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| 02:29.27 | floren | hi all :) |
| 02:29.39 | floren | i have a quick question about g722 |
| 02:29.59 | floren | to enable it all i have to do is add it to allow= i presume |
| 02:30.13 | floren | in both sip.conf and voicemail.conf |
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| 03:24.24 | igcewieling | yes, assuming your endpoints support g722 |
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| 04:37.33 | floren | thank you |
| 04:37.42 | floren | i have another issue with srtp :( |
| 04:37.53 | floren | while compiling, i get: The SRTP installation appears to be missing or broken |
| 04:38.02 | floren | libsrtp-devel 1.4.4 is installed |
| 04:38.34 | floren | asterisk is looking for srtp/srtp.h |
| 04:39.03 | floren | do i have to soft link the binary into addons/srtp? |
| 04:39.56 | floren | i'm making with %{__make} %{?_smp_mflags} ASTCFLAGS="%{optflags} -fPIC" |
| 04:40.23 | floren | -PIC should be enough to compile, but it does not find the library |
| 04:45.39 | ChannelZ | is it at the link stage? |
| 04:47.16 | floren | http://codepad.org/01HqaEs7 |
| 04:47.34 | floren | it finds srtp_init |
| 04:47.49 | floren | but look at the srtp/srtp.h, bob |
| 04:48.31 | floren | i guess i have to soft link it in a dir? |
| 04:49.12 | floren | # find /usr -name srtp.h |
| 04:49.12 | floren | /usr/include/srtp/srtp.h |
| 04:49.12 | floren | /usr/include/openssl/srtp.h |
| 04:49.26 | floren | is there the lib |
| 04:49.35 | floren | but asterisk does not see it |
| 04:51.06 | floren | the includedir is ok --includedir=/usr/include |
| 04:51.14 | ChannelZ | interesting it says (cached) no |
| 04:51.59 | floren | any trick in your sleeve to get it rolling |
| 04:52.10 | floren | i dont see anything wrong |
| 04:52.54 | ChannelZ | Do you have a config.cache in your build dir? |
| 04:53.05 | floren | let me check |
| 04:53.22 | floren | no |
| 04:53.36 | floren | config.guess is the first |
| 04:54.59 | floren | my options: http://codepad.org/fRRkAmvO |
| 04:55.33 | floren | they are in alphabetical order |
| 04:56.26 | ChannelZ | Dunno, I'd 'make clean' and then reconfigure and see if it finds the header correctly |
| 04:56.45 | floren | ya |
| 04:56.54 | floren | a real headache :) |
| 04:57.09 | floren | i got everything neat except srtp |
| 04:57.24 | floren | i guess that's why all distros dropped that option |
| 04:57.56 | floren | or maybe because the encrytion restrictions |
| 04:58.38 | ChannelZ | make clean and configure is a headache? |
| 04:58.57 | ChannelZ | D'oh. The Ubuntu forums got hacked. |
| 04:59.00 | floren | no not at all |
| 04:59.10 | floren | heh let me see |
| 04:59.21 | floren | they run vbulletin, is normal they get hacket |
| 04:59.36 | floren | vbulletin is probably the most insecure software you have out there |
| 05:02.23 | ChannelZ | or phpBB. or Wordpress. or... |
| 05:02.30 | floren | ya |
| 05:02.38 | floren | i think the most secure is xenforo |
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| 05:11.28 | hanuman | hi |
| 05:12.24 | hanuman | how can i write asterisk manual installation script in ubuntu, can you give any sugessions |
| 05:13.55 | ChannelZ | ./configure;make;make install |
| 05:15.52 | hanuman | k that is after installing dependencies , diownloading asterisk then doing those things but all dependencies install through script only |
| 05:17.23 | ChannelZ | well minimally you'd need gcc, g++, make, libcurses-dev, libxml2-dev, libsqlite3-dev, libssl-dev plus others depending on what other features you might build |
| 05:17.39 | ChannelZ | libuuid-dev now in 11.5 |
| 05:19.14 | hanuman | that all are installing through sudo apt-get .. using commnd prompt |
| 05:19.27 | ChannelZ | sure |
| 05:19.33 | hanuman | but i want to install all this using one script |
| 05:20.05 | ChannelZ | although this is less of an "installer" than an automated build |
| 05:20.36 | hanuman | yeah |
| 05:20.49 | hanuman | i know python can i write that in python |
| 05:20.55 | ChannelZ | if you just want to build a newer package than they provide, well, grab the source package and dig around for their build options |
| 05:21.25 | ChannelZ | then you could just build a package of your own and shove it somewhere and install it on wherever |
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| 05:43.20 | hanuman | how can i wirte automatic asterisk installation script |
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| 05:46.34 | hanuman_ | how can i write asterisk autimated installation script in ubuntu |
| 05:47.37 | hanuman_ | how can i write asterisk autimated installation script in ubuntu |
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| 06:36.18 | bulkorok | hi |
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| 08:10.02 | danfromuk | Hi. I understand that nat=yes is going. Whats the replacement for it? |
| 08:10.28 | Gugge | where did you see it was going? |
| 08:10.41 | Chainsaw | The changes document for 1.8 I suspect. |
| 08:10.52 | danfromuk | The CLI shows a warning |
| 08:10.59 | danfromuk | " sip_parse_nat_option: nat=yes is deprecated" |
| 08:11.21 | Gugge | and what does the sample sip.conf in the same version say about nat= ? |
| 08:12.14 | Gugge | or the changes file? :) |
| 08:12.21 | danfromuk | gives a few options but its not obvious which one I should use. |
| 08:12.28 | danfromuk | changes or upgrade file? |
| 08:12.31 | Gugge | or the "New in 11" on the wiki :) |
| 08:12.38 | danfromuk | i didnt see any mention in the upgrade file. |
| 08:12.57 | danfromuk | ill check out the wiki |
| 08:13.30 | Gugge | the CHANGES file mentions nat=yes :) |
| 08:13.32 | danfromuk | Got it |
| 08:13.36 | Gugge | and so does the wiki :) |
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| 08:42.16 | izbushka_ | hi |
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| 08:44.10 | izbushka_ | I don't understand what "n" option does in Queue application. No retries on the timeout... What's the difference between Queue(manager,n) and Queue(manager)? |
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| 11:06.20 | jm|laptop | hello :) |
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| 11:10.51 | jm|laptop | I'm using 1.8 with perl AGI. When I use exec("Dial","xyz"); I am seeing the correct CALLERID(name) - in this case from GSM network (i.e. calling in from my mobile phone) it's using my TO header in CALLERID(num), I think |
| 11:11.14 | danfromuk | Hi, I've just upgraded from 1.8 to 11 and now one sippeer can't seem to connect. Its sending the REGISTER and seems to be doing things properly by supplying the authorisation details, but asterisk doesnt seem to process the login. |
| 11:11.21 | danfromuk | Here is the SIP debug. http://pastebin.com/y5zn0iXT |
| 11:12.55 | jm|laptop | e.g. Receiving new incoming call from "07712777777" <sip:linphone@81.8.1.8:5060>, assigned id 12 |
| 11:15.53 | danfromuk | Does anyone know what the problem could be? It doesnt appear to be a nat issue. |
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| 11:27.17 | jm|laptop | RST. It was linphone being stupid. |
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| 11:30.31 | MikeH | Hi, my messages log seems to have been erased on reboot - is this stored/moved elsewhere or simply wiped? I'm trying to hunt down an issue where SIP trunks go down overnight and requires a reboot to get them back. |
| 11:31.27 | danfromuk | MikeH, check you havent done anything with logrotate that could be deleting logs each night. |
| 11:32.38 | MikeH | I've not - this is a third party asterisk based system, but their support is useless. There is no rotate strategy in logger.conf |
| 11:33.23 | danfromuk | Rotate is not in logger.conf |
| 11:33.38 | danfromuk | Look at /etc/logrotate.conf or /etc/logrotate.d/ |
| 11:37.09 | MikeH | rotate 1 |
| 11:37.37 | MikeH | I ought to up that, but there is nothing rotated currently. I wonder if there is a script that empties them on boot |
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| 11:40.14 | MikeH | oh hang on, /var is mounted on loopback :/ |
| 11:40.43 | MikeH | oh wait, /var/log is tmpfs - that's helpful |
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| 12:17.19 | Cuzner | Wow nickserv is SLOW today. |
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| 12:38.00 | BorjaGVO | Hi guys. Where or how can I see what information is stored inside astdb? |
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| 12:41.40 | leifmadsen | BorjaGVO: http://www.asteriskdocs.org |
| 12:42.02 | leifmadsen | BorjaGVO: http://asteriskdocs.org/en/3rd_Edition/asterisk-book-html-chunk/asterisk-CHP-6-SECT-6.html |
| 12:42.18 | BorjaGVO | Thank you |
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| 12:59.25 | Cuzner | smacks leifmadsen on the ass |
| 12:59.33 | leifmadsen | w0000! |
| 12:59.44 | leifmadsen | backhands Cuzner across the chest like Rick Flair |
| 13:00.16 | jmetro | Wooo! |
| 13:00.22 | Cuzner | k, someone really needs to clean up the gateway repos |
| 13:00.32 | leifmadsen | Cuzner: don't get me started :D |
| 13:03.34 | Cuzner | seriously, which one is it? core, core-1.8, core-1.8a, core-gateway, core-gateway-1.8, core-trunking? |
| 13:03.35 | Cuzner | wtf |
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| 13:07.46 | psanders | @BorjaGVO: What version of asterisk are you using? |
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| 13:16.03 | julgr | I'm running into an issue with T.38. Wondering if anyone knows of a patch to have T.38 calls use the same port as the initial voice session of a call. |
| 13:19.46 | file | I know of no patch for Asterisk to have it do that, but if the remote side uses the same port we don't care |
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| 13:41.53 | waj | When dialling through a SIP trunk the From address contains: "XXXXXX@anonymous.invalid". Is there a way to change that? I tried with fromuser and fromdomain already. |
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| 13:45.08 | [TK]D-Fender | your caller presentation is blocked |
| 13:45.54 | [TK]D-Fender | https://wiki.asterisk.org/wiki/display/AST/Function_CALLERPRES |
| 13:48.20 | LooserOuting | if a caller sends an INVITE with only ilbc mode 20, and asterisk ist compiled with ilbc mode 30, asterisk will respond with 200 OK and ilbc mode 30 in SDP. Is this the right behavior ? |
| 13:48.54 | LooserOuting | What happens wehn one asterisk compiled with ilbc mode 20 calls another asterisk with ilbc mode 30 ? |
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| 13:57.59 | [404] | I am currently working on a pre-existing system which I am running test against. I am having issues getting the dtmf from the answering side to the caller side. Would anyone happen to know any quick configs for me to check |
| 13:59.44 | [TK]D-Fender | My guess... the configs of the devices that you are using |
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| 14:02.01 | [404] | ok, will check it out, using linphone |
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| 14:10.28 | *** mode/#asterisk [+o sruffell] by ChanServ |
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| 14:34.37 | floren | hi all |
| 14:35.21 | floren | if you have a chance, please have a look at my srtp compiling issues: http://forums.asterisk.org/viewtopic.php?f=1&t=87434 |
| 14:39.16 | jmetro | why srtp? |
| 14:40.04 | floren | is a challenge to get it working properly, that's all |
| 14:40.11 | jmetro | also your error tells you what failed |
| 14:40.11 | floren | sop i learn more about it |
| 14:40.36 | floren | ya it tells me, but the libs are there |
| 14:40.55 | leifmadsen | floren: pretty sure ATDG has the documentation about how to compile in the srtp libraries |
| 14:41.06 | floren | checking srtp/srtp.h usability... no |
| 14:41.14 | floren | is it related to version? |
| 14:41.24 | floren | running latest CVS srtp |
| 14:41.25 | igcewieling | maybe you are missing the headers? |
| 14:41.44 | floren | no, the headers are there |
| 14:42.18 | igcewieling | floren: I guess you need to find out what about /usr/include/srtp/srtp.h which Asterisk does not like |
| 14:42.20 | floren | hi leifmadsen, i'm not familiar what ATDG means |
| 14:42.41 | [TK]D-Fender | ~book |
| 14:42.41 | infobot | Asterisk: The Definitive Guide, 4th Edition (ISBN 1-4493-3242-0) available at http://oreilly.com/catalog/0636920025894 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and a version is available for reading online at http://www.asteriskdocs.org/ or see ~buybook |
| 14:43.15 | Cuzner | is a founding member of the leifmadsen fanclub |
| 14:43.35 | Cuzner | membership is only a one-time payment of $600 payable directly to me. |
| 14:44.12 | floren | ahh, i see |
| 14:44.15 | Cuzner | i accept litecoins - LakNAWNT742VqJfMiKw8s7jMqthhEFm3wz |
| 14:44.28 | slav3_kitten | i'm only charging 580 |
| 14:44.30 | jmetro | Cuzner: is it waivable if we sit with you on that black leather couch in your office? |
| 14:44.34 | floren | i read that already, they only mention: ensure that you have both OpenSSL and LibSRTP installed |
| 14:44.43 | floren | which i did |
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| 14:45.37 | *** mode/#asterisk [+o pabelanger] by ChanServ |
| 14:46.26 | Cuzner | jmetro: it's actually a loveseat, and no the membership fee is unwaivable. |
| 14:47.02 | jmetro | Cuzner: i'm gonna need that tape back <.< |
| 14:47.19 | *** join/#asterisk ChkDigit (~u388mw@74.3.144.66) |
| 14:47.40 | *** join/#asterisk dash_ (~d45h@unaffiliated/dash-/x-7576607) |
| 14:48.29 | *** join/#asterisk polysics (~Adium@host209-73-dynamic.45-79-r.retail.telecomitalia.it) |
| 14:48.33 | polysics | hello |
| 14:48.38 | floren | the libsrtp rpm I built are made with centos standards and pkgconfig, shareable. i tried to open this as bug report but it was refused |
| 14:48.54 | *** part/#asterisk waj (~Adium@host78.181-1-46.telecom.net.ar) |
| 14:48.54 | polysics | does a semi-official/maintained repo for Ubuntu exist? |
| 14:49.10 | [TK]D-Fender | Maintained... by them. Official? Not by us. |
| 14:49.12 | floren | on my side, there is no way to debug the libsrtp issue |
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| 14:50.29 | polysics | [TK]D-Fender: I mean, can I avoid having to make my own Asterisk 11 pckages somehow? |
| 14:50.41 | [TK]D-Fender | polysics: I don't see any others... soo... |
| 14:50.43 | Weezey | I don't think directmedia=no is working on my systems, is there a CLI command to verify it's being picked up from the Realtime config? |
| 14:52.00 | igcewieling | polysics: you could do what almost everyone here does an install from source. |
| 14:52.19 | igcewieling | Weezey: "sip show settings" "sip show peer X" |
| 14:52.46 | polysics | production machines don't run build tools |
| 14:53.29 | Weezey | Direct RTP setup? = directmedia ? |
| 14:53.55 | floren | rtp is related packets enxryption |
| 14:54.34 | mjordan | directmedia refers to whether or not Asterisk will re-negotiate the media stream to flow between two SIP endpoints once a call is established. directrtpsetup is something else and is slightly weird and strange. |
| 14:54.34 | floren | there is a good tutorial on asterisk wiki but they explain everything except how to properly install srtp |
| 14:55.16 | Weezey | mjordan: yeah, I want make sure asterisk stays in line and does not re-invite. |
| 14:55.20 | floren | from what it looks, asterisk is dependent on a specific version of srtp, which it should not, that explains why i created the bug |
| 14:55.43 | Weezey | having an issue where users on one box who were migrated from another are being rejected when they call the other. |
| 14:56.22 | Weezey | (Failed to authenticate on INVITE to ...) |
| 14:56.37 | igcewieling | Weezey: you are looking in the wrong place. |
| 14:56.46 | igcewieling | you would not get that error on a simple reinvite issue |
| 14:57.01 | *** part/#asterisk polysics (~Adium@host209-73-dynamic.45-79-r.retail.telecomitalia.it) |
| 14:57.16 | [TK]D-Fender | indeed |
| 14:57.43 | Weezey | igcewieling: dang. I know I can delete the old account on the box they are calling and it will go throuhg |
| 14:57.45 | Weezey | through |
| 14:58.00 | Weezey | I'd prefer to have it there as a backup so I can pop them back to it should I need to though. |
| 14:58.14 | igcewieling | you didn't do something silly like assign the old and the new IP to the same hostname, did you? |
| 14:58.22 | Weezey | no |
| 14:58.29 | Weezey | hmm |
| 14:58.37 | Weezey | maybe in asterisk it is |
| 14:58.43 | Weezey | nice catch. |
| 14:58.48 | Weezey | in real DNS it isn't |
| 14:59.15 | igcewieling | we had a problem where we set blah-our.domain.in to have multuple a-records. when Asterisk reinvited (for direct media or because of an expired session) the packets sometimes went to the wrong server. |
| 15:00.39 | Weezey | it shouldn't re-invite though, I would think that the call should be from the server to the extension, not from the user's peer to the extension |
| 15:01.46 | Weezey | I want the server to stay in the call. |
| 15:02.15 | igcewieling | Weezey: reinvites happens for many reasons. They are only a problem if it sets up direct media |
| 15:03.09 | *** join/#asterisk caveat- (hoax@shell.bshellz.net) |
| 15:04.43 | Weezey | so how do I make it allow the call in? I changed the fromdomain setting and it's still the same thing. |
| 15:05.24 | Weezey | (other than deleting all of the migrated peers) |
| 15:06.13 | Katty | hello my asterisk does not work at all how to fix plz |
| 15:06.21 | leifmadsen | Katty: I think you have to press then any key |
| 15:06.59 | Weezey | Katty: did you dial the reset sequence once it first booted? |
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| 15:14.58 | Weezey | peer vs. friend? |
| 15:20.03 | Weezey | looks like that was it |
| 15:24.40 | Weezey | any harm in things being peers vs friends? |
| 15:32.43 | Katty | you have to help friends move. |
| 15:32.51 | Katty | and console them when their significant other rips their heart out, with beer. |
| 15:32.57 | mic_ | (rotfl) |
| 15:32.58 | Katty | peers, not so much. you can just ignore them. |
| 15:34.13 | mic_ | Hello. I deployed today a small modification in the dialplan - MixMonitor() namely |
| 15:34.44 | mic_ | and people started complaining about some mysql issues |
| 15:35.23 | mic_ | do you have any funny i/o bottleneck experiences with recording calls? |
| 15:35.59 | Greenlight | I always record to a ramdisk |
| 15:36.16 | Greenlight | To avoid any IO issues effecting the system |
| 15:36.30 | Greenlight | Otherwise you'll thrash the hell outta the disks |
| 15:36.42 | mic_ | Greenlight: I plan to switch it off tomorrow completely |
| 15:36.44 | Greenlight | Far better to hit the disk only once for the whole call when it completes |
| 15:37.00 | mic_ | Greenlight: just to see how it's going to react |
| 15:37.20 | mic_ | and right now I am trying to gather some more data (been to google obviously already ;) |
| 15:37.25 | Greenlight | If you've a single spindle of spinning rust, and are recording calls directly to it, AND accessing mysql from there.... |
| 15:37.36 | Greenlight | World of pain |
| 15:37.40 | mic_ | I smell |
| 15:37.44 | mic_ | well, to make it worse |
| 15:37.53 | mic_ | there's a lot of mysql activity, call recording |
| 15:37.58 | mic_ | then they are compressed to ogg |
| 15:38.11 | mic_ | and all on a pair of RAID1 drives. |
| 15:38.24 | Greenlight | decent BBU controller ? |
| 15:38.27 | Greenlight | or softraid? |
| 15:38.33 | mic_ | softraid, centos 6.4 64bit |
| 15:38.39 | Greenlight | More pain |
| 15:38.43 | mic_ | yep, that I know |
| 15:38.51 | mic_ | ok - I will certainly switch it off. |
| 15:38.58 | mic_ | (and let us hope that was it) |
| 15:39.03 | Greenlight | No need to turn if off, just write to ramdisk |
| 15:39.08 | Greenlight | Then when the call ends move to disk |
| 15:39.27 | Greenlight | The volume of data is quote low, it's the frequent small writes |
| 15:40.10 | Greenlight | We record hundreds of calls simultanously to ramdisk, and move to proper disk after, and no issues |
| 15:40.48 | Greenlight | Leave the mp3 conversion to overnight though |
| 15:40.54 | mic_ | Do you recommend moving them all at once, say every 30 minutes? |
| 15:40.55 | WIMPy | Greenlight: You onviousely don't have users making 12h calls. |
| 15:41.04 | Greenlight | WIMPy: No, I don't :) |
| 15:41.12 | Greenlight | But still, we've 192Gb of RAM on that box... |
| 15:41.28 | Greenlight | mic_: Get them outta ram ASAP imo |
| 15:41.31 | WIMPy | Ok, even that should fit then. |
| 15:41.40 | Greenlight | We do a system() inside the dialplan to move them |
| 15:41.43 | [TK]D-Fender | Do it at end-of-call |
| 15:41.49 | mic_ | Greenlight: ok, that's how it works right now - python scripts moves them to a different place once the recording is ready. |
| 15:42.08 | Greenlight | Literally, StopMixMonitor(), then System(mv ....) |
| 15:42.17 | WIMPy | You can use rsync to limit bw. |
| 15:42.36 | mic_ | Well, I am also throwing some garbage to the database |
| 15:42.55 | mic_ | so I will stick with that python script - but it's executed just after. |
| 15:43.07 | mic_ | thanks a lot, I will implement this on the dev box in a moment. |
| 15:43.19 | Greenlight | Good luck, hope it sorts your issue! |
| 15:45.00 | mic_ | And encoding to ogg |
| 15:45.06 | mic_ | could simply be done overnight anyway. |
| 15:45.20 | Greenlight | Depends on volume of calls, but yea |
| 15:45.32 | mic_ | it's not that bad - ca. 1000 calls per working day |
| 15:45.45 | WIMPy | If you have an issue with i/o but not with CPU, it might help to do it immediately. |
| 15:45.59 | Greenlight | I'd be tempted to just "nice" encode them to ogg/mp3 |
| 15:46.08 | mic_ | hehe |
| 15:46.10 | mic_ | (sec.) |
| 15:46.24 | WIMPy | Yes, good use for nice. |
| 15:46.24 | mic_ | os.system("/bin/nice /usr/bin/oggenc -o %s/%s.ogg %s/%s" % (target_dir,dest_file,record_dir,file_path)) |
| 15:46.31 | Greenlight | Yes ^^ |
| 15:46.32 | mic_ | I thgought about it :) |
| 15:46.49 | [TK]D-Fender | mic_: How many simultaneous calls do you run? |
| 15:46.53 | mic_ | 6-8 |
| 15:47.07 | mic_ | and there could be some people in the queue. |
| 15:47.15 | [TK]D-Fender | mic_: Can't believe this load should require you to delay transcoding like that |
| 15:47.38 | Greenlight | That's what we're saying; just to it immediately |
| 15:47.49 | [TK]D-Fender | mic_: end of call script to convert and off-load the recording should be more than fine. |
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| 15:48.36 | WIMPy | Might be a good idea to queue stuff. But you need to monitor that hen as well. |
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| 15:48.57 | mic_ | I am not so worried about CPU here - quad xeon |
| 15:49.12 | Greenlight | Let "nice" and the scheduler take care of all that |
| 15:49.28 | Greenlight | At that volume of calls he won't have issues |
| 15:49.38 | mic_ | ok, more coffee and I'll be hacking that on the dev box. |
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| 15:57.17 | mic_ | Greenlight: are you using ramdisk set during boot or tmpfs? |
| 15:57.35 | Greenlight | A mix between /dev/shm and ones created at boot time |
| 15:57.58 | Greenlight | To be honest, /dev/shm works fine, I couldn't see any performance difference between different methods |
| 15:59.07 | Greenlight | (That uses tmpfs) |
| 15:59.51 | mic_ | Ok |
| 16:01.04 | Greenlight | If you want more control over the size and permissions etc, you can do something like http://pastebin.com/BSCqLH9c |
| 16:01.09 | Greenlight | At startup |
| 16:04.46 | mic_ | Nice - I am currently checking tmpfs |
| 16:04.49 | mic_ | as it does not require reboot. |
| 16:05.55 | Greenlight | Also - we have a script that runs nightly to move anything that might somehow have got left in the ramdisk to proper hard disk |
| 16:06.01 | Greenlight | Just incase :) |
| 16:06.28 | Greenlight | We've a folder called "orphanms" where those lost calls get put |
| 16:06.32 | Greenlight | *orphans |
| 16:07.12 | Greenlight | And every now and again IT can save the day for QA by retrieving a call.... |
| 16:07.57 | mic_ | :) |
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| 16:11.48 | mic_ | ok |
| 16:11.58 | mic_ | tmpfs, ogg+nice - works on both dev & production. |
| 16:14.17 | Greenlight | Excellent |
| 16:14.24 | Greenlight | Should alleviate your IO issues |
| 16:14.38 | mic_ | I hope so, but the real answer will come tomorrow |
| 16:14.43 | mic_ | during peak hours. |
| 16:15.38 | Greenlight | It will :) |
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| 16:54.51 | wolrah | Is there a generally accepted "right" way to do redundancy in Asterisk for a F/OSS setup? Just looking for a pointer to where to start, I can figure it out from there, but all my searches have turned up commercial stuff and outdated discussions from 2006. |
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| 16:57.50 | navaismo | what is F/OSS? |
| 16:57.55 | navaismo | sorry my ignorance |
| 16:58.00 | wolrah | free/open source |
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| 16:58.26 | leifmadsen | wolrah: define "redundancy" |
| 16:58.41 | Mango45 | What level of redundancy are you after? Are you looking for "server A goes down, phone automatically registers to server B"? |
| 16:59.05 | wolrah | leifmadsen: I'd be fine with two servers that sync dialplans and phones could register to either functionally |
| 16:59.06 | navaismo | wolrah, thanks, and yes what kind of solution you are looking? |
| 16:59.33 | leifmadsen | wolrah: look at LinuxHA for a hot-swap solution, and a shared FS like MooseFS |
| 16:59.42 | mic_ | wolrah: and pacemaker |
| 16:59.50 | jmetro | "my existing phone, server, and cable melt but i can still make calls" |
| 16:59.55 | leifmadsen | then deploy with something like Chef or Puppet for reproduceability |
| 16:59.58 | Cuzner | MooseFS! |
| 17:00.17 | Cuzner | no atime patch |
| 17:01.03 | leifmadsen | it's pretty easy to keep the systems mostly in sync though if you deploy with DevOps |
| 17:01.24 | leifmadsen | you might need to rsync voicemail or something though using something like unison and such |
| 17:01.24 | mic_ | (that MooseFS looks really _good_( |
| 17:01.35 | leifmadsen | depends how you plan to sync the data |
| 17:01.58 | Cuzner | mic_: it works pretty well |
| 17:02.26 | wolrah | I figured I'd use IMAP-hosted voicemail and just handle keeping that available separately |
| 17:03.26 | leifmadsen | could do that as well |
| 17:03.38 | wolrah | I used to do that in a former Broadsoft-adminning life, so that was the familiar answer for that problem |
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| 17:12.53 | wolrah | if I'm not looking to use the seamless virtual IP setup and am happy with the results of DNS SRV priority, is the cluster stuff even needed or just the shared FS? |
| 17:15.34 | mic_ | you probably can do your own sync and some state maintenance |
| 17:16.05 | mic_ | but for example pacemaker comes with a really good framework, where you can just use building blocks instead. |
| 17:17.16 | wolrah | ok, i'll keep that in mind |
| 17:18.20 | leifmadsen | if you're thinking of just using multiple systems in parallel, then you might need to front the asterisk boxes with a proxy of some sort like kamailio |
| 17:20.37 | wolrah | would that be to make the transition smoother by allowing the phones to not see the registration changes? |
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| 17:35.12 | jmetro | i think thats the goal there wolrah, they can bounce to any asterisk server in the cluster and not know which one it is. |
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| 17:35.32 | [404] | [TK]D-Fender: Turns out with the issue of dtmf not being detected, once i setup the dtmfmode set to auto, everything worked fine. For some reason it didn't like rfc2833 coming from linphone or ekiga |
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| 18:00.25 | cmendes0101 | Aside from dialstatus using dial(). Is there a way to return the billsec for that dial attempt? CDR(billsec) seems to be the current channels but I need the Dials atttempt |
| 18:06.24 | jmetro | arent there no billsec until the call is connected anyway. |
| 18:09.21 | cmendes0101 | Well not of that channel but the outgoing call with Dial. I think I might have found something. looking in to it |
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| 18:10.56 | igcewieling | ANSWEREDTIME is what you are looking for |
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| 18:14.27 | navaismo | it show the actual time of the call right? I was thinking he wants the time that dial take before the answer |
| 18:15.00 | igcewieling | The time from answer to hangup, yes |
| 18:15.36 | igcewieling | if you want to know the predial time then set a variable with the current timestamp, then compare that with dialedtime - answeredtime |
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| 18:57.03 | cmendes0101 | igcewieling: Thanks, that worked for me |
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| 19:41.39 | ageis | my extensions.conf is not getting loaded, jsut extensions.ael. list of things to check? |
| 19:42.15 | igcewieling | ageis: inforrectly formatted extensions.conf |
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| 19:44.02 | ageis | aha, got it |
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| 19:54.11 | leifmadsen | igcewieling: inforrectly! |
| 19:56.24 | jmetro | Inforrectly sounds like Infobot solved someones issue. |
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| 20:19.23 | floren | could anyone spare a bt of their busy time and help me troubleshoot the rstp compile issue i have on hand, with asterisk? |
| 20:19.49 | floren | i googled a lot but i don't see anywhere a clear reference about the issue |
| 20:21.06 | newtonr | floren: pastebin the output from the compilation attempt and I'll take a glance real quick |
| 20:21.13 | floren | i opened a thread into forums: http://forums.asterisk.org/viewtopic.php?f=1&t=87434 |
| 20:21.28 | floren | thanks newtonr |
| 20:21.44 | floren | just for the record, i use the official rpm created on fedora 19 |
| 20:22.01 | floren | i compile it for centos 6 x86_64 |
| 20:22.20 | floren | it is available for testing at rpm.axivo.com |
| 20:23.49 | floren | i did also set a -fPIC flag, just to be safe |
| 20:24.54 | floren | pkgconfig reports the proper location also |
| 20:33.38 | floren | i'm going to recompile srtp with --enable-pic |
| 20:33.47 | floren | i'll report back if this solved the issue |
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| 21:08.35 | newtonr | floren: hmm, yeah IDK. Did you try with setting your path in "--with-srtp=" specifically, as opposed to just using the "--prefix=" ? |
| 21:09.05 | floren | i just tried now, --with-srtp=%{_includedir} |
| 21:09.09 | floren | still failing :( |
| 21:09.39 | floren | i'm recompiling libsrtp as we speak, we will get this nailed for sure :) |
| 21:10.05 | floren | i just dont understand why is not viewed by asterisk |
| 21:10.22 | floren | libsrp si definitelly compiled with -fPIC for visibility |
| 21:12.01 | floren | newtonr: i think i nail it |
| 21:12.28 | floren | i used on libsrtp --program-prefix=%{_prefix} |
| 21:12.39 | floren | that looks like made it visible for asterisk |
| 21:12.42 | jmetro | complicated |
| 21:12.47 | floren | i never saw this before |
| 21:13.03 | floren | because program-prefix pull the info from --prefix |
| 21:13.32 | floren | and you dont need to define prefix as is pushed by default |
| 21:14.10 | himura_x_ | hello |
| 21:14.11 | floren | at least the packages will be available for others to use :D |
| 21:14.24 | floren | what a pain in the derierre |
| 21:14.43 | navaismo | hehe |
| 21:15.29 | himura_x_ | somebody could help me with a question |
| 21:15.31 | floren | gettin' ready to party with res_srtp |
| 21:15.45 | WIMPy | himura_x_: Not unless you ask it. |
| 21:15.53 | navaismo | ~ask |
| 21:15.54 | infobot | Questions in the channel should be specific, informative, complete, concise, and on-topic. Don't ask if you can ask a question first. Don't ask if a person is there; just ask what you intended to ask them. Better questions more frequently yield better answers. We are all here voluntarily or against our will. |
| 21:16.05 | newtonr | floren: yeah I don't understand why you had to do that :\ |
| 21:16.44 | newtonr | floren: make sure you update your forum post so someone else will find it when they Google it :D |
| 21:16.50 | himura_x_ | i need connect one hipath 3550 with server asterisk |
| 21:17.04 | floren | newtonr: will do :) |
| 21:17.21 | floren | for some reason nobody replies to my questions on forums :( |
| 21:17.27 | floren | they are not complicated, heh |
| 21:18.04 | WIMPy | floren: Becauyse you usually need an account to do so? |
| 21:18.16 | floren | also, using program-refix does not requires anymore to define a specific --with-srtp yay |
| 21:18.20 | navaismo | reason one: nobody knows about your issue |
| 21:18.38 | floren | http://forums.asterisk.org/viewtopic.php?f=1&t=87434 |
| 21:18.41 | navaismo | reason two: many person comile instead using rpms and dont mess with rpms |
| 21:18.44 | floren | i have an account, no? |
| 21:18.51 | floren | or did i missed something? |
| 21:19.03 | navaismo | reason three: they dont want to answer |
| 21:19.15 | navaismo | reason four: the dont know what are you talking about... |
| 21:19.16 | WIMPy | floren: Be someone with the answer probably hasn't. |
| 21:19.20 | navaismo | and so on... |
| 21:19.29 | floren | i'm new so i guess i did not upset anyone... plus i'm a very nice person :) |
| 21:20.00 | navaismo | ¬¬ |
| 21:20.04 | WIMPy | himura_x_: That was not a question. |
| 21:20.06 | floren | i'm glad i nailed srtp though, i got it by looking at the source :) |
| 21:20.33 | floren | i saw no reference to prefix and imediatelly figured is got to be this the case |
| 21:21.24 | floren | asterisk compiled perfect, i got srtp enabled :D |
| 21:21.39 | floren | now is time to play with the asterisk tutorial |
| 21:22.04 | floren | i will be in paradise if voip.ms does enables srtp |
| 21:22.11 | floren | and g722 |
| 21:22.13 | floren | :) |
| 21:22.26 | jmetro | g722 \o/ |
| 21:22.32 | floren | :D |
| 21:22.40 | WIMPy | I'm pretty sure someone will mke sure that doesn;t happen. |
| 21:22.41 | floren | i use it for voicemail, what a difference |
| 21:23.11 | floren | i was playing the ulaw asterisk files, then switched to g722 i was like WOW |
| 21:23.28 | himura_x_ | I really need ways to connect the hipath with asterisk, I have reviewed that is connected via a sip trunk but i dont access to hipath there any other way |
| 21:23.37 | floren | same bandwidth, what a diff in sound quality |
| 21:24.37 | WIMPy | himura_x_: That's still not a question. And it's probably cheaper to connect via BRI/PRI. |
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| 21:25.51 | newtonr | himura_x_: Asterisk could connect with the hipath via any voice communications protocol that they both support. I have no idea what the hipath supports. |
| 21:26.29 | floren | newtonr: http://forums.asterisk.org/viewtopic.php?f=1&t=87434&p=189833#p189833 :) |
| 21:26.51 | floren | i'm going to drop my centos5 repo |
| 21:26.56 | floren | way too much work |
| 21:27.08 | floren | plus soon redhat7 will make surface |
| 21:27.13 | himura_x_ | WIMPy is the same way dahdi |
| 21:27.14 | floren | nto worth the effort |
| 21:27.40 | jmetro | redhat is gross, debian ftw |
| 21:27.45 | floren | :D |
| 21:27.58 | floren | i manage at weok over 3,000 linux boxes :) |
| 21:28.01 | floren | got used to it |
| 21:28.05 | floren | work* |
| 21:28.40 | WIMPy | himura_x_: Sorry, my glass shpere is still not working. I have no idea what you might be looking for. |
| 21:29.10 | floren | WIMPy: what was the issue you were working on yesterday that you had to patch? |
| 21:29.24 | floren | i saw it quickly on chan |
| 21:29.45 | WIMPy | ASTERISK-22183 |
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| 21:30.18 | WIMPy | He, noone expanding that? |
| 21:30.21 | floren | ya, i remember now |
| 21:30.28 | floren | it looks like others have the smae issue? |
| 21:30.59 | WIMPy | Everyone who "hacks" their phone "line". |
| 21:31.08 | himura_x_ | WIMPy lets write well the situation |
| 21:31.39 | floren | :) |
| 21:32.37 | WIMPy | Connecting Asterisk to a SIP switch via ISDN seems just wrong, but does work better so far :-( |
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| 21:33.17 | fullstop | Hi all. Any recommendations for analog telephone adapters? I only need one FXS port. |
| 21:33.46 | Mango45 | OBi100 |
| 21:34.04 | Mango45 | Designed by the same engineers as PAP2T, SPA2102, and friends. |
| 21:34.16 | fullstop | looking |
| 21:34.26 | floren | i have a cisco spa112 collecting dust |
| 21:34.32 | Mango45 | Good use for it |
| 21:34.51 | floren | since i plan to use asterisk fax now :) |
| 21:35.02 | Mango45 | :) |
| 21:35.20 | WIMPy | Don't remind me of that. I should take a look at fax as well :-( |
| 21:35.21 | fullstop | looks decent |
| 21:35.31 | fullstop | I gave up on t.38 fax |
| 21:35.49 | fullstop | but this is for home and I don't deal with faxes here. |
| 21:36.12 | Mango45 | It rocks. I have an OBi110 (contains an FXS port also) and it's pretty great. I don't even use Asterisk at home any more as its dial plan is so powerful. (Still use Asterisk at work.) |
| 21:36.23 | floren | after 2 weeks of insane hours, i managed to get my cisco 8961 and 9971 working perfectly with asterisk :) |
| 21:36.29 | floren | what a ride |
| 21:37.08 | floren | those phones are so nice, to bad their sip is proprietary/broken |
| 21:37.28 | Mango45 | Only downside to the OBis is no configurable jitter buffer. |
| 21:37.41 | floren | i still have a retransmit issue that i cannot figure why is hapenning |
| 21:37.49 | Mango45 | Alarm systems are reported to have issues. |
| 21:37.55 | floren | as usual nobody alswered to my forum thread :D |
| 21:37.59 | WIMPy | That's just the way SIP (doesn't) work. Get used to it or stay away from it. |
| 21:38.25 | floren | http://forums.asterisk.org/viewtopic.php?f=1&t=87333 |
| 21:38.36 | floren | pushing my luck, in case someone smart knows the answer :) |
| 21:38.54 | floren | so far today i'm lucky, i get things fixed :D |
| 21:39.06 | WIMPy | *and* want's to use a forum. |
| 21:39.16 | WIMPy | -' |
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| 21:40.06 | floren | WIMPy: fix my retransmission issue please :D |
| 21:40.35 | WIMPy | I try to keep out of the VOIP stuff as much as possible. |
| 21:40.42 | floren | :) |
| 21:40.44 | WIMPy | Obviousely withpout too much success :-( |
| 21:41.03 | floren | heh |
| 21:42.20 | WIMPy | For the weekend I need a box to complement a Hipath. That will be completely non-IP. |
| 21:42.47 | WIMPy | I'm not even sure I need to add Asterisk to it. |
| 21:43.37 | WIMPy | Lets see if that will be less painfull than other experiments. |
| 21:43.41 | floren | Unknown value '' found in build_tools/menuselect-deps for NATIVE_ARCH |
| 21:43.53 | floren | does anyone know thew accepted values? |
| 21:43.59 | floren | x64? |
| 21:44.09 | floren | it will save me the trouble to scan the code |
| 21:47.11 | himura_x_ | WINPy what I need is to connect a hipath with asterisk,my problem is because I have not worked with analog trunk, I want to know if operate like two sip trunk Asterisk or working as a DAHDI PRI ?? |
| 21:48.33 | WIMPy | himura_x_: You are throwing in random bits of information and not asking a clear question, nor do you state what exactely you have and what you want to do. No help possible this way. |
| 21:48.33 | himura_x_ | WINPy Now my question is understood apologize for any inconvenience |
| 21:48.45 | himura_x_ | ok |
| 21:59.07 | floren | i wonder if i should define the AST_NATIVE_ARCH |
| 21:59.32 | WIMPy | Or just ARCH? |
| 21:59.33 | floren | is expecting a value, and by default it returns a null so that's why i get a complaint |
| 22:00.14 | floren | $as_echo "user CFLAGS present" >&6; } |
| 22:00.32 | floren | AST_NATIVE_ARCH= |
| 22:00.36 | floren | so that explains it |
| 22:00.41 | floren | i'll leave it like that |
| 22:01.25 | floren | it does nto break anything but it should be abool value instead of 1 or null |
| 22:02.13 | floren | menuselect-depts complains because: NATIVE_ARCH=@AST_NATIVE_ARCH@ |
| 22:02.39 | floren | since is null, it throws an error |
| 22:02.54 | Qwell | What error, and what is the result of the error? |
| 22:03.06 | floren | my compiling day is done for today, building the latest packages as we speak :) |
| 22:03.35 | floren | Qwell: is not actually an error, more like a warning, let me paste it |
| 22:05.15 | floren | http://codepad.org/TIcQU3yw |
| 22:05.20 | floren | self explanatory |
| 22:05.51 | floren | look in menuselect-depts.in, last line |
| 22:06.18 | floren | then in configure and configure.ac |
| 22:06.58 | floren | line 16255 for example in configure |
| 22:07.37 | floren | AST_NATIVE_ARCH= |
| 22:07.47 | floren | i would set a 0 value there |
| 22:07.50 | floren | any anywhere else |
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| 22:37.02 | himura_x_ | Hey, some one can help me ... |
| 22:37.14 | himura_x_ | maybe i don't explain very good ... |
| 22:37.30 | WIMPy | I'd rather say you didn't explain at all. |
| 22:37.47 | WIMPy | Describe what you have and what you want. And be as precise as possible. |
| 22:37.47 | himura_x_ | yes Wimpy |
| 22:37.55 | himura_x_ | i try to do my best right now |
| 22:38.07 | himura_x_ | with my english :) |
| 22:38.54 | himura_x_ | I have a Siemens HiPath 3550, that have 1 E1 connected |
| 22:39.10 | newtonr | himura_x_: *connected to what? |
| 22:39.22 | himura_x_ | 1 E1 to the Siemens HiPath 3550 |
| 22:39.44 | himura_x_ | Siemens is a Central |
| 22:40.06 | newtonr | the Siemens is hosted at your service provider? |
| 22:40.32 | himura_x_ | yes |
| 22:40.37 | WIMPy | One E1 from your telco to the Hipath? |
| 22:40.42 | himura_x_ | yes |
| 22:40.51 | WIMPy | Ok, go on. |
| 22:40.59 | himura_x_ | that's right one E1 from my telco to the HiPath |
| 22:41.02 | himura_x_ | so |
| 22:41.24 | himura_x_ | i need comunicated an Asterisk Server with the HiPath |
| 22:41.55 | himura_x_ | i think put the E1 to the asterisk with a sangoma card |
| 22:42.35 | WIMPy | The most obvious choice would be to put it between the Line and the Hipath. |
| 22:42.41 | himura_x_ | but i don't know how connect the Asterisk with the Hipath to don't lose the annex |
| 22:42.42 | WIMPy | But what do you want to do with Asterisk? |
| 22:43.03 | himura_x_ | i need to put on the asterisk some IVR |
| 22:43.04 | WIMPy | Through the E1. |
| 22:43.16 | WIMPy | Telco <> Asterisk <> Hipath. |
| 22:43.24 | himura_x_ | <PROTECTED> |
| 22:43.37 | WIMPy | Use a 2 dual E1 card. |
| 22:43.49 | *** join/#asterisk bigDoggy (brads@2600:3c00::f03c:91ff:fe70:f0a9) |
| 22:43.52 | himura_x_ | ohhh ise |
| 22:43.54 | himura_x_ | i see |
| 22:44.04 | himura_x_ | so with the E1 dual |
| 22:44.19 | himura_x_ | on the asterisk i send to the Hipath |
| 22:44.31 | himura_x_ | perfect |
| 22:44.39 | WIMPy | yes |
| 22:44.55 | bigDoggy | where is the sms outgoing spooler dir? I don't hav a /var/spool/asterisk/outgoing |
| 22:45.09 | bigDoggy | Asterisk 11.2.2 |
| 22:45.36 | himura_x_ | thanks a lot WIMPy |
| 22:45.39 | WIMPy | himura_x_: it's about the same as I'm currently setting up. Just that my patient os connected via two BRIs only. |
| 22:47.45 | *** join/#asterisk ooggetuigen (~m@120-163-ftth.onsbrabantnet.nl) |
| 22:47.51 | ooggetuigen | hello all |
| 22:48.13 | ooggetuigen | someone here who has a list of dutch outbound routing patterns? |
| 22:50.35 | navaismo | _X. |
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| 22:51.44 | ooggetuigen | what you mean navaismo |
| 22:52.03 | WIMPy | ooggetuigen: I guess that was the answer to your question. |
| 22:54.28 | ooggetuigen | hahah |
| 22:54.35 | Katty | hi kids |
| 22:54.43 | ooggetuigen | kiddys |
| 22:54.44 | ooggetuigen | ;) |
| 22:54.49 | WIMPy | is not sure what's funny about that. |
| 22:55.00 | Katty | WIMPy: answer me this. |
| 22:55.06 | Katty | WIMPy: why is 2 + 2 not 4? |
| 22:55.34 | WIMPy | you're using a base<=4? |
| 22:55.46 | Katty | well. i guess that's one good answer. |
| 22:55.53 | Katty | ok, you win! |
| 22:56.08 | WIMPy | What did I win? |
| 22:56.32 | Katty | a lifetime supply of rice a roni, the san francisco treat! |
| 22:57.00 | WIMPy | FUp for collection, I assume? |
| 22:57.10 | Katty | hmm? |
| 22:57.20 | floren | hi Katty |
| 22:57.27 | Katty | hello, floren |
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| 23:07.37 | igcewieling | Katty: Knowing you, I expected "free hugs for life" to be the prize 8-| |
| 23:10.36 | Katty | lol |
| 23:10.40 | Katty | nah |
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| 23:52.52 | file | falls into existence |
| 23:53.14 | Katty | catches file and deposits into reality |
| 23:55.32 | file | Katty, hello! |
| 23:55.51 | Katty | hello cupcake! |