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| 00:44.04 | Shaan | can anyone recommend any good enterprise softphones |
| 00:44.11 | Shaan | other then resource intensive bria |
| 00:44.17 | Shaan | havent tried zoiper communicator or 3cx |
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| 01:15.19 | themrrobert | Wow no navaismo today? |
| 01:17.40 | themrrobert | So I have a strange issue. I have asterisk 11.17.1, setting up for WebRTC. I also have a STUN/TURN server using CoTurn. It works fine for devices behind a regular NAT, like my home internet, and other peoples homes, however it doesn't work in an office that's behind a pfSense firewall. I've traced all the SIP logs, it looks like it's all configured properly, there just isn't any audio flowing |
| 01:17.40 | themrrobert | in either direction. I've tried all different combinations of TURN/STUN servers. (Using SIPML5 btw) I'll put together some details |
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| 01:45.48 | themrrobert | here is a call log from SIPML5, and my sip+rtp.conf Outbound Call: http://pastebin.ca/3002473 & RTP + SIP.conf http://pastebin.ca/3002478 |
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| 02:05.39 | Shaan | other then resource intensive bria |
| 02:05.43 | Shaan | can anyone recommend any good enterprise softphones |
| 02:44.24 | drmessano | What is an "enterprise" softphone |
| 02:46.37 | carrar | Blink Pro |
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| 02:47.48 | WIMPy | Isn't pro<enterprise? |
| 02:47.55 | carrar | heh |
| 02:47.59 | carrar | or zoiper |
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| 02:49.19 | drmessano | Does it have an developer system integrator version? |
| 02:49.26 | drmessano | a |
| 02:49.44 | drmessano | I hate the absolute hell out of software adjectives |
| 02:50.32 | carrar | http://www.counterpath.com/enterprise-solutions/ |
| 02:50.33 | carrar | heh |
| 02:53.08 | carrar | having the whole automated provisioning thing down is key for enterprises |
| 02:54.23 | drmessano | Possibly, though one could argue that ANY softphone that stores its settings in the registry could be deployed with group policy templates enabling essentially the same mass configuration |
| 02:54.32 | drmessano | and by "one", I mean, ME |
| 02:55.13 | carrar | that sounds windowish |
| 02:55.14 | WIMPy | registry? |
| 02:55.30 | carrar | wedding registry |
| 02:55.31 | WIMPy | Are there still people left using Windows? |
| 02:55.56 | carrar | not here in Redmond |
| 02:56.17 | drmessano | I assume he means "Windows" when he says "Enterprise softphone". Only Windows people speak like that |
| 02:56.46 | drmessano | As in |
| 02:56.52 | WIMPy | I'm pretty sure I've heard that from RedHat as well. |
| 02:57.25 | drmessano | "I am running Windows 7 Black Edition and I need a keygen for ____ softphone Ultra Enterprise Edition" |
| 02:58.14 | carrar | needz linux keygen!! |
| 02:58.21 | drmessano | Well, I am not saying it's exclusive to Windows. But in the Windows world, everything is "Basic, Standard, Pro, and Enterprise" |
| 02:58.40 | drmessano | Like, Small, Medium, Large, and Gut Buster |
| 02:59.44 | drmessano | Enterprise pretty much means "The one with all the stuff that you should probably Pirate" |
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| 03:14.37 | phix | WIMPy:! Can you help me out mate? |
| 03:14.57 | phix | I lost the dahdi init script :( |
| 03:18.25 | phix | WIMPy: I have /etc under git revision control so I have checked it out again :D although I am still getting errors when loading the module: 4 is not a valid channel number. |
| 03:18.28 | phix | Missing '/sys/bus/astribanks/drivers/xppdrv/sync' |
| 03:18.42 | phix | Did they start to count from 0 instead of 1? |
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| 03:20.27 | phix | gang? |
| 03:24.40 | drmessano | Sorry, I was reinstall Windows 69 Enterprise Edition |
| 03:25.17 | carrar | You have a key for that?!! |
| 03:25.31 | drmessano | I have a keygenloadermod thing |
| 03:26.01 | drmessano | Its straight from M$ ... 100% legit |
| 03:26.30 | carrar | I hear they do that for all their B2B deals |
| 03:27.12 | drmessano | I need to change my Twitter profile to read "Windows pays my bills. Linux makes my coffee" |
| 03:28.08 | phix | drmessano!!! |
| 03:28.17 | drmessano | sup |
| 03:28.40 | phix | drmessano: Linux pays my bills and makes me coffee |
| 03:30.38 | drmessano | yeah thats not my life yet |
| 03:32.03 | drmessano | Maybe one day |
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| 04:08.32 | phix | drmessano: perhaps |
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| 04:38.08 | tuxd00d | chuckes at drmessano âs âGut Busterâ comparison. |
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| 10:08.41 | WildPikachu | does anyone know if the behavior of the MONITOR_EXEC channel variable has changed? apon upgrading from 1.8 to 1.11 it seems that its not being run anymore despite it being set in my dialplan |
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| 12:09.33 | themrrobert | WildPikachu it should work. You sure you aren't using mixmonitor? |
| 12:10.09 | themrrobert | pastebin the relevent dialplan? and maybe a test call verbose logging |
| 12:10.50 | WildPikachu | themrrobert, I did a diff between my configs and both have monitor-type: MixMonitor in queues.conf |
| 12:10.56 | WildPikachu | sec, I'll pastebin |
| 12:11.59 | WildPikachu | http://paste.linuxassist.net/view/9228e541 <= thats what I've been using for the past 5+ years :) |
| 12:12.16 | WildPikachu | I get the .wav's just it never runs my script after upgrading |
| 12:13.03 | WildPikachu | ok, themrrobert I see what you mean, its changed in the recent versions of asterisk, I'll check up on it |
| 12:15.14 | themrrobert | yes in queues.conf, mixmonitor uses the new[er] mixmonitor function which doesn't run the command in MONITOR_EXEC |
| 12:15.49 | WildPikachu | themrrobert, what I am doing is using that I pasted as the call comes in before its queued, right? so I should revert to the old behavior then by commenting out the line in queues? |
| 12:17.15 | WildPikachu | themrrobert, oddly enough the queues.conf file comments still say it should be executed, hrmm |
| 12:18.20 | zekoZeko | it seems i got rid of my "recordings contain nasty clipping" problem by going from Monitor to MixMonitor |
| 12:18.29 | themrrobert | well i know the purpose of the MONITOR_EXEC is to mix the 2 legs of the call |
| 12:18.40 | themrrobert | i'm trying to lookup what i've done to solve the same issue |
| 12:18.51 | WildPikachu | thanks themrrobert :) |
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| 12:19.41 | zekoZeko | i have no idea how the hell it happened, but somehow the recording got screwed up before mixing, i was saving the original unmixed recordings too and they contained the clipping already |
| 12:19.49 | themrrobert | what happens when you make a call? Is it showing that its attempting to execute the file or not? |
| 12:21.50 | WildPikachu | themrrobert, busy testing that right now |
| 12:24.35 | themrrobert | WildPikachu can you not simply change your code from Monitor to MixMonitor(filename.ext,,/var/lib/asterisk/agi-bin/wav2mp3) and change the script so it only deals with a pre-mixed file instead of two individual files? |
| 12:27.01 | themrrobert | WildPikachu or change the queues.conf to use monitor-format=wav then the MONITOR_EXEC should trigger, be sure to set __MONITOR_EXEC if it's outside the originating channel, and it must be called BEFORE the queue application |
| 12:28.28 | themrrobert | also it would be more useful if i could more of the dialplan, i have no idea what context that macro is called in, or where in relation to the rest of the call. i would like to see all of the dialplan that is executed on a particular call, as well as the verbose logging, that is, if you don't figure it out on your own first ;) |
| 12:28.34 | WildPikachu | themrrobert, busy seeing if I can move to mixmonitor now |
| 12:28.45 | themrrobert | alright, i'll try to check back in a little bit |
| 12:28.56 | WildPikachu | zekoZeko, I had some clipping issues with normal monitor too which is what got me onto this :) |
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| 12:29.19 | zekoZeko | my issues were very strange |
| 12:29.27 | zekoZeko | i had asterisk in a container (LXC) |
| 12:29.37 | zekoZeko | and hardware was not very loaded |
| 12:29.44 | zekoZeko | not CPU, not I/O |
| 12:29.58 | zekoZeko | and it worked for a few months |
| 12:30.01 | zekoZeko | just well |
| 12:30.07 | zekoZeko | then it started doing horrible clipping |
| 12:30.19 | zekoZeko | i installed asterisk in it's own physical machine |
| 12:30.25 | zekoZeko | it got *a little* better |
| 12:30.29 | zekoZeko | but not yet OK |
| 12:30.41 | zekoZeko | and it's not like it's a bunch of calls |
| 12:30.48 | zekoZeko | 5-10 recordings/day |
| 12:30.54 | zekoZeko | and <100 calls a day i guess |
| 12:30.58 | zekoZeko | nothing at all |
| 12:31.16 | themrrobert | nope lol i get over 20k recordings per day on some of my systems |
| 12:31.16 | zekoZeko | I'll see how it goes, but it seems OK so far |
| 12:31.30 | themrrobert | so < 100 is really nothing |
| 12:32.03 | zekoZeko | it's a small radio station, the recordings are journalists, and the others are regular calls by people |
| 12:32.16 | zekoZeko | but yeah, the standalone machine had load close to 0 all the time |
| 12:32.27 | zekoZeko | and a SSD, so shouldn't be an issue with I/O |
| 12:32.43 | themrrobert | definitely not, could be a driver issue tho |
| 12:33.01 | themrrobert | i know its not likely |
| 12:33.07 | zekoZeko | but just out of curiosity i tried using MixMonitor instead of Monitor and it works, after i've done a bunch of other debugging. |
| 12:33.21 | zekoZeko | what driver? I'm SIP only... |
| 12:33.30 | themrrobert | i meant the disk driver |
| 12:33.32 | zekoZeko | oh |
| 12:33.42 | themrrobert | but i was just playing devils advocate |
| 12:33.44 | zekoZeko | well the first machine (that had asterisk running in container) |
| 12:33.54 | zekoZeko | was at the same time running a bunch of other stuff |
| 12:33.58 | zekoZeko | without trouble |
| 12:34.30 | zekoZeko | We'll see, it's still early to say i'm out of the dark. |
| 12:34.58 | themrrobert | check out this post from 2005, this is when i first starting using linux and hearing about asterisk: https://issues.asterisk.org/jira/browse/ASTERISK-5672 |
| 12:35.09 | themrrobert | not that post, i mean 2005 is when |
| 12:35.30 | themrrobert | its come a long way.. except my experience with asterisk 13 is that it's unstable with WebRTC |
| 12:35.53 | themrrobert | after only a few test calls, it crapped out, and wouldn't process any more calls until i restarted it, wasn't even going to waste my time with something that touchy |
| 12:36.19 | zekoZeko | interesting |
| 12:36.27 | themrrobert | thats the latest version available, 13.3.2, of course, i was using res_http and chan_sip, so it might be a bug in either of those modules |
| 12:36.34 | zekoZeko | i'm on 11 |
| 12:36.35 | themrrobert | i know that pjsip is the new thing |
| 12:36.42 | themrrobert | but i can't switch to it yet |
| 12:36.51 | zekoZeko | was 11.7 at first (ubuntu 14.04) but I've built myself packages with latest 11 now |
| 12:36.53 | themrrobert | yes 11 has been MUCH more stable |
| 12:36.58 | themrrobert | than 13 |
| 12:37.13 | zekoZeko | just in case it could be some bug that was fixed in the meantime. |
| 12:37.14 | themrrobert | i tried multiple times, always crashed after just a few calls |
| 12:37.15 | zekoZeko | :) |
| 12:37.36 | themrrobert | yea always best imo to compile yourself |
| 12:37.44 | themrrobert | mission critical packages anywy |
| 12:37.52 | zekoZeko | no, it sucks |
| 12:37.54 | zekoZeko | i mean |
| 12:37.57 | zekoZeko | not that it's a problem |
| 12:38.10 | zekoZeko | but repeatable processes are hard to do when you build your own stuff |
| 12:38.15 | zekoZeko | err, hard(er) |
| 12:38.42 | zekoZeko | i'm trying to make all machines i do these days to be rebuildable with one command. |
| 12:38.57 | zekoZeko | but that's why I built myself .deb packages with the latest version |
| 12:39.21 | themrrobert | yea, i don't have time for that, well, i do, but development outpaces it, so my images + scripts are outdated almost immediately |
| 12:40.01 | themrrobert | i have a custom AWS image i built that i deploy frequently, and wrote a few setup scripts for it, but of course, each one often needs custom care |
| 12:40.01 | zekoZeko | it's possible to have stuff done that way, but harder |
| 12:40.11 | zekoZeko | pays dividends when you have to do similar stuff again though. |
| 12:40.35 | zekoZeko | just use some config management thingy, puppet or whatever suits you. |
| 12:40.36 | themrrobert | yea that extra few hours i put into that stuff has saved me dozens of hours down the road even if i had to adjust it later |
| 12:40.40 | zekoZeko | i use saltstack myself. |
| 12:40.58 | themrrobert | i'm still on the fence, i think i'm leaning more towards chef, but i was originally a puppeter |
| 12:41.50 | themrrobert | saltstack is constantly pushing themselves on me, every conference i go to -.- |
| 12:41.50 | zekoZeko | hehe |
| 12:41.50 | zekoZeko | i love it |
| 12:41.50 | zekoZeko | i mean, of the big four (puppet, chef, ansible, salt) |
| 12:41.50 | zekoZeko | i threw away ruby-based stuff |
| 12:42.07 | zekoZeko | because i won't learn another language to be able to customize stuff i need on a daily basis |
| 12:42.17 | zekoZeko | and of the two left, i prefer salt |
| 12:42.17 | themrrobert | i will take any excuse to learn a new language ;) |
| 12:42.33 | zekoZeko | the thing is, i don't have time to learn _ALL_ languages :) |
| 12:42.48 | zekoZeko | and I don't program nearly enough so I'll pick one and use it if at all possible |
| 12:42.53 | themrrobert | i wish i did lol, i've neglected them for so long, recently i've been MAKING time |
| 12:42.54 | zekoZeko | it's python for me |
| 12:43.31 | themrrobert | lol i won't say what i'm thinking on that one |
| 12:43.52 | themrrobert | suffice to say, i hadn't touched perl in about 14 years, so i've been getting back into it, and having a lot of fun |
| 12:45.04 | themrrobert | man i remember starting to program when i was 11, and it was super impressive when i was like 14 because i could do amazing things by then, but now i'm almost 30 and somehow it doesn't seem as impressive to me anymore |
| 12:45.10 | zekoZeko | ugh... I've been fixing some stuff in Sympa a few months ago... and then I remembered why I stopped doing perl (even though sympa is relatively cleanly written...) |
| 12:45.36 | themrrobert | haha yea my favorite thing about perl is you can do so much with so little typing |
| 12:45.54 | themrrobert | but if you aren't familiar with it, it looks like gibberish |
| 12:45.54 | zekoZeko | except when you have to fix or add to it later and you're not a monk... |
| 12:47.31 | themrrobert | yea i must admit, i'm not a monk yet, hoping to be, if i find the time.. its quick to start things, takes longer to finish. the thing i hate most about perl is how it can be "abused" to do things indirectly so the code is extremly unreadable |
| 12:47.45 | themrrobert | but i also that love it, its a love/hate relationship with perl pretty much on all sides |
| 12:50.05 | *** join/#asterisk Vizx (~vlungu@37.233.19.32) |
| 12:50.22 | Vizx | Hi everyone |
| 12:52.02 | Vizx | Could someone help me to install app_mysql.so on my asterisk13 ? |
| 12:53.05 | Vizx | What package should be installed in ubuntu trusty to be able to enable "app_mysql.so" from menu select |
| 12:53.07 | Vizx | ? |
| 12:53.31 | themrrobert | wow app_mysql is VERY deprecated |
| 12:53.45 | themrrobert | why not use res_odbc and func_odbc? |
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| 12:54.09 | themrrobert | u sure you mean app_mysql and not cdr_mysql? |
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| 12:54.30 | themrrobert | if you have legacy dialplan i guess you could mean app_mysql |
| 12:55.46 | Vizx | I'v got an error: |
| 12:55.47 | Vizx | [2015-05-14 12:55:25] WARNING[1489][C-00000001]: pbx.c:4910 pbx_extension_helper: No application 'MYSQL' for extension (ivr_all, s, 3) |
| 12:55.48 | themrrobert | if you really want it, apt-get install libmysqlclient-dev mysql-client |
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| 12:56.13 | themrrobert | then you will need to ./configure && make menuselect |
| 12:56.16 | themrrobert | and enable app_mysql |
| 12:56.25 | [TK]D-Fender | Preferred means is either func_odbc or do it in AGI |
| 12:56.29 | themrrobert | but seriously its time to move to func_odbc |
| 12:56.34 | Vizx | same => n,MYSQL(Query resultid ${connid} CALL yopeso.is_holiday('${IsHolidayTime}', ${CCOUNTRY})) |
| 12:56.43 | Vizx | this is an example of my dialplan |
| 12:58.42 | Vizx | I thing was some issue with regular expression regarding func_odbc , that why it was done using app_mysql |
| 13:00.09 | themrrobert | can always use AGI, then you can use PERL or PHP which both handle regex pretty well |
| 13:00.16 | themrrobert | Perl* |
| 13:01.30 | Vizx | <PROTECTED> |
| 13:01.54 | Vizx | and thanks for advice , I will try to use agi or func_odbc |
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| 13:29.43 | voip | greetings |
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| 14:13.52 | carrar | greetings |
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| 15:54.50 | Dpunkt | I'm trying to get PJSIP working with the Deutsche Telekom. Asterisk sits behind NAT, Chan_Sip works in both directions. With PJSIP i can't dial out, every invite is answered with "403 Unauthorized". My Config is http://pastebin.com/pYW31GTS |
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| 15:56.03 | Dpunkt | A Sipgate account is working fine with PJSip |
| 15:58.36 | jvhester | Dpunkt: to be clear calls from Deutsche Telekom to Asterisk/PJSIP are working? |
| 15:59.23 | Dpunkt | Yes, but only using a Anonymous Endpoint due to load-balancing on their side |
| 15:59.49 | file | what is the sip.conf equivalent? |
| 15:59.51 | *** join/#asterisk thiagoc (b3dee04e@gateway/web/freenode/ip.179.222.224.78) |
| 15:59.51 | babak | Hi,Asterisk can use DNS SRV to send outgoing sip calls loadshare to 2 providers ? |
| 16:01.24 | WIMPy | Dpunkt: Maybe you can find another DTAG user in #asterisk-de. |
| 16:01.47 | thiagoc | hi, my asterisk was running fine until 2 days ago, now sometimes I don't get audio on calls (no nat) |
| 16:02.18 | thiagoc | nothing changed on the network |
| 16:02.32 | thiagoc | I tried to replace the switch but the problem remains |
| 16:02.39 | Dpunkt | Sip.conf http://pastebin.com/sdXPJJzQ |
| 16:04.24 | jvhester | Dpunkt: maybe from_user in the endpoint section? |
| 16:06.35 | thiagoc | I think it's a problem on the sip channel or in Asterisk |
| 16:06.54 | *** join/#asterisk tristero (~al.f.zero@unaffiliated/transfinite) |
| 16:07.17 | Dpunkt | from_user makes no difference |
| 16:07.35 | Dpunkt | i can copy the pjsip logger output to pastebin |
| 16:07.59 | jvhester | Dpunkt: your chan_sip peer has username= which I *think* defined user section of the 'From' header in messages going to that endpoint. |
| 16:08.04 | thiagoc | when I run "sip reload" it doesn't show anything for some seconds |
| 16:08.35 | thiagoc | and my peers suddenly go unreachable, but I can ping them |
| 16:09.05 | Dpunkt | Ill upload the log of PJSip, then take one with chan_sip to compare thenm |
| 16:09.10 | thiagoc | please someone can give me a clue? |
| 16:09.38 | jvhester | Dpunkt: pjsip logger output of the second INVITE from Asterisk to the ITSP might give a clue. |
| 16:10.15 | Dpunkt | there is only one invite |
| 16:10.52 | Dpunkt | invite -> 403 -> ack |
| 16:11.03 | *** join/#asterisk svm_invictvs (~patricktw@unaffiliated/svminvictvs/x-938456) |
| 16:11.17 | WIMPy | Then compare the 1st INVITEs. |
| 16:11.21 | Dpunkt | ok |
| 16:13.47 | thiagoc | actually it's chan_sip, the problem occurs with pjsip also |
| 16:13.56 | thiagoc | so I think it's asterisk related |
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| 16:18.10 | jvhester | Dpunkt: what does the Dial() do on those outbound calls? |
| 16:18.21 | jvhester | <PROTECTED> |
| 16:18.39 | jvhester | or PJSIP/${EXTEN}@trunk-telekom ? |
| 16:18.57 | Dpunkt | exten => _0X.,n,Dial(PJSIP/${EXTEN:1}@trunk-telekom,,W) |
| 16:21.56 | jvhester | registration works? (CLI> pjsip list registrations) |
| 16:23.30 | Dpunkt | Shown as Registered |
| 16:24.47 | Dpunkt | Chan sip: http://pastebin.com/LkYBistE PJSIP: http://pastebin.com/wm2fcDTA |
| 16:26.01 | Dpunkt | The From Headers look different |
| 16:26.28 | Dpunkt | pjsip uses my interal ip there?? |
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| 16:31.54 | jvhester | and it appears you've already tried from_domain in the endpoint section though it is commented out in the pjsip.conf pastebin |
| 16:32.40 | jvhester | oh wait, looks like sip.conf has tel.t-online.de while pjsip.conf has tel.telekom.de |
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| 16:34.35 | Dpunkt | sorry, lost wifi |
| 16:34.58 | jvhester | no worries, did you get the last message about from_domain? |
| 16:35.40 | jvhester | looks like sip.conf has fromdomain=tel.t-online.de while pjsip.conf has from_domain=tel.telekom.de |
| 16:35.57 | Dpunkt | nop did not get that |
| 16:37.16 | Dpunkt | Thank you |
| 16:37.23 | Dpunkt | That solved the Problem |
| 16:38.00 | jvhester | glad I could help |
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| 16:38.27 | Dpunkt | I have written the external_adress into the pjsip conf |
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| 16:39.21 | Dpunkt | but we are having a dynamical ip, do i have to change the file everytime? |
| 16:40.57 | Dpunkt | ok, just removed the line and still working |
| 16:48.49 | jvhester | Dpunkt: you mean external_signaling_address in transport section? |
| 16:49.47 | Dpunkt | Yes, this one |
| 16:52.48 | *** join/#asterisk Sudravirodhin (~Sudraviro@162.218.22.124) |
| 16:58.07 | jvhester | good question. When you say "removed the line and still working", did you do a complete restart of Asterisk? |
| 17:07.04 | thiagoc | what can cause a SIP extension becomes unreachable if I still can ping the IP? |
| 17:07.51 | WIMPy | The application has gone |
| 17:08.05 | WIMPy | Some firewall being evil. |
| 17:08.25 | WIMPy | An address change. |
| 17:09.01 | thiagoc | none of these in this case |
| 17:09.41 | jvhester | thiagoc: what's the ms response time on the ping? |
| 17:09.58 | doop | what kind of phone is at that IP |
| 17:10.02 | thiagoc | ~20ms or less |
| 17:10.04 | thiagoc | x-lite |
| 17:11.00 | thiagoc | when I run "sip reload" it hangs for about 10 seconds |
| 17:11.22 | thiagoc | I think there's some problem but I don't know what to look |
| 17:11.25 | WIMPy | That's most likely a DNS issue. |
| 17:12.57 | thiagoc | how can I disable dns lookup? |
| 17:13.15 | thiagoc | srvlookup=no |
| 17:13.16 | thiagoc | ? |
| 17:13.30 | drmessano | set up a local DNS cache |
| 17:13.49 | drmessano | like a minimal config DNSMASQ |
| 17:14.52 | thiagoc | drmessano: I will do that. but this can cause this strange sip being unreachable? |
| 17:15.04 | [TK]D-Fender | I'm betting a simple failure to even have a keep-alive <- |
| 17:15.21 | [TK]D-Fender | thiagoc: "sip show peer THEPEER" <---- |
| 17:15.28 | [TK]D-Fender | thiagoc: Show us your settings for it |
| 17:15.32 | [TK]D-Fender | ~pb |
| 17:15.32 | infobot | A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few: http://www.pastebin.com, http://pastebin.ca, http://channels.debian.net/paste, http://paste.lisp.org, http://bin.cakephp.org/; or install pastebinit with yum or aptitude. |
| 17:15.32 | drmessano | thiagoc, no one said these were the same issue |
| 17:15.34 | [TK]D-Fender | ^^^ |
| 17:16.45 | thiagoc | http://pastebin.com/raw.php?i=Tg5gsuSq |
| 17:19.11 | [TK]D-Fender | That seems fine so far. |
| 17:20.41 | [TK]D-Fender | Is this device actually remote from your server? |
| 17:21.54 | thiagoc | it's on the same network, actually on the same router |
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| 17:24.06 | [TK]D-Fender | thiagoc: What's the IP on the PC? |
| 17:29.53 | thiagoc | 192.150.10.103 |
| 17:31.28 | thiagoc | drmessano: I have a local dnsmasq configured and I'm using it on resolv.conf, but "sip reload" still hangs for 10-15 sec |
| 17:32.00 | thiagoc | drmessano: sorry, verbose was 0 |
| 17:32.06 | thiagoc | it's working now |
| 17:32.30 | drmessano | Never install * without a local DNS cache |
| 17:32.48 | [TK]D-Fender | thiagoc: Both your PC and server are each on PUBLIC IP's directly? |
| 17:32.55 | drmessano | In the words of Sean Connery in The Hunt for Red October |
| 17:33.24 | drmessano | "Ryan, some things in here don't react well to bullets" |
| 17:37.28 | thiagoc | [TK]D-Fender: they're behind the same router |
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| 17:37.52 | [TK]D-Fender | thiagoc: Those look like public IP's.. what other router are you talking about? |
| 17:38.03 | [TK]D-Fender | Neither is NAT'd are they? |
| 17:38.55 | thiagoc | [TK]D-Fender: my customer use this subnet on the local network (yes, I know) |
| 17:39.09 | drmessano | uhhhh |
| 17:39.24 | [TK]D-Fender | You means AS their "local"? |
| 17:39.33 | thiagoc | yes |
| 17:39.36 | [TK]D-Fender | As in... "all public direct routable"? |
| 17:40.13 | thiagoc | sorry, I don't get it |
| 17:40.32 | drmessano | I think he means as a private address range |
| 17:41.02 | thiagoc | yes drmessano |
| 17:42.19 | [TK]D-Fender | Well if you are loosing it in a subnet that is effectively only switched, then your issue is most likely tied toa firewall on the client side |
| 17:42.25 | [TK]D-Fender | or other odd issue |
| 17:43.01 | thiagoc | yes, I bet on the "odd issue" |
| 17:44.11 | thiagoc | now it's working again, I disabled NetworkManager and setup a local dnsmasq |
| 17:45.04 | [TK]D-Fender | Are you even ujsing hostnames anywhere? |
| 17:45.23 | thiagoc | no, that's why it's weird |
| 17:46.00 | thiagoc | the main issue was no audio on the sip endpoints |
| 17:46.25 | *** join/#asterisk robmal (r@91.231.70.61) |
| 17:46.33 | thiagoc | Now it's working, but I don't know if these changes I made have relation |
| 17:46.59 | drmessano | Lack of DNS resolution breaks chan_sip in bad ways |
| 17:47.16 | [TK]D-Fender | thiagoc: since your range is directly public, assuming no NAT on the server itself you should only have to prevent re-invites, and set the remote ones to "nat=yes" |
| 17:47.43 | [TK]D-Fender | drmessano: he's not even using hostnames, and the server seem fixed.... |
| 17:47.53 | drmessano | Doesnt matter |
| 17:48.56 | drmessano | Kill whatever DNS resolution is there and chan_sip does horrible things. Always has |
| 17:48.58 | [TK]D-Fender | thiagoc: We should see * trying to send the options packets it's currently trying to. When it frops off you should look at the SIP debug.... |
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| 17:50.41 | thiagoc | ok, thanks [TK]D-Fender and drmessano |
| 17:51.01 | drmessano | https://issues.asterisk.org/jira/browse/ASTERISK-21378 <-- This details it a bit more |
| 17:51.59 | drmessano | That shows a specific hang on a DNS lookup failing and causing chan_sip to go stupid.. but it's not the only way to reproduce |
| 17:59.59 | Shaan | can anyone recommend any good enterprise softphones |
| 18:02.36 | [TK]D-Fender | Shaan: All softphones suck, some only slightly less than others |
| 18:02.59 | Shaan | okay suggest a good wifi mobile phone then pleaSE |
| 18:03.58 | [TK]D-Fender | ~wifivoip |
| 18:03.58 | infobot | [~wifivoip] Wi-Fi (802.11a/b/g) VoIP phones on the market may possess any of the following drawbacks : poor battery life, limited range, lack of call features (many lack transfer, etc), poor or no NAT traversal, shoddy construction, etc. Because of this they are usually not recommended. Some alternatives are : VoIP ATA + cordless, DECT w/ VoIP base, etc |
| 18:04.27 | [TK]D-Fender | If you're looking for wireless within an office, then SIP + DECT is the way to go |
| 18:05.23 | Shaan | wow wow lol |
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| 18:09.06 | Shaan | [TK]D-Fender what about wifi desktop phones? |
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| 18:20.24 | Sudravirodhin | I can't post large messages, can I? |
| 18:21.59 | Sudravirodhin | I understand it goes against how Attended Transfers are supposed to work (via CLID) but is there any way to force Asterisk to change the outbound caller ID for the atxfer feature? Like a command I can't think of to change the local peer's caller ID? Reason I ask is because I've exhausted my options and notice that Asterisk uses Local tech during the atxfer. Apologies for the wall. |
| 18:27.39 | [TK]D-Fender | Shaan: You really have no wiring to use? |
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| 18:31.57 | [TK]D-Fender | Shaan: There are a tiny handful of SIP desk phones with Wifi. Not that great, but they'd do the job if that's what you need |
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| 18:36.09 | Shaan | [TK]D-Fender i wanted it to be softphone actually or a wifi phone so i can travel with it |
| 18:36.43 | [TK]D-Fender | Then I believe Bria works OK on Android. |
| 18:37.20 | robmal | Try csipsimple. |
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| 18:51.20 | Shaan | [TK]D-Fender i dont need it to be on android can be windows |
| 18:51.31 | Shaan | i just want very light weight simple sip phone |
| 18:51.39 | Shaan | brias eyebeam is soo blogggy |
| 18:52.14 | robmal | Shaan: http://icanblink.com/ ? |
| 18:53.16 | [TK]D-Fender | Shaan: figured you didn't want to run it on a laptop... |
| 18:53.47 | [TK]D-Fender | Shaan: laptop based is just .... bleh period. At which point there are other basic choices that work technically fine... Zoiper, etc |
| 18:53.48 | [TK]D-Fender | ~softphone |
| 18:53.49 | infobot | [~softphone] A soft-phone is a program that lets your computer use a compatible audio interface (usually a speaker&mic, or headset), and optionally a webcam, to place VoIP calls with a given protocol (SIP/IAX/H.323/MGCP/etc). For links enter ~zoiper , ~xlite , ~twinkle , ~bria , ~eyebeam , ~ekiga |
| 18:55.06 | babak | Hi,can Asterisk use DNS SRV to send outgoing sip calls loadshare to 2 providers ? |
| 19:00.16 | Micc | Could I use ARI to listen to device state change events and then use that in a web app to see state of everything in pretty much real time? |
| 19:00.49 | Shaan | ~zoiper |
| 19:00.49 | infobot | [~zoiper] Zoiper (formerly known as Idefisk) is a free SIP and IAX soft-phone for Windows, Mac OS X, and Linux that can be found at http://www.zoiper.com |
| 19:00.56 | Shaan | !bria |
| 19:00.58 | Shaan | ~bria |
| 19:00.59 | infobot | [~bria] Bria is a NON-free SIP soft-phone for Windows, Mac OS X, and Linux that can be downloaded from http://www.counterpath.com . This includes licensed audio & video codecs and is full-featured. |
| 19:01.04 | Shaan | ~ekiga |
| 19:01.04 | infobot | [~ekiga] Ekiga is an OSS SIP soft-phone for Windows, Mac OS X, and Linux (requires GTK+ which is included with the binary releases) that can be found at http://www.ekiga.org |
| 19:08.16 | [TK]D-Fender | Micc: No, you'd use AMI, not ARI |
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| 19:25.34 | Micc | TKD, but I hate AMI. |
| 19:26.13 | Micc | TKD, its been so buggy in the past, but I suppose it got an overhaul too recently. |
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| 19:34.29 | Resnik | good evening |
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| 21:16.34 | fireman_biff | Hi, I'm using an IAX2 trunk with DUNDi on Asterisk 11.13.0. Several times every day IAX2 stops working (the DUNDi lookup is fine). IAX2 debugging shows that it's trying to resend the same message multiple times, and tcpdump shows that the traffic never reaches the far side. 99% of the time I am able to fix the problem by unloading chan_iax2.so and then loading it again. Any ideas? |
| 21:22.57 | carrar | put the reload command in a crontab :) |
| 21:23.15 | carrar | SOLVED!! |
| 21:25.23 | fireman_biff | carrar: Yeah, that's what I have now (actually an unload/load, cause just a reload isn't good enough), but I was hoping for an actual fix. |
| 21:25.28 | lanning | 0 0 * * * init 6 |
| 21:25.52 | carrar | If you see the traffic leaving the server and never making it to the far side one might think it;s the circuit |
| 21:26.21 | fireman_biff | Except that un/loading the IAX2 module fixes it |
| 21:26.45 | fireman_biff | Actually 100% of the time, I said 99% percent before but apparently that other time was an iptables issue |
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| 21:28.21 | fireman_biff | lanning: it happens several times a day, so a nightly reboot won't help. |
| 21:29.04 | lanning | well that can be modified... :P 0 * * * * init 6 |
| 21:29.21 | fireman_biff | :P |
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| 22:58.39 | Katty | TURTLE SOUP |
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| 23:37.37 | phix | KAT! |
| 23:39.52 | carrar | SOUP!! |
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| 23:48.54 | phix | CHUNKY!!!! |
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