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| 00:39.48 | plut0 | i'm trying to override the contect for a peer but it seems to use the context from [general] instead, what am i doing wrong? |
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| 00:55.33 | [TK]D-Fender | pluyou are failing to match your peer in the first place. Or you made a typo somewhere. Or you didn't apply config changes you should have. Or 42. |
| 00:56.24 | leifmadsen | plut0: sounds like your device isn't authenticating correctly then and is coming in as a guest |
| 00:58.28 | plut0 | if i disable allowguest i see this in the logs, Sending fake auth rejection for device |
| 01:02.27 | plut0 | what am i missing? |
| 01:04.44 | [TK]D-Fender | You are not SHOWINS us configs an SIP DEBUG from your failed call attempts |
| 01:04.51 | [TK]D-Fender | ~pb |
| 01:04.51 | 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. |
| 01:04.53 | [TK]D-Fender | ^^^ |
| 01:09.50 | plut0 | [TK]D-Fender: http://pastebin.com/pwS6C8Eq |
| 01:09.55 | plut0 | let me know what else you need |
| 01:15.23 | [TK]D-Fender | [21:04][TK]D-FenderYou are not SHOWINS us configs an SIP DEBUG from your failed call attempts |
| 01:16.56 | plut0 | i don't see a 'sip debug' command available |
| 01:17.34 | [TK]D-Fender | "sip set debug on" |
| 01:17.50 | plut0 | ahh |
| 01:26.54 | plut0 | hmm |
| 01:31.34 | [TK]D-Fender | heads out for the night |
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| 01:52.42 | sirsquishy | AsteriskNow 1.8 to Asterisknow2.0.2 now i get one way audio, the ONLY difference is the version. what am I missing... |
| 01:57.29 | sirsquishy | i know im soooo descriptive; so SIP ATA to SIP Softphone, one way audio, SIP Softphone through SIP Trunk to Cell Number/landline one way audio; on the 1.8(which none of the device settings changed) it worked fine. The Asterisk box is hosted on the public Internet, no NAT and no 'firewall' involved. |
| 01:58.10 | sirsquishy | NAT is on the end points, however, aside from my Android Softphone which is over the verizon public network |
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| 04:30.20 | Korolev | hey guys, cant find an answer to this problem anywhere |
| 04:30.53 | Korolev | trying to do MP3Player(blah), all I get is: app_mp3.c:133 timed_read: Poll timed out/errored out with 0 |
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| 06:25.41 | bulkorok | hi |
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| 07:02.20 | schmidts | good morning |
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| 07:10.54 | ChannelZ | TGIF! |
| 07:14.59 | Glonet | Hello, i have a problem with my asterisk. When my internet is going down and up again, my peer want register himself back. When i change my ip of my Asterisk my peer is registered to my provider... I get an registration timed out |
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| 07:24.07 | schmidts | ChannelZ +1 |
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| 07:51.09 | FlashDeluxe | hi! i got a problem. If i call my cell phone (i am using a leading "0") everything works fine eg 001577123456. But if i add the area code of my country (germany), the call breaks up e.g. 000491577123455. Here is the output http://paste.debian.net/201746/ Can somebody help me please? |
| 07:57.15 | FlashDeluxe | is that maybe the problem CAPI/g1/00491577123456? Should the "0049" be converted or should that work? |
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| 08:12.18 | jacekowski | hi people |
| 08:12.54 | Glonet | Hello, i have a problem with my asterisk. When my internet is going down and up again, my peer want register himself back. When i change my ip of my Asterisk my peer is registered to my provider... I get an registration timed out |
| 08:13.17 | jacekowski | i'm looking for some .pdf about asterisk that i could show to the management |
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| 08:52.31 | davlefou | Hi, i try to use voicemail.conf |
| 08:53.49 | nW44b | hi davlefou (hum hum) |
| 08:54.14 | davlefou | Actually, i try to send the mail for notification |
| 08:58.56 | nW44b | davlefou: and ? |
| 09:02.52 | davlefou | But, i don't know if a have to do something configuration of sendmail. |
| 09:03.16 | davlefou | or i can give mail option in asterisk? |
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| 09:21.04 | Glonet | somebody who can help me out with my problem? |
| 09:21.54 | kaldemar | what is your problem? |
| 09:22.41 | kaldemar | where do you see a registration timeout? |
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| 09:26.53 | Glonet | in my Centos |
| 09:26.58 | Glonet | and in my logs |
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| 09:28.01 | Glonet | when i change my ip of my asteisk my peer is registered again |
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| 13:25.55 | kchehab | i was using res_mysql to get data ex exten => _X.,1,MYSQL(Connect connid localhost |
| 13:26.10 | kchehab | now iam using res_odbc i should use same way |
| 13:26.22 | kchehab | or its has another syntax |
| 13:27.25 | leifmadsen | kchehab: different syntax -- you don't have to setup the connection |
| 13:27.50 | leifmadsen | http://www.asteriskdocs.org/en/3rd_Edition/asterisk-book-html-chunk/installing_configuring_odbc.html |
| 13:27.53 | leifmadsen | start there |
| 13:27.59 | leifmadsen | then look at the example usage of func_odbc |
| 13:28.18 | leifmadsen | http://www.asteriskdocs.org/en/3rd_Edition/asterisk-book-html-chunk/Database_id287624.html |
| 13:29.36 | kchehab | thansk |
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| 13:37.30 | carrar | happy happy |
| 13:37.34 | carrar | happy happy FRIDAY |
| 13:38.16 | carrar | gives Katty the happy friday nudge |
| 13:39.01 | carrar | someone write my documentation for me!! |
| 13:41.44 | carrar | fires up LaTeX |
| 13:45.23 | kchehab | leifmadsen please can you check where is the error ,i can do a query using rres_odbc ,my config locateed at http://pastebin.com/bdrXeBVh |
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| 13:46.53 | leifmadsen | kchehab: ODBC_WHITELIST |
| 13:46.57 | leifmadsen | functions must be entirely uppercase |
| 13:47.26 | leifmadsen | kchehab: also get rid of that MYSQL() stuff as it's not at all necessary |
| 13:47.30 | kchehab | how can i check that [whitelist] i add in fun_odbc.conf is loaded in memory |
| 13:47.34 | leifmadsen | ODBC will handle the connection setup and teardown |
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| 13:47.43 | leifmadsen | kchehab: core show function ODBC_WHITELIST |
| 13:50.10 | kchehab | leifmadsen core show function ODBC_whitelist shows the function in lower case |
| 13:50.18 | leifmadsen | that is invalid |
| 13:50.21 | leifmadsen | make everything uppercase |
| 13:51.24 | kchehab | leifmadsen i did the upper case in fun_odbc.conf and in config but still not working |
| 13:51.40 | kchehab | leifmadsen is their a deeper debug mode to check the problem ? |
| 13:52.16 | leifmadsen | nope, you need to look at the sql server side with query debug |
| 13:52.21 | leifmadsen | you also haven't shown what the function returns |
| 13:53.16 | kchehab | leifmadsen how can see the function returns please check http://pastebin.com/4Q2DM05z |
| 13:53.32 | leifmadsen | use a NoOp() or Verbose() |
| 13:53.35 | kchehab | leifmadsen can i do a test for the function |
| 13:54.22 | kchehab | please can you specify more i fi print a noop ( ?????) what should i add inside to get the mysql return |
| 13:54.33 | leifmadsen | NoOp(${ODBC_WHITELIST(...)}) |
| 13:54.39 | leifmadsen | replace ... with something appropriate |
| 13:56.14 | kchehab | you mean after exten => _96771.,1,GotoIf($[${ODBC_WHITELIST(${EXTEN})}]?conn,1:fail,1) ,exten => _96771.,2,NoOp(${ODBC_WHITELIST(967712171717)}) ? |
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| 13:57.13 | leifmadsen | before |
| 13:57.18 | leifmadsen | if it's after it'll never hit |
| 13:57.25 | kchehab | leifmadsen yes sure it returns NoOp("SIP/office-00000012", "") in new stack |
| 13:57.45 | kchehab | office is the name of the incoming call from trunk |
| 13:58.45 | kchehab | leifmadsen it returns nothing |
| 13:59.13 | leifmadsen | then there probably isn't any rows being returned |
| 13:59.26 | leifmadsen | look at the ODBC_ROWS channel variable |
| 13:59.42 | kchehab | leifmadsen i will check something i sec |
| 14:00.01 | leifmadsen | it's working as expected since nothing is returned |
| 14:05.32 | kchehab | leifmadsen i do |
| 14:05.32 | kchehab | Database changed |
| 14:05.32 | kchehab | mysql> SELECT `number` FROM `Yemen` WHERE `number` ='967712171717' |
| 14:05.32 | kchehab | <PROTECTED> |
| 14:05.46 | kchehab | and there is a return of 4 rows |
| 14:06.59 | mcolombo | hi all! |
| 14:07.05 | carrar | well hello |
| 14:07.35 | mcolombo | i would insert in the SDP : rtpmap: 18, anybody can help me? |
| 14:07.43 | mcolombo | thanks |
| 14:10.26 | kchehab | leifmadsen i find this in the ful debug res_odbc.c: Class 'asterisk-connector' not found! |
| 14:11.14 | kchehab | leifmadsen but odbc show all |
| 14:11.33 | kchehab | <PROTECTED> |
| 14:12.25 | kchehab | leifmadsen i change the DSN to asterisk and it worked thanks |
| 14:19.45 | *** part/#asterisk sekil (~sekil@78.24.104.73) |
| 14:23.18 | kchehab | leifmadsen in res_odbc.conf limit=1 |
| 14:23.26 | kchehab | mean just one query per sec ? |
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| 14:26.03 | *** join/#asterisk invadeLTD (~invadeLTD@178.78.119.76) |
| 14:26.44 | invadeLTD | hey, wondering if anyone can point me to some good resources for understanding the output from "pri debug" and "pri intense debug" asterisk CLI commands |
| 14:27.30 | dwayne | invadeLTD, http://www.corp.att.com/cpetesting/techrefs/41459.html |
| 14:28.15 | invadeLTD | great :) thanks a lot dwayne this looks like exactly what i was searching for |
| 14:28.44 | dwayne | invadeLTD, no problem. Enjoy |
| 14:29.19 | tzanger | fjfjfjfjf |
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| 14:56.29 | xSmurf | Is it at all possible to compile extra modules to add to an rpm install? I can't seem to get the configs options right and always endup with "this module was compiled with different options" error (even when I build from the rpm source) |
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| 15:02.35 | kaldemar | xSmurf: get the sources from your distribution. |
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| 15:04.35 | xSmurf | kaldemar: yes that's what I did (rpm source) |
| 15:04.43 | xSmurf | built with rpmbuild |
| 15:05.13 | kaldemar | xSmurf: you better compile everything while you're at it. that would be a custom install anyway. |
| 15:05.32 | xSmurf | I'd much rather have only a couple of modules to track |
| 15:05.47 | xSmurf | modules that are not 100% mission critical and can fail, while the rest can't |
| 15:06.36 | kaldemar | i wouldn't trust pre-packaged binaries any more than self-compiled ones. |
| 15:06.58 | xSmurf | it's not about trust, it's about having to maintain it |
| 15:07.21 | ijpalmer | hi all, I have a snom phone connected to asterisk 1.8.5. This phone receives inbound calls from everywhere correctly, except from one location where I get the message 'Got sip reponse 400 bad request back from ipaddress of phone' Anyone have any idea what's going on |
| 15:09.02 | xSmurf | also I'm intrigued as to why rebuilding a pkg from it's source would yield different configuration options |
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| 15:11.20 | invadeLTD | ijpalmer, would need more info but in cases of inbound calls failing the first place to look is the dialplan |
| 15:13.07 | invadeLTD | sounds like the IP addressing is going wrong somewhere, if you really want to see the guts of the problem try doing a packet capture and looking at the messages in something like wireshark |
| 15:14.06 | ijpalmer | invadeLTD: thanks for the reply, I don't think it's the dialplan as normally it works. The call comes across an ISDN line from the telco, except when the call comes from this one location it fails |
| 15:14.44 | invadeLTD | ok, so the context the ISDN lines sit in works, where is this this other location? |
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| 15:16.47 | ijpalmer | invadeLTD: the other location is an office with a cisco pbx sending calls across the isdn line |
| 15:17.08 | invadeLTD | okay and in the asterisk logs you see this call come into the isdn lines? |
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| 15:19.02 | ijpalmer | invadeLTD: yeah I watch the asterisk cli AND CAN SEE THE CALL HIT THE DIALPLAN IT |
| 15:19.18 | ijpalmer | invaddeLTD: sorry fat fingers |
| 15:19.27 | invadeLTD | lol, no worries |
| 15:19.32 | ijpalmer | I'll start again |
| 15:21.57 | ijpalmer | i watch the call come in and hit the dialplan it's routed through right up to the point it tries to call the handset, it then gets a message saying IAX2channel Requested special control 20, SIP/channel. Followed by Got SIP response 400 bad request back from handset ip address |
| 15:22.14 | *** join/#asterisk Hive (~Hive@173-165-205-1-jacksonville.hfc.comcastbusiness.net) |
| 15:22.32 | invadeLTD | iv not used snom myslef but this thread mentions problems with SNOM and sip encryption http://www.elastix.org/index.php/en/component/kunena/31-asterisk/92863-incoming-call-got-sip-response-400-qbad-requestq.html |
| 15:24.33 | invadeLTD | also, are you using full logging with max verbosity and debug? that a sip trace are really needed for any deeper troubleshooting |
| 15:24.51 | ijpalmer | invadeLTD: i did read that thread which lead me on to upgrade the Snom's firmware, still no joy |
| 15:25.48 | ijpalmer | invadeLTD: you are correct I need to turn on sip debug to really see what's going on. I may also route the call into a recorded message to take the handset out of the equation |
| 15:27.10 | invadeLTD | yeah i would try taking the handset out of the equation, it mauy be something external to asterisk causing the problem in which case - packet capture the traffic and look at the sip messages |
| 15:29.07 | ijpalmer | invadeLTD: thanks a little more investigation on my part I think, but you've given me some ideas of where to go cheers |
| 15:29.15 | invadeLTD | no problem |
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| 15:54.13 | *** join/#asterisk onebitxajax (~onebitxaj@unaffiliated/onebitxajax) |
| 15:54.16 | onebitxajax | hi to all |
| 15:54.27 | onebitxajax | is there any guide to configura a conference room? |
| 15:54.28 | onebitxajax | . |
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| 15:58.48 | invadeLTD | http://www.voip-info.org/wiki/view/Asterisk+cmd+MeetMe |
| 15:59.06 | invadeLTD | onebitxajax: thats what you need, the meetme app |
| 16:00.33 | WIMPy | Better use https://wiki.asterisk.org/wiki/display/AST/ConfBridge+10 |
| 16:01.09 | invadeLTD | WIMPy: thanks your right, conf bridge is whats used in the most up to date asterisk installs |
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| 16:02.11 | onebitxajax | invadeLTD: WIMPy thanks |
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| 16:42.11 | *** join/#asterisk devCloud (~devand@209.5.190.9) |
| 16:42.54 | devCloud | hello. |
| 16:44.44 | devCloud | i am having an issue building asterisk 1.8.11-cert7 using a menuselect.makeopts file copied to both ~/.asterisk.makeopts and /etc/asterisk.makeopts |
| 16:45.08 | devCloud | anyone here that might know why that would be? |
| 16:45.14 | drmessano | What is the issue? |
| 16:45.20 | devCloud | it doesn't use the options. |
| 16:45.22 | devCloud | lol. |
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| 16:45.47 | drmessano | Hang on.. Why do you haver it in those locations? |
| 16:46.12 | drmessano | Where are you building it from? |
| 16:47.04 | devCloud | According to UPGRADE-1.4.txt in the top dir of the asterisk installation folder, "If you would like to save your choices and have them applied against all |
| 16:47.04 | devCloud | builds, the file can be copied to '~/.asterisk.makeopts' or |
| 16:47.05 | devCloud | '/etc/asterisk.makeopts'." |
| 16:48.14 | drmessano | Have you looked at that? https://issues.asterisk.org/jira/browse/ASTERISK-18137 |
| 16:48.54 | devCloud | i had, but i also found sources that said it was resolved in the following build. |
| 16:50.45 | devCloud | what i'm trying to do is write an installation script that will install asterisk with all of the modules that i want (and their dependencies before them obvs.) from a fresh install of RHEL 6.1 |
| 16:50.58 | devCloud | other than this, it works flawlessly. |
| 16:52.44 | devCloud | one thing i've found -does- work, is if i enter makeselect and exit with "x", and then copy over my menuselect.makeopts file with the desired modules setup to the top dir of the installation, it works as intended. |
| 16:52.52 | devCloud | but even if i run |
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| 16:54.00 | devCloud | make menuselect -& |
| 16:54.02 | devCloud | and then killall -9 make |
| 16:54.10 | devCloud | it doesn't work. |
| 16:54.46 | devCloud | i'm guessing it must write the necessary config to tell make to pull those makeopts on exit. |
| 16:56.55 | devCloud | do you know if it is possible to tell my bash script to run "make menuselect" and then input "x" to it after say… 10-15 seconds? |
| 16:58.59 | *** join/#asterisk brdude (~brdude@12.155.183.30) |
| 16:59.09 | carrar | Why not just edit the file that program creates? |
| 16:59.40 | devCloud | well, i would love to… but when you run make menuselect it waits for user input. |
| 17:00.01 | carrar | which is why you would edit the file instead of running menuselect |
| 17:00.22 | carrar | I'm assuming you can do that of course |
| 17:00.43 | devCloud | i'm not certain what all it changes. |
| 17:00.59 | devCloud | i know it creates the menuselect.makeopts and menuselect.makedeps files. |
| 17:01.46 | devCloud | but i'm fairly certain it also changes something inside of the makefile, which i'm not very fluent in. |
| 17:02.13 | devCloud | is it possible to run menuselect in a hyper-verbose mode? |
| 17:03.49 | carrar | Might want to read the README file |
| 17:03.55 | carrar | in the menuselect directory |
| 17:04.30 | carrar | You'll fine your answers there |
| 17:04.33 | carrar | find |
| 17:05.55 | devCloud | You sir, are a star. |
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| 17:06.03 | devCloud | RTFM, amirite? |
| 17:07.33 | carrar | puts his star over his iphone camera |
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| 17:13.30 | devCloud | alternatively, apparently i can just "cp -r /root/install_files/asterisk/menuselect* ." after i run ./configure and ./contrib/scripts/get_mp3_source.sh, and that works. |
| 17:13.35 | devCloud | \o/ |
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| 18:12.19 | GeoGeek | Should I ask questions about IncrediblePBX here or is there a better place for that? |
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| 18:14.17 | GeoGeek | Nevermind....found #vuc.... |
| 18:14.25 | Qwell | vuc is completely unrelated... |
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| 18:44.39 | jmls1 | anyone else running large volume of calls through asterisk-11 ? |
| 18:45.02 | jmls1 | wanna compare notes ? We're now at uptime 2 days, 11 hours, 12 minutes, 15 seconds, 39717 calls processed |
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| 18:49.32 | Merlin | you mean 1.8.11? |
| 18:49.53 | Merlin | i dunno what you mean |
| 18:51.00 | mjordan | Merlin: I think he means Asterisk 11 :-) |
| 18:51.42 | Merlin | i.e. the version with digium ip phone support? |
| 18:52.56 | mjordan | Merlin: no, although Asterisk 11 is compatible with res_digium_phones. |
| 18:53.20 | Merlin | why would someone use asterisk 1.8 vs. 10/11 ? |
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| 18:56.47 | drmessano | Thats a good question |
| 18:56.50 | drmessano | WHy would they |
| 18:58.05 | Katty | hi kids. |
| 18:58.23 | Merlin | yeah |
| 18:58.43 | Merlin | so i asked this question yesterday, and someone responded, but with an answer that didn't make sense, so i'm going to ask again (sorry for the repeat) |
| 18:58.49 | Merlin | what is the right way to enforce a global wrap-up time for agents in multiple queues in asterisk 1.6? is it shared_lastcall in the queue config, or wrapuptime in agents.conf ? |
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| 19:03.01 | Katty | glomps carrar |
| 19:03.09 | Katty | carrar: HAPPI FRIDAI |
| 19:03.28 | Katty | carrar: ermagerd!! freder!!! |
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| 19:24.29 | carrar | Katty!! |
| 19:24.30 | carrar | Yes |
| 19:24.33 | carrar | Thanks |
| 19:24.41 | carrar | HAPPY F R I D A Y 2u2 |
| 19:24.42 | Katty | :> |
| 19:26.18 | carrar | Katty |
| 19:26.37 | carrar | Katty walk over to someone at digium in the registration dept and smack them on the head for me |
| 19:26.53 | carrar | it doesn't matter who |
| 19:27.41 | carrar | man I need lunch |
| 19:27.43 | carrar | & |
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| 20:16.00 | rrittgarn1 | afternoon gentlemen |
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| 21:38.18 | Micc | is there a way to get the number of people in a confbridge from the dialplan? |
| 21:39.36 | mjordan | Micc: https://wiki.asterisk.org/wiki/display/AST/Asterisk+11+Function_CONFBRIDGE_INFO |
| 21:39.39 | mjordan | use the parties option |
| 21:39.59 | Micc | thanks |
| 21:40.12 | Micc | is this only in asterisk 11? |
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