| 00:02.14 | *** join/#asterisk ipengineer (~zconkle@108-206-120-218.lightspeed.mckntx.sbcglobal.net) |
| 00:03.07 | Worsedah | Thanks for the help guys |
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| 01:14.04 | wormEZ | hi, i have downloaded and installed astersikNOW and installed with the freePBX (opetion 1) every time i try to change the admin password, it gives me a SQL error, is this normal / common ? |
| 01:14.25 | WIMPy | Try to ask on #freepbx |
| 01:14.33 | wormEZ | sorry and thanks wimpy :-) |
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| 01:15.08 | wormEZ | hi, i have downloaded and installed astersikNOW and installed with the freePBX (opetion 1) every time i try to change the admin password, it gives me a SQL error, is this normal / common ? |
| 01:15.17 | wormEZ | shit, wrong channel, sorry |
| 01:15.27 | wormEZ | still getting used to this irc client! |
| 01:17.39 | wormEZ | in the interium however, i must say that everything i have read about asterisks makes it sure look like a nice product, im not really sold on the whole webUI for it though, do you think that is really actually a good plan ??? |
| 01:18.03 | WIMPy | Depends on your plans. |
| 01:18.15 | WIMPy | If you want a quich start, a GUI helps. |
| 01:18.39 | WIMPy | If you want to understand what's going on, skip it. |
| 01:20.24 | wormEZ | i would like to be able to understand things at a personal level, but it also have to be easy to maintain (and easy means that i can go on holidays one day... |
| 01:21.10 | WIMPy | If you use a GUI you will have a hard time changing stuff yourself. |
| 01:21.26 | wormEZ | one of the things i have not looked into yet is intergration with ldap / odbc / open directory / etc - i presume it will exist because of how well it all presents |
| 01:24.02 | wormEZ | i actually set it up working following the instructions for editing the config files, with voice mail, (yes still basic) within about two hours, 5 extensions, no outbound calls, afterwards i installed the freePBX WUI and have not been able to understand much past looking at a browser window thinking *where* is the section i added the sip registration and the extension's and then realised i had better read the doco for the WUI then i di |
| 01:24.03 | wormEZ | scovered the error on the sql... then i though MAYBE i did something, but i dont think i did |
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| 03:45.09 | vader-- | Can anyone tell me if this is a valid number to send to flowroute to dial someone in the UK: 441617779554 |
| 03:46.10 | ChannelZ | from where? don't they usually have to start with 0 to dial international |
| 03:46.33 | ChannelZ | (says the guy who never calls international, so...) |
| 03:46.41 | vader-- | ya im not sure |
| 03:46.47 | vader-- | here is what im getting in the cli |
| 03:46.48 | vader-- | http://pastebin.com/yZCAa9iJ |
| 03:47.14 | vader-- | this is what they recommend as a dial plan |
| 03:47.16 | vader-- | exten => _011.,1,Dial(SIP/${EXTEN:3}@flowroute) |
| 03:47.16 | vader-- | ;dialing format - SIP/{countrycode}{number}@flowroute |
| 03:47.36 | ChannelZ | oh.. well you are dialing 011.. |
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| 03:49.19 | ChannelZ | but you're stripping off the 011 somewhere (is this a script generating the dial? Is that why you're using local channels?) |
| 03:49.53 | vader-- | ya |
| 03:50.39 | ChannelZ | but either way no idea what flowroute in particular wants |
| 03:51.23 | ChannelZ | One random thought is do they have some sort of international disable switch? With Vitelity I have to go in and allow international calls in my account |
| 03:52.36 | ChannelZ | https://support.flowroute.com/entries/22587729-What-dialing-format-should-I-use-when-sending-calls- "We use the E.164 format: [CountryCode][Number]" |
| 03:53.32 | ChannelZ | So yeah check and see if it's just an account access problem or something. Although I don't actually see indication that your call is failing. |
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| 10:16.37 | skrusty | morning |
| 10:17.17 | mirela666 | morning |
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| 10:24.04 | Greenlight | hmm silly question, but I want to do asterisk -rx "dialplan reload" and have the verbose output sent to the CLI, how do I go about this. I'm sure I've had this setup before, and *thought* it was the verbose setting in asterisk.conf, but that doesn't seem to help |
| 10:24.04 | michael_work | anyone using asterisk on debian and can advice what is more proper way "inittab respawn" or safe_Asterisk? |
| 10:24.35 | michael_work | Greenlight, logger.conf -> console (not sure) |
| 10:25.48 | Greenlight | Yes, logger.conf has console => notice,warning,error,degbug |
| 10:26.39 | mirela666 | Greenlight: maybe your verbosity is less then 5 on * 11+ you need verbosity 5+ |
| 10:26.53 | mirela666 | something like that |
| 10:27.10 | Greenlight | Oh, I was using 3... which I thought would do it. Let me try 5 |
| 10:27.28 | mirela666 | I was stuck on the same thing |
| 10:27.49 | Greenlight | na... still nothing other than "Dialplan reloaded" ;/ |
| 10:28.23 | mirela666 | hm |
| 10:28.26 | Chainsaw | Greenlight: I'm getting a bit more console output with a -cdnrx "dialplan reload" |
| 10:28.33 | Chainsaw | Greenlight: But indeed, it feels like it's holding back. |
| 10:28.48 | Greenlight | Ahh I didn't think to check of other switches... hold on |
| 10:29.24 | Greenlight | Wait, doesn't "c" start it from scratch |
| 10:29.39 | Chainsaw | Greenlight: That specific instance, yes. |
| 10:29.47 | Chainsaw | Greenlight: But not your main daemon. |
| 10:29.52 | Greenlight | But wouldn't that start it in console mode |
| 10:30.33 | Greenlight | I use that if I'm getting startup issues and the daemon ins't running to start it |
| 10:31.51 | Chainsaw | Greenlight: And capital R instead of r is insufficient? |
| 10:32.12 | Chainsaw | Greenlight: Hrm, no. I misread that. |
| 10:32.38 | Greenlight | Yea, just the basic output, none of the actual dialplan reloading info |
| 10:32.57 | Greenlight | Like if I was at the console normally, with say verbose 3, and did a "dialplan reload" |
| 10:33.01 | Greenlight | All that info that it spits out. |
| 10:33.03 | Greenlight | I want that ;) |
| 10:33.15 | Chainsaw | Greenlight: I understand what you're wanting to do. And it seems like such a trivial request. |
| 10:33.33 | Chainsaw | Greenlight: But I am going to have to let you win. Trial & error here suggests it is not possible. |
| 10:33.35 | Greenlight | QUiet. I presume it was a lack of caffine and someone would point it out to me :) |
| 10:33.49 | Greenlight | *presumed |
| 10:33.54 | Greenlight | *quite |
| 10:34.03 | Greenlight | Arg, can't type today. |
| 10:34.05 | Chainsaw | Greenlight: They're all staying quiet. Which is generally a bad sign. |
| 10:34.25 | Chainsaw | Greenlight: There must be someone with a machine gun in the back who mows down anyone who says "I don't know". |
| 10:34.34 | Greenlight | :) |
| 10:34.55 | Greenlight | I'm almost *sure* I've seen it output that stuff here before. |
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| 10:38.08 | mirela666 | so are you running * as a deamon? what does ps aux | grep asterisk says |
| 10:47.32 | Greenlight | Yes, as deamon |
| 10:48.09 | Greenlight | [root@asterisk ~]# ps aux | grep asterisk |
| 10:48.09 | Greenlight | root 2245 0.0 0.0 108164 764 ? S Jan06 0:00 /bin/sh /usr/sbin/safe_asterisk |
| 10:48.09 | Greenlight | asterisk 2249 33.0 0.2 4408128 54584 ? Sl Jan06 741:35 /usr/sbin/asterisk -f -U asterisk -vvvg -c |
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| 11:00.06 | catphish | i've got a problem with res_fax_spandsp being unable to create a timing source, i believe this is because asterisk was started and the modules were loaded prior to dahdi's kernel modules being loaded, is there any way i might be able to fix this without restarting asterisk as a whole? |
| 11:00.35 | catphish | loading the dadhdi modules fixed meetme immediately, but not spandsp |
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| 11:01.34 | catphish | attempting to unload res_fax_spandsp fails with "'res_fax_spandsp.so' has use count 32" but i'm not sure what is using it |
| 11:07.19 | bsdice | <PROTECTED> |
| 11:12.32 | Chainsaw | catphish: And a module show has nothing else T38 or fax related? |
| 11:13.10 | Chainsaw | catphish: The res(ource) modules tend to be low-level plumbing that have high-level consumers. |
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| 11:14.23 | catphish | Chainsaw: i have res_fax.so and res_fax_spandsp.so |
| 11:14.32 | catphish | res_fax.so has a use count of 1 |
| 11:15.15 | Chainsaw | catphish: If res_fax.so is the only consumer you might be able to unload that. |
| 11:15.28 | Chainsaw | catphish: If not, consider who the consumer of res_fax is, unload that, then unload res_fax, then unload res_fax_spandsp. |
| 11:15.55 | catphish | "Soft unload failed, 'res_fax.so' has use count 1" - i'm not sure what would be using that though |
| 11:16.39 | Chainsaw | catphish: That information is not recorded, so you will have to make an educated guess. |
| 11:17.21 | catphish | ok, thanks |
| 11:17.55 | Chainsaw | catphish: The use count will drop to 0 when you get it right. |
| 11:18.26 | catphish | ok, thanks, i'm afraid i have no clue what's using it though |
| 11:18.35 | wdoekes | res_fax is used by res_fax_spandsp, and res_fax_spandsp is probably used by one or more open calls. you could try to disable fax in the chan_* drivers |
| 11:18.44 | catphish | the use count 32 makes me suspicious because that's the number of failed fax calls |
| 11:19.00 | catphish | i suspect something hasn't closed cleanly |
| 11:19.32 | wdoekes | that might be the case too. if you can reproduce that on a recent asterisk, that would be something to file a bug report about |
| 11:19.39 | catphish | there definitely aren't 32 genuinely open fax channels |
| 11:19.48 | catphish | sure, i'm afraid my asterisk is old |
| 11:20.05 | catphish | embarrassingly so |
| 11:20.35 | Chainsaw | That will reduce the chances of a bug report being actioned, yes. |
| 11:20.50 | catphish | the bug may even already be fixed :) |
| 11:20.55 | catphish | (assuming it is indeed a bug) |
| 11:20.56 | Chainsaw | Precisely. |
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| 11:23.45 | Blashyrkh | how do i find out what is the default setting for a setting in an asterisk conf? in my case iam wondering if overlapdial in chan_dahdi is by default off or on? |
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| 11:39.38 | Neoti | Blashyrkh: you can look at the sample .conf files |
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| 12:04.03 | michael_work | let me be annoying and as k again. anyone uses debian and have asterisk from source? and would advice to use initba for respawn or safe_asterisk ? |
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| 12:39.19 | ipalmer | Hello all I'm running asterisk 1.8 and have had a weird issue with the sound across a NAT. I'm using SIP realtime and had both endpoints set to directmedia and natting set to yes. I could hear no sound, I then set directmedia on the internal LAN to no (the only endpoint who could hear sound had this setting) still no sound. I then configured my dialplan to record internal calls using the monitor app and the sound magically worked, remove the call rec |
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| 12:42.04 | gusto | so |
| 12:42.50 | gusto | up 1 day /sys/fs/ext4/mmcblk0p5/session_write_kbytes 188 |
| 12:42.56 | gusto | looks good, would i say |
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| 14:11.36 | Katty | drmessano: yes, yes it was VERY cute. |
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| 14:38.16 | gmalsack | in need of those who can think outside the box!!!!!!!! |
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| 14:39.01 | file | o.o |
| 14:39.16 | Chainsaw | Wow, that was very loud. So many exclamation marks. |
| 14:39.46 | gmalsack | have the same queue on 2 different asterisk machines. same phones registered to both asterisk machines. need phone 1 to not ring when a call comes into queue on asterisk 2 while phone 1 is on a queue call from asterisk 1 |
| 14:41.27 | gmalsack | both asterisk boxes are running realtime looking at the same mysql db. was hoping phone status would be held in the realtime db rather than astdb but that doesn't seem to be the case, so now I'm in need of an "outside the box" solution/workaround.... |
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| 14:43.30 | [TK]D-Fender | gmalsack: use a composite hint with a distributed device state between the 2 boxes as the member state device on each queue |
| 14:44.02 | [TK]D-Fender | Not really "outside the box"... this is pretty normal as things go |
| 14:45.12 | gmalsack | [TK]D-Fender: cool. now, can you state that in english... lol |
| 14:45.30 | gmalsack | what is a composite hint? |
| 14:46.00 | [TK]D-Fender | "distributed device state" "asterisk pubsub" < - GOOGLE |
| 14:46.22 | gmalsack | [TK]D-Fender: AWESOME!!!! you sooooo da man... thanks buddy! |
| 14:46.30 | [TK]D-Fender | exten blah,hint,something&somethingelse |
| 14:48.43 | [TK]D-Fender | + => |
| 14:53.19 | *** join/#asterisk ipengineer (~zconkle@static-71-252-134-63.dllstx.fios.verizon.net) |
| 14:53.52 | *** join/#asterisk bananapie (~david@static-dslcom4-242.express.oricom.ca) |
| 14:54.55 | bananapie | Hello, I am using the format 'SIP/user@domain' for some calls , I want these calls to go through a proxy. I don't want my other peers to go through a proxy. How can I tell asterisk to send sip calls without a peer via a proxy? |
| 14:55.00 | *** join/#asterisk protocoldoug (~doug@unaffiliated/protocoldoug) |
| 14:55.55 | *** join/#asterisk leedm777 (~leedm777@nat/digium/x-sqbvdlkqtwqrmuwz) |
| 14:56.18 | bananapie | I tried setting outboundproxy=10.0.0.3 in sip.conf general and outboundproxy= in my peers, but it is ignoring my outboundproxy= because it is empty. |
| 14:56.41 | [TK]D-Fender | bananapie: Set your peer's proxy as the host |
| 14:57.01 | bananapie | but when I call 'sip/user@domain' using entries from e164, there is no peer |
| 14:57.35 | [TK]D-Fender | [09:56]bananapieI tried setting outboundproxy=10.0.0.3 in sip.conf general and outboundproxy= in my peers, but it is ignoring my outboundproxy= because it is empty. <- don't leave it empty |
| 14:58.07 | bananapie | but for my peers ( local phones and stuff ) there is no proxy. Phones are directly registered with host=dynamic. |
| 14:58.33 | *** join/#asterisk Hilverdays (~jhilverda@office.voys.nl) |
| 15:01.30 | Katty | eyes bananapie with a fork |
| 15:01.44 | bananapie | I did something horribly wrong didn't I ? |
| 15:01.48 | [TK]D-Fender | Guess that's just not happening with chan_sip on a single instance |
| 15:02.44 | Katty | pokes at bananapie |
| 15:05.14 | *** join/#asterisk newtonr (~newtonr@nat/digium/x-blwymmxdqscbuipk) |
| 15:05.15 | *** mode/#asterisk [+o newtonr] by ChanServ |
| 15:10.38 | Blashyrkh | when i make changes in the sip.conf is a sip reload enough for asterisk to read the changes? |
| 15:13.19 | Qwell | Blashyrkh: for most things |
| 15:16.25 | gmalsack | http://pastebin.com/aNhYmNii 7011 is actually on a call.... thoughts? |
| 15:17.19 | [TK]D-Fender | gmalsack: Where do we see you setting the state device for the member? |
| 15:17.23 | newtonr | gmalsack, callcounter enabled for those peers? |
| 15:17.24 | *** join/#asterisk Bkhan (dd847442@gateway/web/freenode/ip.221.132.116.66) |
| 15:17.33 | [TK]D-Fender | that too |
| 15:17.41 | Bkhan | Hi |
| 15:17.57 | Bkhan | Starting asterisk: Unable to connect to remote asterisk (does /var/run/asterisk/asterisk.ctl exist? |
| 15:18.14 | newtonr | Bkhan, does it exist? |
| 15:18.21 | [TK]D-Fender | Asterisk isn't running, or the user you are running as has no permission to connect |
| 15:18.26 | Bkhan | I am getting this error please help. Due to it no module is working |
| 15:18.28 | Bkhan | yes |
| 15:18.42 | [TK]D-Fender | go start Asterisk MANUALLY and see what happens |
| 15:19.10 | Bkhan | Should i delete and recreate it |
| 15:19.15 | Bkhan | yes |
| 15:19.36 | gmalsack | [TK]D-Fender: don't understand what setting a state device means... |
| 15:20.14 | newtonr | gmalsack, he might mean the state interface? as in "; syntax: member => interface,[,penalty][,membername][,state_interface][,ringinuse]" |
| 15:20.23 | [TK]D-Fender | gmalsack: Verify the setting newtonr told you to confirm. Set up a dialplan hit to show that it is tracking the state properly. |
| 15:20.43 | *** join/#asterisk navaismo (~navaismo@201.124.147.198) |
| 15:20.52 | [TK]D-Fender | gmalsack: And read the sample confg & app instructions for dynamic members if appropriate. |
| 15:21.04 | [TK]D-Fender | gmalsack: But validate what newtonr suggested first |
| 15:21.25 | gmalsack | queues and dynamic members are being held in mysql not config files... |
| 15:21.31 | Bkhan | Module 'chan_bridge.so' could not be loaded. |
| 15:21.38 | Bkhan | no permission error |
| 15:22.32 | gmalsack | callcounter is NOT enabled... ;-) I'll try that and let you know... |
| 15:22.33 | [TK]D-Fender | Bkhan: Is * running? |
| 15:22.44 | [TK]D-Fender | gmalsack: that's a DOA right there... |
| 15:22.53 | Bkhan | no i use asterisk -c |
| 15:23.08 | [TK]D-Fender | Bkhan: and....? |
| 15:23.57 | Bkhan | otherwise asterisk is running |
| 15:25.26 | Bkhan | should i delete asterisk.ctl file? |
| 15:25.39 | newtonr | Bkhan, so Asterisk is running now, you can connect to it, but the chan_bridge.so module won't load? Look at logger.conf, turn up DEBUG and VERBOSE in your full log, then go look at where it fails to load. You should see why. |
| 15:25.50 | [TK]D-Fender | Bkhan: Stop *. Restart it as a sevice |
| 15:26.03 | Bkhan | I did many times |
| 15:26.09 | Bkhan | even restart server |
| 15:26.14 | [TK]D-Fender | Bkhan: verify that the process it running. Wait a few seconds. check AGAIN. verify that the process number is the same. Then try connecting |
| 15:26.23 | gmalsack | ALTER TABLE `pbx`.`sippeers` CHANGE COLUMN `callcounter` `callcounter` ENUM('yes','no') NULL DEFAULT 'yes' ; <-- for the win!!! |
| 15:26.49 | *** join/#asterisk bulkorok (~Benjamin@85.183.61.47) |
| 15:27.08 | [TK]D-Fender | gmalsack: Or.... you could ADD your records right in the first place. But that's probably jsut me being silly again... |
| 15:27.34 | *** part/#asterisk bulkorok (~Benjamin@85.183.61.47) |
| 15:28.16 | *** join/#asterisk thecardsmith (~doug@unaffiliated/protocoldoug) |
| 15:28.18 | gmalsack | http://pastebin.com/eCadC9vk <- :-( |
| 15:29.08 | [TK]D-Fender | gmalsack: I see no peer dump (to prove it got reloaded) and no hint to validate against. |
| 15:29.33 | bananapie | can I specify the outbound proxy in the dial command ? I looked in chan_sip.c and I don't see any possibility of this |
| 15:30.56 | gmalsack | [TK]D-Fender: how would I do a peer dump using realtime??? sip show peers doesn't display peers, and sip show peer 7006 says peer 7006 not found.... |
| 15:32.30 | [TK]D-Fender | gmalsack: rtccache=yes |
| 15:34.29 | Bkhan | 5246 is process id |
| 15:34.46 | Bkhan | which is same |
| 15:34.53 | [TK]D-Fender | Bkhan: PASTEBIN <- |
| 15:35.05 | Bkhan | ok |
| 15:35.41 | *** join/#asterisk vlad_sta_ (~vlad_star@109.188.126.101) |
| 15:36.23 | newtonr | Bkhan, To clarify, so you run asterisk with "asterisk -c" and it starts up (a process is running), but leaves you at the linux shell? |
| 15:37.04 | Bkhan | yes |
| 15:37.14 | [TK]D-Fender | newtonr: Not where he should be if he following directions. |
| 15:37.19 | Bkhan | tehn use ctrl+c to quit |
| 15:37.22 | [TK]D-Fender | newtonr: what are the odds... |
| 15:37.42 | [TK]D-Fender | he's* |
| 15:40.00 | newtonr | [TK]D-Fender, low, but maybe something is broken |
| 15:40.10 | newtonr | Bkhan, you ctrl+c to quit what? |
| 15:40.34 | *** join/#asterisk ipengineer (~zconkle@static-71-252-134-63.dllstx.fios.verizon.net) |
| 15:40.37 | Bkhan | from asterisk console when i run it from asterisk -cv |
| 15:41.11 | newtonr | Bkhan, oh.. that contradicts what you said about it leaving you at the linux shell |
| 15:41.32 | newtonr | Bkhan, so asterisk runs, and you do connect to the Asterisk CLI ? |
| 15:41.32 | gmalsack | [TK]D-Fender: http://pastebin.com/mu0tX6HH >- peer dump |
| 15:41.53 | Bkhan | yes when from asterisk -c |
| 15:42.04 | Bkhan | and also connects with asterisk -rvv |
| 15:42.18 | Bkhan | but v limited commands |
| 15:42.23 | [TK]D-Fender | So it connects now? |
| 15:42.25 | newtonr | Bkhan, ... what is the current issue then? |
| 15:42.43 | Bkhan | when we user asterisk -rvvv |
| 15:42.49 | Bkhan | no sip command |
| 15:43.03 | Bkhan | and i am not able to load or unload any module |
| 15:43.30 | Qwell | Bkhan: What, specifically, are you trying to do? |
| 15:43.32 | newtonr | Bkhan, so you need to follow what I mentioned about logger.conf and getting a verbose and debug log on startup to see what is and isn't loading |
| 15:44.10 | Bkhan | newtonr: ok |
| 15:46.28 | newtonr | Bkhan, i'd get one log without "DEBUG" type messages and one with it. Look at the one without it first as there can be a lot of noise if you are not used to looking at it |
| 15:47.33 | newtonr | Bkhan, if you are not getting SIP commands , odds are chan_sip.so isn't loaded. You should search through the log for chan_sip related messages. Feel free to pastebin the log and we can help |
| 15:48.01 | newtonr | Bkhan, when on the actual CLI you can also run a command like "module show like chan_sip.so" to see if that module is loaded or running |
| 15:48.49 | Bkhan | http://pastebin.com/T7euptjP |
| 15:48.53 | Bkhan | please check |
| 15:49.33 | [TK]D-Fender | go verify the owners on your modules |
| 15:52.48 | Bkhan | i am logging with root |
| 15:53.10 | [TK]D-Fender | not what I asked |
| 15:53.28 | [TK]D-Fender | if you started * as a service it may not be running it as root |
| 15:54.32 | Bkhan | actually i am using elastix |
| 15:54.47 | newtonr | Bkhan, go to the "/usr/lib64/asterisk/modules/" directory and verify the file permissions make sense with which user is actually running Asterisk |
| 15:55.29 | newtonr | and go ahead and pastebin the output of the commands you run when doing that so that we can have a look too. |
| 15:57.00 | gmalsack | [TK]D-Fender: http://pastebin.com/ZYGGg2wW <- hints showing but not working..... |
| 15:57.37 | [TK]D-Fender | gmalsack: I don't see callounter as being set yet... |
| 15:57.44 | [TK]D-Fender | gmalsack: I'd reload the peer... |
| 15:57.52 | Bkhan | ok |
| 16:03.46 | *** join/#asterisk zigg (~matt@unaffiliated/zigg) |
| 16:03.49 | *** join/#asterisk paulc (~root@unaffiliated/paulc) |
| 16:04.44 | gmalsack | [TK]D-Fender: http://pastebin.com/YGZGfnh4 <- peer db record, peer prune load prune load.... still doesn't seem to work right.... :-( |
| 16:05.13 | Bkhan | should i install same version of asterisk on it ? |
| 16:05.22 | Bkhan | with tar.gz |
| 16:06.20 | *** join/#asterisk _Corey_ (~chatzilla@50-200-184-54-static.hfc.comcastbusiness.net) |
| 16:07.09 | [TK]D-Fender | gmalsack: I'm not seeing it reload from your attempts and I'm wondering on the cache status now... |
| 16:07.23 | [TK]D-Fender | gmalsack: DE-REGISTER THE RE-REGISTER AND SEE IF IT COMES UP ANY DIFFERENT |
| 16:07.34 | [TK]D-Fender | darn caps |
| 16:09.28 | newtonr | Bkhan, i wouldn't go there yet.. first you'll want to verify file ownership like TK suggested and pastebin the commands you run to check all of that |
| 16:09.48 | Bkhan | ok |
| 16:11.18 | Bkhan | http://pastebin.com/2ee65Mje |
| 16:11.27 | Bkhan | its modules |
| 16:13.15 | newtonr | Bkhan, okay now, show us the output of "ps -aux | grep -i asterisk" |
| 16:13.27 | [TK]D-Fender | Bkhan: all owned by ROOT> |
| 16:13.33 | [TK]D-Fender | Bkhan: this is bad |
| 16:13.47 | [TK]D-Fender | Bkhan: amportal chown <--------------- |
| 16:13.48 | Bkhan | yes all owned by root |
| 16:13.52 | [TK]D-Fender | BAD |
| 16:13.59 | *** join/#asterisk jansiva (~janaki@118.102.128.225) |
| 16:14.04 | [TK]D-Fender | elastix does NOT run * as root |
| 16:14.22 | [TK]D-Fender | You've clealry screwed up an attempt to manually upgrade your system |
| 16:14.34 | [TK]D-Fender | run the fixup I gave you and verify |
| 16:18.08 | *** join/#asterisk gerhard7 (~gerhard7@77-172-47-159.ip.telfort.nl) |
| 16:24.01 | *** join/#asterisk zerick (~eocrospom@190.187.21.53) |
| 16:25.06 | *** join/#asterisk marlinc (~marlinc@ip565fa73c.direct-adsl.nl) |
| 16:28.57 | gmalsack | [TK]D-Fender: newtonr: Thanks guys.... I at least now have hints working on 1 server. Now to setup distribution. would you recommend openais or xmpp? Those are the 2 primary methods I found through google. |
| 16:30.15 | *** join/#asterisk lorsungcu (~anonymous@67.138.198.66) |
| 16:31.19 | vader-- | would love to know why this flowroute connection doesn't seem to be working for some UK numbers... |
| 16:31.43 | vader-- | Here is a sample number we are sending to flowroute to dial the UK 441617779554 |
| 16:32.55 | vader-- | This is the cli output: http://pastebin.com/yZCAa9iJ |
| 16:32.56 | gmalsack | vader--: I was having a problem in mobile al usa with a did giving busy signal as well this morning. when I checked for sip traffic coming from flowroute there was nothing hitting my server. my guess is routing issues.... |
| 16:34.37 | WIMPy | vader--: Looks perfectely ok to me. What's your issue with that call? |
| 16:34.45 | vader-- | it's not connecting |
| 16:34.53 | vader-- | basically the system sets up a conference room |
| 16:34.56 | vader-- | it doesn't a sip phone |
| 16:35.08 | vader-- | then it dials a phone number and connects them |
| 16:35.14 | vader-- | it's not making the connection |
| 16:35.20 | WIMPy | That's not what the log says. |
| 16:35.58 | WIMPy | You're using AMD and that fails. But that's a rather different story. |
| 16:37.16 | [TK]D-Fender | vader--: You're wondering why your call doesn't work... and aren't including SIP DEBUG |
| 16:37.47 | *** join/#asterisk Mango45 (~Mango45@209.89.209.166) |
| 16:38.08 | WIMPy | ... "making progress" ... "answered" ... What more do you need? |
| 16:38.26 | *** join/#asterisk jasonwert (~w3rt@75-134-81-98.static.aldl.mi.charter.com) |
| 16:38.40 | Mango45 | Slightly related question to before. I need to register to a server to receive calls. What is the best way to register to some other server, only if the first registration fails? |
| 16:41.20 | [TK]D-Fender | WIMPy: well there is that too... and "not making connection" could mean a few things. That's what you get for vague descriptions. |
| 16:41.42 | [TK]D-Fender | Mango45: There is no means for this. |
| 16:41.49 | Mango45 | Fair enough. |
| 16:46.48 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.127.133) |
| 16:56.30 | gmalsack | request from call center manager: can you pause them after 5 minutes of inactivity and when they log back into their phone they go to the back of the Line for Round Robin (i.e. rrmemory) |
| 17:00.12 | Bkhan | <PROTECTED> |
| 17:00.31 | Bkhan | Issue was some one has changed is /etc/asterisk files |
| 17:03.01 | [TK]D-Fender | gmalsack: RRmemory tracks where it left off... it doesn't change the order |
| 17:03.15 | [TK]D-Fender | gmalsack: You'd have to use dynamic memebers and actuall REMOVE them |
| 17:03.59 | gmalsack | [TK]D-Fender: exactly.... I just love what these guys come up with.... lol |
| 17:04.14 | *** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/chainsaw) |
| 17:08.00 | *** join/#asterisk jacobkiers (~jacobkier@jacobkiers.net) |
| 17:14.50 | *** join/#asterisk darkbasic_ (~quassel@niko.linuxsystems.it) |
| 17:20.40 | *** join/#asterisk darkbasic (~quassel@niko.linuxsystems.it) |
| 17:21.35 | *** join/#asterisk dfighter (~someone@arcemu/staff/dfighter) |
| 17:25.26 | *** join/#asterisk danjenkins_ (~dan@cpc65687-folk2-2-0-cust207.1-2.cable.virginmedia.com) |
| 17:27.11 | vader-- | tk, ever use goautodial? |
| 17:31.21 | *** join/#asterisk danjenkins_ (~dan@nat/digium/x-alpkmaarydeskfkf) |
| 17:31.36 | DigiDaz | I have a bunch of 'hung' invites still showing on my console, apart from restarting asterisk is there any way to get rid of them? |
| 17:33.37 | newtonr | DigiDaz, can you pastebin the command output to show us what you mean |
| 17:34.18 | [TK]D-Fender | vader--: nope |
| 17:35.10 | DigiDaz | newtonr: Basically a sip show channels gives: 161 active SIP dialogs |
| 17:35.47 | newtonr | DigiDaz, that could be normal depending on what your system is doing |
| 17:35.58 | DigiDaz | Of which about 8 are actually acks with loads of invites from about 6 hours ago |
| 17:36.02 | Katty | looks in |
| 17:36.15 | newtonr | DigiDaz, yeah that part sounds abnormal |
| 17:36.41 | newtonr | DigiDaz, what exact version of Asterisk? |
| 17:36.47 | DigiDaz | It was someone trying some sort of attack this morning, I was hoping they would time out |
| 17:37.01 | *** join/#asterisk marlinc (~marlinc@ip565fa73c.direct-adsl.nl) |
| 17:37.16 | DigiDaz | Asterisk 1.8.11.1 |
| 17:39.47 | *** join/#asterisk alsuren (~dlaban@host86-130-163-226.range86-130.btcentralplus.com) |
| 17:40.54 | newtonr | DigiDaz, If you can't restart, you could try just reloading chan_sip to see if that gets rid of them. As for them sticking around, that sounds buggy. May be a fix in a newer version if it is a bug. |
| 17:42.30 | newtonr | From talking with one of the developers, in a normal scenario where the remote party ends the call, they should stick around for about 32 seconds, and if Asterisk is ending the call then they should go away immediately. Therefore six hours does sound strange. |
| 17:44.12 | newtonr | DigiDaz, You probably want to upgrade to the latest 1.8 regardless, for various security fixes, so I'd make that a next step. Then you ever see stuff sticking around again, file a bug on the tracker. |
| 17:44.14 | DigiDaz | Cool, thanks, I may as well restart asterisk as reload chan_sip but I'll have to look into it, its starting to become a pain. I have homer hooked up to that box so I'll have a look at the sip dialogue but I believe fail2ban just blocked their IP |
| 17:45.35 | DigiDaz | i'm just messing with putting kamailio in front of it at the moment so I should nail it soon |
| 17:49.06 | newtonr | DigiDaz, you might also search JIRA for possibly related issues before upgrading. https://issues.asterisk.org/jira/browse/ASTERISK-23047 |
| 17:49.26 | *** join/#asterisk mic__ (~mic@90.184.7.102) |
| 17:49.33 | newtonr | You'll probably need to find out more specifically from your logs what exactly those particular calls were doing |
| 17:49.42 | DigiDaz | Cool, I have to be careful because its an a2billing box |
| 17:49.42 | newtonr | I'm off for a bit! |
| 17:49.56 | DigiDaz | thanks, cya! |
| 17:49.59 | newtonr | np |
| 17:51.39 | *** join/#asterisk rrittgarn (~rrittgarn@75-150-221-196-Illinois.hfc.comcastbusiness.net) |
| 18:01.12 | *** join/#asterisk darkbasic_ (~quassel@niko.linuxsystems.it) |
| 18:10.30 | *** join/#asterisk darkbasic (~quassel@niko.linuxsystems.it) |
| 18:13.07 | *** join/#asterisk puzzled (~patrick@puzzled.xs4all.nl) |
| 18:15.39 | *** join/#asterisk afidegnum (~chatzilla@41.207.162.253) |
| 18:15.47 | afidegnum | hello, pls what s the difference between asteris 11 and 1.8 ? |
| 18:16.23 | Qwell | 9.2 |
| 18:16.30 | WIMPy | https://wiki.asterisk.org/wiki/display/AST/New+in+11 |
| 18:17.40 | rrittgarn | when using feature codes to kick off automixmon(call recording), Is there a variable that says what the file name that was used? Or do i have to do a custom application map to make that work? |
| 18:18.23 | afidegnum | any anwer |
| 18:19.20 | WIMPy | afidegnum: Was that a question? If so what more answers do you need? |
| 18:19.48 | [TK]D-Fender | afidegnum: there is a CHANGELOG and UPGRADE.TXT to describe the major and minor changes. Read them |
| 18:19.59 | afidegnum | I am trying to install asterisk on debian squeeze so I wnt to know which is hte appropriate one to isntall |
| 18:20.23 | [TK]D-Fender | afidegnum: 11 has many major improvements over 1.8 |
| 18:20.31 | [TK]D-Fender | afidegnum: if you are loking at 11 there is no point in looking at 1.8 |
| 18:20.47 | afidegnum | ok |
| 18:32.32 | *** join/#asterisk jhlavacek (~jirka@87.89.218.63) |
| 18:38.22 | rrittgarn | Fender, you seem to have most of the answers to my questions, happen to know the answer to my Previous one? |
| 18:39.03 | [TK]D-Fender | rrittgarn: When? Where? |
| 18:39.26 | rrittgarn | (12:17:40 PM) rrittgarn: when using feature codes to kick off automixmon(call recording), Is there a variable that says what the file name that was used? Or do i have to do a custom application map to make that work? |
| 18:39.43 | [TK]D-Fender | [13:39][TK]D-Fenderrrittgarn: When? Where? |
| 18:40.36 | rrittgarn | on a dial with the 'X' option set. |
| 18:41.00 | [TK]D-Fender | Lets try this again.... |
| 18:41.14 | [TK]D-Fender | WHEN are you expecting to try and get your hands on this information in the first place? |
| 18:41.33 | rrittgarn | after the call is fine. And thanks for the clarification |
| 18:41.36 | [TK]D-Fender | From WHERE? |
| 18:42.09 | rrittgarn | goal is to either set or read from a variable, pass the path to a script that will email the recording to the person who hit the record button |
| 18:42.09 | [TK]D-Fender | After the call.... the call is over and there is no "variable" |
| 18:42.18 | rrittgarn | in the exten=>h |
| 18:42.37 | [TK]D-Fender | You have the filename itself which is based on the channelid (uniqueid) which ties it to CDR |
| 18:42.57 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.127.146) |
| 18:44.41 | rrittgarn | auto-1389204332-6309385300-16309385325 |
| 18:44.41 | rrittgarn | so if i looked for auto-${CHANNEL(uniqueid)}-ConnectedLineNum-DNID's existence that about covers it then? |
| 19:01.12 | *** join/#asterisk jasonwert (~w3rt@75-134-81-98.static.aldl.mi.charter.com) |
| 19:03.31 | *** join/#asterisk gauner1986 (~Adium@f055005165.adsl.alicedsl.de) |
| 19:09.46 | gauner1986 | hey guys. i'm on Asterisk 11.7.0 on fritzbox/freetz (mips).. for some reason asterisk is not loading my extensions.conf and none of my dialplans are loaded. config.list shows a few config files, but extensions.conf is not one of them. the file IS accessible for asterisk, i have checked that. does anyone have a clue what the problem could be? |
| 19:10.40 | gauner1986 | haven't found anything about extensions.conf or any related error in debug output of asterisk -vvvv |
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| 19:31.21 | Synx|hm | Anyone familiar with .vce NMS24 files? I need to figure out how to convert from .wav to .vce in NMS24 format |
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| 19:45.37 | ipengineer | In Asterisk 12 what would be the best way to subscribe to device state changes? I saw the res_pubsub module but am not finding much documentation. I basically want to get events anytime a devices status changes |
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| 19:50.29 | Qwell | Synx|hm: Sounds like a question for NMS. |
| 19:50.37 | Qwell | Proprietary files, and all... |
| 19:51.18 | Synx|hm | Qwell: yes that's what it seems. Sounds like I'll be contacting Dialogic directly soon then, bummer. |
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| 20:13.15 | kristoffelloos | I'm wondering what monitoring tools you all use, I need to monitor some asterisk server & opensips, looking into Nagios + cacti or Zabbix, what should I use?! Any ideas or tips? |
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| 20:21.32 | jeffspeff | kristoffelloos, i use opennms |
| 20:21.48 | jeffspeff | it's documented in ~book on how to configure |
| 20:21.55 | jeffspeff | ~book |
| 20:21.55 | infobot | Asterisk: The Definitive Guide, 4th Edition (ISBN 1-4493-3242-0) available at http://oreilly.com/catalog/0636920025894 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and a version is available for reading online at http://www.asteriskdocs.org/ or see ~buybook |
| 20:26.16 | kristoffelloos | Thank you jeffspeff |
| 20:27.14 | leifmadsen | I think it was documented in the 3rd edition, but we removed it in the 4th edition |
| 20:27.15 | leifmadsen | if I remember correctly |
| 20:27.29 | jeffspeff | leifmadsen, ok. 3rd is the one i read it in. i haven't read the 4th edition yet |
| 20:27.38 | leifmadsen | +1 |
| 20:51.19 | drmessano | I bought the ebook yesterday of the 4th one |
| 20:51.33 | drmessano | Half off on OReilly. Booyah |
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| 22:51.55 | bsdice | Good news |
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| 23:49.18 | bananapie | Is it possible for a SIP header to set the language in asterisk if I don't read the SIP header in the dialplan ? |
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