs |
| 03:25.37 | ctooley | TXLink doesn't have online signup but they have better standard rates than a lot of places |
| 03:25.50 | jimmy_deanPB | really, and all of these places are good with Asterisk? |
| 03:26.03 | ctooley | And they have recently merged with CommPartners who sponsor voip-info.org |
| 03:26.10 | ctooley | jimmy_deanPB, yep |
| 03:26.26 | jimmy_deanPB | interesting |
| 03:26.44 | jimmy_deanPB | it's kinda hard looking for voip providers it seems |
| 03:26.51 | harryvv | Thats a nifty name commpartners |
| 03:26.55 | ctooley | Tell TXLink that Chris Tooley at gNumber sent you. They'll take good care of you and they're reliable |
| 03:27.26 | jimmy_deanPB | ctooley, ok, excellent...I'll send them an email right now |
| 03:27.28 | harryvv | I need a company to sell me portable 604 area code numbers. |
| 03:28.07 | ManxPower | I don't suppose anyone has installed a small formfactor / small power usage PC into their car to play MP3s? |
| 03:28.07 | ctooley | They had some reliability issues in the past, but the merger has given them the opportunity to make most of those go away. Asterlink is a little quicker at doing stuff if you're just doing outbound though |
| 03:28.09 | harryvv | I mean a company that does portability and sell DIDs |
| 03:28.23 | ManxPower | harryvv, tried Teliax? |
| 03:28.26 | ctooley | harryvv, TXLink does that |
| 03:28.34 | harryvv | xo does also? |
| 03:28.38 | ctooley | We ordered 10,000 DIDs from them at one point |
| 03:28.41 | ManxPower | A person from Teliax Support is on this channel often. |
| 03:28.42 | harryvv | wow |
| 03:28.49 | SkramX | Yeah, I am also looking for just one DID. |
| 03:28.52 | harryvv | teliax ? |
| 03:29.00 | SkramX | ~teliax |
| 03:29.06 | SkramX | They are a VOIP Provider |
| 03:29.12 | harryvv | ctcooley you mean teliax ? |
| 03:29.14 | ctooley | I ordered mine from TXLink |
| 03:29.24 | ManxPower | teliax.com |
| 03:29.26 | SkramX | ctooley: will txlink sell just one? |
| 03:29.31 | ManxPower | or teliax.net or something like that |
| 03:29.42 | ctooley | SkramX, I'm pretty sure they would |
| 03:29.55 | SkramX | Okay |
| 03:30.07 | ctooley | you were looking for DIDs in 604? |
| 03:30.16 | harryvv | yes |
| 03:30.33 | harryvv | anything for vancouver bc |
| 03:30.39 | ManxPower | Oh! |
| 03:30.49 | ManxPower | Sorry, I assumed 604 was USA. Silly me! |
| 03:30.53 | ctooley | they dont' have any current ones for 604 but BC shouldn't be an issue. |
| 03:31.23 | jimmy_deanPB | ctooley, ok, I emailed the txlink guys |
| 03:31.33 | SkramX | ctooley: are you a reseller? |
| 03:31.42 | jimmy_deanPB | ctooley, by the way, know if they have any Indianapolis DIDs? |
| 03:31.55 | ctooley | SkramX, nope. They have an interface for customers to provision your own. |
| 03:32.06 | ctooley | you can take up available and they'll charge you for them. |
| 03:32.07 | ManxPower | ctooley, You'll have trouble finding a carrier with "unlimited" inbound DIDs. |
| 03:32.15 | ManxPower | Why not get a toll free for the same per min cost. |
| 03:32.40 | ctooley | ManxPower, toll free isn't the same per minute cost I have |
| 03:33.02 | ManxPower | ctooley, with 10,000 DIDs it would not be 8-) |
| 03:33.31 | ctooley | ManxPower, if I have 10,000 DID's there's a reason I had the did's versus one toll free |
| 03:33.42 | *** join/#asterisk [hC] (n=hardcore@S0106000e9b96114f.vf.shawcable.net) |
| 03:33.53 | ManxPower | ctooley, *nod* |
| 03:34.00 | hhoffman | what is the dundi-e164 stuff in the example extensions.conf? |
| 03:34.52 | ctooley | ManxPower, Luckily for us the new product is all outbound calling so DID's aren't useful. |
| 03:35.03 | hhoffman | Teliax has been good for me :-) |
| 03:35.28 | ctooley | We use one of our Toll Free numbers as CID but don't have to pay the much higher toll free connection rate. |
| 03:35.36 | ManxPower | All Internet Telcos suck. Teliax is one of the ones that suck less. |
| 03:36.07 | ctooley | Which is why our servers are colo'd at the facilities with our providers. :) |
| 03:36.27 | ctooley | We use SIP interconnect to them but it's over a private GigE LAN |
| 03:37.06 | ctooley | The other little company I work on uses Internet service to Asterlink, but when we get a bit bigger I'll be asking them to let me ship them the server. |
| 03:37.42 | ManxPower | I don't use a lot of VoIPoI (VoIP over Internet) |
| 03:37.53 | ManxPower | Mostly it's the local LAN or companu wide WAN |
| 03:37.54 | ctooley | jimmy_deanPB, what are the area codes in Indy? |
| 03:38.10 | SkramX | Where can I get free TTS.. I need to make a proof of concept system... |
| 03:38.18 | harryvv | need 604 dids |
| 03:38.45 | jimmy_deanPB | ctooley, 317 is the main one |
| 03:39.10 | SkramX | harryvv: how many? |
| 03:39.18 | shmaltz | ~tts |
| 03:39.19 | jbot | tts is, like, time to sleep |
| 03:39.23 | ctooley | jimmy_deanPB, From their available did list 317: 25 available |
| 03:39.30 | harryvv | sk, no limit...dpends on how many i sell |
| 03:39.35 | SkramX | jbot: tts is also text to speech |
| 03:39.36 | jbot | SkramX: okay |
| 03:39.37 | jimmy_deanPB | ctooley, where did you look that up at? |
| 03:39.39 | SkramX | ~tts |
| 03:39.41 | jbot | it has been said that tts is time to sleep, or text to speech |
| 03:39.52 | SkramX | shmaltz: better? |
| 03:40.02 | shmaltz | SkarmX, much :) |
| 03:40.05 | jimmy_deanPB | ctooley, assuming their rates are good, they sound like the provider for me |
| 03:40.25 | ctooley | jimmy_deanPB, once you are signed up with them they have a "VoIP Control" web interface. |
| 03:40.33 | jimmy_deanPB | very nice |
| 03:40.37 | jimmy_deanPB | where are they out of? |
| 03:41.03 | ctooley | jimmy_deanPB, Dallas, Las Vegas, New York, Chicago, a couple of other cities. |
| 03:41.20 | shmaltz | What a ride: |
| 03:41.22 | shmaltz | http://news.yahoo.com/s/ap/20051209/ap_on_re_us/midway_accident;_ylt=AiycVHV8GgBou1W4aqorIQCs0NUE;_ylu=X3oDMTA2Z2szazkxBHNlYwN0bQ-- |
| 03:41.30 | jimmy_deanPB | not bad, though it sounds like there's a business opportunity to become a top VOIP provider in the Indy area |
| 03:41.53 | ctooley | jimmy_deanPB, I wouldn't suggest it. |
| 03:42.05 | jimmy_deanPB | any particular reasons? |
| 03:42.18 | jimmy_deanPB | too crowded of a market? |
| 03:42.41 | ctooley | jimmy_deanPB, All the little CLECs are scrambling to start providing SIP Termination/Origination and the big boys are already doing it (though you have to be big to be interesting) |
| 03:42.44 | *** join/#asterisk |dennis| (i=dennis@200.32.215.84) |
| 03:42.57 | jimmy_deanPB | yeah, it sounds like it |
| 03:43.13 | jimmy_deanPB | large barriers to entry in other words |
| 03:43.20 | ctooley | They've got infrastructure and experience. By the time you get your stuff working, you wont' have a market left to sell to |
| 03:43.43 | jimmy_deanPB | it'd be a fun venture nonetheless :) |
| 03:43.49 | jimmy_deanPB | I bet you could do it all with Asterisk |
| 03:44.04 | ManxPower | Doing an ITSP is VERY much like starting an ISP. i.e. you either have to install locall numbers or you have to pay a big carrier for numbers for DID. If you just want to provide toll calling - well there are 50 billion of those companies around. |
| 03:44.39 | jimmy_deanPB | very true |
| 03:44.48 | ctooley | jimmy_deanPB, Asterisk isn't particularly well suited to running a CLEC and unless you know how to do the development, you're screwed. |
| 03:45.14 | jimmy_deanPB | hmm, so what would you use to run a CLEC? |
| 03:45.22 | ctooley | jimmy_deanPB, and no offense met, but if you don't know the players in your market already... you're not ready to start a CLEC |
| 03:45.23 | jimmy_deanPB | commercial stuff? |
| 03:45.37 | jimmy_deanPB | ctooley, oh I won't do it, it was just a fun idea |
| 03:45.50 | *** join/#asterisk konfuzed (n=KonfuzeD@H129.C72.B0.tor.eicat.ca) |
| 03:46.47 | ctooley | or SER doing most of your call routing, using Asterisk for apps (voicemail, IVR, etc) |
| 03:46.55 | *** join/#asterisk bjohnson_ (n=bjohnson@i216-58-58-83.cybersurf.com) |
| 03:46.59 | jimmy_deanPB | SER? |
| 03:47.51 | ManxPower | jimmy_deanPB, SER is software specially designed for SIP and only for SIP call routing. |
| 03:48.45 | ManxPower | Unless you are a carrier you don't usually need to worry about SER |
| 03:50.01 | dokhench | sip express router |
| 03:50.59 | *** join/#asterisk bmg505 (n=leon@dsl-146-31-32.telkomadsl.co.za) |
| 03:51.01 | ManxPower | Has everyone here read the SECURITY file in the Asterisk source? If not, you should do so right now. |
| 03:51.24 | jimmy_deanPB | thanks for your help ctooley and ManxPower |
| 03:51.32 | ctooley | jimmy_deanPB, no problem |
| 03:51.44 | *** join/#asterisk morale (i=russell@S010600111155e117.cg.shawcable.net) |
| 03:52.01 | jimmy_deanPB | ctooley, hopefully txlink can help me out tomorrow |
| 03:52.50 | jimmy_deanPB | anyway, night all |
| 03:52.54 | ctooley | jimmy_deanPB, if they don't email you right back, give them a call in the morning. |
| 03:52.57 | SkramX | http://les.net/ |
| 03:53.14 | SkramX | They buy right from Level3 |
| 03:53.57 | ctooley | SkramX, so do a lot of people. We have a SIP Termination contract with Level3 |
| 03:54.48 | SkramX | Cool, Cool. |
| 03:54.58 | SkramX | I have an associate/frient who is working on getting one with them |
| 03:55.00 | SkramX | and some others |
| 03:55.03 | ctooley | anyway... needsome sleep |
| 03:55.08 | ctooley | later folks |
| 03:55.12 | ManxPower | As far as I can tell most companies with DIDs nationally buy from Level3 |
| 03:55.32 | ManxPower | I can't even get to les.net but that's not suprizing. |
| 03:55.41 | alephcom | les.net does have some local DIDs but I think that's only in Manitoba |
| 03:56.07 | SkramX | alephcom: what? |
| 04:02.10 | alephcom | I know that les.net does have pstn connectivity but I'm not sure where all. I think every other than Manitoba is level3 with them. |
| 04:02.17 | dokhench | silly question here, but how are blocks of numbers actually bought/sold? what is accessed to find out if a number is avail/inuse, what telco has it, etc? is this through telcordia? |
| 04:06.12 | dokhench | anyone? |
| 04:07.53 | *** join/#asterisk stkn_ (i=nobody@gentoo/developer/pdpc.active.stkn) |
| 04:09.32 | hhoffman | if I have something like "[internal] include => local" "[local] exten => _9215XXXXXX,1,Dial(${PSTNTRUNK}/${EXTEN:-1})" what do I need to do to carry across the context? I'm getting the error "request '92157727898@internal' does not exist" |
| 04:10.56 | file[laptop] | hhoffman: you do realize that the exten => line won't match that number? |
| 04:11.03 | file[laptop] | the number is one digit too long |
| 04:12.33 | hhoffman | huh |
| 04:12.37 | hhoffman | :-/ |
| 04:13.05 | dokhench | '92157727898@internal' does not exist" has too many digits to match. |
| 04:13.51 | hhoffman | doesn't the -1 say to just take off the 9? |
| 04:14.09 | SkramX | 2) |
| 04:14.09 | SkramX | The United States foreign policy doctrines are a combination of Wilsonianism with a mix of both realism and hegemony. Through the .war on terror. and attempts to bring peace to the Middle East, the Wilsonianism aspect is evident. |
| 04:14.09 | file[laptop] | uh no |
| 04:14.13 | file[laptop] | ${EXTEN:1} does that |
| 04:14.13 | SkramX | <PROTECTED> |
| 04:14.18 | SkramX | Wooops! |
| 04:14.19 | dokhench | 215-XXX-XXX != 215-772-7898 |
| 04:14.19 | hhoffman | shit! |
| 04:14.32 | alephcom | Thanks, SKramX :-) |
| 04:14.35 | file[laptop] | and that's for dialing out, it's not for pattern matching |
| 04:14.40 | SkramX | alephcom: no problem. |
| 04:14.48 | file[laptop] | see what dokhench said and you shall see... |
| 04:14.49 | SkramX | i can send you the rest of my paper later.. |
| 04:15.04 | *** join/#asterisk artmeister (n=artmeist@c-71-56-71-9.hsd1.ga.comcast.net) |
| 04:16.08 | distortion | http://www.force10networks.com/images/E1200-lg1.jpg |
| 04:16.21 | distortion | wow. 1260 gige ports |
| 04:17.09 | *** join/#asterisk wasim (n=wasim@pdpc/supporter/active/wasim) |
| 04:19.29 | dokhench | nanpa is the telco world equiv of iana (in north america), right? |
| 04:19.49 | SkramX | WTF, today I saw someone selling 360 DID's.. |
| 04:21.51 | artmeister | anybody here have much experience running asterisk on a virtual server? |
| 04:22.04 | distortion | dok: nanpa manages the number planning for north america- area code splits, nxx assignments- im not familiar with iana |
| 04:22.08 | SkramX | artmeister: i do! i do! |
| 04:22.08 | dokhench | art: you mean user mode linux? |
| 04:22.15 | heroine | like an user mode linux or a xen host ? |
| 04:22.28 | SkramX | http://linux-vserver.org rocks |
| 04:22.41 | hhoffman | thanks guys, fixed it :-) |
| 04:22.41 | dokhench | you'll get interrupt problems if your virtual box isn't beefy enough. |
| 04:22.47 | znoG | anyone know why chan_modem is no longer built by default? |
| 04:22.51 | SkramX | hmm |
| 04:22.56 | SkramX | it works great on my vpshost |
| 04:22.57 | znoG | with asterisk 1.2.1 |
| 04:23.01 | SkramX | i have a couple customers running asterisk |
| 04:23.14 | artmeister | Cool, I'll have to check that out. I'm running an installation on a godaddy vserver |
| 04:23.28 | SkramX | Hmm |
| 04:23.44 | SkramX | i bet they have that vpshost packed and latency like crazy |
| 04:23.45 | SkramX | heh |
| 04:24.03 | artmeister | no, aroun 45 to 50 ms |
| 04:24.12 | artmeister | very little jitter |
| 04:24.16 | SkramX | hmm |
| 04:24.16 | SkramX | okay |
| 04:24.47 | dokhench | best bet is do go dedicated. isn't that much more monthly for something like a serverbeach box vs a uml box. |
| 04:25.10 | artmeister | only strange problem is that sometimes it takes really long to go from a single digit menu in one context to another.... |
| 04:25.26 | *** join/#asterisk MikeJ[Laptop] (n=ircatjer@d14-69-8-30.try.wideopenwest.com) |
| 04:25.45 | artmeister | was wondering if it could be related to the time-slicing... |
| 04:29.03 | *** join/#asterisk colinm_ (n=colol@VDSL-130-13-8-235.PHNX.QWEST.NET) |
| 04:30.06 | *** part/#asterisk colinm_ (n=colol@VDSL-130-13-8-235.PHNX.QWEST.NET) |
| 04:31.30 | ManxPower | znoG, because it never worked right anyway and has been depracated. |
| 04:31.57 | ManxPower | znoG, You didn't read the UPGRADE.txt notes? |
| 04:33.23 | znoG | ManxPower: yep just did and read' what it said :) |
| 04:33.39 | znoG | ManxPower: i upgraded to 1.2.1 hoping the distinctive ring stuff was fixed, but doesn't look like it. |
| 04:42.59 | sbingner | znoG: whats wrong w/ the distingtive rung stuff? |
| 04:44.43 | znoG | sbingner: when a call comes in, Asterisk doesn't wait a ring or two to determine the pattern, so it comes in with 0,0,0 every time |
| 04:45.07 | Katty | hi lads. |
| 04:45.15 | znoG | at least the type of ring my provider sends me, at the start of the second ring you can see it rings 3 times quickly, where a normal ring is just a standard ring |
| 04:45.23 | znoG | s/see/hear |
| 04:45.31 | morale | has anyone setup postgresql+voicemail yet? that app_voicemail.c thing.. |
| 04:45.57 | *** join/#asterisk SplasPood (n=jwb@brooklyn.paravolve.net) |
| 04:48.40 | *** join/#asterisk gnosys (n=ksford@griffin2.GnoSys.us) |
| 04:53.31 | *** join/#asterisk salaud (n=salaud@h-64-105-253-69.sttnwaho.covad.net) |
| 04:53.54 | *** join/#asterisk tclineks (n=tclineks@ppp-70-252-171-228.dsl.tpkaks.swbell.net) |
| 04:54.00 | salaud | Anyone know if you can pass multiple e-mail addresses in the e-mail address field in voicemail.conf? |
| 04:54.37 | tclineks | salaud: what happens when you try? |
| 04:54.55 | *** join/#asterisk implicit (n=implicit@ip70-181-114-97.oc.oc.cox.net) |
| 04:55.00 | implicit | anyone done http provisioning on grandstreams? |
| 04:55.17 | salaud | Haven't gone through all the testing yet. Just curious to see if I could get a shortcircuit to a bunch of testing on a live system |
| 04:55.31 | sbingner | znoG: I thought you could set asterisk to wait and it would work properly |
| 04:55.52 | sbingner | problem then is of course that you don't know it's ringing till like the 2nd or 3rd ring |
| 04:56.10 | loud | separate them with commas. |
| 04:57.03 | salaud | loud: you talking to me about the commas? |
| 04:57.08 | loud | yes. |
| 04:57.50 | salaud | loud: It looks like there is an <options> parameter behind the <pageremail> ... How would you specify the <options> parameter... whatever that does? |
| 04:58.20 | salaud | loud: I mean if you just put a bunch of email addresses with commas... how would you specify <options> |
| 04:58.32 | tclineks | I'm using asterisk as my answering machine for voicemail -- I have an ht-286 ATA but haven't set it up yet (*horrible* quality with the thing currently) so I'm using my PSTN phone to catch calls before asterisk answers ( exten => s,1,Ringing : exten => s,2,Wait,10 : exten => s,3,Answer() ) |
| 04:58.38 | wasim | "" |
| 04:59.11 | tclineks | The problem is that asterisk will pick up partway through a current call -- How can I have it detect the line state and not pick up if I have with my PSTN -- Is this possible? |
| 04:59.46 | salaud | loud: how would you specify the pager-email separately with e-mail addresses with commas? I was thinking (since it's perl) maybe putting 'email1,email2' etc. |
| 05:05.17 | *** join/#asterisk IronHelix (n=irc@ool-45785cfe.dyn.optonline.net) |
| 05:17.37 | *** join/#asterisk kuku5 (i=kuku@c-67-175-218-223.hsd1.il.comcast.net) |
| 05:18.40 | *** join/#asterisk mog_home (n=mogorman@user-24-236-84-48.knology.net) |
| 05:19.25 | kuku5 | I can't get the msg light to light on these cisco 7960 |
| 05:19.32 | kuku5 | Any suggestions? |
| 05:19.50 | Qwell | got a mailbox= line in sip.conf? |
| 05:20.03 | kuku5 | yes |
| 05:20.10 | Qwell | Got any voicemail? |
| 05:20.16 | kuku5 | ailbox=14 |
| 05:20.16 | kuku5 | allow=ulaw |
| 05:20.16 | kuku5 | allow=alaw |
| 05:20.20 | kuku5 | of course |
| 05:20.20 | Qwell | ailbox? |
| 05:20.30 | kuku5 | bad copy |
| 05:20.39 | kuku5 | mailbox=14 |
| 05:20.46 | Qwell | what happens when you go into voicemailmain for 14? |
| 05:20.51 | Qwell | Do you get messages? |
| 05:20.57 | kuku5 | you have 20 messages |
| 05:22.10 | kuku5 | what else |
| 05:22.38 | Qwell | It's registering properly and everything? |
| 05:22.44 | kuku5 | yup |
| 05:24.13 | kuku5 | no light |
| 05:24.26 | kuku5 | no stutter ( after setting it on the phone ) |
| 05:24.34 | *** join/#asterisk r0d3nt|m (n=RatMan@tinfoilhat.net) |
| 05:26.59 | kuku5 | anyone ? |
| 05:27.09 | mog_home | anyone ? |
| 05:27.12 | mog_home | bueler... |
| 05:27.36 | *** join/#asterisk qw3rty (n=qw3rty@c-67-167-79-57.hsd1.il.comcast.net) |
| 05:28.31 | SkramX | mog_home: lol |
| 05:28.36 | qw3rty | for voicemail and music on hold, what would crappy sound have to do with... CPU, Memory, Disk or Audio Drivers? |
| 05:29.07 | SkramX | well, if the cpu is tied up or there is no mememory free, then itll jitter, but thats more b/w |
| 05:30.07 | qw3rty | I am running on a PIII 700Mhz with 256MB RAM and I get crappy audio... not a big deal as it's just for my house |
| 05:30.45 | kuku5 | qqwell: bad context :) |
| 05:30.51 | kuku5 | iin voicemail.conf |
| 05:31.00 | kuku5 | the files were in the different folder. |
| 05:31.02 | kuku5 | <PROTECTED> |
| 05:31.04 | Qwell | kuku5: yeah, having the right context helps |
| 05:31.41 | kuku5 | you could press the messages button to get to it |
| 05:31.48 | kuku5 | thats what got me confused :) |
| 05:33.03 | *** join/#asterisk jahani2 (n=k@adsl-175-47-192-81.adsl.iam.net.ma) |
| 05:41.09 | *** join/#asterisk santiago (n=santiago@208.195.215.160) |
| 05:46.19 | litage | what exactly is DUNDi? |
| 05:46.39 | mog_home | dynamic unified number discovery |
| 05:49.12 | litage | mog_home: what does that mean? |
| 05:49.26 | wasim | http://www.dundi.com/ |
| 05:49.43 | litage | thanks |
| 05:50.39 | *** join/#asterisk magic_1 (n=maig_1@dsl-165-149-85.telkomadsl.co.za) |
| 05:51.07 | mog_home | lol |
| 05:51.13 | mog_home | think of it like enum |
| 05:51.19 | mog_home | without a verisign |
| 05:51.22 | mog_home | its p2p |
| 05:51.27 | mog_home | or like gnutella |
| 05:52.51 | denon | or al qeida |
| 05:53.46 | litage | dundi sounds awesome. it'd be nice to not have another megalomaniac company like verisign =P |
| 05:55.36 | Insanity5 | Is there a win32 client for asterisk that shows things like who's claling, helps you transfer calls, offers a phone directory, etc? Kind of like "interaction client" for many commercial systems. |
| 05:55.48 | denon | yes |
| 05:55.49 | denon | dozens of them |
| 05:55.54 | denon | hit the wiki |
| 05:55.59 | Insanity5 | denon - win32? |
| 05:56.04 | denon | yes |
| 05:56.08 | denon | windows-based, flash based |
| 05:56.09 | denon | you name it |
| 05:56.16 | Insanity5 | Now some switchboard manager, something aimed at the end user |
| 05:56.18 | denon | there are so many that even I dont know them all .. so go google |
| 05:56.20 | denon | yes |
| 05:56.20 | Insanity5 | win32 = not html/iexplorer |
| 05:56.24 | denon | there are all of the above |
| 05:56.30 | denon | there are native win32, running .net |
| 05:56.30 | denon | etc |
| 05:57.53 | litage | rather than list every client in sip.conf, etc., is it possible to tell asterisk to query a database to determine client information? |
| 05:57.55 | Insanity5 | What was the wiki url? Wasn't it like voipinfo.org? |
| 05:58.03 | litage | Insanity5: voip-info.org |
| 05:58.06 | denon | http://www.voip-info.org/ |
| 05:58.52 | Insanity5 | yuck, google search :( |
| 05:59.26 | *** join/#asterisk tuxinator_linuxM (n=spabin@70-32-106-248.ontrca.adelphia.net) |
| 05:59.37 | Insanity5 | denon - This is what I want: http://www.voip-info.org/tiki-pagehistory.php?page=Asterisk+bounty+windows+manager&source=7 |
| 05:59.45 | Insanity5 | denon - I don't need a sip client :) |
| 06:00.04 | wasim | litage: yes, its called realtime |
| 06:00.34 | denon | Insanity5: I know exactly what you want |
| 06:00.42 | denon | and it exists. |
| 06:00.48 | denon | I opened that bounty a long, long time ago |
| 06:00.55 | Insanity5 | What is one sample? I can't believe a bounty would exist for it |
| 06:01.10 | Insanity5 | Cause this google search is giving me pain :( |
| 06:01.45 | mog_home | bounty for waht |
| 06:01.51 | Insanity5 | this looks ok, but the links are dead: http://www.voip-info.org/wiki-Asterisk+Call+Manager+for+Windows |
| 06:02.04 | Insanity5 | mog_home - http://www.voip-info.org/tiki-pagehistory.php?page=Asterisk+bounty+windows+manager&source=7 |
| 06:02.22 | mog_home | i would need a windows box.... |
| 06:02.24 | jahani2 | how to configure outgoing call without prefix? |
| 06:02.24 | mog_home | so no dice |
| 06:02.55 | denon | Insanity5: google for IpswitchBoard |
| 06:03.14 | Insanity5 | thx |
| 06:03.26 | denon | http://ipswitchboard.thorben.dk/ |
| 06:03.40 | Insanity5 | yuck, gotta register to d/l :( |
| 06:04.16 | denon | uh heh |
| 06:04.17 | denon | you're welcome |
| 06:04.22 | wasim | what do you expect for winblows |
| 06:04.24 | Insanity5 | Just the job for mailinator.com |
| 06:04.31 | Insanity5 | :) |
| 06:05.31 | Insanity5 | .net 2.0 required... beginning 20 some meg download. |
| 06:05.34 | litage | thanks wasim. looking up realtime now |
| 06:09.38 | mog_home | ew .net |
| 06:11.31 | Insanity5 | The framework is quite annoying. |
| 06:11.45 | Insanity5 | All the capabilities to be ported to linux, but of course ms never will :) |
| 06:12.16 | *** join/#asterisk sekhmet_ (n=pez@ppp-70-226-134-230.dsl.mdsnwi.ameritech.net) |
| 06:14.24 | artmeister | can anone tell me how I can tell an application to ignore a specific charecter within the () ? |
| 06:15.41 | wasim | artmeister: : |
| 06:16.07 | Qwell | artmeister: explain |
| 06:16.19 | wasim | artmeister: and the ${EXTEN:1:-5} type ... readme.variables |
| 06:16.31 | jahani2 | is that correct to make outgoing calls ? exten => _0XXXXXXXX,1,Dial(SIP/${EXTEN}@192.168.0.9,20) |
| 06:18.13 | artmeister | Qwell: for example, if I had: exten => Voiecemail(sean@vmcontext) |
| 06:18.16 | Insanity5 | How hard is the upgarde from a cvs-head build (slightly pre 1.2 final) to the current release? |
| 06:18.31 | artmeister | the voicemail app would think I wanted to use the 's' option |
| 06:19.31 | Qwell | It should only think it's option s, if it were something like |
| 06:19.37 | Qwell | Voicemail(|sean@vmcontext) |
| 06:19.43 | Qwell | but, since there is no |... |
| 06:19.52 | *** join/#asterisk scardinal (n=supreme@port816.ds1-suoe.adsl.cybercity.dk) |
| 06:20.17 | Qwell | artmeister: what is happening? |
| 06:20.41 | Qwell | ooohhhh... |
| 06:21.05 | Qwell | hmm, that's odd |
| 06:21.12 | Qwell | I guess voicemail boxes can only be numeric |
| 06:21.23 | Qwell | because, yes, s1234 as a mailbox is valid |
| 06:21.53 | artmeister | ok, I have exten => 10,1,Record(/var/www/vhosts/somedomain.com/httpdocs/comments/${CALLERIDNUM}${ANSWEREDTIME}:wav) |
| 06:22.22 | *** join/#asterisk AFK1 (n=itsme@203.81.239.35) |
| 06:22.27 | artmeister | but, when I try to record, there is a problem with the '.' |
| 06:22.28 | AFK1 | hi all,, |
| 06:22.57 | AFK1 | hey can anyone gimme a price idea for Digium 2400 |
| 06:22.59 | AFK1 | port |
| 06:23.06 | Qwell | What .? |
| 06:23.06 | mog_home | depends how its configured |
| 06:23.35 | AFK1 | 2400 card with 24 FXS ports? |
| 06:23.37 | artmeister | Qwell: the one in 'somedomain.com' |
| 06:24.27 | artmeister | I get an error message in the CLI: 8 23:20:00 WARNING[23170]: file.c:978 ast_writefile: No such format 'com/httpdocs/comments/4045201596:wav' |
| 06:24.27 | artmeister | Dec 8 23:20:00 WARNING[23170]: app_record.c:248 record_exec: Could not create file /var/www/vhosts/somedomain |
| 06:27.50 | Qwell | artmeister: Why are you doing :wav? |
| 06:28.46 | artmeister | good question, I'm using an example in a book |
| 06:29.05 | Qwell | That was a smudge |
| 06:29.20 | Qwell | coffee maybe |
| 06:30.15 | *** join/#asterisk nagl (n=nagl@213.235.241.6) |
| 06:30.21 | ManxPower | artmeister, Try exten => 10,1,Record("/var/www/vhosts/somedomain.com/httpdocs/comments/${CALLERIDNUM}${ANSWEREDTIME}":wav) or exten => 10,1,Record(/var/www/vhosts/somedomain\.com/httpdocs/comments/${CALLERIDNUM}${ANSWEREDTIME}:wav) |
| 06:30.39 | Qwell | ManxPower: It should check on the last . |
| 06:30.52 | Qwell | So, simply /var/www/vhosts/somedomain.com/httpdocs/comments/${CALLERIDNUM}${ANSWEREDTIME}.wav |
| 06:30.58 | ManxPower | Qwell, It should not check any . since you don't need an extension. |
| 06:31.16 | Qwell | filename.format |
| 06:31.24 | Qwell | per show application record |
| 06:31.49 | ManxPower | Qwell, Um, different versions of record work differently. |
| 06:31.54 | ManxPower | artmeister, what version of Asterisk? |
| 06:32.11 | ManxPower | 1.0.9: Record(filename:format|silence[|maxduration][|option]) |
| 06:32.12 | Qwell | With the error he's getting, it's clearly trying to use anything after the . as the format |
| 06:32.34 | artmeister | I'm on 1.2 |
| 06:32.45 | artmeister | the \ doesn't work |
| 06:32.54 | ManxPower | 1.2: Record(filename.format|silence[|maxduration][|options]) |
| 06:33.00 | Qwell | mine should - if not, then it's a bug |
| 06:33.06 | ManxPower | artmeister, "show application record" |
| 06:33.10 | artmeister | nor do 'text' ot `text` or "text" |
| 06:33.43 | *** join/#asterisk krisguy (n=krisguy@h216-170-039-057.adsl.navix.net) |
| 06:33.43 | ManxPower | artmeister, report it on bugs.digium.com |
| 06:34.39 | Qwell | didn't try mine? ;/ |
| 06:34.59 | artmeister | ok, thanks. Just FYI... Should I typically be able to use quotes or \ to ignore charecters? |
| 06:35.05 | artmeister | in other applications.... |
| 06:35.20 | artmeister | or rather FMI? |
| 06:35.25 | ManxPower | artmeister, no, but it was worth a try 8-) |
| 06:36.01 | artmeister | ok, thans for the help on a newb question! |
| 06:36.35 | ManxPower | artmeister, quick fix is ln -s /var/www/vhosts/somedomain.com /var/www/vhosts/somedomain-com |
| 06:36.51 | Qwell | I guarantee mine will work |
| 06:36.56 | Qwell | It's using strrchr |
| 06:36.59 | ManxPower | then exten => 10,1,Record(/var/www/vhosts/somedomain-com/httpdocs/comments/${CALLERIDNUM}${ANSWEREDTIME}:wav) |
| 06:37.22 | artmeister | thanks, that's even better than the cron job I was about to write for moving these files.... |
| 06:37.24 | Qwell | ext = strrchr(filename, '.'); /* to support filename with a . in the filename, not format */ |
| 06:37.56 | ManxPower | artmeister, did you fix your exten line? |
| 06:38.08 | ManxPower | since the format you were using is for older asterisk's |
| 06:38.20 | Qwell | That format will still work with 1.2 |
| 06:38.27 | Qwell | however, not if there is a . |
| 06:38.31 | ManxPower | Qwell, Yesh - if there's not a . |
| 06:38.37 | Qwell | so, abc:wav would work, abc.def:wav would not |
| 06:38.38 | Qwell | BUT |
| 06:38.40 | Qwell | abc.def.wav will |
| 06:38.52 | ManxPower | Really, you should ALWAYS check "show application foo" to see command docs |
| 06:43.58 | jahani2 | qlq parle francais? |
| 06:44.11 | artmeister | Manx: it works! Thanks for the pointers! |
| 06:44.45 | ManxPower | "show applications" is your friend |
| 06:52.19 | *** join/#asterisk sigterm (i=sigterm@devious.info) |
| 06:55.39 | *** join/#asterisk benjk (n=benjamin@f8a01-0357.din.or.jp) |
| 07:08.25 | *** join/#asterisk rene- (n=rene-@201.137.84.89) |
| 07:10.25 | rene- | hello, does anyone know what the irc nick of kristian kielhofner is? |
| 07:13.45 | Fendor_ | Hm. |
| 07:13.50 | Fendor_ | NaughtyE1 |
| 07:14.13 | *** join/#asterisk chibski (n=chibski@user-24-236-120-147.knology.net) |
| 07:16.17 | chibski | With asterisk is there support for seamless transfer of active sip channels to a different server if the sip client has a lower latency to server A over server B? |
| 07:19.23 | rene- | thx |
| 07:20.42 | *** join/#asterisk lme (n=lme@gw-r-ifc.infoclip.fr) |
| 07:20.54 | rene- | chibski never heard of it |
| 07:26.10 | *** part/#asterisk rene- (n=rene-@201.137.84.89) |
| 07:31.02 | *** join/#asterisk rene- (n=rene-@201.137.84.89) |
| 07:32.47 | rene- | in an [1..4] scale how would you rate the reliability of an asterisk system that ran from an usb stick, cdrom, ata hd or scsi hd? |
| 07:32.58 | rene- | all other things being equal |
| 07:35.11 | *** part/#asterisk rene- (n=rene-@201.137.84.89) |
| 07:35.19 | *** join/#asterisk rene- (n=rene-@201.137.84.89) |
| 07:46.24 | rene- | does R2 (unicall) work over tdmoe? |
| 08:03.26 | *** join/#asterisk lorinc (n=ang@caracas-1492.adsl.interware.hu) |
| 08:05.05 | *** join/#asterisk psk (n=psk@golia.caltanet.it) |
| 08:08.22 | *** join/#asterisk af_ (n=af@ip-142-84.sn1.eutelia.it) |
| 08:15.32 | *** join/#asterisk chapeaurouge (n=chap@85.201.81.201) |
| 08:19.46 | salaud | jahani2: je parle francais |
| 08:20.06 | chapeaurouge | with a name like that, no wonder |
| 08:20.14 | lme | ... |
| 08:20.16 | salaud | chapeaurouge: ;) |
| 08:20.40 | salaud | chapeaurouge: with a name like that, you probably do too... |
| 08:20.55 | chapeaurouge | a little bit |
| 08:23.05 | salaud | jahani2: Il faut que je me tire... desole.. |
| 08:23.12 | salaud | night all |
| 08:24.03 | lme | nice example :) |
| 08:24.20 | chapeaurouge | lol |
| 08:24.28 | lme | how to promote french speaking worldwide |
| 08:24.44 | chapeaurouge | yea.. i like the way my gf do it better :-D |
| 08:25.12 | lme | Audiard is dead... ! |
| 08:26.49 | *** join/#asterisk indego (n=chris@floyd.gms.lu) |
| 08:28.41 | *** join/#asterisk sofh (n=ok@203.101.180.103) |
| 08:33.14 | lme | well !!!! |
| 08:33.37 | lme | gone to burn my ML110 server.... This one is like a piece of sh.. |
| 08:35.20 | *** join/#asterisk Gunnar (n=gunnar@bkkb-gw.bitcon.no) |
| 08:45.18 | sofh | hello all |
| 08:45.45 | sofh | did somebody try qualcomm CDMA phone(USB) with asterisk ? |
| 08:46.47 | sofh | :-S |
| 08:48.44 | knight_ | anyone have a csv of country codes? |
| 08:49.05 | knight_ | i gotta convert some voip csv rate sheets to a new csv for my billing system |
| 08:49.36 | *** join/#asterisk EriSan (n=erisan@151.8.109.102) |
| 08:50.40 | *** join/#asterisk KriS83 (n=KriS@212.202.141.92) |
| 08:50.51 | KriS83 | Hello @ all |
| 09:07.07 | *** join/#asterisk frix (i=frix@p54A85A3B.dip.t-dialin.net) |
| 09:07.18 | frix | good morning. |
| 09:09.07 | frix | i have a strange problem. i use asterisk with isdn and i have connected a teledat 2ab which makes me use old analog telephones. everything works just fine, i can make internal and external calls from both phones |
| 09:09.33 | iDunno | doesn't sound like a problem yet ;) |
| 09:09.52 | frix | but as soon as i connect an isdn phone and add it to my config, every incoming call gets denied after one ring |
| 09:10.02 | frix | there it is :) |
| 09:11.47 | frix | and the internal calls dont work either anymore |
| 09:12.10 | frix | sorry about my english, btw |
| 09:17.22 | *** join/#asterisk id_root (n=Miranda@195.239.37.3) |
| 09:18.11 | *** join/#asterisk [Airwolf] (n=airwolf@attilla.nl) |
| 09:30.39 | heroine | i'm upgrading from asterisk-1.0 to 1.2 .. but can't find the mysql support for voicemail in addonds-1.2 .. should i use the code from addons-1.0.x ? |
| 09:32.41 | *** join/#asterisk taec (n=phil@eventhorizon.hosting365.ie) |
| 09:36.02 | *** join/#asterisk dudes (n=dudes@12-215-32-62.client.mchsi.com) |
| 09:46.12 | *** join/#asterisk rumba (n=ropawa@cpe-68-201-147-176.sw.res.rr.com) |
| 09:47.06 | mrtwister | heroine: yes, you need upgrade everything to 1.2 |
| 09:48.31 | knight_ | heh, writing utility scripts to reformat csv == boring!!! |
| 09:55.16 | joelsolanki | hello all: anybody using cisco ata 186 with asterisk ...i m not getting clid in csv cdr. i get in the softphone xlite & hardware device linksys pap2 but not in cisco ata. |
| 09:55.27 | joelsolanki | is there any configuration for that in cisco ata ? |
| 09:59.25 | joelsolanki | anyhints ? |
| 09:59.35 | joelsolanki | any body awake here ? |
| 09:59.38 | joelsolanki | :( |
| 10:01.11 | webmind | no |
| 10:01.37 | lme | zzZZZZzzzzZZZZZZzzzZZZZZ |
| 10:01.38 | joelsolanki | :) |
| 10:02.11 | *** join/#asterisk jaike (i=aa@210.5.118.254) |
| 10:02.41 | joelsolanki | anybody knows asterisk + cisco ata config. i m not getting src while using cisco ata |
| 10:03.17 | joelsolanki | I need " SRC " in csc cdr but when using cisco ata 186 it is not getting. while using linksys pap2 it is working. |
| 10:03.23 | joelsolanki | any idea guys / |
| 10:03.36 | lme | no sorry |
| 10:03.47 | jaike | ours work well...came with default config |
| 10:04.04 | *** join/#asterisk newl (n=newlook@203-59-61-62.dyn.iinet.net.au) |
| 10:06.28 | jaike | anyone know any sip/iax providers supporting g729? with rates around 1c/min, our company is doing 100,000mins a month expected to reach 300,000 by mid next year and were looking for a stable provider |
| 10:08.18 | *** join/#asterisk Abbas (i=Abbas@203.81.199.94) |
| 10:10.48 | knight_ | Salvage: 3747 => 6374752(Armenia - Mobile/Special Services => Philippines-Digitel) |
| 10:10.58 | knight_ | lol, that's not quite the same country codes :) |
| 10:11.23 | knight_ | guess my salvaging code needs manual confirmation :( |
| 10:14.05 | *** join/#asterisk pa (n=Paolo@unaffiliated/pa) |
| 10:14.07 | joelsolanki | jaike : what is your cisco ata version. mine is 2.15 |
| 10:17.50 | veepster_ | guys, will I need any special hardware just to test asterisk out? or can I test basic functionality on any linux box? |
| 10:19.31 | *** join/#asterisk Tili (i=Tili@202-133-67-44-dialup.sat.net.pk) |
| 10:21.13 | veepster_ | and does it work on 2.6 linux kernel? |
| 10:23.21 | lme | veepster_: at least, you should have an fxo interface |
| 10:23.36 | lme | and yes it's working with 2.6 |
| 10:25.21 | *** join/#asterisk sofh (n=ok@203.101.180.103) |
| 10:25.26 | sofh | hello all.. |
| 10:25.29 | sofh | howz everybody.. |
| 10:25.48 | sofh | i need some help..plz give me an idea if anybody can.. |
| 10:26.04 | sofh | how to integerate a CDMA(wireless) phone with asterisk ? |
| 10:26.07 | *** join/#asterisk Junbug (n=Administ@lan.webunited.net) |
| 10:26.46 | sofh | :-S |
| 10:27.01 | knight_ | Salvage: 2449 => 24491(Angola - Mobile/Special Services => Angola-Mobile) |
| 10:27.01 | knight_ | <PROTECTED> |
| 10:27.03 | knight_ | there we go!!! |
| 10:27.44 | sofh | anybody active there ...? |
| 10:27.57 | knight_ | sofh, what are you trying to do? |
| 10:28.45 | sofh | i'ev a CDMA phone..its wireless both for data and voice |
| 10:29.05 | sofh | i want to integarate it with asterisk... |
| 10:29.16 | knight_ | Be more specific... How do you want to integrate it? |
| 10:29.19 | sofh | may be as FXS or FXO even ? |
| 10:29.36 | knight_ | Your chances of doing that is unlikely. |
| 10:29.59 | sofh | hmm.. |
| 10:30.07 | knight_ | You'd need atleast a way to get a dialtone via mic/headphone jack to use as an analog FXS |
| 10:30.18 | knight_ | Or a custom USB driver |
| 10:30.22 | knight_ | which is realllllly slim |
| 10:30.36 | sofh | but its usb driver is installed in linux |
| 10:30.47 | sofh | it gets it automatically.... |
| 10:32.06 | knight_ | right, but that usb driver probably doesnt have audio in/out and phone control available easily |
| 10:32.39 | sofh | true... |
| 10:32.41 | knight_ | you could write a custom asterisk device module if that device makes a /dev/dsp* available, and some sort of control interface |
| 10:32.54 | sofh | i also notice it is installed as A MODEM Only..no voice modem is there.. |
| 10:32.58 | knight_ | yep |
| 10:33.13 | sofh | its in dev like /dev/ttyACM0 as its usb interface |
| 10:33.22 | knight_ | most phone's usb drivers allow you to control very little |
| 10:33.50 | knight_ | what kind of phone is it? |
| 10:33.52 | Tili | knight_: there are plenty of GSM to FXO devices available. there should be same for CDMA somewhere |
| 10:34.01 | knight_ | Tili, are there? |
| 10:34.03 | Tili | like Nokia Premicell |
| 10:34.07 | sofh | it doesnt conatin any SIM card... |
| 10:34.20 | sofh | its axes telecom's CDMA phone |
| 10:34.51 | Tili | knight: yeah plenty of those stuff. they have SIM cards. premicells take 2 cards. there was one with 4 sim cards with ISDN interface |
| 10:35.06 | Tili | let me check boookmarks |
| 10:35.22 | sofh | if it had some GSM phone..then i must find some GSM GW and interface card like E1 .. |
| 10:35.34 | knight_ | right, but that's not an interface between a cell phone and asterisk |
| 10:35.36 | knight_ | it's a cell phone itself |
| 10:36.14 | sofh | so now according to you..the nearest possiblity is to get the voice from CDMA via sound card ? |
| 10:36.49 | knight_ | sofh, with an EXISTING cell phone device, yes |
| 10:36.59 | knight_ | and even then, i'm not sure how you plan to do that |
| 10:37.01 | Tili | well if you want to use your own phone than its tough. but using wireless GW is easier |
| 10:37.14 | knight_ | or how you plan to control the phone to dial, answer, etc |
| 10:37.45 | sofh | hmm..thats realy going kinda typical |
| 10:38.28 | sofh | i must forget it :( |
| 10:38.59 | sofh | actually in my area..CDMA is offering free calls (local) to other CDMA phone of same company |
| 10:39.10 | sofh | so i though if we can integarte CDMA phone with asterisk..everything is done then :) |
| 10:39.16 | sofh | but this usb interface is ...just like.. |
| 10:39.36 | knight_ | sofh, get a CDMA to FXO interface like Tili mentioned |
| 10:39.51 | knight_ | you wont be able to use your existing phone, but you can use this device which is a phone of its own |
| 10:40.23 | sofh | sorry i didn't get...would you please explain a little more ? |
| 10:40.39 | Tili | sofh: it would be tough to use your phone like that. |
| 10:40.56 | sofh | yeah i understand that...becasue of only usb interface is there.. |
| 10:41.16 | Tili | sofh: there are devices that can take your SIM card and give you an FXO out interface which you can put into asterisk's TDM cards etc |
| 10:41.26 | knight_ | some phones would work... like PDA phones and other smartphones |
| 10:41.34 | *** join/#asterisk pa (n=Paolo@unaffiliated/pa) |
| 10:41.39 | knight_ | you could write/install a SIP server on it |
| 10:42.07 | Tili | another way would be to get a CDMA to SIP gateway. |
| 10:42.15 | Tili | knight_: you really hate sofh |
| 10:42.48 | knight_ | ;) |
| 10:43.01 | sofh | not realy Tili :P |
| 10:43.16 | sofh | any way thanks guys.. |
| 10:43.24 | sofh | hope to get see you again.. |
| 10:43.26 | sofh | T~C |
| 10:43.27 | sofh | bbye |
| 10:43.33 | Tili | again its same thing as Wireless GW but instead of FXO you use ethernet |
| 10:43.49 | knight_ | it would be 100 times as expensive and time consuming |
| 10:44.06 | knight_ | as the CDMA/GSM-to-FXO method |
| 10:50.08 | knight_ | anyway, back to writing this rate card conversion code :) |
| 10:50.15 | Tili | knight_: its not very expensive. its like 300 dollars from EBay. i once saw a premicell but didnt buy it as that project went bust also |
| 10:50.34 | knight_ | Tili, ahh these are products that went bust? |
| 10:50.38 | Tili | it had 2 sims |
| 10:51.16 | Tili | good luck |
| 10:51.36 | Tili | no our project went bust. those devices are fine |
| 10:52.03 | knight_ | I also cant imagine that most cell companies allow you to pass traffic commercially |
| 10:52.07 | knight_ | i.e. reselling |
| 11:00.12 | *** join/#asterisk zotz (n=zotz@24.231.47.168) |
| 11:07.30 | *** join/#asterisk expat_iain (n=expat_ia@194.204.99.131) |
| 11:07.52 | frix | sry, must ask once more... |
| 11:08.57 | frix | i have an teledat2 a/b connected to the hfc-s. if i plug in an isdn phone also, my analog phones dont work anymore. they get hungup immediately when i call them |
| 11:09.23 | frix | on internal as well as incoming calls |
| 11:10.25 | *** join/#asterisk pif (n=ldm@zenon.apartia.fr) |
| 11:14.27 | *** join/#asterisk SoloFlyer (n=soloflye@59.167.146.54) |
| 11:14.29 | *** join/#asterisk pa (n=Paolo@unaffiliated/pa) |
| 11:14.40 | SoloFlyer | Hiya |
| 11:14.49 | KriS83 | I have a stupid question... the number I dial is saved in ${EXTEN} in which var is the extension saved that I am calling from? |
| 11:15.36 | SoloFlyer | the callid being displayed or the actual extension? |
| 11:15.57 | KriS83 | the actual extension |
| 11:16.04 | KriS83 | I want to set the callerid |
| 11:16.19 | KriS83 | depending on the extension that is making the call |
| 11:16.37 | SoloFlyer | ahh |
| 11:16.39 | SoloFlyer | i see |
| 11:17.19 | cypromis | <PROTECTED> |
| 11:17.21 | cypromis | :) |
| 11:17.38 | KriS83 | ? |
| 11:17.52 | KriS83 | thats the answer? /w 7 ? meaning? |
| 11:18.00 | SoloFlyer | no thats not the answer |
| 11:18.03 | KriS83 | k |
| 11:18.04 | KriS83 | :) |
| 11:18.05 | SoloFlyer | im looking it up |
| 11:18.09 | SoloFlyer | gimme 5 |
| 11:18.13 | KriS83 | oh.. thanks |
| 11:18.15 | KriS83 | sure |
| 11:18.30 | SoloFlyer | i cant remember it exactly but i have it |
| 11:18.34 | SoloFlyer | i think... |
| 11:18.52 | SoloFlyer | exntesions.conf it tooo biiigg!!! |
| 11:19.24 | KriS83 | yeah same here |
| 11:21.36 | KriS83 | And then another question... is it possible to ask for a input within a queue, else kick caller from queue? I know I can have it so that if a caller presses a key he is transfered to "content=" but I need it the otherway round basicly. If caller presses "$key[1-9] stay in queue |
| 11:23.54 | SoloFlyer | have a inital que |
| 11:24.03 | SoloFlyer | if they press a button send them to the wait que |
| 11:24.14 | KriS83 | hmm.. good point... |
| 11:24.17 | SoloFlyer | if they dont kickem out of que to reception |
| 11:24.37 | SoloFlyer | you might be able to do what you asked but i dont know... |
| 11:24.50 | KriS83 | Can I kick a caller if he is in the queue $X minutes? |
| 11:24.59 | SoloFlyer | yeah |
| 11:25.15 | SoloFlyer | but i cant remember the command |
| 11:25.16 | SoloFlyer | lol |
| 11:25.22 | *** join/#asterisk nesys (n=nesys@2001:1418:1a6:0:20d:93ff:fe28:3ef8) |
| 11:25.30 | KriS83 | hmm k... I'll checkup on that then |
| 11:25.36 | pif | any happy 1.2+misdn user around? |
| 11:25.38 | KriS83 | the idea is good anyway.. thanks for that |
| 11:27.25 | SoloFlyer | actually its a bit of a work around... if i remember correctly |
| 11:27.35 | SoloFlyer | you use timeout from something else... |
| 11:27.59 | SoloFlyer | IIRC |
| 11:29.35 | *** join/#asterisk Brumle (n=brumle@brumle.com) |
| 11:37.21 | *** join/#asterisk _4d4m_ (n=adam@212-14-101-159.adsl.legend.co.uk) |
| 11:37.26 | *** join/#asterisk remibreval (i=Remek@APuteaux-153-1-35-171.w82-124.abo.wanadoo.fr) |
| 11:37.30 | remibreval | Hello everyone |
| 11:37.39 | remibreval | I have simple question about asterisk console |
| 11:37.53 | remibreval | to debug I tap : asterisk -vvvvvvvr |
| 11:38.01 | remibreval | I use sip debug and stuff like that |
| 11:38.18 | remibreval | when all is OK, should I do something or is ok if I just log out ??? |
| 11:41.53 | *** join/#asterisk RENZ0T (i=renzo_ac@200.60.63.197) |
| 11:51.39 | SoloFlyer | KriS83: i cant find it |
| 11:51.40 | SoloFlyer | sorry |
| 11:52.19 | SoloFlyer | i cant find it anywhere... not on my boxes and not on the internet... |
| 11:53.06 | SoloFlyer | but you could just setup special contexts for each user.. and then have that special context set a variable and the go to normal context |
| 11:53.19 | SoloFlyer | or you could write a patch :) |
| 11:55.13 | remibreval | how do you stop Asterisk ??! |
| 11:55.26 | remibreval | (without stopping computer) |
| 11:55.36 | remibreval | and with a clean method .... |
| 11:55.39 | SoloFlyer | lol |
| 11:55.47 | remibreval | :) |
| 11:55.51 | remibreval | do you have any idea ? |
| 11:56.04 | *** join/#asterisk pooh_ (n=hfwang@cust.15.241.adsl.cistron.nl) |
| 11:57.05 | SoloFlyer | a little reading for you http://www.voip-info.org/wiki/view/Asterisk+CLI :) |
| 11:57.23 | remibreval | arff, I'm stupid, I didn't find this topic on the wiki :-( |
| 11:57.25 | remibreval | thanks |
| 11:57.37 | SoloFlyer | but i think you are looking from stop now :) |
| 11:58.23 | SoloFlyer | it will be one of the stop commands anyway... |
| 11:58.34 | *** join/#asterisk ful|work (n=fulgas@209.8.233.106) |
| 11:58.55 | remibreval | yep exaclty |
| 11:59.04 | remibreval | thanks it's what I was searching for |
| 11:59.09 | SoloFlyer | :) |
| 11:59.09 | remibreval | by the way, after using asterisk -vvvvvvvr |
| 11:59.20 | remibreval | should I do something before loging out |
| 11:59.40 | SoloFlyer | like what... |
| 11:59.42 | remibreval | I mean, I use aste -vvv... to debug, but when all is ok, I should put back verbosity to 1 ? |
| 11:59.49 | remibreval | does it change perfs ? |
| 11:59.56 | SoloFlyer | oh no |
| 12:00.11 | SoloFlyer | nah |
| 12:00.13 | remibreval | Ok, so I just log out and it's ok :-) |
| 12:00.19 | SoloFlyer | yeah |
| 12:00.26 | remibreval | Great great great :-) |
| 12:00.49 | SoloFlyer | the -vvvvvvv only gives extra debugging information within the console |
| 12:01.02 | SoloFlyer | once you close the console it doesnt matter |
| 12:03.06 | remibreval | and asterisk just call the CLI... so when log out it's ok |
| 12:03.27 | SoloFlyer | yes |
| 12:03.41 | SoloFlyer | lol... Stress less :) |
| 12:04.12 | *** join/#asterisk coppice (n=chatzill@142.198.17.210.dyn.pacific.net.hk) |
| 12:06.26 | remibreval | exaclty !!! |
| 12:07.39 | SoloFlyer | KriS83: ${callingexten} <--- aparently |
| 12:07.42 | *** part/#asterisk jaike (i=aa@210.5.118.254) |
| 12:08.09 | remibreval | I go eat, see you !!! |
| 12:08.13 | SoloFlyer | bye |
| 12:08.21 | remibreval | bye |
| 12:09.21 | SoloFlyer | nm that is someone saying they would like that feature :) |
| 12:14.02 | SoloFlyer | it would be very easy to add the feature to asterisk... |
| 12:14.19 | SoloFlyer | but i currently dont have the required carefactor... |
| 12:14.20 | *** part/#asterisk RENZ0T (i=renzo_ac@200.60.63.197) |
| 12:23.00 | expat_iain | What is causing me to receive the messages |
| 12:23.06 | expat_iain | "Primary D-Channel on span 1 up" |
| 12:23.13 | expat_iain | ...once per second on console?? |
| 12:23.25 | expat_iain | It's driving me nuts |
| 12:24.08 | *** part/#asterisk nesys (n=nesys@2001:1418:1a6:0:20d:93ff:fe28:3ef8) |
| 12:26.05 | *** join/#asterisk transporter (n=transpor@66-23-211-29.clients.speedfactory.net) |
| 12:27.39 | JunK-Y | expat_iain: u see all the bchannels resetting too? |
| 12:27.55 | JunK-Y | prolly ur resetinterval in ur zapata.conf |
| 12:29.52 | KriS83 | SoloFlyer, I'll try that thx |
| 12:33.03 | *** join/#asterisk gnosys (n=ksford@ip68-9-201-250.ri.ri.cox.net) |
| 12:33.28 | *** join/#asterisk amir_ (n=amir@gentoo/developer/amir) |
| 12:33.54 | *** join/#asterisk olivier_ (n=olivier_@obs92-4-82-239-116-113.fbx.proxad.net) |
| 12:34.55 | SoloFlyer | cd Asteriskgeeks |
| 12:35.12 | SoloFlyer | arg wrong window :) |
| 12:35.34 | *** part/#asterisk MatsK (n=Administ@55.80-203-80.nextgentel.com) |
| 12:37.25 | gnosys | Anybody here run Asterisk 1.0.9-r2 from portage on Gentoo? |
| 12:37.27 | p1tst0p | hi, i have an asterisk box in a DMZ, behind a internet connection... i can connect IAX <--> IAX to my friends asterisk box in the states and get 2 way audio.. however, if my friend connects to my asterisk using X-Lite, we only get 1 way audio |
| 12:37.52 | SoloFlyer | xlite using sip...? |
| 12:38.18 | p1tst0p | Yeh SoloFlyer, im assuming this is Nat issues. |
| 12:38.21 | *** join/#asterisk RoyK (n=roy@213.160.242.93) |
| 12:38.56 | p1tst0p | and if so, does that mean IAX doesnt get effected |
| 12:38.58 | SoloFlyer | which way is the one way audio? |
| 12:39.17 | *** join/#asterisk r0d3nt (n=RatMan@tinfoilhat.net) |
| 12:39.20 | p1tst0p | he can hear me, and im on the same lan as the ast box |
| 12:39.32 | SoloFlyer | sip is very bad with nats |
| 12:39.36 | SoloFlyer | iax isnt |
| 12:39.47 | p1tst0p | whats a good IAX client, |
| 12:40.13 | SoloFlyer | tell sip its external ip |
| 12:40.21 | SoloFlyer | tell asterisk its external ip |
| 12:40.26 | SoloFlyer | arg |
| 12:40.33 | SoloFlyer | tell asterisk its external SIP ip |
| 12:40.36 | expat_iain | Junk-Y: Nope, just the D channel |
| 12:40.38 | SoloFlyer | yay got it :) |
| 12:40.40 | remi-eating | SoloFlyer, I have solved this with canreinvite=no |
| 12:41.07 | remi-eating | the only pb, is that with X-lite, when I'm on same lan, it works only if I have canreinvite=yes... |
| 12:41.24 | remi-eating | So I have to change th config in fuction i'm outside or inside the same lan... |
| 12:41.46 | *** part/#asterisk _4d4m_ (n=adam@212-14-101-159.adsl.legend.co.uk) |
| 12:41.49 | SoloFlyer | remi its because the asterisk box isnt putting the correct ip address in the sip packets |
| 12:41.52 | p1tst0p | remi-eating, already got canreinvite=no set for the sip client |
| 12:42.28 | *** join/#asterisk _4d4m_ (n=adam@212-14-101-159.adsl.legend.co.uk) |
| 12:42.32 | remib | SoloFlyer, yep, but as far as I put nat=yes, it should do something more intellignent no ??? |
| 12:42.39 | SoloFlyer | no |
| 12:42.46 | p1tst0p | SoloFlyer, do you mean set the bindaddr in sip.conf ? |
| 12:42.54 | p1tst0p | SoloFlyer, which is currently set to 0.0.0.0 |
| 12:43.03 | SoloFlyer | no leave that at 0.0.0.0 |
| 12:43.15 | SoloFlyer | is the asterisk box directly connected to the internet |
| 12:43.15 | remib | I can't understand why in remote lan (I mean, not lan of asterix) it ONLY works between 2 X-lite with canreinvite=yes, and no audio with canreinvite=no... |
| 12:43.52 | p1tst0p | SoloFlyer, the Ast box is in a DMZ |
| 12:43.59 | SoloFlyer | ok |
| 12:44.14 | SoloFlyer | gimme a sec |
| 12:45.05 | p1tst0p | no problem mate |
| 12:45.47 | remib | p1st0p, in sip.conf externip=your_public_IP |
| 12:45.47 | remib | localnet=your_lan_IP (I have 192.168.0.0/255.255.255.0) |
| 12:46.21 | sivana | is app_groupcount.so now obselete with SVN HEAD? |
| 12:46.50 | SoloFlyer | set externip=232.321.32.321 or what ever yer external ip is |
| 12:47.00 | SoloFlyer | in sip.conf for his logon |
| 12:47.11 | SoloFlyer | and leave nat=yes :) |
| 12:47.29 | p1tst0p | ok lets try this |
| 12:47.57 | SoloFlyer | no sorry |
| 12:48.04 | SoloFlyer | put it under the general config |
| 12:48.14 | SoloFlyer | under general in sip config |
| 12:48.20 | SoloFlyer | and nat=yes in his config |
| 12:48.24 | gnosys | Ok, so nobody runs Asterisk 1.0.9-r2 from portage on Gentoo...... Does anybody here run Asterisk on Gentoo at all? |
| 12:48.32 | *** join/#asterisk MrEntropy (n=entropy@ppp230-139.lns2.adl4.internode.on.net) |
| 12:48.36 | SoloFlyer | Debian :) |
| 12:48.40 | MrEntropy | yo |
| 12:48.46 | MrEntropy | what version of zaptel will compile under gcc 4.0.2? zaptel 1.0.10 does not |
| 12:48.47 | p1tst0p | gnosys mines on Gentoo |
| 12:48.52 | p1tst0p | from CVS mind |
| 12:49.12 | gnosys | p1st0p... do you do anything special in building in from CVS? |
| 12:49.22 | gnosys | building *it* |
| 12:49.38 | *** join/#asterisk Zach^^ (i=chaos@dialup-4.224.84.174.Dial1.Cincinnati1.Level3.net) |
| 12:49.53 | remib | when canreinvite works well ? No chance behind NAT ? |
| 12:50.08 | SoloFlyer | i have no idea why canreinvite works for you lol |
| 12:50.14 | blop | hum, i got this on some incoming calls on an fxo :-- Starting simple switch on 'Zap/4-1' NOTICE[6439]: chan_zap.c:6227 ss_thread: Got event 18 (Ring Begin)... NOTICE[6439]: callerid.c:322 callerid_feed: Caller*ID failed checksum NOTICE[6439]: chan_zap.c:6227 ss_thread: Got event 2 (Ring/Answered)... , any clue ? |
| 12:50.16 | SoloFlyer | that isnt what canreinvite is ment to do :) |
| 12:51.02 | Zach^^ | anyone here that can help me with voipjet? |
| 12:51.04 | SoloFlyer | looks like asterisk doesnt like the caller*id it gets |
| 12:51.16 | remib | it should make a P2P connection no ?? |
| 12:51.24 | KriS83 | SoloFlyer, that was not it.. ${callingexten} is empty :( |
| 12:51.35 | Romik_ | Zach^^: email fastsupport@voipjet.com they will answer...in 24 hours.. |
| 12:51.44 | SoloFlyer | kris said about 5 lines after i said that |
| 12:52.03 | Zach^^ | Romik_ i have and no reply 48hrs later |
| 12:52.18 | Romik_ | zach: so wait...they answer. |
| 12:52.28 | SoloFlyer | (22:41:02) SoloFlyer: nm that is someone saying they would like that feature :) |
| 12:52.44 | knight_ | voipjet hardly replies FAST :) |
| 12:52.45 | SoloFlyer | sorry... |
| 12:52.52 | Zach^^ | Romik_ i am getting all circuits busy when i try to dial out.... is that my problem or on there end? |
| 12:53.00 | remib | At least, canre.. shoud try to make a P2P connection. But It should cancel if no signal (I mean, I would be great if it was like that, something like : canreinvite=try :-) |
| 12:54.02 | SoloFlyer | remib canreinvite will work if asterisk doesnt and both clients know their correct external ip but asterik doesnt know its correct external ip |
| 12:54.21 | SoloFlyer | but its still not what canreinvite is suppose to be fore |
| 12:54.50 | SoloFlyer | its there to reduce latency induced by asterisk proxying the connection |
| 12:54.57 | Romik_ | zach: i spend with them near $250 a week |
| 12:55.17 | Romik_ | zach: i do not hear about from them |
| 12:55.38 | Romik_ | zach: about problem....we terminate all |
| 12:55.48 | Romik_ | zach: all north american calls via them |
| 12:56.04 | remib | Solo, yep, both reduce latency and also footprint of your server no ? Because if you have several call via *, it could saturate it... |
| 12:56.10 | remib | and the bandwith... |
| 12:57.26 | SoloFlyer | proxying calls doesnt use many resources other than bandwidth.... unless it has to transcode or do something special with the stream |
| 12:57.55 | SoloFlyer | but yeah it does reduce load on server :) |
| 12:58.28 | SoloFlyer | KriS83: are you there? |
| 12:59.26 | remib | If I have 2 remote LAN, I should put 2 asterisk box so ?? |
| 12:59.33 | p1tst0p | SoloFlyer still one way audio. hm |
| 12:59.46 | knight_ | iax2 iax2 iax2 |
| 13:00.10 | SoloFlyer | knight yes but he wants to support sip aswell |
| 13:00.18 | remib | p1tst0p, use SIP DEBUG and pastebin it... |
| 13:00.31 | knight_ | then he needs either a public IP, or a SIP Proxy |
| 13:00.39 | knight_ | siproxyd works great |
| 13:00.45 | knight_ | for outbound sip |
| 13:00.53 | knight_ | over nat |
| 13:00.54 | p1tst0p | SoloFlyer, sorted it |
| 13:01.00 | knight_ | but siproxyd needs to be outside the nat |
| 13:01.07 | SoloFlyer | knight stop |
| 13:01.08 | knight_ | on the firewall |
| 13:01.13 | SoloFlyer | STOP! |
| 13:01.28 | *** join/#asterisk amir_ (n=amir@gentoo/developer/amir) |
| 13:01.50 | remib | Ant STUN ? |
| 13:01.57 | p1tst0p | SoloFlyer, it was the localnet= that someone previousley suggested ! |
| 13:02.06 | KriS83 | SoloFlyer, yes I'm here |
| 13:02.12 | Ahrimanes | is there any way to control what asterisk does when a # is sent via dtmf? |
| 13:02.13 | knight_ | Sorry, didn't mean to steal your fire. |
| 13:02.41 | SoloFlyer | no you were just stating the obvious, it was annoying |
| 13:03.08 | knight_ | About 10 lines up you were stating the obvious too. This guy needs to read the damn Wiki. |
| 13:05.19 | blop | does 'astagidir => /usr/share/asterisk/agi-bin' (asterisk.conf) still works in 1.2 ? it seems to be ignored |
| 13:06.20 | remib | Do you think Stun is a good solution for Nat pb ? or Iax2 is more elegant (if supported by phone) |
| 13:06.56 | Ahrimanes | stun is much more widespread.. and works well |
| 13:07.37 | SoloFlyer | iax2 wrox |
| 13:07.58 | SoloFlyer | but sip is more widespread |
| 13:07.59 | Ahrimanes | no doubt, but a real shortage on phones supporting it |
| 13:08.34 | remib | yep oki :) |
| 13:08.46 | SoloFlyer | everyone that supports iax2 supports sip (it it feels like it anyway) |
| 13:09.33 | *** join/#asterisk fugitivo (n=ajf@209.13.244.233) |
| 13:09.44 | remib | yep, but the opposite is not true !! |
| 13:09.49 | SoloFlyer | yeah |
| 13:09.50 | fugitivo | morning |
| 13:10.02 | SoloFlyer | afternoon |
| 13:10.15 | SoloFlyer | well technically.. its morning here |
| 13:10.16 | SoloFlyer | lol |
| 13:10.24 | knight_ | here too |
| 13:10.26 | knight_ | 5am |
| 13:10.32 | fugitivo | 5am? |
| 13:10.43 | fugitivo | whay are you doing awake? |
| 13:11.33 | knight_ | making a lot of progress on some code i'm writing |
| 13:11.36 | SoloFlyer | <PROTECTED> |
| 13:11.39 | knight_ | otherwise normally i'm in bed |
| 13:11.49 | SoloFlyer | :) |
| 13:11.56 | remib | Ok, thanks. It is more clear now for me |
| 13:12.17 | knight_ | :) |
| 13:12.34 | SoloFlyer | they need to make sip2... |
| 13:12.39 | remib | lol |
| 13:12.59 | SoloFlyer | NAT friendly :) |
| 13:13.18 | knight_ | funny that you say that, some broken code in astbill tried to access SIP2/ |
| 13:13.34 | SoloFlyer | really? |
| 13:13.40 | fugitivo | nice :) |
| 13:13.52 | knight_ | hah yeah |
| 13:14.08 | *** join/#asterisk wunderkin (i=kev@ip68-226-113-228.ph.ph.cox.net) |
| 13:14.47 | SoloFlyer | i think i mean SIP3 |
| 13:15.10 | remib | well in france it is 2 PM |
| 13:15.23 | *** join/#asterisk r0d3nt|m (n=RatMan@tinfoilhat.net) |
| 13:15.44 | lme | 3 hours to go ! |
| 13:16.37 | remib | LOL : http://www.google.fr/search?q=SIP3&hl=fr&rls=GGLD%2CGGLD%3A2005-05%2CGGLD%3Afr |
| 13:17.00 | SoloFlyer | :) |
| 13:17.05 | SoloFlyer | ok SIP4 ? |
| 13:18.20 | *** join/#asterisk amir_ (n=amir@gentoo/developer/amir) |
| 13:18.52 | SoloFlyer | i cant belive that they hardcoded ip addresses into sip |
| 13:19.06 | SoloFlyer | its such and amature thing to do |
| 13:21.12 | fugitivo | what do you mean with ip addresses into sip? |
| 13:21.29 | heroine | I'm looking throught asterisk-1.2 for voicemail mysql support but from what i can understand it seems that mysql support moved to an odbc support .. is that right .. ? or i miss understanding something ? |
| 13:21.51 | SoloFlyer | sip packets have the source/destination ip addresses in the packet data |
| 13:22.07 | KriS83 | heroine, I'd say it's RealTime support |
| 13:22.15 | KriS83 | Search for REaltime and voicemail |
| 13:22.27 | knight_ | realtime+++++ |
| 13:22.28 | *** join/#asterisk [TK]D-Fender (n=joe@toronto-HSE-ppp4122655.sympatico.ca) |
| 13:22.57 | *** join/#asterisk znoG (n=gs@OL101-122.fibertel.com.ar) |
| 13:24.26 | heroine | KriS83: thanx for the tip . will take a look at that |
| 13:28.40 | KriS83 | np |
| 13:31.43 | *** part/#asterisk SoloFlyer (n=soloflye@59.167.146.54) |
| 13:33.07 | *** join/#asterisk areski (n=areski@113.Red-83-55-99.dynamicIP.rima-tde.net) |
| 13:34.13 | docelmo | Good Moring #asterisk!!!!!!!!!!!!1 |
| 13:34.50 | *** join/#asterisk arguile (i=user224@66.38.201.234) |
| 13:35.01 | docelmo | or not. |
| 13:40.51 | *** join/#asterisk PoWeRKiLL (n=PoWeRKiL@84.205.154.241) |
| 13:42.22 | MassiveBlue | good morning, my local time is 14:42 :) |
| 13:46.23 | *** join/#asterisk amir_ (n=amir@gentoo/developer/amir) |
| 13:47.45 | *** join/#asterisk pa (n=Paolo@unaffiliated/pa) |
| 13:51.43 | docelmo | Hay bub.. it might be almost 3pm your time but its almost 9am GMT-5 here |
| 13:52.55 | *** join/#asterisk Assid (n=assid@203.115.64.62) |
| 13:53.46 | *** join/#asterisk postwait (n=jesus@catapult.omniti.com) |
| 13:53.56 | postwait | Anyone here use voipjet? |
| 13:54.18 | postwait | I have an old IP in my asterisk set up and it no longer works. |
| 13:54.35 | postwait | Looking for voipjet's new york IP address and cant' find it online |
| 13:54.46 | *** join/#asterisk locksy (n=locksy@mrtg.sisgroup.com.au) |
| 13:55.17 | *** join/#asterisk Spacebar (n=stingray@stingr.net) |
| 13:55.30 | *** join/#asterisk grimse (n=grimse@p5481F8CD.dip.t-dialin.net) |
| 13:55.42 | Spacebar | hmm anybody can answer newbie queue question? :) |
| 13:55.58 | [TK]D-Fender | shoot |
| 13:56.16 | Spacebar | I'm trying to use queue without Agents (adding dynamic members directly) |
| 13:56.46 | Spacebar | but this way it distributes second calls to members |
| 13:56.58 | Spacebar | i.e. second call is ringing while 1st in progress |
| 13:57.06 | pooh_ | postwait: 216.118.117.46 |
| 13:57.20 | [TK]D-Fender | Hmmm, seems to be a problem with knowing if an agent is on the phone already |
| 13:57.35 | [TK]D-Fender | using a call-back context right? |
| 13:57.53 | Spacebar | [TK]D-Fender: no. I'm not using agent channel at all |
| 13:58.09 | [TK]D-Fender | how does it add? Direct tech/#? |
| 13:58.17 | postwait | pooh_: That IP no workie for me. |
| 13:58.21 | Spacebar | i'm using "add queue member IAX2/310 to call01" |
| 13:58.28 | postwait | That's what I had... it went silent about 6 days ago. |
| 13:58.41 | [TK]D-Fender | Spacebar : ok, something I haven't tried yet.. sorry :/ |
| 13:58.48 | postwait | I've been crawling mailing lists and found: 66.246.220.19 |
| 13:58.55 | postwait | which works and has 20ms of latency for me. |
| 13:59.03 | postwait | Just want to sanity check it... |
| 13:59.05 | Spacebar | show queues say "In use" on that members so it detects it as in use |
| 13:59.46 | docelmo | postwait, its on their site |
| 13:59.56 | *** join/#asterisk synthetiq (n=roger@64.201.13.50) |
| 14:00.07 | Spacebar | but inside app_queue.c there are some obfuscated logic containing "stillgoing" parameter |
| 14:00.21 | Zach^^ | how can i make an extension fwd to and outside line? |
| 14:01.27 | znoG | postwait: i'm using 64.34.45.100 for voipjet |
| 14:02.54 | Katty | znoG: exten => extensionyouwanttouse,1,Dial(Zap/g1/wwSomethingPhoneNumber) |
| 14:03.10 | Zach^^ | and someone help me setup voipjet with amp? |
| 14:03.13 | Katty | ^ Zach^^ |
| 14:04.05 | Zach^^ | Katty |
| 14:04.18 | Katty | Zach^^: see above. |
| 14:04.27 | Katty | Zach^^: accidently went to znoG |
| 14:05.15 | Zach^^ | Katty but i need to get voipjet to work first.... |
| 14:05.28 | Katty | Zach^^: then copy down that line and get back to it later. |
| 14:05.41 | Katty | Zach^^: if you didn't want it, you shouldn't have asked for it. |
| 14:06.08 | Zach^^ | when i try to dialout i get the error all circuits arebusy |
| 14:08.33 | lme | anybody with junghanns's cards experience here ? |
| 14:08.44 | *** join/#asterisk negatendo (n=negatend@c-24-9-136-152.hsd1.co.comcast.net) |
| 14:08.55 | KriS83 | lme, I have some.. but I wouldn't call it "experiance" :) |
| 14:09.27 | lme | kris83 : got problems with quadbri and layer 1 activation here... |
| 14:09.52 | KriS83 | I have 2 quadBRI setup and running |
| 14:09.58 | lme | I can pass & receive calls, but layer one always goes down between calls |
| 14:10.23 | *** join/#asterisk jahani (n=k@adsl-175-47-192-81.adsl.iam.net.ma) |
| 14:11.00 | *** join/#asterisk heison (n=heison@ns.somanetworks.com) |
| 14:11.55 | KriS83 | lme, I'm new to Asterisk, so I guess I'm no good help... |
| 14:12.08 | KriS83 | I'm fighting myself thru also |
| 14:12.47 | iCEBrkr | KriS83: worthless!! |
| 14:13.01 | docelmo | Can one restart udev w/o rebooting? |
| 14:13.12 | bkw__ | hup it |
| 14:13.19 | iCEBrkr | docelmo: I just rerun udevstart |
| 14:13.27 | bkw__ | or that :P |
| 14:13.31 | KriS83 | iCEBrkr, huh? |
| 14:13.45 | iCEBrkr | KriS83: I was give'n ya shit :P |
| 14:13.47 | docelmo | Im running Redhat tho |
| 14:13.50 | tzafrir_laptop | KriS83, what distro? |
| 14:13.59 | KriS83 | tzafrir_laptop, CentOS |
| 14:14.16 | iCEBrkr | docelmo: So? udevstart should still be an executable. |
| 14:14.23 | iCEBrkr | docelmo: It looks SuSE like, I know. |
| 14:14.51 | KriS83 | I tried Debian, but seems the BriStuff don't like Debian. |
| 14:15.32 | docelmo | Kickass dude.. |
| 14:15.36 | docelmo | thanks |
| 14:15.51 | lme | kris83 : which version of bristuff r u using ? |
| 14:15.51 | iCEBrkr | docelmo: you mean I actually helped you with something for a change? |
| 14:16.06 | KriS83 | lme, latest available |
| 14:16.11 | *** join/#asterisk amir_ (n=amir@gentoo/developer/amir) |
| 14:16.15 | KriS83 | for * 1.2.0 |
| 14:16.18 | iDunno | the debian version of asterisk in sarge has the bristuff in it already. |
| 14:16.28 | iDunno | but that's 1.0.7 rather than 1.2.0 |
| 14:16.42 | iDunno | there's 1.2.0 in debian unstable, not checked if that's bristuff'd |
| 14:17.23 | docelmo | More or less.. Now that your head will swell.. |
| 14:17.35 | iCEBrkr | docelmo: haha, naww man, I ain't like that |
| 14:17.35 | docelmo | Ztdummy now workS!!! |
| 14:17.40 | Katty | yay! |
| 14:17.44 | *** join/#asterisk bjohnson (n=bjohnson@jecinc.tor.istop.com) |
| 14:18.33 | lme | kris83: 0.3.0-PRE-1c |
| 14:18.46 | docelmo | if it only wasnt december I would have my TDM11B for my house.. |
| 14:18.52 | docelmo | ohh hay Katty.. Whats shaking? |
| 14:18.52 | *** join/#asterisk lehel (n=lehel@82.79.20.17) |
| 14:19.00 | lehel | hey |
| 14:19.08 | lme | KriS83: and got no problems with your layer1 ? it's seems that mine looks like a yoyo |
| 14:21.00 | docelmo | all good.. Im just recompiling Asterisk for the Zaptel options.. I just wish I could get my damn SPA3000 to work correctly.. Inbound works of all things.. But I cant make a call out.. Keeps bitching about authentication.. Zaptel cards are SO much easier to install |
| 14:21.01 | *** join/#asterisk kimosabe (n=kimosabe@201.135.10.173) |
| 14:21.05 | KriS83 | lme, nop |
| 14:21.25 | docelmo | Anyone wanna buy a SPA2000 and 3000 and help me fund my move towards a TDM411B? |
| 14:21.27 | lme | kris83: damn... WHich signalling r u using to connect to your telco ? |
| 14:21.52 | iCEBrkr | docelmo: No thanks. I got SPAs coming out my ass.. |
| 14:21.54 | lme | i must put pri_cpe_ptmp here. But it's sounds funny to me to do multipoint... |
| 14:22.11 | KriS83 | lme, signalling = bri_cpe_ptmp |
| 14:22.20 | lme | well... :) |
| 14:22.29 | docelmo | No.. you.. Anyone.. :) |
| 14:22.38 | iCEBrkr | :) |
| 14:22.55 | kimosabe | does any one know what 30 did cost with t-1 for resels of phone numbers |
| 14:23.36 | iCEBrkr | docelmo: Thing is, I'm not sure how it all happened. I have 4 SPA2000's and I really only need 1. |
| 14:24.15 | iCEBrkr | http://www.ksta.de/ks/images/mdsBild/1132660403592l.jpg |
| 14:24.16 | iCEBrkr | Oh geesh |
| 14:24.20 | *** join/#asterisk javar (n=javar@69.79.133.185) |
| 14:24.22 | lme | bouhouhouuuuuuuuuu |
| 14:25.18 | docelmo | I wanna get my TDM board for my house then I will be happy as HELL! |
| 14:25.30 | KriS83 | lme, works? :) |
| 14:25.53 | lme | kris83 : no... I doesn't understant errors message.... |
| 14:26.11 | lme | KriS83: qozap: not re-activating layer1 span 0 |
| 14:26.39 | lme | KriS83: and... according to my zap config, my first span is 1.... |
| 14:26.47 | kimosabe | sip device 1 conects to asterisk box then asterisk box is always connected to iconnect acounnt registered and all i dial 9 then the number i wish 2 dial it says on comand line interface dialing 713xxxxx@iconnect.com then all it does is hang up and fast bussy |
| 14:26.56 | [TK]D-Fender | docelm0 : pastebin your sip & extensions and I might be able to help you with that... |
| 14:27.05 | lme | KriS83: if i receive a call, layer 1 span 0 goes up.... |
| 14:27.33 | znoG | kimosabe: which country are you from? |
| 14:27.37 | kimosabe | mexico |
| 14:27.38 | lme | KriS83: gonna shot myself and buy 100 analogs phones with 100 analogs lines from my telco |
| 14:27.48 | KriS83 | :) |
| 14:27.55 | znoG | kimosabe: ya me parecÃa, no entendà del todo lo que quisiste decir. Cual es el problema? |
| 14:28.41 | *** join/#asterisk puzzled (n=patrick@puzzled.xs4all.nl) |
| 14:28.55 | docelmo | I think the problem is in the SPA3000 but ok I will do it here in a sec.. In a conference now |
| 14:29.04 | [TK]D-Fender | np |
| 14:29.05 | puzzled | morning |
| 14:29.16 | kimosabe | znog i have a sipura that makes calls amongstmy other sipuras with no problem i have an iconnect acount configured in my asterisk box i want for my sipuras 2 take that acount from iconnect and use it |
| 14:29.21 | lme | ßðŋsdgfsdđgßdfŋ€«¶{t«er |
| 14:29.29 | zigman | what ? ;) |
| 14:29.30 | lehel | ;) |
| 14:29.38 | javar | hola znoG |
| 14:30.01 | lme | nothing, that was my head hitting my keyboard... |
| 14:30.39 | znoG | kimosabe: so you just need to configure your extensions.conf to do so |
| 14:30.57 | kimosabe | let me paste bin what i have in my extensions.conf |
| 14:32.03 | MassiveBlue | i get the following error when im calling from a misdn-channel to an iax2-channel "Dec 9 15:26:38 NOTICE[5760]: channel.c:1903 ast_read: Dropping incompatible voice frame on IAX2/4939329-3 of format alaw since our native format has changed to ulaw" |
| 14:33.06 | znoG | kimosabe: into a pastebin, I hope. :) |
| 14:33.08 | *** join/#asterisk paryl (n=paryl@216-201-177-82.res.logixcom.net) |
| 14:33.11 | MassiveBlue | sry, i mean "from a CAPI-Channel" |
| 14:33.28 | paryl | oh what a beautiful morning |
| 14:34.40 | Spacebar | [TK]D-Fender: well, Agent channel is strictly one-call (it gets busy when talking) while IAX2 just marked as "in use" |
| 14:34.48 | kimosabe | znog http://pastebin.com/455453 |
| 14:34.53 | paryl | is there a way to automatically log an agent out of a queue if they don't answer a call? |
| 14:35.21 | paryl | i thought a read about it a long time ago, but i can't find anything about it |
| 14:35.45 | Spacebar | [TK]D-Fender: but the problem stays when I transfer from agent to other extension I must do this via transfer channel too |
| 14:35.49 | Spacebar | heh |
| 14:36.33 | *** join/#asterisk Flauto (n=zhao@c-24-14-197-214.hsd1.il.comcast.net) |
| 14:36.39 | znoG | kimosabe: you probably want EXTEN:1, not EXTEN-1 |
| 14:37.43 | *** join/#asterisk RoyK (n=roy@55.80-202-161.nextgentel.com) |
| 14:37.47 | [TK]D-Fender | Spacebar : ok you are clearly working way out of my league :) |
| 14:37.57 | kimosabe | ok but EXTEN:1 must i declare it ? |
| 14:37.59 | KriS83 | Could someone help me on this problem: http://pastebin.ca/33092... Asterisk wants to forward me to the extension ${EXTEN} in [outbound]... and I don't know why :( |
| 14:38.16 | znoG | kimosabe: no. |
| 14:38.33 | *** join/#asterisk bjohnson (n=bjohnson@jecinc.tor.istop.com) |
| 14:38.50 | Spacebar | [TK]D-Fender: ;( |
| 14:38.59 | kimosabe | znog thanks |
| 14:39.09 | Katty | znoG: you remind me of eggnog. |
| 14:39.16 | KriS83 | All I want to do is achive that the caller from the Phone 069.... is connected to the extension he/she dialed, but with the CallerID given... |
| 14:40.21 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 14:41.04 | *** join/#asterisk pengyong (n=lala@222.185.22.73) |
| 14:41.41 | gnosys | recommendations on the best economy-to-mid-price-range hardware SIP phones for working with asterisk? I like my two Cisco 7960's thus far, but I only need that much functionality in a SIP phone for two desktops, whereas I need at least two other phones, maybe as many as 7 or 8 other economical phones, but want to make sure they work well with asterisk and have good firmware and configuration support. Thoughts? |
| 14:41.55 | [TK]D-Fender | KriS83 : whats with the "/06941903117" ? |
| 14:42.01 | Katty | [TK]D-Fender: hi! |
| 14:42.10 | Flauto | hey guys, when i make update under cvs, do i have to reconpile and install astersk and zaptel and libpri? |
| 14:42.22 | gnosys | yes. |
| 14:42.23 | [TK]D-Fender | Katty : y0 |
| 14:42.51 | Flauto | thanks |
| 14:42.56 | [TK]D-Fender | gnosys : Need PoE? 2nd lan port? |
| 14:43.19 | gnosys | Fender: huh? Need phones... |
| 14:43.28 | Katty | gnosys: what type of phones. |
| 14:43.38 | Katty | gnosys: power over ethernet... |
| 14:43.51 | Katty | gnosys: second lan port is very handy if you've got computers nearby |
| 14:43.54 | [TK]D-Fender | gnosys : I'm talking about specifig functionality. Are they to be plugged inline with a pc? use PoE or a power brick, etc.. |
| 14:43.56 | gnosys | don't need PoE (thanks for elaborating on that) |
| 14:44.08 | KriS83 | [TK]D-Fender, Thats to explictly say this extension is for the phone 06941903117 |
| 14:44.14 | gnosys | yes, 2nd lan port would be nice then... |
| 14:44.14 | Katty | gnosys: 2nd lan port is a bit like a fax....line into fax, line out from fax to something else. |
| 14:44.23 | [TK]D-Fender | I like my SPA-941. Polycom 501 is also a very good bet for you |
| 14:45.20 | KriS83 | But if I call that extension, it wants to transfer me to exten => ${EXTEN}... but I can't create an extension for every phonecall I want to make :) |
| 14:45.23 | *** join/#asterisk UlbabraB (n=caplaz@host241-43.pool8172.interbusiness.it) |
| 14:45.42 | gnosys | Polycom 501 for me? |
| 14:45.47 | [TK]D-Fender | yeah |
| 14:45.52 | gnosys | Ok. Thanks. |
| 14:45.55 | Katty | gnosys: those are nice ones :> |
| 14:46.05 | Katty | gnosys: with ulaw 8mono for ringtones :> |
| 14:46.07 | [TK]D-Fender | 601 > realllllyy nice phone |
| 14:46.16 | gnosys | Are they pricey? I'm on a budget for these phones. My 7960's were my high-end phones. |
| 14:46.17 | Katty | gnosys: we have polycom 500s here (= |
| 14:46.26 | Katty | gnosys: voip-supply has a price for you |
| 14:46.28 | [TK]D-Fender | Katty : I ripped off the Cisco "24" one for mine :) |
| 14:46.36 | Katty | gnosys: voip-supply.com i mean |
| 14:49.55 | iCEBrkr | Is it time to go home yet? |
| 14:50.05 | [TK]D-Fender | gnosys : 501's can be had for +/- $170 |
| 14:50.27 | *** join/#asterisk DrDeke (i=dekemar@deculator.engin.umich.edu) |
| 14:50.42 | [TK]D-Fender | http://voipstore.atacomm.com/Shops/ViewItem.aspx/27934028032-44253921024.htm |
| 14:50.50 | gnosys | [browsing voip-supply.com...] Hmmm... Looks like the 501 has way more functionality than I need for these phones... looks like it's a similar phone to the 7960... how does the 301 work with asterisk? I prolly only need 1-3 lines at most and limited functionality for these phones. |
| 14:51.24 | }btorch{ | is there a festival example config file somewhere in the festival install ? |
| 14:51.27 | }btorch{ | I can't find it |
| 14:51.56 | gnosys | are those wall- and desk-mount? |
| 14:52.28 | gnosys | thanks for the pointer Fender |
| 14:52.29 | *** join/#asterisk mesfet (n=mesfet@host130-204.pool82188.interbusiness.it) |
| 14:54.00 | mesfet | Hi. I'm trying a HFC isdn card with mISDN + asterisk... callings are working, but with NO AUDIO. Some ideas about? |
| 14:54.36 | gnosys | Hey, how does PoE work anyway? Do I need a special switch that sends the power and data over ethernet? And aside from that, it's the same as any other switch? Are there limitations on cable-type and cable-length? Pointers to reading material on this? |
| 14:55.46 | [TK]D-Fender | gnosys : You need a special switch or a PoE injector which plugs in-line. I never suggest injectors unless you only need a FEW ports |
| 14:56.02 | [TK]D-Fender | Apparently cat5+ works just fine |
| 14:56.18 | [TK]D-Fender | Look at the D-Link DES-1526 |
| 14:57.04 | [TK]D-Fender | gnosys : the price diff betweent he 301 and 501 isn't much, but the functionality gai is HUGE. |
| 14:57.06 | gnosys | My setup now has both server computers and client computers and ip phones plugged into a basic dumb D-Link switch. could I do the same if I had a special switch? the power over ethernet wouldn't cause problems for the computers? |
| 14:57.09 | [TK]D-Fender | gain* |
| 14:57.50 | [TK]D-Fender | gnosys : PoE (802.11af) auto-detects if the plugged device wants power and only provides if it does |
| 14:58.17 | gnosys | neato... |
| 14:58.33 | [TK]D-Fender | 301 has no speaker-phone, and a much more limited feature set (2 calls, etc). |
| 14:59.03 | *** join/#asterisk anthm (n=anthmct@CPE-69-76-83-52.wi.res.rr.com) |
| 14:59.03 | *** mode/#asterisk [+o anthm] by ChanServ |
| 14:59.10 | [TK]D-Fender | Almost too low end to consider unless you're going all polycom in a bigger layout and need several budget phones. What are they going to be used for? |
| 14:59.41 | gnosys | good points Fender |
| 15:00.03 | [TK]D-Fender | 802.3af... my bad... |
| 15:00.30 | DrDeke | Man, 802.11af would be sweet ;). Power over Wireless Ethernet! |
| 15:00.34 | gnosys | they'd be used for very short, one-on-one conversations, never a conference call. |
| 15:00.49 | *** join/#asterisk bjohnson (n=bjohnson@jecinc.tor.istop.com) |
| 15:01.02 | gnosys | DrDeke: I think they have that already... it's called a microwave oven! ;-) |
| 15:01.07 | DrDeke | lol yea |
| 15:01.22 | *** join/#asterisk ManxPower (n=ewieling@adsl-67-65-233-194.dsl.lgvwtx.swbell.net) |
| 15:01.25 | *** join/#asterisk klictel (n=klictel@207.107.208.137) |
| 15:01.56 | [TK]D-Fender | gnosys : well.... I'd still say either the 501 or SPA-941. the 941 is really nice.... |
| 15:02.07 | [TK]D-Fender | and will save you a little bit |
| 15:02.23 | gnosys | So, am I unlikely then to get below a $100 price point on a budget SIP phone? Looks that way from what I'm hearing. |
| 15:02.40 | *** join/#asterisk nagl (n=nagl@213.235.241.6) |
| 15:02.45 | DrDeke | gnosys: You can definitely get a budgetone 101 or 102 for less than $100, but the question is, even though you can do that; would you really want to? |
| 15:03.05 | [TK]D-Fender | "Just say no! (to BarbieTones!)" |
| 15:03.23 | [TK]D-Fender | GS = SUCK |
| 15:03.25 | gnosys | Well, I don't need much functionality... maybe only one line even... |
| 15:03.27 | *** join/#asterisk bjohnson_ (n=bjohnson@jecinc.tor.istop.com) |
| 15:03.42 | blitzrage | [TK]D-Fender: amen! |
| 15:03.55 | gnosys | GS??? |
| 15:04.01 | blitzrage | grandstream |
| 15:04.07 | [TK]D-Fender | I'd suggest a 301 over any GS. |
| 15:04.09 | gnosys | ok.. good2know |
| 15:04.38 | blitzrage | [TK]D-Fender: aye -- or Sipura (or, the new Cisco/Linksys (which is a Sipura repackaged)) looks really good -- only $150 |
| 15:05.16 | [TK]D-Fender | [10:01] <[TK]D-Fender> gnosys : well.... I'd still say either the 501 or SPA-941. the 941 is really nice.... |
| 15:05.21 | gnosys | how do i get firmware and config files and docs? with the cisco 7960's, i bought them new on eBay and was bummed to learn that I had to pay extra for a license from Cisco to get the firmware upgrades and config files. |
| 15:05.24 | [TK]D-Fender | I own one :) |
| 15:05.37 | lehel | destructure |
| 15:05.52 | [TK]D-Fender | Polycom firmware = free. They are like Cisco - the BS :D |
| 15:06.03 | gnosys | ??? |
| 15:06.16 | gnosys | my cisco firmware wasn't free.... |
| 15:06.19 | [TK]D-Fender | Cisco = Licensed costly gear |
| 15:06.34 | gnosys | not so for polycom? |
| 15:06.34 | blitzrage | [TK]D-Fender: free? I guess if you buy new... but it seems like a bitch to find (from Polycom -- you can find it on websites by searching google) |
| 15:06.35 | [TK]D-Fender | gnosys : I was talking about Polycom, sorry |
| 15:07.11 | [TK]D-Fender | blitzrage : its just that Polycom doesn't shove it on their website, they want you to go through a reseller (they don't want the hassle) |
| 15:07.14 | lehel | [TK]D-Fender, i wondered a solution would be between two eu countries, 2 * server, routed through iax2, with isdn cards, like fritz! and a tdm card? |
| 15:07.23 | blitzrage | [TK]D-Fender: gotcha |
| 15:07.51 | bjohnson_ | If these remind you of yourself, it's a good bet you are an engineer. |
| 15:07.56 | bjohnson_ | - At Christmas, it goes without saying that you will be the one to |
| 15:07.56 | bjohnson_ | find the burnt-out bulb in the string. |
| 15:07.57 | [TK]D-Fender | blitzrage : I have 4 SIP application version files here and 3 Bootrom :) |
| 15:08.06 | bjohnson_ | - The salespeople at Circuit City can't answer any of your questions. |
| 15:08.11 | [TK]D-Fender | blitzrage : Happy to pass your way if you need. |
| 15:08.18 | bjohnson_ | - You bought your wife a new CD ROM for her birthday. |
| 15:08.27 | bjohnson_ | - You can quote scenes from any Monty Python movie. |
| 15:08.39 | blitzrage | [TK]D-Fender: coolio -- I need to get my phone to boot off my tftp (again -- HD crashed after I got it working ;)) and the dialplan in the phone is f00ked (dials immediately when I dial 151) |
| 15:08.47 | bjohnson_ | - You have ever saved the power cord from a broken appliance. |
| 15:08.48 | blitzrage | [TK]D-Fender: please so! I have the IP500 |
| 15:08.54 | bjohnson_ | - You have more friends on the Internet than in real life. |
| 15:09.01 | lehel | any hint? |
| 15:09.11 | [TK]D-Fender | blitzrage : tftp = suck, ftp = good |
| 15:09.11 | ManxPower | ~doc |
| 15:09.13 | jbot | Documentation can be found at http://digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or http://www.voip-info.org/wiki-Asterisk or http://www.asteriskdocs.org or http://www.oreilly.com/catalog/asterisk |
| 15:09.14 | ManxPower | ~docs |
| 15:09.15 | jbot | somebody said docs was probably Documentation can be found at http://www.digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or http://www.voip-info.org/wiki-Asterisk or http://www.asteriskdocs.org or http://www.oreilly.com/catalog/asterisk or http://www.asteriskguru.com |
| 15:09.15 | DrDeke | gnosys: I had a BT102 for a while, but my cousin's wife demanded that I sell it to her so she can call her friends and family in Peru, Italy and Germany on the cheap. It "worked" but I was not overly impressed with it. It didn't actually crash on me like I have heard from some people, but it acts kind of flaky. |
| 15:09.17 | ManxPower | ~mailinglist |
| 15:09.18 | jbot | mailinglist is, like, Search Asterisk mailing lists by prepending site:lists.digium.com to your Google search. Browse the mailing list archive at http://lists.digium.com/ or search through it at http://www.asteriskguru.com/archives/search.php |
| 15:09.24 | blitzrage | uhhh... |
| 15:09.27 | bjohnson_ | - You're in the back seat of your car, she's looking wistfully at the |
| 15:09.27 | bjohnson_ | moon, and you're trying to locate a geosynchronous satellite. |
| 15:09.30 | [TK]D-Fender | blitzrage : What SIP version do you want? |
| 15:09.39 | blitzrage | [TK]D-Fender: whatever you recommend :) |
| 15:09.40 | bjohnson_ | - You've ever tried to repair a $5 radio. |
| 15:09.49 | bjohnson_ | that pretty much describes me |
| 15:09.53 | gnosys | thanks for that anecdote, DrDeke, about the BT102 |
| 15:10.18 | blitzrage | since we're flooding docs anyways... :) |
| 15:10.18 | blitzrage | ~thebook |
| 15:10.19 | jbot | extra, extra, read all about it, thebook is Asterisk: The Future of Telephony, released under the Creative Commons license and available at http://www.asteriskdocs.org << Read the book online! |
| 15:10.24 | Meaty | How can i change voicemail.conf for do not send email nofication on voicemail message ? |
| 15:10.33 | DrDeke | np |
| 15:10.50 | angler | Meaty, remove the email address |
| 15:11.18 | azzie | where can I get original .wav sounds from *, not .gsm ? need to recompress them to a different codec and not loose quality... |
| 15:11.27 | blitzrage | azzie: don't exist |
| 15:11.29 | Meaty | :S angler. .. Thx |
| 15:11.58 | lehel | a solution would be between two locations, 2 * server, routed through iax2, with isdn cards, like fritz! and a tdm card? |
| 15:12.07 | azzie | blitzrage, okey... so i gotta redo it from different prompts like cisco or quintum... thanks |
| 15:12.21 | blitzrage | yep |
| 15:13.03 | *** join/#asterisk stkn (n=stkn@gentoo/developer/pdpc.active.stkn) |
| 15:15.50 | gnosys | [TK]D-Fender: is that Polycom firmware you're talking with blitzrage about? for the 501? If it can be had through the "fair-use" clause here, then maybe it's not so bad that the firmware isn't free... which version do you like for the 501? maybe I'll pick up some of the 501's after all... |
| 15:17.09 | Flauto | anyone has any idea about web voicemail. i had no problem with it when i was using mandriva 2005 now, i use mandriva 2006, when i make webvmail, it seems working but i can not access it on the web, it is telling me server error |
| 15:17.12 | Flauto | any idea? |
| 15:17.41 | *** join/#asterisk santiago (n=santiago@208.195.215.160) |
| 15:18.20 | Flauto | when i look at the file under /var/www/cgi-bin, it is red |
| 15:21.32 | gnosys | I've noticed that the book, TFOT, is a tad out of date on some points... is the webvmail still a good make target with fully functional software installed? I haven't seen it discussed except in the book, TFOT. |
| 15:21.45 | gnosys | I mean in 1.2.1? |
| 15:22.24 | [TK]D-Fender | gnosys : the firmware IS free, its jsut the Polycom doesn't want you bugging them DIRECTLY for it. |
| 15:22.40 | gnosys | so how do you get it then? voip-supply.com? |
| 15:23.02 | [TK]D-Fender | gnosys : exactly or anyone else who has it. You ARE entitled to it, its jsut they don't want the effort. |
| 15:23.18 | gnosys | oh, i see... that is different from cisco... |
| 15:23.27 | DrDeke | In other words get a cheap VoIP account and keep calling them and yelling until they give it to you? :) |
| 15:23.49 | [TK]D-Fender | gnosys : they entire difference. Cisco - BS :) |
| 15:24.01 | gnosys | BS? Bullshit? |
| 15:24.25 | DrDeke | Oh hey speaking of VoIP... (Imagine that!): Are there any free-of-charge SIP or IAX clients for PocketPC/WindowsMobile? |
| 15:24.39 | Flauto | anyone would help me with the webvmail? |
| 15:24.52 | gnosys | anyone here using webvmail with a recent * version? |
| 15:25.35 | mog_home | make webvmail |
| 15:25.35 | Flauto | mog, i did |
| 15:25.35 | Flauto | but it is not working |
| 15:25.35 | gnosys | what version of * are you using Flauto? |
| 15:25.39 | Flauto | i am using cvs and just made an update |
| 15:25.48 | gnosys | cvs-head? |
| 15:25.48 | mesfet | Flauto, look at apache logs. |
| 15:26.01 | mesfet | Flauto, you'll need to permit cgi-bin from that directory |
| 15:26.13 | mog_home | that and do you have perl-suid |
| 15:26.21 | mishehu | cvs-head means that you give asterisk head. |
| 15:26.40 | gnosys | <chuckle> |
| 15:26.47 | [TK]D-Fender | [10:24] <gnosys> BS? Bullshit? <- INDEED |
| 15:26.51 | Flauto | how can i do it |
| 15:27.00 | gnosys | i thought they were using Subversion rather than CVS. |
| 15:27.23 | af_ | which way I can run asterisk and samba in the same pc? |
| 15:27.53 | gnosys | [TK]D-Fender, are you saying that you think Cisco's products are bullshit or that they policy about their software is bullshit? Just curious. I'm pretty happy with my 7960 phones, but I don't have much experience with SIP hardware phones so... |
| 15:28.12 | *** part/#asterisk pooh_ (n=hfwang@cust.15.241.adsl.cistron.nl) |
| 15:28.57 | DrDeke | gnosys: Not that you asked me, but of the Cisco products I have used (no VoIP), the products have worked extremely well, but the software/firmware/updates/etc have been extremely annoying and difficult to set up, and even more annoying and difficult to acquire. |
| 15:29.17 | gnosys | that's been my experience (though limited) also |
| 15:29.42 | [TK]D-Fender | gnosys : Hardware = good, plicy = suck |
| 15:29.59 | gnosys | I get the general impression that Polycom is very highly regarded on both counts in here though. True? |
| 15:30.20 | gnosys | Would anyone say that it's perhaps the most popular SIP phone for use with *? |
| 15:30.45 | Nugget | do you really care what's the most popular, or do you care what's best? :) |
| 15:30.57 | Nugget | the honda civic is a "popular" car. |
| 15:31.06 | gnosys | best of course, but in here, i suspect best ~= most popular... no? |
| 15:31.10 | DrDeke | Do you want it done fast, or do you want it done right? I JUST WANT IT DONE... DAMN! MY PHONES! |
| 15:31.12 | mog_home | mmm honda civic |
| 15:31.18 | [TK]D-Fender | gnosys : Fair assessment. I've worked with 50x & 60x Polycom's and run 26 60x's here |
| 15:31.31 | [TK]D-Fender | Rock solid phones and very flexible |
| 15:31.47 | gnosys | ok. good2know |
| 15:31.59 | gnosys | can i wall-mount the 501s? |
| 15:32.10 | *** part/#asterisk lme (n=lme@gw-r-ifc.infoclip.fr) |
| 15:32.12 | [TK]D-Fender | gnosys : I'd say that with Polycom's new acceptance of * as a platform, their $/value is very good. its makes a good combo for business use. |
| 15:32.23 | gnosys | ok |
| 15:32.41 | synthetiq | anyone know if ser config can be reloaded with out restarting ser? |
| 15:32.48 | Flauto | mesfet, did you see my message? |
| 15:33.18 | [TK]D-Fender | gnosys : NOt sure about the wall mounting.. there'd have to be a special-order part to allow that from what I see in the frame |
| 15:33.29 | gnosys | otay... |
| 15:34.16 | *** join/#asterisk hugov6 (n=foo@p54AD63D9.dip.t-dialin.net) |
| 15:34.19 | hugov6 | hiho |
| 15:34.30 | [TK]D-Fender | I bought Uniden UIP-200's for that here (higher risk of being vandalized and I wanted to minimize the liability too) |
| 15:34.46 | [TK]D-Fender | But I wouldn't recommend them for anything else. |
| 15:34.57 | ManxPower | The polycom phones can be wall mounted. |
| 15:35.09 | ManxPower | you flip around the stand for wall mount |
| 15:35.12 | gnosys | ManxPower: need special hardware? |
| 15:35.17 | ManxPower | gnosys, no |
| 15:35.21 | gnosys | thanks. |
| 15:35.24 | [TK]D-Fender | ManxPower : It's not apparent when I look at mine and I didn't get any plates to support it. |
| 15:35.42 | ManxPower | [TK]D-Fender, take off the plastic base, rotate 180 degrees. |
| 15:35.50 | ManxPower | It should be in the quickstart guide. |
| 15:36.38 | hugov6 | q: i have to match on extension 12340 and extension 123412 123499 and so on. now i tried _1234XX (wont match on 12340) and _1234. (wont match on 123412/99) i dont get it working with both. got someone a hint? |
| 15:37.05 | gnosys | What's the general consensus in here about getting SIP phones from ebay vendors? Bad? Ok? I did this once for my two 7960 phones and had no problems, but if I want the Polycom firmware, maybe I need to go to a licensed reseller? |
| 15:37.21 | [TK]D-Fender | ManxPower : not on a 60x.... maybe the 50x..... |
| 15:37.37 | ManxPower | _1234X will match any 5 digit extension starting with 1234 |
| 15:37.39 | [TK]D-Fender | gnosys : I'd be happy to pass mine on to you. |
| 15:37.57 | gnosys | thanks [TK]D-Fender |
| 15:38.08 | hugov6 | ManxPower: it has to match 5-6 digit extensions |
| 15:38.22 | *** join/#asterisk rajiv (n=irc@gentoo/developer/rajiv) |
| 15:38.27 | ManxPower | hugo-v6, well that's not going to work well. |
| 15:38.28 | [TK]D-Fender | hugo-v6 : _1234. |
| 15:38.59 | hugov6 | tkd-fender: wont match on 123499 for example |
| 15:39.09 | ManxPower | yes, _1234. will work, but if you are on Zap you'll have to wait for a timeout. |
| 15:39.12 | hugov6 | tkd: already tried that |
| 15:39.13 | *** join/#asterisk azid (n=janne@hus051a.gronstenen.se) |
| 15:39.26 | hugov6 | ManxPower: it wont work here |
| 15:39.33 | ManxPower | hugo-v6, _1234. will match any length starting with 1234 |
| 15:39.40 | ManxPower | hugo-v6, then you are doing something else wrong. |
| 15:39.44 | hugov6 | ManxPower: wher i can set the timeout |
| 15:39.55 | ManxPower | hugo-v6, "show application DigitTimeout |
| 15:39.58 | lehel | i want to use some analog phones to connect to PSTN, if not TDM what else? |
| 15:40.05 | *** part/#asterisk MaD-DaRiUs (n=ian@S01060050ba8804cd.vn.shawcable.net) |
| 15:40.35 | *** join/#asterisk funxion (n=nunya@mtnuser.icgws.com) |
| 15:40.43 | *** join/#asterisk Hmmhesays (n=Neg@72.24.227.83) |
| 15:41.04 | af_ | I mean, if samba is running, voice is chopping |
| 15:41.22 | ManxPower | af_, That's not suprizing. |
| 15:41.30 | lehel | .. |
| 15:41.45 | af_ | ManxPower, I guess there is some way to do it |
| 15:41.53 | af_ | like use some of 2.6 |
| 15:42.10 | af_ | I am wondering if there is some I could read for |
| 15:42.20 | hugov6 | pastebin is slow for me atm *wait* |
| 15:43.05 | ManxPower | af_, You can't change reality by wishing really hard. |
| 15:44.42 | hugov6 | ManxPower: btw: concerning af_'s question. i want to run asterisk and samba on the same machine too, whats the problem befor i encounter it? |
| 15:44.43 | *** join/#asterisk MikeJ__ (n=ircatjer@d14-69-8-30.try.wideopenwest.com) |
| 15:44.56 | af_ | it's just that asterisk must be prioritized some way different that samba |
| 15:45.11 | many | renice -19 |
| 15:45.16 | mog_home | ? |
| 15:45.39 | iCEBrkr | Get a real machine, geesh :P |
| 15:46.12 | iCEBrkr | I have web, mail, ftp running on my Asterisk box, I don't have issues. |
| 15:46.24 | iCEBrkr | model name : Intel(R) Pentium(R) 4 CPU 1300MHz |
| 15:47.54 | [TK]D-Fender | I run X (KDE), Samba, *, and plenty more on my AMD2000+ box without any problems.... |
| 15:49.11 | funxion | apt-cache search make |
| 15:49.14 | funxion | lol |
| 15:49.21 | funxion | wrong screen |
| 15:50.29 | funxion | its gcc no |
| 15:50.32 | hugov6 | iCEBrkr: well then it should work on the p4 3ghz cpu with 2mb cache. |
| 15:51.09 | *** join/#asterisk sysdebug (n=sysdebug@200.163.193.247) |
| 15:51.26 | hugov6 | back to my problem. i tried _1234X. a few mins ago. wont work also. now im trying this digittimeout |
| 15:52.12 | *** join/#asterisk oogle (n=jart@justin.ctlinc.com) |
| 15:52.56 | *** join/#asterisk A-Tuin (n=a-tuin@steves.ip.v4.me.uk) |
| 15:53.39 | gnosys | hey, if i missed it then i apologize, but are folks using webvmail with 1.2.1 ? is it fully functional? |
| 15:54.19 | fugitivo | what is webvmail? |
| 15:55.15 | heroine | webvmail is a cgi script distributed with asterisk to allow user to retreive their voicemail messages throught a webserver |
| 15:55.16 | xheliox | Web access to your voice mail. |
| 15:55.37 | heroine | (and was found vulnerable to some security issue recently) |
| 15:55.57 | fugitivo | distributed with asterisk? where is it? |
| 15:56.13 | heroine | don't know .. :) |
| 15:56.33 | Nugget | it's in ./contrib/scripts/ |
| 15:56.51 | fugitivo | Nugget: thanks |
| 15:57.07 | fugitivo | i don't have it |
| 15:57.12 | fugitivo | does it come with 1.2 ? |
| 15:57.12 | Nugget | it's called vmail.cgi |
| 15:57.21 | fugitivo | i see it :) |
| 15:57.49 | paryl | i set a timeout in queues.conf, but use "Queue(4|t|||20)" in extensions.conf, will it use the timeout in the dial string or the conf file? |
| 16:00.03 | [TK]D-Fender | paryl : in queues.conf thats the timeout before trying other agents when the first doesn't answer. in Extensions that limits the max time in queue PERIOD |
| 16:00.15 | [TK]D-Fender | Good for kicking people out after a while to leave VM. |
| 16:00.17 | *** join/#asterisk NDT (i=NDT@cpe-24-195-219-101.nycap.res.rr.com) |
| 16:00.36 | *** join/#asterisk _Madar (n=tophe@219-84-129-163-adsl-tpe.static.so-net.net.tw) |
| 16:00.39 | paryl | aha... that makes sense :) |
| 16:00.49 | gnosys | so are folks in here using webvmail? is it still vulnerable to the security issue? do folks like it? |
| 16:01.56 | xheliox | Yes. Probably. And ]I love it. |
| 16:02.36 | gnosys | are you using 1.2.1? 1.2? |
| 16:02.44 | paryl | though... did anyone see my question earlier about automatically logging off an agent who doesn't answer? |
| 16:02.48 | paryl | is that possible? |
| 16:02.56 | NDT | Hey guys...question...We toll free number for a trunk group...We have a bunch of numbers pointed to the toll free...The carrier we get the numbers from turns a sig data field on their end to yes so we can read the dialed number 10 digit string (Billing this way by matching number to account) Is there anyway in asterisk for me to read this 10 didgit string sent without a gatekeeper? |
| 16:03.36 | gnosys | xheliox: what version of * are you using? |
| 16:03.55 | xheliox | 1.2.1 |
| 16:03.58 | *** join/#asterisk Seldon1975 (n=someone@gatekeeper.radintl.com) |
| 16:04.09 | gnosys | anybody else care to comment on webvmail ? |
| 16:05.01 | *** join/#asterisk pa (n=Paolo@unaffiliated/pa) |
| 16:05.04 | fugitivo | gnosys: i'm trying to make it work |
| 16:05.58 | gnosys | what distribution are you using? TFOT claims that it will only work with Redhat unless you modify the make file. Just curious. What about you xheliox? What distro? |
| 16:06.07 | fugitivo | me? |
| 16:06.11 | gnosys | yep |
| 16:06.14 | fugitivo | i'm using a home made distro |
| 16:06.18 | gnosys | ? |
| 16:06.26 | fugitivo | linuxfromscratch |
| 16:06.32 | gnosys | ah! |
| 16:06.32 | hugov6 | if this wont work soon ill got a gun and go to beronet to kill ppl. |
| 16:06.52 | hugov6 | many: save the beronet ppl, try to support me ;)) |
| 16:08.01 | gnosys | xheliox: what linux distribution are you running your webvmail on? |
| 16:08.25 | *** join/#asterisk Samoied (n=Samoied@popeye.opens.com.br) |
| 16:09.30 | many | buysangoma! |
| 16:11.00 | [TK]D-Fender | many : Already have... twice :) |
| 16:11.09 | coppice | many: is this a brokerage thing? :-) |
| 16:11.20 | many | hugo just asked me to support beronet. |
| 16:11.31 | many | so.. i fled in another direction :-P |
| 16:11.53 | hugov6 | many: grrr ;) |
| 16:12.36 | hugov6 | many: u dont have to support them, only saving them would be enough or ill go and do some assassination |
| 16:12.52 | many | so, 'sup? |
| 16:13.01 | Seldon1975 | this is odd - my system was all set up so that dialling 1111 would play the tt-weasels file, but I rebooted and now when I dial 1111 the message 'Playing tt-weasels' still comes up in the * console, but it stalls there and doesnt play the file |
| 16:13.26 | [TK]D-Fender | Seldon1975 : Pastebin your extensions.conf |
| 16:13.35 | *** join/#asterisk arguile (n=arguile@66.38.201.234) |
| 16:14.37 | Seldon1975 | Fender ok - 1 sec |
| 16:14.56 | hugov6 | jep |
| 16:15.00 | hugov6 | ewindow |
| 16:15.17 | *** join/#asterisk jaiger (n=jaiger@fire.innovationsw.com) |
| 16:16.48 | Seldon1975 | D-Fender: http://pastebin.com/455588 |
| 16:17.41 | [TK]D-Fender | looks good... maybe its your phone setup. |
| 16:18.51 | *** join/#asterisk rking (n=rking@ip68-1-234-152.dl.dl.cox.net) |
| 16:19.00 | Seldon1975 | the odd thing is this whole setup has been working for a couple of days |
| 16:19.15 | Seldon1975 | ive been calling between extensions without problems |
| 16:19.32 | *** join/#asterisk jeffik (n=Jeff@CPE0011505c92d3-CM014350000760.cpe.net.cable.rogers.com) |
| 16:19.54 | *** join/#asterisk Darwin35 (n=Darwin@sta-208-139-193-162.rockynet.com) |
| 16:19.56 | Seldon1975 | could you check that paste again - I readded the lines for other extensions that I had deleted originally - could they be interfering? |
| 16:20.20 | funxion | man |
| 16:20.29 | *** join/#asterisk KranZ (n=user@sme.bestline.net) |
| 16:21.48 | [TK]D-Fender | Seldon1975 : nothing to recheck you need to give me the new # |
| 16:21.59 | Darwin35 | http://pastebin.ca/33102 I need help with converting from astdb to odbc/postgress |
| 16:22.19 | Darwin35 | not sure I got it right |
| 16:22.28 | [TK]D-Fender | I see it... it still looks good |
| 16:23.12 | Seldon1975 | thaks D |
| 16:23.23 | Seldon1975 | how frustrating |
| 16:23.30 | [TK]D-Fender | Darwin35 : Looks a lot like my STDEXTEN :D |
| 16:24.14 | Darwin35 | I wrote this one but at thepoint I am moving to odbc from astdb and I dont think I got it right |
| 16:24.14 | [TK]D-Fender | http://pastebin.ca/33103 |
| 16:26.01 | [TK]D-Fender | I didn't think ODBC sould use DB1 style stuff... thought it was only SQL |
| 16:27.43 | [TK]D-Fender | WIERD |
| 16:27.51 | Darwin35 | it uses odbcput get and del |
| 16:28.03 | Darwin35 | but making all the right calls |
| 16:28.45 | jeffik | hello all: have some questions about mandrake iso discs |
| 16:28.52 | Darwin35 | burn it |
| 16:28.58 | Darwin35 | use it as a coaster |
| 16:29.10 | Darwin35 | wall art |
| 16:29.17 | [TK]D-Fender | Frisbee! |
| 16:29.40 | jeffik | Darwin35: wny? |
| 16:29.44 | jeffik | why |
| 16:30.37 | Darwin35 | so I would say use centos for asterisk if yo use linux |
| 16:30.45 | Darwin35 | else use FreeBSD |
| 16:31.10 | jeffik | Darwin35: i am new to linux, what i want to do is control traffic on a shared wifi router |
| 16:31.58 | Darwin35 | everything we have is fbsd based |
| 16:34.09 | jeffik | Darwin35: fbsd? |
| 16:34.38 | Darwin35 | FreeBSD |
| 16:34.53 | loud | that might explain why teliax has better service than the rest. |
| 16:35.01 | Darwin35 | heheh |
| 16:35.12 | Darwin35 | How did you figure out thats where I worked |
| 16:35.21 | *** join/#asterisk Ayano (n=Ayano_@adsl-70-245-190-90.dsl.spfdmo.swbell.net) |
| 16:35.27 | loud | rockynet. i exchange emails the whole day with david. |
| 16:35.27 | Darwin35 | I am the tech/noc monkey |
| 16:35.35 | loud | cool :) |
| 16:35.35 | Darwin35 | hehhehe |
| 16:35.46 | h3x0r | some asshole emailed me asking if they could switch over from teliax |
| 16:35.50 | Zach^^ | is there a voip company that allows free tolfree calls? |
| 16:35.51 | h3x0r | and they are in some bum fuck egypt place |
| 16:36.00 | h3x0r | and im like, well it wouldnt be much different coz we use level3 dids here too |
| 16:36.02 | h3x0r | so get lost |
| 16:36.03 | h3x0r | haha |
| 16:36.04 | *** join/#asterisk nagl (n=nagl@213.235.241.6) |
| 16:36.11 | Zach^^ | outbound calls that is |
| 16:36.14 | h3x0r | s/here/there/ |
| 16:36.46 | Darwin35 | I stand by the company I work for . we are about stabilty not options and addons |
| 16:36.55 | Darwin35 | but we are adding features |
| 16:36.59 | Darwin35 | as we go |
| 16:37.19 | h3x0r | are you guys using ser or asterisk? |
| 16:37.24 | Darwin35 | both |
| 16:37.28 | h3x0r | yeah |
| 16:37.35 | loud | i can compare teliax with my 8 pris easily. |
| 16:37.37 | h3x0r | b2bua on the did's? |
| 16:38.11 | h3x0r | i bet this guy was trying to run g.711 on his dialup |
| 16:38.20 | Darwin35 | who |
| 16:38.30 | loud | the egyptian |
| 16:38.35 | Darwin35 | heheh |
| 16:38.51 | Darwin35 | brb call |
| 16:38.53 | *** join/#asterisk dalabera (n=dalabera@146.82.190.164) |
| 16:39.00 | dalabera | heelo guys |
| 16:39.30 | Seldon1975 | D-Fender hmm I just rebooted my machine and now Weasels is back |
| 16:39.33 | Seldon1975 | not sure what was wrong |
| 16:40.12 | dalabera | Is there any way to passthrough the disconnect or unallocated messages tones to the caller when using ISDN EI |
| 16:41.16 | KranZ | weasels have eaten our phone system |
| 16:41.24 | iDunno | cool. |
| 16:41.38 | iDunno | you need to get the badgers in to eat the weasels. |
| 16:41.53 | [TK]D-Fender | Seldon1975 : Why do you keep rebooting? |
| 16:42.07 | file[laptop] | VoIP with Vonage! |
| 16:42.14 | [TK]D-Fender | The most * should even need is a "restart now" |
| 16:42.26 | KranZ | wonage! |
| 16:42.41 | *** join/#asterisk coppice (n=chatzill@142.198.17.210.dyn.pacific.net.hk) |
| 16:42.43 | Darwin35 | wronage |
| 16:43.05 | brad_mssw | yeah, switched from vonage to teliax ... far fewer issues |
| 16:43.13 | *** part/#asterisk Ayano (n=Ayano_@adsl-70-245-190-90.dsl.spfdmo.swbell.net) |
| 16:43.14 | brad_mssw | though vonage wouldn't let us go direct sip into * ... |
| 16:43.30 | brad_mssw | even with their business plus plans |
| 16:44.03 | dalabera | I'm using E1 euroisdn and when calling certain numbers I got disconnected instead of hearing the message over the phone |
| 16:44.22 | [TK]D-Fender | file[laptop] : just left you VM :D |
| 16:44.22 | dalabera | is there any way to deactivate this ? |
| 16:44.28 | file[laptop] | [TK]D-Fender: oh no! |
| 16:44.40 | [TK]D-Fender | Routed from work - home - you :D |
| 16:44.54 | Darwin35 | brb have to go blow up a efax server |
| 16:44.56 | [TK]D-Fender | All SIP :D |
| 16:46.50 | file[desk] | [TK]D-Fender: 8223! |
| 16:47.01 | [TK]D-Fender | My work ext # |
| 16:47.29 | file[desk] | now you should route it back the other way! |
| 16:47.56 | *** join/#asterisk Math` (n=math@modemcable148.4-81-70.mc.videotron.ca) |
| 16:48.09 | [TK]D-Fender | I register from home to work as an ext here. My home phone forwards on busy/noanswer to a work DID which then calls my reg through SIP. |
| 16:48.13 | DrDeke | Does anyone know of a free-of-charge SIP or IAX client for PocketPC or Windows Mobile? |
| 16:48.33 | file[desk] | so what will happen if I call your home ext? |
| 16:48.42 | [TK]D-Fender | file[desk] : I don't give people calling my home the ability to call work #'s, just home ones... and accidentally, YOU :) |
| 16:48.56 | file[desk] | lol |
| 16:48.59 | [TK]D-Fender | You'd jsut get VM |
| 16:49.06 | [TK]D-Fender | maybe I can add you... |
| 16:49.35 | Darwin35 | and the no speak english calls come rolling in |
| 16:50.45 | brad_mssw | Darwin35: how many of you are there over there ... on the live support, i've only ever gotten david and richard |
| 16:51.04 | brad_mssw | and I take it david is like the main guy or something |
| 16:51.34 | [TK]D-Fender | file[laptop] : try now :) |
| 16:52.12 | file[desk] | [TK]D-Fender: k lemme look and add this here |
| 16:54.11 | *** part/#asterisk mog_home (n=mogorman@user-24-236-84-48.knology.net) |
| 16:54.30 | Darwin35 | I am richard |
| 16:54.40 | brad_mssw | ah |
| 16:54.40 | *** join/#asterisk g__ (n=g@itd01fw-fibre.itdepartment.com) |
| 16:54.59 | Darwin35 | David is the boss |
| 16:55.28 | loud | i wonder how do you manage that, people not speaking english and all |
| 16:55.29 | file[desk] | and I made up my mind... I'm wasting all my time... here I go again |
| 16:55.33 | *** part/#asterisk DrDeke (i=dekemar@deculator.engin.umich.edu) |
| 16:55.36 | Darwin35 | we are here in the office and we have other that work remote |
| 16:55.37 | *** join/#asterisk DrDeke (i=dekemar@deculator.engin.umich.edu) |
| 16:55.37 | DrDeke | oops |
| 16:56.06 | file[desk] | I was born to walk alone... |
| 16:56.34 | Darwin35 | brb |
| 16:58.41 | g__ | [TK]D-Fender, are you around? |
| 16:59.04 | [TK]D-Fender | yup |
| 16:59.35 | g__ | [TK]D-Fender, I heard a rumour you're familiar with the polycom sip phones.. |
| 16:59.50 | Darwin35 | we take the outback the noc to the learn english 101 room. and enroll them for a 6 week class |
| 16:59.56 | [TK]D-Fender | g__ : that I am |
| 17:00.30 | g__ | [TK]D-Fender: we're having a problem with * agents transfering calls using their polycom phones.. 50% of the time their callers get lost as soon as they push the 'transfer' button.. got any ideas what might be happening? |
| 17:01.16 | jaiger | g__, what do the * logs/console say? |
| 17:01.48 | *** join/#asterisk ManxPower (n=ewieling@adsl-67-65-233-194.dsl.lgvwtx.swbell.net) |
| 17:01.54 | jaiger | g__, we have polycom phones (over a year) with no transfer problems |
| 17:01.56 | [TK]D-Fender | not a clue.... |
| 17:02.39 | jaiger | [TK]D-Fender, do you recommend the latest polycom firmwares with *? I'm still using 1.3.1 but want to upgrade |
| 17:03.15 | g__ | Jaiger: not much. I've compared the console logs of working transfers to broken transfers and there's no difference what soever. |
| 17:04.00 | g__ | Specifically, it's only happening when our technicians attempt to transfer a call they answered from the queue. |
| 17:04.10 | jaiger | g__, does it affect all phones or stay with one phone? does it follow a network port? etc |
| 17:04.10 | g__ | Transfers in general seem to work ok. |
| 17:04.21 | [TK]D-Fender | 1.3.1?! OMG, 1.5.2 suggested... |
| 17:04.39 | jaiger | hmm, we have some queues but most calls come in 'regular' |
| 17:04.45 | g__ | jaiger: we've seen it happen on several different phones. |
| 17:05.07 | g__ | Speficially, it's a queue call answered by an Agent channel. |
| 17:05.11 | Darwin35 | http://pastebin.ca/33102 I still need help changing toodbc ... |
| 17:05.22 | *** join/#asterisk ian_k (n=ian@gateway.digium.com) |
| 17:05.35 | g__ | We used to do without agents, and the problem only surphaced after the change. |
| 17:06.21 | g__ | Does anyone know what happens when you push the "transfer" button on a Polycom phone? I presume it puts the caller on hold and makes another outgoing call. |
| 17:06.29 | *** join/#asterisk Utah_Dave (n=boucha@0-1pool139-158.nas28.salt-lake-city1.ut.us.da.qwest.net) |
| 17:07.03 | g__ | jaiger: we're running 1.5.3.. it's ok. |
| 17:08.27 | jaiger | g__, I thought 'transfer' was handled internally to the Polycom phone |
| 17:09.12 | funxion | anyone know why I would get a segmentation fault while making? |
| 17:09.19 | g__ | jaiger: I think you're right. But * has to do something to provide the hold music. |
| 17:09.35 | jaiger | true |
| 17:09.50 | g__ | And the first sign a transfer is going to fail is the caller gets silence instead of hold music. |
| 17:09.50 | funxion | chan_sip.c:5484: internal compiler error: Segmentation fault |
| 17:10.16 | jaiger | g__, hmm, is there a problem with your hold music? |
| 17:11.24 | g__ | jaiger: I don't think so. It would be worth looking at except the caller is then left off-hook talking to no one until they hang up. |
| 17:11.51 | g__ | .. I mean it suggests something else is happening. |
| 17:12.05 | jaiger | anything in dmesg say indicating a problem spawning your hold-music process (out of mem, etc)? |
| 17:12.06 | *** join/#asterisk kokey (n=ubunture@201.153.63.79) |
| 17:12.32 | jaiger | or syslog, daemon.log etc |
| 17:12.37 | g__ | Nothing since the last system boot.. |
| 17:14.10 | *** join/#asterisk fulgas (n=fulgas@209.8.233.106) |
| 17:14.48 | g__ | I double-checked daemon.log and syslog.. good thought though. |
| 17:15.32 | jaiger | how does the cpu/memory/disk look when it happens (eg. cpu spike, heavy swapping) |
| 17:16.25 | g__ | The machine has plenty of memory, and the cpu usage appears pretty constant. Also, other phone calls aren't interrupted... |
| 17:17.23 | g__ | Does anyone know who maintains the Agent channel? |
| 17:17.36 | *** join/#asterisk cpatry (n=grepmoo@65.39.228.5) |
| 17:19.37 | *** join/#asterisk _4d4m_ (n=adam@212-14-101-159.adsl.legend.co.uk) |
| 17:21.44 | KranZ | hey g__ |
| 17:21.52 | KranZ | you gettin your ?s answered? |
| 17:21.56 | g__ | Hey KranZ.. how are you today? |
| 17:22.07 | KranZ | warmer, its only 34 |
| 17:22.51 | g__ | Not bad.. it must be -5 or -10 here. |
| 17:23.17 | g__ | Anyways, I'm begining to think it's not likely a polycom problem.. |
| 17:23.33 | [TK]D-Fender | Upgrade the firmware... |
| 17:23.36 | g__ | My gut feeling is it's the agent code itself.. do you know who maintains that? |
| 17:23.53 | ManxPower | g__, You are having problems with Agents transfering? |
| 17:23.55 | g__ | [TK]D-Fender: we have allready.. to 1.5.3. |
| 17:24.09 | ManxPower | g__, using AgentCallbackLogin? |
| 17:24.27 | ManxPower | That's been a problem for at least 6 months. Report it as a bug. |
| 17:24.38 | ManxPower | We stopped using AgentCallbackLogin because of this issue. |
| 17:24.39 | g__ | I'm surprised no one else has.. |
| 17:25.14 | ManxPower | g__, AgentCallbacklogin sucks. There are frequently work arounds so you don't have to use it. |
| 17:25.45 | g__ | Thanks ManxPower.. I'll file a bug report now. |
| 17:25.50 | ManxPower | We don't use it. We use Queues with member= rather than agent= |
| 17:26.20 | ManxPower | the member= sends the call to a line on the phone that is not used for other things. The user can put that line in DND if they don't want to get queue calls. |
| 17:26.20 | g__ | We used to not use agents, but we got tired of getting phone calls while allready on the phone with a customer. |
| 17:26.28 | jaiger | yeah, we don't login to our queues |
| 17:27.01 | jaiger | and we have dedicated 'queue' lines registered on all phones |
| 17:27.24 | g__ | I wish we could do that.. |
| 17:27.52 | g__ | How do you get them to avoid rining people's desk phones of people who aren't there? |
| 17:28.36 | jaiger | the downside seems to be "60 missed calls" as it seems that each queue ring is a new call |
| 17:28.59 | jaiger | plus that, phones for people who aren't in still ring the queue line |
| 17:29.10 | *** join/#asterisk Kokey (n=Kokey@201.153.63.79) |
| 17:29.53 | jaiger | my partner didn't want to bother with logging in to the queues, figured it would never happen |
| 17:30.12 | jaiger | so this is our "always logged in" queue |
| 17:30.13 | *** join/#asterisk generalhan (i=general_@ip67-90-64-2.z64-90-67.customer.algx.net) |
| 17:30.18 | generalhan | whats going on everyone ? |
| 17:30.21 | g__ | Interesting. Would it be nice if the polycom's agent support was supported by *? |
| 17:31.46 | generalhan | i have a question for anyone who has used the monitor function ... my calls going to 2 different extensions (and only those 2) are being recorded on 2 seperate channels, an "in.wav" and an "out.wav. where i can only hear my employee on the out.wav, and the caller on the in.wav. any idea why it would be recording them like this ? |
| 17:31.51 | generalhan | anyone ever seen this before ? |
| 17:31.54 | [TK]D-Fender | g__ : yup.. I'm not holding my breath though. I have been in direct contact with Polycom's programming division and maybe can work something out.... |
| 17:31.58 | af_ | I guess I can run samba and asterisk in 2.6 |
| 17:33.47 | g__ | [TK]D-Fender: On the other hand, I'd prefer Polycom fix basic functionality. Do you have as many problems as we do configuring new phones for use? I find they have to be rebooted several times before they pick up their configuration properly. If you're trying to use specific ringtones and graphics, you'll need to reboot even more. |
| 17:34.58 | *** join/#asterisk MikeJ[Laptop] (n=ircatjer@d14-69-8-30.try.wideopenwest.com) |
| 17:35.10 | af_ | I guess that 2.6 do some to assign cpu slices to processes that needs it |
| 17:35.37 | MstlyHrmls | g__: what protocol are you using? |
| 17:35.51 | g__ | ManxPower: what category should I make it? "ACD"? |
| 17:36.07 | g__ | SIP, IAX2, tcp/ip.. |
| 17:36.29 | MstlyHrmls | g__: heh, for the polycom configuring... |
| 17:37.03 | g__ | Oh, ftp with proftpd. |
| 17:37.16 | jaiger | g__, I don't have too many problems configuring a phone for the first time but I don't use custom graphics or ring tones either. I boot 2 or 3 times |
| 17:38.03 | MstlyHrmls | g__: what BootROM/App version? |
| 17:38.13 | *** join/#asterisk heison (n=heison@gw-yyz1.somanetworks.com) |
| 17:38.15 | g__ | I'm using 1.5.3 and the old bootrom. |
| 17:38.32 | g__ | We don't yet have a need for the new bootrom.. |
| 17:38.33 | MstlyHrmls | 2.6.1? |
| 17:38.53 | g__ | Yup |
| 17:39.32 | MstlyHrmls | interesting. I've seen that once and a while with a similar setup, but not constantly. Course I only use a few ringtones and no custom graphics |
| 17:42.07 | [TK]D-Fender | g__ : new phones are never a problem. I jsut copy & paste the 2 config files, change the ID & password & MicroBrowser target and come back in 5 minutes. Boot-rom, SIP and config refresh done and ready to use |
| 17:42.24 | KranZ | is there a progressinband equivalent for mgcp? |
| 17:42.33 | g__ | By "new phone" do you mean the 501 serries? |
| 17:42.54 | jaiger | [TK]D-Fender, that sounds about what I do |
| 17:44.07 | *** join/#asterisk jhiver (n=jhiver@AStDenis-105-1-4-4.w193-253.abo.wanadoo.fr) |
| 17:44.11 | jhiver | ~seen Zeek |
| 17:44.15 | jbot | zeek <~zeekk@gw.dhivehinet.net.mv> was last seen on IRC in channel #asterisk, 286d 10h 50m 28s ago, saying: 'does anybody here use firefly?'. |
| 17:44.34 | [TK]D-Fender | g__ : I use 60x exclusively here |
| 17:44.47 | *** join/#asterisk shmaltz (n=chatzill@mail.dmaven.com) |
| 17:45.38 | jaiger | g__, I've used 60x, 50x and 30x |
| 17:45.59 | *** join/#asterisk EriSan (n=erisan@81-174-42-85.f5.ngi.it) |
| 17:46.20 | shmaltz | I'd rather eat it: |
| 17:46.22 | shmaltz | http://www.breitbart.com/news/2005/12/09/051209141924.flu6l9pn.html |
| 17:46.39 | g__ | I ment 500 vs. 501. |
| 17:47.17 | g__ | [TK]D-Fender: Nice.. I wouldn't mind a 601 with an expansion panel myself. |
| 17:48.26 | shmaltz | I like that liitle new feature from gmail, with the rss on top |
| 17:48.38 | shmaltz | it advertises VoipSupply.com every few minutes |
| 17:49.54 | g__ | ManxPower: can I quote you in the bug report? |
| 17:50.10 | shmaltz | g___, what bug? |
| 17:50.15 | ManxPower | g__, if you want to, but I can't test any fixes. |
| 17:50.22 | ManxPower | g__, you should ask on the mailinglist too. |
| 17:51.01 | g__ | ManxPower: Of course.. but that you observed it might lend credability to me because it supports the existance of the bug. |
| 17:51.06 | ManxPower | *nod* |
| 17:51.19 | g__ | I promise I won't quote you as saying "AgentCallbacklogin is crap" |
| 17:53.28 | *** join/#asterisk keith80403 (n=keith804@24-56-188-49.co.warpdriveonline.com) |
| 17:53.39 | g__ | ManxPower: I was just about to quote this guy called "Eric Wieling" from the mailing list as well.. |
| 17:53.49 | ManxPower | 8-) That would be me. |
| 17:53.58 | g__ | Thanks for replying to my email the other day :) |
| 17:54.12 | ManxPower | which one was that? |
| 17:54.27 | ManxPower | Same issue? |
| 17:54.27 | g__ | The one that asked if you'd reported it as a bug yet.. |
| 17:54.29 | g__ | yup |
| 17:54.30 | ManxPower | Ah. |
| 17:55.07 | shmaltz | interesting, I guess someone will put up a bounty for this: |
| 17:55.09 | shmaltz | http://news.yahoo.com/s/cmp/20051209/tc_cmp/174907945;_ylt=Ap3dX_7rNad.1eO9qT2F13qor7oF;_ylu=X3oDMTBjMHVqMTQ4BHNlYwN5bnN1YmNhdA-- |
| 17:56.19 | ManxPower | I thought BBN was bought by GE Networks, which was bought by Sprint, which was bought by UUNet |
| 17:57.50 | shmaltz | holdon |
| 17:57.59 | shmaltz | sprint is owned by uunet? |
| 17:58.16 | g__ | shmaltz: the AgentCallbacklogin transfer-call bug we've been discussing. |
| 17:58.31 | shmaltz | g__, I realized |
| 17:59.57 | generalhan | can anyone refer me to some place that has a macro script for the soxmix command? my recordings are getting stuck in the "*-in.wav" and "*-out.wav" and they wont combine anymore. i need to find a way to force this to happen ? any one have any suggestions ? |
| 18:00.28 | pif | you don't need somix, use 'm' in monitor() |
| 18:00.46 | generalhan | i have the ",m" in my monitor call |
| 18:00.51 | g__ | Done! Bug 0005962.. |
| 18:00.53 | KranZ | uunet is owned by sprint |
| 18:00.54 | generalhan | its not working on 2 of my extensions |
| 18:01.08 | KranZ | er |
| 18:01.11 | KranZ | no MCI owned UUNET |
| 18:01.18 | KranZ | i think verizon bought MCI |
| 18:01.19 | pif | because of spaces in filenames |
| 18:01.35 | pif | fix that |
| 18:01.37 | g__ | MCI still ownz uunet. |
| 18:01.49 | g__ | I know because I had to call them recently and gave up. |
| 18:01.52 | generalhan | well i have the monitor() being called extaly the same way as in all 30 of my extensions, and only 2 of them arent combining |
| 18:01.56 | KranZ | yeah -owned |
| 18:02.25 | *** join/#asterisk t0ke (n=toke@51.Red-83-46-136.dynamicIP.rima-tde.net) |
| 18:04.09 | generalhan | pif: http://generalhan.pastebin.ca/33108 there is my extensions.conf for those 2 extensions, it is set up the same way all my others are and they work fine, could you take a look and tell me if you notice anything out of place ? |
| 18:05.25 | KranZ | the space |
| 18:05.27 | KranZ | incomming/ ${EXTEN}-${TIMESTAMP} |
| 18:05.37 | *** part/#asterisk cpatry (n=grepmoo@65.39.228.5) |
| 18:05.38 | [TK]D-Fender | SPACE in there |
| 18:05.40 | generalhan | CRAP! how come that works on the others and not on this one |
| 18:05.44 | generalhan | these 2 rather |
| 18:05.52 | KranZ | any reason you have the space? |
| 18:05.58 | generalhan | all the other 28 work fine ... |
| 18:06.14 | pif | generalhan : yes, space in filename |
| 18:06.17 | generalhan | well there WAS a reason for the space for my external DB that was pulling these, wanted the space in there |
| 18:06.24 | KranZ | you could prolly do some _7XXX extensions and combine them |
| 18:06.30 | [TK]D-Fender | generalhan : those extens are SCREAMING to be macro'd |
| 18:06.40 | file[desk] | I'm the master today! The master of fixing problems! |
| 18:06.40 | generalhan | ill fix it and see if that changes anything |
| 18:06.41 | pif | generalhan : you need macros |
| 18:06.51 | pif | of course it will fix it |
| 18:06.57 | generalhan | im not really familiar wiht macros |
| 18:07.01 | Nugget | obviously :) |
| 18:07.17 | generalhan | what do you mean "of course it will fix it" ? 28 of my 30 extensions are set up this EXACT way and they all work fine |
| 18:07.18 | pif | without even seeing your code I told you "space in filename" |
| 18:07.22 | [TK]D-Fender | generalhan : Here, have some pain :D http://pastebin.ca/33103 |
| 18:07.40 | Nugget | while we're piling on, you also misspelled "incoming" |
| 18:07.41 | [TK]D-Fender | generalhan : A quick mod or 30 and it'll record too :D |
| 18:08.10 | *** join/#asterisk drbrown (n=keith@user-0cdvefr.cable.mindspring.com) |
| 18:08.16 | generalhan | Nugget: LOL i know about that .. i have an "incoming" and an "incomming" |
| 18:08.47 | pif | I have an "incommmming" for HBs |
| 18:08.48 | drbrown | has anyone had any problems routing sip over an openvpn connection, the phone rings, I can pickup, but cannot transmit voice |
| 18:09.00 | generalhan | [TK]: OMG .. i cant do that stuff ! HAHAHAHA |
| 18:09.03 | brad_mssw | hmm, my extensions.conf is 16k |
| 18:09.09 | drbrown | could this be an rtp prob???? |
| 18:09.26 | [TK]D-Fender | brad_mssw : how much is "filler" though? |
| 18:09.54 | brad_mssw | [TK]D-Fender: heh, not much ... i'm bad about commenting |
| 18:09.58 | [TK]D-Fender | generalhan : Check the WIKI for the stdexten macro sample and base yours on that. |
| 18:10.15 | brad_mssw | [TK]D-Fender: unless you're meaning 20k lines ... then no, i'm not close to that ... 20kbytes, yes |
| 18:10.22 | [TK]D-Fender | brad_mssw : I'm ok with commenting, but minimally. I tend to write stuff that explains itself. |
| 18:10.24 | generalhan | [TK]: thanks ! ill check it out (dont know if i can learn this stuff that quickly though) LOL. |
| 18:10.49 | brad_mssw | need to move to AEL though .... |
| 18:10.56 | pif | generalhan : the time you spend debugging redundant code would be better spent on macros |
| 18:11.02 | [TK]D-Fender | brad_mssw : Actuall its only 12.8k... its just that docelm0 here always pokes fun at me about it even being that big... |
| 18:11.14 | pif | redundant == bugs |
| 18:11.24 | [TK]D-Fender | AEL = waste. Nothing you can't do in std extensions.conf and thats what it gets parsed itno anyways. |
| 18:11.28 | shmaltz | brad_mssw, why AEL, whats wrong with plain old extensions.conf? |
| 18:11.32 | generalhan | pif -- and everyone: i agree with you, its just that i have a ton on my plate right now that my boss is riding me about, i just dont know if i have time to figure this all out right now |
| 18:11.39 | generalhan | but ill definately take a look |
| 18:11.42 | brad_mssw | shmaltz: i prefer actual structured code |
| 18:11.50 | brad_mssw | shmaltz: easier to read |
| 18:12.03 | shmaltz | brad_mssw, and extensions.conf? that's not structured? |
| 18:12.11 | shmaltz | how so? |
| 18:12.25 | brad_mssw | shmaltz: yeah, it's structured, like basica is structured |
| 18:12.26 | [TK]D-Fender | it only LOOKS like structured code.... seen any "GOTO"'s in there? ;) |
| 18:12.27 | pif | I went the AEL route but went back |
| 18:12.41 | brad_mssw | shmaltz: numbering everything and crap |
| 18:12.49 | pif | it's all converted back to standard crap by * |
| 18:13.05 | [TK]D-Fender | I though about it till I learned that it only gets parsed back and doesn't offer any new functionailty, just syntax and I"m ok with extensions.conf |
| 18:13.08 | shmaltz | brad_mssw, u dont' have to number anymore, you can use n |
| 18:13.28 | brad_mssw | shmaltz: heh, yeah, still on 1.0.x though ... plan on going 1.2.1 this weekend when no one is using the phones |
| 18:13.29 | pif | the 'n' prio is nice in 1.2 |
| 18:14.05 | shmaltz | anybody tried their echocans? |
| 18:14.07 | shmaltz | http://www.nmscommunications.com/NetSolutions/VoiceQuality/VQProducts/default.htm |
| 18:15.05 | [TK]D-Fender | shmaltz : over what kind of tech? |
| 18:15.13 | *** join/#asterisk RaYmAn-Bx (i=rayman@x1-6-00-40-63-da-39-3f.k191.webspeed.dk) |
| 18:15.14 | *** join/#asterisk florz (i=nobody@2001:1a50:503c:0:0:0:0:1) |
| 18:15.22 | shmaltz | [TK]D-Fender, tellabs |
| 18:15.36 | [TK]D-Fender | in-line T1? |
| 18:16.37 | *** join/#asterisk vel_paris (n=vel@gam75-1-81-57-22-59.fbx.proxad.net) |
| 18:19.32 | generalhan | im looking at the wiki for the stdexten macro ... where are ${ARG1} and ${DIALSTATUS} defined ? or are the preset variables ? |
| 18:19.47 | generalhan | or are they preset rather |
| 18:19.55 | *** join/#asterisk johnnyq (n=johnnyq@cr-5.pitdc1.pa.stargate.net) |
| 18:20.24 | *** join/#asterisk m160858 (n=jsaenz@200.89.12.46) |
| 18:20.32 | m160858 | hi |
| 18:20.39 | [TK]D-Fender | generalhan : ${ARG[number}} referes to parameters you PASS to your macro. ${DIALSTATUS} is set by the DIAL command after it executes automatically. |
| 18:20.44 | m160858 | i need to record the incoming calls |
| 18:21.03 | m160858 | this option record_out=Always, is ok? |
| 18:21.07 | generalhan | [TK]: so they ARE pre determined before i call on them. yes ? |
| 18:21.18 | generalhan | well for DIALSTATUS anyway |
| 18:21.29 | johnnyq | Hello I was wondering if someone could help me with finding a solution that will dial a bunch of number and play a prerecrded message, something that will auth via SIP would be great |
| 18:21.30 | [TK]D-Fender | so - exten => 1234,1,Macro(mymacroname,param1,param2,param3,......) |
| 18:22.01 | [TK]D-Fender | generalhan : You call your macro and PASS it the details and it will process based on those. |
| 18:22.18 | generalhan | lol ... ok, still need more research i think ! LOL |
| 18:22.21 | m160858 | hello? |
| 18:22.29 | [TK]D-Fender | Not too much, just look at how STDEXT is called. |
| 18:23.03 | pif | generalhan : normally you pass nothing to stdexten and use ${MACRO_EXTEN} inside |
| 18:23.14 | pif | which is the EXTEN you come from |
| 18:23.18 | generalhan | [TK]: and your [macro-stdexten] context is defined in extensions.conf ? |
| 18:23.24 | [TK]D-Fender | http://www.voip-info.org/wiki/view/Stdexten+macro |
| 18:23.40 | [TK]D-Fender | this is a sample you would add to extensions.conf |
| 18:23.49 | generalhan | yea thats what im looking at now ! lol |
| 18:23.55 | Seldon1975 | NOTICE[3314] app_dial.c:1010 dial_exec_full: Unable to create channel of type 'Zap' (cause 17 - user busy) |
| 18:24.01 | Seldon1975 | can someone tell me what this means |
| 18:24.04 | *** join/#asterisk haribole (n=hariom10@bi01p1.nc.us.ibm.com) |
| 18:24.08 | Seldon1975 | it happens when I try to dial out |
| 18:24.23 | *** join/#asterisk ful|work (n=fulgas@209.8.233.106) |
| 18:24.35 | *** join/#asterisk SplasPood (i=jwb@ludicrous.paravolve.net) |
| 18:24.36 | pif | Seldon1975 : what does "zap show channels" say? |
| 18:24.50 | [TK]D-Fender | and you would call it like "exten => _7xxx,1,Macro(STDEXTEN,SIP/${EXTEN},${EXTEN}) for an extension to dial its SIP device and use thew same # for the voicemail box |
| 18:25.42 | Seldon1975 | all channels blank |
| 18:25.42 | haribole | someone: how do I execute an agi after hangup, saw that it can be done use h,1,AGI, but I am using 800xxxxxx,7,hangup |
| 18:25.45 | file[desk] | all your pbx are belong to... ME! |
| 18:25.56 | *** part/#asterisk m160858 (n=jsaenz@200.89.12.46) |
| 18:26.05 | [TK]D-Fender | take off every zig! |
| 18:26.43 | Seldon1975 | all your base |
| 18:26.44 | Seldon1975 | base |
| 18:27.33 | *** join/#asterisk arguile (i=user224@66.38.201.234) |
| 18:27.35 | *** join/#asterisk jake1932 (n=jake1932@pool-68-162-123-228.phil.east.verizon.net) |
| 18:28.48 | *** part/#asterisk g__ (n=g@itd01fw-fibre.itdepartment.com) |
| 18:29.08 | generalhan | pif / [TK]: will these macros help my monitor() ?? cause taking the space out of there didnt solve anything. it is still being recorded into two files |
| 18:31.15 | [TK]D-Fender | missing SOXMIX maybe... |
| 18:36.22 | zoa | somebody go test the jitter buffer! |
| 18:37.19 | generalhan | [TK]: but SOXMIX is doing its thing on 28/30 extensions, why would it just pick these two ? |
| 18:37.39 | [TK]D-Fender | Bad path? |
| 18:42.28 | *** join/#asterisk Cinen (n=Cinen@vpn.triadtelecom.com) |
| 18:42.48 | *** join/#asterisk florz (i=nobody@2001:1a50:503c:0:0:0:0:1) |
| 18:43.55 | Seldon1975 | inh my dialplan |
| 18:43.58 | Seldon1975 | if i have: |
| 18:44.02 | *** part/#asterisk johnnyq (n=johnnyq@cr-5.pitdc1.pa.stargate.net) |
| 18:44.27 | Seldon1975 | exten=>_9.,1,Dial(Zap/24/{exten:1}) |
| 18:44.42 | Seldon1975 | and I dial 94162722222 |
| 18:44.57 | Seldon1975 | should asterisk call 4162722222 |
| 18:45.07 | Seldon1975 | ie: missing out the first digit (9) |
| 18:45.21 | Seldon1975 | becasue it seems to be including the 9 when it dials out |
| 18:45.40 | azzie | it should |
| 18:46.18 | Seldon1975 | hmm |
| 18:47.28 | *** join/#asterisk t0ke (n=t0ke@51.Red-83-46-136.dynamicIP.rima-tde.net) |
| 18:48.53 | Seldon1975 | any ides for me to try? |
| 18:50.25 | [TK]D-Fender | UPPERCASE <----- |
| 18:50.30 | [TK]D-Fender | it is case sensitive |
| 18:50.38 | Seldon1975 | ohhh |
| 18:50.41 | [TK]D-Fender | And add the $ you are missing. |
| 18:50.48 | funxion | anyone know why I would get segmentation fault when trying to compile * |
| 18:50.49 | Seldon1975 | aha |
| 18:50.55 | [TK]D-Fender | exten=>_9.,1,Dial(Zap/24/${EXTEN:1}) |
| 18:50.58 | Seldon1975 | ok thanks D-Fender |
| 18:51.03 | funxion | I've never had these probs b4 |
| 18:51.07 | Seldon1975 | doh! |
| 18:51.14 | [TK]D-Fender | And why use /24 sepcifically? |
| 18:51.31 | [TK]D-Fender | You're on a PRI right? |
| 18:51.45 | funxion | its an ARM9 processor |
| 18:51.47 | Darwin35 | so anyone going to proof read and point to me what I did wrong yet ? |
| 18:51.51 | funxion | would that make a diff? |
| 18:51.53 | Darwin35 | http://pastebin.ca/33102 I still need help changing toodbc ... |
| 18:52.57 | funxion | In file included from aestab.c:42: |
| 18:52.57 | funxion | aesopt.h:992: internal compiler error: Segmentation fault |
| 18:53.34 | Corydon-w | Darwin35: since when was ODBCget a function? |
| 18:55.07 | Corydon-w | and third priority doesn't make any sense |
| 18:56.07 | Corydon-w | You're using functions that aren't part of Asterisk, and you think we should be able to tell you what you're doing wrong? |
| 18:56.26 | *** join/#asterisk ard (n=ard@2001:7b8:32d:0:20c:6eff:fe18:d11f) |
| 18:57.26 | funxion | anyone? |
| 18:57.49 | Corydon-w | funxion: you have bad memory |
| 18:57.56 | funxion | really |
| 18:58.03 | Corydon-w | funxion: or another hardware fault |
| 18:58.13 | file[desk] | if you're getting segmentation faults with your compiler, it's not the fault of asterisk... |
| 18:58.15 | Corydon-w | overheating is another possibility |
| 18:58.21 | funxion | the box functions perfectly otherwise |
| 18:58.29 | funxion | its a flash absed debian box |
| 18:58.43 | file[desk] | ever hear of cross compiling? |
| 18:58.49 | Corydon-w | Yeah, things that tax the memory and CPU tend to expose problems like that |
| 18:58.55 | Romik_ | anybody uses ldap with asterisk? |
| 18:59.31 | funxion | hmm |
| 18:59.33 | funxion | thnx |
| 18:59.36 | funxion | both of you |
| 18:59.42 | waddy | isdn -> Bri Card -> Aterisk --- Should the card be in TE or NT mode? |
| 18:59.50 | Corydon-w | Most operations don't stress either memory or the CPU |
| 19:00.05 | Seldon1975 | D-Fender, I have exten=>_9.,1,Dial(Zap/24/${EXTEN:1}) |
| 19:00.12 | Seldon1975 | and its still including the 9 when it dials out |
| 19:00.17 | Seldon1975 | oops |
| 19:00.18 | Seldon1975 | no |
| 19:00.21 | Seldon1975 | my bad |
| 19:02.03 | docelmo | YIPPIE! |
| 19:05.44 | *** join/#asterisk ReD-MaN (i=redman@dhcp-0-2-b3-9a-4a-5b.cpe.quickclic.net) |
| 19:07.42 | *** join/#asterisk splatone (n=splat1@rrcs-24-172-35-197.midsouth.biz.rr.com) |
| 19:13.15 | Darwin35 | its listed inthe wiki |
| 19:13.21 | *** join/#asterisk pryk (n=tmalkut@fw.orasoft.net.pl) |
| 19:14.42 | Darwin35 | it works fine with astdb |
| 19:14.55 | Darwin35 | but just moving to odbc is changing things |
| 19:20.33 | *** join/#asterisk darby_t (i=darby_t@dkl128.neoplus.adsl.tpnet.pl) |
| 19:20.34 | *** part/#asterisk Samoied (n=Samoied@popeye.opens.com.br) |
| 19:20.52 | *** join/#asterisk kredford (n=merida@201.138.18.144) |
| 19:21.32 | kredford | good afternoon |
| 19:22.12 | kredford | I am a new user to asterisk and have download and installed on suse 10.0 |
| 19:22.46 | kredford | if anyone can help me with a zaptel issue I would appricate it |
| 19:22.54 | [TK]D-Fender | Whats the issue? |
| 19:23.25 | *** part/#asterisk eye69 (i=magnus@upcore.net) |
| 19:24.05 | *** join/#asterisk YoMama (n=r00t@pcp02689850pcs.roylok01.mi.comcast.net) |
| 19:25.04 | YoMama | so...i got a tricky one for people...if I'm using CFIM/CFBS, is there a way to get a BLF to light up when the phone is forwarded? |
| 19:25.09 | YoMama | how sweet would that be? |
| 19:25.36 | kredford | It worked that time sorry.. |
| 19:25.41 | *** join/#asterisk L|NUX (i=linux@202.5.131.32) |
| 19:27.26 | kredford | but If I can move to theory I have a sip ata which I want to attached.. I shouldn't need the zaptel module |
| 19:29.02 | *** join/#asterisk zapa (n=zapa@200.39.202.5) |
| 19:29.31 | YoMama | can i call a macro from a macro? |
| 19:36.32 | *** join/#asterisk burnproof (n=burnproo@210.213.244.76) |
| 19:36.46 | *** join/#asterisk gnosys (n=ksford@ip68-9-201-250.ri.ri.cox.net) |
| 19:36.46 | burnproof | hi, good day guys :) |
| 19:37.01 | *** join/#asterisk azop (i=curt@twisted.bluecherry.net) |
| 19:37.13 | burnproof | has anyone alive out there, can i ask a few question please |
| 19:37.23 | azop | Does asterisk support Vonage without purchasing a 'softphone' account? |
| 19:37.54 | [TK]D-Fender | YoMama : macro to macro? sure |
| 19:38.04 | [TK]D-Fender | Just remember they jump back like a stack |
| 19:38.21 | burnproof | how can i accept inconming call coming from internet / another asterisk server |
| 19:38.30 | [TK]D-Fender | azop : If you know what the connection details are for the account, sure |
| 19:38.47 | azop | [TK]D-Fender: would this be public information? :P |
| 19:38.48 | YoMama | what exactly is callback voicemail? |
| 19:38.48 | *** join/#asterisk Seldon1975 (n=someone@gatekeeper.radintl.com) |
| 19:38.52 | [TK]D-Fender | burnproof : Depends what the calling side uses |
| 19:39.21 | burnproof | [TK]D-Fender: e.g another asterisk server ? |
| 19:39.49 | *** join/#asterisk zeedo (n=zeedo@reboot-robot.net) |
| 19:39.51 | *** join/#asterisk zapa (n=zapa@200.39.202.5) |
| 19:40.06 | [TK]D-Fender | azop : Vonage tends to lock their ATA's and NOT want to give it out. |
| 19:40.40 | [TK]D-Fender | burnproof : Several ways to do it. SIP / IAX (with / without registrations) |
| 19:41.04 | burnproof | [TK]D-Fender: i prefer it via sip with registration of course :) |
| 19:41.15 | [TK]D-Fender | My home server connects to my work as though it ws a phone in the work server. |
| 19:41.24 | [TK]D-Fender | thats 1 easy way. |
| 19:41.36 | [TK]D-Fender | * > * is what IAX is good for. |
| 19:42.27 | burnproof | [TK]D-Fender: oic |
| 19:43.06 | [TK]D-Fender | look up "dual servers" on the WIKI. its what I did. |
| 19:43.24 | *** join/#asterisk harryvv (n=none@S010600a0c93f6f7e.vs.shawcable.net) |
| 19:43.34 | burnproof | [TK]D-Fender: i'll dig into it thanks bro. |
| 19:43.38 | *** join/#asterisk fulgas (n=fulgas@a81-84-117-84.cpe.netcabo.pt) |
| 19:45.17 | YoMama | Fender: any ideas of how to get a BLF indicator to go off when someone has forwarded their phone? |
| 19:46.13 | [TK]D-Fender | How is the forwarding enabled? |
| 19:46.20 | [TK]D-Fender | Dialplan or on the phone? |
| 19:47.26 | YoMama | i have a GXP-2000 |
| 19:47.35 | YoMama | there's no forwarding feature on the phone..so i'd have to do it from the dialplan |
| 19:47.43 | YoMama | which means for a light to light up on this phone..you'd have to use BLF |
| 19:47.58 | [TK]D-Fender | BLF = no-go, but you could use MWI.... |
| 19:48.07 | zapa | hi all iam having troubles with a norstart and asterisk with e&m W any clue WARNING[21966]: chan_zap.c:5351 ss_thread: getdtmf on channel 2: takes to long to hear a simulated dial tone |
| 19:50.12 | YoMama | TKD-Fender: hmmm |
| 19:50.56 | YoMama | Fender: how do i manually trigger MWI? and could i make it solid light for forward and blinking for an actual MWI? |
| 19:51.41 | [TK]D-Fender | You can only basically fake having a message, not how the phone will react to it. |
| 19:51.56 | YoMama | ah |
| 19:51.57 | [TK]D-Fender | unless there is some sort of SIP header message youcan pass a GS |
| 19:52.35 | YoMama | or maybe use this big honkin' screen they got on here |
| 19:52.43 | justinu | i think that GXP2000 supports BLF's with a beta firmware |
| 19:52.50 | YoMama | justinu: it does |
| 19:53.13 | justinu | but you'd have to modify the * code that sends out the notifies to the phone |
| 19:53.20 | justinu | to hook into the call forwarding status in the dialplan |
| 19:53.24 | justinu | which seems pretty fubar'd |
| 19:54.58 | YoMama | it's just be nice to give people an indication that their call has been forwarded |
| 19:55.06 | YoMama | err..that their extension has been |
| 19:55.08 | justinu | yeah, i get what you're after |
| 19:55.16 | justinu | you could certainly do it |
| 19:55.18 | YoMama | same with DND |
| 19:55.31 | YoMama | most traditional PBXes have lights for DND and forwarded extension |
| 19:55.40 | justinu | yeah... it's just that the gxp2000 kinda sucks |
| 19:55.45 | justinu | not enough features |
| 19:56.02 | YoMama | well, hopefully GS comes out with some fancier firmware |
| 19:56.08 | justinu | agreed |
| 19:56.14 | YoMama | it's got a huge display...lots of buttons...should be able to do lots with it |
| 19:56.25 | justinu | yep |
| 19:56.27 | YoMama | although..i'll have to say they made one really fatal mistake in it already..that's irreversible |
| 19:56.32 | YoMama | and makes it pretty useless as a business phone |
| 19:56.35 | justinu | i wanted a gxp-2000 that ran sipura firmware |
| 19:56.40 | YoMama | if the headset is plugged in, the phone won't ring thru the speaker |
| 19:56.52 | YoMama | and it's hardware |
| 19:56.58 | justinu | wow, that sucks |
| 19:57.01 | YoMama | yes it does |
| 19:57.17 | [TK]D-Fender | YoMama : I just tested the MWI method on my polycom's. Works well. |
| 19:57.23 | YoMama | so think about it in a call center..no one would hear their phoen ring unless they were a) wearing their headsets or b) had the headset disconnected |
| 19:57.38 | [TK]D-Fender | justinu : Get an SPA-941 :) |
| 19:57.42 | YoMama | Fender: but u can't tell if you have voicemail or a forwarded phone |
| 19:57.51 | justinu | yeah, but 941 is 150 bucks |
| 19:57.53 | justinu | :( |
| 19:58.00 | justinu | gxp2000 is 85 bucks |
| 19:58.02 | [TK]D-Fender | I bought one... very nice |
| 19:58.13 | [TK]D-Fender | GXP = fit for Ken & Barbie |
| 19:58.16 | YoMama | justinu: yup...that's how much i paid for my GXP-2000..it's not bad...there are just some big missing features |
| 19:58.22 | [TK]D-Fender | You do get what you pay for. |
| 19:58.27 | YoMama | yep u do |
| 19:58.32 | justinu | yep |
| 19:59.20 | justinu | i've tried about all of them except the new 941 |
| 19:59.30 | *** join/#asterisk Sp14t (n=splat1@rrcs-24-172-35-197.midsouth.biz.rr.com) |
| 19:59.37 | funxion | y |
| 19:59.40 | justinu | got a snom 360, aastra 480, polycom 501/601, 841, etc.... |
| 20:00.34 | Sp14t | can you dial directly to a IP address on cisco phones? |
| 20:00.40 | [TK]D-Fender | The 941 = Baby Cisco. Can be provisioned multiple ways, with Sipura's great web interface for those who want that. Best of all worlds in that sense. Poly / Cisco has a better speakerphone, but its better than most of the rest and that handset is great |
| 20:01.01 | justinu | i'm really impressed with the sipura software |
| 20:01.10 | justinu | super customizable |
| 20:01.16 | justinu | great hacker phone |
| 20:01.36 | GXTi | omghax |
| 20:01.59 | YoMama | when a phone is on DND...it returns the call as unavailable? |
| 20:02.14 | YoMama | as if it had rung and no one answered? |
| 20:02.28 | justinu | fender: you have polycoms? |
| 20:03.51 | [TK]D-Fender | YoMama : depends on the phone and its config |
| 20:04.06 | [TK]D-Fender | justinu : yup, 600/601's here |
| 20:04.31 | YoMama | Fender: ah..figured it out...i didn't have n+101 setup so it just went to n+1 |
| 20:04.33 | YoMama | works now |
| 20:04.45 | justinu | fender: no 501s? i was gonna ask you if you knew how to get the 501's working over PoE without having to buy that 40 dollar adapter |
| 20:05.19 | [TK]D-Fender | you need an adapter. I'm sure its possible to make on, but for the ahssle thats why they charge 40$ |
| 20:05.30 | justinu | do you have any technical details? |
| 20:05.49 | justinu | because I found some adapters to make cisco 7960s work with standard 802.3af PoE |
| 20:05.50 | [TK]D-Fender | justinu : nope, but I'm sure its not hard... |
| 20:05.53 | justinu | and they're only 20 bucks |
| 20:06.00 | [TK]D-Fender | there may be aftermarket ones... |
| 20:06.43 | [TK]D-Fender | I got 600's because I wanted PoE and for the differnce in price from the 500 it isn't much. And that payed for the microbrowser :D |
| 20:06.46 | justinu | yeah |
| 20:06.57 | *** join/#asterisk batphone (n=batphone@69.15.174.114) |
| 20:06.59 | justinu | even tho they're awesome |
| 20:07.01 | batphone | hel0p! |
| 20:07.16 | batphone | i need a mobo that can handle four 4 port pri cards! |
| 20:07.37 | batphone | its hard to find one that can handle all the transcoding w/o blips and chirps |
| 20:07.38 | [TK]D-Fender | batphone : Digium recommends no more than 2 period. |
| 20:07.45 | justinu | yomama: FYI, polycom returns "486 busy here" when DND is set. |
| 20:07.51 | mog_work | 3 with echo can |
| 20:07.55 | harryvv | I know that its best to use only one port for a pri card but has anyone tried more then one port without irq conflicts? |
| 20:07.55 | mog_work | and good box |
| 20:08.00 | mog_work | but why do that |
| 20:08.00 | batphone | [TK]D-Fender, its been done with 6 |
| 20:08.02 | mog_work | just get another box |
| 20:08.03 | [TK]D-Fender | justinu : I believe even that is configurable.... |
| 20:08.05 | batphone | i just need to know what mobo |
| 20:08.16 | justinu | fender: i wouldn't doubt it... the polycom config is pretty big |
| 20:08.19 | mog_work | why not get two boxes batphone |
| 20:08.19 | justinu | but, on that topic! |
| 20:08.20 | YoMama | justinu: the GXPs do too..i just wish there was a way to "indicate" immediate call forwarding |
| 20:08.25 | batphone | harryvv, i have a box running with 4 w/o irq conflicts |
| 20:08.26 | mog_work | failover is good |
| 20:08.36 | batphone | mog_work, fractional ds3 from AT&T wont allow it |
| 20:08.45 | harryvv | batphone how many ports is it running? |
| 20:08.47 | batphone | we have to terminate it in one box to make a long story short |
| 20:08.47 | justinu | fender: do you know if there's an easy way to make polycom play a reorder tone when the far end hangs up? my users complain that they can't tell when someone hangs up on them (too lazy to look at the display) |
| 20:08.50 | YoMama | hmm..is there anyway to set an extension as "in use" through the dialplan without actually using it? |
| 20:08.51 | mog_work | yeah i hear that |
| 20:08.54 | justinu | "but the old system did it!" |
| 20:08.55 | batphone | 16 ports total, 16 T1's.. |
| 20:09.01 | [TK]D-Fender | YoMama : GS's dev team seem open to suggestions for firmware updates... |
| 20:09.03 | mog_work | well you can do it batphone |
| 20:09.09 | mog_work | its just not easy |
| 20:09.19 | harryvv | batphone what kind of biz are you running? |
| 20:09.27 | batphone | call center |
| 20:09.37 | harryvv | thats good how long now? |
| 20:09.48 | coppice | unlike most of the phone makers, GS actually control their own software, so they can add stuff easily |
| 20:09.51 | batphone | harryvv, like 6 months |
| 20:09.51 | YoMama | Fender: apparently...i've been paying close attention to the wiki on the GXP...lots of activity |
| 20:10.00 | batphone | harryvv, we just need to start rolling with it |
| 20:10.01 | harryvv | patphone no issues yet? |
| 20:10.11 | harryvv | batphone so it was a test box then |
| 20:10.13 | batphone | harryvv, just this, our other boxes with 2 and 3 cards work fine |
| 20:10.28 | YoMama | u could use BLF if you could "trick" the asterisk box into marking an extension as in use and then setting one of the BLF indicators to use that extension as an indication of call forward |
| 20:10.47 | batphone | harryvv, no we have 4 others but we want to consolidate them into 2 boxes so we can fill up a single rack with a PBX capable of running a small city |
| 20:10.48 | harryvv | so thats 23x16 which is 368 ports |
| 20:10.55 | YoMama | the question is...is there a command to mark an extension as in use? |
| 20:10.59 | batphone | 393 total..yes |
| 20:11.05 | harryvv | k |
| 20:11.09 | batphone | 383.. |
| 20:11.10 | justinu | YoMama: yeah, of course |
| 20:11.16 | YoMama | justinu: what's that? |
| 20:11.17 | justinu | oh, you mean from the dialplan? |
| 20:11.22 | YoMama | justinu: yeah |
| 20:11.23 | justinu | i was thinking at the code level |
| 20:11.27 | YoMama | justinu: haha..no |
| 20:11.34 | harryvv | how did u get 383 |
| 20:11.38 | YoMama | justinu: i'm just learning asterisk..when i know a bit more...i'll help with the code :) |
| 20:11.48 | justinu | YoMama: I could be wrong, but I'm betting that you'll need to modify the source to do what you want |
| 20:11.57 | justinu | do you know C? |
| 20:12.02 | harryvv | what codec are you using batphone? |
| 20:12.10 | YoMama | justinu: i've known C since i was 13..(I'm 32 now) |
| 20:12.16 | YoMama | maybe 14 |
| 20:12.18 | batphone | 1 sec |
| 20:12.29 | justinu | YoMama: ditto (except i'm 29) |
| 20:12.33 | YoMama | i just don't like driver development..but i'm real good at apps |
| 20:12.57 | batphone | no codec |
| 20:12.58 | justinu | asterisk is kinda of a mess, but you'll eventually come to terms with the "structure" |
| 20:13.07 | batphone | in 1 zap channel and out the other |
| 20:13.10 | batphone | plays gsm.. |
| 20:13.14 | batphone | no RTP traffic |
| 20:13.23 | harryvv | mm whats the compression of gsm anyone? |
| 20:13.32 | batphone | no voip communication, all tdm |
| 20:13.33 | [TK]D-Fender | 13kbps |
| 20:13.35 | harryvv | what bandwith? i know its used alot on phone |
| 20:13.38 | YoMama | one thing i still gotta debug is why i cannot hear my * server play the AA message when i call in thru my sip proxy |
| 20:13.51 | harryvv | k |
| 20:14.01 | harryvv | batphone all tdm I see |
| 20:14.03 | YoMama | if it were a firewall problem...you'd be able to hear the message..but u wouldn't be able to respond...not the other way aroudn..whcih is what's happening |
| 20:14.19 | harryvv | since when do call centers use ordinary phones? |
| 20:14.31 | justinu | YoMama: not necissarily |
| 20:15.06 | [TK]D-Fender | harryvv : define "ordinary" |
| 20:15.07 | *** join/#asterisk tdonahue (n=tdonahue@208.51.101.201) |
| 20:15.26 | harryvv | TK, tdm as ordinary tdm phones like what everyone has. |
| 20:15.36 | justinu | POTS phone |
| 20:15.40 | YoMama | justinu: well, something weird is going on |
| 20:15.41 | harryvv | same thing |
| 20:15.41 | justinu | is a better term |
| 20:15.42 | harryvv | :) |
| 20:15.51 | [TK]D-Fender | harryvv : And why not? Agents don't necessarily need much of anything special. A call is a call is a call... |
| 20:15.54 | justinu | YoMama: we could get into debugging it |
| 20:15.56 | YoMama | *sigh*...good ol' debug sip time |
| 20:16.08 | harryvv | yea probebly tru |
| 20:16.09 | harryvv | true |
| 20:16.14 | [TK]D-Fender | VERY true |
| 20:16.35 | [TK]D-Fender | My guys are just spectacularly overequiped now :) |
| 20:16.38 | YoMama | justinu: maybe later :)....i never call into the sip proxy so if it's broked...no big deal...but i'll wanna fix it one day just so i know what the solution is |
| 20:16.42 | [TK]D-Fender | IP 600's the lot of them! |
| 20:16.59 | YoMama | the weird thing is..when the sip proxy functions as a peer..it works great |
| 20:17.04 | *** join/#asterisk splatone (n=splat1@rrcs-24-172-35-197.midsouth.biz.rr.com) |
| 20:17.30 | [TK]D-Fender | 1 reg, 1 line key, 1 call per key, 2 speed dials for PAUSE/UNPAUSE, 3 buddy-watched presence line keeps for their neighbours. |
| 20:17.43 | [TK]D-Fender | justinu : no thanks :D |
| 20:17.53 | justinu | those lusers would never know |
| 20:18.11 | *** join/#asterisk MikeJ[Laptop] (n=ircatjer@d14-69-8-30.try.wideopenwest.com) |
| 20:18.15 | [TK]D-Fender | justinu : sure they would! I make extensive use of the MicroBrowser! |
| 20:18.15 | Nugget | http://flightaware.com/live/flight/N9563Z <-- check out the aircraft type |
| 20:18.22 | [TK]D-Fender | SPECIFCALLY for my call center! |
| 20:18.36 | justinu | nugget: you a pilot? |
| 20:18.58 | Nugget | student pilot, so no, not really. :) |
| 20:19.16 | justinu | cool, it's well worth it |
| 20:19.33 | *** join/#asterisk synthetiq (n=roger@64.201.13.50) |
| 20:19.34 | Nugget | I'm enjoying it. I've got about 40 hours |
| 20:19.47 | justinu | i've got a single/multi engine cert |
| 20:19.53 | justinu | working on Instrument rating |
| 20:19.55 | Nugget | ifr yet? |
| 20:20.09 | Nugget | cool |
| 20:20.22 | *** join/#asterisk kink0 (n=k@62.37.205.161) |
| 20:20.25 | kink0 | buenas noches |
| 20:20.28 | Nugget | it's really not very useful without the instrument rating |
| 20:20.40 | justinu | true, but IFR is a totally different thing |
| 20:20.46 | Nugget | yeah |
| 20:20.55 | kink0 | somebody knows about some hardware to take PRI E1 in one side, and 30 BRI on the other side ? |
| 20:20.56 | Nugget | several of my friends have gone on to get their instrument rating. |
| 20:20.57 | batphone | how does linux work on dual core p4? |
| 20:20.59 | Nugget | I'm just a slacker. :) |
| 20:21.19 | [TK]D-Fender | I'm a Slacker :) |
| 20:21.30 | justinu | kink0: i remembe ra while back, there was channel banks that would take PRI and break it out into BRI |
| 20:21.59 | justinu | but the format of the PRI was slightly funky |
| 20:22.19 | kink0 | justinu, yes, that is what I am seeking, because I bougth quad T1/E1 from digium, and now I need to connect to BRI ports |
| 20:22.34 | justinu | most of us here are in north america |
| 20:22.38 | justinu | so we dunno much about BRI |
| 20:23.08 | harryvv | batphone any echo problems? |
| 20:23.25 | batphone | not that i am aware of |
| 20:23.32 | harryvv | sounds good |
| 20:23.32 | *** join/#asterisk m160858 (n=jsaenz@200.89.12.46) |
| 20:23.34 | justinu | kink0: http://www.betterbox.co.uk/acatalog/Product_List__ISDN_PRI_BRI_Channel_Bank_1157.html |
| 20:23.37 | batphone | harryvv, it turns out that this particular box will be doing transcoding after all |
| 20:23.39 | m160858 | hi everyone |
| 20:23.46 | harryvv | i see |
| 20:23.54 | Seldon1975 | are there examples of dialplans set up so that callers from outside get a recording "Welcome,..." and then can dial an extension number? |
| 20:23.59 | m160858 | i want to know how record the calls out |
| 20:24.15 | harryvv | There is a local city hall that has constant problems with its nortel networks voip system. going down or echo or other problems. |
| 20:25.18 | jake1932 | m160858: http://www.voip-info.org/wiki-Asterisk+record+calls |
| 20:25.24 | jake1932 | take your pick |
| 20:26.49 | [TK]D-Fender | justinu : OUCH on the $ for that channel bank... |
| 20:26.59 | *** join/#asterisk Gunnar (n=gunnar@34.80-203-209.nextgentel.com) |
| 20:27.06 | justinu | yeah, BRI stuff is $$$ |
| 20:27.08 | jake1932 | i wonder what the markup is on those things |
| 20:27.14 | justinu | but that was just the first hit on google |
| 20:27.16 | [TK]D-Fender | I used to have echo.... that really sucked... not not anymore :) Life is good.. |
| 20:27.46 | jake1932 | ..good ... goood.. damnit |
| 20:28.30 | *** part/#asterisk bjohnson (n=bjohnson@jecinc.tor.istop.com) |
| 20:28.39 | jake1932 | [TK]D-Fender: completely gone? |
| 20:28.42 | *** join/#asterisk eKo1 (n=bernd@metrored-gw.tropicohn.com) |
| 20:28.54 | eKo1 | quick ?: what is the purpose of sip_notify? |
| 20:29.07 | justinu | to reboot your phones |
| 20:29.09 | jake1932 | lol |
| 20:29.10 | justinu | resync configs |
| 20:29.13 | jake1932 | not quick enough |
| 20:29.13 | *** join/#asterisk eKo1 (n=bernd@metrored-gw.tropicohn.com) |
| 20:29.18 | [TK]D-Fender | jake1932 : OBLITERATED :) |
| 20:29.30 | justinu | [12:29] justinu: to reboot your phones |
| 20:29.30 | justinu | [12:29] jake1932: lol |
| 20:29.30 | justinu | [12:29] justinu: resync configs |
| 20:29.30 | [TK]D-Fender | I think once in the past month. |
| 20:29.42 | justinu | fender: what was your echo problem? |
| 20:29.45 | jake1932 | <PROTECTED> |
| 20:29.58 | m160858 | i did it, but doesn't work |
| 20:29.59 | [TK]D-Fender | jake1932 : only on ATA's |
| 20:30.06 | jake1932 | ok |
| 20:30.06 | [TK]D-Fender | Running PRI in. |
| 20:30.19 | jake1932 | what did it? |
| 20:30.34 | harryvv | okay what is the reason that cidcw would not be working on my system? |
| 20:30.43 | [TK]D-Fender | jake1932 : My new card :D |
| 20:30.54 | jake1932 | xmas present? |
| 20:31.15 | harryvv | has anyone had issues with cidcw not working on a asterisk system?: |
| 20:31.36 | harryvv | tk, not not support digium? |
| 20:31.50 | *** join/#asterisk zotz (n=zotz@24.231.47.168) |
| 20:32.07 | splatone | is the cisco phones the only ones with the xml interface? |
| 20:32.16 | justinu | polycom uses xml also |
| 20:32.49 | [TK]D-Fender | harryvv : I had 2 TE405P's each being unable to synch with the telco's clock, with constant frame slips, horrible echo, and caused problems with my Intel NIC's. It got to a point where I sad "&^% that..." |
| 20:32.53 | splatone | so you can pull a directory listing on them? |
| 20:32.58 | m160858 | i did this http://www.voip-info.org/wiki-Asterisk+record+calls |
| 20:33.02 | m160858 | but doesn't works |
| 20:33.18 | justinu | fender: i bought an A101u for a client |
| 20:33.23 | justinu | you think i'll have those issues with that card? |
| 20:33.33 | jake1932 | m160858: you did what? there are several options |
| 20:33.36 | harryvv | tkfender okay so this is a totally different company then. |
| 20:33.41 | *** join/#asterisk paryl (n=paryl@216-201-177-82.res.logixcom.net) |
| 20:33.52 | [TK]D-Fender | justinu : Doubt it highly. I also run a Sangoma S518 ADSL card at home on the same WanPipe driver. Its gold... |
| 20:33.57 | m160858 | for web, ln -s /var/spool/asterisk/monitor /var/www/html/recordings/monitor |
| 20:34.02 | justinu | fender: ok, i hope so |
| 20:34.07 | YoMama | who does the absolute cheapest SIP proxy for both inbound/outbound in the US? |
| 20:34.29 | justinu | i specifically bought the Sangoma because of people complaining about the digium cards |
| 20:34.33 | m160858 | into sip.conf ... add the follow option |
| 20:34.36 | m160858 | record_out=Always .. |
| 20:34.37 | [TK]D-Fender | justinu : Its just a few more little steps at the beginning, but after that its "set & forget" |
| 20:34.47 | harryvv | normally a fast busy signal on a DID is indicative of problems within the voip service? |
| 20:34.52 | justinu | yeah, i took a look at the sangoma install instructions ,seemed easy enough |
| 20:34.58 | [TK]D-Fender | And No IRQ problems or PCI voltage concerns, etc... |
| 20:34.58 | paryl | this isn't making sense. i have timeout=10 in queues.conf and timeout=15 in agents.conf... once a queue starts ringing an extension, it neither times out or logs the agent off. |
| 20:35.05 | Seldon1975 | are there examples of dialplans set up so that callers from outside get a recording "Welcome,..." and then can dial an extension number? |
| 20:35.12 | [TK]D-Fender | justinu : I"m a schmuck and I figured it all out. |
| 20:35.22 | [TK]D-Fender | and their tech support is GODLY. |
| 20:35.29 | justinu | fender: well, you were probably one of the guys that convinced me |
| 20:35.29 | funxion | hmm |
| 20:35.45 | funxion | I have had no problems with digium cards at all |
| 20:35.46 | [TK]D-Fender | Cool |
| 20:35.53 | jake1932 | m160858: why didn't you just use Monitor? |
| 20:35.55 | [TK]D-Fender | funxion : more power to you... |
| 20:35.57 | justinu | i've heard some people have zero problems with digium |
| 20:36.05 | justinu | and some people obviously do have problems with digium |
| 20:36.08 | *** join/#asterisk bleck (i=kris@dsl-202-72-161-61.wa.westnet.com.au) |
| 20:36.12 | justinu | but I have yet to hear anyone complain about problems with Sangoma. |
| 20:36.34 | mog_work | ill complain.... |
| 20:36.36 | m160858 | like this exten => _1XXXXXXXXXX,4,Monitor(gsm, ${EXTEN}) ? |
| 20:36.36 | mog_work | ^_^ |
| 20:36.43 | justinu | mog_work: go ahead :P |
| 20:36.44 | m160858 | i try, but nothing yet |
| 20:36.56 | [TK]D-Fender | justinu : And Sangoma has been in the business far longer and their stuff is multi-platform as well. I support commodity telcom in all aspects. |
| 20:37.04 | jake1932 | m160858: does it say it's monitoring? |
| 20:37.04 | justinu | Fender: i'm with you there |
| 20:37.05 | mog_work | they knock asterisk |
| 20:37.10 | mog_work | thats not cool in my book |
| 20:37.14 | mog_work | but meh |
| 20:37.20 | justinu | bias :) |
| 20:37.25 | justinu | it's normal |
| 20:37.52 | bleck | I am trying to get MWI working from asterisk to callmanager express, and i have put in callmanager express's telephony section "mwi sip-server 192.168.1.2 unsolicited", but it still says this (sip debug on asterisk) Call Leg/Transaction Does Not Exist |
| 20:38.03 | [TK]D-Fender | The best part of my entire VoIP setup at the office is no 1 piece of hardware owns my ass.... EVERYTHING is replacable here. |
| 20:38.19 | justinu | do you have spares? |
| 20:38.36 | justinu | i don't even get to work near my equipment... it's 45 miles away |
| 20:38.37 | [TK]D-Fender | justinu : as in spares for every important piece? |
| 20:38.46 | justinu | yeah |
| 20:38.52 | [TK]D-Fender | like T1 cards, extra phones, etc? |
| 20:38.58 | splatone | has anyone seen or used a cellphone with a sip softphone on it w/ wifi. Im looking at purchasing a couple of phones and wondering if this is possible. |
| 20:39.03 | justinu | yeah, or standby servers |
| 20:39.15 | [TK]D-Fender | everything except the card, but I still have my TE405P handy which my vendor didn't pick upyet |
| 20:39.29 | [TK]D-Fender | no standby. Im prepparing for "Plan B" |
| 20:39.31 | bleck | it is annoying, as this page explains the error, and says to put in that line in the telephony section of callmanager and it should work -> http://home.comcast.net/~kurtwp2/cme/telephony.htm |
| 20:39.32 | justinu | splatone: they have that kind of thing in Asia |
| 20:39.49 | harryvv | splatone, if the cell phone has wifi capability and allows you to install third party voip software its possible. |
| 20:39.57 | justinu | Fender: I set up a load balancing SER proxy in from of my * machines (media servers) |
| 20:40.05 | splatone | would the quality suck because its a softphone? |
| 20:40.13 | harryvv | not really |
| 20:40.18 | bleck | anyone got mwi working through cisco equip? |
| 20:40.18 | harryvv | at least on a pc |
| 20:40.28 | [TK]D-Fender | justinu : I'm a liux hack, for which keeping this system running is only part of my job. I'm not "there' yet... |
| 20:40.30 | bleck | s/cisco/cisco callmanager/ |
| 20:40.52 | m160858 | no, i try to use cmd monitor |
| 20:40.55 | [TK]D-Fender | Linux* I know very little, but succeed at a lot surprisingly still |
| 20:41.05 | splatone | we are looking at getting a couple of these http://www.gsmarena.com/i_mate_sp5m-1268.php |
| 20:41.14 | m160858 | i try too, to record from de web |
| 20:41.18 | justinu | Fender: that's because it's intuitive :P |
| 20:41.27 | m160858 | and doesn't record |
| 20:41.59 | jake1932 | m160858: any error when you run the Monitor cmd? |
| 20:42.08 | [TK]D-Fender | justinu : not sure about that... I'll chalk it up to me just being "lucky" |
| 20:42.11 | jake1932 | in the CLI? |
| 20:42.48 | justinu | anyways, ser makes load balancing super easy |
| 20:42.56 | *** join/#asterisk pryk (n=tmalkut@fw.orasoft.net.pl) |
| 20:42.57 | KranZ | justinu: indeed |
| 20:42.58 | m160858 | any error |
| 20:43.04 | justinu | so if you ever want to do such, look into it |
| 20:43.13 | KranZ | justinu: are your * boxes on a private lan? |
| 20:43.15 | m160858 | just doesn't record |
| 20:43.32 | justinu | kranz: yeah, but the SER proxies are on public IPs |
| 20:43.33 | [TK]D-Fender | justinu : my company runs ScopServ's GUI (the only way I could convince them here) and I am preparing a backup plan in case it "dies" on me somehow. Thats my backup plan. New HD w Slackware, *, and enough to keep this place running as close to identically as possible. |
| 20:43.39 | justinu | they live outside the firewall |
| 20:43.49 | KranZ | justinu: as they should |
| 20:43.49 | [TK]D-Fender | I should learn SER sometime.... |
| 20:44.00 | [TK]D-Fender | or something like it. |
| 20:44.07 | justinu | i tried to get SER working on private IPs too |
| 20:44.13 | justinu | but it's not ready for that yet |
| 20:44.15 | KranZ | SER is easier |
| 20:44.36 | justinu | i've also got SER doing SIP TCP to UDP proxy, since * only speaks UDP |
| 20:44.59 | KranZ | what uses tcp? |
| 20:45.05 | justinu | level3 |
| 20:45.18 | m160858 | i have to install .. some soft extra ? |
| 20:45.31 | *** join/#asterisk |dennis| (i=dennis@200.32.215.84) |
| 20:46.17 | p1tst0p | am i able to send notification of voicemail to X-Lite client? |
| 20:47.27 | *** join/#asterisk kavit (n=kavit@ppp167-252-96.static.internode.on.net) |
| 20:48.17 | [TK]D-Fender | p1tst0p : As in MWI? |
| 20:48.29 | Seldon1975 | can someone tell me where to find a Festival RPM? |
| 20:48.32 | p1tst0p | [TK]D-Fender, yep mate WMI |
| 20:48.50 | KranZ | rpm...eww |
| 20:48.52 | azzie | what is this codec? rtpmap: 13 CN/8000 |
| 20:49.06 | [TK]D-Fender | p1tst0p : does that already... |
| 20:49.10 | p1tst0p | i have a friend that uses xlite to my ast box, i wanted to let him know when he had voicemail |
| 20:49.25 | Seldon1975 | Kranz well if theres an easier way to install it... |
| 20:49.34 | [TK]D-Fender | p1tst0p : make sure his mailbox is indicated in sip.conf |
| 20:49.38 | harryvv | I wonder if i can get away with selling some voip connections though my cable :) |
| 20:50.13 | Seldon1975 | anyone know an open SNTP server? |
| 20:50.19 | harryvv | im in canada |
| 20:50.45 | harryvv | TK, obviosly I probebly cannot unless its only for long distance service. |
| 20:51.21 | harryvv | What would cause fast bussys on my DID ? |
| 20:51.24 | p1tst0p | TK, i dont suppose you know if the WMI works for avaya 4602IP phones do you ! |
| 20:51.32 | harryvv | Calling the DID number I get fast bussys |
| 20:51.36 | kavit | I have a X100p clone, after making a call the cpu usage shoots up... is this normal? |
| 20:52.37 | [TK]D-Fender | p1tst0p : not a clue on Avaya |
| 20:52.44 | Seldon1975 | anyone know an open SNTP server? |
| 20:52.53 | Seldon1975 | anyone know where to get a Festival RPM? |
| 20:53.05 | [TK]D-Fender | SNTP? |
| 20:53.07 | *** join/#asterisk darby_t (i=darby_t@dkk25.neoplus.adsl.tpnet.pl) |
| 20:53.15 | Seldon1975 | simple network time server |
| 20:53.47 | [TK]D-Fender | NTPD <- |
| 20:54.08 | *** join/#asterisk stkn (n=stkn@gentoo/developer/pdpc.active.stkn) |
| 20:54.25 | Seldon1975 | oh |
| 20:54.26 | eKo1 | quick ?: what is the purpose of sip_notify? |
| 20:54.29 | Seldon1975 | open servers? |
| 20:54.35 | test34 | Seldon1975, type: time server in google |
| 20:54.40 | Seldon1975 | my Polycom needs a SNTP server |
| 20:54.45 | *** part/#asterisk m160858 (n=jsaenz@200.89.12.46) |
| 20:55.27 | Seldon1975 | not NTPD |
| 20:55.31 | [TK]D-Fender | Seldon1975 : Just point them to pool.ntp.org <- |
| 20:55.39 | Seldon1975 | ok thanks |
| 20:55.47 | [TK]D-Fender | ntpd is a NTP daemon... |
| 20:55.55 | [TK]D-Fender | if you want to do it internally. |
| 20:56.02 | [TK]D-Fender | and yes BOTH work. |
| 20:56.10 | Seldon1975 | aha |
| 20:56.12 | Seldon1975 | great |
| 20:56.40 | YoMama | anyone here use fwdOUT? |
| 20:57.17 | ManxPower | ~docs |
| 20:57.18 | jbot | i heard docs is probably Documentation can be found at http://www.digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or http://www.voip-info.org/wiki-Asterisk or http://www.asteriskdocs.org or http://www.oreilly.com/catalog/asterisk or http://www.asteriskguru.com |
| 20:57.23 | ManxPower | ~mailinglist |
| 20:57.24 | jbot | i guess mailinglist is Search Asterisk mailing lists by prepending site:lists.digium.com to your Google search. Browse the mailing list archive at http://lists.digium.com/ or search through it at http://www.asteriskguru.com/archives/search.php |
| 20:57.36 | YoMama | Manx: hey |
| 20:58.02 | YoMama | i haven't even tried setting it up..it just sounds like a cool concept..i'm wondering if it is actually practical |
| 20:58.55 | KranZ | what cisco phones are anyone using? |
| 20:59.28 | paryl | i asked this above... does anyone have any ideas? : i have timeout=10 in queues.conf and timeout=15 in agents.conf... once a queue starts ringing an extension, it neither times out or logs the agent off. |
| 20:59.53 | ManxPower | KranZ, Polycom |
| 21:00.03 | p1tst0p | TK, setting the mailbox in sip.conf for the avaya phone doesnt indicate the WMI... Oddnes. |
| 21:00.06 | paryl | actually.. the second one is autologoff = 15 |
| 21:00.15 | [TK]D-Fender | paryl : there's another option that kicks out agents who don't answer. Those options only control how long till it tries the next guy... |
| 21:00.29 | [TK]D-Fender | p1tst0p : Using SIP? |
| 21:00.35 | p1tst0p | [TK]D-Fender yup |
| 21:00.52 | [TK]D-Fender | p1tst0p : Can't speak for them... largely propriety BS... |
| 21:00.53 | *** join/#asterisk SeanSmith44502 (i=SeanSmit@phnxapanas75poola89.phnx.uswest.net) |
| 21:01.36 | paryl | [TK]D-Fender: yeah, i meant autologoff. but timeout doesn't seem to be working. the call comes out of moh, starts ringing the extension, and just rings indefinitely |
| 21:01.51 | p1tst0p | [TK]D-Fender, ya, i work with avaya pbx's.... i re imaged this 4602 to be sip compatable.. only thing i dont see working is WMI. |
| 21:02.05 | SeanSmith44502 | Is there any reason I would get a bunch of static when calling a FXO port from a FXS on the same card...Asterisk says it is a "Native Bridge" |
| 21:02.34 | ManxPower | SeanSmith44502, native bridge has nothing to do with your problem. Call Digium, you may have a bad port or module. |
| 21:03.04 | SeanSmith44502 | ManxPower even though FXS to SIP is fine and SIP to FXO is fine? |
| 21:03.12 | ManxPower | SeanSmith44502, Correct. |
| 21:03.16 | Katty | if the /entire/ conversation is echoy, which value do i need to change? |
| 21:03.18 | Katty | tx or rx? |
| 21:03.26 | SeanSmith44502 | ManxPower...Thanks. |
| 21:03.27 | ManxPower | Katty, tx in the direction of the PSTN |
| 21:03.43 | ManxPower | also echocancel=yes and echotraining-900 |
| 21:03.49 | ManxPower | =900 that is |
| 21:04.06 | SeanSmith44502 | ManxPower....Do you know if it is possible to break into a bridged call? |
| 21:04.18 | ManxPower | SeanSmith44502, Define "break into". |
| 21:04.21 | [TK]D-Fender | SeanSmith44502 : ZapBarge |
| 21:04.26 | *** part/#asterisk kredford (n=merida@201.138.18.144) |
| 21:05.12 | KranZ | SeanSmith44502: sure, i've done it |
| 21:05.34 | *** join/#asterisk P4C0 (n=nousefor@200.124.22.34) |
| 21:05.35 | ManxPower | If you mean "connect to the call to listen and talk" the answer is "no" |
| 21:05.40 | P4C0 | hello guys |
| 21:05.40 | SeanSmith44502 | ManxPower...I have my system configured PTSN->Panasonic 624 PBX->Asterisk....when call comes in asterisk sends to SIP channel....SIP user answers the phone but would like to transfer to another user on the PBX. |
| 21:05.46 | ManxPower | IF you mean "listen to the call without being able to talk" then the answer is yesd |
| 21:06.16 | ManxPower | SeanSmith44502, then you press the TRANSFER button on your SIP device. |
| 21:06.27 | ManxPower | Check the docs for your SIP device, of course. |
| 21:06.43 | ManxPower | And that is not called "break into" thats called "transferring" |
| 21:07.51 | SeanSmith44502 | ManxPower...when hitting transfer on my SIP device (Grandstream b101...i know....not the best choice) I cannot seem to get it to work. |
| 21:08.02 | P4C0 | hello guys I'm planning to put an asterisk server inside a nat.. (I have access to the nat/firewall device) I have read on asteriskguru and similar about nat issue with asterisk but I'm not sure what ports should I redirect or open? any urls or suggestions will be appreciated |
| 21:08.21 | SeanSmith44502 | ManxPower...I need to have the ZAP FXO Send a flash and then dial the 3 digit extn and hangup. |
| 21:09.13 | [TK]D-Fender | P4C0 : Forward UDP 5060, 10000-20000 to your box, add your public ip to "externip" in sip.conf and configure "localnet" there too. |
| 21:09.44 | Katty | ManxPower: i think i'll keep my echocancel=64 (= |
| 21:09.46 | bsdfreak | anyone here have a sipura 3000 answering pstn calls? |
| 21:10.00 | [TK]D-Fender | bsdfreak : I do. |
| 21:10.15 | bsdfreak | have you ever had someone on the other end say something and the sipura thought the vocal was a dtmf tone and echoed it? |
| 21:10.42 | P4C0 | [TK]D-Fender, and that's all? |
| 21:10.47 | [TK]D-Fender | nope, can't say that I have, though I haven't gotten that much PSTN use out of mine yet |
| 21:10.54 | bsdfreak | oh ok |
| 21:10.57 | bsdfreak | =\ |
| 21:10.57 | [TK]D-Fender | P4C0 : Thats pretty much it,. |
| 21:11.09 | P4C0 | [TK]D-Fender, all udp no tcp? |
| 21:11.18 | file[desk] | meep |
| 21:11.32 | bsdfreak | well guess i'll have to figure that out |
| 21:11.33 | bsdfreak | heh |
| 21:11.49 | bsdfreak | i know it's the sipura doing it tho |
| 21:12.17 | [TK]D-Fender | I think theres an option in there to tweak the DTMF sensitivity or something like that... |
| 21:12.40 | bsdfreak | hmm |
| 21:12.41 | bsdfreak | good call |
| 21:13.30 | [TK]D-Fender | :) |
| 21:13.40 | [TK]D-Fender | P4C0: not for SIP |
| 21:13.43 | *** join/#asterisk Sedorox (i=brandon@smartserv/cna/Sedorox) |
| 21:14.12 | P4C0 | [TK]D-Fender, thanks, what about stun server? do I need that? |
| 21:16.46 | KranZ | P4C0: doesnt hurt |
| 21:23.20 | [TK]D-Fender | no need. |
| 21:23.42 | [TK]D-Fender | It CAN help in certain cases but * doesn't do STUN |
| 21:29.29 | *** join/#asterisk anthm (n=anthmct@CPE-69-76-83-52.wi.res.rr.com) [NETSPLIT VICTIM] |
| 21:29.29 | *** join/#asterisk Sedorox (i=brandon@smartserv/cna/Sedorox) [NETSPLIT VICTIM] |
| 21:29.29 | *** join/#asterisk P4C0 (n=nousefor@200.124.22.34) [NETSPLIT VICTIM] |
| 21:29.29 | *** join/#asterisk stkn (n=stkn@gentoo/developer/pdpc.active.stkn) [NETSPLIT VICTIM] |
| 21:29.30 | *** join/#asterisk kavit (n=kavit@ppp167-252-96.static.internode.on.net) [NETSPLIT VICTIM] |
| 21:29.30 | *** join/#asterisk |dennis| (i=dennis@200.32.215.84) [NETSPLIT VICTIM] |
| 21:29.30 | *** join/#asterisk paryl (n=paryl@216-201-177-82.res.logixcom.net) [NETSPLIT VICTIM] |
| 21:29.30 | *** join/#asterisk zotz (n=zotz@24.231.47.168) [NETSPLIT VICTIM] |
| 21:29.30 | *** join/#asterisk eKo1 (n=bernd@metrored-gw.tropicohn.com) [NETSPLIT VICTIM] |
| 21:29.30 | *** join/#asterisk synthetiq (n=roger@64.201.13.50) [NETSPLIT VICTIM] |
| 21:29.30 | *** join/#asterisk splatone (n=splat1@rrcs-24-172-35-197.midsouth.biz.rr.com) [NETSPLIT VICTIM] |
| 21:29.30 | *** join/#asterisk tdonahue (n=tdonahue@208.51.101.201) [NETSPLIT VICTIM] |
| 21:29.31 | *** join/#asterisk batphone (n=batphone@69.15.174.114) [NETSPLIT VICTIM] |
| 21:29.31 | *** join/#asterisk fulgas (n=fulgas@a81-84-117-84.cpe.netcabo.pt) [NETSPLIT VICTIM] |
| 21:29.31 | *** join/#asterisk zapa (n=zapa@200.39.202.5) [NETSPLIT VICTIM] |
| 21:29.31 | *** join/#asterisk zeedo (n=zeedo@reboot-robot.net) [NETSPLIT VICTIM] |
| 21:29.31 | *** join/#asterisk Seldon1975 (n=someone@gatekeeper.radintl.com) [NETSPLIT VICTIM] |
| 21:29.31 | *** join/#asterisk azop (i=curt@twisted.bluecherry.net) [NETSPLIT VICTIM] |
| 21:29.31 | *** join/#asterisk gnosys (n=ksford@ip68-9-201-250.ri.ri.cox.net) [NETSPLIT VICTIM] |
| 21:29.31 | *** join/#asterisk ReD-MaN (i=redman@dhcp-0-2-b3-9a-4a-5b.cpe.quickclic.net) [NETSPLIT VICTIM] |
| 21:29.32 | *** join/#asterisk ard (n=ard@2001:7b8:32d:0:20c:6eff:fe18:d11f) [NETSPLIT VICTIM] |
| 21:29.32 | *** join/#asterisk florz (i=nobody@2001:1a50:503c:0:0:0:0:1) [NETSPLIT VICTIM] |
| 21:29.32 | *** join/#asterisk Cinen (n=Cinen@vpn.triadtelecom.com) [NETSPLIT VICTIM] |
| 21:29.32 | *** join/#asterisk jake1932 (n=jake1932@pool-68-162-123-228.phil.east.verizon.net) [NETSPLIT VICTIM] |
| 21:29.32 | *** join/#asterisk arguile (i=user224@66.38.201.234) [NETSPLIT VICTIM] |
| 21:29.32 | *** join/#asterisk RaYmAn-Bx (i=rayman@x1-6-00-40-63-da-39-3f.k191.webspeed.dk) [NETSPLIT VICTIM] |
| 21:29.32 | *** join/#asterisk drbrown (n=keith@user-0cdvefr.cable.mindspring.com) [NETSPLIT VICTIM] |
| 21:29.33 | *** join/#asterisk keith80403 (n=keith804@24-56-188-49.co.warpdriveonline.com) [NETSPLIT VICTIM] |
| 21:29.33 | *** join/#asterisk Utah_Dave (n=boucha@0-1pool139-158.nas28.salt-lake-city1.ut.us.da.qwest.net) [NETSPLIT VICTIM] |
| 21:29.33 | *** join/#asterisk Math` (n=math@modemcable148.4-81-70.mc.videotron.ca) [NETSPLIT VICTIM] |
| 21:29.33 | *** join/#asterisk nagl (n=nagl@213.235.241.6) [NETSPLIT VICTIM] |
| 21:29.33 | *** join/#asterisk KranZ (n=user@sme.bestline.net) [NETSPLIT VICTIM] |
| 21:29.33 | *** join/#asterisk jeffik (n=Jeff@CPE0011505c92d3-CM014350000760.cpe.net.cable.rogers.com) [NETSPLIT VICTIM] |
| 21:29.33 | *** join/#asterisk jaiger (n=jaiger@fire.innovationsw.com) [NETSPLIT VICTIM] |
| 21:29.34 | *** join/#asterisk _Madar (n=tophe@219-84-129-163-adsl-tpe.static.so-net.net.tw) [NETSPLIT VICTIM] |
| 21:29.34 | *** join/#asterisk A-Tuin (n=a-tuin@steves.ip.v4.me.uk) [NETSPLIT VICTIM] |
| 21:29.34 | *** join/#asterisk sysdebug (n=sysdebug@200.163.193.247) [NETSPLIT VICTIM] |
| 21:29.34 | *** join/#asterisk Hmmhesays (n=Neg@72.24.227.83) [NETSPLIT VICTIM] |
| 21:29.34 | *** join/#asterisk azid (n=janne@hus051a.gronstenen.se) [NETSPLIT VICTIM] |
| 21:29.34 | *** join/#asterisk santiago (n=santiago@208.195.215.160) [NETSPLIT VICTIM] |
| 21:29.34 | *** join/#asterisk bjohnson_ (n=bjohnson@jecinc.tor.istop.com) [NETSPLIT VICTIM] |
| 21:29.34 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) [NETSPLIT VICTIM] |
| 21:29.35 | *** join/#asterisk puzzled (n=patrick@puzzled.xs4all.nl) [NETSPLIT VICTIM] |
| 21:29.35 | *** join/#asterisk javar (n=javar@69.79.133.185) [NETSPLIT VICTIM] |
| 21:29.35 | *** join/#asterisk lehel (n=lehel@82.79.20.17) [NETSPLIT VICTIM] |
| 21:29.35 | *** join/#asterisk amir_ (n=amir@gentoo/developer/amir) [NETSPLIT VICTIM] |
| 21:29.35 | *** join/#asterisk locksy (n=locksy@mrtg.sisgroup.com.au) [NETSPLIT VICTIM] |
| 21:29.35 | *** join/#asterisk postwait (n=jesus@catapult.omniti.com) [NETSPLIT VICTIM] |
| 21:29.35 | *** join/#asterisk [TK]D-Fender (n=joe@toronto-HSE-ppp4122655.sympatico.ca) [NETSPLIT VICTIM] |
| 21:29.35 | *** mode/#asterisk [+o anthm] by irc.freenode.net |
| 21:29.36 | *** join/#asterisk wunderkin (i=kev@ip68-226-113-228.ph.ph.cox.net) [NETSPLIT VICTIM] |
| 21:29.36 | *** join/#asterisk Brumle (n=brumle@brumle.com) [NETSPLIT VICTIM] |
| 21:29.36 | *** join/#asterisk Junbug (n=Administ@lan.webunited.net) [NETSPLIT VICTIM] |
| 21:29.36 | *** join/#asterisk taec (n=phil@eventhorizon.hosting365.ie) [NETSPLIT VICTIM] |
| 21:29.36 | *** join/#asterisk chapeaurouge (n=chap@85.201.81.201) [NETSPLIT VICTIM] |
| 21:29.36 | *** join/#asterisk af_ (n=af@ip-142-84.sn1.eutelia.it) [NETSPLIT VICTIM] |
| 21:29.36 | *** join/#asterisk lorinc (n=ang@caracas-1492.adsl.interware.hu) [NETSPLIT VICTIM] |
| 21:29.36 | *** join/#asterisk benjk (n=benjamin@f8a01-0357.din.or.jp) [NETSPLIT VICTIM] |
| 21:29.37 | *** join/#asterisk sigterm (i=sigterm@devious.info) [NETSPLIT VICTIM] |
| 21:29.37 | *** join/#asterisk krisguy (n=krisguy@h216-170-039-057.adsl.navix.net) [NETSPLIT VICTIM] |
| 21:29.37 | *** join/#asterisk scardinal (n=supreme@port816.ds1-suoe.adsl.cybercity.dk) [NETSPLIT VICTIM] |
| 21:29.37 | *** join/#asterisk kuku5 (i=kuku@c-67-175-218-223.hsd1.il.comcast.net) [NETSPLIT VICTIM] |
| 21:29.37 | *** join/#asterisk IronHelix (n=irc@ool-45785cfe.dyn.optonline.net) [NETSPLIT VICTIM] |
| 21:29.37 | *** join/#asterisk wasim (n=wasim@pdpc/supporter/active/wasim) [NETSPLIT VICTIM] |
| 21:29.37 | *** join/#asterisk morale (i=russell@S010600111155e117.cg.shawcable.net) [NETSPLIT VICTIM] |
| 21:29.37 | *** join/#asterisk bmg505 (n=leon@dsl-146-31-32.telkomadsl.co.za) [NETSPLIT VICTIM] |
| 21:29.38 | *** join/#asterisk Syrus_ (n=pascal@tahiti.mpl.rullier.net) [NETSPLIT VICTIM] |
| 21:29.38 | *** join/#asterisk Insanity5 (n=feaw@ip68-111-5-23.sv.om.cox.net) [NETSPLIT VICTIM] |
| 21:29.38 | *** join/#asterisk kFuQ (n=somedude@c-67-185-114-199.hsd1.wa.comcast.net) [NETSPLIT VICTIM] |
| 21:29.38 | *** join/#asterisk }btorch{ (n=kvirc@208.63.19.172) [NETSPLIT VICTIM] |
| 21:29.38 | *** join/#asterisk De_Mon (n=de_mon@fl-69-69-147-198.dyn.sprint-hsd.net) [NETSPLIT VICTIM] |
| 21:29.38 | *** join/#asterisk tessier (n=treed@rrcs-67-53-110-66.west.biz.rr.com) [NETSPLIT VICTIM] |
| 21:29.38 | *** join/#asterisk mog_work (n=mogorman@gateway.digium.com) [NETSPLIT VICTIM] |
| 21:29.38 | *** join/#asterisk InfideNino (i=InfideNi@164-233.surfsnel.dsl.internl.net) [NETSPLIT VICTIM] |
| 21:29.39 | *** join/#asterisk Meaty (n=cp_simbu@office.abi.ca) [NETSPLIT VICTIM] |
| 21:29.39 | *** join/#asterisk tainted- (n=somewher@mail.k2usa.com) [NETSPLIT VICTIM] |
| 21:29.39 | *** join/#asterisk toddf (n=toddf@ns0.fries.net) [NETSPLIT VICTIM] |
| 21:29.39 | *** join/#asterisk Koshatul (n=evangeli@ip157-65-132.cust.bit.net.au) [NETSPLIT VICTIM] |
| 21:29.39 | *** join/#asterisk vexorg (n=vexorg@CPE000021ded913-CM001225419164.cpe.net.cable.rogers.com) [NETSPLIT VICTIM] |
| 21:29.39 | *** join/#asterisk devel (n=devel@wiggum.digitalcoven.com) [NETSPLIT VICTIM] |
| 21:29.39 | *** join/#asterisk Romik_ (n=romik_@1.fix.netvision.net.il) [NETSPLIT VICTIM] |
| 21:29.40 | *** join/#asterisk mountie (n=mountie@CPEdeaddeaddead-CM000a739acaa4.cpe.net.cable.rogers.com) [NETSPLIT VICTIM] |
| 21:29.40 | *** join/#asterisk paljas (n=paljas@tuxtown.xs4all.nl) [NETSPLIT VICTIM] |
| 21:29.40 | *** join/#asterisk waddy (n=waddy@83.218.4.231) [NETSPLIT VICTIM] |
| 21:29.40 | *** join/#asterisk bkw__ (n=brian@68.32.112.142) [NETSPLIT VICTIM] |
| 21:29.40 | *** join/#asterisk destructure (n=irish@rrcs-24-173-126-174.se.biz.rr.com) [NETSPLIT VICTIM] |
| 21:29.40 | *** join/#asterisk brettnem (n=brettnem@72.29.102.158) [NETSPLIT VICTIM] |
| 21:29.40 | *** join/#asterisk xheliox (n=jeff@user-0c6s3h2.cable.mindspring.com) [NETSPLIT VICTIM] |
| 21:29.40 | *** join/#asterisk Mother (n=mother@93.Red-80-32-127.staticIP.rima-tde.net) [NETSPLIT VICTIM] |
| 21:29.41 | *** join/#asterisk Dibbler_ (n=Dibbler@snaddy.plus.com) [NETSPLIT VICTIM] |
| 21:29.41 | *** join/#asterisk brimstone (n=brimston@pdpc/sponsor/digium/brimstone) [NETSPLIT VICTIM] |
| 21:29.41 | *** join/#asterisk vaewyn (i=freeman@mail.parrishmachine.com) [NETSPLIT VICTIM] |
| 21:29.41 | *** join/#asterisk f0urtyfive (n=noone@pcp02538578pcs.crosky01.pa.comcast.net) [NETSPLIT VICTIM] |
| 21:29.41 | *** join/#asterisk wundaboy (n=asdf@c-67-164-107-68.hsd1.or.comcast.net) [NETSPLIT VICTIM] |
| 21:29.41 | *** join/#asterisk mjmac (n=mjmac@pdpc/supporter/active/mjmac) [NETSPLIT VICTIM] |
| 21:29.41 | *** join/#asterisk cypromis (n=michael@asterisk.pl) [NETSPLIT VICTIM] |
| 21:29.41 | *** join/#asterisk Juggie (i=agony@CPE00c049d9f271-CM00137186c8d8.cpe.net.cable.rogers.com) [NETSPLIT VICTIM] |
| 21:29.42 | *** join/#asterisk niZon (n=ilt@S0106deadbeefbeef.wp.shawcable.net) [NETSPLIT VICTIM] |
| 21:29.42 | *** join/#asterisk Smi|k (n=smilk@adsl-66-159-200-157.dslextreme.com) [NETSPLIT VICTIM] |
| 21:29.42 | *** join/#asterisk perlmonkee (i=oginvu@c-24-22-125-15.hsd1.or.comcast.net) [NETSPLIT VICTIM] |
| 21:29.42 | *** join/#asterisk p1tst0p (n=admin@82-38-106-54.cable.ubr03.donc.blueyonder.co.uk) [NETSPLIT VICTIM] |
| 21:29.42 | *** join/#asterisk jtodd (n=jtodd@ti.fox-den.com) [NETSPLIT VICTIM] |
| 21:29.42 | *** join/#asterisk Godsey (n=admin@pdpc/supporter/sustaining/Godsey) [NETSPLIT VICTIM] |
| 21:29.42 | *** join/#asterisk upsite (n=upsite@wls.swh.uni-halle.de) [NETSPLIT VICTIM] |
| 21:29.43 | *** join/#asterisk dogtanian (i=dogtania@eye.ham.zee.walroos.com) [NETSPLIT VICTIM] |
| 21:29.43 | *** join/#asterisk stevek (n=stevek@slim-eth0.horizonlive.net) [NETSPLIT VICTIM] |
| 21:29.43 | *** join/#asterisk rjuan (n=rjuan@194.224.169.27) [NETSPLIT VICTIM] |
| 21:29.43 | *** join/#asterisk j4m3s (n=debbie@gateway.digium.com) [NETSPLIT VICTIM] |
| 21:29.43 | *** join/#asterisk MGSsancho (n=user@adsl-67-126-140-54.dsl.irvnca.pacbell.net) [NETSPLIT VICTIM] |
| 21:29.43 | *** join/#asterisk mrtwister (n=andrius@cable-9-42.cgates.lt) [NETSPLIT VICTIM] |
| 21:29.43 | *** join/#asterisk veto (i=mdkuser@cpe-66-69-38-192.satx.res.rr.com) [NETSPLIT VICTIM] |
| 21:29.43 | *** join/#asterisk Gerriall (n=NonYa@209.42.198.18) [NETSPLIT VICTIM] |
| 21:29.44 | *** join/#asterisk steff (n=steff@80.125.254.220) [NETSPLIT VICTIM] |
| 21:29.44 | *** join/#asterisk Wi_Fi (n=OUT@c-24-127-12-85.hsd1.ca.comcast.net) [NETSPLIT VICTIM] |
| 21:29.44 | *** join/#asterisk fenlander (n=fenlande@82.152.81.57) [NETSPLIT VICTIM] |
| 21:29.44 | *** join/#asterisk redeer (n=redeer@host241-43.pool8172.interbusiness.it) [NETSPLIT VICTIM] |
| 21:29.44 | *** join/#asterisk enota (i=dimka@freelsd.net) [NETSPLIT VICTIM] |
| 21:29.44 | *** join/#asterisk Qwell (n=north@24-205-180-81.dhcp.wsco.ca.charter.com) [NETSPLIT VICTIM] |
| 21:29.44 | *** join/#asterisk Jesster (i=jesster@jesster.org) [NETSPLIT VICTIM] |
| 21:29.45 | *** join/#asterisk Faithful (n=Faithful@202-6-145-116.ip.adam.com.au) [NETSPLIT VICTIM] |
| 21:29.45 | *** join/#asterisk docelm0 (n=brian@static-71-251-95-3.tampfl.fios.verizon.net) [NETSPLIT VICTIM] |
| 21:29.45 | *** join/#asterisk izo (n=izo@193.202.114.43) [NETSPLIT VICTIM] |
| 21:29.45 | *** join/#asterisk trixter (n=trixter@65.172.209.246) [NETSPLIT VICTIM] |
| 21:29.45 | *** join/#asterisk tainted_ (n=identd@71.143.144.147) [NETSPLIT VICTIM] |
| 21:29.45 | *** join/#asterisk _Thor (i=_Thor@user-vc8fl7n.biz.mindspring.com) [NETSPLIT VICTIM] |
| 21:29.45 | *** join/#asterisk denon (i=denon@synapse.subneural.net) [NETSPLIT VICTIM] |
| 21:29.45 | *** join/#asterisk malcolmd (n=malcolmd@pdpc/sponsor/digium/malcolmd) [NETSPLIT VICTIM] |
| 21:29.46 | *** join/#asterisk dsfr (n=dsfr@pdpc/sponsor/digium/dsfr) [NETSPLIT VICTIM] |
| 21:29.46 | *** join/#asterisk DrukenWrk (i=Druken@67.69.139.226) [NETSPLIT VICTIM] |
| 21:29.46 | *** join/#asterisk Katty (n=angela@68-112-15-110.dhcp.cpgr.mo.charter.com) [NETSPLIT VICTIM] |
| 21:29.46 | *** join/#asterisk LapTop006 (n=laptop00@sparc006.chriskaine.com.au) [NETSPLIT VICTIM] |
| 21:29.46 | *** join/#asterisk syle (n=blah@unaffiliated/syle) [NETSPLIT VICTIM] |
| 21:29.46 | *** join/#asterisk Sebbbbb (n=sebastia@einstein.f0o.de) [NETSPLIT VICTIM] |
| 21:29.46 | *** join/#asterisk duckz (n=duckz@193.192.46.26) [NETSPLIT VICTIM] |
| 21:29.46 | *** join/#asterisk g0mb0 (n=test@external.micom.mng.net) [NETSPLIT VICTIM] |
| 21:29.47 | *** join/#asterisk welby (n=welby@tollcross.wheely-bin.co.uk) [NETSPLIT VICTIM] |
| 21:29.47 | *** join/#asterisk espenh (i=holm@blackedge.org) [NETSPLIT VICTIM] |
| 21:29.47 | *** join/#asterisk file[laptop] (n=jcolp@mctnnbsa24w-142167049176.nb.aliant.net) [NETSPLIT VICTIM] |
| 21:29.47 | *** join/#asterisk genmud (n=genmud@ip68-98-82-206.ph.ph.cox.net) [NETSPLIT VICTIM] |
| 21:29.47 | *** join/#asterisk gst (i=gst@eris.sysfrog.org) [NETSPLIT VICTIM] |
| 21:29.47 | *** join/#asterisk mutilator (n=animenod@65.111.201.79) [NETSPLIT VICTIM] |
| 21:29.47 | *** join/#asterisk Damin (n=damin@nucleus.nacs.net) [NETSPLIT VICTIM] |
| 21:29.48 | *** join/#asterisk ^Howler (n=wwolfe@12.33.170.149) [NETSPLIT VICTIM] |
| 21:29.48 | *** join/#asterisk emrah (n=emrah@knsrv1-zrh8048.net1.kavun.ch) [NETSPLIT VICTIM] |
| 21:29.48 | *** join/#asterisk xkev (n=kevin@orbit.xmission.com) [NETSPLIT VICTIM] |
| 21:29.48 | *** join/#asterisk kippi (n=chris@untrust-gct.equinoxit.net) [NETSPLIT VICTIM] |
| 21:29.48 | *** join/#asterisk FaithX (n=FaithX@202-6-145-116.ip.adam.com.au) [NETSPLIT VICTIM] |
| 21:29.48 | *** join/#asterisk azzie (n=az@azzie.net) [NETSPLIT VICTIM] |
| 21:29.48 | *** join/#asterisk {zombie} (i=zombie@soulasylum.penguincare.com.au) [NETSPLIT VICTIM] |
| 21:29.48 | *** join/#asterisk SwK[Work] (n=SwK@64.89.118.139) [NETSPLIT VICTIM] |
| 21:29.49 | *** join/#asterisk tclark_ (n=TC@S0106000f66c5d294.gv.shawcable.net) [NETSPLIT VICTIM] |
| 21:29.49 | *** join/#asterisk maik (n=maik@bfs.cs.uni-sb.de) [NETSPLIT VICTIM] |
| 21:29.49 | *** join/#asterisk DougRoyer (i=doug@70-67.69-92-cpe.cableone.net) [NETSPLIT VICTIM] |
| 21:29.49 | *** mode/#asterisk [+o denon] by irc.freenode.net |
| 21:29.49 | *** join/#asterisk greendisease (n=jack@fedora/greendisease) [NETSPLIT VICTIM] |
| 21:29.49 | *** join/#asterisk heroine (n=heroine@www.helheim.net) [NETSPLIT VICTIM] |
| 21:29.49 | *** join/#asterisk iCEBrkr (n=icebrkr@6532244hfc169.tampabay.res.rr.com) [NETSPLIT VICTIM] |
| 21:29.49 | *** join/#asterisk kll (i=kll@85.195.4.225) [NETSPLIT VICTIM] |
| 21:29.49 | *** join/#asterisk VxJasonxV (n=jason@unaffiliated/VxJasonxV) [NETSPLIT VICTIM] |
| 21:29.50 | *** join/#asterisk Exion (n=jon@odin.utilityisp.com) [NETSPLIT VICTIM] |
| 21:29.50 | *** join/#asterisk joelsolanki (i=joelsola@202.160.161.93) [NETSPLIT VICTIM] |
| 21:29.50 | *** join/#asterisk h3x0r (n=h3xor@64.192.116.16) [NETSPLIT VICTIM] |
| 21:29.50 | *** join/#asterisk sbingner (n=thanotos@pdpc/supporter/sustaining/sbingner) [NETSPLIT VICTIM] |
| 21:29.50 | *** join/#asterisk alrs (n=lars@69-160-242-101.vnnyca.adelphia.net) [NETSPLIT VICTIM] |
| 21:29.50 | *** join/#asterisk Fendor_ (i=yotam@line105-64.adsl.actcom.co.il) [NETSPLIT VICTIM] |
| 21:29.50 | *** join/#asterisk twisted (n=twisted@asterisk/friend-and-developer/pdpc.professional.twisted) [NETSPLIT VICTIM] |
| 21:29.51 | *** join/#asterisk dokhench (n=dochench@adsl-156-13-196.bna.bellsouth.net) [NETSPLIT VICTIM] |
| 21:29.51 | *** join/#asterisk loud (n=ariel@cypher.punk.net) [NETSPLIT VICTIM] |
| 21:29.51 | *** join/#asterisk newsmafia (n=paulr@wsip-68-15-19-142.sd.sd.cox.net) [NETSPLIT VICTIM] |
| 21:29.51 | *** join/#asterisk hugo-v6 (n=hugo@ns1.bundesunixminister.de) [NETSPLIT VICTIM] |
| 21:29.51 | *** join/#asterisk asteriskgeeks (n=SIPdawg@pbxtech.com) [NETSPLIT VICTIM] |
| 21:30.55 | *** join/#asterisk docelmo (n=docelmo@66.237.242.41.ptr.us.xo.net) [NETSPLIT VICTIM] |
| 21:30.55 | *** join/#asterisk brookshire (n=nubb@gateway.digium.com) [NETSPLIT VICTIM] |
| 21:30.55 | *** join/#asterisk jontow (i=jontow@secure-bsd.be) |
| 21:30.55 | *** join/#asterisk knight_ (i=[U2FsdGV@blackhole.phunc.com) [NETSPLIT VICTIM] |
| 21:30.55 | *** join/#asterisk MajestiK (n=MajestiK@S0106000024c058cc.ed.shawcable.net) [NETSPLIT VICTIM] |
| 21:30.56 | *** join/#asterisk ben_d (n=ben@cpe-66-66-209-96.rochester.res.rr.com) [NETSPLIT VICTIM] |
| 21:30.56 | *** join/#asterisk MstlyHrmls (n=mh@melbourne.mostly-harmless.ca) [NETSPLIT VICTIM] |
| 21:30.56 | *** join/#asterisk marv[work] (n=timr@border0hsv.asterisksgi.com) [NETSPLIT VICTIM] |
| 21:30.56 | *** join/#asterisk macTijn (i=martijn@linda.net.insecure.nl) [NETSPLIT VICTIM] |
| 21:30.56 | *** join/#asterisk many (i=many@krikkit.ukeer.de) [NETSPLIT VICTIM] |
| 21:30.56 | *** join/#asterisk Falle (i=falstaf@voip-forum.se) [NETSPLIT VICTIM] |
| 21:30.56 | *** join/#asterisk GXTi (n=scuba@freenode/developer/GXTi) [NETSPLIT VICTIM] |
| 21:30.56 | *** join/#asterisk jwr (i=[sU70RO3@202-44-174-4.nexnet.net.au) [NETSPLIT VICTIM] |
| 21:30.57 | *** join/#asterisk hypa7ia (i=hypatia@wsip-24-234-241-145.lv.lv.cox.net) [NETSPLIT VICTIM] |
| 21:30.57 | *** join/#asterisk sivana (n=sivana@mixdown.ca) [NETSPLIT VICTIM] |
| 21:30.57 | *** join/#asterisk kiko69 (n=Keith@kauai.sys.pas.earthlink.net) [NETSPLIT VICTIM] |
| 21:30.57 | *** join/#asterisk SkramX (n=skramy@vistech.org) [NETSPLIT VICTIM] |
| 21:30.57 | *** join/#asterisk ST-3 (i=ser@dipsy.tch.org) [NETSPLIT VICTIM] |
| 21:30.57 | *** join/#asterisk msw (n=msw@rdu-nat.rpath.com) [NETSPLIT VICTIM] |
| 21:30.57 | *** join/#asterisk lilo (i=levin@freenode/staff/pdpc.levin) |
| 21:30.58 | *** join/#asterisk danalien (n=danalien@unaffiliated/danalien) [NETSPLIT VICTIM] |
| 21:30.58 | *** join/#asterisk Delvar (n=irc@host-83-146-53-34.bulldogdsl.com) [NETSPLIT VICTIM] |
| 21:30.58 | *** join/#asterisk Poincare (n=jefffnod@195.207.137.89) [NETSPLIT VICTIM] |
| 21:30.58 | *** join/#asterisk feist (n=feist@nat-pool-msp.redhat.com) [NETSPLIT VICTIM] |
| 21:30.58 | *** join/#asterisk jarrod (i=jarrod@juniperyour.net) [NETSPLIT VICTIM] |
| 21:30.58 | *** join/#asterisk Money5ack (i=moneysac@wer.will.spontanficken.de) [NETSPLIT VICTIM] |
| 21:30.58 | *** join/#asterisk nitram (i=foo@superblob.com) [NETSPLIT VICTIM] |
| 21:30.58 | *** join/#asterisk flynux (i=n85yb89@pingou.in) [NETSPLIT VICTIM] |
| 21:30.59 | *** join/#asterisk andyd (n=andy@elvis.nosignal.org) [NETSPLIT VICTIM] |
| 21:30.59 | *** join/#asterisk joe (n=nnnnnnnj@ip66-107-33-195.z33-107-66.customer.algx.net) [NETSPLIT VICTIM] |
| 21:30.59 | *** join/#asterisk JonR800 (i=jon@p1mp.org) [NETSPLIT VICTIM] |
| 21:30.59 | *** join/#asterisk rocket (n=rocket@gentoo/developer/rocket) [NETSPLIT VICTIM] |
| 21:30.59 | *** join/#asterisk zigman (i=zigman@irc.zigman.de) [NETSPLIT VICTIM] |
| 21:30.59 | *** join/#asterisk malverian[work] (n=pawalls@pawalls.teamgleim.com) [NETSPLIT VICTIM] |
| 21:30.59 | *** mode/#asterisk [+o twisted] by irc.freenode.net |
| 21:30.59 | *** join/#asterisk bsdfreak (n=alex@daemon.bsdgeek.com) [NETSPLIT VICTIM] |
| 21:30.59 | *** join/#asterisk eldu (i=damajor@tuxmania.org) [NETSPLIT VICTIM] |
| 21:31.00 | *** join/#asterisk KryoStoffer (n=kri@helium.kri.dk) [NETSPLIT VICTIM] |
| 21:31.00 | *** join/#asterisk jlewis (i=jlewis@solo.atlantic.net) [NETSPLIT VICTIM] |
| 21:31.00 | *** join/#asterisk Ahrimanes (n=michael@aronsen.dk) [NETSPLIT VICTIM] |
| 21:31.00 | *** join/#asterisk kaldemar (n=kalde@130.233.228.9) [NETSPLIT VICTIM] |
| 21:31.00 | *** join/#asterisk ApEtc (i=apetc@ip68-3-225-51.ph.ph.cox.net) [NETSPLIT VICTIM] |
| 21:31.00 | *** join/#asterisk Dibbler (n=Dibbler@zidane.pi-net.net) [NETSPLIT VICTIM] |
| 21:31.00 | *** join/#asterisk jojo (n=jojo@c83-253-39-20.bredband.comhem.se) [NETSPLIT VICTIM] |
| 21:31.01 | *** join/#asterisk Tjief (i=Tjief@r0x0r.dk) [NETSPLIT VICTIM] |
| 21:31.01 | *** join/#asterisk Snake-Eyes (n=blog@203.220.55.70) [NETSPLIT VICTIM] |
| 21:31.01 | *** join/#asterisk smurf (n=smurf@debian/developer/smurf) [NETSPLIT VICTIM] |
| 21:31.01 | *** join/#asterisk _m_ (n=m@fbta199.fbta.uni-karlsruhe.de) [NETSPLIT VICTIM] |
| 21:31.01 | *** join/#asterisk rbd (n=robbyd@24-181-194-064.dhcp.hckr.nc.charter.com) [NETSPLIT VICTIM] |
| 21:31.01 | *** join/#asterisk _Sam-- (n=sam@phone2.kneedraggers.com) [NETSPLIT VICTIM] |
| 21:31.01 | *** join/#asterisk Chuji (i=Chuji@pcp09930052pcs.tulipgrove.tn.nash.comcast.net) [NETSPLIT VICTIM] |
| 21:31.01 | *** join/#asterisk CoolAcid (n=jason@216.99.98.39) [NETSPLIT VICTIM] |
| 21:31.02 | *** join/#asterisk A-Tuin|work (n=A-Tuin@nat.office.legend.net.uk) [NETSPLIT VICTIM] |
| 21:31.02 | *** join/#asterisk pfn (n=pfnguyen@netblock-66-245-252-239.dslextreme.com) [NETSPLIT VICTIM] |
| 21:31.02 | *** join/#asterisk iDunno (i=brettp@stef.sommitrealweird.co.uk) [NETSPLIT VICTIM] |
| 21:31.02 | *** join/#asterisk wildcard0 (n=generic@S0106006097e16040.vc.shawcable.net) [NETSPLIT VICTIM] |
| 21:31.03 | *** join/#asterisk jayk- (i=jayk@lasziv.reprehensible.net) [NETSPLIT VICTIM] |
| 21:31.03 | *** join/#asterisk david-home (n=david@gw-djc.davidcoulson.net) [NETSPLIT VICTIM] |
| 21:31.03 | *** join/#asterisk mishehu (i=mishehu@cshells.shavedgoats.net) [NETSPLIT VICTIM] |
| 21:31.03 | *** join/#asterisk ender (n=me@fedora/ender) [NETSPLIT VICTIM] |
| 21:31.03 | *** join/#asterisk infinity1 (n=brendon@solara.netcal.com) [NETSPLIT VICTIM] |
| 21:31.03 | *** join/#asterisk Pazzo (n=Pazzo@host130-250.pool8172.interbusiness.it) [NETSPLIT VICTIM] |
| 21:31.03 | *** join/#asterisk _Lyfe_ (n=lyfe@broke.the.gsnservers.com) [NETSPLIT VICTIM] |
| 21:31.04 | *** join/#asterisk hohum (i=corbe@snoop.burghcom.com) [NETSPLIT VICTIM] |
| 21:31.04 | *** join/#asterisk kram (n=mark@pdpc/sponsor/digium/kram) [NETSPLIT VICTIM] |
| 21:31.04 | *** join/#asterisk marcus2 (i=marcus@pompeii.outer.org) [NETSPLIT VICTIM] |
| 21:31.04 | *** join/#asterisk CoaxD (i=coax@shell1.cornernet.com) [NETSPLIT VICTIM] |
| 21:31.04 | *** mode/#asterisk [+o kram] by irc.freenode.net |
| 21:31.18 | [TK]D-Fender | b00m! |
| 21:31.18 | *** join/#asterisk IronHelixz (n=irc@ool-45785cfe.dyn.optonline.net) |
| 21:31.18 | KranZ | they made a channelized ds3 card yet? |
| 21:32.05 | *** join/#asterisk h3x0r4t0r (n=h3xor@64.192.116.16) |
| 21:32.05 | KranZ | ./slap dmwaters |
| 21:32.05 | p1tst0p | eheh |
| 21:32.06 | *** join/#asterisk file (n=joshnet@mctnnbsa24w-142167049176.nb.aliant.net) |
| 21:32.06 | *** join/#asterisk jimmy_deanPB (n=jhodapp@adsl-70-228-242-126.dsl.ipltin.ameritech.net) |
| 21:32.07 | batphone | [TK]D-Fender, i dont see much more than 4 port PRI |
| 21:32.07 | [TK]D-Fender | http://www.sangoma.com/products/p_aft-et3-specs.htm |
| 21:32.07 | batphone | ahh |
| 21:32.07 | [TK]D-Fender | But not channelized yet... from what it appears |
| 21:32.07 | KranZ | has anyone put more than 2 quad t1 cards in a box? |
| 21:32.07 | KranZ | on the same bus |
| 21:32.08 | batphone | freebsd!! |
| 21:32.08 | batphone | KranZ, ive put 4 |
| 21:32.08 | KranZ | 4 quads? |
| 21:32.08 | batphone | with some success.. i have a flaky mobo though |
| 21:32.08 | jimmy_deanPB | I have an incoming Zap channel for an analog line that is behaving weirdly. I call from my cell phone to the Zap line, I see it coming in on the Asterisk console verbose view, it seems to answer and follows my dialplan perfectly, but I don't hear any of what I should on my cell phone - it actually keeps ringing on my cell phone side almost like Asterisk forgot to answer...yet "Answer" is clearly being executed |
| 21:32.08 | batphone | yep |
| 21:32.08 | jimmy_deanPB | Any thoughts? |
| 21:32.08 | *** join/#asterisk file[laptop] (n=jcolp@mctnnbsa24w-142167049176.nb.aliant.net) |
| 21:32.09 | KranZ | no irq/bus issues with slips and timing? |
| 21:32.09 | batphone | KranZ, it took about 2 days to iron out the irq thing |
| 21:32.09 | trixter | is the owner of anatifero.us here? I forgot who you were :/ |
| 21:32.09 | batphone | KranZ, i disabled APIC in the bios and set one of the card slots to be reserved |
| 21:32.09 | batphone | KranZ, otherwise it would try to share an irq with another card |
| 21:32.10 | KranZ | i've had issues where the onboard sata would steal bw from the quad card |
| 21:32.10 | [TK]D-Fender | IRQ problems? Voltage problems? not in my world :D |
| 21:32.10 | KranZ | caused HDLC (6) aborts and slips |
| 21:32.10 | batphone | KranZ, where are you seeing that? |
| 21:32.10 | [TK]D-Fender | ok, time to get the heck outta here.. bbiab |
| 21:32.10 | KranZ | then i got a pci sata controller with the same chip as the onboard one and moved the raid to that card |
| 21:32.10 | KranZ | fixed the problems |
| 21:32.10 | batphone | [TK]D-Fender, digium tech told me that the 4 port cards consume 2 watts |
| 21:32.10 | batphone | KranZ, what mobo??? |
| 21:32.53 | KranZ | tyan |
| 21:32.53 | batphone | model? |
| 21:32.53 | jimmy_deanPB | Anyone have thoughts on the Zap channel weirdness? |
| 21:32.53 | KranZ | k8sdpro |
| 21:32.54 | KranZ | the 1 32bit port shares the same bus as the onboard sata controller |
| 21:33.42 | KranZ | my next t1 card purchase will be a 64bit card |
| 21:33.42 | batphone | KranZ, ahh but no gigabit NIC |
| 21:33.42 | *** join/#asterisk lilo (i=levin@freenode/staff/pdpc.levin) |
| 21:33.43 | KranZ | actually, 2 gbit and 1 10/100 |
| 21:34.28 | batphone | KranZ, what kind of bios? might try reserving that slot if possible |
| 21:34.55 | KranZ | batphone: problem's resolved |
| 21:34.55 | KranZ | its a bus issue |
| 21:34.55 | KranZ | not an irq |
| 21:34.55 | KranZ | (was) |
| 21:34.55 | batphone | oh |
| 21:34.55 | batphone | what did you do to change the bus? |
| 21:35.40 | KranZ | bought a pci sata card w/ the same chipset as the onboard |
| 21:35.40 | KranZ | disabled the onboard sata |
| 21:35.40 | batphone | so you are _not_ sharing interrupts now? |
| 21:35.40 | KranZ | i never was |
| 21:35.40 | batphone | oh |
| 21:35.40 | batphone | ive been awake for 2 days... |
| 21:35.41 | *** join/#asterisk Seldon19751 (n=someone@gatekeeper.radintl.com) |
| 21:36.07 | KranZ | the sata was robbing bandwidth on the bus |
| 21:36.33 | batphone | i see |
| 21:36.33 | batphone | i was getting chirps and garbling on calls |
| 21:37.02 | KranZ | this is a normal occurance, but the t1 cards are realtime cards and cant wait like a nic card |
| 21:37.02 | jimmy_deanPB | So it the Zap/3 channel is being answered by *, yet you wouldn't know this from the cell phone side...it keeps on ringing. On *'s side, it looks like it thinks it's answered and everything |
| 21:37.02 | batphone | i couldnt tell what was going on though |
| 21:37.02 | KranZ | same |
| 21:37.02 | KranZ | my spans would drop if i had heavy disk activity |
| 21:37.03 | batphone | AH |
| 21:37.03 | batphone | man |
| 21:37.03 | batphone | ok |
| 21:37.03 | KranZ | especially when i ran hdparm -Tt /dev/sda |
| 21:37.26 | KranZ | game over at that point |
| 21:37.26 | batphone | i was doing ls -lR / |
| 21:37.26 | batphone | that would pretty much kill zaptel |
| 21:37.26 | KranZ | yup |
| 21:37.37 | KranZ | it did |
| 21:37.39 | batphone | but in a big box like that its the equivalent of having lots of calls |
| 21:37.53 | batphone | my deal is this: when i would ls -lsR /proc i would get the same thing |
| 21:37.53 | batphone | this is not disk activity |
| 21:38.00 | batphone | i wouldnt think |
| 21:38.08 | KranZ | you still have the prob? |
| 21:38.12 | batphone | yeah |
| 21:38.25 | KranZ | try hdparm -tT /dev/yourharddisk |
| 21:38.25 | *** part/#asterisk YoMama (n=r00t@pcp02689850pcs.roylok01.mi.comcast.net) |
| 21:38.29 | batphone | im ordering one of those boards right now |
| 21:38.37 | KranZ | does a read test |
| 21:39.30 | *** join/#asterisk keith80403_ (n=keith804@24-56-188-49.co.warpdriveonline.com) |
| 21:39.35 | *** join/#asterisk kavit (n=kavit@ppp167-252-96.static.internode.on.net) |
| 21:39.40 | batphone | KranZ, do you use te405p? |
| 21:39.49 | bleck | gah! I just can't figure this out. |
| 21:39.50 | batphone | these are 32 bit cards you have right? |
| 21:40.20 | bleck | how do you show debug info in IOS? |
| 21:40.45 | batphone | KranZ, i cant really experiment on it right now |
| 21:40.47 | bleck | i've done "debug ccsip messages" but nothing appears. |
| 21:40.54 | batphone | kranz its in production |
| 21:41.00 | batphone | minus a card |
| 21:41.05 | bleck | i've done debug ccsip <most options> |
| 21:41.30 | *** join/#asterisk ben_d (n=ben@cpe-66-66-209-96.rochester.res.rr.com) |
| 21:41.31 | bleck | whats goin on with this channel? |
| 21:41.37 | *** join/#asterisk tessier (n=treed@rrcs-67-53-110-66.west.biz.rr.com) |
| 21:42.04 | batphone | bleck, fw upgrades |
| 21:42.06 | *** join/#asterisk marv[work] (n=timr@64.89.118.139) |
| 21:42.16 | *** join/#asterisk Pazzo (n=Pazzo@host130-250.pool8172.interbusiness.it) |
| 21:42.17 | *** join/#asterisk ipso (n=ipso@d207-81-249-35.bchsia.telus.net) |
| 21:42.17 | *** join/#asterisk synthetiq (n=roger@64.201.13.50) |
| 21:42.26 | bleck | aah. |
| 21:42.34 | kavit | my X100P clone card is acting up i get a FXO PCI MASTER ABORT errors, google and mailing list tell me nothing, may I please get some assistance or a link if someone has encountered this problem before |
| 21:42.37 | *** join/#asterisk krisguy (n=krisguy@h216-170-039-057.adsl.navix.net) |
| 21:42.38 | *** join/#asterisk xheliox (n=jeff@user-0c6s3h2.cable.mindspring.com) |
| 21:42.52 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 21:43.41 | bsdfreak | heh |
| 21:43.44 | bsdfreak | holy crapola |
| 21:43.53 | batphone | yeah ahah |
| 21:43.55 | batphone | err0r |
| 21:43.57 | *** join/#asterisk benjk (n=benjamin@f8a01-0357.din.or.jp) |
| 21:44.07 | *** join/#asterisk kant (n=bernd@metrored-gw.tropicohn.com) |
| 21:44.12 | *** join/#asterisk maik (n=maik@bfs.cs.uni-sb.de) |
| 21:45.37 | *** join/#asterisk jaiger (n=jaiger@fire.innovationsw.com) |
| 21:45.57 | *** join/#asterisk vexorg (n=vexorg@CPE000021ded913-CM001225419164.cpe.net.cable.rogers.com) |
| 21:46.47 | *** join/#asterisk _santiago_ (n=santiago@208.195.215.160) |
| 21:48.02 | *** join/#asterisk mjmac (n=mjmac@pdpc/supporter/active/mjmac) |
| 21:52.48 | *** join/#asterisk newl (n=newlook@203-59-61-62.dyn.iinet.net.au) [NETSPLIT VICTIM] |
| 21:52.48 | *** join/#asterisk BhaalWK (i=bhaal@freenode/staff/bhaal) [NETSPLIT VICTIM] |
| 21:52.48 | *** join/#asterisk juice (n=juice@mo-67-77-176-14.dyn.sprint-hsd.net) [NETSPLIT VICTIM] |
| 21:53.39 | *** join/#asterisk _Madar (n=tophe@219-84-129-163-adsl-tpe.static.so-net.net.tw) |
| 21:54.46 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 22:01.53 | *** join/#asterisk scubasteve (n=steve@cpe-071-065-215-219.nc.res.rr.com) |
| 22:02.08 | scubasteve | Can someone tell me what the purpose of ASTERISK_GPL_KEY is? |
| 22:02.41 | *** join/#asterisk phifli (n=mike@ip70-180-108-164.no.no.cox.net) |
| 22:02.55 | phifli | ok i rebooted my boxs out of tyhe blue and now when i do AEL reload it says |
| 22:02.55 | phifli | it cannot merge contexts |
| 22:03.00 | phifli | or contexts cannot be merged, wtf? |
| 22:03.09 | phifli | its like after my macros at the top it stops loading AEL completely |
| 22:04.28 | phifli | and my main incoming-did context won't get loaded |
| 22:04.40 | phifli | whats the deal with that???? |
| 22:04.53 | *** join/#asterisk Dr-Linux (n=Nothing@202.125.141.6) |
| 22:05.06 | phifli | ssays "Requested contexts didn't get merged" |
| 22:05.50 | SkramX | Werid! |
| 22:05.51 | SkramX | Just wanted to let you know you were just left a 0:03 long message (number 4) |
| 22:05.59 | SkramX | in mailbox 0 from local |
| 22:05.59 | SkramX | from "local"? |
| 22:06.22 | phifli | hey how are you |
| 22:06.26 | phifli | sorry having some serious issues |
| 22:06.29 | SkramX | me? |
| 22:06.30 | phifli | after they are fixewd sure |
| 22:06.40 | SkramX | phifli: whats your porblems? |
| 22:06.43 | SkramX | *problems |
| 22:06.54 | phifli | my AEL wont interpret |
| 22:07.00 | phifli | says requested contexts cannot be merged |
| 22:07.05 | phifli | after it loads the macros from the top |
| 22:07.11 | SkramX | hmm I havent messed with AEL. |
| 22:07.11 | phifli | it never had a problem but i decided to reboot |
| 22:07.16 | phifli | damn |
| 22:07.28 | phifli | i need thi sfixed ASAP! |
| 22:07.39 | *** join/#asterisk Flauto (n=zhao@c-24-14-197-214.hsd1.il.comcast.net) |
| 22:07.47 | SkramX | phifli: ill research |
| 22:08.06 | phifli | thanks so much so will i |
| 22:08.13 | _Sam-- | does the 't' timeout rule in extensions apply if all channels are busy? for example if i have an extension dial SIP/1 & SIP/2, if they are both busy what rule would i use to specify what to do? |
| 22:08.26 | *** join/#asterisk [hC] (n=hardcore@S0106000e9b96114f.vf.shawcable.net) |
| 22:09.30 | ManxPower | phifli, check bugs.digium.com to see if it's a known issue, also search the mailinglist archive |
| 22:09.43 | ManxPower | _Sam--, no. |
| 22:09.44 | SkramX | _Sam--: I think you would just put a exten => witht he next s,#, |
| 22:09.59 | SkramX | t is if it timesout, like it does not receive input or whatnot |
| 22:10.10 | Dr-Linux | exten => 4444,1,Wait(2) |
| 22:10.10 | Dr-Linux | exten => 4444,2,directory(default) |
| 22:10.19 | ManxPower | exten => t is called when Asterisk is waiting for an extension in an IVR (WaitExten, Background, Playback, etc). |
| 22:10.22 | SkramX | Dr-Linux: Yes? |
| 22:10.35 | Dr-Linux | when i dial 4444 its say "welcome message" but i can't heard first work |
| 22:10.49 | ManxPower | Dr-Linux, answer, then wait, then the rest |
| 22:11.20 | Dr-Linux | oo, Answer(what-here?) ? |
| 22:11.22 | Dr-Linux | or nothing? |
| 22:11.36 | ManxPower | exten => 4444,1,Answer |
| 22:11.44 | Dr-Linux | acha |
| 22:12.32 | _Sam-- | so if i use something like this... exten => 1234 1,Dial(sip/1&SIP/2) ....how do i implement the 's' rule ? |
| 22:12.36 | _Sam-- | i understand the t rules |
| 22:12.37 | _Sam-- | and use them |
| 22:13.12 | Flauto | is anyone using jajah? |
| 22:13.43 | _Sam-- | just put in s rule in for the same context as the 1234 extension? |
| 22:13.57 | *** join/#asterisk stkn_ (n=stkn@gentoo/developer/pdpc.active.stkn) |
| 22:14.05 | phifli | ok |
| 22:14.21 | phifli | ael won't finish loading after my madcros. says requested contexts cannot be merge |
| 22:14.24 | phifli | this is very odd |
| 22:14.34 | Seldon19751 | can someone tell me how to set up extensions.conf so incoming callers hear a welcome message and then can dial an extension? |
| 22:14.44 | Dr-Linux | ManxPower: thanks its working fine now, |
| 22:14.45 | Dr-Linux | exten => 4444,1,Answer |
| 22:14.45 | Dr-Linux | exten => 4444,2,Wait(2) |
| 22:14.45 | Dr-Linux | exten => 4444,3,directory(default) |
| 22:14.48 | Dr-Linux | its all |
| 22:14.59 | ManxPower | Wait(.5) should be enough. |
| 22:15.41 | Dr-Linux | ManxPower: should i add anything in last, like 4444,4,Hangup(congestion) |
| 22:15.43 | Dr-Linux | ? |
| 22:15.44 | _Sam-- | Seldon19751: Backgroun(messageyouwntopla) |
| 22:15.47 | _Sam-- | Background |
| 22:16.23 | ManxPower | exten => 4444,4,1,Congestion(15) |
| 22:16.27 | ManxPower | ..er.. |
| 22:16.30 | Dr-Linux | :S |
| 22:16.32 | ManxPower | exten => 4444,4,Congestion(15) |
| 22:16.52 | ManxPower | I'm assuming you are running 1.2, Dr-Linux |
| 22:16.58 | Seldon19751 | Sam: ok, then what |
| 22:17.06 | Dr-Linux | ManxPower: yes, how you know? |
| 22:17.15 | Seldon19751 | fo them to dial an exxtension |
| 22:17.17 | _Sam-- | then some pattern matching works good |
| 22:17.26 | ManxPower | Dr-Linux, I don't, but I don't think Congestion supports a timeout on 1.0 |
| 22:17.31 | _Sam-- | exten => _XXX Dial($exten) or something like that |
| 22:17.32 | Seldon19751 | all my extensions are 200-299 |
| 22:17.44 | SkramX | okay |
| 22:17.46 | Dr-Linux | ManxPower: i dont' know much about this call by name feature, |
| 22:18.02 | Dr-Linux | ManxPower: ooo i didn't know that |
| 22:18.06 | _Sam-- | sledon: an exmple of how i do it (sorry for paste) |
| 22:18.06 | _Sam-- | exten => s,1,Wait,1 |
| 22:18.07 | _Sam-- | exten => s,2,Answer |
| 22:18.07 | _Sam-- | exten => s,3,DigitTimeout,5 |
| 22:18.07 | _Sam-- | exten => s,4,ResponseTimeout,9 |
| 22:18.07 | _Sam-- | exten => s,5,Wait,1 |
| 22:18.08 | _Sam-- | exten => s,6,BackGround(closed) ; First IVR menuing waits for input |
| 22:18.10 | _Sam-- | #exten => s,6,BackGround(salesmeeting) ; First IVR menuing waits for input |
| 22:18.17 | Seldon19751 | thanks! |
| 22:18.18 | _Sam-- | exten => _XXX,1,Goto(default,${EXTEN},1) |
| 22:18.48 | Dr-Linux | ManxPower: so what you suggest , i should use this congestion option in last pirority or not? |
| 22:19.11 | _Sam-- | you will probably want these in there too: |
| 22:19.11 | _Sam-- | exten => t,1,Hangup |
| 22:19.11 | _Sam-- | exten => _XXX,1,Goto(default,${EXTEN},1) |
| 22:19.11 | _Sam-- | exten => i,1,Playback(invalid) |
| 22:19.11 | _Sam-- | exten => i,2,Goto(closed,s,3) |
| 22:19.27 | _Sam-- | something that handles invalid extensions and timeouts |
| 22:19.56 | phifli | anyone have any ideas/issues with AEL? |
| 22:20.07 | phifli | i rebooted and now when i run asterisk it wont load my context.. says it cannot be merged wtf |
| 22:21.16 | Seldon19751 | sam: thanks |
| 22:21.21 | phifli | i wonder if its a memmory issue |
| 22:21.28 | _Sam-- | sure thing, good luck |
| 22:25.36 | _Sam-- | ManxPower : what does the 's' rules represent...what does the s mean |
| 22:25.44 | _Sam-- | i know, t, i, but not what s means |
| 22:25.55 | Dr-Linux | s = begin |
| 22:26.08 | Dr-Linux | start |
| 22:26.08 | *** join/#asterisk gnosys (n=ksford@ip68-9-201-250.ri.ri.cox.net) |
| 22:27.18 | _Sam-- | so you could use this syntax: exten => 1234,s,1 Dial(blah/sip)...or it has to be exten => s,1,Dial(blah/sip) |
| 22:27.39 | _Sam-- | because im not following how i would use the s rule in a context that has multiple DIDs |
| 22:27.39 | *** join/#asterisk shido6 (n=shido6@d221-68-216.commercial.cgocable.net) |
| 22:27.45 | Dr-Linux | phifli: did you check you all logs files? i.e system log? "dmesg" * logs, inttrups etc, ? |
| 22:28.24 | phifli | yes |
| 22:28.25 | Dr-Linux | _Sam--: your 2nd choice looks fine |
| 22:28.43 | phifli | it just stops after loading modules |
| 22:28.46 | phifli | wheh i take modules out.. |
| 22:28.57 | phifli | it says requested contexts cannot be merged |
| 22:29.02 | phifli | thats in pbx.c |
| 22:29.25 | phifli | lasttmp |
| 22:29.27 | phifli | wonder what that is |
| 22:30.06 | Dr-Linux | :S |
| 22:30.21 | *** join/#asterisk Kokey (n=Kokey@201.153.63.79) |
| 22:30.28 | Dr-Linux | save your data backup |
| 22:30.40 | ManxPower | exten => s is NOT a "catchall" it's more of a "catch nothing" i.e. it only catches calls that have no destination info. A "catchall" would be exten => _. but that would catch extensions that are not numbers (like o, i, t, T, h, etc). A catch all number extensions would be something like exten => _X. |
| 22:30.43 | ManxPower | From a posting to the mailinglist |
| 22:30.46 | Dr-Linux | and try to recompile |
| 22:31.03 | ManxPower | When a call comes into Asterisk (PSTN, VoIP, etc) and call has NO information as to what extension to route to then Asterisk will try sending the call to extension => s |
| 22:31.03 | ManxPower | In practice this only happens if you have a voice T-1 (Not PRI) with no DIDs, or if you have an analog FXO port. |
| 22:31.08 | ManxPower | Also from a mailinglist post |
| 22:32.59 | Dr-Linux | ManxPower: should i use "hangup" after the congestion? |
| 22:33.10 | Dr-Linux | bcoz its still connected after the congestion, |
| 22:33.11 | phifli | wonder what that is |
| 22:33.13 | phifli | err |
| 22:33.34 | Dr-Linux | exten => 4444,1,Answer |
| 22:33.34 | Dr-Linux | exten => 4444,2,Wait(.5) |
| 22:33.34 | Dr-Linux | exten => 4444,3,directory(default) |
| 22:33.34 | Dr-Linux | exten => 4444,4,Congestion(15) |
| 22:33.35 | ManxPower | Dr-Linux, no. See "show application congestion" Pay special attention as to what the timeout does. |
| 22:33.37 | *** join/#asterisk fugitivo (n=ajf@209.13.244.233) |
| 22:33.42 | phifli | so in other words |
| 22:33.44 | phifli | im doing something wrong? |
| 22:33.48 | phifli | or him? |
| 22:33.58 | ManxPower | phifli, I'll bet your problem is a typoe |
| 22:34.00 | phifli | my shit worked fine till i rebooted |
| 22:34.08 | phifli | then i had to startup mysql |
| 22:34.10 | phifli | to get it going |
| 22:34.17 | ManxPower | phifli, did you do a reload the last time you made a change? |
| 22:34.18 | phifli | now i realize ael reload wont go past macros |
| 22:34.21 | phifli | ahuh |
| 22:34.25 | _Sam-- | manx thanks. |
| 22:34.44 | phifli | im loading from a backup tho |
| 22:34.46 | Dr-Linux | reboot is never a solution using linux, |
| 22:34.57 | phifli | we;l; |
| 22:34.57 | phifli | well |
| 22:35.07 | phifli | i didnt add things to boot yet on purpose |
| 22:35.09 | phifli | chkconfig |
| 22:35.11 | phifli | in fedora |
| 22:35.30 | Dr-Linux | you are using FC ? |
| 22:35.31 | Dr-Linux | ooo |
| 22:35.41 | phifli | i sue * |
| 22:35.42 | phifli | use |
| 22:35.45 | ManxPower | phifli, What is the EXACT error message? |
| 22:37.04 | phifli | that requested contedxts cannot be merged |
| 22:37.16 | phifli | sec now im having issues connewcting to my box grr |
| 22:37.58 | *** join/#asterisk razu (n=razu@213-35-173-39-dsl.prn.estpak.ee) |
| 22:38.18 | ManxPower | phifli, so if I do a source code search for "that requested contedxts cannot be merged" I would get a hit? |
| 22:38.27 | ManxPower | Perhaps you can just PASTE the error message? |
| 22:39.07 | *** join/#asterisk The-Dark (n=The-Dark@p508E2C50.dip0.t-ipconnect.de) |
| 22:39.18 | *** part/#asterisk The-Dark (n=The-Dark@p508E2C50.dip0.t-ipconnect.de) |
| 22:39.25 | *** join/#asterisk The-Dark (n=The-Dark@p508E2C50.dip0.t-ipconnect.de) |
| 22:39.25 | phifli | yeah if i can get to the config again |
| 22:39.28 | phifli | its in pbx./c |
| 22:39.29 | phifli | pbx.c |
| 22:39.32 | phifli | and yes its the OHNLY hit |
| 22:39.33 | phifli | ONLY |
| 22:39.38 | phifli | search for cannot be merged |
| 22:39.47 | phifli | lasttmp i think might be the prob |
| 22:40.19 | The-Dark | hello |
| 22:40.30 | ManxPower | [root@fs-1 asterisk-1.2.0-rc2]# grep -r "cannot be merged" * |
| 22:40.30 | ManxPower | [root@fs-1 asterisk-1.2.0-rc2]# |
| 22:40.34 | SkramX | Hiya |
| 22:40.50 | ManxPower | Without the exact text of the error message I cannot help you. |
| 22:40.57 | The-Dark | maybe someone could help me? |
| 22:40.57 | p1tst0p | what would i use, to set my phone to busy mode, so no one can call me.. like say, a send to voicemail or something. |
| 22:41.36 | Seldon19751 | with the Record() function, how do you end the recording from the handset? |
| 22:41.51 | The-Dark | i have some probs with zaphfc and a gigaset 1054isdn |
| 22:42.05 | ManxPower | Seldon19751, You are an idiot if you do not check for that information in "show application record" |
| 22:42.15 | *** join/#asterisk RaYmAn-Bx (i=rayman@x1-6-00-40-63-da-39-3f.k191.webspeed.dk) |
| 22:42.36 | *** join/#asterisk morale (i=russell@S010600111155e117.cg.shawcable.net) |
| 22:43.00 | Seldon19751 | thx for the abuse |
| 22:43.01 | phifli | look for merg in pbx.c |
| 22:44.37 | Seldon19751 | "show application record" tells me it's #, but this seemed not to work |
| 22:44.43 | Seldon19751 | ill try again |
| 22:45.14 | The-Dark | is there someone havin time to help me? |
| 22:45.38 | Seldon19751 | The-Dark if you're a novice don't risk asking Manx |
| 22:45.50 | *** join/#asterisk brimstone (n=brimston@pdpc/sponsor/digium/brimstone) |
| 22:45.57 | phifli | is there ahny way to compile my shit into a normal extensions file fom AEL |
| 22:45.59 | phifli | and dump it |
| 22:46.04 | phifli | i dont wan tthis shit happening again later |
| 22:47.06 | The-Dark | no i'm not a novice... |
| 22:47.21 | The-Dark | my asterisk workin fine |
| 22:47.32 | The-Dark | but not with a gigaset |
| 22:49.53 | The-Dark | so my prob is that the mobile handsets only rings once and asterisk says zap is busy |
| 22:50.36 | The-Dark | the reverse way works fine, calling a analog handset from mobile set |
| 22:50.59 | *** join/#asterisk [hC] (n=hardcore@S0106000e9b96114f.vf.shawcable.net) |
| 22:51.52 | *** join/#asterisk marc32344 (n=marc3234@Toronto-HSE-ppp3762675.sympatico.ca) |
| 22:52.22 | The-Dark | no ideas? |
| 22:54.40 | Dr-Linux | anyone tell me please, what does this error mean? |
| 22:54.41 | Dr-Linux | Dec 9 14:48:14 WARNING[12671]: chan_sip.c:1064 __sip_xmit: sip_xmit of 0xb74ce9d8 (len 435) to 192.168.0.33:-1 returned 5060: Operation not permitted |
| 22:55.57 | *** join/#asterisk [Airwolf] (n=airwolf@attilla.nl) |
| 22:57.41 | *** part/#asterisk splatone (n=splat1@rrcs-24-172-35-197.midsouth.biz.rr.com) |
| 22:59.51 | The-Dark | can noone help me? |
| 22:59.52 | SkramX | ctooley around? |
| 22:59.53 | SkramX | fuck |
| 23:01.10 | Darwin35 | no everyone in here is beyond help |
| 23:02.11 | SkramX | The-Dark: what do you need? |
| 23:03.09 | Darwin35 | a life awife and three head of sheep 2 cows and a hog for breeding |
| 23:03.10 | The-Dark | i've postet above, i have probs to get a gigaset workin with zaphfc in NT |
| 23:03.31 | Darwin35 | nt |
| 23:04.00 | *** join/#asterisk UyCaRumBa (n=administ@200.121.130.49) |
| 23:04.12 | The-Dark | my ta-33 works fine at internal s0 (the hfc) |
| 23:04.24 | *** join/#asterisk Druken (n=druken@CPE00121716da99-CM000e5cde4ca2.cpe.net.cable.rogers.com) |
| 23:04.42 | The-Dark | but my gigaset rings once and asterisk says zap busy |
| 23:05.15 | [hC] | hey if i make update from a cvs head tree, will it still be up to date with svn? or do i have to actually go get a new svn checkout altogether? |
| 23:05.20 | *** join/#asterisk Cresl1n (n=matt@gateway.digium.com) |
| 23:05.34 | The-Dark | a call form gigaset to ta-33 analog handset works |
| 23:06.34 | *** join/#asterisk ldnblk (n=Just@212.183.128.185) |
| 23:06.38 | The-Dark | but not the reverse way (only one ring then busy) |
| 23:06.47 | *** join/#asterisk hardwire (n=nhardwir@66-230-102-166-cdsl-rb1.nwc.acsalaska.net) |
| 23:06.57 | *** join/#asterisk heison (n=heison@CPE000a01d49e6f-CM00111a59bec2.cpe.net.cable.rogers.com) |
| 23:07.02 | hardwire | you monkey lovin squid munchers! |
| 23:07.35 | Druken | jelious? |
| 23:08.11 | Druken | hmm... that don't look right... |
| 23:08.27 | hardwire | gelulose :) |
| 23:08.53 | hardwire | you cattle rattling corn farmer. |
| 23:09.00 | tzanger | haha |
| 23:09.14 | tzanger | monkey-lovin' squid muncher |
| 23:09.25 | *** join/#asterisk negatendo (n=negatend@c-24-9-136-152.hsd1.co.comcast.net) |
| 23:09.31 | *** join/#asterisk Inv_arp (n=junya@c-66-176-211-109.hsd1.fl.comcast.net) |
| 23:09.41 | hardwire | I hate my life.. SHOOT ME! |
| 23:10.05 | hardwire | he wasted around $6k this month |
| 23:10.11 | docelm0 | Whats cookin? |
| 23:10.16 | tzanger | heh |
| 23:10.20 | tzanger | squid apparently |
| 23:10.24 | hardwire | mmm |
| 23:10.34 | hardwire | so this italian place around here (Anchorage) has amazing calamari |
| 23:10.40 | docelm0 | huh? 6K? On what? |
| 23:10.42 | hardwire | he never reuses the oil he pan fries it in |
| 23:11.09 | tzanger | nice |
| 23:11.35 | hardwire | docelm0: well.. he wanted to interface 4 analog phones to a PBX that we are going to replace soon because none of its functinos work.. only the hunt groups. |
| 23:11.43 | hardwire | so he bought 4 $480 ATA's for it |
| 23:11.43 | morale | w |
| 23:11.51 | hardwire | vs just putting those phones on the 4 lines coming into the pbx. |
| 23:11.53 | tzanger | wow |
| 23:11.59 | tzanger | what ata is $480? |
| 23:12.03 | hardwire | dta |
| 23:12.21 | hardwire | actually.. some engenius tech nortel -> analog converter |
| 23:12.23 | tzanger | digitla telepnone adaptor? |
| 23:12.24 | tzanger | ahh |
| 23:12.32 | tzanger | I have a telebridge hooked up to my norstar |
| 23:12.46 | hardwire | and then he squandered all this money on this wireless phone system.. to put on top of those 4 lines. |
| 23:12.49 | tzanger | DP00 |
| 23:12.49 | tzanger | DMS"Dec 9 6:11 pm " |
| 23:12.49 | hardwire | and.. we already have one. |
| 23:12.54 | hardwire | so.. |
| 23:12.57 | tzanger | nice |
| 23:13.00 | *** join/#asterisk m160858 (n=jsaenz@200.89.12.46) |
| 23:13.00 | hardwire | he is just throwing money away at this point |
| 23:13.05 | SkramX | dumnass. |
| 23:13.06 | SkramX | he |
| 23:13.06 | SkramX | h |
| 23:13.07 | hardwire | first off.. by going from pots to pbx to pots |
| 23:13.09 | m160858 | hi |
| 23:13.12 | hardwire | when pots to phone will work just fine |
| 23:13.27 | hardwire | then making double purchases |
| 23:13.34 | m160858 | i've a member on the queues |
| 23:14.03 | m160858 | how can i do to recording wav file with the name of the member? |
| 23:14.07 | m160858 | what VAR |
| 23:15.04 | hardwire | tzanger: it just make me crazy |
| 23:15.08 | hardwire | !!! CRAZY !!! |
| 23:15.34 | hardwire | esp when I just spent so much money making an asterisk PBX that goes in place of this pbx he is trying to interface $1800 in hardware too. |
| 23:15.39 | hardwire | thats more than the damn * box. |
| 23:15.46 | hardwire | CRAZY! |
| 23:15.48 | hardwire | YOU HEAR ME! |
| 23:16.01 | hardwire | dude |
| 23:16.01 | file | get ahold of yourself |
| 23:16.02 | hardwire | CRAZY! |
| 23:16.03 | hardwire | I can't |
| 23:16.31 | hardwire | any formulation of sense I make, and then speak only seems to effect people that have a slight understanding of what I want. |
| 23:16.40 | hardwire | and this guy.. its all mumble jumped before his brain gets ahold of it. |
| 23:16.44 | The-Dark | still waitn |
| 23:16.52 | hardwire | in The-Dark ? |
| 23:16.54 | hardwire | hehe |
| 23:17.42 | m160858 | hello? i've 8 extensions ... then using the same number for call out |
| 23:18.23 | m160858 | i want to record the calls out, but i can't record a wav file for member of my queue |
| 23:18.34 | The-Dark | yeah |
| 23:19.36 | m160858 | what var i should use |
| 23:19.40 | The-Dark | no ideas? |
| 23:21.23 | *** join/#asterisk Delvar (n=irc@host-83-146-53-34.bulldogdsl.com) |
| 23:22.45 | *** join/#asterisk MGSsancho (n=user@adsl-67-126-140-54.dsl.irvnca.pacbell.net) |
| 23:25.04 | *** join/#asterisk mikayle (i=user@nolmstd-cadent1-68-169-97-60.clvdoh.adelphia.net) |
| 23:25.30 | m160858 | no ideas? |
| 23:26.21 | *** join/#asterisk kietlak (n=kietlak@11-mo3-6.acn.waw.pl) |
| 23:26.46 | *** join/#asterisk stkn_ (n=stkn@gentoo/developer/pdpc.active.stkn) |
| 23:27.29 | m160858 | hello? |
| 23:27.33 | CleanerX | ? |
| 23:27.59 | CleanerX | verbose...? |
| 23:28.08 | p1tst0p | how does one turn on *78 / *79 DND service ? |
| 23:28.50 | m160858 | I want to record the ougoing calls by each member of my queues, somebody have some idea? |
| 23:30.20 | m160858 | it seems that Fridays all go away of party |
| 23:30.25 | CleanerX | p1tst0p, depends on the kind of phone you use |
| 23:31.03 | CleanerX | p1tst0p, normally the phones should implement that |
| 23:31.28 | CleanerX | p1tst0p, so you need to configure them to act upon these numbers |
| 23:32.18 | *** part/#asterisk UyCaRumBa (n=administ@200.121.130.49) |
| 23:33.04 | p1tst0p | CleanerX, hmm i wanna use DND on my X-Lite, and my Avaya 4602 hard phone |
| 23:33.22 | [hC] | man, iax2 seems to be VERY sensitive to one way audio problems |
| 23:33.48 | bsdfreak | yep |
| 23:34.18 | [hC] | Im almost debating using sip to interconnect sites instead |
| 23:36.07 | bsdfreak | yeah |
| 23:36.26 | *** join/#asterisk iguy (n=iguy@rrcs-67-53-152-36.west.biz.rr.com) |
| 23:36.29 | bsdfreak | i only use iax2 if sip fails |
| 23:37.22 | [hC] | yeah.. plus sip will allow me to do RTP bridging, without harming my billing data |
| 23:37.42 | [hC] | cause i dont even see iax error messages as to why this happens 'sometimes' |
| 23:38.19 | [hC] | hmm.. i dont suppose anyone runs t1/e1 cards by the company whos name i shall not mention, on asterisk 1.2? |
| 23:39.57 | m160858 | somebody help me? |
| 23:40.23 | *** join/#asterisk Qwell (i=north@outboxes.com) |
| 23:42.59 | *** join/#asterisk |Vulture| (n=V@c-69-180-67-53.hsd1.fl.comcast.net) |
| 23:43.26 | *** join/#asterisk zotz (n=zotz@24.231.47.168) |
| 23:43.52 | |Vulture| | Anyone know how to deal with a large number of AGI scripts in the <defunct> state? |
| 23:44.29 | *** join/#asterisk dsfr (n=dsfr@pdpc/sponsor/digium/dsfr) |
| 23:44.40 | Qwell | |Vulture|: Don't let them get that way. :) |
| 23:44.45 | GXTi | restart asterisk |
| 23:45.01 | |Vulture| | Qwell: is there a trick to writing the script? adding some sort of timeout? |
| 23:45.02 | Qwell | They're probably not returning, or returning poorly |
| 23:45.14 | GXTi | youd expect asterisk would reap them properly |
| 23:45.25 | GXTi | its poor design on its part, not necessarily yours |
| 23:45.54 | |Vulture| | oh well there goes like 6 month uptime heheh |
| 23:45.57 | Qwell | I'd disagree with that. If you have a crappy script...asterisk shouldn't know/care... |
| 23:46.35 | Qwell | If I may draw an analogy: You write a poor C program with infinite loops, and memory leaks, and the like |
| 23:46.43 | Qwell | Is it gcc's job to fix it for you? Absolutely not. |
| 23:47.33 | |Vulture| | its strange another box with the same agi script has a 4 month uptime with no runaway procs |
| 23:48.19 | Qwell | That isn't so abnormal. If it's doing something with libraries that are different, or perhaps a file is missing...I don't know |
| 23:48.30 | *** join/#asterisk nvrs (i=RUR@toronto-HSE-ppp4257648.sympatico.ca) |
| 23:49.22 | |Vulture| | yea I guess its good ol debug time |
| 23:50.47 | *** join/#asterisk slayer192 (n=Blah@208.188.175.186) |
| 23:50.59 | bsdfreak | heh |
| 23:51.14 | bsdfreak | iax is really sensitive to high loads too evidently |
| 23:51.26 | Qwell | bsdfreak: it is...it's kind of a known issue |
| 23:51.35 | [hC] | sup qwell |
| 23:51.39 | Qwell | [hC]: y0 |
| 23:51.43 | Qwell | <-- at work |
| 23:51.59 | [hC] | Qwell: <-- at work at home.. i moved back to vancouver... after living in miami for a few years, i am FREEZING right now |
| 23:51.59 | [hC] | haha |
| 23:52.06 | Qwell | haha, sucker |
| 23:52.31 | [hC] | its 38 degrees F right now |
| 23:52.37 | Qwell | oh, that's nothing |
| 23:52.43 | [hC] | yeah its not horrible |
| 23:52.44 | Qwell | though...from miami...yeah |
| 23:52.48 | [hC] | not warm either |
| 23:52.54 | Qwell | I'd bitch too if it were < 60F :p |
| 23:52.58 | [hC] | im going to costa rica to do an asterisk install in jan tho |
| 23:53.02 | Qwell | wow |
| 23:53.06 | [hC] | woot |
| 23:53.07 | Qwell | need help? ;] |
| 23:53.12 | [hC] | heheh |
| 23:53.23 | [hC] | i used to live there, gonna go visit my gf, and her office is moving into a new building so we're gonna contract them out |
| 23:53.26 | Druken | i could do with a little costa |
| 23:53.31 | shido6 | cacacacosta |
| 23:53.34 | shido6 | rararar rico |
| 23:53.39 | shido6 | rica |
| 23:53.41 | Qwell | used to live in costa rica? you get around... |
| 23:53.46 | |Vulture| | hmm thats not good... when you boot a system and just see "GRUB" that usually means that grub is currupt and needs to be reinstalled correct? |
| 23:53.48 | shido6 | so do the viri |
| 23:53.50 | [hC] | yeah i travelled a bit the last 3-4 years.. |
| 23:54.32 | [hC] | so i got my osx86 box working basically 100%, but now im gonna upgrade to 10.4.3 which is supposedly alot better.. we'll see |
| 23:54.32 | [hC] | heh |
| 23:54.53 | [hC] | the issues i have right now are very minimal |
| 23:55.24 | slayer192 | yum osx86 |
| 23:55.32 | Qwell | yum install osx86? |
| 23:55.47 | Druken | angler: what kinda hookup ?? i hear prositutes are inexpensive there... |
| 23:55.51 | slayer192 | naaa... apt-get |
| 23:56.12 | Qwell | yum works better across platforms |
| 23:56.21 | angler | Druken, lol... i need voip account to dial locally in germany :) |
| 23:56.42 | Druken | just need termination ? |
| 23:56.57 | slayer192 | it's been a while since I've played with yum, I hear it has gotten better |
| 23:57.18 | Qwell | yum is okay. I like it on my sparc. I'd use apt over yum on x86 though |
| 23:57.32 | angler | Druken, termination and origination... |
| 23:57.34 | bsdfreak | qwell: yes I know it is |
| 23:57.40 | bsdfreak | that's why I'm avoiding iax2 unless I can't. |
| 23:57.44 | bsdfreak | sip->iax2->pstn |
| 23:57.46 | bsdfreak | heh |
| 23:57.47 | Qwell | bsdfreak: use sccp :P |
| 23:57.52 | bsdfreak | what is sccp |
| 23:57.53 | bsdfreak | heh |
| 23:57.57 | Qwell | cisco |
| 23:58.00 | [hC] | heh |
| 23:58.05 | bsdfreak | bah |
| 23:58.07 | Qwell | I <3 sccp |
| 23:58.08 | MstlyHrmls | Morse Code... |
| 23:58.09 | bsdfreak | :p |
| 23:58.13 | bsdfreak | i don't have a cisco phone or ata |
| 23:58.18 | Qwell | get one |
| 23:58.22 | [hC] | yeah i think im going to install 1.2.1 across the board rather than running various svn trunk releases, and move my interconnects to sip instead of iax2 |
| 23:58.22 | bsdfreak | meh |
| 23:58.29 | [hC] | iax2 has too many load/one way audio problems it seems |
| 23:58.34 | [hC] | and not always reproducable |
| 23:58.38 | [hC] | very frustrating |
| 23:58.43 | Qwell | one way audio on iax? never seen that |
| 23:58.57 | bsdfreak | i have |
| 23:58.59 | bsdfreak | but rarely |
| 23:59.08 | [hC] | yeah i get it sometimes when i have a call come into one * box, forward to another via iax2, then back to the first * box via iax2 again then out pstn |
| 23:59.20 | Qwell | oh...yeah...maybe |
| 23:59.22 | [hC] | if i dont use iax2, it works fine |
| 23:59.33 | bsdfreak | heh |
