| 00:03.28 | *** part/#asterisk kiscokid (n=ron@208.106.33.66) |
| 00:03.45 | N9URK | come out come out where ever you are! |
| 00:03.54 | bkw_ | you don't auth or register in extensions.conf |
| 00:04.00 | bkw_ | sip.conf maybe |
| 00:04.05 | N9URK | I screwed that one up |
| 00:04.06 | bkw_ | but surely not extensions.conf |
| 00:04.10 | N9URK | you are right |
| 00:04.12 | N9URK | too much work |
| 00:04.17 | N9URK | I do have it in sip.conf |
| 00:04.19 | bkw_ | any reason you're going to 1.4? |
| 00:04.29 | N9URK | an error of transcription |
| 00:04.49 | N9URK | bkw_ I am installing * on a new machine so I figure it best to go with the most up to date branch |
| 00:04.52 | bkw_ | if 1.4 doesn't have a must have feature you need stick with 1.2 |
| 00:05.02 | N9URK | bkw_ why is that? |
| 00:05.08 | bkw_ | 1.4 in my opinion is worse than 1.2.. 1.6 I hope will be better. |
| 00:05.11 | N9URK | I want to use the gui at some point |
| 00:05.30 | bkw_ | I don't touch 1.4 with a 100ft pole |
| 00:05.48 | N9URK | anyone else on here hate 1.4? |
| 00:07.11 | N9URK | have you used it? |
| 00:08.51 | N9URK | bkw_ have you used it? |
| 00:14.16 | *** join/#asterisk Obergandhi83 (n=Obergand@P449c.p.pppool.de) |
| 00:16.34 | *** join/#asterisk [hC] (n=hardcore@S01060016b61c8983.vf.shawcable.net) |
| 00:20.31 | *** join/#asterisk Smith-ub (n=biolacre@81-228-28-154-o1110.telia.com) |
| 00:21.47 | xuser_ | anyone used a orion gsm gateway with a quintum dx? |
| 00:26.17 | N9URK | can anyone help me with sip.conf in *1.4.4? |
| 00:26.48 | *** join/#asterisk [hC] (n=hardcore@S01060016b61c8983.vf.shawcable.net) |
| 00:26.58 | *** join/#asterisk J4k3 (n=jsuter@openwrt.us) |
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| 00:52.18 | *** join/#asterisk fr500 (n=andres@200.63.228.67) |
| 00:52.24 | fr500 | i got a question |
| 00:52.48 | fr500 | assume i have an asterisk server on a central office, and phones on remote locations of my organization |
| 00:53.18 | fr500 | when i do a local call (like two phones on a remote location) |
| 00:53.37 | fr500 | does traffic go thru the wan links, or only the call establishment |
| 01:01.26 | *** join/#asterisk rikstah (n=rick@rhamnett.plus.com) |
| 01:02.13 | *** join/#asterisk ivanfm (n=ivanfm@c93481ec.virtua.com.br) |
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| 01:35.38 | *** join/#asterisk hrmphh (i=patrick@notchill.com) |
| 01:35.45 | hrmphh | where are users' recorded greetings stored? |
| 01:35.58 | fr500 | assume i have an asterisk server on a central office, and phones on remote locations of my organization |
| 01:36.00 | fr500 | when i do a local call (like two phones on a remote location) |
| 01:36.17 | fr500 | does voice traffic go thru the wan links, or only the call establishment |
| 01:36.54 | hrmphh | like when the user records his voice |
| 01:36.55 | hrmphh | where is that kept |
| 01:36.57 | hrmphh | his name rather |
| 01:38.38 | *** join/#asterisk l3jj (n=chatzill@dsl093-001-038.det1.dsl.speakeasy.net) |
| 01:41.11 | *** join/#asterisk ivanfm (n=ivanfm@c93481ec.virtua.com.br) |
| 01:44.29 | JT | fr500: depends on settings |
| 01:44.51 | fr500 | JT: hmmm |
| 01:44.57 | fr500 | can you please elaborate |
| 01:46.14 | JT | fr500: canreinvite=yes in sip.conf allows endpoints to talk directly to each other for the media side of things |
| 01:46.27 | JT | but may not work if nat or call recording is involved |
| 01:51.06 | fr500 | ohh |
| 01:51.06 | fr500 | i see |
| 01:51.17 | hrmphh | where are users' recorded greetings stored? |
| 01:51.28 | JT | anyway, bbl |
| 01:52.01 | fr500 | jt thanks |
| 01:55.49 | *** join/#asterisk anthm (n=anthm@CPE-72-131-113-50.wi.res.rr.com) |
| 01:55.49 | *** mode/#asterisk [+o anthm] by ChanServ |
| 02:06.20 | *** join/#asterisk inventor_ (n=foo@ns2.drpcfix.com) |
| 02:10.14 | *** join/#asterisk tengulre (n=tengulre@222.90.66.10) |
| 02:10.29 | tengulre | hi,all |
| 02:10.56 | tengulre | does the asterisk-1.4 supportted mp3 format when musiconhold? |
| 02:19.15 | *** join/#asterisk kiscokid (n=Ron@adsl-216-101-109-187.dsl.snfc21.pacbell.net) |
| 02:21.25 | *** join/#asterisk Maxxed (i=foobar@65.59.245.122) |
| 02:21.42 | Maxxed | damn, im trying to upgrade a 7940 cisco to a sip image |
| 02:21.43 | Maxxed | its not picking up the files |
| 02:21.49 | Maxxed | well, it picked up the OS79XX.TXT, SEP<MAC>.cnf.xml, and XMLDefault.cnf.xml |
| 02:21.58 | Maxxed | i forget what kinda trick is done to get it to pick the old bin up |
| 02:22.01 | *** join/#asterisk mosty (n=mostyn@60-241-198-194.static.tpgi.com.au) |
| 02:22.12 | Maxxed | i have some of these real old bin's |
| 02:22.19 | Maxxed | P0S30203.bin |
| 02:22.39 | Maxxed | i remember there being something diffrent you do to the OS79XX.TXT |
| 02:22.42 | Maxxed | i just cant remember what |
| 02:23.44 | Maxxed | maybe my permisions r hosed.. |
| 02:23.46 | Maxxed | hum.. brb |
| 02:24.33 | Maxxed | ah thats what it looks like, ok i see the 19 RRQ "OS79XX.TXT" octet |
| 02:25.27 | Maxxed | but i dont actualy see the file transfer with tcpdump |
| 02:25.28 | Maxxed | bah, il be back in a few im gona go eat |
| 02:25.28 | Maxxed | i needa clear mind here, i know its something stupid :p |
| 02:29.30 | tengulre | Maxxed: Nobody here! |
| 02:30.05 | tengulre | today is sat. |
| 02:32.56 | *** join/#asterisk chenjm (n=chenjm@221.221.210.51) |
| 02:37.43 | *** join/#asterisk rikstah (n=rick@rhamnett.plus.com) |
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| 02:41.46 | kiscokid | let me look at my Cisco setup |
| 02:42.42 | *** join/#asterisk DaveCanoe (n=Dave@24.114.252.226) |
| 02:45.52 | kiscokid | seems like the name of the bin has to be in OS79XX.TXT and SIPDefault.cnf |
| 02:53.40 | lee_is_me | does anyone have a kavorkian apparatus available? I think I would rather drink HCL before banging my head against the wall with this TDM card any longer... |
| 02:54.00 | lee_is_me | it's literally driving me nuts :) |
| 02:54.46 | lee_is_me | I keep getting this error: ZT_CHANCONFIG failed on channel 3: No such device or address (6) |
| 02:55.16 | mosty | lee_is_me, is the module in correctly? |
| 02:55.40 | lee_is_me | I believe it is. I can try and pull it and check it for the 20th time today ;) |
| 02:56.01 | lee_is_me | maybe the card is not any good... |
| 02:56.58 | lee_is_me | what the heck, let me power down the pc and check it again... |
| 02:57.36 | lee_is_me | <I have had the absolute worst luck with pc's for this customer> |
| 02:58.38 | lee_is_me | both modules (3 and 4) seem to be nicely seated on the card... |
| 02:59.38 | lee_is_me | here is my info, btw ===> http://www.pastebin.ca/496145 |
| 03:00.28 | lee_is_me | TKD-Fender was helping me, but I had to reinstall linux which took a while... |
| 03:01.11 | *** join/#asterisk DrukenLPY (n=jdumais@CPE001346f4961f-CM00137189cb0c.cpe.net.cable.rogers.com) |
| 03:01.42 | *** join/#asterisk fujin (i=aj@unaffiliated/fujin) |
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| 03:07.21 | lee_is_me | on boot up, i get "no functioning hardware in /proc/zaptel" or something to that effect (scrolls too fast to be exact).... |
| 03:07.52 | *** part/#asterisk chenjm (n=chenjm@221.221.210.51) |
| 03:08.46 | kiscokid | you can see it with the dmesg command |
| 03:09.24 | kiscokid | sounds like the kernel is not seeing the card |
| 03:09.52 | lee_is_me | kiscokid: it's a lot of output and I'm still fairly new to linux. Pastebin is here: http://www.pastebin.ca/496145 |
| 03:10.06 | lee_is_me | lspci sees the card |
| 03:10.49 | *** join/#asterisk shinao1 (n=shinao1@196.3.63.252) |
| 03:10.57 | lee_is_me | i just moved the modules to 1-2 from 3-4 and am trying that... |
| 03:11.52 | lee_is_me | after moving the modules, i get a different error where it thinks i am using the the wrong interface |
| 03:12.10 | lee_is_me | i'm going to add my zaptel.conf to that pastebin now... |
| 03:14.42 | lee_is_me | I just modified the pastebin to include my zaptel.conf ===> http://www.pastebin.ca/496192 |
| 03:15.05 | mosty | if you move the modules, you have to change zaptel.conf |
| 03:15.40 | lee_is_me | FXO's are configured with fxsks, right? |
| 03:15.52 | mosty | yes |
| 03:15.53 | JunK-Y | correct |
| 03:16.07 | lee_is_me | I did, zaptel was configured fxsks=3-4 before |
| 03:16.37 | lee_is_me | i just changed it to what's in the pastebin and now get the error about not using the correct interface types |
| 03:16.40 | *** join/#asterisk aptura (n=jondoe@S010600a0c93f6f7e.vs.shawcable.net) |
| 03:17.02 | lee_is_me | ZT_CHANCONFIG failed on channel 1: Invalid argument (22) |
| 03:17.03 | lee_is_me | Did you forget that FXS interfaces are configured with FXO signalling |
| 03:17.04 | lee_is_me | and that FXO interfaces use FXS signalling? |
| 03:17.30 | lee_is_me | and FXO cards are red, correct? |
| 03:17.30 | aptura | nice |
| 03:17.47 | *** join/#asterisk SplasPood (n=nijwb@schizophrenia.paravolve.net) [NETSPLIT VICTIM] |
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| 03:18.06 | *** join/#asterisk Delvar (n=Delvar@host-83-146-53-46.bulldogdsl.com) [NETSPLIT VICTIM] |
| 03:19.02 | lee_is_me | i will do what ever is in my power to talk customers out of using analog lines from now on...way too much trouble with these things for me |
| 03:19.03 | *** join/#asterisk anderiv (n=anderiv@207-67-87-34.static.twtelecom.net) [NETSPLIT VICTIM] |
| 03:19.03 | *** join/#asterisk ixx (i=foobar@cpe-70-112-123-132.austin.res.rr.com) [NETSPLIT VICTIM] |
| 03:19.03 | *** join/#asterisk Nugget (i=nugget@dazed.notslacker.com) [NETSPLIT VICTIM] |
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| 03:19.42 | *** join/#asterisk tessier (n=treed@kernel-panic/sex-machines) [NETSPLIT VICTIM] |
| 03:19.42 | kiscokid | what about reliability? |
| 03:19.42 | lee_is_me | of course, many cases, customer wants what customer wants... |
| 03:19.42 | aptura | lee analog has its place |
| 03:19.42 | *** join/#asterisk SplasPood (n=nnijwb@schizophrenia.paravolve.net) [NETSPLIT VICTIM] |
| 03:19.42 | aptura | lee and thats how you keep a customer. Give them what thay want. |
| 03:19.47 | lee_is_me | aptura: i agree. i'm just frustrated right now because I'm going on my 7 th hour trying to get this card to work ;) |
| 03:20.10 | aptura | which card |
| 03:20.18 | kiscokid | people keep telling me that you can't get 99999 reliability unless you use analog |
| 03:20.27 | lee_is_me | TDM02b |
| 03:20.30 | aptura | kisco thats true |
| 03:20.54 | lee_is_me | i've seen lots of people asking about this error via google, but nothing that leads to a resolution unfortunately. |
| 03:21.20 | kiscokid | Its a Digium card? |
| 03:39.48 | *** join/#asterisk jbot (i=ibot@pdpc/supporter/active/TimRiker/bot/apt) |
| 03:39.49 | *** topic/#asterisk is Asterisk: The Open Source PBX -=- Asterisk 1.4.4 (April 27, 2007) Asterisk 1.2.18 (April 24, 2007), Zaptel 1.2.17.1, 1.4.2.1 (April 25, 2007) -=- Other fun channels: #asterisk-gui, #asterisknow, #asterisk-commits -=- Join #freepbx for freepbx/#trixbox for trixbox support. |
| 03:41.37 | *** join/#asterisk _omer (n=omer@lhr-mp-dig-p11-198.brain.net.pk) |
| 03:41.50 | kiscokid | lee: you did the kernel config right? |
| 03:44.18 | *** join/#asterisk coppice (n=chatzill@96.199.17.210.dyn.pacific.net.hk) |
| 03:45.18 | _omer | hello |
| 03:45.27 | _omer | anyhelp or suggestions? http://www.pastebin.ca/496232 |
| 03:46.40 | mosty | omer: why didn't you just ask that here, in one line? |
| 03:47.41 | *** part/#asterisk kiscokid (n=Ron@adsl-216-101-109-187.dsl.snfc21.pacbell.net) |
| 03:48.12 | _omer | It was just to clear everything.....otherwise...it starts further discussions |
| 03:48.21 | _omer | Is there a way to get to know the Call status in realtime? (Not on disconnection) |
| 03:49.58 | mosty | maybe the M(x) option of the dial command does what you want |
| 03:51.59 | _omer | any other way around? |
| 03:52.23 | mosty | not sure. what's wrong with that way? |
| 03:53.16 | _omer | I wont come to know, which SIP PEER Answered the call |
| 03:54.06 | *** join/#asterisk bmg505 (n=leon@196.209.183.132) |
| 03:55.26 | mosty | for statistics of some sort? perhaps a queue would help |
| 03:56.03 | *** join/#asterisk tschafer (n=tschafer@207.241.143.246) |
| 04:03.34 | *** join/#asterisk scardinal (n=supreme@port816.ds1-suoe.adsl.cybercity.dk) |
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| 04:22.54 | *** join/#asterisk nighty^ (n=nighty@sushi.rural-networks.com) |
| 04:38.28 | *** join/#asterisk naxxtor (n=naxxtor@vm209.contextshift.co.uk) |
| 04:43.32 | *** join/#asterisk mightnare (n=mike@p6060-ipad03motosinmat.mie.ocn.ne.jp) |
| 04:44.31 | *** join/#asterisk Gunnar (n=gunnar@nat.sigmasoft.com) |
| 04:50.04 | [TK]D-Fender | _omer, use the M option in Dial to make a DB entry |
| 04:50.21 | *** part/#asterisk mightnare (n=mike@p6060-ipad03motosinmat.mie.ocn.ne.jp) |
| 04:53.47 | *** join/#asterisk kiscokid (n=Ron@adsl-216-101-109-187.dsl.snfc21.pacbell.net) |
| 04:59.03 | *** join/#asterisk voltagex (n=voltagex@121.79.15.236) |
| 05:01.30 | voltagex | so naxxtor... |
| 05:09.29 | *** join/#asterisk rmayorga (n=rmayorga@unaffiliated/rmayorga) |
| 05:10.57 | *** join/#asterisk mightnare (n=mike@p2022-ipad05motosinmat.mie.ocn.ne.jp) |
| 05:18.00 | naxxtor | yes voltagex |
| 05:18.36 | *** join/#asterisk asteriskguy (n=learnast@cpe-75-80-111-113.socal.res.rr.com) |
| 05:18.56 | asteriskguy | hello all |
| 05:20.25 | carrar | HELO . |
| 05:23.11 | voltagex | naxxtor: are you going to explain the situation? |
| 05:25.28 | naxxtor | yeah sure |
| 05:26.03 | naxxtor | we're trying to work out why one of our extensions has got a huge delay on it before it gets connected. we've looked for waits and things that might incurr processing time but can't find anything |
| 05:29.28 | voltagex | the delay varies between 2 and 4 seconds. It's not transferring between boxes or anything like that. |
| 05:30.00 | mosty | what dialplan command is running during the pause? |
| 05:31.15 | voltagex | should be Playback...naxxtor? |
| 05:31.15 | *** join/#asterisk dseeb_ (n=dcb@CPE-124-177-38-240.vic.bigpond.net.au) |
| 05:31.29 | naxxtor | uhh |
| 05:31.40 | voltagex | Background() - we've checked that there's no pause in the actual file |
| 05:31.50 | naxxtor | yeah that's it |
| 05:32.00 | naxxtor | Background() then wait() |
| 05:32.23 | voltagex | but the delay is *before* the Background starts playing |
| 05:33.19 | naxxtor | indeed |
| 05:33.30 | *** join/#asterisk Mavvie (n=edwin@ppp1-208.lns1.syd7.internode.on.net) |
| 05:33.54 | [TK]D-Fender | naxxtor, ditch the "wait" |
| 05:34.43 | [TK]D-Fender | naxxtor, and it might help to know what kind of call is being answered. |
| 05:35.04 | naxxtor | any kind of call |
| 05:35.12 | naxxtor | exten => 4,1,Goto(info,s,1) <-- that's where it's coming from |
| 05:35.53 | voltagex | [TK]D-Fender: ? That's running after the Background, shouldn't there be a delay after, not before the Background? |
| 05:36.15 | naxxtor | also, it used to have an Answer before it |
| 05:36.21 | naxxtor | which we removed to no effect |
| 05:37.02 | [TK]D-Fender | naxxtor, Is is this delay BEFORE you see the background being called? |
| 05:37.19 | naxxtor | hang on lemme check |
| 05:37.23 | naxxtor | i seem to remember so |
| 05:37.54 | naxxtor | yes |
| 05:38.01 | [TK]D-Fender | naxxtor, pastebin the cli output of a problem call from beginning to end at verbose 10 |
| 05:38.21 | [TK]D-Fender | naxxtor, and does removing the WAIT help? |
| 05:38.29 | *** part/#asterisk kiscokid (n=Ron@adsl-216-101-109-187.dsl.snfc21.pacbell.net) |
| 05:40.11 | naxxtor | nope |
| 05:40.16 | naxxtor | we tried that# |
| 05:40.19 | [TK]D-Fender | naxxtor, ok, pastebin it |
| 05:41.36 | naxxtor | http://hvr.pastebin.co.uk/14625 |
| 05:41.50 | naxxtor | it's probably got some other crap mixed in there too |
| 05:42.41 | carrar | TK, do you know if you can use a gosubif inside of a dial(Local/blah) and ultimately return back to the Dial? |
| 05:43.53 | carrar | using a goto inside of a Dial(Local/blah) definately kills the return back to Dial(Local/blah) |
| 05:44.18 | carrar | I need to test that |
| 05:44.56 | *** join/#asterisk jeffik (n=Valued@206-248-128-28.dsl.teksavvy.com) |
| 05:45.20 | [TK]D-Fender | Maxxed, I count 3 waits in there before your background kicks in |
| 05:45.40 | [TK]D-Fender | Maxxed, plus AGI delay of unknown value |
| 05:45.45 | voltagex | [TK]D-Fender: you mean naxxtor right? |
| 05:45.58 | [TK]D-Fender | oops, yeah |
| 05:46.05 | [TK]D-Fender | naxxtor, rather |
| 05:46.25 | voltagex | [TK]D-Fender: ok, I'll help him find them, I have a shell on that box. |
| 05:46.30 | voltagex | thanks |
| 05:46.33 | [TK]D-Fender | carrar, Your local channel is its own little world. |
| 05:46.40 | voltagex | we didn't see those waits |
| 05:47.20 | [TK]D-Fender | voltagex, the pastebin is pretty damn blatant.... how blind can one be? :) |
| 05:47.22 | naxxtor | we can't see them in the plan either |
| 05:48.40 | naxxtor | are you talking about the ones going to the SIP connection? |
| 05:48.43 | naxxtor | cause that wasn't me |
| 05:49.55 | voltagex | naxxtor: you might have to shut off outside access to the box until we can work this out |
| 05:50.06 | naxxtor | hmm |
| 05:50.24 | voltagex | [TK]D-Fender: we only care about the delay from the first ivr to the info extension |
| 05:50.33 | voltagex | but yeah naxxtor , the cdr agi must be slow |
| 05:51.17 | voltagex | [TK]D-Fender: the delay happens between line 33 and 36 in that output |
| 05:51.23 | *** part/#asterisk joat (n=joat@ip70-160-147-169.hr.hr.cox.net) |
| 05:51.37 | voltagex | there is other junk in that, sorry. |
| 05:51.52 | naxxtor | as i said earlier |
| 05:56.09 | *** join/#asterisk Assid (n=assid@59.165.14.35) |
| 06:10.31 | *** join/#asterisk DarKnesS_WolF (n=wolf@unaffiliated/sherif) |
| 06:12.38 | *** join/#asterisk mihinomenest (n=argh@cerebus.clandestineresearch.com) |
| 06:13.51 | *** join/#asterisk steven|c (n=steve@66-215-217-140.dhcp.rvsd.ca.charter.com) |
| 06:14.34 | steven|c | Is it possible to setup asterisk to have several sip providers and randomly choose one, or switch to another if one is down? |
| 06:15.12 | DarKnesS_WolF | steven|c: yes |
| 06:16.51 | *** join/#asterisk masked (i=masked@shell.iinet.net.au) |
| 06:17.05 | steven|c | any hints/tips that can get me on my way |
| 06:17.55 | [TK]D-Fender | steven|c, Dial one, check the DIALSTATUS variable. Determine what you want to do based on the result. Wash-rinse-repeat |
| 06:18.46 | *** join/#asterisk Strom_M (n=strom@12.175.45.197) |
| 06:21.35 | asteriskguy | what is a local channel? |
| 06:21.55 | asteriskguy | and why would you use it? |
| 06:22.25 | [TK]D-Fender | asteriskguy, Go look it up on the WIKI. too big a question for here |
| 06:22.47 | asteriskguy | ok |
| 06:23.29 | coppice | a big question, eh? the answer must be 42 |
| 06:27.29 | DarKnesS_WolF | coppice: no 24 + a mirror :-) |
| 06:44.34 | *** join/#asterisk [Airwolf] (n=airwolf@voip.ymav.nl) |
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| 07:06.54 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 07:17.24 | *** join/#asterisk tzafrir_laptop (n=tzafrir@bzq-179-75-202.static.bezeqint.net) |
| 07:20.34 | tessier | An rtp packet is normally sent every 20ms right? |
| 07:22.55 | *** join/#asterisk remmo (n=junk@smack.isp.net.au) |
| 07:23.26 | *** join/#asterisk DrukenLPY (n=jdumais@CPE001346f4961f-CM00137189cb0c.cpe.net.cable.rogers.com) |
| 07:28.17 | *** join/#asterisk mistermocha (n=chef@adsl-75-22-50-160.dsl.irvnca.sbcglobal.net) |
| 07:28.54 | mistermocha | I'm having a wacky time trying to find UK dialplans... any suggestions? |
| 07:31.16 | *** join/#asterisk Shaun2222 (n=shaun@ip68-4-212-221.oc.oc.cox.net) |
| 07:38.13 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 07:39.27 | *** join/#asterisk oej (n=olle@cust-IP-19.data.tre.se) |
| 07:44.50 | *** join/#asterisk dudes (n=nixtux@66-216-226-114.dhcp.stcd.mn.charter.com) |
| 07:45.17 | *** join/#asterisk DaveCanoe (n=Dave@24.114.252.226) |
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| 07:52.19 | *** join/#asterisk reber (n=reber@cl-158.dub-01.ie.sixxs.net) |
| 07:56.31 | mistermocha | I'm having a wacky time trying to find UK dialplans... any suggestions? |
| 07:58.39 | *** join/#asterisk hansin321 (n=eric@c-71-196-138-170.hsd1.co.comcast.net) |
| 08:06.45 | *** join/#asterisk CrazyTux[m] (n=CrazyTux@cpe-24-162-48-94.houston.res.rr.com) |
| 08:10.36 | *** join/#asterisk brussel_ (n=brussel@72.130.172.213) |
| 08:25.05 | *** join/#asterisk [Airwolf] (n=airwolf@voip.ymav.nl) |
| 08:27.29 | *** join/#asterisk saftsack (n=oliver@p54A7D1FA.dip.t-dialin.net) |
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| 08:35.17 | *** join/#asterisk UlbabraB (n=salama@host241-43-static.72-81-b.business.telecomitalia.it) |
| 08:35.55 | *** join/#asterisk [Airwolf] (n=airwolf@voip.ymav.nl) |
| 08:42.02 | *** join/#asterisk cnbrk (n=canberk@teknopet.com) |
| 08:44.33 | dudes | <mistermocha> - UK Dialplan? |
| 08:46.41 | dudes | there really isn't much difference between UK and US dialplans |
| 08:46.52 | dudes | just how you express extensions |
| 08:47.09 | dudes | like US, _NXXNXXNXXX |
| 08:48.25 | dudes | but the wiki does have some info on UK http://www.voip-info.org/wiki/view/UK+Asterisk+Details |
| 08:49.50 | *** part/#asterisk voltagex (n=voltagex@121.79.15.236) |
| 08:51.13 | *** join/#asterisk saftsack (n=oliver@p54a7e621.dip.t-dialin.net) |
| 08:51.19 | dudes | http://www.voip-info.org/wiki/view/Asterisk+Dialplan+Patterns will be helpful |
| 08:51.53 | dudes | What is crazy is Greek numbers |
| 08:51.54 | dudes | heh |
| 08:55.40 | *** join/#asterisk adorah (n=Michael@87.69.57.73.cable.012.net.il) |
| 09:02.46 | *** join/#asterisk ploieel (n=ploieel@Fb284.f.ppp-pool.de) |
| 09:03.35 | ploieel | Die Geister, die ich rief... |
| 09:04.29 | dudes | yea I bet |
| 09:06.51 | ploieel | Hier bin ich, ich kann nicht anders! |
| 09:09.29 | *** join/#asterisk lorinc (n=ang@pool-8230.adsl.interware.hu) |
| 09:10.26 | dudes | Sprechen Sie hier Englisch. |
| 09:13.05 | *** join/#asterisk saftsack (n=oliver@p54a7e5a7.dip.t-dialin.net) |
| 09:17.24 | *** join/#asterisk drako (n=ljd@nelug/coreteam/luisjose) |
| 09:22.08 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 09:24.34 | ploieel | Die Geister, die ich rief... |
| 09:25.26 | dudes | Geist was? |
| 09:27.00 | *** join/#asterisk vgster (n=vgster@cpc2-ledn1-0-0-cust826.leed.cable.ntl.com) |
| 09:30.48 | dudes | póg mo asal Gearmánach |
| 09:42.04 | *** join/#asterisk keulin (n=cray@nat30-2-88-160-17-233.fbx.proxad.net) |
| 10:09.06 | *** join/#asterisk XIN01OZ (n=12@thegeneral.phatservers.com) |
| 10:19.07 | *** join/#asterisk lorinc (n=ang@pool-7210.adsl.interware.hu) |
| 10:25.04 | *** join/#asterisk bkw_ (i=brian@adsl-70-143-39-207.dsl.tul2ok.sbcglobal.net) |
| 10:32.23 | *** join/#asterisk cnbrk (n=canberk@teknopet.com) |
| 10:32.25 | cnbrk | hello |
| 10:32.41 | cnbrk | i have a problem, my asterisk server is not registering with my spa400 device |
| 10:33.44 | cnbrk | do you have any idea why |
| 10:36.42 | *** join/#asterisk keulin (n=cray@nat30-2-88-160-17-233.fbx.proxad.net) |
| 10:50.34 | *** join/#asterisk yannj_fr (n=yannj@vpn.intelunix.fr) |
| 11:12.27 | *** join/#asterisk CrazyTux[m] (n=CrazyTux@cpe-24-162-48-94.houston.res.rr.com) |
| 11:23.25 | robin_sz | cnbrk, try watching an the CLI with asterisk -rvvv and see what it says when the SPA400 tries to register |
| 11:24.10 | robin_sz | cnbrk, umm, the SPA400 ... is this one of the SPA's that sits between your network and the internet |
| 11:24.24 | robin_sz | ie it has two sockets, LAN and INTERNET? |
| 11:27.14 | robin_sz | I assume you have CAREFULLY followed the instructions in http://voxilla.com/voxilla-tips/voxilla-tips/the-linksys-spa400-and-asterisk-886.html |
| 11:30.00 | cnbrk | yeah i did |
| 11:30.11 | cnbrk | hey robin_sz are u still there |
| 11:36.54 | robin_sz | cnbrk, yeah, just |
| 11:39.58 | *** join/#asterisk danalien (n=danalien@unaffiliated/danalien) |
| 11:40.13 | *** join/#asterisk BS2 (n=bertus@snws.demon.nl) |
| 11:41.07 | cnbrk | so |
| 11:41.22 | cnbrk | the problem is, my asterisk server has to register with spa400 |
| 11:41.29 | cnbrk | spa400 will not register with my asterisk |
| 11:41.31 | cnbrk | so |
| 11:41.33 | cnbrk | allow=ulaw |
| 11:41.33 | cnbrk | canreinvite=no |
| 11:41.33 | cnbrk | context=from-pstn |
| 11:41.33 | cnbrk | disallow=all |
| 11:41.33 | cnbrk | host= Your SPA400 IP address |
| 11:41.34 | cnbrk | insecure=very |
| 11:41.36 | cnbrk | nat=no |
| 11:41.38 | cnbrk | port=5060 |
| 11:41.40 | cnbrk | qualify=yes |
| 11:41.42 | cnbrk | type=peer |
| 11:41.48 | cnbrk | this is the sip.conf file |
| 11:41.58 | cnbrk | and i have a line for register => blabla... |
| 11:42.04 | cnbrk | to make asterisk register with my spa400 |
| 11:42.08 | cnbrk | however it is not registering |
| 11:44.00 | robin_sz | sip debug peer <ip of spa400> |
| 11:44.09 | robin_sz | see if its actually communicating |
| 11:46.00 | cnbrk | sec |
| 11:47.15 | BS2 | cnbrk: does the debug say anything about codec negotiation ? |
| 11:49.14 | cnbrk | :( |
| 11:49.19 | cnbrk | when i call the spa400 |
| 11:49.26 | cnbrk | it is not communicating with asterisk |
| 11:49.38 | cnbrk | my asterisk is not registerig with spa400, sip show registry shows nothing |
| 11:50.17 | BS2 | it should register before you call it |
| 11:50.32 | cnbrk | yes |
| 11:50.40 | cnbrk | no idea why it doesn't though |
| 11:51.09 | BS2 | so when you reset the spa400, and run the debug on asterisk.. what do you see |
| 11:51.13 | *** join/#asterisk saftsack (n=oliver@pd9e0705b.dip.t-dialin.net) |
| 11:52.00 | cnbrk | just a sec |
| 11:55.47 | cnbrk | http://pastebin.ca/496690 |
| 11:55.59 | *** join/#asterisk friedrich| (n=friedric@e177251000.adsl.alicedsl.de) |
| 11:58.46 | cnbrk | what do you think |
| 12:00.30 | BS2 | did you call one of the spa400 number during debug, or did you just reset it ? |
| 12:00.45 | cnbrk | this is when i call |
| 12:00.49 | BS2 | ok |
| 12:01.02 | *** join/#asterisk zotz (n=zotz@24.244.163.157) |
| 12:01.53 | BS2 | try changing to type=friend and move disallow=all above allow=ulaw |
| 12:02.24 | cnbrk | i did that before |
| 12:02.29 | cnbrk | nothing happened the same results... |
| 12:04.18 | BS2 | take a look at http://www.voipsupply.com/product_info.php?products_id=1692 |
| 12:06.20 | cnbrk | i used the same guide :( |
| 12:07.30 | BS2 | look at the spa400 config |
| 12:08.36 | cnbrk | sorry? |
| 12:13.14 | BS2 | you are trying to setup your aps400 (4 port analog gateway) to communicate to your asterisk box .. right ? |
| 12:13.29 | cnbrk | yes |
| 12:14.27 | BS2 | every line on the spa400 should register on the asterisk box |
| 12:14.51 | cnbrk | the spa 400 is not a device which will register with asterix |
| 12:14.55 | cnbrk | asterisk |
| 12:15.02 | cnbrk | it is a device which asterisk registers with |
| 12:15.16 | cnbrk | it only allows connections and registrations to be made |
| 12:15.23 | cnbrk | it is a sip server |
| 12:16.11 | *** join/#asterisk Strom_M (n=strom@12.175.45.203) |
| 12:16.16 | Assid | err |
| 12:16.23 | Assid | anyone know any toll free providers |
| 12:16.33 | Assid | tollfreegateway.com isnt working |
| 12:17.09 | BS2 | but did you check the passwords on the spa400, codecs to use etc.. |
| 12:18.13 | cnbrk | yeah |
| 12:18.16 | cnbrk | everything is set |
| 12:18.22 | cnbrk | no passwords on spa400 |
| 12:18.24 | cnbrk | only a username |
| 12:18.28 | cnbrk | which is also set to spa400 |
| 12:18.30 | cnbrk | codecs are set |
| 12:18.42 | cnbrk | it is not placing a call on asterisk, so no codec problem |
| 12:18.47 | cnbrk | also no password... |
| 12:19.25 | BS2 | some sip devices dont register when the allowed codec list is not the same on both sides |
| 12:20.03 | BS2 | at the asterisk side you only use ulaw ,,, how about the spa400 side |
| 12:20.21 | cnbrk | ulaw too, but i'll recheck |
| 12:24.09 | cnbrk | now i set both of them to g729 |
| 12:24.09 | cnbrk | still no change |
| 12:24.42 | BS2 | when on the asterisk box what does sip show registry show ? |
| 12:25.28 | cnbrk | no :( |
| 12:25.42 | BS2 | doesnt show anything ? |
| 12:25.46 | cnbrk | no |
| 12:25.57 | cnbrk | *CLI> sip show registry |
| 12:25.57 | cnbrk | Host Username Refresh State |
| 12:25.57 | cnbrk | *CLI> |
| 12:26.11 | Strom_M | sip show users or sip show peers |
| 12:26.15 | Strom_M | not sip show registry |
| 12:27.04 | BS2 | what version of asterisk are you running ? |
| 12:27.55 | cnbrk | 1.2.17 |
| 12:28.30 | BS2 | what does sip show peers show |
| 12:28.58 | *** join/#asterisk RypPn (i=TuMbL@rosscom.demon.co.uk) |
| 12:29.14 | BS2 | is there any sign of the configured spa400 registration, even when is not registered |
| 12:30.39 | cnbrk | sip show peers |
| 12:30.39 | cnbrk | Name/username Host Dyn Nat ACL Port Status |
| 12:30.39 | cnbrk | spa400 192.168.2.170 5060 OK (61 ms) |
| 12:31.03 | RypPn | afternoon all, I've successfully completed my testing using a x100p-clone and would like to move to production. can someone recommend a replacement with better audio quality for one fxo? |
| 12:31.52 | BS2 | cnbrk: that looks ok |
| 12:32.02 | cnbrk | yeah, however calls are not forwarded to my asterisk |
| 12:33.55 | *** join/#asterisk saftsack (n=oliver@pd9e0705b.dip.t-dialin.net) |
| 12:35.10 | cnbrk | uff |
| 12:35.15 | cnbrk | will this work with my asterisk |
| 12:36.22 | BS2 | is there any kind of a dialplan config on the spa400 web interface ? |
| 12:38.14 | cnbrk | no |
| 12:38.50 | BS2 | ok try insecure=invite,port in place of very |
| 12:39.03 | *** join/#asterisk Nivex (n=kjotte@user-0ce2kma.cable.mindspring.com) |
| 12:39.04 | BS2 | and type= friend |
| 12:39.09 | cnbrk | just a sec |
| 12:39.28 | *** join/#asterisk Tomasu (i=moose@S010600045a4c73cc.ed.shawcable.net) |
| 12:39.55 | Tomasu | I recall hearing that 1.4.x has some form of included web based administration interface? Is this true? |
| 12:39.58 | *** join/#asterisk KmiKze (n=KmiKze@brutus.fragtime.net) |
| 12:40.42 | BS2 | Tamsu: do you mean AsteriskNOW ?? |
| 12:40.42 | *** join/#asterisk Fieldy (i=jQs2W9Qb@gentoo/contributor/Fieldy) |
| 12:41.21 | Tomasu | BS2: I have no idea. I introduced a friend to asterisk, and he installed a more recent version (1.4.x) whereas I was still on 1.2.x, and he mentioned there was a bundled web admin.. |
| 12:42.42 | BS2 | you can have a webinterface for 1.4x but when you install AsteriskNOW it comes with the webadmin. |
| 12:42.52 | Tomasu | is that a full distro? |
| 12:42.58 | BS2 | yep |
| 12:43.21 | Tomasu | I currently use asterisk on a "server" that already does other things, so I can't just wipe it. |
| 12:43.51 | cnbrk | still problems |
| 12:43.58 | BS2 | no you dont have to |
| 12:44.15 | Tomasu | oh, so its not a standalone unix/linux distro? |
| 12:44.21 | Tomasu | just another version of asterisk? |
| 12:44.22 | BS2 | you can install mysql realtime config and FreePBX on your current asterisk |
| 12:45.04 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 12:45.12 | BS2 | I havent installed AsteriskNOW, but as far as I have seen its a cd ISO and it installes a complete system for you |
| 12:45.47 | Tomasu | yeah, and I can't just do that on my box ;) |
| 12:46.01 | BS2 | so when you dont want to wipe your current system take a look at FreePBX for you current asterisk install |
| 12:46.27 | Tomasu | ahh. ok, I thought FreePBX was another full distro... |
| 12:46.34 | Tomasu | shows how much I know |
| 12:46.38 | Tomasu | thanks |
| 12:46.54 | Tomasu | so would you suggest I upgrade to 1.4? |
| 12:49.08 | Tomasu | nice. the screenshots on the freepbx wiki are broken ::) |
| 12:52.24 | *** join/#asterisk ecam (n=ecam@bb121-6-102-119.singnet.com.sg) |
| 12:52.40 | ecam | hey people |
| 12:53.26 | ecam | im not sure if this is under asterisk, but is this error due to asterisk or a config error or something? |
| 12:53.27 | ecam | Reason: Q.850;cause=100;text="Invalid information element contents" |
| 12:53.36 | ecam | how do i fix it? |
| 12:59.30 | Tomasu | damn asterisk builds fast from source. |
| 12:59.45 | Tomasu | then again I have a icrecream cluster... |
| 13:00.32 | *** join/#asterisk coppice (n=chatzill@96.199.17.210.dyn.pacific.net.hk) |
| 13:07.15 | *** join/#asterisk SuperID (n=gary@c-65-96-225-97.hsd1.ma.comcast.net) |
| 13:20.00 | *** join/#asterisk fakhir (n=Fakhir@unaffiliated/fakhir) |
| 13:24.13 | *** join/#asterisk Superbaloo (n=Superbal@home.superbaloo.net) |
| 13:24.15 | Superbaloo | Hello |
| 13:24.40 | Superbaloo | I have some problem to configure my pabx |
| 13:24.54 | Superbaloo | i'm using a sip account provided |
| 13:24.58 | Superbaloo | by my isp |
| 13:25.32 | Superbaloo | i'm creating two connection on it called freephonie_out1 and freephonie_out2 |
| 13:26.09 | Superbaloo | i want to "load balance" the outgoing connection on them |
| 13:26.36 | Superbaloo | but if freephonie_out1 is busy, asterisk is still trying to use it |
| 13:27.01 | Superbaloo | here is my extension.conf |
| 13:27.04 | Superbaloo | exten => _0.,1,ChanIsAvail(SIP/freephonie_out2&SIP/freephonie_out1|js) |
| 13:27.04 | Superbaloo | exten => _0.,2,Dial(${CUT(AVAILCHAN||1)}/${EXTEN}) |
| 13:27.04 | Superbaloo | exten => _0.,3,Hangup |
| 13:27.30 | Superbaloo | i can't figure it out why it doesn't work :/ |
| 13:28.39 | ecam | anyone knows what this means? |
| 13:28.39 | ecam | Reason: Q.850;cause=100;text="Invalid information element contents" |
| 13:38.38 | ploieel | Hier bin ich, ich kann nicht anders! |
| 13:38.45 | *** join/#asterisk Strom_M (n=strom@206.166.206.34) |
| 13:53.23 | tzafrir_laptop | Superbaloo, why CUT? IIRC ChanIsAvail also provides another variable with the dial target |
| 13:55.36 | Corydon76-home | Superbaloo: I'd suggest using groups, instead of trying to rely on the sip call-limits |
| 13:57.20 | tzafrir_laptop | Superbaloo, AVAILORIGCHAN |
| 13:57.20 | Superbaloo | tzafrir_laptop, ? |
| 13:57.20 | Superbaloo | hum |
| 13:57.21 | Corydon76-home | i.e. Set(GROUP()=foo) Dial(SIP/${EXTEN}@${IF($[${GROUP(foo)} > 1]?freephonie_out2:freephonie_out1)} |
| 13:59.03 | Superbaloo | Corydon76-home, don't understand this command :/ |
| 13:59.14 | Superbaloo | what can i do with this ? |
| 13:59.29 | Corydon76-home | Try it |
| 13:59.50 | tzafrir_laptop | Set and Dial on the same line? |
| 14:00.04 | Corydon76-home | tzafrir_laptop: obviously, not |
| 14:01.26 | Superbaloo | Corydon76-home, doesn't work :/ |
| 14:01.27 | Corydon76-home | But if you associate groups with your channel usage, you can instantly tell how many of each are in use at any particular time |
| 14:01.49 | Superbaloo | exten => _0.,1,Set(GROUP()=foo) Dial(SIP/${EXTEN}@${IF($[${GROUP(foo)} > 1]?freephonie_out2:freephonie_out1)} |
| 14:01.50 | Superbaloo | exten => _0.,2,Hangup |
| 14:01.59 | Superbaloo | i'm using it like this |
| 14:02.08 | Corydon76-home | That should be 1, 2, and 3, not 1 and 2. |
| 14:02.20 | Strom_M | or 1, n, and n ;) |
| 14:02.26 | Superbaloo | :P |
| 14:02.45 | *** join/#asterisk cspot (i=cspot@ip68-1-63-100.pn.at.cox.net) |
| 14:03.03 | *** join/#asterisk DaveCanoe (n=Dave@137.122.39.10) |
| 14:03.07 | Corydon76-home | As this is a line-oriented medium, I wanted to give both lines at the same time, to avoid interlacing them with other people's text |
| 14:03.09 | Superbaloo | Corydon76-home, [May 19 16:02:15] WARNING[31625]: func_logic.c:106 acf_if: Syntax IF(<expr>?[<true>][:<false>]) |
| 14:03.44 | Superbaloo | <PROTECTED> |
| 14:03.45 | Superbaloo | no ? |
| 14:03.52 | Corydon76-home | NO! |
| 14:04.01 | Strom_M | cocks. |
| 14:04.30 | Corydon76-home | Are you just a cut-n-paste lackey or do you actually understand extensions.conf logic? |
| 14:05.16 | Superbaloo | Corydon76-home, i don't understand every lines in the default extensions.conf (most of them are commented) |
| 14:05.17 | *** join/#asterisk var1 (n=james@77-97-208-6.cable.ubr11.edin.blueyonder.co.uk) |
| 14:05.34 | Corydon76-home | Nevermind, then. Forget I said anything. |
| 14:05.45 | Superbaloo | but i understand every lines i typed but yours |
| 14:07.47 | Superbaloo | Corydon76-home, i don't even can find a documentation for the "group" command :/ |
| 14:07.57 | Strom_M | show function GROUP |
| 14:09.06 | Superbaloo | Strom_M, where ? |
| 14:09.09 | Superbaloo | in the cli? |
| 14:09.12 | Strom_M | at the asterisk CLI |
| 14:09.25 | Strom_M | or possibly on a vintage 1962 Teletype |
| 14:09.27 | Strom_M | it's your choice |
| 14:09.40 | Superbaloo | Kanette*CLI> show function group |
| 14:09.40 | Superbaloo | No such command 'show function group' (type 'help' for help) |
| 14:09.43 | Superbaloo | :/ |
| 14:09.50 | Strom_M | for fuck's sake, use tab-complete |
| 14:09.54 | Strom_M | GROUP, not group |
| 14:10.27 | Superbaloo | show function GROUP |
| 14:10.27 | Superbaloo | No such command 'show function GROUP' (type 'help' for help) |
| 14:10.28 | Superbaloo | same |
| 14:10.36 | Superbaloo | i tryed both |
| 14:10.39 | Strom_M | what version of asterisk are you running? |
| 14:10.43 | Superbaloo | 1.4 |
| 14:10.55 | Strom_M | cocks |
| 14:10.57 | Superbaloo | SVN-trunk-r61788 |
| 14:11.03 | Strom_M | uh |
| 14:11.03 | Corydon76-home | That's not 1.4 |
| 14:11.05 | Strom_M | that's not 1.4 |
| 14:11.10 | Superbaloo | hum |
| 14:11.35 | Corydon76-home | You should pay attention to that little notice at startup that says you should NOT USE THIS FOR PRODUCTION |
| 14:11.59 | *** join/#asterisk ariel_ (n=ariel_@70-46-87-154.ftl.fdn.com) |
| 14:12.04 | Strom_M | Corydon76-home: obviously he can't even read the documentation, much less the notices |
| 14:28.27 | Superbaloo | ok i'm now on 1.4.4 |
| 14:29.17 | Superbaloo | i can show function |
| 14:29.33 | Superbaloo | but your line is still not working |
| 14:32.27 | *** join/#asterisk Fieldy (i=8KMj2FiJ@gentoo/contributor/Fieldy) |
| 14:39.14 | *** join/#asterisk bintut (n=bintut@cm199.gamma182.maxonline.com.sg) |
| 14:40.30 | bintut | hello all.. anyone running asterisk on debian etch? i just apt-get'ed asterisk and zaptel but the wtcdm module is not loaded.. does it mean that i have to create a kernel module of zaptel to my kernel? |
| 14:49.18 | *** join/#asterisk `Sean (i=Un1x@CPE000c248d137c-CM00111ae601f8.cpe.net.cable.rogers.com) |
| 14:50.58 | *** join/#asterisk `Sean (i=Un1x@CPE000c248d137c-CM00111ae601f8.cpe.net.cable.rogers.com) |
| 14:56.48 | bbryant | bintut, you should try "modprobe wtcdm" |
| 14:58.41 | Qwell | wctdm :) |
| 15:01.25 | Qwell | man, I seriously can't wait for Monday... |
| 15:03.35 | file | why is that now? ^_^ |
| 15:06.22 | *** join/#asterisk rmayorga (n=rmayorga@unaffiliated/rmayorga) |
| 15:08.28 | *** join/#asterisk MrTelephone (n=test@bas13-toronto63-1242371594.dsl.bell.ca) |
| 15:08.36 | MrTelephone | hey |
| 15:09.12 | var1 | hey MrTelephone |
| 15:09.23 | Qwell | file: no reason :D |
| 15:09.27 | MrTelephone | hows it goin |
| 15:10.34 | MrTelephone | whoever uses sangoma pri use BETA drivers if you run channel banks that operate using RBS loopstart/groundstart signalling |
| 15:10.58 | *** join/#asterisk ploieel (n=ploieel@Fb24a.f.ppp-pool.de) |
| 15:11.40 | MrTelephone | anyone have the pdf manual for the Norstar CICS |
| 15:13.17 | *** join/#asterisk zotz (n=zotz@24.244.163.157) |
| 15:15.20 | MrTelephone | im lazy this morning, doesn anyone have a short script already made to strip the area code off local calls |
| 15:15.33 | Qwell | umm |
| 15:15.55 | Qwell | exten => 256NXXXXXX,1,Dial(${EXTEN:3}) ? |
| 15:16.14 | MrTelephone | inbound calls |
| 15:16.24 | MrTelephone | the phone company is sending all 10 digits on local calls |
| 15:16.26 | MrTelephone | ?? |
| 15:16.29 | MrTelephone | on the callerid |
| 15:16.49 | MrTelephone | my cisco ata186 adaptor strips the 807 automatically and I really don't know how it knows the 807 is local |
| 15:16.58 | MrTelephone | the adaptor doesn't know what the phone number is |
| 15:17.20 | MrTelephone | you know what I mean? |
| 15:17.28 | MrTelephone | 807 shows up but my analog phone shoes it (807) |
| 15:17.41 | MrTelephone | so if I hit redial it dials only the last 8 digits |
| 15:20.02 | MrTelephone | need if regex calleridnum =~ 807229{/d,4} then calleridnum = $EXTEN:3 |
| 15:20.42 | MrTelephone | it bugs me how the cisco ata figures it out :-/ |
| 15:23.14 | MrTelephone | maybe it was because I used the did for the loginid |
| 15:28.43 | var1 | what does status unmonitored mean? and why does my device not show up as registered? |
| 15:28.54 | coppice | interesting - http://www.magiclink.cn/en/other/Asterisk.html |
| 15:29.36 | MrTelephone | qualify=no in your config |
| 15:29.50 | MrTelephone | what device do you have? |
| 15:30.16 | MrTelephone | device hasn't sent register request |
| 15:30.51 | var1 | I have 2 devices the sipgate Netgear box and the X-Lite softfone, the softfone is unmonitered. |
| 15:31.21 | var1 | the Netgear box registers just fine. |
| 15:31.23 | *** join/#asterisk angryuser (n=Miranda@df01t2-62-34-203-156.d4.club-internet.fr) |
| 15:31.43 | MrTelephone | try and make a call with the softfone |
| 15:31.55 | MrTelephone | see if it registers when you make the call |
| 15:32.53 | var1 | nope I tried calling 1000 and call was not approved and it didn't register. |
| 15:33.19 | var1 | pretty sure it was working last night, I only changed 1 lline of text. |
| 15:33.47 | var1 | I mean it registered last night. |
| 15:36.53 | var1 | Is there a softfone you could recommend where I can see the messages as it attempts to register? |
| 15:37.16 | angryuser | i ned somebody to help me with b410p isdn crd from digium |
| 15:37.54 | angryuser | my port's go down after i call out with any of them, or receive first call |
| 15:38.05 | angryuser | http://www.pastebin.ca/496959 more details |
| 15:38.22 | MrTelephone | ${REGEX("^807229/d/d/d/d" ${CALLERIDNUM}) |
| 15:38.37 | MrTelephone | is that right? |
| 15:38.37 | angryuser | i am near to make this card fly out of the window |
| 15:39.02 | MrTelephone | what line of text did u change |
| 15:39.04 | var1 | hehe flying cards |
| 15:39.24 | MrTelephone | var1, eyebeam is good but they want money |
| 15:42.39 | SwK | xLite == eyebeam - a few features - cost |
| 15:43.15 | MrTelephone | xlite works in windows? |
| 15:43.40 | var1 | thx I'm looking at options. |
| 15:43.51 | MrTelephone | - operating system diversity - cosmetic appearance |
| 15:43.57 | MrTelephone | haha |
| 15:44.01 | MrTelephone | i have to try it |
| 15:44.19 | *** join/#asterisk hmepas (n=hmepas@ppp85-140-77-110.pppoe.mtu-net.ru) |
| 15:45.43 | MrTelephone | i installed cups and made a few perl scripts that go through cdr and bill and the bills automatically get printed or emailed |
| 15:45.46 | MrTelephone | it kicks ass |
| 15:45.58 | MrTelephone | now i can go suntanning and everything will run itself |
| 15:46.00 | MrTelephone | yeah right |
| 15:46.06 | hmepas | hi! two questions which complete asterisk doc you could advice novice user to start from. And the second: i have two sip phones, added them to asterisk they could call each over but don't listen any voice in line even when asterisk reports it's "native bridge" (no code translation). |
| 15:46.12 | SwK | xlite works on everything... windows mac and linux |
| 15:46.14 | hmepas | any suggestion? |
| 15:46.30 | SwK | jbot, book? |
| 15:46.42 | jbot | book is, like, a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 15:46.42 | SwK | ~book |
| 15:46.44 | jbot | from memory, book is a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 15:46.44 | MrTelephone | hmepas, one way audio? |
| 15:46.59 | hmepas | ty, SwK |
| 15:47.08 | MrTelephone | i have a sangoma dual port t1 card and it has a hard time supporting 2 t1s |
| 15:47.19 | hmepas | MrTelephone: what's ur mean? Nope i need both way audio. |
| 15:47.21 | MrTelephone | i wonder how the 4 port even works |
| 15:47.50 | hmepas | it's one soft phone on my PC and another is HW D-Link phone. I could call each other by asterisk on my server but can't listen anything. |
| 15:47.57 | MrTelephone | hmepas, NAT causes a lot of problems with audio.. usually one person can hear the other |
| 15:48.10 | MrTelephone | but then that person cannot hear them |
| 15:48.24 | hmepas | nope. no voice at all. |
| 15:48.35 | SwK | hmepas is the hardphone doing G729? |
| 15:48.36 | MrTelephone | are they on a local network |
| 15:48.42 | MrTelephone | make sure they are both dowin ulaw |
| 15:48.45 | SwK | yeah |
| 15:48.46 | MrTelephone | doing ulaw |
| 15:48.52 | SwK | disallow=all |
| 15:48.55 | SwK | allow=aluw |
| 15:49.07 | SwK | on the sip definitions for each device |
| 15:49.28 | hmepas | k. will try. donnu which codec both uses. and yes, one is from local network and the other from internet via NAT. |
| 15:49.47 | SwK | hmepas: NAT can cause headaches |
| 15:50.47 | hmepas | btw i tried to turn nat support off and use STUN. |
| 15:50.47 | SwK | and Device -> NAT -> internet -> NAT -> asterisk is the worst |
| 15:50.47 | hmepas | but via sniffer any way i saw asterisk trying to send packets to local address |
| 15:51.22 | SwK | so the softphone is on a public IP? |
| 15:51.35 | hmepas | SwK yes but via NAT. |
| 15:51.39 | SwK | hah |
| 15:51.48 | hmepas | and Device is private IP. |
| 15:51.50 | SwK | thats double natting and is a PITA to get working correctly |
| 15:52.23 | *** part/#asterisk SuperID (n=gary@c-65-96-225-97.hsd1.ma.comcast.net) |
| 15:52.24 | hmepas | why double? My Device not NATing |
| 15:52.33 | hmepas | it's use asterisk as a proxy |
| 15:52.54 | hmepas | as well as soft phone so NAT is only between soft phone and asterisk server. |
| 15:53.11 | MrTelephone | exten => s,1,Set(CALLERIDNUM=${IF( $[${REGEX("^807229/d/d/d/d" ${CALLERIDNUM})}]? ${EXTEN:3} : ${EXTEN})}) |
| 15:53.17 | MrTelephone | can someone comment on this |
| 15:53.31 | MrTelephone | what a mess with brackets |
| 15:53.32 | MrTelephone | lol |
| 15:54.00 | hmepas | bah... it's that's poor PDF books without content links... =( |
| 15:54.04 | MrTelephone | qwell ur the expert, think it will work? |
| 15:54.54 | tzafrir_laptop | MrTelephone, the ${EXTEN} in that line is 's', and ${EXTEN:3} is thus mpty |
| 15:55.23 | SwK | yeah what tzafrir_laptop said |
| 15:56.06 | hmepas | btw, another stupid things i setup special number with context which only sayin one, two, three, four one the line and when i am callin from soft phone i am hear that numbers while can't hear it from HW phone. |
| 15:59.55 | MrTelephone | really? |
| 16:00.03 | MrTelephone | s,1 is ther first line on an incoming call |
| 16:00.06 | MrTelephone | oh |
| 16:00.07 | MrTelephone | haha |
| 16:00.10 | MrTelephone | ok |
| 16:00.33 | MrTelephone | :3 will strip three chars off any veriable? |
| 16:05.04 | Corydon76-home | :3 starts at offset 3, yes |
| 16:05.47 | Corydon76-home | The first optional argument to a variable name is the offset, and the second optional argument is the length |
| 16:06.10 | steven|c | qwerty |
| 16:06.28 | steven|c | how stable would you consider asterisk |
| 16:06.43 | MrTelephone | very stable |
| 16:06.53 | steven|c | something i can setup and never touch? |
| 16:07.04 | MrTelephone | if I would have screwed in my pri card properly I'd be running for a month without any hitches |
| 16:07.16 | MrTelephone | no you have to do computer maintanence |
| 16:07.28 | MrTelephone | if you want something u never want to touch buy a norstar or nortel system |
| 16:07.32 | Corydon76-home | Add/change/delete |
| 16:07.48 | Corydon76-home | Every telephone system needs maintenance |
| 16:09.21 | *** join/#asterisk Koshatul (n=evangeli@ppp188-33.static.internode.on.net) |
| 16:09.52 | angryuser | has anybody has this? http://bugs.digium.com/view.php?id=9400#bugnotes misdn related problems |
| 16:11.55 | *** join/#asterisk bluelinq (n=bluelinq@dsl-7-36.cofs.net) |
| 16:12.04 | bluelinq | hello everyone |
| 16:12.15 | bluelinq | <PROTECTED> |
| 16:12.15 | bluelinq | <PROTECTED> |
| 16:12.15 | bluelinq | <PROTECTED> |
| 16:12.18 | *** join/#asterisk e-Regg4e (i=ganja@201.225.15.26) |
| 16:16.02 | cnbrk | hello everyone |
| 16:16.10 | cnbrk | who has experience with spa400 and asterisk? |
| 16:17.46 | e-Regg4e | i just have experience with spa3102 |
| 16:17.59 | e-Regg4e | i bought recently a pap2 and i want to do something that i don't know if its posible |
| 16:18.16 | cnbrk | well i need somebody to help me with spa400 and asterisk configuration for incoming calls |
| 16:18.24 | cnbrk | all the google articles failed :( |
| 16:18.26 | e-Regg4e | instead of connect analog phones to line1 and line2 |
| 16:18.40 | e-Regg4e | i want to connect line1 and line2 to an analog pbx (panasonic) |
| 16:18.53 | e-Regg4e | any ideas? |
| 16:19.43 | e-Regg4e | cnbrk: incoming calls from? |
| 16:19.49 | MrTelephone | revised... exten => s,1,Set(CALLERIDNUM=${IF( $[${REGEX("^807229/d/d/d/d" ${CALLERIDNUM})}]? ${$CALLERIDNUM:3} : ${CALLERIDNUM})}) |
| 16:22.02 | MrTelephone | is it normal to see your memory slowly become all used up after a few weeks of asterisk running? |
| 16:23.56 | cnbrk | incoming calls from pstn |
| 16:23.59 | cnbrk | to asterisk |
| 16:24.02 | cnbrk | they are not working |
| 16:24.16 | cnbrk | and i definetely need them working with my asterisk :( |
| 16:24.22 | MrTelephone | lol |
| 16:24.26 | MrTelephone | yeah its really important |
| 16:24.37 | MrTelephone | how do they get to your asterisk box? |
| 16:24.46 | MrTelephone | analog pri? |
| 16:25.20 | *** join/#asterisk Strom_M (n=strom@65.42.208.134) |
| 16:25.20 | e-Regg4e | cnbrk: are you able to make calls with spa? |
| 16:25.34 | e-Regg4e | asterisk->pstn? |
| 16:26.33 | e-Regg4e | i have a spa3102 in my house, i receive calls from pstn -> asterisk |
| 16:26.37 | e-Regg4e | and i have a cool IVR |
| 16:26.53 | cnbrk | asterisk to pstn working |
| 16:26.56 | e-Regg4e | when you call to my house, the IVR asks with who you want to talk |
| 16:26.58 | *** join/#asterisk sasch (n=sasch@host102-30-static.107-82-b.business.telecomitalia.it) |
| 16:27.03 | cnbrk | however, i am not going to use it for dialing out purposes |
| 16:27.08 | cnbrk | i need it for dial in |
| 16:27.32 | cnbrk | because i have an agreement with a voip termination company |
| 16:27.43 | cnbrk | and i need the current pstn lines to not to change my phone numbers |
| 16:27.49 | e-Regg4e | cnbrk: can i get access to your spa400? so i can help you |
| 16:27.55 | cnbrk | of course |
| 16:28.07 | cnbrk | just a second |
| 16:28.19 | *** join/#asterisk rhombus (n=rhombus@dsl-cap-66-18-218-36-cgy.nucleus.com) |
| 16:29.08 | rhombus | Anybody here with a Sangoma A200 card? |
| 16:29.43 | MrTelephone | yes |
| 16:29.47 | MrTelephone | i mean no |
| 16:29.52 | rhombus | no? |
| 16:29.53 | rhombus | yes? |
| 16:29.55 | _charly_ | sorry that i ask here, but is there an irc channel or a bugtracker for the misdn kernel driver? |
| 16:29.55 | rhombus | which is it? |
| 16:30.01 | Strom_M | no one has ever bought that card in the history of the universe |
| 16:30.03 | MrTelephone | getting it confused with an a201d |
| 16:30.05 | Strom_M | EVAR |
| 16:30.10 | cnbrk | e-Regg4e, http://85.105.49.124:26000 |
| 16:30.18 | rhombus | WHO HAS ONE |
| 16:30.26 | rhombus | cmoncmocmon |
| 16:30.36 | e-Regg4e | cnbrk: send me de l/p via notice or /msg |
| 16:30.57 | MrTelephone | rhombus whats wrong with the peice of poop? |
| 16:30.59 | rhombus | I just put the card in, and it works, but I hear a strange rushing noise when two Zap channels are bridged |
| 16:31.12 | MrTelephone | rushing? |
| 16:31.32 | rhombus | like static, only smoother... as though one of the parties is in a noisy room |
| 16:31.45 | MrTelephone | like shhhhhhhh |
| 16:31.48 | e-Regg4e | cnbrk: i'll check my configuration, wait a minute |
| 16:31.50 | rhombus | yeah |
| 16:31.58 | MrTelephone | interesting |
| 16:32.10 | rhombus | in fact, it almost sounds like fan noise, but there are no extensions in the riser room |
| 16:32.39 | MrTelephone | fxs or fxo? |
| 16:32.49 | rhombus | well, one is fxs, and the other fxo |
| 16:33.13 | MrTelephone | sorry i can't even make a guess at to what is wrong |
| 16:33.21 | MrTelephone | very strange |
| 16:33.35 | MrTelephone | fxs -> fxo calls are noisy? |
| 16:33.39 | rhombus | yeah |
| 16:33.43 | MrTelephone | did you try a sip -> fxo? |
| 16:33.48 | rhombus | not yet |
| 16:34.02 | MrTelephone | try installing xlite or something and see what happens |
| 16:34.08 | rhombus | haven't set up the SIP phones yet |
| 16:34.18 | MrTelephone | then you'll know if its a zap bridge problem or a fxo problem |
| 16:34.30 | rhombus | I also need to try calling a few more land lines |
| 16:34.45 | MrTelephone | did you try hooking up a regular phone |
| 16:34.57 | MrTelephone | hook up a regular phone, clear the dial tone by pressing a digit |
| 16:35.03 | MrTelephone | and listen to see if there is junk on the line |
| 16:35.16 | MrTelephone | its possible that the fxo card is amplifying noise thats already on th eline |
| 16:35.38 | rhombus | okay, that's good advice |
| 16:35.57 | MrTelephone | just something i would try to narrow it down a little |
| 16:35.57 | Strom_M | rhombus: do you have rxgain set? |
| 16:36.27 | rhombus | Strom_M: to 0.0 |
| 16:36.53 | MrTelephone | did you do yor own cablign rhombus? |
| 16:36.54 | Strom_M | hm, ok... |
| 16:37.03 | MrTelephone | cableling |
| 16:37.03 | rhombus | MrTelephone: yes |
| 16:37.15 | MrTelephone | would reversed ring and tip lines cause that? |
| 16:37.22 | rhombus | I'm wondering about that |
| 16:37.24 | hmepas | channel.c: Unable to find a codec translation path from g723 to ulaw |
| 16:37.26 | Strom_M | no; reversed polarity generally doesnt cause noise |
| 16:37.28 | rhombus | that was my first thought, actually |
| 16:37.40 | hmepas | while i set disallow=all;allow=ulaw for all peerss. |
| 16:37.42 | Strom_M | rhombus: what kind of cable did you use for the voice circuit? |
| 16:37.51 | rhombus | cat 5e |
| 16:38.12 | Strom_M | no kinks, sharp bends, or parallel runs with 120vAC wiring? |
| 16:38.23 | Strom_M | nothing a professional installer would cringe over? |
| 16:38.29 | MrTelephone | lol |
| 16:38.56 | MrTelephone | cringe |
| 16:38.56 | rhombus | no, in fact the wiring was done by a pro, all I did was the cross-connects at the panel |
| 16:39.13 | MrTelephone | i've seen some shadey stuff in the last day or two |
| 16:39.22 | MrTelephone | had to replac cable tv at a customers home |
| 16:39.25 | MrTelephone | some people are pigs |
| 16:39.28 | Strom_M | rhombus: and I'm assuming you know how to properly use a 66 / 110 block and jumpers and so on... |
| 16:39.30 | rhombus | and the lines were used with non-asterisk setup before and apart from the odd crackle, it was clean |
| 16:40.44 | rhombus | Strom_M: well, as far as somebody who has not worked as a lineman, I know how to use 66, 110 and Bix punch tools |
| 16:40.48 | MrTelephone | Cable Guy Joke: Drop -> coax -> 2 way splitter -> rg 59 -> 4 way splitter -> rg 59 -> 3 way splitter -> TV "why is my cable fuzzy?" |
| 16:40.51 | Strom_M | rhombus: i'd go with MrTelephone's advice and try a SIP phone as well as trying an analog phone connected directly to the circuit... |
| 16:41.01 | rhombus | okay |
| 16:41.38 | MrTelephone | yeah im curious about the phone thing |
| 16:42.03 | MrTelephone | have you guys ever seen those orange surge supressors modules that telco companies use? |
| 16:42.04 | rhombus | will try that... (when I tested the line before attaching the Asterisk system, it sounded clean) |
| 16:42.22 | Strom_M | MrTelephone: you mean the fuses on the protector block? |
| 16:42.28 | cnbrk | any ideas |
| 16:42.34 | MrTelephone | yeah just double chekc then use a sip phone.. if the sip phone is clean then its something with your fxs to fxo bridge |
| 16:42.44 | cnbrk | Name/username Host Dyn Nat ACL Port Status |
| 16:42.44 | cnbrk | spa400 192.168.2.170 5060 OK (65 ms) |
| 16:42.50 | cnbrk | but i can't receive calls from spa400 |
| 16:42.54 | MrTelephone | strom_m, im trying to think of the name.. sneak protectors |
| 16:43.24 | MrTelephone | sneak voltage or something? |
| 16:43.33 | Strom_M | ive never heard that term |
| 16:43.43 | MrTelephone | sneak current protection |
| 16:43.46 | MrTelephone | pardon me |
| 16:43.46 | Strom_M | and unfortunately my library of BSPs isn't easily accessible right now |
| 16:43.50 | MrTelephone | i dunno i see them lots out here |
| 16:43.55 | *** join/#asterisk pagilmon (i=paketecu@77.209.29.190) |
| 16:44.11 | pagilmon | hello |
| 16:44.28 | pagilmon | can anybody help a spanish one?P |
| 16:44.28 | Strom_M | cocks |
| 16:44.29 | pagilmon | :) |
| 16:44.35 | pagilmon | sorry |
| 16:44.46 | Strom_M | pagilmon: mi espanol no esta mas bueno, pero esta asi asi |
| 16:44.47 | MrTelephone | google says sneak current protection is the same as using your inline fuses but its for really small applications where there is little current |
| 16:44.57 | pagilmon | hombre hablas muy bien |
| 16:45.04 | e-Regg4e | pagilmon: que paso? |
| 16:45.05 | pagilmon | quieres practicar? |
| 16:45.08 | Strom_M | vamos a #asterisk-es |
| 16:45.15 | MrTelephone | they are cool lookin boxes.. I'll take a picture of one strom and show you soemtime |
| 16:45.22 | pagilmon | perdon, pero soy nuevo en irc y no manejo bien |
| 16:45.24 | pagilmon | como voy? |
| 16:45.33 | pagilmon | si quereis lo cuento aqui... |
| 16:45.34 | MrTelephone | #asterisk-mexican? |
| 16:45.37 | Strom_M | <PROTECTED> |
| 16:45.38 | e-Regg4e | pagilmon: hasta que no preguntes, vas mal :) |
| 16:45.40 | pagilmon | ok |
| 16:45.41 | Strom_M | MrTelephone: uh, no |
| 16:45.50 | pagilmon | me voy a asterisk es... |
| 16:47.58 | MrTelephone | does anyone want to buy a mazda 3 |
| 16:49.10 | e-Regg4e | clear |
| 16:51.05 | *** join/#asterisk mihinomenest (i=SyPD@cerebus.clandestineresearch.com) |
| 16:51.50 | *** join/#asterisk keulin (n=cray@nat30-2-88-160-17-233.fbx.proxad.net) |
| 16:55.47 | *** join/#asterisk justinnnnnn (n=justinn@d58-110-73-172.meb4.vic.optusnet.com.au) |
| 16:56.02 | justinnnnnn | hmm |
| 16:56.17 | justinnnnnn | anyone no if that guy implicit still comes here ? |
| 16:57.11 | Strom_M | shirley eff ewe use homonyms wear they donut make cents, wii will understand ewe. |
| 16:57.14 | pagilmon | MrTelephone: can you help a spanish friend??? |
| 16:57.17 | pipwerk | ~seen implicit |
| 16:57.43 | jbot | implicit <n=implicit@lgb-static-66.18.140.106.mpowercom.net> was last seen on IRC in channel #asterisk, 189d 16h 22m 3s ago, saying: 'debina is a smart girl'. |
| 16:57.43 | justinnnnnn | ~seen implicit |
| 16:57.47 | jbot | implicit <n=implicit@lgb-static-66.18.140.106.mpowercom.net> was last seen on IRC in channel #asterisk, 189d 16h 22m 8s ago, saying: 'debina is a smart girl'. |
| 16:57.47 | justinnnnnn | ... |
| 16:57.49 | pipwerk | be more patient |
| 16:57.55 | justinnnnnn | hehe sorry :P |
| 16:57.58 | justinnnnnn | thanks |
| 16:58.27 | *** join/#asterisk ManxPower (n=manxpowe@dpc67142183150.direcpc.com) |
| 16:58.29 | *** join/#asterisk roro9 (n=nicky_la@129.163-200-80.adsl-dyn.isp.belgacom.be) |
| 16:58.36 | *** part/#asterisk roro9 (n=nicky_la@129.163-200-80.adsl-dyn.isp.belgacom.be) |
| 17:00.51 | *** join/#asterisk tclark (n=TC@S0106000f66c5d294.gv.shawcable.net) |
| 17:02.14 | MrTelephone | pagilmon, i might be able to help |
| 17:02.29 | MrTelephone | i can't speak spanish |
| 17:02.34 | *** part/#asterisk justinnnnnn (n=justinn@d58-110-73-172.meb4.vic.optusnet.com.au) |
| 17:02.45 | MrTelephone | im not racist to skin color but i do beleive everyone should speak english :-/ |
| 17:02.52 | MrTelephone | or one common language |
| 17:03.09 | Strom_M | well, if that's the case, start learning Chinese really quick;y |
| 17:03.19 | Strom_M | it's the most commonly spoken language on the planet |
| 17:03.37 | ManxPower | Everyone should learn Esperanto |
| 17:03.42 | Strom_M | yes! |
| 17:04.02 | MrTelephone | remember the story in the bible where some guys were building a huge tower to reach god.. and they got so close that god said, "because of this im going to make you all speak different languages", and then the towers couldn't be finished because noone could communicate |
| 17:04.08 | Strom_M | or maybe everyone should stop being geocentric racist fuckwad twits |
| 17:04.32 | MrTelephone | chinese has a real bad written language |
| 17:04.39 | *** join/#asterisk var1 (n=james@77-97-208-6.cable.ubr11.edin.blueyonder.co.uk) |
| 17:04.45 | hmepas | k. i have one HW phone, 2 soft phones and one asterisk servers. Soft phones call each other and listen each other. HW phone can call, can recieve calls but cant' talk (didn't send or recieve voice) what to check? =) |
| 17:04.58 | hmepas | tried to force them all to ulaw. |
| 17:05.10 | MrTelephone | we will all learn french so we can sleep with more women |
| 17:07.19 | *** join/#asterisk Superbaloo (n=Superbal@esj.f3.fr) |
| 17:07.54 | hmepas | seems like you have a lot of troubles with asterisk too, MrTel =) |
| 17:10.23 | MrTelephone | lol |
| 17:11.51 | *** part/#asterisk Smith-ub (n=biolacre@81-228-28-154-o1110.telia.com) |
| 17:17.00 | var1 | if only our French teacher had promoted that in school. |
| 17:17.51 | *** join/#asterisk Gpl_Source (n=The_natu@unaffiliated/gplsource) |
| 17:23.14 | *** join/#asterisk BSD_tech (n=BSDTech@adsl-69-230-174-37.dsl.irvnca.pacbell.net) |
| 17:23.19 | BSD_tech | ok morning |
| 17:23.27 | BSD_tech | anyone here working on the gui ? |
| 17:23.35 | BSD_tech | that works for digium ? |
| 17:31.40 | robin_sz | MrTelephone, you siad "chinese has a real bad written language" ... I presume you know noting of chinese then? |
| 17:32.23 | *** join/#asterisk Strom_M (n=strom@65.42.208.134) |
| 17:32.32 | *** join/#asterisk DaveCanoe (n=Dave@137.122.39.10) |
| 17:33.41 | *** join/#asterisk ploieel (n=ploieel@Fb307.f.ppp-pool.de) |
| 17:33.51 | *** join/#asterisk Op3r (n=op3r@121.97.211.188) |
| 17:34.18 | robin_sz | MrTelephone, if you knew even the most basic facts about China, you would know its written language is one of its big plus points ... althought there are many spoken dialects, many so different they cannot be understood by others, there is one common fact, the written language, that means that chinese people from different areas who cannot understand what the other is saying, can at least understand what they write |
| 17:35.02 | robin_sz | MrTelephone, its that fact that is going to enable China to expand more rapidly than it would otherwise have done. |
| 17:36.04 | Op3r | is chanspy stable now?" |
| 17:36.51 | dudes | in 1.4.2/3 we managed crashes cause of it |
| 17:37.19 | dudes | but in 1.2.13/17 (somewhere in there) has a pretty stable version |
| 17:37.59 | rhombus | I have this pattern in a dial command: _NXXXXXX |
| 17:38.11 | rhombus | the Zap channel won't take a number starting with 2 |
| 17:38.17 | rhombus | what's wrong here? |
| 17:38.24 | dudes | http://www.voip-info.org/wiki/view/Asterisk+Dialplan+Patterns will be helpful |
| 17:39.01 | rhombus | according to that, a 2 should be no problem |
| 17:39.09 | rhombus | 255 2525 should work |
| 17:39.13 | rhombus | but it doesn't |
| 17:39.19 | dudes | try a _X and see what happens |
| 17:39.24 | dudes | what version, BTW? |
| 17:39.28 | rhombus | 1.2.17 |
| 17:39.46 | dudes | should work, but you'll find with Asterisk, things don't always work as they should |
| 17:40.23 | rhombus | hmn... that's not a very satisfying answer -- it has worked in other installations... |
| 17:40.29 | rhombus | bizarre. |
| 17:40.41 | dudes | safisfying and working are two different things |
| 17:40.48 | *** join/#asterisk cr4z3d (n=cr4z3d@ip70-162-119-225.ph.ph.cox.net) |
| 17:40.53 | dudes | I'm not suggesting you "use" X I"m saying test it to see if it works |
| 17:40.55 | carrar | How are you sure some other pattern is not grabbing it? |
| 17:41.06 | dudes | that too |
| 17:41.11 | dudes | like _X., or something |
| 17:41.20 | rhombus | okay, here's some weirdness |
| 17:41.28 | carrar | or some pattern in another context that is included in the current |
| 17:41.40 | rhombus | if I dial a number, say 508 2985, this is what Asterisk shows: |
| 17:41.56 | Strom_M | rhombus: _NXXXXXX will only match a seven digit number starting with 2-9 |
| 17:42.10 | rhombus | so why can I not dial a number starting with 2? |
| 17:42.20 | cr4z3d | cuz that's what N means |
| 17:42.20 | Strom_M | pastebin your whole dialplan |
| 17:42.22 | carrar | past the entire context in pastbin |
| 17:42.26 | carrar | heh |
| 17:42.54 | rhombus | cr4z3d: you just contradicted everything everyone else is saying |
| 17:44.02 | rhombus | http://pastebin.ca/497155 |
| 17:44.18 | Strom_M | rhombus: and what context is the phone set to? |
| 17:44.22 | cr4z3d | rhombus, http://www.voip-info.org/wiki/index.php?page=Asterisk+Dialplan+Patterns N matches any digit 2-9 that's why it has to start with 2-9 |
| 17:44.41 | rhombus | cr4z3d: thank you for actually reading my message. |
| 17:44.55 | rhombus | cr4z3d: I CANNOT DIAL numbers starting with 2, that is the problem |
| 17:45.08 | cr4z3d | oh |
| 17:45.10 | Strom_M | rhombus: i'm trying to help you. will you answer my questions? |
| 17:45.15 | rhombus | Strom_M: sure |
| 17:45.17 | rhombus | one second |
| 17:45.19 | carrar | Are there any calls on channel 1 of your ZAP? |
| 17:45.22 | Strom_M | cocks |
| 17:45.54 | rhombus | The phone is in context from-internal |
| 17:45.59 | Strom_M | ok |
| 17:46.02 | rhombus | it's a Zap channel |
| 17:46.09 | Strom_M | what error message do you get? |
| 17:46.22 | rhombus | I think it's not picking up the DTMF properly... maybe I have my polarity reversed |
| 17:46.32 | Strom_M | uh |
| 17:46.32 | rhombus | the set is an old one |
| 17:46.51 | rhombus | well, when I dial a number starting with 5, it records two 5s |
| 17:46.55 | Strom_M | if it's a polarity sensitive set, you won't here any DTMF at all. |
| 17:47.04 | rhombus | Strom_M: right |
| 17:47.07 | Strom_M | er |
| 17:47.08 | Strom_M | hear |
| 17:47.16 | Strom_M | jesus, what the hell is wrong with me |
| 17:47.31 | carrar | I'll paste bin that |
| 17:47.32 | mvanbaak | Strom_M: better question: what isn't wrong with me |
| 17:47.33 | rhombus | I know what's wrong -- you are a human being |
| 17:47.40 | Strom_M | rhombus: pastebin your zapata.conf |
| 17:47.40 | carrar | too much to paste |
| 17:47.41 | mvanbaak | list will be shorter ;) |
| 17:47.45 | rhombus | okay |
| 17:48.20 | carrar | rhombus, whats the error you get? |
| 17:48.29 | carrar | Also try another phone that the keys are not stuck |
| 17:49.35 | carrar | test with a softphone |
| 17:49.35 | carrar | xlite works well for that |
| 17:49.35 | rhombus | http://pastebin.ca/497163 |
| 17:49.37 | rhombus | carrar: hang on, I'll show you |
| 17:50.16 | carrar | try sending out Zap/g1 |
| 17:50.20 | carrar | er |
| 17:50.21 | *** part/#asterisk mog (n=mog@c-71-207-214-192.hsd1.al.comcast.net) |
| 17:50.22 | carrar | try sending out Zap/g0 |
| 17:50.38 | rhombus | before I do that, get this noise: |
| 17:51.25 | Strom_M | rhombus: what error message does asterisk give you? |
| 17:51.26 | rhombus | if I dial 288 2128, nothing happens... until I dial an additional digit |
| 17:51.26 | carrar | ERROR PLEASE |
| 17:51.26 | Strom_M | rhombus: what error message does asterisk give you? |
| 17:51.26 | rhombus | with verbosity at 4 there is no error at all |
| 17:51.35 | carrar | then whats the problem |
| 17:51.42 | rhombus | the call is hungup |
| 17:51.43 | Strom_M | cocks. |
| 17:51.51 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 17:52.04 | carrar | use a softphone to rule out a bad phone |
| 17:52.14 | rhombus | the phone works fine when it's on a straight analog line |
| 17:52.33 | Strom_M | use a softphone to rule out a piece of shit interface card ;) |
| 17:52.37 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 17:52.39 | carrar | hahah |
| 17:53.06 | rhombus | so, do you guys want to hear the behavior, or do you have more instructions for me? |
| 17:53.17 | Strom_M | use a softphone |
| 17:53.20 | Strom_M | use a softphone |
| 17:53.24 | Strom_M | and also use a softphone |
| 17:53.45 | rhombus | and what will that tell me? |
| 17:54.03 | Strom_M | whether or not the problem is with the fxs port |
| 17:54.08 | rhombus | I don't think this is a port problem. |
| 17:54.15 | rhombus | Because of the behavior |
| 17:54.19 | carrar | ok then, then you solved it |
| 17:54.22 | Strom_M | well how about TRYING IT instead of just making up theories? |
| 17:54.37 | Strom_M | no, wait, that would be too easy |
| 17:55.21 | rhombus | because it is too easy to blame hardware when 90% of problems are caused by configuration |
| 17:55.33 | Strom_M | well your configuration looks right to me |
| 17:55.46 | rhombus | it's like when you troubleshoot a car -- you don't waste your time looking for fuel system problems until you have exhausted all the ignition system causes |
| 17:56.04 | Strom_M | so are you going to listen to me, or are you just here because you enjoy complaining/ |
| 17:56.37 | rhombus | Actually, I had some more information to share, but it seems you're the one not interested in hearing it, irrespective of whether or not it is relevant |
| 17:56.42 | carrar | rhombus, you phone should be plugged into a FXS port, but your _NXXXXXX dial plan is not in a FXS dialplan |
| 17:56.47 | var1 | use a softphone like me. |
| 17:56.55 | carrar | it's in a FXO |
| 17:56.58 | *** join/#asterisk znoG (n=gs@235-180-235-201.fibertel.com.ar) |
| 17:57.26 | Strom_M | Fieldy: i have nothing better to do while i'm waiting to change planes |
| 17:57.46 | rhombus | Okay, I am going to say my bit, then I will try a softphone |
| 17:57.56 | carrar | rhombus, what port is your Analog phone (the one you are dialing from) pluged into? |
| 17:58.00 | rhombus | When I dial a number starting with 5, the system logs two 5s |
| 17:58.03 | rhombus | it's on Zap/8 |
| 17:58.12 | carrar | FXO port? |
| 17:58.27 | Strom_M | carrar: it's an fxs port; you configure those with fxo signalling |
| 17:58.34 | carrar | oh |
| 17:58.35 | rhombus | FXS port -- I wouldn't be getting a dial tone if it were on an FXO |
| 17:58.54 | rhombus | when I dial a number starting with 2 but followed by a 0, I get a fast busy |
| 17:58.57 | carrar | ok nm |
| 17:59.03 | carrar | go back to the softphone idea |
| 17:59.05 | carrar | heh |
| 17:59.35 | *** join/#asterisk adorah (n=Michael@87.69.57.73.cable.012.net.il) |
| 17:59.40 | rhombus | if I dial a number starting with 2, the call doesn't patch until I have dialed 8 digits, after which the call is bridged on Zap/1 using the last 7 digits |
| 17:59.53 | Strom_M | rhombus: like I said earlier, it sounds like digit interpretation issues with the FXS card. I would suggest you try a sip phone to see if those issues go away. |
| 18:00.48 | rhombus | Strom_M: I agree with you -- what I need to know is if there are some card settings that need to change to make it behave differently |
| 18:00.56 | carrar | so put a patter like this in there exten => 2,1,Dial(Zap/1/2882128) as a test |
| 18:01.06 | Strom_M | carrar: shut up. |
| 18:01.13 | rhombus | Strom_M: thank you |
| 18:01.46 | Strom_M | rhombus: i'd make sure that my theory is reasonably accurate before working out a solution |
| 18:03.29 | rhombus | okay -- softphone, then |
| 18:05.52 | mistermocha | morning all |
| 18:05.54 | rhombus | i have to restart; be back in a moment |
| 18:05.56 | *** join/#asterisk DarKnesS_WolF (n=wolf@unaffiliated/sherif) |
| 18:05.57 | *** part/#asterisk rhombus (n=rhombus@dsl-cap-66-18-218-36-cgy.nucleus.com) |
| 18:05.57 | mistermocha | I'm having a wacky time trying to find UK dialplans... any suggestions? |
| 18:06.19 | Strom_M | mistermocha: um, what do you mean "UK dialplans"? |
| 18:06.26 | Strom_M | go to ofcom and look it up yourself :) |
| 18:06.44 | mistermocha | ofcom - you've given me a place to look... thanks! |
| 18:09.41 | *** join/#asterisk pagilmon (i=paketecu@77.209.13.93) |
| 18:10.08 | pagilmon | any experienced guy configuring extensions.conf or zapata.conf?? |
| 18:10.27 | *** join/#asterisk rhombus (n=rhombus@dsl-cap-66-18-218-36-cgy.nucleus.com) |
| 18:15.50 | *** join/#asterisk Smith-ub (n=biolacre@81-228-28-154-o1110.telia.com) |
| 18:19.12 | rhombus | okay -- it works fine with Xlite |
| 18:19.30 | Strom_M | alright |
| 18:19.38 | Strom_M | try relaxdtmf=true in zapata.conf |
| 18:20.07 | rhombus | Strom_M: when I take calls on this channel, there is a slight shhhhhhh sound |
| 18:20.16 | rhombus | I wonder if that is messing up the DTMF detection |
| 18:20.18 | Strom_M | maybe youve got a bad card |
| 18:20.25 | rhombus | :( |
| 18:20.30 | rhombus | sigh |
| 18:20.44 | rhombus | okay, let me try this with relaxdtmf=true |
| 18:21.50 | rhombus | to apply those changes I only need to restart asterisk, right? |
| 18:22.42 | rhombus | okay, it works with relaxdtmf=true |
| 18:22.54 | Strom_M | technically, you only need to reload the zapata channel driver, but I don't know in what special ways the sangoma driver fucks everything up |
| 18:23.41 | rhombus | this FXS module might still be defective... first, there's the rushing sound, second, the port LED on the second of the two FXS ports is dim |
| 18:27.28 | rhombus | argh... I just reset wanrouter and now I don't get a dial tone anymore... maybe this card is humped after all |
| 18:27.31 | rhombus | ARGH |
| 18:28.21 | *** join/#asterisk crochat (n=crochat@84-74-150-141.dclient.hispeed.ch) |
| 18:28.25 | pagilmon | can i have help, please? |
| 18:28.34 | Corydon76-home | ~ask |
| 18:28.36 | jbot | Questions in the channel should be specific, informative, complete, concise, and on-topic. Don't ask if you can ask a question first. Don't ask if a person is there, just ask what you intended to ask them. Better questions more frequently yield better answers. We are all here voluntarily or against our will. |
| 18:28.36 | rhombus | no |
| 18:28.51 | *** join/#asterisk rmayorga (n=rmayorga@unaffiliated/rmayorga) |
| 18:29.49 | pagilmon | sorry |
| 18:30.00 | rhombus | just kidding. |
| 18:30.04 | rhombus | just ask your question. |
| 18:30.15 | pagilmon | right |
| 18:30.48 | pagilmon | i want to configure extensions.conf and zapata.conf to use the first avalaible channel in a tdm04 with 4 fxo ports... |
| 18:30.52 | pagilmon | is it possible? |
| 18:31.02 | Corydon76-home | Your question must also rhyme and be in iambic pentameter |
| 18:31.04 | pagilmon | i am using 4 gsm gateways |
| 18:31.13 | pagilmon | \::=)) |
| 18:31.31 | pagilmon | Corydon76-home: very good |
| 18:31.35 | Corydon76-home | See the group definition in zapata.conf and use Dial(Zap/g1/${EXTEN}) |
| 18:31.49 | Corydon76-home | the group number is 1 or whatever you define it to be |
| 18:32.04 | pagilmon | umhh |
| 18:32.25 | pagilmon | my main problem is i am using asterisknow, and it uses some kind of macros.. |
| 18:32.31 | pagilmon | so i am a bit loosed!! |
| 18:32.37 | Corydon76-home | Then you want #asterisknow |
| 18:32.40 | pagilmon | i am a newbbie in linux and astisk |
| 18:32.45 | Corydon76-home | or #asterisk-gui |
| 18:32.47 | pagilmon | yes i am asking |
| 18:32.52 | pagilmon | but no answer |
| 18:33.00 | pagilmon | i don't know where the people is |
| 18:33.02 | Corydon76-home | Well, those are the people who know |
| 18:33.09 | pagilmon | thanks, anyway |
| 18:33.42 | Corydon76-home | Considering it's the weekend, they've probably off with their families |
| 18:33.48 | pagilmon | jajajaaj |
| 18:33.51 | pagilmon | yes maybe |
| 18:34.30 | pagilmon | ok, thanks i will follow your instructions and will do the same: playing with my son!!! thanks to remember |
| 18:37.41 | *** join/#asterisk DrRighteous (n=DrRighte@ool-44c7ad06.dyn.optonline.net) |
| 18:41.30 | *** join/#asterisk r0d3nt (i=nobody@foster.stonedcoder.org) |
| 18:43.50 | var1 | I have kphone working now as a replacement for xlite. |
| 18:46.27 | var1 | but it's not registering here is the output debug messages from kphone. http://pastebin.ca/497260 |
| 18:47.03 | var1 | my server is on 192.168.1.4 and my client is 192.168.1.7 |
| 18:47.27 | ManxPower | The local radio station got a new call sign. WDRM |
| 18:49.19 | *** join/#asterisk cnbrk (n=canberk@212.156.213.131) |
| 18:49.29 | cnbrk | hi |
| 18:49.38 | cnbrk | who has an experience with spa400 and asterisk? |
| 18:51.22 | *** join/#asterisk ToyMan (n=Stuart@user-12lcqol.cable.mindspring.com) |
| 18:52.45 | *** join/#asterisk eltech (n=eltech@ool-457c9ece.dyn.optonline.net) |
| 18:52.59 | var1 | someone mentioned qualify=yes is important, sorry I read it as quality |
| 18:53.14 | sbingner | lol |
| 18:54.07 | sbingner | I don't see qualify=yes would matter unless you're already registering fine and don't want it to just ring when the phone isn't there |
| 18:57.08 | *** join/#asterisk zotz (n=zotz@24.244.163.157) |
| 19:01.47 | *** join/#asterisk eltech (n=eltech@ool-457c9ece.dyn.optonline.net) |
| 19:02.24 | *** join/#asterisk wunderkin (i=wunderki@ip68-108-204-139.ph.ph.cox.net) |
| 19:12.52 | cnbrk | ok friends, my asterisk is not registering with my spa400 |
| 19:12.57 | cnbrk | it is not sending the register command |
| 19:12.59 | cnbrk | why do you think is this |
| 19:15.22 | *** join/#asterisk zogulus (n=zogulus@58.98.adsl.brightview.com) |
| 19:17.24 | var1 | thanks to whoever sent this link http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 previously, I'm reading the book now. |
| 19:18.30 | *** part/#asterisk DrRighteous (n=DrRighte@ool-44c7ad06.dyn.optonline.net) |
| 19:22.32 | *** join/#asterisk Kamileon (n=Kamileon@c-67-177-121-110.hsd1.in.comcast.net) |
| 19:24.54 | *** join/#asterisk vgster (n=vgster@cpc2-ledn1-0-0-cust826.leed.cable.ntl.com) |
| 19:25.45 | *** join/#asterisk IOscanner (n=IOscanne@cpe-76-187-194-128.tx.res.rr.com) |
| 19:25.53 | *** join/#asterisk jmacz (n=jmacz@190.25.35.171) |
| 19:26.06 | *** part/#asterisk pagilmon (i=paketecu@77.209.13.93) |
| 19:26.26 | IOscanner | I am having a problem with database get. I have an entry in the database /SIP/Registry/71453 |
| 19:26.54 | IOscanner | I used to verify it with database get /Registry 71453 |
| 19:27.04 | IOscanner | That doesn't work any more. |
| 19:27.22 | IOscanner | I thought database get /SIP/Registry 71453 would work, but it doesn't |
| 19:27.26 | IOscanner | any ideas? |
| 19:29.12 | cnbrk | ok friends, my asterisk is not registering with my spa400 |
| 19:29.13 | cnbrk | why do you think is this |
| 19:29.16 | cnbrk | it is not sending the register command |
| 19:29.26 | bkruse | ~thebook |
| 19:29.38 | jbot | i guess thebook is a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 19:29.38 | bkruse | ~book |
| 19:29.44 | jbot | i guess book is a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 19:29.44 | bkruse | what is called dangit |
| 19:29.44 | bkruse | there we go |
| 19:31.08 | *** join/#asterisk ploieel (n=ploieel@Fb371.f.ppp-pool.de) |
| 19:36.25 | IOscanner | found it |
| 19:37.11 | *** join/#asterisk crochat (n=crochat@84-74-150-141.dclient.hispeed.ch) |
| 19:39.39 | Corydon76-home | cnbrk: shouldn't that be the other way around? |
| 19:39.55 | Corydon76-home | generally a SIP phone registers to Asterisk, not the other way around |
| 19:45.42 | *** join/#asterisk aptura (n=jondoe@S010600a0c93f6f7e.vs.shawcable.net) |
| 19:50.40 | cnbrk | spa400 is not a sip phone |
| 19:50.44 | cnbrk | it is a fxo gateway |
| 19:50.49 | cnbrk | and asterisk has to register with it |
| 19:51.00 | cnbrk | not the usual way |
| 19:51.33 | Corydon76-home | Oh, do you have a register line in the [general] section of sip.conf? |
| 19:51.37 | cnbrk | yes i do |
| 19:51.46 | cnbrk | however when i type sip show registry |
| 19:51.48 | cnbrk | nothing shows |
| 19:52.22 | cnbrk | it's like asterisk is ignoring the register=> line |
| 19:53.52 | Corydon76-home | cnbrk: any errors on reload? |
| 19:54.31 | cnbrk | yeah i can put the whole debug to pastebin |
| 19:54.34 | cnbrk | just a second |
| 19:54.47 | *** join/#asterisk nuonguy (n=john@c-24-6-175-26.hsd1.ca.comcast.net) |
| 20:02.45 | *** join/#asterisk Gunnar (n=gunnar@nat.sigmasoft.com) |
| 20:03.47 | Corydon76-home | cnbrk: ??? |
| 20:07.08 | cnbrk | http://pastebin.ca/497344 |
| 20:07.15 | cnbrk | http://pastebin.ca/497344 |
| 20:07.23 | cnbrk | this is the debug of the startup |
| 20:11.18 | *** join/#asterisk arafath (n=shaon@202.56.7.163) |
| 20:11.27 | cnbrk | Corydon76-home any ideas |
| 20:11.52 | *** join/#asterisk techie (n=gus@voip.routedsystems.com) |
| 20:13.02 | arafath | do any one have idea about libss7+TE407+ 1 signal channel for 4 E1 |
| 20:13.16 | Corydon76-home | cnbrk: can you copy+paste your register line? |
| 20:13.23 | cnbrk | sure one sec |
| 20:13.26 | *** join/#asterisk _VoiceMeUp_COM (n=_VoiceMe@145-27.mc.cite.net) |
| 20:13.59 | arafath | i m not familer with register line |
| 20:14.47 | cnbrk | i'll really order something to the person who solves my problem :) |
| 20:14.49 | arafath | cnbrk> can u help me? |
| 20:15.04 | Corydon76-home | arafath: libss7 is not supported in any release of Asterisk. It is still considered experimental |
| 20:15.09 | cnbrk | arafath sorry i don't even have a clue what you are talking about :( |
| 20:16.02 | arafath | is there any alternative idea? |
| 20:16.26 | Corydon76-home | Alternative to what? |
| 20:16.48 | arafath | altenative of libss7 |
| 20:17.17 | Corydon76-home | Well, there's libpri, but you need a PRI link, not an SS7 link, to use it |
| 20:17.19 | arafath | asterisk-ss7 |
| 20:17.40 | arafath | what i will do if i want to use ss7 link? |
| 20:17.54 | Corydon76-home | Then you become an expert |
| 20:18.34 | Corydon76-home | The vast majority of the Asterisk experts here have never had an opportunity to get an SS7 link, so they will be unable to help you, either |
| 20:18.44 | cnbrk | so Corydon76-home, it is register =>spa400@192.168.2.170/spa400 |
| 20:20.03 | Corydon76-home | So your username is "spa400" in the spa400? |
| 20:20.14 | arafath | having trouble that my telco is giving me 4 E1 connection but only one signal channel, and my 1st e1 work fine, but 2 to 4th e1 got only ringing tone but no voice from both side |
| 20:20.27 | cnbrk | yes |
| 20:20.51 | Corydon76-home | cnbrk: are you absolutely sure that the register line is in the [general] section? |
| 20:21.06 | Corydon76-home | cnbrk: because it won't work if it's defined in a peer or user section |
| 20:21.31 | cnbrk | let me check |
| 20:22.51 | arafath | Corydon: my telco is giving me 4 E1 connection but only one signal channel, and my 1st e1 work fine, but 2 to 4th e1 got only ringing tone but no voice from both side |
| 20:23.20 | Corydon76-home | arafath: I CANNOT HELP YOU |
| 20:23.39 | arafath | ok thanx Corydone |
| 20:23.56 | arafath | is there anyone who can help me? |
| 20:24.29 | cnbrk | Corydon76-home |
| 20:24.39 | cnbrk | it was at the very bottom of the sip.conf |
| 20:25.18 | cnbrk | i'm trying now, i put the line right after the general section |
| 20:26.39 | cnbrk | hey Corydon76-home, u r really the best!! |
| 20:26.43 | cnbrk | now it tried to register |
| 20:26.50 | cnbrk | however it failed |
| 20:27.00 | Corydon76-home | Probably because of lack of password |
| 20:27.36 | cnbrk | *CLI> REGISTER attempt 1 to spa400@192.168.2.170 |
| 20:27.37 | cnbrk | May 19 23:25:55 WARNING[2319]: chan_sip.c:9957 handle_response_register: Forbidden - wrong password on authentication for REGISTER for 'spa400' to '192.168.2.170' |
| 20:27.44 | Corydon76-home | Bingo |
| 20:27.54 | cnbrk | yes but it has no password though :( |
| 20:28.10 | Corydon76-home | arafath: please go to http://pastebin.ca and paste the contents of your zapata.conf |
| 20:28.36 | arafath | thanx Corydon |
| 20:29.32 | cnbrk | Corydon76-home, there is no password |
| 20:29.44 | Corydon76-home | cnbrk: the remote end thinks you do |
| 20:29.55 | cnbrk | yes but it's strange... |
| 20:30.49 | cnbrk | how can i make the register line send the password too |
| 20:31.00 | cnbrk | register=spa400:password |
| 20:31.03 | cnbrk | @blabla? |
| 20:32.24 | Corydon76-home | cnbrk: correct |
| 20:33.18 | tzafrir_laptop | arafath, look at the trunkgroups section in sample zapata.conf for starters. |
| 20:33.35 | cnbrk | register=user:password@ip/what is this?? at the end |
| 20:33.39 | Corydon76-home | tzafrir_laptop: ss7 uses trunkgroups? |
| 20:34.18 | tzafrir_laptop | don't knowabout ss7. But this is how you get one signlling channel for all trunks |
| 20:34.24 | Corydon76-home | cnbrk: it's the extension at which it registers |
| 20:34.48 | Corydon76-home | as in, dialplan extension |
| 20:35.06 | Corydon76-home | Generally only used for Asterisk registrations |
| 20:35.08 | tzafrir_laptop | as for ss7: nothing I can help there |
| 20:35.43 | Corydon76-home | tzafrir_laptop: that's why I asked him to pastebin his zapata.conf. From Matt's comments, it looks like the signalling trunk is set differently for SS7 |
| 20:36.09 | Corydon76-home | but with no SS7 trunk here, it's kind of difficult to be sure |
| 20:38.49 | cnbrk | it's really strange |
| 20:38.54 | Corydon76-home | and if I don't get an answer soon WRT the pastebin, I'm afraid he's going to be out my guesswork, too. |
| 20:38.59 | cnbrk | google says there is no password |
| 20:39.18 | arafath | i pasted that |
| 20:39.29 | Corydon76-home | arafath: to what link? |
| 20:39.29 | *** join/#asterisk HockeyInJune (i=HockeyIn@pool-68-161-185-98.ny325.east.verizon.net) |
| 20:42.26 | Corydon76-home | I'm off. |
| 20:43.07 | *** part/#asterisk HockeyInJune (i=HockeyIn@pool-68-161-185-98.ny325.east.verizon.net) |
| 20:44.59 | arafath | corydon: http://pastebin.ca/497398 |
| 20:50.12 | *** join/#asterisk BS2 (n=BS@2001:960:68f:201:211:d8ff:fe9a:6b0b) |
| 20:52.09 | tzafrir_laptop | arafath, you don't even have [channels] in your /etc/asterisk/zapata.conf |
| 20:52.13 | *** join/#asterisk Math` (n=seb@modemcable037.229-56-74.mc.videotron.ca) |
| 20:52.45 | tzafrir_laptop | you should put that line at the begining of the section that describes the channels |
| 20:52.59 | tzafrir_laptop | See the sample configuration file |
| 20:53.41 | tzafrir_laptop | that said, I still have no idea what should be done for ss7 |
| 20:53.49 | arafath | pls wait let me see |
| 20:54.07 | *** part/#asterisk rhombus (n=rhombus@dsl-cap-66-18-218-36-cgy.nucleus.com) |
| 20:59.43 | *** join/#asterisk anonymouz666 (n=anonymou@200.218.193.6) |
| 21:00.38 | *** join/#asterisk [TK]D-Fender (n=Joe@64.235.216.2) |
| 21:01.50 | arafath | tzafrir: what is the purpose of [channels] |
| 21:02.11 | *** join/#asterisk alexpe (n=alex@cev75-1-81-57-14-91.fbx.proxad.net) |
| 21:17.21 | arafath | !ping |
| 21:17.48 | *** join/#asterisk mrichmanM (n=richmanm@68.178.77.110) |
| 21:23.37 | robin_sz | arafath, channels are where you come to bug people about stuff |
| 21:24.59 | arafath | Corydon: my telco is giving me 4 E1 connection but only one signal channel, and my 1st e1 work fine, but 2 to 4th e1 got only ringing tone but no voice from both side |
| 21:25.24 | arafath | robin: my telco is giving me 4 E1 connection but only one signal channel, and my 1st e1 work fine, but 2 to 4th e1 got only ringing tone but no voice from both side |
| 21:25.42 | arafath | can u help me on that issue? |
| 21:29.54 | ManxPower | arafath: Perhaps they are using NFAS? |
| 21:30.17 | ManxPower | If you have "signal" channels, then you may have a PRI |
| 21:31.44 | *** join/#asterisk Zorix (n=Brandon@c-76-101-72-47.hsd1.fl.comcast.net) |
| 21:32.39 | arafath | but E1 also have signal channel |
| 21:32.57 | Zorix | i know this is probably not an easy task but what do i need to do to get modem support working.. i am using the 711 codec which is uncompressed 64 kbit and am am trying to link two machines over a sipura analog adapter box and the handshake always fails |
| 21:33.03 | ManxPower | arafath: No, they do not. |
| 21:33.12 | ManxPower | A channelized E-1 does not have a signalling channel |
| 21:33.53 | ManxPower | A PRI over E-1 does have a signalling channel (D-Channel), on an E-1 it would be channel 16 |
| 21:33.56 | arafath | can u pls see my conf file http://pastebin.ca/497398 |
| 21:34.32 | ManxPower | Sorry, I cannot help you with SS7 |
| 21:34.33 | arafath | yes i have D-channel and which is 16 |
| 21:35.14 | *** part/#asterisk BS2 (n=BS@2001:960:68f:201:211:d8ff:fe9a:6b0b) |
| 21:35.16 | arafath | thanx Manx |
| 21:35.16 | ManxPower | It sounds like you are using NFAS. See the sample config files for NFAS. I have never used NFAS |
| 21:35.38 | arafath | me too |
| 21:35.48 | arafath | where i can find the sample? |
| 21:37.16 | ManxPower | <PROTECTED> |
| 21:37.35 | arafath | oh ok |
| 21:38.30 | arafath | thanx |
| 21:38.40 | *** join/#asterisk blitzrage (n=blitzrag@asterisk/documenteur-extraordinaire/blitzrage) |
| 21:38.52 | ManxPower | Zorix: I doubt you will ever be able to make that work |
| 21:39.18 | blitzrage | making things work is for suckers |
| 21:39.57 | Zorix | ManxPower the adapter says it has fax/modem support why wouldnt it work |
| 21:40.03 | Zorix | its on a local lan not over the internet |
| 21:41.23 | ManxPower | The adapter says it has modem support or fax support? |
| 21:41.40 | ManxPower | Zorix: lock the carrier speeds to 9600, what is what faxes use. |
| 21:41.58 | ManxPower | not the PORT speed, the CARRIER speed. |
| 21:42.12 | Zorix | yes it says that.. but it should negotiate it to 9600 if its all thats supported |
| 21:42.46 | anonymouz666 | winter...sucks! |
| 21:46.25 | cnbrk | uff |
| 21:46.29 | cnbrk | spa400 is asking for password |
| 21:50.46 | robin_sz | quelle suprise |
| 21:51.03 | *** join/#asterisk RavuAlHemio (n=Ondra@M1335P004.adsl.highway.telekom.at) |
| 21:51.08 | robin_sz | ManxPower, faxes use 14,400 |
| 21:51.19 | RavuAlHemio | Hey people... |
| 21:51.25 | robin_sz | hey dude |
| 21:51.55 | RavuAlHemio | Does anybody know if the Digium TDM400P plays by the European standard EN 300 001? |
| 21:52.42 | RavuAlHemio | The Austrian POTS doesn't need any actual certification but everything you connect to the PSTN has to abide by that norm... |
| 21:54.02 | cnbrk | allrite guys |
| 21:54.07 | cnbrk | whenever i try to register with spa400 |
| 21:54.08 | cnbrk | i get |
| 21:54.14 | cnbrk | May 19 23:25:55 WARNING[2319]: chan_sip.c:9957 handle_response_register: Forbidden - wrong password on authentication for REGISTER for 'spa400' to '192.168.2.170' |
| 21:54.39 | cnbrk | as i know spa400 does not require a password |
| 21:54.45 | cnbrk | so what else can the problem be? |
| 21:55.54 | Zorix | ManxPower, i got it, had to turn off echo cancellation |
| 21:56.08 | *** join/#asterisk aptura (n=jondoe@S010600a0c93f6f7e.vs.shawcable.net) |
| 21:56.41 | explidous | There are a number of settings that make fax/Modem somewhat possible.... |
| 21:56.51 | *** join/#asterisk docelmo (n=vircuser@c-76-99-157-112.hsd1.de.comcast.net) |
| 21:57.28 | robin_sz | cnbrk, looks like your spa400 needs a password |
| 21:57.54 | Zorix | explidous i disabled a lot of the settings in there one by one to make it work and the last one was echo cancellation that made it work |
| 21:59.00 | *** join/#asterisk RavuAlHemio (n=Ondra@M1432P022.adsl.highway.telekom.at) |
| 21:59.08 | explidous | Zorix: Jitter buffer is one of the other ones and it usually helps to hard set carrier speed on both sides... sometimes the delay is too long to make the handshaking work.. |
| 21:59.10 | RavuAlHemio | Sorry, bolted... :( |
| 21:59.23 | cnbrk | robin_sz, will it only be caused by a wrong passwd? |
| 21:59.26 | cnbrk | nothing else? |
| 22:00.32 | robin_sz | or wrong username |
| 22:02.08 | Zorix | explidous yea i turned off jitter first |
| 22:02.30 | Zorix | it worked at the modem's connect speed of 28800 i dont have any others to test with |
| 22:02.47 | *** join/#asterisk stimpie (n=michiel@ip565faf27.direct-adsl.nl) |
| 22:03.12 | RavuAlHemio | Can anybody here answer my question? I might have missed it or it might not have gone through since I got bolted shortly after. |
| 22:03.39 | Zorix | sorry dont have any experience with that hardware |
| 22:04.23 | *** join/#asterisk osiris (n=osiris@c-71-205-27-131.hsd1.mi.comcast.net) |
| 22:07.16 | RavuAlHemio | 'Kay, I might end up contacting Digium or somebody... |
| 22:09.22 | Maxxed | damn im having probs unlocking a 7940 |
| 22:09.33 | Maxxed | im trying to set the addr for the tftp server |
| 22:09.40 | Maxxed | and the **# isnt doing anything |
| 22:10.08 | Maxxed | any recomendations |
| 22:16.28 | *** join/#asterisk marc\cba (n=marc@cpc2-whit2-0-0-cust200.cdif.cable.ntl.com) |
| 22:17.30 | var1 | does anybody know the trixbox (vmware image) username and password? |
| 22:19.45 | Maxxed | oh nevermind, i forget the new firmware its with the unlock config thinger and the defualt passwd is cisco :p |
| 22:28.21 | *** join/#asterisk N9URK_lithium (n=leonard@cpe-065-184-160-244.ec.res.rr.com) |
| 22:29.35 | N9URK_lithium | HI All, what kind of routers/swithes do you all use for your * installation. We are going to have 5-10 devices and are trying to figure out what is the best way to go for performace for that number of devices. tnx |
| 22:31.25 | *** join/#asterisk Cybertoy (n=cybertoy@dsl254-123-112.nyc1.dsl.speakeasy.net) |
| 22:32.26 | Cybertoy | hi there... I have a transfer problem... can someone help me? I'm trying to transfer an incoming call to another internal extension .. but asterisk can't transfer it as the extension I'm transferring to is not in the default context.. the phone I'm transfering from though can dial into that context... |
| 22:32.34 | Cybertoy | is there a seperate transfer context I have to define somewhere? |
| 22:32.50 | Sweeper | no |
| 22:32.58 | N9URK_lithium | do you have t in your exten() line/ |
| 22:32.58 | Sweeper | your contexts just have to make sense :P |
| 22:33.07 | Cybertoy | heh |
| 22:33.23 | Cybertoy | I do have t ... |
| 22:33.43 | N9URK_lithium | hmmm, not too sure then |
| 22:33.45 | N9URK_lithium | sorry |
| 22:33.46 | explidous | Zorix: basically all Modems support the slower speeds you just need the right AT command to set them... what did it actually connect at? |
| 22:33.49 | Cybertoy | actually let me clarify.. I'm using the sip device to transfer... |
| 22:34.02 | Cybertoy | so either a ata or a cisco phone or whatever... |
| 22:34.11 | Zorix | explidous, 28800 |
| 22:34.22 | Cybertoy | it's attended transfer.. so I'm even able to call the extension... but upon hitting the transfer key it's failing... |
| 22:34.30 | Cybertoy | ailed SIP Transfer to non-existing extension 1234567 in context default |
| 22:34.36 | Cybertoy | that's what I see in sip debug... |
| 22:34.53 | Cybertoy | so it's failing as that extension is not in the default context.... I don't want it to be there either. |
| 22:40.24 | N9URK_lithium | what kind of routers do you guys use for *? |
| 22:41.24 | robin_sz | any? |
| 22:41.40 | robin_sz | anyting with VLAN support |
| 22:41.50 | N9URK_lithium | what specific one are you using? |
| 22:41.56 | Math` | routing is routing.... use one with QoS support |
| 22:42.15 | N9URK_lithium | not so |
| 22:42.18 | robin_sz | QoS or VLAN tagging, there are many ways |
| 22:43.43 | robin_sz | I use 3com switches with vlansupprt, phones on a separate vlan |
| 22:44.05 | *** join/#asterisk CVirus (n=GoD@196.205.192.216) |
| 22:44.17 | CVirus | How is the majority of the softphones are configured ? |
| 22:44.27 | robin_sz | from a gui |
| 22:44.35 | CVirus | robin_sz: a web based GUI ? |
| 22:44.41 | robin_sz | no |
| 22:44.50 | N9URK_lithium | yeah bring up the ip your phone registered in your browser |
| 22:45.07 | CVirus | N9URK_lithium: how can I assign an IP to it ? |
| 22:45.08 | robin_sz | N9URK_lithium, he said softphones |
| 22:45.15 | CVirus | N9URK_lithium: if I got no dhcp server in my network |
| 22:45.23 | CVirus | oops |
| 22:45.24 | N9URK_lithium | robin_sz, CVirus oops sorry, misread that |
| 22:45.28 | CVirus | I meant SIP phones actually |
| 22:45.30 | CVirus | sorry |
| 22:45.32 | robin_sz | ahh |
| 22:45.45 | robin_sz | well, assing it an ip from the menu |
| 22:45.53 | N9URK_lithium | what phone do you have? |
| 22:45.54 | robin_sz | or get a dhcp server |
| 22:46.03 | CVirus | robin_sz: what menu ? |
| 22:46.07 | CVirus | N9URK_lithium: don't have one yet ? |
| 22:46.14 | robin_sz | lcd screen on phone? |
| 22:46.18 | CVirus | I see |
| 22:46.23 | CVirus | does all phones support this ? |
| 22:46.28 | N9URK_lithium | robin_sz, do you use managed or unmanaged switches |
| 22:46.30 | robin_sz | what phone is it? |
| 22:46.36 | N9URK_lithium | if you don't have a phone what makes you ask? CVirus |
| 22:46.39 | CVirus | haven't bought a phone yet |
| 22:46.39 | robin_sz | N9URK_lithium, managed |
| 22:46.45 | CVirus | N9URK_lithium: just asking |
| 22:46.56 | N9URK_lithium | CVirus, ok, a grandstream is pretty easy to configure |
| 22:46.58 | robin_sz | well, thye all allow you to set an ip .. afaik |
| 22:47.03 | N9URK_lithium | there is a menu button on the phone |
| 22:47.11 | CVirus | I see |
| 22:47.12 | robin_sz | grandstream? |
| 22:47.21 | N9URK_lithium | and it will let you change the ip |
| 22:47.27 | robin_sz | thats not a make of phone |
| 22:47.38 | CVirus | grandstream budget tone |
| 22:47.39 | robin_sz | the GXP2000 is it? |
| 22:47.48 | robin_sz | the budget tone is not too bad |
| 22:48.02 | robin_sz | but avoid the GXP2000 .. its a crock of shit |
| 22:48.03 | N9URK_lithium | I havea GXP2000 |
| 22:48.10 | robin_sz | I have 4 |
| 22:48.12 | robin_sz | all shit |
| 22:48.15 | N9URK_lithium | I have 2 of them and they work fine here |
| 22:48.29 | N9URK_lithium | A friend has had trouble with the BT |
| 22:48.31 | robin_sz | none of mine can even keep the display working for a day |
| 22:48.38 | N9URK_lithium | robin_sz, yikes |
| 22:48.46 | robin_sz | shittiest phones on the planet |
| 22:48.57 | N9URK_lithium | robin_sz, not had the problem. What do you use now? |
| 22:49.01 | robin_sz | I use Snom/Elmeg 190s now |
| 22:49.08 | robin_sz | or Snom 320/360 |
| 22:49.21 | *** join/#asterisk cspot (i=cspot@ip68-1-63-100.pn.at.cox.net) |
| 22:49.45 | CVirus | how about the budgetone 101 ? |
| 22:50.21 | robin_sz | http://www.voip-info.org/wiki/view/GXP-2000 |
| 22:50.22 | N9URK_lithium | robin_sz, why didn't you return them then? |
| 22:50.59 | robin_sz | yeah, should have |
| 22:51.19 | robin_sz | read the wiki there ... see the major bugs in current release |
| 22:52.20 | N9URK_lithium | I will read that, thanks for posting it |
| 22:54.49 | N9URK_lithium | robin_sz, is a 3com switch worth the more money than a similar Linksys one? |
| 22:55.23 | robin_sz | no eyed deer |
| 22:55.33 | robin_sz | i buy mine on ebay |
| 22:55.46 | robin_sz | 50~100 $ max |
| 22:55.49 | N9URK_lithium | that's funny, I will have to use that sometime |
| 22:56.00 | N9URK_lithium | I will look for one, great idea |
| 22:56.10 | N9URK_lithium | what model do you have? |
| 22:56.18 | robin_sz | various |
| 22:56.31 | robin_sz | 3300 |
| 22:56.36 | robin_sz | mainly ... |
| 22:56.46 | robin_sz | 24 port 10/100 switch |
| 22:56.52 | Cybertoy | ok .. I can answer my own question now ... :) |
| 22:57.11 | Cybertoy | there is indeed a channel variable called ${TRANSFER_CONTEXT} ... that one's set to "default" per default. |
| 22:58.44 | robin_sz | n9 serch for 3c16980A |
| 22:59.25 | N9URK_lithium | looking |
| 23:01.25 | N9URK_lithium | nothing came up on google or ebay |
| 23:01.26 | *** join/#asterisk Koshatul (n=evangeli@ppp188-33.static.internode.on.net) |
| 23:02.21 | N9URK_lithium | robin_sz, what do you use for dhcp with your 3300? |
| 23:03.00 | ManxPower | Cybertoy: README.variables is your frined |
| 23:03.01 | robin_sz | linux |
| 23:03.17 | robin_sz | N9URK_lithium, i run a dhcp server on the * box |
| 23:03.54 | robin_sz | its on its own vlan, so it just deals with the phones |
| 23:06.30 | N9URK_lithium | robin_sz, how difficult was it to get your dhcp server going? |
| 23:06.34 | Cybertoy | manx, tnx.. I'll look into that... although I discovered the channel variable it doesn't seem to work... |
| 23:06.38 | N9URK_lithium | robin_sz, I am reading about it now |
| 23:07.49 | mrichmanM | Is there anyone here that could help me setup my pri |
| 23:09.04 | Cybertoy | Manx, HA! ... I need __ ...... :) |
| 23:32.51 | *** join/#asterisk killfill_ (n=killfill@201.238.233.3) |
| 23:33.35 | killfill_ | hey, my te110p is only lettting me call (to the outside) ones. i.e. if i call with a secon phone, it says all channels are busy |
| 23:33.37 | killfill_ | where is that set?.. |
| 23:33.38 | *** join/#asterisk rikstah (n=rick@rhamnett.plus.com) |
| 23:35.34 | Math` | check your span definitions |
| 23:36.08 | killfill_ | got "channel=>1-15,17-31" in zapata.conf.. |
| 23:36.22 | killfill_ | and span=1,0,0,ccs,hdb3,crc4 in zaptel.. |
| 23:42.04 | *** join/#asterisk sav_ (n=dm@2013677155.nit.megazon.com.br) |
| 23:43.41 | *** part/#asterisk Cybertoy (n=cybertoy@dsl254-123-112.nyc1.dsl.speakeasy.net) |
| 23:50.04 | *** join/#asterisk e-Regg4e (n=ariel@201.224.94.114) |
| 23:51.36 | Maxxed | can anybody recommend a good sip provider? |
| 23:51.57 | Maxxed | i wana stick the ol pbx up at the colo and get rid of this pri |
| 23:54.05 | e-Regg4e | try voipjet |
| 23:54.08 | e-Regg4e | is iax |
| 23:55.25 | Math` | ya but it does outbound only |
| 23:55.49 | e-Regg4e | right |
| 23:56.08 | Maxxed | well, iax is better than sip anyway right? |
| 23:56.31 | e-Regg4e | iax is to connect 2 asterisks |
| 23:56.32 | Math` | not necessarily it depends |
| 23:56.47 | Math` | but in itself its great... just that more people support sip |
| 23:57.20 | Maxxed | yep |