| 00:02.42 | pdtpatrick__ | anyone got experience with openvxml and asterisk ? |
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| 00:19.59 | Eitan | so im having a silly issue, phone quality is great, but hodl music has pretty bad static |
| 00:20.03 | Eitan | tried both mp3 and .wav |
| 00:20.17 | Eitan | files sounds fine if i play from comp, just not through the phone |
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| 01:16.38 | fagiani | anyone knows how to avoid Sangoma card to take the eth0 place if the card is being used to voice purposes? |
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| 01:37.17 | p3nguin | fagiani: You mean that the card is an Ethernet card? |
| 01:37.40 | fagiani | p3nguin: not really this is very odd |
| 01:37.59 | fagiani | because somehow i am not being able to set the ethernet card as eth0 it always get eth1 |
| 01:38.19 | fagiani | and i issued kudzu --probe and i get device: eth0 to be Sangoma E1 card :P |
| 01:42.27 | p3nguin | Can you get the output of "lspci -nnqv" for the card and pastebin it? |
| 01:43.28 | WiretapWork_ | p3nguin, the sangoma cards do indeed declare themselves as network interfaces |
| 01:43.48 | WiretapWork_ | though usually with wXpY |
| 01:44.17 | p3nguin | If the card were mine, I would create a couple of udev rules to make sure my actual NIC was always the device name I wanted it to be and the Sangoma card is always something different. |
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| 01:55.59 | LemensTS | I can call out a pots line from a polycom 321 just fine, but people calling in to it, I get like 1-2 seconds of bad echo from myself |
| 01:57.22 | WiretapWork_ | fix your echo cancellation |
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| 02:25.33 | Tulga | I connected asterisk box to E1 and configured 3333 number. but when call 3333 nothing happen in asterisk. only busy caller side |
| 02:25.53 | Tulga | line and configuration looks ok. checked 10 times |
| 02:28.32 | p3nguin | Do you know in what context you created extension 3333? Can you tell me what extension 3333 is supposed to do? |
| 02:31.14 | Tulga | extension created to ring SIP account |
| 02:31.25 | kaldemar | Tulga: where did you configure the number? what device are you using to call? where is the device connected to? |
| 02:31.39 | p3nguin | That's half of the info I asked for. |
| 02:32.11 | Tulga | kaldemar: routed number in operator and using cell phone to call it |
| 02:33.16 | Tulga | p3nguin: problem is I cannot see any debug or error message in asterisk |
| 02:33.23 | kaldemar | Tulga: if you see nothing in asterisk, then you're not getting the call. |
| 02:33.25 | p3nguin | I didn't ask that. |
| 02:33.43 | Tulga | 3333 to receive call |
| 02:33.49 | Tulga | like ACD |
| 02:34.13 | Tulga | kaldemar: yes. but route is ok, E1 connection is OK (ISDN) |
| 02:34.27 | Tulga | how to troubleshoot it? |
| 02:35.54 | kaldemar | enable verbosity and pri debug. if you still don't see anything, your line is not ok or you're not getting a call from the operator. |
| 02:36.13 | *** join/#asterisk coppice (~chatzilla@m121-202-47-110.smartone-vodafone.com) |
| 02:36.57 | kaldemar | can you make outbound calls through the PRI? |
| 02:37.27 | Tulga | kaldemar: I cannot make outbound calls |
| 02:37.32 | Tulga | it is only incoming |
| 02:37.46 | hobodave | finally fixed my mysterious asterisk 1.8.4 problem |
| 02:38.01 | hobodave | solution? install 1.8.4.2 |
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| 03:19.21 | zachsis | any freepbx people here? Does anyone know how to enable 'groundstart' as the signalling mode on FreePBX? I can hand edit it, but the file gets modified every time i do anything with the GUI |
| 03:19.37 | zachsis | and groundstart isn't an option. only 'loopstart' and 'kewlstart' |
| 03:19.40 | seraphie | #freepbx |
| 03:19.46 | zachsis | ty |
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| 04:25.46 | JuStIcIa_ | hi guys, one question for the high availability i just need only VRRP on my two asterisk ? |
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| 04:55.22 | doneir | hey all, i have E1 PRI config info from my telco, and have no alarms on the E1 line. However, the d-channel does not seem to be coming up even though I have the correct settings. My pri span shows: http://pastebin.com/Wdd1CxiQ . My systems.conf is: http://pastebin.com/hvEEuz8u . My chan_dahdi.conf is: http://pastebin.com/v6nM8xVP |
| 04:55.51 | doneir | the remote switch is a siemens EWSD DSS1 (euroisdn) |
| 04:57.15 | doneir | pri debug just shows SABME sends |
| 04:57.59 | doneir | i've removed the accunet line form the chan_dahdi.conf , was just trying that option out to no avail |
| 05:04.20 | *** join/#asterisk timahvo1 (~rogue@41.223.57.77) |
| 05:07.28 | doneir | can the loadzone/defaultzone in systems.conf stop the ability to up the d-channel? |
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| 05:24.41 | kleszcz | morning |
| 05:25.21 | doneir | afternoon |
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| 05:41.13 | doneir | i've tried using hardhdlc instead of dchan to no avail |
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| 06:24.53 | doneir | well, sent to telco a query to see if their d-channel is indeed on 16 (would be strange if it weren't) and to make sure the sircuit isn't in 'adminstrative down' mode |
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| 06:41.38 | WIMPy | doneir: The loadzone is only used for generating tones AFAIK. Shouldn't matter technically. |
| 06:43.00 | WIMPy | But if you don't receive ANYTHING at all, there's somethign else going on. |
| 06:46.46 | doneir | okay great, good to rule that out |
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| 07:49.27 | cneb3000 | good morning sunshine |
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| 08:21.47 | mandla | Hello... |
| 08:23.00 | WIMPy | mandla: Good morning. How's your server today? |
| 08:24.05 | mandla | WIMPy, im trying to reinstall the asterisk server. |
| 08:25.07 | WIMPy | Have you confirmed that everything is working? |
| 08:28.38 | mandla | WIMPy, yah everything is working, how do i install dahdi on asteriskNow? |
| 08:30.41 | *** join/#asterisk irroot (~irroot@41.53.15.228) |
| 08:31.26 | irroot | hi there mandla mzanzi is not warm today you well |
| 08:35.18 | *** join/#asterisk Wiretap (~Wiretap@unaffiliated/wiretap) |
| 08:36.25 | *** join/#asterisk k3asd` (~k3asd`@pdemichele.broker.freenet6.net) |
| 08:37.06 | k3asd` | hi |
| 08:37.22 | irroot | morning folks |
| 08:43.39 | cneb3000 | how do, irroot |
| 08:45.54 | *** join/#asterisk hetii (~Grzegorz@194.181.154.25) |
| 08:45.58 | hetii | Hello :) |
| 08:46.14 | cneb3000 | hey hetii |
| 08:47.23 | hetii | I try to use t38modem + hylafax + * and something is strange for me |
| 08:48.51 | hetii | i run t38modem ninary with those params: /usr/bin/t38modem --no-h323 -u T38modem${COUNTER} ... so the user is T38modem0 on asterisk by freepbx i set extenstion but instead of name T38modem0 i put some digits |
| 08:49.28 | hetii | when i do sip show peers i see: 100600 127.0.0.1 A 6060 OK (3 ms) |
| 08:49.53 | hetii | so why the status is 3ms when the user don`t math :/ |
| 08:51.33 | *** join/#asterisk GreatSUN (~greatsun@188-22-185-129.adsl.highway.telekom.at) |
| 08:51.40 | GreatSUN | hi all |
| 08:51.48 | WIMPy | Because it doesn't matter. |
| 08:52.26 | *** join/#asterisk irroot (~irroot@197.108.65.133) |
| 08:52.35 | kaldemar | hetii: host and port matters for qualify. |
| 08:53.00 | *** join/#asterisk jacobkiers_ (4d3c9662@gateway/web/freenode/ip.77.60.150.98) |
| 08:53.24 | hetii | and this is only required without any authorization ? |
| 08:53.35 | *** join/#asterisk engrxyz (~puitpyitr@212.23.51.7) |
| 08:53.48 | WIMPy | It's just like a ping. |
| 08:53.56 | hetii | i seee |
| 08:53.58 | hetii | ok |
| 08:56.38 | mandla | irroot, Yah im good. Its only that my problem still persists, even after i reinstalled CentOS AsteriskNow sever, the server still cant detect the astribank. |
| 08:56.45 | hetii | strange even when user don`t match, when i call to this extension i see some logs on t38modem side :/ |
| 08:58.00 | hetii | http://pastebin.ca/2077207 |
| 08:58.14 | kaldemar | hetii: the status has little to do with calling, and the name has nothing to do with authentication of outbound calls. |
| 08:58.15 | hetii | on both case when the user math or not i do.. |
| 08:58.22 | irroot | mandla you on the centos command line ?? |
| 08:58.22 | *** join/#asterisk TimeRider (~steve@188-220-33-123.zone11.bethere.co.uk) |
| 08:58.25 | irroot | as root |
| 08:58.53 | kaldemar | hetii: is something not working? |
| 08:59.32 | mandla | irroot, no but i can quickly go there. |
| 08:59.46 | irroot | yeah |
| 09:00.18 | tzafrir_laptop | mandla, please pastebin the output of: dahdi_hardware -v |
| 09:00.57 | irroot | and lsusb |
| 09:01.14 | hetii | to be onest i jast set t38modem+ asterisk, hylafax is not set with this modem yet so i don`t know if i can test it somehow on this stage. |
| 09:01.31 | hetii | *just |
| 09:01.47 | *** join/#asterisk sgimeno (~chatzilla@163.117.206.10) |
| 09:01.54 | irroot | heti you using --old-asn |
| 09:02.04 | *** join/#asterisk timahvo1 (~rogue@41.223.57.75) |
| 09:02.28 | irroot | had endless problems with it without that option using v2 here |
| 09:02.48 | hetii | no, i remove this option because i got from t38modem message that this argument is unknow |
| 09:03.05 | hetii | i use version 1.2 |
| 09:03.52 | irroot | ah ok could explain it thx |
| 09:04.31 | mandla | tzafrir_laptop, Where should i run that command? |
| 09:05.54 | mandla | usb:002/002 xpp_usb- e4e4:1160 Astribank-modular no-firmware |
| 09:06.01 | mandla | Thats the output |
| 09:06.13 | hetii | also --sip-redundancywas unknow... |
| 09:06.41 | irroot | ...:~# |
| 09:11.35 | *** join/#asterisk cneb3000 (~Ben@87.127.15.145) |
| 09:12.11 | cneb3000 | Hey guys.. just doing some general research at the minute. in particular regards to security and Asterisk. |
| 09:12.48 | cneb3000 | I found something called astmanproxy, which seems good at what it does. are there any other products like that.. in the vain of 'asterisk security' that are worth looking at? |
| 09:13.00 | cneb3000 | not looking for anything in particular, just useful stuff out there. |
| 09:13.22 | irroot | heti it works well have used it for testing T38 gateway to mISDN / DAHDI |
| 09:14.49 | cneb3000 | ooo.. failtoban looks cool.. |
| 09:15.18 | irroot | mandla you can look at the output of dmesg to see errors too |
| 09:15.51 | *** join/#asterisk fhmiv (~fhmiv@adsl-065-005-242-247.sip.ags.bellsouth.net) |
| 09:15.58 | irroot | cneb3000 yeah i use iptables limit too |
| 09:16.09 | irroot | its a little tricky to set up but works great |
| 09:16.24 | cneb3000 | googles |
| 09:17.07 | irroot | /sbin/iptables -A TARPIT -j RETURN -m state --state ESTABLISHED -m limit --limit 2/s --limit-burst 5 |
| 09:17.09 | irroot | /sbin/iptables -A TARPIT -j RETURN -m state --state NEW -m limit --limit 2/s --limit-burst 5 |
| 09:17.11 | irroot | /sbin/iptables -A TARPIT -j LOG -m recent --rcheck --seconds 30 --hitcount 20 --name RATELIM -m limit --limit 6/minute --limit-burst 1 --log-prefix "RATELIM " --log-level debug |
| 09:17.12 | irroot | /sbin/iptables -A TARPIT -j DENY -m recent --name RATELIM --update --seconds 30 --hitcount 20 |
| 09:17.53 | irroot | it screws up brute force password attacks at high packet rates |
| 09:18.02 | cneb3000 | I see yes |
| 09:18.08 | cneb3000 | hmm, that is useful thanks :) |
| 09:18.48 | irroot | /sbin/iptables -A TARPIT -j RETURN -m recent --name RATELIM --set |
| 09:19.23 | cneb3000 | thanks irroot, i'll have a play |
| 09:20.01 | irroot | also helps for pop3 / smtp / .... |
| 09:20.09 | WIMPy | irroot: Nice idea, but doesn't work for SIP. I fell for that as well. |
| 09:20.12 | tzafrir_laptop | Hmm... mandla's issue seems to be https://issues.asterisk.org/jira/browse/ANOW-143 |
| 09:20.55 | WIMPy | After the first reply from Asterisk, all probes from the haxx0rs will be treated as ESTABLISHED. |
| 09:21.04 | irroot | WIMpy how ?? |
| 09:21.13 | irroot | ah i must double check this |
| 09:22.12 | WIMPy | That's the bad thing about UDP. "connections" are guesswork. |
| 09:22.16 | irroot | thx for heads up |
| 09:22.47 | irroot | ah but wait |
| 09:23.03 | irroot | im looking at new and established connec |
| 09:24.00 | irroot | not as per default allow established in ... |
| 09:24.52 | WIMPy | Oh, right. I didn't read the whole thing because of the waraparond in the irc window. You can leave that part out and only use one line, then. |
| 09:25.38 | WIMPy | But the usual lecture about shooting yourself in the foot applies, off course. |
| 09:26.18 | irroot | dude i only have one foot left :P |
| 09:26.24 | WIMPy | You just make it easier to be DOSed. |
| 09:26.44 | irroot | yeah need to watch it carefully |
| 09:27.31 | WIMPy | A lot easier that the usual fail2ban approach. |
| 09:27.52 | irroot | its one of those catch 22 things |
| 09:28.14 | Haraken | would a P4 with 2GB of ram on a raid1 be sufficient for an asterisk/freepbx server running five channels? |
| 09:28.16 | irroot | possibility of DOS is more but reduces a finacial risk |
| 09:29.02 | WIMPy | Haraken: Only if it has a lot of other things to do as well. |
| 09:29.08 | irroot | Haraken sounds bout right im using Atom 1.8 [DC] with 2 BRI [mISDN] and couple sips |
| 09:29.39 | Haraken | server would definitely be dedicated to asterisk, just wanted to make sure |
| 09:29.50 | irroot | have some P4's with 1Gb ram out there still |
| 09:30.30 | irroot | Haraken run squid and samba on the atom's for small use they rock |
| 09:31.01 | Haraken | irroot, atoms would work for a 5 channel setup as well? |
| 09:31.31 | irroot | the ones i get are < 2000R +/- 200 euro and are usb only |
| 09:32.42 | Haraken | i see |
| 09:33.19 | *** join/#asterisk binbash_ (~peter@a83-161-240-87.adsl.xs4all.nl) |
| 09:34.28 | *** join/#asterisk ironm (~ironm@fwj00.e-fon.ch) |
| 09:40.03 | irroot | mandla you come right ?? |
| 09:40.29 | *** join/#asterisk caveat- (hoax@newshell1.bshellz.net) |
| 09:40.44 | hetii | 2011/06/09 11:40:15.258ttyT38-0(i...0xed712700--> read ERROR -1 Input/output error |
| 09:40.49 | hetii | 2011/06/09 11:40:15.268ttyT38-0(b...0xf652e700PseudoModemPty::OpenPty removed link /dev/ttyT38-0 -> /dev/pts/2 |
| 09:40.49 | hetii | 2011/06/09 11:40:15.268ttyT38-0(b...0xf652e700PseudoModemPty::OpenPty read ERROR -1 Input/output error |
| 09:40.58 | hetii | any clue what can be wrong ? |
| 09:41.27 | hetii | this was from log when i restart hylafax or try send fax |
| 09:42.55 | mandla | fxload -t fx2 -D /dev/bus/usb/MMM/NNN -I /usr/share/dahdi/USB_FW.hex, is this how firmware are installed on AsteriskNow? |
| 09:45.48 | irroot | heti im using h323plus not opal sorry cant relate to pastebin in v2 there is psudo modem support im using it need to -p +/dev/ttyF0 to create the dev |
| 09:55.07 | hetii | hmm |
| 10:02.22 | *** join/#asterisk sigius (~sigius@93-125-185-45.dsl.alice.nl) |
| 10:05.14 | sigius | I am having a weird issue with my agi script (php). The very first $agi->get_variable I do returns a code 510 (invalid or unknown command) regardless. Subsequent get_variables work ok. If Imove my get_variable's around again the same thing: the first one returns a code 510, the others work ok |
| 10:05.27 | sigius | any ideas anyone? |
| 10:05.40 | *** join/#asterisk coppice (~chatzilla@m121-202-89-162.smartone-vodafone.com) |
| 10:05.57 | WIMPy | That framework you're using is broken? |
| 10:06.43 | Haraken | hetii, look like a drive error |
| 10:07.05 | Haraken | hetii, do you see similar input/output errors in /var/log/messages? |
| 10:08.58 | hetii | this is what i got when i restart hylafax from - http://pastebin.ca/2077234 |
| 10:09.23 | hetii | message file, dmesg, syslog don`t show any errors about it |
| 10:10.03 | Haraken | smartctl -a -son -data /dev/sda |
| 10:10.09 | Haraken | replace /dev/sda with your drive |
| 10:10.18 | Haraken | paste output to pastebin, just want to be sure |
| 10:10.49 | Wiretap | obviously qualify on TCP peers is fixed in 1.8.4 |
| 10:10.50 | Wiretap | thank god |
| 10:15.10 | *** join/#asterisk ihor (~Miranda@194.44.15.90) |
| 10:19.44 | *** join/#asterisk schmidts (~schmidts@Lo20-rt01.lm33.sil.at) |
| 10:19.46 | schmidts | hello |
| 10:20.41 | *** join/#asterisk sekil (~sekil@80.93.247.26) |
| 10:25.30 | *** join/#asterisk orn (~orn@rtr1.sh23.sip.is) |
| 10:32.49 | hetii | hmm |
| 10:33.20 | hetii | maybe the reson is that my t38modem command line arguments have no information about registerig |
| 10:33.22 | hetii | so ... |
| 10:35.58 | sigius | sigius: answering my own question (to some extent anyway), the problem was gone after I removed some whitespace after the closing bracket ?> in a agi script of mine |
| 10:36.38 | *** join/#asterisk naif (~naif@host227-105-static.46-85-b.business.telecomitalia.it) |
| 10:36.40 | sigius | no idea how thats works but i guess one should never use a closing bracket in a php file |
| 10:36.41 | naif | Hi |
| 10:37.00 | naif | ANY ITALIAN there? I am looking in Milan for full time employment of an Asterisk Knolwedgable person in Italy/Milan |
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| 10:47.03 | hetii | ok i need some person who successfully set up hylafax_t38modem + asterisk |
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| 11:24.49 | hetii | hmm Got SIP response 502 "Bad Gateway" back from 127.0.0.1:6060 |
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| 11:39.28 | mandla | tzafrir_laptop, in the Astribank are phone lines from the PSTN connected in the BRI ports |
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| 11:44.31 | tzafrir_laptop | mandla, yes. Those should be TE BRI ports |
| 11:45.11 | mandla | Ok. Thanx |
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| 12:04.32 | hetii | Q: on each extensinon user i have argument port, for what exactly it is? |
| 12:04.46 | hetii | listening for * or endpoint ? |
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| 12:10.57 | irroot | hi heti still at it ?? |
| 12:12.32 | hetii | yes |
| 12:12.41 | hetii | its so crazy ... |
| 12:13.32 | hetii | first of all on google i don`t see that anybody use register string, i don`t suppose that * allow without registerig |
| 12:14.01 | WIMPy | Beware! It might be contageous. ;-) |
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| 12:16.47 | hetii | but when i set register string i got just |
| 12:16.51 | hetii | REGISTER transaction id=z9hG4bK8e2bdff3-ff90-e011-8a77-001a4d70ee47 timeout, making retry 5, timeout 4.000 |
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| 12:37.22 | NourSs | HEllo |
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| 12:38.08 | mandla | One last thing, is there an installable X-lite for Ubuntu 11.04? |
| 12:38.24 | *** part/#asterisk ihor (~Miranda@194.44.15.90) |
| 12:38.26 | cneb3000 | todays the day I learn to code in Perl. |
| 12:38.30 | cneb3000 | stands tall |
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| 12:39.11 | cneb3000 | mandla: I dont think there is... is that something you could try running in Wine? |
| 12:39.29 | wdoekes2 | cneb3000/mandla: I've ran x-lite on ubuntu 10.04 |
| 12:39.43 | cneb3000 | ^^^^ apparently there is.. http://forums.counterpath.com/viewtopic.php?f=6&t=16915 |
| 12:40.35 | NourSs | Anyone know a site specialized for sell a Digium Card ? |
| 12:40.40 | wdoekes2 | although last time I ran it, I had to REJECT some (xtensoft) IP because it would timeout on that |
| 12:40.57 | NourSs | I have a TE122 and a TDM411 for sell |
| 12:41.22 | cneb3000 | does the Digium forum have a bit to sell stuff on? |
| 12:41.22 | WIMPy | NourSs: Which country? |
| 12:41.45 | NourSs | France, but can ship all country ;) |
| 12:41.46 | WIMPy | NourSs: Oh, that way. Well, i think ebay has some sort of monopoly there. |
| 12:42.18 | NourSs | Ok.. already on ebay .. |
| 12:42.22 | *** join/#asterisk mintos (~mvaliyav@nat/redhat/x-emgjzdclrojkpawz) |
| 12:42.28 | WIMPy | NourSs: List it on ebay.com. The US guys pay a lot more for them :-) |
| 12:43.19 | NourSs | For example : http://cgi.ebay.fr/ws/eBayISAPI.dll?ViewItem&item=110698890013&ssPageName=ADME:L:LCA:FR:1123 |
| 12:43.53 | NourSs | Not expensive ;-) |
| 12:44.25 | WIMPy | See above. |
| 12:44.58 | WIMPy | IThat price is unlikely to get a bid in the EU. |
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| 12:45.27 | NourSs | You kknow ? |
| 12:45.37 | anonymouz666 | seanbright: hello |
| 12:46.27 | WIMPy | I'm watching that stuff from time to time and there's a huuuuge price difference between US and EU. |
| 12:47.12 | NourSs | In US i can sell it for ? |
| 12:47.24 | mandla | wdoekes2, how did you get it to work, how did you install it? In Wine it doesnt work at all. |
| 12:47.46 | WIMPy | NourSs: Use the search function. |
| 12:48.58 | WIMPy | mandla: Don't use wine. I don't know about packages, but there's definitely a linux version. |
| 12:49.59 | mandla | WIMPy, ok il look it up. |
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| 13:01.34 | infernix | My asterisk setup is replying with a SIP/2.0 401 Unauthorized only on incoming calls (INVITE) from my voip provider, outgoing is working fine. the incoming call doesn't even seem to hit the dialplan. what could be wrong? |
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| 13:03.12 | infernix | sip show registry shows that it's registered just fine, too |
| 13:03.31 | merlin8282 | Hello here ! |
| 13:04.09 | merlin8282 | I've a problem: I don't know how to detect if a call is answered by an automatic responder. |
| 13:04.17 | cneb3000 | infernix: have a look at this and see if it helps? --> http://www.voipuser.org/forum_topic_17788.html |
| 13:04.54 | *** join/#asterisk ^-^ (~atan@unaffiliated/atan) |
| 13:04.56 | WIMPy | infernix: They probably don't authenticate towards you, so you have to configure them as peers with their IP. |
| 13:05.00 | merlin8282 | Imagine: an important support call comes in. Asterisk is configured to let ring the first support mobile phone 12 seconds, then if no answer it rings the second mobile. |
| 13:05.12 | ^-^ | I am slightly confused why ReceiveFAX() isn't making the file it should be in /var/spool/asterisk/fax |
| 13:05.18 | ^-^ | Actually, it doesn't make any file. |
| 13:05.36 | merlin8282 | But what if the first mobile phone is turned off ? An automatic responder answers and Asterisk is considering it's OK. :/ |
| 13:05.44 | WIMPy | merlin8282: What about the option to accept the call via DTMF keypress? |
| 13:05.45 | ^-^ | Console shows it should make a .tif in there but there's nothing showing up. It's owned by asterisk:asterisk and set to 755. |
| 13:06.17 | merlin8282 | WIMPy: ah ! Good idea ! I'll search this way |
| 13:07.43 | *** join/#asterisk bl0b` (~bl0b@host27-89-dynamic.1-87-r.retail.telecomitalia.it) |
| 13:08.08 | infernix | wth, uncommenting the secret fixes it, but that will break outgoing calls. |
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| 13:08.12 | *** mode/#asterisk [+o Deeewayne] by ChanServ |
| 13:08.53 | WIMPy | infernix: You need two entries for in and out. |
| 13:09.28 | infernix | since when? this worked for like 2 years |
| 13:09.45 | anonymouz666 | I never needed two entries for in and out. |
| 13:09.46 | WIMPy | And what changed? |
| 13:10.03 | WIMPy | Maybe a insecure=invite will do? |
| 13:10.07 | bl0b` | Hi |
| 13:10.12 | infernix | i installed i6nets' VXI module |
| 13:10.12 | *** part/#asterisk NourSs (~gholzinge@LAubervilliers-151-13-22-64.w217-128.abo.wanadoo.fr) |
| 13:10.18 | infernix | talk about weird correlation |
| 13:10.21 | ^-^ | looks at libspandsp-dev |
| 13:10.24 | ^-^ | Think I got it, thanks anyway |
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| 13:10.35 | bl0b` | Someone know how to set the dahdi for an ISDN PRI to use the 3.1khz |
| 13:10.40 | bl0b` | codec ? |
| 13:11.17 | WIMPy | bl0b`: It always does. It's the only it supports. |
| 13:11.30 | bl0b` | it request the speech |
| 13:11.36 | bl0b` | istead of 3.1 |
| 13:11.39 | infernix | WIMPy: that looks better, changed insecure=very to invite and kept the secret. thanks |
| 13:12.15 | WIMPy | bl0b`: That was the wrong question then. The codec is the same. You want it to request another bearer capability. |
| 13:12.19 | Katty | guten morgan |
| 13:12.24 | bl0b` | WIMPy yes sorry |
| 13:13.05 | *** join/#asterisk m_tadeu (~quassel@89.181.66.231) |
| 13:13.13 | WIMPy | bl0b`: I was on that topic not long ago. I found an App SetTransferCapability (IIRC), which doesn;t seem to exist any more. |
| 13:13.22 | Katty | distributes hugs and muffins. |
| 13:13.32 | WIMPy | bl0b`: So I have no idea, how to do it with dahdi. |
| 13:13.46 | WIMPy | infernix: very is deprecated. Use port,invite. |
| 13:14.10 | bl0b` | WIMPy my * is 1.4 so prolly it have it? |
| 13:14.41 | WIMPy | bl0b`: I guess you might be lucky. |
| 13:15.17 | bl0b` | WIMPy tnx |
| 13:15.22 | WIMPy | And I'd still like to know how it's done on a current version, BTW. |
| 13:17.32 | malcolmd | the CHANNEL function has the transfercapability setting: https://wiki.asterisk.org/wiki/display/AST/Function_CHANNEL |
| 13:18.15 | WIMPy | Ah, thank you malcolmd. |
| 13:18.21 | malcolmd | np :) |
| 13:19.13 | bl0b` | malcolmd but theris a way to support both ? for incoming calls? |
| 13:19.41 | WIMPy | bl0b`: For incomming calls you shouldn't have to care. |
| 13:20.20 | bl0b` | WIMPy I should, due that some ppl when call me hear the telco message for unexistent number. |
| 13:20.23 | *** part/#asterisk LemensTS (~matthew@adsl-70-238-163-82.dsl.stlsmo.sbcglobal.net) |
| 13:20.32 | WIMPy | That's the standard problem when ppl try to send faxes. |
| 13:21.15 | WIMPy | bl0b`: That must be something else. Can you reproduce that? |
| 13:21.37 | bl0b` | WIMPy not now, but yes |
| 13:22.17 | WIMPy | Ok, if you can't find any hint when looking at the *CLI with verbose and debug turned up, try a pri debug. |
| 13:22.43 | bl0b` | okz |
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| 13:26.37 | WIMPy | Katty: It's Guten MorgEn |
| 13:26.49 | bl0b` | ok seems that the telco didn told the right information to me |
| 13:27.01 | Katty | oh? |
| 13:27.03 | Katty | good to know |
| 13:27.08 | Katty | updates enteries |
| 13:27.16 | bl0b` | -- Requested transfer capability: 0x10 - 3K1AUDIO .. but still reject the call |
| 13:28.20 | WIMPy | bl0b`: Is there a matching extension? If not, Asterisk would signal a number unassigned. |
| 13:29.00 | bl0b` | WIMPy yes, for incoming... due that if other ppl call me, it work, only some callers can't reach me |
| 13:29.32 | WIMPy | bl0b`: On the same number? |
| 13:29.41 | bl0b` | WIMPy yes |
| 13:30.02 | bl0b` | but for outgoing I didnt undertand, now I'v forced the 3.1khz and still did not work |
| 13:30.03 | WIMPy | Sounds fishy. But you can see both in pri debug? |
| 13:30.19 | bl0b` | for the outgoing I'v the log now on the screen |
| 13:30.58 | WIMPy | bl0b`: It shouldn't matter much which BC you use for phone calls. |
| 13:31.05 | WIMPy | What cause code do you get? |
| 13:32.09 | bl0b` | < Message type: DISCONNECT (69) |
| 13:32.25 | bl0b` | < Ext: 1 Cause: Normal, unspecified (31), |
| 13:32.46 | WIMPy | Did I mention today, that I hate it displaying them decimal? |
| 13:33.15 | WIMPy | 0x45, "Requested facility not implemented" |
| 13:33.37 | WIMPy | You seem to send some information, you shouldn't. |
| 13:33.58 | WIMPy | Maybe a redirectiong number? I think I've ssen that before. |
| 13:34.29 | WIMPy | Oh. Read the wrong bit. Doh. |
| 13:35.13 | WIMPy | What's the location? |
| 13:35.47 | bl0b` | wait a mom... I'm calling an unexistent number ! doh |
| 13:36.27 | WIMPy | Hmm. It should tell you that. |
| 13:38.03 | bl0b` | ok I'v double checked |
| 13:38.05 | bl0b` | the number is correct |
| 13:38.18 | bl0b` | but prolly they have some problem with their phone line |
| 13:39.03 | WIMPy | You should see that as well. |
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| 13:49.31 | bl0b` | ok fixed it |
| 13:49.42 | bl0b` | The outgoing now work with the settransfercapability |
| 13:49.52 | bl0b` | and the incoming problem was the overlapdial |
| 13:55.24 | bl0b` | WIMPy thanks for the help :) |
| 13:56.01 | WIMPy | bl0b`: That sounds like a misconfuguration athe the other end. |
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| 13:57.38 | bl0b` | WIMPy Telecom Italia really sucks |
| 13:57.45 | bl0b` | they do the things how they like |
| 13:57.53 | WIMPy | No, the called party. |
| 13:58.06 | bl0b` | the called party is a rtg |
| 13:58.16 | bl0b` | didnt have too much to configure |
| 13:58.36 | *** join/#asterisk jsjc (~Adium@226.Red-80-33-236.staticIP.rima-tde.net) |
| 13:58.36 | sxpert | bl0b`: well, this being italy, do you expect anything to work as documented ? |
| 13:59.08 | WIMPy | What's rtg? |
| 13:59.18 | jsjc | I have a question I have a few SIP trunks that allow me a 1 ree simultaneous call as soon as I start 2nd call I need to pay |
| 13:59.37 | jsjc | is it possible for me in dialplan to check if that peer is being used? |
| 13:59.37 | bl0b` | sxpert there no |
| 14:00.19 | bl0b` | sxpert but with bt for example the things work smoth, with telecom I get always a lot of trouble |
| 14:00.56 | WIMPy | jsjc: The Group function might be your friend. |
| 14:01.16 | sxpert | bl0b`: guess there's a pattern there ;) |
| 14:01.24 | jsjc | group. will go an have a look to it. |
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| 14:16.27 | bl0b` | WIMPy maybe call-limit parameter in sip.conf can't help him? |
| 14:16.45 | *** join/#asterisk sekil (~sekil@80.93.247.26) |
| 14:16.51 | WIMPy | Yes, but that's deprecated. |
| 14:17.12 | bl0b` | ah ok |
| 14:18.20 | WIMPy | But I'm not sure it works for outgoing. |
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| 14:40.45 | irroot | evening peeps |
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| 14:49.48 | mick_laptop | anyone know of a way to setup just an autoattendant w/ a "press 1 for X and 2 for Y" and have it ring one number if 1 is pressed and another for 2 |
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| 14:51.57 | bchia | mick_laptop there is a free video on this page that describes basic IVR set up including dialing a phone when DTMF is sent http://www.digium.com/en/training/courses/#essentials |
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| 15:00.11 | bl0b` | byebye |
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| 15:04.27 | WIMPy | Now I didn't find out what (a?) rtg is. |
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| 15:18.53 | cneb3000 | relax everyone. I wrote my first scripts in Perl. |
| 15:18.55 | cneb3000 | we'll be OK now. |
| 15:19.32 | WIMPy | So you learned a lot about your keyboard mappings? :-) |
| 15:19.44 | cneb3000 | >_> |
| 15:19.46 | cneb3000 | wut? |
| 15:20.29 | cneb3000 | ...google prevails! that seems useful. |
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| 15:38.59 | asterisk0001 | Hello #Asterisk |
| 15:39.17 | asterisk0001 | We have strange issue with MWI |
| 15:39.25 | WIMPy | Hello <asterisk0001> |
| 15:39.34 | asterisk0001 | Somehow asterisk 1.8 not sending MWI notification to polycom501 |
| 15:39.48 | irroot | Punjabi curry 4 dinner if it dont make a grown man cry it aint .... ideal for winter |
| 15:39.56 | asterisk0001 | I ran tcpdump and sip debug |
| 15:40.43 | asterisk0001 | I do have mailbox=xxx@defaults |
| 15:41.01 | asterisk0001 | But if i restart asterisk my asterisk will send NOTIFY to ll phones |
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| 15:43.20 | WIMPy | irroot: Hah. Nothing beats a good salad with that new EHEC spice. |
| 15:45.48 | *** join/#asterisk methodvon (~methodvon@static-70-109-53-76.clppva.fios.verizon.net) |
| 15:47.35 | irroot | ?? WIMPy be kind rewind and explain |
| 15:49.56 | WIMPy | Well until recently, real men didn't eat salad. Now it's the other way round. |
| 15:50.12 | cneb3000 | what changed in the world? |
| 15:50.22 | Qwell | WIMPy: not eating salad is in again |
| 15:50.27 | WIMPy | EHEC |
| 15:51.00 | irroot | ah E-Coli ROFLMAO |
| 15:52.08 | methodvon | Hello. I'm looking to implement a phone verification system using Asterisk. I am looking for a friendly nudge in the right direction. I've spent time on Google, but coming up short. Looking for third-party products or assistance, as creating from scratch will take me to long. Familiar with Asterisk, but new to this field. |
| 15:52.29 | Qwell | verification of what? |
| 15:52.37 | Qwell | That sounds very simple. |
| 15:53.02 | cneb3000 | you mean like going through numbers sequentially to check if they're active? |
| 15:53.26 | methodvon | Qwell: upon signing up for our service, user receives a call from our PBX, hears a verification code, user then enters code on site |
| 15:53.45 | Qwell | Sounds like about 5 minutes worth of work. O.o |
| 15:54.13 | methodvon | basically similar to what Google Voice does when verifying additional numbers |
| 15:54.44 | Qwell | Generate a number, store it in a database, call somebody, use SayDigits, have the website use the same database. |
| 15:54.58 | irroot | methodvon did it with SMS sent from func_curl :) |
| 15:55.47 | irroot | methodvon if you can script it in perl/python/php you can use agi |
| 15:57.20 | asterisk0001 | Hey Guys any idea about MWI ? |
| 15:59.41 | methodvon | Qwell: cool. sounds like its possible with some elbow grease. thx |
| 16:00.09 | methodvon | irroot: I read about Adhearsion but I'm not familiar enough with Ruby. Thx for the input. |
| 16:00.18 | tzafrir_laptop | Off-topic: Trying to help someone with a E72 phone connect to our PBX: |
| 16:00.50 | tzafrir_laptop | how can I make the device unregister from an account temporarily to access some extra settings? |
| 16:03.28 | *** join/#asterisk mmontgomery (4a29a932@gateway/web/freenode/ip.74.41.169.50) |
| 16:04.12 | mmontgomery | I am seeking help using a NEC DT700 with asterisk, I cannot find anything about this on the web. |
| 16:10.22 | adyn | I need up upgrade from 1.2.17 to 1.2.40 I looked over the changelogs and don't see anything drastic that could break my system. Any issues other than that's an old old version? I don't have the option to move to later versions atm. |
| 16:10.24 | *** join/#asterisk afink (~afink@204.26.87.226) |
| 16:17.27 | mmontgomery | no clue |
| 16:17.52 | mmontgomery | any ideas on how to get a nec dt700 working with asterisk? |
| 16:27.23 | *** join/#asterisk gerhard7 (~gerhard7@82-171-103-215.ip.telfort.nl) |
| 16:27.46 | mmontgomery | any ideas on how to get a nec dt700 working with asterisk? |
| 16:31.51 | *** join/#asterisk citywok (~citywok@67-134-194-33.dia.static.qwest.net) |
| 16:32.04 | *** join/#asterisk BarthezZ (~bart@ip51cd391a.speed.planet.nl) |
| 16:33.13 | *** join/#asterisk scud_ (~scud@192.151.88.132) |
| 16:35.31 | *** part/#asterisk scud_ (~scud@192.151.88.132) |
| 16:36.32 | *** join/#asterisk fhmiv (~fhmiv@c-67-173-205-151.hsd1.ga.comcast.net) |
| 16:37.09 | *** join/#asterisk agentace (~agentace@72-28-175-001-dhcp.aik.sc.atlanticbb.net) |
| 16:41.27 | mmontgomery | any ideas on how to get a nec dt700 working with asterisk? |
| 16:41.52 | agentace | Greetings! I'm curious whether anyone here has any experience using asterisk as a standalone voicemail server for another PBX (Samsung DCS) which currently access its voicemail server over analog lines. |
| 16:42.31 | anonymouz666 | anyone in here is aware of any patch that support version 1.4. to block incoming collect calls in ISDN-PRI ? |
| 16:46.19 | *** join/#asterisk jacobkiers (~jacobkier@82-168-215-74.ip.telfort.nl) |
| 16:47.11 | *** join/#asterisk jsjc (~Adium@209.254.20.95.dynamic.jazztel.es) |
| 16:47.22 | *** join/#asterisk leroybuckingham (637c543e@gateway/web/freenode/ip.99.124.84.62) |
| 16:50.00 | *** join/#asterisk dushadow (~dushadow@12.54.128.226) |
| 16:51.53 | mmontgomery | any ideas on how to get a nec dt700 working with asterisk? |
| 16:52.11 | *** join/#asterisk pdtpatrick__ (~pdtpatric@mainstwan.farheap.com) |
| 16:54.19 | pdtpatrick__ | Good morning/Afternoon. Can someone please help me w/ this |
| 16:54.20 | pdtpatrick__ | http://pastebin.com/mnpwfyN0 |
| 16:57.07 | *** join/#asterisk nimbiotics (~chatzilla@adsl-74-225-240-252.mia.bellsouth.net) |
| 16:59.14 | nimbiotics | hello everyone. Im not an asterisk user but I dont know where else to go with my issue. using ubuntu 11.04, I'm trying to setup twinkle so I can make and receive calls with my Google Voice account, but I really dont know where to start. Please help me. TIA! |
| 17:12.41 | Nugget | On that same subject, can anyone help me with information on how to bleed the brake fluid on my 2007 Porsche? |
| 17:14.21 | Deeewayne | Nugget, pump the brakes after you take them off the rotor |
| 17:18.34 | *** join/#asterisk irroot (~irroot@41.55.33.102) |
| 17:19.46 | nimbiotics | c'mon guys, I can only guess most people here know about voip and the such ... guess i'm wrong ... this is the clowns channel |
| 17:22.32 | Qwell | ~asteriskwiki |
| 17:22.32 | infobot | methinks asteriskwiki is http://wiki.asterisk.org |
| 17:22.32 | *** join/#asterisk sereal (~sereal@2001:4830:116e:1:1e6f:65ff:fec3:cd0d) |
| 17:22.43 | sereal | anyone have any experience configuring polycom phones? |
| 17:22.48 | afink | yup |
| 17:23.06 | sereal | Whats considered a softkey? |
| 17:23.23 | afink | I think its a programmable key on the phone |
| 17:23.39 | sereal | is it a physical button on the phone, or is it a changable button that shows on the lcd? |
| 17:23.41 | dr0ck | the row of buttons below the display |
| 17:23.43 | sereal | or are they the same? |
| 17:23.53 | dr0ck | they correspond, yes |
| 17:23.58 | ^-^ | slaps nimbiotics around a bit with a large trout |
| 17:24.24 | Nugget | I don't even know what "twinkle |
| 17:24.26 | sereal | ah okay. specifically i'm trying to map the messages key to dial the correct extension to a asterisk pbx |
| 17:24.26 | Nugget | is. |
| 17:24.38 | Nugget | but go ahead and try insulting the channel, that'll surely enlist our support. |
| 17:24.45 | sereal | I wasn't sure if hardbuttons were specifically defined else where. |
| 17:24.52 | nimbiotics | Nugget, twinkle is a softphone for linux |
| 17:25.43 | Katty | telnet |
| 17:25.43 | Nugget | telnet is eeeeeeevil! |
| 17:25.45 | Katty | I WIN |
| 17:25.49 | Nugget | silly katty |
| 17:25.51 | Katty | hugs Nugget |
| 17:26.01 | Katty | Nugget: thanks for the comment |
| 17:26.12 | Nugget | I just made the comment because you're beautiful. |
| 17:26.24 | Katty | :/ |
| 17:26.26 | Katty | did you really? |
| 17:26.32 | Nugget | huggles Katty |
| 17:27.52 | *** join/#asterisk TimeRider (~steve@188-220-33-123.zone11.bethere.co.uk) |
| 17:28.19 | dushadow | my asterisk system will stop working. In the logs it shows app_dial.c: Unable to create channel of type 'SIP' (cause 20 - Unknown) |
| 17:28.23 | dushadow | any ideas? |
| 17:29.16 | Katty | i've seen that before |
| 17:29.44 | dushadow | when that error pops up, non of the phones work. |
| 17:29.58 | dushadow | katty: do you know the cause and fix? |
| 17:30.01 | Katty | i found it's usually becasue the server can't reach the phones |
| 17:30.38 | Katty | sometimes networking issues |
| 17:30.39 | Katty | routing issues |
| 17:30.44 | Nugget | yeah, or trying to route a call to a SIP peer that's not registered |
| 17:31.14 | Katty | sip show peers might be useful |
| 17:31.23 | Katty | if that's still the syntax these days |
| 17:31.52 | dushadow | thanks. I'll check those out. it may take be a bit to figure where everything is locate. |
| 17:31.57 | dushadow | kinda new to asterisk |
| 17:32.33 | *** join/#asterisk coppice (~chatzilla@210.17.219.183) |
| 17:33.33 | *** join/#asterisk hobodave (~hobodave@pdpc/supporter/professional/hobodave) |
| 17:34.59 | *** join/#asterisk leifmadsen (~Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 17:34.59 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 17:35.11 | Katty | hi leif |
| 17:35.22 | leifmadsen | waves |
| 17:35.24 | leifmadsen | huh, can you not use G722 through a Local channel? |
| 17:35.28 | Katty | leif, tell me something |
| 17:35.35 | leifmadsen | I get [Jun 9 10:34:23] NOTICE[15758]: channel.c:4128 __ast_read: Dropping incompatible voice frame on Local/205@QueuedReception-1591;2 of format g722 since our native format has changed to 0x4 (ulaw) |
| 17:35.37 | Katty | since you're all pretty and attractive and stuff |
| 17:35.44 | leifmadsen | o.O |
| 17:35.44 | irroot | for the record customers that check sql queries from cdr's / queue_logs with calculators suck but usefull at pointing out errors |
| 17:35.47 | Katty | how do you manage to not be a complete, shallow, rude, dooface |
| 17:35.50 | Katty | duface |
| 17:35.50 | leifmadsen | checks to make sure he is in the right channel :) |
| 17:35.52 | Katty | however ou spell it |
| 17:36.05 | Katty | jerkface, there we go |
| 17:36.08 | leifmadsen | Katty: who says I'm not? :) |
| 17:36.14 | Katty | you don't seem like one. |
| 17:36.18 | Katty | are you one? |
| 17:36.26 | leifmadsen | heh, I'm not sure... guess I grew up ugly :) |
| 17:36.36 | Katty | that's your secret? |
| 17:36.38 | irroot | leifmadsen there is some nasties with local and audio formats |
| 17:36.42 | Katty | really? |
| 17:36.47 | leifmadsen | irroot: ya... looks like I can't use g722 with Local channels :( |
| 17:36.48 | irroot | is the g722 a result of a reinvite somewhere ?? |
| 17:36.55 | leifmadsen | nope |
| 17:37.06 | leifmadsen | here I'll show you the one line |
| 17:37.15 | coppice | I invite everyone to use G.722 |
| 17:37.27 | leifmadsen | so I use this: same => n,Dial(Local/206@QueuedReception&Local/205@QueuedReception&Local/203@QueuedReception,45) |
| 17:37.37 | leifmadsen | oh wait, idea |
| 17:37.41 | irroot | mmm there is a patch i put in to change the format when the proxied chan changes |
| 17:37.45 | leifmadsen | nevermind |
| 17:37.58 | leifmadsen | ok so all end points have allow=g722 as the first codec |
| 17:37.59 | leifmadsen | ulaw 2nd |
| 17:38.19 | leifmadsen | all the dialplan does in the Local channel is Dial(SIP/0004f2040001) |
| 17:38.30 | leifmadsen | that's for 206 |
| 17:38.46 | leifmadsen | 205 and 203 have Wait(18) and Wait(24) respectively, followed by a Dial() like above |
| 17:38.55 | irroot | leifmadsen https://reviewboard.asterisk.org/r/1137/ |
| 17:38.55 | leifmadsen | that's the whole dialplan, and I get that NOTICE flying on the screen |
| 17:39.13 | leifmadsen | irroot: ok, let me try patching and see if it helps any |
| 17:40.38 | irroot | if it applies |
| 17:40.51 | irroot | it may not be in your case |
| 17:41.13 | leifmadsen | right |
| 17:41.16 | leifmadsen | easy to test though |
| 17:42.07 | irroot | what this does is try force local1/2 to be what the original channel is |
| 17:42.15 | irroot | and if it changes it changes with |
| 17:42.20 | leifmadsen | bah! failed patch :) |
| 17:42.24 | leifmadsen | looks at failed hunk |
| 17:42.32 | irroot | yeah indeed thought it might |
| 17:42.53 | irroot | hold on ill post new one |
| 17:42.56 | leifmadsen | awesome thx! |
| 17:43.07 | irroot | this is trunk i can give you 1.8 ?? |
| 17:43.54 | irroot | there is a difference |
| 17:44.04 | leifmadsen | this is 1.8 yes |
| 17:44.09 | leifmadsen | latest branch (since yesterday) |
| 17:44.13 | leifmadsen | brb, gotta go activate call forwarding |
| 17:45.17 | *** join/#asterisk saxa (~sasa@189.26.255.43) |
| 17:46.51 | *** join/#asterisk saxa (~sasa@189.26.255.43) |
| 17:55.46 | *** join/#asterisk actionboy (~actionboy@38.103.96.75) |
| 17:56.54 | actionboy | I have a digium T220P connected to my T1 phone lines, on the connection (in /proc/dahdi/2) the error counts for CRC4 and e-bit are slowly increasing, is this indicative of a problem, or is some amount of errors expected? phones appear to be working normally |
| 18:06.40 | *** part/#asterisk nimbiotics (~chatzilla@adsl-74-225-240-252.mia.bellsouth.net) |
| 18:11.55 | *** join/#asterisk binbash_ (~peter@a83-161-240-87.adsl.xs4all.nl) |
| 18:12.06 | *** join/#asterisk brainiac (~mhauss@c-68-52-249-113.hsd1.tn.comcast.net) |
| 18:17.22 | asterisk0001 | Hey Gusy! |
| 18:17.33 | asterisk0001 | any idea about delayed MWI notification ? |
| 18:18.14 | coppice | actionboy: CRC4 is an E1 only thing |
| 18:26.55 | actionboy | coppice, I suppose I could have an E1 connection, I'm honestly not sure what it is, how might I determine that? (I didn't set this thing up, I just inherited it) |
| 18:33.34 | *** join/#asterisk hetii (~hetii@87.99.51.172) |
| 18:33.40 | hetii | Hello :) |
| 18:34.02 | hetii | me again :) |
| 18:36.25 | *** join/#asterisk silkcut (silkcut@dan75-6-82-239-15-163.fbx.proxad.net) |
| 18:38.01 | *** join/#asterisk binbash_ (~peter@a83-161-240-87.adsl.xs4all.nl) |
| 18:38.49 | Katty | i need a beer too |
| 18:38.54 | Katty | who wants to go get some beers? |
| 18:39.50 | _Corey_ | I'm in... it's too hot to work |
| 18:40.02 | hetii | i know this feelings |
| 18:40.05 | *** join/#asterisk binbash_ (~peter@a83-161-240-87.adsl.xs4all.nl) |
| 18:40.29 | snoow | hey guys, is there a way I can map an extension to go to a specific outgoing route within asterisk / elastix? |
| 18:40.34 | hetii | who wants help me with t38modem :) ? |
| 18:40.56 | hetii | snoow: it is. check you outbounds routes |
| 18:42.24 | snoow | hey Hetii, I dont see anything that will allow me to map it to the actual extension though, in the outgoing routes properties |
| 18:42.41 | hetii | caller id you can use |
| 18:42.44 | *** join/#asterisk imox1234 (~imox1234@85.106.78.22) |
| 18:44.36 | *** join/#asterisk MarKsaitis (~MarKsaiti@cpc7-pool12-2-0-cust219.15-1.cable.virginmedia.com) |
| 18:46.43 | *** join/#asterisk binbash_ (~peter@a83-161-240-87.adsl.xs4all.nl) |
| 18:47.25 | *** join/#asterisk binbash_ (~peter@a83-161-240-87.adsl.xs4all.nl) |
| 18:47.57 | snoow | thanks hetii, at least now I know where to poke around until i get it behaving properly |
| 18:48.02 | *** join/#asterisk Dovid (~Dovid@213.8.121.90) |
| 18:48.05 | Qwell | snoow: In the future, try #freepbx |
| 18:48.17 | Qwell | Asterisk has no concept of things like "outgoing routes". |
| 18:48.25 | Qwell | Those are an invention of GUIs. |
| 18:48.55 | snoow | alright, my apologies, i saw the irc channel on the asterisk website |
| 18:49.12 | irroot | passes Katty a castle lager |
| 18:49.14 | Qwell | But you aren't using Asterisk. FreePBX is using Asterisk. |
| 18:49.29 | Katty | isn't sure what that is, but tries it regardless! |
| 18:52.08 | *** join/#asterisk mmontgomery (4a29a932@gateway/web/freenode/ip.74.41.169.50) |
| 18:52.40 | mmontgomery | Does anyone know how to setup dt700 NEC phones to work with asterisk? |
| 18:53.10 | Qwell | mmontgomery: Same as any other SIP phone. |
| 18:53.16 | Qwell | ~book |
| 18:53.16 | infobot | For more information about the Asterisk book, see ~thebook |
| 18:53.27 | Qwell | infobot: You stupid bot... |
| 18:53.28 | Qwell | ~thebook |
| 18:53.29 | infobot | Asterisk: The Definitive Guide, 3rd Edition (ISBN 0-596-51734-3) available at http://oreilly.com/catalog/9780596517342. Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and is available for reading online at http://ofps.oreilly.com/titles/9780596517342/, or ~buybook |
| 18:57.58 | *** join/#asterisk espent (~espent@46.80-203-219.nextgentel.com) |
| 18:58.29 | JerJer | gooie |
| 18:58.44 | irroot | naand jerjer |
| 18:58.59 | hetii | please check that: http://pastebin.com/WfRLQJSX |
| 19:01.15 | *** join/#asterisk oliver1 (~oliver@95.116.190.2) |
| 19:05.29 | *** join/#asterisk binbash_ (~peter@a83-161-240-87.adsl.xs4all.nl) |
| 19:06.33 | mmontgomery | Does anyone know how to setup dt700 NEC phones to work with asterisk? |
| 19:07.30 | *** join/#asterisk binbash_ (~peter@a83-161-240-87.adsl.xs4all.nl) |
| 19:07.34 | p3nguin | (1353.10) <@Qwell> mmontgomery: Same as any other SIP phone. |
| 19:08.30 | anonymouz666 | seanbright: do you remember if there is a patch to make 1.4 block incoming collects calls in ISDN-PRI. |
| 19:08.37 | anonymouz666 | s/./? |
| 19:08.42 | *** join/#asterisk NightMonkey (~NightMonk@pdpc/supporter/professional/nightmonkey) |
| 19:08.55 | citywok | anonymouz666: don't forget the trailing / |
| 19:09.09 | anonymouz666 | I saw that you did some patches to this feature |
| 19:09.38 | anonymouz666 | oh yeah |
| 19:12.10 | *** join/#asterisk hobodave_ (~hobodave@pdpc/supporter/professional/hobodave) |
| 19:13.43 | *** join/#asterisk cVsup (~cVsup@189.107.142.245) |
| 19:14.46 | seanbright | anonymouz666: 1.4? i doubt it |
| 19:15.56 | seanbright | i added a small patch to libpri that detected reverse charge indication, but that was so long ago i don't even remember it |
| 19:17.46 | *** join/#asterisk grEvenX (~even@cC0FD00C3.dhcp.bluecom.no) |
| 19:18.29 | mmontgomery | dt700 allows for sip setting but isn't working with asterisk |
| 19:19.47 | Defraz | That is a good question how can you block incoming collect calls. |
| 19:20.04 | seanbright | my microphone must not be working |
| 19:21.01 | brainiac | I bought two copies of Asterisk: The Definitive Guide, 3rd Ed. |
| 19:21.48 | Qwell | brainiac: So you can read in stereo? |
| 19:22.39 | brainiac | no. I thank amazon... I wish I could though |
| 19:22.56 | *** join/#asterisk vinhdizzo (~vinh@dhcp-053179.ics.uci.edu) |
| 19:24.03 | hetii | its incredible... cu: open (/dev/ttyT38-0): Permission denied, cu: ttyT38-0: Line in use |
| 19:24.18 | *** part/#asterisk mick_laptop (~mick@clamwin/admin/mickhome) |
| 19:24.23 | hetii | and before i was able to sen AT command now not |
| 19:24.39 | hetii | and there is nothing that i change ... :/ |
| 19:24.46 | hetii | wrr #@!@# |
| 19:25.00 | hetii | even after restart i still have that. |
| 19:26.53 | anonymouz666 | seanbright: libpri itself alerts the chan_dahdi of that? (by setting some var?) |
| 19:27.09 | seanbright | anonymouz666: correct |
| 19:27.23 | seanbright | you can use CHANNEL(reversecharge) |
| 19:27.41 | seanbright | which will be a 1 if it is a collect call, if i remember correctly |
| 19:27.50 | seanbright | you have to have a recent version of asterisk though, it's not in 1.4 |
| 19:28.13 | anonymouz666 | i need to backport, you think that will be hard? |
| 19:28.26 | seanbright | probably |
| 19:28.50 | seanbright | i kid. i don't remember the extent of the asterisk changes. |
| 19:30.53 | *** join/#asterisk jkroon (~jkroon@dsl-242-10-219.telkomadsl.co.za) |
| 19:31.38 | anonymouz666 | seanbright: will search for the commit then |
| 19:32.02 | anonymouz666 | basically two files were modified q931.c and chan_dahdi.c |
| 19:42.37 | *** join/#asterisk tamiel (~tamiel@ip-28.net-81-220-88.toulouse.rev.numericable.fr) |
| 19:43.02 | asterisk0001 | hey guys! |
| 19:43.19 | asterisk0001 | anybody know about mwiexpiry=3600 |
| 19:43.24 | asterisk0001 | at sip.conf ? |
| 19:46.31 | *** join/#asterisk vfabi (~fabi@host-static-188-237-242-50.moldtelecom.md) |
| 19:47.43 | asterisk0001 | No one know about that |
| 19:47.53 | *** part/#asterisk asterisk0001 (~root@216.204.211.243) |
| 19:49.21 | *** join/#asterisk Bryanstein (~Bryanstei@shellium/admin/bryanstein) |
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| 20:06.04 | *** join/#asterisk asterisk0001 (~root@216.204.211.243) |
| 20:06.17 | asterisk0001 | http://pastebin.com/xV9GAhnW |
| 20:07.37 | asterisk0001 | how to change SIP subscription expiry ? |
| 20:07.52 | *** join/#asterisk Bryanstein (~Bryanstei@shellium/admin/bryanstein) |
| 20:09.43 | hetii | i remove t38modem at all and install all from last sources |
| 20:10.02 | hetii | if this will not help than i don`t know what to do :... :) |
| 20:10.22 | *** join/#asterisk justdave (~dave@unaffiliated/justdave) |
| 20:10.29 | irroot | hetii good luck |
| 20:10.44 | hetii | thx :> |
| 20:11.33 | irroot | asterisk0001 have you looked pollmailboxes / pollfreq options in voicemail.conf |
| 20:12.07 | irroot | hetii im using the h323plus build of v2 so cant help ya with opal |
| 20:12.12 | irroot | good night all |
| 20:12.52 | hetii | good night you too :) |
| 20:13.11 | hetii | btw the question is do i need opal at all ? when i will use it with * |
| 20:13.28 | irroot | hetii yes for SIP |
| 20:13.45 | irroot | you can use OPAL or H323Plus for H.323 |
| 20:13.58 | irroot | T38Modem is nothing by itself |
| 20:14.08 | irroot | it needs one or the other library |
| 20:14.12 | hetii | but sip for provider or just * ? |
| 20:14.34 | hetii | i mean endpoint provider :) |
| 20:14.47 | hetii | ok i don`t want to stop you if you go sleep :> |
| 20:15.25 | irroot | if the endpoint is sip yes it does either asterisk or something else ie gateway |
| 20:17.31 | *** join/#asterisk tzafrir_laptop (~tzafrir@bzq-218-155-146.cablep.bezeqint.net) |
| 20:17.41 | Kobaz | there goes the internets |
| 20:18.04 | *** join/#asterisk jaytee (~jforde051@unaffiliated/jaytee) |
| 20:21.05 | Kobaz | come back internets |
| 20:23.51 | jaytee | where did the internets go? |
| 20:24.01 | Kobaz | cablemodem was out for a bit |
| 20:24.07 | Kobaz | switched to 3g |
| 20:24.12 | Kobaz | now it's back |
| 20:25.06 | jaytee | I've been amusing myself with the Les Paul doodle on Google's page |
| 20:27.29 | _Corey_ | I was hving fun with that earlier while on a long conf. call |
| 20:31.19 | *** join/#asterisk puzzled (~patrick@83.163.53.136) |
| 20:35.02 | *** join/#asterisk WiretapWork_ (~Wiretap@unaffiliated/wiretap) |
| 20:38.54 | *** join/#asterisk TimeRider (~steve@188-220-33-123.zone11.bethere.co.uk) |
| 20:41.35 | Kobaz | heh yeah the guitar thing |
| 20:41.42 | Kobaz | i need to go get strings |
| 20:43.38 | *** join/#asterisk anthm (~anthm@freeswitch/developer/anthm) |
| 20:44.34 | *** join/#asterisk luckman212 (~do-not-re@rrcs-50-74-138-150.nyc.biz.rr.com) |
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| 21:25.35 | *** join/#asterisk war9407 (war@c-71-62-61-45.hsd1.va.comcast.net) |
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| 21:37.17 | *** join/#asterisk tendril (~aphid@n159-178-232-130.hsc.xlate.ufl.edu) |
| 21:38.17 | tendril | does anyone know how to tell if a bunch of "wrong timestamps" on an rtp stream from my voip provider the cause of choppy audio? |
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| 21:48.07 | *** mode/#asterisk [+o Corydon76-home] by ChanServ |
| 21:53.45 | Katty | grumpy mcgrumperson needs antigrumpity device. |
| 21:54.09 | Katty | a nice long sit at the river, mayhaps |
| 21:54.14 | Qwell | grumps at Katty |
| 21:54.27 | Katty | hugs Qwell |
| 21:54.47 | Katty | Qwell: would you like to join me for some de-grumping therapy? |
| 21:54.57 | Qwell | sure |
| 21:56.12 | *** part/#asterisk jacobkiers (~jacobkier@82-168-215-74.ip.telfort.nl) |
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| 22:01.15 | WiretapWork_ | is there a way to make asterisk send notifies before a reload? |
| 22:01.41 | WiretapWork_ | I have a bunch of Cisco phones and I want them to reregister on a reload (theyre TCP peers and for some reason their information is lost on a reload) |
| 22:14.58 | *** join/#asterisk n0tk (~n0tk@gateway/tor-sasl/n0tk) |
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| 22:39.39 | voxio | does anyone know whats going wrong here: |
| 22:39.41 | voxio | http://pastie.org/private/9wdqbaafypl5rckxadihw |
| 22:40.27 | voxio | the sip phone is in the DMZ on a remote DSL line |
| 22:41.13 | *** join/#asterisk ChannelZ (channelz@burner.com) |
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| 22:49.03 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
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| 23:50.00 | Eitan | hey guys, how u would go bout copying a working version of asterisk on a centos box, to a fresh install of centos |
| 23:50.04 | Eitan | all the settings and everything |
| 23:54.00 | *** join/#asterisk chalcedony (~llhull@unaffiliated/chalcedony) |