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| 00:16.31 | *** join/#asterisk joobie (~joobie@mx01.anric.com.au) |
| 00:16.37 | joobie | hey guys.. silly / noob question |
| 00:16.55 | joobie | alaw / ulaw.. 64 kbit/s is that 32kbit/s per upstream and 32kbit/s per downstream? |
| 00:17.04 | joobie | or 128kbit/s for total |
| 00:17.14 | joobie | trying to figure out actual bandwidth use for a call made for 1 minute on alaw vs g729 |
| 00:17.28 | WIMPy | It's always per direction. |
| 00:17.37 | joobie | ahh k |
| 00:17.46 | joobie | so if i wanted to calculate 'rough' bandwidth usage for a call for 1 min |
| 00:17.52 | joobie | just 128kbit * 60 ? |
| 00:18.21 | WIMPy | yes |
| 00:18.24 | joobie | thanks |
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| 00:21.48 | carrar | joobie, 128k would be two calls |
| 00:22.07 | carrar | roughly |
| 00:22.23 | carrar | 1 call ulaw/alaw works out to be about 88kbps |
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| 01:40.21 | maxagaz | Hi, |
| 01:40.38 | maxagaz | how to have TLS support when compiling Asterisk 1.8.2 ? |
| 01:41.19 | maxagaz | Isn't there something to check in the menuselect to enable it ? |
| 01:43.08 | *** join/#asterisk tzafrir_laptop (~tzafrir@bzq-218-155-147.cablep.bezeqint.net) |
| 01:53.10 | pabelanger | maxagaz: check if res_srtp.so is enabled |
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| 01:56.21 | *** join/#asterisk asteriskmonkey (~darlek@74-51-38-204.telnetcommunications.com) |
| 01:56.45 | asteriskmonkey | whats the max size a jitterbuffer can be for iax? |
| 02:02.45 | *** join/#asterisk DrkShadow (~andrew@host-72-175-240-62.static.bresnan.net) |
| 02:03.55 | DrkShadow | hey, I have a slew of Ass-tra 57i's and I try dialing an international number. The number turns out to not work, the server replies with 503 "Service Unavailable", and ... the phone _immediately_ tries again, 10 times in under 20 seconds, until I get 603 "Declined" and an e-mail from my phone provider saying they've cut off international service for odd dialing forms. ANyone know what's up? |
| 02:05.59 | maxagaz | pabelanger: ok, so it goes all together with strp... thanks! |
| 02:08.34 | pabelanger | maxagaz: my mistake, TLS support is dependent on OpenSSL-dev, IIRC. |
| 02:09.04 | maxagaz | pabelanger: how to make sure it is installed then ? |
| 02:09.24 | pabelanger | maxagaz: res_crypto.so |
| 02:09.26 | maxagaz | pabelanger: I mean compiled in Asterisk ? |
| 02:09.34 | maxagaz | pabelanger: ok, thanks! |
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| 02:44.56 | Jouva | So if I have a softphone behind a NAT and * on a public server, what do I have to do to let audio from behind the NAT through? Port forwarding? Or is there a way to configure the client and asterisk to get it through? Or? |
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| 02:47.51 | ChannelZ | no you pretty much need port forwarding, unless the firewall understands SIP packets and can watch the RTP ports being requested and do it its self |
| 02:54.27 | Jouva | hmmm, it might actually. I'm running DD-WRT here |
| 02:54.31 | Jouva | So lemme check that |
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| 02:59.08 | *** join/#asterisk mzb (~mzb@ppp108-88.static.internode.on.net) |
| 03:00.08 | mzb | Hi all, can I create a Dial() (or similar) that continues the context after either 1) the called number rings 'n' times, OR 2) the call is answered? |
| 03:01.24 | WIMPy | 'core show application dial' |
| 03:01.40 | mzb | Absolute timeout and a timeout in the Dial() kind of work, but don't allow for mobile phone 'seek' times |
| 03:02.07 | mzb | ie: all I'm trying to do is pass caller ID to a mobile phone and then hangup |
| 03:03.53 | mzb | I guess 'G' helps the answer issue |
| 03:04.28 | mzb | oh ... L |
| 03:04.30 | mzb | hmm |
| 03:04.45 | WIMPy | If you want to hangup when the call is answered, L would be it. |
| 03:08.47 | mzb | k, thanks ... trying it now |
| 03:10.13 | Jouva | argh trying to read how to use milkfish but the milkfish wiki is just broken right now |
| 03:12.32 | antiwire | the sip proxy? |
| 03:12.41 | Jouva | yes |
| 03:13.29 | antiwire | I've used siproxd |
| 03:14.05 | antiwire | siproxd with a dyndns dynamic hostname |
| 03:14.14 | Jouva | yeah but this is built into my dd-wrt router |
| 03:14.25 | antiwire | yep |
| 03:14.36 | antiwire | I never got it working well on my wrt54gs |
| 03:15.28 | mzb | hrmm ... or maybe S(x) |
| 03:15.58 | Jouva | argh :/ |
| 03:21.03 | mzb | no, can't get L, G or S to make any difference |
| 03:21.34 | mzb | err |
| 03:21.54 | *** join/#asterisk Russ (foobar@ip70-176-251-1.ph.ph.cox.net) |
| 03:24.58 | mzb | I've probably got something wrong with the context |
| 03:27.17 | maxagaz | asterisk can't load srtp module when selinux is set as 'enforcing', how to make an selinux rule to make it work ? |
| 03:27.43 | maxagaz | (instead of disabling selinux) |
| 03:27.46 | mzb | http://openpaste.org/en/24461/ |
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| 03:30.41 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
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| 04:24.09 | antiwire | Hey, should I be adjusting tx or rx gain if I am a SIP client and a person calls in over copper and I hear myself echoing? |
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| 04:33.27 | Carl0s- | Haha... Just for a laugh I tried to /join #porn .. |
| 04:35.17 | rushowr | hello all! Quick question, no issue, I just happen to be finally be migrating to func_ODBC instead of using a library of macros I made a few years back for accessing MySQL. I've searched fairly exhaustively and come up with very little information on retrieving multiple columns from a query, using Set(ARRAY(var1,var2,var3)=${ODBC_SOMEQUERY()})...My problem is, I can clearly see the 3 values... |
| 04:35.19 | rushowr | ...come back (by looking at the verbose log of the Set command) but var2 and var3 do not contain the values they should |
| 04:36.05 | rushowr | if need be I can provide a code snippet of the actual code as well as the related odbc entry and the log entry |
| 04:36.48 | rushowr | looks sheepish for asking for help on IRC twice in two days...That is probably more than he's ever requested assistance since 2004 |
| 04:39.51 | antiwire | any ideas for stopping echo on the SIP softphone end of a call that is incoming on copper/FXO? |
| 04:40.24 | *** join/#asterisk fofware (~Fabian@200.82.52.7) |
| 04:40.42 | rushowr | have you recorded the call before it connects to the softphone, to see if asterisk "hears" the echo? |
| 04:40.59 | rushowr | that'll help you determine if it's the softphone, the network, or Asterisk |
| 04:41.30 | rushowr | i.e. Answer()...MixMonitor(...) |
| 04:41.38 | rushowr | ? sorry, just spitballing |
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| 04:43.31 | antiwire | From the SIP client dialing out over the copper, I also hear echo during the outbound ring back |
| 04:43.55 | antiwire | but the phone on the other end of the copper PSTN doesn't have any echo |
| 04:44.39 | rushowr | ok, so, we know that it's on the "near" leg |
| 04:44.57 | rushowr | however, is there echo on calls using the softphone but NOT the copper? |
| 04:45.03 | antiwire | no |
| 04:45.19 | antiwire | only on calls that cross between copper and SIP |
| 04:45.22 | rushowr | is the FXO going through a channelbank or a card? |
| 04:45.36 | rushowr | (direct card port or a channel bank with t1/e1 card)( |
| 04:45.49 | antiwire | nope, standard POTS line going into the Sangoma U100 USBFXO |
| 04:45.56 | rushowr | tries to remember how he fixed this a couple years back when he was doin' a call center |
| 04:46.00 | rushowr | hrm...one sec mate |
| 04:46.12 | rushowr | another quick one |
| 04:46.22 | rushowr | well, nevermind |
| 04:46.27 | rushowr | you probably haven't tried it |
| 04:46.31 | rushowr | one sec |
| 04:47.12 | rushowr | do you have more than one sip softphone or other sip endpoint? if so, do they ALL result in echo on calls over the FXO, or just one? |
| 04:47.34 | antiwire | I'll test that right now |
| 04:47.45 | rushowr | right on |
| 04:48.51 | rushowr | also, check chan_dahdi.conf |
| 04:48.57 | rushowr | echocancel=yes ? |
| 04:49.17 | antiwire | yep, that's set |
| 04:49.33 | rushowr | right on |
| 04:50.15 | rushowr | you may need to tweak some of the settings in that section. Another thing you may end up having to do is contacting the circuit provider, I know I've had to work with a couple of 'em in the past to nail down a troublesome echo |
| 04:51.46 | rushowr | I've got to run mate, good luck1 |
| 04:51.48 | rushowr | ! |
| 05:03.28 | antiwire | no echo between two sip clients |
| 05:04.50 | antiwire | same echo with a different sip client and same POTS line |
| 05:05.11 | antiwire | the sip clients can hear themselves but the POTS end is fine |
| 05:14.52 | antiwire | could it be that the Sangoma U100 USBFXO adapter is just a piece of crap? |
| 05:14.54 | *** join/#asterisk ChannelZ (channelz@burner.com) |
| 05:14.59 | antiwire | could it be that simple? |
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| 05:24.12 | antiwire | I restarted the whole box and it's fine now... |
| 05:24.16 | antiwire | *rage* |
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| 05:42.34 | Corydon76-home | USB adapters tend to have issues with static electricity. This is why Digium discontinued its USB adapter several years ago |
| 05:43.04 | Corydon76-home | You might try adjusting environment controls to boost the humidity, such that static discharges aren't a problem |
| 05:44.06 | *** join/#asterisk antiwire (~antiwire@unaffiliated/antiwire) |
| 05:49.30 | Russ | or maybe a shorter cable? |
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| 06:09.17 | maxagaz | why do I get this error on a fresh install of Asterisk : |
| 06:09.21 | maxagaz | loader.c: Error loading module 'res_pktccops': /usr/lib/asterisk/modules/res_pktccops.so: cannot open shared object file: No such file or directory |
| 06:09.22 | maxagaz | ? |
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| 06:33.56 | verywiseman | i want to make mobile extension where at any extension , i can enter my extension and password to change current extension to my extension , how can i do that? |
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| 07:08.52 | ChannelZ | google asterisk hot-desking for ideas |
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| 07:18.14 | AreolaMonster | hello, I was under the impression that exten => s,1,Answer() |
| 07:18.14 | AreolaMonster | exten => s,1,Answer() |
| 07:18.18 | AreolaMonster | hello I was under the impression that exten => s,1,Answer() would pick up any call that didn't match a number in extensions.conf but I keep getting an error that the extension is not found. I'm using the default asterisk install on debian. |
| 07:18.33 | AreolaMonster | sorry for the multple paste |
| 07:18.43 | AreolaMonster | can someone clarify this for me please |
| 07:31.58 | ChannelZ | no. s is a special extension used mostly when an extension is not given |
| 07:32.58 | ChannelZ | like on a DAHDI channel coming from an analog interface card, or a SIP call with no extension, etc. |
| 07:34.46 | AreolaMonster | I have a sip trunk configured and I can see the call reach the asterisk machine but immediately gets rejected, the next priority is to playblack(hello-world) |
| 07:35.28 | ChannelZ | pastebin some console output of the call |
| 07:35.33 | ChannelZ | with verbose on 3 or so |
| 07:35.50 | AreolaMonster | ok, will do. thank you |
| 07:41.32 | AreolaMonster | http://pastebin.com/xigCY2pD |
| 07:44.03 | carrar | exten => _1NXXNXXXXXX,1,Answer() |
| 07:45.04 | carrar | or exten => _X!,1,Answer() |
| 07:45.32 | AreolaMonster | cool, let me try that |
| 07:46.23 | ChannelZ | or better yet exten => 13234441234,1,Answer() etc since that appears to be your DID |
| 07:46.37 | ChannelZ | </asumption> |
| 07:48.41 | AreolaMonster | it's not, thanks that works. I thought s would simply take any # |
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| 08:05.18 | AreolaMonster | thanks for all the help, very much appreciated. |
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| 08:07.01 | verywiseman | i make this dial plan http://www.fpaste.org/RtEk/ , when i enter password 2011, it is said password is incorrect , why? |
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| 08:15.28 | Polysics | hello |
| 08:15.43 | Polysics | where can i get some more info on why SIP users can't log in? |
| 08:16.08 | Polysics | the system was working ok, but now is rejecting all SIP users, i am using realtime |
| 08:16.31 | tamiel | Polysics: sip debug ? |
| 08:16.34 | Polysics | it correctly says wrong password to those that exist in the table, and no matching peer if i make up a non-existent one, so it does speak to the db |
| 08:17.06 | Polysics | asterisk 1.6.2 |
| 08:17.09 | tamiel | Polysics: log sql requests to see queries |
| 08:17.10 | Polysics | checking debug |
| 08:18.19 | Polysics | how do i log sql? debug doesn't say anything mor than i know |
| 08:18.22 | tamiel | Polysics: and when sip register is failing, replay sql query manually to show result |
| 08:19.00 | tamiel | Polysics: depends on your database |
| 08:19.10 | Polysics | using mysql |
| 08:22.03 | tamiel | Polysics: log=/var/log/mysqld.log in [mysqld] section for example |
| 08:22.16 | tamiel | Polysics: and restart mysql server |
| 08:22.48 | tamiel | Polysics: user running mysql must have write access to /var/log/mysqld.log |
| 08:23.13 | tamiel | Polysics: be carefull, this is only for debugging purpose, don't use in production |
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| 08:24.31 | nunne | Is anyone else experiencing problems with getting MOH and announcement files to play in queues under asterisk 1.8? And anyone have a working fix for it? :) (Using the CentOS yum repo) |
| 08:26.03 | Polysics | tamiel: there isn't any way to display it in the console? |
| 08:26.04 | Polysics | weird |
| 08:26.22 | verywiseman | i make this dial plan http://www.fpaste.org/RtEk/ , when i enter password 2011, it is said password is incorrect , why? |
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| 08:30.04 | tamiel | Polysics: in asterisk console you mean ? |
| 08:30.24 | Polysics | tamiel: yes, does any way exist? |
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| 08:34.31 | tamiel | Polysics: I don't remember this |
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| 08:34.42 | tamiel | Polysics: Are you using odbc ? |
| 08:35.04 | Polysics | tamiel: no, res_mysql |
| 08:35.30 | nunne | verywiseman: is dtmf working as it should? try read and then saydigits to see if it's read correctly |
| 08:36.58 | verywiseman | nunne, ok |
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| 08:40.23 | schmidts | good morning |
| 08:40.36 | E-bola | Morning |
| 08:40.41 | tamiel | Polysics: it seems you can't |
| 08:41.51 | tamiel | Polysics: you must do debugging in mysql |
| 08:42.07 | Polysics | tamiel: ok, already set it up then, thanks |
| 08:43.30 | *** join/#asterisk v1s (~v1s@202.84.107.67) |
| 08:45.15 | v1s | if I have some packet loss on my network is it beter to set the codec packet sizes bigger or smaller? |
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| 08:48.13 | schmidts | vls smaller, cause if you loose one packet you will loose less of the stream |
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| 09:14.20 | v1s | what would be next codec up from g729 for quality and bandwidht? |
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| 09:20.50 | tobi- | thanks for the link leifmadsen ill check it out |
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| 10:26.22 | asterisk-learner | hi, if i have HPET but running a centos core of 2.6.18, is ztdummy going to use it (HPET) or do i have to modify smthg else ? |
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| 10:40.49 | E-bola | Can somebody verify that notifyringing is meant to control whether hints also work with the ringing state? The docs are a bit vague |
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| 10:43.53 | ijpalmer | H, I have * 1.6.2.6 installed. Part of my dialplan has an incoing call which is recorded to a wav file, then the dialplan plays back the file, the trouble is the file it looks for when playing back the file is a slin not wav file so it can't find the file. How do I get playback to play the wav file. Thanks |
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| 11:13.44 | krion | isn't in the topic 2011/01/18 instead of 2010/01/18 ? |
| 11:16.19 | E-bola | hehe yep |
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| 11:38.08 | tuxx- | pommm |
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| 11:58.00 | wdoekes2 | ijpalmer: it's not actually looking for a .slin file though |
| 11:58.31 | wdoekes2 | something else is probalbly wrong, wrong bitrate, etc.. |
| 11:59.22 | wdoekes2 | and yes.. that message is awfully confusing |
| 12:00.08 | wdoekes2 | perhaps wrong path, or path with extension |
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| 12:06.35 | ijpalmer | wdoekes2: thanks, it does look like the worng path, but the only difference in the path is between the extensions, when the file is recorded, it is recorded with a wav extension when it tries to playback it has a slin extension |
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| 12:35.31 | joachim_- | What is the best solution for redundancy, when running multiple asterisk servers & mysql servers ? |
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| 12:59.52 | Chainsaw | Qwell, leifmadsen: 2001:470:e0d4::e9 appears to be unreachable (downloads.asterisk.org). It falls back to IPv4 eventually, which is up. Who should I bother about this please? |
| 13:00.11 | leifmadsen | Chainsaw: kpfleming |
| 13:00.21 | Chainsaw | leifmadsen: Thank you. |
| 13:00.35 | Chainsaw | leifmadsen: File a bug or e-mail? |
| 13:00.39 | leifmadsen | email |
| 13:00.47 | Chainsaw | leifmadsen: Cheers. Assuming @digium.com |
| 13:00.55 | leifmadsen | there should be a support page on digium.com |
| 13:00.56 | leifmadsen | or asterisk.org |
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| 13:02.35 | joachim_- | Im trying to find information about using kamailio as a sip-router.. Any1 familiar with this? I have multiple asterisk servers and multiple mysql servers. Is a SIP proxy the way to go here? |
| 13:15.42 | tobi- | ringinuse=no in queue settings seems to have to effect, inuse agents still get the call waiting, thats intended? |
| 13:15.57 | tobi- | seems to have _no_ effect |
| 13:19.32 | *** join/#asterisk binbash_ (~peter@ip4da5c213.direct-adsl.nl) |
| 13:20.17 | Tozz_ | if I am not mistaken |
| 13:20.21 | Tozz_ | you have to set calllimit= too |
| 13:20.24 | Tozz_ | for ringinuse to work |
| 13:22.41 | binbash_ | Hmm, i used the fixrouting script from voip-info.org but a got a issue with that with calls going to another subnet.. |
| 13:22.46 | binbash_ | Anybody had that issue :-)? |
| 13:25.39 | Tozz_ | another subnet? |
| 13:25.46 | leifmadsen | tobi-: what version of asterisk? |
| 13:26.06 | leifmadsen | ~newbook |
| 13:26.06 | infobot | Help review the 3rd edition of the Asterisk book published by O'Reilly at http://ofps.oreilly.com/titles/9780596517342/ Released under a Creative Commons license: http://creativecommons.org/licenses/by-nc-nd/3.0/us/ |
| 13:26.19 | leifmadsen | that site discusses how to get ringinuse working in the ACD chapter |
| 13:27.50 | leifmadsen | (the trick is making sure sip.conf has callcounter=yes enabled) |
| 13:29.28 | tobi- | leifmadsen: its 1.8.2.1 |
| 13:29.38 | leifmadsen | then that documentation will apply to you |
| 13:30.27 | E-bola | Is it correct that there was only 1 change from 1.8.2rc1 to final? |
| 13:30.35 | binbash_ | Tozz_ yeah so |
| 13:30.40 | tobi- | alright thanks for the hint |
| 13:30.41 | binbash_ | my Asterisk is 10.100.2.15 |
| 13:30.48 | binbash_ | and it needs to find a phone in 10.100.3.15 |
| 13:30.53 | binbash_ | but the rewrite fix on voip-info |
| 13:30.59 | leifmadsen | E-bola: yes, that's what the ChangeLog says |
| 13:31.16 | binbash_ | only works with 10.100.2.0/24 |
| 13:31.25 | binbash_ | http://www.voip-info.org/wiki/view/Asterisk+High+Availability+Solutions |
| 13:31.56 | binbash_ | When i enable it like that, so rewrite the from ip to my failover IP, i only have audio in my subnet .. so 10.100.2.x and not 10.100.3.x |
| 13:34.07 | *** join/#asterisk i_heart_asterisk (~chatzilla@firewall-a.buf.ny.i-evolve.net) |
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| 13:37.40 | tobi- | thanks leifmadsen that did it |
| 13:37.50 | leifmadsen | of course it did, yay for documentation! :) |
| 13:37.58 | tobi- | true :) |
| 13:41.32 | joachim_- | Im trying to find information about using kamailio as a sip-router.. Any1 familiar with this? I have multiple asterisk servers and multiple mysql servers. Is a SIP proxy the way to go here? |
| 13:41.41 | binbash_ | Nobody :-)? then i'll try the mailing list. |
| 13:41.52 | binbash_ | joachim_ i have some yeah, i use opensips as a sip router. |
| 13:42.41 | joachim_- | and then u use multiple asterisks with replicated configs and database servers? |
| 13:42.47 | binbash_ | yup |
| 13:43.09 | joachim_- | how long downtime u get if one server go down? |
| 13:43.28 | binbash_ | <PROTECTED> |
| 13:44.33 | joachim_- | Nice! I've been trying to use Kamailio but cant say i've had any luck... Is opensips pure proxy? |
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| 13:45.27 | binbash_ | Yeah :-) i only catch the subcribe stuff, and the rest goes to one of the asterisk boxes |
| 13:45.59 | binbash_ | so i register my phones on opensips, but all the actually calling and stuff is done by one of my * boxes |
| 13:46.48 | joachim_- | Hmmm. I cant do that, we administer the servers from a self made gui, i need to have all the customer accounts in mysql databases |
| 13:47.39 | binbash_ | I know, |
| 13:47.41 | *** join/#asterisk slidesinger (~slidesing@c-68-44-99-163.hsd1.nj.comcast.net) |
| 13:47.42 | binbash_ | I have that :-) |
| 13:47.54 | binbash_ | the subscriber table from opensips is a view of the asterisk table |
| 13:48.27 | joachim_- | OK. That could work |
| 13:49.29 | bip | ppI have been requested the following feature: when some caller ID (telephone number) calls it should receive no answers but trigger an event like sending an email sayng that a call was received from that number. Can this be done with asterisk ? any hint in ther right direction will be appreciated thanks! |
| 13:51.24 | Tozz_ | do an IF on the callerid if it matches some string |
| 13:51.26 | leifmadsen | exten => _NXXNXXXXXX/4165551212,1,System(mutt -s "HELP" < /tmp/my_message.txt) |
| 13:51.30 | leifmadsen | same => n,Hangup() |
| 13:51.34 | Tozz_ | what leifmadsen says ;) |
| 13:51.34 | leifmadsen | bip: see above |
| 13:52.02 | bip | yes thanks |
| 13:52.46 | binbash_ | joachim_ i'm sure it works :P. |
| 13:54.46 | *** join/#asterisk Kobaz (~kobaz@its.kobaz.net) |
| 13:54.49 | Kobaz | is there something like iperf for windows |
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| 14:12.23 | tobi- | is there a good pstn-sip provider located in france? |
| 14:13.34 | *** join/#asterisk af_ (~getsmart@78.134.22.194) |
| 14:14.41 | Tozz_ | tobi-: if you are not in france you probably wont have any luck. France telecom rules require that you have an address in france |
| 14:14.47 | Tozz_ | to get a .fr DID number |
| 14:15.35 | tobi- | i see thanks |
| 14:20.45 | Katty | goooooooooooooooooooooooooooood morning all you wonderful people!!!! |
| 14:21.34 | Chainsaw | Good morning Katty! *hug* |
| 14:22.41 | Tozz_ | good morning? its 15:22! |
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| 14:23.03 | Katty | hugs Chainsaw |
| 14:23.09 | Katty | Naikrovek: sorry to hear you are feeling poorly sir )= |
| 14:23.25 | Katty | Naikrovek: hope your head gets better quickly |
| 14:25.04 | Katty | infobot: seen hmmhesays |
| 14:25.14 | infobot | hmmhesays <~hmmhesays@24-116-107-203.cpe.cableone.net> was last seen on IRC in channel #asterisk, 132d 18h 46m 50s ago, saying: '*hand'. |
| 14:26.39 | Kobaz | it's the Katty! |
| 14:27.04 | Katty | hugs on Kobaz |
| 14:27.08 | Kobaz | yays |
| 14:27.08 | Katty | tis. |
| 14:27.16 | Kobaz | do you know if there's an iperf type program for windows |
| 14:27.42 | Katty | what is iperf |
| 14:28.08 | tzanger | it is a client/server thing that measures network performance between two computers |
| 14:28.23 | Katty | hmm. |
| 14:28.27 | Katty | like...bandwidth? |
| 14:28.56 | Chainsaw | Kobaz: There is a port of the real thing apparently: http://linhost.info/2010/02/iperf-on-windows/ |
| 14:29.43 | *** join/#asterisk millerx (~millerx@rrcs-24-73-84-250.se.biz.rr.com) |
| 14:29.45 | Katty | i think there's ms network monitor |
| 14:31.27 | Kobaz | oh mm |
| 14:31.38 | Kobaz | the iperf official site is just a *nix version |
| 14:31.41 | Kobaz | i'll take a look |
| 14:33.18 | Chainsaw | Kobaz: Note the comments lower down, they moved the download URL. |
| 14:34.05 | Katty | Kobaz: http://www.microsoft.com/downloads/en/details.aspx?FamilyID=983b941d-06cb-4658-b7f6-3088333d062f&displaylang=en <- personally i'd just use wireshark |
| 14:34.08 | Kobaz | k |
| 14:34.16 | Katty | Kobaz: but that's probably not what you're looking for really |
| 14:34.18 | Kobaz | wireshark for bandwidth testing? |
| 14:34.30 | Kobaz | i'm having a problem with an adtran 1234poe switch |
| 14:34.35 | Kobaz | i think i know what the problem is |
| 14:34.36 | Katty | well the network monitor has a time offset column |
| 14:34.53 | Kobaz | i need like port-to-port throughput,latency,jitter |
| 14:35.03 | Chainsaw | Kobaz: Well the filtering language in Wireshark is pretty powerful. Someone's probably written a bandwidth meter in it by now :) |
| 14:35.06 | Kobaz | the poe switch is duplicating all the traffic on all the vlans |
| 14:35.30 | Kobaz | like, if someone is sending 100mbit/sec from port 1 to port 2... well that traffic is also going to all the other ports |
| 14:35.39 | Kobaz | and killing throughput for everyone |
| 14:35.47 | Kobaz | i can see it in jnettop |
| 14:36.04 | Kobaz | there's traffic that's a souce and dest that's not my server going through that hic |
| 14:36.08 | Kobaz | nic |
| 14:37.31 | Kobaz | my server is 192.168.1.201 and with jettop or tcpdump i'm seeing stuff like 192.168.1.57 <-> 192.168.1.48 that's not broadcast traffic |
| 14:37.55 | Kobaz | looks like it's acting as a hub and not a switch, it's broadcasting traffic to all ports |
| 14:41.25 | *** join/#asterisk PMantis (~sswitzer@cpe-67-244-159-142.rochester.res.rr.com) |
| 14:43.06 | Kobaz | the arp table is empty |
| 14:43.11 | Kobaz | except for one entry |
| 14:43.48 | E-bola | if if its acting like a hub its broken, case solved? |
| 14:43.55 | *** join/#asterisk cesar_CR (~cesar@201.192.86.30) |
| 14:43.56 | Chainsaw | Or the CAM is full. |
| 14:44.13 | Chainsaw | In which case flooding the traffic back out of everything except the port it came from is correct & expected behaviour. |
| 14:44.31 | E-bola | Chainsaw: i didnt think new switches still did it that way? |
| 14:44.43 | Chainsaw | E-bola: What other way could they use? |
| 14:44.52 | E-bola | drop the traffic |
| 14:45.01 | E-bola | since the other method is very abuse related |
| 14:45.27 | Chainsaw | E-bola: Your suggestion would break multicast. |
| 14:45.28 | PMantis | Hi guys. I recently upgrades a customer system from 1.6.1.6 to 1.6.2.15 and they are now seeing frequent dropped calls at 30 seconds. So far, as I understand it, the issue always involves a soft phone, but is with more than one computer. Is there anything known with 1.6.2.15? |
| 14:45.48 | Chainsaw | PMantis: Drops after ~3 minutes tend to be silence detection. |
| 14:45.51 | blee | PMantis: eyebeam? |
| 14:45.56 | Chainsaw | PMantis: Still worth checking that. Please turn it off. |
| 14:46.29 | Kobaz | E-bola: that seems a little broken.. but the arp table being empty is a clear indication something is wrong somewhere |
| 14:46.53 | Kobaz | E-bola: configure each workstation to drop the traffic too? |
| 14:47.09 | Kobaz | that's not gonna fix anything, it's still killing the throughput for every other device on the network |
| 14:47.11 | E-bola | Kobaz: huh? |
| 14:47.32 | E-bola | I was simply commenting on the concept fo switches turning into hub's if you flood them |
| 14:48.14 | PMantis | Chainsaw, Sometimes the drops occur while he's leaving someone a voicemail, sometimes, they're in conversation. Doesn't seem to be silence detection, but I adjusted lots of timeouts when this was first reported. One sec... |
| 14:48.21 | PMantis | blee, Guess I should ask... |
| 14:48.43 | Kobaz | E-bola: the arp table is empty |
| 14:48.50 | Kobaz | there's no port-macaddress associations |
| 14:48.57 | Kobaz | so it's going to broadcast to all ports |
| 14:48.59 | Chainsaw | E-bola: The main thing is that the CAM should be appropriately sized for the traffic flow. |
| 14:49.07 | Kobaz | arp might be disabled or something? |
| 14:49.13 | Kobaz | i just put in a ticket with adtran now |
| 14:49.51 | E-bola | Kobaz: did u try to just restart the switch? |
| 14:49.56 | Kobaz | hah |
| 14:50.14 | Kobaz | restart the switch? during hours? with 40 people on the phone? |
| 14:50.30 | Chainsaw | Turning things off and back on again works. Even on recalcitrant hydraulic lifts. |
| 14:51.13 | E-bola | Kobaz: just say somebody tripped in a wire :) |
| 14:51.18 | E-bola | it wont be offline for long anyway :0) |
| 14:51.23 | devmod | If I want all outbound sip traffic to go through a sip proxy, what do I have to do besides setting outboundproxy=xxxx on sip.conf ? |
| 14:52.16 | *** part/#asterisk TobSnyder (~schneider@dslb-088-073-184-208.pools.arcor-ip.net) |
| 14:52.19 | *** join/#asterisk espiceland (~erin@nat/digium/x-tknffcaguortuzyd) |
| 14:52.34 | *** join/#asterisk Tim_Toady (~moi@178.128.134.192.dsl.dyn.forthnet.gr) |
| 14:53.59 | i_heart_fs | hi |
| 14:55.37 | *** join/#asterisk jamko (~chatzilla@173-162-11-81-naples.hfc.comcastbusiness.net) |
| 14:59.04 | blee | PMantis: what? |
| 14:59.21 | blee | PMantis: oh im so sorry, i forgot |
| 14:59.36 | blee | Yes find out, there IS a setting in eyebeam where if RTCP isnt being transmitted it will drop the call |
| 14:59.40 | blee | or something along those lines |
| 15:00.09 | PMantis | Chainsaw, for example, almost 2 weeks ago I set: rtptimeout=360, session-expires=1800, and a couple others. |
| 15:00.14 | PMantis | blee, right. |
| 15:00.57 | blee | PMantis: that setting drove me crazy for a bit, let me know how that works out |
| 15:01.26 | PMantis | blee, set debug on... so far I see lots of Polycom and XLite. |
| 15:01.58 | blee | Yep xlite/eyebeam same difference |
| 15:02.59 | PMantis | blee, Oh, I don't use softphones a lot - especially not on Windows. didn't realize that connection. LOL |
| 15:03.52 | blee | Im afraid I cant help you as far as finding the setting |
| 15:03.59 | blee | none of my machines are on windows |
| 15:04.08 | blee | there IS a setting in xlite that says, disconnect call if theres no RTCP |
| 15:04.20 | blee | that 30 second mark sounds oddly familiar |
| 15:04.32 | *** join/#asterisk chopp (~chopp@unaffiliated/chopp) |
| 15:04.38 | Chainsaw | blee: Yes, silence suppression. |
| 15:04.55 | *** join/#asterisk dajhorn (~dajhorn@adsl-75-17-124-41.dsl.rcsntx.sbcglobal.net) |
| 15:06.53 | *** join/#asterisk proute (~AnthonyCB@mail.sysun-technologies.com) |
| 15:06.57 | proute | Hello |
| 15:08.13 | proute | I use asterisk 1.6.2.16 with FFA. So, when the fax sender send me a fax it works fine if the fax sender don't use ecm mode. If the fax sender use ecm mode, fax receive doesn't work. How can I fix this issue? I work with dahdi 2.4 via pstn connection |
| 15:08.18 | proute | thanks for your help |
| 15:10.00 | *** join/#asterisk JonnyD_work (~Jon@cpe-071-075-036-057.carolina.res.rr.com) |
| 15:11.03 | *** join/#asterisk telnettech (~telnettec@216.49.139.56) |
| 15:11.30 | telnettech | Ey Yall |
| 15:12.21 | telnettech | was wondering what people do to help alleviate the effects of SIPVicious and sipscuser DDOS attacks? |
| 15:12.44 | *** join/#asterisk atan3 (~atan@unaffiliated/atan) |
| 15:12.58 | telnettech | I have put the offenders on an ACL on my router but that takes up too much time to site and watch the traffic all day or so |
| 15:13.15 | beek | telnettech: I've personally hired a hit man to track the offenders down and kill them. |
| 15:13.17 | *** join/#asterisk pinoyskull (~pinoyskul@112.198.64.80) |
| 15:14.04 | telnettech | funny Beek |
| 15:14.18 | leifmadsen | telnettech: I use fail2ban |
| 15:14.27 | leifmadsen | documented here: |
| 15:14.29 | leifmadsen | ~newbook |
| 15:14.29 | infobot | Help review the 3rd edition of the Asterisk book published by O'Reilly at http://ofps.oreilly.com/titles/9780596517342/ Released under a Creative Commons license: http://creativecommons.org/licenses/by-nc-nd/3.0/us/ |
| 15:14.37 | beek | telnettech: What leifmadsen said for me as well. |
| 15:14.43 | Chainsaw | telnettech: fail2ban. |
| 15:14.43 | leifmadsen | beek: ohai! |
| 15:14.50 | beek | Hello leifmadsen! |
| 15:14.51 | leifmadsen | works quite well |
| 15:14.56 | Chainsaw | telnettech: Controlling your firewall and the sipvicious crasher. |
| 15:15.48 | *** join/#asterisk frigidzephyr (~rnewton@nat/digium/x-oaroosdfjozqkubl) |
| 15:16.19 | *** join/#asterisk putnopvut (~putnopvut@asterisk/master-of-queues/mmichelson) |
| 15:16.19 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 15:16.37 | telnettech | We have installed a DDOS equipment and are still trying to fine tune it but it appears that the offenders are attacking our custoemrs directly and I dont find it feasible to spend $18,000 per customer to put this equipment on each site |
| 15:17.40 | leifmadsen | fail2ban is free :) |
| 15:17.52 | leifmadsen | works great, just modifies iptables based on rules you configure |
| 15:18.21 | Tozz_ | mm, good point. we've had hacked SIP Accounts, but that could indeed be solved by fail2ban |
| 15:18.25 | Chainsaw | leifmadsen: Yeah, submits to DroneBL as well for me. |
| 15:18.59 | leifmadsen | Tozz_: stronger passwords and stronger usernames helps a lot with that kind of thing :) |
| 15:18.59 | Chainsaw | leifmadsen: For the good of the internet. Now if only Asterisk had DNSBL support... it could benefit from that DroneBL submission everywhere. |
| 15:19.09 | telnettech | leif: that is good on our side but they are not attacking us....they are attacking our customer's....we have implemented ACL on the routers |
| 15:19.17 | leifmadsen | that seems like an outside problem separate of asterisk |
| 15:19.30 | Tozz_ | leifmadsen: it was my test account ;) |
| 15:19.32 | Tozz_ | test/test |
| 15:19.37 | leifmadsen | telnettech: I don't see why you don't install it on your customers as well |
| 15:19.38 | Tozz_ | so basicly my fault ;) |
| 15:19.45 | leifmadsen | Tozz_: 110% your fault |
| 15:19.49 | telnettech | we dont control the customer pbx |
| 15:19.53 | leifmadsen | Tozz_: even test accounts should be secured |
| 15:20.05 | Tozz_ | yeah i noticed ;) |
| 15:20.06 | leifmadsen | telnettech: then there isn't much you can do other than to build your own appliance with fail2ban |
| 15:20.27 | leifmadsen | just build a simple appliance with fail2ban on it, then put that in front instead of the $18k machine |
| 15:20.30 | leifmadsen | problem solved |
| 15:20.33 | leifmadsen | NEXT!!! |
| 15:20.35 | telnettech | i was just looking for general info related to VOIP |
| 15:20.50 | leifmadsen | telnettech: DDOS is a standard problem, separate from VOIP |
| 15:20.56 | leifmadsen | you can only do so much |
| 15:21.09 | leifmadsen | if someone overwhelms your link, there isn't anything you can do about that |
| 15:21.24 | leifmadsen | goes back to bug triage |
| 15:21.27 | *** join/#asterisk Defraz (~Defraz@63-226-95-152.dia.static.qwest.net) |
| 15:21.34 | tzanger | leifmadsen: I would think that a sip registration proxy with fail2ban would be a good start, although ideally you need to block those attempts at the far end of your pipe |
| 15:21.51 | tzanger | fail2ban is awesom |
| 15:21.53 | tzanger | e |
| 15:22.12 | leifmadsen | tzanger: that'd be even better, if you have an ISP who will allow you to call in and get things blocked from their end |
| 15:22.26 | *** join/#asterisk JonnyD_work (~Jon@12.222.63.34) |
| 15:23.14 | tzanger | agreed |
| 15:23.16 | tzanger | how goes? |
| 15:23.39 | thews | I had to make a firewall whitelist for our sip servers, and make our home agents use dyndns |
| 15:23.48 | leifmadsen | tzanger: not bad! just working on bugs right now |
| 15:24.01 | *** join/#asterisk Cain` (~Geek@unaffiliated/cain) |
| 15:24.06 | tzanger | sounds like fun |
| 15:24.24 | leifmadsen | thews: I have a client that makes their home agents login to their email, then they scan the logs for the IP address and allow only those found IP addresses (which expire after 24 hours) |
| 15:24.34 | leifmadsen | so basically each agent checks their email each morning, then logs in |
| 15:24.43 | leifmadsen | I'm sure there are other ways, but I found that a clever way |
| 15:24.54 | *** join/#asterisk Faustov (user@gentoo/user/faustov) |
| 15:25.05 | JonnyD_work | does anyone know how queues handle what agents get called first? |
| 15:25.20 | leifmadsen | JonnyD_work: depends how you configure it |
| 15:25.43 | JonnyD_work | how are the agents handled? the problem i am seeing is my agents with the lowest ext number are getting calls first, not based on when they logged in |
| 15:25.55 | JonnyD_work | i am using rrmemory |
| 15:25.58 | *** join/#asterisk werdan7 (~w7@freenode/staff/wikimedia.werdan7) |
| 15:26.03 | thews | that would work, but then I would have to script into our firewall unless I just did it on the client firewall of the sip server |
| 15:26.20 | thews | currently I just add dns whitelist to the datacenter firewall, don't run any firewalls on individual servers |
| 15:27.09 | leifmadsen | JonnyD_work: rrmemory is based on when they were last called |
| 15:27.10 | thews | I could also grab their ips from the custom ajax call controller we made |
| 15:27.13 | *** join/#asterisk Tech_Travis (~Travis_El@cpe-76-168-191-127.socal.res.rr.com) |
| 15:27.20 | thews | never thought about that |
| 15:27.56 | JonnyD_work | leifmadsen: right but first thing in the morning how does it determin who gets called first? |
| 15:29.17 | leifmadsen | not sure |
| 15:29.21 | leifmadsen | never bothered to look at that |
| 15:29.35 | leifmadsen | you could control how it happens with queuerules.conf |
| 15:29.48 | leifmadsen | beyond that, I've never worried about who got called first |
| 15:30.05 | leifmadsen | I'd imagine it'd be the order in which they logged in, or whatever the order shows in 'queue show' |
| 15:30.14 | leifmadsen | but that's just a guess, you'd have to look at the code |
| 15:33.04 | *** join/#asterisk atan (~atan@unaffiliated/atan) |
| 15:34.05 | JonnyD_work | thats what i though it would do but it seems to be sending them based on the lowest ext number |
| 15:34.14 | JonnyD_work | i will have to look at queuerules |
| 15:34.28 | jamko | Morning. On the attacks subject. How is it that some attackers are able to get the sip user ids for my devices? If an attackers is not on my network segment, they shouldn't be able to sniff my traffic, so is asterisk giving a response to indicate a correct hit on a user id? |
| 15:35.29 | *** join/#asterisk rgsteele (~rgsteele@apo.aweber.com) |
| 15:35.34 | JonnyD_work | when was queuerules introduced? |
| 15:35.38 | JonnyD_work | i am on 1.4.31 |
| 15:35.55 | thews | on my roundrobin setup I have priority on users, and then it calls the one who hasn't been called in the longest time (first logged in gets called first) |
| 15:36.35 | JonnyD_work | can you add priority to dynamic agents? |
| 15:36.40 | thews | if you want to punish people who take too long on calls use a fewestcalls setup |
| 15:37.04 | thews | how are they dynamic? |
| 15:37.16 | *** join/#asterisk lftsy (~lftsy@pul-lav-fw-so-01-x1.vtxnet.net) |
| 15:38.08 | *** join/#asterisk w0ls0n (~w0ls0n@221-114-181-66.dsl.sacoriver.net) |
| 15:38.37 | w0ls0n | can someone suggest an inexpensive ip phone that will work with *? Used is totally fine and 1 line is preferred. |
| 15:38.57 | thews | strategy=rrmemory |
| 15:38.57 | thews | member => Local/#401@neoagent/n,3 |
| 15:39.02 | thews | that's how mine are setup |
| 15:39.09 | thews | last number is the priority |
| 15:39.42 | JonnyD_work | i have my agents login and out |
| 15:39.49 | thews | mine too |
| 15:39.51 | *** join/#asterisk The_Boy_Wonder (~vossel@asterisk/batman-developer/dvossel) |
| 15:40.03 | thews | they log in as available to take calls from queue |
| 15:40.08 | JonnyD_work | but you add the agents in queues.conf? |
| 15:40.23 | thews | software adds it |
| 15:40.55 | thews | everything is software/db driven |
| 15:41.00 | thews | we have a complicated environment |
| 15:44.43 | *** join/#asterisk asterisk-learner (~chatzilla@77.42.241.114) |
| 15:47.20 | *** join/#asterisk rushowr (~rushowr@99-28-31-100.lightspeed.stlsmo.sbcglobal.net) |
| 15:47.26 | rushowr | greetings everyone |
| 15:48.06 | rushowr | thought I'd come by and see if anyone can help with a minor question. Anyone here use FUNC_ODBC and ARRAY() to retrieve multiple columns from a query? |
| 15:50.03 | *** join/#asterisk coppice (~chatzilla@m121-202-108-195.smartone-vodafone.com) |
| 15:51.44 | asterisk-learner | ping |
| 15:52.27 | leifmadsen | pong |
| 15:52.39 | leifmadsen | rushowr: yes, works great :) |
| 15:52.54 | rushowr | hey! Leif! |
| 15:53.10 | leifmadsen | Set(ARRAY(col1,col2)=${ODBC_GET(${values})}) |
| 15:53.21 | rushowr | hey real quick mate, I've got to apologize for my stupidity the other day with the missing => :P |
| 15:53.42 | rushowr | now, your code snip above, I used and get an error...one sec |
| 15:53.43 | *** join/#asterisk ClintGoudie-Nice (~clint@smtp.callware.com) |
| 15:53.52 | rushowr | I'll pull the code, log, query |
| 15:54.00 | *** join/#asterisk russellb (~russell@asterisk/digium-open-source-team-lead/russellb) |
| 15:54.00 | *** mode/#asterisk [+o russellb] by ChanServ |
| 15:54.24 | ClintGoudie-Nice | Hola all. |
| 15:54.51 | ClintGoudie-Nice | Is there a link/guide anywhere for configuring SRTP and TLS in asterisk 1.8? |
| 15:55.22 | rushowr | @leif sorry mate, it'll take a sec to dig it back up |
| 15:55.24 | leifmadsen | ~newbook |
| 15:55.24 | infobot | Help review the 3rd edition of the Asterisk book published by O'Reilly at http://ofps.oreilly.com/titles/9780596517342/ Released under a Creative Commons license: http://creativecommons.org/licenses/by-nc-nd/3.0/us/ |
| 15:55.27 | leifmadsen | ClintGoudie-Nice: see above |
| 15:55.34 | *** join/#asterisk Faustov (user@gentoo/user/faustov) |
| 15:56.18 | *** join/#asterisk FiReSTaRT (~dlyh@unaffiliated/firestart) |
| 15:57.25 | rushowr | @leif :: query line in func_odbc.conf : SELECT cnamid, age, cname from cnam WHERE btn = '${ARG1}' |
| 15:57.28 | ClintGoudie-Nice | thanks for the link. I'll start reading now |
| 15:58.11 | leifmadsen | exten => start,1,Set(ARRAY(cnamid,age,cname)=${ODBC_GET(first_arg)}) |
| 15:58.31 | rushowr | right |
| 15:58.46 | rushowr | weird |
| 15:59.16 | rushowr | lol, nevermind, I started to say the ODBC_GET was wroing, but then I realized you don't know what I named it and were substituting LOL |
| 15:59.21 | rushowr | here's the related log entry |
| 16:00.47 | rushowr | [Jan 18 22:33:37] VERBOSE[15728] pbx.c: -- Executing [+17322761300@getcnam:8] Set("SIP/sip1-inbound-00000007", "ARRAY(cnamid,age,cname)=60069,-215,Cell Phone MO") in new stack |
| 16:00.49 | rushowr | [Jan 18 22:33:37] WARNING[15728] pbx.c: MSet: ignoring entry '-215' with no '=' (in +17322761300@getcnam:8 |
| 16:01.30 | *** join/#asterisk bmoraca_work (~bmoraca@66-242-174-254.ceres.bvn.net) |
| 16:01.34 | leifmadsen | why are you using MSet? |
| 16:01.40 | rushowr | I'm not |
| 16:01.41 | leifmadsen | guesses AEL |
| 16:01.44 | rushowr | nope |
| 16:01.48 | rushowr | nice try though |
| 16:01.52 | leifmadsen | dialplan? |
| 16:01.54 | rushowr | I'm using: |
| 16:02.00 | *** join/#asterisk drudge` (drudge@unaffiliated/drudge/x-837452) |
| 16:02.01 | leifmadsen | dialplan says you're using MSet |
| 16:02.38 | rushowr | Dunno mate, I used the command Set(ARRAY(cnamid,age,cnameid)=ODBC_CNAMGET(${BTN})) |
| 16:03.03 | rushowr | and it's not being called via a macro or anything, straight context within standard ast dialplan |
| 16:03.26 | rushowr | odd eh? |
| 16:03.32 | leifmadsen | rushowr: you're not calling ODBC right! |
| 16:03.38 | leifmadsen | if that is actually the command you're using |
| 16:03.41 | leifmadsen | look at my example again |
| 16:03.44 | leifmadsen | tell me what you have missing |
| 16:04.00 | *** join/#asterisk _Corey_ (~chatzilla@64.121.4.75) |
| 16:04.01 | rushowr | hrm.....ok, let me see |
| 16:04.02 | leifmadsen | otherwise, please show the actual command and not just a typed version of it |
| 16:04.09 | rushowr | yeah, let me grab the original code |
| 16:04.14 | rushowr | that was just me typing |
| 16:04.15 | leifmadsen | always show the original code |
| 16:04.20 | rushowr | (all my other odbc calls work ;-) ) |
| 16:04.27 | leifmadsen | what version? |
| 16:05.04 | *** join/#asterisk AnhVT (~tuananht@58.187.60.60) |
| 16:05.06 | *** join/#asterisk b0gatyr (~b0gatyr@unaffiliated/b0gatyr) |
| 16:05.12 | *** join/#asterisk BugKhaM (~BugKhaM@125.25.10.129.adsl.dynamic.totbb.net) |
| 16:05.14 | rushowr | 1.6.2.14 |
| 16:05.31 | rushowr | Set(cnamid=${ODBC_CNAMCHECK_CNAMID(${BTN})}) |
| 16:05.33 | rushowr | oops |
| 16:05.57 | rushowr | I had to modify the code to make it work in the system, let me dig through the revisions |
| 16:06.25 | b0gatyr | Hi everyone, im trying to set verbosity on my asterisk box so i can see when someone places a call, i have tried setting verbosity to 10 but i dont see any messages on the console am i missing smth? |
| 16:06.50 | rushowr | lol, ok the command I typed was right except for using the VALUE of the ODBC call ${ODBC_CNAMGET(${BTN})} |
| 16:07.07 | *** join/#asterisk timahvo1 (~rogue@41.223.57.74) |
| 16:07.24 | leifmadsen | rushowr: just show me the exact command (the whole line) |
| 16:07.38 | *** join/#asterisk JonnyD_work_ (~Jon@cpe-071-075-036-057.carolina.res.rr.com) |
| 16:08.30 | leifmadsen | rushowr: I'd rather you just msg in here and not directly |
| 16:08.31 | leifmadsen | looks fine |
| 16:08.42 | rushowr | ah sorry mate |
| 16:08.52 | rushowr | yeah, so you see why I'm confused ;-) |
| 16:08.54 | leifmadsen | other people may have more time to help |
| 16:09.17 | rushowr | thanks mate for your time |
| 16:09.19 | rushowr | :D |
| 16:09.25 | rushowr | slainte / proost / cheers |
| 16:10.03 | leifmadsen | skoal |
| 16:10.42 | rushowr | ah, nice! I'd forgotten that one (although I'm guessing it's your native tongue, as sláinte is mine :D |
| 16:10.56 | leifmadsen | nah, english is my native tongue |
| 16:11.02 | leifmadsen | I don't speak danish unfortunately |
| 16:11.07 | rushowr | neither do I |
| 16:11.10 | rushowr | :P |
| 16:11.14 | rushowr | just Irish and English |
| 16:12.02 | AdvoWork | if im trying to dial a number, i have a dialtone, but it just hangs and doesnt call, what logs would that be in? struggling to find anything.. |
| 16:12.14 | rushowr | you can try /var/log/asterisk/full |
| 16:12.26 | rushowr | however, you might end up needing to do a traffic capture |
| 16:12.57 | rushowr | since it sounds like it may not have executed anything on Asterisk |
| 16:14.57 | DarkRift | well what he can do is core set debug 7 and core set verbose 7 in the CLI and see what happens when he dials that number |
| 16:15.19 | DarkRift | He'll have debug information in the CLI |
| 16:15.57 | rushowr | note: nothing past 3 results in extra infro |
| 16:16.30 | rushowr | also, if the call is not actually being executed on the Asterisk server, due to various possible issues, he'll need to see the traffic cap |
| 16:16.52 | DarkRift | Yeah but it may be a function/app failing |
| 16:16.58 | rushowr | true |
| 16:17.06 | rushowr | hence why I told him to check the full log |
| 16:17.29 | rushowr | and then offered a possible next step if he sees nothing there |
| 16:17.38 | DarkRift | I'm not going to fight, I was just saying he could see it live in the CLI |
| 16:17.49 | rushowr | cheers |
| 16:18.55 | leifmadsen | rushowr: actually the number is '5' |
| 16:19.08 | leifmadsen | most things show up in 3 and 4 though |
| 16:19.17 | rushowr | odd, I've never seen any improvement in using anything above 3 |
| 16:19.33 | rushowr | but, I believe ya mate |
| 16:19.43 | rushowr | is that in verbose or debug or both? |
| 16:20.59 | w0ls0n | hehe |
| 16:21.18 | *** part/#asterisk w0ls0n (~w0ls0n@221-114-181-66.dsl.sacoriver.net) |
| 16:21.22 | rushowr | I'll read the source |
| 16:22.29 | rushowr | well mates, I'm off, I'll stop by later |
| 16:22.32 | rushowr | Is féidir do lá dul chomh réidh agus is an craiceann ar bhean álainn |
| 16:22.34 | *** part/#asterisk rushowr (~rushowr@99-28-31-100.lightspeed.stlsmo.sbcglobal.net) |
| 16:23.10 | devmod | Is there a way to negotiate or even just change the bitrate on incoming calls ? (besides sip.conf's maxcallbitrate) |
| 16:23.12 | b0gatyr | hi everyone, how can I get asterisk CLI console to show messages of the dialed numbers when someone places a call? |
| 16:23.17 | b0gatyr | is that set with verbosity? |
| 16:23.40 | leifmadsen | devmod: change the codecs allowed? |
| 16:23.54 | leifmadsen | b0gatyr: use Verbose() |
| 16:24.01 | *** join/#asterisk tuxxie (~Ryan@rrcs-70-63-90-226.midsouth.biz.rr.com) |
| 16:24.11 | leifmadsen | set to level 1 or 2 if you don't want to see anything but your Verbose() messages |
| 16:24.16 | leifmadsen | then 'core set verbose 2' |
| 16:24.54 | devmod | leifmadsen, I guess im talking about b= on the invite and 200ok's SDP |
| 16:25.14 | leifmadsen | not that I'm aware of. Pretty sure all the bit rates are static in Asterisk. |
| 16:25.27 | devmod | leifmadsen, ohh ok got it |
| 16:25.29 | leifmadsen | to modify headers you need to use something like OpenSIPS to modify the headers before sending them out |
| 16:25.44 | leifmadsen | although it probably wouldn't help you much since the bitrate is static for the codecs |
| 16:25.54 | devmod | leifmadsen, not video codecs |
| 16:25.59 | leifmadsen | you can only change the payload size with something like: allow=speex:30 |
| 16:26.15 | devmod | leifmadsen, I should have said I am dealing with video codecs |
| 16:26.20 | leifmadsen | I still don't think so |
| 16:26.30 | leifmadsen | video support is limited in asterisk |
| 16:26.34 | b0gatyr | leifmadsen: im setting verbosity with : core set verbose 10 but i still dont see anything =\ |
| 16:26.40 | devmod | leifmadsen, yeah I understand |
| 16:26.58 | leifmadsen | b0gatyr: I've seen people talk about that issue before, but have never been able to get enough information to know why that is the case. It's very rare. Sorry, not sure. |
| 16:27.43 | tuxxie | i am having issues on call transfers. once the call transfer is made the call auto is only one way. I get "Spawn extension (main, 4050, 1) exited non-zero on 'SIP/0004F2A601A2-00018549<ZOMBIE>" in my error logs. What does ZOMBIE mean? |
| 16:28.42 | b0gatyr | leifmadsen: hmm, weird i just typed in the command again and worked.. |
| 16:28.50 | leifmadsen | sounds like a typo the first time |
| 16:28.58 | tuxxie | does that mean tha the person transfering the call has exited the call? |
| 16:32.02 | *** join/#asterisk gerhard7 (~gerhard7@212-123-146-122.ip.telfort.nl) |
| 16:32.49 | leifmadsen | tuxxie: just means a channel was killed, which was probably the channel that originally tranferred the call |
| 16:33.42 | *** join/#asterisk BugKhaM (~BugKhaM@199.255.209.74) |
| 16:35.25 | AdvoWork | DarkRift, do you know how id do a traffic capture? |
| 16:35.59 | leifmadsen | use tshark |
| 16:36.03 | DarkRift | The traffic capture would be with ngrep or wireshark, as for ngrep, I've never used it |
| 16:36.03 | tuxxie | leifmadsen: so the transfer-er hungup prior to the transfer being completed? or zombie message is to be expected? |
| 16:36.30 | leifmadsen | tuxxie: hard to say based on the lack of information, but it just means there was a channel that was a zombie (dead channel) that was killed |
| 16:38.42 | tuxxie | leifmadsen: my next line in my messages log is "Executing [h@main:1] Hangup("SIP/0004F2A601A2-00018549<ZOMBIE>", "") " does that help show the reason? |
| 16:41.02 | leifmadsen | no it just means the 'h' extension was executed, which happens when a channel is hung up |
| 16:42.55 | tuxxie | leifmadsen: I am seeing a large amount of calls with this status. show this be a reason for alarm? |
| 16:43.28 | leifmadsen | only if it's affecting something |
| 16:44.00 | *** join/#asterisk oej (~olle@109.58.251.215) |
| 16:45.30 | AdvoWork | my full log shows a few: [Jan 19 16:44:33] NOTICE[2065] chan_sip.c: -- Registration for '12345@sip.whatever.co.uk' timed out, trying again (Attempt #18215) any suggestions? thats a trunk, everything seems to work though... |
| 16:49.26 | *** join/#asterisk eject_ck (~eject_ck@haitek.cosmonova.net.ua) |
| 16:49.48 | eject_ck | can someone suggest sip client for android 2.2 |
| 16:50.50 | *** join/#asterisk Fruchthoernschen (~Fruchthoe@trir-5d80024a.pool.mediaWays.net) |
| 16:51.57 | *** join/#asterisk jsjc (~jsjc@13.Red-80-37-165.staticIP.rima-tde.net) |
| 16:52.06 | jsjc | I am having some trouble to faxdetect over SIP. |
| 16:52.21 | jsjc | is there any codeec I must have while detect? |
| 16:53.12 | leifmadsen | ulaw |
| 16:53.31 | leifmadsen | can't use anything other than ulaw (or alaw) for even somewhat reliable faxing |
| 16:58.47 | raden | can never get fax to work :( |
| 17:00.53 | *** join/#asterisk andylockran (~andylockr@genesis.zrmt.com) |
| 17:01.04 | andylockran | guys, what platform/dist is AsteriskNow built on .. and what's it's 'auto-update' mechanism? |
| 17:01.14 | drmessano | CentOS and Yum |
| 17:01.24 | andylockran | drmessano: thanks. |
| 17:02.15 | drmessano | It's a pretty unmolested build of CentOS as well, not hacked into some completely new distro with lots of defaults changed or any of that nonsense.. |
| 17:02.30 | andylockran | drmessano: I've been using asterisk for about 5 years on gentoo, but their version has slipped quite far behind. I'm not doing anything particularly unique with my setup, and hopefully having a community of users with the same setup (eg, AsteriskNOW) would be a more sensible way of supporting the system. |
| 17:02.55 | andylockran | Is FreePBX kept updated on AsteriskNow too ? |
| 17:03.02 | drmessano | Yes |
| 17:03.36 | drmessano | FreePBX RPM's are updated as needed, and newer FreePBX also has a module admin that works just as well |
| 17:04.48 | andylockran | awesome. Thanks for your feedback. |
| 17:07.07 | drmessano | There's lots of online support as well.. Depending on the hour of day, you can get support in here, but there's also #asterisknow and #freepbx. #freepbx is pretty active, and a lot of us are running CentOS as well, so we generally will support more than *just* the FreePBX bits of an AsteriskNOW box if it's within reason |
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| 17:11.02 | thews | on polycom's website they say that the newest release for the 501s is 3.1.4 but there is also the vvx 4.1.4 bootrom |
| 17:11.18 | thews | http://downloads.polycom.com/voice/voip/sip_sw_releases_matrix.html |
| 17:12.04 | thews | sorry 3.1.6, is the bootrom vvx 4.1.4 and the sip software version 3.1.6 ? |
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| 17:16.08 | thews | nevermind, I understand |
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| 17:24.06 | PMantis | blee, I talked to my customer, and they DID have the RTCP option checked off in X-Lite, set to 30 seconds. I think that was the solution. Thanks! |
| 17:33.22 | b0gatyr | folks im having this really weird problem i'm calling an extension that is at a remote office sometimes works and sometimes the call does not go through, network connection seems fine what can it be? any help is appreciated |
| 17:35.41 | *** join/#asterisk peep (637c543e@gateway/web/freenode/ip.99.124.84.62) |
| 17:36.30 | peep | can anyone tell me how/if its possible to silence the notifications in the console when you make a request to AJAM? |
| 17:37.17 | kaldemar | b0gatyr: by extension do you mean a VoIP phone? find out what happens in asterisk's CLI when a call doesn't go through. probably network issues. |
| 17:40.37 | b0gatyr | kaldemar: yup VoIP phone, i'm able to ping the remote cisco callmanager at the remote office and asterisk CLI does not even show up anything when I try placing a call and i have set verbose to at least 10, when it goes through i do see it on the CLI.. =\ |
| 17:41.02 | peep | specifically the lines like 'HTTP Manager add header...' |
| 17:44.14 | kaldemar | b0gatyr: so you are dialing phone -> asterisk -> cisco -> phone ? |
| 17:45.48 | plundra | Uhm, can is it possible to some how enable sip debug for all but <given ip-range>? |
| 17:46.10 | plundra | Or all but <some peers> |
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| 17:47.22 | b0gatyr | kaldemar: yup |
| 17:47.39 | kaldemar | b0gatyr: if you don't see _anything_ on the asterisk CLI with verbosity when you make the call, then there is a problem between the first phone and asterisk. |
| 17:49.20 | b0gatyr | kaldemar: right now everything is working fine again let's see how long it lasts |
| 17:49.45 | kaldemar | plundra: only by setting debug on the wanted peers one by one. probably not what you're looking for. |
| 17:51.09 | plundra | kaldemar: Unfortunatly no, I need to clear out all the noise of known/working traffic. |
| 17:52.32 | kaldemar | plundra: you could make a script that enables and disables the debug for a subset of peers. for example using -rx parameter to give the CLI commands. |
| 17:54.10 | plundra | kaldemar: I'm a bit earlier then that actually, have no peer set up yet etc., wanted to see what it gave me. |
| 17:54.24 | plundra | Might tcpdump and filter that way, look at the traffic in wireshark. |
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| 18:21.12 | boch | hi all |
| 18:22.26 | boch | one question: how is the proper way to delete all database entries from a tree with value=X , from dialplan, any idea? |
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| 18:44.41 | NephFL | hello, can anyone here help me out with the basics of cisco? Specifically is the CCA for the UC540 a web app or seperate installed app? |
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| 18:54.06 | silvestre_id | Hi to all, i was trying to put 'linkedid' by using cdr_odbc with no sucess. I created the field after 'uniqueid'. The 'linkedid' using cel_odbc already works. Someone know configure cdr_odbc or table of cdr correctly? asterisk-1.8.2.1 |
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| 19:52.54 | mateu | boch: DBdeltree() ? |
| 19:53.48 | boch | mateu, that will delete the whole tree, not only the entries with value=X |
| 19:54.08 | mateu | ah, that's how your question is to be interpreted. I wasn't sure. |
| 19:57.04 | leifmadsen | boch: use a While() loop |
| 19:57.19 | leifmadsen | check if it has the value you want, delete, iterate |
| 19:57.26 | leifmadsen | something like that |
| 19:57.37 | leifmadsen | there is no DBWhereValueEquals() |
| 19:58.17 | boch | leifmadsen, can i do that on dialplan ? |
| 20:06.16 | leifmadsen | boch: if you know the DB structure, sure |
| 20:06.26 | leifmadsen | there is nothing that will let you return the existing DB structure that I can see |
| 20:06.44 | leifmadsen | you can use DB_EXISTS() to check for existing values in trees |
| 20:07.03 | boch | i wonder how to cycle all entries in a tree |
| 20:07.11 | leifmadsen | use While() |
| 20:07.17 | leifmadsen | you need to know the tree structure already though |
| 20:07.23 | leifmadsen | (which I just said) |
| 20:07.36 | boch | what you mean? |
| 20:07.58 | leifmadsen | I mean you can't just return the tree structure. You need to program it in statically. |
| 20:08.10 | boch | the tree im using is a simple AVP |
| 20:08.17 | leifmadsen | I have no idea what AVP means |
| 20:08.26 | boch | attribute value pair |
| 20:08.46 | leifmadsen | you're not listening |
| 20:09.00 | boch | sorry i cant understand what you mean |
| 20:09.02 | leifmadsen | you need to know the structure prior to looping, because you can't just return the structure from the dialplan |
| 20:09.26 | b0gatyr | mmm i love me some steak! |
| 20:09.43 | leifmadsen | the only way I can think of doing that is via something like writing the output of: asterisk -rx "database show" to a file, then looping through that to know the structure of the database and check everything |
| 20:09.59 | leifmadsen | but at that point you're better off just using either a relational database or an external script |
| 20:10.13 | boch | for example i need to know all entries in the tree |
| 20:10.15 | boch | got it |
| 20:10.20 | leifmadsen | exactly |
| 20:10.21 | leifmadsen | that's the problem |
| 20:10.22 | boch | thank you |
| 20:13.10 | mateu | yeah, i'm surprised there isn't something from the dialplan like 'database show family' so one can get all keys in a family to loop over. |
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| 20:14.46 | leifmadsen | mateu: like 'database showkey' ? |
| 20:15.25 | leifmadsen | actually |
| 20:15.26 | leifmadsen | nevermind |
| 20:15.29 | leifmadsen | showkey is a single |
| 20:15.35 | leifmadsen | database shows everything |
| 20:16.22 | mateu | i mean something from the dial plan that mimics 'database show <family>' |
| 20:16.41 | leifmadsen | guess no one has found that important enough to program :) |
| 20:16.52 | leifmadsen | at that point you should probably just use a relational database... |
| 20:17.10 | mateu | i dunno |
| 20:17.16 | leifmadsen | me either |
| 20:17.16 | mateu | seems pretty basic to me. |
| 20:17.19 | leifmadsen | sure does |
| 20:17.24 | leifmadsen | no one has programmed it though |
| 20:17.28 | leifmadsen | shrugs |
| 20:17.44 | mateu | ok, well at least we know how it currently stands. thanks leifmadsen |
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| 20:28.52 | Corydon76-home | leifmadsen: something like HASHKEYS() ? |
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| 20:30.11 | leifmadsen | Corydon76-home: ummm, I was thinking more like DUNDI_QUERY() and DUNDI_RESULT() |
| 20:30.31 | leifmadsen | although HASHKEYS() might work |
| 20:30.57 | leifmadsen | actually ya, looking at it, similar to HASHKEYS() |
| 20:31.01 | leifmadsen | DBKEYS() I guess? |
| 20:31.45 | Corydon76-home | So with no argument, retrieves families, with an argument, retrieves keys of that family? |
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| 20:34.02 | leifmadsen | ya |
| 20:34.15 | leifmadsen | how would you iterate through layers of them? |
| 20:34.30 | leifmadsen | i.e. family/key/key/key ? |
| 20:34.43 | Corydon76-home | Essentially, yes |
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| 21:04.04 | *** join/#asterisk fullstop (~fullstop@static-173-210-91-4.ngn.onecommunications.net) |
| 21:04.37 | fullstop | Is anyone here familiar with UniMRCP? |
| 21:07.19 | pabelanger | ~ask |
| 21:07.20 | infobot | Questions in the channel should be specific, informative, complete, concise, and on-topic. Don't ask if you can ask a question first. Don't ask if a person is there; just ask what you intended to ask them. Better questions more frequently yield better answers. We are all here voluntarily or against our will. |
| 21:07.42 | fullstop | Okay. |
| 21:08.40 | fullstop | I'm attempting to configure asterisk to do ASR with the UniMRCP asterisk module talking to a vestec ASR server. The documentation seems to be.. thin, and I'm not exactly sure how to get started. |
| 21:14.49 | drmessano | https://groups.google.com/forum/?pli=1#!forum/unimrcp <-- They have a google group |
| 21:14.57 | drmessano | Probably a better shot than asking here |
| 21:35.33 | ClintGoudie-Nice | is there a way to activate the MWI light from the asterisk console? |
| 21:37.26 | tzanger | ClintGoudie-Nice: there is but it's not noice |
| 21:37.27 | tzanger | er nice |
| 21:37.59 | ClintGoudie-Nice | Just to make sure it's working |
| 21:38.42 | tzanger | ClintGoudie-Nice: you create a .txt file in /var/spool/asterisk/voicemail/[DOMAIN]/[MBOX]/INBOX/ |
| 21:38.50 | tzanger | it should light up fairly quickly |
| 21:39.27 | tzanger | like msg0000.txt |
| 21:39.29 | ClintGoudie-Nice | I haven't set up voicemail :P |
| 21:39.38 | tzanger | ClintGoudie-Nice: then no, there is no way to light up MWI |
| 21:40.02 | *** join/#asterisk anthm (~anthm@freeswitch/developer/anthm) |
| 21:40.23 | ClintGoudie-Nice | I've got a device that's sending a SIP notify into asterisk to light the MWI on a phone, but asterisk is responding with 489 Bad event. |
| 21:40.55 | tzanger | probably because VM is not set up... I'm not an expert on SIP |
| 21:41.02 | ClintGoudie-Nice | I'm just trying to figure out if asterisk is outright blocking it, or if it's not getting to the phone and asterisk is informing the sip sender it's blocked. |
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| 21:46.59 | jsjc | is there anyway to change some .gsm recording to ulaw or alaw format? |
| 21:48.27 | fullstop | ClintGoudie-Nice: I do this... but it's not exactly pretty. |
| 21:48.48 | fullstop | I use netcat and craft a sip message to send to the set. |
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| 21:52.25 | fullstop | ClintGoudie-Nice: I use this to light up the MWI light on a set attached to a second server |
| 21:53.25 | citywok | tzanger: the filesystem polling option has to be enabled for that to work |
| 21:54.08 | tzanger | citywok: not really, I am pretty sure Asterisk will fallback to manually polling as opposed to inotify/dnotify if it's not present |
| 21:55.24 | citywok | on my system without the file system polling option enabled it doesn't check for deleted voicemails to turn off the mwi light. i'd assume the reverse is true for creating files. |
| 21:55.31 | fullstop | I did not have any luck with creating a file in the directory w/1.6.2.13 |
| 21:55.54 | citywok | i use an XML app to access/delete voicemails, so i have to use filesys polling to turn off the light if a file is deleted from the browser |
| 22:01.54 | citywok | fullstop: what do you mean you didnt have luck creating the file? |
| 22:02.23 | fullstop | citywok: creating a .txt file in /var/spool/asterisk/voicemail/[DOMAIN]/[MBOX]/INBOX/ did nothing for me. |
| 22:02.39 | ClintGoudie-Nice | fullstop: I assume if I crafted the packet perfect for the handset it would light, but I'm trying to figure out where the breakdown is happening. |
| 22:03.13 | citywok | add these lines to your voicemail.conf pollmailboxes=yes; pollfreq=60 |
| 22:03.18 | citywok | reload voicemal |
| 22:04.08 | ClintGoudie-Nice | In a way, I'm crafting the sip message on the device, but I'm getting back a 489 bad event |
| 22:04.08 | fullstop | citywok: that ship has sailed. I'm crafting sip messages in extern notify in voicemail |
| 22:04.31 | fullstop | ClintGoudie-Nice: Make sure that your content-length is correct |
| 22:04.53 | citywok | asterisk by default won't notice the new txt file, so it wont know to create the notification |
| 22:06.42 | ClintGoudie-Nice | yeah, the length should be fine |
| 22:06.57 | fullstop | ClintGoudie-Nice: what phone? |
| 22:07.04 | fullstop | I did my testing w/polycom |
| 22:07.29 | ClintGoudie-Nice | I've put a wireshark dump of the two packets here: http://pastebin.com/hpsBYQQ7 |
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| 22:08.49 | fullstop | I don't think that is the proper format for the messages... |
| 22:09.06 | ClintGoudie-Nice | fullstop: what would you expect to be different? |
| 22:09.19 | fullstop | well, maybe it is.. but I have Messages-Waiting:, :Message-Account: and Voice-Message: in my payload. |
| 22:10.13 | ClintGoudie-Nice | in your Event list? |
| 22:10.51 | fullstop | ClintGoudie-Nice: Do you send exactly this in your payload? "Messages-Waiting:yes." ? |
| 22:12.01 | ClintGoudie-Nice | Messages-Waiting:yes. |
| 22:12.06 | ClintGoudie-Nice | just like so |
| 22:12.15 | fullstop | ClintGoudie-Nice: http://pastebin.com/0vTsDZAM This is what mine would look like if I had 1 new message and 2 old messages. |
| 22:13.21 | ClintGoudie-Nice | hmm |
| 22:13.32 | fullstop | and I believe that the content-length includes a newline at the end |
| 22:17.38 | fullstop | ClintGoudie-Nice: Since you already have wireshark handy, leave a voicemail in that mailbox and watch the wire as asterisk sends the sip message to the set. |
| 22:19.58 | _Corey_ | I just checked a script I wrote to send those out a long time ago |
| 22:20.13 | _Corey_ | the length is the SDP part |
| 22:20.44 | ClintGoudie-Nice | ah, I'm the one sending the notify to asterisk. I'm expecting asterisk to take the notify and route it to the set |
| 22:20.56 | _Corey_ | oh |
| 22:21.02 | _Corey_ | hmm |
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| 22:22.25 | _Corey_ | I missed your earlier messages--and I could be wrong--but I don't believe that will work |
| 22:22.39 | fullstop | ClintGoudie-Nice: I don't think that asterisk will do that for you. |
| 22:22.42 | _Corey_ | Asterisk isn't a SIP proxy, so it won't just relay everything you send it |
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| 22:23.06 | fullstop | ClintGoudie-Nice: I parse the output of sip show peers and snag the ip address and port to send to. |
| 22:23.10 | peep | Could anyone tell me how to suppress the console notifications from AJAM operations being executed? |
| 22:23.19 | fullstop | like I said, ugly. |
| 22:25.44 | fullstop | ClintGoudie-Nice: Asterisk 1.8 might be a little more helpful -- there is at least a "sip notify clear-mwi" command |
| 22:27.57 | fullstop | maybe not.. it looks pretty static. |
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| 22:41.49 | Kobaz | anyone know how the who-hung-up patchset is doing? |
| 22:44.37 | russellb | Kobaz: i don't even know what patchset that is, heh |
| 22:44.57 | Kobaz | i think oej is working on it |
| 22:45.58 | russellb | ah. |
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| 22:49.31 | peep | Could anyone tell me how to suppress the console notifications from AJAM operations being executed? |
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| 22:51.32 | russellb | peep: displayconnects=no in manager.conf I think |
| 22:52.08 | jsjc | I have an IVR recorded in .gsm and faxdetect is having a few issues (I think .gsm could be the issue) is there a way to convert .gsm to .ulaw or .alaw? How can I make sure that alaw or ulaw are being forced in my sip incoming calls? |
| 22:53.01 | peep | russellb: Hrm I tried that one already, that seemed to work for regular AMI connections but I dont think it supresses AJAM |
| 22:55.23 | Quadrant | jsjc: use sox |
| 22:55.45 | JonnyD_work | does anyone know how rrmemory decides who gets the first call? i though it was based on who logged in first but that does not seem to be the case |
| 22:55.57 | Quadrant | and secondly, disallow=all allow=ulaw |
| 22:56.25 | Quadrant | you dont really want alaw unless you have non-north american trunks |
| 22:56.54 | russellb | peep: what is the text of the message? |
| 22:57.56 | peep | "> HTTP Manager add header action: login" |
| 22:58.19 | peep | russellb: thats just an example but you get 4-5 per action you execute :( |
| 22:58.28 | russellb | ah, looks like there is no way to turn it off other than setting verbose < 4 |
| 22:58.33 | russellb | that looks annoying |
| 22:58.57 | peep | russelb: yeah especially when you have lots and lots of AJAM calls happening at all times |
| 22:59.00 | russellb | what version are you using? |
| 22:59.16 | peep | 1.6.2.13 |
| 22:59.47 | ClintGoudie-Nice | Why wouldn't asterisk forward it on for me though? It has the knowledge and registration of the handsets. |
| 22:59.59 | peep | perhaps its time to flex this developer license and start pulling C apart :( |
| 23:00.07 | russellb | peep: k, I'll change it. in the meantime, search for "HTTP manager add header" in main/manager.c, and you can remove that line of code |
| 23:00.23 | peep | russellb: oh wow, you're the man |
| 23:00.25 | ClintGoudie-Nice | This sip device ultimately wouldn't even know if a given extension has a handset or what it's IP might be |
| 23:00.49 | russellb | peep: starts with ... ast_verb(4, "HTTP manager add header ...... |
| 23:00.55 | russellb | something like that |
| 23:01.19 | peep | russellb: you sir just made my day. |
| 23:01.54 | russellb | :-) |
| 23:02.07 | drmessano | He makes everyones day |
| 23:02.15 | drmessano | russellb = the man |
| 23:03.50 | drmessano | I tried to tell him that the other day, but he didn't hear me from behind the bushes in front of his place again. In too much of a hurry, I guess, off to make those Asterisk donuts we all enjoy. |
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| 23:42.56 | carrar | drmessano, PICS!!! of Asterisk donuts!! |
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| 23:58.52 | sinergycomm | hey anoyone here? |
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