| 00:01.50 | *** join/#asterisk duki (n=duki@host-85-27-58-159.brutele.be) |
| 00:03.11 | *** part/#asterisk atomicd (n=Atomicd@74-206-0-80.static-ip.m.telepacific.net) |
| 00:05.44 | *** join/#asterisk J4k3 (n=jsuter@openwrt.us) |
| 00:09.38 | Tagor | [TK]D-Fender >> On the internet people are only using the macro to for example drop the call. Is there also a way to execute a shell script using a macro? |
| 00:10.06 | [TK]D-Fender | Tagor, it executes... DIALPLAN. Use your imagination for more than 1/2 a second.... |
| 00:13.57 | russellb | TokyoJimu: asterisk will not hang. you will lose CDR records, though. |
| 00:14.28 | TokyoJimu | russellb: Thanks. Losing records is fine. Asterisk hanging would not be. |
| 00:15.10 | russellb | though a couple people have been working on a nice patch so that you won't lose records, either ... |
| 00:15.20 | russellb | that would be a 1.6 thing, though. |
| 00:15.21 | pepse | hey, i think someone asked this before, can you accept DTMF while asterisk is ringing an extension? |
| 00:16.09 | pepse | for home use type thing, have asterisk pick up phone right away as it usually does and ring the extension that rings all the house phones, but you can press a number to do something else |
| 00:16.45 | *** join/#asterisk duki (n=duki@host-85-27-58-159.brutele.be) |
| 00:17.00 | *** join/#asterisk craigk (n=ckowald@58.174.122.198) |
| 00:19.49 | *** join/#asterisk shido6 (n=shido6@204.126.120.132) |
| 00:22.59 | *** join/#asterisk DrAk0 (n=ljd@nelug/coreteam/luisjose) |
| 00:32.22 | *** join/#asterisk ming_zym (i=ming_zym@nat/yahoo/x-d8fa17fa38edb897) |
| 00:34.18 | alpha232 | cdr is such a pain |
| 00:38.49 | *** join/#asterisk jmesquita (n=jmesquit@201.20.243.195.user.ajato.com.br) |
| 00:39.34 | blitzrage | I love CDRs |
| 00:39.57 | [hC] | i love CDR's when they're right :) |
| 00:40.27 | tzanger | heh |
| 00:41.36 | Tagor | How can I get the number that answered when I call several numbers with Dial()? |
| 00:41.49 | Tagor | ${CALLERID} outputs some change info |
| 00:41.51 | tzanger | Tagor: BRIDGEPEER? |
| 00:41.55 | Tagor | *strange |
| 00:43.10 | Tagor | tzanger >> That var seems to be empty |
| 00:43.21 | tzanger | that may not be the exact var |
| 00:43.24 | tzanger | look it up in vars.txt |
| 00:43.33 | tzanger | I thinkt hat's what it is |
| 00:43.48 | Tagor | vars.txt? Where can I find that doc? |
| 00:43.54 | tzanger | source dir, under docs/ |
| 00:44.15 | *** part/#asterisk ming_zym (i=ming_zym@nat/yahoo/x-d8fa17fa38edb897) |
| 00:45.14 | Tagor | According to voip-info it's ${DIALEDPEERNUMBER} |
| 00:45.18 | Tagor | But that var is also empty |
| 00:47.21 | *** join/#asterisk shido6 (n=shido6@204.126.120.132) |
| 00:47.52 | Tagor | Ah seems to be a bug in Asterisk |
| 00:48.32 | alpha232 | is there a way to generate a call from the console, say call party X as party Y and then Call Party Y as Party X and join them together |
| 00:50.32 | blitzrage | alpha232: don't think so |
| 00:51.31 | alpha232 | blitzrage: maybe via ami or somthing |
| 00:51.45 | blitzrage | well ya... but that's not CLI :) |
| 00:51.50 | alpha232 | ugh my god these config files are a pita |
| 00:52.08 | [hC] | Sweettttt... Parking hints working great in 1.2 with a simple devstate patch |
| 00:52.12 | alpha232 | and the nominclature |
| 00:52.50 | blitzrage | really? i never noticed that |
| 00:53.56 | alpha232 | blitzrage: well I mean i'm coming from the world of MAT's and this is far more convoluted than a NEC NEAX 2400 |
| 00:54.45 | alpha232 | and to have thinks like... "friend" |
| 00:55.07 | blitzrage | and about a billion times more powerful :) |
| 00:56.55 | alpha232 | only by virtue that it is extensible |
| 00:58.05 | alpha232 | i mean, why is a SIP that can do inbound and outbound calls called friend? |
| 00:58.30 | alpha232 | or is that purely to say someone who can call "into" the system and "thru" the system |
| 01:00.17 | [hC] | in and out yes. |
| 01:00.36 | [hC] | its means peer and user together |
| 01:00.37 | [TK]D-Fender | That idea is mashed in 1.4 |
| 01:01.17 | [hC] | [TK]D-Fender: that parkingi patch for hints worked perfectly, that i found in the bugtracker for 1.2 |
| 01:01.19 | [hC] | easy peasy. |
| 01:01.24 | TJNII | ~sipnat |
| 01:01.25 | jbot | [~sipnat] Quick guide on configuring * + SIP behind NAT : http://www.aocomputing.net/?p=3 otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
| 01:01.28 | [TK]D-Fender | [hC], lucky find... |
| 01:02.12 | alpha232 | [hC]: ok so the idea is friend is allowed to IN, OUT and Through |
| 01:02.38 | TJNII | Completely off topic, but a roll of tiolet paper soaked in kerosene inside a jack-o-lantern works really well. |
| 01:02.41 | TJNII | That is all. |
| 01:02.47 | J4k3 | TJNII: RIGHT ON. |
| 01:03.23 | [hC] | alpha232: sure if that makes sense to you. in and out also implies through, really. |
| 01:03.40 | alpha232 | [hC]: well in a way it doesn't though |
| 01:03.41 | J4k3 | pumpkin inferno |
| 01:04.05 | J4k3 | I suggest filling the pumpkin with everclear, lighting it, and throwing it at something that needs to be covered with burning everclear |
| 01:04.15 | TJNII | Heheh. Possibly |
| 01:04.25 | J4k3 | pile of brush, fallen down barn, police cars, owner of the rival team's lexus, etc. |
| 01:04.50 | TJNII | I have 2 (well, about 1 3/4 now) of gunked kerosene thats not good for too much else. So: burning pumpkin! |
| 01:05.00 | *** join/#asterisk MrBIOS (n=aperez@38.104.128.78) |
| 01:05.08 | TJNII | s/) of/) gallons of/ |
| 01:05.13 | alpha232 | bargh i just botched my apt-get |
| 01:05.38 | MrBIOS | hi there, I'm wondering if it's possible to use Altigen-branded IP710 H.323 phones with an Asterisk switch |
| 01:09.03 | [TK]D-Fender | MrBIOS, *'s H.323 kinda blows. If you don't have those phones yet don't buy them |
| 01:09.14 | MrBIOS | we have an office full of them |
| 01:09.18 | MrBIOS | and yess, I fully realize that |
| 01:09.23 | MrBIOS | but we're growing at 10% a month. |
| 01:09.36 | alpha232 | MrBIOS: you're better off with tin cans and wet string |
| 01:10.14 | MrBIOS | I've more or less come to that conclusion on my own |
| 01:10.21 | alpha232 | ARGGH wtf |
| 01:15.29 | MrBIOS | I need ~50 handsets, anyone recommend some handsets that are cheap and do SIP/IAX2? |
| 01:15.40 | BBHoss | heh |
| 01:15.46 | *** join/#asterisk lowlevel (n=Stuart@CPE0017f2e2a3e8-CM000f9f7d6742.cpe.net.cable.rogers.com) |
| 01:15.48 | alpha232 | ~cheap |
| 01:15.49 | jbot | rumour has it, cheap is a bad idea. If you're setting up an Asterisk system, one of the wisest pieces of advice you can take is this: DON'T BE A CHEAPSKATE. |
| 01:15.54 | BBHoss | youre shit out of luck for iax2 handsets |
| 01:16.04 | MrBIOS | we've been quoted $10K for more altigen licenses. |
| 01:16.11 | MrBIOS | so anything up to 10k would be "cheap" |
| 01:16.22 | BBHoss | just buy 50 polycom 340s |
| 01:16.34 | alpha232 | thats $200 a saetr |
| 01:16.36 | alpha232 | set |
| 01:16.40 | alpha232 | why not go softphone? |
| 01:16.51 | MrBIOS | politics |
| 01:16.52 | BBHoss | my bad |
| 01:16.54 | BBHoss | 430 |
| 01:17.00 | BBHoss | softphones suckzorz |
| 01:17.08 | MrBIOS | I think so yes |
| 01:17.16 | MrBIOS | polycom 430s are $200 new? |
| 01:17.23 | docelm0 | If you want to be cheap.. Get some GXP2000's.. but you will pay for it.. |
| 01:17.28 | TJNII | Allright, I'm trying to allow incoming SIP connections to my * server which is behind a NAT. I've forwarded 5060 and my RTP range (10000 - 15000) for SIP. Any ports I forgot? |
| 01:17.28 | docelm0 | ~gs |
| 01:17.29 | jbot | GrandSuck phones & gateways are cheap junk which should be avoided with extreme prejudice. |
| 01:17.57 | docelm0 | ya 15001 to 20000 UDP |
| 01:18.27 | TJNII | docelm0: I dropped the range from to 10k-20k to 10k-15k |
| 01:18.43 | TJNII | So thet should be OK. |
| 01:18.49 | docelm0 | You are like the 3rd person I spoke today that has but their asterisk box behind a nat.. its a pain in the ass and dont recommend it.. |
| 01:18.51 | docelm0 | Yes |
| 01:19.16 | TJNII | docelm0: I know, but it's how my network is set up. |
| 01:19.35 | docelm0 | although 1.4's STUN config is promising.. I havent tried it yet tho |
| 01:19.49 | *** join/#asterisk jksM (n=jks@87.57.88.86) |
| 01:20.50 | TJNII | Yea, I'm still running 1.2 because I'm trying not to deviate from my distros package manager, but STUN probably would make my life easier. I may upgrade for it. |
| 01:20.58 | BBHoss | asterisk behind NAT is only a PITA if you want other stuff to go over the rtp ports and sip ports |
| 01:21.49 | docelm0 | why not just build a linux router and stick asterisk on it. |
| 01:21.58 | TJNII | Well, hopefully the problem was that I didn't have the rtp ports forwarded. NTP and TFTP both worked to the server, SIP not so much. The server didn't even show a connection was attempted. |
| 01:22.16 | TJNII | docelm0: Because I pay for electricity and don't want another computer running 24/7 |
| 01:22.20 | [TK]D-Fender | MrBIOS, almost no point to the IP 430 anymore. |
| 01:22.31 | MrBIOS | what would you use? |
| 01:22.31 | TJNII | Though this machine is getting overloaded so I may change my stance on that. |
| 01:22.33 | [TK]D-Fender | MrBIOS, And IP 320 (the entry point) is now about $85 |
| 01:22.44 | *** join/#asterisk MrKeuner (n=kudo@unaffiliated/mrkeuner) |
| 01:22.49 | [TK]D-Fender | MrBIOS, www.telephonydepot.com |
| 01:23.08 | docelm0 | TJNII you could easily use ASTLinux to do the same. |
| 01:23.20 | MrKeuner | Hello, can I use asterisk as a sip client? Will it be an overkill for that? |
| 01:23.23 | BBHoss | if polycoms are too cheap you could go with some cheap aastras |
| 01:23.29 | [TK]D-Fender | MrBIOS, if you need a passthrough port, then IP 330. If you need passthrough & NON POE, maybe IP 501 then. |
| 01:23.36 | BBHoss | i have a 9113i or w/e for 79.99 |
| 01:23.51 | [TK]D-Fender | BBHoss, Polycoms too cheap? How si that possible? :) |
| 01:24.04 | TJNII | docelm0: I fail to see what I need ASTLinux for. |
| 01:24.20 | *** join/#asterisk vizo (n=Administ@c-71-58-223-233.hsd1.nj.comcast.net) |
| 01:24.48 | vizo | hello |
| 01:24.51 | BBHoss | my bad, just woke up :) |
| 01:24.54 | [TK]D-Fender | TJNII, just follow that first guide..... |
| 01:24.56 | docelm0 | use it as a linux router also. It runs from 9V DC. Cant complain about power then |
| 01:24.58 | jksM | anyone experienced any differences between spandsp releases? (like for example between 0.0.4pre4 and the newest pre11) |
| 01:24.59 | BBHoss | too expensive i mean!!! |
| 01:25.01 | [TK]D-Fender | TJNII, Should be all you need. |
| 01:25.19 | alpha232 | docelm0: url? |
| 01:25.20 | BBHoss | anybody here ever got dundi working? |
| 01:25.31 | docelm0 | alpha232 URL for what? |
| 01:25.31 | [TK]D-Fender | BBHoss, I wouldn't touch anything lower than a 9133 in Aastra's line, and even then... EW... |
| 01:25.47 | TJNII | [TK]D-Fender: Hopefully. I did find something I missed. Now I just need someone to get home so I can get them to try and log in. |
| 01:25.50 | alpha232 | docelm0: that piece of equipment... that works as a router and runs from 9vdc |
| 01:26.01 | BBHoss | well some people can't always have the best, and IMO, aastras are better than ~gs |
| 01:26.54 | [TK]D-Fender | BBHoss, very true, but in budget range I'd pick a Linksys over a low end Aastra (sub 480i / 55i+) |
| 01:27.06 | *** join/#asterisk ming_zy1 (i=ming_zym@nat/yahoo/x-583772549a190f25) |
| 01:27.18 | jksM | anyone have ideas on what to test when sending out faxes via app_txfax doesn't work, although receiving faxes with app_rxfax do? |
| 01:27.21 | [TK]D-Fender | BBHoss, And geographically speaking Linksys' price point is stable low world-wide |
| 01:27.25 | vizo | so I have decide to build a asterisk box for home and was wondering if anyone has any tips on installing it on gentoo. |
| 01:27.36 | TJNII | docelm0: Okay, that is cool, then. |
| 01:27.37 | BBHoss | the 480is are getting dirt cheap too, ive never actually tried the 9113i, just suggesting |
| 01:27.47 | docelm0 | Well it can do tons of shit.. Not just that.. :) Um, astlinux is a distro for running Asterisk on a soekris board |
| 01:27.47 | [TK]D-Fender | vizo, Nope. Just have the dependencies installed and any distro is as good as the next |
| 01:28.00 | docelm0 | the site is www.astlinux.org |
| 01:28.04 | TJNII | vizo: Just emerge it and set it up. It's easy. |
| 01:28.06 | [TK]D-Fender | BBHoss, You mean 9112. the other is a 9133. There is no 9113 :) |
| 01:28.13 | BBHoss | yeah yeah |
| 01:28.22 | BBHoss | like i said, just woke up :) |
| 01:28.30 | TJNII | docelm0: Yea, once I clicked on the hardware link it made a lot more sense. |
| 01:28.31 | [TK]D-Fender | BBHoss, Caffeinate :p |
| 01:28.37 | BBHoss | definitely |
| 01:28.55 | BBHoss | TJNII: how recent is the emerge file? |
| 01:29.19 | vizo | cool, one question come to mind should I use alsa in the kernel or emerge the drivers? |
| 01:29.56 | fujin_ | #gentoo? |
| 01:30.01 | fujin_ | I always emerge alsa personally |
| 01:30.37 | alpha232 | i wish there was an easy way to get an softmodem to act as an FXO card |
| 01:30.54 | alpha232 | i mean it has almost all the guts |
| 01:31.54 | TJNII | BBHoss: Still in 1.2, last I checked |
| 01:31.58 | fujin_ | lol softmodem |
| 01:31.58 | fujin_ | sif |
| 01:32.08 | [TK]D-Fender | vizo, * doesn't need ALSA for anything... |
| 01:32.11 | TJNII | BBHoss: Though it looks like they are in the final stages of grooming 1.4 |
| 01:32.22 | BBHoss | hmm |
| 01:32.38 | vizo | [TK]D-Fender: no? |
| 01:32.51 | BBHoss | there are a few issues with the DTMF and some media gateways, since it doesnt send a DURATION |
| 01:33.00 | BBHoss | you know if they patched it? |
| 01:33.01 | alpha232 | fujin_: not everyone can afford a $100 card |
| 01:33.07 | fujin_ | get a better job |
| 01:33.08 | fujin_ | ~cheap |
| 01:33.08 | jbot | hmm... cheap is a bad idea. If you're setting up an Asterisk system, one of the wisest pieces of advice you can take is this: DON'T BE A CHEAPSKATE. |
| 01:33.17 | TJNII | vizo: * only talks to alsa for console calls and special stuff. It doesn't need it. |
| 01:33.18 | fujin_ | hell, don't use an FXO card for a start |
| 01:33.19 | fujin_ | that's just dumb. |
| 01:33.24 | BBHoss | heh |
| 01:34.07 | alpha232 | never knew there was so much toilet paper in the world |
| 01:34.51 | TJNII | Speaking of toilet paper, I need to put another roll in the jack-o-lantern.... |
| 01:35.42 | alpha232 | too cheap to buy a candle |
| 01:36.14 | J4k3 | I'd just integrate a bottle of WD40 and a spark plug |
| 01:36.22 | J4k3 | about every 5 seconds make it go *WOOSH* |
| 01:37.00 | alpha232 | lol |
| 01:37.16 | alpha232 | J4k3: need somthing more volitile |
| 01:37.31 | alpha232 | J4k3: compressed alcohol |
| 01:39.48 | J4k3 | alpha232: yeah, that works |
| 01:40.08 | TJNII | alpha232: the toilet paper has been soaking in kerosene for the last 6 hours. ^.^ |
| 01:41.04 | alpha232 | ahh |
| 01:41.07 | alpha232 | smokey |
| 01:41.23 | J4k3 | hey, amsoil just called and asked why you're burning their bypass filters |
| 01:41.24 | J4k3 | :D |
| 01:41.44 | J4k3 | (any excuse to poke fun of scamsoil) |
| 01:42.55 | alpha232 | lol |
| 01:46.50 | alpha232 | so lets talk about voice modems and getting them to wokr ;) |
| 01:49.03 | TJNII | Hmmm... I have a query. I have a couple friend who run * servers connecting over IAX. When the call, I see the extension of their phone on the caller ID. I'd like to append a prefix to that number so instead of XXXX I see YXXXX. Near as I can tell callerid= in iax.conf simply overrides the callerid. Can you point me in the right direction for this? |
| 01:50.57 | J4k3 | alpha232: save yourself lots of effort and just don't. :) |
| 01:51.03 | J4k3 | death to pots |
| 01:51.58 | *** join/#asterisk grndslm (n=grndslm@24-116-87-97.cpe.cableone.net) |
| 01:51.59 | alpha232 | J4k3: ummm then find me a very affordable BRI card that supports NI1 and Asterisk |
| 01:52.34 | alpha232 | i would much rather have my BRI line than POTS |
| 01:53.20 | *** join/#asterisk jsoftw_ (n=Administ@60.234.135.124) |
| 01:53.57 | [TK]D-Fender | TJNII, Set(CALLERID(num)=12345${CALLERID(num)}) |
| 01:54.46 | MrBIOS | are the HD polycoms compatible with asterisk yet? |
| 01:54.47 | TJNII | [TK]D-Fender: But that's for extensions.conf, correct? |
| 01:55.26 | TJNII | hmmmm... I suppose a script couls do it.... |
| 01:55.38 | MrBIOS | or do they only function in "non HD" mode? |
| 01:56.03 | [hC] | MrBIOS: they work in HD mode |
| 01:56.06 | *** join/#asterisk lowlevel (n=Stuart@CPE0017f2e2a3e8-CM000f9f7d6742.cpe.net.cable.rogers.com) |
| 01:56.18 | MrBIOS | the price difference isn't that huge |
| 01:56.26 | [TK]D-Fender | TJNII, naturally. |
| 01:59.21 | MrBIOS | do the cisco SIP phones work with Asterisk? |
| 01:59.33 | [hC] | MrBIOS: yes. |
| 01:59.38 | MrBIOS | are they any good? |
| 02:00.03 | [hC] | Yes, they're alright. Upgrading them and provisioning them -can- be a bit ofa headache but overall they're good. |
| 02:00.51 | mocker | Having a problem with Asterisk not waiting for me to enter a full extension during a background wait. |
| 02:00.56 | fujin_ | the easiest to provision phones I'v ever done are Linksys |
| 02:01.03 | fujin_ | the spa9x2 series |
| 02:01.07 | alpha232 | that's the one thing that Asterisk doesn't have, Online Activation |
| 02:01.21 | fujin_ | online activation? |
| 02:01.23 | fujin_ | what the hell? |
| 02:01.42 | MrKeuner | pap2t-na is this device locked? |
| 02:01.55 | [hC] | MrKeuner: shouldnt be, hence the NA |
| 02:01.57 | fujin_ | mocker: have you set the device timeout too short? |
| 02:01.59 | alpha232 | TJNII: allowing the asterisk to auto-provision a phone |
| 02:02.02 | fujin_ | err, dievce timeout |
| 02:02.10 | fujin_ | waitexten or equivalent |
| 02:02.14 | fujin_ | alpha232: that's not asterisks job |
| 02:02.20 | mocker | fugin: DigitTimeout,5 |
| 02:02.22 | [hC] | alpha232: asteirsk doesnt do it, but a tftp/ftp/http server can. thats how everyone does it. |
| 02:02.33 | TJNII | alpha232: I use tftp for that. |
| 02:02.36 | [hC] | alpha232: some web gui's will do the work for you, other people do it by hand |
| 02:02.44 | fujin_ | mocker: is it waiting 5 secs? |
| 02:02.56 | mocker | No. |
| 02:03.06 | mocker | I have extension 101, and IVR menu optoin 1 |
| 02:03.08 | fujin_ | I'm oldschool, I use WaitExten(5) |
| 02:03.22 | fujin_ | have never bothered to try digittimeout |
| 02:03.26 | mocker | I try to dial 101, but 1 gets recognized first. |
| 02:03.34 | MrBIOS | any other vendors doing the HD-audio schtick? |
| 02:04.09 | mocker | fujin: WaitExten doesn't listen in the background though, does it? |
| 02:05.29 | MrKeuner | do you guys recommend this telephonedepot? are there better alternatives? |
| 02:05.50 | fujin_ | mocker: yes, it does |
| 02:05.54 | fujin_ | that's how I do my IVR's, with waitexten |
| 02:05.58 | fujin_ | and they're all working super. |
| 02:07.48 | mocker | So users can enter options mid menu? |
| 02:08.10 | jksM | anyone has experience in debugging app_txfax/app_rxfax? |
| 02:10.23 | MrBIOS | is there an asterisk H.323 guide out there? |
| 02:11.04 | [hC] | probably but people try not to use h323 since it sucks |
| 02:11.22 | MrBIOS | sure I understand that |
| 02:11.36 | MrBIOS | this would be a short-term, stop-gap solution |
| 02:12.17 | [hC] | well, im sure theres info out there, but i cant help you with it, sorry |
| 02:14.44 | alpha232 | anyone here currently using FWD? |
| 02:15.31 | alpha232 | I signed up but can't register :/ |
| 02:15.51 | alpha232 | and you're quite SOL if you can't connect as you can't get help any other way |
| 02:16.15 | MooingLemur | fwd.pulver.com:5060 593941 105 Registered |
| 02:16.35 | alpha232 | hmm i was told to use iax.fwdnet.net |
| 02:16.42 | MooingLemur | and alpha232: were you on EFNet, say, 6 years ago? |
| 02:16.50 | MooingLemur | I'm using SIP. |
| 02:16.50 | alpha232 | oh dear |
| 02:17.06 | alpha232 | long time no see :) |
| 02:17.25 | alpha232 | I just recognised the nick ;) considering it's on my aim i should have caught it sooner |
| 02:17.30 | MooingLemur | hah |
| 02:17.59 | alpha232 | i went incommunicado for a while, worked on sustaining a relationship, almsot 2 years so far |
| 02:18.30 | TJNII | Pfft. Who needs women when you've got a computer. |
| 02:18.37 | MooingLemur | I remember back then you were setting callerid on a PRI and calling me |
| 02:18.52 | alpha232 | ROFFL |
| 02:18.53 | alpha232 | yep |
| 02:19.00 | TJNII | Note he said at, not with. |
| 02:19.31 | alpha232 | TJNII: don't grin too much or people might get the wrong or possibly right idea |
| 02:19.42 | TJNII | Haha |
| 02:19.50 | *** part/#asterisk dan__t (i=dan@neener.neener.org) |
| 02:19.57 | [hC] | hahaha nice domain. |
| 02:20.49 | MooingLemur | hmm.. not that I've used FWD at all in the last 2+ years |
| 02:21.16 | *** join/#asterisk watchy (n=watchy@c-68-51-54-72.hsd1.ar.comcast.net) |
| 02:21.24 | watchy | is that xorcom worker dude here? |
| 02:21.35 | alpha232 | my call may be monitored or recorded?!? and I'm the only one in the conference? |
| 02:21.40 | alpha232 | MooingLemur: good hold music |
| 02:21.47 | watchy | anyone here use Xorcom? |
| 02:21.51 | *** join/#asterisk dlynes (n=dlynes@d154-20-9-152.bchsia.telus.net) |
| 02:22.00 | mocker | fujin: waitexten doesn't seem to work like that. |
| 02:22.05 | *** part/#asterisk mocker (n=user@198.247.173.227) |
| 02:22.05 | alpha232 | MooingLemur: can i steal it? |
| 02:22.09 | *** join/#asterisk mocker (n=user@198.247.173.227) |
| 02:22.14 | alpha232 | MooingLemur: is that enya? |
| 02:22.28 | MooingLemur | haha.. I forgot I sent FWD calls to the meetme |
| 02:22.38 | mocker | Does anyone know how to increase the amount of time Background waits between keypresses? |
| 02:22.53 | MooingLemur | something about digittimeout? |
| 02:22.54 | alpha232 | MooingLemur: not a bad idea though |
| 02:23.10 | watchy | Anyone able to help me with Xorcom? |
| 02:23.13 | MooingLemur | alpha232: the hold music is stock with asterisk-sounds |
| 02:23.17 | mocker | I've tried DigitTimeout and ResponseTimeout |
| 02:23.18 | MooingLemur | I'm pretty sure |
| 02:23.36 | alpha232 | really |
| 02:23.38 | mocker | But it sees pattern 1 and matches. |
| 02:23.38 | alpha232 | interesting |
| 02:23.43 | mocker | Doesn't let me go to 101 |
| 02:24.02 | alpha232 | ahh much better, GSM sucked ass on that call |
| 02:24.08 | alpha232 | I still get studder though hrrm |
| 02:24.09 | MooingLemur | -[/:#]- ls /var/lib/asterisk/mohmp3/ |
| 02:24.09 | MooingLemur | fpm-calm-river.mp3 fpm-sunshine.mp3 fpm-world-mix.mp3 |
| 02:24.51 | alpha232 | oh helps if i wasn't running azureus at full tilt |
| 02:25.00 | MooingLemur | heh |
| 02:25.21 | watchy | Anyone able to help me with Xorcom? |
| 02:25.46 | alpha232 | MooingLemur: so how ya been? still living up here? |
| 02:27.09 | *** join/#asterisk r0d3nt (i=nobody@foster.stonedcoder.org) |
| 02:27.14 | *** join/#asterisk DarkRift (i=dark@bas10-montreal02-1177581747.dsl.bell.ca) |
| 02:27.15 | MrBIOS | anyone here have any experience to know how much time it takes to get a previously-existant data T1 converted into a PRI? |
| 02:27.27 | MooingLemur | alpha232: Arizona, for the last 4.5 years |
| 02:27.51 | alpha232 | wow |
| 02:27.54 | *** join/#asterisk Teln12100 (i=hello123@bas2-toronto12-1167861631.dsl.bell.ca) |
| 02:28.12 | alpha232 | finally settledown? |
| 02:32.39 | alpha232 | hrrm big latency ugh |
| 02:33.05 | mocker | Guh, this is killing me. |
| 02:33.13 | mocker | Never had this problem before. |
| 02:37.32 | watchy | if anyone helps me with a Xorcom Astribank i'll give you a hug |
| 02:38.13 | alpha232 | watchy: asl? |
| 02:38.37 | watchy | 27/m/super sex |
| 02:38.38 | watchy | y |
| 02:39.07 | J4k3 | sechsee |
| 02:39.36 | *** join/#asterisk ghento (n=ghento@64.180.85.230) |
| 02:39.40 | watchy | yea see jake knows all about my sechseeness |
| 02:41.12 | alpha232 | is there a reason why when listening to the audio on the console/dsp it's distorted? |
| 02:48.06 | TJNII | alpha232: Hold music or voice? |
| 02:49.11 | *** join/#asterisk macros73_ (n=cs@c-24-3-246-27.hsd1.mn.comcast.net) |
| 02:49.14 | *** join/#asterisk vAd0r (n=IceChat7@65.67.210.121) |
| 02:49.25 | vAd0r | can someone help me configuring my aastra 9133i phone |
| 02:49.31 | vAd0r | i have the 3 files in the directory |
| 02:49.35 | vAd0r | i want to use http to download the config file |
| 02:50.02 | vAd0r | it wont let me type in the http path on the phone. It only lets me do numbers |
| 02:50.25 | TJNII | ....yea, the IP address of the server |
| 02:50.35 | vAd0r | thats the thing |
| 02:50.41 | vAd0r | it is a virtual host |
| 02:51.34 | TJNII | Well, it probably doesn't do name resolution so you may need a server with a fixed ip. |
| 02:51.46 | TJNII | Though I may be wrong, but if it only allows numbers I doubt it. |
| 02:53.06 | alpha232 | TJNII: pre-recorded messages |
| 02:53.52 | TJNII | alpha232: I don't know, then. Only the console, or your phones too? |
| 02:54.00 | alpha232 | just the console, weird |
| 02:54.41 | alpha232 | hmm i thought all the SIP traffic went through the switch |
| 02:55.19 | alpha232 | i called myself from one machine to another via FWD and it actually connected me directly instead of through FWD |
| 02:56.27 | mocker | I hate stupid typos that kill my free time. |
| 02:56.28 | mocker | :( |
| 02:56.40 | alpha232 | ugh i have the feeling my line can't support voip pft |
| 02:57.09 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 02:57.28 | vAd0r | what is the debug command in asterisk -r |
| 02:57.34 | vAd0r | seems like debug sip and verbose |
| 02:58.13 | mocker | set debug 20 |
| 02:58.16 | mocker | set verbose 20 |
| 02:58.17 | mocker | etc.. |
| 03:01.31 | *** join/#asterisk ZX81 (n=matt@202.49.106.158) |
| 03:04.44 | *** join/#asterisk bkruse_home (n=kruz@76.73.154.120) |
| 03:04.58 | vAd0r | ok this is so frustrating |
| 03:05.03 | TJNII | alpha232: If you don't have canreinvite=no your phones will try and connect directly. |
| 03:05.04 | vAd0r | i cant get this phone to register |
| 03:05.36 | vAd0r | i created the aastra.cfg file and the mac.cfg file and point his phone to my pub ip and i dont even see it attempt |
| 03:05.42 | vAd0r | in the debug |
| 03:05.57 | TJNII | In the asterisk debug? |
| 03:06.02 | vAd0r | right |
| 03:06.03 | vAd0r | nothin |
| 03:06.18 | vAd0r | what do i need to set on the phone |
| 03:06.25 | vAd0r | just tftp and put my ip |
| 03:06.29 | TJNII | Well, no it wouldn't. The configs are on a http server so it won't talk to asterisk to get the configs |
| 03:06.35 | TJNII | Check the httpd logs |
| 03:06.41 | vAd0r | i changed them |
| 03:06.50 | vAd0r | or im confused what im suppose to do |
| 03:06.56 | vAd0r | i moved them to tftpboot |
| 03:07.09 | TJNII | So you're running a tftp server? |
| 03:07.13 | vAd0r | yes |
| 03:07.16 | vAd0r | both i think |
| 03:07.47 | TJNII | You should be able to look at the logs or start the tftp server in a debug mode to see if the phone is downloading the configs. |
| 03:08.20 | TJNII | Feh. No trick-or-treaters. I'm disappointed. |
| 03:08.28 | vAd0r | i never get any |
| 03:08.28 | alpha232 | TJNII: no pre-teens for you |
| 03:08.32 | vAd0r | im on a culdesac |
| 03:08.55 | vAd0r | should i do tftp or http |
| 03:09.33 | TJNII | I use tftp, but I think http is viewed as more secure |
| 03:09.47 | vAd0r | do i just need port 69 open |
| 03:09.49 | TJNII | Though if you set up tftp right it shouldn't be a problem |
| 03:09.59 | TJNII | 69 fot tftp, yes |
| 03:10.02 | vAd0r | how do i check to see if tftp is running |
| 03:10.27 | TJNII | pgrep is uesfull for that |
| 03:10.36 | vAd0r | can you tell me |
| 03:10.38 | TJNII | s/uesfull/useful/ |
| 03:10.42 | vAd0r | how to use it |
| 03:11.03 | TJNII | Also it depends on which flavor of tftp you have installed |
| 03:11.22 | vAd0r | dont know |
| 03:11.26 | vAd0r | what i have |
| 03:11.42 | alpha232 | argh, how come when I add, exten => 9999,2,Milliwatt() |
| 03:11.54 | alpha232 | I still get No such extension '9999' in context 'local' |
| 03:12.04 | TJNII | vAd0r: then I suggest you find that out. You can't start the tftp server if you don't know what it is. |
| 03:12.30 | TJNII | alpha232: no extensions reload, perhaps? |
| 03:12.56 | TJNII | Also I assume you have 9999,1 defined |
| 03:13.25 | [TK]D-Fender | ~assume |
| 03:13.26 | jbot | About assumptions : http://www.moviewavs.com/php/sounds/?id=gog&media=WAVS&type=Movies&movie=Long_Kiss_Goodnight"e=assume.txt&file=assume.wav |
| 03:13.27 | alpha232 | where do i define it? |
| 03:13.28 | [TK]D-Fender | ^^^^^^^^^^^^^^ |
| 03:13.43 | [TK]D-Fender | alpha232, how about right ABOVE 9999,2? :) |
| 03:14.23 | TJNII | [TK]D-Fender: Yes, I know about assumptions, that's why I ASKED HIM if he did it. |
| 03:14.40 | *** join/#asterisk l2trace99 (n=asd@219.172.8.67.cfl.res.rr.com) |
| 03:14.41 | [TK]D-Fender | <TJNII> Also I assume you have 9999,1 defined <---------- |
| 03:14.47 | [TK]D-Fender | TJNII, nice try :) |
| 03:15.12 | [TK]D-Fender | TJNII, "A" for effort, "F" for form :p |
| 03:15.49 | alpha232 | HHGTG |
| 03:15.49 | watchy | punch me in the face tk i bought 2 xorcoms |
| 03:15.59 | alpha232 | watchy: can i have one? |
| 03:16.15 | watchy | i dunno i finnaly just got the linux box to see one |
| 03:16.30 | alpha232 | TJNII: hrrm reload doesn't reload extensions i guess |
| 03:16.40 | alpha232 | have to do it via extensions reload |
| 03:17.46 | watchy | i'm about to goto backyard burger and get me a burger |
| 03:19.24 | [TK]D-Fender | watchy, Nah... you'll be punching yourself before long... |
| 03:21.15 | alpha232 | OUCH |
| 03:21.21 | alpha232 | nothing like an ear full of milliwatt |
| 03:24.56 | alpha232 | hrrm... |
| 03:24.58 | alpha232 | Oct 31 23:24:19 WARNING[8835]: chan_zap.c:913 zt_open: Unable to open '/dev/zap/pseudo': No such file or directory |
| 03:25.19 | watchy | tk: well the lead tech where i work wanted to try em |
| 03:25.41 | *** join/#asterisk inv_arp[work] (n=junya@c-76-26-24-197.hsd1.fl.comcast.net) |
| 03:30.48 | J4k3 | wow, trixbox is turning into some extra nazi shite. |
| 03:31.23 | [TK]D-Fender | J4k3, I hat it when shit goes commie :) |
| 03:32.40 | *** join/#asterisk antlers (n=antlers@ip70-173-89-173.lv.lv.cox.net) |
| 03:35.48 | variable_office | is there any way to get the username of the calling user? |
| 03:36.05 | keith4_ | milliwatt? |
| 03:36.35 | variable_office | keith4, directed towards me? |
| 03:37.06 | flenders | variable_office: 14:21 < alpha232> nothing like an ear full of milliwatt |
| 03:37.20 | keith4_ | variable_office: no |
| 03:37.24 | keith4_ | yah, what flenders said |
| 03:37.32 | alpha232 | flenders: que? |
| 03:37.58 | variable_office | ah, gotcha |
| 03:37.59 | alpha232 | lol |
| 03:38.09 | keith4_ | alpha232: what is this "milliwatt" you speak of? |
| 03:38.18 | variable_office | well any idea on my question? the callerid is not set to the username unfortunatly |
| 03:38.21 | J4k3 | 936-687-9905 = milliwatt fo yo ass |
| 03:38.36 | flenders | keith4_: it's a very annoying sound that you use to tweak the gains on analog lines |
| 03:38.37 | *** join/#asterisk GeneG (n=GeneG@bas3-toronto01-1177779508.dsl.bell.ca) |
| 03:39.23 | flenders | keith4_: add Milliwatt() to your dialplan and you'll know what it is |
| 03:39.34 | keith4_ | ... ok |
| 03:39.48 | GeneG | Apologies if this is a very old question, but how do I get around the "bad magic number" seg fault in astobj2.c with * 1.4.13 on Mac OS X 10.4? |
| 03:40.01 | alpha232 | keith4_: just make sure you turn down the volume |
| 03:40.02 | flenders | ~milliwatt |
| 03:40.05 | J4k3 | GeneG: I could make a suggestion |
| 03:40.09 | GeneG | Please |
| 03:40.18 | J4k3 | GeneG: but if you think macs are keen, you probably won't like it ;) |
| 03:40.27 | keith4_ | "don't use OS X"? |
| 03:40.27 | GeneG | :) |
| 03:40.28 | flenders | keith4_: yeah, otherwise you'll go deaf |
| 03:40.36 | J4k3 | keith4_: ding ding |
| 03:40.43 | alpha232 | 1khz at 0db |
| 03:40.44 | keith4_ | i need to find an analog phone, first, i guess? |
| 03:40.54 | J4k3 | it works fine from my cellphone |
| 03:40.54 | keith4_ | whoa |
| 03:40.54 | keith4_ | hmm |
| 03:40.56 | alpha232 | and 0db on voip is reallllly 0db |
| 03:41.07 | J4k3 | quite annoying |
| 03:41.14 | J4k3 | if I want to run a cellphone battery down |
| 03:41.21 | J4k3 | I dial the local milliwatt and throw the phone in the microwave |
| 03:41.25 | alpha232 | haha |
| 03:41.32 | alpha232 | good rf block |
| 03:41.41 | J4k3 | the microwave's cage works good enough to force the phone to max txpower to get back to the local cdma 800 tower. |
| 03:41.48 | keith4_ | and what's the next step in this calibrating process? |
| 03:42.05 | J4k3 | keith4_: somebody to turn the screw on the line card ;) |
| 03:42.21 | flenders | keith4_: change the gains on zapata.conf |
| 03:42.22 | keith4_ | now you're just making shit up |
| 03:42.29 | J4k3 | keith4_: really |
| 03:42.35 | J4k3 | basically |
| 03:42.45 | J4k3 | the linesman comes to your house with his little worthless meter |
| 03:42.52 | J4k3 | clips it to your line, clips his butt set to the meter |
| 03:43.08 | J4k3 | and dials milliwatt and quiet termination (936-687-9905 and 936-687-9906, for me) |
| 03:43.18 | J4k3 | and takes power readings at each |
| 03:43.28 | keith4_ | to set a high and low |
| 03:43.32 | J4k3 | then, if the CO is manned, calls the CO and has the line adjusted up/down so many db |
| 03:43.41 | J4k3 | now |
| 03:43.43 | *** join/#asterisk CunningPike (n=CunningP@S010600095b33697f.vc.shawcable.net) |
| 03:43.44 | J4k3 | the only problem with this is |
| 03:43.58 | J4k3 | the test is done only at 1khz (1004 hz I think is the exact #, but I may be wrong) |
| 03:44.10 | J4k3 | so if the line is weak above or below, its untested |
| 03:44.25 | J4k3 | my local phone lines are delivered through a line-side remote, and everything over 2700 hz is mush |
| 03:44.38 | keith4_ | bet that makes faxing tricky |
| 03:44.52 | J4k3 | it makes any sort of data-ish activity tricky |
| 03:44.56 | J4k3 | 2400 works fine |
| 03:45.04 | alpha232 | lol |
| 03:45.10 | J4k3 | I can usually squeeze 21.6k out of them with a usr courier |
| 03:45.12 | alpha232 | J4k3: what kind of lineside remote are they using |
| 03:45.35 | J4k3 | alpha232: I know I don't have it right, but what springs into my mind is "american anaconda timespan 64" |
| 03:45.48 | J4k3 | that seems right |
| 03:45.54 | J4k3 | or close enough |
| 03:46.10 | keith4_ | hmmm |
| 03:46.15 | keith4_ | this looks bad |
| 03:46.16 | keith4_ | Oct 28 18:15:19 ERROR[10649] chan_sip.c: SIP MESSAGE JUST IGNORED: ACK |
| 03:46.16 | keith4_ | Oct 28 18:15:19 ERROR[10649] chan_sip.c: BAD! BAD! BAD! |
| 03:46.27 | J4k3 | chan_sip.c needs a spanking |
| 03:46.28 | J4k3 | hawt |
| 03:46.50 | alpha232 | lol |
| 03:46.55 | *** join/#asterisk DarkRift (i=dark@bas10-montreal02-1177581830.dsl.bell.ca) |
| 03:47.05 | keith4_ | bad chan_sip, bad! |
| 03:47.08 | J4k3 | all I know is this remote is old as shit... OLD. |
| 03:47.22 | J4k3 | they haven't been able to buy parts for it for years |
| 03:47.34 | alpha232 | J4k3: so destroy it |
| 03:47.35 | J4k3 | one of the old timer inside plant guys uses a soldering iron and mouser.com to keep the shit alive. |
| 03:47.40 | J4k3 | why bother? |
| 03:47.45 | alpha232 | J4k3: get a new SLC |
| 03:47.46 | J4k3 | I got a T1 and vitelity.net |
| 03:47.53 | J4k3 | err an internet T1 |
| 03:48.07 | alpha232 | ah |
| 03:48.10 | J4k3 | far cheaper ;) |
| 03:48.21 | alpha232 | i have a wireless T1 but it seems to suck for VOIP |
| 03:48.37 | J4k3 | sucky |
| 03:48.49 | J4k3 | wireless and T1 don't usually go together |
| 03:48.57 | J4k3 | unless its some old cylink hardware |
| 03:48.58 | J4k3 | ;) |
| 03:49.01 | keith4_ | jitter and voip don't, either |
| 03:49.11 | keith4_ | so, i hear the milliwatt tone.... |
| 03:49.25 | J4k3 | eh, I have customers a couple 802.11 hops out from here that use vonage with no problems |
| 03:49.41 | J4k3 | 11a backhauls, 11g (ofdm-only) APs, generally |
| 03:50.01 | J4k3 | and, I mean, its vonage... *boggle* |
| 03:50.05 | keith4_ | a buddy of mine works for a point-to-point microwave ISP... and they have clients who try to use vonage, with varying degrees of success |
| 03:50.23 | J4k3 | yeah, it has a lot to do with the routing between you and them |
| 03:50.27 | keith4_ | 5.6Ghz unlicensed backhauls |
| 03:50.32 | J4k3 | I'm sure it'll probably get crappy after I change to qwest |
| 03:50.42 | alpha232 | J4k3: it's Navini |
| 03:50.46 | keith4_ | some of these idiots are 10 to 12 miles from the closest omni |
| 03:50.54 | J4k3 | navini's wanna-be wimax stuff? |
| 03:51.08 | alpha232 | J4k3: it's better than most of the wimax stuff |
| 03:51.09 | J4k3 | that stuff has some really weird latency jitters from what I've read |
| 03:51.13 | J4k3 | it angers people |
| 03:51.31 | alpha232 | J4k3: got an echo? |
| 03:51.36 | J4k3 | alpha232: not I. |
| 03:51.42 | J4k3 | ;) |
| 03:51.50 | keith4_ | so what is this milliwatt tone telling me? |
| 03:51.52 | keith4_ | i still don't get it |
| 03:52.24 | alpha232 | keith4_: lol |
| 03:52.26 | J4k3 | --- 12.197.131.7 ping statistics --- |
| 03:52.26 | J4k3 | 25 packets transmitted, 25 received, 0% packet loss, time 6023ms |
| 03:52.26 | J4k3 | rtt min/avg/max/mdev = 4.300/4.769/8.301/0.863 ms, pipe 2 |
| 03:52.26 | J4k3 | [jsuter@asterisk1 ~]$ |
| 03:52.30 | J4k3 | thats a 2-hop customer. |
| 03:52.34 | alpha232 | ouch |
| 03:53.14 | variable_office | or is there possibly a way to make voicemailmain auto load with the mailbox parameter for the calling user/ |
| 03:53.15 | variable_office | ? |
| 03:53.32 | alpha232 | rtt min/avg/max/mdev = 151.117/151.357/151.794/0.545 ms |
| 03:53.34 | alpha232 | thats me |
| 03:53.34 | keith4_ | variable_office yes |
| 03:53.42 | J4k3 | alpha232: daaaaamn |
| 03:53.55 | GeneG | Sorry to interrupt again: for anyone who is interested, I solved my magic number segfault by removing a pre-1.4.13 module I was using (app_conference.so) |
| 03:53.57 | alpha232 | yep |
| 03:54.10 | variable_office | keith4, how to do this? you will help me out alot if you know how! |
| 03:54.29 | J4k3 | alpha232: jesus... my EVDO phone can beat that under optimal circumstances. |
| 03:54.30 | keith4_ | variable_office: i have no idea, but I seem to recall reading about it.... either in the book or the wiki |
| 03:54.49 | J4k3 | and very low local network usage |
| 03:54.56 | variable_office | there are only three instances of voicemailmain in the book and none of them are it |
| 03:54.57 | keith4_ | GeneG: i have nothing against using macs.... i'm on an ibook right now.. but why run asterisk on os x? |
| 03:55.01 | alpha232 | funny |
| 03:55.04 | variable_office | the voip-info site doesnt have it either |
| 03:55.26 | GeneG | keith4, I needed an extra Mac for testing purposes and I figure since it's sitting idle for 99% of the time why not use it for * |
| 03:55.45 | GeneG | There's no real load so for the most part it works just fine |
| 03:55.53 | alpha232 | J4k3: the nice thing is, the more data i push the faster it gets |
| 03:56.05 | keith4_ | my "testing purposes" asterisk box is a trash-picked P3 600 |
| 03:56.37 | keith4_ | GeneG: sure, except that you have to compile everything yourself, and there aren't drivers for lots of stuff |
| 03:56.38 | GeneG | The only problem is that OS X isn't really treated as a first class citizen by the community, so there seem to be issues with every upgrade |
| 03:57.23 | GeneG | keith4, I don't mind the compiling and the lack of drivers isn't an issue because our environment is pretty stable (100% VoIP). I just don't like it when stuff breaks. |
| 03:57.30 | GeneG | Like now |
| 03:57.37 | J4k3 | alpha232: yeah, you actually get allocated timeslots. |
| 03:57.38 | variable_office | there should be some other way to assign voicemail mailboxes besides CID info |
| 03:57.39 | GeneG | (I just got that segfault again) |
| 03:57.54 | J4k3 | alpha232: I don't like polled-like-a-nazi networks |
| 03:58.03 | J4k3 | alpha232: it leads to lots of unnessasary latency. |
| 03:58.24 | *** join/#asterisk k-man (n=jason@unaffiliated/k-man) |
| 03:58.31 | k-man | hello |
| 03:58.37 | keith4_ | variable_office: isn't that how cell phone companies do it? |
| 03:59.01 | alpha232 | J4k3: well it's not polled perse |
| 03:59.18 | k-man | why would one want to store voicemail messages on an imap server? |
| 03:59.21 | alpha232 | but it does use multiple channels |
| 03:59.28 | variable_office | keith4, i have multiple sip users that have the same CID but different mailboxes |
| 03:59.29 | alpha232 | k-man: because it's the perfect interface to it |
| 03:59.45 | J4k3 | alpha232: yeah, there are multiple channels and timeslots across them... its assigned both |
| 03:59.53 | alpha232 | keith4_: unified messaging |
| 03:59.56 | k-man | alpha232, does that mean you can then retrieve your voicemail via the same imap server? |
| 04:00.01 | J4k3 | alpha232: and theres all sorts of nice processing gain that can be done with multiple antennas and such with navini. |
| 04:00.06 | alpha232 | k-man: sure, why not |
| 04:00.15 | J4k3 | (although I've read some really pissed off people concerning the processing gain part...) |
| 04:00.17 | k-man | i dunno... i'm just wondering |
| 04:00.35 | alpha232 | J4k3: dunno, all i know is i get about T1 down and half up |
| 04:00.43 | alpha232 | and I can drive around town and have connectivity |
| 04:01.01 | *** join/#asterisk nighty^ (n=nighty@sushi.rural-networks.com) |
| 04:01.36 | J4k3 | alpha232: sounds like what I get with evdo, pretty much |
| 04:01.47 | alpha232 | and how much do you pay? |
| 04:01.48 | J4k3 | except I get the evdo for $19.95/month, and replace the word "country" with "town" :) |
| 04:01.58 | J4k3 | well, I pay $0 now, but its vzw |
| 04:02.05 | alpha232 | lol |
| 04:02.11 | *** join/#asterisk bmg505 (n=leon@196.209.183.44) |
| 04:02.15 | J4k3 | and I'm switching from my en mou account with vzw, to a $20/mo legitimate account with alltel. |
| 04:02.24 | J4k3 | no stupid 5GB/month usage restriction on evdo, etc. |
| 04:02.33 | J4k3 | vzw is serious data nazis. |
| 04:02.44 | watchy | nawzi |
| 04:02.49 | alpha232 | J4k3: when I want to do somthing latency sensitve i just setup a ping between here and the other router with half second delay |
| 04:02.50 | J4k3 | naws-ie |
| 04:02.52 | watchy | how much u pay for unlimited data jakes |
| 04:03.11 | J4k3 | watchy: from verizon? since I only ever use it at night and the weekends, nothing above my normal voice contract ;) |
| 04:03.12 | watchy | i just ordered a laptop with built in hsdpa |
| 04:03.34 | J4k3 | of course, verizon has started charging me enough bullshit fees to pay for a data account, hence why I'm switching to alltel. |
| 04:03.46 | J4k3 | watchy: I prefer using a handset |
| 04:03.50 | J4k3 | for the access device |
| 04:03.54 | watchy | ah |
| 04:04.01 | watchy | i'm poor i only got an iphone |
| 04:04.04 | J4k3 | with bluetooth, you may take a small performance hit but you can put the phone somewhere 'inconvienent' for a couple hours |
| 04:04.05 | watchy | its only edge:( |
| 04:04.08 | J4k3 | and get data service |
| 04:04.16 | J4k3 | iphones are cute |
| 04:04.23 | J4k3 | my friend had his iphone bt'd through my vx9800 |
| 04:04.25 | watchy | i love it |
| 04:04.27 | J4k3 | so he could surf via evdo |
| 04:04.35 | watchy | but it edge sux |
| 04:04.52 | J4k3 | all mobile data sucks |
| 04:05.06 | J4k3 | hopefully your hspda laptop has fallback |
| 04:05.14 | watchy | 802.11n |
| 04:05.22 | J4k3 | I meant gsm/gprs/edge fallback |
| 04:05.36 | watchy | oh yea it does |
| 04:06.11 | alpha232 | Interesting |
| 04:06.21 | alpha232 | FWD rejects my call to the milliwatt number |
| 04:06.22 | watchy | i got my xorcoms working |
| 04:06.42 | watchy | but i am super awesome |
| 04:06.45 | J4k3 | alpha232: on g729, vitelity only gives me the first tone. |
| 04:06.54 | J4k3 | on g711, it just works. |
| 04:07.20 | alpha232 | J4k3: well by setting up a ping feed i get almost flawless audio |
| 04:07.31 | alpha232 | my only problem is that my laptop microphone is located near the fan |
| 04:07.32 | alpha232 | lol |
| 04:07.44 | alpha232 | hey MooingLemur you here? |
| 04:07.55 | alpha232 | where do I get the default hold music |
| 04:13.59 | J4k3 | alpha232: sip isn't enough traffic to give you slots? |
| 04:15.29 | fujin_ | anyone know the difference between soundpoint ip330 and ip320? |
| 04:15.37 | fujin_ | oh nm |
| 04:15.38 | fujin_ | i got it |
| 04:15.39 | fujin_ | the pic shows |
| 04:18.42 | *** join/#asterisk stkn_ (n=stkn@gentoo/developer/pdpc.active.stkn) |
| 04:19.32 | alpha232 | J4k3: guess not :) |
| 04:19.41 | J4k3 | alpha232: ghetto |
| 04:19.45 | alpha232 | lol |
| 04:19.55 | alpha232 | well a simple half second ping works fine |
| 04:20.10 | alpha232 | 24 bytes from 198.144.136.1: icmp_seq=1868 ttl=255 time=86.8 ms |
| 04:20.52 | *** join/#asterisk PepOSX (n=pepOSX@190.72.144.104) |
| 04:21.36 | J4k3 | alpha232: man... thats still outrageously high... |
| 04:23.06 | alpha232 | lol |
| 04:23.49 | alpha232 | omg i am so going to kill someone right now |
| 04:24.10 | *** join/#asterisk rnovotny22 (n=ro085181@c-69-180-187-19.hsd1.mn.comcast.net) |
| 04:25.05 | watchy | why |
| 04:25.19 | alpha232 | because someone from the office called because he is inept |
| 04:26.08 | alpha232 | and he needs to know how to take a test |
| 04:27.21 | *** join/#asterisk Alowishus (n=jpenix@70.230.199.141) |
| 04:28.29 | J4k3 | what I need to do is buy a server to run as my dedicated sql server |
| 04:28.37 | J4k3 | I keep thinking small and that keeps making my brain fold |
| 04:28.54 | J4k3 | a dedicated optimized db box would solve a lot of my mental pain |
| 04:29.07 | J4k3 | also, can one store voicemail to database with *? |
| 04:29.37 | J4k3 | ahh here we go |
| 04:30.07 | alpha232 | lol |
| 04:30.11 | J4k3 | thats sexy. |
| 04:33.50 | J4k3 | http://img.photobucket.com/albums/1003/josethebean/67fmm2a.gif |
| 04:36.43 | *** part/#asterisk rnovotny22 (n=ro085181@c-69-180-187-19.hsd1.mn.comcast.net) |
| 04:47.18 | *** join/#asterisk lowlevel (n=Stuart@CPE0017f2e2a3e8-CM000f9f7d6742.cpe.net.cable.rogers.com) |
| 04:47.23 | Alowishus | haha what's the context of that |
| 04:48.59 | J4k3 | I don't know |
| 04:49.07 | J4k3 | it seems I've seen the video on youtube |
| 04:49.17 | J4k3 | its like the school football star being interviewed on holoween |
| 04:49.26 | J4k3 | and this guy jumps out of nowhere and the guy just drops him |
| 04:49.28 | J4k3 | straight up |
| 04:49.36 | J4k3 | and you hear it, he followed through quite well |
| 04:49.46 | J4k3 | then backs up and starts smiling |
| 04:51.04 | J4k3 | alpha232: 1-hop customer: 100 packets transmitted, 100 received, 0% packet loss, time 1304ms |
| 04:51.07 | J4k3 | rtt min/avg/max/mdev = 2.078/2.372/5.153/0.653 ms, pipe 2 |
| 04:51.22 | J4k3 | 12 miles, one hop |
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| 04:52.45 | *** part/#asterisk Igbothom_III (n=Hilton@office.quarkgroup.com.au) |
| 04:57.43 | alpha232 | lol |
| 04:57.54 | alpha232 | yeah and thats a wired connection |
| 04:58.03 | *** join/#asterisk lukus (n=luke@202.172.122.210) |
| 04:59.44 | alpha232 | haha all my hold music is playing in slow motion |
| 04:59.46 | alpha232 | ROFL |
| 05:02.01 | *** join/#asterisk linagee (n=linagee@about/linux/staff/linagee) |
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| 05:06.03 | J4k3 | alpha232: wired? nope. |
| 05:06.11 | J4k3 | alpha232: thats straight up 20 mhz 802.11g |
| 05:06.13 | J4k3 | ;) |
| 05:06.24 | J4k3 | off an *omni* no less. |
| 05:06.28 | J4k3 | (hpol) |
| 05:06.30 | alpha232 | lol |
| 05:06.40 | J4k3 | 24 dBi parabolic on the far end. |
| 05:06.49 | alpha232 | lol |
| 05:07.02 | J4k3 | but thats also over pppoe bottled down to 768k/384k |
| 05:07.28 | alpha232 | lol |
| 05:08.04 | alpha232 | anyways |
| 05:08.20 | alpha232 | i got my hold music working now... what i need next is a way to control it lol |
| 05:09.05 | alpha232 | hmm |
| 05:09.06 | alpha232 | Nov 1 01:08:48 WARNING[12794]: res_musiconhold.c:852 moh_register: Unable to open pseudo channel for timing... Sound may be choppy. |
| 05:09.15 | J4k3 | chopped *and* screwed? |
| 05:09.25 | alpha232 | naw actually it sounds pretty good |
| 05:09.29 | J4k3 | haha |
| 05:11.35 | *** join/#asterisk quidpro (n=quid@CPE00131078ba5d-CM000f9f7eff1e.cpe.net.cable.rogers.com) |
| 05:11.49 | *** part/#asterisk quidpro (n=quid@CPE00131078ba5d-CM000f9f7eff1e.cpe.net.cable.rogers.com) |
| 05:12.04 | alpha232 | hmm now i am just trying to figure out how things work with inbound sip and where it goes |
| 05:22.29 | *** join/#asterisk TestMaster (n=TestMast@xplr-ts-w10-208-114-135-205.barrettxplore.com) |
| 05:23.01 | TestMaster | hello all quick question i am trying to install asterisk but its requesting zaptel but i cant install zaptel with out zaptel hardware |
| 05:23.34 | *** join/#asterisk anujsingh (n=root@59.90.65.14) |
| 05:26.22 | [TK]D-Fender | TestMaster, Sure you can, thats what ZTDUMMY is for. |
| 05:27.12 | watchy | goto bed tk |
| 05:27.48 | alpha232 | hmm anyway to do Midi for hold music? |
| 05:27.52 | alpha232 | *grin* |
| 05:28.16 | J4k3 | yeah |
| 05:28.23 | J4k3 | get one of those nice roland midi boxes |
| 05:28.24 | [TK]D-Fender | alpha232, Sure, stream from ALSA after rendering MIDI playback. |
| 05:28.28 | J4k3 | and hook it to your line-in |
| 05:28.33 | [TK]D-Fender | watchy, I found me a new toy :) |
| 05:28.36 | J4k3 | and make some mp3s ;) |
| 05:29.09 | [TK]D-Fender | And with the dwindling US$ cheaper still :D |
| 05:29.09 | alpha232 | hmm |
| 05:29.12 | alpha232 | no |
| 05:29.16 | tzafrir_home | you can use the original (non-native) moh to stream moh from a command |
| 05:29.23 | alpha232 | what i need is a midi player that dumps to stdout |
| 05:29.30 | tzafrir_home | and then stream it from something that plays midi |
| 05:30.03 | alpha232 | timidty |
| 05:30.21 | J4k3 | at least give them .mod |
| 05:30.30 | J4k3 | don't be totally evil! |
| 05:31.02 | alpha232 | rofl |
| 05:31.10 | alpha232 | midi is good with the right sound fonts |
| 05:31.34 | [TK]D-Fender | ok, it IS checkout time for me... later all |
| 05:32.06 | alpha232 | lol i think that scared him |
| 05:33.45 | J4k3 | haha no |
| 05:33.55 | J4k3 | (although that sounds pretty damned interesting!) |
| 05:34.07 | alpha232 | you don't know the half of it ;) |
| 05:34.10 | J4k3 | haha |
| 05:34.37 | alpha232 | argh this is difficult to understand the dial plan is so convoluted |
| 05:34.39 | J4k3 | mostly just feeling like shit because I'm going to spend 2 days under my car and about $600 on various crap it needs to make a 700 mile roundtrip to see my girlfriend |
| 05:34.46 | J4k3 | and to pass inspection (it ran out 34 minutes ago) |
| 05:34.54 | alpha232 | yours too? |
| 05:35.03 | alpha232 | mine ran out 1:34 ago |
| 05:35.08 | J4k3 | haha |
| 05:35.19 | alpha232 | and thats nothing, i drove 1700 miles, 27.5 hours to go see my boyfriend |
| 05:35.29 | J4k3 | I've got four bald tires and a fried cv joint |
| 05:35.55 | alpha232 | i need new tierods and my sidewalls are freying |
| 05:35.56 | J4k3 | and a failing cat I think |
| 05:36.20 | alpha232 | do you have FWD? |
| 05:36.29 | J4k3 | http://www.intrastar.net/~jsuter/stormy/1024071506.jpg = my junk in a junkyard! |
| 05:36.43 | J4k3 | yeah |
| 05:37.45 | alpha232 | hrrm this was working before now it doesnt wtf |
| 05:37.56 | J4k3 | its a hater |
| 05:38.07 | J4k3 | providing you with unreasonably large quantities of hate. |
| 05:38.07 | alpha232 | can you call 872868 |
| 05:38.12 | alpha232 | mm it is |
| 05:38.45 | J4k3 | I'm currently updating my old-ass trixbox box. |
| 05:38.49 | J4k3 | I Suspect its going to just explode. |
| 05:40.15 | alpha232 | k i'm outie |
| 05:50.02 | *** part/#asterisk Alowishus (n=jpenix@70.230.199.141) |
| 05:52.29 | *** join/#asterisk BBHoss_Work (n=hoss@216.186.235.254) |
| 05:57.19 | *** join/#asterisk J4k3 (n=jsuter@openwrt.us) |
| 05:57.26 | J4k3 | yay for scheduled maintenance. |
| 05:57.40 | *** join/#asterisk LakeSolon (n=blake@64-83-209-86.dhcp.stcd.mn.charter.com) |
| 05:58.53 | watchy | ill stab u |
| 05:58.59 | J4k3 | I'll stab u back |
| 05:59.09 | J4k3 | with my machete, man. |
| 05:59.09 | watchy | i just make install zaptel |
| 05:59.17 | watchy | without backing up nt zaptel.conf |
| 05:59.18 | J4k3 | I'll make install vxworks |
| 05:59.21 | watchy | is that bad |
| 05:59.24 | J4k3 | in your brain |
| 05:59.46 | J4k3 | normally apps don't overwrite a conf if its populated alreay |
| 05:59.47 | watchy | did the make install back it up anywhere? |
| 06:00.21 | watchy | well you would think that |
| 06:00.43 | J4k3 | if it didn't |
| 06:00.46 | J4k3 | that shit ain't cool |
| 06:00.56 | watchy | -rw-r--r-- 1 root root 9526 Oct 31 23:49 zaptel.conf |
| 06:01.13 | watchy | i think i got owned |
| 06:01.22 | J4k3 | thats fucked up |
| 06:01.51 | watchy | that was at a clients site to |
| 06:02.01 | watchy | that i've never seen their /etc/zaptel.conf |
| 06:02.01 | watchy | heh |
| 06:02.20 | watchy | oh well the other tech be here in an hour or so |
| 06:02.29 | watchy | i guess he can put that back it aint to serious i don't think |
| 06:03.10 | watchy | aslong as * don't restart or the system don't |
| 06:03.13 | watchy | i think it'll work fine |
| 06:03.58 | *** join/#asterisk Igbothom_III (n=Hilton@office.quarkgroup.com.au) |
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| 06:08.24 | *** part/#asterisk pussfeller (n=friendof@CPE-75-87-115-142.kc.res.rr.com) |
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| 06:47.00 | dlynes | watchy: dood...tar is your friend |
| 06:51.38 | *** join/#asterisk Juggie (n=Juggie@CPE001601df17fb-CM001a6682d7b6.cpe.net.cable.rogers.com) |
| 06:52.28 | Maliuta | star if you want to deal with acl's |
| 06:57.04 | tzafrir | watchy, ping |
| 07:06.04 | watchy | yo |
| 07:08.13 | *** join/#asterisk sergee (n=serg@voip1.west-call.com) |
| 07:13.59 | J4k3 | oh wow, switchvox |
| 07:14.08 | J4k3 | not only does it rhyme with trixbox, its the same crappy idea. |
| 07:14.35 | Igbothom_III | do they install *all* apps as user "trixbox" like trixbox does? |
| 07:14.43 | Igbothom_III | that's so bad |
| 07:15.52 | J4k3 | I don't find any user 'trixbox' on my old 1.2.0 trixbox install |
| 07:16.44 | Igbothom_III | all apps are running under that account |
| 07:17.53 | J4k3 | this is all running under the user 'asterisk' |
| 07:17.53 | J4k3 | ;) |
| 07:17.53 | Igbothom_III | ok |
| 07:17.53 | Igbothom_III | whatever :) |
| 07:17.53 | Igbothom_III | all running under one user |
| 07:17.53 | J4k3 | yeah |
| 07:17.53 | Igbothom_III | including SugarCRM |
| 07:17.59 | Igbothom_III | it is poor form |
| 07:18.09 | J4k3 | its trixbox... |
| 07:18.10 | J4k3 | ;) |
| 07:19.21 | Igbothom_III | agreed |
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| 07:50.05 | anujsingh | is astertest broken ? has anyone tried astertest? |
| 07:50.54 | anujsingh | i can see graphs when i dial manually , but not automatically same as shown in documentation of astertest. |
| 08:00.12 | *** part/#asterisk kiscokid (n=Ron_Laut@adsl-216-101-109-187.dsl.snfc21.pacbell.net) |
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| 08:08.32 | FlatFoot | morning all |
| 08:09.38 | *** join/#asterisk |omni| (n=rob@c-67-185-56-106.hsd1.wa.comcast.net) |
| 08:10.14 | *** join/#asterisk CBU[^_^]M`` (n=love@210.213.140.51) |
| 08:10.46 | FlatFoot | just been asked to quote for an install comprising 395 phones and * server |
| 08:10.59 | watchy | man thats pretty big |
| 08:11.04 | FlatFoot | anyone got any recommendations for size of server hardware etc |
| 08:11.15 | CBU[^_^]M`` | my xlite still cant connect using my internet IP huhuhuh |
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| 09:18.48 | marc7 | which subversion branch can I download the additional sound files that this page talks about: http://www.voip-info.org/wiki/view/Asterisk+sound+files+additional ? |
| 09:19.41 | marc7 | ah, i see. from http://downloads.digium.com/pub/telephony/sounds/ |
| 09:30.27 | roxlu | does someone knows if it is possible to 'tweak' the sound quality of a voip phone? |
| 09:34.21 | stimpie | roxlu, what do you mean by 'tweak' |
| 09:34.25 | stimpie | improve? |
| 09:34.38 | roxlu | yes |
| 09:36.01 | agx | roxlu, which phone? |
| 09:36.27 | roxlu | atcom 230 |
| 09:36.43 | sehh | q: can someone please remind me: FXO and FXS, which one connects to the telco and which one connects to a normal phone? |
| 09:37.04 | agx | roxlu, uhm, try different codec, some telephone is pretty buggy with codecs; for example GXP2000<-->GSM very bad |
| 09:38.10 | *** join/#asterisk _ys (n=yuri@236-069.nat.mns.ru) |
| 09:38.31 | roxlu | okay |
| 09:38.40 | roxlu | what codec is preferred |
| 09:40.17 | JT | atcom is a cheapie isn't it |
| 09:40.25 | agx | JT, yes, few $$$ |
| 09:40.29 | roxlu | well... 75,- euro :-) |
| 09:40.35 | agx | roxlu, G.711 A/U if you have bandwidth |
| 09:40.55 | roxlu | agx: how much bandwidth? |
| 09:41.27 | agx | roxlu, www.voip-info.org has a table about codec comparisition |
| 09:41.42 | roxlu | thanks |
| 09:41.46 | JT | G.711 is 85kbit/s with SIP overhead |
| 09:41.52 | JT | 64kbit/s for the codec alone |
| 09:44.16 | roxlu | my 'codec1' is now G.729 |
| 09:45.15 | JT | get a decent phone if you want to use G.729 |
| 09:45.31 | roxlu | ah so g.711 is the one I need to try? |
| 09:45.38 | JT | right |
| 09:45.48 | JT | it's the codec the pstn uses |
| 09:47.45 | agx | roxlu, if you want to use G729 with * you're forced to buy a license too, since there are patents issue on the codec (and on G723.1) |
| 09:48.33 | roxlu | oh |
| 09:49.52 | JT | G.723 sounds like arse anyway |
| 09:51.04 | stimpie | anybody knows something about the outboundproxy paramater in sip.conf I cant seem to get it working |
| 09:53.19 | agx | JT, i was trying to have a phone use 2 different codecs when calling * or passthru to a provider but seems the actual codec negotiation does not handle this, only way is using Set(SIP_CODEC=g729) on outbound calls |
| 09:54.33 | roxlu | and iLBC ? |
| 09:54.58 | JT | good luck finding a decent phone that supports that |
| 09:55.04 | JT | it pulls tonnes of cpu too |
| 09:55.10 | roxlu | ah |
| 09:55.29 | roxlu | and can't I change/tweak the "audio frames" settings for example? or the jitter size? |
| 09:56.26 | stimpie | roxlu, do a trace first and let wireshark give you some rtp statistics |
| 09:56.36 | agx | roxlu, for phones the only way is G711 A/U and G729 |
| 09:56.37 | roxlu | ok |
| 09:56.46 | JT | roxlu: is this only a lan? |
| 09:56.51 | agx | roxlu, for providers uses whatever save bandwith GSM, etc. |
| 09:56.54 | roxlu | no external |
| 09:57.01 | JT | oh ok |
| 09:57.22 | stimpie | from my experience the jitter buffer almost never causes audio quality problems |
| 09:57.51 | roxlu | stimpie: it's now "0" ? |
| 09:58.00 | stimpie | thats small |
| 09:58.04 | roxlu | :-0 |
| 09:58.08 | roxlu | what does that mean? |
| 09:58.23 | stimpie | dont know your phone it could mean 'automatic' |
| 09:58.38 | stimpie | dont think it means 0 |
| 09:58.45 | roxlu | no I mean whatdoes the jitter size and audio frames mean? |
| 09:59.08 | stimpie | normaly ever 20ms (depening on codec) a packet arrives |
| 09:59.25 | stimpie | if your network gets congested it could take 2ms extra |
| 10:00.06 | stimpie | jitter is the variation in the arrival time of packets |
| 10:01.12 | roxlu | ok |
| 10:02.38 | stimpie | unless 0 means automatic it could cause issues |
| 10:02.47 | roxlu | ok |
| 10:03.06 | roxlu | what would be a number to try? |
| 10:03.24 | stimpie | which codec? |
| 10:03.31 | *** join/#asterisk sergee (n=serg@voip1.west-call.com) |
| 10:03.43 | roxlu | g711a |
| 10:04.11 | stimpie | frame size = 02 |
| 10:04.16 | stimpie | 20 |
| 10:04.18 | roxlu | okay thanks |
| 10:04.26 | stimpie | so try multiples of 20 |
| 10:05.08 | stimpie | try keeping you total delay under 200ms |
| 10:09.10 | *** join/#asterisk eurodrew555 (n=eurodrew@83-244-164-130.cust-83.exponential-e.net) |
| 10:16.04 | *** join/#asterisk Dj_InstincT (n=marcus@87-194-124-8.bethere.co.uk) |
| 10:20.19 | eurodrew555 | anyone running asterisk svn TRUNK here ? |
| 10:21.08 | eurodrew555 | wondering if anyone can give some feedback as to it's stability versus SVN branch 1.4 |
| 10:32.21 | *** join/#asterisk |omni| (n=rob@c-67-185-56-106.hsd1.wa.comcast.net) |
| 10:35.00 | *** join/#asterisk cpm (n=Chip@pdpc/supporter/active/cpm) |
| 10:35.49 | *** join/#asterisk clandmeter (n=Carlo@81.175.82.2) |
| 10:37.12 | *** join/#asterisk fs-locaweb (i=fs-locaw@200.234.206.130) |
| 10:43.53 | *** join/#asterisk linxroute (n=VietPhon@222.252.108.5) |
| 10:50.17 | *** join/#asterisk _ys (i=ys@91.151.196.254) |
| 11:01.41 | *** join/#asterisk duki (n=duki@host-85-27-58-159.brutele.be) |
| 11:01.57 | J4k3 | DUKI (tm) |
| 11:01.59 | J4k3 | haha |
| 11:02.50 | J4k3 | uhm |
| 11:04.49 | *** join/#asterisk caio1982 (i=caio1982@CAcert-br/caio1982) |
| 11:06.38 | *** join/#asterisk myiagy (n=myiagy@201.56.112.108) |
| 11:25.51 | duki | hi J4k3 . |
| 11:31.02 | sehh | q: can someone please remind me: FXO and FXS, which one connects to the telco and which one connects to a normal phone? |
| 11:31.12 | Strom_M | FXO port points at the telco |
| 11:31.35 | Strom_M | remember: foreign exchange OFFICE |
| 11:35.46 | *** join/#asterisk waKKu (n=eurulo@unaffiliated/wakku) |
| 11:45.24 | *** join/#asterisk BeeBuu (n=chatzill@125.95.249.38) |
| 11:46.26 | BeeBuu | how can i set * answer income calling without delay (du....du)? |
| 11:50.11 | destructure | asterisk picks up the call immediately unless you Tell it to delay |
| 11:51.11 | BeeBuu | destructure:but when i call in ,i heard DU.... and * picks up... |
| 11:51.29 | *** join/#asterisk denon (n=denon@tooth.decay.org) |
| 11:51.29 | *** mode/#asterisk [+o denon] by ChanServ |
| 11:51.43 | *** join/#asterisk fordfrog (n=fordfrog@gentoo/developer/fordfrog) |
| 11:52.14 | BeeBuu | which config file let * delay? |
| 11:53.04 | *** join/#asterisk af_ (n=getsmart@88-149-240-124.dynamic.ngi.it) |
| 11:53.17 | BeeBuu | denon: can you help me please? |
| 11:54.29 | BeeBuu | which config file make * answer delay? |
| 11:54.56 | *** join/#asterisk allankardec (n=root@20150068138.user.veloxzone.com.br) |
| 11:55.36 | *** join/#asterisk duki (n=duki@host-85-27-58-159.brutele.be) |
| 11:55.46 | linxroute | immediate=yes |
| 11:55.50 | linxroute | zapata.conf |
| 11:57.04 | BeeBuu | Oh,asteriskNOW default: immediate=no |
| 11:57.16 | BeeBuu | thanks,linxroute |
| 11:58.44 | BeeBuu | linxroute: have i reload zapdata.conf? |
| 12:02.12 | destructure | oh, are we talking answering a "call", or dialing? |
| 12:02.39 | BeeBuu | answer a call in |
| 12:02.57 | destructure | sip? zap? iax? |
| 12:03.09 | BeeBuu | destructure: zap |
| 12:03.52 | destructure | and is the issue that you are hearing ringback before asterisk picks up? |
| 12:04.16 | ai-a | asterisk just stopped. asterisk -r fails to connect,, asterisk restart failed to stop some as its stopped.. see (http://pastbin.ca/756696) what logs should i view to find reasons for the quitting of the service ? |
| 12:04.34 | BeeBuu | i don't want to hear any ring when * picks up,can i do that? |
| 12:04.56 | ai-a | http://pastebin.ca/756696 |
| 12:05.09 | ai-a | BeeBuu: turn off your ringer. |
| 12:05.29 | BeeBuu | ai-a: how? which config file? |
| 12:05.38 | destructure | what kind of zap hardware? |
| 12:05.43 | ai-a | BeeBuu: you mean calling 'into' asterisk, you hear rinning in your ear.. but you want to hear what instead ? |
| 12:06.23 | BeeBuu | destructure: TDM400 |
| 12:07.07 | BeeBuu | ai-a: i don't want hear anything |
| 12:07.28 | *** join/#asterisk blitzrage (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 12:07.28 | *** mode/#asterisk [+o blitzrage] by ChanServ |
| 12:08.00 | linxroute | well |
| 12:08.13 | linxroute | this really depends on your telco lines |
| 12:08.28 | linxroute | needs to do |
| 12:08.37 | linxroute | fxotune -s 3 |
| 12:08.46 | linxroute | if you have three modules |
| 12:09.15 | BeeBuu | linxroute: could you teach me how to do? |
| 12:09.20 | destructure | nod. asterisk will answer as soon as it can, but you have call set up time. the remote side is probably playing ringback until the call is negotiated. Because it's analog, I think you are going to have trouble getting rid of All ringback |
| 12:09.50 | linxroute | well |
| 12:09.51 | linxroute | mine |
| 12:09.56 | linxroute | it's still ring |
| 12:09.57 | ai-a | BeeBuu: its their phone that is ringing, not the PBX. however you can answer the call, and play a empty wav file in an IVR / QUEUE. |
| 12:10.00 | linxroute | but only once |
| 12:10.07 | BeeBuu | hm.. that means at lease ring once? |
| 12:10.12 | linxroute | is that okay for you beebuu |
| 12:10.15 | linxroute | like i said |
| 12:10.21 | linxroute | it's really depends on your telco |
| 12:10.29 | linxroute | like provide callerid |
| 12:10.35 | linxroute | with tdm400 |
| 12:10.40 | linxroute | some country like mine |
| 12:10.57 | BeeBuu | ya, as mine |
| 12:11.02 | linxroute | it's still need to ring once then the caller id show up |
| 12:11.08 | *** join/#asterisk wick2o (n=wick2o@72.25.0.101.static.dejazzd.com) |
| 12:11.40 | linxroute | i tried with diffirent PXB |
| 12:11.47 | linxroute | to simulate the CO lines |
| 12:11.54 | linxroute | with Panasonix |
| 12:12.07 | linxroute | i can get it pickup immediately |
| 12:12.22 | linxroute | without a single ring |
| 12:13.24 | BeeBuu | what's CO lines? |
| 12:13.31 | linxroute | ( FXO ) |
| 12:13.38 | linxroute | Central Office line |
| 12:13.50 | linxroute | the line that telco bring to your office or your house |
| 12:14.06 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 12:14.06 | linxroute | where r u from, mate |
| 12:14.07 | destructure | the CO is probably where your rinback is coming from |
| 12:15.35 | BeeBuu | so if i want that,i must try another PXB? |
| 12:15.47 | linxroute | nope |
| 12:15.52 | linxroute | not really |
| 12:16.01 | linxroute | what's country you from ? |
| 12:16.09 | BeeBuu | CHINA |
| 12:16.16 | linxroute | oh |
| 12:16.16 | destructure | ni hao! |
| 12:16.37 | BeeBuu | hi,destructure |
| 12:17.02 | linxroute | => from vietnam |
| 12:17.21 | linxroute | beebuu , what's is your computer hardware ? |
| 12:17.28 | linxroute | this might effect as well |
| 12:18.00 | BeeBuu | <PROTECTED> |
| 12:18.01 | BeeBuu | Linux localhost.localdomain 2.6.19.7-0.4.gcc3.4.x86.i686 #1 Tue May 1 13:00:52 EDT 2007 i686 i686 i386 GNU/Linux |
| 12:18.03 | BeeBuu | Uptime: |
| 12:18.05 | BeeBuu | 20:17:37 up 1:35, 1 user, |
| 12:18.06 | BeeBuu | Load Average: 0.00, 0.00, 0.00 |
| 12:18.08 | BeeBuu | Asterisk Build: |
| 12:18.09 | BeeBuu | Asterisk 1.4.5 |
| 12:18.11 | BeeBuu | Asterisk GUI-version Revision: 11255 $ |
| 12:18.12 | BeeBuu | Server Date & TimeZone: |
| 12:18.14 | BeeBuu | ËÄ 11ÔÂ 1 20:17:37 CST 2007 |
| 12:18.15 | BeeBuu | Hostname: |
| 12:18.17 | BeeBuu | localhost.localdomain |
| 12:18.21 | linxroute | no |
| 12:18.26 | linxroute | i meant HARDWARE |
| 12:18.33 | *** join/#asterisk L|NUX (n=linux@unaffiliated/lnux/x-10290) |
| 12:18.33 | linxroute | like speed of your CPU |
| 12:18.36 | linxroute | RAM |
| 12:18.38 | linxroute | HDD |
| 12:18.44 | linxroute | things like that |
| 12:19.21 | linxroute | anyway |
| 12:19.22 | BeeBuu | Memory Usage: |
| 12:19.23 | BeeBuu | <PROTECTED> |
| 12:19.25 | BeeBuu | Mem: 246636 170260 76376 0 25856 64696 |
| 12:19.26 | BeeBuu | -/+ buffers/cache: 79708 166928 |
| 12:19.28 | BeeBuu | Swap: 522072 0 522072 |
| 12:19.51 | linxroute | does it still ring now ? |
| 12:20.05 | destructure | ~pb |
| 12:20.07 | jbot | A Pastebin is a place to paste your stuff without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste |
| 12:20.12 | BeeBuu | yes,it ring once |
| 12:20.59 | BeeBuu | destructure: i got that. |
| 12:21.29 | *** join/#asterisk dlynes (n=dlynes@d154-20-9-152.bchsia.telus.net) |
| 12:21.34 | *** join/#asterisk Uatec (n=uatecuk@adsl.ntsols.com) |
| 12:21.36 | Uatec | Hello |
| 12:21.46 | dlynes | hello, uatec |
| 12:21.52 | Uatec | I'm dialing a SIP trunk that I have setup between my asterisk box and another one |
| 12:21.58 | Uatec | but when i dial it says: Got SIP response 482 "Loop Detected" back from 192.168.232.52 |
| 12:22.03 | linxroute | try to call several times |
| 12:22.03 | Uatec | where that IP is my asterisk box |
| 12:22.05 | *** join/#asterisk lirakis (n=eric@65.200.191.253) |
| 12:22.11 | linxroute | you will not hear it ring |
| 12:22.14 | Uatec | it's essentially saying "lookback detected from localhost" |
| 12:22.18 | *** join/#asterisk styelz (n=yoohoo@2001:388:c098:0:0:0:0:1) |
| 12:22.24 | dlynes | Does anyone have an example of a zaptel.conf/zapata.conf file setup I can use, if I want to have another asterisk box connect via a pri channel? |
| 12:23.02 | *** join/#asterisk pepo-- (n=pepOSX@190.72.144.104) |
| 12:23.24 | dlynes | Uatec: that's because you've got something set up in your dialplan to send any call coming in on your sip channel back out on your sip channel to the same destination as the original source |
| 12:23.40 | Uatec | oh how sstrange |
| 12:23.50 | Uatec | i was dialing "SIP/asterisk/123" |
| 12:23.52 | dlynes | Uatec: in this case, the source and target are both local host |
| 12:23.59 | Uatec | but there is no sip account called "asterisk" |
| 12:24.10 | Uatec | so i don't know how that was working, even to the exten of trying to setup a loop back |
| 12:24.36 | dlynes | Uatec: try sip show peers and sip show users |
| 12:24.46 | dlynes | Uatec: you might see one listed as 'asterisk' |
| 12:26.12 | *** join/#asterisk saftsack (n=saftsack@pD9E0551F.dip.t-dialin.net) |
| 12:26.12 | *** join/#asterisk ming_zym (n=ming_zym@123.103.29.249) |
| 12:29.31 | *** join/#asterisk kclaussen (n=kclausse@204.13.224.242) |
| 12:39.34 | alpha232 | hrrrm |
| 12:39.55 | *** join/#asterisk fiXXXerMet (n=kjohnson@69.85.26.2) |
| 12:40.36 | *** join/#asterisk bantu (n=Miranda@p54A3295D.dip0.t-ipconnect.de) |
| 12:45.00 | *** join/#asterisk oej (n=olle@64.251.114.2) |
| 12:46.39 | *** join/#asterisk oej_ (n=olle@64.251.114.2) |
| 12:48.04 | *** join/#asterisk [TK]D-Fender (n=joe@MTRLPQ02-1177745839.sdsl.bell.ca) |
| 12:49.05 | *** join/#asterisk anonymouz666 (n=anonymou@201.19.138.80) |
| 12:56.47 | *** join/#asterisk blop (n=blop@16-186-112-217.dyn.adsl.belcenter.be) |
| 12:56.55 | blop | hey |
| 12:57.21 | *** join/#asterisk Tagor (n=none@s55928c6d.adsl.wanadoo.nl) |
| 12:57.37 | blop | where can i find the source of the IAXy firmware available here : http://svn.digium.com/view/asterisk/trunk/contrib/firmware/iax/iaxy.bin?view=log ? |
| 12:59.37 | Tagor | Hi |
| 12:59.45 | Tagor | I am trying to fetch the number of the called party |
| 12:59.50 | Tagor | But for some reason it goes wrong |
| 12:59.59 | Tagor | Does anyone know what I am doing wrong? |
| 13:00.00 | Tagor | http://rafb.net/p/JdvBkN95.html |
| 13:00.16 | Tagor | I copied the part of my extensions.conf and what the CLI says |
| 13:00.34 | Corydon76-dig | blop: you cannot get at the source of the iaxy firmware |
| 13:00.38 | Tagor | For some reason this worked yesterday, don't ask me why :P |
| 13:01.50 | [TK]D-Fender | Tagor: exten => s,2,Set(FOO1=${CUT(FOO1,-,1}) <---- bad brackets. |
| 13:02.09 | blop | Corydon76-dig: too bad :( why is that closed source ? |
| 13:02.39 | Corydon76-dig | You were expecting it to be in C? |
| 13:03.08 | blop | another language is good too :p |
| 13:03.17 | Corydon76-dig | How about assembly? |
| 13:03.32 | blop | i like asm :) |
| 13:03.50 | Corydon76-dig | Well, there you go |
| 13:04.10 | Tagor | [TK]D-Fender >> I got this from http://www.voip-info.org/wiki/view/DIALEDPEERNUMBER |
| 13:04.15 | blop | but i doubt they've written the whole firmware in ASM .. :) |
| 13:04.15 | Corydon76-dig | The main reason it's not open source is to avoid clones |
| 13:04.34 | Tagor | [TK]D-Fender >> So I guess it's wrong there too? It should be Set(FOO1=${CUT(FOO1,-,1)}) right? |
| 13:04.39 | [TK]D-Fender | Tagor: So? they made a mistake and you copied it. |
| 13:04.41 | Corydon76-dig | The protocol it runs is open, though |
| 13:04.45 | blop | yeah: i tought of that too :) |
| 13:04.55 | [TK]D-Fender | Tagor: Sorry you need a working brain of your own. |
| 13:05.23 | alpha232 | hmm are there some better ways of handeling MOH? |
| 13:05.43 | blop | but it doesnt seem to be much work done lately on the iaxy firmware .. :) |
| 13:06.03 | Corydon76-dig | blop: the only way to get the source of the IAXy is to execute an agreement with Digium sales. I don't know what criteria, but that's where you'll have to start |
| 13:06.07 | tzafrir | blop, go chase after the source of that free iax2 device chipset |
| 13:06.19 | tzafrir | (ga168? I forgot the name) |
| 13:06.28 | [TK]D-Fender | PA1668 |
| 13:06.33 | blop | some says it could perform dns lookups :p |
| 13:06.37 | [TK]D-Fender | pa1688 rather |
| 13:06.43 | tzafrir | supposed to be free, but for some reason you cannot download it from anywhere |
| 13:06.47 | tzafrir | right |
| 13:07.24 | blop | :( |
| 13:07.25 | Corydon76-dig | You can download chips, now? How times have changed. ;-) |
| 13:07.52 | pigpen | Corydon76-dig, where can I download a new quad core? |
| 13:07.53 | pigpen | :) |
| 13:08.01 | blop | also, i'm having a caller id issue with the iaxy |
| 13:08.20 | tzafrir | I think transmeta tried that idea, but noone really bought it |
| 13:08.40 | blop | which cidsignaling is the iaxy supposed to use ? |
| 13:09.14 | Corydon76-dig | blop: bell |
| 13:09.14 | blop | :( |
| 13:09.14 | Corydon76-dig | and no, you can't change it |
| 13:09.24 | blop | too bad :( |
| 13:09.29 | pigpen | This is a bit beyond my function knowledge. Is there any way to combine these two "cuts" into a sinle line? |
| 13:09.29 | pigpen | exten => 123,n,Set(EXTINFO18=${CUT(EXTINFO,":",18)}) ; where EXTINFO is a realtime call |
| 13:09.29 | pigpen | exten => 123,n,Set(FOLLOWME=${CUT(EXTINFO18,"=",2)}) |
| 13:09.40 | [TK]D-Fender | Corydon76-dig: I'm just waiting for my replacement Heisenberg compensator's to arrive so I can put my Transporter back into service again :) |
| 13:10.05 | blop | why cant we change that? is it hardware dependent ? |
| 13:10.21 | Corydon76-dig | In a word, ys |
| 13:10.25 | Corydon76-dig | yes |
| 13:10.31 | blop | dam :( |
| 13:10.37 | [TK]D-Fender | IAXY = suck |
| 13:10.47 | Corydon76-dig | iaxy works fine |
| 13:10.59 | Corydon76-dig | I'm using them to control a PA system over fiber |
| 13:11.12 | Tagor | [TK]D-Fender >> I know changed it to above which I think is correct. But then I get this output: http://rafb.net/p/satp6l94.html |
| 13:11.28 | Tagor | [TK]D-Fender >> so it only outputs the extension instead of the called number |
| 13:11.40 | Corydon76-dig | shower time |
| 13:11.48 | blop | and what about the new hardware revision s101l? same issues? |
| 13:12.03 | tzafrir | BTW: Is there any other device besides the iaxy that uses iax provisioning? |
| 13:12.32 | pigpen | tzafrir, sure, build you own. |
| 13:12.41 | [TK]D-Fender | Tagor: ENTIRELY wrong wording there... |
| 13:12.44 | pigpen | but commercially, not many. |
| 13:13.01 | Tagor | [TK]D-Fender >> sorry this one is correct: http://rafb.net/p/myGeY526.html |
| 13:13.09 | tzafrir | pigpen, I considered it slightly but then noticed it supports one account per device |
| 13:13.14 | [TK]D-Fender | Tagor: And you have no clue what you're doing. The channel name does NOT contain the NUMBER you dial. |
| 13:14.05 | pigpen | tzafrir, you could get a soekris, load it up, and stick in a 4 port fxs card....now you do trunking back to the mothership. |
| 13:14.16 | Tagor | [TK]D-Fender >> According to voip-info this is the way to get the called number instead of using the broken var DIALEDPEERNUMBER |
| 13:14.33 | [TK]D-Fender | Tagor: -- Executing Set("SIP/sipprovider-3fd0", "FOO1=sipprovider-3fd0") in new stack <--- you see the channel name in every CLI execution of your dialplan there. it does NOT hold the number. You are completely wASTING YOUR TIME AND LOOKING A LITTLE SLOW... |
| 13:15.19 | Tagor | [TK]D-Fender >> 'FOO1 will now contain the dialed peernumber of the current SIP channel' .... voip-info says |
| 13:15.48 | mocker | [TK]D-Fender: I spent 45 minutes trying to figure out a dialplan customer last night and it was just a damn typo. |
| 13:15.53 | Tagor | [TK]D-Fender >> Or do I misunderstand something? |
| 13:15.58 | mocker | er s/customer/error/ |
| 13:16.17 | [TK]D-Fender | Tagor: vOip-INFO IS **wrong** |
| 13:16.43 | [TK]D-Fender | Tagor: Wake up and realize what you're DOING and that that article is contradicting itself. |
| 13:16.45 | destructure | morning, [TK]D-Fender |
| 13:17.05 | Tagor | [TK]D-Fender >> What else would be the way to get the number in a macro? |
| 13:17.20 | [TK]D-Fender | Tagor: Page title says "DIALEDPEERNUMBER" and in the TEXT it says : This variable should contain the number of the called party. Unfortunately it is broken for now and only contains the second part of the internal/temporary channel information (the part after the technology ie gs1-8b21 if the DIALEDPEERNAME was 'SIP/gs1-8b21). |
| 13:17.32 | [TK]D-Fender | Tagor: DIALEDPEERNAME! |
| 13:17.39 | *** join/#asterisk chodorenko (n=chodoren@etm005.nl.ded.neolocation.net) |
| 13:17.45 | chodorenko | Hi All |
| 13:18.10 | [TK]D-Fender | destructure: *yawn* |
| 13:19.18 | chodorenko | plese get me link to manual howe to use subscriptions in asterisk |
| 13:20.27 | pigpen | hmm..I don't feel like a secretary today. |
| 13:20.56 | cpm | you'll do it! and you'll like it! |
| 13:21.10 | destructure | please get me some coffee |
| 13:21.32 | chodorenko | i add in sip.conf line "Subscribecontext=subscr_cont" , in context i add line for hint my number |
| 13:22.32 | chodorenko | and call to different number , in sip show subscriptions my status is idle, Whu not change to dial? |
| 13:22.50 | pigpen | cpm, yeah..no kidding. |
| 13:22.52 | pigpen | sugar in that? |
| 13:23.10 | cpm | no thank you. |
| 13:23.32 | Tagor | [TK]D-Fender >> ${DIALEDPEERNAME} is also empty |
| 13:23.50 | mocker | I <3 sangoma vs digium threads on asterisk-users |
| 13:23.54 | destructure | soy milk, no sugar |
| 13:23.54 | mocker | They are always so entertaining. |
| 13:25.51 | Tagor | [TK]D-Fender >> voip-info also says; Unfortunately it is broken for now |
| 13:26.16 | pigpen | well, I normally don't ask the same question twice in the same day, in fear of getting flamed, but, is it possible to "nest" two cut functions in the same line (ie: extension priority)? |
| 13:26.47 | destructure | pigpen: have you tried it? |
| 13:26.49 | pigpen | Just trying to clean/shorten/simplify my dialplan. |
| 13:26.59 | destructure | it sounds easy to try |
| 13:27.31 | destructure | do you want them literally (nested (inside each other)) or just two in one line |
| 13:27.44 | pigpen | No..I don't know what the dam symbol is to do it? Bash is "|" |
| 13:27.48 | destructure | well, not each other |
| 13:28.03 | alpha232 | anyone know what the source was for the default MOH? |
| 13:28.06 | destructure | wait, you want to pipe it? heh |
| 13:28.37 | pigpen | ie: |
| 13:28.38 | pigpen | exten => 123,n,Set(EXTINFO18=${CUT(EXTINFO,":",18)}) |
| 13:28.38 | pigpen | exten => 123,n,Set(FOLLOWME=${CUT(EXTINFO18,"=",2)}) |
| 13:28.48 | pigpen | ^^^make one line |
| 13:28.51 | Tagor | Anyone know how to get the called number? |
| 13:29.07 | *** join/#asterisk socken23 (n=socken@ip-213-189-154-029.fix.magnet.ch) |
| 13:29.08 | destructure | try it out. report back the result |
| 13:29.15 | destructure | it should be easy enough to try |
| 13:29.39 | *** join/#asterisk felix_da_catz (n=felix@PixNationalGroup.phonoscope.com) |
| 13:29.53 | pigpen | Tagor, it would be the ${EXTEN} variable. |
| 13:29.58 | *** join/#asterisk coppice (n=chatzill@39.192.17.210.dyn.pacific.net.hk) |
| 13:30.10 | socken23 | Hi! I can't start asterisk. 'chan_zap.c: unable to open channel 1: no such device or address' but the card is here, modules are loaded.... |
| 13:30.12 | socken23 | any idea? |
| 13:30.13 | krdian_ | Tagor: try to get it from CDR |
| 13:30.21 | pigpen | destructure, so pipe may be it here...hm.. |
| 13:30.51 | Katty | [TK]D-Fender: Starting Zap/10-1 at zapG207,s,1 failed so falling back to exten 's' <- i need help with this... |
| 13:32.24 | pigpen | Katty, something doesn't look right with the "zapG207,s,1" and the 's' is the fall back extension when all else fails. |
| 13:32.45 | Katty | so that's extensions.conf it's yipping about. |
| 13:32.46 | pigpen | I haven't seen TK for a bit. |
| 13:32.51 | pigpen | yeah. |
| 13:32.54 | Katty | kk |
| 13:33.33 | *** join/#asterisk gardo (n=gardo@124.107.39.47) |
| 13:33.34 | *** join/#asterisk michael-i (n=michael-@W88f2.w.pppool.de) |
| 13:33.42 | Katty | there we go.. |
| 13:34.06 | pigpen | cool. |
| 13:35.17 | Katty | problem solved (= |
| 13:35.35 | pigpen | no prob..I am useful once and a while. :) |
| 13:37.30 | tzafrir | socken23, cat /proc/zaptel/1 |
| 13:37.53 | Katty | if i want *80 to go the voicemail program |
| 13:37.55 | socken23 | Span 1: ZTDUMMY/1 "ZTDUMMY/1 1" |
| 13:37.58 | Katty | it's just exten => *80,1,etc |
| 13:37.59 | socken23 | that looks bad :-) |
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| 13:38.42 | tzafrir | socken23, ls /proc/zaptel |
| 13:38.49 | [TK]D-Fender | Katty: Fix your context in zapata.conf |
| 13:38.54 | [TK]D-Fender | Katty: (Mew) |
| 13:38.54 | socken23 | tzafrir: 1 |
| 13:39.02 | pigpen | Katty, exten => *98,n,VoiceMailMain(@from-sip) |
| 13:39.06 | [TK]D-Fender | pigpen: I'm here daily, and started an hour ago |
| 13:39.08 | pigpen | ^^something like that possibly. |
| 13:39.23 | tzafrir | right. So the driver has not registered a span. Probably has not loaded or did not find the device |
| 13:39.29 | tzafrir | socken23, what device is that? |
| 13:39.39 | pigpen | Yeah...I figured you were going to meditate after the "DIALEDPEERNAME" thing. |
| 13:39.42 | socken23 | a junghanns with two channels |
| 13:39.50 | socken23 | BRIDuo |
| 13:40.03 | [TK]D-Fender | Tagor: Those variables don't even exist in 1.4 anymore. |
| 13:40.19 | alpha232 | hrrm nothing like finding out your pre-installed music on hold is copyrighted |
| 13:40.22 | tzafrir | socken23, what version of bristuff do you use? |
| 13:40.47 | [TK]D-Fender | Tagor: And the WIKI has a lot of 1.0 stuff that is useless now. |
| 13:40.53 | tzafrir | some older versions of qozap still did not recognize that newer hardware |
| 13:41.18 | socken23 | tzafrir: mhh.. where can I look that up? |
| 13:41.18 | alpha232 | morning tzafrir |
| 13:41.24 | Katty | [TK]D-Fender: my context in zapata was fine |
| 13:41.32 | tzafrir | socken23, what version of zaptel is it? |
| 13:41.35 | Katty | [TK]D-Fender: but it was fixed, nonetheless. |
| 13:42.52 | Katty | [TK]D-Fender: also! |
| 13:42.55 | Katty | [TK]D-Fender: good morning!! :P |
| 13:43.01 | Kobaz | [TK]D-Fender: poke |
| 13:43.25 | Kobaz | [TK]D-Fender: so the polycom 400's i have use H323... |
| 13:43.35 | *** join/#asterisk anthm (n=anthm@CPE-72-131-113-50.wi.res.rr.com) |
| 13:43.35 | *** mode/#asterisk [+o anthm] by ChanServ |
| 13:43.35 | alpha232 | hmmm Is there a way to capture digits while on hold? |
| 13:43.45 | Katty | morning anthm! |
| 13:43.49 | [TK]D-Fender | Kobaz: FUBAR-d. Big-time. |
| 13:43.53 | Kobaz | alpha232: trying to detect angry customers? |
| 13:43.53 | anthm | hiya |
| 13:43.58 | Kobaz | [TK]D-Fender: but but |
| 13:44.13 | [TK]D-Fender | alpha232: Nope. |
| 13:44.31 | mocker | Maybe a fake hold dropping them into a background? |
| 13:44.32 | [TK]D-Fender | Kobaz: If you like anal, I don't want to know about it.... |
| 13:44.35 | socken23 | tzafrir: I'm still searching for the info :-) |
| 13:44.45 | Kobaz | [TK]D-Fender: they were originally produced for shoretel |
| 13:44.52 | alpha232 | Kobaz: naw more like change hold music |
| 13:45.02 | [TK]D-Fender | Kobaz: Then consider them "beached" |
| 13:45.07 | mocker | Press 1 for easy listening, Press 2 for death metal. |
| 13:45.10 | tzafrir | socken23, I asked you some very specific questions |
| 13:45.11 | alpha232 | mocker: not qauite |
| 13:45.17 | Kobaz | alpha232: press 1 for modern rock |
| 13:45.22 | alpha232 | mocker: more like, while it's playing allowing them to change channels |
| 13:45.42 | socken23 | tzafrir: OK, here we are: zaptel 1.2.10 |
| 13:45.55 | alpha232 | so they have the choice of say, channels 1-6 |
| 13:46.03 | alpha232 | and can change during the hold |
| 13:46.23 | socken23 | tzafrir: Aha, and bri-stuff 0.3.0-PRE-1y |
| 13:46.23 | tzafrir | socken23, try a more recent version of Zaptel / bristuff |
| 13:46.33 | tzafrir | socken23, check the changelog: |
| 13:46.52 | tzafrir | http://updates.xorcom.com/astribank/bristuff/CHANGES |
| 13:47.05 | socken23 | Thanks, I'll check that |
| 13:47.30 | Kobaz | [TK]D-Fender: :( |
| 13:47.35 | Kobaz | [TK]D-Fender: free phones :( |
| 13:47.38 | tzafrir | You can also try a newer version of bristuff simply using http://updates.xorcom.com/iso/live-1.1.0.iso |
| 13:48.05 | [TK]D-Fender | Kobaz: Like I said, go ask for your money back :) |
| 13:48.39 | socken23 | tzafrir: Ah, that's a good idea... |
| 13:48.44 | socken23 | That will save me some time! |
| 13:49.22 | alpha232 | so who wants to talk BRI NI1 :D |
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| 13:53.43 | Kobaz | [TK]D-Fender: heh |
| 14:00.34 | *** part/#asterisk myiagy (n=myiagy@201.56.112.108) |
| 14:02.55 | socken23 | mhh.. strange, I'm trying to install a new version of BRIstuff now. But 'genzaptelconf -sdMv' is not working! |
| 14:03.02 | socken23 | -sdM are not valid options! |
| 14:04.28 | *** join/#asterisk mindCrime (n=chatzill@66.83.208.218.nw.nuvox.net) |
| 14:09.00 | socken23 | aha, running the ./install.sh script returns './compile.sh: line 78: syntax error near unexpected token 'fi' ' .. is this a bug?? |
| 14:11.00 | BeeBuu | please someone teach me how to playback callerid? |
| 14:11.37 | pigpen | BeeBuu, see the wiki... |
| 14:12.56 | BeeBuu | voip-info.org ? |
| 14:13.01 | pigpen | yeah. |
| 14:14.33 | [TK]D-Fender | ~book |
| 14:14.34 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf |
| 14:14.35 | socken23 | it was indeed a bug... found in config.sh |
| 14:15.11 | pigpen | [TK]D-Fender, any way to nest "cut" functions? after this I give up and do it one per line. |
| 14:15.43 | [TK]D-Fender | pigpen: Dunno... |
| 14:16.02 | pigpen | k. If I find anything revolutionary, I'll let you know. |
| 14:16.02 | pigpen | :) |
| 14:16.21 | blitzrage | pigpen: just nest them.... |
| 14:16.49 | blitzrage | Set(MY_VAL=${CUT(CUT(foo_var,-,3),&,2)}) |
| 14:17.06 | pigpen | Thank You. |
| 14:17.14 | pigpen | Thank You. |
| 14:17.26 | pigpen | I have been trying them in parallel, not inside each other. |
| 14:17.31 | blitzrage | ya, that won't work |
| 14:17.40 | blitzrage | you have to pass the value to the other CUT() function |
| 14:18.00 | pigpen | I am trying to split up a realtime result, but I have been doing it in 2 lines. |
| 14:18.10 | blitzrage | nothing wrong with doing it in 2 lines |
| 14:18.16 | blitzrage | probably makes it more clear anyways |
| 14:18.31 | blitzrage | although if you look at my DP's, I always nest functions :) |
| 14:18.59 | pigpen | yeah, nesting makes more sense to me...just call the realtime once, then the final value once. |
| 14:19.26 | blitzrage | well, you can just call RT once and save the value into a variable then parse on it |
| 14:19.37 | pigpen | right. |
| 14:19.42 | pigpen | no need to call realtime each time. |
| 14:19.49 | pigpen | have you worked much with realtime? |
| 14:20.39 | socken23 | hey guys, I still can't start asterisk with the newest bristuff drivers... I get a 'ZT_SPANCONFIG failed on span 1: Invalid argument (22)' |
| 14:21.38 | socken23 | and cat /proc/zaptel/1 still only shows a ZTDUMMY device... |
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| 14:22.28 | *** join/#asterisk jcmoore (n=jcmoore@unaffiliated/tgrman) |
| 14:22.28 | pigpen | blitzrage, thanks again, that worked great. |
| 14:23.39 | cpm | you see a guru around here, you kick his *ss |
| 14:23.51 | *** join/#asterisk mocker (n=user@198.247.173.227) |
| 14:24.04 | [TK]D-Fender | pigpen: I have a guru or two on the back-burner but they are "job based" only :) |
| 14:24.15 | mocker | Is rxgain on a tdm400p the volume a user hears for people who talk to them? |
| 14:24.34 | blitzrage | pigpen: I know a thing or two about realtime, but like [TK]D-Fender said, I'd like to get paid :) |
| 14:24.41 | blitzrage | <-- evil consultant |
| 14:25.35 | ai-a | where's an expert ? |
| 14:25.45 | tzanger | nobody is an expert here |
| 14:25.49 | *** join/#asterisk fskrotzki (n=fskrot@host198.textwise.com) |
| 14:25.49 | tzanger | we're all just consultants |
| 14:25.56 | blitzrage | :) |
| 14:26.30 | coppice | i feel the need to shower if I call myself a consultant |
| 14:26.44 | FlatFoot | ooooh dirty word |
| 14:26.50 | blitzrage | I just sit in the shower all day with my water proof laptop |
| 14:27.15 | ai-a | my boss considers an expect to be someone that will know something |
| 14:28.01 | coppice | we all know something. the question is whether its something relevant |
| 14:28.15 | blitzrage | I know lots about realtime, but not every single thing |
| 14:28.22 | tzanger | blitzrage: damn, man, that's a picture I just did not want this morning |
| 14:28.39 | blitzrage | tzanger: don't lie. no one believes you. |
| 14:29.00 | tzanger | ok I lied, I didn't just not want the picture this morning. I didn't want it, ever |
| 14:29.16 | blitzrage | :) |
| 14:30.04 | cpm | my boss considers anything written by anyone who gushes over Apple offerings to be an expert |
| 14:30.55 | tzanger | so put asterisk on a mac mini |
| 14:31.13 | pigpen | My question is not functional related..I am running it in production. |
| 14:31.19 | [TK]D-Fender | And prove him wrong as can be in 1 shot! :p |
| 14:31.27 | *** join/#asterisk linxroute (i=linxrout@117.0.17.61) |
| 14:31.34 | pigpen | My issue is that it seems that it broke after 1.4.11 |
| 14:31.51 | pigpen | is this confimed, did something break, something change? |
| 14:32.55 | blitzrage | pigpen: your question does not have an answer -- please restate with specifics |
| 14:33.07 | blitzrage | "seems that it broke" is slightly too general |
| 14:33.26 | pigpen | ok..seems realtime broke (ie: does not work, fails, el no worke) after 1.4.11 |
| 14:33.32 | tzanger | blitzrage: what kind of consultant are you? the clear anser is "perhaps." |
| 14:33.40 | blitzrage | tzanger: yes. |
| 14:33.56 | pigpen | shit..I used "seems" |
| 14:34.05 | pigpen | ok..realtime broke (ie: does not work, fails, el no worke) after 1.4.11 |
| 14:34.08 | blitzrage | I've used realtime after 1.4.11 with no issues |
| 14:34.22 | pigpen | mysl or postgres |
| 14:34.23 | pigpen | ? |
| 14:34.34 | blitzrage | I just used it yesterday with the code from the audiohooks-1.4 branch |
| 14:34.38 | blitzrage | pgsql of course |
| 14:34.40 | cpm | [TK]D-Fender, proof is meaningless to someone who reads advertising copy (read 'reviews') and considers them to be peer reviewed scientific papers |
| 14:34.43 | pigpen | same here. |
| 14:34.53 | pigpen | hmm... |
| 14:34.59 | [TK]D-Fender | cpm : |
| 14:35.19 | pigpen | did any syntax change? |
| 14:35.42 | [TK]D-Fender | cpm : "peer pressure" (or complete lack thereof) One great benifit of my God complex :D |
| 14:35.51 | cpm | [TK]D-Fender, good! |
| 14:35.59 | cpm | handy, that. |
| 14:36.39 | coppice | pier pressure - will I be in time for the next ferry |
| 14:36.53 | cpm | nice |
| 14:37.11 | coppice | pier to pier is very important for us |
| 14:38.21 | coppice | though since they built the tunnel its no longer critical |
| 14:38.50 | pigpen | blitzrage, yeah, I had it all running fine up through 1.4.11, then upgraded to 1.4.13 using the same patches and hooks as always, and the realtime call never makes it to postgres (via postgres debug). |
| 14:39.15 | pigpen | after beating my head against it for 2 hours, downgraded to 1.4.11 and all is fine. |
| 14:39.36 | pigpen | So to your knowledge, no syntax change? |
| 14:39.57 | blitzrage | none |
| 14:40.08 | pigpen | k....tks, helps get one piece out of the puzzle. |
| 14:40.58 | pigpen | fyi, I like getting paid too... |
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| 14:52.03 | anonymouz666 | chan_zap.c: Failed to read gains: Invalid argument -> this sucks. |
| 14:52.14 | anonymouz666 | can't find where is the error. |
| 14:52.33 | anonymouz666 | http://lists.digium.com/pipermail/asterisk-users/2007-January/176653.html -> same thing |
| 14:52.39 | *** join/#asterisk peanut- (n=tokarev@cpe-70-113-100-193.austin.res.rr.com) |
| 14:53.11 | Katty | anonymouz666: :> |
| 14:53.31 | [TK]D-Fender | anonymouz666: pastebin your zapata.conf |
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| 14:56.48 | Katty | twisted: ping? |
| 14:57.42 | anonymouz666 | [TK]D-Fender: http://pastebin.ca/757671 |
| 14:59.14 | anonymouz666 | I can't find anything wrong with this file. |
| 14:59.58 | [TK]D-Fender | anonymouz666: not sure about JB stuff belonging in there...and your channels could be cleaned up |
| 15:00.16 | anonymouz666 | I know about the X-Y, but that's not the issue here |
| 15:00.34 | blitzrage | anonymouz666: fyi -- you could simplify that last line with: channel => 1-31 |
| 15:00.49 | blitzrage | oh... you've got 16 missing in there :) |
| 15:01.04 | blitzrage | then 2 lines would do it :) |
| 15:01.49 | anonymouz666 | that's bizarre chan_zap.c can't just read the gains. |
| 15:01.57 | anonymouz666 | tzafrir? |
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| 15:02.12 | tzafrir | anonymouz666, hmmm |
| 15:02.59 | Katty | blitzrage: ! |
| 15:03.02 | Katty | blitzrage: YOU! |
| 15:03.05 | Katty | blitzrage: are about to be HUGGED |
| 15:03.06 | blitzrage | Katty: ME! |
| 15:03.10 | anonymouz666 | any suggestion? |
| 15:03.31 | *** join/#asterisk putnopvut (i=putnopvu@nat/digium/x-98a1f7b0089ab97d) |
| 15:04.20 | blitzrage | lol |
| 15:04.31 | Katty | blitzrage: you're not in huntsville, are you? |
| 15:04.37 | blitzrage | nopers |
| 15:04.38 | Katty | or canada? |
| 15:04.40 | blitzrage | unfortunately not |
| 15:04.42 | blitzrage | I'm in Toronto |
| 15:04.43 | Katty | oh. |
| 15:04.44 | Katty | i see. |
| 15:05.13 | Katty | for some reason, i couldn't remember ^_- |
| 15:05.21 | Katty | but that comes as no surprise (= |
| 15:05.38 | tzafrir | anonymouz666, asterisk built with a nnon-matching zaptel.h vs. the one that built the running zaptel kernel modules? |
| 15:09.39 | Katty | anonymouz666: i have a suggestion! |
| 15:09.42 | anonymouz666 | tzafrir: nops. |
| 15:09.53 | Katty | anonymouz666: i suggest goign to the park! |
| 15:09.57 | Katty | anonymouz666: and feeding DUCKS |
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| 15:11.24 | tzafrir | Katty, this channel is not about Gentoo |
| 15:11.35 | tzafrir | We don't support Gentoo |
| 15:12.43 | tzafrir | anonymouz666, do you see any errors in the kernel logs? |
| 15:12.47 | tzafrir | What distro is it? |
| 15:12.54 | anonymouz666 | centos 5 |
| 15:13.10 | tzafrir | Then the kernel logs go to /var/log/messages |
| 15:14.02 | anonymouz666 | no error in the kernel logs. |
| 15:14.07 | anonymouz666 | dmesg output ok. |
| 15:14.17 | anonymouz666 | the /var/log/messages OK |
| 15:14.24 | anonymouz666 | the card is working. |
| 15:16.34 | anonymouz666 | tzafrir: res = ioctl(fd, ZT_GETGAINS, &g); |
| 15:16.39 | anonymouz666 | how we get the ZT_GETGAINS? |
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| 15:20.06 | tzafrir | anonymouz666, this is a #define from zaptel.h |
| 15:22.15 | anonymouz666 | I see |
| 15:22.37 | Katty | tzafrir: what's gentoo have to do with ducks? |
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| 15:24.28 | *** mode/#asterisk [+o russellb] by ChanServ |
| 15:24.34 | tzafrir | hmmm.... I thought it was a duck. Aparatly it is a penguin. |
| 15:24.37 | tzafrir | oh well |
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| 15:25.51 | anonymouz666 | tzafrir: does make sense rmmod and then modprobe again? |
| 15:26.08 | pigpen | tzafrir, well, you may not support Gentoo. |
| 15:26.43 | Katty | just because it's Not Supported doesn't mean it Won't Work |
| 15:27.15 | tzafrir | anonymouz666, maybe. not sure |
| 15:27.32 | tzafrir | pigpen, I sure don't, for the lack of knowledge |
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| 15:28.24 | pigpen | yeah, I leave the core crap to my Business Partner...a Gentoo Dev. |
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| 15:28.51 | pigpen | I tried to keep up, but not doing it everyday became a pain. |
| 15:28.52 | *** join/#asterisk Maliuta_ (n=nikolai@c210-49-69-241.rochd2.qld.optusnet.com.au) |
| 15:29.41 | pigpen | I know he is planning on setting up a special portage tree for us to use that will limit the crap that is not needed on a server platform. |
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| 15:33.58 | BeeBuu | why i had no res_mysql.conf file? |
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| 15:41.04 | FlatFoot | ~consultant |
| 15:41.05 | jbot | Hire a consultant. |
| 15:41.05 | FlatFoot | ~ consultant |
| 15:41.06 | jbot | Hire a consultant. |
| 15:41.06 | FlatFoot | oops wrong room |
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| 15:56.25 | naitram | anyone used AMI with events to log call activity to database? |
| 16:00.19 | naitram | anyone here used AMI much? need advise |
| 16:02.29 | naitram | tap, tap, tap. Is this thing on? someone give me a mic check please. Test 1,2,3... |
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| 16:08.44 | alrs | !killall links |
| 16:09.48 | Maliuta_ | links: no process killed |
| 16:14.32 | irule | kill `pidof links` |
| 16:17.52 | *** join/#asterisk IPetrov (i=IPetrov@ppp85-140-234-50.pppoe.mtu-net.ru) |
| 16:18.41 | Maliuta_ | kill: usage: kill [-s sigspec | -n signum | -sigspec] pid | jobspec ... or kill -l [sigspec] |
| 16:19.03 | Maliuta_ | (in other terms pidof returned nothing) |
| 16:19.31 | naitram | anyone here used AMI to do call logging |
| 16:23.25 | *** join/#asterisk ct2clay (n=ct2clay@65-60-106-98.static-ip.telepacific.net) |
| 16:24.05 | *** join/#asterisk c0rnflake (n=c0rnflak@38.112.4.210) |
| 16:26.44 | *** join/#asterisk saftsack (n=saftsack@pD9E0551F.dip.t-dialin.net) |
| 16:27.26 | anonymouz666 | tzafrir: I just give up with zaptel 1.2.19. I will upgrade to 1.2.21 and try. |
| 16:27.28 | *** join/#asterisk ct2clay (n=ct2clay@65-60-106-98.static-ip.telepacific.net) |
| 16:27.37 | IPetrov | Hi. Anyone use russels patch for bug #8413 - with atxfer? |
| 16:27.40 | ct2clay | is anyone familiar with vodavi phones? |
| 16:30.36 | *** join/#asterisk FuriousGeorge (n=email@ool-4354d123.dyn.optonline.net) |
| 16:30.40 | FuriousGeorge | hey all |
| 16:32.36 | FuriousGeorge | I never was able to get any of the servers i install asterisk on to work right. i used to think there was something wrong with me, but ive learned not to let it affect my self-esteem :) http://www.pastebin.ca/757782 |
| 16:32.39 | FuriousGeorge | seriously though |
| 16:32.56 | FuriousGeorge | every now and then zap channels get locked up, as shown in that pastebin |
| 16:33.23 | FuriousGeorge | no one is on the phone, but of course, we can not receive incomming calls because the system is tieing up all the land lines |
| 16:34.19 | tzafrir | does a soft hangup fix things? |
| 16:34.25 | FuriousGeorge | tzafrir: yup |
| 16:34.25 | waKKu | folks... have some way to force an unregistering from a SIP exten ? ?? |
| 16:35.59 | *** join/#asterisk flujan (n=flujan@200.160.115.20) |
| 16:36.02 | FuriousGeorge | i recently upgraded asterisk, and the phones firmware |
| 16:36.20 | FuriousGeorge | but im starting to get convinced im doomed to be stuck with these issues forever |
| 16:36.48 | FuriousGeorge | its not crashing/deadlocking anymore, but all the asterisk servers now have issues with detecting channel hangups |
| 16:37.24 | FuriousGeorge | the other ones tend to have it happen with internal lines only, which is ok because then the blf just gets stuck, no big deal |
| 16:37.36 | *** join/#asterisk tsurko (n=tsurko@150-190.go.evo.bg) |
| 16:37.40 | FuriousGeorge | but when your phone system prevents you from taking calls once every two weeks, that is a problem |
| 16:37.50 | tsurko | hello |
| 16:37.57 | FuriousGeorge | thank god we are in the age of cell phones so we have people to call us and tell us when are lines are down |
| 16:38.04 | FuriousGeorge | embarassing, but better than nothing |
| 16:39.09 | FuriousGeorge | i got snom360 phones, sangoma a200 cards, and asterisk 1.4.12 |
| 16:39.48 | tsurko | i want to connect an asterisk box with analog PBX, so the outgoing calls to be routed through asterisk. If I use FXO/FXS interface to connect them how many paralell outgoing calls will be possible - just one? |
| 16:40.18 | *** join/#asterisk JackEStorm (n=no@ip68-225-77-136.no.no.cox.net) |
| 16:41.52 | *** join/#asterisk lbt (n=david@i-83-67-36-194.freedom2surf.net) |
| 16:42.02 | *** join/#asterisk macros73_ (n=cs@c-24-3-246-27.hsd1.mn.comcast.net) |
| 16:43.52 | *** join/#asterisk SM0TVI (i=teodor@c83-253-122-78.bredband.comhem.se) |
| 16:44.11 | FuriousGeorge | im on the phone with sangoma, and they say they have 'no control' over when how or if asterisk detects hangup |
| 16:44.56 | lbt | hi - no-one in #asterisk-gui, can someone here help? |
| 16:45.05 | lbt | I've installed the gui but keep gettin "move along..." |
| 16:45.13 | lbt | can't find any logs, errors etc |
| 16:45.20 | [TK]D-Fender | tsurko: Well if you want to pass calls THROUGH *, then 1 FXS & 1 FXS per call. |
| 16:45.30 | [TK]D-Fender | lbt : .... |
| 16:45.37 | [TK]D-Fender | lbt : .... move along :) |
| 16:45.48 | outtolunc | sounds like a failed access attempt 'warning message' <G> |
| 16:46.09 | lbt | [TK]D-Fender: Ho ho ;) |
| 16:46.16 | tsurko | [TK]D-Fender, 1fxs on the PBX and one on * ? |
| 16:46.21 | lbt | I can't get over how it knows my name is Dave... |
| 16:46.46 | [TK]D-Fender | tsurko: well for every port on your PBX you want to use in-line you need 1 each on * |
| 16:47.54 | lbt | "http show status" works and the other urls ".../asterisk/httpstatus" work; just not those in static-http... |
| 16:48.07 | *** join/#asterisk sasch (n=info@host117-234-static.4-79-b.business.telecomitalia.it) |
| 16:48.12 | sasch | hi all |
| 16:48.15 | *** join/#asterisk Spida (n=timo@spinnennetz.org) |
| 16:48.21 | Spida | hi |
| 16:48.30 | lbt | hello |
| 16:48.55 | *** join/#asterisk FreezeS (n=bla@82.208.157.125) |
| 16:49.02 | FreezeS | hey guys |
| 16:49.27 | lbt | can any of you new arrivals help with a -gui problem? |
| 16:49.36 | lbt | "http show status" works and the other urls ".../asterisk/httpstatus" work; just not those in static-http... |
| 16:49.38 | FreezeS | I made a php agi script that needs to hang up a call and then do some db queries |
| 16:49.45 | Spida | is there a minimal config example for a zaptel with a isdn-line and a sip-account to use that? |
| 16:49.53 | FreezeS | how can I ignore SIGHUP ? |
| 16:50.42 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 16:50.45 | [TK]D-Fender | lbt: The answer isn't changing. This is NOT a GUI support channel |
| 16:51.24 | [TK]D-Fender | FreezeS: use DeadAgi |
| 16:52.10 | *** join/#asterisk GreggB (n=GreggB@66.206.86.107) |
| 16:52.40 | *** part/#asterisk chodorenko (n=chodoren@etm005.nl.ded.neolocation.net) |
| 16:53.26 | FreezeS | [TK]D-Fender: I am using deadagi |
| 16:53.36 | [TK]D-Fender | FreezeS: :/ |
| 16:53.48 | FreezeS | but, even if I don't explicitly run Answer, it still answers my call |
| 16:53.56 | FreezeS | and I want to run first Hangup |
| 16:54.32 | lbt | [TK]D-Fender: Thanks for your help mate. Nice. |
| 16:55.08 | FreezeS | I was thinking about something else |
| 16:55.30 | FreezeS | my call comes in to exten => _000999.,1,DeadAgi(quickdial.php) |
| 16:55.50 | FreezeS | can I run the script after I'm hanging up ? |
| 16:56.09 | TJNII | FreezeS: fork a child? |
| 16:56.10 | *** join/#asterisk lvl`home (n=lvl@101.248.dhcp.lht.hhs.nl) |
| 16:56.12 | [TK]D-Fender | FreezeS: Treid catching it in "h" yet? |
| 16:56.23 | FreezeS | my current dialplan is the following: |
| 16:56.29 | FreezeS | exten => _000999.,1,HangUp |
| 16:56.29 | FreezeS | exten => _000999.,h,DeadAgi(quickdial.php) |
| 16:56.35 | [TK]D-Fender | FreezeS: Actually... not a bad idea... just fork a LOCAL channel for it |
| 16:56.37 | FreezeS | doesn't work |
| 16:56.44 | FreezeS | how should I do that ? |
| 16:56.51 | lbt | how do I detect asterisk logs for a module? |
| 16:56.54 | [TK]D-Fender | FreezeS: "h" is NOT a priority, it is an Asterisk Standard EXTENSION |
| 16:57.06 | FreezeS | yes, I was just reading about it |
| 16:57.13 | FreezeS | and not reading it completely it seems :) |
| 16:57.15 | *** join/#asterisk EtienneG (n=etienne@ubuntu/member/EtienneG) |
| 16:57.23 | EtienneG | hello everybody |
| 16:57.27 | FreezeS | it looked too nice to be true.. |
| 16:57.31 | lbt | I've set core debug to a high value but http (not the -gui [TK]D-Fender) isn't logging to messages |
| 16:57.40 | EtienneG | I would need a little help using a IAXy |
| 16:58.21 | EtienneG | For example, how would we put a call on hold using a standard analog handset connected to a IAXy ? |
| 16:58.24 | *** join/#asterisk Ritzerisk (n=Ritztech@24.120.190.66) |
| 16:58.49 | EtienneG | I tried flash, but then, the phone keep ringing when I hung up |
| 16:58.58 | [TK]D-Fender | EtienneG: I seriously doubt it has that functionality |
| 16:59.11 | FreezeS | TJNII: so how would I fork a child ? |
| 16:59.17 | EtienneG | [TK]D-Fender, it is advertised in the brochure ... ! |
| 16:59.48 | [TK]D-Fender | EtienneG: got a link to the manual? |
| 17:00.05 | TJNII | FreezeS: Depends on the language, most have some version of fork() |
| 17:00.21 | TJNII | FreezeS: You'll have to read about it. I haven't dealt with fork() in a few years. |
| 17:00.43 | EtienneG | [TK]D-Fender, brochure: http://www.digium.com/elqNow/elqRedir.htm?ref=http%3A%2F%2Fdigium.com%2Fen%2Fdocs/S101I/IAXy.pdf, manual: http://www.digium.com/elqNow/elqRedir.htm?ref=http%3A%2F%2Fdigium.com%2Fen%2Fdocs/S101I/Iaxy_Installation_Guide.pdf |
| 17:01.07 | coppice | I seldom deal with fork. I generally use chopsticks |
| 17:01.08 | EtienneG | basically, the manual is just about provisionning |
| 17:01.21 | *** join/#asterisk penguinFunk (n=penguin@unaffiliated/penguinfunk) |
| 17:02.01 | Spida | is there a minimal config example for a zaptel with a isdn-line and a sip-account to use that? |
| 17:02.25 | *** join/#asterisk Alowishus (n=jpenix@fwsdo.projectdesign.com) |
| 17:02.28 | penguinFunk | what kind of isdn line? |
| 17:02.34 | penguinFunk | BRI or PRI ? |
| 17:02.43 | penguinFunk | isdn30 or isdn2 ? |
| 17:02.47 | *** join/#asterisk Xen^ (n=linux@unaffiliated/lnux/x-10290) |
| 17:02.52 | Spida | euroisdn, external |
| 17:02.55 | [TK]D-Fender | EtienneG: maybe the constant ringing is part of the "hold".... |
| 17:03.03 | *** join/#asterisk davixx (n=davixx@85.69.218.190) |
| 17:03.04 | Spida | two b-channels |
| 17:03.05 | penguinFunk | how many channels? |
| 17:03.06 | [TK]D-Fender | EtienneG: How long have you had it? |
| 17:03.11 | EtienneG | [TK]D-Fender, hopefully not! |
| 17:03.12 | penguinFunk | that will be BRI then |
| 17:03.31 | EtienneG | [TK]D-Fender, I just mad eit working this morning, had it gathering dust for a few weeks |
| 17:03.35 | *** join/#asterisk msetim (n=marcos@200.195.161.164) |
| 17:03.45 | msetim | Hi guys |
| 17:03.56 | penguinFunk | Spida: http://www.voip-info.org/wiki/index.php?page=Asterisk+config+zapata.conf |
| 17:03.57 | msetim | where can I read about asterisk deadlocks, I would like to undersdant it :) |
| 17:04.00 | penguinFunk | scroll down to BRI |
| 17:04.01 | [TK]D-Fender | EtienneG: Oh, stuck with it then.... oh well... co e-mail Digium support and they should ahve the answer for you... |
| 17:04.34 | EtienneG | [TK]D-Fender, yeah, that is my plan B, but IRC is generally faster than vendor support |
| 17:04.54 | penguinFunk | Spida: and this http://www.voipinfo.org/wiki/index.php?page=Asterisk+config+zaptel.conf |
| 17:04.54 | [TK]D-Fender | EtienneG: Variably yes, but IAXY use is kinda rare... |
| 17:04.58 | *** join/#asterisk weaver_88 (n=cell_491@pool-71-101-7-200.tampfl.dsl-w.verizon.net) |
| 17:05.22 | BBHoss | EtienneG, what are you trying to do with the iaxy? |
| 17:05.36 | FreezeS | forking in php seems quite problematic (hackish) |
| 17:05.39 | BBHoss | ive got a bunch of them |
| 17:05.45 | FreezeS | isn't there a better way to do it ? |
| 17:05.45 | EtienneG | BBHoss, just remote extension ... I like the easy NAT traversal of IAX |
| 17:05.48 | msetim | sorry guys, I will ask it in asterisk-dev |
| 17:05.52 | weaver_88 | Hello, Ive got some questions you you guys dont mind |
| 17:05.56 | BBHoss | yes i agree |
| 17:05.59 | EtienneG | although the IAXy is not the most featureful ATA around |
| 17:06.03 | BBHoss | heh |
| 17:06.12 | BBHoss | it gets A job done |
| 17:06.17 | weaver_88 | if* |
| 17:06.27 | BBHoss | you want to put a call on hold |
| 17:06.28 | EtienneG | BBHoss, any idea on blind transfer and call hold ? |
| 17:06.32 | BBHoss | hmm |
| 17:06.36 | EtienneG | er, yeah, indeed |
| 17:06.40 | BBHoss | usually its # or ## |
| 17:06.48 | BBHoss | to transfer |
| 17:07.00 | EtienneG | lemme try that |
| 17:07.01 | BBHoss | but you may have to set it |
| 17:07.28 | *** join/#asterisk peanut- (n=tokarev@cpe-70-113-100-193.austin.res.rr.com) |
| 17:08.08 | EtienneG | BBHoss, you got blind transfer right dude ... ##ext# |
| 17:08.16 | BBHoss | check features.conf |
| 17:08.36 | EtienneG | actually, no trailing # |
| 17:08.39 | BBHoss | yeah |
| 17:08.40 | weaver_88 | So ive got an office running a definity g3v8 with 60 extensions. I need to set up some sort of call center type control with total call recording. the asterisk software is capable of this right? If so what other kind of hardware would I need? |
| 17:08.42 | BBHoss | it times out |
| 17:08.52 | EtienneG | BBHoss, thanks a bunch for the tip |
| 17:08.56 | BBHoss | sure |
| 17:09.10 | BBHoss | i have a few of these things, they aren't the best but they can help you in a pinch |
| 17:09.32 | BBHoss | i wish digium or someone would make an IAX device with a few more features, maybe more ports |
| 17:09.43 | BBHoss | weaver_88: yes this is very possible |
| 17:09.44 | Maliuta_ | weaver_88: read the book |
| 17:10.18 | weaver_88 | what book? |
| 17:10.21 | Spida | penguinFunk: thx, reading... |
| 17:10.35 | BBHoss | im not sure if you'll be able to use the definity with it, as im not experienced with avaya |
| 17:10.37 | *** join/#asterisk ocgltd (n=support@CPE004063e0ee74-CM00159a010632.cpe.net.cable.rogers.com) |
| 17:10.39 | Maliuta_ | !book |
| 17:10.42 | BBHoss | ~ book |
| 17:10.43 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf |
| 17:10.58 | Maliuta_ | BBHoss: so I got the character wrong :P |
| 17:11.05 | BBHoss | heh |
| 17:11.35 | *** join/#asterisk dlynes (n=dlynes@d154-20-9-152.bchsia.telus.net) |
| 17:12.22 | ocgltd | Can someone offer help connecting an asterisk box to a Nortel Option 61 via a T1 with PRI. We have hit a problem we cannot overcome; specifically, the Nortel is asking for the ROSE information element (IE) over the PRI connection. This causes libpri to drop the connection, with cause 100. The Nortel cannot "turn off" asking for ROSE. Does anyone know how to overcome this? |
| 17:12.22 | BBHoss | good luck buddy |
| 17:12.27 | ocgltd | and...the firmware on the Nortel is too old to support QSIG. |
| 17:13.13 | BBHoss | can you not possibly upgrade the firm? |
| 17:13.41 | ocgltd | That's an $80k solution...(domino effect of upgrading cpu, then line cards, etc) |
| 17:14.56 | coppice | $80k is roughly the price of a new manual for a meridian :-) |
| 17:15.11 | *** join/#asterisk DrAk0 (n=ljd@nelug/coreteam/luisjose) |
| 17:15.12 | BBHoss | ooh |
| 17:15.13 | [TK]D-Fender | ocgltd: It'd cost about $10k to completely replace your setup with a Polycom & * solution |
| 17:15.18 | BBHoss | yeah |
| 17:15.29 | BBHoss | how many lines/extensions you using now |
| 17:15.39 | [TK]D-Fender | ocgltd: in hardware. add a bit for probable rewiring. |
| 17:16.41 | BBHoss | hell, a rack of * servers with quad t1 cards in each could pull off just about anything, especially with channel banks etc |
| 17:17.21 | BBHoss | if you wanted to use the old-skool phones, which probably wouldnt work with nortel phones |
| 17:18.09 | *** join/#asterisk kannan (n=kannan@61.11.72.127) |
| 17:18.20 | *** join/#asterisk bkruse (i=bkruse@nat/digium/x-4eba4c0444defb72) |
| 17:20.22 | BBHoss | did everyone die all the sudden? |
| 17:20.40 | ocgltd | Yeah I know...but it's for a emergency medical center - no way I want to take on that liability! |
| 17:21.48 | Alan_Hicks | ocgltd: Sub it out. |
| 17:21.50 | BBHoss | you could always hire someone if you don't fell comfortable |
| 17:21.55 | BBHoss | damn, beat me |
| 17:22.13 | Spida | according to http://www.voipinfo.org/wiki/index.php?page=Asterisk+config+zaptel.conf ztcfg -vvvv should exit silently... is http://rafb.net/p/70TmnV75.html ok, too? |
| 17:22.51 | Alan_Hicks | There's plenty of people here who can do the job, and offer consulting services, or you could refer the client to another group for a small take of the profits. |
| 17:23.04 | Alan_Hicks | NOTE: I'm not one of those people. :^) |
| 17:24.00 | destructure | http://www.lloyds.com/ |
| 17:25.29 | *** join/#asterisk techie (n=techie@adsl-76-214-18-225.dsl.lsan03.sbcglobal.net) |
| 17:25.36 | ocgltd | we're still at the pilot stage - they want to see * can work (handing calls from the Nortel). The way the pilot is going there won't be a job :( |
| 17:26.08 | BBHoss | hmm |
| 17:28.49 | ocgltd | Anyone here a libpri expert? Perhaps a Nortel expert? |
| 17:29.01 | Qwell | ~ask |
| 17:29.01 | jbot | Questions in the channel should be specific, informative, complete, concise, and on-topic. Don't ask if you can ask a question first. Don't ask if a person is there, just ask what you intended to ask them. Better questions more frequently yield better answers. We are all here voluntarily or against our will. |
| 17:29.14 | ocgltd | I can't believe I'm the first person to experience this PRI limitation! There are lots of nortel switches out there |
| 17:29.57 | destructure | send me one and I'll try it out |
| 17:31.37 | [TK]D-Fender | ocgltd: This is something you'll have to do some SERIOUS consulting to get done if even... |
| 17:32.19 | Spida | penguinFunk: is http://rafb.net/p/70TmnV75.html ok? |
| 17:33.17 | *** join/#asterisk Op3r (n=edwin@125.212.120.184) |
| 17:36.28 | *** join/#asterisk ghento (n=ghento@64.180.85.230) |
| 17:37.30 | *** join/#asterisk ManxPower (n=manxpowe@120.sub-75-201-68.myvzw.com) |
| 17:38.42 | ocgltd | We see it is a big opportunity...with budget to match. This would include $ to cover legal costs should an outage cost lives etc....so we are taking it seriously. We want to demonstrate stability & functionality first, earn their trust, and then build up slowly. Over time, replacing the nortel |
| 17:38.53 | BBHoss | hmm |
| 17:39.24 | BBHoss | i see your predicament |
| 17:40.04 | BBHoss | what kind of setup do you have? |
| 17:40.30 | BBHoss | like outgoing/incoming lines, how many extensions you have, mobile phones, etc |
| 17:41.01 | ocgltd | For the pilot, 10 wireless, 10 wired. |
| 17:41.07 | BBHoss | ahh |
| 17:41.14 | BBHoss | wireless :) |
| 17:41.56 | BBHoss | how do you have your outgoing trunks setup right now |
| 17:41.58 | ocgltd | We went spectralink/polycom because of congestion management using SVP protocol. We've seen many failed wireless implementation (which we got to clean up) |
| 17:42.01 | *** join/#asterisk drwelby (n=mpfister@mail.enplan.com) |
| 17:42.10 | *** join/#asterisk k-man_ (n=jason@unaffiliated/k-man) |
| 17:42.24 | BBHoss | spectralinks a bitch to setup |
| 17:42.28 | ocgltd | Initial trunk was to Nortel for moving extensions to *. That's were we got stuck |
| 17:42.50 | BBHoss | hmm |
| 17:42.55 | ocgltd | Yes - spectralinks can be tough be we have done enough that we have saved profiles... |
| 17:43.07 | BBHoss | so you have already purchased? |
| 17:43.40 | EtienneG | ok, back to my IAXy wos, I can fake call hold with call parking. However ... |
| 17:43.54 | EtienneG | it does not for outbound calls! |
| 17:44.17 | BBHoss | so you can't call out and put them on hold correct? |
| 17:44.18 | EtienneG | ie, if I dial my IAXy, I can park the call from the handset connected to it |
| 17:44.28 | EtienneG | BBHoss, correct |
| 17:44.31 | BBHoss | ok |
| 17:44.41 | BBHoss | you will probably need to change your dial options |
| 17:44.43 | Qwell | add a 't' (or was it 'T'?) to your outgoing Dial() |
| 17:44.49 | BBHoss | yes |
| 17:44.52 | BBHoss | hmm |
| 17:44.56 | *** join/#asterisk oej (n=olle@64.251.114.2) |
| 17:45.01 | EtienneG | using FreePBX ... any idea where to set that ? |
| 17:45.07 | Qwell | #freepbx |
| 17:45.10 | EtienneG | I'll check |
| 17:45.10 | BBHoss | General Settings i think |
| 17:45.25 | *** join/#asterisk wwalker (n=wwalker@pdpc/supporter/sustaining/wwalker) |
| 17:46.31 | wwalker | I'm working on an existing dial plan written by someone long gone. it has "_**1707" I've looked at the Asterisk+Dialplan+Patterns page on the wiki and there is no mention of * as a pattern character. Anyone know what gives? |
| 17:46.48 | BBHoss | * |
| 17:47.09 | BBHoss | i think |
| 17:47.34 | [TK]D-Fender | wwalker: * = * |
| 17:47.34 | blitzrage | wwalker: it just means it matches the * literally |
| 17:47.48 | blitzrage | in fact -- nothing about what you wrote there is actually a pattern at all :) |
| 17:47.53 | blitzrage | the _ is superflouous |
| 17:47.55 | [TK]D-Fender | blitzrage: SHHH!!! |
| 17:49.23 | EtienneG | no luck with dial option t |
| 17:49.25 | EtienneG | or T |
| 17:49.25 | wwalker | blitzrage: so, it can never match? |
| 17:49.39 | blitzrage | it just means it'll only match **1707 |
| 17:49.46 | wwalker | (which wouldn't surprise me after looking at the rest of the dialplan |
| 17:49.47 | blitzrage | hence the _ is superfluous |
| 17:49.58 | blitzrage | there is nothing in there that is a "pattern" |
| 17:50.07 | blitzrage | it would work the same without the _ |
| 17:50.08 | BBHoss | did you find where it went? |
| 17:51.03 | wwalker | agreed, I don't see any way I'd have EXTEN = **1707. Is there something in asterisk that would prepend ** to an exten for some reason? |
| 17:51.14 | blitzrage | nope |
| 17:51.20 | blitzrage | not unless you told it to, or someone dialed it |
| 17:51.57 | Spida | genzaptelconf detects the signalling on my bri euroisdn line as bri_cpe, but asterisk doesn't recognize this... |
| 17:52.21 | Spida | if I try to set it to fsxs_ks, I get "Signalling requested on channel 1 is FXS Kewlstart but line is in PRI Signalling signalling" |
| 17:52.45 | Spida | so, how should I configure signalling? |
| 17:53.09 | tzanger | Spida: you have to make sure asterisk is stopped and re-run ztcfg |
| 17:53.16 | tzanger | and then make sure zapata.conf is correct |
| 17:53.21 | tzanger | zaptel.conf != zapata.conf |
| 17:55.26 | BBHoss | bbiab |
| 17:55.31 | wwalker | blitzrage: thanks for the confirmation that this dialplan is total crap |
| 17:55.38 | blitzrage | lol |
| 17:55.58 | *** join/#asterisk Buhntz (i=Boones@port-212-202-42-6.dynamic.qsc.de) |
| 17:56.06 | JackEStorm | ugh, other than voicemail.conf, is there any other place in the configs that the timezone is 'set'/'configured'?? the Date: in the voimail emails are off by 5hours (sooner), and the playback is off by 5hours too in voicemail, but the .txt file is correct |
| 17:56.14 | EtienneG | Qwell, that was it, big thanks |
| 17:57.22 | EtienneG | now, there's an r in outbound Dial options, and it make FreePBX complain about possible breakedness ... is there any good reason to keep that (I was not the one who did the original setup) |
| 17:58.38 | [TK]D-Fender | EtienneG: No |
| 17:59.00 | blitzrage | 'r' is unnecessary |
| 17:59.19 | EtienneG | why would one use that ? |
| 17:59.38 | blitzrage | to generate ringing on a line that doesn't pass back ringing information -- i.e. pretty much never |
| 17:59.57 | EtienneG | blitzrage, I see |
| 18:00.13 | EtienneG | in fact, it is detrimental |
| 18:00.28 | blitzrage | yes -- because it causes ringing to be generated when a line might not be ringing |
| 18:00.29 | EtienneG | I guess you would want to know when the line does not ring, right |
| 18:00.35 | blitzrage | yes |
| 18:00.40 | EtienneG | hummmm |
| 18:00.47 | EtienneG | that may explain some funkiness |
| 18:01.33 | EtienneG | will check with the guy who set up the PBX in the first place |
| 18:03.05 | EtienneG | anyway, here is another one |
| 18:03.26 | EtienneG | call received from IAX trunk tend to get a second or two clipped at the beginning |
| 18:03.33 | EtienneG | is there a way to avoid that ? |
| 18:04.48 | EtienneG | longer buffer, maybe ... |
| 18:10.00 | [TK]D-Fender | wwalker: there is nothing wrong with a dialplan exten starting with **... that'd be what I'd do to prefix an internal function.... |
| 18:14.06 | *** part/#asterisk naitram (n=chatzill@216.77.58.40) |
| 18:17.53 | wwalker | [TK]D-Fender: so what would the code look like that changed EXTEN to be prefixed with **? I don't see anything like that in this extensions.conf |
| 18:17.55 | *** join/#asterisk spatulamaan (n=ggilmore@207.188.8.252) |
| 18:18.06 | variable_office | does anyone use the monitor app? |
| 18:18.24 | variable_office | specifically with the m flag? |
| 18:18.40 | [TK]D-Fender | wwalker: what you showed us WAS a smple of an exten that did that. I'm not sure what you're getting at with your question... |
| 18:18.45 | Corydon76-vcch | wwalker: Goto(**${EXTEN},1) |
| 18:18.54 | variable_office | cause the two channels are not being mixed together in sync |
| 18:19.48 | wwalker | Corydon76-vcch: thx, I'll double check for lines like that |
| 18:21.28 | wwalker | no such code (nothing with ** in it anywhere except tha lines trying to match it, so it must be cruft from some past iteration |
| 18:21.31 | [TK]D-Fender | wwalker: Why don't you just pastbin it all up? |
| 18:22.42 | wwalker | thousands of lines in extensions.conf and thousands more rows in realtime :( |
| 18:23.08 | mocker | thousands of lines??? |
| 18:23.10 | mocker | wow. |
| 18:23.30 | wwalker | yes, nightmarish. and only a couple hundred extensions total.... |
| 18:24.30 | mocker | wwalker: You do know about pattern matching, right? :) |
| 18:24.57 | wwalker | where would I find docs on the word "check" as used in dial plans? the word is so common that I can't find where it's used |
| 18:25.46 | wwalker | mocker: yes. I didn't write this. I was asked to fix it. I just have to figure out what it is currently doing so my 10 times smaller plan will maintain the same functionality. |
| 18:25.59 | mocker | wwalker: I hear ya. |
| 18:26.04 | mocker | Just messin' around. |
| 18:26.27 | mocker | I remember my first dialplain I had no idea that I could match extensions instead of typing them all out. |
| 18:26.36 | mocker | It grew very quickly. |
| 18:26.37 | *** join/#asterisk generalhan (n=asd@ip67-90-64-2.z64-90-67.customer.algx.net) |
| 18:29.12 | wwalker | mine too. |
| 18:31.47 | *** join/#asterisk ghtdak (n=ghtdak@75-165-47-59.tukw.qwest.net) |
| 18:35.38 | generalhan | anyone know if there is a more specific way to get Agent information from the CLI, other than show Agents? 'show agent <Agent>' is not valid |
| 18:36.32 | generalhan | i have a little app i just wrote and i would like it to come back and inform me if i am currently logged on or not, but its not going to be fun to run the 'show agents' command and then parse out the line with my Agent number in it, then parse that out again to tell me if im logged in or now |
| 18:36.33 | generalhan | not |
| 18:36.55 | *** join/#asterisk msetim (n=marcos@200.195.161.164) |
| 18:37.29 | *** join/#asterisk Schumie (n=Steve@cpc1-oxfd2-0-0-cust25.oxfd.cable.ntl.com) |
| 18:37.54 | Trevor_b | Is there a good way to trackdown why an IVR is doing a Hangup when verbosity 99 doesnt give any helpfull information? |
| 18:39.42 | mocker | Trevor_b: When is it doing the hangup? |
| 18:40.09 | blitzrage | if you have Background(), it'll hangup if there are no more priorities to execute after the file is done playing |
| 18:40.13 | Trevor_b | I Background a message on my 3rd menu deep and then instead of waiting for input (for the third damn time) it decides to hangup on me. |
| 18:40.14 | blitzrage | you need to use WaitExten() |
| 18:40.25 | mocker | yup, exactly. |
| 18:40.35 | blitzrage | that is normal behaviour -- it's called autofallthrough |
| 18:40.47 | Trevor_b | hmm, interesting, the other 2 menus have no further priorities and dont do it. ill swap all those out with WaitExten's. |
| 18:40.56 | Trevor_b | thanks |
| 18:41.03 | mocker | Trevor_b: You can still use background. |
| 18:41.04 | blitzrage | no no no |
| 18:41.08 | mocker | You just need something at the end. |
| 18:41.13 | blitzrage | you just add WaitExten() after the Background() |
| 18:41.18 | blitzrage | you still need Background() to play the file |
| 18:41.41 | *** join/#asterisk PC_Clone (n=pc_clone@209-252-117-210.ip.mcleodusa.net) |
| 18:41.42 | Trevor_b | yeah, I meant I have multiple menues that DO NOT use WaitExten after Background, have to change them all, lest i want to see this crop up in other parts ;) |
| 18:42.56 | mocker | Ahh, good idea. ;)_ |
| 18:43.08 | mocker | Wow, typo emoticon. |
| 18:43.18 | Trevor_b | !&*($! |
| 18:43.30 | Trevor_b | STill hangsup on me even while im in the WaitExten window... |
| 18:44.09 | mocker | Trevor_b: Can you pastebin? |
| 18:44.15 | Trevor_b | sure |
| 18:44.33 | mocker | Make sure you don't have any messed up priority #s. |
| 18:45.43 | Trevor_b | http://pastebin.com/d46c97987 |
| 18:46.33 | Trevor_b | I put the entire dialplan in from the application-system (first menu) to the end. The problem happens in application-site-validation, after Background(application-ivr3) it hangs up, even with WaitExten. The Pattern match of _X. seems to work fine 2 times before it. |
| 18:46.37 | *** join/#asterisk duki (n=duki@host-85-27-58-159.brutele.be) |
| 18:46.57 | Spida | does http://rafb.net/p/LoyIW692.html mean that asterisk thinks it had two cards with 2 isdn bri lines with 2 channels each? |
| 18:47.03 | PC_Clone | hi all, I'm trying to get a pri up and running with asterisk, but I cannot dial out using the zap group. When I use a specific channel it all works as planned |
| 18:47.53 | *** join/#asterisk shambala (n=chatzill@189.141.27.130) |
| 18:47.55 | Trevor_b | PC_Clone: your channels have the group defined in /etc/asterisk/zapata.conf correctly? |
| 18:48.13 | shambala | hola alguien de mexico que me pueda ayudar con mi asterisk |
| 18:48.24 | *** join/#asterisk adeel (n=adeel@c-24-7-132-155.hsd1.ca.comcast.net) |
| 18:49.01 | PC_Clone | Trevor_b: I believe so |
| 18:49.05 | mocker | Trevor_b: Can you throw some NoOps in to make sure you are routing to site-validation? |
| 18:49.09 | PC_Clone | one sec and I'll put it to paste bin |
| 18:49.11 | shambala | please helpme with my asterisk |
| 18:49.21 | mocker | Like 'NoOp(About to WaitExten) |
| 18:49.25 | *** join/#asterisk sigmounte (n=sigmount@bai59-1-88-172-80-96.fbx.proxad.net) |
| 18:49.33 | mocker | NoOp(Just did muh WaitExten |
| 18:49.34 | mocker | ) |
| 18:49.35 | mocker | etc.. |
| 18:49.54 | Trevor_b | so toss 'loggin' info before and after the problem priority? |
| 18:50.04 | Trevor_b | s/loggin/log/ |
| 18:50.15 | sigmounte | hello , i'm trying to use moh , but look like my extention.conf is not good , cause when calling i have the moh but the phone doesn't ring , anyone have an example of conf ? |
| 18:50.41 | mocker | Trevor_b: Yeah, just see what it's doing. |
| 18:50.46 | *** join/#asterisk _ys (n=yuri@80.70.236.69) |
| 18:50.48 | PC_Clone | Trevor_b: http://pastebin.com/ma5ab427 |
| 18:50.50 | mocker | Make sure you're hitting it, etc.. |
| 18:52.33 | Trevor_b | PC_Clone: http://pastebin.com/d23bcf966 |
| 18:52.39 | Trevor_b | make that change, reload and test |
| 18:53.08 | shambala | hello |
| 18:53.22 | shambala | some one from mexico |
| 18:53.38 | shambala | i need help with my asterisk |
| 18:53.49 | mocker | ~question |
| 18:53.50 | jbot | somebody said question was If you have a question and want people to give useful answers, make sure you have read this first: http://www.catb.org/~esr/faqs/smart-questions.html |
| 18:54.14 | PC_Clone | Trevor_b: thanks...didn't know the placement of the group statement mattered |
| 18:54.19 | Trevor_b | mocker: I NoOP before Background, after background, before WaitExten, and after. I see them all extept the one after WaitExten, goes straight from WaitExten to Hangup, skipping the NoOp.... |
| 18:54.30 | Trevor_b | PC_Clone: I think thats it, let me know ;) |
| 18:55.13 | Trevor_b | PC_Clone: All mine have group above the channel definition, since you can have lots of channel definitions. |
| 18:55.33 | shambala | ok i install my asterisk and works ok all my extensions are ready but if i want to conect a remote user out of my office this user can login to my asteris |
| 18:55.51 | adeel | while using * 1.4, and sip realtime, it seems to default my DTMF settings to inband...i've tried putting 'dtmf=auto' in my sip.conf file, but that doesn't seem to be doing anything...how else can i override it? |
| 18:56.01 | *** join/#asterisk jedaustin (n=chatzill@austin-j.its.dist.maricopa.edu) |
| 18:56.17 | jedaustin | Anyone here from Xorcom? |
| 18:56.20 | *** join/#asterisk Maxxed (i=foobar@65.59.245.122) |
| 18:56.24 | Maxxed | sup fellas :) |
| 18:56.29 | tzafrir | jedaustin, yes |
| 18:57.05 | *** join/#asterisk r0d3nt (i=nobody@foster.stonedcoder.org) |
| 18:57.28 | _x86_ | hmm, i set my caller ID on a channel before Dial'ing over IAX2 to another asterisk box, but when the second asterisk box calls the end-station (polycom IP501), it sees the caller ID as the second asterisk box's IP address? |
| 18:57.38 | _x86_ | anyone know what i'm doing wrong? |
| 18:58.30 | _x86_ | even when zap station ZA calls sip station SA within the same asterisk server, SA sees the caller ID of the IP of the asterisk box |
| 18:58.30 | Trevor_b | mocker: http://pastebin.com/d24070887 I put the console output at the bottom as well. |
| 18:59.24 | mocker | And it doesn't wait three seconds? |
| 19:00.15 | Trevor_b | No it waits 3 seconds, but then does hangup instead of the Last NoOp saying it FINISHED WaitExten |
| 19:00.41 | Trevor_b | wonder if it doesnt like the single digit. |
| 19:01.27 | Trevor_b | IM A @&$ IDIOT |
| 19:01.33 | Trevor_b | its a pattern problem |
| 19:02.17 | mocker | Trevor_b: Where at? |
| 19:02.26 | Trevor_b | I had _X. and was trying to do single digit "sites" in the menu |
| 19:03.42 | Trevor_b | needed to make the application-site-validation pattern match be exten => _. not _X. |
| 19:03.50 | *** join/#asterisk NDT (n=vb6xweb@cpe-74-70-217-136.nycap.res.rr.com) |
| 19:04.06 | mocker | Trevor_b: Don't feel bad, I spent 45 minutes last night fighting an IVR problem. |
| 19:04.21 | mocker | Only to find out I had didn't start a priority w/ 1 at one point. |
| 19:04.21 | [TK]D-Fender | Trevor_b: EW |
| 19:04.22 | Trevor_b | sweet! |
| 19:04.32 | [TK]D-Fender | Trevor_b: _. = MISTAKE. |
| 19:04.38 | *** join/#asterisk sasch (n=info@host117-234-static.4-79-b.business.telecomitalia.it) |
| 19:04.46 | Trevor_b | whats a better way to limit a pattern to 1-3 digits? |
| 19:04.49 | [TK]D-Fender | Trevor_b: just ADD a "exten=> _X,1" |
| 19:05.05 | [TK]D-Fender | Trevor_b: do 3 matches. |
| 19:05.06 | Trevor_b | do 3 pattern matches instead? |
| 19:05.08 | Trevor_b | cool |
| 19:05.09 | Trevor_b | thanks |
| 19:05.11 | sasch | i have one problem with asterisk .... i compile asterisk with bristuff.... but now say this http://pastebin.ca/757909 |
| 19:05.13 | sasch | why ?? |
| 19:05.45 | *** join/#asterisk _ys (n=yuri@236-069.nat.mns.ru) |
| 19:07.57 | *** join/#asterisk |omni| (n=rob@69.28.39.99) |
| 19:09.23 | *** part/#asterisk Alowishus (n=jpenix@fwsdo.projectdesign.com) |
| 19:17.45 | *** join/#asterisk saftsack (n=saftsack@pD9E0551F.dip.t-dialin.net) |
| 19:18.09 | *** part/#asterisk NDT (n=vb6xweb@cpe-74-70-217-136.nycap.res.rr.com) |
| 19:18.25 | sigmounte | my moh doesn't play when dialing , but work when using queue , what's wrong ? |
| 19:21.55 | blitzrage | sigmounte: really? Someone complained about that today, but I could not reproduce it |
| 19:22.04 | blitzrage | it plays between calling agents right? |
| 19:22.11 | blitzrage | see bug 11095 |
| 19:22.21 | blitzrage | add some information that might help if possible to reproduce |
| 19:22.26 | blitzrage | I'm outta here for a bit |
| 19:23.38 | `Sauron | What the hell is this elqNow download crap all about |
| 19:24.05 | sigmounte | blitzrage, i can hear the moh using the MusicOnHold cmd , but it doesn't let the next command (dial) to worl |
| 19:24.57 | blitzrage | next command Dial() you mean? |
| 19:25.03 | blitzrage | I don't understand what you're saying... |
| 19:25.11 | *** join/#asterisk keulin (n=cray@ifth-pdcr2.infotheme.fr) |
| 19:25.22 | `Sauron | and I love how ftp2.digium.com gives a connection refused. |
| 19:25.23 | blitzrage | you mean from in a Queue()? |
| 19:26.44 | sigmounte | blitzrage, i'm doing this 1,Answer() 2,MusicOnHold 3,Dial(SIP/myphone,60) 4,Voicemail ...... |
| 19:28.01 | drwelby | Is # officially a terminating character or "dial now" when dialing or transferring an extension? |
| 19:28.07 | drwelby | Never noticed it before |
| 19:28.11 | *** join/#asterisk davixx (n=davixx@85.69.218.190) |
| 19:28.58 | `Sauron | Is it safe to assume that * is 64bit-safe? |
| 19:30.06 | davevg-btwtech | sigmounte, can you try Dial(SIP/myphone,60,m) instead of the musiconhold line |
| 19:30.07 | putnopvut | `Sauron: yes, most of the devs use 64-bit machines when coding/testing |
| 19:31.08 | sigmounte | thanks you! |
| 19:31.31 | sigmounte | yes working !! |
| 19:32.28 | tzafrir | `Sauron, downloads.digium.com ? s/ftp/http/ ? |
| 19:33.16 | brookshire | http only |
| 19:33.30 | *** join/#asterisk k-man__ (n=jason@ppp244-232.static.internode.on.net) |
| 19:33.34 | tzafrir | `Sauron, that script looks like "please don't mirror us" |
| 19:33.40 | `Sauron | tzafrir: http://www.asterisk.org/downloads |
| 19:33.48 | brookshire | `Sauron: http://downlaods.digium.com |
| 19:33.49 | `Sauron | if you try to download anything from there |
| 19:34.09 | `Sauron | brookshire: Tried that. Nothing. |
| 19:34.14 | `Sauron | http://downloads.digium.com/pub/asterisk/ |
| 19:34.16 | brookshire | helps to spell |
| 19:34.18 | brookshire | http://downloads.digium.com/ |
| 19:34.29 | brookshire | works for me |
| 19:34.35 | `Sauron | all of the links have the elqNow/elqRedir.html?blahadasdfasfd script |
| 19:34.45 | `Sauron | which just gives me a blank page |
| 19:34.48 | `Sauron | and nothing else |
| 19:34.52 | tzafrir | because you have javascript support |
| 19:35.06 | `Sauron | Huh? |
| 19:35.19 | `Sauron | It refrains from working if I have JS enabled? |
| 19:35.32 | brookshire | disabled |
| 19:35.34 | tzafrir | s/if/unless/ |
| 19:35.48 | `Sauron | I very much have JS enabled in this browser. |
| 19:36.01 | `Sauron | so I'm calling BS |
| 19:36.11 | tzafrir | yes, and all links go through this wierd redirection script |
| 19:36.36 | `Sauron | And I'm saying it doesn't work (right). |
| 19:36.46 | `Sauron | Doesn't matter, I got the stuff downloaded |
| 19:37.01 | tzafrir | Anybody for a quick greasmonkey module for it? ;-) |
| 19:40.39 | mocker | lftp http://downloads.digium.com/ doesn't work. :( |
| 19:40.55 | *** join/#asterisk agx (n=badpengu@81-174-47-187.dynamic.ngi.it) |
| 19:46.01 | sigmounte | sorry , but multiple call using the cmd (dial) what is the syntaxe ? |
| 19:46.40 | hads | & |
| 19:47.04 | sigmounte | thanks |
| 19:48.25 | Katty | doobydoobydo |
| 19:50.32 | JerJer | toke toke |
| 19:52.45 | *** part/#asterisk techie (n=techie@adsl-76-214-18-225.dsl.lsan03.sbcglobal.net) |
| 19:53.00 | _x86_ | heya Katty |
| 19:53.38 | Katty | herro |
| 19:55.33 | anonymouz666 | JerJer: did you fix the tcp2udp stuff? |
| 20:09.06 | *** part/#asterisk shambala (n=chatzill@189.141.27.130) |
| 20:18.10 | Katty | 2 more hours! i can do this! |
| 20:19.10 | JerJer | anonymouz666: that customer hasn't complained, so I am going to presume so |
| 20:19.17 | anonymouz666 | Katty! |
| 20:19.40 | anonymouz666 | JerJer: it is very easy to do with t_relay() |
| 20:20.08 | JerJer | I haven't bothered to figure out what I need to pass t_relay |
| 20:20.19 | JerJer | so i used the append trick |
| 20:20.33 | anonymouz666 | transport=blah ? |
| 20:21.23 | JerJer | yup |
| 20:23.42 | *** join/#asterisk sacitec (n=tobi@189.149.99.172) |
| 20:24.31 | *** part/#asterisk myiagy (n=myiagy@189.4.71.73) |
| 20:25.42 | *** join/#asterisk [hC] (n=hardcore@66.119.167.162) |
| 20:28.57 | *** join/#asterisk marl_ (n=matt@82-40-218-233.cable.ubr01.dunb.blueyonder.co.uk) |
| 20:29.34 | *** join/#asterisk duki (n=duki@host-85-27-58-159.brutele.be) |
| 20:29.53 | marl_ | anyone know of a simple web frontend to the monitor recordings (and posibly voicemail) that does not require asterisk to be using a database for any kind of storage? |
| 20:30.20 | marl_ | hav been trying to get ARI working, but it keeps complaining about needing mysql databases :( |
| 20:30.21 | mocker | marl_: I think ari may suite your needs. |
| 20:30.26 | mocker | Oh. |
| 20:30.47 | mocker | I forget why it needs a database. |
| 20:31.06 | marl_ | from wat i read then it should work without databases, but i cant seem to get it to work, it is using the database for CDR |
| 20:31.08 | *** part/#asterisk ghtdak (n=ghtdak@75-165-47-59.tukw.qwest.net) |
| 20:32.32 | *** join/#asterisk variable_office (n=variable@208.73.60.2) |
| 20:32.58 | marl_ | anyone else any ideas? doesnt have to be complicated, just basic access to the call monitor files |
| 20:34.54 | vlt | Hello. I have an HFC4S card. Would you reccommend using it with Junghanns's qozap module and zaptel or with misdn? |
| 20:37.08 | *** join/#asterisk [hC] (n=hardcore@66.119.167.162) |
| 20:40.48 | *** join/#asterisk slykens (n=slykens@mail.verimedservices.com) |
| 20:42.31 | *** join/#asterisk sehh (n=sehh@cust-224-67.on1.ontelecoms.gr) |
| 20:43.24 | slykens | hello all... struggling with a pri over tdmoe setup... tdmoe looks like it is up but when channels are set in zapata.conf * won't load the zap channel driver... not sure what to check next... |
| 20:44.42 | *** part/#asterisk lirakis (n=eric@65.200.191.253) |
| 20:47.50 | *** join/#asterisk CBU[^_^]M`` (n=love@210.213.140.51) |
| 20:49.12 | *** join/#asterisk [TK]D-Fender (n=Joe@64.235.216.2) |
| 20:50.25 | *** join/#asterisk anthm (n=anthm@CPE-72-131-113-50.wi.res.rr.com) |
| 20:50.25 | *** mode/#asterisk [+o anthm] by ChanServ |
| 20:50.43 | *** join/#asterisk sigmounte (n=sigmount@bai59-1-88-172-80-96.fbx.proxad.net) |
| 20:50.56 | sigmounte | anyone use busy lamp field with grandstream ? |
| 20:56.31 | tzafrir | slykens, it is not asterisk's job to load drivers |
| 20:57.00 | tzafrir | have you set up /etc/zaptel.conf properly? |
| 20:57.14 | tzafrir | not sure though what it takes to set up TDMoE |
| 20:57.57 | *** join/#asterisk blq (n=Bl@dslb-088-064-135-076.pools.arcor-ip.net) |
| 20:58.10 | tzafrir | slykens, anyway, I figure you should pastebin relevant configs and /proc/zaptel/* |
| 20:59.48 | *** join/#asterisk jsaunders (n=nevermin@70.70.0.33) |
| 21:01.14 | slykens | i misspoke, the zaptel drivers are loaded and the "span" looks happily configured by ztcfg, just * won't talk to the zap drivers when the channels are specified... i think it is realted to the order in which the modules are being loaded so i'm pursuing it that way atm... |
| 21:02.53 | *** join/#asterisk karleeto (n=root@207.191.91.242) |
| 21:03.03 | karleeto | [TK]D-Fender: u here> |
| 21:03.32 | karleeto | whats that sip notify command you can send to a phone remotely to restart it? |
| 21:03.50 | [TK]D-Fender | karleeto, its in the WIKI, I've never done it personally |
| 21:04.11 | *** part/#asterisk slykens (n=slykens@mail.verimedservices.com) |
| 21:06.37 | scooby2 | whats the preferred card for 3 t1's? looking for something with hardware based echo cancellation |
| 21:07.04 | Qwell | scooby2: one of the Digium TE4xxp models |
| 21:07.46 | mocker | karleeto: For polycoms it's sip notify polycom check-config or something |
| 21:07.57 | mocker | I usually tab complete it. |
| 21:08.44 | *** part/#asterisk vlt (n=dm@suez.musketa.de) |
| 21:10.28 | *** join/#asterisk saftsack (n=saftsack@pD9E0551F.dip.t-dialin.net) |
| 21:12.57 | sigmounte | anyone use busy lampfield ?? |
| 21:14.07 | mocker | sigmounte: Yup. |
| 21:14.58 | sigmounte | yes ! |
| 21:15.11 | scooby2 | Qwell: thats what I figured thanks. Now to figure out of this pci slot is 3.3v or 5v |
| 21:15.26 | Qwell | scooby2: if you've got pcie, there are pcie versions |
| 21:15.33 | mocker | sigmounte: Ask questions in channel please. |
| 21:15.45 | mocker | I usually IRC in 5-10 second bursts, with many minutes in between. |
| 21:15.53 | mocker | :) |
| 21:15.59 | sigmounte | lol |
| 21:16.37 | sigmounte | ok , i think i've configured every thing on asterisk and on my gxp 2000 , but the blf is not working , and i don't know where to look for diagnostic |
| 21:16.40 | scooby2 | its an older dell. pe1850. just cracked the lid and looked like pci only. Currently has a sangoma a101 in it |
| 21:17.11 | mocker | sigmounte: Have you setup hints? |
| 21:17.21 | mocker | I've only done it on polycoms before. |
| 21:17.21 | sigmounte | currently the 3 led i've configured has blf are "on" , but nothing append when one of the user make a call |
| 21:17.36 | Qwell | scooby2: it shouldn't be difficult to find out the voltage - just look at the notches in the card |
| 21:17.49 | agx | sigmounte, do "sip show subscriptions" to check it subscribed the extensions |
| 21:18.20 | sigmounte | mocker, i've make an context named "blf" with hints like this : exten => 101,hint,Dial(SIP/sigmounte-grandstream) |
| 21:18.28 | agx | sigmounte, if you restart * it loose subscription, you have to set the phone to resubscribe/reregister or reboot it |
| 21:19.00 | sigmounte | agx, the command look like to work , can i post here one of the line ? |
| 21:19.19 | mocker | Er, that hint line doesn't look right. |
| 21:19.20 | sigmounte | 192.168.0.1 sigmounte- 436f83123bd 103@blf Idle pidf+xml <none> 6 active SIP subscriptions |
| 21:19.22 | mocker | let me check mine. |
| 21:19.32 | sigmounte | thanks mocker |
| 21:19.35 | *** join/#asterisk L2SHO (n=adam@fw01ext.voicepulse.com) |
| 21:19.39 | mocker | exten => 101,hint,SIP/101 |
| 21:19.49 | generalhan | anyone that has worked with an app using the manager API, looking at the samples i was able to send commands to *, but when trying to get the results back it gives me WAY more than i want to see. since the session is logged in, anything written to the CLI is posted back to me ... anyone have a way to stop this from happening.. to only give me the results i ask for and thats it ? |
| 21:19.49 | L2SHO | I'm having problems with my IVR in asteriskNOW, when I call in from my cell phone is sounds terrible, but if I call from en extension it sounds perfect |
| 21:19.50 | sigmounte | ok let change it |
| 21:19.52 | mocker | Pretty sure your hint shouldn't have a dial. |
| 21:20.15 | sigmounte | yes , it was in my mind , but not sure |
| 21:20.46 | *** join/#asterisk DarkRift (i=dark@bas10-montreal02-1177581830.dsl.bell.ca) |
| 21:21.48 | sigmounte | ok rebooting my grand stream ... |
| 21:22.09 | mocker | sigmounte: Did you reload your extensions as well? |
| 21:22.31 | GreggB | mocker: my blf hint doesn't have dial either (looks just like mocker's). Though the phone still doesn't seem to notice when when watched user makes a call either. Didn't you say you were using a GPX-2000 - that's what I'm using, and a while back stumbled across some docs which simply claimed it didn't work. |
| 21:23.08 | ct2clay | my trixbox pro is not gettting an address to access it at... its the VMware version... does anyone know how i can fix this? |
| 21:23.08 | mocker | You have to make sure the hints are in the same context that the phones register too. |
| 21:23.11 | sigmounte | mocker, yes , ok it look like to work better |
| 21:23.29 | mocker | sigmounte: Good to hear. |
| 21:23.35 | mocker | ~trixbox |
| 21:23.36 | jbot | well, trixbox is a full linux distro that includes FreePBX, and other 3rd party add-ons. It is all this extra stuff which makes trixbox seriously painful to support and hence you will find little help here for it. Try asking in #trixbox or on their forums & WIKI at http://www.trixbox.org |
| 21:25.00 | GreggB | Hmm, my extensions.conf starts with [stations] |
| 21:25.01 | GreggB | include => hint |
| 21:25.14 | sigmounte | mocker, is it normal the blf led only change when communication betwen two phone , and not also when the phone hang up ? |
| 21:26.33 | GreggB | mocker: and of-course [hint] is where I've packed all entries I wanted to watch with blf |
| 21:27.21 | mocker | GreggB: Where do your phones register to though? |
| 21:27.31 | mocker | In your sip.conf do you have a context=stations ? |
| 21:27.37 | mocker | For each phone? |
| 21:28.35 | *** join/#asterisk mosty (i=mostyn@60-241-198-194.static.tpgi.com.au) |
| 21:29.23 | GreggB | mocker: hmm, in the sip.conf that's the default context, but for the actual phone's SIP entry it's usersOI, which has an include => stations as the first entry after [usersOI] |
| 21:30.10 | mocker | GreggB: Since it's just an include, might try dumping that include into there as well. |
| 21:30.17 | mocker | It can't hurt (famous last words) |
| 21:30.25 | Spida | what is the difference between "=" and "=>"? |
| 21:30.33 | mocker | Spida: > |
| 21:30.48 | Spida | semantically? |
| 21:30.50 | *** join/#asterisk duki (n=duki@host-85-27-58-159.brutele.be) |
| 21:32.03 | mocker | Spida: One is used for extensions, one is used for setting parameters? |
| 21:32.23 | GreggB | mocker: well, that'll take a bit of time, as [stations] is where I've packed the inbound call routing details. Oy...the joy of learning as you go. It's time to clean up these configs... |
| 21:32.40 | JT | there are conventions for their use, but they're both interchangeable |
| 21:33.52 | GreggB | mocker: or did you mean dumping my [hint] data under userOI? (the phone's defined context) |
| 21:34.07 | mocker | GreggB: Just your hint data. |
| 21:35.41 | karleeto | [TK]D-Fender: sip notify XXX 'polycom-check-cfg' |
| 21:36.01 | mocker | karleeto: Hey, that's what I said. :) |
| 21:36.06 | [TK]D-Fender | karleeto, I always found pulling the network cable effective :) |
| 21:36.06 | mocker | Or something close I think. |
| 21:36.24 | mocker | You can also hold down volume -/+ mute and messages |
| 21:36.26 | Spida | thx |
| 21:37.42 | *** join/#asterisk duki (n=duki@host-85-27-58-159.brutele.be) |
| 21:38.33 | mosty | i have a box with a sangoma card in it, which does not appear in the output from lspci. the ports are lit up green (FXS ports). could it simply be that the card has become unseated? |
| 21:38.41 | *** join/#asterisk TokyoJimu (n=TokyoJim@ikebukuro.nccom.com) |
| 21:38.54 | mocker | mosty: did it ever work? |
| 21:39.05 | mosty | mocker, yes |
| 21:39.08 | *** join/#asterisk DarkRift (i=dark@bas10-montreal02-1177581830.dsl.bell.ca) |
| 21:39.30 | mocker | Hmm, did you upgrade kernels or reboot recently? |
| 21:39.57 | *** join/#asterisk duki (n=duki@host-85-27-58-159.brutele.be) |
| 21:40.10 | mosty | mocker, i shipped this box onsite (interstate) a few months ago. the tech interstate has only just turned it on. it was working when i sent it... |
| 21:40.34 | mocker | mosty: Did you screw the card in? :P |
| 21:40.37 | mosty | so yes there was a reboot, but no kernel upgrade (yet, i was thinking about trying that |
| 21:40.49 | mosty | it's a dell server- screwless |
| 21:40.51 | mocker | Might as well check the card, reseating is easy. |
| 21:41.44 | mosty | but a new kernel version shouldn't stop the card from appearing in lspci would it? i think it must be lower level than that |
| 21:43.19 | *** join/#asterisk k-man (n=jason@unaffiliated/k-man) |
| 21:45.23 | mocker | Yeah, shouldn't. |
| 21:45.50 | GreggB | mocker: hmph, it's still misbehaving. For example, sip show subscriptions shows: 10.0.10.219 3700 0589e48974d 3721@hint Idle dialog-info+xml <none> |
| 21:46.18 | mocker | GreggB: Is that more than before? |
| 21:46.46 | GreggB | mocker: though the GXP's key mode is "Asterisk BLF", the LED doesnt change color when 3721 makes or receives a call |
| 21:47.04 | GreggB | mocker: nope, same behavior - it has always shown a subscription |
| 21:48.20 | mocker | When you make a call does asterisk show a state change? |
| 21:48.40 | mocker | <PROTECTED> |
| 21:48.53 | mocker | Something like that? |
| 21:49.00 | *** join/#asterisk tristezo2k (n=seba@200.117.247.43) |
| 21:49.07 | mocker | (And you've set your phones to have those extensions listed on them?) |
| 21:49.38 | GreggB | good question...brb |
| 21:49.47 | tristezo2k | Hi everybody. |
| 21:49.48 | tristezo2k | I have some calls placed though an * that do not appear on the cdr details. |
| 21:49.51 | tristezo2k | Is this possible? |
| 21:50.48 | tristezo2k | I may change my question to, in which cirmcustances can this happen? :P |
| 21:51.30 | Spida | how can I check if zaptel is configured correctly? |
| 21:51.37 | mosty | tristezo2k, the dialplan can use this command: http://www.voip-info.org/wiki/view/Asterisk+cmd+NoCDR which will not produce a cdr entry for the call |
| 21:51.51 | mosty | Spida, plug in phones/lines and see if they work? |
| 21:52.08 | GreggB | mocker: for example, you mean if I were to call into a watched number, the "last state" under sip show subscriptions should change from something other than "idle"? If so, it just remains at "idle", even during a call. |
| 21:52.09 | mosty | also, check the output from ztcfg -vvvv |
| 21:52.23 | tristezo2k | mosty: thanks, but I *need* that cdr. I mean, they should be there, |
| 21:52.28 | tristezo2k | and they are not :P |
| 21:52.41 | Katty | oh noes. |
| 21:52.44 | mosty | tristezo2k, then don't use NoCDR() in your dialplan |
| 21:52.45 | Katty | i have a HORRIBLE problem |
| 21:52.51 | Katty | i just don't know what to do!!! |
| 21:52.58 | mocker | GreggB: Yeah. |
| 21:53.06 | mocker | It should change to no longer be idle. |
| 21:53.22 | tristezo2k | I usually have 2000-22000 calls per day. Today I recived a cdr from my pstn company, |
| 21:53.22 | tristezo2k | with some calls that do not appear on the asterisk cdr. |
| 21:53.45 | mosty | tristezo2k, how many? |
| 21:53.46 | *** join/#asterisk l2trace99 (n=asd@75.112.133.235) |
| 21:53.55 | tristezo2k | sorry, 2000-2200 |
| 21:54.02 | tristezo2k | I was just willing :) |
| 21:54.07 | mosty | tristezo2k, how many are not in your cdr though? |
| 21:54.10 | mocker | GreggB: I think Grandstream puts out new firmware like every other month. might wanna check for an update. :) |
| 21:54.19 | tristezo2k | I noticed just six. |
| 21:54.22 | GreggB | mocker: I wonder what I've done wrong then, as both SIP users are in the same context at this point. |
| 21:54.38 | tristezo2k | we are using asterisk 1.4.11 + psql for cdr |
| 21:54.42 | Spida | mosty: no, I am trying to debug a problem.... |
| 21:54.50 | mosty | tristezo2k, ok that's not so bad. it could be something simple like if you kill asterisk before it writes a cdr to the file (or db), then that cdr is lost |
| 21:54.54 | tristezo2k | those calls are not in the db not in the cdr-csv |
| 21:55.09 | Spida | mosty: for example, I get http://rafb.net/p/LoyIW692.html, even though I have only ONE card.... |
| 21:55.16 | codefreeze | tristezo2k: do these calls involve a transfer of some kind? |
| 21:55.21 | Spida | with ONE line (and two channels) |
| 21:55.27 | tristezo2k | well... that can be, but I am quite sure that asterisk was alive all the time. |
| 21:55.41 | mosty | spida: what does ztcfg -vvvv show? |
| 21:56.28 | tristezo2k | codefreeze: well, yes... and no... they are bridged |
| 21:56.30 | sigmounte | mocker, thanks for the help , but do you know why the blf led work only for incoming call and not for outgoing ? |
| 21:56.42 | tristezo2k | I have an AGI that selects the best provider |
| 21:56.54 | tristezo2k | then I dispatch the call properly. |
| 21:57.09 | Spida | it did show http://rafb.net/p/70TmnV75.html when I started, now I get http://rafb.net/p/AZyTTR19.html |
| 21:57.13 | mocker | sigmounte: Umm, not really. |
| 21:57.22 | codefreeze | tristezo2k: There's some known probs with cdr's in xfer situations, like where a gets a call, hits #100 or something, etc. |
| 21:57.37 | GreggB | mocker: sure enough, www.grandstreamsucks.com lists yet another update - looks like a nasty beta though (random lockups, and BLF reliability issues)... I just wish I could find a phone which could be used as an operator "console", and be done with the GXP all-together. |
| 21:58.14 | mocker | GreggB: Looked at Polycom's stuff? |
| 21:58.29 | mocker | GreggB: (Hint, I've never used a Grandstream. :)) |
| 21:58.46 | GreggB | Getting a IP4000 in a few days if that's what you mean :-) |
| 21:59.24 | mocker | GreggB: You need some 650s. :) |
| 22:00.32 | GreggB | mocker: I've been primarly looking for something where the operator can do two things she deems critical (cant say I disagree with her too much either): 1) quickly/instantly see an internal user's line status. 2) Transfer calls with the press of as few buttons as possible (ie: like pushing transfer, and then hitting the user's monitored button). |
| 22:00.33 | *** join/#asterisk k-man_ (n=jason@unaffiliated/k-man) |
| 22:01.12 | mocker | I've never tried the transfer to monitor button, but the sidecars w/ polycom are pretty awesome. |
| 22:01.24 | GreggB | mocker: right now she's got a list of extensions taped to the phone, and while that works alright for now - it's a hackup in this day-and-age. |
| 22:01.32 | mocker | GreggB: http://www.digiumcards.com/images/digium_cards_images/sidecar_for_601.jpg |
| 22:01.55 | GreggB | mocker: ohhh, pretty... |
| 22:02.04 | mocker | Time for home. |
| 22:02.08 | mocker | later all. |
| 22:02.52 | GreggB | mocker: thanks for the help |
| 22:04.00 | *** join/#asterisk k-man__ (n=jason@ppp244-232.static.internode.on.net) |
| 22:04.45 | Spida | mosty: what do you read from the ztcfg -vvvv output? |
| 22:05.45 | mosty | spida: looks right to me |
| 22:07.19 | Spida | mosty: so, why do I get two cards and no channels in asterisk? http://rafb.net/p/LoyIW692.html |
| 22:07.37 | mosty | spida: perhaps your zapata.conf is wrong? |
| 22:08.00 | ManxPower | Spida: did you use genzaptelconf? |
| 22:09.13 | Spida | ManxPower: yes |
| 22:09.40 | *** join/#asterisk ccaron (n=cedric@85-218-16-115.dclient.lsne.ch) |
| 22:09.54 | ManxPower | Spida: and did you import the generated file into /etc/asterisk/zapata.conf |
| 22:10.05 | tristezo2k | codefreeze: sorry, I had to leave a while. |
| 22:10.14 | ManxPower | because Asterisk will only read /etc/asterisk/zaptel.conf |
| 22:10.15 | Spida | ManxPower: yes |
| 22:10.35 | tristezo2k | ok... I will try to reproduce this. |
| 22:10.39 | Spida | ManxPower: /etc/asterisk/zapata.conf or /etc/asterisk/zaptel.conf ? |
| 22:10.45 | tristezo2k | Is there any known workarround? |
| 22:10.50 | codefreeze | It's OK. Just thought I'd let you know about the cdr 'weakness'. |
| 22:11.13 | Spida | I thought zaptel.conf was in /etc, and zapata.conf in /etc/asterisk? |
| 22:12.44 | mosty | codefreeze, what is the problem with cdr and transferred calls? |
| 22:13.07 | ManxPower | Spida: Sorry, the files are /etc/zaptel.conf (ztcfg) and /etc/asterisk/zapata.conf (asterisk) |
| 22:13.24 | l2trace99 | anyone know of a way besides the manager interface to capture sip registration events ? |
| 22:13.25 | codefreeze | In most cases, one of the legs of the call won't get show up in the cdr's. |
| 22:13.27 | Spida | ManxPower: ok, thats what I understood. |
| 22:13.50 | mosty | codefreeze, is there a fix? |
| 22:14.02 | *** join/#asterisk duki (n=duki@host-85-27-58-159.brutele.be) |
| 22:14.12 | codefreeze | working on it. Over a week so far of labor, maybe another to go. |
| 22:14.17 | *** join/#asterisk mindCrime (n=chatzill@rrcs-96-10-17-117.se.biz.rr.com) |
| 22:14.24 | ManxPower | codefreeze: do it will never be in 1.4 or 1.2 |
| 22:14.48 | mosty | codefreeze, which version(s) of asterisk? |
| 22:15.00 | codefreeze | Well, I'm aiming at fixing the bugs in 1.4 and trunk. |
| 22:15.12 | mosty | how about 1.2 ? |
| 22:15.17 | codefreeze | mosty: ALL versions of asterisk are subject to this bug. |
| 22:15.18 | ManxPower | if it takes 2 weeks to fix the bug then it is too much of a change to go into 1.2 or 1.4 |
| 22:15.59 | codefreeze | ManxPower: 1.2 is totally out of the picture. And if folks don't want it in 1.4, it won't show up there. |
| 22:16.34 | ManxPower | codefreeze: "want" has nothing to do with it. There are pretty strict guidelines as to what goes into a released version of Asterisk |
| 22:17.15 | codefreeze | true. It's a lot of work, but actually fairly small tweaks. |
| 22:17.28 | ManxPower | released versions of Asterisk are feature frozen. I guess it comes down to Digium deciding if your work is a "new feature" or a "bug fix". There have been times that a bug fix would have been too intrusive in the code and it was not put into the released code. |
| 22:18.02 | mosty | codefreeze, do you have a link to the bug on bugs.digium.com ? |
| 22:18.14 | *** part/#asterisk hads (n=hads@reef80.anchor.net.au) |
| 22:18.27 | codefreeze | True, ManxPower, we'll have to decide when I'm done, if they'll only go in trunk, or in 1.4 and trunk. |
| 22:18.37 | tristezo2k | codefreeze: Yes, please, I would like to try to reproduce this cdr bug. |
| 22:19.07 | tristezo2k | and report more data if I found it. |
| 22:19.33 | ManxPower | I'm just so glad I don't bill for calls. |
| 22:19.39 | codefreeze | try 10927, 11093 |
| 22:19.49 | tristezo2k | ManxPower: Yes, it is a pain in the 455... |
| 22:20.24 | codefreeze | In general, do ANY kind of transfer with ANY kind of interface, and look at what CDR's come out. Then, shake your head. and sigh. |
| 22:20.57 | tristezo2k | well... I did not mention that my provider serves me by sip.. |
| 22:21.03 | tristezo2k | it is not an zap interface. |
| 22:21.25 | codefreeze | doesn't matter what interface, they ALL have the same sets of probs. |
| 22:21.57 | *** join/#asterisk kand (n=kanderso@139.148.188.72.cfl.res.rr.com) |
| 22:24.45 | kand | occasionally my * 1.4 will hang at any audio file playback, I can either wait a while and it will start working or restart zaptel and it works immediately. Any ideas why? |
| 22:26.10 | mosty | codefreeze, 11093 says it did not occur on versions of asterisk earlier than 1.4.13, are you trying to fix a more general problem with xfer cdr's? |
| 22:26.40 | codefreeze | yes, mosty |
| 22:27.08 | Spida | genzaptelconf has put signalling=bri_cpe in zapata-channels.conf . however, asterisk doesn't seem to know this signalling. what do I have to put in there (and in zapata.conf) for signalling? I believe that I need fxs_ks to connect to the pstn, but if I use it on both places I get chan_zap.c:7013 mkintf: Signalling requested on channel 1 is FXS Kewlstart but line is in PRI Signalling signalling |
| 22:27.10 | mosty | codefreeze, ok well good luck :) |
| 22:27.37 | mosty | spida: fxs_ks is only for analogue lines- BRI is not analogue |
| 22:27.59 | codefreeze | thanks! I have over a dozen test cases in different sequences and permutations, and it's quite tricky to get them all to come out just right. |
| 22:28.01 | mosty | spida: do you have a BRI or PRI ISDN card/connection? |
| 22:28.27 | JT | Spida: using bristuff? |
| 22:28.28 | Spida | mosty: oh; i didn't know that. its a pri-line |
| 22:28.32 | Spida | aeh |
| 22:28.33 | Spida | bri |
| 22:28.38 | Spida | just 2 channels |
| 22:28.47 | JT | Spida: using bristuff? |
| 22:29.07 | Spida | no (at least not knowingly) |
| 22:29.13 | JT | i swear i told you last time, there is no such signalling as bri_cpe |
| 22:29.34 | Spida | thats whats asterisk tells me, too. |
| 22:29.41 | Spida | but what should I use? |
| 22:29.46 | JT | Spida: well what driver are you using for what card, and what channel driver? |
| 22:29.49 | *** join/#asterisk [hC] (n=hardcore@66.119.167.163) |
| 22:30.03 | JT | make sure your asterisk version is bristuff patched first |
| 22:30.23 | Spida | zaphfc |
| 22:30.37 | mosty | the only BRI card i have experience with is the digium one, which uses misdn, not zaptel |
| 22:30.43 | J4k3 | I can run sip over this pretty damned well... I'd venture to say better than AT&T's GSM around here (which isn't saying much)... ftp://ftp.intrastar.net/www/IRC/carpics/ |
| 22:30.44 | Spida | zaptel, for a hfc card. |
| 22:30.47 | J4k3 | err |
| 22:30.48 | JT | misdn is the worst ever |
| 22:31.03 | J4k3 | http://www.intrastar.net/~jsuter/IRC/carpics/ |
| 22:31.04 | J4k3 | that |
| 22:31.12 | mosty | JT, i have two production boxes using misdn, no issues |
| 22:31.24 | JT | mosty: trust me, misdn sucks major balls |
| 22:31.27 | MrBIOS | anyone here have any H.323 experience with asterisk? |
| 22:31.52 | JT | Spida: zaphfc is a bristuff driver usually, not sure how well it works without bristuff |
| 22:32.18 | mosty | JT, i heard that sangoma have a BRI card now, but there are only two in australia. i will gladly use that when it becomes available |
| 22:33.25 | Spida | JT: I didn't know I had to use bristuff... |
| 22:34.39 | JT | Spida: i don't know if you have to, i still thought you need some sort of zaptel or chan_zap patch |
| 22:34.51 | JT | mosty: yeah it sounds a lot better |
| 22:35.00 | JT | mosty: the even easier option is PRI |
| 22:35.33 | kand | anyone know why the audio playback cmd would occasionally hang indiffently? |
| 22:35.54 | JT | what is "indiffently" |
| 22:36.00 | mosty | JT: if you need more than a certain number of channels, of course |
| 22:36.18 | mosty | kand, are you using ztdummy? |
| 22:36.20 | kand | Until a new call or after restarting zaptel |
| 22:36.22 | kand | yes |
| 22:36.26 | JT | mosty: the price point is around 5 or 6 channels. |
| 22:37.05 | mosty | yeah, i have a few boxes with 6 channels |
| 22:37.07 | JT | i'd make it 5 channels, for the avoidance of hassle |
| 22:37.10 | kand | mosty: yes |
| 22:37.16 | JT | if you need 5 or more channels in australia, get a PRI |
| 22:37.55 | mosty | kand, ztdummy sucks. especially on dell's. i would recommend getting a card that provides zaptel timing, like a cheap used x100p clone, if you have to |
| 22:38.23 | *** join/#asterisk craigk (n=ckowald@58.174.122.198) |
| 22:38.41 | kand | I had the same thought today and ordered a single span, but this box has been running for months with no problems |
| 22:38.51 | kand | Just wondering why |
| 22:39.00 | ManxPower | kand: why do you need a timing source? |
| 22:39.28 | kand | No, sorry, why it started having problems |
| 22:39.30 | mosty | are you trunking or using meetme? |
| 22:39.51 | ManxPower | specifically are you using IAX2 trunking or meetme? |
| 22:39.59 | kand | yes, IAX2 and I know it depends on zaptel |
| 22:40.14 | mosty | you could try disabling iax trunking in the meantime |
| 22:40.20 | ManxPower | iax2 does not depend on zaptel, only the trunking option does. |
| 22:40.34 | kand | not really an option. I didnt know that |
| 22:40.46 | kand | nm, I can could turn off trunking |
| 22:41.22 | ManxPower | IAX2 trunking helps save bandwidth used by UDP packet headers by putting the audio from more than 1 call in a single UDP packet, this only helps if you have more than 2 calls going between the same two machines. |
| 22:41.49 | kand | One last question, same time this happened had about 2 calls with one way audio after hold, do you think it is related? |
| 22:42.01 | ManxPower | and turning trunking on would actually use more bandwidth if you have 1 or 2 calls between the same two machines. |
| 22:42.50 | kand | I was aware of that, these usally have four or five |
| 22:42.59 | *** join/#asterisk cspot (i=cspot@ip68-1-63-100.pn.at.cox.net) |
| 22:43.45 | ManxPower | I thought ztdummy in 1.4x used the Linux kernel RTC if it was on kernel 2.6x |
| 22:44.04 | kand | I am under that impression too |
| 22:44.30 | ManxPower | what version are you using? |
| 22:44.38 | Spida | so, what signalling should I use for a isdn bri line to pstn? |
| 22:44.45 | kand | latest |
| 22:44.52 | kand | 1.4.13 |
| 22:44.54 | ManxPower | Spida: whatever the docs say. |
| 22:44.54 | JT | Spida: check with the telco if the line is PTP or PTMP |
| 22:45.12 | ManxPower | JT: aren't most PTP? |
| 22:45.15 | JT | no. |
| 22:45.21 | Spida | JT: and depending on that I choose between which alternatives? |
| 22:45.33 | JT | it depends on telco and how the line was provisioned |
| 22:45.38 | JT | most in australia ar PTMP |
| 22:45.40 | JT | are |
| 22:46.20 | *** join/#asterisk [hC] (n=hardcore@66.119.167.162) |
| 22:46.22 | Spida | does anybody know what I need in germany? |
| 22:46.46 | JT | probably PTMP |
| 22:46.50 | JT | but RING YOUR TELCO |
| 22:47.11 | JT | bri_net_ptmp |
| 22:47.16 | JT | err |
| 22:47.24 | JT | bri_cpe_ptmp i mean |
| 22:48.13 | *** join/#asterisk karleeto (n=karl@207.191.91.242) |
| 22:48.28 | Spida | looks much better now.... |
| 22:50.04 | Spida | zap show channels lists pseudo and 2 channels. |
| 22:51.06 | Spida | does "== Primary D-Channel on span 1 down" mean that I chose the wrong signalling? |
| 22:51.20 | mosty | quite possibly |
| 22:51.24 | JT | it means it thinks the channel is down |
| 22:51.30 | JT | does it ever go up? |
| 22:52.11 | Spida | if it does, asterisk doesn't tell me... |
| 22:52.32 | Spida | it just repeatedly tells me that is/goes down |
| 22:52.51 | *** join/#asterisk dserban (n=dserban@S010600195b2f0f96.ok.shawcable.net) |
| 22:52.52 | JT | right |
| 22:52.56 | JT | pri intense debug span 1 |
| 22:53.23 | *** join/#asterisk DanielHolth (n=chatzill@gleim-gw.atlantic.net) |
| 22:53.44 | dserban | Hmm, who should I look for as a VOIP provider? Need something like 30 channels for long distance dialing internationally as well as incoming toll-free lines. Everyone I look at seems to be residential oriented, is there anyone out there someone can recommend? |
| 22:54.34 | dserban | I'm in canada and our PRI provider has been down for almost a full day :(. Need to look at other options. |
| 22:54.49 | dserban | Seeing that we're a call center... it starting to suck |
| 22:55.17 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 22:55.39 | JT | long term... get a second pri provider? |
| 22:55.47 | karleeto | http://rafb.net/p/gBiXtv89.html |
| 22:56.34 | karleeto | is a multisite asterisk setup in a triangle (three boxes with 2 trunks to the others on each box) impossible? |
| 22:57.15 | karleeto | i did this in a pretty straight foreward matter, and now I'm having this problem and don't some why, someone please read my post |
| 22:57.21 | karleeto | PLEASE ;) |
| 22:57.22 | JT | don't think of them as trunks if they're just voip connections |
| 22:58.27 | DanielHolth | Does asterisk-java have an irc channel? The manager connection's login() method unexpectedly never returns. |
| 22:59.01 | mosty | karleeto, so you can't call from box A to box B? (ignore box C for now.) what happens when you try calling A->B? |
| 22:59.25 | mosty | karleeto, and does calling B->A work? |
| 22:59.45 | karleeto | i figured it out!!!!! |
| 22:59.47 | mosty | methinks your dialplan/iax.conf don't match up properly |
| 23:00.20 | GreggB | Is there any way to obtain state information on sip peers (ie: ringing, busy, in-use, etc...) without limiting the number of calls that a sip peer can make at a time (ie: without having to set call-limit to 1)? |
| 23:00.52 | GreggB | dserban: ~itsp |
| 23:01.01 | GreggB | ~itsp |
| 23:01.02 | jbot | itsp is, like, an Internet Telephony Service Provider, or a "VoIP Phone Company". |
| 23:01.23 | mosty | greggb: take a look at BLF (snom, polycom, some others) support it |
| 23:01.48 | J4k3 | ~itsybitsyspider |
| 23:02.02 | GreggB | oy... that used to list Teliax as a "...sucks less than most others..." provider. I guess that comment got pulled recently - any ideas why? |
| 23:02.23 | J4k3 | becuase this is #asterisk not #voipspam? |
| 23:02.30 | tristezo2k | regards! |
| 23:02.32 | *** part/#asterisk tristezo2k (n=seba@200.117.247.43) |
| 23:02.33 | mosty | GreggB, http://www.hurdman.net/mirror/voip-info/wiki/view/Asterisk+presence.html |
| 23:02.50 | GreggB | mosty: that's exactly what I'm trying to get working, but I thought BLF relied on SIP hint feedback? |
| 23:02.59 | mosty | yes, it does |
| 23:04.43 | GreggB | J4k3: fair enough... dserban: well, I happily use Teliax.com though YMMV |
| 23:05.35 | GreggB | mosty: So from my understanding of this, I must restrict each "monitored" |
| 23:05.58 | GreggB | mosty: sip peer to effectively just one usable line then |
| 23:06.44 | mosty | GreggB, with asterisk 1.4 maybe (i'm not sure). i only use it with 1.2 |
| 23:08.04 | GreggB | mosty: ouch - that sucks then. Our desktop phones can support four "lines", I'd hate to explain to a user that in order for the operator to see if they're on the line or not, that I'd have to restrict them to just one call at a time. |
| 23:08.35 | kand | GreggB: You simply need to set a limit, it doesnt have to be one |
| 23:09.27 | GreggB | kand: I better take a closer took again, because I tried setting a limit higher than 1 before, and the SIP peer's status didnt change. |
| 23:10.25 | kand | GreggB: I set all mine to 40, and it works for polycom 'buddy watch' and grandstream BLF |
| 23:11.48 | kand | GreggB: check the general section of sip.conf for --- notifyringin=yes notifyhold=no limitonpeers=yes |
| 23:12.16 | kand | GreggB: then each peer --- call-limit=40 |
| 23:12.54 | GreggB | kand: yes, yes, yes (I love templates)...brb |
| 23:16.26 | kand | GreggB: in that case does each extension have a (exten => 120,hint,SIP/001120) and when are you watching the extension (not the sip channel if the name is different)? |
| 23:17.23 | GreggB | kand: well, that actually worked (setting call-limit=5 in my case), I tried this around 4-5 months ago, and that setting didnt change the peer's status... of-course I was using an older version of 1.4 at the time...oy |
| 23:17.53 | kand | GreggB: excellent |
| 23:24.52 | *** join/#asterisk mordaunt (n=mordaunt@unaffiliated/mordaunt) |
| 23:27.12 | *** join/#asterisk atomicd (n=Atomicd@74-206-0-80.static-ip.m.telepacific.net) |
| 23:28.46 | *** part/#asterisk atomicd (n=Atomicd@74-206-0-80.static-ip.m.telepacific.net) |
| 23:37.18 | *** join/#asterisk cnbrk (n=asd@host-19-4.mielccc.clients.pavlovmedia.com) |
| 23:37.21 | cnbrk | hi |
| 23:37.31 | fujin_ | HELLO cnbrk |
| 23:37.34 | fujin_ | how are you going |
| 23:37.37 | *** join/#asterisk mightnare (n=mightnar@s230165.ppp.asahi-net.or.jp) |
| 23:38.51 | cnbrk | i have a client connected to my asterisk server, the client is not behind a firewall. connected directly to internet, the pap2 that my client using is plugged directly to the network outlet on the wall provided by the resident management and has a real ip, port 1 of pap2 is working, however, port 2 can call and receive calls but no sound on calls at all. No sound is sent or received. Why do you think is this? |
| 23:39.02 | *** join/#asterisk Runlvl (n=gustavo@190.2.40.57) |
| 23:39.03 | cnbrk | ] |
| 23:39.03 | cnbrk | [Nov 2 01:30:13] WARNING[74630]: chan_sip.c:1938 retrans_pkt: Maximum retries exceeded on transmission b20112d3-3b9c16a7@76.10.16.220 for seqno 102 (Critical Response) |
| 23:39.03 | cnbrk | [Nov 2 01:30:13] WARNING[74630]: chan_sip.c:1962 retrans_pkt: Hanging up call b20112d3-3b9c16a7@76.10.16.220 - no reply to our critical packet. |
| 23:39.04 | Runlvl | Estan invitados a http://www.asterisk-la.org , comunidad de Asterisk en espaņol :D |
| 23:39.04 | GreggB | kand: sorry, PHB came by |
| 23:39.23 | l2trace99 | anyone know of a good sip/jabber client ? |
| 23:39.35 | GreggB | kand: very cool - thanks. Now I'm off to see if theres a way for her to do blind-transfers by just hitting transfer and then the associated button for that destination user (using a GXP2000) |
| 23:41.31 | JT | cnbrk: is your asterisk server on nat? |
| 23:43.31 | *** join/#asterisk anthm (n=anthm@CPE-72-131-113-50.wi.res.rr.com) |
| 23:43.31 | *** mode/#asterisk [+o anthm] by ChanServ |
| 23:45.50 | GreggB | hmm... he's not really a PHB, I wonder what an anti-PHB is called, as my boss is actually knowledgeable and highly-competent... I'm drifting OT though... |
| 23:50.22 | *** join/#asterisk DarkRift (i=dark@bas10-montreal02-1177581830.dsl.bell.ca) |
| 23:51.21 | Spida | JT: after issuing "pri intense debug span 1" I get no more messages saying that thte d-channel is down. |
| 23:51.50 | Spida | but after restarting asterisk, I get these messages again.... |
| 23:52.56 | *** part/#asterisk zerohalo (n=zeroHalo@pool-71-255-162-167.bstnma.east.verizon.net) |
| 23:53.50 | *** join/#asterisk ylon (n=ylon@cpe-76-181-182-10.columbus.res.rr.com) |
| 23:54.45 | ylon | I was told that it was impossible to totally remove echo from an asterisk phone system. When the users pick up and start talking they are very much upset with the echo that happens at first and I wanted to see if someone here might know of some solutions |
| 23:56.03 | *** join/#asterisk rnovotny22 (n=root@000-119-180.area2.spcsdns.net) |
| 23:56.46 | *** join/#asterisk lowlevel (n=Stuart@CPE0017f2e2a3e8-CM000f9f7d6742.cpe.net.cable.rogers.com) |
| 23:57.32 | kand | ylon: is the echo on zaptel channels? |
| 23:58.36 | ylon | apparently, I know that the person who put this together used zaptel |
| 23:58.45 | *** join/#asterisk hads (n=hads@reef80.anchor.net.au) |
| 23:59.58 | kand | ylon: Then this may help you http://www.asterisktutorials.com/videos/echo/movie.html |
