| 00:00.05 | bmoraca_work | lol |
| 00:00.49 | bmoraca_work | hey manxpower, who do you get CNAM from? |
| 00:01.32 | manxpower | Accudata or some such if my boss ever signs the NDA. AsteriskCNAM.com is what I think we used for our prototyping |
| 00:01.51 | bmoraca_work | i wish they were cheaper |
| 00:02.40 | bmoraca_work | right now, if i want CNAM, it's $0.65/mo per DID for unlimited lookups. i'd prefer to have a one-off lookup for DIDs because a lot of my customers don't use their DIDs very often. |
| 00:03.11 | *** join/#asterisk obnauticus (~obnauticu@about/windows/regular/obnauticus) |
| 00:03.37 | fenrus | CNAM is some kind of caller-id-database ? |
| 00:03.38 | bmoraca_work | so i can't justify enabling CNAM on 100 DIDs when they take a combined total of 20 calls per day on them, but i know that i'm going to start getting CNAM complaints from certain customers soon |
| 00:03.38 | manxpower | asteriskcnam is something like 1.2/cents/lookup |
| 00:03.48 | manxpower | fenrus: cnam IS the callerid database |
| 00:04.03 | bmoraca_work | $0.009/lookup |
| 00:04.09 | bmoraca_work | but they're the cheapest i've found |
| 00:04.10 | fenrus | manxpower, in the usa i guesS=? |
| 00:04.12 | manxpower | yeah, that's it. |
| 00:04.24 | manxpower | fenrus: correct, but other countries may have similar things |
| 00:05.10 | fenrus | there's a couple of hacks to make asterisk look up numbers towards some free number-database-services in sweden. |
| 00:05.27 | manxpower | bmoraca_work: for people with lots of DIDs or want CNAM we usually put in a PRI via the ILEC |
| 00:05.43 | *** join/#asterisk Professional (~Pro@unaffiliated/shani) |
| 00:06.09 | bmoraca_work | unfortunately, i don't have that luxury (not big enough yet, not a CLEC) |
| 00:06.24 | bmoraca_work | primarily VOIP trunking |
| 00:07.11 | Professional | can any one help me with the weblink which simply explain the asterisk implementation with PD and usage , thanx |
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| 00:09.18 | bmoraca_work | i should just implement a cache and use my upstream provider for high-traffic numbers and asteriskcnam for DIDs |
| 00:09.41 | bmoraca_work | can never have enough func_odbc! |
| 00:10.03 | manxpower | bmoraca_work: I did that. Used sqlite, actually |
| 00:10.17 | manxpower | the all powerful lid_cache.sqlite3! |
| 00:10.33 | bmoraca_work | fun stuff |
| 00:10.38 | bmoraca_work | well, it's time to go home |
| 00:10.44 | bmoraca_work | until tomorrow! |
| 00:10.58 | manxpower | it was interesting. It is best to understand your traffic before spending the time one something like that. |
| 00:16.02 | grandpapadot | manxpower: Ok, I get variable inheritance, but how would I use it with CALLERID(xxx)? I tried Set(_CALLERID(num)=12345) but no go ... |
| 00:16.12 | grandpapadot | .. on the local child channel |
| 00:16.35 | manxpower | or maybe __MY_CALLERID_NUM=${CALLERID(num)} |
| 00:16.49 | manxpower | and do that in the parent channel before your Local/ Dial |
| 00:17.14 | *** part/#asterisk eskaypey (~Adium@unaffiliated/eskaypey) |
| 00:17.54 | grandpapadot | So no way to use inheritance on CALLERID()? |
| 00:19.49 | grandpapadot | The reason is I need to make this compatible with the child Local channel CALLERID(dnid) basically, so I need to set _CALLERID(DNID) in the parent ... There are literrally thousands of files I would have to update for customers to get a new variable set for DNID |
| 00:19.54 | grandpapadot | hrm... |
| 00:28.23 | *** join/#asterisk circut (~circut@c-71-57-110-244.hsd1.il.comcast.net) |
| 00:28.55 | circut | hey all, i just installed an FXS module on my TDM410 |
| 00:29.06 | grandpapadot | .. but that looks like my option, lol, DANGIT! |
| 00:29.12 | circut | ive got an analog phone plugged into it which gets a dialtone when picked up |
| 00:29.19 | circut | but the second i press a number i get a hangup |
| 00:29.34 | circut | nothing in the logs or on the console indicating whats wrong |
| 00:34.52 | *** join/#asterisk hipitihop (~denis@203.132.229.236) |
| 00:36.41 | hipitihop | is there a standard voice prompt for dealing with anonymous cid calls ? |
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| 00:39.21 | Nugget | I like tt-allbusy for that |
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| 00:42.45 | hipitihop | I would like something more specific like "We don't accept anonymous calls, please enable caller id and call back, goodbye" |
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| 00:46.13 | circut | wow |
| 00:46.29 | circut | after removing my head from the southern portion of my anatomy, i determined the problem was with contexts ;/ |
| 00:46.33 | circut | thanks guys |
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| 00:54.04 | Jumpie | haha |
| 00:54.11 | Jumpie | head to rectum contact will do that every time |
| 00:55.31 | p3nguin | Better than a2m, I guess. |
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| 01:47.29 | aschneiderg | Hey everybody; I've been searching a stable Android for the HTC Diamond all the web around. There's a lot of forks. I'm using the XDA-Forums and Connect-UTB's but apps are crashing all the time. Do you guys know of a better version? I understand there's a bunch of work on progress about it. |
| 01:54.16 | leifmadsen | aschneiderg: that doesn't seem like an Asterisk question, unless you misspoke |
| 01:54.32 | aschneiderg | (ouch) |
| 01:54.35 | aschneiderg | sorry all |
| 01:54.39 | aschneiderg | cheers |
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| 02:09.11 | carrar | Just add "... for Asterisk" to the end |
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| 02:50.59 | CoolCat2012 | hi all, could some give me some light on this http://forums.whirlpool.net.au/forum-replies-archive.cfm/1025313.html ? |
| 02:51.45 | CoolCat2012 | the post before the last one |
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| 03:12.22 | TJNII | CoolCat2012: So you're using Trixbox / FreePBX? |
| 03:14.16 | ChannelZ | The PBX of hookers |
| 03:14.31 | CoolCat2012 | freepbx |
| 03:14.40 | CoolCat2012 | (elastix bundled) |
| 03:14.40 | TJNII | CoolCat2012: #freepbx |
| 03:14.44 | TJNII | ~freepbx |
| 03:14.45 | infobot | [~freepbx] FreePBX is unable to be supported here. It is made up of complex dialplans and scripts which can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking there, or fpbx |
| 03:15.12 | CoolCat2012 | TJNII thanks! =o) |
| 03:15.59 | TJNII | ChannelZ: Once again you do not disappoint. Kudos. |
| 03:17.37 | ChannelZ | Sorry it's a Pavlovian response every time I hear "Trixbox" |
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| 03:33.46 | manxpower | ~freepbx |
| 03:33.47 | infobot | [~freepbx] FreePBX is unable to be supported here. It is made up of complex dialplans and scripts which can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking there, or fpbx |
| 03:33.53 | manxpower | ChannelZ: You too, eh? |
| 03:34.16 | manxpower | I sometimes can't help myself. my fingers type that by themselves, I swear! |
| 03:34.24 | CoolCat2012 | =o/ |
| 03:34.45 | CoolCat2012 | i would touch asterisk without something like freepbx |
| 03:34.51 | CoolCat2012 | *wouldnt |
| 03:35.15 | CoolCat2012 | (its not easy to setup those text file from the ground) |
| 03:35.18 | manxpower | CoolCat2012: then you are one of the few on this channel that feel that way. |
| 03:35.40 | CoolCat2012 | manxpower im just be sincerity, no offense. |
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| 03:35.57 | CoolCat2012 | i would love to say that i do control asterisk, but im far from that! |
| 03:36.42 | manxpower | "Like meth, you don't control it, trixbox controls you" |
| 03:37.26 | CoolCat2012 | no, i dont use trixbox.... |
| 03:42.46 | CoolCat2012 | cya people, night all! |
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| 04:33.55 | ChannelZ | wonders why he has The Price Is Right theme stuck in his head |
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| 05:05.59 | carrar | if you were raided by the FBI they would not even let you keep the quit message |
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| 06:14.20 | boodu | ciao |
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| 06:18.56 | Knightfal | ANyone know what might cause the following : DEBUG[2622] chan_dahdi.c: Failed to read gains: Invalid argument |
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| 06:26.09 | kaldemar | Knightfal: an invalid argument in your configuration? |
| 06:26.57 | Knightfal | Ya Im looking into it. I have a few pstn gateways and all configs are the same its strange. |
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| 06:32.04 | ChannelZ | farts a happy little tune |
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| 08:07.43 | petern_ | ponders his pri issue |
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| 08:19.29 | elzid | Morning all, have a question about ices 0.4 and its seeming inability to stream live audio - can it only stream playlists? I'm trying to avoid upgrading to ices 2.0 because it'll break legacy setups... |
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| 08:29.29 | petern_ | hmm, is "Restarting T203 counter" normal? |
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| 08:44.48 | binbash_ | Hi everyone, |
| 08:45.12 | binbash_ | i'm trying to intergrate fax in to my pbx, but i'm not finding that much info |
| 08:45.30 | binbash_ | i have a sip trunk from my provider, and i would like to do fax2e-mail on my pbx |
| 08:45.46 | Jumpie | its a bear to setup sometimes |
| 08:45.53 | Jumpie | there are things like hylafax and astrafax |
| 08:45.57 | Jumpie | but i have had best luck using an ata |
| 08:46.17 | binbash_ | ata? |
| 08:47.26 | binbash_ | hmm ah a google on ata made some clear |
| 08:47.33 | binbash_ | yeah because, we have like 100 numbers |
| 08:47.40 | binbash_ | and we route those numbers directly to phones |
| 08:47.54 | binbash_ | but it would be great if those numbers could also recieve fax |
| 08:48.09 | *** join/#asterisk joobie (~joobie@CPE-124-180-193-8.lns7.lon.bigpond.net.au) |
| 08:48.11 | binbash_ | so that it would detect that's a fax, and then e-mail it to the e-mail address that goes with the number |
| 08:48.26 | binbash_ | or am i on a mission impossible :D? |
| 08:49.20 | joobie | hey guys.. i have a monitoring system that i want to integrate to asterisk to track how many DAHDI channels are in use.. anyone got a good idea on how to do this? |
| 08:49.52 | joobie | i can do passive checks, where every X mins my monitoring system runs a cmd in asterisk and stores the value (this can grab the "current used lines") .. but it runs at a set frequency only (like every 30 seconds) |
| 08:50.22 | joobie | alternatively i can do active checks, where something connects to my monitoring box and pushes a value |
| 08:51.12 | joobie | is there a way in asterisk i can run a script every time i get a call on dahdi? i wanted ot do it in a way that ran as an independant thread to the call being handled |
| 08:51.23 | joobie | so if there was a delay in the script or sumthen, it wouldnt impact the call in any way |
| 08:51.49 | joobie | putting it in the dialplan i was thinking would hold up the call progression if there was a delay with the script running |
| 08:51.57 | WIMPy | joobie: You could listen on AMI. |
| 08:54.31 | joobie | hmm |
| 08:54.32 | joobie | good idea |
| 08:54.41 | joobie | thanks WIMPy |
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| 09:22.06 | joobie | hmm |
| 09:22.10 | joobie | trying to manually go through the AMi |
| 09:22.13 | joobie | via telnet |
| 09:22.22 | joobie | im putting \r\n at the end of each command, cant seem to login tho |
| 09:22.26 | joobie | is that right for the linefeed? |
| 09:23.19 | WIMPy | IIRC: In theory yes, but in practice it doesn't matter. |
| 09:25.16 | kaldemar | joobie: just press enter twice. |
| 09:28.13 | joobie | tried that |
| 09:28.34 | joobie | ok wak |
| 09:28.36 | joobie | it's working now |
| 09:28.49 | joobie | odd |
| 09:28.53 | joobie | maybe user error :P |
| 09:28.58 | joobie | so hmm |
| 09:29.08 | joobie | to monitor each time a dahdi call comes in |
| 09:29.14 | joobie | wats a good cmd |
| 09:29.19 | joobie | and out |
| 09:30.05 | joobie | like i guess when i connect to the AMI, i want to be able to see when a DADHI call comes in and ends.. and likewise a sip call |
| 09:30.57 | WIMPy | From some distant location im Memory, I'd say newchannel and hangup. |
| 09:33.45 | WIMPy | There might have been a catch however, where channels get renamed. |
| 09:34.28 | WIMPy | Or was that only the ID? |
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| 10:25.16 | AlanW | Good day. I having some _fun_ with 1.6. I moved from 1.4 and the extensions.conf is not working. I want to strip right down removing all the demo/crude from the default one. |
| 10:25.59 | AlanW | i have: exten => _0.,1,Dial(IAX2/------etc----) |
| 10:26.27 | AlanW | but when i attempt to dial anything with 0, asterisk takes over after 3digits. i can't find who is hijacking that dialplan rule |
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| 11:16.18 | joobie | guys is tehre an * AMI command that I can run to see the total number of DAHDI channels in use? or even llist the channels in use? |
| 11:17.06 | joobie | i have 1.4 btw |
| 11:17.11 | joobie | noticed 1.6 has these inbuilt :/ |
| 11:19.25 | kaldemar | joobie: "Action: Command" lets you use CLI commands. http://www.voip-info.org/wiki/view/Asterisk+Manager+API+Action+Command |
| 11:21.35 | joobie | kaldemar, ya, but not sure which cmd in there i can use |
| 11:22.10 | joobie | dahdi show channels vaguely does it.. but the only way to tell that the channel is in use is to check if it has an extension specified.. which im not sure is the best way |
| 11:24.24 | leifmadsen | joobie: if 1.4 doesn't have the ability to do that, you could use GROUP() and GROUP_COUNT() in your dialplan to track the number of calls |
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| 11:33.18 | manxpower | AlanW: read the UPGRADE*.txt files to learn about the various changes |
| 11:33.36 | AlanW | thank you |
| 11:35.10 | manxpower | If you are using SIP phones then the phone has it's own dialplan |
| 11:37.30 | AlanW | mmm its the polycom handset, and it was working fine under 1.4, now its not letting me dial more than 015 before it cuts off. so i am thinking 1.6 is the reason here. |
| 11:37.41 | manxpower | unlikely |
| 11:38.26 | manxpower | remember each digit is not sent to asterisk as you dial it. The phone collects digits and when it thinks it has enough, it send them all to Asterisk at once. |
| 11:40.04 | AlanW | well i _want_ to believe you. but all the handsets are doing this, cisco/polycom, and they weren't doing this under 1.4 |
| 11:40.14 | AlanW | so extensions.conf is playing a factor here i am sure. |
| 11:40.42 | *** join/#asterisk joako_ (~joako@opensuse/member/joak0) |
| 11:41.40 | AlanW | also since moving 1.6, it seems to be way more sensitive to NAT phones; as my handsets are showing UNREACHABLE status way more than they use to. |
| 11:45.30 | joobie | leifmadsen, do i have to unset group() for a channel when it hangs up or is that automatic? |
| 11:45.47 | leifmadsen | joobie: automatic |
| 11:46.09 | manxpower | joobie: "core show function GROUP" does not say? |
| 11:46.26 | joobie | leifmadsen, can group_count() be pulled from the asterisk console? |
| 11:46.34 | joobie | manxpower, duno |
| 11:46.37 | leifmadsen | joobie: not to my knowledge |
| 11:46.44 | joobie | doh.. that could be a prob |
| 11:46.57 | manxpower | joobie: you might want to read the docs for the functions/applications you are using. |
| 11:47.27 | *** join/#asterisk zomenox (~brock@254120.firewall.winthrop.edu) |
| 11:47.43 | joobie | im thinking of using the AMI to monitor the channels in realtime, so i can instantly update my monitoring system when it sees a new call / sees a call hangup.. but i need an additional check to run say every 5 mins, just to confirm the channels are in sync.. like every 5 mins, "how many channels are actually in use" type thing, to ensure i dont fall out of sync |
| 11:48.15 | joobie | manxpower, that's an interesting idea |
| 11:48.36 | manxpower | joobie: radical, I know. |
| 11:48.46 | joobie | :) |
| 11:48.52 | joobie | manxpower, word has it you are the SMS guru |
| 11:49.23 | joobie | leifmadsen, btw thanks for the knowledge on group*() |
| 11:50.41 | manxpower | joobie: I am not an SMS guru. I just know the answers to the most common problem that people have with SMS and Asterisk. |
| 11:51.11 | joobie | manxpower, i was looking into integrating SMS into asterisk the other day.. needed a way to get an SMS into asterisk. The only decent way I could find was to hook up a PSTN line that supports SMS (managed to find 1 AU provider that does this) |
| 11:51.12 | manxpower | That problem is "app_sms is not working". |
| 11:51.18 | joobie | is this the only way to suck in an SMS into * ? |
| 11:51.59 | manxpower | joobie: no, but it is the only way app_sms supports SMS i.e. over PSTN lines. |
| 11:52.10 | joobie | ahh |
| 11:52.22 | joobie | but app_sms doesnt even work over PSTN lines? |
| 11:52.25 | manxpower | if yo want SMS some other way, there are a zillion and 25 sites out there with web interfaces for sending SMS. |
| 11:52.34 | joobie | yea |
| 11:52.38 | manxpower | joobie: no, app_sms ONLY WORK S ON PSTN LINES |
| 11:52.39 | joobie | the sending part is fine |
| 11:52.43 | joobie | the receiving is a problem |
| 11:52.47 | joobie | ahh |
| 11:53.08 | joobie | does app_sms work for sending SMS over PSTN also? |
| 11:53.52 | manxpower | people seem to think app_sms is some magical SMS thing. It is not. It simply sends and receives SMS using FSK (think 1200 baud modem) over PSTN lines. The fact that there are NO PSTN SMS providers in the USA seems to confuse many people. |
| 11:54.14 | manxpower | joobie: app_sms ONLY supports SMS over PSTN. It supports no other type of SMS |
| 11:54.55 | joobie | http://www.telstra.com.au/homephone/features_services/talking-text.html# |
| 11:55.03 | manxpower | they really should rename app_sms to some thing like app_esti4582 or something like that. |
| 11:55.03 | joobie | that is the product i was looking at integrating to |
| 11:55.56 | manxpower | joobie: does the service use PSTN lines? |
| 11:56.23 | joobie | nod |
| 11:56.50 | joobie | it's a little limited though - it will only work between telstra products |
| 11:56.58 | joobie | ie. you can't sms across to another carrier |
| 11:58.04 | manxpower | if it uses SMS (real SMS, not some marketing product for text messaging) over PSTN lines then it should work with app_sms |
| 11:58.07 | *** join/#asterisk ruben23 (~ITadmin@122.55.48.243) |
| 11:58.18 | joobie | what is the protocol |
| 11:58.25 | joobie | like is there a formal standard |
| 11:58.32 | manxpower | standby |
| 11:58.41 | joobie | knowing these ass bandits |
| 11:58.55 | joobie | it will use a modified version |
| 11:59.16 | manxpower | ETSI ES 201 912 |
| 11:59.25 | manxpower | As documented in "core show application SMS" |
| 11:59.30 | joobie | ahh .. good old ETSI ES 201 912 |
| 11:59.36 | joobie | hey |
| 11:59.43 | joobie | that core show application thingie came handy again |
| 11:59.47 | joobie | 2 times in a row |
| 11:59.53 | joobie | you might be onto something manxpower |
| 12:00.01 | manxpower | joobie: it is the most useful command in asterisk. start using it. |
| 12:01.02 | joobie | manxpower, so what have you used app_sms for before? what type of setup |
| 12:01.17 | manxpower | joobie: I live in the USA. I have never used app_SMS. |
| 12:01.22 | manxpower | ~manxpower |
| 12:01.23 | infobot | ManxPower has been using Asterisk in production since late 2001. Currently works at InterGlobe Communications, a CLEC based in NYC with service in NY, NJ, FL, and TX. http://www.nyigc.com |
| 12:01.28 | manxpower | But I've been around long enough to know this stuff |
| 12:01.54 | joobie | .. "Telstra use ETSI standard ES 201 912 for their fixed line sms product, |
| 12:01.55 | joobie | which is what most fixed line sms implementations use in other |
| 12:01.55 | joobie | countries. |
| 12:01.55 | joobie | " |
| 12:02.43 | *** join/#asterisk rttrey (~trey@andc-office-fw.atlantic.net) |
| 12:02.44 | manxpower | most people seem to think app_sms is for MOBILE SMS |
| 12:04.24 | joobie | fuk knos how u would try and hook up a mobile telephone to asterisk |
| 12:04.50 | manxpower | "very poorly" is usually the answer to that |
| 12:05.31 | joobie | werd |
| 12:06.35 | joobie | these ass bandits at telstra are the only ones in AU with the product |
| 12:06.41 | joobie | but they resitrct it so it only works on their network |
| 12:06.47 | joobie | sucks, |
| 12:07.06 | manxpower | many providers have a public SMSC |
| 12:09.10 | joobie | i dont know too much about SMSC's |
| 12:09.15 | joobie | except that it was a number in my phone |
| 12:09.19 | joobie | under SMS settings |
| 12:09.21 | joobie | on my old nokia |
| 12:09.24 | joobie | about 10 years ago |
| 12:10.55 | joobie | presuming it's the router for the SMS's though |
| 12:11.20 | joobie | telstra will probably restrict the DST of the numbers to be their mobile numbers |
| 12:11.27 | joobie | unless it's from a telstra mobile number |
| 12:12.10 | *** part/#asterisk rttrey (~trey@andc-office-fw.atlantic.net) |
| 12:15.36 | *** join/#asterisk [TK]D-Fender (~chatzilla@216.191.106.163) |
| 12:16.35 | joobie | hello [TK]D-Fender |
| 12:17.20 | *** join/#asterisk jaytee (~jforde051@unaffiliated/jaytee) |
| 12:17.36 | *** join/#asterisk kruemeltee (~Maddin@port-92-198-62-82.static.qsc.de) |
| 12:18.54 | kruemeltee | hello everybody :-) |
| 12:19.03 | jaytee | hi |
| 12:20.36 | joobie | night guys |
| 12:23.40 | *** join/#asterisk luckyaba (~lucky@ip72-194-218-169.sb.sd.cox.net) |
| 12:24.13 | *** join/#asterisk soman (~somnath@stargate.starnet.fi) |
| 12:27.47 | *** join/#asterisk ralonso (~a@76.Red-81-43-207.staticIP.rima-tde.net) |
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| 12:48.49 | *** join/#asterisk joako (~joako@opensuse/member/joak0) |
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| 13:06.59 | *** join/#asterisk muiro (~muiro@unaffiliated/muiro) |
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| 13:11.43 | Katty | hi |
| 13:12.21 | jaytee | hi Katty |
| 13:12.56 | *** join/#asterisk rdircio (~admin@189.242.22.188) |
| 13:17.30 | *** join/#asterisk Deeewayne (~dwayne@75-150-14-49-Atlanta.hfc.comcastbusiness.net) |
| 13:17.30 | *** mode/#asterisk [+o Deeewayne] by ChanServ |
| 13:18.10 | elzid | guys - anyone know how to convert ices2 ogg output to mp3 on the fly? |
| 13:19.29 | *** join/#asterisk utahsaint (~utahsaint@mail.ntegratedsolutions.com) |
| 13:21.00 | ruben23 | hi |
| 13:25.23 | *** join/#asterisk davido1 (~davido1@p54B0A898.dip0.t-ipconnect.de) |
| 13:25.42 | davido1 | hello room |
| 13:26.44 | *** join/#asterisk gr0mit (~tim@extrt.txrx.org.uk) |
| 13:27.29 | davido1 | I'd like to send a custom notify message to my sip peers... is that possible from asterisk? |
| 13:28.25 | pabelanger | Anybody know if the Persian language is supported? |
| 13:28.46 | pabelanger | or something similar in grammar syntax |
| 13:32.04 | kaldemar | davido1: "help sip notify" in CLI. sample sip_nofity.conf has some message types. |
| 13:32.08 | tzafrir_laptop | pabelanger, in say.conf? say.c ? |
| 13:32.19 | pabelanger | tzafrir_laptop: yar! |
| 13:32.33 | tzafrir_laptop | Which of the two? (IIRC: not in either) |
| 13:32.53 | tzafrir_laptop | Anyway, look in say.c |
| 13:33.37 | davido1 | kaldemar: Yeah, I've used that before, but now I want to send it directly from the dialplan... What I want, actually, is to display some text in a peer without having to make a call... |
| 13:33.38 | pabelanger | tzafrir_laptop: ya, currently check, figured I'm ping the channel |
| 13:34.11 | manxpower | davido1: sounds like you are setting yourself up for failure |
| 13:34.29 | kaldemar | davido1: i've used sipsak for that. calling cli commands from dialplan is a bit ugly. |
| 13:34.39 | davido1 | manxpower: i know :(... hahaha, but it's not me who asked for it :p |
| 13:36.14 | davido1 | kaldemar: yes, I've usesd sipsak for that too... but i wanted to know if there's an alternative. Something a bit prettier than s => { System(run-command);} to execute a script that would use sipsak to send the text... |
| 13:36.56 | manxpower | davido1: any thing you do for this will be a hack. |
| 13:37.10 | davido1 | manxpower: Maybe I can display text in another way apart from sending a notify? |
| 13:37.34 | davido1 | manxpower: SipAddHeaders wouldn't help or? |
| 13:37.45 | *** join/#asterisk Naikrovek (~jjohnson@unaffiliated/naikrovek) |
| 13:38.42 | davido1 | manxpower: Ah, but then I would have to call the peer... Hmm... No good |
| 13:38.46 | davido1 | Oh well |
| 13:38.50 | *** join/#asterisk putnopvut (~putnopvut@asterisk/master-of-queues/mmichelson) |
| 13:38.55 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 13:39.54 | *** join/#asterisk anthm (~anthm@freeswitch/developer/anthm) |
| 13:40.27 | manxpower | davido1: um, that is TOTALLY dependent on the PHONE. |
| 13:41.23 | elzid | ices anyone? how to convert ogg stream into mp3 on the fly for live traffic? |
| 13:41.35 | davido1 | manxpower: But, independently of the phone, if I want to send something to it without calling, I have to use notify, is it? |
| 13:41.36 | ruben23 | i got sample number 02828260357 where this needs to be added with 44(area code) and strip the 0 value which it will be 442828260357 at the final stage of dialing.. |
| 13:41.58 | ruben23 | how do i setup the exten sion of that |
| 13:43.32 | manxpower | davido1: um, that is TOTALLY dependent on the PHONE. |
| 13:44.28 | davido1 | manxpower: Okok... hehe... What do you know about Snom phones? |
| 13:44.53 | manxpower | davido1: no. |
| 13:44.57 | manxpower | ~answers |
| 13:44.58 | infobot | rumour has it, answers is Asterisk Book: http://astbook.asteriskdocs.org & SIP w/ NAT: http://www.aocomputing.net/?p=3 & Wiki: http://www.voip-info.org & docs: /path/to/src/asterisk/doc & 1.4 .vs. 1.6.0 .vs. 1.6.1: : /path/to/src/asterisk/UPGRADE*.txt |
| 13:45.02 | manxpower | I bet there's info in the web |
| 13:45.26 | manxpower | 44${EXTEN:1} |
| 13:45.51 | davido1 | manxpower: Ok, thanks anyway =) |
| 13:46.23 | ruben23 | manxpower: any sample extensions patter |
| 13:46.39 | *** join/#asterisk ktwilight[m] (~ktwilight@91.179.82.183) |
| 13:47.07 | *** join/#asterisk freezey (~trees@static-64-61-84-174.isp.broadviewnet.net) |
| 13:47.14 | freezey | so i have a conf set with meetme and internally it works how would i set it to be able to be dialed from outside? it comes into my phone system but says not in service |
| 13:47.24 | ruben23 | i get this work------->exten => _44.,1,Dial(SIP/${EXTEN}@<carrier ip>) |
| 13:48.34 | ruben23 | <PROTECTED> |
| 13:49.16 | manxpower | ruben23: I don't see anything that could hold the number you provided. |
| 13:49.34 | manxpower | it won't be EXTEN, since your pattern would not match 02828260357 |
| 13:49.59 | manxpower | maybe you need something like exten => _0.,1,Goto( 44${EXTEN:1},1) |
| 13:50.49 | ruben23 | manxpower: i just need to remove 0 then i can add up prefix 44 + number without zero then im ok.. |
| 13:51.00 | carrar | manxpower, it's like you've read the book or something |
| 13:51.01 | *** join/#asterisk grEvenX (~even@apb9hb.ip.ssc.net) |
| 13:51.10 | carrar | :) |
| 13:51.23 | carrar | How do you know this mojo |
| 13:51.33 | manxpower | carrar: more like read channelvariables.txt before they removed the .txt version and replaced it with a .tex file. |
| 13:51.44 | manxpower | that's where you learn this stuff. |
| 13:51.45 | carrar | heh |
| 13:51.51 | jaytee | i hate the tex version |
| 13:52.13 | manxpower | jaytee: I rant about the conversion to .tex about once a month. Never does any good, but it does make me feel better. |
| 13:52.34 | *** part/#asterisk Slashman (~Slash@ariane.fimasys.com) |
| 13:52.36 | jaytee | manxpower, yeah I bitched a couple times and then resigned myself to it |
| 13:52.40 | carrar | formulate a EMail and we'll all send it once a month |
| 13:52.50 | jaytee | and I've really got nothing against people with latex fetishes :-) |
| 13:53.03 | carrar | depends who is in the laex |
| 13:53.06 | carrar | latex |
| 13:53.07 | jaytee | hehe |
| 13:53.49 | carrar | Perhaps alison |
| 13:53.55 | carrar | can make audio versions of the txt files |
| 13:54.11 | manxpower | jaytee: but the info is in the asterisk.pdf! |
| 13:56.58 | *** join/#asterisk photographe (~nick656@173.176.84.78) |
| 13:57.43 | jaytee | manxpower, thanks for pointing that out! I didn't realize that was in the pdf. |
| 13:57.50 | jaytee | just started browsing it. |
| 13:57.55 | freezey | anybody have any idea why i cant dial my conference bridge from the outside? if i won that DID block shoulnt it append my NPA-NXX- to the extension? |
| 13:58.23 | leifmadsen | for those who want TeX to text, I found the steps easily with google |
| 13:59.00 | leifmadsen | cd /usr/src/asterisk/docs/tex ; latex asterisk.tex ; catdvi -e 1 -U asterisk.dvi | sed -re "s/\[U\+2022\]/*/g" | sed -re "s/([^^[:space:]])\s+/\1 /g" > asterisk-docs.txt |
| 13:59.05 | leifmadsen | less asterisk-docs.txt |
| 13:59.06 | leifmadsen | done |
| 13:59.20 | manxpower | freezey: you should ask on #freepbx |
| 13:59.21 | carrar | assuming you have catdvi |
| 13:59.30 | leifmadsen | carrar: because apt-get install catdvi was hard? |
| 13:59.36 | carrar | yes |
| 13:59.39 | manxpower | leifmadsen: I need to download 105MB of RPMs first |
| 13:59.49 | leifmadsen | apparently I'm not on ignore anymore |
| 14:00.04 | jaytee | i would never ignore you |
| 14:00.07 | carrar | heh |
| 14:00.08 | manxpower | leifmadsen: I'm on my personal account using Pidgin, which does not have a /ignore. |
| 14:00.19 | manxpower | Also you've not been coming in, bitching about us and then leaving. |
| 14:00.58 | manxpower | I am glad that the command to convert from tex to txt is so easy. I'm sure all the n00bs already know that command. |
| 14:01.08 | leifmadsen | I didn't know about that either |
| 14:01.12 | leifmadsen | I just googled and it was the 2nd link |
| 14:01.16 | leifmadsen | "convert tex to text" |
| 14:01.28 | leifmadsen | or "convert tex to ascii" |
| 14:01.41 | manxpower | My problem with .tex files is not that *I* have problems reading them. It's that the poor n00bs that can barely type "less file.txt" have problems |
| 14:02.12 | manxpower | we need to lower the barriers to reading the docs, not raise them. |
| 14:02.20 | leifmadsen | mkrelease is the script we use to generate our releases, and it is available publicly. Anyone is welcome to submit a simple patch to add the converted text files. |
| 14:03.03 | manxpower | leifmadsen: thank you. I will keep that in mind and might even post a bounty for it. |
| 14:03.36 | manxpower | is such a patch likely to be accepted? |
| 14:03.40 | tzafrir_laptop | leifmadsen, that still requires the tex stuff installed |
| 14:03.49 | tzafrir_laptop | (for running latex and catdvi) |
| 14:03.57 | leifmadsen | tzafrir_laptop: welcome to 3 minutes ago |
| 14:04.03 | manxpower | tzafrir_laptop: if I understand it correctly, it would only require it on the digium machines that build the releases? |
| 14:04.19 | leifmadsen | I already have that stuff installed on the machine I use to make releases |
| 14:04.20 | *** join/#asterisk jmacz (~jmacz@190.144.75.22) |
| 14:04.27 | manxpower | and I don't care if Digium has to install 100MB of stuff just to do something. |
| 14:05.04 | tzafrir_laptop | well, the machine building proper packages has to have tex anyway, to build asterisk.pdf |
| 14:05.09 | leifmadsen | I would even install 1GB of stuff is necessary |
| 14:05.11 | manxpower | jaytee: one of my other pet peeves is that Digium isn't confident enough in their own software to run the latest release on their corporate PBX. |
| 14:05.12 | leifmadsen | it's just disk space |
| 14:05.29 | leifmadsen | manxpower: digium runs switchvox |
| 14:05.52 | manxpower | <PROTECTED> |
| 14:06.03 | manxpower | better? |
| 14:06.08 | leifmadsen | whatever |
| 14:06.09 | jaytee | what's that expression about eating the dogfood that's used in software development? |
| 14:06.09 | tzafrir_laptop | leifmadsen, if you build in a chroot builder (do you?) , extra build dependencies mean extra build time, eventually |
| 14:06.22 | manxpower | jaytee: I think they are all cat people. 8-) |
| 14:06.32 | leifmadsen | tzafrir_laptop: I build in a virtual machine dedicated to building releases. That's all it does. |
| 14:06.45 | manxpower | jaytee: anything I deploy to customers is first tested on our own corporate people. |
| 14:06.49 | tzafrir_laptop | manxpower, so the cats are eating the dogfood? |
| 14:07.07 | leifmadsen | tzafrir_laptop: I don't build Asterisk on this machine, I simply use it for packaging the source code into .tar.gz files |
| 14:07.07 | manxpower | tzafrir_laptop: no, that's the point. nobody is eating the dog food except the customers. |
| 14:07.35 | manxpower | ok, not customers, but end users |
| 14:07.37 | jaytee | it was something Microsoft changed in their policies after the Vista embarrassment where they in most areas run the latest test versions to try to catch more issues before release |
| 14:07.52 | jaytee | something like "eating your own dogfood" |
| 14:08.09 | manxpower | jaytee: a radical shift in software development 8-) |
| 14:08.12 | tzafrir_laptop | jaytee, huh? "eating their own dogfood" appears in articles about MS from way before Vista |
| 14:08.50 | jaytee | yeah, I just googled. I'm seeing stuff from 1988 |
| 14:09.05 | jaytee | obviously they should get out of the dogfood business :-) |
| 14:09.43 | leifmadsen | manxpower: actually the change would go in build_tools/prep_tarball of the asterisk source |
| 14:11.06 | jaytee | anyone care to give a "guesstimate" of the number of Asterisk servers in production in the US and Canada only? and worldwide? |
| 14:12.06 | leifmadsen | 100 hundred thousand million |
| 14:12.20 | Chainsaw | One billion doll... eh, machines. |
| 14:13.08 | jaytee | ok, so lebenty-leven it is! |
| 14:16.03 | elzid | sorry guys - this isn't a repeat question - this is more to do with the * Ices() function than the linux ices package: there's an * cmd called Ices(), it outputs an ogg format audio stream and I need to convert it to mp3 on the fly - can I just run a System call and execute a batch ices command with pipes through lame etc? My assumption here is that Ices() in * effectively calls ices bin on the OS. Any ideas anyone |
| 14:16.16 | [TK]D-Fender | that's 10 million. |
| 14:16.20 | [TK]D-Fender | Math FAIL |
| 14:17.40 | [TK]D-Fender | (10 million million that is) |
| 14:18.10 | [TK]D-Fender | wonders if there is a fixed name for that... |
| 14:19.27 | Naikrovek | quadrillion? |
| 14:19.56 | Naikrovek | 10 trillion |
| 14:20.01 | Naikrovek | my math is not to be trusted, btw |
| 14:25.42 | Woody2143 | Hey all, I'm looking for a point in the right direction. I want to add a param to the SIP From header. I see the SIPAddHeader for adding headers but is there a function for manipulating existing headers? |
| 14:27.04 | *** join/#asterisk rgsteele (~rgsteele@207.106.239.81) |
| 14:27.32 | [TK]D-Fender | Woody2143: No |
| 14:28.30 | Woody2143 | I figured as much, I'd been searching around for awhile. Thank you. |
| 14:29.41 | *** join/#asterisk cesar_CR (~cesar@190.10.115.176) |
| 14:33.00 | manxpower | Woody2143: What *specific* thing do you want to do to the From headeR? |
| 14:33.35 | manxpower | there is no general way to modify existing headers, but there are config options for specific things like the From domain, etc. |
| 14:34.10 | Woody2143 | add 'pstn-params:80'; eg From: <sip:+12015550000@192.168.0.1:5060;pstn-params=80> |
| 14:34.53 | manxpower | Woody2143: That's not going to happen. 8-| |
| 14:35.00 | Woody2143 | :) |
| 14:35.02 | Woody2143 | No worries |
| 14:35.43 | manxpower | Woody2143: header modification and stuff like that is the function of a SIP Proxy, which Asterisk is not. You could easily do it with something like SER/OpenSIPPS/OpenSER or whatever they changed the name to this week |
| 14:36.26 | *** join/#asterisk FreezeS (~KVIrc@82.208.157.125) |
| 14:36.42 | FreezeS | hi guys |
| 14:36.54 | Woody2143 | Cool, thanks for the information max |
| 14:37.08 | FreezeS | I need to know the codecs of a call |
| 14:37.28 | FreezeS | the codec that comes in, the one that goes out and whether it was transcoded or not |
| 14:37.45 | *** join/#asterisk The_Boy_Wonder (~vossel@asterisk/batman-developer/dvossel) |
| 14:38.12 | FreezeS | right now I can get the codec that comes in and if it's transformed into SLIN or not |
| 14:38.27 | FreezeS | but I can't get the codec that goes out of asterisk |
| 14:39.50 | *** join/#asterisk moy (~chatzilla@bas1-unionville55-1177733627.dsl.bell.ca) |
| 14:40.14 | manxpower | "sip show channels" |
| 14:40.42 | FreezeS | manxpower: I need that in an agi script ran in deadagi |
| 14:40.48 | FreezeS | sorry, I should have mentioned that |
| 14:41.02 | davido1 | see you guys |
| 14:41.03 | *** part/#asterisk davido1 (~davido1@p54B0A898.dip0.t-ipconnect.de) |
| 14:41.25 | manxpower | FreezeS: read channelvariables.tex There should some SIP related variables that may have that info. I doubt you will be able to do this. |
| 14:41.42 | *** join/#asterisk codefreeze-lap (~murf@mail.parsetree.com) |
| 14:42.52 | sbrath | I moved all my SIP devices from users.conf to sip.conf and now I have 1 extension that's getting a time limit. Is their some sort of default timelimit of like 20 minutes? Can it be defined somewhere other than the Dial() ? |
| 14:43.13 | manxpower | sbrath: no. |
| 14:43.37 | manxpower | it can be set with the TIMEOUT stuff, but since you wrote the dialplan you should know. |
| 14:43.40 | sbrath | Wierd, at about 13 minutes she says she gets a "Beep" and then at 18 minutes it cuts the channel. |
| 14:43.47 | sbrath | I guess is could be the ATA |
| 14:43.52 | manxpower | sbrath: pastebin the problem call |
| 14:44.08 | *** join/#asterisk Linuturk (~linuturk@unaffiliated/linuturk) |
| 14:44.51 | FreezeS | manxpower: right now I'm using CHANNEL(audioreadformat) but this has only the data for the A channel |
| 14:46.54 | [TK]D-Fender | FreezeS: Go make an AGI that will use AMI to poll the channels for this info |
| 14:47.25 | FreezeS | [TK]D-Fender: that's exactly what I was trying to avoid |
| 14:47.51 | *** join/#asterisk rare1980_ (~rare1980@12.25.228.67) |
| 14:47.52 | *** join/#asterisk KavanS (~KavanS@static-173-50-141-22.ptldor.fios.verizon.net) |
| 14:49.19 | *** join/#asterisk mbranca (~matteo@host139-217-static.224-95-b.business.telecomitalia.it) |
| 14:50.01 | sbrath | http://pastebin.com/ZrQfrtGt |
| 14:51.51 | [TK]D-Fender | FreezeS: Actually... if you're in the dialplan.. there is only an "A" channel |
| 14:52.06 | manxpower | sbrath: it does not look like a dialplan / config issue. |
| 14:52.25 | FreezeS | [TK]D-Fender: thanks |
| 14:52.39 | [TK]D-Fender | sbrath: Try again with SIP DEBUG enabled for that peer so we can see who initiates the hangup, and for what reason. |
| 14:54.59 | sbrath | good idea. |
| 14:56.30 | *** join/#asterisk xAvia (~matt@cpe-69-204-26-134.buffalo.res.rr.com) |
| 14:56.59 | *** join/#asterisk jpeeler (~jpeeler@asterisk/digium-software-dev/jpeeler) |
| 14:57.09 | pentanol | bmoraca_work you're even use func_odbc? it working for you properly? on which asterisk version? presumably you can argue this behaviour? http://codepad.org/GbrN0BGj |
| 14:59.05 | [TK]D-Fender | pentanol: Going on well over a week and I have yet to see the proof I asked for |
| 15:00.07 | *** join/#asterisk SaintJean (~xyz@64.229.75.107) |
| 15:00.08 | pentanol | I see, but I still can't evaluate it |
| 15:00.25 | [TK]D-Fender | pentanol: Show us the isql backup. |
| 15:00.56 | pentanol | db is clear |
| 15:01.14 | SaintJean | Hi, is the variable ${DIALEDPEERNUMBER} still broken? |
| 15:01.48 | pentanol | in first it should check out entries and then add something news |
| 15:04.34 | *** join/#asterisk Polysics (~Luca@95.237.66.55) |
| 15:05.17 | Polysics | hello |
| 15:07.07 | bmoraca_work | wow, someone asking me a question completely out of the blue! |
| 15:07.26 | pentanol | hm? |
| 15:07.29 | bmoraca_work | i haven't been in this channel in 15 hours, lol |
| 15:07.45 | pentanol | you're don't like that colour? |
| 15:08.02 | bmoraca_work | pentanol: does your query execute outside of asterisk's ODBC connector? |
| 15:08.14 | pentanol | of course |
| 15:08.20 | [TK]D-Fender | bmoraca_work: I've asked for this repeatedly. I never got it |
| 15:08.37 | bmoraca_work | pentanol: interesting, because your DISTINCT syntax is INCORRECT. |
| 15:08.51 | bmoraca_work | pentanol: http://dev.mysql.com/doc/refman/5.1/en/select.html |
| 15:09.08 | pentanol | Unable to execute query [SELECT t_dst FROM av_trunks limit 1]\n".. |
| 15:09.28 | pentanol | holy crap |
| 15:09.45 | bmoraca_work | you've got other problems then. |
| 15:10.05 | pentanol | readsql=SELECT t_dst FROM av_trunks limit 1 |
| 15:10.07 | *** join/#asterisk southtel_ (~slester@68-114-19-101.dhcp.gwnt.ga.charter.com) |
| 15:10.48 | southtel_ | Hey everyone. |
| 15:10.48 | bent_screwdriver | how to add an extra field to the CDR for mysql storage? i created the field recorded in the cdr table, in mysql, and put this in the dialplan: exten => s,n,Set(CDR(recorded)=1) (ast 1.6.2) |
| 15:11.16 | leifmadsen | bent_screwdriver: you're using cdr_adaptive_odbc ? |
| 15:11.30 | pentanol | bmoraca_work could you show your func_odbc.conf.... where readsql... |
| 15:11.55 | bent_screwdriver | @leifmadsen: i'm not sure |
| 15:12.36 | leifmadsen | me either |
| 15:13.12 | bent_screwdriver | @leifmadsen: how could i find out. i just yum'ed asterisk and asterisk addons and created the cdr table |
| 15:13.19 | carrar | booleans are for wusses ;) |
| 15:13.21 | leifmadsen | then you probably don't |
| 15:13.31 | leifmadsen | ls /usr/lib/asterisk/modules/cdr_adaptive_odbc.* |
| 15:13.31 | bmoraca_work | pentanol: not sure what good it will do |
| 15:13.32 | *** join/#asterisk outtolunc (~me@c-67-160-192-210.hsd1.ca.comcast.net) |
| 15:14.20 | pentanol | on which * version? |
| 15:14.26 | pentanol | odbc version also... |
| 15:14.39 | bmoraca_work | 1.6.2.0, but i've used it on 1.4 as well without issue |
| 15:14.45 | pentanol | where did you get this odbc? |
| 15:14.46 | southtel_ | Has anyone out there dealt with heavy static on a PRI line? |
| 15:15.10 | *** join/#asterisk smooth_penguin (~smoove@59.95.21.228) |
| 15:15.34 | leifmadsen | static on a PRI? seems odd ;) |
| 15:15.38 | leifmadsen | it'd a digital circuit |
| 15:15.41 | leifmadsen | it's*( |
| 15:16.17 | *** join/#asterisk southtel_ (~slester@68-114-19-101.dhcp.gwnt.ga.charter.com) |
| 15:16.39 | bent_screwdriver | @leifmadsen: no such file or directory so i guess not. do i need to add the custom field to cdr_custom.conf? |
| 15:16.54 | *** join/#asterisk atis_work (~atis_work@193.238.212.171) |
| 15:17.03 | leifmadsen | not sure -- it probably isn't going to do what you expect then |
| 15:17.10 | leifmadsen | cdr_custom is for writing to a CSV file |
| 15:17.14 | pentanol | what from you've installed these odbc drivers? |
| 15:17.15 | southtel_ | I've got an older PBX connected to a PRI, and on the majority of outbound calls, there's heavy static, but only on the external end. |
| 15:17.43 | pentanol | bmoraca_work poke |
| 15:21.15 | bmoraca_work | pentanol: "yum install unixODBC mysql-odbc-devel" |
| 15:21.20 | pentanol | great |
| 15:21.29 | bmoraca_work | er |
| 15:21.36 | bmoraca_work | mysql-connector-odbc or something |
| 15:21.50 | *** join/#asterisk corretico (~laguilar@201.201.46.106) |
| 15:23.55 | beefpastry | saw an old post that said alsa support doesn't work...is that still the case? |
| 15:24.43 | [TK]D-Fender | beefpastry: with what? |
| 15:25.02 | idespinner | southtel_, my guess is that its possibly a bad card |
| 15:25.11 | *** join/#asterisk dandre (~daniel@ble59-2-81-56-122-47.fbx.proxad.net) |
| 15:25.15 | beefpastry | trying to rig an overhead pager to see if it works better than paging through my polycoms |
| 15:25.46 | beefpastry | the echo in the main office from the polycoms is a little distracting |
| 15:26.24 | [TK]D-Fender | beefpastry: have you considered trying it? |
| 15:26.25 | *** join/#asterisk xpot (~james@70-91-210-233-BusName-Utah.hfc.comcastbusiness.net) |
| 15:27.22 | beefpastry | working on it...tried dsp because of the post, but was getting busy when dialing |
| 15:27.45 | p3nguin | What about using record and playback instead of paging real time? |
| 15:28.35 | leifmadsen | beefpastry: when you do the paging do you mute the phones too? I think you can do that... |
| 15:29.36 | Faustov | is there any simple way to send SMS from asterisk CLI? |
| 15:29.40 | southtel_ | idespinner, interesting thought...any experience with that? A card going partially bad like that? |
| 15:29.59 | Faustov | is there any simple way to send SMS from asterisk CLI through a SIP provider? |
| 15:30.44 | leifmadsen | no. |
| 15:30.49 | [TK]D-Fender | Faustov: No. Asterisk is NOT an SMS platform |
| 15:30.57 | beefpastry | I overrode the default (freepbx) paging dialplan to not try phones in use...it's more an office design issue (and perhaps a networking issue)...there is a slight delay from the pager to the pagees so the echo comes from hearing the real voice before the page comes through |
| 15:31.16 | leifmadsen | beefpastry: ahh --- use earplugs :) |
| 15:31.31 | beefpastry | My office is in a different area...not my problem ;) |
| 15:31.47 | p3nguin | Or use the phone like a hand mic instead of a phone (with the ear piece on your ear). |
| 15:32.01 | beefpastry | But the secretaries complaining is probably a worse annoyance. |
| 15:32.06 | *** join/#asterisk Da-Geek (~Da-Geek@80.235.230.186) |
| 15:32.06 | *** join/#asterisk Akiraa (~Akiraaaa@79.112.35.181) |
| 15:32.22 | p3nguin | Okay, maybe I still have no idea what the actual problem is. |
| 15:33.06 | *** join/#asterisk CunningPike (~CunningPi@204.239.8.157) |
| 15:33.27 | Faustov | thanks |
| 15:34.33 | p3nguin | Alright, now I got it. If you use the record/playback idea, the delay between the real voice and the page will be much larger, so the annoyance would be lessened. |
| 15:35.44 | bent_screwdriver | @leifmadsen: just figured out that you have to add the key and value to cdr_manager.conf. also a restart was required as a reload didn't take it. |
| 15:36.20 | leifmadsen | cdr_manager.conf? that doesn't make much sense since it seemed you were trying to write a custom field to your database |
| 15:36.27 | leifmadsen | at least that's how I read it |
| 15:36.40 | leifmadsen | it's more likely the restart did the trick and not the cdr_manager.conf change |
| 15:36.59 | beefpastry | but the time it would leave customers parked wouldn't be desirable |
| 15:38.14 | bent_screwdriver | @leifmadsen: good point. i'll take it out of manager and see if it still pops the field, just out of curiosity |
| 15:39.59 | Naikrovek | i thought the internet had everything |
| 15:40.12 | Naikrovek | i want a funny poem about faxing to put on my fax cover sheet |
| 15:40.24 | Naikrovek | guess i'll have to write one |
| 15:40.43 | Naikrovek | this ... could take time |
| 15:41.51 | coppice | OH FAX! |
| 15:42.03 | Naikrovek | that's a good poem title |
| 15:42.11 | Naikrovek | maybe a limerick would be better |
| 15:42.22 | leifmadsen | Naikrovek: write a haiku! |
| 15:42.31 | Naikrovek | i can barely write my name |
| 15:42.45 | p3nguin | beefpastry: 5 seconds to record the page, 5 seconds to playback the page. Seems okay to me. |
| 15:42.50 | Naikrovek | i'll give it a go, though |
| 15:43.25 | beefpastry | yeah...try to get my secretaries to keep a page to 5 seconds and I'll buy you a big steak dinner. |
| 15:43.36 | p3nguin | hahaha |
| 15:43.56 | p3nguin | I would expect 7 seconds max. |
| 15:44.22 | p3nguin | A person should not feel uncomfortable on hold for at least the first 30 seconds. |
| 15:44.38 | bent_screwdriver | @leifmadsen: you're right. i commented out the key/value, restarted *, and the field is still populating. must have just needed to add field to db, add Set(CDR(field)) to dp, and restart. thx for your help. |
| 15:44.42 | p3nguin | I'm usually okay for a couple minutes. |
| 15:51.44 | *** join/#asterisk asheron (~asheron@190.98.10.210) |
| 15:51.49 | Get_The_Fish | hey leif, I think that I found a bug in LDAP realtime sippeers... is documenting the behavior enough to get a bug report started? |
| 15:51.59 | *** part/#asterisk asheron (~asheron@190.98.10.210) |
| 15:52.02 | *** join/#asterisk asheron (~asheron@190.98.10.210) |
| 15:52.11 | leifmadsen | Get_The_Fish: open the bug, state how to reproduce the issue, and what the bug is |
| 15:52.32 | Naikrovek | here's what i came up with. the bandwidth order fax: http://pastie.org/928064 |
| 15:52.51 | bmoraca_work | mysql-connector-odbc or something |
| 15:52.53 | bmoraca_work | erm |
| 15:52.58 | Get_The_Fish | I never know how much detail to put in bug reports, and I think that two things are related to each other... I dont want to waste anyone's time either |
| 15:53.09 | asheron | i cant get digium asterisk gui to work with asterisk, does anyone has this problem to ? |
| 15:53.34 | *** join/#asterisk [T]ank (~ckwall@77.sub-75-252-51.myvzw.com) |
| 15:53.35 | bmoraca_work | asheron: #asterisk-gui or #asterisknow for help with that |
| 15:53.43 | asheron | bmoraca_work: thx |
| 15:53.50 | beefpastry | Our paging requires a little more information due to the nature of the business...10 seconds is a more likely goal for us, but we're also a service company so promptness with calls is imperative |
| 15:54.36 | [TK]D-Fender | #asterisk-gui , not #asterisknow . That distro stopped using the old OS, and the old GUI. Neither is really supported anymore |
| 15:55.20 | bmoraca_work | well there you go |
| 15:56.31 | smooth_penguin | hey, how do I figure if a asterisk training/cert center is legitimate or not? |
| 15:56.53 | d1b | smooth_penguin: ask a few good questions |
| 15:56.58 | leifmadsen | smooth_penguin: request references |
| 15:57.59 | malcolmd | http://www.digium.com/en/training/partners/partners.php |
| 15:58.06 | *** join/#asterisk michael-i (~michael-i@p3EE28B59.dip0.t-ipconnect.de) |
| 15:58.26 | Qwell | malcolmd FTW |
| 15:58.31 | Qwell | lurker :) |
| 15:58.50 | malcolmd | yeah, i did that yesterday, too ;) popped in with a URL and disappeared again, silently... |
| 15:59.58 | bmoraca_work | not a single US center listed there |
| 15:59.59 | bmoraca_work | curious |
| 16:00.00 | smooth_penguin | malcolmd, ok they are listed there but I wasnt able to find them using the training locator |
| 16:00.03 | smooth_penguin | hence I asked |
| 16:00.06 | asheron | is there a gui like asterisk gui that works with 1.6 version of asterisk ? |
| 16:00.40 | smooth_penguin | http://www.digium.com/en/training/locator/ |
| 16:00.55 | bmoraca_work | it seems asterisk's international appeal is greater than its US appeal...either that or US telco engineers are smart enough not to need training (HAHAHA) |
| 16:01.19 | smooth_penguin | This is wrt "Enterux Solutions Private Limitd" |
| 16:01.46 | smooth_penguin | I was looking for a Indian center |
| 16:02.26 | smooth_penguin | but wasnt able to find one through the locator - came across this center by chance while googling around |
| 16:03.31 | Naikrovek | seriously? no reviews on my office fax cover sheet poetry? http://pastie.org/928064 |
| 16:03.38 | malcolmd | smooth_penguin: means enterux doesn't have an upcoming class that they've listed with us - or that we've failed to properly register their notice of such an upcoming class in our system |
| 16:03.56 | smooth_penguin | malcolmd, http://www.entvoice.com/training/index.php?name=set |
| 16:04.17 | smooth_penguin | Im just trying to be totally sure Im paying the right folks |
| 16:04.24 | manxpower | asheron: virtually nobody here uses a GUI with Asterisk. Those people are in OTHER channels. |
| 16:04.52 | d1b | what's a gui ? |
| 16:05.21 | bmoraca_work | hmmm...i wonder how much it costs to take the dCAP exam and whether it's worth it from a career point of view.... |
| 16:05.36 | leifmadsen | d1b: globally unique identifier |
| 16:05.42 | asheron | manxpower: oke, i will have to stop using it then, and learn more about it |
| 16:05.50 | malcolmd | smooth_penguin: re: the url. interesting. i'll alert the training folks here. drop me an e-mail with your e-mail if you want them to get back to you directly. |
| 16:05.57 | bmoraca_work | leifmadsen: isn't that GUID? |
| 16:05.57 | d1b | leifmadsen: that's gln |
| 16:06.04 | p3nguin | Before we get too far away from the CDR topic... when should duration and billsec values be different? Reviewing my csv, the two values always match. |
| 16:06.09 | leifmadsen | lets not be pedantic here, I was having fun :) |
| 16:06.14 | bmoraca_work | lol |
| 16:06.24 | d1b | manxpower: http://www.asterisk.org/asterisknow |
| 16:06.26 | manxpower | asheron: no reason to not use a GUI. This is simply not the place to get support for GUIs |
| 16:06.49 | smooth_penguin | malcolmd, ok thanks, Ive spoken to enterux, and they just had unconvincing reasons |
| 16:07.18 | manxpower | d1b: No, I'm not going to start using a GUI |
| 16:07.21 | bmoraca_work | p3nguin: duration would be the life of the channel where billsec is the length of time the call was connected. if you have lots of ringing time, billsec will be less. if you always immediately answer the call (findme or IVR, etc), then they'll always be the same |
| 16:07.38 | p3nguin | Ah, okay. |
| 16:07.40 | bmoraca_work | or if you're using analog channels, they'll likely always be the same too |
| 16:07.46 | bmoraca_work | at least, that's what i've always held it to be |
| 16:07.52 | bmoraca_work | if i'm wrong, someone will correct me, i'm sure |
| 16:08.10 | manxpower | bmoraca_work: Dialing on analog takes a few seconds |
| 16:08.13 | d1b | oh sorry i read it wrong |
| 16:08.28 | d1b | asheron: http://www.asterisk.org/asterisknow |
| 16:09.32 | *** join/#asterisk nny (~Scott@64.203.239.83) |
| 16:10.21 | asheron | d1b: i cant you that, i have a dedicated server |
| 16:10.41 | d1b | right. |
| 16:10.58 | *** join/#asterisk TimeRider (~steve@109.224.131.68) |
| 16:11.10 | bmoraca_work | asheron: nothing stoping you from using the asterisknow repo to load the same utilities it uses (freepbx) |
| 16:12.01 | asheron | bmoraca_work: i will look into freepbx |
| 16:12.45 | *** join/#asterisk rdircio (~admin@189.242.22.188) |
| 16:13.16 | leifmadsen | M17220 |
| 16:13.18 | MuffinMan | [ready for testing] [Asterisk] Documentation 0017220: [patch] Add ability to generate an ASCII document from the TeX files reported by lmadsen https://issues.asterisk.org/view.php?id=17220 |
| 16:13.20 | leifmadsen | manxpower: ^^^^ |
| 16:13.49 | Naikrovek | nice |
| 16:14.17 | smooth_penguin | malcolmd, Ive sent the mail |
| 16:14.31 | d1b | ooo mantis |
| 16:15.59 | malcolmd | smooth_penguin: got it; replied :D |
| 16:16.21 | [T]ank | i am finding that AMD as i have it configured is most often times correctly detecting machine and human correctly. But when it detects machine, its still playing the recorded message to early. Its playing the message while the voicemail greeting is playing. When i use: exten => s,n(mach),WaitForSilence(2500) it causes the call to hang up. What am I doing wrong? Here is my extension and how i have it set up: http://pastebin.com/sQHaZLmD and here |
| 16:16.24 | smooth_penguin | thanks :) |
| 16:16.33 | [T]ank | could anyone possibly suggest what I am doing wrong? |
| 16:16.45 | malcolmd | smooth_penguin: thank you for pointing it out |
| 16:17.08 | [T]ank | actually in the examples i gave above, it doesnt look like its playing the message at all |
| 16:17.10 | Naikrovek | [T]ank: AMD is a black art; i don't think anyone has mastered it yet |
| 16:17.15 | leifmadsen | [T]ank: I've not heard of too many people who have had luck with AMD() unfortunately |
| 16:17.16 | smooth_penguin | malcolmd, oh well np, infact Im hoping they are legit so I can get certified ;P |
| 16:17.26 | leifmadsen | Naikrovek: black art++ |
| 16:17.52 | *** join/#asterisk slidesinger (~slidesing@c-68-44-99-50.hsd1.nj.comcast.net) |
| 16:18.12 | [T]ank | thats too bad. it could do so much |
| 16:22.37 | *** join/#asterisk Sargun (~Sargun@atarack/Staff/Sargun) |
| 16:23.00 | *** join/#asterisk lhz (~shrekz@c-b9aa72d5.021-158-73746f34.cust.bredbandsbolaget.se) |
| 16:25.58 | *** join/#asterisk dajhorn (~dajhorn@adsl-75-17-124-26.dsl.rcsntx.sbcglobal.net) |
| 16:27.00 | tzafrir_laptop | OK. Updated spandsp git repo. Also included there the scripts for importing tarballs. http://gitorious.org/spandsp |
| 16:28.17 | hardwire | I kinda wish there was a userspace dundi query system |
| 16:28.25 | leifmadsen | userspace? |
| 16:28.32 | hardwire | leifmadsen: I suppose I could use AMI |
| 16:28.36 | leifmadsen | hardwire: why not use the CLI? |
| 16:28.42 | leifmadsen | hardwire: dundi lookup foo@bar bypass |
| 16:28.44 | hardwire | leifmadsen: because |
| 16:28.52 | hardwire | :P |
| 16:29.03 | leifmadsen | that's about as userspace as it gets :) |
| 16:29.06 | hardwire | leifmadsen: I also want a standalone dundi server.. I do very .. very .. evil things with dundi. |
| 16:29.13 | hardwire | very. |
| 16:29.17 | leifmadsen | welcome to the club |
| 16:29.20 | hardwire | nono |
| 16:29.25 | leifmadsen | mine kills kittens |
| 16:29.26 | hardwire | I'm talking 4 goats in a day |
| 16:29.35 | leifmadsen | goats aren't cute like kittens |
| 16:29.46 | hardwire | kittens have a spawn point.. |
| 16:29.49 | hardwire | goats are finite. |
| 16:30.20 | tzafrir_laptop | after you killed them: sure they are |
| 16:31.49 | *** join/#asterisk korcan (~kshamoun@ip65-44-169-89.z169-44-65.customer.algx.net) |
| 16:31.54 | *** join/#asterisk ruyo (~psantos@195.23.253.223) |
| 16:32.16 | hardwire | leifmadsen: any insight into "realtime" dundi mappings? |
| 16:32.25 | hardwire | please say "oh.. that just plain works". |
| 16:33.59 | leifmadsen | hardwire: I'm not sure what you mean |
| 16:34.16 | leifmadsen | like using realtime to store the mappings? doesn't exist afaik |
| 16:34.28 | hardwire | yar |
| 16:34.34 | hardwire | it does if I use extconfig I suppose. |
| 16:34.44 | leifmadsen | aye |
| 16:34.46 | hardwire | aye |
| 16:34.50 | leifmadsen | static realtime I guess :) |
| 16:34.55 | hardwire | super cool |
| 16:34.59 | leifmadsen | thank you |
| 16:35.06 | hardwire | oh you're so very welcome. |
| 16:35.20 | leifmadsen | aaaaaaand scene. |
| 16:35.27 | hardwire | yeh |
| 16:35.29 | hardwire | *applause* |
| 16:35.47 | leifmadsen | lunch! |
| 16:35.55 | hardwire | erm.. I think the biggest hurdle in what I wanted to use dundi for was overcoming how the results are returned |
| 16:35.58 | hardwire | I needed RR |
| 16:36.31 | hardwire | so I ended up using dundiquery in the dialplan and reimplementing the dundi switch |
| 16:37.10 | hardwire | I'm still reading the dundi code to figure out the best method of returning a round robin offset list to the switch for equal weight results. |
| 16:37.24 | hardwire | I'm guessing I should be doing this the same way DNS works. |
| 16:43.47 | *** join/#asterisk ariel_ (~chatzilla@63.214.236.169) |
| 16:44.30 | *** join/#asterisk Ad-Hoc (~nimbus@62.1.237.46.dsl.dyn.forthnet.gr) |
| 16:45.52 | *** part/#asterisk ehsjoar (~ehsjoar@c-98-245-155-132.hsd1.co.comcast.net) |
| 16:55.28 | manxpower | leifmadsen: you have removed one of the few things about Asterisk itself that I thought was a real and serious barrier to n00bs. |
| 16:55.47 | manxpower | I should have put "easily fixable" in there. THANK YOU. |
| 16:59.28 | hardwire | which? |
| 17:00.26 | *** join/#asterisk GameGamer43|Mac (~GameGamer@CPE-65-27-76-78.new.res.rr.com) |
| 17:01.23 | leifmadsen | manxpower: ya took me about 2 hours, but I got it |
| 17:02.37 | *** join/#asterisk TSM (~the_softw@fw-lon1.wenn.com) |
| 17:02.44 | Naikrovek | hardware the .tex documentation |
| 17:02.48 | Naikrovek | ugh |
| 17:02.54 | Naikrovek | hardwire: the .tex documentation |
| 17:03.20 | hardwire | Naikrovek: .tex documentation was a n00barrier? |
| 17:03.24 | hardwire | hmm |
| 17:03.33 | Naikrovek | how many of your users know how to compile .tex |
| 17:03.38 | Naikrovek | into a pdf or dvi document |
| 17:03.47 | leifmadsen | make asterisk.pdf :) |
| 17:04.01 | Naikrovek | if latex or tex are installed... |
| 17:04.15 | Naikrovek | not saying tex is a bad idea |
| 17:04.17 | Naikrovek | my resume is in .tex |
| 17:04.22 | Naikrovek | but newbs don't grok tex |
| 17:06.14 | *** part/#asterisk codefreeze-lap (~murf@mail.parsetree.com) |
| 17:07.18 | hardwire | ok so.. my project over the next week or so is to smartly change how sip reload handles sip reregistration. |
| 17:07.56 | hardwire | I need to add a 'sip registration reload' command for forced registration reload and only reregister if a peers data has been invalidated in any way during a 'sip reload' |
| 17:08.09 | hardwire | this is mostly due to problems I'm having with broadvoice |
| 17:08.35 | hardwire | which.. at this point.. are easier to handle using python + twistedsip as a registration UAC. |
| 17:08.57 | *** join/#asterisk oej (~olle@ns.webway.se) |
| 17:09.18 | hardwire | they have a 30 second registration timeout and limit (per auth name/or ip if not present) 10 attempts per second. |
| 17:09.51 | hardwire | so sip reload unfortunately causes sip reregistrations to happen.. and for some reason asterisk likes to register a good half dozen times per registration for some reason. |
| 17:10.42 | hardwire | so.. yeh.. even an idle system has trouble keeping 8 registrations to broadvoice active without hitting their limit problem |
| 17:10.49 | hardwire | their solution is to use DNS-SRV |
| 17:11.10 | hardwire | however that means I need to have a broadvoice peer that allows the entire gamut of broadvoice gateways access in through a single peer. |
| 17:11.14 | hardwire | sigh |
| 17:11.36 | hardwire | right now I have to manually update /etc/hosts before each sip reload |
| 17:11.37 | hardwire | :P |
| 17:12.40 | bmoraca_work | why? |
| 17:14.33 | bmoraca_work | hardwire: why not just use realtime sip peers? i haven't done a sip reload on my main asterisk box in months. |
| 17:15.07 | hardwire | bmoraca_work: even without sip reload I run into this problem after a few days and network connection problems. |
| 17:16.31 | bmoraca_work | sounds like broadvoice needs to fix their shit |
| 17:17.40 | hardwire | according to broadvoice.. their shit is awesome. |
| 17:18.07 | hardwire | because every other UAC supports DNS-SRV and when it fails with one proxy it re-registers to the other. |
| 17:18.39 | *** join/#asterisk iCEBrkr (~icebrkr@72.251.206.106) |
| 17:18.44 | *** join/#asterisk oej (~olle@ns.webway.se) |
| 17:18.52 | [TK]D-Fender | accoding to others, our awesome is shit |
| 17:19.23 | hardwire | I can't seem to find a good SIP proxy that supports DNS-SRV and acts as a UAC |
| 17:19.32 | hardwire | however.. I got simpleopal to work almost well enough |
| 17:19.43 | hardwire | and I'd have to run some tests to see if it should use another IP or not |
| 17:20.14 | hardwire | simpleopal unfortunately doesn't support changing the IAX port that it can use.. and SIP redirection is broken |
| 17:21.23 | *** join/#asterisk spiceycurry (~mikecurry@proxy.hostopia.com) |
| 17:22.05 | bmoraca_work | i'm going to be testing multiple asterisk boxes, sharing a single RT database, with dundi between them and DNS SRV distributing client connections between them...i hope it works :P |
| 17:22.15 | spiceycurry | is there a way to automate the make menuselect for asterisk off hand? |
| 17:22.19 | hardwire | bmoraca_work: good luck |
| 17:22.24 | hardwire | I've done something similar |
| 17:22.41 | hardwire | I had to write some custom dialplan for stdexten macros |
| 17:22.47 | hardwire | it collated results from dundi_query into a dialgroup |
| 17:23.00 | hardwire | that way multiple phones could register with the same extension at all asterisk nodes |
| 17:23.19 | hardwire | I set up an amazing system for doing a completely distributed asterisk setup |
| 17:23.21 | hardwire | and my boss said no |
| 17:23.27 | hardwire | he couldn't understand the dialplan anymore |
| 17:23.28 | hardwire | sigh |
| 17:23.33 | hardwire | sooo.. it's all flatfile now :) |
| 17:23.37 | Slugs_ | ;/ |
| 17:23.55 | spiceycurry | SHIZ, just ran into the 64-bit problem I think with chan_ooh323 |
| 17:24.16 | spiceycurry | Should I be running asterisk on a 32-bit system? |
| 17:24.52 | bmoraca_work | hardwire: my benefit is that my system doesn't get phones registered directly to it, but rather i use it as a switch among many other asterisk boxes |
| 17:25.10 | bmoraca_work | so they'll only ever be registeres to one at a time |
| 17:25.35 | spiceycurry | Anyone have problems compiling asterisk with the chan_ooh323 errors? |
| 17:25.51 | freezey | whats the dial plan look like so the user doesnt have to dial one to get dial out |
| 17:25.58 | freezey | i have NXXNXXXXX |
| 17:26.01 | freezey | that wasnt doing it |
| 17:26.14 | p3nguin | doesn't have to dial one? What does that mean? |
| 17:26.23 | p3nguin | On, dial a 1 on the number? |
| 17:26.33 | freezey | yeah like 1=800=44444 |
| 17:26.35 | freezey | crap like that |
| 17:26.37 | hardwire | bmoraca_work: exactly |
| 17:26.52 | spiceycurry | Do I have voice? |
| 17:26.52 | hardwire | bmoraca_work: the beni to me was that people could register their phones on the closest proxy |
| 17:26.57 | hardwire | they could literally take them from site to site |
| 17:27.01 | hardwire | or hotdesk one already on site |
| 17:27.02 | p3nguin | something like exten => _NXXNXXXXXX,1,Dial(SIP/itsppeername/1${EXTEN}) |
| 17:27.06 | hardwire | and I wouldn't have to do lots of crazy |
| 17:27.06 | spiceycurry | test.. |
| 17:27.39 | freezey | p3nguin: yeah i had that but a change reverted and now its not working |
| 17:27.47 | spiceycurry | Can anyone read this? |
| 17:27.56 | p3nguin | freezey: Show me the failure. |
| 17:28.10 | bmoraca_work | hardwire: well, that'll be an eventual goal (geographically distributed POPs) |
| 17:28.10 | p3nguin | Otherwise, I maintain that this is how it is done. |
| 17:28.11 | freezey | p3nguin: these are the ones i have now 1|NXXNXXXXXX NXXNXXXXXX 1NXXNXXXXXX 81NXXNXXXXXX |
| 17:28.18 | hardwire | bmoraca_work: indeed |
| 17:28.19 | bmoraca_work | but we'lre a bit aways from there now |
| 17:28.23 | hardwire | may your efforts be fruitful and multiply. |
| 17:28.35 | hardwire | it was pretty easy to set up the initial part |
| 17:28.45 | hardwire | I even got distributed queues and conferencing working |
| 17:28.53 | hardwire | the only real problem was pulling back reports. |
| 17:29.00 | hardwire | also.. iax2 transfers are THE BOMB |
| 17:29.05 | bmoraca_work | i've already confirmed that the shared RT database works great |
| 17:29.12 | hardwire | hello call.. goodbye call.. mwa ha ha ha ha |
| 17:29.22 | hardwire | bmoraca_work: distributed DBs can be a pita |
| 17:29.27 | bmoraca_work | yeah |
| 17:29.30 | hardwire | I'd recommend something like SymettricDS |
| 17:29.30 | bmoraca_work | that's the next issue... |
| 17:29.33 | hardwire | for sanity sake |
| 17:29.38 | hardwire | it's free |
| 17:29.39 | bmoraca_work | symettricds? |
| 17:29.57 | bmoraca_work | i was trying to get mysql clustering set up, but after losing half my hair, i gave up |
| 17:29.58 | hardwire | bmoraca_work: it hooks into your tables and adds triggers to them.. |
| 17:30.08 | hardwire | then performs differential sync |
| 17:30.16 | hardwire | sql independent. |
| 17:30.18 | bmoraca_work | interesting... |
| 17:30.19 | freezey | p3nguin: so i am not sure why that would happen if i have those in my dialplan |
| 17:30.22 | freezey | itw as working earlier |
| 17:30.23 | freezey | now just stopped |
| 17:30.29 | hardwire | bmoraca_work: yes.. it works really well.. it's java based tho |
| 17:30.31 | p3nguin | freezey: Show me the failure. |
| 17:30.35 | bmoraca_work | ick |
| 17:30.52 | bmoraca_work | well, if it can do symmettric replication, that'll work great |
| 17:31.05 | bmoraca_work | if i can time replication to only happen ever 5 minutes or something, that'd be great, too |
| 17:31.07 | freezey | p3nguin: http://pastebin.com/HaDudd8p |
| 17:31.20 | bmoraca_work | it'd be great for distributing points of failure as well |
| 17:31.21 | bmoraca_work | hmmm |
| 17:31.25 | bmoraca_work | this might be a great solution |
| 17:31.53 | p3nguin | freezey: FreePBX? |
| 17:31.57 | freezey | yeah |
| 17:32.03 | p3nguin | freezey: There's yer problem. |
| 17:32.10 | freezey | stupid thing reverts changes every god dam time you do anything |
| 17:32.20 | p3nguin | Yeah, that's how it works. |
| 17:32.22 | freezey | i had to put this up in order for people who do not understand howto use CLI can add users etc |
| 17:32.30 | p3nguin | You aren't supposed to alter the config files by hand. |
| 17:32.33 | freezey | this is more oof a headache |
| 17:32.38 | freezey | yeah but the interface blows chunks |
| 17:32.38 | freezey | lol |
| 17:32.44 | p3nguin | Stop using FreePBX. |
| 17:32.51 | freezey | cant |
| 17:32.53 | freezey | work |
| 17:33.01 | freezey | if i leave nobody will know howto use the phones |
| 17:33.01 | freezey | lol |
| 17:33.04 | p3nguin | Then learn to use it correctly. |
| 17:33.13 | freezey | sucks |
| 17:33.25 | freezey | so i gues i am going over to that chan to ask them eh? |
| 17:33.35 | p3nguin | That's where I would start. |
| 17:33.37 | freezey | k |
| 17:33.38 | freezey | thanks |
| 17:33.55 | p3nguin | If you weren't using FreePBX, we could solve it here. |
| 17:34.17 | *** join/#asterisk soman (~somnath@dsl-jklbrasgw1-fe19fb00-113.dhcp.inet.fi) |
| 17:34.24 | freezey | k |
| 17:34.58 | bmoraca_work | hardwire: do you know if symmetricds runs on windows? |
| 17:35.17 | hardwire | bmoraca_work: it's java! |
| 17:35.20 | hardwire | I believe it does |
| 17:35.30 | hardwire | as well as interfaces to MsSQL |
| 17:35.39 | *** join/#asterisk atis_work (~atis_work@193.238.212.171) |
| 17:35.43 | *** join/#asterisk anthm (~anthm@freeswitch/developer/anthm) |
| 17:35.51 | bmoraca_work | asterisk extconfig realtime doesn't like mssql, lol |
| 17:36.10 | hardwire | I use realtime ODBC |
| 17:36.19 | hardwire | and MySQL |
| 17:36.21 | hardwire | it's nice |
| 17:36.35 | bmoraca_work | res_odbc and extconfig here...no problems at all |
| 17:36.51 | bmoraca_work | i don't use realtime extensions, though. only SIP. |
| 17:36.54 | hardwire | oh.. doesn't like mssql |
| 17:36.55 | hardwire | meh! |
| 17:37.09 | hardwire | also.. symmetricDS can be queried ala HTTP |
| 17:37.11 | bmoraca_work | it worked, for the most part, but it couldn't see some of the table names |
| 17:37.13 | hardwire | so you could use res_curl :) |
| 17:37.17 | bmoraca_work | ick |
| 17:37.21 | hardwire | DO IT@ |
| 17:37.24 | hardwire | DO IT NOW! |
| 17:37.43 | hardwire | I should really be focusing on getting hylafax set up |
| 17:37.43 | hardwire | bbl |
| 17:37.44 | bmoraca_work | i'd just like to replicate from one central database to a local instance of mysql on each proxy |
| 17:38.04 | hardwire | yeh.. use something that does that on behalf of the built in replication. |
| 17:38.22 | hardwire | trust me |
| 17:38.25 | hardwire | :P |
| 17:38.47 | hardwire | I just tossed a mysql master/master solution because it kept finding problems and crashing to a halt |
| 17:38.55 | hardwire | and in the mean time queries were being made on the outdated host. |
| 17:38.58 | *** join/#asterisk oej (~olle@ns.webway.se) |
| 17:39.03 | bmoraca_work | it looks like this SymmetricDS does what i need |
| 17:39.15 | hardwire | bmoraca_work: yeh.. I tested it for a while and it appeared sane |
| 17:39.22 | hardwire | but it's a bit abstract |
| 17:39.31 | hardwire | it took me a while to get the hang of ut |
| 17:39.32 | hardwire | it |
| 17:39.46 | hardwire | bbl |
| 17:41.26 | *** join/#asterisk moos3 (~rgenthner@rrcs-24-39-23-74.nys.biz.rr.com) |
| 17:44.18 | bent_screwdriver | anyone know where to get soft cat5/6 so i can make some patch cables like those that come with Polycoms? |
| 17:44.35 | [TK]D-Fender | bent_screwdriver: huh? |
| 17:44.51 | fenrus | just buy pre-fabricated patch cables? |
| 17:45.05 | bent_screwdriver | [TK]D-Fender: those patch cables that come with Polycoms are much more flexible |
| 17:45.41 | bent_screwdriver | fenrus: okay, who sells pre-fab patch cables that are soft/flexible like those that come with Polycoms? |
| 17:45.47 | p3nguin | bent_screwdriver: They are likely to be regular Cat 5e stranded patch cables. You can make your own or get them from CDW. |
| 17:46.01 | bmoraca_work | the user guide's only 60 pages... |
| 17:46.19 | *** join/#asterisk Peste_Bubonica (~eduardo.f@189-47-176-158.dsl.telesp.net.br) |
| 17:46.22 | Peste_Bubonica | Hi all |
| 17:46.25 | bent_screwdriver | p3nguin: i have 4 boxes of cat5 and 2 cat6 and the cable is much more ridgid.... |
| 17:46.34 | p3nguin | bent_screwdriver: Are they stranded? |
| 17:46.52 | *** join/#asterisk cusco (~trilili@2001:0:53aa:64c:2448:408e:2ac0:762d) |
| 17:46.54 | cusco | hi.. |
| 17:47.00 | bmoraca_work | bent_screwdriver: any patch cable made with stranded cable will be. |
| 17:47.15 | bmoraca_work | bent_screwdriver: for bulk patch cables, i buy them from PI Manufacturing (www.pimfg.com) |
| 17:47.16 | cusco | I have a queue and clients wating in queue, and asterisk is not dialing to members |
| 17:47.20 | p3nguin | That's what I'm saying. |
| 17:47.20 | *** join/#asterisk flyankur (~Zod@125.19.237.34) |
| 17:47.49 | moos3 | whats a good linux sip or iax soft phone to use |
| 17:47.52 | bmoraca_work | bent_screwdriver: per spec, though, patch cables should ALWAYS be stranded. spec calls for 90m of solid core, flanked by 5m of stranded on either side. |
| 17:47.54 | bent_screwdriver | p3nguin: they are. i'll get some that aren't and see if that works. thanks p3nguin and bmoraca_work |
| 17:48.03 | cusco | http://paste.debian.net/70084/ |
| 17:48.07 | cusco | I don't know what to do |
| 17:48.20 | p3nguin | Solid wire is usually more stiff than stranded, and usually isn't used for patch cables. |
| 17:48.32 | [TK]D-Fender | cusco: You might want to unpause some members |
| 17:49.06 | Peste_Bubonica | Im making testes on a IAX2 channel, on two asterisk boxes 1.6.2.6. The latency between each server is about 20ms, 25ms MAX. I can hear a MP3 clear, and without issues over this channel,using MP3Player(), but a simple application that uses Playback stalls, like a voicemail application, or a macro that use 3 or 4 playbacks. The dialplan stalls, and I need to hangup the call |
| 17:49.37 | bent_screwdriver | p3nguin: okay, i just noticed the polycom cables have info on them so maybe i can start there.... |
| 17:49.39 | *** join/#asterisk [SySteM] (~antoine@aqu33-6-88-168-80-163.fbx.proxad.net) |
| 17:49.42 | [SySteM] | Hello |
| 17:49.55 | [SySteM] | I search some help about asterisk 1.4 and app_fax on debian lenny |
| 17:50.31 | [SySteM] | [Apr 21 19:31:39] WARNING[5593]: /usr/src/asterisk/agx-ast-addons/app-spandsp/app_fax.c:524 fax_run: TXFAX: Channel INF is NULL, i will continue... |
| 17:50.32 | [SySteM] | [Apr 21 19:31:39] WARNING[5593]: /usr/src/asterisk/agx-ast-addons/app-spandsp/app_fax.c:500 fax_run: TXFAX: Channel has been hanged at fax. |
| 17:50.32 | [SySteM] | [Apr 21 19:31:39] ERROR[5593]: /usr/src/asterisk/agx-ast-addons/app-spandsp/app_fax.c:214 phase_e_handler: [FaxSent ERROR] result (49) The call dropped prematurely. |
| 17:50.37 | p3nguin | ~pb |
| 17:50.38 | infobot | [~pb] A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://bin.cakephp.org/ , http://asterisk.pastey.net/ , or install pastebinit with yum or aptitude. |
| 17:50.44 | p3nguin | [system]: pastebin |
| 17:50.44 | cusco | [TK]D-Fender: ok they were paused for other reason only for a couple of seconds |
| 17:50.47 | cusco | http://paste.debian.net/70085/ |
| 17:50.48 | [SySteM] | sorry. |
| 17:50.49 | cusco | here they are not.-- |
| 17:50.52 | cusco | I don't knwo what to do |
| 17:51.24 | [SySteM] | asterisk call number on txfax() and... nothing.. its blank |
| 17:51.40 | [SySteM] | 20 sec next.. hangug and 'got theses lines on my console view |
| 17:53.34 | [TK]D-Fender | susUNPAUSE someone |
| 17:53.39 | [TK]D-Fender | cusco: UNPAUSE someone |
| 17:53.48 | *** join/#asterisk githogori (~githogori@catmint.mail-abuse.org) |
| 17:54.21 | cusco | 18:50 < cusco> http://paste.debian.net/70085/ |
| 17:54.24 | cusco | they are not paused! |
| 17:54.44 | cusco | they were paused because they were answering other queues |
| 17:54.53 | cusco | this specific queue was not being answered |
| 17:55.14 | *** join/#asterisk jehovah (~bisconer@unaffiliated/jehovah) |
| 17:56.25 | *** join/#asterisk Sipster (~Sipster@modemcable045.5-200-24.mc.videotron.ca) |
| 17:56.41 | *** join/#asterisk githogori (~githogori@catmint.mail-abuse.org) |
| 17:59.06 | *** join/#asterisk uqlev (~yuriy@91.184.221.31) |
| 17:59.51 | *** join/#asterisk githogori (~githogori@catmint.mail-abuse.org) |
| 18:02.55 | *** join/#asterisk slawek (~slawek@chello089072183060.chello.pl) |
| 18:03.57 | *** join/#asterisk ehsjoar (~ehsjoar@c-98-245-155-132.hsd1.co.comcast.net) |
| 18:04.57 | *** join/#asterisk theHub (~theHub@69.177.93.21) |
| 18:05.32 | *** join/#asterisk fskrotzki_ (~fskrotzki@cpe-74-74-245-250.rochester.res.rr.com) |
| 18:09.18 | [TK]D-Fender | cusco: Then you should ahve stopped it from calling them |
| 18:15.34 | hardwire | wonders why asteriskcallbacklogin died for realz. |
| 18:15.36 | hardwire | err |
| 18:15.40 | hardwire | agentcallbacklogin |
| 18:16.34 | *** join/#asterisk atis_work (~atis_work@193.238.212.171) |
| 18:17.44 | *** join/#asterisk azertyuio (~chatzilla@bgl93-7-88-189-218-150.fbx.proxad.net) |
| 18:17.53 | *** join/#asterisk GameGamer43|Mac (~GameGamer@cpe-74-65-36-91.rochester.res.rr.com) |
| 18:19.27 | azertyuio | hi |
| 18:19.30 | azertyuio | i can't see asterisk on my sys |
| 18:19.32 | azertyuio | on freepbx it says asterisk not running it is a critical error |
| 18:19.44 | *** join/#asterisk oej (~olle@ns.webway.se) |
| 18:19.54 | azertyuio | what too do ? |
| 18:21.08 | p3nguin | You have two choices. |
| 18:21.26 | azertyuio | yes |
| 18:21.47 | p3nguin | 1) Ask in the freepbx channel, 2) Stop using freepbx. |
| 18:22.41 | azertyuio | why have to stop using freepbx ? |
| 18:22.53 | p3nguin | because we don't support it here. |
| 18:23.17 | jehovah | whats the defualt login for FreePBX? |
| 18:23.27 | p3nguin | ~freepbx |
| 18:23.28 | infobot | [~freepbx] FreePBX is unable to be supported here. It is made up of complex dialplans and scripts which can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking there, or fpbx |
| 18:23.28 | Qwell | jehovah: #freepbx |
| 18:23.31 | Corydon76-dig | and if they're not going to support it, then you really don't have much of a choice |
| 18:24.15 | *** join/#asterisk knctrnl (~aembrey@76.164.169.130) |
| 18:24.21 | p3nguin | Yep. If the result of #1 is unsatisfactory, the only remaining thing is #2. |
| 18:25.54 | *** join/#asterisk kannan (~kannan@118.102.142.210) |
| 18:26.17 | *** join/#asterisk QaDeS (~mklaus@p54A1ACB2.dip0.t-ipconnect.de) |
| 18:26.47 | knctrnl | I have been reading about this topic and I have read that it has to be supported by the phone provider and want to know if my info is correct. Is it possible to configure asterisk to receive SMS messages on a landline? |
| 18:27.12 | leifmadsen | knctrnl: in certain countries under the correct circumstances |
| 18:27.21 | knctrnl | US? |
| 18:27.22 | leifmadsen | in Canada or USA -- no. |
| 18:27.26 | knctrnl | ahh |
| 18:27.42 | leifmadsen | it works in Germany afaik |
| 18:28.21 | *** join/#asterisk Z_God (~julius@wlan237200.mobiel.utwente.nl) |
| 18:28.30 | *** join/#asterisk FlaPer87 (~FlaPer87@unaffiliated/flaper87) |
| 18:28.55 | FlaPer87 | hey guys, is it possible to know (with asterisk) if the number I'm dialing is a mobile number? |
| 18:29.17 | Qwell | FlaPer87: not without something external that can provide a list |
| 18:29.21 | Qwell | ie; no |
| 18:29.26 | p3nguin | If you implemented some type of database, sure. |
| 18:29.45 | p3nguin | Asterisk alone... refer to qwell's answer. |
| 18:30.00 | FlaPer87 | thanks |
| 18:30.03 | *** part/#asterisk azertyuio (~chatzilla@bgl93-7-88-189-218-150.fbx.proxad.net) |
| 18:30.15 | FlaPer87 | I don't care if I get the inf after or before calling |
| 18:30.22 | FlaPer87 | actually after would be better |
| 18:30.25 | p3nguin | And even with the database, there is nothing to ensure a mobile number is going to a mobile phone these days. |
| 18:30.36 | p3nguin | For example, my Google Voice number is a mobile number. |
| 18:30.59 | *** join/#asterisk fskrotzki_ (~fskrotzki@cpe-74-74-245-250.rochester.res.rr.com) |
| 18:31.18 | *** join/#asterisk Alagar (~Administr@122.164.33.50) |
| 18:31.19 | leifmadsen | p3nguin: one of my VoIP numbers was a mobile phone number I ported a while ago |
| 18:31.25 | p3nguin | nod |
| 18:31.44 | *** join/#asterisk vgster (~vgster@94-194-190-189.zone8.bethere.co.uk) |
| 18:31.49 | FlaPer87 | I see |
| 18:31.51 | FlaPer87 | thanks |
| 18:32.01 | p3nguin | Within the past few years, knowing for sure that a landline number went to a landline phone and mobile to mobile was much more accurate. |
| 18:32.18 | *** join/#asterisk azertyuio (~chatzilla@bgl93-7-88-189-218-150.fbx.proxad.net) |
| 18:32.20 | p3nguin | I mean, prior to the last few years it was easier. |
| 18:32.21 | azertyuio | hi |
| 18:32.29 | azertyuio | i connect connect to asterisk |
| 18:32.37 | p3nguin | Within the last few, things have evolved too much. |
| 18:32.38 | Qwell | azertyuio: #freepbx. |
| 18:32.44 | *** join/#asterisk RobH (~robh@2620:0:860:2:21e:c2ff:fe03:2465) |
| 18:32.52 | azertyuio | noone there |
| 18:32.57 | Qwell | not our problem. |
| 18:33.02 | azertyuio | and also for me its logical |
| 18:33.04 | p3nguin | Doesn't mean we suddenly support it here. |
| 18:33.14 | azertyuio | the problem is from asterisk |
| 18:33.21 | leifmadsen | #asterisk is not #freepbx tier 2 support |
| 18:33.22 | p3nguin | Okay, show us. |
| 18:33.34 | azertyuio | that's why freepbx don't support ? |
| 18:33.35 | p3nguin | If it's an asterisk problem, maybe we can fix it. |
| 18:33.40 | Qwell | p3nguin: <td>Asterisk ERROR</td> |
| 18:33.47 | p3nguin | lol |
| 18:33.52 | Qwell | you think i'm kidding, sir. |
| 18:33.54 | p3nguin | That's a freepbx problem. |
| 18:33.59 | p3nguin | No, I believe you. |
| 18:34.09 | azertyuio | what i give the exact error ? |
| 18:35.52 | p3nguin | Provide an asterisk error and we might be able to help, provided that the error is not caused by freepbx. |
| 18:36.20 | p3nguin | You may paste the asterisk error into pastebin.com. |
| 18:36.29 | azertyuio | http://paste.ubuntu.com/419976/ this is what i got |
| 18:36.46 | p3nguin | Okay, asterisk it not started. |
| 18:36.53 | p3nguin | Reboot the computer and try again. |
| 18:37.09 | azertyuio | from one week i got this error |
| 18:37.20 | azertyuio | i reboot system several time |
| 18:37.30 | p3nguin | need more info |
| 18:37.36 | leifmadsen | means asterisk doesn't start up on reboot |
| 18:37.49 | Qwell | leifmadsen: freepbx starts it |
| 18:37.52 | leifmadsen | run "asterisk -c" and look at what is causing asterisk to fail to start |
| 18:38.12 | leifmadsen | Qwell: ah, then he's scuppered probably |
| 18:40.20 | azertyuio | plz wait |
| 18:41.18 | azertyuio | http://paste.ubuntu.com/419981/ |
| 18:41.27 | azertyuio | this what i got for asterisk -c |
| 18:42.32 | p3nguin | as root, run "namei -mx /var/run/asterisk/asterisk.pid" and paste the output in pastebin. |
| 18:42.58 | kannan | I have an autodialer app with asterisk ; i get a lot of circuit-busy errors from the SIP provider, this happens on and off , sometimes the autodialler goes great. Also, the problem , its only on UK numbgers, and USA numbers goes fine, the SIP service provider says it is a dialler problem, how can i start troubleshooting ? |
| 18:43.25 | azertyuio | are you sure for " namei - mx " |
| 18:43.28 | azertyuio | ? |
| 18:43.38 | azertyuio | my distro is ubutnu |
| 18:43.40 | azertyuio | ubuntu |
| 18:43.55 | p3nguin | sudo namei -mx /var/run/asterisk/asterisk.pid |
| 18:44.03 | *** join/#asterisk TimeRider (~steve@109.224.131.68) |
| 18:44.23 | azertyuio | namei: failed to stat: /var/run/asterisk/asterisk.pid: No such file or directory |
| 18:44.58 | p3nguin | hmm... even if the file doesn't exist, it should have output the permissions on each directory in the past. |
| 18:45.01 | p3nguin | path |
| 18:45.25 | azertyuio | so ? |
| 18:45.39 | azertyuio | what do you think ? |
| 18:45.47 | p3nguin | Check them each manually with ls -dl |
| 18:46.00 | azertyuio | everythings was working perfectly from 3 week |
| 18:46.09 | azertyuio | the problem come after reboot my system |
| 18:46.10 | p3nguin | but now it doesn't |
| 18:46.15 | azertyuio | yes |
| 18:46.20 | p3nguin | so you don't need to keep saying that it was working before. |
| 18:46.44 | azertyuio | ls -dl |
| 18:46.45 | azertyuio | drwx------ 6 root root 4096 Apr 19 12:55 . |
| 18:46.50 | p3nguin | sigh |
| 18:46.55 | p3nguin | ls -dl /var |
| 18:46.59 | p3nguin | ls -dl /var/run |
| 18:47.01 | p3nguin | ls -dl /var/run/asterisk |
| 18:47.14 | p3nguin | c'mon, man... throw me a friggin' bone, here. |
| 18:47.32 | Qwell | Ubuntu clears /var/run/ on boot. |
| 18:48.00 | p3nguin | adds that to his list of reasons to not use Ubuntu |
| 18:48.03 | leifmadsen | mkdir /var/run/asterisk |
| 18:48.39 | azertyuio | this is what i got http://paste.ubuntu.com/419986/ |
| 18:49.24 | p3nguin | do like leifmadsen suggested to create the dir. |
| 18:49.38 | azertyuio | ok |
| 18:49.50 | azertyuio | done |
| 18:49.55 | Qwell | WHO THE CRAP COLORIZED MY NANO?! DIAFCF UBUNTU |
| 18:50.02 | Qwell | wtf |
| 18:50.20 | p3nguin | Now start asterisk with asterisk -c like before. |
| 18:51.17 | hardwire | hmm. |
| 18:51.38 | Kobaz | nanobots did it |
| 18:52.09 | azertyuio | http://paste.ubuntu.com/419987/ |
| 18:52.14 | voxter | Im sad that there was a patch to remove colorized logging from asterisk logfiles and not make it an option :( It was nice to be able to tail / less in color when looking back through older logs. |
| 18:53.06 | leifmadsen | voxter: I had no idea such an application even existed to allow that |
| 18:53.17 | leifmadsen | voxter: I'd argue that is a regression! |
| 18:53.25 | p3nguin | Asterisk is not thread safe. ? Does this mean you need to run safe_asterisk or whatever it's called? |
| 18:53.40 | voxter | leifmadsen: it was enabled "by mistake" where the color codes were being written to the logfiles. tzafrir decided to remove it and not allow it as an option. |
| 18:53.41 | Slugs_ | Im trying to use grep to find an extension 8xxx, and i only want each match to appaer once |
| 18:54.23 | leifmadsen | voxter: can you submit an issue for that? I'd argue that is a regression to be honest |
| 18:54.25 | Naikrovek | voxter: it caused CPU spikes. the fix is to cure the spikes and leave it an option, IMO but of course I'm free to submit a patch |
| 18:55.14 | voxter | Naikrovek: oh really! Hmm. thats no good. maybe the code that caused cpu spikes could be looked at, but having color codes available when debugging is super handy. |
| 18:55.18 | azertyuio | what have i to do concratly ? |
| 18:55.24 | *** join/#asterisk lanning (~lanning@208.87.235.224) |
| 18:55.27 | azertyuio | excatly ? |
| 18:55.37 | Naikrovek | voxter: yeah someone was talking about it in here a couple days ago |
| 18:55.54 | Naikrovek | found out that it was the colorized logging option that was causing the cpu spikes |
| 18:56.32 | voxter | leifmadsen: the decision to remove it (and patch) is here: https://issues.asterisk.org/view.php?id=16786 |
| 18:56.42 | voxter | leifmadsen: whats the best course of action? should i do something about it, or? |
| 18:57.26 | voxter | i was wrong, it was tilghman not tzafrir |
| 18:57.30 | *** join/#asterisk aidinb (~Aidin@71-94-148-218.static.mtpk.ca.charter.com) |
| 18:57.31 | Kobaz | Naikrovek: really? |
| 18:57.47 | leifmadsen | voxter: well I don't particularly care that much about it (and would prefer no colour tags in the logs) but if it was something you found useful, that tends to get into the realm of a regression (loss of functionality). The thing to do is probably to file a new issue stating it should be configurable, and to link to the original issue. |
| 18:57.54 | Naikrovek | Kobaz: yeah |
| 18:57.58 | Naikrovek | searches his logs |
| 18:58.15 | Kobaz | wiggity |
| 18:58.19 | voxter | Naikrovek: if you could find that info ill submit it with my bug report |
| 18:58.20 | Kobaz | how much of a spike? |
| 18:58.38 | Naikrovek | Kobaz: don't remember off hand, looking it up now |
| 18:58.55 | azertyuio | how to run safe_asterisk ? |
| 18:59.17 | leifmadsen | /usr/sbin/safe_asterisk |
| 18:59.34 | p3nguin | hard stuff, there |
| 18:59.57 | *** join/#asterisk tzafrir_laptop (~tzafrir@bzq-218-155-148.cablep.bezeqint.net) |
| 19:00.13 | *** part/#asterisk FlaPer87 (~FlaPer87@unaffiliated/flaper87) |
| 19:00.25 | Kobaz | heh |
| 19:00.28 | Kobaz | indeed |
| 19:01.29 | azertyuio | http://paste.ubuntu.com/419991/ |
| 19:01.33 | azertyuio | this is it |
| 19:04.18 | azertyuio | so there is no solution for that ? |
| 19:05.21 | *** join/#asterisk sahafeez (~sahafeez@65-119-47-100.dia.static.qwest.net) |
| 19:08.07 | slawek | Hi everyone, I am a student and new to asterisk, I am trying to modify audio in Asterisk - i would like to hide some information in voice audio(using steganography) |
| 19:08.07 | slawek | while talking using PSTN network. How could I modify the audio that i being sent from one end point to another? I mean i would |
| 19:08.07 | slawek | like to be able only to modify one persons voice on one side and read that on the other side of the call, to look for information. |
| 19:08.07 | slawek | I have to boxes with asterisk connected by fxs/fxo interfaces. |
| 19:08.38 | slawek | i don't now how to start - I am looking for maybe some example code or info where to look for. |
| 19:08.42 | Naikrovek | Kobaz: still looking |
| 19:08.48 | [TK]D-Fender | azertyuio: You are running #freepbx . You do not use saf_asterisk. Continue your support in there. |
| 19:09.03 | leifmadsen | slawek: I think you want to look at JACK |
| 19:09.15 | leifmadsen | slawek: you can connect to JACK from Asterisk using app_jack.c |
| 19:09.27 | slawek | can i use jack to get audio from one person? |
| 19:09.41 | slawek | because i now checked the audiohooks |
| 19:09.56 | slawek | and i get the mixed audio in a channel (i think) |
| 19:09.59 | leifmadsen | slawek: potentially. I'm not sure how the implementation is in JACK, but if you want to manipulate audio outside of Asterisk, I think JACK is the right approach there |
| 19:10.36 | *** join/#asterisk mpe (~mpe@94.127.49.1) |
| 19:10.55 | slawek | well I can manipulate it in asterisk not necessarily in an another program |
| 19:11.06 | leifmadsen | slawek: looking at the documention in app_jack.c it seems to allow you to send input and output audio separately |
| 19:11.10 | slawek | i tried to write a dialplan function based on func_volume |
| 19:11.44 | leifmadsen | beyond that, this may be a question for #asterisk-dev if you're doing actual C coding |
| 19:12.03 | leifmadsen | passing encrypted data via steganography in phone calls seems really neat to me :) |
| 19:12.17 | slawek | oh i didn't know that there is such a group thanks :) |
| 19:12.22 | leifmadsen | :D |
| 19:12.24 | *** join/#asterisk jmacz (~jmacz@190.144.75.22) |
| 19:13.31 | Naikrovek | argh i can't find that conversation anywhere |
| 19:13.50 | Naikrovek | the word "color" is used a hell of a lot in the channels i lurk in |
| 19:14.03 | Naikrovek | but i KNOW someone had a cpu spike problem because of colorized logging a few days ago |
| 19:14.03 | azertyuio | how to install gnome on hosted server ? |
| 19:14.15 | leifmadsen | not an #asterisk problem |
| 19:14.36 | azertyuio | lol i m sorry |
| 19:14.37 | Naikrovek | azertyuio: same way you do it on a non-hosted server. but we don't know how and we can't help you with that |
| 19:14.37 | p3nguin | naikrovek: Did you make sure you tried the variation "colour" as well? |
| 19:14.45 | Naikrovek | p3nguin: ooh |
| 19:14.52 | *** part/#asterisk azertyuio (~chatzilla@bgl93-7-88-189-218-150.fbx.proxad.net) |
| 19:14.54 | *** join/#asterisk Mango (~iMango@d154-20-86-138.bchsia.telus.net) |
| 19:15.06 | Mango | vi /etc/asterisk/sip.conf |
| 19:15.09 | Mango | whoops |
| 19:15.29 | *** join/#asterisk diegomad (~mad@190.146.200.120) |
| 19:15.57 | Naikrovek | p3nguin: no good :/ |
| 19:16.11 | ecrane | Mango: Could have been worse, could have been rm /etc/asterisk/sip.conf |
| 19:16.28 | Mango | :) |
| 19:16.34 | jaytee | i remember that conversation about the color text in the CLI but I can't remember the date it took place. |
| 19:16.41 | Mango | Trillian insists on popping up the chat window when I log in. |
| 19:16.47 | jaytee | otherwise you could search on rikers.org |
| 19:16.59 | Naikrovek | jaytee: i have the logs but i can't find it |
| 19:17.12 | jaytee | it was over a week ago |
| 19:17.20 | Naikrovek | i was gonna say it was about a week ago |
| 19:17.27 | Naikrovek | either way i can't find it with grep |
| 19:17.31 | Naikrovek | will have to READ oh the horror |
| 19:17.49 | jaytee | rikers.org logs this chat and lets you pick by date but doesn't let you search |
| 19:18.03 | *** join/#asterisk atis_work (~atis_work@193.238.212.171) |
| 19:20.57 | Naikrovek | aah |
| 19:21.49 | Naikrovek | not in my logs because it happened over a weekend. |
| 19:21.49 | Naikrovek | http://ibot.rikers.org/%23asterisk/20100410.html.gz |
| 19:21.49 | voxter | Hmmm. |
| 19:21.49 | Naikrovek | granpapadot was having the issue |
| 19:21.49 | Naikrovek | TJNII was the one who mentioned the CPU spikes |
| 19:22.36 | Naikrovek | yes that's it |
| 19:22.39 | Naikrovek | some color option |
| 19:22.54 | voxter | damn, i forgot to click advanced and tag it as a regression, and now i cant edit the issue? lame |
| 19:23.01 | Mango | note to self |
| 19:23.08 | Mango | my wife is not impressed by Playback(tt-monkeys) |
| 19:23.17 | Naikrovek | conversation about color options continued here: http://ibot.rikers.org/%23asterisk/20100411.html.gz |
| 19:23.29 | Get_The_Fish | so, is it helpful to include the full log of the asterisk startup in a bug report if it's not related to the startup of asterisk? I was thinking it would show whoever is looking the modules that are loaded, but it might just be spam.. what do you think? |
| 19:23.37 | Naikrovek | a color option turned on caused cpu spikes |
| 19:24.21 | Get_The_Fish | I think a helpful CLI feature would be something like "show tech support", which spits out useful info such as the version, loaded modules, and whatever else may be helpful |
| 19:25.31 | voxter | Naikrovek: thanks for the URL. Am i understanding this correctly that if asterisk is launched in color CLI mode at ALL (let alone colorized logfile writing) that this happens? |
| 19:25.40 | leifmadsen | the command would actually be 'core show tech support' :) |
| 19:25.56 | Naikrovek | voxter: ask granpapadot or whatever his name is when/if he shows up |
| 19:26.09 | leifmadsen | ~seen grandpapadot |
| 19:26.18 | infobot | grandpapadot <~nonya@99-175-248-81.lightspeed.brhmal.sbcglobal.net> was last seen on IRC in channel #asterisk, 18h 57m 12s ago, saying: '.. but that looks like my option, lol, DANGIT!'. |
| 19:26.34 | Naikrovek | must be a different timezone guy |
| 19:26.40 | Naikrovek | i rarely see him anymore |
| 19:26.41 | Get_The_Fish | lol yeah, something like that |
| 19:26.57 | jaytee | I was just reading the logs and it seems to only have happened when * was initialized in the init script |
| 19:27.03 | Get_The_Fish | I think it might be helpful in quite a few instances |
| 19:27.21 | Naikrovek | jaytee: if it's a bug with how ansi colors are handled it won't be limited to startup |
| 19:27.29 | Naikrovek | jaytee: but yes you're right |
| 19:27.33 | Naikrovek | that's where his problem was |
| 19:27.37 | voxter | jaytee: yeah, when some sort of COLOR variable is set.. but, isnt everyone's CLI color enabled by default? |
| 19:28.16 | jaytee | at the beginning of the next days logs grandpapadot mentions #COLOR=yes |
| 19:28.25 | Naikrovek | yeah the 11th |
| 19:28.33 | jaytee | I've never had spikes on any of my * boxes from the color option being set |
| 19:28.37 | Naikrovek | according to infobot's sense of time |
| 19:29.24 | jaytee | grandpapadot said he was going to look in mantis and submit a bug report if it wasn't already in there. |
| 19:29.46 | *** join/#asterisk MmixX (mmixx@61.14.191.143) |
| 19:31.01 | Naikrovek | this is some #asterisk cooperation right here |
| 19:31.07 | Naikrovek | this color thing |
| 19:31.11 | Get_The_Fish | so, would a full log of the asterisk startup be helpful in a bug report, if the bug in question isnt related to a module not loading or something else startup related, or would it just be more crap to sift through? |
| 19:31.12 | *** join/#asterisk jeffgus (~jeffgus@2002:ad33:b504::1) |
| 19:31.13 | Naikrovek | maybe it's for real, maybe it's not |
| 19:31.29 | jaytee | http://lists.digium.com/pipermail/asterisk-bugs/2010-March/072279.html |
| 19:31.36 | voxter | Im so confused, this COLOR variable is not set on centos' stuff, but when it is set on debian, what does it do differently? |
| 19:32.03 | voxter | well. that answers that question. |
| 19:32.22 | Naikrovek | that's a debian bug, not asterisk |
| 19:32.28 | Naikrovek | and a lame bug at that |
| 19:32.41 | leifmadsen | Get_The_Fish: a module not loading on startup is rarely a bug in Asterisk -- it's usually a load order thing that can be corrected in modules.conf |
| 19:32.45 | voxter | well, its not a debian bug per se |
| 19:33.01 | voxter | its that debian assumed that passing -c to asterisk meant "color" when it means "console" |
| 19:33.12 | voxter | it actually has nothing to do with color at all. |
| 19:33.13 | Naikrovek | well that script is part of the asterisk package provided by debian repositories |
| 19:33.15 | jaytee | no, it's an init script bug and leif already posted a fix for it |
| 19:33.32 | Naikrovek | okay |
| 19:33.33 | voxter | The actual issue is that debian tries to request an attached console to asterisk in an init script. |
| 19:33.38 | leifmadsen | 'make config' with the latest checkout of a branch |
| 19:33.42 | voxter | it just so happens that the variable is "called" COLOR |
| 19:33.47 | Get_The_Fish | right right, but I am asking if showing the full asterisk startup is helpful for a run of the mill bug report in something that occurs after a startup... I was thinking that people seeing all the modules loaded, conf files, etc might be helpful to at least have on hand |
| 19:34.02 | leifmadsen | Get_The_Fish: not typically useful |
| 19:34.33 | Get_The_Fish | gotcha, just more crap to sift through... ok, just checking |
| 19:34.35 | bmoraca_work | mmmm...deli cut roast beef on whole wheat ciabata bread with spicy mustard and pickles...mmmm |
| 19:35.30 | voxter | leifmadsen: can you mark 17222 as a regression please? I missed it when filling out the form for the first time and cant find how to edit it, i dont believe i can. |
| 19:35.42 | voxter | leifmadsen: or is that something that you only do once you decide it is infact a regression.... :P |
| 19:36.28 | leifmadsen | voxter: done |
| 19:36.40 | Naikrovek | i wanna learn to park a car like this: http://i.imgur.com/2KhIt.gif |
| 19:36.48 | manxpower | "220 packages / 246MB to update, continue?" I think someone has not updated their system in a while. |
| 19:37.42 | bmoraca_work | Naikrovek: www.thatwillbuffout.com |
| 19:38.17 | Naikrovek | bmoraca_work: awesome |
| 19:39.41 | bmoraca_work | Naikrovek: learn to park like this and I'll give you a cookie: http://thatwillbuffout.com/2010/04/18/funny-car-photos-telephone-poles/ |
| 19:39.49 | voxter | leifmadsen: thanks! I also asked murf about fixing the parking lot thing, i think he might look at it, but not for quite a while. anyone else you know looking to fix this and finally get real multiple parking lots working? the feature is pointless without it almost, and its SO useful! |
| 19:40.08 | Naikrovek | bmoraca_work: i love the title of that post: "aww, he thinks he's telephone poles" |
| 19:40.17 | leifmadsen | Naikrovek: weird, that link must be going around everywhere today |
| 19:40.28 | *** join/#asterisk oej (~olle@ns.webway.se) |
| 19:40.29 | Naikrovek | leifmadsen: reddit.com |
| 19:40.35 | leifmadsen | ah |
| 19:40.35 | Naikrovek | that's where i saw it |
| 19:40.43 | Naikrovek | lots and lots and lots of people frequent that site |
| 19:40.51 | bmoraca_work | Naikrovek: www.thereifixedit.com is another great site |
| 19:41.02 | p3nguin | I like that one. |
| 19:41.14 | p3nguin | kludges everywhere. |
| 19:43.47 | Naikrovek | wow |
| 19:46.29 | *** join/#asterisk Naikrovek (~jjohnson@unaffiliated/naikrovek) |
| 19:46.43 | *** join/#asterisk snapple42 (~snapple42@h216-18-80-131.gtconnect.net) |
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| 20:06.39 | Peste_Bubonica | Some can Help me? I have two servers connected via IAX2. When I try to access a voicemail(or another application that uses playbacks) from a server to another, via a SIP Phone, the call always stalls, in the middle of the dialplan. Somethings is played, then, the call stalls, and I need to hangup... If I connect this SIP Phone directly to the other server, using a vpn for example, I can access the voicemail normally |
| 20:07.21 | *** join/#asterisk binbash_ (~peter@ip4da53781.direct-adsl.nl) |
| 20:07.32 | Peste_Bubonica | Ive tryied to connect the boxes using sip to, for tests, and the call stalls two. only works, if I connect the SIP Phone directly to the server that I want to talk. I can bridge the calls with my localservers |
| 20:08.00 | *** join/#asterisk QaDeS (~mklaus@p54A1AD95.dip0.t-ipconnect.de) |
| 20:09.04 | *** join/#asterisk Defraz (~tim@69.1.183.94) |
| 20:09.56 | bent_screwdriver | call parking internally: when user A calls user B and user B parks the call it stops in the dialplan. If user A parks a call they made it continues through the dail plan, onto the next priority. Is this normal/expected? |
| 20:11.05 | bent_screwdriver | * 1.6.2 |
| 20:12.44 | hardwire | anybody ever found a DECT dual-line headset? |
| 20:15.40 | Mango | I can start Asterisk from the command line but not via cron. The cron log says it started, but it didn't. Any ideas? |
| 20:16.18 | fenrus | check the logs. |
| 20:16.24 | fenrus | why would you want to start asterisk from cron ? |
| 20:16.53 | leifmadsen | that seems like the wrong approach... :) |
| 20:16.59 | ecrane | fenrus: lol.. good question |
| 20:17.11 | fenrus | ;) |
| 20:17.37 | fenrus | i'd start it with init-scripts and perhaps use cron as keepalive.. |
| 20:17.43 | paulc | isn't that why we have safe_asterisk script? |
| 20:18.37 | Mango | fenrus, ok, tell me more please :) |
| 20:19.03 | fenrus | Mango, what distro do you use ? |
| 20:19.06 | *** part/#asterisk Get_The_Fish (~Get_The_F@c-24-8-50-199.hsd1.co.comcast.net) |
| 20:19.09 | Mango | CentOS |
| 20:19.12 | *** join/#asterisk farkus_ (chatzilla@cpe-72-225-212-219.nyc.res.rr.com) |
| 20:19.33 | fenrus | Mango, try reading http://www.voip-info.org/wiki/view/Asterisk+Starting+and+Stopping |
| 20:19.38 | Mango | THanks! |
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| 20:33.15 | *** join/#asterisk raden_work (~jon@69-179-99-17.stat.centurytel.net) |
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| 20:34.09 | raden_work | can someone tell me what " Operator Service Provider" means on a telco bill Calls 4 Amount $ 169.80 |
| 20:34.50 | Qwell | raden_work: "operator" |
| 20:35.00 | raden_work | why would it be $170 ? |
| 20:35.10 | Qwell | raden_work: note the quotes. |
| 20:35.13 | Qwell | "operator" |
| 20:35.32 | Qwell | Sounds like a premium-toll call. |
| 20:35.52 | raden_work | its under usage charges |
| 20:35.57 | raden_work | 0 mins |
| 20:36.01 | raden_work | qty 4 |
| 20:36.05 | raden_work | $169.80 |
| 20:36.21 | jaytee | "It's a trap!" |
| 20:36.32 | pabelanger | raden_work: Ask your telco? |
| 20:36.36 | paulc | "And I'd have gotten away with it - if it wasn't for those pesky kids!" |
| 20:38.58 | *** join/#asterisk farkus_ (chatzilla@cpe-72-225-212-219.nyc.res.rr.com) |
| 20:40.24 | ecrane | raden_work: Nobody dialed '0' and asked an operator to help them make a call, did they? Voicemail hacker? |
| 20:40.44 | raden_work | i looked it up |
| 20:41.00 | raden_work | its basically long distance service through a non traditional company |
| 20:41.20 | Mango | like 101-55-66 or somesuch? |
| 20:43.37 | jaytee | raden_work, does it show what numbers were dialed? |
| 20:44.14 | *** join/#asterisk Failrar (~Failrar@5ED66E6D.cable.ziggo.nl) |
| 20:48.09 | raden_work | jaytee nothing just says operator service provider min: 0 amount " 169.80 |
| 20:49.27 | jaytee | raden_work, I'd dispute it with your telco then. |
| 20:50.02 | raden_work | its a customer who wanted to give us a quote for a VOIP system |
| 20:50.09 | raden_work | alot of charges ive never seen before |
| 20:50.27 | raden_work | 10 lines costing them over 1500 a month with less than 3000 min of outbound volume |
| 20:53.10 | jaytee | raden_work, I'm confused. this is one of their bills or one of yours? |
| 20:53.22 | raden_work | their bills |
| 20:53.41 | raden_work | just all sorts of odd charges on it |
| 20:53.47 | jaytee | so they're being overcharged then |
| 20:54.14 | raden_work | 1b LN for unlimited EAS 2 @ 49.95 |
| 20:54.20 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.218.194) |
| 20:54.29 | raden_work | Business EAS 10 @ 19.75 |
| 20:54.33 | p3nguin | There's no need to blame it on [tk]d-fender |
| 20:54.43 | p3nguin | Oh, hi there, [tk]d-fender. |
| 20:54.44 | raden_work | yes there is |
| 20:55.20 | jaytee | back before Worldcom went bust a company I worked at had merged with a company in New Jersey that had MCI Worldcom service. I had to go over their bills and they were getting charged $5 a minute for a fax call from NJ to Indiana. |
| 21:01.42 | *** join/#asterisk Tim_Toady (~moi@77.49.29.230.dsl.dyn.forthnet.gr) |
| 21:06.34 | *** join/#asterisk miamiseb (~seb@c-75-74-27-128.hsd1.fl.comcast.net) |
| 21:06.59 | idespinner | is the asterisk-gui project dead or still going strong? |
| 21:07.06 | p3nguin | death to gui |
| 21:07.31 | idespinner | the project or just in general? |
| 21:07.43 | miamiseb | Hi all, I've got ztdummy loaded but when I try to join a conference I get WARNING[10516]: app_meetme.c:1097 build_conf: Unable to open pseudo device. Asterisk 1.6.2.6 |
| 21:08.27 | p3nguin | miamiseb: You need a timging device that works with your version of Asterisk. Remove zaptel and install dahdi, then load dahdi_dummy rather than ztdummy. |
| 21:08.35 | miamiseb | ah ha. |
| 21:08.46 | p3nguin | timing, that is |
| 21:09.00 | miamiseb | Must I recompile dahdi or is that module built already? modprobe dahdi_dummy = no dice |
| 21:09.11 | [TK]D-Fender | idespinner: Cryogenics w/ freezer-burn |
| 21:09.20 | p3nguin | You'll need dahdi installed before you compile asterisk. |
| 21:09.54 | miamiseb | I used a iso similar to asterisknow, thirdlane, which I'm pretty sure has dahdi built in. |
| 21:10.39 | p3nguin | Then I have no idea what else to tell you. Maybe someone else is interested. |
| 21:11.02 | idespinner | i'm still a little lost on the answers, are you telling me its just on hold for a really long time and may never come back? |
| 21:11.03 | miamiseb | lol |
| 21:11.43 | miamiseb | The dahdi modules were built for previous versions of the kernel, but not in /lib/modules for this kernel version, maybe I can find that source laying around or just recompile em myself |
| 21:11.55 | miamiseb | thanks anyway for pointing me in the right direction p3nguin |
| 21:12.20 | p3nguin | Sounds like you know how to get there from here. :) |
| 21:15.08 | miamiseb | Yup, compile and installed for latest version and now getting /lib/modules/2.6.18-164.11.1.el5/dahdi/dahdi_dummy.ko lsof doesn't show it in use though. |
| 21:15.13 | miamiseb | goes to search |
| 21:15.27 | miamiseb | Ermm getting FATAL: Error inserting dahdi_dummy (/lib/modules/2.6.18-164.11.1.el5/dahdi/dahdi_dummy.ko): Device or resource busy |
| 21:15.40 | p3nguin | Usually you can update via package manager and get things of compatible versions. |
| 21:16.45 | miamiseb | rmmod ztdummy and zaptel followed by the modprobe worked. Yeah, I've got yum keeping me up to date but couldn't find the packages for zaptel or dahdi there even though I've got the asterisk-current repo |
| 21:16.50 | miamiseb | shrugs. |
| 21:18.13 | *** join/#asterisk Torrieri (~Torrieri@nelug/crew/torrieri) |
| 21:19.25 | *** join/#asterisk putnopvut (~putnopvut@asterisk/master-of-queues/mmichelson) |
| 21:19.25 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 21:24.14 | *** join/#asterisk VaGoNeTaS (~cchavez@200.111.138.170) |
| 21:24.20 | VaGoNeTaS | hello |
| 21:24.23 | VaGoNeTaS | every1 |
| 21:24.57 | VaGoNeTaS | who knows which is the package required for cdr_pgsql ?? |
| 21:25.09 | VaGoNeTaS | i thought it was pslib-dev but it wasnt |
| 21:25.32 | p3nguin | yum search pgsql |
| 21:27.39 | VaGoNeTaS | libpq-dev |
| 21:27.43 | VaGoNeTaS | that's the ne |
| 21:27.45 | Katty | collapses somewhere |
| 21:27.46 | VaGoNeTaS | *one |
| 21:28.11 | miamiseb | Bah, I'm getting WARNING[14211]: app_meetme.c:1097 build_conf: Unable to open pseudo device even though dahdi_test can open the psuedo device and dahdi_dummy module is loaded, any other ideas? |
| 21:28.29 | Katty | what a day |
| 21:30.11 | miamiseb | dahdi show status also shows the dahdi_dummy but has an alarm as unconfigured, is that normal? |
| 21:30.26 | miamiseb | Katty, difficult times? |
| 21:30.34 | *** join/#asterisk cosmicwombat (~cosmicwom@69.7.44.68) |
| 21:31.08 | Katty | no, not difficult |
| 21:31.14 | Katty | just tedious |
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| 21:31.41 | *** part/#asterisk ehsjoar (~ehsjoar@c-98-245-155-132.hsd1.co.comcast.net) |
| 21:36.46 | *** join/#asterisk devoid (yiffstar66@unaffiliated/devemo) |
| 21:46.17 | jaytee | hi Katty |
| 21:47.34 | *** join/#asterisk citywok (~chatzilla@67-134-194-33.dia.static.qwest.net) |
| 21:47.56 | citywok | does anybody have a good recommendation for wholesale LD? 200k/mi/mo. Tired of bandwidth.com's idiocy |
| 21:51.21 | leifmadsen | odd, I always had good luck with bandwidth.com, but I do very simple things for that customer |
| 21:51.25 | leifmadsen | citywok: Level 3? |
| 21:51.36 | leifmadsen | or do they require a million minutes a month now? |
| 21:52.05 | leifmadsen | or you could try Global Crossing... |
| 21:53.04 | citywok | Hmm. I've only worked with VP, flowroute, and bandwidth.com. bandwidth.com has been the cheapest & best call quality, but as a company they suck. Now i have to prefix 011 in front of all calls to hawaii, alaska, and puerto rico. |
| 21:53.31 | citywok | i'll check in to L3 & GC. thanks for the suggestions |
| 21:55.41 | miamiseb | bandwidth is the largest level 3 reseller, so your already with level3 indirectly, if only because you won't meet the ridiculous minimum commitment with lvl3 |
| 21:56.11 | citywok | bandwidth has horrible customer service and i'm tired of being dicked around by them |
| 21:56.23 | *** join/#asterisk dennisG (~dennisG@2002:541e:88d0:0:213:2ff:fe56:e380) |
| 21:56.54 | citywok | but the call quality has been much better than i've gotten with VP or flowroute |
| 21:57.02 | miamiseb | So if anyone is familiar with meetme in 1.6, I should really only have to install and get working dahdi_dummy (because I have no real timing) and make sure that chan_dahdi loads and shows the dummy channel right? |
| 21:57.36 | *** join/#asterisk s4msung (~s4msung@dice.s4msung.de) |
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| 22:02.01 | *** part/#asterisk dennisG (~dennisG@2002:541e:88d0:0:213:2ff:fe56:e380) |
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| 22:07.20 | raden_work | how does one go about getting in a phone book or having the correct phone number on the internet if they use VOIP ? |
| 22:08.05 | hardwire | you pay them |
| 22:08.41 | bmoraca_work | wow |
| 22:08.53 | bmoraca_work | func_odbc does NOT like it when your ODBC user doesn't have privs |
| 22:09.04 | bmoraca_work | catastrophic asterisk fail! |
| 22:09.24 | hardwire | segfawlt? |
| 22:09.32 | bmoraca_work | possibly |
| 22:12.02 | bmoraca_work | my brilliant cnam caching uses two queries and that's it! it's excellent! |
| 22:16.41 | *** join/#asterisk tkrn (~tkrn@WS1-DSL-208-102-253-13.fuse.net) |
| 22:19.24 | raden_work | is there a phone that has a line indicator for like 10 extensions ???? |
| 22:19.32 | raden_work | without having a addon caddy on the side ? |
| 22:19.38 | p3nguin | Phones don't care about extensions. |
| 22:20.04 | *** join/#asterisk blaines (~blaines@75-171-88-163.phnx.qwest.net) |
| 22:20.24 | p3nguin | unless you're talking about BLF |
| 22:21.39 | *** join/#asterisk blaines (~blaines@75-171-88-163.phnx.qwest.net) |
| 22:21.59 | *** join/#asterisk `paul (~paul@112.201.212.74) |
| 22:22.04 | p3nguin | blaines: C'mon, dude, take us off auto-join until you can fix your client. |
| 22:22.05 | raden_work | yea BLF |
| 22:22.21 | raden_work | like need to know if 100-110 is on the phone |
| 22:22.31 | raden_work | is there a LCD phone or a phone that has 10 BLF's ? |
| 22:22.33 | p3nguin | then the answer is no. You would need a lamp for every buddy that you want to subscribe to. |
| 22:22.49 | `paul | if i have an old versioned (1.4) asterisk built from source and i want to update it to latest 1.4 whats the proper way to do it? |
| 22:22.59 | p3nguin | Oh, there could be one with 10 on it, but I don't know the model. |
| 22:23.08 | miamiseb | The cisco had a addon, that can do it |
| 22:23.14 | bmoraca_work | raden_work: Adtran makes a SIP phone with 12 line apperance buttons that can be used for BLF |
| 22:23.16 | miamiseb | lemme go look at my model number |
| 22:23.25 | p3nguin | 7914 |
| 22:23.32 | bmoraca_work | not usable with SIP firmware |
| 22:23.35 | p3nguin | and he wants to do it without a sidecar. |
| 22:23.39 | miamiseb | 7914 |
| 22:23.59 | miamiseb | bah |
| 22:24.20 | p3nguin | There could be some phone out there that has 10+ lamps. |
| 22:24.21 | raden_work | how many BLF ? |
| 22:24.31 | blaines | ? |
| 22:24.45 | blaines | how many times did I connect? |
| 22:25.06 | p3nguin | blaines: This time, just once. |
| 22:25.20 | p3nguin | blaines: Usually 20-100 times, though. |
| 22:25.29 | miamiseb | Or just use something that is completely digital and scroll up down |
| 22:25.54 | blaines | p3nguin: i dunno what there is to fix... but the connection is flaky depending on where I'm at |
| 22:25.59 | bmoraca_work | raden_work: Adtran has a phone with 12 line buttons that can optionally be used for BLF. try that. |
| 22:26.10 | raden_work | im looking at them all |
| 22:27.20 | bmoraca_work | lol |
| 22:27.30 | bmoraca_work | "POLYCOM INC" just called one of my customers |
| 22:30.21 | miamiseb | from google: Max sendq exceeded is when the server is flooding the client, not the |
| 22:30.21 | miamiseb | <PROTECTED> |
| 22:30.21 | miamiseb | using this mechanism. Doing a /who on a channel with 500 users is likely |
| 22:30.21 | miamiseb | to cause this |
| 22:30.33 | miamiseb | errm, sorry for the scroll |
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| 22:52.27 | *** join/#asterisk The_Boy_Wonder (~vossel@asterisk/batman-developer/dvossel) |
| 22:59.01 | miamiseb | Have a good night all. |
| 23:03.05 | *** join/#asterisk jks (jks@193.189.93.254) |
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| 23:05.01 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
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| 23:05.47 | *** part/#asterisk GameGamer43|Mac (~GameGamer@cpe-74-65-36-91.rochester.res.rr.com) |
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| 23:26.43 | citywok | my aastra 6757i has 12 buttons that can do BLF |
| 23:27.22 | *** join/#asterisk parim (~pari@chlorine.dhcp.rose-hulman.edu) |
| 23:27.54 | parim | hey guys, how well does asterisk run on the wrt54g? |
| 23:30.22 | p3nguin | parim: Try it and let us know. |
| 23:31.49 | parim | ok p3nguin, i will |
| 23:32.05 | parim | this is my first trial with asterisk |
| 23:32.26 | p3nguin | Wouldn't a normal computer be a better idea for a novice? |
| 23:33.15 | parim | i am going to try it out on a gentoo machine and if i get that to work then i am going to switch to the wrg54g |
| 23:33.26 | parim | sorry wrt54g |
| 23:33.54 | [TK]D-Fender | parim: You mean on a device that aspires to the power of my wrist-watch? Yeah.. thats a fair testing ground.... |
| 23:34.52 | TJNII | I know Gentoo is kind of the red-headed stephild of the Linux world, but comparing it to a wrtist watch is a bit harsh. |
| 23:35.59 | parim | TJNII: i have been using gentoo as my primary machine for about 2years now, |
| 23:36.39 | *** join/#asterisk astrutt (astrutt@pinky.ratman.org) |
| 23:41.28 | *** join/#asterisk tzafrir (~tzafrir@212.179.75.202) |
| 23:49.00 | citywok | i think he was comparing the wrt54g to a wristwatch |
| 23:49.08 | citywok | b/c it has a 200mhz processor that can barely do 30mbit |
| 23:49.31 | citywok | 15mbit of traffic and i bet the call would be choppy just b/c the proc would be peaking |
| 23:50.27 | Naikrovek | bmoraca_work: why would polycom call one of your customers i wonder |
| 23:50.41 | citywok | b/c they are advertising on an unrelated matter? |
| 23:50.51 | citywok | i got called by polycom a few weeks ago... |
| 23:51.43 | manxpower | ~answers |
| 23:51.43 | infobot | somebody said answers was Asterisk Book: http://astbook.asteriskdocs.org & SIP w/ NAT: http://www.aocomputing.net/?p=3 & Wiki: http://www.voip-info.org & docs: /path/to/src/asterisk/doc & 1.4 .vs. 1.6.0 .vs. 1.6.1: : /path/to/src/asterisk/UPGRADE*.txt |
| 23:54.47 | parim | is the spa-3000 still the cheapest device with both FXS and FXO ? |
| 23:58.19 | *** join/#asterisk JAMMAN2110 (~james@unaffiliated/jamman2110) |
