| 00:00.22 | Joel | carrar, awk '{if ($7 == "cn") { print $3 }}' country-ipv4.lst is a smidge cleaner, as well. |
| 00:00.42 | Joel | if you insist on bash :) |
| 00:00.48 | carrar | heh |
| 00:00.51 | carrar | adding |
| 00:00.51 | Joel | "bash" |
| 00:00.59 | Joel | it's really just awk. |
| 00:01.17 | carrar | I never went back to make it better, just was my inital step by step striping through the data |
| 00:01.29 | Joel | could probably pot it in a sed one liner too |
| 00:01.44 | carrar | I like CIDR blocks |
| 00:02.02 | Joel | then change $3 to $5 |
| 00:02.04 | carrar | $3 is the wrong field |
| 00:02.06 | carrar | :) |
| 00:02.25 | carrar | then I run that through the perl cisco ACL module |
| 00:02.32 | carrar | and create the actuall ACL |
| 00:03.11 | carrar | NetAddr::IP |
| 00:03.19 | carrar | works well |
| 00:03.41 | carrar | and wala, you have a nice updated ACL on your cisco router |
| 00:05.31 | jaytee | don't you mean voila? |
| 00:06.02 | carrar | I mean whatever I am thinking and don't write!! |
| 00:08.01 | carrar | I've actually set aside of IP's that are not used, and block China from those IP's and have been loggin hits and ports |
| 00:08.05 | carrar | Yesterdays Total Hits from China: 21141 |
| 00:08.40 | carrar | with port 1434 udp being the most popular yesterday with 10,698 hits |
| 00:10.14 | *** join/#asterisk pkecastillo (~pirruar@190.113.141.122) |
| 00:10.17 | carrar | and IP 122.225.100.154 doing 2,807 of those hits |
| 00:10.19 | pkecastillo | hello guys |
| 00:10.33 | *** join/#asterisk mr_ian (~mr_ian@S0106001b63f49383.du.shawcable.net) |
| 00:10.37 | carrar | HARRO |
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| 00:13.09 | *** join/#asterisk binbash_ (~peter@ip4da53781.direct-adsl.nl) |
| 00:13.27 | Joel | sed -e 's|.*:\s\(.*\)\s: cn.*|\1|' -e 's|.*[a-zA-Z].*||' country-ipv4.lst mostly there |
| 00:13.32 | Joel | probably a nicer way though |
| 00:14.05 | carrar | that spits out nothing on my box |
| 00:14.22 | carrar | just spaces |
| 00:14.23 | *** join/#asterisk diegomad (mad@190.159.87.34) |
| 00:14.23 | Joel | *shrug* |
| 00:14.54 | carrar | but points for making the line with lots of regex :) |
| 00:15.13 | *** join/#asterisk shimizu (~shimizu@87.241.161.23) |
| 00:16.06 | shimizu | Good day everyone |
| 00:16.23 | shimizu | I have a question regarding software design |
| 00:16.37 | carrar | code in assembly language |
| 00:17.42 | shimizu | hehe :) I have'to design a realtime billing on django, so is it correct to wait for cdr-s on AMI? |
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| 00:18.40 | shimizu | carrar: that would be too much :) |
| 00:20.07 | shimizu | Or should i use mongo-db instead, asterisk has support of it |
| 00:20.48 | carrar | PostgreSQL~ |
| 00:20.52 | carrar | FTW |
| 00:21.25 | hardwire | mongo-db? |
| 00:23.23 | Joel | carrar, sed -e 's|.*:\s\(.*\)\s: cn.*|\1|' -e 's|.*[a-zA-Z].*||' -e '/^$/d' country-ipv4.lst |
| 00:23.29 | Joel | not happy that's it three patterns... |
| 00:23.50 | carrar | spits out 6 # symbols |
| 00:23.55 | carrar | heh |
| 00:24.07 | Joel | may have to be tweaked for whatever format you have of this file |
| 00:24.16 | carrar | it's ok, mine works fine :) |
| 00:24.49 | Joel | replacing the space to the left of cn with a \s might make it more portable. |
| 00:25.26 | carrar | Mine's easier to read alos :) |
| 00:25.27 | carrar | also |
| 00:25.33 | *** join/#asterisk arpu (~arpu@chello062178159144.10.14.univie.teleweb.at) |
| 00:25.53 | carrar | and shorter |
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| 00:31.02 | shimizu | hardwire: yeah http://www.mongodb.org/ http://github.com/FlaPer87/cdr_mongodb |
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| 00:31.09 | *** join/#asterisk c0dyhi11 (~c0dyhi11@ip70-190-105-213.ph.ph.cox.net) |
| 00:31.36 | c0dyhi11 | Is there a way to dial outbound over a dahdi trunk straight from the CLI? |
| 00:32.02 | c0dyhi11 | I'm tryin to trouble shoot some dahdi problems and i have no clue where to begin. |
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| 00:35.44 | shimizu | What is the validation rule for sip username ? |
| 00:36.40 | c0dyhi11 | are you asking me that? |
| 00:36.59 | shimizu | if you know the answer :) |
| 00:37.00 | boodu | hello |
| 00:37.03 | carrar | c0dyhi11, like originate? |
| 00:37.24 | c0dyhi11 | originate is a command in the CLI? |
| 00:37.28 | carrar | yes |
| 00:38.46 | jaytee | is there a limit to the number of characters in a labeled priority? I would think there must be |
| 00:38.49 | *** join/#asterisk raden_work (~jon@69-179-99-17.stat.centurytel.net) |
| 00:40.09 | c0dyhi11 | it says the usage is "channel originate <tech/data> application <appname> [appdata]" |
| 00:40.16 | c0dyhi11 | I'm not sure what that means. |
| 00:40.22 | c0dyhi11 | the channel is 1 |
| 00:40.37 | c0dyhi11 | is the application the phone number? |
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| 00:41.54 | Joel | c0dyhi11, show applications |
| 00:42.16 | joobie | hmm.. Joel, any idea where the "num line keys" option is on a polycom 600 ? |
| 00:42.22 | joobie | cant seem to locate it in the web config |
| 00:42.25 | joobie | 320 had it....... |
| 00:42.41 | c0dyhi11 | show applications says "no such command" |
| 00:42.56 | Joel | joobie, don't have access to a 600 to check. |
| 00:43.08 | Joel | c0dyhi11, then I guess you have to do what people did back in the old days |
| 00:43.40 | c0dyhi11 | that doesn't sound fun. |
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| 00:43.47 | Joel | c0dyhi11, research. |
| 00:43.48 | carrar | joobie, program your polycom via ftp config files |
| 00:43.57 | carrar | not the web interface |
| 00:44.25 | Joel | carrar, p.s. my awk solution owns your stringed up mess :) |
| 00:44.36 | carrar | haha whatevah! |
| 00:44.52 | carrar | Mine is simple, shorter, easier to read :) |
| 00:45.10 | carrar | and works |
| 00:45.33 | Joel | you fail and counting then |
| 00:45.41 | Joel | and my awk solution works as well |
| 00:45.48 | Joel | s/and/at/ |
| 00:45.56 | Joel | 79 > 39 |
| 00:46.38 | coppice | nah, you'll feel greater at 39 than at 79 |
| 00:46.42 | carrar | ok here are the two lines as I least read |
| 00:46.51 | carrar | grep ": cn :" country-ipv4.lst | awk -F: '{print $2}' | sed "s/ //g" | sort -n |
| 00:46.52 | carrar | sed -e 's|.*:\s\(.*\)\s: cn.*|\1|' -e 's|.*[a-zA-Z].*||' -e '/^$/d' country-ipv4.lst |
| 00:47.23 | carrar | Your's looks 6 chars longer and isn't even sorted!! |
| 00:47.47 | Joel | mine can be golfed down to 30, and yours just to 71 with no real work |
| 00:47.54 | Joel | awk '{if($7=="cn"){print $3}}' |
| 00:47.58 | Joel | if you call sed awk |
| 00:48.02 | Joel | then I'm really worried about you |
| 00:49.53 | carrar | never mentioned that |
| 00:50.16 | Jumpie | can somebody help me out with some dahdi issues hehe |
| 00:50.27 | Jumpie | i originally had trixbox...scrapped it and reinstalled clean centos+freepbx |
| 00:50.28 | carrar | So that line is 6 characters shorter |
| 00:50.35 | Joel | carrar, indeed I did, and you even acknowledged it. If you need me to continue to own you, I will ;) |
| 00:50.43 | Jumpie | but...the issue is regarding that i dont have a /dev/dahdi, so dahdi_cfg and dahdi_Genconf dont work |
| 00:50.58 | Jumpie | i have dahditools and its latest..i think im missing somethin simple |
| 00:50.58 | Joel | Jumpie, who did you buy the card from? |
| 00:51.04 | carrar | I've never claimed to be the AWk master, simple provided a solution to a problem |
| 00:51.24 | Jumpie | from a vendor i have always before..the card is fine. its a tdm400p..and it worked with trixbox |
| 00:51.24 | carrar | Always ways to improove everyones output |
| 00:51.28 | Jumpie | the card is not the issue |
| 00:51.41 | Jumpie | i think im just missin some files |
| 00:51.50 | Joel | Jumpie thanks for jumping down my throat, but that's not the angle I was taking things. |
| 00:52.02 | Jumpie | lol sorry..just people keep sayin is the card fine..yes it is |
| 00:52.09 | Jumpie | lspci shows it correctly and functional |
| 00:52.13 | Joel | Jumpie, put the output of lsmod and dmesg on pastebin. |
| 00:52.17 | Joel | Jumpie, and lspci |
| 00:52.19 | Jumpie | k sec |
| 00:52.23 | Joel | lspci -vvv would be nice |
| 00:52.36 | Jumpie | you want full dmesg output? |
| 00:52.41 | Joel | Jumpie, would be nice. |
| 00:52.43 | Jumpie | k sec |
| 00:52.48 | Joel | lspci doesn't indicate functionality per say, either. |
| 00:52.53 | Joel | Just hardware seen on the pci bus. |
| 00:53.11 | Jumpie | true..well i guess i was sayin at least somehow/somewaht the os sees the card hehe |
| 00:53.14 | Jumpie | ill post the -vv |
| 00:53.16 | Jumpie | er vvv |
| 00:53.20 | Joel | carrar, agreed, which is why I gave you alternatives to stimulate your thought process |
| 00:53.33 | carrar | I'm stimulated |
| 00:53.41 | Joel | the more shit you string together, the more chances of something breaking |
| 00:54.38 | Jumpie | hmmm |
| 00:54.41 | Jumpie | dmesg is past my buffer |
| 00:54.48 | Joel | Jumpie, /var/log/messages |
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| 00:55.19 | Jumpie | is ther a way to 'select all' within nano or vi? |
| 00:55.23 | Jumpie | so i can paste it all into pastebin |
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| 00:57.57 | Jumpie | nm i figured out a way |
| 00:58.56 | Jumpie | http://jumpie.pastebin.com/3Ba7MBSH dmsg |
| 00:59.35 | Jumpie | http://jumpie.pastebin.com/th1KH2Wn lsmod |
| 01:01.12 | Joel | what do you see missing in the second link? |
| 01:01.44 | Jumpie | http://jumpie.pastebin.com/AraZZp9d lspci |
| 01:01.49 | Jumpie | its not there ..i nkow |
| 01:01.53 | Jumpie | says unknown module |
| 01:02.03 | Joel | what says unkown module? |
| 01:02.03 | Jumpie | yet /etc/dahdi exists..i even tried to unblacklist the device |
| 01:02.20 | Jumpie | trying to do lsmod dahdi or lsmod wcdtm |
| 01:02.30 | Jumpie | er whatever it is...i used the right one just cant remember it offhand |
| 01:02.50 | Joel | what happens when you type modprobe dahdi |
| 01:03.00 | Jumpie | whatever libraries were pulled on my trixbox install..it worked fine so i think its somethin basic im missing |
| 01:03.09 | Jumpie | fatal module dahdi not found |
| 01:03.14 | Joel | there's your issue. |
| 01:03.17 | Jumpie | so /etc/dahdi exists /dev/dahdi doesnt |
| 01:03.28 | Jumpie | but...i have the latest dahdi tools |
| 01:03.32 | Joel | so? |
| 01:03.33 | Jumpie | can i maybe blow it away redo it? |
| 01:03.43 | Jumpie | joel..i am 99.99% sure my card is fine |
| 01:03.51 | Joel | relax. |
| 01:03.53 | Jumpie | ok |
| 01:03.55 | Jumpie | sorry |
| 01:03.56 | Joel | are you done freaking out and would you like more help? |
| 01:03.58 | Jumpie | just frustrated :( |
| 01:04.05 | Jumpie | freakout mode: disabled |
| 01:04.16 | Joel | rpm -ql dahdi-tools |
| 01:04.29 | Joel | now tell me, do those looks like TOOLS or do they look like MODULES? |
| 01:04.51 | Jumpie | it looks to me all relevant paths on the system pertaining to dahdi in some way |
| 01:04.59 | Joel | now |
| 01:05.03 | Joel | revisit my question |
| 01:05.04 | Jumpie | they are files |
| 01:05.06 | Joel | and pick one of the two options |
| 01:05.11 | Jumpie | i'd say tools |
| 01:05.16 | Joel | you can do this, I believe in you. |
| 01:05.18 | Joel | great. |
| 01:05.20 | Jumpie | yay me |
| 01:05.22 | Joel | So what do you think is missing then? |
| 01:05.25 | ChannelZ | They look like yummy gumdrops to me |
| 01:05.38 | Jumpie | it looks to me, something somehow hasnt created the module? |
| 01:05.44 | ChannelZ | Yeah. Maybe you. |
| 01:05.48 | Jumpie | and the actual interface to the card isnt recognized? |
| 01:05.58 | Joel | Jumpie, and if we know the tools come from the TOOLS rpm, do you think just maybe there is an rpm for modules? |
| 01:05.59 | Jumpie | keep in mind, while im not a super linux nub, im not super guru either |
| 01:06.04 | Jumpie | joel...bingo |
| 01:06.07 | Jumpie | hah |
| 01:06.25 | Jumpie | strange they werent pulled from the getgo |
| 01:06.34 | Joel | You have gotten this solved much easier by a) keeping an open mind and not FREAKING out. b) providing a clear and concise description of the issue contains TONS of detail |
| 01:06.37 | Jumpie | joel i dont remember the exact syntax, i can search for relevance right? |
| 01:06.44 | Jumpie | yes, i appreciate it |
| 01:06.45 | Joel | yum search dahdi |
| 01:06.46 | Jumpie | and you have been patient |
| 01:07.04 | carrar | c) by eating gumdrops |
| 01:07.06 | Jumpie | ok, a lot of stuff starting with kmod-dahdilinux.... |
| 01:07.19 | Joel | Jumpie, are you using centos's repos for asterisk? |
| 01:07.26 | Jumpie | kmod-dahdi-linux.i686 i thnk? |
| 01:07.34 | Jumpie | i believe so |
| 01:07.40 | Jumpie | regular, not beta, etc |
| 01:07.41 | Joel | Jumpie, most likely, yes, try it. |
| 01:07.45 | Jumpie | k sec |
| 01:08.31 | Joel | If that doesn't work, then follow the instructions on this page: http://www.asterisk.org/downloads/yum |
| 01:08.40 | Joel | centos patches things HEAVILY |
| 01:08.57 | Jumpie | hmm |
| 01:09.03 | Joel | the 5.4 release kernel contains well over 3,000 patches against it. |
| 01:09.05 | Jumpie | Package kmod-dahdi-linux-2.3.0-1_centos5.2.6.18_164.15.1.el5.i686 already installed and latest version |
| 01:09.05 | Jumpie | Nothing to do |
| 01:09.22 | Joel | dunno, I wouldn't use any of centos's default rpms. |
| 01:09.39 | Jumpie | so the issue is the repo is probably wrong |
| 01:09.54 | Joel | The issue is you don't have kernel modules installed for dahdi that match the running kernel. |
| 01:10.15 | Joel | if you see a version in your yum search that match your running kernel, feel free to try them. |
| 01:10.15 | Jumpie | so when i did a yum update is probably when it got broken? |
| 01:10.30 | Joel | I've given you a link that should hopefully take care of it |
| 01:10.35 | Joel | on that note, good night, I am going home. |
| 01:10.42 | Jumpie | thanks |
| 01:11.01 | Joel | (if you bought the card from digium (tdm400 is old, btw) they would have been able to help you with all of this for free I believe) |
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| 01:14.43 | Jumpie | hmm ok...i have the correct repos |
| 01:14.57 | Jumpie | per that url..i already have those 2 files with the correct info |
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| 01:23.39 | carrar | jaytee |
| 01:23.46 | carrar | did you want to know the context length? |
| 01:24.42 | carrar | #define AST_MAX_CONTEXT80/*!< Max length of a context */ |
| 01:24.48 | carrar | 80 |
| 01:24.52 | Jumpie | yay it works |
| 01:24.58 | Jumpie | i had 2.6.18-164.15.1.el5PAE and the PAE version wasnt the one installed |
| 01:25.07 | carrar | include/asterisk/channel.h |
| 01:25.35 | jaytee | carrar, thanks |
| 01:25.39 | Jumpie | i kinda wish i could get rid of older versions though |
| 01:26.33 | jaytee | that's helpful but I was wondering about the length of labeled priorities. what file did you find that #define? it may have other useful info |
| 01:26.52 | carrar | I pasted the file name |
| 01:26.55 | jaytee | oh, I see the file name abovie. thanks |
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| 01:46.40 | Ropeguru | Evening all |
| 01:46.57 | Ropeguru | Anyone here that might be able to help with an AsteriskNow install issue?? |
| 01:48.05 | TJNII | Ropeguru: You should really seek help in #asterisknow |
| 01:48.12 | TJNII | ~asterisknow |
| 01:48.13 | infobot | it has been said that asterisknow is based on Asterisk, but is difficult to support in #asterisk for a number of reasons. Please seek support in #asterisknow instead. |
| 01:49.00 | Ropeguru | Thaks.. I did not see a reference to #asterisknow on their website. Only #asterisk. Will slide on over there. Thanks |
| 01:49.24 | *** part/#asterisk Ropeguru (~ropeguru@173-13-39-1-Pennsylvania.hfc.comcastbusiness.net) |
| 01:51.01 | Jumpie | can anybody help me? im a bit confused and not sure if this is beyond the scope of this channel, i have also installed freepbx and in /etc/asterisk/manager.conf i have changed the secret, thinking this is what controls the admin account |
| 01:51.13 | Jumpie | but that isnt being passed correctly, mor is the wrong account |
| 01:51.20 | Jumpie | because admin/admin is the only thing still accepted, which i cant have |
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| 01:58.16 | ChannelZ | shits on freepbx |
| 01:58.55 | pabelanger | Jumpie: #freepbx |
| 02:00.22 | TJNII | can't remember who got all butthurt last night after being told to seek support there |
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| 02:06.20 | Jumpie | pabelanger they kinda idle haha |
| 02:06.20 | Jumpie | just curious if anybody knew off hand |
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| 02:12.08 | c0dyhi11 | has anyone tried to run an analog line over Magic Jack? |
| 02:17.00 | drfreeze | Hi |
| 02:17.41 | drfreeze | Anyone know if you have to explicitly compile in echo cancellation in dahdi? |
| 02:18.16 | drfreeze | I get this error right after the call is answered |
| 02:18.16 | drfreeze | [Apr 28 21:14:50] WARNING[5452]: chan_dahdi.c:1776 dahdi_enable_ec: Unable to enable echo cancellation on channel 1 (No such device) |
| 02:21.24 | *** join/#asterisk Andras888 (~60fa1253@gateway/web/freenode/x-lciyjcihfyvitlot) |
| 02:22.00 | ChannelZ | no software ec should be there.. what are you trying to use? |
| 02:22.36 | drfreeze | http://pastie.textmate.org/private/vjfickqzqjvz3ybmerlctq |
| 02:24.12 | drfreeze | no ec should be 'where'? |
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| 02:25.47 | ChannelZ | hmm |
| 02:26.15 | ChannelZ | Sorry, I meant to say "no, software EC should be there.." |
| 02:26.56 | drfreeze | by 'there', you mean in dahdi? |
| 02:27.40 | *** join/#asterisk path (path@gateway/shell/bshellz.net/x-rzunlowbcyaukjhd) |
| 02:28.42 | ChannelZ | yes |
| 02:28.50 | ChannelZ | lsmod |grep echocan |
| 02:31.15 | drfreeze | lsmod has nothing with echo |
| 02:31.26 | drfreeze | lsmod | grep -i echo #=> nonthing |
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| 02:34.16 | Andras888 | Hi All! |
| 02:34.42 | Jed__ | hello |
| 02:35.12 | Andras888 | Hello Jed__ |
| 02:36.14 | ChannelZ | Oh really? |
| 02:36.25 | ChannelZ | lsmod |grep mg2 |
| 02:36.25 | ChannelZ | dahdi_echocan_mg2 7688 4 |
| 02:37.12 | Andras888 | I have a question regarding phone setup in Asterisk... 2 of them seems registered, but cannot call one-another... |
| 02:37.19 | Jumpie | hmm this seems like a serious issue |
| 02:38.15 | ChannelZ | Andras888: Is it setup in your dialplan? Are you getting errors or indications on the console? |
| 02:38.35 | Andras888 | sip.conf and dialplan has 2 entries each... |
| 02:39.15 | Andras888 | I get SIP channel errors for unknown reason (20)? |
| 02:39.56 | Andras888 | Must I specify the extension in the phone's web setup, or the call should complete based on the dialplan entries? |
| 02:40.18 | pabelanger | drfreeze: lsmod will tell you if your echo cancel module is loaded. If it returns nothing then you don't have it loaded. |
| 02:40.32 | pabelanger | drfreeze: IE: check your system.conf file for dahdi |
| 02:40.49 | ChannelZ | Andras888: Pastebin some ACTUAL console output |
| 02:41.18 | drfreeze | pabelanger: how do I load it? |
| 02:41.44 | pabelanger | dahdi should load to depending on your system.conf settings. |
| 02:42.46 | Andras888 | ChannelZ: I am not currently at work where the problem is, but have remote access. Is there a log file for earlier errors from the afternoon? |
| 02:42.51 | drfreeze | I guess my next question is how do I find the name? |
| 02:43.24 | pabelanger | drfreeze: It is all in the README file |
| 02:43.25 | pabelanger | http://svn.digium.com/svn/dahdi/linux/trunk/README |
| 02:51.59 | ChannelZ | drfreeze: did you build the dahdi drivers yourself? |
| 02:54.02 | drfreeze | pabelanger: hmmm, all I'm seeing is: <<_echo_cancellers,Echo cancellers>> and <<_tone_zones,tone-zones>> are |
| 02:54.05 | drfreeze | handled separately later. |
| 02:54.49 | *** join/#asterisk maxagaz (~maxagaz@soho2.i-xanadu.com) |
| 02:54.50 | drfreeze | wctdm is loaded |
| 02:55.46 | drfreeze | so far teh only solution I see online is a guy who bought a hardware echo canceller. :) |
| 02:56.15 | ChannelZ | re: did you build the drivers yourself? |
| 02:56.55 | drfreeze | yes. from 2.2.1+2.2.1 |
| 02:57.32 | ChannelZ | And do you have them turned on in /etc/dahdi/system.conf ? |
| 02:58.43 | ChannelZ | IE "echocanceller=mg2,1-4" |
| 02:58.53 | *** part/#asterisk Mhaddog (~Mhaddog@adsl-64-223-140.mia.bellsouth.net) |
| 02:59.05 | drfreeze | I have them in /etc/asterisk/chan_dahdi.conf |
| 02:59.31 | ChannelZ | echocanceller=xxx is not valid there |
| 03:00.32 | ChannelZ | You tell dahdi what canceller it should use per channel in /etc/dahdi/system.conf and then tweak parameters for cancellation in general in chan_dahdi |
| 03:01.33 | ChannelZ | fix your config, restart the drivers, it should start up the right echo cancel module |
| 03:02.16 | drfreeze | ok, ran dahdi_genconf and it generated: http://pastie.textmate.org/private/qinjpk0ynqpftegunbqtrw |
| 03:02.51 | *** join/#asterisk chilicuil (~chilicuil@unaffiliated/chilicuil) |
| 03:02.52 | ChannelZ | yes.. which is /etc/dahdi/system.conf, NOT /etc/asterisk/chan_dahdi.conf |
| 03:03.03 | drfreeze | restarted asterisk, but not drivers |
| 03:03.03 | *** join/#asterisk rossand (~aross@CPE000c413a19a3-CM00159a025ad4.cpe.net.cable.rogers.com) |
| 03:03.14 | drfreeze | ye sto /etc/dahdi/system.conf |
| 03:06.16 | ChannelZ | ..and? |
| 03:06.33 | drfreeze | [Apr 28 22:05:26] WARNING[7791]: chan_dahdi.c:1776 dahdi_enable_ec: Unable to enable echo cancellation on channel 1 (No such device) |
| 03:06.50 | ChannelZ | Could it possibly have something to do with... <drfreeze> restarted asterisk, but not drivers |
| 03:07.20 | drfreeze | I did a modprobe -r wctdm; modprobe wctdm |
| 03:07.22 | ChannelZ | When I says <ChannelZ> fix your config, restart the drivers .... |
| 03:07.38 | ChannelZ | stop asterisk, stop the drivers, restart them, see if it loads the mg2 module on its own |
| 03:08.26 | ChannelZ | as in '/etc/init.d/dahdi stop' or whatnot depending on your distro |
| 03:11.06 | drfreeze | not seeing any new moduels loaded |
| 03:11.41 | *** join/#asterisk baddragon (yiffstar66@unaffiliated/devemo) |
| 03:12.50 | ChannelZ | What does dahdi_cfg -t -v show you |
| 03:13.27 | drfreeze | ChannelZ: seems like it is fixed now |
| 03:14.31 | drfreeze | http://pastie.textmate.org/private/fxrypozqqkppksltau6jw |
| 03:14.36 | drfreeze | ChannelZ: thanks |
| 03:14.59 | *** join/#asterisk boodu (~boodu@175.158.129.128) |
| 03:15.22 | ChannelZ | sure - you really need to start the drivers with the init script as it will run dahdi_cfg for you and other magic based on config files |
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| 03:24.35 | *** part/#asterisk Andras888 (~60fa1253@gateway/web/freenode/x-findhjoikxpzmxma) |
| 03:24.58 | drfreeze | ok, it's been awhile since I have done analog, but if I wanted to use the default echo canceller, how do I enable echocan in /etc/dahdi/system.conf - is there a name for the default cancellor? |
| 03:27.20 | [TK]D-Fender | drrmg2 |
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| 03:29.14 | *** part/#asterisk jaskew (~jdaskew@netblock-75-79-182-110.dslextreme.com) |
| 03:32.17 | [TK]D-Fender | drfreeze: mg2 |
| 03:32.50 | carrar | w00t |
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| 03:46.55 | murdock_ut | Has anyone noticed that if you do a module reload and then do a dialplan reload that asterisk does weird things? |
| 03:48.09 | joobie | murdock_ut, no.. what weird things? |
| 03:48.29 | carrar | time goes backwards for me |
| 03:48.32 | Jumpie | mysql is evil |
| 03:48.40 | carrar | PostgreSQL FTW |
| 03:48.42 | Jumpie | whoever designed the syntax needs to be covered in honey and ants |
| 03:49.17 | murdock_ut | Well normally when I run dialplan reload I see a bunch of stuff. Well I don't after a module reload. Running 1.6.1.18 |
| 03:49.34 | murdock_ut | It acts like I didn't do anything. |
| 03:51.17 | murdock_ut | And when I look at the messages log I see things like this: [Apr 28 15:41:18] WARNING[24044] pbx.c: Unable to register extension '0', priority 1 in 'system_ivr', already in use |
| 03:51.54 | murdock_ut | There is an entry for every line in my dialplan that says it's already in use. |
| 03:52.08 | murdock_ut | I think that is weird. |
| 03:52.28 | [TK]D-Fender | murdock_ut: what module? |
| 03:53.03 | murdock_ut | I guess all of them since I type "module reload" |
| 03:54.04 | [TK]D-Fender | murdock_ut: that isn't a command you should be issuing blind like that. |
| 03:54.11 | [TK]D-Fender | murdock_ut: how about something SANE... |
| 03:54.35 | murdock_ut | [TK]D-Fender: must be habit from the "reload" days of 1.2 |
| 03:54.54 | [TK]D-Fender | murdock_ut: Well we still have "reload" |
| 03:56.35 | murdock_ut | If I do that it does the same thing as if I did a module reload. Things get messed up. |
| 03:58.11 | *** join/#asterisk manxpower (~ewieling@61.sub-75-254-27.myvzw.com) |
| 04:00.15 | *** join/#asterisk ChannelZ (channelz@burner.com) |
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| 05:02.29 | spenguin[work] | TEST |
| 05:05.29 | *** join/#asterisk romb (~romb@unaffiliated/romb-work/x-7222485) |
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| 05:22.28 | ChannelZ | WIN! |
| 05:23.38 | *** join/#asterisk Defraz (~Defraz@c72co-edge-router.fuzecore.com) |
| 05:29.32 | *** join/#asterisk Tim_Toady (~moi@77.49.26.91.dsl.dyn.forthnet.gr) |
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| 05:31.40 | jsjc | I did compile and installed dahdi before |
| 05:31.48 | jsjc | now I want to remove absolutely everything to start fresh... |
| 05:31.57 | jsjc | becaus ei am getting some weird errors |
| 05:32.07 | jsjc | how i can know what to delete? |
| 05:32.13 | Jumpie | lol you are the 3rd dahdi stress situation |
| 05:36.09 | *** join/#asterisk romb (~romb@unaffiliated/romb-work/x-7222485) |
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| 05:41.52 | carrar | Just re-install the OS :) |
| 05:41.58 | carrar | probably faster |
| 05:42.30 | *** join/#asterisk Defraz (~Defraz@c72co-edge-router.fuzecore.com) |
| 05:43.07 | carrar | All these people suffering from DSS (Dahdi Stress Syndrome) |
| 05:43.13 | carrar | DSDS |
| 05:43.27 | carrar | DSDS (Dahdi Stress Disorder Syndrome) |
| 05:44.35 | Jumpie | haha |
| 05:44.40 | Jumpie | jsjc..is the card detected? |
| 05:44.45 | Jumpie | is the service started, etc |
| 05:44.51 | Jumpie | can you run dahdi_config -v ? |
| 05:45.06 | jsjc | Jumpie it was not now |
| 05:45.10 | jsjc | so I want to clear that old version |
| 05:45.15 | jsjc | and install fresh 2.3.0 |
| 05:45.26 | Jumpie | well i had issues with an older version |
| 05:45.29 | Jumpie | i really didnt 'clear' it |
| 05:45.35 | Jumpie | but i downloaded the correct one |
| 05:45.42 | Jumpie | what is your kernel version? |
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| 06:33.00 | ChannelZ | whispers "make uninstall" |
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| 06:40.19 | Imo | hello, i have installed an OpenVZ Centos and i have installed asterisk. but asterisk dont lunch on start up. i have set the runlevel on 3 and 5. in normaly centos works well but. what can i do ? |
| 06:44.49 | *** join/#asterisk AndyF (~a@212.248.238.62) |
| 06:45.54 | Imo | can somebody help me ? |
| 06:50.25 | Tim_Toady | Imo run the init script by hand and see if thes some error |
| 06:50.48 | Tim_Toady | u installed from source? |
| 06:51.29 | *** join/#asterisk ChannelZ (channelz@burner.com) |
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| 08:04.58 | *** join/#asterisk Polysics (~luca@host113-41-static.25-87-b.business.telecomitalia.it) |
| 08:05.00 | Polysics | hello |
| 08:05.24 | Polysics | anyone knows why when i do a call and log it on CDR, the timer starts from when the phone rings? |
| 08:05.40 | Polysics | instead of when the call is actually picked up? |
| 08:05.52 | Polysics | might have something to do with using the "m" option on Dial? |
| 08:06.56 | carrar | You looking at duration or billsec? |
| 08:09.35 | Polysics | billsec |
| 08:09.39 | Polysics | since it made sense :-) |
| 08:09.46 | carrar | don't answer the channel |
| 08:09.50 | carrar | remove the MOH |
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| 08:10.50 | Polysics | hmm, then how do i provide the message to the user? |
| 08:10.57 | Polysics | i am required to play that message |
| 08:11.08 | carrar | what message |
| 08:11.09 | Polysics | but then again, i suppose that the message answers the channel |
| 08:11.22 | Polysics | "you are waiting to be connected to the desired operator" |
| 08:11.34 | carrar | yeah the call is connected at that point |
| 08:12.52 | Polysics | so i suppose CDR is useless for me, and I have to use some sort of AMI parsing |
| 08:13.18 | carrar | or create your own logfiles |
| 08:13.20 | Polysics | also because i could use the CDR event to at least detect the call's end, but it would not tell me anything about if it was answered or not |
| 08:13.22 | gelo | in cdr you have a field with the call duration from the moment when it were answered |
| 08:13.53 | Polysics | gelo, yes, but since I have MoH on the channel, it starts from when MoH starts, that is, at the very beginning |
| 08:14.55 | *** join/#asterisk denon (~denon@synapse.subneural.net) |
| 08:14.55 | *** mode/#asterisk [+o denon] by ChanServ |
| 08:15.22 | gelo | ok, then you must create your own logfiles, like carrar said |
| 08:15.35 | Imo | i get this error Starting asterisk: Cannot find specified TTY (9) |
| 08:15.35 | Imo | <PROTECTED> |
| 08:17.28 | Polysics | how do i create my own logfiles? AMI? |
| 08:18.43 | carrar | What do you want to log? |
| 08:18.45 | *** join/#asterisk Sargun (~Sargun@atarack/Staff/Sargun) |
| 08:18.55 | carrar | or track |
| 08:19.45 | Polysics | i need to log the duration of calls coming in both to direct numbers and to queues that have the same people in them |
| 08:19.53 | Polysics | this is a translation service |
| 08:20.32 | Polysics | people either know the code of the single person they want to talk to, or just ask for a language they need translation for |
| 08:21.21 | carrar | probably want to parse AMI events |
| 08:21.25 | Polysics | a secondary function i will have to sort out somehow is: if an operator is talking on a queue and gets called directly, is there a way to enqueue that call to that operator? |
| 08:21.43 | Polysics | yes, i probably need to react to bridge and hangup events |
| 08:22.26 | Polysics | the single-user queue thing is probably more complicated |
| 08:22.49 | *** join/#asterisk romb (~romb@unaffiliated/romb-work/x-7222485) |
| 08:26.06 | gelo | you could use local channel in queue, set the moment when the dial is done in cdr |
| 08:28.12 | Polysics | yes, but it will still start the timer when it dials, not when it is answered |
| 08:28.15 | Polysics | or not? |
| 08:29.28 | carrar | M(x): Executes the macro (x) upon connect of the call |
| 08:31.51 | gelo | or G() if you prefer |
| 08:36.38 | *** join/#asterisk raj-darkmystery (~test@114.143.184.114) |
| 08:36.48 | raj-darkmystery | hi friends |
| 08:36.57 | carrar | HARRO |
| 08:37.01 | raj-darkmystery | need some help with asterisk troubeshooting |
| 08:37.52 | raj-darkmystery | for outgoing calls cant here anything untill 28 sec passed, specially unable to here outgoing ring |
| 08:38.24 | *** join/#asterisk dandre (~daniel@ble59-2-81-56-122-47.fbx.proxad.net) |
| 08:38.57 | *** join/#asterisk dandre (~daniel@ble59-2-81-56-122-47.fbx.proxad.net) |
| 08:39.27 | raj-darkmystery | anyone?? |
| 08:41.21 | raj-darkmystery | hello expertise... need a li'l help |
| 08:47.52 | *** join/#asterisk joobie (~joobie@CPE-124-180-8-218.lns1.lon.bigpond.net.au) |
| 08:48.56 | spenguin[work] | raj-darkmystery: check the console |
| 08:48.59 | spenguin[work] | logs |
| 08:49.02 | spenguin[work] | whats happening |
| 08:49.23 | raj-darkmystery | hi spenguin[work] |
| 08:49.47 | raj-darkmystery | thanks for your resonse.. actually i am facing some problem with ringing |
| 08:50.19 | raj-darkmystery | if anyone calls out then no one can hear if its ringing or not.. if not answered then after 28 sec can hear ringing volume |
| 08:50.54 | raj-darkmystery | hope that you are getting what problem i am facing spenguin[work] |
| 08:52.19 | *** join/#asterisk gr0mit (~tim@extrt.txrx.org.uk) |
| 08:53.39 | raj-darkmystery | are you there spenguin[work] ? |
| 08:55.49 | *** join/#asterisk c0rnoTa (~c0rnoTa@178.176.143.130) |
| 08:56.27 | Jumpie | hey guys |
| 08:56.45 | Jumpie | is there a particular 'faxing' module one should get? |
| 08:57.05 | Imo | i get this error on startup "Starting asterisk: Cannot find specified TTY (9)" what can i do ? |
| 09:00.01 | *** join/#asterisk TimeRider (~steve@89.242.206.56) |
| 09:04.31 | *** join/#asterisk raj-darkmystery (~test@114.143.184.114) |
| 09:04.54 | raj-darkmystery | anyone here.. need li'l help with asterisk |
| 09:05.27 | petern_ | press shift-8? |
| 09:06.22 | spenguin[work] | raj-darkmystery: but you dont hear it ringing, does it right? |
| 09:06.28 | spenguin[work] | s/right/ring |
| 09:07.05 | raj-darkmystery | yes spenguin[work] .. i cant here if its ringing or not but if the person receives the call then i can here his/hers voice properly |
| 09:07.38 | raj-darkmystery | issue is i just cant here the ring for outgoing calls |
| 09:07.49 | *** join/#asterisk cjk (~cjk@85.93.217.128) |
| 09:08.07 | raj-darkmystery | i can here ring after 28 secs |
| 09:08.16 | cjk | hi, does anyone know a basic wlan voip phone that works (battery life) and maybe transfers |
| 09:08.46 | Jumpie | i bought one of these free fax for asterisk licenses off digium's site for "zero dollars" but...all i get is a confirmation and thank you email...am i supposed to get some key shortly? |
| 09:09.02 | Jumpie | lol..i was impatient |
| 09:09.12 | *** join/#asterisk sergey (~sergey@ua0zeh.iks.ru) |
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| 09:09.51 | raj-darkmystery | hi kaldemar |
| 09:10.08 | raj-darkmystery | hope u still can remember me |
| 09:11.01 | raj-darkmystery | 'm again facing last times problem.. ringing after 28 secs :-/ calling is fine but cant here outgoing calls ring |
| 09:11.17 | spenguin[work] | eh, sorry Im just buried under work "[ |
| 09:11.18 | spenguin[work] | :p |
| 09:11.46 | raj-darkmystery | thats fine spenguin[work] at least you tried to help.. thanks for that |
| 09:12.18 | raj-darkmystery | but let me know spenguin[work] , if you figure out the cause of the problem |
| 09:12.51 | raj-darkmystery | kaldemar, are you there? :-/ |
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| 09:14.08 | *** join/#asterisk dinesh___ (~dinesh@84-73-120-175.dclient.hispeed.ch) |
| 09:14.39 | dinesh___ | hi folks, is it possible to specify different allowed codecs for different "register => .." entries in sip.conf ? |
| 09:14.51 | *** join/#asterisk romb (~romb@unaffiliated/romb-work/x-7222485) |
| 09:15.07 | dinesh___ | i have on incoming number that handles g729,gsm,g726,ulaw,alaw, but another just ulaw,alaw |
| 09:15.27 | dinesh___ | so i'd rather no have to limit the codecs in [general] |
| 09:15.34 | dinesh___ | rather not* |
| 09:15.46 | *** join/#asterisk Z_God (~julius@wlan234127.mobiel.utwente.nl) |
| 09:16.11 | raj-darkmystery | problem with outgoing call ring.. anyone knows how to solve the issue? |
| 09:20.47 | Jumpie | so can you actually use gmail servers for outgoing email options in asterisk? or is it assumed you are the admin of some corporate email server? |
| 09:22.37 | *** join/#asterisk mr_ian (~mr_ian@S0106001b63f49383.du.shawcable.net) |
| 09:24.13 | raj-darkmystery | <PROTECTED> |
| 09:24.36 | gelo | Jumpie asterisk just sends mail, doesn't bother about how your mail server is configured |
| 09:24.47 | gelo | so you can use whatever you want |
| 09:25.00 | Jumpie | gelo but the problem is how some ISPS allow you do send out on smtp |
| 09:25.05 | Jumpie | i.e. being cox/comcast |
| 09:25.28 | gelo | that's not an asterisk problem |
| 09:25.33 | gelo | but a network problem |
| 09:26.02 | Jumpie | true |
| 09:26.04 | Jumpie | :) |
| 09:26.18 | Jumpie | so it just needs an smtp host/password |
| 09:26.19 | Jumpie | thast it |
| 09:26.27 | Jumpie | its up to you to be sure its possible form your network/isp |
| 09:26.29 | spenguin[work] | hey, Ive been trying to understand wat g1,g2 means in this case |
| 09:26.38 | Jumpie | spenguin[work] , groups? |
| 09:26.39 | spenguin[work] | Dial(DAHDI/g1/${CALLERID(dnid):1},40,Ttg) |
| 09:26.49 | spenguin[work] | whats g1, g2? |
| 09:26.49 | Jumpie | its a group |
| 09:26.53 | spenguin[work] | group of? |
| 09:26.54 | Jumpie | which is a collection of channels |
| 09:27.03 | spenguin[work] | oh ok |
| 09:27.18 | gelo | in fact, 1 or 2 is the group |
| 09:27.19 | Jumpie | look at /etc/asterisk/dahdi-channels.conf |
| 09:27.22 | spenguin[work] | channel =>1-15,17-31 |
| 09:27.27 | spenguin[work] | that grouping? |
| 09:27.31 | gelo | g means the way it dials the group |
| 09:27.32 | Jumpie | also check out your trunks, and how you ahve it set |
| 09:27.53 | Jumpie | gelo but you can specify indiv channels too tho if i recall |
| 09:28.06 | gelo | of course |
| 09:28.09 | spenguin[work] | bchan=1-15,17-31 ? |
| 09:28.23 | gelo | but you can make dial(DAHDI/G1/.... |
| 09:28.23 | spenguin[work] | so thats two groups? |
| 09:29.06 | gelo | and that would be dialing the group's channels from uppest (if this word exists) to lowest |
| 09:29.07 | raj-darkmystery | <PROTECTED> |
| 09:30.42 | gelo | ...highest would have been better :P |
| 09:31.22 | Jumpie | lol |
| 09:35.27 | *** join/#asterisk OrNix (~ornix@178.49.0.149) |
| 09:36.04 | *** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/atheme.member.chainsaw) |
| 09:38.50 | dinesh___ | heh weird ulaw sounds actually pretty good |
| 09:39.22 | dinesh___ | perhaps better than g729 , gsm and g726 |
| 09:39.49 | Chainsaw | That's expected, yes. |
| 09:40.10 | Chainsaw | ulaw is only barely compressed and it uses a fair amount of bandwidth. |
| 09:40.26 | dinesh___ | cool then |
| 09:40.31 | Jumpie | are default allows decent? |
| 09:40.41 | Jumpie | i cant remember if ulaw is allowed by default on extensions |
| 09:40.47 | dinesh___ | i'm only interested in quality |
| 09:40.56 | Jumpie | important to me too |
| 09:40.57 | Jumpie | hehe |
| 09:42.51 | dinesh___ | but in the general case, more bandwidth doesn't necesseraly mean better quality |
| 09:43.23 | dinesh___ | old codecs might be way worse than newer ones, but in this case ulaw is indeed great |
| 09:51.11 | *** join/#asterisk romb (~romb@unaffiliated/romb-work/x-7222485) |
| 09:51.32 | Jumpie | im curious if there is a way to see a voicemail for another extension on a phone...im sure there is |
| 09:51.47 | Jumpie | the idea is..a house based system only has one did...and its gonna ring all 7 extensions as a ringgroup |
| 09:51.55 | Jumpie | nobody is gonna 'leave a message' for the kitchen or for the basement, etc |
| 09:52.05 | Jumpie | so i want any given phone to be able to see if a general voicemail has been left |
| 09:52.08 | Jumpie | this a pain to setup? |
| 09:53.18 | gelo | you mean mwi? |
| 09:53.59 | *** join/#asterisk sulex (~sulex@95.236.116.143) |
| 09:54.06 | Jumpie | yea but can you query that on an extension you arent registered on? |
| 09:54.08 | *** join/#asterisk ickmund (~magnus@cli-5b7ee15c.bcn.adamo.es) |
| 09:54.17 | Jumpie | like, im deferring the 'house' general voicemail to lets say extension 501 |
| 09:54.30 | Jumpie | but i'd like any phone in the house to see if there is a voicemail on that extension |
| 09:54.36 | Jumpie | simple, or funky php script? |
| 09:54.55 | *** join/#asterisk kruemeltee (~Maddin@port-92-198-62-82.static.qsc.de) |
| 09:55.02 | kruemeltee | hello everybody |
| 09:55.14 | gelo | afaik asterisk just sends a notify to the extension with the same number as the given voicemail |
| 09:55.38 | kaldemar | Jumpie: a simple extension. core show application VoiceMailMain. |
| 09:55.50 | kruemeltee | is there any way to get the SIP number of the agent wo takes the call from the queue within the filename of MixMonitor? |
| 09:55.51 | gelo | if you want further functionality, you must use "externnotify" option, which allows you to execute a script/program |
| 09:56.23 | gelo | as asterisk does not support publish |
| 09:56.43 | gelo | talking about sip |
| 09:59.32 | *** join/#asterisk mikkel (~mikkel@84-238-113-66.u.parknet.dk) |
| 10:00.51 | *** join/#asterisk Jumpie (~lah@c-76-111-80-110.hsd1.va.comcast.net) |
| 10:01.30 | Jumpie | comcast fart |
| 10:03.01 | *** join/#asterisk jaytee (~jforde051@unaffiliated/jaytee) |
| 10:03.47 | *** join/#asterisk vader-- (~me@c-71-225-201-226.hsd1.nj.comcast.net) |
| 10:07.10 | *** join/#asterisk dr_gogeta86 (~fisgro@host82-53-static.58-88-b.business.telecomitalia.it) |
| 10:08.54 | *** join/#asterisk raj-darkmystery (~test@114.143.184.114) |
| 10:09.04 | raj-darkmystery | problem with outgoing call ring.. |
| 10:09.14 | raj-darkmystery | have googled a lot bt no luck |
| 10:11.23 | *** join/#asterisk pinoyskull_ (~pinoyskul@122.55.80.205) |
| 10:11.28 | *** join/#asterisk Faithful (~Faithful@202.6.145.116) |
| 10:11.37 | dinesh___ | so what? |
| 10:11.44 | dinesh___ | what you're saying doesn't make any sense |
| 10:13.40 | Jumpie | lol |
| 10:17.38 | *** join/#asterisk hehol (~hehol@buero-gw.dortmund.loca.net) |
| 10:21.35 | *** join/#asterisk af_ (~getsmart@88-149-240-255.dynamic.ngi.it) |
| 10:22.45 | spenguin[work] | from the asterisk console is it possible to make calls as a certain user? |
| 10:23.39 | *** join/#asterisk UQlev (~yuriy@212.50.99.8) |
| 10:24.30 | Jumpie | may have to user some perl script |
| 10:25.42 | Jumpie | https://issues.asterisk.org/view.php?id=5973 |
| 10:25.48 | Jumpie | see if ther is some guidance ther for ya |
| 10:26.22 | kaldemar | spenguin[work]: yes if you make an extension that modifies the caller id. |
| 10:27.26 | kaldemar | help console dial will tell some more about dialing from command line. |
| 10:32.20 | spenguin[work] | kaldemar: Im on asterisk 1.6.2 |
| 10:43.39 | *** join/#asterisk ccesario (~ccesario@189-19-6-236.dsl.telesp.net.br) |
| 10:49.04 | *** join/#asterisk puzzled (~patrick@puzzled.xs4all.nl) |
| 10:54.12 | *** join/#asterisk romb (~romb@unaffiliated/romb-work/x-7222485) |
| 10:56.04 | *** join/#asterisk daya (~daya@113.199.168.185) |
| 10:57.43 | daya | My asterisk server is hosted in one of the public IP, the clients(PAP2 devices) are behind the nat firewall, the phone only ring but I can't hear any audio |
| 10:57.59 | daya | I have set nat=route in sip.conf |
| 10:58.13 | daya | Is there is any extra config in sip.conf |
| 11:00.11 | gelo | canreinvite=no |
| 11:03.59 | kaldemar | ~sipnat |
| 11:04.00 | infobot | [~sipnat] Quick guide on configuring * + SIP behind NAT : http://www.aocomputing.net/?p=3 , otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
| 11:04.04 | kaldemar | daya: ^^ |
| 11:04.08 | *** join/#asterisk zorp75ck (~zorp75ck@pool-71-162-42-177.altnpa.east.verizon.net) |
| 11:04.40 | daya | gelo, isn't it default |
| 11:05.31 | gelo | i don't trust defaults, i check everything myself |
| 11:06.03 | joobie | anyone played with http://www.polycom.com/products/telepresence_video/video_conference_systems/personal_systems/vvx1500d.html ? |
| 11:06.52 | *** join/#asterisk hc_e (~hc@pdpc/supporter/active/hc-e) |
| 11:06.54 | hc_e | hi |
| 11:06.54 | daya | gelo, ok I set careinvite=no, do I set nat=route or yes |
| 11:07.09 | leifmadsen | canreinvite=yes is the default |
| 11:08.05 | hc_e | I've a question about chan_sip of the latest asterisk rc (1.6.2.7-rc2). I'm running it on an openvz guest and notice that under certain conditions the usage of 'dgramrcvbuf' rises significantly. |
| 11:08.34 | hc_e | I suspect this happens when the connection between asterisk and sip client is poor / suffers from high packet loss. Can anyone confirm this? |
| 11:09.32 | gelo | yeah, canreinvite=yes makes more sense as default value |
| 11:10.19 | daya | gelo, My asterisk server is in public IP, and pap clients are behind nat, do I need to set nat=yes in sip.conf |
| 11:10.54 | Gido-E | daya, does it work without? |
| 11:12.38 | daya | Gido-E, Its not working in both cases |
| 11:12.54 | kaldemar | daya: check the guide above. you only need nat=yes for the clients in sip.conf, not under general. |
| 11:13.09 | daya | Gido-E, but when I put nat=yes, the ring cames but no conversation |
| 11:14.45 | Gido-E | daya ok, then the RTP stream is not good forwarded or blocked. Check your firewalls and NAT helpers. |
| 11:15.21 | *** join/#asterisk raj-darkmystery (~test@114.143.184.114) |
| 11:16.10 | *** join/#asterisk ChaosDragon (~bjastles@67.204.47.157) |
| 11:17.07 | raj-darkmystery | trying to deploy asterisk server over centos 5.1, if i fire make for dahdi-linux, getting following error.. not getting how to resolve the issue :-/ |
| 11:17.27 | *** join/#asterisk stmaher (~stephen@80.68.89.200) |
| 11:17.27 | raj-darkmystery | You do not appear to have the sources for the 2.6.18-164.el5PAE kernel installed |
| 11:17.30 | stmaher | hi guys.. |
| 11:18.01 | stmaher | is there a way in asterisk to force rtp through asterisk? and bind the sdp outbound from the asterisk box to tell rtp to goto a specific ip address? |
| 11:18.03 | gelo | raj-darkmystery try installing the sources... |
| 11:18.15 | joobie | guys anyone been able ot integrate skype video into asterisk? |
| 11:18.27 | *** join/#asterisk Pazzo (~ugelt@reserved-225136.rol.raiffeisen.net) |
| 11:18.29 | joobie | want to have a handset registered on my asterisk box but the option to skype video call.. just duno if it's possible |
| 11:18.29 | stmaher | im getting an issue where asterisk is issuing out the public IP address of the company and not its internal 10.55.7.1 address |
| 11:18.30 | raj-darkmystery | gelo, i already did that have installed kernel-PAE-devel but still the same error |
| 11:18.32 | stmaher | in the sdp |
| 11:18.36 | joobie | found a page on skype saying that they do SIP |
| 11:18.37 | joobie | but not video |
| 11:18.57 | gelo | did you update kernel recently? did you reboot the machine? |
| 11:19.18 | raj-darkmystery | yes i did |
| 11:20.00 | raj-darkmystery | or if anyone can provide with accurate instructions for deployment, gelo |
| 11:20.24 | gelo | sorry, i'm a debian user |
| 11:21.10 | raj-darkmystery | gelo, me also.. thtz why got stuck :( |
| 11:21.28 | *** join/#asterisk torrio (~f3k@p54A2AAD7.dip.t-dialin.net) |
| 11:21.49 | raj-darkmystery | anyone can help out with this issue or with proper instructions over asterisk deployment with dahdi using redfone |
| 11:22.00 | raj-darkmystery | issue: You do not appear to have the sources for the 2.6.18-164.el5PAE kernel installed |
| 11:23.06 | *** join/#asterisk TimeRider (~steve@109.224.131.68) |
| 11:23.36 | torrio | hey guys, is it possible to run asterisk and softphone still for development issues on one host ? |
| 11:24.38 | ChaosDragon | hello all, any one have experiance writing dsp code ... |
| 11:27.34 | raj-darkmystery | kaldemar, are you there? |
| 11:31.45 | coppice | statistics suggest about 50,000 people have experience writing dsp code |
| 11:32.51 | leifmadsen | and what percentage of that were successful at it? :) |
| 11:33.20 | eject_ck | Hi all |
| 11:33.36 | ChaosDragon | ok so let me rephrase is there anyone that can point me to documentation on writing some new dsp for an asterisk ap ;) |
| 11:33.38 | eject_ck | I'm getting warning "[Apr 29 13:32:57] WARNING[29967]: res_rtp_asterisk.c:1912 ast_rtp_read: RTP Read too short" when I'm trying to send FAx |
| 11:33.50 | leifmadsen | torrio: yes -- run on different SIP port |
| 11:33.54 | leifmadsen | ports* |
| 11:33.56 | *** join/#asterisk The-Bat (~The-Bat@59.162.86.164) |
| 11:34.06 | leifmadsen | (i.e. asterisk on 5060 and the phone on 5061) |
| 11:34.23 | leifmadsen | heads off to work on the 3rd edition of TFoT |
| 11:34.33 | coppice | i don't think they gather stats like that. surveys don't usually ask "do you do X?" and "are you incompetant?" |
| 11:35.58 | coppice | eject_ck: you are probably getting UDPTL on your RTP port |
| 11:39.05 | daya | Gido-E, what are the change that I need to set in pap2 linksys adaper if asterisk is in public ip and client are behind the nat |
| 11:39.22 | daya | Gido-E, I have set nat=yes in sip.conf |
| 11:39.32 | fenrus | <PROTECTED> |
| 11:40.51 | coppice | ChaosDragon: You're question seems vague |
| 11:42.07 | ChaosDragon | I would like to extend the apt_rpt app by adding a few new funtions to it that would require some dsp |
| 11:42.46 | coppice | are you a DSP engineer, or are you looking for one? |
| 11:43.18 | *** join/#asterisk thebaddragon (yiffstar66@unaffiliated/devemo) |
| 11:45.03 | hc_e | my problem seems to be that the asterisk rtp code doesn't call recvfrom(2) in time in some situations, so the dgram buffer fills up until it reaches its limit, temporarily making the whole openvz guest offline |
| 11:45.24 | hc_e | I don't know what to do - this problem seems to be present even in the current release candidate - any suggestions? |
| 11:45.42 | ChaosDragon | depends on how complicated it is I always like to learn new skills |
| 11:45.54 | *** join/#asterisk [Outcast] (~james@64.202.62.5) |
| 11:45.59 | [Outcast] | I was wondering if someone could look at http://pastebin.com/BDT5A4yg and tell me why asterisk will not obey the 302? it should send another reinvite to the contact in the redirect message right? |
| 11:46.03 | [Outcast] | or do I need to modify the to URI as well? |
| 11:46.24 | *** join/#asterisk nitram (foo@superblob.com) |
| 11:46.41 | *** join/#asterisk rossand (~aross@CPE000c413a19a3-CM00159a025ad4.cpe.net.cable.rogers.com) |
| 11:46.52 | coppice | ChaosDragon: so you are really looking for a book on basic DSP? |
| 11:47.35 | ChaosDragon | that or example code |
| 11:48.30 | eject_ck | coppice: "you are probably getting UDPTL on your RTP port" what does it mean for me ? |
| 11:50.23 | eject_ck | I have Asterisk behind firewall and forwarded all the needed ports |
| 11:51.29 | eject_ck | Can someone help with my problem http://pastebin.com/FjGh5axc |
| 11:51.37 | eject_ck | there is verbose log for my session |
| 11:52.22 | eject_ck | I have extension in extensions.conf and call file which I'm copying to spool/outgoing then call invokes and I see teh log records |
| 11:54.09 | coppice | ChaosDragon:example code demands a specific requirement. A good beginner's book is understanding dsp by Richard Lyons |
| 11:54.33 | *** part/#asterisk hc_e (~hc@pdpc/supporter/active/hc-e) |
| 11:54.54 | coppice | eject_ck: UDPTL means T.38 packets |
| 11:55.37 | *** join/#asterisk gr0mit (tim@corerouter-29bh.hi-wifi.co.uk) |
| 11:56.14 | eject_ck | coppice: yes I know :), but why I see teh warnings ? |
| 11:56.37 | eject_ck | Btw, I have my * box behind NAT. Can this be a problem ? |
| 11:56.42 | ChaosDragon | I don't know how familier you are with radio there is a tone control method we use that is called CTCSS and another one DCS I would like to add this detection in to app_rpt |
| 11:57.55 | *** join/#asterisk [TK]D-Fender (~chatzilla@216.191.106.163) |
| 12:04.11 | *** join/#asterisk ChrisWi (~admin@mx2.wwserver.net) |
| 12:05.45 | *** join/#asterisk maximo (~maximo@CPE001217b1920e-CM00111ade9528.cpe.net.cable.rogers.com) |
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| 12:07.49 | eject_ck | There is all of my configs and call file |
| 12:07.49 | eject_ck | http://pastebin.com/4c3XFcmN |
| 12:08.23 | eject_ck | I'm getting call on fax machine (where I wanna to SendFax) |
| 12:08.33 | eject_ck | then nothing happens |
| 12:09.09 | *** join/#asterisk nighty^ (~nighty@x122091.ppp.asahi-net.or.jp) |
| 12:09.11 | coppice | basic tone detection is pretty easy. robust detection rather less so. good luck |
| 12:10.06 | *** join/#asterisk kamh (~kamh@xdsl-1814.wroclaw.dialog.net.pl) |
| 12:10.30 | *** join/#asterisk agx (~Antonio@host63-216-static.34-88-b.business.telecomitalia.it) |
| 12:11.50 | agx | i've 2 SIP accounts registered with the same "host=xxx" BUT incoming call are always matched using the "host" value; instead how do i config sip.conf to match INVITEs using the phone number (fromuser) ? |
| 12:11.54 | ChaosDragon | It would be similar to the DTMF detetion |
| 12:14.20 | coppice | maybe. it depends on the actual tones, and what else might be present with the tones |
| 12:16.07 | *** join/#asterisk rttrey (~trey@andc-office-fw.atlantic.net) |
| 12:18.05 | *** part/#asterisk rttrey (~trey@andc-office-fw.atlantic.net) |
| 12:20.23 | *** join/#asterisk Professional (~Pro@unaffiliated/shani) |
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| 12:42.00 | agx | does "type=user" in sip.conf works in a differents from 1.4 to 1.6? |
| 12:42.12 | agx | different way* |
| 12:42.12 | *** join/#asterisk HorizonXP (~xitij@69-165-161-132.dsl.teksavvy.com) |
| 12:46.29 | *** join/#asterisk krion (~seb@unaffiliated/krion) |
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| 12:58.01 | highvoltz | Is it possible to assign a different ringtone to a specific incoming DID? |
| 12:58.09 | *** join/#asterisk Mhaddog (~Mhaddog@adsl-64-223-140.mia.bellsouth.net) |
| 12:59.01 | *** join/#asterisk Mhaddog (~Mhaddog@adsl-64-223-140.mia.bellsouth.net) |
| 12:59.23 | *** part/#asterisk Mhaddog (~Mhaddog@adsl-64-223-140.mia.bellsouth.net) |
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| 13:00.30 | [Outcast] | http://www.opensips.org/Resources/DocsTutRedirect |
| 13:01.12 | *** join/#asterisk Mhaddog (~Mhaddog@adsl-64-223-140.mia.bellsouth.net) |
| 13:01.52 | *** join/#asterisk Mhaddog (~Mhaddog@adsl-64-223-140.mia.bellsouth.net) |
| 13:02.11 | *** part/#asterisk Mhaddog (~Mhaddog@adsl-64-223-140.mia.bellsouth.net) |
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| 13:04.02 | Scorcerer | how can i enable users to text-chat through asterisk ? |
| 13:04.21 | [Outcast] | oops wrong room |
| 13:04.33 | *** join/#asterisk muiro (~muiro@unaffiliated/muiro) |
| 13:07.48 | *** join/#asterisk waa (~waa@balrog.credipar.com.br) |
| 13:11.14 | *** join/#asterisk spiceycurry (~mcurry@proxy.hostopia.com) |
| 13:12.25 | spiceycurry | I am doing a "sip show peers" and it is showing 1 online - however my x-lite softphone cannot connect ot the asterisk server (getting a registration error: 408 request time out). So how could 1 sip be online? |
| 13:13.18 | [Outcast] | spiceycurry: are you behind nat? |
| 13:13.43 | spiceycurry | hmm, it is on the local network. however, let me check my mac firewall settings |
| 13:13.48 | agx | spiceycurry: when you go offline without unregistering then you're online untile asterisk send you an OPTIONS (qualify=yes) and you don't reply |
| 13:15.58 | spiceycurry | Here are my logs: http://pastebin.com/4GKcdquK |
| 13:16.14 | spiceycurry | says failed to write on 34 broken pipe |
| 13:16.21 | spiceycurry | (read the bottom portion) |
| 13:16.51 | *** join/#asterisk Arsenick (~y@69.70.231.230) |
| 13:17.45 | spiceycurry | very strange |
| 13:19.22 | spiceycurry | is this a DNS prob, or could it just be my settings? |
| 13:20.40 | *** join/#asterisk waa (~waa@balrog.credipar.com.br) |
| 13:21.23 | *** join/#asterisk brad_mssw (~brad@shop.monetra.com) |
| 13:23.16 | spiceycurry | Getting a registration error: 408 - Request Timeout, here is my log. http://pastebin.com/4GKcdquK What should I try? |
| 13:24.39 | Arsenick | Hi all, we have a 1800 number on our PSTN line and this 1800 line go in the queue setup by our provider, I would like to know if there's a way to match a pattern and do something special with call from this line, I mean is it possible to know this call come from the 1800 line ? even if the call enter in the same zap as the other calls ? |
| 13:25.32 | *** join/#asterisk LND (~LND@89.192.172.70) |
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| 13:26.12 | *** mode/#asterisk [+o malcolmd] by ChanServ |
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| 13:27.04 | *** part/#asterisk muiro (~muiro@unaffiliated/muiro) |
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| 13:30.09 | eject_ck | I see in messages T38FaxUdpEC in udptl.conf is no longer supported; use the t38pt_udptl configuration option in sip.conf instead. |
| 13:31.34 | eject_ck | where get description of available options for sip.conf and udptl ? |
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| 13:35.19 | *** join/#asterisk bipolar (~bipolar@offsitesysadmin.com) |
| 13:36.42 | leifmadsen | eject_ck: look in the sample files (<asterisk_src>/configs/*.sample) |
| 13:36.46 | *** join/#asterisk jmacz (~jmacz@190.144.75.22) |
| 13:38.55 | kruemeltee | is it possible to create a custom variable for each sip account for using within the dialplan? If so, how? |
| 13:39.24 | manxpower | kruemeltee: see setvar= in sip.conf.sample |
| 13:39.34 | kruemeltee | okay ... thanks :-) |
| 13:40.18 | spiceycurry | can someone share with me via pastebin, their iptables file for asterisk? |
| 13:42.35 | *** join/#asterisk dajhorn (~dajhorn@adsl-75-17-124-26.dsl.rcsntx.sbcglobal.net) |
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| 13:50.06 | *** part/#asterisk c0rnoTa (~c0rnoTa@178.176.143.130) |
| 13:50.17 | moos3 | is it possible to fax over SIP |
| 13:50.33 | Naikrovek | moos3: yes |
| 13:51.12 | moos3 | i just tried and got no carrier detected ideas? |
| 13:51.21 | moos3 | i'm using hylafax with iaxmodems |
| 13:51.59 | Naikrovek | "hi is it possible to drive a car on the road" "yes" "my car won't go ideas" |
| 13:52.02 | Naikrovek | going to need more than that |
| 13:52.35 | *** join/#asterisk UQlev (~yuriy@212.50.99.8) |
| 13:52.40 | Naikrovek | need some error messages, things of that nature |
| 13:52.49 | moos3 | ok I can recieve just fine over my PRI, when I try to send out over the pri the fax fails |
| 13:53.04 | Naikrovek | what codec are you using |
| 13:53.10 | Naikrovek | if not G711 you're likely to have issues |
| 13:53.20 | moos3 | ok |
| 13:53.53 | moos3 | my iaxmodems are using alaw |
| 13:53.57 | moos3 | for the codex |
| 13:54.01 | Naikrovek | okay that's fine then |
| 13:54.44 | Naikrovek | unfortunately you've exhausted my knowledge of faxing using asterisk. hang out, though, someone will read this and ask you more questions. |
| 13:56.14 | *** join/#asterisk jaytee (~jforde051@unaffiliated/jaytee) |
| 13:56.35 | *** part/#asterisk agx (~Antonio@host63-216-static.34-88-b.business.telecomitalia.it) |
| 13:57.59 | moos3 | here is whats in the console for asterisk http://pastebin.org/192187 |
| 14:00.49 | *** join/#asterisk leifmadsen (~Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 14:00.49 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 14:05.59 | *** join/#asterisk moy (~chatzilla@bas1-unionville55-1177733627.dsl.bell.ca) |
| 14:07.09 | Yudaisrael1984 | can someone explain to me how asterisk is sending out packets with no nat when nat is set to yes |
| 14:07.09 | *** join/#asterisk gelpg (~chatzilla@dsl51B619C3.pool.t-online.hu) |
| 14:07.09 | [Outcast] | can you pastebin your config: sip.conf? |
| 14:07.09 | [TK]D-Fender | Yudaisrael1984: perhaps you should SHOW US |
| 14:07.09 | Yudaisrael1984 | ok |
| 14:07.44 | *** join/#asterisk The_Boy_Wonder (~vossel@asterisk/batman-developer/dvossel) |
| 14:08.12 | Yudaisrael1984 | how can i get it if its realtime? |
| 14:08.20 | Yudaisrael1984 | how can i copy relatime to u? |
| 14:09.05 | gelpg | hi, my B400P ISDN card permanently changes the master |
| 14:09.13 | gelpg | like this from syslog: http://pastebin.com/zk1sqbNM |
| 14:09.26 | [Outcast] | in that case....what is the ip address in the contact field in DB? |
| 14:09.36 | gelpg | do you have any idea how to fix this? |
| 14:11.20 | [Outcast] | gelpg: what is problem you are having....that log snippet tell me nothing. |
| 14:11.54 | [Outcast] | gelpg: nevermind didn't see the line above |
| 14:12.10 | gelpg | Outcast: i have 3 ISDN ptp line and I can't dial out |
| 14:12.41 | gelpg | Outcast: the dahdi_tool shows all the spans are red |
| 14:13.19 | *** join/#asterisk werdan7 (~w7@freenode/staff/wikimedia.werdan7) |
| 14:13.20 | gelpg | Outcast: I tested the card in another place and the card worked with the same config |
| 14:13.28 | beefpastry | Has anybody attempted to use a sip warning header to prompt a pop-up on polycoms? |
| 14:13.53 | gelpg | Outcast: so it is a telco problem or I should fix something? |
| 14:15.20 | [Outcast] | gelpg:hmmmm.....could possible be...if it work on a different set of ISDNs that are exactly the same...I would call my provider to trouble shoot. |
| 14:16.36 | gelpg | Outcast: and what should I tell them? What is the problem? |
| 14:17.17 | [Outcast] | gelpg: It seems like a signaling problem on the dchans |
| 14:18.00 | [Outcast] | gelpg: but that is only a guess |
| 14:18.44 | gelpg | Outcast: thanks, I'll try |
| 14:18.59 | *** join/#asterisk putnopvut (~putnopvut@asterisk/master-of-queues/mmichelson) |
| 14:18.59 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 14:21.06 | manxpower | If the span is RED, then you do not have a line connected to the card |
| 14:22.22 | gelpg | manxpower: phisically I have |
| 14:22.59 | gelo | manxpower not necessarily, there are providers which save energy disconnecting l2 |
| 14:23.14 | coppice | but emotionally there may not be closure |
| 14:23.35 | manxpower | gelpg: RED means "no layer 1" |
| 14:24.00 | [Outcast] | maxpower: if there is no carrier signal on the dchan it will be red as well |
| 14:24.07 | gelo | ok, all layers then :) |
| 14:24.17 | manxpower | [Outcast]: as far as I'm concerned that is layer 1 |
| 14:24.37 | manxpower | no d-channel would not cause a red alarm |
| 14:25.00 | manxpower | d-channel is a higher level protocol |
| 14:26.17 | [Outcast] | manxpower: yes it would |
| 14:26.37 | *** join/#asterisk sun28 (~light@sun28.ipfw.su) |
| 14:27.20 | coppice | Outcast: a red alarm would cause no D-channel, not the other way around |
| 14:27.27 | Chainsaw | beefpastry: It's documented in the Polycom SIP administrators guide; I've toyed with the idea but not implemented it. You can cause popups or a lightning bolt (as a more more info marker). |
| 14:27.58 | *** join/#asterisk ctooley (~ctooley@254.sub-75-255-118.myvzw.com) |
| 14:28.24 | beefpastry | I've been playing with it for the last day...I know the header is going out, but the polycom is unaffected (and it should be provisioned correctly) |
| 14:29.07 | leifmadsen | beefpastry: never knew such a thing existed -- sounds useful if it works |
| 14:29.09 | gelpg | manxpower: I disconnected all the cabels but the first one |
| 14:29.16 | Chainsaw | beefpastry: What mode did you opt for? Direct or passive? |
| 14:29.25 | gelpg | manxpower: and i have this: http://pastebin.com/BZNXxmJf |
| 14:29.43 | manxpower | gelpg: now take that cable and plug it into an ISDN phone and confirm that the line is not working |
| 14:29.55 | gelpg | manxpower: the cabel is good, another ptmp works well with that |
| 14:30.14 | [Outcast] | feel free to correct if I am wrong....if I understand what you are tell me...if you plug a wink t1 in a port that was configured for pri it would not cause a red alarm? |
| 14:30.21 | gelpg | manxpower: it is a ptp line, it won't work with an isdn phone |
| 14:30.25 | manxpower | [Outcast]: correct. |
| 14:30.31 | gelpg | manxpower: as far as i know |
| 14:31.00 | manxpower | gelpg: you have a line problem. you can waste days trying to diagnose it or you can call the trouble into the telco |
| 14:31.21 | beefpastry | Chainsaw: you talking about web content...active or passive? |
| 14:31.51 | Chainsaw | beefpastry: Yes, that's the only "popup" I know you can trigger on a Polycom with a SIP header. |
| 14:31.52 | *** join/#asterisk TJ^ (~TJ@193.47.83.49) |
| 14:32.07 | Chainsaw | beefpastry: You basically command the microbrowser. |
| 14:33.35 | [Outcast] | manxpower and coppice: just labbed it, I stand corrected and little more educated...thanks :) |
| 14:33.54 | [Outcast] | gelpg: as stated you should still contact our provider |
| 14:38.36 | *** join/#asterisk rare1980_ (~rare1980@12.25.228.67) |
| 14:38.45 | *** join/#asterisk Slugs_ (~yeah@c-76-97-217-69.hsd1.ga.comcast.net) |
| 14:39.25 | gelo | gelpg wait a minute, do you have bri or pri lines? |
| 14:39.38 | gelpg | gelo: bri |
| 14:39.52 | gelpg | gelo: signalling: bri_cpe |
| 14:41.00 | gelo | sorry, but in your last paste i read "PRI got event". Is it DAHDI doesn't mind if it has pri or bri? |
| 14:41.38 | gelpg | gelo: dahdi sent it in CLI |
| 14:41.49 | *** join/#asterisk clintc (~clintc@n128-227-179-127.xlate.ufl.edu) |
| 14:44.21 | gelo | I know, I'm just a little surprised I hadn't notice that before... |
| 14:44.24 | beefpastry | Chainsaw: that was my next thing to look into...figured if I had asterisk doing what it was supposed to do and the polycom was set up as the Admin guide says, there had to be a missing part of the admin guide. |
| 14:45.00 | Naikrovek | there is a polycom manual on how to configure the phones for use with asterisk. maybe that'll help? |
| 14:45.15 | Naikrovek | i haven't read it and I dunno what you guys are working on so i don't know if your situation is covered in there |
| 14:50.36 | *** join/#asterisk jpeeler (~jpeeler@asterisk/digium-software-dev/jpeeler) |
| 14:52.17 | beefpastry | Naikrovek: the UsingPolycomswithAsterisk guide primarily talks about efks and things to override default settings that disagree with * |
| 14:52.32 | Naikrovek | ah |
| 14:52.50 | Naikrovek | yeah i was just looking through it (I printed it out weeks ago) and i'm of no help again :) |
| 14:53.22 | beefpastry | It's a handy guide for starters, though. ;) |
| 14:55.11 | *** join/#asterisk anthm (~anthm@freeswitch/developer/anthm) |
| 14:55.28 | *** join/#asterisk outtolunc (~me@c-67-160-192-210.hsd1.ca.comcast.net) |
| 14:58.24 | *** join/#asterisk Deeewayne (~dwayne@75-150-14-49-Atlanta.hfc.comcastbusiness.net) |
| 14:58.24 | *** mode/#asterisk [+o Deeewayne] by ChanServ |
| 14:58.30 | beefpastry | Chainsaw: mb.ssawc.call.mode=Active and voIpProt.SIP.header.warning.enable=1 the header is being sent ("Warning: 399..."), but no luck so far. |
| 14:59.36 | *** join/#asterisk spiceycurry (~mcurry@proxy.hostopia.com) |
| 15:00.24 | spiceycurry | Setup my iptables, all working now. My x-lite connected to my server. However, when I dial a number, I get "Call Failed: Not Found". Where could Iook to get more info about what is happening? |
| 15:01.07 | Gugge | whats written in the asterisk cli when you dial? |
| 15:01.23 | spiceycurry | I will see |
| 15:01.27 | [Outcast] | spiceycurry: dialplan? |
| 15:01.56 | spiceycurry | haha |
| 15:02.01 | spiceycurry | extension not found |
| 15:02.23 | spiceycurry | I suppose I am missing something in another conf file |
| 15:02.31 | [TK]D-Fender | spiceycurry: extensions.conf <------- |
| 15:02.38 | spiceycurry | sweet, thanks so much! :D |
| 15:02.40 | [TK]D-Fender | spiceycurry: Which is 90% of Asterisk |
| 15:05.24 | *** join/#asterisk soman (~somnath@stargate.starnet.fi) |
| 15:06.42 | *** join/#asterisk lordvadr (~something@jose-tc.ctc.biz) |
| 15:10.06 | *** join/#asterisk kombi (~kombi@port-92-198-15-96.static.qsc.de) |
| 15:11.11 | kombi | if it says "Unable to create channel of type 'foobar'", what is happening? |
| 15:11.28 | [TK]D-Fender | kombi: Never heard of chan_foobar before |
| 15:11.44 | [TK]D-Fender | kombi: How about SHOWING US THE PROBLEM? |
| 15:12.01 | kombi | you won't like it... I try to get sccp running |
| 15:12.28 | outtolunc | he's mikey, he doesn't like anything |
| 15:12.28 | [TK]D-Fender | kombi: that's only part of the picture |
| 15:12.42 | kombi | true.. wait, I'll post the output |
| 15:18.00 | *** join/#asterisk ChannelZ (channelz@burner.com) |
| 15:21.18 | spiceycurry | My colleague tells me that I could make an outgoing call by specifiying zap, however, I am using dahdi. What is the equivalent of 'zap' ? |
| 15:21.38 | ChannelZ | DAHDI/ |
| 15:22.04 | spiceycurry | ok |
| 15:22.15 | spiceycurry | do I need the trailing / ? |
| 15:22.23 | ChannelZ | like DAHDI/1/5551212 |
| 15:22.31 | spiceycurry | ok gret, thanks |
| 15:23.15 | kombi | [TK]D-Fender, everyone: http://pastebin.se/201138 |
| 15:23.59 | pabelanger | Anybody have any experience with http://www.rtpproxy.org/wiki/AsteriskCodecNegotiationPatch |
| 15:24.55 | pabelanger | Was looking for feed back from people running it. |
| 15:25.35 | gelo | pabelanger: i tried it some time ago. didn't like it, there were more calls not working than before |
| 15:25.52 | gelo | but of course, i may have done some things wrong back then |
| 15:26.47 | pabelanger | gelo: Ya, know some people take are using it, but I don't really understand the benefit for them. |
| 15:28.12 | gelo | pabelanger: yeah, i tried it thinking that it would really improve codec negotiation, but the calls just failed in different ways |
| 15:28.48 | *** join/#asterisk Sargun (~Sargun@atarack/Staff/Sargun) |
| 15:28.51 | spiceycurry | Does this make sense to dial out for area code 1 (226) ? http://pastebin.com/1a0Hg33T |
| 15:29.46 | pabelanger | gelo: It seems to me, if you locked down which codec you wanted asterisk to use, there would no reason to use the patch. |
| 15:32.05 | gelo | pabelanger: right. That's why i'm not using it |
| 15:33.20 | pabelanger | spiceycurry: What does your provider require. |
| 15:33.54 | spiceycurry | good question |
| 15:34.14 | spiceycurry | I was provided with almost nothing for info. I think we are using telus |
| 15:35.16 | *** join/#asterisk gr0mit (~tim@extrt.txrx.org.uk) |
| 15:35.38 | pabelanger | spiceycurry: try passing 10 digits, see what happens. Otherwise you need to call your provided and see what they need |
| 15:35.51 | spiceycurry | ok great, I will give that a try |
| 15:35.53 | spiceycurry | thanks |
| 15:36.44 | spiceycurry | Here is the error: [Apr 29 06:37:47] NOTICE[5489]: chan_sip.c:20059 handle_request_invite: Call from 'mike' to extension '2262082003' rejected because extension not found |
| 15:37.47 | *** join/#asterisk henry-nicolas (~d940f005@gateway/web/freenode/x-gmdoavtnkhurodjd) |
| 15:38.09 | kombi | [TK]D-Fender: did you take a look? |
| 15:38.13 | leifmadsen | spiceycurry: means the context that is being used to match the extension does not contain a pattern match or that extension |
| 15:38.23 | spiceycurry | ok, thanks again |
| 15:39.00 | *** part/#asterisk gelo (~gelo@209.138.60.213.dynamic.mundo-r.com) |
| 15:42.45 | *** join/#asterisk anthm (~anthm@freeswitch/developer/anthm) |
| 15:43.47 | henry-nicolas | Hi *, against you, what's the best web interface for asterisk currently ? supporting contexts, ivr, extension, call group/conference/forwarding/queues and it should also be able to make your coffe :) I would like to get that interface to only push data in a database and not to get other daemons running just for the web interface. |
| 15:46.57 | *** join/#asterisk Jibbs (~Jibbs@cpe-69-207-58-188.buffalo.res.rr.com) |
| 15:47.35 | Jibbs | hi everyone... i have an asterisk (trixbox) install and for some reason my outbound calls are dropping around 32-35 seconds consistently... hardware phone OR softphone... any insight on this would be GREATLY appreciated |
| 15:48.11 | *** join/#asterisk pinoyskull (~pinoyskul@112.198.64.91) |
| 15:49.07 | spiceycurry | Where would I register a dahdi channel, as I am getting these messages: |
| 15:49.08 | spiceycurry | WARNING[5887]: channel.c:4035 ast_request: No channel type registered for 'DAHDI' |
| 15:49.19 | spiceycurry | WARNING[5887]: app_dial.c:1745 dial_exec_full: Unable to create channel of type 'DAHDI' (cause 66 - Ch... |
| 15:50.00 | spiceycurry | My span: OK T2XXP (PCI) Card 0 Span 1 |
| 15:52.10 | tamiel | spiceycurry: try "module load chan_dahdi.so" |
| 15:52.16 | spiceycurry | ok sec |
| 15:52.47 | spiceycurry | Unable to load module chan_dahdi.so |
| 15:53.02 | spiceycurry | and loader.c:794 load_resource: Module 'chan_dahdi.so' could not be loaded. |
| 15:54.38 | tamiel | did you install dahdi and configure it (/etc/dahdi/*.conf and /etc.asterisk/chan_dahdi.conf) ? |
| 15:54.59 | spiceycurry | I believe so, but I will look again |
| 15:55.20 | spiceycurry | my conf files are there |
| 15:55.46 | Jibbs | is there a list of ports to forward by my router for asterisk? i've seen about a dozen websites and they're all slightly different |
| 15:55.51 | [TK]D-Fender | spiceycurry: Show us. also do "dahdi show channels" from * CLI |
| 15:55.58 | [TK]D-Fender | ~sipnat |
| 15:55.59 | infobot | [~sipnat] Quick guide on configuring * + SIP behind NAT : http://www.aocomputing.net/?p=3 , otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
| 15:56.08 | [TK]D-Fender | Jibbs: ^^^ |
| 15:57.03 | spiceycurry | doing /etc/init.d/dahdi status works, and shows all pri |
| 15:57.32 | spiceycurry | 1 - 23 = Clear, 24 = HDLCFCS |
| 15:59.12 | Jibbs | its not a NAT firewall (at least not natively) its a netgear router lol |
| 15:59.22 | [TK]D-Fender | spiceycurry: that is NOT what I asked yuo to do |
| 15:59.37 | [TK]D-Fender | spiceycurry: that has no impact on whever ASTERISK is configured to use DAHDI at all |
| 15:59.54 | [TK]D-Fender | Jibbs: same info is in there |
| 16:03.48 | *** join/#asterisk Mhaddog_ (~Mhaddog@adsl-072-149-063-056.sip.bct.bellsouth.net) |
| 16:03.50 | *** join/#asterisk kartik (~koolkarti@117.207.81.83) |
| 16:09.10 | Jibbs | ok yeah i have it set up ok... |
| 16:09.50 | Jibbs | does anyone know how to disable UAC in asterisk? |
| 16:10.27 | florz | don't load chan_sip |
| 16:10.51 | Jibbs | ok then how do i do that? lol sorry learning here |
| 16:12.15 | *** join/#asterisk iscario (~div@laureades.davout.pck.nerim.net) |
| 16:13.47 | *** join/#asterisk jasonwert-work (~jasonwert@99-27-170-70.lightspeed.cicril.sbcglobal.net) |
| 16:14.51 | leifmadsen | Jibbs: noload => chan_sip.so in modules.conf |
| 16:15.48 | iscario | hi, here is my extensions.conf (i omit general and globals) and my problem is that the "s" name seems not to work.... if i put 123 instead, it works, but not with 's', is it normal ? |
| 16:15.54 | iscario | http://pastebin.com/AbGzUtrD |
| 16:16.00 | leifmadsen | iscario: 's' is not a catch all |
| 16:16.14 | leifmadsen | if that's what you're thinking it does |
| 16:16.32 | leifmadsen | 's' will not match 123 |
| 16:16.36 | iscario | that was what i understood ;) leifmadsen |
| 16:16.44 | leifmadsen | you misunderstood incorrectly |
| 16:16.47 | iscario | what does the 's' is for then ? |
| 16:17.05 | Jibbs | so my asterisk is sending a "BYE" 30 seconds after my outgoing calls connect... any reason why that is? |
| 16:17.08 | leifmadsen | 's' means "start" and is for match calls that do not send an extension request (i.e. analog lines) |
| 16:17.31 | leifmadsen | iscario: use a real pattern match to do what you want |
| 16:17.33 | leifmadsen | _XXX |
| 16:17.44 | leifmadsen | where did you read that 's' is a catch-all? |
| 16:17.52 | jaytee | i thought s stood for "shit! I can't find this number anywhere." |
| 16:18.28 | iscario | oh i see leifmadsen , then what is the wildcard ? * ? |
| 16:18.43 | iscario | or maybe i should continue reading the doc^^ |
| 16:18.49 | leifmadsen | iscario: you need to read some basic dialplan documentation |
| 16:18.59 | leifmadsen | yes, you're asking Asterisk 101 questions |
| 16:19.06 | jaytee | jokers and one-eyed jacks are wild |
| 16:19.06 | iscario | ok no problem! thx leifmadsen |
| 16:19.25 | iscario | ok thx jaytee |
| 16:19.39 | Jibbs | why would me asterisk server be sending a "BYE" to my phone after 30 seconds of being connected ? |
| 16:19.54 | leifmadsen | Jibbs: not getting a response back or not getting audio? |
| 16:20.06 | Jibbs | i've been talking to people and just poof |
| 16:20.26 | Jibbs | doesn't happen on internal calls |
| 16:20.33 | leifmadsen | check the sip trace to see if the phone's response is getting back to asterisk |
| 16:20.38 | leifmadsen | sounds like a NAT issue |
| 16:20.49 | leifmadsen | someone isn't responding to something that needs to be responded to |
| 16:20.49 | Jibbs | how can i check that? |
| 16:20.54 | leifmadsen | sip set debug on |
| 16:21.08 | Jibbs | that's gonne be fun reading through that text lol |
| 16:21.29 | Jibbs | ok its on |
| 16:21.33 | Jibbs | what am i looking for? |
| 16:21.42 | *** join/#asterisk sysreq (~sysreq@unaffiliated/sysreq) |
| 16:21.50 | leifmadsen | well, once you understand how SIP works, it'll be easier to understand what is going on |
| 16:22.00 | leifmadsen | I'd suggest reading the first 30 pages of the SIP RFC |
| 16:22.11 | leifmadsen | it does a good job of giving an overview of how SIP works |
| 16:22.23 | Jibbs | right i'm looking for some sort of acknowledgement that some sort of packet is coming back to determine the call is still active, coorect? |
| 16:22.27 | leifmadsen | yes |
| 16:22.43 | leifmadsen | you might see asterisk retry sending a request multiple times |
| 16:23.00 | leifmadsen | if that is the case, then you know the phone isn't getting the requests correctly, or isn't responding |
| 16:23.13 | leifmadsen | (the issue could really be anything, but that's where I'd start) |
| 16:24.26 | Joel | Jibbs, fwiw, I myself am in complete agreement with leifmadsen. |
| 16:24.46 | Jibbs | yeah i'm sure its server and or firewall related |
| 16:24.54 | *** part/#asterisk atis_work (~atis_work@193.238.212.171) |
| 16:24.57 | Jibbs | its not the phone tried 2 phones, and a soft phone |
| 16:25.07 | Jibbs | this is the 2nd computer its happening with |
| 16:25.11 | *** join/#asterisk atis_work (~atis_work@193.238.212.171) |
| 16:25.22 | kruemeltee | bye bye ... :-) Have to go home now :-) |
| 16:25.27 | Joel | yeah, it's almost always networking related, but because everyone's network is different, there is no sure fire answer. |
| 16:25.36 | Jibbs | i'm just trying to get some guidance |
| 16:26.05 | *** join/#asterisk dohd (~Xaa@nala.dohd.org) |
| 16:26.56 | *** join/#asterisk atis_work (~atis_work@193.238.212.171) |
| 16:27.04 | [TK]D-Fender | Jibbs: I'm not seeing SIP DEBUG of your failed call anywhere |
| 16:27.40 | Jibbs | i'm trying to isolate it... i have 3 phones so i'm turning 2 off so they dont interfere, i will pastebin something shortly |
| 16:28.46 | Jibbs | Retransmitting #4 (NAT) to 192.168.1.25:5060: .... how can i stop this lol i turned the phone off and its just going crazy spamming me with that |
| 16:29.38 | dohd | hi all |
| 16:30.06 | Joel | Jibbs, that's the exact clue leifmadsen was telling you to look for. |
| 16:30.16 | Jibbs | no thats from the unplugged phone |
| 16:30.17 | pabelanger | sip set debug off |
| 16:30.26 | pabelanger | sip set debug ip <phone IP> |
| 16:30.39 | Joel | Jibbs, I'm not here to argue with you, if you don't believe me, that's a-ok :) |
| 16:30.52 | Jibbs | lol no i get it... |
| 16:31.12 | Joel | signal to noise ratio limit hit for the day, switching to lurking. |
| 16:31.16 | Jibbs | i'm just telling you that i unplugged the phoen so my debug window was clean, that IP is from the phone i just unplugged and wasnt happening before |
| 16:32.31 | pabelanger | Jibbs: And that will not stop asterisk from sending SIP packets to the phone, since it still exists in your sip.conf file |
| 16:32.53 | Jibbs | ok i get that |
| 16:33.09 | Jibbs | alright i did what you said pabelanger... i set the sip debug to the ip of my softphone |
| 16:33.26 | pabelanger | Jibbs: reproduce your issue |
| 16:33.32 | Jibbs | i did |
| 16:33.50 | pabelanger | ~pb |
| 16:33.50 | 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.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. |
| 16:33.55 | Jibbs | http://pastebin.com/SEZCeFC9 |
| 16:34.28 | Jibbs | i'm not sure if i missed something... is there a way for me to send my debug to a log? |
| 16:34.52 | Joel | I would just use ngrep myself |
| 16:35.32 | pabelanger | we need a !collectdebug setting :) |
| 16:35.33 | pabelanger | http://svn.digium.com/svn/asterisk/trunk/doc/HOWTO_collect_debug_information.txt |
| 16:36.09 | pabelanger | ~debug |
| 16:36.10 | infobot | ACTION DeBuggers $1 |
| 16:36.10 | Jibbs | ok let me do that real quick |
| 16:37.56 | Jibbs | should i also have debugging on the external IP? |
| 16:38.02 | Jibbs | not just my soft phone? |
| 16:38.12 | *** join/#asterisk moos3 (~rgenthner@pool-72-73-117-158.ptldme.east.myfairpoint.net) |
| 16:38.46 | pabelanger | Jibbs: yes |
| 16:38.55 | Jibbs | pabelanger may i PM you my pastebin? |
| 16:39.14 | Joel | Yes, pm it, make sure as few eyes as possible are able to help you :P |
| 16:39.48 | Jibbs | joel.. i just really dont want to expose my cell # is that alright? |
| 16:39.57 | Joel | Jibbs, you can mask your cell # |
| 16:40.05 | Jibbs | yes i know |
| 16:40.10 | Joel | Jibbs, just do a simple search and replace. |
| 16:40.14 | Jibbs | that requires search and replace! :P |
| 16:40.44 | Jibbs | lol its 7 megs |
| 16:40.48 | Jibbs | of text |
| 16:40.52 | Jibbs | for liek 30 seconds |
| 16:42.17 | Joel | perl -pie 's/1800goaluv/1800notmine/g' trace |
| 16:42.25 | Jibbs | should i try a different level of verbosity? |
| 16:43.49 | pabelanger | infobot: debug is a method of collecting logs allowing others help troubleshoot an issue. Refer to doc/HOWTO_collect_debug_information.txt or http://svn.digium.com/svn/asterisk/trunk/doc/HOWTO_collect_debug_information.txt |
| 16:43.50 | infobot | ...but debug is already something else... |
| 16:44.06 | pabelanger | infobot: collectdebug is a method of collecting logs allowing others help troubleshoot an issue. Refer to doc/HOWTO_collect_debug_information.txt or http://svn.digium.com/svn/asterisk/trunk/doc/HOWTO_collect_debug_information.txt |
| 16:44.07 | infobot | pabelanger: okay |
| 16:44.19 | pabelanger | ~collectdebug |
| 16:44.19 | infobot | rumour has it, collectdebug is a method of collecting logs allowing others help troubleshoot an issue. Refer to doc/HOWTO_collect_debug_information.txt or http://svn.digium.com/svn/asterisk/trunk/doc/HOWTO_collect_debug_information.txt |
| 16:46.12 | Jibbs | pabelanger this log is massive |
| 16:47.43 | pabelanger | Jibbs: then trim it down. Just before you place your call 'logger rotate', reproduce issue, then 'logger rotate' again. |
| 16:48.04 | Jibbs | yeah in the 30 seconds of me logging the file is up to 7 megs |
| 16:48.14 | Jibbs | i mean its not terrible but its too big for pastebin it seems |
| 16:48.22 | *** join/#asterisk diegomad (~mad@190.146.200.120) |
| 16:49.41 | pabelanger | You will have to figure out what other garbage is listed in the log file and unload modules. |
| 16:50.04 | Jibbs | i went to verbose 1 for a sec |
| 16:50.15 | pabelanger | 7MB for 30seconds seems way to much for 1 sip channel |
| 16:50.28 | Jibbs | http://pastebin.com/r6PTjAGW |
| 16:50.30 | *** join/#asterisk grEvenX (~even@cC0FD00C3.dhcp.bluecom.no) |
| 16:52.36 | *** join/#asterisk gr0mit (~tim@extrt.txrx.org.uk) |
| 16:54.10 | *** join/#asterisk lvlolvlo (~lvlolvlo@unaffiliated/lvlolvlo) |
| 16:54.18 | Jibbs | http://pastebin.com/Diha2ZyR this is from my asterisk box |
| 16:54.46 | *** join/#asterisk drmessano (~nonya@pdpc/supporter/active/drmessano) |
| 16:54.53 | Jibbs | (other ip) |
| 16:55.29 | Jibbs | yeah its only on outbound calls i can receive just fine |
| 16:56.35 | pabelanger | Jibbs: What is your actual issue? |
| 16:56.45 | *** join/#asterisk moos3 (~rgenthner@216.52.121.66) |
| 16:56.53 | Jibbs | my outgoing calls, without fail, disconnect after 30 ish seconds |
| 16:57.08 | moos3 | any one have ideas on my issues? |
| 16:57.21 | [TK]D-Fender | Jibbs: And where is this outbound call to look at? Please do NOT include all the extra spam debug. basic verbose + SIP DEBUG only |
| 16:57.37 | Joel | moos3, what issue? does it hurt when you pee? or? |
| 16:57.52 | Jibbs | ok well i followed that link pabelanger posted it said to set verbosity to 15 so what should it be set to? |
| 16:58.43 | [TK]D-Fender | Jibbs: you have core debug enabled. Don't. Next, show us a COMPLETE call, with your sip.conf masking only passwords |
| 16:58.50 | moos3 | Joel, http://pastebin.org/192545 http://pastebin.org/192558 |
| 16:58.50 | *** join/#asterisk sun28 (~light@sun28.ipfw.su) |
| 16:59.15 | moos3 | Joel, i can't get faxes to go out, but i got them working coming in |
| 16:59.33 | Jibbs | core set verbose 15 .... is that what you want to see? |
| 17:02.34 | Jibbs | http://pastebin.com/utvKzTbq |
| 17:04.23 | Jibbs | is that better? |
| 17:08.40 | pabelanger | Jibbs: your setup for NAT is wrong. |
| 17:09.05 | pabelanger | post your sip.conf file |
| 17:10.34 | Jibbs | show's a bunch of includes... http://pastebin.com/kj6BTeDu |
| 17:17.18 | pabelanger | Jibbs: better get some help from #freepbx, since you are running the GUI. Not sure the settings you need for it |
| 17:17.39 | Jibbs | actually i just think i figured it out by accident |
| 17:17.53 | Jibbs | and to be honest i feel like a turd |
| 17:18.34 | Jibbs | i was hunting through my router settings and there is some sort of "Disable SIP ALG" setting in there, i turned that on and it seems to be working... not sure what SIP ALG is but it seems to have done the trick |
| 17:18.49 | lvlolvlo | @Jibbs SIP ALG = SIP Application Layer Gateway |
| 17:19.16 | lvlolvlo | it is designed to assist VoIP (SIP) traffic transverse over your NAT'ed network |
| 17:19.18 | Qwell | ALG = break SIP |
| 17:19.22 | Qwell | in all cases. |
| 17:19.23 | Jibbs | lol i see that now |
| 17:19.33 | Qwell | there are 0 working ALG implementations. |
| 17:19.34 | lvlolvlo | however, it will more than often break SIP/VoIP on your network |
| 17:19.47 | lvlolvlo | agreed |
| 17:20.22 | Jibbs | well i must say thank you to all that helped me |
| 17:26.23 | *** join/#asterisk voxter (~voxter@S010600090f53ea17.vc.shawcable.net) |
| 17:29.34 | *** join/#asterisk Ad-Hoc (~nimbus@62.1.237.17.dsl.dyn.forthnet.gr) |
| 17:31.47 | *** join/#asterisk Tim_Toady (~moi@77.49.26.91.dsl.dyn.forthnet.gr) |
| 17:32.30 | *** join/#asterisk theron (~theron@ip244.scolloc.lh.net) |
| 17:34.17 | moos3 | Joel, any ideas why the call would fail |
| 17:34.56 | Joel | moos3, no idea, I haven't done a whole lot w/ faxing. |
| 17:38.12 | moos3 | Joel, ok cool |
| 17:38.51 | *** join/#asterisk Jumpie (~lah@c-69-255-192-97.hsd1.dc.comcast.net) |
| 17:40.12 | *** join/#asterisk phobosd (~phobosd@69.175.66.195) |
| 17:40.26 | phobosd | hey, any good asterisk devs out there want to help me with writing some statistical software |
| 17:40.33 | phobosd | will pay, etc |
| 17:40.57 | pabelanger | phobosd: #asterisk-consultants |
| 17:41.23 | phobosd | thanks |
| 17:42.14 | Jumpie | heh didnt know of such a channel |
| 17:42.26 | *** join/#asterisk voxter (~voxter@S010600090f53ea17.vc.shawcable.net) |
| 17:44.39 | outtolunc | i stopped hangin out in there due to lack of screen space and visitors |
| 17:47.40 | Joel | probably better off trying to find a solution that already fits, or comes close... |
| 17:47.55 | *** part/#asterisk Jibbs (~Jibbs@cpe-69-207-58-188.buffalo.res.rr.com) |
| 17:49.04 | *** join/#asterisk lhz (~shrekz@c-b9aa72d5.021-158-73746f34.cust.bredbandsbolaget.se) |
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| 18:05.04 | *** join/#asterisk jaskew (~jdaskew@netblock-75-79-182-110.dslextreme.com) |
| 18:07.15 | *** join/#asterisk TimeRider (steve@5ac93073.bb.sky.com) |
| 18:07.47 | *** join/#asterisk Defraz (~Defraz@corp.fuzecore.com) |
| 18:09.19 | *** join/#asterisk pa (~pa@unaffiliated/pa) |
| 18:20.45 | Arsenick | Hi all, How can i see the number dialed to get into my system asterisk ? is this possible ? I want to match a pattern when a call is comming from a specific number.. |
| 18:21.38 | Joel | Arsenick, on most t1's, and most voip providers you can, yes. |
| 18:23.16 | Arsenick | Joel, and on classic PSTN line ? because we have a single number here, ppl dial 666-7777 and the telco automaticly send the call to one of the 6 phonne line.. so it's never incomming in the same ZAP channel... I don't know how can I say hey the incomming call was for the 1800 line, move it to the queue... |
| 18:23.35 | Arsenick | Joel, btw thanks for your awnser |
| 18:23.42 | Joel | Arsenick, no, the telco doesn't sent the number dialed over a true pstn line, there is no way to send this info. |
| 18:24.02 | Arsenick | oops.. |
| 18:25.17 | [TK]D-Fender | Arsenick: Analog lines "just ring". Thre is no DID on them except in very rare implementations |
| 18:25.26 | Arsenick | ok |
| 18:25.43 | Arsenick | so the best way to do this will probably to "move" the 1800 line to a VOIP provider.. |
| 18:25.56 | moos3 | can anyone help me with hylafax not sending faxes out? |
| 18:26.13 | *** join/#asterisk mnick86 (~Matthias@whhem00002.cip.uni-regensburg.de) |
| 18:26.18 | Arsenick | and then I'll be able to handle the destination caller id.. |
| 18:26.32 | Arsenick | moos3, 95% chances are that your modem is wedged ? |
| 18:26.33 | Arsenick | :p |
| 18:27.11 | moos3 | Arsenick, its a iaxmodem, now do i un wedge it |
| 18:29.32 | Arsenick | lol, I was just kidding.. I didn't work a lot with hylafax, the only thing I remember was the damned serial modem was always wedged.. |
| 18:29.37 | Arsenick | good luck! |
| 18:29.58 | Arsenick | Thanks Joel and [TK]D-Fender for your help |
| 18:34.00 | moos3 | Arsenick, thansk |
| 18:35.26 | Joel | Arsenick, correct, moving it to a voip provider may be your answer. Although a pri or isdn may provide you with better call quality and slas. |
| 18:35.54 | moos3 | anyone good with hylafax and iaxmodems |
| 18:37.57 | *** join/#asterisk rocksfrow (~kyle@static-66-16-158-235.dsl.cavtel.net) |
| 18:38.09 | rocksfrow | does anybody here use asterisk for a SMS gateway? |
| 18:38.42 | manxpower | rocksfrow: have you read the SMS info in /doc ? |
| 18:38.48 | manxpower | well doc/ |
| 18:39.16 | rocksfrow | yeah |
| 18:39.23 | rocksfrow | you can use a cell phone to do it right |
| 18:39.30 | rocksfrow | thats why im asking if anybody has experience with it |
| 18:39.59 | rocksfrow | but the cell phone option probably sucks. |
| 18:40.03 | rocksfrow | slow as hell i would imagine |
| 18:40.32 | rocksfrow | i've been using clickatell, and have been happy until a month ago their short codes got blocked, they've been down for over a month |
| 18:40.35 | rocksfrow | crazy. |
| 18:41.24 | manxpower | the string "cell" and "mobile" do not appear in my sms.txt |
| 18:41.54 | manxpower | have you read the output of "core show application sms" ? |
| 18:42.56 | manxpower | Which ETSI ES 201 912 will you be using? |
| 18:43.11 | manxpower | ..er.. which ETSI ES 201 912 service provider will you be using? |
| 18:43.33 | manxpower | (as you know from the docs, ETSI ES 201 912 is the only "sms" protocol app_sms supports) |
| 18:46.03 | rocksfrow | yeah, my question was preliminary, "does anybody use asterisk for an sms gateway" |
| 18:46.42 | rocksfrow | more or less asking HOW GOOD it works, not HOW it works |
| 18:46.44 | rocksfrow | but thx manx |
| 18:47.19 | *** join/#asterisk jeffgus (~jeffgus@2002:ad33:b504::1) |
| 18:47.25 | Joel | rocksfrow, the best sms gateways to use all use email to operate. |
| 18:47.36 | manxpower | *nod* I was inquiring to find out if you were one of the very small number of people that live in an area that Asterisk's SMS support is compatible with. |
| 18:47.48 | rocksfrow | Joel, eh..not sure about that |
| 18:48.17 | rocksfrow | Joel, i have read about em though.. |
| 18:48.17 | Joel | rocksfrow, sorry, if you do any amount of worthwhile volume they are. |
| 18:48.27 | rocksfrow | Joel, why so? |
| 18:48.43 | rocksfrow | Joel, most messages are one-offs, not bulk |
| 18:48.45 | Joel | rocksfrow, it sounds like you need to do some basic research on available sms gateway options. |
| 18:48.47 | rocksfrow | (for me) |
| 18:48.53 | rocksfrow | Joel, lol. |
| 18:48.57 | Joel | well in that case then use a cellphone. |
| 18:49.06 | Joel | or a cell modem and a sim card. |
| 18:49.35 | rocksfrow | Joel, i'm very aware of available sms gateways, I'm basically just curious on if anybody uses their asterisk boxes to send SMS, and how their experience with it has been |
| 18:49.53 | manxpower | I've seen people here spend three hours learning about Asterisk's SMS support, only to realize that since they live in the USA/Canada they can't use app_sms |
| 18:49.54 | Joel | I've used asterisk to send sms |
| 18:49.57 | Joel | works great. |
| 18:50.05 | Joel | although I've only used it between asterisk machines. |
| 18:50.33 | rocksfrow | manxpower, yeah..that's why i was asking about the cell phone method, bc i did read that app_sms has limited support |
| 18:50.53 | rocksfrow | right.. |
| 18:51.01 | rocksfrow | I'm blown away by Clickatell |
| 18:51.06 | rocksfrow | my account has been down for a month and a half |
| 18:51.12 | rocksfrow | and support pretty much just says, sorry gotta wait lol |
| 18:51.28 | rocksfrow | i think they had some clients spamming like crazy which in turn got most of their shared codes blocked |
| 18:51.44 | rocksfrow | i'm looking at FastSMS now |
| 18:52.40 | rocksfrow | hey guys so I should add 0800NXXXXXX to my dial patterns in order to call UK 800 numbers, right? |
| 18:52.47 | rocksfrow | i've never even heard of 0800 numbers. lol |
| 18:53.38 | rocksfrow | looks like they have 0-808 as well |
| 18:54.51 | rocksfrow | interesting.. |
| 18:56.24 | rocksfrow | <PROTECTED> |
| 18:57.47 | rocksfrow | i guess that makes sense my PRI isn't going to let me call UK toll-free numbers from the US? |
| 18:58.16 | *** join/#asterisk Yudaisrael1984 (~Yuda@bzq-79-177-133-59.red.bezeqint.net) |
| 18:58.41 | Yudaisrael1984 | is there anyone here willing to help me with a mystery in asterisk that might have a man in the middle involved |
| 18:58.47 | p3nguin | rocksfrow: Are you allowed to call internationally? |
| 18:58.58 | Yudaisrael1984 | i pasted to pastebin a debug of an ATA i need to know if it looks like theres a man in the middle |
| 18:59.04 | *** join/#asterisk mpd (~chatzilla@70.28.49.95) |
| 18:59.21 | Yudaisrael1984 | http://pastebin.com/qHkzvbw5 |
| 18:59.45 | rocksfrow | p3nguin, i was just about to play with that...but the company doesn't have a local # posted to call |
| 18:59.56 | Yudaisrael1984 | the symptom is that the call gets cut off in middle |
| 18:59.56 | Yudaisrael1984 | and on the server the debug is retransmitting to the correct ip of the ATA |
| 19:00.37 | rocksfrow | anybody have an international # i can test calling? |
| 19:00.43 | rocksfrow | im in the USA |
| 19:02.38 | *** join/#asterisk lnd (~lnd@92.41.122.98.sub.mbb.three.co.uk) |
| 19:03.19 | Yudaisrael1984 | hello anyone??? |
| 19:03.47 | rocksfrow | nice.. |
| 19:03.47 | paulc | rocksfrow: UK 0800 numbers - some are 6 digit, some are 7 |
| 19:03.49 | *** join/#asterisk ikariW (~ikariW@74-92-245-181-Utah.hfc.comcastbusiness.net) |
| 19:03.51 | p3nguin | rocksfrow: You'll have to have a matching extension for it, but you can call 44 1223 770 016 to test international calling. |
| 19:03.53 | rocksfrow | p3nguin, yes. international calls wokr. |
| 19:03.58 | rocksfrow | p3nguin, i just called a hotel in UK :-p |
| 19:04.02 | paulc | rocksfrow: call +44 1534 888222 to get my voicemail in the UK |
| 19:04.05 | p3nguin | rocksfrow: That'll work, too. |
| 19:04.06 | rocksfrow | p3nguin, i added the internatinoal dialling pattern..and it worked great. |
| 19:04.16 | rocksfrow | paulc, thanks bro, already found a hotel # to test :-p |
| 19:04.42 | rocksfrow | so, it makes sense that UK toll-free's don't work. |
| 19:04.42 | p3nguin | rocksfrow: Add one matching the toll-free numbers, too, and see if it works. |
| 19:04.48 | rocksfrow | p3nguin, i did. |
| 19:04.52 | rocksfrow | did you see the line i pasted? |
| 19:05.00 | *** join/#asterisk citrus2 (~citrus2@mail.serviceobjects.com) |
| 19:05.01 | rocksfrow | the call gets passed, then gets a signal to hang up |
| 19:05.05 | rocksfrow | assuming from the PRI |
| 19:05.26 | p3nguin | <rocksfrow> hey guys so I should add 0800NXXXXXX to my dial patterns in order to call UK 800 numbers, right? <-- no, that is not a valid pattern |
| 19:05.32 | rocksfrow | p3nguin, -- Channel 0/1, span 1 got hangup request, cause 16 |
| 19:05.47 | rocksfrow | p3nguin, that pattern isnt valid? |
| 19:05.48 | p3nguin | Patterns must begin with _ |
| 19:05.52 | rocksfrow | .... |
| 19:05.56 | rocksfrow | p3nguin, no.. |
| 19:06.00 | p3nguin | yes. |
| 19:06.03 | rocksfrow | p3nguin, probably should mention i'm using freePBX :-p |
| 19:06.04 | citrus2 | i just set up a asterisk box in the amazon EC2 cloud. when it calls my phone and plays a file its really stuttery any reason this may be? ping times look great from the server.. |
| 19:06.08 | rocksfrow | that's probably why we're not matching up. |
| 19:06.54 | p3nguin | I don't know anything about FreePBX, since this isn't the appropriate place for it. But anyway, extension patterns begin with an underscore. |
| 19:07.10 | *** join/#asterisk moos3 (~rgenthner@216.52.121.66) |
| 19:07.14 | rocksfrow | p3nguin, i guess freePBX is putting the _'s |
| 19:07.22 | rocksfrow | the way they have it is a textbox you separate each pattern by a new line, |
| 19:07.22 | manxpower | citrus2: what is the JITTER |
| 19:07.24 | p3nguin | let's hope |
| 19:07.34 | rocksfrow | p3nguin, i know it's valid, bc the others i am using |
| 19:07.40 | rocksfrow | and the pattern worked |
| 19:07.41 | rocksfrow | lol.. |
| 19:07.52 | rocksfrow | before i add it, my phone will say invalid # or w/e |
| 19:07.57 | rocksfrow | then i'll add it..it dials..but the PRI just hangs up |
| 19:08.22 | rocksfrow | and from asterisk CLI i see, "Channel 0/1, span 1 got hangup request, cause 16" |
| 19:08.38 | citrus2 | manxpower, i don't understand your question |
| 19:08.41 | rocksfrow | p3nguin, i wouldn't expect a UK toll free to work |
| 19:08.47 | p3nguin | yeah |
| 19:08.49 | rocksfrow | you can't call a US toll-free from UK |
| 19:10.18 | manxpower | citrus2: look in the doc/ directory of the Asterisk source directory, there should be at least one, maybe two documents on jitter. |
| 19:12.32 | ikariW | Why would my database be logging the destination number as "s" in the database after an upgrade to 1.6? |
| 19:12.52 | citrus2 | manxpower, i don't see any jitter docs i see jabber, but that is obviously not it |
| 19:19.24 | *** join/#asterisk ruben23 (~ITadmin@122.55.48.243) |
| 19:22.21 | *** part/#asterisk Mhaddog_ (~Mhaddog@adsl-072-149-063-056.sip.bct.bellsouth.net) |
| 19:22.25 | *** join/#asterisk sat-man (~jlupresto@c-174-52-20-94.hsd1.ut.comcast.net) |
| 19:22.35 | *** join/#asterisk TimeRider (~steve@5ac7b3ed.bb.sky.com) |
| 19:23.08 | [Outcast] | anyone else notice google not working? |
| 19:24.19 | Yudaisrael1984 | hello anyone here to help?? |
| 19:24.52 | Yudaisrael1984 | http://pastebin.com/qHkzvbw5 |
| 19:24.53 | jaytee | Google works just fine for me |
| 19:26.36 | *** join/#asterisk Jumpie (~lah@c-69-255-192-97.hsd1.dc.comcast.net) |
| 19:26.45 | Jumpie | hey guys |
| 19:27.04 | Jumpie | on a tdm400p card....the port assignment is port 1 is at the very top right? farthest from mobo i think somebody said? |
| 19:27.29 | Qwell | it's the one with a '1' on the port. |
| 19:27.32 | *** join/#asterisk corretico (~laguilar@201.201.46.106) |
| 19:28.15 | Jumpie | mine arent la bleed |
| 19:28.18 | Jumpie | labeled |
| 19:28.24 | *** part/#asterisk gelo (~gelo@143.128.165.83.dynamic.mundo-r.com) |
| 19:28.38 | Jumpie | or if it is...its behind the metal of the slot opening |
| 19:29.41 | Jumpie | ah ok.. |
| 19:30.07 | pabelanger | Yudaisrael1984: User-Agent: ITM4L Softswitch ? |
| 19:30.35 | pabelanger | how is your log related to Asterisk? |
| 19:30.36 | Yudaisrael1984 | thats the switch |
| 19:30.44 | Yudaisrael1984 | thats the ATA |
| 19:30.47 | Yudaisrael1984 | the end user |
| 19:30.59 | Yudaisrael1984 | calls get disconnected |
| 19:31.28 | Yudaisrael1984 | u can see that after the begining of a session it gets the packets from a different ip |
| 19:34.07 | sat-man | I have a polycom ip650 on an asterisk box with the time/date just flashing. Can't check my call log on the phone because it can't figure out the time/date. Rebooted and unplugged. Any tips? |
| 19:34.21 | pabelanger | Yudaisrael1984: I'll ask again, how is this related to Asterisk? |
| 19:34.52 | Yudaisrael1984 | because its connecting to a asterisk server |
| 19:35.02 | Yudaisrael1984 | all my calls are cutting off |
| 19:35.26 | Yudaisrael1984 | so i did a debug on the end user as well as on my asterisk |
| 19:35.38 | Yudaisrael1984 | how can i keep a debug clean?? |
| 19:36.06 | pabelanger | Yudaisrael1984: Then post the debug logs from Asterisk, not your end user. |
| 19:36.22 | sat-man | I have a polycom ip650 on an asterisk box with the time/date just flashing. Can't check my call log on the phone because it can't figure out the time/date. Rebooted and unplugged. Any tips? |
| 19:36.45 | Yudaisrael1984 | ok how can i do that and keep it clean |
| 19:36.56 | pabelanger | sat-man: patience young grasshopper. |
| 19:37.21 | leifmadsen | sat-man: means you need to setup the SNTP Address |
| 19:37.31 | [TK]D-Fender | sat-man: Set up its SNTP server |
| 19:41.03 | moos3 | has anyone ever gotten iaxmodems to send faxes out over SIP |
| 19:41.29 | *** join/#asterisk jart (~jart@c-76-23-206-246.hsd1.ct.comcast.net) |
| 19:41.32 | leifmadsen | o.O |
| 19:41.43 | p3nguin | sure |
| 19:42.26 | moos3 | so far, i'm failing |
| 19:42.49 | moos3 | i can do it from sendfax on the cli faxes no issue, but from a hylafax client |
| 19:43.18 | p3nguin | So the probably isn't iaxmodem. |
| 19:44.23 | moos3 | well it tries to send it but returns no carrier detected or no answer from remote |
| 19:44.50 | moos3 | I watch the call happen i watch it answer but just requeues |
| 19:46.43 | moos3 | i have been working this all day, but can't figure out why its failing |
| 19:47.33 | moos3 | i'm open to all kinds of suggestions |
| 19:48.33 | *** join/#asterisk rossand (~aross@CPE000c413a19a3-CM00159a025ad4.cpe.net.cable.rogers.com) |
| 19:49.17 | *** join/#asterisk jtodd (e6fzz18czl@asterisk/community-director-and-tie-dye-shirt-lover/jtodd) |
| 19:49.17 | *** mode/#asterisk [+o jtodd] by ChanServ |
| 19:50.47 | *** join/#asterisk Tim_Toady (~moi@77.49.61.52.dsl.dyn.forthnet.gr) |
| 19:50.55 | moos3 | i have tried this exten => _X.,1,Dial(${PRIPORTS}/${EXTEN}), exten => _X.,1,Dial(local/${EXTEN}), not sure what else to try |
| 19:54.01 | *** join/#asterisk jeffgus (~jeffgus@2002:ad33:b504::1) |
| 19:56.40 | *** join/#asterisk hfb (~hfb@pool-98-112-210-75.lsanca.dsl-w.verizon.net) |
| 19:57.31 | [TK]D-Fender | moos3: You have 2 priority "1"'s for that. |
| 19:59.01 | *** join/#asterisk timholum (~chatzilla@65.209.186.58) |
| 19:59.48 | *** join/#asterisk spiceycurry (~mcurry@proxy.hostopia.com) |
| 19:59.57 | timholum | Hello, I am wondering if anyone knows of a way that I can make someone type a number in order to accept a call? |
| 19:59.57 | moos3 | [TK]D-Fender, what do you mean |
| 20:00.36 | spiceycurry | where could I find instructions installing asterisk 1.6.2 with DAHDI and LibPRI... the asterisk book states Zapta/ZT stuff, and I do not want to confuse anything. |
| 20:00.58 | [TK]D-Fender | [15:50]<moos3>i have tried this exten => _X.,1,Dial(${PRIPORTS}/${EXTEN}), exten => _X.,1,Dial(local/${EXTEN}), not sure what else to try <-- two exten lines with PRIORITY 1 |
| 20:01.00 | leifmadsen | timholum: look at the M() option of Dial() |
| 20:01.21 | timholum | leifmadsen: Thanks |
| 20:01.31 | [Outcast] | voip-info.org |
| 20:01.38 | *** join/#asterisk anthm (~anthm@freeswitch/developer/anthm) |
| 20:02.14 | spiceycurry | ty |
| 20:04.19 | *** join/#asterisk thebaddragon (yiffstar66@unaffiliated/devemo) |
| 20:04.48 | Yudaisrael1984 | anyone know of a asterisk consultant who would help solve a issue with payment? |
| 20:06.21 | *** join/#asterisk rare1980_ (~rare1980@115.186.4.96) |
| 20:06.48 | moos3 | [TK]D-Fender, not i only try one of those at a time |
| 20:08.06 | *** join/#asterisk ChUbB (~IceChat7@cpc2-aztw12-0-0-cust229.aztw.cable.virginmedia.com) |
| 20:12.46 | spiceycurry | Well... |
| 20:12.50 | spiceycurry | TAKE THAT: âDigium Asterisk Hardware Device Interfaceâ, itâs pronounced âDaddyâ |
| 20:13.11 | spiceycurry | should I put daddy on my box? |
| 20:14.16 | [TK]D-Fender | spiceycurry: MOMMY puts DAHDI in her box.... |
| 20:14.36 | spiceycurry | oh, I see. It all makes sense to me now! :D |
| 20:15.00 | moos3 | [TK]D-Fender, here is what my fail notice looks like from hylafax http://pastebin.org/192545 |
| 20:15.03 | [Outcast] | a joke about mounting stuff comes to mind |
| 20:15.04 | spiceycurry | I knew this channel was not so dry lol |
| 20:15.32 | spiceycurry | Outcast, you can get more information on the mount command by typing: mount man |
| 20:15.39 | spiceycurry | or wait, man mount |
| 20:15.41 | spiceycurry | :O |
| 20:15.44 | [TK]D-Fender | ~sex |
| 20:15.45 | infobot | [~sex] updatedb; locate; talk; date; cd; strip; look; touch; finger; unzip; uptime; gawk; head; emerge --oneshot condom; mount; fsck; gasp; more; yes; yes; yes; more; umount; emerge -C condom; make clean; sleep; |
| 20:15.56 | spiceycurry | lol wtf |
| 20:16.23 | Scorcerer | i'm trying to install asterisk GUI (2.0.4) to my asterisk (1.6.2.6) and all i get after logging in is "The GUI does not have necessary privileges. Please check the manager permissions for the user !" |
| 20:16.38 | [TK]D-Fender | Scorcerer: #asterisk-gui <---- not supported here |
| 20:16.44 | Scorcerer | ah, thanks |
| 20:17.16 | [TK]D-Fender | Scorcerer: Actually... we should say "not supported anywhere" in that there isn't an active maintainer. |
| 20:18.27 | Scorcerer | i'm still thinking that it isn't some kind of big bug, but rather a small one so i can use gui for learning and stuff :> |
| 20:21.17 | [TK]D-Fender | Scorcerer: As I said, it is unmaintained and has a lot of "holes" in its implementation. Not sure what you expect to learn from it, or achieve using it, but best of luck |
| 20:21.54 | Scorcerer | thanks, looks like i'm gonna need it :D |
| 20:23.02 | Scorcerer | and out of curiosity, is there any gui-web-etc app you recommend, or only CLI and documentation ? |
| 20:24.06 | *** join/#asterisk Alagar (~Administr@122.164.41.66) |
| 20:24.14 | *** part/#asterisk Joel (~jschuweil@unaffiliated/joel) |
| 20:24.15 | [TK]D-Fender | Scorcerer: FreePBX is maintained at least and far more complete |
| 20:24.19 | [TK]D-Fender | ~freepbx |
| 20:24.20 | infobot | [~freepbx] FreePBX is unable to be supported here. It is made up of complex dialplans and scripts which can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking there, or fpbx |
| 20:25.39 | *** join/#asterisk iratik (~itariki@74.223.41.171.nw.nuvox.net) |
| 20:25.40 | Scorcerer | but isn't it some fork of asterisk ? |
| 20:26.02 | ChannelZ | no |
| 20:26.23 | ChannelZ | It's a sort-of front end to it |
| 20:26.25 | iratik | Guys I need your help pretty badly. Asterisk seems to be taking up way too much CPU time. How can i figure out what is causing asterisk to be so busy? |
| 20:26.40 | ariel_ | top |
| 20:26.47 | Scorcerer | ah, thanks, i'll look into it :) |
| 20:27.03 | *** join/#asterisk PuroOsso (~PuroOsso@unaffiliated/puroosso) |
| 20:27.52 | iratik | is there a top for asterisk so i can find out exactly what inside asterisk is causing the main process to take so much cpu? |
| 20:28.16 | [TK]D-Fender | iratik: What ver again? |
| 20:28.27 | iratik | omg its god |
| 20:28.44 | iratik | 1.2.30.2 |
| 20:28.54 | ChannelZ | type H to see threads though that might not be of any help |
| 20:28.57 | leifmadsen | o.O |
| 20:29.10 | iratik | http://pastebin.com/hNVH016W |
| 20:29.10 | [TK]D-Fender | iratik: Oh... unsupported. Best of luck with that |
| 20:29.17 | ChannelZ | but I'll echo the 'eeks', kick 1.2 to the curb |
| 20:29.23 | [TK]D-Fender | checkout time, BBL |
| 20:29.35 | iratik | Does 1.2 have problems with cpu usage? |
| 20:29.47 | spiceycurry | is "Fax for Asterisk" with T.38 able to be run on 64 bit? For some reason I was told only could be done on 32 bit |
| 20:30.25 | Qwell | spiceycurry: yes, as of semi-recently |
| 20:30.32 | spiceycurry | ok cool |
| 20:30.34 | spiceycurry | thanks |
| 20:30.49 | ChannelZ | only under 1.6.2 yes? |
| 20:31.07 | spiceycurry | I am using 1.6.2 |
| 20:31.08 | ChannelZ | oh no I see an x64 for 1.6.1.5+, nm |
| 20:31.25 | ChannelZ | well then you're in luck :) |
| 20:32.24 | ChannelZ | iratik: it has problems with being really old |
| 20:35.56 | *** part/#asterisk jaytee (~jforde051@unaffiliated/jaytee) |
| 20:42.46 | *** join/#asterisk binbash_ (~peter@ip4da53781.direct-adsl.nl) |
| 20:43.03 | *** part/#asterisk spiceycurry (~mcurry@proxy.hostopia.com) |
| 20:49.23 | *** join/#asterisk patrb (~asdf@64-150-178-3.kansascity.abac.net) |
| 20:49.26 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.218.194) |
| 20:49.50 | *** join/#asterisk binbash_ (~peter@ip4da53781.direct-adsl.nl) |
| 20:50.05 | patrb | Anyone know of a good failover solution if servers are in 2 different data centers? |
| 20:50.26 | idespinner | would dundi work? |
| 20:50.38 | patrb | I've been trying with dundi |
| 20:50.40 | idespinner | its kind of a broad question |
| 20:51.07 | idespinner | polycoms can handle multiple registrations and backup servers |
| 20:51.13 | patrb | whatd id like is for an extension to be on both servers, have all calls to that extension go to server A, if server A goes down..it fails over to server B |
| 20:51.31 | patrb | dont really care about my sip registrations....just this single extension |
| 20:52.16 | patrb | I've tried this by putting the extension in different dundi priorities on both servers...but no luck |
| 20:58.55 | [TK]D-Fender | patrb: Basic dialplan on each and a peer on each side |
| 20:59.51 | *** join/#asterisk romb (~romb@unaffiliated/romb-work/x-7222485) |
| 21:00.04 | manxpower | wash it frequently |
| 21:00.29 | manxpower | I really should make sure I'm not reading the scroll back before I respond to statement 4 hours old. |
| 21:02.57 | patrb | [TK]D-Fender, thats what I was thinking...I guess I have a fundamental misunderstanding of what dundi priorities are for |
| 21:03.28 | *** join/#asterisk lanning (~lanning@208.87.235.224) |
| 21:05.14 | *** join/#asterisk eliel (~eliels@201.234.94.226) |
| 21:06.51 | eliel | hello, is there any condition/reason that makes asterisk only send rtp (while reproducing moh for example), only when it receives a rtp packet? i am having sound problems and asterisk ~1.4.21 is sending the rtp only when it receives a packet if there is no audio coming into asterisk it doesn't send rtp |
| 21:08.35 | *** join/#asterisk Toerkeium (~Miranda@201.216.206.221) |
| 21:08.42 | Toerkeium | hello guys |
| 21:09.26 | Toerkeium | does anyone knows why the caller hear his voice with eco? I'm using SJPhone, a Sound blaster 24 bits sound card and windows OS, obviously as server I'm using asterisk |
| 21:09.30 | Jumpie | this siemens gigaset a580 ip are fun to deploy |
| 21:09.46 | Jumpie | the web gui is impossible to use though utnil you initiate a base station firmware upgrade from handset |
| 21:12.27 | *** part/#asterisk ikariW (~ikariW@74-92-245-181-Utah.hfc.comcastbusiness.net) |
| 21:23.26 | p3nguin | So what you're saying is that Siemens ships useless phones. |
| 21:24.42 | *** join/#asterisk KNERD (~KNERD@129.113.46.109) |
| 21:26.40 | *** join/#asterisk tzafrir (~tzafrir@212.179.75.202) |
| 21:26.51 | *** join/#asterisk MT`AwAy (~MagicalTu@2001:41d0:2:973::aeb) |
| 21:27.06 | Jumpie | heh well..they have good call quality |
| 21:27.12 | Jumpie | i just think the initial setup is cumbersome... |
| 21:27.16 | Jumpie | not complex..just cumbersome |
| 21:27.44 | *** join/#asterisk aidinb (~Aidin@71-94-148-218.static.mtpk.ca.charter.com) |
| 21:28.11 | *** join/#asterisk jaytee (~jforde051@unaffiliated/jaytee) |
| 21:28.22 | MT`AwAy | hi, is there a way to remove the ;received= in the Via: header for reply to incoming calls? I see there's a patch for asterisk 1.2 for that but I'm running 1.6 ... |
| 21:30.42 | *** join/#asterisk jaskew (~jdaskew@netblock-75-79-182-110.dslextreme.com) |
| 21:33.48 | Jumpie | these seimans can only do 2 channels...so i am tryin to do a mesh setup across 2 base stations with 3 phones |
| 21:33.55 | Jumpie | 2 devices can be registered with the same credentials right? |
| 21:34.33 | leifmadsen | no |
| 21:34.47 | leifmadsen | 1 device per peer definition |
| 21:34.51 | Jumpie | hmm |
| 21:34.57 | leifmadsen | otherwise the registrations fight |
| 21:35.03 | leifmadsen | asterisk is not a proxy |
| 21:35.09 | Jumpie | well im tryin to make it so i can effectively make 3 calls on 3 wireless headsets |
| 21:35.19 | Jumpie | and i was told i can register multiple base stations to 1 headset |
| 21:35.19 | p3nguin | mt`away: Check the SIP_HEADER function. |
| 21:35.26 | Jumpie | 1. for coverage and 2. for round robin |
| 21:35.38 | Jumpie | tryin to figure out how im gonna do this sip wise |
| 21:35.40 | leifmadsen | that'd be dependent on the base station itself |
| 21:35.42 | leifmadsen | not asterisk |
| 21:35.42 | p3nguin | I wouldn't even know how to go about doing one handset to multiple bases. |
| 21:35.53 | MT`AwAy | I modified channels/chan_sip.c with the original "no ;received=" patch, I just had to modify it a bit to make it work |
| 21:36.54 | Jumpie | leifmadsen yea..but you have to setup the possible sip accounts to be 'doled out' to the headsets |
| 21:36.57 | Jumpie | at the bsase station |
| 21:37.25 | Jumpie | the idea is that that either base station can handle extensions a, b, c |
| 21:37.29 | Jumpie | but not the same time |
| 21:37.38 | leifmadsen | ok, but they have to all register individually |
| 21:37.50 | leifmadsen | to different peer/friend definitions |
| 21:38.14 | Jumpie | but in the end...there would be "2" devices registerd as a, b, c, respectively right? are you sayhing as long as its not at the same time? |
| 21:38.35 | p3nguin | It could be possible that the handsets do the registering and the bases are just dumb bridges. |
| 21:38.49 | Jumpie | thats how it works |
| 21:38.53 | Jumpie | dect only to handsets |
| 21:38.54 | leifmadsen | I have no idea how your system works. I'm just saying what Asterisk expects. |
| 21:38.59 | Jumpie | right..i understand |
| 21:39.02 | Jumpie | asterisk only sees the base station |
| 21:39.05 | leifmadsen | right |
| 21:39.10 | Jumpie | can register up to 6 extensions per base |
| 21:39.23 | p3nguin | If Asterisk only sees the bases, then they are not just dumb bridges. |
| 21:39.35 | p3nguin | You can register 6 handsets per base. |
| 21:39.41 | p3nguin | not extensions, handsets. |
| 21:39.42 | Jumpie | right, 2 concurrent channels |
| 21:39.52 | Jumpie | well, you can also do 6 sip extension :) |
| 21:39.57 | Jumpie | maybe its 8.... |
| 21:39.58 | p3nguin | negative |
| 21:40.17 | Jumpie | im showing 6 entries |
| 21:40.30 | Jumpie | for sip connections |
| 21:40.37 | Jumpie | maybe its coincedence? |
| 21:40.38 | p3nguin | sip devices, sure, but the bases don't give a rat's ass about the extensions. |
| 21:40.54 | Jumpie | oh oh..well |
| 21:40.57 | Jumpie | yea i see what your saying |
| 21:41.04 | Jumpie | i was just logically associating them mentally :P |
| 21:41.17 | p3nguin | The bases support six handsets. |
| 21:42.02 | p3nguin | Without seeing the base configuration, you could probably have a single extension ring all six handsets. |
| 21:42.57 | Jumpie | yea, but only 2 per base station can actually talk :( |
| 21:43.03 | Jumpie | well and...a 3rd on pots |
| 21:43.04 | *** join/#asterisk jaskew (~jdaskew@netblock-75-79-182-110.dslextreme.com) |
| 21:43.18 | Jumpie | i think it seems a silly limitation to allow 6 headsets per base |
| 21:43.49 | Jumpie | and not 6 concurrent channels |
| 21:44.37 | p3nguin | I agree, but that's how they built the hardware. |
| 21:44.40 | *** join/#asterisk murdock_ut (~chatzilla@mail.kimballequipment.com) |
| 21:54.56 | *** part/#asterisk mnick86 (~Matthias@whhem00002.cip.uni-regensburg.de) |
| 21:58.19 | Jumpie | interesting, gui is much faster in ie |
| 21:58.23 | Jumpie | fail @ seimens developres |
| 22:01.27 | *** join/#asterisk Akiraaa (~Akiraaaa@79.112.33.79) |
| 22:02.48 | Jumpie | hmm...these sip registrations are being rejected as unauthorized |
| 22:02.55 | Jumpie | even though im using the correct info...same as i did in the aastras |
| 22:02.58 | Jumpie | laame |
| 22:03.21 | *** join/#asterisk ikariW (~ikariW@74-92-245-181-Utah.hfc.comcastbusiness.net) |
| 22:05.45 | ikariW | Why would my CDR log "s" as the "dest" instead of the actual destination number? |
| 22:05.59 | Jumpie | oh...now i think it works |
| 22:06.31 | Jumpie | could invalid rtp port range effect even a sip registration? |
| 22:07.30 | leifmadsen | RTP != SIP |
| 22:07.30 | ChannelZ | newp |
| 22:07.30 | leifmadsen | they are different protocols |
| 22:07.30 | Jumpie | yea..i know..just... only thing i changed |
| 22:07.30 | Jumpie | and now its magically registered |
| 22:07.31 | Jumpie | hehe |
| 22:08.54 | ikariW | I'd even take just a hint . . . or an RTFM with a reference to a section? |
| 22:10.00 | Jumpie | ah...i thin i know what i did |
| 22:10.13 | Jumpie | is it normal for ip dect devices to have god aweful ping times when they are 5 feet away?100-120ms |
| 22:10.24 | Jumpie | ikariW i think there are some tweaks you can do |
| 22:10.36 | Jumpie | what does s refer to again? hehe |
| 22:10.51 | ikariW | Jumpie: ha ha. Right. |
| 22:11.00 | ikariW | Jumpie: Any idea on what I'd need to change? |
| 22:11.44 | ikariW | Jumpie: I wonder if I'm not returning from a gosub right or something. So I'm ending on s. |
| 22:12.19 | ikariW | Jumpie: But I don't see that anywhere in my dial plan. (no Goto(failure)) |
| 22:14.36 | ikariW | All of the src for outbound are correct. Also, all of the src/dest for inbound are correct. |
| 22:18.12 | *** join/#asterisk luckyaba (~lucky@ip72-194-218-169.sb.sd.cox.net) |
| 22:20.05 | ikariW | Well, If no one knows, does anyone want to make wild accusations? |
| 22:20.09 | ikariW | Congress? Congress is incorrectly setting dst in my CDR? ;) |
| 22:22.20 | *** join/#asterisk theshadow (~theshadow@c-24-8-143-181.hsd1.co.comcast.net) |
| 22:23.24 | theshadow | I've read through most of the O'Reilly PDF but I haven't run across what I need to be able to do. Can a SIP phone register anonymously or just be assigned a random ID? |
| 22:24.23 | theshadow | To put it into context we have dialing nodes (machines) that we need to test our software with, part of this is getting them to dial our desk phones. So it would be nice if they could register w/out having to be in the sip.conf file. |
| 22:24.33 | *** join/#asterisk Greek-Boy (~Greek-B0y@41.188.154.137) |
| 22:28.00 | *** join/#asterisk joobie (~joobie@mx01.anric.com.au) |
| 22:30.42 | [TK]D-Fender | theshadow: "autocreatepeer=yes" |
| 22:31.18 | theshadow | D-Fender ty |
| 22:34.22 | *** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/atheme.member.chainsaw) |
| 22:35.25 | *** join/#asterisk riddlebox (~riddlebox@75-132-225-75.dhcp.stls.mo.charter.com) |
| 22:35.45 | hardwire | riddle me this... what's in the box? |
| 22:35.52 | hardwire | wait.. that's not a riddle. |
| 22:35.55 | hardwire | just tell me whats in the box. |
| 22:37.02 | leifmadsen | hardwire: step 1) get a box, step 2) cut a whole in the box, step 3) put your... |
| 22:37.06 | leifmadsen | oh wait, not what you meant |
| 22:37.13 | leifmadsen | s/whole/hole/ |
| 22:37.21 | Qwell | leifmadsen: ... |
| 22:37.22 | hardwire | offers leifmadsen a riddlehole |
| 22:37.26 | leifmadsen | Qwell: weee! |
| 22:37.44 | Qwell | leifmadsen: step 4) have her open the box |
| 22:37.50 | leifmadsen | yay! |
| 22:38.01 | paulc | @leifmadsen: I'm back at my desk after a deluge of meetings - get my tweet? |
| 22:38.39 | hardwire | it's so not friday. |
| 22:38.43 | leifmadsen | paulc: yep! I DM'd you |
| 22:38.53 | leifmadsen | paulc: just said to shoot me an email :) |
| 22:39.00 | leifmadsen | hardwire: it certainly is not... |
| 22:39.04 | leifmadsen | I wished it was friday yesterday |
| 22:39.11 | leifmadsen | these 12 hr days are taking their toll |
| 22:39.13 | leifmadsen | tole? |
| 22:39.14 | hardwire | today is sorta my friday.. taking tomorrow off |
| 22:39.16 | leifmadsen | tadpole! |
| 22:39.37 | Slugs_ | i wish it was sat everyday |
| 22:39.52 | leifmadsen | Slugs_: amen! |
| 22:40.07 | leifmadsen | I wonder if that's what it's like being homeless |
| 22:40.10 | leifmadsen | they might be onto something there |
| 22:40.13 | Slugs_ | hehe; |
| 22:41.06 | Slugs_ | hardwire, i finally got that proj done |
| 22:41.29 | MT`AwAy | yay, the patch works |
| 22:42.08 | MT`AwAy | btw is there any way to make a feature request for asterisk? I got a patch, it would just need a config param to make things "clean" |
| 22:43.28 | *** join/#asterisk lnd (~lnd@92.41.108.226.sub.mbb.three.co.uk) |
| 22:44.15 | paulc | @leifmadsen: Ok cool - I'll drop you an email tonight once I'm home and free from the day job :) |
| 22:44.23 | leifmadsen | ok coolio :) |
| 22:44.39 | leifmadsen | paulc: include your skills, what you can do (and want to do), and your rate |
| 22:44.54 | leifmadsen | MT`AwAy: feature request, no -- feature with patch, yes |
| 22:45.24 | leifmadsen | MT`AwAy: if you have a patch that at least mostly implements what you want to accomplish, you can file it with the issue tracker at https://issues.asterisk.org |
| 22:45.55 | MT`AwAy | leifmadsen, problem is I have no idea how to add a sip per-user/peer parameter |
| 22:46.12 | *** join/#asterisk devdvd (~twister19@173-31-160-214.client.mchsi.com) |
| 22:48.20 | paulc | @leifmadsen will do sir *doffs hat* |
| 22:49.28 | Scorcerer | text chat embedded in some clients (like ekiga od eyebeam) should work out of the box (assuming that clients can call each other ? os there is some option i should enable? |
| 22:50.25 | *** join/#asterisk ariel_ (~chatzilla@c-24-127-196-248.hsd1.fl.comcast.net) |
| 22:51.06 | *** join/#asterisk riddlebox (~riddlebox@75-132-225-75.dhcp.stls.mo.charter.com) |
| 22:52.06 | *** join/#asterisk TheDavidFactor (~chatzilla@c-68-34-116-180.hsd1.md.comcast.net) |
| 22:52.44 | [TK]D-Fender | Scorcerer: * is NOT a text messaging platform and will not pass these on |
| 22:53.22 | Scorcerer | ok, thanks |
| 22:55.22 | theshadow | [TK]D-Fender: Alright they can now log in anonymously, and I can even call those nodes from my desk phone, but they can't seem to dial any of the defined extensions. |
| 22:56.04 | *** join/#asterisk KNERD (~KNERD@129.113.46.109) |
| 22:56.20 | [TK]D-Fender | thePerhaps you should look where the calls are being SENT |
| 22:59.34 | *** join/#asterisk Cain (~Geek@unaffiliated/cain) |
| 22:59.58 | theshadow | [TK]D-Fender: Could you elaborate a bit, in my extensions.conf I have the context loading [internal] within it I have the defined dial plan for extensions. Within the sip.conf file I have context set to default and in the [default] section I have type=friend host=dynamic context=internal |
| 23:00.14 | *** part/#asterisk MT`AwAy (~MagicalTu@2001:41d0:2:973::aeb) |
| 23:02.04 | [TK]D-Fender | theshadow: context=default is where it will look in the DIALPLAN. |
| 23:02.14 | [TK]D-Fender | theshadow: it is not a reference to a PEER to use as a template |
| 23:02.35 | [TK]D-Fender | theshadow: the call would eb looking in [default] in extensions.conf, not "incoming" |
| 23:02.39 | [TK]D-Fender | or "internal" |
| 23:02.51 | theshadow | i see |
| 23:03.01 | hardwire | Slugs_: howso? |
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| 23:04.49 | *** part/#asterisk ikariW (~ikariW@74-92-245-181-Utah.hfc.comcastbusiness.net) |
| 23:05.13 | TJNII | "browser based IDE" Grnahhhhhhh.... |
| 23:11.27 | Jumpie | hmmm |
| 23:11.42 | Jumpie | i was thinking the siemens gigaset a580ip could have headsets registered to multiple base stations |
| 23:11.46 | Jumpie | was hoping for a redundant type setup |
| 23:14.57 | [TK]D-Fender | bbl |
| 23:15.48 | p3nguin | ~a580 |
| 23:15.49 | infobot | [~A580] The Siemens Gigaset A580 IP with ECO DECT technology is multi-line so you are free to register up to 6 handsets for 6 SIP accounts from different providers and make up to 3 calls in parallel: 2 VoIP calls and 1 fixed-line call. See http://gigaset.com/hq/en/product/GIGASETA580IP.html for details. Cost is about $70USD for one handset with base. Extra A58H handsets are around $40USD each. |
| 23:16.17 | p3nguin | Does their product page mention anything about that? Maybe there's a data sheet? |
| 23:21.19 | Jumpie | p3nguin i had thought they did..and i talked to a guy in here a few days ago that used them extensively and said he thought you could |
| 23:21.21 | Jumpie | i mean its not a huge deal |
| 23:21.32 | Jumpie | i'll just dedicate 2 headsets to one and the 3rd to the other |
| 23:21.38 | Jumpie | its why i got the 2nd base station |
| 23:21.59 | Jumpie | i get that the 6 headset capability is more for accessibility |
| 23:22.15 | Jumpie | but customer wants to be able to theoretically use all 3 at a given time..which cant be done tied to a single base station so i got 2 :D |
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| 23:26.24 | Jumpie | hmm...it does seem to be holding the registration on both base stations..but not at the same time |
| 23:26.26 | p3nguin | I would need to get some handsets and couple bases before I could guess how to set up the things. |
| 23:26.33 | Jumpie | it seems like there is a back and forth registration |
| 23:26.44 | Jumpie | i wonder if its available on demand if channel is unavail |
| 23:27.04 | Jumpie | i figure 3 sip registrations semi chattiness isnt too bad |
| 23:27.13 | Jumpie | its not that hard actually...now i have the hang of it |
| 23:27.23 | Jumpie | the key is you have to upgrade th base firmware FIRST from the headset |
| 23:27.28 | Jumpie | or its pretty unaccessible |
| 23:27.35 | Jumpie | and..apparently IE is much better than ff |
| 23:27.43 | Jumpie | like bya factor of 100 |
| 23:28.15 | p3nguin | WHAT?! |
| 23:28.34 | ChannelZ | is a 'u' with a line over it pronounced like 'you' ? |
| 23:29.01 | Jumpie | i cant remember all the funny symbol types heh |
| 23:29.07 | Jumpie | i do remember what an umlat is :P |
| 23:29.18 | p3nguin | Has firefox gone that far down that IE is now better? |
| 23:29.39 | Jumpie | it has somethin to dow ith the javascripting bs |
| 23:29.44 | Jumpie | i mean normally i have to use ff on most things |
| 23:29.50 | p3nguin | Oh, bad web devs. Got it. |
| 23:29.51 | Jumpie | this is first time i've had to 'resort' to ie over ff haha |
| 23:29.53 | Jumpie | yeah |
| 23:29.58 | Jumpie | a ton of forum complaints |
| 23:30.15 | p3nguin | What about if you used IE Tab to load the IE engine in firefox? Would that solve it? |
| 23:30.17 | Jumpie | also it tries to auto reg you forsome trial gigaset voip thing..which is rather chatty at first |
| 23:30.24 | Jumpie | p3nguin heh haven tried that |
| 23:30.26 | Jumpie | went the easy route |
| 23:30.40 | Jumpie | i remember using that once..you could simulate several browsers couldnt you |
| 23:30.46 | p3nguin | right click, click use IE. Not much easier than that. |
| 23:30.56 | Jumpie | <PROTECTED> |
| 23:30.56 | Jumpie | <PROTECTED> |
| 23:30.58 | Jumpie | hehe |
| 23:31.02 | Jumpie | like...every minute |
| 23:31.04 | Jumpie | they are swapping |
| 23:31.05 | TJNII | Mozilla's software is so bloated nowadays. I've stopped using both Firefox and Thunderbird. |
| 23:31.15 | Jumpie | not to mentnion every time they claim to fix the memory issue |
| 23:31.20 | Jumpie | its still eatin like 800mb with a few tabs |
| 23:31.38 | Jumpie | invite/register traffic isnt actually too bad overhead wise is it? |
| 23:31.42 | Jumpie | if 3 did it every minute |
| 23:31.48 | Jumpie | would that be a considerable network hit? |
| 23:32.13 | Jumpie | im still deciding if i want to disassociate the 3rd headset from the first baes |
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| 23:47.24 | ruben23 | hi why do i experience choppy line even my bandwidht is 10 Mbps and im only using around 1Mb on my voice traffic..any explantion. |
| 23:47.48 | ChannelZ | latency latency latency |
| 23:48.30 | ruben23 | ChannelZ:my latency is steady... |
| 23:48.48 | ChannelZ | Doesn't really matter how fast the packets can get from one place to another if they're arriving at inconsistent intervals |
| 23:49.57 | ruben23 | ChannelZ: any possible correction i can do on my end.. |
| 23:50.37 | ChannelZ | You can try toying with your jitter buffer if that is indeed what is going on |
| 23:51.57 | ruben23 | ChannelZ: jitter is..? |
| 23:52.23 | ChannelZ | when packets show up inconsistently |
| 23:53.09 | ruben23 | like having latency 220ms then drop 240ms then again 230ms then back to 220ms |
| 23:54.45 | ChannelZ | Between you and the remote end? That seems a little high |
| 23:56.36 | Jumpie | lol i cant seem to unregister this handset from the original base station |
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