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| 02:17.43 | ChannelZ | Wow. Lots of pingy-pongy but no talky-talky |
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| 02:20.02 | SeRi | I just got a Gigaset C610A-IP |
| 02:20.10 | SeRi | Not bad.... Is ok. |
| 02:21.35 | WIMPy | I've been told they have interoperability issues since they've become an independant company. |
| 02:22.34 | SeRi | Cant say right now much about it. I just got it two days ago. |
| 02:22.49 | SeRi | What I know is that there is a huge delay on the key press when I call my vmail |
| 02:23.27 | *** join/#asterisk cmendes0101 (~cmendes01@wtnl.corp.tierra.net) |
| 02:23.40 | SeRi | like when inputing my password |
| 02:23.48 | SeRi | not sure why.... but is there |
| 02:28.30 | carrar | don't call your vm |
| 02:28.33 | carrar | FIXED |
| 02:28.52 | carrar | anything else I can resolve? |
| 02:31.37 | ChannelZ | It's just people wanthing things anyway. |
| 02:32.08 | carrar | All you need is a sound card and a soldering iron |
| 02:34.20 | [TK]D-Fender | ~savemoney |
| 02:34.20 | infobot | <Gremlin> It's okay, but I was hoping to save some money with a soldering iron and a bunch of sound cards. |
| 02:34.27 | [TK]D-Fender | \o/ |
| 02:44.30 | SeRi | lol |
| 02:44.44 | SeRi | I wont use the phone at all. how about that :) |
| 02:44.54 | carrar | woah |
| 02:45.05 | *** join/#asterisk deo (~dnepangue@203.177.214.75) |
| 02:45.09 | carrar | PICS of you smashing the phone |
| 02:45.11 | carrar | !! |
| 02:45.16 | SeRi | Is the kitchen phone and nobody needs to be calling the vmail excpet me and I dot from my office |
| 02:45.30 | SeRi | LOL |
| 02:46.02 | SeRi | well zZZz time. cya guys |
| 02:46.06 | SeRi | g/n |
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| 06:20.19 | schmidts | good morning |
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| 06:20.40 | Jaxon007_ | On my asterisk server has one SIP trunk connected to another asterisk machine , i want to know using ChanISAvail application will i get the status of that trunk ( means whether trunk is busy or unavailable ..etc.) |
| 06:20.42 | Jaxon007_ | ? |
| 06:21.40 | schmidts | Jaxon007_ only if you set a limit to it in the sip conf, like limit=1 then you should get a busy if this limit is reached |
| 06:24.16 | mirela666 | call-limit=1 , alowing that trunk to have only one call at the time |
| 06:24.57 | Jaxon007_ | @schmidts: thanks, will i get the status if the channel is unavailable...And is there any other way to get the status of trunk?..my scenario is i will originate the only if the trunk is available .. |
| 06:27.51 | kaldemar | Jaxon007_: func DEVICE_STATE |
| 06:30.32 | schmidts | Jaxon007_ imho you need two ways of checking it, with func DEVICE_STATE you get the state of the trunk and if you have qualify enabled you will also see if its unreachable and also should see an inuse state when there is a call active |
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| 06:59.46 | Jaxon007_ | Is there any method to know how many live calls are on a particular trunk?. |
| 07:02.35 | schmidts | Jaxon007_ you have to use GROUP_COUNT for it |
| 07:04.02 | WIMPy | SIPPEER(currcalls) |
| 07:05.02 | schmidts | WIMPy curcalls with only one r ;) |
| 07:05.08 | mirela666 | cool func :) |
| 07:05.16 | WIMPy | right |
| 07:07.56 | kaldemar | curcalls is only available if call-limit is set. the call-limit option was marked deprecated in 1.6.0. the GROUP functions are favored. |
| 07:08.10 | kaldemar | quite a handy field though. |
| 07:08.53 | WIMPy | Don't trust any documentation. Maybe it works with callcounters. |
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| 07:44.41 | Jaxon007_ | can I use group_count() from AMI, i know its possible in originating call from ami and call the group_count application . but its take more cpu resources if i check the channel count for every call. suppose we have 1000 calls to dial . |
| 07:47.10 | WIMPy | If you're on AMI you can just count yourself. |
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| 07:49.04 | kaldemar | Jaxon007_: manager show command Getvar |
| 07:50.04 | kaldemar | i'm not absolutely sure if you can execute functions with that, but worth a try. |
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| 08:33.10 | mcolombo | hi all |
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| 09:04.04 | mcolombo | it's possible to configure asterisk to send RE-INVITE instead UPDATE messages? |
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| 09:15.00 | schmidts | mcolombo asterisk would do this if update messages arent supported from the client |
| 09:22.59 | mcolombo | ok, but this update are sent by asterisk for change codec |
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| 09:23.04 | mcolombo | G711 to T.38 |
| 09:24.31 | schmidts | an update for codec? never seen this |
| 09:24.33 | mcolombo | is fine to send these messages, but the provider asks me to send re-invite not update |
| 09:25.34 | schmidts | AFAIK is an UPDATE only for changing headers but not for changing sdp stuff |
| 09:25.41 | schmidts | which version do you use? |
| 09:26.10 | mcolombo | i'm using SVN-oej-darjeeling-prack-1.8-r369882M- |
| 09:26.15 | mcolombo | for prack support |
| 09:26.30 | schmidts | hmm ok |
| 09:27.08 | schmidts | oej is doing some trainings this week so i dont know if he would answer when i ask him about this ;) |
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| 09:27.39 | schmidts | ok, it looks like update is ok to change the sdp, havent know this ;) |
| 09:28.16 | mcolombo | ok, no problem |
| 09:28.18 | mcolombo | thanks |
| 09:30.03 | jmls | wdoekes: what's the best firmware for the cisco ? |
| 09:30.27 | wdoekes | no idea, the latest? |
| 09:30.41 | schmidts | jmls which cisco? |
| 09:30.47 | jmls | 79XX |
| 09:30.59 | schmidts | ah ok, dont know ;) i only use the SPAs |
| 09:31.26 | jmls | wdoekes: you suggested upgrading the firmware :) |
| 09:31.42 | wdoekes | you were sending the REGISTERs without From-tag? |
| 09:31.54 | jmls | uh huh :-D |
| 09:32.15 | wdoekes | any firmware from this decade would do, I think? |
| 09:32.18 | jmls | well, not me personally, but my phones were ... |
| 09:32.22 | jmls | hahhhahahahahhaha |
| 09:32.42 | jmls | that's only 18 months :) |
| 09:33.07 | schmidts | mcolombo i will take a look at the code |
| 09:37.07 | schmidts | mcolombo you have to set the canreinvite or directmedia setting for your trunk to something else than update, than it shouldnt send an update anymore |
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| 09:57.41 | fredericve | Hi, I'm running asterisk 1.8.17.0, and I want callers to be able to join a queue, even when there is only one member that is currently in the paused state. |
| 09:57.48 | fredericve | how would I go about that? |
| 09:58.19 | fredericve | Right now I see it joining the queue, but immediately leaving again as well |
| 09:59.44 | fredericve | Ok nevermind I found the solution. |
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| 10:05.54 | k610 | asterisk link to odbc linked to redis someone tried this ? |
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| 10:17.42 | mcolombo | schmidts : Thanks, i will try it |
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| 10:38.24 | salz212 | hi, it there any way to get SIP METHODs in a call in Asterisk through some variable? |
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| 10:45.33 | kaldemar | salz212: be more elaborate |
| 10:46.41 | salz212 | okay, I want to get 183 and 180 Responses in a variable while giving fake ringing. This is required to STOP the custom FAS |
| 10:48.26 | salz212 | Am I clear? let me know if you have any question. |
| 10:51.58 | kaldemar | when you Dial something, the dialplan execution sticks in the Dial app. there's nothing you can do in dialplan until you get an answer. |
| 10:54.25 | schmidts | salz212 maybe the M option of dial can help you, but this is only called WHEN the other side answers the call, early media wouldnt execute this imho |
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| 11:07.14 | salz212 | what if I want when the other side .. starts ringing? that is what I want.. I am fake ringing till I get remote ringing. |
| 11:08.12 | schmidts | salz212 why do you want it that way? that would sound annoying hearing one ring in a different sound until the other ringing starts |
| 11:08.28 | schmidts | and how long do you think it takes until the other side starts to ring? |
| 11:08.44 | Greenlight | Using the AMI Bridge action, is there a way to prevent the the calling channel hanging up when the called channel hangs up, and have it go back into the dialplan ? |
| 11:08.46 | schmidts | if you would hear 1 or 2 seconds silence until you hear the ringing from the other side would for me sounds better |
| 11:10.32 | Greenlight | Or alternatively, is there a way to acomplish the same thing - join two arbritary channels, and then have them not hangup after the call ends? |
| 11:11.31 | salz212 | yeh you are right but mine is custom scenario... In case of SIP call I am sending signal to the remote side to get registered on SIP .. menwhile I am doing FAS. |
| 11:12.13 | schmidts | salz212 then you should take a look at progress but i am not sure if this could help you |
| 11:12.45 | salz212 | I am few things in mind... will try some external control to this. |
| 11:16.37 | *** join/#asterisk mathis_ (~crytic@lvps176-28-10-76.dedicated.hosteurope.de) |
| 11:16.54 | mathis_ | hello everyone :) |
| 11:17.31 | mathis_ | has anyone in here ever worked with the siren14 codec, especially with decoding recordings done by asterisk in siren14? |
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| 11:54.47 | Jaxon007_ | what will b the default volume level when a call land in Meetme conerence ?..when increase talking volume in meemte how much level it increase s? |
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| 13:14.20 | wonderworld | if i'd like to test the performance of a conference system, which tool would you recommend? i stumbled upon SIPp and was wondering if it's the best choice. |
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| 13:20.29 | mathis_ | I know I asked this earlier but there was a lot of joining going on afterwards :) has anyone in here ever worked with the siren14 codec? |
| 13:20.42 | mathis_ | and maybe knows how to decode recordings done by Asterisk in siren14 |
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| 13:22.20 | twanny796 | what is the bandwith needed for a sip phone with G711? |
| 13:23.32 | carrar | 88k |
| 13:23.45 | carrar | ~88k |
| 13:23.57 | mathis_ | well, G.711 is 64kbit/s |
| 13:24.02 | mathis_ | that plus a little overhead |
| 13:24.07 | mathis_ | I think 88kbit/s sounds about right :) |
| 13:24.13 | mathis_ | maybe a bit less |
| 13:25.19 | twanny796 | is 64kbit/s for both sides or should it be times two ?? |
| 13:25.26 | leifmadsen | 64k + SIP overhead |
| 13:25.40 | leifmadsen | it's right around that anyways |
| 13:27.26 | [TK]D-Fender | 80kbs |
| 13:27.30 | [TK]D-Fender | bkps |
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| 13:29.00 | malcolmd | mathis_: i think i saw your forums post about that; sorry, i don't have an answer for you. |
| 13:30.44 | carrar | From a Cisco switch counter the traffic looks like roughly |
| 13:30.44 | carrar | <PROTECTED> |
| 13:30.44 | carrar | <PROTECTED> |
| 13:30.47 | mathis_ | malcolmd: yeah, not trying to hurry anyone, but I thought it might help asking there, too ;) |
| 13:31.08 | mathis_ | malcolmd: you wouldn't happen to know who encoded the music on hold files in siren14, would you? |
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| 13:40.44 | leifmadsen | my guess is they were done automatically by a scrit |
| 13:40.45 | leifmadsen | script* |
| 13:40.50 | leifmadsen | from a source .wav file |
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| 13:43.32 | mathis_ | probably, yeah. the commands in that script would be highly interesting though ;) |
| 13:43.48 | mathis_ | although I don't think that there is any cli option I haven't tried yet for the decode |
| 13:43.51 | mathis_ | *decoder |
| 13:46.40 | malcolmd | mathis_: repotools has a soundtools package. it converts from .wav to the target format |
| 13:47.06 | malcolmd | http://svn.digium.com/svn/repotools/ |
| 13:47.12 | malcolmd | sound_tools, that is |
| 13:47.22 | leifmadsen | heh |
| 13:47.26 | leifmadsen | yes, open source ftw :D |
| 13:47.47 | mathis_ | leifmadsen: amen to that. :) |
| 13:47.58 | leifmadsen | those are the scripts I used to run to build audio files |
| 13:47.58 | mathis_ | malcolmd: thanks a lot, I'll take a look at that |
| 13:48.03 | leifmadsen | someone else runs those now :) |
| 13:48.07 | mathis_ | :) |
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| 13:49.30 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 13:55.25 | mathis_ | mhm. the converter script would like to cll make32kraw and makesiren14 |
| 13:55.31 | mathis_ | which is not in repotools |
| 13:57.33 | mathis_ | arrr. |
| 13:57.35 | mathis_ | pebcak. |
| 13:57.53 | jmetro | rtfm pebkac!! |
| 13:57.58 | mathis_ | :) |
| 13:58.03 | jmetro | XD |
| 13:58.23 | mathis_ | well, half pebcak. make32kraw is a function, makesiren14 not |
| 13:59.27 | *** join/#asterisk bjornts (~BTS@it010226.klientdrift.uib.no) |
| 14:02.50 | salz212 | guys is there a way of getting SIP METHODs in AMI? |
| 14:03.39 | salz212 | I am listenting to 5038 port all of it... Can't see any message do I have to use Command in AMI and SIP_HEADER function to do this? |
| 14:06.44 | *** join/#asterisk vlad_starkov (~vlad_star@c-50-136-210-10.hsd1.ca.comcast.net) |
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| 14:11.55 | *** mode/#asterisk [+o sruffell] by ChanServ |
| 14:13.26 | mathis_ | leifmadsen: let me guess, makesiren14 is not included due to Polycom license issues? ;) |
| 14:13.47 | leifmadsen | mathis_: might be possible |
| 14:14.00 | leifmadsen | I don't really remember :) |
| 14:15.02 | mathis_ | :) yeah judging from the svn log it has been over 2 years |
| 14:16.10 | mathis_ | if only the source code they supply would compile on linux |
| 14:16.50 | mathis_ | but that would not make that big of a difference I guess. the sound files encoded using the windows binary play just fine in asterisk, its just the recordings I can not decode |
| 14:22.10 | *** join/#asterisk jmls (~julian@host217-36-208-155.in-addr.btopenworld.com) |
| 14:22.35 | jmls | just been asked to explain what this means |
| 14:22.37 | jmls | WARNING[22945][C-00004962]: translate.c:343 framein: no samples for lintoulaw |
| 14:24.34 | *** join/#asterisk brdude (~brdude@c-24-7-76-160.hsd1.ca.comcast.net) |
| 14:27.03 | *** join/#asterisk Defraz (~Defraz@mail.pocatellochildren.com) |
| 14:32.48 | AkkerKid | yawns |
| 14:35.20 | *** join/#asterisk vlad_starkov (~vlad_star@c-50-136-210-10.hsd1.ca.comcast.net) |
| 14:35.56 | jmetro | jmls: based on context and the error message, it looks like asterisk has no way to convert from Lin to Ulaw sound |
| 14:37.13 | Qwell | jmetro: That's not accurate. |
| 14:37.26 | Qwell | It has a way, it's just unable to with that channel. |
| 14:37.39 | mjordan | jmls: what was the channel type? |
| 14:38.07 | mjordan | (and ooo - 4962 channels created on an Asterisk 11 system... nifty!) |
| 14:38.16 | mjordan | actually, 4962 calls. |
| 14:38.34 | mathis_ | nice! |
| 14:38.41 | mathis_ | is the machine still breathing? ;) |
| 14:38.46 | jmls | simultaneous calls ? |
| 14:38.51 | mjordan | jmls: no |
| 14:39.30 | mjordan | jmls: but your call ID there was 00004962. That's a monotonically increasing number, which - while it doesn't necessarily translate perfectly to channels/calls, is a good enough number for government work |
| 14:39.46 | jmls | pah. core show calls : 18 active calls, 24730 calls processed |
| 14:39.46 | *** join/#asterisk deo (~dnepangue@112.198.79.53) |
| 14:39.50 | mathis_ | ah. |
| 14:39.52 | *** part/#asterisk deo (~dnepangue@112.198.79.53) |
| 14:39.54 | mjordan | :-) |
| 14:40.35 | jmls | mjordan: channel types are all sip. |
| 14:40.36 | jmls | now. |
| 14:40.39 | jmls | :) |
| 14:41.21 | mathis_ | leifmadsen: do you think that I have a chance on a positive answer if I contact Digium about this? |
| 14:43.44 | leifmadsen | mathis_: I don't know what you mean by that question |
| 14:43.44 | *** join/#asterisk kchehab (~ck@77.42.241.66) |
| 14:43.45 | kchehab | hi |
| 14:44.00 | *** part/#asterisk Jaxon007_ (~root@123.252.144.92) |
| 14:44.31 | kchehab | i have too many connections in mysql processlist while using and application exten => _96771.,1,MYSQL(Connect connid localhost root xxx whitelist) |
| 14:44.36 | *** join/#asterisk sertaconay (~sertacona@unaffiliated/sertaconay) |
| 14:45.01 | leifmadsen | kchehab: it's typically recommended that you use res_odbc, not res_mysql |
| 14:45.26 | leifmadsen | then use the func_odbc stuff to make connections and execute SQL |
| 14:45.29 | *** join/#asterisk kchehab (~ck@77.42.241.66) |
| 14:45.34 | kchehab | res_odbc |
| 14:46.12 | leifmadsen | kchehab: http://asteriskdocs.org/en/3rd_Edition/asterisk-book-html-chunk/installing_configuring_odbc.html |
| 14:47.00 | kchehab | leifmadsen res_mysql cause this error ? |
| 14:47.11 | leifmadsen | I don't know what the "error" is |
| 14:47.24 | leifmadsen | I'm just saying, using the MYSQL() application is not really recommended |
| 14:48.00 | kchehab | leifmadsen my problem that i alot of sessions in processlist has sleep mode |
| 14:48.02 | leifmadsen | res_odbc will likely manage the connections better since you don't do it from the dialplan manually |
| 14:48.32 | mathis_ | leifmadsen: well, I don't pay for commercial asterisk support but have a question regarding a codec they (digium) provide, so I am not sure if they will be offering any kind of help |
| 14:48.45 | mathis_ | gaah, not really sure what I mean myself :) |
| 14:49.01 | leifmadsen | mathis_: you could use the contact form on asterisk.org and find out I guess |
| 14:49.11 | leifmadsen | you may or may not receive a response |
| 14:50.45 | *** join/#asterisk vfabi (~fabi@host-static-93-116-255-140.moldtelecom.md) |
| 14:51.08 | mathis_ | true |
| 14:51.58 | kchehab | leifmadsen odbc can work on asterisk 1.6 |
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| 14:52.27 | leifmadsen | kchehab: and 1.4 |
| 14:52.50 | leifmadsen | bites his tongue about his pet-peeve about 1.6 not being a version |
| 14:52.56 | leifmadsen | oh wait, I guess I didn't |
| 14:53.19 | pabelanger | I always though app_directory would loop the main menu if no input was detected. But, it just drops the call |
| 14:55.10 | kchehab | leifmadsen i cant find res_odbc.so |
| 14:55.26 | kchehab | leifmadsen can i install it using yum as i am using elastix distro |
| 14:55.37 | leifmadsen | you're on your own then :) |
| 14:55.45 | leifmadsen | knows nothing about non-vanilla asterisk |
| 14:55.46 | [TK]D-Fender | kchehab, ask them. We are not responsible for their packaging. |
| 14:56.06 | kchehab | ok thanks |
| 14:58.51 | kchehab | [TK]D-Fender far away from my distro but why i can find alot of sleep sessions in mysql proccess list and i am opening and closing the connection each time |
| 15:00.14 | *** join/#asterisk bchia (~Adium@nat/digium/x-mnvyutlvcejramgl) |
| 15:00.19 | [TK]D-Fender | kchehab, You have told us nothing specific about your installation. We have nothing to comment on |
| 15:01.56 | kchehab | [TK]D-Fender please check my syntax |
| 15:02.01 | kchehab | [TK]D-Fender at http://pastebin.com/CQdfaaU6 |
| 15:06.48 | *** join/#asterisk wonderworld (~ww@dsdf-4d0a1b22.pool.mediaWays.net) |
| 15:07.18 | [TK]D-Fender | kchehab, Nothing to check for.... if you have a process leak, it's somewhere else. the dialplan should not being doing damage all by itself. |
| 15:12.53 | *** join/#asterisk vinhdizzo (~vinh@vqn-portege.ics.uci.edu) |
| 15:13.22 | pabelanger | leifmadsen, You using Directory() in any dialplans? |
| 15:13.42 | leifmadsen | yep, on Roberts I think |
| 15:14.01 | pabelanger | Hmm |
| 15:14.07 | leifmadsen | I could be wrong though |
| 15:14.40 | pabelanger | I cannot figure out how / why Directory will drop the call and not fall through the dialplan on no DTMF entry on the dir-intro greeting |
| 15:14.55 | leifmadsen | option? |
| 15:16.12 | pabelanger | not that I see |
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| 15:26.55 | jmls1 | mjordan: 25832 calls processed. Still hanging in there ;) |
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| 15:51.17 | johoja | hi guys, i'm having an issue with outgiong calls, they get dropped after about 30 seconds -- i have the debug log from here http://pastebin.com/EesJdUEr |
| 15:51.35 | johoja | it looks like the peer making the call does not respond to the Invite being sent by asterisk |
| 15:51.48 | leifmadsen | sounds like a firewall issue perhaps |
| 15:51.52 | leifmadsen | asterisk behind NAT? |
| 15:51.59 | johoja | asterisk is public |
| 15:52.05 | [TK]D-Fender | johoja, <--- Transmitting (NAT) to 192.168.75.10:5060 ---> <- why is a Class-C PRIVATE IP listed as NAT? |
| 15:52.22 | [TK]D-Fender | johoja, Contact: <sip:16136008510@152.48.1.105:5060> <- You are sending them a PUBLIC IP to respond to. |
| 15:52.29 | leifmadsen | possible routing issue then? the phone is likely responding, but it's not getting routed back to asterisk |
| 15:52.45 | johoja | It's a bit of a strange setup |
| 15:52.48 | leifmadsen | asterisk will not keep the call up unless it gets an acknowledgement |
| 15:52.56 | leifmadsen | suggests makes it less strange |
| 15:53.03 | leifmadsen | s/makes/making/ |
| 15:53.06 | johoja | the 152.48.1.105 is reachable and is the asterisk server ip. |
| 15:53.11 | [TK]D-Fender | Retransmitting #4 (NAT) to 192.168.75.10:5060: |
| 15:53.12 | johoja | teh peer is reaching that ip using an ipsec tunnel. |
| 15:53.39 | leifmadsen | ya, so routing issue likely |
| 15:53.50 | johoja | there is an ipsec tunnel from 152.48.1.105 <-> 192.168.75.0/24 |
| 15:53.50 | leifmadsen | default gateway might be routing it correctly |
| 15:53.54 | [TK]D-Fender | If it's effectively direct you need to set you peer as NOT being NAT'd |
| 15:54.19 | johoja | there is a router at 192.168.75.* |
| 15:54.23 | johoja | and the peer is behind that router. |
| 15:54.28 | leifmadsen | that would make sense, yes |
| 15:54.34 | leifmadsen | what kind of router? |
| 15:54.36 | johoja | the asterisk server is public |
| 15:54.38 | johoja | its just a d-link |
| 15:54.41 | johoja | dsl modem. |
| 15:54.50 | leifmadsen | it's not likely the asterisk server that is the problem |
| 15:54.57 | leifmadsen | it's the d-link probably not routing the response correctly |
| 15:55.06 | johoja | the peer is a linksys pap2t |
| 15:55.26 | johoja | should i try truning nat=off ? |
| 15:55.30 | leifmadsen | I have no other suggestions other than to verify your routing and that responses on the far end network are right |
| 15:55.59 | jmetro | home network eh |
| 15:57.00 | johoja | any other suggestiosn ? |
| 15:57.08 | johoja | as far as i know the far end is reachable... |
| 15:57.32 | johoja | for both sides. |
| 15:59.07 | wonderworld | how can i find out which timing module asterisk is using currently? |
| 15:59.22 | johoja | would having nat=on be wrong? I thought in this case it would be right, since the peer is effectifly behind asterisk. |
| 15:59.50 | jmetro | behind asterisk from a call standpoint, not from a network standpoint. |
| 16:01.40 | *** join/#asterisk Galen (~Galen@rrcs-24-43-20-118.west.biz.rr.com) |
| 16:03.21 | johoja | jmetro: could you explain ? |
| 16:03.45 | johoja | jmetro: from what i see in the sip headers, the pap2 should respond to the invite |
| 16:03.52 | johoja | using 16136008510@152.48.1.105:5060 |
| 16:03.59 | johoja | which is a reachable ip ... ? |
| 16:04.07 | johoja | and that is where asterisk is running. |
| 16:05.05 | navaismo | wonderworld: try with: module show like timing |
| 16:06.18 | wonderworld | navaismo: thanks, that worked |
| 16:18.58 | AkkerKid | Hi everyone! I have a physical fax machine connected to an Openvox FX(S/O) card and on the other end is a SIP Trunk running G.711 ulaw. Hylafax runs great but the physical fax complains about line quality. Both connect to the outside world via the SIP trunk. What can I do to improve "line quality" for the physical fax machine? |
| 16:19.14 | kresp0 | Hi there, |
| 16:19.15 | kresp0 | We are trying to make a call using some voipprovider, and for some reason asterisk tries to make the call using the same "From" and "To", instead of ${EXTEN} |
| 16:19.29 | kresp0 | This is the line that makes the call on extensions.conf: |
| 16:19.29 | kresp0 | exten => _00349ZXXXXXXX,1,Dial(SIP/voipcheap-brian,${EXTEN}) |
| 16:19.36 | kresp0 | And here is the full sip log for the failed call: |
| 16:19.37 | kresp0 | http://pastebin.com/Xc3bsH03 |
| 16:19.46 | n3hxs | AkkerKid, you might bump the gain up a bit for the fax machine. |
| 16:21.11 | AkkerKid | n3hxs: is that a dahdi setting I need to adjust or is it more in the drivers of the card? |
| 16:22.03 | kresp0 | as you can see, both fields (from and to) points to <sip:myuser@sip.voipcheap.com> |
| 16:22.04 | AkkerKid | There's also maybe 200ft of cat5 from the card to the machine too. I don't know if that makes a difference, though |
| 16:23.18 | [TK]D-Fender | AkkerKid, If you're not using T.38 expect failures. Lots of failure |
| 16:24.20 | AkkerKid | D-Fender: unfortunately my ITSP doesn't yet support T.38 so I'm stuck with minimising failure with ulaw. |
| 16:25.03 | [TK]D-Fender | kresp0, exten => _00349ZXXXXXXX,1,Dial(SIP/voipcheap-brian,${EXTEN}) <--- there is a COMMA there where there shouldn't be..... |
| 16:25.38 | kresp0 | >.< |
| 16:26.08 | kresp0 | thank you [TK]D-Fender |
| 16:26.10 | kresp0 | again |
| 16:26.32 | [TK]D-Fender | kresp0, And while we're at it ... you set your peer to them as NAT=YES. Almost never is it appropriate to do that |
| 16:26.52 | *** join/#asterisk TimeRider (~steve@host81-136-216-215.in-addr.btopenworld.com) |
| 16:27.23 | [TK]D-Fender | ITSP's are almost never in a situation that you shuold have that set to YES, and in many cases where their SIP is separate from their media server it will cause audio to fail and calls to die |
| 16:28.31 | Qwell | drmessano: Have some codecs. |
| 16:30.56 | *** join/#asterisk vfabi (~fabi@host-static-89-41-121-42.moldtelecom.md) |
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| 16:46.21 | drmessano | YAY!!! |
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| 16:53.54 | kresp0 | [TK]D-Fender, I forgot to say: thanks again and fixed both config errors ;) |
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| 17:15.53 | [TK]D-Fender | kresp0, Glad to hear. |
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| 17:51.21 | *** join/#asterisk Teduardo (~DW-10297@dhcp92.cmh.ee.net) |
| 17:51.54 | Teduardo | Does anyone know of an autoattendant that you can use with asterisk that will ask the caller for a 'support code' and when they enter it to a lookup on an external resource? |
| 17:53.35 | [TK]D-Fender | Teduardo, Yes, it's called "Asterisk" |
| 17:54.33 | *** join/#asterisk elico (~Thunderbi@109.64.221.76) |
| 17:55.45 | Teduardo | ah, I figured the AA was handled by a specific module, I'll do more research, apparently with shoretel to do what I described above the software costs $35,000 |
| 17:55.53 | Teduardo | So... yea |
| 17:56.04 | Qwell | Teduardo: I'll help you do it for $34,000 |
| 17:56.38 | sirsquishy | lol |
| 17:56.43 | leifmadsen | I'm only $25k |
| 17:56.45 | *** join/#asterisk pabelanger (~pabelange@asterisk/contributor-and-bug-marshal/pabelanger) |
| 17:56.45 | *** mode/#asterisk [+o pabelanger] by ChanServ |
| 17:57.03 | sirsquishy | i still cant beleive shoretel has no backup to the HQ server or the DVS's |
| 17:57.09 | Qwell | $22k. Anybody that offers to do it for less is clearly using outsourced labor. |
| 17:57.19 | Qwell | Probably outsourced to Canada. |
| 17:57.43 | pabelanger | sirsquishy, why? outage? |
| 17:57.50 | Teduardo | Shoretel is crazy anyway in the sense that if you dont use one of like 5 SIP providers you need an adtran to do SIP still |
| 17:57.51 | leifmadsen | Qwell: lolz |
| 17:57.58 | Qwell | leifmadsen: <3 |
| 17:57.59 | Teduardo | its like almost 2013 |
| 17:58.11 | sirsquishy | well with an MPLS setup if a site was to go down that has a DVS anything tied to that DVS (other sites on the MPLS) dont route calls :-) |
| 17:58.40 | sirsquishy | if the HQ site were to go down, the entire MPLS shoretel cloud is dead |
| 17:58.41 | sirsquishy | :-) |
| 17:58.52 | sirsquishy | its really nice how their N+1 doesnt work as described |
| 17:59.09 | Teduardo | shoretel is alot better than the uc5xx system we were using by cisco tho |
| 17:59.22 | sirsquishy | Cisco sucks anyway |
| 17:59.31 | sirsquishy | call manager was and is garbage |
| 17:59.49 | Teduardo | still kind of pisses me off that you need an adtran to do VoIP and the ECC software is so expensive |
| 17:59.53 | Teduardo | with shoretel |
| 17:59.56 | Teduardo | but oh well whatever =D |
| 18:00.20 | sirsquishy | ECC is a joke |
| 18:00.28 | anonymouz666 | [TK]D-Fender: Trying to configure a PolycomSoundStationIP-SSIP_5000, but there's no button to save the config. any clue why? |
| 18:00.29 | sirsquishy | ECC has to be at the same location as your agent |
| 18:00.41 | *** join/#asterisk pabelanger (~pabelange@asterisk/contributor-and-bug-marshal/pabelanger) |
| 18:00.41 | *** mode/#asterisk [+o pabelanger] by ChanServ |
| 18:00.47 | sirsquishy | and you do not need an adtran to SIP to Shortel, i do it with an asterisk box NP |
| 18:00.51 | [TK]D-Fender | anonymouz666, Save where? |
| 18:00.58 | [TK]D-Fender | anonymouz666, From where? How? |
| 18:01.07 | anonymouz666 | save the changes I did to the phone |
| 18:01.25 | Teduardo | sirsquishy: ah I was told that unless you use one of their approved sip trunk providers you need a external SIP handler and have to connect it using the trunk ports |
| 18:01.32 | anonymouz666 | there's a submit button, but when I apply the old config back |
| 18:01.33 | sirsquishy | nope |
| 18:01.42 | sirsquishy | i use voip.ms to asterisk to shoretel |
| 18:01.44 | sirsquishy | works quite well |
| 18:01.46 | anonymouz666 | bizarre |
| 18:01.48 | [TK]D-Fender | anonymouz666, There is no "export" for full configs. manual overrides can be saved from provisioning however |
| 18:02.12 | Teduardo | Ah, I just wanted our broadvox trunks to terminate directly in the shoretel box and the installers told me to suck it |
| 18:02.21 | anonymouz666 | I am in WEB config... I just want to change the auth ID |
| 18:02.28 | sirsquishy | physcal trunks or SIP? |
| 18:02.31 | [TK]D-Fender | Poeple configuring Polycom phones via the web interface should be dragged out and shot. Survivors should be shot AGAIN. |
| 18:02.33 | Teduardo | SIP |
| 18:02.50 | anonymouz666 | [TK]D-Fender: It's just a username |
| 18:02.53 | *** join/#asterisk pabelanger (~pabelange@asterisk/contributor-and-bug-marshal/pabelanger) |
| 18:02.53 | *** mode/#asterisk [+o pabelanger] by ChanServ |
| 18:02.53 | sirsquishy | dump them into Asterisk then build a intratrunk between that and Shoretel on Conference ports. |
| 18:02.55 | [TK]D-Fender | Shoretel = suck. Proprietary ~= suck |
| 18:03.00 | sirsquishy | ^ |
| 18:03.51 | Teduardo | thats really no different than using an adtran between the shoretel box and the sip trunk except that it's asterix instead of adtran either way it sucks that you have to do any of that and that it can't just connect to the sip trunk directly |
| 18:04.04 | Qwell | What's asterix? |
| 18:04.13 | sirsquishy | the term is siperator IIRC |
| 18:04.23 | carrar | TK, BUT ShoreTel now has a AMAZING new feature where you can make calls from your cell phone that showup as your PBX Phone number now!!! :) |
| 18:04.26 | Teduardo | asterisk sorry typo |
| 18:04.36 | sirsquishy | i can do that on asterisk :-) |
| 18:04.40 | carrar | yeah |
| 18:04.43 | carrar | it's just funny |
| 18:04.52 | Qwell | carrar: You mean Google voice? |
| 18:04.53 | sirsquishy | i know |
| 18:04.53 | carrar | Cause ShoreTel is all over it |
| 18:04.55 | Qwell | s/v/V/ |
| 18:05.38 | carrar | not to mention switchvox has had that in their iphone app for a long time |
| 18:05.49 | carrar | doesn't work as nice but it works |
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| 18:08.19 | [TK]D-Fender | Shoretel = Beached whale |
| 18:09.33 | carrar | They have orange on their boxes |
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| 18:55.38 | justnulling2 | during update i have overwritten sip.conf file by mistake, and there is no back up, is there a way to dump relevant data from running asterisk (as it wasn't reloaded yet) into sip.conf, or it must be done by hand? |
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| 19:00.54 | leifmadsen | wow... |
| 19:01.44 | leifmadsen | there is no sip export commands that I'm aware of |
| 19:01.53 | leifmadsen | you'll have to re-create the file it sounds like |
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| 19:06.15 | jmetro | there is a dialplan export |
| 19:07.52 | leifmadsen | right :) |
| 19:08.03 | leifmadsen | not useful for sip.conf unfortunately |
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| 19:09.52 | jpsharp | backups are for chumps and people with no sense of adventure. |
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| 19:11.28 | pabelanger | ~book |
| 19:11.28 | infobot | Asterisk: The Definitive Guide, 3rd Edition (ISBN 0-596-51734-3) available at http://oreilly.com/catalog/9780596517342 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and is available for reading online at http://www.asteriskdocs.org/ or see ~buybook |
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| 19:29.36 | justnulling2 | ok let me do it by hand, i ran sip show registry (doesn't dump password :( and sip show users is there anything that can be dumped that goes into sip.conf? |
| 19:31.19 | mjordan | justnulling2: what version of Asterisk are you on? |
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| 19:55.52 | bobb_WU | Anyone know how to troubleshoot a message like this? X-Asterisk-HangupCause: Protocol error, unspecified X-Asterisk-HangupCauseCode: 111 |
| 19:56.16 | justnulling2 | mjordan: 10.9 |
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| 20:19.09 | nny | what file is used to set USER and GROUP for asterisk to run as in 1.8? |
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| 20:23.14 | elguero | nny: asterisk.conf |
| 20:24.10 | nny | elguero: runuser = asterisk rungroup = asterisk ? |
| 20:24.43 | nny | yeah got it |
| 20:24.55 | elguero | nny: yep |
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| 20:30.42 | barbosa2 | what the best TTS for portuguese ? |
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| 20:43.38 | mjordan | justnulling2: theoretically, the astdb underlying Asterisk 10 is sqlite3. |
| 20:44.15 | mjordan | justnulling2: this means you can use the sqlite3 command line utility to open it and get the key/value pairs for your SIP endpoints - should be in table astdb |
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| 20:46.57 | justnulling2 | mjordan: where is the file located? thank in advanced, have to run now will check when get back |
| 20:47.02 | ruunix | hi there .. |
| 20:47.09 | ruunix | just a liittle question . |
| 20:47.40 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.216.2) |
| 20:48.00 | ruunix | i have register:passw@ip/extension ... but the extension is in another server ... what i need to look in google ? |
| 20:49.31 | mjordan | justnulling2: /var/lib/asterisk. It will only contain the currently registered peers, but its probably better than nothing |
| 20:49.53 | mjordan | (by default, by the way - your system configuration can change where the astdb is stored) |
| 20:50.34 | ruunix | may be exten => 600,1,Dial(SIP/user:passwd@host) |
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| 20:54.34 | ruunix | somebody ? |
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| 20:57.45 | kresp0 | hi ruunix |
| 20:58.34 | ruunix | hi |
| 20:58.47 | ruunix | i m tryng to setup a remote extension .. |
| 20:58.51 | ruunix | in extensions.conf |
| 21:00.14 | kresp0 | ruunix, please read this faq: |
| 21:00.15 | kresp0 | http://www.catb.org/esr/faqs/smart-questions.html |
| 21:00.58 | kresp0 | a bit large, I know. But I think that is essential to know how to ask for help on the internet |
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| 21:01.32 | nny | how do I set dahdi to have asterisk perms on the /dev/dahdi files? |
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| 21:02.16 | ruunix | Thanks .. maybe another time .. now i have a little call center down... lets google... |
| 21:02.27 | sirsquishy | haha |
| 21:02.33 | sirsquishy | more like, lets get fired |
| 21:03.13 | ruunix | or something like that too |
| 21:03.33 | kresp0 | damn sorry ruunix, I tought u just joined the channel and asking for someone, but that was I xD |
| 21:08.24 | nny | what's the method for setting the user dahdi runs as? |
| 21:09.25 | jpsharp | dahdi runs either as a kernel module or as the same user asterisk runs as. |
| 21:10.36 | nny | the dev nodes have root/root right now |
| 21:10.47 | nny | and it gets reset on reboot to root root if i modify them |
| 21:11.22 | nny | so I guess my question is "how do I force those nodes to be asterisk.asterisk on creation" |
| 21:14.36 | nny | god damn it doesn't get any easier to secure an asterisk install |
| 21:16.46 | nny | just gonna make a shell script that corrects perms before asterisk starts.. sigh |
| 21:23.09 | nny | hmm i'd much rather have dahdi's dev nodes have the proper permissions on creation. I see udev rules exist, but they don't seem to apply to this setup (centos based). What's the alternative way to have those nodes created properly |
| 21:23.18 | elguero | nny: http://astbook.asteriskdocs.org/en/3rd_Edition/asterisk-book-html/asterisk-book.html |
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| 21:23.27 | nny | ? |
| 21:23.27 | jmetro | ~book |
| 21:23.27 | infobot | Asterisk: The Definitive Guide, 3rd Edition (ISBN 0-596-51734-3) available at http://oreilly.com/catalog/9780596517342 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and is available for reading online at http://www.asteriskdocs.org/ or see ~buybook |
| 21:23.33 | nny | I have the book |
| 21:23.36 | elguero | thanks jmetro |
| 21:23.42 | jmetro | for future reference elguero |
| 21:23.47 | nny | it doesn't state how to create dev nodes properly |
| 21:23.49 | elguero | nny: it covers what you want to know |
| 21:24.22 | nny | ffs |
| 21:24.29 | nny | I see udev rules exist, but they don't seem to apply to this setup (centos based). |
| 21:24.37 | nny | the book states that the udev rules should apply |
| 21:24.41 | nny | therefor, it doesn't have what I need |
| 21:24.53 | elguero | did you look under "setting file permissions" |
| 21:25.05 | elguero | I think it does have what you need under that section |
| 21:25.18 | nny | lol |
| 21:25.31 | nny | i must be speaking french.. no, the udev rules aren't applied |
| 21:25.35 | elguero | nny: sorry, wasn't parsing |
| 21:25.40 | nny | under the section setting file permissions |
| 21:25.58 | elguero | attitude coming through irc shut that part of my brain off |
| 21:26.03 | elguero | sorry |
| 21:26.18 | nny | assuming this is a distro issue. Yeah i must not speak clearly when given answers I could find on my own. Thanks |
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| 21:44.16 | Merlin | what is the right way to enforce a global wrap-up time for agents in multiple queues in asterisk 1.6? is it shared_lastcall in the queue config, or wrapuptime in agents.conf ? |
| 21:46.00 | [TK]D-Fender | wrapuptime isn't from agents, it's a queue parm |
| 21:46.09 | [TK]D-Fender | and perhaps can be specified unger [general] |
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| 22:30.58 | elguero | nny: been messing around with dahdi.rules... it wasn't changing the owner for me either on CentOS 5.8... I renamed the file to 96-dahdi.rules... it worked... I renamed it back to dahdi.rules and seems to be working for me changing user and group... not sure why it wasn't working at first but it is now for me on CentOS... hope that helps |
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