| 00:10.51 | *** join/#asterisk fauxalliance (~fauxallia@142.163.152.120) |
| 00:11.44 | *** join/#asterisk jaytee (~jforde051@unaffiliated/jaytee) |
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| 00:32.21 | Maxus2 | hi asterisk people, was the guy who wrote the "Asterisk: The Definitive Guide" in this channel? |
| 00:42.13 | jaytee | actually two of the authors are in this channel |
| 00:43.15 | Maxus2 | just trying to under stand in the dundi config this line: ${IF($[${DB_EXISTS(phones/${NUMBER}/device)}]?${DB(phones/${NUMBER}/device)}:None)} |
| 00:43.25 | Maxus2 | why is it being added to the mapping? |
| 00:43.58 | Maxus2 | is it to convert the number to the device? |
| 00:44.10 | jaytee | dunno, don't have a copy of the new book yet |
| 00:44.32 | Maxus2 | http://ofps.oreilly.com/titles/9780596517342/asterisk-CHP-5.html |
| 00:44.36 | Maxus2 | you can read it there |
| 00:47.57 | *** join/#asterisk guilhermebr (~Guilherme@187.116.201.171) |
| 00:48.29 | jaytee | Maxus, just below that line is a fairly straightforward answer as to what that line does. |
| 00:49.10 | Maxus2 | hi jaytee, saw that, but why is it being done? |
| 00:49.51 | Maxus2 | just so "none" can be returned when a device isn't there? |
| 00:51.30 | jaytee | or to return the actual number if it is in the database |
| 00:53.32 | Maxus2 | but surely if your checking a number that might be on another machine that would stop the request from going through as the number wouldn't be in the local machines dataabase? |
| 00:55.10 | jaytee | well, I don't use DUNDI and I haven't read through that whole chapter. |
| 00:55.46 | jaytee | not sure if leif is actually available or just logged in and away from his computer |
| 00:56.06 | jaytee | he would be able to explain the logic behind the examples in the book. |
| 00:56.56 | Maxus2 | cool, was more directing it to the authors more than anything |
| 00:56.57 | Maxus2 | :) |
| 01:00.26 | *** join/#asterisk killown (~killown@unaffiliated/killown) |
| 01:00.26 | jaytee | they're usually pretty active in here but then again they're human, not cylon so they have to eat and do other stuff sometime. |
| 01:01.49 | Maxus2 | haha, yeah :) |
| 01:02.03 | Maxus2 | we need more cylon devs. |
| 01:02.24 | jaytee | as long as they look like 6 I'm down with it |
| 01:02.42 | Maxus2 | haha that makes 2 of us. |
| 01:03.21 | jaytee | although I did like the one skin job played by Lucy Lawless. |
| 01:04.16 | Maxus2 | yeah, most of the female cylons were hotties. |
| 01:04.23 | jaytee | when she played Xena I used to wish the show was X rated and she'd end up getting it on with her sidekick whose name I can't remember. |
| 01:04.31 | Maxus2 | looking forward to the new series |
| 01:04.41 | Maxus2 | oh yeah the side kick was tidy. |
| 01:04.43 | jaytee | Caprica? it's been good so far |
| 01:04.50 | Maxus2 | haven't thought about that show in ages. |
| 01:04.55 | jaytee | tidy? hmmmm, that's a cool expression |
| 01:05.02 | Maxus2 | capricas cancelled :( |
| 01:05.13 | jaytee | what new series then? |
| 01:05.18 | Maxus2 | there is a new one, chrome and metal i think |
| 01:05.25 | Maxus2 | chrome and blood sorry |
| 01:05.39 | Maxus2 | http://en.wikipedia.org/wiki/Battlestar_Galactica:_Blood_%26_Chrome |
| 01:05.47 | jaytee | cool! |
| 01:05.52 | Maxus2 | yeah :) |
| 01:06.02 | jaytee | wish I had HBO, I can't watch Game of Thrones :-( |
| 01:06.15 | Maxus2 | haven't seen that oen |
| 01:07.09 | jaytee | wow, that's an awesome concept for a spinoff series. I'll be lookin forward to that. |
| 01:07.20 | jaytee | have you ever watched Firefly? |
| 01:07.22 | Maxus2 | yeah looks good. |
| 01:07.37 | Maxus2 | shame about caprica, it was just getting good |
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| 01:52.05 | zeropoint46 | Fauxalliance, you around? |
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| 02:35.37 | Maxus2 | is there a way in an if statement to check if something is connected before dialing it? |
| 02:36.01 | *** join/#asterisk Defraz (~Defraz@184-155-137-27.cpe.cableone.net) |
| 02:37.51 | kaldemar | Maxus2: if you're still on DUNDi, take a few steps back and look at regexten. |
| 02:39.31 | Maxus2 | yep im using that |
| 02:40.02 | Maxus2 | my problems is i have a set of out goign connections, but i need to check if one is valid and connected, if its not default to the next int he list |
| 02:40.28 | Maxus2 | im using realtime and im not allowed to put anything into the config files all has to be in the db. |
| 02:40.46 | Maxus2 | been struggling with this for a coupel weeks now :( |
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| 02:43.47 | *** join/#asterisk kaushal (~kaushal@182.72.14.170) |
| 02:43.53 | kaushal | hi |
| 02:43.57 | kaushal | Just curious to know what is CRC4 and NCRC4 |
| 03:02.22 | kaldemar | Maxus2: i'm not sure i fully understand your issue, but the IF is probably intended for use in a case where a query is likely to return multiple results. |
| 03:02.51 | kaushal | Anyone around here ? |
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| 03:04.28 | ideaman | Anyone: what is the easiest way to record outbound calls? |
| 03:04.45 | Maxus2 | Hi kaldemar, we have a collection of dundi boxes acting as a cluster, each node has a different provider connected, i want to check if a given provider hasn't failed before placing a call, through it. its complex and a bit hard to explain |
| 03:07.39 | ideaman | anyone...? |
| 03:08.06 | *** join/#asterisk suxx (~Oleg@113486930.convex.ru) |
| 03:08.39 | suxx | hi all! |
| 03:12.41 | kaldemar | Maxus2: ideally you either need something to monitor that or do the checking in the originating box. |
| 03:13.52 | kaldemar | ideaman: use app MixMonitor in your dialplan. |
| 03:30.11 | ideaman | thanks |
| 03:31.28 | Maxus2 | kaldemar, how would i check from originating box? |
| 03:31.50 | teloniusz | hi guys. I'm doing stress tests on my asterisk with perl sip clients |
| 03:32.29 | teloniusz | and after some time with ~100 sip channels used it appears that there are active connections on my asterisk which are not active on the caller |
| 03:33.07 | teloniusz | there's voice menu running on the asterisk side |
| 03:34.14 | Maxus2 | perhaps a missing hangup, im only guessing? |
| 03:34.39 | teloniusz | there seems to be rtp communication, but no sip, tshark shows only 'port unreachable' responses to my asterisk OPTIONS requests |
| 03:35.05 | teloniusz | Maxus2: maybe, but it's only sometimes |
| 03:35.28 | Maxus2 | what version are you using? |
| 03:35.35 | Maxus2 | or asterisk that is? |
| 03:35.36 | teloniusz | and my problem is, how to configure asterisk to discard such connections? This totally messes up my billing |
| 03:35.44 | teloniusz | 1.6.2, debian package |
| 03:37.11 | Maxus2 | hmm sorry not sure what the answer is, still learning myself. |
| 03:38.10 | teloniusz | and it seems taht it's not only a problem with my caller script, sometimes I see external connections with duration ~10 hours |
| 03:38.32 | kaldemar | Maxus2: by dialing unless you use something custom. autokill in iax and timers in sip would allow it to be a bit faster. |
| 03:39.24 | Maxus2 | yeah so if i do a dail(device,10, g) type thing so if the call times out it just moves on? |
| 03:40.30 | kaldemar | ,10 is probably not what you want. |
| 03:57.30 | Maxus2 | what would i do instead of ,10 kaldemar? |
| 04:00.20 | *** join/#asterisk Cesare (~Adium@creati59.lnk.telstra.net) |
| 04:04.12 | kaldemar | Maxus2: use autokill with iax and timers with sip. |
| 04:04.42 | Maxus2 | oh okay |
| 04:04.49 | Maxus2 | just doesn't seem reliable. |
| 04:05.10 | Maxus2 | if i diconnect a machine it still think its there for up to five-ten minutes |
| 04:08.55 | *** part/#asterisk benngard (~mabe@h30n2-g-ml-a11.ias.bredband.telia.com) |
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| 06:15.29 | dr00d | hi all - i have an extension specified in extensions_custom.conf in the from-internal-custom context ? |
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| 06:16.57 | SiNGLer | dr00d: why should we know if you have an extension? :) |
| 06:17.04 | dr00d | ... which uses read, set etc asterisk commands and it works fine - if i put a similar extension under the from-pstn-custom context the asterisk commands dont run - any ideas anyone ? |
| 06:17.12 | kaldemar | does not see a question |
| 06:17.13 | dr00d | sorry there was more coming |
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| 06:18.24 | kaldemar | how do they not run? does the call not land in your extension? |
| 06:18.25 | SiNGLer | dr00d: pastebin working and not working contexts and asterisk verbose output |
| 06:18.36 | dr00d | ok |
| 06:18.59 | dr00d | you want me to past it all in here ? |
| 06:19.23 | SiNGLer | pastebin.com |
| 06:19.33 | Maxus2 | http://pastebin.com/ |
| 06:19.33 | SiNGLer | or similar service |
| 06:21.48 | dr00d | ok i pasted the extensions - rtp asterisk |
| 06:22.20 | SiNGLer | paste link into channel :) |
| 06:22.46 | dr00d | http://pastebin.com/ana08h9W |
| 06:23.21 | dr00d | the hangup line is missing in the lower extension |
| 06:23.27 | dr00d | but it is actually in there |
| 06:24.06 | dr00d | it seems that some apps dont run when used under the from-pstn-custom context |
| 06:24.16 | SiNGLer | and what about asterisk verbose output? |
| 06:24.22 | dr00d | ok im doing that now |
| 06:24.38 | kaldemar | dr00d: all apps run if the call matches the extension. regardless of the context. |
| 06:27.42 | *** join/#asterisk schmidts (~schmidts@Lo20-rt01.lm33.sil.at) |
| 06:27.45 | schmidts | good morning |
| 06:27.49 | dr00d | http://pastebin.com/JC26jXA2 |
| 06:28.15 | dr00d | thats the v asterisk output - and that is what i thought but i cant get it to work - the commands wont run |
| 06:28.52 | dr00d | if i used SayDigits that works |
| 06:29.51 | dr00d | and the last paste is for when i call the extension 899 that doesnt work |
| 06:30.40 | dr00d | also - ext 600 is a sip extension - 899 is an inbound route from a trunk connected to an fxo gateway |
| 06:30.52 | kaldemar | dr00d: the call never goes to that context. |
| 06:31.52 | dr00d | i have swift installled - if i put a line of code with swift - it runs ok so the call does seem to be going to that context |
| 06:32.42 | kaldemar | what you pasted goes to from-internal. |
| 06:32.51 | dr00d | hmmm |
| 06:33.31 | dr00d | ok ill comne in from another angle |
| 06:33.43 | kaldemar | maybe you should ask in #freepbx since your issue seems to be purely related to dialplan structure. |
| 06:33.54 | dr00d | ok thanks |
| 06:34.50 | jacc0 | goodmorning |
| 06:38.53 | Maxus2 | morning |
| 06:40.42 | dr00d | i dont think they want to help me :( |
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| 08:34.09 | devil_evoxxx | hi all |
| 08:34.51 | devil_evoxxx | i've installed and succesfully configured 3 B410PF card on pc |
| 08:35.06 | devil_evoxxx | i need to make some test, with a isdn phone |
| 08:36.51 | devil_evoxxx | but i read on b410p datasheet that the card does not supply power to phone. I think that i need a PWR400 ( digium ) but there isnt this power adapter in any store i know. |
| 08:38.01 | devil_evoxxx | can i power on the phone, using pin 7 and 8 of isdn cable? |
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| 08:49.04 | dabukalam | I have an ubuntu server, a landline phone, and an old PCI modem. Is that enough to be able to answer my home phone on my android via Wi-Fi? |
| 08:49.53 | henk | i think you'd have to substitute the pci modem with a telephony card... |
| 08:50.23 | kaldemar | or an ATA with an FXO. |
| 08:56.08 | dabukalam | kaldemar: would a LOL or a WTF work? |
| 08:56.12 | dabukalam | kaldemar: :P |
| 08:57.32 | devil_evoxxx | you need an ata with fxo or a thelephony card as henk say.. |
| 08:57.50 | devil_evoxxx | next , you have only to route the incoming call from pstn to your android phone |
| 08:57.52 | kaldemar | not quite as well. :) ATA as in analog telephone adapter and FXO as in foreign exchange office. an ATA is a device that connects analog telephony lines to VoIP, and an FXO port is like a phone, i.e. is is used to connect to an analog line. |
| 08:59.33 | kaldemar | ~thebook |
| 08:59.33 | infobot | Asterisk: The Definitive Guide, 3rd Edition (ISBN 0-596-51734-3) available at http://oreilly.com/catalog/9780596517342. Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and is available for reading online at http://ofps.oreilly.com/titles/9780596517342/ |
| 09:01.44 | devil_evoxxx | ops, i want to say fxs |
| 09:01.47 | dabukalam | how much would that set me back? |
| 09:01.49 | dabukalam | $$ |
| 09:01.58 | dabukalam | also, can i manage asterisk through a web interface |
| 09:04.04 | devil_evoxxx | i don't like web interface :) but if you need, you can |
| 09:06.44 | kaldemar | dabukalam: under $100 |
| 09:12.16 | dabukalam | There must be a way to do it with a modem no? I mean in the end you're talking about an RJ11 jack in the back of a PC |
| 09:12.58 | dabukalam | There must be some sort of driver which will manipulate the modem to communicate with asterisk... Or am I talking out of my ass? |
| 09:15.07 | *** join/#asterisk vfabi (~fabi@h2-92.faust.net.ua) |
| 09:15.07 | devil_evoxxx | i not say if you can use a common pci modem. I always use Digium card (BRI/PRI) and the drive is provided by DAHDI |
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| 09:19.50 | devil_evoxxx | anyone know how i can power a isdn phone connected on a B410PF Digium Card? |
| 09:20.30 | devil_evoxxx | the phone is not self-powered :( |
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| 09:49.21 | puzzled | devil_evoxxx: not sure but if the card has a power connector then I assume it should be possible to hook up an isdn phone. powering a phone requires more power than the pci/pcie slot can provide hence the need for a separate power connector on the card |
| 09:52.34 | devil_evoxxx | thankyou for reply puzzled :) i have read in the datasheet that if i connect a isdn phone i need to use a PWR400 (digium) but this item has never been producted. I need to make some test before install the system and make it in production |
| 09:52.49 | devil_evoxxx | have you got any idea for testing? |
| 09:53.29 | puzzled | nope as I don't have an isdn phone and I only have a Sangoma B700 isdn card |
| 09:54.27 | *** join/#asterisk Lantizia (~Lantizia@erebus.seaquake.net) |
| 10:00.59 | devil_evoxxx | and how you made your test? I have to make a mediagw for provide a 4 nt ports .I'm converting an old isdn pbx for making/receive calls over VoIP |
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| 10:39.10 | puzzled | devil_evoxxx: I'm just using a live isdn line for initial testing. if I need more I'll hook up an old ISDN switch. but I don't use ISDN phones so that is completely different |
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| 10:51.06 | *** join/#asterisk shamelessn00b (~chatzilla@203.175.64.42) |
| 10:51.12 | shamelessn00b | voip-info.org down? |
| 10:52.05 | *** join/#asterisk SoulRaven (SoulRaven@79.117.213.217) |
| 10:52.46 | shamelessn00b | ~book |
| 10:52.46 | infobot | [~book] Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://astbook.asteriskdocs.org or see ~buybook, or http://ofps.oreilly.com/ |
| 10:53.05 | shamelessn00b | ~newbook |
| 10:53.06 | infobot | Please see ~thebook for more information about Asterisk: The Definitive Guide |
| 10:53.13 | shamelessn00b | ~thebook |
| 10:53.13 | infobot | Asterisk: The Definitive Guide, 3rd Edition (ISBN 0-596-51734-3) available at http://oreilly.com/catalog/9780596517342. Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and is available for reading online at http://ofps.oreilly.com/titles/9780596517342/ |
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| 11:01.29 | ironm | Good afternoon - please allow me one question - is it possible to log a CDR-entry with the start time/date even the call is still active (just to calculate current costs including still running calls)? .. maybe with a special flag to recognize accomplished and still active calls. Thank you in advance for any hints. |
| 11:03.01 | *** join/#asterisk moltar_net (~Roman@180.183.200.118) |
| 11:03.44 | moltar_net | This might seem like a noob question, and it's a bit off topic, but it's difficult to Google this one. Where do DIDs come from? Who manages the pool and assigns distributes them to the VOIP companies? Thanks! |
| 11:05.04 | ironm | moltar_net, what do you mean with DIDs ? |
| 11:06.13 | moltar_net | the phone number |
| 11:06.26 | moltar_net | where do VOIP companies get the pools of numbers to assign to customers? |
| 11:06.37 | *** join/#asterisk MrSmurf (~MrSmurf@unaffiliated/mrsmurf) |
| 11:06.51 | moltar_net | e.g. i go to voip.ms and they present me with 10 options for my area code. where did they get these numbers from? |
| 11:07.03 | moltar_net | I want to go to the source to see if they have better numbers :) |
| 11:07.50 | *** join/#asterisk vfabi (~fabi@h2-92.faust.net.ua) |
| 11:08.52 | moltar_net | P.S. this is in regards to Canadian numbers |
| 11:09.22 | moltar_net | e.g. 416-900 number pool has been opened up a year or so ago ... many numbers available. I want to see what is there up for grabs. |
| 11:09.33 | moltar_net | Is there like a central database that has the info on all numbers? |
| 11:10.17 | kaldemar | moltar_net: governments usually assign number blocks to telcos through an agency. |
| 11:10.40 | kaldemar | or civil service deparment or whatever you call those in english. |
| 11:11.17 | kaldemar | telcos may give pools further to service providers. |
| 11:11.35 | moltar_net | telcos as in major players like Bell etc? |
| 11:13.34 | kaldemar | telcos, despite their market share. it's up to the officials to decide how it works in your country. |
| 11:14.01 | kaldemar | s/despite/regardless of/ |
| 11:15.56 | moltar_net | but there must be some kind of central database for looking up which telco owns the number... otherwise how does a pbx know where to send the connection to... something like DNS for the web. |
| 11:23.07 | kaldemar | moltar_net: yes there is. find the officials that take care of numbering. a PBX doesn't know crap about anything, it just sends calls out a connection as it is instructed. delivery in PSTN is another thing. |
| 11:27.53 | *** join/#asterisk sourcode (~code@ppp-58-8-73-183.revip2.asianet.co.th) |
| 11:41.29 | *** join/#asterisk WiretapSeven (~Wiretap@unaffiliated/wiretap) |
| 11:43.00 | dandre | hello, |
| 11:43.05 | dandre | I have an fxo port connected to a subscriber line. How can I from a sip phone connected to my asterisk box send a flash hook to the line |
| 11:43.17 | dandre | <PROTECTED> |
| 11:44.49 | *** join/#asterisk jg1234 (~jan@dslc-082-082-037-188.pools.arcor-ip.net) |
| 11:45.43 | *** join/#asterisk vfabi (~fabi@h2-92.faust.net.ua) |
| 11:46.11 | leifmadsen | is it enabled in features.conf? |
| 11:46.25 | leifmadsen | alternatively you can create a custom feature code which uses the Flash() application |
| 11:46.26 | jg1234 | is there a substitute for "dialplan save"/"save dialplan" ? |
| 11:46.37 | leifmadsen | ya, save it from a real editor |
| 11:47.01 | jg1234 | haha |
| 11:47.08 | leifmadsen | if you save it from the asterisk CLI you will remove all comments from the file |
| 11:47.08 | *** join/#asterisk Faustov (user@gentoo/user/faustov) |
| 11:47.14 | jg1234 | i was printing/scanning ist the whole time |
| 11:47.48 | jg1234 | i dont care about the comments |
| 11:48.13 | leifmadsen | perhaps you can define why you're looking for something other than 'dialplan save' |
| 11:48.23 | leifmadsen | or what it doesn't do for you that you're expecting it to? |
| 11:48.32 | jg1234 | because it doesnt exist |
| 11:49.05 | leifmadsen | what version are you using? |
| 11:49.19 | jg1234 | 1.8.3.1 |
| 11:49.38 | schmidts | leifmadsen Congratulation for the cookbock ;) |
| 11:50.14 | WiretapSeven | glad it was 'cook' and not 'book' that you misspelled |
| 11:50.27 | WiretapSeven | err |
| 11:50.29 | WiretapSeven | spelled right |
| 11:50.31 | WiretapSeven | fuck I'm out of it |
| 11:51.24 | schmidts | WiretapSeven :D damn i thought about this 3 times if i write cook right and then :D |
| 11:51.49 | WiretapSeven | 'cookbook', a 'cookbock' doesn't exist and a 'cockbook' is something totally different :P |
| 11:51.59 | leifmadsen | jg1234: it does exist in the code, but I think there is a small bug here, let me see if I can create a patch and you can test it |
| 11:52.35 | dandre | leifmadsen: I thought *0 was directly handled by dahdi or zap channel |
| 11:52.43 | schmidts | WiretapSeven yes but i was afraid of writing cook wrong and didnt thought about book again ;) |
| 11:52.56 | WiretapSeven | tehehehe |
| 11:53.33 | schmidts | WiretapSeven typical case of a Freud typo |
| 11:53.38 | leifmadsen | dandre: I wouldn't know about that sorry |
| 11:53.45 | leifmadsen | dandre: I'm just telling you how you might enable it |
| 11:53.50 | jg1234 | leifmadsen: ok thx |
| 11:54.15 | dandre | ok |
| 11:54.29 | leifmadsen | jg1234: http://pastebin.com/AbgbqecN |
| 11:54.38 | leifmadsen | jg1234: untested |
| 11:54.50 | dandre | but I don't see what to put in features.conf |
| 11:54.52 | leifmadsen | (it does compile fine though) |
| 11:55.28 | leifmadsen | dandre: use an [applicationmap] which triggers the Flash() application |
| 11:55.34 | *** join/#asterisk freeedrich| (~eeePC@hansaserver.de) |
| 11:55.42 | leifmadsen | there are several examples of how to create application maps in the file |
| 11:56.09 | *** join/#asterisk radic (~radic@dslb-178-002-235-185.pools.arcor-ip.net) |
| 11:58.24 | jg1234 | Dialplan successfully saved into '/etc/asterisk/extensions.conf' |
| 11:58.36 | jg1234 | leifmadsen: thx again |
| 11:58.45 | leifmadsen | jg1234: yay! |
| 11:58.54 | leifmadsen | codes again! |
| 11:58.59 | leifmadsen | I'm not even a developer :) |
| 12:00.08 | dandre | ok |
| 12:00.38 | *** join/#asterisk Sertys (~sertys@89.252.247.42) |
| 12:00.38 | *** join/#asterisk ariel_ (~chatzilla@unaffiliated/abatista) |
| 12:00.38 | *** join/#asterisk Praise (~Fat@unaffiliated/praise) |
| 12:00.38 | *** join/#asterisk seanbright (~sean@asterisk/contributor-and-bug-marshal/seanbright) |
| 12:00.41 | leifmadsen | jg1234: you can also specify a path to save it to a different file |
| 12:00.49 | leifmadsen | jg1234: dialplan save /etc/asterisk/extensions.conf.new |
| 12:01.25 | jg1234 | leifmadsen: ok |
| 12:04.31 | *** join/#asterisk file (~file@asterisk/developer-and-muffin-lover/file) |
| 12:04.32 | *** mode/#asterisk [+o file] by ChanServ |
| 12:05.02 | leifmadsen | file: you should get your vanity mask updated |
| 12:12.17 | leifmadsen | M19140 |
| 12:12.22 | leifmadsen | oops wrong room |
| 12:18.58 | *** join/#asterisk TimeRider (~steve@188-220-33-123.zone11.bethere.co.uk) |
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| 12:45.11 | *** join/#asterisk logicwrath (~no@mail.vistitude.com) |
| 12:50.14 | *** join/#asterisk Valen (~Valen@ppp121-44-184-217.lns20.syd7.internode.on.net) |
| 12:50.31 | Valen | having some odd permissions problems (i think) with freepbx |
| 12:50.52 | Valen | how do i find out what file asterisk is actually trying to play when it goes to play a message? |
| 12:50.57 | logicwrath | i think amportal chown will check and set permissions check google on that |
| 12:51.19 | Valen | done that, its some weird problem with symlinks and other stuff i think |
| 12:51.24 | Valen | its in ubuntu 10.10 |
| 12:51.27 | logicwrath | check your CLI when the message gets played |
| 12:51.40 | logicwrath | it should tell you |
| 12:51.43 | Valen | it all works fairly well, except for recordings |
| 12:51.48 | Valen | it just gives a name not the path |
| 12:52.15 | *** join/#asterisk eduzimrs (~eduzimrs@201.22.86.124.static.gvt.net.br) |
| 12:53.00 | logicwrath | i think most sounds are in /var/lib/asterisk/sounds |
| 12:53.46 | Valen | I'm having issues getting them into there from the phone type recording thing to start with |
| 12:54.15 | Valen | it creates a file in /var/spool/asterisk/tmp owned by asterisk:asterisk rw |
| 12:54.23 | logicwrath | http://www.voip-info.org/wiki/view/Asterisk+sound+files |
| 12:55.04 | logicwrath | there should be a custom folder |
| 12:55.17 | logicwrath | you could try making the folder and see if that helps |
| 12:55.32 | Valen | then freepbx (running as www-data) cant move it into the /var/lib/asterisk/sounds/custom/ |
| 12:55.39 | Valen | says it cant read it |
| 12:56.30 | *** join/#asterisk orn (~orn@rtr1.sh23.sip.is) |
| 12:56.51 | Valen | so i chmod 777 it and it no longer whines about that |
| 12:56.55 | Valen | and moves it there |
| 12:57.11 | Valen | hmm but i wonder if thats not the location |
| 12:57.18 | Valen | (that it should be going) |
| 12:57.33 | logicwrath | #freepbx |
| 12:57.58 | Valen | heh probably |
| 12:58.29 | Valen | I was hoping there was some debug i could turn on and get asterisk itself to say "trying to play /var/lib/foo.wav but its not there doofus" |
| 12:58.45 | logicwrath | you can grep /var/log/asterisk/full |
| 12:58.55 | *** join/#asterisk clintc (~clintc@n128-227-125-7.xlate.ufl.edu) |
| 12:58.57 | Valen | doesn't seem to appear there |
| 12:59.16 | Valen | hmm I think it should perhaps be putting them into /usr/local/share/asterisk/sounds |
| 12:59.31 | logicwrath | your in the wrong channel |
| 12:59.34 | logicwrath | #freepbx |
| 12:59.41 | Valen | fairynuff |
| 13:00.05 | Valen | any idea why the "full" log wouldn't be showing much detail or is that a freepbx issue again |
| 13:00.28 | logicwrath | likely has to do with your debugging settings |
| 13:00.51 | kaldemar | Valen: it shows what logger.conf is told. |
| 13:01.56 | *** part/#asterisk clintc (~clintc@n128-227-125-7.xlate.ufl.edu) |
| 13:02.00 | *** join/#asterisk c0rnoTa (~c0rnoTa@78.24.154.190) |
| 13:04.23 | Valen | thanks anyway, think I'm going to call it for tonight, see what tomorrow brings |
| 13:05.26 | Valen | yeah definatly some path/permissions thing |
| 13:05.37 | Valen | if i stick it into a non /custom it seems to work |
| 13:05.44 | Valen | erugh i hate permissions |
| 13:06.02 | Valen | thanks anyway logicwrath and kaldemar, its appreciated |
| 13:10.32 | *** join/#asterisk SeTTleR (~bernd@p5DDEFC0D.dip.t-dialin.net) |
| 13:11.06 | *** join/#asterisk leifmadsen (~Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 13:11.06 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 13:15.08 | mocker | ~book |
| 13:15.08 | infobot | [~book] Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://astbook.asteriskdocs.org or see ~buybook, or http://ofps.oreilly.com/ |
| 13:17.08 | mocker | Anyone know if the PDF for the 3rd edition is going to be freely downloadable? |
| 13:23.50 | *** join/#asterisk killown (~killown@unaffiliated/killown) |
| 13:24.20 | leifmadsen | mocker: no PDF unless purchased -- only HTML |
| 13:24.26 | leifmadsen | ~thebook |
| 13:24.26 | infobot | Asterisk: The Definitive Guide, 3rd Edition (ISBN 0-596-51734-3) available at http://oreilly.com/catalog/9780596517342. Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and is available for reading online at http://ofps.oreilly.com/titles/9780596517342/ |
| 13:24.52 | leifmadsen | infobot: no book is <reply> For more information about the Asterisk book, see ~thebook |
| 13:24.52 | infobot | leifmadsen: okay |
| 13:25.05 | leifmadsen | ~buybook |
| 13:25.05 | infobot | [~buybook] You can buy "Asterisk the Future of Telephony" at http://www.oreilly.com/catalog/9780596510480/ so go buy it SERIOUSLY |
| 13:25.51 | leifmadsen | infobot: no, buybook is <reply> You can buy "Asterisk: The Definitive Guide" at hhttp://oreilly.com/catalog/9780596517342 so go buy it SERIOUSLY |
| 13:25.51 | infobot | okay, leifmadsen |
| 13:26.11 | leifmadsen | infobot: thebook is also ~buybook |
| 13:26.11 | infobot | okay, leifmadsen |
| 13:34.07 | *** join/#asterisk Faustov (user@gentoo/user/faustov) |
| 13:34.40 | *** join/#asterisk cerberus_za (~coert@196-210-206-216.dynamic.isadsl.co.za) |
| 13:36.54 | serafie | ~buybook |
| 13:36.55 | infobot | You can buy "Asterisk: The Definitive Guide" at hhttp://oreilly.com/catalog/9780596517342 so go buy it SERIOUSLY |
| 13:36.59 | serafie | leifmadsen: typo ^ |
| 13:37.06 | serafie | hhttp |
| 13:37.27 | leifmadsen | infobot: no, buybook is <reply> You can buy "Asterisk: The Definitive Guide" at http://oreilly.com/catalog/9780596517342 so go buy it SERIOUSLY |
| 13:37.27 | infobot | leifmadsen: okay |
| 13:37.29 | leifmadsen | thanks! |
| 13:44.40 | *** join/#asterisk killown (~killown@unaffiliated/killown) |
| 13:45.05 | *** join/#asterisk killown (~killown@unaffiliated/killown) |
| 13:46.24 | *** join/#asterisk chandoo (~chandoo@ool-4a5a2824.dyn.optonline.net) |
| 13:47.19 | *** join/#asterisk mallchin (~mallchin@mail.dataproservices.co.uk) |
| 13:47.31 | *** join/#asterisk jkroon (~jkroon@196-210-167-126.dynamic.isadsl.co.za) |
| 13:51.40 | *** join/#asterisk noobAsterisk (~matheus@200.17.112.83) |
| 13:51.54 | noobAsterisk | Hello everybody! |
| 13:52.07 | noobAsterisk | I'm in trouble here |
| 13:52.40 | noobAsterisk | Does anyone know about device called IPO-11? |
| 13:53.03 | noobAsterisk | AMe Optimal Technology |
| 13:53.21 | *** join/#asterisk FiReSTaRT (~dlyh@unaffiliated/firestart) |
| 13:54.35 | noobAsterisk | I'm trying to use it to connect a normal phone on usb |
| 13:54.37 | *** part/#asterisk benngard (~mabe@213.88.138.230) |
| 13:55.03 | noobAsterisk | To use VoIP |
| 13:55.08 | noobAsterisk | Any help? |
| 13:55.27 | noobAsterisk | Sorry bad english I'm brazilian |
| 13:55.30 | *** join/#asterisk garymc (~chatzilla@host86-148-189-105.range86-148.btcentralplus.com) |
| 13:57.26 | *** join/#asterisk Freeaqingme| (~dolf@dsl-083-247-011-232.solcon.nl) |
| 14:02.14 | leifmadsen | noobAsterisk: never heard of it |
| 14:02.59 | JonathanRose | noobAsterisk: It's also early in the morning for most of us, so please don't be too put off by the lack of responses. |
| 14:03.18 | *** join/#asterisk putnopvut (~putnopvut@asterisk/master-of-queues/mmichelson) |
| 14:03.18 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 14:03.21 | JonathanRose | And I've never heard of it, but on the other hand, I haven't heard of lots of things. |
| 14:05.19 | noobAsterisk | Jonathan, IPO-11 is a device RJ11-USB |
| 14:06.06 | noobAsterisk | Have heard about this kind of devices? |
| 14:08.09 | *** join/#asterisk deuast (~steigelr@business-213-023-245-200.static.arcor-ip.net) |
| 14:09.23 | deuast | exit |
| 14:09.27 | deuast | quit |
| 14:09.35 | leifmadsen | fail |
| 14:09.50 | *** join/#asterisk killown (~killown@unaffiliated/killown) |
| 14:11.31 | *** join/#asterisk beek (~klinebl@pdpc/supporter/bronze/beek) |
| 14:12.03 | kaldemar | noobAsterisk: seems to be just an adapter that can be used with a soft phone that supports the used API. |
| 14:12.34 | kaldemar | noobAsterisk: what are you trying to do with it? |
| 14:16.00 | noobAsterisk | I'm trying to connect a normal phone on USB port, but, on Ubuntu, this device dont work properly |
| 14:16.48 | *** join/#asterisk Praise (~Fat@unaffiliated/praise) |
| 14:20.10 | kaldemar | noobAsterisk: how do you expect it to work? |
| 14:21.48 | leifmadsen | Ubuntu would have to support it no? |
| 14:23.24 | kaldemar | ubuntu would need a driver for it and a soft phone that can use it. |
| 14:23.41 | noobAsterisk | I'm expect make and received call by a normal phone plugged on Usb |
| 14:24.21 | *** join/#asterisk deuast (~steigelr@business-213-023-245-200.static.arcor-ip.net) |
| 14:24.53 | noobAsterisk | leifmadsen, At the moment I dont find |
| 14:24.56 | *** join/#asterisk Defraz (~Defraz@tim.spudnik.com) |
| 14:25.40 | noobAsterisk | I dont find any drivers to make the IPO-11 works on Ubuntu |
| 14:25.53 | noobAsterisk | Have a hint? |
| 14:26.15 | leifmadsen | none at all |
| 14:26.47 | dandre | I am trying to setup features and that doesn't work |
| 14:27.09 | dandre | I have enabled them in features.conf |
| 14:27.38 | dandre | set the channel variable __DYNAMIC_FEATURES |
| 14:28.11 | dandre | and the feature is never used |
| 14:28.12 | kaldemar | noobAsterisk: a driver is not enough. in addition you would need a soft phone that supports it. |
| 14:28.30 | dandre | the *8 default feature works perfectly |
| 14:28.38 | *** join/#asterisk Devon_ (~chatzilla@63.214.236.169) |
| 14:28.45 | dandre | but not mine |
| 14:28.53 | *** join/#asterisk brainiac (~brainiac@208.86.215.38) |
| 14:29.13 | kaldemar | dandre: what do you see in CLI when it doesn't work? |
| 14:29.32 | dandre | nothing |
| 14:32.40 | *** join/#asterisk titter (~Justin@c-98-208-153-148.hsd1.fl.comcast.net) |
| 14:33.28 | dandre | http://pastebin.fr/11106 |
| 14:34.15 | dandre | but once the call is established, #9 dos nothing and give no trace in cli |
| 14:35.48 | *** join/#asterisk dewman (~dewman@68-188-190-218.dhcp.bycy.mi.charter.com) |
| 14:36.33 | Defraz | If I change something in the chan_dahdi.conf file do I need to reload asterisk or will a simple config reload make the changes? |
| 14:36.51 | kaldemar | dandre: before it |
| 14:37.04 | kaldemar | Defraz: depends on what you change. |
| 14:37.12 | *** part/#asterisk dewman (~dewman@68-188-190-218.dhcp.bycy.mi.charter.com) |
| 14:37.25 | Defraz | rxgain and txgain and relaxdtmf |
| 14:37.55 | kaldemar | iirc, a reload should be enough. |
| 14:38.01 | Defraz | that is what I thought. |
| 14:38.29 | Defraz | do I need to set the echotraining if the card has built in echo canceling? |
| 14:39.49 | Defraz | Just having some DTFM issues that I can't figure out. When I use the T1/PRI via the router it works great when I use the TE205P card dtmf works about half the time. |
| 14:39.57 | dandre | kaldemar: here is the trace: http://pastebin.fr/11107 |
| 14:39.58 | Defraz | and my quality isn't very good. |
| 14:41.46 | leifmadsen | what module provides Asterisk CLI command 'channel originate' again? |
| 14:43.31 | dandre | and after a reload: |
| 14:43.34 | dandre | <PROTECTED> |
| 14:44.06 | *** join/#asterisk ariel_ (~chatzilla@unaffiliated/abatista) |
| 14:44.07 | *** join/#asterisk Tim_Toady (~moi@79.103.50.246.dsl.dyn.forthnet.gr) |
| 14:44.16 | leifmadsen | answer: res_clioriginate.c |
| 14:44.52 | mallchin | hi, I'm having an issue with calls not hanging up properly, it is intermittent, any idea on the best way to track it down? :) |
| 14:45.45 | mallchin | Calls are incoming zap to asterisk box 1 -> sip to asterisk box 2 -> iax2 to asterisk box 3 -> zap outgoing to pstn landline |
| 14:45.49 | *** join/#asterisk deuast (~irc2@business-213-023-245-200.static.arcor-ip.net) |
| 14:45.55 | BlackBishop | [Sep 14 04:21:07] WARNING[30383]: chan_sip.c:15274 receive_message: Received message to <sip:0000000234@xxx.xxx.ro> from <sip:dex@xxx.xxx.ro>;tag=MtwEL3pMINnMY7wWmCITVPZvvGojsP.I, dropped it... |
| 14:46.05 | BlackBishop | <PROTECTED> |
| 14:46.09 | BlackBishop | <PROTECTED> |
| 14:46.14 | BlackBishop | any way I can make something with this ? |
| 14:46.25 | BlackBishop | as in .. when asterisk gets a message ... to pass it to an app ... |
| 14:46.34 | BlackBishop | or execute a command with some args... |
| 14:47.39 | *** part/#asterisk moltar_net (~Roman@180.183.200.118) |
| 14:48.03 | leifmadsen | BlackBishop: not really, that's more of a SIP proxy job |
| 14:48.53 | BlackBishop | so I have to install a sip proxy to do stuff with messages |
| 14:48.58 | BlackBishop | and the rest to pass to asterisk ? |
| 14:49.40 | *** join/#asterisk jaytee (~jforde051@unaffiliated/jaytee) |
| 14:49.46 | BlackBishop | any recomandations ? |
| 14:50.49 | *** join/#asterisk The_Boy_Wonder (~vossel@asterisk/batman-developer/dvossel) |
| 14:51.01 | leifmadsen | there are several SIP proxies. Asterisk is a B2BUA and is not really designed to work with low level SIP messages directly |
| 14:52.20 | BlackBishop | do you recommend one that's easy to set up for what I want to do ? |
| 14:53.03 | *** join/#asterisk dimm (~appleworm@unaffiliated/dimm) |
| 14:55.37 | *** join/#asterisk seanbright (~sean@asterisk/contributor-and-bug-marshal/seanbright) |
| 14:58.03 | aberrios | hmm, the "PDA Module" in Digiums branded back packs are just the right size for a Samsung Galaxy Pad... coincidence..?? ;) |
| 15:00.39 | BlackBishop | yes. |
| 15:01.01 | aberrios | I'm wondering what "PDA's" Digium had in mind. |
| 15:01.28 | *** join/#asterisk anthm (~anthm@freeswitch/developer/anthm) |
| 15:03.05 | BlackBishop | the blonde ? .. I have a nice blonde PDA .. |
| 15:03.13 | BlackBishop | ( if only she wasn't already married .. ) |
| 15:04.16 | *** join/#asterisk luckman212 (~irc@pool-173-77-253-141.nycmny.fios.verizon.net) |
| 15:05.15 | henk | digitally blonde? |
| 15:07.02 | *** join/#asterisk freckle (~jon@87.127.248.44) |
| 15:09.03 | luckman212 | Is there a setting that allows/disallows anonymous SIP calls from the local network? I've enabled anonymous SIP on my ast 1.8 pbx(to allow direct SIP-SIP dialing via URI) and it works as long as the call has the EXTERNAL (wan) ip in the SDP. so a call to xxx@my.wan.ip.addr works but the same call to xxx@10.10.10.20 (asterisk lan ip) fails with "401 UNAUTHORIZED" |
| 15:09.19 | luckman212 | im tearing out many hairs over this |
| 15:11.35 | Tozz_ | anyone here that knows a DiD provider in Belarus? (White Russia) |
| 15:11.45 | leifmadsen | luckman212: you could use permit and deny settings |
| 15:12.02 | luckman212 | leifmadsen: you know what, stepping into this room has given me some magical powers |
| 15:12.12 | luckman212 | leifmadsen: i just figured out what it was |
| 15:12.14 | leifmadsen | :) |
| 15:12.15 | leifmadsen | nice |
| 15:12.22 | leifmadsen | Tozz_: now I want a white russian |
| 15:12.41 | luckman212 | leifmadsen: by the way, i just bought the new book, congratulations on that!! an amazing work |
| 15:12.58 | leifmadsen | awesome! thanks! Cookbook or The Definitive Guide? |
| 15:13.04 | luckman212 | Definitive Guide |
| 15:13.09 | leifmadsen | most excellent |
| 15:13.17 | leifmadsen | we're very proud of that book |
| 15:13.21 | leifmadsen | a lot of good info in there |
| 15:13.39 | BlackBishop | I think I'll get it too |
| 15:13.50 | luckman212 | everyone Must Buy That Book |
| 15:13.56 | leifmadsen | if you like it don't be afraid to write a review on Amazon for us! |
| 15:14.11 | leifmadsen | high reviews helps our ranking and sales |
| 15:14.20 | leifmadsen | and more sales means we have the ambition to write another book :) |
| 15:14.23 | luckman212 | oh will do that later today |
| 15:14.38 | BlackBishop | damn, so I'll have to buy the new one to see what's in it then ! |
| 15:14.38 | BlackBishop | :)) |
| 15:14.41 | leifmadsen | luckily you can review it now since you can read it at http://ofps.oreilly.com |
| 15:14.53 | BlackBishop | I'd just like a .diff from the old book if a new one appears ! |
| 15:14.54 | BlackBishop | :)) |
| 15:15.07 | aberrios | BlackBishop: :) |
| 15:15.17 | luckman212 | BlackBishop: i have both and I think the .diff would be bigger than the book itself |
| 15:15.22 | luckman212 | if that even makes sense |
| 15:15.25 | beek | leifmadsen: I just got a note from Amazon saying that they expect to get it 4/26... |
| 15:15.25 | leifmadsen | it would :) |
| 15:15.43 | luckman212 | beek: you can get the e-book now (thats what I did) |
| 15:15.43 | leifmadsen | beek: ya it was a little bit delayed -- it will start shipping first week of May from what I've been told |
| 15:15.56 | leifmadsen | luckman212: it was pretty much completely rewritten |
| 15:16.09 | BlackBishop | well .. a .patch then ! |
| 15:16.10 | BlackBishop | :) |
| 15:16.45 | leifmadsen | a patch would not apply |
| 15:16.53 | leifmadsen | it would literally be larger than the book itself |
| 15:17.12 | mallchin | tipex and a biro? |
| 15:17.12 | leifmadsen | that's because none of the chapters are in their original form, and many new ones were added |
| 15:17.15 | BlackBishop | that's it .. I'm moving to ael ! |
| 15:17.18 | leifmadsen | :) |
| 15:17.26 | BlackBishop | this exten => thing gets too hard to read |
| 15:17.27 | leifmadsen | you won't find any AEL code in A:TDG :) |
| 15:17.36 | BlackBishop | google is my friend .. |
| 15:17.40 | jaytee | I think the book should have come with a fold out poster photo of Leif and Russell :-) |
| 15:17.43 | BlackBishop | all I need is to get the syntax right |
| 15:17.45 | beek | AEL for ever! |
| 15:17.53 | BlackBishop | it looks more like C/php/whatever |
| 15:17.56 | BlackBishop | so I can read it easyer |
| 15:17.59 | BlackBishop | easier |
| 15:18.09 | luckman212 | leifmadsen: so the deal with my original question on SIP URI and why it was failing... seems like it *might* almost classify as a bug. Basically if a client (softphone in my case) is registered as an extension to the pbx, but attempts an anonymous call, the server will reject it as 401 Unauthorized so the "fix" is that in order to make anon SIP calls to a LAN-local pbx, you cannot be regg'ed at the same IP |
| 15:18.14 | luckman212 | does that make sense |
| 15:18.14 | *** join/#asterisk benngard (~mabe@h30n2-g-ml-a11.ias.bredband.telia.com) |
| 15:18.41 | luckman212 | I was regged at x703 but was testing sip uri dialing via an anonymous (non-regged) account from the same ip |
| 15:18.58 | luckman212 | as soon as I unregged the other extension, calls went thru fine |
| 15:19.30 | leifmadsen | will catch up after this meeting |
| 15:20.01 | leifmadsen | luckman212: actually that makes sense to me because asterisk will try to match based on IP first |
| 15:20.08 | leifmadsen | that's just kind of how SIP works |
| 15:20.19 | leifmadsen | chan_sip specifically (not the SIP protocol) |
| 15:20.30 | *** join/#asterisk marlowe (~marlowe@ip68-100-150-54.dc.dc.cox.net) |
| 15:20.32 | leifmadsen | I don't think it would count as a bug, but rather more of a, "that's how it's implemented" issue |
| 15:20.36 | luckman212 | hmm interesting, yes its kind of a rare case I guess that would almost never occur in the real world |
| 15:20.58 | luckman212 | my page detailing those is getting rather long however <grin> |
| 15:21.20 | serafie | BlackBishop: the "same" syntax makes dialplans easier to read. :) |
| 15:22.10 | *** join/#asterisk chandoo (~chandoo@ool-4a5a2824.dyn.optonline.net) |
| 15:22.22 | BlackBishop | even better |
| 15:22.42 | leifmadsen | I <3 same => |
| 15:23.14 | serafie | mee tooo |
| 15:23.35 | flashdeluxe | hi! is there any gui which is easy to handle for customers (e.g. for adding SIP Clients or activating call forwarding to an own number)? |
| 15:23.37 | luckman212 | we need a 3rd book: "Asterisk Oddities: A Comprehensive Guide to Quirks, Gotchas, Snags & Snafus" |
| 15:23.48 | leifmadsen | flashdeluxe: Asterisk GUI! |
| 15:24.06 | leifmadsen | luckman212: write me up a list of them and I'll create a chapter in the cookbook |
| 15:24.30 | flashdeluxe | leifmadsen: is it stable so far? I heard that it is a little bit buggy.. |
| 15:24.37 | luckman212 | leifmadsen: I am working on an Evernote notebook that will be a compilation of them. I will share it with you when that's ready |
| 15:25.16 | serafie | flashdeluxe: there have been tons of bugfixes recently. |
| 15:25.39 | serafie | the SVN version is fairly stable. If you are using 1.6.0 or greater, I would suggest waiting a week. |
| 15:27.13 | *** join/#asterisk ks3 (~ksandy@74.203.195.1) |
| 15:27.20 | luckman212 | I am planning to update my test server to 1.8.3.2 today along with the new DAHDI 2.4.1.2 -- gonna be a big day |
| 15:27.31 | flashdeluxe | serafie: i am using asterisk 1.8 with capi |
| 15:28.08 | *** join/#asterisk like_a_horse (~like_a_ho@firect.saao.ac.za) |
| 15:28.25 | serafie | 1.8 support is not strong, but it will be much better soon. |
| 15:29.04 | serafie | I hesitate to say "all fixed," but it will be close to all fixed. |
| 15:29.05 | flashdeluxe | serafie: i have nothing to loose, i will try it out :) |
| 15:29.22 | leifmadsen | s/loose/lose/ |
| 15:29.30 | benngard | lot of fun things to try in 1.8 :) |
| 15:31.29 | *** join/#asterisk Tech_Travis (~Travis_El@cpe-76-168-191-127.socal.res.rr.com) |
| 15:31.33 | benngard | is gonna play with the calendar functions tonight, and the beginning of qsig support in ooh323 :) |
| 15:31.36 | leifmadsen | 1.8 is the win |
| 15:31.47 | jaytee | is same => only in 1.8 or is it available in 1.6.2? |
| 15:31.53 | leifmadsen | 1.6.2 + |
| 15:32.11 | like_a_horse | hi all, i'm quite new at this but hoping someone can point me in the right direction. I want to implement a self service extension that ppl can call to change their pin. I was going to use astdb to store the pins but i'm battling to find a decent structure to store them. I want to use 8 digit pins that include the persons 4 digit ext and then another 4 digit code. So something like 40041234. Is there a well known way to do this? I'm been play |
| 15:32.12 | like_a_horse | ing with the authenticate application but not having too much luck.. |
| 15:32.19 | Qwell | Nobody uses 1.6.2 anymore. |
| 15:33.48 | leifmadsen | Security fixes only in 2 more days! |
| 15:33.58 | leifmadsen | then I can close hundreds of issues on mantis |
| 15:38.27 | like_a_horse | dang.. have to run... |
| 15:40.42 | *** join/#asterisk iprouteth0 (~iprouteth@unaffiliated/iprouteth0) |
| 15:47.33 | ariel_ | I still use 1.6 |
| 15:47.42 | leifmadsen | 1.6 is not a branch |
| 15:47.56 | leifmadsen | ~asterisk16 |
| 15:47.56 | infobot | new features in Asterisk 1.6 are listed at http://svn.digium.com/view/asterisk/trunk/CHANGES?view=markup |
| 15:49.03 | leifmadsen | infobot: no, asterisk16 is <reply> Asterisk 1.6 is not a branch or a version. Asterisk 1.6.0, 1.6.1 and 1.6.2 are major version changes, much like Asterisk 1.2, 1.4, and 1.8 are. Please be more specific about the branch you are using. Information about Asterisk support is available at ~asterisk-versions |
| 15:49.03 | infobot | okay, leifmadsen |
| 15:49.07 | leifmadsen | ~asterisk-versions |
| 15:49.07 | infobot | Information about Asterisk maintenance support and when branches will move into security fix only mode, and eventually end-of-life is available at https://wiki.asterisk.org/wiki/display/AST/Asterisk+Versions |
| 15:52.30 | freckle | got a weird issue with 1.6.2.17.2 |
| 15:52.54 | freckle | it just stops responding to signalling and audio drops randomly |
| 15:53.10 | *** join/#asterisk dimm (~appleworm@unaffiliated/dimm) |
| 15:53.24 | freckle | I mean all audio on all calls... killing and restarting is the only way to get it going again.. logs reveal nothing |
| 16:01.29 | leifmadsen | freckle: sounds a lot like a deadlock to me |
| 16:02.08 | leifmadsen | check the wiki articles on wiki.asterisk.org under Development > Debugging and get a backtrace and 'core show locks', then attach that data to an issue at http://issues.asterisk.org |
| 16:04.17 | *** join/#asterisk aberrios (~aberrios@195.171.4.82) |
| 16:09.46 | *** join/#asterisk oej (~olle@2001:470:1f15:d79:f159:f24a:7aab:2ac3) |
| 16:10.07 | *** join/#asterisk Poincare (~jefffnode@2001:470:d6b3:4::2) |
| 16:12.04 | freckle | leifmadsen: 'core show locks' results in No such command |
| 16:13.01 | Qwell | freckle: You have to enable it. See the page he referred to. |
| 16:13.12 | freckle | oh ok |
| 16:16.34 | freckle | ok guys I didn't enable "DONT_OPTIMZIE' when I compliled... is that going to be an issue getting a good backtrace? |
| 16:17.19 | leifmadsen | yep |
| 16:17.31 | freckle | so I need to recomile? |
| 16:17.42 | freckle | recompile |
| 16:19.07 | *** join/#asterisk [T]ank (~Tank@206.71.78.158) |
| 16:21.16 | [T]ank | when I do a dial from one asterisk server to another, the call bridges between the two servers... that is working just like it should. What I want it to do is send the call from one server to another without bridging. I tried using 'Transfer', that did not work. here is the command i tried using: exten => 11839,n,Transfer(SIP/${EXTEN}@ICR) |
| 16:21.33 | [T]ank | exten => 11839,n,Dial(SIP/${EXTEN}@ICR) works just fine. not sure I know how to use the transfer command. |
| 16:23.03 | [T]ank | any help would be appreciated |
| 16:23.37 | [T]ank | im not sure that Transfer() is even what i want to be using. |
| 16:24.38 | leifmadsen | Transfer() before Answer() (or audio) will do a 302 Redirect |
| 16:25.15 | [T]ank | let me pastebin a few things and have you look at what i have done. |
| 16:25.17 | [T]ank | just a moment |
| 16:25.35 | *** join/#asterisk vinhdizzo (~vinh@169.234.7.56) |
| 16:27.03 | *** join/#asterisk [loy] (~nobody@95.72.112.115) |
| 16:31.28 | [T]ank | crap... pastebin is blocked by our network admins now... cant paste anything. LAME!!! |
| 16:33.55 | Freeaqingme| | omg |
| 16:33.58 | Freeaqingme| | why would they block that? |
| 16:34.04 | Freeaqingme| | www.pastie.org is your friend? |
| 16:35.31 | *** join/#asterisk dimm (~appleworm@unaffiliated/dimm) |
| 16:37.28 | [T]ank | ability for random employees to share sensitive data, im sure. |
| 16:37.28 | luckman212 | guys is it possible to allow SIP INVITES only from the Trunks that I have registered with? |
| 16:38.11 | luckman212 | I want to allow anyone to SIP dial joe@mydomain.com but I don't want them (the sip spammers) to be able to dial joe@173.55.22.11 |
| 16:38.52 | luckman212 | when I set "allowexternaldomains=no" in my sip.conf, that works but I can then no longer receive inbound calls from my SIP TRUNKS because I guess they are addressed to from-trunk@173.55.22.11 |
| 16:40.29 | [T]ank | @leifmadsen: transfer() before answer(). basically, what i am doing is playing an audio file, like an auto attendand. Then on a button press, it send to the other server. what did you mean by transfer before answer? when 1 is pressed for example, the only line i have is the dial command. that is the one i want to change to be a transfer. so, there is no answer() command on that extension. am |
| 16:40.29 | [T]ank | i following you correctly? |
| 16:41.24 | [T]ank | here is what i get when it runs the transfer command: |
| 16:41.55 | [T]ank | -- Executing [transfer@Avaya_SIP:1] Transfer("SIP/<IPADDRESS Removed>-10593500", "SIP/11839@ICR") in new stack |
| 16:41.55 | [T]ank | <PROTECTED> |
| 16:42.32 | [T]ank | but if i change transfer( to dial( it works just fine |
| 16:43.46 | *** join/#asterisk Poincare (~jefffnode@2001:470:d6b3:4::2) |
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| 16:55.18 | *** join/#asterisk jkroon (~jkroon@dsl-241-228-173.telkomadsl.co.za) |
| 16:55.25 | *** join/#asterisk killown (~killown@unaffiliated/killown) |
| 16:55.52 | jkroon | hi guys, with just wct4xxp loaded (for PRI channels), is it possible to check whether we can use the remote end as a timing source? |
| 16:56.56 | jkroon | or does the value of the timing thing in /etc/dahdi/system.conf not really matter in this case? (needs to be a value from 0 to 4 where 0 is supposed to indicate that we provide timing to the peer, and 1 to 4 indicates priority of using that particular link for timing - or at least, that's how I understand it ...) |
| 17:07.07 | *** join/#asterisk dimm (~appleworm@unaffiliated/dimm) |
| 17:07.45 | *** join/#asterisk Hyper-Core (~lol@h174.117.31.71.dynamic.ip.windstream.net) |
| 17:13.31 | freckle | exit |
| 17:14.00 | Freeaqingme| | what's up with people typing 'exit' to quit? :/ |
| 17:14.11 | *** part/#asterisk joshaidan (~brianj@S0106001c1023e838.tb.shawcable.net) |
| 17:15.35 | *** part/#asterisk Hyper-Core (~lol@h174.117.31.71.dynamic.ip.windstream.net) |
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| 18:10.19 | Kobaz | makes some soup with options |
| 18:10.21 | Kobaz | er |
| 18:16.51 | *** mode/#asterisk [-q antoasla_!*@*] by Qwell |
| 18:18.29 | *** mode/#asterisk [-q fauxalliance!*@*] by Qwell |
| 18:34.37 | *** join/#asterisk chuckf (~chuckf@ubuntu/member/chuckf) |
| 18:35.54 | benngard | no i think i wm right |
| 18:36.06 | benngard | wrong chat :( |
| 18:38.43 | *** join/#asterisk dimm (~appleworm@unaffiliated/dimm) |
| 18:40.32 | [T]ank | when i do a exten => 11839,1,Dial(SIP/${EXTEN}@ICR) my call goes through just fine, but when i do a exten => 11839,1,Transfer(SIP/${EXTEN}@ICR) I actually get a Auto fallthrough, channel 'SIP/<ipaddress removed>-10593500' status is 'UNKNOWN' |
| 18:40.35 | [T]ank | what could cause this? |
| 18:47.27 | *** join/#asterisk afink (~afink@204.26.87.226) |
| 18:48.22 | *** join/#asterisk Delemas (~rhardy@vortex.webcon.ca) |
| 18:55.57 | afink | hello everyone, I just got some aastra phones for the first time and I am having some trouble with them. I can receive calls just find but I can't make any outgoing calls. Here is a paste of sip show peer 1007 (aastra phone) http://pastebin.com/raw.php?i=v4Eyrbzk |
| 18:56.53 | Delemas | I need 1.4 Asterisk server to allow connections to two external IPs. I can always ping both. Asterisk is listening on all addresses and has only one marked as externalip in sip.conf. Other than firewall rules what could be blocking connections? |
| 18:58.07 | *** join/#asterisk joshaidan (~brianj@host-216-26-197-55.tbaytel.net) |
| 18:58.30 | afink | Delemas: is selinux enabled? |
| 18:58.50 | Delemas | It's really weird because when it can't connect, it just times out and nothing is shown in the logs. uhm I don't think selinux is enabled but I'll check... |
| 18:59.02 | afink | I've had problems with selinux doing strange things even though the firewall is disabled / configured correctly |
| 18:59.42 | Delemas | selinuxenabled returns 1 which is not enabled... |
| 19:02.35 | Delemas | It's weird... From my LAN (listed as localnet) I can connect a SIP soft phone via EXTIP2 but not EXTIP1. From the Internet I can connect a SIP soft phone via EXTIP1 but not EXTIP2. externalip=EXTIP1 is set. |
| 19:02.54 | *** join/#asterisk oej (~olle@ns.webway.se) |
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| 19:40.01 | [T]ank | if all i want to do is recieve a call from one server and send it to another, avoiding bridged channels, is transfer() the correct command? Or is something else more appropriate? |
| 19:42.15 | leifmadsen | [T]ank: canreinvite |
| 19:42.48 | [T]ank | canreinvite in the sip.conf on both servers, right? |
| 19:43.28 | *** join/#asterisk corretico (~luis@201.201.44.82) |
| 19:49.36 | [T]ank | now with reinvite set... i am getting Failed to authenticate on INVITE to... blah. the call connects, but ends immediatly. if i set canreinvite to no, then the call works. so i know the dialplan is good. it is just something to do with the sip auth or something like that |
| 19:52.09 | jkroon | Delemas, udp always sends with the IP associated with the outbound route :). |
| 19:52.26 | jkroon | it takes some work to get around that and is not in the domain of asterisk. |
| 19:54.48 | jkroon | well, actually i've argued that it can be fixed in asterisk by creating multiple udp sockets - one for each local IP and then sending on the same socket the packet has been received, but this complicates things for rtp tremendously. I've worked around it by writing a routing daemon that sniffs all sip and iax/2 traffic and adjusts the routing table in real time (adds a bunch of /32 destination entries with appropriate src values). |
| 19:57.41 | *** join/#asterisk vfabi (~fabi@h2-92.faust.net.ua) |
| 20:04.13 | *** join/#asterisk golam (4531720a@gateway/web/freenode/ip.69.49.114.10) |
| 20:06.59 | golam | Hi, can anyone help me with asterisk fax issue? problem: asterisk thinks it generated the tif file and sets status=SUCCESSFUL but the tif was not generated in the first place. This problem does not happen all the time, only 1 percent of the time |
| 20:10.15 | Delemas | I finally figured out it was a routing issue. Communication to EXTIP1 was being replied to using EXTIP2 totally confusing the sender. |
| 20:29.08 | *** join/#asterisk Dr-Linux (~Dr-Linux@182.177.181.209) |
| 20:31.38 | Dr-Linux | I uprgaded to 1.6.2.17.x but still asterisk process CPU goes high and user face bad voice quality, this is happening on my 6 servers |
| 20:31.57 | Dr-Linux | I tried alot of things but no fix, anyone suggest me what should i do? |
| 20:33.38 | *** join/#asterisk fhmiv (~fhmiv@c-67-173-205-151.hsd1.ga.comcast.net) |
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| 20:43.44 | Dr-Linux | any comment on my question? |
| 20:45.30 | *** join/#asterisk Deeewayne (~dwayne@75-150-14-49-Atlanta.hfc.comcastbusiness.net) |
| 20:45.30 | *** mode/#asterisk [+o Deeewayne] by ChanServ |
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| 20:53.30 | Dr-Linux | no one active |
| 20:53.48 | *** join/#asterisk GTXComm (~John@72.128.62.30) |
| 20:59.24 | *** join/#asterisk killown (~killown@unaffiliated/killown) |
| 21:01.12 | GTXComm | Hello all |
| 21:01.50 | *** join/#asterisk dimm (~appleworm@unaffiliated/dimm) |
| 21:02.29 | *** join/#asterisk slowfuse (~slowfuse@c-98-247-250-122.hsd1.wa.comcast.net) |
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| 21:14.25 | pabelanger | Dr-Linux: why did you upgrade? Did you test prior to moving into production? |
| 21:18.55 | Dr-Linux | pabelanger: yes, i checked it as well, there was no issue |
| 21:19.48 | Dr-Linux | pabelanger: becasue 1.6.1.0 is old and some other issues, i've asterisk support as well, they said they do not support older version |
| 21:19.50 | Dr-Linux | so i upgrade |
| 21:20.12 | [T]ank | no matter what i set up. the call will not reinvite. i have a simple exten => 11839,1,Dial(SIP/{EXTEN}@ICR) that is the only line. in sip.conf on both servers i have canreinvite set to yes and nat set to no. i have insecure=invite. what could be keeping this from doing a reinvite? Is there a way to force it? |
| 21:21.15 | pabelanger | Dr-Linux: lots of things affect CPU. EG: transcoding audio |
| 21:22.34 | pabelanger | [T]ank: s/canreinvite/directmedia |
| 21:22.37 | Dr-Linux | so what you suggest? |
| 21:22.40 | *** join/#asterisk TimeRider (~steve@188-220-33-123.zone11.bethere.co.uk) |
| 21:23.30 | pabelanger | Dr-Linux: what is your system doing? |
| 21:23.49 | [T]ank | @pabelanger: not sure what you just told me to do. |
| 21:24.18 | pabelanger | [T]ank: canreinvite is not an option in sip.conf, it has been replaced with directmedia |
| 21:24.34 | [T]ank | directmedia=yes? |
| 21:24.37 | pabelanger | well, it depends on which version of Asterisk you are using |
| 21:24.51 | *** join/#asterisk cusco_ (~tralala@a89-152-96-250.cpe.netcabo.pt) |
| 21:24.53 | cusco_ | hi |
| 21:25.00 | cusco_ | I just compiled a copy of latest 1.6 |
| 21:25.18 | cusco_ | module load cdr_csv.so |
| 21:25.26 | cusco_ | Unable to load module cdr_csv.so |
| 21:25.26 | pabelanger | [T]ank: yes, it is enabled by default. So if your endpoints support it, media will go directly between end points |
| 21:25.27 | [T]ank | 1.4 |
| 21:25.32 | cusco_ | Command 'module load cdr_csv.so ' failed. |
| 21:25.34 | cusco_ | why? |
| 21:26.39 | cusco_ | :( |
| 21:27.15 | pabelanger | cusco_: is the module compiled? |
| 21:27.19 | [T]ank | @pabelanger: so, i have enabled that, and i am still doing bridging when i make the call. |
| 21:28.01 | pabelanger | [T]ank: do you endpoints support reinvites? |
| 21:28.10 | pabelanger | s/you/your/ |
| 21:29.13 | [T]ank | the call is coming into asterisk from an Avaya PBX which supports reinvites. from there, the call is sent to a second asterisk server. so Avaya->asterisk1->asterisk2. i want the end result to be avaya->asterisk2 |
| 21:30.54 | pabelanger | [T]ank: how do you know RTP is not being redirected from Ayaya to asterisk2? |
| 21:31.20 | pabelanger | ~connectdebug |
| 21:31.25 | pabelanger | ~collectdebug |
| 21:31.26 | infobot | [collectdebug] a method of collecting logs allowing others help troubleshoot an issue. Refer to https://wiki.asterisk.org/wiki/display/AST/Collecting+Debug+Information |
| 21:31.26 | [T]ank | well... i may not understand at all what i am doing here... my end goal is that the channels are freed from asterisk1. while the call is connected, i do a core show channels and see the bridged channels on asterisk1 |
| 21:32.15 | pabelanger | [T]ank: Yes, that channel will be bridged, because you are using Dial(), if you want Asterisk1 to be removed from the call control, you need to use Transfer() |
| 21:32.36 | pabelanger | bridging channels does not mean RTP is also bridged |
| 21:33.31 | [T]ank | ok... getting bounced back and forth a bit. was trying to do transfer and someone told me that i should be doing reinvites. |
| 21:33.51 | pabelanger | [T]ank: You want to do a SIP REFER, not a REINVITE |
| 21:34.12 | [T]ank | so... what i am running into is while dial is working perfectly... if i change just the command from dial to transfer with no other changes, it sais status unknown like the other server is offline |
| 21:35.00 | [T]ank | getting the exact error |
| 21:35.02 | [T]ank | sec |
| 21:35.08 | pabelanger | [T]ank: Yes, so now you need to make sure your Avaya support a SIP REFER. |
| 21:35.58 | [T]ank | hmmm ok. |
| 21:36.06 | [T]ank | well, i already copied it, so here is the error: |
| 21:36.24 | [T]ank | <PROTECTED> |
| 21:36.25 | [T]ank | <PROTECTED> |
| 21:37.22 | pabelanger | [T]ank: That does not tell us much, you'll need to get a debug log, with 'sip set debug on' to see what is happening |
| 21:37.24 | pabelanger | ~collectdebug |
| 21:37.25 | infobot | i heard collectdebug is a method of collecting logs allowing others help troubleshoot an issue. Refer to https://wiki.asterisk.org/wiki/display/AST/Collecting+Debug+Information |
| 21:37.32 | pabelanger | [T]ank: ^ good place to start |
| 21:37.59 | [T]ank | yeah... the network is so locked down i cant pastebin them up. :-( |
| 21:38.29 | wdoekes2 | wow.. you get irc but not http? |
| 21:38.42 | pabelanger | [T]ank: not much we can do then |
| 21:39.12 | *** join/#asterisk killown (~killown@unaffiliated/killown) |
| 21:39.20 | [T]ank | yeah, i know... thats the part that really sucks. Could have probably already had this resolved. gonna have to wait till i am home and on my own network. |
| 21:39.44 | [T]ank | but... thanks for the help so far... looking into the SIP REFER for Avaya |
| 21:43.12 | *** join/#asterisk chandoo (~chandoo@ool-4a5a2824.dyn.optonline.net) |
| 21:43.16 | *** join/#asterisk FiReSTaRT (~dlyh@unaffiliated/firestart) |
| 21:45.29 | *** join/#asterisk dimm (~appleworm@unaffiliated/dimm) |
| 21:47.01 | *** join/#asterisk Failrar (~Failrar@5ED66E6D.cm-7-7b.dynamic.ziggo.nl) |
| 21:52.30 | jdoe | hey, can someone tell me why this happens? |
| 21:52.47 | jdoe | [Apr 19 14:28:51] NOTICE[29266]: channel.c:4046 __ast_read: Dropping incompatible voice frame on IAX2/vancouver-746 of format slin since our native format has changed to 0x2 (gsm) |
| 21:52.50 | cusco_ | pabelanger: yes its compiled |
| 21:53.00 | jdoe | both ends of the trunk are 1.8.3, both have disallow=all, allow=gsm |
| 21:53.05 | cusco_ | I just compiled for the first time in this machine |
| 21:53.22 | cusco_ | its there in /usr/lib/asterisk/modules with right permissions and ownership |
| 21:55.13 | Qwell | cusco_: add a noload line for it in modules.conf, then load it manually after startup. any errors? |
| 21:56.35 | Qwell | If there are errors, fix them and remove the noload. If not, I don't see why you're trying to load it manually. |
| 22:00.22 | devil_evoxxx | anyone know how i can power a isdn phone connected on a B410PF Digium Card? |
| 22:04.38 | devil_evoxxx | have you got any idea for testing? |
| 22:05.47 | *** join/#asterisk killown (~killown@unaffiliated/killown) |
| 22:07.02 | *** join/#asterisk zerohalo (~zerohalo@c-66-31-42-200.hsd1.ma.comcast.net) |
| 22:08.54 | cusco_ | Qwell: same happens |
| 22:09.01 | cusco_ | Unable to load module cdr_csv.so |
| 22:09.01 | cusco_ | Command 'module load cdr_csv.so ' failed. |
| 22:10.43 | *** join/#asterisk marlowe (~marlowe@ip68-100-150-54.dc.dc.cox.net) |
| 22:10.44 | pabelanger | cusco_: no warnings or errors? Did you check /var/log/asterisk/messages |
| 22:10.51 | pabelanger | does cdr.conf exists? |
| 22:12.36 | cusco_ | pabelanger: yes I did |
| 22:18.19 | cusco_ | ok its something to do with configurations |
| 22:18.28 | cusco_ | I just deleted all /etc/asterisk/* and make samples |
| 22:18.32 | cusco_ | and it loads now |
| 22:18.41 | cusco_ | how do I know wich configuration it does not like?? |
| 22:19.46 | *** join/#asterisk fhmiv (~fhmiv@adsl-065-005-242-247.sip.ags.bellsouth.net) |
| 22:22.03 | leifmadsen | cusco_: I'd copy the cdr.conf.sample file from the source over top of the one that is there and try it with the stock file |
| 22:23.10 | cusco_ | I also have cdr_mysql module tho not using it.. |
| 22:27.26 | *** join/#asterisk zong123 (4531720a@gateway/web/freenode/ip.69.49.114.10) |
| 22:29.40 | cusco_ | leifmadsen: ok some configuration in cdr.conf right |
| 22:30.35 | cusco_ | o.O |
| 22:32.33 | cusco_ | but.. |
| 22:32.39 | cusco_ | all I have in old cdr.conf is: |
| 22:32.40 | cusco_ | [general] |
| 22:32.41 | cusco_ | enable=yes |
| 22:32.41 | cusco_ | unanswered = no |
| 22:32.41 | cusco_ | endbeforehexten=yes |
| 22:32.56 | cusco_ | what is wrong? |
| 22:33.06 | cusco_ | ah the space round = |
| 22:33.07 | cusco_ | <PROTECTED> |
| 22:33.18 | leifmadsen | no idea what is wrong |
| 22:33.23 | leifmadsen | that's why I suggested just using a stock template |
| 22:34.33 | cusco_ | ow I need the [csv] stuff |
| 22:34.37 | cusco_ | seems like |
| 22:34.41 | cusco_ | thanks for that leifmadsen |
| 22:34.54 | leifmadsen | always bring it back to basics |
| 22:35.24 | cusco_ | hehe |
| 22:35.45 | cusco_ | I just want to use the cdr billsec and duration on thye h => exten |
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