| 00:00.18 | gushi | [Sep 6 06:17:50] NOTICE[51232] channel.c: Dropping incompatible voice frame on IAX2/isc-ast-1253 of format ulaw since our native format has changed to 0x1 (g723) |
| 00:00.44 | gushi | So for some reason, asterisk is allowing a format that it can't transcode |
| 00:00.49 | gushi | I guess? |
| 00:01.13 | navaismo | so in your peer definition use disallow=all \n allow=ulaw \n allow=ulaw \n allow=gsm |
| 00:01.49 | navaismo | ja replace one of the allow=ualw with allow=alaw |
| 00:02.01 | navaismo | stupid fingers |
| 00:03.30 | *** join/#asterisk xpot-mobile (~james@166-70-100-198.ip.xmission.com) |
| 00:05.08 | gushi | Okay, cool, I put that on the iax trunk to my office and all is now good. |
| 00:12.05 | gushi | Thanks! |
| 00:13.00 | navaismo | anytime |
| 00:15.21 | *** join/#asterisk xpot-mobile (~james@166-70-100-198.ip.xmission.com) |
| 00:19.09 | *** join/#asterisk NightMonkey (~NightMonk@pdpc/supporter/professional/nightmonkey) |
| 00:43.15 | *** part/#asterisk nwxi (~nwxi@client-86-23-125-82.brhm.adsl.virginmedia.com) |
| 00:49.06 | *** join/#asterisk jblack (~jblack@pool-71-173-14-99.sctnpa.east.verizon.net) |
| 00:55.41 | *** join/#asterisk xpot-mobile (~james@166-70-100-198.ip.xmission.com) |
| 01:07.11 | *** join/#asterisk ex-parrot (~ex-parrot@2401:f000:3:0:21a:4dff:fe0d:7e59) |
| 01:07.28 | ex-parrot | hi all.... I have the country selected in indications.conf and the right ring cadence but I still seem to just get a US ring cadence |
| 01:07.30 | ex-parrot | anyone know why? |
| 01:07.52 | ex-parrot | running 1.4.21 |
| 01:08.31 | p3nguin | What did you write in the general section of indications.conf? |
| 01:09.32 | *** join/#asterisk BuenGenio (~Gene@059148208218.ctinets.com) |
| 01:09.57 | ex-parrot | excuse this paste but it's only two lines: |
| 01:09.58 | ex-parrot | [general] |
| 01:09.59 | ex-parrot | country=nz |
| 01:10.53 | ex-parrot | then the NZ section has ringcadence = 400,200,400,2000 |
| 01:10.55 | p3nguin | And you have your [nz] section defined below? |
| 01:10.56 | ex-parrot | which is correct |
| 01:11.03 | ex-parrot | yes |
| 01:11.35 | p3nguin | Did you run "module reload res_indications.so" ? |
| 01:11.46 | ex-parrot | yes |
| 01:11.52 | ex-parrot | I also restarted asterisk entirely |
| 01:12.21 | p3nguin | And what are you doing to hear ring tones? |
| 01:12.45 | *** join/#asterisk xpot-mobile (~james@166-70-100-198.ip.xmission.com) |
| 01:13.03 | ex-parrot | when I dial out another SIP channel I hear the US cadence ring until the outbound call picks up |
| 01:13.35 | p3nguin | So you are calling a phone number via ITSP? |
| 01:14.07 | ex-parrot | not sure what you mean by that |
| 01:14.18 | ex-parrot | I'm calling a PSTN number via a SIP trunk provider |
| 01:14.27 | ex-parrot | is it possible that they're providing the ringing cadence? I wasn't sure what generated it |
| 01:17.46 | p3nguin | Since there's no such thing as a SIP trunk, I'll assume that was an affirmative to my question. |
| 01:18.15 | ex-parrot | p3nguin: you'll have to excuse my ignorance of the relevant terms :/ that's what the provider calls it |
| 01:18.22 | *** join/#asterisk BuenGenio (~Gene@059148208218.ctinets.com) |
| 01:18.28 | p3nguin | If you have another phone on your system, call it and see if you get US or NZ ringing. |
| 01:18.43 | ex-parrot | gives me US cadence |
| 01:19.21 | p3nguin | I don't know why it won't apply the change. |
| 01:19.23 | p3nguin | |
| 01:19.25 | p3nguin | |
| 01:19.48 | ex-parrot | and those characters don't render on my machine :) it's not a big deal, just a cosmetic issue. everything still works fine |
| 01:20.10 | p3nguin | What characters are you talking about? |
| 01:20.30 | ex-parrot | (13:19:31) p3nguin: |
| 01:20.54 | *** join/#asterisk DrDigital (~mmurphy@gallery/DrDigital) |
| 01:20.58 | p3nguin | The line feeds can be disregarded. They were not part of the sentence I typed to you. |
| 01:21.06 | ex-parrot | uhm, ok |
| 01:21.37 | ex-parrot | thanks. |
| 01:21.38 | *** part/#asterisk ex-parrot (~ex-parrot@2401:f000:3:0:21a:4dff:fe0d:7e59) |
| 01:21.40 | p3nguin | I haven't tried the NZ indications before, but I have tried UK, and it worked for me. |
| 01:23.02 | *** join/#asterisk delki8 (~delki8@189.5.136.31) |
| 01:29.49 | *** join/#asterisk FainaUkraina (~Gene@059148208218.ctinets.com) |
| 01:33.36 | *** join/#asterisk adolfomaltez (~taro@190.62.239.123) |
| 01:49.49 | *** join/#asterisk gxdssoft (~gxdssoft@190.234.163.109) |
| 02:12.00 | *** join/#asterisk nix8n82-phone (~AndChat@65.161.180.230) |
| 02:17.28 | *** join/#asterisk Yurka_8400 (~chatzilla@91.202.181.34) |
| 02:23.31 | *** join/#asterisk xpot-mobile (~james@166-70-100-198.ip.xmission.com) |
| 02:25.03 | *** join/#asterisk gxdssoft (~gxdssoft@190.234.163.109) |
| 02:37.51 | *** join/#asterisk greenwolf (greenwolf@cpe-74-77-221-5.buffalo.res.rr.com) |
| 02:41.19 | *** join/#asterisk Juggie (~Juggie@unaffiliated/juggie) |
| 02:50.52 | *** join/#asterisk _mwoodj_ (~mwoodj@pdpc/sponsor/digium/hyper-eye) |
| 02:51.48 | *** join/#asterisk xpot-mobile (~james@166-70-100-198.ip.xmission.com) |
| 03:17.03 | *** join/#asterisk nobodyshome (~nobodysho@bas10-kitchener06-1279412163.dsl.bell.ca) |
| 03:17.24 | dijib | can someone tell me if this make finished or nnot? |
| 03:17.26 | dijib | http://pastebin.com/UxqkrakX |
| 03:18.34 | p3nguin | Looks okay to me. |
| 03:18.38 | dijib | ok then |
| 03:19.08 | dijib | not sure how to configure this one |
| 03:30.10 | p3nguin | You probably don't even need it. |
| 03:35.18 | *** join/#asterisk seraphie (~erin@75.76.38.159) |
| 03:39.40 | *** join/#asterisk bmg505 (~leon@196-209-99-56.dynamic.isadsl.co.za) |
| 03:40.24 | *** join/#asterisk ajkaanbal (~ajkaanbal@189.143.102.103) |
| 03:48.06 | *** part/#asterisk JunK-Y (~junky@pdpc/supporter/active/junk-y) |
| 03:54.02 | dijib | dont even need what? |
| 03:58.09 | penguin | i dont understand why asterisk keeps handling all the rtp streams on the server and not resending th INVITE header on to the next destionation out my trunk |
| 03:58.20 | penguin | to my provider and connect that call to the called extension |
| 04:00.32 | dijib | no clue |
| 04:00.38 | dijib | even what the problem is |
| 04:05.40 | penguin | the problem is asterisk seems to want to handle the rtp media during every call |
| 04:05.58 | penguin | i send calls form openser to be handled to asterisk for outbound calls |
| 04:06.18 | penguin | once asterisk sets up that outbound call with the callee its supporse to forware that traffic between those to enities |
| 04:06.30 | penguin | and not have asterisk stream rtp packets or media |
| 04:06.43 | penguin | canreinvite=yes does not work i have already enabled it |
| 04:07.10 | dijib | your light years beyond me there p3nguin |
| 04:07.49 | p3nguin | What'd I do? |
| 04:14.10 | ChannelZ | what version of asterisk? "canreinvite" is no longer the config directive |
| 04:15.39 | ChannelZ | and * will stay in the media stream if you're doing something like recording the channel or have other options in Dial that might cause it to need to "listen" for dtmf etc |
| 04:16.45 | ChannelZ | meh |
| 04:24.59 | *** join/#asterisk CoffeeIV (~rgr@rrcs-71-42-183-81.sw.biz.rr.com) |
| 04:32.41 | *** join/#asterisk gerhard7 (~gerhard7@82-171-103-215.ip.telfort.nl) |
| 05:05.26 | *** join/#asterisk vinhdizzo (~vinh@pool-173-60-112-55.lsanca.fios.verizon.net) |
| 05:10.04 | *** join/#asterisk tyman_ (~tyler@173-12-219-189-Fresno.hfc.comcastbusiness.net) |
| 05:10.29 | *** join/#asterisk adeeln (~adeel@184.175.36.92) |
| 05:12.24 | *** join/#asterisk mickecarlsson (~Micke@h10n3c1o1101.bredband.skanova.com) |
| 05:12.38 | *** part/#asterisk tyman (~tyler@173-12-219-189-Fresno.hfc.comcastbusiness.net) |
| 05:13.41 | *** join/#asterisk mocker (~mocker@206.55.118.84) |
| 05:13.41 | *** join/#asterisk psilikon (~joel@rrcs-24-73-84-158.se.biz.rr.com) |
| 05:13.53 | *** join/#asterisk sunfone (~jeff@c-67-184-36-158.hsd1.il.comcast.net) |
| 05:14.12 | *** join/#asterisk dmz (~dmz@64.203.235.49.dyn-cm-pool-34.pool.hargray.net) |
| 05:14.20 | *** join/#asterisk _Shadowfax_ (dan@neu.cow.org) |
| 05:14.37 | *** join/#asterisk aberrios (~aberrios@195.171.4.82) |
| 05:14.47 | *** join/#asterisk tzafrir (~tzafrir@212.179.75.202) |
| 05:16.11 | *** join/#asterisk den512 (~dradon@carbon.gonicus.de) |
| 05:21.58 | *** join/#asterisk jblack (~jblack@pool-71-173-14-99.sctnpa.east.verizon.net) |
| 05:47.32 | *** join/#asterisk Ean (ean@unaffiliated/ean) |
| 06:13.17 | *** join/#asterisk Ad-Hoc (~nimbus@194.30.241.105) |
| 06:18.17 | *** join/#asterisk m29poff (~m29poff@ARennes-258-1-260-17.w81-53.abo.wanadoo.fr) |
| 06:18.36 | *** join/#asterisk c0rnoTa (~c0rnoTa@78.24.154.190) |
| 06:18.49 | *** part/#asterisk c0rnoTa (~c0rnoTa@78.24.154.190) |
| 06:27.06 | *** join/#asterisk freeman_u (~freeman@193.110.114.54) |
| 06:37.12 | *** join/#asterisk irroot (~irroot@pbx.distrotech.co.za) |
| 06:42.30 | nunne | I have a customer that needs a good mid-size cisco router for having an asterisk (1.4) pbx behind it. they have, today, an cisco ASA-5510 (and i have had problems with how cisco ASA-5505 handles SIP+NAT), so Im expecting the ASA-5510 to behave in the same way. Does anyone know a good "replacement" router from Cisco that can handle SIP (without SIP-ALG whatever, just plain port-forwards) well?? |
| 06:48.09 | *** join/#asterisk Subdolus (dexterity@creep.bur.st) |
| 06:51.21 | nunne | Or does the ASA-5510 have some sort of SIP-ALG that you can turn off that makes it behave "normally"? gratefull for any answer. not good at cisco products :P |
| 06:53.07 | *** join/#asterisk schmidts (~schmidts@Lo20-rt01.lm33.sil.at) |
| 06:53.09 | schmidts | good morning |
| 06:53.54 | irroot | nunne you refering to a connection tracker / packet mangler to modify the sip for you without need for STUN and the like ?? |
| 06:54.02 | irroot | morning there |
| 06:54.33 | WIMPy | Goo morning. |
| 06:55.09 | WIMPy | The good thing about Cisco routers is that if you made them do what you want, they will probably continue to do so forever. |
| 06:55.33 | WIMPy | The bad thing is that it might be difficult or even impossible to do so. |
| 06:57.12 | *** join/#asterisk BuenGenio (~Gene@awork025232.netvigator.com) |
| 06:59.25 | nunne | irroot: no, the asterisk + provider will handle NAT-support. So basically i just need the router to forward the traffic good :) |
| 06:59.26 | schmidts | Wimpy the only real bad thing i know about cisco routers is that they also have a SIP ALG enabled by default and the option to turn it off is not so easy to find, cause you cannot see it in the config |
| 06:59.39 | *** join/#asterisk FainaUkraina (~Gene@awork066061.netvigator.com) |
| 07:00.14 | schmidts | nunne if you have a cisco router you might have the problem with this sip ALG stuff |
| 07:00.19 | nunne | I had a customer with Cisco ASA-5505, but it worked terrible.. but I used the management tool and i never found anything about SIP-ALG / SIP-support.. So I just thought that it re-writes it by default and there is nothing to do about it.. But maybe there is a way to turn off this "feature" |
| 07:00.42 | nunne | and I read that the ASA-5510 is the same as 5505, just for more "people" :P |
| 07:01.43 | schmidts | nunne just go to the telnet console of the router enter conf t |
| 07:01.54 | schmidts | no ip nat service sip udp port 5060 |
| 07:01.59 | schmidts | exit |
| 07:02.01 | schmidts | write memory |
| 07:02.44 | *** join/#asterisk samuelsapps (~samuel@202.137.7.242) |
| 07:03.08 | schmidts | also the same for tcp if you use it |
| 07:03.10 | nunne | aha, seems simple enough :) perfect! thanks.. and after that i can be pretty sure it will just NAT it like a dumb little router? ;) |
| 07:03.19 | schmidts | yes ;) |
| 07:03.25 | nunne | perfect :D thanks alot! |
| 07:03.29 | schmidts | your welcome ;) |
| 07:04.29 | irroot | schmidts id normally charge 1k [100E] for that service :P |
| 07:06.39 | schmidts | irroot only if i can get access to that router and do like i work one hour on it ;) |
| 07:07.04 | *** join/#asterisk ickmund (~ickmund@cli-5b7e85e2.bcn.adamo.es) |
| 07:09.34 | *** join/#asterisk hehol (~hehol@2001:1438:1009:200:89d4:def7:67d5:eef2) |
| 07:12.39 | *** join/#asterisk Wiretap7 (~Wiretap@unaffiliated/wiretap) |
| 07:20.11 | *** join/#asterisk ollii (~risker@2001:470:1f15:1384:62eb:69ff:fe31:b4) |
| 07:25.55 | *** join/#asterisk Azrael808 (~peter@212.161.9.162) |
| 07:30.46 | *** join/#asterisk tzafrir_laptop (~tzafrir@local.xorcom.com) |
| 07:30.47 | *** join/#asterisk cyford (~cyford@c-24-30-15-39.hsd1.ga.comcast.net) |
| 07:32.02 | *** join/#asterisk nix8n82-phone (~AndChat@65.161.180.230) |
| 07:33.48 | *** join/#asterisk tamiel (~tamiel@213.30.183.226) |
| 07:35.47 | *** join/#asterisk Nasga (~Nasga@186.212.10.93.rev.sfr.net) |
| 07:39.05 | *** join/#asterisk c0rnoTa (~c0rnoTa@78.24.154.190) |
| 07:56.18 | *** part/#asterisk c0rnoTa (~c0rnoTa@78.24.154.190) |
| 08:06.25 | *** join/#asterisk m29poff (~m29poff@ARennes-258-1-260-17.w81-53.abo.wanadoo.fr) |
| 08:10.36 | *** join/#asterisk BuenGenio (~Gene@059148208218.ctinets.com) |
| 08:11.19 | *** join/#asterisk chrisjunkie (~Chris@2001:4428:22d:2:208:2ff:fe7e:6312) |
| 08:11.26 | *** join/#asterisk TimeRider (~steve@188-220-33-123.zone11.bethere.co.uk) |
| 08:19.29 | *** join/#asterisk greenwolf (penguin@cpe-74-77-221-5.buffalo.res.rr.com) |
| 08:21.35 | ollii | hey...ive got a question about asterisk 1.8.5 and dahdi 2.5.0 about fax/voice switching |
| 08:21.41 | ollii | If the call is an fax, it will automatically jump into the fax extension IF your fax call comes in thru a ZAP channel (i.e. not a CAPI channel). |
| 08:21.47 | drzed | i'm having a Pickup() problem: it simply does not work and the log says: No channel found 1073741824. |
| 08:21.52 | ollii | is that also true for dahdi? |
| 08:21.57 | ollii | source is: http://www.voip-info.org/wiki/view/Asterisk+fax |
| 08:22.32 | ollii | below is an entry about dahdi but im not sure if that really works |
| 08:23.13 | irroot | drzed trace pb please |
| 08:23.21 | drzed | how does asterisk come to this channel id (=2^30)? core show channels shows no such channel ... |
| 08:24.39 | WIMPy | ollii: dahdi should do everything zap did. |
| 08:25.22 | drzed | irroot: core show verbose tells me this: ;Executing [*830@lokal:2] Verbose("SIP/34-b6a73e50", "1|34 wants to pickup from extension 30") in new stack |
| 08:25.25 | drzed | Executing [*830@lokal:3] PickUp("SIP/34-b6a73e50", "30") in new stack |
| 08:25.28 | drzed | No channel found 1073741824. |
| 08:25.33 | irroot | ~pb |
| 08:25.33 | infobot | [~pb] A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few: http://www.pastebin.com , http://pastebin.us , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://bin.cakephp.org/ , http://asterisk.pastey.net/ ; or install pastebinit with yum or aptitude. |
| 08:25.46 | irroot | 1073741824 = -1 or looks like it |
| 08:26.50 | drzed | irroot: http://nopaste.voric.com/paste.php?f=q3ex9p |
| 08:27.09 | drzed | i'm calling 30 from 35 and try to pick the call with 34 |
| 08:29.10 | drzed | irroot: http://nopaste.voric.com/paste.php?f=xi1joy <--- sip && extensions conf |
| 08:30.42 | *** join/#asterisk singler (~singler@81-7-123-162.static.zebra.lt) |
| 08:34.51 | *** join/#asterisk nicola_pav (~chatzilla@62.219.238.196) |
| 08:35.43 | nicola_pav | hello. anyone familiar with tshark? I want to use tshark to capture SIP packets. I want only to capture SIP NOTIFY only. Is this possible? any hints please? |
| 08:38.33 | wdoekes2 | nicola_pav: ngrep might be what you want |
| 08:38.56 | wdoekes2 | e.g. ngrep 'CSeq: \d+ NOTIFY' udp |
| 08:40.43 | nicola_pav | wdoekes2: thank u |
| 08:41.18 | Maliuta | nicola_pav: tshark? what's wrong with tcpdump? |
| 08:41.37 | Maliuta | nicola_pav: or is your regex foo not that good? ;P |
| 08:42.47 | drzed | irroot: any idea what the problem could be? |
| 08:43.15 | irroot | drzed does not make sense at the moment looking at code |
| 08:43.33 | irroot | Maliuta nicola_pav sometimes sip debug in asterisk is all you need |
| 08:43.45 | *** join/#asterisk Faustov (~fst@gentoo/user/faustov) |
| 08:44.24 | irroot | drzed version of ast you using ?? |
| 08:46.26 | Maliuta | irroot: I know that, I use it all the time |
| 08:46.58 | Maliuta | irroot: but if you're doing something on a router, for example, tcpdump is the original and the best :) |
| 08:47.43 | irroot | Maliuta indeed i use tcpdump daily at least but for the normal run of mill sip tracing asterisk has the goods |
| 08:54.35 | drzed | irroot: 1.4.21.2~dfsg-3+lenny5 |
| 08:55.30 | irroot | ok that is why i cant find error in code im sorry wont help you with 1.4 im rolling out 10 soon and only going to focus on 1.8/10 |
| 08:55.32 | *** join/#asterisk jkroon (~jkroon@dsl-241-232-76.telkomadsl.co.za) |
| 08:55.45 | irroot | jkroon holla come right ? |
| 08:57.35 | wdoekes2 | jkroon: tried gdb -ex generate-core-file -ex detach -ex quit yet? |
| 09:04.16 | *** join/#asterisk Wiretap (~Wiretap@unaffiliated/wiretap) |
| 09:08.54 | greenwolf | anyone here familiar with rtpproxy for |
| 09:13.54 | ollii | WIMPy: thanks |
| 09:15.01 | *** join/#asterisk stix (~stix@193.89.191.209) |
| 09:16.33 | greenwolf | dooes anyone know anything about rtpproxy? |
| 09:16.49 | *** join/#asterisk kaii (~kh@ciphron.de) |
| 09:17.21 | kaldemar | greenwolf: if you have a specific question, just ask it. you're more likely to get answers. |
| 09:18.49 | greenwolf | when i start rtpproxy with rtpproxy -F -l 192.168.1.101:8889 |
| 09:18.54 | greenwolf | i receive rtpproxy: setbindhost: Name or service not known |
| 09:19.04 | greenwolf | how do i get rtpproxy to start up? |
| 09:19.56 | *** join/#asterisk sgimeno (~sgimeno@163.117.206.10) |
| 09:22.27 | kaii | greenwolf: -l is only for the listen address, not port. you should really read the manpage http://www.digipedia.pl/man/doc/view/rtpproxy.8/ |
| 09:22.42 | jkroon | irroot, no I did not. |
| 09:23.32 | jkroon | wdoekes2, that would be really cool, now how do I get that to happen at _exactly_ the moment I detect the error condition? By the time any kind of pipe process picks up the log entry the stack has long since unwinded and the core dump becomes useless. |
| 09:24.04 | greenwolf | kaii: now i get rtpproxy: can't find ID for the user: 74.77.221.5: Success |
| 09:24.13 | greenwolf | does this mean rtpproxy server has started? |
| 09:25.32 | wdoekes2 | hm.. that's another matter. you want the bt of a different thread than the one you're detecting it in, right? |
| 09:26.19 | *** join/#asterisk xnfinite (~xnfinite@62.82.191.170.static.user.ono.com) |
| 09:28.34 | *** join/#asterisk sekil (~sekil@78.24.104.73) |
| 09:33.21 | wdoekes2 | jkroon: you could try to system() the gdb command at log-write-time. (use at own risk!) |
| 09:41.08 | schmidts | i need some help with dahdi and sangoma |
| 09:41.33 | schmidts | i got this errors: http://pastebin.com/SBb0ceL6 |
| 09:41.40 | schmidts | any ideas what could be the problem? |
| 09:43.11 | Gugge | asterisk[608]: segfault at 31c ip 0000000000453ca6 sp 00007f729433e180 error 4 in asterisk[400000+187000] <- is that good? :) |
| 09:43.56 | wdoekes2 | segfault is never good :) |
| 09:44.07 | schmidts | further information: http://pastebin.com/jcCxC35b |
| 09:44.48 | irroot | segfault nom nom |
| 09:46.25 | irroot | schmidts what the problem looks like its setup and some lines are dead ?? |
| 09:48.01 | irroot | any thing in wanrouter log / dmesg |
| 09:49.17 | *** join/#asterisk garymc (~chatzilla@host81-148-107-162.in-addr.btopenworld.com) |
| 09:51.10 | StaRetji | Folks |
| 09:51.31 | StaRetji | my system date/time : Tue Sep 6 11:51:23 CEST 2011 |
| 09:51.47 | StaRetji | but asterisk shows two hours less, 09:51 |
| 09:51.58 | StaRetji | can someone help me out fixing this, thx |
| 09:52.04 | jkroon | wdoekes2, no, i want the backtrace for the specific thread that is executing. |
| 09:52.15 | wdoekes2 | Gugge: echo 0000000000453ca6 | addr2line -Cfe /usr/sbin/asterisk |
| 09:52.46 | jkroon | I suspect the backtrace(3) and backtrace_sumbols_fd(3) is the way to go. |
| 09:53.04 | Gugge | wdoekes2: nice - asterisk/main/channel.c:4897 |
| 09:53.22 | irroot | StaRetji ntpd ?? |
| 09:53.27 | jkroon | ouch! Gugge that one could hurt a lot. |
| 09:53.35 | wdoekes2 | jkroon: that sounds about right |
| 09:53.39 | irroot | StaRetji and or timezone settings |
| 09:53.53 | Gugge | jkroon: asterisk stops when i hit it :) |
| 09:54.10 | irroot | Gugge what vero |
| 09:54.35 | Gugge | 1.6.2.11 |
| 09:54.42 | StaRetji | irroot: ntpd, ehm, you mean I have to have it installed for asterisk to take correct time? |
| 09:54.46 | Gugge | its int oldstate = chan->_state; in ast_setstate() |
| 09:54.49 | StaRetji | I mean, my system time is okay |
| 09:55.04 | irroot | yeah saw system time is good |
| 09:56.39 | jkroon | StaRetji, did you change the timezone settings after starting asterisk? |
| 09:56.45 | StaRetji | nope |
| 09:56.56 | jkroon | did you explicitly set a timezone for asterisk? |
| 09:56.57 | StaRetji | that is timezone settings old 4 years |
| 09:57.07 | StaRetji | asterisk installed 2 years ago lol |
| 09:57.09 | jkroon | or are you telling asterisk to use gmt perhaps? |
| 09:57.15 | StaRetji | but I never needed before |
| 09:57.34 | StaRetji | where can I check that? |
| 09:57.51 | StaRetji | I'm having real time inbound calls which I have to keep record |
| 09:58.08 | jkroon | asterisk.conf iirc. |
| 09:58.11 | StaRetji | and now I have problem with this time difference |
| 09:58.59 | StaRetji | http://pastebin.com/K8dLQMJ8 |
| 09:59.03 | StaRetji | asterisk.conf |
| 09:59.34 | wdoekes2 | Gugge: you want to try latest 1.6.2 (or .17, because .19 has refer issues) before investigating further |
| 09:59.49 | wdoekes2 | .. and enable core dumps |
| 10:00.15 | *** join/#asterisk black187 (5d67162a@gateway/web/freenode/ip.93.103.22.42) |
| 10:00.18 | Gugge | wdoekes2: im going to try to see if i can find out what causes it to happen first :) |
| 10:00.33 | Gugge | but i guess i should enable core dumps first :) |
| 10:00.39 | wdoekes2 | indeed |
| 10:01.48 | black187 | Hello, one question I was having deadlocks in Asterisk 1.8.5.0 when a reload was made. I will upgrade to 1.8.6.0 to see if this was fixed - and this will be done in production system. Can I run "make" and "make install" when Asterisk is still running, or do I have to shut down Asterisk before upgrading? |
| 10:03.07 | jkroon | StaRetji, where are you seeing that time difference exactly? |
| 10:03.20 | Gugge | wdoekes2: can i enable it while its running, or do i have to start it with -g? |
| 10:03.44 | ollii | black187: make is okay...i would trigger "make install " after stopping asterisk |
| 10:03.52 | StaRetji | jkroon: in /var/log/asterisk/cdr-csv/Master.csv |
| 10:03.53 | wdoekes2 | start with -g (or proper ulimit settings) |
| 10:03.55 | ollii | just for my own nerves...maybe its okay |
| 10:04.08 | black187 | Okay, will do that... |
| 10:04.39 | wdoekes2 | ollii: make install is "guaranteed" (no warranty) to be okay |
| 10:04.53 | wdoekes2 | .. while running |
| 10:05.38 | ollii | wdoekes2: "no warranty" <3 |
| 10:05.39 | wdoekes2 | you'll want to make install early, because it goes ahead and downloads sound files first.. which can take up quite some time |
| 10:05.41 | black187 | So it's no problem making upgrade with Asterisk running. Then I just make asterisk service restart and voila - new version? |
| 10:05.57 | ollii | wdoekes2: it sounds like an upgrade so i think downloading sounds should not be neccessary |
| 10:06.17 | ollii | make menuselect => remove "X" |
| 10:06.57 | wdoekes2 | black187: affirmative |
| 10:07.19 | StaRetji | jkroon: I have debian and it seems that ntp is working correctly |
| 10:07.39 | black187 | Great, thanks guys!!! |
| 10:07.56 | wdoekes2 | StaRetji: were does asterisk show this? |
| 10:08.40 | StaRetji | wdoekes2: no, it is debian console. I ran ntpq -p and got *mirror.be.gbxs. 192.87.106.2 2 u 289 1024 377 20.481 0.844 0.106 |
| 10:09.22 | jkroon | StaRetji, weird one. restart asterisk perhaps and see if it fixes it? |
| 10:09.41 | jkroon | the only place i can find tz settings for ast now is in voicemail.conf |
| 10:09.52 | wdoekes2 | the bad time, I mean, StaRetji. if in cdrs. did you put usegmtime=yes in the conf? |
| 10:10.36 | StaRetji | jkroon: Asterisk is restarted several times in few months, while system time is correct for years |
| 10:11.07 | StaRetji | wdoekes2: which conf would that be. I never set it, expect if it was there by default |
| 10:11.11 | wdoekes2 | cdr.conf |
| 10:11.24 | jkroon | In terms of UTC/GMT±X, what is CEST? |
| 10:11.29 | wdoekes2 | +2 |
| 10:11.51 | jkroon | ok, then in cdr.conf, or the logger you are using switch off usegmtime. |
| 10:12.30 | StaRetji | usegmtime=yes |
| 10:12.35 | StaRetji | I should put no? |
| 10:12.37 | StaRetji | right? |
| 10:13.28 | jkroon | ok, i can reproduce the fdopen error on a non-production box !! |
| 10:15.38 | irroot | jkroon win |
| 10:15.44 | irroot | now to kill it |
| 10:16.09 | StaRetji | jkroon: should I change usegmtime=yes to usegmtime=no |
| 10:16.09 | StaRetji | ? |
| 10:16.21 | jkroon | usegmtime=no |
| 10:16.41 | jkroon | irroot, big win. no, now to go to a big meeting where i decide if it's worth it to fix it. |
| 10:16.49 | jkroon | btw, i'd like to get a quote from you. |
| 10:18.30 | jkroon | wdoekes2, how do I set a breakpoint in ast?!? |
| 10:18.37 | jkroon | gdb actually - been a while. |
| 10:19.16 | irroot | jkroon quote 'Be sure to eat something red today.' |
| 10:19.26 | jkroon | ehe |
| 10:19.56 | irroot | I'll get a bunch of monkeys, dress 'em up, and make 'em reenact the Civil |
| 10:19.58 | irroot | War! Heh, heh, heh! |
| 10:20.00 | irroot | -- Homer Simpson |
| 10:20.02 | irroot | Homer the Great |
| 10:21.24 | wdoekes2 | break funcname |
| 10:21.50 | wdoekes2 | (and del 1 to remove breakpoint 1) |
| 10:21.51 | jkroon | wdoekes2, thanks! got it. |
| 10:23.55 | StaRetji | irroot: jkroon and wdoekes2 BIG THX FOLKS! |
| 10:23.59 | StaRetji | works ;) |
| 10:24.24 | *** join/#asterisk c0rnoTa (~c0rnoTa@78.24.154.190) |
| 10:26.00 | *** join/#asterisk TimeRider (~steve@188-220-33-123.zone11.bethere.co.uk) |
| 10:35.43 | schmidts | irroot there are only two lines connected, 1 and 5 |
| 10:36.02 | irroot | ok looks good then ?? |
| 10:36.29 | irroot | what the problem |
| 10:37.48 | *** join/#asterisk chrisjunkie (~Chris@2001:4428:22d:2:208:2ff:fe7e:6312) |
| 10:39.20 | *** join/#asterisk black187 (5d67162a@gateway/web/freenode/ip.93.103.22.42) |
| 10:39.44 | black187 | does anybody here have issues with deadlock on reload (ast 1.8.5.0 and 1.8.6.0)? |
| 10:40.34 | schmidts | irroot it looks like the connection flaps but i guess this is something like a wiring problem |
| 10:41.22 | schmidts | black187 sounds like this problem: https://reviewboard.asterisk.org/r/1407/ |
| 10:42.53 | *** join/#asterisk fisted (~fisted@unaffiliated/fisted) |
| 10:44.12 | black187 | schmidts: hmm, no reload issue there. Or did I miis it? |
| 10:45.00 | schmidts | black187 sorry got the wrong one |
| 10:45.24 | black187 | Do you have the right one? :D |
| 10:47.10 | schmidts | nope sorry, i just thought it was the same but its segfaulting not deadlock, do you have any traces? |
| 10:47.56 | black187 | No, it occurs occasionaly - once a hour, or once a week... Strange :( |
| 10:48.31 | black187 | Reload ussualy works, then sometimes it just hangs and asterisk deadlock occurs :( |
| 10:49.18 | irroot | black187 core show locks / coredump |
| 10:49.53 | schmidts | black187 i guess the deadlock happens somewhere before and when you do the reload everything got stuck cause of the deadlock |
| 10:51.09 | black187 | Yeah, I will try to get the exact moment of deadlock, then I can do backtrace... |
| 11:01.29 | *** join/#asterisk c0rnoTa (~c0rnoTa@78.24.154.190) |
| 11:05.10 | *** part/#asterisk c0rnoTa (~c0rnoTa@78.24.154.190) |
| 11:16.32 | *** join/#asterisk rgsteele (~rgsteele@apo.aweber.com) |
| 11:25.11 | *** join/#asterisk oej (~olle@static-213-115-251-100.sme.bredbandsbolaget.se) |
| 11:25.48 | *** join/#asterisk c0rnoTa (~c0rnoTa@78.24.154.190) |
| 11:27.18 | *** part/#asterisk c0rnoTa (~c0rnoTa@78.24.154.190) |
| 11:34.26 | *** join/#asterisk e7e5 (~rudenko@188.134.2.33) |
| 11:39.16 | *** join/#asterisk atheos (~atheos@208.119.68.16) |
| 11:55.37 | *** join/#asterisk billmania (~bill@38.98.130.98) |
| 12:10.03 | *** join/#asterisk wesphillips (~wphill04@137-237-233-124.harris.com) |
| 12:11.07 | *** join/#asterisk willianmazzardo (~willian@187.53.232.107) |
| 12:11.15 | willianmazzardo | tzafrir_laptop: hi ... good morning ;) |
| 12:11.24 | tzafrir_laptop | hi |
| 12:11.35 | willianmazzardo | do you have the oslec patch for dahdi 2.5.0? |
| 12:12.20 | irroot | hwillianmazzardo its simple to patch against kernel oslec you got the staging echo mod ?? |
| 12:13.04 | tzafrir_laptop | willianmazzardo, yes |
| 12:13.08 | willianmazzardo | irroot: i have a patch, by tzafrir_laptop who I apply in dahdi's file |
| 12:13.35 | willianmazzardo | tzafrir_laptop: can you send me? |
| 12:13.38 | willianmazzardo | ;) |
| 12:14.26 | tzafrir_laptop | http://anonscm.debian.org/viewvc/pkg-voip/dahdi-linux/trunk/debian/patches/dahdi_linux_extra?view=co |
| 12:14.33 | tzafrir_laptop | http://anonscm.debian.org/viewvc/pkg-voip/dahdi-linux/trunk/debian/patches/dahdi_linux_extra?view=log |
| 12:14.56 | tzafrir_laptop | Though it includes a few extra drivers |
| 12:15.00 | *** join/#asterisk leifmadsen (~Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 12:15.00 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 12:16.22 | irroot | http://pastebin.com/RRCY8LnE tzafrir_laptop willianmazzardo <- oslec hack |
| 12:16.41 | willianmazzardo | irroot: thank you... tzafrir_laptop too ;) |
| 12:19.44 | awk | man 1.8 is buggy, pickup causes such problems, with orphan channels. |
| 12:20.06 | awk | who finds 1.6 stable? I totaly skipped this branch |
| 12:20.24 | irroot | awk that pickup thing is fixed now in SVN |
| 12:21.09 | irroot | and not any deadlocks to speak of |
| 12:21.30 | awk | thanks irroot |
| 12:21.56 | *** join/#asterisk eduzimrs (~eduzimrs@201.22.86.124.static.gvt.net.br) |
| 12:26.48 | *** join/#asterisk jkroon (~jkroon@dsl-241-232-76.telkomadsl.co.za) |
| 12:27.24 | *** join/#asterisk TimeRider (~steve@188-220-33-123.zone11.bethere.co.uk) |
| 12:27.47 | anonymouz666 | irroot: hello |
| 12:28.00 | irroot | hello |
| 12:28.02 | anonymouz666 | irroot: I am working to update the 1.4 to 1.8 soon |
| 12:28.11 | anonymouz666 | I am just waiting for 1.8.7 release |
| 12:28.25 | anonymouz666 | yesterday I tried to svn co the 1.8 branch and didn't work |
| 12:28.48 | irroot | ah ok well the rc1 will be out soon |
| 12:29.09 | anonymouz666 | irroot: in what condition the app_queue deadlock occurs? |
| 12:29.17 | anonymouz666 | when the queue weight is set? |
| 12:29.45 | irroot | leifmadsen will know the secret date |
| 12:30.07 | leifmadsen | 1.8.7.0-rc1 will be out early this week |
| 12:30.08 | irroot | anonymouz666 no its reported on JIRA under heavy load |
| 12:30.16 | leifmadsen | it will be a snapshot of whatever is in SVN 1.8 branch at the time |
| 12:30.39 | irroot | there any blockers ?? |
| 12:31.04 | anonymouz666 | in JIRA, If I look correctly, there is one blocker |
| 12:31.25 | leifmadsen | irroot: https://issues.asterisk.org/jira/browse/ASTERISK-18393 |
| 12:32.19 | anonymouz666 | leifmadsen: that user comment will be considered for this release? |
| 12:33.15 | irroot | ah cool that SQL thing is a clean reversion IMHO |
| 12:33.49 | irroot | who is on it ?? tilgman ? |
| 12:34.02 | anonymouz666 | corydon76 is the sql master of asterisk. |
| 12:34.48 | leifmadsen | irroot: no one right now -- you're welcome to commit if you wish |
| 12:34.52 | irroot | indeed |
| 12:35.15 | irroot | leifmadsen ill RB it in a bit |
| 12:35.26 | leifmadsen | irroot: great thanks! |
| 12:37.26 | anonymouz666 | leifmadsen: I am preparing a nice stuff with realtime queues and openais - nice project |
| 12:37.46 | anonymouz666 | a big distributed call center setup |
| 12:38.03 | anonymouz666 | more than 300 seats |
| 12:44.04 | StaRetji | need help folks |
| 12:44.12 | StaRetji | I have ivr |
| 12:44.18 | StaRetji | with 1 did |
| 12:44.25 | StaRetji | and 10 extensions behind |
| 12:44.48 | StaRetji | can someone help setting if extension number1 is busy, try extension number2? |
| 12:44.51 | StaRetji | thx folks |
| 12:45.03 | StaRetji | 1 did have 10 channels of course |
| 12:47.22 | StaRetji | I fail to find example on google, anyone, any tip/help is highly appreciated :) |
| 12:48.39 | *** join/#asterisk iulhk (carryoff@117.102.40.148) |
| 12:50.03 | cyford | star try followme |
| 12:50.13 | willianmazzardo | StaRetji: you can use something like this: |
| 12:50.28 | willianmazzardo | exten => DID,1,Dial(SIP/ext1,50,tT) |
| 12:50.38 | willianmazzardo | exten => DID,n,Dial(SIP/ext2,60,tT) |
| 12:50.42 | *** join/#asterisk chuckf (~chuckf@ubuntu/member/chuckf) |
| 12:51.04 | willianmazzardo | when SIP/ext1 is offline or busy ... the call goes to next extension ... in this case, SIP/ext2 |
| 12:51.04 | iulhk | hi all, i am sample_a.agi i want to call sample_b.agi from sample_a.agi , is there any command for it ? |
| 12:51.26 | StaRetji | willianmazzardo: thx, I forget to mention, that did is calling one extension, sorry, and then context leads caller to final extension |
| 12:51.39 | StaRetji | problem I have, I've set 1 extension and all calls are going to one person |
| 12:51.45 | StaRetji | and that person now hates me lol |
| 12:51.56 | willianmazzardo | StaRetji: ehehehe |
| 12:51.59 | StaRetji | lol |
| 12:52.24 | StaRetji | so, did > localext> ivr > ext1 |
| 12:52.39 | cyford | set them up to go to a que then make her priority and if she dont answer it will role over |
| 12:52.49 | StaRetji | and I need did > localext> ivr > ext1, if busy ext2, if busy ext3 and so on |
| 12:52.56 | StaRetji | something like that |
| 12:53.24 | StaRetji | cyford: easy said then done lol, I don't have a clue how to do that :) |
| 12:53.35 | StaRetji | cyford: thx for reply |
| 12:54.03 | *** join/#asterisk ollii (~risker@2001:470:1f15:1384:62eb:69ff:fe31:b4) |
| 12:54.05 | StaRetji | cyford: also, she answers |
| 12:54.17 | StaRetji | and now other callers can't get her |
| 12:54.46 | StaRetji | but they need to be bypassed to ext2, if that one busy (on the line) ext3 and so on |
| 12:55.13 | cyford | she only has a one line phone? |
| 12:55.54 | StaRetji | yes |
| 12:55.59 | StaRetji | 10 person 10 phones |
| 12:56.24 | StaRetji | exten => 3,3(call212001),Goto(default,212001,1) |
| 12:56.45 | StaRetji | but I have 212002 to 212010 |
| 12:56.50 | StaRetji | also |
| 12:57.16 | StaRetji | willianmazzardo: can I do something with this extension, similar to your example? |
| 12:58.10 | StaRetji | omg, she called me and said she is gonna break my legs |
| 12:58.11 | StaRetji | lol |
| 12:58.35 | willianmazzardo | StaRetji: sorry ... I can`t help you right now ... :S |
| 12:58.36 | StaRetji | I guess I have to fix it soon |
| 12:58.41 | StaRetji | willianmazzardo: no problem |
| 12:59.32 | cyford | well i cheat and use freepbx a nice gui.. so my commands arnt that bright but i accomplish by using ques |
| 13:00.19 | cyford | but if she is set too answer alot of calls i would setup a 6 line phone for her |
| 13:01.14 | StaRetji | problem is, she is suppose to be relaxed, 10 phones ring equally |
| 13:01.31 | StaRetji | now, she is on the line all the time, other 9 polish nails |
| 13:01.32 | StaRetji | hehehehe |
| 13:01.40 | *** join/#asterisk SuPrSluG (~SuPrSluG@8.22.96.106) |
| 13:02.02 | cyford | what version of asterisk you using |
| 13:02.20 | StaRetji | 1.4 |
| 13:03.49 | iulhk | anybody guide how to call agi from another agi ? |
| 13:04.55 | StaRetji | cyford: is it something like this: |
| 13:04.56 | StaRetji | exten => 502,3,Dial(SIP/502,20,trj) |
| 13:04.58 | StaRetji | trj? |
| 13:05.19 | cyford | star you really want to go with queues, i think changing dial plans for this will drive you mad when ques is made for it |
| 13:05.21 | *** join/#asterisk serafie (~erin@nat/digium/x-ugwarhxbwiigsqcy) |
| 13:06.02 | cyford | make a que, give it an exstenstion,,.. then setup 10 users to be agents in that que |
| 13:06.10 | cyford | http://www.orderlyq.com/asteriskqueues.html |
| 13:06.12 | StaRetji | got it |
| 13:06.39 | cyford | there should be a queue.conf file right |
| 13:07.09 | *** join/#asterisk oej (~olle@static-213-115-251-100.sme.bredbandsbolaget.se) |
| 13:07.19 | StaRetji | yes |
| 13:07.36 | StaRetji | I'm reading it, not sure if I'm capable of making it work, but I'm trying |
| 13:08.04 | *** join/#asterisk mandla (~mandla@168.167.180.161) |
| 13:20.57 | p3nguin | staretji: You shouldn't use option r unless you know why you're using it. |
| 13:21.28 | p3nguin | and j isn't needed in most situations. |
| 13:21.42 | p3nguin | and if you have SIP phones, you also don't need t. |
| 13:21.52 | p3nguin | So now you're down to no options. |
| 13:22.01 | StaRetji | p3nguin: got it, thx |
| 13:22.11 | p3nguin | You also should consider not using numbered priorities. |
| 13:22.39 | p3nguin | So instead of 502,3, you would have 502,n for every priority that isn't 1. |
| 13:23.38 | p3nguin | staretji: See this example http://pastebin.com/Piqv4Egj |
| 13:24.18 | Maliuta | or you could just use ael or lua :) |
| 13:24.35 | Maliuta | bees non-helpful helpful :) |
| 13:25.22 | StaRetji | p3nguin: Maliuta, thx folks, you are really kind |
| 13:25.58 | wdoekes2 | p3nguin: why all the trailing semi-colons? |
| 13:26.09 | p3nguin | end of line |
| 13:26.22 | Maliuta | I do all my extensions in ael, it's more flexible for me and easier for me to read and update |
| 13:26.39 | p3nguin | not required in .conf, but I use them anyway. |
| 13:26.56 | Maliuta | safer to use them than not |
| 13:27.02 | wdoekes2 | yuck @ unecessary chars |
| 13:27.15 | Maliuta | never know what may try to read the file and then barf all over your install |
| 13:27.39 | p3nguin | Yeah, yuck, a few hundred semi-colons to look at. Pfft. |
| 13:28.15 | p3nguin | If you don't like them, don't look at them. |
| 13:28.32 | wdoekes2 | can't.. *must* *open* *every* *link* |
| 13:28.33 | Maliuta | covers the delicate ears of Katty before saying the next bit ... |
| 13:28.42 | Maliuta | AEL is the dogs bollocks! |
| 13:28.57 | Maliuta | (that's a good thing people) |
| 13:36.45 | *** join/#asterisk FiReSTaRT (~dlyh@unaffiliated/firestart) |
| 13:36.53 | leifmadsen | Maliuta: meh |
| 13:37.37 | leifmadsen | wdoekes2: lol |
| 13:38.13 | *** join/#asterisk bchia (~Adium@nat/digium/x-ckrwdnqnmucgzvdy) |
| 13:39.36 | *** join/#asterisk asilva (~andre@2801:88:1000:2::12) |
| 13:40.35 | *** join/#asterisk otwieracz (~gonet9@v6.gen2.org) |
| 13:40.36 | otwieracz | Hello. |
| 13:40.38 | asilva | Does anyone know if dCAP is under v1.8 ? |
| 13:40.52 | otwieracz | I have problem with Asterisk on my CentOS 5 - there's no app_meetme. |
| 13:41.00 | ollii | dahdi available? |
| 13:41.15 | otwieracz | No⦠|
| 13:41.19 | ollii | thats it |
| 13:41.33 | ollii | meetme requires dahdi |
| 13:41.55 | bchia | asilva as of today dCAP is on v1.6.2 |
| 13:42.12 | *** join/#asterisk _Corey_ (~chatzilla@64.121.4.75) |
| 13:42.20 | otwieracz | <PROTECTED> |
| 13:42.26 | otwieracz | Dependency problems⦠|
| 13:42.26 | asilva | bchia, oh thanks!!! do you know if the diaplans that is need under labs can be one in AEL ? |
| 13:42.50 | *** join/#asterisk coppice (~chatzilla@116.92.16.50) |
| 13:44.39 | otwieracz | ollii: what should I do? |
| 13:44.54 | bchia | There's is a practical section of the test where an Asterisk server is configured. As of today as long as the configuration works then the AEL dialplan is just fine. (Using a GUI to configure the dialplan is not allowed) you can find the most up-to-date information here http://www.digium.com/en/training/certifications/dcap.php |
| 13:45.38 | asilva | bchia, big thanks again!!! |
| 13:46.14 | ollii | otwieracz: install dahdi-firmware i think |
| 13:46.23 | otwieracz | There's no dahdi-firmware. |
| 13:46.42 | *** join/#asterisk moy (~moy@173.239.155.74) |
| 13:47.15 | ollii | http://techspotting.org/how-to-install-asterisk-on-centos-5/ |
| 13:47.17 | *** join/#asterisk bchia (~Adium@nat/digium/x-tnpygjbpqwctiqle) |
| 13:47.19 | otwieracz | Ok, solved. |
| 13:47.28 | ollii | installing dahdi from scratch would resolve it |
| 13:47.30 | ollii | fine |
| 13:47.42 | ollii | you have to ./configure asterisk again |
| 13:48.13 | otwieracz | I added new repo. |
| 13:48.25 | otwieracz | http://www.asterisk.org/downloads/yum |
| 13:48.49 | ollii | that would also a solution ;) |
| 13:56.30 | *** join/#asterisk mintos (~mvaliyav@nat/redhat/x-nwwhntgesqvuomnc) |
| 14:07.14 | *** join/#asterisk jblack (~jblack@pool-71-173-14-99.sctnpa.east.verizon.net) |
| 14:07.50 | *** join/#asterisk irroot (~irroot@196-215-2-62.dynamic.isadsl.co.za) |
| 14:27.36 | *** join/#asterisk putnopvut (~putnopvut@asterisk/master-of-queues/mmichelson) |
| 14:27.36 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 14:27.45 | otwieracz | ollii: but, why with exten => 4,1,MeetMe(,rDMpc) I'm still getting invalid conference pin? |
| 14:27.46 | *** join/#asterisk KavanS (~KavanS@LINBIT/KavanS) |
| 14:29.21 | *** join/#asterisk vinhdizzo (~vinh@pool-173-60-112-55.lsanca.fios.verizon.net) |
| 14:30.34 | *** join/#asterisk ipc9 (~ipc9@173-162-245-206-NewEngland.hfc.comcastbusiness.net) |
| 14:30.39 | *** join/#asterisk Tech_Travis (~Travis_El@cpe-76-168-191-127.socal.res.rr.com) |
| 14:31.26 | leifmadsen | otwieracz: have you configured meetme.conf? |
| 14:31.31 | otwieracz | Yes. |
| 14:31.39 | leifmadsen | did you reload app_meetme? |
| 14:31.51 | otwieracz | But, ,rDMpc shouldn't allow to dynamically create conference rooms? |
| 14:31.54 | leifmadsen | no |
| 14:31.56 | leifmadsen | that's 'd' |
| 14:32.10 | leifmadsen | and you'd have to specify the conference room number too |
| 14:33.01 | leifmadsen | and specify the pin if you want to use ' |
| 14:33.04 | leifmadsen | 'D" |
| 14:33.22 | leifmadsen | MeetMe(1000,D,1234) |
| 14:34.36 | ipc9 | i have an audio card with 8 different channels, is there a way that asterisk can be configured to listen to each channel independently depending on what extension is dialed? For instance i would like to setup extension 1001 to be channel 1 on the sound card, and extension 1002 to be channel 2, etc... |
| 14:34.45 | Gugge | hmm, doesnt D ask for the pin when the first member connects, and uses that for everyone else? |
| 14:35.52 | SuPrSluG | will asterisk execute the variables int the dial string in the order they are listed ie -> exten => 222,1,Dial(SIP/150,,D(w14#),M(hangupcall) . I'm trying to get a door phone to answer dial the buzzer to open and then hangup. |
| 14:36.50 | Gugge | SuPrSluG: cant you just set a max time on the call? |
| 14:37.34 | *** join/#asterisk Rico29 (~rico@sar-b123.olm.fr) |
| 14:37.37 | Rico29 | hello all |
| 14:37.50 | *** join/#asterisk The_Boy_Wonder (~vossel@asterisk/batman-developer/dvossel) |
| 14:37.57 | SuPrSluG | Gugge, that's an idea too. Thnx. |
| 14:38.31 | Rico29 | i've got a problem with sip channels not closing properly |
| 14:38.36 | Rico29 | and not closing at all |
| 14:38.59 | Rico29 | even if I do a "channel request hangup" |
| 14:39.11 | Rico29 | has anyone hear anything about this bug ? |
| 14:39.30 | Rico29 | I use svn version (release 329529) |
| 14:39.39 | *** join/#asterisk Fritz09 (~Adium@pop1-3802.catv.wtnet.de) |
| 14:39.59 | otwieracz | leifmadsen: exten => 4,1,MeetMe(401,rDMpc,4401) |
| 14:40.11 | otwieracz | meetme.conf: conf => 401,4401 |
| 14:40.11 | Kobaz | anyone want a grandstream fxo |
| 14:40.16 | Kobaz | http://www.ebay.com/itm/110739957639?ssPageName=STRK:MESELX:IT&_trksid=p3984.m1555.l2649 |
| 14:40.25 | leifmadsen | otwieracz: if you're using a dynamic conference, there is no need to configure meetme.conf |
| 14:40.31 | otwieracz | leifmadsen: and: |
| 14:40.32 | otwieracz | <PROTECTED> |
| 14:40.32 | leifmadsen | (that's kind of the point of dynamic conferences) |
| 14:40.58 | leifmadsen | otwieracz: not enough information |
| 14:41.09 | leifmadsen | all I can tell from that line is that the call hung up |
| 14:41.29 | otwieracz | leifmadsen: yes, but there's nothing else⦠|
| 14:41.37 | leifmadsen | shrugs |
| 14:41.41 | Kobaz | there's always something else |
| 14:42.08 | Kobaz | that one thing you neglected to paste will be the thing that tells what's wrong |
| 14:42.34 | *** join/#asterisk aidinb (~aidin@unaffiliated/aidinb) |
| 14:42.39 | *** part/#asterisk The_Boy_Wonder (~vossel@asterisk/batman-developer/dvossel) |
| 14:43.17 | otwieracz | http://www.nopaste.pl/13w8 |
| 14:45.46 | *** join/#asterisk mjordan (~mjordan@nat/digium/x-wzihadgerynlpesr) |
| 14:46.20 | *** join/#asterisk gmcharlt (~gmchart`@pdpc/support/active/gmcharlt) |
| 14:46.50 | *** join/#asterisk irroot (~irroot@pbx.distrotech.co.za) |
| 14:54.45 | Rico29 | does anybody have the same problem than me ? |
| 14:55.49 | cyford | rico29 i have experienced this in nat inviroments |
| 14:56.11 | Rico29 | cyford, it appears on no-nat peers too |
| 14:58.42 | cyford | which version of astersik are you using |
| 14:59.09 | Rico29 | Asterisk SVN-branch-1.8-r329529 |
| 15:00.07 | Rico29 | I had to move to svn because of bug ASTERISK-17760 |
| 15:00.18 | cyford | an you saying when a extention dials another extenstion it doesnt hangup after one party leaves |
| 15:00.44 | Rico29 | it doesn't happen everytime |
| 15:01.13 | Rico29 | but sometimes, the sip channel is not closed, and the phone stays "inuse", even if it hanged up |
| 15:01.26 | *** join/#asterisk binbash_ (~peter@server.digitog.nl) |
| 15:01.41 | Rico29 | restart of the phone doesn't solve the problemù |
| 15:01.49 | Rico29 | only a restart of asterisk solves it |
| 15:02.54 | *** join/#asterisk garymc (~chatzilla@host81-148-107-162.in-addr.btopenworld.com) |
| 15:04.31 | *** join/#asterisk irroot (~irroot@pbx.distrotech.co.za) |
| 15:04.50 | Rico29 | should I stay on an SVN version, or should I go to the latest stable release ? |
| 15:04.59 | *** join/#asterisk Stese (~Stese@adsl.ntsols.com) |
| 15:05.22 | Rico29 | -> is the SVN branch enough stable to run in prod environment ? |
| 15:05.34 | *** join/#asterisk navaismo (~navaismo@187.170.4.20) |
| 15:05.54 | Kobaz | otwieracz: do you have dahdi loaded? |
| 15:06.26 | Stese | hey all, hopefully a quick question... I want to use different hold music, depending on what context the call is in when MoH is activated. at the moment, we have no specific configuration for MoH, so when it's activated, the 'default' is used. How do I change this behaviour? |
| 15:06.52 | ollii | what tells you: module load chan_dahdi.so |
| 15:07.40 | otwieracz | Kobaz: yes⦠|
| 15:08.24 | ollii | Stese: SetMusicOnHold(class) prior 1.6 and now it is: Set(CHANNEL(musicclass)=class) |
| 15:08.50 | Kobaz | otwieracz: pastebin ls -al /dev/dahdi |
| 15:09.29 | otwieracz | Hmm. What kmod I should load? |
| 15:10.09 | Stese | ollii : so I need to just have that in each context where I require a different MoH to play (i have added the 'context' in musiconhold.conf) |
| 15:10.38 | ollii | you could add it also to your Dial |
| 15:11.16 | Stese | OlliI > ok, thanks for the prompt response! |
| 15:11.21 | ollii | np |
| 15:12.26 | *** join/#asterisk Defraz (~Defraz@corp.fuzecore.com) |
| 15:13.16 | *** join/#asterisk granola (~david@stg.maculon.com) |
| 15:13.20 | granola | hi! |
| 15:13.38 | otwieracz | Nope, I can't load dahdi module beucase of xen kernel⦠|
| 15:13.45 | granola | i'm researching geo routing options for routing calls and wanted to get some ideas or places to start |
| 15:13.59 | granola | we currently do zipcode based but i'm wondering if there is a better way |
| 15:17.40 | leifmadsen | otwieracz: then you can't use MeetMe |
| 15:17.53 | leifmadsen | otwieracz: you'll need to use timerfd and ConfBridge() |
| 15:18.00 | leifmadsen | res_timing_timerfd* |
| 15:18.09 | leifmadsen | (again, that assumes your kernel supports timerfd) |
| 15:19.24 | *** join/#asterisk garymc (~chatzilla@host81-148-107-162.in-addr.btopenworld.com) |
| 15:19.29 | otwieracz | HMm⦠|
| 15:19.36 | otwieracz | So, try app_confbridge? |
| 15:21.10 | ollii | otwieracz: it might be possible to rebuild your rpms to use your xen kernel source |
| 15:21.24 | ollii | but using app_confbridge would be easier for your |
| 15:22.00 | leifmadsen | otwieracz: since app_confbridge can use res_timing_timerfd instead of res_timing_dahdi, then yes, you'll need to use ConfBridge instead |
| 15:23.08 | otwieracz | OK, thanks. |
| 15:32.07 | Kobaz | leifmadsen: and it assumes you're using a new enough asterisk that doesn't have the really bad timerfd problems... is the patch in 1.8.6.0? |
| 15:34.19 | irroot | Kobaz leifmadsen think its due for 1.8.7 ?? |
| 15:39.24 | *** join/#asterisk ferdna (~ferdna@cpe-67-10-220-35.elp.res.rr.com) |
| 15:39.36 | iulhk | will anybody guide how to call agi from another agi ? |
| 15:40.50 | Rico29 | $AGI->exec('AGI','agi_name.pl) doesn't work ? |
| 15:41.28 | iulhk | <Rico29>: thanx .. let me check |
| 15:41.49 | Rico29 | it's in perl |
| 15:42.01 | iulhk | yes |
| 15:42.08 | Rico29 | but I don't know why you want to use en agi inside an agi |
| 15:42.14 | Rico29 | you should use functions... |
| 15:43.35 | leifmadsen | irroot: Kobaz: 1.8.7.0-rc1 |
| 15:45.18 | *** join/#asterisk KavanS (~KavanS@LINBIT/KavanS) |
| 15:45.42 | anonymouz666 | irroot: do you think it's good to start 1.8 app_queue with our deadlock patch? |
| 15:45.51 | anonymouz666 | that piece of code really could DL? |
| 15:46.09 | anonymouz666 | because queues will be used heavily and the load will be high |
| 15:46.18 | anonymouz666 | exactly where the symptons of the reporter |
| 15:46.24 | irroot | anonymouz666 id appreciate testing of it |
| 15:47.29 | irroot | https://issues.asterisk.org/jira/browse/ASTERISK-18101 |
| 15:48.06 | *** join/#asterisk jwiggins (~James@gateway/tor-sasl/jwiggins) |
| 15:48.49 | irroot | anonymouz666 please note there were not only app_queue problems |
| 15:49.13 | irroot | We've had two lockups in the past 2 weeks. These have been the biggest uptimes between dealocks since rolling out 1.8. |
| 15:49.25 | irroot | that is the part with app_queue |
| 15:55.19 | eja | is it bad form to use _. to match any extension on an incoming context? |
| 15:55.48 | p3nguin | Yes, very bad. |
| 15:56.05 | p3nguin | Because it matches ALL extensions, not just the ones you want to match. |
| 15:56.13 | p3nguin | If you need to match most phone numbers, use _X. instead. |
| 15:56.44 | eja | ahh ok. |
| 15:57.10 | leifmadsen | there is even a readme file in the source you should read... |
| 15:57.31 | leifmadsen | it's called: README-SERIOUSLY.bestpractices.txt |
| 15:57.35 | p3nguin | _X. will match any one digit plus one or more characters. That covers most numbers you'll receive calls on. |
| 15:57.49 | leifmadsen | s/digit/numeric digit/ |
| 15:58.08 | p3nguin | Is there such a thing as a digit which is non numeric? |
| 15:58.19 | leifmadsen | yes, fingers are sometimes referred to as digits |
| 15:58.23 | leifmadsen | :) |
| 15:58.31 | p3nguin | Well it won't match fingers, so that's out. |
| 15:58.41 | p3nguin | "Call my finger." |
| 15:59.26 | Stese | p3nguin > which finger? |
| 15:59.27 | Stese | :P |
| 15:59.28 | p3nguin | "Hey there, pretty lady... you think I can get your fingers?" |
| 15:59.44 | Stese | "can I give you my finger?" |
| 15:59.46 | Gugge | maybe just one of them :P |
| 15:59.49 | navaismo | O_o |
| 16:01.14 | p3nguin | draws a hand turkey on a napkin |
| 16:01.16 | jkroon | hi guys, I'd like to bind chan_sip on both port 5060 and 5070 - any ideas if this is possible? |
| 16:01.32 | jkroon | broken wireless NAT device that's dropping port 5060 traffic ... ubnt ftl. |
| 16:01.54 | p3nguin | I'd forward it at the firewall and leave poor old asterisk alone. |
| 16:02.05 | Stese | wait, wireless? |
| 16:02.25 | jkroon | p3nguin, done, but asterisk sends a Contact: <sip:asterisk@my.ip:5060> and back into the same situation. |
| 16:02.31 | p3nguin | d'oh |
| 16:02.32 | *** join/#asterisk Monya (~Monya@2001:470:27:864::2) |
| 16:02.33 | jkroon | Stese, ask no questions be told no lies. |
| 16:02.47 | *** join/#asterisk oej (~olle@ns.webway.se) |
| 16:02.52 | Stese | isn't the overhead on wireless traffic particulary bad for RTP? |
| 16:03.20 | p3nguin | Try adding a second bindport line with the other port. |
| 16:03.31 | Qwell | Stese: it can be |
| 16:04.14 | Stese | always been my gut instinct to avoid it! :)... then my boss gets a load of wirless n 270mbit kit for our office to use Lync on |
| 16:04.45 | Qwell | Stese: it's mitigable. |
| 16:04.55 | *** part/#asterisk granola (~david@stg.maculon.com) |
| 16:05.06 | p3nguin | Ick, a second bindport line blew away the first one for me. |
| 16:05.15 | Qwell | p3nguin: Don't do that then. |
| 16:05.21 | p3nguin | undoes it |
| 16:07.10 | p3nguin | I'll refer back to my previous suggestion of doing it at the firewall. |
| 16:07.13 | *** join/#asterisk omgwtf (hater@lolunix.org) |
| 16:07.33 | p3nguin | jkroon: Assuming you're using iptables, what rule(s) did you add to try to make it work? |
| 16:07.35 | jaytee | hmmmm |
| 16:08.07 | omgwtf | Is there any way to make it where the queue member has to press a key after answering a call from the queue to accept the call? |
| 16:08.40 | jkroon | iptables -t nat -A PREROUTING -i eth0 -p udp --dport 5070 -j REDIRECT --to-ports 5060 |
| 16:08.54 | jkroon | p3nguin, trying the two udpbindport lines trick now. |
| 16:09.10 | p3nguin | It failed hard for me. |
| 16:09.27 | p3nguin | It killed 5060 when I added a second line, making it listen only on the second listed port. |
| 16:10.08 | p3nguin | That's probably the iptables rule I would have used, so I guess we have to wait for someone who actually uses multiple ports to speak up. |
| 16:10.26 | jkroon | p3nguin, same here. |
| 16:10.43 | *** join/#asterisk jwiggins (~James@gateway/tor-sasl/jwiggins) |
| 16:13.46 | *** join/#asterisk toddejohnson (~toddejohn@adsl-68-23-193-222.dsl.spfdil.ameritech.net) |
| 16:15.05 | *** join/#asterisk talntid (~erict@li93-153.members.linode.com) |
| 16:15.18 | *** part/#asterisk talntid (~erict@li93-153.members.linode.com) |
| 16:22.56 | *** join/#asterisk devmikey (~irc@unaffiliated/devmikey) |
| 16:25.35 | *** join/#asterisk hehol (~hehol@2001:1438:1009:200:416:47e6:73f8:aea7) |
| 16:27.14 | *** join/#asterisk slidesinger (~slidesing@c-68-44-99-163.hsd1.nj.comcast.net) |
| 16:32.15 | *** part/#asterisk willianmazzardo (~willian@187.53.232.107) |
| 16:35.36 | toddejohnson | Can someone help me find the right contact person or place to contact for the asterisk yum repo? The kernel modules aren't complied for the CR(http://lists.centos.org/pipermail/centos-announce/2011-August/017689.html) repo. |
| 16:36.15 | *** join/#asterisk grantm (~grant@68.142.138.4) |
| 16:36.20 | p3nguin | toddejohnson: https://wiki.asterisk.org/wiki/display/AST/Asterisk+Packages |
| 16:37.05 | p3nguin | I guess you could post your comment there, or you could tell pabelanger and/or qwell here. |
| 16:37.49 | toddejohnson | p3nguin: will do thankyou. |
| 16:38.30 | Qwell | correct, it isn't. |
| 16:38.59 | Qwell | - We hope to have the 5.7 Release ready in the next 7 to 10 days |
| 16:38.59 | Qwell | - Once 5.7 is released, the 5.6/cr/ repositories will be removed |
| 16:39.06 | Qwell | It's now been 3 weeks. |
| 16:39.21 | Qwell | CR should have only lasted a week. Hardly worth spending my time on. |
| 16:40.19 | toddejohnson | My opinion too. The repo includes the httpd fixes and many other security related patches to my understanding though. I hate to leave a sys unpatched. |
| 16:40.52 | toddejohnson | Plus wouldn't this give you a head start on compiling for 5.7? |
| 16:47.28 | samandiriel | I have a quick question concerning voicemail |
| 16:47.38 | p3nguin | ~ask |
| 16:47.38 | infobot | 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. |
| 16:48.00 | samandiriel | is it possible to have asterisk store the Message Information file data in a mysql db as well as writing it to the file? |
| 16:48.59 | samandiriel | eg,I'd like to see the info written to /var/spool/asterisk/voicemail/default/5014/INBOX/msg0001.txt also written to a mysql database table |
| 16:49.33 | *** join/#asterisk paulc (~paulc@unaffiliated/paulc) |
| 16:50.22 | samandiriel | it would make sense to me that there would be a feature for this, but I haven't located one. I'd rather not have to write a custom odbc function and have to call it in the dialplan if there's something already available |
| 16:50.58 | *** join/#asterisk Naikrovek (~mbluth@unaffiliated/naikrovek) |
| 16:54.05 | leifmadsen | samandiriel: just use voicemail in realtime, configured in extconfig.conf |
| 16:54.09 | leifmadsen | that's what you want |
| 16:56.29 | *** join/#asterisk garymc (~chatzilla@host81-148-107-162.in-addr.btopenworld.com) |
| 16:56.42 | omgwtf | I have a situation where I add a cell phone to my queue through addqueuemember. Since its a cell voicemail will pickup. Is there a way to make certain queue members press a button to answer a call? |
| 16:57.23 | samandiriel | leifmadsen: wouldn't that also store the vmails themselves in the DB as opposed to files? |
| 16:57.39 | leifmadsen | only if you enabled ODBC storage |
| 16:57.44 | *** join/#asterisk bitbandit (~taggmcd@c-98-202-116-168.hsd1.ut.comcast.net) |
| 16:57.57 | leifmadsen | I'm pretty sure you can just store the data in the DB separate of the binary object |
| 17:02.14 | *** join/#asterisk devmikey (~irc@unaffiliated/devmikey) |
| 17:02.19 | samandiriel | hm. from what I can see, you need to create table "voicemessages" to get this data, which requires column "recording", a longblob |
| 17:02.52 | samandiriel | so I'm thinking not |
| 17:02.55 | *** join/#asterisk DrDigital (~mmurphy@gallery/DrDigital) |
| 17:04.42 | *** join/#asterisk Atriks (~Hatrix@2a01:e34:ee51:4670:9c6:2bde:2970:ed41) |
| 17:04.44 | Atriks | Hello |
| 17:07.02 | *** join/#asterisk brdude (~brdude@c-24-5-194-184.hsd1.ca.comcast.net) |
| 17:07.19 | Atriks | I've just bought a modem card, but I can't find any tutorial about it with Asterisk, So I try here... |
| 17:07.48 | devmikey | Hi, question: I'm having a problem with my SIP connections in that when I place a call, it seems to make the call but my VoIP phone software says it 'calling', then a minute later, it seems to reconnect and starts actually timing the call. Any idea why? I cant access the asterisk logs but I do have my phone client logs. |
| 17:07.51 | devmikey | http://pastebin.com/qx2Hk8qf |
| 17:07.52 | Atriks | My card is a Peabird V92 |
| 17:09.03 | p3nguin | devmikey: Maybe you have a goofy dial plan. |
| 17:10.16 | devmikey | The only thing I can think of is that I'm behind a NAT router |
| 17:10.49 | devmikey | And I cant do port forwarding |
| 17:10.53 | p3nguin | you don't have access to asterisk? |
| 17:11.01 | devmikey | not to my server, no |
| 17:11.16 | raden | Naikrovek, Yo bro |
| 17:11.23 | raden | hugs Katty |
| 17:11.24 | Naikrovek | yo |
| 17:12.01 | raden | how goes it |
| 17:12.26 | Naikrovek | goes well |
| 17:12.29 | Naikrovek | planning new infrastructure |
| 17:12.32 | Naikrovek | always fun |
| 17:12.40 | raden | Sweet whats going on these days ? |
| 17:13.35 | raden | im working on a rather large network at moment 6 switches trunked 2 routers 5 servers 5 wifi radios on one place and a 150' tower going up with more radios |
| 17:13.49 | Naikrovek | neat |
| 17:15.18 | raden | interesting and a learning experiance to say the least |
| 17:15.37 | raden | Naikrovek, The dark side is getting me starting to actually like win 7 :( |
| 17:15.56 | Naikrovek | well win7 is a good os |
| 17:15.58 | Naikrovek | no question |
| 17:21.21 | *** join/#asterisk irroot (~irroot@197.174.72.225) |
| 17:23.27 | *** join/#asterisk brad_mssw (~brad@shop.monetra.com) |
| 17:23.57 | anonymouz666 | hard rock punta cana |
| 17:24.01 | anonymouz666 | nice place to go |
| 17:25.47 | devmikey | Is there a better channel to ask in? |
| 17:27.37 | *** join/#asterisk anthm (~anthm@freeswitch/developer/anthm) |
| 17:28.22 | p3nguin | I don't see how it can easily be troubleshot without seeing the entire picture. |
| 17:29.48 | *** join/#asterisk atan (~atan@unaffiliated/atan) |
| 17:30.00 | pabelanger | Atriks: A 'modem card' will not work with Asterisk. |
| 17:30.05 | pabelanger | you need a tdm card |
| 17:30.07 | pabelanger | ~tdm |
| 17:30.07 | infobot | tdm is probably Time Division Multiplexing. It is a scheme in which numerous signals are combined for transmission on a single communications line or channel. Each signal is broken up into many segments, each having very short duration. |
| 17:30.26 | atan | Seeking cheap adapter to connect to my copper pair so I can connect with Asterisk so I can route calls to it. What's the device of choice these days? |
| 17:30.34 | Atriks | sure ? |
| 17:30.48 | Atriks | A modem card works to make calls, but asterisk doesn't support them ? wtf |
| 17:31.15 | atan | Different sounds I do believe, Atriks |
| 17:32.56 | n3hxs | I am running an openvox card. |
| 17:33.30 | n3hxs | but depends on your description of "cheap" |
| 17:34.31 | Atriks | A friend has tested a modem card |
| 17:34.32 | Atriks | It works |
| 17:34.51 | Atriks | But you're saying that it doesn't work with asterisk ? ... |
| 17:35.03 | Atriks | I don't wanna pay 4584574545574455$ for a card |
| 17:35.04 | leifmadsen | only a very specific type of modem can work -- not just any modem |
| 17:35.17 | leifmadsen | and they don't work very well at all |
| 17:35.25 | Atriks | Yeah I know |
| 17:35.30 | atan | n3hxs, well like $30 would be awesome if I could :) I have a SPA1001 but I guess I got the wrong unit. I'm trying to connect my VOIP phone to a copper line. |
| 17:35.32 | leifmadsen | you're much better off with an ATA |
| 17:35.36 | Atriks | But V92 are the "best" modems for that |
| 17:35.37 | irroot | correct and if it works on windows not garunteed to work on asterisk |
| 17:36.08 | Atriks | I bought one, now I want to tes tit |
| 17:36.08 | p3nguin | atan: The SPA-3102 has both FXS and FXO port, and they cost around $70 US. |
| 17:36.11 | Atriks | test it |
| 17:36.15 | irroot | look at a Linksys 3102 it has 1FXO 1FXS |
| 17:36.20 | irroot | snap |
| 17:36.24 | p3nguin | I like tes tit better. |
| 17:36.26 | leifmadsen | Atriks: it's not just v.92 -- it's a very specific chipset on the modem -- one of the motorola chipsets. The problem is because of the drivers. Anyone with the ability to write a suite of drivers for modem has not done it because it takes a lot of effort. |
| 17:36.59 | atan | Was the SPA-3102 one of those sketchy unlocked things from Vonage that reboots every few hours and causes all hell? |
| 17:37.18 | leifmadsen | not all of them are unlocked vonage models |
| 17:37.26 | p3nguin | I've been running a 3102 for quite some time without ever noticing it has restarted on its own. |
| 17:37.32 | leifmadsen | p3nguin: same here |
| 17:37.36 | irroot | vonage used mostly 2102 |
| 17:38.00 | irroot | there is a setting on all the linksys regarding resync / reset |
| 17:38.01 | atan | Well I remember in the past getting a bunch of linksys ATAs and the things rebooted and dropped calls, err, what a mess that was! |
| 17:38.04 | p3nguin | or PAP2 |
| 17:38.07 | irroot | under provisioning |
| 17:38.12 | atan | p3nguin, I think it was a PAP2, yes. |
| 17:38.28 | p3nguin | I can't think of any reason Vonage would give you an adapter with an FXO port on it. |
| 17:38.49 | irroot | PAP2 is a piece of Sh1t it does not transcode on 2 ports simultaneously nor supports fax |
| 17:39.04 | p3nguin | It shouldn't need to transcode at all on any ports. |
| 17:39.11 | atan | So I want the SPA3102 then by the sounds of it :-) |
| 17:39.13 | p3nguin | And The PAP2T is supposed to support fax. |
| 17:39.42 | irroot | p3nguin dbl check not T.38 only G711 on lan it hurts |
| 17:40.12 | irroot | atan indeed with asterisk-10 + 3102 you can even fax properly with T.38 :P |
| 17:40.48 | atan | I have not upgraded to 10 just yet but soon I will. When I have time to fix anything that might break on me. My dialplan is a complete mess and queries mysql for each call and such. Blarg. |
| 17:40.53 | irroot | atan for more than 1 port look at the digium cards or similar |
| 17:41.30 | atan | Really, what I want to get on the go eventually is to build call features into the call itself somehow so the ATA adapters can do call transfers, hold, call park, 3-way without any issue. Is that even possible? =\ |
| 17:42.20 | irroot | atan there are those features on the ATA but require supporting dialplan |
| 17:42.27 | atan | But I need to rig up my CDR to catch those calls and link them to customer accounts first so I don't put my foot in my mouth :-) anyhow, I will look at getting the SPA3102 for now then seems it would float my boat |
| 17:42.49 | irroot | atan also CEL will help |
| 17:42.54 | atan | What is CEL? |
| 17:43.24 | p3nguin | You can apply account codes to each device, you know. No rigging necessary. |
| 17:43.52 | irroot | Call Event Logging |
| 17:43.55 | atan | p3nguin, right now I have some account code set on the sip peers which puts it into the cdr which is what I use for billing |
| 17:44.32 | irroot | atan also linkedid vs uniqueid is a nice one in the CDR a second call can be linked |
| 17:45.17 | atan | I really need to spend some time looking in to it. I know the one big thing I'd kill to have is move my SIP users from the static file into a MySQL database. |
| 17:45.57 | atan | Blarg! You guys are too helpful sometimes :-) all these new features make my head spin I want them all :D just not enough time to learn it just yet |
| 17:46.01 | atan | books 3 weeks off work |
| 17:46.54 | Atriks | leifmadsen, fu |
| 17:47.05 | Atriks | I have the reference |
| 17:47.10 | Atriks | But I don't find it on internet |
| 17:50.35 | leifmadsen | did I just get told to eff off? |
| 17:50.40 | atan | ... |
| 17:50.52 | *** join/#asterisk devmikey (~irc@unaffiliated/devmikey) |
| 17:51.01 | p3nguin | I kind of wondered that, myself. |
| 17:51.37 | irroot | leifmadsen perhaps some translation ?? |
| 17:51.52 | leifmadsen | I'm unfamiliar with the word 'fu' |
| 17:52.02 | atan | Chinese for "I love you" |
| 17:53.07 | atan | Or maybe Fordham University in Bronx, NY. |
| 17:53.12 | leifmadsen | that is a totally different meaning then :) |
| 17:53.44 | atan | ..hah, urbandictionary says Short for "fur", as a horrible insult to say to one person (and only one person). When "fu" is said to someone, it means you hope someone has an allergic reaction to fur. |
| 17:54.07 | p3nguin | That's pretty stupid. |
| 17:54.11 | Atriks | fu for FFFFFUUUUUU |
| 17:54.28 | p3nguin | It's almost as stupid as writing "u" in place of "you." |
| 17:55.04 | irroot | im a cunning linguist but know nothing about language |
| 17:55.35 | leifmadsen | Atriks: as in, "fuck you" ? |
| 17:55.38 | p3nguin | I don't mind cunnilingus, either. |
| 17:55.40 | Atriks | no |
| 17:56.02 | p3nguin | Please enlighten us as to what "fu" means to you. |
| 17:56.40 | n3hxs | short for foo |
| 17:56.52 | p3nguin | I don't think so. |
| 17:57.00 | p3nguin | fo would be short for foo. |
| 17:57.04 | coppice | fu == 夫 |
| 17:57.11 | Kobaz | i think he hopes you are allergic to fur |
| 17:57.20 | *** join/#asterisk brdude (~brdude@12.155.183.30) |
| 17:57.34 | leifmadsen | 夫 == ? (in English) |
| 17:57.43 | *** join/#asterisk angryuser_laptop (~angryuser@90-156-167-83.reverse.alphalink.fr) |
| 17:58.10 | coppice | its half of å夫 == kung fu |
| 17:58.26 | coppice | could mean wife under some conditions |
| 17:58.44 | Kobaz | all i see is a box with 52/2b |
| 17:59.36 | coppice | how parochial |
| 17:59.39 | anonymouz666 | omg coppice speaks chinese |
| 17:59.42 | anonymouz666 | :P |
| 18:00.21 | coppice | quite a few people around here can |
| 18:00.28 | Kobaz | omgmioplashafsksahfforreels |
| 18:00.51 | anonymouz666 | just like portuguese then. I would change the default language of my country to English/Spanish would be more useful |
| 18:01.00 | anonymouz666 | more people in the world would understand us |
| 18:01.23 | n3hxs | So, spanglish? anonymouz666 |
| 18:02.01 | anonymouz666 | lol |
| 18:02.19 | anonymouz666 | s/\//or |
| 18:02.33 | p3nguin | so close. |
| 18:04.02 | irroot | anonymouz666 our neighbours speak Portuguese [country] but we not big friends with the Mozambican refugees |
| 18:04.38 | *** join/#asterisk trumee (~trumee@cpc2-cmbg7-0-0-cust855.5-4.cable.virginmedia.com) |
| 18:05.21 | trumee | i have bought an x100p card for timing purpose. How do i make sure asterisk is using this card rather than dummy timing source? |
| 18:05.36 | coppice | we have neighbours that can speak just about any language you can think of |
| 18:06.37 | *** join/#asterisk DrDigital (~mmurphy@gallery/DrDigital) |
| 18:08.46 | anonymouz666 | irroot: very very few countries know that language |
| 18:09.17 | anonymouz666 | there are few in Africa, Brasil and Portugal. |
| 18:09.36 | coppice | brazil is big enough to make it a major language |
| 18:10.15 | irroot | anonymouz666 yeah well down here its Angola and Mozambique we also have a large expat Portuguese contingent my aunt married a Portuguese man and cousins are fluent |
| 18:10.21 | trumee | what is the module for dahdi dummy timing source? |
| 18:11.20 | anonymouz666 | irroot: are you fluent? |
| 18:11.59 | irroot | <PROTECTED> |
| 18:13.13 | *** join/#asterisk lvlolvlo (~lvlolvlo@unaffiliated/lvlolvlo) |
| 18:13.50 | anonymouz666 | 11? too many |
| 18:13.55 | trumee | is anybody using dahdi? |
| 18:14.05 | coppice | there are over a hundred in india |
| 18:14.59 | billmania | trumee: I used DAHDI with my PRIs. |
| 18:14.59 | anonymouz666 | trumee: lots of people, including me |
| 18:15.11 | trumee | i want to use an x100p card as a timing source. What do i need to do? |
| 18:16.11 | trumee | i loaded the module 'modprobe wcfxo'. How do i find out whether dahdi_test is using the card rather than the dummy timing source |
| 18:18.03 | trumee | billmania: anonymouz666:? |
| 18:18.18 | trumee | is this a stupid question? |
| 18:18.34 | billmania | I have only used the PRI cards from Digium and haven't even played around much with those - they just work for me. |
| 18:19.31 | n3hxs | trumee, try pulling the card to see if the dummy loads. Maybe it sees the card and the dummy doesn't load. |
| 18:20.51 | trumee | n3hxs: i dont even have a dahdi_dummy module on my system |
| 18:21.10 | p3nguin | The dummy module is no longer separate. |
| 18:21.27 | n3hxs | You can then run the dahdi_test without the card to see if it responds. |
| 18:21.47 | trumee | n3hxs: yes without the card it responds. |
| 18:21.58 | p3nguin | The dummy is included in the dahdi module rathan than the old dahdi_dummy module. |
| 18:22.10 | p3nguin | s/rathan/rather/ |
| 18:22.14 | n3hxs | See... what do I know! ;) |
| 18:23.22 | *** join/#asterisk voxter (~hardcore@macpro.daytonhome.voxter.net) |
| 18:23.27 | *** join/#asterisk DaveCanoe (~Dave@2001:1928:0:ffff::2) |
| 18:23.29 | *** join/#asterisk TimeRider (~steve@188-220-33-123.zone11.bethere.co.uk) |
| 18:23.55 | trumee | p3nguin: ok, if i do a 'modprobe wcfxo', i see both dahdi and wcfxo have loaded. Does it mean that dahdi_test will use the card rather than dummy source now? |
| 18:24.17 | p3nguin | If the hardware is present, I'd guess it will be used. |
| 18:24.36 | voxter | Anyone use cisco 7942's with SIP firmware and asterisk? Any suggestions on which firmware version to use, preferably one that will work behind NAT, but if thats just not possible, which version (non nat) is most compatible? |
| 18:25.06 | trumee | p3nguin: thanks, i dont need to setup anything in asterisk now? |
| 18:25.24 | p3nguin | I have no idea, because I do not use dahdi hardware. |
| 18:26.35 | trumee | dahdi_test is giving a worst of 99.987. Even with a hardware timing source, i cant get a 100! |
| 18:29.42 | *** part/#asterisk mjordan (~mjordan@nat/digium/x-wzihadgerynlpesr) |
| 18:29.58 | *** join/#asterisk mjordan (~mjordan@nat/digium/x-cremkbnskdnxycyq) |
| 18:30.20 | *** part/#asterisk mjordan (~mjordan@nat/digium/x-cremkbnskdnxycyq) |
| 18:31.03 | *** join/#asterisk mjordan (~mjordan@nat/digium/x-fvhsywkiaxhybfog) |
| 18:37.38 | *** join/#asterisk jwiggins (~James@gateway/tor-sasl/jwiggins) |
| 18:38.15 | *** join/#asterisk garymc (~chatzilla@host109-155-155-5.range109-155.btcentralplus.com) |
| 18:43.28 | ectospasm | trumee: that's not indicative of a problem |
| 18:43.43 | ectospasm | it's quite rare to see 100% on dahdi_test |
| 18:44.00 | trumee | ectospasm: oh is it. then its fine. |
| 18:44.41 | ectospasm | trumee: yes. An average around 90% is really bad, means you've got system problems. |
| 18:46.21 | *** join/#asterisk vinhdizzo (~vinh@pool-173-60-112-55.lsanca.fios.verizon.net) |
| 18:46.37 | trumee | ectospasm: i presently have cpu frequency scaling disabled on my system. With the addition of this hardware timing source can i turn power saving on? |
| 18:47.24 | trumee | My main issue has been that my Nokia N900 phone sometimes have a choppy audio. So i am trying to tighten all loose ends |
| 18:47.58 | trumee | The softsip clients and ATA's have worked well though |
| 18:51.42 | ectospasm | trumee: I dunno how to answer that, as what you describe doesn't sound like a timing issue. |
| 18:52.23 | *** join/#asterisk Defraz (~Defraz@corp.fuzecore.com) |
| 18:55.56 | *** join/#asterisk bmoraca_work (~bmoraca@66-242-174-254.ceres.bvn.net) |
| 18:56.13 | bmoraca_work | has anyone here implemented SLA with a SIP trunk before? |
| 18:59.11 | *** join/#asterisk nny (~SM@cpe-174-107-223-014.sc.res.rr.com) |
| 19:00.46 | nny | have an asterisk install just "restarting" in the middle of some dialplan testing. This looks like a bug (1.8.3) is there any reason why the dialplan could be at fault? i would think this isn't possible unless an app is bugged in this specific version |
| 19:02.03 | p3nguin | Upgrade to the current version and test again. |
| 19:02.34 | nny | yeah my sentiments as well, thanks |
| 19:07.29 | *** join/#asterisk tzafrir_laptop (~tzafrir@bzq-218-155-170.cablep.bezeqint.net) |
| 19:11.55 | atan | What's the sound quality like on the Polycom IP 300 phone using a headset? |
| 19:16.12 | omgwtf | I have a situation where I add a cell phone to my queue through addqueuemember. Since its a cell voicemail will pickup. Is there a way to make certain queue members press a button to answer a call? |
| 19:16.35 | chuckf | trumee: the n900 is a wireless device correct? If so, it is likely the wireless connection causing the choppiness. I know I have a laptop when wired no issues, but wireless introduces choppiness at times. |
| 19:16.50 | bmoraca_work | i guess devstate is pretty much necessary to properly implement SLA using only SIP channels...hrm |
| 19:17.02 | trumee | chuckf: right |
| 19:18.00 | _Corey_ | chuckf: Dial a local channel as the queue member, then use a Dial macro M() (see 'core show help dial' for more info there) to let the called party accept |
| 19:19.52 | chuckf | omgwtf: ^^ that was for you |
| 19:20.04 | _Corey_ | whoops |
| 19:20.05 | _Corey_ | lol |
| 19:20.18 | chuckf | _Corey_: not a problem |
| 19:20.29 | _Corey_ | I'm just having my afternoon coffee, still a little fuzzy ;) |
| 19:28.00 | omgwtf | _Corey_: The queue member I'm adding is based on their callerid. Would there be anyway to do this dynamically? |
| 19:28.24 | *** join/#asterisk sulex (~sulex@pdpc/supporter/professional/sulex) |
| 19:29.14 | p3nguin | core show applications like agent |
| 19:31.36 | _Corey_ | omgwtf: Yeah, I guess you could do AddQueueMember(queuename,Local/${WHATEVER}@ack-call) |
| 19:32.19 | _Corey_ | p3nguin: He's looking to ring an agent's cell and then have them acknowledge the call |
| 19:33.04 | *** join/#asterisk BuenGenio (~Gene@cm61-10-82-188.hkcable.com.hk) |
| 19:34.48 | omgwtf | _Corey_: Wouldn't I need an entry for exten => ${WHATEVER} in extension.conf? I'd like to have zero maintenance on this..... oh nm I got you |
| 19:35.14 | *** join/#asterisk DrDigital (~mmurphy@gallery/DrDigital) |
| 19:35.16 | *** part/#asterisk _Shadowfax_ (dan@neu.cow.org) |
| 19:35.21 | _Corey_ | omgwtf: To save time, here is an example: http://pastebin.com/BLqUB0v2 |
| 19:39.06 | omgwtf | _Corey_: Thanks very much I think this solution is what I am looking for. I'll let you know how it turns out. |
| 19:39.18 | _Corey_ | Yeah, no problem |
| 19:39.42 | *** join/#asterisk tzafrir_laptop (~tzafrir@bzq-218-155-147.cablep.bezeqint.net) |
| 19:45.47 | *** join/#asterisk eduzimrs (~eduzimrs@201.22.86.124.static.gvt.net.br) |
| 19:54.58 | *** join/#asterisk evil_gordita (robert@ip70-188-50-186.rn.hr.cox.net) |
| 19:59.51 | *** join/#asterisk _Corey_ (~chatzilla@64.121.4.75) |
| 20:14.07 | *** join/#asterisk cyford (~cyford@c-24-30-15-39.hsd1.ga.comcast.net) |
| 20:16.30 | *** join/#asterisk garymc (~chatzilla@host109-155-155-5.range109-155.btcentralplus.com) |
| 20:16.53 | *** join/#asterisk gogasca (~Adium@nat/cisco/x-fydxlvtfnrvhtdtc) |
| 20:19.54 | WIMPy | StaRetji: I can't find your feature request to upload the patch to. |
| 20:21.40 | StaRetji | sorry WIMPy, had rough day, almost get fired |
| 20:22.03 | StaRetji | I looked for a minute, but could see where am I suppose to fill feature request |
| 20:22.13 | StaRetji | couldn't* |
| 20:23.30 | WIMPy | issues.asterisk.org. |
| 20:27.28 | leifmadsen | https://issues.asterisk.org/jira |
| 20:30.01 | *** part/#asterisk angryuser_laptop (~angryuser@90-156-167-83.reverse.alphalink.fr) |
| 20:41.50 | *** join/#asterisk shadowapex (~William@adsl-99-36-142-6.dsl.irvnca.sbcglobal.net) |
| 20:43.15 | shadowapex | Does anyone know by chance how to set up a failover destination in Asterisk using a2billing for a sip_buddy? |
| 20:43.51 | *** join/#asterisk grEvenX (~even@cC0FD00C3.dhcp.bluecom.no) |
| 20:43.56 | shadowapex | So if a call fails to go through to a particular sip_buddy, it tries another sip_buddy account? |
| 20:51.36 | *** join/#asterisk garymc (~chatzilla@host109-155-155-5.range109-155.btcentralplus.com) |
| 20:52.44 | *** join/#asterisk malcolmd (~malcolmd@pdpc/sponsor/digium/malcolmd) |
| 20:52.44 | *** mode/#asterisk [+o malcolmd] by ChanServ |
| 20:58.18 | *** join/#asterisk jayson_r (~jayson@cpe-071-076-046-081.sc.res.rr.com) |
| 21:03.27 | *** join/#asterisk cyford (~cyford@c-24-30-15-39.hsd1.ga.comcast.net) |
| 21:15.03 | *** join/#asterisk vinhdizzo (~vinh@dhcp-v000-219.mobile.uci.edu) |
| 21:20.14 | p3nguin | staretji: Why did you almost get fired? |
| 21:21.15 | *** join/#asterisk SuPrSluG (~SuPrSluG@8.22.96.106) |
| 21:22.59 | StaRetji | for some mysql database |
| 21:23.02 | StaRetji | not for asterisk |
| 21:23.10 | *** join/#asterisk slaz (~slaz@cpe-72-185-69-140.tampabay.res.rr.com) |
| 21:25.48 | *** join/#asterisk sjobeck (~sjobeck@173-164-83-17-Oregon.hfc.comcastbusiness.net) |
| 21:26.43 | *** join/#asterisk DrDigital (~mmurphy@gallery/DrDigital) |
| 21:28.11 | *** join/#asterisk rdahlin_1 (~rdahlin_1@78-73-19-238-no168.tbcn.telia.com) |
| 21:31.01 | *** join/#asterisk TimeRider (~steve@188-220-33-123.zone11.bethere.co.uk) |
| 21:31.32 | *** join/#asterisk rdahlin_1 (~rdahlin_1@2001:16d8:cc97:1:8508:efe9:2c89:2e58) |
| 21:32.03 | _Corey_ | Anyone going to ITEXPO in Austin next week? |
| 21:55.16 | *** join/#asterisk billmania (~bill@38.98.130.98) |
| 21:56.31 | *** join/#asterisk Micc (~Micc@c-98-232-46-178.hsd1.wa.comcast.net) |
| 21:59.58 | *** part/#asterisk shadowapex (~William@adsl-99-36-142-6.dsl.irvnca.sbcglobal.net) |
| 22:01.44 | *** join/#asterisk ipc9 (~ipc9@173-162-245-206-NewEngland.hfc.comcastbusiness.net) |
| 22:02.25 | ipc9 | is there a way to specify multiple input channels in alsa.conf? |
| 22:03.26 | *** part/#asterisk ipc9 (~ipc9@173-162-245-206-NewEngland.hfc.comcastbusiness.net) |
| 22:04.01 | *** join/#asterisk ipc9 (~ipc9@173-162-245-206-NewEngland.hfc.comcastbusiness.net) |
| 22:04.14 | ipc9 | ok |
| 22:05.17 | *** join/#asterisk Defraz (~Defraz@corp.fuzecore.com) |
| 22:05.44 | ipc9 | not sure if my last message sent. Is there a way to specify multiple input channels in alsa.conf? |
| 22:12.37 | *** join/#asterisk DrDigital (~mmurphy@gallery/DrDigital) |
| 22:19.18 | *** join/#asterisk f2knight (~ben@c-76-115-44-207.hsd1.or.comcast.net) |
| 22:20.51 | f2knight | Q: Using AEL, is there a way to do an IF statement more along the lines of something like this.... if (LEN(${myvar} !=16) {...}; ? Basiclly I am looking for a not equals seems all the docs point to using only =, >, < operators only. |
| 22:21.11 | *** join/#asterisk nix8n82-phone (~AndChat@65.161.180.230) |
| 22:24.09 | f2knight | NVM, I can just do an }else{ to catch it. |
| 22:24.56 | *** part/#asterisk ipc9 (~ipc9@173-162-245-206-NewEngland.hfc.comcastbusiness.net) |
| 22:25.21 | p3nguin | What's wrong with != in the condition? |
| 22:32.54 | *** join/#asterisk cyford (~cyford@c-24-30-15-39.hsd1.ga.comcast.net) |
| 22:37.00 | WIMPy | Can someone enlighten me if it is possible to use directmedia with SRTP? Is there anything special about that? |
| 22:37.59 | *** join/#asterisk lanning (~lanning@173-8-187-197-SFBA.hfc.comcastbusiness.net) |
| 22:49.39 | *** join/#asterisk fisted (~fisted@unaffiliated/fisted) |
| 22:53.15 | *** join/#asterisk puzzled (~patrick@puzzled.xs4all.nl) |
| 23:23.36 | f2knight | p3nguin, sorry It gave invalid syntax errors. |
| 23:25.28 | p3nguin | Care to show me exactly what you did? Using != in conditions works as expected for me and almost everyone else here. |
| 23:30.48 | *** join/#asterisk kaushal (~kaushal@14.97.11.192) |
| 23:31.23 | kaushal | leifmadsen: hi |
| 23:39.10 | kaushal | I am still unable to resolve the Error 101 issue. Can someone please suggest me how do i go about in fixing this issue ? |
| 23:43.41 | f2knight | p3nguin, the statement was if ( LEN(${myvar}) != 16) {...} |
| 23:44.02 | p3nguin | What should happen if the len of the var is not 16? |
| 23:44.40 | p3nguin | What should happen if the len of the var is 16? |
| 23:44.50 | f2knight | p3nguin, I worked around the issue by just doing this if (${ LEN(${myvar}) } = 16 ) {...}else{...} |
| 23:46.26 | f2knight | p3nguin, I was checking to see if the user entered 16 digits (the number on a credit card) if they did just continue on my app flow if not loopback and ask for it again. I got it working though. Just didnt use != prob. for the best because I can check for 2 conditions and use a verbose on the CLI in both cases so I know whats up. |
| 23:50.54 | p3nguin | exten => s,1(start),Read(whatever); |
| 23:50.55 | p3nguin | exten => s,n,GotoIf($[${LEN(${whatever})} != 16]?start); |
| 23:50.55 | p3nguin | exten => s,n,NoOp(do more stuff here); |
| 23:51.38 | *** join/#asterisk nwxi (~nwxi@client-86-23-125-82.brhm.adsl.virginmedia.com) |
| 23:51.56 | nwxi | anyone know of good sources for replacement sound samples |
| 23:54.01 | *** join/#asterisk DrDigi (~mmurphy@50-73-49-97-static.hfc.comcastbusiness.net) |
| 23:59.19 | kaushal | checking in again for the query ? |
| 23:59.26 | leifmadsen | kaushal: uhhh hi? |
| 23:59.39 | kaushal | leifmadsen: hi |