| 00:00.44 | saint_ | igcewieling: good call, thanks |
| 00:01.48 | igcewieling | this is why you do not want to be aggressive with noloading modules until you have a stable PBX and dialplan |
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| 00:20.32 | saint_ | igcewieling: i never touched the noload i think. i just used what was recommended in the default asteriskdocs.org |
| 00:26.29 | saint_ | igcewieling: I'm not using AddQueueMember (Im not using any call center stuff). How do i find out what module is addqueuemeber in, so i can add it in the noload list ? |
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| 02:43.19 | LLckfan | Does any1 know how to stop Shockwave flash from crashing? I have uninstalled both Flash and my browser (Chrome), installed both from a fresh download, and scanned my computer (come up clean). Everything is updated |
| 03:01.24 | WIMPy | [sr]: Yes, I have it installed. |
| 03:05.52 | WIMPy | transfers himself to bed. |
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| 07:41.24 | Rico29 | hi all |
| 07:41.39 | Rico29 | is there way, in sip.conf, to limit concurrent outgoing calls for a sip peer ? |
| 07:41.53 | Rico29 | sip.conf, or anywhere else... |
| 07:42.07 | kaldemar | GROUP functions in dialplan. |
| 07:45.10 | Rico29 | kaldemar> will take a look thanks |
| 07:45.15 | apb1963 | I'm trying to use AMI to originate a call, but it's giving me an error. I have a feeling it has something to do with the channel, but I don't really know what I'm looking for. Input & output here: http://ix.io/5In |
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| 08:29.50 | [sr] | WIMPy: how stable is it? chan_dongle, it doesn't compile on my instalation :( |
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| 08:31.48 | stevenm | Hey has anyone else noticed that polycoms need the "tftp://" part to be in the option 66 string? i'm sure it used to be it'd accept it as just an IP |
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| 10:29.53 | msaraiva | What's the best way to debug a possible memory leakage on asterisk? |
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| 13:19.43 | lugubrious | does anyone know why SIP extensions might be timing out and refusing to connect to an Asterisk box which is on the same subnet as the extensions? |
| 13:20.20 | Greenlight | Firewall? |
| 13:20.28 | lugubrious | I've tried various NAT options in both 'extension.conf' and 'sip.conf' to no avail |
| 13:20.36 | lugubrious | There is no firewall between the hosts |
| 13:20.56 | Greenlight | Enable SIP debug and have a look if it's hitting the box |
| 13:20.58 | lugubrious | Unless FreePBX enables a firewall by default :P |
| 13:21.24 | Greenlight | Why not check, to be sure |
| 13:22.26 | lugubrious | It doesn't appear as though FreePBX has any sort of firewall enabled/available |
| 13:22.51 | Greenlight | FreePBX wouldn't, but the box may |
| 13:22.58 | Greenlight | What's "iptables -L" say |
| 13:23.18 | lugubrious | The box doesn't either |
| 13:23.36 | lugubrious | It's my first foray into the world of Asterisk/VoIP and I'm decidedly unimpressed with FreePBX |
| 13:23.55 | lugubrious | not least because it's still based on kernel version 2.6.32 |
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| 13:24.28 | Greenlight | Well, that's presumably as the all-in-one installer uses CentOS |
| 13:25.06 | lugubrious | it appeared to be based upon CentOS, yeah |
| 13:25.47 | Greenlight | You want something reliable for a PBX, not something cutting edge :) |
| 13:26.26 | Greenlight | As for your issue, you should enable sip debug, and watch for the phones trying to register |
| 13:26.28 | lugubrious | true, but I did have to futz around with the VGA mode so that I could get the framebuffer to display something other than a blank screen |
| 13:27.32 | lugubrious | and this is on a >4 y.o. laptop which has an Intel "GPU"; not exactly cutting edge hardware... |
| 13:27.53 | lugubrious | Greenlight: Am I correct in assuming that the way to enable SIP debug is to use the Asterisk console? |
| 13:28.03 | Greenlight | Yup |
| 13:28.42 | johnwigley | Hello, anyone here able to do a paid (hopefully small) patch to app queue to change it's behaviour so that when it's in the ringall strategy, it will call all members with penalty between min penalty and max penalty instead of the current behaviour where it will only call members with the lowest penalty even if a max and min penalty are specified? |
| 13:29.51 | Greenlight | Why not just set those memebers to the same penalty ? |
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| 13:32.51 | johnwigley | What I'm trying to do is to achieve a custom call distribution strategy where we use give the call to the agent with lowest talktime and then if they don't answer within say 10 seconds then switches to a ring all strategy within the same queue - which is obviously not possible using app queue on it's own. So what we do is to have database triggers which watch for the real time writes to the queue log, and then calculate the st |
| 13:33.39 | lugubrious | I'm just getting "No such command 'debug'" or "No such command 'sip debug'" |
| 13:34.00 | lugubrious | I've logged into the machine and have run `asterisk -vr` |
| 13:34.04 | Greenlight | sip set debug on |
| 13:34.10 | Greenlight | core set verbose 9 |
| 13:34.35 | johnwigley | number of calls etc, and then dynamically switches the members penalties so that they have the lowest penalty, then we have a queue penalty rule in asterisk that for the first 10 seconds waiting time has the max penalty set 10, and the min to 1, so for the first 10 secs the designated agent rings, and then after that the penalty rule changes so that agents from penalty 1 - 100 are included, so that all ring. |
| 13:34.49 | lugubrious | Greenlight: Thanks |
| 13:35.46 | johnwigley | the problem is that because the targetted agent has penalty 10 and everyone else penalty 50, after the first 10 seconds still only the lowest penalty agent rings, because even though and max and min penalty are specified * will only call the lowest. |
| 13:37.01 | johnwigley | What I'm looking for is a patch that will make it behave so that when a max and min penalty are specified it will call all those within that penalty range not just the lowest. |
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| 13:38.11 | *** join/#asterisk matrix1233 (~matrix123@41.228.41.152) |
| 13:38.16 | matrix1233 | hi |
| 13:39.22 | matrix1233 | i have asterisk Asterisk 1.8.11.1-1digium1~lucid and when cunstumer sent 500 call on the same time the asterisk halt and freeze |
| 13:43.15 | lugubrious | Greenlight: After running `sip set debug on` and even `sip reload`, I still don't get any messages in the console when the extension which I am trying to configure tries to register. Is this normal or an indication that the extension is not actually contacting the Asterisk server? |
| 13:43.43 | Greenlight | Yea, looks like it's not even hitting the box |
| 13:44.21 | Greenlight | Did you have verbosity turned up |
| 13:44.27 | Greenlight | "core set verbose 9" |
| 13:44.51 | matrix1233 | how can i optimize my asterisk to resolve the problem of freezing |
| 13:45.26 | Greenlight | matrix1233: I would imagine raising an issue with Digium support, with detailed logs would be your fastest path to resolution |
| 13:46.15 | lugubrious | Greenlight: yep, I turned the verbosity up |
| 13:46.40 | Greenlight | And "iptables -L" shows everything set to ACCEPT ? |
| 13:46.41 | lugubrious | hmm... I think I shall have break out tcpdump :P |
| 13:46.46 | lugubrious | Greenlight: Yep |
| 13:46.54 | Greenlight | Yup - tcpdump away |
| 13:47.08 | lugubrious | I'm suspecting a dodgy ATA / dodgy firmware |
| 13:47.18 | Greenlight | Yea, could well be |
| 13:47.22 | lugubrious | I might have to try a softphone or something else to test it |
| 13:47.28 | lugubrious | Thanks for the help though |
| 13:47.36 | Greenlight | No probs - good luck! |
| 13:51.05 | Rico29 | I've got a problem with CLI>realtime update2 function |
| 13:51.18 | Rico29 | I want to reset a value to it's default value |
| 13:51.39 | Rico29 | but instead it saves "DEFAULT" string in the field |
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| 13:59.19 | igcewieling | try setting it to NULL |
| 14:00.20 | igcewieling | lugubrious: make SURE you have a local caching name server. Asterisk does a DNS lookup on each incoming connection. |
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| 14:11.52 | Rahail | Hi i got some issue with iax2 every time i hit reload |
| 14:12.00 | Rahail | i lose i iax2 conf info |
| 14:12.08 | Rahail | then when i reload it agian it come back |
| 14:12.09 | Rahail | any ide ahwy |
| 14:13.42 | Rahail | can some one help me please |
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| 14:18.59 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 14:19.52 | Rahail | any idea |
| 14:20.07 | Rahail | hello when asterisk drop all the conf on reload then when i relaod it it work |
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| 14:21.00 | Rahail | WIMPy |
| 14:21.02 | Rahail | you around |
| 14:21.10 | igcewieling | Rahail: You know that IAX2 is not a very popular protocol and so you know very few people can help you. |
| 14:21.31 | Rahail | i know but still i am sure some one is there exprince more then me |
| 14:21.48 | igcewieling | For a bug like that upgrade your Asterisk. |
| 14:22.04 | igcewieling | Either 1) this is a major bug in Asterisk or 2) your report is not valid and something else is going on too. |
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| 14:22.06 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 14:22.18 | igcewieling | I suspect option 2, but I can't prove it. |
| 14:22.28 | Rahail | me 2 |
| 14:22.36 | Rahail | but i do not know why when i reload vi cron |
| 14:22.38 | Rahail | i lose conf |
| 14:22.43 | Rahail | then when i reload again it come back |
| 14:22.48 | Rahail | then i reload agian its gone |
| 14:22.51 | Rahail | like that |
| 14:23.48 | Qwell | What version of Asterisk? |
| 14:23.59 | Rahail | 1.8.21.0 |
| 14:24.17 | lugubrious | Greenlight: Are you still around? |
| 14:24.20 | Qwell | show us logs |
| 14:24.25 | igcewieling | "reload vi cron"?? |
| 14:24.38 | Rahail | crontab |
| 14:24.43 | Rahail | i ws keep losign regsitraion |
| 14:24.50 | Rahail | so some one recomend me to do relooad |
| 14:24.59 | igcewieling | you reload iax by issuing "reload iax2" or similar in the Asterisk CLI. |
| 14:25.01 | Rahail | so i did /usr/sbin/asterisk -rx "iax2 reload" |
| 14:25.28 | lugubrious | Now I have a new problem: I've got a registered SIP trunk and have set up an outbound route but whenever I attempt to make a call using said route/trunk, the call doesn't complete and I just get silence from the phone/extension |
| 14:25.32 | Rahail | Qwell what log you refering to .. |
| 14:25.45 | igcewieling | and with NO other changes iax2 breaks and when you issue the command again with no other changes iax2 comes back |
| 14:25.52 | Rahail | yes |
| 14:25.58 | Qwell | Rahail: Whatever makes you think things are broken. |
| 14:26.05 | igcewieling | Rahail: the asterisk console logs showing the reload, the iax2 failures and the reload it again and see the iax2 success |
| 14:26.30 | johnwigley | Is there anyone here with experience of creating/submitting patches to Asterisk, I need a really simple one line patch doing to app_queue but I haven't got the experience to compile the code or to submit the patch for inclusion? Quite happy to paypal a thank you. |
| 14:26.45 | Rahail | Qwell i am montiring it now every 2 min i set it up to reload |
| 14:26.50 | Rahail | letsee what happned |
| 14:27.21 | igcewieling | johnwigley: 1) if it is a new feature it will be 1 - 2 years before the patch is in a released version, assuming it is accepted. If you just need a patch for YOUR asterisk there are lots of people who can do that |
| 14:28.01 | igcewieling | If it is obviously a bug fix then your chances are better at getting it included faster |
| 14:28.26 | igcewieling | Rahail: define "monitoring" |
| 14:28.52 | johnwigley | igcewieling: Ideally it would suit me to have it included so that I could update easier, but if that's the timeframe, then I'd be much happier if someone could just compile me a modified version of app_queue with this single change in it. Is that something you could do? |
| 14:28.53 | Rahail | sorry speell wrong |
| 14:29.19 | *** join/#asterisk Faustov (user@gentoo/user/faustov) |
| 14:29.36 | igcewieling | johnwigley: heck no, I stay out of code as much as I can. |
| 14:29.46 | igcewieling | And queues are horribly complex |
| 14:30.14 | igcewieling | johnwigley: you can post to the asterisk-biz mailing list. If it is a small feature post it as a bounty |
| 14:30.20 | johnwigley | Do you know anyone who'd be able to do it? I know the patch that I want, and it's just a single condition check removed, all they'd need to do is to compile it. |
| 14:30.40 | johnwigley | I even know the line number ;) |
| 14:32.11 | igcewieling | I've been using Asterisk for over 10 years and there are about 4 lines of my code in current Asterisk releases |
| 14:32.58 | apb1963 | johnwigley: how is it that you know the line number but don't know how to compile it? |
| 14:34.26 | *** join/#asterisk mquin (~mquin@freenode/staff/mquin) |
| 14:34.35 | Rahail | Qwell igcewieling i found it |
| 14:34.36 | johnwigley | apb1963: because I've got experience as an assembly programmer in windows, so I can read code fine and identify what I need to change, what I don't know is anything about linux, creating a build environment, or even how to go about building something like asterisk. |
| 14:34.40 | Rahail | i twas my mistake |
| 14:34.49 | Rahail | I had cron sync iax2 conf from our main server |
| 14:35.01 | Rahail | both were same time because of that iax2 conf was keep missing |
| 14:35.06 | Rahail | 2nd reload coming up like that |
| 14:35.09 | *** join/#asterisk serafie1 (~erin@nat/digium/x-bzzsqymkhbfolapv) |
| 14:35.13 | Rahail | i turn of the cron for sync now its workign |
| 14:35.48 | igcewieling | johnwigley: What version of Asterisk. |
| 14:35.55 | *** join/#asterisk blizzow (~jburns@173-8-237-25-Colorado.hfc.comcastbusiness.net) |
| 14:35.56 | johnwigley | 11.3 |
| 14:36.06 | igcewieling | johnwigley: SOMEONE built Asterisk for you. |
| 14:37.45 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.216.2) |
| 14:38.10 | apb1963 | johnwigley: Have you actually edited the code and made the change? |
| 14:39.14 | *** join/#asterisk darkbasic (~quassel@niko.linuxsystems.it) |
| 14:40.44 | apb1963 | igcewieling: How's your AMI? |
| 14:41.00 | igcewieling | apb1963: just good enough to do what I need to do. |
| 14:41.22 | apb1963 | well that's generally good enough :) |
| 14:41.24 | igcewieling | apb1963: sounds like johnwigley installed from package or something -- poor guy. |
| 14:42.08 | apb1963 | igcewieling: where did he get an 11.3 package? |
| 14:42.08 | *** join/#asterisk asilva (~asilva@2801:88:1000:2::12) |
| 14:42.22 | igcewieling | apb1963: no idea. Only idiots install Asterisk from a package |
| 14:43.17 | apb1963 | igcewieling: Now now... beeing noob doesn't make one an idiot |
| 14:43.21 | *** join/#asterisk sruffell (~sruffell@asterisk/the-kernel-guy/sruffell) |
| 14:43.21 | *** mode/#asterisk [+o sruffell] by ChanServ |
| 14:43.49 | igcewieling | apb1963: correct, it doesn't. Even n00bs install Asterisk from source. |
| 14:43.49 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.127.131) |
| 14:43.56 | apb1963 | igcewieling: this may come as a surprise to you, but Asterisk isn't simple. |
| 14:44.00 | Greenlight | Indeed igcewieling no need to be insulting |
| 14:44.40 | igcewieling | apb1963: Your're right, it is complex and you want to build it with YOUR options, not the options and parts some random packager thinks you need. |
| 14:45.25 | igcewieling | I don't know if this is still the case, but for a while packagers didn't even include the DOCS. Other times packagers REQUIRED DAHDI, other times they compile without DAHDI so you can't use DAHDI at all. |
| 14:45.39 | Greenlight | To be fair, I don't think building from source is *that* complex. Unfortunatlely some distros push people in the package direction without them knowning any better |
| 14:46.04 | *** join/#asterisk TimeRider (~steve@host81-136-216-215.in-addr.btopenworld.com) |
| 14:46.05 | igcewieling | Greenlight: it isn't complex. |
| 14:46.21 | matrix1233 | hi |
| 14:46.22 | Greenlight | It's like 5 commands, and selecting some options from a menu |
| 14:46.31 | matrix1233 | <PROTECTED> |
| 14:46.34 | Greenlight | Google's always on hand as well :) |
| 14:46.35 | matrix1233 | whats means |
| 14:46.51 | Greenlight | matrix1233: Sounds like you've ran out of file destriptors |
| 14:47.00 | Greenlight | *descriptors |
| 14:47.14 | apb1963 | sells file descriptors for $10 each |
| 14:47.28 | *** join/#asterisk SuperNull (~FreeManof@24-148-101-238.ip.mhcable.com) |
| 14:47.37 | Greenlight | That's a good price, you should buy them from him, before he runs out! |
| 14:47.44 | apb1963 | limited supply |
| 14:47.50 | igcewieling | Unless you have a LOT of calls, you won't run out of file descriptors unless something is very very wrong on your system |
| 14:47.50 | *** join/#asterisk Defraz (~Defraz@mail.pocatellochildren.com) |
| 14:48.06 | *** join/#asterisk coppice (~chatzilla@123203240234.ctinets.com) |
| 14:48.28 | apb1963 | buy 5, get 2 free |
| 14:49.49 | SuperNull | hey guys, trying to work on my MWI and i cant seem to get it to trigger by doing the file trick in the /var/spool/asterisk/voicemail/<context>/<extension>/INBOX/<msgFile> ? |
| 14:50.11 | igcewieling | SuperNull: make sure you have mailboxpolling enabled in sip.conf |
| 14:50.11 | matrix1233 | Greenlight haw can i resolve this ? |
| 14:51.27 | Greenlight | matrix1233: As igcewieling suggests, you really shouldn't be coming close to using them all up unless you've got loads of calls or an odd system. It's likely an indicator of another problem. |
| 14:51.59 | igcewieling | matrix1233: when you get that message, how many calls are shown in "core show channels" and "sip show channels" |
| 14:52.17 | igcewieling | matrix1233: also what version of Asterisk |
| 14:52.33 | igcewieling | Has anyone found safe_asterisk issues with 11.4? |
| 14:52.50 | apb1963 | such as? |
| 14:53.25 | igcewieling | apb1963: such as asterisk not starting and safe_asterisk waiting 4 seconds and restarting it without errors shown |
| 14:53.34 | SuperNull | icgewieling thanks i will check. |
| 14:53.36 | igcewieling | rinse, repeat |
| 14:53.51 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.127.131) |
| 14:54.16 | apb1963 | not I. But then I"m running 11.2.1 |
| 14:54.47 | apb1963 | However sometimes I get an issue where asterisk doesn't show errors |
| 14:54.50 | Rico29 | igcewieling> does not work |
| 14:54.59 | Rico29 | for the "realtime update2" |
| 14:55.55 | *** join/#asterisk blizzow (~jburns@173-8-237-25-Colorado.hfc.comcastbusiness.net) |
| 14:55.57 | igcewieling | apb1963: this started with the new 11.4 |
| 14:56.18 | igcewieling | Rico29: did you quote it? |
| 14:57.17 | Rico29 | igcewieling> I've tried both |
| 14:57.28 | matrix1233 | igcewieling i have 150 call now and i have this error, it can be with this number of call ? |
| 14:57.29 | Rico29 | it writes in database exactly what I write in CLUI |
| 14:57.30 | Rico29 | CLI |
| 14:57.48 | igcewieling | matrix1233: 150 calls or 150 channels? |
| 14:58.24 | igcewieling | Rico29: "NULL" is the value in our realtime SQL databases for items which use the globally configured valie |
| 14:58.43 | matrix1233 | igcewieling now exactly 80 calls and 150 channels |
| 14:58.46 | igcewieling | Actually NULL not "NULL" one is a null, one ins a string of 4 chars N U L and L |
| 14:59.04 | Greenlight | matrix1233: Execute "ulimit -a" as the user asterisk is running as |
| 14:59.25 | igcewieling | matrix1233: we routinely have 100+ calls on our servers |
| 14:59.35 | Greenlight | Whats the "open files" set to |
| 14:59.40 | johnwigley | igcewieling / apb1963: It's worse than installing from a package I'm afraid, the asterisk install has actually come from a idiot proof distro called Xivo. So I've not even needed to install the os separately ;) |
| 14:59.44 | Rico29 | igcewieling> yes it is, but if I do a realtime update2 <table> ... "NULL", it does not set the field value to NULL |
| 14:59.55 | Rico29 | it writes "NULL" as value for the field |
| 14:59.58 | matrix1233 | igcewieling i have this |
| 14:59.58 | igcewieling | johnwigley: then you need to contact your distro |
| 14:59.59 | matrix1233 | core file size (blocks, -c) 0 |
| 14:59.59 | matrix1233 | data seg size (kbytes, -d) unlimited |
| 14:59.59 | matrix1233 | scheduling priority (-e) 20 |
| 14:59.59 | matrix1233 | file size (blocks, -f) unlimited |
| 14:59.59 | matrix1233 | pending signals (-i) 16382 |
| 14:59.59 | matrix1233 | max locked memory (kbytes, -l) 64 |
| 14:59.59 | matrix1233 | max memory size (kbytes, -m) unlimited |
| 15:00.00 | apb1963 | that's a package |
| 15:00.00 | matrix1233 | open files (-n) 1024 |
| 15:00.00 | matrix1233 | pipe size (512 bytes, -p) 8 |
| 15:00.01 | matrix1233 | POSIX message queues (bytes, -q) 819200 |
| 15:00.01 | matrix1233 | real-time priority (-r) 0 |
| 15:00.02 | matrix1233 | stack size (kbytes, -s) 8192 |
| 15:00.02 | matrix1233 | cpu time (seconds, -t) unlimited |
| 15:00.03 | Greenlight | _pb |
| 15:00.03 | matrix1233 | max user processes (-u) unlimited |
| 15:00.03 | matrix1233 | virtual memory (kbytes, -v) unlimited |
| 15:00.07 | Greenlight | ~pb |
| 15:00.08 | 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. |
| 15:00.08 | carrar | w00t |
| 15:00.11 | Rico29 | matrix1233> do you know pastebins..? |
| 15:00.26 | Greenlight | 1024 open files max |
| 15:00.26 | matrix1233 | ah sorry |
| 15:01.02 | Greenlight | I *guess* if you're recording as well, or doing other things on server, you could be reaching that |
| 15:01.11 | matrix1233 | so i have to increase the number is it it ? |
| 15:01.40 | apb1963 | that's not very many |
| 15:02.11 | Greenlight | I have my boxes set at 6k |
| 15:02.38 | matrix1233 | how can i set it |
| 15:02.46 | igcewieling | johnwigley: I doubt anyone is willing to rebuild the packages for you (which is what you need to do and will likely be overwritten in the next update of your distro |
| 15:02.55 | apb1963 | if only the error message indicated how to increase them! |
| 15:02.56 | Greenlight | matrix1233: Run "lsof | wc -l" |
| 15:03.25 | matrix1233 | i have this 3831 |
| 15:03.45 | apb1963 | wonders if 3831 is more than 1024 |
| 15:04.14 | Greenlight | Yea, perhaps wise to increase that limit |
| 15:04.15 | apb1963 | hmm... carry the 3... |
| 15:04.29 | Greenlight | goes for the calculator |
| 15:04.54 | apb1963 | divide by the time the sun rises in the west... |
| 15:05.05 | matrix1233 | ok but what is tha command to do this |
| 15:05.28 | Greenlight | In CentOS it's in limits.conf iirc |
| 15:05.41 | matrix1233 | i will put it to 6k like you Greenlight |
| 15:05.45 | Greenlight | In,umm, /etc/security off the top of my head |
| 15:05.51 | Greenlight | COuld be wrong though |
| 15:05.58 | matrix1233 | am on ubunto |
| 15:06.06 | johnwigley | igcewieling: ok, sorry I thought it'd be an easy thing for someone to do with an already configured build environment, but I don't even know enough about how the shared libraries work in linux to know if you could build it on say a debian box and then run it on a centos install. |
| 15:06.30 | Greenlight | johnwigley: You'll be pleasently suprised how easy it is to build asterisk |
| 15:06.47 | apb1963 | johnwigley: Have you made the change and just need it compiled, or do you need the file edited for you too? |
| 15:06.58 | johnwigley | I've made the change but just need it compiling |
| 15:07.05 | igcewieling | the problem is even if you build it yourself, you are still running it on an Asterisk distro, which could overwrite your changes |
| 15:07.20 | Greenlight | Indeed - first thing, is install from scratch |
| 15:07.25 | apb1963 | igcewieling: Only if he reinstalls the distro |
| 15:07.28 | Greenlight | And remove the distro, so it doesn't overwrite |
| 15:07.35 | igcewieling | johnwigley: there are hundreds of documents on Google with instructions on installing Asterisk |
| 15:07.47 | johnwigley | The xivo distro that I'm using has a lot of other functionality in it, so I can't easily build from scratch without integrating all the stuff they've done into it. |
| 15:07.49 | igcewieling | apb1963: one "yum update" away. We ran into this. |
| 15:07.59 | Greenlight | mmhmm |
| 15:08.08 | igcewieling | the distro updated Asterisk and overwrote our custom zaptel stuff |
| 15:08.18 | Greenlight | That's why it's a PITA |
| 15:08.37 | johnwigley | It has a whole cti system, web interface etc, with as far as I understand a whole bunch of patches into asterisk to do what they need to. |
| 15:08.38 | apb1963 | igcewieling: oh you mean the distro... I thought you meant the package. nvm |
| 15:08.51 | *** join/#asterisk darkbasic (~quassel@niko.linuxsystems.it) |
| 15:09.04 | apb1963 | johnwigley: Why aren't you asking for help on #xivo? |
| 15:09.21 | igcewieling | apb1963: johnwigley said he was running an Asterisk distro, but in my situaion we were running the Trixbox RPM install. |
| 15:09.46 | igcewieling | took me most of an evening to resolve the version mismatches between asterisk, the asterisk modules, and zaptel |
| 15:09.54 | apb1963 | igcewieling: Trix are for kids! |
| 15:10.26 | igcewieling | apb1963: the server was from long before I arrived at the company. |
| 15:10.42 | apb1963 | isn't that always the case? lol |
| 15:11.28 | apb1963 | well, I imagine it depends on what you yum update |
| 15:11.47 | igcewieling | apb1963: not here. at least 3/4 of our asterisk boxes were installed by me. |
| 15:11.53 | apb1963 | If you don't yum asterisk modules, I'd think you'd probably be ok... especially in johnwigley's case |
| 15:12.45 | *** join/#asterisk malcolmd (~malcolmd@pdpc/sponsor/digium/malcolmd) |
| 15:12.45 | *** mode/#asterisk [+o malcolmd] by ChanServ |
| 15:12.55 | johnwigley | I presumed that there would be lots of people on here who were experienced in customising asterisk and who would be able to easily apply a patch and compile, rather than a distro support channel where you've got a lot of questions such as people struggling to connect a trunk rather than very technical issues. |
| 15:12.57 | apb1963 | looks like xivo is from france |
| 15:13.17 | johnwigley | it's developed in canada by a french company. |
| 15:13.23 | *** join/#asterisk brian98 (~brian98@unaffiliated/brian98) |
| 15:13.40 | igcewieling | johnwigley: there are, they all customize STOCK Asterisk |
| 15:13.57 | apb1963 | You never see Canadian companies in France |
| 15:14.52 | igcewieling | Q: What is the first major thing new users of Asterisk distros discover? A: That they should have installed from source. |
| 15:15.01 | Greenlight | +1 |
| 15:15.27 | apb1963 | johnwigley: make your changes, then type "make". See what happens. |
| 15:16.20 | apb1963 | I just can't stand the suffering anymore.... think of the children. |
| 15:17.18 | apb1963 | quickly adds a disclaimer. For compilation advice, see a licensed professional. |
| 15:17.42 | *** join/#asterisk darkbasic (~quassel@niko.linuxsystems.it) |
| 15:18.02 | *** join/#asterisk edge (~edge@74-84-85-54.client.mchsi.com) |
| 15:18.16 | [TK]D-Fender | igcewieling: They are using a complete GUI'd solution... when it addresses other needs sometimes you find there is some bad bundled in with the good. |
| 15:18.26 | [TK]D-Fender | and it's not necessarily the first thing they find |
| 15:18.38 | [TK]D-Fender | That tends to be "this GUI sure removes a lot of choices" |
| 15:18.47 | igcewieling | [TK]D-Fender: the first major thing they learn |
| 15:19.00 | [TK]D-Fender | and "generic Asterisk support groups don't know it and can't help me" |
| 15:19.08 | igcewieling | Hmm, OK, I concede your point [TK]D-Fender |
| 15:19.09 | johnwigley | apb1963: I don't think I'll have the source on the downloaded install cd, let alone a c compiler ;) it's meant as a production deployment cd rather anything else I think. |
| 15:20.00 | johnwigley | apb1963: Make - command not found, gcc - command not found. |
| 15:20.02 | apb1963 | no.... source.... ?? |
| 15:20.15 | apb1963 | I have a silly question |
| 15:20.19 | igcewieling | <PROTECTED> |
| 15:20.27 | apb1963 | How did you make the change to the code if there's no source? |
| 15:20.34 | johnwigley | I wouldn't have expected it to be honest, if you download trixbox would you expect that to have a build environment in it? |
| 15:20.37 | igcewieling | NOT suitable for people who want to change |
| 15:20.39 | [TK]D-Fender | johnwigley: How attached are you to their solution BTW? |
| 15:21.09 | johnwigley | very, as I use things like their windows cti client which has an operator console, queue statistics, IM, all that sort of stuff in it. |
| 15:21.16 | apb1963 | Forget making the change... how did you FIND what line to change? |
| 15:21.22 | igcewieling | johnwigley: You mean FreePBX. FreePBX has a pre-built distro DVD, but you can also download just FreePBX and install it on your own server you built yourself. |
| 15:21.59 | igcewieling | johnwigley: have you considered giving up on the idea of patching Asterisk and trying to see if you can work around the issue some other way? |
| 15:22.24 | apb1963 | that's a good point... what is it you're trying to change? |
| 15:22.39 | apb1963 | or better yet... what are you trying to accomplish? |
| 15:22.49 | [TK]D-Fender | johnwigley: I highly recommend you deploy it on another system for your experimentation then so you don't risk blowing it up. |
| 15:23.00 | johnwigley | apb1963: I assumed that the queue strategy implementations would be in app_queue.c, downloaded it and then spent 20 minutes searching for where the functionality would likely be, and then another 20 mins understanding how the logic worked, and then pinned down exactly what I needed to change to make it behave as I needed, |
| 15:23.18 | *** join/#asterisk zerohalo (~zerohalo@74.61.196.236) |
| 15:23.25 | [TK]D-Fender | johnwigley: and take note that ones you override their packaging you need to make sure that it doesn't try to assert itself over you afterwards |
| 15:23.33 | *** join/#asterisk Vince-0 (~quassel@196-209-246-90.dynamic.isadsl.co.za) |
| 15:23.39 | *** part/#asterisk igcewieling (~igcewieli@ip98-183-25-31.pn.at.cox.net) |
| 15:24.53 | edge | I have a Cisco SPA50x phone, in the features.conf I've setup #1 as being blind transfer and inside the phone under regional I've set #1 to also be blind transfer. On a Speed dial button I dial #1123 and it sends the #1 to Asterisk and that wasn't expected. How can I either handle this #1 not being dropped by the phone or the server in the dial plan? |
| 15:25.19 | johnwigley | igcewieling: I've tried for ages to work around it, but come to the conclusion that the only way to get it to behave is to make a small change to the way the ringall queueing strategy selects members to call. |
| 15:27.11 | Greenlight | johnwigley: Could you not have a queue failover into another queue to acomplish what you require? |
| 15:29.27 | johnwigley | Greenlight: I could do that, but that then screws the queue statistics which xivo provides, and also complicates logging in and out as I'd have to patch xivo so that agents logged into the failover queues when they logged into the primaries etc. |
| 15:29.47 | Greenlight | Ahh gotcha |
| 15:29.59 | *** join/#asterisk chris_n (~Chris@koha/developer/chris-n) |
| 15:31.53 | johnwigley | What's annoying is that I can fix this with a single line change removing a check condition which took half an hour to find, however to actually get it compiled looks like it's going to take three days to install another linux distro, work out how to set up the build environment, learn how to check out code from the repositories, then learn how to integrate the xivo patches into it, and finally compile it. It's this part that |
| 15:31.54 | [TK]D-Fender | edge: You have a real SIP phone. You shouldn't be using crappy DTMF-based transfers at all... |
| 15:32.23 | johnwigley | what they're doing, with an established build environment to just compile the app_queue module for me in 5 mins :) |
| 15:33.34 | Greenlight | johnwigley: You could try and buiild and compile the same version as you've got running and then swap in the module, but YMMV |
| 15:33.35 | edge | [TK]D-Fender, am i using the syntax wrong. I'm trying to make it so while on a call I can hit a extension button on my call attendant module with a blind transfer. I have this: fnc=sd;ext=#1*9121@$PROXY; The #1 tells the phone i want to blind transfer, but asterisk gets the #1 and that i didn't think would happen |
| 15:33.56 | Greenlight | johnwigley: Literally it takes <5 minutes to download, extract and build asterisk. |
| 15:34.27 | johnwigley | greenlight: ok, I think I'm going to have to bite the bullet and do it. Thanks for all the suggestions. |
| 15:34.45 | Greenlight | But you're taking the chance that they've not made their own customisations to app_queue which you'll remove |
| 15:36.07 | johnwigley | Greenlight: quite possibly, I'll try and work out how to download their patches and integrate them. |
| 15:36.41 | [TK]D-Fender | edge: SIP phones should not be using coding like that to indicate a transfer... |
| 15:38.18 | *** join/#asterisk zerohalo (~zerohalo@74.61.196.236) |
| 15:41.44 | *** join/#asterisk darkbasic_ (~quassel@niko.linuxsystems.it) |
| 15:44.37 | apb1963 | I'm trying to use AMI to originate a call, but it's giving me an error. |
| 15:45.23 | apb1963 | Response: Error Message: Originate failed |
| 15:46.02 | *** join/#asterisk darkbasic (~quassel@niko.linuxsystems.it) |
| 15:46.27 | Greenlight | Pastebin what you're sending |
| 15:46.52 | Greenlight | And the full response |
| 15:47.07 | apb1963 | Input & output here: http://ix.io/5In |
| 15:48.29 | Greenlight | Are you sending the login, then waiting for the response, then sending the originate? |
| 15:48.44 | apb1963 | yes |
| 15:48.53 | Greenlight | Lets see the dialplan ? |
| 15:51.07 | Greenlight | Does 6618365653@ext-meetme-custom 100% exist? |
| 15:51.22 | Greenlight | Be it directly or by a pattern match |
| 15:51.40 | *** join/#asterisk ChadAragorn (~ChadArago@63-235-131-194.dia.static.qwest.net) |
| 15:52.18 | *** join/#asterisk ChadAragorn (~ChadArago@63-235-131-194.dia.static.qwest.net) |
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| 15:55.51 | apb1963 | well that's what I'm a bit confused about...... the first part of it is the number it's trying to dial |
| 15:56.36 | apb1963 | the ext-meetme-custom ... yes, it's a context |
| 15:56.41 | apb1963 | http://ix.io/5IG |
| 15:58.08 | Qwell | Which pattern (hint: there's only one there) do you expect to match that phone number? |
| 15:58.45 | Greenlight | Indeed |
| 15:59.33 | SuperNull | alright. looks like my server is not checking to see if there are any real voicemail files. it does sometimes send a mwi but they are all off and 0 messages (When 2 files exist) |
| 16:07.00 | *** join/#asterisk bchia (~Adium@nat/digium/x-wbojtpxyroellhza) |
| 16:07.46 | *** join/#asterisk igcewieling (~igcewieli@ip98-183-25-31.pn.at.cox.net) |
| 16:08.24 | bchia | Does anyone know where the Asterisk CLI stores its command history? (In particular, can I clear the *CLI history in a manner like `history -c` on the shell?) |
| 16:08.39 | Qwell | bchia: ~/.asterisk_history |
| 16:09.21 | bchia | lol, why didn't I think of that? thanks! |
| 16:11.15 | *** join/#asterisk navaismo (~navaismo@189.191.248.18) |
| 16:15.58 | SuperNull | igcwieling i set that pollin up (voicemail.conf actually... btw) its .. almost like its looking in the wrong place it keeps sending out mwis of 0/no ... |
| 16:17.55 | *** join/#asterisk tadspoles (~tadspoles@nc-184-3-103-49.dhcp.embarqhsd.net) |
| 16:17.57 | asilva | ASTERISK-21787 anyone looking at this ? |
| 16:17.58 | LieutPants | [ASTERISK-21787] [Status: Triage] No IAX2 communication either user/peer or friend accounts - https://issues.asterisk.org/jira/browse/ASTERISK-21787 |
| 16:18.47 | navaismo | deja vú?? |
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| 16:37.46 | SuperNull | just here for the lulz ? |
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| 16:55.27 | apb1963 | Qwell, Greenlight: Thank you both! |
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| 17:22.04 | navaismo | hey asilva do you want to try the connection between your server and mine(11.2.0)? |
| 17:25.36 | *** join/#asterisk HolTech (~IceChat77@pool-74-107-104-2.bltmmd.fios.verizon.net) |
| 17:26.03 | igcewieling | Asterisk 11.4: Has anyone seen a problem where safe_asterisk fails to start Asterisk and keeps trying? asterisk -fvvv works just fine as does -fvvv |
| 17:26.07 | igcewieling | ,,er,,, -cvvv |
| 17:26.39 | eirirs | was gonna ask what the difference was |
| 17:26.39 | eirirs | hehe |
| 17:28.21 | HolTech | So, I'm using a .gsm for some background music in my dialplan, sounds great from a landline but when I call via a cell phone it's choppy. The cell phone is calling to the POTS line, so I don't think this can be a codec thing. Any help ? |
| 17:29.45 | igcewieling | HolTech: the number your calling on Asterisk is POTS or SIP? |
| 17:29.53 | *** join/#asterisk vlad_starkov (~vlad_star@91.206.59.140) |
| 17:29.59 | igcewieling | remember GSM in asterisk has nothing whatsoever to do with cell phones |
| 17:30.27 | igcewieling | Once the call leaves the cell network it is converted to ulaw or alaw (depending on your country) |
| 17:30.29 | HolTech | POTS, right so I'm not sure if maybe I need to encode the .gsm file better or something |
| 17:30.44 | HolTech | but either way its going out on the pots as analog right ? |
| 17:30.57 | igcewieling | HolTech: so you are calling a POTS port (Digium or Sangoma?) ports on your Asterisk box from a land line and a cell phone? |
| 17:32.11 | [TK]D-Fender | HolTech: if you're going to POTS then you should be using G.711 encodings |
| 17:32.28 | [TK]D-Fender | HolTech: Because GSM is already lower quality than POTS baseline |
| 17:32.34 | HolTech | Sorry, let me explain a little clearer. My asterisk is connected to the POTS via an ATA ( sipura 3000). When I call the phone number from my cell phone ( tested with another cell also ), it sounds choppy. If I call the number from some other landline it sounds crisp |
| 17:32.55 | [TK]D-Fender | HolTech: And when the cell carrier gets the audio they'll DOUBLE compress it. |
| 17:33.01 | [TK]D-Fender | HolTech: Hence "bad" |
| 17:34.17 | HolTech | [TK]D-Fender: Am I right to understand that the cell carrier is recieving an analog signal, and then compressing it? |
| 17:34.26 | [TK]D-Fender | HolTech: Of course |
| 17:34.37 | igcewieling | Generally MUSIC over highly compressed codecs like GSM and G729 sounds bad. If your "music" is not very melodic with lots of instruments it may sound fine. |
| 17:34.39 | [TK]D-Fender | Cell < POTS quality. |
| 17:35.01 | [TK]D-Fender | And conversion is bad |
| 17:35.03 | [TK]D-Fender | Any* |
| 17:35.12 | [TK]D-Fender | So reduce those. |
| 17:35.14 | HolTech | So, what is a recommended codec for music on hold, etc, g.711 like you said ? |
| 17:35.27 | [TK]D-Fender | you should be using G.711 to your SPA and your recodings should be in the same |
| 17:35.39 | igcewieling | Addigio For Strings may sound like crap where joan jett's "I love rock n roll" may sound pretty good. |
| 17:36.08 | igcewieling | HolTech: unless you NEED compressed codecs, don't use them. |
| 17:36.29 | HolTech | Understood. Thanks guys ! |
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| 19:21.26 | *** join/#asterisk Lars_G (~Lars@unaffiliated/lars-g/x-000001) |
| 19:21.28 | Lars_G | Hi |
| 19:21.41 | Lars_G | Hast anyone worked with VG1X-8 Gateways? are they reliable? |
| 19:23.01 | *** join/#asterisk serafie (~erin@nat/digium/x-yxedymornmxdeqhk) |
| 19:23.59 | Qwell | Lars_G: "VOPTech"? Highly unlikely. |
| 19:25.55 | Lars_G | :( |
| 19:25.57 | Lars_G | Thanks |
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| 19:28.59 | Lars_G | What about grandstream? |
| 19:29.31 | Lars_G | I'm looking at the Gxe5028 but it looks like a whole integrated pbx, I'm unsure if I can stack several as gateways for an Asterisk install |
| 19:29.55 | igcewieling | Lars_G: the cheapest boxes are not going to be the most reliable boxes |
| 19:30.12 | igcewieling | get a quintum or adtran or cisco |
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| 19:31.12 | Lars_G | Hmmm, cisco costs 10 times more.... I will never have that budget pass. |
| 19:31.27 | Lars_G | And the others, can't be goten locally, if they can't be obtained locally, I can't buy them |
| 19:31.54 | Lars_G | I might end up getting a decent machine, with three digium cards stocked with FXOs |
| 19:33.03 | igcewieling | you are going to make comprises since you are working in a hostile enviroment. |
| 19:33.30 | igcewieling | Adtran can be gotten almost anywhere. |
| 19:35.16 | Lars_G | hmmm |
| 19:35.22 | igcewieling | Europe, Middle East and Africa (EMEA) |
| 19:35.22 | igcewieling | To locate a channel partner in EMEA, contact us at sales.emea@adtran.com. |
| 19:36.33 | Lars_G | igcewieling: if they don't have a partner in my country (Venezuela) I will NOT be able to get their products. |
| 19:36.38 | Lars_G | I'll mail them anyhow. |
| 19:36.53 | igcewieling | Americas (Outside USA) |
| 19:36.54 | igcewieling | To locate a channel partner in the Americas, contact us at sales.ai@adtran.com. |
| 19:43.17 | *** join/#asterisk BrokenArrow (~BrokenArr@unaffiliated/brokenarrow) |
| 19:43.58 | *** join/#asterisk wonderworld (~w@ip-176-199-167-15.unitymediagroup.de) |
| 19:44.44 | wonderworld | hey, is VP8 video support planed for asterisk? |
| 19:44.59 | pabelanger | patches welcome |
| 19:45.09 | pabelanger | but seriously, not that I know of |
| 19:45.59 | pabelanger | I would think pass-thru support would be pretty straightforward |
| 19:46.07 | pabelanger | but doubt any transcoding would be done |
| 19:47.01 | Lars_G | Are there hardware devices using vp8? |
| 19:47.29 | wonderworld | pass-thru would be a great start. experimenting with webRTC. audio works great. was wondering about video. |
| 19:48.19 | navaismo | wonderworld, check the section VP8 Passtrough from this link http://linux.autostatic.com/asterisk-and-sipml5-interoperability |
| 19:48.54 | navaismo | I have applied that patch but never tested Im not using Video yet |
| 19:50.30 | wonderworld | navaismo: yes, i know about this setup, but it requires an extra daemon on server side to do the transcoding |
| 19:51.00 | wonderworld | becuase people made it work with patches i was wondering if there is official devlopement going on |
| 19:53.58 | navaismo | you mean the webrtc2sip? hmmm If i recall the patch work without the media gateway |
| 19:55.49 | wonderworld | ohh nice, didn't know that. the problem with the patch for me was, that is for a specific older asterisk version, so it might not be future proof. |
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| 20:20.26 | navaismo | wonderworld, seems to work with latest asterisk, at least no errors when applied the patch and loading sip |
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| 20:45.09 | wonderworld | navaismo: thanks, will try to use video then |
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| 21:50.21 | wonderworld | hey, i compiled asterisk with srtp support. setting encryption=yes in sip.conf, should encrypt the media path now. calls work but how would i check if they are really encrypted? |
| 21:55.33 | *** join/#asterisk t (tom@freenode/staff/tomaw) |
| 21:55.34 | Tim_Toady | wonderworld: tcpdump during a call is the most reliable way |
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| 22:06.48 | wonderworld | Tim_Toady: thanks, could you give me a hint on how to find out which port to sniff? as i understand, the media channels are created dynamically. how would i find out which port a call is using? |
| 22:07.20 | navaismo | sniff for udp then in wireshark you can filter by voip calls |
| 22:08.02 | *** join/#asterisk TimeRider (~steve@timerider.plus.com) |
| 22:09.32 | [TK]D-Fender | I would consider looking at SIP DEBUG which tells you what is negotiated... |
| 22:10.46 | wonderworld | thanks. found the port with netstat -aup | grep asterisk |
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| 22:15.46 | navaismo | srtp only encrypt the signalling and is needed zrtp for encrypt the media? |
| 22:16.24 | WIMPy | No, both encrypt media. |
| 22:16.29 | WIMPy | only. |
| 22:16.54 | navaismo | media=voice? |
| 22:17.02 | WIMPy | srtp for one link, ztrp end-to-end. |
| 22:17.09 | WIMPy | Whatever. |
| 22:17.29 | navaismo | got it |
| 22:18.09 | wonderworld | ok i tcpdumped two calls, one with encryption=on , one with encryption=off. the output seems to be the same random gibberish to me in both cases |
| 22:20.12 | navaismo | how are you seeing the info of tcpdump? |
| 22:20.49 | wonderworld | i did -> tcpdump udp -w ./capture |
| 22:21.19 | wonderworld | after that looked at it with a text editor |
| 22:24.23 | navaismo | I guess you need to open it with wireshark |
| 22:27.22 | wonderworld | ok, i checked SIP debug. got this. does this mean it works? http://pastebin.com/WWmVti8C |
| 22:28.28 | navaismo | something is wrong with pastebin here can't open it |
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| 22:40.28 | igcewieling | Ya know what would be really awesome? a func_json_encode and a func_json_decode for Asterisk |
| 22:40.46 | WIMPy | Write it. |
| 22:41.24 | igcewieling | WIMPy: I'm an admin, not a programmer |
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| 23:30.01 | *** join/#asterisk ruben231 (~OpenDial@112.198.90.198) |
| 23:30.39 | ruben231 | hi guys install asterisk 1.8 on centos and i get this issue during make -----> http://pastebin.com/gPhD9RfF <--------any idea guys..? |
| 23:30.44 | ruben231 | anyone..? |
| 23:31.53 | WIMPy | Did you do what it told you to do? |
| 23:35.57 | ruben231 | <PROTECTED> |
| 23:36.57 | WIMPy | Did you read the text you pastebinned? |
| 23:39.24 | ruben231 | <PROTECTED> |
| 23:41.17 | ruben231 | :-(help |
| 23:41.40 | [TK]D-Fender | remove the existing menuselect.makeopts <---------- |
| 23:43.19 | ruben231 | [TK]D-Fender: where should i find this..? |
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| 23:45.46 | ruben231 | guys i delete on the asterisk source file, thanks and its compiling now..Thank guys again. |