| 00:00.19 | sevard | If you downloaded the zaptel source the man pages are in the doc directory under your source directory. |
| 00:00.23 | sevard | I can't be more specific than that |
| 00:01.50 | knarfly | sevard: I installed * and zaptel from the ports collection...not source...I'll get the source and gunzip it to see what's in there. Thanks |
| 00:04.07 | jtexter3 | knarfly: I'm not familiar with ports, never touched BSD. But, from your description, sounds like you haven't setup /etc/zaptel.conf (or equivalent) |
| 00:04.58 | jtexter3 | knarfly: Also, zttool is for show status, ztcfg is what loads the configuration |
| 00:06.47 | knarfly | jtexter3: Yes I have edited zaptel.conf. FreeBSD is quite different than Linux setup. |
| 00:07.11 | jtexter3 | knarfly: can you pastebin your zaptel.conf? |
| 00:07.26 | knarfly | jtexter3: I've tried ztcfg. It only shows me that 0 channels are configured. |
| 00:07.51 | knarfly | jtexter3: I followed the examples in * TFOT manual |
| 00:08.01 | jtexter3 | knarfly: What modules are you loading? |
| 00:08.43 | *** join/#asterisk seebs (n=seebs@216-243-131-210.static.iphouse.net) |
| 00:09.13 | seebs | Hey, dumb newbie question. I have a maximally-trivial Asterisk setup which just monitors calls. If I try to dial *xx, such as *69 or *57, I get a rapid-busy and no monitoring record starts. |
| 00:09.35 | seebs | I assume this means that there's something I have to do to tell Asterisk to allow asterisks in phone numbers? |
| 00:10.20 | seebs | exten => _X., 1, NoOp(Line 1 outbound to ${EXTEN}) <-- should this work? |
| 00:10.52 | knarfly | jtexter3: I only have the three simple lines shown in the example: |
| 00:10.53 | knarfly | fxoks=2 |
| 00:10.56 | knarfly | loadzone=us |
| 00:10.56 | knarfly | defaultzone=us |
| 00:11.04 | jtexter3 | seebs: that doesn't match your pattern. You need exten => _*X.,1,NoOp(Line 1 outbound to ${EXTEN}) |
| 00:11.13 | seebs | Huh. |
| 00:11.34 | sevard | seebs: seriously, it's a series of tubes. |
| 00:11.39 | seebs | Doh! |
| 00:11.42 | seebs | X is any digit. |
| 00:12.11 | jtexter3 | knarfly: assuming that you only have card, you need fxoks=1 |
| 00:12.18 | seebs | How do I match "any string of buttons at all", without special cases for * and #? |
| 00:13.06 | jtexter3 | seebs: I would use three rules. Technicall, you can do exten => _., but that does a lot more than you may realize. I would stick with _X, _*X, and _#X |
| 00:13.27 | seebs | Eep. |
| 00:13.33 | jtexter3 | seebs: with the dots of course( fat fingered the enter key ) |
| 00:13.44 | seebs | This means that my four rules (one for each line and each direction) become 12 rules, with WAY more potential for errors. |
| 00:13.47 | seebs | *sigh* |
| 00:14.08 | seebs | Can I put [#*] in brackets, and expect them to work? |
| 00:14.09 | jtexter3 | seebs: Use a macro |
| 00:14.17 | sevard | you'll find that you may want hundreds of rules. |
| 00:14.43 | jtexter3 | seebs: I can't say I've ever tried [#*]. i would interested to see if it works |
| 00:14.50 | jtexter3 | Though in theory it should |
| 00:15.15 | *** join/#asterisk juanjoc (n=juanjoc@248-32-235-201.fibertel.com.ar) |
| 00:15.26 | knarfly | jtexter3: Yes, I[m sorry I didn't mention that I just tried that too. |
| 00:15.35 | seebs | Okay, I'll try it. |
| 00:15.45 | seebs | My new rule is _[1-9#*]. |
| 00:15.47 | jtexter3 | knarfly: What modules are you loading? |
| 00:17.26 | jtexter3 | seebs: you forgot 0 |
| 00:17.40 | knarfly | jtexter3: zaptel.ko, qozap.ko, wcfxo.ko, wcfxs.ko, wct1xxp.ko, wct4xxp.ko, wctw11xp.ko, ztdummy.ko |
| 00:17.42 | knarfly | The standard stuff that loads with zaptel.sh |
| 00:18.05 | seebs | Fixed! |
| 00:18.16 | knarfly | jtexter3: wcte11xp.ko |
| 00:18.23 | sevard | iirc you should only be loading the module for your card |
| 00:18.24 | seebs | And as I suspected, the lying !@#$@$ "search engine optimization" people managed to bribe a phone company to be untraceable. |
| 00:18.43 | sevard | seebs: -what- |
| 00:19.36 | knarfly | sevard: so I should only load zaptel.ko and wcfxo.ko? |
| 00:19.46 | *** join/#asterisk Dibbler_ (n=Dibbler@dsl-217-155-254-174.zen.co.uk) |
| 00:19.57 | *** part/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net) |
| 00:20.21 | seebs | I needed to patch my asterisk setup so I could dial *57 to try to trace a call I got recently. |
| 00:20.32 | seebs | The only thing I use asterisk for is to get call records so I can sue telemarketers. |
| 00:20.45 | seebs | I think that's a very noble goal. :) |
| 00:23.31 | *** join/#asterisk ariel_ (n=Ariel@dsl-20-177.cofs.net) |
| 00:27.25 | *** join/#asterisk SplasPood (n=jwb@brooklyn.paravolve.net) |
| 00:33.00 | *** join/#asterisk rene1 (n=rene1@gea-gye-internet.telconet.net) |
| 00:33.22 | rene1 | ~seen jmls |
| 00:33.39 | jbot | jmls <n=asterisk@62.49.235.130> was last seen on IRC in channel #asterisk, 6h 37m 3s ago, saying: 'damn. too slow.'. |
| 00:33.39 | rene1 | hey |
| 00:34.24 | rene1 | i have seen that most softphones issue some sort of unregister to asterisk when they are shut down |
| 00:35.23 | rene1 | what i wonder if it was possible to run some commands when this unregister happens. like a dialplan context |
| 00:35.54 | rene1 | my main problem is that when i have one dynamic member go offline (a voip dynamic member) i would like to remove it from the queue |
| 00:38.00 | *** join/#asterisk sumasuma (n=sumkji@61.14.86.23) |
| 00:39.01 | rene1 | my member is a local channel |
| 00:39.14 | HaMYaI | hi, is there any way to specify a timeout for agi scripts? |
| 00:41.15 | rene1 | you could try with a timeout in that context and a t extension |
| 00:43.18 | rene1 | if i use normal voip members at least they show as unavailable |
| 00:43.43 | rene1 | i think i might have an unavailable extension in the local channel context to do the disconnection |
| 00:44.47 | HaMYaI | rene1: that wouldn't work for agi scripts |
| 00:45.12 | HaMYaI | I tried to set set_time_limit() for my php script but it just doesn't include time spent on most of the agi activities |
| 00:45.41 | kink0 | anyone who uses Page() ? |
| 00:46.03 | kink0 | I get the call when I do page(Zap/g1/mi_mobile_phone_number) |
| 00:46.16 | kink0 | but after I pick the phone, the system hangs |
| 00:46.27 | kink0 | hangs the line |
| 00:47.00 | kink0 | <PROTECTED> |
| 00:47.00 | kink0 | <PROTECTED> |
| 00:53.03 | *** join/#asterisk tuck3r (n=tuck3r@unaffiliated/tuck3r) |
| 01:03.00 | *** part/#asterisk rene1 (n=rene1@gea-gye-internet.telconet.net) |
| 01:04.38 | *** join/#asterisk spako (i=spakowsk@georgew.vcnet.com.br) |
| 01:08.42 | *** join/#asterisk synthetiq (n=tampon@ip68-9-234-50.ri.ri.cox.net) |
| 01:08.42 | *** join/#asterisk paolob (n=donpaolo@pri-214-b7.codetel.net.do) |
| 01:08.42 | *** part/#asterisk paolob (n=donpaolo@pri-214-b7.codetel.net.do) |
| 01:14.28 | *** join/#asterisk Katty (n=Angela@dialup-4.244.183.202.Dial1.StLouis1.Level3.net) |
| 01:16.00 | Katty | hihi. |
| 01:17.49 | qzxcd | hi |
| 01:18.07 | *** join/#asterisk _DAW (n=_DAW@adsl-222-12-239.msy.bellsouth.net) |
| 01:18.56 | ariel_ | Katty, hello, hope all is well. |
| 01:19.04 | *** join/#asterisk Defraz (n=t0tal@fw.centrisys.com) |
| 01:19.05 | Katty | ariel_: hey you (= |
| 01:19.11 | Katty | ariel_: all is going well...how're you stranger? |
| 01:19.13 | _DAW | hey everyone |
| 01:19.38 | Katty | hey _DAW |
| 01:19.39 | Defraz | Could someone help me figure out this log file. I can't seem to see why my fax detect hangs up right away. |
| 01:19.42 | ariel_ | Katty, fine. just been working allot... |
| 01:19.43 | *** join/#asterisk Native_Mind (n=ajb@c-69-142-19-57.hsd1.nj.comcast.net) |
| 01:19.47 | Defraz | I have the logs posted to pastebin. |
| 01:20.04 | Katty | ariel_: tell me about it...we're moving to a new building, and i've got 3 client installs going right now. |
| 01:20.47 | Katty | ariel_: and a house to clean :< |
| 01:20.51 | ariel_ | 3 clients wow, Sounds like my setup Setting up 1 new office and moving 3 call centers to a different building. |
| 01:21.10 | Katty | they're not asterisk clients. (= |
| 01:21.22 | sevard | heh. |
| 01:21.46 | Katty | but i've got a secondary asterisk box to setup for upstairs soon. |
| 01:21.49 | Katty | and a bes server. |
| 01:21.56 | ariel_ | that is good |
| 01:22.01 | Katty | and at least 6 other servers they're waiting for me to setup :< |
| 01:22.13 | Katty | don't have the time...people are pissy. |
| 01:22.17 | sevard | Send some to me. |
| 01:22.26 | Katty | sevard: i care about my clients. |
| 01:22.31 | sevard | I do too. |
| 01:22.33 | Katty | sevard: so you can't have them. |
| 01:22.40 | sevard | I care about your client's hardware. |
| 01:22.44 | ariel_ | I need a programmer. Things are too busy where I am at. But it's hard to find a good programmer that is asterisk aware. |
| 01:22.51 | Katty | we have a programmer (= |
| 01:22.57 | Katty | just hired him in fact... |
| 01:23.03 | ariel_ | nice |
| 01:23.07 | Katty | file: how're you dear? |
| 01:23.08 | sevard | ariel_: what languages? |
| 01:23.09 | ariel_ | I need one down here. |
| 01:23.13 | file | Katty: well! |
| 01:23.19 | Katty | file: deep. |
| 01:23.21 | ariel_ | Java, C, php depends |
| 01:23.35 | jtexter3 | ariel_: where you located? |
| 01:23.35 | sevard | ariel_: would Java, C, and Perl do? |
| 01:23.44 | ariel_ | sevard, yes |
| 01:23.51 | Katty | file: :< |
| 01:23.52 | sevard | May I message you? |
| 01:23.53 | ariel_ | South Florida |
| 01:24.15 | ariel_ | sevard, yes |
| 01:24.34 | Katty | ariel_: in other mews, i'm gonna be all grown up (again) soon! |
| 01:24.38 | jtexter3 | too bad. I would love to do more with asterisk, but I'm in Oklahoma |
| 01:24.58 | ariel_ | Katty, well in my view you were always grown up. |
| 01:25.13 | ariel_ | jtexter3, relocate |
| 01:25.19 | Katty | ariel_: minor detail. |
| 01:25.43 | jtexter3 | if only it were that simple. Wife in pharmacy school sort of prevents that |
| 01:25.55 | Katty | jtexter3: yay! a wife! |
| 01:25.56 | ariel_ | ahh |
| 01:26.08 | jtexter3 | yep, a wife is a great thing! |
| 01:26.52 | Katty | any kids yet? |
| 01:27.10 | jtexter3 | no, waiting for her to graduate, which happens in June |
| 01:27.19 | Katty | good thinking. |
| 01:28.29 | Katty | i'm still toying with the idea of kids. sorta. |
| 01:28.36 | Katty | can't make up my mind if i actually want any or not. |
| 01:29.06 | ariel_ | kids are great |
| 01:29.13 | jtexter3 | We want 3-4. Her sister just had her first kid, so we get to play with him without the hassle of diapers, sleepless nights, etc :D |
| 01:29.21 | Katty | eep! |
| 01:29.31 | Katty | 4? you're nuts. |
| 01:29.44 | Katty | but hey, more the merrier i guess (= |
| 01:29.50 | jtexter3 | So, what type of development are people doing with Asterisk? GUI's? Custom modules? AGI? A little bit of everything? |
| 01:30.27 | jtexter3 | She wants girls too, so if that happens I'll be at the deep end of the hormone pool ;-) |
| 01:30.58 | xachen | Why is bkw_ banned? |
| 01:32.03 | ariel_ | I have all girls |
| 01:32.12 | ariel_ | don't do it get some boys into the mix |
| 01:32.34 | Katty | but girls are so much fun. |
| 01:32.40 | jtexter3 | If I have any say, I'll have at least one boy. Gotta carry on the family name |
| 01:32.40 | Katty | you can play dress up, and tea parties. |
| 01:32.50 | Katty | and bake cookies with mamma. |
| 01:32.53 | ariel_ | jtexter3, Well programming is more for the reporting and queue management for us. We need also to do reporting and data processing between different databases. |
| 01:32.58 | Katty | and brunette pigtails running around chasing dogs. |
| 01:34.12 | jtexter3 | All of my cousins are boys, so I'm sure I'm doomed for 100% girls |
| 01:34.21 | Katty | all the little boys i've ever seen are just rowdy and noisy |
| 01:34.30 | jtexter3 | Many people here going to Astricon? |
| 01:35.20 | Katty | I doubt i'll have time. |
| 01:35.56 | *** join/#asterisk Pegasus_Epsilon (i=pegasus@198.174.233.133) |
| 01:36.01 | Pegasus_Epsilon | wonky network. |
| 01:37.12 | sevard | wonkity. |
| 01:37.14 | Katty | you know how when a person enters a room, it shows their hostname thingy |
| 01:37.31 | Katty | if it has just an ip, does that mean they don't have reverse dns lookup? |
| 01:37.48 | Pegasus_Epsilon | that means that the server they connected to failed to look it up |
| 01:37.55 | Katty | oh ah. |
| 01:37.56 | Katty | k'then. |
| 01:38.09 | wunderkin | dont everyone has reverse |
| 01:38.12 | wunderkin | er |
| 01:38.13 | Pegasus_Epsilon | no |
| 01:38.15 | Katty | nope |
| 01:38.18 | wunderkin | i meant not |
| 01:38.32 | sevard | for the record Katty, it's "channel" not "room" |
| 01:38.34 | Katty | there's another one too |
| 01:38.43 | Pegasus_Epsilon | most people have reverse DNS support, but not administration rights |
| 01:38.44 | Katty | sevard: i've been using irc for 9 years |
| 01:38.50 | Katty | sevard: i'll call it a channel if i so desire |
| 01:38.54 | Katty | sevard: or i'll call it a room |
| 01:39.07 | sevard | If you've been using IRC for 9 years you ought to be used to calling it a channel by now. |
| 01:39.12 | wunderkin | heh |
| 01:39.28 | Katty | sevard: run along and pester someone else (= |
| 01:39.32 | Katty | Pegasus_Epsilon: something like spif |
| 01:39.43 | Pegasus_Epsilon | what? |
| 01:39.44 | Katty | Pegasus_Epsilon: or sdif |
| 01:39.53 | Katty | Pegasus_Epsilon: it's on dnsreport.com |
| 01:39.54 | wunderkin | lmnop? |
| 01:39.59 | Pegasus_Epsilon | qwert |
| 01:40.02 | Katty | Pegasus_Epsilon: another one of those spamy fighting things. |
| 01:40.05 | Katty | nopenope, starts with an s. |
| 01:40.06 | *** join/#asterisk pabluss (n=aquicamb@200.75.1.29) |
| 01:40.32 | Defraz | Has anyone seen this error? No application 'rxfax' for extension |
| 01:40.39 | Pegasus_Epsilon | Katty: i have no idea what the hell you're talking about. and i'm not terribly surprised, or upset by that. |
| 01:40.48 | Katty | Pegasus_Epsilon: i'll go get it. |
| 01:40.59 | wunderkin | i know what shes referring to but cant remember the name either |
| 01:41.02 | pabluss | :( problems... sip connection ready but there isn't voice between extensions |
| 01:41.22 | Pegasus_Epsilon | Defraz: you have a file extension associated with a program called rxfax, which isn't installed |
| 01:41.35 | pabluss | i've just opened 5060 udp sip port and 10000:20000 udp port for rtp |
| 01:41.36 | Defraz | hmmmm |
| 01:41.39 | pabluss | hmmm |
| 01:42.22 | Katty | spf record! |
| 01:42.27 | Katty | wunderkin: that's what it was (= |
| 01:42.29 | *** join/#asterisk sevard (n=sev@c-67-188-173-23.hsd1.ca.comcast.net) |
| 01:42.33 | wunderkin | ok |
| 01:42.52 | *** join/#asterisk quid246 (n=quid24@CPE00131078ba5d-CM000f9f7eff1e.cpe.net.cable.rogers.com) |
| 01:42.55 | Katty | Pegasus_Epsilon: i know what i'm talking about, even if i can't remember names ;) |
| 01:42.59 | pabluss | hi quid246 |
| 01:43.17 | sevard | Nasa is playing this old video from the early 90s. It's for children. They just taught the kids that Pluto is a planet. |
| 01:43.18 | Pegasus_Epsilon | yes, but i don't know what you're talking about, but for some reason, you're talking to me |
| 01:43.26 | Pegasus_Epsilon | even though i don't really care about your spam toys |
| 01:43.28 | pabluss | quid246: problem... only i can make sip connection between extensions but no voice |
| 01:43.29 | sevard | It's not okay to play old material, NASA. |
| 01:43.54 | pabluss | quid246: i don't know what's the wrong |
| 01:43.56 | Pegasus_Epsilon | sevard: what'd they call it now, a protoplanet? |
| 01:44.00 | Katty | Pegasus_Epsilon: i see. |
| 01:44.19 | Katty | i'll talk to file then. he's not snobby. heh |
| 01:44.26 | quid246 | pabluss: Are the connections on the same subnet? |
| 01:44.29 | Pegasus_Epsilon | does file call it a "room"? |
| 01:44.38 | pabluss | quid246: no |
| 01:44.52 | pabluss | quid246: in severals lans |
| 01:44.56 | sevard | Close, a dwarf planet. |
| 01:45.04 | Pegasus_Epsilon | ah |
| 01:45.10 | pabluss | quid246: i get this information in sip.conf file |
| 01:45.10 | quid246 | pabluss: turn off all firewalling between the lans.. and try it. |
| 01:45.30 | sevard | Pegasus_Epsilon: they really had no other choice, either dwarf pluto or promote several other rocks to planet status. |
| 01:45.34 | pabluss | quid246: i tunr off my adsl modem firewall |
| 01:45.43 | Pegasus_Epsilon | like haley's comet |
| 01:45.51 | pabluss | quid246: and turn off my work server firewall |
| 01:46.04 | sevard | it's stupid anyway, let's all get fired up and angry about how we name things. whatever, it doesn't change the actual object.. just how we reference it. |
| 01:46.32 | Katty | Pegasus_Epsilon: amazing. |
| 01:46.45 | Katty | Pegasus_Epsilon: i didn't think that when guys grew up, they still acted like kids. |
| 01:47.00 | Katty | Pegasus_Epsilon: don't let me rain on your parade though. |
| 01:47.25 | sevard | I didn't know we were going to degrade outself to child status by becoming defensive and rolling out with the sexist insults. |
| 01:47.31 | sevard | don't lst us rain on your prarde, though. |
| 01:47.35 | Katty | haha. |
| 01:47.38 | sevard | s/lst/let/g |
| 01:47.39 | Pegasus_Epsilon | at least i have the good sense and experience to realize that women are generally stupid twats until they're about 45. |
| 01:48.00 | Pegasus_Epsilon | prardes rule. |
| 01:48.12 | Katty | you can have the last comment, if it makes you feel better :) |
| 01:48.21 | sevard | Thanks, you suck. |
| 01:48.31 | Katty | feel better? |
| 01:48.38 | sevard | Absolutly: you suck. |
| 01:48.51 | Katty | fantastic. life's too short to waste on nonsense. |
| 01:48.57 | sevard | he. |
| 01:55.11 | Defraz | do I have to compile anything with asterisk to do faxing |
| 01:56.26 | Pegasus_Epsilon | only if you want to send/receive faxes with asterisk itself |
| 01:59.52 | Defraz | hmm cuz it looks like it is looking for rxfax extention |
| 02:00.01 | Defraz | would that be in asterisks it's self? |
| 02:01.53 | Pegasus_Epsilon | Defraz: i presume you're running windows |
| 02:09.22 | *** part/#asterisk pabluss (n=aquicamb@200.75.1.29) |
| 02:09.33 | *** join/#asterisk RoyK (n=roy@gprs-ggsn6-nat.mobil.telenor.no) |
| 02:12.53 | sevard | man... i love people that ask for help and then argue with you. |
| 02:13.04 | Pegasus_Epsilon | heh |
| 02:13.25 | Pegasus_Epsilon | the best part is when you turn out to be right anyway |
| 02:13.37 | sevard | my friend came to me and asked me how to tell if his cdrom was fried, i gave him the run down, 'if it's not this then it's this, if it's not that then it's this' |
| 02:14.02 | sevard | well no shit you are right, it's why they came to you asking for help in the first place |
| 02:14.13 | sevard | they knew you'd be right but holy hell they hate being told they are wrong. |
| 02:14.16 | *** join/#asterisk justinu|laptop (n=Justin@dsl081-083-093.lax1.dsl.speakeasy.net) |
| 02:14.35 | Pegasus_Epsilon | i mean more like "i think it's a DNS problem" |
| 02:14.45 | sevard | hey justinu|laptop |
| 02:15.08 | sevard | Pegasus_Epsilon: you mean, "we have slow ping responses.. i thin kit's a DNS problem" |
| 02:15.12 | justinu|laptop | yo |
| 02:15.15 | Pegasus_Epsilon | yes |
| 02:20.30 | sevard | oh sweet |
| 02:20.35 | sevard | i invited my gf over for dinner |
| 02:20.38 | sevard | and she looked at my food |
| 02:20.53 | sevard | and decided that i should take her out for dinner. |
| 02:21.08 | Pegasus_Epsilon | heh |
| 02:21.14 | sevard | awesome. |
| 02:21.35 | sevard | :/ later. |
| 02:33.10 | *** join/#asterisk kristalino (n=kristali@LNeuilly-152-21-126-73.w193-253.abo.wanadoo.fr) |
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| 02:35.58 | *** mode/#asterisk [+o denon] by irc.freenode.net |
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| 02:53.26 | Aces1Up | does anyone here have experience interfacing cell-phones to asterisk? |
| 02:56.46 | *** join/#asterisk osiris (n=osiris@c-71-205-27-131.hsd1.mi.comcast.net) |
| 03:00.07 | Jason99 | Aces1Up: what exactly are you trying to do? |
| 03:01.06 | *** join/#asterisk TESTER2 (n=Cyber@modemcable082.42-81-70.mc.videotron.ca) |
| 03:02.08 | TESTER2 | Hi, someone knows if chan_bluetooth is still developped somewhere? |
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| 03:25.05 | kronic | I'm having an issue with this dialplan: http://pastebin.com/799811 |
| 03:25.13 | kronic | seems that the ${TRUNK} variable isn't used |
| 03:25.23 | kronic | I tried a NoOp with no result in the CLI |
| 03:26.23 | Corydon76-home | Perhaps you should try a Verbose |
| 03:26.49 | kronic | yeah, tried that, set to 5 |
| 03:27.10 | *** join/#asterisk angom_h (n=Angel@red-corp-200.38.15.105.telnor.net) |
| 03:32.13 | Jason99 | kronic: from what I can see it wouldn't be calling your SET lines because it would match another dial plan first |
| 03:32.36 | Corydon76-home | Try level 0, instead |
| 03:33.17 | Corydon76-home | or the Log() application |
| 03:33.59 | Corydon76-home | If you do verbose(5,...) you need to do a 'set verbose 5' on the command line |
| 03:34.01 | *** part/#asterisk hyphen (n=hyphen@71.224.213.97) |
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| 03:40.58 | kronic | Jason99: are you referring to the context and priorities? |
| 03:41.42 | Jason99 | kronic: If calling the [my-context], I believe it will look for a match within [iax-peers] and [agent-login] first. If a match is found it wouldn't call the SET(TRUNK=...).. |
| 03:42.03 | *** join/#asterisk linuxbangalore (n=karsansu@59.92.194.240) |
| 03:42.25 | linuxbangalore | hi I am getting the following error |
| 03:42.26 | linuxbangalore | Got SUBSCRIBE for extension 1234@testsip from 192.168.1.100, but there is no hint for that extension |
| 03:43.22 | linuxbangalore | no hint? how can debug this error |
| 03:50.32 | kronic | Jason99: I read up on it, seems that you need to place the inclusion above the exten processing, yet no difference :S |
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| 03:51.21 | vader192 | Has anyone had any expericence with GR303 and asterisk? |
| 03:55.31 | Jason99 | kronic: for example, if you dialed 130000, it would simply call Dial() and would never call your two Set() lines. |
| 03:56.11 | kronic | think I've screwed up the priorities |
| 03:56.52 | Jason99 | I'll post a possible solution on the paste-bin |
| 03:56.57 | kronic | cheer |
| 03:57.01 | kronic | *cheers mate |
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| 04:00.34 | Jason99 | kronic: have a look |
| 04:02.48 | kronic | arhh, thanks |
| 04:04.25 | Jason99 | does it work? |
| 04:04.45 | kronic | editing now |
| 04:04.52 | Jason99 | ok |
| 04:08.56 | kronic | having issues matching an outbound number in the dial plan |
| 04:09.01 | kronic | doesn't match 0XXX... numbers |
| 04:10.47 | *** join/#asterisk L|NUX (n=linux@unaffiliated/lnux/x-10290) |
| 04:13.49 | Jason99 | kronic: check the paste-bin again, i commented out lines in [iax-peers] |
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| 04:23.20 | Jason99 | kronic: I did that because it would have called 1,Ringing, 2,Wait(1) and then 3,HangUp.. It would have never matched the _0XXX. |
| 04:27.55 | sev | wtf |
| 04:28.11 | Aces1Up | what software can i use to interface my bluetooth cellphone to asterisk, and send receive calls through bluetooth? or is there a usb interface that i can use to interface my cell-phone to asterisk? |
| 04:28.23 | sev | nick name recovery wtf |
| 04:30.37 | *** part/#asterisk TESTER2 (n=Cyber@modemcable082.42-81-70.mc.videotron.ca) |
| 04:32.13 | *** join/#asterisk littleball (n=littleba@bb58-185-248-174.singnet.com.sg) |
| 04:32.24 | littleball | hello, any one using h323 in asterisk? |
| 04:32.41 | littleball | is the h323 channel implementation stable? |
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| 04:42.31 | JimJty | does anyone know where recordings are stored (directory)? |
| 04:42.46 | kronic | /var/spool/asterisk/monitor by default |
| 04:43.15 | JimJty | does that include voice mail recordings? |
| 04:43.53 | kronic | have a look in /var/spool/asterisk |
| 04:43.59 | kronic | there's monitor and voicemail |
| 04:44.05 | JimJty | ahh yes i see thanks for the help |
| 04:45.58 | *** part/#asterisk westie42 (n=westie42@12-226-85-195.client.mchsi.com) |
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| 04:49.25 | HaMYaI_ | is there a way to have agi script time out? |
| 04:49.41 | *** join/#asterisk Corydon-w (n=tilghman@pdpc/supporter/sustaining/Corydon76-home) |
| 04:49.41 | *** mode/#asterisk [+o Corydon-w] by ChanServ |
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| 04:58.06 | [TK]D-Fender | kronic : What you put in the pastebin won't work because youa re trying to pattern match against ".", and didn't put the "_" in front. |
| 04:58.59 | [TK]D-Fender | kronic : and you don't even DO anything in there. you'd set 2 vars and then the channel would hang up even if it WERE to match |
| 05:01.52 | [TK]D-Fender | kronic : Also you are setting variables whose lifespan is that of the overall channel which never gets to the point of using them for anything. |
| 05:03.10 | kronic | I'm basically wanting to set a var called TRUNK so I can reuse a particular dialplan across multiple trunks |
| 05:04.11 | [TK]D-Fender | kronic : what you should do is make a macro to do you actual dialing and use a pattern match to decide what to call the macro and pass it. |
| 05:18.00 | kronic | [TK]D-Fender: that's not what I'm after though |
| 05:18.41 | kronic | [TK]D-Fender: because that's going to require repetition of the matches etc. |
| 05:20.10 | HaMYaI_ | what's the effect of Set(TIMEOUT(absolute)=timeout)? |
| 05:21.42 | HaMYaI_ | I set it to 10 secs in my agi script and sleep for 20 secs, the call just doesn't hangup when 10th sec is reached |
| 05:24.26 | [TK]D-Fender | kronic :quite the opposite actually..... |
| 05:30.38 | *** join/#asterisk Winkie (n=urmom@cpc2-stre3-0-0-cust344.bagu.cable.ntl.com) |
| 05:32.19 | sevard | Two universal remotes. Each has the ability to control one device :| |
| 05:45.55 | Snake-Eyes | is there a way to shorten the time it takes Asterisk to recieve incoming call from zap (tdm card) to ring the phone? I've used ringtimeout=2300 but I was wondering if theres something better. |
| 05:47.16 | *** join/#asterisk Chris-NB (n=chris@ng1.kurtkrenn.com) |
| 05:50.06 | hads | Snake-Eyes: Disable callerid |
| 05:51.42 | Snake-Eyes | hads, so callerid slows up the process by quite abit ? |
| 05:52.22 | hads | On analog yes. Asterisk will wait for the callerid to be received. |
| 05:53.55 | Snake-Eyes | can callerid be sent along analog lines ? (or does it depend on your carrier) |
| 05:54.57 | hads | Yes it can, and yes it does. |
| 05:56.05 | Snake-Eyes | hmm, so if is use callerid i have to live with 1-2 rings before asterisk starts ringing the ip phone |
| 06:00.08 | hads | Yes. |
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| 06:12.05 | Snake-Eyes | thanks hads |
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| 06:31.05 | littleball | anyone here using h323 channels ? |
| 06:31.32 | alexburke | Hello, world! I'm having trouble with outbound CID being lost. The Set works for CALLERID(all), but then when the SIP INVITE goes out, it includes: From: "Unknown" <sip:Unknown@1.2.3.4>;tag=stuff |
| 06:31.42 | alexburke | I'm running 1.2.12.1 with FreePBX 2.1.3. |
| 06:47.26 | *** join/#asterisk juanjoc (n=juanjoc@248-32-235-201.fibertel.com.ar) |
| 06:56.50 | marcus2 | so is there a decent/reliable/long term iax provider selling 800/888 DIDs? |
| 07:00.47 | *** join/#asterisk eject_ck (n=eject@mail.interlink.ck.ua) |
| 07:06.10 | *** join/#asterisk alexburke (n=Octothor@dsl-205-210-52-82.kingston.net) |
| 07:06.19 | alexburke | Hello, world! I'm having trouble with outbound CID being lost. The Set works for CALLERID(all), but then when the SIP INVITE goes out, it includes: From: "Unknown" <sip:Unknown@1.2.3.4>;tag=stuff |
| 07:06.24 | alexburke | Any thoughts as to what could be causing it? |
| 07:06.40 | alexburke | I'm running Asterisk 1.2.12.1 with FreePBX 2.1.3. |
| 07:14.48 | *** join/#asterisk qdk (n=qdk@213.150.62.32) |
| 07:17.02 | *** join/#asterisk psk (n=psk@golia.caltanet.it) |
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| 07:26.42 | alexburke | anyone awake? |
| 07:29.18 | eject_ck | Hi all! I'm newbie and have troubles with configuring Zap interface please see my logs http://pastebin.ca/190678 |
| 07:34.08 | stephane_ | jour |
| 07:42.21 | *** join/#asterisk lorinc (n=ang@caracas-1177.adsl.interware.hu) |
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| 08:08.27 | *** join/#asterisk KaZeR (n=kazer@220.204.98-84.rev.gaoland.net) |
| 08:08.45 | KaZeR | hi there! |
| 08:09.01 | *** join/#asterisk Star568 (i=Star@69-167-150-120.lmdaca.adelphia.net) |
| 08:09.41 | Star568 | anybody here give me a suggestion which prepaid package i should go for? |
| 08:13.43 | *** join/#asterisk Cyt (n=Cyt@85.75.176.202) |
| 08:20.36 | E-bola | are u kidding? |
| 08:22.18 | *** join/#asterisk HaMYaI (n=HaMYaI@ppp-58.8.14.32.revip2.asianet.co.th) |
| 08:23.40 | benjk | Star568, did you forget already what I told you? |
| 08:30.30 | *** join/#asterisk Zouzou (n=test@193.227.186.146) |
| 08:30.59 | Zouzou | what are the default codecs used by asterisk if nothing is mentionned in ip.conf |
| 08:31.27 | Zouzou | *sip.conf |
| 08:33.28 | DarKnesS_WolF | Zouzou: try a call and then show sip channle blabla |
| 08:33.52 | KaZeR | i'd like to test asterix with a rs232 modem, or a soundcard, whatever does not require buying a specific card at the moment. is it possible? |
| 08:36.30 | *** join/#asterisk HarryR (n=Harry@host-83-146-53-46.bulldogdsl.com) |
| 08:36.50 | Cyt | Hi, please is there any dialplan application which waits for the CALLED PART to press anykey? |
| 08:38.55 | benjk | KaZeR, chan_modem can use some voicemodems BUT ... |
| 08:39.00 | benjk | a) it has been deprecated |
| 08:39.06 | benjk | b) it is half duplex only |
| 08:39.36 | *** join/#asterisk eject_ck (n=eject@mail.interlink.ck.ua) |
| 08:39.50 | KaZeR | benjk, ok, thanks |
| 08:40.01 | KaZeR | &) deprecated, as doesn't work and won't be fixed? |
| 08:40.28 | KaZeR | b) doesn't matter at the moment, i'm thinkig of bying a zapteck card, but i'd like to test a little bit asterisk first |
| 08:41.27 | benjk | deprecated as "was removed from the distribution as of Asterisk 1.2" |
| 08:41.56 | benjk | half duplex means you cannot talk in both directions at the same time |
| 08:42.09 | benjk | you can only use it for answering machine mode |
| 08:42.37 | KaZeR | benjk, that's what i'm wanting at the moment, answering machine |
| 08:42.52 | benjk | well, check chan_modem then |
| 08:42.58 | KaZeR | ok thanks :) |
| 08:43.00 | benjk | maybe you can port it to Asterisk 1.2 |
| 08:43.17 | benjk | it works with a few voice modems on Asterisk 1.0.x |
| 08:43.28 | benjk | only very few modems though |
| 08:43.45 | KaZeR | well, if i can get it to work, i'll probably buy a zapteck card |
| 08:43.55 | benjk | but you can probably change the AT commands so it will work with your modem |
| 08:44.00 | benjk | but it has to be a voice modem |
| 08:44.57 | benjk | for experimenting only (not for any real use) you could buy one of those PCI modems with the MD3200 chipset, also known as X100P cards |
| 08:45.17 | benjk | they are about 15-25 USD on eBay |
| 08:45.56 | benjk | there is a driver for them "wcfxo" that lets you use them as an FXO port (for connecting to a POTS line) |
| 08:46.31 | benjk | they totally suck because the chips are no longer manufactured and the cards you can get now are either made from refurbished chips or reject chips |
| 08:46.45 | benjk | but for learning they are probably ok |
| 08:51.00 | alexburke | How can I increase the amount of verboseness at the asterisk console? I've tried "set verbose 15" but I'm still not seeing step-by-step output from AGI scripts |
| 08:51.07 | alexburke | dialparties.agi being the main one I'm interested in |
| 08:51.46 | benjk | I don't think asterisk's verbosity influences the AGI at all |
| 08:51.54 | benjk | because the AGI is external to Asterisk |
| 08:52.08 | benjk | you are calling an external script to execute outside of Asterisk |
| 08:52.13 | alexburke | right. |
| 08:52.17 | alexburke | but I used to be able to see that stuff. |
| 08:52.45 | benjk | you probably want to study the code of that agi script |
| 08:52.53 | alexburke | like, for example: http://www.trixbox.org/modules/newbb/viewtopic.php?topic_id=2839&forum=2&viewmode=flat&order=ASC&start=20#forumpost12401 |
| 08:54.34 | alexburke | see what i mean? |
| 08:55.49 | alexburke | benjk? |
| 08:55.59 | KaZeR | benjk, X100P is what i'll probably buy :) |
| 08:56.54 | benjk | KaZeR, just remember, its TOTAL CRAP, only good as a simulation device for experimenting/learning |
| 08:57.33 | *** join/#asterisk aiksa[LV] (n=aiksa[LV@83.223.131.104) |
| 08:57.50 | KaZeR | benjk, ok thanks for the tip. this is what i'm going to do now, just play a bit and leanr |
| 08:57.52 | benjk | alexburke, I don't know much about this, I don't use AGI, but I know how the dialplan engine passes control to an AGI |
| 08:58.24 | E-bola | http://pastebin.com/799937 |
| 08:58.41 | E-bola | can any1 tell me why that hangsup the call after the first dial command? |
| 08:58.47 | E-bola | it should go on and then dial the other number |
| 09:00.39 | HarryR | because you're starting at 100.. |
| 09:01.14 | E-bola | nah im not its just a cutout of a bigger plan |
| 09:01.35 | E-bola | its once it reaches 101 and then after the 40 seconds timeout |
| 09:01.36 | HarryR | ah |
| 09:01.36 | E-bola | it dies |
| 09:02.05 | E-bola | -- Got SIP response 486 "Busy Here" back from 213.150.48.226 |
| 09:02.05 | E-bola | <PROTECTED> |
| 09:02.05 | E-bola | 2006-10-04 10:50:15 DEBUG[15648]: chan_sip.c:2426 sip_hangup: update_call_counter(Backup112) - decrement call limit counter |
| 09:02.05 | E-bola | <PROTECTED> |
| 09:02.06 | E-bola | <PROTECTED> |
| 09:02.08 | E-bola | 2006-10-04 10:50:15 DEBUG[15648]: app_dial.c:1619 dial_exec_full: Exiting with DIALSTATUS=BUSY. |
| 09:02.10 | E-bola | <PROTECTED> |
| 09:02.35 | *** join/#asterisk abatista (n=Ariel@dsl-20-177.cofs.net) |
| 09:03.20 | hank | whats the difference between pci_cpe and pri_net? could one say like bri_cpe is for te mode and pri_net for nt mode? |
| 09:05.48 | alexburke | I'm running Asterisk 1.2.12.1 and FreePBX 2.1.3. I'm having trouble with outbound CID being lost. The Set works for CALLERID(all), but then when the SIP INVITE goes out, it includes: From: "Unknown" <sip:Unknown@1.2.3.4>;tag=stuff -- any thoughts as to what could be causing this, or places I could look for clues? |
| 09:06.41 | *** join/#asterisk zeedo (n=zeedo@reboot-robot.net) |
| 09:12.51 | qdk | alexburke: so how are your reading skills? |
| 09:12.59 | *** join/#asterisk kink0 (n=kinko@pluton.interec.com) |
| 09:13.12 | kink0 | g.morning |
| 09:13.35 | kink0 | anyone have experience with Page() ? |
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| 09:18.25 | *** join/#asterisk b|aise (n=blaise@12-210-245-153.client.mchsi.com) |
| 09:19.10 | b|aise | pardon, are there any free sip endpoints? |
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| 09:24.59 | *** part/#asterisk alexburke (n=Octothor@dsl-205-210-52-82.kingston.net) |
| 09:26.46 | *** join/#asterisk kronic (n=gnorman@mail.stabat.com) |
| 09:26.56 | kronic | hey guys, I'm having some issues with the manager interface, I |
| 09:27.17 | kronic | *I've defined my bindaddr, but netstat reports that its bound only to localhost.localdomain |
| 09:27.31 | kronic | so I can't externally connect |
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| 09:43.34 | *** part/#asterisk kronic (n=gnorman@mail.stabat.com) |
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| 09:54.09 | *** join/#asterisk oQPa (i=name@70.Red-83-33-148.dynamicIP.rima-tde.net) |
| 09:54.15 | lilalinux | hey guys |
| 09:54.32 | lilalinux | can I forward an incoming call to 2 sip clients, and who ever answers first, gets the call? |
| 09:58.46 | Makenshi | lilalinux, yes. |
| 09:59.50 | lilalinux | Makenshi: do you have a starter? |
| 10:00.09 | lilalinux | I search G for ringdown, but didn't find domething usefull |
| 10:07.13 | hwt | how can i use the Sipaddheader and -getheader functions in AEL? |
| 10:07.18 | *** join/#asterisk HaMYaI (n=HaMYaI@ppp-58.8.14.32.revip2.asianet.co.th) |
| 10:07.42 | Makenshi | lilalinux, Dial(SIP/123,SIP/234) will route the call to two extensions |
| 10:07.48 | Makenshi | something like that, anyway |
| 10:09.21 | *** join/#asterisk saftsack (n=oliver@p54A7C886.dip.t-dialin.net) |
| 10:09.57 | *** join/#asterisk jmls (n=asterisk@62.49.235.130) |
| 10:16.24 | *** join/#asterisk mtaht4 (n=m@65.242.184.131) |
| 10:17.01 | *** join/#asterisk nassy (n=nassy@207-38-197-201.c3-0.wsd-ubr1.qens-wsd.ny.cable.rcn.com) |
| 10:22.02 | *** join/#asterisk toonz (n=dexter@dsl-145-255-202.telkomadsl.co.za) |
| 10:22.06 | toonz | howsit howsit :) |
| 10:22.57 | toonz | anyone familiar with CDR stuff on asterisk ? ... I have a situation, where sometimes the channel and dstchannel, are the wrong way around. and yeah, it really is. i'm shure. |
| 10:23.58 | *** join/#asterisk epvdw (n=epvdw@tttelecom.xs4all.nl) |
| 10:25.38 | epvdw | Hello all! I have an Asterisk system running CentOS 4.4 and Asterisk 1.2.12.1 using a Digium B410P BRI card. It crashes with a kernel panic as soon as the first call on mISDN hangs up |
| 10:28.15 | *** join/#asterisk in-pt (n=lokesh@estrela.nortenet.pt) |
| 10:29.20 | saftsack | buy a patton gateway with bri functionality. you save time with bad isdn drivers this way ;) |
| 10:30.47 | *** join/#asterisk scanna (n=scannach@81-174-16-211.f5.ngi.it) |
| 10:31.20 | toonz | epvdw: for lack of a better reply, I would suggest you try the bristuffed drivers from www.junghanns.net. the last thing you wanna hear when u have a piece of hardware, is someone telling you to buy something else. |
| 10:33.38 | epvdw | we just switched from multiple cheap HFC board to the expensive Digium Quad solution with onboard echo-cancellation, expecting it to be plug and play (as it is a Digium!!) |
| 10:34.06 | MrChimpy | normal mistake. digium h/w not very good. |
| 10:34.25 | epvdw | with the cheap HFC board we used the bri-stuffed drivers (but in the Netherlands is giving us Telco issues with continues sync) |
| 10:39.41 | *** join/#asterisk Mandus (n=aasmundo@simula-015.simula.no) |
| 10:40.04 | saftsack | epvdw, did you experienced echo with the cheap hfc card? |
| 10:41.25 | florz | a better question would be: Are there any analog phones connected to it without any EC in between? |
| 10:41.57 | florz | Otherwise it's probably the wrong place to do the EC anyway ... |
| 10:43.16 | florz | Plus, IIRC there was some solution to that problem with KPN, even though I don't remember what, except that it didn't involve the need for new hardware ;-) |
| 10:50.32 | *** part/#asterisk mtaht4 (n=m@65.242.184.131) |
| 10:52.20 | *** join/#asterisk grEvenX (n=even@pc107-130.ktv.no) |
| 11:01.30 | kink0 | anyone have experience with Page() ? |
| 11:01.54 | kink0 | I am trying to originate fews calls to the PSTN network and put all called in a meeting |
| 11:02.28 | kink0 | appears my Page(Zap/g1/number1&Zap/r1/number2) runs, but call is hanged just about 2 or 3 seconds |
| 11:03.30 | kink0 | I have try to add Dial() next priority after Page, but that appears doesn't work |
| 11:03.47 | kink0 | pretending to hold the call while the meeting is alive. |
| 11:04.06 | DarKnesS_WolF | kink0: what page do ? |
| 11:04.43 | kink0 | DarKnesS_WolF: calls a Tech/extension or more than one, and place all in a new meeting room |
| 11:04.50 | *** join/#asterisk ivanfm (n=ivanfm@c93481ec.virtua.com.br) |
| 11:10.57 | *** join/#asterisk olivier__ (n=olivier@obs92-4-82-239-116-113.fbx.proxad.net) |
| 11:13.00 | *** join/#asterisk olivier__ (n=olivier@obs92-4-82-239-116-113.fbx.proxad.net) |
| 11:13.25 | epvdw | saftsack, (sorry I was out away for a while) but yes we did experience echo... probably because of using multiple board and timing issues |
| 11:16.27 | DarKnesS_WolF | kink0: it's a a broadcast app them ? |
| 11:16.46 | *** join/#asterisk inspired (n=mikael@85.221.7.59) |
| 11:17.03 | kink0 | DarKnesS_WolF: well, you can use to broadcast if you like, then don't use option d and all users will be muted |
| 11:17.38 | kink0 | DarKnesS_WolF: my problem is that works fine for just fews seconds, then I got Hangup even if no one party has hanged |
| 11:18.27 | DarKnesS_WolF | kink0: did u check ur full log ? |
| 11:19.06 | epvdw | florz, KPN is not happy with us when we request continues line sync... the are not willing to enable it, though we need it when installing our Asterisk build at customer premises |
| 11:19.39 | epvdw | and no, there are no analog lines without EC |
| 11:20.09 | tzafrir | What exactly is that continues line sync? |
| 11:20.36 | *** join/#asterisk [Outcast] (n=bill@222-154-84-179.jetstream.xtra.co.nz) |
| 11:20.40 | kink0 | DarKnesS_WolF: is hard to check a full debug since there a lot of other calls in progress in the same asterisk, and I don't know what channel name will be assigned in next test call |
| 11:24.53 | epvdw | line sync is necessary on the KPN network when using (the HFC bri-stuffed) Asterisk and ISDN in The Netherlands, otherwise connectivity is copletely gone at frequent intervals |
| 11:25.06 | *** join/#asterisk saftsack (n=saftsack@p54A7C87D.dip.t-dialin.net) |
| 11:25.38 | *** join/#asterisk cpm (n=Chip@pdpc/supporter/active/cpm) |
| 11:27.39 | epvdw | hi saftsack, (sorry I was also out away for a while) but yes we did experience echo... probably because of using multiple board and timing issues |
| 11:28.06 | saftsack | epvdw, yes |
| 11:28.43 | *** join/#asterisk c0w (n=sean@212.9.103.220) |
| 11:29.38 | c0w | hello all just a quick question i've moved from ooh323 to just the h323 channel and i now have a problem with special characters breaking the dialing |
| 11:30.08 | HaMYaI | anyone know why the volume is very low with 16-bit mono wav format? |
| 11:30.12 | c0w | Executing Dial("Zap/33-1", "h323/329015949@192.168.1.23 |
| 11:30.12 | c0w | |15") in new stack |
| 11:30.19 | c0w | which then does this |
| 11:30.24 | HaMYaI | it's louder with gsm |
| 11:30.42 | c0w | does anyone know if asterisk can trip the special chars in the dialplan like a trim function? |
| 11:30.53 | c0w | or does anyone know of a quick and easy way to to it in C |
| 11:32.53 | florz | epvdw: @echo: Well, that's relatively unlikely to be the cause. Actually, it's not the cause of the echo anyway, it only could be the cause for the echo becoming audible. |
| 11:33.20 | florz | epvdw: do you have any clue where the echo originates? |
| 11:35.13 | florz | tzafrir: KPN somehow is "shutting down the d channel" every few minutes, and it's somehow possible to request them to not do that. I don't have any exact clue of what's going on there, either, though ... |
| 11:35.18 | epvdw | florz: I'm not sure, but I do hear myself echo when calling PSTN from my Speed Touch 2030 phone (through asterisk) |
| 11:36.17 | florz | epvdw: Ahh, IC, yeah, in that kind of setup that could be. So you have 4 BRI PSTN lines? |
| 11:36.22 | *** join/#asterisk grEvenX (n=even@pc107-130.ktv.no) |
| 11:36.40 | epvdw | I read somewhere that KPN is using 'EuroISDN+' wich is only used in The Netherlands.... as using DTMF for Caller-ID.... |
| 11:36.41 | florz | epvdw: The source is probably some analog connection at the remote end |
| 11:36.57 | benjk | what's EuroISDN+ |
| 11:36.58 | benjk | ? |
| 11:37.12 | epvdw | at the moment we have 3 BRI lines |
| 11:37.54 | florz | epvdw: Even though a "single card" or something won't help with that - it's only the EC that could help |
| 11:37.55 | epvdw | benjk: good question, apparently the cause of some compatibility issues with KPN/Asterisk |
| 11:38.16 | benjk | no way BRI will use DTMF for caller ID, it always comes over the D channel |
| 11:38.41 | epvdw | florz: yeah, I had de B410P running for a few days, but now it keeps giving kernel panics after 1 call |
| 11:39.08 | epvdw | florx: with the B410P echo was practically gone |
| 11:39.35 | *** join/#asterisk Bobcat991966 (n=chatzill@cpe-069-132-138-111.carolina.res.rr.com) |
| 11:39.52 | florz | epvdw: Yeah, but because of the EC, not because it's a single card or something |
| 11:40.54 | epvdw | florz: ok, but i have tried Asterisk EC to counter it to no avail |
| 11:41.37 | florz | epvdw: IC |
| 11:41.54 | epvdw | florz: anyway, because of the single board achitecture and EC, reduing IRQ conflicts, timings issues etc, I thought it was a good idea to go for the B410P.... unfortunately this is not so stable with mISDN (in my case) |
| 11:45.22 | kink0 | anyone ussing Page() ? |
| 11:45.42 | tzafrir | florz, so this is something that needs working around in chan_zap? |
| 11:45.59 | florz | tzafrir: If I knew =:-) |
| 11:46.23 | florz | tzafrir: I mean, if I knew what the problem actually is, maybe I could say :-) |
| 11:46.29 | kink0 | I got hangup when I use Page(Zap/g1/number1&Zap/g1/number2,d) while both, number1 and number2 are in conversation, even if I use Absolutetimeout(0) , what I am doing wrong ? |
| 11:46.39 | epvdw | FYI, we also had to make source change to q931.c in libpri in order to function correctly with KPN and Bri-stuffed |
| 11:49.28 | florz | epvdw: Timing in the few-millisecond-range is not an issue when you are switching to VoIP (and echo becomes audible anyway) anyway - so, basically, you either need short latency _or_ echo cancellation, but not both. |
| 11:50.55 | florz | epvdw: But you don't have any detiled information as to what the problem with KPN actually is, at the protocol level? |
| 11:51.04 | florz | detailed even |
| 11:51.06 | tzafrir | epvdw, I wonder what it would take to use that card with zaptel drivers |
| 11:51.15 | tzafrir | The horrors |
| 11:51.33 | DarKnesS_WolF | kink0: i tested page i got the same results |
| 11:51.46 | DarKnesS_WolF | kink0: after few sec both parts closed |
| 11:51.57 | DarKnesS_WolF | and few sec * crash du to losing mpg123 proc. |
| 11:52.07 | DarKnesS_WolF | kink0: 1.2.12.1 ? |
| 11:52.08 | kink0 | DarKnesS_WolF: what asterisk version are you ussing ? |
| 11:52.16 | kink0 | yes, same 1.2.12.1 |
| 11:52.19 | tzafrir | DarKnesS_WolF, don't use mpg123 |
| 11:52.46 | *** join/#asterisk docelmo (n=vircuser@c-69-138-91-104.hsd1.de.comcast.net) |
| 11:52.47 | kink0 | I am not ussing any mpg123, just Page , no any background or so |
| 11:53.04 | kink0 | DarKnesS_WolF: you got any message or just Hangup ? |
| 11:53.19 | DarKnesS_WolF | kink0: hangup i'm checking the logs. |
| 11:53.21 | DarKnesS_WolF | tzafrir: why ? |
| 11:53.25 | DarKnesS_WolF | what should i use native? |
| 11:53.57 | tzafrir | DarKnesS_WolF, Better performance on most cases (unless you have a heavy load of on-hold callers) |
| 11:54.11 | *** join/#asterisk joelsolanki (i=joelsola@202.160.161.94) |
| 11:54.36 | tzafrir | DarKnesS_WolF, and you can give the held caller the message from the start |
| 11:55.00 | tzafrir | DarKnesS_WolF, however it won't stream from remote servers |
| 11:55.08 | epvdw | florz: the problem with the cheap HFC + Bri-stuffed is continues line sync has to be enabled (otherwise connected hardware will only work for several minutes) because of the way S/T bus is activated/deactivated also see my post (http://sourceforge.net/forum/forum.php?thread_id=1437025&forum_id=420324) as irwinpoe with regard to the code change necessary |
| 11:55.26 | *** join/#asterisk mtaht4 (n=m@h-72-244-145-197.phlapafg.covad.net) |
| 11:55.51 | epvdw | florz: problem 2 ofcource now is the mISDN (which seems to work OK with KPN) and B410P ... |
| 11:56.07 | kink0 | DarKnesS_WolF: are you ussing real Zap or pseudo for meeting ? |
| 11:56.27 | kink0 | I am ussing real, even I set threecall=1 in zapata.conf |
| 11:56.56 | kink0 | I have also to add some priority like Dial, Wait, Park ... and so after the Page, but are ignored |
| 11:57.11 | *** join/#asterisk brif8 (n=brif8@ns1.ttienterprises.org) |
| 11:57.13 | DarKnesS_WolF | kink0: seems it's the same ... kink0 yes i do u have zap channels but i was using sip only in this page issue |
| 11:57.26 | *** part/#asterisk cpm (n=Chip@pdpc/supporter/active/cpm) |
| 11:57.32 | *** join/#asterisk zotz (n=zotz@24.244.163.225) |
| 11:57.42 | epvdw | florz: with the continues syn ON we receive messages like "received TEI check request for TEI = 96" al the time |
| 11:57.43 | DarKnesS_WolF | tzafrir: ok thx i'll try native method |
| 11:58.11 | epvdw | this is some sort of keep-alive |
| 11:58.14 | kink0 | well, I still researching , now time for lunch |
| 11:58.29 | kink0 | DarKnesS_WolF: if you discover something, please leave me a mesg |
| 11:58.47 | kink0 | see u later |
| 11:59.38 | DarKnesS_WolF | kink0: okay |
| 12:00.36 | florz | epvdw: Well, or you could try to make the zaptel EC work for you :-) |
| 12:01.10 | epvdw | florz: how would attempt to do that? |
| 12:01.13 | florz | epvdw: so, that "patch" solves the KPN problems? |
| 12:01.47 | florz | epvdw: Well, simply try out each and every available echo canceller with a huge tail length at forst |
| 12:01.53 | florz | s/forst/first/ |
| 12:02.13 | *** join/#asterisk jgoo (n=e4b80e21@foodtecsolutions.com) |
| 12:04.12 | epvdw | ok, I have tried the echocancel and echtraining settings, but never changed the default echo-canceller in asterisk source (but apparently the EC has been improved in the latest * builds) |
| 12:05.10 | epvdw | but the line-sync is becoming an big issue also, when installing * for customers.... |
| 12:07.36 | florz | epvdw: Well, maybe you should try to find out what the exact problem is and then fix it :-) |
| 12:07.38 | HarryR | Hmm, anybody got large fastagi applications? |
| 12:08.08 | *** join/#asterisk DanTMG (n=danielga@203-206-234-123.dyn.iinet.net.au) |
| 12:08.49 | florz | epvdw: I guess many KPN customers woul love to see a fix for that :-) |
| 12:10.49 | *** join/#asterisk jeffik (n=Jeff@dialup-4.247.245.203.Dial1.Tampa1.Level3.net) |
| 12:11.24 | *** join/#asterisk RoyK (n=roy@80.239.107.70) |
| 12:12.18 | *** join/#asterisk aiksa[LV] (n=aiksa[LV@83.223.131.104) |
| 12:12.42 | xnon | hello friends |
| 12:12.55 | xnon | i have a GrandStream Handy Tone 286 |
| 12:13.38 | *** join/#asterisk Modcuts (n=bob@lan.proporta.com) |
| 12:13.47 | RoyK | xnon: so nice for you |
| 12:13.57 | xnon | when other sip phone is calling to my phone HandyTone go to a voicemail option |
| 12:14.12 | xnon | and my phone cant ring |
| 12:14.51 | xnon | i think that my phone is down conection latter a few minutos or hours |
| 12:14.58 | xnon | in your activity |
| 12:15.42 | xnon | the status phone is ONLINE its ok, but in the call go to a voicemail option i dont understand! |
| 12:15.58 | xnon | RoyK: cant you help me? |
| 12:16.05 | xnon | sorry my english is not so god |
| 12:16.08 | xnon | good |
| 12:16.36 | RoyK | no. sorry. |
| 12:18.08 | xnon | for what is Register Expiration?????? |
| 12:18.20 | xnon | is in 3600 seconds! |
| 12:24.55 | *** join/#asterisk bjohnson (n=bjohnson@jecinc.tor.istop.com) |
| 12:34.49 | saftsack | does anyone of you use t38modem? |
| 12:35.53 | *** join/#asterisk [TK]D-Fender (n=joe@MTRLPQ02-1177745839.sdsl.bell.ca) |
| 12:37.17 | *** join/#asterisk coppice (n=chatzill@229.166.17.210.dyn.pacific.net.hk) |
| 12:37.48 | saftsack | RoyK, i dont ask here because i use * i ask here because i want to know if t38modem is stable ^^ |
| 12:38.16 | RoyK | I beleive t38modem is stable, yes, but I have no idea if it's stable with asterisk |
| 12:39.59 | sivana | anyone work with the Cisco 7960? |
| 12:40.29 | sivana | I just got one, first time, and the display only says "Configuring IP" |
| 12:42.57 | *** join/#asterisk jtexter3 (n=jtexter3@COX-66-210-197-34-static.coxinet.net) |
| 12:45.23 | tzanger | morning [TK]D-Fender, coppice |
| 12:45.52 | RoyK | sivana: that means Configuring Internet Pornography. See http://uncyclopedia.org/wiki/IP for more info |
| 12:47.08 | *** join/#asterisk stugster (n=Stuart@80-192-33-73.stb.ubr09.edin.blueyonder.co.uk) |
| 12:47.16 | stugster | anyone know where i can find complete newbie help? |
| 12:47.21 | coppice | i never heard of anyone getting t38modem to work with * |
| 12:47.42 | coppice | tzanger hi |
| 12:51.21 | *** join/#asterisk ambriento (n=cyndi@200-158-14-51.dsl.telesp.net.br) |
| 12:53.05 | [TK]D-Fender | *yawn* |
| 12:53.10 | [TK]D-Fender | mornin' |
| 12:53.45 | *** join/#asterisk tparcina (n=tparcina@lns01-2323.dsl.iskon.hr) |
| 12:54.11 | tparcina | i have problem with iax2 registration |
| 12:54.46 | tparcina | my * has two ip address, and both of them are defined as bindaddr in iax.conf |
| 12:54.57 | tparcina | one address is internal and another is external |
| 12:55.40 | [TK]D-Fender | tparcina: You should bind as 0.0.0.0. I'm not sure that if you do otherwise it will only bind to the LAST one mentioned. |
| 12:55.49 | tparcina | problem is when another asterisk, from public IP, tries to register with first *, first * returns internal ip address and not external |
| 12:56.08 | stugster | do i NEED a soundcard to get asterisk working? |
| 12:56.15 | *** join/#asterisk lowlevel (n=Stuart@bigbrother.vermeulens.com) |
| 12:56.22 | [TK]D-Fender | stugster: Not at all. |
| 12:56.22 | tparcina | in iax.conf.sample they say that * will use all which are defined |
| 12:56.25 | stugster | cool :D |
| 12:56.42 | [TK]D-Fender | stugster: You don't need almost ANYTHING, though I might suggest a NIC ;) |
| 12:57.09 | stugster | lol |
| 12:57.12 | stugster | i gots that;) |
| 12:57.19 | stugster | i've just installed it, so im off to read the manual |
| 12:57.27 | stugster | i need to set it up for voice activated prompts |
| 12:57.35 | tparcina | [TK]D-Fender: but, as you said, it can work even without nic :) |
| 12:58.19 | tparcina | [TK]D-Fender: is there any other way i can tell * to, for this specific user, use public (second) ip addres? |
| 12:58.59 | stugster | rather than getting a digital phone, can i connect modem and connect fone to that? |
| 12:59.15 | tparcina | [TK]D-Fender: all i need is that * replies using same IP addres (in header) as which the request has arrived |
| 12:59.38 | tparcina | stugster: you can't use modem |
| 12:59.48 | stugster | k |
| 13:00.10 | tparcina | bit you can use fxs ata ot figium tdm 400 card |
| 13:00.26 | [TK]D-Fender | stugster: Start with a soft-phone (X-Lite for Windows, Ekiga for Linux) |
| 13:00.54 | stugster | well ideally, i have a USB robotics phone |
| 13:00.56 | stugster | could i use that? |
| 13:01.13 | [TK]D-Fender | stugster: Then when you're ready You can look at getting either an ATA (a little box that will let you use a normal analog phone and will make it into a SIP device), or by a "hard" IP phone. |
| 13:01.28 | [TK]D-Fender | stugster: Depends on the precixes details of that "hone" |
| 13:01.33 | [TK]D-Fender | phone* |
| 13:01.39 | [TK]D-Fender | stugster: Whats the model #? |
| 13:01.42 | stugster | see, im just playin at the moment |
| 13:01.48 | stugster | E337 |
| 13:01.57 | [TK]D-Fender | stugster: Then jsut start with a soft-phone on your PC |
| 13:02.21 | stugster | well i dont know if i even want a phone |
| 13:02.36 | stugster | i want to start it out as an automatic system that simply keeps details of customers/faxes |
| 13:02.46 | stugster | *not faxes, ignore that bit |
| 13:02.48 | tparcina | anybody, how to bind iax to specific port for specific user? |
| 13:03.22 | ambriento | tparcina, I think you can't |
| 13:03.51 | [TK]D-Fender | stugster: Ok, that is a "handest" for use with a PC and is compatible with X-Tens softphones, so yes you can use it |
| 13:03.54 | ambriento | I think you can do it globally only |
| 13:04.23 | [TK]D-Fender | tparcina: Set the port in their peer/user entires |
| 13:04.31 | stugster | http://www.usr-emea.com/products/p-voip-product.asp?prod=voip-9600a&loc=unkg |
| 13:05.07 | [TK]D-Fender | stugster: I would only buy that product if you plan on using it with a laptop. |
| 13:06.02 | stugster | okay |
| 13:06.02 | stugster | as i say, i dont want to be accepting calls just now |
| 13:06.02 | tparcina | [TK]D-Fender> tparcina: Set the port in their peer/user entires - don't understand this |
| 13:06.02 | stugster | I want people to be able to get put through to my asterisk system, and then have options of where they leave a message |
| 13:06.02 | stugster | or hear pre-recorded details |
| 13:06.32 | *** join/#asterisk Ebola (i=1000@81-86-155-65.dsl.pipex.com) |
| 13:06.36 | tparcina | Fender - what did you mean by - Set the port in their peer/user entires |
| 13:08.13 | *** join/#asterisk earthsound (n=one_down@138.26.35.115) |
| 13:08.43 | tparcina | Fender: do you understand what is my problem? |
| 13:09.20 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 13:09.29 | ambriento | [TK]D-Fender, I don't think that is what tparcina needs |
| 13:10.07 | [TK]D-Fender | tparcina: put "port=ASTUPIDNUMBER" in the [peer] and [user] entries for the IAX account! |
| 13:10.18 | [TK]D-Fender | stugster: What would these calls be coming in on? |
| 13:10.33 | stugster | well, i'm working on the Tesco VOIP system |
| 13:10.37 | stugster | it would ALL be via VOIP |
| 13:10.50 | [TK]D-Fender | stugster: What protocol? |
| 13:10.59 | *** join/#asterisk Nebukadneza (n=daddel9@i3ED6FCDB.versanet.de) |
| 13:11.24 | stugster | erm |
| 13:11.40 | stugster | IAX |
| 13:11.46 | [TK]D-Fender | stugster: Ok, can you just give me a link to your providers website? |
| 13:13.17 | stugster | www.tescointernetphone.com :) |
| 13:14.49 | *** join/#asterisk Mercestes (n=Merceste@rrcs-71-41-157-70.sw.biz.rr.com) |
| 13:15.25 | coppice | what next? www.bandcallq.com? :-) |
| 13:16.11 | stugster | lol |
| 13:16.24 | stugster | well if you can find me another provider? |
| 13:17.29 | [TK]D-Fender | stugster: Ok they really don't say what protocol they use, but its likely SIP. If so then you don't need anything extra at all. If they are running a proprietay protocol, you are probably shit outta luck and should look for a new provider. |
| 13:17.34 | *** join/#asterisk puzzled (n=patrick@puzzled.xs4all.nl) |
| 13:17.49 | stugster | lolol |
| 13:17.50 | stugster | thanks :D |
| 13:17.50 | puzzled | morning |
| 13:18.38 | [TK]D-Fender | stugster: Look at these guys http://www.gradwell.co.uk/voip/ |
| 13:19.00 | [TK]D-Fender | stugster: Might be a good option. I don't know the market that well in your area, but a client of mine used them briefly. |
| 13:19.25 | *** join/#asterisk anthm (n=anthmct@CPE-69-76-83-52.wi.res.rr.com) |
| 13:19.25 | *** mode/#asterisk [+o anthm] by ChanServ |
| 13:20.47 | stugster | is there a way to find out the protocol that Tesco use? |
| 13:21.43 | stugster | you may like to know that Tesco VoIP is |
| 13:21.43 | stugster | >IAX2 protocol via a server 202.148.48.108 registered to an Australian |
| 13:21.43 | stugster | >provider. |
| 13:21.46 | *** join/#asterisk jgoo (n=e4b80e21@athedsl-118215.otenet.gr) |
| 13:23.23 | *** join/#asterisk McLazarus (n=mcallist@pool-72-78-32-106.phlapa.east.verizon.net) |
| 13:24.02 | ingenius | ok im update asterisk to last beta version of svn ... but i cant call to my other sip clients ... they are in diferents locations with nat ... im using sipura |
| 13:31.15 | c0w | exit |
| 13:35.52 | *** join/#asterisk Chris-NB (n=chris@ng1.kurtkrenn.com) |
| 13:37.09 | *** part/#asterisk earthsound (n=one_down@138.26.35.115) |
| 13:38.22 | *** join/#asterisk cfh (n=luca@82.193.23.3) |
| 13:39.42 | cfh | hi all, i have problem with Incorrect timing sync on a PRI to a digium TE210P in fact the FAX are: poor quality or failed |
| 13:39.46 | cfh | what can i do ? |
| 13:41.23 | *** join/#asterisk vhatz (n=Vlasis@194.219.121.194) |
| 13:41.33 | ambriento | by FAX you mean facsimile? |
| 13:42.01 | vhatz | hello all |
| 13:42.14 | cfh | FAX are machine that send fax with analog line |
| 13:42.30 | ambriento | cfh, got it |
| 13:42.39 | cfh | I have just config e server voip (* version 1.2.5) to a PRI telecom an to a PRI PBX , and i switch the call from external to internal by asterisk |
| 13:42.50 | cfh | all works good with the voice call, but when the call is a fax sometimes the hwfaxmachine riceve half fax or error |
| 13:43.47 | *** join/#asterisk KaZeR (n=kazer@102.204.98-84.rev.gaoland.net) |
| 13:43.49 | ambriento | cfh, so now you have TELCO<->*1.2.5<->LegacyPBX |
| 13:44.13 | cfh | yes |
| 13:44.56 | *** join/#asterisk stkn (n=stkn@gentoo/developer/pdpc.active.stkn) |
| 13:46.15 | cfh | i m tring to TELCO fax call<> * <>TDM400P <>fax machine |
| 13:47.18 | ambriento | well, digium says that they can't guarantee fax through their boards |
| 13:47.38 | ambriento | it works for some people |
| 13:48.10 | cfh | what can i do ? |
| 13:48.35 | coppice | worry :-) |
| 13:51.10 | X-Rob | ooh |
| 13:51.18 | X-Rob | coppice, just the person I wanted to speak to |
| 13:52.17 | jtexter3 | cfh: What does your dial statement look like to get to the legacy pbx? |
| 13:54.10 | stugster | hey |
| 13:54.21 | stugster | how do i go about setting up the voip tesco line to the asterisk software? |
| 13:54.57 | *** join/#asterisk FaithX (n=FaithX@203.13.37.13) |
| 13:55.59 | *** join/#asterisk uwe (n=uwe@dogbert.palnet.com) |
| 13:56.20 | cfh | jtexter3: sorry but i dont understand |
| 13:56.54 | uwe | hello, im trying to start asterisk, trying with -cvvvvvvvvvvvvv reaches Parsing zapata.conf and exits, i suppose this means there is a syntax error in that file, is there a way to debug it? |
| 13:57.16 | stugster | -vvvc ? |
| 13:57.56 | jtexter3 | cfh: So, a call comes in, you determine it needs to go to your fax machine, so you have a line that is something like: exten => _XXXX,n,Dial(Zap/1). What is the line that dials over to the fax machine? |
| 13:58.03 | uwe | well, yes , that too |
| 13:58.11 | jtexter3 | cfh: You can also pastebin the relevant section of your extensions.conf file |
| 13:58.18 | cfh | ok |
| 13:58.28 | [TK]D-Fender | cfh: If you have synch issues you are pretty dead all-around. You need to fix that. |
| 13:58.40 | uwe | um, well, i think i have an idea why now |
| 13:59.53 | uwe | ive just upgraded my kernel, and i rebuilt zaptel driver, but seems thats not enough to get asterisk working properly, i cant find modules starting with wc*, there sould be , no? |
| 14:00.12 | cfh | D-fender : ok but what can i do to fix that ? |
| 14:02.52 | *** join/#asterisk kink0 (n=k@161.pool62-37-205.static.uni2.es) |
| 14:02.56 | kink0 | hello |
| 14:03.29 | vhatz | hello kink0 |
| 14:03.45 | kink0 | anyone ussing Page() ? |
| 14:03.54 | *** join/#asterisk pifiu-laptop (n=someone@216.5.79.1) |
| 14:03.57 | *** join/#asterisk kristalino (n=kristali@LNeuilly-152-21-126-73.w193-253.abo.wanadoo.fr) |
| 14:04.09 | jtexter3 | kink0: I have a customer that makes extensive use of Page(). |
| 14:04.14 | kink0 | I still having a "little" problem with Page, calls are hanged just fews seconds after all parties are speaking. |
| 14:04.45 | cfh | jtexter3: http://paste.getlinuxhelp.org/5031 |
| 14:04.45 | stugster | anyone from the UK here? |
| 14:05.17 | kink0 | jtexter3, I setup fews extension lines for Page() , but calls hangs while people are speaking. do you use something to get the calls alive ? |
| 14:05.52 | *** join/#asterisk benjk (n=benjamin@f8a01-0357.din.or.jp) |
| 14:07.23 | kink0 | jtexter3, I have tryed seting a priority after Page() for Dial, Wait, Meeme, Park... but that doesn't work, the calls are terminated just on Page, so futher commands and apps are not executed. |
| 14:07.50 | jtexter3 | kink0: I'm using SIP phones, and have them auto answer |
| 14:08.18 | jtexter3 | kink0: The flow of execution is Answer -> AGI which calls PHP script to weed out phones in use -> Page -> Hangup |
| 14:08.33 | kink0 | jtexter3, my case is a bit different, we are ussing calls to the PSTN , to mobile phones. All phones rings and people can meeting while fews seconds. |
| 14:08.39 | *** join/#asterisk _ViperNetworks (n=Nitesh@65.48.63.178) |
| 14:08.52 | *** join/#asterisk Ebola (i=1000@81-86-155-65.dsl.pipex.com) |
| 14:09.01 | _ViperNetworks | Hello Everybody... |
| 14:09.28 | _ViperNetworks | Hows it going [TK]D-Fender |
| 14:09.39 | kink0 | jtexter3, I just use one line like Page(Zap/g1/number1&Zap/g1/number2...) , all phones rings ( where phones are mobile phones ) but after fews seconds are hanged. |
| 14:09.52 | jtexter3 | kink0: Okay. I can't say I've ever used page to the PSTN before. |
| 14:10.09 | _ViperNetworks | I did try the ChanSPY... but how can I tap into a active channel and listen the conversation. |
| 14:10.23 | RoyK | chan_terror |
| 14:10.27 | *** join/#asterisk Modcuts (n=bob@lan.proporta.com) |
| 14:11.17 | kink0 | jtexter3, hmmm, I will try just with SIP now, ussing another asterisk, and bind that SIP channels to route calls to the second asterisk. |
| 14:11.18 | _ViperNetworks | currently, its recording the calls.. but I need to tap into activate calls... any suggestion |
| 14:13.10 | ambriento | cfh, would you paste your zapata.conf ? |
| 14:14.57 | *** join/#asterisk marv[work] (n=timr@br0.asteriasgi.com) |
| 14:15.03 | jtexter3 | cfh: Which context are the fax calls coming in on? isdn-incoming? or pbx-incoming? |
| 14:15.30 | cfh | ambriento : zapata.conf -> http://paste.getlinuxhelp.org/5032 |
| 14:16.47 | cfh | jtexter3 : fax calls coming on ISDN-incoming |
| 14:17.07 | ambriento | while I'm reading it, how do you know you have sync problems? |
| 14:17.42 | jtexter3 | cfh: Remove the tT from the Dial statement. I had this problem with modems at a customer, and when I removed all w, W, t, and T's, it worked without issue |
| 14:17.55 | *** join/#asterisk freebsd_fan (n=ebola@catagiuri305.giuri.unige.it) |
| 14:18.01 | *** join/#asterisk mattchis (n=mattchis@216.54.143.242) |
| 14:18.07 | cfh | ambrimento : because i have read this on zaptel.conf.example : # Incorrect timing sync may cause clicks/noise in the audio, poor quality or failed# faxes, unreliable modem operation, and is a general all round bad thing. |
| 14:18.15 | *** join/#asterisk slobberknocker (n=slobberk@63.149.122.93) |
| 14:18.48 | *** join/#asterisk jcmoore (n=jcmoore@unaffiliated/tgrman) |
| 14:18.59 | ambriento | cfh, I see |
| 14:19.33 | slobberknocker | I am setting the callerid name so that it displays the "name" of the number that was called to the call center agent. That all works fine. But when I do this it changes the caller id information in my database to show only the name. I would like to also record the number that the caller called from. How could I do this? Here is what I have done: http://pastebin.ca/190948 |
| 14:21.03 | ambriento | cfh, check your logs to see if you're having LOS and other alarms on Telco's E1. then you can make sure you're having sync issues |
| 14:21.30 | *** join/#asterisk quid246 (n=quid24@CPE00131078ba5d-CM000f9f7eff1e.cpe.net.cable.rogers.com) |
| 14:22.27 | *** part/#asterisk kmilitzer (n=km@office-gw.westend.com) |
| 14:22.33 | cfh | ambriento: sometimes on cli are : !! Unknown IE 26 (cs6, Unknown Information Element) |
| 14:25.05 | *** join/#asterisk Druken (n=jdumais@CPE000854ddcdb1-CM00137189cb0c.cpe.net.cable.rogers.com) |
| 14:25.17 | ambriento | cfh, port1 of your TE210 is connected to your legacy pbx and port2 to telco, right? |
| 14:25.46 | cfh | <PROTECTED> |
| 14:25.58 | cfh | jtexter3: i have try with a ATA (SIPURA 1001) connetct to hwfaxmachine but the problem persist |
| 14:27.17 | ambriento | cfh, you should try to use span=1,0,0,ccs,hdb3,crc4 and span=2,1,0,ccs,hdb3,crc4 |
| 14:27.39 | cfh | why? |
| 14:27.41 | *** join/#asterisk wunderkin (i=kev@ip68-226-113-228.ph.ph.cox.net) |
| 14:28.41 | stugster | does anyone know any UK VOIP providers that work with Asterisk? |
| 14:29.30 | ambriento | cfh, since the port2 is connected to Telco line, Telco should provide you the clock |
| 14:30.09 | *** join/#asterisk vooduhal (n=vooduhal@tc-proxy2.catt.com) |
| 14:30.40 | vooduhal | Does '*' have any special function in app_queue in queues? We were using * from attended transfer but when in a queue it will disconnect the call. |
| 14:31.00 | *** join/#asterisk dwclarkNU (n=dwclark@208.231.72.129) |
| 14:31.56 | *** join/#asterisk anthonyl (i=anthony@nat/digium/x-8b1d1e6dcac0e6d5) |
| 14:32.22 | ambriento | vooduhal, you have a queue with some callers waiting. Then when you answer them, you want to "attend transfer"'em to another extension |
| 14:32.32 | ambriento | then the call gets disconnected? |
| 14:33.01 | vooduhal | ambriento, yes. |
| 14:33.04 | vooduhal | Yes. |
| 14:33.14 | vooduhal | You press '*' to transfer and the call ends. |
| 14:33.22 | vooduhal | If it's not a queue call, it works perfectly. |
| 14:33.51 | dwclarkNU | trying to configure asterisk to work with broadvoice, i have it behind a cisco pix 501 performing PAT from my single IP...i have the appropriate ports forwarded to asterisk, do i need to enable nat in my sip.conf file? |
| 14:34.08 | ambriento | I see |
| 14:34.20 | *** join/#asterisk quid246 (n=quid24@CPE00131078ba5d-CM000f9f7eff1e.cpe.net.cable.rogers.com) |
| 14:35.35 | *** join/#asterisk juanjoc (n=juanjoc@200.73.189.82) |
| 14:35.43 | ambriento | dwclarkNU, I guess you should |
| 14:36.31 | ambriento | vooduhal, how do you send the callers into the queue |
| 14:36.46 | ambriento | vooduhal, tell me your Queue() line |
| 14:37.18 | vooduhal | exten => s,n(try),Queue(${QUEUE}|t|||${TIMEOUT}) |
| 14:38.16 | ingenius | kink0, hi! |
| 14:38.31 | *** join/#asterisk klasstek (n=nunyobiz@ip67-154-143-190.z143-154-67.customer.algx.net) |
| 14:38.57 | ingenius | kink0, alive ? i have good news :) |
| 14:39.28 | ambriento | vooduhal, check with CLI> show features if you have disconnect mapped to * |
| 14:39.38 | vooduhal | I know Queue has the h option but as you can see, we're not using it. |
| 14:39.53 | ambriento | yeap, I saw that |
| 14:39.57 | vooduhal | ambriento, Great. Found it. |
| 14:40.04 | vooduhal | Pickup *8 *8 |
| 14:40.04 | vooduhal | Blind Transfer # # |
| 14:40.05 | vooduhal | Attended Transfer * |
| 14:40.05 | vooduhal | One Touch Monitor |
| 14:40.05 | vooduhal | Disconnect Call * * |
| 14:40.39 | ambriento | vooduhal, wow map it to another sequence, like *0 or something else |
| 14:40.51 | *** join/#asterisk luke-Jr_work (n=ubuntu@adsl-70-128-250-253.dsl.ksc2mo.swbell.net) |
| 14:41.05 | ambriento | and then reload res_featrures.so |
| 14:41.08 | *** join/#asterisk nvzn (n=nvzn@dormir.dreaming.org) |
| 14:41.16 | ambriento | oops, res_features.so* |
| 14:41.17 | luke-Jr_work | Will 'Goto' within a Macro lose the macro's state? |
| 14:41.27 | luke-Jr_work | eg, will it no longer return to the calling context? |
| 14:41.30 | *** join/#asterisk tsurk0 (n=tsurko@vpn-pppoe-213-240-242-237.megalan.bg) |
| 14:42.53 | vooduhal | Hmmm... How can I change the default? |
| 14:42.55 | vooduhal | Builtin Feature Default Current |
| 14:42.55 | vooduhal | --------------- ------- ------- |
| 14:42.56 | vooduhal | Pickup *8 *8 |
| 14:42.56 | vooduhal | Blind Transfer # # |
| 14:42.56 | vooduhal | Attended Transfer * |
| 14:42.56 | vooduhal | One Touch Monitor |
| 14:42.58 | vooduhal | Disconnect Call * ** |
| 14:43.15 | ambriento | you don't have to worry about the default |
| 14:43.22 | ambriento | the Current is the current :) |
| 14:43.55 | vooduhal | Still doing the same thing. |
| 14:44.13 | ambriento | and try not paste all those lines in here :) use pastebin |
| 14:44.14 | uwe | um, im trying to start asterisk with zaptel card ... its failing and im unable to figure out why, it was working yesterday, i just upgraded the kernel, zttool shows the cards, and it looks fine to me, but asterisk fails when it reaches the parsing of zapata.conf! any hints where i can start? |
| 14:44.19 | ambriento | ~pastebin |
| 14:44.25 | jbot | it has been said that pastebin is a place to paste your stuff without flooding the channel - try http://pastebin.com/ (BROKEN AND SUCKING NUTS), or http://pastebin.ca, or http://channels.debian.net/paste, or for #oe use http://oe.pastebin.com, or http://bzflag.pastebin.ca/, or http://paste.lisp.org/ for the lisp/scheme nuts, or http://paste-it.net |
| 14:45.01 | ambriento | vooduhal, pastebin your CLI> output when you're trying to transfer |
| 14:45.19 | ambriento | if possible, since the beginning of the call |
| 14:45.43 | vooduhal | No need. It's only one line |
| 14:45.45 | vooduhal | Spawn extension (dial-agent, 8156, 1) exited non-zero on 'Local/8156@dial-agent-4d23,2' |
| 14:45.55 | vooduhal | It's still just disconnecting the call. |
| 14:46.04 | vooduhal | verbosity is set to 100. :) |
| 14:47.38 | *** join/#asterisk ghenry (n=ghenry@mail.suretecsystems.com) |
| 14:48.12 | ambriento | vooduhal, good but I think the whole output would be more helpful |
| 14:48.26 | beyond | ambriento, firmeza |
| 14:48.29 | vooduhal | That's it. |
| 14:48.37 | vooduhal | There is no attempt to do anything. |
| 14:48.37 | ambriento | :) beyond |
| 14:48.43 | beyond | ambriento, smack |
| 14:48.45 | beyond | ;-) |
| 14:49.01 | [TK]D-Fender | uwu : if you upgraded your kernel you need to download all the source & headers for it and recompile Zaptel. |
| 14:49.17 | uwe | [TK]D-Fender, i did that |
| 14:49.45 | [TK]D-Fender | uwe : And recompiled zaptel? |
| 14:50.07 | uwe | yes, make clean, make , make install :) |
| 14:50.15 | brif8 | The Dial string (a) which are possible and (b) which is more correct DIAL(SIP/202@gateway/${EXTEN},20) or DIAL(SIP/gateway/{EXTEN},20) where gateway is a context [gateway] in sip.conf ? |
| 14:50.23 | *** join/#asterisk saftsack (n=oliver@p54A7C886.dip.t-dialin.net) |
| 14:50.38 | [TK]D-Fender | uwu : Pastebin the error |
| 14:50.56 | [TK]D-Fender | uwu : Along with "cat /proc/interrupts |
| 14:51.20 | saftsack | [TK]D-Fender, do you know nokias e60? |
| 14:51.35 | [TK]D-Fender | brif8: (b) is the way you should call it |
| 14:51.48 | [TK]D-Fender | saftsack: Nope. |
| 14:52.02 | saftsack | ok but do you have experiences with wifiphones at all? |
| 14:52.25 | [TK]D-Fender | vooduhal: Pastebin your dialplan (ALL of the related bits to the problem,)and the CLI output of a failed call at verbose 10 |
| 14:52.36 | *** join/#asterisk VijayG (i=vijay@203.122.28.121) |
| 14:52.56 | VijayG | hello everyone |
| 14:53.02 | ambriento | vooduhal, try map your attended transfer to another sequence, like *9 |
| 14:53.12 | ambriento | and let's see what happens |
| 14:53.15 | uwe | [TK]D-Fender, well, actually it just reaces == Parsing '/etc/asterisk/zapata.conf': Found and fails there, but i just booted into the older kernel and it failed there too, i dont know why :( |
| 14:53.31 | VijayG | i need to do the conferencing using asterisk, can anyone guide me on that |
| 14:53.52 | *** join/#asterisk sb_mx (n=sb_mx@200.78.229.18) |
| 14:53.57 | VijayG | whats the dialplan required for extensions.conf |
| 14:54.00 | VijayG | i am using sip trunks |
| 14:54.15 | VijayG | what is to be done to initiate conferencing |
| 14:54.38 | *** join/#asterisk xezz (n=dddsd@b-h7.ektheseis.otenet.gr) |
| 14:54.49 | VijayG | uwe: are you using any timing device?? |
| 14:54.57 | [TK]D-Fender | VijayG: http://www.gradwell.co.uk/voip/ |
| 14:55.05 | VijayG | Thanks |
| 14:55.20 | uwe | VijayG, not that i know!! |
| 14:55.32 | marcus2 | is there a decent voip provider that offers toll free DID origination? |
| 14:55.38 | [TK]D-Fender | vooduhal: Wait a set. you are using Zaptel DTMF transfers in Queue calls? * is the hangup key.... |
| 14:55.44 | [TK]D-Fender | sec* |
| 14:56.18 | *** join/#asterisk rpm (n=russell@S01060002b3d10d24.cg.shawcable.net) |
| 14:56.44 | vooduhal | Oh.... |
| 14:56.47 | vooduhal | Yes, I am. |
| 14:57.02 | vooduhal | Can I turn that off? |
| 14:58.14 | uwe | hmm, pastebin is taking forever |
| 14:58.47 | uwe | http://pastebin.com/800065 |
| 14:59.18 | brif8 | [TK]D-Fender Thanks |
| 14:59.42 | [TK]D-Fender | vooduhal: What are you using for phones? |
| 14:59.57 | vooduhal | Polycom 601s |
| 15:00.32 | [TK]D-Fender | vooduhal: Then simply stop using DTMF transfers and just use "normal" transfers |
| 15:00.44 | *** join/#asterisk svenna_ (n=svenna@p548D2539.dip0.t-ipconnect.de) |
| 15:00.46 | vooduhal | Doesn't work with app_queue and QueueMetrics. |
| 15:00.54 | [TK]D-Fender | uwu : Use pastebin.ca from now on.... |
| 15:01.26 | uwe | [TK]D-Fender, ok :) |
| 15:01.56 | kink0 | ingenius, hi !!! |
| 15:02.52 | uwe | hmm, well, i included zapata_additional.conf, and asterisk went past it ... and failed |
| 15:03.17 | uwe | i mean it parsed zapata.conf and parsed zapata_additonal.conf and failed |
| 15:03.55 | [TK]D-Fender | uwu : So you have 2 x TDM400P ? |
| 15:04.27 | uwe | i tried starting asterisk in strace , and the most meaningful thing i found was :ioctl(16, 0x40044a26, 0x7fffff88a76c) = -1 ENXIO (No such device or address) |
| 15:04.30 | [TK]D-Fender | vooduhal: Then you'll have to remap those features in features.conf |
| 15:04.56 | ingenius | kink0, look this -> Asterisk SVN-branch-1.4-r44240 built by root @ zifar on a i686 running Linux on 2006-10-04 10:16:33 UTC |
| 15:06.44 | *** join/#asterisk droops (n=root@adsl-074-245-001-031.sip.jan.bellsouth.net) |
| 15:06.48 | *** join/#asterisk VJG (i=vijay@203.122.28.121) |
| 15:06.57 | VJG | Hello |
| 15:06.59 | ambriento | bbl |
| 15:07.01 | *** join/#asterisk SplasPood (n=jwb@nat-loopback.lga4.us.corp.voxel.net) |
| 15:07.08 | VJG | how can i do conferencing through asterisk |
| 15:07.20 | VJG | can anyone give me the sample configuration for the same |
| 15:07.20 | uwe | yes |
| 15:07.25 | uwe | yes [TK]D-Fender |
| 15:07.40 | kink0 | VJG: I just did , ussing Page() |
| 15:07.48 | qdk | ~meetme |
| 15:08.04 | qdk | how is that bot used again? |
| 15:08.17 | [TK]D-Fender | uwe: So ztfg -vvvvv comes up ok? |
| 15:08.45 | luke-Jr_work | Is there a way to Goto outside of a macro but not disrupt the stack? |
| 15:09.36 | qdk | luke-Jr_work: redesign would probably work. |
| 15:09.59 | uwe | hmm, no , ZT_SPANCONFIG failed on span 1: Invalid argument (22) [TK]D-Fender , thank you, at least i have a hint now :) |
| 15:10.35 | [TK]D-Fender | VJG: http://www.gradwell.co.uk/voip/ |
| 15:10.42 | VJG | kink0: Can i get the sample configuration for that |
| 15:10.45 | uwe | [TK]D-Fender, but i have zaptel module loaded according to lsmod !! |
| 15:10.50 | [TK]D-Fender | VJG: http://www.voip-info.org/wiki/view/Asterisk+cmd+MeetMe |
| 15:10.59 | uwe | or do i need ztdummy too ? |
| 15:11.01 | [TK]D-Fender | VJG: 2ndd link |
| 15:11.03 | [TK]D-Fender | asdhjklhjaskjdhasd |
| 15:11.08 | VJG | ok |
| 15:11.12 | [TK]D-Fender | Stupid &^%@#%^&# IRC client... |
| 15:11.39 | [TK]D-Fender | uwu : I'd suggest redoing the modprobe for both |
| 15:11.41 | *** join/#asterisk _alex_mx_ (n=alex@200.78.229.18) |
| 15:11.48 | [TK]D-Fender | uwu : and pastebin your zaptel.conf |
| 15:11.50 | luke-Jr_work | qdk: ? |
| 15:11.58 | uwe | both zaptel and ztdummy ? |
| 15:12.58 | *** join/#asterisk sevard (n=sev@c-67-188-173-23.hsd1.ca.comcast.net) |
| 15:14.44 | uwe | [TK]D-Fender, http://pastebin.ca/190996 |
| 15:15.41 | [TK]D-Fender | uwu : Why do I see a T1/E1 span definition in there?! |
| 15:16.23 | mut | burn in from your monitor tk |
| 15:16.28 | mut | its GOTTA be |
| 15:16.49 | mut | least thats always my bofh excuse |
| 15:16.51 | [TK]D-Fender | uwu : You only said you had 2 TDM400P's, not a T1 card...... |
| 15:16.52 | uwe | hmm |
| 15:16.57 | mut | "why isn't my login working" |
| 15:17.07 | mut | "well your monitor burned in the login screen so you can't see it" |
| 15:17.14 | [TK]D-Fender | uwu : Unless you've been seriously holding back on me, that doesn't belong there. Comment it out. |
| 15:17.22 | uwe | well, i have this, according to lspci: Network controller: Tiger Jet Network Inc. Tiger3XX Modem/ISDN interface |
| 15:17.31 | quid246 | Hmm... is there a way to prevent * from binding to all interfaces, but only to some (more than one)? |
| 15:17.41 | quid246 | For SIP that is. |
| 15:17.49 | sevard | mut: Somebody needs to write a new calander of BOFH excuses, ones that actually work in 2006 |
| 15:18.28 | sevard | things like "umm, the tape drive ate your data" just don't work anymore. |
| 15:18.55 | mut | heh yea |
| 15:18.57 | quid246 | I'm still using "the punch cards got mangled" |
| 15:19.03 | uwe | the whole line, im commenting the whole line |
| 15:19.03 | sevard | hahaha |
| 15:19.30 | uwe | ok, now modprobe worked |
| 15:19.34 | uwe | :D |
| 15:19.40 | quid246 | Programming by punch cards was probably actually fun in it's day |
| 15:19.42 | vooduhal | Question guys. We're getting " chan_zap.c:8383 pri_dchannel: Ring requested on channel 0/3 already in use on span 1. Hanging up owner." on one of our zap channels that is no longer connected but soft hangup won't do anything and any call that comes over that channel gets a reorder. |
| 15:19.53 | vooduhal | Is there anything else I can do to close the channel? |
| 15:19.59 | sevard | quid246: Yes, if you were insane. |
| 15:20.33 | quid246 | Kind of like playing lotto thousands of times over and over |
| 15:20.42 | uwe | but asterisk still fails [TK]D-Fender |
| 15:20.44 | sevard | my point exactly |
| 15:21.01 | uwe | oh |
| 15:21.15 | uwe | wait, modprobe zaptel fails!! |
| 15:21.27 | uwe | no, it doesnt ... sorry sorry |
| 15:21.36 | sevard | uwe: Hire a consultant |
| 15:21.48 | sevard | There are many here willing to help you for a reasonable fee. |
| 15:21.56 | uwe | ok, now everything works |
| 15:22.47 | uwe | :) sevard , well, im willing to learn, and im trying hard |
| 15:23.04 | quid246 | putting your foot in the wate ris better than paying someone to do it for you |
| 15:23.10 | vooduhal | Anyone have any clue as to how to reset this channel? "chan_zap.c:8383 pri_dchannel: Ring requested on channel 0/3 already in use on span 1. Hanging up owner." |
| 15:23.31 | uwe | thank you very much [TK]D-Fender :), ill now try it again on the newer kernel |
| 15:23.35 | quid246 | voodoo: "stop now" :) |
| 15:23.45 | sevard | quid246: Agreed, but if you can't probe a module... you might want to pay a consultant. |
| 15:23.59 | vooduhal | That won't work so well. :) |
| 15:24.01 | olivier__ | vooduhal> stop and ztcfg . Ive same problem sometimes :/ |
| 15:24.04 | quid246 | sevard: Or learn more about Linux. |
| 15:24.05 | vooduhal | 20 active calls. |
| 15:24.19 | quid246 | All consultants wera t Uwe's point of learnign at one time. |
| 15:25.35 | [TK]D-Fender | quid246: Depends |
| 15:25.43 | uwe | [TK]D-Fender, the last thing i did before it worked was modprobe wctdm .. which i cant see why it used to load and didnt this time... im awfully sorry if i did something before last kernel upgrade that i wasnt aware of and left you with the impression that something broke newly |
| 15:25.53 | uwe | thank you all again :) |
| 15:25.55 | vooduhal | olivier__, no other way to reset it? |
| 15:26.03 | [TK]D-Fender | quid246: Sometimes you're best to have someone who really knows what they're doing set it up so you can learn the right way to do it so your NEXT attempt can be solo. |
| 15:26.13 | [TK]D-Fender | uwe: Quite welcome |
| 15:26.17 | vooduhal | What about zap destroy? |
| 15:26.28 | [TK]D-Fender | vooduhal: AVOID |
| 15:26.41 | [TK]D-Fender | vooduhal: It will trash the channel until you completely restart * |
| 15:26.46 | vooduhal | Is there anything else I can do? We're getting quite a few calls on that channel. |
| 15:27.31 | *** join/#asterisk blitzrage (n=blitzrag@asterisk/documenteur-extraordinaire/blitzrage) |
| 15:27.46 | file | blitzrage: what'cha want me to say?!? |
| 15:28.59 | *** join/#asterisk Amilcar_ (n=xxxxx@201.34.202.17) |
| 15:29.41 | vooduhal | Nothing else I can do? |
| 15:31.08 | *** join/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net) |
| 15:31.08 | *** mode/#asterisk [+o mog] by ChanServ |
| 15:32.37 | *** join/#asterisk jeremy_g (n=jerms@static-213-115-44-90.sme.bredbandsbolaget.se) |
| 15:37.45 | luke-Jr_work | um |
| 15:37.58 | luke-Jr_work | is it just me, or is AEL1's switch unnestable? |
| 15:38.01 | jeremy_g | hi |
| 15:38.11 | jeremy_g | what the fuck is this sipflow product |
| 15:38.44 | jeremy_g | it claims to read tcpdump logs to generate call flow graphs |
| 15:39.17 | quid246 | Sounds cheesy. And can the unnecessayr profanity. |
| 15:39.27 | vhatz | jeremy_g: I think Ethereal is doing this already... |
| 15:39.30 | jeremy_g | but when i shuv the tcpdump -w calllog at my asterisk box, it doesnt show anything |
| 15:39.46 | jeremy_g | vhatz:yeah i found out |
| 15:39.50 | sevard | i, on the other hand, enjoy his unnecessary profanity. |
| 15:40.22 | quid246 | well I haven't used tCPdump much... but I assume it only dumps TCP... and since most sip is UPD, you wont see jack |
| 15:40.24 | jeremy_g | vhatz:how do you save traffic logs using tcpdump in pcap format for ethereal to read them |
| 15:40.40 | jeremy_g | does the simple tcpdump -w shit works |
| 15:40.51 | *** join/#asterisk Ham_Ster (n=Rick@crm.telmatik.com) |
| 15:40.51 | jeremy_g | quid246:wtf, proto udp clearly supported |
| 15:40.58 | luke-Jr_work | quid246: it dumps anything |
| 15:41.03 | quid246 | like I said, I don't use it |
| 15:41.06 | luke-Jr_work | jeremy_g: the name implies otherwise |
| 15:41.46 | Ham_Ster | hello, is there anyone who played with the MYSQL function in the asterisk-addons ? I'm trying to do a query based on a incoming call number |
| 15:42.02 | vhatz | quid246: tcpdump will dump any packet, not just tcp... |
| 15:42.02 | Ham_Ster | and each time I'm getting error that the connection failed |
| 15:42.16 | [TK]D-Fender | LOL, TrixBox got bought! |
| 15:42.17 | vhatz | jeremy_g: you use the -w switch to specify the filename |
| 15:42.18 | jeremy_g | vhatz:yeah thats whats it for |
| 15:42.23 | jeremy_g | yeah |
| 15:42.37 | *** join/#asterisk brodiem (n=brodiem@67.110.68.66.ptr.us.xo.net) |
| 15:42.42 | jeremy_g | but it doesnt work |
| 15:42.46 | jeremy_g | it isnt saving it in pcap format |
| 15:42.47 | vhatz | jeremy_g: for ex: tcpdump -s 65535 -i eth0 -w /tmp/capture.cap will write /tmp/capture.cap file |
| 15:42.51 | jeremy_g | wtf is this pcap format |
| 15:42.58 | sevard | [TK]D-Fender: Somebody bought that POS? |
| 15:43.07 | jeremy_g | vhatz:lemme try this -s this |
| 15:43.24 | [TK]D-Fender | sevard: http://www.fonality.com/press.html |
| 15:43.34 | vhatz | jeremy_g: -s affects the size of the captured packet... it''s best to allow a large value |
| 15:44.01 | quid246 | Trix |
| 15:44.01 | quid246 | NS2-Netaddress: 66.96.218.70 |
| 15:44.05 | quid246 | Oops |
| 15:44.14 | sevard | [TK]D-Fender: I'd just rather the project die |
| 15:44.18 | jeremy_g | vhatz:u have no idea how grateful i am to you |
| 15:44.20 | sevard | but, to each his own. |
| 15:44.26 | jeremy_g | vhatz:thanks mate |
| 15:44.28 | jeremy_g | it worked |
| 15:44.34 | quid246 | TrixBox was interesting when it came out... but like an MS product, it keeps bloating |
| 15:44.36 | vhatz | jeremy_g: you're welcome, anytime |
| 15:44.39 | vhatz | jeremy_g: :) |
| 15:44.51 | jeremy_g | i totally forgot this -s option, the man page is long |
| 15:44.55 | sevard | quid246: It's "unbuntu" for Asterisk |
| 15:45.03 | quid246 | haha |
| 15:45.05 | sevard | I can't wait to see an unbuntu/trixbox release |
| 15:45.19 | quid246 | Yeha, you'll need 2 DVDs for it |
| 15:45.24 | sevard | hahaha. |
| 15:45.29 | [TK]D-Fender | The problem with all these "idiot-proof" setups people keep developing is well.. all the damn idiots if keeps attracting! |
| 15:45.38 | quid246 | DVD-9's |
| 15:45.39 | quid246 | haha |
| 15:45.50 | brodiem | I'm trying to find some info on setting ANI. The wiki doesn't seem to have much... |
| 15:45.51 | sevard | Unbuntu, the NEW Mandrake. TrixBox, the NEW AMP. |
| 15:46.02 | brodiem | ~ani |
| 15:46.04 | jbot | extra, extra, read all about it, ani is Automatic Number Identification Systems |
| 15:46.15 | sevard | These Two Projects Come Together To Create: Sh!tSystem |
| 15:46.25 | sevard | The ultimate in crappy software. |
| 15:47.10 | [TK]D-Fender | I especially "love" the ones who keep thinking its an "Asterisk" bug when they don't fill in essential fields in a product that actually IS pretty well documented. I swear I'm going to slowly become as jaded as RoyK and start spouting "RTFM" to people before long.... |
| 15:47.30 | sevard | [TK]D-Fender: you should switch your name to all caps. |
| 15:47.35 | sevard | s/name/nick/g |
| 15:47.45 | quid246 | There is/was a guy in #trixbox who wrote a book for newbs.... |
| 15:51.34 | *** join/#asterisk AJaymn (i=AJaymn@68-115-17-167.dhcp.mdsn.wi.charter.com) |
| 15:52.09 | AJaymn | When a call comes in one of my SIP trunks the callerid Name is correct, but the # shows up at the trunks # and not the callers.. How do I fix that? |
| 15:53.27 | *** join/#asterisk DonX (i=don@the.lostserver.net) |
| 15:54.09 | DonX | Can someone tell me if this is possible? |
| 15:54.10 | DonX | exten => _6XXXXX,1,AGI(sitenum.agi|${EXTEN}) |
| 15:54.10 | DonX | exten => _6XXXXX,2,Goto(site${SITENUMBER},${SHORTNUMBER},1) |
| 15:55.12 | DonX | sitnum agi parses ${EXTEN} but I can't get asterisk to retreive the values of $SITENUMBER and $SHORTNUMBER |
| 15:55.40 | vhatz | bye all |
| 15:55.41 | vhatz | :) |
| 15:56.27 | *** part/#asterisk vhatz (n=Vlasis@194.219.121.194) |
| 15:57.36 | *** join/#asterisk Ham_Ster (n=Rick@crm.telmatik.com) |
| 15:57.36 | Ham_Ster | damn power failures! |
| 15:57.45 | *** part/#asterisk slobberknocker (n=slobberk@63.149.122.93) |
| 15:58.48 | Ham_Ster | so is there anyone who has experience with the MySQL addon from asterisk-addons ? |
| 15:59.11 | Ham_Ster | i tried the config on voip-info.org but still getting errors |
| 16:00.02 | *** join/#asterisk Ox0F0-0FF (n=pierre@201.38.238.66) |
| 16:02.36 | *** join/#asterisk qdk (n=qdk@0x535eae17.boanxx9.adsl-dhcp.tele.dk) |
| 16:03.35 | *** join/#asterisk hmmhesays (n=hmmhesay@24-117-135-28.cpe.cableone.net) |
| 16:03.59 | *** join/#asterisk pigpen (n=mark@fw.seamans.cc) |
| 16:04.25 | pigpen | Could someone tell me if this error is a "Slip Error" : Got reject for frame 74, but we only have others! |
| 16:04.42 | pigpen | ^^^coming in over zap...passing to IAX. |
| 16:05.20 | *** join/#asterisk oQPa (i=Spain@237.Red-83-44-33.dynamicIP.rima-tde.net) |
| 16:05.44 | pigpen | Sorry... PRI Slip Error... |
| 16:06.21 | jtexter3 | Anybody here ever seen a backtrace like this when asterisk dies with an abort: http://pastebin.ca/191037 |
| 16:06.44 | *** join/#asterisk dovid (n=dovi5988@barak.cellcom.co.il) |
| 16:07.17 | *** join/#asterisk MattH (n=MattH@isengard.chilitech.net) |
| 16:07.18 | pigpen | jtexter3, yuck. Good luck on that... |
| 16:07.30 | pigpen | I will take my pri slip errors over that anyday. |
| 16:07.57 | MattH | Hi.. I have an issue.. i have 2 systems connected via sip.. I can send calls from system B to system A.. however when I try to send a call from system A to system B system B says it can't authenticate the call and fails authenticating the CALLER-ID of the caller rather then the username specified.. any ideas? |
| 16:08.56 | jtexter3 | pigpen: This is the bane of my existence! I'm working on a custom module, and when testing it, asterisk crashes in random places. Always with an abort, always with a similiar, useless backtrace. I think it may actually be related to this bug: http://bugs.digium.com/view.php?id=8013 |
| 16:09.49 | pigpen | jtexter3, hmm..sounds like something is stepping on somethings dick somewhere... |
| 16:10.11 | pigpen | what OS and Kern? |
| 16:10.30 | pigpen | and gcc/glibc ver? |
| 16:11.12 | dovid | ~centosbug |
| 16:11.14 | jbot | from memory, centosbug is a problem with the latest Centos kernels (4.2 and 4.3). To fix it, paste everything inside the quotes into a root shell: "sed -i s/rw_lock/rwlock/ /usr/src/kernels/*/include/linux/spinlock.h" |
| 16:11.47 | blitzrage | file: how about something like "Y'ALL GONNA LOVE THESE ASTERISK HOCKEY JERSEYS" :) |
| 16:12.19 | RoyK | [TK]D-Fender: Repeate The First Message |
| 16:12.34 | [TK]D-Fender | RoyK: What first message? |
| 16:12.41 | RoyK | RTFM :) |
| 16:12.43 | [TK]D-Fender | RoyK: NVM |
| 16:13.27 | jtexter3 | pigpen: CentOS 4.4, glibc 2.3.4-2.25 |
| 16:13.33 | hmmhesays | hahaha |
| 16:13.38 | jtexter3 | dovid: I'll do that now |
| 16:13.40 | file | blitzrage: :D |
| 16:13.47 | hmmhesays | there should be an app that reads a floating point number as a monetary value |
| 16:14.09 | jtexter3 | dovid: I've gotten past that, that just prevents zaptel from compiling |
| 16:14.10 | dovid | huh ? |
| 16:14.27 | [TK]D-Fender | RoyK: Really Tiresome Fucking Moron. |
| 16:14.36 | dovid | i did that just cause i needed the fix. sorry for the mix up. didnt even see ur question |
| 16:14.42 | *** join/#asterisk TESTER2 (n=Cyber@modemcable082.42-81-70.mc.videotron.ca) |
| 16:15.01 | TESTER2 | someone knows if chan_bluetooth is still maintain somewhere? |
| 16:15.10 | jtexter3 | no worries. This has been killing me, so I'm open to just about anything |
| 16:15.27 | [TK]D-Fender | RoyK: Reinstall the ^%$#ing Machine? |
| 16:15.29 | jtexter3 | pigpen: Also, kernel == 2.6.9-42.0.2.ELsmp |
| 16:16.14 | RoyK | [TK]D-Fender: http://karlsbakk.net/fun/ugly-hint.txt |
| 16:17.17 | [TK]D-Fender | RoyK: MEAN |
| 16:17.21 | pigpen | jtexter3, well, I am sorry (and happy) to say, that I don't know the latest info on the Centos kernel bugs. Looks out of date to me as I run Gentoo 2.6.17 Hardened Sources on 64bit Xeon with glibc 2.3.6-r5 |
| 16:17.43 | pigpen | So you may want to see the info regarding centos via dovid... |
| 16:18.13 | dovid | im no centOS guru. just know a bit |
| 16:19.22 | pigpen | dovid, admit it..you are a Centos god. |
| 16:19.28 | [TK]D-Fender | Hey, does CentOS 4.4 have the same Kernel source bug as the older versions, or is there anything new that needs to be taken into account for it? |
| 16:20.12 | dovid | eh. i wish i was |
| 16:20.12 | pigpen | Sorry I don't have much info..I am sheltered by a Gentoo Kernel Dev.... |
| 16:20.28 | _alex_mx_ | have 1.2 and 1.4 running on CentOS 4.4 don't remember any problems compiling either one |
| 16:20.33 | dovid | TK: not that i know of |
| 16:20.44 | dovid | i installed ast on 4.4 with no issues |
| 16:21.00 | pigpen | yeah..but I think he is working with a custom module... |
| 16:21.20 | pigpen | jtexter3 that is... |
| 16:22.00 | dovid | im no progreammer so i am out with that |
| 16:22.18 | dovid | i just go on the devel. list and throw money at the hungry ones |
| 16:22.35 | *** part/#asterisk cfh (n=luca@82.193.23.3) |
| 16:22.40 | pigpen | yeah..same here... |
| 16:23.43 | dovid | and then have em add it to bug tracker so the rest of the world can enjoy. i dont get people that have cust work done and dont commit |
| 16:23.57 | dovid | unless its a biz idea or a product that they wana sell |
| 16:24.09 | jtexter3 | pigpen: We have been using Business Edition. Digium now has a custom linux distro, so I'm downloading that to try it |
| 16:24.15 | *** join/#asterisk Ebola (i=1000@81-86-155-65.dsl.pipex.com) |
| 16:24.16 | pigpen | or greedy with their code... |
| 16:24.30 | pigpen | jtexter3, ah...good luck... |
| 16:24.40 | dovid | jtexter3: y the biz edition ? |
| 16:24.52 | pigpen | If I even think of a different distro than Gentoo, I get hit with a stick. |
| 16:25.23 | jtexter3 | Ah, Gentoo: complete with the gcc screen saver :D |
| 16:25.24 | dovid | lol |
| 16:25.29 | pigpen | but I must admit...all is running great... |
| 16:25.38 | dovid | pigpen: is it the powers above u ? |
| 16:25.40 | pigpen | jtexter3, you know it. |
| 16:25.49 | *** join/#asterisk saftsack (n=saftsack@p54A7C87D.dip.t-dialin.net) |
| 16:26.39 | pigpen | dovid, na...I have a business partner that is a Gentoo Kern dev.....a -HUGE- blessing I might add... |
| 16:26.57 | *** join/#asterisk foxxtrot (n=craig@c-67-185-55-194.hsd1.wa.comcast.net) |
| 16:26.58 | jtexter3 | Either of you guys going to Astricon? |
| 16:27.02 | *** join/#asterisk vader-- (n=johndoe@204.183.88.101) |
| 16:27.10 | pigpen | na..I am too poor...and busy... |
| 16:28.00 | *** join/#asterisk Dude34 (n=Aces1UP@209.101.89.82) |
| 16:32.27 | *** join/#asterisk ESCulapio_ (n=ESCulapi@200.88.44.66) |
| 16:33.19 | *** join/#asterisk Mercestes (n=Merceste@rrcs-71-41-157-70.sw.biz.rr.com) |
| 16:43.25 | *** join/#asterisk websae (n=websae@adsl-64-149-206-121.dsl.milwwi.sbcglobal.net) |
| 16:43.52 | blitzrage | <-- going to astricon |
| 16:44.29 | dovid | lucky u |
| 16:45.24 | file | cause I'm leaving on a jetplane |
| 16:45.29 | file | pretty sure that I'll be back again |
| 16:46.42 | xezz | <PROTECTED> |
| 16:47.17 | carrar | http://ln.ooz.net/27723 |
| 16:48.56 | hmmhesays | file: we do a punk cover of that song |
| 16:48.58 | hmmhesays | I know all the words |
| 16:49.13 | file | neat |
| 16:49.35 | hmmhesays | all my bags are packed and i'm ready to go, i'm standing here outside your door, I hate to wake you up to say goodbye |
| 16:50.06 | hmmhesays | but the dawn is breaking it's early morn, taxi's waiting its blowing its horn, already i'm so lonely I could diiiieeee |
| 16:52.17 | carrar | cool linux based phone: http://www.geekzone.co.nz/content.asp?ContentId=6731 |
| 16:52.25 | carrar | does sip |
| 16:56.00 | *** part/#asterisk TESTER2 (n=Cyber@modemcable082.42-81-70.mc.videotron.ca) |
| 17:08.34 | *** part/#asterisk brif8 (n=brif8@ns1.ttienterprises.org) |
| 17:08.42 | [TK]D-Fender | xezz: Do you own one, or are you looking for people's opinion on it? |
| 17:09.22 | [TK]D-Fender | hmmhesays: "Don't quit your day job" |
| 17:09.23 | [TK]D-Fender | ;) |
| 17:10.12 | *** join/#asterisk ethzer0 (n=torsztyn@204.244.71.19) |
| 17:10.29 | *** join/#asterisk ToTo (n=ToTo@host138-138-dynamic.2-87-r.retail.telecomitalia.it) |
| 17:12.14 | coppice | Leaving on a jet plane |
| 17:12.16 | coppice | Don't know when I'll be back again |
| 17:12.17 | coppice | Oh, babe I hate those flight delays |
| 17:12.38 | blitzrage | ethzer0: w0rd to y0 mama |
| 17:13.08 | *** join/#asterisk pfn (n=pfnguyen@netblock-66-245-252-239.dslextreme.com) |
| 17:13.09 | coppice | carrar: what's cool about it? it looks like a piece of overpriced junk |
| 17:13.34 | *** join/#asterisk rene1 (n=rene1@gea-gye-internet.telconet.net) |
| 17:13.40 | *** join/#asterisk inv_arp[work] (i=junya@c-71-206-88-100.hsd1.fl.comcast.net) |
| 17:14.12 | rene1 | ~seen ^HeLL^ |
| 17:14.32 | jbot | ^hell^ <n=admin@85.137.123.5> was last seen on IRC in channel #asterisk, 20h 41m 21s ago, saying: 'oliviert: tdm400?'. |
| 17:15.08 | RoyK | ~lart himself |
| 17:17.12 | RoyK | is there a 1.4 svn branch? |
| 17:17.25 | blitzrage | yes |
| 17:17.40 | blitzrage | svn.digium.com/svn/asterisk/branches/1.4/ |
| 17:18.10 | file | [TK]D-Fender: run around in circles! |
| 17:18.22 | blitzrage | RoyK: you didn't even bother to look did you? |
| 17:18.39 | *** join/#asterisk Defraz (n=t0tal@71.39.210.81) |
| 17:18.49 | *** join/#asterisk foxxtrot (n=craig@c-67-185-55-194.hsd1.wa.comcast.net) |
| 17:18.58 | file | [TK]D-Fender: yay |
| 17:19.12 | RoyK | blitzrage: for some reason, http://asterisk.org/download doesn't tell |
| 17:19.13 | [TK]D-Fender | \o/ |
| 17:19.41 | [TK]D-Fender | file: Hows progress from Beta to gold going? |
| 17:19.55 | file | [TK]D-Fender: bah |
| 17:20.02 | [TK]D-Fender | file: humbug? |
| 17:20.02 | file | that is all. |
| 17:20.08 | RoyK | [TK]D-Fender: you mean to alpha? |
| 17:20.22 | [TK]D-Fender | RoyK: "says" beta.... |
| 17:21.27 | coppice | stop press: release candidate scores zero votes in key by-election :-) |
| 17:23.02 | *** join/#asterisk DataCompBoy (n=datacomp@gw6.lvs.ru) |
| 17:23.04 | RoyK | 1.0.7 was stable too |
| 17:23.06 | HarryR | what kinda issues were you having with later ones? |
| 17:23.07 | DataCompBoy | Hi |
| 17:23.34 | RoyK | HarryR: first there was an app_queue reload crash, and the latest one is this http://bugs.digium.com/view.php?id=8087 |
| 17:23.49 | coppice | I get weird reports from anything after 1.2.9 about NULL frames being thrown at rxfax, which cause a crash when they are freed |
| 17:23.52 | DataCompBoy | Anybody can describe me, how to monitor status of dialing in console?! :/ looks like sometimes it won't jump to prio+101 even if I have "j" set |
| 17:24.25 | DrkShdw | anyone know what the vonage gateway is, off hand? (for trunk registry) |
| 17:24.28 | DataCompBoy | also, hangup event on LOCAL/...,1 always "Unknown", even if on LOCAL/....,2 is known :( |
| 17:25.30 | marcus2 | so can anyone here recommmend a asterisk-friendly toll-free DID provider? |
| 17:25.32 | *** part/#asterisk dwclarkNU (n=dwclark@208.231.72.129) |
| 17:25.56 | luke-Jr_work | DataCompBoy: prio+101 is obsolete, you shouldn't be using it anyway |
| 17:25.57 | marcus2 | i've already gotten busy circuits twice when i've tried to call my gafachi did |
| 17:26.33 | wmandra | mornint all |
| 17:26.41 | wmandra | sp^ morning |
| 17:27.17 | DataCompBoy | luke-Jr_work: i'm originating calls as AMI Originate command to Local/out, and need to get status of such calls. On success origination, AGI() called, and on +101 called DeadAGI. |
| 17:27.24 | *** join/#asterisk Ryushin (i=proxy@windwalker.openinnovations.com) |
| 17:27.33 | DataCompBoy | luke-Jr_work: or there other way to get error status for dead originations? |
| 17:27.42 | rene1 | i want those too |
| 17:27.48 | luke-Jr_work | switch(${DIALSTATUS}) { |
| 17:27.48 | luke-Jr_work | case BUSY: |
| 17:27.53 | luke-Jr_work | like that DataCompBoy |
| 17:28.01 | rene1 | in the context the originate command connects? |
| 17:28.15 | Ryushin | I'm having a issue, that I'm sure is due to my config, that when a user transfers someone to a different extension, the person that is being transferred doesn't hear it ring. |
| 17:28.26 | luke-Jr_work | dead origination goes to another extension altogether, IIRC |
| 17:29.19 | [TK]D-Fender | luke-jr_ : You must be like the only person out there using AEL........ |
| 17:29.32 | luke-Jr_work | [TK]D-Fender: maybe, but I use it for 100% of my dialplans |
| 17:29.32 | rene1 | luke-Jr-work: but where? i sometimes see the originate failed ami events. however the proxy i am using (RAMI) does not pick them (dont ask why). but if i can get to work this via the dialplan the better |
| 17:29.43 | luke-Jr_work | rene1: set verbose 2000 |
| 17:29.49 | rene1 | ok |
| 17:30.23 | luke-Jr_work | [TK]D-Fender: and I have a very nice macro routing system setup until Asterisk decided to impose its 7-macro-depth-limit on me |
| 17:30.36 | DataCompBoy | luke-Jr_work: Stop. I'm use only .conf, not .ael :) |
| 17:31.05 | luke-Jr_work | DataCompBoy: DIALSTATUS variable should still exist |
| 17:31.23 | [TK]D-Fender | DataCompBoy: Indeed. Time to use GotoIf....... |
| 17:31.31 | luke-Jr_work | even my last .conf used it |
| 17:37.08 | *** join/#asterisk diclophis-work (n=jbardin@65.203.37.58) |
| 17:38.36 | *** join/#asterisk brif8 (n=brif8@ns1.ttienterprises.org) |
| 17:39.21 | DonX | anyone here good with AGI?? |
| 17:40.15 | *** join/#asterisk anarcat (n=anarcat@pdpc/supporter/active/anarcat) |
| 17:40.19 | anarcat | hello |
| 17:40.48 | anarcat | is there a simple solution that would allow me to call an extension, hangup and have asterisk call me back, and that would require external script or messing with the call queue? |
| 17:41.25 | anarcat | i tried this: http://pastebin.ca/191111 |
| 17:41.40 | diclophis-work | DonX... what are you wanting to do? |
| 17:42.13 | anarcat | ... but the above bridges the call with myself and if i hangup, the call just terminates. i tried using the parked calls too |
| 17:42.44 | rene1 | luke-Jr_work: should i use a local channel as the outgoing channel in originate to be able to get a dialstatus |
| 17:42.45 | anarcat | is this the only solution: http://www.voip-info.org/wiki/view/Asterisk+tips+callback ? |
| 17:43.18 | rene1 | diclophis: hey |
| 17:43.25 | diclophis-work | rene1: y0 |
| 17:43.41 | DataCompBoy | luke-Jr_work: ok, look at this, please: http://pastebin.ca/191116 |
| 17:44.04 | diclophis-work | anarcat: what do you want the callback to do? |
| 17:44.15 | DataCompBoy | luke-Jr_work: as you can see, no info about fail! :( event NoOp() after Dial() not called |
| 17:44.40 | anarcat | diclophis-work: i want it to call a certain number and throw it in the default context |
| 17:45.11 | diclophis-work | oh.. check it, look for the G option to the Dial command |
| 17:45.26 | DonX | diclophis-work, when an extension range is dialed I want it to parse out the extension and send it to a certain context |
| 17:46.09 | anarcat | diclophis-work: g or G? |
| 17:46.17 | diclophis-work | there is a 'goto' option |
| 17:46.19 | diclophis-work | i think its G |
| 17:46.29 | anarcat | G(context^exten^pri): If the call is answered, transfer both parties to the specified context and extension. The calling party is transferred to priority x, and the called party to priority x+1. This allows the dialplan to distinguish between the calling and called legs of the call (new in v1.2). |
| 17:46.54 | anarcat | i feel my problem is that the originating channel is not "right"... |
| 17:46.56 | diclophis-work | yea |
| 17:47.02 | DonX | ie, someone from 680125 calls 610333 I want it to do exten => _6XXXXX,2,Goto(site1,10333,1) |
| 17:47.36 | rene1 | DataCompBoy: i have the same prob |
| 17:48.23 | diclophis-work | well that will work... you have ex,1,Dial(NUMBER,timeout,G(context,exten,2)) then ex,2,1,Hangup(), then ex,3,1,Playback(pbx-invalid) |
| 17:48.23 | rene1 | i want to get DialStatus for AMI originated calls |
| 17:48.27 | [TK]D-Fender | anarcat: You can't use "h" with "HANGUP". What you'll want to do is make a small bash script to drop a ".call" file to have it dial you back. Pretty easy stuff |
| 17:48.31 | DataCompBoy | As far, as I see -- there no info even on SIP/vf-out-b681e448 call! while sip show peer vf-out shows messages that from "100 Trying" it changed state to "183 Session progress"... |
| 17:48.31 | diclophis-work | that will dial your number back, and play a sound |
| 17:49.20 | DonX | I'm writing a perl script to do it but I'm unable to figure out what I need to do to get the variables back into extensions.conf |
| 17:49.25 | diclophis-work | DonX: i dont think you need an AGI for that... look into the string functions you can do on variales... i think its possible to do ${EXTEN:-1} or something to strip the first number out |
| 17:50.05 | anarcat | [TK]D-Fender: just what i thought... okay, thanks |
| 17:50.11 | DataCompBoy | rene1: I just want to get any info! anything! I'm got nothing :( nowhere :( even on new dialed channels, looks like asterisk just won't track info :/ |
| 17:50.13 | DonX | cool, I'll check that out |
| 17:50.14 | DonX | thanks |
| 17:51.01 | diclophis-work | <PROTECTED> |
| 17:51.02 | DataCompBoy | may be I have misconfigured something? |
| 17:51.06 | rene1 | may be |
| 17:51.21 | DataCompBoy | rene1: if i' |
| 17:51.21 | rene1 | does your sip peer has qualify=yes? |
| 17:51.28 | DataCompBoy | rene1: yes. |
| 17:51.39 | DataCompBoy | rene1: if i'm just power off my cellphone, then I'm got such: |
| 17:52.33 | diclophis-work | [TK]D-Fender: the h extension doesnt get executed if the dial plan has a Hangup in it |
| 17:52.34 | diclophis-work | ? |
| 17:52.48 | [TK]D-Fender | diclophis-work: Correct. |
| 17:52.53 | diclophis-work | interesting... |
| 17:52.55 | [TK]D-Fender | diclophis-work: Only works when the CALLER hangs up |
| 17:53.34 | *** join/#asterisk teknoprep (i=teknopre@unaffiliated/teknoprep) |
| 17:53.46 | DataCompBoy | rene1: http://pastebin.ca/191124 |
| 17:53.48 | diclophis-work | then how do you do stuff after a call is done? |
| 17:53.52 | diclophis-work | lets say from AGI? |
| 17:53.56 | teknoprep | hi all.. having a problem.. i don't get an idicator when i have a voicemail on my phones... what makes a phone get the indicator? |
| 17:54.48 | DataCompBoy | rene1: So, second leg (Local/1@ps_out-bcea,2) got status (Call Rejected), but leg Local/1@ps_out-bcea,1 -- not. |
| 17:54.54 | *** join/#asterisk SomeJ (n=jasonh@12.19.114.51) |
| 17:55.21 | DataCompBoy | rene1: and this doesn't describe why I don't get status about "ringing" and don't get status if after ringing it was dropped :( |
| 17:55.23 | rene1 | DataCompBoy: but nothing in ${DIALSTATUS} |
| 17:55.43 | DataCompBoy | -- Executing NoOp("Local/1@ps_out-bcea,2", "CONGESTION") in new stack |
| 17:55.59 | rene1 | there you go |
| 17:56.04 | rene1 | that is your dialstatus |
| 17:56.04 | DataCompBoy | rene1: in ${DIALSTATUS} -- "CONGESTION". |
| 17:56.14 | rene1 | what do you want to do? |
| 17:56.44 | *** join/#asterisk ikey1 (n=ikey@203.115.29.2) |
| 17:56.54 | rene1 | you can use it to route your call: http://www.voip-info.org/wiki-Asterisk+variable+DIALSTATUS |
| 17:57.14 | DataCompBoy | rene1: problem if I'm not answer (i.e. phone ringing for some time) and then decline call -- then nothing executed after Dial() ! |
| 17:57.29 | DataCompBoy | rene1: not DeadAGI(), not NoOp() -- nothing! and no status in messages |
| 17:57.58 | rene1 | right now you have your cellphone down right? |
| 17:58.08 | rene1 | and you are calling it via a sip itsp? |
| 17:58.34 | DataCompBoy | Yes |
| 17:59.21 | rene1 | if you dont answer the call you should be getting a noanswer dialstatus |
| 17:59.47 | rene1 | and you say you are getting nothing, is that correct? |
| 18:00.03 | [TK]D-Fender | teknoprep: Depends on the Tech |
| 18:00.17 | DataCompBoy | Yup. The only I get -- is Hangup on Local/1@ps_out-blabla,1 leg. |
| 18:00.30 | rene1 | turn sip debugging |
| 18:00.30 | DataCompBoy | rene1: No messages for ,2 or @vf-out legs! |
| 18:00.38 | DataCompBoy | already. |
| 18:01.03 | *** join/#asterisk toxap (n=toxap@213.227.193.75) |
| 18:01.58 | *** join/#asterisk luchshiy (n=anonymou@d212-53-104-193.cust.tele2.ch) |
| 18:02.11 | DataCompBoy | rene1: moment, i'll pastebin what i'm got in such thing... |
| 18:02.48 | rene1 | in your first pastebin: there are originate failure events |
| 18:03.46 | DataCompBoy | rene1: yes, without any status info! there no info about moment of ringing start, and nothing executed after Dial() command... |
| 18:05.15 | *** join/#asterisk tsurk0 (n=tsurko@vpn-pppoe-213-240-242-237.megalan.bg) |
| 18:06.22 | rene1 | have you tried with a h exten in ps_out_pstn or pst_out_sip? |
| 18:09.04 | *** part/#asterisk jmls (n=asterisk@62.49.235.130) |
| 18:09.39 | DataCompBoy | rene1: http://pastebin.ca/191128 <- here is SIP DEBUG for call |
| 18:09.45 | DataCompBoy | rene1: moment, will try h exten |
| 18:11.32 | rene1 | there is the 487 signal from your gateway |
| 18:11.53 | rene1 | i am not sip expert but seems reasonable 4XX are errors |
| 18:11.55 | *** part/#asterisk brif8 (n=brif8@ns1.ttienterprises.org) |
| 18:12.21 | DataCompBoy | rene1: yes, there... but where reaction from asterisk?! this is a problem!!! |
| 18:12.38 | *** part/#asterisk pablus (n=nn@test.conama.cl) |
| 18:12.43 | DataCompBoy | ooooO!!!!! If i'm add exten -> h,1,NoOp(${..}) --- i'm got -- Executing NoOp("Local/1@ps_out-1008,2", "CANCEL") in new stack |
| 18:12.44 | *** join/#asterisk Alric (n=nbowyer@w-69-148-123-207.stu.swau.edu) |
| 18:12.49 | DataCompBoy | wow! that is what i'm needed: ) |
| 18:13.15 | rene1 | good |
| 18:13.22 | DataCompBoy | Lot of thanks, rene! :) |
| 18:13.27 | Alric | Anyone else seen issues with 1.4beta2 not recognizing ODBC installed on the system (and thus not installing the asterisk/ODBC stuff)? |
| 18:13.27 | rene1 | np |
| 18:13.46 | rene1 | where did you end up placing it? in ps_out_sip? |
| 18:14.47 | rene1 | i have the same problem as you i need to get origination failures but i dont want to get messy with AMI |
| 18:14.50 | DataCompBoy | so, there only one problem left: is possible to get moments of "100 Trying" and "183 Session Progress", and "??? Ringing" -- that really needed for tracking |
| 18:15.04 | rene1 | DataCompBoy: i dont think so |
| 18:15.08 | Alric | Hmmm. I wonder if it has something to do with it being the x86_64 version of unixODBC... |
| 18:15.20 | rene1 | you would need something different |
| 18:15.26 | rene1 | probably doable with SER |
| 18:16.00 | DataCompBoy | rene1: added to ps_out_pstn & ps_out_sip -- so i'm get failures for both cases |
| 18:16.37 | rene1 | good |
| 18:17.08 | rene1 | DataCompBoy: it probably can be done via manager api and newexten or channelstatus |
| 18:17.30 | rene1 | but in the dialplan i dont think so |
| 18:17.51 | DataCompBoy | eh, bad. i was think that AMI messages "newstatus" must contains that changes |
| 18:18.47 | DataCompBoy | rene1: what your problem with AMI? |
| 18:19.19 | *** join/#asterisk pablus (n=nn@test.conama.cl) |
| 18:19.53 | rene1 | mmi am using a library to make interaction easy withami but it doesnt show originate failures so i need local channels and dialstatus like you? |
| 18:20.13 | rene1 | DataCompBoy: yes i think newstatus might have those messages |
| 18:20.32 | rene1 | but calling ami from an AGI? or how would you get those |
| 18:24.13 | *** join/#asterisk oliviert (n=olivier@lap81-1-82-245-68-209.fbx.proxad.net) |
| 18:24.21 | oliviert | hi guys |
| 18:24.51 | oliviert | is one of you did test asterisk with the VOIP PHONES THOMSON 2030 ? |
| 18:26.10 | rene1 | i am also interested in those |
| 18:27.18 | [TK]D-Fender | olivier Specs look ok on it but I'd suggest you look at Polycom or Aastra instead. more well known high quality phones. |
| 18:28.24 | pablus | oliviert i get Sjphone and also i'm searching tester |
| 18:28.33 | pablus | my asterisk server |
| 18:28.38 | pigpen | Oct 4 13:22:56 NOTICE[16954]: chan_zap.c:8194 pri_dchannel: PRI got event: HDLC Abort (6) on Primary D-channel of span 1 |
| 18:28.47 | pablus | if you want talking |
| 18:28.49 | pigpen | ^^^^anyone have any thought? I am thinking slip errorrs... |
| 18:29.02 | oliviert | pablus: is sjphone easy to use ? |
| 18:29.11 | pablus | oliviert: yes |
| 18:29.27 | oliviert | pablus: good quality voice? no cut ? |
| 18:29.29 | pablus | oliviert: if you want let's go to test |
| 18:29.45 | oliviert | well, the voip are in my office |
| 18:29.50 | pablus | oliviert: internet level i never test it |
| 18:30.02 | oliviert | for here i do have to install a softphone |
| 18:30.06 | pablus | yeah in my office too |
| 18:30.12 | oliviert | or if you can call on skype ... |
| 18:30.13 | pablus | but my server is in my house |
| 18:31.00 | pablus | skype i don't have |
| 18:31.00 | oliviert | pablus: let's say we do make a test in about 20mins |
| 18:31.10 | pablus | ok |
| 18:31.21 | pablus | oliviert do you have gtalk |
| 18:31.23 | pablus | ?? |
| 18:36.07 | _alex_mx_ | pigpen, could be slips could also be irq misses |
| 18:36.24 | pigpen | yeah..just read that on the mailing list... |
| 18:37.00 | pigpen | this box is also a mail server...maybe spam/av is being hit hard.... |
| 18:40.11 | _alex_mx_ | pigpen, cat /proc/interrupts and paste |
| 18:40.17 | *** join/#asterisk KaZeR (n=kazer@3.204.98-84.rev.gaoland.net) |
| 18:41.26 | pigpen | http://pastebin.ca/191161 |
| 18:42.30 | _alex_mx_ | dual proc or ht? |
| 18:42.33 | *** join/#asterisk DataCompBoy (n=datacomp@212.17.0.67) |
| 18:43.00 | DataCompBoy | rene1: sorry, disconnect:)); so, what your problem with AMI? |
| 18:43.51 | *** join/#asterisk funkmaster (n=funky1@vhe-364501.sshn.net) |
| 18:45.49 | pigpen | _alex_mx_, It is a 64bit Xeon...and HT support is compiled in. |
| 18:46.16 | pigpen | I want to say it is a single "real" proc |
| 18:49.01 | Alric | Just tried installing 1.4beta2 and found it didn't recognize ODBC and install the ODBC components. Tried 1.2.12.1 and it recognized ODBC just fine. Anyone else have this problem? |
| 18:49.49 | *** join/#asterisk KranZ (n=user@sme.bestline.net) |
| 18:50.33 | *** join/#asterisk mtgh (n=chatzill@dsl093-001-038.det1.dsl.speakeasy.net) |
| 18:50.41 | pigpen | sorry...I haven't had time... |
| 18:51.07 | *** join/#asterisk mattchis (n=mattchis@216.54.143.242) |
| 18:51.51 | anarcat | okay, so i got a callback half-working, but when asterisk is calling me back, i can't enter new extensions |
| 18:51.55 | anarcat | this is my setup: http://wiki.koumbit.net/VoipService/CallBack |
| 18:53.36 | funkmaster | hey ppl, i got an asterisk server and a grandstream 486 box, i configured several phone numbers and i receive incoming calls and can also make outgoing calls, but when ppl call, although my phone rings, they don't hear it ringing, i guess somehitng is wrong in my extensions.conf can some1 take a look and help me please? |
| 18:53.40 | funkmaster | http://pastebin.ca/191169 |
| 18:53.45 | *** join/#asterisk aptura (n=none@S010600a0c93f6f7e.vs.shawcable.net) |
| 18:53.52 | funkmaster | this is my sip.conf and extensions.conf |
| 18:56.39 | *** join/#asterisk paolob (n=donpaolo@pri-214-b7.codetel.net.do) |
| 18:56.53 | *** part/#asterisk paolob (n=donpaolo@pri-214-b7.codetel.net.do) |
| 18:58.07 | anarcat | funkmaster: might be a firewalling problem |
| 18:58.18 | anarcat | tcpdump to see if RTP traffic is going through |
| 18:58.29 | anarcat | i fixed my entering extensions problem, btw |
| 19:00.04 | pigpen | can I stop asterisk from resetting my spans? |
| 19:00.06 | funkmaster | anarcat, i don't think so, cuz it used to work befoer |
| 19:00.13 | pigpen | ie : pri? |
| 19:00.50 | anarcat | now, of course, i have another problem: when I enter the extension i want, my IAX provider refuses the call |
| 19:01.04 | anarcat | whereas it works, normally, when i "just call" |
| 19:01.05 | anarcat | <PROTECTED> |
| 19:01.08 | anarcat | <PROTECTED> |
| 19:02.07 | *** join/#asterisk Mercestes (n=Merceste@rrcs-71-41-157-70.sw.biz.rr.com) |
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| 19:12.29 | *** join/#asterisk mtgh (n=chatzill@dsl093-001-038.det1.dsl.speakeasy.net) |
| 19:12.45 | mtgh | What codec is the best if bandwidth is not an issue....GSM just sounds kinda choppy |
| 19:12.58 | *** part/#asterisk anarcat (n=anarcat@pdpc/supporter/active/anarcat) |
| 19:14.02 | *** join/#asterisk arkangelp (n=arkangel@200.14.68.52) |
| 19:14.03 | *** join/#asterisk DRoBeR (n=DRoBeR@212.145.81.31) |
| 19:14.03 | DRoBeR | Hello |
| 19:14.03 | arkangelp | Hello everyone |
| 19:14.17 | [TK]D-Fender | mtgh: ULAW/ALAW |
| 19:16.27 | _alex_mx_ | pigpen, in zapata.conf with resetinterval=never |
| 19:16.28 | mtgh | which one? |
| 19:16.35 | arkangelp | it´s a simple question, it might be a silly question but I want to know if i´m install asterisk on a server to works as a pbx.. can I get rid of my regular telephone line? |
| 19:17.00 | arkangelp | or I still need a main phone line to work with asterisk? |
| 19:17.44 | arkangelp | like a dedicate line to work with the server where asterisk will run? |
| 19:17.52 | [TK]D-Fender | arkangelp: * can connect to all kinds of things. regular phone lines, ISDN, POTS, T1, VoIP protocols of all kinds. So yes you could swith over to an entirely VoIP terminated solution if you wanted to |
| 19:18.30 | pigpen | _alex_mx_, thanks... |
| 19:19.08 | arkangelp | so , sorry if i insist I don´t need a dedicate line or any sorf of land line to work with asterisk.. just the internet services to get dial tone |
| 19:19.38 | arkangelp | and what about the phone numbers who assing those new numbers on my office? i´m really new in this |
| 19:20.10 | x86 | mtgh: ulaw and alaw are the same codec, just one is big endian and the other is little endian.... they are both uncompressed, and they are both G.711 |
| 19:20.20 | [TK]D-Fender | arkangelp: You can sign up with a VoIP provider to get PSTN service. Who you would choose depends on where you are located. |
| 19:20.37 | x86 | mtgh: you could use either ulaw or alaw and not be able to tell a difference, because there is no difference ;) |
| 19:20.46 | DRoBeR | Hello |
| 19:20.58 | x86 | arkangelp: check out http://voip.shellshark.net/ |
| 19:22.20 | arkangelp | mmm I see, ok the reason that i´m asking these question is because right now i´m working in Valapraiso University which is located in Chile, south america.. and the university wants to switch from a regular phone services to VoIP services |
| 19:22.23 | x86 | mtgh: keep in mind that after IP, UDP, and RTP overhead, you're going to need about 80kbps per call, while using G.711 (either ulaw or alaw) |
| 19:23.02 | *** join/#asterisk luchshiy (n=anonymou@d212-53-104-193.cust.tele2.ch) |
| 19:23.03 | x86 | arkangelp: well they can surely save lots of money by upgrading :) |
| 19:23.27 | [TK]D-Fender | arkangelp: Well like I said you can use * with any kind of technology you want (norma stuff that is) |
| 19:23.29 | x86 | DRoBeR: heya :) |
| 19:23.43 | *** join/#asterisk Op3r (n=Op3r@61.28.130.145) |
| 19:23.53 | arkangelp | yes, thats what I wnat, in chile there is now VoIP service yet.. I used to live in USA and I used to have Vonage the VoIP company |
| 19:23.59 | mtgh | x86: Thanks....teliax dos not like ulaw or alaw.....but gsm does not sound good |
| 19:24.11 | x86 | mtgh: dont use teliax then ;) |
| 19:24.22 | x86 | mtgh: check out http://voip.shellshark.net/ |
| 19:24.23 | Op3r | does anyone knows how to transfer the recordings from /var/spool/asterisk/monitor to another server? |
| 19:24.35 | x86 | mtgh: G.729 also sounds decent, and uses far less bandwidth |
| 19:24.42 | x86 | Op3r: scp |
| 19:25.05 | mtgh | x86: Is that the one you have to pay for? |
| 19:25.34 | Op3r | x86, automatically like every 1 hour or something? can you point me on some howtos? |
| 19:25.47 | mtgh | op3r: check out rsync |
| 19:25.58 | x86 | mtgh: a lot of people recommend them, and i use them personally, yes |
| 19:26.25 | mtgh | https://voip.shellshark.net/cgi-bin/signup.pl |
| 19:26.25 | x86 | Op3r: yeah rsync would be better for doing automatic backups |
| 19:26.30 | mtgh | I would love to use them....but ... |
| 19:26.37 | mtgh | them seem to have some simple perl errors |
| 19:27.09 | Op3r | mtgh, ok thanks |
| 19:28.55 | *** join/#asterisk hi365 (n=hi365@bzq-167-158.dsl.bezeqint.net) |
| 19:30.04 | *** part/#asterisk DRoBeR (n=DRoBeR@212.145.81.31) |
| 19:30.14 | *** join/#asterisk X-Rob_ (n=rob-x@dsl-202-173-151-24.qld.westnet.com.au) |
| 19:30.41 | *** join/#asterisk r0d3nt (n=RatMan@foster.stonedcoder.org) |
| 19:30.41 | hi365 | where do i set the launguge for the "transfer" command? sip.config has language=he and the vm pormpts are he, but the transfer command is still english! |
| 19:31.27 | [TK]D-Fender | hi365: Transfer doesn't have a language... it just ejects the call from your server....... |
| 19:31.46 | x86 | mtgh: it looks like they have fixed it |
| 19:31.50 | hi365 | i meant the promts then u initiate a transfer (# or whatever) |
| 19:32.32 | *** join/#asterisk luchshiy (n=anonymou@d212-53-104-193.cust.tele2.ch) |
| 19:32.58 | hi365 | [TK]D-Fender: i meant the promts then u initiate a transfer (# or whatever) |
| 19:34.17 | *** join/#asterisk luchshiy (n=anonymou@d212-53-104-193.cust.tele2.ch) |
| 19:34.45 | [TK]D-Fender | hi365: Ok, that is not a command, that is a channel function (which should never be used anyways) and should support multiple languages like every other prompt out there. this mean the trnasferring channel's language should be the one it tries to use to prompt them. |
| 19:35.05 | mtgh | x86, they don't have a plan that works for me....thats too bad |
| 19:35.59 | x86 | mtgh: what are you looking for? |
| 19:36.01 | hi365 | [TK]D-Fender: so if im using a SIP phone and the sip.conf has launuge set to "he" then it should be in he? |
| 19:37.37 | mtgh | I need no monthly fees....something like nufone/voicejet/voicepulse/texliax |
| 19:37.47 | x86 | ah i see |
| 19:37.51 | *** join/#asterisk hads (n=hads@mail.nice.net.nz) |
| 19:38.10 | mtgh | (I am with UofM (in Michigan) and we are testing about 10 different vendors |
| 19:38.18 | x86 | mtgh: yeah they dont do that, but they do have TRUE unlimited plans to the US and Canada |
| 19:38.29 | x86 | mtgh: makes sense :) |
| 19:38.30 | mtgh | x86, for one channel at a time? |
| 19:38.47 | mtgh | anyone else have good recomendations other then what I said above? |
| 19:38.53 | x86 | mtgh: residential has one channel, yes, business they offer any channel limit you need |
| 19:39.17 | x86 | quit |
| 19:39.21 | x86 | err wrong window ;) |
| 19:39.37 | [TK]D-Fender | hi365: Correct |
| 19:40.02 | _alex_mx_ | hi365, /var/lib/asterisk/sounds/transfer.wav (.gsm .alaw .ulaw .g729) need to exist in /var/lib/asterisk/sounds/he |
| 19:40.55 | hi365 | _alex_mx_: its here! thats what weird |
| 19:42.49 | [TK]D-Fender | hi365: show us your phone's setup, and the CLI output of a call before, during and after the completion of a transfer |
| 19:44.59 | hi365 | [TK]D-Fender: http://pastebin.ca/191214 phone is grandtream 2000 |
| 19:47.26 | hi365 | [TK]D-Fender: sounds under: /var/lib/asterisk/sounds/he http://pastebin.ca/191216 |
| 19:48.01 | _alex_mx_ | hi365, Playing 'pbx-transfer' (language 'en' |
| 19:48.12 | _alex_mx_ | hi365, missed one :P |
| 19:48.52 | hi365 | _alex_mx_: right. but y? sip.conf language=he |
| 19:49.43 | *** join/#asterisk teknoprep (i=teknopre@unaffiliated/teknoprep) |
| 19:49.52 | teknoprep | hey [TK]D-Fender whats up |
| 19:50.08 | _alex_mx_ | you need pbx-transfer.whatever in your language you don't have it |
| 19:50.16 | teknoprep | [TK]D-Fender, now how are voicemail indication sent to a sip phone ? |
| 19:51.29 | hi365 | _alex_mx_: added pbx-tansfer.gsm but it didnt work |
| 19:52.13 | *** join/#asterisk Death_INC (n=sam@pdpc/supporter/sustaining/sbingner) |
| 19:52.27 | hi365 | _alex_mx_: and the (language 'en') its not even looking for the he file |
| 19:53.15 | _alex_mx_ | hi365, in multi-language systems asterisk will first look in .../sounds/he and if it does not find the file it will always default to english |
| 19:53.39 | hi365 | k. now the file is there. y wont it take it? |
| 19:53.52 | _alex_mx_ | owned by root perhaps_ |
| 19:54.02 | *** join/#asterisk xnon (n=xnon@200.8.87.1) |
| 19:54.40 | hi365 | _alex_mx_ 0777 |
| 19:56.54 | _alex_mx_ | hi365, let me think a sec we had the same problem with just that file and i'm trying to remember... |
| 19:59.27 | hi365 | can it be a problem cause its .gsm? |
| 20:00.51 | *** join/#asterisk Ryushin (i=proxy@windwalker.openinnovations.com) |
| 20:01.21 | *** join/#asterisk beyond (n=beyond@200.192.160.100) |
| 20:02.12 | _alex_mx_ | shouldn't be all your other files are gsm |
| 20:02.33 | hi365 | right. so what the heck??? |
| 20:04.59 | hi365 | can i pm u an email address if u thionk of something? |
| 20:05.08 | *** join/#asterisk QbY (n=Kelvin@cm-64-221-172-192.dhcp.southerncoastalcable.net) |
| 20:05.28 | tzafrir | hi365, where do you have he sound files from, BTW? |
| 20:05.57 | hi365 | ah........ asterisk.org.il. there crap though |
| 20:06.11 | hi365 | they claim theill have better soon. |
| 20:06.17 | hi365 | du u have any other options? |
| 20:06.24 | tzafrir | no |
| 20:06.27 | _alex_mx_ | shot in the dark...are you calling yourself? |
| 20:06.38 | tzafrir | Not only crap, but crap without a clear license |
| 20:06.40 | *** join/#asterisk krondorl (n=krondorl@207.245.14.10) |
| 20:07.00 | hi365 | well if i downloaded it leaggaly its really not my problem |
| 20:07.24 | krondorl | Hi all. |
| 20:07.27 | hi365 | _alex_mx_: no from ext 202 to ext. 201 |
| 20:08.11 | hi365 | tzafrir: then again: do u have better? i think for now its the best there is |
| 20:08.19 | hi365 | even for $$$$$$$$ i havnt found anything |
| 20:08.25 | _alex_mx_ | hi365, your're going through a queue correct? |
| 20:08.25 | tzafrir | No. I don't have anything better |
| 20:08.36 | tzafrir | For $$: contact Tikal |
| 20:08.36 | hi365 | yup |
| 20:08.40 | hi365 | 301 |
| 20:08.49 | hi365 | transfer to q 301 |
| 20:08.59 | _alex_mx_ | try phone to phone and transfer to the queue |
| 20:09.08 | hi365 | tzafrir: whatsw ur affiliation |
| 20:09.09 | hi365 | ? |
| 20:09.27 | tzafrir | Have you verified that the LANGUAGE is 'he' before playing the sound? |
| 20:09.27 | *** join/#asterisk ToxaP (i=ToxaP@194.187.128.88) |
| 20:09.29 | hi365 | and their stuff isnt that much better |
| 20:09.40 | hi365 | in sip.cong? YES |
| 20:09.57 | tzafrir | You can see where I'm from from my hostname. We work a bit with Tikal, but not more than that |
| 20:10.35 | teknoprep | i am having a voicemail indicator problem... i can check voicemail.. but my phones are not indicating wiht the light that i have voicemail |
| 20:11.09 | *** join/#asterisk Emrah (n=Me@adslgva0491.worldcom.ch) |
| 20:11.15 | Emrah | Hi! |
| 20:11.16 | ToxaP | teknoprep, read about MWI |
| 20:11.52 | tzafrir | hi365, No, not in sip.conf: in the dialplan, just before the Playback() |
| 20:11.54 | teknoprep | toxap do you have a good link? |
| 20:13.00 | Emrah | I'm experiencing a strange problem and if it is possible to ask you for a possible solution: The 7, * and mute buton on my Polycom suddenly stoped working. Any idea why? I didn't do anything particular last days... |
| 20:13.03 | hi365 | tzafrir: should be automatic! |
| 20:13.07 | Emrah | sorry for my poor English |
| 20:13.14 | Emrah | Polycom ip500 by the way |
| 20:13.40 | tzafrir | hi365, It should play 'he', right? |
| 20:13.46 | hi365 | right |
| 20:13.51 | ToxaP | teknoprep, No |
| 20:14.17 | Emrah | Doe anyone have an idea? |
| 20:14.20 | Emrah | does* |
| 20:14.45 | ToxaP | teknoprep, In peer configuring write mailbox= peer mailbox |
| 20:14.52 | krondorl | I was wondering if anyone knew how to change the recording time default when someone is leaving a message and if it is possible to send them back to the prompt system (or warn them) when their time is running out?? |
| 20:15.03 | *** join/#asterisk larryr (n=larry@64.56.99.31) |
| 20:15.16 | teknoprep | ToxaP, in peer config ? |
| 20:15.30 | teknoprep | ToxaP, for the extension? |
| 20:15.31 | Emrah | Anyone may help me with my Polycom please? |
| 20:15.52 | ToxaP | teknoprep, Yes, read wthis http://www.voip-info.org/wiki/view/Asterisk+sip+mailbox |
| 20:16.09 | teknoprep | ToxaP, ty |
| 20:16.16 | Emrah | Another problem is that now the dtmf is replaced by a single frequency tone. Just a similar beep for all tuches |
| 20:16.31 | ToxaP | teknoprep, ? |
| 20:16.41 | teknoprep | ToxaP, for the link |
| 20:16.52 | teknoprep | ToxaP, ty = thankyou |
| 20:16.59 | *** join/#asterisk Odie_Flocon (n=chatzill@S01060011953d7c4c.cg.shawcable.net) |
| 20:17.16 | Emrah | No idea? |
| 20:17.34 | *** join/#asterisk Gamercjm (n=chris@pool-71-254-174-163.lsanca.fios.verizon.net) |
| 20:18.13 | Gamercjm | Any one here using Voipjet.com? |
| 20:18.21 | Odie_Flocon | nope what is it? |
| 20:18.32 | Emrah | I was |
| 20:18.33 | Gamercjm | its just a voip provider |
| 20:18.51 | larryr | newbie looking for direction...wanting to know 1) If this is possible and 2)where to start looking. What I need to do is a queue, and when next op is avail, BEFORE transfer, need to announce to caller "Op 26 is taking your call", then do transfer.Can the AGI do this? |
| 20:19.09 | Gamercjm | After the changed there server.. I cant seem to get a call to go through.. |
| 20:19.29 | oQPa | exist a gui for asterisk controll? |
| 20:19.45 | Gamercjm | Emrah: have you used it since they chagned there main server to the test.voipjet.com? |
| 20:19.55 | Emrah | yeah |
| 20:20.01 | Emrah | but many routes are brocken |
| 20:20.18 | ToxaP | teknoprep, ;) I badly talk after English simply |
| 20:20.27 | Emrah | larryr: it is possible even without an AGI script |
| 20:20.54 | Emrah | if you use the local channel and configure your extensions properly it should be possible, but it is a tricky thing |
| 20:21.10 | Gamercjm | I cant make a call at all since then, and the only error is "Call failed to go through, reason 1" |
| 20:21.26 | teknoprep | ToxaP, understand |
| 20:21.32 | Emrah | where are you calling? |
| 20:21.42 | Gamercjm | anywhere.. anything |
| 20:21.50 | Gamercjm | nothing gets through |
| 20:21.51 | Emrah | strange.. hole on let me try |
| 20:22.19 | Emrah | no it works with me |
| 20:22.29 | Emrah | try to replace th test.voipjet.com with the direct IP |
| 20:22.57 | _alex_mx_ | :q! |
| 20:23.06 | Gamercjm | yah i tried, it says the call is accepted from the ip, then format ulaw, then Hungup |
| 20:23.10 | *** join/#asterisk zotz (n=zotz@24.244.163.225) |
| 20:23.21 | Emrah | I have that problem with all other destinations than the US |
| 20:23.23 | Odie_Flocon | is your account up to date? |
| 20:23.27 | Gamercjm | so right when it connects for the call it gets disconnected |
| 20:23.47 | Gamercjm | you mean like international calls? |
| 20:24.19 | Emrah | yeah |
| 20:24.37 | Gamercjm | yah i was never able to make international even when mine was working |
| 20:24.42 | oQPa | the best gui interface is evolution? |
| 20:25.42 | larryr | Emrah:Can you point me somewhere to get started? |
| 20:26.25 | Emrah | larryr: no sorry. I mus leave but I may help you in 30 minutes if you are there |
| 20:26.29 | Emrah | must* |
| 20:26.31 | larryr | ok |
| 20:26.35 | larryr | thanks. |
| 20:29.35 | *** join/#asterisk aptura (n=none@S010600a0c93f6f7e.vs.shawcable.net) |
| 20:30.30 | aptura | Has there been any discussion on the fact that fonality bought out trixbox? I am curios as to the reasons. |
| 20:31.43 | aydiosmio | there was something to puchase? |
| 20:31.46 | aydiosmio | try #freepbx |
| 20:32.10 | QbY | Anyone using the Cisco 7970G? Does it have a SIP Load? |
| 20:33.08 | rene1 | aptura: well trixbox and fonality are similar products |
| 20:36.34 | aptura | Thats a obvios given. |
| 20:36.35 | aptura | :) |
| 20:37.46 | marcus2 | hm, i'm really having zero luck finding a decent toll free did provider |
| 20:38.02 | syzygyBSD | I am trying to detect answering machines with MachineDetect. It will only proceed when I answer if I have the phone muted, event if I am not saying anything (could be white noise, or bad sip phones) |
| 20:38.15 | *** join/#asterisk _VoicePulse (n=contact@unaffiliated/voicepulse) |
| 20:38.20 | *** join/#asterisk bmg505 (n=leon@c1-102-16.rndf.isadsl.co.za) |
| 20:38.22 | syzygyBSD | however, when I send it to voicemail, it detects silence and starts talking right away |
| 20:38.30 | *** join/#asterisk fall0ut (n=tim@c-69-180-250-113.hsd1.tn.comcast.net) |
| 20:39.36 | syzygyBSD | does anyone have this or any machine detection working? |
| 20:41.12 | rene1 | aptura: so why have switchvox sponsoring them |
| 20:41.50 | aptura | Your asking me? |
| 20:43.13 | rene1 | aptura: well they could have done it to bring the trixbox creator expertise to them, ( i bet trixbox has been deployed far more than fonality) |
| 20:44.08 | *** join/#asterisk ShipHead (i=Maf54@gateway/tor/x-3730bf935abfdcb7) |
| 20:44.16 | rene1 | i dont like it either aydiosmio. but it is widely popular |
| 20:44.25 | aydiosmio | Oh I like it |
| 20:44.36 | aydiosmio | but it's not difficult to write a configuratoin wrapper |
| 20:45.04 | aydiosmio | which is all freepbx is |
| 20:45.05 | rene1 | i would say it is difficult to write one that so many people want to use and that is somehow feature complete |
| 20:45.10 | aydiosmio | thus all that trixbox is |
| 20:45.19 | aydiosmio | it's not though |
| 20:45.24 | *** part/#asterisk ShipHead (i=Maf54@gateway/tor/x-3730bf935abfdcb7) |
| 20:45.25 | rene1 | even if it exposes 10% of what asterisk can do |
| 20:45.26 | aydiosmio | you can't manage queue members |
| 20:46.09 | aydiosmio | or even use the default asterisk sounds in an IVR |
| 20:47.32 | rene1 | fonality and switchvox are a lot better |
| 20:47.35 | *** part/#asterisk rene1 (n=rene1@gea-gye-internet.telconet.net) |
| 20:48.23 | DonX | what would be the delimiter to tell cut to seperate each character of a variable? |
| 20:48.44 | *** join/#asterisk rene1 (n=rene1@gea-gye-internet.telconet.net) |
| 20:49.10 | rene1 | ~seen DataCompBoy |
| 20:49.26 | jbot | datacompboy <n=datacomp@212.17.0.67> was last seen on IRC in channel #asterisk, 2h 6m 26s ago, saying: 'rene1: sorry, disconnect:)); so, what your problem with AMI?'. |
| 20:49.36 | diclophis-work | is it possible to have a different context for 2 disticnt PRI span groups? |
| 20:49.39 | DonX | IE: exten => _6XXXXX,1,Set(SITENUMBER=${CUT(EXTEN||2)}) doesn't work |
| 20:51.11 | luke-Jr_work | Gamercjm: you dialed 011 for international? |
| 20:56.54 | *** join/#asterisk FuriousGeorge (n=FuriousG@ool-43536ea8.dyn.optonline.net) |
| 20:58.11 | syzygyBSD | Anyone used MachineDetect or AMD? |
| 20:58.42 | Death_INC | DonX: ${CUT(EXTEN,'',2)}? |
| 20:58.52 | DonX | it wouldn't work |
| 20:58.56 | DonX | uses the default - |
| 20:59.08 | Death_INC | DonX: what are you trying to DO? |
| 20:59.14 | Death_INC | IE. what part do you want out of it? |
| 20:59.34 | syzygyBSD | DonX: what do you want sitenumber to equal? |
| 20:59.41 | Death_INC | ${EXTEN:2:1}? |
| 20:59.42 | DonX | hrmm |
| 21:00.05 | syzygyBSD | if I dial 678912, what number do you want sitenumber to be |
| 21:00.10 | DonX | I thought cut assigned certain pieces into a new veriable |
| 21:00.22 | DonX | site is 7 |
| 21:00.32 | syzygyBSD | then use what Death_INC said |
| 21:00.34 | DonX | 8912 is the ext |
| 21:00.59 | DonX | what does ${EXTEN:2:1} do? |
| 21:01.08 | Death_INC | 8912 = ${EXTEN:3:4} |
| 21:01.20 | syzygyBSD | it starts at the 2 character and goes for 1 |
| 21:01.59 | DonX | oh ok |
| 21:02.01 | DonX | sweet |
| 21:02.07 | DonX | that's exactly what I'm looking for |
| 21:02.14 | syzygyBSD | I know :) |
| 21:02.55 | syzygyBSD | cut needs a delimiter, so if you had a space, colon, something seperating it it could grab the pieces |
| 21:03.56 | syzygyBSD | also if you just want the end of a string, you can use ${EXTEN:3} will grab everything after the third number |
| 21:04.47 | *** join/#asterisk hax (n=hax@httpcraft/hax) |
| 21:07.08 | Death_INC | syzygyBSD: or would it be ${EXTEN:1:1} heh |
| 21:07.08 | rene1 | can chanisavail take a zap group and tell me wether there are available channels in the group? |
| 21:07.36 | syzygyBSD | right death |
| 21:07.39 | Death_INC | EXTEN:2 removes the first 2, doesn't it? |
| 21:07.46 | syzygyBSD | yes |
| 21:08.00 | Death_INC | kk, DonX: sorry heh you want ${EXTEN:1:1} and ${EXTEN:2} |
| 21:08.18 | syzygyBSD | he will figure out that ${EXTEN:2:1} gives him the third not the second quick enough though |
| 21:08.25 | Death_INC | probably heh |
| 21:08.38 | syzygyBSD | unless they are the same number... |
| 21:08.47 | syzygyBSD | why did this start breaking all of the sudden... |
| 21:09.28 | Death_INC | lol |
| 21:11.01 | *** join/#asterisk Ebola (i=1000@81-86-155-65.dsl.pipex.com) |
| 21:12.00 | Death_INC | rene1: try it and let us know? ;) ... I'd guess so |
| 21:12.18 | syzygyBSD | so I am trying to write an app that calls distributers when an order is placed and allows them to confirm it. I want to be able to detect answering machines but Machine detect seems a bit flaky |
| 21:12.47 | Death_INC | syzygyBSD: make something that just listens for the BEEP? |
| 21:12.59 | rene1 | Death_INC: will try later |
| 21:13.00 | Death_INC | or is that what machine detect does? |
| 21:13.08 | syzygyBSD | what if a human answers |
| 21:13.15 | rene1 | is machine detect for IVRs? |
| 21:13.20 | Death_INC | you can just start talking for huiman or machine |
| 21:13.24 | syzygyBSD | machine detect waits for silence |
| 21:13.27 | rene1 | i thought it was for ansering machines |
| 21:13.32 | Death_INC | but if it's machine you'll hear the machine beep and abort |
| 21:13.59 | Death_INC | assuming you can detect the machine beep ;) |
| 21:14.04 | syzygyBSD | hmm, that may work, I would be more of a start over then an abort... |
| 21:15.09 | rene1 | what about requiring called people to send a dtmf tone? |
| 21:15.18 | rene1 | machines wont do that |
| 21:15.41 | rene1 | you are already requiring confirmation |
| 21:16.22 | *** join/#asterisk moon06 (n=michael@cim06-1-82-228-240-97.fbx.proxad.net) |
| 21:18.31 | syzygyBSD | it would be nice if I didn't have to |
| 21:19.08 | QbY | phones with multiple line appearances, how can i prevent the server from handing them another call from the queue when they are already on a call from a queue? |
| 21:22.05 | DonX | sweet |
| 21:22.09 | DonX | that got it up and going |
| 21:22.12 | DonX | thanks for the help |
| 21:22.37 | DonX | btw, is there anyway that I could tell asterisk to look to another voicemail source for a certain user? |
| 21:23.00 | DonX | Like, I have a phone that has two suip registrations to two different servers |
| 21:23.07 | DonX | sip even |
| 21:23.09 | syzygyBSD | QbY: there is a maxcalls I belive, or maxlines |
| 21:23.26 | luke-Jr_work | Zaptel devices automatically try the next available line, correct? (if the one requested is busy/in use) |
| 21:23.26 | DonX | What I'd like to do is send the voicemail to the other server |
| 21:23.46 | syzygyBSD | luke-Jr_work: depends on how you call it |
| 21:23.56 | syzygyBSD | you have to dial a zap group for it to work like that |
| 21:24.16 | luke-Jr_work | hmm |
| 21:25.54 | *** join/#asterisk IronMan2000 (n=kent@65.124.236.252) |
| 21:26.41 | IronMan2000 | Can someone tell me how I can force a call to another extenstion in my asterisk? I need for a call to immediatly route to another extenstion once it gets a call. |
| 21:26.58 | IronMan2000 | maybe an example? |
| 21:27.01 | luke-Jr_work | jump <extension>; |
| 21:27.12 | syzygyBSD | IronMan2000: goto(context,extension,priority) |
| 21:27.34 | IronMan2000 | thanks, I will try that now. |
| 21:28.12 | brodiem | quit |
| 21:29.24 | oQPa | a good free webgui for configure and manage asterisk? |
| 21:29.35 | oQPa | know? |
| 21:29.47 | syzygyBSD | is jump a command? |
| 21:30.13 | oQPa | joint |
| 21:30.27 | sevard | <PROTECTED> |
| 21:30.29 | QbY | syzygyBSD.. IS that a Sip.conf field, or Queue.conf ?? maxcalls/maxlines |
| 21:30.31 | syzygyBSD | oQPa: I have never used one, but I am guessing you are looking for something that does all the work for you |
| 21:30.42 | syzygyBSD | QbY: agent.conf? |
| 21:30.51 | QbY | syzygyBSD.. Not using agent.conf |
| 21:31.10 | QbY | i actually log in the channel .. ie.. SIP/username |
| 21:31.16 | QbY | agents gave me too much headache |
| 21:31.19 | syzygyBSD | really... |
| 21:31.30 | syzygyBSD | using queues without agents, I have never done that |
| 21:31.43 | QbY | yeah.. you can put in the actual extension instead or the sip channel |
| 21:31.48 | oQPa | syzygyBSD, nop.... only a gui |
| 21:32.34 | oQPa | is more easy |
| 21:32.37 | syzygyBSD | oQPa: vi is the only gui I use, I don't know what you really want one for... |
| 21:32.53 | syzygyBSD | I have found gui's to be much harder |
| 21:33.06 | sevard | pico is a hard gui? :) |
| 21:33.15 | syzygyBSD | very |
| 21:33.21 | sevard | you're crazy |
| 21:33.25 | oQPa | pico and vi |
| 21:33.26 | oQPa | XDD |
| 21:33.37 | oQPa | ok ok... nice gui's |
| 21:33.50 | syzygyBSD | at least in vi I can jump to line 5467 with 5 keystrokes |
| 21:34.22 | mog | i dont think thats possible |
| 21:34.28 | mog | maybe 6 |
| 21:34.28 | DonX | hehehe |
| 21:34.28 | syzygyBSD | oQPa: what are you going to be using this for |
| 21:34.32 | syzygyBSD | mog nope 5 |
| 21:34.39 | mog | <PROTECTED> |
| 21:34.42 | DonX | hehe |
| 21:34.45 | oQPa | rapid configure... and no very unix user |
| 21:34.53 | oQPa | simple |
| 21:35.00 | syzygyBSD | mog 5467 shift-G |
| 21:35.08 | mog | thats still 6 |
| 21:35.14 | mog | shift , g, 5,4,6,7 |
| 21:35.34 | syzygyBSD | G is one keystroke |
| 21:36.11 | denon | syzygyBSD: you can do it in pico/nano in less than that |
| 21:36.16 | denon | nano +10 file.txt |
| 21:36.19 | denon | brings you to line 10 |
| 21:36.23 | mog | i would love to see how you hit the shift key and the g key with one finger ^_^ |
| 21:36.23 | oQPa | syzygyBSD, a ivr |
| 21:36.27 | oQPa | simple |
| 21:36.39 | denon | or +10,5 brings you to line 10, column 5 |
| 21:36.43 | syzygyBSD | denon, well you need 2 spaces and a + for the command line option, so it is more |
| 21:37.14 | denon | but it happens faster, because the cursor doesnt need to render twice |
| 21:37.19 | oQPa | and app_rpt radio repeater |
| 21:37.27 | tzafrir | +10 from the command-line also works for vi. Also: +/test_to_find |
| 21:37.41 | syzygyBSD | lol, you can do the same thing in vi, and the UI is lighter so it renders faster... |
| 21:37.51 | denon | editor wars are so 10 years ago.. |
| 21:37.56 | oQPa | ok... i see the manual |
| 21:38.05 | syzygyBSD | lol, ahh, arguing command line text editors |
| 21:38.49 | *** join/#asterisk gambolputty (n=gambolpu@cblmdm72-240-246-145.buckeyecom.net) |
| 21:38.52 | tzafrir | Actually, arbitrary "vi" is a bit limited with command-line editing, as it can only handle one + command |
| 21:38.59 | gambolputty | Hi. Anyone gotten * to work with Jabber? |
| 21:39.07 | denon | well, anyone who thinks they're using vi, is probably using vim anyway |
| 21:39.24 | tzafrir | But vim can handle up to 10 + commands, so you can do some ex editing from the command line. Yicks |
| 21:39.45 | syzygyBSD | true, I haven't used the old vi for 4 years |
| 21:40.01 | denon | Ive got some hpux boxes with real vi on them |
| 21:40.11 | denon | but most modern bsd/linux stuff is vim afaik |
| 21:40.13 | mog | gambolputty, i have ^_^ |
| 21:40.19 | tzafrir | Debian comes with nvi as the default vi |
| 21:41.37 | syzygyBSD | ohh.. i have vi on one of my old freebsd boxes that is retired |
| 21:43.55 | larryr | Can anyone tell me *how* to find out when a call is being transferred to an agent? I want to announce to the caller "Op 26 is taking your call", then do transfer. Just point me in the direction...Thanks |
| 21:44.37 | denon | you could put it in the extension they use to transfer |
| 21:44.59 | larryr | This would be in a queue... |
| 21:48.10 | *** join/#asterisk daysmen3 (n=primus@host81-153-175-92.range81-153.btcentralplus.com) |
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| 21:51.19 | *** join/#asterisk robin_sz (n=you@adsl.redpoint.org.uk) |
| 21:51.30 | robin_sz | greetings. |
| 21:51.45 | blitzrage | salutations |
| 21:52.15 | robin_sz | heres some advice, if you consider buying a Sipura SPA2102 ATA, consider someting more pleasant instead |
| 21:52.17 | robin_sz | like ... |
| 21:52.27 | robin_sz | self circumcision |
| 21:52.44 | robin_sz | hateful device. |
| 21:53.45 | blitzrage | I just stick with Polycom, and no analog adapters |
| 21:54.07 | robin_sz | theres a very good reason for it ... |
| 21:54.22 | robin_sz | SIP to analogue, analogue to DECT |
| 21:54.32 | blitzrage | that sounds like a crappy reason :) |
| 21:54.49 | robin_sz | got a better suggestion? |
| 21:55.04 | robin_sz | and dont say WiFi phones |
| 21:55.44 | robin_sz | see. |
| 21:55.50 | blitzrage | I would never suggest a wifi phone :) |
| 21:56.01 | blitzrage | and you're probably right thats the best thing to do with DECT phones |
| 21:56.04 | blitzrage | damn you Nortel |
| 21:56.08 | robin_sz | heh |
| 21:56.44 | *** part/#asterisk jtexter3 (n=jtexter3@COX-66-210-197-34-static.coxinet.net) |
| 21:56.45 | robin_sz | there is some weird Dutch device I once had, a up-to-6 DECTS on one side, ethernet on the other |
| 21:57.05 | robin_sz | sort of square wall mounted DECT hub thing |
| 21:57.22 | robin_sz | only it only had SCCP at the time, no SIP |
| 21:57.45 | robin_sz | and it chose to hide on the network so I couldnt even see the sodding thing. |
| 21:58.57 | *** join/#asterisk jeffik (n=Jeff@dialup-4.247.245.164.Dial1.Tampa1.Level3.net) |
| 22:02.08 | *** join/#asterisk anonymouz666 (n=anonymou@200.218.193.6) |
| 22:03.34 | *** join/#asterisk brif8 (n=brif8@ns1.ttienterprises.org) |
| 22:03.38 | robin_sz | anyway .. all I actually wanted from this ATA was something that looked like an ethernet port on one side and two phones on the other ... but this thing has routers and bridges and DHCP servers and multiple pages of configuration ... |
| 22:04.26 | robin_sz | and basically way WAY too much complication to make it actually useable :( |
| 22:06.24 | *** part/#asterisk larryr (n=larry@64.56.99.31) |
| 22:06.33 | brif8 | Hi All I have tried following the Clipcomm Asterisk Guide :( Anyone got a Clipcomm CG410 setup with asterisk. Or any type of PSTN Gateway setup with asterisk I can't even get the simple register to work ? |
| 22:12.26 | DonX | anyone have experience with SPA-941's? |
| 22:12.41 | Jason99 | DonX: I've got a couple setup |
| 22:12.48 | DonX | Cool |
| 22:13.07 | DonX | Do you have more than one sip registration on any of them? |
| 22:13.30 | DonX | I'm trying to make it use the same voicemail server for ext 2 that it's using for extension 1 |
| 22:14.28 | Jason99 | ah, i've never tried that.. Are your two registrations setup for the same sip server? |
| 22:14.43 | blitzrage | DonX: thats usually a dialplan solution |
| 22:14.53 | DonX | When I put in the voicemail server and mailbox ID on the ext tab, it doesn't send to voicemail when I call ext2 |
| 22:15.13 | DonX | How can I tell it to look at a different voicemail server with a dialplan? |
| 22:15.34 | blitzrage | I'm not sure what data the phone actually sends, so I can't give you a definitive answer |
| 22:16.06 | blitzrage | but I guess I'm spoiled in that I have a lot of data in a DB that I can say, "hey, this phone should actually use voicemail box <foo>" |
| 22:18.15 | DonX | hehe |
| 22:18.18 | DonX | that's cool |
| 22:18.20 | DonX | thanks anyway |
| 22:19.02 | blitzrage | what extension is the phone trying to call? |
| 22:19.10 | blitzrage | I guess I'm not sure how it is trying to access the VM to begin with |
| 22:19.17 | blitzrage | it must be calling somethign in the DP though? |
| 22:19.28 | *** join/#asterisk angom (n=angom@red-corp-200.79.141.244.telnor.net) |
| 22:20.16 | DonX | well |
| 22:20.44 | DonX | ext1 is configured as sip device 201 on sip server 1 |
| 22:21.02 | DonX | ext2 is configured as sip device 801 on sip server 2 |
| 22:21.13 | DonX | sip server 1 and sip server 2 don't talk to one another |
| 22:21.15 | blitzrage | how are those defined? |
| 22:21.27 | blitzrage | continue with your train of thought first |
| 22:21.30 | DonX | as extensions on the SPA-941 |
| 22:21.34 | DonX | ok |
| 22:21.42 | *** join/#asterisk Adam12 (n=adam@d150-182-137.home.cgocable.net) |
| 22:21.54 | DonX | so, ext1 (sip 201) has voicemail configured on it's asterisk server |
| 22:22.07 | DonX | voicemail 801 doesn't |
| 22:22.28 | blitzrage | are these defined at [201] and [801] on their respective servers? |
| 22:22.39 | blitzrage | in sip.conf |
| 22:22.41 | DonX | I would like to send callers to server 1 when they call into ext2 and the call needs to go to voicemail |
| 22:22.45 | DonX | yeah |
| 22:23.07 | blitzrage | ok, when you hit the VM button on your phone, what is it dialing? |
| 22:23.13 | blitzrage | how it it getting to the VM? |
| 22:23.21 | blitzrage | s/it it/is it |
| 22:23.29 | *** join/#asterisk mattchis (n=mattchis@216.54.143.242) |
| 22:23.43 | DonX | well the first extension will dial 2500 |
| 22:24.00 | DonX | the second extension doesn't have voicemail and I don't want it to |
| 22:24.06 | mattchis | has anyone sucessfully setup Buddy Watch on asterisk? |
| 22:24.47 | blitzrage | DonX: ok, so when you dial 2500 does it prompt you for an extension number, or just a pass? If no extension number, what is the exten => line look like in your DP? |
| 22:25.19 | DonX | correct, it prompts |
| 22:26.08 | blitzrage | ok, so what you want to do on server 2 is just redirect the call to the extension on server 1 via SIP or IAX2 |
| 22:26.24 | blitzrage | so when server 2 see's extension 2500, it shoots it over to server 1 to handle |
| 22:26.34 | DonX | is that the only way to do it |
| 22:26.38 | anonymouz666 | LOST today |
| 22:26.45 | blitzrage | DonX: how else would you expect to do it? |
| 22:26.47 | DonX | the problem is that the provider isn't playing nice |
| 22:27.00 | DonX | so I'm having to setup all of our remote locations cluged |
| 22:27.03 | blitzrage | DonX: you could setup an NFS partition, but that won't work very well across a WAN |
| 22:27.17 | blitzrage | DonX: the data is on server 1, you HAVE to access the data there |
| 22:27.22 | DonX | yeah |
| 22:27.23 | blitzrage | in some fashion |
| 22:27.42 | DonX | I understand that but I can' establish a SIP tunnel to there |
| 22:27.47 | blitzrage | NFS partition or redirecting the call, or putting the information in an ODBC database |
| 22:28.02 | blitzrage | those are basically your 3 options |
| 22:28.12 | blitzrage | use an IAX2 tunnel |
| 22:28.30 | *** join/#asterisk trelane_ (i=trelane@pdpc/supporter/sustaining/trelane) |
| 22:28.38 | blitzrage | or setup a VPN maybe if its the provider that is just filtering traffic |
| 22:28.50 | blitzrage | DonX: I'm not sure what else you would have expected me to say? |
| 22:28.53 | DonX | nah, it's a really messed up scenario |
| 22:29.01 | DonX | I might just tell them to screw off ont he v/m |
| 22:29.06 | DonX | I understand |
| 22:29.11 | DonX | just wanted to verify that |
| 22:29.20 | blitzrage | the data is on server 1. That is a fact. You must access the data that is on server 1. That is also a fact. |
| 22:29.25 | blitzrage | No way to get aroudn that :) |
| 22:29.59 | blitzrage | unless you create some sort of strategy where you rdiff the two voicemail dirs between the servers once an hour or something |
| 22:30.02 | blitzrage | but that is messy |
| 22:30.15 | blitzrage | then technically you could just access it locally |
| 22:30.23 | DonX | yeah |
| 22:32.07 | *** join/#asterisk mattchis (n=mattchis@216.54.143.242) |
| 22:32.48 | mattchis | Can anyone point me to a good article on setting up Polycom Buddy Watch on Asterisk |
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| 22:37.36 | _DAW | hello mates |
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| 23:02.05 | *** join/#asterisk M4dkiss (i=madkiss@madkiss.org) |
| 23:02.08 | M4dkiss | hi all |
| 23:02.33 | nettie | Hi guys, when my voip carrier asterisk server is not reachable for a few minutes, after a couple of retrys my asterisk server doesnt register anymore. sip show registry shows the peer status as "Request Sent". I'm sure the voip carrier server is back online because everytime this happens I do a sip reload and it magically registers. Anyone know what could be wrong please? do I need to specify a timeout somewhere or this might look like a bug |
| 23:02.42 | M4dkiss | Just a short question, I hope somebody can help me out. I have a Cisco 7960, I want it to load its stuff via dhcp. It tells me "Requesting nformation", but on the tftpserver, nothing happens |
| 23:05.22 | Nivex | M4dkiss: is the tftp server the same as the dhcp server? |
| 23:06.09 | M4dkiss | yes |
| 23:09.11 | robin_sz | 13 seperate pages of HTML forms to configure an ATA |
| 23:09.22 | robin_sz | big ones too. thats not normal is it? |
| 23:10.34 | robin_sz | nettie: when you server is not reacable, is it possible your DNS is not reacable either?? |
| 23:10.41 | robin_sz | h |
| 23:10.52 | robin_sz | the missing h's for use above :) |
| 23:11.14 | *** join/#asterisk Tili (n=tili@202.133.65.18) |
| 23:11.22 | nettie | hi robin |
| 23:11.29 | nettie | thanx for helping me out |
| 23:11.55 | nettie | I'm explaining the problem in #asterisk-dev becaus eit looks like a bug imho but I'm not sure. |
| 23:11.58 | M4dkiss | can I test someow whether tftp is running? |
| 23:11.59 | *** join/#asterisk jebba (n=jebba@OL74-217.fibertel.com.ar) |
| 23:11.59 | nettie | the dns is still reachable |
| 23:12.00 | nettie | anyway |
| 23:12.17 | nettie | and the registar resrver hostname can be resolved |
| 23:12.30 | robin_sz | M4dkiss: netstat -tupan | grep tftp |
| 23:12.39 | nettie | if you want I can pm you the messages I'm writing in #asterisk-dev |
| 23:12.41 | robin_sz | or try telnetting to the tftp port on the remote machine |
| 23:12.42 | Nugget | telnet is eeeeeeevil! |
| 23:12.55 | M4dkiss | not open |
| 23:12.56 | M4dkiss | bwaa |
| 23:16.17 | M4dkiss | hm, now it runs but the phone won't use it apparently |
| 23:16.38 | *** join/#asterisk lowlevel (n=Stuart@CPE0050ba71c82f-CM000f9f7d6742.cpe.net.cable.rogers.com) |
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| 23:36.53 | aptura | Which ip phone does not have issues with nat traveral if thay are installed remotly in another office off site? say in a case customer budget? |
| 23:37.08 | aptura | That thay can only afford one pbx and not two of them. |
| 23:38.27 | brif8 | Is a FXO / ATA Gateway configured in sip.conf as a phone or as a VoIP provider ? |
| 23:51.34 | jebba | aptura, you can use an IAX2 ATA |
| 23:51.35 | aptura | jebba I am only installing ip phones. |
| 23:51.35 | benjk | you can use any IAX2 ip phone, then |
| 23:51.36 | *** join/#asterisk DrRighteous (n=DrRighte@ool-457843d1.dyn.optonline.net) |
| 23:52.26 | *** join/#asterisk ast_freak (n=jesse@h69-130-174-77.69-130.unk.tds.net) |
| 23:52.43 | aptura | Well my question concerns the polycom soundpoint series. |
| 23:54.05 | *** join/#asterisk bkw__ (n=brian@adsl-70-234-45-236.dsl.tul2ok.sbcglobal.net) |
| 23:55.08 | benjk | if your pbx is on a public ip and you are not using any total-crap router, then your remote ip phones should just work fine |
| 23:55.32 | benjk | if you do use a total-crap router, get a proper router |
| 23:55.43 | denon | [nat device] |
| 23:55.49 | denon | almost any *router* should be just fine :) |
| 23:55.51 | benjk | if your pbx is behind NAT, build a tunnel |
| 23:56.23 | denon | guess that's just me |
| 23:56.24 | aptura | benjk, you mean a vpn |
| 23:56.39 | denon | no, he means a 1-to-1 nat tunnel |
| 23:56.44 | *** join/#asterisk ariel_ (n=Ariel@dsl-20-177.cofs.net) |
| 23:56.47 | benjk | if your pbx is behind NAT and you can't build a tunnel, set up a remote sip/iax gateway |
| 23:56.57 | benjk | whatever tunnel you prefer |
| 23:57.11 | benjk | as long as it isn't SSL |
| 23:57.30 | *** join/#asterisk teknoprep (n=tekon@unaffiliated/teknoprep) |
| 23:57.30 | benjk | cause that isn't the real thing |
| 23:57.37 | teknoprep | so you guys also against running asterisk in a vm ? |
| 23:57.40 | teknoprep | as in vmware ? |
| 23:57.41 | aptura | benjk, the phone would be installed in a remote office say across town while the pbx is in the company hq. Thats where nat plays a part for the phone. |
| 23:57.59 | benjk | where is the NAT? |
| 23:58.10 | *** join/#asterisk justinu (n=Justin@dsl081-083-093.lax1.dsl.speakeasy.net) |
| 23:58.46 | benjk | justinu hi |